IRC log for #asterisk on 20070503

00:00.50*** join/#asterisk rhombus (n=sfbosch@dsl-cap-66-18-218-36-cgy.nucleus.com)
00:01.05*** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25)
00:01.08rhombusAre Asterisk variable names case sensitive?
00:01.37uwerhombus, you mean those used in agi ?
00:01.51rhombusno, I just mean those used in the dialplan
00:02.01*** join/#asterisk BB|AtWork (n=karl@38.99.18.98)
00:02.26rhombusif I set a custom variable, is it case sensitive? Will ${goat} be the same as ${GOAT} or are they handled as separate variables?
00:02.32km-uwe: yeah, I'm seriously bummed.  I tried it by sox'ng the wav beforehand and it works fine
00:03.02BB|AtWorki have some users complaining of echo problems, but unfortunatly i can't think of a good way to test this. any time i use my mobile phone i don't get anything.  Does any one know of any test numbers i can call?
00:03.06*** join/#asterisk DrukenHME (n=jdumais@CPE000854ddcdb1-CM00137189cb0c.cpe.net.cable.rogers.com)
00:03.15rhombusAha -- I guess they are not.
00:03.19rhombusthanks anyway!
00:03.22*** part/#asterisk rhombus (n=sfbosch@dsl-cap-66-18-218-36-cgy.nucleus.com)
00:03.28BB|AtWorki've been searching for web pages with related info but have not come up with anything so far besides sip addresses
00:05.03uweBB|AtWork, maybe you have issues with irq sharing or inturrupting too much
00:05.13BB|AtWorkhrm
00:06.04uweor network issues
00:06.04uwemaybe ...
00:06.08*** join/#asterisk sbingner (n=sam@pdpc/supporter/sustaining/sbingner)
00:06.08BB|AtWorkBest: 100.000000 -- Worst: 98.437500 -- Average: 99.943441
00:06.19BB|AtWorkits a local physically seperate 10/100 network
00:06.30BB|AtWorkusing a pri t1
00:06.42BB|AtWorkinterupt issue sounds plausible
00:06.45BB|AtWorkit only happens some times
00:06.58BB|AtWorkwhat are bad numbers for zttest?
00:07.06Alex_20storna, i can't see it for now
00:07.30Alex_20but after i see the 223 in the sip.conf
00:07.36Alex_20what i have to do
00:07.40*** join/#asterisk colinm_ (n=colol@VDSL-130-13-104-63.PHNX.QWEST.NET)
00:08.20km-Sucks, I thought I could do this project without having to make asterisk source code changes, but unfortunately AGI just doesn't do the magic I need
00:08.50uweBB|AtWork, how does your /proc/interrupts look like
00:09.42uweive just had similar problems for the last 3 weeks :)
00:09.57*** join/#asterisk kiscokid (n=xxx@208.106.33.66)
00:10.46BB|AtWork10: 1719405387          0          XT-PIC  3w-9xxx, uhci_hcd, uhci_hcd, wcte11xp, wctdm
00:11.18BB|AtWorkhttp://rafb.net/p/3JFv9H73.html
00:11.30kiscokidAnyone know why adding after adding a service provider it won't show up as a choice when adding a new calling rule?
00:11.48*** join/#asterisk l3jj (n=chatzill@dsl093-001-038.det1.dsl.speakeasy.net)
00:12.06kiscokidsorry, I meant to ask that in asterisknow
00:12.20uwei think your  wcte11xp, wctdm are sharing inturrupt with many other stuff
00:12.35stornaalex20_dunno
00:12.47uweand all is hitting the same cpu BB|AtWork
00:12.55BB|AtWorkuwe, looks like its sharing with the 3ware card
00:13.01BB|AtWorkhow would i go about fixing this?
00:13.11uwewait a minute
00:13.30uwethere is the only article i actually found useful
00:13.44uwehttp://www.asteriskguru.com/tutorials/pci_irq_apic_tdm_ticks_te410p_te405p_noise.html
00:14.26uweBB|AtWork, i had to disable some hardware from the bios to get the digium on an inturrupt that is not shared
00:14.40uweand ah, use lspci -vb to check inturrupts too
00:14.45BB|AtWorkthink it would be ok if 2 digium cards shared?
00:14.54BB|AtWorki think i can get the 3ware card onto something else
00:14.55uweno ... i dont think so
00:15.02BB|AtWorkdamn this might be hard
00:15.07uweyes :)
00:16.25uwetry first with smp_affinity for that inturrupt
00:16.54*** join/#asterisk Alex_20 (n=Alex_20@139-62.al.cgocable.ca)
00:17.12uwebecause the eth1 is making also a lot of inturruption
00:17.52*** join/#asterisk [hC] (n=hardcore@mail.metrobridge.com)
00:18.26*** join/#asterisk zotz (n=zotz@24.244.163.157)
00:18.48JTBB|AtWork: definitely not ok for them to share
00:19.12BB|AtWorkhurk
00:19.23JT"hurk"?
00:19.30BB|AtWorkare there any problems with using apic and asterisk?
00:19.43JTBB|AtWork: not that i've found
00:19.51BB|AtWorkfor somereason it seems to be off
00:20.01BB|AtWorki'll have to see what caused that
00:20.50BB|AtWorkyeah tahts probably what the problem is
00:21.43BB|AtWorkwe arn't using the other telephony card yet (its a 4 port fxo) but the raid controller and the pri on the same irq is probably causing trouble
00:21.43uweBB|AtWork, you have to make sure they are not sharing them by checking the output of lspci -vb too
00:22.10BB|AtWorkso i can solve this by either turning off enough devices that they all end up on there own.  or turn on apic?
00:22.18uwebecause having them on diffrent irq when using apic doesnt mean the are not sharing the same inturrupt
00:22.26BB|AtWorkah
00:22.42Alex_20so now, what should i do?
00:22.50BB|AtWorkthis is going to be painful
00:23.03uwethats what it understood in the last two days
00:23.24*** join/#asterisk jtknapp (n=skip@65-126-63-1.dia.static.qwest.net)
00:23.35uweum , not really, simply waiting between reboots :)
00:23.45BB|AtWorkhave to do it out of office hours
00:23.53BB|AtWorkand there are 3 cards sharing irq's :)
00:24.15uweBB|AtWork, ive had this exact experience 2 days ago :D
00:24.28*** part/#asterisk jtknapp (n=skip@65-126-63-1.dia.static.qwest.net)
00:24.29BB|AtWorkgoing to come in way late tomorrow
00:24.30BB|AtWork:)
00:25.24uwehahahaaa ! i swear to the devils this must be dejavu
00:26.18*** join/#asterisk docelmo (n=vircuser@c-76-99-157-112.hsd1.de.comcast.net)
00:26.25BB|AtWork:)
00:26.28*** join/#asterisk |dennis| (n=dennis@200.32.233.82)
00:26.37BB|AtWorkso.... come in late... or go home for a chunk of the day
00:26.41BB|AtWorktough decision
00:27.15uweid choose come in late
00:27.24BB|AtWorkheh i already come in at 9 :)
00:27.27BB|AtWorki'm going to be bored
00:27.42BB|AtWorkbut i think i'll chose that and just stay up till 4 or something
00:28.08uwe4am or 4pm ?
00:29.10*** join/#asterisk Insanity5 (i=Insanity@66-225-36-14.dynamic.tbc.net)
00:29.12Insanity5Does asterisk offer any built in debugging to show jitter/out of order packet information/etc or anything else that may help show a poor call quality problem?
00:30.34BB|AtWorkdid you try set verbosity 99 set debug 99 and pri intense debug (if you have a pri line)
00:31.34kiscokidIs there a way to trace calls going through the dialplan?
