IRC log for #asterisk on 20120304

00:06.15*** join/#asterisk k3asd` (~k3asd@87.18.90.26)
01:12.06*** join/#asterisk Ehtyar (ehtyar@pinky-and-the-brain.will-take.over-the-world.org)
01:12.31Ehtyarhi all. I'm using jitsi to make calls through my asterisk server. if i enable video and try to use it asterisk crashes
01:12.46Ehtyarcan I enable video passthrough at all? googling isn't helping much
01:13.47Ehtyari've tried directrtpsetup=yes
01:13.47Ehtyar<PROTECTED>
01:13.47Ehtyar<PROTECTED>
01:14.49*** join/#asterisk stix (~stix@x1-6-00-1e-2a-87-8d-aa.k1.webspeed.dk)
01:40.57*** join/#asterisk mattwj2002 (~matt@wikisource/pdpc.active.mattwj2002)
01:41.00mattwj2002hi guys
01:41.41mattwj2002what causes garbage on the line?
01:41.55*** join/#asterisk RypPn (~RypPn@unaffiliated/ryppn)
01:46.19*** join/#asterisk xarvox (~xarvox@31.44.231.96)
01:47.11xarvoxhi everybody! does anyone have any experience installing asterisk on a embedded system (such as a dd-wrt router)?
01:47.59mattwj2002I do I do
01:48.05mattwj2002openwrt
01:48.12xarvoxcool :)
01:48.15mattwj2002but
01:48.19xarvoxwhat hardware did you use?
01:48.21mattwj2002I have a question for you
01:48.23mattwj2002first
01:48.25xarvoxk :)
01:48.38mattwj2002what causes garbage on a line? jitter, delay, etc
01:48.58xarvoxhehe if all else equal, i would assume server performance :)
01:49.21mattwj2002oh
01:49.26mattwj2002that might be the router than
01:49.27mattwj2002:(
01:49.35mattwj2002okay well to answer your question
01:49.44mattwj2002NETGEAR WNDR3700
01:49.56mattwj2002dual band gigabit
01:49.57xarvoxhehe its not as much a single question, im mainly curious howto configure the darn thing..
01:50.11mattwj2002oh sure
01:50.26mattwj2002I don't know about dd-wrt
01:50.30xarvoxmy imagined setup is for family use
01:50.35mattwj2002I did openwrt
01:50.44mattwj2002I would assume dd-wrt is pretty similar
01:50.45xarvoxyeah, my hw doesnt support it
01:50.55xarvoxindeed, very similar, but still different
01:51.03mattwj2002okay here is what I did I installed the packages using opkg
01:51.11xarvoxi did with dpkg :)
01:51.12mattwj2002using telnet
01:51.12Nuggettelnet is eeeeeeevil!
01:51.18xarvoxi like it :)
01:51.20mattwj2002I mean ssh
01:51.22mattwj2002sorry
01:51.23mattwj2002:P
01:51.33xarvoxnp :)
01:51.42mattwj2002anyways
01:51.53xarvoxim at the point where the app actually starts, i get to the cli interface
01:52.01mattwj2002created the files and went into the gui and had it auto start
01:52.06xarvoxi can connect to it using asterisk -r
01:52.10mattwj2002nice
01:52.46mattwj2002so what is a particular question?
01:52.46xarvoxbut there is no gui (webpage) and the aditional one i installed (asterisk gui) doesnt work properly
01:52.54mattwj2002right
01:52.58mattwj2002you have to use the cli
01:53.03xarvoxindeed
01:53.22xarvoxbut i have no idea about asterisk.. thats my big problem.. :/
01:53.42xarvoxi do know i want to activate sip
01:54.14mattwj2002xarvox: do you live in the US?
01:54.22xarvoxnope, sweden
01:54.26mattwj2002bummer
01:54.30xarvoxhow come?
01:54.43mattwj2002in the US you can get free us calls incoming and outgoing with google voice
01:54.47mattwj2002sweden ah?
