| 00:04.48 | *** join/#asterisk ldav15 (~ldav15@166.70.157.161) |
| 00:14.41 | *** join/#asterisk BenC[UK] (~bencummin@cpc1-lock3-2-0-cust367.6-1.cable.virginmedia.com) |
| 00:14.53 | BenC[UK] | hi guys |
| 00:15.09 | BenC[UK] | I want to set it up, so that when someones been in the queue for 5 minutes.. they goto voicemail.. |
| 00:15.34 | BenC[UK] | and also have periodic announcements saying thank you for holding |
| 00:15.48 | BenC[UK] | in freepbx, I've set the max wait time to 5 minutes, set the breakout ivr to do the announcements.. |
| 00:15.57 | BenC[UK] | but a customer never gets to vm |
| 00:19.20 | carrar | delete freebpx and install asterisk then come back |
| 00:19.43 | carrar | or try #freepbx |
| 00:20.19 | BenC[UK] | ha. |
| 00:20.39 | carrar | ~freepbx |
| 00:20.39 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there, or fpbx |
| 00:21.01 | carrar | Have a GREAT day! |
| 00:21.18 | *** join/#asterisk aidinb (~Aidin@unaffiliated/aidinb) |
| 00:21.25 | carrar | and be sure to take along a towel |
| 00:22.52 | ldav15 | ~AsteriskNOW |
| 00:22.52 | infobot | [asterisknow] based on Asterisk, but is difficult to support in #asterisk for a number of reasons. Please seek support in #asterisknow instead. |
| 00:23.09 | *** part/#asterisk ldav15 (~ldav15@166.70.157.161) |
| 00:25.02 | carrar | ~asterisk |
| 00:25.02 | infobot | Asterisk is an open source telephony toolkit, or #asterisk on irc.freenode.net, or http://www.asterisk.org/ |
| 00:25.03 | carrar | [Asterisk] due to all the smart ass's in the channel support on this plateform is difficult at best and everything should be taken with a grain of salt. Please try reading the docs supplies with Asterisk or google first before asking here. |
| 00:25.07 | carrar | heh |
| 00:25.17 | leifmadsen | amen |
| 00:25.37 | leifmadsen | carrar: it's hard to wade through the shit, that's for sure |
| 00:25.43 | carrar | heheh |
| 00:27.03 | p3nguin | smart ass's ? |
| 00:27.48 | p3nguin | Someone is certainly a dipshit. |
| 00:28.16 | carrar | don't claim to be a spelling teacher |
| 00:28.26 | carrar | please correct it since it bothers you |
| 00:28.52 | p3nguin | ass's, possession by ass |
| 00:29.07 | leifmadsen | chuckles silently to himself |
| 00:30.14 | carrar | Sounds gay |
| 00:30.48 | carrar | I like my horrid english |
| 00:30.50 | tzanger | that really should be fixed |
| 00:31.19 | tzanger | just replace the ' with e and it's perfect. |
| 00:31.23 | leifmadsen | ok I'm outta here to do something productive like sit on the couch |
| 00:31.26 | tzanger | and s/plateform/platfomr/ |
| 00:31.33 | tzanger | dammit s/plateform/platform/ |
| 00:31.44 | tzanger | s/supplies/supplied/ |
| 00:31.56 | tzanger | nobody's calling you a spelling teacher, but would it hurt to even proofread? |
| 00:32.02 | carrar | yes |
| 00:32.39 | tzanger | sounds like wilful ignorance to me. perhaps you should reconsider putting up the equivalent of infobot if you're not going to put any effort into it anyway. |
| 00:33.01 | carrar | yeah, I actually don't even care |
| 00:33.10 | tzanger | yes, that is obvious. |
| 00:33.13 | carrar | heh |
| 00:33.56 | *** join/#asterisk Faithful (~Faithful@180.194.3.179) |
| 00:35.21 | *** part/#asterisk BenC[UK] (~bencummin@cpc1-lock3-2-0-cust367.6-1.cable.virginmedia.com) |
| 00:46.31 | *** join/#asterisk nephilinturbo (~victor@187.17.152.182) |
| 00:48.07 | nephilinturbo | Hi Guys . I need find the file in asterisk that says: 'The person you are calling is unavailable, please try again' . But cant find it in anywhere ? Somebody knows the location in server ? |
| 00:51.25 | carrar | nephilinturbo, /var/lib/asterisk/sounds ? |
| 00:51.39 | nephilinturbo | no man . its not there |
| 00:51.43 | carrar | or add on sounds |
| 00:52.13 | nephilinturbo | in asterisk logs not said me what file are Playing , so i cant find |
| 00:52.17 | carrar | ./the-party-you-are-calling.gsm |
| 00:52.26 | p3nguin | /var/lib/asterisk/sounds/vm-nobodyavail.wav and /var/lib/asterisk/sounds/please-try-again-later.wav |
| 00:52.35 | nephilinturbo | I need replace this file for another in Portuguese |
| 00:52.54 | nephilinturbo | Thanks p3nguin . Will checj now |
| 00:53.24 | p3nguin | I'm guessing, since I haven't ever heard the exact wording you've described. |
| 00:54.27 | nephilinturbo | no :( its not it |
| 00:55.08 | nephilinturbo | The sound says: 'The person you are calling is unavailable. Please try again' |
| 00:55.19 | nephilinturbo | Its played when tou call a invalid extension |
| 00:55.31 | nephilinturbo | But i cant locate the file |
| 00:55.34 | p3nguin | maybe if you're using freepbx or something. |
| 00:55.48 | p3nguin | If you call an invalid extension on MY systems, you get a busy tone. |
| 00:56.25 | p3nguin | So use lsof to list files in use by asterisk while the sound file is playing. |
| 00:56.37 | nephilinturbo | Thats really. Im using a billing named MOR |
| 00:56.57 | nephilinturbo | how can i use this command please ? |
| 00:58.29 | p3nguin | something like: lsof -u asterisk|grep -i "gsm\|sln\|ulaw\|wav\|mp3" |
| 00:58.52 | p3nguin | That's how I see what is in use. ymmv |
| 00:58.59 | nephilinturbo | hummm .. like netstat .. ok tanks a lot ..will try it |
| 01:19.33 | *** join/#asterisk Montys (~dmartinez@nat/digium/x-hqwpsqkzyzvebqba) |
| 01:19.52 | Montys | elo :) |
| 01:20.13 | thehar | EHLO |
| 01:30.20 | *** join/#asterisk coppice (~chatzilla@m121-202-55-163.smartone-vodafone.com) |
| 01:33.40 | aut | what's the best place to buy soundpoints?" |
| 01:40.55 | *** join/#asterisk tengulre (~tengulre@125.71.208.16) |
| 01:41.11 | thehar | aut: http://www.8774e4voip.com/ |
| 01:45.06 | tengulre | #bash |
| 01:47.22 | *** join/#asterisk b0gatyr_ (~b0gatyr_@adsl-10-106-138.mia.bellsouth.net) |
| 01:53.58 | aut | does anyone know if the soundpoint 550 includes the power adapter? |
| 01:54.03 | aut | seems like it does, but the 335 doe snot? |
| 01:54.52 | p3nguin | Read the description of what is included. Most places sell phones without power bricks, but you can get the bricks for like an additional $10. |
| 01:55.43 | coppice | poe phones don't usually ship with the brick |
| 01:56.02 | aut | the description on voipsupply doesnt say, that's the problem :) |
| 01:56.15 | aut | the data sheet on polycom's site says that the 335 does not include it, but the 550 does |
| 01:59.44 | talntid | hey |
| 01:59.52 | talntid | voipsupple does NOT include the adapters |
| 01:59.58 | talntid | unless the phone is refurb.. |
| 02:00.06 | talntid | i ordered a bunch of 550's from them |
| 02:00.12 | aut | talntid, hrm... that's hella gay |
| 02:00.35 | aut | i'd demand free adapters if i did that since they are obviously scamming |
| 02:02.16 | talntid | they did send me the adapters at no cost... |
| 02:02.27 | talntid | but it may have been because i spent 14k with them.... |
| 02:06.54 | *** join/#asterisk GameGamer43|Mac (~GameGamer@CPE-65-27-76-78.new.res.rr.com) |
| 02:25.11 | *** join/#asterisk pinoyskull (~pinoyskul@122.55.80.205) |
| 02:29.12 | *** join/#asterisk Defraz (~Defraz@corp.fuzecore.com) |
| 02:36.51 | *** join/#asterisk Docteh (~mage@24.86.236.78) |
| 02:40.55 | Docteh | is tcp supported for sip in 1.6.2? |
| 02:43.52 | p3nguin | I think the more important question is why would you want to use it? |
| 02:45.36 | *** join/#asterisk aidinb (~Aidin@unaffiliated/aidinb) |
| 02:46.00 | Docteh | i'm looking at the posted information for sipdroid |
| 02:46.32 | bougyman | it's valuable in some situations, but not many |
| 02:47.10 | bougyman | first you'd have a different path of sip vs rtp |
| 02:47.22 | bougyman | that's the only use-case i've found for it, anyway. |
| 02:52.44 | pabelanger | Docteh: Yes, 1.6.2 support SIP over TCP |
| 02:54.26 | p3nguin | Why would it be beneficial to have a different path for SIP and RTP? |
| 03:03.31 | *** join/#asterisk tengulre (~tengulre@125.71.208.16) |
| 03:08.27 | ChannelZ | they're already different paths |
| 03:14.51 | p3nguin | I guess that's kind of what different port numbers are for, huh? |
| 03:34.59 | *** join/#asterisk coppice (~chatzilla@m121-202-55-163.smartone-vodafone.com) |
| 03:36.35 | *** join/#asterisk evils_unon (~elvis@horace.dionysian-mind.net) |
| 03:38.25 | *** join/#asterisk niekie (~niek@CAcert/Assurer/niekie) |
| 03:55.18 | *** join/#asterisk Defraz (~Defraz@c72co-edge-router.fuzecore.com) |
| 04:41.47 | *** join/#asterisk stoffell (~kristof@dD576AE12.access.telenet.be) |
| 04:43.29 | *** join/#asterisk bluOxigen (~sfds@unaffiliated/bluOxigen) |
| 05:33.50 | *** join/#asterisk timahvo1 (~rogue@41.72.215.94) |
| 05:50.05 | evils_unon | anyone around in here? |
| 05:51.38 | *** join/#asterisk [T]ank (~ckwall@c-71-195-199-101.hsd1.ut.comcast.net) |
| 05:52.31 | [T]ank | so i am trying to get my iptables set up correctly. But not matter what I do i loose my sip peers. could anyone assist in setting this up correctly. |
| 05:52.33 | [T]ank | ? |
| 05:53.02 | [T]ank | i start out with : iptables -A INPUT -p udp -m udp --dport 5060 -j DROP |
| 05:53.43 | [T]ank | then iptables -A INPUT -p udp -m udp -s 1.2.3.4 --dport 5060 -j ACCEPT |
| 05:54.02 | [T]ank | then all of my peers go unreachable |
| 05:54.16 | [T]ank | what am i missing? |
| 06:00.02 | carrar | iptables -A INPUT -j ACCEPT -i eth0 -p udp -s 0.0.0.0/0 --dport 5060 |
| 06:01.48 | carrar | remove the "iptables -A INPUT -p udp -m udp --dport 5060 -j DROP" |
| 06:02.26 | [T]ank | ok, walk me through the thought process... my intention is to block all ips except what i specifically specify. |
| 06:02.41 | carrar | block at the end |
| 06:03.02 | carrar | assuming you are use specific IP's and not 0.0.0.0/0 |
| 06:03.11 | *** join/#asterisk wierdo (~jimmy@212.25.51.150) |
| 06:03.27 | [T]ank | so replace the 0.0.0.0/0 with the specific ip/32, right? |
| 06:03.46 | carrar | if you only have a 1 SIP peer |
| 06:03.51 | carrar | or a few |
| 06:03.57 | [T]ank | i have a few |
| 06:03.58 | carrar | you need specific each IP |
| 06:04.02 | [T]ank | right |
| 06:04.05 | carrar | specify |
| 06:04.25 | carrar | then after the accept rules |
| 06:04.36 | carrar | block everything |
| 06:04.53 | carrar | like iptables -A INPUT -j LOG -i eth0 -p all |
| 06:05.01 | carrar | (if you want to log denies) |
| 06:05.11 | carrar | iptables -A INPUT -j DROP -i eth0 -p all |
| 06:05.16 | carrar | to drop them |
| 06:05.57 | *** join/#asterisk mog (~mog@c-68-62-169-225.hsd1.al.comcast.net) |
| 06:06.01 | carrar | assuming you have other stuff in there to allow established tcp connections |
| 06:06.02 | [T]ank | i need to still allow access to ssh, that is why i was specifying port 5060 |
| 06:06.10 | [T]ank | nothing else in there yet |
| 06:06.21 | [T]ank | just blocking brute force sip attacks for now |
| 06:06.24 | carrar | so then change dport to 22 and -p to tcp |
| 06:06.40 | carrar | and befor dropping everyhing add |
| 06:06.40 | carrar | iptables -A INPUT -j ACCEPT -i eth0 -p tcp ! --syn |
| 06:07.01 | carrar | to allow esteblished connetions |
| 06:07.41 | carrar | you also need to allow the RTP |
| 06:09.48 | *** join/#asterisk weta (~aross@CPE485b390978ce-CM00222ddf42dd.cpe.net.cable.rogers.com) |
| 06:09.50 | carrar | so something like this |
| 06:09.51 | carrar | http://pastebin.ca/1957390 |
| 06:11.55 | *** join/#asterisk bjornts (~Adium@it010225.klientdrift.uib.no) |
| 06:12.58 | [T]ank | ok, so i have that all in... but when i do a iptables -L it takes forever to pull up the output. any ideas why? |
| 06:13.34 | carrar | did you clear out iptables first? |
| 06:13.40 | carrar | tables -F |
| 06:13.43 | carrar | then run the rules |
| 06:14.15 | carrar | http://pastebin.ca/1957391 |
| 06:14.18 | carrar | updated |
| 06:14.27 | [T]ank | i did |
| 06:14.28 | carrar | assuming eth0 is your nic? |
| 06:14.39 | carrar | use -n -L |
| 06:14.45 | [T]ank | ahhh... now all of the peers just went unavailable |
| 06:14.49 | carrar | it's trying to resole DNS |
| 06:14.50 | [T]ank | havnt changed anything |
| 06:15.20 | [T]ank | oh, wait... disregard that |
| 06:15.24 | [T]ank | they all just came back up |
| 06:15.31 | [T]ank | something must have restarted somewhere. |
| 06:15.56 | carrar | I'm assuming your RTP are also the asterisk defaults /etc/asterisk/rtp.conf |
| 06:16.03 | carrar | 10000-20000 |
| 06:16.11 | [T]ank | yeah |
| 06:16.24 | carrar | I wouldn't leave ssh wide open either |
| 06:16.46 | carrar | if nothing else, just open it up to IP's of your country |
| 06:17.02 | carrar | so all ARIN ip's |
| 06:18.20 | carrar | might need to open HTTP (TCP:80) or tftp (UDP:69) if you do any auto provisioning |
| 06:19.42 | *** join/#asterisk ranjan (7aac71cc@gateway/web/freenode/ip.122.172.113.204) |
| 06:19.51 | ranjan | Hi all |
| 06:20.00 | ranjan | started trying out asterisk |
| 06:20.14 | ranjan | how can i playback a wav file in asterisk? |
| 06:20.26 | carrar | playback or background |
| 06:20.35 | carrar | or copy to your local pc |
| 06:20.43 | ranjan | carrar: actually i want boht |
| 06:20.47 | ranjan | i mean both |
| 06:20.58 | carrar | those are the actually commands |
| 06:21.01 | carrar | actual |
| 06:21.16 | carrar | core show application playback |
| 06:21.23 | carrar | core show application background |
| 06:22.01 | ranjan | actually i am getting an error when i give the wav file for playback and background Unable to open ipsr-welcome (format 0x4 (ulaw)): No such file or directory |
| 06:22.14 | wdoekes2 | don't add the ".wav" in the filename |
| 06:22.28 | ranjan | wdoekes2: no i didnt add |
| 06:22.59 | ranjan | i wrote the line like this exten => s,n,Playback(ipsr-welcome) |
| 06:23.05 | wdoekes2 | did you put it in the right dir? |
| 06:23.11 | ranjan | and the filename is ipsr-welcome |
| 06:23.30 | ranjan | i put in the directory where all other sample sound files where located |
| 06:23.55 | wdoekes2 | add the extension to the file |
| 06:24.04 | wdoekes2 | but not in the dialplan |
| 06:25.01 | ranjan | actually the has extension its ipsr-welcome.wav i also tried with ipsr-welcome.ogg both doesnt work |
| 06:25.53 | ranjan | wdoekes2: i found one thing that all the files in the samples folder is .gsm format should i recompile asterisk to make asterisk use wav extension |
| 06:27.00 | wdoekes2 | no.. not if the format_wav.so is loaded |
| 06:27.08 | wdoekes2 | > module show |
| 06:29.04 | ChannelZ | what are the properties of the wav file? |
| 06:29.30 | ChannelZ | If it's stereo, high samplerate, etc. it will barf |
| 06:30.14 | wdoekes2 | barf != No such file, though |
| 06:30.15 | ranjan | ChannelZ: same as other files in that folder |
| 06:30.42 | wdoekes2 | and you can play those files? |
| 06:30.57 | ChannelZ | and it's not owned by root and unreadable by whatever user asterisk runs as on your system, or something? |
| 06:31.48 | ranjan | wdoekes2: i can play those files and the module show shows that format_was is present |
| 06:33.11 | ranjan | ChannelZ: its owned by root |
| 06:33.49 | ranjan | ChannelZ: all other sample files are also owned by root |
| 06:34.33 | *** join/#asterisk Tim_Toady (~moi@193.92.224.201.dsl.dyn.forthnet.gr) |
| 06:34.48 | ranjan | ChannelZ: and asterisk is run from the root prompt |
| 06:35.14 | ChannelZ | then it's either really in the wrong place or you have a type-o or something else going on that we'll probably never figure out |
| 06:36.14 | wdoekes2 | or the file really isn't a wave file |
| 06:37.04 | ranjan | wdoekes2: ChannelZ this is the error i am getting :) |
| 06:37.05 | ranjan | http://pastebin.com/VALgp2hX |
| 06:37.38 | ChannelZ | that's great, but we can't see your filesystem |
| 06:37.39 | ranjan | wdoekes2: actually that is the output of the text to speech software from cepstrals |