00:31.53*** join/#asterisk wolferine (n=profx@unaffiliated/wolferine)
00:32.39Insanity5BB|AtWork - Verbo is at like 4.
00:32.42Insanity5BB|AtWork - IT's SIP
00:32.53Insanity5BB|AtWork - internet is suspect
00:33.11BB|AtWorktcpdump can show you out of order packets i think
00:33.56Insanity5BB|AtWork - reconstructing might be a PITA
00:33.57Insanity5:(
00:35.12*** part/#asterisk kiscokid (n=xxx@208.106.33.66)
00:37.59*** join/#asterisk sbingner (n=sam@pdpc/supporter/sustaining/sbingner)
00:38.09*** join/#asterisk defswork (n=andy@mailgate2.3gcomms.co.uk)
00:40.05*** join/#asterisk fx0 (n=fx0@cypher.punk.net)
00:42.59Alex_20sorry i don't remember :S
00:43.48*** join/#asterisk Avochelm (n=damo@gw-morphett.koalatelecom.com.au)
00:44.11JTBB|AtWork: never share any irq with a digium card
00:44.17JTlet alone a raid controller
00:44.28JTthat and ethernet are about the worse things to share with
00:46.38*** join/#asterisk Alex_20 (n=Aex_20@139-62.al.cgocable.ca)
00:47.25red9012I have problem with dtmf recognition while in a call trasfer mode (ex press 1 accept/refuse call).   DTMF works in all other cases.
00:49.31Alex_20i have zap config and asterisk 1.4.2
00:49.51Alex_20i want to change a ext number of a phone to 223 for 229. how can i do that?
00:50.03Alex_20help me pls
00:53.58*** join/#asterisk FastFeet (n=FastFeet@CPE0013109fd25b-CM000f9fa60d7a.cpe.net.cable.rogers.com)
00:56.59*** part/#asterisk Alex_20 (n=Aex_20@139-62.al.cgocable.ca)
01:00.10blitzrageanyone know how to make Asterisk not load a module again after you tell it to preload in modules.conf?
01:02.08*** join/#asterisk Igbothom_III (n=Hilton@office.quarkit.com.au)
01:02.26*** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25)
01:07.20*** join/#asterisk AndrewGearhart (n=chatzill@h122.183.88.75.ip.alltel.net)
01:08.21AndrewGearhartis it possible to change what extension a call is forwarded to based upon the callerID information?
01:08.50toombaloombayea, theres a variety of ways you can do it
01:08.54toombaloombadepending on what exactly you want to do
01:09.22*** join/#asterisk bintut (n=bintut@203.125.63.150)
01:10.12AndrewGearharttoombaloomba: I was thinking along the lines of if it's one of these numbers: a, b, c... forward to extension X
01:11.40toombaloombaon incoming calls?
01:12.29toombaloombayou could just do If callerid(num) = 123456789
01:12.33toombaloombaor whatever, its easy
01:12.45toombaloombalook on the wiki for syntax
01:13.40AndrewGearharttoombaloomba: k, sorry if I am asking an overly simple question that is well documented. :( was trying to answer a question for somebody over in MythTV
01:13.41toombaloombaor exten => 500/123
01:13.50JTerr that sets callerid
01:14.02JTthat's more like it
01:14.16toombaloombayea JT its not proper syntax
01:14.19toombaloombaits just an idea
01:14.33JT500/123 would work
01:14.43AndrewGearhartbut, yes, the idea is that it would be callerid from the outside line(s)
01:15.56*** join/#asterisk mike38533 (n=omar@dsl092-214-151.atl1.dsl.speakeasy.net)
01:16.12*** join/#asterisk burt75 (n=burt@189.157.128.236)
01:16.18burt75hello guys
01:16.40burt75I need help (payed) with FXO reverse polarity issue , wonder who can help me please :D
01:18.15mike38533Hi all, I am trying to use T.38 faxing with an ATA, by receiving the call with the ATA and sending it over to * with IAXModem. Can anybody give me any pointers on how to configure this? My tests haven't been successful as the fax tone isn't recognized.
01:18.52*** join/#asterisk jm|home (n=jm|home@dilbert.jamiem.com)
01:19.27burt75Looking for a consultant specialized in FXO ports
01:29.33FuriousGeorgeim taking a poll.  i have two identical systems.  one has two tdm400p the other has only one.  They have identical OS and kernel and everything else.  the former deadlocks, the latter does not.  what are the odds replacing the tdm400p (with one sangoma a200 series) will resolve the problem?
01:30.10danphave you tried a memory test?
01:30.16JTtried swapping the cards?
01:30.24danpthat too
01:31.03SwKchecked for shared interupts
01:31.17SwKother stuff running on the box that is starving it for CPU time
01:31.47shido6:)
01:33.48blitzrageSwK: y0!
01:37.58shido6no other res_odbc.so lines in there no autoload yes crap?
01:39.25SwKy0 blitzrage
01:44.09*** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga)
01:47.22*** join/#asterisk tomcontr3 (n=tomcontr@101-77-246-201.adsl.terra.cl)
01:47.38tomcontr3hi,  can anyone help me with the G729 codec?
01:48.15tomcontr3I just got that codec,  and a cofigure  my extentions with  allow=g729  and disallow=all
01:48.28tomcontr3but every time I try to call an extention, I get Service Unavailable
01:48.46blitzrageSwK: have you ever done a preload, then saw that the module loads a 2nd time during the standard module load?
01:49.07tomcontr3how can I do that?
01:49.24*** join/#asterisk hansin321 (n=hansin32@c-71-196-138-170.hsd1.co.comcast.net)
01:49.45shido6hehe
01:50.03blitzrageshido6: we should have been having our conversation out here :)
01:50.10shido6yeah
01:50.48blitzragemaybe just paste the last part of the conversation.... ?
01:50.50shido6i think it would be simpler to add the donotloadagainmodule => module.so but the 2nd version should have a weighted module loading system.
01:50.51blitzrageanyone interested? :)
01:51.14tomcontr3anyone?
01:51.27shido6parts of the system, necessary parts weigh more or are more relevant to the core than others
01:51.48shido6then u have the option to FORCE the order that the modules load
01:52.19shido6that sounds like what you need in loader.c
01:52.32shido6force the order but have the system bitch at you about its relevance.
01:54.52tomcontr3?
01:58.01*** join/#asterisk ccole (n=cole@cpe-24-166-57-30.neo.res.rr.com)
02:02.06LeddyHMhrm
02:02.30LeddyHManyone know which settings change the date/time format on a polycom phone?
02:02.47LeddyHMI'm guessing it's sip.cfg, but can't make heads or tails of which does what
02:02.55*** join/#asterisk l2cache (n=admin@62.180.8.67.cfl.res.rr.com)
02:03.56l2cacheI have a box that can make calls successfully using g729.  Because i have the format_g729 module in usr/lib/asterisk/modules.  Why isn't   background(filename.g729) playing back?
02:03.58justdavelook for the SNTP tag
02:04.10shido6ipmid.cfg
02:04.14LeddyHMthat changes location on the screen?
02:04.20shido6LEddyHM i think its in ipmid.cfg
02:04.35justdaveoh, sorry, I thought you meant setting the time, didn't realize you meant how to display it
02:04.59shido6dattime
02:05.04shido6yes has "Top"
02:05.06shido6in there
02:05.16shido6lcl.datetime.date.dateTop="1"
02:05.32shido6what does lcl.datetime.date.dateBottom="1" do?
02:05.33shido6:)
02:06.33l2cacheany takers?
02:06.53justdaveg729 is a licensed codec, isn't it?  do you have a license?
