IRC log for #asterisk on 20111211

00:00.57p3nguinI prefer to use wiki.asterisk.org for things like this, first.
00:05.30*** join/#asterisk joako_ (~joako@opensuse/member/joak0)
00:05.42Sean-Derhttp://pastebin.com/KZRn4nCe
00:05.47Sean-Derhttp://pastebin.com/6KR4EzkB
00:06.04Sean-Derp3nguin: Does CEL default or do I need to uncomment the fields?
00:07.44p3nguinI'm not sure.  Did you consult the page on CEL at wiki.asterisk.org?
00:08.08Sean-Derhttps://wiki.asterisk.org/wiki/display/AST/CEL+Configuration+Files
00:08.14Sean-DerIts blank?
00:08.22*** join/#asterisk wonderworld (wonderworl@gateway/shell/anapnea.net/x-fzeyzrtmzjxyjrqg)
00:09.56p3nguinYikes!  That's not good.
00:10.16Sean-DerCould it be vandalism? Is there a revert?
00:12.01p3nguinI doubt it is vandalism.
00:12.40Sean-Der[2011-12-10 19:00:44] VERBOSE[3404] cel.c:     -- CEL logging enabled.
00:12.53Sean-DerIn my full log it looks like cel is now enabled
00:13.19Sean-Der[2011-12-10 19:00:44] WARNING[3404] cel_odbc.c: No such connection 'mysql1' in the 'first' section of cel_odbc.conf.  Check res_odbc.conf
00:13.23Sean-DerAny ideas?
00:21.29Sean-Derhmm in odbc what is the 'dsn'?
00:22.08helen_:)
00:29.17*** join/#asterisk TimeRider (~steve@92.40.247.208.threembb.co.uk)
00:30.51Sean-DerIt references your connection in odbc.ini
00:37.55dijibsup everyone
00:37.59Sean-DerOkay I have CEL working. Does anyone know of a decent PHP frontend. Or am I going to have to write my own?
00:39.33*** join/#asterisk rotten777 (~quassel@fl-67-233-25-130.dhcp.embarqhsd.net)
00:41.11rotten777to anyone awake i have a stupid question... i'm trying to make the default incoming calls for my pbx go to a list of playback commands then to a ring group... I'm using trixbox and have it playing moh by default but i would like to insert custom playback commands before the moh... i can't find where trixbox is putting the commands for the incoming calls
00:42.04WIMPy~trixbox
00:42.05infobotTrixbox is unable to be supported here.  It is a closed-source distribtuion of Asterisk and FreePBX containing proprietary code that its users do not have access to, making it difficult to support.  Try joining #trixbox and asking your questions there.
00:42.57rotten777ahh no wonder.. i should have never moved from my old standard asterisk implementation
00:42.59rotten777ugh
00:43.01rotten777thanks wimpy
00:56.13*** join/#asterisk autofsckk (~autofsckk@unaffiliated/autofsckk)
01:08.49*** join/#asterisk KojakWolf (~kojak@kojakwolf.com)
01:53.38dijibanyone wanna hold my p3n15 while i do a pgsql integration.
02:01.35*** join/#asterisk s[X] (~mark@ppp118-208-26-44.lns20.bne1.internode.on.net)
02:05.19carrarOnce you get pgsql installed, maybe then you'll have something for someone to hang onto
02:10.29*** join/#asterisk zerohalo (~zerohalo@74.60.136.128)
02:17.57*** join/#asterisk joako_ (~joako@opensuse/member/joak0)
02:23.03*** join/#asterisk master_of_master (~master_of@p57B52C59.dip.t-dialin.net)
02:58.48*** join/#asterisk LiuYan1 (~LiuYan@222.125.130.16)
03:17.40*** join/#asterisk wonderworld (wonderworl@gateway/shell/anapnea.net/x-paawhanfejrxdazx)
03:41.49SeRicarrar: LOL
03:42.17SeRiwell the bday was fun.... kids had lots of fun
03:45.58*** join/#asterisk radic (~radic@dslb-094-216-229-166.pools.arcor-ip.net)
04:01.51p3nguinseri: I have a conundrum.
04:04.22SeRip3nguin: whats that? :)
04:04.40p3nguinI'm in a bit of a pickle, Dick.
04:05.02SeRilol ok.... whats going on...?
04:05.22p3nguinI'm drawing a blank trying to route some traffic to another router...
04:06.13p3nguinOne router in two subnets (A and B), serving hosts in subnet A...
04:06.29SeRiMhhhh ok..... are you using RIP?
04:06.31p3nguinHosts in subnet A need to have a default gateway of router B.
04:07.00SeRiok I see.
04:07.25p3nguinBut if I explicitly give them the address of router B, they can't reach it because they do not have an address in subnet B.
04:07.40p3nguinSo I give them a default gw of router A.
04:07.52SeRithis is all inside vyatta right?
04:07.55p3nguinBut I don't want traffic to go out router A's internet connection.
04:08.05*** join/#asterisk Naikrovek (~mbluth@unaffiliated/naikrovek)
04:08.24p3nguinYes, but the concept is going to be plain linux routing and iptables.
04:08.42p3nguinShould I NAT it?  Should I try to route it?
04:08.46SeRicompletly doable. I am doing something similar with pf
04:09.00SeRione sec let me pull it up
04:10.37p3nguinI don't see how I can route it because I need to route everything, i.e. 0.0.0.0/0.0.0.0, from hosts on subnet A (192.168.10.0/255.255.255.0) to router B.
04:11.01p3nguinI thought maybe snat was the key, but I just don't know.
04:11.41WIMPyrules are your friend.
04:11.46SeRithe key is to route and to restrict access to servidces or traffic via iptables
04:12.14p3nguinI don't care about restricting anything.
04:12.18SeRiWIMPy: exactly
04:12.27p3nguinI just need to route all traffic to a router that is not on my subnet.
04:12.29WIMPy'ip rule help'
04:12.41SeRiyou can You can route and use ip rules to do what you want
04:12.54SeRior even chains
04:13.13p3nguinShould I do it with nat or firewall?  I'll use the vyatta interface to do it.
04:13.34WIMPyNeither. Do it with routing.
04:13.42p3nguinnat is going to use the nat table of iptables.  firewall is going to use the filter table of iptables.
04:13.46WIMPyPolicy routing.
04:13.51p3nguinOkay.
04:13.58p3nguinLet me look at my policies.
04:15.35p3nguinPerhaps I was over-thinking this.
04:16.00SeRiI would do routing as well. I dont know much about the policy's but I am sure they would do the job as well
04:16.51p3nguinIt has to be based on the source address, though.  That's where I ran into a problem in my mind.
04:17.18WIMPyLook for the LARTC. It's like ~book.
04:17.44WIMPyAdd a from rule and set up a 2nd routing table.
04:18.10WIMPyI'm off for a little hardware change.
04:20.44SeRip3nguin: I never done it the way WIMPy explained. mine is base on routing and rules
04:21.13p3nguinI can't do it that way, anyway.  I have to do it with something that vyatta interface can handle.
04:21.22p3nguinIt's like using freepbx and then trying to edit your own extensions.conf.
04:21.32SeRiahhh I see. :)
04:21.56p3nguinThe vyatta interface will do routing and firewall and nat...
04:22.16SeRigot cha... I need to find the time to load vyatta.
04:23.48p3nguinIt looks like I might have the tools with "route filtering policy."
04:24.13p3nguinset policy access-list 100 rule 10 source ...
04:24.17p3nguinMaybe.
04:24.25SeRiah that looks like the spot
04:25.17p3nguinWait.  This is for RIP.
04:25.30p3nguinI don't care about RIP.  I just need to set some static routing policies.
04:25.46p3nguinIt's one router.  I do not need RIP.
04:25.53SeRiindeed
04:25.59p3nguinI really think NAT is where I need to look.
04:26.05p3nguinsource nat
04:26.29p3nguinAnything from the 192.168.10.0/24 network, send it to the other router.
04:27.05p3nguinI just don't know.  Do you see why I said it was a conundrum?
04:27.14p3nguins/conundrum/pickle/
04:29.17SeRiyes lol
04:30.56SeRijust use nat and set your rules.
