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| 00:38.27 | MaliutaLap | [TK]D-Fender: shouldn't it be set to 8.8.8.8? ;) |
| 00:38.42 | MaliutaLap | and yeah, we're not #freepbx |
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| 01:46.57 | bjhaid | hi, how do I include a custom field (or use the userfield) in a CDR from a call originated via an AMI |
| 02:29.51 | bjhaid | nvm found it |
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| 07:38.47 | CptStabington | good day |
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| 07:41.07 | CptStabington | Quick question. I'm trying to install the sccp module on an AsteriskNOW box. Which rpm should I grab? |
| 07:41.28 | CptStabington | Not quite sure which distro it's based on |
| 07:44.07 | Chainsaw | CptStabington: If the modules & core are not from the same set, they will not work together. |
| 07:44.21 | Chainsaw | CptStabington: So you must find out where your main Asterisk RPM was sourced from. |
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| 08:39.42 | imihaylov | hello! Could someone help how to make Asterisk 1.8 show date variable like 2015_08_08 ? |
| 09:07.35 | WIMPy | imihaylov: Use the STRFTIME function. |
| 09:09.07 | CptStabington | Hm, this is strange. The administration web page on my asteriskNow box is lacking options |
| 09:09.52 | CptStabington | I remember there being loads of menus and whatnot. Now there's only a 'System Admin' thing |
| 09:10.09 | CptStabington | I logged in as root, maybe that's the issue? |
| 09:10.11 | WIMPy | CptStabington: Ask in #freepbx |
| 09:10.24 | CptStabington | Oh there's a channel for that too :D |
| 09:10.47 | WIMPy | GUIs are not supported here. |
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| 09:24.30 | imihaylov | <WIMPy> Thanks, I figured that out... I got it wrong that STRFTIME has veen deprecated and not DATETIME |
| 09:26.09 | *** join/#asterisk denbeiren (~quassel@ptr-178-51-217-247.dyn.mobistar.be) |
| 09:26.46 | denbeiren | is there a starters guide for asterisk (someone who has never worked with any voip related matter) |
| 09:27.22 | WIMPy | ~primer |
| 09:27.22 | infobot | New to asterisk configuration? Check out this primer to get started. http://burner.com/asterisk-primer |
| 09:27.27 | WIMPy | ~book |
| 09:27.27 | 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 |
| 09:28.08 | denbeiren | thx WIMPy |
| 09:28.20 | denbeiren | i'll get started on the primer |
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| 10:47.44 | ipalmer | morning all. Is there a way of answering a call whilst the periodic announcement is being played, I've tested it and can't seem to do it, just wondering if there's a setting I don;t know about. Cheers |
| 10:48.59 | WIMPy | doesn't know any. |
| 10:49.21 | ipalmer | WIMPy: was that in response to me? |
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| 10:49.33 | WIMPy | yes |
| 10:49.46 | ipalmer | WIMPy: ok cheers. Adios |
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| 11:23.17 | CptStabington | Oh dear. I'm in a pickle |
| 11:24.02 | CptStabington | I'm installing the asteriskNOW distro, and it wants me to punch in ctrl-alt-F1 |
| 11:24.18 | CptStabington | I can't. It's inside Hyper-V ._. |
| 11:25.04 | CptStabington | says some firstboot error occured |
| 11:28.38 | CptStabington | aha! I can use ssh, no worries |
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| 12:10.48 | markl | Few days ago I mentioned a strange problem regarding our asterisk 13.5.0 on this channel |
| 12:11.09 | markl | few times a day rtp stream stops and peers are going unreachable |
| 12:11.14 | markl | for 5 - 20 seconds |
| 12:11.57 | markl | monitoring and debugging shows it has nothing to do with network, cpu, IO, timing, RAM |
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| 12:12.07 | markl | anyone an idea what it could be? |
| 12:20.47 | *** join/#asterisk [TK]D-Fender (~chatzilla@216-191-106-163.dedicated.allstream.net) |
| 12:24.02 | Zogot | markl: fixed times? |
| 12:24.51 | Zogot | do you have an asterisk log of the period that it stops? |
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| 12:29.05 | markl | no, not fixed times |
| 12:29.26 | markl | I have a log, but I can't see anything there |
