| 00:06.14 | *** join/#asterisk thansen (n=thansen@70.65.133.117) |
| 00:27.07 | *** join/#asterisk denon (i=denon@sassinak.net) |
| 00:27.07 | *** mode/#asterisk [+o denon] by ChanServ |
| 00:29.57 | *** join/#asterisk ming_zym (n=ming_zym@124.127.101.0) |
| 00:32.20 | *** join/#asterisk jeffgus (n=jeffgus@2002:ad33:b504:0:0:0:0:1) |
| 00:34.06 | *** join/#asterisk obnauticus (n=l@about/windows/regular/obnauticus) |
| 00:41.27 | *** join/#asterisk bimbo (n=oso@200.66.18.220) |
| 00:42.54 | fofware | Other question... May i do a socket connection to Asterisk CLI? |
| 00:45.34 | fofware | Oh!!! yes to 5038 port good |
| 00:47.12 | *** part/#asterisk ruben23 (n=RPL@122.55.48.243) |
| 00:50.38 | p3nguin | What's that port for? |
| 00:51.43 | p3nguin | asterisk server console? |
| 00:53.41 | *** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
| 00:54.52 | drmessano | AMI |
| 01:03.21 | p3nguin | Does the AMI have to be running to use the console? |
| 01:04.40 | drmessano | O.o |
| 01:04.49 | drmessano | They're two entirely different things |
| 01:05.14 | p3nguin | The commenting in the top of manager.conf prompted the question. |
| 01:12.10 | fofware | p3nguin: yes it's the port of manager.conf and you can make a socket connetion and use like CLI |
| 01:12.41 | fofware | Ok guys I'm going to a party, have a good night |
| 01:12.49 | *** part/#asterisk WinZ (n=winz@82.146.61.218) |
| 01:18.45 | *** join/#asterisk matrix1233 (n=Administ@41.230.76.58) |
| 01:19.50 | matrix1233 | hello |
| 01:20.07 | matrix1233 | i have a problem :s |
| 01:20.17 | matrix1233 | i have this message : 1253948916 NOTICE[6550]: rtp.c:566 ast_rtp_read: Unknown RTP codec 96 received |
| 01:20.28 | matrix1233 | when i send a DTMF |
| 01:22.45 | matrix1233 | hello |
| 01:24.04 | matrix1233 | alo |
| 01:26.09 | *** join/#asterisk Defraz (n=Defraz@c72co-edge-router.fuzecore.com) |
| 01:30.49 | *** join/#asterisk manxpower (n=EWIELING@24.42.221.26) |
| 01:38.55 | ChannelZ | matrix1233: what kind of phone? |
| 01:41.10 | bimbo | hello, here's the scenario I've got: one asterisk server, 4 pstn lines connected to it and 2 other client computers connected via ethernet to the asterisk server each handling at most 2 of the 4 pstn lines, what would you recommend for this setup? |
| 01:42.39 | matrix1233 | ChannelZ: i have a trunk |
| 01:42.59 | drmessano | He's fighting an old asterisk bug, cant upgrade, cant recompile, before you waste your time |
| 01:43.01 | matrix1233 | ChannelZ: sip trunk asteriks 1.2 |
| 01:43.23 | matrix1233 | drmessano: yes lol |
| 01:44.00 | ChannelZ | ah. So can't fix the problem. |
| 01:44.01 | matrix1233 | drmessano: thx for your help i wil found a solution witout recomple |
| 01:44.26 | drmessano | You can patch rtp.c without recompiling? |
| 01:44.56 | ChannelZ | bimbo: In terms of what? |
| 01:44.58 | p3nguin | I guess you could patch it, but it wouldn't do much good until you did recompile it. |
| 01:45.12 | drmessano | heh |
| 01:45.16 | drmessano | Yeah |
| 01:45.31 | ChannelZ | bimbo: I'm running 4 POTS lines on a $150 computer from newegg and a Digium TDM800 |
| 01:45.39 | drmessano | .... |
| 01:45.51 | drmessano | You spent $150 on yours? |
| 01:46.01 | p3nguin | You can recompile just rtp, right? |
| 01:46.23 | ChannelZ | I think, it was over a year ago |
| 01:46.26 | drmessano | Big fuckin spender.. I got like $75 and some sexual favors I rather not relive invested in mine |
| 01:46.55 | drmessano | Sure, i took one for the team.. but it was a Celeron D.. |
| 01:47.04 | bimbo | ChannelZ: I mean, can this be actually done with asterisk? what would the client need? (asterisk is needed in order to grab the caller id and integrate that to a system through php AGI) |
| 01:47.06 | drmessano | Those are good, right? |
| 01:47.13 | p3nguin | lolno |
| 01:47.27 | drmessano | No? :( |
| 01:47.30 | p3nguin | I mean, uh, yeah. |
| 01:47.33 | matrix1233 | drmessano: perhaps i can install the same asterisk in anadher machine and i patch it and after i copy the chosen asterisk file |
| 01:48.23 | drmessano | matrix1233: Wont you need app_fluxcapacitor to go back 3 years in time? |
| 01:48.38 | ChannelZ | bimbo: yes asterisk can do it with the proper hardware (a TDM card of some sort for the POTS lines) |
| 01:50.37 | bimbo | ChannelZ: ok thank you, any wireless headset you recommend? |
| 01:51.04 | ChannelZ | no sorry I don't use them |
| 01:51.55 | matrix1233 | drmessano: i can't for now update my asterisk beacause this server is on production .. it work in real time |
| 01:52.13 | *** join/#asterisk ming_zym (n=ming_zym@124.127.101.0) |
| 01:52.29 | p3nguin | You could fix the problem and then restart it gracefully. |
| 01:52.33 | drmessano | Correction, it _mostly_ work in real time.. It would appear that "Shits broke" |
| 01:52.36 | matrix1233 | drmessano: for now there are 60 Teleoperator running |
| 01:54.14 | *** join/#asterisk ming_zym (n=ming_zym@124.127.101.0) |
| 01:54.29 | ChannelZ | well you have to restart it eventually |
| 01:54.59 | p3nguin | Or you can restart it gracefully and it'll restart when possible, no? |
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| 02:09.58 | ChannelZ | Wahoo! a complete distro update and * recompile later and my skypeforasterisk module is finally loading |
| 02:14.39 | *** join/#asterisk misteranonymous (n=tyler@udp115943uds.hawaiiantel.net) |
| 02:32.18 | *** part/#asterisk matrix1233 (n=Administ@41.230.76.58) |
| 02:34.17 | *** join/#asterisk darkdrgn2k (n=darkdrgn@bas2-toronto44-1176437585.dsl.bell.ca) |
| 02:34.24 | darkdrgn2k | hey guys where can i find the skype trunk module |
| 02:34.30 | darkdrgn2k | (i know its beta.. its for play not production) |
| 02:40.17 | Tim_Toady | its not beta anymore, you can buy it from digium |
| 02:48.17 | *** join/#asterisk darkdrgn2k (n=darkdrgn@bas2-toronto44-1176437585.dsl.bell.ca) |
| 02:48.26 | darkdrgn2k | tim_Toady: is there a "freeware" version? |
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| 02:48.47 | Tim_Toady | no |
| 02:49.05 | darkdrgn2k | dam prorpiatary protocols |
| 02:49.47 | darkdrgn2k | how much is it? |
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| 02:57.00 | dreadmaul | I remember reading something about letting the rtp channel redirect while the control channel stays with * to monitor the breakdown. How what that done? |
| 02:57.15 | dreadmaul | was* |
| 03:17.21 | ChannelZ | hmm is there a way to get 'core show channels' to not truncate the channel name? |
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| 03:33.26 | *** join/#asterisk AeroCloud (n=aero@ip98-165-113-125.ph.ph.cox.net) |
| 03:34.02 | AeroCloud | question: in dialplan, I am using t,1,Hangup, but on timeout, it is not hanging up |
| 03:34.04 | ChannelZ | oh, how wonderfully counter-intuitive. 'core show channels concise' actually _doesn't_ truncate the channel where as 'verbose' does. |
| 03:34.13 | AeroCloud | any ideas? |
| 03:34.23 | ChannelZ | What's it doing instead? |
| 03:34.29 | AeroCloud | recalling |
| 03:34.32 | *** join/#asterisk n3glv (n=n3glv@pool-71-240-8-118.pitt.east.verizon.net) |
| 03:34.40 | n3glv | hi guys |
| 03:34.42 | AeroCloud | repeating the s,1 |
| 03:34.58 | n3glv | I'm trying to get a lenny debian system running |
| 03:35.10 | n3glv | it's all good execpt for this dahdi module error |
| 03:35.17 | n3glv | fatal error inserting dahdi invalid module format |
| 03:35.19 | ChannelZ | hmm.. and the exten => t,1,Hangup() is in the same context? |
| 03:35.51 | n3glv | I see lots of ref to it on google, but no fixes so far. |
| 03:35.57 | AeroCloud | yeah |
| 03:36.18 | ChannelZ | what is the last step in your 's' extension? |
| 03:36.35 | AeroCloud | hangup |
| 03:36.45 | AeroCloud | step before is AGI(script) |
| 03:37.41 | p3nguin | Can you paste that entire context into a pastebin? |
| 03:37.59 | AeroCloud | I force a timelimit of 3 minutes to dial a number |
| 03:38.15 | AeroCloud | then it times out on a person, if they dont dial in 3 minutes, its going back to s,1 |
| 03:39.54 | ChannelZ | yeah it's hard to tell without seeing the dialplan |
| 03:40.08 | p3nguin | the whole context, not just parts of it. |
| 03:40.47 | AeroCloud | got it |
| 03:40.53 | AeroCloud | T,1,hangup |
| 03:41.03 | AeroCloud | need both t,1,hangup and T,1,hangup |
| 03:41.15 | p3nguin | hmm |
| 03:42.35 | AeroCloud | dialplan is simple |
| 03:43.20 | AeroCloud | s,1,Answer - s,2,AGI(script) - s,3,hangup - t,1,hangup - T,1,hangup - h,1,DeadAGI(script) - h,2,hangup |
| 03:43.47 | *** join/#asterisk coppice (n=chatzill@157.202.17.210.dyn.pacific.net.hk) |
| 03:44.11 | AeroCloud | the h,1 script is to cleanup custom call detail |
| 03:44.18 | ChannelZ | what is the AGI script doing? |
| 03:44.25 | ChannelZ | the T timeout is a timeout for the entire call |
| 03:44.41 | AeroCloud | T timeout is when absolute timeout is reached |
| 03:45.02 | AeroCloud | I set 3 minutes to dial a number, then set 120 minutes once a valid number is dialed |
| 03:45.34 | AeroCloud | its working perfectly now, thanx |
