| 00:05.49 | *** part/#asterisk beek (~klinebl@pdpc/supporter/bronze/beek) |
| 00:14.10 | *** join/#asterisk serafie (~erin@76.73.167.231) |
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| 01:41.12 | *** join/#asterisk muzak (~muzak@184.71.181.166) |
| 01:42.14 | muzak | Hi All, I am trying to run asterisk along aside a program called dash. I am very new to asterisk. Is it relatively easy to program a shortcut say #111 and it plays a beep, as if recording a vm? |
| 01:42.55 | WIMPy | You just want to play a beep? |
| 01:43.22 | muzak | yes whenever a someone dials a certain command |
| 01:43.41 | muzak | we are making a small call centre and we want to make the beep not the person receiving the call. |
| 01:43.43 | WIMPy | Nothing easier than that. |
| 01:44.16 | muzak | oh? |
| 01:44.47 | WIMPy | has a feeling that this is not the full story at all. |
| 01:45.27 | muzak | well the full story is a lot longer :P |
| 01:45.54 | muzak | however at this point i just want to make it easy and i want to learn how i make a certain command like #111 or w.e play a beep tone |
| 01:46.43 | WIMPy | exten => #111,1,Playback(beep) |
| 01:46.50 | WIMPy | That's the whole story. |
| 01:47.28 | muzak | so just put that in extensions.conf? |
| 01:47.35 | WIMPy | yes |
| 01:48.02 | WIMPy | In to the context that's configured for your phone, that is. |
| 01:49.05 | muzak | pardon? |
| 01:49.57 | WIMPy | extensions.conf contains (potentially) lots of contexts. You need to put it in to the right one. |
| 01:50.09 | muzak | what if my extensions.conf is empty |
| 01:50.15 | muzak | i take it i am looking at the wrong one? |
| 01:50.33 | WIMPy | Then you have to fill it with something. |
| 01:50.50 | muzak | you mean aside from the line you gave me? |
| 01:51.07 | WIMPy | Yes. At least the name of a context. |
| 01:51.19 | WIMPy | For a start, try the |
| 01:51.24 | WIMPy | ~book |
| 01:51.24 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 01:52.19 | muzak | Oh i def. plan on doing a lot more reading. the person who normally does the asterisk stuff for us isnt available right now |
| 01:53.57 | NovceGuru | WIMPy is for hire |
| 01:54.22 | *** join/#asterisk droemel (~droemel@p4FCACA3A.dip.t-dialin.net) |
| 01:55.43 | WIMPy | Yes. NovceGuru is my manager. |
| 01:59.51 | SeRi | WIMPy: any experience with the Obi110 device? |
| 02:00.07 | WIMPy | Is the deal still not completed? I think I need a new manager. |
| 02:00.29 | SeRi | lol |
| 02:00.33 | WIMPy | nope |
| 02:00.35 | *** join/#asterisk gg608f (~Adium@c-67-180-129-182.hsd1.ca.comcast.net) |
| 02:00.39 | SeRi | ok. Thanks. |
| 02:02.27 | NovceGuru | lawyers are drawing up a contract hang tight |
| 02:21.53 | *** join/#asterisk ledoktre (~chatzilla@omni-01-161.cable.netins.net) |
| 02:23.19 | ledoktre | question fellas - just had someone mention they are trying to enter a "#" sign while on a call, and it initiates an asterisk transfer. They want to be able to transfer an outbound call to other users, park, or conf call, so I can't just disable the transfer ability in the dial plan. I am thinking of changing the # sign to ## or something, how do you guys handle a scenario like this? |
| 02:30.22 | WIMPy | Get decent phones? |
| 02:36.39 | ledoktre | Ive got Polycom IP 550 phones |
| 02:37.08 | WIMPy | They should allow you to do real transfers. |
| 02:37.10 | *** join/#asterisk bchia (~Adium@nat/digium/x-kavyvukdahenthdz) |
| 02:37.13 | ledoktre | I'm using the # feature for transferring to parked call extension, transferring to conf call macro, etc |
| 02:37.47 | WIMPy | Disable that and use the phones transfer button. |
| 02:38.47 | ledoktre | You got any link on using that button with *? Ive not had much luck finding documentation on Polycom configurations (XML) |
| 02:39.07 | WIMPy | Don't they have a manual? |
| 02:40.00 | ledoktre | Of course. But they don't contain any information on programming their XML configuration files |
| 02:40.50 | WIMPy | There must be one. |
| 02:41.07 | *** join/#asterisk tzafrir_laptop (~tzafrir@212.179.75.202) |
| 02:41.21 | WIMPy | But I guess someone here will tell you in a while. |
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| 03:07.19 | slav3_kitten | <PROTECTED> |
| 03:08.12 | ectospasm | i prefer debian, but their packages tend to be out of date |
| 03:08.34 | ectospasm | slav3_kitten: ^ |
| 03:09.09 | slav3_kitten | my systems are all debian, they may be out of date but are always stable an well tested |
| 03:10.36 | ectospasm | then you get to decide: stability or features (noting that they are not mutually exclusive) |
| 03:11.45 | ectospasm | difference between 1.8LTS, 10, or 11LTS(new) |
| 03:12.21 | slav3_kitten | i'll go debian again, just wasn't sure if there was a better or worse dstro |
| 03:12.54 | ectospasm | neutral, go with what you know |
| 03:14.14 | slav3_kitten | unix? |
| 03:17.49 | slav3_kitten | this is all in vain if this hdd is dead an an no-go :| |
| 03:20.52 | *** join/#asterisk gg608f (~Adium@c-67-180-129-182.hsd1.ca.comcast.net) |
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| 03:22.21 | slav3_kitten | ectospasm, gimme a high five! |
| 03:22.44 | slav3_kitten | 800mhz, 512 ram, 20gb hdd. i have an asterisk server :D |
| 03:29.53 | *** join/#asterisk moy (~moy@UNVLON55-1176057127.sdsl.bell.ca) |
| 03:32.41 | *** join/#asterisk cmendes0101 (~nn@96.247.7.202) |
| 03:34.32 | cmendes0101 | Out of nowhere one of my asterisk boxes has started to receive SIP 480 Temp Unavailable. This happens like about 20% of outbound calls on 2 different providers. That reply is coming from the provider and should be the asterisk box right? |
| 03:35.47 | slav3_kitten | cmendes0101, did this start happening in relation to the super storm? |
| 03:37.17 | cmendes0101 | It barely happened today about 8 hours ago |
| 03:37.23 | cmendes0101 | started* |
| 03:38.05 | slav3_kitten | i've got no clue honestly. i'm just starting my first install |
| 03:39.56 | slav3_kitten | i know there were some major voip providers who got sunk in the storm |
| 03:46.17 | cmendes0101 | yah noticed a little connectivity problem for a little bit but that was displaying as a different issue. I believe the provider is located in midwest and the server is also midwest so hoping the storm is unrelated |
| 04:03.52 | *** join/#asterisk bchia (~Adium@user-24-236-95-16.knology.net) |
| 04:03.52 | ChannelZ | Damnation! |
| 04:04.36 | WIMPy | Is the call number as used in log messages accessible form the dialplan? |
| 04:10.49 | ChannelZ | Anyone farting around with Asterisk 11? |
| 04:11.36 | WIMPy | hands ChannelZ a can of beans. |
| 04:15.23 | ChannelZ | My console color is no more. |
| 04:15.50 | WIMPy | Mine hasn't changed. |
| 04:16.24 | ChannelZ | It doesn't work under screen anymore here. |
| 04:17.54 | WIMPy | Same colour with screen as well. |
| 04:18.31 | ChannelZ | hmmm let me try something |
| 04:20.55 | ChannelZ | ok.. if the main process is started under screen it doesn't like the terminal type. I sort of don't remember it being like this, but ok... |
| 04:28.38 | ChannelZ | Hummm.. but my 1.8 box at work doesn't behave like that. |
| 04:31.47 | kikohnl | I've been running 11 since Monday, on a Dell Mini 10, one Cisco 7941, works great! |
| 04:32.19 | WIMPy | Mine has just lost sip support. |
| 04:32.26 | ChannelZ | Nice. |
| 04:32.52 | WIMPy | Or rtp to be exact. The use of ffmpeg didn;t seem like a good idea from the start. |
| 04:33.11 | kikohnl | currently with broadvoice, but looking to move |
| 04:33.22 | ChannelZ | If I Noop(${ENV(TERM)}) I get "screen" on both boxes yet no color on the * 11 box. |
| 04:33.43 | *** join/#asterisk serafie (~erin@76.73.167.231) |
| 04:34.48 | kikohnl | Color works for me with * 11, Ubuntu 12.0 |
| 04:35.11 | ChannelZ | under what terminal? |
| 04:35.14 | kikohnl | not as much color as 10.9 but there is still some |
| 04:35.16 | kikohnl | zsh |
| 04:35.29 | ChannelZ | echo $TERM |
| 04:35.31 | *** join/#asterisk samuel_sappa (~samuel_sa@202.137.7.252) |
| 04:35.57 | kikohnl | xterm |
| 04:36.07 | kikohnl | ssh'd from a Mac |
