irclog2html for #asterisk on 20070202

00:01.29DrukenLPYi'd go with a database call on a variable for holiday's....
00:02.12DrukenLPYas for checking to see if a file exsists or not.. well, i highly doubt it.. i personally don't know of a way
00:02.42*** part/#asterisk mkrufky (n=mk@unaffiliated/mkrufky)
00:03.19*** join/#asterisk a1fa (n=a1fa@unaffiliated/a1fa)
00:03.30*** join/#asterisk map7 (n=map7@teksup41.lnk.telstra.net)
00:03.51a1fais there a guy that works for kneedraggers?
00:03.59map7how do i get asterisk to check for call files in the spool directory more frequently?
00:04.14JTthat sounds like a company that does contract killings
00:04.57coppicewhy are they called contract kilings? you can't exactly take them to court for breach of contract
00:05.10JTverbal contract :)
00:05.33JThey, it may not be a legal contract, but it works roughly like any other contract
00:05.51IOscannerif you are in Texas verbal is legal
00:05.56coppicewell, the idea is for the population to contract
00:06.25JTverbal contracts are legal in a lot of places, they are just *Extremely* hard to enforce
00:06.30JTone person's word against another's
00:06.40JTcoppice: indeed
00:06.50a1faJT : motorcycle gear
00:07.01a1faJT : the owner usually hangs out here.. i was gonna bitch him out
00:07.01JTah ok
00:07.01IOscanneryep just a phone recording of the verbal agreement
00:07.07JTa1fa: i se
00:07.08IOscannerAsterisk recording ;)
00:07.21a1faJT : they dont have a return policy on sale items
00:07.29JTIOscanner: illegal in most places without all parties to the conversation being informed
00:07.37a1faJT : and my leather suite is 1 size bigger than it should =)
00:07.38JTIOscanner: unless there's a warrant
00:07.43IOscanneryep
00:07.57IOscannerrecord on demand
00:08.12a1farb
00:08.12rudholmin the US, some states are single-party consent, and some are both-party consent.
00:08.49rudholmwhen a phone call crosses jurisdictions that have different laws, it's unclear, but to be safe, you should follow the stricter jurisdiction's laws.
00:08.57DrukenLPYrudholm: what about a call from a single to a double? hehehe
00:10.26rudholmspeaking of knee dragging, my knee sliders are worn down
00:12.57*** join/#asterisk errr (n=errr@fedora/errr)
00:12.57*** join/#asterisk booray (n=ray@adsl-71-156-59-223.dsl.irvnca.sbcglobal.net)
00:14.24boorayquiet in here...
00:14.35IOscannerTime for food
00:14.52booraymmm... food
00:14.56*** join/#asterisk Snake-Eyes (n=blog@203.220.55.70)
00:15.15IOscannerThen a good drink of some port and relax
00:15.21coppicehum, breakfast
00:15.45IOscannerbreakfast for dinner....mmmmm
00:16.51coppicedinner at 8:15AM...mmmmm
00:16.56JTit would be nice for incoming calls to do a greypages match for CDRs :)
00:17.12*** join/#asterisk XChris (n=JChris@cssgate1.wccnet.org)
00:17.26boorayso I have a question for anyone who's not eating at the moment...  in the oreilly PDF it says to compile asterisk as i586 on VIA chipset mobos as opposed to i686.  I haven't found anything else out there supporting this.. is it still true with latest versions of things?
00:17.52coppicethat is specifically for the VIA C3 processor
00:18.03IOscannerommmmmmmmmommmm
00:18.23IOscannerI think they have a via C3 processor option in the Make file don't they?
00:18.43IOscannerI have built on C3 and yes you have to modify the Make file to get it to build
00:19.00booraygotcha.  thanks guys.. in this case just a via chipset and not a c3
00:19.56IOscannerI think so I don't remember.  If it will not bild with Via it should build with i586
00:20.18IOscannerjust make sure you comment out the intel or MMX stuff
00:20.37IOscannerthat  VIA chipset doesn't like that
00:20.46boorayEven on a P4 processor?
00:20.51coppiceyes it does. MMX is fine wth the VIA chips
00:21.19IOscannerI made a cluster of Via C3 $99 boxes a few years ago and they are still running
00:21.51booraywhat, running distributed.net?
00:22.00IOscannerwell it didn't last year
00:23.00IOscannerNo VRRP and linux VirtualFS
00:26.20Nuggetyay distributed.net  :)
00:26.29boorayha, nugget lurking
00:28.19DrukenLPYJT: greypages?
00:28.36JTreverse phonebook lookup
00:28.47DrukenLPYoh...
00:28.58DrukenLPYi'd love to get the database for that :)
00:29.07JTi have a database for australia
00:29.24DrukenLPYnice...
00:30.00DrukenLPYprolly only one telco in aussie right?
00:30.04JTonly just listed numbers
00:30.11JTthere's lots of telcos, one major one
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00:30.55DrukenLPYi'd like to find one for canada... however, i think it would be a couple gigs of data :)
00:31.39JTprobably, depending on how efficient the format is
00:32.16DrukenLPYwhat is the database you have in ?
00:32.35JTcsv
00:32.54DrukenLPYyou haven't imported it into like postgres of mysql ?
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00:33.14JTnup, i could if i could be bothered to :P
00:33.22DrukenLPYhehehe
00:33.26DrukenLPYhow big if the file?
00:33.31DrukenLPYs/if/is
00:33.42JTlet me log into the machine and check
00:34.35JTabout 600MB including both the business and residential file
00:34.47DrukenLPYthat's not bad at all...
00:35.15DrukenLPYwelp, i'm out gotta go pickup the wife
00:35.36JTi just grep the file at the moment
00:35.51JTi could event access it as sql with perl's DBD::CSV
00:35.57JTs/event/even/
00:36.05JTno worries
00:39.25*** join/#asterisk LeddyHM (n=NONE@polar.artica.net)
00:41.51LeddyHMOur Asterisk vendor decided to go awol, so I'm trying to pick up the pieces and learn asterisk as we need to add a few extensions. From what I can tell I just need to edit sip.conf, extensions.conf, and voicemail.conf
00:41.55LeddyHMdoes that sound about right?
00:42.52boorayhooray for awol vendors...  I'm in a similar boat, except I'm the new vendor learning everything from scratch.  I could probably tell you in a few hours after I've finished mastering it.  :-/
00:43.16LeddyHMThat would be us as well
00:43.44LeddyHMunfortunately they used gentoo, which I'm not familiar with. Just another wrench to get past
00:44.43Carp1can someone help me setup SellVoIP in my iax,conf?
00:47.11boorayLeddyHM: I think help is out to lunch...
00:47.30LeddyHMprolly awol ;)
00:48.13*** join/#asterisk angler (i=angler@pdpc/sponsor/digium/angler)
00:48.41LeddyHMI'm also toying with the idea of starting over, on a platform we are more familiar with.... but baby steps
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00:49.18boorayLeddyHM: that's what I'm doing
00:50.23LeddyHMthe nice thing at least is you have all your configurations
00:51.02booraytrue, but they're requesting that the old ones only be used for reference.. and at this point I agree
00:51.16boorayand if I can set up this system from scratch, it lets me offer that as a real service in the future
00:52.16LeddyHMtrue
00:52.18wulfy814I'm having a very strange issue, inability to dial internal three digit extensions on Polycom 430's and 601's
00:52.29LeddyHMwe're ust trying to keep ours working w/o downtime
00:52.42wulfy814I see nothing if I punch 110 and pickup the handset and look at the CLI
00:52.52booraynew box, two test phones, weekend install
00:52.54wulfy814if I dial 8500 I see it going to VM
00:53.03perdwulfy814 turn on debugging and paste it to pastebin.ca
00:53.03wulfy814incoming PSTN calls are received fine
00:53.11perdwith your extensions.conf
00:53.13perdand sip.conf
00:53.19wulfy814perd sip debugging?
00:53.20Carp1can someone help me setup SellVoIP in my iax,conf?
00:53.25perdyeah
00:53.34wulfy814ok
00:54.27*** join/#asterisk denon (i=denon@synapse.subneural.net)
00:54.27*** mode/#asterisk [+o denon] by ChanServ
00:55.00Carp1i signed up with SellVoIP but I dont know what server to connect to!! :(
00:55.49*** join/#asterisk orlock (i=jwr@202.44.174.4.static.nexnet.net.au)
00:56.06orlockDoes anybody have any advice/tools for diagnosing poor voice quality using SIP/RTP?
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01:01.25wulfy814perd I think I got it, I wasn't including my phones context in my out context for the phones
01:01.30wulfy814I imagine that would do it
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01:09.40demigod2khi guys
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01:13.45Carp1can someone help me setup SellVoIP in my iax,conf?
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01:22.14joe[TK]D-Fender: ping
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01:27.06Carp1does anyone have an iax.conf example for SellVoIP?
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01:33.19Carp1some of a b
01:33.25Carp1i am getting the same error again
01:33.35Carp1rejected connect attempt
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01:40.33Carp1this time its no auth found though
01:40.38Carp1can someone help me?
01:41.53J4k3is there any sort of website where voip providers register their rates to specific countries?
01:42.05J4k3if not, I'm gonna quit doing what I'm doing and register a domain
01:43.24J4k3I hit a voip forum site and you've got people claiming to do very-much-below 1c/minute termination to the US and canada, lots of them... so I know there are cheap termination minutes to be had ;)
01:43.48[TK]D-FenderJ4k3 : I've never heard of a site that gathers pricing info like that....
01:43.59[TK]D-Fenderjoe : pong
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01:44.31orlockbloody verizon
01:44.56J4k3[TK]D-Fender: damn...  that has to change.
01:45.14J4k3I mean geez... theres pricewatch, froogle, etc..
01:46.53Carp1If you are seeing 'No Authority Found' in IAX debug trying to receive inbound calls. Make sure that for your inbound settings on our trunk the context = 'youraccountnumber'.
01:46.55joe[TK]D-Fender: email sent to you...
01:46.58Carp1Does anyone understand this?
01:48.02[TK]D-Fenderjoe : jsut read
01:50.46[TK]D-FenderJ4k3 : ITSP's don't format themselves in a way that webspiders like those can grab stuff easily if at all
01:51.10JTwe have voip rate comparison sites in australia
01:51.17JTbut they are updated manually by people i believe
01:52.25[TK]D-FenderJT : Oh.. you mean masochists ;)
01:52.25Carp1Can someone please help me with my no auth found error?
01:52.25*** join/#asterisk Lbase (n=Bub@fl-67-76-7-198.dyn.embarqhsd.net)
01:52.38[TK]D-FenderCarp1 : pastebin your IAX.conf
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01:53.07Strom_Cyo
01:53.11Carp1sure
01:53.13Carp1one minute please.
01:53.38JT[TK]D-Fender: it's not that hard really
01:54.17[TK]D-FenderStrom_C : y0
01:55.16Carp1shit
01:55.22Carp1can you delete a pastebin?
01:55.32orlockhah
01:55.35orlockwhoops :)
01:55.35k-manis there any doco about using asterisk and sip from behind a NAT?
01:55.42JTonly if you put an expiry on it
01:55.46Carp1damn
01:55.48Carp1i didnt :(
01:55.49JTit will delete itself then
01:55.53Carp1i have to change my pass
01:56.03Carp1http://pastebin.ca/336485
01:56.08JTis there any reason you insist on iax, Carp1 ?
01:56.40Carp1not really.
01:57.03Carp1I am already using SIP on my VoIP home phone service
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01:57.49[TK]D-FenderCarp1 : Ok, that context you wrote has NO authentication infor in there.....
01:58.30Carp1What do you mean?
01:58.37Carp1I need to put the host= in there also?
01:59.59davieyWhere in the conf files would a trunk dial plan go??
02:00.26SplasPoodhey anyone ever write a script to parse through sip debug output?   Cause if not I'm about to...  need to sort out all the stuff that didn't involve a specific IP..
02:01.02*** join/#asterisk Defraz (n=t0tal@24-116-152-177.cpe.cableone.net)
02:02.28JTdaviey: what is a trunk dialplan?
02:02.55*** part/#asterisk squish102 (n=squish10@cpe-024-074-100-250.carolina.res.rr.com)
02:03.07[TK]D-FenderCarp1 : your userid at sellvoip ISN'T sellvoip....
02:03.23Carp1Oh, I get it now
02:03.23davieyadding or removing digits to fit, so you dial a local number and asterisk adds the international format to it
02:03.30Carp1I didnt know it had to be
02:03.35davieyi think it is in extensions.conf
02:03.44Carp1let me try
02:04.03[TK]D-FenderCarp1 : that "friend" you set up is your identity and needs the auth credentials.  did you ask sellvoip for a sample config?
02:04.08JTyeah extensions.conf is the dialplan
02:04.57Carp1yeah, earlier
02:05.02Carp1they never emailed me back
02:05.24Carp1and U cant find an exampe online anywhere
02:05.51JTmost providers support sip better, unless they promote iax heavily
02:06.11Carp1I will try SIP
02:06.27Carp1I've never connected to a provider using SIP before
02:06.30Carp1only local phones.
02:06.38k-manhow does one  turn off sip debugging?
02:06.55The_DoC^thats odd, I can't get asterisk to detect the x100p, I installed
02:07.07JTsip no debug
02:07.11k-manah
02:07.12k-mantahns
02:07.53Carp1I found an sip.conf example
02:07.55Carp1let me try it
02:13.46Carp1ok
02:14.45Carp1now I get a "failed to authenticate user "CELL PHONE   NY" <sip:+1xxxxxxxxxx@72.5.55.200>;tag=as161deb1c
02:15.48Carp1that us a notice
02:15.55Carp1right above that is a warning
02:16.03Carp1username mismatch
02:16.31Carp1after that is says digest has <s>
02:18.05JTyou've had no luck at all so far with connecting to an itsp have you?
02:18.14Carp1nope
02:18.17Carp13rd provider.
02:18.26Carp1NuFone, Teliax, now SellVoIP
02:18.47JTlike
02:18.59JTdo you want to pay someone to fix it? :P
02:19.16Carp1if I wasnt poor,  most definately
02:19.47JTheh
02:19.51Carp1:-\
02:23.51Carp1i might be able to throw $5 out....i know its not alot lol.....but what can I say
02:25.15k-mani still can't make a sip call
02:25.21k-mani can't work out why....
02:25.56The_DoC^ugg, why do I have such bad luck. ordered what I thought was a original x100p and it turns out its a clone that down't work
02:26.14The_DoC^doesn't even
02:27.27HushPeThe_DoC^: i had a clone, waste of time and $$$, i got a digium card to play with, never looked back :)
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02:28.26The_DoC^what I really want is a standalone fxo
02:28.58The_DoC^that way I can keep my asterisk box a 1u
02:31.21[TK]D-FenderThe_DoC^ : Your 1U has no slots?
02:32.13k-manany idea why i would get this error when i try and make a sip call? <--- SIP read from 203.2.134.1:5060 --->
02:32.13k-manSIP/2.0 408 Request Timeout
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02:33.39The_DoC^[TK]D-Fender: network card is in the open slot
02:34.19JTCarp1: i'd rather do it for free than take $5 :)
02:34.40Carp1$10?
02:34.53Carp1thats as high as I can go (i hate to say that lol)
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02:39.55k-mando i need a sip_nat.conf file?
02:40.07Carp1are you using freepbx?
02:40.12k-manno
02:40.19JTCarp1: i can give it a go, a little busy atm
02:40.27Carp1Thanks alot.
02:40.49The_DoC^can you use isdn hardware on a standard pstn line?
02:40.50Carp1Shall I PM you the login details, and when you have time, you can do it.
02:41.11JTThe_DoC^: what, like an isdn card on an analogue line?
02:41.13JTCarp1: ok
02:41.34The_DoC^yes JT
02:42.17orlock...
02:43.52JTThe_DoC^: no.
02:44.45orlockgrr
02:46.17JTCarp1: do you have details of your provider?
02:47.54The_DoC^oh how I hate that I am a cheap bastard
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02:50.07[TK]D-Fenderk-man : that 408 is sounding like a NAT issue....
02:50.16k-manhmm
02:50.23k-man[TK]D-Fender, i did the port forwarding
02:50.34[TK]D-Fenderk-man : pastebi your [general] section of sip.conf again
02:50.35k-manso that part shold be working
02:51.01[TK]D-Fenderk-man : I gave you a PILE of settings to adjust the other day... not just port forwarding....
02:51.10k-manyeah, i did those too
02:51.19k-manat least i think i did all the ones you gave me
02:52.20k-manhttp://pastebin.ca/336562
02:54.25k-man[TK]D-Fender, the error come up very quickly, could it be some sort of authentication issue or the port asterisk connects to is refusing connection?
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02:56.18[TK]D-Fenderk-man : http://pastebin.ca/336566
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03:00.10k-man[TK]D-Fender, why does the jason one need nat=yes? that is for my internal phone no?
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03:03.18Carp1DOes anyone have a SIPURA?
03:03.25k-mansipura what?
03:04.12Carp1hardware
03:04.45k-mani have a linksys phone... which appears to also be a sipura
03:04.53k-mani'm not sure of the link between sipura and linksys
03:05.09k-man[TK]D-Fender, i get the same error still
03:05.25[TK]D-Fenderk-man : * will look at the IP the phone is using and apply the externip automatically as needed.
03:05.42[TK]D-Fenderk-man : this helps if you want to move your phone to a friends place for instance for a demo
03:05.50k-manoh, i see
03:05.55JTlinksys bough sipura
03:06.05k-manjt, oh... interesting
03:06.22JTcisco own linksys
03:06.38J4k3and cisco smells funny
03:06.44k-mancisco have too much money
03:06.50The_DoC^I am using 2 pap2's and a rt31p2
03:07.00Carp1lol
03:09.22NivexI've got an SPA-2000
03:10.16Carp1i have a 2100
03:11.22The_DoC^I wan't a spa-3000 or 3102
03:11.51Carp1i have a q tho
03:11.53Carp1i cant edit any information on it
03:11.55Carp1it just shows it.
03:12.41The_DoC^do you have a admin login?
03:13.25Carp1I dont know
03:14.20Carp1i own the hardware
03:14.36Carp1but I bought it through the company that i get my phone from
03:14.47Carp1and I just called and said I want information to it because I own it
03:15.01Carp1he said the username was user, and the pass was the 11 digit phone number.
