| 00:03.46 | ChannelZ | in |
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| 01:31.50 | eyesec | Has anyone used the Chinese FXO/FXS cards from ebay in their Asterisk boxes?? |
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| 01:33.47 | [TK]D-Fender | "the chinese" precludes there being a multitude of actual manufacturers as though it was a singular state-run plant |
| 01:34.44 | ChannelZ | The clones from china send all your calls to the chineese government |
| 01:37.53 | eyesec | ohh right though to APT1? haha |
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| 01:38.30 | eyesec | Can you recommend a good manufacturer? Sorry I'm a real Asterisk noob |
| 01:39.19 | eyesec | Sorry they seem to look the same as if they're coming from one factory |
| 01:39.51 | [TK]D-Fender | What are your actual needs? |
| 01:39.54 | MaliutaLap | shudders |
| 01:39.59 | ChannelZ | They probably are |
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| 01:40.13 | MaliutaLap | I shelled for a Digium TDM400 when I started with * |
| 01:40.23 | eyesec | 4 FXO ports |
| 01:40.32 | eyesec | for 4 POTS lines |
| 01:40.42 | eyesec | Expensive? |
| 01:40.46 | MaliutaLap | 1 FXS, 1 FXO. It's now got two of both and I only need 1 FXS :) |
| 01:40.57 | MaliutaLap | ish, but at the time I had the money |
| 01:41.23 | MaliutaLap | I was still surprised to see second hand module going for like $100 on ebay |
| 01:41.38 | MaliutaLap | makes the card look like $600 or something |
| 01:41.48 | eyesec | There are just so many copies it's hard to know which is which |
| 01:41.54 | [TK]D-Fender | http://www.telephonydepot.com/Catalog/Sangoma/B600D-Analog-Voice-Card |
| 01:41.56 | eyesec | ebay.com/itm/tdm400p-asterisk-card-fxo-card-tdm400-for-voip-ip-pbx-/150517307956 |
| 01:42.27 | [TK]D-Fender | Or http://www.telephonydepot.com/Catalog/Digium-TDM410P |
| 01:42.28 | eyesec | I'm basically just worried about getting it working and then the echo issues |
| 01:42.38 | MaliutaLap | if you found a card that took the digium modules I could do you a decent deal on the two fxo units |
| 01:42.51 | ChannelZ | I have a TDM800, 7 years old, software EC... not had a problem |
| 01:43.05 | [TK]D-Fender | eyesec: Correct. Get one of the el-cheapo's and you're playing Russian Roulette for service on a Chinese knock-off. Oh the irony |
| 01:43.13 | MaliutaLap | like $60 inc postage ($10 will get you express post anywhere in .au) |
| 01:46.34 | eyesec | Thanks for your help |
| 01:47.00 | [TK]D-Fender | eyesec: Aussie? |
| 01:47.09 | eyesec | Yerp! |
| 01:47.38 | [TK]D-Fender | eyesec: Then if you care last I heard your telco's were nasty as far as what they'll allow you to use on their lines (certifications-wise) |
| 01:47.57 | [TK]D-Fender | Telstra IIRC was one of the worst |
| 01:48.02 | eyesec | It's actually not for here |
| 01:48.24 | eyesec | I'm setting it up in Asia.. where nobody gives a F**K |
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| 01:48.40 | *** mode/#asterisk [+o Qwell] by ChanServ |
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| 01:49.05 | eyesec | that's interesting about telstra though! |
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| 01:50.02 | [TK]D-Fender | This is somewhat dated second-hand info so I would go verify exactly what the state of affairs is. I know AU CID & CDS is tricky.... |
| 01:50.08 | [TK]D-Fender | ~cds |
| 01:50.08 | infobot | [~cds] Call Discconect Supervision is a service placed on analog lines to be able to signal you that that the calling party has hung up. This is typically done either by a momentary battery cut, or by a polarity reversal on the line. |
| 01:50.38 | eyesec | ohh okay |
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| 03:00.35 | drmessano | weee |
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| 08:01.41 | WIMPy | It smells like someonw weed in te channel. |
| 08:02.38 | ChannelZ | That might have been me. I was cold. |
| 08:06.34 | mirela666 | is stonned |
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| 10:35.38 | whizzi | I have an odd situation with Asterisk 11.6.1 and incoming SIP. It seems to me Asterisk is random choosing a context where to deliver when having multiple logins on the same endpoint |
| 10:39.15 | whizzi | http://pastebin.com/Qu0LSJae is a part of my sip.conf |
| 10:41.06 | whizzi | In 80% of the cases, a called number (which should end in cust2 for example) ends in the first context, and being send to the context incoming (where it will actually will end in this customer) |
| 10:41.48 | whizzi | in the other 20%, an incoming phone call for cust1 ends in the context of cust2 |
| 10:41.53 | *** part/#asterisk LiuYan (~LiuYan@211.154.128.133) |
| 10:42.33 | whizzi | just immediately it goes to cust2-inkomend while it should be ending in cust1-inkomend (preferred) or even incoming (where it will be switched anyway) |
| 10:42.45 | whizzi | does anyone have an explanation for this? |
| 10:43.24 | whizzi | the current solution is not really the best, sending all [customers] to context=incoming |
| 10:43.40 | whizzi | which allows customers to steal each others nr |
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| 12:02.51 | Runedhand | Hello people. |
| 12:03.01 | eyesec | hello! |
| 12:03.13 | Runedhand | Wow that was fast. I'm not used to an instant response! |
| 12:03.27 | Chainsaw | We have efficient greeters. |
| 12:03.32 | Chainsaw | They're expensive, but it's worth it. |
| 12:03.35 | Runedhand | Hahahaha |
| 12:03.38 | Runedhand | Indeed. |
| 12:03.40 | eyesec | Haha I just happened to be looking through the previous chat |
| 12:03.58 | eyesec | we sure are! |
| 12:04.07 | Runedhand | I have a question. |
| 12:04.27 | Chainsaw | Runedhand: Please proceed. |
| 12:05.12 | Runedhand | I'm using Asterisk 1.8, I'm connecting to AMI and I can't remember (or find any reference to) the event that's fired when a caller leaves the queue after hanging up. |
| 12:06.50 | Runedhand | I guess it started with "Queue" and I just can't find anything. |
| 12:06.51 | Runedhand | Oh wow. |
| 12:07.12 | *** join/#asterisk gringo (~gringo@unaffiliated/gringo) |
| 12:07.27 | eyesec | Sorry, I'm a newbie |
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| 12:07.41 | Runedhand | No worries, so am I. |
| 12:07.50 | Chainsaw | Runedhand: Does "manager show events" have it? |
| 12:08.03 | Runedhand | Hang on, let me see. |
| 12:09.06 | Runedhand | >OriginateResponse ParkedCallGiveUp ParkedCallTimeOut |
| 12:09.54 | Runedhand | Could be ParketCallGiveUp? |
| 12:10.08 | *** join/#asterisk luckman212 (~luckman21@unaffiliated/luckman212) |
| 12:10.18 | Runedhand | No, it can't. No reference to queues. |
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| 12:21.11 | whizzi | Runedhand: I may be wrong here, but my guess is that it just ends in a exten => h,1, |