| 06:38.18 | ranjan | ChannelZ: what should i do for that? i want to get it work somehow :( |
| 06:38.37 | ChannelZ | so now you tell us the whole story |
| 06:39.01 | ChannelZ | What version of * is this |
| 06:39.28 | ranjan | ChannelZ: i am running a fedora minimal installation on virtual box and its the latest stable version asterisk 1.6.2 |
| 06:39.45 | *** join/#asterisk cnu (cnu@the.ultimate.lamer.la) |
| 06:40.09 | ChannelZ | so you're probably using the language prefix? IE your sounds are in /var/lib/asterisk/sounds/en/* or somesuch |
| 06:40.25 | ranjan | ChannelZ: and i am on a LAN and my virtualbox instance also has an IP and configured in bridge mode |
| 06:40.30 | wdoekes2 | ranjan: I don't know your software, but the file should begin with RIFF |
| 06:40.31 | wdoekes2 | hd myfile.wav | head -n1 |
| 06:40.31 | wdoekes2 | 00000000 52 49 46 46 a2 14 00 00 57 41 56 45 66 6d 74 20 |RIFF....WAVEfmt | |
| 06:41.05 | *** join/#asterisk lftsy (~lftsy@install.deckpoint.ch) |
| 06:41.14 | ranjan | actually my sounds are in /usr/local/var/lib/asterisk/sounds/en/* coz i installed with a --prefix=/usr/local option |
| 06:41.18 | ChannelZ | just use 'file'... file myfile.wav |
| 06:41.53 | carrar | file ipsr-welcome |
| 06:41.58 | carrar | file ipsr-welcome.wav |
| 06:42.20 | ranjan | carrar: ChannelZ wdoekes2 #file ipsr-welcome.wav output is ipsr-welcome.wav: RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, mono 16000 Hz |
| 06:42.26 | carrar | heh |
| 06:42.42 | carrar | try 8000 Hz |
| 06:42.51 | ChannelZ | well that's a problem. Make it 8kHz or try renaming it to .slin instead |
| 06:43.03 | ChannelZ | or what is it called.. slin16 or something? I forget |
| 06:43.10 | ChannelZ | but just make it 8kHz |
| 06:43.43 | ranjan | ChannelZ: ok, but why is it showing ipsr-welcome does not exist in any format |
| 06:44.08 | ChannelZ | Anyways, what was I saying earlier about it being the wrong samplerate etc? Hmmmmmm................... |
| 06:44.13 | carrar | do you have a acceptle .gsm format for it to use? |
| 06:44.39 | ranjan | ChannelZ: ok :) |
| 06:45.25 | ChannelZ | ranjan: because * auto-selects the file to play based on the channel's codec, so it searches for many filenames. If it can't understand the file or doesn't see one it likes, it's not necessarily a failure as it looks for something else. |
| 06:45.48 | ChannelZ | In your case the file is not the right type, gets skipped, and then ultimately the whole thing fails because there's nothing else to do. |
| 06:46.38 | ranjan | ChannelZ: ok :) let me try with a 800Hz sample rate |
| 06:46.44 | ChannelZ | So as far as * is concerned, it didn't ultimately find a file to play and tells you "File ipsr-welcome does not exist in any format" |
| 06:48.08 | *** join/#asterisk UQlev (~Yuriy@212.50.99.8) |
| 06:49.59 | carrar | 800Hz = 8000Hz |
| 06:50.03 | carrar | BTW |
| 06:50.04 | carrar | :) |
| 06:50.44 | ChannelZ | 800 will sound supar-sexay |
| 06:56.00 | *** join/#asterisk lftsy (~lftsy@install.deckpoint.ch) |
| 06:57.53 | *** join/#asterisk c0rnoTa (~c0rnoTa@178.177.146.70) |
| 06:57.59 | *** part/#asterisk c0rnoTa (~c0rnoTa@178.177.146.70) |
| 07:01.50 | *** join/#asterisk stoffell (~kristof@fw.catsanddogs.com) |
| 07:02.42 | *** join/#asterisk ukine (~ukine@14-145.97-97.tampabay.res.rr.com) |
| 07:06.03 | *** join/#asterisk waschtl (~waschtl@3ed8a58a.d.d9tcloud.de) |
| 07:07.23 | *** join/#asterisk `paul (~kutimoy@119.93.15.82) |
| 07:07.44 | `paul | is it ok if i delete /var/lib/astdb then restart asterisk? |
| 07:11.55 | *** join/#asterisk v1s (~v1s@2001:5c0:110a:1400:5ab0:35ff:fe84:d147) |
| 07:12.17 | ChannelZ | yeah if you've got something against the astdb.. |
| 07:12.56 | v1s | whats the best way to handle dialing for multiple lines? Like try line1 if busy use line2 if busy use line3 if busy try line 1 again etc.. |
| 07:13.31 | *** join/#asterisk markit (~marco@88-149-177-66.staticnet.ngi.it) |
| 07:13.37 | *** join/#asterisk mintos (~mvaliyav@nat/redhat/x-ljkiusaywrwgnrjx) |
| 07:14.16 | markit | hi, time ago, if I remember correctly, sounds for release like 1.6 were in http://downloads.asterisk.org/pub/telephony/sounds/releases/, but can't find anymore. Anyone can tell me where has been moved? Would love to find 1.8-related also |
| 07:14.18 | *** join/#asterisk shamelessn00b (~chatzilla@58-65-172-114.nayatel.pk) |
| 07:14.31 | *** join/#asterisk mpe (~mpe@94.127.49.1) |
| 07:15.29 | v1s | markit: http://downloads.asterisk.org/pub/telephony/sounds/ |
| 07:16.14 | markit | v1s: is where I've looked at, but maybe I'm blind, cant find them, you do?+ |
| 07:19.04 | markit | btw, not much talk here... no european users/devs? :) |
| 07:19.36 | v1s | markit: script I use to get the sounds for 1.6.2 |
| 07:19.38 | *** join/#asterisk creativx (~creadurex@226.62-97-205.bkkb.no) |
| 07:19.40 | v1s | http://pastebin.com/dQwSytP1 |
| 07:20.22 | markit | v1s: thanks, but I don't think works since the files are NOT there, or am I wrong? |
| 07:20.46 | v1s | they are there. just its by type |
| 07:20.54 | v1s | like wav alaw gsm etc.. |
| 07:21.13 | markit | where is the release number? let me check again |
| 07:21.49 | markit | v1s: please, provide me the url of a 1.6 or 1.8 sound file you (but not me) can find there |
| 07:22.21 | v1s | core or extras? |
| 07:22.29 | markit | core |
| 07:22.36 | v1s | what format? |
| 07:22.40 | markit | I can only find gsm-current, not gsm-1.6 or whatever |
| 07:22.55 | markit | well, gsm for start... is only that I can't see them listed |
| 07:23.08 | markit | if you point me the right place, I will grab the other format easely |
| 07:23.34 | markit | I don't understand if I'm blind or there is a misunderstandment with you, that's why I ask a url, so I will be sure :) |
| 07:23.48 | v1s | http://downloads.asterisk.org/pub/telephony/sounds/asterisk-core-sounds-en-gsm-current.tar.gz |
| 07:24.15 | markit | v1s: how can you tell that are for 1.6 release? or for 1.8? or 1.4 that is the current version? |
| 07:24.53 | v1s | 1.6.2 is current |
| 07:25.21 | markit | oh, I see... and what about 1.8? I provide the italian set, so time to time I check for modifications |
| 07:26.13 | markit | I do remember that time ago there was a dir with all the rel. number in the filename, something like asterisk-core-sounds-en-gsm-1.4.22.tar.gz |
| 07:26.19 | markit | or 1.6.0 |
| 07:26.20 | markit | etc |
| 07:26.24 | v1s | I am going to take a wild guess and just say that they probably the same sounds files for 1.8 but I am not sure to be honest I just use 1.6.2 |
| 07:26.42 | markit | v1s: ok, thanks a lot for your help and pacience :) |
| 07:26.48 | *** join/#asterisk lost_soul (~noymfb@cpe-67-249-141-227.twcny.res.rr.com) |
| 07:27.26 | puzzled | markit: http://downloads.asterisk.org/pub/telephony/sounds/releases/ |
| 07:27.28 | markit | I'll check if issue https://issues.asterisk.org/view.php?id=16825 is still valid |
| 07:27.59 | markit | puzzled: thanks, is where I looked at, but found only 1.4.x |
| 07:28.13 | markit | and was looking for 1.6.x AND 1.8.x, possibly |
| 07:28.56 | puzzled | markit: think those are the latest. never heard that there were different sound files for 1.4, 1.6 and 1.8 |
| 07:29.27 | *** join/#asterisk tamiel (~tamiel@213.30.183.226) |
| 07:30.01 | markit | 2009-03-09 I downloaded asterisk-1.6.0.6.tar.gz for instance |
| 07:30.10 | markit | and asterisk-1.6.1-rc1.tar.gz |
| 07:31.12 | markit | I would be very happy if they unified the sounds file, since I'm providing italian translation is much easier cover the whole release set if unified |
| 07:37.34 | Docteh | unified how? |
| 07:37.44 | Docteh | i think they're already unified :-/ |
| 07:42.35 | *** join/#asterisk hrhrhr (~c1@unaffiliated/hrhrhr) |
| 07:43.34 | Docteh | markit: the version numbers for the sounds are different from the asterisk version numbers. 1.4.33 1.6.2.9 and 1.8.0-rc3 all include asterisk-core-sounds-en-gsm-1.4.19.tar.gz |
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| 08:02.48 | markit | Docteh: good to know, thanks |
| 08:02.57 | markit | Docteh: but how can I discover this fact? |
| 08:04.00 | *** part/#asterisk `paul (~kutimoy@119.93.15.82) |
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| 08:06.14 | joobie | anyone in the UK who has played with asterisk? |
| 08:07.36 | henk | there are a few million people in the uk, i guess one of them will have... |
| 08:07.51 | *** join/#asterisk UQlev (~Yuriy@212.50.99.8) |
| 08:11.26 | markit | henk: I think he suspects this fact also, wondering if at least one is here and willing to reply ;P |
| 08:11.46 | henk | oh :-p |
| 08:12.18 | *** join/#asterisk _mwoodj_ (~mwoodj@pdpc/sponsor/digium/hyper-eye) |
| 08:12.42 | henk | i wonder what difference the country makes for him... |
| 08:13.19 | *** join/#asterisk Faustov (~fst@gentoo/user/faustov) |
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| 08:14.34 | markit | henk: well, if he asks, I'm sure there is a valid (at least for him) reason :) |
| 08:15.06 | markit | as general feeling, I dislike when I ask something and instead of giving me an answer, I'm flooding with question about why I ask it that way, lol |
| 08:15.25 | markit | I remember at the beginning of my GNU/Linux experience that I had to shutdown teh system with crontab |
| 08:15.45 | markit | and did not know if there was a command to shutdown (was a previous M$slave, sigh) |
| 08:15.50 | markit | and I asked in a channel |
| 08:16.04 | henk | yeah, me too, but in the end i usually get a result better than expected just because people question my asking... |
| 08:16.24 | markit | well, they started ask me "why do you want to do so? GNU/Linux never needs to be shut down" and something like that |
| 08:16.35 | markit | go figure... |
| 08:16.36 | *** join/#asterisk krion (~seb@unaffiliated/krion) |
| 08:16.37 | henk | lol |
| 08:16.42 | markit | then I started mastering google ;P |
| 08:16.58 | markit | henk: yes, me too, but I thing people should FIRST answer, then question |
| 08:17.08 | *** join/#asterisk mintos (~mvaliyav@nat/redhat/x-ollvgvufjelhiqnx) |
| 08:17.10 | creativx | first you marry, then you statoil |
| 08:17.21 | markit | like "the command is shoutdown -h now", but keep in mind that is not M$crap, you don't have to reboot often |
| 08:17.24 | markit | or something like that |
| 08:17.35 | henk | well, you asked a concrete question to which there is at least one answer. that's not quite what joobie did... and he got k-lined? wonder what for 'g' |
| 08:18.01 | *** join/#asterisk mintos (~mvaliyav@nat/redhat/x-lpqwegrnqpvuvfen) |
| 08:18.31 | markit | henk: oh, just a general conversation with you in a silent channel, don't want to be "against" you of course |
| 08:18.59 | markit | btw, I'm not english native speaker, so take my workds in the best and friendly way you can, even if they could not sound so |
| 08:19.08 | henk | ah ok :) yes, as long as there is a concrete question there should be a concrete answer, not just question bashing at least... |
| 08:19.09 | markit | it's because I've a limited vocabulary |
| 08:19.23 | henk | welcome to the club, i'm german |
| 08:19.25 | markit | i.e. don't know what "statoil" means, nor google language aids |
| 08:19.36 | henk | italian |
| 08:19.39 | henk | you are |
| 08:19.43 | *** join/#asterisk visik7 (~Adium@unaffiliated/visik7) |
| 08:19.43 | henk | right? |
| 08:19.53 | markit | henk: mmmm german people, as far as I've seen, in average are MUCH better english speakers than italians |
| 08:19.58 | markit | yep :) |
| 08:20.37 | *** join/#asterisk joobie (~joobie@CPE-58-170-204-28.lns7.win.bigpond.net.au) |
| 08:20.38 | henk | now that you say it: i have barely heard italians speak english... only in movies i think. |
| 08:21.32 | henk | i have no idea what statoil means either... |
| 08:21.37 | henk | creativx: enlighten us please :) |
| 08:28.05 | *** join/#asterisk mintos (~mvaliyav@nat/redhat/x-nbrjdimemjarccjf) |
| 08:29.47 | creativx | was just an old commercial that came to mind |
| 08:29.49 | creativx | anyways |
| 08:29.55 | creativx | i want to kick soxmix in the behind |
| 08:30.59 | *** join/#asterisk avruser (~user@94.158.60.12) |
| 08:32.03 | avruser | hello |
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| 08:39.02 | henk | creativx: oh ok |
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| 10:22.07 | avruser | Hello, Asterisk users! |
| 10:24.50 | zamba | hi there |
| 10:29.51 | avruser | Sorry, is there anybody, who faced with chan_mobile? |
| 10:38.48 | *** join/#asterisk ningia (~gain@109.69.131.226) |
| 10:40.15 | v1s | avruser: what u trying to do ? |
| 10:40.18 | v1s | or what is the prob |
| 10:44.33 | avruser | I'm compiled and installed chan_mobile. found appropriate bluetooth dongle, nikia phone. All ok, but when i trying to make call, i get kernel panic. |
| 10:46.14 | avruser | dongle csr, phone 6230i. |
| 10:49.15 | *** join/#asterisk mintos (~mvaliyav@nat/redhat/x-kplbjibecnytpebg) |
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| 10:51.15 | BenC[UK] | hi, can anyone help me with phpagi ? |
| 10:51.47 | BenC[UK] | im trying to send a call to a queue.. |
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| 11:17.04 | *** join/#asterisk Naikrovek (~jjohnson@unaffiliated/naikrovek) |
| 11:18.11 | *** join/#asterisk Yoe (~wouter@samba.grep.be) |
| 11:18.24 | Yoe | how can I tell asterisk which soundcard to use on the console? |
| 11:18.51 | Naikrovek | soundcard? |
| 11:19.33 | Yoe | ah, through alsa.conf :) |
| 11:19.47 | Naikrovek | alsa.conf? |
| 11:19.58 | Yoe | Naikrovek: yes, you can have asterisk act as a softphone by way of the 'dial' command |
| 11:20.08 | Naikrovek | command? |
| 11:20.11 | Naikrovek | (lol) |
| 11:20.13 | Yoe | Naikrovek: fuck you |
| 11:20.17 | Naikrovek | you? |
| 11:20.37 | Naikrovek | aw shit now i got the giggles |
| 11:20.53 | Naikrovek | time for some team fortress 2. that'll piss me off and put me back where i belong |
| 11:25.13 | *** part/#asterisk Yoe (~wouter@samba.grep.be) |
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| 11:26.22 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
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| 11:45.03 | nephilinturbo | Hi Guys. I cant find in anywhere that audio file that says: 'The person you are calling is unavailable, please try again' . Its played when you called an invalid extension |
| 11:45.12 | nephilinturbo | Someone knows the name of file ? |
| 11:45.39 | henk | check your dialplan... |
| 11:46.15 | nephilinturbo | Man , i dont have problems with dialplan. I want find that file to translate it |
| 11:46.22 | nephilinturbo | for users , when called wrong |
| 11:46.24 | *** join/#asterisk mintos (~mvaliyav@nat/redhat/x-pibzfilaniuiuimr) |
| 11:46.48 | nephilinturbo | I tranlated 99% of my server files , but cant find only that |
| 11:48.49 | nephilinturbo | somebody knows ? the location oh this file ? |
| 11:49.00 | nephilinturbo | The person you are calling is unavailable, please try again |
| 11:49.27 | henk | nephilinturbo: your dialplan is responsible for the file being played, so check it and you will know the filename... |
| 11:54.58 | nephilinturbo | No .. Im usind an free-pbx that Playback the files from DB. I cant find que file |
| 11:55.09 | markit | nephilinturbo: with the audio file, there is a .txt file with names and phrase in english |
| 11:55.24 | markit | maybe you can search for "the person" in it and find the file/s |
| 11:57.05 | markit | nephilinturbo: maybe is afreepbx stuff, in additional sound files |
| 11:57.13 | markit | in core I can't find that phrase |
| 11:57.27 | markit | btw, what language are you translating (re-reconrding) in? |
