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00:12.29 | lost_soul | jblack: Merry Christmas |
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00:49.18 | bcrisp | ho ho ho |
00:50.54 | bcrisp | echo |
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01:01.35 | bcrisp | uh oh, p3nguin has competition... |
01:03.31 | *** join/#asterisk gme30066 (n=gme@173.160.69.30) |
01:05.44 | gme30066 | hello! are there any asterisk iptables gurus around? |
01:06.08 | bcrisp | gme30066, got a Q? |
01:06.26 | gme30066 | Yeah, would like to find out if these rules make any sense: http://skitch.com/gellenburg/nc6ck/firewall-rules |
01:07.25 | bcrisp | gme30066, im pretty sure there are some articles posted on suggested settings for * |
01:07.29 | bcrisp | iptables |
01:07.48 | bcrisp | i.e. open hole for manager if you use it |
01:08.12 | bcrisp | gme30066, check this out: http://www.voip-info.org/wiki/view/Asterisk+firewall+rules |
01:08.17 | gme30066 | i'm allowing everything from my internal lan. only want to open what's req'd for asterisk from external |
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01:09.44 | gme30066 | k... hmmm. with the exception of getting iax & iax2 confused everything is pretty much the same then. hmm. i wonder why the system keeps freezing then when i login. |
01:09.45 | bcrisp | gme30066, its gonna depend on what types of interfaces you need etc |
01:09.57 | bcrisp | freezing? |
01:11.25 | gme30066 | yeah. PIAF. with iptables disabled everything is rock solid, with those rules I listed above enabled every once in a while all soft phones and hard phones drop, my trunks drop, and everything freezes for about 90 secs and then comes back so I'm assuming it's firewall related (since nothing happens with /etc/init.d/iptables stop) :-) |
01:11.41 | bcrisp | oh thats strange |
01:11.56 | bcrisp | you have hardware firewall set up? |
01:12.29 | gme30066 | no. my network setup doesn't allow for a hardware firewall for this specific instance. |
01:12.45 | bcrisp | what kind of environment? |
01:12.53 | bcrisp | are you using EC2? |
01:14.04 | gme30066 | way too complicated to go into. it involves comcast (it's craptastic!), VMs (but no EC2), bridged routing, RFC1918 addresses, and publicly routable addresses, and a /28 network. |
01:14.21 | bcrisp | holy crap |
01:14.26 | bcrisp | thats a lot of variables |
01:14.38 | bcrisp | could it be nat-related? |
01:14.40 | gme30066 | nah. i've got things trimmed down. |
01:14.50 | gme30066 | eth0 is publicly routable. |
01:16.07 | gme30066 | It's default-gw is the gateway of the router. |
01:16.24 | bcrisp | interesting, id suggest asking some of the veterans here like leifmadsen |
01:16.39 | bcrisp | im pretty new and not a networking expert by any means |
01:18.59 | gme30066 | that's why i'm here. ;-) |
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01:31.23 | fofware | Merry Christmas every Body!!!!!!!!! |
01:37.23 | p3nguin | So what was the question about networking? |
01:37.37 | p3nguin | Hoping for a condensed version. |
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01:41.06 | gme30066 | :-) Do these rules make sense? http://skitch.com/gellenburg/nc6q2/webmin-1.500-on-sip.ellenburg.net-centos-linux-5.4 |
01:44.11 | p3nguin | The SIP range doesn't make sense to me; I think it only needs port 5060. The IAX and IAX2 ports don't make sense, because as far as I know, IAX2 uses 4569. |
01:44.18 | p3nguin | Other than that, it seems okay. |
01:44.58 | p3nguin | IAX 1 used 5036. |
01:45.32 | gme30066 | i'm not an iptables expert, but from what you can tell that only allows those ports right? nothing else will be allowed through am I reading that correctly? |
01:45.53 | bcrisp | p3nguin, whats odd is he says it will work fine then all of a sudden everything drops |
