IRC log for #asterisk on 20070507

00:00.33Math`but u can use a channel as a modem :P
00:01.31kink0http://www.pastebin.ca/474749
00:12.16*** join/#asterisk infernix (i=nix@unaffiliated/infernix)
00:13.13*** join/#asterisk |dennis| (n=dennis@200.32.233.82)
00:15.19kink0JT did you gotten ?
00:15.38JTkink0: haven't seen your extensions.conf yet
00:15.49kink0ahh ok, one sec
00:16.31JT~thebook
00:16.32jbotrumour has it, thebook is a book called  Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11
00:16.35JThmm
00:17.33*** join/#asterisk sbingner (n=sam@pdpc/supporter/sustaining/sbingner)
00:20.16kink0http://www.pastebin.ca/474780
00:29.58kink0JT ?
00:30.14*** join/#asterisk sonet (n=darrnh@CPE-144-133-204-78.nsw.bigpond.net.au)
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00:30.28hacimjoaovianna: ?
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00:31.21JTkink0: what is this: Dial(Zap/r1/${EXTEN:2}) ;,,M(test))
00:31.29sonethi [TK]D-fender, im working with neville with his asterisk setup, the one generating the glibc error
00:31.34JTsyntax appears erroneous
00:32.11kink0forgot after ;
00:32.51JTi don't think the syntax is right with the space and ;
00:33.09kink0just Dial to zap, group 1, exten minus 2 first digits. That works
00:33.22kink0after the ; is ignored
00:36.01*** join/#asterisk santiago (n=santiago@debian/developer/santiago)
00:36.54JTkink0: you should fix it up though
00:37.52kink0ok, that was remaining for a Macro I used, I can clean these comments
00:40.59Stridernzl[TK]D-Fender - PING!!
00:43.08LeddyHMPONG!!
00:45.00*** join/#asterisk Brijn (n=bas@S010600e0b601c51e.vn.shawcable.net)
00:46.41kink0time for sleep !! Thanks !!
00:51.03BrijnHi all, moved to 1.4, and it now complains that Meetme is not available. Show applicatiuons also doesn't show Meetme.. What could be wrong?
00:56.21*** join/#asterisk sharp (n=sharp@dsl092-234-217.phl1.dsl.speakeasy.net)
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01:00.00*** join/#asterisk sumasuma (n=kurukko@61.14.86.23)
01:00.24sumasumais there is any pcix FXO card for asterisk ?
01:00.44JTyou mean pci-e?
01:02.40kiscokidare there any cheap (100<) fxo cards?
01:03.22JTno, not that are any good
01:05.15sumasumayes, PCIX
01:05.36JTgrr
01:05.36sumasumaPCI-Express
01:05.43JTPCI-Express is NOT PCI-X
01:05.56sumasumai c
01:05.57JTcompletely different slots
01:06.09sumasumait is PCI Express
01:06.16JTsumasuma: digium don't make pci-e cards
01:06.21sumasumaAny FXO Card on PCI Express ?
01:06.22JTsome other companies do
01:06.24sumasumafor asterisk
01:06.25sumasumai c
01:06.27xhelioxsumasuma: Sangoma does. ;)
01:06.29sumasumaSangoma ?
01:06.43sumasumaDo they do for FXO lines or only E1/T1 ?
01:06.57JTboth i think, check their web site
01:07.25xhelioxYeah, both.
01:07.38xhelioxI love Sangoma. Though going with them may not always be the "safest" bet.
01:08.38xhelioxAnd for fear of being killed in my sleep, I won't elaborate. :)
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01:17.45DefrazAnyone try using AudioCodes?
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01:28.09docelmoDefraz once I gave up in 10 minutes..  It was a bigger headache to leard to configure than cisco was the first time
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01:30.12docelmoQuestion..  Anyone here experienced with Asterisk and Comfort Noise?
01:30.22*** join/#asterisk pariah (n=j0sh@unaffiliated/pariah)
01:30.38docelmoI have a customer who keeps having his MOH die out when someone calls into his IVR and gets sent to the queue
01:32.39ghentoHi folks -  I'm wondering if anyone could point me in the right direction - i'm looking to add an overall trigger, if '0' is pressed it will go straight to the operator, no matter what area in the extensions.conf the user is in.  Any help would be much appreciated
01:33.15JTdocelmo: what about CNG?
01:34.12docelmoCompressed Natural Gas?
01:34.26docelmoI have it frequently..  :)
01:34.28JTcomfort noise generation
01:34.58docelmohmm..   I dont know anything about it.  I am looking for a fix to keep the MOH playing like it is supposed to
01:35.17docelmoRight now it fades in and out depending on if someone talks into the phone while on hold
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01:46.31*** join/#asterisk DocHolliday (i=RgRabbit@gateway/gpg-tor/key-0x0E4F6D6C)
01:47.48docelmoJT Any thoughts?
01:48.14JTexplain the call flow
01:48.22JTcalling what to what with what technology?
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01:50.28docelmoPSTN (Having Fading Audio) -> Vendor -> MVTS II -> Asterisk(App_Queue)
01:50.49JTvendor, mtvs ii?
01:50.52docelmoIm trying to kill total support for VAD on MVTS but it still keeps telling me this: process_rfc3389: Comfort noise support incomplete in Asterisk (RFC 3389). Please turn off on client if possible.
01:50.56docelmoMy Toll free Vendor
01:51.02docelmoMVTS II is my switch
01:51.22JTso the fading audio is actually in the other direction?
01:51.35JTcalling from what type of connection to the pstn?
01:52.31docelmoAll SIP
01:52.41JTyou said pstn
01:52.41docelmoThe PSTN is hearing the fading audio
01:53.03docelmoCan I IM you a phone number so you can hear what I am talking about?
01:53.20JTi know what you mean
01:53.26JTyou need to explain the setup better
01:53.30docelmook..
01:53.36JTwhat sort of pstn connection?
01:54.12docelmoCell phone or Home Phone -> Called Toll Free -> Routed to TF Vendor -> MVTS II(My switch) -> Asterisk(Customer's App_Queue)
01:54.29JTcell phone acts the same as home phone?
01:54.32docelmoyep
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01:55.03JTmake sense for it to fade in and out on cell phone
01:55.03docelmoJust fades in and out..  I have done a RTP debug and found that when no audio packets are coming to the asterisk box it doesnt send any of the MOH packets to the caller
01:55.07JTbut landline is weird
01:55.12JTah
01:55.25docelmoI was told this is a CN or VAD issue
01:55.36docelmoI am trying to figure out if I can resolve it in asterisk some where
01:56.27JTdoes the asterisk box have any zaptel timing?
01:56.32docelmonope
01:56.38docelmoI can install it if need be
01:56.57docelmowell ZTdummy
01:57.00JTi think MoH needs zap timing
01:57.18docelmoLet me google..  This is the first time I have heard this
01:57.36docelmoWell its app_queue not so much MOH even tho MOH has the same issue
01:57.43JTmaybe it also has a bug when VAD is turned on, i dunno
01:58.53Strom_Masterisk doesn't support VAD
01:59.00docelmoI know
01:59.04Strom_Mso turn it off and your problems will be (mostly) solved
01:59.15JTStrom_M: it usually works ok through, it just doesn't do CNG when VAD occurs
01:59.17*** join/#asterisk |dennis| (n=dennis@200.32.233.82)
01:59.19docelmoWell thats what I am trying to figure out on my switch
02:00.38DocHollidayany wholesale providers in the channel?
02:01.46docelmoyes
02:01.50skruukIf anybody has successfully built zaptel drivers on rhel4, consider messaging me...
02:02.00docelmoMolten Telecom here what can I do for you?
02:02.19DocHollidaywe started a discussion a few weeks ago over IRC but you ran away :)
02:02.28skruukDoc, good to see you.
02:02.30DocHollidaywould you like to continue it?
02:02.41skruukI dunno, you up for it?
02:02.45docelmoSorry I am always doing 100000000 things..   Like right now this VAD issue for one of my customers..
02:02.49docelmoSure..  Fire away
02:02.50DocHollidayheya skruuk, hows it going?
02:02.57DocHollidaymay i pm you?
02:03.01docelmoIf I dont answer right off its cause I am kicking my switch
02:03.03docelmosure
02:03.03skruukabsolutely.
02:04.03DocHollidayskruuk, i think our conversations got confused :(
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02:16.52[Outcast]is there away to get asteris to always send the Proxy-Authorization with every invite?
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02:25.59dc3aesI am reading that ${TIMESTAMP} is depreciated but the recommended alternative is not working either
02:28.41LeddyHMdepreciated?
02:28.47LeddyHMis that like not appreciated ;)
02:28.56dc3aeshaha
02:29.04*** join/#asterisk ohadz (n=ohad@cpe-69-203-27-50.nyc.res.rr.com)
02:29.18dc3aesdamn. i always spell it that way
02:29.25dc3aesjust had to confirm I was in fact wrong
02:29.34LeddyHMspellcheck
02:29.36dc3aesthe wiki says "${TIMESTAMP}: Current date time in the format: YYYYMMDD-HHMMSS This is deprecated as of Asterisk 1.4, instead use :${STRFTIME(${EPOCH},,%Y%m%d-%H%M%S)})"
02:29.42Qwelldepreciated is a word
02:29.54Qwellit isn't the right word, but it is a word
02:29.56LeddyHMyes I know
02:30.01dc3aesaww i feel a bit better then :P
02:30.01ohadzhi, i'm trying to setup asterisk on my home machine. does anyone know of a good doc/man for setting asterisk ubuntu 7.04?
02:30.05Qwellso then how is a spell check going to help?..
02:30.10LeddyHMI was more making fund at the lack of the correct word he was trying to use
02:30.45LeddyHMspell check also includes grammar check too
02:30.51LeddyHMdepending on your mechanism
02:31.49LeddyHMI am only here for comic relief as I know nothing about *
02:32.03LeddyHMbut I'm not even good at that according too Qwell ;)
02:32.14LeddyHMso I will digress
02:32.18Qwelloh the irony
02:32.21dc3aesyou are making fund of me? lol
02:32.42QwellLeddyHM: according too me, eh?
02:32.44dc3aesgrrrr asterisk has been the most frustrating but rewarding project ive decided to involve myself in
02:33.05LeddyHMhow is it rewarding?
02:33.28dc3aesha
02:33.54dc3aesim trying now to record all the calls as a filename with timestamp and dialed/or/calling party..
02:34.16LeddyHMthat didn't answer my question :)
02:34.51*** join/#asterisk infernix (i=nix@unaffiliated/infernix)
02:35.02dc3aesohadz: I would start here: http://www.voip-info.org/wiki/index.php?page=Asterisk#HowtosandTutorials   but there appears to be tons of heaping piles of contradicting documentation lol
02:35.02LeddyHMI'm actually interested to hear the reason
02:35.31ohadzdc3aes, thanks
02:35.33dc3aesso far my system is working enough that it is useful.. :) does that appear rewarding enough ? lol
02:35.53LeddyHMit means something different to everyone
02:35.57dc3aesohadz: call in sick for the next few days and get a few pots of coffee going
02:35.59LeddyHMI was just curious is all
02:36.00dc3aesya for sure
02:36.02*** join/#asterisk SGM (n=stoyan@213.91.216.130)
02:36.16ohadzdc3aes, i wish:)
02:36.22LeddyHMI only learned it out of necessity
02:36.29SGMhi guys
02:36.31dc3aestheres a factor of necessity here
02:37.18dc3aesI have a home office that also acts as a satellite office of my other office.. so I need a phone system that appears to be one system and acts accordingly.. I also need to uhhhmmm set the callerid since we are a PI firm and sometimes need to modify our identity :)
02:39.06SGMim wondering if there's some way to set environment variables per user/group in sip.conf or iax.conf per user/peer
02:39.47SGMor some better way to do user groupping for accounting purposes
02:39.52SGMexcept accountcode
02:40.00SGMso i can view per user and per group usage
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02:41.06ghentoHi folks -  I'm wondering if anyone could point me in the right direction - i'm looking to add an overall trigger, if '0' is pressed it will go straight to the operator, no matter what area in the extensions.conf the user is in.  Any help would be much appreciated
02:42.36*** join/#asterisk infernix (i=nix@unaffiliated/infernix)
02:44.41LeddyHMour voip provider went awol, and we were having issues, and needed some modifications to boot
02:45.31LeddyHMnot voip, * consultant
02:45.47LeddyHMthey did our implementation and went belly up
02:48.53Hymiedoh
02:50.06dc3aeswow.. got the recording working pretty good... now need to figure out why it records the in/out as seperate files.. I guess because it really is two streams as far as asterisk is concerned
02:50.17groogs[h]ghento: you have to add it to every context manually.  #include would be a good way..
02:50.17docelmoLeddyHM Molten, Inc does Asterisk consult and has dCAP's on staff
02:50.33LeddyHMTK has been assisting us
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02:51.42ghentothanks groogs, i'll take a look at it
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02:52.28docelmoTK?
02:53.50stridernzldocelmo: he means [TK]D-Fender I'm sure
02:53.58docelmoahh ya good guy
02:54.14stridernzldocelmo: he is!
02:55.13docelmoWe have bumped heads a few times on questions
02:55.24docelmowhat I dont know he can ususaly get me unstuck
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02:56.03stridernzlI think he has helped alot of people ..  I know he has helped me alot so certainly i can't speak ill of him
02:56.16MrTelephonewhat would be the problem if your getting overruns and frame errors in ifconfig
02:56.25LeddyHM[TK]D-Fender
02:56.54MrTelephone<PROTECTED>
02:56.54MrTelephone<PROTECTED>
02:57.04JTMrTelephone: your ethernet connection is dodgy, most likely
02:57.15MrTelephonethats for the t1
02:57.35JTMrTelephone: why would the T1 show up in ifconfig?
02:57.51MrTelephonesangoma device driver
02:58.03JTsplit voice/data t1?
02:58.10MrTelephoneno just voice
02:58.27JThmm
02:58.29MrTelephonethe digium cards don't have ifconfig stats?
02:58.40JTmaybe you are experiencing framslips
02:58.43JTframeslips
02:58.43JTnup
02:58.54MrTelephonethis is the link to the adit 600
02:58.55MrTelephone<PROTECTED>
02:58.55MrTelephone<PROTECTED>
02:59.29MrTelephone23 thousand errors out of 279 million packets.. hmm
02:59.34MrTelephonestill should be 0
03:03.43MrTelephoneits like .008 %
03:03.49MrTelephoneerror rate
03:04.53JT<PROTECTED>
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03:08.57MrTelephoneyeah everything points to an interrupt problem
03:09.13MrTelephoneI have this riser card in there and I don't think it works properly
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03:10.26MrTelephoneit plugs into a 64-bit pci and has two 32-bit size ram chip looking cards on wires...
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03:11.14JTyour sangoma card looks like a ram chip?
03:13.12MrTelephonenot there is a riser card off the mainboard with 3 64-bit pci slots
03:13.21MrTelephonebut I don't think its working like it should
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03:15.28JTi see
03:15.47JTMrTelephone: is it a dual span card with 1 to a telco and 1 to and adit/
03:18.39MrTelephoneyeah
03:18.54MrTelephoneI was going to try it
03:19.08MrTelephoneI never hooked up any phones to the adit yet because they didn't ship an ampehnol with it
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03:21.58MrTelephonethe riser card looks like this http://www.plinkusa.net/web2u32riser3-3.htm except its 64-bit
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03:23.13MrTelephoneword of advice. don't frig around with 2u or 1u crap
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03:23.39JToften there's no choice
03:24.04JTbut that makes me think there's nothing wrong with the riser, if the other span has no errors
03:25.04MrTelephonemaybe becuas ethere is little traffic on it
03:25.09MrTelephonegood point
03:25.20MrTelephonemaybe I'll disable the second span and see what happens first
03:25.40MrTelephoneits a 20km fibre run, maybe something is wrong with the transport
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03:26.01MrTelephoneI can't really notice anything when I'm on the phone
03:26.15MrTelephonebut there are some frames going missing
03:26.32JTwhat about zttest?
03:27.12MrTelephonemostly 100%
03:27.22MrTelephone100.000000% 100.000000% 100.000000% 100.000000% 100.000000% 99.987793% 100.000000%
03:27.22MrTelephone100.000000% 99.987793% 100.000000% 100.000000%
03:27.23JThmm
03:27.27JTsounds good
03:27.36MrTelephonenot sure if it works properly with sangoma
03:27.57MrTelephoneits really wierd
03:28.10MrTelephoneIt said change the cable but its a 10 foot cat5 cable
03:28.25MrTelephonemaybe it just needs to be rebooted, heh
03:28.39JTwhat said change the cable?
03:28.53MrTelephoneI was dinking around and I noticed I never screwed the card in tight.. it moved and there was a console error saying PCI DEVICE ERROR
03:29.04MrTelephonesangoma support website said change the cable
03:29.18MrTelephoneor cheque for irq problems
03:29.41MrTelephoneI have 2 cards in that server.. one is an adaptec raid5 but it BARELY requests an interrupt
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03:33.01MrTelephonejt, how do you check for line conditions with the digium t1 card?
03:33.12MrTelephonejust look for the alarms?
03:33.37Mavviepri show spans
03:33.43JTMrTelephone: or frame slips/bad zttest results
03:35.30MrTelephoneI have this connection to the adit 600.. I find it interesting that you don't configure asterisk as you would with the telco company.. you actually configure fxo_ls signalling even tho that your dealing with a t1 link
03:35.53JTumm
03:36.01JTit's channelised T1
03:36.12JTit basically is acting as analogue over a digital link :)
03:36.21JTdoesn't use PRI ISDN signalling
03:37.17MrTelephonethats pretty cool then that these devices work with asterisk's signalling
03:38.09MrTelephonethe adit 600 is a nifty little device
03:38.34MrTelephoneI want to try one of those t1 mux things from rad over a 900mhz wireless link
03:38.52MrTelephoneI'm scared it won't work though, have you tried doing any t1 over wireless?
03:39.02JTnope :)
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03:40.25MrTelephonecarrier error in ifconfig is pretty bad and there is one of those
03:40.44MrTelephonefive of them rather
03:41.24MrTelephonei know there were a couple dropped calls I heard about
03:41.28MrTelephoneso I'm guessing thats why
03:44.35MrTelephoneone guy on the web posted this... and asked if it was bad
03:44.36MrTelephoneRX packets:1136725015 errors:0 dropped:0 overruns:0 frame:0
03:44.36MrTelephoneTX packets:1195140164 errors:0 dropped:0 overruns:0 carrier:1110109378
03:44.38MrTelephonehaha
04:05.02*** join/#asterisk Avero (n=no@24.96.142.67)
04:06.30AveroIs there a way in app voicemail to "force" a user to set their box up if it hasn't been set up yet?
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04:21.51QwellAvero: forcegreeting=yes and forcename=yes, I believe
04:21.59Qwellbut the password has to be set to the same as their mailbox
04:22.15QwellI think it might be forcegreet, actually..  check the voicemail.conf sample
04:24.14AveroYeah, there it is. I thought I remembered being able to do that. Thanks!
04:29.25docelmoQwell got any experience with MOH and Timing issues or VAD?   I need a work around for VAD or CN
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04:49.41DefrazI have a phone system here at work, the system answered calls, it had voice mail. I am replacing it with an Audio Codes gateway and asterisk. The old system hung up when it needed to if the caller hung up, but the asterisk and audiocodes doesn'thang up and leaves the fx0 port open.
04:49.58DefrazMy question is should the audiocodes drop the call like the old system.
04:52.16Strom_MDefraz: you have to set the fxo port as "ks"
04:53.51DefrazI can't figure that out on the Audio Codes fx0 gateway mp-118
04:54.31Defrazkewlstart
04:58.37*** join/#asterisk ExR90 (n=exr9001@cpe-76-166-105-25.socal.res.rr.com)
04:58.44Strom_Mit's FXO
04:58.46Strom_Mnot fx0
05:00.50ExR90I have googled for some time on this without luck. Are there any known issues with asterisk-1.4.4 and festival 1.95? I get WARNING[31901]: app_festival.c:393 festival_exec: festival_client: gethostbyname failed
05:03.18ExR90I saw a reference to it in an old bug but it was closed many versions ago being fixed. My festival.conf has all options commented out. I have tried it with them uncommented as well
05:10.03*** join/#asterisk stimpie (n=michiel@ip565faf27.direct-adsl.nl)
05:11.05*** join/#asterisk arrenlex (n=em@S01060040052da362.ed.shawcable.net)
05:11.17arrenlexIs there some list of cool but useless things one can do with asterisk?
05:11.39JTi think there's on or two on the wiki
05:11.41JT~thewiki
05:11.42jbotit has been said that thewiki is at http://www.voip-info.org/wiki-Asterisk
05:14.09arrenlexCool beans. Thanks.
05:14.10*** part/#asterisk arrenlex (n=em@S01060040052da362.ed.shawcable.net)
05:14.25ExR90Now if only the bot had an answer for my Festival issue
05:14.50*** join/#asterisk Joe_CoT (i=joe_cot@powerade.dreamhost.com)
05:16.10Joe_CoThey, so I'm setting up asterisk on my box, and I can't register my sip soft phone. On the soft phone end, i see "Joe, registration failed: 404 Not found", and in the asterisk log, i see "chan_sip.c:11245 handle_request_register: Registration from '"Joe" <sip:202@192.168.1.25>' failed for '192.168.1.2' - Username/auth name mismatch"
05:16.11*** join/#asterisk NOT_guru (n=NOT_guru@209.145.181.55)
05:16.55NOT_guruany festival guru's about?
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05:24.18[Outcast]is there away to get asteris to always send the Proxy-Authorization with every invite?
05:33.11ohadzwhat is the difference between 1.4.4 and 1.2.x * versions?
05:33.16ohadzshould i download 1.4.4?
05:34.21*** join/#asterisk adorah (n=Michael@87.69.57.73.cable.012.net.il)
05:36.49docelmoa shit load
05:36.55docelmoexpecially if you are developing
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05:44.01adorahיאללה להתעורר להתעורר יש עבודההיום:)
05:44.10adorahoops..
05:44.30adorahwrong window..hehe
05:45.28Defrazif my old pbx worked with disconnecting a call, then my asterisk box with a fx0 gateway should too right.
05:45.36DefrazKewl start is just voltage drop right?
05:45.54Strom_MDefraz: it's FXO
05:46.00Strom_MForeign eXchange Office
05:46.06Defrazyes FXO sorry
05:46.34Strom_MDefraz: asterisk will work fine; it's whether your FXO port can recognize it
05:46.59Defrazyea it is an AudioCodes
05:47.12DefrazI only 3 options
05:47.13yonahw-workcan anyone assist with the following error?
05:47.13yonahw-workres_config_mysql.c:669 mysql_reconnect: MySQL RealTime: Unable to select database:  asterisk. Still Connected (1049).
05:47.13yonahw-workSegmentation fault (core dumped)
05:47.33Strom_MDefraz: what are your options?
05:47.34yonahw-workthe database exists and i can connect to it using the same credentials as asterisk
05:49.28DefrazStrom_M: Prolarity Reversal, Current Disconnect, Silence Detection.
