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| 00:36.29 | Ice_Strike | Which most popular Linux distro that Asterisk is installed on? |
| 00:44.09 | igcewieling | Ice_Strike: generally "The one you are most familiar with.". I prefer CentOS. |
| 00:44.39 | Ice_Strike | igcewieling Do you CentOS as desktop as well? |
| 00:45.18 | igcewieling | Ice_Strike: I currently use fedora, but only because of inertia. |
| 00:48.56 | Ice_Strike | Cool |
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| 01:19.10 | irule | please remind me, an auto.-installer for debian or ubuntu? |
| 01:19.43 | Ice_Strike | yum install spandsp-devel freetds-devel iksemel-devel libsqlite3x-devel radiusclient-ng-devel portaudio-devel libresample-devel gmime22-devel sqlite2-devel jack-audio-connection-kit-devel |
| 01:19.53 | Ice_Strike | Why it wont install? see http://pastebin.com/pctvzMRy |
| 01:20.14 | Ice_Strike | This has to be installed before I install Asterisk |
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| 01:35.54 | WIMPy | Ice_Strike: Yu need all that stuff? |
| 01:36.02 | WIMPy | +o |
| 01:36.30 | Ice_Strike | WIMPy ./contrib/scripts/install_prereq test |
| 01:36.34 | Ice_Strike | Tell me I do. |
| 01:37.14 | WIMPy | Depends on what modules you want to build. But a few of them don't ring a bell for anything. |
| 01:37.47 | Ice_Strike | I just wont to install Asterisk as normal |
| 01:37.52 | Ice_Strike | want* |
| 01:38.10 | Ice_Strike | configure: WARNING: *** Asterisk now uses SQLite3 for the internal Asterisk database. |
| 01:38.10 | Ice_Strike | configure: WARNING: *** Please install the SQLite3 development package. |
| 01:38.16 | WIMPy | Define "normal". |
| 01:38.22 | Ice_Strike | default install |
| 01:38.28 | WIMPy | You definitely need sqlite3. |
| 01:39.06 | WIMPy | That script doesn't work here. |
| 01:39.42 | Ice_Strike | Got it installed |
| 01:39.49 | Ice_Strike | install_prereq suck really |
| 01:41.39 | Ice_Strike | What did asterisk use before without internal Asterisk database ? |
| 01:41.50 | WIMPy | The only one from your list I've got installed is sqlite3. |
| 01:42.02 | WIMPy | berkleydb |
| 01:42.29 | Ice_Strike | WIMPy Im installing from CentOS 6.3 64bit |
| 01:42.52 | Ice_Strike | What distro do you use? |
| 01:43.16 | WIMPy | Pretty much Slackware. |
| 01:44.04 | WIMPy | You can just run menuselect and if something you want is unavailable, it will tell you what you need to instal to get it. |
| 01:45.56 | Ice_Strike | I've got it installed :) |
| 01:46.09 | Ice_Strike | I was using old version in the past |
| 01:46.15 | Ice_Strike | 4 and 6 |
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| 06:20.35 | jploh | anyone here can help me troubleshoot webrtc with aasterisk? |
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| 07:10.31 | nivenh | hiya, anyone know a good MoIP service provider or VoIP service that supports G.711? I need to dialout with a dial-up analogue modem |
| 07:10.34 | nivenh | ? |
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| 07:15.52 | ChannelZ | Moip? |
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| 08:01.06 | zamba | i want to send an email for all missed calls.. is this possible? |
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| 08:06.46 | schmidts | good morning |
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| 08:07.27 | din3sh | gd mrning all |
| 08:07.49 | din3sh | when running patlooptest , i get this "Device or resource busy" |
| 08:08.19 | zamba | i'm thinking i maybe should start using macros soon |
| 08:08.53 | kaldemar | zamba: sure, do it in your dialplan after Dial. DIALSTATUS contents will help you with that. |
| 08:09.27 | din3sh | kaldemar: how do we run patlooptest on digium card? |
| 08:09.50 | zamba | my dialplan is SIMPLE.. so i think i might need some help setting this up |
| 08:12.12 | kaldemar | zamba: use Gosub instead of Macro. |
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| 08:12.27 | kaldemar | however, neither are needed for what you want. |
| 08:12.38 | zamba | can you show me an example? |
| 08:14.23 | kaldemar | ExecIf($["${DIALSTATUS}" = "NOANSWER"]?System(/path/to/script/that/sends/email arg1 arg2...)) |
| 08:14.53 | kaldemar | but you may need something else to get an e-mail address etc. |
| 08:15.51 | kaldemar | din3sh: do you have asterisk running with chan_dahdi loaded? |
| 08:17.59 | kaldemar | din3sh: btw, if you bothered to stuff a simple "patlooptest" search in google, the first result is digium's knowledgebase article for running patlooptest. |
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| 08:18.29 | din3sh | have followed the guide |
| 08:18.39 | din3sh | actually |
| 08:19.17 | din3sh | yes asterisk is running |
| 08:19.33 | zamba | what about writing a gosub with varying number of extensions? |
| 08:19.36 | din3sh | ahh |
| 08:19.43 | din3sh | got it, asterisk should be stopped |
| 08:19.46 | din3sh | thnks |
| 08:20.06 | zamba | which distro has the best support for asterisk? |
| 08:20.09 | kaldemar | zamba: your question makes no sense. |
| 08:20.25 | zamba | varying number of peers, i mean |
| 08:20.36 | zamba | for some of my extensions we want to dial three different peers |
| 08:20.40 | zamba | but for others only one |
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| 08:23.10 | din3sh | ues queues |
| 08:23.17 | din3sh | *use |
| 08:26.23 | zamba | centos a good distro for running asterisk? |
| 08:26.48 | zamba | i've been running on ubuntu previously |
| 08:26.54 | jzaw | zamba, what ever distro youre comfortable with admining |
| 08:27.14 | jzaw | then go with ubunt |
| 08:27.47 | zamba | well, i'm also interested in having a relatively updated asterisk version |
| 08:27.56 | jzaw | then compile from svn |
| 08:30.08 | jzaw | zamba, is this for home or business does it need to be stable? |
