IRC log for #asterisk on 20111014

00:02.21SeRiI have set to drop silently.
00:02.37SeRiusually hangs most scanners or attempt
00:03.25p3nguinAs far as I know, pf only blocks or accepts.  There's nothing in the middle.  Of each of those, you can log or not log.
00:04.39*** join/#asterisk luckman212_ (~do-not-re@static-108-46-165-162.nycmny.fios.verizon.net)
00:05.01luckman212_anybody in here have any experience configuring Patton SmartNode gateways?
00:29.38*** join/#asterisk seraphie (~erin@75.76.38.159)
00:30.15*** join/#asterisk coppice (~chatzilla@121.203.226.162)
00:32.34*** join/#asterisk p3nguin (~xwQ5kwYl6@cobalt.esxi.hosts.a2infotech.com)
00:32.35*** join/#asterisk luckman212 (~irc@pool-108-41-8-176.nycmny.fios.verizon.net)
00:39.11cuscohi...
00:39.31cuscop3nguin: im trying to test gtalk now, but calls from gtalk to asterisk are not comming in
00:39.39cuscocontext is set..
00:39.52cuscoI don't know what Im missing
00:40.10cuscojabber debug shows the call in
00:48.08*** join/#asterisk snuff-work (~snuffy@210.9.82.197)
00:48.39p3nguinI'd have to see your jabber.conf, gtalk.conf, and the relevant context in extensions.conf.
00:50.05cuscook hold
00:52.17*** join/#asterisk RypPn (~RypPn@unaffiliated/ryppn)
00:53.21cuscop3nguin: http://paste.debian.net/136342/
00:54.43p3nguinI see two things that stand out to me.
00:55.25p3nguinYou specify a context of google-in-guest, but you don't have that context in the dial plan.
00:55.52cuscowell the buddy calling is not a guest
00:55.54p3nguinAnd the usernames in jabber.conf are typically yourid@gmail.com/Talk
00:56.12cusco/Talk?
00:56.50p3nguinDid you bother reading the wiki that explains how to set it up so it works?
00:58.42cuscoyes
00:58.52cuscowell http://www.voip-info.org/wiki/view/Asterisk+Google+Talk
00:59.03cuscoright /talk
00:59.58cuscotho it works dialing to buddies
01:01.29cuscoso I did those two thingies
01:01.35cuscostill not getting the call
01:01.45cuscoi replicated the context and named it google-in-guest
01:04.11*** join/#asterisk p3nguin (~xwQ5kwYl6@cobalt.esxi.hosts.a2infotech.com)
01:04.17cuscowb
01:04.21cuscoguess u didn't read
01:04.50cuscoso I added the context (replicated from google-in and changed the name) and added /Talk
01:04.55cuscostill not getting the call in
01:04.57cuscodunno why
01:05.04cuscoit shows on jabber debug
01:05.14*** part/#asterisk snuff-work (~snuffy@210.9.82.197)
01:06.51cuscoit shows: to="merdosa@gmail.com/TalkD4B81223
01:07.00cuscoinstead of /Talk
01:07.02cuscoalone..
01:07.18cuscoI dunno what to look for
01:07.35p3nguinI followed the wiki and mine works.
01:07.47cusco:(
01:09.18p3nguinI didn't notice if you said you followed the wiki or not.
01:09.44cuscoI read several howtos but I just checked everything from the wiki
01:09.59cuscoI obviously missed te /Talk
01:10.08cuscoI can make calls froma sterisk to gtalk
01:10.20cuscobut I don't see incomming
01:11.36p3nguinDid you switch your calls to go to google chat only?
01:11.45p3nguinAnd don't login to chat in your email.
01:12.36cuscoswitch calls to go to google chat only?
01:12.42cuscodidn't understand that question
01:12.57cuscoI logged in at a time, but closed the browser (2 days ago)
01:13.08cuscoI can login and click to terminate all other sessions... hold
01:13.21p3nguinWhen you login on voice.google.com, you go to the setup...
01:13.30p3nguinGo to the section where you set your phone number to forward calls.
01:13.34cuscobut asterisk should be the only one logged in
01:13.47p3nguinUncheck any forwarding numbers.  Check only Google Chat.
01:13.53cuscoow... I don't think I did let me check
01:14.12*** join/#asterisk Naikrovek (~mbluth@unaffiliated/naikrovek)
01:14.43Naikrovekit is amazing to me how much being an IRC op can get to someone's head...
01:14.53Naikrovekno one in here, though.  at least not to my knowledge.
01:15.29Naikrovekone of the ops in #minecraft is *drunk* with 'power'.  lol
01:16.47cuscoI don't find any setup options on voice, besides the billing
01:18.37p3nguinClick on your GV phone number.  Click on the Calls tab.
01:18.47*** join/#asterisk vinhdizzo (~vinh@vqn-portege.ics.uci.edu)
01:19.01cuscoyes I found that on google help, but no... wait I don't have a gv phone number
01:19.06cuscoI need to click upgrade first?
01:19.33p3nguinI'm confused.  How were you planning to get calls without a phone number?
01:19.58cuscoerr... I was trying only gtalk not google voice
01:20.30cuscoso... gmail has chat box, and it allows talk (with a browser plugin)
01:20.39cuscoI was trying to make that work with asterisk
01:20.59cuscoand it works asterisk -> google talk
01:21.18cuscok let me click on upgrade
01:21.21p3nguinAh, I thought you were asking about google voice.
01:22.08p3nguinI'd guess you'd have to make sure you don't have any other clients logged in, and then any calls to chat would arrive on your only logged-in client (asterisk).
01:22.26cuscook Im guessing the same
01:22.43cuscojabber debug does read the incomming call
01:23.44cuscoit says this is the only location
01:23.46cuscoI logged out
01:24.11cuscoow
01:24.14cuscoim getting something now
01:24.21cuscoyay!
01:25.24*** join/#asterisk skrull (skr@189.73.184.138)
01:26.48p3nguinDid you find out there was another client connected?
01:27.10cuscook it works
01:27.19cuscono it said only me
01:27.20cusco...
01:27.22cuscobut well
01:27.25cuscothat did something
01:27.30cuscologging in and clicking logout
01:27.30p3nguinWhat made it go?
01:27.37cuscothat
01:27.39cuscolol
01:27.44cuscothanks for patience :D
01:28.08cusconow I still have to check this out when outside nat on both ends
01:29.22cuscoand my objective is to be able to let people use gmail plugin as sort of a softphone
01:30.33cuscoabout google voice...
01:30.45cuscoit costs money to have a number, right?
01:30.52p3nguinNo.
01:34.00cuscohmm google is returning error in my request
01:34.05cuscowill look at that another time
01:34.57cuscoalso... no jabber recieve chat unless i'm in a call right?
01:39.21*** join/#asterisk cnu (cnu@the.ultimate.lamer.la)
01:40.55cuscohttp://i.imgur.com/76RkJ.png
01:41.52p3nguinI'm not sure, to be honest.
01:42.29cuscook
01:43.39*** join/#asterisk ggd (~ggd@pool-173-72-204-39.clppva.fios.verizon.net)
01:48.36*** join/#asterisk adolfomaltez (~taro@190.62.254.245)
01:49.01*** join/#asterisk KavanS (~KavanS@LINBIT/KavanS)
02:11.10*** join/#asterisk p3nguin (~xwQ5kwYl6@cobalt.esxi.hosts.a2infotech.com)
02:36.25*** join/#asterisk aberrios (~aberrios@195.171.4.82)
02:39.11*** join/#asterisk p3nguin (~xwQ5kwYl6@cobalt.esxi.hosts.a2infotech.com)
02:41.32*** join/#asterisk troyt (~troyt@PariahZero.broker.freenet6.net)
02:57.33*** join/#asterisk dandate2 (~dan@124.6.157.210)
02:57.43*** join/#asterisk jayson_r (~jayson@cpe-071-076-046-081.sc.res.rr.com)
02:59.36dandate2for some reason when a DID provider's forwarding is set to sip@ip  my inbound routes don't recognize it, but if i remove the sip@ then it works?
03:00.06*** join/#asterisk p3nguin (~xwQ5kwYl6@cobalt.esxi.hosts.a2infotech.com)
03:02.23*** join/#asterisk jayson_r (~jayson@cpe-071-076-046-081.sc.res.rr.com)
03:05.39*** join/#asterisk SeRi (~seriosly@c-76-31-169-54.hsd1.tx.comcast.net)
03:18.38*** join/#asterisk ChannelZ (channelz@burner.com)
03:20.20*** join/#asterisk radic (~radic@dslb-178-007-133-244.pools.arcor-ip.net)
03:21.44*** join/#asterisk p3nguin (~xwQ5kwYl6@cobalt.esxi.hosts.a2infotech.com)
03:26.42*** join/#asterisk seraphie (~erin@75.76.38.159)
03:34.58SeRip3nguin, you avail?
03:36.55*** join/#asterisk adolfomaltez (~taro@190.62.206.205)
03:39.47*** join/#asterisk Rufus (Rufus@unaffiliated/rufus)
03:43.15*** join/#asterisk adolfomaltez (~taro@190.62.239.193)
03:50.11*** join/#asterisk p3nguin (~xwQ5kwYl6@cobalt.esxi.hosts.a2infotech.com)
03:51.41p3nguinseri: I'm here if you can get through the DoS.
03:55.55SeRiwtf
03:55.58SeRiseriously?
03:56.15SeRip3nguin, whats going on?
03:56.32SeRisomebody DDoS you?
03:56.48p3nguinDisgruntled employee.
03:57.01p3nguinNo, DoS.
03:57.09p3nguinDDoS is from multiple hosts.  This attack is from a single host.
03:57.18SeRibastard
03:57.30p3nguinIt's just a high-bandwidth UDP flood.
03:57.37SunTsup3nguin: then ask your ISP to blockhole that ip
03:57.45SunTsublackhole even
03:57.58p3nguinI wish it were that easy.
03:58.23SeRi^^
03:58.30p3nguinseri: Yeah, once I can prove he's behind the attack, he'll probably be fired.
03:58.35*** join/#asterisk dijib (~nobodysho@bas10-kitchener06-1176001555.dsl.bell.ca)
03:58.45SeRiI called once comcast with a similar issue and they had no clue wtf whats going on lol
03:58.53SeRithey told me to email security...
03:58.59p3nguinYep, they don't know anything.
03:59.07p3nguinDenial of what?  :/
03:59.11SeRilmao
03:59.24p3nguinOr, "What kind of service?"
03:59.41p3nguinDenial of Service, sir.
03:59.48p3nguin"I'm sorry, we don't... have that."
04:00.17SeRirofl
04:00.24dijibp3nguin, is the voip.ms customer portal getting ddos'ed or something or is it working for you? i cant get into order another iNum DID
04:00.36SeRidijib, working for me
04:00.49p3nguinNah, they're hammering one of my systems.
04:00.49SeRiI was just there getting a 5th number
04:00.50dijibsrsly
04:01.01dijibwho is?
04:01.12p3nguinI changed the IP address, so they are tracking it by my web site host name.
04:01.31p3nguinI'm not going to say his name until I have the proof in my hand that he is responsible for it.
04:01.52dijibyikes. then sue his ass
04:02.11SeRidijib, not that simple... the law does not care unless you have a few mill in loses
04:02.16p3nguinHe is pissed because I refuse to give him root access to a server that he used to have root access to.