05:49.37Strom_Moh, lets see
05:49.50Strom_Mthis is a tough one
05:50.04Strom_Myour polarity doesn't reverse
05:50.17Strom_Mand...what happens when the other end hangs up?
05:50.25Strom_MTHE BATTERY DROPS
05:50.39docelmoCan anyone tell me if MOH in APP_QUEUE uses zaptel for timing?
05:50.42DefrazYes it should drop
05:50.52Defrazbut the call just stays up.
05:51.07Strom_MDefraz: you do have it set to "current disconnect", right?
05:51.10Defrazand they leave a 10 minute voicemail or they sit in queue forever till soemone answers and of course nobody is there.
05:51.22Defrazyes current disconnect is enabled.
05:51.31Strom_Mwell then your gateway just blows dead yaks
05:51.37Defrazbut I don't see any of that in the log.
05:51.39*** part/#asterisk mattwj2005 (n=Matt@user-38q4155.cable.mindspring.com)
05:52.53DefrazTypical. I always go on the forms and check things out and blam I get crap.
05:53.31Strom_Mget a digium card and save yourself the headache :)
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05:56.58DefrazYea, I guess so.
05:57.06*** join/#asterisk phalacee (n=Sunforge@202.3.110.33)
05:57.12DefrazThis was as expensive as a digium card.
05:57.14Defrazoh well.
05:57.19DefrazI will figure it out.
06:02.30yonahw-workis there an advantage to using the odbc driver for realtime rather than the mysql driver? (I am attempting to connect to a mysql db)
06:07.51*** join/#asterisk tessier (n=treed@kernel-panic/sex-machines)
06:08.12tessierI wonder how many I've done in the last 3 years. I've lost count.
06:10.33Joe_CoTso ... how easy is it to set up asterisk without freepbx? I've been tearing my hair out all day trying to get both working right
06:11.13tessierI just set it up without freepbx.
06:11.24tessierBut I remember when I was first starting out. Asterisk has a learning curve for sure.
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06:12.00yonahw-workcan anyone help me with some realtime issues?
06:12.13yonahw-workasterisk can not connect to the database although i can using the same info
06:13.06VioBytewhat host name/ip did you give asterisk to connect to
06:13.07VioByte?
06:13.12yonahw-worklocalhost
06:13.33yonahw-workdoes it need to be contained in a string?
06:13.37VioBytepaste up some info on pastebin.com
06:13.50VioBytethe output of where the problem is
06:14.05yonahw-workwill do
06:15.04Joe_CoTyeah, same problem i'm having :-/
06:16.19JTuse pastebin.ca not .com
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06:18.03yonahw-workJT: thanks i just figured out that .com was not working right and am on my way to .ca
06:18.31yonahw-workhttp://www.pastebin.ca/475065
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06:24.51VioByteasterisk. Still Connected
06:24.53VioByteodd
06:24.59VioBytenever seen that before
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06:26.26VioBytecheck mysql "show processlist;"
06:26.33VioBytesee if asterisk is still connected
06:29.34VioByteseems like its already connected to mysql
06:29.46VioBytemaybe somthing in extconfig.conf is using the same table
06:31.12yonahw-workcould it be a problem if cdr_mysql is using the same database?
06:31.26yonahw-workdifferent table of course
06:31.28VioByteshouldnt be
06:32.26yonahw-workhmmm
06:32.38yonahw-workthere is nothing else in extconfig.conf i just set it up for this
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06:33.26yonahw-workwell i gotta run but thanks for taking a look
06:33.34VioBytei use AGI+PHP for my mysql needs :) so i'm not much of help on this problem :( sorry
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07:06.07notnytdoes anyone know where in the source the caller id update from the agent info takes place?
07:06.27tessierbkw_: What's the name of that asterisk fork project you've been working on?
07:06.49AsteriskGuy99Hello all, I'm an Asterisk Newbie.
07:07.00AsteriskGuy99I've just finishing installing Asterisk on a Ubuntu v7.04 server.
07:07.09tessierAsteriskGuy99: Hi Asterisk Newbie.
07:07.17AsteriskGuy99(v1.4.4 Compiled from source)
07:07.35AsteriskGuy99I wanted to get your opinions on what the best GUI is
07:07.46AsteriskGuy99(GPL or Freeware GUI) for Asterisk
07:07.54AsteriskGuy99for web based configuration
07:08.03tessieremacs-x is my favorite gui
07:08.04tessieroh
07:08.08tessierweb gui
07:08.10tessierThey all suck.
07:08.15tessierLearn to use a text editor. :)
07:08.32AsteriskGuy99Well I don't know Asterisk yet, and
07:08.39AsteriskGuy99I want an easy way to configure it
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07:08.59AsteriskGuy99I've heard of freePBX, etc
07:09.03JTAsteriskGuy99: take a look at the book
07:09.05AsteriskGuy99but I don't know which one is the best
07:09.06JT~thebook
07:09.07jbotextra, extra, read all about it, thebook is a book called  Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11
07:09.22JTyou'll find very little support for freepbx
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07:09.58AsteriskGuy99ok
07:10.08AsteriskGuy99Ya, the screenshots didn't even come up there, so
07:10.19AsteriskGuy99I was wondering if it was discontinued
07:10.37JTif you use freepbx of trixbox, you won't be able to come here for help
07:10.57JTthey're not, but they don't seem to support anyone
07:12.17AsteriskGuy99I see
07:12.46AsteriskGuy99I was looking for something like this:
07:12.47AsteriskGuy99http://www.asterisknow.org/image/tid/55
07:13.02AsteriskGuy99But I know that Asterisk Now is a complete install
07:13.31AsteriskGuy99and since I've already installed Asterisk on a Ubuntu server, I was just looking for a web based configuartion part
07:13.47JTspend a bit of time reading some of the book, you won't need a gui then
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07:15.12AsteriskGuy99JT - ok, I'll start RTFMing it then :)
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07:16.32pseudorgot a problem with H323 installation (both native h323 and oh323)
07:17.15pseudorhave someone had a deal with the h323 installation?
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07:29.55notnytdoes anyone know where in the source the caller id update from the agent info takes place?
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08:11.43sitxudoes anyone work with ALIWEI devices?
08:12.00AsteriskGuy99ok I have a simple newbie question
08:12.24AsteriskGuy99where should the Asterisk "confs" directory normally be located?
08:15.15sitxuat "/etc/asterisk"
08:16.24AsteriskGuy99ok
08:16.32AsteriskGuy99because I'm not seeing any directories in there at all
08:16.38AsteriskGuy99am I supposed to manually create them all?
08:17.08AsteriskGuy99I was hoping to see a /sounds directory
08:17.13AsteriskGuy99a /mohmp3 directory
08:17.14AsteriskGuy99etc
08:17.24AsteriskGuy99I just did a fresh install of Asterisk v1.4.4
08:17.28mostyAsteriskGuy99: does /etc/asterisk exist?
08:17.37AsteriskGuy99Absolutely
08:17.44AsteriskGuy99There are a number of configuration files in there
08:17.47AsteriskGuy99but no directories
08:17.56mostyAsteriskGuy99: see /etc/asterisk/asterisk.conf
08:18.05JTerr those files shouldn't be in /etc
08:18.12JTthose directories, even
08:18.42AsteriskGuy99There are 64 .conf files in /etc/asterisk
08:19.00AsteriskGuy99JT - Where should they be?
08:19.03JTright...
08:19.11AsteriskGuy99They were put there by the installer I think
08:19.39JT/var/lib/asterisk/sounds
08:19.51AsteriskGuy99Thank you JT
08:19.54AsteriskGuy99Let me look there
08:20.09JTAsteriskGuy99: /etc is only for config files
08:20.11JTnothing else
08:21.03mostyAsteriskGuy99: look inside that file i mentioned, it says where the directories are
08:22.17AsteriskGuy99Thanks mosty
08:22.20AsteriskGuy99Will do
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08:27.21AsteriskGuy99In /var/lib/asterisk the modules directory doesn't exist
08:27.24AsteriskGuy99should I just create it?
08:27.35AsteriskGuy99the asterisk.conf file says that's where it should be
08:29.07mostyit should be created as part of the asterisk install process, do not create it manually
08:29.16AsteriskGuy99hmmm
08:29.18AsteriskGuy99It wasn't
08:29.25AsteriskGuy99I installed Asterisk and the Addons
08:29.50mostydid you look in that file i mentioned? is that where you got the location from?
08:29.55AsteriskGuy99Yes
08:29.56AsteriskGuy99It is
08:30.51mostyread the asterisk install doc again, make sure that you ran all the required make commands
08:31.08AsteriskGuy99I'm 98% sure that I did
08:31.12AsteriskGuy99I was very careful
08:31.29mostytry again one more time, i want you to be 100% sure
08:31.54AsteriskGuy99ok
08:32.07tzafrirthe modules dir is usually under /usr/lib/asterisk
08:32.43tzafrirnot /var/lib/asterisk
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08:40.08notnytdoes anyone know where in the source the caller id update from the agent info takes place?
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08:45.14marcelverhagenhello everybody
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08:49.56jeremy_ghi
08:50.09jeremy_gi am reading a sip log from asterisk
08:50.15jeremy_gi need something that could color code it
08:50.20jeremy_gits very hard to read otherwise
08:52.29jeremy_gsip debug
08:52.42jeremy_gin logger.conf verbose messages are directed to a file
08:53.02jeremy_gvi /var/log/asterisk/sip-log and it be all color coded
08:54.17tzafriris vi actually vim?
08:54.33tzafririf so, this is a matter of writing a vim syntax file
08:56.18jeremy_gyes its vim
08:56.35jeremy_gand i am explore if someone has already encountered such a problem and done something about it
08:59.04jeremy_gexploring :@
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09:14.24JThrm
09:14.30JTjust got a cisco 7905
09:14.44JTdamn the underside of the handset is made from a horrible shiny plastic
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09:15.52skirmishaguys any idea how to configure chanspy
09:16.19skirmishai mean how can i set which channel to be monitor and is it only possible to monitor 1 channel at a time
09:18.17mostyskirmisha: chanspy monitors two channels at a time, if the call is bridged
09:19.08mostypress * to cycle through channels, or there is an option to the dialplan command which limits the call to a specific channel or group of channels
09:20.15mostyhttp://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+ChanSpy
09:23.00skirmishamosty ahh so i just need to make exten to go to chanspy right?
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09:45.24matt_does anybody here use voipdiscount ?
09:45.32matt_or any of the related companys
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09:55.13Nobbieheya =)
09:57.11Nobbiehow would one setup asterisk in such a way that you have 2 asterisk servers in different locations, and users can login to eighther of the 2, and still be contactable from both ? is it possible without a SIP Proxy ?
09:59.06mostydial(SIP/box1/ext&SIP/box2/ext) ?
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10:01.58kink0hi
10:02.19pseudorwhat is the problem with h323 in *? Do they hate each other?
10:02.20kink0anybody knows why the form Dial(Zap/g1/ww${EXTEN}) returns CC38 (INVALID NUMBER FORMAT) ?
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10:10.05Nobbiemosty: looking for something cleaner, potentially there could be more then 2 boxes when more locations are added
10:10.19mostykink0: what's that wW doing there?
10:10.41mostypseudor: http://www.voip-info.org/wiki-Asterisk+H323+channels
10:12.30mostyNobbie: dundi?
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10:14.15kink0mosty: just add 1 sec wait time to the zap channel
10:14.55mostykink0: are you sure about that syntax? looks wrong to me.
10:15.33mostywhat do you mean by wait time?  isn't that normally done with the Wait dialplan command?
10:16.31kink0http://www.voip-info.org/wiki/index.php?page=Asterisk+Zap+channels
10:16.41kink0exten => s,1,Dial(Zap/2/ww5551234)  ; Wait 1 second then dial 5551234 on channel 2
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10:18.32mostykink0: try using Wait instead
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10:53.22pseudormosty: yaeh, I know this. Have you tried to install the native h323?
10:55.41mostyno. everybody pretty much just uses sip/iax these days
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11:22.16nicoxHello, is anybody there who made some ss7-links with chan_ss7?
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11:36.08ManxPowerI'm really tired of people trying to do EC on voip only calls.
11:36.26*** join/#asterisk crochat (n=crochat@84-74-150-141.dclient.hispeed.ch)
11:36.33nicox<PROTECTED>
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12:05.44*** join/#asterisk BruXo (n=celio@c91192a6.static.bhz.virtua.com.br)
12:08.00jeremy_gnicox:i wish i could :(
12:10.38CBU[^_^]M``hello
12:15.25*** join/#asterisk crochat (n=crochat@84-74-150-141.dclient.hispeed.ch)
12:17.23*** join/#asterisk Crescendo_ (n=martinda@adsl-072-151-080-148.sip.rmo.bellsouth.net)
12:17.32nicoxanybody there who made some ss7-links with chan_ss7?
12:17.56*** part/#asterisk raptorra1 (n=rathomps@cpe-66-25-25-138.houston.res.rr.com)
12:18.07ManxPowerAPPARENTLY NOT!
12:18.19ManxPowernicox: very very few people use SS7 with Asterisk
12:20.38CBU[^_^]M``hello... anyone here used portech products?
12:24.19*** join/#asterisk explidous (n=explidou@rrcs-24-173-134-222.se.biz.rr.com)
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13:25.53b00gzHey guys, I have an IVR file and the begining of it gets messed up when its played by Asterisk but if I play it in a media player it sounds fine.  The file is a mono 800hz with a 32bit float...
13:27.07*** join/#asterisk unixlike (n=spid3r@31.67.modemcable.oricom.ca)
13:27.50coppiceb00gz: every time you ask that question the answer is the same
13:28.56ManxPowercoppice: The answer is "put an Answer and Wait(1) before the Playback or Bakcground line"?
13:29.29ManxPoweror is the answer "Asterisk expects 8000hz, not 800hz and 16 bit, not 32bit"?
13:30.19coppiceI think he gets both those issues raised each time
13:30.52ManxPowercoppice: So many idiots, so little time.
13:31.15*** join/#asterisk Fieldy (i=x4P2W30y@gentoo/contributor/Fieldy)
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13:36.05*** join/#asterisk sbuntin (n=chatzill@mail.kalleo.net)
13:36.51sbuntinI am trying to use fxotune and I keep on getting "couldn't fill input buffer".  How do I address this?
13:38.14*** join/#asterisk phillipk (n=pkey@fw.datafax.net)
13:38.15ManxPowersbuntin: what card model?
13:39.13b00gzManxPower: how to I put a Wait before playback?
13:39.25ManxPowerexten => 666,1,Answer
13:39.32ManxPowerexten => 666,n,Wait(1)
13:39.43ManxPowerexten => 666,n,Playback(yoursoundfile)
13:39.56*** part/#asterisk bertrand^ (n=bertrand@ATuileries-151-1-27-71.w82-123.abo.wanadoo.fr)
13:40.03b00gzManxPower: this goes in extension.conf?
13:40.10ManxPowerb00gz: yes.
13:40.23b00gzManxPower: Thanks so much!
13:41.02*** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca)
13:41.09ManxPowerb00gz: Your Asterisk project will fail terribly if you do not read up and understand Asterisk
13:41.10ManxPower~book
13:41.12jbotsomebody said book was a book called  Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11
13:42.01sbuntinI have a TDM400P
13:42.21sbuntinManxPower: it has four FXO ports
13:42.56*** join/#asterisk Capps- (n=andrew@67-67-242-2.ded.swbell.net)
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13:44.41*** join/#asterisk ZaVoid (n=zavoid@c-67-165-25-195.hsd1.ct.comcast.net)
13:44.47ZaVoidmorning everyone
13:44.57unixlikegood morning
13:45.00ManxPowersbuntin: I don't have any suggestions
13:45.43ZaVoidanyone have any idea what it means whenthe following flashes by in console 15 times a minute or so
13:45.51ZaVoid<PROTECTED>
13:45.51ZaVoid<PROTECTED>
13:46.02sbuntinManxPower:  is it that bad?  hasn't anyone had any issues with this?
13:46.23Putzzomg
13:46.30Putzzit means what it says
13:46.48iCEBrkrZaVoid: It means someones connecting/disconnecting from the asterisk CLI
13:46.58ZaVoidhmm
13:47.04iCEBrkrPutzz: Crazy talk!)*$#
13:47.17Putzz;-)
13:47.50ZaVoidis there any way to tell who/what ip?
13:48.33iCEBrkrYou could start by seeing who has an account the machine running Asterisk
13:48.44docelmoiCEBrkr what the fuc* do you know bout asterisk..   newb.,.
13:48.53redaxhi
13:48.59iCEBrkrdocelmo: yea, really, WTF do I know?!!?
13:49.09iCEBrkr<- newb to the c0re!
13:49.19redaxis there a text version of the asterisk/sounds files?
13:49.28iCEBrkrhuh? text version?
13:49.30docelmoiCEBrkr this is true..
13:49.45redaxlike pbx-invalid: I am sorry, that's not a valid extension.  Please try again.
13:49.53docelmotoo newbish for anyone..  but what you do know is BOCA sucks
13:49.54docelmo:)
13:49.55iCEBrkrredax: cat /var/lib/asterisk/sounds/goodbye.gsm
13:50.04ManxPowerredax: sounds.txt
13:50.07iCEBrkrdocelmo: Yes. Boca sucks.. I know that much
13:50.12redaxoh really? thanks
13:50.28docelmohaha
13:50.44docelmoIf you would have only figured that out last year
13:50.59ZaVoidyeah the only people that have access to the accutns on the box aren't doing
13:51.04ZaVoidthats why i'm asking for other ideas
13:51.07iCEBrkrdocelmo: damnit, give me a Los Angelas DID or I'll stomper your little car.
13:51.21redaxManxPower: hm. where's that sounds.txt?
13:51.24iCEBrkrZaVoid: They're lying.
13:51.27docelmowhich one do you want?  I have 500 of them
13:51.28ManxPowerredax: asterisk source
13:51.28ZaVoidno they are no
13:51.29ZaVoidt
13:51.48ZaVoidso is there anyway to debug it in the console and see where the connection is coming from?
13:51.50redaxah. toplevel.
13:51.53redaxthanks
13:52.13iCEBrkrdocelmo: 562 areacode
13:52.27docelmoI probably have it..  Why who needs it?
13:52.39iCEBrkrdocelmo: my gold-digger GF
13:52.49docelmoI pleed the 5th
13:52.52iCEBrkrlol
13:53.09docelmoI figured you would have married her by now
13:53.14iCEBrkrF THAT
13:53.19iCEBrkrAre you crazy?
13:53.30docelmobeing she has the potiental to make your paycheck to look like chump change
13:53.35iCEBrkrdocelmo: I'm moving back to Tampa
13:53.41[TK]D-Fenderdocelmo: Keep pleading the 5th ;)
13:53.59docelmoya Matt Florell told me you missed it
13:54.06[TK]D-Fenderdocelmo: And here I was thinking his emplyer did a good enough job of that ;)
13:54.14iCEBrkrdocelmo: I miss his boyish good looks
13:54.55docelmohmmmm
13:55.06iCEBrkrCrazy!! I'll be able to finally participate in the Asterisk Users Group I kick-started.
13:55.11docelmoSo you bringing regin back with you?
13:55.18*** join/#asterisk jtknapp (n=skip@65-126-63-1.dia.static.qwest.net)
13:55.26docelmoThen matt and I built up
13:55.30iCEBrkrdocelmo: Yea, man, The return policy states I have to put her back where I got her.
13:55.58docelmoeh..  So yer gonna bounce when you get back
13:56.16iCEBrkrSeriously.  I got a house lined up in Larghetto. 3bed/2ba + 1 car garage.. 10mins from Indian rocks beach
13:56.26iCEBrkrdocelmo: The free ride is over!
13:57.26docelmohaha..  Just air out yer shit on the channel
13:57.29iCEBrkrI scored job with pay increase and my rent goes down about $200.  With having a garage, I won't have to pay for storage, so it's about $300/mo savings
13:57.46docelmoGood deal
13:57.49iCEBrkrdocelmo: oh, I wouldn't want to REALLY bore people
13:57.51docelmowho did you get hired with?
13:58.04iCEBrkrdocelmo: Oh man, you'll laugh your ass off
13:58.17iCEBrkrdocelmo: It's some tiny little shop.  I'll be doing work for the darkside.
13:58.23*** join/#asterisk DarylVOIP (n=daryl@c-71-224-42-97.hsd1.pa.comcast.net)
13:58.34iCEBrkrdocelmo: .NET/ASP and Flex/Flash stuff.
13:58.43docelmohaha..  Well air yer shit ot so people over seas see how we handle our shit
13:58.46docelmoewww
13:58.49iCEBrkrhaha
13:58.54*** join/#asterisk SwK (n=SwK@70.158.103.10)
13:59.01docelmoKEN!
13:59.06iCEBrkrdocelmo: naa man, it's all good. I've been wanting to learn Flash/Flex.
13:59.32docelmoeh..  whatever works for ya
13:59.53iCEBrkrdocelmo: I feel as if it's 'expanding' my horizons.
14:00.17iCEBrkrThe guy even asked "Um, you got all this opensource stuff here, are you sure you want to make the career change?"
14:00.25docelmoupgrade to GF2.0
14:00.27*** part/#asterisk nicox (n=nicox@83-64-42-210.prater.xdsl-line.inode.at)
14:00.34iCEBrkrcode is code. It all sucks.
14:00.37iCEBrkrsome just sucks less
14:00.40*** part/#asterisk jtknapp (n=skip@65-126-63-1.dia.static.qwest.net)
14:00.41SwKwhats up elmo
14:00.58docelmonada..  just chilling turning up my non-rboc route
14:01.32ManxPowerIf you have problems with app_girlfriend then try app_boyfriend.
14:01.41*** join/#asterisk phillipk (n=pkey@fw.datafax.net)
14:01.58*** join/#asterisk PseudoNim (n=pseudo@74.57.2.150)
14:01.59docelmoManxPower thats quite fine
14:02.03PseudoNimhey all
14:02.18iCEBrkrhaha
14:02.21docelmoManxPower say I saw you posted something about res_musiconhold awhile back
14:02.29ManxPowerdocelmo: what did I say?
14:02.51docelmoI am looking for a work around for VAD/CN.  Im having issues with MOH
14:03.00PseudoNimi'm trying to set up a callback system with DISA... so far, i managed to get asterisk to identify the call, call me back, identify me with a password and give me disa. however, when i try to dial out after getting a dialtone, i get a fast busy after the first digit i press.
14:03.04ManxPowerdocelmo: the solution to VAD/CN is to not use it.
14:03.05docelmoWell app_queue the audio fades in and out
14:03.21iCEBrkrhaha
14:03.22PseudoNimit must be a dialplan thing, it probably doesn't know how to dial out..... but i'm too clueless about asterisk to know. can anyone advise? hehe
14:03.29docelmoIm trying to get it shut off but I need a work around for time being know any?
14:03.39ManxPowerdocelmo: there is no workaround
14:03.54docelmoeh..  crap
14:03.56iCEBrkrPseudoNim: after all that jumping through those hoops, it won't dial out?! lol
14:04.10ManxPowerI guess the "workaround" would be to totally redesign and recode how Asterisk handles audio, timing, etc.
