| 00:00.10 | [TK]D-Fender | JAMMAN2110: No.... plenty of kiwis around here ;) |
| 00:00.20 | JAMMAN2110 | Shhh |
| 00:00.28 | JAMMAN2110 | Dont dilute my reality |
| 00:00.31 | JAMMAN2110 | ;) |
| 00:01.35 | [TK]D-Fender | JAMMAN2110: It was never that concentrated anyway.... |
| 00:05.14 | hardwire | kiwi concentrate? |
| 00:08.24 | JAMMAN2110 | You can buy kiwi fruit concentrate |
| 00:08.25 | JAMMAN2110 | Does that count? |
| 00:12.29 | hardwire | I have this crazy organic key lime juice that is super potent |
| 00:15.02 | *** join/#asterisk riddlebox (n=user@75-132-225-75.dhcp.stls.mo.charter.com) |
| 00:15.12 | *** join/#asterisk Katty (n=asterisk@mail.copi-rite.com) |
| 00:15.12 | Katty | hai |
| 00:15.28 | hardwire | is nomming a klondike bar |
| 00:16.09 | manxpower | hardwire: do it in private, dude! |
| 00:16.22 | Katty | i have a tuna patty and hmm. not sure what to call this side dish. |
| 00:16.38 | Katty | taste of home calls it Pepper Parmesan Beans |
| 00:16.43 | Katty | but it's horribly undescriptive. |
| 00:17.16 | Katty | red pepper, onion, garlic, green beens, basil, and parmesan. |
| 00:18.01 | Katty | looks kinda christmasy |
| 00:21.07 | *** join/#asterisk btsteve (n=tstevens@24-196-234-39.dhcp.gwnt.ga.charter.com) |
| 00:22.08 | ardor | "Then restart the Festival server. " |
| 00:22.10 | ardor | how do i do that? |
| 00:23.09 | TJNII | Well, how did you start it in the first place? |
| 00:23.18 | hardwire | heh |
| 00:23.28 | ardor | just installed it with apt-get |
| 00:23.45 | hardwire | /etc/init.d is yer buddy |
| 00:24.11 | TJNII | I don't think debian installs an init script for festival |
| 00:24.27 | TJNII | You can start it in a terminal, see man festival |
| 00:27.13 | Katty | hmm. i made the tuna patties with cornmeal this time. turned out pretty awesome (= |
| 00:27.47 | TJNII | I should make cornbread this evening |
| 00:27.53 | TJNII | Before the mix and the eggs go bad. |
| 00:28.00 | Katty | i've been debating cornbread here lately. |
| 00:28.11 | *** join/#asterisk bbt (n=sam@180.189.138.55) |
| 00:28.25 | TJNII | I love cornbread, but I'm so lazy when it comes to cooking I never make it. |
| 00:29.11 | TJNII | My diet consists mostly of cold-cut sandwitches. Not because I can't cook, but because I'm too lazy. :P |
| 00:29.24 | Katty | that's most unfortunate. |
| 00:31.32 | manxpower | I don't think so. The less time spent cooking means more time spent doing other stuff. |
| 00:31.33 | Katty | so many wonderful things you're missing out on |
| 00:32.03 | manxpower | I usually just nuke something out of the freezer. |
| 00:33.07 | TJNII | I think I'll go make dinner. And tonight I have to actually cook something, because I've eaten everything that doesn't need to be cooked. |
| 00:33.45 | *** join/#asterisk alexshell (n=abc@unaffiliated/alexshell) |
| 00:35.08 | ardor | anyone seen this before? |
| 00:35.08 | ardor | http://pastebin.ca/1662760 |
| 00:35.52 | manxpower | ardor: I think that means "your computer is too damn slow" or "you're computer is too overloaded with stuff" |
| 00:37.16 | ardor | manxpower: nothing goes crazy in top, barely moves. |
| 00:38.00 | Katty | someone explain to me why people ask for advice |
| 00:38.03 | Katty | and then want to argue about it |
| 00:38.41 | Katty | i just can't seem to wrap my brain around it. |
| 00:39.32 | ardor | Katty: he said he thinks, not that he knows. |
| 00:39.50 | Katty | well you certainly don't know |
| 00:40.01 | Katty | and a thought is certainly better than nothing |
| 00:40.05 | ardor | Katty: Right, Thats why I am asking, What are you doing? |
| 00:42.17 | Katty | ... |
| 00:45.22 | *** join/#asterisk Iamnacho (i=Iamnacho@ip98-186-180-143.ks.ks.cox.net) |
| 00:48.00 | *** join/#asterisk chendy (n=chatzill@119.123.211.213) |
| 00:58.46 | *** join/#asterisk Kumbang (n=kumbang@rusnas.paume.itb.ac.id) |
| 00:58.49 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 01:02.28 | *** join/#asterisk thazza (n=thazza@ntechm.lnk.telstra.net) |
| 01:09.19 | *** join/#asterisk xpot (n=xpot@70-91-210-233-BusName-Utah.hfc.comcastbusiness.net) |
| 01:17.57 | *** part/#asterisk thazza (n=thazza@ntechm.lnk.telstra.net) |
| 01:30.11 | *** join/#asterisk Gokee2_Extra (n=gokee2@173-10-74-246-BusName-Washington.hfc.comcastbusiness.net) |
| 01:31.41 | *** join/#asterisk xpot-mobile (n=xpot@70-91-210-233-BusName-Utah.hfc.comcastbusiness.net) |
| 01:33.22 | Gokee2_Extra | Hello all, is there a way in users.conf to set what mailbox a person has and define it in voicemail.conf? I tried creating a voicemail.conf user and then setting mailbox = exten_of_person_in_voicemail.conf but it brought me into the main menu. So I tried hasvoicemail = yes but that created a new voicemailbox for the user instead of the one setup in voicemail.conf |
| 01:35.19 | manxpower | Gokee2_Extra: the only people that use users.conf are GUIs. |
| 01:35.50 | [TK]D-Fender | ~users.conf |
| 01:35.51 | infobot | [users.conf] an Asterisk configuration file that was primarily created for the AsteriskGUI project. It is intended as a simple configuration interface for users with basic PBX functionality, not as a replacement for other configuration methods. |
| 01:36.14 | [TK]D-Fender | Gokee2_Extra: FORGET voicemail.conf. Forget sip.conf... that POS users.conf owns your config. |
| 01:36.31 | [TK]D-Fender | Gokee2_Extra: It fakes everything out. |
| 01:36.48 | [TK]D-Fender | ... |
| 01:36.56 | [TK]D-Fender | Apparently more botlet changes... |
| 01:37.05 | *** join/#asterisk bbt (n=sam@180.189.138.55) |
| 01:37.11 | manxpower | [TK]D-Fender: can you lock your bot? |
| 01:37.22 | [TK]D-Fender | manxpower: Its not actually mine... |
| 01:38.31 | Gokee2_Extra | [TK]D-Fender, Ahhh... The other day I decided to play around in asteriskGUI and it showed me users.conf.... I seem to have got most stuff working in it but I need the phones to all share a common voicemailbox |
| 01:38.43 | manxpower | ~asterisk-gui |
| 01:38.44 | infobot | [~asterisk-gui] The Asterisk-GUI is an open source project lead by Digium. It's client-driven, and its only dependency is Asterisk itself. Check out the new version from svn here: $ svn co http://svn.digium.com/svn/asterisk-gui/branches/2.0 asterisk-gui-2.0. For support go to #asterisk-gui |
| 01:39.22 | Gokee2_Extra | was currently playing around in the config files not clicking around in the gui |
| 01:39.42 | [TK]D-Fender | Gokee2_Extra: going to take a lot of manual work at best |
| 01:39.50 | manxpower | Gokee2_Extra: The GUI writes config files that can't be easily supported here. |
| 01:40.28 | Gokee2_Extra | Ahh... I read the chapter on the gui in the asterisk book and it sounded nice being able to use the config files and the gui... |
| 01:40.31 | manxpower | Written by hand an extension can be as little as one line in extensions.conf. Written by the GUI an extension can be as little as 25 lines in extensions.conf. |
| 01:41.37 | [TK]D-Fender | manxpower: users.conf does so much more than that.... |
| 01:41.49 | hardwire | oh u 2. |
| 01:41.50 | hardwire | err |
| 01:41.52 | hardwire | o u 2. |
| 01:41.53 | hardwire | heh |
| 01:41.56 | manxpower | [TK]D-Fender: I may have to look at users.conf for my stuff. |
| 01:42.00 | hardwire | just get a motel and get it over with. |
| 01:42.10 | Gokee2_Extra | But the gui can be used by other people so I don't have to always tweek the config files :) |
| 01:42.14 | manxpower | hardwire: I don't think I'm [TK]D-Fender |
| 01:42.14 | [TK]D-Fender | manxpower: Oh I'm not advocate... |
| 01:42.15 | Gokee2_Extra | tweak* |
| 01:42.16 | manxpower | s type |
| 01:42.30 | hardwire | aww.. how does that make you feelz? |
| 01:42.47 | [TK]D-Fender | manxpower: Its a 1-way trip to toasterville which makes FreePBX look like a deluxe pleasure cruise |
| 01:42.57 | manxpower | hardwire: I doubt he's my type, but I'm pretty sure I'm not his type. |
| 01:43.02 | hardwire | heh |
| 01:43.04 | hardwire | how PC! |
| 01:43.06 | hardwire | I dig it. |
| 01:43.15 | manxpower | <-- not into women |
| 01:43.28 | [TK]D-Fender | <- keeps trying to get into women... success varies |
| 01:43.45 | [TK]D-Fender | uNF! |
| 01:43.52 | Gokee2_Extra | [TK]D-Fender, Wow... you make it sound so good. |
| 01:43.53 | hardwire | manxpower: oooh I had no clue :) |
| 01:44.22 | hardwire | the sassyness should have been a clue tho. |
| 01:45.06 | manxpower | [TK]D-Fender: The stuff I'm working on will require modifications of Asterisk config files using an external script. |
| 01:45.36 | [TK]D-Fender | manxpower: I'm sure your wheel will be much rounder... |
| 01:46.14 | hardwire | rawr? |
| 01:46.27 | manxpower | [TK]D-Fender: I re-implemented those legendary macros in AEL2. They are about 20% of the size of the extensions.conf version. |
| 01:46.53 | manxpower | AEL2 rocks! |
| 01:50.28 | *** join/#asterisk Caplain (i=shayne@2001:470:5:fb:a1c7:96f5:9ebf:292a) |
| 01:51.09 | *** join/#asterisk Caplain (i=shayne@2001:470:5:fb:a1c7:96f5:9ebf:292a) |
| 01:51.47 | [TK]D-Fender | manxpower: Visibly... |
| 01:52.09 | [TK]D-Fender | manxpower: Then again they weren't my definition os sane to begin with. |
| 01:52.21 | [TK]D-Fender | manxpower: and efficiency-wise is only worse with AEL2 |
| 01:52.33 | *** join/#asterisk Caplain (i=shayne@2001:470:5:fb:a1c7:96f5:9ebf:292a) |
| 01:52.46 | Caplain | there we go |
| 01:53.06 | Katty | peeks in |
| 01:53.08 | hardwire | is cyber-stalking manxpower |
| 01:53.24 | Caplain | how |
| 01:53.26 | Caplain | hot |
| 01:53.27 | Caplain | omg |
| 01:54.10 | Katty | ^_- |
| 01:54.25 | hardwire | finds photographic evidence of existance |
| 01:57.37 | manxpower | [TK]D-Fender: *nod* I feel that it's better to be able to maintain the dialplan than to try to squeeze out every ounce of performance out of the server. |
| 01:59.43 | [TK]D-Fender | manxpower: As long as you're past the debugging and "shared code" phase it might work. Butu it has functional shortcomings giving it compiles back |
| 02:00.29 | Gokee2_Extra | So I saw a old post (2004) about voicemail folders being hard coded in. Is this still true? |
| 02:00.32 | hardwire | Butu? |
| 02:02.28 | manxpower | [TK]D-Fender: I do agree that it makes debugging harder. |
| 02:02.49 | manxpower | Gokee2_Extra: URL? |
| 02:04.09 | *** join/#asterisk Iamnach0 (i=Iamnacho@ip98-186-180-143.ks.ks.cox.net) |
| 02:05.53 | Gokee2_Extra | manxpower, Sure one sec, this computer is going crazy cause I rebooted my server which hosts all the user home directory's through a nfs share (I really need to find a better way to do central home dir's) |
| 02:06.18 | Gokee2_Extra | Ah here it is http://forums.digium.com/viewtopic.php?t=1318 |
| 02:06.58 | *** join/#asterisk ming_zym (n=ming_zym@220.181.34.232) |
| 02:07.19 | manxpower | Gokee2_Extra: I believe that is still the case. |
| 02:07.41 | Gokee2_Extra | manxpower, Ok, too bad. Thanks :) |
| 02:08.47 | manxpower | Gokee2_Extra: 1.6+ has MiniVM suite of apps that let you build a voicemail system in the dialplan |
| 02:09.02 | [TK]D-Fender | "While it is possible to remove these folders from disk (they are only created if somebody is using/has used them), changing or removing them from the voicemail application may be a bit of a coding effort." |
| 02:09.46 | [TK]D-Fender | Gokee2Please ignore 4 year old crap that references 1.2 as BETA |
| 02:09.57 | Gokee2_Extra | Mostly I would like the voicemail to just start playing! When I call it. Eg no "You have bla bla press one bla bla...." |
| 02:10.08 | ardor | Gokee2: or you can use RealTime to build your voicemail conf in mysql |
| 02:10.26 | Gokee2_Extra | realtime? |
| 02:10.51 | *** join/#asterisk slinksh0t (n=slinksh0@c-76-108-31-194.hsd1.fl.comcast.net) |
| 02:10.53 | ardor | http://www.voip-info.org/wiki/view/Asterisk+RealTime |
| 02:12.20 | *** join/#asterisk xpot (n=xpot@70-91-210-233-BusName-Utah.hfc.comcastbusiness.net) |
| 02:12.25 | Gokee2_Extra | Ah I see |
| 02:13.01 | Gokee2_Extra | Thinks he will just stick with the standard voice mail and improve the web interface some |
| 02:13.22 | p3nguin_ | Voicemail has a web interface? |
| 02:13.59 | Gokee2_Extra | p3nguin_, Ya |
| 02:14.10 | p3nguin_ | You sure? |
| 02:14.26 | Gokee2_Extra | p3nguin_, Yep vmail.cgi |
| 02:14.36 | p3nguin_ | looks |
| 02:15.03 | p3nguin_ | I found the file in my sources, but apparently it didn't install. |
| 02:15.20 | Gokee2_Extra | p3nguin_, If you are compiling from source you do something like make vmail or such... Not sure, I got it with the debian package and figured out where it wanted to be to be happy |
| 02:21.16 | *** join/#asterisk Baelig (n=chatzill@user-0c8h5rq.cable.mindspring.com) |
| 02:21.24 | *** join/#asterisk KavanS (n=KavanS@static-173-50-141-22.ptldor.fios.verizon.net) |
| 02:23.03 | Baelig | a place i use to do work for was using a windows app that registered with asterisk and showed extensions online/busy/etc, but I can't seem to recall it's name. i think the UI was blue/silver. ring any bells? |
| 02:23.20 | manxpower | Baelig: "flash operator panel" |
| 02:24.39 | Baelig | manxpower: that looks interesting, but this was a native windows app. |
| 02:24.54 | *** join/#asterisk Iamnacho (i=Iamnacho@ip98-186-180-143.ks.ks.cox.net) |
| 02:25.34 | manxpower | Baelig: I'm not aware of any such app |
| 02:26.22 | Baelig | phooey... ok, thanks. |
| 02:26.41 | dan__t | all you can eat sushi and sake bombers for $40 :/ |
| 02:26.44 | dan__t | i hurt |
| 02:28.11 | hardwire | TOOMUCHTUUUUUUNAAAA |
| 02:28.22 | dan__t | fact. |
| 02:30.55 | b14ck | sup everyone |
| 02:36.13 | *** join/#asterisk maxagaz (n=maxagaz@soho2.i-xanadu.com) |
| 02:45.52 | *** join/#asterisk Iamnach0 (i=Iamnacho@ip98-186-180-143.ks.ks.cox.net) |
| 02:55.26 | cosmicwombat | iSymphony |
| 02:55.51 | *** join/#asterisk Knightfal (n=Knightfa@75.142.144.171) |
| 02:58.02 | Knightfal | Hey Guys.. In Asterisk 1.4.26 can I change what is written to queue_log when UnpauseQueueMember(Agent/blah) is triggered? Could I possibly just make it not write to queue_log and then I can just define my own entry. Any Ideas? |
| 02:58.16 | *** join/#asterisk KavanS (n=KavanS@static-173-50-141-22.ptldor.fios.verizon.net) |
| 03:02.15 | [TK]D-Fender | Knightfal: 1: mod the source 2: make a script to automate stripping the entries. 3:use a DB and write a trigger to drop the adds |
| 03:03.25 | *** join/#asterisk nighty^ (n=nighty@210.188.173.245) |
| 03:04.45 | carrar | That simple! |
| 03:05.36 | *** part/#asterisk Baelig (n=chatzill@user-0c8h5rq.cable.mindspring.com) |
| 03:07.12 | Knightfal | Thanks TK all great Ideas. UnpauseQueueMember is an application correct. But where does it live in the source. I was actually just poking around a bit for it. |
| 03:08.12 | [TK]D-Fender | Knightfal: Keep poking. There aren't enough wrong choices for this to take as long to find as to answer |
| 03:09.35 | ChannelZ | GIYF (Grep Is Your Friend!) |
| 03:10.38 | Knightfal | ya |
| 03:10.47 | Knightfal | Thanks Guys |
| 03:10.49 | Knightfal | :) |
| 03:10.58 | Knightfal | Hows other things going? |
| 03:11.09 | *** join/#asterisk voipmonk (n=voipmonk@dsl-67-55-17-41.acanac.net) |
| 03:24.56 | Gokee2_Extra | I have two incoming PSTN lines. Is there any way to get my sip phones to light up when the analog line is busy? |
| 03:25.19 | [TK]D-Fender | Gokee2_Extra: What phone? |
| 03:25.30 | Gokee2_Extra | [TK]D-Fender, Polycom 321 |
| 03:25.40 | [TK]D-Fender | Gokee2_Extra: They don't support presence |
| 03:25.50 | [TK]D-Fender | Gokee2Only the higher models |
| 03:26.36 | Gokee2_Extra | [TK]D-Fender, Ah... Ok.... Thats quite a downgrade from our old analog system not to know when the line is in use.... :/ I guess we have to live with it though eh |
| 03:26.58 | [TK]D-Fender | Gokee2_Extra: 4XX + support it. |
| 03:27.19 | [TK]D-Fender | Gokee2_Extra: you COULD use a microbrowser page to indicate this on interval |
| 03:28.21 | Gokee2_Extra | [TK]D-Fender, Hmm, I will have to look into the microbrowser page thing. Its too bad we have the 321's already. |
| 03:28.51 | [TK]D-Fender | Gokee2_Extra: they are great phones but not enough buttons for this sort of thing anyway |
| 03:29.48 | Gokee2_Extra | Ya, I am already noticing a lack of buttons. I also don't see a easy way to display the time of last call after a call or in a one button press kinda way |
| 03:30.56 | [TK]D-Fender | Gokee2_Extra: the call history works just fine |
| 03:33.19 | Gokee2_Extra | [TK]D-Fender, My user is kinda annoyed at the prospect of having to hit 4 buttons after every call |
| 03:35.04 | Katty | peers |
| 03:35.35 | p3nguin_ | Where do I configure automon? I see where I can set the keys to activate it, but I don't see where to set the file type or the monitor app it uses. |
| 03:36.21 | [TK]D-Fender | Gokee2_Extra: Feel free to make that part of your custom MicroBrowser Idle script for them too |
| 03:36.28 | russellb | automon is just a feature in features.conf, that you enable from your dialplan |
| 03:36.30 | *** join/#asterisk lowlevel (n=Stuart@lowlevel.ca) |