04:31.26dijibhows the whiskey tonight ?
04:31.30dijibSeRi:
04:32.18SeRino whiskey :)
04:32.27SeRikids bday party today :)
04:32.39dijibi guess you didnt see my last post
04:33.01dijibi thought that already happend?
04:33.07dijibwas reading sometime... i dont know
04:33.19dijibanyways none of my business
04:33.28dijibcontinue, sup asterisk?
04:33.44SeRi?????
04:33.51dijibi gottah git some pgsql going
04:34.06dijiband am without a effin clue, ossmosis aint working either
04:34.11dijib~osmosis
04:34.12infobot[~osmosis] Osmosis is the act of beating yourself on the head repeatedly with THE BOOK, until some measure of absorption has occured ... or at least until your unconsciousness restores peace to the channel ...
04:35.18*** join/#asterisk tyman (~tyman@173-12-219-189-Fresno.hfc.comcastbusiness.net)
04:35.34SeRidijib: for what?
04:36.06SeRip3nguin: making progress>?
04:36.22p3nguinI'm looking at the SNAT configuration to see if I can use it for this.
04:37.50SeRiI see,
04:39.21SeRihow does vyatta use snat? as source nat?
04:39.43*** join/#asterisk OldMonk (~raju@120.56.163.142)
04:39.45OldMonkhi
04:40.18SeRipqsql is not hard at all.
04:41.40OldMonki'm running an agi, which is executing fine.  near the end, i call $agi->set_extension(1); extension 1 exists in the context.  the agi debug shows <SIP/1001-00000030>AGI Rx << SET EXTENSION 1; however, on completion of the agi, the call ends with: Auto fallthrough, channel 'SIP/1001-00000031' status is 'UNKNOWN'
04:42.10OldMonkif i set_extension($invalid), the i extension is invoked normally
04:42.16p3nguinsnat is source nat, configured with set service nat rule <rule> type source
04:42.53SeRihttp://www.vyattawiki.net/wiki/NAT#Source_NAT
04:43.20SeRip3nguin: many company's have their own defenition of snat thats whay I ask :)
04:53.10*** join/#asterisk joako_ (~joako@opensuse/member/joak0)
04:58.01Naikrovekmy daughter's hamster is going insane.  run run run.  all night long.  run run run run run.
05:04.31*** join/#asterisk KingDavidNYC (~Chris1232@pool-74-96-172-188.washdc.fios.verizon.net)
05:05.30KingDavidNYChello can someone please help me to identify the ip address of the incoming call?
05:05.40KingDavidNYCvia the sip.conf
05:07.08KingDavidNYCam I the only one at home in front of a computer tonite?
05:07.32*** join/#asterisk nix8n82 (~hmg@75-174-154-76.chyn.qwest.net)
05:09.31p3nguinseri: http://imagebin.org/index.php?mode=image&id=188014
05:10.38SeRip3nguin: that indeed looks a bit complicated.
05:11.02p3nguinIt actually seems like a very simple scenario, but the solution seems hard to come by.
05:11.25SeRiKingDavidNYC: what do you mean via sip.conf? If somebody is calling you via sip than do sip show channells in cli and you will see there IP.
05:11.38p3nguinkingdavidnyc: You can't identify an active call by looking at a conf.  You can, however, use "sip show channels" to see addresses of calls.
05:11.52SeRis/tehre/their/
05:12.03SeRishit
05:12.25KingDavidNYCseri: I mean, how can I create a sip.conf entry to route calls from on specific IP addres to an specific context
05:12.30SeRisorry for my fucked up spelling... is one of those nigths again
05:13.29KingDavidNYCseri: the first thing I would do would be to get the ip address from the sip header, when the call comes in, but I was wondering if the other way was the proper way to do it
05:14.26SeRiKingDavidNYC: you dont have to get it from the sip headers as posted from p3nguin and me you do sip show channels in cli and you can see the ip
05:15.30p3nguinIf you just want to create a peer entry to match based on IP address, create a new peer definition in sip.conf using host=THE-IP-ADDRESS
05:15.46p3nguinThis entry will be for a host which will not register to you, but will send calls to you.
05:16.07p3nguinIt will be based on the IP address when you use type=peer.
05:16.29KingDavidNYCI did, but I am getting a weird error..
05:16.40KingDavidNYCI created a sip.con entry
05:17.37KingDavidNYCand I am getting a ton of warnings that say "registration from x failed, no matching peer found
05:18.11KingDavidNYCI created a very simple entry:
05:18.20KingDavidNYC[Harry]
05:18.24KingDavidNYCtype=peer
05:18.49SeRishow us the error
05:18.51SeRi~pb
05:18.51infobot[~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.
05:18.55KingDavidNYChost=the-ip-I want to authorize
05:20.15dijibholy cow m pissed
05:20.17dijibim
05:20.28dijiband eating mmm rice krispy squares
05:20.34dijiband beer
05:20.40dijibsleemans
05:20.42p3nguinkingdavidnyc: Are you trying to create a peer entry for a phone which will register to you?
05:20.49KingDavidNYChttp://pastebin.com/wxhRHRBf
05:21.16KingDavidNYCp3nguin: no, no registration :) ... I just want to validate the ip
05:21.19p3nguinDon't get into a habit of trying to hide information when you're asking for help.
05:21.41p3nguinThis error you're using says a phone is trying to register.
05:21.54SeRi^^
05:22.16KingDavidNYCp3nguin: I just want to receive incoming calls and authorize them by ip
05:22.45KingDavidNYCp3nguin: that's the weird part... I am not trying to register
05:22.54SeRidijib: you ok?
05:23.08p3nguinTo create an entry for any random devie which will not register to you, make it something like this:  http://pastebin.com/GEd3YtyH
05:23.11*** join/#asterisk s[X] (~mark@ppp118-208-26-44.lns20.bne1.internode.on.net)
05:23.59SeRiKingDavidNYC: thats not what the error is saying.... The error defently is showing a peer trying to register
05:24.05SeRi[Dec 11 05:19:24] NOTICE[18971]: chan_sip.c:22318 handle_request_register: Registration from '"harry" <sip:harry@x.x.x.x>' failed for 'y.y.y.y' - No matching peer found
05:25.25KingDavidNYCseri/p3nguin: okay guys, sorry, it can then most likely be a sip phone that is trying to register, but I still dont know how to create the entry in sip.conf to authorize incoming calls
05:25.47p3nguinAlready showed you.
05:25.53SeRip3nguin: +1
05:25.59p3nguinhttp://pastebin.com/GEd3YtyH   <----
05:26.38p3nguinCalls from THE-IP-ADDRESS will match this entry, and no authentication should be performed after that.
05:26.45dijiboh yeah SeRi im good
05:26.54dijibjust doing a case swap on an x86
05:27.03dijibjust finished the swap. yet to plug it in
05:27.26SeRidijib: cool. good luck :) dont burn the mbo :P
05:27.37KingDavidNYCp3nguin: I am going crazy :)... harry is not a phone, it is a sip.conf entry....why in the world is it trying to register?
05:27.46dijibburn out the mobo..... this guy new?
05:27.58dijibits all done, all scres accounted for.
05:28.04KingDavidNYCp3nguin: I promise you it is not a phone
05:28.24p3nguinWhatever it is, it is sending registrations.
05:28.35dijibi need a new i7
05:28.36dijib:(
05:28.48p3nguinYou _could_ allow it to register to you if you wanted.
05:28.57SeRiKingDavidNYC: maybe the source trying to call you is nor calling you instead is trying to register?
05:29.10SeRis/nor/not/
05:29.28p3nguinChange to host=dynamic and let it register.
05:29.49p3nguinBe sure to set a context that it can't do any harm.
05:29.49dijibp3nguin: you are an amaizing asterisk guide, and you must charge people $40 paypal to talk to you.
05:29.55dijibyour that informed
05:30.06KingDavidNYCseri: yes, I think so... the source is another asterisk box, but there is no entry ther called harry
05:30.14p3nguinThey won't pay.  They'll just call me names and continue to ask questions.
05:30.16dijibawemazing
05:30.25dijibwant $5?
05:30.55p3nguinI'll accept any cash that anyone wants to offer.