| 12:22.19 | whizzi | that's how I see it here in my configuration. It sends to Queue and the next line is in my case going to a Macro(hangup-call) |
| 12:26.32 | Runedhand | Well, I'm not sure. I just need the event's name so I can simply hook it up to a handler. |
| 12:26.33 | Runedhand | :/ |
| 12:30.33 | *** join/#asterisk eduardonunesp (~eduardonu@187.103.104.42) |
| 12:30.47 | eduardonunesp | hi everybody |
| 12:31.25 | Runedhand | Hello there. |
| 12:31.42 | eduardonunesp | someone have experienced a consumption of 10gb with asterisk ? |
| 12:31.58 | Runedhand | Definitely not me. |
| 12:32.07 | eduardonunesp | i think that is too much |
| 12:32.17 | eduardonunesp | seems like a leak |
| 12:32.27 | whizzi | unless you record all your phone calls ;) |
| 12:32.44 | eduardonunesp | i'm talking about ram memory |
| 12:33.05 | Runedhand | Sounds like a waterfall than a leak, imho. |
| 12:33.17 | whizzi | that's a flood even |
| 12:33.22 | whizzi | 10GB is extremely much |
| 12:33.25 | whizzi | memory |
| 12:34.08 | whizzi | I use 240MB memory here |
| 12:34.12 | eduardonunesp | yeah i fear that |
| 12:34.27 | whizzi | 962 sip peers [Monitored: 778 online, 183 offline Unmonitored: 1 online, 0 offline] |
| 12:34.55 | Runedhand | Wow, that's a lot of numbers there, whizzi. |
| 12:35.08 | tuxx- | sounds like a tsunami! |
| 12:35.09 | whizzi | that includes queues, recording of conversations, CDR processing etc etc |
| 12:35.14 | eduardonunesp | wow |
| 12:35.35 | eduardonunesp | i'm using asterisk 1.4.44 |
| 12:36.48 | whizzi | eduardonunesp: you should consider upgrading to Asterisk 1.8 or asterisk 11 |
| 12:37.23 | eduardonunesp | whizzi, yeah i now that, but that is a operation very complicated |
| 12:37.39 | whizzi | eduardonunesp: I'm aware of it, I've did it a few times ;) |
| 12:37.49 | eduardonunesp | i'm just inherits that server from another guy |
| 12:37.52 | whizzi | so you know, Asterisk 1.8 -> 11 is fairly easy |
| 12:38.17 | Runedhand | Wait, hang on. I'm not on 1.8 |
| 12:38.23 | Runedhand | I'm on 11 or something. |
| 12:38.34 | whizzi | 1.4->1.8 is really a tough one, especially if you use deprecated commands |
| 12:38.39 | Runedhand | 11.3.0 |
| 12:39.09 | eduardonunesp | whizzi, yeah it's really tough |
| 12:39.11 | whizzi | I'm running this high-usage system on 11.6.1 at the moment |
| 12:39.29 | whizzi | probably update it to 11.7 soon |
| 12:39.58 | Runedhand | So, is there a place I can see the whole list of events? |
| 12:40.20 | whizzi | if only somebody could explain why my incoming calls seems to end up on random contexts |
| 12:40.51 | whizzi | Runedhand: http://downloads.asterisk.org/pub/telephony/asterisk/releases/asterisk-11.7.0-summary.html |
| 12:41.11 | whizzi | or even http://downloads.asterisk.org/pub/telephony/asterisk/releases/ |
| 12:41.33 | eduardonunesp | well thanks i will consider upgrade that server soon |
| 12:41.41 | Runedhand | Yeah, but these are changelogs. <_< I need AMI Events. |
| 12:41.56 | eduardonunesp | but i will check all deprecates first |
| 12:42.00 | Runedhand | You know, ones like "QueueMemberAdded". |
| 12:42.20 | whizzi | ah, sorry.. my bad |
| 12:42.40 | Runedhand | voip-info.org doesn't really help. |
| 12:42.55 | eduardonunesp | there is a huge list in voip-info |
| 12:42.56 | whizzi | http://www.voip-info.org/wiki/view/asterisk+manager+events ? |
| 12:43.11 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.127.162) |
| 12:43.23 | Runedhand | Yeah, that doesn't have what I'm looking for. |
| 12:43.28 | coreyf1513 | Runedhand: https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+AMI+Events |
| 12:43.52 | eduardonunesp | you did ! is that |
| 12:43.54 | Runedhand | QueueCallerAbandon. Thanks a bunch coreyf1513. |
| 12:44.10 | whizzi | I wonder who's in charge of voip-info since it seems to have stopped updating since 1.8.x |
| 12:44.25 | Runedhand | whizzi, I've got no idea, but it's a shitty site anyway. |
| 12:44.29 | Runedhand | Bad layout. |
| 12:44.38 | whizzi | it's just very old |
| 12:44.46 | whizzi | it used to be a good informative site |
| 12:44.59 | whizzi | but nowadays it's better to look on wiki.asterisk.org anyway |
| 12:45.03 | Runedhand | It's stuck between the awesomeness of '90s and the modernness of '10s. |
| 12:45.17 | whizzi | :D something like that |
| 12:45.19 | Runedhand | No flame gifs, but no jquery either. |
| 12:45.33 | whizzi | at least I can still visit it with my 1995 browser |
| 12:45.51 | Runedhand | I visit it once every month. |
| 12:47.22 | whizzi | kinda odd of voip-info that they have a massive banner of 3CX promoting a Easy to Install&Manager Software PBX for windows |
| 12:48.29 | eduardonunesp | it's a tsunami of banners |
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| 13:12.48 | Chainsaw | whizzi: It's probably working very well for them. "...none of these examples work, stupid Asteris... ah, 3CX you say?" |
| 13:14.12 | whizzi | lol, probably :P |
| 13:16.30 | *** join/#asterisk luckman212 (~luckman21@unaffiliated/luckman212) |
| 13:22.03 | *** join/#asterisk Weezey (~ohno@i.am.weezey.com) |
| 13:36.16 | *** join/#asterisk vk4akp (~vk4akp@c114-77-243-39.ipswc3.qld.optusnet.com.au) |
| 13:36.53 | vk4akp | Greetings Oh Great Asterisk Guru's! |
| 13:37.59 | *** join/#asterisk spengler1 (~spengler@pool-74-106-240-211.bltmmd.fios.verizon.net) |
| 13:38.03 | vk4akp | I have a question that begs your attention. |
| 13:38.58 | spengler1 | i have 4 pots lines ; is there any logic that will allow asterisk check line availability and dial out on the first available line? |
| 13:40.18 | vk4akp | I used to use VoXaLoT a few years ago. But now it is gone. :( Is there a way to still get an inbound VoIP connection to Asterisk without VoXaLoT? And also still support Sipbroker codes? |
| 13:42.46 | WIMPy | spengler1: Put them in to a grou and dial out the group. |
| 13:46.08 | eduardonunesp | hi |
| 13:46.25 | eduardonunesp | i'm just look at htop and we have many agi calls |
| 13:46.42 | Chainsaw | spengler1: And if those lines are analog, make sure they are set up for disconnect supervision. |
| 13:47.14 | spengler1 | yes they are analog |
| 13:47.16 | eduardonunesp | whizzi, |
| 13:47.30 | eduardonunesp | Runedhand, whizzi , ping |
| 13:47.38 | whizzi | whut? |
| 13:47.42 | spengler1 | when you say disconnect supervision do you mean busy count? |
| 13:47.59 | eduardonunesp | i'm just look at htop and we have many agi calls |
| 13:48.21 | WIMPy | spengler1: No. a way for yor interface to detect that a call has ended. |
| 13:48.37 | eduardonunesp | each consumes an average of 220M |
| 13:48.42 | WIMPy | Like e.g. polarity reversal. |
| 13:49.00 | Runedhand | eduardonunesp, I won't be much of help, I just use AMI to gather info for the front end. :/ |
| 13:49.01 | eduardonunesp | whizzi, so i have counted at least 40 agis calls |
| 13:49.08 | spengler1 | WIMPy ; i did configure it for busydisconnect=yes |
