IRC log for #asterisk on 20091225

00:07.34*** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir)
00:12.29lost_souljblack: Merry Christmas
00:34.29*** join/#asterisk dahl (i=dahl@4.84-234-222.customer.lyse.net)
00:48.39*** join/#asterisk bcrisp (i=bcrisp@ip72-222-167-229.ph.ph.cox.net)
00:49.18bcrispho ho ho
00:50.54bcrispecho
00:51.15*** join/#asterisk smooth_penguin (n=smoove@59.95.11.234)
01:01.35bcrispuh oh, p3nguin has competition...
01:03.31*** join/#asterisk gme30066 (n=gme@173.160.69.30)
01:05.44gme30066hello! are there any asterisk iptables gurus around?
01:06.08bcrispgme30066, got a Q?
01:06.26gme30066Yeah, would like to find out if these rules make any sense: http://skitch.com/gellenburg/nc6ck/firewall-rules
01:07.25bcrispgme30066, im pretty sure there are some articles posted on suggested settings for *
01:07.29bcrispiptables
01:07.48bcrispi.e. open hole for manager if you use it
01:08.12bcrispgme30066, check this out: http://www.voip-info.org/wiki/view/Asterisk+firewall+rules
01:08.17gme30066i'm allowing everything from my internal lan. only want to open what's req'd for asterisk from external
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01:09.44gme30066k... 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.
01:09.45bcrispgme30066, its gonna depend on what types of interfaces you need etc
01:09.57bcrispfreezing?
01:11.25gme30066yeah. 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) :-)
01:11.41bcrispoh thats strange
01:11.56bcrispyou have hardware firewall set up?
01:12.29gme30066no. my network setup doesn't allow for a hardware firewall for this specific instance.
01:12.45bcrispwhat kind of environment?
01:12.53bcrispare you using EC2?
01:14.04gme30066way 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.
01:14.21bcrispholy crap
01:14.26bcrispthats a lot of variables
01:14.38bcrispcould it be nat-related?
01:14.40gme30066nah. i've got things trimmed down.
01:14.50gme30066eth0 is publicly routable.
01:16.07gme30066It's default-gw is the gateway of the router.
01:16.24bcrispinteresting, id suggest asking some of the veterans here like leifmadsen
01:16.39bcrispim pretty new and not a networking expert by any means
01:18.59gme30066that's why i'm here. ;-)
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01:31.23fofwareMerry Christmas every Body!!!!!!!!!
01:37.23p3nguinSo what was the question about networking?
01:37.37p3nguinHoping for a condensed version.
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01:41.06gme30066:-) Do these rules make sense? http://skitch.com/gellenburg/nc6q2/webmin-1.500-on-sip.ellenburg.net-centos-linux-5.4
01:44.11p3nguinThe 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.
01:44.18p3nguinOther than that, it seems okay.
01:44.58p3nguinIAX 1 used 5036.
01:45.32gme30066i'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?
01:45.53bcrispp3nguin, whats odd is he says it will work fine then all of a sudden everything drops
01:45.54p3nguinThat's what it seems, but unless you show me iptables -L -nv, I can't say for sure.
01:46.16bcrispdoesnt sound like an iptable issue to moi
01:47.19gme30066iptables -L -nv: http://skitch.com/gellenburg/nc6xb/terminal-ssh-bash-basic-ttys001-129x22-2
01:48.51p3nguinThis will only allow those ports to connect to the machine where you ran that command.
01:49.18jblackI'm lonely and bored and a little sad.
01:49.23jblackLet's all go mug santas
01:49.24p3nguinThe INPUT policy is DROP, so only the accepted ports will be allowed.  Everything else will be ignored.
01:49.37bcrispjblack: go rent bad santa
01:49.57jayteebilly bob?
01:50.03jblackI don't feel a need to justify mugging santa. I just want to mug 'em.
01:51.37p3nguinAnd 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.
01:52.46gme30066do you think it's secure?
01:53.27p3nguinI 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.
01:53.35p3nguin"connections," sorry
01:53.54jblackcan't remember if there is a sip awareness module.
01:54.02gme30066related, established?
01:54.32p3nguiniptables -I INPUT -m state --state RELATED,ESTABLISHED -j ACCEPT
01:55.01p3nguinThe security of Asterisk doesn't really depend on the firewall.
01:55.16gme30066does it matter if I'm not NAT'ed?
01:55.22p3nguinYes.
