| 00:01.48 | ChannelZ | ps ax |grep asterisk |
| 00:03.13 | overyander | avahi 397 0.0 0.0 28072 1672 ? Ss 04:19 0:00 avahi-daemon: running [asterisk-1.local] root 6092 0.0 0.0 115212 816 ? S 19:01 0:00 /bin/sh /usr/sbin/safe_asterisk |
| 00:03.13 | overyander | root 6094 1.4 0.7 3075248 59164 ? Sl 19:01 0:51 /usr/sbin/asterisk -f -vvvg -c |
| 00:03.14 | overyander | root 6551 0.0 0.0 112640 964 pts/0 S+ 20:02 0:00 grep --color=auto asterisk |
| 00:12.37 | *** join/#asterisk phormulate (~phormulat@unaffiliated/phormulate) |
| 00:37.12 | ChannelZ | So safe_asterisk runs the real asterisk and relaunches it if it dies. Though I dunno why sysctl knows the right pid of the actual asterisk and not safe_asterisk which is what it's actually running |
| 00:47.10 | ChannelZ | but I'm avoiding systemd for as long as possible |
| 00:49.47 | WIMPy | Sounds like a good idea to me. |
| 00:51.57 | overyander | so, you sticking with centos 6.5 or older releases of debian? |
| 00:52.38 | WIMPy | is using Slackware, which seems to be safe for the moment. |
| 00:53.51 | overyander | despite the process child warning, it all seems to be running ok |
| 01:39.15 | *** join/#asterisk Tenhi (~tenhi@178.18.241.180) |
| 01:49.49 | *** join/#asterisk evilman_work (~evilman@87.244.6.228) |
| 02:00.25 | *** join/#asterisk frek818 (~frek818@172.56.17.39) |
| 02:02.59 | *** join/#asterisk pppingme (~pppingme@unaffiliated/pppingme) |
| 02:15.15 | overyander | ~book |
| 02:15.15 | 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 |
| 02:17.18 | overyander | so, the online version of the boook only applies to asterisk 1.8? |
| 02:18.14 | WIMPy | It applies to all versions, but you certainly won't find information about chan_pjsip for example. |
| 02:19.30 | overyander | the 4th edition would have pjsip stuff wouldn't it? when i go to www.asteriskdocs.org i don't see any links to view the 4th edition online |
| 02:20.19 | overyander | it so happens that i'm trying to learn about pjsip. just did a fresh install of * 13 and thought it would be better to learn pjsip and start using it instead of chan_sip |
| 02:20.24 | WIMPy | You can read it, there's just no pdf download. |
| 02:20.50 | WIMPy | wiki.asterisk.org is the place for you then. |
| 02:21.58 | overyander | i was reading the links for the different pjsip modules, just thought something in more of a book format would help put all the pieces together for me. |
| 02:22.40 | WIMPy | I don't think there's anything more "booky", yet. |
| 03:16.11 | *** join/#asterisk cyford (support@c-73-137-1-6.hsd1.ga.comcast.net) |
| 03:16.31 | *** join/#asterisk vader- (~Adium@pool-71-175-67-97.phlapa.fios.verizon.net) |
| 03:16.55 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 03:45.23 | overyander | are there any reasons to switch to pjsip from sip other than for the sake of embracing the new? |
| 03:52.36 | ChannelZ | There are some features that are worthwhile to some |
| 03:54.25 | ChannelZ | like multiple registrations to a device, multiple interface binding support |
| 03:55.32 | overyander | can you do realtime sip registrations yet in pjsip or do you still have to use the flat file for that? |
| 04:02.33 | ChannelZ | I think you can |
| 04:17.14 | moe` | <ChannelZ> like multiple registrations to a device, multiple interface binding support |
| 04:17.21 | moe` | that kinda gives me a woody. :) |
| 04:17.40 | *** join/#asterisk protem` (~protem@unaffiliated/protem) |
| 04:19.55 | overyander | so, the multiple registrations aspect. could that be used to register the same peer on two different * boxes for failover or load balancing? |
| 04:22.08 | moe` | I doubt you'll get that much joy from it. |
| 04:22.31 | moe` | in fact, I'm sure you will not. |
| 04:24.34 | overyander | i was thinking about multiple * systems, using a shared realtime db in mulit-master mode using gluster-fs to keep the two systems exactly in sync |
| 04:29.44 | moe` | I'm not expert, but I can't see that ending well. |
| 04:31.34 | overyander | lol |
| 04:34.38 | moe` | you'd best ask the folks known as file or malcolmd or ChannelZ or [TK]D-Fender of items so detailed. |
| 04:39.21 | ChannelZ | The multiple registrations is the other way around; multiple devices can be registered to the same SIP peer. So dialing PJSIP/Joe might ring 5 different devices |
| 04:41.25 | ChannelZ | For load balancing you're probably better off putting a generic SIP proxy in front of your Asterisk boxes |
| 04:42.41 | overyander | ok. i was getting excited for a moment. :P |
| 04:43.50 | overyander | i can see how that would really help for those offices that still want to have 'line 1' 'line 2' etc. on all their phones |
| 04:44.13 | overyander | it'd make it easier than configuring pickup groups |
| 04:45.16 | drmessano | In that application, you could just use multiple devices with chan_sip just as easily |
| 04:47.08 | overyander | so, i guess pjsip/joe ringing multiple devices would be the equivelant of me having multiple extensions configured with the same internal CID so i can use my desk phone, softphone on laptop and softphone on my cell phone? |
| 04:47.54 | ChannelZ | well right now to do it you'd have to Dial(SIP/Phone1&SIP/Phone2&SIP/Phone3) etc |
| 04:48.53 | ChannelZ | If you want to add a 4th, you edit your dialplan. With multiple addresses on a PJSIP endpoint, your dialplan doesn't have to care how many or what each specific peer is called |
| 04:49.17 | drmessano | I think it's "neat" that PJSIP permits multiple regs, and for some, that would consolidate such a scenario.. but inevitably, something would come up to where "I just want to ring the softphone on my cell" and Asterisk doesnt do that yet |
| 04:49.59 | drmessano | I think PJSIP does, but last I checked, it's all or nothing in Asterisk |