| 13:49.33 | whizzi | eduardonunesp: what kind of agi-calls are they? agi-mysql? agi-php? |
| 13:49.34 | WIMPy | spengler1: That's said to be dangerous. |
| 13:49.43 | eduardonunesp | Runedhand, thanks, we are using AGI with PHP to make many things |
| 13:49.49 | eduardonunesp | whizzi, agi-php |
| 13:50.08 | Runedhand | Maybe you should check your php scripts, then. |
| 13:50.18 | spengler1 | WIMPy ; yes i was reading that. i coupled it with busycount=3 to avoid premature disconnects |
| 13:50.31 | whizzi | eduardonunesp: What Runedhand said |
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| 13:50.41 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 13:50.53 | whizzi | probably your PHP is waiting for input or just hangs somehow |
| 13:50.55 | Runedhand | Maybe you forgot to call $asterisk->disconnect(); |
| 13:51.07 | WIMPy | spengler1: Check with your telco what they can offer. |
| 13:51.09 | eduardonunesp | $asterisk->disconnect ? |
| 13:51.12 | eduardonunesp | for each agi call |
| 13:51.17 | spengler1 | WIMPy ; i installed the system today ; everything seems to be working great so far ; since telephony is so important to this company i chose POTs for the reliability |
| 13:51.20 | whizzi | Runedhand: It should not use that much memory then ;) |
| 13:51.30 | Runedhand | Once you're done, you should disconnect. |
| 13:51.55 | whizzi | eduardonunesp: also check your php.ini for timeouts and things ;) |
| 13:52.03 | whizzi | memory limit |
| 13:52.03 | Runedhand | whizzi, if he forgot, and that opened a thousand agi-php stuff, it's possible. |
| 13:52.52 | whizzi | agree, if we're talking thousands, then sure ;) |
| 13:53.32 | eduardonunesp | whizzi, default_socket_timeout = 60 |
| 13:53.40 | eduardonunesp | whizzi, in php.ini |
| 13:54.42 | whizzi | and memory-limit ? |
| 13:54.48 | eduardonunesp | whizzi, someone put a huge limit |
| 13:54.53 | eduardonunesp | whizzi, of 1024 |
| 13:56.19 | spengler1 | WIMPy ; so my outbound context would look like this exten => _9.,1,Dial(DAHDI/g0/${EXTEN:1} |
| 13:57.23 | whizzi | eduardonunesp: I'd just check the agi scripts, it's hanging somewhere there |
| 13:57.58 | *** join/#asterisk protocoldoug (~doug@unaffiliated/protocoldoug) |
| 13:58.16 | eduardonunesp | whizzi, yours too ? |
| 13:58.18 | *** join/#asterisk thecardsmith (~doug@unaffiliated/protocoldoug) |
| 13:58.50 | whizzi | well, memory limit could be much less |
| 13:59.02 | whizzi | 1024M is kinda much for simple text scripts |
| 13:59.11 | whizzi | 64MB is usually enough |
| 13:59.13 | *** join/#asterisk thecardsmith (~doug@unaffiliated/protocoldoug) |
| 13:59.38 | eduardonunesp | whizzi, sure, but i'm informed that exists some reports in system that consumes too much |
| 13:59.46 | *** join/#asterisk creativx (~creadurex@226.62-97-205.bkkb.no) |
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| 14:04.47 | whizzi | eduardonunesp: it's not good agi-scripts use that amount of memory. So something in that code is wrong in some way, could be the closing-issue (asterisk->disconnect(); ) or something else (loops? DB queries taking too long) |
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| 14:05.32 | eduardonunesp | $asterisk->disconnect is a call from php-agi, right ? |
| 14:05.32 | WIMPy | spengler1: Possible, if you want to cut off the first digit. |
| 14:05.44 | *** join/#asterisk amessina (~amessina@2001:470:c1dc:7779:d6be:d9ff:fe8d:7c1e) |
| 14:06.16 | Runedhand | eduardonunesp, yes |
| 14:06.36 | Runedhand | Whatever the name of the php-agi instance is. |
| 14:06.46 | Runedhand | can be $ast, $foo whatever. |
| 14:06.49 | eduardonunesp | Runedhand, the system using a library from phpagi project |
| 14:07.08 | eduardonunesp | http://phpagi.sourceforge.net/ |
| 14:07.15 | Runedhand | We use the same. |
| 14:07.25 | eduardonunesp | so i'm looking for disconnect |
| 14:07.35 | eduardonunesp | and haven't found |
| 14:07.49 | Runedhand | $astman = new AGI_AsteriskManager(); |
| 14:08.00 | Runedhand | This is my variable for asterisk stuff. |
| 14:09.55 | eduardonunesp | here we have a mix of phpagi and zend |
| 14:10.18 | *** join/#asterisk sekil (~sekil@78.24.104.73) |
| 14:10.21 | Runedhand | Brb. |
| 14:10.41 | eduardonunesp | so i've searched in sources looking for disconnect statement, and nothing |
| 14:11.04 | eduardonunesp | i think the problem must be much database calls in AGIs |
| 14:13.08 | eduardonunesp | need to take a break for lunch, come back in a bit, thanks |
| 14:13.28 | whizzi | or fix it both ;) |
| 14:13.35 | whizzi | $astman->disconnect(); should be doing the trick then at the end of the AGI-script |
| 14:15.38 | *** join/#asterisk JonathanD (~JonathanD@freenode/staff/jonathand) |
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| 14:28.08 | Runedhand | I don't know how your php-agi works then. |
| 14:28.19 | Runedhand | Without connection and disconnection functions, it seems a bit weird. |
| 14:28.49 | WIMPy | wonders if you guys are talking about the same thing at all. |
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| 14:32.25 | Runedhand | eduardonunesp: |
| 14:32.39 | Runedhand | $astman->connect(); |
| 14:32.39 | Runedhand | $astman->database_put("QPENALTY/".$kuyrukno, "dynmemberonly", "no"); |
| 14:32.39 | Runedhand | $astman->disconnect(); |
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| 15:22.31 | *** join/#asterisk BinaryMaster (~IceChat9@rrcs-97-76-15-66.se.biz.rr.com) |
| 15:22.58 | BinaryMaster | Question is there a way to watch traffice to and from a specific peer only? |
| 15:23.14 | eduardonunesp | Runedhand, but this is for the AMI client, ins't ? |
| 15:23.35 | Runedhand | eduardonunesp, that's for AGI. |
| 15:23.42 | Runedhand | PHPAGI, to be specific. |
| 15:23.43 | *** join/#asterisk luckman212 (~luckman21@unaffiliated/luckman212) |
| 15:23.53 | eduardonunesp | Runedhand, right, let me see |
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| 15:27.37 | eduardonunesp | Runedhand, i don't see any disconnect line in sources for PHPAGI |
| 15:27.58 | Runedhand | Try calling it from a php file. |
| 15:27.59 | eduardonunesp | Runedhand, only a function with name disconnect, which closes a socket |
| 15:28.12 | Runedhand | Yeah, that's the one.ç |
| 15:28.27 | Runedhand | You should disconnect your webserver from asterisk once you're done |
| 15:29.35 | eduardonunesp | Runedhand, there is no web server connections made to asterisk, they are made from dial plan when call a AGI function |
| 15:29.54 | Runedhand | Well... I can't help with that then. :/ |
| 15:30.28 | eduardonunesp | Runedhand, like exten => s,1,AGI(script.php) |
| 15:31.17 | eduardonunesp | Runedhand, no problem, i appreciate your effort |
| 15:31.49 | Runedhand | Sorry. :( |
| 15:32.04 | eduardonunesp | Runedhand, so is possible that problem is too much db calls in scripts |
| 15:33.13 | Runedhand | Could be. |
