| 00:12.08 | *** join/#asterisk keycruncher (~Adium@c-174-55-112-94.hsd1.pa.comcast.net) |
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| 00:51.18 | koffel | how can i block attacks on my asterisk box |
| 00:52.40 | WIMPy | Disconnect from the internet. |
| 00:54.56 | koffel | besides that |
| 01:04.10 | Carlos_PHX_ | We use Fail2Ban with great success. |
| 01:06.37 | koffel | i tried fail2ban |
| 01:06.43 | koffel | and they do call from |
| 01:06.46 | *** join/#asterisk deo (~deo@122.53.72.218) |
| 01:06.47 | koffel | attack |
| 01:12.55 | koffel | carlos here my log http://pastebin.com/sVc95hMD |
| 01:26.35 | resist0r | koffel: iptables might be a start |
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| 01:43.22 | *** mode/#asterisk [+o pabelanger] by ChanServ |
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| 02:07.50 | deo | guys good day, how do we change the record file name in asterisk? the files that are in /var/spool/asterisk/monitor/xxxxxxxx.wav |
| 02:08.23 | deo | i need to rename it to whatever extension it is, and if its outgoing call or incoming call.. any ideas? thanks |
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| 02:13.35 | ChannelZ | depends |
| 02:14.27 | ChannelZ | How are you recording? |
| 02:14.43 | ChannelZ | Monitor and MixMonitor let you specify the base filename, you can put whatever you want in that |
| 02:18.17 | Katty | oh it's too late for asterisk stuffs. |
| 02:18.26 | Katty | ChannelZ: you should be eating cookies and drinking beer. |
| 02:19.45 | ChannelZ | I just finished a Drumstick if that counts |
| 02:19.58 | Katty | that's a start. |
| 02:20.46 | Katty | i'm plotting garlic parmesan wings next week. should be tasty. |
| 02:21.18 | ChannelZ | Mmmmm |
| 02:27.04 | deo | ChannelZ: thanks for the tip.. ill try that.. |
| 02:45.49 | *** join/#asterisk Neptu (~Hej@c213-89-24-62.bredband.comhem.se) |
| 02:47.02 | carrar | garlic wtf |
| 02:47.09 | carrar | err |
| 02:47.13 | carrar | garlic ftw |
| 02:47.17 | carrar | heh |
| 02:47.46 | Katty | both legit statements. |
| 02:47.50 | Katty | depending on the recipe. |
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| 02:53.07 | carrar | the problem with garlic |
| 02:53.14 | carrar | is that it doesn't aggregate RSS feeds |
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| 03:38.05 | deo | guys, is it possible to modify the call recording format to use the real name instead of extension number? |
| 03:43.08 | ChannelZ | huh? |
| 03:43.27 | navaismo | ae you using a GUI to do that? |
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| 08:32.23 | ChannelZ | Wow. Almost 5 hours and not a peep out of anyone. Rare. |
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| 10:14.21 | sedeki | i'm trying to send an AGI command to the server. is this correct? "DIAL SIP/VoIPProvider/###,150" where ### is a telephone number |
| 10:32.15 | Greenlight | Hmm....45gig logfile... guess Asterisk doesn't auto rotate then |
| 10:32.55 | *** join/#asterisk DarkKnightCZ (~lukas@2001:718:1001:110:a017:6b32:24ce:8f02) |
| 10:33.36 | DarkKnightCZ | hi, i've just installed asterisk in debian, enabled http in http.conf, but when i try to access localhost:8088, it throws 403 error... if i enable static content, it throws 404, any ideas? |
| 10:40.13 | wdoekes | DarkKnightCZ: sprintf(path, "%s/static-http/%s", ast_config_AST_DATA_DIR, uri); |
| 10:40.16 | wdoekes | <PROTECTED> |
| 10:40.19 | wdoekes | <PROTECTED> |
| 10:41.01 | DarkKnightCZ | yeeah, static-http is empty :) |
| 10:41.43 | wdoekes | uri is also empty, so that should be fine.. but it it stat'able by the asterisk-user? |
| 10:42.28 | DarkKnightCZ | how do i check that? i've created testing user "admin" in manager.d/admin.conf |
| 10:42.41 | wdoekes | is asterisk running as root? |
| 10:42.49 | DarkKnightCZ | it should |
| 10:43.38 | wdoekes | and you have /var/lib/asterisk/static-http? put dummy.txt there and try to access localhost:8088/dummy.txt |
| 10:44.17 | wdoekes | .. because if it's a directory, you'll get a 404 again ;) |
| 10:44.18 | wdoekes | <PROTECTED> |
| 10:44.19 | wdoekes | <PROTECTED> |
| 10:44.44 | DarkKnightCZ | static-http is directory, but it is in /usr/share/asterisk (debian) |
| 10:46.06 | wdoekes | see datadir in asterisk.conf |
| 10:46.17 | wdoekes | and see redirect = / /static/config/index.html in http.conf |
| 10:46.36 | DarkKnightCZ | hmm, http show status shows only /httpstatus, /phoneprov/, /static (which works), redirect is all right |