| 04:36.46 | ChannelZ | actually I guess that's not what I really wanted to know. |
| 04:37.11 | ChannelZ | it's the terminal type the original asterisk process was run under |
| 04:38.48 | ChannelZ | I see some notes about color mods in 11, I wonder if something broke for certain terminal types that used to work. |
| 04:39.44 | kikohnl | I ssh in, my shell is zsh, sudo /etc/init.d/asterisk start; sudo asterisk -r |
| 04:40.47 | ChannelZ | ok so it was started from that same shell whose type was xterm. So that one works. |
| 04:40.56 | kikohnl | yep |
| 04:41.24 | WIMPy | has got rtp working again. |
| 04:41.30 | ChannelZ | Without screen, my normal type SSH'd in with putty is "linux" and that works |
| 04:41.30 | kikohnl | but there is much less coloring than there use to be, I only see it for certain apps, like Dial |
| 04:41.55 | ChannelZ | but "screen" doesn't for some reason. |
| 04:42.04 | WIMPy | Set your TERM in scree to linux. |
| 04:43.17 | ChannelZ | yeah or I can start it outside of screen, but I'm more curious why it doesn't work anymore. |
| 04:43.34 | ChannelZ | I'm basically doing the exact same thing on my work box and it's fine (running 1.8) |
| 05:15.35 | *** join/#asterisk ectospasm (~ectospasm@unaffiliated/ectospasm) |
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| 05:39.04 | *** mode/#asterisk [+o Qwell] by ChanServ |
| 05:59.07 | ChannelZ | Grrph. |
| 06:09.05 | slav3_kitten | awesome debian... awesome, your net installer is made of fail sauce |
| 06:11.00 | *** join/#asterisk LiuYan (~LiuYan@211.154.128.171) |
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| 06:15.02 | slav3_kitten | should just say 'is your comp old and made with an intel nic, if so e100 firmware is non-free ha ha ha' |
| 06:15.04 | slav3_kitten | instead it bitches to feed it non-free firmware and has no wpa2 support |
| 06:25.07 | slav3_kitten | huzzah flash drive located, drivers inbound |
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| 06:59.22 | v0lZy | lo |
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| 07:22.51 | bulkorok | hi |
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| 07:46.17 | slav3_kitten | so how does caller ID work with sip DIDs |
| 07:57.21 | *** join/#asterisk evilman_home (kvirc@2.94.90.227) |
| 07:59.18 | ChannelZ | I guess the answer is "it does" |
| 08:01.09 | slav3_kitten | well that's not helpful |
| 08:01.27 | slav3_kitten | nor was google but i've been drinking and it's very late so i think i am the problem there |
| 08:02.36 | slav3_kitten | offers a hard cider to ChannelZ |
| 08:03.21 | kaldemar | slav3_kitten: there's a header called From in SIP. |
| 08:03.36 | slav3_kitten | nods |
| 08:03.51 | slav3_kitten | makes sense, want a cider an smoke? |
| 08:06.00 | *** join/#asterisk nantou (~phonetic@gateway/tor-sasl/martinphone) |
| 08:06.08 | kaldemar | thanks but no thanks. |
| 08:07.05 | slav3_kitten | i'm going to quit asking stupid questions and wanderto sleep |
| 08:07.31 | *** join/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 08:07.37 | slav3_kitten | thanks for induldeing my insanity |
| 08:08.09 | *** part/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
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| 08:24.21 | hebber | I'm using asterisk to record prompts - they get long silence in front and trailing. Does anyone have some references on how to edit the files? I tried to use Audacity, but ended up with background noise. |
| 08:24.26 | *** join/#asterisk coreyf1513 (~cfarrell@ws2.cfware.com) |
| 08:24.52 | ChannelZ | depends on what format you are recording them in |
| 08:25.12 | hebber | Using SIP alaw, 8khz |
| 08:25.14 | ChannelZ | It's usually easier to just record them outside of Asterisk entirely |
| 08:26.09 | hebber | Would it be possible to crop the files without recompressing them? |
| 08:26.22 | hebber | or resample |
| 08:26.32 | *** join/#asterisk Rokfan (~Rokfan@D522448D.static.ziggozakelijk.nl) |
| 08:26.55 | ChannelZ | yeah.. ffmpeg and sox probably can |
| 08:27.24 | hebber | ok, thanks will check that - thanks ChannelZ |
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| 08:29.03 | ChannelZ | ffmpeg -s 00:00:01.500 -i whatever.ulaw -t 00:00:05 new-whatever.ulaw |
| 08:29.47 | ChannelZ | ideally would skip over the first 1.5 sec and stop 5 seconds thereafter |
| 08:30.32 | hebber | You make it easy :) thanks again |
| 08:30.59 | kaldemar | hebber: sox even has an effect to remove silence in the beginning, end or middle of a sample. |
| 08:31.23 | kaldemar | hebber: see "silence" under "Supported Effects" on the sox man page. |
| 08:32.01 | ChannelZ | (and sorry I meant -ss not -s) |
| 08:32.26 | hebber | kaldemar and ChannelZ - thanks for great tips |
| 08:34.23 | *** join/#asterisk latv (~latv@109.238.244.73) |
| 08:34.59 | ChannelZ | mind you I didn't actually test this :) Since ulaw is headerless ffmpeg might need to be told a few more specific things about the file, like sample rate and bits |
| 08:35.35 | hebber | np :) I will learn by trying |
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| 08:42.33 | ChannelZ | FWIW I just imported a ulaw file into Audacity... |
| 08:43.23 | hebber | and could you crop the file and resample the file back with out a lot of noise? |
| 08:44.51 | ChannelZ | well it sounds fine, let me try exporting it now |
| 08:44.59 | hebber | I used Wav Windows and 8 bit unsigned |
| 08:45.35 | ChannelZ | that's not the same.. although if you put it back into asterisk as .wav it should work (I think, or maybe the wav format handler only accepts 16-bit.. I can't remember) |
| 08:45.38 | hebber | 16 bit signed wasn't compatible with Asterisk |
| 08:46.02 | ChannelZ | It still has to be mono, 8000hz |
| 08:48.05 | hebber | Yes, I'm fairly sure I had it in mono 8khz, and with 16bit Asterisk wouldn't play it, but played it with 8bit unsigned - but with background noise. Even the Asterisk recording source file is really good quality |
| 08:48.52 | ChannelZ | well I just re-exported as ulaw and its fine |
| 08:49.12 | hebber | 8khz, mono with 16bit? |
| 08:49.29 | ChannelZ | Go to Export, select 'other uncompressed files' for the type, then click the options button and select RAW (headerless) and u-law encoding |
| 08:49.39 | ChannelZ | No, original 8 bit of the ulaw file |
| 08:49.48 | hebber | aha thanks - thumbs up |
| 08:52.06 | ChannelZ | sure, have fun. Bed time for me |
| 08:55.59 | *** join/#asterisk Russ (~russ@pool-74-100-57-85.lsanca.fios.verizon.net) |
| 08:56.15 | Russ | the copper wires that come to my house, how do I find out who services them? |
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| 09:17.54 | x2s | Hi. This might be an unusual question. But why did asterisk change it's name from 1.8 directly to 10.x? |
| 09:20.11 | kaldemar | x2s: http://blogs.digium.com/2011/07/21/the-evolution-of-asterisk-or-how-we-arrived-at-asterisk-10/ |
| 09:21.16 | *** join/#asterisk danfromuk (~IceChat77@2.26.254.105) |
| 09:22.57 | x2s | kaldemar: thank you! |
| 09:35.12 | x2s | And I like that decision :) |
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| 09:56.45 | ghost75 | is ext s only valid in macros? |
| 10:09.09 | bitwize | Hi! Im having som eproblems installing prerequired packages on CentOS6, yum cannot find 6 of the prerequired packages (libsqlite3x-devel, gmime22-devel etc.) Anybody know a good repository containing these packages? |
| 10:12.55 | bitwize | The packages are not included in the rpmforge repo either... |
| 10:16.06 | kaldemar | ghost75: no. |
| 10:16.14 | coreyf1513 | bitwize: that rpm isn't for centos 6, the correct packages are sqlite-devel and gmime-devel.. sqlite is part of centos, gmime-devel is in epel (probably in others too) |
| 10:19.36 | bitwize | coreyf1513: OK, thanks! I suppose the same issue relates to the other 4 missing packages. I will search for the alternativ package names. |
| 10:21.38 | coreyf1513 | bitwize: to use that rpm you will have to fix the spec in the source rpm and rebuild.. |
| 10:23.49 | bitwize | coreyf1513: Thanks! |
