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| 02:56.21 | wyoung | WOooo!! The correct answer was: originateAction.setVariable("__SIPADDHEADER51", "Call-Info: sip:\\;answer-after=0"); |
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| 03:16.41 | gruetzkopf | the toolchain says max clock is 70MHZ (for a 4-core quad-issue design), it still works at 100MHz |
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| 06:10.13 | gavimobile | I understand there are a few methods to setup faxing with asterisk. I currently have asterisks 11 up and running. I also have a spa2102 spa adapter with t.38 support. what would be the best method to use for sending faxes with asterisk? |
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| 10:06.29 | Quadrant | is gerrit down ? |
| 10:06.56 | Quadrant | [root@axcent src]# git clone -b 11 http://gerrit.asterisk.org/asterisk asterisk-11 |
| 10:06.56 | Quadrant | Initialized empty Git repository in /usr/src/asterisk-11/.git/ |
| 10:06.56 | Quadrant | error: RPC failed; result=22, HTTP code = 405 |
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| 16:27.58 | nickgaw | Hi, With the asterisk packages in order to get the sample pbx working where it has the english voices working as if I dial using the cli an extention I get the message language not registerd what package should I install in order to make sure the sample stuff works? |
| 16:28.24 | nickgaw | I am using on my laptop debian stretch |
| 16:29.33 | lvlinux | nickgaw: you mean the apt-get packages? |
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| 16:30.21 | nickgaw | Yes as I was wanting to test out on my local system not the centos server if asterisk cli could connect to that test server and do basic things locally. |
| 16:31.23 | lvlinux | ah. k. I think the soundfiles are in the package "asterisk-core-sounds-en" |
| 16:31.43 | lvlinux | so try that with "apt-get install asterisk-core-sounds-en" |
| 16:32.34 | lvlinux | How are you dialing with the cli? originate? |
| 16:33.53 | nickgaw | That package was already installed if I do console dial 6000 which I believe is the congrats message I get nothing. |
| 16:35.45 | lvlinux | Hmm, I'm not familiar with dialing on the console using methods other than originate. I think they deprecated that after v 1.8 |
| 16:36.03 | nickgaw | ok that might be the issue. |
| 16:36.40 | lvlinux | but i think 1.8 is the version of that package you have with Debian testing. |
| 16:37.09 | lvlinux | does it give any text output? |
| 16:37.15 | lvlinux | or just goes back to prompt? |
| 16:37.32 | nickgaw | lots of text output about dialing into channels. it is version 13. |
| 16:37.42 | lvlinux | You may need to "core set verbose 4" to get it to show details. |
| 16:37.43 | lvlinux | ah ok. |
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| 16:38.03 | lvlinux | really? debian testing has v13? wow didn't realize it was that recent. |
| 16:38.18 | nickgaw | 13.8.2 |
| 16:38.48 | lvlinux | wow yeah thats just a version or two behind current |
| 16:39.27 | nickgaw | Is version 0.1.0 the earliest version or can I get earlier versions then that in source form? |
| 16:40.04 | lvlinux | The earliest I've seen available is version 1.0 so I don't know. Why? |
| 16:40.45 | nickgaw | just wanted to look at how it has evolved in source as version 0.1.0 won't compile on debian stretch. |
| 16:41.29 | lvlinux | Yes I suspect there would be trouble compiling anything below 1.8 or 1.6 on a modern distro. |
| 16:42.22 | nickgaw | what is the latest svn branch other then 13 if I wanted to get the latest development code for testing on debian? |
| 16:42.30 | lvlinux | Do you have a hardphone yet or are you going to start with a softphone? |
| 16:43.23 | lvlinux | I think they are working on v14, which is probably what you get in svn |
| 16:43.45 | nickgaw | Still looking for a good hardware phone probably a sip phone that connects to the router as we have two numbers so we would want two phones one per number should I look at getting the same brand or different to know what phone goes to what number? |
| 16:44.31 | nickgaw | What is a good softphone that is command line and works under linux and windows? |
| 16:45.09 | lvlinux | I don't know about windows, but PJSUA is a good one for Linux. |