| 12:30.44 | WIMPy | Nothing on vmstat? |
| 12:31.50 | markl | i'll try to upload the vmstat log |
| 12:42.12 | Zogot | is there a way to get the length of a recording from MixMonitor? |
| 12:42.31 | markl | vmstat log: https://www.dropbox.com/s/jw168bqn9es5s5c/vmstat.log?dl=0 |
| 12:42.50 | markl | issue occured at 09:29:59 |
| 12:56.58 | [TK]D-Fender | Zogot: either set a timer before you start and compare, or shell out and stat it |
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| 13:08.31 | Zogot | [TK]D-Fender: yeh, we had that idea a few moments ago, except we can't keep track of both sides with that. if we support both ends recording as h exten doesn't fire for the callee |
| 13:09.38 | [TK]D-Fender | Depends |
| 13:09.51 | [TK]D-Fender | if you're using Dial there is an option for that |
| 13:10.11 | Zogot | trying from the feature codes |
| 13:10.29 | Zogot | record => *3,self,Gosub(call_recording,s,1) |
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| 13:52.56 | markl | WIMPy, Zogot any idea? |
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| 15:11.10 | *** join/#asterisk bkervaski (18c50696@gateway/web/freenode/ip.24.197.6.150) |
| 15:13.44 | bkervaski | Hey guys .. I have a small utility Asterisk box that runs a non-public phone system for a bunch of our locations and it works great. The Asterisk servers sit behind NAT, and when I have the public port set to 5060 everything rocks right along. When I try to set an alternate public port other than 5060 then clients from the outside get dropped after just a few seconds of a call. Is there a setting for the public NAT port like th |
| 15:15.03 | [TK]D-Fender | You don't get to use multiple ports on the same channel driver |
| 15:15.05 | *** join/#asterisk tzafrir (~tzafrir@local.xorcom.com) |
| 15:15.12 | [TK]D-Fender | And forwarding will not work |
| 15:15.26 | [TK]D-Fender | because SIP specifies the contact port to use in the payload itself |
| 15:15.46 | [TK]D-Fender | so rewrite will just en up getting you no answers at all and things drop off the map |
| 15:16.04 | bkervaski | I have it mapped on the firewall, public port 6050, local net port 5060 ... |
| 15:16.19 | [TK]D-Fender | as I just said this will fail tragically |
| 15:16.24 | [TK]D-Fender | do NOT attempt this |
| 15:16.24 | bkervaski | Got it. |
| 15:16.42 | [TK]D-Fender | because the content of the packets will say "I'm on 5060" still and NOT whatever this remap port is |
| 15:16.45 | bkervaski | Okay, so basically just set an alternat SIP port all around and just use that ... |
| 15:16.54 | [TK]D-Fender | Why are you doing this at all? |
| 15:16.55 | bkervaski | Yep. |
| 15:18.21 | bkervaski | Well .. I have local SIP clients and remote SIP clients that all need to connect .. some local net, some out on the internet, etc. But, having 5060 open just invites all those random attempts from hackers (or whatever). I can't have an IP whitelist, I can't do VPNs. There's nothing mission critical with these servers, just relaying calls, not even connected to anything else. It's utilitarian really and works great except for t |
| 15:18.46 | bkervaski | Changing the SIP port at least makes it harder on port scanners |
| 15:19.00 | bkervaski | Changing the [public] SIP port .. not considering just changing it all around |
| 15:19.06 | bkervaski | not = now |
| 15:20.22 | [TK]D-Fender | That is not proper security. |
| 15:20.35 | glNito | If your IP is public I'd almost guarantee you will be scanned and attacked by scanners, you're probably better off just working on your firewall settings |
| 15:20.38 | [TK]D-Fender | just deal with firewalling and flood attacks like everyone else |
| 15:22.35 | bkervaski | Yep, understood, thanks for the help :) |
| 15:23.22 | bkervaski | (and I can't believe you're still hanging out here, last time I signed in to IRC was like 2009/10ish and it was #asterisk) |
| 15:24.45 | *** join/#asterisk coppice (~chatzilla@123203240102.ctinets.com) |
| 15:24.54 | lennard | but the logging is so messy ;) |
| 15:25.24 | [TK]D-Fender | Time keeps on slippin', slippin.... |
| 15:26.26 | lennard | I just replaced my first Dial(,,M(...)) with Dial(,,U(...)), yay! |