| 10:47.57 | DarkKnightCZ | datadir is /usr/share/asterisk, so it should be ok... but normal user doesnt have enough permissions to access /static |
| 10:50.54 | DarkKnightCZ | wdoekes: you can check it remotely, what it does - http://sysq.cz:8088/ |
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| 10:51.35 | DarkKnightCZ | at that server, there is clean install, only thing edited is http.conf set to enabled and binding to 0.0.0.0 |
| 10:52.52 | wdoekes | is there a config/index.html then? |
| 10:53.40 | DarkKnightCZ | nope... in /usr/share/asterisk is only conf folder with bunch of sample files |
| 10:54.00 | wdoekes | ok.. so you expect index.html to be served out of thin air? |
| 10:54.40 | DarkKnightCZ | well, i don't know, i thought there would be some sample web admin |
| 10:56.16 | wdoekes | in my default asterisk there are some sample files in static-http (not index.html, but mantest.html for instance) |
| 10:56.37 | wdoekes | if there are no files, you won't get anything.. that should speak for itself |
| 10:57.00 | DarkKnightCZ | yeeah, so wrong debian package then :) |
| 11:00.22 | *** join/#asterisk Sorcier_FXK (~nssystem@unaffiliated/sorcierfxk) |
| 11:01.14 | wdoekes | DarkKnightCZ: don't overestimate what you can do with the sample files. if you're looking for something to administer your asterisk config with, you're going to be disappointed |
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| 11:14.02 | *** join/#asterisk DarkKnightCZ (~lukas@2001:718:1001:110:6a5d:43ff:fe39:2308) |
| 11:14.52 | DarkKnightCZ | ok, so there is no default asterisk frontend, right? |
| 11:15.39 | file | there isn't |
| 11:16.08 | DarkKnightCZ | so, any recommendations? |
| 11:24.36 | sedeki | i'm using AGI. i've set it up to play some music. i have to make three calls to it before it plays music |
| 11:28.47 | sedeki | asterisk says it is playing music but it isn't |
| 11:31.50 | wdoekes | 12:01 < wdoekes> DarkKnightCZ: don't overestimate what you can do with the sample files. if you're looking for something to administer your asterisk config with, you're going to be disappointed |
| 11:31.55 | wdoekes | ~gui |
| 11:31.55 | infobot | gui is, like, (Graphical User Interface) Any system that uses graphics to represent the functions of a program. All Windows operating systems are GUI. or point-and-drool, double-click, drag-and-drop, such as http://toastytech.com/guis/miscwin1xp.png. Of course Real Programmers use the command line interface. See cli |
| 11:32.05 | wdoekes | ~freepbx |
| 11:32.05 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 11:32.12 | wdoekes | ~asterisknow |
| 11:32.12 | infobot | hmm... asterisknow is based on Asterisk, but is difficult to support in #asterisk for a number of reasons. Please seek support in #asterisknow instead. |
| 11:32.15 | wdoekes | ~elastix |
| 11:32.15 | infobot | elastix is probably a full linux distro that includes FreePBX and other 3rd party add-ons. It is all this extra stuff which makes Elastix VERY difficult to support, and is not supported in #asterisk. Try asking in #Elastix or on their forums at http://www.elastix.org |
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| 11:32.48 | DarkKnightCZ | wdoekes: yes, i know, but i'm making school project, so gui should be enough to do with :) |
| 11:33.05 | wdoekes | that's fine. |
| 11:33.13 | wdoekes | but then you're in the wrong channel :) |
| 11:33.37 | DarkKnightCZ | yes :) so #freepbx... i didn't know it wasnt part of asterisk :) thanks |
| 11:36.32 | *** part/#asterisk DarkKnightCZ (~lukas@2001:718:1001:110:6a5d:43ff:fe39:2308) |
| 11:36.41 | kaldemar | sedeki: use pastebin to show what really happens. enable verbosity and agi debug. |
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| 12:29.56 | Rumbles | Hi guys, do you know how I can check whether jitter buffer is being enforced on a particular call? I believe it is set up correctly, but I'm seeing latency spikes and the EU is reporting call quality issues |
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| 13:16.32 | igcewieling | Rumbles: the jitter buffer is not usually needed since endpoints dejitter. Is there a specific reason you are looking at the JB? |
| 13:16.56 | igcewieling | Rumbles: you might be able to find the info in "core show channel <channelname>" |
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| 13:23.22 | eXcAliBuR | looks around silently |
| 13:23.36 | eXcAliBuR | :} |