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| 10:48.11 | angryuser | MArc from sangoma is here ? |
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| 11:00.08 | ghost75 | is there are way to see the output of a failed system cmd ? |
| 11:00.36 | ghost75 | verbose 3 is showing only the cmd to be executed |
| 11:02.11 | *** join/#asterisk kannan (~chatzilla@122.164.40.112) |
| 11:04.13 | kaldemar | ghost75: how are you executing the system command? |
| 11:04.22 | *** join/#asterisk kchehab (~ck@77.42.241.66) |
| 11:04.24 | kchehab | hi |
| 11:05.06 | ghost75 | from dialplan like system(bla) |
| 11:05.54 | kchehab | i have a problem that some of the cdrs in a2billing table have a negative value which means that its not limiting the max call duration related to the customer amount ,and each use can make 1 concrent call only |
| 11:07.26 | *** join/#asterisk sekil (~sekil@78.24.104.73) |
| 11:07.35 | ghost75 | i typed in the command manualy in bash with user asterisk and it worked |
| 11:08.19 | ghost75 | but not over dialplan |
| 11:09.15 | ghost75 | [Nov 2 12:02:49] -- Executing [h@spamtrap:1] System("SIP/10-00000023", "echo "Anruf von 10 mit Spamscore wurde geblockt weil in Blacklist oder Score 7-9." | mutt -s "[PBX]: Geblockter Anruf von 10" -a /usr/share/asterisk/sounds/recordings/honeypotfrom10on2012-11-02-in.wav -- trechber@intern.net") in new stack |
| 11:09.59 | kchehab | is there any bug in a2billing you know |
| 11:10.19 | kaldemar | ghost75: you won't get command output to dialplan with System directly. if you want that, use func SHELL instead. |
| 11:11.36 | kaldemar | ghost75: first guess is that there is a path issue, use full path for mutt. |
| 11:12.01 | ghost75 | ok will try |
| 11:13.29 | kchehab | ? |
| 11:19.07 | ghost75 | hmm now i know the output is: could not send message |
| 11:21.53 | ghost75 | maybe because of " |
| 11:25.51 | *** join/#asterisk fisted_ (~fisted@unaffiliated/fisted) |
| 11:26.17 | coreyf1513 | ghost75: are you running selinux or something else that could cause mutt to fail under asterisk's security context? selinux can prevent connections to port 25 |
| 11:26.44 | ghost75 | no, i use also other system commands without problem |
| 11:27.48 | ghost75 | even this fails: exten => h,1,Set(spamresult=${SHELL(echo "test" | /usr/bin/mutt -s "test" -- trechber@intern.net)}) |
| 11:28.49 | ghost75 | on bash this works: sudo -u asterisk echo "test" | /usr/bin/mutt -s "test" -- trechber@intern.net |
| 11:30.05 | kaldemar | try it in CLI. !<command> |
| 11:30.25 | ghost75 | this works in cli ? |
| 11:30.30 | kaldemar | something working in bash by using sudo as another user is no proof of anything. |
| 11:30.53 | kaldemar | yes, you can run shell commands in asterisk's CLI if you prepend them with !. |
| 11:31.11 | ghost75 | thats cool feature |
| 11:31.39 | ghost75 | that worked also |
| 11:32.02 | Rokfan | ghost75: I would recommend to use an agi script for mailing |
| 11:32.05 | kaldemar | what version of astrisk are you using? |
| 11:32.10 | ghost75 | 1.6.2 |
| 11:32.32 | ghost75 | i think those " are maybe ... |
| 11:32.38 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 11:32.39 | kaldemar | escape the quotes with \ |
| 11:33.21 | kaldemar | Rokfan: what is the benefit for using AGI in executing a shell command? |
| 11:35.33 | Rokfan | We had a lot of stability issues when we used shell commands directly in the dialplan |
| 11:36.08 | Rokfan | but maybe that had to do with the Asterisk version |
| 11:36.49 | ghost75 | which commands you were using |
| 11:38.07 | ghost75 | i had problems that the cpu went 100% and i had to reboot |
| 11:39.28 | Rokfan | I would have to look that up. I only took over development of our application from a colleague recently |
| 11:40.21 | ghost75 | exten => h,1,Set(spamresult=${SHELL(echo \"test\" | /usr/bin/mutt -s \"test\" -- trechber@intern.net)}) |
| 11:40.25 | ghost75 | it doesnt like me |
| 11:41.25 | *** join/#asterisk serafie (~erin@76.73.167.231) |
| 11:43.38 | *** join/#asterisk Praise (~Fat@unaffiliated/praise) |
| 11:48.23 | *** join/#asterisk Praise (~Fat@unaffiliated/praise) |
| 11:51.28 | *** join/#asterisk jblack (~jblack@100.sub-70-192-130.myvzw.com) |
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| 11:54.20 | *** join/#asterisk blee (~blee@72.188.117.219) |
| 11:55.07 | kaldemar | ghost75: maybe your asterisk user does not have the permissions to send mail. |
| 11:55.21 | ghost75 | sudo could send |
| 11:55.26 | ghost75 | i try now agi |
| 11:55.39 | kaldemar | your sudo was for the echo, not for mutt. |
| 11:55.49 | *** join/#asterisk blee (~blee@72.188.117.219) |
| 11:56.12 | ghost75 | will not do for both commands? |
| 11:56.24 | Rokfan | but it did work on the cli? |
| 11:56.30 | ghost75 | yes |
| 11:57.03 | kaldemar | after pipe comes another command. |
| 11:59.54 | *** join/#asterisk Praise (~Fat@unaffiliated/praise) |
| 12:01.32 | Rokfan | I don't think that's the problem, but you try echo "test" | sudo -u asterisk /usr/bin/mutt ... |
| 12:04.01 | ghost75 | server*CLI> ! echo "test" | sudo -u asterisk /usr/bin/mutt -s "test" -- trechber@intern.net |
| 12:04.05 | ghost75 | works as well |
| 12:05.06 | *** join/#asterisk Praise- (~Fat@unaffiliated/praise) |
| 12:05.25 | *** join/#asterisk amessina (~amessina@2001:470:c1dc:7779:d6be:d9ff:fe8d:7c1e) |
| 12:05.32 | kaldemar | is the output still "could not send message"? |
| 12:05.58 | ghost75 | yes |
| 12:06.02 | ghost75 | also agi doesnt work |
| 12:06.33 | coreyf1513 | ghost75: what is the output of ${SHELL(echo $USER)} ? |
| 12:08.11 | coreyf1513 | ghost75: sorry ${SHELL(whoami)} |
| 12:08.33 | ghost75 | server*CLI> ! echo $USER |
| 12:08.35 | ghost75 | root |
| 12:08.36 | ghost75 | huh? |
| 12:08.44 | kaldemar | coreyf1513: +1 |
| 12:09.41 | coreyf1513 | hmmm... on my server echo $USER outputted 'root', but whoami said asterisk.. maybe this is an environmental variable issue for mutt? |
| 12:10.33 | ghost75 | could be |
| 12:10.55 | ghost75 | mutt can send attachments |
| 12:11.48 | *** join/#asterisk Praise (~Fat@unaffiliated/praise) |
| 12:13.05 | ghost75 | with mail command i cannot send attachment |
| 12:13.21 | Rokfan | I have used a php agi script to send mail. Php should also be able to send attachments.. |
| 12:13.43 | ghost75 | i have agi bash script and failed also |
| 12:14.23 | Rokfan | mmm it really doesn't like you |
| 12:14.39 | kchehab | Guys why i can find negative values in the cutomer accounts in mya2billing |
| 12:14.53 | ghost75 | i wonder what they use to send voicemails |
| 12:14.56 | kchehab | and each customer can limited to one concurent call |
| 12:18.14 | ghost75 | at least he could http://www.pbxinaflash.com/community/index.php?threads/how-to-mail-attachments-with-asterisk.3715/ |
| 12:19.54 | Rokfan | Does the mutt log tell you anything? |
| 12:20.17 | ghost75 | hmm have to find |
| 12:21.40 | ghost75 | maybe it needs a homedir? |
| 12:21.51 | ghost75 | If you have mutt built with debug, the -d switch will enable logging to |
| 12:21.51 | ghost75 | ~/.muttdebug0 |
| 12:22.05 | ghost75 | asterisk user has no homedir |
| 12:22.48 | *** join/#asterisk danfromuk (~IceChat77@2.26.254.105) |
| 12:24.45 | ghost75 | :/var/lib/asterisk:/bin/false is set |
| 12:26.13 | *** join/#asterisk Mango45 (~Mango45@d209-89-214-26.abhsia.telus.net) |
| 12:26.51 | coreyf1513 | weird, I just tried ${SHELL(readlink -f ~)}, which i expected to see /var/lib/asterisk, but it said /root |
| 12:28.14 | Mango45 | I have a peer that sends a corrupted invite after exactly 30 minutes, and causes the call to drop because my Asterisk thinks audio is at a different ip. I don't have control over that peer so I can't change its behaviour. Is there any way for me to mitigate that? |
| 12:28.47 | Mango45 | Can I say to Asterisk, "After the call is already established, if the peer says audio is at some random IP, it is lying." |
| 12:31.11 | *** join/#asterisk elico (~Thunderbi@109.64.221.76) |
| 12:34.19 | coreyf1513 | ghost75: try echo "test" | env -i /usr/bin/mutt ... |