05:31.03dijibemail
05:31.13dijibppal?
05:31.27p3nguinI'll pm it to you.
05:32.01dijibif i can remember how seri explained multiwindow to me.
05:32.05dijibalt+ huh?
05:32.06dijiblol
05:32.16p3nguinWhich window am I on?
05:32.36SeRiKingDavidNYC: unless you are been the middle box there is no reason why another asterisk box would call you. I think what is happening is that the other asterisk box is not trying to call you via sip is actually trying to register.
05:32.53p3nguinIf I went to window 8, use Alt+8
05:32.58p3nguinor
05:33.01p3nguin/win 8
05:33.16SeRiatl+8 = awesomeness
05:33.24SeRiI love hot keys
05:33.29KingDavidNYCseri: I think so... I got one more box to check
05:33.31p3nguinYou can also use Alt+ right arrow or left arrow.
05:33.52SeRiKingDavidNYC: You are confused and looks like you have it misconfigured.
05:34.02p3nguinThis routing thing is really irritating me.
05:34.14SeRip3nguin: yeap :)
05:34.18dijiblol
05:34.22dijibthere we go
05:34.23p3nguinhttp://imagebin.org/index.php?mode=image&id=188014
05:34.29KingDavidNYCseri: actually, I am beginning to think my box is being attacked
05:34.32p3nguinThe topology is simple.
05:34.32dijib$5.05 canadian comming your way.
05:34.39p3nguinThe task is difficult.
05:35.17p3nguinAmount: $5.00 CAD
05:35.24p3nguinvia eCheck
05:35.38SeRip3nguin: a vlan would have solve this problem in a simple way....
05:36.00p3nguinBut then I'll need two vlan capable switches, I think.
05:36.08p3nguinActually, I know I will.
05:36.27p3nguinBecause in reality, I have more than just the one PC on subnet A.
05:36.46KingDavidNYCdijib: are you seriously sending $5?
05:37.29p3nguinI have a second switch connected to the first switch.  The second switch has systems in both subnets A and B, as well.
05:37.47KingDavidNYCdijib: the man deserves at least $100
05:38.06SeRiKingDavidNYC: go for it :)
05:38.49KingDavidNYCseri: serious, where I come from, $5 is an insult
05:38.54SeRiI dont think two switches.. one would suffice.
05:39.17SeRiKingDavidNYC: Is not the money but the thought that counts. dijib did a good thing in return.
05:40.20SeRi5 dollars is some nice beer money :)
05:41.56KingDavidNYCseri: okay, let's just say it is better I dont give my opinion on that one :)
05:42.04SeRiI do agree p3nguin deserves a lot more but this all part of information sharing so we are all in the same pege here and thats to help on another.
05:42.07ChannelZgoes for a Slurpee
05:42.28ChannelZ(and it's not really "sharing" if you pay)
05:42.49SeRi^^
05:43.15SeRidonation's are welcome
05:44.09p3nguinI'll have to have another switch to tag traffic from the hosts on my other switch.
05:44.10KingDavidNYCquestion: on this one:chan_sip.c:22318 handle_request_register: Registration from '"harry" <sip:harry@x.x.x.x>' failed for 'y.y.y.y' - No matching peer found
05:44.31KingDavidNYCwho is trying to register?  x.x.x.x or y.y.y.y??
05:44.43p3nguinYyyy
05:44.44SeRi"other switch" <----- didnt know that one :)
05:44.48ChannelZhelen_: you ever get your thing working BTW?  I wandered off
05:44.54KingDavidNYCcause for the life of me I dont know whery the y.y.y.y ip comes from
05:45.14p3nguinXxxx is your asterisk.
05:45.23KingDavidNYCthat's right
05:45.24SeRiKingDavidNYC: what is sad is that you are letting that happen. mis configured * for sure.
05:45.52KingDavidNYCthat means I am being attacked?
05:46.22ChannelZIf they're not expected IPs....
05:46.26SeRirandom public IP's should not be trying to register to your box unless you let it happen
05:46.38p3nguinSeRi: I have a core switch with both routers attached, and another switch daisy chained.
05:47.13KingDavidNYCgreat... the only thing that puzzles me is why it says "harry"?... harry was the name of my sip.conf context
05:47.19p3nguinThe second switch has hosts attached which belong in different vlans.
05:47.35ChannelZThey can use any name they want (to try)
05:47.54ChannelZAre you sure this isn't a softphone or something of your own? :P
05:47.55SeRiKingDavidNYC: look in to iptables, fail2ban and alwaysauthreject=yes
05:47.55p3nguinSo I need to tag at that switch by the port.
05:48.09KingDavidNYCbut harry is the name of one of my contexts....
05:48.16ChannelZso?
05:48.33ChannelZEither a lucky guess or someone knows more about your system than they probably should
05:48.48KingDavidNYCexactly
05:48.52p3nguinThat's your fault for having corresponding words.
05:49.14KingDavidNYCI only have one more x-lite phone in my office computer
05:49.27KingDavidNYCdont think so
05:49.46p3nguinIf you want the Harry phone to register, create a peer for Harry.
05:50.28ChannelZand then make an exten called 'balls'
05:50.36ChannelZmmheh heh hmm heh
05:51.17KingDavidNYCp3guin: thanks p3nguin
05:56.15*** join/#asterisk WIMPy (~wimpy@e183095026.adsl.alicedsl.de)
05:56.17SeRiChannelZ: hahahahah!
05:56.28SeRip3nguin: +1
05:56.41SeRiI bet fluffy is the password
05:56.52p3nguinor 'sweaty'
05:56.57SeRihahaha
05:57.27SeRip3nguin: ok I am starting to see the full pic your top....
05:58.27p3nguinhttp://pastebin.com/GEd3YtyH
06:01.02SeRihahahah p3nguin I got one for you!
06:02.09p3nguinDoes it involve routing traffic from 192.168.10.0/24 our another gateway on a different subnet?
06:02.28p3nguins/our/out/
06:06.31SeRip3nguin: http://i106.photobucket.com/albums/m260/xcom7/omfg.jpg
06:06.45SeRithe pic was an ad
06:06.54SeRilol
06:08.28p3nguinThe chick pic was an advertisement?
06:09.58SeRiyes
06:10.08SeRihahaha
06:10.23SeRisomething about texas dl
06:16.39dijibKingDavidNYC: yes i did send him $5 and yes he is worth $1000's
06:16.42dijibno argument
06:17.13dijibp3nguin must explain to me how and why he exists.
06:17.18*** join/#asterisk AmirBehzad (~behzad@87.248.136.180)
06:17.32dijibfuck texas you guys are gun nutz
06:17.40SeRifuck yea
06:18.08SeRipussy's...
06:18.14SeRilol
06:18.15dijiblol
06:18.21dijibdude ive been drinking
06:18.32dijiband therefor im kinof drunk
06:18.41dijiband yeh i need to smoke one im thinking
06:19.35nix8n82smoke one what?
06:20.57*** join/#asterisk gerhard7 (~gerhard7@82-171-103-215.ip.telfort.nl)
06:22.39[TK]D-Fenderdijib: You mean this is you .... BEFORE DRUGS?
06:23.59*** join/#asterisk p3nguin (~xwQ5kwYl6@cobalt.esxi.hosts.a2infotech.com)
06:24.33p3nguinThat is so irritating.
06:26.44SeRi[TK]D-Fender: LMAO
06:26.50SeRip3nguin: whats going on?
06:27.45p3nguinIt's kind of hard to explain exactly, but the gist of it is that when certain clients disconnect from my session, it locks up screen and kills IRC temporarily.
06:28.45p3nguinI don't know the cause, but I do know that it happens on more than one FreeBSD system running screen with irssi in it.
06:37.46p3nguinMaybe I could supernet this and make it work.
06:37.53SeRiweird...
06:38.57SeRimaybe... wouldnt that be a bit to much?
06:39.12p3nguinIt's ridiculous that there isn't some way to route based on source address.
06:39.35WIMPyGet a real Linux.
06:39.50p3nguinI see you aren't a fan of Debian.
06:39.54WIMPyYou don't use FreePBX either, do you?
06:40.03p3nguinI don't use FreePBX.