| 13:23.44 | eXcAliBuR | lets talk about sound for a second |
| 13:24.13 | eXcAliBuR | i have audacity and I recorded a nice line from google translate, because I like how it sounds. |
| 13:24.19 | eXcAliBuR | tried to export as gsm, didn't work |
| 13:24.26 | eXcAliBuR | wav doesn't work either |
| 13:24.47 | eXcAliBuR | when I say doesnt work, asterisk screams at me |
| 13:25.01 | eXcAliBuR | how are others doing their sound? |
| 13:25.31 | mirela666 | eXcAliBuR: single channell 8000 sample rate 8k hz |
| 13:25.33 | mirela666 | ? |
| 13:26.19 | mirela666 | I do them row/headerless .alaw formt |
| 13:26.53 | eXcAliBuR | i think i need plugins |
| 13:27.41 | mirela666 | which version of Audacity you have |
| 13:28.39 | eXcAliBuR | 2.0.3 |
| 13:30.08 | mirela666 | cool, so you change project rate to 8000 in bottom left corner |
| 13:30.52 | [TK]D-Fender | No, leave the project at it's best and change your EXPORT settings |
| 13:31.08 | mirela666 | record and export file as raw/headerless |
| 13:31.24 | mirela666 | ok, probably better |
| 13:31.55 | [TK]D-Fender | mirela666, and GSM 6.10 wave is a good choice (.wav) |
| 13:32.02 | mirela666 | encoding alaw or gsm (6.10) |
| 13:32.03 | eXcAliBuR | i don't see raw option |
| 13:32.06 | [TK]D-Fender | mirela666, make sure it's NAMED ".wav" |
| 13:32.35 | mirela666 | header > raw |
| 13:32.46 | Rumbles | thanks igcewieling, the latency on the sip device is going from 50ms at good times up to 300ms at bad times, meaning they are reporting choppy calls |
| 13:33.07 | eXcAliBuR | ok found it |
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| 13:33.21 | mirela666 | [TK]D-Fender: is it better than alaw? |
| 13:33.34 | [TK]D-Fender | mirela666, No, but it should work. |
| 13:33.36 | eXcAliBuR | has a extension of .raw now |
| 13:33.39 | eXcAliBuR | is that ok? |
| 13:33.42 | mirela666 | [TK]D-Fender: oki |
| 13:33.43 | [TK]D-Fender | no |
| 13:33.51 | [TK]D-Fender | * processes based on the extension |
| 13:34.36 | mirela666 | eXcAliBuR: if you chose gsm give it wav, if you chose alaw give it .alaw |
| 13:36.00 | mirela666 | alaw files are played directly and wav is copied to .slim and then played by Asterisk |
| 13:36.05 | mirela666 | I think |
| 13:36.24 | eXcAliBuR | it played in super slow motion |
| 13:36.31 | eXcAliBuR | progress |
| 13:36.33 | eXcAliBuR | yay |
| 13:38.20 | mirela666 | eXcAliBuR: ok, try with project rate of 8000 and see if it;s ok |
| 13:40.47 | eXcAliBuR | better |
| 13:40.52 | eXcAliBuR | but still slow |
| 13:47.01 | eXcAliBuR | all better |
| 13:47.02 | eXcAliBuR | :) |
| 13:47.11 | eXcAliBuR | i had to resample it |
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| 13:52.00 | *** mode/#asterisk [+o mjordan] by ChanServ |
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| 13:58.44 | eXcAliBuR | hmmm |
| 13:58.48 | eXcAliBuR | crashed my laptop |
| 13:58.53 | eXcAliBuR | yikes |
| 14:00.43 | eXcAliBuR | new question, i made a folder fr in the sound folder because i have french also. how do i tell asterisk to look there? |
| 14:00.51 | eXcAliBuR | or it's only 1 folder |
| 14:00.53 | eXcAliBuR | n not two? |
| 14:01.02 | eXcAliBuR | i'll just use one |
| 14:03.25 | *** join/#asterisk FireAndIce (~FireAndIc@123.201.161.26) |
| 14:05.53 | eXcAliBuR | hmmm my client said I make them sound handicap :( |
| 14:06.57 | Greenlight | Hmm Asterisk deadlocked again. This time I've got full debug loggin, and DONT_OPTIMISE set in the backtrace |
| 14:07.36 | *** join/#asterisk garymc (~chatzilla@host81-139-87-129.in-addr.btopenworld.com) |
| 14:09.01 | file | grab what you did before, and trim the log down to where it looks like stuff went funky |
| 14:10.13 | Greenlight | So you don't want all 15gigs of the log :) |
| 14:10.21 | file | I *could* |
| 14:10.26 | file | if you have a place to put it, that is |
| 14:11.32 | Greenlight | Yea, sure I can stick it somewhere. I'll see what I can trim down first, but it's likely going to be over the limits of pastebin. |
| 14:12.04 | file | aye |
| 14:12.23 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-altfjxbmfijpsekq) |
| 14:12.23 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 14:12.48 | Greenlight | Mind giveing me the IP you'll be accessign from so I can allow through to where I'll dump the logs etc ? |
| 14:13.07 | file | IPv4 I assume? |
| 14:13.10 | Greenlight | Yea |
| 14:13.13 | file | 156.34.244.17 |
| 14:13.16 | Greenlight | CHeers |
| 14:13.30 | Greenlight | Ok - give me 5 mins - again, thanks :) |