14:04.17PseudoNimiCEBrkr: i know eh? :P
14:04.21docelmoohh nevermind
14:04.28docelmoI was hoping for something simple
14:04.31iCEBrkrPseudoNim: core set verbose 9
14:04.34PseudoNimiCEBrkr: all the faqs i find point to using WAMP. but it doesn't work for me for whatever reason, so i'm trying to do everything manually
14:04.45ManxPowerWAMP?
14:04.49*** join/#asterisk tsurko (n=tsurko@77.70.24.142)
14:04.51docelmoI was reading somewhere that by setting up timing with Zaptel it could fix it..
14:04.51ManxPowerIs that like WINMP?
14:05.02ManxPowerdocelmo: what version of Asterisk
14:05.13iCEBrkrManxPower: Yea, that's how you stream MP3s.
14:05.43ManxPowerI thought you streamed MP3s using Icecast/Shoutcast
14:05.56PseudoNimicebri: er, no such command core. i'm using 1.2 btw
14:05.58docelmoManxPower 1.4.4
14:06.09iCEBrkrPseudoNim: oh.. then just 'set verbose 9'
14:06.19iCEBrkrPseudoNim: and then try having it dialout
14:07.48PseudoNimiCEBrkr: the only way for me to do that is to do the disa thing, since i'm running this off a remote server with nothing connected to it locally (just a fyi)
14:08.13PseudoNimiCEBrkr: i don't see anything radically wrong when i press the first digit though, hm
14:08.21PseudoNimit complains about ulaw vs alaw
14:08.42PseudoNimshould i pastebin everything around that event?
14:09.02iCEBrkrPseudoNim: I just figured you'd see it run out of priorities or have a missing context or something
14:09.19iCEBrkrYou should be able to see it Dial()
14:09.28iCEBrkrif it doesn't get that far, then you know something else is hosed.
14:10.11PseudoNimnope, it doesn't even try Dial()
14:10.18PseudoNimi bet my extensions.conf is hosed.
14:10.55iCEBrkrYOu should be able to see where things 'stop working'
14:11.00*** join/#asterisk Ifaistos (n=stelios@ipa226.211.tellas.gr)
14:11.21PseudoNimmy guess is that it simply doesn't see an exten to dial anywhere, so it doesn't know what to do with the #'s that i enter
14:11.33PseudoNimi mean i do have a Dial() in my incoming context, but i don't even know if it gets to it.
14:11.36PseudoNim<-- asterisk tard.
14:14.03*** part/#asterisk BruXo (n=celio@c91192a6.static.bhz.virtua.com.br)
14:15.36iCEBrkrIt needs to be June 1st.
14:15.43iCEBrkrSo I can get the heck outta this shithole
14:16.01crochatHello
14:16.22crochatWhat about t.38 fax and Asterisk 1.2 ?
14:16.30iCEBrkrPseudoNim: Well really, get into the CLI and set verbose 3 and you should see the call progress and where it's failing
14:17.03crochatI saw that it doesn't work with app_fax, but is there another possibility ?
14:17.54iCEBrkrcrochat: Are you attempting to do fax's with VoIP?
14:19.26crochatiCEBrkr: I want to install a mail2fax and fax2mail solution, as well as use a real fax machine in my LAN with VoIP... is that possible with Asterisk 1.2 ?
14:19.36*** join/#asterisk slmnhq (n=salmanh@denali.asti-usa.com)
14:20.04iCEBrkrcrochat: I haven't had any luck with faxing.. I gave up now that I don't even have POTS into my asterisk box
14:20.04*** join/#asterisk rmayorga (i=rmyorg@unaffiliated/rmayorga)
14:20.36mostycrochat: hylafax works well, i don't bother doing fax with asterisk
14:20.59crochatmosty: Hylafax with iaxmodem ?
14:21.37mostycrochat: no hylafax with a real modem
14:21.55*** join/#asterisk Zeeek (n=IceChat7@pdpc/supporter/active/Zeeek)
14:22.46crochatmosty: Yeah, but if you haven't any PSTN ? I really must do fax over IP ! So is that possible with hylafax and Asterisk 1.2 ?
14:23.00*** join/#asterisk sajith (n=user@203.187.143.130)
14:23.13mostycrochat: fax over voip does not work well
14:23.14[TK]D-Fendercrochat: FORGET about Fax over VoIP.  Your failure rate is going to be craptastic
14:23.15*** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com)
14:23.15*** mode/#asterisk [+o anthm] by ChanServ
14:23.38iCEBrkrhehe craptastic
14:24.21Zeeekwhat a concept. Start from a prinout, scan it, send it over a phone to voip where it's digitized agaion, then try to recompose it into a printable image
14:24.23[TK]D-FenderiCEBrkr: my favourite-ist word EVAR!
14:25.00*** join/#asterisk SomeOne1 (n=SomeOne1@pool-71-126-150-144.washdc.fios.verizon.net)
14:25.08[TK]D-FenderZeeek: And a pass through the Ronco food dehydrator & paste maker to boot ;)
14:25.13[TK]D-Fenderpasta*
14:25.33Strom_Mand then retrieve it with the popeil pocket fisherman
14:25.37Zeeekof course, a Polycom will receive faxes even before it boots up
14:25.44SomeOne1for SIP, can you set the host= to like multiple or a range of IP addresses?
14:26.29[TK]D-FenderZeeek: Polycom is so AWESOME they can receive faxes before they are even SENT (thanks to its new chan_fluxcapacitor.so plugin!)
14:27.01mostySomeOne1: that wouldn't make much sense
14:27.06ZeeekNot to mention the configurable pertinence levels
14:27.08coppiceZeeek: pity it can't manage the same after it boots up
14:27.32Zeeekcoppice who waits that long to see?
14:29.25[TK]D-FenderStrom_M: Don't worry.... a little GLH with make that fax pass as normal before the eyes of the public!
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14:34.09*** join/#asterisk Gled|Work (n=gled@LPuteaux-151-42-17-115.w193-252.abo.wanadoo.fr)
14:34.25Gled|WorkHi, Is there anyone there familiar with asterisk and snmp ?
14:35.09Gled|WorkI'm having something strange, as soon as i issue an snmpwalk command, the snmpd segfaults
14:35.30Gled|Workand by the output, i think this is the asterisk snmp sub agent causing this trouble
14:38.58*** join/#asterisk axisys (n=axisys@155.70.141.45)
14:39.44*** join/#asterisk agile (n=mike@68-189-222-201.dhcp.ftwo.tx.charter.com)
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14:44.36tzafrirGled|Work, I'm not familiar, but the version of asterisk etc. my help
14:45.36Gled|Workyes, I'm running SVN-branch-1.4-r62331M
14:45.56*** join/#asterisk crochat (n=crochat@84-74-150-141.dclient.hispeed.ch)
14:46.01Gled|Workwith NET-SNMP version:  5.3.1
14:47.55*** join/#asterisk `pariah (n=josh@unaffiliated/pariah)
14:49.29*** join/#asterisk wunderkin (n=wunderki@dslstat-ppp-95.fastq.com)
14:50.58Gled|Workall dependancies are satisfied, but i can't get any output from the snmp subagent
14:54.01Gled|Workwhat happens exactly when issuing snmpwalk command is that i have outputs from the general state of the system, but it seems that as soon as snmpd wants to connect to asterisk subagent, and/or get data from it, it segfaults.
14:54.04*** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com)
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15:03.45menaroHi everbody. I got a problem transfering a call properly within an AGI-script, postfixed dialstring with "|t". It sets the context and extension correctly, but sets priority to 0. Anybody familiar with this? Thanks
15:08.35*** join/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl)
15:09.34mostytried setting the priority explicitly to 1?
15:10.13menaromosty: Yes, I've tried that :)
15:11.12iCEBrkrSo what are cheap options for SMS in the US?
15:11.33menaromosty: So punching #5 on transfer will give "Spawn extension (someextension, 5, 0)"
15:12.33*** join/#asterisk vgster (n=vgster@cpc2-ledn1-0-0-cust826.leed.cable.ntl.com)
15:12.38*** part/#asterisk sajith (n=user@203.187.143.130)
15:13.22ZeeekiCEBrkr for what?
15:14.02iCEBrkrZeeek: To send SMS messages. I was check'n out bayhams... *shrug*
15:14.13iCEBrkrThere's got to be something besides them.
15:14.17Zeeekfor what purpose, business or pleasure
15:14.33iCEBrkrPleasure?  How about tinkering?
15:14.34iCEBrkr<PROTECTED>
15:14.41Zeeeka lot of sites send sms now, stuff like twitter
15:14.54Zeeektinkering, twitter and jaiku for sure
15:14.57*** join/#asterisk hfb (n=hfb@pool-72-67-156-130.lsanca.dsl-w.verizon.net)
15:15.05Zeeekyou can send a zillion free sms
15:15.14Zeeekskype does it but not free
15:15.18iCEBrkrSo what? I write something using CURL() to interface Asterisk to Twitter?
15:15.35ZeeekiCEBrkr easily, the API is public
15:15.44iCEBrkrhrrrm
15:15.46iCEBrkrInteresting
15:15.57Zeeekalthough that would be for tinkering cause they are down every time some new star starts posting
15:16.22Zeeekhttp://twitter.com/help/api
15:16.30iCEBrkrZeeek: most of the stuff I do with asterisk is for the challenge and of course a lot of proof of concept work.
15:16.33Gled|WorkI found out what happens when using snmpwalk
15:16.44ZeeekiCEBrkr I love to play with SMS and asterisk
15:16.49iCEBrkrSNMP walk it out?
15:16.59ZeeekI get an sms whenever there's a vmail for example
15:17.10Zeeekbut that's thru my orange account
15:17.27ZeeekI also can receive sms on our fixed line to asterisk
15:17.30iCEBrkrI can send SMS to my T-Mobile account via email. But what fun is that?
15:17.32Zeeekso I can send it commands
15:18.12Zeeekhow about getting an sms sent every time the tempera&ture drops below a certain point?
15:18.22iCEBrkrha
15:18.42*** join/#asterisk Mercestes (n=Merceste@rrcs-71-41-157-70.sw.biz.rr.com)
15:18.44Zeeekthe thing with twitter is, you can set it send an sms every time one of your "friends" posts
15:18.50Zeeekvery handy for texting
15:18.54Zeeektesting
15:19.15Zeeekhttp://groups.google.com/group/twitter-development-talk/web/api-documentation
15:19.15*** join/#asterisk Visual_E (n=Visual@unaffiliated/visuale/x-000000001)
15:19.21Zeeekhere are some curl examples
15:20.08ZaVoidi still don't see the point to twittering
15:20.09Zeeekactually, you can have asterisk post stuff to twitter for the wolrd to see
15:20.14ZaVoidwho wants to know if your at the store or not
15:20.19ZeeekI don't eaith it's a waste of bandwidth
15:20.29ZeeekI agree it's silly
15:20.41Zeeekbut so is testing sms with asterisk for fun
15:20.47Zeeekwhich I do
15:20.51iCEBrkrYo! Where you at?
15:20.56iCEBrkrI know where you're at, where are YOU at?
15:21.11Putzzyou know where im at
15:21.13Putzzheh
15:21.16ZeeekI'd rather have that than cell discussions of the same nature next to me in a restaurant
15:21.16iCEBrkr:)
15:21.47ZeeekiCEBrkr you can use aim or IM to have a sms sent to you thru twitter
15:26.48Gled|Worksnmpwalk fails to get values from the system and from asterisk at the same time
15:28.11Mercesteslol
15:28.34Zeeek*
15:28.42Zeeekshit, soory
15:28.43drfreezeHello
15:28.49MercestesNo your not, Zeeek.
15:28.51drfreezeAnyone use nufone with sip?
15:28.55Zeeekyes
15:29.12drfreezeZeeek: what that a yes to nufone?
15:29.15drfreeze*was
15:29.33Zeeekit was, but in fact that's not true. I use IAX with them, I forgot
15:29.50drfreezeZeeek: does iax work well for you?
15:29.51ZaVoidiax is broken in 1.4.xx :(
15:29.59Zeeekdracosilv very well, yeah
15:30.00drfreezehmm,
15:30.06ZeeekI use 1.2
15:30.12ZaVoidyeah i'm going back to 1.2
15:30.18Zeeekoh yeah?
15:30.22ZeeekI never left ;)
15:30.25ZaVoidsmart man
15:30.29ZaVoid1.2 has issues for me too though
15:30.33Zeeekdrfreeze yeah they're fine
15:30.35ZaVoidlike once a day it core dumps :(
15:31.13drfreezeusing 1.4.2
15:31.27drfreezeSo, 1.4.2 and nufone with iax == no worky?
15:31.37ZeeekI don't know since I use 1.2
15:32.04*** join/#asterisk NovceGuru (n=asdf@oh-71-50-248-25.dhcp.embarqhsd.net)
15:32.26iCEBrkr1.4.4 works fine with VoicePulse IAX
15:32.29*** join/#asterisk irule (n=irule@189.164.43.19)
15:33.47drfreezewow, I just installed 1.4.2. We're already to 1.4.4
15:34.40ZaVoidiCEBrkr:  what kind of volume you putting on it?
15:35.02ZaVoidi'm running about 20 concurrent iax channels and it just stops processing iax registrations after about 20 hours it seems
15:35.10ZaVoidless if the volume grows.. oin 1.4.2
15:35.14drfreezeiCEBrkr: can you make multiple outbound calls with voicepulse?
15:36.06iCEBrkrdrfreeze: yeah
15:36.17iCEBrkrZaVoid: haha, It's my home phone... So very little.
15:36.30PseudoNimhow can i make asterisk ask a user to enter a # with his phone, and then store that in a variable?
15:36.51PseudoNim(i want them to enter a # to be called back at)
15:36.53iCEBrkrPseudoNim: www.voip-info.org has all your answers :P
15:37.04PseudoNimiCEBrkr: fair enough =) btw, i managed to make it dial, hehe
15:37.05iCEBrkrPseudoNim: and it'll generated a lot of questions too! :)
15:37.26iCEBrkrvoip-info is kind of out of date, but it'll push you in the right direction
15:37.29*** join/#asterisk nasls_lsa (n=chatzill@87.203.68.253)
15:38.07Zeeekthe book would help too
15:38.13*** join/#asterisk jsolares (n=jsolares@206.113.226.107)
15:38.16SomeOne1mosty
15:38.25drfreezeAnyone using telasipo?
15:38.28drfreeze*telasip
15:38.43Zeeeknever hoid of 'em
15:39.25SomeOne1for SIP, can you set the host= to like multiple or a range of IP addresses? (i.e. if someone might be connecting from like 202.232.125.4,.5,.6 and .7 only)
15:39.49Strom_MSomeOne1: you don't set host= for inbound calls
15:40.21mostySomeOne1: firewall it off if you want
15:42.20*** join/#asterisk qdk (n=qdk@0x50c627be.bynxx11.adsl-dhcp.tele.dk)
15:42.38MindTheGapWARNING[5039]: res_config_ldap.c:1655 parse_config: No directory port found, using 389 as default.
15:42.42MindTheGapWARNING[5039]: res_config_ldap.c:1767 ldap_reconnect: bind failed: Invalid DN syntax
15:43.02*** join/#asterisk astawerksdotcom (n=astawerk@cpe-75-179-164-7.woh.res.rr.com)
15:43.03MindTheGapbasedn="dc=lpj,dc=com,dc=br"            ; Base DN
15:43.03MindTheGappass=lpj2005            ; Bind password
15:43.03MindTheGapuser="cn=caio,dc=lpj,dc=com,dc=br"              ; Bind DN
15:43.07PutzzOMG
15:43.10Putzz~pb
15:43.12jbotpb is probably a place to paste your stuff without flooding the channel - try  http://pastebin.ca, or http://channels.debian.net/paste
15:43.14MindTheGapwhats wrong?
15:43.39MindTheGapsorry jbot just 5 lines... :)
15:44.06Zeeekso you don't know about the 4 line rule then?
15:45.05MindTheGapwell, actually it was 2 lines of error then 3 lines of config... shame no one talked in between... :)
15:45.37Zeeekldap lines cout as two each though
15:45.52Putzzhehe
15:46.03Mercestesyou also pasted you rpassword to the channel.
15:46.15*** join/#asterisk mcf3782 (n=mfreeman@209.117.160.3)
15:46.18iCEBrkrHAX0RZ
15:46.30Mercestesand you didn't even tell us what FILE you pasted that out of.
15:46.33MindTheGaphehehe... yes, its a test pass... anyway, i get invalid DN syntax but i think it is correct
15:46.40MindTheGapres_ldap.conf
15:47.57ZeeekI auto-ignored myself once by having flood protection on and pasting 4 lines
15:48.36MindTheGapldap works ok, got samba and other stuff hooked in using the same baseDN and bindDN
15:48.40Zeeekdrfreeze are you looking for a provider
15:50.16SomeOne1because i can limit it with the host= thing
15:50.32SomeOne1i dont want any random joe shmoe be able to call into some of my contexts
15:50.38SomeOne1based on the number theyre dialing
15:50.57iCEBrkrSomeOne1: that's what passwords are for
15:51.14Strom_Mthat's what deny= is for
15:52.36MindTheGapand if i set port=389 on res_ldap i get:
15:52.52MindTheGapERROR[5075]: res_config_ldap.c:1744 ldap_reconnect: Failed to init ldap connection to l:389. Check debug for more info.
15:53.03MercestesMindTheGap:  Your pastes look nothign like this.  http://www.voip-info.org/wiki/view/Asterisk+config+ldap.conf
15:53.28SomeOne1passwords = too much overhead
15:53.35SomeOne1because call volume will be VEYR high
15:53.37SomeOne1production box
15:53.47SomeOne1easier just to match by source IP
15:53.58SomeOne1pssh that sucks
15:54.02SomeOne1i should code it myself
15:54.03drfreezeZeeek: yes
15:54.06iCEBrkrSomeOne1: Does your Asterisk box even work yet?
15:54.09ManxPowerSomeOne1: I've never gotten it to match by IP address.
15:54.11SomeOne1to allow host= have soemthing like 192.168.0.1/24
15:54.19ManxPowerSomeOne1: Asterisk always allowed all ips to connect
15:54.29ManxPowerSomeOne1: you cannot have a netmask in host
15:54.29SomeOne1no
15:54.29*** join/#asterisk shido6 (n=shido6@d221-68-200.commercial.cgocable.net)
15:54.30SomeOne1like
15:54.35SomeOne1if i have a context
15:54.41ManxPowerhost= is for an IP address
15:54.41SomeOne1[my-private-context]
15:54.51SomeOne1and i only want some IPs to go directly to that
15:55.15iCEBrkrSomeOne1: You're overcomplicating thing man
15:55.19mostySomeOne1: how many simultaneous calls are you expecting?
15:55.25SomeOne1my-private-context is in the dialplan
15:55.37SomeOne1[registrer] host=blah/12;
15:55.47SomeOne1context=my-private-context;
15:55.47SomeOne1brb
15:55.52ManxPowerSomeOne1: as you know that won't work
15:56.07ManxPowerhost= does not support a netmask
15:56.16ManxPowerunless that has changed at some point recently
15:56.28iCEBrkrManxPower: Come'on man, HAX0RZ!! will hijack his VoIP!!!
15:56.34mostyhost implies a netmask off 255.255.255.255
15:57.30iCEBrkrbecause you know, his users are going to change their phone settings to use the uber-seekret-c0d3z
15:57.36MindTheGapMercestes, probably because im not using ldapget but realtime ldap driver from asterisk svn, anyway heres the config: http://pastebin.ca/475647
16:03.23*** join/#asterisk boch (n=fran@190.48.213.43)
16:04.48*** join/#asterisk jnfuller (n=joshfull@209.121.25.42)
16:04.49tzangerhmm, do DWDM systems use FEC?  i.e. the fast telco optical links... I would imagine so, no?
16:05.26jnfullerHi, is there any way to put ;cic=### in an asterisk sip invite?
16:05.35tzangerI mean they claim a BER of 1.8x10^9 (1 bit error in 17 years)
16:07.09tzangerer 1.059x10^-18 rather
16:07.19tzanger1.8x10^9 was at 1bps, not 1.76Tbps :-)
16:07.23pipwerkDWDM is a passive multiplexer
16:07.40pipwerkso fec or anything is very unlikely
16:07.47tzangerpipwerk: this owuld be before it hits the fiber; it'd be the convolutional encoder that sends the bits to the DWDM MUX
16:08.04ManxPowerBTW, I am now available for disaster recovery testing.  I have discovered that if I am near a server it is almost guaranteed to fail, even though I did not touch the box.  I'm trying to make money from this odd thing.
16:08.11*** join/#asterisk logan|w (i=nothing@workstation.frippers.com)
16:08.17tzangeri.e. if you want to send "11011011010110" to the other side, you don't send that to the MUX, you convolve it first, IIRC
16:08.19iCEBrkrManxPower: haha
16:08.48*** join/#asterisk ploieel (n=ploieel@Fb39f.f.ppp-pool.de)
16:08.50ManxPoweriCEBrkr: I was standing 1 ft from the mail server.  Within 24 hrs we had a fatal HD crash on the machine.
16:08.51pipwerktzanger: unless you use colored lasers
16:08.52tzangerManxPower: htat's easy... make money telling people they need to buy $foo high availability software, where $foo is your friend who has the opposite effect on hardware
16:08.58wunderkinselling your body to science again, ManxPower
16:09.21ManxPowerwunderkin: Nothing new.  I sell it for most other things as well.
16:09.36iCEBrkrslut
16:10.03ManxPowerWe did discover that if you replace the mail server with a different machine then Outlook won't show you any NEW messages, just the messages before the server change.
16:10.25ManxPowerThe poor helpdesk perosn had to re-setup 500 machines
16:10.36*** join/#asterisk supjigatr (n=syslod@152.53.16.10)
16:10.43Qwell[]ManxPower: heh
16:10.49Qwell[]ManxPower: they did that at my last job
16:10.51Qwell[]...
16:10.55Qwell[]we had 150,000 employees
16:11.20supjigatrAny MAXTNT routing gurus?  I'm having issues using my MAXTNT for both SIP gateway and routing dialup calls to a portmaster.
16:11.32jnfullerI'm wondering about the custom uri options mentioned in chan_sip.c and wondering how I can invoke these.
16:12.26shido6yes
16:12.32shido6you have to smack the maxtnt
16:12.42shido6show me your config
16:13.04*** join/#asterisk Dimik (n=Dimik@unaffiliated/dimik)
16:14.53jnfullerDoes anyone know if SetVar(_URI_OPTIONS=) mentioned in the patches is the syntax for extensions.conf in the 1.4.3+ code? If so, could I use that option to set a cic=### for my initial invite?
16:16.26supjigatrshido6:You want pastbin?
16:16.58*** join/#asterisk aptura (n=jondoe@S010600a0c93f6f7e.vs.shawcable.net)
16:16.59tzangersupjigatr: that's a pain
16:18.17apturamorning. I know wifi has been given a bit of a bad rap when doing wifi/voip but want to know if the utstartcom 6700 is any better or worse among the other wifi phones? Somone is moving and wants to give me the phone with the remaining contract.
16:18.30supjigatrtzanger: Everything works except sometimes my sip calls try to get routed to the portmaster.