| 08:30.45 | zamba | it needs to be stable, yeah |
| 08:30.49 | zamba | it's for business |
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| 08:31.06 | jzaw | maybe look at vanilla debian and apt-get |
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| 08:32.05 | zamba | but isn't centos a distro more aimed at stability and "enterprise"? |
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| 08:33.15 | jzaw | id say debian is conservative and stable ... cant go wrong |
| 08:33.41 | jzaw | if you fancy learning a new distro and playing by all means try centos |
| 08:33.50 | jzaw | you may like it ... if not ... you have your answer |
| 08:35.48 | jzaw | zamba, there is no right answer ... and there are a 100+1 ways to skin a cat |
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| 08:43.12 | din3sh | i have centos running around 20 prod servers |
| 08:44.01 | jploh | i have playback in an extension but when called from sip via websockets, i don't hear anything |
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| 09:42.25 | din3sh | if i swap a digital card in my server, do i have to re-install the os again? |
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| 09:43.11 | kaldemar | din3sh: digital card? |
| 09:43.46 | din3sh | digium E1 card |
| 09:44.25 | kaldemar | that's like asking if you need to buy a new computer if you change your keyboard. |
| 09:44.33 | din3sh | lol |
| 09:45.00 | kaldemar | no, you don't need to re-install your OS. :) |
| 09:45.22 | WIMPy | Sometimes you need a new computer for a new keyboard. |
| 09:45.36 | din3sh | stupid question maybe |
| 09:46.03 | din3sh | I should remove the dahdi driver 1st? |
| 09:46.16 | din3sh | then remove the car physically? |
| 09:46.24 | kaldemar | WIMPy: adapters, adapters. :) |
| 09:46.36 | WIMPy | If available. |
| 09:47.08 | kaldemar | din3sh: how about powering off the machine? |
| 09:47.15 | WIMPy | din3sh: Just replace the card, reboot, and see how it magically works as before. |
| 09:47.22 | WIMPy | Or hopefully not exactely as before. |
| 09:47.23 | din3sh | WIMPy: will try to remove the 4th generation with a 5th generation one |
| 09:47.53 | kaldemar | it should just work. |
| 09:48.19 | din3sh | 4th generation card* |
| 09:48.44 | WIMPy | But it's probably a good idea to shutdown and power off, indeed. |
| 09:49.07 | din3sh | not sure if the hdlc errors are caused by the card though |
| 09:50.01 | WIMPy | Put the other card in to another PC. Twice the fun and more chance for an answer. |
| 09:50.08 | din3sh | another stupid question: does stacking a telehony server on another server without proper use of rack rails impact voice? |
| 09:57.50 | jploh | it depends how many you stack, consult your chassis manual on maximum weight |
| 09:58.09 | jploh | also, not recommended |
| 09:59.21 | din3sh | can that impact voice quality? |
| 10:00.48 | kaldemar | depending on the technology, and electrical crappiness of your devices, it may. |
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| 11:09.13 | jkroon | hi guys, what to do with rather long commands that I need to pass to the asterisk cli? (* 11.2.1) |
| 11:09.24 | jkroon | I recall having this issue with 1.8.X too but thought it would be fixed with 11.2.1 |
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| 11:35.27 | v0lZy | hey |
| 11:35.51 | v0lZy | kaldemar, WIMPy , anyone else, can somone help me with g729 |
| 11:36.10 | v0lZy | Im trying to help someone configure a sort of bridge |
| 11:36.33 | v0lZy | I want asterisk to dial to another pbx and use the g729 codec |
| 11:36.52 | v0lZy | now by default, i had allow=ulaw, allow=alaw, allow=gsm in that order |
| 11:36.57 | v0lZy | then added g729 to the top |
| 11:37.12 | bulkorok | you need an extra licence for g729 |
| 11:37.14 | v0lZy | when i make a call and to sip show channels i notice that codec is alway 'nothing' |
| 11:37.21 | v0lZy | i searched around a big |
| 11:37.23 | v0lZy | bit* |
| 11:37.34 | v0lZy | and saw that there is a bcg729 package |
| 11:38.10 | v0lZy | which is some sort of linphone developed encoded/decoder with lots of stuff ranging from video to images to sound in the package content |
| 11:38.23 | v0lZy | (its a hefty 140mb extracted) |
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| 11:38.32 | v0lZy | is it possible to use this with asterisk? |
| 11:38.42 | v0lZy | and this is pass through config |
| 11:38.42 | bulkorok | could be... but without g729 asterisk module you won't have g729 suipport in asterisk |
| 11:39.04 | v0lZy | bulkork and the module isnt part of the default install? |
| 11:39.10 | bulkorok | nope |
| 11:39.11 | v0lZy | how do i check.. list modules? |
| 11:39.18 | bulkorok | core show translation |
| 11:39.26 | kaldemar | afaik g729 passthrough works without the codec. |
| 11:39.31 | bulkorok | shows you all codecs your asterisk can talk |
| 11:40.06 | Ice_Strike | I need to a new dedicated server for a callcentre company.. At the moment just 20 seats and it may expand in the future |
| 11:40.07 | Ice_Strike | See http://pastebin.com/vCJe8Cy4 |
| 11:40.13 | Ice_Strike | Are the spec ok for the asterisk? |
| 11:40.59 | bulkorok | why not 8gb ram? |
| 11:41.03 | v0lZy | kaldemar: I receive a call and then match the number and do dial(Sip/... @server..) |
| 11:41.18 | Ice_Strike | Yea could do 8GB |
| 11:41.25 | v0lZy | kaldemar: any tips how to configure this to work... |
| 11:41.47 | Ice_Strike | 2 x 120GB SSD Drive for operating system, asterisk, and MySQL |
| 11:42.03 | Ice_Strike | 2 x 1TB drive for backup to be safe. |
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| 11:42.43 | v0lZy | i have format_g729.so in my /usr/lib/asterisk/modules |
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| 11:43.56 | v0lZy | i have [modules] autoload=yes in modules.conf too.. |
| 11:44.10 | kaldemar | does the call not work? |