| 14:13.37 | file | yw |
| 14:13.59 | *** join/#asterisk jsjc (~Adium@226.Red-80-33-236.staticIP.rima-tde.net) |
| 14:14.57 | *** join/#asterisk lorsungcu (~anonymous@65.103.31.33) |
| 14:16.45 | *** join/#asterisk apocn (~apo@unaffiliated/apocn) |
| 14:18.22 | apocn | Hello all, I have this simple scenario. What can I use to make it work? (I've tried with a sip proxy without success): http://pastebin.com/PCixisEg |
| 14:20.33 | Greenlight | Backtrace @ : http://paste.ubuntu.com/5634153/ Still working on the logfile. |
| 14:22.16 | Greenlight | Wow audiohook.c: is quite spammy in the logs |
| 14:22.21 | [TK]D-Fender | apocn, Port forward the appropriate ports. Tell * your WAN IP. Set the peers to NOT allow re-invites. DONE |
| 14:22.47 | apocn | ok |
| 14:29.04 | *** join/#asterisk gonewage (~gonewage@smtp.rwrlaw.com) |
| 14:29.40 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.124.47) |
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| 14:36.24 | Greenlight | Most of the log appears to be filled with this: http://paste.ubuntu.com/5634197/ Is that of any relevance, and if not is there a fancy way to trim it out? |
| 14:38.44 | Rumbles | igcewieling, I don't see any info in the "core show channel SIP/......." output about jitter buffer, do you know if that should produce jitter buffer info? |
| 14:39.10 | *** part/#asterisk mirela666 (~Thunderbi@212.200.146.253) |
| 14:40.41 | file | Greenlight, you can trim it out |
| 14:43.26 | *** join/#asterisk gonewage (~gonewage@smtp.rwrlaw.com) |
| 14:45.24 | igcewieling | Rumbles: jitter buffers are used so seldom with Asterisk I don't know for sure. |
| 14:45.48 | igcewieling | Rumbles: do you have reinvites disabled? |
| 14:46.41 | Greenlight | Okay, down to 2gigs for the last 1 1/2 hrs prior to deadlock |
| 14:47.04 | *** join/#asterisk vinhdizzo (~vinh@ip-64-134-229-70.public.wayport.net) |
| 14:49.50 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.124.47) |
| 14:51.16 | igcewieling | Found what appears to be a bug in Adtran media GWs. We use SRV records for load balancing, but we also had two IP addresses set up on the A record for the hostname (sans the _sip._udp). The adtrans were updating the A record DURING a call and using the NEW IP for new SIP packets on existing calls. Hilarity ensues (for negative values of hilarity) |
| 14:57.43 | Rumbles | canreinvite is set to no igcewieling, we are using FreePBX on the system, and it is set in there for Jitter Buffer to be forced on, but we're still seeing voice issues, so I wanted to ensure his calls were using jitter buffer |
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| 15:16.35 | *** join/#asterisk gerhard7 (~gerhard7@77-172-47-159.ip.telfort.nl) |
| 15:16.54 | apocn | [TK]D-Fender, it worked and now I'm registered. But RTP won't go through. I did DNAT to the rtpstart - rtpend ports as well. |
| 15:17.19 | *** join/#asterisk navaismo (~navaismo@189.241.122.125) |
| 15:17.53 | [TK]D-Fender | apocn, Show us your configs masking only passwords |
| 15:17.55 | [TK]D-Fender | ~pb |
| 15:17.56 | infobot | A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 15:17.57 | [TK]D-Fender | ^^^ |
| 15:18.08 | [TK]D-Fender | sip.conf <- |
| 15:18.27 | [TK]D-Fender | [general ] + peers, and the SIP DEBUG for a failed call |
| 15:19.03 | *** join/#asterisk ruben231 (~OpenDial@112.198.90.97) |
| 15:20.41 | ruben231 | hi guys any idea how to completely remove dahdi..? on my asterisk server |
| 15:22.55 | ruben231 | dahdi_test -------------->Opened pseudo dahdi interface, measuring accuracy... 4.693% 4.689% 4.688% 4.689% 4.691% 4.689% 4.689% 4.690% <----------------seems issue coz with working install it dispplay 99.9%99% |
| 15:23.01 | ruben231 | any idea guys..? |
| 15:25.12 | ruben231 | dahdi_test ------>Opened pseudo dahdi interface, measuring accuracy...99.606% 99.613% 99.449% 99.643% 99.433% 99.614% <--------------working value (no audio issue) |
| 15:26.21 | *** join/#asterisk glaz (strke@hiro.glaciuz.com) |
| 15:27.01 | WIMPy | Remove it or jsut don't use it. |
| 15:27.22 | glaz | now that I've upgrade asterisk from 1.4 to 11, I seem to have a hard time googling for config examples. I'm trying to configure CDR, I've read that CDR mysql is now native, any place I could find a write up for this? |
| 15:28.18 | Qwell | glaz: You read incorrectly. |
| 15:28.33 | apocn | [TK]D-Fender, it worked after I put nat=yes in my sip.confÂ… thanks a lot for your help |