04:03.03dijibi think the law... well here anyways would be interested in hearing the case even for a couple of hundred
04:03.09SeRip3nguin, fuck him... look at him tomorrow and with an evil looked laugh at him and whisper (n00b)
04:03.33p3nguinThe DoS is coming from utoronto.ca
04:03.59p3nguinI called the abuse phone number and left some messages, but got no response yet.  I figure tomorrow I need to make a few more calls.
04:04.00SeRidijib, seriously? cyber crime for a few hundred bucks?
04:04.19SeRip3nguin, good luck. I know it can be painful
04:04.20dijibsure why not? at least small claims
04:04.24dijibutoronto.ca
04:04.32dijibis the person you think is there there?
04:05.26SeRilol mhhhh ill stop there. read a bit about cyber crime. small claims can not touch it.
04:05.48SeRiany who p3nguin can you defer it with pf?
04:05.52p3nguinOnce I can prove who is responsible, I will file criminal charges for endangering my family by DoSing my system where I have an internet phone system.
04:06.04SeRiwell that you can do
04:06.33p3nguinI have better luck absorbing the packets rather than blocking them.
04:06.34SeRirendering emergency services useless. Thta has a criminal intend
04:06.39p3nguinI accept them and do not reply.
04:07.07p3nguinBy blocking them, the impact is greater.
04:07.08SeRip3nguin, Cant be much. I mean most be coming from a consumer line given that you can still be online
04:07.15SeRimust*
04:07.36p3nguinNo, it's from utoronto.ca
04:07.36dijibim 2hours from uofT
04:07.52p3nguinI'm sure they have a fairly large pipe.
04:08.02SeRimhhhhhh
04:08.40dijibyeah one of the big pipes goes to front street which is about 3-6 blocks away
04:08.58*** join/#asterisk r33dtard (~r33dtard@gateway/tor-sasl/r33dtard)
04:09.26dijibutoronto.ca is university of toronto or u of T
04:09.33SeRiis pondering
04:09.46SeRip3nguin, shut the source down.
04:09.47p3nguinI knew he was upset, but I didn't know he was a packet kiddie.
04:09.56SeRiyou can call the IT dep in the edu
04:10.04dijibcant you iptables fail2ban him?
04:10.32SunTsudijib: as long as his uplink is filled up he can't do anything on his side
04:10.47p3nguinHe's upset because his boss told me to change the root password and keep him off the server.
04:10.47p3nguinSo I put it in lockdown.
04:11.07SeRiSunTsu, you beat me to it
04:11.08SeRilol
04:11.25dijibthen ifconfig eth0 down
04:11.28SunTsuU of Toronto = big pipe, p3nguin = considerably smaller pipe -> congestion -> bummer
04:11.30dijibno?
04:12.01SeRidijib, look at what SunTsu is saying
04:12.23SeRithe only people that can put a stop at it is the ISP by defering the traffic or shutting the source down
04:13.19SunTsup3nguin: I'd try to get in touch with your ISP's NOC/NMC/whatevertheycallit, ask them to nullroute it. 1st level support normally will put you through if you throw around terms they don't understand
04:13.24SeRithe damage is done. DoS is to take the target down by not allowing him to be online. some times it causes larger issue than that like crahsed systems etc...
04:13.31p3nguinI wouldn't be surprised if he's here watching what I'm saying about him.
04:13.34dijibnetwork operations
04:13.48p3nguinHe was on here every day until I didn't give him what he wanted, then he quit.
04:14.50p3nguinBut when you're a moron who does not necessarily need to be touching computers, bosses tend to give other people your job.
04:15.51SunTsup3nguin: looks like a wise choice to not have him being uid0 on your box
04:16.01p3nguinI use the interface he's hitting for connecting to IRC.  If I shut it down, I won't be able to talk to you.
04:16.46SeRithat sucks... what a punk.....
04:16.56SunTsup3nguin: and he'll prolly move on to another ip you own
04:17.14p3nguinOr as he put it, his box.
04:17.17p3nguinlamer
04:17.29p3nguinHe has done two so far.
04:17.41SeRip3nguin, well let me entreating you with some dial plan glory.... :P
04:17.42SeRihttp://pastebin.com/Unu1Ur0r
04:19.07SeRip3nguin, I hope you can catch the little fucker and burn him.
04:19.09p3nguinTwo things to say about that.
04:19.48SeRihahaha I knew it was all fucked up :P I thought I put it out there for you to laugh
04:20.14SeRiwell I ddint know. I just knew you where going to say something :)
04:20.19p3nguinLine 3 does not make much sense to me.  And Dial(Tech/peer/exten)
04:22.19p3nguinIf you fix the pipe... line 3 says to me, if the DIALSTATIS is CHANUNAVAIL, go to a priority label the same as whatever number you just entered; if it is something else, go to the failover label, where you should change the syntax of the dial.
04:22.24SeRiwell I am just trying to come up with a good way to fail over :)
04:22.30SeRinot doing a good job at it I guess :P
04:22.56p3nguinWhat will your failover be?
04:23.07p3nguinIAX2 on VoIP.ms?
04:23.18SeRias a test yes
04:23.23SeRijust testing now
04:23.35SeRieventually at some point in the extended future another provider :)
04:23.44p3nguinDial(IAX2/voipms/1${EXTEN})
04:23.54p3nguinWhere voipms is a peer in iax.conf.
04:24.25SeRiyes I just didt it that way for now. I clear all the "cluster fuck" :)
04:24.52p3nguinIf you don't have time to do it right the first time, when will you have time to redo it later?
04:25.42SeRiwell I am just experimenting to see if it works. I have to learn some how ;)
04:38.03drudge-goneman
04:38.09drudge-gone.....effen china
04:38.23drudge-goneand thier effen stunts at 930pm
04:38.42drudge-gonei told this customer not to cheap out on a firewall
04:38.59ChannelZthe whole country should be disconnected from the net
04:39.08drudge-goneand afrinic
04:39.11drudge-goneagreed
04:39.31drudge-gonesome customer got hacked
04:39.43drudge-gonei got called home from the bar, *sigh*
04:40.42drudge-gonethey are lucky they dont pay for INTL or LD... locla only
04:41.05drudge-gonethey are too small to afford some $15k+ phone bill for being hacked
04:46.38dijibnuke em
04:50.21*** join/#asterisk SteveWilliams (~SteveWill@122.160.52.178)
04:50.56SteveWilliamsHi All!
04:51.41SteveWilliamsIs there a way I can call an HTTP page from the dialplan in asterisk?
04:58.10*** join/#asterisk seraphie (~erin@75.76.38.159)
05:00.27[TK]D-FenderSteveWilliams: System(curl http://....)
05:03.51dijibfor what though?
05:05.20*** join/#asterisk kaldemar (~kaldemar@unaffiliated/kaldemar)
05:11.56*** join/#asterisk werdan7 (~w7@freenode/staff/wikimedia.werdan7)
05:21.50*** join/#asterisk adeel (~adeel@184.175.36.92)
05:25.52*** join/#asterisk freeman_u (~freeman@193.110.114.54)
05:38.03*** join/#asterisk KavanS (~KavanS@LINBIT/KavanS)
05:44.22*** join/#asterisk mintos (mvaliyav@nat/redhat/x-pitehovgvivdfyid)
05:56.45*** join/#asterisk d_preston215 (~chatzilla@173-12-4-137-panjde.hfc.comcastbusiness.net)
06:11.05*** join/#asterisk Ad-Hoc (~nimbus@194.30.241.105)
06:16.03*** join/#asterisk schmidts (~schmidts@Lo20-rt01.lm33.sil.at)
06:16.04schmidtsgood morning
06:19.56*** join/#asterisk freeman_u (~freeman@193.110.114.54)
06:24.22*** join/#asterisk sgimeno (~sgimeno@163.117.206.10)
06:26.11*** join/#asterisk sgimeno (~sgimeno@163.117.206.10)
06:39.54*** join/#asterisk oej (~olle@ns.webway.se)
06:39.59*** join/#asterisk cerberus_za (~coert@41-134-110-162.dsl.mweb.co.za)
06:59.07*** join/#asterisk darkbasic (~quassel@host37-245-static.119-2-b.business.telecomitalia.it)
07:02.07*** join/#asterisk waschtl (~waschtl@HSI-KBW-078-043-090-014.hsi4.kabel-badenwuerttemberg.de)
07:03.58*** join/#asterisk ollii (~risker@2001:470:1f15:1384:62eb:69ff:fe31:b4)
07:04.28*** join/#asterisk elred_ (~elred_@unaffiliated/elred-/x-5010831)
07:11.44*** join/#asterisk gerhard7 (~gerhard7@82-171-103-215.ip.telfort.nl)
07:14.19*** join/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190)
07:15.00*** join/#asterisk ChannelZ (channelz@burner.com)
07:21.34*** part/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190)
07:22.19*** join/#asterisk asterisk-Tester (~RAMYT@193.227.170.247)
07:25.32*** join/#asterisk Azrael808 (~peter@212.161.9.162)
07:30.49*** join/#asterisk gr0mit (~tim@extrt.txrx.org.uk)
07:34.55*** join/#asterisk Hanumaan (~Hanumaan@92.74.225.193)
07:35.03*** join/#asterisk darkbasic (~quassel@host37-245-static.119-2-b.business.telecomitalia.it)
07:47.05*** join/#asterisk k3asd` (~k3asd@static-94-32-127-180.clienti.tiscali.it)
07:50.59*** join/#asterisk tamiel (~tamiel@213.30.183.226)
07:55.35*** join/#asterisk irroot (~irroot@pbx.distrotech.co.za)
08:02.45irrootWIMPy ASTERISK_VERSION 10808 <- 1.8 but any SVN has 999999 this is confuzzeling
08:18.40*** join/#asterisk Faustov (~fst@gentoo/user/faustov)
08:25.13*** join/#asterisk Nasga (~Nasga@186.212.10.93.rev.sfr.net)
08:28.06*** join/#asterisk sunfone (~jeff@c-67-184-36-158.hsd1.il.comcast.net)
08:40.14*** join/#asterisk engrxyz (~fgdfgfdg@212.23.51.7)
08:47.43*** join/#asterisk Dovid (~Dovid@213.8.121.90)
08:48.02*** join/#asterisk jkroon (~jkroon@41.55.231.116)
08:56.30*** join/#asterisk eject_ck (~eject@62.205.134.210)
09:04.38*** join/#asterisk SteveWilliams (~SteveWill@122.160.52.178)
09:05.03SteveWilliamsHi All! Good Morning!
09:07.34irrootSteveWilliams top o the mornin
09:10.58SteveWilliamsI would like to access the ${EXTEN} of a context from another context. Is that possible? Please help.
09:11.47SteveWilliamsThe other context is a macro and being called by the first context whose ${EXTEN} I would like to access.
09:17.59*** join/#asterisk SteveWilliams (~SteveWill@122.160.52.178)
09:18.41SteveWilliamsI would like to access the ${EXTEN} of a context from another context. Is that possible? Please help.
09:18.44SteveWilliamsThe other context is a macro and being called by the first context whose ${EXTEN} I would like to access.