06:40.23WIMPyErr, if you say, Debinan can;t do it, I may have to retink if it's Linux at all.
06:40.50WIMPyI thought you were tryin vyatta or however that's spelled.
06:40.59p3nguinIt is Vyatta.
06:41.26p3nguinIt's a regular Linux, but with an interface that combines everything into it rather than using a bunch of separate tools.
06:42.48p3nguinIt could be possible that the interface has the ability to accomplish this task, but without knowing what direction to go, I don't know what commands I need to enter into vyatta.
06:42.49WIMPy... which doesn;t allow you to do what you want.
06:43.00p3nguinI tried natting it, but that didn't work.
06:43.31p3nguinThe firewall does not perform that sort of operation, so that's no good.
06:43.49p3nguinRouting tables are based on destination, so that's out of the question.
06:44.23WIMPyYes, but as I said befor I left, you can have multiple routing tables.
06:44.44WIMPyAnd you can have rules (like FROM) to decide which table to use.
06:44.52p3nguinIs there such a thing as source-based routing?
06:45.21p3nguinConventional routing is destination based.
06:45.35WIMPyThat's ehere the rules come in.
06:45.49WIMPy'ip rule' ...
06:48.17p3nguinI'm not sure if I can access that from vyatta.  I know it does use 'ip' for some things, but I would have to find out what commands exist for ip rules.
06:48.37WIMPy'ip rule help'
06:49.56p3nguinI can't do it manually.  That's exactly the same as hand-editing extensions.conf when you use FreePBX.  DO NOT DO IT.
06:50.35WIMPyDo you see a pattern?
06:52.09SeRi:/
06:54.38SeRiha! GV will stay free for local US calls in 2012!
06:54.55SeRiI knew it. there is no way they can become a paid for service been beta
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07:42.02p3nguinDammit.  I thought I had it solved, then ran into a command MISSING.
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07:48.03SeRip3nguin: ?
07:48.33p3nguinA command was renamed without notification in the documentation.
07:48.37p3nguinI found it.
07:48.52SeRiah
07:49.03p3nguinThey renamed "enable-source-based-routing" to "per-packet-balancing"
07:49.11SeRi:/
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07:51.06p3nguinBut it still does not work!
07:52.24SeRi:(
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07:53.57WIMPyWhy am I not surprised that dahdi won't compile?
07:55.33ChannelZI don't know.  Why?
07:56.10WIMPyI upgraded to the latest Kernel before I put the card in.
07:57.12ChannelZIs it some strange new kernel from the future?
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07:57.52WIMPyWhat linux-stable gave me, 3.2.0-rc1.
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07:58.04WIMPy4 rcs behind.
07:58.59WIMPyBut I do wonder why rc1 is considered stable.
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08:01.38ChannelZis still in 2.6-land
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08:03.54elliot98is there anyway anymore to submit a bug for * 1.4?
08:04.19WIMPyYes, but noone wil care.
08:04.19elliot98just upgraded to a new version and there seems some regression
08:04.26WIMPyl
08:04.35elliot98version .37 to .42
08:04.47elliot98so versions are doing something
08:04.55WIMPy1.4 and 1.6 are EOL
08:05.21elliot98true true, but it's a stable system that's been on 1.4 for like forever
08:06.08elliot98so what are the newer versions of 1.4 for?
08:06.12WIMPyThen why did you change it?
08:06.25elliot98to fix another issue
08:07.28elliot98that issue seems to be fixed in the later version, but another thing comes up
08:08.35p3nguinYou'll be lucky if even a serious security vulnerability in 1.4 will receive attention.
08:09.52elliot98but versions of 1.4 are still being released
08:10.26elliot98is it possible to run 2 instances of asterisk on the same maching?
08:11.06WIMPyIf you configure them carefully, yes.
08:13.19elliot98but probably better to set up virtualbox or some sort of viruatlization...do virutalization and PRIs play well together?
08:13.36p3nguinIf your issue is a security problem, it may get attention.  Nothing else really matters.  1.4 has been in security mode for over six months already.
08:13.37WIMPyProbably not
08:14.29WIMPyI had serious problems last WE with a Digium PRI card.
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08:15.24WIMPyErr, vitout any virtualisation that is.
08:15.57p3nguinseri: I had to give up on it for now.  I just changed subnet A to be the same as subnet B so I can assign router B to the dhcp clients.
08:15.58elliot98so it could be the card, not the upgrade?
08:16.40SeRisorry I was not much of help... :(
08:16.48WIMPyelliot98: ?
08:17.11elliot98WIMPy: the mention of having problems with the PRI card
08:17.56WIMPyThat was a dahdi or card issue. But why do you think tha could have anything to do with your upgrade?
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08:18.22WIMPyBut I wouldn't use PRI with pre 1.8 versions.
08:18.22elliot98WIMPy: we've reverted back and forth and it occurs only on the upgrade
08:18.37elliot98diffing chan_dahdi.c with the old version yields a lot of changes
08:18.50elliot98so someone's still in there tinkering with code
08:19.03SeRip3nguin: http://www.ebay.com/itm/Polycom-2201-12560-001-SoundPoint-IP-560-SIP-Phone-/260913553715?pt=LH_DefaultDomain_0&hash=item3cbfa8e133  <---- g722
08:19.17WIMPyAre both the old and the new issue PRI related?
08:20.38elliot98unfortunately  yes, so I can't just pick and choose some modules
08:20.43elliot98one is dtmf
08:20.58elliot98the other is that the card suddenly stops incoming calls
08:21.20WIMPyIsn't DTMF done by the card?
08:21.24elliot98and need to run a "dahdi restart" to correct things
08:21.37SeRip3nguin: http://www.ebay.com/itm/Plantronics-CS70-NC-Headset-System-W-HL10-Lifter-/250951359133?pt=LH_DefaultDomain_0&hash=item3a6dddda9d
08:21.48elliot98it is?
08:22.09elliot98how do you set that up?
08:22.48WIMPyMaybe teh Digium cards need the DSP module for that.
08:23.19elliot98perhaps it has it, but I have yet find any documentation regarding setting it up
08:24.14WIMPyHmm. It could be it's disabled by default. I think I stumbled upon something there.
08:24.26WIMPyIt's a module parameter.
08:24.46elliot98a module paramer in chan_dahdi.c?
08:24.51elliot98*parameter
08:25.09WIMPyThe kernel module for your card.
08:25.24elliot98libpri or dahdi?
08:25.49p3nguinDomino's to the rescue.
08:26.07elliot98but asterisk would still need to be configured somehow to work with it
08:26.27WIMPyWith what?
08:26.37SeRip3nguin: lol
08:26.41SeRipizza?
08:27.04elliot98WIMPy: with the onboard DTMF analysis
08:27.22WIMPyWhy do you think it might not do so?
08:27.34SeRip3nguin: should I put a bid on that 560?
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08:27.45SeRiI can move my 501 to a new home. :)
08:28.34elliot98WIMPy: well, for starters, we don't want asterisk to also try to diagnose DTMF, because double DTMF can result
08:29.14WIMPyOff course I don;t KNOW what Digium does, but I would expect them to support their hardware in Asterisk.
08:30.16elliot98seemingly, but how it's done...there must be some doc somewhere
08:30.17p3nguinYes, pizza -- I'm pretty friggin' hungry.
08:30.26SeRinom nom nom
08:30.35p3nguinWhat will you to with the 501?
08:30.41elliot98and also, should PRIs be set to inband or rfc2832?
08:31.00SeRiIll send it to its "original" home :P
08:31.11WIMPyelliot98: The doc is called modinfo
08:31.20p3nguinBack into the closet?
08:31.23elliot98where?
08:31.28elliot98in dahdi?
08:31.39SeRinah ill send it to you. I did offered it to you some time ago.
08:31.48WIMPyAnd PRIs usually use inband DTMF.
08:31.54SeRiI just didnt come threw :(
08:31.55WIMPyAt least if it's peer to peer.
08:32.11SeRiIll place my bid at 130.00
08:32.13p3nguinWill you benefit from the 560?
08:32.18SeRiYes ser.
08:32.19elliot98what else is there besides peer to peer?