01:45.54 | p3nguin | That's what it seems, but unless you show me iptables -L -nv, I can't say for sure. |
01:46.16 | bcrisp | doesnt sound like an iptable issue to moi |
01:47.19 | gme30066 | iptables -L -nv: http://skitch.com/gellenburg/nc6xb/terminal-ssh-bash-basic-ttys001-129x22-2 |
01:48.51 | p3nguin | This will only allow those ports to connect to the machine where you ran that command. |
01:49.18 | jblack | I'm lonely and bored and a little sad. |
01:49.23 | jblack | Let's all go mug santas |
01:49.24 | p3nguin | The INPUT policy is DROP, so only the accepted ports will be allowed. Everything else will be ignored. |
01:49.37 | bcrisp | jblack: go rent bad santa |
01:49.57 | jaytee | billy bob? |
01:50.03 | jblack | I don't feel a need to justify mugging santa. I just want to mug 'em. |
01:51.37 | p3nguin | And since the OUTPUT chain's policy is ACCEPT, the two rules you have are not doing much good... and nothing originating on the machine where the command was entered will be blocked when going out. |
01:52.46 | gme30066 | do you think it's secure? |
01:53.27 | p3nguin | I also typically enter a RELATED,ESTABLISHED rule to help keep state. I realize we are dealing with UDP, but iptables is still capable of keeping states of connectionless protocol connections. |
01:53.35 | p3nguin | "connections," sorry |
01:53.54 | jblack | can't remember if there is a sip awareness module. |
01:54.02 | gme30066 | related, established? |
01:54.32 | p3nguin | iptables -I INPUT -m state --state RELATED,ESTABLISHED -j ACCEPT |
01:55.01 | p3nguin | The security of Asterisk doesn't really depend on the firewall. |
01:55.16 | gme30066 | does it matter if I'm not NAT'ed? |
01:55.22 | p3nguin | Yes. |
01:55.54 | p3nguin | Because if the asterisk machine is behind NAT, your firewall rules should be at the NAT server. |
01:56.04 | gme30066 | That related, established rule I assume that should go at the bottom? |
01:56.27 | gme30066 | (yeah the box isn't nat'ed so I guess it's right that the rules are actually sitting on the * box itself.)? |
01:57.30 | gme30066 | I see something here about mod conntrack: http://www.calivia.com/iptables-sip-conntrack-nat but that looks to be only for if you're natting your asterisk box. |
01:58.54 | p3nguin | If the * box is sitting on the internet with a public IP address, then your rules are pretty good (except I still don't understand why you have a range for SIP, when SIP is port 5060). |
01:59.39 | bcrisp | you just need to make sure SIP and media |
02:00.05 | gme30066 | i saw something somewhere that had SIP as a range of IP addresses. I'll just make it 5060. |
02:00.12 | gme30066 | Not using the media. |
02:00.19 | jblack | gme30066: rtp.conf ? |
02:00.37 | p3nguin | You'll also want to make sure your rtp.conf ports match up to your range of 10000-20000. |
02:01.05 | gme30066 | no, it was from a google search looking for the ports asterisk uses for firewalls |
02:01.05 | jblack | you probably want to knock that down.. unless you actually think you could have 5,000 calls at once. |
02:01.07 | p3nguin | That's the typical port range, by the way. |
02:01.59 | p3nguin | For RTP, you need two ports per call, so you can figure it out by how many calls you expect to have at any given time. |
02:02.06 | gme30066 | i'm not worried about the range so much, more about someone finding a back door and getting in through the public IP. there's other back doors for sure. |
02:02.31 | jblack | then tighten that up. |
02:04.10 | gme30066 | i've got a /28 from Comcast, and the way I've got this SMC D8014 configured the gateway address is acting as the NAT for all the RFC1918 addresses (workstations and internal servers) on my network. The rest of the public IPs are for servers and services since I actually can't do any NAT'ing through this router anymore (i'd have to give up my default-gw NAT). |