09:23.25SteveWilliams'
09:27.03SteveWilliams'
09:28.42GuggeSET(IWANTTOUSETHIS=${EXTEN}) before you call the macro
09:28.52Guggeor something like that :)
09:34.05irrootindeed EXTEN is for the "line" in the dialplan its excuting at the time
09:34.21irrootyou will need to set it or pass it as a ARG to the macro
09:34.52irrootMacro(mymacro, ${EXTEN})
09:35.14irrootrefer to it as ${ARG1} in the macro
09:35.58*** join/#asterisk SteveWilliams (~SteveWill@122.160.52.178)
09:36.16SteveWilliams'
09:38.50SteveWilliams<PROTECTED>
09:39.08kaldemaryou don't need to pass it to the macro, just use MACRO_CONTEXT, MACRO_EXTEN, and MACRO_PRIORITY. they store the calling ones.
09:39.23SteveWilliamsokay
09:39.32kaldemarwhat do you mean by access?
09:40.20SteveWilliamsi am dialing a number from one context. when it connects, i pass it to another context which makes the callee listen to a voice message
09:40.35kaldemarand then?
09:40.55SteveWilliamsi would like the macro retrieve the phone number and log it onto the database
09:41.08SteveWilliamsthe my sql database
09:41.13kaldemarwhy do you want the macro to do it?
09:42.07SteveWilliamsi would like the macro to store the number in my mysql database along with the timestamp etc
09:42.27kaldemarSteveWilliams: i repeat, why do you want the macro to do it? why not do it in the extension?
09:43.17kaldemarif you store the things you want in the extension already, you can play a file with the A() option of Dial application. and you won't even need a macro.
09:43.45kaldemarbut if you insist in using a macro, you have the original extension stored in MACRO_EXTENSION variable.
09:43.54SteveWilliamsthe macro gathers some data from the callee. i want that to be stored with the phone number
09:44.05SteveWilliamsin my sql database
09:45.04SteveWilliamsi have multiple files to play and various Read functions to go with them
09:52.59SteveWilliamsThanks, I used the MACRO_EXTEN. It works! Thank you kaldemar !
10:02.49*** join/#asterisk ayrjola (~ayrjola@89.18.236.11)
10:09.30*** join/#asterisk Hyperbyte (jan@middelkoop.cc)
10:11.09HyperbyteHi!  I have a dialplan for incoming calls, which dials all phones (peers) until someone picks up
10:11.52HyperbyteI would like to make it so that, it doesn't try to call the phones that are already in a call.  Right now when I'm in a call, and Asterisk receives another, it keeps trying to ring me.
10:12.29HyperbyteI'm considering programming a feature into our softphones, which rejects calls when already in one, but I'd prefer to do this on the server side.
10:14.09kaldemarHyperbyte: first Dial(Local/exten@contex&Local/exten2@context...) and then use GROUP functions in each extension to limit calls to 1.
10:14.46kaldemarsounds like you could a queue though.
10:16.37Hyperbytekaldemar, if I limit calls to 1 like that, can they still make outgoing calls when already in a call?
10:18.46*** join/#asterisk MarKsaitis (~MarKsaiti@195.59.185.18)
10:31.37*** join/#asterisk irroot (~irroot@pbx.distrotech.co.za)
10:32.20kaldemarHyperbyte: if you want to.
10:33.21kaldemarHyperbyte: if you set and check a group only in the extension, it has no effect to calls from the phones.
10:44.02*** join/#asterisk petern_ (~petern@lachesis.fuzzle.org)
10:45.30petern_hi, do any systems exist for speech recognition with asterisk, for data entry type situations where the input could feasably be anything, not waiting for a list a keywords...
10:48.02*** join/#asterisk wonderworld (~ww@port-92-201-118-207.dynamic.qsc.de)
10:48.35*** join/#asterisk puzzled (~patrick@puzzled.xs4all.nl)
11:13.30*** join/#asterisk ChrisInSydney (~Chris@60-242-81-231.tpgi.com.au)
11:14.48*** join/#asterisk gerhard7 (~gerhard7@82-171-103-215.ip.telfort.nl)
11:22.08*** join/#asterisk jacc0 (~jacc0@D522448D.static.ziggozakelijk.nl)
11:22.12jacc0hoi all
11:22.17jacc0almost weekend!!!
11:22.24jacc0:)
11:27.19*** join/#asterisk jsjc_ (5021ece2@gateway/web/freenode/ip.80.33.236.226)
11:28.55*** join/#asterisk TimeRider (~steve@host-92-27-131-175.static.as13285.net)
11:29.19jsjc_hello I wonder I have this issue Unable to create channel of type 'DAHDI' (cause 0 - Unknown)
11:29.23jsjc_because dahdi channel is already in use
11:29.45jsjc_is there nyway I could in my dialplan create a condition if DIAL returns that error the dial trough this other channel?
11:37.21ayrjolacheck channel variable DIALSTATUS after dial
11:41.20kaldemarjsjc_: what is the other channel?
11:52.51*** join/#asterisk timahvo1 (~rogue@41.215.1.35)
12:00.42*** join/#asterisk rgsteele (~rgsteele@apo.aweber.com)
12:09.21jsjc_kaldemar: it is a sip channel.
12:11.02*** join/#asterisk Dovid (~Dovid@213.8.121.90)
12:12.24*** join/#asterisk nunne (~nunne@static-213-115-116-75.sme.bredbandsbolaget.se)
12:12.40jsjc_ayrjola: what you mean by checking the variale after dial? could you give me an example?
12:12.54*** join/#asterisk [TK]D-Fender (~aoulton@216.191.106.162)
12:13.12nunneDoes anyone know a good way to make asterisk issue a shell script/program etc. upon recieving a new voicemail (thinking of sms notification with the help of wget to a online sms service)
12:13.58ayrjolaif dialstatus==CONGESTION, then DIAL(SIP/.....)
12:14.28ayrjolacheck in asterisk cli> core show application Dial
12:14.44ayrjolathere you get information about DIALSTATUS values
12:16.42ayrjolaif I remember correctly DIALSTATUS reports congestion if no DAHDI channels available
12:16.50*** join/#asterisk Naikrovek (~mbluth@unaffiliated/naikrovek)
12:24.02nunneahh, just found externnotify.. no worries :)
12:24.56*** join/#asterisk xSmurf (~MrSmurf@unaffiliated/mrsmurf)
12:32.42*** join/#asterisk irroot (~irroot@pbx.distrotech.co.za)
12:44.54*** join/#asterisk wesphillips (~wphill04@137-237-233-124.harris.com)
12:46.02*** join/#asterisk bukansesiapa (~me@115.178.55.123)
12:47.08*** join/#asterisk seanbright (~sean@asterisk/contributor-and-bug-marshal/seanbright)
12:50.57*** join/#asterisk SuperNull (~FreeManof@24-148-101-238.ip.mhcable.com)
12:59.48*** join/#asterisk Vilius_Invade (~Vilius_In@178.78.119.76)
13:00.13Vilius_Invadehi Guys, is it possible to setup loopback SIP trunk?
13:00.35*** join/#asterisk SteveWilliams (~SteveWill@122.160.52.178)
13:02.06SteveWilliamsI have added a function to the func_odbc.conf file. Even after a cold reboot, asterisk says that the function does not exist. Please help
13:03.37wdoekes2forget the prefix?
13:05.37*** join/#asterisk ickmund (~ickmund@cli-5b7e85e2.bcn.adamo.es)
13:07.58*** join/#asterisk SteveWilliams (~SteveWill@122.160.52.178)
13:08.15SteveWilliamsI have added a function to the func_odbc.conf file. Even after a cold reboot, asterisk says that the function does not exist. Please help
13:09.00SteveWilliams'
13:11.21kaldemar~ask
13:11.21infobotQuestions in the channel should be specific, informative, complete, concise, and on-topic.  Don't ask if you can ask a question first.  Don't ask if a person is there; just ask what you intended to ask them.  Better questions more frequently yield better answers.  We are all here voluntarily or against our will.
13:11.30kaldemarshow your dialplan and the function.
13:12.53sehhhey people
13:12.55sehhearlier today, I was asking for help with sending DTMF to the phone line, in order to enable/disable various features provided by my phone provider. I'm using ISDN lines with mISDN (chan_misdn). We found that its possible to do that with _MISDN_KEYPAD. Unfortunately, while this seems to work for Germany, it doesn't for me. The phone provider says that I've sent an invalid command.
13:13.02sehhany help would be appreciated please
13:17.53*** join/#asterisk sekil (~sekil@78.24.104.73)
13:20.11*** join/#asterisk mjordan (~mjordan@nat/digium/x-qhwfreaaufkyntwd)
13:24.58*** join/#asterisk azv4 (~azv4@host-66-202-7-218.pit.choiceone.net)
13:27.41*** part/#asterisk asterisk-Tester (~RAMYT@193.227.170.247)
13:29.16*** join/#asterisk Azrael808 (~peter@212.161.9.162)
13:31.33*** join/#asterisk p3nguin (~xwQ5kwYl6@cobalt.esxi.hosts.a2infotech.com)
13:32.41*** join/#asterisk k3asd` (~k3asd@static-94-32-127-180.clienti.tiscali.it)
13:37.38*** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson)
13:37.38*** mode/#asterisk [+o putnopvut] by ChanServ
13:39.46*** join/#asterisk aberrios (~aberrios@195.171.4.82)
13:47.29*** join/#asterisk CrossWired (~chatzilla@65.210.186.34)
13:54.50*** join/#asterisk jonmasters (~jcm@edison.jonmasters.org)
14:01.19*** join/#asterisk anonymouz666 (~anonymouz@187-28-37-118.poolip.RJO.embratel.net.br)
14:01.25anonymouz666hello
14:01.55anonymouz6661.8.8.0-rc1 is a great version.
14:02.16anonymouz666irroot: ping
14:02.29*** join/#asterisk adyn (~adyn@pdpc/supporter/active/adyn)
14:02.37irrootpong
14:03.07irrootindeed its pretty awesome had no issues reported by customers so far
14:03.57anonymouz666irroot: I have seen one behaviour I think its a limitation about the feature wrapuptime used by app_queue, but when you have multiple agents and multiple queues, the wrapuptime is not "global"
14:04.56anonymouz666the agents keep getting calls from queue A in the next sec, if they just hangup the call from Queue B.
14:05.32irrootah that is something to be worked on indeed
14:05.57puzzledalmost seems most questions are about queues these days
14:06.23anonymouz666irroot: do you think that will be hard to do it, or do you know any patches that already does that ?
14:07.22irrootnot know of any patches i put it on my work flow
14:07.31jacc0I have some problems with timing in a brand new asterisk install (asterisk-1.8.7.0 +  dahdi-2.5.0.1 on Debian)
14:07.41*** join/#asterisk master_of_master (~master_of@p57B53526.dip.t-dialin.net)
14:07.48jacc0Failed to open timing fd
14:07.48jacc0Command 'timing test' failed.
14:08.24jacc0I deselected timerfd in make menuselect
14:09.22jacc0and selected dahdi
14:09.24jacc0:S
14:09.42jacc0I think it is the new dahdi failing
14:10.34p3nguinWorks for me.
14:10.41jacc0I've made the same configuation about 10 times in the past weeks; now I'm using the new dahdi and it fails
14:13.01anonymouz666irroot: I have a setup under heavy load with exactly this problem, I would be glad to test when you finish, just let me know.
