IRC log for #asterisk on 20100509

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00:07.52Dovidhi
00:09.21saxahi
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00:14.40carrarhi
00:23.26TJNIIhi
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00:31.08Doviduit
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00:58.40saxaMay  9 03:15:32 quadserv kernel: wctdm24xxp 0000:03:05.0: Port 1: Installed -- AUTO FXO (FCC mode)
00:58.56saxaanybody knows to explain whta means FCC mode ?
00:59.12saxaI tried to load also the driver with opermode=ITALY
00:59.24saxabut it seems it doesnt work
01:00.08saxaat least it asterisk doesnt get any signal from the card when I load the module with that option
01:00.24saxaany ideas ? Is this just some kind of misconfiguration ?
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01:54.59p3nguinIf MagicJack has been adding over 250,000 new customers a month for at least the past two years, how do we have so many people in here not using them?
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01:56.18sawgoodha!
01:56.24sawgoodIts Magic!
01:57.10sawgoodwhat is a URL to 'paste' a .PDF drawing (like pastebin.org)
01:57.15sawgoodI totally forgot what its called
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01:58.47Mark22sawgood: you could "use" mibbit for that if I am correct
01:59.33*** join/#asterisk romb_work (~romb@unaffiliated/romb-work/x-7222485)
02:00.20p3nguinsawgood: If it is an image, use imagebin.org or something similar.
02:00.25sawgoodnice
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02:18.15sawgoodhttp://imagebin.org/96159
02:18.31sawgoodIf anyone could look at this drawing and give me your opinion on how I can make it work, I'd appreciate it
02:19.43sawgoodI'd like to make Asterisk Box A ... have a connection to Asterisk Box B ...
02:23.58Corydon76-digsawgood: Take a look at openvpn
02:24.19Corydon76-digespecially if either machine is not on the public IP network
02:24.46sawgoodbox Asterisk boxes are on the Internet
02:24.59TJNIIThat's a ... unique ... way to draw a phone.
02:25.13sawgoodyeah ... funny ... but I saw another guy do it once
02:25.34sawgoodCould I simply create a trunk from Box A to Box B?
02:25.48sawgoodSIP trunk on Box A which 'registers' on Box B?
02:25.54TJNIIYou can just connect them with SIP or IAX.
02:25.57Corydon76-digSIP peer, but yes
02:26.22sawgoodyes, a SIP peer from Box A to Box B ...
02:26.56Corydon76-digDial(${EXTEN}@boxb)
02:27.05sawgoodnice ... thank you
02:27.35Corydon76-digMake sure you're using good practices, like authentication with a STRONG password on both ends
02:28.03sawgoodCorydon76-dig: so, basically all that this is ... is a standard SIP trunk between the two boxes?
02:28.12Corydon76-digyep
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03:06.35*** join/#asterisk Vin73 (~vinodp@sbis4101.otago.ac.nz)
03:07.09Vin73hi, I am facing an issue with dahdi installation
03:07.38Vin73dahdi-linux-complete-2.3.0
03:08.47Vin73The issue is that when I issue the command make all as root, the installation script tries to connect to downloads.digium.com, but fails
03:09.09Vin73However, I can connect to the portal from my browser window...
03:09.18Vin73Is this a bug?
03:16.51*** join/#asterisk neurosys (~neurosys@c-71-196-20-208.hsd1.fl.comcast.net)
03:17.55Vin73hello, has any of the list members faced this problem before?
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03:19.42pabelangerpastebin output
03:19.57*** join/#asterisk Vin73 (~vinodp@sbis4101.otago.ac.nz)
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03:20.23Vin73hi, I am facing an issue with dahdi installation
03:20.33Vin73dahdi-linux-complete-2.3.0
03:20.50pabelangerpastebin output
03:20.56Vin73The issue is that when I issue the command make all as root, the installation script tries to connect to downloads.digium.com, but fails
03:21.11Vin73However, I can connect to the portal from my browser window...
03:21.33pabelanger~pb
03:21.33infobot[~pb] 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/ , http://asterisk.pastey.net/ , or install pastebinit with yum or aptitude.
03:21.46pabelangerVin73: ^^^
03:21.59Vin73hi pabel
03:22.41Vin73I am sorry about overstepping the limit
03:22.51Vin73But has anyone seen this problem before?