03:15.47The_DoC^try http://ip.of.2100/admin/
03:16.21Carp1404 Not Found !
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03:31.54Carp1why does my SIPURA not have an admin login link? lol
03:32.48The_DoC^is it a locked system like the PAP2's?
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03:33.55CrashSysWould Dial(Console/DSP,,a(beep)) play the beep sound to both parties or just one side?
03:33.57[TK]D-FenderCarp1 : Have you tried the DTMF "reset to factory defaults" code?
03:39.23The_DoC^**** 73738# is factory reset
03:42.20CrashSysDoes chan_oss work properly with the ALSA API for OSS?
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03:50.22k-man[TK]D-Fender, i'm getting this error now: *CLI>
03:50.23k-man*CLI>
03:50.23k-man<--- SIP read from 10.0.2.201:5060 --->
03:50.23k-manINVITE sip:2@10.0.2.231 SIP/2.0
03:50.27k-manVia: SIP/2.0/UDP 10.0.2.201:5060;branch=z9hG4bK-60a26c6d
03:50.29k-manFrom: "Jason" <sip:jason@10.0.2.231>;tag=c034c1eb6df1dc75o0
03:50.31k-manTo: <sip:2@10.0.2.231>
03:50.33k-manCall-ID: d2c38fa3-d8beaa1d@10.0.2.201
03:50.35k-manCSeq: 101 INVITE
03:50.37k-manMax-Forwards: 70
03:50.39k-manContact: "Jason" <sip:jason@10.0.2.201:5060>
03:50.41k-manExpires: 240
03:50.43k-manUser-Agent: Linksys/SPA942-5.1.5
03:50.45k-manContent-Length: 395
03:50.47k-manAllow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
03:50.49k-manSupported: replaces
03:50.51k-manContent-Type: application/sdp
03:50.53k-manv=0
03:50.57k-mano=- 16644837 16644837 IN IP4 10.0.2.201
03:50.59k-mans=-
03:51.01k-manc=IN IP4 10.0.2.201
03:51.03k-mant=0 0
03:51.05k-manm=audio 16462 RTP/AVP 18 0 2 4 8 96 97 98 101
03:51.06JTwtf
03:51.07k-mana=rtpmap:18 G729a/8000
03:51.07JTSTOP
03:51.09k-mana=rtpmap:0 PCMU/8000
03:51.11k-mana=rtpmap:2 G726-32/8000
03:51.13k-mana=rtpmap:4 G723/8000
03:51.13CrashSys~pb
03:51.15jbothmm... pb is a place to paste your stuff without flooding the channel - try  http://pastebin.ca, or http://channels.debian.net/paste
03:51.15k-mana=rtpmap:8 PCMA/8000
03:51.17k-mana=rtpmap:96 G726-40/8000
03:51.17JTk-man: bad boy
03:51.19k-mana=rtpmap:97 G726-24/8000
03:51.21k-mana=rtpmap:98 G726-16/8000
03:51.22CrashSysdie
03:51.23k-mana=rtpmap:101 telephone-event/8000
03:51.23CrashSysDIE
03:51.27k-mana=fmtp:101 0-15
03:51.27JTargh
03:51.29k-mana=ptime:30
03:51.31k-mana=sendrecv
03:51.33k-man<------------->
03:51.35k-man--- (14 headers 18 lines) ---
03:51.37k-manSending to 10.0.2.201 : 5060 (NAT)
03:51.39k-manUsing INVITE request as basis request - d2c38fa3-d8beaa1d@10.0.2.201
03:51.41k-man<--- Reliably Transmitting (NAT) to 10.0.2.201:5060 --->
03:51.43k-manSIP/2.0 407 Proxy Authentication Required
03:51.45k-manVia: SIP/2.0/UDP 10.0.2.201:5060;branch=z9hG4bK-60a26c6d;received=10.0.2.201
03:51.47k-manFrom: "Jason" <sip:jason@10.0.2.231>;tag=c034c1eb6df1dc75o0
03:51.49k-manTo: <sip:2@10.0.2.231>;tag=as47f8849f
03:51.51k-manCall-ID: d2c38fa3-d8beaa1d@10.0.2.201
03:51.53k-manCSeq: 101 INVITE
03:51.57k-manUser-Agent: Asterisk PBX
03:52.16k-manAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
03:52.16k-manSupported: replaces
03:52.17k-manProxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="72d0c555"
03:52.17k-manContent-Length: 0
03:52.17CrashSysWhy is it an e-mail?
03:52.17k-man<------------>
03:52.17k-manScheduling destruction of SIP dialog 'd2c38fa3-d8beaa1d@10.0.2.201' in 32000 ms (Method: INVITE)
03:52.17k-manFound user 'jason'
03:52.17JTk-man: are you on drugs?
03:52.17k-man<--- SIP read from 10.0.2.201:5060 --->
03:52.17k-manACK sip:2@10.0.2.231 SIP/2.0
03:52.17CrashSyssomeone kill me
03:52.18k-manVia: SIP/2.0/UDP 10.0.2.201:5060;branch=z9hG4bK-60a26c6d
03:52.18JTCrashSys: what is an email?
03:52.19k-manFrom: "Jason" <sip:jason@10.0.2.231>;tag=c034c1eb6df1dc75o0
03:52.21k-manTo: <sip:2@10.0.2.231>;tag=as47f8849f
03:52.23k-manCall-ID: d2c38fa3-d8beaa1d@10.0.2.201
03:52.24CrashSysjt: EXACTLY!
03:52.27k-manCSeq: 101 ACK
03:52.29k-mancrap
03:52.29JTCrashSys: ??
03:52.31k-mansorry guys
03:52.33k-mansorry
03:52.35k-manyes
03:52.37k-manit was my fault
03:52.39k-manit was an accidtental paste
03:52.41k-mani selected 1 line
03:52.43k-manand forgot to copy it
03:52.45k-manbefore pasting
03:52.46CrashSysUse pastebin
03:52.47k-mansorry sorry sorry
03:52.51k-manno
03:52.53k-manjt, no...
03:52.57k-manJT, maybe i should be
03:52.58*** join/#asterisk topping (n=topping@c-69-181-217-16.hsd1.ca.comcast.net)
03:53.03JTCrashSys: what do you mean email?
03:53.38CrashSysJT: That looked like e-mail headers
03:53.53JTCrashSys: looks like you've never looked at sip debug before
03:53.56JTthat's sip
03:53.59CrashSysOhh
03:54.00CrashSysnope
03:54.17CrashSysI dont program the stuff, I just script it all and configure it...
03:54.32*** join/#asterisk drako (n=ljd@unaffiliated/luisjose)
03:54.34JTusually configuration involves debugging :P
03:54.59CrashSysI must be lucky... I figure it out from the console with -vvvvv
03:55.07JTheh
03:55.15CrashSysthe errors/warnings it gives are usually enough for the small systems I do
03:55.22JTit doesn't give enough info if there's a weird protocol problem
03:55.27*** part/#asterisk topping (n=topping@c-69-181-217-16.hsd1.ca.comcast.net)
03:55.53CrashSysYeah, but i'm basically doing key-systems... ZAP to POTS, with SIP phones...
03:56.08CrashSysnothing really extravagant about it
03:56.34JTshrug, i've needed to use all the various channel debug modes for even the simplest setups
03:56.50k-manok, i'll try again
03:56.55JTi guess if your configuration always works, there may be not so much need
03:57.03CrashSysMaybe my set-up's are fubar'd and I just dont know it... but they work...
03:57.06k-manis this error significant? SIP/2.0 405 Method Not Allowed
03:57.19CrashSysSIP use the same codes at HTML?
03:57.29JTcloser to email than htm
03:57.32CrashSysahhhh
03:57.39JTerror codes i guess are similar to http
03:57.57CrashSys404 = Not There, 405 = denied, etc etc...
03:58.04JTyes
03:58.14*** join/#asterisk kavit (n=kavit@ppp244-74.static.internode.on.net)
03:58.27JTk-man: mayb, dunno
03:58.36CrashSyslearn something everyday in here :)
04:00.48CrashSysMan, that sucks, there's no way to play a file to the caller and callee witht he dial command upon connect :(
04:00.54CrashSysguess that has to do with the bridging
04:00.56CrashSyshmmm
04:01.20JTyeah i think that is the case
04:01.25The_DoC^I just need to give up on setting up a pbx and stick to wireless networking
04:01.36JTasterisk is pretty inflexible in what it can do once a calle is bridged
04:01.50JTThe_DoC^: hah
04:01.50CrashSysI guess if I play the beep before issuing the dial the lag it takes them to start talking (that whole second) will be OK
04:02.08CrashSysYeah, and I dont feel like using meetme/conference/etc to do it :D
04:02.11JTwhat's the beep?
04:02.17CrashSysbeep.gsm
04:02.21JTfor?
04:02.22CrashSysIt's for overhead paging
04:02.23*** join/#asterisk oej (n=olle@216.64.24.250)
04:02.26JTok
04:02.40CrashSysSo you dial an extension, you hear the beep on the phone and the overhead, to let you know to start talking...
04:03.10JTyeah not sure how you'd do that
04:03.14CrashSysCause now there's no beep and they dont know when to talk other then to wait 5 seconds (cause that's about how long it takes to build a call up through the A200 to the PSTN)
04:03.39CrashSyswell the beep is 1 second... I can playback(beep), then dial(console/dsp,,A(beep))...
04:04.00CrashSysthere is a potential 1-second dead-spot after their beep plays that they can talk and nothing comes out...
04:04.03CrashSysbut i'll risk it
04:04.20Strom_CCrashSys: are you trying to get the overhead pager to beep before the person talks?
04:04.25CrashSysThey're not rappers, they cant talk that much in 1 second...
04:04.31CrashSysStrom: Yeah, but also the handset
04:04.43CrashSysCause the office isn't in the warehouse, and hence, cant hear the beep
04:05.19JTCrashSys: it's a hack, but you could try using the L dial option
04:05.20Strom_Cdo this
04:05.44CrashSysSo I need to play the beep to both parties upon connect... but the closest I can get to that is playback(beep) to the caller, then play the beep to the console dial(console/dsp/answer,,A(beep))
04:05.55CrashSysthat's the best solution i've found so far...
04:06.14JTCrashSys: look at L
04:06.20JTyou can play a file on connect
04:06.38Carp1so....anyone know why there is no admin interface on my SIPURA?
04:07.45[TK]D-FenderCarp1 : have you done the full reset as suggested?
04:08.09Carp1I didnt see that message, sorry....I really cant becasue its my home phone number on port 1
04:08.18Carp1but I cant see where the information was stored in there
04:08.23Carp1you cant edit any values either
04:08.32Carp1just shows status basicallyt
04:08.34The_DoC^locked
04:08.41JTyou using http or telnet or serial?
04:08.41Nuggettelnet is eeeeeeevil!
04:08.45Carp1hmmm
04:08.54JTfriggen Nugget's stupid script
04:08.56JTtelnet
04:08.57[TK]D-FenderCarp1 : if you're locked out the answer is to flush the config.  If you don't like it, at least stop whining about it.
04:09.13The_DoC^**** 73738# is factory reset
04:09.16CrashSysJT: Hmmm... looks interesting... I guess I set a 5-minute limit and the limit_connect...
04:09.25Carp1I'm not whining :)
04:09.28Carp1I just didnt know
04:09.42JTCrashSys: not sure if it can play to both, now that i look at it :(
04:09.45Carp1I would reset it if I knew what values to put back in :)
04:10.16[TK]D-FenderCarp1 : If you don't know that they you're flying COMPLETELY blind.
04:10.28[TK]D-FenderCarp1 : Clueless and crippled on both fronts.
04:10.33k-mancan someone have a look at this and tell me why they think i am unable to connect to nodephone to make a call please? http://pastebin.ca/336635
04:10.42Carp1dont know that they what?
04:10.44Carp1lol
04:11.02JTCarp1 may have a provider that restricts the sip info that's on his locked device
04:11.12*** join/#asterisk Igbothom_III (n=Hilton@office.quarkit.com.au)
04:11.32The_DoC^I ride the short bus when it comes to this stuff, thats why I idle and read alot
04:11.34Carp1I see
04:11.56Carp1I wanted to get in just because I want to use port 2 for an asterisk extennsion
04:12.04The_DoC^vonage does the locked equipment
04:12.25JTk-man: sip.conf pls
04:12.30The_DoC^evil bastards
04:12.31danpi'd like to flag calls that came in via zap channels somehow so when they ring the receptionist and subsequently get transferred to someone else i can make them ring differently. would the best way to do that be setting an inhertied variable and checking it?
04:12.31[TK]D-Fenderk-man : please pastebin your [general] section again
04:12.38k-man[TK]D-Fender, ok
04:12.40danpsomething like __FROM_ZAP
04:12.41k-manhang on
04:12.50k-manits still giving that timeout error
04:12.57k-mani don't understand where its coming from
04:13.17[TK]D-Fenderk-man : You keep repeating that.  WE HEARD YOU ALREADY.
04:13.27k-man[TK]D-Fender, oh... sorry
04:14.18CrashSysSpecial Variables... that means I define them in the dial cmd or as a global variable in the extensions.conf?
04:14.42JTCarp1: answer is you probably can't
04:15.05Carp1Yeah, you're probably right
04:15.17Carp1I am going to see if I can get the information from them to reprogram is
04:15.19Carp1it*
04:16.18k-manhttp://pastebin.ca/336638
04:18.10*** join/#asterisk GreyFoxx (i=greg@216.83.31.88)
04:18.12[TK]D-Fenderk-man : Double check your IP and forwarding
04:18.18k-manok
04:18.56JTwhy are you forwarding/
04:19.09k-mani am behind NAT
04:19.10GreyFoxxCan anyone point me to some documention about storing my astdb data in a mysql database ? (If anyone's doing that)
04:19.13k-manbrb
04:19.52*** join/#asterisk litage (n=nick@203.220.55.70)
04:20.23[TK]D-FenderGreyFoxx : Not sure if that possible.  BDB isn't SQL compatable
04:20.54GreyFoxxI figured as much, I was hoping someone might have a patch or something to do it :)
04:21.15Qwellthere is on the tracker I think
04:21.39JTk-man: i have never needed to do port forwarding to connect an asterisk server behind NAT to an account on a SIP provider
04:21.39GreyFoxxOh? What's the URL to that? I'll go check it out
04:22.09Qwellbugs.digium.com
04:22.14GreyFoxxthanks
04:23.14[TK]D-FenderJT : Thats interesting, first I've heard of it without some other form of proxying going on.
04:23.32JTi just set nat=yes
04:24.25JTobviously forwarding is required to connect to an asterisk server behind nat, completely inititated from the outside world
04:24.58*** join/#asterisk litage (n=nick@203.220.55.70)
04:27.10[TK]D-FenderJT : I guess you like your isolation :)
04:28.25JThrm?
04:28.33JTwell if i'm registered, inbound calls work fine
04:29.12[TK]D-FenderJT : Funny those ports should close down behind you ....
04:29.32JThey, i'm sure exactly why it works, but it does :)
04:30.13JTdone this both on linux gateways and some adsl modems with inbuilt routers
04:31.15orlockgrrrrr, ok, voip quality is giving me the shits
04:31.17[TK]D-FenderCould be the gateway is excessively SIP aware...
04:31.28orlockRTP analysis with ethereal all looks fine
04:31.38J4k3orlock: are you testing with some bs softphone, or real gear?
04:31.38orlockproblem is with inbound voice to somebody on a DSL pipe
04:31.43*** join/#asterisk thoughtpolice (n=austin@ip70-185-140-61.lu.dl.cox.net)
04:31.45JTmy linux gateway doesn't has sip_conntrack, and is years old
04:31.46orlockJ4k3: real gear
04:32.09orlockJ4k3: Linksys phones usually
04:32.27orlockiaccording to the packet dumps, all looks fine
04:32.50J4k3orlock: ahh, you're not a noob obviously, so you know a bit more than me ;)
04:32.58JT[TK]D-Fender: i guess this is why sip providers spend 10s of thousands of dollars on expensive session border controllers that have all sorts of tricks for NAT busting
04:33.00orlocki cannot see any technical reason in the dumps for the problem they are having
04:33.02J4k3I just know... "don't trust softphones for performance testing AT ALL"
04:33.07*** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal)
04:33.07*** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal)
04:33.45orlockhmm
04:33.58orlocki am checking the data pre-asterisk as it comes in on the external interface
04:34.10orlockmaybe i should dump the packets on the internal
04:34.21k-manjt, oh... so its not a port forwarding issue then?
04:35.08*** join/#asterisk CrashHD (n=crashhd@c-67-182-170-132.hsd1.ca.comcast.net)
04:35.13JTk-man: it might be, i don't know
04:35.23JTk-man: i suggest you do packet dumping or something
04:35.31JTto see if you can see the rtp stream at all
04:35.36k-manpacket dumping
04:35.54orlocktcpdump and wireshark
04:36.00orlockget to know them
04:36.05orlockthey are your buddies
04:37.05k-manok
04:39.55*** join/#asterisk CunningPike (n=CunningP@S010600095b33697f.vc.shawcable.net)
04:42.00Strom_Chey CunningPike
04:42.08CunningPikeHi, Strom_C
04:42.10CunningPikeAll set?
04:42.21CrashSysAnyone got any ideas on a console-based alsa mixer that lets me see sound level being output? (trying to remotely see if the soundcard is making noise)
04:42.45Strom_CCunningPike: pretty much, just doing a few last things
04:42.46Qwellalsamixer
04:42.49Qwelloh, wait, no
04:42.53CrashSysheh
04:43.02CrashSyswait, that sets volume...
04:43.03Qwellbut it is a console-based alsa mixer ;)
04:43.12CunningPikeStrom_C: Great - what time does your flight get in?
04:44.16CrashSysQwell: Yeah... wonder if it will show me a VU meter...
04:45.18*** part/#asterisk GreyFoxx (i=greg@216.83.31.88)
04:46.29olsenalsamixer is nice and looks nice
04:46.41JTbut does it do what he needs?
04:46.43orlockgrrrr
04:46.52orlockinternal traffic looks the same as external as it leaves asterisk
04:47.49olsenJT: what does he needs?