| 15:35.07 | BinaryMaster | Question: Any ideas what to look for if a remote "NAT" peer can connect to the Asterisk server can call another peer on the system but there is no audio |
| 15:35.50 | *** join/#asterisk luckman212 (~luckman21@unaffiliated/luckman212) |
| 15:37.38 | eduardonunesp | BinaryMaster, int's a problem with Reinvites ? |
| 15:39.01 | BinaryMaster | I'm not sure I have way too much out put in the console to isolate the two peers |
| 15:39.26 | geeksteve | do a tcpdump of the sip traffic, make sure the SDP looks sensible |
| 15:40.17 | BinaryMaster | okay |
| 15:42.08 | BinaryMaster | apparently they can initiate a call to me but I am unable to initiate to them |
| 15:42.56 | *** join/#asterisk phr0zen (~phr0zen@blk-7-133-0.eastlink.ca) |
| 15:43.43 | Penguin | No audio is almost always misconfigured nat settings in asterisk, SIP ALG on a router, or nat settings being enabled on the phone. |
| 15:43.55 | BinaryMaster | everything was working earlier today, but our Internet provider changed our ip addresses since then we have had the problems |
| 15:44.15 | Penguin | Fix your externaddr value in asterisk. |
| 15:45.52 | BinaryMaster | I don't see where that setting is |
| 15:45.59 | Penguin | sip.conf |
| 15:46.10 | Penguin | It's part of configuring asterisk for use with nat. |
| 15:47.45 | BinaryMaster | would that have been auto configured if I was using freepbx |
| 15:47.53 | Penguin | ~freepbx |
| 15:47.53 | 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 |
| 15:48.02 | Penguin | No clue. |
| 15:48.36 | BinaryMaster | awesome, I hate that I'm the one in charge of this stupid phone system |
| 15:48.46 | phr0zen | for free pbx, install the module 'asterisk sip settings' |
| 15:49.17 | phr0zen | from there you can configure all of your NAT settings including internal and external networks |
| 15:49.22 | geeksteve | probably in sip_nat.conf, but depends on who did it and where really |
| 15:49.31 | geeksteve | could be in sip*_custom.conf |
| 15:49.43 | Penguin | binarymaster: I think phr0zen and geeksteve are willing to help you in #FreePBX |
| 15:50.00 | BinaryMaster | Thank you |
| 15:50.00 | whizzi | grep *your old ip address* /etc/asterisk/* |
| 15:50.05 | geeksteve | also works |
| 15:50.22 | geeksteve | :) |
| 15:50.34 | whizzi | change that, reload sip in asterisk and you're good to go |
| 15:50.45 | geeksteve | +1, that's the most likely cause |
| 15:51.19 | Penguin | whizzi: If it is configured by FreePBX, that may not be what he should do. |
| 15:51.39 | whizzi | agree, but it's the quick solution |
| 15:51.59 | whizzi | and only if his ISP only changed IP-address |
| 15:51.59 | geeksteve | and if the option wasn't obvious in a gui, then it's probably been done by hand |
| 15:52.44 | phr0zen | the freepbx option is found in the module "asterisk sip settings" which then adds the external ip to "sip_general_additional.conf" |
| 15:52.51 | BinaryMaster | sweet found it |
| 15:52.57 | BinaryMaster | thank you guys!!!! |
| 15:53.13 | geeksteve | no prob. just make sure it's not in a file with 'do not edit' type comments in it |
| 15:53.17 | geeksteve | or it may be nuked by the gui |
| 15:53.34 | geeksteve | they're usually obviously labelled if that's the case |
| 15:53.41 | Penguin | We don't have those types of files in asterisk, so you're discussing it in the wrong channel. |
| 15:54.06 | geeksteve | It's an asterisk based system, and it's not like it's so busy we're disrupting other conversations ;) |
| 15:54.25 | whizzi | he left already |
| 15:54.37 | geeksteve | not suprising, probably running for cover |
| 15:54.41 | phr0zen | lol |
| 15:54.50 | phr0zen | i do need some help over in freepbx :) |
| 15:55.11 | WIMPy | I have an Indali which is based on Asterisk. Do O get support for that here too? |
| 15:55.28 | WIMPy | I |
| 15:55.50 | whizzi | I guess if the question is really in Asterisk, I don't see why not |
| 15:56.07 | *** join/#asterisk luckman212 (~luckman21@unaffiliated/luckman212) |
| 15:56.10 | geeksteve | never heard of it, but i'm sure we'll give it a go |
| 15:56.28 | whizzi | speaking of Asterisk, I still have a very strange issue |
| 15:57.16 | whizzi | http://pastebin.com/Qu0LSJae |
| 15:57.18 | WIMPy | No, the only thing Asterisk related thing is that I have NFI what Version it is. |
| 15:57.46 | whizzi | my incoming phone calls seem to go to random contexts |
| 15:58.18 | Penguin | Then the call is not matching the peer entry. |
| 15:58.37 | geeksteve | mmmkay.. could it be that sip.prov.nl (for example) resolves to multiple addresses? |
| 15:58.45 | WIMPy | It's matching the host which is identical. |
| 15:58.52 | Penguin | whizzi: You have multiple entries with the same host. |
| 15:58.54 | geeksteve | could try defining by IP. never found asterisk to be great where dns is concerned |
| 15:59.05 | geeksteve | guessing that's where he's sanitised it |
| 15:59.08 | whizzi | sip.prov.nl point to 2 addresses |
| 15:59.09 | geeksteve | insanely :P |
| 15:59.12 | geeksteve | hmm |
| 15:59.15 | Penguin | whizzi: You can't do that. |
| 15:59.20 | geeksteve | try static config of those IP addresses as different peers |
| 15:59.25 | geeksteve | [peer1] |
| 15:59.27 | geeksteve | host=123.123.123.123 |
| 15:59.31 | geeksteve | [peer2] |
| 15:59.35 | geeksteve | host=234.234.234.234 |
| 15:59.36 | Penguin | You have THREE peers with the same host. |
| 15:59.37 | geeksteve | and the rest the same |
| 15:59.38 | Penguin | Three. |
| 15:59.46 | Penguin | Not one or two, but three. |
| 16:00.02 | whizzi | it's a Kameleo server |
| 16:00.07 | whizzi | sip.prov.nl |
| 16:00.10 | Penguin | You can't do that. |
| 16:00.33 | Penguin | That system isn't using a username, I'm sure. |
| 16:00.39 | geeksteve | asterisk will resolve the dns entry once at reload, so it'll work for one IP |
| 16:00.47 | geeksteve | you need to define each IP as a peer |
| 16:00.51 | geeksteve | and hope they dont change them ;) |
| 16:01.07 | Penguin | geeksteve: He has THREE peers with the same host and the host resolves to TWO different addresses. It's not going to work. |
| 16:01.13 | geeksteve | i see that |
| 16:01.17 | geeksteve | and that's why i'm telling him how to configure it |
| 16:01.34 | Penguin | If they only have two addresses, you can't configure three peers that way. |
| 16:02.00 | whizzi | shall I scare you even more then |
| 16:02.08 | geeksteve | i never said to make three. i said he needs as many peers as they have IP addresses |
| 16:02.31 | geeksteve | whizzi: scare me |
| 16:02.41 | whizzi | 287 entries |
| 16:02.55 | geeksteve | O_o |
| 16:02.57 | geeksteve | uh |
| 16:03.03 | whizzi | [st-x2-custID] |
| 16:03.22 | whizzi | context=custID-inkomend |
| 16:03.27 | whizzi | type=friend |
| 16:03.29 | geeksteve | if it's only a single carrier you have, have a peer that matches everything, then restrict with iptables instead? would have to do some fun dialplan to sort that out though... |
| 16:03.30 | geeksteve | not the best idea |