| 15:29.02 | glaz | Qwell: any place I should look to setup mysql CDR for asterisk 11? |
| 15:29.35 | ruben231 | WIMPy: how do i completely remove it..? |
| 15:29.42 | *** join/#asterisk Greenlight (~email@cpc1-dund9-0-0-cust142.16-4.cable.virginmedia.com) |
| 15:29.47 | Qwell | glaz: Don't use MySQL CDRs. Use cdr_odbc, or one of the others. |
| 15:29.57 | ruben231 | so i can install lower version of dahdi to amke it work somehow |
| 15:30.00 | glaz | will it write to a mysql table? |
| 15:30.10 | WIMPy | ruben231: How did you install it? |
| 15:30.34 | Greenlight | file: Sorry my pc decided that opening that logfile would hard crash it /sigh Windows sometimes infuriates me. Did I miss anything? |
| 15:30.40 | file | no |
| 15:30.42 | file | nothing yet |
| 15:32.12 | navaismo | glaz, http://lmgtfy.com/?q=cdr+mysql+asterisk check the 5th result |
| 15:32.26 | glaz | I did that already |
| 15:33.15 | *** join/#asterisk gonewage (~gonewage@smtp.rwrlaw.com) |
| 15:33.48 | navaismo | so? |
| 15:34.12 | glaz | Seems like odbc can write to a MSSQL, trying to find if it can write to a MySQL |
| 15:34.35 | file | it can. |
| 15:34.36 | navaismo | yes you can use ODBC or deprecated cdr_mysql |
| 15:34.58 | glaz | Using MySQL for CDR records is supported by using ODBC and the cdr_adaptive_odbc module (depends on res_odbc). |
| 15:35.48 | navaismo | now search with odbc mysql asterisk |
| 15:36.22 | navaismo | one result ---> http://edmundlong.com/edsBlog/odbc-cdr-with-asterisk/ |
| 15:36.34 | glaz | edmunglong.com is not responding to me |
| 15:36.50 | glaz | oh now it is, nvmd. |
| 15:41.19 | *** join/#asterisk sipman (~slane@client-216.114.57.167.tx.skybeam.com) |
| 15:41.55 | *** join/#asterisk blee (~blee@68.204.217.123) |
| 15:48.28 | glaz | now comes packages conflits |
| 15:48.52 | glaz | this write up tells me to install unixodbc, libmyodbc and iodbc |
| 15:49.09 | glaz | iodbc wont work with libmyodbc, it now requires libmyodbc2 |
| 15:49.20 | glaz | and unixodbc wont work with libmyodbc2 |
| 15:49.23 | glaz | suggestions? |
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| 16:00.52 | igcewieling | glaz: Distro? |
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| 16:52.50 | linocisco | hello all |
| 16:53.12 | glaz | igcewieling: sorry I was at lunch, let met very, iirc ubuntu 12.04 LTS |
| 16:53.31 | glaz | igcewieling: yes, Ubuntu 12.04.2 LTS |
| 16:54.04 | igcewieling | ah, sorry, can't help with that distro |
| 16:55.21 | glaz | igcewieling: I think I'm screwed. |
| 16:55.35 | glaz | can I use the obsolete cdr_mysql addon ? |
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| 16:57.56 | linocisco | hi all, I am to configure IVR for bus/Airplan schedule. Management want to put annoucement mp3 files themselves. How can I just create simple folder so that they can put easily? |
| 16:59.21 | navaismo | glaz yes you can |
| 16:59.22 | igcewieling | linocisco: you do not want to do that. Convert your files to a format Asterisk support nativly |
| 16:59.38 | navaismo | linocisco, use samba |
| 17:00.22 | linocisco | igcewieling, whatever file format, it doesnt matter. I would like to know where to create which folder and how to give file path |
| 17:00.31 | linocisco | so that they can access easily |
| 17:01.28 | WIMPy | That's not an Asterisk question. |
| 17:01.43 | igcewieling | Create a folder in /home/sounds and run Playback(/home/sounds/thesoundfilename) |
| 17:02.01 | igcewieling | The rest is a Linux question |
| 17:02.14 | linocisco | igcewieling, thanks bro |
| 17:02.15 | glaz | navaismo: ok, but in make menuconfig the option seems disabled, it shows xxx, I can't select it |
| 17:02.54 | WIMPy | glaz: And at the bottom it tells you why. |
| 17:03.10 | glaz | really? didn't see that, let me check |
| 17:03.26 | linocisco | igcewieling, how can we make sure that inserted new audio file is updated or annouced real time without having to stop functioning of running asterisk? |
| 17:05.08 | WIMPy | It just happens. The downside is that you might get side effects when updating a file while it's playing. |
| 17:05.29 | *** join/#asterisk gonewage (~gonewage@smtp.rwrlaw.com) |
| 17:06.03 | *** join/#asterisk threesome (~threesome@ip-94-113-12-74.net.upcbroadband.cz) |
| 17:06.33 | glaz | it says depending on mysqlclient |
| 17:06.35 | glaz | I have it installed |
| 17:06.56 | linocisco | WIMPy, how can I avoid? |
| 17:07.16 | *** part/#asterisk gonewage (~gonewage@smtp.rwrlaw.com) |
| 17:07.40 | igcewieling | linocisco: it is avoid automatically internally by Asterisk |