08:32.21SeRi6 lines and HD
08:32.26WIMPyNo, it's the dirver. modinfo that one.
08:32.36SeRiI been wanting more lines. I actually need them
08:32.42p3nguinHow many lines does the 501 have?
08:32.46SeRi3
08:33.14WIMPyelliot98: Terminal to switch. That would certainly be keypad.
08:33.43p3nguinDo you configure an account for each line key, or just use them for speed dials and stuff like that?
08:34.25SeRispeed dials and stuff like that.... But you could use each one for one individual account
08:34.30elliot98keypad? need to bone up on PRI stuff
08:35.13p3nguinI'm not all that familiar with how Polycom deals with the accounts and line keys.
08:35.45WIMPyelliot98: Is the DTMF issue the old or the new one? And what's the other?
08:35.47SeRiRight now #1 Hot #2 Brother Office #3 Brother cell
08:36.05elliot98the DTMF is ongoing, the new is issue occurs after the upgrade
08:36.12SeRiIts very simple actualy..
08:36.20SeRis/its/it's/
08:36.41SeRiinfobot: you drunk?
08:38.57elliot98how does one make any changes in the settings after seeing the params in modinfo?
08:39.24WIMPyuse them as parameter when you load the module,
08:39.48WIMPyOr put them in the modprobe.d somewhere.
08:39.53elliot98in insmod?
08:40.16WIMPyIf you're still using insmod, yes.
08:40.41elliot98what is dtmfthreshold?
08:42.30WIMPyProbably either a level or a duration. No idea.
08:43.34WIMPyThe default seems to be 1000.
08:47.52WIMPyThe lower the value the more hits.
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08:55.02pdtpatrick__Question .. which is more powerful... AEL or LUA ?
09:06.45elliot98is there a way to restart a specific channel is Dahdi?
09:07.43WIMPyNo
09:08.00WIMPyDid you only upgrade Asterisk or libpri as well?
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09:11.43elliot98WIMPy: upgraded to latest libpri and dahdi
09:12.13WIMPyThen you should try the new Asterisk with the old libpri.
09:13.45elliot98WIMPy: hehe...I only upgraded after the old libpri/dahdi was failing.  Thought that was the problem, but aparently not so
09:13.49elliot98sorry
09:14.04elliot98only upgrade libpri/dahdi after asterisk started to fail
09:14.18elliot98so tried all variations already
09:14.22WIMPystarted to fail? What does that mean?
09:14.33elliot98WIMPy: that the pri lines started to fail
09:15.02WIMPyYou are aware that you need to rebuild Asterisk after changing libpri?
09:15.49elliot98WIMPy: did a full reinstallation after the libpri update
09:16.21elliot98the chan_dahdi.c code has significant changes there...don't know who to contact to find out what happened
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10:24.26pdtpatrick__WIMPy: if ur PRI is active, once u load chan_dahdi.so .. it should pick up all available channels and resume working (if you disabled prior) right?
10:25.17WIMPyDisabled? How?
10:25.57pdtpatrick__well the PRI was shutdown by provider for a bit so i unloaded the module and switched over to using SIP
10:26.04pdtpatrick__but then looks like they will be turning it back on
10:26.11pdtpatrick__so i should just have to load the same module
10:26.27pdtpatrick__and the channels should (if active) should start working again right?
10:26.44WIMPyyes
10:26.59pdtpatrick__cool deal. Thanks
10:27.02WIMPyOr you just load it and wait until it works.
10:27.14WIMPyIt shopuld find out by itself.
10:27.50pdtpatrick__will try that
10:27.52kaldemarand there's no need to unload the module.
10:28.12pdtpatrick__well i didn't want all those channel unavailable messages to the screen
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12:50.26helen_ChannelZ: No. :(
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18:11.11ChannelZhelen_: so what does the console say when you try to call?  I don't see anything blaringly wrong in your config you posted so something else has to be at play
18:14.55helen_ChannelZ: http://pastebin.com/MwNdT5C7
18:15.32ChannelZneed more verbose output, but the first line pretty much means whatever device you were trying to call is not online or doesn't exist
18:16.00helen_It definately exists.
18:16.02helen_1min...
18:16.11ChannelZcore set verbose 3
18:16.19helen_ChannelZ: It is set to that.
18:16.32WIMPyCheck the device with 'sip show peers'.
18:16.36ChannelZthen your console is not outputting verbose because there should be more than that
18:18.36helen_WIMPy: I told Tom to turn on his client
18:18.53helen_and it's coming up unspecified under host on his extension.
18:18.56helen_sighs
18:19.11ChannelZsee
18:19.15ChannelZtold you so
18:19.35helen_ChannelZ: ?
18:19.52helen_I don't get it...
18:20.03helen_Everyone else has been having problems connecting apart from me.
18:20.05ChannelZ"the device doesn't exist or is not online"
18:20.12helen_it exists
18:20.19helen_just not online
18:20.27ChannelZyes, like I said, again
18:20.38ChannelZYou can't call devices you can't reach
18:20.46ChannelZso mystery there
18:20.53ChannelZs/so/no/
18:21.04helen_It seems Tom dialed the voice maqil extension
18:21.07helen_[Dec 11 21:20:10] NOTICE[10126]: chan_sip.c:20163 handle_request_invite: Call from '' to extension '4242' rejected because extension not found.
18:21.14ChannelZNo he tried to call 4242
18:21.26helen_yes
18:21.36helen_which is the voice mail extension
18:21.50helen_Call from ''
18:21.50ChannelZapparently not, because it is "not found"
18:21.54helen_what is that all about?
18:22.08helen_ChannelZ: But extension 4242 is there
18:22.12helen_because I can call it
18:22.13ChannelZNo user name/anonymous SIP
18:22.25ChannelZContext is everything.
18:22.42helen_Ok
18:22.46helen_Forget that bit
18:22.53helen_How do I make it so I can call Tom
18:22.56helen_and he can call me?
18:23.03helen_We need this urgently.
18:23.04ChannelZEither that was a random person on the net trying that exten, or the other guy's device isn't configured right/your sip.conf isn't configured right and it's not matching a peer and not sending it to the right context.
18:23.24ChannelZI don't even know who Tom is, that's new.  Your old config referred to 101, 102, 103..
18:23.25helen_ChannelZ: YOu just said my configs are fine earlier.
18:23.36helen_ChannelZ: Tom = 101
18:23.41helen_Helen = 102
18:23.51helen_Sorry for being confusing.
18:23.54ChannelZsip show peer 101
18:24.22ChannelZor not even that.  just do "sip show peers"
18:24.23ChannelZIs 101's Host "unspecified"?
18:25.18helen_ChannelZ: Doesn't say.
18:25.26ChannelZWhat do you mean it doesn't say
18:25.45helen_ChannelZ: THere is no host thingie in there.
18:25.59ChannelZsip show peers    ?
18:26.20helen_Unspecified
18:26.50ChannelZSo his phone is not registering.  You've got to fix that.  Whether because his device is not configured right, or it's being blocked by a firewall on his or your side, I couldn't say.
18:27.21helen_ChannelZ: Not my side
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18:27.46WIMPyIf that anonymous call to 4242 was from there, it must be missing the right configuration.
18:29.11ChannelZWhat is 101?  A softphone?
18:29.24helen_ChannelZ: Yes
18:29.29helen_It's my softphone
18:29.40ChannelZI thought it was Tom's
18:29.42ChannelZ:P
18:29.47helen_no?
18:29.52helen_Tom's is 102
18:29.57ChannelZIn event it's not configured right.
18:30.02ChannelZthen why did you say
18:30.03ChannelZ<helen_> ChannelZ: Tom = 101
18:30.04ChannelZ<helen_> Helen = 102
18:30.14helen_I got it the wrong way around.
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18:31.26ChannelZsighs
18:31.33helen_Is it me or is this a hard task?
18:31.53ChannelZIt's you.  No offense meant
18:32.01ChannelZIt's only hard if you have bizzarre networking setups to work around
18:32.16ChannelZSo I don't know what device you're dialing, and which from.
18:32.26Neptuhej I have a headache trying to make work a sip provider like voipgain.com...
18:32.34helen_I used to use Axon PBX but it's not free
18:32.39helen_and it worked fine
18:32.47helen_But asterisk is too complex for me.