| 11:44.33 | v0lZy | nope |
| 11:44.44 | v0lZy | at no point do i see any codecs set |
| 11:44.53 | kaldemar | neither do i. |
| 11:44.55 | v0lZy | but if i remove allow=g729 |
| 11:45.10 | v0lZy | the call gets to the other side i think |
| 11:45.15 | kaldemar | i don't even see a call. actually, i see nothing. |
| 11:45.16 | v0lZy | one moment, let me confirm the exact situation |
| 11:45.27 | v0lZy | kaldemar: im doing sip show channels |
| 11:45.44 | kaldemar | and little good will it do for you. |
| 11:45.47 | *** join/#asterisk vlad_starkov (~vlad_star@79.104.7.235) |
| 11:45.48 | v0lZy | it shows 2 channels but always, the codec is <nothing> |
| 11:46.23 | kaldemar | what issue are you experiencing in the call and where is the sip debug for it? |
| 11:47.04 | v0lZy | i mean .. .format <nothing> |
| 11:47.30 | v0lZy | no audio in the call (i think thats a firewall thing right now, i onyl have the outgoing asterisk here...) |
| 11:47.37 | v0lZy | we connect |
| 11:47.37 | v0lZy | but no audio |
| 11:47.52 | kaldemar | show whole outputs instead of single words. |
| 11:48.11 | v0lZy | one moment |
| 11:48.17 | kaldemar | and fix your firewall thing before deciding it to be a codec issue. |
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| 11:49.02 | jacekowski | hi peopl |
| 11:50.55 | v0lZy | kaldemar: http://bpaste.net/show/eYPytVZWC0csfQwwDeDv/ |
| 11:52.03 | v0lZy | if i put allow=g729 above allow=alaw and allow=ulaw, the phone call doesnt connect the phones. If i comment allow=g729 or put it at the bottom, the phones ring and upon answering there is no audio... which is just a tiny little bit better then with allow=g729 |
| 11:52.08 | kaldemar | v0lZy: at this point you should know what sip debug is. |
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| 11:52.54 | v0lZy | yeah i know but before i go into that (Since its a lot of output) i wanted to show what im doing |
| 11:53.32 | v0lZy | kaldemar: do i have to do anything else to use g729 except allow=g729 ... i installed the bcg729 package |
| 11:53.42 | jacekowski | does anybody know if multicast paging works with digium phones? |
| 11:53.46 | v0lZy | and i have a format_g729.so file in /asterisk/modules... |
| 11:54.50 | v0lZy | kaldemar: put short, im lost as to how to enable this .... and the 'pass-through' thing... we're actually receiving a call and then doing a direct sip dial to another pbx. |
| 11:56.10 | v0lZy | like so:exten => ourpublicnum,1,Dial(SIP/somextension@sip.4gcall.us,,Tt) |
| 12:06.59 | v0lZy | kaldemar: ok, I got ulaw and alaw working sound both ways, everything fine |
| 12:07.11 | v0lZy | this is me not knowing how to configure g729 issue :D |
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| 12:18.31 | v0lZy | kaldemar: Ok, i read up on stuff |
| 12:18.36 | v0lZy | I understand the pass through thing now |
| 12:18.49 | v0lZy | if asterisk doesnt recode from one codec to another, thats passthrough |
| 12:19.02 | v0lZy | ill confirm with the provider if they can set to g729 |
| 12:19.20 | v0lZy | then i suppose i dont need any more changes in my asterisk configuration |
| 12:19.31 | v0lZy | if the other side im calling is on g729 as well |
| 12:31.03 | v0lZy | figured it out |
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| 13:37.13 | v0lZy | kaldemar: If I am building IVR with Background() and WaitExten |
| 13:37.32 | v0lZy | kaldemar: Is this still considered pass through? |
| 13:37.58 | v0lZy | kaldemar: I.e.: user calls XXX, XXX answers with IVR, then dials YYY based on user's input |
| 13:38.04 | v0lZy | XXX and YYY are g729 |
| 13:38.29 | v0lZy | can asterisk play sound file to the calling user... and listen for user's choice |
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| 13:55.15 | v0lZy | anyone around? |
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| 13:56.24 | [TK]D-Fender | no |
| 13:56.54 | v0lZy | :S |
| 13:57.01 | v0lZy | Hi [TK]D-Fender |
| 13:57.23 | v0lZy | I'm trying to do pass-through with g720 .. and i successfully set it up |
| 13:57.28 | v0lZy | now i want to put IVR into it |
| 13:57.33 | kaldemar | v0lZy: no you're not. |
| 13:57.41 | v0lZy | but ... is that still pass-thru? |
| 13:57.53 | v0lZy | And... if i playback something, does it have to be g729 |
| 13:58.07 | v0lZy | kaldemar: I have the IVR working, but without prompts being played |
| 13:58.50 | v0lZy | g729* |
| 14:00.22 | kaldemar | sounds more like you don't have it working. if you have non-g729 prompts, would you reckon something is just passed through? |
| 14:00.42 | v0lZy | i suppose its not |
| 14:00.53 | v0lZy | it would have to recode |
| 14:00.55 | v0lZy | orz |
| 14:01.55 | [TK]D-Fender | v0lZy, It's pass-through so long as * does not have to transcode anything at all. No ringing, prompts, etc |
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| 14:02.37 | v0lZy | ok |
| 14:02.38 | v0lZy | so |
| 14:02.49 | v0lZy | i need to put my wav files into g729 |
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| 14:09.05 | apurvtwr | I have written a context in asterisk for making outbound calls. I wish to load test my asterisk server with multiple simultaneous calls using this context. Is there a tool for that? Quick google search lead me to SIPp but it seems to make outbound calls, what I need is a tool that acts a sync for calls |
| 14:09.30 | apurvtwr | Any suggestions? |
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| 14:11.03 | [TK]D-Fender | apurvtwr, "acts as a sync"? SIPP is for load testing just as you asked. Running simultaneous calls... |
| 14:11.28 | [TK]D-Fender | apurvtwr, "but it seems to make outbound calls" , "I wish to load test my asterisk server with multiple simultaneous calls" |
| 14:11.47 | [TK]D-Fender | apurvtwr, It does exactly what you just asked and you said as much in the same line. |