01:55.54p3nguinBecause if the asterisk machine is behind NAT, your firewall rules should be at the NAT server.
01:56.04gme30066That related, established rule I assume that should go at the bottom?
01:56.27gme30066(yeah the box isn't nat'ed so I guess it's right that the rules are actually sitting on the * box itself.)?
01:57.30gme30066I 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.
01:58.54p3nguinIf 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).
01:59.39bcrispyou just need to make sure SIP and media
02:00.05gme30066i saw something somewhere that had SIP as a range of IP addresses. I'll just make it 5060.
02:00.12gme30066Not using the media.
02:00.19jblackgme30066: rtp.conf ?
02:00.37p3nguinYou'll also want to make sure your rtp.conf ports match up to your range of 10000-20000.
02:01.05gme30066no, it was from a google search looking for the ports asterisk uses for firewalls
02:01.05jblackyou probably want to knock that down.. unless you actually think you could have 5,000 calls at once.
02:01.07p3nguinThat's the typical port range, by the way.
02:01.59p3nguinFor 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.
02:02.06gme30066i'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.
02:02.31jblackthen tighten that up.
02:04.10gme30066i'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).
02:05.22gme30066rest of my servers are running FreeBSD, Solaris, or Mac OS X. IPFilter I know. I'm a complete n00b when it comes to iptables.
02:07.08p3nguinAdd the rule that I gave you earlier, as well as alter the SIP port, and you should be fine on port configuration.
02:07.34gme30066that should be the last rule then?
02:07.45gme30066that related, established one?
02:08.35p3nguinNah, I would probably put it long about the 1st.
02:09.11p3nguinIn my own configuration, I first allow localhost, then allow icmp, then set the related established, then the rest.
02:10.00p3nguinI 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.
02:12.41p3nguinIf you determine that it is better or more effective to put it later, let me know.
02:14.29gme30066I think I just found my hanging problem.
02:15.25gme30066I was allowing outbound DNS queries, but there wasn't a rule for the subsequent reply from the DNS server for lookups.
02:16.40p3nguinrelated,established would cover that.
02:16.58p3nguinsince it would be "related" and all.  :)
02:17.15gme30066yeah I suppose so. :-) I've gone ahead and added implicit rules though j/i/c
02:17.40p3nguinI don't agree with that being the correct way to handle it.
02:17.56gme30066the implicit rules?
02:18.00p3nguinyeah
02:18.16p3nguinI feel that the correct way is to add the related,established rule.
02:18.47gme30066i see your point. the implicit rule opens up another hole.
02:22.40gme30066sumbitch this is actually starting to come together and work finally. :-)
03:11.46*** join/#asterisk OrNix (n=ornix@l151-249-47.static.cn.ru)
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03:34.54phixhey hey
03:34.57phixwhat's new
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04:03.29NateHBAnyone here familiar with the edgemarc?
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05:02.14Kattydrags in
05:02.34Kattyplops down
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05:12.23bcrisp:/
05:26.44ReDNeQmerry xmas
05:34.15carrarHO HO HO
05:34.19carrarMarry Christmas!
05:37.40coppiceis that what Mrs Claus Christmass did?
05:38.13TJNIII think we should just be friends....
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07:58.35shamelessn00bhey all
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08:15.26ChannelZhi ho
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09:00.41shamelessn00bChannelZ: sup dude
09:00.46shamelessn00bmerry christmas
09:13.30*** part/#asterisk c0rnoTa (n=c0rnoTa@109.188.36.76)
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09:46.49*** join/#asterisk rare1980 (n=rare@115.186.25.235)
09:47.39rare1980hi every one
09:49.10rare1980helloooooooo
09:49.23rare1980is any one online.. in this group??????????????
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10:22.29tzafrir_laptoprare1980 is not on-line in this group
10:52.38shamelessn00bguys
10:52.51shamelessn00bI have this setup, 4 PRIs on 2 machines
10:53.01shamelessn00b(Total 8 PRIs)
10:53.02*** join/#asterisk ChannelZ (i=channelz@burner.com)
10:53.22shamelessn00bI want to initiate 120 calls from 1 machine simultaneously that will land on the other machine
10:53.32shamelessn00bhow can it be done
10:53.51shamelessn00balso, are there some scripts to stress test these systems
11:20.44*** join/#asterisk puzzled (n=foobar@puzzled.xs4all.nl)
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12:22.50fenrusshamelessn00b, use sipp to stress-test SIP.. :)
12:23.10fenrusshamelessn00b, sipp.sf.net
12:23.38shamelessn00bwhat about pri calls
12:27.32shamelessn00bDAHDI
12:27.33shamelessn00bcalls
12:33.33*** join/#asterisk rare1980 (n=rare@115.186.9.201)
12:34.02rare1980hi
12:34.03rare1980all
12:35.28rare1980hellooooo any one alive in this roooooooom
12:35.28rare1980?