| 03:47.59 | ChannelZ | well ok |
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| 04:16.01 | *** mode/#asterisk [+o denon] by ChanServ |
| 04:17.20 | mctweep | hey all, if i want to setup enumlookup's in 1.4, do i need to setup a trunk or would something like exten=> _X.,1,ENUMLOOKUP(+${EXTEN}) work? |
| 04:23.37 | n3glv | fatal error inserting dahdi invalid module format |
| 04:23.43 | n3glv | I'm trying to get a lenny debian system running |
| 04:23.47 | n3glv | I see lots of ref to it on google, but no fixes so far. |
| 04:24.06 | kaldemar | mctweep: lookups have nothing to do with trunks |
| 04:24.41 | kaldemar | n3glv: are you compiling from source? |
| 04:24.52 | n3glv | yes, via a script actually |
| 04:26.15 | kaldemar | i haven't run into any problems compiling it in lenny. |
| 04:26.59 | n3glv | Loading DAHDI hardware modules: FATAL: Error inserting dahdi (/lib/modules/2.6.26-2-amd64/asterisk/dahdi/dahdi.ko |
| 04:27.10 | n3glv | that's not exactly it |
| 04:27.16 | n3glv | its 32bit |
| 04:27.28 | n3glv | but was doing google search and found same error there |
| 04:29.43 | kaldemar | what script are you using to compile? |
| 04:29.58 | n3glv | it's from a lenny mirror, let me get it |
| 04:30.09 | n3glv | (I tried updating it to latest dahdi) |
| 04:30.31 | n3glv | http://www.corenetworks.com.au/wiki/doku.php?id=debian_asterisk_freepbx_script |
| 04:30.41 | n3glv | http://www.corenetworks.com.au/wiki/lib/exe/fetch.php?media=asterisk-freepbx_0.6_dahdi_en_mod_2009-03-3.tar.gz |
| 04:30.46 | kaldemar | is the module compiled for a 32-bit system? |
| 04:31.22 | n3glv | it was made, on this system by the script |
| 04:31.22 | n3glv | like I said, that was someone elses post on a webpage |
| 04:31.26 | n3glv | mine is 32bit all the way |
| 04:31.47 | n3glv | Linux debian-pbx 2.6.26-2-686 #1 SMP Wed Aug 19 06:06:52 UTC 2009 i686 GNU/Linux |
| 04:32.52 | kaldemar | how about compiling the modules yourself? |
| 04:32.57 | n3glv | tried |
| 04:33.15 | n3glv | did dahdi then dahdi-tools asterisk, asterisk-addons etc |
| 04:37.12 | KavanS | is there a way to see if a number is being dialed/currently connected via SIP? via a macro or something? |
| 04:37.29 | KavanS | chan_avail is a nice idea, but I'm not sure it would work...I want to see if an external number is currently connected |
| 04:37.44 | n3glv | sip show channels? |
| 04:38.17 | KavanS | yeah, I want to do this via a macro...i.e. to evaluate whether the person's cell is already on a call |
| 04:38.19 | ChannelZ | you mean programmatically in your dialplan |
| 04:38.23 | KavanS | yep |
| 04:38.44 | *** join/#asterisk MIR100 (n=mir100@24-148-89-37.frg-bsr1.chi-frg.il.cable.rcn.com) |
| 04:40.20 | ChannelZ | hmm |
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| 04:42.38 | n3glv | kaldemar: I can set up for ssh if you would care to take a peek |
| 04:42.47 | kaldemar | KavanS: function DEVICE_STATE |
| 04:42.58 | KavanS | kaldemar, ok googling now |
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| 04:46.02 | kaldemar | KavanS: EXTENSION_STATE might also be useful |
| 04:59.15 | *** part/#asterisk bimbo (n=oso@200.66.18.220) |
| 05:05.18 | mctweep | hmm ok so all i have to do is setup enumlookups in extensions.conf |
| 05:05.58 | mctweep | i must be using the wrong format or something it doesn't pick it up, |
| 05:10.17 | *** part/#asterisk n3glv (n=n3glv@pool-71-240-8-118.pitt.east.verizon.net) |
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| 05:25.05 | mctweep | tryin to test enumlookup , using exten => _XXXXX,1,Set(foo=${ENUMLOOKUP(+14124030895,,,,e164.org)}), according to doco just returns no extension found |
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| 05:35.17 | *** join/#asterisk Cuban0 (n=cubano@190.26.232.46) |
| 05:35.35 | Cuban0 | good nite everyone i have a question about polarity reversal on Digium TDM400 |
| 05:35.43 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 05:36.07 | Cuban0 | in my scenario when remote party answers the call the line makes a polarity reversal |
| 05:36.29 | Cuban0 | and i need my asterisk to report call as ANSWERED when that reversal occurrs |
| 05:36.43 | Cuban0 | now it is counting the call as answered just as it is Dialed |
| 05:40.38 | manxpower | ~answers |
| 05:40.39 | infobot | answers is probably 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 |
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| 07:20.27 | TJNII | So, the problem with VNCing to a box with a identical window manager is that it is very easy to confuse the two, and you might issue a command to the wrong machine. |
| 07:20.37 | TJNII | Like, say, SHUTDOWN. |
| 07:22.14 | ChannelZ | heh oops |
| 07:22.41 | TJNII | doesn't think he had anything important open.... |
| 07:23.12 | ChannelZ | Well you don't now! |
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| 07:39.15 | tzafrir_laptop | TJNII, you also have that problem with ssh |
| 07:39.39 | tzafrir_laptop | This is why you should always have the hostname in the prompt |
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| 07:48.05 | drmessano | I did that once with a pair of windows servers.. Someone had a remote desktop open to another server. So I hop in to reboot the server.. Started the reboot and I see a remote desktop window close and desktop of the other server.. |
| 07:48.21 | drmessano | OOSp |
| 07:48.25 | drmessano | OOPS too |
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| 07:58.25 | TJNII | tzafrir_laptop: I've never had that problem with ssh because I do keep the hostname in the prompt. |
| 07:58.38 | TJNII | Now, screen inside screen inside screen, that leads to confusion. |
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| 09:41.10 | znh | If I see 'RTP packet send to... x.x.x.x.x' does that mean it actually is recevied? |
| 09:41.25 | znh | oh wait I also receive RTP messages |
| 09:41.45 | znh | okay so I can send and receive from the phone. However i'm not getting any audio |
| 09:43.08 | jgoo | Is there a level of debug that will be useful? I want to test making calls inbound, outbound, checking all work, and monitoring any fails. |
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| 09:44.01 | znh | jgoo: rtp debug shows allot |
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| 09:54.59 | jgoo | I don't want to show a lot of anything znh |
| 09:55.06 | jgoo | I want to show one/ two lines per call |
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| 10:12.37 | mctweep | newbie question , in ENUMLOOKUP(+${EXTEN},sip,c)}|, what does the "c" stand for? |
| 10:14.56 | kaldemar | it's a zone suffix. core show function ENUMLOOKUP |
| 10:17.46 | mctweep | aah ok so is that related to enum.conf that contains the domains to lookup |
| 10:23.53 | mctweep | when i use exten => 100,1,Set(foo=${ENUMLOOKUP(+437203001721)}), it just looks it up on e164.arpa, how do i get it to look on e164.org |
| 10:24.56 | kaldemar | have you looked at sample enum.conf? |
| 10:26.05 | kaldemar | e164.arpa will be used by default, you need to enable e164.org in enum.conf and use it explicitly in the lookup. |
| 10:26.38 | kaldemar | make your dialplan do another lookup if you don't get any results from e164.arpa |
| 10:27.26 | mctweep | ok will do ,do i need to specify it in enum.conf to work or can i just use it in extensions.conf by name |
| 10:29.00 | kaldemar | eh, c is not zone suffix. i was looking at the wrong function. "Option 'c' returns an integer count of the number of NAPTRs of a certain RR type." |
| 10:29.25 | kaldemar | you need the search directive in enum.conf |
| 10:29.51 | mctweep | aah ok yea i added search => e164.org but it still seems to be using default zone-suffix e164.arpa |
| 10:30.40 | mctweep | if i just put e164.org in enum.conf and nothing else does that replace the default zone-suffix or it doesn't work like that |
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| 10:44.37 | mctweep | thanks for your help kaldemar:) i'll play around with it |
| 10:52.13 | znh | If RTP packets are received and send, but no audio is heard (echo as test).. what would be going on? |
| 11:01.08 | znh | anyone care to help me out here... |
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| 12:44.15 | WinZ | guys, is it possible to connect an analog phone to asterisk via software modem in laptop? |
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| 12:48.22 | coppice | no |
| 12:49.22 | WinZ | so sad |
| 12:49.27 | WinZ | thank you |
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| 13:44.11 | Zambezi | Any Swede here who know where I can get a blacklist with numbers to telemarkingcompanies? I saw a comment on a homepage with a guy with a huge list of 1614 numbers so some list got to be out there. Collecting manually will take ages. |
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| 13:49.12 | jstew | Hi, is there a way in manager or cli to display parking lot extensions? |
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| 14:08.30 | Zambezi | :q |