| 03:36.37 | russellb | check the Dial() options, it's in there somewhere |
| 03:36.45 | p3nguin_ | Ah, yeah! |
| 03:36.53 | p3nguin_ | w must accept options, then. |
| 03:36.58 | p3nguin_ | Forgot all about that. |
| 03:37.12 | russellb | and if you don't like how the built in automon works, make a DYNAMIC_FEATURE instead |
| 03:41.29 | p3nguin_ | Well, hmm... w doesn't seem to have options, so I guess I'll have to look for something else. |
| 03:42.46 | *** join/#asterisk ToaSty (n=inickX@c-24-18-78-155.hsd1.wa.comcast.net) |
| 03:43.34 | ToaSty | hello |
| 03:43.50 | ToaSty | I have the FreePBX on the screen GUI and its working |
| 03:43.52 | Katty | hello, there. |
| 03:44.00 | ToaSty | whats my next step for me? |
| 03:44.15 | ChannelZ | ToaSty: Read some docs |
| 03:44.23 | Katty | ToaSty: #freepbx might be more useful. |
| 03:44.26 | Katty | ToaSty: we don't use that here. |
| 03:44.29 | ChannelZ | Or maybe even just the topic |
| 03:44.34 | ToaSty | thank you :) |
| 03:55.37 | Gokee2_Extra | [TK]D-Fender, Ya... I guess I need to jump into microbrowsers... How hard can it be to light up a LED? :/ |
| 03:57.09 | *** join/#asterisk mintos (n=mvaliyav@nat/redhat/x-oibhmqofrhxwdote) |
| 03:59.43 | Katty | well i suppose i should head to bed. |
| 03:59.48 | Katty | morning will come early, unfortunately :< |
| 04:02.59 | [TK]D-Fender | Katty: Morning tends to come at the same time all the time here... |
| 04:07.48 | *** join/#asterisk coppice (n=chatzill@171.168.17.210.dyn.pacific.net.hk) |
| 04:08.33 | *** join/#asterisk OrNix (n=ornix@host89-251-107-21.hnet.ru) |
| 04:17.43 | *** part/#asterisk voipmonk (n=voipmonk@dsl-67-55-17-41.acanac.net) |
| 04:33.35 | *** join/#asterisk dkirker (n=dkirker@pcp063428pcs.wireless.calpoly.edu) |
| 04:40.41 | *** join/#asterisk Torrieri (n=Torrieri@nelug/crew/torrieri) |
| 04:42.39 | *** join/#asterisk btsteve (n=tstevens@24-196-234-39.dhcp.gwnt.ga.charter.com) |
| 04:42.54 | *** join/#asterisk AppleBoy (n=AppleBoy@about/cooking/nakedchef/apple/tarts) |
| 04:59.22 | *** join/#asterisk soman (n=somnath@118.102.130.6) |
| 05:24.23 | *** join/#asterisk ecret (n=ecret@CPE001e9002348e-CM001225d8ab30.cpe.net.cable.rogers.com) |
| 05:28.08 | *** join/#asterisk maxagaz (n=maxagaz@soho2.i-xanadu.com) |
| 05:33.46 | *** join/#asterisk JAMMAN2110 (n=james@unaffiliated/jamman2110) |
| 06:02.22 | *** join/#asterisk tommyfun (n=tommyfun@c-24-218-204-226.hsd1.ma.comcast.net) |
| 06:09.07 | *** join/#asterisk [8none1] (n=[8none1]@c-68-52-180-102.hsd1.tn.comcast.net) |
| 06:17.34 | *** join/#asterisk Iamnacho (i=Iamnacho@ip98-186-180-143.ks.ks.cox.net) |
| 06:24.42 | *** join/#asterisk bbt (n=sam@180.189.138.55) |
| 06:43.47 | dan__t | hmm |
| 06:43.54 | dan__t | b14ck, you around? |
| 06:45.52 | dan__t | I was wondering what the limits of a single channel working with bridges was. Can I have one channel part of more than one bridge at any given time? |
| 06:52.05 | *** join/#asterisk chendy (n=chatzill@119.122.163.43) |
| 06:56.13 | *** join/#asterisk baijum (n=baiju@122.166.46.113) |
| 07:03.17 | *** join/#asterisk JAMMAN2110 (n=james@unaffiliated/jamman2110) |
| 07:04.33 | *** join/#asterisk maxagaz (n=maxagaz@soho2.i-xanadu.com) |
| 07:05.14 | *** join/#asterisk Mw3 (i=mw3@mw3.hu) |
| 07:07.24 | *** join/#asterisk lozarythmic (n=lpraties@e1-1.ns500-1.ts.milt.as9105.net) |
| 07:11.31 | *** join/#asterisk Ad-Hoc (n=nimbus@62.169.216.185) |
| 07:13.10 | mchou | holy crap |
| 07:13.28 | mchou | google voice offers local number porting now |
| 07:13.59 | mchou | man, I've been out of it |
| 07:14.11 | mchou | this is just incredible to me |
| 07:16.49 | TJNII | If they offer numbers in your area... |
| 07:17.24 | mchou | TJNII: yup. that's the caveat |
| 07:18.14 | mchou | TJNII: I just don't understand how google is gonna make money on this |
| 07:19.03 | mchou | TJNII: not unless they have plans to turn into a "real" ITSP |
| 07:19.12 | TJNII | It's google |
| 07:19.18 | mchou | lol |
| 07:19.22 | TJNII | They're probably datamining the calls. |
| 07:19.37 | mchou | oh, no doubt |
| 07:19.48 | TJNII | still wants to know how Google knew to advertise Digi-Key to him. |
| 07:19.55 | mchou | but that's a pretty expensive way to dtatmine |
| 07:20.05 | mchou | datamine* |
| 07:20.31 | coppice | every time google has done something new, most people said "how can they make money out of that" |
| 07:20.47 | mchou | coppice: true |
| 07:21.16 | mchou | coppice: but we now understand their business plan in more detail |
| 07:21.54 | mchou | coppice: "targeted & pertinent advertising" |
| 07:22.05 | coppice | you know their existing plan. do you know their future plans? do you know exactly where android is planned to go? |
| 07:22.20 | mchou | android is easy |
| 07:22.39 | coppice | some parts are, but is that the whole picture |
| 07:22.40 | mchou | they want to be thier premier mobile search platform |
| 07:22.49 | mchou | the* |
| 07:22.56 | mchou | that's not hard to understand |
| 07:23.49 | mchou | you don't need to be warren buffet to understand that |
| 07:24.54 | mchou | especially as desktops losse relevance |
| 07:24.57 | coppice | they are doing things of a pretty broad nature in voice and video. I suspect your view of android is only part of their goal |
| 07:24.58 | mchou | lose* |
| 07:26.37 | *** join/#asterisk xrmx__ (n=rm@host59-59-dynamic.14-87-r.retail.telecomitalia.it) |
| 07:29.18 | TJNII | debates whether is would be easier to write his own embedded IP implementation or try and figure somebody else's out.... |
| 07:30.32 | coppice | embedded means many things to many people, but if it means something pretty small to you, you might find http://www.sics.se/~adam/uip/index.php/Main_Page useful |
| 07:31.24 | TJNII | Yea, that's what I was looking at |
| 07:32.13 | TJNII | By embedded I mean running on a 10Mhz mcu with 700 bytes of ram and a nic driver I wrote. |
| 07:33.25 | coppice | most small MCUs have quite limited RAM, making comms a pain. That's changing a bit with the newer chips |
| 07:33.49 | TJNII | Exactly |
| 07:34.16 | TJNII | If I roll it myself I will use the buffer on the NIC and DMA instead of buffering the data on the controller. |
| 07:34.59 | coppice | you have to. 700 is <1 ethernet packet |
| 07:35.26 | TJNII | Exactly |
| 07:35.45 | TJNII | Well, between 0-1 packets, depending on the packet |
| 07:36.02 | coppice | uIP runs on some pretty small MCUs |
| 07:51.34 | *** join/#asterisk baijum (n=baiju@122.166.46.113) |
| 08:04.55 | *** join/#asterisk icyValk77 (n=icyValk7@gateway.ash.thebunker.net) |
| 08:06.02 | *** join/#asterisk niekie (i=quasselc@dreamworld.bergnetworks.com) |
| 08:12.05 | *** join/#asterisk tzafrir__laptop (n=tzafrir@local.xorcom.com) |
| 08:15.47 | *** join/#asterisk tamiel (n=tamiel@213.30.183.226) |
| 08:20.18 | *** join/#asterisk UQlev (n=yuriy@nb11-125.static.cytanet.com.cy) |
| 08:24.07 | *** join/#asterisk sulex (n=sulex@88-149-154-95.static.ngi.it) |
| 08:28.54 | *** join/#asterisk pif (n=ldm@zenon.apartia.fr) |
| 08:31.32 | *** join/#asterisk wam (i=wam@unaffiliated/wam) |
| 08:32.24 | *** join/#asterisk gr0mit (n=tim@2001:8b0:3d9:0:4034:95ed:46d7:3a8c) |
| 08:36.25 | *** join/#asterisk baijum (n=baiju@122.166.46.113) |
| 08:36.58 | *** join/#asterisk [8none1] (n=[8none1]@c-68-52-180-102.hsd1.tn.comcast.net) |
| 08:42.25 | *** join/#asterisk Caplain (i=shayne@2001:470:5:fb:a1c7:96f5:9ebf:292a) |
| 08:49.53 | AppleBoy | which tz is file in? I can't remember |
| 09:00.48 | *** join/#asterisk Da-Geek (n=Da-Geek@ip-217-204-101-67.easynet.co.uk) |
| 09:12.40 | *** join/#asterisk bluOxigen (n=asad@static-host119-73-67-20.link.net.pk) |
| 09:40.50 | *** join/#asterisk cuco (n=Diego@local.xorcom.com) |
| 09:44.55 | *** join/#asterisk Da-Geek (n=Da-Geek@ip-217-204-101-67.easynet.co.uk) |
| 09:48.05 | *** join/#asterisk iksik (i=xk@livedata.pl) |
| 09:56.38 | *** join/#asterisk sergey (n=sergey@sergey-home.iks.ru) |
| 10:00.56 | *** join/#asterisk saftsack (n=oliver@p4FC709EA.dip0.t-ipconnect.de) |
| 10:02.37 | TJNII | is watching old TV shows |
| 10:02.59 | TJNII | You know, if the main character is supposed to be an asian, you'd think they'd get an asian actor. |
| 10:03.10 | TJNII | Even 1940s technology shows that. |
| 10:12.04 | *** join/#asterisk Omorika (n=omorika@89.201.165.226) |
| 10:12.07 | Omorika | hi |
| 10:12.10 | Omorika | a question |
| 10:12.41 | Omorika | is asterisk able to use all available cores on a multicore machine? |
| 10:13.56 | *** join/#asterisk Kchehab (n=kchehab@212.98.141.199) |
| 10:15.36 | *** join/#asterisk Slashman (n=Slash@ariane.fimasys.com) |
| 10:15.43 | Kchehab | how can i know that my Digium, Inc. Wildcard TE405P is running normaly ,how to make a loop test ip-->E1 port 1--->E1 port 2---->sip extension.any one can help me to draw this dialpan |
| 10:20.19 | *** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 10:20.31 | *** join/#asterisk Caplain (i=shayne@2001:470:5:fb:a1c7:96f5:9ebf:292a) |
| 10:25.22 | Kchehab | ? |
| 10:25.48 | tuxcrafter | http://debian.pastebin.com/da7a5d85 |
| 10:25.53 | tuxcrafter | hello everybody |
| 10:26.15 | tuxcrafter | i got an issue that a number forwarding is nog working all the time, sometimes it works but most times it does not |
| 10:26.59 | tuxcrafter | could somebody please help me out? if you need a full sid debug or the full configuration file of the extentions.conf i can pastbin it |
| 10:29.49 | tuxcrafter | i think the problem has someting todo with this part "Got SIP response 482 "Loop Detected" back from" |
| 10:30.10 | tuxcrafter | but i dont know what this exactly means and how to proceed |
| 10:34.41 | *** join/#asterisk TSM2 (n=the_soft@fw-lon1.wenn.com) |
| 10:36.52 | coppice | it means you are trying to route a call round in circles |
| 10:37.36 | JAMMAN2110 | Can anyone lend a hand with my FXO "pass thru" thing? Line has POTS phones and the FXO port, want either the standard phones or a VoIP phone to answer, but asterisk not to answer unless a VoIP phone answers the call. I know it can be done, just cant find details on how, and cant get it working here |
| 10:37.57 | JAMMAN2110 | And no, I cant use an ATA for the POTS phones :/ |
| 10:43.35 | tuxcrafter | coppice: i got a sip debug log form the call http://debian.pastebin.com/d5b98bbd4 0612182441 to 0208910330 that ends with "482 Loop Detected" |
| 10:43.39 | tuxcrafter | how can this be fixed? |
| 10:45.45 | tuxcrafter | this is the compleet extensions.conf http://debian.pastebin.com/d348c9262 |
| 10:46.27 | Kchehab | ,how to make a loop test ip SIPcall-->E1 port 1--->(Loop)E1 port 2---->sip extension local.kindly can any one help me to draw this dialpan in the extensions.conf |
| 10:59.51 | *** join/#asterisk nighty^ (n=nighty@x122091.ppp.asahi-net.or.jp) |
| 11:06.00 | *** join/#asterisk MatBoy (n=MatBoy@wiljewelwetenhe.xs4all.nl) |
| 11:06.33 | *** join/#asterisk cusco (n=trilili@2001:0:53aa:64c:e4:76e7:b2c9:8871) |
| 11:06.35 | cusco | hi |
| 11:06.40 | cusco | [Nov 9 11:03:04] WARNING[11881] /home/murf/asterisk/1.6.1/main/ast_expr2.y: non-numeric argument |
| 11:06.49 | cusco | how can I find out what argument is non-numeric? |
| 11:06.58 | cusco | that file does not exist |
| 11:08.22 | JAMMAN2110 | Better question, are you running asterisk from your home directory? |
| 11:08.43 | cusco | /etc/init.d/asterisk is calling it |
| 11:08.53 | cusco | murf is not my dir |
| 11:09.00 | cusco | it came hardcoded ? |
| 11:09.11 | cusco | there is no /home/murf |
| 11:09.26 | JAMMAN2110 | It shouldnt be calling anything from a home dir |
| 11:09.32 | cusco | so is this a better reply? |
| 11:09.35 | JAMMAN2110 | Unless its been told to |
| 11:10.06 | cusco | look... how can it say some error about "/home/murf/asterisk/1.6.1/main/ast_expr2.y:" other than not found |
| 11:10.07 | JAMMAN2110 | Did you setup asterisk yourself or is it a distro? |
| 11:10.13 | cusco | it is oviously hardcoded |
| 11:10.14 | cusco | myself |
| 11:10.52 | JAMMAN2110 | Where are you getting the error from? |
| 11:11.03 | cusco | full |
| 11:11.19 | cusco | it shows in cli also |
| 11:11.46 | JAMMAN2110 | Has it just started showing up or always been there? |
| 11:11.59 | cusco | not sure |
| 11:12.06 | *** join/#asterisk Tim_Toady (n=moi@adsl138-29.kln.forthnet.gr) |
| 11:12.12 | cusco | it has been here for a while |
| 11:14.28 | cusco | root@perfpbxr:/var/log/asterisk# locate ast_expr2.c |
| 11:14.28 | cusco | /usr/src/CURRENT_asterisk-1.6.1.1/main/ast_expr2.c |
| 11:14.28 | cusco | /usr/src/CURRENT_asterisk-1.6.1.1/utils/ast_expr2.c |
| 11:16.23 | cusco | http://paste.debian.net/51093/ |
| 11:16.43 | Kchehab | any one can help me by trouble shooting my digium card with dahdi |
| 11:16.54 | *** join/#asterisk bluOxigen (n=asad@static-host119-73-67-20.link.net.pk) |
| 11:17.19 | cusco | maybe... |
| 11:18.02 | *** join/#asterisk dkirker (n=dkirker@gateway0.openmobl.com) |
| 11:21.55 | *** join/#asterisk Ad-Hoc (n=nimbus@62.169.216.185) |
| 11:25.45 | *** join/#asterisk ccesario (n=ccesario@189-19-6-236.dsl.telesp.net.br) |
| 11:25.53 | *** join/#asterisk Malkor (n=marco@hlle-d9ba009b.pool.mediaWays.net) |
| 11:30.06 | tzafrir__laptop | cusco, ast_expr2.c is the first one. If you got a build error there, the error is likely to come from elsewhere |
| 11:30.54 | tzafrir__laptop | cusco, "/home/murf" ? with that explicit path? |
| 11:31.45 | tzafrir__laptop | I suspect it was fixed later on |
| 11:32.09 | tzafrir__laptop | Any chance you could try a later 1.6.1.x asterisk version? |
| 11:33.38 | *** join/#asterisk bbt (n=sam@180.189.138.55) |
| 11:34.46 | Kchehab | i get app_dial.c:1528 dial_exec_full: Unable to create channel of type 'DAHDI' (cause 0 - Unknown) |
| 11:34.53 | cusco | tzafrir__laptop: its a production system |
| 11:35.12 | cusco | tzafrir__laptop: We will migrate soon, Im trying to figure out the warnings meanwhile |
| 11:35.37 | tzafrir__laptop | Kchehab, what's the output of: dahdi_hardware; lsdahdi |
| 11:35.38 | tzafrir__laptop | please pastebin |
| 11:35.44 | Kchehab | when recieving a call that should be forwarded to span 1 in the E1 exten=>_X.,1,DIAL(DAHDI/1/${EXTEN}) |
| 11:36.56 | Kchehab | tzafrir__laptop kindly find it http://pastebin.com/m24c85169 |
| 11:38.11 | tzafrir__laptop | Kchehab, channels (in lsdahdi) are not '(In use)' |
| 11:38.13 | *** join/#asterisk mintos (n=mvaliyav@nat/redhat/x-bzwyweiycqwrecpm) |
| 11:38.22 | tzafrir__laptop | Asterisk does not have them defined |
| 11:38.30 | tzafrir__laptop | Please patebin your chan_dahdi.conf |
| 11:40.26 | Kchehab | tzafrir i paste it http://pastebin.com/m4b3315cf |
| 11:41.34 | tzafrir | Kchehab, is asterisk running? |
| 11:41.36 | tzafrir | asterisk -r |
| 11:41.42 | tzafrir | in it: dahdi restart |
| 11:43.15 | Kchehab | i did the output is |
| 11:43.15 | Kchehab | [Nov 9 03:44:26] WARNING[18884]: chan_dahdi.c:11879 pri_dchannel: PRI Error on span 0: We think we're the CPE, but they think they're the CPE too. |
| 11:43.19 | *** join/#asterisk Poincare (n=jefffnod@amp89.ampersant.be) |
| 11:44.22 | *** join/#asterisk icyValk77 (n=icyValk7@213.129.64.4) |
| 11:45.31 | tzafrir | If you connect a loop to yourself, it might be wise to put the second port as pri_net |
| 11:45.36 | tzafrir | rather than pri_cpe |
| 11:45.36 | Kchehab | tzafrir asteisk stopped ,where is the error in my config , |
| 11:46.36 | tzafrir | Kchehab, you asked the question about loopback on the asterisk-users list? |
| 11:47.11 | Kchehab | yes |
| 11:47.53 | Kchehab | tzafrir but now i noticed that Chan_dahdi is not configured well ,and i have problem on it |
| 11:48.41 | Kchehab | since i reload it |
| 11:53.56 | Kchehab | tzafrir rather than pri_cpe which means ? |
| 11:55.50 | *** join/#asterisk icyValk77 (n=icyValk7@gateway.ash.thebunker.net) |
| 11:58.00 | *** join/#asterisk elliot98 (n=windows@unaffiliated/elliot98) |
| 11:58.03 | elliot98 | hello |
| 11:58.12 | elliot98 | what is the difference between Callgroup and pickupgroup? |
| 12:03.28 | elliot98 | how does the system work? |
| 12:03.47 | elliot98 | do I set the callgroup or pickupgroup to answer when dialing *8#? |
| 12:14.24 | *** join/#asterisk Iamnach0 (i=Iamnacho@ip98-186-180-143.ks.ks.cox.net) |
| 12:15.07 | kaldemar | elliot98: members of a pickupgroup can pick up calls to members of the matching callgroup. |
| 12:19.03 | *** join/#asterisk garymc (n=garymc@host81-134-0-102.in-addr.btopenworld.com) |
| 12:19.31 | *** join/#asterisk czindy (n=czindy@91.120.30.42) |
| 12:21.03 | czindy | Hello! I have a problem with group configuration under chan_dahdi.conf. Is it possible to use more than 10 groups? |
| 12:23.46 | czindy | Here is my setting and error message: http://pastebin.com/d7750b2cf |
| 12:24.53 | Kchehab | Is there any management API or mudule or interface to Dahdi ? |
| 12:25.11 | Kchehab | or an applicatin can configure to E1 or SS7 |
| 12:26.20 | *** join/#asterisk bluOxigen (n=asad@static-host119-73-67-20.link.net.pk) |
| 12:28.09 | *** join/#asterisk knctrnl_ (n=aembrey@user-69-1-12-224.knology.net) |