18:32.48ChannelZBut if any of them say 'unspecified' in sip show peers, it's never going to work until you get those devices to register, or hard-code them with a static IP.
18:32.55ChannelZAND the phone has to be configured right as well
18:33.10[TK]D-Fenderhelen_[Dec 11 21:20:10] NOTICE[10126]: chan_sip.c:20163 handle_request_invite: Call from '' to extension '4242' rejected because extension not found.
18:33.12ChannelZWhat softphone is it?
18:33.21helen_ChannelZ: Twinkle
18:33.25[TK]D-Fenderthe '' is a tip-off that the calling agent isn't identifying itself properly
18:33.27helen_me and Tom use twinkle
18:33.36[TK]D-FenderFor which I'd have to ask precisely how this call was dialed
18:33.40[TK]D-Fender^^
18:33.41helen_All OptixLayer staff use twinkle or express talk
18:33.53ChannelZAnd you've added a SIP account to it properly, told it the name (101 or 102) and the password, and the hostname/IP of your Asterisk box?
18:33.54[TK]D-Fenderhelen_: So what precisely did they put in the dial box?
18:34.04Neptuhttp://pastebin.com/vxRWjCRv
18:34.12helen_[TK]D-Fender: ?
18:34.23helen_WHy do you want to know that?
18:34.42helen_Theres 20 of us
18:35.16Neptuanyone can give me hand creating my first  voip peer... im with thos for 20 min and maybe i need a hand here
18:35.19[TK]D-Fenderhelen_: Because that might clearly explain the reson for the call being misidentified
18:35.24[TK]D-Fenderhelen_: details matter
18:35.50Neptuhttp://pastebin.com/vxRWjCRv -> this is my sip configuration they mention i should use a register command aswell but did not work
18:35.51[TK]D-Fenderhelen_: Of course it'd help is we saw the call attempt with SIP debug enabled.
18:36.07helen_[TK]D-Fender: They've been calling me
18:36.11helen_to test the line
18:36.17helen_but haven't gotten through.
18:36.33[TK]D-FendermepWell there is no register in there.  Also almost no provider anywhere should ever be "nat=yes", and should be "type=peer" as well
18:36.52[TK]D-Fenderhelen_: Yes I know.  I just asked you precisely what they put in the dial box to try that call
18:37.02helen_[TK]D-Fender: 101
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18:39.05[TK]D-Fenderhelen_: pastebin your peer configs and the CLI w/ SIP DEBUG enabled for the failed attempt
18:39.53helen_[TK]D-Fender: SIP DEBUG enabled?
18:40.19helen_peer configs as in extensions.conf and sip.conf?
18:40.21[TK]D-Fenderhelen_: "sip set debug on"
18:40.36[TK]D-Fenderhelen_: and yes, sip.conf masking only passwords
18:43.29helen_[TK]D-Fender: http://pastebin.com/3ZTV40GL
18:43.36helen_I removed the passwords
18:44.35[TK]D-Fenderhelen_: Ok, lets look at the failed call now
18:45.52helen_[TK]D-Fender: Screenshot from a staffer.
18:45.55helen_http://magix.megapowers.net/img/wm7YB.png
18:46.28[TK]D-Fenderhelen_: Don't care what their client says.  Look at * CLI w/ SIP debug as I've requested a few times
18:47.22WIMPySmells like a networking issue.
18:48.00helen_[TK]D-Fender: SIP/2.0 401 Unauthorized
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18:49.07[TK]D-Fenderhelen_: Lets try this again....
18:49.11[TK]D-Fenderhelen_: pastebin the call
18:51.03helen_[TK]D-Fender: Theres too much to pastebin.
18:51.36[TK]D-Fenderhelen_: I doubt that  Pastebin can handle a lot....
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18:51.56[TK]D-Fenderhelen_: I doubt that.  Pastebin can handle a lot....
18:52.08[TK]D-FenderBetter... period makes all the difference in there.
18:52.32helen_[TK]D-Fender: YOu said pastebin the call.
18:52.46[TK]D-Fenderhelen_: Yes, and I've seen pastebins well past 3000 lines.
18:53.04[TK]D-Fenderhelen_: So somebody telling me "too much" does hold any weight historically
18:53.13helen_[TK]D-Fender: Can I just pastebin everything?
18:53.25[TK]D-Fenderhelen_: apparently, yes
18:53.30[TK]D-Fender~pb
18:53.30infobot[~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.
18:53.34[TK]D-Fenderwww.pastebin.com
18:53.37[TK]D-Fenderslow bot..
18:53.39[TK]D-Fenderdanit
18:53.41[TK]D-Fenderdangit
18:55.37helen_STUPID ROUTER
18:55.39helen_Ok
18:55.40Neptuhaving the same problem to connecto to an external sip provider
18:55.46helen_[TK]D-Fender: It goes off my terminal
18:56.00[TK]D-Fenderget a bigger scroll-back buffer
18:56.08helen_OMFG
18:56.16[TK]D-FenderNeptu: You should be looking at SIP debug as well.
18:56.17WIMPyUse tee
18:56.29[TK]D-Fenderno need for "tee".
18:56.48helen_I hope this isn't a password
18:56.51helen_Really destroying SIP dialog '
18:56.54helen_after that
18:56.59helen_I hope it isn't a password
18:57.06Neptu[TK]D-Fender: im on it jsut want to be sure my conf file are ok
18:57.18[TK]D-FenderNeptu: Can't be until you actually look.
18:57.50[TK]D-FenderNeptu: And aside from the corrections I already gave you
18:58.30Neptu[TK]D-Fender: Dial(SIP/0046851174812@voipgain) -> this is incorrect?
18:58.55Neptui feel this might be one of the problems
18:59.02[TK]D-FenderNeptu: Not the preferred means, but should be OK.
18:59.08[TK]D-FenderNeptu: And feeling != loking
18:59.11[TK]D-Fenderlooking*
18:59.18helen_Forget it.
18:59.22helen_This is wasting my time.
18:59.28helen_I don't have time to waste.
18:59.58[TK]D-FenderPastebinning a call should have taken 30 seconds
19:00.08[TK]D-FenderTime is being wasted.
19:00.44WIMPyIf you don't have lots of time to waste, Asterisk is not for you.
19:00.45helen_[TK]D-Fender: AMd you didn't answer my question.
19:00.55helen_WIMPy: So then what is for me?
19:00.58[TK]D-Fenderhelen_: What question?
19:01.04helen_There are no other PBXs out there.
19:01.19Neptu[TK]D-Fender: the command is sip debug??
19:01.23helen_13:56 < helen_> Really destroying SIP dialog '
19:01.23helen_13:56 < helen_> after that
19:01.23helen_13:56 < helen_> I hope it isn't a password
19:01.29[TK]D-FenderNeptu: Look up
19:01.31helen_?
19:01.49[TK]D-Fenderhelen_: that is a statement, not a question.  And now that you're considering it one, no.
19:01.56WIMPyhelen_: I don;t know what your needs are.
19:02.14Neptusip set debug on
19:02.17Neptuok
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19:02.51helen_Does anyone know of any other PBX software? Don't you dare tell me to google.
19:02.56helen_I already did that.
19:03.40helen_Ok i'm getting stressed.
19:03.45helen_Time out from the computer.
19:03.56WIMPyhelen_: You will get used to that.
19:04.27WIMPyAnd wikipedia might have a comparative list of available software.
19:05.27WIMPyUsing telephones is constantly getting more complicated.
19:05.42[TK]D-FenderSo rather that showing a call which should take all of 30 seconds... instead we get paranoia, delays no output, and an immediate attempt to bail for something else.
19:05.46WIMPyUnfortunatly without giving more features. Rather the opposite.
19:06.00Neptu[TK]D-Fender:  http://pastebin.com/3HSTnYHF -> Seems the dial is not finding the voipgain peer defined in sip.conf...
19:06.03[TK]D-Fenderlittle speed bump now = full stop
19:06.47Neptu[TK]D-Fender: I did sip show users and i do not see the voipgain peer there
19:06.54SeRi[TK]D-Fender: I am now bidding for a 560 :)
19:07.16[TK]D-FenderNeptu: "sip show peers".