02:06.54shido6?
02:07.06shido6".g729 ?
02:07.07l2cachenot the open source on...i am only using passthrough
02:07.09LeddyHMI have myt date at the bottom, and datetop says 1
02:07.11LeddyHMweird
02:07.14shido6is that the suffix?
02:07.16l2cacheone*
02:07.20l2cacheyes
02:07.23justdavepassthrough works without a license.
02:07.28shido6dont include the suffix
02:07.40l2cachegood tip..(trying now) :)
02:07.57shido6the original filename should have one
02:08.07l2cachegotcah
02:08.10shido6but in the dialplan u dont use it with background or playback
02:08.49justdaveLeddyHM: if you have the manual for the Polycom phones it on page 83
02:08.57justdavesection 4.6.1.3.2
02:09.17justdavethere is no dateBottom
02:09.34justdavedateTop is 1 for date above the time, 0 for time above the date
02:10.58justdavesupposed I should clarify, I'm looking at the manual for IP501 :)  not sure they're all the same
02:11.41justdaveguess it is, it says "SoundPoint®/SoundStation® IP SIP" on the front cover, no model number
02:12.45l2cacheeverytime i try to do a playback it says unable to open "filename" does not exist in any format
02:12.50l2cacheand i know its in the right dir
02:12.59l2cachebecause i can playback gsm fine
02:13.22justdavel2cache: show formats
02:13.30justdavesee if it's listed
02:14.07l2cachei dont have that command...
02:14.15justdavehmm, maybe that's 1.4 only
02:14.23l2cacheim using 1.4
02:14.40l2cacheshow codecs?
02:14.46justdaveshow codecs seems to work
02:14.51l2cacheyeah
02:14.51justdavebut that's not quite the same thing
02:15.34l2cachei have the format_g729 module..and can talk fine...why wont it playback a .729 file then?
02:15.42justdaveshow modules like format
02:15.44l2cachesays does not exist in any format.....maybe its a diff g729
02:15.54justdave.729 or .g729 ?
02:15.55LeddyHMI'm wondering about 501 specifically
02:16.02lee_is_meQuestion: ZT_CHANCONFIG failed on channel 1: No such device or address (6)
02:16.03l2cacheahhh..good point
02:16.06LeddyHMis the doc available for dl?
02:16.12*** join/#asterisk Putzz (n=me@CPE00155824933c-CM00111ae07ac2.cpe.net.cable.rogers.com)
02:16.15lee_is_mei have zaptel.conf and zapata.conf configured
02:16.16l2cache.g279
02:16.24l2cachei need .729
02:16.40l2cachecorrect?
02:17.19lee_is_mebut keep getting the above error with /sbin/ztcfg -vv
02:17.32justdaveLeddyHM: http://www.polycom.com/resource_center/0,,pw-16595-16595,FF.html
02:18.25l2cachei think my problem is im playing back the wrong sound files....wrong g729 files :)
02:18.32justdaveLeddyHM: the Admin Guide is the one you're looking for
02:24.16ccoleWhat is a quality, reputable place to get IAX service through?
02:24.42ccoleSo far, I am pleased with NuFone, but they do not have any 330 area codes :~(
02:24.45LeddyHMfound it thanks
02:24.54LeddyHMeven if my config is contradictory :|
02:37.19*** join/#asterisk sione (i=sione@208-46-202-201.dia.static.qwest.net)
02:38.59sionecan anyone help me trouble shoot a SIP loop problem?
02:39.32*** join/#asterisk sbingner (n=sam@pdpc/supporter/sustaining/sbingner)
02:41.25sioneeveryone is asleep?
02:41.32Putzzzzzzzzzz
02:41.42sionebummer
02:41.53Putzzwhats your problem?
02:42.26sionetrying to get a phone behind asterisk-A to call a phone behind astrisk-b via SIP all the way
02:42.48Putzzok
02:43.03sione37.786956  192.168.0.1 -> 192.168.3.129 SIP/SDP Request: INVITE sip:6001@192.168.3.129, with session description
02:43.12sionetahts the 1st
02:43.17sione<PROTECTED>
02:43.25sione<PROTECTED>
02:43.27Putzzdont flood channel
02:43.38Putzz~pb
02:43.50jboti heard pb is a place to paste your stuff without flooding the channel - try  http://pastebin.ca, or http://channels.debian.net/paste
02:43.50sionetrying to post slowly sorry
02:43.50Putzzpostbin.ca
02:43.51shido6pastebin.ca your post
02:43.51Putzzor whatver u prefer
02:44.03Putzz*pastebin.ca oops
02:44.05sionelet me check it out
02:45.06sionephone (2222) calls (6004) when when 6004 picks up Astersk sends an invite to 2222 with 6004 then asterisk reportd loop and kills the channel
02:46.29*** join/#asterisk bluelinq (n=bluelinq@dsl-7-36.cofs.net)
02:47.03bluelinqhey guys, anyway to make a conference call using a 7960 with chan_sccp wiouth using the meetme stuff?
02:47.35*** join/#asterisk raptorra1 (n=rathomps@cpe-66-25-25-138.houston.res.rr.com)
02:48.17*** join/#asterisk irule (n=irule@189.164.43.19)
02:50.15*** join/#asterisk tenzind (n=tenzind@202.144.144.77)
02:52.15*** join/#asterisk NovceGuru (n=asdf@oh-71-50-248-25.dhcp.embarqhsd.net)
02:52.24l2cachei need a g729 translation codec for a wrt54g...anyone ever heard of this
02:52.36l2cachei know its processor specific so it'd be an awesome find!
02:53.37wunderkini really doubt it would have enough juice for g729
02:53.48l2cachei know..but i need it for testing reasons
02:54.00l2cacheany ideas if anyone's tried it?
02:54.06wunderkinwhat good is testing if it will not even work
02:54.11l2cacheit will
02:54.16l2cacheis a 266mhz box
02:54.16wunderkinha
02:54.24l2cacheand i have a similar box doing that fine
02:54.36raptorra1wunderkin: with out testing one will never know
02:54.37l2cachebut you've never heard of anyone trying then?
02:54.56*** join/#asterisk thoughtpolice (n=austin@c75-111-138-216.plaicmtc01.tx.dh.suddenlink.net)
02:55.27l2cache?
02:55.44raptorra1http://forums.whirlpool.net.au/forum-replies-archive.cfm/356705.html
02:55.48raptorra1might be of interest
02:56.03*** join/#asterisk mosty (i=mostyn@60-241-198-194.static.tpgi.com.au)
02:56.11l2cacheyeah ive seen that one...one guy mentions 729..but no others did
02:56.47raptorra1what processor is in the wrt54g
02:56.57mitcheloca p4
02:57.00l2cachei can passthrough fine..but since the module is processor specific i can't find a translation codec module
02:57.07l2cachemitcheloc you are no help
02:57.13mitcheloc:)
02:57.41l2cache32-bit MIPS architecture processors by broadcom
02:57.50raptorra1I would guess an arm, but I'm using my wrt54g and really don't feel like cracking it
02:58.28l2cachehttp://en.wikipedia.org/wiki/MIPS_architecture
03:00.19l2cachei really wish some one knew about this
03:00.34NormanAtholhey guys i have just set up an IAX trunk but every where i have read that the sending server need type=user and the reciving server need type =peer this dosent work for me i need to have both of the set as the same type could they be guides based on 1.2 or somethign
03:00.50mostywe are intermittently getting some dropouts on iax calls, with trunk_queue: Maximum trunk data space exceeded to <host> messages in the logs. looking at the code there seems to be a fixed upper limit for the amount of iax trunking you can do. is there any way around this?