02:05.22 | gme30066 | rest of my servers are running FreeBSD, Solaris, or Mac OS X. IPFilter I know. I'm a complete n00b when it comes to iptables. |
02:07.08 | p3nguin | Add the rule that I gave you earlier, as well as alter the SIP port, and you should be fine on port configuration. |
02:07.34 | gme30066 | that should be the last rule then? |
02:07.45 | gme30066 | that related, established one? |
02:08.35 | p3nguin | Nah, I would probably put it long about the 1st. |
02:09.11 | p3nguin | In my own configuration, I first allow localhost, then allow icmp, then set the related established, then the rest. |
02:10.00 | p3nguin | I mean, you can put it where ever you want it. If you feel like it is better to have it later in the list, put it last. |
02:12.41 | p3nguin | If you determine that it is better or more effective to put it later, let me know. |
02:14.29 | gme30066 | I think I just found my hanging problem. |
02:15.25 | gme30066 | I was allowing outbound DNS queries, but there wasn't a rule for the subsequent reply from the DNS server for lookups. |
02:16.40 | p3nguin | related,established would cover that. |
02:16.58 | p3nguin | since it would be "related" and all. :) |
02:17.15 | gme30066 | yeah I suppose so. :-) I've gone ahead and added implicit rules though j/i/c |
02:17.40 | p3nguin | I don't agree with that being the correct way to handle it. |
02:17.56 | gme30066 | the implicit rules? |
02:18.00 | p3nguin | yeah |
02:18.16 | p3nguin | I feel that the correct way is to add the related,established rule. |
02:18.47 | gme30066 | i see your point. the implicit rule opens up another hole. |
02:22.40 | gme30066 | sumbitch this is actually starting to come together and work finally. :-) |
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03:34.54 | phix | hey hey |
03:34.57 | phix | what's new |
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04:03.29 | NateHB | Anyone here familiar with the edgemarc? |
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05:02.14 | Katty | drags in |
05:02.34 | Katty | plops down |
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05:12.23 | bcrisp | :/ |
05:26.44 | ReDNeQ | merry xmas |
05:34.15 | carrar | HO HO HO |
05:34.19 | carrar | Marry Christmas! |
05:37.40 | coppice | is that what Mrs Claus Christmass did? |
05:38.13 | TJNII | I think we should just be friends.... |
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07:58.35 | shamelessn00b | hey all |
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08:15.26 | ChannelZ | hi ho |
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09:00.41 | shamelessn00b | ChannelZ: sup dude |
09:00.46 | shamelessn00b | merry christmas |
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09:47.39 | rare1980 | hi every one |
09:49.10 | rare1980 | helloooooooo |
09:49.23 | rare1980 | is any one online.. in this group?????????????? |
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10:22.29 | tzafrir_laptop | rare1980 is not on-line in this group |
10:52.38 | shamelessn00b | guys |
10:52.51 | shamelessn00b | I have this setup, 4 PRIs on 2 machines |
10:53.01 | shamelessn00b | (Total 8 PRIs) |
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10:53.22 | shamelessn00b | I want to initiate 120 calls from 1 machine simultaneously that will land on the other machine |
10:53.32 | shamelessn00b | how can it be done |
10:53.51 | shamelessn00b | also, are there some scripts to stress test these systems |
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12:22.50 | fenrus | shamelessn00b, use sipp to stress-test SIP.. :) |
12:23.10 | fenrus | shamelessn00b, sipp.sf.net |
12:23.38 | shamelessn00b | what about pri calls |
12:27.32 | shamelessn00b | DAHDI |
