| 00:05.37 | *** join/#asterisk m_tadeu (~quassel@89-181-156-133.net.novis.pt) |
| 00:05.39 | pabelanger | [1]sassyn: what are you changing in the build process? |
| 00:10.25 | [1]sassyn | some code and some addons |
| 00:10.27 | [1]sassyn | like mp3 |
| 00:10.35 | [1]sassyn | and some app like fax |
| 00:11.00 | [1]sassyn | which binutils version u using? |
| 00:14.06 | pabelanger | [1]sassyn: pb the complete output of the configure script |
| 00:18.39 | pdtpatrick | Question .. im seeing this error. Has anyone come across such? |
| 00:18.39 | pdtpatrick | [Aug 11 17:17:49] ERROR[31346]: utils.c:1164 ast_carefulwrite: write() returned error: Connection refused |
| 00:18.40 | pdtpatrick | [Aug 11 17:17:49] WARNING[31346]: res_agi.c:1506 launch_netscript: Connect to 'agi://localhost/html/ONP/index.vxml' failed: Connection refused |
| 00:24.18 | *** join/#asterisk logicwrath (~no@c-68-62-24-205.hsd1.mi.comcast.net) |
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| 01:23.42 | *** part/#asterisk shadowapex (~William@adsl-99-107-163-196.dsl.lsan03.sbcglobal.net) |
| 01:46.23 | jeffspeff | i'm using a 7945g ip phone, and trying to edit the SEP<mac>.cnf.xml file to allow the second line button to use the same SIP user and extension as the first line button. I configured my spa504g's to do this, but it had a web interface. any suggestions? |
| 02:02.01 | p3nguin | Last I knew, SEP files weren't for SIP. |
| 02:11.31 | *** join/#asterisk Kumbang (~unknown@180.245.137.5) |
| 02:20.32 | *** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital) |
| 02:22.43 | *** join/#asterisk KavanS (~KavanS@LINBIT/KavanS) |
| 02:27.39 | *** join/#asterisk voiper (75c98a36@gateway/web/freenode/ip.117.201.138.54) |
| 02:28.34 | voiper | can anyone help me with the syntax for sending username and password as part of SIP dial command ? |
| 02:29.15 | *** join/#asterisk bdfoster (~bdfoster@unaffiliated/bdfoster) |
| 02:29.31 | voiper | I tried using this syntax but doesn't seem to work. SIP/extension:password:username@domain.com:5060 |
| 02:31.13 | p3nguin | Did you try Dial(SIP/username:password@domain.com/extension) ? |
| 02:31.26 | voiper | yes i tried that |
| 02:31.29 | voiper | it didn't work either |
| 02:32.46 | voiper | i am trying to send a call from asteirsk 1.4 to trixbox |
| 02:33.26 | *** join/#asterisk coppice (~chatzilla@116.92.38.165) |
| 02:33.43 | voiper | if i send that way in trixbox it is coming as Executing [username:password@from-sip-external:1] |
| 02:33.49 | p3nguin | Any reason you can't define a peer for the other side? It's clear to me that it will be a static setup rather than something you'll be changing often. |
| 02:34.21 | voiper | i am writing an agi to send calls using multiple username and password |
| 02:34.32 | voiper | so a static setup wouldn't work for my scenario |
| 02:39.52 | voiper | thanks for responding p3nguin |
| 02:43.54 | p3nguin | What about Dial(SIP/username:password@domain.com/extension@domain.com) ? |
| 02:45.09 | voiper | i will try that |
| 02:46.55 | voiper | that doesn't seem to work either |
| 02:47.28 | p3nguin | I can't think of anything else. Google didn't have any ideas? |
| 02:48.55 | voiper | no i found few docs with the same syntax but those are failing |
| 02:50.59 | p3nguin | Wait, I thought of one more thing. Dial(SIP/extension@domain.com@username:password@domain.com) |
| 02:51.10 | p3nguin | or Dial(SIP/extension@domain.com:username:password@domain.com) |
| 02:51.40 | voiper | let me try |
| 02:53.43 | voiper | no luck |
| 02:54.07 | p3nguin | I don't know what else to try. |
| 02:55.36 | voiper | thanks for your help |
| 02:56.41 | p3nguin | If possible, ask during daytime business hours in the USA. There are more active people with more experience. |
| 02:57.11 | WIMPy | sip.conf clearly states SIP/user:password@host:port |
| 02:57.48 | p3nguin | Where does the extension go? |
| 02:58.40 | WIMPy | No extension mentioned. |
| 02:59.13 | voiper | yeah i saw that too but where will we send the exten |
| 02:59.56 | WIMPy | Looks like you can only have either user/pass or exten. |
| 03:00.39 | voiper | its very strange though as I could send the same call from a softphone like zoiper without registering |
| 03:00.42 | voiper | and it does work |
| 03:02.53 | p3nguin | That reminds me... is it a feature of SER which requires a user agent to register before it can send calls? Asterisk allows authorized calls without registration first. Is that something asterisk can have configured, or is it the proxy that does it for several ITSPs? |
| 03:03.25 | *** join/#asterisk obnauticus (~obnauticu@about/windows/regular/obnauticus) |
| 03:04.04 | WIMPy | I don't think you can make Asterisk requite registration for calls. |
| 03:04.15 | p3nguin | Must be the proxy that does it, then. |
| 03:04.49 | p3nguin | I learned that with voipms, if I don't register first, they reject an otherwise authorized call. |
| 03:05.03 | p3nguin | Send the registration and then that same call succeeds. |
| 03:05.20 | obnauticus | Anyone here have any idea why Asterisk is not playing this sound file? http://paste.pocoo.org/show/457007/ |
| 03:05.51 | obnauticus | However, it does play this file: http://paste.pocoo.org/show/457008/ |
| 03:06.14 | WIMPy | obnauticus: Because there's mor than just a WAVE chunk in the file and you are using an older Asterisk, perhaps? |
| 03:06.24 | p3nguin | obnauticus: It's all wrong. that file is 44100 Hz stereo, and asterisk require 8000 Hz mono. |
| 03:06.46 | obnauticus | Alright. I will convert it. Sox is not working for me locally right now, though...for whatever reason |
| 03:06.54 | WIMPy | Asterisk doesn't require that. |
| 03:07.31 | p3nguin | So now we can throw random bitrate multi-channel wave files at it? I don't think so, Tim. |
| 03:07.49 | WIMPy | And with Asterisk 10 it might make sense not to downsaple it. But mixing it to one channel should still make sense. |
| 03:07.50 | p3nguin | obnauticus: Compare line 20 of the first paste with line 19 of the second paste. |
| 03:08.40 | WIMPy | But Asterisk didn't like any tags or so in wav files until recently. |
| 03:09.16 | obnauticus | p3nguin, do you know of any way to convert it with Sox? I have version 14.3.2, and I've tried looking online. Nothing I've found has worked. |
| 03:11.16 | *** join/#asterisk voiper (75c98a36@gateway/web/freenode/ip.117.201.138.54) |
| 03:11.28 | obnauticus | p3nguin, it keeps saying the RIFF header was not found :( |
| 03:11.49 | WIMPy | Loos like the file is broken. |
| 03:11.51 | p3nguin | obnauticus: Something like sox TVCampaign.wav -r8000 -c1 -s -w TVCampaign-8k.wav, maybe. |
| 03:12.04 | voiper | sorry got disconnected |
| 03:12.12 | obnauticus | p3nguin, did not work :( |
| 03:12.40 | p3nguin | Any error? |
| 03:12.56 | obnauticus | RIFF header not found. |
| 03:13.03 | obnauticus | well, for your command the syntax was just wrong |
| 03:13.17 | p3nguin | I'll see about that. |
| 03:13.25 | obnauticus | I will upload the sound file somewher |
| 03:13.40 | obnauticus | p3nguin, can I notice the URL to you: |
| 03:13.43 | WIMPy | Try file to find out what kind of file it is. |
| 03:14.13 | obnauticus | WIMPy, AUDIO: 44100 Hz, 2 ch, s16le, 96.0 kbit/6.80% (ratio: 12003->176400) Selected audio codec: [ffwmav2] afm: ffmpeg (DivX audio v2 (FFmpeg)) |
| 03:15.07 | *** join/#asterisk vinhdizzo (~vinh@pool-173-60-112-55.lsanca.fios.verizon.net) |
| 03:15.21 | WIMPy | Looks like you should convert it with mplayer. |
| 03:15.57 | p3nguin | Okay, so I had an extraneous -w in my command. Remove it, and it works. |
| 03:16.32 | obnauticus | p3nguin, it still says the RIFF Header was not found. I think sox is being fed the incorrect audio file |
| 03:16.44 | p3nguin | Give me the file and I'll try it. |
| 03:16.47 | obnauticus | kk |
| 03:16.49 | obnauticus | lemme upload it somewhere |
| 03:16.54 | WIMPy | It might be folled by the name. |
| 03:17.23 | WIMPy | It obviousely isn;t a wav file. Try to give it the right extension or use mplayer. |
| 03:17.40 | WIMPy | NFI what would be correct there. Maybe .mp4? |
| 03:17.41 | obnauticus | That's what I'm thinking. |
| 03:17.52 | p3nguin | Oh, it's not even a wave?! |
| 03:18.00 | p3nguin | DivX audio that we're trying to read as PCM? |
| 03:18.01 | obnauticus | yeah, damn retards recorded this file. lol. |
| 03:18.05 | obnauticus | p3nguin, I didn't do it. |
| 03:18.06 | obnauticus | so |
| 03:18.07 | obnauticus | lol |
| 03:18.25 | p3nguin | sighs in disgust |
| 03:18.55 | obnauticus | http://obnauticus.com/TVCampaign.wav |
| 03:19.26 | p3nguin | Hmm. |
| 03:19.30 | p3nguin | TVCampaign.wav: HTML document text |
| 03:19.34 | obnauticus | ... |
| 03:19.37 | obnauticus | really? lol |
| 03:19.44 | p3nguin | That's going to be even harder to turn into a wave. |
| 03:19.48 | WIMPy | Use maplyer. It doesn't care. |
| 03:19.58 | obnauticus | lol p3nguin i set the wrong permissions |
| 03:19.59 | obnauticus | give me a sec |
| 03:20.06 | obnauticus | WIMPy, what command did you use for mplayer? |
| 03:20.26 | WIMPy | mplayer -ao pcm:file=out.wav thefile |
| 03:20.50 | WIMPy | You can get it faster with extra options, but it will tell you. |
| 03:21.07 | *** join/#asterisk neurosys (~neurosys@76.8.87.100) |
| 03:22.20 | obnauticus | WIMPy, how do I set mono and the rate? |
| 03:22.26 | obnauticus | i.e., mono/8k |
| 03:22.53 | p3nguin | If you can't make mplayer output it correctly, just accept it as whatever it is, then use sox to change it. |
| 03:22.59 | obnauticus | ah ok |
| 03:23.26 | p3nguin | As I said, I just threw in an extra -w in that sox command. |
| 03:23.31 | WIMPy | That was the idea. You can use mencoder, but you have to find the options yourself. |
| 03:43.27 | *** join/#asterisk davlefou (~david@41.225.9.81) |
| 03:52.07 | *** join/#asterisk voiper (75c98a36@gateway/web/freenode/ip.117.201.138.54) |
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| 04:00.27 | obnauticus | How do I manually clear a voicemail box of all of its recordings. |