01:54.53xarvoxah i see.. :)
01:55.03mattwj2002I love Sweden never been there but I have a friend from there
01:55.10mattwj2002sounds nice
01:55.11mattwj2002:)
01:55.12xarvoxthats why i use sip, most ip phone carriers support it and its easy to configure clients
01:55.20mattwj2002cool
01:55.42xarvoxyeah, sweden is nice, alltho the politics is fucking things up, if you escuse my french ;)
01:55.51mattwj2002no problem
01:56.06mattwj2002you should come to the US if you like messed up politics
01:56.07mattwj2002hehe
01:56.12mattwj2002anyways
01:56.19xarvoxheh thats why im staying away.. ;)
01:56.30mattwj2002but we have Google Voice! :D
01:56.37mattwj2002:P
01:56.42xarvoxhaha indeed!
01:57.02mattwj2002sip is generally pretty easy
01:57.13mattwj2002I am going to setup a different asterisk box
01:57.24mattwj2002and see if it fixes my quality issues
01:57.42mattwj2002if it does....I am not going to recommend asterisk on my router in the future
01:57.47mattwj2002another point....
01:57.59mattwj2002at least in openwrt I can't speak for dd-wrt
01:58.08mattwj2002I couldn't get conferencing working :(
01:58.32xarvoxi know they suggest at least 400mhz pc as server.. :/
01:58.36mattwj2002it appeared the meetme package wasn't included
01:58.46xarvoxbut im hoping i can get away with my new asus router
01:59.24mattwj2002who knows....maybe sip will play nicer than gv
01:59.29mattwj2002:)
01:59.45mattwj2002it should you are probably going sip to sip instead of sip to gv
02:00.41xarvoxgv is something i havent tryed, but the issue is with connectivity, the demand of a specific software. and currently, us only... :P
02:31.44*** join/#asterisk fisted (~fisted@unaffiliated/fisted)
02:34.06Henchman21sweet just installed whocalled.us junk_lookup.agi
02:34.26Henchman21send them to my spam context
02:41.49ajgI've had a recent spammer that's forging callerid entries with 3 digit numbers
02:42.03ajgI'll get about 450 calls in the span of 3 minutes or so
02:42.36ajgI can't think of anyone legitimately using a 3 digit callerid.. I imagine it's probably safe to filter those en masse
02:42.58Henchman21good ideae
02:43.22*** join/#asterisk RypPn (~RypPn@unaffiliated/ryppn)
02:43.39ajgI mean all our internal extensions are 4 digits
02:44.09Henchman21i got this goofy CID logger so ill be able to see if anything wierd starts and i can take precautions
02:44.46Henchman21http://wiki.docdroppers.org/index.php?title=Asterisk_Call_Logger
02:44.46ajgjust avoid trixbox and fonality :)
02:45.07Henchman21got it to log all calls to var/log/asterisk/callid.log
02:45.15ajgoh neat
02:45.18ajgI may look into this
02:45.50Henchman21i had to Set(CallerIDString=${CALLERID(all)})
02:46.08Henchman21else it wasnt getting CID info
02:47.43*** part/#asterisk mattwj2002 (~matt@wikisource/pdpc.active.mattwj2002)
02:51.02Henchman21help fix a lil error http://pastebin.com/Y8KtXuJM
02:56.17Henchman21i guess so nevermind
03:16.49*** join/#asterisk jeffgus (~jeffgus@2001:470:f2eb:1::4)
03:40.23p3nguinhenchman21: I don't see that script advertised on their site.  How did you hear about it?
04:12.57Henchman21think its in the about
04:13.05Henchman21lemmie get a link
04:13.29Henchman21http://whocalled.us/about
04:13.36Henchman21yeah scroll down to asterisk scripts
04:19.40Henchman21pretty cool eh?
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04:36.37Henchman21any idea what my fxo port impedence should be?
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04:47.44leroybuckinghamHey guys, I'm trying to compile DAHDI and I'm running into an issue where it just won't build a few of the astribank binaries out of dahdi-tools, (ie, astribank_allow, astribank_tool)
04:48.15leroybuckinghamdo i need to do something to add astribank support?
