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| 01:29.39 | rogersja | what are the required packages to compile asterisk 11.5 with support for postgres (cdr, cel, realtime)? |
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| 01:59.36 | dijib | Penguin: how are you? |
| 01:59.56 | dijib | join #bes |
| 02:03.01 | rogersja | nm… --with-postgres=/usr/pgslq91/ seemes to work |
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| 02:44.25 | hebber | Hi, have a problem with attended transfer *2 (works), but then fails to detect the extension on 11.4 - played with different codecs and my sip.conf use dtmf=rfc2833 |
| 02:44.39 | hebber | dtmfmode=rfc2833 |
| 02:51.31 | igcewieling | enable dtmf logging to verify the dtmf is received correctly. |
| 02:56.00 | hebber | igcewieling: good idea - did that - perfect detection of dtmf both on attentended transfer and extensions - why doesn't it try to transfer then ? |
| 02:56.06 | hebber | scratching my head |
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| 02:59.02 | igcewieling | The moving cursor writes, and having written, blinks on. -- Unknown |
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| 03:00.05 | igcewieling | hebber: at least you now know it isn't a dtmf issue. Read the features.conf.sample in your Asterisk source code, also check "core show application dial" and maybe "core show function CHANNEL" |
| 03:00.07 | hebber | trying to recall what I have changed the last 2 weeks |
| 03:02.08 | hebber | I did reorganize the context 2 days ago - maybe thats where the hiccup is? |
| 03:02.23 | hebber | no it happened before that |
| 03:02.25 | igcewieling | likely |
| 03:03.39 | hebber | thanks for pointing my in a likely direction to where I should start the search |
| 03:22.23 | hebber | igcewieling: its the handling of the local channel that fails to make the transfer - now just need to do the juggling right :) |
| 03:30.25 | hebber | In the dialplan how to detect what type of channel it is? in My case handling of SIP or LOCAL |
| 03:31.21 | hebber | I need to preserve the accountcode and when transferred the local channel looses the accountcode |
| 03:32.58 | hebber | CHANNEL(name)? |
| 03:33.27 | hebber | CHANNEL(channeltype) ofc :) |
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| 04:06.26 | hebber | igcewieling: works like a dream now - by using exception handling on CHANNEL(channeltype)=local |
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| 04:43.17 | asteriskmonkey | is there anyway to see what timezone an asterisk server is running as? |
| 04:43.33 | asteriskmonkey | or the time as asterisk sees it |
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| 05:06.32 | ChannelZ | I imagine 'date' from a shell. |
| 05:16.12 | carrar | You're dating a girl name shell? |
| 05:19.03 | ChannelZ | touch; finger; unzip; mount; yes; yes; yes; |
| 05:21.23 | asteriskmonkey | something is werid , im running asterisk 11.4 with security on |
| 05:21.27 | asteriskmonkey | and im getting in my logs |
| 05:21.36 | asteriskmonkey | NOTICE[23657][C-0000071f]: chan_sip.c:25189 handle_request_invite: Failed to authenticate device 8000<sip:8000@myserverip>;tag=a309607f |
| 05:21.56 | asteriskmonkey | how do i get lines like that to show the ip :/ |
| 05:22.35 | ChannelZ | security log |
| 05:22.42 | asteriskmonkey | its on |
| 05:22.46 | asteriskmonkey | it shows nothing |
| 05:23.03 | ChannelZ | then it's broken.. not writing to it or something |
| 05:23.14 | asteriskmonkey | nope its wrting stuff to it |
| 05:23.25 | asteriskmonkey | is device not a part of sip? |
| 05:23.29 | ChannelZ | what does 'logger show channels' say? |
| 05:23.29 | asteriskmonkey | nm it is |
| 05:23.32 | asteriskmonkey | it says chan sip |
| 05:23.38 | *** join/#asterisk vlad_starkov (~vlad_star@195.68.167.197) |
| 05:23.49 | asteriskmonkey | syslog.local0 Syslog Enabled - NOTICE WARNING ERROR SECURITY |
| 05:24.17 | asteriskmonkey | security is on :/ |
| 05:24.18 | ChannelZ | oh.. well I don't know how that works with syslog |
| 05:24.30 | asteriskmonkey | i have it the same for a flat file |
| 05:24.49 | carrar | asteriskmonkey, look at the SIP Packets |
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| 05:25.03 | asteriskmonkey | i could look at sip packets but the point is to get this working in asterisk |
| 05:25.07 | asteriskmonkey | something is clearly broken |
| 05:25.18 | carrar | Probably could be subscribes |
| 05:25.44 | ChannelZ | then it should be writing InvalidAccountID or InvalidPassword type events |
| 05:26.03 | carrar | Maybe you should install Windows 8.1 instead |
| 05:26.16 | carrar | runs |
| 05:26.37 | asteriskmonkey | dont know if i should open a ticket for this or not |
| 05:26.43 | asteriskmonkey | hence prodding here :/ |
| 05:26.53 | ChannelZ | I'm running 11.4 and it works |
| 05:27.05 | carrar | yeah don't bother looking at what the packets are |
| 05:27.08 | carrar | jsut opentickets |
| 05:27.26 | ChannelZ | he's not trying to debug the error (I don't think) but wondering why it's not logging the auth failure. |
| 05:27.39 | asteriskmonkey | correct |
| 05:27.39 | ChannelZ | IE to ban the bitch |
| 05:27.45 | carrar | find out what it is first |
| 05:27.47 | ChannelZ | which I do |
| 05:28.23 | asteriskmonkey | i seem them going up incrementally |
| 05:28.37 | asteriskmonkey | its a handle_request_invite though |
| 05:28.38 | asteriskmonkey | not a reg |
| 05:29.02 | ChannelZ | I dunno if syslog logging is barfing it up, maybe remove security from that and leave it just on the flat file log and see if it works. |
| 05:29.09 | ChannelZ | All I can say is it does here. |
| 05:29.22 | asteriskmonkey | ive done that |
| 05:29.25 | asteriskmonkey | its the same stuff |
| 05:29.41 | ChannelZ | Is it logging other security events? Like SuccessfulAuth? |
| 05:30.38 | asteriskmonkey | yeah |
| 05:31.01 | asteriskmonkey | this looks like some form of account sniffing via the invite? |
| 05:31.47 | ChannelZ | script kiddies trying anything to make a free call |
| 05:32.03 | carrar | What IP is it coming from |
| 05:32.08 | ChannelZ | though 8000 seems a little arbitrary |
| 05:32.14 | ChannelZ | That's what he's trying to find out |
| 05:32.16 | asteriskmonkey | well its been goign up |
| 05:32.24 | carrar | if only there was a way |
| 05:32.33 | asteriskmonkey | buts its by invite and asterisk is fail for reporting on invite |
| 05:32.43 | asteriskmonkey | i can edit the source and fix |
| 05:32.46 | asteriskmonkey | but thats stupid |
| 05:33.09 | ChannelZ | I just changed the password of a device behind it's back and placed a call and it logged it just fine here. I have no idea why yours isn't. Or you're blind. |
| 05:33.38 | carrar | asteriskmonkey: ngrep -tqd eth0 -W byline -l udp and port 5060 |
| 05:33.45 | asteriskmonkey | yeah ive done that |
| 05:33.49 | asteriskmonkey | i see the ip thats doing it |
| 05:33.51 | carrar | and? |
| 05:34.07 | asteriskmonkey | and? thats not the points, this is an asterisk security vulnerabilty |
| 05:34.11 | carrar | haha |
| 05:34.12 | carrar | ok |
| 05:34.45 | ChannelZ | http://pastebin.com/2QPnLKYB |
| 05:34.57 | ChannelZ | it's not a "vulnerability" |
| 05:38.20 | ChannelZ | You have a peer called [8000]? |
| 05:38.41 | asteriskmonkey | no |
| 05:39.30 | ChannelZ | so no big loss. Firewall the jerk and move on with your life |
| 05:40.05 | asteriskmonkey | yeah its about automation though, if i cant get it reporting right from sec out put :/ |
| 05:40.14 | asteriskmonkey | meh ill just edit the f'ing source and recompile |
| 05:40.16 | asteriskmonkey | :) |
| 05:40.30 | ChannelZ | shrugs |
| 05:40.46 | ChannelZ | should already be generating those events, dunno why. You're really running 11.4.0? |
| 05:42.37 | ChannelZ | maybe your res_security_log is borken. |
| 05:42.47 | carrar | maybe OWNED! |
| 05:43.01 | carrar | 0day! |
| 05:47.19 | asteriskmonkey | yeah |
| 05:47.21 | asteriskmonkey | you know what |
| 05:47.27 | asteriskmonkey | it appears its not spitting out right |
| 05:47.39 | ChannelZ | orrly |
| 05:47.42 | asteriskmonkey | including security in messages lines or systlog lines |
| 05:47.46 | asteriskmonkey | = fail |
| 05:47.50 | asteriskmonkey | have to use staright up |