03:01.00l2cacheif they are talking both ways use type=friend
03:03.30NormanAtholyeah i will be using type friend but then that raises another question for me the sleave sever connects to the master server is ther anything different i need to do in the dial plan if the master server wants to call someone on the slave server or once its registered then it should all work fine
03:03.45l2cacheshould work find both ways
03:03.52*** join/#asterisk stkn__ (n=stkn@gentoo/developer/pdpc.active.stkn)
03:04.28NormanAtholok thanks
03:07.48sioneanyone know why i cant select crc_ccitt in my make menuconfig for kernel 2.6.16?
03:08.51*** join/#asterisk sysreq (n=sysreq@modemcable171.134-81-70.mc.videotron.ca)
03:10.08Insanity5Does anyone know a good method for SIP call quality issues?  Out of order packets/jitter?  An easy method to see these things?
03:10.09sionenever mind, i just answerd my own quetion on that
03:11.20*** part/#asterisk l2cache (n=admin@62.180.8.67.cfl.res.rr.com)
03:12.32sioneweird i just found something out with my SIP problem
03:12.35emirdamadiI need to ask a question guys
03:12.42emirdamadican anyone help me?
03:13.03hansin321Go ahead and just ask.  If someone knows, they will most likely answer.
03:13.04Strom_Mis it still "how do I patch asterisk">
03:13.05Strom_M?
03:13.05JunK-Yemirdamadi: asking a specific question is always much better
03:13.45sioneI find using softphones my looping problem goes away
03:14.43*** join/#asterisk jmacz (n=jmacz@190.24.103.194)
03:15.41*** join/#asterisk hacim (n=micah@debian/developer/micah)
03:16.50hacimcan anyone give me an idea of where I can get a cheap DID in sao paolo, brasil?
03:17.21hansin321emirdamadi:  The reason it is best just to ask the specific question as opposed to asking if anyone can help, is that no one can really know if they can help unless they know what the question is.  Therefore it is just better to ask.  You can't be insured of a response, but that's just the way IRC works...
03:18.27Insanity5Do certain codecs deal better with jitter than others?  IS there an easy way to at the cost of delivery delay, buffer two second of audio each way on the asterisk box or something and have it resort to correct for jitter?
03:20.55mostyInsanity5, jitterbuffer
03:22.05Strom_MTWO SECONDS?!
03:22.05Strom_Mare you completely mad??
03:22.08Strom_Myou can't do two seconds of jitter buffering - no one will be able to converse naturally
03:22.22blitzrageStrom_M: over
03:22.34blitzrageStrom_M: the bird flies at midnight. over.
03:23.03blitzragedamnit.... Jays just lost :(
03:24.11wunderkinthe rooster crowed at 12:01, over
03:24.11Strom_Mthree nine seven zero six
03:24.39Insanity5Strom_M - Well ok, what is normal on a cell phone?  It has to be at least 400 ms at each end.
03:24.59Strom_Mmaximum total allowable call latency is 400ms
03:25.21Strom_Mso if you want to have 400ms of jitterbuffer, that means you have to have 0ms latency between endpoints
03:25.40blitzrageI use 30ms
03:25.48blitzragemight even be 20ms
03:25.52Strom_Mnow that's sanity
03:32.27Insanity5Strom_M - What's the latency on a cell-to-cell call?
03:33.27*** join/#asterisk dlynes_laptop (n=dlynes@d207-216-161-56.bchsia.telus.net)
03:37.13sionewhats the differance from "Packet2Packet Bridging" and "Native bridging" ?
03:38.11Strom_MInsanity5: depends...likely no more than 40-50ms assuming it's within the same metro area
03:38.45Insanity5Strom_M - Ok, will I always wondered why the data cards have 300ms latency to the first hop, minimum.  Cross country calls must be a little nasty then :)
03:38.52sionemysoft phones are showing up as packet2packet bridging and my Linksys PAP2 shows up as Native
03:38.54*** join/#asterisk IguanaNed (n=you@CPE000625db3f84-CM00111ae43f1e.cpe.net.cable.rogers.com)
03:39.15Insanity5Strom_M - WOould updating my ages-old asterisk version help any, or realistically, a perfectly working installation and a crappy internet connection can't be improved much?
03:41.19Insanity5Or even a way to debug jitter and see how bad it is with tethereal/tcpdump would be nice.  I can't find a usable time stamp/ordering determination mechanism, just the time it arrived.
03:41.41sionewireshark can do it
03:42.10*** join/#asterisk Plecebo (n=larry@D-128-208-60-80.dhcp4.washington.edu)
03:42.11Insanity5sione - How so?  I got the packet capture...
03:42.31sioneWireshark has a good SIP/RTP analist<sp>
03:42.44Insanity5Take it and bring it to a win32 machine then?
03:42.50Insanity5No gui on my server, just tethereal
03:43.04sioneya so you have to bring it to a gui computer
03:43.38Insanity5Ok, did not know any such tool exists :)
03:43.39NOT_guruanyone mind my asking a trixbox 2.0 question?
03:44.08sionewireshark is ethereal
03:44.31sioneerthereal is no longer been develop as the name ethereal they moved to wireshark
03:45.18NOT_gurushould I allow kudzu to touch my tdm804p?
03:45.30Insanity5sione - They still call the binary tethereal, even off the new wireshark compile.
03:45.44NOT_gurushould I configure it through kudzu there by creating eth1
03:45.52Insanity5I don't know why... ethereal is wireshark, but tehereal is not twireshark yet :)
03:46.45sioneheh im so use to using ethereal so im glad they didnt change that name
03:49.59FuriousGeorgeanyway, to those people who answered me last time, sorry to repeat myself, but right after i asked the question my roommate pulled me away to look at his failed psu fan
03:50.31*** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25)
03:50.32FuriousGeorgei have two identical systems.  one has two tdm400p the other has only one.  They have identical OS and kernel and everything else.  the former deadlocks, the latter does not.  what are the odds replacing the tdm400p (with one sangoma a200 series) will resolve the problem?
03:50.49FuriousGeorgei have not tried swapping the cards (dont have spare modules/cards)
03:50.59FuriousGeorgeim not sharing interrupts
03:51.15FuriousGeorgeand i ran memtest about 5 months ago
03:51.22FuriousGeorgei guess i should run it again
03:51.26dlynes_laptopFuriousGeorge: sounds like you might have some kind of a hardware issue, possibly with the cpu
03:51.38NOT_guruor check the caps on the board
03:51.44dlynes_laptopFuriousGeorge: are they identical cpus?
03:51.51*** join/#asterisk bmg505 (n=leon@196.209.180.128)
03:51.56dlynes_laptopFuriousGeorge: and are they configured on the bios identically?
03:52.00FuriousGeorgedlynes_laptop: its been my experience that cpus either work or they dont.  generally dont cause instability
03:52.15dlynes_laptopFuriousGeorge: that's not what I'm saying
03:52.16FuriousGeorgedlynes_laptop: bios has to be pretty close to identical, i change few defaults
03:52.21FuriousGeorgesoftware raid
03:52.27FuriousGeorgeoh i get you now
03:52.28NOT_gurubad power in the board ( bad caps ) could cause it
03:52.37FuriousGeorgeNOT_guru: how do i diagnose that?
03:52.46NOT_gurujust look at them
03:52.58dlynes_laptopFuriousGeorge: i'm thinking more like hyperthreading vs non-hyperthreading, smp vs non-smp, and what NOT_guru said, too
03:52.59FuriousGeorgewhat does a bad cap vs a good cap look like?