| 11:57.29 | *** join/#asterisk tzafrir (~tzafrir@bzq-218-155-148.cablep.bezeqint.net) |
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| 12:13.33 | petern_ | Anyone used voice recognition with Asterisk? I'm taking a look at http://www.voip-info.org/wiki/view/Sphinx, but wondered what other systems people might use or recommend. |
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| 12:22.45 | *** join/#asterisk [TK]D-Fender (~chatzilla@216.191.106.163) |
| 12:33.01 | petern_ | Oh, my distro has sphinx2 which is apparently not very good. |
| 12:33.04 | petern_ | Hmm. |
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| 12:39.46 | cjk | hi, has anyone an idea why asterisk opens an increasing number of udp sockets? |
| 12:40.48 | *** join/#asterisk Nwab (~Schnitzel@unaffiliated/benwa) |
| 12:40.54 | Nwab | hi |
| 12:41.29 | Naikrovek | cjk: define "increasing number." UDP is how Asterisk communicates with other phone systems and phones |
| 12:41.29 | [TK]D-Fender | cjk: My guess would be "communication" |
| 12:42.24 | cjk | [TK]D-Fender, i have 0 calls |
| 12:42.41 | [TK]D-Fender | cjk: A lot more goes on than actual calls you know... |
| 12:42.55 | cjk | [TK]D-Fender, like ? |
| 12:43.08 | Naikrovek | communication |
| 12:43.18 | Naikrovek | asterisk talks to other phone systems and phones all the time |
| 12:43.21 | Naikrovek | even when calls are not going on |
| 12:43.25 | [TK]D-Fender | cjk: OPTIONS packets constantly to peers with "qualify", Registrations, Subscriptions, etc |
| 12:43.30 | Naikrovek | phone will say "i'm here" asterisk will say "okay shithead" |
| 12:43.36 | cjk | thats port 5060, no? |
| 12:43.38 | Naikrovek | things like that |
| 12:43.45 | Naikrovek | 5060, and 10000-20000 by default |
| 12:43.58 | [TK]D-Fender | cjk: 5060 is the INBOUND port, not the OUTBOUND |
| 12:44.07 | Naikrovek | you're worrying about things you need not worry about. |
| 12:44.13 | Nwab | is my port forwarding configuration ok for asterisk ? http://paste.debian.net/94423/ |
| 12:44.15 | Naikrovek | take a pill and relax |
| 12:44.17 | cjk | Naikrovek, my system crashes, thats why i worry |
| 12:44.24 | Naikrovek | you failed to mention that |
| 12:44.30 | Naikrovek | the problem, then, is not open UDP ports |
| 12:44.33 | Naikrovek | it's sytem crashes |
| 12:44.36 | Naikrovek | system* |
| 12:44.37 | [TK]D-Fender | cjk: As well as your version. Description of environment, etc |
| 12:44.55 | cjk | ok, one second |
| 12:47.22 | Nwab | anyone ? |
| 12:47.55 | Naikrovek | i dunno iptables very well, Nwab |
| 12:48.05 | Nwab | mmmhh ... |
| 12:48.17 | Naikrovek | in fact, i despise it, but it looks okay to my completely clueless eyes |
| 12:48.29 | Nwab | :) |
| 12:48.45 | Nwab | anyone else ? ;) |
| 12:52.53 | [TK]D-Fender | Nwab: Looks fine. Are you really using all those protocols? |
| 12:53.12 | *** join/#asterisk TobSnyder (~schneider@dslb-088-073-211-188.pools.arcor-ip.net) |
| 12:53.19 | creativx | mixmonitor <3 |
| 12:54.03 | Nwab | [TK]D-Fenderâ not the last one, i will uncomment it. an d what about aix and aix2 ? |
| 12:54.12 | Nwab | *comment |
| 12:55.03 | [TK]D-Fender | Nwab: IAX1 hasn't existed for almost a decade, flush that |
| 12:55.45 | Nwab | [TK]D-Fenderâ mwarf ... ok thanks |
| 12:56.44 | *** join/#asterisk slidesinger (~slidesing@c-174-57-6-126.hsd1.nj.comcast.net) |
| 12:56.49 | leifmadsen | hmmm, anyone have an idea how I can insert a "newline" within the dialplan? e.g.: JabberSend(asterisk,leif@foo.com,This is my message. I want this part on a new line.) |
| 12:58.38 | Naikrovek | hmmm indeed |
| 12:58.51 | Naikrovek | so \n must not work or you wouldn't be asking |
| 12:59.27 | Naikrovek | ...if \n doesn't work try \\n. |
| 13:00.44 | leifmadsen | Naikrovek: heh ya |
| 13:00.49 | leifmadsen | \n just shows up literal |
| 13:00.57 | leifmadsen | as \n, not just 'n' |
| 13:01.07 | Naikrovek | okay well \n won't work then |
| 13:01.08 | Naikrovek | um |
| 13:01.10 | Naikrovek | \\n i mean |
| 13:01.12 | leifmadsen | which makes me think \\n won't work |
| 13:01.13 | [TK]D-Fender | leifmadsen: I don't believe there is a standard encoding method for things like this. You'd probably have to mod JabberSend directly, or use some other scrip |
| 13:01.16 | leifmadsen | aye |
| 13:01.19 | creativx | leifmadsen: i think the choppy audio problem is solved by using mixmonitor instead of monitor.. 1.2 FTW! |
| 13:01.34 | leifmadsen | I stopped using 1.2 years ago :) |
| 13:01.42 | creativx | i have no plans to upgrade! |
| 13:01.42 | creativx | hehe |
| 13:01.48 | leifmadsen | heh, enjoy :) |
| 13:01.55 | creativx | id rather purchase a hosted solution before i touch this beast |
| 13:02.07 | Naikrovek | you could just hire an expert |
| 13:02.14 | creativx | it was fun getting it to work |
| 13:02.29 | creativx | Naikrovek: yes indeed |
| 13:02.44 | creativx | this 1.2 has been with us the last hmm.. 3 years now? |
| 13:05.14 | krion | this is not fully related but i will try |
| 13:05.33 | ningia | hi all |
| 13:05.40 | krion | sip speaking, which signal should mean "the phone is ringing", is 180 right ? |
| 13:05.58 | krion | because i got 2 providers, one who makes the phones ring (back) |
| 13:06.02 | krion | and one who doesn't |
| 13:06.29 | krion | and the only diff i found is that the one who doesn't, doesn't send a 183 ssion Progress SDP ( telephone-event) |
| 13:07.12 | Naikrovek | creativx: well 1.8 is probably going to come out RealSoonNow and it's long term support, so you can settle on that version for a while if you're feeling frisky |
| 13:09.57 | *** join/#asterisk patrick^ (~patrick_@2001:470:b0ea:1:219:21ff:fe4e:f5de) |
| 13:11.30 | cjk | sorry for the delayed answer. I am using/testing asterisk 1.8 beta2 and my system is idle, so no calls at all, maybe 10 test calls per day. I have like 10 phones that register to the system |
| 13:11.52 | cjk | i use netstat -alp | grep -i aster | grep -i udp to check for open udp ports used by asterisk |
| 13:12.04 | cjk | this list increases until too many file descriptors are used |
| 13:12.09 | cjk | the system crashes |
| 13:12.49 | cjk | i do not see any traffic on those ports using tshark |
| 13:13.09 | Naikrovek | cjk: okay |
| 13:13.46 | Naikrovek | maybe a bug report is in order? |
| 13:15.23 | Naikrovek | could probably search through https://issues.asterisk.org/ to see if anyone else has reported it |
| 13:15.42 | Naikrovek | if not, probably going to be a system configuration thing or something so rare that you'll need to provide steps to reproduce |
| 13:15.56 | cjk | Naikrovek, hmmm that process takes too long.... i saw this problem multiple times on the mailing list for asterisk 1.6, I am willing to provide access to my server if someone wants to debug it. opening a bug report has never helped me in any way. issues.asterisk.org looks like a ignore list for me |
| 13:16.09 | Naikrovek | it definitely is not |
| 13:16.19 | Naikrovek | but if that's how you feel, then that's how you feel |
| 13:16.22 | Naikrovek | no one can fault you for that |
| 13:16.35 | leifmadsen | cjk: maybe you try using RC3 instead of beta2? :) |
| 13:16.58 | Naikrovek | wow that's recent |
| 13:17.02 | Naikrovek | yesterday, nice |
| 13:17.08 | leifmadsen | issues.asterisk.org is anything but an ignore list :) I know.. because I triage at least 10 bugs a day |
| 13:17.18 | leifmadsen | Naikrovek: heh ya I just released it :) |
| 13:17.18 | cjk | leifmadsen, sorry, its RC2 |
| 13:17.31 | cjk | leifmadsen, but i am going to upgrade to RC3 right now |
| 13:17.32 | cjk | :) |
| 13:17.59 | Naikrovek | i'm curious to know if this is fixed |
| 13:18.09 | Naikrovek | cjk: let me know if the problem persists in RC3 |
| 13:18.19 | TobSnyder | someone from germany in here? |
| 13:20.28 | *** mode/#asterisk [+o [TK]D-Fender] by ChanServ |
| 13:20.33 | *** topic/#asterisk by [TK]D-Fender -> Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.8.0-rc3 (2010/10/07), 1.6.2.13 (2010/09/15), 1.4.36 (2010/09/15), *-Addons 1.6.2.2, 1.4.12 (2010/09/22), dahdi-linux 2.4.0 (2010/09/01), dahdi-tools 2.4.0 (2010/09/01), libpri 1.4.11.4 (2010/09/01) -=- Related channels: #asterisknow,#switchvox,#freepbx,#asterisk-dev,#asterisk-bugs,#astricon |
| 13:21.39 | *** join/#asterisk jsgoecke (~Adium@c-98-207-5-162.hsd1.ca.comcast.net) |
| 13:21.54 | *** join/#asterisk tzafrir (~tzafrir@bzq-218-155-148.cablep.bezeqint.net) |
| 13:21.58 | leifmadsen | 402! |
| 13:22.04 | jsgoecke | I thought folks here might enjoy this: http://blog.tropo.com/2010/10/08/asterisk-tropo-and-a-single-adhearsion-dialplan/ |
| 13:22.10 | jsgoecke | Good morning everyone! |
| 13:22.27 | leifmadsen | jsgoecke: morn! |
| 13:23.15 | *** mode/#asterisk [-o [TK]D-Fender] by [TK]D-Fender |
| 13:23.57 | *** join/#asterisk Katty (~Katty@mail.copi-rite.com) |
| 13:24.02 | Katty | OHAI |
| 13:24.19 | [TK]D-Fender | Katty: Mew. |
| 13:26.52 | *** join/#asterisk diegoviola (~diego@190.23.77.213) |
| 13:26.57 | *** part/#asterisk diegoviola (~diego@190.23.77.213) |
| 13:30.31 | *** join/#asterisk adyn (~adyn@pdpc/supporter/active/adyn) |
| 13:31.15 | ningia | Hi all. Can I, in my dialplan, jump from a context to another? |
| 13:31.31 | cjk | RC3 runnig, just 2 udp ports open (IAX and SIP) perfekt, so lets wait till the problem appears |
| 13:33.37 | [TK]D-Fender | ningia: "core show application goto" |
| 13:34.26 | cjk | why does asterisk start to listen on ports under 10000 ex: udp 0 0 *:14816 *:* 28006/asterisk |
| 13:34.26 | cjk | udp 0 0 *:14817 *:* 28006/asterisk |
| 13:34.26 | cjk | udp 0 0 *:4331 *:* 28006/asterisk |
| 13:35.24 | Nwab | is it a good idea to upgrade from 1.6 to 1.8 ? Is it stable (enougn) or not yet ? |
| 13:35.52 | henk | cjk: random ports? might be for dns... |
| 13:36.36 | cjk | henk, good idea, i will check this out. thanks for the direction |
| 13:38.53 | Nwab | yes ? no ? |
| 13:40.24 | ningia | [TK]D-Fender: thx... |
| 13:40.46 | [TK]D-Fender | Nwab: Release CANDIDATE. Do we really need to say more? |
| 13:41.17 | Nwab | ok ok |
| 13:41.35 | coppice | I thought americans only had a choice between democrat and republican candidates |
| 13:41.35 | [TK]D-Fender | Nwab: Do you upgrade to the very first release of a new OS branch the day it comes out or wait and see until the first few dozen systems explode? |
| 13:42.06 | [TK]D-Fender | coppice: I'm not American :) |
| 13:42.51 | Nwab | coppiceâ :) |
| 13:43.41 | coppice | [TK]D-Fender you're not a Digium developer |
| 13:44.37 | Nwab | [TK]D-Fenderâ do tou really need an answer ? :) |
| 13:44.44 | Nwab | *you |
| 13:45.31 | Naikrovek | how DARE you |
| 13:45.37 | Naikrovek | okay what's going on here |
| 13:45.49 | *** join/#asterisk asteriskmonkey (~darlek@74-51-38-204.telnetcommunications.com) |
| 13:46.11 | asteriskmonkey | anyone install 1.6.2.13 on freebsd yet? |
| 13:49.11 | TobSnyder | <PROTECTED> |
| 13:49.21 | TobSnyder | I would rather get a message like "No route to destination" or "the number dial does not exist" |
| 13:49.34 | TobSnyder | so is there any way to check ISDN Cause codes and play individual messages? |
| 13:50.58 | *** join/#asterisk frigidzephyr (~rnewton@nat/digium/x-ubbharvqlijzvced) |
| 13:51.23 | asteriskmonkey | TobSnyder: you can use cause codes to redirect to any message you like |
| 13:51.47 | asteriskmonkey | just use a macro that uses the cause code to jump to a message thats more appropriate |
| 13:52.16 | TobSnyder | okay |
| 13:52.35 | TobSnyder | just forgot to say tha I am using freepbx |
| 13:52.44 | TobSnyder | but if it is possible, I will find it there |
| 13:52.55 | *** join/#asterisk rgsteele (~rgsteele@apo.aweber.com) |
| 13:52.59 | [TK]D-Fender | TobSnyder: NOt here <- |
| 13:53.22 | TobSnyder | [TK]D-Fender: just was not quite sure if it is a asterisk or freepbx problem |
| 13:53.27 | TobSnyder | so now I know about |
| 13:53.44 | [TK]D-Fender | TobSnyder: You're asking about choosing differeent reactions to call outcomes. That is DIALPLAN. |
| 13:54.08 | asteriskmonkey | time to scrap you freepbx and learn asterisk :) |
| 13:54.17 | TobSnyder | sure you're all right |
| 13:54.21 | asteriskmonkey | or get used to using the _custom.conf files |
| 13:54.43 | TobSnyder | still I have to admit I am no asterisk guru |
| 13:55.01 | TobSnyder | but I will go on learning |
| 13:55.11 | asteriskmonkey | its no tought at all |
| 13:55.13 | TobSnyder | just for your calender: ask me in 1 year again :D |
| 13:55.34 | asteriskmonkey | if you spend some time could probably ask you i 1 week |
| 13:55.56 | TobSnyder | and time is my major problem ;) |
| 13:57.16 | *** join/#asterisk dacm_work (~dan@host86-166-241-51.range86-166.btcentralplus.com) |
| 13:57.20 | dacm_work | Hmm |
| 13:57.47 | dacm_work | Is it common for people to attempt to break into asterisk servers? |
| 13:57.55 | dacm_work | I guess is means free phone calls. |
| 13:58.05 | *** join/#asterisk waKKu (~blah@unaffiliated/wakku) |
| 13:58.10 | waKKu | hi folks.. |
| 13:58.18 | dacm_work | has someone trying to brute force register.. |
| 13:58.21 | *** join/#asterisk Faithful (~Faithful@180.194.2.50) |
| 13:58.30 | *** join/#asterisk crowbar7 (~JKLOBUCAR@12.2.84.2) |
| 13:58.35 | waKKu | i need to encrypt my voip traffic, what do you recommend? |
| 13:59.09 | dacm_work | Is there a way to make asterisk pause for a while before telling the client they have the wrong password? |
| 13:59.13 | TobSnyder | asteriskmonkey: but anyway thanks |
| 13:59.26 | dacm_work | (This guy could be using a lot of my bandwidth at this rate...) |
| 13:59.53 | asteriskmonkey | you getting sip hacked or dialtone hacked? |
| 13:59.58 | asteriskmonkey | if its sip throw on fail2ban |
| 14:00.45 | dacm_work | asteriskmonkey: SIP, (not even sure what dialtone hacked would be... :-S) |
| 14:01.04 | waKKu | dacm_work you can use "Wait(#)" |
| 14:01.35 | *** join/#asterisk tzafrir (~tzafrir@bzq-218-155-147.cablep.bezeqint.net) |
| 14:01.37 | dacm_work | asteriskmonkey: Thanks for the pointer. |
| 14:01.46 | waKKu | oh, wait.. wrong password for billing... sorry |
| 14:02.03 | Katty | Qwell: i transfered to a different server, and faction changed. |
| 14:02.08 | dacm_work | waKKu: Is it possible to do that when registering SIP clients? |
| 14:02.45 | waKKu | dacm_work not sure, probably not |
| 14:03.53 | dacm_work | Another idea: is it possible to make some users only be able to register if they are on the local network? |
| 14:04.39 | waKKu | you can specify a range for network at host |
| 14:08.32 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 14:08.56 | [TK]D-Fender | dacm_work: Firewall him off. |
| 14:09.11 | [TK]D-Fender | dacm_work: and yes. permit/deny <- |
| 14:09.21 | [TK]D-Fender | dacm_work: Go read the sample configs |
| 14:12.20 | *** join/#asterisk TimeRider (~steve@188-220-33-123.zone11.bethere.co.uk) |
| 14:13.34 | *** join/#asterisk aidinb (~Aidin@unaffiliated/aidinb) |
| 14:14.41 | *** join/#asterisk odenkos (~odenkos@ip-212-081-019-170.static.nextra.sk) |
| 14:15.43 | dacm_work | [TK]D-Fender: Thanks. |
| 14:16.39 | *** join/#asterisk Tech_Travis (~Travis_El@cpe-76-168-191-127.socal.res.rr.com) |
| 14:16.45 | *** join/#asterisk Deeewayne (~dwayne@75-150-14-49-Atlanta.hfc.comcastbusiness.net) |
| 14:16.45 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 14:18.25 | *** join/#asterisk p3nguin (~xwQ5kwYl6@cpe-0210-4bff-fe2b-9074.dhcp.a2infotech.com) |
| 14:19.50 | raden | hugs Katty |
| 14:20.55 | leifmadsen | waKKu: you can encrypt the SIP signalling and RTP media with TLS and SRTP respectively with asterisk 1.8 |
| 14:21.38 | waKKu | hm.. leifmadsen is it dependant of the softphone/ip-phone on the other end, right? |