| 04:51.14 | drmessano | So I still like multiple devices, even if you're using pjsip.. |
| 04:51.17 | *** join/#asterisk mbecroft (~user@ak2.becroft.co.nz) |
| 04:51.18 | ChannelZ | Well it can't work by mind reading. How would you expect it to work? |
| 04:52.30 | ChannelZ | I mean, I don't know how asterisk being involved would change that scenario |
| 04:52.45 | drmessano | The PJSIP stack can address those individual devices, but chan_pjsip doesn't support that yet.. So if it did work, I would expect it to work using that support, at some point. Duh |
| 04:53.22 | overyander | i was referring to dialing pjsip/foobar instead of sip/foobar&sip/foobar1&sip/foobar2 |
| 04:53.25 | drmessano | You dont know how Asterisk being involved would change that? It's called "Feature not yet supported" |
| 04:55.10 | moe` | <ChannelZ> The multiple registrations is the other way around; multiple devices can be registered to the same SIP peer. So dialing PJSIP/Joe might ring 5 different devices .... SO GOOD |
| 04:55.31 | moe` | I just setup my asterisk to "roll over" to other devices, so this feature is frickin' awesome |
| 04:55.48 | drmessano | overyander, yes, I know.. and I was suggesting that if you regged all of your devices to one pjsip registration, and then decided you wanted to create an extension to address ONE of those, you can't |
| 04:56.01 | overyander | i see |
| 04:56.08 | drmessano | moe`, you don't need pjsip to do that.. just ring all of those SIP devices |
| 04:56.25 | moe` | and dialplan that how? |
| 04:56.33 | overyander | drmessano, could you dial a full uri to target an individual device? |
| 04:56.44 | drmessano | SIP/104&SIP/105&SIP/106 |
| 04:56.48 | moe` | right now I have it ring one device, then another, each device having a unique SIP profile |
| 04:56.59 | ChannelZ | I'm just saying that you want to use a feature in a way it wasn't even designed for, so don't use it then. |
| 04:57.15 | drmessano | .... |
| 04:57.17 | drmessano | Whatever |
| 04:57.25 | ChannelZ | If you want things separate, then yeah, duh, they aught to be separate. |
| 04:57.27 | overyander | let's say i have foobar registerd to 192.168.1.10 and 192.168.1.20. could you dial only one of them by specifying foobar@192.168.1.20 from the dialplan? |
| 04:57.42 | drmessano | overyander, no you can't.. not yet |
| 04:57.47 | moe` | exten=>101,1,Dial(SIP/martin-s2,15) |
| 04:57.47 | moe` | exten=>101,2,Dial(SIP/martin-s5,15) |
| 04:57.47 | moe` | exten=>101,3,Dial(SIP/martin-netbook,15) |
| 04:57.47 | moe` | exten=>101,4,Dial(SIP/martin,60) |
| 04:57.53 | moe` | right now I do that, which is ugly |
| 04:57.56 | moe` | nasty |
| 04:58.12 | overyander | agrees |
| 04:58.17 | drmessano | moe`, SIP/martin-s2&SIP/martin-s5&SIP/martin-netbook |
| 04:58.20 | drmessano | ^^^^ |
| 04:58.26 | drmessano | & |
| 04:58.27 | overyander | was about to say the same thing |
| 04:58.31 | moe` | no way |
| 04:58.34 | drmessano | Yes |
| 04:58.42 | ChannelZ | If you want them to all ring simultaneously yes |
| 04:58.42 | moe` | see I'm still very much an asterisk newbie |
| 04:58.52 | moe` | yeah that's exactly what I want |
| 04:59.08 | moe` | how good is that |
| 04:59.12 | moe` | you guys are awesome |
| 04:59.13 | ChannelZ | The way you just showed moe would be for a sequential. IE giving one person the chance to answer for a few seconds before trying someone else |
| 04:59.40 | drmessano | You dont need PJSIP to ring devices simultaneously or to make them the same EXTENSION.. DIAL already supports multiple SIP devices |
| 05:00.07 | ChannelZ | Right. which I said. |
| 05:00.10 | overyander | and setting the same CID(num) on all of them makes them appear as one device when making calls to other extens |
| 05:00.14 | drmessano | PJSIP just has the ability to use ONE REGISTRATION with multiple devices.. which may or may not be useful in some cases |
| 05:00.55 | drmessano | and personally, I think the ability to break out those SIP devices in other ways is far more useful to me |
| 05:01.03 | overyander | can you use pjsip and sip simultaneously? or do you have to choose one or the other |
| 05:01.06 | moe` | does a dialplan reload |
| 05:01.12 | moe` | lets test this |
| 05:01.18 | ChannelZ | You can use them both but they can't both bind to the same SIP port |
| 05:01.25 | overyander | ah |
| 05:01.26 | drmessano | Because I may want 104 to ring SIP/home&SIP/car&SIP/netbook but maybe 904 rings just SIP/netbook |
| 05:01.41 | moe` | HOW GOOD IS THAT |
| 05:01.46 | overyander | lol |
| 05:01.57 | moe` | something so simple just made my day! |
| 05:02.07 | overyander | gotta love * |
| 05:02.20 | moe` | its so good |
| 05:02.29 | moe` | I just last night got flowroute SIP connected to it |
| 05:02.42 | overyander | i use flowroute too, they're pretty good |
| 05:02.49 | moe` | so now I can dial out with a 9 prefix for Skype or 7 prefix for flowroute |
| 05:02.51 | ChannelZ | One annoying thing (to me) about pjsip+asterisk is that it seems to send PROGRESS _immediately_ to the dialing device even before it's tried the remote end |
| 05:03.08 | overyander | moe`, that'd be in your outbound context |
| 05:03.16 | moe` | yeah, that's quite intentional |
| 05:03.28 | moe` | so I can pick which outbound provider I want to use |
| 05:03.33 | overyander | yeah |
| 05:03.52 | drmessano | Not sure why Skype would even be an option |
| 05:04.13 | drmessano | Their per minute pricing is double that of any reasonable ITSP, especially flowroute |
| 05:05.24 | overyander | something like exten=9XXXXXXXXXX,1,Dial(sip/flowroute/${exten}:1) then exten=8XXXXXXXXX,1,Dial(sip/foobar/${exten}:1) |
| 05:05.26 | ChannelZ | I liked it as a novelty to do Skype-to-Skype calls (back when SFA was around) but in the end never even used it more than a couple times |
| 05:05.36 | moe` | yeah its there cuz it was the first SIP provider I plugged into asterisk |