05:25.06leroybuckinghamDerp, libusb-dev.
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11:24.40gain_hi all
11:26.43gain_I've installed asterisk with this guide http://wiki.openvox.cn/index.php/OpenVox_B100P_User_Manual_for_dahdi
11:27.14gain_so, I've compiled it cause my isdn card needs it
11:27.44*** join/#asterisk srini (user@180.215.25.91)
11:28.34gain_but I haven't the init script in /etc/init.d... is that correct?
11:29.19kaldemargain_: "make config" will install an init script.
11:30.27gain_kaldemar: the guide say only make, make install, make samples...
11:30.42gain_if I do make config now, have I to recompile it?
11:31.19kaldemar"make config" won't recompile or cause a need to do so.
11:31.34gain_kaldemar: thanks for the hint
11:40.54gain_and, another question: 'dahdi show channels' doens't show me any channel
11:41.36gain_I think that's because I don't include dahdi_channels.conf in chan_dahdi.conf correctly
11:42.00gain_the guide is not so clearly at this point
11:42.05*** join/#asterisk srini (user@180.215.25.91)
11:42.22sriniHi Room
11:43.00gain_it say to insert 'include dahdi_channels.conf' in chan_dahdi.conf, but don't say at which point of the file and with which syntax
11:46.24kaldemaryou should have #include <file>
11:47.20kaldemarthe inclusion point depends on the contents of your dahdi-channels.conf and chan_dahdi.conf
11:54.59gain_dahdi-channels.conf is created by dahdi_genconf
11:55.21gain_I dunno in which section of the file it must be placed... :(
11:55.41gain_uhm, tried at the beginning, but no success...
11:57.48kaldemartry under [channels]
11:58.20gain_kaldemar: trid it right now, and 'dhadi show channels' returns a list, so seems ok
11:58.22gain_thanks ;)
12:05.06*** join/#asterisk Srini (Srinivasa@180.215.25.91)
12:06.30SriniWhile I am trying to call I get to see :: channel 'SIP/2001-00000003' status is 'UNKNOWN' :: What does that actually mean? DialPlan issues?
12:12.01kaldemarthat doesn't mean much by itself. show CLI output for a whole call.
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14:51.22gain_hi
14:51.37gain_can I make a call directly from the asterisk cli?
14:52.22WIMPy'channel originate'
14:55.43gain_WIMPy: thanks
14:57.54*** join/#asterisk Serees (~Serees@91.183.70.42)
14:59.05Sereeshi, i'm having some issue with a 'legacy' ata... on the FXS connection, somebody here that could help?
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15:00.30Sereesit's very strange but I can recieve and accept incomming calls, and when a call is active voice from handset to other party is transmitted, but not the other way arround (silence on that side)
15:01.01Sereesalso making outcoing calls (unless forced in the ATA with auto offhook dialing) is not possible
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16:52.50gustohi
16:53.05gustosomeone has experiences with ENUM / NAPTR records and Asterisk?
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18:36.45SriniI dont know I can ask this question here or not! Sorry if I am offending anyone.... I have intention though! Is there any IRC channel for vicidial?
18:37.11Srini(I have no intention though)
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19:09.16p3nguinsrini: I don't think there is one for vicidial.
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19:28.37tehrabbitthey, can anyone help me test my fax out?
19:30.16tehrabbittI just need a test fax sent, if anyone can do it let me know
19:34.02gustohey
19:34.08tehrabbittsup
19:34.13gustoi have something
19:34.19gustobut not a fax
19:34.30tehrabbittlol what do you have?
19:34.37gustomy voip is still not working, but i found out something
19:34.40gustohttp://pastebin.com/Wujq54KL
19:34.53gustoi did dig srv _sip._udp.sipproxy.endesha.be-converged.com
19:35.02gustoand you see in the pastebin what i got
19:35.09gustosome ip addresses
19:35.49gustonow the question is ... if this server is sipproxy or registar here -> _sip._udp.sipproxy.endesha.be-converged.com. 8641 IN SRV 1 1 5060 pcscf-brln-de03.endesha.be-converged.com.