| 12:32.35 | elliot98 | gotcha! |
| 12:32.36 | elliot98 | thanks |
| 12:32.48 | elliot98 | does 1.6 or 1.4 still have the maximum number of 63 callgroups? |
| 12:34.50 | *** join/#asterisk garymc (n=garymc@host81-134-0-102.in-addr.btopenworld.com) |
| 12:35.10 | garymc | anyone here with BT in UK isdn ? |
| 12:35.45 | czindy | Ok I found the answer maybe: Groups range from 0 to 63 |
| 12:36.29 | Chainsaw | garymc: Yes, dual BRI presentation going into Patton gateways. |
| 12:36.36 | Chainsaw | garymc: But I believe we have already spoken. |
| 12:42.26 | garymc | we have probably |
| 12:42.49 | garymc | Im just having trouble getting bT to allow me to pass my 0800 as a CLI to my called parties |
| 12:43.17 | *** join/#asterisk baijum (n=baiju@122.166.46.113) |
| 12:44.03 | garymc | my local business team are saying I need to pay for presentaion service and that I ownt beable to display any of my other numbers. Meaning when i dial out from any phone DID it will show my 0800 permanently |
| 12:44.38 | garymc | no matter what i set my CLI to.... |
| 12:45.04 | garymc | and my main number ID would be lost too. Id have to pay for this |
| 12:45.06 | tzafrir | to answer czindy: yes, group number are 0-63 |
| 12:46.08 | tzafrir | Kchehab, you don't need to just configure chan_dahdi. |
| 12:46.16 | tzafrir | You need to configure Asterisk |
| 12:48.24 | *** join/#asterisk deeperror (n=deeperro@d149-67-49-94.try.wideopenwest.com) |
| 12:52.49 | Chainsaw | garymc: Yes, your telco can (and will) filter your outbound number to a specific range. |
| 12:53.44 | Chainsaw | garymc: Updating that filter is extra work, which I'm sure BT will charge for. |
| 12:56.03 | *** join/#asterisk WinZ (n=winz@82.146.61.218) |
| 12:56.48 | garymc | but they tell me i can display 0800 and nothing else. I will lose ability to send my did range and main number |
| 12:57.35 | garymc | that i dont want |
| 12:57.57 | WinZ | guys, what could be the reason for flickering sound (ulaw) when calling *43 for echo test? |
| 12:58.09 | garymc | i want 4 phone extensions to display the 0800 number only. the rest would display the office number |
| 12:58.10 | WinZ | y-o-u a-r-e a-b-o-u-t... |
| 12:58.38 | WinZ | yesterday it was ok, now it's terrible sound for echo test and voicemail |
| 12:59.00 | *** join/#asterisk hugorebelo (n=hugorebe@200-171-132-124.completo.com.br) |
| 13:00.10 | *** join/#asterisk Godfather_ (n=Godfathe@187.Red-83-38-229.dynamicIP.rima-tde.net) |
| 13:01.53 | gr0mit | garymc, what are you trying to do exactly? |
| 13:03.32 | *** join/#asterisk TheDavidFactor (n=chatzill@fw1.safedataisp.net) |
| 13:05.45 | *** join/#asterisk Da-Geek (n=Da-Geek@217.204.101.67) |
| 13:06.23 | garymc | I want to pass my 0800 CID over to customers when i dial out. But only from certain phones. I can do this now with my DID range but BT are discarding the information when i send out the 0800 as a CID |
| 13:07.50 | *** join/#asterisk superbeef (n=superbee@74.84.194.4) |
| 13:08.24 | WinZ | can turning ACPI off somehow affect sound in Asterisk? |
| 13:09.45 | *** join/#asterisk Skeeter- (i=Skeeter-@c216.218.2-65.clta.globetrotter.net) |
| 13:10.05 | Skeeter- | Morning everyone |
| 13:11.10 | Skeeter- | i have about 30 active channel, is there anyway to remove the one not used, I think i used ChanSpy too many time and thats why i got so many channels active |
| 13:13.30 | *** join/#asterisk jamesh1 (n=jhenders@xob.neospire.net) |
| 13:14.21 | gr0mit | garymc, what you really wantit is for your 0800 number just be another option to send out |
| 13:14.36 | gr0mit | but unfortunately, BT won't do that. |
| 13:15.02 | gr0mit | BT presentation numbers are a real pain |
| 13:15.28 | gr0mit | have you thought of making your outgoing calls via VoIP? |
| 13:16.05 | gr0mit | This way, you can send any caller ID which you can demonstrate to a VoIP provider that you own |
| 13:17.36 | garymc | yeah, but thats why we got ISDN30 as the Broadband connection needed was not affordable |
| 13:18.03 | *** join/#asterisk dwery (n=dwery@nslu2-linux/dwery) |
| 13:18.07 | gr0mit | okay, so you need 30 simultaneous calls? |
| 13:18.11 | garymc | yes |
| 13:18.15 | garymc | plus internet usage |
| 13:18.53 | gr0mit | So you need about 3 Mbit/sec symmetrical |
| 13:19.18 | gr0mit | presumably, you got a quote for a 10 meg circuit? |
| 13:19.31 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
| 13:20.25 | garymc | something like that |
| 13:20.45 | gr0mit | you can get some very good deals on 10 Mb circuit's |
| 13:20.49 | garymc | even 4meg up and down was off the clock |
| 13:20.57 | gr0mit | unless you're out in the sticks |
| 13:21.02 | gr0mit | whereabouts are you? |
| 13:21.09 | garymc | yes out in the sticks sort of |
| 13:21.28 | garymc | we are based in Merseyside,but we are based in an old farm |
| 13:21.38 | garymc | which is sort of out the way |
| 13:21.50 | coppice | most of urban britain is based in an old farm |
| 13:22.12 | garymc | yep, but now they aint old farms |
| 13:22.23 | garymc | this still is out of the way from most places |
| 13:22.31 | garymc | like in 400 acres of land |
| 13:22.39 | gr0mit | well just ignore coppice, he hates .uk |
| 13:22.58 | gr0mit | :-) |
| 13:23.01 | coppice | hey, I just stood 2 weeks of it |
| 13:23.12 | gr0mit | i know - and you moaned like anything! |
| 13:23.35 | coppice | i thought i was most restrained |
| 13:23.35 | garymc | you came to the Uk coppice? |
| 13:23.51 | gr0mit | anyhow, PM me your postcode and let me look |
| 13:24.04 | gr0mit | you might get lucky, you never know |
| 13:24.12 | gr0mit | I have a tame ISP |
| 13:24.20 | gr0mit | who actually know what they're talking about |
| 13:25.41 | gr0mit | so, is it all calls you want to send the 0800 on? |
| 13:25.46 | gr0mit | or just some? |
| 13:26.14 | dwery | hello. I need to perform some actions when an incoming call has not been answered because the caller dropped the line. I tried using the "h" extension in different context but it doesn't get called. What should I use? |
| 13:28.12 | [TK]D-Fender | coppice = UK ex-pat |
| 13:28.49 | [TK]D-Fender | dwery: "h" in the context the call is in |
| 13:28.50 | gr0mit | indeed, and always reminds us! |
| 13:28.55 | [TK]D-Fender | dwery: those are the rules |
| 13:28.55 | *** join/#asterisk andres833 (n=andres83@190.144.75.22) |
| 13:28.57 | gr0mit | Some of us have to live here all the time |
| 13:29.10 | [TK]D-Fender | gr0mit: Really? Saying you can't leave? |
| 13:29.19 | *** join/#asterisk [8none1] (n=[8none1]@c-68-52-180-102.hsd1.tn.comcast.net) |
| 13:29.44 | gr0mit | I can leave, but I was born here and one gets used to it |
| 13:30.24 | dwery | [TK]D-Fender: trying again |
| 13:31.18 | *** join/#asterisk donnib (n=mmarines@0x555281d0.adsl.cybercity.dk) |
| 13:31.51 | donnib | hi |
| 13:35.25 | donnib | i have a peer http://pastebin.ca/1663363 which registers fine http://pastebin.ca/1663366 but aftet a while the registration fails http://pastebin.ca/1663367. The peer is a router with incorporated SIP adaptor. The ip of the peer is 192.168.1.1 and the server is 192.168.1.10. Can anybody figure out why it fails registration ? |
| 13:35.25 | dwery | [TK]D-Fender: ty. verbose was at 1 an was ot showing NoOp ! |
| 13:36.00 | donnib | btw, i am running Asterisk 1.4.22 |
| 13:36.46 | donnib | i have replaced my external ip (WAN ip) with X.X.X.X |
| 13:36.53 | donnib | in the abobe pastebin |
| 13:37.05 | donnib | *above |
| 13:37.39 | *** join/#asterisk SirThomas (n=tomc@mail.kendeco.com) |
| 13:39.25 | cusco | hi |
| 13:39.44 | cusco | how to make MONITORED calls accessible to anyone (permissions) ? |
| 13:39.55 | cusco | Monitor(gsm,/var/log/asterisk/asterisk_rec/outbound-${UNIQUEID},mb); |
| 13:40.03 | [TK]D-Fender | donnib: BAD : User-Agent: AVM FRITZ!Box Fon WLAN 7270 54.04.76 (Sep 14 2009) Call-ID: 33E437EE6BD8D920@192.168.1.1 |
| 13:40.24 | donnib | [TK]D-Fender: so this means ? |
| 13:40.35 | [TK]D-Fender | donnib: GOOD : User-Agent: AVM FRITZ!Box Fon WLAN 7270 54.04.67 (Dec 17 2008) Call-ID: B01243FDEBB8E7EA@192.168.178.1 |
| 13:40.43 | *** join/#asterisk mctweep (n=woopa@210-84-11-173.dyn.iinet.net.au) |
| 13:40.44 | donnib | oh |
| 13:40.50 | [TK]D-Fender | donnib: Sure doesn't look like the same device to me <- |
| 13:40.55 | donnib | the version difference ? |
| 13:40.55 | *** join/#asterisk retentiveboy (n=pdugas@atl.pra-corp.com) |
| 13:40.59 | [TK]D-Fender | donnib: And the IP subnet = WTF |
| 13:41.02 | mctweep | hi all, is there a command is asterisk to kick everyone off the server? |
| 13:41.12 | [TK]D-Fender | mctweep: "stop now" |
| 13:41.19 | donnib | well maybe when it reboots it uses another agent ? |
| 13:41.27 | [TK]D-Fender | donnib: BS |
| 13:41.31 | mctweep | but does that stop the server? |
| 13:41.36 | [TK]D-Fender | mctweep: Yes |
| 13:41.37 | donnib | what't wrong with the subnet ? |
| 13:41.47 | [TK]D-Fender | dobWhy the hell are they DIFFENERT? |
| 13:41.47 | mctweep | hmm anything that will get rid of people without stopping the server |
| 13:42.08 | [TK]D-Fender | mctweep: Going one by one and "soft hangup [channel]" |
| 13:42.13 | donnib | dunno why they are different |
| 13:42.16 | donnib | gotta look to see |
| 13:42.26 | donnib | you mean between the server and the peer ? |
| 13:42.36 | [TK]D-Fender | donnib: Better get a clue... UA's aren;t supposed to change. I think you don't know what you're looking at |
| 13:42.56 | donnib | those logs are legit and made by using same device |
| 13:43.08 | donnib | if somebody tells different then it's the adaptor |
| 13:43.13 | donnib | i mean the peer |
| 13:43.26 | mctweep | sorry i'm a noob what do you mean by channel |
| 13:43.48 | [TK]D-Fender | donnib: Your little box could be a PITA POS |
| 13:43.56 | donnib | huh ? |
| 13:44.00 | [TK]D-Fender | mctweep: call = channel |
| 13:44.10 | [TK]D-Fender | donnib: Pain In The Ass Piece Of Shit |
| 13:44.16 | mctweep | aah i dont want to hangup a call just kick a user off the pbx |
| 13:44.16 | donnib | :D |
| 13:44.28 | donnib | it could be |
| 13:44.31 | donnib | hope not |
| 13:44.34 | *** join/#asterisk jmacz (n=jmacz@190.144.75.22) |
| 13:44.42 | mctweep | have a whole lot of unreachable ppl that i wanna get rid of |
| 13:44.42 | [TK]D-Fender | mctweep: what does "kick off" even mean in your case? |
| 13:44.52 | mctweep | boot them off the pbx? |
| 13:45.03 | [TK]D-Fender | mctweep: there is no such thing as a "connection" anyway |
| 13:45.16 | [TK]D-Fender | mctweep: there are only calls <- |
| 13:45.55 | mctweep | not even registered users:P, how long does it take for the unreachables to disapear |
| 13:46.07 | [TK]D-Fender | mctweep: disappear from where? |
| 13:46.17 | DND | guys, what's the difference between snom 360 and 370 aside from bigger LCD on 370? |
| 13:46.42 | mctweep | from sip show peers |
| 13:47.06 | [TK]D-Fender | mctweep: .... its always going to show there. They are a friggen peer because YOU CONFIGURED THEM |
| 13:47.18 | [TK]D-Fender | mctweep: You saing you want to completely remove a peer for all time? |
| 13:47.20 | mctweep | yeaaa but is says UNREACHABLE :P |
| 13:47.34 | mctweep | no just get there status displaying the real thing |
| 13:47.46 | [TK]D-Fender | mctweep: Dumbass, that list doesn't show jsut active ly communicating devices, its show ALL PEERS |
| 13:47.53 | [TK]D-Fender | mctweep: thats its job. |
| 13:47.54 | mctweep | yeaa i can see that |
| 13:48.02 | [TK]D-Fender | mctweep: Then live with it |
| 13:48.13 | mctweep | so its going to sit on unreachable forever:P alright i will live with it |
| 13:48.15 | [TK]D-Fender | mctweep: there is nothing to kcik off. |
| 13:48.38 | [TK]D-Fender | mctweep: Shows all peers. the command wasn't "sip show active peers only" |
| 13:48.42 | *** join/#asterisk jaytee (n=jforde@unaffiliated/jaytee) |
| 13:48.52 | [TK]D-Fender | mctweep: and you're worrying about nothing |
| 13:49.27 | mctweep | probably yeaa trying to register with my TSP but another user whihc is now unreachable was on it before, dunno how longto wait before being able to connect toit, stuck at status request sent |
| 13:50.15 | *** join/#asterisk andres833 (n=andres83@190.144.75.22) |
| 13:50.43 | beek | mornin' [TK]D-Fender |
| 13:51.38 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 13:51.55 | *** join/#asterisk mikkel (n=mikkel@84-238-113-66.u.parknet.dk) |
| 13:51.56 | Kchehab | dahdi E1 config Isdn protocol should be like this for Span 1 span=1,1,0,ccs,hdb3,crc4 |
| 13:52.11 | [TK]D-Fender | mctweep: Registration has nothing to do with your peer |
| 13:52.27 | donnib | let me try to make the logs again, may be i did a mistake |
| 13:52.31 | donnib | brb |
| 13:53.06 | jaytee | morning beek |
| 13:53.08 | mctweep | k thanks |
| 13:53.21 | beek | morning jaytee |
| 13:53.26 | [TK]D-Fender | CRAZY PEOPLE |
| 13:53.35 | [TK]D-Fender | beek / jaytee : Mornin' |
| 13:53.46 | jaytee | morning [TK]D-Fender |
| 13:55.13 | jaytee | I just dread mondays so I think I'm going to listen to alot of Marley and Tosh today :-) |
| 13:55.49 | donnib | ok, here we go again. bad registration http://pastebin.ca/1663390 and good registration http://pastebin.ca/1663391 |
| 13:55.55 | *** join/#asterisk mintos (n=mvaliyav@nat/redhat/x-plogbdnbmbzsvoqe) |
| 13:56.24 | donnib | seems like it must have been my mistake with the UA |
| 13:56.32 | *** join/#asterisk txwikinger (n=quassel@sblug/member/txwikinger) |
| 13:56.41 | txwikinger | Anybody ever seen asterisk crashing (error code 1) without any error messages in logs? |
| 13:59.30 | *** join/#asterisk manxpower (n=ewieling@24.42.221.26) |
| 14:00.00 | manxpower | ~answers |
| 14:00.01 | infobot | well, answers is 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 |
| 14:00.52 | [TK]D-Fender | donnib: remove your port, permit & deny |
| 14:01.18 | *** join/#asterisk KavanS (n=KavanS@static-173-50-141-22.ptldor.fios.verizon.net) |
| 14:01.26 | *** join/#asterisk mumtazah (n=anees@203.82.79.102) |
| 14:03.41 | *** join/#asterisk bluOxigen (n=asad@static-host119-73-67-32.link.net.pk) |
| 14:06.37 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 14:07.34 | *** join/#asterisk Katty (n=asterisk@mail.copi-rite.com) |
| 14:07.45 | *** part/#asterisk mumtazah (n=anees@203.82.79.102) |
| 14:08.04 | [TK]D-Fender | Katty: Mew. |
| 14:08.34 | Katty | hi. |
| 14:08.47 | *** join/#asterisk malcolmd (n=malcolmd@pdpc/sponsor/digium/malcolmd) |
| 14:08.48 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 14:09.19 | *** join/#asterisk ManxPower-work (n=EWieling@24.42.221.26) |
| 14:13.14 | *** part/#asterisk WinZ (n=winz@82.146.61.218) |
| 14:16.40 | *** join/#asterisk dlynes (n=daniel@bas5-hamilton14-1096762451.dsl.bell.ca) |
| 14:16.49 | Skeeter- | How can I close ChanSpy channels??? |
| 14:17.00 | *** part/#asterisk dwery (n=dwery@nslu2-linux/dwery) |
| 14:17.13 | ManxPower-work | Skeeter-: hangup or spy on another channel |
| 14:17.46 | Skeeter- | doesnt work, I got like 30 active channels, and my phone has been online fot 120 hours |
| 14:18.00 | Skeeter- | i rebooted the phone, didnt change anything |
| 14:18.05 | ManxPower-work | Skeeter-: you are not making any sense. |
| 14:18.07 | [TK]D-Fender | Skeeter-: Show us your hangup attempt |
| 14:18.50 | ManxPower-work | Skeeter-: You didn't do something stupid like use exten => _. did you? |
| 14:19.00 | Skeeter- | nope |
| 14:19.02 | Skeeter- | stock configs |
| 14:19.13 | ManxPower-work | Then I guess you'd better start doing what [TK]D-Fender asks. |
| 14:19.15 | *** join/#asterisk voipmonk (n=voipmonk@dsl-67-55-17-41.acanac.net) |
| 14:20.05 | ManxPower-work | Skeeter-: There are NO working stock configs with Asterisk. Again, your statement makes no sense. |
| 14:20.08 | [TK]D-Fender | WTF are "stock configs"? |
| 14:20.20 | ManxPower-work | Smells like a frickin' GUI to me. |
| 14:20.42 | TheDavidFactor | make samples? (which are samples not configs, but still.....) |
| 14:20.44 | [TK]D-Fender | ManxPower-work: Oh I know full well he's a FreePBX user... |
| 14:21.00 | Skeeter- | http://pastebin.com/m41a50bc0 here are the channels |
| 14:21.02 | ManxPower-work | [TK]D-Fender: Ah. So he's asking here just because he's an asshole? |
| 14:21.08 | TheDavidFactor | then why didn't you send him elsewhere? |
| 14:21.30 | [TK]D-Fender | TheDavidFactor: Samples don't have Chanspy, or configured devices, etc. That is just psychotic. And the Samples should never be USED, only looked at on the side for "inspiration" |
| 14:21.41 | ManxPower-work | TheDavidFactor: The sample configs installed with "make samples" are files that try to show all the options, they are not designed to actually work |
| 14:22.09 | [TK]D-Fender | Skeeter-: now show what I ASKED FOR |
| 14:22.24 | Skeeter- | http://pastebin.com/m5737b759 my ext |
| 14:22.27 | [TK]D-Fender | TheDavidFactor: Because his problem isn't a GUI-based one |
| 14:22.28 | TheDavidFactor | well, I've seen at least one installation where someone used the samples and just modified them. All sorts of weird things happened |