| 05:05.50 | moe` | ; --- outbound SIP calls |
| 05:05.51 | moe` | exten=>_9.,1,Dial(SIP/skype/${EXTEN:1}) |
| 05:05.51 | moe` | exten=>_7.,1,Dial(SIP/flowroute/${EXTEN:1}) |
| 05:06.11 | overyander | drmessano, who do you use? their per min pricing is pretty cheap in the US compared to some others i've seen |
| 05:06.22 | overyander | moe`, never do 9. |
| 05:06.28 | moe` | why is that? |
| 05:06.29 | drmessano | I use Flowroute |
| 05:06.37 | moe` | you are not the first person to tell me that |
| 05:06.37 | drmessano | No reason to use anyone else |
| 05:06.54 | drmessano | Skype is nowhere near cheap |
| 05:07.17 | overyander | it's always best practice to set specific patterns for what you're expecting to match. the '.' allows anyhing, not just numbers, but ANY character |
| 05:07.46 | drmessano | Like 2.3cents per minute is 2.5x that of Flowroute |
| 05:07.51 | moe` | yeah I know, regex type things are good friends of mine. |
| 05:07.52 | moe` | :) |
| 05:07.56 | drmessano | How is that even remotely cheap? |
| 05:08.30 | overyander | also, shouldn't you use (...${EXTEN}:1) ? |
| 05:08.35 | drmessano | Anyone above 1.5 cents per minute in the US is expensive |
| 05:08.45 | moe` | overyander: yeah but I could want to call the same number with use skype SIP or flowroute SIP, depending on the target number/location |
| 05:09.34 | moe` | so the 9 prefix was just arbitrary what I remembered from years ago old school PBX's to dial "outside" lines. |
| 05:09.39 | overyander | you were correct. the ":X" goes inside the braces |
| 05:09.50 | overyander | for some reason i was thinking it went on the outside |
| 05:09.52 | moe` | the 7 prefix for flowroute was just because I had to pick a number |
| 05:10.03 | overyander | i know |
| 05:10.11 | overyander | i was referring to the string cutting |
| 05:10.15 | moe` | oh |
| 05:10.33 | moe` | well, it works. I'm not an expert on asterisk config syntax |
| 05:10.38 | overyander | but you really should use 9XXXXXXXXXX instead of 9. |
| 05:10.49 | moe` | yeah I know, but... |
| 05:10.58 | moe` | well, see, where I am, the area code is 4 digits |
| 05:11.03 | moe` | I'm in Curacao |
| 05:11.08 | moe` | prefix 5999 |
| 05:11.16 | overyander | just place an X to match the total number of digits you want to match |
| 05:11.22 | overyander | that's all an X is |
| 05:11.32 | moe` | I know. but then I'd need two exten entries |
| 05:11.40 | moe` | one for 9 and another for 10 digit numbers. |
| 05:11.41 | overyander | 9X says i want to match any 2 digit number beginning with 9 |
| 05:12.01 | overyander | it's better than allowing 9*&KLKJLKJH(&*^YKJH:LKJ |
| 05:12.08 | moe` | again: my area code is 5999 (4 digits) |
| 05:12.30 | moe` | yeah overyander I know the pattern I have can be used for nasty stupid things |
| 05:12.45 | moe` | _9[0-9]+ would be better, even |
| 05:12.59 | drmessano | His box is secure.. im sure he has a BSD firewall |
| 05:13.14 | overyander | :p |
| 05:13.25 | moe` | snickers at drmessano ... its funny cuz its true |
| 05:13.47 | drmessano | Thats like the inside joke of 2015 now |
| 05:13.54 | drmessano | "But I have a BSD firewall.." |
| 05:14.00 | moe` | unixninja.net baby |
| 05:14.02 | drmessano | "It's ok, I have a BSD firewall" |
| 05:14.13 | overyander | you're killing me |
| 05:14.38 | drmessano | "No, she can't be pregnant.. I have a BSD firewall" |
| 05:14.42 | drmessano | Etc |
| 05:14.48 | overyander | that should be on a meme |
| 05:14.54 | drmessano | me rolls eyes back into head |
| 05:15.01 | drmessano | overyander, yes it should be |
| 05:15.59 | overyander | speaking of firewalls... is f2b still the way to secure a box that has to be open to public for sip connections? or has some better technique come alonger? |
| 05:16.08 | overyander | s/alonger/along |
| 05:16.28 | ChannelZ | "secure a box" and "open to the public" does not compute |
| 05:16.34 | overyander | :P |
| 05:16.40 | overyander | relatively secure? |
| 05:16.42 | ChannelZ | but fail2ban is as good a choice as any to try and mitigate |
| 05:17.05 | ChannelZ | disconnecting China from the Internet would be a better way, but unfortunately.. |
| 05:17.14 | drmessano | overyander, http://i.imgur.com/R65tsLy.jpg |
| 05:17.33 | drmessano | overyander, fail2ban is a worthless piece of crap |
| 05:17.46 | drmessano | f2b doesn't secure anything |
| 05:18.05 | overyander | at work i wrote a script that alters the iptables of our * box and allows the external users ip through on udp after they authenticate on the vpn. i tail the firewall logs to determine that they're auth'd and get their public ip. it works like a charm |
| 05:19.01 | drmessano | fail2ban fails miserably during brute force attacks.. because it doesn't stop anything when the attack is fast enough to cripple Asterisk to the point that it doesnt log |
| 05:19.14 | drmessano | Which is actually far less than a total bandwidth exhaustion |
| 05:19.24 | drmessano | We've done some recent testing to support that |
| 05:19.33 | overyander | that makes sense. |
| 05:20.05 | overyander | so use something in iptables itself to trip after X many sessions within a certain timeframe? |
| 05:20.18 | drmessano | Yep, thats exactly the way to go |
| 05:20.44 | drmessano | Look for invites and register |
| 05:20.44 | overyander | i saw an example like that yesterday for a way to help secure ssh... let me see if i can find it |
| 05:21.25 | overyander | i'm not an iptables guru by any means, but i came across this https://wiki.centos.org/HowTos/Network/SecuringSSH look at the second code snippet in section 6 |
| 05:22.07 | drmessano | overyander, http://pastebin.com/7KTLyF3D |
| 05:22.10 | drmessano | Something like that |
| 05:22.12 | overyander | btw, drmessano, this was in #freebsd |