| 16:03.32 | whizzi | host=sip.prov.nl |
| 16:03.47 | geeksteve | is the host= really the same for them all? what's the intention there? |
| 16:03.50 | Penguin | type=friend is supposed to try to match by username first, ip/port second. Be very sure your proxy is passing the username correctly. |
| 16:04.05 | *** join/#asterisk BCS-Satori (~BCS-Sator@2001:470:e169:860:6e3b:e5ff:fe38:da53) |
| 16:04.11 | geeksteve | i'm confused why you have 287 peers for a single carrier |
| 16:04.18 | geeksteve | what's that for exactly? |
| 16:04.21 | whizzi | multi-tennant system |
| 16:04.31 | geeksteve | you doing a peer per number or something? |
| 16:04.40 | geeksteve | or is every tennant being the same ip?! |
| 16:04.44 | Penguin | 287 customers, probably |
| 16:04.50 | geeksteve | indeed, but with the same host? |
| 16:04.53 | Penguin | one single proxy |
| 16:04.55 | whizzi | correct |
| 16:04.58 | geeksteve | eek |
| 16:05.07 | geeksteve | going to need something better than IP to match on then |
| 16:05.16 | whizzi | I haven't made this, I'm only the one who's going to improve it |
| 16:05.16 | Penguin | (1003.50) <Penguin> type=friend is supposed to try to match by username first, ip/port second. Be very sure your proxy is passing the username correctly. |
| 16:05.20 | geeksteve | i'd do a single peer for the proxy, then identify on something else (CLI if you control that?) |
| 16:05.53 | geeksteve | depends what elements the end user controls, if they can edit username/cli then you're boned lol |
| 16:06.00 | Penguin | Identify by command line interface? Weird. |
| 16:06.07 | whizzi | outgoing works perfectly |
| 16:06.10 | geeksteve | CLI - calling line identification |
| 16:06.15 | Penguin | ~cli |
| 16:06.15 | infobot | cli is probably a Command Line Interface, the best form of interface around, of course Call-Level Interface, originally developed by SQL Access Group, but now known as Microsoft's ODBC standard. an x86 assembly instruction Common Language Infrastructure (See mono or .net) |
| 16:06.22 | geeksteve | i believe americans add a D to the end as well |
| 16:06.32 | whizzi | ok, here's the whole situation |
| 16:06.32 | geeksteve | less common in the uk where i happen to be |
| 16:07.09 | whizzi | 1 SIP provider (using Kameleo and failover etc etc) |
| 16:07.12 | geeksteve | ok |
| 16:07.18 | BCS-Satori | Does the http server in asterisk work in 11.7.0? I am trying to work on phoneprov but in my http.conf file I have enabled=yes with the bindaddr and port but when I do a http show status after asterisk restart it says the server is disabled. |
| 16:07.21 | Penguin | Did you ensure that your proxy is passing the user name correctly? |
| 16:07.26 | whizzi | sip.prov.nl has 2 IPv4-addresses |
| 16:07.30 | whizzi | Penguin: Yes :) |
| 16:07.45 | Penguin | Then there shouldn't be any more problem. type=friend matches by username. |
| 16:08.03 | whizzi | every customer has his own whole context-party of they're own |
| 16:08.28 | geeksteve | except using dns for something with two IP addresses, might stuff things up |
| 16:08.28 | whizzi | and they can add or remove peers and users |
| 16:08.33 | geeksteve | ok |
| 16:08.35 | Penguin | I'm not very familiar with Kamailio, so I wouldn't know what to have you check for those settings. |
| 16:08.50 | whizzi | I can't login to that machine.. I wish I could |
| 16:09.10 | whizzi | what I do see, is Asterisk responding in a strange way |
| 16:09.13 | geeksteve | well a good first step is to do a sip trace, and make sure *you* can tell the calls apart |
| 16:09.20 | geeksteve | if you can, then asterisk will be able to eventually |
| 16:09.28 | geeksteve | if you cant - then you need more identifying info :S |
| 16:09.30 | Penguin | Since you've identified the calls going to multiple, seemingly random, contexts, that says to me that you are matching on host and not user name. |
| 16:09.55 | whizzi | outgoing works, customers use their own peer and that's perfectly seperated |
| 16:10.08 | Penguin | Then we don't need to talk about that part of it. |
| 16:10.13 | whizzi | exactly |
| 16:10.22 | whizzi | my problem is with the incoming part |
| 16:10.30 | Penguin | You've mentioned it several times already, though. |
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| 16:10.54 | *** part/#asterisk LiuYan (~LiuYan@222.125.137.85) |
| 16:10.58 | whizzi | Let's say, I have customer 1 who has phonenumer 12345 |
| 16:11.01 | Penguin | Do you have more than one peer entry by the same name? |
| 16:11.03 | geeksteve | make sure sip show peers shows the right IP - i.e. the right one of the two. i'm uncertain what a peer with right username but wrong IP would do |
| 16:11.08 | whizzi | Penguin: No |
| 16:11.24 | whizzi | geeksteve: yes, that's all fine :) |
| 16:11.29 | geeksteve | hmm ok |
| 16:11.50 | whizzi | So, I call nr 12345 and expect to go to context=cust1-inkomend right ? |
| 16:12.03 | whizzi | since cust1 also has that phone number |
| 16:12.10 | geeksteve | does it mis-direct to the same context each time, or is it acutally random? |
| 16:12.23 | whizzi | it's actually random |
| 16:12.50 | whizzi | If I call 12345 I have approx an 80% chance I'll end in context=incoming |
| 16:13.25 | whizzi | which is good, I'll throw the number to the context=cust1-inkomend anyway and that works |
| 16:14.00 | whizzi | but, when the machine is busy (I'm talking about > 30 open channels), there's a chance my nr actually gets into ... context=cust2-inkomend |
| 16:14.07 | WIMPy | And the way to do it. |
| 16:14.26 | WIMPy | If you have multiple numbers on one ITSP, you need extensions, not contexts. |
| 16:14.49 | geeksteve | he has, seperate peers each hits a context |
| 16:15.17 | whizzi | and per customer (cust1 in this example), there's an extension 12345 => |
| 16:16.02 | WIMPy | We,, you could do one peer per ip and number. But apart from being a lot of peers I see no benefits. |
| 16:16.38 | whizzi | the only working solution now is that I throw everything into context=incoming |
| 16:16.43 | whizzi | where it will be seperated |
| 16:16.52 | Penguin | If asterisk cannot match on the user name, it will match on the host. The host is always the same. It is clearly matching on the host and it matches a random peer entry each time. Go fix your kamailio server to send the username. |
| 16:16.58 | geeksteve | is there a downside of doing it in one context? |
| 16:17.08 | geeksteve | pick out on some other attribute, and then push to contexts? |
| 16:17.23 | whizzi | it allows customers to steal each others nrs geeksteve |
| 16:17.41 | whizzi | if cust2 has 98765 as nr |
| 16:17.58 | WIMPy | How so? |
| 16:18.00 | whizzi | and cust1 randomly chooses to put 98765 as incoming extension |
| 16:18.17 | whizzi | cust1 will receive it's phonecall |
| 16:18.34 | WIMPy | Why can they set up ranom incomming numbers? |