16:18.38tzangerdefine "sometimes"
16:18.40*** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman)
16:18.54tzangersounds like you have things set up right, but didn't set up EVERY span
16:18.56shido6there he is
16:19.05shido6shoot it to tzanger
16:19.44tzangershido6: admit it, you were gonna get hte pastebin, MSN me, see what I could do, then present the answer here and take credit, weren't ya.  :-)
16:19.52shido6no
16:19.56shido6i owe you money
16:21.39jnfullerAm I in the wrong room to ask about code and syntax for *?
16:22.05tzangershido6: oh yeah, I forgot about that
16:23.37ManxPowershido6: no hurry.  My life is busy enough.
16:23.39*** part/#asterisk ManxPower (n=manxpowe@stirprop-s4-0-0-21.ndcr2.datasync.net)
16:27.00*** join/#asterisk Uranellus (n=alexande@p57A16263.dip.t-dialin.net)
16:28.25*** join/#asterisk kink0 (n=k@161.pool62-37-205.static.orange.es)
16:28.29kink0hi,
16:29.36kink0any sugestion about why I got many pbx.c Timeout but not h... blah blah ...
16:29.37kink0I know I have not any t exten here, but why is done so much timeouts when over 40 channels in use ?
16:29.37jnfullerThanks, just found the uri_options parser in the code and answered my own question. Looks like this does what I need.
16:29.38jnfullerbye for now
16:29.39Zeeekhello
16:29.49*** join/#asterisk wushin (n=eip@asgard.carspot.com)
16:29.53wushinhello?
16:29.59mcf3782I have a system that's a Trixbox-1.2.3 install. It's running asterisk version 1.2.12.1.  About every 90 seconds, a flood of messages gets logged to /var/log/asterisk/full that look like this line:
16:30.00Zeeekwrong vindo
16:30.01iCEBrkrIt works
16:30.02mcf3782May  7 12:25:36 DEBUG[30285] chan_sip.c: Stopping retransmission on '0e2f969f2f0ffbc200882dd77c8bcc23@172.16.90.20' of Request 102: Match Found
16:30.15mcf3782I don't understand what causes it.
16:30.40mcf3782When this happens, there are about 45 or so of them in the time span of about 1 to 1.5 seconds.
16:30.45mcf3782Anyone have any thoughts?
16:30.48wushinanyone have any experience with the tdm2400 series freaking out and sending power alarms?
16:32.30Uranellushello, is this possible with asterisk: two computers (A and B) .. in A there is a ISDN card and asterisk is installed .. would I be able to phone from B through voip to A and let A make the call through the normal phone line?
16:32.54mostyuranellus, yes
16:33.07errranyone know if there is a way to make asterisk 'reload' from python?
16:33.23DaminiCEBrkr: Yo..
16:33.28wushin'0e2f969f2f0ffbc200882dd77c8bcc23@172.16.90.20' <-- is your sip port open to the internet?
16:33.37iCEBrkrDamin: What's up?
16:33.45Uranellusmosty: ok thanks :)
16:33.50iCEBrkrDamin: Your admin interface is ummm 'working'
16:33.51*** part/#asterisk Uranellus (n=alexande@p57A16263.dip.t-dialin.net)
16:33.56mostyerrr: do asterisk -rx reload, from inside python
16:33.57DaminiCEBrkr: I know that your time is going to be crunched, but is there any way we can get together (You, Ed and I) and drink a beer?
16:34.18errrmosty: just use a system call I guess?
16:34.25iCEBrkrDamin: There's always time for beer.. but I'm 1100mi from Cleveland :)
16:34.32*** join/#asterisk psmaker123 (n=galin@38.112.7.18)
16:34.40DaminiCEBrkr: But I'm going to be in orlando from the 22nd through the 27th..
16:34.45iCEBrkrDamin: oooo.
16:34.52mostyerrr: yes. there may also be a manger command to do it
16:34.52mcf3782There is no path from the Internet through the firewall to the asterisk box... or at least that's what the firewall group tells me.
16:34.53iCEBrkrDamin: let me mark this on my calendar
16:34.56DaminiCEBrkr: And so is Ed.. ;)
16:35.01errrmosty: ok thanks
16:35.11DaminiCEBrkr: Probably best to get together Friday, the 25th or Saturday the 27th..
16:35.22DaminErr... 26th..
16:35.23DaminI think..
16:35.26*** join/#asterisk n00dle (n=ccraft@hillel.springsips.com)
16:35.32iCEBrkrDamin: Oh, well shit dude, I'll actually be headed up that way that weekend
16:35.46DaminiCEBrkr: I'll buy dinner! :)
16:35.55n00dleHi y'all.  Anyone using GXP2000s and BLF?
16:35.58iCEBrkrDamin: haha even better! :)
16:36.08DaminiCEBrkr: Seriously.. let's go find some place where we can devastate a cow..
16:36.08iCEBrkrDamin: I gotta head to Tampa and scope out the new house I'll be renting
16:36.23wushinanyone have any experience with the tdm2400 series freaking out and sending power alarms?
16:36.23iCEBrkrDamin: There's a Brazilian Steak house in Orlando..
16:36.27mcf3782n00dle yes. I have BLF configured on a couple of GXP2000s
16:36.33DaminiCEBrkr: Fogo De Chao? or Texas De Brazil?
16:36.41DaminiCEBrkr: Either would be fine..
16:36.43iCEBrkrDamin: They serve all sorts of beef/chicken on swords and stuff
16:36.55Putzzhmm brazilian steak house....excellent
16:36.55iCEBrkrYea, it's along the lines of Texas De Brazil
16:37.05DaminiCEBrkr: Cool.. let's make it a plan then..
16:37.23n00dlemcf3782: Cool, so... the BLF button allows me to pick up a call ringing on another extension if it was made from inside, but doesn't seem to let me pick up a call coming in from outside... I'm using trixbox, btw.
16:37.26DaminiCEBrkr: Alright.. i gotta run.. :)
16:37.30iCEBrkrlater
16:37.43Putzzu r talking in the open
16:37.46Putzzinvite us also!
16:37.46DaminiCEBrkr: Shoot me your cell number in IM when you get a chance..
16:37.56iCEBrkrroger
16:37.57SomeOne1im gonna modify host=
16:37.59SomeOne1to allow a netmask
16:38.11SomeOne1or wildcards even
16:38.23SomeOne1192.168.0.*
16:38.23mostySomeOne1: how many simultaneous calls are you expecting?
16:38.31SomeOne1250
16:38.47mostySomeOne1: and what kind of termination?
16:38.52iCEBrkrPutzz: haha, You in/near Orlando, FL?
16:39.02SomeOne1simple SIP proxy, no transcoding
16:39.05PutzzI wish im in canada ;-)
16:39.10SomeOne1very low overhead.. and im trying to keep it that way
16:39.11iCEBrkrPutzz: CANADIA!
16:39.19SomeOne1very strong 2 dual core machine
16:39.39mostySomeOne1: but i mean where are you sending calls?
16:39.39SomeOne1dual dual core
16:39.39SomeOne1to another SIP server
16:39.43SomeOne1no zaptel
16:39.46SomeOne1or anything
16:39.48mcf3782I haven't tried a BLF for an external line. But I'd think it would work just the same.
16:40.04mostySomeOne1: sounds like you should use openser instead of asterisk
16:40.06astawerksdotcomwhats wrong with canada?
16:40.12SomeOne1mosty, modifying host= wouldnt be that easy
16:40.24SomeOne1mosty: wait, gotta terminate to h323 :(
16:40.55n00dlemcf3782: No... blf is monitoring the "front desk" extension...
16:40.58SomeOne1i mean it wouldnt be that hard
16:41.06n00dle...a call comes in a line that rings only the front desk...
16:41.08SomeOne1is opernSER more stable?
16:41.23n00dle...and I want to pick it up. BLF is indicating ringing, so I press it...
16:41.33SomeOne1because i dont need 90% of asterisk features for what im doing
16:41.38SomeOne1infact i uninstalled most of the modules
16:41.55mostySomeOne1: openser is more efficient than asterisk, but only supports sip
16:41.58SomeOne1free up some memory and remove possibility of more stuff going wrong
16:42.02n00dle...but if the call to front desk is coming from outside, nothing happens.  If the call to front desk is coming from an inside extension, it works.
16:42.04*** join/#asterisk radovoip (n=radovoip@xd141.sstar.com)
16:42.09*** join/#asterisk Aphelion (n=lk@unaffiliated/lv)
16:42.31wushinSo anyone know any thing that might cause issues with the tdm2400 that would cause it to reset frequently?
16:42.32iCEBrkrmosty: haha, I was thinking OpenSER as well
16:42.55*** join/#asterisk russellb (i=russellb@asterisk/developer-and-stable-maintainer/drumkilla)
16:42.55*** mode/#asterisk [+o russellb] by ChanServ
16:43.22SomeOne1mosty: what if i setup openSER and forward it to SIP/localhost then another process running http://www1.cs.columbia.edu/~kns10/research/gw/
16:43.53CBU[^_^]M``hello... anyone here used portech products?
16:43.55apturaCanada taxes astawerksdotcom
16:45.11mcf3782n00dle - sounds like something perhaps related to contexts in your dial plan.  But I'm far from an expert.
16:46.48SomeOne1okay lets take a different approach
16:47.03SomeOne1ive got a range of IP addresses that i want to go to a specific context in my dialplan
16:47.12n00dleThis getting used to a front end writing my extensions.conf is wearing thing.
16:47.17n00dleerr... "thin" even!
16:47.20mostySomeOne1: i'm not convinced that checking passwords is a major overhead in your setup. have you done any benchmarking?
16:47.51SomeOne1well, youre right, ive been asuming
16:48.32SomeOne1i could do it in my dialplan even
16:48.42mostyyou should do that before wasting time optimizing something that doesn't need it
16:48.49SomeOne1something like GotoIf(${IP} = '192.blah')
16:49.22SomeOne1heh
16:49.48*** join/#asterisk CunningPike (n=CunningP@dhcp-10-234.district.north-van.bc.ca)
16:49.53SomeOne1hows australia, mate?
16:50.05*** join/#asterisk Gouroutrash (n=x@ACaen-151-1-47-53.w86-215.abo.wanadoo.fr)
16:50.10Gouroutrashhellooo
16:50.16mostyit's late here
16:50.28SomeOne1is it winter there these days?
16:50.35mostynot quite
16:50.43SomeOne1is winter longer or summer?
16:50.49SomeOne1do you ever see any kangeroos in the wild?
16:51.41mosty1) i don't know 2) mostly only as roadkill
16:51.51SomeOne1haha
16:51.56SomeOne1thats gotta be awesome
16:51.57carrarsomeone1, what half a circle is larger?
16:52.09Qwell[]carrar: quarter circle
16:52.10tzangerthat reminds me
16:52.20SomeOne1all we see here is deer
16:52.22carrarheh
16:52.22tzangerI was driving up to North Bay once and saw this dead dog at the side of the road
16:52.24SomeOne1and like, squirells
16:52.27SomeOne1which sucks
16:52.36tzanger"man that is one oooooooooooooooogly dog" I thought as I came up on it
16:52.44tzanger"Man that is one big dog..." as I got closer
16:52.51tzanger"Oh... it's a baby moose"
16:52.56Putzzheh
16:52.58PutzzEH!
16:53.45SomeOne1kangeroos
16:53.47SomeOne1i wanna see one
16:54.11Putzzis it true the toilet spins the opposite way? lol
16:54.17Putzz*water in toilet
16:54.41tzangerhahaha
16:54.46tzangeryou got funny toilets man
16:54.50tzangerdo you have to strap yourself down?
16:56.26*** part/#asterisk queuetue (n=scott@70.54.254.134)
16:56.50Putzzlets all phone tzanger see what he is up to ;-) 1-519-....
16:57.09tzangerheh
16:57.13tzangerhow'd you know I was in 519
16:57.13Putzzcan we?
16:57.26Putzz1-519-XXX-2004?
16:57.27Putzz;-)
16:57.46tzangerhmm ok wtf
16:57.53tzangeryou've got it, but now I'm wondering from where
16:57.59Putzz1-519-2XX-2004?
16:58.03iCEBrkrlol
16:58.07Putzz;-)
16:58.09neverbluewhen 'sip debug' is enabled, does Asterisk still have activity (checking that the server is still online/running -- just an example)?
16:58.15iCEBrkrtzanger: They're watching you
16:58.20tzangerindeed they are
16:58.32Putzzyeah the federales are watching
16:58.34tzangerif you want to save the LD though just call my Asterisk box directly
16:58.47iCEBrkryeah, dial by ip
16:58.48iCEBrkrc/lear
16:58.53iCEBrkrhaha
16:59.22Putzztzanger: ring ring
16:59.29shido6:)
16:59.36iCEBrkrtzanger: He probably got your number off a bathroom wall.
16:59.52tzangeriCEBrkr: I asked ou to stop doing that
17:00.03iCEBrkrOh. um..
17:00.04Putzzring ring
17:00.04iCEBrkrsorry
17:00.07Putzz;-)
17:00.13tzangerheh
17:00.20iCEBrkrtzanger: But it's for a GOOD TIME!
17:00.21iCEBrkrc/lear
17:00.22iCEBrkrdamnit
17:00.24Putzzringing?
17:00.33tzangerPutzz: only I will ever know :-)
17:00.55iCEBrkrPutzz: your number has been blacklisted.
17:01.01Putzz0000000000
17:01.02Putzzlol
17:01.14iCEBrkrXXX!
17:01.15SomeOne1only 99 numbers to try
17:01.30Qwell[]SomeOne1: less than that
17:01.36Qwell[]211 != valid
17:01.40SomeOne1really?
17:01.43SomeOne1howd you know?
17:01.47Qwell[]because it can't be
17:01.55Qwell[]X11 is not a valid prefix
17:01.56DefrazWell, I got those AudioCodes working with asterisk. I was almost going to buy a digium card.
17:02.04SomeOne1ahh
17:02.10SomeOne1what about x12?
17:02.12iCEBrkrc0d3z
17:02.13Putzzim sure u can nail it by checking area codes
17:02.17Qwell[]x12 is, sure
17:02.23Putzzthere is only 1 or 2 startiing with 2 in 519
17:02.39SomeOne1hmm
17:02.51SomeOne1or you could just give his phone number to me
17:02.51astawerksdotcom"Digium sale at www.astawerks.com  "!!
17:03.04tzangerno need to be secretive, it's 292.  :-)
17:03.24tzangerwe need an asterisk-biz IRC channel
17:03.36SomeOne1heh
17:03.38iCEBrkrtzanger: another waste of bandwidth? :P
17:03.40astawerksdotcomword  start it homie!
17:03.46tzangerindeed
17:03.46SomeOne1dude arent you scared people will keep calling you now
17:03.50tzangerhell no
17:03.50Putzzwell the answer was on his host mixdown.ca
17:03.54Putzzyour # is on there
17:04.03Putzz;-)
17:04.16SomeOne1519 is in canada??
17:04.16iCEBrkrSomeOne1: Most of us here could really care less about other peoples #s
17:04.21tzangerno I've got another number on the WHOIS for there
17:04.23bochdo you know why my * is not runing the 'h' ext when hang?
17:04.27SomeOne1mine is 1-703-911-2819
17:04.31Putzznot whois
17:04.34Putzzits right on the site
17:04.37MercestesSomeOne1, That's evil.
17:04.40iCEBrkr703, VA
17:04.42tzangerboch: did you use the 'g' flag in Dial() ?
17:04.46astawerksdotcomdc metro
17:04.47Mercestes911, emergency assistance.
17:04.49tzangerSomeOne1: at least make it less obvious
17:04.49Putzzmine is 1-905-898-1221
17:05.04bochtzafrir, it is an answered call
17:05.06tzanger1-703-791-1557
17:05.07tzangeror something
17:05.08SomeOne1heh
17:05.09iCEBrkrI still have my Tampa areacode but I live in Boca Raton
17:05.11SomeOne1in in DC metro
17:05.13SomeOne1im*
17:05.17Qwell[]tzanger: not 1337?
17:05.23tzangerheh no I'm not 1337
17:05.29MercestesQwell[]:  I was thinkin' the same thing.  :D
17:05.33iCEBrkrtzanger: you just don't know it yet
17:05.40tzangerahh yes I forgot about that
17:05.41SomeOne1does that work?
17:05.46SomeOne1like if you dial like, 1-202-911?
17:05.56astawerksdotcomyippe i get to work in pittsburgh today!
17:06.00MercestesSomeOne1, No, not really.  Try it.
17:06.04SomeOne1(202 is Wash DC)
17:06.13tzangerno it doesn't work
17:06.13astawerksdotcom703 is arlington county
17:06.20MercestesYou think they *never* use "911" as part of a number.  That'd be nearly impossible.
17:06.20SomeOne1703 is northern VA
17:06.24tzangerthink of how many phone numbers here start with 519-291-1xxx?
17:06.25tzangera ton
17:06.25SomeOne1its not just arlington county
17:06.29tzanger1000 to be exact
17:06.30SomeOne1757 is virginia beach
17:06.33SomeOne1540 is southern va
17:06.37SomeOne1804 is richmond
17:06.46iCEBrkrI had a friend who lived in Fairfax.. 703.
17:06.46astawerksdotcomva beach is fun.  seen lots of interesting things there
17:06.57SomeOne1iCEBrkr: i live in fairfax!
17:06.59iCEBrkrwell I suppose he was between FairFax and Minassas<sp>
17:07.02dc3aesmy phone company just assigned me a XXX-XXX-0000 number
17:07.07SomeOne1therefor, with the law of transitivity, i must be your friend!
17:07.11dc3aesi swear i thought those were test numbers
17:07.13iCEBrkrSomeOne1: shit
17:07.16tzangerI have a PRI block that's 0001-0030
17:07.17Putzz0000 I ddidnt think they did that
17:07.19tzanger0000 was taken
17:07.26*** join/#asterisk Ifaistos (n=stelios@ipa226.211.tellas.gr)
17:07.31dc3aesi swear, my blackberry has xxx-xxx-0000
17:07.32bochhelp, do you know why my * is not runing the 'h' ext when hang?
17:07.33iCEBrkrtzanger: That's your network number :)
17:07.38tzangeriCEBrkr: hahaha
17:07.47tzangerEnterprise rent-a-car has my network number?
17:07.54iCEBrkrhaha
17:07.56dc3aesnobody ever believes me that its even my number.. and im like lemme call you so you can see
17:07.57tzangerI have a pic of my odometer at 127001km somewhere
17:07.58iCEBrkr0030 is the broadcast
17:08.14SomeOne1boch: do you use the Dial() application
17:08.20PutzzI have 31337 pic on my truck
17:08.21Putzz;-)
17:08.24SomeOne1Dial needs to be told to go and execute more commands
17:08.27Putzzodometer reading
17:08.29SomeOne1after the dial is terminated
17:08.32iCEBrkrI have jackasses old cell number..I keep getting a LOT of random calls from people and collectors
17:08.35tzangerand actually I missed 224xxx, I could have had some multicast addresses in there
17:08.38astawerksdotcommy 97 buick odometer says error!.   im am going to play it off wen i sell it    says it has  5000k
17:08.38tzangeriCEBrkr: I had that
17:08.56bochSomeOne1, but it is an incoming zap call
17:09.05SomeOne1boch: oh i dont know, sorry
17:09.08SomeOne1paste some logs
17:09.11SomeOne1or debug stuff
17:09.14SomeOne1in the pastebin
17:09.20iCEBrkrI'm thinking about getting my number changed, because this is going to turn into some scene from Amazon Women on the Moon.
17:09.31*** join/#asterisk grantm (n=grantm@kolob.wingateservices.com)
17:09.35SomeOne1i want like 1-000-000-0000
17:09.40boch<PROTECTED>
17:09.40bochReally destroying SIP dialog '3039465b-2bfb-db11-8df7-0017318807f2@localhost' Method: BYE
17:09.42SomeOne1can you buy that number?
17:09.48bochthats all..
17:09.56tzangerblitzrage has the best cell phone # ever
17:10.05PutzzI wouldnt care about all 0's but a number thats like 905-111-1111
17:10.07Zeeekthe iPhone is out?
17:10.08Putzzor something like that
17:10.16SomeOne1or like, 1-703-000-0000
17:10.16tzangerit's three digits, 5, 1 and 9, and the other 7 are those digits rearranged
17:10.23tzangerI offered to buy it from him but he's not giving it up
17:10.40*** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-88-152-182-238.red.bezeqint.net)
17:10.46SomeOne1can you import them to like DIDs?
17:10.52SomeOne1like can i import my cell phone number?
17:11.03tzangeryep
17:11.03dc3aesi have harder time with multiple numbers.. you have to count them as you dial.. i always screw em up
17:11.27tzangerin fact, I have 5 or six DIDs that were once business FL1 lines, and one (my old residential #) converted to a  business line and made a DID when I got the PRI
17:11.47SomeOne1how much does the PRI cost?
17:11.58tzangerdepends on which "rate group" you're in
17:11.58SomeOne1ahh, youre in canada
17:12.07SomeOne1what about yours?
17:12.12SomeOne1$800/month USD?
17:12.13tzangerI'm in buttfuck nowhere so I'm in the "rape me without lube" rate group
17:12.20SomeOne1haha
17:12.21Apheliono.O
17:12.21iCEBrkrAll I need is a cheap Los Angeles DID
17:12.23Putzz800 a month? no way
17:12.36SomeOne1o.0
17:12.36tzangerI pay about CAD$700 for 15B+D
17:12.44SomeOne1whats that in USD?
17:12.49tzanger*0.9
17:12.50Putzz690
17:12.51Putzzlol
17:13.04iCEBrkrSomeOne1: either way, that's fairly cheap
17:13.06SomeOne1interesting
17:13.07tzangerif I were in Kitchener or Toronto I'd get it for under $400 for a full 23B+D
17:13.09SomeOne1do you need it that way?
17:13.11SomeOne1i mean
17:13.18SomeOne1do you really need it?
17:13.21SomeOne1for business?
17:13.23SomeOne1or something
17:13.37iCEBrkrSomeOne1: he runs an adult party line.. yes.. :P
17:13.42SomeOne1ahhh
17:13.43SomeOne1nice
17:13.48tzangerI love PRI
17:13.50SomeOne1canadian porn
17:13.55SomeOne1phone porn
17:13.56tzangeranalog is teh suck
17:13.58iCEBrkrhence his number being plastered on bathroom walls.
17:14.03tzangeriCEBrkr: hahahaha
17:14.10Zeeekif you wanted to point a friend to your show, you could send them to http://wineloverspage.com/talkshoe/latest.htm
17:14.14NuggetFlightAware is about to graduate to a PRI and I'm trying to wrap my head around the hardware options.  Anyone have any advice?
17:14.26Zeeekok I need to shut this window...
17:14.33SomeOne1verizon runs fiber to my house
17:14.38SomeOne1beat that!
17:14.47SomeOne1its called verizon FiOS
17:14.49*** part/#asterisk Zeeek (n=IceChat7@pdpc/supporter/active/Zeeek)
17:14.50Nuggetmy TDM400P has me a bit spooked by the Digium stuff, although I'd certainly prefer to support them.  Would I be happier with a Sangoma card?
17:14.58Nuggetand is echo cancelling something I want?