03:23.03*** join/#asterisk coppice (~chatzilla@153.166.232.220.dyn.pacific.net.hk)
03:23.10pabelangerVin73: pastebin your output from dahdi
03:23.31Vin73OK, just give me a few minutes
03:30.14Vin73Hi pabl
03:30.17Vin73pabel
03:30.32Vin73here is the pastebin log of dahdi output
03:30.41Vin73make -C linux all
03:30.41Vin73make[1]: Entering directory `/usr/src/dahdi-linux-complete-2.3.0+2.3.0/linux'
03:30.41Vin73make -C drivers/dahdi/firmware firmware-loaders
03:30.41Vin73make[2]: Entering directory `/usr/src/dahdi-linux-complete-2.3.0+2.3.0/linux/drivers/dahdi/firmware'
03:30.41Vin73Attempting to download dahdi-fwload-vpmadt032-1.20.0.tar.gz
03:30.42Vin73--2010-05-09 02:56:27--  http://downloads.digium.com/pub/telephony/firmware/releases/dahdi-fwload-vpmadt032-1.20.0.tar.gz
03:30.42Vin73Resolving downloads.digium.com... 76.164.171.232, 2001:470:e0d4::e8
03:30.43Vin73Connecting to downloads.digium.com|76.164.171.232|:80... failed: Connection timed out.
03:30.43Vin73Connecting to downloads.digium.com|2001:470:e0d4::e8|:80... failed: Network is unreachable.
03:30.43Vin73make[2]: *** [dahdi-fwload-vpmadt032-1.20.0.tar.gz] Error 1
03:30.44Vin73make[2]: Leaving directory `/usr/src/dahdi-linux-complete-2.3.0+2.3.0/linux/drivers/dahdi/firmware'
03:30.44Vin73make[1]: *** [firmware-loaders] Error 2
03:31.37pabelangerVin73: use pastebin
03:31.40pabelanger~pb
03:31.41infobot[~pb] 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/ , http://asterisk.pastey.net/ , or install pastebinit with yum or aptitude.
03:31.49Vin73I just did
03:32.00Vin73What I pasted was the output from pastebin
03:32.10pabelangerno you didn't you flooded the channel.
03:32.14pabelangerpost URL
03:32.31Vin73ok, sorry, here is the URL
03:32.41pabelangerfailed: Network is unreachable
03:32.47Vin73http://pastebin.com/bAguTnxL
03:32.51pabelangerthat is your problem
03:33.10Vin73no, but as I mentioned, there seems to be no problem with the network
03:33.20*** join/#asterisk ManxPower (~manxpower@43.sub-75-254-99.myvzw.com)
03:33.22Vin73I can browse to downloads.digium.com
03:33.52pabelangerthen manually download the file and place it in your source folder
03:33.57Vin73So I am wondering why the script fails with a network error
03:34.07pabelangerbecause there is a network error
03:34.15pabelangereither your side or server side
03:34.46Vin73OK..so I should place the file in the same directory where I am executing make all, right?
03:35.21Vin73hello, pabel?
03:36.10pabelangerpatience young grasshopper
03:36.19pabelangerplace then in your linux/drivers/dahdi/firmware folder
03:36.37pabelangers/then/them/g
03:36.51Vin73ok, thanks...
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04:13.23sawgooddoesn't seem like anyone in FreePBX can help much, so I'll try here ...
04:13.29ChannelZdon't
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04:18.26NivexI've got an IP that's been banging away on my SIP port all night.  Anything I can do besides block the IP?
04:18.59TJNIINot really.  fail2ban is always an option.
04:26.24ChannelZcock blocking
04:27.06NivexI was trying to host block him.  finally blocked him at the firewall and he seems to have finally quit
04:29.12Nivexroot@avalon:/var/log/asterisk# grep '74.212.249.162' messages | wc -l
04:29.12Nivex25824
04:29.27*** join/#asterisk halkun (~halkun@officegateway.ethostream.com)
04:30.49halkunOk, I'm not on version 3 of designing my home network layout. Does this look sane to you?   ------> http://img26.imageshack.us/img26/6513/netc.png
04:31.36halkunfor those following along from last light, I decided to move the * server and make it the firewall/router for the rest of the network
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04:37.52Vin73Hi I get an error while trying to make install dahdi-linux-complete-2.3.0+2.3.0 on Fedora Core
04:38.13Vin73The complete logs have posted at http://pastebin.com/amBjpWb8
04:38.39Vin73Please share your thoughts
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04:42.20Vin73Hi, has anybody seen this error before?