03:53.01NOT_gurulook for bulging ( at the top mainly ) or even worse leaking
03:53.28NOT_guruI always check caps stamped with a + on top
03:53.31FuriousGeorgeboth boxes have identical mb, identicap processor, non-identical 600 watt psus
03:53.49NOT_guruI would check caps first
03:53.57dlynes_laptopFuriousGeorge: caps == capacitors
03:53.59FuriousGeorgenon-identical not-the-cheapest-generic-memory (patriot and ocz ddr400 non-ecc)
03:54.12FuriousGeorgewhat does a bad cap look like tho?
03:54.12NOT_guruafter that I would pull the second tdm from system A and add it to system B and see if system B starts acting up
03:54.22dlynes_laptopFuriousGeorge: he said it'll be leaking or bulging
03:54.23FuriousGeorgeoh bulging
03:54.47dlynes_laptopFuriousGeorge: the capacitors he's referring to will be the electrolytic capacitors, not the ceramic capacitors
03:54.50NOT_guruthanks dlynes_laptop  I made an assumption
03:54.52NOT_gurumy bad
03:55.14FuriousGeorgeelectrolytic = cylindrical
03:55.31dlynes_laptopFuriousGeorge: correct
03:55.31NOT_guruussually metal top with a colored wrap on it
03:55.36dlynes_laptopFuriousGeorge: and they'll be marked with a '-' sign on one side with a white stripe usually
03:56.12FuriousGeorgeNOT_guru: these cards were in a mb that seemed to have fried on the job (didnt check the caps that time), do you guys think the tdm400p were damadged
03:57.01NOT_gurupossible but I don't think likely unless it was an electrical surge that killed it ( lightning )
03:57.16FuriousGeorgecome to think of it i put these in another system and it deadlocked, but at the time (months ago), i assumed it was that the platform there was unreliable (vai platform asus desktop mb 32 bit)
03:57.42NOT_guruyah  I would goto the system thats fine
03:57.52NOT_guruand roll through the boards 1 at a time
03:57.58NOT_guruand find the bad apple
03:58.15NOT_gurutime consuming yes... thorough  yes
03:58.18FuriousGeorgeisnt that kinda unethical, i mean i could do it but they arent the same business :)
03:58.26NOT_guruoh
03:58.37NOT_gurumy bad  sorry  thought these were on a shelf systems
03:59.15NOT_guruwell  I would still pull the boards 1 at a time at the place with a bad one after checking the board
03:59.19*** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn)
03:59.28NOT_gurusee if its just one or the other
03:59.50FuriousGeorgeNOT_guru: tbh, im leaning toward selling those and getting a sangoma a200
04:00.07NOT_gurusure... but only after eliminating the board
04:00.18NOT_gurunew mobo cost less
04:00.19FuriousGeorgeu mean mb right
04:00.24NOT_guruyes
04:00.41NOT_gurusorry for not having best IRC etiquette
04:00.47FuriousGeorgei guess your right
04:00.51NOT_guruI don't IRC often
04:01.04FuriousGeorgei do, and i still do stuff like that all the time
04:01.14NOT_guruI am only here as I have been having fun with a tdm804p
04:01.18FuriousGeorgeNOT_guru: the only thing is that this mb is brand spanking new
04:01.30FuriousGeorgeliterally installed maybe three weeks
04:01.33FuriousGeorgethe old one fried
04:01.36NOT_guruyeah  thats a touphie
04:01.45NOT_gurubut bad boards do ship from time to time
04:01.53NOT_gurueven from the best manufacturers
04:02.00NOT_gurumobos that is
04:02.11FuriousGeorgef* me
04:02.16NOT_guruwhat motherboard you using?
04:02.24NOT_guruwell  I will say
04:02.32NOT_gurugenerally caps don't go that fast
04:02.40FuriousGeorgetyan s2865 series socket 939 nforce with opteron 165
04:02.41NOT_guruif this system is less than 6 months old
04:02.47NOT_gurui would lean not board
04:02.52FuriousGeorgeme too
04:02.53NOT_gurumy
04:02.57NOT_guruthats a nice board
04:03.03FuriousGeorgethe tdm400 are a couple of years old otoh
04:03.05NOT_guruand tyan is a good company in my book
04:03.18FuriousGeorgei like'em for my servers
04:03.26NOT_guruso lets just guess its the suspect tdms
04:03.37IguanaNedanyone here have experience integrating * with legacy PBX?
04:03.38NOT_guruas you mentioned these were in a system that lost the magic smoike
04:04.04FuriousGeorgei was using an asus a8n as a temp when the original one died (yes, the magic smoke was blue ;)
04:04.14NOT_guruLOL
04:04.38*** join/#asterisk k-man (n=jason@unaffiliated/k-man)
04:04.40FuriousGeorgeanyway, i put these tdm400p in that mb and it behaved poorly but i blamed the platform at the time
04:04.43FuriousGeorgemaybe i should rethink
04:04.44NOT_guruanother nice board that an8
04:04.44k-manhey guys
04:04.50FuriousGeorgeIguanaNed: yes and it sucked
04:04.52k-mananyone been to cebit?
04:05.04IguanaNedFG.. thanks
04:05.17FuriousGeorgealthough looking back on it, i dont know if it was the tdm400p which was my bridge
04:05.39NOT_guruiguana  most from what I understand get a t1 card for the lagacy system and the * box
04:05.45FuriousGeorgeif ur legacy pbx supports a t1 interface type job it may be a different story
04:06.03IguanaNedmy * box is off site
04:06.10NOT_guruheh
04:06.20NOT_gurutoo much for my
04:06.24Insanity5Does ICMP being blocked cause any registration problems?  For some reason the host pings back when I try to register.
04:06.27IguanaNedI want to be able to press 8 and have the pbx call my DID over PSTN
04:06.45NOT_guruoh
04:07.01IguanaNedsimilar to how you press 9 for an outside trunk
04:07.17FuriousGeorgeIguanaNed: simle dialplan digit strip
04:07.18NOT_guruso press 8 on lagacy and have it dial into your * or other way around?
04:07.29FuriousGeorge~s/simle/simple
04:07.42IguanaNedpress 8 on legacy
04:08.16IguanaNedproblem is my comapny7 does not have broadband inet connection
04:08.29NOT_guruthen its all on your lagacy at that point from what I can tell as the * box you can have do whatever after the handoff
04:08.32IguanaNedso I have a hosted box with a nice 100 mbit port
04:08.42NOT_gurubut you have did's on the * box?
04:08.48IguanaNedyeah I got the * part down pat
04:08.50FuriousGeorgeexten => _8X.,1,dial(${TRUNK_OUT}) or exten => _9X.,1,dial(${VOIP_OUT}) or something
04:09.11IguanaNedFurious.. no i need to prgram the legacy PBX
04:09.17FuriousGeorgeoh
04:09.18NOT_guruover thinking furious
04:09.19FuriousGeorge:)
04:09.24NOT_guru=)
04:09.32NOT_guruat least your there to help
04:09.43FuriousGeorgetrying to work on my karma
04:09.49NOT_gurubut yah  same thing  just on your lagacy system
04:09.51FuriousGeorgedeadlocks will make you para-religious
04:09.52IguanaNedI have the 900 page programming manual for the PBX but dont know where to start
04:09.55NOT_guruwhat is your lagacy system
04:10.06IguanaNedNEC Elektra IPK
04:10.08NOT_gurufirst place
04:10.20FuriousGeorgeIguanaNed: replacing it is out of the question, huh
04:10.27IguanaNedunfortunately
04:10.34IguanaNedohterwise I would put * in there
04:10.42NOT_gurulook for a consultant... I am not trying to be a jerk... but he may be able to do it in 10 minutes and it could take you days to find the answer
04:10.55IguanaNedNOT_gur... I hear ya
04:10.59Insanity5Every registration request looks like this and the phone won't register.  Sometimes it does this.  I wait a few hours and it starts working again.  Any ideas?  IT displays this many times.