04:48.04CrashSysI want a VU-meter type out display
04:48.12CrashSysto see if my chan_oss setup is making sound
04:48.19CrashSyssince i'm not using chan_alsa anymore
04:48.31JTyou had to scroll up a whole 5-15 lines for that, olsen
04:49.10CrashSysI know chan_alsa worked... but I have to use chan_oss now... and i'm using the ALSA Wrapper for OSS, and just want to make sure (short of driving) that output is being done on the sound card...
04:50.02orlockDoes anybody know what "acceptable" kitter would be?
04:50.03QwellCrashSys: setup a phone to autoansert
04:50.06Qwellautoanswer*
04:50.11Qwellthen make sound
04:50.33CrashSysNo phones in the warehouse that are hooked up, and cant hear it from the office :)
04:50.40CrashSysit's a predicament
04:51.25CrashSysMy two options right now are drive there now, stand outside the warehouse, call in with cellphone, hope it works, drive home (with a rinse-and-repeat if it dont)... Plan B is go there at 6:30am in the morning when they open...
04:51.29CrashSysboth options suck
04:51.57CrashSysI know ALSA works...
04:51.58olsenCrashSys: you are connected remotely to a computer and you need to make sure that it makes sound?
04:52.09JTthe otjher option is to pretend nothing is wrong and go to sleep
04:52.11CrashSysI know chan_alsa worked...
04:52.18CrashSysJT: The thought has crossed my mind :D
04:52.29CrashSysI know the physical connection from sound-card to PA works...
04:52.53CrashSysI just want to verify through console that chan_oss is connecting to the ALSA wrapper correctly and making output...
04:53.14CrashSyschan_alsa worked prior to going to chan_oss...
04:53.37CrashSysIn all likelyhood it works...
04:57.29JTheh
05:07.28olsencan i use ekiga with asterisk and call another computer that has ekiga on it?
05:07.35olsenor another softphone
05:08.18JTyes
05:08.32olsennice
05:13.23*** part/#asterisk JT (n=jon@unaffiliated/jt)
05:13.23*** join/#asterisk JT (n=jon@unaffiliated/jt)
05:15.28*** join/#asterisk topping (n=topping@204.152.97.99)
05:18.07orlockJT: jitter on inbound RTP data.. any suggestions?
05:18.26JTreplace datalink :)
05:20.49fetcheris there a decent web site for network outage reports?
05:21.26JTorlock: sounds like a thorn in your side
05:21.50fetchertrying to find out what's up with Level 3 in Atlanta tonight... looks like a big fiber cut
05:22.32CrashSysAnyone got any suggestions for a simple CDR Log Analyzer that tells me Call Volume, Peak Concurrent Call Volume, and Queue stats?
05:22.47CunningPikefetcher: http://scoreboard.keynote.com/scoreboard/Main.aspx?Login=Y&Username=public&Password=public
05:22.51[TK]D-FenderCrashSys : those are 2 seperate systems.
05:22.59CunningPikefetcher: Try ##level3 as well
05:23.30CrashSysd-fender: well is there one log analyzer package that can look at both?
05:24.19CunningPikeCrashSys: Try asterisk-guru's queue_stats
05:24.25CrashSysok
05:24.26fetcherCunningPike: thanks.
05:24.44CunningPikeCrashSys: There are no packages I know of that do CDR and queue metrics in one - they are two separate logs
05:25.02*** join/#asterisk [hC] (n=hardcore@66.119.167.162)
05:25.41CunningPikefetcher: Hmm - not ##level3......
05:25.42CrashSysok... so Asterisk-Guru's Queue Statistics... any suggestiong on a general CDR analyzer?
05:26.49CunningPikeCrashSys: We rolled our own - import master-csv into MSSQL and then designed some PivotTables in Excel
05:27.53CrashSysHmmm... guess I could do that...
05:28.11CrashSysqueue statistics wont work with MySQL?
05:28.19CrashSysHeh...
05:28.51CunningPikeWhere'd you get that?
05:28.58CrashSyshttp://www.asteriskguru.com/tools/queue_stats.php
05:29.03CrashSyssays it needs postgre
05:29.15CunningPikeOh - maybe it does - I forgret
05:29.25CunningPikeIt's just plug and play - real easy to set up and sue
05:29.28CunningPikeuse, even
05:29.32CrashSysi'll throw it at mysql and hope it doesn't just any special commands
05:29.39CrashSysjust = use
05:29.59CunningPikeNothing wrong with postgres ;)
05:30.07[hC]Anyone have any wifi phone recommendations, hopefully something with decent battery life?
05:30.13CrashSysExcept i'm using MySQL already for zoneminder...
05:30.21[hC]Ive been using the dlink deh-540 clamshell, which is very similar to a linksys, but the battery life isnt so hot.
05:30.47CunningPike[hC]: Some of the Zultys ones are good - not pretty, but work well
05:31.59[hC]Zultys hmm... I'll go check that out.
05:33.04[hC]CunningPike: have any experience with these things? the battery life claims are fantastic
05:33.06*** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal)
05:33.07*** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal)
05:33.22CunningPikeWe have one in our print shop
05:33.39[hC]CunningPike: Price?
05:34.04CunningPike[hC]: Can't recall - not expensive. Less than $200 anyway
05:34.20CunningPike[hC]: I can find out in work tomorrow
05:34.27[hC]CunningPike: Im blown away. Where do i go to get one?  Ive been paying $400+ for dlink/linksys pieces of crap that barely work
05:34.56CunningPike[hC]: Can't remember where we got it - ping me tomorrow and I'll look it ujp
05:35.01CunningPikes/ujp/up/
05:35.06[hC]CunningPike: fantastic. Thanks.
05:35.41CunningPike[hC]: It's ugly - reminiscent of the old 800MHz cordless phones - but it's rugged and it works great
05:35.41[hC]CunningPike: Ive never heard of these guys before..  So you dont have too many issues with this phone?
05:35.50[hC]CunningPike: Thats exactly what I need. Rugged and reliable.
05:36.11[hC]CunningPike: Im not looking for fashion accessories here.  I need like... Service Center phones, consruction worker phones, etc.
05:36.19CunningPike[hC]: This fits the bill then - we have one of the rinky little ones too, but it would never work in a print shop
05:37.02[hC]CunningPike: yeah, im definitely after something with a charging station, good battery time, and reliability and ruggedness.
05:37.13CunningPike[hC]: This should work for you then
05:37.40[hC]CunningPike: I shall order one tomorrow then
05:37.45CunningPike:)
05:38.27[TK]D-FenderWiFi phones = suck. AAL of them.  Some only just a little less than others.
05:38.37[hC][TK]D-Fender: I agree, but this one looks promising.
05:39.28CrashSysHmm... Queue Statistics parses the CSV's eh...
05:40.23CrashSyslooks like it does CDR's too for server call volumes
05:40.28CunningPikeIt's Nugget with his telnet
05:41.00CunningPike:)
05:43.07[TK]D-Fender[hC] : thats the problem with promises.... they usually tun out to be lies ;)
05:43.56[TK]D-FenderCunningPike : I don't do scripts... I manually spew my bile on the unworthy :)
05:44.03[hC][TK]D-Fender: touche. Pikey here says its ok though.
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05:44.21CunningPike[TK]D-Fender: :)
05:46.15[TK]D-Fender[hC] : The world needs guinea pigs, so at least you will find a place :)
05:46.39[hC][TK]D-Fender: Hahaha.
05:46.48[hC][TK]D-Fender: Ive gone thru 4 models already, trust me, i feel your pain.
05:48.07[TK]D-Fender[hC] : I've only had *1* personally, and have handed it to a person who can't tell the difference.
05:48.16[TK]D-FenderI feel little pain from them.
05:48.48*** join/#asterisk xezz (n=asdas@83.235.189.78)
05:48.53[hC][TK]D-Fender: which one did you have to suffer with?
05:49.22[TK]D-FenderI HAVE however been plagued by the Uniden ELT-560 analog flip-phone.  utter GARBE.  they die in so many ways.  I've replaced 2 of them like 5 times in a year
05:49.22[TK]D-FenderGARBAGE
05:49.30JTu..u....uniden!
05:49.40JTsynonymous with rubbish, most of the time
05:49.54xezzhello , is it possible to change cid when calling outbounds ? instead of the standard id come from telco
05:50.19JTyes if over pri or sip to a pri, and telco supports it
05:50.40xezzjt its over pri yes
05:50.50xezzhow can i change the cid ?
05:50.56JTSet(CALLERID(num)=123)
05:51.02xezzi mean...telco isnt responsibble for that ?
05:51.30xezzyes but..when call goes through telco , the standard cid will appear
05:51.52xezzthis command will work only for local calls inside my nerwork
05:51.59JTwell they may not allow it
05:52.03JTno, it will send it out
05:52.12xezzit will send it out
05:52.12JTdo a pri intense debug to prove it for yourself
05:52.30xezzbut the standard id from telco will override
05:52.56JTyes, a lot of telcos won't allow you to set the callerid unless it's a number you own and in some way attached to the connection
05:53.07JTto prevent fraud
05:53.54xezzyeap
05:54.40xezzbut from technical side , i can set the cid number of an extension to 123 i.e right ?
05:55.13xezzallright
05:55.13JTyes
05:56.11xezzhave you setup asterisk on ubuntu machine ?
05:56.17JTno
05:56.36xezzon unix in general
05:56.45JTon debian, yes
05:57.07xezzits kinda same then
05:57.24JTi guess so, it's all linux
06:00.35Strom_Cit's pretty simple on ubuntu
06:00.43J4k3I'd imagine so
06:00.45J4k3its ubunut
06:00.48J4k3er ubuntu
06:01.03J4k3now, for me, it might be hard...considering I've spent the last 4 days trying to get gentoo to install and actually work on boot
06:01.24J4k3been running FreeBSD for >10 years... and I can't get gentoo to work :P
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06:08.26zeeeshhi
06:10.49fetcherJ4k3: where in the boot process does it fail?
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06:16.13xezzanyone got zaphfc module working under unix ?
06:17.04J4k3fetcher: grub fails to start...  I'm installing kubuntu now to see if it has the same problem
06:17.09J4k3it might be a hardware/drive issue
06:19.51J4k3oooh maybe I'm not retarded after all... kubuntu's doing the same.
06:20.08J4k3asrock... add an s, drop the rock... thats what this motherboard is.
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06:22.15xezzi can load zaphfc module now but in ztcfg -vvv output i get 0 channels configured , zttool shows the card but with note UNCONFIGURED
06:24.14xezzany idea?
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06:26.44[TK]D-Fenderxezz : perhpas you should pastebin your zaptel.conf....
06:26.55[TK]D-Fenderperhaps*
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06:31.34xezzthere it is : http://pastebin.ca/336730
06:32.45JTxezz: so there's like nothing configured, is it that surprising it shows as "unconfigured"?
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06:33.12[TK]D-Fender*duh*
06:33.24xezzthat bad ?
06:33.30xezz:(
06:33.49[TK]D-Fenderxezz : Its teling you to your face that you didn't even define a channel in zaptel.conf.  Wht did you THINK was oging to happen?
06:35.34xezzwell , how to define a channel ?
06:35.41[TK]D-Fender~book
06:35.42jbotmethinks book is a book called  Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11
06:36.35xezzi dont see something about zaphfc , in zaptel.conf
06:37.07xezzthanx for book
06:38.18JTxezz: read the documentation in bristuff directory
06:38.22JTtells you how to do it
06:38.32JTmight be in the zaphfc subdirectory
06:39.05xezzgoint to read it right now , thanks for helping
06:39.31[TK]D-Fender~wikis
06:39.32jbothmm... wikis is http://www.voip-info.org
06:43.36xezzi managed to load zaphfc module like make here does : http://www.voip-info.org/wiki/view/Asterisk+BRI-stuffed+Ubuntu+Edgy+Junghanns
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06:50.05xezz<Strom_C> frustration often blinds you :)
06:50.16xezzwise man
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06:50.26xezzthanx all for helping
06:51.22JTdid you get it working?
06:52.51xezzyeah
06:53.21xezzas [TK]D-Fender , i've didn't defined a channel
06:53.36JTyes
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07:19.01ThoMegood morning from germany
07:19.02ThoMe.-)
07:20.17zeeeshdialing through xpro ... at start .. xpro sends fake ring ... but .. after some time it shows ... call failed ... 503.. service unavailable ... ???
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07:32.37zeeeshkisssi ghasti maaa dioooo ... jawab kyon nahi daay rahay
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07:42.56creativxwb
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08:11.32x86anyone know where i can get Phillipines?
08:11.45x86DIDx and DIDWW are both out :(
08:12.30hadsGet Phillipines? I think it's taken.
08:12.41x86err
08:13.07x86i mean get Phillipine DIDs
08:13.17hads:)
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08:20.24fetcherIs there a hard limit on message length with Asterisk voicemail?  (app_voicemail.so)?
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08:35.32niZonanyone have problems with asterisk's native mp3 decoder sounding terrible?
08:35.38niZonas in blips every few seconds
08:35.40niZon?
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08:37.29creativxit might not like the mp3
08:37.33creativxtried with another one
08:39.26niZoni have about 30 in my moh directory, they all do it
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08:41.42clive-do you guys recommend using a 64 bit kernel for asterisk?..... does it perform better than 32 bit ?
08:41.51niZonmpg123 0.59r sounds nice
08:42.24niZoni haven't had much luck with 64 bit
08:42.45clive-nizon what hapenned when you tried 64 bit ?
08:42.56niZoncouldn't get it to compile
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08:43.35clive-I have seen it done, just not sure what they did to compile things
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08:45.24niZonI just went back to 32 bit, it was working fine
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08:50.36x25shi
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08:57.06JTniZon: what are the encoding specifications on the mp3s?
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09:00.36yassinemorning *
09:01.52creativxtop of the morning to you
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09:05.43kutoany user of vicidial? im building a support channel #vicidial , please do join me. thanks
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09:09.40creativxvici suicidal
09:10.35Mw3is there anyone who are using pri trunk lines?
09:10.43endre:)
09:10.58endrei bet you have privacy problems
09:11.13endrehave you tried to call t-com already?
09:11.33Mw3nope, the telco does not support asterisk
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09:13.20niZonJT: various, some are VBR, some are flat 128kbps, most are 44khz sample rate
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09:16.38x25sre
09:17.39x25sin asterisk CLI i can show this: !! Unknown IE 124 (cs5, Unknown Information Element)
09:18.01x25swhat is this message?
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09:20.25creativxinternet explorer in asterisk needs upgrading.
09:20.53Tebilol
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09:23.58JTniZon: argh, do NOT use vbr
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09:24.12JTthere's a doc on the wiki about what the specs of the files should be
09:24.35creativxpreferably just play the caller a 90 hz sine wave
09:24.44creativxensures quick hangup time
09:24.59JTMw3: what is wrong with your pri?
09:25.37Mw3JT: i'd like to hide my caller id for certain calls
09:25.57Mw3JT: but Set(CALLERID(all)="") or SetCallerPres(prohib) does not work
09:26.18x25si have one PRI E1 with TE412P and asterisk cli shows: !! Unknown IE 124 (cs5, Unknown
09:26.19x25s<PROTECTED>
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09:26.48JTx25s: does the link work?
09:26.56x25syes
09:27.05JTx25s: then don't worry about it
09:27.29x25swhy this message?
09:27.42JTit means your telco is sending an Information Element over the D channel that asterisk does not recognise
09:28.17JTMw3: ask if your telco supports it
09:29.41x25swhat information send me  my telco?
09:29.51x25sexcuse for my english
09:31.08JTi don't bloody know, but the point is it doesn't matter if it works
09:31.08JTsearch google for isdn information element
09:31.08JTor q.931 information element
09:31.08JTif you want to know what the term in general means
09:31.09JTyou can also download the q.931 standard from the ITU if you want to see exactly where it comes into it
09:31.15JTand you could look at the libpri source to see what has to happen for that error to be generated
09:31.20JTbut otherwise
09:31.22JTdon't worry about it
09:32.39x25sthanks for your help
09:32.40x25s:)
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09:33.28JTMw3: check the telco supports it
09:33.40JTMw3: then check asterisk is sending it with pri intense debug
09:33.45JTor the other way around
09:33.56JTalso, obviously Set before you Dial
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09:34.42nextimeis pseudo-realtime option usable only if * is launched to run as root user?
09:34.43endreJT: telco doesn't care about the q931 settings
09:35.31JTendre: what?
09:36.12endremw3 just told me
09:36.44endret-com overrides the options in q931
09:39.43JTwell that would be the problem
09:39.48JTcrap telco
09:40.06JTit's one thing overriding false callerid, it's another overriding calling presentation
09:40.45endreyeah
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09:55.36zeeeshusing ... asterisk-1.2.12.1,,, calling through xpro... receiving messenger ... service unavailable .... anybody know????
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10:11.37shadebobhi, I have an SPA400 connected to my asterisk. I have wrote an AGI for manage incoming calls by spa400 port. My actual problem is how to manage outgoing calls by part? anyone have an SPA9000?
10:12.55endre~xpro
10:13.25endreis that the 'new' x100p?
10:13.44endreoh yeah
10:13.48endrethe full version of xlite lol?
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10:41.38Mw3JT: i got it :)
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10:43.12Ast001does anyone have problem with "RTCP transmission error halted" problem in asterisk 1.4.0 ?
10:43.41shadebobhi, it's seem linksys spa400 have open source code. But ftp.linksys.com don't work. Anyone have the SPA400_v1.0.0.2.tgz
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10:45.36FlatFootanyone got some info on setting up the cisco 7905g
10:45.56FlatFootgot one converted to SIP , think i have the settings right BUT
10:46.05_CRC_is anyone able to help me with a compile error on zttranscode.c on zaptel 1.4.0?
10:46.07FlatFootit does not seem to attempt to register
10:46.56_CRC_I bugged it a few days ago, but nothing so far :\
10:47.05_CRC_and I need to get my FXO up and running again :(
10:49.44*** join/#asterisk angler (i=angler@pdpc/sponsor/digium/angler)
10:52.56_CRC_I wanna get the FXO's working again so asterisk stays and Nortel doesn't get a contract :\
10:55.30_CRC_bug report --> http://bugs.digium.com/view.php?id=8945
10:56.32FlatFootbtw it is natted
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11:34.16shadebobanyone have the spa400 source code from linksys?
11:37.05zoaftp://ftp.linksys.com/opensourcecode/spa400/
11:43.24davieyAnybody here use voipcheap.com???