04:42.45halkunhere is the old network map ----------------> http://img85.imageshack.us/img85/5852/neta.png
04:49.04TJNIIYou spelled "Intarweb" wrong.
04:54.32*** join/#asterisk Cain (~Geek@unaffiliated/cain)
04:58.42halkunDid I? Drat! :P
05:11.16TJNIIWhat did you yse to draw that map?
05:11.35Vin73hi, any thoughts on my posting?
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05:21.41halkunTJNII: I used Dia to draw the map
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06:09.47xhelioxwhistles a happy tune
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06:15.52devdvdhi all.
06:16.30xhelioxhello.
06:19.42devdvdim trying to do something with asterisk and im not sure its possible without a custom app.  Just wanted to make sure before i go down that road.  I am doing a support queue system.  I want to be able to bridge another agent onto my call (kind of like 3 way calling) without having to do the bridging on my local phone (not all agent phones will be directly connected sip/iax phones)
06:20.02devdvdactually theres no kind of , its exactly 3 way calling
06:21.29devdvdany thoughts on how i could make asterisk bridge the calls together.
06:21.41devdvdi have a feeling im over complecating this
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06:33.36jdoedevdvd: ghetto way would be to use attended transfers to fake it ;)
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09:54.56onemikehas just tried to install both Asterisk 1.4 and Asterisk 1.6 with the asterisk-gui, both from FreeBSD ports. I'm having major problems getting the gui to work -- the menus on the left only appear after login *sometimes*, and after some kind of weird lengthily delay
09:55.29onemikeFurther, I can't seem to submit changes on some of the dialog boxes in order to actually configure Asterisk from asterisk-gui... it's extremely weird.  Can anyone offer any insight?
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10:17.08onemikeactually the whole gui is running extremely slow
10:23.04*** join/#asterisk rethus (~suther@p5087EA77.dip.t-dialin.net)
10:23.31rethuscan connect on same machine with twinkle to asterisk, but not with remote-machine.
10:23.50rethushave i to set asterisk to except external connections?
10:26.18tzafrir_laptoprethus, are both asterisk and twinkle on the same LAN?
10:26.26tzafrir_laptopah, not
10:26.37tzafrir_laptopis the Asterisk one behind NAT?
10:26.45tzafrir_laptopOr the twinkle one?
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10:27.40tzafrir_laptoponemike, what version of asterisk?
10:31.33rethus<PROTECTED>
10:31.45rethus1.6
10:32.09rethusgot as result: 401 Unauthorized
10:32.51tzafrir_laptoprethus, "401" basically means: "authentication required"
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10:49.13onemikerethus: tried 1.4 and 1.6.0.21 -- same thing
10:49.20onemikeerr, directed to tzafrir_laptop sorry
10:58.15onemikejust tried to reinstall 1.6.0.21, enabled web server but otherwise blank install -- new SVN asterisk-gui
10:58.40onemikeand it's hanging after login on 'checking write permissions for gui folder'
10:59.10onemikeseems to sit there indefinitely
11:02.28tzafrir_laptopDoes it have those permissions?
11:03.56onemikeyes
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13:28.18Weazelhey guys, can anyone tell me if the Linksys PAP2-NA work with T.38  ? (Asterisk 1.6)
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13:33.24coppiceThe PAP2 does not support T.38
13:51.00Weazelno firmware to add support ?
13:53.41coppicenope
13:54.04Weazelcrap
13:54.36coppiceif you want a Linksys ATA that supports T.38 you needs an SPA2102, SPA3102, or one of the bigger ones
13:56.31Weazelthanks for the info
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13:57.53coppiceyou can find a lot of material on the internet that says the PAP2 does support T.38, but then you can find information saying practically anything you want
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14:30.41Weazelwhat does "chan_dahdi.c: Unrecognized pridialplan NPI modifier: a" mean ?
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14:32.19dinesh___hi all, I have an extension rule "exten => _2,1,Set(NUMBER=..)" to implement a kind of speed-dial for the number "2"
14:32.44dinesh___but the problem is that, when I try to dial "2", it is not that extension that is matched, but the one with "exten => 2,1,..." instead
14:32.50dinesh___how to avoid this issue ?