| 04:01.29 | p3nguin | rm will take care of it for you. |
| 04:01.49 | obnauticus | That's what I was thinking but I wanted to make sure. Do you know where they're stored? |
| 04:02.11 | p3nguin | /var/spool/asterisk/voicemail/ |
| 04:06.45 | obnauticus | Thank you |
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| 05:02.01 | *** join/#asterisk singler (~singler@81-7-123-162.static.zebra.lt) |
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| 05:43.18 | *** join/#asterisk schmidts (~schmidts@Lo20-rt01.lm33.sil.at) |
| 05:43.20 | schmidts | good morning |
| 05:48.44 | *** join/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 05:49.04 | ChannelZ | HI! |
| 05:49.47 | WIMPy | lo |
| 05:50.47 | *** part/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 05:57.28 | *** join/#asterisk sunfone (~jeff@c-67-184-36-158.hsd1.il.comcast.net) |
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| 06:07.56 | ChannelZ | Should I put my SIP URI in an <a href...> ? It really only means anything if someone had a softphone and it happened to hook into their browser I guess. |
| 06:08.25 | *** join/#asterisk sunfone (~jeff@c-67-184-36-158.hsd1.il.comcast.net) |
| 06:10.16 | WIMPy | It doesn't hurt. |
| 06:10.47 | WIMPy | But it's quite bad that these kind of things aren't easily set up with modern OSs. |
| 06:10.50 | *** join/#asterisk e7e5 (~rudenko@188.134.2.33) |
| 06:11.33 | ChannelZ | Yeah. Just trying to figure out the best way to put this on the footer of my site |
| 06:12.15 | *** join/#asterisk mintos (~mvaliyav@nat/redhat/x-bjhccfswuzikkgck) |
| 06:14.57 | *** join/#asterisk eject_ck (~eject_ck@62.205.134.210) |
| 06:15.05 | eject_ck | Hi all |
| 06:15.19 | eject_ck | I see tens of messages in console -- Remote UNIX connection disconnected |
| 06:15.30 | eject_ck | why it appears ? |
| 06:15.43 | WIMPy | You are using some GUI? |
| 06:16.12 | eject_ck | no |
| 06:16.35 | WIMPy | Some other util? |
| 06:16.47 | WIMPy | Or did you allow the internet to connect? |
| 06:16.59 | eject_ck | hm, yes .. I have nagios checks :) |
| 06:17.09 | eject_ck | sorry |
| 06:17.18 | eject_ck | how can I suppress this message in console ? |
| 06:17.24 | *** join/#asterisk kleszcz (tick@80.54.23.253) |
| 06:17.54 | WIMPy | I think there's something in manager.conf |
| 06:18.47 | WIMPy | displayconnects might be the one. |
| 06:19.04 | ChannelZ | or turn off manager if you're not using it |
| 06:19.47 | *** join/#asterisk TimeRider (steve@188-220-33-123.zone11.bethere.co.uk) |
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| 06:21.05 | eject_ck | manager is disables |
| 06:21.26 | eject_ck | disabled, I'm using console script to check status with nagios |
| 06:25.42 | ChannelZ | oh, I see |
| 06:26.04 | ChannelZ | I was thinking of the wrong message |
| 06:27.39 | kaldemar | that is because of "asterisk -rx", it has nothing to do with manager. |
| 06:27.51 | ChannelZ | yah |
| 06:28.36 | ChannelZ | It's a verbose message, only way to get rid of it (besides hacking the source) would be to lower verbose to 2 or less |
| 06:28.38 | kaldemar | eject_ck: your choices are to remove the verbosity from main/asterisk.c or live with it. |
| 06:28.43 | eject_ck | :) |
| 06:28.48 | eject_ck | thank you! |
| 06:29.01 | kaldemar | ChannelZ has a point, it is at level >=3. |
| 06:29.14 | eject_ck | verbose 2 is too low for me :) |
| 06:29.24 | eject_ck | Thank you guys! |
| 06:29.35 | ChannelZ | According to the source it's 3 (I think? ast_verb(3,"blah")) but I still see it on 2. Hmm. |
| 06:30.10 | ChannelZ | oh.. nevermind, my test turned it back up to 3. grrph |
| 06:30.10 | kaldemar | hmm.. it is inside an if (!ast_opt_hide_connect)... |
| 06:30.33 | ChannelZ | Yah hadn't looked that one up yet. |
| 06:31.58 | ChannelZ | hideconnect in asterisk.conf |
| 06:32.11 | ChannelZ | Niftty! |
| 06:33.06 | ChannelZ | Cookies for everyone! |
| 06:39.47 | *** join/#asterisk irroot (~irroot@pbx.distrotech.co.za) |
| 06:40.09 | eject_ck | Vilen Dank! |
| 06:41.54 | *** join/#asterisk screenn (~screenn@178.151.86.196) |
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| 07:16.06 | *** join/#asterisk syntaxx (~patvan@unaffiliated/syntaxx) |
| 07:16.27 | syntaxx | what application to use for presence? so it shows the status of the user on a sip client |
| 07:17.45 | irroot | syntaxx device state and hints ... |
| 07:23.15 | syntaxx | irroot, ayt thanks |
| 07:26.18 | syntaxx | irroot, do you have any idea what is the default sip user after doing a make sample? |
| 07:26.31 | syntaxx | is it under users.conf? |
| 07:27.09 | irroot | syntaxx no clue :P have not done that in years |
| 07:27.40 | singler | syntaxx: by default SIP allows guest connections (no users at all) |
| 07:28.01 | syntaxx | singler, guest? like what sip username should i use? |
| 07:28.28 | singler | like Dial(SIP/ip.address/extension) |
| 07:28.37 | singler | no user at all |
| 07:29.21 | syntaxx | singler, i cant seem to login |
| 07:29.44 | singler | login where? |
| 07:29.54 | syntaxx | im using a sip client |
| 07:30.28 | singler | you cannot register to server? you need to do configuration for that |
| 07:31.19 | singler | but if your client can place a call without registration, it should work. But of course it is not safe |
| 07:31.19 | WIMPy | Hmm. What has happened to app_dial? I get a lot of warnings like "left shift count >= width of type". That sounds dangerous. |
| 07:31.41 | syntaxx | singler, ayt |
| 07:32.10 | singler | ? |
| 07:32.12 | syntaxx | singler, i wanted to know the default user so i can check which context should i use for testing |
| 07:33.37 | singler | default context should be used, I think that you should look into configuration files (sip.conf and extensions.conf) |
| 07:33.56 | syntaxx | singler, alright thanks |
| 07:33.58 | singler | there is some demo application by default |
| 07:36.17 | *** join/#asterisk Buklov (~Buklov@mail.sapsun.su) |
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| 08:37.17 | Tuju | any idea / examples how to use conntrack_sip in linux iptables? |
| 08:38.00 | WIMPy | Just load it. |
| 08:38.23 | Tuju | but how it's used in chains? |
| 08:38.49 | WIMPy | Do you understand what connection tracking is? |
| 08:38.54 | Russ | packets will show up as "RELATED" |
| 08:39.29 | WIMPy | The RTP packets, that is. |
| 08:40.41 | WIMPy | Or even more exactely, the first RTP packet. |
| 08:46.07 | Tuju | hmm... |
| 08:46.43 | Tuju | no, i just knew that it will dig deeper inside the packet but not how it affects |
| 08:47.05 | Tuju | my problem is still that new-cisco-with-asterisk problem |
| 08:47.41 | Tuju | cisco sends packets with high src port number and asterisk responds into that port - which cisco does not listen, only 5060. |
| 08:47.51 | WIMPy | It just adds it's findings to /proc/net/nf_conntrack |
| 08:48.16 | Tuju | now i changed that src port number in asterisk firewall end, with src SNAT rule |
| 08:48.30 | Tuju | and asterisk responds back to 5060 as it sees it as source port |
| 08:49.12 | Tuju | but for some reason that packet doesn't appear in cisco's end in firewall anymore - i guess it might screw up NAT state engine whent he portnumbers doesn't match anymore. |
| 08:49.18 | WIMPy | So that phone is behind nat? |
| 08:49.22 | Tuju | yes |
| 08:49.58 | WIMPy | Sure. That would make it a new connection. |
| 08:50.01 | Tuju | cisco phones have a nat setting which i've set Yes. |
| 08:50.20 | WIMPy | You probably shouldn't. |
| 08:50.36 | Tuju | in older models it has helped |
| 08:50.48 | Tuju | but those used src port number differently |
| 08:50.53 | WIMPy | Usually you should only set nat=yes in the peer definition and configure the client as if there was not NAT. |
| 08:51.24 | Tuju | ah, in asterisk end |
| 08:52.41 | Tuju | yup, i've had that nat=yes in asterisk end whole time. |
| 08:52.50 | Tuju | i now switched that off in cisco phone |
| 08:53.52 | WIMPy | Sure. Otherwise you wouldn't have that port issue. |
| 08:56.32 | *** join/#asterisk hehol (~hehol@2001:1438:1009:200:3d:cee0:8ab2:782e) |
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| 08:58.38 | *** join/#asterisk _omer (~omer@182.185.158.87) |
| 08:59.17 | *** join/#asterisk x1user (~x1user@host-212-75-8-69.bbccable.net) |
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| 09:00.03 | _omer | Hello, Can I run any agi script when call is dialed ? actually I want to play User Balance parellal to dial function ... |
| 09:00.08 | Tuju | that didn't help. even i'm able to change that high src port number in service end and able to get asterisk send responses back to 5060, client end fw still changes it back to high port. |
| 09:00.26 | x1user | Hi, i have problem. NOTICE[10488]: chan_sip.c:20276 handle_request_invite: Call from '' to extension '' rejected because extension not found in context 'default'. |
| 09:02.09 | _omer | x1user: use 's' extension in default context ... |
| 09:03.24 | x1user | like exten => _X.,1,s,Answer() |
| 09:05.49 | _omer | no |
| 09:05.58 | _omer | well try this one |
| 09:06.08 | _omer | _X.,1,Answer |
| 09:06.25 | x1user | it was _x.,1,Answer() actually |
| 09:06.29 | x1user | it doesnot works alsow with s |
| 09:07.04 | _omer | add this one too exten = > s,1,Answer |
| 09:08.04 | x1user | doesnot work either |
| 09:08.10 | _omer | what number you are dialing ? |
| 09:08.31 | x1user | my own number witohut prefixes |
| 09:09.00 | _omer | Call from '' to extension '' rejected <----- there is no number |
| 09:09.10 | x1user | i've stripped the number |
| 09:09.31 | x1user | call from 'usernamte to extension 'mynumber' |
| 09:10.18 | _omer | _x.,1,answer should be .... no matters if you dial prefix or not ... |
| 09:10.29 | _omer | it should answer all the calls with any number.. |
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| 09:19.27 | kaldemar | _omer: only ones with one digit and one or more characters. |