19:07.39[TK]D-FenderNeptu: And I did not see your corrected condfigs so I have no idea what you have in there right now.
19:07.55Neptu[TK]D-Fender: I get the same as sip show users 2 sip extension 1001 and 1000
19:08.20Neptu[TK]D-Fender: I pastebin my confs w8
19:08.20[TK]D-FenderSeRi: 5XX is an odd price-point in retail.  If you get an agressive win against it it'd be a very nice phone though
19:08.38[TK]D-FenderNeptu: Never use "sip show users".  Virtually useless.
19:10.13SeRi[TK]D-Fender: 124.00 is the bid up to. I dont want to go over 130.00. That includes free shipping. I have not been able to fins it less than 200.00. I think I am doing ok so far. the 321 came home for 24.50.
19:10.26SeRis/fins/find/
19:10.31[TK]D-FenderSeRi: Good up through 160 maybe...
19:10.36SeRio wow!
19:11.13SeRinice to know. I guess ill hang in there. Thanks [TK]D-Fender
19:11.14[TK]D-FenderSeRi: It is backlit, HD, etc... check out the retail price range between 335, 450, 560, 650
19:11.15Neptu[TK]D-Fender: ai ai never use sip show users... this are my confs... http://pastebin.com/SXv1H4kD
19:11.20[TK]D-FenderSeRi: www.telephonydepot.com
19:13.08SeRi[TK]D-Fender: holly shit! I better hang in there! :D
19:13.16[TK]D-FenderNeptu: From the initial looks of things we should be seeing [voipgain] .... AND [1002].  thats the first and third entries missing.  Something is broken there.
19:13.25[TK]D-FenderNeptu: Make sure you are reloading your config sproperly
19:13.37[TK]D-FenderNeptu: and that you're actually editing the proper file
19:13.45Neptusip reload?
19:13.49Neptuextension reload?
19:14.03[TK]D-FenderNeptu: Should do it.  "reload" should grab everything and I always do anyway
19:14.41Neptudoing it
19:15.27Neptuok i check what can be broken at 1002 and voipgain... hope i can find it im not familiar with the sintax
19:18.58[TK]D-FenderNeptu: Syntax looks OK... tht's why I'm wondering if you are working in the actual proper config file, that permissions haven't gotten screwed up, etc
19:19.15[TK]D-FenderNeptu: "ls -la /etc/asterisk" <- PB
19:19.29Neptuw8
19:19.39Neptuim doing everything as root
19:19.50Neptuand seems strange extension 1002 is not loaded...
19:20.45[TK]D-FenderNeptu: How was * installed?
19:21.02Neptulrwxrwxrwx 1 asterisk asterisk 45 Dec  7 19:28 sip.conf -> /var/www/html/admin/modules/core/etc/sip.conf
19:21.15Neptuis the asterisk now version so should be all in place
19:22.49WIMPyhelen_: Looking for new problems instead of solving existing ones?
19:23.14[TK]D-FenderNeptu: ... you're hand editing a FreePBX system,?
19:23.44Neptu?
19:23.50[TK]D-FenderWIMPy: I'd leave this one alone until ready to concentrate on fixing something...
19:24.02Neptuim writing sip.conf and extensions.conf yes
19:24.09helen_WIMPy: No
19:24.17helen_Looking to switch back to Axon
19:24.18[TK]D-FenderNeptu: that symlink is a dead give-away that you are hand editing the sip.conf on a system that was used with FreePBX to manage it
19:26.37Neptu[TK]D-Fender: menaing i need to do it someother way??
19:27.01[TK]D-FenderNeptu: Meaning you shouldn't be hand-editing file at all and should be using their GUI.
19:27.17Neptummmmmmmmm
19:27.27Neptuok how i get to access the GUI?
19:27.32[TK]D-FenderNeptu: And what you've done already probably breaks it a bit as it is
19:27.41[TK]D-FenderNeptu: via a web-browser...
19:28.16Neptu[TK]D-Fender: user and pass by default?
19:29.05[TK]D-FenderNeptu: "The default username/password is freepbx/fpbx"
19:29.49Neptuok im in
19:29.56Neptulet me familiarize with the enviroment
19:30.03Neptuand see if i make it work
19:30.36[TK]D-FenderNeptu: As I said, you need to undo everything you've dones.. including your dialplan.. this is likely to break things.'
19:30.53Neptuai ai
19:40.11Neptu[TK]D-Fender: the web feels dam anoying...
19:41.24[TK]D-FenderNeptu: Not a common first reaction....
19:42.02[TK]D-FenderNeptu: But if you want to run your system yourself it is very rewarding.  You should re-install your PBX with a base non-gui setup and start from there
19:42.30[TK]D-FenderNeptu: Do backup your previous work though as those are a decent start
19:43.47Neptui cna not even register now my softphone...
19:43.56Neptuand I feel i loosed control
19:44.02Neptuw8
19:44.21SeRip3nguin: you avail?
19:44.31p3nguinBOOM!
19:46.39WIMPyWhat did you break?
19:46.49SeRip3nguin: I am having another issue with voip.ms reseller interface
19:47.00p3nguinwimpy: my Arch Linux
19:47.32SeRihow can a one account for pay pal fees? if the user pays 25.00 dollars they expect 25 dollars....
19:47.50SeRis/a one/one/
19:48.04p3nguinIf you pay $25, that's what you get.
19:49.15SeRiIn their accoutn it shows that but in my pal there is 23. and some change
19:49.55p3nguinIf someone adds funds via portal, why does it end up in your paypal account?
19:50.31SeRiThe reseller portal links the pay pal account to your pay pal account
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20:12.51Neptuthis is a mess
20:15.54SeRip3nguin: Is a bit confusing but looks like I got it figured out.
20:16.06SeRiall deductions happen at the users interface
20:17.11*** join/#asterisk KingDavidNYC (~Chris1232@pool-74-96-172-188.washdc.fios.verizon.net)
20:18.15KingDavidNYChello guys, I am scratching my head here because verbose stops after the first call
20:19.02p3nguinseri: Does that mean that if a person deposits $25, it will show less than $25 in the interface?
20:19.26SeRiNo. I have an open a ticket for that.
20:19.46SeRianother bug. voip.ms has to find a way to account for that for resellers
20:20.16p3nguinIt's almost like you are the very first person to ever try to use the reseller interface.
20:21.11SeRino shit.... I feal the same way. I cant even belive that they would never thought about that. I guess people just didnt bother for a reason
20:24.42SeRithey only thing I can come up with is to include a small extra fee under the monthly fee to account for pay pal fees. but that would be bull shit and corporate like. verzion math at is best.
20:26.50p3nguinLet's say they don't make a deposit every month.  You're either going to be undercharging or overcharging.
20:27.40p3nguinOr if they make a deposit of more or less than you've accounted for, the same could be true.
20:28.01SeRiIndeed
20:28.38SeRiwell lets see how it goes. I am testing the grounds with my family.... I have a few monkeys I guess....
20:30.34KingDavidNYCcan you guys help me? this doesn't make sense.. when I call via a DID, I get verbose, but when I call via x-lite, the first call gets verbose and then no more verbose from calls made to that phone
20:32.17WIMPyKingDavidNYC: The only thing I can tell you is that I have the same issue since about a week.
20:32.34WIMPyWhat version are you using?
20:32.35KingDavidNYCI am using 1.6.2.20
20:33.06KingDavidNYCWIMPy: it doesn't make sense
20:34.15[TK]D-FenderKingDavidNYC: Verbose TO a phone?  * can't be exluding based on some destinatin.
20:34.28[TK]D-FenderKingDavidNYC: You should be showing us both attempts
20:34.51WIMPyKingDavidNYC: Hmm, that's definitely older than my issue.
20:34.52KingDavidNYCI mean, when I initiate calls with x-lite this happens
20:35.10WIMPyBut for me it fails as soon as the first call starts.
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20:35.55KingDavidNYCthe first call fails, and then subsequent calls dont seem to arrive to the server
20:36.10[TK]D-FenderKingDavidNYC: that's the caller, not the callee
20:36.12WIMPyIf it happens I don;t even get any output from things like 'sip show peers' or 'core show application dial' until I restart the remote console.