19:36.05tehrabbittgusto: honestly i'm not sure :-\
19:36.05gusto5060 is the port, so it does not look like proxy to me
19:37.16tehrabbittbbl gotta reboot
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19:58.46gustowhat does it mean when  sip show registry shows "  No Authentication " state?
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20:14.26gustodoes he always use TLS for TCP?
20:16.14gusto<PROTECTED>
20:16.34gustoERROR[9709]: tcptls.c:382 ast_tcptls_client_start: Unable to connect SIP socket to  Connection timed out
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20:56.25Scar-GHello everyone.
20:56.27Scar-GI want Asterisk to disconnect those calls which are being shown as 'connected' while they are actually 'ringing'
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21:13.22gustoi have a problem that i have a @domain.com in the username and either he responds to it as malformed or he takes only the part w/o domain name
21:13.36gustois it possible to have a username with @ in the name?
21:14.19gustois there a way to say to him not to interpretate that @?
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21:30.55p3nguin"he resonds to it"   <--- What are you talking about?
21:33.01Andeei respond to a lot of things
21:34.01p3nguinI have a tendency to respond to too many things.
21:34.58ChrisInSydneyMy ex wife said I never responded
21:35.08ChrisInSydneymornin' all
21:35.36WIMPySaid? Did you silence her?
21:35.39ChrisInSydneyp3nguin. I havent looked at that SPA DND issue yet. I should have some time today to check it out for you
21:36.39ChrisInSydneyWIMPy: Actually, she wasn't as bad as some girls I met after I left...in a hurry....for good reason too...but we'll save that story for therapy....or maybe Jerry Springer
21:37.37ChrisInSydneyThere are girls out there that you look at and you say "Damn, I wont get me som a that" and then they open their mouths and all you want to do is stick something in it
21:38.30ChrisInSydneyI have a good woman now. If I could clone her I would make a fortune
21:39.17WIMPyYou could still rent her out ;-)
21:41.07ChrisInSydneyWIMPy; You made me choke on my coffee
21:41.43WIMPyPress lips firmly together to increase scattering angle.
21:42.24ChrisInSydneyIts times like this I wish I still had CRTs. The glass is so much easier to clean
21:42.57ChrisInSydneyActually thinking about it, there must be a market for "easy to clean" screens
21:43.26WIMPyHmm. I haven't noticed a differens wrt cleaning.
21:43.52Andeeyou can get glass-fronted glossy screens
21:44.02Andeebut epic glare fail
21:44.26ChrisInSydneywhich is fine for those who surf the "glossy sites"
21:44.30ChrisInSydney;)
21:45.01ChrisInSydneywith that, I must type in a more profitable window
21:45.11ChrisInSydneyhaev fun
21:46.01gustoi am still getting SIP/2.0 403 Forbidden
21:46.06gustofrom my VoIP provider
21:46.49p3nguinSend the correct authentication credentials.
21:47.24gustowell, i am doing it all the time
21:48.02WIMPyThey seem to disagree.
21:52.43[sr]howdy WIMPy
21:52.48[sr]have a question for you :)
21:54.25p3nguingusto: Have you already pastebinned your configuration and the sip debug?
21:55.53[sr]WIMPy: its always better to use the kernel modules from mISDN than the vanila kernel, the ones in mISDN are always updated, the one's in the kernel may not be updated right?
21:58.27gustoi configured it according register => [transport://]user[@domain][:secret[:authuser]]@host[:port] and debug shows nothing special but 403 forbidden
21:59.18gustoso udp://user@domain:secret:phonenumber@host
22:07.32gustowhat is better 403 forbidden or 401 unauthorized?