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11:54.04shadebobzoa : ftp.linksys.com doesn't work for me?
11:54.14shadebobcan you try form me please?
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11:54.28m2olufhello all !
11:54.32zoait doesnt work for me either
11:54.35zoaprobably down
11:54.37zoathats the link
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11:55.16x25sre
11:55.22shadebobzoa : and it doesn't exist any mirror....
11:55.27x25smi PRI E1 is down for 1 minute, i see this error in Asterisk CLI:  zaptel Disabled echo canceller because of tone (rx) on channel 6
11:56.43m2olufi try to set language=no in sip.conf after installing norwegian lang files bu the asterisk refuse to speak norwegian. ???
11:58.23corruptordoes any1 know  what to do with Zap state Rsrvd? What leads to this state?
11:59.42x25smy target is a TE412P dont have fxo
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12:04.54m2olufi try to set language=no in sip.conf after installing norwegian lang files bu the asterisk refuse to speak norwegian. of course i have copied lang files to /var/lib/asterisk/sounds/no. anyone?/
12:05.25creativxvoop
12:05.29creativxits not norwegian
12:05.33creativxits neu norwegian
12:05.45creativxand some of the voop files are buggy
12:05.48creativxnot normalized volume
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12:13.42m2olufcreativx: well i should have got some changes in the pbx? (not?)
12:13.48creativxwell yeah
12:13.58creativxbut if i remember correctly
12:14.05creativxi had to cp the files by hand
12:14.53creativxls /var/lib/asterisk/sound/no ?
12:15.18Ast001messages:[Jan 17 12:14:18] ERROR[2803] rtp.c: RTCP RR transmission error to, rtcp halted Success
12:15.18Ast001rtp.c: RTCP RR transmission error to, rtcp halted Success
12:15.18Ast001rtp,c: RTCP transmission error to rtcp halted Success
12:15.40Ast001oh sorry my gnome goes mad :(
12:16.15shadebobanyone have a spa9000?
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12:19.34corruptorAst001: i sometimes get the same message on 1.4.0
12:19.52Ast001well that happend once in a week
12:19.57Ast001in my case
12:20.07alexandrekelleranybody using asterisk-ss7 ?!
12:20.08Ast001and noone can login on system until restart
12:22.05*** part/#asterisk x25s (n=oski@251.Red-80-24-18.staticIP.rima-tde.net)
12:23.27Ast001does anyone know why that transmission error show up ?
12:24.04Ast001what is the problem ? ports from 10000 to 20000 are open in rtcp.conf
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12:32.39E-bolaWhen i do a sip show channels int he asterisk console the User coloum gets cut off
12:32.46E-bolacan i make it show the whole username field?
12:32.51*** join/#asterisk tcseke (n=chatzill@22-36.adsl.etel.hu)
12:32.55E-bolaregardless of how many characters it is?
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12:33.10m2olufcreativx: i do have the files in /var/lib/asterisk/sound/no i guess the ls it to long to paste here :)
12:34.08m2olufcreativx: i've tried to restart and reload asterisk to re-read sip.conf but no luck.
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12:37.05creativxset up a test extension then
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12:51.18LeddyHMOur Asterisk vendor decided to go awol, so I'm trying to pick up the pieces and learn asterisk as we need to add a few extensions. From what I can tell I just need to edit sip.conf, extensions.conf, and voicemail.conf. does that sound about right?
12:51.44ModcutsHow would one setup rule so that if a trunk returns that the channel is unavailable sends the call out on another channel?
12:53.29lilalinuxhow do I make a RewriteRule, that returns a 404?
12:53.29creativxLeddyHM: yes
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12:55.43LeddyHMonce I'm done do I just need to HUP the asterisk process?
12:57.27creativxstop now
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12:58.10AhrimanesLeddyHM, connect to the asterisk console (asterisk -r) and type reload
12:59.18creativxor reload
12:59.19creativxdamn fever
12:59.46LeddyHMdoes that restart the entire process, or just reread config files?
13:01.28creativxjust reread
13:01.38creativxif you do stop now the proc will die
13:02.02LeddyHMahh cool
13:02.19LeddyHMthanks :)
13:03.37creativxnp
13:03.51lilalinuxIf I want to remove a dynamic page example.com/cgi-bin/index.cgi?foobar but still want to leave all other QUERY_STRINGs untouched, would it be wise, to make a RewriteRule with 410 gone?
13:04.26lilalinuxI ask, because if I do that, the returned message is: "The requested resource /cgi-bin/index.cgi is no longer available on this server and there is no forwarding address. Please remove all references to this resource."
13:05.12lilalinuxWouldn't that mark index.cgi completely as gone?
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13:33.16myiagyhi, i'm having some problems with talkoff.. i found that there's a parameter dtmfthreshold that i can adjust.. but i can't seem to find where this parameter goes..
13:33.28myiagydoes anyone know how can i set the dtmf detection sensitivity?
13:34.27coppiceDTMF sensitivity is not something you set
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13:35.19myiagycoppice oh, any ideas then on how to solve this problem?
13:36.03Spudz0rhi guys, just wondering if it was possible to have a custom outgoing thing in from-internal to put the person back to a dialtone using DISA?
13:36.15Osserelaxdtmf=yes  relaxdtmf=no
13:36.35Spudz0rie: when the receiver of the call hangs up, they get a dialtone again, instead of being hung up on.
13:36.45myiagyi've set relaxdtmf=no.. but isn't this option off by default anyways?
13:36.50coppiceif relaxeddtmf is set you might have talkoff issues
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13:38.44myiagythat's the problem.. it's already set to no..
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13:39.13myiagyi found a few reports of people having this trouble with 4E1 cards.. could it be something with the wct4xxp module?
13:39.44coppicethe DTMF detector in * doesn't have talkoff problems if that is set to no. are you sure that is the detector you are using? do you have a hardware card with a DTMF detector?
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13:40.22nas_lslsahello people
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13:41.15coppicemyiagy: do you have one of those cards with the early crappy echo canceller? the DTMF detector on that needs to be turned off
13:41.48myiagycoppice i believe it's an old quad-span card.. i just don't know what model exactly, gonna try to find out
13:42.08myiagywhere do i turn de dtmf detector off?
13:42.42coppicei can't remember
13:42.56myiagyok, i'll look for it..
13:43.02myiagythanks
13:45.07myiagy"you can set vpmdtmfsupport to 0 in wctdm24xxp.c or wct4xxp.c and recompile, or you can specify it as a kernel module option at runtime." according to digium knowledge base :D
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13:53.33susinthshey
13:54.09susinthscan SIP/IAX users be put in mysql database instead of sip/iax.conf?
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14:00.54myiagysusinths yes
14:00.55myiagyhttp://www.voip-info.org/wiki-Asterisk+RealTime
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14:09.35tzafrir_laptopsusinths, yes
14:09.54tzafrir_laptopthere are a number of ways, but generally look into "real time"
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14:12.40susinthsok, thanx guys
14:13.00susinthsi read over it, the realtime explanation
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14:14.00susinthsnot included because of mysql licence change..
14:14.04Simplixhello, is there any changes with the virtual ringtone syntax in the Dial command ?
14:14.37*** join/#asterisk brad_mssw (n=brad@shop.monetra.com)
14:14.40Simplixwith the 1.4.0 version (sorry)
14:15.00QwellSimplix: virtual ringtone?
14:15.03susinthsi'm planning to offer sip accounts to hundreds of members, is mysql the best way?
14:16.02Simplixyes for exemple : Dial(${TRUNK1}/${EXTEN}|120|r)   <= the |r
14:16.39susinthstzafrir_laptop?
14:16.44susinthsmyiagy?
14:17.08Simplixwith the 1.2.14 it works fine .... now with the 1.4 I don't have ringtones any more
14:17.21QwellSimplix: you shouldn't be using r
14:17.35Simplixah ? .... other solution ?
14:18.04tzafrir_laptopsusinths, realtime is generally is another way to get configuration
14:19.00susinthstzafrir_laptop: Like using exten=>MYSQL() command?
14:19.08tzafrir_laptopsusinths, there are basically two "realtime" methods: the real realtime and static realtime
14:19.41tzafrir_laptopstatic realtime only updates your configuration at reload time. non-static, queries the database at call time
14:19.41susinthstzafrir_laptop: i see
14:20.00tzafrir_laptopthis is a tradeoff: update time vs. overhead at call time
14:20.15susinthstzafrir_laptop: ok
14:20.33QwellSimplix: nope, just remove it, and it'll magically work
14:20.51tzafrir_laptopAnd you can also have your own custom stuff: store it in some other method, dump it to a file, and reload
14:20.57SimplixQwell, thx ... i'll test that
14:21.16susinthsi see
14:21.30tzafrir_laptopThis may be better if you want to do some processing to the tored data before sumping it
14:21.32susinthsbut static sounds better for call setup timing..
14:21.42creativxmysql isnt that slow
14:22.00susinthsreally
14:22.05tzafrir_laptopsusinths, if it is better for you, use it. Do you have res_mysql from addons installed?
14:22.26susinthsno, i don't
14:22.29tzafrir_laptopcreativx, but it is an extra point of failure
14:22.37SimplixQwell ok ... it works :) I previously put it for multiple out line .... now it's useless thx again
14:22.39susinthsbut i will inst soon
14:22.50susinthswhats that?
14:23.00creativxtzafrir_laptop: indeed.. and call handling + failure = bad business
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14:23.39susinthsbut i mean, does noting happen to the caller when ast looks up mysql?
14:23.39tzafrir_laptopanyway, I don't have much experince with big databases. So take other people's words for it
14:23.49susinthsok i see
14:24.32susinthsi wonder if people define all the users in sip/iax.conf files?
14:24.38susinthsok :)
14:25.18susinthsflat files scales badly?
14:25.33myiagysusinths you need to install asterisk-addons
14:25.56susinthstzafrir_laptop: Thanks a lot!! Appreciated
14:26.10susinthsmyiagy: I see
14:26.13susinthsi will soon
14:26.24susinthsis mysql module inside addons?
14:26.28myiagyyes
14:26.37susinthsok
14:27.08susinthsfor instance fwd.net must have users in some kind of database
14:28.51susinthsi think i will install openSER too
14:29.18susinthsi read that ast isn't made for pure SIP setup
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14:31.38ez`.
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14:34.45stealthim new to asterisk, my first goal is to get it working.. id like a web based voice mail I understand now there is a web gui for the asterisknow, im  using Gentoo Linux.. would this be found in asterisk addons?
14:35.04Qwellstealth: see topic
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14:36.20stealthoh my bad.. asterisk-gui ..thanks
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14:36.28Defendany one got any recomdations for voip providers for the US? i am looking / testing some and would like to know if there are any ya all recomend
14:37.37x86i run a provider called ShellShark Networks
14:37.37x86https://voip.shellshark.net/
14:37.37x86we're pretty good :)
14:39.44Defendcan i msg you for a secound?
14:39.57x86sure
14:40.38tzafrir_laptopstealth, there is no gui for the voicemail in the asterisk GUI, AFIAK
14:40.48Qwelltzafrir_laptop: you!
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14:41.08stealthwha!?
14:41.09Qwelltzafrir_laptop: I couldn't get the xpp stuff in 1.2 to compile on fc3 yesterday
14:41.14tzafrir_laptopthere is something called vmail.cgi in the asterisk source tree. It is rather buggy and limited
14:41.27tzafrir_laptopQwell, which version? 1.2 trunk?
14:41.31Qwell1.2 svn
14:41.38tzafrir_laptop1.2 SVN?
14:41.41Qwelllatest as of yesterday evening
14:41.59tzafrir_laptopcould you please send/pastebin/whatever the error?
14:42.11QwellI'll be able to in an hour or so
14:42.31tzafrir_laptopstealth, there is also a nice thing called ARI (Asterisk Recording Interface)
14:42.41tzafrir_laptopI'm not sure how actively developed it is
14:44.01tzafrir_laptopQwell, next time if I'm not here, feel free to mail me or open a ticket (there's now an xpp compoennt in zaptel)
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14:44.09jeremy_gis asterisk time now loaded as a module
14:44.14jeremy_gdo i need to patch ast 2
14:44.19jeremy_gnej i dont think so
14:44.29Qwelltzafrir_laptop: will do
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14:51.42shadebobhi, someone have the spa400 gpl source code because linksys ftp doesn't work...thanks
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14:57.25drfreezeHi all
14:57.56drfreezeIs 512MB RAM sufficient for an * system supplying VoIP for about 15 phones?
14:58.18mercestesdrfreeze:  *nix, right?
14:58.24|Vulture|drfreeze: how many concurent connections?
14:58.38zoadrfreeze:its already overkill
14:59.04|Vulture|its true * in its basic form can run on anything lol
14:59.15zoait would probably work with 64mb ram too
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14:59.19mercestesI run * on a linksys wrt5gl
14:59.27mercestesit can handle two whole phone calls.
15:00.36coppiceits the floating point that really limits the current code on a wrt54g
15:01.10*** join/#asterisk tefster (n=ian@smtp.planetbuilders.org)
15:01.11mercestesI used openwrt on the router.
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15:02.00drfreezemercestes: yes, CentOS
15:02.14drfreeze|Vulture|: up to 15
15:02.25tefsterHi. I have a sip peer (incoming FXO to Ethernet gateway) which is registered with Asterisk and to which I can send calls, but any incoming calls get
15:02.29tefsterrefused with "handle_request_invite: Failed to authentic            ate user"
15:02.50tefsterusername,secret,md5secret,authname,and realm all seem correct. what else might i have missed ?
15:03.13drfreezeSince transcoding is not required, I assume the processor and the ram can be light
15:03.20*** part/#asterisk a1fa (n=a1fa@unaffiliated/a1fa)
15:03.46mercestesdrfreeze:  ew..CentOS.  Are you going to bre running any other services?  Web?  Email?
15:04.17RoyKdrfreeze: 512MB is sufficient for 1000 phones, or perhaps 2, the latter if you count real-world memleaks
15:05.15*** part/#asterisk fenlander (n=neils@82.152.81.57)
15:05.44mercestesI don't recall any mem leaks.
15:05.58zoai do :)
15:06.29drfreezemercestes: nope
15:06.39mercestesdrfreeze:  Should be ok.
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15:08.13mercestesdrfreeze:  do some searching on converting your moh, ivr and voicemails to ulaw or whatever codec you are using.  Sound files can cause transcoding too.
15:08.51drfreezemercestes: k
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15:26.01ModcutsAfternoon, Is it possible to send a call out via a second trunk if the first fails?
15:26.31tzangerModcuts: yep
15:26.33tzangerI do it all the time
15:26.44tzangerDial(SIP/${EXTEN}@peer,,g)
15:27.06tzangerGotoIf($[${DIALSTATUS}=CHANUNAVAIL]?tryagain)
15:27.11tzangerGotoIf($[${DIALSTATUS}=CONGESTION]?tryagain)
15:27.28tzangerGoto(postdial,1)
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15:27.46tzanger(tryagain),Dial(Zap/g1/${EXTEN},,g)
15:27.47tzangeretc
15:28.35Modcutslovely
15:28.39Modcutsthank you
15:30.55drfreezemercestes: why the 'ew' on CentOS for Web email?
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15:31.53af_what is web meetme?
15:32.01mercestesdrfreeze:  ew for CentOS and * in general.  I would say it probably works fine for web and email
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15:32.45drfreezemercestes: AAH used to use CentOS. I have not tried trixbox, but would assume it is the same
15:32.55mercestesdrfreeze:  I just see alot of CentOS issues and I'm anti redhat and asterisk to begin with.
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15:33.26drfreezemercestes: your anti asterisk or anit 'rh + asterisk'?
15:33.29mercestesdrfreeze:  AAH was designed for retards.  It doesn't promote the OS much for me.
15:33.52mercestesdrfreeze:  Pro asterisk, anti asterisk on redhat and redhat based distros.
15:33.57drfreezemercestes: retards or lazy/busy people. :)
15:33.59mercestesbut I'm not really anti redhat
15:34.06drfreezemercestes: what distro do you use?
15:34.11mercestesgentoo
15:34.15drfreezeahh
15:34.19drfreezewhat about FC6?
15:34.25mercestesoh, is that gentoo?
15:34.31drfreezeFedora Core
15:34.34mercestesThen no.
15:34.45CrescendoUsing "sip show channels" my caller ID is unreadable. " 0c75e4460bd " - Hex, apparently - why?
15:34.55drfreezeis there a reason to use gentoo if one is not a gentoo bigot?
15:35.26drfreezemercestes: gentoo a good server os?
15:35.29drakoseriously, is there any good SIP or IAX client for Linux?
15:36.02tzafrir_laptopdrako, kiax
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15:36.12mercestesdrfreeze:  Honestly....that's kinda trollish.
15:36.24tzafrir_laptopdrako, also: ekiga is not bad, but I prefer twinkle
15:36.31drfreezemercestes: sorry, just a linux idiot when it comes to all the distros
15:36.46drakotzafrir, ekiga is soo buggy
15:36.48mercestesdrfreeze:  You should use what your comfortable with.  I have a few people that swear by redhat.  If you like CentOS and your comfortable with fixing it, then use it.  I'm comfortable wtih fixing gentoo...8shrugs*
15:36.52drakotwinkle?
15:36.54drakolet me see
15:36.58drfreezemy fried just told me gentoo will maximize performance on a particular processor
15:37.09mercestesyour fried is an idiot.
15:37.20Crescendodrako, you can use wine on most windows clients
15:37.46mercestesparticular "flavors" of gentoo will maximize performance on the systems it is modified for by using the proper cflags/switches/useflags/compile options.
15:37.47drfreezemercestes: why, are you saying that gentoo will not maximize performance over other distros?
15:38.01mercestesand the soure code will be compiled against the specific system youare on, theorhetically, maximizing those binaries for your system alone.
15:38.12mercestesgoogle gentoo ricing for more info on that.
15:38.18tzafrir_laptopdrfreeze, that's rather marginal
15:38.33drfreezemercestes: oh, so gentoo is more like fbsd in that is compiles apps instead of doing binary installs?
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15:38.40drakoCrescendo, oh really? which client do you use with wine that work for prodcution?
15:38.42mercestesbut it's not maximized for any one processor or setup, and....unless you spend half-your life tinkering with it....yea, the difference is nominal.