| 09:20.40 | kaldemar | _omer: if you want to run the script when the dial command is executed, use Dial(Local/exten@scriptcontext&<tech>/...) |
| 09:22.42 | _omer | kaldemar: thanks...let me check |
| 09:24.50 | _omer | kaldemar: Dial(Local/12345@default&SIP/12125552121@trunk) <--- correct ? |
| 09:27.07 | _omer | kaldemar: thanks... I will check it out ... logic looks fine... |
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| 09:28.51 | *** join/#asterisk merlin8282 (~merlin828@AStrasbourg-554-1-218-224.w86-204.abo.wanadoo.fr) |
| 09:28.57 | merlin8282 | Hiho ! |
| 09:29.48 | merlin8282 | Anyone has an idea why the announcement is not played to the queue agent ? "announce = /path/to/wav/file" is set in queues.conf. |
| 09:30.45 | *** join/#asterisk StaRetji (~BigAll@80.93.240.171) |
| 09:31.02 | StaRetji | folks, I need to get did number from telekom provider |
| 09:31.06 | merlin8282 | When the call is coming, it has music, etc. then before being connected the agent has: Playing 'queue-reporthold.slin', 'digits/4.g722' and 'queue-seconds.gsm', then it connects both call legs. The announce is simply ignored ! |
| 09:31.17 | StaRetji | they ask me if I can receive inband RTP |
| 09:31.18 | StaRetji | ? |
| 09:34.06 | kaldemar | inband RTP? are you sure that's what they asked? |
| 09:35.36 | StaRetji | yes |
| 09:35.57 | StaRetji | they have to redirect real phone numbers to my Asterisk server |
| 09:36.02 | StaRetji | and I told them okay |
| 09:36.14 | StaRetji | but I still have chance to call them and tell them I can't |
| 09:36.25 | StaRetji | btw, kaldemar, thx for reply |
| 09:36.30 | kaldemar | inband RTP doesn't make any sense. |
| 09:36.40 | merlin8282 | knows "inband DTMF"... |
| 09:36.47 | kaldemar | are you sure they didn't mean inband DTMF in RTP? |
| 09:36.51 | kaldemar | that asterisk does support. |
| 09:37.07 | StaRetji | I will ask them now |
| 09:37.34 | kaldemar | but RFC2833 of even SIP INFO would be considerably better. |
| 09:39.42 | StaRetji | yes |
| 09:39.59 | StaRetji | they said they can support rfc2833 but they prefer inband |
| 09:40.06 | StaRetji | so, I guess it is only for DTMF |
| 09:40.45 | StaRetji | in that case, shall I do'em a favor and accept inband |
| 09:41.01 | StaRetji | those numbers are for IVR |
| 09:41.16 | StaRetji | so, I don't know what is better for Asterisk |
| 09:41.27 | *** part/#asterisk Russ (foobar@ip70-176-251-1.ph.ph.cox.net) |
| 09:41.38 | kaldemar | your choice. i'd prefer the better solution. which would be RFC2833. |
| 09:41.43 | StaRetji | got it |
| 09:41.52 | StaRetji | I will tell them I prefer RFC2833 |
| 09:41.57 | StaRetji | thx :) |
| 09:42.52 | merlin8282 | http://en.wikipedia.org/wiki/THX ? :p |
| 09:45.25 | StaRetji | hehe |
| 09:45.33 | StaRetji | Thank You!!! |
| 09:45.34 | StaRetji | :) |
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| 10:37.26 | x1user | loader.c:814 load_resource: Module 'chan_mobile.so' could not be loaded. |
| 10:40.41 | kaldemar | when do you get that and what do the lines before it say? |
| 10:41.25 | x1user | loader.c:730 inspect_module: Module 'chan_mobile.so' was not compiled with the same compile-time options as this version of Asterisk. |
| 10:41.39 | x1user | it was working yeastarday and i didnot change anything since |
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| 10:43.39 | *** join/#asterisk devil_evoxxx (~d3v1l@host125-93-dynamic.9-87-r.retail.telecomitalia.it) |
| 10:43.44 | devil_evoxxx | hi all |
| 10:45.19 | kaldemar | x1user: sure. recompile and re-install. |
| 10:46.01 | *** join/#asterisk sohilg (~sohilg434@59.177.35.236) |
| 10:46.10 | sohilg | Hi All... |
| 10:46.21 | *** part/#asterisk didnot (~didnot@unaffiliated/didnot) |
| 10:46.26 | devil_evoxxx | i've got a problem with sip signaling. I'm using asterisk 1.4.37. The scenario is: SIP phone A start the call to SIP Phone B(receiver) and during the call, the PHONE B lost power, and in asterisk CLI i can see that the call is already UP. |
| 10:46.30 | sohilg | facing problems with sip_rouge ...while relaying the calls |
| 10:46.32 | sohilg | pls help |
| 10:46.35 | devil_evoxxx | there are some trick to solve this problema' |
| 10:46.42 | devil_evoxxx | problem? |
| 10:47.05 | kaldemar | devil_evoxxx: what problem? |
| 10:47.20 | devil_evoxxx | that if the phone b lost power, the call in asterisk is still UP |
| 10:48.40 | kaldemar | devil_evoxxx: use RTP timers, they are configured in sip.conf. rtptimeout and rtpholdtimeout. |
| 10:51.53 | devil_evoxxx | thankyou so much! Now i google on this features :) |
| 10:52.23 | kaldemar | just take a look at the sample sip.conf |
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| 11:45.00 | merlin8282 | Anyone has an idea about my problem (announce not played to agent) ? |
| 11:49.08 | kaldemar | did you configure it correctly? do you have an announceoverride in the Queue command? |
| 11:49.32 | merlin8282 | kaldemar: Just tried it with announceoverride: this works. But without, nok. |
| 11:49.50 | merlin8282 | kaldemar: with "announce = /path/to/file" it should work also, no ? |
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| 11:50.39 | kaldemar | don't know about the path, but it should if in the right place. |
| 11:51.30 | merlin8282 | kaldemar: the strange thing is that it does not report any error, such as for example "file not found" or so... |
| 11:51.41 | merlin8282 | anyway. With announceoverride it works, it's then ok. |
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| 12:29.14 | lanmower | lo all |
| 12:29.34 | lanmower | my asterisk-fu is a little rusty, can someone assist me with figuring out why I cant use my trunks channel? |
| 12:30.23 | lanmower | i have a trunk and route configured, and its trying to dial as recommended by the provider. |
| 12:30.33 | lanmower | using the right numbers that is. |
| 12:31.13 | *** join/#asterisk tzafrir (~tzafrir@IGLD-84-229-111-190.inter.net.il) |
| 12:31.50 | lanmower | i'm receiving a congested message from asterisk while the call is being dialled. |
| 12:33.09 | lanmower | my asterisk is behind a firewall, and I have udp and stun ports forwarded back to it, its configured to use nat settings with a dyndns setup to resolve the ip. The sip registrar on the trunk is receiving and allowing my login, based on what I can tell in sip show peers |
| 12:33.12 | lanmower | and sip show registry |
| 12:37.11 | lanmower | should I paste some debug output? |
| 12:39.09 | lanmower | http://pastebin.com/dE89PaXK |
| 12:40.33 | dwayne | Qwell, pop quiz: what's the episode where Bart is enthusiastically singing with the church choir? |
| 12:41.02 | lanmower | http://pastebin.com/hDruSXBf |
| 12:47.28 | lanmower | anyone alive here? |
| 12:48.14 | eject_ck | :-D |
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| 13:01.19 | lanmower | anyway, can I have some advice on how to pinpoint the problem? |
| 13:02.09 | lanmower | sip.wanatel.net:5060 N XXXXXXXX 105 Registered Fri, 12 Aug 2011 14:58:47 |
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| 13:39.48 | StaRetji | omg, lol app_swift is installed and now my asterisk 1.4.2 crashes when I call extension which loads app_swift |
| 13:40.11 | StaRetji | looking at /var/log/asterisk/messages I don't see any error |
| 13:41.20 | StaRetji | can someone help me to debug please |
| 13:41.21 | StaRetji | thx |
| 13:43.17 | leifmadsen | 1.4.2? |
| 13:43.19 | leifmadsen | wow that's crazy old |
| 13:43.33 | kaldemar | StaRetji: is the app supposed to be compatible with your version? |
| 13:44.02 | leifmadsen | 20-Mar-2007 09:22 |
| 13:44.09 | kaldemar | if so, https://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace |
| 13:44.16 | leifmadsen | I'd be shocked if anything third-party worked with an asterisk version that old |
| 13:46.10 | StaRetji | kaldemar: yes |
| 13:46.19 | robl^laptop | leifmadsen: 1.4 .2 is old? I ran across a vmware image where I was staging / testing a pre 1.0 install ;-) |
| 13:46.34 | StaRetji | sorry, it is in production with a2b |
| 13:46.45 | StaRetji | I'm afraid to touch it :/ |
| 13:46.56 | StaRetji | I added app_swift |
| 13:46.58 | StaRetji | and it works |
| 13:47.09 | StaRetji | but it seems if I call extension 777777 |
| 13:47.16 | StaRetji | which starts Swift |
| 13:47.22 | StaRetji | everything is okay, until I hangup |
| 13:47.26 | StaRetji | asterisk crashes |
| 13:47.36 | chazzam | I thought you said it works |
| 13:48.16 | *** join/#asterisk agnogenic (~agnogenic@66.239.124.34.ptr.us.xo.net) |
| 13:48.17 | StaRetji | well, I started asterisk and it works if I don't call 777777 |
| 13:48.30 | StaRetji | if I call 777777, I hear Cepstral voice |
| 13:48.32 | chazzam | so the module loads, but doesn't work? |
| 13:48.40 | StaRetji | it works |
| 13:48.46 | StaRetji | but when I hangup the line |
| 13:48.51 | chazzam | but it crashes, that isn't really working is it? |
| 13:48.52 | StaRetji | asterisk crashes |
| 13:49.02 | StaRetji | well, yes :) |
| 13:49.03 | StaRetji | sorry |
| 13:49.11 | chazzam | =p |
| 13:49.19 | StaRetji | I mean, app_swift does says what it has |
| 13:49.26 | StaRetji | but it crashes asterisk |
| 13:49.28 | chazzam | have you tried finding an older version of app_swift? |
| 13:49.32 | agnogenic | I have a question about the Asterisk yum repo. Is there a time line for adding Centos6 support? |
| 13:49.58 | StaRetji | chazzam: good idea, I installed app_swift 2 |
| 13:50.08 | StaRetji | maybe I should try installing app_swift 1.4 |
| 13:50.33 | chazzam | find one released in about 2009 ? |
| 13:51.00 | StaRetji | okay, will try |
| 13:51.13 | chazzam | or try upgrading |
| 13:51.15 | chazzam | ;p |
| 13:56.58 | StaRetji | chazzam: and kaldemar thx folks, downgrading to app_swift 1.4 seems to fix the problem |
| 13:57.05 | *** join/#asterisk serafie (~erin@nat/digium/x-ekjmlitdivnzmhxr) |
| 13:57.06 | StaRetji | no crashes on hangup |