| 14:22.47 | *** join/#asterisk leifmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 14:22.48 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 14:22.48 | TheDavidFactor | but it did (sort of) work |
| 14:23.03 | [TK]D-Fender | Skeeter-: now show what I ASKED FOR <------------- |
| 14:23.21 | Skeeter- | my ext settings right?? |
| 14:23.29 | *** join/#asterisk DelphiWorld (n=Miranda@41.201.116.181) |
| 14:23.38 | ManxPower-work | (9:18:07 AM) [TK]D-Fender: Skeeter-: Show us your hangup attempt |
| 14:23.50 | [TK]D-Fender | TheDavidFactor: there is no point to starting from the samples. Your dialplan will be completely different, you'll have your own patterns, providers, extensions for "local" devices, and whatever else... |
| 14:23.52 | Skeeter- | oh |
| 14:23.54 | Skeeter- | sec |
| 14:24.02 | [TK]D-Fender | Skeeter-: that's d-ing DIALPLAN. Means jack shit. |
| 14:24.20 | [TK]D-Fender | Skeeter-: Now show me what I ASKED FOR |
| 14:24.32 | *** join/#asterisk Anth8708 (n=Anth8708@client105.jdcc.edu) |
| 14:24.37 | Skeeter- | there u go http://pastebin.com/m3fd24793 |
| 14:24.55 | TheDavidFactor | true |
| 14:25.33 | [TK]D-Fender | Skeeter-: Now show me what I ASKED FOR <----------- |
| 14:26.04 | Skeeter- | well that was my attempt to hangup |
| 14:26.07 | Skeeter- | from the CLI |
| 14:26.10 | ManxPower-work | Looks like a typical Monday. Have fun [TK]D-Fender |
| 14:26.17 | *** part/#asterisk ManxPower-work (n=EWieling@24.42.221.26) |
| 14:26.39 | [TK]D-Fender | Skeeter-: And yes that clearly shows you running FreePBX and not your own setup. |
| 14:26.49 | [TK]D-Fender | Skeeter-: And doesn't NOT show you trying to kill the call from CLI |
| 14:27.29 | Skeeter- | ok, so how am I suppose to kill a call if Hanging up doesnt work |
| 14:27.43 | [TK]D-Fender | Skeeter-: Holy friggen crap.... |
| 14:28.02 | [TK]D-Fender | Skeeter-: "sof hangup [channel] |
| 14:28.06 | [TK]D-Fender | Skeeter-: "soft hangup [channel]" |
| 14:28.27 | [TK]D-Fender | Skeeter-: You jsut showed another call attempt. that has NOTHINg to do with killing off OTHER CALLS |
| 14:29.07 | Skeeter- | thats ok |
| 14:29.16 | Skeeter- | i miss formuled my question then |
| 14:29.25 | Skeeter- | that soft hangup ... cmd |
| 14:29.31 | Skeeter- | thats what i needed |
| 14:29.43 | Skeeter- | Skeeter-> i have about 30 active channel, is there anyway to remove the one not used, I think i used ChanSpy too many time and thats why i got so many channels active |
| 14:30.04 | Skeeter- | my bad again |
| 14:30.05 | [TK]D-Fender | Skeeter-: And somehow you thought placing another call would help? |
| 14:30.10 | Skeeter- | nope |
| 14:30.20 | Skeeter- | i just realize it after 20 calls |
| 14:30.21 | [TK]D-Fender | Skeeter-: well thats the last thing you showed me |
| 14:30.43 | *** join/#asterisk tommyfun (n=tommyfun@c-24-218-204-226.hsd1.ma.comcast.net) |
| 14:30.51 | Skeeter- | u ask for the hangup attemps, i though that you wanted to see what happens when i hangup |
| 14:30.53 | *** part/#asterisk DelphiWorld (n=Miranda@41.201.116.181) |
| 14:31.55 | Skeeter- | If the soft hangup commands doesnt work, it there any other way to kill a channel?? |
| 14:32.22 | [TK]D-Fender | Skeeter-: ....... |
| 14:32.33 | cusco | how can I tell asterisk to write the MONITORED files with system wide read permissions?= |
| 14:33.09 | [TK]D-Fender | cusco: Have monitor call a script to adjust the rights after its finished |
| 14:34.47 | cusco | that sounds good, but im not sure about the syntax in the ael for that... |
| 14:35.38 | cusco | I have for instance: Monitor(gsm,/var/log/asterisk/asterisk_rec/outbound-${UNIQUEID},mb); |
| 14:36.43 | Skeeter- | [TK]D-Fender: your cmd doesnt work in my case |
| 14:37.08 | [TK]D-Fender | Skeeter-: still haven't learned... |
| 14:37.37 | [TK]D-Fender | cusco: the app has the same parameters regraledd of AEL |
| 14:37.49 | [TK]D-Fender | regardless |
| 14:37.54 | [TK]D-Fender | WOW.. its monday all right... |
| 14:38.07 | cusco | so would I call it like System(chmod a+rx gsm,/var/log/asterisk/asterisk_rec/outbound-${UNIQUEID); |
| 14:38.11 | cusco | ? |
| 14:38.23 | cusco | err |
| 14:38.29 | cusco | so would I call it like System(chmod a+rx /var/log/asterisk/asterisk_rec/outbound-${UNIQUEID); |
| 14:39.16 | *** join/#asterisk bluOxigen (n=asad@static-host119-73-77-237.link.net.pk) |
| 14:39.42 | angryuser | can someone help me to compile sangoma bri driver for callweaver, their chan is empty |
| 14:39.49 | angryuser | thank you |
| 14:40.33 | angryuser | Or at least tell me the version which works |
| 14:40.43 | cusco | [TK]D-Fender: as I'm not sure, can you (or somebody else) just confirm me that it would not break my system=? |
| 14:41.10 | Skeeter- | [TK]D-Fender: ill just reboot the server tonight |
| 14:41.38 | [TK]D-Fender | Skeeter-: Yeah... cause thinking = hard |
| 14:41.48 | [TK]D-Fender | cusco: Missing a "}" |
| 14:42.14 | [TK]D-Fender | cusco: Monitor() can call a script directly as opposed to doing it as a dialplans tep. |
| 14:42.35 | [TK]D-Fender | cusco: consider the negative effect of a hangup... System() won't get called |
| 14:43.54 | Skeeter- | im not a pro like you, i havent spend most of my life on asterisk, thats why i am here for help |
| 14:44.30 | [TK]D-Fender | [09:18]<[TK]D-Fender>Skeeter-: Show us your hangup attempt |
| 14:44.47 | [TK]D-Fender | Skeeter-: I asked you to SHOW YOUR HANGUP ATTEMPT over HALF AN HOUR AGO |
| 14:45.01 | [TK]D-Fender | Skeeter-: And all you say is "doesn't work" |
| 14:45.01 | leifmadsen | [TK]D-Fender: have you used mpg123 in MOH situations? Do you know if there are usually multiple mpg123 processes when using it? |
| 14:45.32 | cusco | [TK]D-Fender: sorry yes Im missing a }, where can I read about the syntax, like where to place the script in Monitor() |
| 14:45.43 | Skeeter- | ok, can you clarify the hangup attempt, u want to see the log using soft hangup>?? |
| 14:45.44 | [TK]D-Fender | leifmadsen: IIRC its a single instance and the others hook into the stream... and I've never had to switch back to it |
| 14:45.55 | leifmadsen | thx |
| 14:46.03 | [TK]D-Fender | Skeeter-: I want to see you friggen attemtp to kill the call from CLI. |
| 14:46.11 | [TK]D-Fender | Skeeter-: Seriously... anyone awake in there? |
| 14:46.12 | Skeeter- | thanks |
| 14:46.47 | Skeeter- | http://pastebin.com/m2b073021 |
| 14:47.01 | cusco | im reading at http://www.voip-info.org/wiki/view/Asterisk+cmd+Monitor |
| 14:47.12 | [TK]D-Fender | Skeeter-: And....? |
| 14:47.27 | cusco | it only states Monitor(ext,basename,flags) |
| 14:47.30 | [TK]D-Fender | cusco: WIKI = random value. "core show application meetme" |
| 14:47.37 | [TK]D-Fender | cusco: err.. monitor |
| 14:49.22 | Skeeter- | [TK]D-Fender: the channel is still up, if 'soft hangup [channel]' succed, what should i see |
| 14:49.48 | [TK]D-Fender | Skeeter-: Where do I see this? |
| 14:50.09 | *** join/#asterisk deeperror (n=deeperro@76.226.172.218) |
| 14:50.14 | *** join/#asterisk SuPrSLuG (n=SuPrSluG@firewall-a.buf.ny.i-evolve.net) |
| 14:50.27 | Skeeter- | Requested Hangup on channel 'SIP/260-b58af600' it doesnt say if it succed |
| 14:51.05 | [TK]D-Fender | Skeeter-: Where's the #&^ing pastebin showing me active channels after your God-damned attempt? |
| 14:51.16 | [TK]D-Fender | Skeeter-: this hand holding BS with you is tiresome. |
| 14:52.21 | [TK]D-Fender | [09:50]<Skeeter->Requested Hangup on channel 'SIP/260-b58af600' it doesnt say if it succed <--- where do we see you LOOKING? |
| 14:52.27 | [TK]D-Fender | gah |
| 14:52.59 | Skeeter- | http://pastebin.com/m5509e995 |
| 14:53.01 | Skeeter- | sorry |
| 14:53.16 | Skeeter- | i looked it up for myself, but didnt show it, the channel is still listed |
| 14:55.00 | [TK]D-Fender | Skeeter-: try killing a bunch of them |
| 14:55.23 | Skeeter- | [TK]D-Fender: aight |
| 14:56.20 | Skeeter- | Request that a channel be hung up. The hangup takes effect the next time the driver reads or writes from the channel |
| 14:56.52 | Skeeter- | each channels seems to be linked to another one, does that make sense?? |
| 14:57.13 | [TK]D-Fender | Skeeter-: Spying on yourself in circles. No, little of what you do makes sense |
| 14:57.50 | Skeeter- | i disabled that feature |
| 14:59.13 | eppigy | orning |
| 15:00.55 | donnib | [TK]D-Fender: Sorry, was called to a meeting. you told me to remote the port and permit deny, so i should just leave those settings empty ? |
| 15:01.02 | *** part/#asterisk manxpower (n=ewieling@24.42.221.26) |
| 15:02.07 | donnib | *remove |
| 15:04.23 | Katty | hi eppigy |
| 15:06.45 | *** join/#asterisk slinksh0t (n=slinksh0@c-76-108-31-194.hsd1.fl.comcast.net) |
| 15:08.26 | *** part/#asterisk donnib (n=mmarines@0x555281d0.adsl.cybercity.dk) |
| 15:10.07 | *** join/#asterisk ickmund (n=magnus@ada-bcn-fw01.adamoeurope.com) |
| 15:10.36 | *** join/#asterisk Naikrovek (n=jjohnson@unaffiliated/naikrovek) |
| 15:11.58 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 15:14.39 | *** join/#asterisk bsaxon (n=bsaxon@12.68.234.174) |
| 15:14.53 | *** join/#asterisk Buklov (n=buklov@213.138.71.254) |
| 15:18.08 | Kchehab | i want to test my E1 digium card with 4 ports-i attached port 1 with port 4 by a cross E1 cable(loop back) |
| 15:18.47 | Kchehab | now i am sennding a call to span 1 how to redirect it to span 4 |
| 15:19.13 | Kchehab | in order span 4 to forrward the request as sip invite to 999999@xx.xx.xx.xx |
| 15:20.11 | *** join/#asterisk malcolmd (n=malcolmd@pdpc/sponsor/digium/malcolmd) |
| 15:20.11 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 15:22.06 | *** join/#asterisk baijum (n=baiju@122.166.149.235) |
| 15:22.39 | *** join/#asterisk ManxPower-work (n=EWieling@24.42.221.26) |
| 15:24.28 | Katty | screams, rips hair out |
| 15:24.54 | Katty | breathes |
| 15:28.39 | *** join/#asterisk chriztian (n=chriztia@190.43.68.150) |
| 15:28.50 | elliot98 | what is the maximum number of callgroups? |
| 15:28.55 | elliot98 | is it still 0-63? |
| 15:29.29 | elliot98 | everything ok Katty? |
| 15:30.35 | Katty | no |
| 15:31.14 | *** join/#asterisk ehsjoar (n=ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 15:31.29 | elliot98 | oh, ok, I mean besides your hair being pulle dout |
| 15:31.31 | elliot98 | *pulled |
| 15:31.35 | elliot98 | what happenend? |
| 15:31.39 | Katty | ^_- |
| 15:32.10 | Katty | stupidity happened. |
| 15:34.18 | elliot98 | you...stupid?? |
| 15:35.13 | leifmadsen | HI! |
| 15:35.27 | Katty | heh |
| 15:35.36 | Katty | mister madsen, i'm cranky :< |
| 15:35.45 | Katty | let's hug. |
| 15:35.47 | Katty | hugs leifmadsen |
| 15:36.24 | leifmadsen | acknowledges said hug, and reciprocates with a power hug |
| 15:36.34 | elliot98 | awwww |
| 15:42.50 | *** join/#asterisk putnopvut (n=putnopvu@asterisk/master-of-queues/mmichelson) |
| 15:42.50 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 15:46.40 | Katty | infobot: music on hold? |
| 15:46.47 | Katty | infobot: moh? |
| 15:46.48 | infobot | i heard moh is Music On Hold. Good information about how to set it up in the various possible ways can be found at http://www.voip-info.org/wiki/index.php?page=Asterisk+config+musiconhold.conf, or originally from http://www.freeplaymusic.com |
| 15:46.51 | *** join/#asterisk [8none1] (n=[8none1]@cerberus.franklinamerican.com) |
| 15:47.44 | jaytee | morning Katty *hugs* |
| 15:48.14 | *** join/#asterisk Deeewayne (n=dwayne@75.76.254.162) |
| 15:48.14 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 15:48.21 | Katty | hugs jaytee |
| 15:48.27 | Katty | infobot: SetMusicOnHold? |
| 15:48.44 | Katty | infobot: SetMusicOnHold is http://www.voip-info.org/wiki/view/Asterisk+cmd+SetMusicOnHold |
| 15:48.45 | infobot | okay, Katty |
| 15:49.33 | *** join/#asterisk ppc (n=ppc@64.72.127.39) |
| 15:49.54 | Katty | jaytee: how's your day getting off? |
| 15:50.17 | jaytee | ok so far |
| 15:50.37 | jaytee | had a crappy night, kept waking up during the night |
| 15:50.41 | Katty | :< |
| 15:50.56 | Katty | that's never fun |
| 15:51.12 | hardwire | my day isn't getting off. |
| 15:51.13 | elliot98 | day is just starting for you guys? |
| 15:51.17 | hardwire | I wish it was. |
| 15:51.39 | Katty | well i know i'd feel 100% better if i drank some caffeine. |
| 15:51.48 | Katty | but i know the bad side effects of it, so i'm doing my best to resist. |
| 15:52.17 | elliot98 | catch 22s of life |
| 15:55.33 | Skeeter- | anyone remember the phone number to get a mediatrix IP |
| 15:55.45 | Skeeter- | something like *0*# |
| 15:58.49 | *** join/#asterisk Defraz (n=Defraz@corp.fuzecore.com) |
| 15:59.42 | *** join/#asterisk jpeeler (n=jpeeler@asterisk/digium-software-dev/jpeeler) |
| 16:00.56 | *** join/#asterisk CunningPike (n=CunningP@S01060014bf81366b.vc.shawcable.net) |
| 16:02.19 | *** part/#asterisk deeperror (n=deeperro@76.226.172.218) |
| 16:03.35 | *** join/#asterisk lesouvage (n=lesouvag@82.73.69.76) |
| 16:05.45 | beek | Katty: Hello! |
| 16:06.17 | thehar | Katty: <3 |
| 16:06.25 | thehar | that was in response to your <3 yesterday! |
| 16:06.58 | Katty | :> |
| 16:07.01 | Katty | hugs beek |
| 16:10.59 | *** join/#asterisk af_ (n=getsmart@88-149-240-33.dynamic.ngi.it) |
| 16:11.11 | *** join/#asterisk ayeso (n=chatzill@ext-52.sagetelecom.net) |
| 16:11.49 | ayeso | If i use the manager interface to originate a call, can I put something in the dial plan to echo something back to the manager interface? |
| 16:12.12 | [TK]D-Fender | ayeso: huh? |
| 16:13.55 | ayeso | [TK]D-Fender: So if I telnet to the asterisk manager interface and issue an originate command, it will drop the call to whatever context in extensions.conf i specify. i want to detect if its an answering machine with AMD, and then echo back some info to the connected manager interface. |
| 16:14.24 | ayeso | [TK]D-Fender: looks like i could use getvar |
| 16:14.34 | [TK]D-Fender | ayeso: What "connected manager"? the call is not connected to AMI... |
| 16:15.19 | [TK]D-Fender | ayeso: And there are dialplan apps to send ad-hoc messages over AMI |
| 16:15.34 | *** join/#asterisk nsgn (n=nsgn4@rrcs-24-153-206-251.sw.biz.rr.com) |
| 16:15.34 | *** join/#asterisk ccesario (n=ccesario@189-19-6-236.dsl.telesp.net.br) |
| 16:15.42 | ayeso | well when you connect to the manager interface an originate a call you get a response about what happened, I want to get a custom responce if you will. |
| 16:16.19 | ayeso | [TK]D-Fender: Is there an app for that? I havent seen this... |
| 16:16.26 | nsgn | so is there a way to offer "after hours" ringing ability to employees. we have our system ring straight to voicemail after hours, but we have some employees that would like to press some key on their touchtone phones to be dropped into the actual ringall group after hours |
| 16:16.37 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 16:16.45 | nsgn | stupid question. obviously there's a way, i'm rather curious what the most straightforward way would be |
| 16:16.55 | [TK]D-Fender | ayeso: then your Channel: has to be Local |
| 16:17.12 | ayeso | [TK]D-Fender: It is.... |
| 16:17.16 | [TK]D-Fender | nsgn: Its your dialplan... do whatever you want |
| 16:17.28 | *** join/#asterisk aidinb (n=Aidin@24-182-32-138.static.lnbh.ca.charter.com) |
| 16:17.30 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 16:18.18 | ayeso | [TK]D-Fender: so do you think i should just set a channel variable and then request that var through AMI? or do you know of an applicaton that will trigger the responce? |
| 16:18.36 | nsgn | [TK]D-Fender, currently it's just a day/night control sort of thing. they specifically want a key to be listened for during the outgoing message playback on their main voicemail box. is this a practical approach? |
| 16:18.44 | *** join/#asterisk |Cybex| (n=John@atwork-20.r-212.178.82.atwork.nl) |
| 16:19.29 | [TK]D-Fender | nsgn: voicemail() already has 2 escape options : * & # |
| 16:19.34 | [TK]D-Fender | nsgn: Use them as you wish |
| 16:19.59 | nsgn | oh, nice. thanks. i'm rusty on this stuff |
| 16:20.07 | nsgn | i'll play with those before working on anything more complex |
| 16:21.07 | *** join/#asterisk wonderworld (n=ww@ip-62-143-22-226.unitymediagroup.de) |
| 16:21.10 | *** join/#asterisk spck (n=spck@nwblwi2-l10-1172.nwblwi.tds.net) |
| 16:21.16 | spck | morning all |
| 16:22.12 | spck | anyone have any advice on managing caller id for a business with multiple DID's? |
| 16:22.36 | ManxPower-work | spck: My advice is don't. |
| 16:22.44 | p3nguin | You could use a "main" number for caller ID. |
| 16:22.46 | Katty | i'd ask them how they want it. |
| 16:22.56 | Katty | we have 3 mini companies in our building. |
| 16:23.14 | p3nguin | You could use a number per dept. |
| 16:23.19 | Katty | and the callerid is set for each mini company on the way out. transfers to cellphones callerid isn't touched. it's just resent out the way it came in. |