| 05:22.14 | overyander | <overyander> http://i.imgur.com/R65tsLy.jpg |
| 05:22.14 | overyander | <rennj> har har |
| 05:22.52 | moe` | ipfw on BSD has a connection limit directive that can be applied per source IP. "I have a BSD firewall". :) |
| 05:23.35 | overyander | moe`, you running this at your house or office? |
| 05:23.40 | drmessano | moe`, great, IF you know the source IP.. useless after the attack |
| 05:23.50 | moe` | its stateful |
| 05:24.01 | moe` | so the source IP is known, inherently. |
| 05:24.14 | drmessano | Ok, and what I just posted does the dame |
| 05:24.15 | drmessano | same |
| 05:24.17 | drmessano | In iptables |
| 05:25.16 | moe` | iptables pisses me off. :) That's just my BSDism coming through. |
| 05:25.58 | overyander | moe`, you running this at your house or office? |
| 05:26.07 | drmessano | Then overyander and the rest of us can use it, and you can keep telling everyone about your BSD Firewall |
| 05:26.17 | drmessano | *cough* troll *cough* |
| 05:26.26 | overyander | :/ |
| 05:26.49 | moe` | asterisk (voip.unixninja) is on a box at the colo where I work, but unixninja.net runs from my living room. |
| 05:27.04 | drmessano | It secures his entire basement |
| 05:27.43 | overyander | moe`, so how would you suggest those of us using vps' secure a public server behind a bsd fw? |
| 05:28.37 | overyander | my * vps is exposed directly to the internet... nothing i can do about that. i can however use the tools within my distro to harden the crap out of it. |
| 05:29.01 | drmessano | yum install bsd-firewall |
| 05:29.04 | drmessano | No, fails |
| 05:29.16 | drmessano | sudo apt-get install bsd-firewall |
| 05:29.18 | drmessano | nope |
| 05:29.21 | drmessano | :(((((((((((((((((((((((((((((((((( |
| 05:29.54 | overyander | lol, you scared me for a sec. i was thinking "yum install bsd-firewall ??? wtf??? " |
| 05:30.45 | moe` | ahha |
| 05:30.47 | overyander | drmessano, do you have a code snippet or link to an example on how to use iptables to look for the notify and registers as you were suggesting? |
| 05:30.47 | moe` | gold |
| 05:31.30 | drmessano | I already posted it |
| 05:31.35 | drmessano | overyander, http://pastebin.com/7KTLyF3D |
| 05:31.41 | drmessano | ^^^^ |
| 05:31.47 | overyander | ahh, missed that |
| 05:31.49 | overyander | thanks |
| 05:31.51 | drmessano | Also |
| 05:31.52 | drmessano | http://i.imgur.com/5lbiMuq.jpg |
| 05:32.00 | moe` | just updated my dialplans to ring all my devices at once, not rollover between them. |
| 05:32.04 | overyander | oh god |
| 05:32.10 | moe` | still, I'm stuck on that, how good |
| 05:32.56 | moe` | so yeah, two android phones, android tablet, netbook, and desktop. ringy dingy dingy all over the damned place. |
| 05:33.41 | overyander | drmessano, so that actually looks at the contents of the packet for those strings? |
| 05:33.50 | drmessano | overyander, yes |
| 05:34.10 | moe` | there's something about that, having a weed stash and a BSD firewall. |
| 05:34.12 | overyander | how badly does that effect performance (net and cpu) ? |
| 05:35.26 | drmessano | overyander, never noticed a hit.. i'm sure it's there, but i've taken massive attacks on tiny VPSs that I could easily wipe out with SIPp and didn't move the needle |
| 05:36.07 | overyander | nice |
| 05:36.22 | overyander | so, you use that instead of f2b? |
| 05:36.29 | overyander | or both? |
| 05:36.42 | drmessano | I've thrown like 100 calls per second at a t1.micro instance on Amazon and didn't see the CPU rise |
| 05:37.51 | drmessano | F2B is useless.. I DO see a CPU hit from all that log scanning, especially during an attack |
| 05:37.59 | drmessano | I havent run F2B in years |
| 05:38.21 | drmessano | Besides |
| 05:38.27 | drmessano | The logic is ALL wrong |
| 05:38.38 | drmessano | If I stop a SIP attack for even 60 seconds |
| 05:38.45 | drmessano | Most bots will go on their way.. gone forever |
| 05:38.57 | drmessano | That IP will most likely NEVER show up in my log again |
| 05:38.59 | drmessano | Why ban it? |
| 05:39.18 | drmessano | WHy not just drop packets from it for 60 seconds, let it go on, then quietly allow it again |
| 05:39.50 | drmessano | Seems like all those added rules would add CPU overhead.. thousands of banned IPs |
| 05:40.41 | overyander | i can see the logic in that. plus if you have a legit user trying with a bad password or something you don't have to admin the server to unban them |
| 05:40.44 | overyander | http://imgur.com/BJbJ2au |
| 05:41.30 | drmessano | HA |
| 05:41.59 | drmessano | Yeah, I totally agree from the bad password end too |
| 05:42.23 | drmessano | I chose the limits there due to the load on my VPS'ed Asterisk instances |
| 05:42.40 | moe` | ahha, BSD firewall for the win. |
| 05:42.49 | drmessano | That seemed like a MAX based on worst case from one IP, which was power loss, phones come back up, all try to register at the same time |
| 05:43.21 | drmessano | Even if they were too much, 60 seconds later they can try again |
| 05:43.29 | *** join/#asterisk elitas (~elitas@213.226.135.203) |
| 05:43.32 | overyander | which most phones will do |
| 05:43.40 | drmessano | With Fail2ban, you're fixing things |
| 05:44.07 | moe` | hey, speaking of phones.... :) .... what hardware SIP phones would you guys recommend? |
| 05:44.09 | drmessano | Too much work.. and I like to make things easy... better coding, better scripting = less work |
| 05:44.40 | moe` | I usually like cisco hardware for network goodies, but dunno about their phones |
| 05:45.48 | overyander | moe`, if you want a nightmare, go ahead and use cisco phones with * |
| 05:46.02 | overyander | i have had a really good experience with polycom's |
| 05:46.04 | drmessano | overyander, http://i.imgur.com/5lbiMuq.jpg |
| 05:46.16 | overyander | LOL |
| 05:46.24 | drmessano | Hang on |