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| 13:59.34 | treborsux | I am so excited |
| 14:00.06 | chazzam | heh, yay! |
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| 14:01.07 | treborsux | I ordered 25 polycom 501s last night and 2 560s |
| 14:01.21 | pabelanger | treborsux: I see, you just can't hide it! |
| 14:01.40 | lirakis | sup snizwidgets |
| 14:02.25 | treborsux | you know you know you know i just cant hide it |
| 14:02.35 | treborsux | good by merlin legends!!! |
| 14:03.13 | lirakis | Yeah! |
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| 14:11.14 | treborsux | merlin legend processor SMASH SMASH |
| 14:11.22 | treborsux | THis is my first change of one. |
| 14:11.37 | treborsux | I have 6 more to go |
| 14:11.40 | merlin8282 | . |
| 14:11.42 | treborsux | at our car dealerships |
| 14:12.01 | treborsux | the caps are getting so bad on cards i switch the procs monthly |
| 14:12.07 | treborsux | so over it! |
| 14:12.13 | treborsux | thank you asterisk! |
| 14:12.38 | treborsux | freepbx is kewl but what else do you guys recomend to use with asterisk? |
| 14:12.53 | treborsux | what is easiest? |
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| 14:13.38 | chazzam | flat text + templates? |
| 14:13.59 | chazzam | and "same" |
| 14:14.42 | treborsux | what is the diffrence between a ip 500 and ip 501 polycom |
| 14:14.52 | chazzam | shrugs |
| 14:14.57 | chazzam | Their website doesn't say? |
| 14:15.10 | treborsux | i bought 30 polycom 501s and he says he does not have enough and wants to give me 3 500s instead is that ok? |
| 14:15.31 | treborsux | looking on polycom now |
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| 14:20.15 | defswork | 1 |
| 14:20.25 | leifmadsen | treborsux: no -- the 500 is not the same as the 501 |
| 14:20.36 | leifmadsen | (it has significantly less memory and you can't use the latest bootroms) |
| 14:21.04 | leifmadsen | treborsux: tell him to either not send the extra 500's, or get a significant discount as you won't be able to upgrade them past a certain point |
| 14:21.23 | defswork | wonders why polycoms |
| 14:21.33 | leifmadsen | because they are rock solid? |
| 14:21.42 | kaldemar | treborsux: 501 has more memory. both are discontinued though. |
| 14:21.54 | leifmadsen | treborsux: ^^^ what kaldemar said |
| 14:22.08 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 14:22.26 | treborsux | He has offered instaed of 30 501s to send 28 501s and 3 500s for total of 31 |
| 14:22.37 | leifmadsen | treborsux: don't do it |
| 14:22.39 | treborsux | i have svery simple setup I think i will be ok |
| 14:22.45 | leifmadsen | honestly you should be using the 550's |
| 14:22.59 | leifmadsen | treborsux: I've heard that before until you want to do something and have to replace the phones :) |
| 14:23.12 | treborsux | it will never change |
| 14:23.53 | leifmadsen | That's what she said! |
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| 14:24.51 | coppice | in business "it will never change" == "we're going out of business this month" |
| 14:25.12 | leifmadsen | coppice: :) |
| 14:25.38 | treborsux | cant afford 550 |
| 14:25.45 | treborsux | 501 is the pricepoint |
| 14:26.02 | treborsux | answer calls transfer calls talk over vpn that is all i need |
| 14:26.11 | chazzam | ask the guy sending them to you to give you a discount on the three and give you 550's instead of 500s |
| 14:26.19 | chazzam | because you need at least equally capable, not less |
| 14:26.29 | *** join/#asterisk Godfather_ (~estanteri@90.162.114.94) |
| 14:26.29 | chazzam | and he can't fulfill your demand |
| 14:28.15 | coppice | telephony never fulfills |
| 14:29.36 | anonymouz666 | Mr. Corleone has joined this channel |
| 14:29.40 | treborsux | told him he could only substitue 550s |
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| 14:38.49 | agnogenic | I've set up a small asterisk server for testing, and am wondering if there are any voip providers that I could setup for around $5 |
| 14:39.31 | leifmadsen | agnogenic: you might be able to try out voip.ms as I think you can just pre-pay whatever amount you want |
| 14:39.46 | leifmadsen | usually the DID is going to be the thing that is the most "expensive" |
| 14:40.00 | agnogenic | Do they have any gotchas? |
| 14:40.18 | *** join/#asterisk _Corey_ (~chatzilla@64.121.4.75) |
| 14:40.19 | leifmadsen | I don't know -- read the fine print |
| 14:40.30 | leifmadsen | I don't understand what "gotchas" are |
| 14:41.37 | agnogenic | Its an idiom.. for example.. "Our service is only $5 a month... with a onetime $50 setup fee" |
| 14:42.08 | agnogenic | usually stuff hidden in fine print. Thank you for the recommendation though. I will check them out. |
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| 14:48.37 | Kobaz | leifmadsen: a gottcha is like, you think you have it right, but then you get swindled |
| 14:48.43 | coppice | fine print is similar to fine art - there's more con men than good guys |
| 14:49.27 | Kobaz | the 'gottcha' is a rewritten form of 'got-ya'.. but when you say 'got-ya' really fast, it sounds like gottcha |
| 14:49.59 | coppice | gotcha == "do you have any tea" |
| 14:50.01 | leifmadsen | Kobaz: doesn't no matter how fast I say it :) |
| 14:50.48 | Kobaz | faster! |
| 14:51.04 | Kobaz | and more mumbled |
| 14:51.14 | Kobaz | and skip syllables |
| 14:51.37 | chuckf | thinking you have it right and then you get swindled, that's not a gottcha. That's you makeing an assumption and being wrong when there were fees you didn't see |
| 14:51.38 | *** join/#asterisk Tech_Travis (~Travis_El@cpe-76-168-191-127.socal.res.rr.com) |
| 14:52.16 | *** join/#asterisk ezano (~ezano@sto93-2-82-228-142-248.fbx.proxad.net) |
| 14:52.22 | ezano | hi, |
| 14:53.09 | ezano | I really need to know how to get the real caller id on a dialplan please ? |
| 14:53.12 | Kobaz | http://en.wikipedia.org/wiki/Gotcha |
| 14:53.19 | Kobaz | Gotcha and I gotcha are relaxed pronunciations of "I've got you", usually referring to an unexpected capture or discovery. Gotcha is a common colloquialism meaning to understand or comprehend. |
| 14:53.43 | Kobaz | fees you didn't see would fit that definition |
| 14:53.59 | *** join/#asterisk cerienjean (~iper@95.138.77.91) |
| 14:54.15 | Kobaz | expected discovery |
| 14:54.23 | Kobaz | *un |
| 14:54.31 | chuckf | Kobaz: no they wouldn't, if they are spelled out in the contract and you didn't see them, that's your fault for not reading carefully |
| 14:54.45 | Kobaz | it's still an unexpected discovery |
| 14:54.53 | chuckf | but not a gottcha |
| 14:56.11 | ezano | humpf, this is a troll chan or a support chan here |
| 14:56.42 | Kobaz | ~asterisk |
| 14:56.43 | infobot | Asterisk is an open source telephony toolkit, or #asterisk on irc.freenode.net, or http://www.asterisk.org/ |
| 14:57.35 | *** join/#asterisk prometheanfire (~mthode@rrcs-24-173-105-84.sw.biz.rr.com) |
| 14:57.44 | malcolmd | caller id is retrieved using the callerid function: https://wiki.asterisk.org/wiki/display/AST/Function_CALLERID |
| 14:57.58 | malcolmd | ...or manipulated |
| 14:58.06 | prometheanfire | when is 1.4 not getting any fixes anymore? |
| 14:58.25 | malcolmd | prometheanfire: as of the current release; it's the last one |
| 14:58.26 | leifmadsen | prometheanfire: about a month ago |
| 14:58.37 | leifmadsen | ~asteriskversioning |
| 14:58.37 | infobot | extra, extra, read all about it, asteriskversioning is http://www.asterisk.org/asterisk-versions |
| 14:58.43 | prometheanfire | ah, thanks |
| 14:58.45 | ezano | no but |
| 14:58.50 | prometheanfire | sec too? |
| 14:58.54 | ezano | this is not what I want |
| 14:59.03 | leifmadsen | prometheanfire: documented on the wiki per that link |
| 14:59.05 | prometheanfire | the bot needs to be updated to the new url https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions |
| 14:59.07 | leifmadsen | updates infobot |
| 14:59.10 | prometheanfire | :D |
| 14:59.13 | leifmadsen | prometheanfire: yes I see that :) |
| 14:59.32 | ezano | if I set a callerid on (a2billing) and I call a number with a softphone I want to get the number of the softphone |
| 14:59.40 | ezano | and not the callerid define befor |
| 14:59.45 | leifmadsen | infobot: no, asteriskversioning is <reply> Asterisk versions and their support levels are documented at https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions |
| 14:59.45 | infobot | leifmadsen: okay |
| 15:00.36 | malcolmd | i don't know how a2billing works. if the softphone is tied to asterisk, you control its caller id. |
| 15:01.09 | *** part/#asterisk prometheanfire (~mthode@rrcs-24-173-105-84.sw.biz.rr.com) |
| 15:02.08 | ezano | yes, but I want to get the softphone number into the dialplan |
| 15:02.32 | ezano | I've not choice, but I don't know how to make that |
| 15:02.58 | malcolmd | okay....so make the first thing you do be to copy the callerid from the caller id function to some other variable, and then use it as you see fit. |
| 15:04.54 | *** join/#asterisk cerienjean (~iper@ALamentin-106-1-37-20.w90-43.abo.wanadoo.fr) |
| 15:04.54 | ezano | hum that will doesn't work |
| 15:05.54 | ezano | I want to use the softphone number to call VoiceMailMain($mybastardcallid@context) and so bypass the prompt to access voicemail |
| 15:07.14 | ezano | I tried many variable but nothing works |
| 15:08.01 | p3nguin | Are you guessing randomly? |
| 15:08.42 | ezano | obviously not |
| 15:08.53 | p3nguin | There is no variable that is your callerid. |
| 15:08.58 | p3nguin | At least not until you create it. |
| 15:10.30 | ezano | yes but I know that => MAVAR=${CALLERID(num)} or other than num but don't work |
| 15:11.04 | ezano | that return the callerid define into a2billing and not the softphone number |
| 15:11.08 | p3nguin | You also can't set a global in the globals section to contain caller ID. |
| 15:11.23 | leifmadsen | ya that |
| 15:11.31 | p3nguin | I can't understand what you're actually trying to accomplish. |