| 16:25.34 | nsgn | [TK]D-Fender, seems from my toying that 0 is a third escape from voicemail. within the proper context it should dial to a "receptionist". all i can't determine is how to configure where the hell it dials to |
| 16:25.57 | [TK]D-Fender | nshSorry, meant * & 0 |
| 16:26.22 | [TK]D-Fender | nsgn: "o" <- Asterisk Standard Extension. |
| 16:26.30 | [TK]D-Fender | nsgn: Dialplan 101 |
| 16:26.56 | russellb | [TK]D-Fender: everything you said was fine, until that last comment |
| 16:26.58 | russellb | unnecessary. |
| 16:27.42 | [TK]D-Fender | russellb: gotta know your Asterisk Standard Extensions :) |
| 16:28.03 | nsgn | russellb, thanks. [TK]D-Fender, its cool, i appreciate any help offered |
| 16:28.04 | russellb | If you can't help people without additional comments to belittle them, then you should not help. |
| 16:28.08 | [TK]D-Fender | considers hijacking "Gotta Catch 'em All" from that sill Japanimation show... |
| 16:28.22 | [TK]D-Fender | silly* |
| 16:28.24 | russellb | That was a pretty minor instance as compared to others, but it drives me nuts. |
| 16:29.11 | [TK]D-Fender | russellb: yeah, I have my quirks. goes over better with some than others, but I do try to work at it... |
| 16:29.25 | russellb | Thank you. |
| 16:29.49 | [TK]D-Fender | russellb: We're cool... as always ... |
| 16:29.54 | leifmadsen | agrees with russellb |
| 16:30.19 | [TK]D-Fender | leifmadsen: ditto :p |
| 16:30.35 | [TK]D-Fender | leifmadsen: Cryin' shame you're off in TO... I could use a drummer :) |
| 16:30.39 | ManxPower-work | I found that taking a month or three off from helping here was good. |
| 16:30.50 | ManxPower-work | Take a vacation [TK]D-Fender |
| 16:30.57 | eppigy | NEIN |
| 16:31.07 | [TK]D-Fender | ManxPower-work: And your propensity to leaving as soon as the risk factor mounts helps ;) |
| 16:31.08 | *** part/#asterisk ehsjoar (n=ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 16:31.15 | leifmadsen | [TK]D-Fender: I have electronic drums and midi inputs into my computer; just send me some recordings and I'll lay a track of them |
| 16:31.18 | leifmadsen | over* |
| 16:31.24 | [TK]D-Fender | leifmadsen: Live baby! |
| 16:31.38 | leifmadsen | shrugs |
| 16:31.45 | ManxPower-work | [TK]D-Fender: I think you mean "as soon as the stupidity factor mounts" |
| 16:31.49 | [TK]D-Fender | leifmadsen: pm an e-mail and I'll send you the book I'm working on... |
| 16:32.01 | [TK]D-Fender | ManxPower-work: Your words.... |
| 16:32.20 | *** join/#asterisk nsgn (n=nsgn4@rrcs-24-153-206-251.sw.biz.rr.com) |
| 16:32.21 | leifmadsen | [TK]D-Fender: my first name at my full name dot com |
| 16:32.54 | ManxPower-work | The only questions we ever seem to get here are ones that could easily be answered by reading some docs or are so wildly insane nobody should be trying to do whatever they are trying to do. |
| 16:33.12 | russellb | If you can't handle that, you're welcome to leave |
| 16:33.20 | ManxPower-work | russellb: And I frequently do. |
| 16:33.31 | russellb | k :-) |
| 16:33.40 | eppigy | i love you all |
| 16:34.25 | ManxPower-work | I must admit it was quite funny to see a new Asterisk user who would not even listen to advice saying that [TK]D-Fender doesn't know what he's talking about. |
| 16:34.32 | ManxPower-work | (that was yesterday) |
| 16:36.05 | [TK]D-Fender | ManxPower-work: Trick is knowing when to defer. I do this on just about anything coppice talks about, you for higher networking, any dev for the nitty gritty code bits, etc. |
| 16:36.24 | [TK]D-Fender | ManxPower-work: I tend to speak about things I know about and announce where my knowledge ends. |
| 16:36.52 | Qwell | [TK]D-Fender: You're...working on a book? |
| 16:36.56 | *** join/#asterisk thazza (n=thazza@124-254-81-140-static-dsl.ispone.net.au) |
| 16:37.41 | ManxPower-work | [TK]D-Fender: I try to replace me saying to someone "You're a fsckin' idiot" with "/part #asterisk". My success in doing that varies, so you know. |
| 16:37.57 | *** join/#asterisk ryduh (n=ryduh@204.16.143.186) |
| 16:39.12 | dlynes | If the asterisk server is running as 'root', is there a way to connect to it, and run a couple of commands as a non-privileged user? |
| 16:39.27 | ManxPower-work | Qwell: Titled "OCD and #Asterisk: A Love Story". |
| 16:39.29 | ManxPower-work | ducks |
| 16:39.30 | *** join/#asterisk DelphiWorld (n=Miranda@196.20.95.121) |
| 16:39.44 | dlynes | i.e. say if I wanted to check the status of asteriskdb keys, or manipulate them? |
| 16:39.53 | DelphiWorld | what [weather-wakeup] mean? |
| 16:40.08 | dlynes | DelphiWorld: it's a context in a given configuration file |
| 16:40.29 | DelphiWorld | dlynes: right but i see sevral extension |
| 16:40.36 | DelphiWorld | i called *6223 |
| 16:40.42 | DelphiWorld | and is requested a time |
| 16:40.43 | yang | Has anyone come accross IPv6 supported phones ? |
| 16:40.43 | dlynes | DelphiWorld: and? |
| 16:40.44 | [TK]D-Fender | Qwell: Playlist for jamming. |
| 16:40.57 | DelphiWorld | is called me i answered it but i heare just moh |
| 16:40.57 | [TK]D-Fender | Qwell: Though I should probably author a book for * |
| 16:41.07 | dlynes | DelphiWorld: do you have a question? |
| 16:41.11 | leifmadsen | dlynes: that's what 'sudo' is for |
| 16:41.30 | DelphiWorld | dlynes: i want to know what *6223 do |
| 16:41.34 | [TK]D-Fender | leifmadsen: E-mail sent BTW |
| 16:41.39 | dlynes | DelphiWorld: and maybe some useful information? |
| 16:41.46 | [TK]D-Fender | DelphiWorld: Does whatever you coded it to |
| 16:41.57 | DelphiWorld | [TK]D-Fender: ;) |
| 16:42.06 | DelphiWorld | see: exten => *61,2,AGI(nv-weather.php) |
| 16:42.07 | dlynes | DelphiWorld: we can't tell you what it does, unless we actually see some code (that doesn't mean copy and paste to the channel, either.) |
| 16:42.08 | p3nguin | Read the dialplan. |
| 16:42.11 | DelphiWorld | i see it in elastix |
| 16:42.15 | dlynes | ~pb |
| 16:42.16 | infobot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , or apt-get install pastebinit |
| 16:42.25 | [TK]D-Fender | DelphiWorld: GUI's are not supported here <- |
| 16:42.26 | ManxPower-work | ~elastix |
| 16:42.27 | infobot | methinks elastix is a full linux distro that includes FreePBX and other 3rd party add-ons. It is all this extra stuff which makes Elastix VERY difficult to support, and is not supported in #asterisk. Try asking in #Elastix or on their forums at http://www.elastix.org |
| 16:42.33 | [TK]D-Fender | DelphiWorld: Go ask them what their code does |
| 16:42.35 | dlynes | DelphiWorld: wrong channel...you want to /join #elastix and/or /join #freepbx |
| 16:42.46 | DelphiWorld | ;) |
| 16:42.47 | DelphiWorld | ok |
| 16:43.13 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 16:43.35 | [TK]D-Fender | [11:37]<ManxPower-work>[TK]D-Fender: I try to replace me saying to someone "You're a fsckin' idiot" with "/part #asterisk". My success in doing that varies, so you know. <- Actually... I think it does you some good... far more successful than not |
| 16:43.37 | p3nguin | leifmadsen: Isn't there some way to configure "users" in asterisk so that you don't have to give "sudo asterisk" rights to lowly users? |
| 16:43.40 | dlynes | leifmadsen: ok, so the '-s' option doesn't help me out with that then, eh? |
| 16:43.54 | leifmadsen | nope |
| 16:43.59 | dlynes | leifmadsen: or what p3nguin said, too |
| 16:44.05 | ManxPower-work | p3nguin: lowly users should not have access to the Asterisk CLI. |
| 16:44.08 | *** part/#asterisk DelphiWorld (n=Miranda@196.20.95.121) |
| 16:44.08 | leifmadsen | there is not concept of "users" in asterisk -- there is just the asterisk process |
| 16:44.28 | leifmadsen | the rest is an OS issue |
| 16:46.45 | *** join/#asterisk andres833 (n=andres83@190.144.75.22) |
| 16:46.54 | [TK]D-Fender | p3nguin: "Users" shouldn't have access to CLI |
| 16:47.12 | leifmadsen | p3nguin: btw -- https://issues.asterisk.org/view.php?id=11123 |
| 16:47.16 | leifmadsen | dlynes: also see above |
| 16:47.30 | leifmadsen | try asterisk 1.6.2, and see the cli_permissions.conf file |
| 16:48.16 | [TK]D-Fender | leifmadsen: Ok... increasingly cool... |
| 16:48.37 | *** join/#asterisk Scunizi (n=mark@69.199.151.114) |
| 16:51.19 | *** join/#asterisk puzzled (n=patrick@188.91.218.81) |
| 16:54.35 | *** part/#asterisk thazza (n=thazza@124-254-81-140-static-dsl.ispone.net.au) |
| 16:56.33 | Anth8708 | hey guys, does anyone know if support for polycom's enhanced blf is going to be scheduled for work any time soon? This says no (as of July), but I was hoping that perhaps something had changed and it hadn't been updated: https://issues.asterisk.org/view.php?id=10354 |
| 16:57.17 | Anth8708 | It would be an amazing thing for many people, being able to have a "buddy" you can actually hear ringing and do a PU by pressing the line keky |
| 16:57.24 | spck | anyone know what CALLERID(DNID) is for? |
| 16:57.57 | *** join/#asterisk Failrar (n=Failrar@2001:470:1f15:316:2a0:d1ff:fe4e:e802) |
| 16:57.58 | Anth8708 | spck: http://www.voip-info.org/wiki/view/Asterisk+Detailed+Variable+List |
| 16:58.08 | *** join/#asterisk ReDNeQ (i=ReDNeQ@70.114.229.58) |
| 16:59.13 | *** join/#asterisk SirThomas (n=tomc@mail.kendeco.com) |
| 17:00.40 | dlynes | leifmadsen: thanks |
| 17:00.42 | p3nguin | spck: http://www.voip-info.org/wiki/view/Asterisk+func+callerid |
| 17:00.51 | dlynes | leifmadsen: 1.6.2 is out now? |
| 17:00.59 | leifmadsen | dlynes: 1.6.2.0-rc4 is out |
| 17:01.08 | leifmadsen | and I'm spinning a new set of RCs right now |
| 17:01.20 | dlynes | leifmadsen: ah, ok |
| 17:01.29 | leifmadsen | 1.6.2.0 full should be out in the next couple of weeks hopefully |
| 17:01.38 | leifmadsen | that's my general feeling anyways |
| 17:01.52 | Qwell | dlynes: do me a favor.. attach a console log of trying to start Asterisk, for issue 15846 |
| 17:01.57 | dlynes | leifmadsen: cool...so when that happens, 1.4 will be an unmaintained branch? |
| 17:02.03 | leifmadsen | dlynes: not at all |
| 17:02.26 | *** join/#asterisk raden_work (n=jon@69-179-99-17.stat.centurytel.net) |
| 17:02.42 | cusco | hi [TK]D-Fender, sorry I was away back then... yea I just figured out I cannot just place the System(chmod blah) after the Monitor, the call hasn't been hung up yet |
| 17:03.17 | dlynes | leifmadsen: ah....how many branches is digium planning to maintain, simultaneously? |
| 17:03.20 | leifmadsen | dlynes: please see this thread to understand: http://lists.digium.com/pipermail/asterisk-dev/2009-October/040082.html |
| 17:03.28 | cusco | core show application monitor doesn't stat exacly where can I place a shell command unless I use other mixer than sox ?! |
| 17:03.48 | dlynes | Qwell: ok...probably won't happen today...I'll need some time to install it on a virtualbox or something so that it doesn't affect my production boxes |
| 17:04.05 | *** join/#asterisk TiToyz (n=TiToyz@aut75-5-82-239-181-57.fbx.proxad.net) |
| 17:04.34 | dlynes | Qwell: if it doesn't happen this week, i'll get it done Monday or Tuesday of next week for sure |
| 17:05.10 | Qwell | I'm pretty sure it isn't reading your asterisk.conf at all |
| 17:05.34 | *** join/#asterisk Tim_Toady (n=moi@adsl148-100.ath.forthnet.gr) |
| 17:05.47 | *** join/#asterisk CGMChris (n=chris@74.143.228.142) |
| 17:05.55 | dlynes | Qwell: that's what I think, too |
| 17:06.04 | Qwell | which would be a permissions issue... |
| 17:06.53 | dlynes | Qwell: if it's a permissions issue (it doesn't make any sense, because it was owned by asterisk:asterisk and chmod 644), it should spit out an error message to that effect |
| 17:07.18 | *** join/#asterisk saxa (n=sasa@host242-95-static.223-217-b.business.telecomitalia.it) |
| 17:07.28 | Qwell | and the /etc/asterisk/ dir permissions? |
| 17:07.44 | CGMChris | At the risk of sounding like a complete noob, I have to ask: Does anyone know of a beep-detect function that could be used in conjunction with AMD() on outbound calls to wait for a beep before playing a message to an answering machine? WaitForSilence(2000) seems to be the most commonly used method, but in my testing it just doesnt work well enough. Many people leave 3-5 seconds of silence at the end of their voicemail greetings. Thoughts? |
| 17:07.58 | dlynes | Qwell: if I remember correctly, it was spitting out an error message about permissions to do with /var/run/asterisk/, which didn't make any sense either, because it was asterisk:asterisk and 0755 |
| 17:08.03 | dlynes | Qwell: same thing |
| 17:08.09 | Qwell | same? as in 644? |
| 17:08.46 | dlynes | Qwell: /var/run/asterisk, /etc/asterisk /var/spool/asterisk /var/lib/asterisk /usr/lib/asterisk all had permissions for asterisk:asterisk, directories had 0755, and files had 0644 |
| 17:09.09 | Qwell | where is asterisk.conf? |
| 17:09.11 | dlynes | Qwell: i didn't change permissions on /usr/sbin/asterisk /usr/sbin/safe_asterisk or /usr/sbin/rasterisk |
| 17:09.15 | dlynes | Qwell: /etc/asterisk |
| 17:09.32 | jeff | trying to use WaitForSilence() with an outgoing call to a Google Voice recipient, using Monitor() seems to show that google voice is supervising the call when it first starts ringing. |
| 17:09.36 | Qwell | /etc/asterisk/asterisk.conf? |
| 17:09.41 | dlynes | Qwell: correct |
| 17:09.58 | jeff | which is rather frustrating. anyone else see similar with outgoing calls to google voice? |
| 17:10.00 | dlynes | Qwell: and everything was compiled from source |
| 17:10.15 | jeff | i did a bit of (heh) googleing, but haven't found much yet. |
| 17:10.33 | dlynes | Qwell: what I will do when I redo it, is give you an ls -alR dump and an asterisk startup dump |
| 17:10.53 | *** join/#asterisk DelphiWorld (n=Miranda@196.20.95.121) |
| 17:10.53 | saxa | hello, I have a question, and it is, why do I get this kind of error by e-mail ? See http://pastebin.ca/1663628 , It seems that it wants to send the message to a wrong mailbox. My /etc/asterisk/voicemail.conf has the following entry: 101 => 1234,Sasa Ostrouska,sasa@brastrak.com.br,attach=yes |
| 17:11.19 | saxa | It seems that it doesnt parse well the voicemail.conf line |
| 17:11.29 | DelphiWorld | [TK]D-Fender: if i get no audio in a SIP call, what i need to do? |
| 17:12.09 | dlynes | saxa: I would think you didn't write it well....did you mean to have two email addresses: "Sasa Ostrouska" and "sasa@brastrak.com.br"? |
| 17:12.53 | saxa | dlynes: ok, but why it tries to send it to attach=yes ? |
| 17:14.02 | dlynes | saxa: oops...sorry...it'll take a pager address of 'sasa@brastrak.com.br' and an email address of 'attach=yes' |
| 17:14.12 | saxa | yup |
| 17:14.14 | dlynes | saxa: forgot the second field was the name of the voicemail entry |
| 17:14.22 | saxa | yes |
| 17:14.36 | CGMChris | DelphiWorld: Firewall? Check "sip set debug" |
| 17:14.49 | dlynes | saxa: if you want it to work, you need 101 => 1234,Sasa Ostrouska,,sasa@brastrack.com.br,attach=yes |
| 17:14.54 | DelphiWorld | CGMChris: ok |
| 17:14.56 | saxa | so this means that , attach=yes should be an mail address of form sasa@machine ? |
| 17:14.58 | dlynes | saxa: notice the second comma? |
| 17:15.05 | saxa | yup |
| 17:15.07 | saxa | got it |
| 17:15.10 | saxa | thx |
| 17:15.17 | saxa | let me try it ou |
| 17:15.19 | saxa | out |
| 17:15.20 | dlynes | saxa: asterisk is very picky about formatting |
| 17:15.22 | *** join/#asterisk ehsjoar (n=ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 17:15.33 | dlynes | saxa: even extra spaces where they don't belong will make something not work |
| 17:16.10 | saxa | dlynes: i can understand that, no problem, but the examples i looked at where made as the mine is, so therefore i reproduced sombodys else error :) |
| 17:16.13 | dlynes | saxa: attach=yes could actually be a valid email address on some systems, if it's a local user |
| 17:16.30 | *** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun) |
| 17:16.30 | dlynes | saxa: local users don't need domains |
| 17:16.44 | saxa | dlynes: but then attach=yes is doesnt needed anymore ? |
| 17:16.46 | dlynes | saxa: however, that being said, i've never heard of an email address with an '=' in the name, either |
| 17:16.56 | saxa | :) |
| 17:17.02 | dlynes | saxa: attach=yes is only needed if you want to send the voicemail as an attachment |
| 17:17.10 | saxa | thats what i want |
| 17:17.23 | saxa | but in any case its in the last field ? |
| 17:17.25 | DelphiWorld | CGMChris: Really destroying SIP dialog '35644f6011f7ded26e704da52157835a@192.168.1.3' Meth |
| 17:17.25 | DelphiWorld | od: OPTIONS |
| 17:17.26 | dlynes | saxa: and it'll be attached to the second email address...the first email address (that you're skipping) does not get a voicemail attachment |
| 17:17.29 | dlynes | saxa: correct |
| 17:17.45 | saxa | dlynes: thx, let me put that comma in the file |
| 17:17.52 | *** join/#asterisk TheDavidFactor (n=chatzill@fw1.safedataisp.net) |
| 17:17.56 | *** join/#asterisk casix (n=casix@xenpbxedifici.adamvozip.es) |
| 17:17.58 | casix | hello |
| 17:18.05 | dlynes | saxa: or if you don't want any emails: 101 => 1234,Sasa Ostrouska,,,attach=no |