17:15.01tzangerNugget: just get a T1 card WITH ECHO CAN or get a tellabs can off ebay and do some wiring
17:15.08tzangerdon't go for cheap echo can, and don't go for software echo can
17:15.20Nuggethow can I tell which is cheap and which is expensive?
17:15.21tzangerTE405 and 407 have been ROCK SOLID for me
17:15.30NuggetI'm not particularly price-sensitive
17:15.38tzangerthat's a quadspan though
17:15.55*** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir)
17:15.57Nuggetthat's encouraging to hear
17:15.58tzangerI think Sangoma's the only one with a single/dual span with HW echo can... I think
17:16.06SomeOne1Qwell: how come youre OP in this channel
17:16.17tzangerbut yeah $80 for a tellabs 64ms echo can on ebay and some wiring, and you should be golden with a TE110P
17:16.24Qwell[]SomeOne1: why wouldn't I be?
17:16.28MercestesSomeOne1, because he's 31337
17:16.35SomeOne1i dunno
17:16.42SomeOne1i want to be OP
17:16.45tzanger*HIS* phone number ends in 1337.  :-)
17:16.52SomeOne1heh
17:17.05SomeOne1Qwell: does it feel powerful?
17:17.17PutzzQwell is tha man
17:17.23Nuggetit would appear that a TE212P is my minimum entry to get the DSP-based echo cancellation in a Digium card.
17:17.25PutzzQwell u work for digium dont u?
17:17.30Qwell[]Putzz: yes
17:17.37NuggetI really don't understand the products, though
17:17.37SomeOne1Qwell: can i get a job?
17:17.39tzangerNugget: make sure you get Octasic echo can, not the Oki one...
17:17.50Nuggetis a 120 better than a 212?  20 is higher than 12.  :)
17:17.51filetzanger: we only sell the Octasic one now
17:17.57Qwell[]SomeOne1: you have to apply first...
17:18.02tzangerfile: that's excellent :-)
17:18.02iCEBrkrNugget: does it cost more? of course it's better!
17:18.05Nuggetheh
17:18.07SomeOne1resumes@digium.com ?
17:18.15SomeOne1i got a lot of c++ exp
17:18.19iCEBrkrSomeOne1: um, that's not what he meant.
17:18.23Qwell[]SomeOne1: or jobs@digium.com, or you could look on the website...
17:18.29iCEBrkrSomeOne1: you gotta get your 'knees dirty' I think is what he was talking about
17:18.32Qwell[]it's kinda a test...and so far, you're failing it :P
17:18.41SomeOne1:(
17:19.10SomeOne1are they even hiring really?
17:19.26SomeOne1have you ever met mark spencer?
17:19.29Qwell[]SomeOne1: again - you could look on the web site...
17:19.29tzangerI have
17:19.33tzangerhad dinner with him even
17:19.46iCEBrkrtzanger: liar, you're not cool enough for Mark
17:19.50Qwell[]What a silly question. :P
17:19.52tzangerthen last week I had dinner with file and kpfleming too
17:20.11tzangeriCEBrkr: heh
17:20.13fileI will neither confirm or deny that
17:20.14iCEBrkrYeah well, I drank pitchers of beer with Damin.
17:20.22iCEBrkrfile: good man
17:20.30tzanger:-)
17:20.49tzangerfile drank pop...  I think he'd be an absolute scream if I could get him drunk
17:20.53SomeOne1Qwell: so you HAVE met him?
17:20.56SomeOne1he is cool?
17:21.01fileI get giddy and happy...
17:21.03Qwell[]SomeOne1: of course I've met him
17:21.05fileand laugh easily.
17:21.09tzangerfile: heh
17:21.19tzangerso not much different from normal, except you can't code as fast
17:21.25SomeOne1is he full of himself?
17:21.26filequite
17:21.35iCEBrkrtzanger: come'on, you heard about Kristian when he got all liquored up?
17:21.40tzangerno...
17:21.40Aphelionugh, i need help... a client wants me to create a website login to show the minutes used by an accountcode, but when i asked how one would properly authenticate the user he suggested the voicemail password for one of the extensions.....
17:21.44tzangerI'm out of the loop on these things
17:21.55Qwell[]iCEBrkr: oh man, that boy can drink
17:21.59iCEBrkrtzanger: haha, he blew chunks in docelmo's car.
17:22.00Aphelionexperts: what's the clean and proper way to do what i've been asked to do?
17:22.04tzangerewwwwwwwwwwwwwwww
17:22.08tzangerthat doesn't sound like he can drink
17:22.19iCEBrkrtzanger: Well, from what i hear, he was drinking alllll night
17:22.19Aphelionwow, interesting conversation >_>
17:22.20Qwell[]well, see...here's the thing
17:22.22iCEBrkrAphelion: heheh
17:22.28Qwell[]when you chug like a 5th of vodka...
17:22.34tzangerI get loud(er) and speak more easily, unless I've been drinking gin
17:22.40tzangerin which case I get mean
17:22.51filetzanger r0x0rz
17:22.51*** join/#asterisk fakhir (i=fakhir@unaffiliated/fakhir)
17:22.52SomeOne1i had 8 shots of 151 in 10 minutes once
17:22.55SomeOne1puked my guts out
17:23.00SomeOne1but at least i won the bet :)
17:23.01tzangeruh yeah...
17:23.01iCEBrkrno good
17:23.01tzangerheh
17:23.04*** join/#asterisk alexhopper (n=a27386@mctnnbsa24w-142167035146.pppoe-dynamic.nb.aliant.net)
17:23.04iCEBrkrew
17:23.25iCEBrkrI'll make it to a Astricon one year. :-/
17:23.29SomeOne1got props from everyone and felt cool for a while
17:23.30tzangerindeed
17:23.31SomeOne1thats all
17:23.36tzangerI need to get to more taug meets
17:23.42Putzzanyone here went to IT360?
17:23.46Putzzexpo
17:23.47filePutzz: yes
17:23.48SomeOne1you guys are so cool
17:23.51tzangerI did
17:23.54SomeOne1all about expos
17:23.55tzangerthat's where I met file and kpfleming
17:23.56SomeOne1and drinking
17:23.56PutzzI was there ;-)
17:23.58SomeOne1and meeting up
17:23.59tzangerand a bunch of thers too
17:24.04tzangerPutzz: who are you?
17:24.06iCEBrkr...some how I missed the VoIP Conference here in Ft. Lauderdale.
17:24.09PutzzI was in the Taug meeting too
17:24.13Qwell[]tzanger: you coming to astridevcon?
17:24.14tzangerI wonder if I met you and didn't realize it
17:24.22tzangerQwell[]: I don't think I can get the time off
17:24.22Putzzpossibly
17:24.26SomeOne1<PROTECTED>
17:24.27Qwell[]suck
17:24.36Qwell[]SomeOne1: eh?
17:24.36tzangerPutzz: you probably bet me, I was on the panel about asterisk war stories
17:24.41SomeOne1yep
17:24.41tzangerer met
17:24.44iCEBrkrtzanger: i'll take your place, I'm just as much of a hack as you :-P  hahahaha
17:24.50tzangerhahahaha
17:24.52Qwell[]file: I bet you a tzanger that you can't...
17:24.52tzangerthere we go
17:24.54Qwell[]something
17:24.59*** join/#asterisk joebob777as7 (n=richard@65.103.68.176)
17:25.00tzangeryou bet me what?
17:25.01iCEBrkrlol
17:25.02tzangerhahaha
17:25.03filelol
17:25.14tzangerdamn I'm getting passed around as currency now
17:25.14SomeOne1okay gotta get backj to work
17:25.14Qwell[]tzanger: nothing, I just used you as currency
17:25.25SomeOne1im an oracle developer
17:25.26fileQwell[]: I'll trade you tzanger for a muffin
17:25.26SomeOne1actually
17:25.35tzangerman and not much currency either
17:25.48Qwell[]well, it *is* a chocolate muffin
17:25.51PutzzCAD currency less taxes
17:25.52Putzzheh
17:25.56Qwell[]that's gotta count for something
17:26.08*** join/#asterisk nDuff (n=ccd@fw2.isgenesis.com)
17:26.08tzangerheh
17:26.10SomeOne1are you guys going to the porn expo in Las Vegas?
17:26.10iCEBrkrAll I gotta say is
17:26.12iCEBrkrMOOSE PENIS
17:26.21*** join/#asterisk SwK (n=SwK@wsip-68-98-207-182.ks.ok.cox.net)
17:26.21neverbluewhen 'sip debug' is enabled, does Asterisk still have activity (checking that the server is still online/running -- just an example)?
17:26.38SomeOne1neverblue: of course
17:26.42neverblueok
17:26.46mostyneverblue: turning debug on doesn't turn anything off
17:26.52SomeOne1are you guys going to the porn expo in Las Vegas?
17:26.58Mercestesneverblue, except your girlfriend
17:26.59SomeOne1mosty speaks?!
17:27.08SomeOne1i wish i had a girlfriend
17:27.10neverbluemosty I wasnt implying that
17:27.13SomeOne1im gonan find one at the porn expo
17:27.31SomeOne1PORN EXPO
17:28.02neverblue--- (0 headers 0 lines) Nat keepalive --- <--- this is confirming the connection?
17:28.26fileneverblue: that is a SIP device sending an empty packet to keep the NAT mapping alive
17:28.57SomeOne1why wont anyone acknowledge the porn expo thing
17:29.01neverblueDestroying call '3aaff3806dae60b77e55c0981556081d@192.168.99.76'
17:29.01neverblueDestroying call '6619d89a164dbd1576ae509f301420ae@127.0.1.1'
17:29.04neverblueand that is?
17:29.16filean internal debug message indicating that the SIP dialog is gone now
17:29.19SomeOne1neverblue: i'll tell you if you go to the porn expo
17:29.19nDuffI'm trying to add some dynamic features to the [applicationmap] section of features.conf, but a "module reload res_features" says nothing about them (and they don't show up in "show features" afterwards). Is there anything obviously wrong about my expectations?
17:29.27fileand a message that totally freaks out non-developers
17:29.48Mercestesand me.
17:29.50neverbluefile, that was a reponse to my question?
17:29.51*** join/#asterisk Math` (n=seb@modemcable234.87-70-69.static.videotron.ca)
17:29.54fileneverblue: yes
17:29.55neverblueresponse*
17:30.08neverbluethanks file
17:30.14Math`can iax2 register => lines be included in realtime?
17:30.24SomeOne1sip > iax > h323
17:30.25fileMath`: no.
17:30.26SomeOne1h323 sucks
17:30.27SomeOne1i hate it
17:30.42Math`file: any particular architecture reason why that is?
17:30.57SomeOne1Math`: whats the square root of -1
17:31.03*** join/#asterisk SwK (n=SwK@wsip-68-98-207-182.ks.ok.cox.net)
17:31.04tzangerSomeOne1: i
17:31.04Math`i
17:31.04Math`:P
17:31.23Putzzipeeppee
17:31.27Corydon-wSomeOne1: because implementing would make the realtime dialplan even slower
17:31.29Putzzsry im bored
17:31.43tzangerPutzz: so where at the taug meeting were you
17:31.44Mercestesasterisk 1.2.13.  any bugs in detecting DTMF?
17:31.50Corydon-wSomeOne1: and it's god-awful slow enough already
17:32.03SomeOne1Corydon-w: you mean putting matching for IP= in the dialplan?
17:32.07Math`Corydon-w: or at least make it load up those register string when you prune realtime
17:32.14fileMercestes: not enough info, you haven't actually said using what technology
17:32.41Math`anyways, I'll just write a file and send an iax2 reload via ami for now
17:32.44Mercestesfile:  Sure, incoming PRI using a cell phone.
17:33.18Corydon-wCell phones are notorious for not being within specs for DTMF
17:33.25Mercestesfile:  I'm testing reports that the extensions dialing doesn't work for our menu.  Out of about 20 times of dialing "4915" I got one detection of "4911" instead.
17:33.50MercestesCorydon-w, Agreed, but these issues are on various CLEC land lines.  Qwest, Cox, and one "Southern Telephone."
17:33.51Math`I wonder why cells are like that tought.... is it sent inband over gsm?
17:33.59MercestesCorydon-w, and I hav enothing else to test with
17:34.22Corydon-wNo, cell phones send it digitally and it's translated to DTMF at the landline
17:34.40Math`thats better... so theoretically it shouldnt matter
17:34.43MercestesCorydon-w:  Ok, and exactly at what point are you disagreeing with me??
17:34.48Putzztzanger: far back somewhere
17:34.49Putzzu?
17:34.52joebob777as7can someone help me I have some simple questions... I am wanting to have 3 phone lines and about six phones in our new office. What hardware should I get? Should i get a voip router? etc... and what phones do you guys recommend?
17:34.52tzangerhaha
17:34.54tzangerright up front
17:35.09PutzzI was one of the first to walk in
17:35.13iCEBrkrI can't use my cellphones speaker phone option when dialing Asterisks voicemail... DTMF gets all jacked up
17:35.18tzangerthen what were you doing at the back?
17:35.30Corydon-wMercestes: have you tried relaxdtmf=yes yet?
17:35.38*** join/#asterisk nybble (n=nybble@about/apple/performa/nybble)
17:35.40Putzzdunno
17:35.47MercestesCorydon-w, Hm...Lemme check
17:36.18Putzztzanger: u dont usually go to taug meetings do u?
17:36.25tzangerPutzz: not often no
17:36.28tzangertrying ot go to more
17:36.53Putzzwe all go to the bar after
17:36.55MercestesCorydon-w, um.. No, no I haven't.  if that works I guess I'll have to be nice to you. :)  (for awhile)
17:36.55Putzz;-)
17:36.57*** join/#asterisk potsboy (n=jsoap@c5-386-1.vic.dial.mweb.co.za)
17:37.42*** join/#asterisk dacter (n=dlittrel@207.200.33.213)
17:37.44*** join/#asterisk SwK_ (n=SwK@wsip-68-98-207-182.ks.ok.cox.net)
17:37.59tzangerPutzz: oh I know
17:38.03tzangerI went to the first few
17:38.11tzangerstopped going for a long time
17:38.15tzangerthen went again at it360
17:38.21dacterquestion... is it possible to make a softphone-to-softphone call after editing nothing more than sip.conf?
17:38.23tzangerI wasn't going ot be at it360 but someone convinced me to go
17:38.25Putzzright on
17:38.32Putzzit was worth it
17:38.43tzangerit's not bad
17:38.54Putzzcould have been better tho
17:38.55tzangerI think Jim's got me convinced to do more talks
17:38.57MercestesCorydon-w, Thanks, I'll give it a go.
17:39.49[TK]D-Fenderdacter: Not throught Asterisk, no.  Without a dialplan you aren't getting ANYWHERE.
17:41.03file[TK]D-Fender: !!!
17:42.23[TK]D-Fenderfile: I DON'T WANT TO BE AT WORK!
17:42.36file[TK]D-Fender: :(
17:42.38file[TK]D-Fender: GO HOME!
17:42.40joebob777as7can someone help me I have some simple questions... I am wanting to have 3 phone lines and about six phones in our new office. What hardware should I get? Should i get a voip router? etc... and what phones do you guys recommend?
17:43.42*** join/#asterisk sevard (i=chuck-th@adsl-71-129-115-242.dsl.irvnca.pacbell.net)
17:45.46apturaTK work will not allow you to remote in?
17:49.39Mercestesaptura:  Of course, after hours after he's already worked his 8 hours on location.
17:50.08*** join/#asterisk xpot (n=jim@c-71-195-241-115.hsd1.ut.comcast.net)
17:50.49*** join/#asterisk oej (n=olle@apollo.webway.se)
17:51.03*** join/#asterisk tsurko (n=tsurko@77.70.24.142)
17:51.07[TK]D-Fenderaptura: Was a play on words for an in-joke I have between file & blitzrage .... which he has clearly missed!
17:52.09apturaTK I see :) with price of fuel reaching 4.75 per imp gallon people in this province are starting to change driving habits ect.
17:54.48fileI miss nothing.
17:55.08*** join/#asterisk Hmmhesays (n=Neg@24-117-131-41.cpe.cableone.net)
17:55.17*** join/#asterisk IOscanner (n=IOscanne@216.88.109.2)
17:55.49*** join/#asterisk jsolares (n=jsolares@206.113.226.107)
17:56.37Putzzany of u guys know a sip softphone for PDA other then SJphone?
18:00.12DefrazX-Ten
18:00.15Defrazmakes one right?
18:00.46[TK]D-Fenderaptura: I live so close to work that I don't really care much, and I'm bringing my bike into the shop for a complete tune-up.  Will care even less shortly :)
18:02.33iCEBrkrIf riding my motorcycle wasn't such a chore, I'd ride to work
18:02.53iCEBrkrBut having to gear up, sit in traffic, sweat and then change when I get to work.. yuck
18:02.54Hmmhesaysi used to ride mine to work all the time
18:03.13iCEBrkroh, and since I'm in an apartment i have to go get my bike out of storage each time
18:03.17iCEBrkrpain!!
18:04.12Hmmhesaysi need a new battery for my dell
18:04.47neverbluecan someone help a noob, i just have a few general questions, (I have read the documentation)
18:04.56iCEBrkrsure you did
18:04.57neverbluein pm preferably
18:05.46Putzzin pm?
18:05.51Putzzeveryone is here to learn
18:05.54Putzzshare with all of us
18:05.55Putzz;-)
18:06.02Putzzlearn/share
18:06.08Hmmhesaysi'm here to ridicule
18:06.10iCEBrkrneverblue: yea, don't but a putzz
18:06.11apturaTK Same here. I am just a bit fustrated by city planners to not really have dedicated bike paths though our city. Also we have a BIG ethnic minority that has pretty much a run of the city. The norm here is thay buy these small homes and then crush them with a dozzer and build huge 4-5 thousand square foot homes in its place. 20,000 plus homes here have had this happen to them. Seattle on the other hand had great access to dedicated tra
18:07.05apturaIF this was the case for the founder of Efnet he would perhaps be still alive today.
18:07.17apturaI mean freenode
18:07.29iCEBrkrVrrrrrrrrrrrrrrrrrrrrroooooooooom
18:07.46apturaice ride a motorcyle?
18:07.50iCEBrkraptura: yeah
18:07.59[TK]D-Fenderaptura: Luckily the path between me and the office is pretty clear
18:08.09*** join/#asterisk anonymouz666 (n=anonymou@re366.compuland.com.br)
18:08.17iCEBrkrand I won't get into the politics and options of Freeload.. err. I mean, Freenode
18:08.22iCEBrkrerr
18:08.24iCEBrkropinions
18:08.25apturaI have plans to make a hybred cross diesel motorcyle that is three wheeled and enclosed like a car.
18:08.42iCEBrkraptura: you mean like a T-Rex? :)
18:08.45Corydon-wiCEBrkr: the guy at the center of that controversy is DEAD
18:08.58iCEBrkrCorydon-w: I know this..
18:09.07apturanot really like T-rex its going to have a very low Cd paracitic drag ratio.
18:09.13Corydon-wLet him rest
18:09.36iCEBrkrI repeat.. I won't get into the politics and opinions of this.
18:09.50dc3aesI just had a conversation with someone on my asterisk box and 50% of the call was garbage due to compression artifact/delays/etc.. im curious now that im in the asterisk game.. if I can analyze logs as to why.. I have an 8mbps down, 3mbps up broadband here and i was using IAX2
18:10.07*** join/#asterisk axisys (n=axisys@155.70.141.45)
18:10.09iCEBrkrdc3aes: Turn off your Bittorrents
18:10.12dc3aeshaahhha
18:10.14dc3aessssshhhh
18:10.16[TK]D-Fenderdc3aes: 3mbps UPSTREAM?!
18:10.16apturaI run biodiesel out of fustration with the dam oil companies
18:10.18*** join/#asterisk adorah (n=Michael@87.69.57.73.cable.012.net.il)
18:10.20dc3aesits only 1 doing 10kbit
18:10.43[TK]D-Fenderdc3aes: Have you considered it might be the OTHER END's fault>?
18:10.44neverblueis there any "better" reference materials online other than the voip-info.org's 'The Asterisk Manager API' for using php with asterisk via the Manager?
18:11.17dc3aesdamn.. i lied.. its 10mb down, 1mb up.. I swear they changed that on me.. reading their marketing now
18:11.23dc3aes[TK]D-Fender: i called a landline
18:11.35dc3aesim using nufone as the peer/trunk
18:11.36[TK]D-Fenderdc3aes: Via what?
18:11.41iCEBrkrWell regardless, pretty much and broadband connection should work just fine for VoIP
18:11.43[TK]D-FenderHrm.
18:12.06dc3aesthis is true.. however dont get me started with my vonage problems last year lol...
18:12.07[TK]D-Fenderdc3aes: Try another link of some kind.
18:12.22potsboyneverblue, what are you trying to achieve and with the manager?
18:12.35dc3aesya I will try my les.net trunk.. i used it for 2 hours last night and was like wow this is good... two calls today during "business hours" and its junk.. so i suspect inet congestion
18:12.45*** join/#asterisk lpaz (n=paz@c83-251-203-30.bredband.comhem.se)
18:12.49neverbluepotsboy, using 'Action: Originate'
18:13.34dc3aesI just wish there was a way of logging disruptions..
18:13.53potsboywhy you mention php?? are you scripting a orginate?
18:13.56*** part/#asterisk Joe_CoT (i=joe_cot@powerade.dreamhost.com)
18:14.21neverbluepotsboy, do you know of any other reference materials on the Manager?
18:14.30xkevanyone have stats on residential usage for cost analysis?  local minutes, inbound minutes, US long distance minute averages
18:14.32*** join/#asterisk alexpe (n=alex@cev75-1-81-57-14-91.fbx.proxad.net)
18:14.52*** join/#asterisk DeeJayTwo (n=deejay2@office.abi.ca)
18:15.06potsboyi have done some work in perl ... check cpan, an have used a custom hacked up astmanproxy.. that the best i can suggest
18:15.12*** join/#asterisk SwK (n=SwK@65.192.110.34)
18:15.31DeeJayTwohas anybody used siproxd for simple proxying to an asterisk system?
18:15.33neverbluepotsboy, what does it do?
18:16.01DeeJayTwoI've got IP phones behind a NAT box which also runs siproxd but I can send calls from asterisk to these phones (by the public NAT IP)
18:16.02potsboyyou can control via the AMI or proxy it
18:16.15PseudoNimhm
18:16.21neverbluepotsboy, not sure thats what I am looking for
18:16.23PseudoNimis there any reason why asterisk would disconnect when i'm calling Red()
18:16.25PseudoNimread()
18:16.30neverbluepotsboy, thanks though
18:16.39PseudoNimif i dial the numbers fast, it disconnects without getting the entire string, and if i dial slowly, it works fine
18:16.44potsboywell am a little rusty at mind reading these days
18:17.05neverbluewhat is that suppose to mean?
18:17.09LeddyHMAnyone know when color was introduced in the cli?
18:17.27apturaman this version is bugy. I need to upgrade it. zap went down unknown to me and now its up.
18:17.35*** join/#asterisk digus (n=digus@206.222.110.30)
18:17.38LeddyHMI have a 1.2.4 that has no color, trying to figure out if it was a command line switch, or version change that implemented that feature
18:17.52Qwell[]LeddyHM: 1.2.4?