12:27.33 | shamelessn00b | calls |
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12:34.02 | rare1980 | hi |
12:34.03 | rare1980 | all |
12:35.28 | rare1980 | hellooooo any one alive in this roooooooom |
12:35.28 | rare1980 | ? |
12:42.32 | fenrus | rare1980, yea? |
12:42.43 | fenrus | shamelessn00b, dunno :) |
13:22.48 | tzafrir__laptop | infobot, tell rare1980 about ask |
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13:23.53 | rare1980 | HI FENRUS |
13:23.54 | rare1980 | u still there? |
13:24.57 | fenrus | rare1980, yes. |
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13:28.01 | rare1980 | i want to that how can we configure |
13:28.07 | rare1980 | predictive dialer in asterisk |
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13:29.56 | tzafrir__laptop | rare1980, any specific predictive dialer? |
13:36.06 | fenrus | i've no clue about that.. but i guess you'll just have to read up about it |
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13:50.52 | benngard | any1 here that have tested avaya 9650 r2.5? |
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14:36.35 | mazpe | how can i setup MeetMe to have a conference in Listen Only mode exepct for admin or marked users? |
14:36.44 | mazpe | everyone else who joins should be muted. |
14:37.02 | mazpe | users should be muted at all times. |
14:41.25 | [TK]D-Fender | mazpe: "core show application meetme" <- |
14:41.48 | [TK]D-Fender | mazpe: all of the options are right there.... |
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14:57.49 | grandpapadot | Merry X-Mas peeps. |
15:02.54 | ariel_ | Morning folks, hope everyone has a great Merry Christmas |
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15:43.31 | drmessano | Merry Asterisk |
15:50.27 | grandpapadot | lol |
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16:22.59 | bcrisp | merry christmas |
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17:08.31 | teknoprep | does anyone know how to do a tcpdump so i can analyze it in wireshark ? |
17:08.39 | teknoprep | i forget how to dump it off to pcap |
17:15.38 | mazpe | tcpdump -i xl0 -s 1500 -w file.name |
17:16.47 | dlynes | teknoprep: btw |
17:17.01 | teknoprep | yo |
17:18.02 | dlynes | teknoprep: a better solution is sipdumppcap or something like that |
17:18.23 | dlynes | teknoprep: it only dumps sip convos, and dumps each convo into a separate pcap file |
17:18.41 | teknoprep | where would i pick tha tup at ? |
17:19.36 | dlynes | oh yeah... pcapsipdump |
17:20.07 | dlynes | http://pcapsipdump.sf.net/ |
17:20.52 | dlynes | teknoprep: default mode is to create a subdirectory and dump each conversation into a separate pcap file |
17:21.02 | dlynes | teknoprep: it ignores any other type of traffic |
17:21.17 | dlynes | teknoprep: it also dumps all the rtp traffic into the same file |
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17:21.34 | teknoprep | well thats nice |
17:23.04 | teknoprep | i love wireshark |
17:23.08 | teknoprep | decoding call's is fun |
17:23.20 | teknoprep | too bad it doesn't do G729 |
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17:28.54 | teknoprep | yeah but wireshark seperates the calls for you |
17:29.02 | teknoprep | why would i need it to be done at the file level ? |
17:31.03 | tzafrir__laptop | teknoprep, mazpe: tcpdump -s0 instead of -s<whatever> - just grab everything |
17:31.27 | teknoprep | ok |
17:31.31 | teknoprep | -s0 ? |
17:31.44 | tzafrir__laptop | just gets everything. |
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17:41.29 | wwalker | I just enabled jitterbuffer and did a sip reload but "sip show settings" doesn't show any jb stuff. how can I see the jitterbuffer settings? |
17:41.56 | wwalker | (doing SIP/RTP over a high loss network bytes) |
17:44.20 | Pan3D | merry christmas |