| 14:21.53 | leifmadsen | waKKu: it is dependent on the other end being able to support such protocols, yes |
| 14:21.58 | leifmadsen | otherwise the only other way is via VPN |
| 14:22.18 | leifmadsen | it's hard for me to have a conversation with someone who speaks Russian when I don't speak it. |
| 14:22.22 | *** join/#asterisk n3hxs (~HAMming@c-76-29-19-100.hsd1.il.comcast.net) |
| 14:22.35 | Naikrovek | <-- amazed by how quickly people change their opinion of him when they find that he's an Atheist. "I like that guy!" He believes in evolution, not creation. "I don't like that guy!" |
| 14:22.56 | waKKu | hahaa |
| 14:23.08 | waKKu | leifmadsen thanks for your effort ;) |
| 14:23.08 | leifmadsen | Naikrovek: don't worry, I don't hate you any more than I already did :) |
| 14:23.15 | Naikrovek | aww |
| 14:23.22 | Naikrovek | feelin the love from leifmadsen |
| 14:23.26 | leifmadsen | <3 |
| 14:23.28 | Nwab | :) |
| 14:23.38 | Katty | ohai |
| 14:23.45 | Katty | i has loves to share! |
| 14:23.49 | Katty | hugs on Naikrovek |
| 14:23.51 | Katty | hugs on leifmadsen |
| 14:23.57 | leifmadsen | w00t! |
| 14:24.19 | Naikrovek | reciprocates... a little too much, creeping Katty out slightly. |
| 14:24.27 | p3nguin | Is it commonplace (a lot of people do it) to run IP phones through a VPN when connecting to a PBX over the public internet? |
| 14:25.58 | leifmadsen | hard to say, I've never taken a poll |
| 14:27.13 | Katty | Naikrovek: it is hard to creep me out, sir |
| 14:27.17 | *** join/#asterisk rrb3942 (~rbullock@208.34.105.161) |
| 14:27.32 | Katty | p3nguin: i do that often. |
| 14:28.04 | *** join/#asterisk tzafrir (~tzafrir@bzq-218-155-147.cablep.bezeqint.net) |
| 14:29.48 | Naikrovek | p3nguin: yes |
| 14:32.30 | *** join/#asterisk JamesHarrison (~jharr@hometree.mmmetrics.co.uk) |
| 14:33.24 | p3nguin | What's the benefit? Are you worried that people are going to capture your RTP streams and replay them, hearing what you're talking about? |
| 14:34.40 | mmlj4 | you're not paranoid if they really are out to get you |
| 14:37.44 | Naikrovek | well |
| 14:37.47 | Naikrovek | you're still paranoid |
| 14:37.49 | Naikrovek | it's just justified |
| 14:37.50 | *** join/#asterisk Bakey (61411212@gateway/web/freenode/ip.97.65.18.18) |
| 14:38.21 | Naikrovek | p3nguin: benefit here is that it's much easier to make it work over a vpn with no NAT involved than to lose the VPN and fsck with NAT |
| 14:38.24 | Bakey | hello #asterisk |
| 14:38.25 | Naikrovek | we had the VPN up anyway |
| 14:38.38 | p3nguin | Alright, that's a valid reason. |
| 14:38.51 | Bakey | question about locally hosted #asterisk vs. using a provider |
| 14:39.09 | Naikrovek | so we have two offices in India and one connects via VPN, the other does not |
| 14:39.19 | p3nguin | bakey: Asterisk is Asterisk, regardless of its location. |
| 14:39.20 | Naikrovek | guess which one works reliably? the VPN phones never fail, |
| 14:39.33 | Naikrovek | the NAT over public internet phones haven't worked for more than 10 minutes at a time |
| 14:39.47 | Bakey | p3nguin - thanks for that, but I have more details I'm trying to type out |
| 14:40.04 | Naikrovek | type 'em! |
| 14:40.09 | Naikrovek | :P |
| 14:40.13 | Bakey | if we have a really distributed user base, at what point does it make more sense to have this hosted by someone else versus bringing it in house? |
| 14:40.29 | Bakey | currently 85 users, 65 of which are remote |
| 14:40.31 | Bakey | give or take |
| 14:40.35 | Naikrovek | no vpn involved? |
| 14:40.38 | Bakey | we have a 4xT1 here for Internet |
| 14:40.42 | p3nguin | Network uptime and bandwidth capacity will be your factors. |
| 14:40.45 | Bakey | not for phones unless we have to |
| 14:40.49 | Naikrovek | i worked for a place like that once |
| 14:40.51 | Naikrovek | it was nice |
| 14:40.57 | Naikrovek | next wave logistics |
| 14:41.01 | Naikrovek | java development |
| 14:41.04 | Bakey | should I bring this in house or host it somewhere? |
| 14:41.05 | Naikrovek | ahh, those were the days |
| 14:41.19 | p3nguin | I'd do it in-house. |
| 14:41.24 | Naikrovek | Bakey: you'll get the same performance either way, except with hosting it in house, it's cheaper |
| 14:41.39 | *** join/#asterisk Flyser__ (~flyser@unaffiliated/flyser) |
| 14:41.40 | Naikrovek | a little more work on your part, but much cheaper |
| 14:41.46 | Naikrovek | i'd do in-house as well |
| 14:41.50 | p3nguin | Are you going to be using an ITSP for calls, or will you connect directly to the PSTN? |
| 14:42.02 | Qwell | Katty: O.o |
| 14:42.21 | Bakey | p3nguin ?? I see that you typed some acronyms, but my simplisting SysAdmin brain doesn't process that :) |
| 14:42.26 | Bakey | I have 0 VOIP experience |
| 14:42.28 | Bakey | :( |
| 14:42.34 | [TK]D-Fender | ~itsp |
| 14:42.35 | infobot | [~itsp] An ITSP is an Internet Telephony Service Provider (or VoIP telephone company). They allow you to either SEND calls to the PSTN (this is called termination), RECEIVE calls from the PSTN (called origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs. |
| 14:42.36 | [TK]D-Fender | ~pstn |
| 14:42.36 | infobot | pstn is, like, Public Switched Telephone Network, or "please stop the nonsense" |
| 14:42.44 | mmlj4 | VOIP++ # now it's 1 |
| 14:43.13 | Naikrovek | ITSP = IP voice service provider. PSTN = regular phone service, as in analog lines to your premise |
| 14:43.21 | Bakey | also, that was supposed to be *simplistic |
| 14:43.35 | Bakey | what are the trade offs? |
| 14:43.41 | Bakey | ITSP vs PSTN |
| 14:43.46 | Naikrovek | asterisk can connect to the first with just data service, like 4 T1s, and can connect to the second with IP telephony hardware in the asterisk server |
| 14:44.23 | Naikrovek | i would recommend going with voip service and tossing some QoS on your routers |
| 14:44.23 | Bakey | I don't think we'd be opposed to either, and I'd almost prefer dedicating some sort of circuit to the phones anyway |
| 14:44.33 | *** mode/#asterisk [+b *!~diego@190.23.77.213] by Qwell |
| 14:44.36 | Qwell | Not welcome here. |
| 14:44.39 | leifmadsen | indeed |
| 14:44.47 | Naikrovek | what |
| 14:44.50 | Naikrovek | i missed something |
| 14:44.56 | Qwell | no you didnt |
| 14:45.11 | leifmadsen | he has bad history with the project |
| 14:45.14 | Flyser__ | Hi, I am using asterisk 1.6 for my SIP<->ISDN gateway with an avm Fritz!card and fcpci+chan_capi. Unfortunately the fcpci is not developed anymore and it's becoming increasingly difficult to use fcpci with recent kernels. I was going to try the mISDN driver with chan_lcr, but the fritzcard is not supported by lcr. What can I do to keep my gateway running? |
| 14:45.17 | Bakey | Naikrovek - can 4xT1 handle that number of users OK? |
| 14:45.42 | Naikrovek | Bakey: you can get i think 50 calls per T1 with the G729 codec. Maybe it's 49 |
| 14:46.15 | Bakey | But that's also assuming no other network traffic, right? |
| 14:46.26 | leifmadsen | right |
| 14:46.35 | Bakey | man I'm such a VOIP noob |
| 14:46.36 | Naikrovek | G711 is the regular, for the most part uncompressed codec. if G711 = raw .wav, G729 = MP3 |
| 14:46.42 | Bakey | no idea what to recommend here |
| 14:46.49 | Naikrovek | that's okay |
| 14:46.53 | raden | 48 calls |
| 14:47.05 | logicwrath | does insecure=very work like port,invite in asterisk 1.6.2.12 |
| 14:47.26 | Naikrovek | i have tons of patience for newbs and i'll help you all i can |
| 14:48.02 | hrhrhr | you can tell when the patience starts running out tho |
| 14:48.08 | *** join/#asterisk luckyaba (~lucky@ip72-194-218-169.sb.sd.cox.net) |
| 14:48.12 | hrhrhr | when the replies START LOOKING LIKE THIS |
| 14:48.16 | Naikrovek | HOW DARE YOU |
| 14:48.25 | hrhrhr | for [TK]D-Fender, that's about 46 seconds into the channel |
| 14:48.26 | leifmadsen | logicwrath: invite=port,invite works like invite=very used to, yes |
| 14:48.40 | logicwrath | does invite=very still work though in that version? |
| 14:48.45 | logicwrath | err insecure=very |
| 14:48.49 | logicwrath | or does it do nothing |
| 14:48.52 | mmlj4 | we all know someone here that starts out with the caps, then settles down once he's figured out you're not a moron |
| 14:48.55 | leifmadsen | logicwrath: not sure, check CHANGES and UPGRADE.txt |
| 14:49.54 | Naikrovek | Bakey: you have choices so now you need to line them all up and compare pros and cons |
| 14:50.15 | *** join/#asterisk UQlev (~Yuriy@212.50.99.8) |
| 14:50.41 | Bakey | Naikrovek, I just need to understand the choices better so I know what the pros and cons even are :p |
| 14:51.08 | Naikrovek | hosted vs in-house, and for in-house do you want to connect to the phone system via IP or via server hardwaer |
| 14:51.11 | p3nguin | insecure=very probably doesn't work; use port,invite |
| 14:51.21 | Bakey | right |
| 14:51.40 | dacm_work | Does anyone know if a delay can be set for failed SIP registrations? (SSH does this I believe.) |
| 14:52.03 | Bakey | Is there any way to test this before I go full-bore replacement? |
| 14:52.10 | Naikrovek | Bakey: yes |
| 14:52.22 | Naikrovek | gets ready to start plugging AsteriskNOW |
| 14:52.35 | Bakey | googles AsteriskNOW |
| 14:52.49 | Naikrovek | Bakey: you familiar with virtual machines? |
| 14:53.27 | p3nguin | bakey: The typical reason for making one choice over the other is for cost reasons. |
| 14:53.32 | Katty | http://www.food.com/recipe/pineapple-black-bean-enchiladas-154388 <- Dinner |
| 14:54.09 | Naikrovek | mmm |
| 14:54.17 | *** join/#asterisk tzafrir (~tzafrir@212.179.75.202) |
| 14:54.52 | p3nguin | bakey: Some people use a single T1 connected to their Asterisk system and make up to 24 PSTN calls at a time. Other people use a data connection and send as many calls to their ITSP as they have bandwidth for. |
| 14:55.29 | Katty | Naikrovek: yesh. |
| 14:55.34 | Katty | Naikrovek: i'm thinking it will be quite yum. |
| 14:55.48 | *** join/#asterisk bsaxon (~bsaxon@12.107.149.61) |
| 14:56.34 | *** part/#asterisk asteriskmonkey (~darlek@74-51-38-204.telnetcommunications.com) |
| 14:57.02 | p3nguin | Can I expense tools when I buy them without having to depreciate them over time? |
| 14:58.03 | krion | i'm running Asterisk 1.6.2.10 |
| 14:58.16 | krion | what do i need to enable in order to get "core show locks" |
| 14:58.21 | Naikrovek | p3nguin: ...? i think you WANT them to depreciate |
| 14:58.50 | leifmadsen | krion: enable DEBUG_THREADS in menuselect and recompile |
| 14:58.51 | Naikrovek | you can write the initial cost of the tool and the depreciation off on taxes |
| 14:58.55 | Naikrovek | if you're in the US anyway |
| 14:58.57 | Bakey | Naikrovek, yes I'm familiar with VMs |
| 14:59.08 | Bakey | I have a 3 node vSphere 4 cluster |
| 14:59.31 | p3nguin | I'm not much of an accountant. I'd prefer not to have to depreciate assets. |
| 14:59.44 | Naikrovek | Bakey: okay so download AsteriskNOW and put it on a VM. install it with and without FreePBX |
| 14:59.58 | Naikrovek | p3nguin: they're going to depreciate whether you account for it or not. |
| 15:00.46 | Bakey | Naikrovek, I guess what I don't understand is where I get dial tone from? |
| 15:01.04 | Naikrovek | on a voip phone, usually it's the phone itself that generates dial tone |
| 15:01.11 | p3nguin | bakey: If you have SIP phones, they produce a dialtone when you connect them to the call control device. |
| 15:01.24 | Bakey | so I could hook up 2 local phones and make internal calls |
| 15:01.25 | Naikrovek | but if your'e asking where the phone service comes from you can get a "sip trunk" from places for very cheap |
| 15:01.28 | Naikrovek | Bakey: yes |
| 15:01.31 | Bakey | but there's no way to test an external one |
| 15:01.31 | p3nguin | bakey: you could, yes. |
| 15:02.02 | p3nguin | bakey: What country are you in? |
| 15:02.03 | Naikrovek | you can connect your test system to voip service providers (ITSPs) often for free for a small time |
| 15:02.06 | Bakey | US |
| 15:02.15 | Naikrovek | ooh, I'm in the US. what state |
| 15:02.20 | Bakey | Alabama |
| 15:02.22 | Bakey | Birmingham |
| 15:02.51 | p3nguin | Do you want to have an unmetered service where you pay a flat rate every month, or do you want to pay a per-minute rate on calls? |
| 15:03.35 | *** join/#asterisk pgarcia (~chatzilla@yoda.kanatek.com) |
| 15:03.46 | Naikrovek | someone needs to maybe call him and do this over the phone |
| 15:03.48 | p3nguin | Do you have both inbound and outbound calling, or mostly inbound, or mostly outbound? |
| 15:03.55 | Naikrovek | fingers cant' type as fast as mouths can ask and answer questions |
| 15:04.12 | Bakey | lol, both p3nguin |
| 15:04.20 | Bakey | I have a question up on reddit/r/sysadmin |
| 15:04.23 | Bakey | and over on Spiceworks |
| 15:04.27 | p3nguin | Now you skipped a question. |
| 15:04.33 | krion | leifmadsen: thanks |
| 15:04.38 | Bakey | and unmetered I think |
| 15:04.48 | p3nguin | So you have LOTS of calls. |
| 15:04.51 | p3nguin | or will have. |
| 15:04.52 | Bakey | yes |
| 15:05.00 | Bakey | because we're so distributed |
| 15:05.10 | Bakey | we have multiple conference bridges a day |
| 15:05.34 | p3nguin | Okay, so you need a good unmetered flat rate service. naikrovek might have the answer for an ITSP for you. |
| 15:05.39 | Bakey | is downloading AsteriskNOW |
| 15:06.03 | Naikrovek | well |
| 15:06.03 | pgarcia | Hi, I'm using Asterisk 1.4, and I'm testing using SIP INFO messages as dtmfmode. My phone is set to use INFO only, as my sip.conf. When the phone calls asterisk, it doesn't detect the DTMFs. Looking at the SIP debug, I've noticed that the Non-codec capabilities (dtmf) shows that asterisk has nothing configured. |
| 15:06.25 | Naikrovek | if he's giving service to those 65 phones from his asterisk system, those conf bridges won't cost him anything |
| 15:06.33 | pgarcia | anybody knows why? checking 'sip show settings' shows me that dtmfmode is INFO. Any idea? |
| 15:06.44 | [TK]D-Fender | pgarcia: Maybe you should be dumping your peer from CLI, pastebinning your configs, and showing us this SIP DEBUG....... |
| 15:06.48 | p3nguin | pgarcia: Consider using a better dtmfmode. |
| 15:07.31 | pgarcia | p3nguin, my default is rfc2833, but I have to test all modes |
| 15:08.07 | pgarcia | I just don't understand why asterisk side shows no dtmfmode in the SIP negotiation.... |
| 15:08.56 | pgarcia | my ip phone is in the same network/provider and can use INFO. Asterisk SIP logs shows that the phone is asking for INFO, but asterisk cannot do it. |
| 15:09.41 | *** join/#asterisk DelphiWorld (~Delphi@41.200.12.252) |
| 15:09.52 | p3nguin | bakey: How many calls will be on/from the PSTN? That's the amount of calls you need to worry about paying for. Calls to/from phones all connected to your Asterisk system don't cost you. |
| 15:10.04 | DelphiWorld | hi |
| 15:10.12 | Bakey | p3nguin I'll have to do some digging there |
| 15:10.22 | Bakey | not sure how to track that with our current solution |
| 15:10.26 | *** part/#asterisk DelphiWorld (~Delphi@41.200.12.252) |
| 15:10.34 | Bakey | (CommPartners VOIP) |
| 15:10.54 | pgarcia | is it possible to combine options in dtmfmode, like inband|info, for example? |
| 15:10.58 | p3nguin | bakey: Your 80 people could all talk to each other from their IP phones without having any phone service provider involved. |
| 15:12.06 | Bakey | p3nguin, right. I'm not sure how to track those outside calls though. Have to dig around and see what that volume actually looks like |
| 15:12.49 | p3nguin | bakey: You can't even guess how many calls are to or from people outside of your organization? |
| 15:13.19 | Bakey | p3nguin, how could I? Like I said, most of our users work from home. Who knows what they do all day with their phones. |