20:36.30[TK]D-FenderKingDavidNYC: Details and debug please
20:36.36WIMPyThe calls don;t work or you just don;t see them?
20:37.05WIMPyAnd why have I been unable to press ' for quite some months now?
20:37.32p3nguinMaybe your ' moved.
20:38.41WIMPyI didn't change keyboard this cenury.
20:39.29Neptu[TK]D-Fender: reinstaling * from scratch...
20:40.37Nepturecomend freepxb or asterisk gui??
20:41.14KingDavidNYCI got it!!  I was sending the calls from the x-lite phone to a different context!
20:41.39NeptuKingDavidNYC: congratz!
20:41.45[TK]D-FenderWhich should have shown up in CLI like everything else
20:42.09p3nguinneptu: No.
20:42.29KingDavidNYCTK: yes TK, that's what has me puzzled
20:43.13KingDavidNYCTK: but I am not going to bother....got a route to have ready in a couple of hours
20:43.19KingDavidNYCthanks a lot guys
20:44.32p3nguinPaper route?
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20:48.02KingDavidNYCwholesale route
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20:49.05KingDavidNYCnowadays asterisk people get so low paid that we have to take anything that comes our way :)
20:50.39[TK]D-Fender"wholesale route" sounds like about 10 minutes work on the outside...
20:51.45KingDavidNYCWEll... they had a2billing, and since I did not want to spend the time learning it, I am now spending the time writing some basic least cost routing and routing scripts
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20:52.36KingDavidNYCI figure they may come in handy next time
20:53.33p3nguinI need to do that, too.
20:54.03p3nguinI need to compare the cost based on destination and route the call accordingly.
20:54.48KingDavidNYCI am using mysql+php
20:57.39p3nguinI don't code, so I'll end up hiring someone to do it for me if I can't do it with basic sql commands and a unix-like shell.
20:58.38KingDavidNYCI might jsut give you my code
21:05.18p3nguinseri: Does a chili pepper get more intense spiciness as it gets older and dries out, or does it get milder?
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21:28.17elliot98gives a double arm wave
21:28.46elliot98ok, so I need to submit a bug for Asterisk 1.4
21:28.51elliot98is that even possible?
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21:34.48patrickodI'm having trouble debugging two soft clients on the same network. One is registering and functioning correctly the other keeps getting 408 errors and timeouts
21:35.53patrickodI can switch credentials and the same one still works whil the other continues to fail
21:43.54patrickodscratch that. For some reason it's working now. weird
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22:23.11*** topic/#asterisk is Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 10.0.0-rc2 (2011/11/15), 1.8.7.2 (2011/12/08), dahdi-linux 2.5.0.2 (2011/10/21), dahdi-tools 2.5.0.2 (2011/10/21), libpri 1.4.12 (2011/07/06) -=- Visit the official Asterisk wiki: wiki.asterisk.org
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22:27.09Neptuhej trying freepbx and i think i almost have it ready but still my trunk seems is not connecting properly... http://pastebin.com/FXUPWdpE
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22:28.05Neptuany ideas?
22:28.24beccaraanyone here got post dial delay logging with asterisk 1.8?
22:29.29WIMPybeccara: Can you describe what you're after?
22:29.52beccaraI'm wanting to log post dial delays into the CDR's
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22:30.10WIMPyWhat delay?
22:30.37WIMPyFrom dialling to getting a response? To answer? Or what exactely?
22:30.50beccaraPDD, The time it takes from sending the invite to a trunk carrier and the carrier responding
22:31.20beccarathe only thing floating around is patchs for asterisk 1.2
22:31.44p3nguin~freepbx
22:31.44infobot[~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
22:31.47p3nguinneptu: ^^^^^^^^^^^^^
22:31.51WIMPyI don't see any way to find out about that.
22:32.09beccarait's been done, it's possible
22:32.24beccarait was under 1.2 and FPBX seems to have those sorts of options aswell
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22:34.50Neptup3nguin: hej donno how to solve this I added the registration chain but the trunks are not working...
22:35.32p3nguin~freepbx
22:35.32infobot[~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there, or fpbx
22:35.34p3nguinneptu: ^^^^^^^^^^^^^
22:35.56p3nguinThis is a FreePBX concern.  And that does not concern us here in #asterisk.
22:36.40Neptuok, the problem is the same... need to configure a trunk and have it operational
22:36.44p3nguinNow did you understand what I'm saying?
22:36.53Neptuyep
22:37.15p3nguinSo you'll go over there and ask for help configuring your system by FreePBX, yes?
22:37.34Neptucan u help me configure the trunck over sip.cfg?
22:37.56p3nguinWhat part of "FreePBX is unable to be supported here" are you having the most trouble with?
22:38.06p3nguinIt's not an asterisk issue.
22:38.11p3nguinYou aren't using asterisk.
22:38.15Neptuim not talking about freeçPBX i mean from editing the files
22:38.28Neptu;)
22:38.33p3nguinYou're using FreePBX.  You don't edit files when you're using FreePBX.
22:38.56Neptuwell they are included as sip_additions.cfg as far as I saw
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22:39.05Neptuso mix configurations are possible
22:39.12p3nguinGo ask in the appropriate channel.  This is an asterisk channel.
22:40.21p3nguinYou don't go to an automobile repair center and ask for help laying carpet in your bedroom.
22:41.26Neptup3nguin: Im asking you to help me to fix sip.cfg file WITHOUT freepbx to have a peer configration... that is all
22:41.32p3nguinCan't be done.
22:41.38Neptu?
22:41.44p3nguinYou use FreePBX.  That's all there is to it.
22:41.54WIMPyIf you start editing that, it will the reason why it fails.
22:41.58p3nguinIf you didn't use FreePBX, I'd help you edit sip.conf.
22:42.07Neptuok if i install a fresh version of asterisk without freepbx can u help me then?
22:42.08p3nguinBut You don't use Asterisk, so I can't help you.
22:42.32Neptuok give me 10 min i get a new fresh compy without freepbx
22:42.37Neptu;)
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22:45.13beccaraseems like asterisk just doesn't have PDD, Freeswitch does http://wiki.freeswitch.org/wiki/PDD
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22:45.54p3nguinYou're trying to measure the post-dial delay?
22:46.18beccarayep
22:46.21beccarain cdr's
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23:09.43SeRidamn kids where going crazy dialing numbers.
23:09.56SeRiI am going to have to restrict outbound calls to those in the db only
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23:16.29SeRip3nguin: you avail?
23:16.50p3nguinsure.
23:17.12SeRihttp://www.ebay.com/itm/250951359133?ssPageName=STRK:MEWAX:IT&_trksid=p3984.m1438.l2649
23:17.14p3nguinDo you want them to be able to dial anything at all?
23:17.17SeRiwhat do you think?
23:17.50SeRip3nguin: no all I want them to dial is a 4 digit ext and 911
23:18.07p3nguinCreate a new dial plan for their phone.
23:18.14SeRigot it
23:18.48p3nguinVery simple.  Limit it only to patterns or explicit extensions you want them to dial.  Do not include other contexts that have the ability to dial other extensions.
23:19.03SeRigot it.
23:19.09SeRiI think I am on the right track
23:19.12SeRiThanks
23:19.16SeRidid you see my link?
23:19.18p3nguinFor example, [kids] context, include "internals" and an extension for 911.
23:19.29SeRiindeed
23:19.35SeRi+1
23:19.52p3nguinIf you like that sort of headset and it isn't too expensive for you, grab it.
23:20.39SeRiseems resonable... I am just to sure about the in ear thing...
23:20.56SeRimy ears dont like that
23:21.07SeRibut for 40 dollars I migth give it a spin
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23:25.19p3nguinYou can always try to resell it if you don't like it.
23:26.07SeRitrue.
23:26.16SeRiI told my brother to buy it for me :P
23:26.19SeRilol
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23:48.14p3nguinI just keep running into limitations of Vyatta.
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23:49.14p3nguinFirst the lack of source-based routing and/or policy-based routing, now it's a problem with hairpin NAT (reflective NAT) not working when there's a dynamic WAN IP address.

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