17:45.10 | wwalker | Pan3D: merry christmas |
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19:21.59 | Nugget | http://thereifixedit.com/2009/12/25/a-tower-ing-christmas-tree/ |
19:23.15 | *** join/#asterisk GameGamer43 (n=GameGame@CPE-65-27-76-78.new.res.rr.com) |
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19:42.36 | Akiraa | What's the difference between "public SIP" and "private SIP"? |
19:42.46 | hardwire | accessability |
19:43.14 | Kobaz | mmm |
19:43.32 | Kobaz | still getting random crashes with sip on 1.6.0.19 |
19:43.55 | *** join/#asterisk denon (n=denon@synapse.subneural.net) |
19:43.57 | *** mode/#asterisk [+o denon] by ChanServ |
19:44.04 | Akiraa | hardwire: as in, private SIP would sit on the LAN, while public SIP on a WAN? |
19:44.18 | *** join/#asterisk denon (n=denon@synapse.subneural.net) |
19:44.18 | *** mode/#asterisk [+o denon] by ChanServ |
19:44.25 | hardwire | Akiraa: as in a private website would be accessable to whomever has access. |
19:44.32 | hardwire | same with sip |
19:44.41 | *** join/#asterisk denon (n=denon@synapse.subneural.net) |
19:44.41 | *** mode/#asterisk [+o denon] by ChanServ |
19:45.22 | hardwire | Akiraa: why do you ask? I never refer to them in that way |
19:45.25 | *** join/#asterisk denon (n=denon@synapse.subneural.net) |
19:45.25 | *** mode/#asterisk [+o denon] by ChanServ |
19:45.44 | hardwire | denon: your system.. is down.. your system.. is down.. |
19:45.51 | Akiraa | hardwire: some internal settings in a few phones I have ATM |
19:45.58 | hardwire | Akiraa: which? |
19:47.48 | Akiraa | hardwire: http://www.5111soft.com/productview.php?pid=7&id=1 -- the cheapest voip phones I could find, really |
19:49.13 | hardwire | hah |
19:49.17 | hardwire | yeh.. they over engrished |
19:49.25 | hardwire | Private Register/Private Proxy |
19:49.44 | hardwire | and then Enable Private Register |
19:49.58 | hardwire | I can see how that would be weird |
19:50.13 | denon | looks at hardwire |
19:50.22 | *** join/#asterisk Tim_Toady (n=moi@77.49.123.166.dsl.dyn.forthnet.gr) |
19:50.37 | hardwire | denon: you've disconnected/reconnected at least 10000000 times |
19:50.49 | hardwire | ok so maybe just 5 |
19:50.51 | hardwire | :P |
19:50.59 | hardwire | just thought I'd let you know :P |
19:51.05 | denon | hm, maybe someone's jackin with my irc socks |
19:51.09 | denon | sorry bout that |
19:52.31 | hardwire | Akiraa: http://www.5111soft.com/largerview.php?id=18 |
19:52.34 | hardwire | that thing is crazy |
19:52.37 | *** join/#asterisk denon (n=denon@synapse.subneural.net) |
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19:53.00 | *** join/#asterisk denon (n=denon@synapse.subneural.net) |
19:53.00 | *** mode/#asterisk [+o denon] by ChanServ |
19:53.23 | *** join/#asterisk denon (n=denon@synapse.subneural.net) |
19:53.23 | *** mode/#asterisk [+o denon] by ChanServ |
19:53.38 | Akiraa | hardwire: crazy how? |
19:55.05 | Katty | ferretcam is on. |
19:55.23 | hardwire | crazy crazy |
20:00.28 | *** join/#asterisk debuggerboy (n=anish@117.196.163.239) |
20:02.01 | Katty | boyfriend is cranky today |
20:02.20 | gme30066 | Hi! Is anyone familiar with IPKall? I've got an inbound route setup for a specific SIP URI but I'm getting "The number you have dialed is not in service" when I attempt to dial the IPKall DID that was given to me. Any thoughts? :-) |
20:04.07 | hardwire | Katty: kick him in the nuts |
20:04.14 | Katty | that's just make him more pissy |
20:04.20 | hardwire | gme30066: kick them in the nuts |
20:04.20 | *** join/#asterisk vitaminx (n=vitaminx@89.130.31.1) |
20:04.41 | gme30066 | :sigh: |
20:05.06 | hardwire | how are you originating the call? |
20:05.24 | Katty | hardwire: he has his headphones on, with this scowl on his face. |