| 05:47.55 | asteriskmonkey | security => security :/ |
| 05:48.05 | ChannelZ | didn't I suggest that and you said you already were? |
| 05:48.12 | asteriskmonkey | you suggest flat file |
| 05:48.21 | asteriskmonkey | I was doing flat file in messages |
| 05:48.31 | ChannelZ | eh? |
| 05:48.45 | asteriskmonkey | thats bs though, i stated security in the other named logs it should have borked the proper outs in all of then |
| 05:48.52 | carrar | opening 1 file twice? |
| 05:49.03 | asteriskmonkey | no :/ |
| 05:49.29 | asteriskmonkey | its a good gotcha i guess, ill probably submit minor bug report then |
| 05:49.33 | asteriskmonkey | thanks :) |
| 05:56.25 | carrar | asteriskmonkey, you running opensips? |
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| 06:00.23 | asteriskmonkey | carrar, no not for this instance |
| 06:00.42 | carrar | yeah I was just seeing you had opensips & freeswitch |
| 06:01.06 | asteriskmonkey | i do have freeswitch |
| 06:01.18 | asteriskmonkey | and kamailo |
| 06:02.11 | asteriskmonkey | looks in his stack of floppys, look a copy of Yate |
| 06:02.32 | carrar | no sipwitch? |
| 06:02.34 | carrar | wtf |
| 06:03.43 | asteriskmonkey | i do all my load balancing with bash, sed and awk :) |
| 06:03.45 | asteriskmonkey | lol |
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| 09:28.38 | floren | morning all :) |
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| 09:33.22 | ThothCastel | what is a cdr ? |
| 09:37.53 | floren | call data records |
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| 09:37.59 | floren | ~book |
| 09:37.59 | infobot | Asterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
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| 09:50.48 | mic_ | Greenlight: the issue still happens |
| 09:51.06 | mic_ | Greenlight: it's not every time - it happens every now and then. i.e. over the time of 4 hours ca. 20 times |
| 09:51.18 | mic_ | (and yes, now I am recording to RAM) |
| 09:51.32 | mic_ | I am thinking, that maybe moving it to the hdd+transcoding |
| 09:51.42 | mic_ | could be an issue? (especially when a few calls "build up") |
| 09:52.40 | mic_ | I will switch off recording tonight to see tomorrow what's the problem. |
| 09:52.55 | mic_ | (I'm also going there tomorrow to be on-site and possibly localize the issue) |
| 09:58.51 | floren | should i compile asterisk with G711_NEW_ALGORITHM? |
| 09:59.06 | floren | it is "slower" but cleaner |
| 09:59.38 | floren | let me know your results, if you tried that option |
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| 10:09.55 | hanuman | hi |
| 10:10.39 | hanuman | i am new to asterisk how can i check asterisk installation working fine and load test |
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| 10:15.11 | hanuman | i am new to asterisk how can i check asterisk installation working fine and load test |
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| 10:44.03 | kchehab | hi |
| 10:45.15 | kchehab | i am using asterisk 11 ,when UA registered through UDP it works fine ,but when it use TLS cant dial *98 SIP/2.0 401 Unauthorized as its registred |
| 10:46.22 | kchehab | how can i solve that ? |
| 10:49.33 | floren | leaving for work, another bright day in telecom :) |
| 10:50.59 | hanuman | i am new to asterisk how can i check asterisk installation working fine and load test |
| 10:53.20 | kchehab | hanuman you can use sipp to test it |
| 10:54.42 | hanuman | kchehab: any other knowledge i need to know |
| 10:55.07 | hanuman | kchehab: asterisk is enough for that tool |
| 10:59.07 | kchehab | hanuman check core parameters and sip setting using asterisk shell ex: sip show settings |
| 11:10.31 | ThothCastel | anybody familiar with 3cx softphone? |
| 11:11.47 | ThothCastel | I have my asterisk pbx running on a virtual machine and need to install 3cx softphone on a computer to use the asterisk pbx, I downloaded 3cx but it's installing the pbx server for windows ! Do I need to install the pbx for windows? |
| 11:27.29 | ThothCastel | help pls |
| 11:27.45 | ThothCastel | I only need the 3cx softphone client |
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| 11:34.24 | kchehab | any idea why do asterisk send 401 unauthorized when trying to call the voice mail 98 while the UA is registered ?? |
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| 11:39.17 | mic_ | kchehab: sipp -> thx! (I actually was looking for something like that) |
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| 11:43.10 | mic_ | funny thing - sipp is called sip-tester under ubuntu |
| 11:49.32 | kchehab | mic_ yes |
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| 11:57.11 | linuxasterisk | Would anyone want to help me with a basic set up question? I've never run into this problem before using the dedicated asterisk distros, but I wanted to use my existing box to set up asterisk and I set it up from source. I cannot register sip phones. Debug info: http://pastebin.com/Ez8uqs0M |
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| 12:26.46 | izbushka | hi |
| 12:26.57 | izbushka | is it possible to check in dialplan whether sip user is registered or not? |
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| 12:27.35 | WIMPy | core show function SIPPEER |
| 12:27.57 | izbushka | thanks |
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| 12:42.12 | The_Miser | never thought of that ... (seriously) |
| 12:42.23 | The_Miser | Oh heck wrong window. :( |
| 12:43.05 | kchehab | Guys just any one experince TLS with asterisk 11.3 |
| 12:44.47 | WIMPy | hasn't tried TLS for a long time. |
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| 12:55.28 | d45h | I am looking for an VoIP interface in Python which would support audio data passing and getting realtime. What would you suggest me? |
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| 12:56.42 | [TK]D-Fender | d45h: http://tinyurl.com/khz9t2h |
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| 12:58.58 | d45h | [TK]D-Fender: :) I am googling and programming 5 days now with no success, just want to know if something doesn't missing me |
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| 12:59.09 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 12:59.31 | eject_ck1 | Hi guy, I have cisco 7960 phone (only 1 line configure) and in case if I'm on call and someone call me I see that new call (call wait). On other phones I don't see and caller hear busy tone. Is this particular phone feature or configuration ? How can enable call waiting for all extensions? |
| 12:59.39 | [TK]D-Fender | d45h: tons of libraries right there. It's been done. It's been documented and took less than 3 seconds to Google. |
| 12:59.54 | d45h | [TK]D-Fender: Which one? |
| 13:00.03 | eject_ck1 | Another question I wasn't able to find answer is how can I play some sound to caller in case extension busy ? |
| 13:00.15 | [TK]D-Fender | d45h: I'm sure they all work. |
| 13:00.29 | d45h | [TK]D-Fender: Name one please |
| 13:00.33 | WIMPy | eject_ck1: Unless you use the deprecated call-limit, it's a phone feature. |
| 13:01.01 | [TK]D-Fender | d45h: I see tons of names right from the results of that Google search. And that's without actually even following the links |
| 13:01.15 | [TK]D-Fender | d45h: Can you not see library names and full app names? |
| 13:02.04 | [TK]D-Fender | eject_ck1: Your phone has to be set up to accept more calls (hence call-waiting) |
| 13:02.15 | [TK]D-Fender | eject_ck1: This is between your phone... and your setup of it. |
| 13:02.41 | *** join/#asterisk jetlag (~jetlag@pool-71-168-203-184.cmdnnj.east.verizon.net) |
| 13:03.36 | eject_ck1 | I'm not using call-limit |
| 13:03.45 | d45h | [TK]D-Fender: ok, otherwise, I tried PJSIP Python binding, there is no support for custom port in Python, so you cannot do that. Also tired telepathy-python, it has exactly what I need (using GStreamer library, it's great library), but it is no longer working and supported. Developers suggested me to use gobject-introspection with telepathy-glib, but there are no examples, no support, and I have no idea how to make it work. Others are derivates |
| 13:03.47 | d45h | <PROTECTED> |
| 13:04.50 | *** join/#asterisk ruben231 (~OpenDial@112.198.79.151) |