| 15:29.16 | igcewieling | bkervaski: I solve that issue on our systems by allowing all incoming requests. If the call is a guest call (not auth'd) I route the call to an IVR designed to waste their time. |
| 15:29.55 | igcewieling | I find even after days of attackes, once I set that up attacks went way down. |
| 15:30.46 | [TK]D-Fender | that solves 1 type of attacker |
| 15:31.00 | igcewieling | [TK]D-Fender: there are others? 8-| |
| 15:31.13 | [TK]D-Fender | Next is the one that tries to reg first THEn places calls thus ccounting as "authed" |
| 15:31.34 | [TK]D-Fender | Then there are the ones who just flood you with requests and effectively DoS you |
| 15:31.40 | igcewieling | [TK]D-Fender: I set autocreatepeer and make the default context be the IVR. |
| 15:31.50 | *** join/#asterisk ipengineer (~zconkle@static-71-252-134-63.dllstx.fios.verizon.net) |
| 15:35.40 | igcewieling | [TK]D-Fender: I find that, once successful, those floods stop. |
| 15:38.32 | [TK]D-Fender | suitably dirty.... unless they actually hit an existing peername |
| 15:38.59 | igcewieling | For known attack scripts, I also use some iptables rules. See http://pastebin.ca/3185555 |
| 15:39.11 | igcewieling | [TK]D-Fender: we mainly auth on ip address |
| 15:39.41 | igcewieling | I also to additional checks in my AGI script. |
| 15:41.29 | glNito | igcewieling:You can also add |
| 15:41.30 | glNito | -A SIPSCAN -m string --string "sipcli/v1.8" --algo bm --to 15000 -j DROP |
| 15:41.48 | igcewieling | glNito: Thanks. |
| 15:42.48 | bkervaski | I think I'm just going to periodically scan the logs and put the IP addresses in iptables .. or just do a white list .. which will be a pain .. but even though they aren't getting anywhere we already have scanners taxing the systme to the point of uselessness .. and it doesn't help we hav a 100mb connection |
| 15:45.42 | [TK]D-Fender | bkervaski: Here's a solution for those as well... http://pastebin.com/p1ViMFPK |
| 15:48.05 | bkervaski | Interesting ... thanks again |
| 15:48.50 | bkervaski | Ok, I'm out, see you guys in 3 years ;) |
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| 16:32.26 | glNito | Wow, this scanner generated 43624 CDRs in 3 hours this morning |
| 16:34.55 | igcewieling | glNito: Hmm? |
| 16:35.01 | igcewieling | SIP scanner? |
| 16:35.18 | glNito | I think so? There's no common hostname and it has been trying WORDS and extensions |
| 16:35.51 | *** join/#asterisk drab (~administr@23-24-198-190-static.hfc.comcastbusiness.net) |
| 16:36.08 | igcewieling | ah. |
| 16:36.27 | igcewieling | 011 9011 99011 999999999011, etc? |
| 16:38.18 | drab | has anybody seen sporadic missing audio with zoiper clients? this is on an iphone on the same network of an extension I'm trying to call (voip lg nortel phone). using asterisk 11. No NAT involved and dumping debug logs on astersik I see no errors, the call went to voicemail and all, but on the iphone I never even heard it rining, complete silence (the other phone was ringing tho) |
| 16:39.00 | drab | but it's sporadic, maybe 1 call out of 10 or something. the iphone is connected through an acces point that goes to the same switch the phone and the asterisk server are connected to |
| 16:39.06 | glNito | no, it doesn't look like international, I am just seeing the attacker's INVITE (in this case, "fries@PBX_IP") then we reply with 11 200 OKs |
| 16:42.17 | glNito | I think it kept going because it received 200 OK when trying words |
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| 16:56.18 | igcewieling | Can anyone reccoment a command line utility to search pcaps for specific to or from in a group of files? As far as I can tell sipgrep doesn't appear to support wildcard specifications |
| 16:56.40 | eschmidbauer | does anyone know a carrier that does outbound cname on toll free numbers? |
| 16:57.08 | igcewieling | eschmidbauer: your question makes little sense. |
| 16:58.47 | eschmidbauer | what don't you understand? |
| 17:00.05 | igcewieling | eschmidbauer: CNAM is an inbound service. |
| 17:00.30 | igcewieling | The receiving carrier of the call looks up the CNAM. |
| 17:00.53 | eschmidbauer | there is an outbound service too |
| 17:02.12 | *** join/#asterisk darkbasic_ (~quassel@niko.linuxsystems.it) |