14:33.40dinesh___the latter extension "exten => 2,1,.." is one that is not supposed to be reached directly by the phone, the idea is that I should only be able to reach it using Goto
14:35.06*** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485)
14:36.38[TK]D-Fenderdinesh___: You shouldn't be using "_2", but rather "2" instead.  There is no pattern there, it is a fixed value.
14:36.56[TK]D-Fenderdinesh___: And you aren't showing us enough to verify what you're saying.
14:37.16dinesh___ok, but how can I then mask some extensions so that they can't be dialed directly ?
14:38.44dinesh___http://pastebin.com/j5drdBAP
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14:39.09dinesh___this is what I am doing, I have some pattern matching, which share the same logic, so I Goto() the shared logic
14:39.29dinesh___however the shared logic (the exten 2 here), should not be directly accessible
14:40.11dinesh___so right now if someone picks up the phone and dials 2, it goes to that logic, without the ${BUSINESS} variable set, which is of course not right
14:40.41[TK]D-Fenderdinesh___: put it in another CONTEXT
14:40.57[TK]D-Fenderdinesh___: Or name it something that can't be dialed instead.
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14:41.55dinesh___good, I'll put the private extensions in another context then, thanks a lot
14:50.21CainHi, sorry for this offtopic question, i have 8 SPA922 IPPhones configured with the sameway as a sip phones of asterisk, i have 3 of them that forget last config and restore an old configuration, is there anyway to hard write configuration for those phones or any other way to get them remember the last configuration
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14:56.19WeazelWhy does freepbx redirects me to     DAHDI/G1/fax,300,TtWw        while trying to send fax out through ATA > PRI
14:58.30[TK]D-FenderWeazel: ...
14:58.32[TK]D-Fender~freepbx
14:58.32infobot[~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
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15:41.12xLPI have a problem with an old Asterisk, which I'm not allowed to upgrade (1.4.26.1) : upon Hangup(), I have a DeagAGI script, which runs fine... however, no SIP "BYE" will be sent until my AGI script returns... Has anyone seen that before?
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15:54.02iscariohi, just a question abt the 'host' parameter of sip.conf : if i want to allow a peer to connect to my server from different domain or with different ip address (let's say for example a sip user which is used to be connected from different cybercafe), my only solution is to put host=dynamic , right ?
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16:03.24[TK]D-Fenderiscario: correct
16:03.46iscariothank you [TK]D-Fender
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16:43.44cuscohello
16:45.00cuscowe are having loads of IAX warnings like http://paste.debian.net/72675/
16:45.16cuscoeverytime a warning like this comes there is "cuts" in the call
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17:06.45ChannelZcusco: network issue probably.. either not enough bandwidth to support the calls, or some latency or other issue causing packets to show up more out of order than usual..
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17:20.16cuscoChannelZ: ok thanks for info
17:20.20cuscoanother question
17:20.26cuscoa que defined on queues.conf
17:20.32cuscothere are members onit
17:20.54cusconow when I call the dialplan, dialplan has queue(name of queue);
17:21.03cuscomy softphone is using gsm codec
17:21.12cuscoI know this can be configured on the peer
17:21.26cuscobut how do I set p a priority?
17:21.36cuscoliek, first alaw, then ulaw, then gsm
17:22.31ChannelZwell in sip.conf I think the order you allow them in is the preferred order it will try to negotiate, but I honestly don't know for sure
17:22.52cuscook thnaks
17:28.35cuscoone more question...
17:28.36cusco[May  9 18:27:39] NOTICE[25181]: rtp.c:1778 ast_rtp_read: Unknown RTP codec 126 received from '0.0.0.0'
17:28.53cuscois tis a simple signal sent that can be ignored
17:29.12cuscolike a codec misconfigured in a softphone
17:29.19cuscoor is it something bad in a call?
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17:31.31ChannelZhmm that sounds familiar
17:32.27ChannelZis this x-lite?
17:33.11cuscoyes
17:34.25ChannelZI seem to remember reading something about x-lite not really following the spec and attempting to use a codec even though it was negotiated not to, or something like that
17:34.40ChannelZI don't know which #126 is specifically
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17:39.52ChannelZah.. apparently it's no actual codec, they are using it as keep-alives - http://forums.counterpath.com/viewtopic.php?f=6&t=12552
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17:46.21cuscoright... gotcha ChannelZ
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18:16.11halkunDumb Networking question... I'm putting my asterisk server together now. I seem to have 4 extra network cards. I'm using my server also as a gateway. Can I use the extra network cards as a kind of switch?