14:13.34jacc0i use dahdi-dummy
14:13.34*** join/#asterisk Fritz09 (~Adium@pop1-1489.catv.wtnet.de)
14:14.23p3nguinThere's no dahdi dummy in dahdi 2.5.0.1.
14:14.37irrootmodule show like timing
14:14.52anonymouz666but that would be a fix or a feature?
14:14.56anonymouz666:P
14:15.11irrootmake sure its not loaded maybe from previous build deselecting from menuselect does not stop old modules loading if installed
14:15.30*** join/#asterisk clintc (~clintc@128.227.109.39)
14:15.37irrootdahdi_dummy is no longer standalone module dahdi core has the timing magic
14:15.39jacc0i did : rm /usr/lib/asterisk/modules/res_timing_timerfd.so
14:15.44irrootso all you need is to load dahdi
14:16.03*** part/#asterisk clintc (~clintc@128.227.109.39)
14:16.25irroottesting with faxes dahdi timing is best timerfd is not as good and pthread almost unusable
14:17.07jacc0how do I 'load dahdi'
14:17.17jacc0cli> load dahdi ?
14:17.25p3nguinmodprobe dahdi
14:17.51jacc0okay, so i need to :  modprobe dahdi;modprobe dahdi-dummy
14:18.01jacc0or can I leave out dahdi-dummy?
14:18.12p3nguin(0914.23) <p3nguin> There's no dahdi dummy in dahdi 2.5.0.1.
14:18.40jacc0it doesn't give an error when I do: modprobe dahdi-dummy
14:18.43p3nguinIf dahdi, timerfd, and pthread are all installed... If timerfd is not available (perhaps because it was unloaded), will dahdi be used automatically?
14:19.09jacc0any other, no-exsisting name would fail
14:19.24p3nguinIf you try to modprobe a module that does not exist, there should be some type of report saying it does not exist.  That leads me to believe that you have old dahdi parts lying around.
14:19.53p3nguinmodprobe -l dahdi\*
14:20.25p3nguinIf you see dahdi/dahdi_dummy.ko, you have stale modules that should be uninstalled.
14:21.12puzzledold parts lying around is the reason why it makes sense to use a package management system like deb or rpm instead of installing src
14:21.24p3nguinIndeed.
14:21.30jacc0dahdi/dahdi_dummy.ko is not there
14:21.59kaldemarfrom dahdi-base.c: MODULE_ALIAS("dahdi_dummy");
14:22.21p3nguinMaybe modprobe doesn't report failures from missing modules or something.  I really thought it did.
14:22.30jacc0it does
14:22.49jacc0but no error when doing modprobe dahdi-dummy
14:22.58jacc0I'll try the older version
14:23.28p3nguinYou'd be better off removing any old shit you have, installing the latest version, modprobe dahdi, and move on.
14:24.05jacc0there is no old shit; it's a clean install
14:24.24p3nguinGood luck!
14:24.30p3nguinThere's nothing else I can tell you.
14:24.37jacc0okay, thnx
14:24.43*** join/#asterisk The_Boy_Wonder (~vossel@asterisk/batman-developer/dvossel)
14:25.17*** join/#asterisk devyll (~devyll@thpallady.net.hostway.ro)
14:25.50kaldemarmaybe my alias comment didn't connect. dahdi_dummy is an alias for the core dahdi module. if you modprobe dahdi-dummy or dahdi_dummy, it just loads the core module.
14:25.54devyllquestion: what os is recommended for asterisk 1.8 ? centos ? debian?
14:26.08kaldemardevyll: one that you're comfortable with.
14:26.13atheosdevyll the one you're most familiar with
14:26.21devyllok centos 6? 5.5?
14:26.48devyll5.7?
14:26.58[TK]D-Fender8.6.7.5.3.0.9.?
14:27.22devyll:) got it
14:27.24devyllthanks
14:31.25*** join/#asterisk ferdna (~ferdna@cpe-67-10-220-35.elp.res.rr.com)
14:31.26*** join/#asterisk pigpen (~mark@fw.seamans.cc)
14:39.11*** join/#asterisk cyborg-one (1000@31.31.110.253)
14:44.26*** join/#asterisk Tim_Toady (~fuzzy@77.49.252.192.dsl.dyn.forthnet.gr)
14:52.06*** join/#asterisk miztic (~miztic@75-149-203-105-Illinois.hfc.comcastbusiness.net)
14:52.45*** join/#asterisk dandre (~daniel@ble59-2-81-56-122-47.fbx.proxad.net)
14:57.15*** join/#asterisk jasonwert (~w3rt@adsl-99-27-170-70.dsl.klmzmi.sbcglobal.net)
15:00.04*** join/#asterisk Azrael808 (~peter@212.161.9.162)
15:00.19*** join/#asterisk _Corey_ (~chatzilla@64.121.4.75)
15:01.38*** join/#asterisk navaismo (~navaismo@189.146.55.171)
15:02.20*** join/#asterisk twitchnln (~Adium@adsl-184-37-51-181.asm.bellsouth.net)
15:02.32twitchnlnmorning
15:03.12twitchnlnhas anyone ever setup a grandstream gxw4104 with specific outbound rules as to which channel extension dials out on?
15:03.12navaismomorning
15:03.54*** join/#asterisk jsarrel (~IceChat77@66-191-161-122.dhcp.gnvl.sc.charter.com)
15:06.52navaismotwitchnln use differents contexts
15:12.13*** join/#asterisk KavanS (~KavanS@LINBIT/KavanS)
15:13.11*** join/#asterisk FLeiXiuS (~FLeiXiuS@unaffiliated/fleixius)
15:13.32FLeiXiuSIs it possible for asterisk to use the same RTP ports if the client is connecting from the same IP address and the same SIP user?
15:15.54ChannelZnot that I'm aware of
15:16.36[TK]D-Fenderno
15:18.15FLeiXiuSIm having an issue where two of my clients connect
15:18.19FLeiXiuSand hear the same voice call
15:18.26FLeiXiuSEven though they dialed 2 separate extensions
15:18.47FLeiXiuS2 SIP users, dialing 2 separate conference rooms, listening to the same audio.
15:19.18FLeiXiuSIm lost on debugging ideas, figured they were using duplicate RTP ports
15:19.50ChannelZnot really possible
15:19.59Naikrovekyou have the port range choked down too far, or you have a configuration problem, or you have some networking gear with real issues, or you have some networking gear that is trying to be "smart" and to "help" you.
15:20.14ChannelZAre you SURE they're in different conferences?
15:20.27Naikrovekmaybe someone bridged the conf's together.
15:20.47FLeiXiuSChannelZ, yes, I can see meetme dropping them in 2 diff conferences.
15:20.59FLeiXiuSI minimized the config, very short only a few lines
15:21.17p3nguinWhat would cause some outbound calls through my ITSP to have normal ringing but other calls through the same ITSP, same iax peer entry, and same extension pattern to not have any ringing sound?
15:21.20ChannelZAre both people behind the same firewall?
15:21.21[TK]D-Fendertwitchnln, Go read the manual's section mentioning "Prefix To Specify Port"
15:21.35FLeiXiuSChannelZ, yes, same machine.  It's a web SIP client.
15:22.13ChannelZtwo people are using the same computer?
15:22.42FLeiXiuSChannelZ, They are connecting to the same web server, web server has a java applet which allows them to listen to a conference room.
15:23.21FLeiXiuSThis applet is given instructions for which conf room to drop them in.
15:23.36FLeiXiuSIN asterisk, I can see the conf room being dialed and entered by the client.  But the audio is exactly the same.
15:24.18FLeiXiuSBUT!  When I reboot asterisk, the audio is different as you would expect.
15:24.59*** join/#asterisk hobodave (~hobodave@pdpc/supporter/professional/hobodave)
15:26.02FLeiXiuSIf I dial from a soft sip client, the audio is different.  It seems that connecting with the same server and the same sip user causes problems after like 10-20 calls
15:29.03*** join/#asterisk navaismo (~shaka@fixed-203-96-202.iusacell.net)
15:30.09*** part/#asterisk navaismo (~shaka@fixed-203-96-202.iusacell.net)
15:30.31*** join/#asterisk MarKsaitis (~MarKsaiti@cpc4-rdng22-2-0-cust932.15-3.cable.virginmedia.com)
15:30.38*** join/#asterisk navaismo (~navaismo@fixed-203-96-202.iusacell.net)
15:37.22jsarreldtmf signaling for analog lines is inband...correct?
15:38.11irrootjsarrel if by analog you refering to FXO POTS lines yes
15:38.25jsarrelyes, ty
15:43.52*** join/#asterisk vinhdizzo (~vinh@vqn-portege.ics.uci.edu)
15:43.55*** part/#asterisk mjordan (~mjordan@nat/digium/x-qhwfreaaufkyntwd)
15:44.06*** join/#asterisk mjordan (~mjordan@nat/digium/x-qhwfreaaufkyntwd)
15:47.20*** join/#asterisk Dibbler (~Dibbler@87-194-103-72.bethere.co.uk)
15:51.40*** part/#asterisk russellb (~russellb@asterisk/contributor-and-cool-guy/russellb)
15:57.59*** join/#asterisk Greenlight (~Wullie@cpc2-dund11-2-0-cust994.sgyl.cable.virginmedia.com)
15:59.24GreenlightHowdy folks. Was wondering if someone could help with a question regarding jitterbuffers. The jitterbuffers configured using sip.conf and iax.conf, when should these be used, since most endpoints already have a jitterbuffer?
15:59.53*** join/#asterisk owmyeyes (~hlau@fwny2.delivery.com)
16:01.51GreenlightReason I'm asking is I've a couple of Asterisk boxes connected via IAX trunk and seem to be getting choppy and jittery audio
16:06.52*** join/#asterisk timahvo1 (~rogue@41.215.1.35)
16:08.06*** join/#asterisk cerberus_za (~coert@8ta-151-39-40.telkomadsl.co.za)
16:09.31navaismoGreenlight maybe your bad audio its for bad network quality
16:10.26GreenlightNaa, it's only happening for calls into ConfBridge from the remote asterisk trunk. Direct calls are fine, and other calls into confbridge are fine
16:10.58GreenlightI've tried jitterbuffer on and off for the trunk and on and off for the confbridge
16:11.42GreenlightWas just wondering exactly what I should be setting - In my mind I only need the confbridge jitter buffer and nothing on the IAX trunk, but that doesn't seem to work ;/
16:12.37navaismoenable trunking in the iax, use gsm codec or g729 to verify your network load isnt a issue
16:13.03GreenlightIt's using trunked g729 already
16:14.18GreenlightRunning at about 15% usage of the upstream bandwidth
16:15.13GreenlightOh well - thanks anyways - Guess some more trial and error testing over the weekend :)
16:16.32navaismonp
16:17.19*** join/#asterisk hardwire (~spencersr@cl-36.anc-01.us.sixxs.net)
16:17.20hardwiremeh
16:19.58ChannelZFLeiXiuS: is this Java thing a server/daemon or do multiple instances just run as web clients fire it up?
16:25.48*** join/#asterisk jblack (~jblack@pool-71-173-1-251.sctnpa.east.verizon.net)
16:26.06jblackHi. I seem to remember there's a problem with calling a macro from inside of macros. is that true?