| 05:46.38 | drmessano | That was the old one |
| 05:46.40 | drmessano | http://i.imgur.com/OR1KOQr.jpg |
| 05:46.41 | drmessano | There |
| 05:47.00 | overyander | nice! |
| 05:49.44 | overyander | drmessano, what's the purpose of iptables -A SIP -m hashlimit --hashlimit 6/sec --hashlimit-mode srcip,dstport --hashlimit-name tunnel_limit -j ACCEPT |
| 05:51.45 | drmessano | Overall rate limiting for SIP, per IP |
| 06:01.35 | *** join/#asterisk qdel (~Rholk@bny92-1-82-67-178-101.fbx.proxad.net) |
| 06:12.39 | overyander | drmessano, have you thought about modifying the iptables rule you shared to only send to that chain if the src network is within the US? |
| 06:13.41 | drmessano | No, but not a bad idea |
| 06:16.36 | overyander | i'm hollaring at the guys in #netfilter now to get their opinion on doing a subnet filter to only send US networks to the SIP iptables chain and to drop all other networks. that would really mitigate the amount of systems * is even exposed to |
| 06:17.13 | overyander | i have a list of the subnets in the US. if the iptables gurus don't think it's a bad idea, i'll make the tables and share it with ya |
| 06:18.40 | drmessano | That would be great! |
| 06:19.07 | *** join/#asterisk elijah_ (~elijah@174-26-43-54.phnx.qwest.net) |
| 06:21.03 | elijah_ | I've just built asterisk on Ubuntu 14.04 but when I run 'asterisk -vvvc' I get nothing back |
| 06:21.19 | elijah_ | It's like it's immediately dying but I can't see why, any ideas? |
| 06:24.24 | overyander | elijah_, have you checked the logs? |
| 06:25.29 | overyander | drmessano, guess how many subnets there are in the US? |
| 06:25.40 | overyander | 48,416 |
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| 06:30.40 | elijah_ | Which logs? |
| 06:30.54 | elijah_ | I'm running it with -c so all output should go to the console right? |
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| 06:32.48 | overyander | whichever logs you have configured in /etc/asterisk/logger.conf |
| 06:33.15 | overyander | you would think it is all going to console, the fact that it's not is very weird and i can't think of why it wouldn't... unless you didn't compile something correctly |
| 06:33.52 | overyander | i also don't use debian or ubuntu, so i have no ideas there. i use centos and fedora |
| 06:33.52 | elijah_ | Right now logger.conf just has console => notice,warning,error |
| 06:34.07 | elijah_ | Yeah, I'd think it would at least crash or something if it's built wrong |
| 06:34.18 | overyander | apparently it is crashing |
| 06:34.50 | elijah_ | Right, but with the -c switch shouldn't it report the error to the console? |
| 06:35.03 | elijah_ | I'm just trying to find something to go on |
| 06:35.31 | overyander | have you tried editing the verbosity and debug levels in /etc/asterisk/asterisk.conf and then just running asterisk -c ? |
| 06:36.17 | elijah_ | hmm no, let me see how to do that... |
| 06:36.28 | overyander | vim /etc/asterisk/asterisk.conf |
| 06:36.40 | overyander | edit verbose = |
| 06:36.44 | overyander | edit debug = |
| 06:36.54 | overyander | press escape key then :wq! |
| 06:37.01 | overyander | asterisk -c |
| 06:37.20 | elijah_ | verbose = 3? |
| 06:37.34 | overyander | higher |
| 06:37.40 | overyander | i have mine set for 99. LOL |
| 06:37.42 | elijah_ | 10? |
| 06:37.44 | overyander | the more info i get the better |
| 06:37.44 | elijah_ | oh haha ok |
| 06:38.10 | elijah_ | yeah still nothing |
| 06:38.21 | elijah_ | The log dir is set to /var/log/asterisk |
| 06:38.31 | elijah_ | I'm tailing the messages file in there but it's totally empty |
| 06:39.02 | overyander | did you by chance install from apt-get previously then try compiling from source? |
| 06:39.29 | overyander | did you read the install instructions or just try to do it based on typical make/make install logic? |
| 06:40.11 | elijah_ | I followed a writeup I found that was supposed to be fore ubuntu |
| 06:40.30 | overyander | how old was it? |
| 06:40.31 | elijah_ | no apt-get install for asterisk, some of the dependencies were installed that way |
| 06:40.41 | elijah_ | http://wiki.freepbx.org/display/HTGS/Installing+FreePBX+12+on+Ubuntu+Server+14.04+LTS#InstallingFreePBX12onUbuntuServer14.04LTS-InstallRequiredDependencies |
| 06:40.42 | overyander | was it specifically written for the version you tried compiling? |
| 06:40.48 | elijah_ | It was that, yeah for ubuntu 14.04 |
| 06:41.03 | elijah_ | I mean it all seemed pretty straightforward |
| 06:41.10 | overyander | :( |
| 06:41.12 | elijah_ | I'm just kinda surprised there is no error at all |
| 06:41.26 | overyander | that link is for installing freepbx, not asterisk |
| 06:41.40 | elijah_ | well yeah but I just used the parts that apply to asterisk |
| 06:41.44 | overyander | 2 completely different things |
| 06:41.51 | overyander | freepbx hacks the crap out of asterisk |
| 06:41.52 | elijah_ | freepbx uses asterisk tho... |
| 06:42.22 | elijah_ | They used the plain asterisk source for the part I followed, I didn't apply any patches or anything |
| 06:42.52 | elijah_ | there were a couple others I referenced, trying to find them |
| 06:42.55 | elijah_ | again |
| 06:43.19 | elijah_ | http://www.ipcomms.net/sample-device-configurations/41-asterisk/181-install-asterisk-13-on-ubuntu-debian |
| 06:44.09 | overyander | i would stick with official documentation |
| 06:44.11 | overyander | https://wiki.asterisk.org/wiki/display/AST/Installing+Asterisk+From+Source |
| 06:45.16 | elijah_ | Use version 11, not 13? |
| 06:46.33 | overyander | use the latest version |
| 06:46.44 | elijah_ | ok |
| 06:46.57 | elijah_ | I'll try over from the beginning I guess |
| 06:46.57 | overyander | 11 was used in the code examples because that was probably the latest out at the time. |
| 06:47.12 | overyander | the install instructions are pretty much the same |