| 16:18.47 | whizzi | because they all end in the same extension |
| 16:18.50 | whizzi | incoming |
| 16:19.16 | whizzi | the extension incoming includes all customers/*/extensions.conf |
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| 16:19.34 | *** mode/#asterisk [+o newtonr] by ChanServ |
| 16:19.54 | geeksteve | dont define customer numbers in the shared context then |
| 16:20.06 | geeksteve | do that in the customer one, which you goto based on a suitable attribute |
| 16:20.20 | geeksteve | i'd really look at sip traces and make sure the users are identifyable |
| 16:20.26 | WIMPy | I'd go and think about what users should be able to configure. |
| 16:21.12 | geeksteve | indeed, some sort of sanity checking for extension creation would be wise anyway |
| 16:21.13 | *** join/#asterisk thecardsmith (~doug@unaffiliated/protocoldoug) |
| 16:21.57 | Penguin | Or just fix the kamailio server and move on to something else. |
| 16:22.08 | whizzi | geeksteve: that is actually one of the things which are happening |
| 16:22.56 | whizzi | geeksteve: the problem is that for some reason, asterisk seems to think the incoming peer is full or so and just throws it at a random context |
| 16:23.21 | eduardonunesp | worth using the dundi ? |
| 16:23.22 | Penguin | Are you not paying attention or what? |
| 16:23.31 | Penguin | (1016.52) <Penguin> If asterisk cannot match on the user name, it will match on the host. The host is always the same. It is clearly matching on the host and it matches a random peer entry each time. Go fix your kamailio server to send the username. |
| 16:23.34 | geeksteve | oh back in your box |
| 16:23.38 | whizzi | Penguin: I'll definately look into that Kamalio-server, that could explain it a lot |
| 16:23.39 | Penguin | READ and UNDERSTAND what I am telling you. |
| 16:23.54 | whizzi | I am mate ;) |
| 16:24.40 | Penguin | If the kamailio server sends the user name on each call, the problem will be solved. |
| 16:24.58 | whizzi | aye, that's the only explanation |
| 16:25.09 | eduardonunesp | anyone here already used DANDI ? |
| 16:25.10 | WIMPy | What problem? |
| 16:25.15 | eduardonunesp | ops DUNDi |
| 16:25.34 | Penguin | He's using type=friend. That tells asterisk to match calls on the USER NAME. |
| 16:25.41 | WIMPy | The only "problem" I see are wrong expectations. |
| 16:26.09 | Penguin | But if the user name is not there, asterisk will match by IP/port. In this case, the host is the same on all the peer entries. |
| 16:26.10 | WIMPy | eduardonunesp: Ask real questions, not meta questions. |
| 16:26.17 | Penguin | So asterisk is doing what it is supposed to be doing. |
| 16:26.36 | whizzi | I never doubt Asterisk, I only doubt the configuration ;) |
| 16:26.58 | WIMPy | Beginners mistake |
| 16:27.20 | whizzi | but Penguin has a point, I'll do some traces and see what I actually get from those Kamilio machines |
| 16:27.40 | eduardonunesp | WIMPy, sorry i heard about DUNDi, and i want to know if worst using asterisk with cluster configuration |
| 16:27.51 | whizzi | it would immediately explain why Asterisk is matching on hostname (cause that's what it's doing) instead of username |
| 16:28.37 | Penguin | I have the same problem with an ITSP. I want to have multiple accounts on asterisk, but the ITSP does not send username information. |
| 16:29.00 | WIMPy | There are many ways to do clusters, althoug I guess you want a grid. |
| 16:30.06 | whizzi | Penguin: ok, than I think I found the explanation I was looking for |
| 16:30.12 | whizzi | thanks for your advice all :) |
| 16:30.56 | *** join/#asterisk luckman212 (~luckman21@unaffiliated/luckman212) |
| 16:31.31 | whizzi | anyway, my boss is not paying me for this so I'll head off home.. I'll probably be around tomorrow again here to see if I also can help people with the knowledge I have :) |
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| 16:35.15 | BCS-Satori | Does the http server in asterisk work in 11.7.0? I am trying to work on phoneprov but in my http.conf file I have enabled=yes with the bindaddr and port but when I do a http show status after asterisk restart it says the server is disabled. |
| 16:35.49 | eduardonunesp | WIMPy, i want to make an experiment of make more then one machine with asterisk act like one machine |
| 16:36.01 | eduardonunesp | WIMPy, ins't a cluster conf ? |
| 16:36.13 | Penguin | bcs-satori: If you find that it does not work, file a bug report. |
| 16:36.36 | WIMPy | A cluster is a bunch of machines running one operating system. |
| 16:37.53 | phr0zen | gonna ask here as freepbx is kinda dead, but it is an asterisk function that i am inquiring about (i can get the freepbx specific stuff later) |
| 16:37.59 | phr0zen | I have a questions regarding call recording after transfering. Currently if I take and inbound queue call, the call records as it is supposed to. I then transfer the call to another queue, but the recording does not stop, it continues to record. What I want, is for the initial call to record, then stop when transfered, then record if the second queue is set to record. i did some googling and am not sure if I am on the right path here, but |
| 16:37.59 | phr0zen | would setting AUDIOHOOK_INHERIT from yes to no accomplish this? (found in extensions_additional.conf) |
| 16:38.02 | eduardonunesp | WIMPy, cluster are usually connected to each other through fast local area networks ("LAN"), with each node (computer used as a server) running its own instance of an operating system. |
| 16:39.03 | Penguin | phr0zen: That's what AUDIOHOOK_INHERIT does. It makes Monitor or MixMonitor keep recording. |
| 16:39.15 | WIMPy | eduardonunesp: That's what used to be called grid computing. |
| 16:39.23 | *** part/#asterisk whizzi (~whizzi@82-171-3-8.ip.telfort.nl) |
| 16:39.36 | Penguin | Set(AUDIOHOOK_INHERIT(MixMonitor)=yes), for example |
| 16:40.02 | Penguin | Remove the Set() or change the value to no. |
| 16:40.27 | eduardonunesp | WIMPy, ok, my doubt is about the perform and configuration of DUNDi |
| 16:40.29 | tm1000 | anyone using pjsip in A12? |
| 16:40.34 | Chainsaw | Is it a semicolon for comments in sip.conf? Doesn't seem to like hashing out a register statement... |
| 16:40.41 | Chainsaw | tm1000: Still on A11 here, sorry. |
| 16:40.57 | Penguin | chainsaw: Yes, semi colon for commenting in almost every asterisk config file. |
| 16:41.04 | Chainsaw | Penguin: Very good, let's try that. |
| 16:41.05 | eduardonunesp | WIMPy, is really perform as one super server |
| 16:41.07 | file | tm1000, occasionally |
| 16:41.18 | tm1000 | file: didnt want to bother the all mighty Josh ;-) |
| 16:41.25 | eduardonunesp | WIMPy, making a bunch of machines seems only one asterisk server |
| 16:41.27 | tm1000 | using pjsip show endpoints |
| 16:41.30 | file | tm1000, pfft |
| 16:41.37 | tm1000 | and seeing: Invalid 1 |
| 16:41.50 | tm1000 | but I dont know if that means invalid config or invalid because no endpoint is connected |