18:17.53Qwell[]upgrade
18:17.57potsboyneverblue, leave it i am a little cranky.. apologies
18:18.10LeddyHMqwell: I am in the process of trying to get it approved :)
18:18.25LeddyHMbut in the interim, was wondering if it was an available option
18:19.07LeddyHMit is much easier to read hence the asking :)
18:19.31*** join/#asterisk yannj_fr (n=yannj@choisy.intelunix.fr)
18:19.38yannj_frHi all
18:19.49*** join/#asterisk jm|laptop (n=jm|home@dilbert.jamiem.com)
18:20.28yannj_fris ther somebody that have enought time to answer my questions about asterisk module developpement
18:20.49neverblueanyone else, know of some good documentation on the Asterisk Manager? (other than the API on voip-info.net)
18:21.08Math`what else do u want
18:21.22Hmmhesaysthats probably the best documentation  you'll get
18:21.22Math`you can also show manager command [xxx]
18:21.28Math`or core show manager command.... if you are in 1.4
18:21.29Hmmhesaysit is pretty thorough
18:22.33neverblueso there is nothing else?
18:22.40neverbluethat you know of...
18:23.15dc3aessheesh.. i figured it out I think.. I had commented out the nufone trunk to use the Les.Net trunk last night.. I went back to nufone and its working fine so its 100% les.net's end..
18:24.03MindTheGapERROR[5121] res_config_ldap.c: Failed to init ldap connection to l:389. Check debug for more info.
18:24.11*** join/#asterisk jm|laptop (n=jm|home@dilbert.jamiem.com)
18:24.17MindTheGapanyone got this?
18:24.38Math`Check debug for more info.
18:24.41MindTheGapheres the res_ldap.conf; http://pastebin.ca/475647
18:24.41Math`try that
18:26.31MindTheGaplike with -vvvvvvvv ?
18:26.36shido6cool
18:26.42shido61 pt for us
18:26.43*** join/#asterisk phillipk (n=pkey@fw.datafax.net)
18:26.50Math`MindTheGap: 1.2 or 1.4
18:26.52MindTheGapshows nothing more than this
18:26.57MindTheGap1.4.4
18:27.11Math`<PROTECTED>
18:27.15*** join/#asterisk Gpl_Source (n=The_natu@unaffiliated/gplsource)
18:27.18Math`and then try reloading the module
18:27.36Math`module reload res_config_ldap
18:29.13MindTheGapParsing '/etc/asterisk/res_ldap.conf': Found
18:29.13MindTheGap[May  7 16:30:44] ERROR[5154]: res_config_ldap.c:1744 ldap_reconnect: Failed to init ldap connection to l:389. Check debug for more info.
18:29.13MindTheGap[May  7 16:30:44] WARNING[5154]: res_config_ldap.c:1606 reload: Couldn't establish connection. Check debug.
18:30.03Math`MindTheGap: check the end of /var/log/asterisk/full
18:30.06Math`it should contain more info
18:31.37MindTheGapait got asterisk/full just /messages and theres nothing more there too
18:32.41*** join/#asterisk johnchristopher (n=grump@70.151.90.227)
18:32.49MindTheGapaint got asterisk/full just asterisk/messages
18:34.45*** join/#asterisk |dennis| (n=dennis@200.32.233.82)
18:35.08MindTheGapif i comment out "port=389" on res_ldap.conf i get:
18:35.26MindTheGap[May  7 16:36:29] WARNING[5234]: res_config_ldap.c:1655 parse_config: No directory port found, using 389 as default.
18:35.26MindTheGap[May  7 16:36:29] WARNING[5234]: res_config_ldap.c:1767 ldap_reconnect: bind failed: Invalid DN syntax
18:35.26MindTheGap[May  7 16:36:29] WARNING[5234]: res_config_ldap.c:1606 reload: Couldn't establish connection. Check debug.
18:35.27anonymouz666${CALLERIDNUM} is read-only var?
18:35.41anonymouz666I need to set ${CALLERIDNUM:1} on CDR
18:35.46MindTheGapwierd
18:35.54*** join/#asterisk SoftIce (n=bongo@vc-196-207-45-253.3g.vodacom.co.za)
18:36.23Math`anonymouz666: thats deprecated... use ${CALLERID(number)}
18:36.45anonymouz666ok, even with that.
18:37.15anonymouz666${CALLERID(number):1}
18:37.43anonymouz666the CDR does not seem to care about the :1
18:37.44SoftIcehi, hmmm, can somebody tell me of some billing system, that has multi user level, so I can have 1 db for instance and have a user level for each user with say a softphone and to be able to check their balance
18:37.57SoftIcea2billing, doesn't have multi level
18:40.18*** join/#asterisk Deeewayne (i=dwayne@nat/digium/x-b1445f4a7ea123c7)
18:41.09*** join/#asterisk FuriousGeorge (n=brian@ool-43536ea8.dyn.optonline.net)
18:42.27*** join/#asterisk Uranellus (n=alexande@p57A16263.dip.t-dialin.net)
18:42.56*** join/#asterisk ryguillian (i=rhayes@numbertwo.midphase.com)
18:42.57Uranellushow do I figure if the modem card I have works with asterisk ? I've got a bcm94212
18:43.09Qwell[]Uranellus: it doesn't
18:43.15anonymouz666the src in CDR does not change
18:43.17anonymouz666the callerid is ok
18:43.28UranellusQwell[]: is there a list somewhere ?
18:43.36Qwell[]Uranellus: x100p
18:43.44Qwell[]that's the list.  any we don't recommend using it
18:43.47Qwell[]and*
18:44.18UranellusQwell[]: what kind of card would be recommended ?
18:44.19jerso what do you recommend using?
18:44.23Math`for a single line if you want my opinion you're better off with an external gateway... which is going to cost less than digium's tdm400p
18:44.28Qwell[]Uranellus: What do you need?
18:44.44Math`or else the tdm400p can have 4 modules (either fxo(line) or fxs(phone))
18:45.30UranellusQwell[]: actually I  just have 2 pcs at home.. one near the phone jack .. now i wanted to phone from the other comp through the one close to the phone jack .. so I don't really know what I need for this ..
18:45.54idoi heard ryguillian has a huge ...
18:45.57Qwell[]Uranellus: You don't even need asterisk for that
18:45.59idoprocessor
18:46.00Qwell[]just softphones
18:46.17*** join/#asterisk dacter (n=dlittrel@207.200.33.213)
18:46.21Hmmhesaysgeebus linux wireless can be a pain in the ass
18:46.37bkruseQwell[]: res_56kmodem?
18:46.51bkruselol
18:46.54bkrusebut its not my birthday!
18:46.55bkruse:P
18:46.58Qwell[];)
18:46.59bkruseHmmhesays: yes it can :[ madwifi!
18:47.18Qwell[]I will kill you.
18:47.23Qwell[]I will kill you until you die.
18:47.26MindTheGapMath, if i set "host=10.0.0.222,ldaptest.lpj.com.br" as sugented on the .conf file the error is:
18:47.41MindTheGap<PROTECTED>
18:47.42UranellusQwell[]: well just a softphone .. any other hints? :/
18:47.48bkruseQwell[]: or i can just go get another one
18:47.53Qwell[]bkruse: that'd work :p
18:47.55bkruseomg, Qwell does the dollar work now?
18:48.00Qwell[]no idea
18:48.03bkruseCURSES
18:48.05Qwell[]I bought mine from the gas station, heh
18:48.10bkrusegah, owned.
18:48.11Math`MindTheGap: connection fails... check your syntaxes
18:48.16MindTheGapif only one argument is given, like host=10.0.0.222 it says:
18:48.19Qwell[]probably not though
18:48.22Qwell[]stupid vendor
18:48.30Qwell[]they suck as much as the one we had at the Atrium...
18:48.34Strom_Mbkruse, Qwell[]: I'll be in lolabama again next week
18:48.35Qwell[]...it's probably the same one ;/
18:48.40MindTheGap<PROTECTED>
18:48.51Qwell[]Strom_M: nice
18:48.56Qwell[]bootcamp?
18:49.18bkruseStrom_M: want to go get some white milk from cheeburger?
18:49.22bkruseand go bowling?
18:49.33bkruseQwell[]: totally, same lazy dude i bet.
18:49.35bkruseour new building wont be like that
18:49.41bkruseme and russellb are making a lemonade stand
18:49.45Qwell[]5c?
18:50.01MindTheGapMath, the conf is here: http://pastebin.ca/475647 would you please take a look?
18:50.01bkruseyes, and we will always accept dollars, even crinkly onces
18:50.03bkruseones*
18:50.10Math`MindTheGap: I dont use ldap... so I cant help you thee
18:50.12Math`there*
18:50.20MindTheGapoh, sorry...
18:50.23filebkruse: will you always hand out lemonade? not randomly other drinks?
18:50.35MindTheGapanyone usnfg ldap?
18:50.40MindTheGapusing*
18:50.49*** join/#asterisk nasls_lsa (n=chatzill@87.203.68.253)
18:50.54bkrusefile: possibly drpepper and vault.
18:50.59*** join/#asterisk fbffff (n=fbffff@r74-193-79-81.pfvlcmta01.grtntx.tl.dh.suddenlink.net)
18:51.08Qwell[]bkruse: rockstar!
18:51.16bkruseQwell[]: we totally should!
18:51.17Qwell[]and redbull, naturally
18:51.20bkruseof courser.
18:51.22bkrusecourse*
18:51.22bkrusewhat do you prefere jcolp!?? muffin flavored soft drinks?
18:51.33Qwell[]and tab for mog
18:51.35Qwell[]you'd make a killing
18:52.38Math`I dont know why muffins have a special meaning here
18:53.47Uranellussry if this is OT: If I have a modem how can I make phone calls through that modem ? (on one pc)
18:54.01Math`Uranellus: grab a software that supports it and use it?
18:54.18UranellusMath`: well what to search for?
18:54.41Math`well considering you are looking for a voice modem software I'd look for... voice modem software
18:55.54*** join/#asterisk monstertruck (n=monstert@74.167.124.204)
18:56.09UranellusMath`: well I'll try it .. :)
18:56.25monstertruckhi
18:56.37monstertruckhere's a question
18:56.48monstertrucki have an iax2 channel coming from didww
18:56.55[TK]D-FenderUranellus: Your modem is almost garaunteed WORTHLESS to *.  Time to actually shop for COMPATIBLE hardware.
18:57.03monstertruckand an outgoing iax2 channel to another * server
18:57.23monstertruckthe second server places the call on the pstn
18:57.37monstertruckafter the second server asnwers the iax2 channel
18:57.44Uranellus[TK]D-Fender: well I also have some ISDN cards (I'm from germany)..
18:57.47monstertruckboth channels are hung up on the first server
18:57.55monstertruckbut the call is still alive
18:58.12monstertruckand the second server shows one iax2 channel alive
18:58.27*** join/#asterisk |dennis| (n=dennis@200.32.233.82)
18:58.51[TK]D-FenderUranellus: Well go read up on what it takes to sue them with * on the WIKI
18:58.53[TK]D-Fender~wikiws
18:58.57[TK]D-Fender~wikis
18:58.58jbotwell, wikis is http://www.voip-info.org
18:59.03monstertruckany idea why?
18:59.13Uranellus[TK]D-Fender: I'll see what I find .. tanks
18:59.36*** join/#asterisk james4765 (n=james476@pool-71-127-129-139.rcmdva.east.verizon.net)
19:02.07*** join/#asterisk darkmug (n=dennis@143.106.167.234)
19:03.19monstertruckok, i figured half of it
19:03.25monstertrucknow the question is
19:03.32monstertrucki have this setup
19:04.11monstertruckPSTN (didww) -iax2-> *1 -iax2-> *2 ->PSTN
19:04.43monstertruckwhen *2 receives the call, *1 is taken out of the equation and didww connects directly to *2
19:05.10monstertruckbut i need all calls to go through *1, because there is where the billing system is
19:05.25Strom_Mmonstertruck: notransfer=yes
19:05.30Strom_Mor transfer=no in 1.4+
19:05.43monstertruckStrom_M, in iax2.conf?
19:05.47Strom_Miax.conf
19:05.54monstertruckStrom_M, thanks
19:05.55Strom_Mfor the iax peer/user/friend entries, eys
19:06.51Hmmhesaysyum is sooooo slow
19:07.01Math`yum sucks
19:07.10Hmmhesaysit is what fedora uses though
19:07.21Math`yeah thats the main reason I dont use fedora
19:07.25Hmmhesayswhich makes me want to vomit
19:07.29Hmmhesaysbut thats ok
19:07.33Math`lol
19:07.39Hmmhesaysfor the most part I don't mind fedora for a desktop
19:07.39Math`have a muffin, it helps it go through
19:07.53Math`I was a debian guy and I moved on to ubuntu
19:08.10Hmmhesaysubuntu: linux for people who don't know how to use linux
19:08.14VioBytemandriva*
19:08.26Hmmhesaysmy roomate runs it for the express reason "he doesn't like microsoft"
19:09.08Hmmhesayswhich is just stupid
19:09.33Hmmhesayshe can't really give me any valid reasons he doesn't like microsoft
19:09.35*** join/#asterisk jmacz (n=jmacz@190.24.97.247)
19:09.58Math`lol those guys
19:09.58Hmmhesaysas most people who use linux for that reason
19:10.11Math`I use linux as server because its flexible and I know how it works
19:10.21Hmmhesaysbingo
19:10.26Math`but as desktop I dont care whatever OS I run
19:10.29Hmmhesaysand you can take advantage of that flexibility
19:10.38Math`currently running winxp especially when I have to code for that platform
19:10.49HmmhesaysI use winxp for gaming and it rocks
19:10.54Math`hehe
19:11.16bkrusei use wine. thank you.
19:11.30bkruseno, im just playing, sometimes wine just DOESNT WORK. in that case i dual boot on the gaming machine :P
19:11.32wunderkini have recently found that the mac commercial about vista is highly understated
19:11.39Hmmhesaysand my games run better bkruse
19:11.39Hmmhesaysls
19:11.48bkruseHmmhesays: lol, it will
19:12.02Hmmhesaysok where do I find 4k_STACK option in the kernel config
19:12.16neverbluei can tell you why I dont like M$ and windows
19:12.23neverblueif you need a few reasons...
19:12.28bkruse/lib/modules/`uname -r`/build/Makefile i believe
19:12.58Hmmhesaysthere are reasons to dislike microsoft yes, but to use linux because that is your sole reason...
19:13.06Hmmhesaysand you can't back it up with a valid argument
19:13.19Hmmhesaysthen you are just ignorant
19:14.10neverblueim having an issue receiving calls
19:14.31*** join/#asterisk kFuQ (n=somedude@c-67-185-123-34.hsd1.wa.comcast.net)
19:14.56neverbluei have two "sections" in my sip.conf, one for my phone, the other for the server I am connecting to (all are on my LAN)
19:15.25*** join/#asterisk bbryant (i=brett@nat/digium/x-9e5f12e87bfc3a88)
19:15.43neverblueif i am having imcoming issues, this could be either sections, as they are both listed as type=friend, is that correct?
19:15.47*** join/#asterisk jgw2001 (n=jgw@87-194-118-242.bethere.co.uk)
19:15.51jgw2001hello
19:16.00dacterbkruse... you get warcraft going on wine?
19:17.09bkrusei have
19:17.11bkrusedont play, though
19:17.18bkrusei just did it for fun, im such a nerd ;[
19:17.27wunderkin*snort snort*
19:17.31dacterheh.
19:20.47*** join/#asterisk goldenear (n=goldenea@2001:6f8:392:1:213:2ff:fe4a:53a7)
19:20.47iCEBrkrOH LOOK! IT'S THE NERD HERD!
19:21.51*** part/#asterisk Uranellus (n=alexande@p57A16263.dip.t-dialin.net)
19:21.58goldenearhi I've got a problem with sip.conf and I try to figure out if it's a bug from asterisk or a problem with the dns :
19:22.09*** part/#asterisk mcf3782 (n=mfreeman@209.117.160.3)
19:22.30*** join/#asterisk robin_sz (n=robin@rapid2.gotadsl.co.uk)
19:22.40robin_szmeep?
19:22.57goldenearI have 6 sip accounts (at different voip providers) and 2 of them only work if host=ip and not if host=username
19:23.35goldenearwhat could explain this
19:24.00goldenearwith host=1.2.3.4 it works (sip show peers show the peer/account)
19:24.22*** join/#asterisk BSD_Tech (n=bsdtech@adsl-69-230-166-20.dsl.irvnca.pacbell.net)
19:24.29goldenearwith host=hostname.domain.net it doesn't (sip show peers doesn't show the peer/account)
19:24.44*** join/#asterisk Blackthorn (i=blacktho@72.236.88.10)
19:26.32BlackthornHi. I an * box with a pri and I want to send an incoming pri call to another * box. I have tried to follow some directions and read on the wiki to setup the iax -- iax. but my remote server rejects
19:26.58Blackthornthe call stating the conext is in "banned".  Which is the guest user account with conext "banned"
19:28.59jmaczHi, anybody knows a way to provide CallerID information about incoming calls to agents using AgentLogin?
19:31.09Blackthornso i'm guessing the remote server that i'm sending hte call too dosn't have the sending server registered.. so how would i correct that?
19:31.09jmaczI've tried several of the ones listed here: http://www.voip-info.org/wiki/view/Asterisk+call+notification, but most involve Dialplan and AGI entries and those which use the mgr api don't have agent notifications (in the sense of AgentLogin), but only for common SIP channels (phones)
19:31.09blitzragewas someone looking for me earlier?
19:31.47*** join/#asterisk kavit (n=kavit@ppp167-236-231.static.internode.on.net)
19:32.05[TK]D-Fenderjmacz: No.
19:32.22*** join/#asterisk toot (n=toot@84.19.255.123)
19:33.15*** join/#asterisk sbingner (n=sam@pdpc/supporter/sustaining/sbingner)
19:33.35jmacz[TK]D-Fender, guess I was afraid of that answer.
19:34.02[TK]D-Fenderjmacz: Your phone is ON a call and the only thin you get is a "beep".  the TFB is to be expected :)
19:34.14*** join/#asterisk Avero (n=Avero@216.186.253.120)
19:35.14jmacz[TK]D-Fender, Yep, that's the thing, I tried some other notification methods (POP up windows), msgs using netcat and YAAC, but none of them seems to work with agents events in the mgr :(
19:35.47AveroI'm trying to find a way to get custom CDR fields in MySQL when using cdr_mysql. I found two bugs (0006519 and 0006384) with patches for it. Does anyone know if either of these were patched into the 1.2.18 release?
19:36.32*** part/#asterisk goldenear (n=goldenea@2001:6f8:392:1:213:2ff:fe4a:53a7)
19:39.33*** join/#asterisk rmayorga (i=rmyorg@unaffiliated/rmayorga)
19:40.39*** part/#asterisk ctooley (n=ctooley@209.33.108.198)
19:45.09codefreezeAvero: probably not yet. Doing most of the fixes in 1.4 right now, sort of experimental. Minimal damage.
19:45.26*** join/#asterisk |dennis| (n=dennis@200.32.233.82)
19:46.50anonymouz666for an IVR system... should I use 1.2 or 1.4?
19:46.57anonymouz666what you would use?
19:47.04blitzrageAvero: looks like it was never merged, and closed due to inactivity
19:47.19*** join/#asterisk sysreq (n=sysreq@modemcable171.134-81-70.mc.videotron.ca)
19:47.25anonymouz666blitzrage is crazy :)
19:47.28blitzrageit's true
19:47.30JunK-Yi use 1.4 too.
19:47.45blitzrage1.4.4 is actually a pretty good release
19:47.55anonymouz666I will use 1.4 too, on the first segfault I will sent the BT to you guys :)
19:47.57blitzrage1.4 I'd say is stable enough to start using in production now
19:48.09JunK-Yanonymouz666: great, sounds a deal :)
19:48.20blitzrageanonymouz666: sure, you can send it to me. It won't mean anything to me, but you can send it anyways
19:48.29anonymouz666heh.
19:48.49*** join/#asterisk drazak (i=drazak@65.36.176.140)
19:48.52blitzrageugh... I think I need a nap[
19:48.57pipwerki've been using 1.4 since it has been released, never had a problem
19:49.01blitzrageif I'm not gonna work, I might as well do that
19:49.08blitzragethen I can work tonight
19:49.49*** join/#asterisk joeTSUNAMI (n=joeTSUNA@unaffiliated/joetsunami)
19:51.56Gouroutrashso do i pipwerk
19:52.06Gouroutrash1.4 b3 and now 1.4.4
19:52.16Gouroutrashmmh i've a question
19:52.43Gouroutrashif i upgrade, for example, asterisk 1.4.0 for a 1.4.4
19:53.46Gouroutrashi need to recompile only asterisk
19:53.54Gouroutrashor asterisk + zaptel + libpri ?
19:54.01pipwerkGouroutrash: seems sensible to upgrade
19:54.10Nuggetwell seeing as there's also a newer zaptel, you'll want to upgrade it as well.
19:54.18Gouroutrashyes of course :)
19:54.28Gouroutrashmmh ok
19:54.31Gouroutrashanother situation
19:54.39Gouroutrashi compile a new kernel
19:55.01Gouroutrashi need to recompile asterisk or not ?
19:55.11Nuggetno, but you would need to rebuild zaptel.
19:55.14Gouroutrash(modules..)
19:55.20Gouroutrashmmh ok
19:55.37Gouroutrashjust a clean and make make install
19:55.40Nuggetzaptel is specific to the version of linux that it finds in /usr/src/linux when it is built.
19:55.57Gouroutrashokkk
19:56.15Gouroutrash(sorry for my english, i'm french)
19:56.29Gouroutrash(french with a new dumb president)
19:57.24*** part/#asterisk PseudoNim (n=pseudo@74.57.2.150)
19:57.38*** join/#asterisk saftsack (n=saftsack@pD9E07C30.dip.t-dialin.net)
19:57.40NuggetMDR  :)
19:57.42pipwerkit can't be worse than bush </politics>
19:57.55anonymouz666NY timezone is EST or EDT?
19:58.03NuggetEDT at the moment.
19:58.18Gouroutrashour new president is called "little bush" in france :)
19:58.25Nugget"ET" if you want to be precise, "EST5EDT" if you're setting $TZ
19:58.28Gouroutrashor "bush junior"
19:58.40Nuggetand you'll want to make sure your zoneinfo is current.
19:58.57anonymouz666Nugget: the NTP server answer EDT... but there's a guy that is EST. I live in Brazil...I really don't know.
19:59.05NuggetNTP is timezone agnostic.
19:59.15NuggetNTP is strictly UTC.  Timezone conversions are local.
19:59.22joeTSUNAMIhi guys.  i'm running asterisk 1.2.14 and have a problem with dropped calls when trying to do a "blind transfer".  I'm not sure how to troubleshoot this..
19:59.36anonymouz666Nugget: the system is configured with america/new york timezone.
19:59.42Nuggetthat's irrelevant to NTP.
20:00.12anonymouz666ok, then is it wrong ?
20:00.19NuggetIs what wrong?
20:00.25anonymouz666Mon May  7 16:00:10 EDT 2007
20:00.33Nuggetthat is correct.