16:27.16*** join/#asterisk Azrael808 (~peter@212.161.9.162)
16:33.08*** join/#asterisk aidinb (~aidin@unaffiliated/aidinb)
16:41.40*** join/#asterisk SuperNull (~FreeManof@24-148-101-238.ip.mhcable.com)
16:43.13*** join/#asterisk dre (~dre@69.38.200.246)
16:58.51*** join/#asterisk mpe (~mpe@31.25.23.177)
16:58.51hardwiresorta wants to make a cdr_cpickle
16:59.42*** join/#asterisk mpe (~mpe@31.25.23.177)
17:00.43FLeiXiuSChannelZ, nope, it runs locally on the users end in a JVM
17:07.34*** join/#asterisk thehar (thehar@diddlebox.thehar.com)
17:11.05*** join/#asterisk eject_ck (~eject@h17.240.159.dialup.iptcom.net)
17:12.41*** join/#asterisk gerhard7 (~gerhard7@82-171-103-215.ip.telfort.nl)
17:13.20*** join/#asterisk eject_ck (~eject@h17.240.159.dialup.iptcom.net)
17:18.35*** join/#asterisk eject_ck (~eject@h17.240.159.dialup.iptcom.net)
17:21.25*** join/#asterisk eject_ck (~eject@h17.240.159.dialup.iptcom.net)
17:26.11*** join/#asterisk _Corey_ (~chatzilla@64.121.4.75)
17:27.05*** join/#asterisk eject_ck (~eject@h17.240.159.dialup.iptcom.net)
17:30.57*** join/#asterisk ChrisInSydney (~Chris@60-242-81-231.tpgi.com.au)
17:32.18*** join/#asterisk eject_ck (~eject@h17.240.159.dialup.iptcom.net)
17:35.24byroncI'd like to create a dialplan extension from the console like this "dialplan add extension _X.,hint,${CUT(REALTIME(presence_hints,key,${CONTEXT}:${EXTEN},:,:),:,4)} into foo" but when I run the command I find an extension like this: '_X.' =>          hint: ${CUT
17:35.33byroncHow would I escape that value correctly?
17:36.10byroncThe real goal is to not have any of that substitution evaluate until a hint lookup on the context is done?
17:36.54[TK]D-FenderBryanstein, You can't just put a function there.
17:38.41[TK]D-FenderBryanstein, hints are fed on dialplan load and don't evaluate
17:38.46*** join/#asterisk gmaruzz (~gmaruzz@2-225-249-20.ip178.fastwebnet.it)
17:39.01*** part/#asterisk gmaruzz (~gmaruzz@2-225-249-20.ip178.fastwebnet.it)
17:39.07*** join/#asterisk eject_ck (~eject@h17.240.159.dialup.iptcom.net)
17:40.52*** join/#asterisk blizzow (~jburns@67.50.165.58)
17:42.25*** join/#asterisk gxdssoft (~gxdssoft@201.230.122.17)
17:43.09byronc[TK]D-Fender: I have those hints working out of realtime: http://pastebin.com/yqxVLPcX
17:43.39byroncWhen a request is made for any hint on that context, it does a realtime lookup to find the right hint value
17:44.06byroncBut I'd like to be able to add the generic hint from the console instead of having to run "dialplan reload"
17:50.29*** join/#asterisk eject_ck (~eject@h17.240.159.dialup.iptcom.net)
17:50.37[TK]D-FenderBryanstein, Hrm
17:51.30*** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger)
17:51.30*** mode/#asterisk [+o pabelanger] by ChanServ
17:53.50*** join/#asterisk eject_ck (~eject@h17.240.159.dialup.iptcom.net)
17:56.10*** join/#asterisk r33dtard (~r33dtard@gateway/tor-sasl/r33dtard)
17:58.21*** join/#asterisk eject_ck (~eject@h17.240.159.dialup.iptcom.net)
17:58.23*** join/#asterisk evil_gordita (robert@ip70-188-50-186.rn.hr.cox.net)
18:03.51*** join/#asterisk eject_ck (~eject@h17.240.159.dialup.iptcom.net)
18:05.48*** join/#asterisk eject_ck (~eject@h17.240.159.dialup.iptcom.net)
18:08.43*** join/#asterisk eject_ck (~eject@h17.240.159.dialup.iptcom.net)
18:13.53*** join/#asterisk eject_ck (~eject@213.159.240.17)
18:19.20*** join/#asterisk eject_ck (~eject@h17.240.159.dialup.iptcom.net)
18:21.11*** join/#asterisk eject_ck (~eject@h17.240.159.dialup.iptcom.net)
18:23.18*** join/#asterisk eject_ck (~eject@h17.240.159.dialup.iptcom.net)
18:26.59*** join/#asterisk _mwoodj_ (~mwoodj@pdpc/sponsor/digium/hyper-eye)
18:28.14*** part/#asterisk twitchnln (~Adium@adsl-184-37-51-181.asm.bellsouth.net)
18:29.46*** join/#asterisk eject_ck (~eject@h17.240.159.dialup.iptcom.net)
18:31.35*** join/#asterisk twitchnln (~Adium@adsl-184-37-51-181.asm.bellsouth.net)
18:33.30*** join/#asterisk eject_ck (~eject@h17.240.159.dialup.iptcom.net)
18:34.07*** part/#asterisk eject_ck (~eject@h17.240.159.dialup.iptcom.net)
18:39.30*** join/#asterisk ferdna (~ferdna@cpe-67-10-220-35.elp.res.rr.com)
18:53.46*** join/#asterisk rdahlin_1 (~rdahlin_1@78-73-19-238-no168.tbcn.telia.com)
19:05.46*** join/#asterisk zerohalo (~zerohalo@173-13-92-27-NewEngland.hfc.comcastbusiness.net)
19:08.59*** join/#asterisk gerhard7 (~gerhard7@82-171-103-215.ip.telfort.nl)
19:09.49*** join/#asterisk coppice (~chatzilla@183178203098.ctinets.com)
19:13.42*** join/#asterisk Kastegir (~Kastegir@c-67-167-43-204.hsd1.il.comcast.net)
19:14.53Kastegirhello, I am trying to get my TDM410P cards to work. I have fully functional FXO ports but no dial tone on the FXS ports. Can anyone help?
19:21.18[TK]D-FenderKastegir, Did you plug the molex in?
19:21.47Kastegiryes, first thing I checked
19:22.22[TK]D-Fenderpastebin your configs.
19:22.25[TK]D-Fender? pb
19:22.32[TK]D-Fender~pb
19:22.32infobot[~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.us , 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.
19:23.36*** join/#asterisk timahvo1 (~rogue@41.223.56.19)
19:23.54KastegirI'm a little new at this, which config dop you want?
19:24.16*** join/#asterisk fskrotzki_ (~fskrotzki@cpe-67-253-245-174.rochester.res.rr.com)
19:27.19[TK]D-Fenderall of the DAHDI one.
19:27.21[TK]D-Fenders
19:30.00*** join/#asterisk coppice (~chatzilla@183178203098.ctinets.com)
19:31.48Kastegirjust paste the link?
19:33.58Kastegirhttp://pastebin.com/hhBRYwNC
19:34.53Kastegirhttp://pastebin.com/kqVmzRGa
19:38.24*** join/#asterisk g_r_eek (~g_r_eek@173-9-142-122-Miami.FL.hfc.comcastbusiness.net)
19:45.06navaismono fxo_ks configuration do you create the dadhi extension too?
19:45.39[TK]D-FenderKastegir, #include chan_dahdi_additional.conf <--- where is this in a PB?
19:45.59Kobazhaha
19:46.16Kobazwhen you change a route to a sip device in cisco call manager, it drops every call to that device
19:46.23[TK]D-FenderKastegir, So far you have not defined an extension for those DAHDI channels
19:50.25*** join/#asterisk kerx (~kerx@76-240-161-60.lightspeed.irvnca.sbcglobal.net)
19:51.00*** join/#asterisk lcat (~lcat@187.45.254.55)
19:55.29*** join/#asterisk Rufus (Rufus@unaffiliated/rufus)
20:00.09Kobaz[TK]D-Fender: isn't that great?
20:00.23[TK]D-Fenderindeed
20:00.27Kastegirsorry
20:00.29Kastegirhttp://pastebin.com/GJ7rsZsu
20:00.31*** join/#asterisk mpe_ (~mpe@31.25.23.177)
20:03.21[TK]D-FenderKastegir, pastebin "dahdi show channels" from * CLI and "dahdi_cfg -vvvv" from OS CLI
20:05.04Kastegirhttp://pastebin.com/fL6E2fT3
20:06.05Kastegirhttp://pastebin.com/nxj86f2D
20:06.53[TK]D-FenderKastegir, "ls -la /etc/asterisk"
20:09.06Kastegirhttp://pastebin.com/qDsGnUu0
20:11.37[TK]D-Fenderroot has perms in there.
20:11.40[TK]D-Fendernot good.
20:12.00KastegirI know, its a test box
20:12.08[TK]D-Fenderchown -R asterisk:asterisk /etc/asterisk
20:13.59Kastegirdone
20:19.43[TK]D-Fenderrestart *
20:23.21Kastegirseriously.... it was a permissions issue?
20:23.25*** join/#asterisk TimeRider (~steve@188-220-33-123.zone11.bethere.co.uk)
20:23.27Kastegirthat worked
20:23.29[TK]D-FenderIs it working?
20:23.34[TK]D-FenderThat would do it
20:23.43*** part/#asterisk wesphillips (~wphill04@137-237-233-124.harris.com)
20:23.48KastegirI will song of your greatness by the fire.... thank you
20:23.53Kastegirsing even
20:24.04[TK]D-Fender* runs as "asterisk".  fail to load the config that specifies that port due to privileges = fail
20:24.15[TK]D-FenderKastegir, You're welcome
20:24.51[TK]D-FenderAnd on that note, it's checkout time.  Later all...
20:24.55Kastegirdidnt even occure to me that the file it made it couldnt read
20:28.20*** join/#asterisk _mwoodj_ (~mwoodj@pdpc/sponsor/digium/hyper-eye)
20:31.03*** part/#asterisk Kastegir (~Kastegir@c-67-167-43-204.hsd1.il.comcast.net)
20:45.41*** join/#asterisk ChrisInSydney (~Chris@120.152.62.230)
20:45.51*** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194)
20:46.44*** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw)
20:52.12*** join/#asterisk hobodave (~hobodave@pdpc/supporter/professional/hobodave)
21:21.05SuperNullHey Guys, anyone ever see asterisk have a megaton socket connects: asterisk  24873        root  504u     sock                0,6       0t0     640986 can't identify protocol
21:21.19SuperNullerrr i will pastebin
21:22.29*** join/#asterisk dijib (~nobodysho@bas10-kitchener06-1176001555.dsl.bell.ca)
21:22.36dijibp3nguin, still getting ddos'd
21:22.37dijib?
21:23.27p3nguinDoS, not DDoS.  And not right now.
21:23.32SuperNullmight try fail2ban .. i have been waiting for the right situation of DOS. Could probably prevent a lot of failed login attempts
21:23.39dijibnot distributed/
21:23.40dijib?
21:23.51p3nguinfail2ban is useless in this case.
21:23.55dijibthats what im saying SuperNull china or jp wouldnt have it lastnight
21:23.59dijibim like pppffft
21:24.12dijibbounce the interface and bam
21:24.17dijibmore bacon
21:24.21dijiband bacon
21:24.25SuperNullswear to god.. our company got raped by russia telekom for like 40k in international ..