04:11.00NOT_guruand 10 should be relatively cheap
04:11.00Insanity5284.907180 66.233.151.86 -> 66.225.32.67 SIP Request: REGISTER sip:sip.domain.com
04:11.00Insanity5284.907368 66.225.32.67 -> 66.233.151.86 ICMP Destination unreachable (Port unreachable)
04:11.02IguanaNedi know it is prbbly simple
04:11.22NOT_guruI agree   we have a merlin at work
04:11.24NOT_guruI hate the ting
04:11.28Insanity5That is a ethereal capture from the SIP server side
04:11.44NOT_guruI am putting a * box at a remote office as a demo
04:12.05NOT_guruwhat is your phone system model
04:12.15NOT_gurumight be able to find an answer
04:12.27IguanaNedIPK Elektra Eliant IPK
04:13.14*** join/#asterisk Juggie (n=Juggie@CPE00c049d9f271-CM000a73a18a20.cpe.net.cable.rogers.com)
04:14.58FuriousGeorgeInsanity5: did you change the domain to sip.domain.com
04:15.15Insanity5yes
04:15.16Insanity5:)
04:15.20NOT_guruiguanna  thats uhm
04:15.32NOT_guruI can't find a single doc on it
04:15.40Insanity5It does that over and over furious.  I can't figure it out.
04:16.25FuriousGeorgeis there nat?
04:16.55FuriousGeorgeInsanity5: ?
04:17.03Insanity5yes
04:17.09Insanity5On client side only.
04:17.26FuriousGeorgeshouldnt matter then
04:18.01FuriousGeorgewhats the client?
04:18.37Insanity5hacked vonage box
04:19.07Insanity5User-Agent: Linksys/PAP2-3.1.3(LS)
04:19.38FuriousGeorgeur sure its registering w/ the right port, right?
04:19.43FuriousGeorge5060
04:20.04Insanity5yes
04:20.13Insanity5eterheal cap shows that
04:20.28FuriousGeorgehave you tried x-lite or something just to be sure?
04:21.59Insanity5nope, but could
04:22.07Insanity5It'll work kind of randomly if I just wait :)
04:22.20FuriousGeorgeoh, thats always grat
04:22.23FuriousGeorgegreat*
04:22.39FuriousGeorgegot qualify = yes in the sip.conf peer section
04:22.47Insanity5gah, it's working again
04:23.22FuriousGeorgei would try qualify first then i would try x-lite
04:23.29FuriousGeorgeim gonna make some eggs brb
04:24.48Insanity5Where can you get a reasonably priced DID?
04:25.02Insanity5208 area code :)
04:25.13Insanity5That's the whole state of Idaho...
04:26.19*** join/#asterisk b00gz (n=b00gz@d233-124-245.col.wideopenwest.com)
04:27.29*** join/#asterisk antlers (n=antlers@ip70-173-89-173.lv.lv.cox.net)
04:32.29Insanity5What kind of jitter can I make work in a call?  This has near zero jitter inbound, but outbound from SIP device to server it's high -- 20-40 ms.
04:32.29Insanity5http://img501.imageshack.us/img501/8897/jitteryj7.jpg
04:33.35Strom_Myou have to jitterbuffer on the receiving end
04:34.26Insanity5Why is there no jitter on the one leg but there is on the other?  I need jitter buffer at server?
04:34.54Insanity5And even if a jitter buffer was present -- would having the jitter buffer change the tethereal capture at all?  Or it would just make that jitter tolerable, even though it would stay.
04:36.24NOT_guruI have just upgraded my zaptel drivers so my system could see the tdm804p... kudzu wants to make it eth1
04:36.42NOT_guruI am new to digium cards larger than the 100p  and suggestions
04:37.03NOT_guruoh  I am mainly a BSD guy to so the kudzu thing is new to me
04:38.46Insanity5I really need a packet guru :(.
04:39.32*** join/#asterisk sione (i=sione@208-46-202-201.dia.static.qwest.net)
04:39.52sionehow you limit a SIP user to have only one channel
04:40.21sioneso they cant do 3 way calling or forward calls
04:43.29*** join/#asterisk kuku5 (n=kuku5@c-71-201-219-72.hsd1.il.comcast.net)
04:43.32Juggiehttp://www.voip-info.org/wiki/index.php?page=Asterisk+config+sip.conf
04:43.44kuku5I'm getting massive errors trying to compile rxfax - any takers ?
04:44.00Insanity5Strom_M - You still here?  Can you offer any advice on my packet delay problem?
04:44.11Insanity5Strom_M - you said receiving end but I'm not sure what you meant :)
04:44.34*** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25)
04:46.41*** join/#asterisk Mavvie (n=edwin@ppp22-11.lns1.adl2.internode.on.net)
04:55.39*** join/#asterisk CunningPike (n=CunningP@S010600095b33697f.vc.shawcable.net)
04:57.38*** join/#asterisk dasuberdavid (n=root@user-69-1-11-117.knology.net)
04:58.47*** join/#asterisk Cabal (n=Cabal@unaffiliated/cabal)
05:04.26*** join/#asterisk cgb1911 (n=cgb1911@kim.netcraft.com.au)
05:05.15cgb1911Anyone around able to assist with a SIP peer "Maximum retries exceeded on transmission" problem?
05:06.24cgb1911as in , during an active call that's been operating for minutes, a reinvite occurs from the external peer, Asterisk sends 200 OK, extnernal peer ACK's the OK, asterisk appears to ignore it, retransmitting the 200 OK 6 times before dropping the call?
05:17.42*** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk)
05:21.10*** part/#asterisk dasuberdavid (n=root@user-69-1-11-117.knology.net)
05:22.36kuku5If I load up 1.4, can I use the 1.2 configs ?
05:27.32CunningPikekuku5: Sort of - but quite a bit changed, so you're best bet is to read UPGRADE.txt
05:27.43CunningPikes/you're/your/
05:31.09*** part/#asterisk burt75 (n=burt@189.157.128.236)
05:32.31*** join/#asterisk Zipper_32 (n=None@d154-5-87-181.bchsia.telus.net)
05:34.07Zipper_32Has anybody had experience, or could anybody point me in the direction of an asterisk setup with an intercom / paging system which plays music on a continuous basis until interrupted by pages? I currently have a paging system enabled through the soundcard on my asterisk server, but I now need music playing at all times in the mean time.
05:35.10pipwerkhmmm, intresting
05:35.32Zipper_32Quite interesting.
05:36.48Zipper_32I suppose I could have it route through some other device which will play the music from any source until the page causes an interrupt and takes over... but I was hoping Asterisk would be capable of doing the trick.
05:36.56*** join/#asterisk yidiyuehan (n=yidiyueh@58.185.253.70)
05:37.23yidiyuehanhi, anyone can explain the callgroup= and pickupgroup= ? i am a bit confused with these two lines.
05:37.46pipwerkmaybe you could have some script pause or mute the playing music from your dialplan and unmute after?