| 16:45.50 | lvlinux | You don't have to have two phones if you don't want to, but you can. It's up to you. Asterisk can be made to operate however you like. |
| 16:46.36 | lvlinux | Course if you need to take two calls at once, two phones might be easier to keep up with instead of using hold on one phone. |
| 16:46.59 | nickgaw | I would like to have two phones as we have two numbers and I would like to have each phone work with each number. |
| 16:47.14 | lvlinux | That'll work. |
| 16:47.19 | nickgaw | one for one number the other for the other one. |
| 16:48.16 | lvlinux | Just out of curiosity, what is your need for the setup you want, with the call recording on independent channels and the two separate voice menus, etc.? |
| 16:50.49 | nickgaw | The purpose for the menus is for each number I want it to ask the user if they want to talk to someone they can dial their extension and if not it would just transfer to the default extension for that number and for the two channel recordings mainly for interest and as I do a lot of recording of blindness conference calls in my area and like to have my own copy I would like my voice to be on another channel from their voice. |
| 16:51.29 | lvlinux | Ah ok. Yes that makes sense. |
| 16:52.15 | lvlinux | Hows your reading of the book coming along? Is it still working ok with your reader? |
| 16:53.29 | nickgaw | yes I transfered the book to one of these external media players that can read text documents as I converted the pdf to html for easey reading and so I can read on one device and do the computer without having to jump between windows. |
| 16:54.36 | lvlinux | I'm thinking mainly of the code segments, does it know how to distinguish stuff like that from regular text so that it doesn't try to pronounce things like Dial(PJSIP/kfs8,65,mk) as a sentence? |
| 16:54.55 | nickgaw | In the credits file for that music as the files are in sln format and I would like them in good quality mp3 or wav if possible would writing the authors on these musical beds about getting higher quality versions for my own listening pleasure be the best thing to do? |
| 16:57.29 | lvlinux | you could try it. But sln is the same quality as wav, just in a different container. |
| 16:57.45 | lvlinux | So I don't think you'd get anything of any better quality. |
| 16:57.51 | nickgaw | How can I play sln files? |
| 16:58.01 | lvlinux | do you have ffplay? |
| 16:58.11 | lvlinux | (part of ffmpeg) |
| 16:58.28 | lvlinux | you can do it with that. Maybe with VLC too. |
| 16:58.40 | nickgaw | I was going to play them on windows do you know if winamp has a sln plugin? |
| 16:58.45 | lvlinux | Or you can convert them to WAV with ffmpeg or SOX. |
| 16:58.59 | lvlinux | Have no idea about Winamp. I don't use Windows at all anymore. |
| 16:59.16 | lvlinux | Converting them would probably be your best bet. |
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| 17:00.47 | nickgaw | I have sox installed on my debian system how can I convert them to wav files at the same bit rate and quality? |
| 17:03.07 | lvlinux | not sure with sox |
| 17:03.22 | nickgaw | What tool do you mainly use? |
| 17:03.40 | lvlinux | ffmpeg |
| 17:06.15 | nickgaw | Do you play sln files in ffmpeg or just convert? |
| 17:07.10 | lvlinux | i personally never use them---for asterisk I always use g.722 or g.711u |
| 17:07.28 | nickgaw | Is g.722 the higher quality? |
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| 17:07.38 | lvlinux | not than sln. |
| 17:07.53 | nickgaw | is it higher then g.711? |
| 17:07.58 | lvlinux | yes |
| 17:08.29 | lvlinux | g.711 is standard telephone quality |
| 17:08.39 | lvlinux | g.722 is wideband HD Voice telephone quality |
| 17:09.09 | lvlinux | sln is uncompressed PCM, similar to a CD (but it can be lower quality than CD) |
| 17:09.44 | lvlinux | sln is like what you'd normallly find in a WAV file, only it doesn't have the WAV file header and format information in it. |
| 17:10.36 | nickgaw | For my server probably using g.722 would be best how can I convert files to this format? |
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| 17:11.35 | lvlinux | You mean files other than the ones you already downloaded for Asterisk? |
| 17:11.45 | lvlinux | (which come in g.722 versions already) |
| 17:12.03 | lvlinux | You can convert them with ffmpeg (I don't think sox will convert to g.722) |