| 15:13.52 | p3nguin | bakey: Some people can guess based on the nature of the business. |
| 15:14.44 | Bakey | p3nguin, I'm not comfortable recommending a solution based on a swag, I'd like to dig around and see actual numbers. ;) Maybe I'm overly cautious? |
| 15:15.16 | p3nguin | bakey: I was only curious for the purpose of this brainstorm session. |
| 15:15.27 | Bakey | oh |
| 15:15.30 | *** join/#asterisk lauris (~la@unaffiliated/lauris) |
| 15:15.53 | p3nguin | bakey: You'll ultimately make your decisions based on your calculated needs. |
| 15:16.00 | Bakey | are we talking concurrent calls? |
| 15:16.05 | Bakey | or volume per day? |
| 15:16.50 | p3nguin | I'm only trying to find out if you have a lot of "outside calls" going on. |
| 15:16.58 | lauris | hi, i'm looking for 1xFXO port SIP-compatible device solution |
| 15:17.05 | Bakey | I'd say around 100ish per day |
| 15:17.09 | p3nguin | If all your calls are "inside calls," then you don't need a phone company at all. |
| 15:17.15 | Bakey | probably no more than 15-20 on concurrently at any given time |
| 15:17.18 | lauris | any recommendations ? |
| 15:17.56 | Naikrovek | ATA |
| 15:18.05 | lauris | which one ? |
| 15:18.29 | Naikrovek | excellent question. I have no idea. Usually the one recommended in here is a Linksys model, but I dunno the model number. |
| 15:18.35 | lauris | i found that SPA3102 has 1 FXO, but i'm not sure whether it's suitable in my case |
| 15:18.38 | p3nguin | SPA-3102 |
| 15:18.48 | p3nguin | What is your case? |
| 15:18.48 | bougyman | ew |
| 15:18.52 | bougyman | worst. device. ever. |
| 15:19.26 | Naikrovek | ever? wow |
| 15:19.28 | Naikrovek | that's a huge claim |
| 15:19.47 | bougyman | it has a bunch of fatal bugs in the firmware. |
| 15:20.00 | lauris | p3nguin, usualy those ATAs have FXO only for the purpose of "live line" so that FXS port could call through FXO |
| 15:20.01 | bougyman | i had to work around some for a while before I replaced them with a pci FXO card. |
| 15:20.05 | bougyman | http://gitorious.org/spa3102-invite-packet-scrubber < the workaround. |
| 15:20.23 | lauris | but what i'm looking for is ATA which would register FXO port as seperate peer |
| 15:20.25 | bougyman | even linksys/cisco engineers are mad that fix hasn't made it into the firmware, it's been known over a year and a half. |
| 15:20.35 | lauris | so that i could both terminate and answer calls from telco |
| 15:20.35 | p3nguin | lauris: That's what the SPA-3102 does. |
| 15:20.52 | p3nguin | It has a peer for both ports. |
| 15:20.52 | lauris | not only through fxs line |
| 15:20.55 | lauris | but also through asterisk |
| 15:21.11 | lauris | is it capable of doing this ? |
| 15:21.16 | p3nguin | yes |
| 15:21.20 | bougyman | somewhate. |
| 15:21.28 | p3nguin | That's exactly how I use mine. |
| 15:21.50 | lauris | boygyman said it's worst device ever :) |
| 15:21.55 | bougyman | so long as you don't mind the echo and lack of inbound caller ID (without using a workaround to fix inbound caller id name packets) |
| 15:21.56 | lauris | bougyman, can you elaborate on that ? |
| 15:22.08 | lauris | clid is not an issue |
| 15:22.13 | p3nguin | Huh, my caller ID works just fine. |
| 15:22.20 | p3nguin | I'm not in India, though. |
| 15:22.34 | bougyman | nor were my installs. |
| 15:22.36 | lauris | my main concern is whether i'll be able to accept call on FXO |
| 15:22.44 | bougyman | nor the cisco engineers complaining about the same issue. |
| 15:22.48 | p3nguin | lauris: It does. |
| 15:22.49 | lauris | and then dial to soft client |
| 15:22.50 | p3nguin | lauris: That's what its purpose is. |
| 15:22.56 | p3nguin | IT DOES THAT! |
| 15:22.59 | lauris | thanks! |
| 15:23.00 | lauris | :) |
| 15:23.17 | bougyman | http://homecommunity.cisco.com/t5/VoIP-Adapters/SPA-3102-sending-bad-quot-From-quot-field/m-p/229531 |
| 15:23.26 | p3nguin | VoIP-to-PSTN gateway and PSTN-to-VoIP gateway |
| 15:23.39 | bougyman | two year old fatal bug still in the firmware. |
| 15:23.51 | lauris | i thought that FXO port is limited for the use from FXS port only |
| 15:24.12 | p3nguin | no |
| 15:24.33 | lauris | but if it registers two ports as seperate peers, everything is great :) |
| 15:24.46 | [TK]D-Fender | [11:21]<bougyman>so long as you don't mind the echo and lack of inbound caller ID (without using a workaround to fix inbound caller id name packets) <- I never had CID issues, and Echo isn't a set problem... some people have more issues than others. Mine was fine... |
| 15:24.52 | lauris | and what about echo ? |
| 15:25.01 | [TK]D-Fender | luaYMMV <- |
| 15:25.03 | p3nguin | I use mine primarily to connect a phone (on the FXS port) to use with Asterisk, but I have a dialplan on it to send certain calls out the FXO port onto the PSTN. |
| 15:25.06 | bougyman | [TK]D-Fender: as noted above, the CLID is a known issue, the spa sends a broken packet. |
| 15:25.08 | [TK]D-Fender | lauris: YMMV <- |
| 15:25.14 | bougyman | [TK]D-Fender: if you've never had a problem, you've just never noticed. |
| 15:25.24 | Naikrovek | another huge claim |
| 15:25.28 | bougyman | my customers noticed for me, there were no signs in asterisk or freeswitch of the failure. |
| 15:25.32 | Naikrovek | someone ate his wheaties |
| 15:25.35 | [TK]D-Fender | bougyman: This was on older firmwares mind you... have touched in 2 years |
| 15:25.41 | lauris | thnx again, i'll buy it |
| 15:25.42 | p3nguin | I also sometimes send calls from the PSTN into Asterisk with it. |
| 15:25.44 | bougyman | Naikrovek: i just sent the documentation of the problem in the cisco.com forum. |
| 15:25.50 | Naikrovek | i believe you |
| 15:25.53 | bougyman | it's known, it's documented, it's unfixed. |
| 15:25.55 | Naikrovek | it's not that |
| 15:25.56 | p3nguin | I also sometimes send calls from the PSTN straight to the phone. |
| 15:26.02 | p3nguin | THIS IS WHAT THE DEVICE IS MADE TO DO. |
| 15:26.08 | [TK]D-Fender | Naikrovek: From what I saw his point is legit.... much newer mind you. |
| 15:26.31 | Naikrovek | okay so get the spa from ebay not the store |
| 15:27.00 | p3nguin | I got mine on ebay. It works fine. |
| 15:27.07 | p3nguin | No, I lied. I got mine on Amazon. |
| 15:27.09 | lauris | not much of a price diff |
| 15:27.14 | p3nguin | $70 |
| 15:27.15 | bougyman | i wish i could flash earlier firmware on it. |
| 15:27.17 | lauris | ebay compared to local distributor |
| 15:27.22 | bougyman | i don't even know the point of firmware that works. |
| 15:27.33 | p3nguin | I have a 3.something firmware on mine. |
| 15:27.42 | bougyman | i know all 5.1+ is broken. |
| 15:27.44 | Docteh | the older ones were better? dang mine has a cisco logo on it |
| 15:28.00 | bougyman | you have 4 year old firmware? |
| 15:28.04 | bougyman | or 5 year old? |
| 15:28.21 | Docteh | spa3102 with 3.3.6(gw) |
| 15:28.43 | lauris | i'm now browsing cisco's website |
| 15:28.48 | p3nguin | 3.3.6(GW) |
| 15:28.50 | lauris | there is 5.1.10 firmware for this device |
| 15:28.52 | p3nguin | That's what I have. |
| 15:28.59 | bougyman | lauris: yes, 5.1 is the broken one. |
| 15:29.06 | lauris | dated 19/Apr/2010 |
| 15:29.07 | p3nguin | 3.3.6(GW) works fine for me. |
| 15:29.07 | lauris | broken ? |
| 15:29.09 | bougyman | if you can find one like these fellas have maybe you can avoid the brokeness. |
| 15:29.23 | lauris | how exactly broken ? |
| 15:29.53 | bougyman | sends a broken SIP packet if the CallerID(name) has any special character in it. |
| 15:30.07 | bougyman | the sip stack rejects it without sending it to the sip application, in most stacks. |
| 15:30.25 | bougyman | if you don't use callerid, you just have to adjust for the echo |
| 15:30.33 | bougyman | its echo cancel and adaptive echo cancel are horrible. |
| 15:30.43 | bougyman | most sites recommend turning them off. |
| 15:31.07 | bougyman | with a lot of adjustment i got it just barely tolerable quality-wise, but long calls would still get echo. |
| 15:31.13 | *** join/#asterisk Mhaddog (~Mhaddog@z65-50-116-17.ips.direcpath.com) |
| 15:31.21 | lauris | yes, i plan to use it for long calls |
| 15:32.01 | lauris | i'd ilke to use my flatrate home line for terminating calls to home country when i'm abroad |
| 15:32.06 | lauris | like* |
| 15:32.37 | p3nguin | That's sort of what I do with mine. |
| 15:32.56 | p3nguin | I use it to make local calls on POTS via Asterisk. |
| 15:33.21 | bougyman | lauris: don't bother buying one with 5.1, i have a boxful. |
| 15:33.25 | lauris | but you can also answer calls on POTS and then process them through Asterisk, right ? |
| 15:33.36 | bougyman | will sell for 1/2 what you can find them for anywhere else. |
| 15:33.41 | p3nguin | lauris: That's correct. |
| 15:33.51 | lauris | bougyman, no way to downgrade it ? |
| 15:34.03 | bougyman | not accd to any docs i've seen thus far. |
| 15:34.08 | bougyman | i've seen some people brick theirs trying to. |
| 15:34.09 | bougyman | :( |
| 15:34.15 | p3nguin | lauris: Mine actually rings the connected phone for a designated amount of seconds before sending the call into Asterisk over the Ethernet port. |
| 15:34.45 | lauris | you mean connected phone through FXS ? |
| 15:34.54 | p3nguin | That gives the person seitting at the phone a chance to answer first. |
| 15:35.01 | p3nguin | yes |
| 15:35.04 | lauris | that's not an issue in my case |
| 15:35.20 | [TK]D-Fender | lauris: The FXO & FXS ports on the SPA work independent of each other. |
| 15:35.29 | p3nguin | That's what I've been saying. |
| 15:35.33 | [TK]D-Fender | lauris: And * can do whatever it wants with either |
| 15:35.40 | lauris | p3nguin, is timeout period configurable ? |
| 15:35.48 | p3nguin | lauris: sure |
| 15:36.08 | [TK]D-Fender | lauris: You can also set the device to pass calls through direct based ont he # dialed, etc. The FXO & FXS also bridge on power failure |
| 15:36.16 | [TK]D-Fender | lauris: Wich is a very good thing |
| 15:37.58 | lauris | great. |
| 15:38.31 | lauris | is there an alternative for |
| 15:38.31 | lauris | spa3102 which does not face problems mentioned above ? |
| 15:38.36 | Bakey | is installing AsteriskNOW |
| 15:38.58 | Kobaz | asterisklater |
| 15:39.20 | *** join/#asterisk bmoraca_work (~bmoraca@66-242-174-254.ceres.bvn.net) |
| 15:39.24 | p3nguin | lauris: Is your Asterisk system at the location where your phone line comes in? |
| 15:39.45 | lauris | i don't have it yet but i have a home server where i can install it |
| 15:40.20 | [TK]D-Fender | Nothing vaguely close in price range that isn't shit for other reasons. The CID issue only occurs on ones with special chars which may represent only the smallest fraction of calls you get. The echo is a VARIABLE issue and you may have no problems at all. |
| 15:40.54 | lauris | there will be no CIDs with special char, i'm sure about this |
| 15:41.07 | lauris | telco send either national or international numbering plan |
| 15:41.11 | bougyman | for home use it may in fact be fine. |
| 15:41.22 | lauris | but echo could be the issue |
| 15:41.23 | bougyman | we had ours installed in small office settings (lawyers mostly) |
| 15:41.28 | [TK]D-Fender | lauris: I'd say grab one for what its worth |
| 15:41.58 | lauris | i will, local distributor have it for 74$ |
| 15:42.54 | Docteh | so are the newer spa3102's coming with the 5.x firmware? |
| 15:43.16 | lauris | it's not written in product description |
| 15:43.20 | [TK]D-Fender | lauris: http://www.telephonydepot.com/Catalog/Cisco-Analog-Adapters/Linksys-SPA3102 |
| 15:43.38 | lauris | [TK]D-Fender, i'm outside US |
| 15:45.21 | lauris | S/H+customs will cost me more than buying this device at local distributor |
| 15:46.00 | [TK]D-Fender | lauris: OK |
| 15:46.11 | lauris | ok, next question :) |
| 15:47.24 | lauris | what device would you recommend if i need GSM instead of FXO |
| 15:47.39 | Naikrovek | ............ |
| 15:47.42 | Naikrovek | lol |
| 15:47.45 | lauris | :-) |
| 15:47.54 | Naikrovek | chan_bluetooth |
| 15:48.03 | lauris | are you serious ? :-) |
| 15:48.22 | Naikrovek | are you asking because "wow there's a way to do that?!" or are you asking because "this dude is a fool." |
| 15:48.35 | Naikrovek | either way, I guess I'm serious |
| 15:48.38 | lauris | because of 'wow!' |
| 15:48.38 | lauris | :) |
| 15:48.57 | lauris | ok, i'll google on this topic my self |
| 15:49.04 | Naikrovek | yes, chan_bluetooth is real |
| 15:49.08 | Naikrovek | and yes it works |
| 15:49.09 | lauris | http://www.voip-info.org/wiki/view/Asterisk+Bluetooth+channels |
| 15:49.11 | lauris | niceee.... |
| 15:49.11 | Naikrovek | dunno hwo supported it is |
| 15:49.26 | lauris | will test it right now :) |
| 15:49.42 | *** join/#asterisk wcselby (~wcselby@pdpc/supporter/professional/wcselby) |
| 15:49.45 | wcselby | o/ |
| 15:49.49 | Naikrovek | but I think it would be cool to bring cell phone home, put it on charger, within reach of blue tooth module on home asterisk server, then have all the phones ring when a call comes in |
| 15:50.02 | lauris | i have an old nokia for this purpose |
| 15:50.15 | Naikrovek | and be able to make a call over the cell from any of those phones |
| 15:50.19 | lauris | which i no longer use |
| 15:50.35 | Naikrovek | nokia has that neat popport connector |
| 15:50.45 | Naikrovek | i remember a product sold that did what i just described for nokia phones |
| 15:51.01 | Naikrovek | plug your nokia into this thing and plug that thing into your home phone wiring |
| 15:51.03 | Naikrovek | done. |
| 15:51.32 | Naikrovek | well they used to have that popport connector |
| 15:51.37 | Naikrovek | new ones don't have it i don't think |
| 15:52.00 | lauris | it's nokia 6021 |
| 15:52.19 | lauris | i like the idea of using BT for this purpose |
| 15:55.00 | *** join/#asterisk rrb3942 (~rbullock@208.34.105.161) |
| 15:56.43 | *** join/#asterisk drudge` (tacos@unaffiliated/drudge/x-837452) |
| 15:57.41 | *** part/#asterisk dandre (~daniel@ble59-2-81-56-122-47.fbx.proxad.net) |
| 15:59.34 | *** join/#asterisk jaskew (~jdaskew@netblock-75-79-182-110.dslextreme.com) |
| 16:04.55 | Bakey | head asplodes. |
| 16:05.38 | Bakey | asteriskNOW installed, time to slog through some documentation |
| 16:05.46 | Qwell | documewhatnow? |
| 16:05.52 | tzanger | Qwell: haha |
| 16:06.36 | Qwell | I need a mini army. Who wants to help me get my name back? |
| 16:06.42 | Qwell | Time for an assault. |
| 16:06.45 | tzanger | your name? |
| 16:06.48 | Qwell | my name. |
| 16:06.49 | Qwell | qwell.com |
| 16:06.58 | leifmadsen | Qwell: I'm in! |
| 16:06.58 | thehar | me me |
| 16:07.00 | Qwell | jerks trademarked it. |
| 16:07.02 | thehar | pick me! |
| 16:07.11 | leifmadsen | I will donate $20 to the fund |
| 16:07.19 | Qwell | I'm going to work on a plan with Google this weekend. Stay tuned! |
| 16:08.07 | leifmadsen | tunes in |
| 16:08.13 | *** join/#asterisk BenC[UK]_ (~bencummin@cpc1-lock3-2-0-cust367.6-1.cable.virginmedia.com) |
| 16:08.52 | *** join/#asterisk Defraz (~Defraz@corp.fuzecore.com) |
| 16:08.59 | tzanger | hmm, qwell.com looks like something easily done with asterisk... you sure you didn't set it up in your sleep? |
| 16:09.15 | Qwell | exactly. |
| 16:09.23 | Qwell | I call prior use. |
| 16:10.11 | leifmadsen | Qwell: heck ya prior use |
| 16:10.15 | leifmadsen | I'll testify! |
| 16:10.29 | Qwell | tzanger: Want to play my lawyer on TV? |
| 16:18.55 | Naikrovek | well, qwell, the Finders Keepers Treaty of 1859 is going to be working against you, here. |
| 16:19.34 | chazzam | why are you holding peoples phones Qwell ? |
| 16:19.50 | Qwell | chazzam: Hostage. |
| 16:20.10 | chazzam | you should just make your on/off button wipe their device using exchange |
| 16:20.18 | tzanger | do I get to wear an eye patch? |