15:38.46tzafrir_laptopYou'll get most performance boost from rebuilding 5 or 6 basic packages
15:39.09tzafrir_laptopAn the kernel and libc are already optimized in most distros. Not to mention stuff like openssl
15:39.09drfreezemercestes: this is good info, thanks
15:39.13mercestesdrfreeze:  emerge downloads source and custom compiles it on your server, yes.
15:40.05Crescendodrako, idefisk works well
15:40.06mercestesdrfreeze:  I personally like the ability to do an emerge asterisk libpri zaptel asterisk-addons asterisk-sounds mysql apache nullmailer and go home for the evening and wake up to an almost complete install.
15:40.15mercestesdrfreeze:  But that's just me.  Some like doing manual compiles.
15:40.19tzafrir_laptopFrom what I hear in sane distro-fights , the thing people most love about gentoo (the distro, not the file manager) is the ability to easily customize anything
15:40.43drfreezemercestes: sounds like my friends in FreeBSD land
15:40.54mercestesyea, heard good things about Fbsd.
15:40.56mercestesnever tried it.
15:41.02mercestesOpenBSD turned me off to BSD in general tho.
15:41.02tzafrir_laptopmercestes, with Debian it is 'apt-get install', wait 5-10 minutes, answer a few question , and it's done
15:41.09tzafrir_laptop(that's the theory)
15:41.43mercesteshells, I remember th efirst time I found "yum" on redhat....
15:42.17mercestesit's like spending 20 years on the keyboard and suddenly finding yoru mouse.
15:42.41Makenshii didnt think yum shipped on red hat, it's been up2date as long as i've known
15:42.50mercestes....I also remember the first time yum broke my library dependencies.  ..it was like, going "I hate you you damn mouse!"
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15:44.54mercestesI guess Redhat is good if you use the cds and install the GUI.  but...if I want a GUI I use Kororaa, which is gentoo but with retardo-install cds
15:45.04mercestesbut, yea </distro war>
15:45.59mercestesdrfreeze:  People have gotten * to work on CentOs.  :)  So there is a case history of success.
15:45.59tzafrir_laptopmercestes, yum is highly inefficient
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15:46.00tzafrir_laptoptry apt on the same system, and it will work much better
15:46.00mercestestzafrir_laptop:  It quite efficently hosed my library dependencies and, cooincidentally, my system.
15:46.00coppiceyum could teach microsoft how to use memory :-)
15:46.17mercestes....yum could teach microsoft how to update a gfx driver too.. =/
15:46.45tzafrir_laptopmy Fedora ontact tells me that it generally undergoes optimizations
15:47.02tzafrir_laptopBut there are more efficient tools. apt, for instance.
15:47.02w0ls0nHi all. My boss was mentioning to me about buying a new phone system. We currently have 4 trunk lines. What kind of box would I need to have for a PBX for asterisk with voicemail?
15:47.24mercestessomething with a 4 port card in it.
15:47.31tzafrir_laptopHosing library dependencies is a matter of broken packaging. I'll have to see an example to attest the source of that
15:47.35mercestes4 trunk lines is 4 individual lines?
15:47.41w0ls0nyea
15:48.02mercestesyea, 4 port analog card then...and any descent server system will do.
15:48.19mercestes....actually that 486 your using as a door stop would work if you did it right.
15:48.32mercestesHell, i could install linux on a linksys router and make it work just off a router.
15:48.36mercestesfor 4 lines.
15:48.49mercestesjust replace the ethernet ports with fxs ports...
15:49.53w0ls0nhmmm
15:49.55mercestesw0ls0n:  Yea, if your running 4 analog, then hardware really isn't an issue.
15:49.59w0ls0nok
15:50.11w0ls0nasterisk also includes voicemail?
15:50.16mercestesbuilt in.
15:50.25w0ls0nhmmm
15:50.32w0ls0nanyone using asterisk on FreeBSD?
15:50.46mercestesI've heard it can be done.....I've heard it can be difficult to get going.
15:51.02mutpoor w0ls0n
15:51.04mutsticking to bsd
15:51.08w0ls0npfft
15:51.11w0ls0nBSD rocks
15:51.13mutwhat a trooper
15:51.20mercestesnot for asterisk....lol
15:51.23CrescendoUsing "sip show channels" my caller ID is unreadable. " 0c75e4460bd " - Hex, apparently - why?
15:51.36mutyou'll have more trouble that its worth
15:51.39w0ls0nwell I'll just have to make this work
15:51.44mutplus you'll want it on a dedicated box anyway
15:51.47w0ls0nyep
15:52.00w0ls0nhow much storage will you think I need?
15:52.09mercesteshow many voicemails do you want to keep?
15:52.11mutuh
15:52.13w0ls0nright
15:52.29mut2gig?
15:52.39mercestesmut:  aw, splurge a bit...2.5 gig
15:52.44w0ls0nok, what is your opinion on the best distro to run it on
15:52.47mutok ok
15:52.50mutdebian ofcourse
15:52.55mut;)
15:53.05mercestesw0ls0n:  http://www.voip-info.org/wiki/view/Asterisk+FreeBSD
15:53.09mercestesGentoo.
15:53.16mercestessomeone else will vote CentOS.
15:54.02muti scoff at gentoo
15:54.02mercestesI scoff at debian.
15:54.02mutheh
15:54.02mercestesPolycom wants RedHat Enterprise for a support contract
15:54.02w0ls0nty mercestes
15:54.02mercestesWhich.....CentOS is a rebranded RHEL.
15:54.04w0ls0nso what kind of hardware will I need other than just the basic stuff
15:54.13mercestesother guys will tell you RH sucks for asterisk.
15:54.23mercestesThe answer is, use whatever distro you are *most* comfortable with.
15:54.39w0ls0nFreeBSD for me :-)
15:54.46mercestesw0ls0n:  Either an FXS or FXO card.  I think you need FXO to plug into POTS lines.
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15:54.54w0ls0nok
15:55.09Teelidoes anybody know who is mattf. he posted a patch for USERUSERINFO field in Q931
15:55.12w0ls0nwell we can prob get roadrunner and kill the DSL line once this is up and going
15:55.14TeeliI cant get it to work
15:55.22mercestes...
15:55.36mercestesyour internet service has nothing to do with your usage of 4 analog lines.
15:56.01mercestesunless you intend to replace those 4 analog lines with VoIP instead of POTS, and i fyou do that, roadrunner is likely *not* the way to do it.
15:56.41w0ls0nwell with the money we save we can upgrade our current ISP
15:56.47mercestes....then again...DSL isn't the way to do it either if your referring to the consumer level broadband service.
15:56.54w0ls0nwe spend $400-$600 a month on our phone bill
15:57.02mercestesThen get a T1
15:57.42w0ls0nIs there a way to keep my current # ?
15:57.55mercestesw0ls0n:  It's called porting your number.  yes.
15:57.59w0ls0nok
15:58.14w0ls0nMy boss just made this #1 priority
15:58.32mercestesI suggest a contractor then..;)
15:58.41w0ls0nI ahve tons of free time
15:58.55mercestesI suggest voip-info.org and google then.
15:59.07w0ls0noh yea
15:59.16w0ls0nthanks
15:59.19mercestesnp
15:59.22mercestesgood luck
15:59.31w0ls0n:-)
16:00.58x86anyone ever trunked vlans to a polycom IP-601?
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16:01.28x86trying to run two vlans to the polycom with 802.1q from a cisco switch
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16:01.48x86i want the phone to use one of the vlans, and the PC switch port to use the other vlan
16:01.52x86anyone ever do this?
16:02.27aydiosmiocan't you assign VLANs based on MAC address or IP Address?
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16:04.07drakokiax does not make good in my laptop
16:04.10drakoi need someting better
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16:08.04x86aydiosmio: no
16:08.28x86aydiosmio: you assign them per switch port on the cisco switch, or trunk multiple vlans on the same switch port
16:08.37x86aydiosmio: in my case, i'm trunking to the phone
16:08.56mercestesx86:  and how will the cisco differentiate between the PC and the phone coming across on the same wire??
16:09.06Defendis there a feature list/change log for 1.4 out there yet?
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16:10.56mercestesx86:  Just use seperate subnets.  No reason computers and phones need to share a subnet or be natted together.  Put the computers on 192.168 and the phones on 10.10 or something.
16:10.58x86mercestes: 802.1q tagging, done by the switch on the polycom
16:11.14x86mercestes: i AM using different subnets, that's why i'm using vlans ;)
16:11.26mercestesso what's the problem?
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16:15.22x86mercestes: i want the polycom to know how to tag each port on it's own switch
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16:17.02kFuQAdding an iTunes Telephone Controller to Your Asterisk PBX     http://nerdvittles.com/index.php?p=159   rofl
16:17.19Qwell[]what?
16:17.34mercestesx86:  Did you try google?
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16:22.45sevard"Then you'll need a rock-solid Asterisk system. We recommend TrixBox 1.2.3"
16:22.53tzanger:-)
16:22.55CrescendoUsing "sip show channels" my caller ID is unreadable. " 0c75e4460bd " - Hex, apparently - why?
16:23.00mercestes...
16:23.09tzangerCrescendo: probably a pointer to the string
16:23.09x86sevard: hah
16:23.22Crescendosevard, wish we would all get along, though ... :P
16:23.28in-ptHi all
16:23.44w0ls0nIs there a web interface for asterisk once I get it all setup and running?
16:23.45Crescendotzanger, our voip provider is saying that we're not providing it properly.
16:23.45in-pti am unable to see cdr_addon_mysql.so module compiled with asterisk-1.4
16:23.57tzangersip debug might help
16:24.00mercestesw0ls0n:  not in this channel there's not.
16:24.04in-ptis it present in this ver of asterisk or obsoleted
16:24.14CrescendoWe did a packet trace on a call connection, same thing
16:24.34sevardthis borrowed laptop is LOUD
16:24.51CrescendoIs there any optional encryption for CID or anything that might be breaking this?
16:25.44sevardCrescendo: where are you setting CNAM in your dialplan? paste that line
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16:33.50Crescendoin sip_additional.conf?
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16:37.22sevardnevermind then.
16:42.51Defendi have a user who is reporting that when an incoming call comes it it is almost like it is being delayed and cutting of the few secounds of the caller any ideas?
16:43.52tzangerDefend: packet trace
16:44.37CrescendoDefend, I've had similiar issues with outbound calls. :)
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16:44.52CrescendoThey can't hear us for a second or two.
16:45.18BrokenNozeanyone tried wireless handsets with asterisk give me any recommendations?
16:45.27mercestesDefend:  Router issue most likely.  could be a timeout where first attempt to establish connect times out and second attempt succeeds.
16:45.37mercestesBrokenNoze:  Yea, don't use them.
16:46.15coppiceif you want a wireless phone don't use 802.11. Use a DECT IP phone
16:46.27BrokenNozemercestes : yeah, I thought so. have a client that want's 75 wireless handsets. assuming there's no WAY you've got enough bandwidth?
16:46.28Defendbut were all lan here :/
16:47.02BrokenNozeDECT IP, don't think it'll give me the range
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16:58.28Dr-Linuxguys, a little question
16:58.28Dr-Linux<PROTECTED>
16:58.29Dr-Linux<PROTECTED>
16:58.29Dr-Linux<PROTECTED>
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16:59.10Dr-Linuxin above case caller still hears rings, where caller should hear MOH. What could be the reason?
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17:06.40danphuh. polycom's offering version 2.1.0 of the SIP software for download
17:07.20Qwell[]directly?
17:07.34wunderkinbeta testing?
17:07.37danpahh, i guess not. it has a link for partners to log into
17:07.43danpbut i hadn't seen that before
17:07.49wunderkinis it a release?
17:09.30Dr-Linuxany clue for my question?
17:11.10stubertDr-Linux: Are you using the "m" switch in the dial command?
17:11.29Dr-Linuxstubert: no
17:12.00stubertThen how are you executing the music on hold in for the call?
17:12.01wunderkinyeah i guess so, lets see if i can get it to check the release notes
17:12.03Dr-Linuxstubert: i checked on my 4 different asterisk servers, but one server seems to be in problem
17:12.44Dr-Linuxstubert: basically i'm using queues
17:12.58stubertOh, then I don't know...
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17:18.17Dr-Linuxstrange, since it's printing on CLI, that "Started music on hold, class "default"  but i still hear rings
17:18.19Dr-Linux<PROTECTED>
17:18.19Dr-Linux<PROTECTED>
17:18.19Dr-Linux<PROTECTED>
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17:20.39mercestesDr-Linux:  There are queue optiosn to play ringing instead of MoH.  r in particular.
17:21.33Dr-Linuxmercestes: i can't see any ringing option in queues.conf configuration, even i matched it with my other servers
17:21.49Dr-Linuxalso i checked the process "mpg123" is running
17:21.54sevardDr-Linux: can you play MoH on a regular call?
17:22.01mercestesDr-Linux:  the ringing option is |r  :)
17:22.08mercestesyea, try MusicOnHold() and see what it does
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17:22.36Dr-Linuxmercestes: yes i know but i didn't use that ,r option
17:22.51jjshoeI have a card with HW echo can that requires echocanel=yes in zapata, what can I do to make sure software echo can is turned off?
17:22.57Dr-Linuxmercestes: that's good idea, let's see if i can test MOH on my this production server
17:27.06Defendi got a question is there a way to modify this exten => *,2,Page(SIP/3218x1&SIP/3219x1&SIP/3220x1) ; add all your devices here  so it does all extentions that start with 3 and are 3 digits long?
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17:28.13flujanhi guys... I am having a strange behavior with asterisk realtime odbc and postgresql
17:28.34flujansometimes the jitter from my iax clients goes up to 1236
17:28.36flujanops
17:28.50Dr-Linuxmercestes: how can i set the time in such way it should play music for long: exten => 7272,n,MusicOnHold()
17:28.54flujanI dunno what is causing this behavior.
17:28.56mercestesDefend:  You mean like _3XXX?  You'd have to build a command in AGI to do that.
17:29.10mercestesDr-Linux:  default should be indefinite.
17:29.12flujanthe database is also used by my crm system.
17:29.24flujandoes the database access can make asterisk have this strange behavior?
17:30.13mercestesDr-Linux:  if it doesnt work and you are using defaults....check yoru hardware.  memtest and fsck.
17:30.23Dr-Linuxmercestes: no, it ends fast and goes to next priroriy
17:30.36mercestesDr-Linux:  then something is terribly wrong.
17:30.46Defendok cool gives me a place to start thanks mercestes
17:31.04Dr-Linuxhhm..
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17:33.13Dr-Linuxmercestes: sometime it plays MOH for long, then voice disappears
17:33.30Dr-Linuxmercestes: maybe something is wrong with mpg123 player :S
17:36.11jjshoeechocancel=yes
17:36.14jjshoeoops
17:36.17sevardDr-Linux: ls -l $(which mpg123)
17:36.23jjshoewhat release of zaptel is mg2 in?
17:37.54Qwell[]jjshoe: should be in 1.2 at least
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17:47.43jjshoeQwell[] ?
17:47.50Qwell[]mg2
17:48.00jjshoeoh, it's in zaptel-1.2.3
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17:53.57fetchsterAnyone know anything about Snap (snapanumber.com)?
17:54.32b11d|bblOh! Snap!
17:54.34b11d|bbl:|
17:55.22fetchsterI didn't think people could say that anymore after the dumb BMW commercials...
17:56.22b11d|bbli dunno.. i dont watch television
17:56.37b11d|bblthe few shows I do like, I pull off "the internet" and they are commerical free
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17:58.50[[blah]asfdok, so i am trying to dial an sip extension from one server to another. I have a peer and a user entry on each server for the connection sip show peers from each server says that they are connected. i have a dial command from the server i am dialing from it is formatted dial(SIP/peer_name/extension_on_other_server). is that the correct format and way to do this?
17:59.10b11d|bbldont you need to do IAX for inter-asterisk dialing?
17:59.15b11d|bblmy experience on this is limited, i may be wrong..
17:59.31[[blah]asfdit can be done with both
17:59.34b11d|bblcool
17:59.53FuriousGeorgei need a usb "conference microphone" that can cancel echo like people are chasing it
18:00.55b11d|bblget an old school 1940's radio microphone :)
18:01.40FuriousGeorgea tin scan a string and an inline usb adapter, perhaps?
18:02.01b11d|bblyou know it.. maybe a solar panel from a calculator for power :)
18:02.35FuriousGeorgehey dont knock photovoltaic solar cells
18:02.43b11d|bblwho was knocking them?
18:02.49b11d|bbli said we could use one
18:02.54b11d|bblnot "lets go and badmouth solar energy"
18:03.01Qwell[]solar...ha
18:03.09Qwell[]it's like...ambient
18:03.12b11d|bblhahaha
18:03.32b11d|bblmaybe we should construct a device powerd from the ambient "gravity" too
18:03.45FuriousGeorgei heard they are using nanotech to get the silicon receptors so tiny (4 nm) they catch light in the ir spectrum, and be dispersed in a paint and applied to my entire body
18:03.45Qwell[]b11d|bbl: ...
18:03.47Qwell[]that...
18:03.50Qwell[]could work
18:03.55b11d|bbl:P
18:04.23b11d|bblbrb
18:04.58FuriousGeorgea stirling engine runs on heat and part of its cycles uses gravity to help it drop a piston and compress air.  that's why they are so efficient
18:05.54*** join/#asterisk greendisease (n=jack@fedora/greendisease)
18:09.26*** join/#asterisk Burgwork (n=corey@ubuntu/member/burgundavia)
18:09.46Burgworkis there a central list for independant contractors who setup asterisk?
18:09.54Qwell[]~asterisk consultants
18:10.30FuriousGeorgeim a little scared.  a dude wants an extension in a conference room, where he has a pc and a big screen, which he wants me to put a softphone on.  what im terrified is gonna be the case is, if he actually uses this thing as a video phone in a room full of people, how bad is the echo gonna be
18:10.30Qwell[]jbot: Why do you hate me so?
18:10.36FuriousGeorgeBurgwork: on the wiki
18:10.36Qwell[]Burgwork: You can search voip-info.org
18:10.36FuriousGeorgesearch asterisk consultants
18:10.48BurgworkFuriousGeorge: by the wiki, you mean voip-info?