| 17:08.07 | WIMPy | How could Asterisk do that? |
| 17:08.13 | linocisco | igcewieling, so no need to shutdown asterisk? goood |
| 17:08.33 | igcewieling | WIMPy: In Linux when you have a file open and it is deleted, the application won't see it gone until the file is closed. |
| 17:08.37 | igcewieling | linocisco: correct. |
| 17:08.59 | igcewieling | if you are overly paranoid then upload to a temp folder on the same filesystem and the mv the files to the correct location |
| 17:09.16 | WIMPy | Yes, but if you update an existing file that's a different matter. |
| 17:09.29 | igcewieling | ah, I see your point. |
| 17:10.08 | WIMPy | That's the safe way, yes. |
| 17:11.49 | glaz | meh, I just realised I don't have mysql.h, I need to install libmysqlclient-dev |
| 17:12.52 | igcewieling | linocisco: NONE of this is Asterisk specific. |
| 17:12.56 | igcewieling | ~doc |
| 17:12.56 | infobot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk |
| 17:13.30 | linocisco | ~doc |
| 17:13.30 | infobot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk |
| 17:20.48 | glaz | igcewieling, navaismo: it works, thank you. |
| 17:27.14 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.126.69) |
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| 17:52.01 | *** join/#asterisk eXcAliBuR (~eXcAliBuR@206.162.174.6) |
| 17:52.04 | eXcAliBuR | i'm back |
| 17:52.08 | eXcAliBuR | but i don't need support |
| 17:52.11 | eXcAliBuR | just came to chill |
| 17:52.12 | eXcAliBuR | :} |
| 17:52.15 | eXcAliBuR | i'm nice like that |
| 17:57.03 | ChannelZ | Let's get drunk and screw! |
| 17:57.29 | ChannelZ | I mean, good afternoon |
| 18:03.00 | igcewieling | ChannelZ: http://xkcd.com/330/ |
| 18:14.06 | eXcAliBuR | ;D |
| 18:14.28 | eXcAliBuR | I know this is so like 1990's but ChannelZ: asl ? |
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| 18:32.17 | Greenlight | Anyone else in UK got internet connectivity issues as of a few moments ago ? |
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| 18:44.11 | glaz | after using Dial() with the Fg option (continue dialplan execution after caller/callee hangs up), I lose the variable I've set, is there a way to keep it? |
| 18:47.16 | [TK]D-Fender | g only continues on CALLEE hanging up |
| 18:47.35 | glaz | and F on caller, right? |
| 18:48.06 | glaz | so if I put both it should continue whomever hangsup right? |
| 18:48.09 | [TK]D-Fender | Where do you see this documented? |
| 18:48.18 | glaz | here, https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Application_Dial |
| 18:49.03 | [TK]D-Fender | glaz, that other channel has INHERITED variables. |
| 18:49.09 | [TK]D-Fender | not the original names |
| 18:49.16 | [TK]D-Fender | Read up on your inheritance basics |
| 18:49.24 | glaz | All I want to do is UnPauseQueueMember() on the caller interface |
| 18:49.52 | glaz | but I can't do UnPauseQueueMember(,SIP/${CDR(src)}) |
| 18:50.02 | [TK]D-Fender | perhaps you could back things up a bit and show us the bigger picture... |
| 18:50.04 | glaz | since the CDR(src) becomes the callee |
| 18:50.20 | glaz | ok, let me pastebin that |
| 18:51.42 | glaz | http://pastebin.ca/2337555 |
| 18:52.54 | glaz | if I use ${UNPAUSEID}), the variable is unset, if I use ${CDR(src)}, like I do to pause the member, the variable becomes the callee number |
| 18:55.58 | [TK]D-Fender | glaz, foget BOTH options and just use "h" <- |
| 18:56.03 | [TK]D-Fender | forget* |
| 18:56.08 | [TK]D-Fender | You are gdoing it the hard way... |
| 18:57.30 | glaz | h - Allow the called party to hang up by sending the DTMF sequence defined for disconnect in features.conf. |
| 18:57.46 | glaz | oh you mean exten h ? |
| 18:58.23 | glaz | Let me try this |
| 18:58.31 | glaz | h per context, right ? |
| 19:01.16 | glaz | <PROTECTED> |
| 19:01.22 | glaz | it works, thanks [TK]D-Fender |
| 19:01.56 | [TK]D-Fender | glaz, You're welcome |
| 19:03.40 | glaz | I was really doing it the wrong way. |
| 19:03.48 | glaz | exten => h,1,UnpauseQueueMember(,SIP/${CDR(src)}) |
| 19:03.59 | glaz | oops, oh well, this worked |
| 19:04.45 | glaz | or should I use ${CALLERID(dnid)} ? |
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| 19:31.58 | poseidon | When I do a wget for action login, with '--save-cookies cookies.txt' I get authentication accepted. However, when I go to do action=ping, using that same cookie, I get 'Message: Authentication Required' |