21:24.49dijibnuts
21:24.53SuperNullyeah.
21:25.06SuperNullwe upgraded to 10gig and level3 said 'ehh okay it didnt happen'
21:25.10dijibhow do you have a context security zone... im just home user nuub
21:25.22dijibehhh canadiances?
21:25.44p3nguinNot distributed.
21:25.54dijibsee i dont see the corporate itsp pricing schema
21:25.57SuperNullwhat are they doing p3nguin ?
21:26.01p3nguinJust a UDP flood from a single host.
21:26.01dijibone source. u of t
21:26.19dijibi think iptables and fail2ban would have worked
21:26.21dijiblast night
21:26.22SuperNullis it saturating your uplinks ?
21:26.24dijibstill flooding?
21:26.34p3nguinI called the CIO today and left a message.  I then called the helpdesk and the minion was supposed to pass along the info to a manager.
21:26.49p3nguinIt's saturating my downlink.
21:26.54dijibany inpact on the traffic now?
21:27.07p3nguinIt's not happening right now.
21:27.23SuperNullmulti-homed using bgp ?
21:27.28p3nguinAnd no, iptables and fail2ban would not have worked.  Please don't recommend things you don't understand.
21:27.34dijibbgp?
21:27.51dijibim like shoot. lets go plinking
21:27.52SuperNulli do understand now .. obviously firewalling it would not prevent udp flooding. unless it was stopped pre-downstream router
21:28.23dijibwell then im getting an elightenment bya considertion from jp
21:28.30p3nguinMy remark was directed at dijib's suggestion that "iptables and fail2ban would have worked."
21:28.31dijiband no im not qualified
21:28.53SuperNullcurious .. how much traffic ?
21:29.01SuperNullgigs or megs ?
21:29.04p3nguingigs
21:29.11SuperNullyeah thats a whore of a problem to deal with.
21:29.47dijibi still dont understand why blocking that fqdn wouldnt work... but again im a nuuub
21:30.08SuperNullwe had 12 gig coming at us one night.. because some kid pissed off someone on call of duty who had the ability to dos him.. 12 gigabit going to a 5megabit cable modem.
21:30.42dijibability = bandwidth ?
21:30.45p3nguindijib: You can block anything you want, but it still uses up your bandwidth to get it to you... where you block it.
21:31.06dijibok then/// still need to say no... dont rape me
21:31.07p3nguinThe only solution is to have it blocked up stream.
21:31.15dijibtalk to the makers of IOS for that one
21:31.35p3nguinAnd iptables won't work because it's not a Linux box.
21:32.02SuperNullits not all bandwidth... with good transit providers they can mitigate things before they get to you.. but it may not be a quick thing it took level3 30 minutes from our call in
21:32.13p3nguinAnd fail2ban won't work because that's not what it does.  fail2ban detects certain types of things in log files and responds by blocking traffic from the host.
21:32.31p3nguinBut I already identified the host.
21:32.40SuperNullfail2ban utilizes iptables so its kind of a dillema ... i thought you had random peoples trying to hack sip accounts.. we get that all the time.
21:32.51p3nguinI can use fail2ban with pf if I want.
21:32.54*** join/#asterisk vinhdizzo (~vinh@vqn-portege.ics.uci.edu)
21:33.04SuperNulldoes it work with pf ?
21:33.07p3nguinSure.
21:33.10SeRip3nguin, you got good news for me? :)
21:33.11SuperNulli dont use *bsd so
21:33.20p3nguinseri: Maybe.
21:33.27SeRi?
21:34.04p3nguinseri: The SIM arrived, but I'm not sure if it will do any good.  It is my understanding that I'm supposed to use a SIM which was used to activate another iPhone.
21:34.04SuperNullmy least favorite type of dos is email tho .. people get pissed at even a 5 minute delay in email delivery.
21:34.20dijibhow would i build a security perimeter for outound calling in my dialplan using contexts?
21:34.36dijibget them using sip.
21:35.13SeRip3nguin, Try it. report back. I never had to use an original sim to use ny iphone.
21:35.19dijibp3nguin, jailbrak that iphone h'esuze
21:35.29SeRiany*
21:35.34dijibjailbreak
21:35.40SeRip3nguin, you still dealing with the DoS?
21:36.16p3nguinThe DoS I get is just a bandwidth-consuming, CPU-hogging, log-filling type of attack.  But I disabled logging, so that took care of log filling.  I accept the traffic, so that took care of some of the CPU hogging.  I accept the traffic but block replies, so that took care of part of the bandwidth consuming part.
21:36.40p3nguinNot yet today.  He usually starts it at 6 PM Eastern time.
21:36.58*** join/#asterisk corretico (~luis@201.201.44.82)
21:37.02SuperNullwhy not block inbound so no reply is generated period.
21:37.03p3nguinI have an idea.  :)
21:37.27SeRip3nguin, let me also add that if that would be the case with your iphone than you are sol... that would be hard to find unless you buy a sim unlocker from china
21:40.03SuperNullanyone ever see anything like this from lsof output for asterisk: http://pastebin.com/qe04JnZn
21:41.14SuperNullwe use mysql for realtime as well as some mysql lookups in dialplan im wondering if its getting stuck but.. oddly.. its all tcp not unix sockets.
21:41.54*** join/#asterisk vinhdizzo (~vinh@vqn-portege.ics.uci.edu)
21:44.14SuperNullwonders if p3nguin did to much firewalling
21:45.43*** join/#asterisk ChrisInSydney (~Chris@101.170.108.129)
21:46.00SuperNullChris where are ya from ?
21:46.56*** join/#asterisk p3nguin (~xwQ5kwYl6@cobalt.esxi.hosts.a2infotech.com)
21:47.57ChrisInSydneySydney
21:48.16ChrisInSydneyHey P3nguin: The bridge is was still up 10 minsago
21:48.28ChrisInSydneyEastern Beaches
21:48.48ChrisInSydneySomewhere between Bondi and Maroubra
21:48.58SuperNullrainin like a mofo here.
21:48.59ChrisInSydneyThats as close as I'll let ya
21:49.13ChrisInSydneyyep. You in au too ?
21:49.53SuperNullnooo actually upstate New York..
21:49.55SeRip3nguin, you avail?
21:49.58*** join/#asterisk nix8n82-phone (~AndChat@75-174-137-216.chyn.qwest.net)
21:50.06ChrisInSydneyahh
21:50.10ChrisInSydneyrainin here too
21:50.12ChrisInSydneynot too bad
21:50.45ChrisInSydneybut enough to make the trip to swimming for the little one a little scary
21:52.06ChrisInSydneyVUC is still up. 3 people
21:52.19*** join/#asterisk devil_evoxxx (~d3v1l@host79-37-dynamic.4-87-r.retail.telecomitalia.it)
21:52.24devil_evoxxxhi all guy
21:52.29devil_evoxxxi've got a question...
21:52.45ChrisInSydneyOne is me at home, that leaves 2 other die hards
21:53.01ChrisInSydneydevil_evoxxx: Ask away
21:53.02devil_evoxxxi serve to a small provider a sip interconnection for terminating sip traffic
21:53.07devil_evoxxxahahah ok :)
21:53.14ChrisInSydneyand...
21:53.24devil_evoxxxi bill the call after with a cron
21:53.26devil_evoxxxevery 10 minute
21:53.31SuperNulluhg.
21:53.33SuperNullmessy.
21:53.38SuperNullRadius baby.
21:53.38devil_evoxxxbut, is not safe, because if the call remain up..i'm fucked
21:53.46ChrisInSydneyyep
21:53.51ChrisInSydneytake up music instead
21:53.59ChrisInSydneyguitar. I'd recommend as a start
21:54.08SuperNullrockband for complete newbs.
21:54.12SuperNullplus you get drums too
21:54.15ChrisInSydneyand you thunk I am joking :0)
21:54.31devil_evoxxx:(
21:54.35devil_evoxxxin my case
21:54.50SuperNulldevil_evoxxx i can tell you the evils of text file based billing FIRST HAND.
21:55.11devil_evoxxxy
21:55.13devil_evoxxxyes
21:55.16SuperNullmy company loses billing all the damn time because the previous guy implemented the most redic text based CDR system that merges everything.. monthly.. and then runs perl scripts against it
21:55.18SuperNullwell
21:55.28SuperNullthe way your doing it is even more painful
21:55.39devil_evoxxxthe only wai for make it safe is using radius?
21:55.43SuperNullbut possibly less likely to lose stuff than ours.. or lose long calls.
21:55.57ChrisInSydneyI concur with SuperNull
21:55.59SuperNullehhh radius is just real sexy for that kind of thing. its been around for ever.. and you can easily run searchs against it quickly
21:56.34ChrisInSydneyI've not had much to do with billing, but the standard CDR stuff is not really your friend
21:56.44devil_evoxxxand i can send "disconnect" when is time..
21:56.48SuperNullcall starts - radius sends a call start. call stops radius sends a call stop .. and calculatues things for you.. then you just have to make reports instead of working on some kind of ghetto rigged text file situation
21:57.03ChrisInSydneyFine for stats, and aproximates, but billing is a nightmware
21:57.13SuperNulltext based cdr is the devil for end points.
21:57.21devil_evoxxxthere is some example for handlig this?
21:57.26SuperNullehhh...
21:57.27ChrisInSydneyHey, some of the best music came out of the Ghetto
21:57.43*** join/#asterisk vinhdizzo (~vinh@vqn-portege.ics.uci.edu)
21:58.28SuperNulldevil_evoxxx all i can say is.. you will wish you could find me on irc again later to thank me if you are serious about not losing billed seconds.
21:58.30devil_evoxxxehm but i've got another question
21:58.32*** join/#asterisk bn-7bc (bjarne@pdpc/supporter/active/bn-7bc)
21:58.49ChrisInSydneythats OK. Ask away
21:59.02devil_evoxxxSuperNull: is not the time for losing billed second..the problem that i've just observerd is
21:59.29devil_evoxxxon TRUNK_1 i receive the call from the other is
21:59.31devil_evoxxxisp
21:59.48devil_evoxxxthat, i receive a call...and it'start
22:00.07devil_evoxxx(let 's suppose that
22:00.32devil_evoxxxthe call duration is 240 second
22:00.33SuperNullsupposes
22:00.42devil_evoxxxafter 1 second when call 1 start, another call start
22:01.05ChrisInSydney...and...
22:01.19SuperNullhey man .. bable fish cant translate that quick ;-)
22:02.38p3nguinseri: iTunes says I need to try again later or contact customer care.
22:02.58p3nguinIn other words, it doesn't work.
22:02.59*** join/#asterisk bn-7bc (bjarne@pdpc/supporter/active/bn-7bc)
22:04.15ChrisInSydneynope, they dont
22:04.17ChrisInSydneyBRB
22:04.28devil_evoxxxi'm back
22:04.55*** part/#asterisk mjordan (~mjordan@nat/digium/x-qhwfreaaufkyntwd)
22:05.19p3nguinseri: http://support.apple.com/kb/ht3406
22:06.21devil_evoxxxso, if i use a "prepayed" structure, how can i calculate the amount of second
22:06.29devil_evoxxxof two concurrent call?
22:06.36devil_evoxxxwith one call , ok, no problem
22:06.44devil_evoxxxbut with >2?