| 15:11.35 | leifmadsen | because ${CALLERID(num)} is a channel level function |
| 15:11.38 | leifmadsen | not a global function |
| 15:12.02 | ezano | okay leifmadsen, |
| 15:12.14 | leifmadsen | you can't know what the callerid is until you know what it is |
| 15:12.26 | leifmadsen | asterisk doesn't have res_prediction yet :) |
| 15:13.00 | ezano | hum, |
| 15:13.24 | ezano | but a softphone which call a number ? |
| 15:13.44 | ezano | it not send any data ? like his number ? |
| 15:13.48 | *** join/#asterisk KavanS (~KavanS@LINBIT/KavanS) |
| 15:14.09 | p3nguin | Set it to a variable in dial plan and then use that variable later. Put into an extension Set(myCID=${CALLERID(num)}) |
| 15:14.38 | malcolmd | <PROTECTED> |
| 15:15.00 | malcolmd | or see p3nguin ^ :D |
| 15:15.25 | *** join/#asterisk vinhdizzo (~vinh@pool-173-60-112-55.lsanca.fios.verizon.net) |
| 15:15.32 | ezano | I tried that yet :) |
| 15:15.45 | ezano | don't work :D |
| 15:16.11 | p3nguin | If you don't need to carry that data around, there's not really any reason to put it into a variable in the first place; just call the function directly within the app like leifmadsen pointed out in his example. |
| 15:16.31 | ezano | oh malcolmd I don't use voicemail.conf, I developped entirly a system to support voicemail in realtime |
| 15:16.36 | leifmadsen | I think malcolmd pointed it out, but ya that :) |
| 15:16.54 | p3nguin | I'll reitterate: I still have no flippin' idea what you're trying to accomplish. |
| 15:16.59 | leifmadsen | ezano: you developed it instead of using the existing developed voicemail in realtime"? :) |
| 15:17.02 | leifmadsen | p3nguin: +1 |
| 15:17.41 | ezano | the existing voicemail in realtime is not implemented correctly with a2billing |
| 15:17.47 | p3nguin | leifmadsen: I thought malcolmd said to put it into a variable, and then you used it directly. |
| 15:17.57 | p3nguin | but it doesn't matter, really. |
| 15:17.59 | ezano | (this is a2billing that I modify not asterisk ^^) |
| 15:18.05 | leifmadsen | I dont' think I used it directly I just wrote out the function name :) |
| 15:18.19 | p3nguin | That's what I mean by using directly. |
| 15:18.22 | leifmadsen | oic |
| 15:18.24 | leifmadsen | then continue on :) |
| 15:19.00 | p3nguin | I often catch myself setting variables to the data output by a function, only to use the variable within the next few lines of dial plan... |
| 15:19.14 | ezano | <p3nguin> I'll reitterate: I still have no flippin' idea what you're trying to accomplish. // ok example is more easy |
| 15:19.30 | ezano | imagine this dialplan |
| 15:19.30 | p3nguin | I smack myself, and then delete the setting of the variable, to put the function where I was previously using the variable value. |
| 15:19.33 | *** join/#asterisk cerberus_za (~coert@196-215-13-234.dynamic.isadsl.co.za) |
| 15:20.20 | ezano | exten => 5555,1,VoiceMailMain(${CALLERID(name)}@default) |
| 15:20.48 | ezano | if inside a2billing I set the callerid of an account at 1337 |
| 15:21.06 | ezano | and I call 5555, callerid will be 1337 |
| 15:21.32 | ezano | and not the softphone number which is the username of account sip/iax |
| 15:21.52 | ezano | it's better ? (my english is very limited so ...) |
| 15:22.03 | p3nguin | such as in exten => s,n,Set(myVar=${CALLERID(num)}) exten => s,n,VoiceMailMain(${myVar}@context) ... setting the variable was a waste of time and creates an unncessary line in dialplan. |
| 15:22.19 | Gugge | so you want the loginname of the sip-device, and not the callerid ? |
| 15:22.30 | ezano | voilaaaa |
| 15:22.35 | ezano | yes |
| 15:22.42 | Gugge | why do you ask about callerid stuff then? :) |
| 15:23.18 | ezano | because I developped a2billing, not asterisk it's confuse sometimes ^^ |
| 15:24.22 | ezano | so it's possible to get that ? :) |
| 15:25.09 | Gugge | no idea :) |
| 15:25.25 | Gugge | you could use setvar to set it to some var in sip.conf / realtime |
| 15:25.36 | Gugge | but i assume there is a way to read it without too :) |
| 15:26.48 | ezano | yeah I tried to use setvar without success |
| 15:26.54 | Gugge | maybe CHANNEL(peername) |
| 15:27.42 | p3nguin | If setvar didn't work, you did it wrong. |
| 15:27.49 | ezano | yes I know |
| 15:28.31 | ezano | but I don't understand how to use setvar |
| 15:28.56 | p3nguin | In the peer entry in sip.conf: setvar=someVar=value |
| 15:29.34 | p3nguin | In the dialplan, use ${someVar} to find out value. |
| 15:29.46 | ezano | yes but this is the same problem no ? |
| 15:29.55 | ezano | how get value now ? |
| 15:30.20 | p3nguin | You'll need to set the value when you write the setvar line. |
| 15:30.37 | leifmadsen | then just access it from the dialplan like any other channel variable |
| 15:31.10 | ezano | yes but I'll need one line by sip account ? |
| 15:31.20 | leifmadsen | huh? |
| 15:31.31 | Gugge | yes, each peer needs its own setvar entry |
| 15:31.35 | p3nguin | You'll need to ADD it to every account that you want to use the variable and have a value. |
| 15:31.41 | ezano | because value is hardcoded into sip.conf |
| 15:31.45 | leifmadsen | the variable you set in sip.conf with setvar will be set with the value you configured in sip.conf, and will be available when the peer creates a channel |
| 15:31.58 | leifmadsen | [my_peer] |
| 15:32.04 | leifmadsen | setvar=my_var=this_is_awesome |
| 15:32.06 | leifmadsen | <PROTECTED> |
| 15:32.08 | leifmadsen | extnesions.conf |
| 15:32.09 | Gugge | but if you need the peername, as far as i can tell, CHANNEL(peername) gives you that |
| 15:32.11 | ezano | okay |
| 15:32.16 | leifmadsen | Verbose(2,${my_var}) |
| 15:32.23 | leifmadsen | output would be "this_is_awesome" |
| 15:33.31 | sunfone | good morning all... |
| 15:33.33 | ezano | but I can't make that, I want a system which works entirly whith database, so one line by person is not possible. thanks |
| 15:33.45 | ezano | but I'll find another way I hope |
| 15:34.07 | ezano | Gugge: don't work ^^ |
| 15:34.12 | p3nguin | setvar can't be used with realtime? |
| 15:34.14 | *** join/#asterisk cerienjean (~iper@ALamentin-106-1-37-20.w90-43.abo.wanadoo.fr) |
| 15:34.28 | sunfone | off topic... but still kind of telephony... :) My mother in law is taking a trip to Paris next week, and wonders if she will be able to get a pay-as-you-go cell phone at the airport or very nearby... any advice? |
| 15:34.35 | Gugge | setvar works fine with realtime |
| 15:35.13 | p3nguin | That's what I thought. |
| 15:35.42 | Gugge | just set the field "setvar" to "somevar=value1;somevar2=value2" |
| 15:35.44 | Gugge | done |
| 15:36.55 | ezano | yes but I don't want anything hardcoded into sip.conf or others file |
| 15:37.01 | Gugge | then dont |
| 15:37.04 | Gugge | put a setvar in the db |
| 15:37.25 | Gugge | you can even make a view that automatically makes the setvar from the username field |
| 15:39.13 | Gugge | what output does ${CHANNEL(peername)} give you? |
| 15:39.48 | ezano | nothing |
| 15:39.56 | ezano | '' |
| 15:40.04 | Gugge | and which asterisk version do you use? |
| 15:40.47 | ezano | the last |
| 15:40.51 | Gugge | 10 beta? |
| 15:40.57 | ezano | 1.9 |
| 15:41.01 | Gugge | 1.9? |
| 15:41.19 | Gugge | that is some kind of strange version, as the one before 10 is 1.8 |
| 15:41.30 | p3nguin | This is some freaky Twilight Zone shit right here. |
| 15:41.37 | ezano | wait ^^ |
| 15:41.58 | *** join/#asterisk CryptixOverdrive (~cryptix@215.sub-174-252-80.myvzw.com) |
| 15:42.06 | ezano | yes 1.8 ^^ (a2billing -> 1.9) |
| 15:42.37 | *** join/#asterisk BenC[UK] (~bcummins@cpc2-lock3-2-0-cust77.6-1.cable.virginmedia.com) |
| 15:43.19 | BenC[UK] | hi guys - I am getting "no samples for g729tolin" loads in the console when using the g729 plugin with official licences... any ideas how to stop it? |
| 15:43.26 | Gugge | strange, peername is shown in core show function CHANNEL on both my 1.6.2 and 10 beta |
| 15:43.33 | Gugge | i would assume it is in 1.8 too |
| 15:43.43 | Gugge | but i guess you are stuck with setvar then |
| 15:44.15 | p3nguin | I bet it's there in the 1.8 branch too. |
| 15:45.05 | Gugge | p3nguin: but apparently not in the version ezano is running :) |
| 15:45.20 | p3nguin | He's a special case. |
| 15:47.12 | Gugge | maybe :) |
| 15:52.22 | ezano | peerip and channeltype send nothing too |
| 15:52.46 | ezano | yes I reload my dialplan ^^ |
| 15:53.17 | ezano | must be leave o/ |
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| 17:56.22 | *** join/#asterisk AlecTaylor (~AlecTaylo@unaffiliated/alectaylor) |
| 17:56.23 | AlecTaylor | hi |
| 17:58.41 | AlecTaylor | I want my website to show a call-in button for connecting directly into a conference call (via Flex/Flash or Javascript/JQuery or Java). Backend must require username auth and moderation capabilities. Which Asterisk and/or other packages do I need for this? |
| 18:00.16 | _Corey_ | AlecTaylor: You may want to look at Zoiper. They have a web-based embeddable softphone. |
| 18:02.04 | AlecTaylor | Hmm, they don't seem to be free. I have seen various free open-source SIP clients, but I'm confused at how to setup the Asterisk backend. Also, moderation and authentication seem to be equally confusing... |
| 18:02.34 | _Corey_ | I haven't seen a free embeddable one... |
| 18:08.59 | *** join/#asterisk caveat- (~false@newshell1.bshellz.net) |
| 18:10.26 | sunfone | Anyone know if this is valid: |
| 18:10.30 | sunfone | include => night|20:00-8:59|mon-fri|*|* |
| 18:10.45 | sunfone | i.e., can the time wrap around to the next morning and do what you would expect? |
| 18:11.52 | *** join/#asterisk moy (~moy@CPE002719f00364-CM0026f3a1701d.cpe.net.cable.rogers.com) |
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| 18:14.41 | Gugge | sunfone: if i had the need, i would just try :) |
| 18:15.13 | sunfone | kind of hard to test ;) |
| 18:15.33 | Gugge | why? |
| 18:15.35 | Gugge | set the time |