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18:17.55p3nguin_yes
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18:18.18orionihi there, ho to connect an old phone/fax to an asterisk server so it can send=receive fax ?
18:18.32orionip3nguin hi there
18:18.57halkunCool, I'll load up the server then...
18:19.30p3nguin_Just because it is possible does not mean it is the best thing to do, though.
18:20.01p3nguin_Especially if you have an actual switch that you could use instead.
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18:24.51zambawhen sending users to my voicemail, how can i omit the playback of the vm-intro file first?
18:25.19zambai'm running my asterisk for norwegian users, so the english version of vm-intro makes no sense to the callers
18:25.27p3nguin_core show application VoiceMail
18:25.45zambaah!
18:25.47zambathanks
18:26.10p3nguin_Or change the language (which I do not know how to do).
18:26.13zambais there a way to set this as default for all calls to VoiceMail()
18:26.42p3nguin_It's done in the dialplan, so it will do whatever you tell it to do.
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18:45.55halkunI'l mooking up using mutile netwrok cards as a switch... I'm not coming up with anything
18:46.09saxais dahdi-linux-2.3.0 the last stable one ? Is it recomended for use ?
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18:46.18p3nguin_It's because they aren't really a switch.  You said "like a switch."
18:46.23halkunin fact some say it's I need a crossover cable?
18:46.37p3nguin_Cross-over depends on other factors.
18:47.30halkunWell, the problme is I don't have a switch, but I have 5 network cards. I'm trying to make the asterisk server into a router too
18:47.58halkunso I can plg in my compter/Access point/PS3 into it too
18:48.18p3nguin_You can connect your modem to one port and computers to other ports.
18:48.24TJNIIYou want to google bridging
18:49.00TJNIIThat will give you the info to make the cards into a "psudo-switch"
18:49.24halkunwhy can't I make it into a "switch-switch" :)
18:49.35p3nguin_because it isn't a switch.
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18:49.47orionihalkun have a look at bridging under linux
18:49.57halkunI thought a switch is pretty much a splitter
18:50.01p3nguin_"like a switch" is the best you'll be able to do with a bunch of NICs.
18:50.31halkunOk, I'll look up bridging
18:50.35TJNIIA switch isn't a splitter.  Modern switches do a hell of a lot.
18:51.01p3nguin_A hub is more like a splitter, but not a lot of people use hubs anymore.
18:51.35TJNIIstill uses hubs.
18:51.43TJNIIFor very specific applications, mind you.
18:54.03dohdonce had the nice situation of having to try and make a switch work like a hub... took a little effort :(
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18:54.52dohdbut a pc can be turned into a switch
18:55.02dohdyou'll just never get the performance of a decent switch
18:55.08dohdwith the ease of use of a decent switch
18:55.23dohdand it'll take you a hell of a lot more time than just buying a decent switch
18:55.55dohdand if it's just for home purposes, you can easily buy just a simple switch for 20euro or something
18:56.06dohdso why bother messing with a pc
18:56.18dohdwaste of time and energy
18:56.43p3nguin_He already has five NICs in his box and he doesn't already have a switch.
18:57.48p3nguin_But yeah, you could easily bridge four NICs into one, assign bridge0 an IP address for the LAN, and then connect computers to those ports.
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19:00.23TJNIIThis strikes me as one of those "If you have to ask you probably don't know enough to pull it off" scenarios, though.  Is it worth the time needed to learn and set it up properly?  i'm all for learning, but don't you have a router you can dumb down to a switch?
19:01.07p3nguin_Most "routers" for home use have switches built in, anyway.
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19:01.15TJNIIExactly my point.
19:01.33TJNIII have a couple SOHO routers I use as switches.
19:01.47p3nguin_Just don't plug anything in on the WAN/Internet port, and you've got yourself a 4-port switch.
19:01.59TJNIIYou have to turn off DHCP on it, too.
19:02.09p3nguin_Good idea.
19:02.14TJNIIOtherwise, yea, just ignore the WAN port.
19:03.09dohdthat seems like the only way nowadays to just have a wireless accesspoint
19:03.17TJNIIYep.
19:03.30dohdwhich kinda sucks
19:03.34TJNIIYep.