| 16:20.25 | Qwell | tzanger: if'n you'd like |
| 16:20.36 | tzanger | <presses fingertips together> excellent |
| 16:24.14 | *** join/#asterisk ritzt3ch (~ritzt3ch@ip70-189-221-21.lv.lv.cox.net) |
| 16:24.59 | ritzt3ch | im curious on how sip providers do it ( do the somehow interface to level-3 with some sort of SS7 style / Softswitch ? |
| 16:27.56 | p3nguin | do it? DO WHAT? |
| 16:28.08 | TobSnyder | when using external conference system I have to enter a pin - but some digits are doubled. e.g. the pin is 071641 and it received 07166411 |
| 16:28.42 | TobSnyder | outgoing call is via ISDN, my telephone set is an analog telephone set connected via Astribank (Xorcom) |
| 16:30.28 | talntid | qwell, you don't like my website? :\ |
| 16:30.39 | TobSnyder | could this be a configuration issue on my box (have heared from somethin like relaxdtmf IIRC??) |
| 16:30.47 | talntid | i needed a catchy name. |
| 16:33.03 | talntid | i should also note, that you are the metaphorical secretary for hundreds of users... |
| 16:38.49 | drmessano | First, you misspell "talented", then you steal Qwell's good name for commercial endeavor |
| 16:38.55 | drmessano | I am so calling Stallman |
| 16:39.49 | talntid | i misspelled talented, because i race an Eagle Talon, and talented would not fit on the license plate... I had to make sacrifices. |
| 16:40.45 | Naikrovek | ritzt3ch: SS7 is the protocol used within telcos. voip providers bridge that SS7 stuff to SIP and sometimes AIX |
| 16:40.55 | talntid | and hey, I she said she was 18! |
| 16:41.01 | drmessano | Naikrovek, or IAX |
| 16:41.01 | talntid | she told me she was! |
| 16:41.04 | TobSnyder | when using exten => 972,1,Dial(DAHDI/r1/01803001179,180,D(w0w7w1w6w4w1)) it works |
| 16:41.07 | Naikrovek | yes IAX sorry |
| 16:41.31 | *** join/#asterisk visik7 (~Adium@unaffiliated/visik7) |
| 17:02.37 | TobSnyder | if someone got any hint on my DTMF Problem: please let me know |
| 17:02.46 | TobSnyder | I am only again soon ;) |
| 17:03.31 | TobSnyder | /s/only/online |
| 17:04.25 | p3nguin | Oh, so close! |
| 17:04.42 | p3nguin | s/so/you were so/ |
| 17:05.19 | ritzt3ch | So SS7 used within telcos what devices would it be called to as the vOIP Providers use / Softswitches / Freeswitch ? |
| 17:05.53 | ritzt3ch | or are they often refered to as Class 5 switches |
| 17:07.20 | *** join/#asterisk Ad-Hoc (~nimbus@62.1.109.144.dsl.dyn.forthnet.gr) |
| 17:07.30 | TobSnyder | s/only/online |
| 17:07.45 | TobSnyder | whatever ;) |
| 17:07.48 | TobSnyder | see u later |
| 17:08.28 | *** join/#asterisk fraudory (d56be9a9@gateway/web/freenode/ip.213.107.233.169) |
| 17:08.50 | p3nguin | later |
| 17:08.59 | p3nguin | s/later/see ya/ |
| 17:09.00 | fraudory | Hi all - I've got a question regarding asterisk and failover configs that I'm hoping someone may be able to advise me on? |
| 17:09.32 | p3nguin | fraudory: Ask your question or no one will help you. |
| 17:10.51 | fraudory | I've currently got two identical asterisk setup running on two separate servers, on two separate networks. One of which I want to use as the 'main' server, and second as a failover for calls to be routed through in the event the main one were offline. I've investigated setting up dns srv records, which seem to work... but not all softphones seem to play nicely with? |
| 17:11.02 | fraudory | Is there a better way I could be looking at approaching this? |
| 17:16.07 | *** join/#asterisk jaxyeh (~jaxyeh@c-69-250-52-161.hsd1.md.comcast.net) |
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| 17:37.50 | Katty | herroes |
| 17:38.03 | carrar | ARRO! |
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| 18:08.15 | ANurmi | I have uncommented the "context=incoming" line in chan_dahdi.conf and have added a section to the extensions.conf, saved both files and reload dahdi and the dialplan, but upon calling the pstn from my cell phone it is still going to the default context, are there any steps i'm missing? |
| 18:08.47 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 18:08.47 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 18:10.08 | ANurmi | Does hierarchy matter in the extensions.conf since default is directly above incoming context? |
| 18:10.30 | p3nguin | In that case, it should not matter. It does matter when you start doing includes, though. |
| 18:10.32 | citywok | position in the file doesn't matter |
| 18:10.43 | citywok | ^^ what he said |
| 18:11.12 | ANurmi | i'm strictly doing testing playbacks and echos in the incoming context. |
| 18:11.42 | p3nguin | Then it won't matter if it is at the top, bottom, or anywhere in the middle. |
| 18:12.36 | p3nguin | pbx_config probably puts them in order anyway. |
| 18:12.54 | p3nguin | alpha-numeric ordering |
| 18:14.29 | ANurmi | what is the command to turn down verbosity on the CLI? |
| 18:14.55 | p3nguin | core set verbose <new level> |
| 18:15.03 | *** join/#asterisk Cain` (~Geek@unaffiliated/cain) |
| 18:15.08 | p3nguin | core set verbose 0 for example |
| 18:15.17 | citywok | or asterisk -rvvvvvv |
| 18:15.18 | ANurmi | perfect |
| 18:16.33 | ANurmi | I am also getting this output when dialing in "Starting DAHDI/5-1 at DID_trunk_1,s,1 failed so falling back to exten 's'" |
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| 18:16.56 | p3nguin | That seems like the same thing. |
| 18:17.14 | p3nguin | Starting at s, it fails, so fall back to s. |
| 18:17.48 | p3nguin | DID_trunk_1 is the context, by the way. |
| 18:17.59 | ANurmi | followed immediately by " Starting DAHDI/5-1 at DID_trunk_1,s,1 still failed so falling back to context 'default'" |
| 18:18.10 | p3nguin | There's the problem. |
| 18:18.35 | ANurmi | do I need to define the 's' extension? |
| 18:18.48 | p3nguin | dialplan show DID_trunk_1 |
| 18:19.07 | p3nguin | s certainly isn't magic. If it doesn't exist, you can't use it. |
| 18:19.22 | ANurmi | where do I creat a reference to that/ |
| 18:19.29 | p3nguin | extensions.conf |
| 18:19.48 | p3nguin | run "dialplan show DID_trunk_1" though. |
| 18:19.53 | p3nguin | I bet it doesn't even exist. |
| 18:19.58 | [TK]D-Fender | ANurmi: GUI's are not supported in here. Please use #asterisk-gui and realize that it is UNMAINTAINED and help will be sparse |
| 18:20.15 | ANurmi | There is no existence of 'DID_trunk_1' context |
| 18:20.16 | ANurmi | Command 'dialplan show DID_trunk_1' failed. |
| 18:20.34 | p3nguin | Herein lies the problem you've described. |
| 18:20.49 | p3nguin | Create the context. Create the appropriate exten. |
| 18:21.18 | *** join/#asterisk KavanS (~KavanS@unaffiliated/kavans) |
| 18:23.46 | ANurmi | thanks p3nguin, I'll have to do a bit more studying. |
| 18:24.00 | p3nguin | ~book |
| 18:24.00 | 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 |
| 18:24.20 | p3nguin | The info on how to create contexts and extensions is in The Book. |
| 18:24.28 | ANurmi | oh yes, its open on the desktop currently |
| 18:25.14 | p3nguin | Start looking at page 119. |
| 18:27.56 | *** part/#asterisk jsgoecke (~Adium@c-98-207-5-162.hsd1.ca.comcast.net) |
| 18:30.40 | *** join/#asterisk jsgoecke (~Adium@c-98-207-5-162.hsd1.ca.comcast.net) |
| 18:31.15 | jsgoecke | How does one call a extensions function over AGI, like this one in particular? http://www.voip-info.org/wiki/index.php?page=Asterisk+func+sip_header |
| 18:31.22 | jsgoecke | Clear in the extensions.conf, but not over AGI |
| 18:35.01 | [TK]D-Fender | jsEXEC <- |
| 18:35.05 | *** part/#asterisk dan__t (~dant@vpn.withparity.net) |
| 18:35.07 | [TK]D-Fender | jsgoecke: EXEC <- |
| 18:35.30 | jsgoecke | Let me try again, trying so many combos maybe I left out the obvious one |
| 18:37.58 | jsgoecke | So how would that look here? http://www.voip-info.org/wiki/index.php?page=Asterisk+func+sip_header |
| 18:38.06 | jsgoecke | EXEC SIP_HEADER⦠? |
| 18:38.13 | jsgoecke | This is not a normal command |
| 18:38.21 | *** part/#asterisk Bakey-meeting (61411212@gateway/web/freenode/ip.97.65.18.18) |
| 18:38.28 | [TK]D-Fender | jdoe: Exec calls a dialplan app. That isn't one. So call it like you would int he dialplan |
| 18:39.16 | jsgoecke | I tried 'EXEC Set(foo=${SIP_HEADER(headername)}' and no joy |
| 18:39.22 | jsgoecke | Let me play a bit more |
| 18:41.53 | jsgoecke | Yeah, just not sure how best to do it⦠http://gist.github.com/617293 |
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| 18:52.12 | jdoe | huh? |
| 18:52.13 | jdoe | oh. |
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| 19:00.50 | *** join/#asterisk DrDamnit (~michael@173-165-161-161-atlanta.hfc.comcastbusiness.net) |
| 19:01.20 | DrDamnit | I run the SVN versions of asterisk. core show version gives me the svn version. how do I know if it is 1.4,1.6,or 1.8? |
| 19:01.44 | citywok | what svn trunk did you check it out of? lol |
| 19:01.52 | p3nguin | It doesn't say the version number? |
| 19:02.00 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 19:02.10 | DrDamnit | just got trunk, which is 1.6 (I think). Asterisk SVN-trunk-r290939 |
| 19:02.15 | DrDamnit | Just want to confirm. |
| 19:02.37 | jsgoecke | So, just can not figure out the right sequence to tease out this read only variable⦠http://www.voip-info.org/wiki/view/Asterisk+func+sip_header |
| 19:02.40 | jsgoecke | Makes me sad |
| 19:02.53 | citywok | js what are you trying to do? lol |
| 19:03.40 | visik7 | hi anyone using iaxmodem and mgetty? I dunno how to configure mgetty to pickup the call T0:23:respawn:/sbin/mgetty ttyUSB0 -x4 -D /dev/ttyUSB0 works for a real modem but T0:23:respawn:/sbin/mgetty ttyIAX -x4 -r -s 38400 -F /dev/ttyIAX doesn't with iaxmodem |
| 19:03.42 | jsgoecke | Read a SIP Header off of the current channel |
| 19:03.49 | DrDamnit | p3nguin: core show version give me: "Asterisk SVN-trunk-r290939 built by root @ voip on a i686 running Linux on 2010-10-08 18:25:49 UTC" |
| 19:03.50 | jsgoecke | SIPAddHeader works fine |
| 19:03.58 | jsgoecke | Goes on its merry way |
| 19:04.01 | [TK]D-Fender | jsoAnd where do I see that it even exists? |
| 19:04.14 | [TK]D-Fender | jsgoecke: And where do I see that it even exists? |
| 19:04.27 | jsgoecke | Well, should I just try and pull a standard one? |
| 19:04.29 | jsgoecke | Like the To header? |
| 19:05.04 | [TK]D-Fender | jsgoecke: Heathier start at least. Given that you still don't think LOOKING at what you think you should be able to pull is a good idea |
| 19:05.39 | jsgoecke | Well, I see the header on a far side SIP endpoint when I set it, so I know it gets set |
| 19:05.57 | jsgoecke | But, for illustrative purposes I will stick with a standard one |
| 19:06.35 | [TK]D-Fender | jsgoecke: What would be illustrative would be LOOKING AT THE CALL |
| 19:06.48 | jsgoecke | https://gist.github.com/55d1cebf63890f1be551 |
| 19:06.56 | jsgoecke | I don't understand what you are driving at? |
| 19:07.54 | jsgoecke | BTW, line 7 is me outputing the value of foobar |
| 19:08.40 | jsgoecke | So this does not work over AGI EXEC SET "foobar=${SIP_HEADER(To)}" |
| 19:08.58 | jsgoecke | Since it sets foobar to the literal value rather than the variable on the channel |
| 19:09.29 | *** join/#asterisk stoffell (~kristof@dD576AE12.access.telenet.be) |
| 19:10.57 | [TK]D-Fender | jsgoecke: Try a NoOp |
| 19:11.40 | [TK]D-Fender | jsgoecke: if nothing parses, then this is a serious oversight |
| 19:14.29 | jsgoecke | Ok, I will try to exec a NoOp and see what happens, although in other contexts the Exec works fine for things like voicemail and the like |
| 19:15.03 | jsgoecke | What is anti-pattern on this one is how you have to set that SIP_HEADER to a variable as opposed to just having it return its value |
| 19:15.47 | Qwell | You don't have to set it. You can "return" it just fine. |
| 19:15.54 | jsgoecke | k, let me try that |
| 19:15.54 | Qwell | Dial(${ANYFUNC(args)}) would work just fine |
| 19:16.14 | jsgoecke | NoOp works fine AGI Script Executing Application: (NoOp) Options: (foobar) |
| 19:16.38 | Qwell | so NoOp the header |
| 19:17.36 | jsgoecke | Adding: -- AGI Script Executing Application: (SIPADDHEADER) Options: (x-ahn-test: rubyrox) |
| 19:17.50 | thehar | 5/win 11 |
| 19:17.52 | *** join/#asterisk bluOxigen (~sfds@unaffiliated/bluOxigen) |
| 19:17.53 | thehar | fail |
| 19:17.56 | thehar | <PROTECTED> |
| 19:18.20 | jsgoecke | Get, not so good: -- AGI Script Executing Application: (NoOp) Options: (${SIP_HEADER(x-ahn-test)}) |
| 19:18.25 | jsgoecke | Literal value again |
| 19:20.22 | Qwell | What version of Asterisk is this? |
| 19:20.47 | jsgoecke | Tried on 1.4 and 1.6 with the same results |
| 19:21.16 | thehar | inhales tomatoe bisque |
| 19:21.19 | thehar | Katty: did you make some? |
| 19:22.33 | jsgoecke | Also, is SIPAddHeader sticking around in 1.6/1.8? Seems vague on that point. |
| 19:22.55 | thehar | hey! i use SIPAddHeader! |
| 19:23.05 | jsgoecke | thehar, cool, how do you read the headers? |
| 19:23.16 | thehar | o change alert-fino |
| 19:23.18 | citywok | i use it as well, but don't ahve any need to read it back. |
| 19:23.27 | thehar | s/fino/info/ |
| 19:23.38 | jsgoecke | yeah, my use case is not to write and then read it |
| 19:23.45 | jsgoecke | But I need to have a method in Adhearsion that does both |
| 19:23.55 | jsgoecke | Writing one worked a charm |
| 19:24.02 | jsgoecke | Reading over AGI is proving troublesome |
| 19:24.15 | jsgoecke | Let me try this |
| 19:27.34 | jsgoecke | No workie either, since indeed SIP_HEADER is a func and not an app http://gist.github.com/617375 |
| 19:31.22 | Qwell | isn't there an AGI thing to just retrieve a variable? |
| 19:32.12 | jsgoecke | yes |
| 19:32.14 | jsgoecke | GET VARIABLE |
| 19:32.19 | Qwell | so use that |
| 19:32.20 | jsgoecke | But SIP_HEADER is not a var, it is a func |
| 19:32.28 | jsgoecke | Tried that too, will show you the result |
| 19:32.31 | Qwell | a func acts the same as a variable |
| 19:32.59 | *** join/#asterisk hfb (~hfb@pool-98-119-37-144.lsanca.dsl-w.verizon.net) |
| 19:33.04 | [TK]D-Fender | Qwell: What he seems to show is that * won't parse vars out of the "Data" yuo pass which is just wrong.... |
| 19:33.16 | jsgoecke | DEBUG agi: >>> EXEC GET VARIABLE "SIP_HEADER(x-ahn-test)" |
| 19:33.16 | jsgoecke | DEBUG agi: <<< 200 result=-2 |
| 19:33.35 | jsgoecke | But I don't get it back (tried To as well) |
| 19:33.45 | jsgoecke | Wait, let me try something else, the EXEC is in there |
| 19:34.23 | *** join/#asterisk rethus (~suther@p5087EEB0.dip.t-dialin.net) |
| 19:34.48 | jsgoecke | Ah, that is what was throwing me off, the extra EXEC |
| 19:34.50 | rethus | i have configured fail2ban like described here in the fail2ban wiki: http://www.fail2ban.org/wiki/index.php/Asterisk |
| 19:34.52 | jsgoecke | Thanks for humoring me on this: |
| 19:34.52 | jsgoecke | DEBUG agi: >>> GET VARIABLE "SIP_HEADER(To)" |
| 19:34.52 | jsgoecke | DEBUG agi: <<< 200 result=1 (<sip:1000@sandite.orl.voxeo.net>) |
| 19:35.01 | rethus | but fail2ban don't ban the hackers. |
| 19:35.18 | citywok | lol rethus. why dont you describe what's failing :P |
| 19:35.32 | Qwell | citywok: It's failing to ban. |
| 19:35.33 | citywok | i'm just going to guess you didn't configure it properly |
| 19:35.38 | Qwell | Sounds like the app does exactly what it says it'll do. |
| 19:35.56 | citywok | lol. it works pretty well on my install, took 5 minutes to get it going. |
| 19:35.57 | rethus | :D |
| 19:36.17 | citywok | one IP has been trying to brute force me for nearly a week and racked up a huge amount of bandwidth (20gb or so) |
| 19:36.28 | rethus | but seems, i'm not the only one with this problem... see the last answer in this wikipage |
| 19:36.56 | rethus | citywok: maybe you can help me a bit. |