| 17:12.19 | nickgaw | yes like my ivr message rather then mp3 format. |
| 17:13.39 | lvlinux | Well if you already have it recorded, you can do this: |
| 17:13.41 | lvlinux | ffmpeg -i /tmp/sample.wav -ar 16000 -acodec g722 /tmp/sample.g722 |
| 17:14.01 | lvlinux | But if you don't already, I'd just use the phone to record it straight into g722. |
| 17:14.32 | lvlinux | (with Asterisk I mean) |
| 17:15.00 | lvlinux | gtg. might be back later. |
| 17:15.21 | nickgaw | Currently they are in mp3 recorded with a good quality recorder with not a lot of noise. I did install the mp3 support with the asterisk server. |
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| 19:52.50 | darkdev | Hello! I'm newbie with Asterisk, please tell me, how can I decrease time between extension is going from one command to next? The main goal is to record voice right after played file, but it is little delay (about 2 seconds). And how can I run parallel commands? Thank you very much! |
| 19:59.50 | lvlinux | darkdev: I'm not sure about how to solve your 2 second delay, however, one way to sortof run parallel commands is using local channels (although that might not help with what you are needing specifically). |
| 20:01.00 | darkdev | lvlinux: Thank you! Can you give an url to article or example, which can describe, how to do it? |
| 20:02.04 | lvlinux | well, you use the Dial() application to call multiple extensions at once: |
| 20:02.39 | lvlinux | exten => 1234,1,Dial(Local/456@context&Local/789@context) |
| 20:03.10 | lvlinux | so when something enters the dialplan at exten 1234, it get's split and goes to both 456@context and 789@context simultaneously. |
| 20:04.35 | darkdev | Oh.. It is very good, thank you very much! So, don't you know, how to decrease time between that, for example? |
| 20:04.39 | darkdev | exten => en,n,Playback(at-tone-time-exactly) |
| 20:04.39 | darkdev | exten => en,n,SayUnixTime(${FutureTime},,IM 'and' S 'seconds' p) |
| 20:04.57 | lvlinux | More info: https://wiki.asterisk.org/wiki/display/AST/Local+Channel+Examples |
| 20:05.25 | lvlinux | Not sure about decreasing the time though. |
| 20:07.08 | darkdev | lvlinux: thank you for your help! :) |
| 20:07.32 | lvlinux | no problem! |
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| 21:38.58 | xochilpili | hi all |
| 21:39.18 | xochilpili | im trying to connect 2 asterisk servers using this how-to: http://www.asteriskdocs.org/en/2nd_Edition/asterisk-book-html-chunk/connecting_two_asterisk.html |
| 21:40.04 | xochilpili | then when i call from server_A to server_B, in CLi on server_a just said: Called SIP/server_b/1100 |
| 21:40.28 | xochilpili | but nothing else, is not calling |
| 21:40.36 | xochilpili | and server_b i have nothing |
| 21:40.58 | xochilpili | sip show registry shows me the registratio to that server_b |
| 21:42.07 | xochilpili | i have disabled firewall in both servers, and both servers in sip show registry output "regitered" |
| 21:42.37 | xochilpili | any hand? |
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| 21:53.04 | wuffi600 | hi |
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| 22:06.21 | [TK]D-Fender | xochilpili, Prove the packets are making it |
| 22:11.04 | xochilpili | [TK]D-Fender, now i got another issue; chan_sip.c:16630 check_auth: username mismatch, have <100>, digest has <server_b> |
| 22:11.43 | xochilpili | as in the how-to guide, using register => user:pass@server_a/user_server_b |
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| 22:11.58 | xochilpili | must be solve this but it isnt |
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| 22:38.10 | [TK]D-Fender | Clearly a name mismatch with peers using that host |
| 22:38.20 | [TK]D-Fender | having MULTIOPLE would be a clear screwup |
| 22:38.28 | [TK]D-Fender | Also Never link to some guide you supposedly followed |
| 22:38.45 | [TK]D-Fender | Because we have no reason to believe you did so properly, or that something else isn't happening |
| 22:38.54 | [TK]D-Fender | Sho show something useful |
| 22:57.39 | *** join/#asterisk Oatmeal (~Suzeanne@75-103-145-152.ccrtc.com) |
| 23:00.45 | *** join/#asterisk klow (~textual@c-67-161-106-89.hsd1.wa.comcast.net) |
| 23:29.31 | *** join/#asterisk Navion (~navion@rivendell/users/navion) |
| 23:51.36 | *** join/#asterisk wil_syd2 (~wil_syd@c110-20-159-70.rivrw10.nsw.optusnet.com.au) |