12:42.32fenrusrare1980, yea?
12:42.43fenrusshamelessn00b, dunno :)
13:22.48tzafrir__laptopinfobot, tell rare1980 about ask
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13:23.53rare1980HI FENRUS
13:23.54rare1980u still there?
13:24.57fenrusrare1980, yes.
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13:28.01rare1980i want to that how can we configure
13:28.07rare1980predictive dialer in asterisk
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13:29.56tzafrir__laptoprare1980, any specific predictive dialer?
13:36.06fenrusi've no clue about that.. but i guess you'll just have to read up about it
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13:50.52benngardany1 here that have tested avaya 9650 r2.5?
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14:36.35mazpehow can i setup MeetMe to have a conference in Listen Only mode exepct for admin or marked users?
14:36.44mazpeeveryone else who joins should be muted.
14:37.02mazpeusers should be muted at all times.
14:41.25[TK]D-Fendermazpe: "core show application meetme" <-
14:41.48[TK]D-Fendermazpe: all of the options are right there....
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14:57.49grandpapadotMerry X-Mas peeps.
15:02.54ariel_Morning folks, hope everyone has a great Merry Christmas
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15:43.31drmessanoMerry Asterisk
15:50.27grandpapadotlol
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16:22.59bcrispmerry christmas
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17:08.17*** join/#asterisk teknoprep (n=Chris@unaffiliated/teknoprep)
17:08.31teknoprepdoes anyone know how to do a tcpdump so i can analyze it in wireshark ?
17:08.39teknoprepi forget how to dump it off to pcap
17:15.38mazpetcpdump -i xl0 -s 1500 -w file.name
17:16.47dlynesteknoprep: btw
17:17.01teknoprepyo
17:18.02dlynesteknoprep: a better solution is sipdumppcap or something like that
17:18.23dlynesteknoprep: it only dumps sip convos, and dumps each convo into a separate pcap file
17:18.41teknoprepwhere would i pick tha tup at ?
17:19.36dlynesoh yeah... pcapsipdump
17:20.07dlyneshttp://pcapsipdump.sf.net/
17:20.52dlynesteknoprep: default mode is to create a subdirectory and dump each conversation into a separate pcap file
17:21.02dlynesteknoprep: it ignores any other type of traffic
17:21.17dlynesteknoprep: it also dumps all the rtp traffic into the same file
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17:21.34teknoprepwell thats nice
17:23.04teknoprepi love wireshark
17:23.08teknoprepdecoding call's is fun
17:23.20teknopreptoo bad it doesn't do G729
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17:28.54teknoprepyeah but wireshark seperates the calls for you
17:29.02teknoprepwhy would i need it to be done at the file level ?
17:31.03tzafrir__laptopteknoprep, mazpe: tcpdump -s0 instead of -s<whatever> - just grab everything
17:31.27teknoprepok
17:31.31teknoprep-s0 ?
17:31.44tzafrir__laptopjust gets everything.
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17:41.29wwalkerI 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?
17:41.56wwalker(doing SIP/RTP over a high loss network bytes)
17:44.20Pan3Dmerry christmas
17:45.10wwalkerPan3D: merry christmas
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19:21.59Nuggethttp://thereifixedit.com/2009/12/25/a-tower-ing-christmas-tree/
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19:42.36AkiraaWhat's the difference between "public SIP" and "private SIP"?
19:42.46hardwireaccessability
19:43.14Kobazmmm
19:43.32Kobazstill 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
19:44.04Akiraahardwire: 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)
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19:44.25hardwireAkiraa: as in a private website would be accessable to whomever has access.
19:44.32hardwiresame with sip
19:44.41*** join/#asterisk denon (n=denon@synapse.subneural.net)
19:44.41*** mode/#asterisk [+o denon] by ChanServ
19:45.22hardwireAkiraa: why do you ask?  I never refer to them in that way
19:45.25*** join/#asterisk denon (n=denon@synapse.subneural.net)
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19:45.44hardwiredenon: your system.. is down.. your system.. is down..