| 16:41.51 | phr0zen | so here is the scenario, caller1 -> queue1 -> answered by agent1 (caller1 and agent1 recording now). agent1 -> conference call to queue2 with caller1 (warm transfering) -> caller1 and agent1 still being recorded -> agent2 answers the call, agent1 hangs up -> caller 1 and agent2 still being recording in same file |
| 16:41.51 | WIMPy | eduardonunesp: You probably wat to look at dynamit realtime as well. |
| 16:42.06 | file | tm1000, you can never have an invalid config |
| 16:42.08 | Penguin | phr0zen: You don't need to repeat. I gave you the answer. |
| 16:42.17 | tm1000 | file: http://pastebin.com/2Qga7TAf |
| 16:42.25 | phr0zen | was just clarifying to make sure we are on the same page, sorry |
| 16:42.35 | phr0zen | we do transfers a bit differently for this particular scenario |
| 16:42.36 | file | tm1000, that's the device state |
| 16:42.48 | tm1000 | file: so since no endpoint is connected it says invalid? |
| 16:42.49 | file | tm1000, uno momento |
| 16:43.02 | Penguin | phr0zen: It does not inherit by default, so delete the Set() that is doing it. Or change the value to no. |
| 16:43.11 | phr0zen | ok perfect |
| 16:43.11 | eduardonunesp | WIMPy, sorry i don't get it |
| 16:43.11 | file | tm1000, https://issues.asterisk.org/jira/browse/ASTERISK-23065 might be what you are hitting |
| 16:43.33 | phr0zen | i think freepbx does do that by default (i did not change that), which confused me a bit since i thought it was default no |
| 16:43.33 | Penguin | phr0zen: You can stop recording and restart it on a different call leg. |
| 16:43.48 | Penguin | phr0zen: I don't speak FreePBX. |
| 16:44.07 | phr0zen | i know, that is why i was just asking about the asterisk function and not specifically freepbx |
| 16:44.13 | Penguin | In asterisk, it is not set until you set it. |
| 16:44.15 | WIMPy | eduardonunesp: You probably want to store the Asterisk configuration in a central database. |
| 16:44.30 | phr0zen | penguin: thanks again, i will try setting that to no and see how it goes |
| 16:44.37 | eduardonunesp | WIMPy, yes |
| 16:44.53 | file | tm1000, could be extended to general device state incorrectness on initial start |
| 16:44.57 | eduardonunesp | WIMPy, and distribute the load |
| 16:47.14 | Penguin | phr0zen: core show function AUDIOHOOK_INHERIT |
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| 16:59.07 | phr0zen | thanks penguin |
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| 17:32.21 | Penguin | luckman212: Please fix your connection. |
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| 17:43.55 | Penguin | luckman212: Please fix your connection. |
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| 18:01.24 | *** join/#asterisk eduzimrs (~eduzimrs@mail.aytycrm.com.br) |
| 18:02.21 | eduzimrs | hi, anyone knows how can i monitor a sip trunk with my voip provider? |
| 18:02.47 | eduzimrs | maybe using sip signaling |
| 18:03.13 | *** join/#asterisk cmendes0101 (~cmendes01@office.phone.com) |
| 18:03.42 | tm1000 | file: thanks. I havent actually tried to register an enpoint yet. So when I get to that point I'll look through that ticket |
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| 18:07.00 | newtonr | eduzimrs, what do you mean by monitor? what do you want to know or do with the SIP trunk? |
| 18:08.46 | eduzimrs | @newtonr, sry i mean i´d like to monitor with zabbix, but im trying to find out how to do it by using sip signaling, which i believe is the best and trusted way |
| 18:11.28 | phr0zen | eduzimrs: look into this perhaps? http://www.communig8.com/articles/67-technical/149-using-zabbix-to-monitor-asterisk-sip-trunks |
| 18:11.33 | phr0zen | i just googled what you asked lol |
| 18:12.44 | newtonr | eduzimrs, I am not familiar with Zabbix. However I just googled as phr0zen did and found at least five or six useful looking sources of information on your question including the link phr0zen provided |
| 18:13.43 | eduzimrs | phr0zen, ok but, there are many providers that does not use registration and block icmp, thats why im asking its not easy that way |
| 18:14.24 | eduzimrs | using SIP signaling i be sure that the response will always came |
| 18:16.20 | *** join/#asterisk luckman212 (~luckman21@unaffiliated/luckman212) |
| 18:16.42 | phr0zen | not sure then, might be a question for your provider |
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| 18:20.10 | _Corey_ | eduzimrs: We use sipsak with Nagios in the manor you describe. I believe it uses an OPTIONS message and times the response. You should be able to do something similar. |
| 18:21.29 | eduzimrs | _Corey_, ill try this out tks |
| 18:24.05 | eduzimrs | _Corey_, can u paste the arguments passed to sipsak ? |
| 18:24.56 | _Corey_ | eduzimrs: Sorry, I wasn't the one who set it up and would have to research it over here. |
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| 18:30.02 | eduzimrs | ok |
| 18:37.26 | *** join/#asterisk ChannelZ-Wk (~bobm@spark.idolum.com) |
| 18:45.22 | jeffspeff | what's a good (preferably free) tool or suite to pen-test my asterisk box? |
| 18:46.33 | ChannelZ-Wk | mmmhehmm penetration mmheh hehmmheh |
| 18:46.43 | jeffspeff | lol |
| 18:47.05 | ChannelZ-Wk | Put your system on the public internet. People will test it for you |
| 18:47.20 | jeffspeff | yeah, i know that. |
| 18:47.31 | jeffspeff | but i'd rather not foot the bill if they make it in |
| 18:47.39 | jeffspeff | just a personal preference |
| 18:48.29 | ChannelZ-Wk | you can play with sipvicious |
| 18:49.20 | jeffspeff | i tried putting that on my laptop, but couldn't get it to run |
| 18:49.37 | jeffspeff | i forget the error. and yes, i had python installed and in the system path |
| 18:49.50 | jeffspeff | i just figured that sipvicious maybe wasn't a good tool |
| 18:50.02 | ChannelZ-Wk | But I'd say turn off allowguest if you don't need anonymous calls, make sure your peer names aren't stupid (like "100") and have good secrets if you use dynamic peers, and run fail2ban to keep an eye on things like multiple failed registrations, etc. |
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| 18:51.21 | jeffspeff | well, peer names aren't 100, but they are the exten numbers. and fail2ban is what i'm working on setting up. i wanted a way to test it out though |
| 18:53.52 | ChannelZ-Wk | Well I get that extension numbers for peer names makes it easy to write lazy dialplans, but it's not necessarily the best idea. All day long I see registration attempts using numeric names |
| 18:54.08 | ChannelZ-Wk | In any event I can throw some bogus registrations your way |
| 18:54.50 | jeffspeff | but after your 2nd failed attempt, your ip gets blocked |
| 18:55.24 | ChannelZ-Wk | Isn't that the point? |
| 18:55.45 | ChannelZ-Wk | You said you wanted to test it |
| 18:55.48 | jeffspeff | lol, yes |