| 12:39.40 | coreyf1513 | ghost75: sorry it would actually be: ${SHELL(env -i bash -ic 'echo "test"|/usr/bin/mutt ...')} |
| 12:40.09 | ghost75 | the fuuuuuuuuuuuuuuuuuuuuuuuuuuuuuuuuuuuuuuuuuuuuuuuu |
| 12:40.13 | ghost75 | it worked! |
| 12:40.16 | coreyf1513 | that will atleast ensure that mutt goes to the right place for ~ |
| 12:40.35 | ghost75 | env -i /usr/bin/mutt was enough |
| 12:40.44 | ghost75 | thanks a bunch |
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| 13:37.11 | Katty | sways |
| 13:37.29 | Mango45 | oscillates |
| 13:46.32 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 13:46.33 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 13:46.58 | *** join/#asterisk x1user (~User@212.36.13.6) |
| 13:47.31 | x1user | What happened to Asterisk 11, no sip in the CLI ? |
| 13:48.29 | file | it's there, provided chan_sip.so loaded as expected |
| 13:57.18 | *** join/#asterisk felipealmeida (~user@mvx-187-16-79-187.mundivox.com) |
| 14:01.17 | *** join/#asterisk [TK]D-Fender (~TK]D-Fend@216.191.106.162) |
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| 14:08.37 | leifmadsen | file: +1 |
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| 14:15.04 | *** join/#asterisk Micc (~Mic@c-24-19-33-189.hsd1.wa.comcast.net) |
| 14:15.26 | Micc | I don't much like the new asterisk.org site. |
| 14:15.59 | ghost75 | is it possible to use variable from external file? |
| 14:16.22 | Kobaz | okay so |
| 14:16.30 | Kobaz | i have this polycom phone that keeps requesting dhcp |
| 14:16.34 | Kobaz | and wont take it |
| 14:16.43 | Kobaz | andit's just sitting there waiting for network to initalize |
| 14:18.25 | file | Micc, I think you are the first person I've heard since launched who says they don't much like it |
| 14:18.51 | file | Micc, any particular reason? |
| 14:20.34 | Micc | @file, I can't find what I'm looking for easily |
| 14:20.44 | file | what are you looking for? |
| 14:20.48 | Micc | mainly the releases. |
| 14:21.19 | Micc | used to be really easy to see what the latest releases were and get to the notes quickly. |
| 14:22.25 | jmetro | http://www.asterisk.org/downloads ? |
| 14:22.25 | Micc | the only reason I went to the site was to see what the latest versions were and if there are any new security bulletins |
| 14:22.45 | jmetro | links on the left - security advisories, source code, news. |
| 14:23.20 | Micc | I guess the news is what I was looking for then. |
| 14:23.49 | *** join/#asterisk j4m3s_ (~j4m3s_@pdpc/supporter/active/j4m3s) |
| 14:24.00 | Micc | Now I'm happy that I've found the news page. |
| 14:24.18 | *** join/#asterisk rgsteele (~rgsteele@apo.aweber.com) |
| 14:24.50 | Micc | I might like the news and security together, but this is fine. |
| 14:25.55 | Micc | now that I know, I think it is growing on me. |
| 14:26.26 | mjordan | :-) |
| 14:27.13 | file | jmetro, thanks for showing Micc the way ^_^ |
| 14:27.34 | file | to the LIGHT! er, downloads and such |
| 14:29.29 | *** join/#asterisk bsdfreak (~ninja@ninja.paranode.net) |
| 14:29.56 | [TK]D-Fender | ghost75, variables don't come from files. They exist live in the dialplan. |
| 14:30.25 | Kobaz | [TK]D-Fender: oh... yay |
| 14:30.26 | Kobaz | you're here |
| 14:30.53 | Kobaz | did you see my polycom question above? I know you play with polycom a lot |
| 14:31.03 | ghost75 | i thought like to manipulate dialplan from external |
| 14:31.17 | *** join/#asterisk Defraz (~Defraz@mail.pocatellochildren.com) |
| 14:31.23 | [TK]D-Fender | ghost75, doesn't sound like you're talking about "variables" here... |
| 14:31.29 | bsdfreak | Hi. I'm running Asterisk 1.4.44 on FreeBSD 9.0p3 (x86). I am getting persistent Operation not permitted and Failed to lock path errors when trying to leave voicemails, but the /var/spool/asterisk/voicemail/default directory and subdirectories are owned by the user asterisk, which also runs the asterisk process. There are no lock files (hidden or otherwise) in these directories. I've been fighting this issue for a bit and would love some fresh insi |
| 14:31.34 | [TK]D-Fender | ghost75, More like "constants" |
| 14:31.58 | Katty | Hi. my name is Katty. and i've been running asterisk for 7 years. |
| 14:32.09 | [TK]D-Fender | Kobaz, not sure on this... checked if it's VLAN'd away from your DHCP scope? |
| 14:32.10 | Katty | i decided to check myself into therapy, so that's why i'm here |
| 14:34.26 | Kobaz | [TK]D-Fender: the weird thing is the vlan is good. And i see the dhcp request/response in the server log |
| 14:34.35 | Kobaz | [TK]D-Fender: the dhcp server is running on exactly one vlan, vlan 50 |
| 14:34.57 | [TK]D-Fender | Kobaz, I don't have any direct experience with those kinds of issues beyond a concept of what I suggested |
| 14:37.03 | Kobaz | yeah |
| 14:37.19 | Kobaz | i've seen this problem before. and power cycling the phone has fixed it |
| 14:37.23 | Kobaz | but not in this case |
| 14:37.44 | bsdfreak | I'm seeing this error immediately proceeding the others: utils.c: write() returned error: Broken pipe |
| 14:37.46 | [TK]D-Fender | I'd suggest up/down-grading the bootrom hoping that helps zero it out |
| 14:40.03 | x2s | When setting up a fax receiving service, what would you recommend: Using asterisk 1.8 with spandsp (from debian wheezy) or getting packages for asterisk 11? |
| 14:40.05 | n3hxs | Katty, no know therapists lurking here. |
| 14:40.37 | bsdfreak | i am a therapist. |
| 14:40.59 | n3hxs | Well, they you go! ask and one pops up. |
| 14:42.24 | Katty | dear therapist, i need hugs |
| 14:43.46 | Mango45 | offers Katty to borrow his one-year-old. |
| 14:44.30 | Katty | screams, runs the other way |
| 14:44.38 | Mango45 | :( |
| 14:44.43 | Mango45 | He's adorable. And he gives hugs. |
| 14:44.45 | *** join/#asterisk tyrrexrrg (~roger@200.75.45.194) |
| 14:44.48 | Katty | he is also a child. |
| 14:44.53 | Katty | miniture hooman. |
| 14:45.04 | Katty | children are scary! |
| 14:45.16 | Mango45 | I'm teaching him to write dial plans. |
| 14:45.38 | Katty | :< |
| 14:45.43 | Katty | you should be teaching him how to catch a ball. |
| 14:45.53 | Katty | he should have cool girlfriends. |
| 14:45.58 | Mango45 | I'm saving that for after he can walk. |
| 14:46.21 | Mango45 | <-- can write dial plans, can't really throw a ball, has a cool girlfriend. :D |
| 14:46.31 | Katty | yeah, that's totally not legit. |
| 14:46.37 | bsdfreak | yeah, no |
| 14:46.38 | Katty | cause my boy can't catch a ball either. |
| 14:46.40 | bsdfreak | we dont do that here |
| 14:46.45 | Katty | but he's totally fab at vmware |
| 14:47.08 | Mango45 | :) |
| 14:47.18 | Katty | ok, go back to teaching dial plan. |
| 14:47.25 | Mango45 | hehe |
| 14:47.34 | Mango45 | We covered Javascript when he was 2 days old. |
| 14:47.37 | Katty | does he have a fisher price phone?! |
| 14:47.47 | Katty | something noisy, with light up buttons |
| 14:47.47 | Mango45 | OMG |
| 14:47.53 | Mango45 | hits himself in the head |
| 14:48.02 | Katty | plz escort self to toys r us immediately |
| 14:48.05 | Mango45 | He has two phones. One is Fisher-Price and the other is VTech. |
| 14:48.08 | Mango45 | BUT |
| 14:48.21 | Mango45 | The Fisher-Price phone is a rotary phone, and THE ROTARY DIAL GOES THE WRONG WAY. |
| 14:48.29 | Katty | fail. |
| 14:48.36 | Mango45 | Oh! It gets better. |
| 14:48.38 | Katty | i prefer rotary engines to rotary phones. |
| 14:48.44 | Katty | flutterflutterflutter. |
| 14:48.44 | Mango45 | The VTech phone plays RANDOM DTMF. It doesn't play the proper DTMF for the buttons you press. |
| 14:48.55 | *** join/#asterisk mintos (mvaliyav@nat/redhat/x-lqgdwglldzglpgfp) |
| 14:49.04 | Mango45 | WTF. |
| 14:49.07 | Katty | that engineer does not deserve a hug |
| 14:50.17 | [TK]D-Fender | x2s, * 11 isn't even a week old yet. You may want to hold back just a little for the first kinks to come out... |
| 14:50.21 | bsdfreak | hoo |
| 14:50.22 | bsdfreak | hah |
| 14:50.32 | Katty | [TK]D-Fender: that's what your best friend said? |
| 14:52.15 | [TK]D-Fender | Katty, Shhh this is a family show ;) |
| 14:52.38 | Katty | not when i'm here!!! |