| 17:18.06 | TSM2 | im trying to get my polys to accept 0 as a number it can dial the operator with, at the moment it just gives a new dialtone and if you then press 0 again it sends 00 |
| 17:18.19 | CGMChris | DelphiWorld: The call connects but no audio in or out? |
| 17:19.09 | DelphiWorld | CGMChris: no, now is droped |
| 17:19.15 | DelphiWorld | CGMChris: but other call is no audio |
| 17:19.50 | DelphiWorld | how to show active channels? |
| 17:19.57 | casix | I have a problems with the voicemail. My users are like this hel-333 and his voicemail is 333 in context hel. How can I make thant the asterisk tells the user hel-333 that have voicemails? how can I relation the extension hel-333 with his voicemail [hel] 333 => ... |
| 17:19.58 | casix | ?? |
| 17:20.28 | saxa | dlynes: thx |
| 17:20.29 | CGMChris | DelphiWorld: Your SIP provider may require you to use "insecure=port,invite" in your sip.conf settings. Set "qualify=yes" to see if you're getting registered properly. "sip show peers" will show information about your peers (devices + providers) |
| 17:20.48 | ManxPower-work | casix: ${EXTEN:4} would strip off the first 4 chars |
| 17:21.18 | DelphiWorld | CGMChris: that don't require reg, just trunking using Ip Auth |
| 17:22.02 | *** join/#asterisk afink (n=afink@204.26.87.226) |
| 17:22.11 | ManxPower-work | ~siptrunk |
| 17:22.12 | infobot | i heard siptrunk is To set a SIP peer/friend/user as a trunk add either trunk=yes or wombat=yes (they both do the same thing) in the peer/friend/user definition in sip.conf |
| 17:22.15 | casix | ManxPower-work: I know how to send the call to the voicemail but after the phone of the user don't know the user have voicemails to warm the user |
| 17:22.26 | CGMChris | DelphiWorld: did you properly specify your external ip in sip.conf? http://www.voip-info.org/tiki-index.php?page=Asterisk+SIP+externip |
| 17:22.37 | ManxPower-work | casix: Is English not your first language? |
| 17:22.37 | *** join/#asterisk ingcomrbr (n=ingcomrb@189.162.161.66) |
| 17:22.58 | casix | ManxPower-work: is not |
| 17:23.08 | DelphiWorld | CGMChris: no, i will do it |
| 17:23.12 | ingcomrbr | hi there. |
| 17:23.14 | ingcomrbr | I need re-read configs since an extension.. |
| 17:23.25 | ingcomrbr | where Do I need set it? |
| 17:23.33 | ingcomrbr | <PROTECTED> |
| 17:23.42 | ingcomrbr | I need the same but as code. |
| 17:23.50 | ingcomrbr | to press since an extension |
| 17:23.57 | ingcomrbr | by example, whether I press *666 I need that all configs being re-read |
| 17:23.59 | ManxPower-work | casix: you would set the mailbox= option in sip.conf to specify which mailbox to monitor for that peer |
| 17:24.08 | *** join/#asterisk wollie (n=quassel@190.106.64.22) |
| 17:24.11 | casix | thx :) |
| 17:24.15 | ingcomrbr | Does Somebody know do that? |
| 17:24.22 | ManxPower-work | ingcomrbr: you might have better luck asking on #freepbx or #trixbox. |
| 17:24.46 | ingcomrbr | ok |
| 17:24.47 | ingcomrbr | thanks |
| 17:25.06 | wollie | hi all, I have a question I can't find the answer for: how can I change the tone that sounds into the phone when it's ringing? So not the ringtone of the phone, but the zoom that comes every few seconds? |
| 17:25.15 | afink | Is there a way to have agents login to call groups? This would be different than a queue b/c the person logged into the group wouldn't be on in a q just when a phone call comes in, it would be routed to the people that are logged in. |
| 17:25.23 | leifmadsen | wollie: indications.conf |
| 17:25.30 | DelphiWorld | CGMChris: can you give me the ext sip ip string? |
| 17:25.55 | *** join/#asterisk boch (n=fran@200.61.191.9) |
| 17:26.02 | ManxPower-work | leifmadsen: that's only for inband indications, isn't it? |
| 17:26.12 | leifmadsen | ManxPower-work: isn't that what he was asking about? |
| 17:26.30 | ManxPower-work | leifmadsen: he did not say one way or the other. |
| 17:26.32 | boch | can i playback on SIP channels without answering the call? im not using skip neither noanswer option |
| 17:26.46 | wollie | @leifmadsen: thanks, I'll have a look at that. I couldn't find it,perhaps I was using the wrong questions in Google. Must be because English is not my native language ;) |
| 17:26.46 | DelphiWorld | cgmexternip=196.20.95.121 |
| 17:26.48 | DelphiWorld | right? |
| 17:27.01 | ManxPower-work | boch: use the noanswer option |
| 17:27.03 | DelphiWorld | externip=196.20.95.121 |
| 17:28.45 | *** join/#asterisk Tim_Toady (n=moi@adsl249-13.kln.forthnet.gr) |
| 17:30.25 | boch | ManxPower-work, thanks, and can i send a 5xx reply instead a 603 declined ? |
| 17:30.42 | *** part/#asterisk DelphiWorld (n=Miranda@196.20.95.121) |
| 17:31.28 | ManxPower-work | boch: Hangup() takes an optional HANGUPCAUSE (as you can see by "core show application hangup"). There is an RFC that shows the mappings between Q.931 cause codes and SIP codes. |
| 17:31.46 | boch | ManxPower-work, thanks again |
| 17:32.38 | [TK]D-Fender | [12:03]<cusco>core show application monitor doesn't stat exacly where can I place a shell command unless I use other mixer than sox ?! <- CORRECT |
| 17:33.37 | TSM2 | boch: http://www.voip-info.org/wiki/index.php?page=Asterisk+variable+hangupcause |
| 17:33.57 | *** join/#asterisk moy (n=moy@74.12.130.190) |
| 17:34.11 | TSM2 | ive come across this as i wanted to give a diffrent message based on hangupcause code |
| 17:34.11 | boch | TSM2, thanks |
| 17:35.05 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 17:35.05 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 17:36.20 | *** join/#asterisk circut (n=erik@173-15-94-125-Illinois.hfc.comcastbusiness.net) |
| 17:36.41 | circut | hey all, can anyone offer me some advice on debugging POTS line quality issues? |
| 17:36.44 | superbeef | does tos tagging work properly in asterisk 1.4.current or does it need to be ran as root to tag the packet? |
| 17:37.16 | ManxPower-work | superbeef: you need libcap-devel which allows non-root processes to set their ToS bits. |
| 17:37.31 | ManxPower-work | see "./configure --help" |
| 17:37.41 | superbeef | ManxPower-work: will i have to recompile asterisk? |
| 17:37.49 | hardwire | is everything normal again? |
| 17:39.36 | ManxPower-work | superbeef: yes |
| 17:39.44 | *** join/#asterisk errotan (n=errotan@81.0.115.119) |
| 17:39.50 | superbeef | ManxPower-work: great info thanks for your help |
| 17:40.17 | Katty | jeebus i feel like el crapola. |
| 17:41.44 | hardwire | cat crap |
| 17:46.42 | bmoraca | anyone have experience running Asterisk on an Intel Atom processor? |
| 17:47.06 | cusco | [TK]D-Fender: so making a script that besides calling sox will also chmod... ok thanks |
| 17:48.34 | hardwire | bmoraca: you'd think it wouldn't be that different |
| 17:49.01 | hardwire | I run it on Via C3 and Geode GX |
| 17:49.10 | bmoraca | hardwire: i'm more concerned with performance, based on the fact that the Atom is an in-order processor and has pretty crappy FP performance |
| 17:49.16 | hardwire | It's slow.. but usable :) An Atom should be like a slow P4. |
| 17:50.18 | *** join/#asterisk ariel_ (n=chatzill@63.214.236.169) |
| 17:50.26 | hardwire | bmoraca: so don't use anything non-fixed point |
| 17:50.51 | hardwire | it should be able to handle 6-7 g729 streams no problem if a Via c3 1ghz can. |
| 17:51.41 | bmoraca | that's what i'm hoping...the office is tiny, 6 phones and 4 trunks only. i'm probably going to use a 4-port digium echo cancelled call. |
| 17:52.02 | bmoraca | card |
| 17:52.05 | bmoraca | can't type |
| 17:52.20 | hardwire | pots card? |
| 17:52.29 | bmoraca | yes |
| 17:52.33 | hardwire | gotcha |
| 17:52.47 | bmoraca | nah, i thought 4 T1s should be fine :P |
| 17:52.48 | hardwire | echocan should be fine as well. The cost is too high for echocan on board |
| 17:52.50 | hardwire | imho |
| 17:53.29 | hardwire | go buy an acer laptop with an atom processor in it as a "work expense" and do some research :) |
| 17:53.32 | hardwire | send me one too |
| 17:53.36 | bmoraca | my experience with AT&T and hte wiring out here is that if you're not doing echo cancellation onboard, you'll hear yourself 4 times for every one word you say |
| 17:53.44 | bmoraca | lol |
| 17:54.12 | hardwire | doh |
| 17:54.27 | bmoraca | yeah...it's pretty bad |
| 17:54.33 | hardwire | is watching surrogates. bbl |
| 17:56.59 | *** join/#asterisk donnib (n=donnib@0x555281d0.adsl.cybercity.dk) |
| 17:57.10 | donnib | still with registration problems |
| 17:57.13 | angryuser | anyone from sangoma here ? (or a good sangomas BRI guru ) ? |
| 17:58.23 | moy | angryuser: with your nick, anyone from Sangoma would be scared to talk to you |
| 17:59.08 | *** join/#asterisk TimToady_ (n=moi@adsl92-40.kln.forthnet.gr) |
| 17:59.24 | *** join/#asterisk aidinb (n=Aidin@24-182-32-138.static.lnbh.ca.charter.com) |
| 17:59.29 | Qwell | moy: :p |
| 17:59.40 | angryuser | i see there is no one |
| 17:59.47 | Qwell | ~ask |
| 17:59.48 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 18:00.02 | [TK]D-Fender | angAnd the fact you're asking this to support it with CallWeaver.... well gimme a sec to gather the pitchforks & villagers :p |
| 18:00.11 | [TK]D-Fender | angryuser: ^ |
| 18:00.14 | aidinb | oh dear, not the villagers |
| 18:01.04 | angryuser | crap i have a realy strange one, when calling in with bri i got sangoma_mgd[2125]: CALL INCOMING: Enqueue Error Sent SIGBOOST_EVENT_CALL_START_NACK [w1g1] And the call is not redirected to the callweaver .... |
| 18:01.36 | Qwell | angryuser: #callweaver |
| 18:01.49 | [TK]D-Fender | angryuser: Call up Sangoma support |
| 18:02.14 | angryuser | Qwell, [TK]D-Fender yea, i think i am doing that |
| 18:02.37 | angryuser | will be* |
| 18:02.39 | angryuser | xD |
| 18:03.41 | donnib | [TK]D-Fender: i did what you told me earlier today to remove the 0.0.0.0/0.0.0.0 under permit/deny and i also left blank the port. do you have other ideas to my earlier problem with the registration ? |
| 18:03.48 | hardwire | the callweaver |
| 18:03.52 | hardwire | excellent |
| 18:03.55 | hardwire | rubs palms together |
| 18:04.14 | donnib | unfortunately it did not work. |
| 18:04.36 | ManxPower-work | donnib: REMOVE the option, don't just blank it out. |
| 18:04.37 | [TK]D-Fender | donnib: I said REMOVE THE LINES. Not "leave blank" |
| 18:05.21 | donnib | well i guess that i can´t do that :( since i am running free... damn it |
| 18:09.37 | *** join/#asterisk rpm (n=Russell_@S0106000c29898b7e.cg.shawcable.net) |
| 18:10.39 | rpm | Does IAX realtime "accountcodes" not work? I cannot seem to find out why the ${ACCOUNTCODES} variable is not populated when I initiate a call from an account which has an account code set. |
| 18:10.50 | rpm | I've used it with SIP before, but never IAX2 |
| 18:11.42 | [TK]D-Fender | rpm: no "S" |
| 18:12.19 | rpm | I mean "accountcode" sorry and ${ACCOUNTCODE} |
| 18:12.46 | *** join/#asterisk Kobaz (n=kobaz@its.kobaz.net) |
| 18:13.43 | Kobaz | is there a way to ad-hocly turn on mwi on a sip phone, without needing to move to asterisk 1.6.2 for the mwi application |
| 18:14.09 | ManxPower-work | Kobaz: touch a msgxxx.txt file in the user's mailbox |
| 18:14.38 | Kobaz | yeah |
| 18:14.40 | Kobaz | that's not reliable |
| 18:14.53 | *** part/#asterisk Scunizi (n=mark@69.199.151.114) |
| 18:15.31 | ManxPower-work | Kobaz: It's exactly how Asterisk does it |
| 18:15.47 | Kobaz | somehow whatever asterisk does anyways works |
| 18:15.58 | Kobaz | when I do it, 90% of the time, the light won't come on |
| 18:16.12 | Kobaz | i have ti restart asterisk a half dozen times, and then it will come on |
| 18:16.37 | Kobaz | reloading voicemail doesn't do it either |
| 18:17.11 | *** join/#asterisk slinksh0t_ (n=slinksh0@c-76-108-31-194.hsd1.fl.comcast.net) |
| 18:17.32 | Kobaz | i need to indicate a call is in queue, on a polycom phone |
| 18:17.46 | Kobaz | i've tried the mwi approach before, but it just isn't reliable |
| 18:17.47 | [TK]D-Fender | Kobaz: FFS use presence or the MicroBrowser |
| 18:18.05 | Kobaz | yeah, i was thinking about the microbrowser |
| 18:18.12 | [TK]D-Fender | Kobaz: What are you doing messing with MWI as an indicator? That's worst possible option. |
| 18:18.32 | Kobaz | [TK]D-Fender: flashing red light is a perfect attention grabber |
| 18:18.42 | Kobaz | and none of these phones have voicemail |
| 18:18.43 | [TK]D-Fender | Kobaz: What model? |
| 18:18.48 | Kobaz | 330 |
| 18:18.57 | [TK]D-Fender | Kobaz: MicroBroser is it. |
| 18:19.12 | [TK]D-Fender | Kobaz: 3XX don't support presence |
| 18:19.26 | Kobaz | ah |
| 18:19.39 | *** join/#asterisk luckyaba (n=lucky@ip72-194-215-55.sb.sd.cox.net) |
| 18:20.22 | *** join/#asterisk Iamnacho (i=Iamnacho@ip98-186-180-143.ks.ks.cox.net) |
| 18:21.39 | Kobaz | mmm, is there another way to turn on the blinking light other than mwi? |
| 18:22.45 | Kobaz | i can do the microbrowser thing, but the mwi light is just so perfect |
| 18:25.52 | Kobaz | ah crap |
| 18:25.54 | Kobaz | this won't work |
| 18:26.09 | carrar | Just put a dozen strobe lights on the users desk |
| 18:26.15 | Kobaz | Microbrowser that allows browsing of simple XHTML web pages on the phone's LCD screen. In addition the microbrowser can also be used to display information on the phone's idle screen |
| 18:26.25 | rpm | Is anyone using IAX with realtime and accountcodes? It's not reading the 'accountcode' field in my MySQL table... Neither the ${ACCOUNTCODE} variable or ${CDR(accountcode)} |
| 18:26.26 | carrar | flashing at random intervuls |
| 18:26.33 | Kobaz | The provisioning variables are mb.idleDisplay.home="(url)" and mb.idleDisplay.refresh="(seconds"). |
| 18:26.42 | Kobaz | i need it to display at all times, not just idle |
| 18:27.00 | Kobaz | carrar: just might have to do that |
| 18:27.25 | Kobaz | so i'm back to using mwi |
| 18:27.43 | Kobaz | i was thinking of doing a thing where i make a fake call to line 2, as notification |
| 18:27.44 | *** join/#asterisk centrex (n=pornstar@173-24-237-231.client.mchsi.com) |
| 18:27.46 | *** join/#asterisk WinZ (n=winz@82.146.61.218) |
| 18:28.10 | Kobaz | but the problem is, the display changes and locks things up for 10 seconds when the call comes in, if you don't answer it |
| 18:28.54 | [TK]D-Fender | rpm: .... PASTEBIN <- |
| 18:29.11 | WinZ | guys, if I can call some numbers, but with other numbers I get "All circuits are busy" message through only one trunk -- the problem is on the provider's side? |
| 18:29.42 | WinZ | I can call the US, but can't call some mobile numbers in Europe |
| 18:29.44 | centrex | Regarding queues, I have sip agents that were receiving calls when already on the line in a queue. The call-limit was set to 50 for the sip peer so it was showing them as not in use. However, when I set the call-limit to 1, they can no longer transfer calls. I'm looking for a solution where I can only have one call go to a queue member, but also allow them to transfer out. If I set the call limit to 2 I believe it would still all |
| 18:29.44 | centrex | ow 2 calls per queue member, correct? |
| 18:29.51 | [TK]D-Fender | WinZ: Show us something to LOOK AT |
| 18:30.07 | [TK]D-Fender | ~pb |
| 18:30.08 | infobot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , or apt-get install pastebinit |
| 18:30.08 | *** join/#asterisk raden_work (n=jon@69-179-99-17.stat.centurytel.net) |
| 18:30.09 | [TK]D-Fender | ^^^^^^^6 |
| 18:30.11 | [TK]D-Fender | ^^^^^^^ |
| 18:30.11 | rpm | [TK]D-Fender: http://pastebin.com/m3143770b should work |
| 18:30.53 | centrex | Is there a way to set it so queue members can only receive one call from the queue, but also allow them to transfer calls and dial out while on another call as well? |
| 18:31.11 | [TK]D-Fender | rpm: variable was deprecated for the function. Set the function |
| 18:35.28 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 18:35.37 | rpm | [TK]D-Fender: ${CDR(accountcode)} still doesn't work - I shouldn't need to initially set the ${CDR(accountcode)=....} as it is set in the 'iax_buddies' table. |
| 18:36.05 | WinZ | [TK]D-Fender, http://pastebin.com/d52d0b280 - regarding the "all circuits are busy" message |
| 18:36.45 | *** part/#asterisk lmadsen[testnet] (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 18:38.30 | [TK]D-Fender | rpm: I don't see that.. |
| 18:39.31 | [TK]D-Fender | WinZ: -- Got SIP response 480 "Temporarily Unavailable - Cannot Complete Call" back from 66.33.157.12 <-- from your provider |
| 18:39.36 | WinZ | yes yes |
| 18:39.41 | WinZ | I'm lookin at it too.. |
| 18:40.23 | WinZ | thanks, [TK]D-Fender, at least I know where the problem is |
| 18:41.24 | *** join/#asterisk hatoff (n=hatline@unaffiliated/hatoff) |
| 18:41.55 | *** join/#asterisk s34n (n=chatzill@ip-208-76-93-125.mvdsl.com) |
| 18:42.34 | hatoff | I have made my first steps into Asterisk. Is there a simple way to configure it to autoanswer so I can check if it is correctly installed? |
| 18:44.10 | [TK]D-Fender | hatoff: Make a simple exten to do something like Answer, Playback(), and then hangup |