| 17:06.57 | glNito | "Setting Caller ID Name (CNAM) on your PBX has no effect in the U.S. and most other countries. Only the CID number is transmitted to the terminating carrier. The terminating carrier then does a lookup in the Line Information Database (LIDB) to determine the name associated with the number. Some carriers only look to free databases, because there is |
| 17:06.57 | glNito | <PROTECTED> |
| 17:08.44 | igcewieling | There you go. |
| 17:12.18 | eschmidbauer | your carrier sets the CNAM info in the database |
| 17:12.24 | eschmidbauer | for future lookups |
| 17:12.30 | eschmidbauer | that is what I mean by outbound |
| 17:20.23 | igcewieling | I don't know if there is an LIDB for Toll Free. |
| 17:21.04 | eschmidbauer | i think you are correct |
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| 18:10.46 | pankid | How can I troubleshoot asterisk manager not running? Doesnt look like anything is binding to port 5038 and freepbx is unable to make any changes because it can't access asterisk manger |
| 18:11.52 | cusco | netstat -tapn|grep 5038 - anything? |
| 18:13.01 | cusco | in asterisk cli: manager show settings |
| 18:13.06 | pankid | oh wait,yeah |
| 18:13.08 | pankid | [root@pbx asterisk]# netstat -tapn|grep 5038 |
| 18:13.08 | pankid | tcp 0 0 192.168.3.2:5038 0.0.0.0:* LISTEN 23208/asterisk |
| 18:13.21 | cusco | so, its listening |
| 18:13.25 | nunopereira | pankid, pastebin with it |
| 18:13.57 | cusco | but its fixed on IP 192.168.3.2 (instead of 0.0.0.0) |
| 18:14.03 | pankid | https://paste.debian.net/314995/ |
| 18:14.08 | cusco | tried: nc 192.168.3.2 5038 |
| 18:14.09 | cusco | ? |
| 18:14.16 | pankid | is it because it is binding to a 192.168 address instead of 127.0.0.1? |
| 18:14.31 | cusco | seems all good |
| 18:14.35 | cusco | because of manager.conf ? |
| 18:14.35 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 18:14.36 | glNito | it should resolve |
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| 18:15.05 | cusco | bindaddr = 0.0.0.0 |
| 18:16.45 | pankid | cusco: that did it |
| 18:16.46 | pankid | thanks a ton |
| 18:16.55 | pankid | wonder how that got changed |
| 18:18.06 | cusco | dunno, I've never used freepbx |
| 18:18.36 | WIMPy | Does it have a random failure button? |
| 18:19.08 | pankid | I really dont like freepbx. |
| 18:19.29 | WIMPy | You're in the right channel then. |
| 18:19.34 | cusco | :) |
| 18:26.12 | [TK]D-Fender | cuscobindaddr = 0.0.0.0 <- do NOT do this unless you have external servers that you need to access AMI |
| 18:31.44 | Milenco | yup, better bind to 127.0.0.1 |
| 18:36.59 | *** join/#asterisk Stabington (~CptStabin@unaffiliated/cptstabington) |
| 18:37.22 | Stabington | Good day |
| 18:38.43 | Milenco | hello |
| 18:39.25 | Stabington | So, I'm trying to get this DHCP option 66 thing to work.. and I've managed to create a SEP<MAC>.cnf.xml file inside my tftp server. |
| 18:40.04 | Stabington | Do I need another file or something? The phone doesn't seem to pick it up |
| 18:40.26 | Milenco | this isn't really related to asterisk. what kind of phone are you using? |
| 18:40.31 | WIMPy | Did you configure youre DHCP server? |
| 18:40.41 | Milenco | and what dhcp server are you using? |
| 18:40.58 | Stabington | I haven't configured it yet. It's a Meraki Z1 |
| 18:41.03 | Milenco | if you do a tcpdump during dhcp negotiations, what do you see? |
| 18:41.11 | Stabington | But, I have punched in the TFTP server manually into the phone |
| 18:42.03 | Stabington | So it should, in theory, go ahead and fetch the config file |
| 18:42.14 | [TK]D-Fender | <PROTECTED> |
| 18:42.20 | Stabington | Yeap |
| 18:42.21 | Milenco | i dont have experience with those phones. normally, you specify option 66 to a tftp (or sometimes http) server and make sure the file the phone is looking for is there |
| 18:42.23 | [TK]D-Fender | are you sure this is appropriate to your phone? |
| 18:42.26 | WIMPy | then you asked the wrong question. |
| 18:43.03 | Stabington | It is an SCCP phone |
| 18:43.56 | [TK]D-Fender | ok, I just got phone vs router reveresed there |
| 18:45.45 | Stabington | My end goal is to use option 66 to configure the phones, but that will come later. Now I just want the phone to pick up whatever is on the TFTP server |
| 18:47.44 | Stabington | oh! I think I see the problem. |