| 14:52.57 | mjordan | x2s: the other question you need to ask yourself is whether or not you need T.38 gateway. Asterisk 11 has that functionality; 1.8 does not. |
| 14:53.08 | [TK]D-Fender | Indeed... |
| 14:53.45 | [TK]D-Fender | Actually, * 10+ has it... but 10 is on a shorter track to EOL |
| 14:53.49 | *** join/#asterisk blitzrage (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 14:53.50 | *** mode/#asterisk [+o blitzrage] by ChanServ |
| 14:53.56 | [TK]D-Fender | How long does it have? A year? |
| 14:53.59 | x2s | I just want to receive and send them. Not relaying them :) |
| 14:54.04 | Katty | blitzrage: i like your crib, yo |
| 14:54.10 | blitzrage | indeed :) |
| 14:54.13 | [TK]D-Fender | x2s, it does matter... |
| 14:54.46 | [TK]D-Fender | x2s, If you need T.38 on one leg and need to hit DAHDI on the other then you'll need this... |
| 14:56.05 | x2s | It's far easier. (or worse, decide yourself ;) I want to use asterisk behind a Lancom 1722, which receives the calls/faxes over ISDN and forwareds them over SIP |
| 14:56.14 | bsdfreak | phew |
| 14:56.26 | x2s | I'm not sure which codec is used here |
| 14:56.34 | mjordan | [TK]D-Fender: Asterisk 10 is Security Fix only in December |
| 14:57.01 | jeffspeff | on asterisk 1.8 and 10.x i used nat=yes for all my peers and it allowed them to register multiple different devices/extensions/sip users behind the same nat'd external ip. with asterisk ver 11 net=yes is deprecated, so what nat setting should be used to replace it? i've tried comedia and force_rport. comedia is only letting 1 device register behind a single external IP and force_rport allows multiple devices to register simultaneously but there is o |
| 14:57.02 | jeffspeff | nly 1-way audio ( I called the user, I can hear the user, the user cannot hear me). |
| 14:59.05 | [TK]D-Fender | x2s, What are you going to send faxes from? |
| 14:59.29 | x2s | [TK]D-Fender: directly with asterisk. |
| 14:59.34 | *** join/#asterisk felipealmeida (~user@mvx-187-16-79-187.mundivox.com) |
| 14:59.37 | [TK]D-Fender | x2s, because if the Lancom supports T.38 on its side then * can receive those to a file in 1.8- just fine |
| 14:59.44 | x2s | I'm planing to build an email-to-fax interface |
| 14:59.53 | [TK]D-Fender | x2s, And if you want to send, IIRC it should do that as well... |
| 15:00.14 | [TK]D-Fender | x2s, Then 1.8 should be good for now... |
| 15:01.34 | x2s | I'm not sure how the lancom handles this kind of connections. Maybe it just forwars the audio stream from its ISDN interface. The documentation isn't very good on such details. |
| 15:03.44 | x2s | oh, just found something. Looks like you can configure fax forwarding over SIP as t.38. |
| 15:05.45 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 15:05.45 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 15:07.11 | mjordan | jeffspeff: yes is the same thing as specifying force_rport,comedia |
| 15:07.18 | blitzrage | x2s: yes, might be worth to read the fax chapter in ATDG |
| 15:09.01 | jeffspeff | mjordan, you need to specify both options? |
| 15:09.13 | mjordan | jeffspeff: yes |
| 15:09.25 | jeffspeff | mjordan, ok, thanks i'll give that a shot |
| 15:09.33 | mjordan | the reason why the setting is deprecated (which only means its not recommended) is that the nat setting allows options to be combined |
| 15:09.50 | mjordan | setting nat=yes doesn't really tell you what you're doing however, which is to set force_rport and comedia together |
| 15:10.34 | mjordan | the warning message that gets spit out onto the CLI should tell you what to use in your config instead of 'nat=yes' |
| 15:10.36 | [TK]D-Fender | jeffspeff, Most importantly you also need to disable reinvites (directmedia=no) |
| 15:10.38 | [TK]D-Fender | ^ |
| 15:11.07 | jeffspeff | thanks [TK]D-Fender. i learned the hard way a while back about directmedia. :) |
| 15:14.46 | x2s | blitzrage: I'm already doing that :) It says it's for 1.8, so it should be right for me. |
| 15:15.05 | x2s | [TK]D-Fender: Thanks for your help. |
| 15:16.28 | jeffspeff | mjordan, thanks for you help as well. |
| 15:20.55 | mjordan | jeffspeff: np |
| 15:20.55 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 15:20.56 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 15:30.27 | slav3_kitten | the book could be written a bit better... |
| 15:33.07 | *** join/#asterisk TomCat2 (~TomCat@static-72-77-202-50.tampfl.fios.verizon.net) |
| 15:34.58 | TomCat2 | Hey guys - I'm trying to find information on how to add some tones to the beginning of any incoming calls before they will dial my SIP device in order to defeat some robo dialers - can't recall what those tones are called. Can someone remind me what they are called, or better yet, if you have a readily available example for what I'm trying to do, would gladly review that as well. |
| 15:35.28 | [TK]D-Fender | "core show application playtones" |
| 15:35.38 | [TK]D-Fender | SIT is what you're looking for |
| 15:35.46 | [TK]D-Fender | Which ZapATeller does as well |
| 15:36.48 | TomCat2 | SIT - that's right... Looking at ZapATeller - thanks |
| 15:37.41 | TomCat2 | campaign offices got my number and won't let it go anymore - and it's getting kind of disruptive to work and I don't expect it to let up until after Nov 6 |
| 15:37.57 | TomCat2 | as I'm sure many are experiencing the same issue |
| 15:38.10 | TomCat2 | at least here in the states |
| 15:38.29 | *** join/#asterisk gg608f (~Adium@c-67-180-129-182.hsd1.ca.comcast.net) |
| 15:38.30 | [TK]D-Fender | No, I'm sure plenty of people are being hammered.... |
| 15:39.25 | *** join/#asterisk vinhdizzo (~vinh@dhcp-v025-135.mobile.uci.edu) |
| 15:40.03 | slav3_kitten | yay for elections :D |
| 15:41.01 | TomCat2 | it's just a sad state of affairs when contributing to a particular campaign, it results in getting called 2-3 times a day from that very same campaign office asking you to vote for their candidate |
| 15:41.25 | Mango45 | I worked on a campaign once |
| 15:41.41 | Mango45 | My boss was like, "WTF, I have received five calls from us so far." |
| 15:41.41 | TomCat2 | reminds me of WalMart playing 'Shop at WalMart for low prices' commercials on their screen inside of the store |
| 15:41.59 | Mango45 | Their phone lists are that mangled. |
| 15:42.32 | TomCat2 | I'm just hoping their robo dialer will trigger on the SIT and remove me |
| 15:42.32 | Mango45 | "Hello! I am calling to remind you that November 27 is voting day..." "It's me idiot, stfu." |
| 15:42.45 | TomCat2 | LOL |
| 15:43.02 | ghost75 | what the hell is that: GotoIf($[foo${DB(CF/${EXTEN})} != foo]?normal:forward) |
| 15:44.10 | slav3_kitten | god i need to stop smoking |
| 15:46.06 | [TK]D-Fender | ghost75, a check to see if it's blank |
| 15:46.20 | ghost75 | if the db is empty then foo = foo ? |
| 15:46.57 | [TK]D-Fender | ghost75, beacuse an empt var will be LITERALLY blank on the left side there which will cause an error. "foo" is just literal text so that when the var ends up being blank there is something there |
| 15:47.16 | Qwell | quoting both sides would be better |
| 15:47.29 | [TK]D-Fender | well ... more sane anyway... |
| 15:47.32 | [TK]D-Fender | "looking" |
| 15:47.33 | Qwell | really, just use ISNULL |
| 15:47.53 | Qwell | GotoIf($[${ISNULL(${DB(CF/${EXTEN})})}]?normal:forward) |
| 15:48.06 | Qwell | err, maybe swap the args there. whatever |
| 15:48.08 | WIMPy | Or DB_EXISTS? |
| 15:48.14 | Qwell | WIMPy: you win |
| 15:48.20 | [TK]D-Fender | Qwell, no need for $[] |
| 15:48.21 | Qwell | actually, no, it could exist but be empty |
| 15:48.51 | WIMPy | Yes. Depends on how you clear it. |
| 15:49.39 | ghost75 | Set(DB(CF/${CALLERID(num)})=${EXTEN}) <- is that valid? |
| 15:51.02 | [TK]D-Fender | yes |
| 15:52.13 | *** join/#asterisk TimeRider (~steve@188-220-34-144.dsl.cnl.uk.net) |
| 15:53.57 | slav3_kitten | [TK]D-Fender, i'm still reading the book (it's a big book) but i have a caller id question. can you set it to send outbound caller names as anything you want dependent on the extension dialing? |