| 14:09.09 | Zambezi | Sorry. Misprint. Tried to close Vim. |
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| 14:20.17 | Gugge | Zambezi, doesnt telemarketing call with hidden numbers in sweden? |
| 14:20.21 | Gugge | i know they do in denmark |
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| 14:39.42 | Zambezi | Gugge: I heard that too and I hate it. But at least it works for now. What if goverment made it illegal to with hidden number from telemarketing? Impossible probably. |
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| 14:51.06 | florz | it is in .de |
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| 15:02.05 | DigitalFlux-AFK | Hello Guys |
| 15:02.55 | DigitalFlux-AFK | i need there is a solution to make asterisk store SIP users information and CDR in a MySQL Database .. |
| 15:02.59 | DigitalFlux-AFK | any hints ? |
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| 15:09.37 | _ShrikE | DigitalFlux-AFK: realtime |
| 15:09.37 | Sandheaver | store data of the user in the CDRDB, not just hte extension |
| 15:09.38 | Sandheaver | ? |
| 15:09.56 | _ShrikE | DigitalFlux-AFK: cdr_mysql |
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| 15:40.47 | DigitalFlux-AFK | _ShrikE: Thanks, i am researching it right now .. |
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| 15:54.32 | *** join/#asterisk jgoo (n=floops__@athedsl-4538203.home.otenet.gr) |
| 15:54.38 | jgoo | When did misdn change to xml configuration files? |
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| 16:03.19 | hardwire | ew |
| 16:04.15 | drmessano | XML is so yesterday |
| 16:06.54 | p3nguin | What's the new fad? |
| 16:09.05 | jgoo | yaml |
| 16:09.08 | jgoo | json |
| 16:09.12 | jgoo | @ p3nguin |
| 16:09.40 | jgoo | not really fads, but used to better effect. And the examples are so redundant. It just makes me want to download a copy of it onto a usb stick so I can burn it |
| 16:13.48 | drmessano | All configs should be in HTML |
| 16:13.56 | Zambezi | I'm totally novice with this so I might prefer Trixbox, but Asterisk on my Debian will make it one less computer which is good. Is it still possible to fairly easy adapt/use script, addons made for Trixbox in Asterisk? Like blacklisting phonenumbers etc. |
| 16:14.59 | drmessano | Asterisk has no native GUI, so any GUI addons will not work.. Trixbox uses Asterisk, so any tricks in the config files are obviously the same |
| 16:16.47 | *** join/#asterisk xuser (n=xuser@unaffiliated/xuser) |
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| 16:17.26 | Zambezi | drmessano: I prefer CLI before GUI using Irssi, Bitlbee, rtorrent, Mutt. Sometimes even even, growisofs, wodim. You got the point... But it's a bit scary using IP-based phone since my connecting started to be a bit shacky. |
| 16:17.37 | drmessano | ok |
| 16:19.00 | p3nguin | Login, run screen, open the editor and your config. If you lose a connection, just login again and resume the screen session. |
| 16:19.19 | jgoo | With PAP2Ts - one one port, if it is engaged, and I call that number, it still rings - is this the Call Waiting Serv setting it has? |
| 16:19.31 | p3nguin | The changes don't take effect until after you save the files and reload something. |
| 16:20.51 | drmessano | No its the call waiting setting |
| 16:21.17 | p3nguin | The phone rings while you are using it? |
| 16:21.21 | p3nguin | Is that even possible? |
| 16:21.22 | drmessano | Why not just disable it for the extension? |
| 16:21.28 | drmessano | *call waiting" |
| 16:21.46 | drmessano | Disable it in Asterisk |
| 16:22.31 | drmessano | I force my ATA's to do very little.. I do everything in the dialplan |
| 16:23.14 | p3nguin | Yeah, just use the ATA to adapt. |
| 16:24.14 | drmessano | and the ritalin to cope? |
| 16:25.26 | p3nguin | or Jack and Coke |
| 16:25.34 | drmessano | In a can? |
| 16:26.03 | drmessano | They had that in Australia.. That was badass |
| 16:26.34 | p3nguin | That would be fun if you could get it as a premix. I've never seen it like that. |
| 16:27.13 | drmessano | One of the two was a namebrand.. It may have been the Jack, and it had "cola" |
| 16:27.51 | p3nguin | Still, Jack and cola off-the-shelf sounds fun. |
| 16:29.00 | drmessano | HA.. http://www.jdcollectorspage.com/cola.html |
| 16:29.03 | drmessano | There you is |
| 16:30.12 | jgoo | drmessano, call waiting - ok. I don't change anything on PAP2Ts at all, just fill in server, username, password |
| 16:30.27 | drmessano | Ok |
| 16:30.32 | drmessano | Nor do I |
| 16:30.47 | jgoo | so, I can disable this globally in asterisk? |
| 16:30.54 | drmessano | ~book |
| 16:30.55 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 16:34.32 | p3nguin | My dad would love to have those bottles. He collects bottles, plus he likes JD. |
| 16:34.43 | *** join/#asterisk [jmc] (n=John@93.45.204.59) |
| 16:35.20 | jgoo | if I set call-limit, while this would work for most clients, I have a few 4 line sip phones - what then? |
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| 16:40.54 | raden | morning |
| 16:41.53 | [TK]D-Fender | jgoo: "you can disable it on the PAP2 or "core show application chanisavail" |
| 16:43.20 | drmessano | 4 line sip phones are FOUR lines |
| 16:43.25 | jgoo | Thanks [TK]D-Fender , I'll disable it across the PAP2s for now |
| 16:43.25 | drmessano | FOUR peers |
| 16:43.48 | drmessano | (for the most part) |
| 16:45.31 | kaldemar | jgoo: don't use call-limit, it's deprecated anyway. use group functions. |
| 16:48.56 | *** join/#asterisk bluOxigen (n=asad@static-host119-73-69-224.link.net.pk) |
| 16:48.57 | jgoo | drmessano, I've not set it up like that - I have one peer, which handles four line (it just worked... ) does that mean it registers 4 times then?!? or something - not has 4 sip addresses? |
| 16:49.11 | jgoo | kaldemar, looks like I need to read more on this |
| 16:49.19 | jgoo | https://issues.asterisk.org/print_bug_page.php?bug_id=13488 <<< This looks like the bug I have been hunting |
| 16:49.58 | jgoo | Anyone using the SimpleEventCorrelator? What was the issues that led up to you using it? Has a newer asterisk fixed this? |
| 16:51.33 | *** join/#asterisk d-k-t (n=D@90.68.111.119.prime.ph.ip.sssworldwide.com) |
| 16:51.57 | jgoo | Anyone up on ISDN? With ISDN cards with NT and TE modes - if you are in the wrong mode, the card will not work right? It isn't as if it will work, but experience intermittent problems? |
| 17:02.43 | p3nguin | If you have one peer for your phone and you configured all four lines with that one peer, then yes, it registers four times. |
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| 17:02.45 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 17:03.18 | jgoo | p3nguin, I guess the phone is defaulted to do that, I just setup one extension for it |
| 17:04.24 | p3nguin | I used to configure my phones to register both lines as one peer, but then I decided to just not use line2 at all. |
| 17:05.50 | *** join/#asterisk sebbl (n=name@HSI-KBW-078-043-155-130.hsi4.kabel-badenwuerttemberg.de) |
| 17:06.10 | sebbl | moin... I search a "mini" dial out tool |
| 17:08.26 | [TK]D-Fender | Phones should not reg 4 times to the same peer... only the LAST will gt the reg |
| 17:11.08 | p3nguin | Initially, the reason I wanted to do it was to allow making an outgoing call on the second line. I later realized I didn't need to do it the way I thought. |
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| 17:29.40 | Zambezi | Can anyone recommend a product like Linksys SPA3102? URL: http://www.dustinhome.se/pd_5010197050.aspx Around same price which is maximum 100€/150$, but half the price would be nice. Two lines like the one I should is awesome, three and I'm sold. |
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| 17:59.55 | jgoo | https://issues.asterisk.org/view.php?id=13491#93173 |
| 18:00.11 | jgoo | geeeeeeeez wow. how can someone program like that? (look at the resolution ...) fffffffffffu- |
| 18:01.21 | jgoo | ok - I have 3 isdn TAs and I they each have 2 analogue ports on them - these get dial tones, can I push them into a PSTN digium card? like the TDM400P |
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| 18:10.59 | ChannelZ | jgoo: sure why not |
| 18:11.09 | *** join/#asterisk zeejee (n=Innocent@110.37.18.53) |
| 18:11.51 | zeejee | Hello |
| 18:12.04 | zeejee | i m having problem in mfcr2 signalling |
| 18:23.43 | zeejee | anybody there ? |
| 18:24.36 | zeejee | i am getting Chan 1 - Protocol error. Reason = Seize Timeout, R2 State = Seize Transmitted, MF state = MF Engine Off, MF Group = Forward MF init, CAS = 0x08 |
| 18:24.54 | zeejee | MFC/R2 protocol error on chan 1: Seize Timeout |
| 18:25.27 | moy | zeejee: nobody replied on the other end |
| 18:25.45 | zeejee | means protocol error ? |
| 18:26.06 | zeejee | and otherend is not configured for mfcr2 ? |
| 18:26.13 | *** join/#asterisk superc (n=superc@port-87-234-216-82.static.qsc.de) |
| 18:26.48 | zeejee | moy: my zone is set on CN |