20:05.32 | Katty | hardwire: and i know when a song goes off, cause it's that loud... |
20:05.43 | Katty | hardwire: but he won't even talk to me when the song goes off. |
20:06.00 | hardwire | aww |
20:06.10 | hardwire | don't make me repeat myself |
20:06.12 | hardwire | :P |
20:06.14 | Katty | i /think/ he's pissed off that i have the ferrets off |
20:06.16 | Katty | err out |
20:06.31 | hardwire | that seems like something to be pissed about |
20:06.40 | Katty | he doesn't like animals. |
20:07.00 | hardwire | I don't really like ferrets.. but they don't piss me off. |
20:07.03 | hardwire | heh |
20:07.14 | Katty | they're interrupting his Quiet Morning |
20:07.18 | Katty | or Quiet Christmas |
20:07.19 | Katty | idk |
20:07.21 | hardwire | :P |
20:07.29 | hardwire | I'm gonna go on a hike/walk/thing |
20:07.30 | hardwire | bbl |
20:07.33 | Katty | k |
20:08.44 | gme30066 | Skype Out -> IPKall DID -> sip://4355362874@173.160.69.25 = Destination Code "s" in FreePBX and the recorded message. There is an inbound route for the SIP DID that goes to one extension (that is active). |
20:09.53 | hardwire | does freepbx recieve the call? |
20:10.24 | hardwire | nm. I hve to go |
20:11.13 | gme30066 | Yes: http://skitch.com/gellenburg/nc964/freepbx-call-detail-reports |
20:16.24 | ManxPower | Remember "s" is NOT a catchall. |
20:16.30 | *** join/#asterisk smooth_penguin (n=smoove@59.95.28.63) |
20:16.45 | ManxPower | It means "port too stupid to send me the number". If the dialed number is received by Asterisk extension "s" will NOT match the call. |
20:16.51 | *** join/#asterisk lost_soul (n=noymfb@cpe-74-71-234-100.twcny.res.rr.com) |
20:17.03 | ManxPower | But of course, I can't tell you how to fix that in FreePBX, as I don't use GUIs |
20:17.07 | ManxPower | ~freepbx |
20:17.08 | 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, or fpbx |
20:20.34 | *** part/#asterisk gme30066 (n=gme@173.160.69.30) |
20:30.52 | [TK]D-Fender | 15:16]<ManxPower>It means "port too stupid to send me the number". If the dialed number is received by Asterisk extension "s" will NOT match the call. <- still off the mark on this... |
20:40.32 | [TK]D-Fender | gmeand CDR is worthless. Go look at an ACTUAL CALL |
20:49.28 | *** join/#asterisk DelphiWorld (n=Miranda@41.201.64.252) |
20:49.44 | DelphiWorld | anyone around here? |
20:57.43 | *** join/#asterisk jrthomerson (n=jthomers@ip70-180-73-193.no.no.cox.net) |
21:01.13 | jrthomerson | Sorry if this is a newb question, but how can I start a monitor from within an Asterisk application? I see that I can include manager.h and send events with manager_event, but I was hoping to use "Action: Monitor" to start the monitor - and I don't see a way to send an action to the manager through the C API. Perhaps there's a better way to start the monitor? |
21:01.42 | *** join/#asterisk NateHB (n=noone@72.34.90.74) |
21:03.02 | jrthomerson | actually, I might have just answered my own question... it looks like I can use ast_monitor_start |
21:04.45 | jrthomerson | I'd still be curious to see if there's a way to send an action to the AMI - so that I could originate a call on a local channel to start a separate channel that did the monitoring (I'm monitoring a conference) |
21:05.21 | jrthomerson | that way it wouldn't depend upon a single channel in the conference (who may get dropped), but would be on a separate channel that would last the duration of the conference |
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22:37.44 | DonX | Has anyone ever sent calls from an asterisk server to an AZACALL's PSTN interface, via SIP? |
22:38.06 | DonX | Device is an old AZATEL AZACALL200 |
22:38.35 | DonX | wondering if I can send calls from asterisk to the device and then out it's PSTN port |
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