| 14:12.09 | *** join/#asterisk elico (~Thunderbi@bzq-79-176-168-23.red.bezeqint.net) |
| 14:17.28 | apurvtwr | [TK]D-Fender SIPp can make calls to a context that I design. So it is acting as the source of the call. In my case, I have a scheduler which will make multiple simultaneous calls, these calls will use the context I want to test. All I need is a tool that would receive these calls. |
| 14:18.59 | apurvtwr | can SIPp be used in this fashion? I might have missed that in it's documentation. |
| 14:19.15 | [TK]D-Fender | apurvtwr, Your description is still lacking. What is this talk about receiving calls? |
| 14:19.42 | [TK]D-Fender | apurvtwr, Next, you don't call "a context". You call your SERVER. What you dialplan DOES is another matter entirely. |
| 14:20.14 | apurvtwr | hm. ok let me explain more, sorry for that. |
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| 14:23.27 | apurvtwr | I am designing a call flow that is like a voice reminder. There is a scheduler which originates call at a certain time and plays a predefined audio to the called party when they pick the call up. |
| 14:24.30 | apurvtwr | So now I have a dial plan and a scheduler. Which I have tested to work for small number of simultaneous calls. |
| 14:24.58 | Greenlight | Just dial into 1 or more other asterisk boxes, which just answer the call and Wait(). |
| 14:27.39 | apurvtwr | thanks Greenlight |
| 14:28.48 | Greenlight | The reason I say perhaps more than 1 other box, is depending on specs and load, you don't want the "test" boxes to be the source of any bottlencks |
| 14:28.51 | Greenlight | *bottlenecks |
| 14:30.21 | apurvtwr | I see. that's great. I can try that. |
| 14:31.43 | Greenlight | Out of interest, how many simultaneous calls you looking to achieve? |
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| 14:36.22 | [TK]D-Fender | apurvtwr, What aspect are you even testing? |
| 14:45.27 | Katty | HI LADS. |
| 14:49.13 | Greenlight | You've not broke it again have you? |
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| 14:58.22 | Katty | broke what? :> |
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| 15:25.39 | Katty | grooves. |
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| 15:50.40 | Ice_Strike | There is no problem to setup Hardware softphone to connect to Asterisk server from Dedicated Server providers? |
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| 15:52.22 | WIMPy | thinks the words in that sentence need some other order. |
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| 15:54.14 | Nugget | heh |
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| 15:55.52 | [TK]D-Fender | <Ice_Strike> There is no problem to setup Hardware softphone to connect to Asterisk server from Dedicated Server providers? <- We all lost our secret decoder rings... please de-gibberfy |
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| 17:33.43 | Greenlight | After the destination channel of a bridge hangs up, where is the source channel supposed to go in the Dialplan? In the past it's always seemed to goto the next priority, however I'm getting odd behavoir now |
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| 17:43.07 | nacho2k | hi guys, I am looking for a sip phone, for IOS with g.726 support, any idea? |
| 17:44.12 | *** join/#asterisk lorsungcu_ (~anonymous@65.103.31.33) |
| 17:44.40 | [TK]D-Fender | nacho2k, Why G.726? |
| 17:44.49 | *** join/#asterisk lorsungcu_ (~anonymous@12.40.176.42) |
| 17:44.53 | nacho2k | yes |
| 17:45.39 | Greenlight | But, *why* |
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| 17:46.38 | [TK]D-Fender | did not ask a binary question |
| 17:46.53 | nacho2k | actually for testing, |
| 17:47.10 | [TK]D-Fender | ... |
| 17:47.15 | Greenlight | You could test with G729 or G711, which are more widespread |
| 17:48.03 | nacho2k | ok, it could be g.729 |
| 17:48.08 | *** join/#asterisk qakhan (~qakhan@208.253.91.58) |
| 17:48.12 | Greenlight | In which case, Bria is available for IOS |
| 17:48.13 | qakhan | hi all |
| 17:48.18 | qakhan | i am getting this message |
| 17:48.19 | qakhan | pbx_substitute_variables_helper_full |
| 17:49.33 | *** join/#asterisk brian98 (~brian98@89.101.198.54) |
| 17:49.39 | brian98 | Hello! |
| 17:49.49 | brian98 | Very basic dialplan help here. |
| 17:49.49 | Greenlight | nacho2k: I have a few customers using Bria (g729/g711) to connect in our systemns from iPhones and iPads, and it works really nicely |
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| 17:50.08 | brian98 | what is wrong with this? |
| 17:50.10 | brian98 | exten => _NXXXXXX,n,Dial(SIP/014${EXTEN:0}@provider) |
| 17:50.11 | brian98 | exten => _0X.,n,Dial(SIP/${EXTEN:0}@provider) |
| 17:50.37 | brian98 | I want to check for 7 digit numbers and prefix with 014 and everything else just gets sent to the provider |
| 17:50.40 | Greenlight | Why do ${EXTEN:0} ? |
| 17:50.49 | [TK]D-Fender | brian98, Aside from the fact you're showing us extens with an "n" priority and no "1" in sight? |
| 17:51.18 | [TK]D-Fender | brian98, ... and the fact ":0" as a substring range is useless. |
| 17:51.31 | brian98 | sorry, the 1 is the very first entry in that context |
| 17:51.44 | brian98 | I'll get rid of the substring. |
| 17:51.47 | [TK]D-Fender | brian98, We don't see that/ |
| 17:52.00 | brian98 | exten => _X.,1,Log(NOTICE,Inbound from PBX -- Dial Outbound) |
| 17:52.06 | brian98 | :) |
| 17:52.16 | [TK]D-Fender | brian98, And youa re asking us what's wrong without showing us enough that we don't doubt what you have so far on sight, and you haven't TOLD us what you are having a problem with |
| 17:52.20 | Greenlight | I think ${EXTEN:0} may cause issues, as it's not zero-indexed |
| 17:52.39 | [TK]D-Fender | brian98, We don't see what call might be lookingt at that... or if it even does |