18:10.58FuriousGeorgei do
18:12.25BurgworkFuriousGeorge, Qwell[]: thanks
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18:18.41EmleyMoorHow temporary is a temporary greeting on voicemail?
18:19.35mogas much as you want
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18:25.23[TK]D-FenderEmleyMoor : It sits there until you remove it.
18:29.00tzafrir_laptop~asterisk consultants
18:29.01jbotit has been said that asterisk consultants is a generic term used to describe Qwell
18:29.09Qwell[]oh, sure
18:29.25moglol
18:29.36mogqwell is people
18:29.45Qwell[]jbot: no, asterisk consultants is a list can be found on the voip-info.org wiki
18:29.47jbotQwell[]: okay
18:29.50tzafrir_laptopanything of more substance to put there?
18:29.53Qwell[]~asterisk consultants
18:29.55jbotasterisk consultants is probably a list can be found on the voip-info.org wiki
18:30.13Qwell[]we don't need no stinking grammar
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18:31.13tzafrir_laptopThere should be a way to avoid the "is a"
18:31.19Qwell[]<reply>
18:31.28b11d|bbl</reply>
18:31.31Qwell[]jbot: no, asterisk consultants is <reply> a list of asterisk consultants can be found on the voip-info.org wiki
18:31.32jbotokay, Qwell[]
18:31.45Qwell[]if somebody wants to go ahead and find the URL for it...
18:32.11tzafrir_laptopjbot, no, asterisk consultants is <reply> a list of Asterisk consultants can be found at http://voip-info.org/wiki/view/Asterisk+consultants
18:32.13jbottzafrir_laptop: okay
18:32.19tzafrir_laptop~asterisk consultants
18:32.21jbota list of Asterisk consultants can be found at http://voip-info.org/wiki/view/Asterisk+consultants
18:32.36Qwell[]there you go :D
18:32.52[[blah]asfdok, i got my calls working. I am calling from one server to another and then out to the world. the call is successful, but it no longer carries the caller ID i set on it. how can I maintain that from server to server?
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18:35.48FuriousGeorge~furiousgeorge
18:35.49jbotextra, extra, read all about it, furiousgeorge is a knife-fighting (cable) monkey last seen with The Man with the Yellow Bat
18:35.52w0ls0nin regards to getting phone calls, How will that work? Say I have a phone number 207-999-8858 and I do the forward for it to my asterisk box, Should I have a static IP?
18:35.59FuriousGeorgekeeps getting funnier
18:37.18Bobthehunteranyregulations on PRI's in pologne ?
18:41.08*** join/#asterisk zotz (n=zotz@24.244.163.157)
18:44.37Dr-Linuxanybody is using native moh?
18:45.35*** join/#asterisk shodan (n=shodan@ip022.99-113-216.pppoe4.joliette.intermonde.net)
18:45.45Dr-Linuxcurrently i'm using mpg123 , it works but after little it auto stop playing. not sure what's wrong
18:47.02EmleyMoorw0ls0n: If you're forwarding directly to Asterisk, presumably you should
18:48.22[TK]D-FenderDr-Linux : In case you missed this for the past 2 years : Since * 1.2 came out about a year & a half ago, almost everybody has switched to native MoH.  Get off your ass and TRY it.  You've been around long enough that you should have just played around with it already.
18:49.50Dr-Linux[TK]D-Fender: you are right, but i never had such problem before
18:50.01nays85anyone know how to get in touch with voxee?
18:53.53*** join/#asterisk MaxeyPad (n=email@74-128-206-77.dhcp.insightbb.com)
18:54.31MaxeyPadI was curious if its possible to configure asterisk to work with vonage. Basically I'd like to setup my own voicemail and hold music. I know a while ago setting this up on vonage was nearly impossible
18:56.10*** join/#asterisk deb_user (n=none@70-59-111-238.albq.qwest.net)
18:56.34deb_usermy users are complaining about a little bit of static on the zap interfaces via a wildcard tdm, can anybody offer any suggestions?
18:56.43sevardMaxeyPad: IIRC vonage will only allow its service to work with its locked devices.
18:56.54sevardMaxeyPad: thusly, vonage does not support BYOD
18:57.13MaxeyPadi see
18:57.22MaxeyPadare there any equivalent services that I can do that with
18:57.42sevardof course, there are a varitey of voip-providers on the voip-info providers page
18:58.18sevardin fact, shellshark.net is one I can recommend
18:58.22deb_userI recommend vitelity communications, personally I've had pretty good experiences
18:58.32MaxeyPadis it super cheap like vonage?
18:58.50*** join/#asterisk juice (n=juice@mo-76-0-47-34.dhcp.embarqhsd.net)
18:58.52sevardCheaper
18:59.00deb_user.015/minute in the us and most of europe
18:59.27deb_userthing i like about vitelity is good international rates, which my business needs
18:59.42deb_userbut, look around for a company that meets your needs, there's a lot of good stuff out there
19:00.00deb_userso, does anybody have any tips for reducing static on zap interfaces?
19:00.13deb_userpersonally I don't care, but my users are bitching
19:07.17Bobthehunterhmm waht the regex to get the peername in channel ? BLAH=CUT(REGEX("/^[A-Z]{3,4}\/(.*)\-(.*)/i" ${CHANNEL}))
19:07.20Bobthehunterdoesnt work
19:07.34Bobthehunterto get foo out of sip/foo-blah
19:07.54Bobthehuntercut can get me the sip/foo .. but then its 2 queries
19:10.12DrukenLPYsevard: can't you go with a softphone thing with vonage and make asterisk work on that ?
19:11.18Carp1I am 100% aware of the statement "you pay for what you get"....however, can someone recommend me a good, reliable cheaper than most IP phone?
19:13.32*** part/#asterisk [[blah]asfd (n=ckwall@63.149.122.93)
19:13.37Dr-Linux[TK]D-Fender: i have just configured the native MOH, but same problem :S
19:14.00Dr-Linuxit works for stops automatically :S
19:14.09booraywhat does BLF stand for?
19:14.28cpmcontet?
19:14.30JoNatetelepathy...*sigh*
19:15.26CunningPikebooray: Busy Line Field
19:15.28SuPrSluGdeb_user,have u tried fxotune?
19:15.28CunningPike~blf
19:15.31jbotfrom memory, blf is Busy Lamp Field, aka little lights next to speed dials that light up when the person is on the phone and blink when that line is ringing.  hint extensions are static mapped to SIP or other channels.
19:15.44DrukenLPYCarp1: cheaper then most... what is your price range?
19:16.03deb_userSuPrSluG: yes, even the version included with the newest 1.4 zaptel drivers
19:16.07booraythank you cunningpike
19:16.09Carp1under 200.
19:16.15DrukenLPYcdn or usd ?
19:16.23*** join/#asterisk J4k3 (i=jsuter@dhcp-12-197-128-58.intrastar.net)
19:16.51SuPrSluGdeb_user,try loading driver like modprobe wcfxs lowpower=1?
19:17.57deb_usersuPrSluG: what does that do?
19:17.57CunningPikejbot is not a dog.......... ;)
19:18.18DrukenLPYjbot is the channel bitch :)
19:19.48[TK]D-Fender~areyouadog
19:19.49jbotbark bark!
19:19.57[TK]D-Fenderah HAAAA!!!!!
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19:23.44fetcherhas anyone tried using iptables & tc to prioritize VoIP on low-bandwidth links (DSL, etc.)?
19:24.36*** join/#asterisk tim0123 (n=cash247@adsl-75-39-213-70.dsl.rcsntx.sbcglobal.net)
19:24.56tim0123Anyone know how to record inbound calls
19:25.10fetchertim0123: Monitor app
19:25.24tim0123On every call
19:26.49joe[TK]D-Fender: wow, you can make ppl bark on demand, impressiv! ;)
19:27.17[TK]D-FenderIf only I could use my powers for good ;)
19:27.24joeindeed ;)
19:28.00tim0123Whats a good way to record all inbound calls?
19:29.22niZonMonitor()
19:30.26Bobthehunterso i cant use reserved chars in cdruserfield.. like ; & * { } and anyother ?
19:30.32Bobthehuntercan i escape
19:31.02Carp1recomendations for IP phone under $200?
19:31.43J4k3my grandstream 101's seem to work alright.  you could buy like 7 for $200
19:31.44J4k3hehe
19:31.53EmleyMoorI think jbot's a pussycat really
19:32.42[TK]D-Fendertim0123 : "show application mixmonitor"
19:32.52J4k3"If you want to have sexual relations with the system administrator, press 8"
19:32.57Carp1lol
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19:33.07*** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal)
19:33.07Carp1besides GS lol.
19:33.07*** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal)
19:33.08[TK]D-FenderGrandSuk should be avoided with extreme prejudice.
19:33.24[TK]D-Fendercarp : Do you/are you planning on PoE?
19:33.45Carp1No, not for this phone.
19:33.55J4k3POE is the art of breaking out two pairs of the cat5, wiring them to a 2.1x5.5 coax power connector, and plugging it in
19:34.10J4k3of course, grandsucks run at 5V, but only 400mA so any sort of reasonable wire length will be acceptable.
19:34.13[TK]D-FenderJ4k3 : Its a cost factor that can change the end-user cost / functioanlity, so it SHOULD be considered
19:34.28[TK]D-FenderCarp1 : Need speakerphone?
19:34.34tim0123Can you set record-in to always to record inbound calls
19:34.44Carp1Not needed, but always nice.
19:34.56[TK]D-Fendertim0123 : thats all done in your dialplan.
19:35.17J4k3[TK]D-Fender: cheap PoE injectors will work "backwards" then all you need is a 2.1x5.5 <-> 2.1x5.5 connector between the "injector" ("dejector"? "Unjector?") and what you want to power.
19:35.20[TK]D-FenderCarp1 : Tell you what, if you want a great general purpose phone you won't regret, get a Polycom IP 501.
19:35.43Carp1Thanks....I'll check it out now.
19:35.58[TK]D-FenderJ4k3 : not quite that easy.... phones do NEGOCIATE PoE, not just throughing voltage down the line.  thats liable to fry non-poe gear...
19:36.12tim0123SO if your monitor a queue how do you keep from monitoring hold time
19:36.18EmleyMoorHow many different ring cadences do you lot use on your Zap phones? (if you have any)
19:36.48J4k3[TK]D-Fender: depends on the PoE implentation.  theres "active" and "passive" POE.  Active POE is the art of shoving power over the data pairs, passive is the art of using the "unused" pairs (which don't exist in a gig-e environment)
19:37.03J4k3if you've got designated cat5 outlets for your phones, passive PoE is perfectly safe.
19:37.18J4k3(else every WISP on earth would be burning down their customers' houses)
19:37.43EmleyMoorI will design my new house's wiring around allowing PoE
19:37.47J4k3the largest problem is these device manufacturers insisting on doing silly crap to the non-data pairs...   either short 'em, or leave 'em open.
19:37.58Carp1the 501 is also wireless?
19:38.50[TK]D-FenderCarp1 : Nope.
19:39.01Carp1Must of looked at the wrong one lol.
19:39.05J4k3oh, and a silly note before I go take a shower and head out for the weekend:  The UT Starcom F1000G seems to work extremely well on g711-alaw, and thats it... :P
19:39.12Carp1http://www.google.com/url?q=http://www.nextdaypc.com/main/products/details.aspx%3FPID%3D2354976%26rsmainid%3DND0130014&fr=AEo97MJo6FzKlIF0QeagPoM2Pvy_01s1pDW5XPcxqXqwllOqju7NVzQAAAAAAAAAAA&sa=X&oi=froogle&ct=result&cd=1&usg=__igW_a3YmCAYcQQIHOAFVTlZqQ78=
19:39.13[TK]D-FenderJ4k3 : Most of the world seems to think 802.3af ... but then again I might be crazy ;)
19:39.31J4k3[TK]D-Fender: 802.3af lists both passive and active.
19:39.43J4k3the only folks pushing active is Cisco.
19:39.47[TK]D-FenderWTF, wireless?  NEWS TO ME.
19:40.19[TK]D-FenderJ4k3 : AND THATS 48V, NOT 24V ....
19:40.24fetcherJ4k3: even an "active" (negotiating) 802.3af device is supposed to accept power on either the spare (blue/brn) or data pairs, if it's really standards compliant
19:40.36fetcherJ4k3: although in practice many don't
19:40.42J4k3[TK]D-Fender: actually passive can be any voltage you want/need.  Thats not actually in the standard.
19:41.11*** join/#asterisk topping (n=topping@h-67-100-91-18.snfccasy.covad.net)
19:41.11J4k3fetcher: yeah.  Technically they should work either way depending on the environment.  Then you meet Cisco.
19:41.13J4k3;)
19:41.40fetcherCisco was selling pre-standard PoE equipment for a while with opposite voltage polarity
19:41.52J4k3Motorola Canopy uses the same f'd up method.
19:42.11J4k3never ever accidentally use a canopy supply on anything else, unless you're in the mood to release smoke.
19:42.31J4k3(the better move is... never use motorola anything)
19:43.13J4k3but yeah.  I blew off PoEing the office when I figured out I could keep each phone up >12 hours with a little 4.5A/12V SLA and a cheap-o charging circuit.
19:43.44J4k3total investment of about $15/phone.
19:44.52fetcherJ4k3: plan on having to replace all those SLAs in a few years after they go bad, though
19:45.00J4k3fetcher: that goes for any sort of battery backup device.
19:45.32J4k3the scary part is some of those UPSes are up to replacement #3
19:45.40J4k3some *OLD* UPSes :)
19:45.59fetcherit's disappointing how little improvement lead-acid batteries have made in 100+ years
19:46.02hadsThat's getting oldish
19:46.22J4k3well...  at least they're the most recyclable battery technology
19:46.29*** join/#asterisk s1gny|wrk (n=s1gny@p549148C8.dip.t-dialin.net)
19:46.30mercestesWe recently replaced all teh batteries in our UPS's.  We had to retrofit a bunch of deep cycle marine batteries because they dno't make those style of batteries anymore.
19:46.39*** part/#asterisk s1gny|wrk (n=s1gny@p549148C8.dip.t-dialin.net)
19:47.00J4k3mercestes: no shame in that, just make sure they're properly vented
19:47.05anonymouz666why does asterisk insist with 407 even with insecure=very? very very strange
19:47.08J4k3unless you paid the insanely big bucks for deep cycle marine SLAs
19:47.45fetcherhttp://www.electrifyingtimes.com/firefly_energy.html
19:47.59fetcher^^^ these should be interesting, when they finally hit the market
19:48.02J4k3they're trying to push SLAs for marine use around here...  apparently the fish and wildlife service doesn't like idjits flipping their boats over in the lake.
19:48.41J4k3its all nonsense til I can buy it ;)
19:48.44mercestesThese were for computerstho...lol.  they were so old they had the "inverted" reciptacles for the old style monitors that plugged directly into your CPU power supply
19:49.07fetcherJ4k3: they should work as a drop-in replacement for SLAs, though.  Same cell voltage & charging characteristics
19:49.52fetchermercestes: those were nice to have.  Annoying that ATX left off the pass-through plug...
19:49.57J4k3I find it funny they're mentioning the worst brand in all of lawn equipment (Electrolux/Husqvarna/Poulan)...  if you find a 10 year old+ poulan or husq part, its WONDERFUL. Everything they sell today is shit.
19:50.03J4k3now they want to sell... battery operated electric shit.
19:50.45J4k3rule #1 with battery technology: You don't work with other vendors.  You release the technology for sale *THEN* work with them.  I find this article sketchy at best.
19:51.12[TK]D-Fenderfetcher : Thats why I've switched all my power backup needs to Mr. Fusion :)
19:52.35rudholm[TK]D-Fender: I hope that's not partially hydrogenated fat!
19:53.13J4k3/you
19:53.25anonymouz666can I use insecure=very on [general] sip.conf?
19:53.33anonymouz666Does not seem to work
19:53.47anonymouz666asterisk is still asking for auth
19:54.58*** join/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl)
19:55.01[TK]D-Fenderrudholm : No, I'd never do anything half asses like that.  its FULLY hydrogenated ;)
19:55.13fetcherrudholm : No, I'd never do anything half asses like that.  its
19:56.18fetcherbleh
19:56.19[TK]D-Fenderassed*
19:56.19fetcherhydrogenated fat == more deuterium? :)
19:56.19mercesteshalf asses?  say it ain't so!!
19:56.20rudholm[TK]D-Fender: ok, fully hydrogenated is ok, since then it's not a trans fat
19:56.57*** join/#asterisk gambolputty (n=gambolpu@cblmdm72-240-185-4.buckeyecom.net)
19:57.24gambolputtyHi.  Anyone heard of a company named "veras" that makes TDM switches?
19:57.30rudholm[TK]D-Fender: but fully hydrogenated fats have a consistency approximating candle wax
19:58.57[TK]D-Fenderrudholm : time to crack up the BTU's :-)
19:59.19[TK]D-Fenderfetcher : No extra neutrons required
20:00.29anonymouz666anyone have an idea why asterisk is asking for auth even with insecure=yes?
20:00.34anonymouz666insecure=very
20:04.22*** join/#asterisk Mad|Cow (n=thirt@74.92.109.205)
20:06.32Mad|CowHas anyone ever experienced any issues using Asterisk realtime with voicemail pins? When I update my pin, it doesnt update my database (so the pin never gets updated). Any pointers on where I might look?
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20:09.39Bobthehunterso waht should it be .. SIP/USER@HOST ? or SIP/HOST/USER
20:13.14hadsYes
20:16.14Carp1first
20:16.24J4k3you're about to enter an echo test!
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20:25.29CunningPikeping Strom_C
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20:35.48[TK]D-FenderBobthehunter : LAST.
20:36.14zotzanyone know of an iax2 client that will run with jacmd?
20:40.41zotzsorry with jackd?
20:41.55gambolputtysip/user@host
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20:44.41Bobthehunterkk
20:45.07Bobthehunterwell asterisk likes beter sip/host/phone
20:45.16Bobthehunters/phone/did
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20:47.03gambolputty<PROTECTED>
20:47.19gambolputtythat's what I am referring to for a command like sip/user@host
20:47.42Bobthehunteryeah
20:47.47Bobthehunterso sip/USER@HOST/DID
20:48.21gambolputtynot sure on the /DID part
20:48.26gambolputtyrefer to the dial command
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21:03.25Defendlooking for some sugestions i want to do multiple operators for the voice mail system when some one presses 0 to do this i think i should do it based off of context for that person but how would i do the correct o extention can you use a gotoif statement to exec a dial statement?