| 18:44.20 | hatoff | I think i'm gettin faster answers on forum lol :) |
| 18:44.27 | hatoff | ah |
| 18:44.31 | hatoff | thanks |
| 18:44.49 | hatoff | but how do I make the exten, in asterisk.conf file in /etc ? |
| 18:44.51 | hatoff | will that work? |
| 18:44.59 | hatoff | or can I do it from the CLI ? |
| 18:46.26 | hatoff | extensions.conf I think |
| 18:48.48 | centrex | Is there any way to have a queue report a device as "in use" when it is on a call without breaking attended transfers with asterisk 1.4? |
| 18:49.50 | centrex | I have a queue where it reports the device is not in use when on a call, unless the device has a call limit set. I only want one call per agent in the queue. But if I set call-limit 1 it breaks transfers |
| 18:50.44 | centrex | If set the sip call-limit to 2, it allows transfers, but then they still get rang from the queue when on a call |
| 18:51.11 | *** join/#asterisk Tim_Toady (n=moi@adsl92-40.kln.forthnet.gr) |
| 18:52.51 | hatoff | centrex, can you tell me where I can put the exten ? |
| 18:53.13 | hatoff | i'm really noob in using asterisk |
| 18:53.22 | centrex | extensions.conf |
| 18:53.33 | hatoff | i just want to see that I've installed it correctly so I want to add an exten to Answer Playback aand then hangup |
| 18:53.37 | hatoff | i've placed it there |
| 18:53.44 | hatoff | how do I reload? |
| 18:53.49 | hatoff | killall -HUP asterisk will be fine!? |
| 18:53.54 | centrex | asterisk -r |
| 18:53.57 | centrex | reload |
| 18:54.07 | *** join/#asterisk t (i=tom@freenode/staff/tomaw) |
| 18:54.11 | hatoff | okie dokie |
| 18:54.22 | hatoff | i'm calling the line but it will not answer |
| 18:55.07 | hatoff | which could be the problem? |
| 18:55.48 | *** part/#asterisk hatoff (n=hatline@unaffiliated/hatoff) |
| 18:56.24 | Katty | had a nap |
| 18:57.25 | s34n | has anyone here used the jack() command? |
| 18:57.58 | *** join/#asterisk donnib (n=donnib@0x555281d0.adsl.cybercity.dk) |
| 19:03.04 | rpm | this makes no sense, CDR(accountcode) is not auto-populated if it is set on the IAX peer/user's account.. Isn't that the point of using an accountcode associated to an IAX peer/user definition? |
| 19:03.06 | hardwire | it does jack for me. |
| 19:05.08 | [TK]D-Fender | hatThat you did not put the exten in a context being accessed by the incoming call. |
| 19:05.15 | ManxPower-work | rpm: put a dummy friend at the end of the iax.conf that points to an invalid context. I bet your incoming call is not matching the peer you think it's matching |
| 19:05.33 | ManxPower-work | and chan_iax likes to match the last entry when it can't find any other matches |
| 19:05.36 | [TK]D-Fender | rpm: I still don't have a comprehensive pastebin... |
| 19:06.05 | rpm | ManxPower-work: I'm using realtime for all of my 'friend' IAX accounts. |
| 19:06.15 | ManxPower-work | rpm: It sucks to be you. |
| 19:06.27 | *** join/#asterisk telnettech (n=telnette@office.callcopy.com) |
| 19:06.58 | Kobaz | chanspy with w doesn't work in 1.6.0.15 |
| 19:08.08 | leifmadsen | Kobaz: that should be fixed in the latest RCs that are not yet put out |
| 19:08.20 | leifmadsen | they will go out in the next 2 hours |
| 19:08.28 | centrex | On my queues they all report that the device isn't in use, even though the caller is on a call. Is there a way to fix this? |
| 19:08.30 | Kobaz | ah okay |
| 19:08.37 | leifmadsen | the RCs put out on Friday should also have the fix, but they cause a crash which is fixed in the new RCs I'm putting out today |
| 19:08.45 | leifmadsen | Kobaz: so you should test with the 1.6.0 branch from SVN |
| 19:08.49 | Kobaz | any time i try something new in asterisk it either doesn't work or crashes :( |
| 19:08.49 | centrex | on asterisk 1.4. |
| 19:08.52 | ManxPower-work | centrex: as talked about in the UPGRADE*.txt files you need to set a call limit for devstate to work |
| 19:09.03 | centrex | ManxPower-work, I did that, but when I set the call-limit to 1 it doesn't allow transfers. |
| 19:09.10 | leifmadsen | centrex: set it to 2 |
| 19:09.13 | ManxPower-work | centrex: so set it to 99 |
| 19:09.25 | leifmadsen | centrex: you can't transfer when you're only allowing 1 call since a transfer is 2 calls |
| 19:09.28 | ManxPower-work | you are not trying to limit calls, you are just trying to make Asterisk keep track of the |
| 19:09.36 | centrex | Oh I see |
| 19:09.45 | centrex | So setting it to two wouldn't allow a 2nd incoming call in? |
| 19:09.56 | leifmadsen | it would.. because it's set to 2... |
| 19:10.02 | leifmadsen | a 3rd call would not be allowed |
| 19:10.09 | leifmadsen | because 3 is bigger than 2 |
| 19:10.13 | centrex | Right, i want to be able to only have one call come in the queue. I dont want them to be on a call and then hve it ring |
| 19:10.25 | leifmadsen | centrex: that is a different functioanlity |
| 19:10.27 | centrex | But I still want them to be able to transfer calls out |
| 19:10.39 | centrex | ringinuse is set to no |
| 19:11.00 | leifmadsen | centrex: that's not what call-limit is for, device state is what controls when a caller is InUse or not |
| 19:11.14 | leifmadsen | setting to a higher call-limit will not break that if configured correctly |
| 19:11.40 | leifmadsen | higher call-limit doesn't mean multiple calls from the queue if you setup the queue to not send calls to agents InUse |
| 19:14.33 | centrex | Okay, i think the problem is I was confused as to fix the problem. It's still not working (not showing the device state in use) even though limitonpeers is set to yes. |
| 19:14.54 | centrex | But the queuemembers are local channels, not sip |
| 19:14.57 | centrex | Could that be the problem? |
| 19:17.46 | centrex | or what else could be stopping it from updating the device state? |
| 19:18.16 | *** join/#asterisk TSM (n=the_soft@87-194-32-212.bethere.co.uk) |
| 19:21.19 | leifmadsen | centrex: if they are Local channels, you need to setup a hint for the local channel which monitors the appropriate SIP device |
| 19:21.36 | centrex | okay thanks |
| 19:25.50 | centrex | leifmadsen, That's for 1.6 only though, correct? |
| 19:26.32 | centrex | from what I'm reading 1.4 doesn't support hints for local channels |
| 19:28.33 | leifmadsen | centrex: sorry, wrong location, it's done in queues.conf |
| 19:28.40 | leifmadsen | from queues.conf.sample: ;member => Local/1000@default,0,John Smith,SIP/1000 |
| 19:30.36 | *** part/#asterisk rpm (n=Russell_@S0106000c29898b7e.cg.shawcable.net) |
| 19:31.49 | *** join/#asterisk Malkor (n=marco@hlle-d9ba009b.pool.mediaWays.net) |
| 19:35.57 | *** join/#asterisk nix8n82 (n=nate@63.162.27.14) |
| 19:38.30 | *** join/#asterisk `paul (n=kutimoy@123-242-230-55.sunnyvisiondatacenter.com) |
| 19:39.30 | `paul | re: vicidial, if i downloaded the latest 1.4 version will the patch of the vicidial on its guide (for 1.4.21.2) still be applicable? |
| 19:48.37 | *** join/#asterisk Winkie (n=urmom@ur.fa.gs) [NETSPLIT VICTIM] |
| 19:49.19 | *** join/#asterisk Winkie (n=urmom@ur.fa.gs) |
| 19:50.09 | *** join/#asterisk fixxxermet (n=lopan@vps.fixertec.net) |
| 19:53.22 | fixxxermet | I am getting a "unable to open psuedo device" error using asterisk and dahdi from svn. http://pastebin.com/d4918b338 is my sip debug output. |
| 19:54.17 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 19:54.27 | *** join/#asterisk voxter (n=voxter@76.77.91.250) |
| 19:54.37 | *** join/#asterisk vgster (n=vgster@94-194-190-189.zone8.bethere.co.uk) |
| 19:55.02 | *** join/#asterisk bluOxigen (n=asad@static-host119-73-66-180.link.net.pk) |
| 19:55.37 | *** join/#asterisk ardnat (i=18e152f7@gateway/web/freenode/x-lpmluxjnqyitzzes) |
| 19:55.40 | ardnat | hey |
| 19:55.52 | ardnat | could someone help me with fowarding in asterisk |
| 19:56.09 | ardnat | im trying to do somthing similar to trapcall |
| 19:56.17 | ardnat | im fowarding an incoming call to a 1800 |
| 19:56.26 | ardnat | so that i can get its real caller id |
| 19:56.38 | ardnat | can somthing like this be done? |
| 19:58.28 | *** join/#asterisk s34n (n=chatzill@ip-208-76-93-125.mvdsl.com) |
| 19:59.03 | s34n | is there a simple way to plug asterisk audio into gstreamer? |
| 20:01.40 | Katty | aww |
| 20:01.49 | Katty | i do believe one of our vendors was being all flirty |
| 20:03.02 | raden_work | Katty, does that make you happy ? |
| 20:03.10 | Katty | it's cute. |
| 20:05.55 | Katty | that nap seemed to help. |
| 20:05.58 | *** join/#asterisk lesouvage (n=lesouvag@82.73.69.76) |
| 20:06.03 | Katty | tho i'm still feeling a bit sickly, and cranky |
| 20:06.24 | thehar | Katty: drink your magic juice you told me about |
| 20:06.45 | Katty | ugah. no more OJ for awhile |
| 20:07.16 | Katty | definately no more vodka for awhile >.< |
| 20:07.22 | thehar | sad |
| 20:07.36 | eppigy | man I felt that way friday night/saturday morning |
| 20:09.59 | *** join/#asterisk keyp (n=keyp@66.184.128.98) |
| 20:10.52 | *** join/#asterisk Skeeter- (i=Skeeter-@190-141.cgocable.ca) |
| 20:12.11 | centrex | leifmadsen, thanks |
| 20:12.33 | fixxxermet | Fixed my error. |
| 20:13.30 | Katty | eppigy: i hear chinese, soda, and tylenol are like the cure all for that. |
| 20:13.45 | Katty | eppigy: luckily, i didn't go as far as Morning Headache |
| 20:13.54 | Katty | eppigy: possibly having something to do with being up illish until 3 >.< |
| 20:17.20 | *** join/#asterisk uqlev (n=yuriy@91.184.221.31) |
| 20:19.40 | *** join/#asterisk wcselby (n=wcselby@216-110-88-194.static.twtelecom.net) |
| 20:19.44 | wcselby | o/ |
| 20:19.44 | *** part/#asterisk fixxxermet (n=lopan@vps.fixertec.net) |
| 20:20.45 | Katty | hi wcselby |
| 20:21.21 | Katty | bye wcselby |
| 20:21.22 | *** join/#asterisk wcselby (n=wcselby@216-110-88-194.static.twtelecom.net) |
| 20:21.25 | wcselby | bleh |
| 20:21.37 | wcselby | hit alt-x on the last letter I was writing in my sentance |
| 20:21.44 | wcselby | have no clue how I did that |
| 20:22.35 | wcselby | anyways, anyone here using the 'd' option with the Dial command to catch dtmf during ringing? |
| 20:24.01 | wcselby | I can't seem to catch the 1-digit extension |
| 20:24.35 | wcselby | inbound call over dahdi -> hit's the proper extension in extensions.conf, which has a Dial(SIP/2625,30,d) statement |
| 20:25.03 | wcselby | begins dialing the SIP/2625 endpoint, but never catches when I press 0 during the ring |
| 20:25.28 | bmoraca | what type of dahdi trunk? maybe try relaxdtmf? have you verified that dtmf works over it in other contexts? |
| 20:25.50 | wcselby | dtmf works in voicemail and the main IVR |
| 20:26.05 | wcselby | just not catching it during the ring |
| 20:26.07 | Katty | rehi wcselby |
| 20:26.17 | wcselby | o/ Katty :) |
| 20:26.25 | Katty | hmm. no, i don't think we use d here in our dial commands. |
| 20:26.31 | ManxPower-work | wcselby: have you done a "core show application dial" to confirm YOUR version Of Asterisk supports that option |
| 20:26.40 | wcselby | ManxPower-work - yes |
| 20:26.47 | ManxPower-work | wcselby: weird |
| 20:27.09 | wcselby | ManxPower-work - I agree |
| 20:27.26 | Katty | stares at teabag. |
| 20:27.30 | wcselby | it's not a critical thing, just one of the board members of the client I'm working for is upset that he has to wait 30 seconds for someone to pickup |
| 20:27.43 | Katty | oh noes. |
| 20:27.44 | beek | Katty: It's not talking to you, is it? |
| 20:27.44 | wcselby | so I'm investigating options now |
| 20:27.46 | Katty | how horrible. |
| 20:27.55 | Katty | beek: well. yes. i mean...it's calling my name. |
| 20:28.05 | beek | :D |
| 20:28.06 | bmoraca | for someone to pick up and do what? |
| 20:28.19 | Katty | d dials dtmf after a person's picks up |
| 20:28.21 | wcselby | bmoraca - to talk, or for it to rollover to voicemail |
| 20:28.21 | Katty | like an auto attendant |
| 20:28.28 | wcselby | Katty - that's D |
| 20:29.01 | bmoraca | how's the ability to capturing dtmf during dialing going to alleviate that? |
| 20:29.19 | Katty | Allow the calling user to dial a 1 digit extension while waiting for a call to be answered. Exit to that extension if it exists in the current context, or the context defined in the EXITCONTEXT variable, if it exists. |
| 20:29.23 | wcselby | bmoraca - he wants to be able to hit 0 to go to the operator instead of waiting either 30 seconds for voicemail |
| 20:29.46 | bmoraca | so give him a direct did to the operator |
| 20:30.04 | wcselby | bmoraca - there are already several ways to do that |
| 20:30.14 | wcselby | he's specifically asked to be able to do it while a line is ringing |
| 20:30.19 | wcselby | before someone has answered |
| 20:30.26 | wcselby | Katty - http://pastebin.com/m5ff44fa3 |
| 20:30.38 | ManxPower-work | wcselby: you can't accept audio until the line is answered. |
| 20:30.43 | ManxPower-work | put an Answer() for the Dial |
| 20:30.44 | *** join/#asterisk Chodorenko (n=chodoren@ext.one.by) |
| 20:31.00 | ManxPower-work | put an Answer() before the Dial() |
| 20:31.25 | ManxPower-work | Since the telco won't actually SEND the DTMF/AUDIO until the line is answered. |
| 20:31.32 | bmoraca | while any line is ringing? that sounds like a terrible idea |
| 20:32.00 | bmoraca | yeah, that too |
| 20:32.25 | wcselby | ManxPower-work - ugh. let me try that |
| 20:32.39 | wcselby | bmoraca - I never said it was a good idea |
| 20:32.50 | wcselby | bmoraca - I said a member of the board is asking for the capability |
| 20:32.58 | ManxPower-work | wcselby: wouldn't it be easier to just switch this guy to decaf? |
| 20:33.15 | wcselby | ManxPower-work - lol, I'm just a consultant investigating options ;) |
| 20:33.40 | bmoraca | tell him to diaf and stop being a prima donna |
| 20:34.07 | wcselby | haha |
| 20:34.25 | wcselby | i've already told him I didn't think it was possible, so if I end up not coming up with a viable way to use it, it's not a big deal |
| 20:34.33 | bmoraca | either way, you won't have to deal with him again :P |
| 20:36.43 | wcselby | ManxPower-work - the Answer() before the Dial() worked |
| 20:37.08 | wcselby | I remember taking all the Answer() out of my inbound extensions at one point, it broke something relating to queues/agents (but I don't remember what) |
| 20:37.28 | ManxPower-work | wcselby: Generally Answer is a bad idea. Except for in this case. |
| 20:37.29 | bmoraca | it also makes your CDRs somewhat inaccurate |
| 20:37.59 | wcselby | yeah, I'm thinking this board member is going to have to live with waiting 30 seconds) |
| 20:39.34 | *** join/#asterisk trumee (i=rs4@cpc2-cmbg15-0-0-cust1000.5-4.cable.virginmedia.com) |
| 20:39.41 | trumee | anybody familiar with spa3102? |
| 20:40.19 | *** join/#asterisk wierdo (n=jimmy@77.78.3.197) |
| 20:40.28 | *** join/#asterisk andres833 (n=andres83@190.144.75.22) |
| 20:40.39 | trumee | <PROTECTED> |
| 20:45.48 | *** join/#asterisk Zokeber (n=Zokeber@unaffiliated/zokeber) |
| 20:46.09 | *** join/#asterisk brezular (n=brezular@adsl-dyn111.91-127-139.t-com.sk) |
| 20:46.17 | *** join/#asterisk fixxxermet (n=lopan@vps.fixertec.net) |
| 20:46.52 | *** join/#asterisk Iamnach0 (i=Iamnacho@ip98-186-180-143.ks.ks.cox.net) |
| 20:48.42 | *** join/#asterisk Ad-Hoc (n=nimbus@ppp57-248.adsl.forthnet.gr) |
| 20:49.08 | *** part/#asterisk WinZ (n=winz@82.146.61.218) |
| 20:53.54 | Ad-Hoc | hi |
| 20:54.12 | jaytee | lo |
| 20:54.36 | *** join/#asterisk xpot-mobile (n=xpot@mx0.synergyconsultant.net) |
| 20:54.42 | wcselby | think it's time to checkout, later |
| 20:54.46 | wcselby | \o |
| 20:55.20 | Katty | http://www.youtube.com/watch?v=JErVP6xLZwg |
| 20:55.30 | Katty | jaytee: let's dance! |
| 20:56.10 | jaytee | dancing is against my religion :-) |
| 20:56.45 | jaytee | I'm a member of the Church of TwoLeftFootism |
| 20:57.36 | TheDavidFactor | members of that Church who do dance end up on one of those "Funniest Videos" TV shows |
| 20:57.37 | Katty | excuse, excuses. |
| 20:57.40 | Katty | dances with jaytee |
| 20:58.20 | eppigy | ALLO |
| 20:59.47 | *** join/#asterisk ardnat (i=18e152f7@gateway/web/freenode/x-fbgzvbfiwtycprbo) |
| 20:59.51 | ardnat | Hey |
| 21:00.15 | ardnat | Can anyone help me transfer a user in my dial plan to an outside phone number while preserving cid records |
| 21:01.21 | Katty | allo dave. |
| 21:01.38 | Katty | ardnat: by what means ar eyou doing the transfer. |
| 21:01.40 | [TK]D-Fender | âWhy don't Baptists have sex standing up ? Because that might lead to dancing.â - Oscar Wilde |
| 21:01.45 | Katty | ardnat: physically. |
| 21:01.57 | ardnat | well i am not sure |
| 21:02.02 | ardnat | my main intention |
| 21:02.03 | Katty | ^_- |
| 21:02.11 | ardnat | is to get the cid of a caller using *67 |
| 21:02.13 | Katty | you don't know how you're transfering the call |
| 21:02.23 | ardnat | no i mean its not implanted yet |
| 21:02.24 | Katty | well i can't help you if you don't know how you're getting the call out. |
| 21:02.33 | ardnat | no i know ways |
| 21:02.42 | ardnat | but im not sure on how to do so |