| 18:15.36 | Gugge | make a call |
| 18:15.37 | Gugge | done |
| 18:15.53 | sunfone | production system... |
| 18:16.05 | Gugge | why would you test on a production system? |
| 18:16.13 | sunfone | exactly |
| 18:16.17 | Gugge | virtual machines are easy to setup |
| 18:16.20 | sunfone | want to know that it works |
| 18:17.24 | sunfone | of course I can do that... was hoping someone just knew, so I didn't have to spend an hour doing it |
| 18:18.06 | atheos | sunfone, just add two includes back to back, one 8pm to midnight, one midnight to 9am |
| 18:18.37 | sunfone | ya, that was my fallback, but voipinfo has that as an example |
| 18:18.39 | atheos | still good to test things before production though |
| 18:19.16 | sunfone | it isn't the end of the world if this particular include doesn't work tonight, so I'll probably just through it in and see what happens |
| 18:20.03 | sunfone | ^through^throw^ |
| 18:24.24 | p3nguin | sunfone: No, that won't work. include wouldn't have any clue what that line of data means. |
| 18:24.36 | *** join/#asterisk caveat- (~false@newshell1.bshellz.net) |
| 18:25.45 | sunfone | kind of leaves saturday mornings undefined too :) |
| 18:26.01 | sunfone | I just split it up into morning and evening... two lines instead of one |
| 18:26.07 | p3nguin | I'll share mine with you. One moment. |
| 18:26.42 | sunfone | I think "TheBook" has an error on this topoc |
| 18:26.46 | sunfone | topic |
| 18:28.57 | p3nguin | http://pastebin.com/DD5HL5Yh |
| 18:29.57 | sunfone | hrmm, well that isn't really conditional contexts |
| 18:30.57 | p3nguin | That's actually exactly what I do. If it falls on the days and hours defined, the call would go to a specified context. |
| 18:31.35 | *** join/#asterisk Bipul (~bipul@unaffiliated/bipul/x-4918593) |
| 18:31.50 | sunfone | it seems you are making it more complicated this way... include is supposed to work as above |
| 18:32.03 | sunfone | (at least with times that don't wrap) |
| 18:32.18 | p3nguin | I'm not complicating it; this is how it works. |
| 18:32.25 | p3nguin | You were just trying to do something that isn't possible. |
| 18:33.10 | *** join/#asterisk Hanumaan (~Hanumaan@dslb-092-075-146-184.pools.arcor-ip.net) |
| 18:33.13 | sunfone | Hmm, lots of examples posted of include having time conditional capabilities - its even in "TheBook" |
| 18:33.32 | sunfone | I was merely wondering if I could wrap the times, as one of the examples I saw on voipinfo did that |
| 18:33.36 | sunfone | and it looked fishy |
| 18:33.44 | p3nguin | I've never seen include => night|20:00-8:59|mon-fri|*|* work. |
| 18:34.22 | sunfone | http://www.voip-info.org/wiki/view/Asterisk+tips+openhours |
| 18:34.26 | p3nguin | If it does, GREAT! I'll modify my notes accordingly. |
| 18:34.50 | sunfone | http://www.the-asterisk-book.com/unstable/einleitung-regex.html |
| 18:35.05 | sunfone | although I just noticed this is labeled "unstable" :):) |
| 18:35.39 | sunfone | woops that wasn't the right URL for the book |
| 18:35.59 | sunfone | http://www.the-asterisk-book.com/unstable/includes-im-dialplan.html |
| 18:36.08 | sunfone | scroll to the bottom |
| 18:36.42 | sunfone | but my problem with this bit is that it includes "night" always, so if it is daytime you end up with two extension "2000" in the same context |
| 18:36.51 | sunfone | Surely the behaviour there is undefined |
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| 18:38.07 | p3nguin | I would say that, if this type of inclusion does work, you've found a method of poor practice. |
| 18:38.32 | sunfone | for the wrapping I would agree |
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| 18:54.31 | sunfone | sigh, all that said, I can't seem to make it work at all |
| 18:54.35 | sunfone | I wonder if it is 1.6+ |
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| 18:58.17 | leifmadsen | sunfone: that "the book" is not ~thebook :) |
| 18:58.24 | techknowlust | hey guys. I'm trying to split some sip configs over a few files so I can give servers default files and define their extensions individually |
| 18:58.34 | techknowlust | is it possible to define a template twice ? |
| 18:58.34 | leifmadsen | techknowlust: #include |
| 18:58.45 | leifmadsen | it will read in the file as many times as you #include it |
| 18:59.12 | Kobaz | i'm really happy with 1.8 so far |
| 18:59.12 | techknowlust | leifmadsen: I'm already using includes, they're very handy |
| 18:59.15 | leifmadsen | techknowlust: then what is your real question? |
| 18:59.20 | leifmadsen | Kobaz: +1 |
| 18:59.25 | Kobaz | +2 |
| 18:59.26 | techknowlust | leifmadsen: if a template is defined twice will all the variables in both be applied to anything that uses that template |
| 18:59.48 | Kobaz | i have one more feature to write and then i'll be really happy |
| 18:59.56 | leifmadsen | techknowlust: I doubt defining it twice will work -- Asterisk would complain -- you should split it into separate templates |
| 19:00.21 | techknowlust | leifmadsen: and presumably you can't have an extension defined by two templates |
| 19:00.23 | leifmadsen | [my_awesome_peer](template1,template2) |
| 19:00.32 | techknowlust | oh you can ? |
| 19:00.32 | leifmadsen | sure you can |
| 19:00.34 | leifmadsen | heck ya you can |
| 19:00.39 | sunfone | leifmadsen: ahh! I didn't know there was an impostor out there! |
| 19:00.42 | techknowlust | that's awesome. thanks! |
| 19:00.45 | leifmadsen | :) |
| 19:00.52 | techknowlust | makes my life much easier now |
| 19:01.07 | techknowlust | many thanks :) |
| 19:01.21 | sunfone | leifmadsen: so do you cover time conditional includes in ~thebook? |
| 19:01.30 | sunfone | is it a 1.6 only feature? |
| 19:01.36 | sunfone | (or 1.6+) |
| 19:01.54 | leifmadsen | sunfone: well the book is 1.8 (latest book) |
| 19:02.04 | sunfone | ahh |
| 19:02.04 | leifmadsen | I've never heard of time based include => though |
| 19:02.14 | sunfone | it doesn't seem to work in 1.4 :) |
| 19:02.26 | *** join/#asterisk _Corey_ (~chatzilla@64.121.4.75) |
| 19:02.30 | sunfone | I don't have any 1.6 or 1.8 loaded... keep meaning to |
| 19:02.58 | hardwire | you should.. it's fun :) |
| 19:03.06 | leifmadsen | skip 1.6.x |
| 19:03.08 | hardwire | 1.8 and CEL are my faaavoooriiiites |
| 19:03.09 | leifmadsen | 1.8 is the bomb |
| 19:03.17 | leifmadsen | I'm excited for Asterisk 10 though :) |
| 19:03.23 | sunfone | ya.. I've heard about some of the wreckage ;) |
| 19:03.28 | p3nguin | I've never heard of it working. |
| 19:03.36 | hardwire | 10? |
| 19:03.36 | leifmadsen | p3nguin: troll |
| 19:03.43 | sunfone | heh |
| 19:03.43 | leifmadsen | ~asterisk10 |
| 19:03.43 | infobot | Asterisk 10 -- http://blogs.digium.com/2011/07/21/the-evolution-of-asterisk-or-how-we-arrived-at-asterisk-10/ |
| 19:03.44 | p3nguin | :( |
| 19:04.16 | leifmadsen | Asterisk 10 == Asterisk 1.10 s/1.// |
| 19:04.17 | sunfone | stuff about call pickup crashing things has me frightened |
| 19:04.29 | leifmadsen | sunfone: that was fixed for me in 1.8.5.0 |
| 19:04.34 | hardwire | no.. p3nguin .. you haven't heard of 10 working? |
| 19:04.42 | leifmadsen | I have several clients using it -- there was some edge case that I think was fixed recently though |
| 19:04.54 | Kobaz | very nice |
| 19:05.07 | leifmadsen | I've used Asterisk 10 for video conferencing and HD voice in ConfBridge() |
| 19:05.11 | leifmadsen | works quite well there |
| 19:05.15 | Kobaz | 1.8 running for two weeks and only using 227megs of ram |
| 19:05.17 | hardwire | I'm interested in ConfBridge |
| 19:05.23 | hardwire | Need to test out 10 soon. |
| 19:05.30 | Kobaz | and 215982 calls processed |
| 19:05.43 | hardwire | Waiting on tzafrir to make deb packages of dailys.. he never will :) |
| 19:05.55 | kn0x | why inflate the versioning numbers |
| 19:06.12 | hardwire | kn0x: change pisses people off.. makes them more productive. |
| 19:06.13 | Kobaz | because they can |
| 19:07.26 | kn0x | hardwire: lost you on that one |
| 19:08.12 | hardwire | kn0x: not really. |
| 19:09.11 | sunfone | penguin: used your pastebin do do what I needed, thanks ;) |
| 19:09.46 | leifmadsen | sunfone: welp, jsmith seems to say that include functionality should work and does exist |
| 19:09.54 | leifmadsen | <jsmith> leifmadsen: Absolutely! |
| 19:09.55 | leifmadsen | <jsmith> leifmadsen: One of the great (unknown) features of the dialplan :-) |
| 19:09.55 | leifmadsen | <jsmith> leifmadsen: Except that you probably now need commas instead of pipes between the fields |
| 19:10.49 | sunfone | leifmadsen: I tried pipes and commas... no worky |
| 19:10.57 | leifmadsen | just might not work anymore? |
| 19:11.10 | sunfone | does he think it should work in 1.4? |
| 19:11.51 | sunfone | I agree it is fantastic if it works |
| 19:12.45 | leifmadsen | he does think it should work |
| 19:13.10 | leifmadsen | could be a bug, hard to say -- not well known or used |
| 19:13.18 | sunfone | right |
| 19:13.20 | sunfone | bummer |
| 19:14.24 | p3nguin | I might try testing it later. |
| 19:15.57 | sunfone | Its nice if you want to conditionally include some call forwards (in my case) without messing with other extensions in the current context |
| 19:16.21 | sunfone | seemed elegant to me, anyway ;) |
| 19:18.19 | p3nguin | Call forwarding is done on my phones, so I don't know. |
| 19:19.41 | WIMPy | Isn't that the main use for AstDB? |
| 19:23.21 | *** join/#asterisk qakhan (~qakhan@180.178.144.12) |
| 19:23.52 | qakhan | hi everyone |
| 19:24.39 | qakhan | can anyone tell me how i enable call forwarding on asterisk i am use asterisk 1.4.38 version |
| 19:25.26 | WIMPy | qakhan: Buid your dilplan |
| 19:25.37 | p3nguin | Press the Fwd key on your phone. |
| 19:25.47 | Kobaz | the dillyplan |
| 19:25.57 | p3nguin | It's not something asterisk does. |
| 19:26.19 | qakhan | can u plz send the code |
| 19:26.20 | p3nguin | The phone provides a deflection of the call. |
| 19:26.41 | p3nguin | plz send the code? What is this, high school? |