| 18:26.55 | zeejee | if i set zone to ITU |
| 18:27.18 | zeejee | then i dont have error but a longggggggggggggggg wait and channel doesnt hangup untill i hangup through CLI |
| 18:27.25 | zeejee | and 1 more thing |
| 18:27.34 | superc | Hi. I use asterisk realtime for voicemail table. after upgrading to 1.6.0.15 asterisk seems not to find the user anymore: leave_voicemail: No entry in voicemail config file for '25' |
| 18:27.51 | zeejee | if i set the zone to CN, incoming indications dont come |
| 18:28.08 | zeejee | when i set zone to ITU, i get indication that a new call is on channel |
| 18:28.15 | zeejee | but it never get answered |
| 18:28.22 | zeejee | and keeps disconnecting and disconnecting |
| 18:28.49 | superc | is there any issue with asterisk and voicemail? |
| 18:28.59 | moy | zeejee: so, you're in china? |
| 18:29.03 | zeejee | no |
| 18:29.08 | zeejee | i m in pakistan |
| 18:29.17 | zeejee | and other end is siemens EWSD |
| 18:29.38 | zeejee | and telco tells me that pri is configured as CAS |
| 18:31.29 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 18:31.59 | moy | zeejee: call files in pastebin for both cn and itu |
| 18:32.21 | zeejee | ok |
| 18:35.01 | superc | any idea why the mailbox is not found with realtime? |
| 18:35.08 | superc | leave_voicemail: No entry in voicemail config file for '25' |
| 18:36.51 | zeejee | http://pastebin.com/m41b62d72 |
| 18:37.23 | KavanS | suggestions for where to buy TDM410 card for good price? |
| 18:38.34 | ChannelZ | try http://www.ipphone-warehouse.com/ |
| 18:38.42 | KavanS | right on |
| 18:39.06 | ChannelZ | I got my TDM card and phones from them |
| 18:40.15 | KavanS | cool |
| 18:41.06 | ChannelZ | course this is US, I didn't ask where you were |
| 18:41.25 | KavanS | yep, in the good ol' u.s. |
| 18:41.30 | ChannelZ | ok :) |
| 18:44.04 | superc | mh... again to voicemail... I'm 1 step further: I used the command in Realtime: Voicemail(s25) ... the s is the option to not to play the 'leave a message....blabla' ... without this 's' option it works. bug? |
| 18:44.35 | [TK]D-Fender | superc: "core show application voicemail" |
| 18:45.04 | zeejee | moy : you there ? |
| 18:45.33 | superc | s - Skip the playback of instructions for leaving a message to the |
| 18:45.33 | superc | <PROTECTED> |
| 18:45.46 | superc | so whats wrong? |
| 18:47.00 | superc | btw: voicemail(25,s) also doesn't work |
| 18:47.02 | superc | leave_voicemail: No entry in voicemail config file for '25,s' |
| 18:47.31 | superc | I'm using the searchcontext=yes option on voicemail.conf... But I also tried voicemail(25@default,s)... no luck |
| 18:49.37 | zeejee | http://pastebin.com/m41b62d72 |
| 18:49.50 | superc | tkd-fender?... so any idea? |
| 18:56.38 | moy | zeejee: give me a minute |
| 18:57.43 | zeejee | moy: sure, take your time. comforting that u r here :) |
| 18:58.54 | [TK]D-Fender | superc: Maybe you should show soem COMPREHENSIVE backup, not just piecemeal |
| 19:00.15 | *** join/#asterisk objorn (n=objorn@unaffiliated/objorn) |
| 19:00.43 | superc | mh? ... what do you mean? |
| 19:01.24 | objorn | how would you setup the phone to communicate by "dialing" into another ip? |
| 19:01.25 | retentiveboy | Where would I start looking into why "core show hints" says a Polycom station is Idle when it's ringing or on a call. Seems it's only changing from Idle when I put it on hold. |
| 19:01.34 | superc | Actually I'm just trying to use the basic asterisk voicemail command. |
| 19:01.49 | objorn | i don't want to pay to use voip, i figure one or two persons can have a server setup for voip communication |
| 19:01.53 | superc | voicemail(s25) should just play a beep and then record the message. It doesn't |
| 19:02.37 | objorn | we still have to pay for internet, but paying to use a voip service sounds ludicrous |
| 19:02.50 | KavanS | objorn, hahaha wtf? |
| 19:03.09 | KavanS | objorn, well if you plan on using voip to dial the rest of the world, you kinda have to pay for it |
| 19:03.16 | moy | zeejee: I don't think they are using MFC R2 .... they may be using DTMF R2 .... have you tried an incoming call? |
| 19:03.18 | KavanS | objorn, but like...asterisk to asterisk, of course is free |
| 19:03.29 | zeejee | yeah |
| 19:03.37 | zeejee | moy : pasted incoming call on pastebin as well |
| 19:03.49 | zeejee | moy : is DTMF R2 signalling is supported ? |
| 19:04.59 | objorn | KavanS: i wasn't thinking of using voip to dial out to the rest of the world, i was thinking of me and a few friends using voip to talk online |
| 19:05.03 | zeejee | moy : http://pastebin.com/m41b62d72 pastebin updated for incoming CLI log |
| 19:05.15 | KavanS | objorn, yep, you can definitely do that |
| 19:05.51 | objorn | but does it have to be asterisk to asterisk? i was thinking that maybe one friend could have twinkle, another ekiga, and i have asterisk |
| 19:06.02 | KavanS | objorn, yeah you can use other sip clients to connect to your asterisk |
| 19:06.14 | *** join/#asterisk leifmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 19:06.14 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 19:06.27 | superc | no idea why voicemail(s25) doesn't work while voicemail(25) does? .. Its no configuration issue, is it? |
| 19:06.32 | objorn | maybe asterisk is overkill for such a simple task, but i plan on having asterisk setup soon for my regular phone as well, so why not, right? it'd keep me from being tied down to the computer when talking |
| 19:06.58 | objorn | so, are there instructions on some kind of wiki for doing this/ |
| 19:07.11 | KavanS | objorn, certainly...I don't like other voice apps unless I have to...using a regular phone makes it nice, then of course being able to use the internet is nice ;) |
| 19:07.47 | objorn | i just want the calls coming into my usb headset, so i can do whatever the heck i want :) |
| 19:08.06 | objorn | within the usb signal range :P |
| 19:10.26 | superc | ok got it... its not like the help function suggests |
| 19:10.42 | superc | you mustn't use voimail(25,s) but voicemail(25|s) |
| 19:11.08 | superc | voicemail(s25) like in 1.2.xx doesn't work at all |
| 19:11.09 | Sandheaver | superc: what version of asterisk |
| 19:11.18 | superc | 1.6.0.15 |
| 19:11.32 | Sandheaver | huh. i thought | as a delimiter went away with 1.4 |
| 19:11.34 | superc | No I use voicemail(25|s) and it works as it should... not good |
| 19:11.35 | Sandheaver | well, if it works. |
| 19:11.51 | superc | probably someone is a coder here and can fix it? |
| 19:11.59 | superc | now, as we know what it is |
| 19:13.59 | Sandheaver | well, maybe it's not a delimiter for that app. maybe it's just how that app works. |
| 19:14.38 | moy | zeejee: sounds like dtmf r2 ... but afaik dtmf r2 exists in different variants as well, you can try this openr2 branch: http://code.google.com/p/openr2/source/browse/#svn/branches/release-1-dtmf |
| 19:14.45 | moy | then in Asterisk |
| 19:14.48 | superc | I'm afraid not |
| 19:14.51 | superc | As of 1.6 |
| 19:14.51 | superc | As of version 1.6 flags must be passed after the box number seperated by a comma. The usage of the pipe (|) symbol is deprecated. Following format is right for 1.6 and higher (tested on 16th of juli 2009 in version 1.6.1.1). |
| 19:14.56 | moy | use mfcr2-1.4 branch |
| 19:14.59 | superc | this is what is written in voip-info.org |
| 19:15.07 | zeejee | moy : ok |
| 19:15.10 | superc | ao actually also s25 should work |
| 19:15.15 | zeejee | moy : do i only need to compile this branch ? |
| 19:15.27 | superc | ...The boxnumber may be preceeded by one or more flags or these may be specified as the second argument... |
| 19:15.56 | moy | yes both, openr2 and asterisk branches |
| 19:16.16 | zeejee | moy : how to download from there ?:) (sorry for the noob ques) |
| 19:16.20 | *** part/#asterisk superc (n=superc@port-87-234-216-82.static.qsc.de) |
| 19:17.08 | moy | svn co http://osvn.digium.com/svn/asterisk/team/moy/mfcr2-1.4 |
| 19:17.13 | moy | svn co http://svn.digium.com/svn/asterisk/team/moy/mfcr2-1.4 |
| 19:17.29 | moy | in Asterisk you need 2 new settings |
| 19:17.36 | zeejee | downloading |
| 19:17.39 | moy | mfcr2_dtmf_detection=yes |
| 19:17.50 | moy | mfcr2_dtmf_dialing=yes |
| 19:18.10 | zeejee | moy : i need to first compile openr2 again and then your branch ? |
| 19:18.55 | moy | svn co http://openr2.googlecode.com/svn/branches/release-1-dtmf |
| 19:19.10 | moy | first compile and re-install the release-1-dtmf branch |
| 19:19.16 | moy | then the asterisk branch |
| 19:19.16 | zeejee | hmm |
| 19:20.28 | zeejee | moy : http://openr2.googlecode.com/svn/branches/release-1-dtmf is installed |
| 19:22.26 | zeejee | moy : signalling=? in chan_dahdi.conf ? |
| 19:23.26 | objorn | http://www.asteriskguru.com/tutorials/ is nice |
| 19:23.40 | objorn | as is http://www.voip-info.org/wiki/index.php?page=Asterisk and http://astbook.asteriskdocs.org/ |
| 19:26.13 | zeejee | moy: signalling=mfcr2 ? in chan_dahdi.conf with this branch ? |