| 17:52.52 | [TK]D-Fender | brian98, Show us complete dialplan and complete call debug. |
| 17:53.08 | brian98 | I'll do a paste bin now. |
| 17:53.24 | [TK]D-Fender | Greenlight, It won't cause issues. |
| 17:53.35 | [TK]D-Fender | Greenlight, And it is effectively the same as not having :0 at all |
| 17:53.39 | Greenlight | It'll just return ${EXTEN} ? |
| 17:53.45 | Greenlight | Fair enough |
| 17:53.52 | qakhan | my asterisk responde very slow |
| 17:53.56 | Greenlight | Wasn't sure - hadn't seen or tried it before :) |
| 17:54.20 | Greenlight | qakhan: Have you turned it up to full power? |
| 17:54.52 | qakhan | when i try to call out it takes 20 secs |
| 17:54.56 | [TK]D-Fender | qakhan, Have you upgraded to a branch that isn't 5 years old and no longer supported? |
| 17:55.37 | qakhan | [TK]D-Fender i have told you before currently i cannot upgrade it |
| 17:55.49 | nacho2k | Greenlight: Thanks Greenlight, I will try it! |
| 17:56.04 | brian98 | [TK]D-Fender: http://paste.debian.net/235292/ |
| 17:56.18 | [TK]D-Fender | qakhan, More like "won't". Which isn't our problem. |
| 17:56.39 | brian98 | If I remove the _NXXXXXX line it dials no problem :/ |
| 17:56.54 | [TK]D-Fender | brian98, You also can't jsut throw around "n" like that. |
| 17:56.57 | *** join/#asterisk chris_n (~Chris@184.7.21.42) |
| 17:57.12 | brian98 | ok... |
| 17:57.14 | [TK]D-Fender | brian98, As the dialplan get's parsed it keeps track of the previous. |
| 17:57.37 | Greenlight | You need to use "goto" based on the Exten, rather than trying to make it branch like that |
| 17:57.41 | [TK]D-Fender | brian98, And the sequence gets broken. Stop trying to use multiple patterns for this. |
| 17:57.51 | [TK]D-Fender | brian98, make each complete unto itself |
| 17:58.04 | qakhan | [TK]D-Fender if i relaod it takes 10 sec to reload |
| 17:58.54 | [TK]D-Fender | qakhan, Let me know when the words "not supported" have sunk in. |
| 17:59.00 | brian98 | best to just goto a different context then if I get a match on the first one? |
| 17:59.00 | Greenlight | qakhan: I doubt anyone's going to try and help if you won't upgrade to a recent and supposed version. |
| 17:59.34 | Greenlight | brian98: You could use labels to acomplish what you're looking to do, and then use GoToIf |
| 17:59.50 | brian98 | ok |
| 17:59.54 | [TK]D-Fender | brian98, What I see in there is simple enough to do within the same context. Stop overlapping your extensions |
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| 18:02.57 | brian98 | [TK]D-Fender: I'm not sure I understand fully the overlapping, is that using N ? |
| 18:03.14 | [TK]D-Fender | brian98, You are trying to span across pattern. Don't |
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| 18:12.14 | volga629 | Hello Everyone, I have some question about fax. https://fpaste.networklab.ca/oN7p/ Got T.38 Re-invite without audio. Keeping RTP active during T.38 session. ? |
| 18:12.47 | MrTAP | bug marshals: can someone review and accept https://issues.asterisk.org/jira/browse/ASTERISK-17523 |
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| 18:20.24 | brian98 | [TK]D-Fender: so I should have 2 labels and use a gotoif based on the matching and do the call in the label? |
| 18:22.07 | file | MrTAP, it's in the queue to get looked at unless a community person wants to take it on sooner |
| 18:22.47 | MrTAP | file: thanks. is this the same queue that it's been in for 2 years, or has the new information put it in a faster moving queue? |
| 18:23.13 | file | same |
| 18:23.58 | MrTAP | :/ any thing that i can do to help speed that up, short of figuring out how to fix it myself? |
| 18:24.27 | file | not really |
| 18:25.22 | MrTAP | alrighty. at a minimum, could you update the affected versions to include teh latest of 1.8 and 11? |
| 18:26.22 | file | done |
| 18:26.25 | gusto | WIMPy: hey, are you there? |
| 18:26.28 | MrTAP | thank you |
| 18:26.33 | file | ...dang it JIRA |
| 18:26.37 | WIMPy | Yes |
| 18:26.54 | gusto | WIMPy: someone wants to mess with me |
| 18:26.55 | file | there. |
| 18:27.10 | gusto | WIMPy: do you know why i think that's the case? |
| 18:27.25 | WIMPy | gusto: That's how life works. Both in the internet and otherwiese. |
| 18:27.26 | gusto | WIMPy: you see how i am connected? |
| 18:27.41 | gusto | WIMPy: I HAVE NO IPV6!!! |
| 18:28.08 | gusto | and lately someone from apple wanted to break in into my account here |
| 18:28.51 | gusto | so, the question is, first someone wants to steal my account, then they disconnect me from IPv6, what happens next? |
| 18:29.06 | gusto | i ll have to check if my asterisk is working ;-) |
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| 18:29.59 | [TK]D-Fender | brian98, Or just make each do it's own thing |
| 18:30.00 | Penguin | [TK]D-Fender: its |
| 18:30.18 | [TK]D-Fender | WAY too fast... |
| 18:30.19 | brian98 | Penguin: grammar nazi |
| 18:30.34 | [TK]D-Fender | I'm wondering if it's botted ;) |
| 18:30.49 | brian98 | It doesnt look like its it |
| 18:30.50 | brian98 | :D |
| 18:31.04 | [TK]D-Fender | it's own |
| 18:31.06 | [TK]D-Fender | ... |
| 18:31.10 | [TK]D-Fender | guess not :) |
| 18:31.13 | brian98 | [TK]D-Fender: , Greenlight: Thanks for help as always! |
| 18:31.19 | brian98 | Bye for now. |
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| 20:28.12 | *** join/#asterisk lorsungcu (~anonymous@12.40.176.42) |
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| 20:31.46 | *** join/#asterisk tompaw (~tompaw@tompaw.xxx) |
| 20:31.49 | tompaw | Hello. |
| 20:32.18 | tompaw | I got a problem with 11.x running on centos. It simply stopped receiving SIP messages. |
| 20:32.21 | lorsungcu | hi |
| 20:32.35 | lorsungcu | pastebin iptables -L |
| 20:32.52 | tompaw | It worked fine for months, and after last restart it's dead. iptables are all opened, not a single rule. |