22:07.45gustowhen i use my username for user i get 403 forbidden
22:08.13gustowhen i use my phone number for user then it takes longer gets a timeout with 401
22:08.20gustounauthorized
22:08.34gustowith username 403 forbidden comes in an instant
22:11.24gustoand when i am doing it exactly other way round with domainname as well in authuser it has the same effect as username in user and so 403 in an instant
22:14.17gustohmm
22:14.43gustoWHAT IF the secret is not a password, but an MD5secret, because it is like bshgsgsgwig_ and so on ...
22:15.44gustobut i think for md5 it is too short
22:15.46*** join/#asterisk Dein (~dein@31.23.243.239)
22:17.51p3nguinI would think 403 is worse.`
22:18.19p3nguinI think 403 means that the user does not exist and therefore is forbidden.
22:18.35p3nguinI think 401 means your user is allowed, but you simply did not authenticate yet.
22:19.02p3nguinI could be misinterpreting the messages, though.
22:21.37gustowhat is this:
22:21.38gustoWWW-Authenticate: Digest realm="ims.telefonica.de",domain="sip:HSS@ims.telefonica.de",nonce="20e6a291b0f00a1bc68208ec7a0cccc2",stale=false,qop="auth",algorithm=MD5
22:21.41gustohm?
22:22.05WIMPy[sr]: Usually the vanilla kernel should be ok.
22:22.47[sr]WIMPy: i may preffer to compile them, what do you say?
22:23.14[sr]WIMPy: this doesn't have any script to load modules like dahdi has right? I have to load them manually, any special order?
22:24.56WIMPyThey should usually be loaded automatically.
22:25.47WIMPyVia some rc.modules, udev or whatever you may have.
22:30.21[sr]strange, they are not
22:35.56*** join/#asterisk netman (netman@195.52.78.188.dynamic.jazztel.es)
22:36.29gustoi just removed the password and authuser and there is no effect
22:36.52gustoso he did not even come to the password section when authorizing with phone number
22:37.15gustoshow registry says that auth was sent
22:37.40gustoand registration times out and tries again
22:37.50[sr]WIMPy: didn't had udev installed ... lol dumb me, it's a new machine from scratch, with both dahdi and LCR, what i alway wanted!
22:38.23*** join/#asterisk tapout (~karen@unaffiliated/tapout)
22:38.38WIMPyBoth at the same time?
22:39.05tapoutanyone know a really cheap.. wifi+sip based phone you can use at your house that is like a handheld that is rechargable?  I looked at kirk brands and they want a server, the handheld and like 500+ bucks
22:39.28[sr]WIMPy: yes
22:39.29gustop3nguin: you were right with your interpretation, when i am putting bullshit or another tel number it does say forbidden in an instant
22:39.57*** join/#asterisk RickCogley (~anonymous@ntkngw522154.kngw.nt.ftth.ppp.infoweb.ne.jp)
23:03.33*** join/#asterisk rossand (~aross@CPE485b390978ce-CM00222ddf42dd.cpe.net.cable.rogers.com)
23:04.16*** join/#asterisk ketas (~ketas@2001:ad0:91f:0:290:27ff:fea6:a66a)
23:04.39[sr]sleep time
23:06.35*** join/#asterisk SunTsu (miyamoto@unaffiliated/suntsu)
23:15.37gustono ideas?
23:21.12tapoutanyone?
23:21.24tapoutanyone know any good hardware that is asterisk compatible for home/personal use?
23:23.54*** join/#asterisk BuenGenio (~Gene@059148208218.ctinets.com)
23:27.01tapoutwhat's the best ip-dect phone that is cheap for home use?
23:33.15ChannelZI don't think I've ever seen a cheap wireless SIP phone
23:38.12ChrisInSydneytapout: The SnomM3 has been discontinued and has been floating around reasonably priced Sub AU$100
23:38.30ChrisInSydneyOtherwise a Siemens Gigaset works OK. C470 does transfer properly
23:38.52ChrisInSydneynot cheap, but better than an ATA and a cordless
23:38.55ChrisInSydneymuch better
23:54.44gustohow do i change Authorization: Digest username?

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