| 19:34.16 | *** join/#asterisk Sorcier_FXK (~nssystem@unaffiliated/sorcierfxk) |
| 19:34.17 | poseidon | I am following directly from http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/AMI-quickstart.html |
| 19:34.41 | *** join/#asterisk TimeRider (~steve@timerider.plus.com) |
| 19:41.56 | glaz | [TK]D-Fender: it's affecting all the dialplan though the h exten, I have some context that I don't want exten h to be executed, any idea? |
| 19:42.05 | *** join/#asterisk AkkerKid (~AkkerKid@23.31.20.201) |
| 19:42.26 | [TK]D-Fender | glaz, Change your context |
| 19:42.40 | [TK]D-Fender | glaz, you should already know your sense of scope... |
| 19:42.42 | glaz | it affects other context too |
| 19:43.23 | glaz | my exten h is in my [outbound-local] context, and calls placed in [office] context are also affected |
| 19:43.49 | [TK]D-Fender | pay attention to where you put it... |
| 19:44.05 | poseidon | 9Any idea why it would be authenticating at first, then failing when I ping? |
| 19:44.11 | glaz | you mean context including other context? |
| 19:45.42 | glaz | [TK]D-Fender: I guess you're trying to point me to my error but I can't see it |
| 19:46.49 | *** join/#asterisk tapout (~tapout@unaffiliated/tapout) |
| 19:47.47 | [TK]D-Fender | glaz, You are using INCLUDES and have forgotten how the seaerch path goes |
| 19:48.11 | [TK]D-Fender | glaz, When you hit your dialplan match... jump to a context that ISN'T INCLUDE-d to do your dirty-work |
| 19:48.43 | glaz | I'm trying to understand what you're saying, as english isn't my first languag |
| 19:49.43 | glaz | so just jump, hangup() and h from this context will be executed? |
| 19:51.10 | igcewieling | generally extensions in your current context are matched before the same extension in include =>'d contexts |
| 19:51.24 | igcewieling | think about that for a few mins and you may see the light |
| 19:52.09 | glaz | That I know |
| 19:52.30 | [TK]D-Fender | glaz, http://pastebin.ca/2337587 |
| 19:52.36 | glaz | but the sip devices sit in a context including a lot of contextes |
| 19:53.10 | glaz | aaaah, I see the light !! |
| 19:53.31 | [TK]D-Fender | And fix the var reference up top |
| 19:53.35 | glaz | yes. |
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| 20:28.17 | poseidon | Anyone here use asterisk ami via http? |
| 20:28.42 | *** join/#asterisk killown (~killown@pdpc/supporter/student/killown) |
| 20:29.35 | killown | could anyone recommend a good cost/benefit fxo/fxs card? |
| 20:29.49 | *** join/#asterisk KNERD (~KNERD@24.175.249.177) |
| 20:31.26 | igcewieling | the main benefit to using FXS and FXO cards is the idiot network admin (who may feel threatended by an IP PBX) doesn't have to be involved. |
| 20:31.36 | igcewieling | at least in my experience. 8-| |
| 20:32.52 | igcewieling | poseidon: is there a reason you want to do AMI over HTTP, AJAX GUI or something like that? |
| 20:33.30 | poseidon | igcewieling: yes, for that reason. |
| 20:34.49 | igcewieling | poseidon: in my experience it is better/easier/more well known to do an ajax call to a script which uses a normal AMI connection. Doing it directly with AJAX/JSON just makes stuff really complicated. |
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| 20:39.06 | navaismo | ~book |
| 20:39.06 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 20:39.44 | *** join/#asterisk iondream (~iondream@host-64-47-124-106.masergy.com) |
| 20:40.00 | iondream | afternoon gentlemen |
| 20:40.33 | leedm777 | poseidon: there was a presentation about AMI over HTTP at AstriCon last year. http://www.tmcnet.com/tmc/videos/default.aspx?vid=7514 |
| 20:42.30 | killown | AEX410 is fine? |
| 20:42.51 | igcewieling | killown: I am personally prefer Sangoma |
| 20:43.10 | igcewieling | killown: price out SPA phones, they might not be much more expensive. |
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| 21:39.34 | *** join/#asterisk zerohalo (~zerohalo@74.61.196.236) |
| 21:43.23 | *** join/#asterisk linocisco (~linocisco@203.81.67.114) |
| 21:43.52 | linocisco | how can I shutdown asterisk from asterisk CLI mode and quit to linux CLI? |
| 21:44.14 | *** join/#asterisk lorsungcu (~anonymous@65.103.31.33) |
| 21:44.35 | [TK]D-Fender | clarify "asterisk CLI mode" |
| 21:45.11 | *** join/#asterisk Rhomber (~david@60-240-245-17.static.tpgi.com.au) |
| 21:45.40 | linocisco | [TK]D-Fender, I am in *CLI> mode and want to gracefully shutdown asterisk and return to linux CLI to shutdown linux properly |