19:03.37dohdif I want a router I get a router
19:03.43dohdif I want an accesspoint I don't want a router
19:04.29TJNIIOn the other hand economies of scale means that by having the one device that meets both needs we can get it cheaper than either of the two seperate devices.
19:04.33dohdwelcome to the wonderful world of the market doing its thing...
19:04.53dohdso you can have all the shitty stuff you don't want for 10 cents less than other shitty stuff you don't want
19:05.09TJNIII think "economies of scale" is the right term.... I should have paid more attention in Econ.
19:05.25dohdTJNII: as long as it does the two things as well as the 2 seperate devices would
19:05.36dohdand there is a market for those all-in-one thingies
19:05.51dohdit just irritates me that it seems to mean that there is only mainstream market left
19:05.57dohdeverything that is not mainstream ceases to exist
19:06.00TJNIIWell, I don't know about your first point but the latter is obviously true.
19:06.27TJNIIIt exists, it is just geared to buisness customers and costs a hell of a lot more.
19:06.34dohdjust like the beloved dsl market in the netherlands...
19:06.53TJNIIThe office I work at is peppered with access points, and I don't want to know how much they cost.
19:07.03dohdsince everyone only knows "price for speed", that means that everything is brought down to the worst thing you can sell for the lesat price
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19:15.23xLPanyone knows how to "propagate" CDR accountcode thru call queues? (I set up accountcode on incoming call, then call goes to a call queue... and there the accountcode disappaear)
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20:37.56saxawhat you people suggest I should use 1.6.0.x , 1.6.1.x or 1.6.2.x version of asterisk ?
20:38.06p3nguin_1.4.30
20:38.19saxaand which dahdi drivers version for TDM410P ?
20:38.49saxap3nguin_: I had 1.4.21 and in fact was working very well
20:39.09saxanow I was trying the 1.6.2.7 and I have problems with dahdi
20:39.31saxaif I load the FCC mode I can get the phone working
20:39.47saxaif I load it with not FCC then its muted
20:40.00saxaasterisk even doesnt load chan_dahdi
20:40.08saxaand I cant understand why
20:41.06saxaenough that I change the loadzone and defaultzone to it and its stops working
20:41.11saxahuh, strange
20:41.29saxabut with FCC I can get the US zone working
20:41.35saxasignal
20:42.01saxap3nguin_: do you use dahdi ?
20:42.07p3nguin_yes
20:42.16saxawhich version ?
20:42.42p3nguin_2.2.1
20:42.54saxado you know if 2.3.0 is safe to use ?
20:43.23saxahow is development stuff numbered ? Is there something like linux kernel was ?
20:43.36saxaodd and even numbering scheme ?
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20:45.55p3nguin_no idea
20:48.10saxaheh, seems I got it working in IT mode
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21:29.22saxahmm, finaly seems I got it working
21:29.50saxait seems I need to use dahdi_cfg -vv to apply the configuration to the driver, correct ?
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22:03.39*** join/#asterisk Chilling_Silence (~Chilling_@smtp.fcsltd.co.nz)
22:04.37Chilling_SilenceI'm having issues with random call drops, and I'm unsure how to go about tracing them. I've done a dozen odd asterisk installs before, and never come across this. Happens with both SIP -> ITSP, as well as calls that use the Patton SN4554 (ISDN -> SIP) device, so it's not the internet connection
22:07.47Chilling_SilenceSome tcpdump logs show semi-interesting things, but I've got nothing solid to go on. It's not internet, it's not a faulty ethernet (We've tested with a secondary USB -> Ethernet device) and I've got identical hardware elsewhere working happily
22:09.21devdvdChilling_Silence: what does sip debug show when a call drops?
22:09.34Chilling_SilenceI've not had sip debug going when a call drops
22:09.52Chilling_SilenceI've had it on, but when I do, I seem to then get an hour or two of "happiness", no call drops or anything
22:10.11Chilling_SilenceProblem is that when that happens, the tcpdump output grows quite rapidly
22:11.15Chilling_SilenceThat's my biggest problem is it's irregular :(
22:11.21pabelangerSo, res_config_odbc or res_config_mysql / res_config_pgsql?
22:12.28devdvd1. you will likely need the sip debug as it is a good place to start because the sip channel seems to be the common factor here.  Possibly some reinvite issue on the phones?