05:38.55yidiyuehanoh no, these two lines are related to remote call pickup
05:39.24yidiyuehanbut i am just not sure what exactly they are meaning
05:47.18*** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290)
05:47.35*** join/#asterisk [hC] (n=hardcore@mail.metrobridge.com)
05:53.10irulehii
05:55.00*** part/#asterisk sevard (i=chuck-th@adsl-71-129-115-242.dsl.irvnca.pacbell.net)
06:36.18*** join/#asterisk Snake-Eyes (n=blog@70.55.220.203.static.comindico.com.au)
06:37.32*** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25)
06:42.34*** join/#asterisk tsurko (n=tsurko@77.70.24.142)
06:46.22*** join/#asterisk squall (n=squall@ns2.squallnetwork.net)
06:47.48dejandinicscan
06:51.53*** join/#asterisk Polis_ttt (n=your@194-237-172-225-no48.business.telia.com)
07:10.47*** join/#asterisk ghenry (n=ghenry@212.159.59.85)
07:11.20ghenrywhere's the best place to get res_config_pgsql.c for * 1.2? Running 1.2.18 here
07:13.28*** join/#asterisk CrazyTux (n=CrazyTux@64.95.219.140)
07:17.54*** join/#asterisk IPmonger (n=ipmonger@63-147-199-228.dia.static.qwest.net)
07:18.12*** part/#asterisk IPmonger (n=ipmonger@63-147-199-228.dia.static.qwest.net)
07:19.37*** join/#asterisk clofilo (n=eSe@87.218.150.72)
07:19.49clofilohi
07:19.59clofiloi have a wireless sip phone
07:20.26clofilowhen i recieved a incoming call i can hear but i cant speak
07:20.36clofilowhen i call all works fine
07:21.23clofiloin the asterisk CLI when i recieved incoming call i can only see this:
07:21.32clofiloSIP/2300-00834e30 answered SIP/0000-00820800
07:21.59clofilobut when i call i can see one more line
07:22.17clofiloNative bridging SIP/2300-00820800 and SIP/0000-00834e30
07:22.30clofiloanybody have idea?
07:29.27*** join/#asterisk af_ (n=getsmart@81-174-46-10.f5.ngi.it)
07:31.14*** join/#asterisk gerhard7 (n=gerhard@82-171-117-191.dsl.ip.tiscali.nl)
07:32.15*** join/#asterisk darkskiez (n=mhb@bb-87-81-62-203.ukonline.co.uk)
07:34.01*** join/#asterisk snook3r (n=ariel@bzq-219-46-202.isdn.bezeqint.net)
07:34.44*** join/#asterisk Keltus (n=Keltus@about/cooking/nakedchef/beefstew/Keltus)
07:41.40*** join/#asterisk [hC] (n=hardcore@S01060016b61c8983.vf.shawcable.net)
07:42.20*** join/#asterisk grEvenX (n=even@ti500720a080-4984.bb.online.no)
07:44.38*** join/#asterisk uppal (n=uppal@2001:618:400:888c:218:deff:fe9f:a77f)
07:48.00*** join/#asterisk Dibbler_ (n=Dibbler@host217-45-198-229.in-addr.btopenworld.com)
07:48.47*** join/#asterisk qdk (n=qdk@213.150.62.32)
07:58.20*** join/#asterisk grEvenX (n=even@ti500720a080-4984.bb.online.no)
08:03.51*** join/#asterisk jm|laptop (n=jm@sentry.flags.co.uk)
08:06.37*** join/#asterisk ptblank (n=MURDER1@cpe-76-175-68-176.socal.res.rr.com)
08:10.26*** join/#asterisk crich1999 (n=crich@pd956852e.dip0.t-ipconnect.de)
08:11.31*** part/#asterisk gerhard7 (n=gerhard@82-171-117-191.dsl.ip.tiscali.nl)
08:15.50*** join/#asterisk nfi|ermes (n=ermsewrk@217.220.121.62)
08:20.52UatecWhat do you call a man covered in leaves?
08:23.18sergeeAdam?
08:23.51hadsNeville?
08:31.35UatecRussel
08:32.43hadshehe
08:33.06*** join/#asterisk ptblank (n=MURDER1@cpe-76-175-68-176.socal.res.rr.com)
08:36.11*** join/#asterisk stkn|work (n=stkn|wor@gentoo/developer/pdpc.active.stkn)
08:48.50*** join/#asterisk sbingner (n=sam@pdpc/supporter/sustaining/sbingner)
08:53.22*** join/#asterisk Exhar (n=Roy@213-73-139-87.cable.quicknet.nl)
08:55.15Polis_ttthow do i get timestamp in cli-console
08:55.50hadsNoOp
08:57.21*** join/#asterisk darkmug (n=Dennis@200.103.148.166)
08:59.17ghenryhow can you get * to compile against a different locatin of openssl?
08:59.23ghenryoen in /usr/local/ssl
09:05.42DarKnesS_WolFwhy asterisk website in the downloads ponts to 1.4.00 beta4 !?
09:05.50*** join/#asterisk vics (n=vics@Brylant.iit.pwr.wroc.pl)
09:06.20DarKnesS_WolFand old 1.2.14
09:14.18*** join/#asterisk sevard (i=chuck-th@adsl-71-129-115-242.dsl.irvnca.pacbell.net)
09:20.58*** join/#asterisk penguinFunk (n=penguin@87.224.86.46)
09:25.11*** join/#asterisk psk (n=psk@golia.caltanet.it)
09:26.12*** join/#asterisk pseudor (n=kvirc@161-118-207-82.ip.ukrtel.net)
09:31.40sergeeghenry: ./configure --help | grep -i ssl
09:31.49pseudorsergee: excuse me, I've got an interesting problem calling from H323 to SIP
09:32.11sergeepseudor: and? :)
09:32.18hadsheh
09:32.34ghenrysergee: not on 1.2
09:32.43pseudorsergee: and want to solve that :) with your help
09:32.51ghenryI hacked the Makefiles in / and res
09:32.55ghenrygot it sorted
09:33.00sergeeoh yes that's an issue..
09:33.09ghenryyeah, sucks
09:33.51pseudorghenry: have you ever called from h323 to sip?
09:34.03ghenrynot yet, sorry pseudor
09:34.12ghenryask away though
09:34.58pseudorghenry: do you know who did?
09:35.06ghenryjust ask
09:35.09*** join/#asterisk tutt9876 (n=tut123@cvl92-2-82-228-144-230.fbx.proxad.net)
09:35.11ghenrymany people listening
09:35.55tutt9876hi: got a problem to find the exact syntax to Dial a user connected with a Sip phone in my dialplan
09:36.18pseudorchan_h323.c:977 ooh323_indicate: Don't know how to indicate condition 17 on ooh323c_o_1
09:36.19pseudorThis message is caused by the request of indication of the state 17:
09:36.19pseudorframe.h:
09:36.19pseudorAST_CONTROL_UNHOLD = 17, /*!< Indicate call is left from hold*/
09:36.19pseudorwhile calling from SIP to H323.
09:36.19pseudorWhile calling from H323 to SIP the sound os absent at the H323 side. Instaed I get the message by chan_h323 module:
09:36.22pseudorDon't know how to deal with mode 0x40 (slin).
09:36.24pseudorAnd after this anothe one by the same module:
09:36.26pseudorDon't know how to indicate condition -1 on ooh323c_2
09:37.05pseudorthat is the problem...
09:37.40tutt9876Dial(SIP/${EXTEN}) return "no Such Host"
09:38.04pseudorthe sound is absent at the side of h323
09:38.13tutt9876So the local sipphone is not dialed
09:38.46tutt9876Must a i put a "localhost" somewhere?
09:39.25pseudorthe call is created but there is no sound at the h323. I can hear from the h323 at the sip side but not in the contrary
09:40.08*** part/#asterisk uppal (n=uppal@2001:618:400:888c:218:deff:fe9f:a77f)
09:41.11JTtutt9876: Dial(SIP/sip.confentry/number)
09:41.31pseudorJT: what about my problem?
09:41.49hadsPushy
09:42.42tutt9876JT: what is "sip.confentry" ?