| 15:54.32 | [TK]D-Fender | slav3_kitten, To whatever you want period... assuming the provider SUPPORTS passing it on |
| 15:55.08 | slav3_kitten | awesome |
| 15:55.26 | slav3_kitten | i have no idea why it's awesome, but it's awesome from a home use fun network standpoint |
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| 15:57.36 | TomCat2 | I'm running 10.6.1 built from sources - getting "No application 'Zapateller' warnings - is Zapateller not in 10.x, or was it renamed? |
| 15:58.47 | [TK]D-Fender | may no longer exist.... check for a direct playtones equivalent |
| 16:00.42 | TomCat2 | ah nvm - just been awhile since I've been in here - forgot to check for noloads |
| 16:03.11 | TomCat2 | but I guess that's not necessarily a bad thing that it just keeps on running w/o any problems and I don't have to be in there all the time tinkering with it... got the SIT working - thanks for keeping me on track |
| 16:10.10 | TomCat2 | while I'm here already - here's one I haven't figured out yet. currently I have DIAL(SIP/1234&SIP12345678910@flowroute-trunk) in my dialplan to simul-ring my ATA and my cell phone when a call comes in. What's the suggested way to do something similar, except that dialing the cell number is delayed by a set number of seconds so I don't answer my office phone and have to have my cell phone |
| 16:10.10 | TomCat2 | already ringing |
| 16:10.50 | TomCat2 | as there seems to be a delay of a few seconds after I answer the office phone (on the ATA) before the cell phone will shut up |
| 16:13.45 | *** join/#asterisk k610 (~Instantbi@cred.epid.ucl.ac.be) |
| 16:17.14 | ghost75 | cant i use * or # char to include at internal extension? |
| 16:26.25 | Qwell | !950/330,!1400/330,!1800/330,0 |
| 16:26.27 | Qwell | looks around |
| 16:27.47 | [TK]D-Fender | <TomCat2> as there seems to be a delay of a few seconds after I answer the office phone (on the ATA) before the cell phone will shut up <- That's your provider. |
| 16:28.05 | [TK]D-Fender | <ghost75> cant i use * or # char to include at internal extension? <- huh? |
| 16:28.26 | ghost75 | like _*95X. |
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| 16:30.51 | fireman_biff | I noticed that dundi/secret only stores the current secret and 1 previous secret. Does that mean I should never set dundi's cache time to more than 2 hours, or will asterisk remember more keys when the cache time is increased? |
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| 16:47.35 | Russ | how can I find out which telco services the copper to my home? |
| 16:48.32 | nightrid3r | check your bill ? |
| 16:48.58 | Russ | I don't have service |
| 16:52.39 | [TK]D-Fender | ghost75, Yes |
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| 16:54.02 | leifmadsen | usually there is a box or something out front for who to call about stuff |
| 16:54.14 | Qwell | leifmadsen: I bet it's 911. |
| 16:54.17 | ghost75 | my phone only dials *95 and skips the appending numbers |
| 16:54.36 | leifmadsen | ghost75: fix dialplan on the phone itself |
| 16:54.51 | leifmadsen | Russ: if in north america, I think you can call 611 or something |
| 16:54.58 | Qwell | Russ: but seriously, if you plug a phone into a jack, any call should hit them |
| 16:55.06 | _Corey_ | I hear 911 likes to get called whenever someone has a question about who their phone company is... or how to use an iPhone, etc. |
| 16:55.24 | Qwell | _Corey_: mcdonalds getting the order wrong |
| 16:55.42 | ghost75 | i think if you call emergency number here you get fine |
| 16:56.21 | _Corey_ | I remember the audio of the calls from the guy with the iPhone was on The Smoking Gun's website last year... the best part was when she said, "yeah, I think one of our offices would be happy to come over now... what's your address" |
| 16:56.32 | _Corey_ | (officers) |
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| 16:57.00 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 16:57.09 | ghost75 | the address was told? |
| 16:57.45 | _Corey_ | lol, that was actually when the guy realized he was in trouble... he said something profane (if I remember correctly) and hung up at that point |
| 16:58.17 | ghost75 | and he got visit from cops |
| 16:58.48 | ghost75 | they should just should such morons |
| 16:58.55 | ghost75 | shoot not should |
| 16:58.57 | Qwell | should them to death. |
| 16:59.44 | _Corey_ | whatever works, really |
| 17:08.01 | Russ | Qwell, leifmadsen, I don't have service |
| 17:08.22 | Russ | I'm in CA and I'd like to get soft-tone enabled for 911 service |
| 17:08.33 | leifmadsen | I have no idea |
| 17:08.38 | leifmadsen | that is highly dependent on area |
| 17:08.49 | leifmadsen | that's the first mention of an area I've seen |
| 17:08.50 | Qwell | it's Verizon. |
| 17:08.50 | Russ | no one seems to want to admit to providing POTS |
| 17:08.59 | Russ | Qwell, I can't find POTS anywhere on their website |
| 17:09.08 | Russ | its all FiOS |
| 17:09.09 | *** join/#asterisk [sr] (~kvirc@pal-213-228-181-48.netvisao.pt) |
| 17:09.11 | [sr] | howdy |
| 17:09.13 | [sr] | hi WIMPy |
| 17:09.23 | [sr] | guys, i have a new card with 12x FXS |
| 17:09.30 | WIMPy | o/~ |
| 17:09.35 | [sr] | but dahdi doesn't recognize it, where can i report this? |
| 17:09.43 | Qwell | http://www22.verizon.com/home/phone/ |
| 17:09.44 | Russ | foresees a very long call with verizon explaining over and over again about copper |
| 17:09.45 | Qwell | next? |
| 17:09.49 | WIMPy | You want to make a functional museum? |
| 17:10.13 | Russ | Qwell, where is the order button? |
| 17:10.42 | Qwell | really? |
| 17:10.57 | Russ | I see one under inside wire maintenance, but I don't want that |
| 17:11.24 | Russ | I see a 'check availability' under verizon triple play, but I don't want that either |
| 17:11.30 | [sr] | WIMPy: any idea? |
| 17:11.35 | Qwell | CALL |
| 17:11.47 | WIMPy | [sr]: Whoever you bought it from? |
| 17:12.02 | Qwell | Or click one of the 8 chat buttons. |
| 17:12.03 | _Corey_ | Russ: You might want that inside wire maintenance... I hear it's a bargain if you put it on every line |
| 17:12.32 | Russ | jebus, they'll love me for my 110 block |
| 17:12.39 | [sr] | WIMPy: openvox |
| 17:13.14 | Qwell | [sr]: Good luck! |
| 17:13.33 | Russ | http://www.flickr.com/photos/russdill/8118450801 |
| 17:15.01 | [sr] | Qwell: :( |
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| 17:30.09 | jeffspeff | Russ, why are you trying to use pots for your 911? We have ours setup via e911 through our ITSP |
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| 17:31.23 | ghost75 | GotoIf($[foo${DB(CF/${EXTEN})} != foo]?normal:forward) <- should this not be without the ! ? |
| 17:32.01 | jeffspeff | ghost75, != means does not equal |
| 17:32.17 | jeffspeff | or think of it like this |
| 17:32.21 | jeffspeff | = is |
| 17:32.24 | jeffspeff | != is not |
| 17:32.32 | ghost75 | when foo=foo then goto forward it means i think |
| 17:33.49 | jeffspeff | no, it means if ${EXTEN} is not foo then forward |
| 17:34.13 | jeffspeff | wait, soryr, you were right |
| 17:34.37 | jeffspeff | i always get it backwards when changing from != and = |
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| 17:35.19 | ghost75 | the argument after ? is the one it goes after match? |
| 17:35.19 | *** join/#asterisk k610 (~Instantbi@host-78-129-3-116.brutele.be) |
| 17:35.39 | jeffspeff | yes |
| 17:35.46 | jeffspeff | so, if ${EXTEN} is not foo then normal if it is foo then forward |
| 17:35.56 | ghost75 | GotoIf($[foo${DB(CF/${EXTEN})} = foo]?normal:forward) |
| 17:36.01 | ghost75 | i think makes more sense ^^ |
| 17:36.08 | jeffspeff | that's how i do mine |
| 17:36.23 | jeffspeff | so, if exten is foo then normal otherwise forward |
| 17:37.22 | ghost75 | when it goes to "normal" |
| 17:37.34 | ghost75 | will it continue with all the lines below? |
| 17:38.18 | jeffspeff | yes, if you use exten=blah,n |
| 17:38.27 | jeffspeff | it depends on your priorities |
| 17:38.40 | ghost75 | exten => _X.,n(normal),Dial(SIP/${EXTEN}) |
| 17:38.45 | ghost75 | exten => _X.,n(forward),NoOp(Anruf fuer ${EXTEN} wird verbunden zu ${DB(CF/${EXTEN})}) |