| 13:05.20 | d45h | [TK]D-Fender: Derivates I tried sofsip-cli (this client has unresolved issue you cannot run it) and simplesip |
| 13:06.35 | d45h | [TK]D-Fender: But maybe you know about some project that support audio data passing and getting realtime? What would you suggest me? |
| 13:08.38 | d45h | I was wondering I could use virtual sound card and pass data there, but it's not good solution. |
| 13:08.47 | *** join/#asterisk Cuzner (~ccuzner@198.41.29.45) |
| 13:09.35 | d45h | It would work, I was able to make work PJSIP, but without custom port - only sound card and WAV. |
| 13:11.34 | *** join/#asterisk matt_ (~matt@ccpc-mwr.bath.ac.uk) |
| 13:11.35 | d45h | I also thought that I could create "LIVE" WAV (adding data on the go), but this cannot be done, because PJSIP is controlling header of WAV. |
| 13:14.04 | d45h | I think it couldn't be done easily in Python right now. [TK]D-Fender: maybe some asterisk override? some other protocol? (not SIP) e.g. jabber? maybe I could find the way in another library, but SIP libraries are not good enough :( |
| 13:14.27 | d45h | monolog :) |
| 13:14.38 | [TK]D-Fender | indeed |
| 13:14.39 | d45h | monologue :) |
| 13:20.04 | leifmadsen | monocles |
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| 13:25.35 | *** join/#asterisk kchehab (~kchehab@77.42.241.66) |
| 13:25.43 | kchehab | [TK]D-Fender hi , |
| 13:26.34 | kchehab | [TK]D-Fender is there a big in asterisk 11.3 since i am using TLS and the client is registered ,but when the client try to call *98 for ex asterisk sip debug shows 401 un authorized |
| 13:27.32 | *** join/#asterisk brad_mssw (~brad@shop.monetra.com) |
| 13:28.18 | mic_ | sanity check - can you blame everything on acpi? ;) |
| 13:28.18 | leifmadsen | kchehab: that doesn't sound like an issue... that sounds like standard authentication |
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| 13:28.32 | [TK]D-Fender | it is. |
| 13:28.38 | leifmadsen | kchehab: you'll need to show some debugging and pointing at what might actually be wrong |
| 13:29.13 | [TK]D-Fender | TLS secures the SIP communication.. it is clealry arriving and being decoded... and Asterisk simply doesn't like what they're saying. |
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| 13:36.54 | kchehab | [TK]D-Fender what exactly want to see sip debug ? or verbose |
| 13:37.34 | [TK]D-Fender | kchehab: Me? Nothing really :) |
| 13:38.11 | kchehab | [TK]D-Fender hehehe i mean if you want to assist me ,i Spent 8 hours of debugging with no output |
| 13:39.20 | kchehab | [TK]D-Fender as i copy paste same tls config from a working 1.8.4 to iy and same out put --successful registration with fail in calling an extension |
| 13:39.58 | fullstop | obi100 arrives tomorrow. |
| 13:40.47 | kchehab | [TK]D-Fender from where do you advise me to restart my debugging or if you can help me if you have an idea |
| 13:41.01 | [TK]D-Fender | kchWell you can go ahead and show a real config from both sides and the full call and maybe someone will have something to say about it. |
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| 13:48.45 | kchehab | [TK]D-Fender please and i will appriciated if you can check my config http://pastebin.com/GBkT1vn6 |
| 13:49.48 | kchehab | [TK]D-Fender thanks in advance |
| 13:50.13 | [TK]D-Fender | cashes his karma and runs |
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| 13:52.00 | [TK]D-Fender | kchHalf a picture doesn't do anybody any good. next I'll show you a catalog picture of a brand new car... and then with just that... you tell me why it crashed. |
| 13:52.24 | kchehab | [TK]D-Fender hehehehe |
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| 13:53.11 | kchehab | [TK]D-Fender really you make me laughh after a hard day |
| 13:53.47 | [TK]D-Fender | kchehab: Your table is an unreadable line-wrap mess. And there is no debug for the actual call |
| 13:53.56 | [TK]D-Fender | kchehab: For starters |
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| 13:56.38 | kchehab | [TK]D-Fender please find my core dump and sip debug in this link http://pastebin.com/fYmWfEBD |
| 13:58.00 | [TK]D-Fender | kchehab: that looks like a 401 challenge that the other sides isn't coming back with an auth for. |
| 13:58.28 | kchehab | [TK]D-Fender you mean my softphone ? |
| 13:59.11 | [TK]D-Fender | yes |
| 13:59.36 | [TK]D-Fender | Who else do think "other side" means... when Asterisk is responding with a 401? |
| 13:59.38 | kchehab | [TK]D-Fender i test it with 1.8.4 same softphone named BRIA and it works even i t remove tls and i set UDP its works too |
| 14:00.09 | [TK]D-Fender | 1.8.4 is very outdated. |
| 14:00.23 | *** join/#asterisk gerhard7 (~gerhard7@77-172-47-159.ip.telfort.nl) |
| 14:00.27 | [TK]D-Fender | If you're asking about "bugs" and running that version.... don't. |
| 14:00.27 | Dovid | is there anyone here in Germany that can do a test call? |
| 14:00.48 | WIMPy | Dovid: Where? What? ...? |
| 14:01.23 | Dovid | WIMPy: I need a test call from a mobile phone to a free phone number. i need the call to be done blocked. i want to see if i still get the caller information so i block certain callers |
| 14:01.28 | kchehab | [TK]D-Fender what can be the solution bro ... |
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| 14:01.45 | kchehab | [TK]D-Fender is there an advanced debug i can do ? |
| 14:02.03 | WIMPy | Dovid: Go on. |
| 14:02.17 | igcewieling | kchehab: The first thing you should do is upgrade to the latest 1.8.x |
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| 14:02.24 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 14:02.30 | kchehab | i am using now 11.3 |
| 14:02.36 | [TK]D-Fender | 11.5 is out |
| 14:02.41 | kchehab | igcewieling 1.8.4 was working fine :) |
| 14:03.09 | igcewieling | kchehab: what carrier? |
| 14:03.25 | kchehab | igcewieling what do you mean by carrier ? |
| 14:03.26 | Dovid | WIMPy: can you call 08005892701 blocked from a mobile phone? |
| 14:03.31 | [TK]D-Fender | kchehab: I don't see the bria side configs. I don't see the non-TLS call for comparison. You aren't being thorought and are providing backup in useless piecemeal fashion. |
| 14:04.05 | igcewieling | kchehab: you are running Bria on ANDROID. Have you tried running it from a PC to eliminate any mobile oddities? |
| 14:05.00 | kchehab | igcewieling i run same config same android to 1.8.4 and it works fine |
| 14:05.01 | WIMPy | Dovid: on the way |
| 14:05.36 | kchehab | [TK]D-Fender if you guide me i will be thanksfull i would do any thing to solve my problem :( |
| 14:06.10 | WIMPy | Dovid: Numbner ends in 0306. |
| 14:08.06 | Dovid | WIMPy: You called with your caller ID blocked? |
| 14:08.15 | WIMPy | yes |
| 14:08.58 | [TK]D-Fender | kchehab: "enknetcom2" <-- why is THIS the "defaultuser"? |
| 14:09.56 | Dovid | Wimpy: can you call again blocked please? |
| 14:10.04 | Dovid | WIMPy: I really apreceate it |
| 14:10.15 | WIMPy | done |
| 14:10.26 | Dovid | checking |
| 14:12.19 | Dovid | WIMPy: Here is the dumpchan: http://pastebin.com/ZxG46jf0 when i tried to Noop the variable ${CallerIDNum} it came as blank. |
| 14:12.34 | *** join/#asterisk russum (~russum@94.139.132.13) |
| 14:12.36 | *** part/#asterisk russum (~russum@94.139.132.13) |
| 14:12.50 | kchehab | [TK]D-Fender just to link the customer to the organization |
| 14:13.31 | WIMPy | Dovid: Err, CALLERID() doesn't give it to you? |
| 14:13.48 | bulkorok | callerid(num)... |
| 14:13.50 | Dovid | WIMPy: I didnt think it owuld cause it's blocked |
| 14:14.01 | kchehab | [TK]D-Fender i print my sip settings and a call with udp and one with tls protocol same softphone http://pastebin.com/kFaeUZT5 |
| 14:14.02 | Dovid | ${CALLERID(num)}) ? |
| 14:14.07 | Dovid | oops |
| 14:14.12 | Dovid | ${CALLERID(num)} |
| 14:14.24 | Dovid | that wont show it cause the caller is calling blocked. |
| 14:14.39 | WIMPy | Err, yes. I shouldn't do so muchg auto-correction ;-) |
| 14:15.08 | Dovid | so I need to get that value in to a variable to check against it. the call is coming in on a SS7 link |
| 14:15.45 | Dovid | WIMPy: Can I ask for another call? |
| 14:16.03 | WIMPy | There it is |
| 14:16.17 | Dovid | WIMPy: Stragne. |
| 14:16.27 | Dovid | it's working. i dotn know why. last I checked it did not |