| 19:26.59 | [TK]D-Fender | [15:15]<superc>ao actually also s25 should work < no, it shouldn't |
| 19:27.21 | [TK]D-Fender | SuPrSluG: your syntax is wrong, jsut like the instructions say, and "|" is not a valid delimiter |
| 19:28.14 | moy | zeejee: yes, just add those 2 parameters I said |
| 19:28.34 | zeejee | moy: wooh, msgs changed |
| 19:28.48 | zeejee | pastebining it in pastebin |
| 19:29.15 | zeejee | moy : http://pastebin.com/m793ef16d |
| 19:31.11 | *** join/#asterisk iksik (i=xk@livedata.pl) |
| 19:31.56 | *** join/#asterisk [8none1] (n=[8none1]@c-68-52-23-77.hsd1.tn.comcast.net) |
| 19:35.54 | *** join/#asterisk smultron (n=smultron@mirbsd/staff/smultron) |
| 19:37.18 | moy | zeejee: enable debugging |
| 19:37.30 | moy | in both logger.conf and chan_dahdi.conf |
| 19:37.36 | moy | all messages enabled |
| 19:37.50 | moy | mfcr2_logging=all in chan_dahdi.conf |
| 19:37.51 | zeejee | debugging is on |
| 19:37.55 | zeejee | its there |
| 19:39.30 | zeejee | moy : let me paste my chan_dahdi.conf |
| 19:40.03 | zeejee | moy : my chan_dahdi.conf http://pastebin.com/m5df03fae |
| 19:42.02 | *** join/#asterisk nightrid3r (n=kvirc@78-20-232-172.access.telenet.be) |
| 19:48.37 | *** part/#asterisk cilkay (n=cilkay@CPE00d0b743a22f-CM0011ae01fcbe.cpe.net.cable.rogers.com) |
| 19:49.06 | retentiveboy | Where would I start looking into why "core show hints" says a Polycom station is Idle when it's ringing or on a call. Seems it's only changing from Idle when I put it on hold. |
| 19:49.42 | zeejee | moy : is that correct ? |
| 19:49.54 | [TK]D-Fender | retentiveboy: perhaps you should be showing us your configs |
| 19:50.04 | retentiveboy | k |
| 19:50.09 | retentiveboy | gimmie a fe :) |
| 19:51.51 | objorn | okay, i have asterisk installed. now what i would like to do is have a test voip call. how would i do this? my point earlier is that it makes no sense to have a sip address provider when you should be able to just use an ip address and some protocol (h.323, jingle, or other) |
| 19:52.26 | *** join/#asterisk luca`gervasi (n=ashura@host122-160-dynamic.42-79-r.retail.telecomitalia.it) |
| 19:52.28 | luca`gervasi | Hallo |
| 19:54.23 | objorn | do i have to go through the http://astbook.asteriskdocs.org/en/2nd_Edition/asterisk-book-html/asterisk-book.html#asterisk-CHP-4 "initial configuration" part of this book if i've already installed asterisk through debian? |
| 19:54.57 | objorn | guess it couldn't hurt, but man, that's one long mofo chapter |
| 19:55.20 | objorn | i guess i have to since i have to create channels |
| 19:57.10 | *** join/#asterisk [8none1] (n=[8none1]@c-68-52-23-77.hsd1.tn.comcast.net) |
| 19:57.22 | zeejee | moy : you need any log ? |
| 19:59.31 | ChannelZ | objorn: yes you still have to configure phones, extensions, hardware, etc |
| 19:59.56 | ChannelZ | the only part you skipped by installing a package was compiling/installing it |
| 20:00.58 | [TK]D-Fender | obWHO are you going to call? |
| 20:01.05 | [TK]D-Fender | objorn: WHO are you going to call? |
| 20:01.51 | *** join/#asterisk manxpower (n=EWIELING@24.42.221.26) |
| 20:02.31 | manxpower | ~answers |
| 20:02.31 | infobot | extra, extra, read all about it, 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 |
| 20:02.53 | luca`gervasi | I got a lot of "WARNING[15275]: app_dial.c:1528 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Unknown)" ... what are they??? |
| 20:03.09 | moy | zeejee: yes, but you need to enable console logging in logger.conf as well |
| 20:03.13 | [TK]D-Fender | lucaGo look at the status of the peer you are calling |
| 20:04.03 | zeejee | moy : i have enabled full in logger.conf |
| 20:04.55 | luca`gervasi | [TK]D-Fender, i use DIAL(internalpeer_N) when i receive a call, but all those peers are not currently registered (they are some accounts used for test and so on)... |
| 20:05.24 | moy | full goes to /var/log/asterisk/full .... enable the same settings for the console => file and then pastebin what you see in the console for an incoming call |
| 20:05.25 | manxpower | luca`gervasi: you can't call a peer if Asterisk doesn't know it's IP address. "sip show peers" |
| 20:05.29 | moy | and then for an outgoing |
| 20:05.33 | [TK]D-Fender | luca`gervasi: If they aren't registered and * has no IP to call, then it can't call. |
| 20:05.39 | ChannelZ | So what you're asking is "I'm trying to call a phone that isn't hooked up and I'm getting a warning in the console" ? |
| 20:05.46 | [TK]D-Fender | luca`gervasi: and that dial statment doesn't look valid |
| 20:06.11 | manxpower | [TK]D-Fender: that Dial couldn't possibly cause the error message he's getting. He lied to us. |
| 20:06.19 | objorn | [TK]D-Fender: that's a wonderful question, probably one of my friends using twinkle or ekiga |
| 20:06.22 | zeejee | moy : http://pastebin.com/m1d55bf4d |
| 20:06.26 | [TK]D-Fender | manxpower: I know... I'm trying to break that down too |
| 20:06.36 | zeejee | moy : this is last lines of var/log/asterisk/full |
| 20:06.50 | manxpower | luca`gervasi: next time paste the ACTUAL Dial line, not what you think it is. |
| 20:07.14 | luca`gervasi | yes, i'm pasting to pastebin, just a sec :D |
| 20:08.07 | *** join/#asterisk jcape (n=jcape@adsl-99-132-249-229.dsl.chcgil.sbcglobal.net) |
| 20:08.28 | luca`gervasi | http://pastebin.com/md13f86f <--- dialplan plus log |
| 20:09.00 | manxpower | luca`gervasi: Your peers are not registered. |
| 20:09.03 | objorn | why buy a sip phone? it can't be too difficult to build your own. it's as simple as voice recording and bluetooth, right? |
| 20:10.05 | [TK]D-Fender | objorn: Right because you can produce something cheaper and better than those whose cost is driven down by mass production and actual competence. |
| 20:10.05 | ChannelZ | You can build your own car too |
| 20:10.18 | [TK]D-Fender | objorn: Go right ahead... try to do it for < 100$.... |
| 20:10.26 | manxpower | I assume that objorn's statement is a troll. |
| 20:10.27 | [TK]D-Fender | objorn: Have fun |
| 20:10.27 | luca`gervasi | manxpower, yes, [TK] says that before. i pasted just because you said that i was lieing :D |
| 20:10.33 | objorn | i bet i could use my neorunner |
| 20:10.47 | manxpower | luca`gervasi: You need to fix the problem with the peers. |
| 20:10.47 | objorn | shouldn't be a prob |
| 20:10.59 | [TK]D-Fender | [16:04]<luca`gervasi>[TK]D-Fender, i use DIAL(internalpeer_N) when i receive a call, but all those peers are not currently registered (they are some accounts used for test and so on)... |
| 20:11.51 | dustybin | rings himself a few more times |
| 20:11.55 | luca`gervasi | 16:04? wow... where are you from? :o 22:10 here in italy :D |
| 20:11.58 | dustybin | brb phone |
| 20:12.00 | [TK]D-Fender | objorn: And that is...? |
| 20:12.29 | [TK]D-Fender | luca`gervasi: That dial command does not look like anything in your pastbin. |
| 20:12.34 | objorn | [TK]D-Fender: the openmokeo neo freerunner |
| 20:12.34 | ChannelZ | Why are you having it dial through all these extensions in series? |
| 20:12.38 | [TK]D-Fender | luca`gervasi: Whichis what manxpower was getting at |
| 20:12.39 | objorn | openmoko* |
| 20:12.49 | [TK]D-Fender | objorn: Good to see you can get the name right |
| 20:13.05 | [TK]D-Fender | objorn: And there could very well be a VoIP client for it already |
| 20:13.07 | luca`gervasi | what do you mean? (sorry, i'm not english) |
| 20:13.26 | [TK]D-Fender | objorn: then again... you already BOUGHT that... not so much "building a voip phone" now is it? |
| 20:13.44 | ChannelZ | Your pastebin has the 'start' extension dialing SIP/1000, then SIP/1001, then SIP/1002, etc. Why are you doing this |
| 20:13.45 | zeejee | moy : do i need spandsp lib ? |
| 20:13.45 | [TK]D-Fender | luca`gervasi: Means you showed us a bullshit sample asking why it doesn't work. |
| 20:13.54 | zeejee | moy : do i need spandsp lib ? |
| 20:13.58 | luca`gervasi | ChannelZ, those are some phones i use (one on my desk, one on the mobile, one on the palmtop some remotes...) |
| 20:14.00 | [TK]D-Fender | luca`gervasi: Now show us the PEER DUMPS. |
| 20:14.20 | luca`gervasi | [TK]D-Fender, peer dumps => show sip peers ? |
| 20:14.25 | ChannelZ | yeah but what is the purpose of dialing them all in series? |
| 20:15.01 | manxpower | zeejee: you need spandsp for fax |
| 20:15.01 | ChannelZ | And if all of the phones are not connected you are going to get those warnings as it tries (which would be normal) |
| 20:15.01 | *** join/#asterisk tnt_ (n=tnt_@142.12-244-81.adsl-dyn.isp.belgacom.be) |
| 20:15.12 | [TK]D-Fender | luca`gervasi: NO. "sip show peer 1000", etc |
| 20:15.17 | manxpower | You need spandsp for Asterisk based fax, not just regular fax to a fax machine. |
| 20:15.26 | tnt_ | Is there a goog tutorial on how to write a channel driver ? (that explains all those call backs :) |
| 20:15.33 | luca`gervasi | ChannelZ, i suppose that would be the correct way to dial multiple phones...how can i do to fix it? :D |
| 20:15.36 | manxpower | tnt_: see #asterisk-dev |