21:05.01[TK]D-FenderDefend : Its just an exten that gets called in the same context as Voicemail is called.  You can whatever you feel like from that point.
21:05.17*** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net)
21:06.13Defendoh so if i change my extentions to break them up to 2 diffrent contexts i can just create an o extention in each context and it will do it correctly?
21:08.24[TK]D-FenderDefend : I didn't say VOICEMAIL context.  I said in your DIALPLAN.
21:08.45Defendthats what i ment
21:08.59[TK]D-FenderDefend : Or if I interpret that differently and you intend to do your dialplan in 2 segments, yes.
21:09.02*** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl)
21:09.21[TK]D-FenderDefend : Though it may be cleaner to do conditional processing in a single place.
21:09.31Bobthehunteranother questions
21:09.43Bobthehunterpark send call back to s,1,.. wich is unbelievably bad
21:09.49Bobthehunteron timeout
21:10.06Defendi am trying to make it so if a person calls a certain dept and press 0 in vm they will hit that depts queue instead of going back to main operator
21:11.04[TK]D-FenderDefend : If you use a single mutli-function macro you can add a parameter for where you'd like the call to go afterwards, or you could just create a second version of things.  Either way works.
21:11.05*** join/#asterisk Gr1ncheux (n=devine@unaffiliated/gr1ncheux)
21:12.10Defendhmm i will read up on macros then maybe cause i would prefer to keep it dynamic as much as possible so i dont have to over duplicate everything
21:24.40ThoMehello
21:24.53ThoMei have asterisk with misdn and a digium card with 4 ports.
21:25.12ThoMeif i ring from external to $msn1 port 1 then ringing my phone
21:25.14ThoMeon port 2
21:25.28ThoMei can from 2 > 1 talk and 1 hear me
21:25.41ThoMebut 1 can not talk to me.. i can't hear what the 1 said.
21:25.51ThoMecan anybody help me please?
21:28.15*** join/#asterisk fall0ut (i=tim@realfuckingnews.com)
21:31.27ThoMeah bridging=no
21:32.52CrazyTuxThoMe, 'kiss me' ?
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21:33.27ThoMeCrazyTux: why?
21:33.38CrazyTuxThoMe, is that what your name means?
21:33.43ThoMeno
21:33.46CrazyTuxah k
21:33.57CrazyTuxI'm pretty sure thats what it means in spanish
21:34.21ThoMe:-)
21:34.59*** join/#asterisk BSDTech (n=RNeese@ppp-69-238-49-231.dsl.irvnca.pacbell.net)
21:36.54*** join/#asterisk i3inary (i=i3inary@ip68-8-91-87.sd.sd.cox.net)
21:38.56i3inaryhey there guys i was wondering if someone could help me out with asterisk 1.4.0 and configuring cdr to write to mysql ...i have followed all of the steps i could dig up on the voip-info.org wiki and i am trying this channel for support next
21:41.28i3inarywhat i know so far is that master.csv is writing properly...but after i configured mysql and added what i thought to be the proper module in modules.conf i still have no records in my table
21:47.28MoobiusI've been interested in implementing this. Is the voip-info.org wiki the best howto there is?
21:49.08i3inaryare you referring to asterisk in general or cdr writing to mysql?
21:49.53[TK]D-Fenderi3inary : Did you compile asterisk-addons as well?
21:50.44*** join/#asterisk Ebola (n=Ebola@host81-151-91-139.range81-151.btcentralplus.com)
21:50.54i3inaryyes i think i did.  i actually used elastix.org's install but i did a ./menuselect/menuselect and i was able to see alot of add-ons with "*" next to them
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21:51.26[TK]D-Fenderi3inary : I can't speak for your means of installation....
21:52.08JoNatei don't very much like * right now
21:52.45synthetiqsofaswitch.org/d/
21:53.02[TK]D-FenderJoNate : Thanks for sharing .... the dorr is still revolving... you can leave at any time...
21:53.11JoNateno no no...
21:53.27JoNateI like * very much
21:53.53JoNateit's just that i'm new to it all...and I've been banging my head trying to figure out why MeetMe wouldn't work...
21:54.00JoNateonly to realize it was never installed
21:54.03[TK]D-FenderJoNate : You have some serious issues with conradictions... perhaps you should see a specialist about that...
21:54.18[TK]D-FenderJoNate : That would be a good tip-off :)
21:55.09JoNateheh...considering I've never used linux or * before and I've got working phones in just a few days, i'm kinda happy...But damn theres alot of stuff to learn
21:55.36[TK]D-Fender"with great power comes great responsibility" - Uncle Ben.
21:56.27JoNate"with great responsibility comes great gray hair" - JoNate
21:57.10JoNateAnd why the hell is Uncle Ben talking about that...He makes rice for god's sake...
21:58.38i3inary[TK]D-Fender: i just reloaded * server and i see the following modules regarding cdr loaded:
21:59.00i3inary<PROTECTED>
21:59.03ThoMecan i send the name and the number to the isdn phone?
21:59.13i3inary[Feb  2 13:52:05] NOTICE[32356]: cdr.c:1092 do_reload: CDR simple logging enabled.
21:59.17ThoMeIF this phone direct on the asterisk (sample: hfc card) connect is?
21:59.28i3inary<PROTECTED>
21:59.35i3inary<PROTECTED>
21:59.36J4k3hey, instant rice IS a powerful thing!
21:59.43[TK]D-Fenderi3inary : www.pastebin.ca
21:59.49[TK]D-Fenderi3inary : do not spam in hree
22:00.11i3inaryok let me check that out...this is first mirc experience..i was wondering how you deal with that
22:00.32[TK]D-FenderJ4k3 : Instant rice is GARBAGE.
22:00.56[TK]D-FenderJ4k3 : whole grain & wild rice.  the only way to fly.
22:01.04JoNatemmmmm wild rice
22:01.10JoNatewith soy sauce...
22:01.13J4k3[TK]D-Fender: well its powerful if you dry it out, dust it, float it in the air and catch it on fire.
22:01.15JoNateand chicken...
22:01.17J4k3ie - grain explosion :)
22:01.30[TK]D-FenderJoNate : A sure-fire sign that you're WHITE.
22:01.41JoNatewhy?
22:01.48Bobthehunterhey wall wahts the Frist Data MErchant Services FDMS rival you guys know ?
22:01.55JoNateI happen to be african-armenian
22:02.18[TK]D-FenderJoNate : Sorry, a generalizaion for NON-ORIENTAL.  My bad.
22:02.41*** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm)
22:02.46JoNateWhy does that make me non-oriental? cause I like soy sauce?
22:02.49hadsGrain explosions are cool
22:03.00[TK]D-FenderJoNate : But to elaborate is is what is typically considered "white-man food" by them.  Just like General Tao, and so much else.  its Americanized.
22:03.12JoNateahhhh
22:03.26[TK]D-FenderJoNate : Just like North American pizza is a bastardization of its true Italian roots.
22:03.34[TK]D-Fender<- Culture nazi :)
22:03.39JoNatemuch the same way as Chicken Parmesan...
22:03.51JoNateconsidering it doesn't use Parmesan...
22:04.05*** join/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl)
22:04.27[TK]D-FenderJoNate : Some do...
22:04.40JoNatesure...but nothing you'd order here in the states...
22:04.44i3inaryFender: thanks for that link.  here are my modules loading http://www.pastebin.ca/337566 ...as you can see obviously im missing mysql...so i think i need to find out where to get that
22:04.50JoNatethey come covered in Mozzerella...
22:04.54[TK]D-FenderBut is the overall dish ethnically indicative?
22:05.08JoNatethose are big words and I'm a small man...
22:05.15JoNateyes, it tastes good...
22:05.20JoNateno it's not true to it's name...
22:05.54[TK]D-FenderJoNate : For the sake of truth in advertising, they probably added a sprinkle :)
22:06.08JoNateI thought that Pizza was invented in China
22:06.22[TK]D-FenderJoNate : No, that'd be pasta.
22:06.36[TK]D-FenderJoNate : Oriental society was never big on bread
22:06.59JoNateI thought I read it somewhere...Oh well...
22:07.24JoNateI'm hungry now
22:07.31JoNateI want sashimi!
22:07.52*** join/#asterisk Crescendo (n=martinda@adsl-072-151-080-148.sip.rmo.bellsouth.net)
22:08.51*** join/#asterisk sahafeez (n=sahafeez@67.109.14.227.ptr.us.xo.net)
22:08.54[TK]D-FenderJoNate : thats on my tp-learn list.  I already make my own maki, but only with the nori on the outside of the roll.  I need to practice the inverse stlye, and then expand.
22:09.06*** join/#asterisk X-Rob (n=Rob@ppp214-210.static.internode.on.net)
22:09.44JoNateyou must be one with the fish young padawan...
22:11.13[TK]D-FenderJoNate : Actually I get to laugh at the entire martial aspect of the Jedi as I've been learning Katori Shinto for the last year :)  My blade may not be of focused light, but it still cuts pretty well :)
22:13.31J4k3if your blades so good, mow my lawn with it
22:13.32J4k3;)
22:14.33[TK]D-FenderJ4k3 : After you're done scrubbing my floor and bathroom!
22:14.50*** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn)
22:15.27anonymouz666i can't believe that * still ask for auth with insecure=very !!!
22:15.28anonymouz666ahhh
22:15.31anonymouz666:S
22:17.48mercestes>.>
22:18.24mercestes:D
22:18.31mercestesanother satisfied customer.
22:18.43*** join/#asterisk s1gny|wrk (n=s1gny@p54917DB3.dip.t-dialin.net)
22:19.13[TK]D-FenderJ4k3 : MEDIUM?!?!!?
22:19.17[TK]D-Fender<- Carnivore
22:19.24*** part/#asterisk BSDTech (n=RNeese@ppp-69-238-49-231.dsl.irvnca.pacbell.net)
22:19.35J4k3yes.  I'm more a medium-rare man, but...  I don't know where either of them have been so I want to be sure to cook the bugs out of them ;)
22:19.37*** join/#asterisk jeebusroxors (n=jeebusro@cpe-75-80-248-142.dc.res.rr.com)
22:19.41Bobthehunter[TK]D-Fender, RAW
22:19.41Bobthehunter;0
22:19.56[TK]D-FenderThere is only ONE kind of beef, and that filet Mignon.  It is to be served blue & seared.  Everything else is a FARCE!
22:20.01J4k3mercestes is not USDA Approved meat!
22:20.14[TK]D-Fendermercestes : the other white meat.
22:20.51J4k3oh...filet mignon?
22:20.52JoNatei need raw...
22:21.10hadsYeah, Fillet is the only beef I usually eat.
22:21.16JoNateraw raw raw...matter of fact...i often ask if they can just bring the cow to the table...
22:21.20[TK]D-FenderJ4k3 : Tourandos!
22:21.22J4k3I prefer ribeye steaks off the grille
22:21.28hadsNot blue though, I like mine cooked.
22:21.36JoNatei like carpacio...
22:21.36J4k3but I'm also a texan, and will likely die of complications of heart disease before the age of 70.
22:21.39JoNateit's delicious...
22:21.52[TK]D-FenderJoNate : Yeah.... bring me the whole damned cow.  I'll carve off what I want.. and RIDE THE REST HOME!
22:22.49JoNatewhy carve anything...just bite into it like a man damn it...
22:22.57J4k3haha
22:23.20J4k3you know, they used to castrate bulls (steers) with their teeth around these parts.
22:23.35J4k3cowboys were a strange bunch.
22:23.36JoNatethats cause your all a bunch of crazies...
22:23.50*** join/#asterisk BSDTech (n=RNeese@ppp-69-238-49-231.dsl.irvnca.pacbell.net)
22:24.03JoNatesee...now I could totally respect GWB if he was crazy enough to do that...It would all make sense...
22:24.12JoNatebut he's got to prove to us that he's that nuts...
22:24.14J4k3well, you've gotta be a pretty hardcore mofo to stick your head between a bull's legs and bite... there...
22:24.19JoNateerr...don't mind the pun...
22:24.51i3inaryquestion: i just ran make in the asterisk-addons-1.4.0 dir and i had the following...should i worry about it? make: *** [config.status] Error 1
22:24.51JoNateyeah for the blue!
22:25.11J4k3its not easy being a minority.
22:25.23*** part/#asterisk BSDTech (n=RNeese@ppp-69-238-49-231.dsl.irvnca.pacbell.net)
22:25.27JoNatewhat do you think? Clinton? Obama? Edwards?
22:25.34J4k3ABB.
22:25.38J4k3(Anybody But Bush)
22:25.44JoNateI thikn Hilary should run with Bill as her Vice...
22:25.55J4k3can't, Bill's got a felony on his record now.
22:26.00JoNatethen she can step down and let the Boy back in the office...
22:26.24JoNatebah...
22:26.27mercestesj4k3:  What part of Texas?
22:26.33J4k3mercestes: near Crockett.
22:26.46J4k3eastern.... amongst the pine trees and weirdos.
22:27.06*** join/#asterisk ping2921 (n=marc3234@206-248-130-152.dsl.teksavvy.com)
22:27.08ping2921Hi,
22:27.15mercestesI'm SE Tx
22:27.16mercestes:)
22:27.18J4k3ahh
22:27.21mercestesyeaa
22:27.31mercestesamongst the pine trees and wierdos
22:27.38ping2921anyone knows how to do a "click to call" that would work with asterisk?
22:27.43J4k3are you pimp c or bun b?  hahaha
22:28.25mercesteshuh?
22:28.42mercestesuhh.....whichever one is the straight one.
22:29.04J4k3oh...  the only thing I know about anything farther east than Houston is... about port arthur rappers, for some unknown reason.
22:29.05i3inaryping2921: i am actually working on a click to call application ....my project is http://click2voice.com
22:29.20EmleyMoorAnyone But Blair/Brown
22:29.30mercestesGalveston.
22:29.44mercestesThere's more eastern stuff farther north
22:29.45EmleyMoorMind you, not keen on Cameron either
22:29.46J4k3mercestes: ahhh!  not bad.
22:30.00J4k3my ex-gf was from Texas City
22:30.02*** join/#asterisk roooaaar (n=helpme@eljakim.xs4all.nl)
22:30.04mercestesbut, we call that lousiana
22:30.21J4k3I spent as little time as possible there.  if I'm driving that far south, I'm not going to spend my time chewing the air when I get there.
22:30.42mercestesYea, I wouldn't mind leaving tx myself.
22:30.44J4k3then theres that damned restraunt that I kept going to... with the bigass crab on the roof... on seawall
22:30.45J4k3guidos?
22:30.58mercestesJoe's Crab Shack?
22:31.01J4k3neg
22:31.04Hmmhesaysanyone in here use squid-cache at all?
22:31.04J4k3f joes crab shack
22:31.07mercestesGuilianis?
22:31.11roooaaarHey guys, can you help me out? Which hardware should I order to hook up an Asterisk server to 2x ISDN lines (Netherlands) ?
22:31.13J4k3yeah that might be it
22:31.26J4k3its been about 5-6 years
22:32.03mercestesroooaaar:  Sangoma/Digium ISDN card interface.
22:32.13mercestesroooaaar:  Just check for ISDN compatibility
22:32.33mercestesThe T1card I had from Sangoma did ISDN if I recall correctly but it wasn't specifically an ISDN card.
22:32.44hadsDepends it it's BRI or PRI
22:32.49roooaaarmercestes: does it also recognize that the two ISDN lines are 'bundled' (don't know the exact English term for this, basically I have 4 paths under the same numbers)
22:32.59hadsBRI
22:33.03JoNateI'd try Mer Cest's T1 Telepathy Card...Inexpensive and super reliable...
22:33.04mercestesYea, i was about to say, unless you mean BRI/PRI
22:33.07*** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal)
22:33.08*** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal)
22:33.32hadsSo you would need either a Digium B410P or an Eicon Diva Server or whatever
22:35.23roooaaarGreat! Thank you. So B410P + Business Edition = 2000$ will get me going, right? I have a 19" server (Dell, 2Gb, 80Gb) lying around, so - except for the clients - I should be ready?
22:38.35*** join/#asterisk jm|home (n=jm@zen.jamiem.com)
22:39.44mercestesWhy business edition?
22:40.11mercestesYea, except for the clients you should be good to go.
22:40.40roooaaarCause we build OS software ourselves and know that sometimes it's nice when people order your stuff + pay for some support.
22:41.12mercestes;P  Yea, I retracted the question.
22:41.23mercestesbut I admire your ethics/support.
22:41.25J4k3good people
22:41.30mercestes(which is why I retracted the qusetion).
22:41.48De_MonGood grief! the 1.4.0 dialplan reminds me of when I switched from dos to linux CLI
22:42.01De_Mon*commandline interfaice*
22:44.04defendya how id 1.4
22:44.14defendis there any docs about features and changes up?
22:44.47defendi have been wanting to try it but at the moment i dont have any extra computers :/
22:44.53[TK]D-Fenderdefend : check the changelog, and I believe there is an upgrade.txt or the like included with the tarball
22:45.06anonymouz666when forwarding a request from openser to asterisk... asterisk get the invite and ask for auth... but I have insecure=very in general.. I configure UAC module to auth an user in asterisk... but asterisk still answer 407 proxy authorization
22:45.29anonymouz666I am fighting with that heh.
22:45.55mercestesanonymouz666:  What ver of asterisk?
22:46.26[TK]D-Fenderanonymouz666 : allowguest=yes
22:46.36anonymouz6661.2.13
22:46.49mercestesGo D-Fender
22:46.51mercestesI was jus tlooking that up
22:47.21mercesteshttp://www.voip-info.org/wiki/view/Realtime+Integration+Of+Asterisk+With+OpenSER
22:47.42anonymouz666[TK]D-Fender I got that in my sip.conf
22:47.53[TK]D-Fenderanonymouz666 : and set a context?
22:48.44anonymouz666yes... the context is sip-sip.
22:49.48anonymouz666* ask for auth... and I send a valid user... through OpenSER
22:49.53anonymouz666* does not seem to accept that
22:53.54anonymouz666I just would like to know why * ask for auth in [general] sip.conf
22:54.00anonymouz666with insecure=very etc etc.