| 14:16.36 | Dovid | can you calla gain NOT blocked so i can see the difference? |
| 14:16.43 | WIMPy | that's what I'd expect. |
| 14:17.15 | WIMPy | You got it. |
| 14:17.27 | [TK]D-Fender | kchehab: Different UserSAgent between those calls, different codec list |
| 14:17.46 | [TK]D-Fender | kchehab: That is not a siple protocol change. What you are showing is completely untrustworthy |
| 14:22.04 | kchehab | [TK]D-Fender what i did is that i change the signalling protocol only ,please can you mark where did you find the diffence |
| 14:22.28 | kchehab | [TK]D-Fender and even if its a codec issue why is the 401 not authorized |
| 14:23.17 | [TK]D-Fender | kchehab: UA string should not be different from these 2 calls. |
| 14:26.25 | *** join/#asterisk jasonwert (~w3rt@96-42-150-164.dhcp.trcy.mi.charter.com) |
| 14:26.34 | kchehab | [TK]D-Fender i will be appriciated if you guide me what to do ,since in the capture i sent you i just do a call with udp ,unregister ,register using TLS and do another call and i didnt get your point by UA string should be different ,please can you specify more or give me the lines in the capture ,that you build you opinion on |
| 14:26.53 | igcewieling | kchehab: ALL SIP transactions start out with a 401, then the client resends the password encoded by the nonce from the first rejection |
| 14:27.22 | *** join/#asterisk dash_ (~d45h@unaffiliated/dash-/x-7576607) |
| 14:27.55 | kchehab | igcewieling thanks alot now i can get your point ,that mean bria is not sending a digest authentication |
| 14:28.28 | kchehab | igcewieling but my question is why its working with other versions ? same config |
| 14:28.29 | igcewieling | kchehab: No. What I'm saying is that you should EXPECT a 401 at the start. |
| 14:28.41 | kchehab | should be a cause right ? |
| 14:28.49 | *** join/#asterisk BorjaGVO (d51beb92@gateway/web/freenode/ip.213.27.235.146) |
| 14:29.00 | igcewieling | kchehab: your setup is so complex I doubt anyone wants to troubleshoot it for you. |
| 14:29.04 | kchehab | igcewieling may it be the .ca .key certificates |
| 14:29.13 | igcewieling | The most you can hope for is little bits of help for specific things |
| 14:29.36 | igcewieling | kchehab: no idea. SIP encrpytion is far, far, too new for us to trust it. |
| 14:29.45 | kchehab | igcewieling or if you can guide me from where i can start |
| 14:30.02 | kchehab | igcewieling ok thanks |
| 14:30.13 | igcewieling | kchehab: the Asterisk book is always a good start, but since SIP encrption is so new there is not a lot of documentation |
| 14:30.16 | BorjaGVO | Hello. I don't know why AMI is returning status UNKNOWN for peers (I've got quelify=yes): http://pastebin.com/MqFa5WF5. Any idea why this can happen? |
| 14:31.00 | igcewieling | BorjaGVO: I doubt AMI is the one returning it. Does sip show peers show the status as unknown too. Also unknown means the peer has not registered so there has been no qualifies happening |
| 14:31.14 | kchehab | please just one more question , i have asterisk installed freepbx , i try yum update and it didnt upgraded ,how can i upgrade the my packaged to let it install 11.53 |
| 14:31.31 | [TK]D-Fender | BorjaGVO: 127.0.0.1 <- IPaddress: -none- <---- maybe this giant sign that nothing has registered to it... |
| 14:31.45 | igcewieling | ~freepbx |
| 14:31.45 | 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 |
| 14:31.55 | igcewieling | that is for you kchehab |
| 14:32.17 | kchehab | heheh i imagine that you may answer me with this |
| 14:32.20 | [TK]D-Fender | kchehab: You showed SIP realtime configs there... FreePBX doesn't support realtime... so what was this first thing you showed us? |
| 14:32.27 | kchehab | at all that alot guys |
| 14:32.59 | igcewieling | I am so glad my boss is as skeptical and cynical of new technology as I am. |
| 14:33.09 | kchehab | [TK]D-Fender without get angry and confused ,i was tetsing new version of elastix |
| 14:33.28 | kchehab | 3.0 |
| 14:33.41 | kchehab | with organizations and ..... |
| 14:33.53 | [TK]D-Fender | kchehab: You are telling us things that completely differ from your description, and your UA doesn't match on the call. NOTHING matches. This is a waste of our time. |
| 14:34.57 | kchehab | [TK]D-Fender trust me till now i didnt get your point by what you are trying to say to me (your UA doesn't match on the call. NOTHING matches. ) |
| 14:35.21 | kchehab | you mean doesnt reply |
| 14:35.27 | [TK]D-Fender | kchehab: Sorry, you've already spent the last bit of trust I had on this. |
| 14:36.14 | kchehab | [TK]D-Fender i didnt lie on you ,i was debugging 11.5 and i didnt use the elastix interface as i print you the databse + config at all thanks |
| 14:36.22 | igcewieling | I miss the days of old when people came in, asked specific questions, and provided useful pastebins. |
| 14:36.45 | kchehab | igcewieling not all people has your knowledge |
| 14:36.46 | [TK]D-Fender | moves on to something more productive |
| 14:37.08 | igcewieling | kchehab: all people can produce cohereent questions and debug if they try. |
| 14:37.17 | igcewieling | moves on as well |
| 14:44.31 | *** join/#asterisk w9sh (~sph@50-79-224-193-static.hfc.comcastbusiness.net) |
| 14:49.45 | BorjaGVO | igcewieling: Yes, the sip show peers shows status as unknown too. |
| 14:53.10 | *** join/#asterisk julgr (~julgr@c-24-143-72-222.customer.broadstripe.net) |
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| 15:05.55 | Dovid | WIMPy: Thanks a lot. it seems the CID always comes in. Can I ask you to call two more times. one blocked once not? both should not work |
| 15:10.28 | *** join/#asterisk jasonwert (~w3rt@75-134-81-98.static.aldl.mi.charter.com) |
| 15:14.52 | WIMPy | Dovid: Negative. I got thru even without CLIR. |
| 15:15.08 | Dovid | WIMpy: one sec. |
| 15:17.44 | *** join/#asterisk bdfoster (~bdfoster@unaffiliated/bdfoster) |
| 15:23.47 | Dovid | WIMPy: Can you try now? |
| 15:25.54 | WIMPy | "The person you have called is temporarily unavailable" in both cases. |
| 15:31.32 | *** join/#asterisk puzzled (~patrick@2001:980:5e31:1:dcaa:d60c:5151:cdf0) |
| 15:40.01 | *** join/#asterisk Thesulac (~Thesulac@82.94.204.46) |
| 15:41.33 | Dovid | Wimpy: Perfect. thanks |
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| 15:55.42 | *** join/#asterisk eject_ck1 (~Eugene@95.67.72.22) |
| 16:01.37 | phix | WIMPy: can you call me? or arnt we on that level yet? |
| 16:06.29 | *** join/#asterisk xn|adept (~pi@rrcs-97-77-208-166.sw.biz.rr.com) |
| 16:09.54 | linuxasterisk | Could anyone want to help me with a basic set up question? I've never run into this problem before using the dedicated asterisk distros, but I wanted to use my existing box (ubuntu 12.04, asterisk 11.5.0, freepbx) to set up asterisk and used the source. I cannot register sip phones. Debug info: http://pastebin.com/Ez8uqs0M |
| 16:12.00 | Qwell | [2013-07-25 07:50:35] NOTICE[13111]: chan_sip.c:27919 handle_request_register: Registration from '"My Name"<sip:0001@192.168.1.10>' failed for '192.168.1.5:41018' - Wrong password |
| 16:12.08 | Qwell | The error is pretty clear... |
| 16:13.10 | *** join/#asterisk rogersja (~rogersja@24.244.23.50) |
| 16:13.13 | *** join/#asterisk threesome (~threesome@ip-94-113-12-233.net.upcbroadband.cz) |
| 16:18.35 | linuxasterisk | @Quell. Unfortunately it is not, because I have made quadrupally clear the secret and password matched. |
| 16:20.21 | linuxasterisk | thanks for trying |
| 16:21.16 | *** join/#asterisk peetaur2 (~peter@x2f04c0d.dyn.telefonica.de) |
| 16:24.40 | igcewieling | linuxasterisk: try a different sip client |
| 16:25.00 | igcewieling | does "sip show peer 0001" show what you expect? |
| 16:27.44 | *** join/#asterisk pgrace (pgrace@vsix.me) |
| 16:28.29 | pgrace | Hi, I'm trying to use features to dynamically pause/unpause mixmonitor. When I execute the feature, it shows that it detects the feature but it then immediately triggers h@ extension. There's no debug to tell me why. How do I go about solving this? |
| 16:30.00 | *** join/#asterisk cmendes0101 (~cmendes01@72.1.46.254) |
| 16:30.09 | *** join/#asterisk navaismo (~navaismo@189.191.233.140) |
| 16:30.47 | pgrace | setting debug 9 on features.c dropped nothing to the debug log. |