| 06:47.16 | elijah_ | But this is seriously weird, I've never seen a program just silently crash like that |
| 06:47.25 | overyander | i would start by going into your source dir and running |
| 06:47.31 | overyander | 'make clean' |
| 06:47.50 | elijah_ | Yeah |
| 06:47.59 | overyander | there's also a bootstrap file you need to run first... not sure if that's mentioned in these docs or not |
| 06:48.05 | elijah_ | yeah I did run that |
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| 07:46.43 | overyander | drmessano, you there? |
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| 07:59.19 | _omer | Does ConfBridge work with all type of channels (SIP/IAX/DAHDI) ? I am planning to replace meetme with confbridge.... |
| 08:00.39 | overyander | confbridge is a an application just like dial,meetme,moh,voicemail, etc. the channel doesn't matter |
| 08:06.23 | _omer | thanks overyander |
| 08:06.51 | overyander | yw |
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| 10:29.07 | aiksa[LV] | hey; anyone here having a bit of experice with ARI? |
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| 10:29.59 | aiksa[LV] | how do I track status of originated call before it has entered stasis application? |
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| 11:35.57 | neeby_goosey_X | Hello everyone. Just wanted to ask if it possible for asterisk to Pickup() inbound channels. Seems like it works for outbound only. |
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| 12:27.40 | [TK]D-Fender | neeby_goosey_X_: this is for "ringing" channels. "outbound" is not a thing |
| 12:29.35 | neeby_goosey_X_ | I tried picking channel that was running `exten => s,1,Ringing();` dialplan with no success. |
| 12:31.31 | [TK]D-Fender | You should probably show us then |
| 12:33.31 | neeby_goosey_X_ | http://pastebin.com/Hg78QmNZ |
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| 12:34.48 | [TK]D-Fender | And the call |
| 12:36.10 | Vinel | Hey [TK]D-Fender. |
| 12:36.42 | [TK]D-Fender | exten => s,1,Set(__PICKUPMARK=1) <- And I'd avoid trying to throw inheritance into this right now... |
| 12:38.43 | neeby_goosey_X_ | Why? Adding inheritance almost never breaks anything. This very line is there because of commented out Dial() command. |
| 12:39.03 | neeby_goosey_X_ | (dial command not shown in my sample) |
| 12:39.14 | [TK]D-Fender | lets see your test |
| 12:39.42 | neeby_goosey_X_ | Will be back in several mins, gotta download those logs. |
| 12:49.17 | neeby_goosey_X_ | Will this log fragment do? |
| 12:49.19 | neeby_goosey_X_ | http://pastebin.com/Umm3CBUS |
| 12:52.39 | neeby_goosey_X_ | Could you explain what was meant by 'ringing channel'? I thought only devices and extensions could have ring statuses. |
| 12:55.19 | neeby_goosey_X_ | Pickup documentation doesn't state anything about types of channels it can deal with either. |
| 12:56.06 | [TK]D-Fender | Not sure if that kind of "ringing" counts" |
| 12:57.05 | WIMPy | How many kinds of "ringing" do you know? |
| 12:57.30 | [TK]D-Fender | passed status from a dial, etc |
| 12:57.39 | neeby_goosey_X_ | [TK]D-Fender: Oh, my bad, pickup docs do state 'ringing channels'. So, how do I make a channel a 'ringing' one? |
| 12:57.54 | [TK]D-Fender | Being in some kind of actual call. |
| 12:57.59 | [TK]D-Fender | Dial, possibly Queue, etc |
| 12:58.25 | WIMPy | It's in ringing state while a device is ringing. |
| 12:59.28 | neeby_goosey_X_ | My example works if I add Dial or Queue to `pickup-seed`, I already tested that. It surprised me that it didn't work without at least one channel creating command. |
| 12:59.57 | neeby_goosey_X_ | Can I somehow fool ast into believing that my channel is ringing? |
| 13:00.14 | neeby_goosey_X_ | DEVICE_STATE()? |
| 13:00.28 | WIMPy | You could try Ringing(). |
| 13:00.29 | [TK]D-Fender | Maybe you should think of another approach and tell us what you are actually trying to do |
| 13:00.44 | [TK]D-Fender | WIMPy: He already did, that's what his test showed fails. |
| 13:00.51 | WIMPy | But once te channel is "up" it's too late. |
| 13:01.07 | WIMPy | ok |
| 13:03.57 | [TK]D-Fender | The extension is typically set on matching channels by the dial application that created the channel. The context is set on matching channels by the channel driver for the device. |
| 13:04.06 | [TK]D-Fender | instructions do infer Dial() specifically |
| 13:05.51 | neeby_goosey_X_ | My goal was to allow users to pick up calls from queue. I solved it by setting PICKUPMARK just before routing inbound calls to the queue. Users then invoke Pickup(<smth>@PICKUPMARK) and get *some* call from the queue. The sample I showed you was my initial attempt to figuring out how Pickup() command works. |
| 13:06.47 | [TK]D-Fender | So it does work with Queue() then? |
| 13:07.17 | Vinel | What choice do I have to create a distributed queue for outbound calls? For instance, 200 agents on Asterisk box 1 and other 200 agents on other Asterisk box but shared same queue? |
| 13:07.32 | WIMPy | I think it works with the Dial executed by Queue. |
| 13:07.48 | neeby_goosey_X_ | Yes, I think so. Although now I gotta test the case with multiple channels having the same PICKUPMARK, such case is not documented anywhere. |
| 13:09.04 | [TK]D-Fender | neeby_goosey_X_: You can bet on it being one of either the oldest, newest, or morest recently changed status that gets grabbed. |
| 13:09.20 | [TK]D-Fender | neeby_goosey_X_: or perhaps alpha-based |
| 13:09.33 | [TK]D-Fender | neeby_goosey_X_: Wouldn't bet on queue order mattering |
| 13:09.33 | WIMPy | Or just random. |
| 13:09.56 | [TK]D-Fender | Pretty sure it wouldn't be actually random. Just not sanything proactical. |
| 13:10.00 | neeby_goosey_X_ | [TK]D-Fender: It would do if it worked any way) |