22:12.40Chilling_Silencedevdvd: We *were* getting malformed packets to either the SIP or the ISDN trunk, but not to the internal SIP phones (A variety of Snom 300, 320s, 360s, Linksys SPA942's and SPA2102s)
22:13.04devdvd2. you say tcpdump grows. what is it saying?
22:13.30Chilling_SilenceOK, because it's happening across all of ^^, not to mention if I have an inbound call setup to go straight through to, say an echo test, or through to MOH, it also happens, so it's not an issue with the endpoints
22:13.31devdvdpabelanger: wont you need a seperate library for odbc on linux?
22:13.48Chilling_Silencedevdvd: It just grows with all the RTP traffic from the other calls that happen
22:14.22devdvdwell then how does that look even semi-interesting?
22:14.23pabelangerdevdvd: each have dependencies on libraries (unixODBC or libmysql-dev)
22:14.40devdvdyea
22:15.18pabelangerUsually use res_config_mysql myself, but wanted to get feedback with res_config_odbc
22:16.39Chilling_Silencedevdvd: Occasionally some of the rtp traffic shows malformed packet errors, apparently it's missing some of the headers or something (I had some help with a 3rd party diagnosing that, dont totally understand it myself)
22:16.45Chilling_SilenceBut mostly it's just boring
22:17.31devdvdwhat is the protocol to your itsp?
22:17.33devdvdsip?
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22:24.43Chainsawphix:
22:25.38Chilling_Silencedevdvd: Yes, SIP. Happens regardless of codec, iLBC, g711 alaw, ulaw, g729
22:25.54Chilling_SilenceSame happens to the SN4554 ISDN <-> SIP box which is on the LAN
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22:34.45booduhello
22:35.34devdvdyou said it happens on echo as well
22:35.37devdvdright?
22:35.42devdvdif you do an echo test
22:35.49devdvdwhere does that echo test come from?
22:36.07devdvdpstn via itsp? or via isdn
22:41.02Chilling_Silencedevdvd: Yes, it happens only on echo tests via ISDN or ITSP. Internally on the LAN, it seems fine
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22:59.02halkunOk, I'm probably in the wrong place, but I was following your guy's advice about setting up my network
22:59.34halkunI was told to make the asterisk server my gateway so I won't have NAT issues
23:01.34Chainsawhalkun: Sounds like a sensible way to avoid the horrors of NAT.,
23:01.53halkunHere is a network map...
23:02.34halkunhttp://img26.imageshack.us/img26/6513/netc.png
23:02.45halkunThis is so I can keep my IP numbers stright
23:03.12*** join/#asterisk jksM (jks@193.189.93.254)
23:03.18halkunthe problem is that I can use the internet fine on the server, but nothing behind it works
23:03.38halkunI can ping the home PC and the home PC can ping the server
23:03.52halkunI can log into the Access point from the home PC
23:03.58halkunbut they can't get out to the internet
23:04.39halkunI used to have a D-link router and it worked fine
23:05.12halkunI think I'm just missing something to "turn on
23:05.25halkunI think I'm just missing something to "turn on" the gateway
23:05.36Belgarathhalkun: you do not have nat enabled
23:05.45*** join/#asterisk jks (jks@193.189.93.254)
23:06.35carrarhttp://www.howtoforge.com/nat_iptables
23:07.09halkunI'm using webmin
23:07.20carrarI'm sorry
23:07.24halkun??
23:07.32Belgarath:))
23:07.48carrarhttp://www.akadia.com/services/pppoe_iptables.html
23:07.58*** join/#asterisk Sargun (~Sargun@atarack/Staff/Sargun)
23:08.11carrarread that
23:08.36carrarand stop using webmin
23:09.30halkunmy pppoe is running fine
23:10.51halkunI used a commad that ran me though a wizard
23:11.25carrarthen wizard out a nat config
23:12.15halkunI used pppoeconf
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23:14.11halkunI'll use the howto you linked above
23:14.32carrargreat, lots of other google links for using pppoe & NAT
23:14.52carrarit's not too hard
23:14.58halkunhuh, I don't have a sysconfig directory
23:15.33halkunI have PPPoE all set up, I just need to turn NAT on
23:15.43carraryou don't "turn on nat"
23:16.10carrarread up
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23:23.26RadenNaikrovek, hey bro
23:23.40NaikrovekRaden: yo
23:23.48Radenwhat you doing online ?