| 19:37.12 | citywok | i followed the instructions on the voip-info page adn they worked as described |
| 19:37.32 | rethus | did you something different as it described on the wikipage? |
| 19:37.47 | rethus | citywok: can u post me the link |
| 19:37.52 | citywok | i did exactly what the instructions said on the voip-info instructions for fail2ban. google. |
| 19:38.36 | rethus | http://www.asterisk-user-group.de/wiki/index.php/Fail2Ban_Asterisk-Log-%C3%9Cberwachung |
| 19:38.37 | rethus | this? |
| 19:39.46 | jsgoecke | I am going to update that page with my learnins' |
| 19:39.56 | *** join/#asterisk erinspice (~erin@207.98.195.107) |
| 19:40.48 | citywok | rethus do you know how to use the internet? |
| 19:40.58 | jsgoecke | Oh, and one other thing, if funcs acts as Vars, then this should not be true: |
| 19:40.58 | jsgoecke | [Oct 8 19:40:32] ERROR[9991]: pbx.c:1567 ast_func_write: Function SIP_HEADER cannot be written to |
| 19:41.01 | jsgoecke | Anyway... |
| 19:41.52 | rethus | citywok: hello? i tell you, that i did it like described in the howto... you say it works for you, know i please you to tell me which howto you have used... |
| 19:42.09 | citywok | i already did, twice. |
| 19:42.35 | rethus | you post a link? oh, i didn't see it |
| 19:43.03 | citywok | no, i'm not going to google the link for you. |
| 19:43.48 | rethus | you kidding me? I tell you twice, that i do the same, but my howtos i've found and try didn't work!!! |
| 19:44.19 | rethus | than i only ask you to tell me which version u use, and if u can see whats wrong on the howto which i posted (abov) |
| 19:45.23 | citywok | * 1.6.2, whatever version of fail2ban is in the debian repo |
| 19:46.23 | rethus | thanks, but i mean which version of howto... so that i can reproduce it step by step and found the error on my configuration |
| 19:46.31 | ANurmi | P3nguin: I looked in my extensions.conf and created the [DID_trunk_1] and reloaded the dial plan, tried dialplan show DID_trunk_1 and this is what is responds with "[ Context 'DID_trunk_1' created by 'pbx_config' ] Include => 'DID_trunk_1_default' [pbx_config] -= 0 extensions (0 priorities) in 1 context. =-" and it is still falling back to default. |
| 19:47.09 | p3nguin | anurmi: Now you just need to create the relevant extension under the context. It was looking for extension 's' |
| 19:47.41 | *** join/#asterisk jtrag-AWAY (~jtrageser@c-174-55-74-243.hsd1.pa.comcast.net) |
| 19:49.37 | ANurmi | isn't the extension s to define the "start" |
| 19:49.48 | p3nguin | no |
| 19:50.02 | p3nguin | ~standard extensions |
| 19:50.03 | infobot | [~asterisk standard extensions] http://www.voip-info.org/wiki/view/Asterisk+standard+extensions |
| 19:51.01 | p3nguin | "for dialplans that enter a context with no other extension information" |
| 19:51.14 | p3nguin | Your call was explicitly looking for s, earlier. |
| 19:52.41 | [TK]D-Fender | In his case "start" applies as it looks like an incominf DAHDI channel call |
| 19:52.57 | p3nguin | my mistake |
| 19:54.51 | p3nguin | While the s might be an abbreviation of start, I rarely think of s as anything other than an extension named s. I don't think of it differently than, say, extension 8004444444. |
| 19:55.21 | [TK]D-Fender | ANurmi: As I told you before, GUI's are not supported here, and that one is not even being maintained. You are heading down a very painful path. |
| 19:55.57 | ANurmi | oh i'm working strictly in the raw extension.conf file |
| 19:55.59 | [TK]D-Fender | p3nguin: It used to have more direct implications before the 1.2 intruduces autofallthrough & waitexten, etc |
| 19:56.41 | [TK]D-Fender | ANurmi: I know you're in ti now, but I also know you're using an AsteriskGUI dialplan |
| 19:57.50 | ANurmi | [TK]D-Fender: what would you recommend at this junction. |
| 19:59.24 | ANurmi | If I delete the testing dialplan I created in the GUI will it wipe it from extensions.conf? |
| 19:59.44 | [TK]D-Fender | ANurmi: What are your personal goals? |
| 20:01.17 | *** join/#asterisk bn-7bc (bjarne@2001:470:dc32:0:6233:4bff:fe0e:bc1) |
| 20:01.22 | ANurmi | I have an 8 port Diggium card with 4 PSTNs and 4 physical phones, and a couple other Softphone extensions, I need to have a 3 layer IVR. |
| 20:03.36 | [TK]D-Fender | ANurmi: Ok, this will be your server to play with and administer? |
| 20:05.49 | ANurmi | it will eventually be implemented into a small business environment |
| 20:07.45 | [TK]D-Fender | ANurmi: But you will be the one administering it by yourself? |
| 20:08.08 | ANurmi | yes |
| 20:12.10 | [TK]D-Fender | ANurmi: Are you looking forward to doing more creative things with your server? Special functionality down the road, complex call routing decisions, etc? |
| 20:12.40 | [TK]D-Fender | ANurmi: And what is your deadline to get it functional? |
| 20:14.45 | ANurmi | [TK]D-Fender: Down the road I would like to set up some custom call intervals based on time. Possibly intergrate with Sage ACT for promotional calling purposes. Deadline for functional is not really set, its been a pet project of mine. |
| 20:16.39 | hamakohako | Hi not completely an asterisk question, but here goes |
| 20:16.39 | [TK]D-Fender | ANurmi: Ok, basic functionality a GUI could do for you an let other people administer. Once you get "interesting" with your CRM, etc then you'll want an environment you can control more. FreePBX is the common GUI route but steals most control from you. AsteriskGUI was more flexible and would ahve opened a lot more doors, but is broken and unmaintained. |
| 20:16.44 | hamakohako | Is it normal to get 80% CPU utilization on a 2811, if we have around 50 g729 SIP calls and 20 g711 SIP calls? the bandwidth we get is only around 3Mbps |
| 20:17.13 | *** join/#asterisk bjornts (~Adium@247.62-97-195.bkkb.no) |
| 20:17.19 | [TK]D-Fender | ANurmi: That in mind I suggest you learn * from scratch which will give you a better understanding of the trade-offs you may make down the line. |
| 20:17.59 | [TK]D-Fender | hamakohako: Aer you also recording? Because you aer doing a lot of transcoding so far as it may seem |
| 20:18.13 | *** join/#asterisk TobSnyder (~Tobias@91-64-185-67-dynip.superkabel.de) |
| 20:19.00 | hamakohako | On the asterisk yes, but what i mean about the 80% utilization is the cisco 2811 |
| 20:19.41 | [TK]D-Fender | hamakohako: Ah yes... missed the implication. No clue... try in #cisco :p |
| 20:20.19 | *** join/#asterisk pabelanger (~pabelange@2607:f2c0:a000:166:218:f3ff:fe51:c71) |
| 20:22.06 | hamakohako | ok thanks, already asked them that, they said its probably not normal. |
| 20:28.16 | ANurmi | [TK]D-Fender: Thanks for the input on the situation, how does building asterisk from source vary from the Centos Asterisk Now install, strictly functionally speaking if I wasn't to use the GUI for administration? |
| 20:31.00 | [TK]D-Fender | ANurmi: the fact it's from a package doesn't really matter, and at elast that repo is kept up to date. it isn't a bad bas. |
| 20:31.04 | [TK]D-Fender | base* |
| 20:31.39 | [TK]D-Fender | ANurmi: Either way if you're doing your own configs and not letting a GUI get in the way you can always back them up easy and transplant to any other server base you want to use |
| 20:32.52 | *** join/#asterisk citywok (~citywok@67-134-194-33.dia.static.qwest.net) |
| 20:33.41 | [TK]D-Fender | ok, checkout time, back later... |
| 20:38.59 | *** join/#asterisk Paulius (~Paulius@unaffiliated/paulius) |
| 20:40.53 | *** join/#asterisk bjornts (~Adium@247.62-97-195.bkkb.no) |
| 20:41.13 | Paulius | Hey guys. Got a quick question. I'm using vo-ip at home and the Internet connection isn't always stable. Both SIP and IAX2 have interesting issues. However, Skype works perfectly. Also any other latency sensitive application doesn't have issues. |
| 20:41.28 | Paulius | Is there any other protocol better than IAX2 or SIP for trunking? |
| 20:41.49 | Paulius | I've got access to machines at a datacenter which I can use to trunk the protocol from home back into SIP/IAX2 to the telco. |
| 20:45.21 | adyn | I use a rather shaky connection with IAX2 with no real issues other than when someone decides to run a truck into a telephone pole taking down internet on the entire block for the weekend... |
| 20:45.23 | p3nguin | IAX2 is the only one that supports trunking at all. |
| 20:45.29 | p3nguin | SIP does not support trunking. |
| 20:47.07 | Paulius | adyn: Well with SIP, I used to have like cut-outs. Like it would cut out and back several times a second. |
| 20:47.19 | Paulius | With IAX2, I even have random repeats and segments that are out of order. |
| 20:47.25 | p3nguin | jitter? |
| 20:47.36 | Paulius | So I would hear the person say a word twice, or even hear what they say out of sequence. |
| 20:47.37 | p3nguin | Install a jitter buffer. |
| 20:47.41 | Paulius | p3nguin: Oh... How? |
| 20:47.42 | *** part/#asterisk rethus (~suther@p5087EEB0.dip.t-dialin.net) |
| 20:47.47 | Paulius | Is it in a config or something? |
| 20:48.00 | p3nguin | If you're using sip.conf, the info is in the sample file. |
| 20:48.07 | Paulius | Interesting. |
| 20:48.18 | Paulius | So you're saying to go back to SIP? |
| 20:48.24 | p3nguin | It can likely be found at voip-info.org, too. |
| 20:48.38 | Paulius | Because I heard IAX2 was much better suited for dsl connections because of its buffers and other features. |
| 20:48.56 | Paulius | voip-info.org actually was the thing that made me try IAX2. It made it sound better than a blowjob. |
| 20:49.06 | p3nguin | Since IAX2 supports trunking, you'll save bandwidth when making multiple calls using IAX2. |
| 20:49.20 | Paulius | p3nguin: Well honestly bandwidth is not a concern at this point. |
| 20:49.27 | p3nguin | SIP does not support trunking, so the usage is linear per call. |
| 20:49.30 | Paulius | Gotcha. |
| 20:49.54 | p3nguin | What do you suggest is causing the issues if it's not related to lack of bandwidth? |
| 20:50.22 | Paulius | p3nguin: My DSL line is just screwy. If I run a ping to a nearby IP, it'll be okay, but some packets will come back with 100 or 200ms once in a while. |
| 20:50.34 | Paulius | And that happens when there's zero bandwidth utilization. |
| 20:51.00 | p3nguin | I'd be concerned that VoIP will never work correctly on it and tell the phone company to fix their broken DSL. |
| 20:51.21 | Jasnejac | Paulius: I had the same problem recently - thought it was Annex M, turned out to be a screwed microfilter |
| 20:51.28 | p3nguin | I would certainly test a large jitter buffer to see what the effects are. |
| 20:51.36 | Paulius | Well but that's the thing. Skype works perfectly on my line. Even at times where I'm having a perfect skype call and sturggling to hear the phone call. |
| 20:51.43 | Paulius | Well see, Bell in Canada sucks. |
| 20:52.10 | Paulius | I've been arguing back and forth with the phone company. They keep insisting that it's on my end. Even though I've changed all the cabling and my modem is a few feet away from the demark. |
| 20:52.29 | Jasnejac | have you changed your filters? |
| 20:52.32 | Paulius | They charge $200 if the technician deems it's a problem on my end. And their technicians are clueless. |
| 20:52.36 | Paulius | Jasnejac: I have no filters. |
| 20:52.49 | Paulius | Jasnejac: Demark is going straight to a jack that's connected to the modem. |
| 20:52.51 | p3nguin | There shouldn't be a filter between the modem and the demarc. |
| 20:53.11 | Paulius | I'm using a dryloop. No phones connected on the same "line". |
| 20:53.12 | Jasnejac | ah - sorry, I'll read all the conversation next time :-( |
| 20:53.13 | p3nguin | The filter belongs on the phones. |
| 20:53.30 | Paulius | Jasnejac: Hey man, no worries. I'm just glad that you're trying to help me. :-) |
| 20:53.56 | Paulius | And everything else works perfectly. So I'm trying to see if I can solve this voip issue in Software. |
| 20:54.15 | Paulius | I found the jitter parameters for SIP. |
| 20:54.22 | *** join/#asterisk ruben23 (~RLACUMBA@121.97.111.142) |
| 20:54.25 | Paulius | But now I need to rechange my stuff from IAX to SIP. Fun fun fun. |
| 20:54.38 | p3nguin | If you run a ping or run mtr for a week (or even just a couple days), you see a lot of packet loss? |
| 20:55.05 | p3nguin | There isn't very much to change to switch from IAX2 to SIP. |
| 20:55.27 | Paulius | p3nguin: I get packet loss once in a while. but not too bad. |
| 20:55.36 | p3nguin | Change the setting on the ITSP's portal, change the extensions from IAX2 to SIP. Done. |
| 20:55.41 | ruben23 | hi guys any im getting tons of this on my asterisk logs---------------> http://pastebin.com/NrztQU74 ---> and a part of it all my recording when i played then dont have any audio or sound i can hear at all. |
| 20:56.05 | Paulius | MTR for a day or a week wouldn't be a good test over here. I'm a geek and do quite a bit of uploading or other internet activity that would just naturally cause the ping to be higher than usual. |
| 20:56.09 | citywok | ruben do you have enough g729 licenses? |
| 20:56.38 | p3nguin | Maybe you actually do have a bandwidth problem. |
| 20:56.53 | Paulius | p3nguin: If you saturate the uplink on a DSL, it's normal for the ping to shoot up. |
| 20:56.56 | p3nguin | Try running a compressed codec between your Asterisk and the ITSP. |
| 20:57.14 | chazzam | I think I picked a bad day to find and follow Qwell on twitter |
| 20:57.26 | p3nguin | If you're saturated, it's no wonder call quality sucks. |
| 20:57.40 | Paulius | p3nguin: No, but... The phone calls are bad even when NOTHING is being transferred. |
| 20:57.51 | Qwell | chazzam: heh |
| 20:58.01 | Qwell | chazzam: Don't worry, I won't add you to the stupid-blackberry-users list :D |
| 20:58.06 | p3nguin | Do some testing. Use another codec that doesn't have such a high cost on your line. |
| 20:58.09 | chazzam | yay! |
| 20:58.22 | Paulius | p3nguin: Yeah, I'll look into it. Going to start with the jitter stuff. |
| 20:58.35 | Paulius | Do some SIP codecs handle bad connections better than others or no? |
| 20:58.35 | p3nguin | Switching to gsm would be a lot quicker. |
| 20:59.16 | Paulius | By the way, my line is 5mbit down and 800kbit up. I'm sure that alone can handle a single call on any codec easily. |
| 20:59.34 | p3nguin | I would think so. |
| 20:59.44 | citywok | as long as you aren't torrenting, yea |
| 20:59.57 | Paulius | Once again, I'm troubleshooting while the link is completely quiet. |
| 20:59.58 | citywok | that should handle a couple ulaw calls @ 800kbit |
| 21:00.38 | p3nguin | Even on my 20x2, I still sometimes get some choppiness, since there is no such thing as QoS on residential class cable modem service. |
| 21:01.05 | Paulius | Well the thing that ticks me off is that... I have a proper router (Cisco 2620) and I'm using a WIC-1ADSL as the "modem". |
| 21:01.13 | Paulius | And the 1ADSL doesn't support like the real QOS. |
| 21:01.30 | Paulius | Or the Cisco easyQOS or whatever they call. |
| 21:01.56 | adyn | QoS is a nightmare for me at the moment :/ |
| 21:02.18 | Paulius | p3nguin: http://www.voip-info.org/wiki/view/Asterisk+readme+iax |
| 21:02.26 | Paulius | check out the "why" section. heheh |
| 21:04.22 | p3nguin | Seems pretty accurate from what I read. |
| 21:06.21 | Paulius | Ah the dreaded circuits busy... |
| 21:06.34 | Paulius | Now I remember why it was painful to switch. Required troubleshooting lol |
| 21:07.07 | p3nguin | /: |
| 21:07.26 | p3nguin | It shouldn't be that hard, especially if you've already done it once. |
| 21:07.46 | Paulius | Okay I think I got it. |
| 21:08.19 | p3nguin | You needn't touch the iax.conf to switch over to sip. |
| 21:08.30 | Paulius | I know. And I didn't. |