| 16:33.15 | linuxasterisk | lgceieling: samething with different client |
| 16:34.19 | linuxasterisk | sip show registry and sip show peers are empty. |
| 16:40.07 | [TK]D-Fender | linuxasterisk: Show us complete SIP debug from * CLI and your actual configs. |
| 16:40.39 | *** join/#asterisk imox (~imox@91-64-148-46-dynip.superkabel.de) |
| 16:42.17 | ulogic | to pgrace: try using "core set debug 9 features" without the ".c" extension |
| 16:42.28 | pgrace | ulogic: thanks! I'll try that now. |
| 16:43.05 | pgrace | ulogic: hmm, still no output to the debug log with that. |
| 16:44.56 | *** join/#asterisk zerick (~eocrospom@190.187.21.53) |
| 16:46.40 | igcewieling | I love routers. With Adtran AOS 10.x if you have multiple ntp server hostnamed defined and if any two hostname resolve to the same IP, the router will crash on boot. |
| 16:47.25 | igcewieling | linuxasterisk: sip show registery has nothing to do with this. |
| 16:47.35 | igcewieling | if sip show peers is empty then you do not have any peers configured |
| 16:49.03 | mic_ | is there any option to set a moh class when putting a call on hold? |
| 16:52.48 | igcewieling | like using the hold button to specify the music class using Morse Code? |
| 16:53.30 | igcewieling | you might be able to create an ugly hack to change the MoH class on a channel using features.conf but it seems like a very Rube Goldberg way to do it. |
| 16:53.32 | [TK]D-Fender | mic_: There is no signal from the phone for "class" |
| 16:53.38 | [TK]D-Fender | mic_: Hold is Hold |
| 16:55.39 | mic_ | [TK]D-Fender: ok. Because it plays by default the class 'default' |
| 16:56.14 | [TK]D-Fender | mic_: * plays whatever you set the channel up for |
| 16:56.28 | [TK]D-Fender | mic_: this was always up to you |
| 16:56.34 | [TK]D-Fender | mic_: it is not chosen "on demand" |
| 16:56.39 | *** join/#asterisk sawgood (~sawgood@unaffiliated/sawgood) |
| 16:57.13 | mic_ | ok |
| 16:59.40 | *** join/#asterisk tharkun (~0@unaffiliated/tharkun) |
| 17:00.21 | igcewieling | You set the music class before the Dial or in the peer or channel config. If you don't do that then the music class named 'default' will be used. |
| 17:00.49 | mic_ | igcewieling: that's what I have noticed now :) |
| 17:01.02 | igcewieling | Setting the MoH class is trivial and standard thing to do Setting it "on demand" is simply crazy talk. |
| 17:01.06 | tharkun | Where can I read a 101 on Asterisk? |
| 17:01.14 | igcewieling | tharkun: the book |
| 17:01.17 | igcewieling | ~book |
| 17:01.17 | infobot | Asterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 17:01.39 | igcewieling | Assuming you mean "101" to be "intro to Asterisk" |
| 17:02.17 | tharkun | igcewieling: Thanks |
| 17:03.02 | *** join/#asterisk jasonwert (~w3rt@75-134-81-98.static.aldl.mi.charter.com) |
| 17:03.19 | navaismo | priority 101? 101 books of asterisk |
| 17:03.35 | navaismo | lol on asterisk |
| 17:03.37 | navaismo | ¿?¿?¿?¿ |
| 17:05.03 | navaismo | OK now we can throw random numbers in a sentence and you can translate it igcewieling...oh im a fan of you |
| 17:05.05 | igcewieling | navaismo: sometimes you just have to guess at what they mean. |
| 17:05.25 | igcewieling | Not really my problem if I get it wrong when the question is confusing |
| 17:05.30 | mic_ | igcewieling: works like a charm. I did not want "on demand" - it was just that simple MusicOnHold(Class) before DIAL(). Thanks. |
| 17:06.01 | tharkun | igcewieling: I would apreciate a pdf for offline read since commuting has no connectivity but I will manage |
| 17:06.49 | igcewieling | mic_: you asked for "on demand" |
| 17:08.07 | *** join/#asterisk xn|adept (~pi@rrcs-97-77-208-166.sw.biz.rr.com) |
| 17:08.43 | [TK]D-Fender | mic_igcewieling: works like a charm. I did not want "on demand" - it was just that simple MusicOnHold(Class) before DIAL(). Thanks. <- and doing it in the dialplan is a last-ditch choice... normally you set this on your DEVICES |
| 17:10.19 | *** join/#asterisk [404] (~404]@12.179.117.114) |
| 17:10.56 | navaismo | really? i always set on dialplan ^ |
| 17:12.44 | *** join/#asterisk peetaur2 (~peter@x2f04c0d.dyn.telefonica.de) |
| 17:13.45 | igcewieling | navaismo: you say tomato, I say rutabaga. |
| 17:14.20 | igcewieling | waits for the response to his rant about our customer service people which he just sent to The Big Boss |
| 17:17.00 | ChannelZ | I say BULLSH** |
| 17:19.39 | navaismo | it |
| 17:30.21 | *** join/#asterisk nantou (~phonetic@gateway/tor-sasl/martinphone) |
| 17:33.03 | *** join/#asterisk mokmeister (~mokmeiste@86-40-247-195-dynamic.b-ras2.lmk.limerick.eircom.net) |
| 17:34.13 | *** join/#asterisk serafie (~erin@nat/digium/x-qfsmlnxcubmkofqt) |
| 17:36.57 | *** join/#asterisk peter (~peter@x2f04c0d.dyn.telefonica.de) |
| 17:53.48 | *** join/#asterisk serafie (~erin@nat/digium/x-aavcgqwfrhxbxrah) |
| 17:54.31 | *** join/#asterisk protocoldoug (~doug@unaffiliated/protocoldoug) |
| 18:18.26 | *** join/#asterisk ooggetuigen (~m@120-163-ftth.onsbrabantnet.nl) |
| 18:18.38 | ooggetuigen | hi all |
| 18:18.43 | ooggetuigen | i'm done with freepbx |
| 18:18.47 | ooggetuigen | i go install plain asterisks |
| 18:18.54 | ooggetuigen | i dont need http sql for it aight? |
| 18:19.00 | tm1000 | ooggetuigen: no |
| 18:19.09 | tm1000 | there is no gui |
| 18:19.14 | tm1000 | its all text files.... |
| 18:19.21 | ooggetuigen | thats great :) |
| 18:19.24 | tm1000 | (or sql if you wished) |
| 18:19.34 | ooggetuigen | you know what happend |
| 18:19.46 | tm1000 | ooggetuigen: no? but im sure you are going to tell me? |
| 18:19.49 | igcewieling | ooggetuigen: you want to read The Book. |
| 18:19.50 | ooggetuigen | i installed freepbx and sometimes i get weird internal calls from an local extension thats unknown |
| 18:19.52 | igcewieling | ~book |
| 18:19.52 | infobot | Asterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 18:20.22 | tm1000 | ooggetuigen: ok? |
| 18:20.24 | igcewieling | ooggetuigen: that issue is unlikely a FreePBX issue |
| 18:20.32 | tm1000 | freepbx makes a lot of internal calls. its how they do time management |
| 18:20.37 | tm1000 | igcewieling: +1 |
| 18:20.47 | ooggetuigen | yes |
| 18:20.52 | ooggetuigen | and thats why i'm done with it :) |
| 18:20.57 | tm1000 | igcewieling: except in the case of time management. where it will check blf through a cron |
| 18:21.06 | tm1000 | ooggetuigen: right. we dont really care about that |
| 18:21.14 | ooggetuigen | call forwarding doesnt work very well with it so |
| 18:21.16 | igcewieling | tm1000: I'd forgotten about that. |
| 18:21.31 | tm1000 | ooggetuigen: if you want to learn asterisk then come here. sure fine. but dont bash freepbx and no one cares that you left it here |
| 18:21.36 | *** join/#asterisk Busstech (~Busstech@41-133-18-37.dsl.mweb.co.za) |
| 18:21.49 | tm1000 | ooggetuigen: you were most likely doing it wrong |
| 18:21.56 | tm1000 | all of the issues you have presented are user error |
| 18:22.05 | tm1000 | but more power to you in using vanilla asterisk |
| 18:22.14 | ooggetuigen | yes |
| 18:22.17 | ooggetuigen | listen up |
| 18:22.24 | ooggetuigen | 10 years back |
| 18:22.26 | ooggetuigen | i installed suse |
| 18:22.31 | tm1000 | oh god |
| 18:22.31 | ooggetuigen | dvd 1 2,3,4,5,6 |
| 18:22.33 | ooggetuigen | system crashed |
| 18:22.34 | tm1000 | Im done |
| 18:22.39 | ooggetuigen | then i installed slackware, 1 cd |
| 18:22.40 | Busstech | Hi room, where can i view the log file contents of asterisk? our server just went down for no reason that is clear to me. After a linux reboot however it was fine |
| 18:22.41 | tm1000 | I dont need a brag-a-thon dude |
| 18:22.43 | ooggetuigen | it was running rock solid since |
| 18:22.52 | *** join/#asterisk k610 (~K610@cable-78.29.241.186.coditel.net) |
| 18:22.57 | tm1000 | Im sure you know linux/unix/ waayyy better than me |
| 18:23.11 | tm1000 | but this is asterisk |
| 18:23.12 | tm1000 | so please stop |
| 18:23.23 | tm1000 | Busstech: /var/log/asterisk/full |
| 18:23.26 | tm1000 | for asterisk |
| 18:23.31 | tm1000 | also semi-system dependent |
| 18:23.33 | Busstech | thanks! |
| 18:24.17 | ooggetuigen | @Busstech, you can also enable debugging on the cli |
| 18:24.35 | ooggetuigen | enter asterisk -r when it runs |