| 19:27.19 | qakhan | sorry buddy but i m like a school boy in asterisk |
| 19:27.24 | WIMPy | AOL |
| 19:27.34 | qakhan | i need ur help |
| 19:27.44 | *** join/#asterisk brdude (~brdude@12.155.183.30) |
| 19:28.01 | Qwell | Please use proper English. If you can't be bothered to spell out words, we can't be bothered to help you. |
| 19:28.07 | WIMPy | Google will show you a lot of exaples. |
| 19:28.51 | qakhan | i tried but cloudnt found any help on google |
| 19:29.14 | qakhan | then i came here in a hope someone will help me |
| 19:30.12 | p3nguin | qakhan: Call forwarding is something your phone does. If you just want to send calls somewhere else, you'll be using Dial() in an extension. |
| 19:30.38 | qakhan | ok |
| 19:31.08 | qakhan | let me tell you what i want to do |
| 19:32.05 | qakhan | if someone call ext 2321 and ext doesnt answer in then call forwarding to user cell phone |
| 19:32.35 | p3nguin | Okay, so exten => 2321,1,... |
| 19:32.48 | p3nguin | That's how extension 2321 starts out. |
| 19:32.57 | p3nguin | We don't need to Answer() it. |
| 19:33.25 | p3nguin | To forward the call to a cell phone, press the Call Forward button on the phone and enter the cell phone number. |
| 19:33.41 | space1nvader | I think what he wants is |
| 19:33.54 | p3nguin | If I say "call forward button" a few more times, will that help you? |
| 19:34.02 | space1nvader | exten => 2321,1,Dial(SIP/2321, 10) |
| 19:34.09 | p3nguin | space fail |
| 19:34.23 | space1nvader | exten => 2321,2,Dial(IAX/trunk/<mobile-number>) |
| 19:35.04 | qakhan | ya right i know that |
| 19:35.09 | WIMPy | If I press the call forward button on my phone it just says Error when it's connected to Asterisk. |
| 19:35.27 | qakhan | but i want it to do dynamic |
| 19:36.27 | qakhan | i have 100 users, some user are required to forward their calls to their cell phone if they are not available on their ext |
| 19:37.20 | qakhan | they dial an ext like 2300 and enter their cell number to be call forward |
| 19:37.26 | p3nguin | wimpy: What kind of broken phone do you use? |
| 19:38.03 | WIMPy | Any standard phone. |
| 19:38.11 | qakhan | did anyone get me? |
| 19:38.55 | p3nguin | What defines it as being a "standard" phone? |
| 19:39.21 | *** join/#asterisk mykhyggz (~col@evolone.org) |
| 19:39.32 | WIMPy | Stuff you can buy in a local consumer electronics store. |
| 19:39.45 | WIMPy | I haven't seen SIP phones there, yet. |
| 19:40.28 | p3nguin | So you use an ATA or a card with an FXS module, I guess. |
| 19:41.09 | p3nguin | The ATA might have a key sequence for call forwarding, but obviously the phone itself can't send a SIP deflection message. |
| 19:41.39 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-kxmntoncwiwnlmrv) |
| 19:41.52 | qakhan | i am using SJphone |
| 19:42.13 | p3nguin | If it's a SIP phone, it probably has a call forwarding option. |
| 19:43.16 | qakhan | exten=s,1,Set(temp=${DB(CFIM/100)}) |
| 19:43.16 | qakhan | <PROTECTED> |
| 19:43.16 | qakhan | <PROTECTED> |
| 19:43.16 | qakhan | <PROTECTED> |
| 19:43.28 | qakhan | i found this on web |
| 19:43.45 | p3nguin | Now... do you have any clue what it means? |
| 19:43.47 | qakhan | and didnt understand y we have to use DB |
| 19:44.07 | qakhan | no dear |
| 19:45.09 | *** join/#asterisk CryptixOverdrive (~cryptix@123.sub-174-255-196.myvzw.com) |
| 19:50.40 | qakhan | u there? |
| 20:03.17 | *** join/#asterisk jeffspeff (~jeffspeff@173-11-144-149-houston.txt.hfc.comcastbusiness.net) |
| 20:03.18 | jeffspeff | on the spa504g you can have the same extension on multiple line keys; which is easily configured via the web gui... however the 7945g has to be configured for sip using .cnf.xml files; how do i accomplish he same feature of the same extension on multiple line keys? |
| 20:11.36 | cerienjean | Hi all... |
| 20:11.56 | cerienjean | I've been doing a load test on asterisk using sipp. I could not really exceed 140 calls |
| 20:12.04 | cerienjean | while I exepect more |
| 20:12.30 | cerienjean | The calls were failing on the cli with: rtcp too many open files |
| 20:12.46 | cerienjean | I've tried to increase the ulimit, no luck |
| 20:13.05 | cerienjean | any suggestion as to articles / pages / info on how to increase the asterisk / linux performances ? |
| 20:15.27 | *** join/#asterisk tzafrir (~tzafrir@bzq-218-155-147.cablep.bezeqint.net) |
| 20:17.48 | malcolmd | what does "ulimit -a" return? |
| 20:19.54 | cerienjean | hi malcom - unfortunately, the access to the server is complicated and I dont have access right now. I was searching 'offline' ideas |
| 20:19.56 | leifmadsen | malcolmd: awesomesauce |
| 20:20.21 | cerienjean | I did a ulimit -n 32768 |
| 20:20.35 | cerienjean | do I need to restart asterisk after such command ? |
| 20:20.39 | Qwell | yes |
| 20:20.44 | cerienjean | ie core stop / asterisk |
| 20:20.45 | cerienjean | tks |
| 20:20.51 | Qwell | The terminal from which Asterisk is run needs to have that set. |
| 20:20.57 | cerienjean | ok - got it |
| 20:21.00 | cerienjean | :m-) |
| 20:21.02 | cerienjean | :-) |
| 20:21.13 | malcolmd | also, if you're starting asterisk from an init script, you need to make sure your init script sets the ulimit before it starts asterisk. |
| 20:21.14 | cerienjean | hence, it did not improve anything ! |
| 20:21.22 | cerienjean | manually for the time being |
| 20:21.29 | malcolmd | cool |
| 20:23.16 | cerienjean | are there any other type of optimizations, (I now linux, but not to optimize the kernel) |
| 20:25.43 | malcolmd | you can run asterisk with a higher priority; you can renice it. don't put it ahead of your ssh or console process though or else logging into the machine remotely could become problematic ;) |
| 20:26.04 | cerienjean | ok - I vaguely know about nice |
| 20:26.26 | cerienjean | but the cpu didnt seem to be the issue, according to top |
| 20:26.42 | malcolmd | here's a great post from Matthew Roth about file limits: http://lists.digium.com/pipermail/asterisk-users/2006-April/147204.html |
| 20:27.33 | cerienjean | thanks ! :-) |
| 20:27.42 | malcolmd | something else is probably amiss then; normally, asterisk goes south when your CPU is otherwise occupied, e.g. you're doing tons of calls and the cpu can't keep up between asterisk and other system tasks |
| 20:27.52 | *** join/#asterisk sustav (~alfa@nat/digium/x-pkynewepxhnmzfom) |
| 20:29.04 | cerienjean | yeah... first tests were doing gsm/g729 transcoding, and that was seriously limiting the number of calls.... by loading appropriate message files, we doubled the number of calls |
| 20:29.09 | cerienjean | and the CPU is not maxed out |
| 20:34.50 | *** join/#asterisk xpot-mobile (~james@166-70-100-198.ip.xmission.com) |
| 20:38.58 | *** join/#asterisk lanning (~lanning@173-8-187-197-SFBA.hfc.comcastbusiness.net) |
| 20:39.48 | *** part/#asterisk wesphillips (~wphill04@137-237-233-124.harris.com) |
| 20:43.42 | *** join/#asterisk JasonL (~jason@216.223.114.3) |
| 20:44.48 | *** join/#asterisk evil_gordita (robert@ip70-188-50-186.rn.hr.cox.net) |
| 20:46.15 | *** join/#asterisk corretico (~luis@201.201.44.82) |
| 20:46.17 | *** join/#asterisk tonsofpcs (~tonsofpcs@cpe-69-205-240-64.stny.res.rr.com) |
| 20:48.35 | *** join/#asterisk dailylinux (~test@88.87.48.55) |
| 20:50.22 | *** join/#asterisk _Corey_ (~chatzilla@64.121.4.75) |
| 20:52.08 | p3nguin | I don't really know my way around jira... is there any serious problem with 1.8.5.0 right now? |
| 20:52.37 | p3nguin | Thinking of upgrading a 1.8.4.4 package to 1.8.5.0 if it's good to go. |
| 20:53.01 | leifmadsen | p3nguin: I'm using 1.8.5.0 in production on at least.... 3-4 systems (customers) where 1.8.4.4 was not stable for me |
| 20:53.16 | p3nguin | Any special patches that you used? |
| 20:53.18 | leifmadsen | nope |
| 20:53.24 | leifmadsen | I try to avoid anything custom |
| 20:53.32 | *** join/#asterisk sustav (~alfa@nat/digium/x-tvnigquraamcxrmn) |
| 20:53.33 | p3nguin | Sounds good. I'll go for 1.8.5.0. |
| 20:53.40 | leifmadsen | ymmv :) |
| 20:53.56 | p3nguin | And when it does, YOU'LL HEAR ABOUT IT! |
| 20:54.21 | p3nguin | I'll post it on your g+ home page. |
| 20:54.28 | leifmadsen | p3nguin: sure! :) |
| 20:54.32 | leifmadsen | rarely checks that |
| 20:54.44 | leifmadsen | but then again I only check my facebook about once a week |
| 20:54.49 | p3nguin | Or maybe I won't be able to, since you never put me into your circle. |
| 20:54.55 | leifmadsen | mwahahahahaha |
| 20:55.11 | leifmadsen | p3nguin: you probably don't show up as p3nguin on my G+ stuff |
| 20:55.47 | p3nguin | I do have that listed under "other names," but I don't know who can see that. |
| 20:56.08 | leifmadsen | I didn't look that closely from my phone :) |
| 20:56.25 | p3nguin | I'd imagine you'd have to actually take the time to view my profile before you'd see it, anyway. |
| 20:56.31 | robl^laptop | leifmadsen: you should have a slightly larger royalty check from O'Reilly next time. I now have a print edition 3rd Ed. sitting next to my 1st and 2nd ;-) |
| 20:56.38 | p3nguin | I don't expect people to check me out just because I add them to a circle. |
| 20:57.01 | *** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital) |
| 20:57.06 | leifmadsen | robl^laptop: w00t another nickle! :) |
| 20:57.14 | p3nguin | booooo |
| 20:57.27 | leifmadsen | robl^laptop: I'm hoping it'll be OK this time around because the book wasn't released in time to hit the last cheque round |
| 20:57.41 | leifmadsen | which means it'll have the maximum amount of time to accumulate royalties :D |
| 20:58.16 | p3nguin | Two words... |
| 20:58.19 | p3nguin | cha ching |
| 20:58.24 | robl^laptop | leifmadsen: nickel!?!? should have been at least 25 cents -- considering it has more pages now. ;-) |
| 20:59.01 | p3nguin | ehhh, okay that was weird. |
| 20:59.06 | leifmadsen | robl^laptop: you'd think so eh?! :) |
| 20:59.16 | leifmadsen | honestly has no idea how much he makes per book.... |