20:00.35anonymouz666NY time
20:00.42*** join/#asterisk trevarthan (n=trevarth@c-71-59-54-137.hsd1.ga.comcast.net)
20:01.22anonymouz666why the guys said that NY tz is GMT-05:00 Eastern Standard Time ?
20:01.26trevarthanHello. I've got two T1s talking to each other. One is from an Intertel machine, the other is Asterisk Zaptel. How do I access DNIS info in Asterisk?
20:01.29Nuggetbecause the guys are idiots.
20:02.05NuggetEST is UTC-5.  EDT is UTC-4.  New York is "EST5EDT" which means that in the winter it's UTC-5 and in the summer it's UTC-4.  It's the summer now.
20:02.05trevarthanI tried just using extensions in the zaptel context, but nothing happens. It only gets the 's' extension.
20:02.09[hC]This is weird, all of a sudden i have to execute "Answer()" on outgoing calls from SCCP phones otherwise i get no receiving audio ...
20:02.59anonymouz666Nugget: thanks for the explanation.
20:03.03Nuggetany time.
20:03.09*** join/#asterisk CrazyTux[m] (n=CrazyTux@cpe-68-201-202-72.houston.res.rr.com)
20:03.38*** join/#asterisk RaYmAn-Bx (i=rayman@skumler.dk)
20:06.59*** join/#asterisk coolbeans (n=null@adsl-074-247-038-249.sip.bhm.bellsouth.net)
20:07.21coolbeansHey guys, for my extensions.conf Dial command, how should I format international calling??
20:07.38joeTSUNAMIanyone aware of any blind transfer issues with Polycom 550 and asterisk?
20:07.53pipwerkcoolbeans: depends on your provider
20:08.25*** part/#asterisk sevard (i=chuck-th@adsl-71-129-115-242.dsl.irvnca.pacbell.net)
20:10.28coolbeanspipwerk: Thanks.
20:11.53coolbeansWhat does the _011. mean in exten => _011.,1,Dial(SIP/${EXTEN}@vitel-outbound)?  More specifically, what does the "period" mean?  Anything after 011?
20:12.22Mercestes<PROTECTED>
20:12.29pipwerkperiod is a wildcard, yes
20:12.37coolbeansAhh! Cool.  Learn somethign new every day! Thanks, guys (gals?).
20:12.53pipwerknaive :P
20:13.00Mercestesindeed.
20:13.16MercestesIRC:  Where men are men, women are men, and  14 year old virgins are undercover FBI agents.
20:13.56[TK]D-Fender^- Knave ;)
20:13.57Mercestes>.>
20:13.57pipwerk^5
20:13.57FuriousGeorgeGIRL=Guy In Real Life ;)
20:13.57Capps-ha
20:13.57[TK]D-Fender...
20:14.13*** join/#asterisk sevard (i=chuck-th@adsl-71-129-115-242.dsl.irvnca.pacbell.net)
20:14.13FuriousGeorgeseriously though, when people claim to be women here in #asterisk.  i believe'em.  why not?
20:14.27FuriousGeorge~lastseen katty
20:14.31FuriousGeorge~lastspoke katty
20:14.37Mercestesone of my internet girlfriends turned out to be a dude.
20:14.41FuriousGeorge~seen katty
20:14.44jbotkatty <n=Katty@hera.copi-rite.com> was last seen on IRC in channel #asterisk, 6d 5h 58m 15s ago, saying: '[TK]D-Fender: cause i think i could make a cronjob to copy them elsewhere, and then delete all of them in a directory.'.
20:15.06Mercestes.....so he admitted to being a guy....and *then* broke up with me to boot.  ...guys suck.
20:15.10FuriousGeorgeMercestes: what tipped you off :)
20:15.18FuriousGeorgelol
20:15.20pipwerkFuriousGeorge: true, this is not #teensex or someting like that ;)
20:15.22MercestesFuriousGeorge, When he said "I'm a guy"
20:15.47FuriousGeorge"are you a guy, or just happy to see me?
20:15.48anonymouz666menuselect is very nice. but I am too lazy to choose one by one the apps, channels, etc.
20:16.39FuriousGeorgeanonymouz666: i think they have group meetings for things like that
20:16.40*** part/#asterisk RaYmAn-Bx (i=rayman@skumler.dk)
20:17.34anonymouz666let's build everything. Someone may cry if I forget to compile.
20:19.04coolbeansAnyone have a working non-US international phone number I can test with?
20:19.37n00dlecool, I have a UK number, if you like.
20:19.44coolbeansn00dle: Please! :)
20:20.14anonymouz666call to me
20:20.15anonymouz666brazil
20:20.25anonymouz666but you will need to speak portuguese
20:21.08*** part/#asterisk trevarthan (n=trevarth@c-71-59-54-137.hsd1.ga.comcast.net)
20:26.12*** join/#asterisk Nivex (n=kjotte@user-0ce2kma.cable.mindspring.com)
20:26.56*** join/#asterisk jm|home (n=jm|home@zen.jamiem.com)
20:28.02*** join/#asterisk santiago (n=santiago@debian/developer/santiago)
20:28.43anonymouz666make[3]: autoconf: Command not found
20:28.44anonymouz666make[3]: *** [config.h] Error 127
20:28.48anonymouz666yum install autoconf?
20:31.24anonymouz666autoconf: no input file
20:31.24anonymouz666make[3]: *** [config.h] Error 1
20:31.31anonymouz666after yum install autoconf
20:31.45anonymouz666what's wrong with ast-addon-package?
20:34.04Hmmhesaysanyone next to a fax machine?
20:35.02coolbeansIf I'm using a wildcard, i.e., 011.,n,whatever, does Set(TIMEOUT(digit)=3) not apply?
20:35.12coolbeansIn 1.2?
20:35.26*** join/#asterisk trevarthan (n=trevarth@c-71-59-54-137.hsd1.ga.comcast.net)
20:36.30trevarthanhello. I'm using an old system that doesn't support the CALLERID application. It's business edition, and I think it might be based on 1.0.x or something. Is there some way to extract all callerid fields without CALLERID(all)?
20:37.39*** join/#asterisk jtexter3 (n=jtexter3@COX-66-210-197-34-static.coxinet.net)
20:37.55jtexter3Anyone here have any luck with a phone behind dd-wrt talking to an Asterisk server over a VPN?
20:38.48NuggetIn my experience, VPNs are pretty damaging to latency and can really ruin the VoIP experience.
20:39.12NuggetI'm sure it's possible to tweak things to mimimize the impact, but it's not straightforward.
20:39.20pipwerktrevarthan: I guess the $CALLERID var will work then
20:39.48NuggetMinimally you'll want to make sure that the UDP voip traffic isn't being tunneled inside a TCP connection
20:39.49pipwerk$CALLERID_NUM nad $CALLERID_NAME it was I believe
20:40.31*** join/#asterisk Zipper_32 (n=None@142.232.142.96)
20:40.33*** join/#asterisk jgw2001 (n=jgw@87-194-118-242.bethere.co.uk)
20:40.33MercestesNugget, Funny, I've found in some cases that VPN enhances VoIp
20:40.52Nuggetinteresting.  any theories about why or how that would work?
20:40.54Mercestesmostly because VPN does use QoS properly
20:41.18Nuggetah, I guess that makes sense
20:41.26MrWuphey
20:41.36*** join/#asterisk jarrod (i=nobody@dont.juniperyour.net)
20:41.51MrWuphow do i go to a certain context if a call is answered successfully after a dial command?
20:41.51Mercestesit also establishes a constant tunnel that makes it hard to nat timeout.  But it only helps in broadband connections and most of my experience has been overseas with VPNs
20:41.51jarrodany ip pbx systems compatible with polycoms that have a presence utility?
20:42.11coolbeansSince international numbers are never the same length, how do you guys handle digit timeouts in 1.2?
20:42.14*** join/#asterisk zotz (n=zotz@24.244.163.157)
20:42.21james4765jtexter3: yep
20:42.56james4765not very good - the dd-wrt cpu adds a lot of lag
20:43.11Zipper_32I have a question regarding T1's/PRI's; I am working for a company that is opening a new location, and they are going to need 8 DID's as voice lines, and 2 separate DID's for Fax and POS terminals. How would I use Asterisk to connect to that PRI and only use 8 channels for Asterisk, but give separate 'lines' for the fax and POS terminals?
20:43.30MercestesIt also helps by elminating NAT.
20:44.07jtexter3james4765: how did you get around the problem of dd-wrt rewriting the IP's in the SIP packets?
20:44.26coolbeansDoes digit timeout not work with extension wildcards in 1.2?  (i.e., _011.)
20:44.54james4765I used the -vpn firmware
20:45.01james4765and used openvpn to tunnel it
20:46.11james4765I've also set up single phones to just map to port 5060 for our phones at home
20:47.46*** part/#asterisk jgw2001 (n=jgw@87-194-118-242.bethere.co.uk)
20:48.43Mercestescoolbeans, the syntax changed from digittimeout to timeout(digits) but it should work.  It would not make any sense for it not to work.
20:48.53Mercestesbeing..that's what digit timeout is for.
20:49.39coolbeansGot it.  My problem is at the phone level and I'm an idiot for not seeing it sooner.
20:49.48tootdd-wrt? :)
20:50.03jtexter3Nugget: If I open my Asterisk box to the outside world, do I need to do anything other than give my user(s) decent passwords?
20:50.07errrwhen I dial a number Im getting a fast busy signal so I know I have something wrong but from the cli when I dial the number nothing shows up at all so I can see where things are going wrong... I have set verbose 1000 but still nothing shows up.. any hints?
20:50.41Nuggetjtexter3: you would also want to (minimally) be very careful about the state of your default context and what access anonymous SIP and IAX users have.
20:51.17Nuggetspecifically, allowguest=no for example, in sip.conf
20:55.04*** join/#asterisk nhudson (n=nhudson@68.113.120.148)
20:57.45*** join/#asterisk Fieldy (i=Kd8tiVCu@gentoo/contributor/Fieldy)
21:00.26Hmmhesayswell that kernel compile when suprisingly smoooth
21:00.31jarrodany ip pbx systems compatible with polycoms that have a presence utility?
21:00.46*** join/#asterisk DocHolliday (i=RgRabbit@gateway/gpg-tor/key-0x0E4F6D6C)
21:00.49trevarthanpipwerk: ${CALLERID} contains nothing ("").
21:00.50Qwell[]jarrod: like...say...asterisk?
21:01.06trevarthanDoes asterisk support ANI?
21:01.10Qwell[]ys
21:01.11Qwell[]yes
21:01.18jarrodwhat presence is available?
21:01.20trevarthanHow about DNIS?
21:01.25Qwell[]yes
21:01.26Hmmhesaysum
21:01.44Qwell[]jarrod: as far as?
21:01.46trevarthanWhere would one get DNIS from business edition asterisk?
21:01.47Hmmhesaysoh ser is just being fan-fscking-tastic today
21:02.10Qwell[]trevarthan: same place as open source asterisk, I imagine
21:02.15jarrodqwell: an interface for an enterprise to view to see if an individual is on the phone?
21:02.16*** join/#asterisk bbryant (i=brett@nat/digium/x-5cad09fa8649bff7)
21:02.27DocHollidaydocelmo?
21:02.28Qwell[]jarrod: that isn't an ip pbx...
21:02.33trevarthanQwell[]: where would that be?
21:02.50jarrodqwell: thats why i asked for an ip pbx that SUPPORTED presence
21:02.52Qwell[]jarrod: BUT, you could use jabber to do that
21:02.57Qwell[]asterisk *does* support presence
21:03.00jarrodand if so, what presence is available
21:03.01*** join/#asterisk zirman (i=zirman@ip194.207.107.216.seg.net)
21:03.06Qwell[]what you are asking for is not "presence"
21:03.07jarrodhence, why i asked what packages it supported
21:03.14trevarthanQwell[]: ${CALLERID(all) yields an application doesn't exist error....
21:03.20jarrodyes it is, i need to see who is on the phone
21:03.30Qwell[]jarrod: and asterisk tracks that
21:03.36Qwell[]what you want, is something that can view that information
21:03.43*** part/#asterisk lpaz (n=paz@c83-251-203-30.bredband.comhem.se)
21:03.48jarrodthats why i said presence utility
21:03.51Qwell[]which isn't an ip pbx
21:03.58jarrod<jarrod> any ip pbx systems compatible with polycoms that have a presence
21:03.59jarrod<PROTECTED>
21:04.02Qwell[]yes, asterisk
21:04.05jarrodthat have a presence utility
21:04.08jarrodwhat is the utility
21:04.14jarrodnot the support, the viewer
21:04.14Qwell[]make one
21:04.22Qwell[]it's trivial
21:04.23jarrodim asking for one that is already in existence
21:04.43*** join/#asterisk spatulamaan (n=ggilmore@207.188.8.252)
21:04.54russellbeh?
21:04.57russellbpresence viewer?
21:05.16Qwell[]that can be anything you want...  do you want a windows app?  web app?  what?
21:05.35jarrodqwell: would prefer a windows app, but i am open to anything available
21:05.39Qwell[]jabber
21:08.38trevarthanDoes anyone know how I can pull DNIS info?
21:09.23Math`CDR(dnis)?
21:10.33*** join/#asterisk Dimik_ (n=Dimik_@unaffiliated/dimik)
21:11.14neverblueok, I can do something (sip/number@server) in my context, but when I try and implement it in php, using Action: Originate, I add that into Exten?
21:11.18neverblueis that correct
21:11.39neverbluedial(sip/number@server)*
21:13.05*** join/#asterisk jgw2001 (n=jgw@87-194-118-242.bethere.co.uk)
21:13.45trevarthanMath: hmmm... I don't see dnis listed. I'll have to try that tomorrow.
21:15.17jgw2001Hekllo, im using asterisk with the webbased destar webbased utility,  ive set up asterisk on a vps server however have issues getting some of the components working such as conference calls,  when a conference call is made it comes up in the logs asUnable to open '/dev/zap/pseudo': Permission denied,     any ideas......
21:15.49Qwell[]jgw2001: you need ztdummy, but it isn't going to work very well on a vps
21:16.38SomeOne1Qwell: i love you
21:16.48Qwell[]SomeOne1: send gifts
21:16.53SomeOne1will do
21:17.02Qwell[]Why do you love me?
21:17.07jgw2001I fail to under stand why its trying to access "zap" when I dont have any zap services installed....
21:17.09SomeOne1just something about you
21:17.09Qwell[]you aren't a stalker, I hope
21:17.18SomeOne1i am, oh, im very much a stalker!
21:17.33Qwell[]well, that's okay, I suppose
21:17.38SomeOne1heh
21:17.38SomeOne1jk
21:17.54SomeOne1(jk = just kidding)
21:19.09SomeOne1gotta go
21:20.35Corydon-wQwell[]: he loves you because you spoon
21:20.41demlakhow to show all registered clients in CLI?
21:20.47Qwell[]Corydon-w: only you <3
21:20.48Qwell[]:P
21:21.07demlakall clients/phones that are currently "online"
21:21.29*** join/#asterisk Mavvie (n=edwin@ppp39-111.lns3.syd7.internode.on.net)
21:21.59*** join/#asterisk Gouroutrash (n=x@ACaen-151-1-16-5.w86-215.abo.wanadoo.fr)
21:22.03Gouroutrashre
21:26.21*** join/#asterisk dc3aes (n=matt@S01060001023fe8ca.no.shawcable.net)
21:28.34*** join/#asterisk DigitalKNK (n=DigitalK@adsl-69-232-146-57.dsl.irvnca.pacbell.net)
21:29.09*** join/#asterisk cspot (i=cspot@ip68-1-63-100.pn.at.cox.net)
21:30.48*** join/#asterisk DeeJayTwo (n=deejay2@office.abi.ca)
21:31.04DeeJayTwoIn a tcpdump, I can't find the password in the register method..
21:31.13DeeJayTwowhere should I see the password for registration?
21:33.55DigitalKNKanyone here using CBeyond SIPConnect?
21:34.00DigitalKNKwith TrixBox :)
21:35.49*** part/#asterisk jtexter3 (n=jtexter3@COX-66-210-197-34-static.coxinet.net)
21:36.25jarroddw
21:37.16*** join/#asterisk hads (n=hads@reef80.anchor.net.au)
21:37.25Hmmhesayssweet I got my linksys wireless card working
21:37.36Math`what chipset
21:37.53Hmmhesaysbroadcom
21:38.17Math`I ordered a Super Range Cardbus
21:38.23Math`300mW atheros chipset
21:38.27Hmmhesaysoops this one is texas instruments
21:38.33Hmmhesaysdoesn't matter though, ndiswrapper rocks
21:38.34Math`with a 7.9dB gain antenna
21:38.37Math`ah you wrapped it
21:38.50Math`acx100?
21:39.04Hmmhesaysyeah
21:39.10Hmmhesayscan you recommend me a good site survey tool?
21:39.14Math`I used the open source driver back in the days
21:39.35Hmmhesaysmost atheros chipsets you don't need the wrap it
21:39.40Hmmhesaysbut this particular linksys driver was free
21:39.43Math`oh
21:39.44Hmmhesays*card was free
21:39.50Math`atheros I'd use madwifi
21:40.03Math`I didnt buy an external antenna and a high power card for no reason :P
21:40.15Hmmhesaysis there any good gui tools like windows has to do site surveys?
21:40.43Math`uhm I used airsnort before
21:41.02Math`there is some gui config tools but I dont know them/use them
21:41.07LeddyHMdig: we use cbeyond, not with trixbox though
21:41.27Hmmhesaysthats one thing I like about windows networking, the wlan config stuff is soooo nice compared to linux
21:42.48mvanbaakHmmhesays: what DE are you using ?
21:43.02mvanbaakgnome? kde?
21:43.55neverbluehow do I record a sound, using my local phone?
21:44.06neverblueso I can play it in my context
21:44.16mvanbaakneverblue: look at the Record() dialplan application
21:44.27Hmmhesaysgnome
21:44.27mvanbaakshow application Record
21:44.39Hmmhesaysiwlist is ok but I would like a pretty graph
21:44.42mvanbaakHmmhesays: I thought there was some tool in gnome to setup stuff
21:45.19mvanbaakthe network manager (or however that is called)
21:45.41*** join/#asterisk dotSlashW (n=HTP@200.80.197.5)
21:45.43mvanbaakiirc it does wireless stuff as well
21:45.44neverblueexten=>456,1,record()
21:45.52neverblueso just that, then 456 on my phone
21:46.00neverbluethen how do i stop the recorrding?
21:46.05mvanbaakneverblue: record needs a file and a format
21:46.09neverblueah#
21:46.24mvanbaakexten => 456,1,Record(/path/to/my/file:gsm)
21:46.26neverblueis gsm more clear than wav?
21:46.28dotSlashWhi ,  I need to connect an * server to a shoreline pbx, any tip or place where I could find some info on that ?
21:46.45mvanbaakneverblue: no, wav is more clear
21:46.48Corydon-wmvanbaak: that should be '.' not ':'
21:46.49neverbluecan I use . as the path?
21:46.56Math`neverblue: always specify a full path
21:47.01mvanbaakeh ?
21:47.12Corydon-wmvanbaak: we stopped using ':' in 1.0
21:47.37Corydon-w1.2 and later use '.' to separate the filename from the format type
21:47.37neverblueexten=>456,1,record(/home/user/wow.gsm)
21:47.40neverbluelike that you mean
21:47.44Corydon-wCorrect
21:47.45mvanbaakCorydon-w: ah
21:47.48mvanbaakCorydon-w: Connected to Asterisk 1.0.9-BRIstuffed-0.2.0-RC8o
21:48.04BSD_Tech1.0.9 wow
21:48.07BSD_Techthats old
21:48.10Corydon-wmvanbaak: eek
21:48.12mvanbaakuhhuh
21:48.14mvanbaakbut it works
21:48.42neverbluehmm that didnt work
21:48.44neverblue:/
21:48.45neverbluelol
21:49.02mvanbaakgoing to upgrade to latest bristuff next week
21:49.10neverbluelong tone (1 sec), then I had a busy signal
21:49.25mvanbaakneverblue: check permissions
21:49.34neverblueits my home dir
21:49.37neverblue:P
21:49.52mvanbaakmake sure the directory you are writing to is there and that asterisk can write in it
21:50.19mvanbaakfor quick testing use /tmp/
21:50.27neverbluegood point
21:51.12mvanbaakok
21:51.17mvanbaakguys, this better?
21:51.18mvanbaakConnected to Asterisk SVN-trunk-r63182M
21:51.43Corydon-wmvanbaak: your choice, but we don't recommend running trunk
21:51.49mvanbaakgheh
21:51.54mvanbaakloads of stuff is broken now
21:51.55mvanbaak;)
21:52.03MrWupguys im having a weird problem
21:52.14mvanbaakmv /etc/asterisk/extensions.conf /etc/asterisk/extensions.conf.old
21:52.22mvanbaakvim /etc/asterisk/extensions.conf
21:52.25mvanbaakstart over ;)
21:52.40MrWupwhen i do a Dial(SIP/102|20|Tt|M(CallSuccess)) the macro macro-CallSuccess isnt executed
21:52.47mvanbaakCorydon-w: I wouldn't even run 1.4 in production yet
21:52.49MrWupasterisk doesnt even try to execute it and fail
21:52.55MrWup(when the call is answered)
21:53.14mvanbaakMrWup: remove the | between Tt and M
21:54.04neverbluehmm, sound isnt playing
21:54.06mvanbaakthe M is just an option for the Dial application, as are t and T
21:54.12MrWupstill wont execute
21:54.14MrWupsame thing
21:54.23neverbluemaybe missing codecs?
21:54.29MrWupoh
21:54.30MrWupno
21:54.32MrWupit works
21:54.37MrWupforgot module reload after altering dialplan
21:54.38MrWupheeh
21:54.40neverblueoh permissions :/
21:54.53neverbluewait
21:54.56neverblueno its not
21:55.02errrI have an aastra 55i when I dial an extension is there a way to make it say the persons name instead of Unknown Name?
21:55.04mvanbaakneverblue: is the file there ?
21:55.09neverbluelisting to a .wav is an x or r ?
21:55.27neverblueyeah, created both files, a .wav and .gsm
21:55.43mvanbaakneverblue: and how do you 'listen' to it ?
21:55.53neverblueexten=>456,1,Playback(/var/lib/asterisk/sounds/intro.gsm)
21:56.09neverbluei changed it to 1,record( to record
21:56.13mvanbaakneverblue: you should Answer() the call first
21:56.14Corydon-wDrop the .gsm on Playback
21:56.16neverbluethen just changed the one word to playback
21:56.19neverblueah
21:56.37neverblueand if I have both, it will play the .wav?
21:56.42neverblueor the .gsm
21:56.43mvanbaakor is that changed as well
21:56.50Qwell[]it'll play the best match for that call
21:56.55Corydon-wWhen you Record, you need to select the format to record for.  When playing back, Asterisk automatically selects the least-cost format.
21:57.10neverblueyes!
21:57.12neverblueplays
21:57.19neverbluecan I remove the # tone at the end?
21:57.21Corydon-wLeast-cost in terms of CPU, for converting.
21:57.31mvanbaakneverblue: with the Answer() ?
21:57.49mvanbaakI'm a bit lost of what is required these days
21:57.59Corydon-wneverblue: are you using a SIP channel or a Zap channel to record?