22:10.35*** join/#asterisk ChrisInSydney (~Chris@60-242-81-231.tpgi.com.au)
22:10.56*** join/#asterisk vinhdizzo (~vinh@vqn-portege.ics.uci.edu)
22:12.55SuperNullerrr i cant speak about prepaid it is not a feature i know of for asterisk .. it may or may not exist as far as i know
22:13.40devil_evoxxxand, a question
22:13.56devil_evoxxxif your small client
22:14.11devil_evoxxxhas been hacked , and make a lot of calls
22:14.15devil_evoxxxhow can you prevent this?
22:15.29*** join/#asterisk Fritz09 (~Adium@pop1-1489.catv.wtnet.de)
22:16.37SuperNullset a call max ? or calls per second limit some how.. when we got hacked it was usually long duration international coming from russia or china
22:16.49*** join/#asterisk lanning (~lanning@173-8-187-197-SFBA.hfc.comcastbusiness.net)
22:16.55SuperNull20 hour call .. like .. seriously.. who does that.
22:17.02devil_evoxxxin my case the ip was from palestina
22:17.07devil_evoxxxand calla special service in Estonia
22:17.15SeRip3nguin, I think you have no choice but  to jailbrake your phone and unlock it.
22:17.15devil_evoxxx1000 Euro burned..
22:17.31ChrisInSydneyOuch :-(
22:17.39devil_evoxxxbecause
22:17.42devil_evoxxxthis fucking client
22:17.47SeRiI never came across any issues with unlocked iphones.
22:17.47devil_evoxxxdon't take care of his hata
22:17.49devil_evoxxxata
22:17.55SeRinone unlocked*
22:18.10devil_evoxxxand..all know that with an exploit you can extract password
22:18.13devil_evoxxxand user..
22:18.50devil_evoxxxit's saying to me that all ata was on default port..with default password
22:18.52devil_evoxxxFUCK..
22:18.56devil_evoxxxi have no word.
22:19.10SuperNullbill em. nuff said. not your problem.
22:19.57SuperNullnot like your provider will give it to ya free.
22:20.10devil_evoxxxeh, it's right
22:20.13SuperNullunless you upgrade from 1gigabit ethernet to 10gigabit .. then maybe..
22:20.38devil_evoxxx..oO we'have recently upgraded to 1 gbit STM-16
22:20.53*** join/#asterisk vinhdizzo (~vinh@vqn-portege.ics.uci.edu)
22:21.03SuperNullwhere is this and .. they dont have native ethernet options yet/
22:21.10p3nguinseri: It can't be jailbroken, it can't be activated without the SIM it's looking for.
22:21.25devil_evoxxxSuperNull: ethernet over sdh too..
22:21.25p3nguinseri: Once I can activate it, I can probably jailbreak it.
22:22.39SuperNulldevil_evoxxx i think your best bet is finding a solution or really planning it out.. it sounds like a lot of work possibly with AGI .. and im guessing because i dont know AGI
22:22.41SuperNullhaha
22:23.22devil_evoxxxi know agi , i've already use
22:23.30SeRip3nguin, http://www.iphonehacks.com/2010/07/how-to-activate-your-iphone-without-official-iphone-carrier-sim.html
22:23.39SeRiRead that ^^
22:23.48SuperNullso if its prepaid why even allow multi-calling at the same time ?
22:24.18SuperNullwhy not just say 'sorry a call with that pin is already active.. go eat a cake'
22:24.38SuperNullusing radius this would be fairly easy..
22:25.12SuperNullthen you just have to worry about max duration of a single call and can probably somehow make that drop after a certain max time.
22:25.20devil_evoxxxSuperNull: because is not only a phone
22:25.22devil_evoxxxis a trunk
22:25.25devil_evoxxxtype=peer
22:25.32SuperNullprepaid trunks ?
22:25.44SuperNullehhh we know of no such thing in the USA.
22:25.51devil_evoxxxy..
22:26.10SuperNullin Soviet Russia VOIP trunk pay for you.
22:26.13SuperNullehh
22:26.17devil_evoxxxif you give a service to a small providere for sip-trunking
22:26.21SuperNullwe bill on usage only and cross our fingers they pay.
22:26.52devil_evoxxxhow you do=
22:26.52devil_evoxxxhow you do?
22:27.10SuperNullwe dont do that type of call service.. we are primarly a VOIP provider for residential docsis modem customers.
22:27.21SuperNullabout 8000-9000 to be exact
22:28.22dijibmy brother bought an iphone 4S today. ive suggested he runs an asterisk implementation on it.
22:28.30SuperNullhahaha
22:28.37SuperNulldo they have such a thing dij?
22:28.47dijibyou could build it yes.
22:28.55dijibpositive
22:28.57SuperNullif you build it.. they will call
22:29.02dijibits got a 1ghz dual core cpu
22:29.15*** join/#asterisk vinhdizzo (~vinh@vqn-portege.ics.uci.edu)
22:29.26dijibthats what im saying... callcenter in a phone. call all your service departments off a queue
22:29.36SuperNullso .. thats an interesting concept..
22:29.45SuperNullphone switch on the phone.
22:29.51dijibword up. mobile terminal scripsts for adding peers
22:30.12SuperNullany ideas if the wifi chip in that thing does WDS ?
22:30.13dijibhe told me he mainly uses it for email sms and web. i said... get an account pat.
22:30.24dijiblol
22:30.29SuperNullhow cool would that be..
22:30.39SuperNullmesh of iphone .. phone switches..
22:30.43dijibthat just is passive mode on the wifi no?
22:30.59SuperNullWDS is like.. an ap that links to another ap .. and .. can still accept users..
22:31.03dijibwds is linksys or cisco tech i think
22:31.30SuperNullehh its an open standard i believe .. if its not.. its documented enough to be supported by many
22:31.39dijibno he would be using the telco's bandwidth... but im sure if there is a less costly path the routing protocal would handle the switchj
22:31.52SuperNulli used to use it with HostAP years ago.. when commercial 802.11 APs were like $1000
22:32.04dijiboh
22:32.10SuperNullyou could dij..
22:32.17dijibdiji
22:32.18SuperNullanyone in range could hop you to the next person in a city.. it would be intense..
22:32.24SuperNulldiji ...
22:32.24dijibas in digital equipment corporation
22:33.01SuperNullthe delimma would be routing tho.. i have used AODV routing before in a beta format.. and its slow as poo.
22:33.20dijibso what do you think, theory on the iphone asterisk server good? also give it dydns
22:33.53SuperNulliphone + thin asterisk + wds + dynamic on demand routing
22:33.59p3nguinwaves at chrisinsydney
22:34.16SuperNullhonestly.. i thought of this before.. for android.. but i dont want to brick my Evo 4G  :-/
22:34.43dijibat least the iphone is restorable easily
22:34.58dijibi would love to alpha on his old iphone 4
22:35.01SuperNulli would be curious to see what kind of call quality you could get on 3G .. lol
22:35.36dijibseeing that your client would be connecting to 127.0.0.1 i would hope good in the 3g band
22:35.50dijibthats what telco uses anyways.. obviously
22:36.14devil_evoxxxguys, i left...good night
22:36.19SuperNullyeah but they can setup QOS service flows of some type im sure, so they have covered their butt
22:36.42SuperNullim leaving for the weekend as well Gents.. i will likely be back when this asterisk box shits from to many open sockets.
22:36.59dijibno way with the service being split with wireless internet packages
22:37.20dijibhave a good one SuperNull
22:44.53ChrisInSydneyThe VUC bridge is still up for you die hards
22:46.47ChrisInSydneySIP:200901@login.zipdx.com (g722, g711)
22:57.14p3nguinI got pulled into making supper, so I'll be away from the conf for a little while.
22:58.56*** join/#asterisk shido6 (~shido6@nat/yahoo/x-nbfnybjamzytdtpx)
22:59.18*** join/#asterisk dijib (~nobodysho@bas10-kitchener06-1176001555.dsl.bell.ca)
23:01.40p3nguinseri: Did you read that page or watch the video?
23:02.08ChrisInSydneyP3nguin. Mans gotta eat
23:02.48p3nguinI intend to do just that.
23:02.59ChrisInSydneyI'm still hacking the data. Cant get crimson editor to do reg expr :-(
23:03.06ChrisInSydneyoh well, it is free
23:03.52ChrisInSydneySo whats for dinner ??
23:03.54dijibso asterisk security perimeteres?
23:04.14dijibchicken leggs and baked potatoes... shaddup beach
23:04.23ChrisInSydneydijib perimeters or parameters ?
23:04.30p3nguinLooks like spaghetti, meat balls, and garlic bread.
23:04.44ChrisInSydneyand a glass of something red
23:05.16ChrisInSydneydijib: SIP:200901@login.zipdx.com (g722, g711)
23:05.23ChrisInSydneyVUC bridge is still up.
23:05.32ChrisInSydneydrop in and have a chat
23:06.09p3nguin?
23:06.15p3nguinWhat do you mean what happened?
23:06.40ChrisInSydneyfelix is out of my sight at the moment. I heard a crash and a whine
23:06.53ChrisInSydneythere is still noise so its not too bad
23:07.06ChrisInSydneyno need to call DOCS (CPS)
23:07.29ChrisInSydneyHe's found his "In The Night Garden" book. hes happy now
23:08.34dijibi just want nobody but someone that authenticated be able to access voipms-outbound context
23:08.42dijiband ChrisInSydney whats this ? mailto:SIP:200901@login.zipdx.com
23:09.02dijibhow did mailto: get there
23:09.23ChrisInSydneydont know
23:09.28ChrisInSydneydial in on SIP
23:09.42dijiba zip?
23:09.47dijibwtf is this p3nguin ?
23:09.47ChrisInSydneySIP:200901@login.zipdx.com
23:10.05dijibSIP/200901@login.zipdx.com
23:10.28dijib?
23:10.30SeRip3nguin, watch the video
23:10.33ChrisInSydneyDial(SIP/200901@login.zipdx.com)
23:11.22p3nguinseri: Did you?
23:12.17SeRio no did not watch the video
23:12.25SeRiwhy?
23:12.34SeRiwhats going on?
23:12.41p3nguinThere's a big red note at the bottom that says the method is worthless as of a year ago and cannot be used now.
23:13.00SeRio damn
23:13.09SeRiThat sucks.
23:14.12SeRip3nguin, http://www.google.com/search?aq=0&oq=jailbraking+iphone+without&sourceid=chrome&ie=UTF-8&q=jailbreaking+iphone+without+sim+card
23:14.32p3nguinchrisinsydney: Who was that person who came on?
23:15.44ChrisInSydneyI think that was Dave Frankl
23:15.50ChrisInSydneyruns ZIPDX
23:15.53p3nguinOh.
23:18.50dijibChrisInSydney, dialing that doesnt work
23:18.59ChrisInSydneywhat happens
23:19.06dijibexten => 6969,1,NoOp();
23:19.07dijibsame => n,Dial(SIP/200901@login.zipdx.com);
23:19.14dijib[Oct 14 19:07:40]   == Using SIP RTP CoS mark 5
23:19.22dijibbusy beep
23:19.22ChrisInSydneycheck the codecs
23:19.26*** join/#asterisk jblack (~jblack@pool-71-173-1-251.sctnpa.east.verizon.net)
23:19.28dijibim using ulaw
23:19.31dijib711
23:19.33ChrisInSydneyshould be OK
23:19.41ChrisInSydneywhats in the cli
23:20.02dijibthat last is in the cli
23:20.07ChrisInSydneyI'll check my configs
23:20.36dijibp3nguin, how am i screwing up with that... and mind im drinking.