| 21:08.48 | p3nguin | Just configure the peer in sip.conf, change the IAX2 in your outbound extensions to SIP, and change the setting on the ITSP. |
| 21:08.57 | Paulius | yup. SIP working. |
| 21:09.09 | Paulius | Gotta find out the code for echo test from my telco. |
| 21:11.16 | Paulius | This is weird. Even with the default SIP settings, the call is much much smoother. |
| 21:12.17 | *** join/#asterisk Defraz (~Defraz@corp.fuzecore.com) |
| 21:13.22 | *** join/#asterisk lauris (~la@unaffiliated/lauris) |
| 21:14.49 | *** join/#asterisk bluOxigen (ssf@unaffiliated/bluOxigen) |
| 21:16.06 | Paulius | Yeah, so far so good. |
| 21:16.06 | Paulius | Thanks for the suggestion p3nguin |
| 21:16.13 | Paulius | I'll implement the buffer if I hear some issues. |
| 21:17.09 | *** join/#asterisk bmoraca_work (~bmoraca@66-242-174-254.ceres.bvn.net) |
| 21:17.14 | bmoraca_work | ahhh, slow friday afternoons ftl |
| 21:17.17 | bmoraca_work | er |
| 21:17.18 | bmoraca_work | ftw |
| 21:17.27 | Paulius | indeeed! |
| 21:19.10 | Katty | ohai |
| 21:19.15 | Paulius | hallo |
| 21:19.20 | Katty | i haz teh bored too |
| 21:21.07 | bmoraca_work | Katty: http://forums.anandtech.com/showpost.php?p=30570905&postcount=13 |
| 21:21.16 | bmoraca_work | wonderful scholastic achievement, that |
| 21:21.22 | bmoraca_work | good read, though |
| 21:22.23 | Katty | the best part: |
| 21:22.24 | Katty | This is page 3. |
| 21:22.45 | bmoraca_work | it was impressive that he didn't receive a failing grade on the paper |
| 21:22.55 | bmoraca_work | 61%...that's passing in a lot of schools, now |
| 21:23.04 | bmoraca_work | and that makes me sad :( |
| 21:23.12 | Paulius | Man, I've collected tons of those funny essays on my site. |
| 21:23.20 | Katty | do you think we'll have a very cold, snowy winter this year? |
| 21:24.33 | *** join/#asterisk n3hxs (~HAMming@c-76-29-19-100.hsd1.il.comcast.net) |
| 21:27.31 | Katty | beek: i got a new nail color |
| 21:27.57 | *** join/#asterisk bcrisp (~bcrisp@wsip-184-191-141-38.ph.ph.cox.net) |
| 21:28.08 | Katty | beek: http://polishaddict.com/wp-content/uploads/2008/10/essie-its-genius.jpg |
| 21:28.20 | bcrisp | "Remote host can't match request NOTIFY to call 'blablablabla'. Giving up." - any clues on this? |
| 21:28.23 | Katty | beek: seems more fallish than the bright pink stuff i was wearing :> |
| 21:28.35 | bcrisp | you still got crittercam? |
| 21:29.18 | Katty | bcrisp: yes. |
| 21:30.44 | bcrisp | i might have to set up my fishcam soon |
| 21:30.52 | Katty | :>>> |
| 21:30.56 | bcrisp | i think you'd like my new green spotted puffer |
| 21:32.36 | carrar | Katty, whats the plans this weekend? |
| 21:32.48 | Katty | bcrisp: tjere is a good chance. |
| 21:32.55 | Katty | carrar: going to see the guardians tonight. |
| 21:33.01 | carrar | oh! |
| 21:33.02 | Katty | carrar: look for some new kitchen chairs. |
| 21:33.10 | Katty | carrar: make pineapple black bean enchiladas |
| 21:33.12 | carrar | What kind of chairs?!? PICS!! |
| 21:33.14 | bcrisp | hm? |
| 21:33.14 | Katty | carrar: and possibly see my mother. |
| 21:33.21 | carrar | woah |
| 21:33.24 | carrar | full weekend |
| 21:33.28 | Katty | mhmm |
| 21:33.34 | Katty | carrar: you also missed my birthday |
| 21:33.37 | Katty | carrar: it was yesterday. |
| 21:33.59 | *** join/#asterisk rethus (~suther@p5087EEB0.dip.t-dialin.net) |
| 21:34.08 | carrar | WOAH |
| 21:34.13 | rethus | how can i get the CDR-Variables of a channel? |
| 21:34.14 | carrar | Are you 21 now? |
| 21:34.32 | carrar | HAPPY BELATED BIRTHDAY |
| 21:35.11 | carrar | ok I won't forget next year :) |
| 21:36.26 | Katty | yes. i am 21 now. again. |
| 21:36.27 | Katty | plus 5. |
| 21:36.40 | carrar | heh |
| 21:36.48 | carrar | Just a youngin! |
| 21:37.10 | Katty | mhnmm |
| 21:37.28 | carrar | Van Halen wrote a fun song when you were born |
| 21:37.36 | carrar | err fun album |
| 21:37.39 | Katty | and i intend to enjoy it for as long as i possibly can. |
| 21:38.54 | carrar | http://en.wikipedia.org/wiki/1984_%28Van_Halen_album%29 |
| 21:38.57 | carrar | good album |
| 21:40.01 | carrar | US Bell System is broken up that year |
| 21:40.49 | *** join/#asterisk Gibby13 (~gibby@204.118.10.244) |
| 21:41.32 | Katty | that is a good album |
| 21:41.44 | carrar | haha |
| 21:41.44 | carrar | August 11, 1984 â United States President Ronald Reagan, during a voice check for a radio broadcast remarks, "My fellow Americans, I'm pleased to tell you today that I've signed legislation that will outlaw Russia forever. We begin bombing in five minutes". |
| 21:42.03 | carrar | that would have been funny to hear |
| 21:42.16 | Katty | :< |
| 21:42.27 | Katty | i don't think so |
| 21:42.35 | *** join/#asterisk csnook (~chris@209-6-38-66.c3-0.smr-ubr1.sbo-smr.ma.cable.rcn.com) |
| 21:42.40 | ANurmi | Also the year apple "made sure 1984 wasn't like the moveie" |
| 21:42.41 | carrar | yeah tensions were pretty high at that time |
| 21:43.24 | carrar | speaking of parents |
| 21:43.28 | carrar | mnie are here |
| 21:43.28 | carrar | bbl |
| 21:43.31 | Katty | bai |
| 21:44.52 | *** part/#asterisk Mhaddog (~Mhaddog@z65-50-116-17.ips.direcpath.com) |
| 21:45.23 | Gibby13 | Hey room, i installed asterisknow and did the upgrade to 2.8 for all the modules... I added a Gerneric SIP Device Extension... basically just added the extension number, a display name and secret... I am trying to register a GXP-2020 to that extension, in the GXp-2020 web interface, i am using account 1, add the account name, sip server and outbound proxy as the asterisknow server, the sip user ID and authenticate ID as the extension number and the secre |
| 21:45.24 | Gibby13 | t as the authenticate password... rebooted the phone however it still does not register... have searched the forums and tried a few things but nothing is helping |
| 21:46.22 | Katty | have you tried cookies? |
| 21:47.07 | Gibby13 | yep, i ate whole box |
| 21:47.16 | Katty | excellent. |
| 21:47.19 | citywok | i hope you aren't diabetic. lol |
| 21:47.19 | Katty | did you then try vodka? |
| 21:47.33 | Gibby13 | not yet, still early on 3rd pot of coffee though |
| 21:47.35 | Katty | citywok: they make sugar free cookies too :> |
| 21:47.43 | citywok | ewwwwwwwwww |
| 21:47.50 | citywok | sugar free?! gross. |
| 21:47.53 | Katty | Gibby13: oh, in that case try some irish cream. |
| 21:47.59 | Katty | citywok: they don't taste that bad, dear. |
| 21:48.10 | Gibby13 | that might help me but not the phone |
| 21:48.25 | Katty | Gibby13: what day is it there? |
| 21:48.30 | citywok | hah, really? i :heart: sugar. maybe i won't when i don't have a 25y/o metabolism. |
| 21:48.33 | Gibby13 | friday |
| 21:48.37 | citywok | Gibby13: it's a grandstream, that's the problem. lol |
| 21:48.38 | Katty | Gibby13: what time. |
| 21:48.49 | citywok | Katty: beer-30. |
| 21:48.59 | Katty | beer is revolting. |
| 21:49.03 | citywok | It's 5oclock somewhere! |
| 21:49.20 | Gibby13 | i was told to get a grandstream, said they would autoprovision with dhcp and network boot, however, i can't use it since i have a tftp server for network booting other devices already |
| 21:49.25 | Gibby13 | 17:47 |
| 21:49.37 | citywok | Ewwww who RECOMMENDED a grandstream? |
| 21:49.38 | citywok | ~phones |
| 21:49.39 | infobot | While personal preference will dictate which phone works best for you, general consensus on a rough order of quality and suggestibility is as follows: Polycom (any), Aastra 480i, Aastra 5i Series, Linksys SPA-9XX, Snom, Cisco 79XX, everything else. Do not consider Grandstream phones. Ever. |
| 21:49.40 | Katty | Well perhaps whoever told you would tell you precisely how to do that. |
| 21:50.06 | Gibby13 | any1 want a grandstream? lol |
| 21:50.16 | citywok | hellll no |
| 21:50.16 | Katty | no thank you. |
| 21:50.26 | citywok | i've got a couple, 3 of 8 died |
| 21:50.31 | Katty | but perhaps someone will have a suggestion for you. |
| 21:50.35 | Katty | you might try again in a bit. |
| 21:50.54 | Katty | personally, i can't think straight i'm so hungry. |
| 21:51.04 | Gibby13 | i have cookies! |
| 21:51.20 | Katty | sweets will not do when i'm hungry. |
| 21:51.32 | Gibby13 | pizza and beer? |
| 21:51.42 | Katty | i don't want to get fat. |
| 21:51.53 | Katty | besides, that probably has meat on it. |
| 21:52.05 | Katty | meat tastes funny. |
| 21:52.43 | Gibby13 | lol |
| 21:53.00 | Katty | Gibby13: dinner -> http://www.food.com/recipeprint.do?rid=154388 |
| 21:53.08 | *** part/#asterisk bsaxon (~bsaxon@12.107.149.61) |
| 21:53.43 | Gibby13 | that looks like what comes up after to much beer |
| 21:54.20 | Gibby13 | so what is wrong with grandstream? |
| 21:54.40 | Katty | http://food.sndimg.com/img/recipes/15/43/88/large/picWRGkyz.jpg <- does it now? |
| 21:54.54 | Katty | http://food.sndimg.com/img/recipes/15/43/88/large/picCTN7AG.jpg <- looks pretty good to me. |
| 21:55.04 | thehar | mmmmmmmmmmmmmmmmmmmmm |
| 21:55.12 | p3nguin | ~gs |
| 21:55.12 | infobot | GrandSuck phones & gateways are cheap junk which should be avoided with extreme prejudice. |
| 21:55.19 | thehar | ^_^ |
| 21:55.20 | p3nguin | ~grandstream |
| 21:55.20 | infobot | rumour has it, grandstream is the Yugo of VoIP hardware. Run... Run away now. Though, therealcircut says that they're not that bad. |
| 21:55.31 | Katty | thehar: would you like to come over for dinner |
| 21:55.37 | thehar | Katty: yes yes i would |
| 21:55.45 | Katty | thehar: excellent. be there at 7. |
| 21:55.45 | thehar | i made some salsa the other day that would go excellent with your dinner |
| 21:55.48 | Gibby13 | ugh, need to find a grandstream IRC room |
| 21:55.54 | p3nguin | hahaha |
| 21:55.55 | thehar | give up on grandstream |
| 21:55.59 | thehar | it is absolutely horrid |
| 21:56.02 | Katty | thehar: bring salsa. i have chips. |
| 21:56.05 | citywok | yea, quite awful actually. |
| 21:56.09 | citywok | aastra ftw :P |
| 21:56.11 | thehar | we deployed 300+ HT286's and replaced everyone of them with PAP2Ts |
| 21:56.19 | thehar | do you like salsa verde? |
| 21:56.23 | Katty | i do. |
| 21:56.25 | thehar | i don't make it for 'white people' |
| 21:56.28 | thehar | i made it HOT HOT HOT |
| 21:56.29 | thehar | haha |
| 21:56.32 | citywok | thehar why the hell do you have 300 adapters? |
| 21:56.33 | Katty | oh em gee. |
| 21:56.36 | Katty | give me the recipe. |
| 21:56.46 | thehar | it was for residential subs |
| 21:56.54 | thehar | most horrible beginning of a deployment evar |
| 21:56.56 | thehar | <PROTECTED> |
| 21:56.57 | *** join/#asterisk erinspice (~erin@207.98.195.107) |
| 21:57.03 | Katty | and i hope there is cumin in it. |
| 21:57.12 | thehar | Katty: there is actually |
| 21:57.12 | citywok | haha, gotcha. could you plug one in to the house line and back-feed the other jacks? lol |
| 21:57.18 | Katty | thehar: :>>>>>>>>>>>>>>>>>>>>>> |
| 21:57.24 | thehar | citywok: yeah |
| 21:58.16 | Katty | tonight is gonna suck so bad. |
| 21:58.28 | citywok | bar time in an hour! |
| 21:58.33 | Katty | Midol time in an hour. |
| 21:58.36 | Katty | then bar time. |
| 21:58.40 | thehar | where did i put that recipe! damn it |
| 21:58.41 | Gibby13 | rofl |
| 21:59.05 | Katty | thehar: you wanna email it to me when you find it? |
| 21:59.19 | Katty | thehar: or /query it |
| 21:59.25 | thehar | i have the mocked up one at home |
| 21:59.29 | thehar | the one i've adjusted to my own needs |
| 21:59.33 | Katty | excellent. |
| 21:59.38 | thehar | but i have the base one i started from here somewhere |
| 21:59.40 | Katty | send me that one when you have a few. |
| 22:00.02 | Katty | i'm off. into the wide blue yonder. |
| 22:00.04 | Katty | poofs. |
| 22:00.16 | thehar | remind me later |
| 22:00.18 | thehar | i can't find it |
| 22:03.29 | Gibby13 | looks like my .conf files are not being updated on the asterisk now side |
| 22:05.37 | Gibby13 | and i get nothing when i got 8088/static/config/index.html on my asterisknow server |
| 22:12.40 | *** join/#asterisk The_Boy_Wonder (~vossel@asterisk/batman-developer/dvossel) |
| 22:14.11 | fraudory | Hi - could anybody shed some light on why, when receiving an incoming call from sipgate, all that asterisk displays in the console is "Using SIP RTP CoS mark 5" and the call is never answered? |
| 22:15.02 | citywok | sip set debug on |
| 22:17.40 | *** join/#asterisk drfreeze (~Jim@207.191.114.82) |
| 22:18.04 | drfreeze | Anyone know what the following problem is from? |
| 22:18.04 | drfreeze | 000043.772|cfg |4|00|Edit|Error 0x380003 attempting stat of /ffs0/local/0004f2290caf-phone_cfg.zzz |
| 22:18.13 | drfreeze | New polycom phones are not registering |
| 22:18.31 | thehar | is this #polycom? |
| 22:18.35 | drfreeze | the app.log file ends with the following |
| 22:18.39 | drfreeze | yes, polycom 550 |
| 22:19.11 | *** join/#asterisk sezuan (bouncer@irc.scheff32.de) |
| 22:19.44 | *** join/#asterisk netmax (~netmax@is.linux-administrator.com) |
| 22:21.59 | *** join/#asterisk chasing`Sol (~rc4@smtp.master-zone.net) |
| 22:42.21 | *** join/#asterisk b0gatyr_ (~b0gatyr_@adsl-10-106-138.mia.bellsouth.net) |
| 22:45.56 | *** join/#asterisk jaskew (~jdaskew@netblock-75-79-182-110.dslextreme.com) |
| 22:46.23 | *** join/#asterisk mpe (~mpe@0xd99d3f8f.customer.cybercity.dk) |
| 22:46.52 | *** join/#asterisk Juggie (~Juggie@CPE001601df17fb-CM001ceac25ada.cpe.net.cable.rogers.com) |
| 22:48.22 | *** join/#asterisk plundra (404@article.se) |
| 22:54.49 | *** part/#asterisk jsgoecke (~Adium@c-98-207-5-162.hsd1.ca.comcast.net) |
| 22:58.17 | Gibby13 | fti, fixed my issues, just had to reboot my asterisk server |
| 23:00.20 | *** join/#asterisk exothermc (~miles@74.85.89.146) |
| 23:03.54 | talntid | asterisk gurus..... when selecting from asterisk_cdr, if i wanted just the NPA/NXX for every area code.. and one NXX in each area code... anyone know how I would acquire this? |
| 23:11.47 | *** join/#asterisk erer (~chatzilla@bgl93-7-88-189-218-150.fbx.proxad.net) |
| 23:11.55 | erer | hello |
| 23:12.27 | erer | any asterisk freelancer herre ? |
| 23:13.10 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 23:21.15 | *** join/#asterisk lauris (~la@unaffiliated/lauris) |
| 23:23.51 | *** join/#asterisk riscphree (~riscphree@h243.104.90.75.dynamic.ip.windstream.net) |
| 23:24.47 | wwalker | erer ... yes |
| 23:25.11 | wwalker | :) |
| 23:25.33 | wwalker | probably 100 or so had he waited a few minutes. |
| 23:27.46 | *** join/#asterisk Faithful (~Faithful@180.194.1.133) |
| 23:38.20 | Jasnejac | wwalker: you scared him off :D |
| 23:40.25 | Jasnejac | would anyone like to share experiences with Sangoma viz-viz Digium? |
| 23:41.41 | drudge` | is asterisk 1.6 compat. with rfc 1149? |
| 23:43.12 | wwalker | Jasnejac: I didn't think he saw my face. oops. |
| 23:43.36 | [TK]D-Fender | Jasnejac: No, but you can go fly a kite ;) |
| 23:43.45 | [TK]D-Fender | drudge`: Rather |
| 23:43.50 | [TK]D-Fender | Jasnejac: Disregard that.. |
| 23:43.55 | drudge` | lol |
| 23:44.00 | Jasnejac | lol. I like flying kites |
| 23:45.42 | Jasnejac | most things have strings attached in life, asterisk included |
| 23:47.04 | Jasnejac | OK, different subject, has anyone played with ews integraton for calendaring? |
| 23:53.28 | *** part/#asterisk [T]ank (~ckwall@c-71-195-199-101.hsd1.ut.comcast.net) |
| 23:55.53 | *** join/#asterisk Khyron (~tim@c-24-15-25-209.hsd1.il.comcast.net) |
| 23:57.03 | Khyron | is this the appropriate place to look for somebody to do an install in chicagoland =D |
| 23:57.58 | drudge` | lol |
| 23:58.09 | drudge` | you cna fly me there back home, ill do it =p |
| 23:58.33 | Khyron | well i dont want to do it im afraid of conflicts with where i work =P |
| 23:59.53 | drudge` | did you get in over your head on some project or something? |