22:54.05anonymouz666does not make sense to me
22:55.39anonymouz666I am using UAC module and blah blah blah just complicating more and more
22:55.45*** join/#asterisk angler (i=angler@pdpc/sponsor/digium/angler)
22:55.55anonymouz666for a simple thing
22:56.16*** join/#asterisk l2cache (n=ghansen@64.128.254.98)
22:57.22*** join/#asterisk MonkeyHugs (n=jojo@63.149.122.93)
23:00.52*** join/#asterisk lba (n=lba@user-12lml5g.cable.mindspring.com)
23:01.26*** join/#asterisk russellb (i=russellb@asterisk/developer-and-stable-maintainer/drumkilla)
23:01.26*** mode/#asterisk [+o russellb] by ChanServ
23:01.53l2cachei need to do a show a "show queue queuename" and have allison read off the agent extensions in the queue
23:02.08l2cacheany idea how i could go about programming that in the extensions.conf?
23:02.30*** join/#asterisk cchristianDraco (n=cchristi@adsl-067-034-103-184.sip.mia.bellsouth.net)
23:02.38cchristianDracohi
23:02.47cchristianDracoI have two questions
23:03.33cchristianDracoone: does Asterisk run on Windows XP, even if I removed alot of the annoying overlays and stuff?
23:04.00cchristianDracotwo: is it easy to use it to talk on a Normal phone line?
23:04.00lbaIs there any way to have a 'wild card' in {expression}?  Like _4XX for extension numbers?  exten => s,n,GotoIf($["${expression}" = "Private"]?SetCIDNum(NOCID),s,1)
23:04.25Strom_CcchristianDraco: first: maybe, but why the hell would you want to
23:04.32cchristianDracouhh
23:04.40Strom_Csecond: you can interface to a phone line with the appropriate hardware
23:04.43cchristianDracobecause I need it to be on Windows
23:04.56cchristianDracoI have a phone hookup port
23:04.59Strom_Cwhy must it be on windows?
23:05.32cchristianDracoso I won't go into it
23:05.52cchristianDracoI just need it to be on windows
23:05.57Strom_Ci don't intend to start drama; i'm just curious what the reason is
23:06.10[TK]D-Fender<PROTECTED>
23:06.14Strom_Cis this for yourself, or is it for multiple users?
23:06.16cchristianDracoWhat else would I use?
23:06.37alrscchristianDraco:  It shouldn't be too much to scrape up a 500mhz p3 somewhere and put the linux on it.
23:06.54Strom_Chell, you can get 500mhz p3 boxes for free half the time
23:07.01cchristianDracoLinux isn't good to me...
23:07.12[TK]D-FendercchristianDraco : Care to elaborate on "phone hookup port'?
23:07.18cchristianDracohmm
23:07.22Strom_Che probably means "cheaptacular modem"
23:07.32alrscchristianDraco:  It isn't 1996, you don't need the stack of floppies
23:07.40cchristianDracoit's a telephone cable thing
23:07.41[TK]D-FenderStrom_C : I want evidence before I execute :)
23:07.43mercestesif linux is too hard........then asterisk is probably not for you
23:07.43alrscchristianDraco: or the $20 Infomagic 6-CDROM set
23:07.46l2cachei dont think so, if i do a system show queue salesfloor | grep SIP/ and work with sed  i believe i can get it to return the extensions back to the cli for reading by allison
23:07.57Strom_CcchristianDraco: in all honesty, if linux is a channelge, then asterisk is probably not for you
23:08.00Strom_Cer
23:08.01Strom_Cchallenge
23:08.07Strom_Ci'm going dyslexic
23:08.07cchristianDracoin the sie of this computer
23:08.12[TK]D-FendercchristianDraco : Congratulations, typically the term 'telephone cable thing" = useless junk as far as Asterisk is concerned
23:08.24cchristianDracook
23:08.38mercestescchristianDraco: Just shop for cheaper phone service.
23:08.40cchristianDracojust answer my fucking questions Linux fuckers
23:08.45mercestes...
23:08.46cchristianDracough
23:08.47alrs:)
23:08.48*** part/#asterisk cchristianDraco (n=cchristi@adsl-067-034-103-184.sip.mia.bellsouth.net)
23:08.48[TK]D-Fenderl2cache : And how do you get variable data BACk to your dialplan?
23:08.59mercestesAww.
23:09.03Strom_Cwhat a charming young man
23:09.03mercesteswe hurt his feelings.
23:09.03l2cachegood question...
23:09.04[TK]D-Fenderl2cache : though yes, you can do BASH w/o AGI to GET the info.
23:09.32l2cacheany ideas?
23:09.45[TK]D-Fenderl2cache : AGI :)
23:09.45mercestessee.....that was me...years ago....and back then, as I stormed out of IRC crying.......i never imagined I would ever be on *this* side of the fence.
23:09.51mercestesand in retrospect.....man was I retarded.
23:09.55l2cache:) thanks man, you think thats the only way???
23:10.11[TK]D-Fendermercestes : Doesnt' take much to realize when YOU are the problem.  He's still simply in denial.
23:10.54[TK]D-Fendermercestes : Plenty of people with 1D10T errors around here...
23:11.10[TK]D-Fendermercestes : waspoint pointing at "you" btw...
23:11.10mercesteslol
23:11.14[TK]D-Fenderwans't
23:11.14[TK]D-Fenderashdkjshd
23:11.16Strom_Cthere's pebkac, and in the phone world, there's also pebdac
23:11.18[TK]D-Fender*blarg*
23:11.21mercestesAww..:)  I know.
23:11.26mercestesyou like me.
23:11.26Strom_Cproblem exists between dial and chair ;)
23:11.47[TK]D-FenderStrom_C : No... he didn't get far enough along to DIAL anything :)
23:12.00Strom_Cyes, i wasnt referring to him
23:12.10*** part/#asterisk l2cache (n=ghansen@64.128.254.98)
23:12.14Strom_Cim referring to the situations when users call me out for major problems that turn out to be them dialing the wrong number or something
23:12.52mercesteswhich happens remarkably often.
23:13.13Strom_Cyes
23:13.16mercestesyou even *touch* the phone system and omg...every click on the line is somethign you did wrong.
23:13.17*** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm)
23:13.47Strom_Cit's fun billing for an hour of travel and an hour minimum of consulting to show up and say "dial carefully"
23:14.24mercestesomg...I dailed the main number, and diled this person..and THEY DIDN'T ANSWER!!! so obviously their phone is not ringing....so I hung up and hit redial...and I got a BUSY signal!  (10 digits +4 digit extension on redial).   This is unacceptable!  *scoffs*
23:15.30lbaIs it possible to use a wild card in GotoIf to change CALLERIDNUM on all extension numbers in the _4XX range?
23:15.39Strom_Csure
23:15.52lbaStrom_C: me?
23:15.57Strom_Cno, the one-eyed monster with the reorder problem
23:15.59lbaStrom_C: How would I go about that?
23:16.06lbaStrom_C: sri
23:16.11mercestes...
23:16.31mercestesyou would......use a wild card in a goto if to change the calleridnum on all extension numbers in the _4xx range.
23:16.38Strom_Cmaybe something like GotoIf($["${EXTEN:0:1}" = "4"}]?label)
23:16.40mercestesyour question had enough information that it was pretty clear exactly on how you intended to do it.
23:17.25Strom_Cbut actually, i think with proper numbering plan engineering, you wouldnt have to use the gotoif
23:17.35lbaActually, I don't really know the proper way to do it.  Can anyone suggest how?
23:17.55[TK]D-FenderGotoIf($["${CALLERID(num):0:1}" = "4"}]?label)
23:17.57lbaStrom_C: Can you elaborate?
23:18.02Strom_Cwhat are you trying to accomplish, really?
23:18.33Strom_CGotoIf($["${CATSEX}" = "${DOGBALLS}"]?dead_hookers)
23:18.38lbaStrom_C: I have Grandstream sip phones that permit a distinctive ring based on CID number.  So I want my Internal calls to have a DR.
23:18.57lbaStrom_C: That means I should change the CID from whatever it is to something like Internal
23:19.06perdyou dont want spaces lba
23:19.07mercesteslba:  So why don't you in sip.conf, set the CID to internal extensions.....
23:19.19mercesteslba:  And then on outbound dialing, set teh CID to whatever external number you want?
23:19.32[TK]D-Fenderlba : I just pasted the answer for you...
23:19.32Strom_Cthat makes more sense :)
23:19.32perdspaces in $[] are bad
23:19.40lbamercestes: Because I use those same extensions to make general calls, not just internal calls
23:19.46perdare they not!?! they are.
23:19.56Strom_Cperd: i've never had trouble with them
23:19.59mercestesperd:  spaces in [] are generally ok as far as I know.
23:20.05perdvoip-info said it would return true always
23:20.09lba[TK]D-Fender: Hi.  Where?
23:20.09Strom_Clba: here's how I do it
23:20.10mercesteslba:  ......right....
23:20.10perdmeh
23:20.24mercesteslba:  I actually confronted that in my response.
23:20.27Strom_Cinternal caller ID on phones is just the extension number
23:20.40Strom_Coutbound calls use Set(CALLERID(num)=xxx)
23:20.42Hmmhesaysdoes anyone know how/where internet explorer caches its http auth info?
23:20.46mercesteslba:  I think you don't want help....you want us to write the answer for you, verbatim, in a copy paste format, right??
23:20.47perdhm i guess im on crack
23:20.49perdoh well
23:21.20lbamercestes: Then I have the same problem in reverse.  It's probably easier to filter the 4XX numbers rather than everything.
23:21.25mercestesHmmhesays:  perfectly on topic!  :D  usually in C:/documents and settings/user/Local Settings/Temporary Internet Folder or something I beleive.
23:21.54Strom_Cperd: then use a pair of GotoIf statements to test for a range of values...
23:22.41*** join/#asterisk angler (i=angler@pdpc/sponsor/digium/angler)
23:22.45lbamercestes: No I want to _understand_.  Checking GotoLf for a range of statements isn't anything I've seen before.  If it can be done, that will help.
23:23.04Strom_Clba: use two statements
23:23.09[TK]D-Fender[18:18] <[TK]D-Fender> GotoIf($["${CALLERID(num):0:1}" = "4"}]?label) <- pay attention
23:23.21Strom_Cone that checks if it's less than 500, and one that checks if it's greater than 399 ;)
23:23.22i3inaryFender: thanks for your info i was able to dl and compile asterisk-addons my configurations must have all been correct because after restarting i now have cdr in my cdr table! thanks!
23:23.31mercesteslba:  fair enough...gimme a sec
23:23.34[TK]D-Fender[18:18] <[TK]D-Fender> GotoIf($["${CALLERID(num):0:1}"="4"}]?label) <- slightly better without whitespace...
23:24.00Strom_C[TK]D-Fender: Shhhhhh!  you'll wake the baby!
23:24.19Strom_C:)
23:24.40lbaSo I need three statements.  Two for the range followed by one to test all numbers in the 4 range for starting with 4 - yes?
23:24.53Strom_Cno
23:24.55mercesteslba: GotoIf($[${CALLERID(num)} > 4699 & ${CALLERID(num) < 4800]?True:false)
23:25.09mercestesworks just like that.
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23:25.33lbamercestes: I've never seen a GotoLf like that.  That would solve my problem nicely.  Thanks very much.
23:25.41mercestesNP.  Good luck..:)
23:29.42fetcherIs there a way to see elapsed-time for calls in progress, from the command line?
23:29.45i3inaryi think i may have another issue...if anyone has any ideas i would appreciate it.  i am using an originate command through the manager api to make a sip call to leg#1 and extension command to make a sip call to leg#2....the originate is not writing any information to cdr...i assume that is by design ...so am i going about bridging my 2 call legs together incorrectly?
23:29.57fetcherlike "show channels", but with a duration column?
23:30.29Strom_Cshow channel [channel]
23:30.38Strom_Ci believe it has duration info
23:30.39Strom_Clemme check
23:31.31fetcherStrom_C: it does, mixed in with a few pages of other info... :)
23:31.39fetcherStrom_C: better than nothing, though.  thanks!
23:31.40mercestesoh well, I'm outie...byes
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23:34.03lbaWhen executing a Dial(SIP/xxx) statement, does Asterisk actually go to the sip.conf xxx channel and then to the context or does it do something completely different?
23:35.09Strom_Cwhat do you mean "and then to the context"?
23:35.17lbaUnfortunately, there seems to be nothing like VERBOSE for sip.conf
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23:35.28lbaStrom_C: The context in the xxx sip.conf entry
23:35.32potential1Anyone looking for a dedicated server>?
23:35.46[TK]D-Fenderlba : that isn't a "context", its just a peer entry.
23:36.11[TK]D-Fenderlba : and the "context=" line in there is for where to send calls COMING from the device.
23:36.13i3inarypotential: i may be by the end of the month depending on my dev progress
23:36.15lba[TK]D-Fender: I'm speaking about the "context=whatever" statement
23:36.23[TK]D-Fenderlba : Answered that too :)
23:36.38lba[TK]D-Fender: You mean outgoing calls right?
23:37.01[TK]D-Fenderlba : calls coming FROM the device TO Asterisk
23:37.56lba[TK]D-Fender: OK.  So this means that Dial() never goes to the context mentioned in the sip entry?
23:39.37lba[TK]D-Fender: Specifically, I have a sip.conf extension 400 which has context=default.  When I Dial(SIP/400) does it go to default?
23:39.56stubertlba: yes
23:40.22ThoMemisdn
23:40.23ThoMeFeb  3 00:39:22 WARNING[16671]: chan_misdn.c:4792 chan_misdn_log: Hold not allowed this port.
23:40.27ThoMehow i can ALLOWED this?
23:41.14lbastubert: Tnx.  That is at the heart of my problem.  Both outside calls and internal intercom type calls go to [default] and I mess with the CID setting but they get changed when I Dial.
23:41.52potential1Anyone looking for a dedicated server to host your asterisk?
23:41.57lbastubert: Can I have two contexts?  I read something about that in Future of Telephony but don't understand it.
23:42.06ThoMehow i can allowed hold for misdn?
23:42.15stubertlba: explain?
23:42.41lbastubert: Using Regext you apparently can have two Regext separated by an &
23:42.41stubertLet's make this simple...
23:42.53lbastubert: ok
23:43.21stubertSo, The use of context is basically to isolate different dial plans
23:43.53lbastubert: yes I get that
23:43.56stubertyou don't want your incoming calls from the outside to be able to dial back out of your system (usualy)
23:44.00ThoMehello?
23:44.06ThoMecan anybody help me with isdn?
23:44.18ThoMehow i can set hold allowed for msidn?
23:44.19lbastubert: Well, I have to do that to make calls to other extensions.
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23:44.39stubertThe use of the context= in the sip.conf file is to specify an entry point for that device when it attaches
23:44.47stubertSo,
23:44.54stubertIf all your extensions
23:45.00stubertare in the same context
23:45.20stubertthen should be able to call them from each other
23:45.33[TK]D-Fenderlba : NO.
23:45.48lbastubert: The extensions also have to be able to make outside calls.
23:46.02[TK]D-Fenderlba : when you Dial(SIP/400) it does NNOWHERE.
23:46.04stubertproviding you have a dialplan for that extension in that context
23:46.32lba[TK]D-Fender: What happens to it?
23:46.45[TK]D-Fenderlba : Dial just calls the DEVICE.  there is no more context, you are already implying your in the dialplan based on the incoming context of the CALLING phone.  the context the CALLED SIP device may have is irrelevent
23:47.28lba[TK]D-Fender: Can I test this?  Something like a VERBOSE statement but in sip.conf?
23:48.18[TK]D-Fenderlba : whats to test?  this is like the laws of physics...
23:48.20ThoMeOsse: hi
23:48.24ThoMelba: huhu
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23:49.05[TK]D-Fenderlba : and no there is no statement like that.  What exactly are you trying to figure out here? I thought I was pretty clear about the nature of who's context matters and when.
23:49.51stubertlba: on a call one sec
23:50.14ThoMestubert: u know mdn?
23:50.16ThoMemisdn
23:50.32lba[TK]D-Fender:  "there is no statement like that".    Referring to what?
23:50.55[TK]D-Fenderlba : "verbose"
23:51.17lba[TK]D-Fender: I use VERBOSE all the time in extensions.conf to trace things.
23:51.55lba[TK]D-Fender: But it doesn't seem to work in sip.conf
23:52.16[TK]D-Fenderlba : thats because lines aren't executed in sip.conf like dialplan
23:53.08[TK]D-Fenderlba : the dialplan is an INTERPRETED file.  sip.conf is just parsed once.
23:53.08lba[TK]D-Fender: OK I figured as much.  Parsed once - understood.
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23:53.14ThoMefor dial intern in us company
23:53.23ThoMeexten => _XX.,1,Dial(Local/${EXTEN:2}@intern)
23:53.25ThoMeis it ok?
23:53.33[TK]D-FenderThoMe : BAD
23:53.41ThoMeif i press 2 digets
23:53.46ThoMethen i will intern dial
23:53.47[TK]D-Fenderyou're in your dialplan already, you shouldn't have to be using chan_local like that.
23:54.03ThoMe[TK]D-Fender: bit _XX is ok?
23:54.19[TK]D-FenderThoMe : They should simply have access to [intern] from whatever other context they may default to.
23:54.49[TK]D-FenderThoMe : You shouldn't even have to declare an exten, you should be INCLUDE - ing them from that other context.
23:56.07ThoMehm.
23:57.10lbaIf I declare regext=foo in sip.conf [sip-phone], can I refer to either of them the same way in extensions.conf?
23:57.22ThoMe[TK]D-Fender: and 2 digits is _XX. or?
23:57.30[TK]D-Fenderlba : "regext" is prcatically worthless.
23:57.48[TK]D-FenderThoMe : again, you should even be MAKING an "exte" line at all.
23:57.53lba[TK]D-Fender: What was it intended for?  And can I refer to it?
23:59.11lba[TK]D-Fender: Is it like an alias?
23:59.12[TK]D-Fenderlba : if you make a context with priorities of "2" and up, regexten will create a priority 1 line for it with NoOp in it IF the device is registered.  this means you can prevent the overall exten from being even dialable if they aren't registered.
23:59.25[TK]D-Fenderlba : Which is in 99% of PBX's entirely worthless.
23:59.42[TK]D-Fenderlba : and in the other 1%... only MOSTLY :)

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