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| 19:24.47 | bsdice | I see about 10 port 5060 scans a day |
| 19:25.00 | bsdice | from all over the map |
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| 19:43.54 | ChannelZ-Wk | yeah I get a ton on the weekend. |
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| 19:54.20 | Penguin | I have a huge network blacklist, so attempts on my asterisk have been reduced. |
| 19:55.33 | WIMPy | I hardly see any attempts any more. |
| 19:55.46 | WIMPy | But I filtered the friendly scnaner. |
| 19:55.55 | bsdice | Penguin blacklist using ipset blacklisting entire countries? I plan to do that |
| 19:56.07 | bsdice | block China, Russia, the Khazaks for good measure too |
| 19:56.31 | bsdice | well actually 5060 is already blocked, because machine is doing stateful UDP |
| 19:56.59 | bsdice | which times out after 180 secs, qualify=50 to get 3 attempts for keepalive into that interval |
| 19:58.29 | bsdice | machine logs blocked attemps anyway (and drops packet) |
| 19:58.30 | Penguin | I keep network and address lists with ipset. I haven't done it by country on this firewall, but I did do that on a web server recently. |
| 19:58.50 | Penguin | I have whitelist and blacklist for both specific addresses and network. |
| 19:58.50 | bsdice | you block entire subnets from whois then? |
| 19:58.53 | Penguin | Yes. |
| 19:59.00 | bsdice | that's what I did with SMTP |
| 19:59.19 | bsdice | Crystone SE spammers were really getting on my nerves |
| 19:59.46 | bsdice | apparently this hoster does not care about spammers so they flock there |
| 19:59.55 | bsdice | just blacklisted entire subnets |
| 20:00.12 | Penguin | I did get tired of doing whois after a while and started blocking /16 if I saw several different addresses from various third octets. |
| 20:00.29 | bsdice | 212.19.32.176/28 554 Sorry, too much spam from you. Please contact us by phone (+49xxx) to resolve issue. |
| 20:00.44 | Penguin | I've got at least one /10 in the list, too. |
| 20:01.31 | Penguin | It's annoying and I get tired of them. I guess I should take the country list from the web server's firewall and apply it to the voip network's firewall. |
| 20:01.33 | bsdice | what an epidemic |
| 20:01.44 | *** join/#asterisk luckman212 (~luckman21@unaffiliated/luckman212) |
| 20:01.58 | bsdice | do you have to allow 5060 udp inward for clients? |
| 20:01.59 | Penguin | I'd guess that would kill off a bunch more. |
| 20:02.20 | bsdice | I just allow 5061 TLS TCP |
| 20:02.44 | Penguin | It's not only phones on the outside, but also needing to keep it open for ISN. |
| 20:03.01 | bsdice | ~isn |
| 20:03.01 | infobot | it has been said that isn is ITAD Subscriber Number (see 'itad'). An ISN is a method of dialing SIP URI's via a standard keypad on a telephone. Because of the alphanumeric nature of SIP URIs, it is difficult to dial them via the keypad on your phone. The use of ISN numbers simplifies this by utilizing DNS lookups to map the ISN number to a domain. See http://www.freenum.org for more information. |
| 20:04.11 | Penguin | I guess I could do away with that. |
| 20:04.28 | Penguin | Or, actually, just firewall off all those other countries anyway. |
| 20:04.37 | Penguin | They don't need to call here via SIP URI. |
| 20:05.05 | bsdice | what country are you in |
| 20:05.37 | *** join/#asterisk hayden_ru (~hayden_ru@0896410724.static.corbina.ru) |
| 20:06.09 | Penguin | USA |
| 20:06.26 | ChannelZ-Wk | PHONE RACIST! |
| 20:06.44 | Penguin | I don't speak Chinese anyway. |
| 20:07.08 | bsdice | Mandarin :) |
| 20:07.41 | ChannelZ-Wk | my 'AHOLES' table has grown fairly large |
| 20:12.43 | *** join/#asterisk Vutral (ss@mirbsd/special/Vutral) |
| 20:15.57 | *** join/#asterisk luckman212 (~luckman21@unaffiliated/luckman212) |
| 20:19.47 | Penguin | 1035 networks added to the network_blacklist. |
| 20:20.27 | *** join/#asterisk Vutral (ss@mirbsd/special/Vutral) |
| 20:21.21 | Penguin | Now it takes a really long time to load the configuration. |
| 20:22.31 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.127.143) |
| 20:23.05 | Penguin | If I can apply multiple lists, maybe I should break that down into several lists separated by country. |
| 20:24.12 | Chainsaw | Or whitelist instead of blacklist. |
| 20:25.01 | Penguin | Block everything and whitelist a select few? |
| 20:25.38 | Penguin | Damn, it's taking a REEEEEEEEEALLY long time to load up. It still isn't done. |
| 20:27.02 | WIMPy | Looks like you're crossed the line where it makes mire sense to just pull the plug. |
| 20:27.12 | Chainsaw | Penguin: That's what I do, yes. |
| 20:27.23 | Chainsaw | Penguin: I don't do it in software though. |
| 20:28.40 | ChannelZ-Wk | I'm only at a couple hundred, ~100 of which are 'permenant' from a DB.. the others are dynamic, from SIP drive-bys and spam |
| 20:31.30 | *** join/#asterisk tzafrir (~tzafrir@bzq-218-28-58.cablep.bezeqint.net) |
| 20:35.50 | Penguin | I was maintaining the list on the border firewall plus whatever fail2ban was doing on the asterisk box. I wanted to change fail2ban's behavior and make it update the border firewall, but I couldn't come up with any good way to do that. |
| 20:38.28 | *** join/#asterisk hayden_ru (~hayden_ru@0896410724.static.corbina.ru) |
| 20:39.36 | ChannelZ-Wk | make the ban action call something on the other machine.. a script or something, or ssh a command |
| 20:40.33 | Penguin | If only it were that simple. |
| 20:40.49 | Penguin | I could do that, but then it would not be stored in the configuration. |
| 20:41.13 | *** join/#asterisk luckman212 (~luckman21@unaffiliated/luckman212) |
| 20:42.00 | Penguin | I suppose with the proper scripting it could be added to the configuration. I would just have to edit the config file inline to add the lines. |
| 20:44.09 | ChannelZ-Wk | Part of my firewall gets generated from a database. My AHOLES table. A script lets me easily add an IP or block with a comment, which adds to the iptables table and shoves it in the database for future reloads |
| 20:44.47 | Penguin | I'm using Vyatta, and everything needs to be maintained in the config.boot file. |
| 20:45.47 | Penguin | I'm told to not run any of the vbash commands via ssh, so I can't just add rules like I would on the console. |
| 20:46.30 | *** join/#asterisk Vutral (~ss@mirbsd/special/Vutral) |
| 20:47.14 | *** join/#asterisk luckman212 (~luckman21@unaffiliated/luckman212) |
| 20:48.59 | ChannelZ-Wk | Sucks for you! I even look at the table packet counter prior to reloading and update a hit counter in the database so I can purge oldies. |
| 20:49.28 | ChannelZ-Wk | ponders lunch |
| 20:49.47 | Penguin | It just takes more maintenance rather than having it done automatically. |
| 20:49.54 | Penguin | It could be worse. |
| 20:59.08 | Penguin | I also have to "sed 's/^/ network /' /tmp/block.tmp" before I feed it to that config file. |
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