| 18:24.36 | ooggetuigen | and then |
| 18:24.42 | tm1000 | ooggetuigen: how would that help! |
| 18:24.46 | Busstech | thanks! what command does that? |
| 18:24.47 | tm1000 | he is asking about what cashed the crash! |
| 18:24.48 | ooggetuigen | sip set debug on |
| 18:24.52 | tm1000 | wtf |
| 18:24.58 | tm1000 | ooggetuigen: stop giving him useless commands |
| 18:25.00 | ooggetuigen | i know :) |
| 18:25.01 | tm1000 | dont turn on sip debug |
| 18:25.20 | tm1000 | Busstech: your crash issues will be in that file. as thats what you asked. dont turn on extra useless information |
| 18:25.28 | ooggetuigen | thats true tm |
| 18:25.33 | Busstech | ok thanks |
| 18:26.04 | Busstech | -bash: cd: var/log/asterisk/full: No such file or directory |
| 18:26.19 | tm1000 | <PROTECTED> |
| 18:26.31 | ooggetuigen | does asterisk run on freebsd too? |
| 18:26.38 | tm1000 | ooggetuigen: yes |
| 18:26.50 | ooggetuigen | stable? |
| 18:26.56 | tm1000 | ooggetuigen: I assume so |
| 18:27.01 | tm1000 | why dont you try |
| 18:27.25 | *** part/#asterisk igcewieling (~igcewieli@ip98-183-25-31.pn.at.cox.net) |
| 18:27.42 | ooggetuigen | well i have read the latest version run better on debian |
| 18:27.49 | ooggetuigen | but i use freebsd for my servers nowdays |
| 18:28.09 | Busstech | Can RAID 10 be a cause of instability? |
| 18:28.11 | *** join/#asterisk sevin (~tpritchet@c-50-132-126-144.hsd1.wa.comcast.net) |
| 18:28.13 | tm1000 | ooggetuigen: actually it would run better on centos |
| 18:28.28 | tm1000 | so I dont know where you read it would run better on debian |
| 18:28.33 | [TK]D-Fender | Busstech: [14:26]Busstech-bash: cd: var/log/asterisk/full: No such file or directory <- full is the file... not a folder |
| 18:28.36 | ooggetuigen | Busstech, check your diskspace with df |
| 18:28.46 | *** part/#asterisk tm1000 (tm1000@2600:3c01::f03c:91ff:fe70:fce6) |
| 18:28.52 | *** join/#asterisk tm1000 (tm1000@2600:3c01::f03c:91ff:fe70:fce6) |
| 18:29.03 | Busstech | df says all good |
| 18:29.13 | ooggetuigen | ok good to hear :) |
| 18:30.42 | file | raises eyebrow |
| 18:30.50 | tm1000 | file: you are missing nothing. get back to work |
| 18:31.09 | file | nothing to see here, move along? |
| 18:31.35 | *** join/#asterisk gerhard7 (~gerhard7@77-172-47-159.ip.telfort.nl) |
| 18:31.36 | ooggetuigen | lol |
| 18:32.03 | *** join/#asterisk dash_ (~d45h@unaffiliated/dash-/x-7576607) |
| 18:32.04 | ooggetuigen | @file download xbot & sent asterisk to users who request |
| 18:32.15 | file | blinks |
| 18:32.24 | tm1000 | file: im just as confused by this guy as you |
| 18:33.01 | ooggetuigen | @file !show_latest asterisk version |
| 18:33.15 | tm1000 | ooggetuigen: what are you talking about |
| 18:33.17 | *** join/#asterisk rgsteele (~rgsteele@12.150.6.65) |
| 18:33.43 | tm1000 | why would you need to know that. its in the topic of the channel |
| 18:35.03 | ooggetuigen | what was it / motd? |
| 18:35.17 | tm1000 | no. its the topic of the channel |
| 18:35.26 | tm1000 | 11.5.0 |
| 18:35.35 | tm1000 | #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 11.5.0 (2013/07/15), 10.12.2 (2013/03/27), 1.8.23.0 (2013/07/15), DAHDI-linux 2.6.2 (2013/03/08), DAHDI-tools 2.6.2 (2013/03/08), libpri 1.4.14 (2012/12/20) -=- Visit the official Asterisk wiki: wiki.asterisk.org |
| 18:35.47 | ooggetuigen | is 11.5.0 stable? |
| 18:35.52 | tm1000 | ooggetuigen: :-\ |
| 18:35.57 | tm1000 | you ask such confusing questions |
| 18:36.03 | tm1000 | no its completely beta. we put it there for laughs |
| 18:36.12 | ooggetuigen | lol |
| 18:36.14 | ooggetuigen | thx |
| 18:36.25 | ooggetuigen | now i laugh |
| 18:36.31 | tm1000 | ooggetuigen: you REALLY need to read the book |
| 18:36.35 | tm1000 | you are going to be SOOO lost |
| 18:37.17 | tm1000 | ooggetuigen: http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html/asterisk-book.html |
| 18:37.20 | ooggetuigen | whats the difference between asterisk and freepbx? The one who your asterisk masters asterisk the one who use freebsd does not :D |
| 18:37.38 | tm1000 | ooggetuigen: what? |
| 18:37.47 | ooggetuigen | *use |
| 18:37.54 | tm1000 | ooggetuigen: Im completely lost |
| 18:37.58 | ooggetuigen | i mean freepbx lol |
| 18:38.01 | tm1000 | ooggetuigen: whats the point of bashing freepbx |
| 18:38.04 | tm1000 | just cut it out |
| 18:38.11 | tm1000 | it amuses no one here |
| 18:38.34 | ooggetuigen | yes |
| 18:38.36 | ooggetuigen | im from netherlands |
| 18:38.37 | ooggetuigen | ;) |
| 18:38.57 | tm1000 | ooggetuigen: just because you hate or dont like freepbx is no reason to come in here and start bashing it. thats my point |
| 18:39.09 | ooggetuigen | hehe ok |
| 18:39.22 | ChannelZ | fanboy |
| 18:39.47 | [TK]D-Fender | That's right ... you should bash on it in #freepbx instead :) |
| 18:39.55 | ooggetuigen | hehe |
| 18:40.00 | tm1000 | ChannelZ: NO |
| 18:40.12 | ooggetuigen | how much server resources does it take when i have 10 users running on asterisk with voicemail |
| 18:40.13 | tm1000 | ChannelZ: its abusive to bash on other projects in unrelated channels |
| 18:40.29 | tm1000 | ooggetuigen: depends on the system at hand |
| 18:40.31 | [TK]D-Fender | ooggetuigen: My watch can do that .. and it's ANALOG |
| 18:40.32 | Gugge | ooggetuigen: 4% |
| 18:40.51 | ooggetuigen | hehe |
| 18:40.51 | tm1000 | ChannelZ: <--hater |
| 18:42.00 | file | hates on Asterisk |
| 18:42.11 | tm1000 | file: good. it makes terrible coffee |
| 18:43.00 | [TK]D-Fender | No, my Asterisk MAKES me coffe, and it's GOOD. You're clearly doing it wrong. Mine is also a jukebox. |
| 18:44.31 | ooggetuigen | lol it should work d-fender when u call a number the cofee device goes on hehe |
| 18:46.36 | [TK]D-Fender | correct |
| 18:49.44 | Katty | incorrect. |
| 18:49.45 | ChannelZ | I've tried to reconfig mine to make coffee but then it just keeps rewriting the config and spits out tea |
| 18:49.50 | Katty | mister fender is never correct. |
| 18:50.05 | Katty | in fact, it's always mister fender's fault. |
| 18:50.06 | ChannelZ | and it goes through like 67 steps just to do it. |
| 18:50.06 | Qwell | ChannelZ: because you didn't read the documentation on how your system is supposed to work |
| 18:50.14 | Katty | Qwell: AND YOU |
| 18:50.21 | Qwell | what did I do?! |
| 18:50.22 | Katty | Qwell: where to start with you... |
| 18:50.39 | Katty | Qwell: well you didn't give me a hug for starters! |
| 18:50.46 | Katty | Qwell: i've been LURKING all DAY |
| 18:50.47 | Qwell | lies and slander |
| 18:51.07 | Katty | hugs Qwell |
| 18:51.40 | Katty | Qwell: you wanna do xmas cards this year? |
| 18:51.52 | Katty | Qwell: 5 months and counting :> |
| 18:52.14 | Qwell | everyone is always so concerned about time |
| 18:52.27 | Katty | yes. especially quitting time and dinner time. |
| 18:52.30 | ChannelZ | Anythony Weiner is sending out xxxmas cards this year |
| 18:52.37 | Qwell | ChannelZ: try again |
| 18:52.45 | Katty | wonders who that is. |
| 18:52.55 | Katty | i feel like this i a name i should know, but it's escaping me |
| 18:53.18 | Qwell | The guy whos namesake was posted all over his twitter. |
| 18:53.23 | ChannelZ | A hack politician who has a propensity for showing his junk on Twitter |
| 18:53.26 | Katty | that...doesn't tell me much |
| 18:53.32 | Katty | oh, i see. |
| 18:53.32 | *** join/#asterisk davlefou (~davlefou@unaffiliated/davlefou) |
| 18:53.59 | *** join/#asterisk peetaur2 (~peter@x2f04c0d.dyn.telefonica.de) |
| 18:54.01 | Katty | actually i'd rather not see :/ |
| 18:54.24 | Katty | don't much care for politicians, much less their. uhh. ..tweets. |
| 18:54.44 | ChannelZ | this guy is lacking on all accounts |
| 18:54.53 | Katty | i don't need to know the details >.< really |
| 18:55.04 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 18:55.05 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 18:55.06 | leifmadsen | 8==============D |
| 18:55.21 | Katty | dagnabbit leif |
| 18:55.34 | Cuzner | yep, that's going on the quotes from the engine room page. |
| 18:55.57 | ChannelZ | at least it's smiling |
| 18:56.05 | Katty | facepalms |
| 18:56.12 | Katty | i disown all of you! except for Qwell |
| 18:56.29 | Katty | and file. |
| 18:56.48 | Katty | and carrar :> |