| 21:02.49 | ardnat | heard of trapcall? |
| 21:02.52 | Katty | nope |
| 21:02.55 | aidinb | trapcall's cool |
| 21:03.05 | aidinb | but theres a site that shows u how to do it with flowroute |
| 21:03.10 | ardnat | you set your call waiting foward to their 1800 number |
| 21:03.16 | aidinb | a youtube vid rather |
| 21:03.21 | ardnat | it fowards the cell phone caller to that number |
| 21:03.34 | ardnat | since all #'s have to provide 1800's with the real cid |
| 21:03.53 | ardnat | it unmask their cid, and fowards their call back to you with the real cid set |
| 21:04.12 | ardnat | i want to do this in asterisk |
| 21:04.22 | aidinb | http://www.youtube.com/watch?v=q3S0RjrXhw0 |
| 21:04.23 | aidinb | there |
| 21:04.24 | aidinb | thats how |
| 21:04.35 | aidinb | easy peasy |
| 21:04.46 | ardnat | xD |
| 21:04.51 | ardnat | only with flowrout though? |
| 21:05.01 | aidinb | only with enterprise class providers |
| 21:05.04 | aidinb | like flowroute |
| 21:05.07 | aidinb | he explains in the video |
| 21:05.08 | ardnat | dang it |
| 21:05.14 | ardnat | im using rapidvox |
| 21:05.17 | aidinb | they have to send the p-assert tag |
| 21:05.47 | aidinb | *shrug* dunno them, worth a shot |
| 21:05.50 | ardnat | kk ty :) |
| 21:05.53 | aidinb | if not, port ur numbers to flowroute |
| 21:06.00 | ardnat | kevin mendiric < FTW |
| 21:06.06 | aidinb | mitnick |
| 21:06.10 | ardnat | yeah lol |
| 21:06.18 | ardnat | do you have a flow account? |
| 21:06.28 | aidinb | yea |
| 21:06.46 | aidinb | put it this way... if mitnick uses it.... |
| 21:06.48 | aidinb | haha |
| 21:06.51 | ardnat | xD |
| 21:06.58 | ardnat | well whats the min deposit |
| 21:07.07 | aidinb | 35 |
| 21:07.15 | ardnat | !!! |
| 21:07.19 | ardnat | darn it |
| 21:07.20 | aidinb | never expires tho |
| 21:07.28 | ardnat | ah well thats too much |
| 21:07.30 | ardnat | im 15 |
| 21:07.32 | ardnat | lmao |
| 21:07.38 | aidinb | smart kid |
| 21:07.39 | aidinb | wow |
| 21:07.44 | ardnat | ah ty lol |
| 21:08.02 | aidinb | u could try asking nicely |
| 21:08.11 | ardnat | hmm |
| 21:08.20 | ardnat | for like a discounted depoit or somthing? |
| 21:08.24 | aidinb | *shrug* |
| 21:08.24 | ardnat | ah got it! |
| 21:08.28 | aidinb | never know |
| 21:08.29 | aidinb | haha |
| 21:08.30 | ardnat | "for a school project " |
| 21:08.33 | ardnat | heh heh |
| 21:08.58 | ardnat | anyways aid quick question |
| 21:09.07 | ardnat | if you set your caller id and ani in asterisk |
| 21:09.13 | ardnat | can you be traced back |
| 21:09.20 | ardnat | you can with the CDR right? |
| 21:09.26 | ardnat | there a cdr number i believe |
| 21:09.38 | aidinb | not always |
| 21:09.49 | aidinb | i think it depends on the carrier really |
| 21:10.12 | ardnat | how could you figure out if they do or dont |
| 21:10.18 | ardnat | like a anic number or somthing |
| 21:10.25 | ardnat | havent seen any that show cdr |
| 21:10.52 | aidinb | im not really sure on that one.. sorry bro |
| 21:11.44 | ardnat | no probs |
| 21:15.55 | leifmadsen | Kobaz: ping? |
| 21:19.39 | Katty | leifmadsen: i think he Nacked. |
| 21:20.48 | leifmadsen | np |
| 21:25.32 | [TK]D-Fender | checkout time, later all |
| 21:26.07 | *** join/#asterisk voipmonk (n=voipmonk@dsl-67-55-17-41.acanac.net) |
| 21:28.13 | *** join/#asterisk ardnat (i=18e152f7@gateway/web/freenode/x-fbgzvbfiwtycprbo) [NETSPLIT VICTIM] |
| 21:37.05 | *** part/#asterisk ehsjoar (n=ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 21:38.26 | *** join/#asterisk p3nguin_ (i=BuhPjX1J@asterisk-klv5.a2infotech.com) |
| 21:38.37 | ardnat | Can anyone help me with transfers in asterisk? |
| 21:40.55 | *** join/#asterisk jeffgus (n=jeffgus@2002:ad33:b504:0:0:0:0:1) |
| 21:41.11 | *** join/#asterisk Zokeber (n=Zokeber@unaffiliated/zokeber) |
| 21:43.08 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 21:45.46 | *** join/#asterisk [TK]D-Fender (n=joeblow@161.216.144.106) |
| 21:48.08 | *** join/#asterisk moodspeed (n=mood@66.49.173.92) |
| 21:57.21 | *** join/#asterisk Defraz (n=Defraz@c72co-edge-router.fuzecore.com) |
| 21:59.07 | *** join/#asterisk Defraz (n=Defraz@c72co-edge-router.fuzecore.com) |
| 21:59.53 | *** join/#asterisk Tuxxie (n=ryan@rrcs-70-63-90-226.midsouth.biz.rr.com) |
| 22:00.32 | Tuxxie | are phone number tied directly to pots lines? |
| 22:00.37 | Tuxxie | ie. |
| 22:01.17 | Tuxxie | if I have an 800 number on a pots line is there a way to two calls going to that number at the same time? |
| 22:01.42 | jblack | No. That's where other techologies come in. |
| 22:01.56 | [TK]D-Fender | tuxcrafter: call-waiting <- |
| 22:02.08 | jblack | Considering the channel, look at voip. Old school, look at ISDN or a T1. |
| 22:02.21 | jblack | THough it may be hard to find isdn these days |
| 22:03.38 | Katty | mister black! |
| 22:03.51 | Tuxxie | Thats what I was thinking also, but you know when managment steps in they dream up stuff that only make sence to managment. ;) |
| 22:04.11 | Tuxxie | make=makes |
| 22:04.20 | Katty | Tuxxie: i've foudn the best way to deal with that is to simply send an email to the telco with the request, and cc them into it. |
| 22:04.32 | Katty | Tuxxie: when the response comes back, they no longer try to argue with you about it. |
| 22:06.03 | Tuxxie | done. however, I was just makeing sure I was right. :) Thanks! |
| 22:06.20 | jblack | Mrs Atty! |
| 22:06.53 | jblack | Mrs? Miss? Ms? Mz? I never can remember which is which |
| 22:07.40 | leifmadsen | Ms is usually safest :) |
| 22:08.36 | jblack | fuckit. From now on, I'll just address anyone making less than a million dollars a year as Serf. |
| 22:08.56 | jblack | Serf leif. Serf James. |
| 22:09.18 | *** join/#asterisk malcolmd (n=malcolmd@pdpc/sponsor/digium/malcolmd) |
| 22:09.24 | jblack | Or wold Peon be better.... |
| 22:09.47 | Katty | atty? |
| 22:10.00 | Katty | that'd still be Miss. |
| 22:10.21 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 22:10.33 | jblack | Nah. Serf Atty. |
| 22:10.51 | jblack | Or Peon. I haven't quite decided yet which is better |
| 22:12.12 | jblack | Ohhh, How about drudge. |
| 22:12.23 | jblack | Drudge James has a ring to it. :) |
| 22:14.49 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 22:16.04 | *** join/#asterisk p3nguin (i=BuhPjX1J@asterisk-klv5.a2infotech.com) |
| 22:16.52 | aidinb | whats that make me.... |
| 22:16.54 | aidinb | Lord? |
| 22:16.59 | aidinb | lol |
| 22:19.51 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 22:21.16 | jblack | I refuse to believe that you make a million dollars a year until you give me a thousand bucks. Until the, you're also a serf. |
| 22:21.41 | Katty | hmm. serf Katty |
| 22:21.43 | Katty | i can deal with that |
| 22:21.56 | Katty | as long as i can find me a duke somewhere. |
| 22:22.03 | Katty | duchess katty has a nice ring to it |
| 22:22.05 | *** join/#asterisk ehsjoar (n=ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 22:23.31 | *** join/#asterisk ingcomrbr (n=ingcomrb@189.164.195.210) |
| 22:32.08 | tuxcrafter | [TK]D-Fender: i saw your "call-waiting <-" hint where you pointing at? |
| 22:39.02 | *** join/#asterisk rpm (n=russell@S0106000c29898b7e.cg.shawcable.net) |
| 22:40.10 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.218.194) |
| 22:40.23 | [TK]D-Fender | Much better |
| 22:46.16 | *** join/#asterisk ruben23 (n=RPL@122.55.48.243) |
| 22:46.28 | s34n | the ices command supposedly dumps audio to stdout |
| 22:46.40 | *** join/#asterisk comradeb14ck (n=comradeb@72.37.252.50) |
| 22:46.45 | comradeb14ck | yo :) |
| 22:47.08 | *** join/#asterisk `Sauron (n=sauron@dsl001-130-033.aus1.dsl.speakeasy.net) |
| 22:47.48 | s34n | I suppose that would be stdout for the original asterisk command that lauched the daemon... |
| 22:50.43 | *** join/#asterisk ZX81 (n=Matt_Rid@121-74-13-183.telstraclear.net) |
| 22:56.05 | *** join/#asterisk ZX81 (n=Matt_Rid@121-74-13-183.telstraclear.net) |
| 23:01.07 | *** join/#asterisk manxpower (n=ewieling@24.42.221.26) |
| 23:06.54 | *** join/#asterisk Godfather_ (n=Godfathe@187.Red-83-38-229.dynamicIP.rima-tde.net) |
| 23:07.27 | Godfather_ | its possibly to call 2 sips at the same time with an extension? |
| 23:07.47 | Qwell | Godfather_: exten => 123,1,Dial(SIP/phone1&SIP/phone2) |
| 23:07.56 | Godfather_ | Qwell, ty |
| 23:08.13 | Qwell | whichever answers first gets the call |
| 23:08.34 | Godfather_ | ok |
| 23:08.49 | *** join/#asterisk wonderworld (n=ww@ip-62-143-22-226.unitymediagroup.de) |
| 23:09.47 | *** part/#asterisk bsaxon (n=bsaxon@12.68.234.174) |
| 23:10.31 | Godfather_ | Qwell, and is there any possibility to have various users on a call? |
| 23:11.28 | Godfather_ | or give me a topic where i can find info about it |
| 23:11.41 | *** join/#asterisk wonderworld (n=ww@ip-62-143-22-226.unitymediagroup.de) |
| 23:12.07 | *** part/#asterisk ZX81 (n=Matt_Rid@121-74-13-183.telstraclear.net) |
| 23:18.52 | *** join/#asterisk Defraz (n=Defraz@corp.fuzecore.com) |
| 23:23.15 | *** join/#asterisk guax (n=guaxinim@unaffiliated/guaxinim) |
| 23:24.46 | ruben23 | hi |
| 23:24.52 | *** join/#asterisk raden_work (n=jon@69-179-99-17.stat.centurytel.net) |
| 23:26.33 | *** join/#asterisk btsteve (n=tstevens@24-196-234-39.dhcp.gwnt.ga.charter.com) |
| 23:26.39 | [TK]D-Fender | Godfather_: That could be a 3-way call. That could be a larger connference. Please describe the flow better |
| 23:27.45 | *** join/#asterisk Defraz (n=Defraz@corp.fuzecore.com) |
| 23:30.15 | Godfather_ | [TK]D-Fender, i'll read about this topic, just learning purpose, ty |
| 23:31.04 | Godfather_ | im just playing with extensions.conf |
| 23:31.08 | [TK]D-Fender | Godfather_: Well you asked a question and it didn't come out very clear as to what you wanted. So describe it again and better and we might be able to advise you |
| 23:31.21 | Godfather_ | [TK]D-Fender, ok |
| 23:31.55 | Godfather_ | image that a phone (pstn call) come to asterisk, and push a extension (100) |
| 23:32.15 | Godfather_ | with this extension i want to able to call all the sip/iax2 users |
| 23:32.21 | Godfather_ | like a normal pstn |
| 23:32.38 | [TK]D-Fender | Godfather_: then that would mean following what Qwell already advised |
| 23:32.46 | Godfather_ | [TK]D-Fender, but.. |
| 23:32.47 | [TK]D-Fender | [18:07]<Qwell>Godfather_: exten => 123,1,Dial(SIP/phone1&SIP/phone2) |
| 23:32.49 | Godfather_ | nope |
| 23:33.00 | [TK]D-Fender | Godfather_: Yes... you'd just have to add them ALL |
| 23:33.01 | Godfather_ | [TK]D-Fender, everbody can hear the same conversation |
| 23:33.03 | Godfather_ | the same channel |
| 23:33.18 | [TK]D-Fender | Godfather_: In your case you want them all to just "join in"? |
| 23:33.35 | Godfather_ | i tried that config, SIP/user1&SIP/user2 and works perfect |
| 23:33.41 | Godfather_ | [TK]D-Fender, yes |
| 23:33.44 | Godfather_ | sorry for the english |
| 23:33.58 | Godfather_ | [TK]D-Fender, i want to emulate a "normal" pstn line |
| 23:34.10 | [TK]D-Fender | Godfather_: So lets say 3 people are ringing. 1 person answers. A is connected to B. C is ringing, but A&B don't hear that. C eventually answers and gets added. same with D... like that? |
| 23:34.34 | [TK]D-Fender | godgodon a normal line the phone stops ringing when the 1st person answers. |
| 23:34.56 | Godfather_ | [TK]D-Fender, the 2ond option |
| 23:35.04 | Godfather_ | if one hang up the line |
| 23:35.13 | Godfather_ | then the others are been able to join |
| 23:35.19 | Godfather_ | but stop ringing his phone |
| 23:35.28 | Godfather_ | like a normal line |
| 23:36.16 | [TK]D-Fender | Godfather_: Ok, that isn't really going to happen... |
| 23:36.26 | Godfather_ | [TK]D-Fender, ? |
| 23:36.39 | [TK]D-Fender | Godfather_: You can't tell a call to stop making sound 1/2 way through calliong it and still be a call that is incoming. |
| 23:37.01 | *** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
| 23:37.06 | Godfather_ | [TK]D-Fender, i dont understand that last |
| 23:37.59 | Godfather_ | I want to emulate the normal phone calls in a house with no IP-telephones |
| 23:38.05 | [TK]D-Fender | Godfather_: Lest say A is the thincoming call. B,C,D are called with Dial. Tehy ALL ring. B answers it. the call is ANSWERED. * can't tell the others to keep ringing only silently |
| 23:38.23 | [TK]D-Fender | Godfather_: is this your home you want to do this to? |
| 23:38.41 | manxpower | Once the call is answered, the other phones CANNOT answer the call. |
| 23:39.03 | Godfather_ | [TK]D-Fender, its trying this at home yes. |
| 23:39.04 | *** join/#asterisk wonderworld (n=ww@ip-62-143-22-226.unitymediagroup.de) |
| 23:39.26 | Godfather_ | manxpower, cant answer, but can join just hanging up the phone? |
| 23:39.33 | manxpower | Godfather_: no! |
| 23:39.34 | Godfather_ | ok |
| 23:39.53 | Godfather_ | manxpower, then the 3-way that [TK]D-Fender said? |
| 23:40.00 | *** join/#asterisk Xetrov` (n=xetrov@unaffiliated/xetrov/x-827361) |
| 23:40.07 | Godfather_ | i really believe i have to read about this topic |
| 23:40.08 | manxpower | Godfather_: Is English not your native language? |
| 23:40.10 | [TK]D-Fender | Godfather_: Answer my previous question. You want to do this at your home? |
| 23:40.14 | [TK]D-Fender | manxpower: Clearly |
| 23:40.16 | Godfather_ | manxpower, obv no... |
| 23:40.39 | Godfather_ | [TK]D-Fender, yes.. |
| 23:40.42 | *** join/#asterisk SaiSoma (n=SaiSoma@adsl-074-167-136-030.sip.mob.bellsouth.net) |
| 23:41.01 | [TK]D-Fender | Godfather_: then just plug all the phones in your houses wiring onto ONE FXS port |
| 23:41.44 | Deeewayne | they all wont fit in one port :-) |
| 23:42.21 | Godfather_ | [TK]D-Fender, i said with an extension |
| 23:42.21 | Godfather_ | ;) |
| 23:42.48 | manxpower | Godfather_: [TK]D-Fender is assuming you are using analog phones. I am assuming you are using SIP phones. |
| 23:43.04 | [TK]D-Fender | stuffs Deeewayne into a non-bio-degradeable Big Mac container and tosses him out the side of his 57' Olds |
| 23:44.04 | Godfather_ | manxpower, using sip phones. |
| 23:44.04 | [TK]D-Fender | Godfather_: Forget SIP phones, and plug your house's phones onto a single ATA FXS port |
| 23:44.04 | [TK]D-Fender | Godfather_: You aren't going to get what you want. |
| 23:44.04 | [TK]D-Fender | (otherwise) |
| 23:44.04 | Godfather_ | then |
| 23:44.12 | Godfather_ | say Its impossible to emulate with sip phones and asterisk a normal line |
| 23:46.07 | [TK]D-Fender | Godfather_: I can imagine a way, but its a LOT of work, and the quality might not be so good.... |
| 23:46.37 | [TK]D-Fender | Godfather_: What is the point of SIP phones when they are all DUMB? |
| 23:46.39 | Godfather_ | [TK]D-Fender, the solution you give me you lost all the extensions of your sips |
| 23:47.00 | [TK]D-Fender | [18:46]<Godfather_>[TK]D-Fender, the solution you give me you lost all the extensions of your sips <- this sentence makes no sense |
| 23:47.22 | Godfather_ | [TK]D-Fender, acutally, i think you dont understand my question |
| 23:47.33 | Godfather_ | let me exlplain it better |
| 23:48.06 | Godfather_ | Image that a person outsite the system call to my house where asterisk is installed |
| 23:48.13 | Godfather_ | then the systen recieve the call |
| 23:48.32 | Godfather_ | and tell him a menu by voice like.. |
| 23:48.42 | Godfather_ | if you want to speak with defender press 101 |
| 23:48.49 | Godfather_ | with ivan press 102 .. etc |
| 23:48.54 | Godfather_ | and the last .. |
| 23:49.06 | Godfather_ | if you want to call all of them press 105 |
| 23:49.15 | *** join/#asterisk Get_The_Fish (n=IceChat7@c-24-8-50-199.hsd1.co.comcast.net) |
| 23:49.29 | Godfather_ | this last is like Dial(SIP/101&SIP/102...) |
| 23:49.43 | [TK]D-Fender | Godfather_: As I said, possible... but a LOT of work. |
| 23:49.44 | Godfather_ | but i want to ring all the SIPS and |
| 23:49.45 | *** join/#asterisk wonderworld (n=ww@ip-62-143-22-226.unitymediagroup.de) |
| 23:49.55 | Godfather_ | the others SIPS are been able to join |
| 23:49.58 | Godfather_ | that call |
| 23:49.59 | [TK]D-Fender | Godfather_: VERY complicated and requires external scripts to make sure things don't hang |
| 23:50.02 | Get_The_Fish | has anyone here worked with the LDAP realtime for SIP peers in 1.6.1.6? |
| 23:50.03 | Godfather_ | and hearing |
| 23:50.09 | Godfather_ | just hanging up the phone |
| 23:50.11 | [TK]D-Fender | Godfather_: and no, Qwell's multi-dial does not do it |
| 23:50.24 | Godfather_ | i know |
| 23:50.27 | Godfather_ | ok |
| 23:50.48 | [TK]D-Fender | Godfather_: VERY complicated and as I described... NOT WORTH IT |
| 23:51.01 | Godfather_ | ok, i forget about it |
| 23:51.49 | *** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun) |
| 23:52.05 | *** join/#asterisk wonderworld (n=ww@ip-62-143-22-226.unitymediagroup.de) |
| 23:52.18 | *** join/#asterisk Amorsen (n=Amorsen@94.127.50.7) |
| 23:57.39 | Get_The_Fish | has anyone here worked with the LDAP realtime for SIP peers in 1.6.1.6? |
| 23:59.14 | *** join/#asterisk voipmonk_ (n=voipmonk@dsl-67-55-17-41.acanac.net) |