| 17:39.02 | ghost75 | so if it jumps to normal, will it continue then with forward? |
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| 17:40.00 | jeffspeff | i don't think so |
| 17:40.25 | jeffspeff | if you put more exten=_X.,n,.... underneath then it will continue with those |
| 17:40.49 | jeffspeff | I have hangup() priorites inbetween mine so that it won't continue where i don't want it to |
| 17:41.13 | ghost75 | just want to avoid forward when not called |
| 17:41.40 | jeffspeff | then put a hangup before the forward rule |
| 17:42.01 | ghost75 | i have voicemail after it |
| 17:42.42 | Russ | jeffspeff, I'd rather have the backup |
| 17:42.45 | jeffspeff | we use followme for people that are constantly in and out of the office, and if they're away for extended periods, then they just set the call-forwarding from the phone itself |
| 17:43.05 | Russ | battery backups only last so long |
| 17:43.09 | Russ | and I don't have generators |
| 17:43.26 | jeffspeff | Russ, true, we use our batteries to keep it running until our desiel generator kicks in |
| 17:43.31 | jeffspeff | oh. lol |
| 17:43.46 | jeffspeff | ghost75, are you familiar with followme? |
| 17:44.13 | ghost75 | no but there are no other people to follow :) |
| 17:44.41 | jeffspeff | ?? |
| 17:45.02 | ghost75 | dont really know what followme means |
| 17:45.13 | jmetro | I'm trying to install app_konference on 1.8 and cant seem to get the module to show up in my Module list within the console =( |
| 17:45.14 | jeffspeff | http://www.voip-info.org/wiki/view/Asterisk+cmd+FollowMe |
| 17:47.21 | jeffspeff | ghost75, our setup is basically: call comes in to my did, it dials my sip phone which times out after 20 seconds. it then goes to the followme configured for my extension, i have my cell number listed in my followme profile, so asterisk asks the person calling me to say their name and press pound. asterisk then calls my cell phone and tells me i have an incoming from from (then plays the persons recorded name to me) and i can press 1 to accept the c |
| 17:47.22 | jeffspeff | all on my cell or press 2 to decline it. if i decline the call then it returns back to the next priority in my dialplan, which is to go to my voicemail. |
| 17:48.31 | ghost75 | is this common in US ? |
| 17:48.56 | jeffspeff | ghost75, i'd guess so |
| 17:49.01 | jeffspeff | it depends on how you implement it |
| 17:49.09 | ghost75 | never heared of that |
| 17:49.27 | jeffspeff | you don't have to have the caller record their name, and there's other options to change |
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| 17:51.00 | jeffspeff | it also solves a problem of having split voicemail. if i were to have the dialplan call my cell phone as the next priority in the event i didn't answer my desk phone, and i decide not to answer my cell phone either, then the person calling gets my cell phone voicemail and not my asteisk voicemail |
| 17:51.17 | ghost75 | that would be good thing for anonymous calls |
| 17:51.32 | jeffspeff | we don't accept anonymous calls either. :) |
| 17:52.07 | jeffspeff | my thought on those is, if you don't want me to know who you are when you're calling, then i probably don't want to hear what you have to say anyways |
| 17:52.13 | ghost75 | many people here do it and they think they are smart |
| 17:52.59 | jeffspeff | anonymous calling has been in existence for what? 40 years now? lol |
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| 17:54.28 | ghost75 | followme works only with a second extension involved? |
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| 17:54.38 | jeffspeff | no |
| 17:54.48 | jeffspeff | you can put any number of extensions or phone numbers |
| 17:55.16 | jeffspeff | some of our employees want their to call their desk phone again, plus their cell phone number at the same time |
| 17:55.28 | jeffspeff | did you read the link i gave you? |
| 17:55.36 | ghost75 | right now |
| 17:56.36 | ghost75 | so if i get anonymous i could send them immediately to followme |
| 17:56.46 | jeffspeff | if you wanted |
| 17:56.58 | jeffspeff | it's your dialplan to structure how you please |
| 17:57.20 | ghost75 | but i think people would hang up |
| 17:57.29 | jeffspeff | then they go to voicemail |
| 17:57.59 | jeffspeff | if you want to talk to me bad enough then you'll wait for the system to track me down |
| 17:58.23 | jeffspeff | otherwise you can leave me a message and i may or may not call you back |
| 17:58.26 | jeffspeff | lol |
| 17:58.42 | jeffspeff | then again, i may or may not answer your call at all regardless of where it calls me |
| 17:59.15 | ghost75 | here they use anonymous not to hide who they, they want to hide their number because they maybe are afraid of spam calls if you see their number |
| 17:59.49 | ghost75 | wow my english is bad |
| 17:59.51 | jeffspeff | lol |
| 18:00.07 | jeffspeff | i work with Indians from India, i understood exactly what you meant |
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| 18:09.03 | rrittgarn | looking for a little input on OpenSIPS vs Kamailio, anybody have a preference? @Astricon people kept saying they were interchangeable, and as they look to be forks of OpenSER i could see that... are there any benefits of either? Like does one have a Web UI so somoene other than myself can look at it and know what its doing? |
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| 18:29.44 | ghost75 | (19:00:23) jeffspeff: i work with Indians from India <- do you also talk to them? |
| 18:30.00 | jeffspeff | ghost75, yes |
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| 18:30.07 | ghost75 | this is like lol |
| 18:30.24 | ghost75 | they talk like machine gun and pronounce everything the same |
| 18:30.49 | jeffspeff | yes they do, and they yell over each other |
| 18:31.00 | jeffspeff | who do you work for? |
| 18:31.24 | ghost75 | for me |
| 18:31.30 | jeffspeff | oh |
| 18:31.40 | ghost75 | but i was once in a bigger company and know that too |
| 18:31.59 | jeffspeff | was it the bigger company that you worked with indians? |
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| 18:32.20 | ghost75 | actually there was just one indian but thats enough |
| 18:32.28 | jeffspeff | yes it is |
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| 18:42.38 | jmetro | Jeffspeff: I was once part of a 2 man team of web developers as the lead developer but couldnt choose who I worked with. My partner was a transfer from india who had been taught how to code in india. Apparently they dont need tabbing or proper variable names or commenting code... |
| 18:44.52 | jeffspeff | jmetro, if it's not in an excel spreadsheet they don't comprehend what they're looking at |
| 18:48.56 | ghost75 | Dial(SIP/${DB(CF/${EXTEN})}@arcor_out1099) |
| 18:49.02 | ghost75 | ^^ thats not correct i assume? |
| 18:50.33 | ghost75 | it tries to dial SIP/@arcor_out1099 |
| 18:51.22 | [TK]D-Fender | ghost75, Which imples that it came back blank. |
| 18:51.54 | [TK]D-Fender | And depends on your idea of "not correct" |
| 18:52.08 | [TK]D-Fender | The SYNTAX is right. Maybe the expectation is wrong. |
| 18:52.29 | cklimos | maybe the family is missing from the db |
| 18:52.45 | *** join/#asterisk muzak (~muzak@184.71.181.166) |
| 18:54.37 | cklimos | try to browse your db from the CLI |
| 18:54.37 | [TK]D-Fender | I've got a bet or two of my own... |
| 18:54.54 | muzak | I am running zoiper and asterisk and recording calls. Is it possible to have a command in zoiper or maybe dial a certain number as the caller that plays a certain sound. I don't want to hang up or anything afterwards. I just want to play a sound when I want in the middle of a call. |
| 18:55.21 | [TK]D-Fender | muzak, features.conf applicationmap <- |
| 18:56.24 | leifmadsen | muzak: check the Asterisk Cookbook for an example of that |
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| 18:58.16 | lvlinux | anybody have any experience with Dialogic cards? |
| 18:58.26 | ghost75 | is in variable EXTEN the internal phone ext or the external number ? |