| 18:56.59 | Cuzner | here's another famous leif quote I added to the page earlier - Project already sold, so we're just making solutions to problems... which are problems to solutions... problutions I like to call them. |
| 18:57.00 | ChannelZ | Good. I ain't nobody's propertaaay |
| 18:57.15 | ooggetuigen | installing debian at the moment :) |
| 18:57.19 | Katty | ooggetuigen: hot. |
| 18:57.20 | leifmadsen | Cuzner: :D |
| 18:57.22 | ooggetuigen | without sql and apache crap |
| 18:57.26 | Cuzner | i now use "problutions" all the time |
| 18:57.27 | leifmadsen | Cuzner: I haz all the answers |
| 18:57.35 | Katty | leifmadsen: i haz all the questions |
| 18:57.41 | Katty | eyebrows at leifmadsen |
| 18:57.46 | leifmadsen | Cuzner: that was actually my first use of that term too :D |
| 18:57.53 | Cuzner | excellent |
| 18:57.59 | leifmadsen | I enjoy it |
| 18:58.20 | Cuzner | i like Tom's quote - We put the "NO" in InNOvation. |
| 18:58.21 | ooggetuigen | he loves it |
| 18:58.45 | leifmadsen | Katty: you put that eyebrow down RIGHT NOW |
| 18:59.32 | Katty | leifmadsen: gah, next you'll be telling me not use your iphone as a chew toy! |
| 18:59.46 | leifmadsen | Katty: I don't know any apple products :) |
| 18:59.52 | leifmadsen | s/know/own/ |
| 18:59.59 | Katty | leifmadsen: well i can introduct you some |
| 19:00.04 | leifmadsen | I'm good thanks |
| 19:00.23 | Katty | <3 |
| 19:00.28 | Qwell | leifmadsen: but this model is 0.021" thinner |
| 19:00.32 | Cuzner | the only problem with an iphone is iTunes, and unfortunately an iphone is pretty useless without that piece of shit software. |
| 19:00.39 | Katty | Qwell: that's what.... nevermind |
| 19:00.39 | file | o.o |
| 19:00.49 | Katty | HI FILE |
| 19:00.55 | file | hihi |
| 19:00.58 | Katty | someone here used to work for apple |
| 19:01.07 | Qwell | file: I CAN SEE YOU |
| 19:01.17 | file | through the walls? |
| 19:01.22 | Qwell | MAYBE |
| 19:01.40 | *** join/#asterisk fakhir (~fakhir@unaffiliated/fakhir) |
| 19:01.45 | ChannelZ | http://lolsnaps.com/news/35291/ |
| 19:02.29 | Katty | he has those new "xray vision" glasses developed by the MIT engineers last month |
| 19:02.58 | Katty | i guess they're not exactly glasses tho |
| 19:03.08 | Katty | it emits wifi signals through a wayll |
| 19:03.20 | *** join/#asterisk igcewieling (~igcewieli@ip98-183-25-31.pn.at.cox.net) |
| 19:03.26 | ChannelZ | so it's more like lasers |
| 19:03.29 | Katty | it reflects off things, and goes back to the device. |
| 19:03.32 | Katty | YES. |
| 19:03.33 | Katty | wifi lasers. |
| 19:03.35 | Katty | pewpew? |
| 19:03.41 | Katty | i see you! |
| 19:03.55 | ChannelZ | so you can beam the cancer directly to your brain |
| 19:04.16 | *** join/#asterisk reconwireless (uid10170@gateway/web/irccloud.com/x-cgnrxwoyjlnktejy) |
| 19:05.34 | Katty | i think they're working with naked mole rats to prevent cancer now (true story) |
| 19:07.25 | Katty | REF http://www.nytimes.com/2013/06/19/science/a-homely-rodent-may-hold-cancer-fighting-clues.html?_r=0 |
| 19:07.46 | Katty | isn't science great?! |
| 19:07.49 | igcewieling | Tech: "I'm at the customer and there is no dialtone at the jack they want to plug the Polycom SoundPoint IP 335 into" |
| 19:08.35 | Katty | ..wait, what? |
| 19:08.44 | igcewieling | calling a naked mole rat "homely" is like calling Genghis Kahn "anti-social" |
| 19:08.48 | igcewieling | Katty: EXACTLY! |
| 19:08.48 | ChannelZ | "and the plug is the wrong size" |
| 19:09.02 | Katty | igcewieling: heh someone here was trying to test a fax port with a polycom phone |
| 19:09.10 | Katty | igcewieling: i didn't say anything |
| 19:09.25 | Katty | igcewieling: i just shook my head and kept on walking while they tried to desperately figure out how it fit into the port |
| 19:09.46 | *** join/#asterisk cstk421 (~cstk421@99-8-81-227.lightspeed.livnmi.sbcglobal.net) |
| 19:10.08 | igcewieling | Katty: sometimes I wish they would allow me to use stun guns on customers. |
| 19:10.56 | Katty | igcewieling: and coworkers?! |
| 19:11.03 | cstk421 | could a fax communication be ended prematurely due to this log comment ? "Restricting registration for peer '1101' to 60 seconds (requested 300)" |
| 19:11.19 | WIMPy | cow-orkers |
| 19:11.31 | Qwell | cstk421: registration has nothing to do with calls, no |
| 19:11.33 | WIMPy | cstk421: no |
| 19:11.40 | cstk421 | darn |
| 19:12.06 | igcewieling | Katty: just the sales department |
| 19:12.29 | mic_ | (I second that) |
| 19:12.48 | cstk421 | i see the fax ansering the call and status changes to "recieving facsimile" and then hangs up and goes idle. No failure report nothing. Log doesnt seem to show anything wierd. Other calls no problem comes through ant transmits fine. |
| 19:12.51 | mic_ | (or at least to mount some electric-shock leashes on them) |
| 19:15.48 | ooggetuigen | *downloading asterisk current 11 |
| 19:19.12 | *** join/#asterisk kaldemar (~kaldemar@unaffiliated/kaldemar) |
| 19:19.12 | *** join/#asterisk Dovid (~Dovid@static-173-63-105-210.nwrknj.fios.verizon.net) |
| 19:19.12 | *** join/#asterisk linuxasterisk (~Steve@ool-435632fe.dyn.optonline.net) |
| 19:19.12 | *** join/#asterisk slav3_kitten (~frankthet@unaffiliated/slav3-kitten/x-0866809) |
| 19:19.12 | *** join/#asterisk dijib (~root@24-231-78-197.eastlink.ca) |
| 19:19.12 | *** join/#asterisk Katty (~Katty@mail.copi-rite.com) |
| 19:19.12 | *** join/#asterisk newtonr (~newtonr@64.34.219.47) |
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| 19:34.19 | igcewieling | cstk421: pastebin the CLI output of a failed call. |
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| 19:54.02 | igcewieling | I wish Chrome didn't suck so much |
| 19:54.40 | ChannelZ | I've never liked it |
| 19:55.03 | igcewieling | ChannelZ: when I standardized on it, Firefox was much slower than Chrome |
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| 19:56.43 | [sr] | WIMPy: are you there? have a master question :p |
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| 19:57.19 | WIMPy | [sr]: You know you should just ask. |
| 19:58.16 | [sr] | just trying to be funny |
| 19:58.40 | [sr] | i have a place where i have a PCI FXS card, and analog phones sync the date+time with asterisk |
| 19:59.07 | WIMPy | didn;t knw such a thing existed. |
| 19:59.20 | [sr] | but with a TA, it doesn't happen, is this a feature? if yes do you know the name so that i can search for it on the TA ? |
| 19:59.26 | ChannelZ | I think time is transmitted in the CID? |
| 19:59.48 | [sr] | ChannelZ: i really have no idea, but it syncronized |
| 20:00.04 | [sr] | in the PCI FXS card, without any extra configuration |
| 20:00.21 | WIMPy | I have no idea. when I last used something Analog, Caller ID wasn;t invented there, yet. |
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| 20:01.04 | igcewieling | WIMPy: it doesn't exist. However, most phones will set their date/time from the date/timestamp from the callerid of the first incoming call |
| 20:01.10 | Cuzner | wow |
| 20:01.45 | [sr] | igcewieling: so its a phone feature, not the PCI CARD or TA? |
| 20:02.03 | igcewieling | [sr]: correct. |
| 20:03.31 | [sr] | hum, for some reason didn't happen this time, phones are the same |
| 20:13.17 | navaismo | pabelanger, your stripe development its a way to replace AMI with queue commands? I've only checked the API reference do you have a "background" page? |
| 20:16.50 | pabelanger | navaismo, Its actually a standalone ACD for to use along side asterisk |
| 20:18.07 | pabelanger | going to be doing a presentation at astricon about it too |
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| 20:24.38 | navaismo | :'( |
| 20:24.59 | navaismo | can't assist to Astricon. |
| 20:34.57 | pabelanger | navaismo, plan to have an alpha release for sept 1st, want to get people testing it out before the astricon demo. |
| 20:35.31 | navaismo | ok we will see, thanks a lot |
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| 20:44.14 | *** join/#asterisk Mon|A|rch (~sbean@72.29.180.35) |
| 20:44.44 | Mon|A|rch | I was wondering if I could get an opinion. I've got a system that auto dials out about 800 times a day using call files |
| 20:44.58 | Mon|A|rch | i was wondering, if I want to bump that up to around 4000 calls, do I need to switch to AMI or something? |