| 20:15.44 | zeejee | manxpower : thanks, but i m using mfcr2 |
| 20:15.53 | ChannelZ | If you want to ring them all at once, you do Dial(SIP/1000&SIP/1001&SIP/1002) etc |
| 20:16.07 | manxpower | zeejee: you may need it for MFCr2 as well |
| 20:16.21 | luca`gervasi | [TK]D-Fender, lot of lines... should i look for something particular or i have to paste them all? |
| 20:16.30 | tnt_ | manxpower: thanks. |
| 20:16.30 | [TK]D-Fender | PASTEBIN |
| 20:16.44 | ChannelZ | How you have it now, it's going to try SIP/1000 - then if no one answers in 10 seconds (or it's not connected) it will try SIP/1002 for 10 seconds... |
| 20:16.55 | zeejee | manxpower : i m using openr2-1.2.0 and i guess spandsp is not necessary for openr2 |
| 20:17.08 | manxpower | zeejee: I know nothing about R2 requirements |
| 20:17.13 | luca`gervasi | ChannelZ, what if (example) SIP/1004 should ring ONLY if SIP/1001 is not available? |
| 20:17.26 | [TK]D-Fender | luca`gervasi: PASTEBIN |
| 20:17.35 | luca`gervasi | [TK]D-Fender, pasting |
| 20:18.02 | manxpower | luca`gervasi: When Dial exits it sets variables HANGUPCAUSE and DIALRESULT You use dialplan logic to do do different things based on the values of those variables. |
| 20:18.06 | *** join/#asterisk KingOfDos (n=irssi@ht1-ix.kingofdos.net) |
| 20:18.17 | ChannelZ | then what you have is sort of fine (but whose going to stay on the line long enough of 1000, 1001, 1002, 1003, and 1004 are all gone but 1005 is there?) |
| 20:18.24 | *** join/#asterisk Slashman (n=Slash@ariane.fimasys.com) |
| 20:18.33 | zeejee | moy : any other log needed moy ? |
| 20:18.37 | [TK]D-Fender | ChannelZ: Waste of time until he looks at WTF is is going on |
| 20:18.38 | *** join/#asterisk tgunr (n=tgunr@cust-66-249-166-11.static.o1.com) |
| 20:18.45 | manxpower | CHANUNAVAIL means "peer can't be contacted", BUSY means "peer available, but can't accept more calls, etc. |
| 20:19.15 | ChannelZ | TK: I think what he's not getting is that the warnings are normal if the peers aren't connected. It's like asking why your phone isn't ringing when it's not plugged into the wall. |
| 20:19.27 | [TK]D-Fender | BUSY = peer contactable and refuse for being unable or unwilling to accept the call. |
| 20:19.50 | luca`gervasi | http://pastebin.com/m276b73bb |
| 20:20.17 | [TK]D-Fender | luca`gervasi: Addr->IP : (Unspecified) Port 5060 <------ not registered |
| 20:20.26 | [TK]D-Fender | luca`gervasi: WHY isn't this device registered? |
| 20:21.36 | luca`gervasi | [TK]D-Fender, sorry, i don't speak english very well, so i didn't make myself clear: some extensions are not registeret and should not (depending on daytime). i'm aware of that. my question is... is there a way to avoid this warning ? |
| 20:21.45 | [TK]D-Fender | luca`gervasi: NO |
| 20:21.50 | ChannelZ | Yeah, don't read it. |
| 20:21.56 | [TK]D-Fender | luca`gervasi: It is telling you it failed. thats what a warning is |
| 20:21.56 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 20:21.57 | tnt_ | Is there any external module to support GSM-AMR & GSM-EFR ? (like there is for g729) |
| 20:22.07 | ChannelZ | Or turn the verbosity of the console down so low it doesn't say anything. |
| 20:22.08 | [TK]D-Fender | luca`gervasi: so IGNORE IT |
| 20:22.13 | luca`gervasi | [TK]D-Fender, ok thanks :D |
| 20:22.22 | *** join/#asterisk ChUbB (n=IceChat7@62-31-213-230.cable.ubr12.aztw.blueyonder.co.uk) |
| 20:22.33 | luca`gervasi | is there a way to make asterisk dial only registered devices? |
| 20:22.39 | ChannelZ | the only other way to avoid it is to put lots of logic in the dialplan to check each peer first before trying to dial it. |
| 20:22.58 | ChannelZ | But it's a stupid amount of work just to supress a harmless warning message on the console |
| 20:23.06 | [TK]D-Fender | "How do I stop seeing these errors?", "CLOSE YOUR FUCKING EYES" |
| 20:23.12 | luca`gervasi | ChannelZ, thanks for the info, i totally agree with you :D |
| 20:23.25 | luca`gervasi | [TK]D-Fender, ok ok... |
| 20:23.47 | [TK]D-Fender | see a lot of nut-jobs out here on the weekend... |
| 20:24.21 | [TK]D-Fender | luca`gervasi: You tell it to take an action, not check first. |
| 20:24.30 | moy | zeejee: no, spandsp is not needed |
| 20:24.32 | [TK]D-Fender | lucanot that the outcome makes any difference |
| 20:24.42 | moy | I embedded the dtmf and mf detectors in openr2 |
| 20:24.57 | zeejee | moy : ok |
| 20:25.04 | zeejee | moy : yeah i have read that |
| 20:25.12 | luca`gervasi | [TK]D-Fender, ok, i think I can close my fucking eyes, now :D |
| 20:25.20 | zeejee | moy : what should i try next ? |
| 20:26.07 | moy | zeejee: I don't see outgoing call attempt |
| 20:26.22 | zeejee | not pasted in pastebin |
| 20:26.37 | zeejee | moy : shld i paste outgoing call attempt as well in pastebin ? |
| 20:27.00 | moy | yes |
| 20:27.58 | *** join/#asterisk uqlev (n=yuriy@91.184.221.31) |
| 20:28.34 | zeejee | moy : http://pastebin.com/m1a515ad3 <--- outgoing call attempt |
| 20:28.57 | zeejee | moy : but call never reached to the called number |
| 20:29.37 | moy | you need to figure out with whoever configured the other side if they are using MF tones or DTMF tones for dialing |
| 20:29.58 | moy | that is, whether they are using MFC R2 or DTMF R2 |
| 20:30.11 | moy | also |
| 20:30.13 | moy | you can try |
| 20:30.17 | moy | using dahdi_monitor |
| 20:30.24 | moy | to record the audio on the line |
| 20:30.37 | zeejee | moy : just run dahdi_monitor ? |
| 20:30.53 | moy | zeejee: read the help for dahdi_monitor |
| 20:30.58 | *** join/#asterisk gr0mit (n=tim@2001:8b0:3d9:0:8561:aca5:8a82:1204) |
| 20:31.05 | moy | then run it, make an incoming call, and then after the call fails |
| 20:31.08 | moy | stop dahdi_monitor |
| 20:31.56 | moy | and put the file in some server where I can download it via ftp or http |
| 20:32.08 | moy | zeejee: and keep the discussion here in #asterisk ... unless you want to pay me for consultancy, in which case we can talk via IM or e-mail :) |
| 20:32.38 | zeejee | moy : ok |
| 20:33.10 | zeejee | moy : i dont mind paying you |
| 20:33.42 | *** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 20:33.53 | zeejee | as far as i will be able to rcv and place calls |
| 20:37.26 | zeejee | moy : i am using this command to monitor |
| 20:37.37 | zeejee | dahdi_monitor 23 -f dahdi-23 |
| 20:37.52 | zeejee | but incoming calls falls on random channel |
| 20:38.22 | moy | then you will need to monitor all of them, with some bash script launching dahdi_monitor on all of them or something like that |
| 20:38.30 | moy | what I need is simply the audio on that channel |
| 20:38.44 | moy | on the incoming channel |
| 20:38.51 | zeejee | ok trying |
| 20:39.17 | zeejee | shld i add .wav after dahdi-23 ? like dahdi_monitor 23 -f dahdi-23.wav ? |
| 20:39.58 | *** join/#asterisk [8none1]_ (n=[8none1]@c-68-52-23-77.hsd1.tn.comcast.net) |
| 20:41.32 | *** part/#asterisk sircolin (n=sircolin@83.216.68.241) |
| 20:41.44 | moy | zeejee: forget the man page, there is better help if you type "dahdi_monitor" without arguments |
| 20:43.06 | moy | zeejee: dahdi_monitor 1 -v -R received-audio.raw |
| 20:44.00 | zeejee | moy : ok, making script |
| 20:45.54 | *** join/#asterisk williammanda (n=william@adsl-234-201-89.cha.bellsouth.net) |
| 20:46.50 | zeejee | moy : where to put the file ? |
| 20:47.21 | zeejee | ok hold |
| 20:47.42 | williammanda | I have a couple of questions concerning asterisk and asterisknow.... |
| 20:47.47 | moy | if I can copy it through scp that's even better |
| 20:48.22 | *** join/#asterisk Polysics (n=Luca@host112-73-dynamic.16-79-r.retail.telecomitalia.it) |
| 20:48.24 | Polysics | hello |
| 20:48.25 | williammanda | Is asterisknow an appliance? |
| 20:48.32 | Polysics | anyone worked with skil-based routing? |
| 20:48.36 | Polysics | *skill |
| 20:48.52 | zeejee | moy : http://208.43.250.118/received-audio.raw |
| 20:49.00 | Polysics | i need to route calls based on two sets of skills: language spoken and field of interest |
| 20:49.14 | williammanda | can it be installed on an existing linux platform and the platform stil have functionality? |
| 20:49.18 | Polysics | i've tried to figure out something that could work using queues, but my head spins |
| 20:49.21 | moy | zeejee: ok give me some minutes |
| 20:49.31 | zeejee | moy : sure, i hope i have done it right |
| 20:49.33 | ChannelZ | williammanda: yes |
| 20:49.33 | Polysics | williammanda, afaik *now is a distribution |
| 20:49.46 | moy | zeejee: you are on E1 right? |
| 20:49.52 | zeejee | moy : yes |
| 20:50.10 | ChannelZ | what Polysics said - *now is somewhat standalone but regular * is just an app |
| 20:50.18 | williammanda | presently I'm trying to install asterisk on a media computer...can I do that with asterisknow? |
| 20:50.34 | Polysics | depends on your general skill level |
| 20:50.47 | Polysics | i like installing * by hand but i'm a control freak :-) |
| 20:51.05 | williammanda | I wish I were that good to do that |
| 20:51.55 | williammanda | I dont' see that much documentation for ubuntu |