19:45.51Akiraahardwire: some internal settings in a few phones I have ATM
19:45.58hardwireAkiraa: which?
19:47.48Akiraahardwire: http://www.5111soft.com/productview.php?pid=7&id=1  -- the cheapest voip phones I could find, really
19:49.13hardwirehah
19:49.17hardwireyeh.. they over engrished
19:49.25hardwirePrivate Register/Private Proxy
19:49.44hardwireand then Enable Private Register
19:49.58hardwireI can see how that would be weird
19:50.13denonlooks at hardwire
19:50.22*** join/#asterisk Tim_Toady (n=moi@77.49.123.166.dsl.dyn.forthnet.gr)
19:50.37hardwiredenon: you've disconnected/reconnected at least 10000000 times
19:50.49hardwireok so maybe just 5
19:50.51hardwire:P
19:50.59hardwirejust thought I'd let you know :P
19:51.05denonhm, maybe someone's jackin with my irc socks
19:51.09denonsorry bout that
19:52.31hardwireAkiraa: http://www.5111soft.com/largerview.php?id=18
19:52.34hardwirethat thing is crazy
19:52.37*** join/#asterisk denon (n=denon@synapse.subneural.net)
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19:53.38Akiraahardwire: crazy how?
19:55.05Kattyferretcam is on.
19:55.23hardwirecrazy crazy
20:00.28*** join/#asterisk debuggerboy (n=anish@117.196.163.239)
20:02.01Kattyboyfriend is cranky today
20:02.20gme30066Hi! 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? :-)
20:04.07hardwireKatty: kick him in the nuts
20:04.14Kattythat's just make him more pissy
20:04.20hardwiregme30066: kick them in the nuts
20:04.20*** join/#asterisk vitaminx (n=vitaminx@89.130.31.1)
20:04.41gme30066:sigh:
20:05.06hardwirehow are you originating the call?
20:05.24Kattyhardwire: he has his headphones on, with this scowl on his face.
20:05.32Kattyhardwire: and i know when a song goes off, cause it's that loud...
20:05.43Kattyhardwire: but he won't even talk to me when the song goes off.
20:06.00hardwireaww
20:06.10hardwiredon't make me repeat myself
20:06.12hardwire:P
20:06.14Kattyi /think/ he's pissed off that i have the ferrets off
20:06.16Kattyerr out
20:06.31hardwirethat seems like something to be pissed about
20:06.40Kattyhe doesn't like animals.
20:07.00hardwireI don't really like ferrets.. but they don't piss me off.
20:07.03hardwireheh
20:07.14Kattythey're interrupting his Quiet Morning
20:07.18Kattyor Quiet Christmas
20:07.19Kattyidk
20:07.21hardwire:P
20:07.29hardwireI'm gonna go on a hike/walk/thing
20:07.30hardwirebbl
20:07.33Kattyk
20:08.44gme30066Skype 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).
20:09.53hardwiredoes freepbx recieve the call?
20:10.24hardwirenm. I hve to go
20:11.13gme30066Yes: http://skitch.com/gellenburg/nc964/freepbx-call-detail-reports
20:16.24ManxPowerRemember "s" is NOT a catchall.
20:16.30*** join/#asterisk smooth_penguin (n=smoove@59.95.28.63)
20:16.45ManxPowerIt 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)
20:17.03ManxPowerBut of course, I can't tell you how to fix that in FreePBX, as I don't use GUIs
20:17.07ManxPower~freepbx
20:17.08infobot[~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-Fender15:16]<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-Fendergmeand CDR is worthless.  Go look at an ACTUAL CALL
20:49.28*** join/#asterisk DelphiWorld (n=Miranda@41.201.64.252)
20:49.44DelphiWorldanyone around here?
20:57.43*** join/#asterisk jrthomerson (n=jthomers@ip70-180-73-193.no.no.cox.net)
21:01.13jrthomersonSorry 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)
21:03.02jrthomersonactually, I might have just answered my own question... it looks like I can use ast_monitor_start
21:04.45jrthomersonI'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)
21:05.21jrthomersonthat 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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22:37.44DonXHas anyone ever sent calls from an asterisk server to an AZACALL's PSTN interface, via SIP?
22:38.06DonXDevice is an old AZATEL AZACALL200
22:38.35DonXwondering if I can send calls from asterisk to the device and then out it's PSTN port
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