| 20:59.20 | leifmadsen | now I'm curious lol |
| 20:59.30 | p3nguin | Right after I said cha ching, a Rally's commercial said "CHA-CHING." |
| 20:59.35 | chazzam | lol, do you get less if we buy it at a discount? |
| 20:59.50 | leifmadsen | yes |
| 21:00.00 | jaytee | p3nguin, quick! what's tonight's megamillions numbers? |
| 21:00.00 | leifmadsen | we get a percentage of what o'reilly gets |
| 21:00.02 | chazzam | well, you do know something about it then |
| 21:00.04 | chazzam | ahh |
| 21:00.11 | *** join/#asterisk Brixius (~Kelly@PDN-VBA.OnvoyInc.fw.onvoy.net) |
| 21:00.26 | chazzam | well... you got a smaller percentage from me then, but I did buy both digital and print |
| 21:00.52 | leifmadsen | lol... I think it's like $1 per book |
| 21:00.53 | p3nguin | jaytee: I wish I knew! I'd take everyone to Rally's or Checkers for lunch tomorrow. |
| 21:01.19 | p3nguin | or Monday |
| 21:01.22 | leifmadsen | (ya, no one is getting rich writing technical books here) |
| 21:01.47 | chazzam | except o'reilly, because they do it in bulk? |
| 21:03.06 | robl^laptop | leifmadsen: so for every 6 books you can go buy a coffee at Starbucks ;-) |
| 21:03.15 | leifmadsen | robl^laptop: yep..... |
| 21:03.35 | leifmadsen | pretty much the hours spent writing books amounts to a new toy every few months |
| 21:04.54 | p3nguin | I'd like to get an outdated iPod touch, if you're looking to get yourself a new 4th gen model. |
| 21:05.00 | p3nguin | Just sayin'. |
| 21:05.35 | leifmadsen | whatever money I get this time around will likely be going directly on the credit card I used to pay for chunks of my wedding :) |
| 21:05.47 | robl^laptop | leifmadsen: any idea on the number of copies sold per edition? it would be intersting metrics to see the trend |
| 21:06.02 | p3nguin | You actually went through with it?! I thought it was a joke. |
| 21:06.21 | leifmadsen | robl^laptop: ya I think I entered all the data into a spreadsheet once.... I'm not sure where I put it now |
| 21:07.43 | *** join/#asterisk albertoandrade (~albertoan@187.59.36.91) |
| 21:08.31 | leifmadsen | robl^laptop: sorry, not sure where I put that spreadsheet :( |
| 21:08.45 | leifmadsen | basically it does this: \ |
| 21:08.52 | leifmadsen | :) |
| 21:09.10 | leifmadsen | if we get lucky it does this.... |
| 21:09.12 | leifmadsen | \ |
| 21:09.13 | leifmadsen | <PROTECTED> |
| 21:09.16 | leifmadsen | <PROTECTED> |
| 21:09.19 | leifmadsen | <PROTECTED> |
| 21:13.04 | *** join/#asterisk raden (~J@66-168-15-100.dhcp.stpt.wi.charter.com) |
| 21:13.12 | raden | Katty, :D :D :D :D |
| 21:17.53 | Kobaz | what should i work on now |
| 21:21.33 | Kobaz | (A) safe ael loading (C) upgrading customers to 1.8 (D) setting up xen (E) put bigger drives in the backup sever (F) write more unit testing or (G) work on one of my 'new' snowblowers |
| 21:21.51 | Kobaz | oh, i missed an option B |
| 21:22.34 | leifmadsen | PROFIT |
| 21:22.51 | Kobaz | yeah |
| 21:22.51 | leifmadsen | votes for {F} |
| 21:23.03 | Kobaz | yeah but F wouldn't benefit you guys |
| 21:23.12 | leifmadsen | WHY NOT?! |
| 21:23.15 | Kobaz | It's my own unit testing for apps |
| 21:23.19 | leifmadsen | pffft |
| 21:23.25 | Kobaz | indirectly it does... i've found lots of bugs in 1.8 with my tests |
| 21:23.27 | leifmadsen | I guess {A} then |
| 21:23.34 | leifmadsen | I still like {F} more |
| 21:23.37 | Kobaz | hehe |
| 21:23.39 | leifmadsen | writes more dialplan |
| 21:23.42 | leifmadsen | doesn't care about AEL |
| 21:23.46 | Kobaz | safe ael loading would be really good for me |
| 21:24.19 | Kobaz | on successful ael load, i want to have asterisk copy it to a seperate spot to be the 'known working good version' |
| 21:24.38 | Kobaz | so if you have a syntax error in your ael, and you restart asterisk, you wont have a totally broken system |
| 21:24.49 | Kobaz | it'll load the last know good files |
| 21:24.54 | Kobaz | known.. |
| 21:25.04 | Kobaz | maybe put it in sqlite |
| 21:25.59 | leifmadsen | Kobaz: do that for dialplan too :) |
| 21:26.34 | *** join/#asterisk anonymouz666 (~anonymouz@187-28-37-118.poolip.RJO.embratel.net.br) |
| 21:27.18 | Kobaz | does asterisk do syntax checking on extensions.conf? |
| 21:27.37 | anonymouz666 | anyone in here already tried to use FUNC_CURL accessing self created cert for HTTPS ? |
| 21:27.41 | anonymouz666 | it just does not work |
| 21:27.53 | Kobaz | i haven't worked with extensions.conf in forever |
| 21:27.54 | anonymouz666 | even with CURLOPT(ssl_verifypeer)=off |
| 21:28.00 | Kobaz | foooorrreeeevvveeeeer |
| 21:32.55 | p3nguin | How little system memory is the LOW_MEMORY asterisk compile option intended to accommodate? |
| 21:33.39 | anonymouz666 | 1 MB like my first computer... 286 |
| 21:33.57 | anonymouz666 | j/k |
| 21:34.38 | anonymouz666 | with a game called "Test drive" and sound coming from PC speaker |
| 21:38.55 | *** join/#asterisk tris (tristan@2001:1868:a00a::4) |
| 21:46.19 | JonathanRose | The Commodore 64 had great sound for what it was. It always amazed me how crappy my first PC sounded coming up from that. |
| 21:46.37 | JonathanRose | Then I got my first sound card and things got better. |
| 21:46.55 | anonymouz666 | Sound Blaster 16? |
| 21:47.03 | JonathanRose | No idea. I was probably 6. |
| 21:47.38 | JonathanRose | By the time I was actually buying my own stuff, sound cards had basically become a thing of the past. |
| 21:47.59 | anonymouz666 | oh yes |
| 21:48.22 | JonathanRose | At the time though, they'd bundle games with pretty much any piece of hardware. |
| 21:48.31 | JonathanRose | So you get a CD Rom drive, it came with a bunch of games. |
| 21:48.34 | JonathanRose | Ditto for sound cards. |
| 21:48.48 | JonathanRose | I think mine came with Jurassic Park. |
| 21:49.22 | JonathanRose | I guess the sound card companies wanted to demonstrate what a pain in the butt it was to actually get the thing working with DOS programs. |
| 21:50.17 | anonymouz666 | magic tree in MSX |
| 21:50.59 | anonymouz666 | MS-DOS 5.0 with xtreegold |
| 21:58.56 | *** part/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 21:59.11 | *** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital) |
| 22:15.56 | leifmadsen | <3 MS-DOS 5.0 |
| 22:18.21 | *** join/#asterisk cerienjean (~iper@95.138.77.91) |
| 22:20.41 | *** join/#asterisk RypPn (~RypPn@unaffiliated/ryppn) |
| 22:36.35 | cerienjean | malcomd / qwell: managed to get access to the box |
| 22:36.57 | cerienjean | I am now handling 400 simultaneous calls (200 in, bridged to 200 out) |
| 22:37.11 | cerienjean | and then no more bandwith on my machine.... |
| 22:37.20 | cerienjean | so it s good.... i was targetting 150 ! |
| 22:37.26 | cerienjean | x2 |
| 22:37.35 | cerienjean | many thanks |
| 22:51.41 | *** join/#asterisk cj (~cjac@2607:ff08:f5:3a::3) |
| 22:51.45 | cj | hey folks |
| 22:51.51 | cj | carrar: I got a tour of the WA EOC yesterday |
| 22:52.16 | cj | pabelanger: have you folks built 1.8.x on sparc64 yet? |
| 22:52.30 | cj | x1user: are you running 1.8.x? |
| 22:52.40 | cj | got an x1 the other day |
| 22:53.00 | x1user | cj: nope |
| 22:53.14 | cj | x1user: what version have you found works best? |
| 22:55.46 | x1user | cj: i am asterisk noob, really. |
| 22:56.00 | cj | and is it still nebs if I run debian on it? |
| 22:56.12 | cj | x1user: yeah, me too. :) |
| 22:56.46 | p3nguin | Is there a good place on IRC to find someone (not a n00b) to build a simple web site? (for pay, of course) |
| 22:57.14 | *** join/#asterisk cerberus_za (~coert@196-215-13-234.dynamic.isadsl.co.za) |
| 22:59.36 | cj | p3nguin: #perl on irc.perl.org prolly |
| 23:01.48 | *** join/#asterisk klarrimore (~klarrimor@adsl-99-146-26-130.dsl.lsan03.sbcglobal.net) |
| 23:03.04 | cj | hi, Tim_Toady! |
| 23:03.52 | cj | sorry I missed your talk this year. I told Gloria I'd give you a copy of the one from 2006, but I've misplaced it. When I find it, I'll take a copy and send it to the address in your whois entry :) |
| 23:06.49 | *** join/#asterisk xpot-mobile (~james@166-70-100-198.ip.xmission.com) |
| 23:16.12 | *** join/#asterisk ferdna (~ferdna@cpe-67-10-220-35.elp.res.rr.com) |
| 23:17.35 | ferdna | can you tell asterisk to listen for RTP at 5004 and 10000 - 20000? |
| 23:27.13 | leifmadsen | nope |
| 23:27.17 | leifmadsen | just start and end ranges |
| 23:27.38 | leifmadsen | tell asterisk to listen in a larger range then filter with iptables |
| 23:28.17 | hardwire | well.. allocate. |
| 23:28.35 | hardwire | thats sort of a problem. |
| 23:28.47 | hardwire | it has no idea a port it allocated is blocked until it's too late. |
| 23:30.27 | *** join/#asterisk eja (~user@75.110.195.31) |
| 23:31.31 | ferdna | leifmadsen, hardwire thank you guys... |
| 23:31.44 | ferdna | leifmadsen, i will try that... thanks =) |
| 23:31.53 | hardwire | err. |
| 23:31.59 | hardwire | best of luck then :) |
| 23:32.13 | eja | hi guys... just wondering about the output of "sip show peers". what constitutes an unmonitored device vs monitored? and online vs offline? |
| 23:32.33 | cj | eja: option=yes, I think |
| 23:32.41 | cj | er, qualify=yes |
| 23:32.46 | cj | and whether the option request comes back |
| 23:32.50 | cj | or ICMP |
| 23:33.05 | cj | open tcpdump and unplug the 8p8c |
| 23:33.07 | cj | watch the result |
| 23:33.19 | eja | does offline mean more like inactive? b/c if it shows up in "sip show peers" it's connected right? |
| 23:33.39 | eja | lol right on cj. first time i've seen someone refer to it as 8p8c instead of rj45 :) |
| 23:35.50 | *** join/#asterisk xpot-mobile (~james@166-70-100-198.ip.xmission.com) |
| 23:43.38 | hardwire | wow.. so that made me look at how rtp ports are selected. |
| 23:44.00 | hardwire | aaand I'll be submitting a patch that reduces the context switching. |
| 23:45.34 | hardwire | apparently rtp port selection gets slower as more ports are in use.. so thats no good. |
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