| 20:51.56 | Polysics | it is not difficult |
| 20:52.08 | Polysics | ifif you know linux a little |
| 20:52.25 | williammanda | I know alittle is be dangerous |
| 20:53.10 | williammanda | maybe you could possible look at a script written by someone....I've been screwing around with it since Friday night |
| 20:53.17 | *** join/#asterisk [8none1] (n=[8none1]@c-68-52-23-77.hsd1.tn.comcast.net) |
| 20:55.10 | moy | zeejee: almost done, hold on a bit more |
| 20:55.16 | zeejee | moy : ok |
| 20:55.20 | retentiveboy | [TK]D-Fender: which configs would be useful for looking at why a station hint isn't showing InUse? |
| 20:55.45 | *** join/#asterisk engrxyz (n=engrxyz@92.237.248.183) |
| 20:55.46 | Polysics | about the skil-based routing, anyone doing that? |
| 20:56.01 | williammanda | http://dudanogueira.com.br/ubuntu/AsteriskOnUbuntuCurrent.sh |
| 20:58.20 | moy | zeejee: that's some funny audio pattern you have there |
| 20:58.32 | zeejee | moy : like ? |
| 20:58.34 | moy | that is incoming call right? |
| 20:58.39 | zeejee | moy : yes |
| 20:58.48 | zeejee | moy : yes, incoming call |
| 20:58.48 | moy | and you stopped dahdi_monitor at which point? |
| 20:58.59 | zeejee | moy : after call disconnected |
| 20:59.21 | moy | it's definitely not DTMF R2 nor MFC R2 |
| 20:59.27 | zeejee | moy : but i have given same filename for all channels |
| 20:59.31 | moy | I bet coppice should know |
| 20:59.50 | moy | but he is not online now |
| 21:00.09 | moy | may be is MFC R1 or some old crap like that |
| 21:00.13 | zeejee | moy : giving same filename for all channels didnt create these pattern ? |
| 21:00.28 | moy | zeejee: mmm may be |
| 21:00.32 | zeejee | moy : telco told me that they are using digital signalling 2 |
| 21:00.41 | moy | zeejee: give a different name |
| 21:00.47 | moy | to discard problems |
| 21:00.49 | moy | and try again |
| 21:00.53 | zeejee | moy : ok |
| 21:01.00 | moy | want to see if we get the same |
| 21:03.20 | *** join/#asterisk fofware (n=fofware@host188.190-136-191.telecom.net.ar) |
| 21:04.18 | zeejee | moy : http://208.43.250.118/received-audio28.raw |
| 21:05.49 | [TK]D-Fender | retentiveboy: sip.conf obviously |
| 21:06.50 | fofware | hello guys, ChannelZ, manxpower |
| 21:07.23 | ChannelZ | AHOY! |
| 21:07.32 | ChannelZ | yarrrr |
| 21:07.33 | fofware | I did solve the problem to send notification mails of VM |
| 21:07.39 | moy | zeejee: this makes more sense |
| 21:08.02 | zeejee | moy : different kind of signalling ? |
| 21:08.14 | fofware | I put in my script a socket connection to Asterisk so I can get all data of client |
| 21:09.20 | moy | well, since openr2 DTMF and MF detectors did not detect the tone there, I assume is some different tone not included, may be bell R1 tones |
| 21:09.23 | fofware | realy thanks ChannelZ and manxpower for your suppot last night |
| 21:09.59 | moy | zeejee: have to go for now, will come back in about 3 hours, and then I may play with your file and spandsp bell R1 detectors to see if that tone you sent me is from bell R2 tone set |
| 21:10.12 | moy | I meant, bell R1 |
| 21:10.28 | zeejee | moy : ok, shld i come here after 3 hours ? |
| 21:10.47 | *** join/#asterisk errotan (n=errotan@62.201.123.79) |
| 21:11.31 | ChannelZ | fofware: glad you figured it out |
| 21:11.35 | moy | if you want to know, yes, cannot guarantee you that I'll come back exactly in 3 hours though, depends on my wife wishes :) .... zeejee what card are you using btw? |
| 21:11.41 | *** join/#asterisk mythicalbox (n=mythical@rrcs-64-183-110-250.west.biz.rr.com) |
| 21:12.12 | zeejee | moy : i m using digium 4 port |
| 21:12.47 | zeejee | moy : can i know the outcome tomorrow ? |
| 21:12.58 | moy | yup, you can ask me tomorrow |
| 21:13.10 | zeejee | moy : ok, i hope it will be done |
| 21:15.19 | zeejee | moy : switch is siemens ewsd btw |
| 21:15.32 | zeejee | from where the pri coming |
| 21:15.32 | retentiveboy | [TK]D-Fender: sip.conf, users.conf, and extensions.conf at http://pastebin.com/m3d046eaf |
| 21:16.56 | [TK]D-Fender | retentiveboy: type=peer , limitonpeers=yes , call-limit=99 |
| 21:19.08 | retentiveboy | [TK]D-Fender: forgive me for being thick but are these additions to users.conf or sip.conf? |
| 21:19.21 | retentiveboy | the limit settings I mean |
| 21:19.25 | [TK]D-Fender | retentiveboy: each peer |
| 21:21.32 | retentiveboy | [TK]D-Fender: that worked. now I gotta go digging in the code to see why. thanks. |
| 21:25.20 | retentiveboy | [TK]D-Fender: FYI "limitonpeers" isn't in the 1.6.1.6 code AFAIK. only found it in the ChangeLog. seems to work without it though. |
| 21:31.35 | *** join/#asterisk Badrobot- (n=Badrobot@cpe-76-173-229-89.socal.res.rr.com) |
| 21:33.18 | *** join/#asterisk sudhir492 (n=sudhir@27.sub-75-199-52.myvzw.com) |
| 21:33.31 | sudhir492 | Hi All |
| 21:33.55 | sudhir492 | I need help in changing Cisco7940 from Skinny to SIP. |
| 21:34.29 | sudhir492 | There are 4 phones in all. 2 are urgent, and 2 can wait |
| 21:36.03 | Chainsaw | sudhir492: There was a good webpage about the process, let me find it. |
| 21:36.25 | Chainsaw | sudhir492: http://www.cisco.com/en/US/products/hw/phones/ps379/products_tech_note09186a0080094584.shtml |
| 21:37.06 | Chainsaw | sudhir492: You will need a TFTP server for this. Also the firmware to do this will need to be downloaded from Cisco. |
| 21:38.40 | sudhir492 | I have dhcpserver and tftpserver both setup. But I do not have the correct firmware or config file |
| 21:40.51 | Chainsaw | sudhir492: Without the firmware file mentioned there, it's not going to work. |
| 21:40.51 | *** join/#asterisk CrashHD (n=CrashHD@65.74.156.109) |
| 21:41.09 | CrashHD | Hello everyone |
| 21:41.27 | *** join/#asterisk guax (n=guaxinim@unaffiliated/guaxinim) |
| 21:41.28 | ChannelZ | Greetings earthling |
| 21:41.35 | CrashHD | :) |
| 21:42.42 | CrashHD | I was curious as to the stability/feature set of the newest branch of asterisk vs what we are currently running currently (1.4.22) |
| 21:43.18 | CrashHD | 1.6.1 I believe? |
| 21:43.22 | ChannelZ | shrugs - I'm still on 1.4 |
| 21:43.32 | Chainsaw | is on 1.6.1.6 |
| 21:43.40 | CrashHD | hows it going Chainsaw? |
| 21:43.44 | CrashHD | is there a big difference? |
| 21:43.45 | Chainsaw | The amount of commands that got deprecated was a bit of a headache when I upgraded from 1.2 |
| 21:43.53 | ChannelZ | I might be blind but I haven't found a list of what is new in 1.6 or why I should/would switch, short of pouring through changelogs |
| 21:44.09 | CrashHD | ChannelZ, I agree. I'm in the same situation |
| 21:44.11 | Chainsaw | (And finding bugs in a dial plan is *HELL* on earth, it must be said) |
| 21:44.28 | CrashHD | ya no kidding |
| 21:44.31 | Chainsaw | Big new features: SIP over TCP, video stream support. |
| 21:44.37 | ChannelZ | I love * but the docs are fairly poor/scattered |
| 21:44.42 | CrashHD | video works on 1.4 |
| 21:44.43 | Chainsaw | As well as SpanDSP integration. |
| 21:44.51 | ChannelZ | It's definately not a package for the faint of heart |
| 21:44.52 | CrashHD | ahh spandsp might be cool |
| 21:45.21 | Chainsaw | *nod* It already has the necessary #ifdef soup for the all the API changes. |
| 21:45.53 | Chainsaw | (SpanDSP is a disaster area when it comes to that, every release seems to incompatibly change something) |
| 21:46.02 | CrashHD | eek |
| 21:46.31 | CrashHD | what about things like better jitter controls |
| 21:46.39 | CrashHD | core calls items |
| 21:46.50 | CrashHD | call parking, etc |
| 21:47.02 | KavanS | does anyone here use video? |
| 21:47.25 | CrashHD | We do on a couple of grandstreams 2000's |
| 21:47.31 | Chainsaw | CrashHD: Well, the good thing about Asterisk integrating SpanDSP support is that it shifts the API burden to them. |
| 21:47.44 | CrashHD | good and bad I guess though |
| 21:47.47 | Chainsaw | CrashHD: Instead of to some external app_rtxfax developer that only intermittently releases. |
| 21:47.56 | CrashHD | lack of control can sometimes be a pain |
| 21:48.39 | Chainsaw | Call parking still seems to work the same. |
| 21:49.00 | Chainsaw | The 'sip show registry' command no longer shows registrations to asterisk, only registrations from asterisk. |
| 21:49.06 | dustybin | Chainsaw, CrashHD, Chainsaw <--- interesting choice of nicks |
| 21:49.14 | *** join/#asterisk sircolin (n=sircolin@my83-216-68-241.mynow.co.uk) |
| 21:49.14 | CrashHD | dustybin: ha |
| 21:49.17 | Chainsaw | dustybin: Oh it could be a lot worse. |
| 21:49.24 | dustybin | aye indeed |
| 21:49.31 | Chainsaw | dustybin: I used to be in one channel with a Chaos and a Chakotay. |
| 21:49.34 | CrashHD | Chainsaw: why wouldn't it show regs to *? |
| 21:49.46 | Chainsaw | dustybin: Not to mention, I'm still in a channel with 3 people that are called Tony IRL. |
| 21:50.03 | Chainsaw | CrashHD: I have no idea. It broke the web interface someone else wrote. |
| 21:50.10 | CrashHD | so much fun |
| 21:50.12 | Chainsaw | CrashHD: Just like the deletion of the dial command on the CLI. |