| 20:45.07 | Mon|A|rch | it seems like call files start to choke up when you use them too much |
| 20:45.55 | WIMPy | AMI is always the clever solution. |
| 20:46.17 | Mon|A|rch | I do like it a lot |
| 20:46.29 | Mon|A|rch | it has an http interface doesn't it? |
| 20:47.00 | WIMPy | There is something. Not sure it's usefull for anything. |
| 20:47.25 | Mon|A|rch | well, we have a web interface based on another server |
| 20:47.33 | Mon|A|rch | seemed like a decent way to originate calls |
| 20:47.39 | Mon|A|rch | just send an http request to the server |
| 20:47.52 | Mon|A|rch | get some http responses via CURL for call data |
| 20:48.03 | WIMPy | I think you get what you want in Asterisk 12. |
| 20:48.16 | Mon|A|rch | only in 1.8 now |
| 20:48.21 | Mon|A|rch | should probably upgrade |
| 20:50.03 | Mon|A|rch | any idea how scary upgrading would be in my situation? |
| 20:50.14 | ChannelZ | Spooooooky! |
| 20:50.14 | WIMPy | Probably not. |
| 20:50.16 | Mon|A|rch | will it break half the dialplan? |
| 20:50.34 | Mon|A|rch | it seems like after 1.6 the dialplan stabilized |
| 20:50.36 | ChannelZ | Not unless you're still using | as a delimeter and priority jumping :P |
| 20:50.42 | Mon|A|rch | lol |
| 20:50.53 | WIMPy | Unless you still use some pre 1.4 compatibility stuff, it should be pretty straightforward. |
| 20:50.54 | Mon|A|rch | well, no pipes |
| 20:51.05 | Mon|A|rch | what do you mean by priority jumping? |
| 20:51.31 | WIMPy | Let applications go to the current priority +100 on error. |
| 20:51.32 | ChannelZ | the old thing where it would jump to n+100 (101?) on busy, etc. |
| 20:51.40 | Mon|A|rch | oh |
| 20:51.41 | Mon|A|rch | ew |
| 20:51.44 | ChannelZ | ancient |
| 20:51.55 | Mon|A|rch | would never do that |
| 20:52.00 | WIMPy | From the darf ageg. |
| 20:52.05 | WIMPy | dark |
| 20:52.12 | Mon|A|rch | well, thanks guys |
| 20:52.18 | ChannelZ | A couple of directives might have changed names in sip.conf, I can't recall.. just read through the UPGRADE docs and make sure you're not using something that might have changed. |
| 20:52.21 | Mon|A|rch | helpful as always |
| 20:52.34 | Mon|A|rch | my sip.conf is pretty basic |
| 20:52.55 | ChannelZ | canreinvite changed but I don't remember when.. |
| 20:54.15 | ChannelZ | or I'm thinking of nat. |
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| 20:56.02 | Mon|A|rch | anyway, WIMPy, 12 has some sort of http AMI? |
| 20:57.34 | WIMPy | I believe that's what it is, yes. I didn't care too much about that part. |
| 20:57.47 | file | ARI is separate from AMI |
| 21:01.36 | ChannelZ | I love a good new acronym. What's ARI? |
| 21:01.51 | file | Asterisk REST Interface |
| 21:01.58 | ChannelZ | Alpaca Registry Inc? |
| 21:02.29 | igcewieling | Asterisk Recordings Interface |
| 21:02.31 | ChannelZ | ahh-so. |
| 21:05.26 | navaismo | Ari sounds like freepbx stuff |
| 21:05.51 | Maliuta | no, no, no ... ARI is that Israeli guy who does weird shite ;) |
| 21:05.53 | ChannelZ | that'd be the Anus Rape Interface |
| 21:06.11 | Mon|A|rch | well, before the ARI (?) ever existed, the AMI could receive http requests for any command |
| 21:07.17 | Maliuta | Mon|A|rch: that's the Israeli fecking with you :P |
| 21:12.08 | *** join/#asterisk rrittgarn (~rrittgarn@75-150-221-205-Illinois.hfc.comcastbusiness.net) |
| 21:13.06 | rrittgarn | anybody have fail2ban working with asterisk? currently fighting a regex that works outside of fail2ban-regex but fails inside... |
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| 21:14.01 | Maliuta | rrittgarn: no, it's something I have been working. That and an effective regex for postfix/cluebringer |
| 21:14.43 | rrittgarn | thus far i've been beating my head against the desk. Looking to build this regex into Fail2Ban: (NOTICE)..*?.(chan_sip)..*?.((?:(?:25[0-5]|2[0-4][0-9]|[01]?[0-9][0-9]?)\.){3}(?:25[0-5]|2[0-4][0-9]|[01]?[0-9][0-9]?))(?![\d])..*?.('\d+')..*?.( rejected because extension not found in context )..*?.('dead-end') |
| 21:15.08 | rrittgarn | where the IP match gets replaced with <HOST> thus stopping the most common 'attack' against my boxes |
| 21:15.56 | ChannelZ | I use NOTICE.* chan_sip\.c.* Call from '.*' \(<HOST>:[0-9]+\) to extension '.*' rejected .* extension not found |
| 21:16.15 | Maliuta | I'd look at cutting chunks out of that and focusing on the most common occurring msg in the logs. |
| 21:16.25 | Maliuta | ChannelZ: I might just nick that one |
| 21:16.35 | ChannelZ | please do |
| 21:17.33 | rrittgarn | thats pretty close to what i was going for... only reason for doing the 'dead-end' part is if somehow I have a user thats dialing to something i don't have setup (missing an exten=>i) they wouldnt get banned... but that looks quite useful |
| 21:18.06 | Maliuta | ChannelZ: now if you could just sort out the regex I need for my smtpd logs ... ;) |
| 21:18.44 | Maliuta | rrittgarn: servers them right for not using the system properly ;) |
| 21:19.21 | Maliuta | rrittgarn: and it you're using IPSET for the jails it's about a 30 char fix to unban them |
| 21:20.29 | Maliuta | s/it/if/ |
| 21:20.33 | ChannelZ | I use it for my qmail smtpd logs |
| 21:20.47 | petn-randall | I don't know for sure, but couldn't you abuse that regex to DoS someone? After all, SIP is most of the time UDP. |
| 21:21.09 | Maliuta | Champi: qmail? people still use that hunk'o'junk |
| 21:21.31 | petn-randall | Is a handshake needed with difficult to guess values to trigger a login failure? If not, you can easily stop people from making SIP calls. |
| 21:21.37 | Maliuta | ChannelZ: I didn't even think it was still under dev/support. |
| 21:21.38 | ChannelZ | It's been running my mail for a very long time perfectly fine |
| 21:22.01 | Maliuta | ChannelZ: I am also biting my tongue about DJB |
| 21:22.11 | ChannelZ | he's a pompous jerk |
| 21:22.23 | ChannelZ | but his mailer is decent. |
| 21:22.38 | Maliuta | I was more thinking of golf balls and hoses ... and some references to goats |
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| 21:23.10 | Maliuta | ChannelZ: I never liked it's config ... I was a deep sendmail geek before going over to postfix |
| 21:23.49 | ChannelZ | well that says a lot.. you're averse to configuration that makes sense, coming from sendmail |
| 21:24.55 | Maliuta | ahh, I never used m4. I used to hand edit the deep files |
| 21:25.12 | Maliuta | and postfix has more conf options than you can blink at |
| 21:25.27 | ChannelZ | sendmail is what drove me to qmail 15 years ago or whenever it was (I don't even remember) |
| 21:25.57 | ChannelZ | I will at some point switch to postfix but it's an "if it ain't broke" situation |
| 21:26.15 | rrittgarn | ChannelZ - Thank you... with a little modification, that did exactly what i wanted it to do. |
| 21:27.12 | Maliuta | ChannelZ: to call me "adverse to configuration" is like calling a member of 22 SAS a soft touch |
| 21:27.14 | ChannelZ | hurray! |
| 21:27.25 | Maliuta | ChannelZ: I am a conf/control freak |
| 21:28.03 | ChannelZ | I said averse to configuration that "makes sense" |
| 21:28.12 | ChannelZ | Running sendmail is like programming assembler |
| 21:28.31 | Maliuta | I liek that one |
| 21:28.44 | Maliuta | s/liek/like/ |
| 21:28.57 | ChannelZ | (and security wise about as dangerous back in the day) |
| 21:29.43 | Maliuta | I always thought the m4 confing route was to complicated. I just liked to find the file that held the thing I wanted and bend it to my will manually |
| 21:30.08 | Maliuta | yeah, security was a big problem. Although none of my systems got breached |
| 21:30.34 | Maliuta | The one MTA I don't understand people using is Exim |
| 21:34.16 | ChannelZ | It seems to be the default for some distros but I don't know anything about it |
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| 22:28.44 | cstk421 | how would one go about removing elastix and using hylafax on its own on a elastix / asterisk distro ? |
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| 22:39.59 | navaismo | ~elastix |
| 22:39.59 | infobot | extra, extra, read all about it, elastix is 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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