| 21:46.05 | [TK]D-Fender | How did you GET there? |
| 21:46.37 | linocisco | [TK]D-Fender, It is my first installation as per http://blogs.digium.com/2012/11/14/how-to-install-asterisk-11-on-ubuntu-12-4-lts/ |
| 21:47.17 | [TK]D-Fender | linHow did YOU just get to * CLI? |
| 21:47.27 | [TK]D-Fender | What command did you do precisely |
| 21:48.29 | *** join/#asterisk dxrt (~dxrt@unaffiliated/dxrt) |
| 21:48.30 | linocisco | [TK]D-Fender, I think "/etc/init.d/asterisk start" |
| 21:48.43 | [TK]D-Fender | that didn't get you to CLI |
| 21:48.46 | [TK]D-Fender | that starts * as a daemon |
| 21:49.06 | linocisco | [TK]D-Fender, and then asterisk -rvvv |
| 21:49.10 | [TK]D-Fender | At which point the assumption is that you did "asterisk -r" to CONNECT to it |
| 21:49.35 | *** join/#asterisk bpriddy (~bpriddy@ipv4.host.stabbyspazzout.net) |
| 21:49.39 | [TK]D-Fender | so do "exit" to get back to CLI and "/etc/init.d/asterisk stop" |
| 21:49.49 | linocisco | [TK]D-Fender, thanks |
| 21:52.23 | *** join/#asterisk bandroidx (~bandroidx@205.185.117.117) |
| 21:56.09 | *** join/#asterisk ruben231 (~OpenDial@112.198.90.97) |
| 21:57.00 | *** part/#asterisk ruben231 (~OpenDial@112.198.90.97) |
| 21:58.20 | *** join/#asterisk ariel_ (uid3533@pdpc/supporter/active/abatista) |
| 21:58.27 | ariel_ | hello folks |
| 22:00.30 | ariel_ | I have a strange issue which maybe a simple way to fix it. I normally don't set a bindport = or port= setting. I have a sip provider that is sending me traffic on ports 5061 to 5064, but I can't seem get this to work. Asterisk 1.8.8 I have tried port=5060-5964 but it fails if I put port=5062 when it's send from the on 5062 it works.. any ideas on how to |
| 22:00.31 | ariel_ | fix this? |
| 22:02.05 | *** join/#asterisk sipman (~slane@71-14-128-129.dhcp.ftwo.tx.charter.com) |
| 22:02.08 | ariel_ | s/port=5060-5964/port=5060-5064 |
| 22:02.52 | [TK]D-Fender | no |
| 22:03.21 | [TK]D-Fender | port is a PEER/USER option and is for outbound comms. |
| 22:03.30 | [TK]D-Fender | bindpeer is the * listening port |
| 22:03.42 | [TK]D-Fender | and is one port only, not a range. |
| 22:03.58 | *** join/#asterisk HmdP_Mobile (~HmdP_Mobi@D9799130.cm-3-2c.dynamic.ziggo.nl) |
| 22:04.53 | Tim_Toady | u can redirect the traffic on specific ports using iptables, but it will be a bit ulgy |
| 22:05.28 | ariel_ | [TK]D-Fender: ok how can I allow for a range ports to listen too? |
| 22:07.25 | *** join/#asterisk lvlolvlo (~lvlolvlo@unaffiliated/lvlolvlo) |
| 22:07.33 | [TK]D-Fender | ariel_: nothing in Asterisk |
| 22:07.40 | [TK]D-Fender | ariel_: Other proxy software |
| 22:07.49 | [TK]D-Fender | ariel_: Prepare to gang-pile layers |
| 22:07.59 | [TK]D-Fender | WINTER IS COMING |
| 22:08.43 | drmessano | This is the Winter of your discount tent |
| 22:09.14 | [TK]D-Fender | CHEAP TENTS! |
| 22:09.26 | [TK]D-Fender | and homonyms! |
| 22:14.05 | *** join/#asterisk ruben231 (~OpenDial@112.198.90.97) |
| 22:16.25 | ruben231 | hi guys..can i ask how do i reduces the number of channels on my asterisk currently its running 13 and wanted to reduce to 2 only |
| 22:16.45 | ruben231 | im using voip |
| 22:17.13 | navaismo | vague! |
| 22:17.37 | [TK]D-Fender | Be more vague! Come back and don't remind us you're using VoIP! |
| 22:18.30 | [TK]D-Fender | Want to reduce channels? HANG UP. :) |
| 22:18.53 | igcewieling | ruben231: search for "call-limit" on voip-info.org |
| 22:19.39 | igcewieling | navaismo: vague or ESL (or both)? |
| 22:20.29 | navaismo | call-limit isnt deprecated? |
| 22:21.08 | *** join/#asterisk lorsungcu (~anonymous@65.103.31.33) |
| 22:28.48 | ruben231 | igcewieling:thanks |
| 22:37.18 | carrar | nom nom melonpan |
| 22:40.01 | *** join/#asterisk Dovid (~Dovid@ool-43523983.dyn.optonline.net) |
| 22:40.39 | igcewieling | navaismo: teaching him GROUP() and GROUP_COUNT() is more than I want to commit to at the moment. |
| 22:43.42 | navaismo | he |
| 22:45.04 | navaismo | dont you love the deprecated in asterisk still working like my cdr_mysql |
| 22:45.25 | newtonr | call-limit is deprecated |
| 22:46.41 | newtonr | sip.conf says to use group counters |
| 22:47.49 | newtonr | somehow missed igcewieling's comment |
| 22:50.55 | ruben231 | <PROTECTED> |
| 22:51.12 | newtonr | ruben231: what igcewieling said |
| 23:12.40 | *** join/#asterisk CunningPike (~CunningPi@d28-23-24-84.dim.wideopenwest.com) |
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| 23:51.57 | *** part/#asterisk ruben231 (~OpenDial@112.198.90.97) |