| 20:33.08 | tompaw | wireshark showing the SIP messages hitting the server just fine/ |
| 20:33.17 | wdoekes | tompaw: module reload chan_sip.so ? |
| 20:33.23 | WIMPy | What has been automatically updated in the meantime? |
| 20:33.49 | tompaw | http://pastebin.com/XEvrD99y |
| 20:33.57 | paulc | selinux: enabled or disabled? |
| 20:34.14 | wdoekes | we believed you when you said your iptables was clean |
| 20:34.16 | tompaw | WIMPy: nothing, it was a power outage. |
| 20:34.23 | tompaw | paulc: disabled |
| 20:34.48 | tompaw | I can access the manager just fine, but the SIP stack seems to be cut off. |
| 20:35.03 | wdoekes | netstat -tulpen | grep asterisk |
| 20:35.16 | wdoekes | does it show 5060? |
| 20:35.21 | wdoekes | what does module reload chan_sip say? |
| 20:35.22 | tompaw | yes |
| 20:35.32 | tompaw | it's bound to 0.0.0.0:5060 |
| 20:36.00 | tompaw | Previous SIP reload not yet done |
| 20:36.12 | wdoekes | well.. there's your problem |
| 20:36.26 | *** join/#asterisk lorsungcu_ (~anonymous@65.103.31.33) |
| 20:36.57 | Chainsaw | You've likely got a "bad" peer in your list that it is getting stuck on. Disable a few and see how you get on. |
| 20:37.42 | tompaw | ok |
| 20:38.33 | Chainsaw | Once you get your SIP channel driver stuck like that, you'll need to restart the core to unwedge it. There's a recent commit to fix that, but it's not made it into a release yet. |
| 20:38.48 | tompaw | I tried restarting both asterisk and the whole server |
| 20:39.05 | tompaw | But it doesn't help, and there's no error messages that would suggest what's wrong... |
| 20:40.09 | WIMPy | You probably have to kill -9 it. |
| 20:40.13 | wdoekes | increase debug level before starting asterisk |
| 20:40.30 | wdoekes | and make sure the logs go somewhere |
| 20:40.33 | WIMPy | Then start it in the foreground with verbose and debug to see what's going on. |
| 20:40.34 | pbxbrian | tompaw: some DNS issue? |
| 20:41.00 | pbxbrian | tompaw: is some peer address in sip.conf not resolving? |
| 20:41.37 | tompaw | pbxbrian: pinged all of them, resolve just fine... |
| 20:42.51 | tompaw | I'm getting Asterisk Ready followed by " == Parsing '/etc/asterisk/cli.conf': Found" |
| 20:43.18 | Chainsaw | tompaw: With verbose & debug please. |
| 20:43.45 | wdoekes | and generate some sip, e.g. with sipsak |
| 20:44.43 | *** join/#asterisk BrokenArrow (~BrokenArr@unaffiliated/brokenarrow) |
| 20:45.13 | tompaw | Chainsaw: that's with -d -vvvvvvvvvvv. And the server is being bombarder by the people trying to log in... |
| 20:45.18 | tompaw | bombarded even |
| 20:45.46 | Chainsaw | tompaw: If you're getting more then just that one line, you're summarising. We want raw data. |
| 20:46.05 | wdoekes | (and perhaps a couple of more -d ) |
| 20:46.58 | Chainsaw | (Yes, that would be nice too.) |
| 20:50.39 | tompaw | http://tompaw.pl/afck.txt |
| 20:55.42 | Chainsaw | tompaw: You're using realtime Asterisk and your PostgreSQL database is unhappy? |
| 20:56.32 | Chainsaw | tompaw: PostgreSQL RealTime: Failed to query 'stl_sipagent@manganium'. |
| 20:56.43 | *** join/#asterisk Wiretap (~wiretap@unaffiliated/wiretap) |
| 20:56.50 | Chainsaw | tompaw: That is where the train is derailing. The rest of the log is just the fallen carriages screeching along the rails. |
| 20:57.51 | *** join/#asterisk pbxbrian (~pbxbrian@unaffiliated/brian98) |
| 20:57.51 | *** join/#asterisk brian98 (~brian98@unaffiliated/brian98) |
| 21:01.40 | Chainsaw | tompaw: So that PostgreSQL server probably isn't running, or you can't resolve the DNS for that. |
| 21:02.02 | lorsungcu | TOMPAW COME BACK |
| 21:02.32 | tompaw | I am |
| 21:02.33 | tompaw | I am |
| 21:02.34 | tompaw | sorry |
| 21:02.37 | lorsungcu | whew |
| 21:02.41 | tompaw | no |
| 21:03.02 | tompaw | psql is not the reason, this is a new query for callback dids, but it has always been crashing |
| 21:03.24 | tompaw | I only use them for REGISTERing agents, so I use a stripped down table, which is good enough |
| 21:03.38 | tompaw | been having that since day 1, it never blocked SIP stack from being loaded... |
| 21:04.18 | tompaw | Never, ever, never. I can actually add the missing column just to prove it :P |
| 21:04.37 | lorsungcu | do it |
| 21:06.42 | *** join/#asterisk bpietro (~bpietro@82.51.236.132) |
| 21:07.45 | tompaw | wow, that helped, I see the SIP errors now in console |
| 21:08.00 | tompaw | why the FCUK was this not a terminal error for 6 months and then out of a sudden it kills the whole thing? |
| 21:08.07 | ^rage^ | hi! |
| 21:08.29 | Chainsaw | So who was right? |
| 21:08.35 | Chainsaw | Hello ^rage^. |
| 21:08.42 | ^rage^ | i have strange problem with Dial and multiple targets |
| 21:08.44 | lorsungcu | that was Chainsaw i think |
| 21:08.49 | Chainsaw | ^rage^: Please elaborate. |
| 21:09.15 | lorsungcu | no way. lets try and troubleshoot rage with no additional information |
| 21:09.35 | Chainsaw | lorsungcu: Only if you do the music and the "Level 2" overlay text. |
| 21:09.48 | lorsungcu | rofl |
| 21:10.08 | lorsungcu | tompaw: could it be that you never reloaded in months and months? |
| 21:13.15 | ^rage^ | scheme: pbx => ispsip. on pbx i have sip peer(1007 for example). I'm doing Dial(SIP/1007&SIP/some_mobile_number_here) on pbx, but mobile phone is switch off and in simple case( Dial(SIP/some_mobile_number_here) i hear message from vodafone) |
| 21:13.58 | ^rage^ | but with multiple dial i just hear silence. no any tones and no sounds from vodafone |
| 21:14.02 | Chainsaw | ^rage^: If SIP/1007 answers first, that call "wins" and the Vodafone channel is dropped. |
| 21:14.23 | lorsungcu | i guess wed want to know what the goal is? |
| 21:14.38 | ^rage^ | Chainsaw: yep. but i recieve sip message "progress" from ispsip |
| 21:14.55 | WIMPy | ^rage^: Great thing. Must mean that you don;t pay for the announcement. |