21:58.02neverbluewhen I record(), i need to press the # to stop the recording, correct?
21:58.05neverbluesip
21:58.15mvanbaakI always start my IVR with an Answer() and a Wait(1)
21:58.18Corydon-wneverblue: change it to out of band DTMF (like SIP INFO)
21:58.27neverblue:/
21:58.33neverbluei have no idea what that means lol
21:58.35Corydon-wThe tone should never show up in the recording, then
21:58.44Corydon-wCheck your SIP config
21:58.51Corydon-wIt's probably set to inband DTMF
21:59.21Corydon-wand in sip.conf, dtmfmode=info
21:59.50neverblueall dtmf lines are commented it out
21:59.54neverblueso its set to the default
22:00.00Mavviehow can I see if a linux is running in 32 or 64 bit mode?
22:00.26neverblueCorydon-w, so do I need to set that =info into each of my "sections"?
22:00.30mvanbaakMavvie: what's the output of uname -a
22:00.36Corydon-wneverblue: correct
22:00.40neverbluethanks
22:01.00Mavviemvanbaak: /msg to prevent spamming here.
22:01.08mvanbaakMavvie: that is 32bit modus
22:01.20mvanbaaki686 i686 i386
22:01.21Mavviemvanbaak: thanks!
22:01.23neverblueCorydon-w, and that is to assist with recording?
22:01.44Mavviemvanbaak: aha, now that's an interesting thing to remember.
22:02.00*** join/#asterisk bkruse (i=bkruse@nat/digium/x-57e23c591792b21b)
22:02.03*** join/#asterisk zm23 (n=chatzill@zaara.cuit.columbia.edu)
22:02.07bkruseDeeewayne: is that you!!!?
22:02.12bkrusethe love is automatic, zoom zoom zoom
22:02.34mvanbaakMavvie: 64 bit kernel will report: x86_64
22:02.37file:D
22:02.39Corydon-wneverblue: it's actually better signalling
22:02.41fileI sent it to him
22:02.42Deeewayneyes it is I
22:02.47bkrusefile: lies!
22:02.53filecould you be my supernova girl?
22:02.56bkrusefile++
22:02.59mvanbaakMavvie: check my uname -a output sent in private
22:03.03bkruseyou get mad cool points for that
22:03.06filemake my heart go
22:03.09fileboom boom
22:03.11filemy supernova girl
22:03.26Mavviemvanbaak: yeah, realized it was the same as on FreeBSD.
22:03.30Deeewaynehere's the jam
22:03.31Mavviemvanbaak: just didn't know how to get the info :-)
22:03.41neverblueCorydon-w, so I need to re-record my message again
22:03.48neverbluethats what I was trying to get at
22:04.11zm23hello all.  I am having trouble with call transfers using openser as the sip proxy forwarding calls to asterisk
22:04.12mvanbaakneverblue: you should get a sexy sounding woman to record your prompts ;)
22:04.13Corydon-wneverblue: correct
22:04.30neverbluemvanbaak, your g/f available :)
22:04.48mvanbaakneverblue: sure. if you can afford that ;)
22:04.53Corydon-wneverblue: or you could import them into an audio editing program, just be sure to save back out as 8000Hz, single channel
22:05.00DeeewayneWOPR: do you want to play a game?
22:05.07zm23i'm trying to use the call park feature.. openser directs the call to asterisk and i hear back the lot number 701.  but when i complete the transfer, caller hears 702
22:05.13neverblueCorydon-w, im not that picky
22:05.14neverbluelol
22:05.28Corydon-wneverblue: that's the rate required for Asterisk format files
22:05.42mvanbaakneverblue: actually my gf recorded all our customers sounds
22:05.48neverblueyes, but im not the pciky about my own voice
22:06.00neverbluemvanbaak, pron hotline?
22:06.02neverbluelmao
22:06.05Corydon-wOr you could record prompts with Cepstral
22:06.15mvanbaakit's fun to call one of our customers and hear my gf babble some $random_greeting
22:06.18mvanbaaklol neverblue
22:06.24zm23help anyone!
22:06.33Corydon-w~ask
22:06.42jbotQuestions in the channel should be specific, informative, complete, concise, and on-topic.  Don't ask if you can ask a question first.  Don't ask if a person is there, just ask what you intended to ask them.  Better questions more frequently yield better answers.  We are all here voluntarily or against our will.
22:06.42mvanbaakor use the weasel sound files ;)
22:06.50*** join/#asterisk MrChicken (n=Dorphals@200.71.58.39)
22:06.56MrChickenHello.
22:07.09zm23i'm trying to use the call park feature.. openser directs the call to asterisk and i hear back the lot number 701. but when i complete the transfer, caller hears 702
22:07.12MrChickenI just installed g729 on my server, however I have some echo problems
22:07.23MrChickenI can hear myself speaking on the phone
22:07.26mvanbaakzm23: any console logs ?
22:07.48*** join/#asterisk danalien (n=danalien@unaffiliated/danalien)
22:07.50Gouroutrashonly when you use g729 ?
22:07.56*** join/#asterisk sysreq (n=sysreq@modemcable171.134-81-70.mc.videotron.ca)
22:08.10MrChickenyes
22:08.16MrChickenUlaw seems to work fine
22:09.01Gouroutrashmmh
22:09.10neverblueyes, still getting the # press at the end
22:09.23Gouroutrashthe two phones use g729 ?
22:09.30neverblueso I should just record, then hangup I guess (physically hang up the phone I mean)
22:09.41mvanbaakneverblue: that's how we do it
22:09.56Gouroutrashis it just phone 1 <=> asterisk <=> phone 2 ?
22:09.58MrChickenuhhh no
22:10.02zm23mvanbaak nothing special on teh console log jast says added extensions 701 priority 1 to parkedcalls.  I think it may have to do with teh actual "call tarnsfer" method
22:10.06MrChickenactually one uses ulaw and the other gsm
22:10.24Gouroutrashtry with g729 on both
22:10.30MrChickenhrmmm lemme seeeee
22:10.35Gouroutrashand core show translation, nothing special ?
22:11.19Gouroutrashg729 free or "official" ?
22:11.27Gouroutrash(i use free version and no problem)
22:11.28MrChickenfree
22:11.37MrChicken:D
22:11.47Gouroutrashyou choose the right version ?
22:11.55Gouroutrash(sse, no-sse, sse3 ...)
22:11.56MrChickenI think so...
22:12.16Gouroutrashi think there is a troobleshooting section on the website
22:12.24Gouroutrashproblems with g729 ...
22:12.37Gouroutrashversions gcc4 and icc
22:13.04zm23mvanbaak my phone is using REFER method  but it seems like when i complete teh transfer, asterisk does connect to the other phone but extension priority starts over, and phone 1 is not disconnected
22:13.15*** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2)
22:13.20zm23mvanbaak i am using polycom phones
22:13.50MrChickenI did not know to which sse to choose
22:13.58MrChickenso I choose the one that said nothing about sse
22:14.00MrChicken:P
22:14.14Gouroutrashcat /proc/cpu
22:14.26Gouroutrashyou can see the supported extensions
22:15.05Gouroutrashtry other version :)
22:15.05*** join/#asterisk jazzanova (n=boris@S010600146cfc7d5b.vc.shawcable.net)
22:15.06jazzanovahi
22:15.17Gouroutrashno echo problem reported in troobleshooting section
22:15.24mvanbaakzm23: I have no experience wiht polycom
22:15.27jazzanovai need to call people in Vancouver, BC, Canada. What's a good VOIP sip provider to use with Asterisk ?
22:15.28mvanbaakneither with openser
22:15.39neverbluewhere is the default /sounds dir for record?
22:15.51mvanbaakso I'm afraid I wont be of any help there zm23
22:16.03mvanbaakneverblue: /var/lib/asterisk/sounds/
22:16.28mvanbaakon a default source install that is
22:16.50mvanbaakyou can change that in /etc/asterisk/asterisk.conf
22:17.05jazzanovawhats a good SIP provider for asterisk, period ?
22:17.17mvanbaakjazzanova: voop
22:17.29zm23mvanbaak in general:  is there anythign that i need to configure in asterisk related to call transfers.
22:17.30mvanbaakthey do iax as well
22:17.45*** part/#asterisk jgw2001 (n=jgw@87-194-118-242.bethere.co.uk)
22:17.51neverblueso playback(intro) should access  /var/lib/asterisk/sounds/intro.gsm
22:17.56*** join/#asterisk kFuQ (n=somedude@c-67-185-123-34.hsd1.wa.comcast.net)
22:17.59mvanbaakzm23: not that I know. you have to use the t and/or T in your Dial command
22:18.07neverbluewonder why its not
22:18.19neverbluesound dir is setup in asterisj.conf?
22:18.20mvanbaakneverblue: should be
22:18.24neverblueasterisk*
22:18.24mvanbaakyeah
22:18.33*** join/#asterisk |dennis| (n=dennis@200.32.233.82)
22:18.57jazzanovamvanbaak: tanks
22:19.27neverblue"/var/lib/asterisk/sounds/" not found
22:19.33neverbluewasnt found in asterisk.conf
22:19.36neverblueoh well
22:20.05mvanbaakneverblue: astvarlibdir => /var/lib/asterisk
22:20.27jazzanovamvanbaak: what can you recommendfor canada ?
22:20.34mvanbaakit will use $astvarlibdir/sounds
22:20.38*** part/#asterisk FuriousGeorge (n=brian@ool-43536ea8.dyn.optonline.net)
22:20.55mvanbaakjazzanova: no idea. I'm in .nl and my ITSP has nice prices for canada
22:21.01mvanbaakso I never looked into it
22:21.17mvanbaakzm23: ur welcome
22:21.27mvanbaakzm23: I hope you find what the problem is
22:22.42MrChickenCan anybody give me a hand upgrading the GXP2000 firmware?
22:22.44MrChickenThnx
22:23.04Qwell[]MrChicken: Do you have a goat handy?
22:23.19[TK]D-Fenderzm23, You do NOT need to use "tT" in your dial statement, you should be able to use the native SIP transfer on your phone jsut fine
22:24.18mvanbaakif your phone supports it yeah
22:24.43mvanbaakbe sure to enable call-waiting on the phone when you want to use native SIP transfers
22:25.07zm23[TK]D-Fender i am not using dial command at all.  dialing 700 from teh phone, openser directs that call to asterisk so asterisk itself gets a call for 700
22:25.22*** join/#asterisk nomadsoul (n=nomadsou@unaffiliated/nomadsoul)
22:25.36zm23this results in call being parked and message is played back that parked at 701 etc
22:26.10zm23completing call transfer should hangup phone1 that dialed 700 and connect the caller (another voip phone) to asterisk.
22:26.35nomadsoulhi
22:26.43zm23what happens is that phone 2 is connected to asterisk as if it connected for the first time and get 702 played back.  meanwhile phone 1 does not disconenct
22:27.01zm23both phones are polycom and they do support transfer and have call waiting enalbled as well.
22:27.27zm23phones use REFER method
22:27.49shido6reefer
22:27.59mvanbaakbut you hove openser in between right ?
22:28.06zm23yes
22:28.38zm23so there is no dial command in extensions.conf, just include parkedcalls
22:29.28mvanbaakI never used openser
22:29.30mvanbaaksorry
22:29.40*** join/#asterisk Mavvie (n=edwin@ppp39-111.lns3.syd7.internode.on.net)
22:29.43Mavviejoin #postfix
22:29.48mvanbaakMavvie: why ?
22:30.07Mavviesorry. the / was in another window.
22:30.16[TK]D-Fenderzm23, So you do [transfer] , 700 , (hear the parking lot position) , [transfer], and then the call does NOT get released?
22:30.51zm23[TK]D-Fender that is correct.  and it is not limited to this scenario.
22:31.05*** join/#asterisk Here_And_There (n=Here_And@pool-71-244-103-43.phlapa.fios.verizon.net)
22:31.09[TK]D-Fenderzm23, hmm
22:31.51*** join/#asterisk marcan (i=1337@198.Red-83-54-248.dynamicIP.rima-tde.net)
22:32.11zm23[TK]D-Fender i maybe missing somethign to make the actual transfer happen
22:32.30*** join/#asterisk `pariah (n=josh@unaffiliated/pariah)
22:32.51zm23in another scenario, i get a call, hit transfer, dial asterisk, asterisk is processiong the call.. goign through the extensions.conf, I hit transfer again
22:33.11*** join/#asterisk [hC] (n=hardcore@66.119.167.162)
22:33.31zm23and call is not disconnected, however caller1 also gets connected to asterisk and hears prompts from priority 1
22:33.49zm23not from the priority wehre i tried to complete the transfer
22:34.54[TK]D-Fenderzm23, ok that last part made no sense.  You'd have ot show some CLI output and your configs
22:35.03MrWupis there any way at all to keep variables in a channel which has gone zombie?
22:35.14MrWupwhen you do an attended transfer with asterisk, e.g. SIP1 calls SIP2, SIP2 presses * and transfers SIP1 to SIP3
22:35.26MrWupSIP2 then drops out and goes zombie
22:35.32MrWupand you lose all the channel variables
22:35.43MrWupwhich is a pain if you need to do anything with the phone which dropped out
22:35.53MrWupcause in the h extension you have no variables to play with
22:37.41*** join/#asterisk disgrntld57 (n=sdf@CPE-65-30-153-8.wi.res.rr.com)
22:38.20disgrntld57so what is the best soft free (open source would be better) SIP softphone?
22:39.09mvanbaakdisgrntld57: x-lite or sjphone
22:39.43mvanbaakI use x-lite on linux and osx
22:40.02disgrntld57cool
22:40.03disgrntld57thanks
22:40.39[TK]D-Fenderidefisk > xlite
22:40.48[TK]D-Fenderit has native transfer feature
22:40.56mvanbaakyeah, but it's IAX
22:40.58mvanbaaknot sip
22:41.01[TK]D-Fendersmaller AND support IAX2 if you so desire
22:41.08[TK]D-Fendermvanbaak, BOTH <-
22:41.17[TK]D-Fendermvanbaak, wake up to TODAY :)
22:41.36mvanbaakhello world
22:42.14mvanbaakremind me to log off from freenode when the first bottle of wine is empty ok ?
22:42.17Math`[TK]D-Fender: they removed the transfer from x-lite
22:42.25Math`to "encourage" people to buy the professional version
22:43.05[TK]D-FenderMath`, I am well aware of that which is why I'm suggesting idefisk and made a point to HIGHLIGHT that fact
22:43.14mvanbaakI think I spent too much time on the chan_skinny driver lately
22:43.32Math`ahhhh sorry about that misreading
22:43.48[TK]D-Fendermvanbaak, even are clearance prices, its just means you sold your soul at a premium ;)
22:44.00MrWupanyone?
22:44.01mvanbaakgheh
22:44.17mvanbaakbut at least my cisco phones are now working great with asterisk again
22:44.20Math`mvanbaak: for that to happen you need to send us BOTTLE_STATUS messages
22:44.26mvanbaakand no more buggy chan_sccp
22:44.35mvanbaaklol Math`
22:44.37mvanbaakgood idea
22:44.58*** join/#asterisk CrazyTux[m] (n=CrazyTux@cpe-24-162-48-94.houston.res.rr.com)
22:48.41neverblueto setup Ekiga to work with Asterisk, who has to give up port 5060?
22:48.52[hC]Can polycoms ring the phone's ringer when a call comes in on, say, line2 - while you're on line1, instead of playing a call waiting tone in your ear (and disrupting audio)?
22:50.04[TK]D-Fender[hC], nope.
22:50.25[hC][TK]D-Fender: poo.
22:50.25[TK]D-Fender[hC], Its in-line audio or nothing
22:50.39[hC][TK]D-Fender: polycom is starting to kind of annoy me with some of their limitations in business use.
22:51.55*** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org)
22:51.56[TK]D-Fender[hC], Lets say you're on a call and some ass keeps calling you and you don't want to answer, and your CALLER hears your phone ringing off the hook (no subtle irony here!) and gets annoyed.  Or feels you are ignoring people, and maybe he'll be next.  Or those working around you getting pissed off at the constant noise....
22:52.25[TK]D-Fender[hC], All of the above exellent reasons why your request should be BURNED AT THE STAKE.
22:52.40neverbluecan I listen to my voice main via a browser using php?
22:52.43neverbluemail*
22:52.51[TK]D-Fender[hC], What phone out there has sunch an "Annoy Mode" on it?
22:52.55[hC][TK]D-Fender: I agree, but lets say you sell these phones to people who are used to dealing with that, and their 65 year old secretary bitches at you ever day to make it work the way shed like it to, and you have to say 'oh yeah these new phones you bought dont do what you want'
22:53.13[hC][TK]D-Fender: old key systems that i upgrade basically 100% of my clients from. :P
22:53.43[TK]D-Fender[hC], waitasec.. you're the one asking to ramp up that bitch's CW beep with something deafening, no? :)
22:53.45[hC]I agree with you, but theres just a few things that i wish were at least an option to get needy clients off my back without having to tell them it cant be done.
22:54.11[TK]D-Fender[hC], Thell them it can't be done or they'll walk all over you till you give up the farm.
22:54.20[hC][TK]D-Fender: i dont want it to be louder, she just doesnt want CW in her ear that cuts off what the caller is saying while it beeps, shed rather it be a low audible ring from the base of the handset
22:54.31[TK]D-Fenderneverblue, PHP is a scripting languange, not a PHONE.
22:55.00[TK]D-Fenderneverblue, Next thing we know, you'll be asking how to talk into your mouse......
22:55.36[TK]D-Fender[hC], get her a 2nd phone and ring both simultaneously all the time and silence the beep.
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22:55.52[hC][TK]D-Fender: hahaha. good call :)
22:56.17[hC][TK]D-Fender: you're the guy that implemented a contact directory using the microbrowser right?
22:56.30[TK]D-Fender[hC], I've done a bunch of things like that, yes
22:56.53[hC][TK]D-Fender: can you dial from the browser? (I mean, i know its possible, but did you set it up that way?)
22:57.15[TK]D-Fender[hC], yup
22:57.31mvanbaaklatero all
22:57.44[hC][TK]D-Fender: would i be able to snag a copy from you? I need to try implementing the same thing.
22:58.00neverblue[TK]D-Fender, lmao
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23:00.49neverblueto setup Ekiga to work with Asterisk, who has to give up port 5060?
23:01.18BSD_Techwhat is ekiga
23:01.34neverbluesoftphone
23:01.49BSD_Techno one has to give up port 5060
23:02.00lesouvageInbound calls come in a queue and agents can login into the queue ans start answering the phones. The caller ID shown on Idefisk is the last account in IAX.conf instead of the number of the calling party.  What can I do to fix this?
23:02.04BSD_Tech5060 is just the port you send your reg request to
23:02.05neverbluethey will both use that port though
23:02.36BSD_Techset one to use 5060 and 1 to use 5061
23:02.51BSD_Techit shold ajustable in the softphone
23:02.56neverblueits not
23:02.59neverbluei dont think
23:03.13BSD_Techthen its a pisspoor soft phone
23:05.23BSD_Techits gnomemeeting rebranded
23:05.27BSD_Techit is settable
23:05.45BSD_Techyou need to look in the configs
23:08.19lesouvageI have no idea why but I change the last account, reload, turn it back to the original and reload and now it is working ok.
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23:37.29kink0any idea why Zap fails over 60 calls from the same peer ?
23:37.40Qwell[]zap doesn't have peers
23:38.12kink0I know, but if I carry lets say 58 calls from one peer
23:38.30kink0all calls over 60 are returned as cause 47 from zap
23:38.58kink0but if I send 50 calls from that peer, and then 50 more calls from another one pees, I get 100 calls
23:40.11n00dleUm... Zap only has channels...
23:41.35kink0n00dle, yes,  I understand, but why when peer 1 wwhen has about 58 consecutive calls fails in ZAp dialing , while other peers can still dialing more calls in the same exten ?
23:41.54justdaveyou can only have as many simultaneous calls on Zap as you have channels available.
23:42.18kink0justdave, I have 120 channels, but I can not get over 60 from the same peer.
23:42.26n00dlekink0: What hardware and software are you running?
23:42.27justdaveahh, ok.
23:42.29kink0I can add other peer here, and the lets up 120 calls
23:43.01kink0n00dle, Digium TE412 on Dual Xeon 3.2/2 SATA Supermicro
23:43.39kink0yes, by peer, no more than 60 simultaneous calls
23:43.41n00dleAll 4 E1s to same provider, then...
23:43.58n00dle...with all 120 channels terminated with that provider.
23:44.08kink0n00dle, yes, and are grouped. I have also tryed to do not groupes, in 4 groups, but the same
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23:46.08n00dleUnfortunately, I only have analog and SIP/IAX trunks... no experience with * and T1/E1.
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23:46.58kink0i c
23:47.13n00dleNo PRI available here. :(
23:47.50kink0well... see u tomorrow
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23:48.09n00dlePerhaps... it's nearly quitting time here.
23:48.21fiber0ptiWhat does the follow message in the CLI mean: "Got SIP response 500 "Internal Server Error" back from <ip address>"
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23:49.23justdavefiber0pti: probably means you have a Polycom phone and you can probably ignore the message
23:49.47fiber0ptithanks
23:49.53justdaveIIRC it has something to do with status notifications and the Polycom phone doesn't quite understand them
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23:50.21justdave(it understands part of it because the icons do the right thing on the phone, but that's just what I've heard)
23:50.25nomadsoulhi
23:51.37nomadsouli just discovered asterisk and i think it is very amazing
23:51.40nomadsoulbut
23:51.40justdavewe have lots of Polycoms, and get that periodically from several of them, but they otherwise seem to work fine
23:51.51justdavethere's always a "but" :)
23:52.08nomadsoulcan i use normal modems instead of using the $500 cards that you see on asterisk webshop?
23:52.18nomadsouljust for testing
23:52.47nomadsoul?
23:53.04justdaveas long as they have voice capability and either work with the zaptel drivers or provide channel interfaces for Asterisk, sure
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23:53.27nomadsoulmmm
23:53.41justdaveI've seen older Wildcards on sale on eBay for cheap
23:53.44justdavebut YMMV
23:53.58nomadsoulwhat is a channel interface?
23:54.00nomadsoul:P
23:54.05nomadsouli-m pretty noob
23:54.08justdavea chan_XXXX.so file for asterisk to load
23:54.17justdavethat provides the interface between asterisk and a hardware driver
23:54.20nomadsouland what is this used for?
23:54.28nomadsoulok
23:54.52justdave(or between Asterisk and a network protocol, in the case of sip, iax, and friends)
23:55.23nomadsouland just another question
23:55.28nomadsouli swear it is the last
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23:56.27nomadsoulmmm nothing
23:56.30jlcoxHi all, I have a probe with 1.4.4 on gentoo. It all works on a local net but I get no audio when connected via the internet?
23:56.32nomadsouli-ve already checked :P
23:57.34jlcoxis there a config line i am missing ?
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23:58.14bochwould be a heavy load an * with 120 calls, with many MYSQL() querys/playback/record/bridge in the dialplan?
23:59.08nomadsouljustdave: sorry, i was thinking... can i do some testing without a Wildcard? i mean all over ip?

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