23:23.55RadenI ordered that book :) should be here wednesday
23:24.12Naikrovekwhat book i forgot
23:24.19Naikrovekmerging vm snapshots
23:24.20Naikrovekwell
23:24.27Naikrovekwaiting for them to merge so i can start the VM again
23:24.49RadenLOL
23:24.51Radenthe pilot one
23:24.54Naikrovekoh yeah
23:24.57Naikrovekthat's a good one
23:24.58Radenfound on ebay for 28
23:25.02Naikroveknice
23:25.28Radenfound a flight school 30 miles away that trains with 172 intro flight $30 for an hour
23:25.33Naikrovekbingo
23:25.47Naikroveki found out my bladecenter doesn't have 2000W power supplies like I thought it did
23:25.52Radenthey have commercial pilot training and everything there look pretty repuatable
23:25.53Naikrovekso i got a new blade i can't use yet
23:26.05RadenOh wonderful
23:26.22Naikrovekyep
23:26.35RadenNeed to upgrade ?
23:26.36Naikrovekfound the power supplies I need on ebay for 45 but not sure bosses will go for htat
23:26.43Naikrovekyeah need to upgrade two of them at least (out of four)
23:26.53Radenwhy they dont like ebay ?
23:27.09Naikrovekbecause no one wants to use their own credit card for that kind of thing
23:27.09Naikrovekso\
23:27.16Naikrovekthey'll wind up paying $699 each
23:27.31Radenjust so it can be billed to the company
23:27.32Naikrovekwhatever it's not my money
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23:27.35Naikrovekyeah
23:27.36Radenwhy does the company not have a card ?
23:27.43Naikrovekbecause
23:27.48Radenwell ill buy them and bill them :)
23:27.49Naikrovekthe person that pays the bills never pays them on time
23:27.50NaikrovekEVER
23:27.58RadenLMAO
23:28.05Radentypical business
23:28.19Naikrovekshe's so ruthless when people owe her
23:28.27Naikrovekone day late and its one phone call per hour
23:28.34Naikrovekbut she blows people off for months
23:28.38RadenLMAO
23:28.42RadenROFL
23:29.11Naikrovekso as much as i'd like it if you bought the supplies, no way jose
23:29.37Naikrovekjust going to have to convince them to buy them on ebay or let a $6k blade sit there unused
23:29.39Radenwhy are they so horrid about paying lack of mulla ?
23:30.56halkunAHA!
23:31.12Naikrovekit's not "they" it's the accountant
23:31.17Naikrovekthat's just how she is
23:31.19halkunHere is how ou tuen NAT on  ---> /sbin/iptables -t nat -A POSTROUTING -s 192.168.0.0/255.255.255.0 -d 0/0 -j MASQUERADE
23:31.26Naikrovek"if you're loose with the money you're broke!"
23:31.49carraror need ot make more
23:31.59Naikrovekwe make plenty
23:32.01Naikrovekshe's just insance
23:32.02Naikrovekum
23:32.03Naikrovekinsane
23:32.07halkuntook a while to find it
23:32.48Naikrovekhalkun: yeah that makes your box a router i think
23:33.11halkunnow to reboot and see if it sticks :)
23:36.56ChannelZnot unless you put it somewhere to make it stick..
23:41.43Naikrovekjust was a guy in ##hardware who wanted to know how to fix components on motherboards.  eventually the conversation boiled down to "what's a multimeter"
23:42.03Naikrovekheh
23:48.45*** join/#asterisk halkun (~chatzilla@mke-66-97-112-72.milwpc.com)
23:48.55halkunYAY! Nat works!
23:49.47halkunnow, the question is. Should I plug in the 3102?
23:50.19halkunor should I wait till tomorrow to go though that headache
23:50.47halkunWell, maybe I should install asterisk on the server :P
23:52.46*** join/#asterisk romb_work (~romb@unaffiliated/romb-work/x-7222485)
23:56.51*** join/#asterisk doctorray (~ray@static-71-177-137-76.lsanca.fios.verizon.net)
23:58.04doctorrayI've had an issue a couple of times today where a group of phones is ringing (Dial SIP/101&SIP/102...etc) and when a phone is answered, it continues to ring.  using all Polycoms IP650.  anyone hear of that before?
23:58.11doctorrayI haven't been able to watch the console when it happens

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