| 18:50.05 | Stabington | yup.. I apparently didn't have to type in SEP when I typed in the MAC address in freepbx. now the file is named SEPSEP<MAC>.cnf.xml :D |
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| 18:50.54 | drmessano | DUCKDUCK<GOOSE |
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| 19:17.06 | Stabington | Cool. Now it's not listening on port 2000 for some reason |
| 19:18.16 | Stabington | I have no idea what I'm doing :D |
| 19:18.50 | Stabington | Literally grabbed like "Hey, you're a linux nerd. Get this thing up and running" |
| 19:23.36 | Stabington | Do I need some gypsy magic to make it listen on port 2000 |
| 19:27.01 | [TK]D-Fender | Is the channel driver there and loaded? |
| 19:27.19 | Stabington | yeap |
| 19:28.01 | [TK]D-Fender | Show us |
| 19:28.12 | [TK]D-Fender | As well as your netstat to match |
| 19:29.32 | Stabington | At least FreePBX claims it is enabled |
| 19:29.53 | [TK]D-Fender | that measn... somewhat less than nothing to me... |
| 19:30.13 | Stabington | How can I check this from the command line |
| 19:30.14 | [TK]D-Fender | And you should be asking in their channel if the GUI is involved with the fact of things not being configured right |
| 19:30.42 | [TK]D-Fender | "module load chan_sccp.so" |
| 19:30.46 | [TK]D-Fender | try to load the module. |
| 19:30.50 | [TK]D-Fender | and see if it complains |
| 19:31.30 | Stabington | oh, it sure does. |
| 19:31.47 | Stabington | FreePBX has been nothing but a hazzle so far >.> |
| 19:32.23 | [TK]D-Fender | Things to note, Cisco = pain, and SCCP = MISTAKE. |
| 19:32.33 | Stabington | Didn't properly configure the xml either |
| 19:32.48 | [TK]D-Fender | What didn't configure the XML properly? |
| 19:33.07 | Stabington | The sccp manager for freepbx |
| 19:33.58 | [TK]D-Fender | Never heard of.... |
| 19:34.10 | Stabington | WARNING[2334]: loader.c:502 load_dynamic_module: Error loading module 'chan_sccp.so': /usr/lib64/asterisk/modules/chan_sccp.so: undefined symbol: __ast_module_unref |
| 19:34.21 | [TK]D-Fender | looks like compile error <- |
| 19:34.43 | WIMPy | Sounds like a broken installation. |
| 19:34.47 | Stabington | Huh, all seemed fine during compilation |
| 19:34.54 | [TK]D-Fender | So not a FreePBX issue |
| 19:35.26 | WIMPy | Do you have multiple Asterisk versions instlled? |
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| 19:36.30 | Stabington | That would be weird. It's literally AsteriskNOW out of the box |
| 19:36.47 | Stabington | version 11 is running |
| 19:37.07 | WIMPy | Since when do you compile yourself there? |
| 19:37.30 | Stabington | I had to ._. sccp module weren't included |
| 19:39.29 | Stabington | or is it..? Crap, it might have just not been in FreePBX |
| 19:40.16 | WIMPy | If you try to modify FreePBX, you should go to #freepbx. Even if I don't know if that helps. |
| 19:41.04 | Stabington | Isn't that just the front-end though |
| 19:41.52 | WIMPy | Technically, yes. But who knows with what Asterisk options it works or doesn't. |
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| 19:48.36 | pavelsc | Hello. Can anyone help me with ami event ConfbridgeListRooms? it returns event with parameters "actionid":"1444316684478","conference":"192","parties":"1","marked":"1". Where 192 is extension+customer_id. Is there a way to receive it as "19_2" ? |
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| 19:49.16 | WIMPy | You gave the conference its name. |
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| 19:52.05 | pavelsc | there is 'name' field in mysql table for conferences, but it is no unique |
| 19:53.19 | WIMPy | I have no idea what you're talking about. You call the ConfBridge application with a name from your dialplan. That's what you get. |
| 19:53.20 | [TK]D-Fender | There is no such thing as a customer ID in a conference |
| 19:53.38 | [TK]D-Fender | There is only the properties of the conference as created |
| 19:53.52 | WIMPy | How you make up that name is up to you. |
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| 19:55.21 | pavelsc | Ahh, got it! Thank you guys. |
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| 23:47.46 | prppedro | Good night, people... |
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