23:21.10ChrisInSydneyexten => 882,1,Dial(SIP/200901@login.zipdx.com)
23:21.33p3nguin:) I use 882 as well.
23:22.00p3nguinMaybe it won't let in new calls?
23:22.45ChrisInSydneyI jts checked. It does
23:22.50ChrisInSydneyhjust
23:22.54ChrisInSydneyjust
23:22.54p3nguinNope, it let me call in again.  It even gave me a warning that said my ID was in use.
23:23.08ChrisInSydneyI have a differnt ID
23:23.14p3nguinI only have one.
23:23.17ChrisInSydneyactually, I'm not registered
23:23.23p3nguinI dial in with my PIN.
23:24.24ChrisInSydneyoops. My son has just plugged in the barcode reader
23:24.43ChrisInSydney606449067835
23:24.48*** join/#asterisk dandate2 (~dan@124.6.157.210)
23:24.51ChrisInSydneyhe he he
23:25.06ChrisInSydney606449067835
23:25.14p3nguinhaha
23:25.28ChrisInSydneyCHAE44025B
23:25.37p3nguinThe first time I plugged in mine, I had IRC in focus and it did the same thing.
23:25.37ChrisInSydneyI'm going to get him a storemans job
23:25.57p3nguinI think I printed my wife's driver license number in IRC.
23:26.02ChrisInSydneyoops
23:26.07p3nguin:/
23:26.13p3nguinI didn't know it was going to do that.
23:26.24ChrisInSydneythewse are just prouct serials
23:26.25p3nguinI plugged it in and started looking for things to scan.
23:26.28dandate2so i got a problem, i got this DID that my inbound routes won't recognize, in core show channels it just says SIP/64.136.174.30-00
23:26.41p3nguinThat's normal.
23:26.47p3nguinSo what's the actual problem?
23:26.49dandate2how can i get inbound routes to recognize  it?
23:27.10dandate2it just sends it to the default any DID route
23:27.28p3nguinAre you sending SIP registrations to that ITSP?
23:27.38p3nguinAnd we don't know what "inbound route" means.
23:27.44ChrisInSydneydandate2: exten => 882,1,Dial(SIP/200901@login.zipdx.com)
23:27.50dandate2oh its a freepbx term
23:27.54dandate2i guess i should go there!
23:27.55p3nguin~freepbx
23:27.55infobot[~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
23:28.11p3nguinI can help you fix the peer in asterisk.
23:28.11ChrisInSydneyThe VUC bridge is still up
23:28.14ChrisInSydneyjoin us
23:28.46p3nguinI figure you just need to send the correct extension in your register statement and create said extension in the relevant context to accept calls.
23:29.19p3nguinUsually the extension is the same as the DID number, i.e. your phone number.
23:30.22dandate2right, i experienced this problem with another ITSP. if the DID was forwarded as #@ip it worked, but if it was sip:#@ip it wouldn't
23:30.26dijibi think i need to register asd a peer in sip.conf no?
23:30.46ChrisInSydneycorrect. Or have a friend and user
23:31.43p3nguinYou don't need a peer entry in sip.conf to call a SIP URI.
23:32.26p3nguinIf you wanted to do Dial(SIP/zipdx/200901), then you'd need a peer entry for zipdx.
23:32.30ChrisInSydneyYou should be able to call a SIP URI that support anonamous connects
23:32.46ChrisInSydneyso long as it is as a SIP URI
23:32.53p3nguinTo dial the URI, just dial it like chrisinsydney said.
23:32.59ChrisInSydneySIP/name@host.whatever
23:33.06ChrisInSydneyexten => 882,1,Dial(SIP/200901@login.zipdx.com)
23:33.29ChrisInSydneyif you have freepbx, you can stick it into a custom extension
23:33.43ChrisInSydneyor hack the dial plan. Just dont apply any more extensiosn
23:35.33ChrisInSydneydijib: Whats happening ? Still no luck ??
23:35.36p3nguinIf it still doesn't work, core set verbose 3, make a call to it, copy/paste the stuff in the pastebin.
23:35.53p3nguinIf that doesn't show where the problem is, then we'll go on to the sip debug.
23:36.58p3nguinI hear someone talking from the other room, but I don't know what it is.  I hope you weren't talking to me; I'm near the kitchen.
23:37.06ChrisInSydneynope
23:37.17ChrisInSydney[probably just to myself
23:37.39p3nguinor the child
23:37.43ChrisInSydneymaybe it was the voices in my head. If they tell me to act normal, things seem to go OK
23:39.49*** join/#asterisk drmessano (~nonya@pdpc/supporter/active/drmessano)
23:40.34*** join/#asterisk kdmessano (~nonya@unaffiliated/kdmessano)
23:40.39ChrisInSydneydrmessano: Join us.
23:40.54ChrisInSydneyexten => 882,1,Dial(SIP/200901@login.zipdx.com)
23:41.00ChrisInSydneyVUC bridge is still up
23:41.11*** join/#asterisk Naikrovek (~mbluth@unaffiliated/naikrovek)
23:41.13p3nguinI'm still wondering about the "recording" thing that was mentioned.
23:41.23p3nguinI don't understand why it would be recording.
23:41.37dijibwhat recording thing?
23:41.58dijibstill no luck ChrisInSydney
23:42.03ChrisInSydneypaste the CLI
23:42.15p3nguinGood luck getting that from him.
23:42.22*** join/#asterisk shido6 (~shido6@nat/yahoo/x-ohgcodxihklyoaui)
23:42.27ChrisInSydneydijib: the VUC is recorded
23:42.29dijibsuch a pesemist
23:42.31p3nguinI've decided to ask once and if I don't get it, fuck it.
23:42.35ChrisInSydneyThey pick the best bits out for the podcast
23:42.38dijib[Oct 14 19:12:54]     -- Registered SIP '300' at 74.198.9.174:49941
23:42.38dijib[Oct 14 19:16:54]   == Using SIP RTP CoS mark 5
23:42.38dijib[Oct 14 19:31:29]   == Using SIP RTP CoS mark 5
23:42.39dijibflood
23:42.40p3nguinI don't care enough to keep asking for it.
23:43.03ChrisInSydneystrange
23:43.19ChrisInSydneywhat is the order of codecs in your SIP.conf
23:43.46p3nguinI'd rather see something relevant -- what I asked for.
23:44.49dijibwhat did you ask for ?
23:45.28p3nguin(1835.29) <p3nguin> If it still doesn't work, core set verbose 3, make a call to it, copy/paste the stuff in the pastebin.
23:45.54*** join/#asterisk PhoenixMage (~Phoenix@CPE-120-146-192-94.static.vic.bigpond.net.au)
23:46.55p3nguinYou know, standard procedure to get help.
23:47.05PhoenixMageHi guys, I am having probs with a Cisco 7975 with SIP firmware. It can receive calls fine but when I try to make a call it attempts to dial but fails on the first digit as if that is the whole number
23:47.18PhoenixMageSame deal on dialing back a missed call
23:47.39PhoenixMageAny ideas what I am forgetting
23:48.15p3nguinDo not lift the handset.  Dial the number, then press the Dial key.
23:48.18p3nguinDoes that work?
23:48.33PhoenixMagenope same issue
23:48.51ChrisInSydneyjimmy*CLI> core set verbose 10
23:48.51ChrisInSydneyVerbosity was 28 and is now 10
23:48.51ChrisInSydney<PROTECTED>
23:48.51ChrisInSydney<PROTECTED>
23:48.51ChrisInSydney[Oct 15 10:47:52] NOTICE[1887]: chan_sip.c:12724 handle_response_peerpoke: Peer 'DBC-extn201' is now Reachable. (76ms / 2000ms)
23:48.52ChrisInSydney<PROTECTED>
23:48.53p3nguincore set verbose 3
23:48.54ChrisInSydney<PROTECTED>
23:48.58ChrisInSydney<PROTECTED>
23:48.58p3nguinmake a call.
23:48.59ChrisInSydney<PROTECTED>
23:49.01ChrisInSydney<PROTECTED>
23:49.03ChrisInSydney[Oct 15 10:48:00] NOTICE[1887]: chan_sip.c:15820 handle_request: Unknown SIP command 'UPDATE' from '76.74.151.123'
23:49.05p3nguinWhy the flood?
23:49.08p3nguin~pb
23:49.08infobot[~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.us , 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.
23:49.14p3nguinpastebin   ^^^
23:49.58ChrisInSydneyI recon its codecs
23:50.28p3nguinUntil he starts the troubleshooting procedure that we all use here, I don't care to guess what the problem is.
23:50.31ChrisInSydney<PhoenixMage> Jump onto the VUC bridge. Its still up
23:51.00ChrisInSydney<PhoenixMage>: exten => 882,1,Dial(SIP/200901@login.zipdx.com)
23:51.22PhoenixMage[Oct 15 10:50:35] NOTICE[17332]: chan_sip.c:21358 handle_request_invite: Call from 'button1contact' to extension '3' rejected because extension not found in context 'LocalSets'.
23:51.33PhoenixMagethe number i am dialing is 3610
23:51.56ChrisInSydneyPhoenixMage: dialplan.xml ??
23:52.20ChrisInSydneybrb
23:52.26PhoenixMageChrisInSydney: using the same context and dialplan for my iphone and ipad without issue
23:52.28p3nguinIf you're not getting a dial tone first, but dialing the number and then pressing the Dial key, the dialplan file should not be consulted.
23:52.44p3nguinYour iPhone and iPad do not use the Cisco phone dialplan xml file.
23:53.07PhoenixMageI dont have a dialplan.xml
23:53.13p3nguinMaybe you should get one.
23:53.20PhoenixMagedidnt know I needed one
23:53.30PhoenixMageeverything talks about the SEP.xml files
23:53.35PhoenixMageany doco?
23:53.39p3nguinSEP is something else.
23:54.24ChrisInSydneyI'll find some. I'll be back.
23:54.36ChrisInSydneyJoin us on the bridge if you can
23:54.39p3nguinThe dialplan file tells the phone when to "send" the call, based on the number entered.
23:54.49ChrisInSydneyI'm the last ine left and its getting lonely :-/
23:55.01p3nguinIf you don't have one, it should just take a really long time to send the call.
23:55.11ChrisInSydneyneed to attend to family
23:55.13ChrisInSydney5 mins
23:55.56*** join/#asterisk chalcedony (~llhull@unaffiliated/chalcedony)
23:56.04PhoenixMageI'll bbs I have to get my haircut
23:56.11PhoenixMagethanks for the help though
23:56.27ChrisInSydney<PhoenixMage>: There is some stuff on 7970s in the  voip-info website. Explains dialplan.xml
23:56.47p3nguinNot having one should not cause the described problem, though.
23:56.56ChrisInSydneyhair cut ??
23:57.02ChrisInSydneyh ehe he
23:57.03p3nguinBut it will be nice to have it later, when things are working correctly.
23:57.08PhoenixMageyep I am overdu
23:57.09PhoenixMagee
23:57.14ChrisInSydneyhelps you see what you're dialling
23:57.48ChrisInSydneynah. Let it grow. Goi hang oiut on wall street

Generated by irclog2html.pl Modified by Tim Riker to work with infobot.