| 21:15.15 | tompaw | lorsungcu: nope |
| 21:15.31 | tompaw | lorsungcu: I updated kernel 1 month ago and rebooted then |
| 21:15.41 | WIMPy | suggests "make money fast" |
| 21:15.56 | ^rage^ | WIMPy: yep. but i hear nothing. dial don't try call SIP/1007 after recieve progress from ispsip |
| 21:16.33 | Chainsaw | ^rage^: Are you trying to connect SIP/1007 and the vodafone mobile together? |
| 21:16.40 | ^rage^ | Chainsaw: yes |
| 21:16.43 | WIMPy | ^rage^: You mean the local ccall is dropped? |
| 21:16.51 | Chainsaw | ^rage^: Because the dial command you gave means "call both SIP/1007 and vodafone and connect me with whoever answers first". |
| 21:16.58 | lorsungcu | ^^ lorsungcu> |
| 21:16.58 | lorsungcu | i guess wed want to know what the goal is? |
| 21:17.04 | ^rage^ | Chainsaw: progress != answer |
| 21:17.10 | Chainsaw | ^rage^: I am quite aware, thank you. |
| 21:17.47 | ^rage^ | Chainsaw: ok, if * mean progress as answer, why * don't pass sound to calling party? |
| 21:17.49 | Chainsaw | ^rage^: What I am attempting to explain is that what you want to happen and what you are asking to Asterisk to do are not the same. |
| 21:17.51 | tompaw | Previous SIP reload not yet done |
| 21:17.55 | tompaw | The f*ck! |
| 21:18.46 | lorsungcu | still, tompaw ? |
| 21:18.50 | tompaw | yeah |
| 21:18.57 | Chainsaw | ^rage^: No, progress doesn't mean answer. Progress means "number recognised, call being connected". Generally you would expect ringing to follow. |
| 21:19.02 | tompaw | it parsed the postgres, created the realtime peers (which I don't need) and it dies. |
| 21:19.03 | lorsungcu | but you are receiving SIP messages now, anyway? |
| 21:19.18 | tompaw | nope, I was receiving them for like 1 minute and then it went back to this limbo |
| 21:19.31 | tompaw | got verbo 99, sip debug on, console debug on, and see nothing. |
| 21:19.44 | lorsungcu | can you remove the realtime stuff from your dial plan entirely? |
| 21:20.29 | tompaw | I think I'll just get another server and set up everything from scratch, adding it all one by one. |
| 21:20.36 | WIMPy | ChanServ: That's proceeding. |
| 21:20.45 | tompaw | But this is a major fuckup for me, no idea what cause it. |
| 21:20.50 | Chainsaw | WIMPy: Why are you involving Chanserv in this? |
| 21:21.05 | WIMPy | oops |
| 21:21.28 | *** join/#asterisk yang (yang@freenode/sponsor/fsf.member.yang) |
| 21:21.43 | *** join/#asterisk phunguy (santas@will.one.day.hack-the-pla.net) |
| 21:21.48 | Chainsaw | -- SIP/XXX-isdngate-00001c86 is making progress passing it to SIP/XX007-00001c85 |
| 21:21.58 | yang | Hm, any idea where could I define DNS servers in the Polycom 321 menu ? |
| 21:22.28 | Chainsaw | yang: I provision my 670/7000 handsets over HTTP, sorry. Wouldn't know. |
| 21:22.36 | yang | yes |
| 21:22.42 | yang | I am looking in HTTP/GUI |
| 21:22.56 | yang | but yeah maybe I should go into the menu on the handset itself |
| 21:22.57 | lorsungcu | what firmware, yang |
| 21:23.26 | yang | well |
| 21:23.34 | yang | I am not sure where I can check for firmware |
| 21:23.38 | yang | its the original one |
| 21:23.41 | yang | for that phone |
| 21:23.47 | lorsungcu | old? with red and black lines? |
| 21:24.02 | lorsungcu | or the new kind with fancy graphics and grey background |
| 21:24.25 | yang | right |
| 21:24.28 | yang | old |
| 21:25.04 | ^rage^ | Chainsaw: but * mean progress as answer when i use the dial with multiple targets. In general, this is a very controversial decision. What if both the target sent progress? whose voice channel, we must bridge? |
| 21:25.33 | lorsungcu | whatever comes first, ^rage^ |
| 21:26.08 | Chainsaw | lorsungcu: It sounds to me like ^rage^ wants to Originate, not Call with two targets. |
| 21:28.22 | ^rage^ | lorsungcu: ook. good. now it is not working. it's not working because it is all about answer, not progress. progress used for strange things like ring-back(called party hear music, not beep-beeeep...) |
| 21:28.47 | ^rage^ | i think about use confbridge |
| 21:29.10 | yang | lorsungcu: I found the options on tha hardphone menu, if you happen to recall about the HTTP/GUI you can tell me |
| 21:29.12 | WIMPy | ^rage^: Maybe you should give a FULL description of what you want to do. |
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| 21:30.39 | lorsungcu | yang: honestly i could've sworn it was just under the network menu, but if not then i don't recall, sorry |
| 21:30.47 | lorsungcu | yang: only use the new firmware now |
| 21:32.08 | *** join/#asterisk dxd828 (~dxd828@213.205.240.196) |
| 21:32.54 | yang | Do you happen to know if its possible to use Polycom phone as an alarm-clock trigger also ? |
| 21:32.57 | yang | it has the time |
| 21:34.11 | tompaw | Hi. How do I find out what causes "Previous SIP reload not yet done"? |
| 21:35.03 | ^rage^ | WIMPy: create two peers(for example 100 & 101). one of them make a dialplan: exten=>100,1,Progress ; exten=>100,n,Background(some_sound) and call Dial(SIP/100@remote_peer&SIP/101). asterisk mean progress as answer, but not bridge both channels. |
| 21:35.03 | *** join/#asterisk navaismo (~navaismo@189.241.118.172) |
| 21:35.34 | yang | And which would be like the last tested phones for around 100-150 EUR per handset which are prooved as good ? Does Polycom still owns a reputation nowdays ? |
| 21:35.36 | WIMPy | tompaw: You have to wanth it and see how far the previous relod got. |
| 21:35.41 | yang | that is - wired phones |
| 21:35.43 | Chainsaw | tompaw: I would say "incorrect use of realtime peer configs", but then I'd be repeating myself. |
| 21:36.18 | tompaw | Chainsaw: but they all are reported as created just fine |
| 21:36.22 | WIMPy | ^rage^: That was not what I asekd for. What exactely do you want to do. |
| 21:36.31 | tompaw | it lists every single one of them, then goes on to other stuff and dies... |