| 13:10.50 | neeby_goosey_X_ | [TK]D-Fender: Happily, I don't need the queue ordering influence my Pickup(); |
| 13:17.22 | neeby_goosey_X_ | Thanks for suggestions. |
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| 14:11.41 | Vinel | Asterisk RealTime Extensions, it I create a new Extention in one of Asterisk box, can it be available automatically in other Asterisk boxes? |
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| 15:56.53 | zaf | anyone familiar with digium phones and smart_blf? now matter what i try, the phones don't even seem to attempt to subscribe to other peers' status |
| 15:58.35 | zaf | my xml files are very close to the examples in the wiki |
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| 17:22.21 | reveal | how can i remove unathorized attempts for asterisk from people trying to connect to external ip |
| 17:42.25 | WIMPy | zaf: Have you configured Asterisk correctly? Do you have hints and are the phones allowed to subscribe? |
| 17:42.49 | WIMPy | reveal: Remove? From where/what? |
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| 17:51.53 | zaf | WIMPy, as far as I can tell they are, the hints show up correctly afaict. but i'm not seeing any attempts at all to subscribe in sip debug |
| 17:52.05 | zaf | other than them subscribing to themselves |
| 17:52.23 | WIMPy | To themselves? |
| 17:53.47 | zaf | i see a SUBSCRIBE from the phone with a destination of its own extension |
| 17:54.16 | WIMPy | Voicemail? |
| 17:59.20 | zaf | i'm not sure why they subscribe to themselves, maybe voicemail |
| 18:00.34 | zaf | i really think it's something with the configuration of the phones that i'm missing |
| 18:01.01 | WIMPy | There isn't much to configure. |
| 18:01.06 | WIMPy | Or much that can be configured. |
| 18:01.58 | zaf | i get the phonebook, but i get no blinky lights |
| 18:02.54 | WIMPy | <contact first_name="conf" last_name="0" subscribe_to="11990"><numbers><number dial="11990" /></numbers></contact> |
| 18:03.14 | WIMPy | That's an example of what works for me. |
| 18:04.14 | WIMPy | Hmmm. Do you have the contacts in the "blf" group? |
| 18:04.42 | zaf | i have the group name as "Directory", if that matters |
| 18:04.57 | WIMPy | I think it does. |
| 18:05.03 | zaf | i haven't seen any references to the name of the group having any significance |
| 18:05.07 | zaf | but i'll try changing it |
| 18:05.42 | WIMPy | I can't relly remember, but it would seem to me lie that's not a name I came up with. |
| 18:07.14 | zaf | no change.. |
| 18:07.31 | zaf | do you have something else in your files that references a group named blf? |
| 18:07.54 | WIMPy | SYS "Wimp_CommandWindow",-1QUIT |
| 18:07.59 | WIMPy | oops |
| 18:08.23 | WIMPy | Actually, yes. |
| 18:08.33 | WIMPy | <setting id="blf_contact_group" value="blf" /> |
| 18:11.00 | zaf | you're not using DPMA, right? |
| 18:11.14 | WIMPy | correct |
| 18:15.29 | zaf | dammit, no change |
| 18:18.18 | reveal | WIMPy: connections i see that are 401 from CLI |
| 18:18.51 | reveal | WIMPy: http://pastebin.com/0c0fberH |
| 18:19.28 | WIMPy | Turn off debug. |
| 18:19.52 | WIMPy | If you have a service reachable from the internet, people will try to (ab)use it. |
| 18:20.22 | reveal | i see |
| 18:20.52 | WIMPy | Some people recommend using fail2ban to limit the number of attempts. |
| 18:21.06 | reveal | ive been also using iptables |
| 18:23.32 | zaf | WIMPy, that's smart_blf right? |
| 18:24.04 | WIMPy | I don't think so. |
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| 19:09.58 | fornax | Hi all. I have problem where I really need help since I have no idea how to debug it. I'm quite familiar with asterisk but am lost at this moment. |
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| 19:10.43 | [TK]D-Fender | ~ask |
| 19:10.43 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 19:11.27 | fornax | give me a second to formulate the answer, I do not want to flood the channel and so write in small pieces... |
| 19:11.44 | fornax | When I make an outbound call from my gentoo asterisk (11.17.1), the called number rings, but when the other person answers, the call is terminated |
| 19:12.20 | fornax | On my side I have no sound and it looks like nothing happens. The line did not hang up |
| 19:12.25 | WIMPy | Nothing unusual so far. How do you call? |
| 19:12.38 | fornax | When I now call again, the call is established correctly |
| 19:13.22 | [TK]D-Fender | Show us the call |
| 19:13.24 | [TK]D-Fender | * CLI |
| 19:13.28 | [TK]D-Fender | "sip set debug on" |
| 19:13.28 | fornax | I have a fritzbox which cannot be jailbraiked and I register the individual fritz box lines as trunks |
| 19:13.29 | [TK]D-Fender | ~pb |
| 19:13.30 | infobot | A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 19:13.32 | [TK]D-Fender | ^^^ |
| 19:13.53 | [TK]D-Fender | your FB should not be touching those accounts. They could be fighting with * over them |
| 19:14.01 | [TK]D-Fender | and you would be causing yoruself issuez |
| 19:14.09 | WIMPy | Since when do you have to jailbreak (or whatever) a Fritzbox? |
| 19:14.11 | fornax | I enabled debug and can send you the file. Unfortunately after the first unsuccessful call everything works perfect, so it is hard to reproduce... |
| 19:15.06 | fornax | WIMPy I would like to jailbeak the box to get the credentials to directly register the trunks with o2 instead of using the fritzbox to register at o2 and use asterisk to register at the fritzbox... |
| 19:15.08 | robmal | Set shorter registration time on the phone. |
| 19:16.33 | WIMPy | Either download te config and get a decoder for it or log in and use the provided commands to display the configuration. |
| 19:18.33 | fornax | Give me some moment, the debug file is quite big and I want to find the part where the call occured |