| 18:58.35 | jmetro | aw someones gonna pull the ~poll |
| 18:58.39 | muzak | Thanks guys will try |
| 19:01.53 | ghost75 | i tried the first example here: http://das-asterisk-buch.de/1.6/section-callforwarding.html |
| 19:03.13 | ghost75 | is this even working with external numbers? |
| 19:04.38 | jeffspeff | ghost75, if i were you i would read up on this a little more, rather than trying to implement other examples |
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| 19:06.26 | ghost75 | i have now in astdb: /CF/10 : numbertoforward |
| 19:06.44 | ghost75 | this will work only if somebody dials 10? |
| 19:06.53 | leifmadsen | yes |
| 19:07.00 | leifmadsen | since that's the only entry likely |
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| 19:07.13 | ghost75 | and if i get call from outside ? |
| 19:07.52 | *** join/#asterisk [sr] (~kvirc@pal-213-228-181-48.netvisao.pt) |
| 19:08.06 | [sr] | guys, for software echo it's mg2, and for hardware ? |
| 19:15.18 | cklimos | if you call from outside and you EXTEN is numbertoforward then you will go to forward |
| 19:16.45 | ghost75 | i should enable some NoOp to see what is going on |
| 19:17.04 | cklimos | excellent idea |
| 19:21.30 | [sr] | i can see that it shows hardware assisted HDLC, but i'm not sure if i have to configure anothing else for it |
| 19:21.32 | [sr] | do i? |
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| 19:25.28 | slav3_kitten | ok i think i have all the info i need from the phones, time to start figuring out how to make the cisco configs for it |
| 19:26.43 | ghost75 | this seems to be empty in my db: ${DB(CF/${EXTEN})} |
| 19:27.53 | ghost75 | this is in db: http://pastebin.com/z3vRmyiK |
| 19:28.43 | ghost75 | why its empty |
| 19:29.07 | ghost75 | i am confused |
| 19:31.46 | jeffspeff | ghost75, you have to set the value in the db before you can call it from the db... like i said. you really need to read about why the code does what it does |
| 19:32.26 | ghost75 | its in the db like above |
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| 19:40.11 | jeffspeff | ghost75, seriously you need to read |
| 19:40.23 | jeffspeff | ${EXTEN} isn't a value it's a variable |
| 19:40.31 | jeffspeff | ~book |
| 19:40.31 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 19:40.33 | ghost75 | my head is already full because of reading |
| 19:40.46 | jeffspeff | then you're not reading the right things |
| 19:40.51 | jeffspeff | read that book |
| 19:40.58 | jeffspeff | at least reference the book |
| 19:41.09 | cklimos | what is the value of your EXTEN when you call from outside? |
| 19:41.33 | ghost75 | this is the question |
| 19:41.43 | jeffspeff | cklimos, he's trying to call exten from the asterisk db but never sets the value. he's copy pasting examples expecting them to work |
| 19:41.49 | jeffspeff | but they're incomplete examples |
| 19:41.51 | cklimos | well Noop it |
| 19:41.58 | cklimos | agree with jeff |
| 19:42.07 | jeffspeff | ~book |
| 19:42.08 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
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| 19:45.22 | ghost75 | (20:41:25) cklimos: what is the value of your EXTEN when you call from outside? <- its my external number |
| 19:46.25 | ghost75 | and yes that example doesnt work at all |
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| 19:51.35 | jeffspeff | ghost75, the example works, but it's not complete. please read so that you will understand what it is lacking and why it doesn't work |
| 19:52.22 | ghost75 | oh than i have to read another month or so |
| 19:53.06 | jeffspeff | asterisk isn't something that you can just pick up and run with within a day or so. it takes time and patience. it's a framework to develop a pbx not a pbx server |
| 19:53.29 | cklimos | i've been reading since 2003 |
| 19:53.37 | WIMPy | Or at least some pepole say so. |
| 19:53.42 | ghost75 | thats why i started with asterisk from scratch and not like freepbx |
| 19:53.51 | jeffspeff | clarificiation*** it is a framework to develop a pbx, it is not a software server that you can install and have a pbx up ang going |
| 19:54.30 | WIMPy | I'm not yet convinved you can build a PBX with it, but I'n only 5 years in. |
| 19:54.39 | jeffspeff | lol |
| 19:54.58 | ghost75 | very mighty software it is |
| 19:55.16 | WIMPy | It certainly is. |
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| 19:55.57 | ghost75 | i really admire people who this as fulltime job |
| 19:56.02 | cklimos | itis flexibilty and openness make it very powerful but also easy to break if you don`t know what you`re doing |
| 19:56.21 | WIMPy | ghost75: I don't think you do :-) |
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| 19:57.06 | ghost75 | i did lots of server stuff but asterisk is really challenging me |
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| 20:00.34 | ghost75 | so it needs to be like /CF/myfullexternalnumber : forwardnumber |
| 20:00.37 | ghost75 | ? |
| 20:01.20 | jeffspeff | are you asking about the db entry structure or the gotoif syntax or how you put the external number into the db or what? |
| 20:01.36 | ghost75 | how it should be in the db |
| 20:02.04 | jeffspeff | why are you trying to use the db for this in the first place? |
| 20:02.24 | ghost75 | i want to control it over web :) |
| 20:02.32 | cklimos | your myfullexternalnumber should be the same as the value of your EXTEN and it will dial forwardnumber using local channel |
| 20:02.57 | ghost75 | ah thank now i understood |
| 20:03.10 | jeffspeff | external applications can't connect to the internal asterisk database. you want to user some other database engine (mysql, postgresql, etc.) to store the info |
| 20:03.47 | ghost75 | it works with asterisk -rx "command" |
| 20:03.52 | WIMPy | I do it via AMI and AstDB. |
| 20:03.58 | ghost75 | or ami yes |
| 20:04.10 | cklimos | ghost don`t settle for quick understanding... you will not go far like that |
| 20:04.33 | cklimos | just a friendly advice |
| 20:04.44 | jeffspeff | ghost75, using asterisk -rx "command" for database lookups isn't anything close to effecient |
| 20:05.05 | ghost75 | i already have phonebook in my webbrowser and cdr |
| 20:05.33 | jeffspeff | if you're struggling with dialplan syntax i would stay away from integrating ami into web until you have a better grasp of asterisk |
| 20:06.05 | cklimos | agree |
| 20:07.56 | _Corey_ | cklimos: Long time no see... Hope you're well :-) |
| 20:08.49 | cklimos | Corey yeah true... it has been a long while since I came here. All is well :) Consumed by work like usual |
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| 20:42.52 | _Corey_ | http://www.asterisk.org/downloads is busted |
| 20:43.11 | ghost75 | cklimos: do you use starastapi and php ? |
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| 20:59.03 | pabelanger | mjordan: ^^ |
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| 21:09.04 | Russ | hmmm..was looking into using an old cell phone for backup 911 service, and found celliax, but it seems dead |
| 21:09.14 | Russ | is there some new way of connecting cell phones to asterisk? |
| 21:09.30 | dfgas-cr48 | anyone have a download link for current asterisk source? |
| 21:10.19 | dfgas-cr48 | i use my old android phones |
| 21:10.30 | dfgas-cr48 | website down atm |
| 21:12.15 | coreyf1513 | http://packages.asterisk.org is online |
| 21:13.14 | drmessano | http://downloads.asterisk.org/pub/telephony/asterisk/ is online |
| 21:14.51 | _Corey_ | anyone have the link to the AsteriskNow ISOs? |
| 21:16.10 | Russ | locates chan_mobile |
| 21:19.12 | dfgas-cr48 | thinking i downloaded the wrong one or something |
| 21:19.27 | _Corey_ | guess not... :-) I'll wait for the site to get resolved |
| 21:19.31 | Russ | hmm...chan_mobile needs a mildly smart phone, no alsa support |
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| 23:30.32 | Micc | Is there anyway to get the list of ConfBridge channels from dialplan? |
| 23:30.58 | Micc | in asterisk 11 |
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