| 21:50.21 | CrashHD | thats part of why I'm scared of pushing to newer releases |
| 21:50.29 | CrashHD | all the cleanup and bug catches that start getting reported to me |
| 21:50.43 | Chainsaw | CrashHD: A lot of bugs only get fixed in 1.6 |
| 21:50.49 | Chainsaw | CrashHD: So sooner or later, you're going to have to deal with it. |
| 21:50.54 | CrashHD | ya true |
| 21:51.00 | Chainsaw | CrashHD: What I did was deploy Asterisk 1.6.1.6 on a separate box with my dial plan. |
| 21:51.17 | Chainsaw | CrashHD: And I connected a single phone. Trying to exercise all the dialplan paths whilst running asterisk -dddddddddddddddddvvvvvvvvvvvvvvvvvv |
| 21:51.32 | CrashHD | yang, lots of manually regression testing |
| 21:51.34 | CrashHD | *ya |
| 21:51.36 | Chainsaw | (Because without that it'll never admit to you where the bug actually *is* in the dial plan, just that it exists) |
| 21:52.01 | CrashHD | any improvements on jitter control? |
| 21:52.02 | Chainsaw | It'll also generate such a humongous amount of output on a live system that you'd never want to run it there. |
| 21:52.08 | Chainsaw | No idea about that, sorry. |
| 21:52.22 | CrashHD | I need a reboot, brb. |
| 21:52.25 | Chainsaw | k |
| 21:53.49 | ChannelZ | I really should at least go mow the back yard |
| 21:55.14 | Chainsaw | Yeah, now's the time ChannelZ. Nothing interesting going on. |
| 21:55.23 | ChannelZ | heh |
| 21:57.03 | *** join/#asterisk grEvenX (n=even@cB78D5AC1.dhcp.bluecom.no) |
| 21:59.09 | *** join/#asterisk CrashHD (n=CrashHD@65.74.156.109) |
| 21:59.23 | Chainsaw | wb CrashHD |
| 21:59.28 | CrashHD | ty |
| 22:00.06 | CrashHD | any good cti interfaces out for * these days? |
| 22:03.19 | *** join/#asterisk ecret (n=ecret@CPE001e9002348e-CM001225d8ab30.cpe.net.cable.rogers.com) |
| 22:11.32 | williammanda | sorry to ask this again...can I install asterisknow and still have the functionality of my present system? |
| 22:12.54 | [TK]D-Fender | williammanda: AsteriskNOW is a complete distro and wipes your HD |
| 22:13.31 | *** join/#asterisk CrashHD (n=CrashHD@65.74.156.109) |
| 22:14.09 | [TK]D-Fender | williammanda: Asterisk is an application you can install on an existing system. |
| 22:15.17 | *** join/#asterisk robl^ (n=robl^@c-98-197-98-39.hsd1.tx.comcast.net) |
| 22:15.47 | williammanda | ok... thats not what I want...ty |
| 22:16.13 | williammanda | asterisknow ....that is... |
| 22:16.54 | williammanda | there doesn't seem to be a newbee friendly install for ubuntu |
| 22:17.18 | [TK]D-Fender | williammanda: Installation is hardly an issue. Learning it is another matter |
| 22:17.19 | drmessano | No, Ubuntu sucks, and is generally hated by everyone except soccer moms and 14 yr olds |
| 22:17.28 | CrashHD | hahah |
| 22:17.35 | ChannelZ | oh here we go |
| 22:17.38 | williammanda | oh man |
| 22:17.42 | CrashHD | runs away |
| 22:17.44 | [TK]D-Fender | drmessano: Are the 14yr olds necessarily soccer-kids? |
| 22:18.04 | drmessano | [TK]D-Fender: No, usually high school dropout potheads |
| 22:18.08 | ChannelZ | Can't you get the same web UI thingy from AsteriskNOW |
| 22:18.17 | drmessano | Just like Debian users |
| 22:18.19 | williammanda | well if you are switching over from windows...its not bad |
| 22:18.26 | [TK]D-Fender | ChannelZ: More stuff to install |
| 22:18.32 | CrashHD | ubuntu can help with a transition from win to linux |
| 22:18.36 | ChannelZ | yeah but that's easy |
| 22:18.56 | ChannelZ | I mean you can install the Asterisk package in ubuntu but its pretty dead simple to compile yourself (configure, make, make install...) |
| 22:19.06 | [TK]D-Fender | williammanda: And there are only several dozen easily Google-able guides on how to do this. So get cracking |
| 22:19.13 | drmessano | Ubuntu makes it easy to install a 4 yr old asterisk package |
| 22:19.23 | drmessano | Just look for the AOL icon |
| 22:19.28 | ChannelZ | yeah which is why I say just build the source |
| 22:19.34 | williammanda | easy for you guys |
| 22:19.42 | drmessano | "Youve got Asterisk" |
| 22:19.48 | drmessano | grumbles |
| 22:20.20 | drmessano | Oh, my Kubuntu ISO is finished.. See you losers later |
| 22:20.26 | ChannelZ | Well williammanda I'm not trying to be rude but configuring/running Asterisk is no cakewalk either so if installing is too difficult you're going to be in for it |
| 22:20.51 | drmessano | ~book |
| 22:20.51 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 22:20.54 | [TK]D-Fender | williammanda: time to get reading |
| 22:20.58 | drmessano | Read it all, first |
| 22:21.17 | drmessano | If you dont understand it, you're halfway there.. if it all makes sense, youre a fucking liar and we dont want you here |
| 22:21.32 | drmessano | First rule of Asterisk: Asterisk doesnt make sense.. |
| 22:21.52 | ChannelZ | I should be called *#@? |
| 22:23.02 | williammanda | thanks guys....I'm willing to do the work....but thanks again for the slap down....you guys make every windows user want to switch |
| 22:23.06 | drmessano | Asterisk should be called "badass telephony engine thats 90% COBOL-like ad nauseum programming and 10% plain english" |
| 22:23.22 | [TK]D-Fender | williammanda: Umm.. this isn't ##linux or #ubuntu |
| 22:23.24 | CrashHD | not sure it's that bad |
| 22:23.30 | CrashHD | but ya * takes some getting use to |
| 22:23.34 | drmessano | We're not here to convince you to switch from Windows |
| 22:23.38 | ChannelZ | It's not a slapdown, just a reality check, that all of this requires a bit of adventure on the part of the user |
| 22:23.45 | ChannelZ | But it's free, so you get what you pay for |
| 22:23.52 | drmessano | This is #asterisk, not ##Skype and not ##newbuntu |
| 22:23.54 | [TK]D-Fender | williammanda: We don't care about destop users switching an many here don't use *NIX as a destop |
| 22:24.18 | CrashHD | is away -( working )- at 03:24p -( P:On / L:On )- |
| 22:24.23 | drmessano | I use Windows XP... No one had to beg me to switch from Windows to anything |
| 22:24.28 | ChannelZ | CrashHD: :) I like * actually, I haven't cursed it much |
| 22:24.37 | CrashHD | I played with freeswitch a bit |
| 22:25.12 | CrashHD | found that asterisk is just too simple in comparison |
| 22:46.08 | *** join/#asterisk [8none1] (n=[8none1]@c-68-52-23-77.hsd1.tn.comcast.net) |
| 22:47.23 | *** join/#asterisk sahX (n=Bawbatos@65-119-47-100.dia.static.qwest.net) |
| 23:05.22 | box2 | i use linux on my desktop because i hate getting anything accomplished in under 3 hours |
| 23:05.32 | *** join/#asterisk sahX (n=Bawbatos@65-119-47-100.dia.static.qwest.net) |
| 23:05.35 | box2 | makes me feel like i havn't worked hard enough for what i get |
| 23:05.41 | drmessano | HAHAHAHAH |
| 23:05.43 | drmessano | No shit |
| 23:05.59 | drmessano | Its not free wifi if your meal is still warm when you finally get connected |
| 23:06.17 | box2 | heh |
| 23:08.34 | drmessano | Linux only needs to be rebooted when you make a change significant enough to require a reboot.. or when you're doing something minor that requires that signficant change as a dependency of the minor change youre attempting to make. |
| 23:08.38 | drmessano | OOPS, wait, what? |
| 23:11.40 | nextime | drmessano : if you really want, you can also change kernel without reboot at all with linux :) |
| 23:12.19 | nextime | Takapa: linux doesn't need to be rebooted, it is just more confortable to do a reboot if you are changing the kernel |
| 23:12.24 | nextime | ops |
| 23:12.30 | nextime | s/takapa/drmessano |
| 23:13.29 | box2 | how do you unload your kernel and load a new kernel without a reboot |
| 23:13.36 | nextime | box2 : kexec |
| 23:15.11 | *** join/#asterisk sudhir492 (n=sudhir@222.sub-75-199-61.myvzw.com) |
| 23:16.15 | sudhir492 | Chainsaw, are you still there |
| 23:16.45 | Chainsaw | Of course. |
| 23:18.34 | box2 | kexec looks dangerous and unstable as hell |
| 23:19.38 | nextime | box2 : as i was saying, it is more confortable to make a regular reboot if you are changing the kernel. Anyway, in some environments, for example where you need no downtime at all, kexec is a way to minimize it even in case of a kernel change |
| 23:19.58 | [TK]D-Fender | n3gI presume this is so I can use the mask you are uncertain how to apply... that so? |
| 23:20.09 | nextime | and it come to be usefull to boot systems on usb if you have a machine that doesn't support booting from usb, and so on |
| 23:26.42 | *** join/#asterisk Alfio (n=Alfio@190.94.44.214) |
| 23:30.37 | sudhir492 | Can anyone help me with converting Cisco 7940, from Skinny to SIP. I am offering dinner+beer |
| 23:31.01 | *** join/#asterisk Defraz (n=Defraz@c72co-edge-router.fuzecore.com) |
| 23:48.04 | Sandheaver | sudhir492: just upload the sip firmware to the phone via tftp or whatever cisco phones use |
| 23:50.39 | *** join/#asterisk ltd_wk (i=z@patwk.transact.net.au) |
| 23:51.02 | *** join/#asterisk markt-9 (n=markt9@d27-96-218-35.nap.wideopenwest.com) |
| 23:54.26 | *** join/#asterisk obnauticus (n=obnautic@about/windows/regular/obnauticus) |