| 19:18.41 | fornax | which is your recommended paste service? |
| 19:23.21 | fornax | Here is the log: http://pastebin.com/zAB4SqEe |
| 19:25.32 | fornax | When you seachr for Executing you see the call, which worked. The call before did not even executethe call, the line simply hung up |
| 19:26.15 | [TK]D-Fender | Dialing OUT your FB? |
| 19:26.42 | fornax | Yes, my phone ist registered to asterisk, and asterisk uses a line of the fritzbox to dial out |
| 19:26.55 | [TK]D-Fender | What kind of line? |
| 19:27.07 | WIMPy | Which seems to succeed. |
| 19:27.30 | [TK]D-Fender | It does get ack'd then answered |
| 19:27.49 | fornax | [TK]D-Fender It is a registered with "register" in my sip.conf |
| 19:27.53 | [TK]D-Fender | media address seem sane as well |
| 19:28.07 | [TK]D-Fender | fornax: No, I asked WHAT it is using on the FB to dial out. |
| 19:28.51 | fornax | The fritz box is using voip to register at o2, or what do you mean? |
| 19:30.31 | fornax | [TK]D-Fender You get a preconfigured FritzBox from o2, which registeres via DSL and VoIP at O2. You can then use your sip phones to register at your fritzbox or plug in your isdn or analog phones. But the box bridges to VoiP, I think they use SIP, but you cannot hack the box anymore to get more information |
| 19:30.51 | [TK]D-Fender | Ok, it passes the call call through to your provider. |
| 19:30.59 | [TK]D-Fender | as a B2BUA |
| 19:31.09 | [TK]D-Fender | So was this call you showed a FAILURE? |
| 19:31.26 | fornax | [TK]D-Fender The first call did not work, the second did work |
| 19:31.48 | WIMPy | They use a FB with special F |
| 19:31.49 | fornax | [TK]D-Fender It is always the first call that fails. When you then call a second time, it works without problems |
| 19:31.55 | WIMPy | oops |
| 19:31.57 | [TK]D-Fender | User-Agent: Sinus 501V021890000000 <- first call, this side terminated without error |
| 19:32.00 | WIMPy | They use a FB with special firmware? |
| 19:32.19 | fornax | WIMPy Yes, they hava a custom firmware and no way to reflash ist or change it |
| 19:32.41 | WIMPy | There's always a way. |
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| 19:33.19 | fornax | WIMPy The VoiP functionality is provisioned via O2. Some time ago there was some way, but when I got the box, the Fritz OS has been upgraded and currently I do not think there is a real way |
| 19:36.10 | fornax | WIMPy I can not find any error. But when you call the first time, nothing happens. Do you have an other idea how to debug it? |
| 19:36.26 | WIMPy | Luckily it looks like that practice will soon become illegal. |
| 19:37.00 | WIMPy | I don't see anything go wrong. So I can't say. |
| 19:37.25 | fornax | WIMPy Absolutely, and I really look forward to it. Probably an rtp problem? |
| 19:37.50 | WIMPy | I don't see any problem. |
| 19:38.02 | WIMPy | Maybe your Sinus has some sort of issue? |
| 19:38.03 | fornax | WIMPy Or something goes into some standy after a while and need a call to wakeup first? |
| 19:39.13 | fornax | WIMPy I think I can really easy test this. And I think I already tested it. I have a CISCO phone that is connected via sccp, and I think it did not work either. This means the problem needs to be on the trunk side to the fritzbox |
| 19:41.49 | fornax | WIMPy Okay, this worked, but that does not say anything since it takes quite some time until it stops working |
| 19:45.24 | fornax | WIMPy The interesting thing is that there also seems to be a problem with echo cancelling. But this does not really seem to be related. |
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| 19:47.19 | mutilator | is there any built in mechanism to rotate MOH messages? "if you're having an issue with X, please try Y" |
| 19:48.35 | zaf | mutilator, you can have multiple files, all that's edited in musiconhold.conf |
| 19:49.08 | fornax | [TK]D-Fender Probably I should try another sip hardware and check if this happens too. But I think it should be related to the new fritzbox voip trunks. Before I had a quadbri to call to the outside and this worked. With the FritzBox I now have the problem. But it does not explain why my sip hone terminated without error, but the phone does not signalize that the call has ended |
| 19:50.06 | mutilator | yea but nothing out of the box to handle adding/editing that |
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| 19:53.28 | zaf | mutilator, you could write a script to move files around |
| 19:54.01 | mutilator | yea |
| 19:54.01 | mutilator | k |
| 19:54.14 | mutilator | prob just go with one of these streaming services |
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| 21:50.07 | malcolmd | re: digium phones. to make sure subscriptions are happening, the contacts file group_name needs to be the same as the phone config's blf_contact_group. plz to ask Support (digium.com/support) if you're running into issues. :) they're able to be much more responsive than i am. |
| 21:51.30 | zaf | malcolmd, yeah, had a ticket open since friday, but they didnt seem to be too familiar with the xml |
| 21:54.14 | malcolmd | basic concept isn't that different from how it's done in DPMA land. in DPMA-land contacts files and blf_items files are still just XML files. in DPMA they're pulled in via specific key-pair values for the type=phone. in xml land they're declared the phone's xml config file. |
| 21:55.02 | zaf | yeah, that's what i've learned in the past few hours, heh |
| 21:59.23 | zaf | <phonebooks> |
| 21:59.23 | zaf | <PROTECTED> |
| 21:59.38 | zaf | <PROTECTED> |
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| 22:25.00 | HurricaneHernand | Just wondering if anyone in here is using Yealink phones |
| 22:26.36 | HurricaneHernand | And perhaps if they could help me with multicast paging, I can't seem to get it to work |
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