| 21:36.51 | Chainsaw | ^rage^: Again, dial A & B means call A *or* B. |
| 21:36.57 | Chainsaw | ^rage^: It does not mean "connect A with B". |
| 21:37.18 | ^rage^ | Chainsaw: i know. |
| 21:38.02 | lorsungcu | tompaw: paste bin the new output |
| 21:38.08 | lorsungcu | since adding the column in postgre |
| 21:39.11 | ^rage^ | WIMPy: I want to hear an audio message from Vodafone and the beep-beep of causing local peer. |
| 21:39.35 | lorsungcu | ^rage^: what? |
| 21:39.58 | WIMPy | ^rage^: At the same time? How is that supposed to work? |
| 21:39.58 | *** join/#asterisk ashd (~ashleyd@94-195-121-125.zone9.bethere.co.uk) |
| 21:40.28 | ^rage^ | I'm just doing dial (sip / mobile_number @ ispsip), progress normally processed and I hear the message the cellular operator. |
| 21:40.46 | tompaw | lorsungcu: http://tompaw.pl/afck2.txt |
| 21:41.36 | ^rage^ | if you add another one called peer - I can hear the silence. no beeps, no messages from the cellular operator. |
| 21:41.50 | WIMPy | ^rage^: Sure. If there's only one destination, the situation is easy. |
| 21:44.23 | ^rage^ | WIMPy: yep. |
| 21:44.39 | wdoekes | tompaw: I see no debug, only verbose. try noload=>chan_sip.so in modules.conf and then wait for boot, and then module load chan_sip.so (after setting debug to the appropriate high level) |
| 21:45.15 | tompaw | wdoekes: it was like -ddddddddddd -vvvvvvvvvvvv |
| 21:45.20 | ^rage^ | WIMPy: confbridge + two dial |
| 21:45.43 | wdoekes | perhaps debug is not on your log |
| 21:45.45 | wdoekes | see logger.conf |
| 21:46.15 | WIMPy | ^rage^: And hope that Confbridge accepts audio from unanswered cahnnels. |
| 21:48.00 | ashd | Hi. Anyone worked with smsq in here? |
| 22:03.46 | *** join/#asterisk dpilon (~dpilon@c-50-138-178-238.hsd1.ct.comcast.net) |
| 22:06.13 | Chainsaw | wonders if anyone has written a dnsbl support module for Asterisk yet |
| 22:08.57 | *** join/#asterisk fakhir (~fakhir@unaffiliated/fakhir) |
| 22:16.43 | drmessano | Chainsaw: That would rock |
| 22:17.25 | Chainsaw | drmessano: I have a netblock of Palestinians quite convinced that I offer free calls to Israel. |
| 22:17.38 | Chainsaw | drmessano: It's all very politically relevant, but I'm a bit tired of seeing them try. |
| 22:20.28 | drmessano | I think it would be great to see a baked in solution involving centralized blacklisting. It would be a big step ahead |
| 22:20.38 | pbxbrian | Chainsaw: What is it with those crazy .ps peoples |
| 22:21.33 | ChannelZ | To lob rockets across the border, press 1. |
| 22:21.42 | pbxbrian | Chainsaw: they have some new net blocks. 5.135.0.0/16 5.11.40.0/21 |
| 22:21.59 | Chainsaw | pbxbrian: This is 37.8 at it again. |
| 22:22.18 | pbxbrian | 37.8.0.0/17 ? |
| 22:22.32 | Chainsaw | 37.8.0.0/20, yeah. |
| 22:22.34 | Chainsaw | Gaza something. |
| 22:22.40 | pbxbrian | Chainsaw: its /17 |
| 22:22.59 | Chainsaw | pbxbrian: So far both callers fit in sub-route 1.1 - BSA-GAZA, which is a /20 |
| 22:23.06 | pbxbrian | Chainsaw: ok |
| 22:23.40 | pbxbrian | Chainsaw: I just drop everything in iptables from .ps ip addresses on any public facing sip servers |
| 22:23.57 | Chainsaw | pbxbrian: I have two French server operators at it as well. |
| 22:24.05 | pbxbrian | Chainsaw: ovh? |
| 22:24.10 | Chainsaw | pbxbrian: How'd you guess? |
| 22:24.52 | pbxbrian | Chainsaw: Hetzner were hosting palgsm.ps also |
| 22:25.08 | pbxbrian | Chainsaw: but they have shut them down now I think. |
| 22:25.09 | pbxbrian | C |
| 22:25.19 | *** join/#asterisk TimeRider (~steve@timerider.plus.com) |
| 22:26.10 | ChannelZ | I get it from anywhere and everywhere. AWS, Rackspace, Russia.. though the last one I see in my console was palestine |
| 22:26.15 | ChannelZ | 37.75.209.73 |
| 22:26.29 | pbxbrian | ChannelZ: 37.75.212.0/22 |
| 22:26.46 | pbxbrian | ChannelZ: That was quite recent.. |
| 22:26.52 | ChannelZ | and the one before that was in 213.244.66.0/24 |
| 22:27.14 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 22:27.40 | pbxbrian | ChannelZ: Haven't come across that block, added! |
| 22:28.16 | ChannelZ | Is there already a sip-centric BL somewhere? |
| 22:28.31 | ChannelZ | I mean writing an interface isn't much of a problem. |
| 22:28.46 | Chainsaw | DroneBL seems appropriate. |
| 22:29.02 | pbxbrian | There is something I was reading about a while back. |
| 22:29.45 | pbxbrian | siprbl google throws this back http://www.opentek.ca/secure-your-pbx-servers-with-our-sippot-rbl-client |
| 22:30.25 | pbxbrian | there is an israeli company doing something similar for $ |
| 22:30.55 | ChannelZ | hmm |
| 22:31.39 | pbxbrian | it is really open to abuse. |
| 22:33.11 | WIMPy | BTW: Is there something special going on here or can someone else confirm that a Goto() to an invalid destination will use the i extension on the destination context and not in the current context as |
| 22:33.16 | WIMPy | the documentation says. |
| 22:35.01 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 22:35.01 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 22:36.17 | lorsungcu | just checked some of my servers, also getting lots of 37.8 attempts. |
| 22:36.43 | pbxbrian | lorsungcu: just drop em |
| 22:36.48 | lorsungcu | already am |
| 22:37.23 | pbxbrian | There is a twitter account that posts sipvicous attempts |
| 22:44.38 | WIMPy | And since about 2 weeks BLF seems to be a bit hit or miss. I didn't see anything that looked relatrd to me in the changelog however. |
| 22:45.05 | pbxbrian | VOIP Fraud.. http://voipsa.org/blog/2010/09/29/voip-attackers-sometimes-they-come-back/ |
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| 22:46.59 | *** mode/#asterisk [+o pabelanger] by ChanServ |
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| 23:00.17 | Ice_Strike | Hello |
| 23:01.17 | lorsungcu | hi |
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