09:43.38JTtutt9876: the name of the relevant entry in sip.conf
09:44.05hadsOr just Dial(SIP/801) if 801 is registered.
09:44.25JTdepends if he needs to dial a number or not
09:44.58tutt9876801 is not registred, and don't really know what to get in sip.conf
09:45.21JTtutt9876: you need to read the book
09:45.22JT~thebook
09:45.32jbotwell, thebook is a book called  Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11
09:46.12tutt9876I have no section in sip.conf
09:46.25JTadd one, would be the logical idea
09:46.56hadsheh
09:47.05tutt9876With what data?
09:47.11tutt9876which
09:47.13pseudorjbot: do you have any idea about my problem?
09:47.17jbotyes, I have any idea about my problem.
09:47.22hadstutt9876: Read   the   book
09:48.00tutt9876ok will make some tries, thanks
09:48.05*** part/#asterisk tutt9876 (n=tut123@cvl92-2-82-228-144-230.fbx.proxad.net)
09:48.24pseudorJT: need help
09:48.40JTpseudor: and?
09:48.58pseudorJT: the problem with SIP and H323
09:49.22JTpseudor: i don't like people constantly soliciting me for help in here when i've never spoken to them before
09:49.30JTi don't have the necessary h.323 experience
09:49.42JTmost people don't use h.323 with asterisk
09:50.12penguinFunkis h.323 better than sip ?
09:50.16penguinFunkwe use sip everywhere
09:50.34JTdebatable, but probably not, especially for an end user/pbx perspective
09:51.25pseudorpenguinFunk: I just want to provide full compatibility, that is why I check calls from H323 to SIP
09:51.41penguinFunkcant say we have any trouble with sip, if it's not broken don't fix it i guess.
09:51.50hadsQuite
09:51.54JTpseudor: asterisk may not be the best choice then
09:52.09JTthere are at least 3 different h.323 channel drivers available for asterisk also
09:52.31pseudorJT: I use OOH323 from addons-1.4.1
09:53.38*** join/#asterisk Iamnacho (i=Iamnacho@ip68-103-153-140.ks.ok.cox.net)
09:54.22*** join/#asterisk clona (n=Atle@82.196.214.14)
09:55.20clonaYo Guys, In extentions conf, you have ${CALLERID(num)} wich is the caller's number.. What is the variable for the Called party ? I cant use ${EXTEN) because it's changed before I get where I need it
09:55.44JTincoming call from what technology?
09:55.47clonasip
09:55.52hadsAssign it to something before it changes
09:56.07clonaI tried doing
09:56.11clonaexten => _X.,1, Answer()
09:56.11clonaexten => _X.,2, set(${ruri} = ${EXTEN})
09:56.11clonaexten => _X.,3, Goto(xMenu,xMenu,1)
09:56.18clonabut.. in xMenu I cant get out $ruri
09:56.24clonaor ${ruri}
09:57.00JTincorrect Set syntax
09:57.06clonaoh
09:57.15clonahow should it look like ? :$
09:57.32JTwell you're setting the variable, not trying to read the resukt if ut
09:57.52hadsshow application set
09:57.57clonaI try in [xMenu]
09:58.56JTclona: also be consistant
09:59.11JTapplications start with capitals normalls
09:59.29clonaHmm okey
09:59.30hadsAlso watch your whitespace
09:59.33clona<- I'm new to asterisk :$
09:59.38JTno spaces between priority and application command
09:59.41JT~thebook
09:59.42jbotthebook is, like, a book called  Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11
10:00.22hadsAnd in the Set command, you
10:00.39hads'll likely get whitespace in your variable or something
10:01.24clonaHmm.. I now swaped it so there is no whitespaces there
10:01.56clonain xMenu (will change it) I have exten => xMenu,1, agi(test.agi,${ruri})
10:02.12clonabut, I'll try somewhat more:D
10:03.07clonaah.. Now I see what I did wrong :$
10:03.11clonaSTUPID me
10:03.28clonaset(ruri=${EXTEN}) :S
10:08.25*** join/#asterisk stimpie (n=michiel@ip565faf27.direct-adsl.nl)
10:09.31*** join/#asterisk |dennis| (n=dennis@200.32.233.82)
10:11.29*** join/#asterisk eltech (i=G00Ds@ool-457c94a3.dyn.optonline.net)
10:12.42*** join/#asterisk skirmisha (n=viki@87-126-55-7.btc-net.bg)
10:12.47skirmishahello guys
10:14.05skirmishaany ideas if type=friend i can send more than 1 call
10:14.47hadsYes
10:14.53*** join/#asterisk infernix (i=nix@unaffiliated/infernix)
10:15.22skirmishathis is default
10:16.36skirmishathen why i get Everyone is busy/congested at this time
10:16.54skirmishaand when i change type=peer all is working fine
10:20.00*** join/#asterisk iZev (n=Zevensof@220-253-16-60.VIC.netspace.net.au)
10:26.33*** join/#asterisk Zeeek (n=Zeeek@pdpc/supporter/active/Zeeek)
10:31.59*** join/#asterisk Zeeek (n=Zeeek@pdpc/supporter/active/Zeeek)
10:40.41*** join/#asterisk nasls_lsa (n=chatzill@athedsl-212159.home.otenet.gr)
10:46.01*** join/#asterisk Cinen (n=Cinen@208.70.20.33)
10:57.02*** part/#asterisk iZev (n=Zevensof@220-253-16-60.VIC.netspace.net.au)
11:09.16*** join/#asterisk tcseke (n=chatzill@22-36.adsl.etel.hu)
11:11.22VecI need to convert a wav (I think a-law) file to a wav49 file using sox, not sure how to ?, must U convert it to gsm and then rename it, that does not sound right ?
11:11.29*** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose)
11:11.38*** join/#asterisk k-man__ (n=jason@ppp244-232.static.internode.on.net)
11:12.43penguinFunkwhat's wrong with a standard wav file ?
11:13.07penguinFunktype: man sox
11:13.33*** join/#asterisk coppice (n=chatzill@94.143.17.210.dyn.pacific.net.hk)
11:14.52VecpenguinFunk : its already a pcm wav file, seems like I must convert it to gsm then rename it to WAV, according to http://lists.digium.com/pipermail/asterisk-users/2007-January/176577.html
11:16.20JTthat sounds wrong
11:16.22*** join/#asterisk Zeeek (n=Zeeek@pdpc/supporter/active/Zeeek)
11:16.25JTit must be a mono wave of 16kHz
11:16.31JTfor optimum performance
11:16.32DrukenHMEwhat are you trying to accomplish?
11:17.06Zeeekladies and gentlemen of earth
11:17.37Zeeekis it possible for asterisk to bow out, i.e., leave a call by connecting two SIP channels?
11:17.40*** join/#asterisk |dennis| (n=dennis@200.32.233.82)
11:17.59JTyou can reinvite media, but it still does signalling
11:18.00DrukenHMEof course....
11:19.02Zeeekthe setup is, I receive the incoming SIP call, I dial a SIP number and send some DTMF. Then, I want to leave the media stream because I am too far from each end
11:19.11ZeeekI don't see how to do this
11:19.24VecI am trying to overite a voicemail greeting which is stored in WAV49 format, I have a wave file which was recorded using the record app ?
11:20.14ZeeekJT I don't control the two endpoints so I don't see how to reinvite?
11:20.38JTZeeek: canreinvite=yes
11:21.47Zeeekadded to sip.conf in the two peer entries?
11:22.00Zeeekwouldn't that happen before the DTMF from dial()
11:25.36*** join/#asterisk DrukenLPY (n=jdumais@74.115.68.16)
11:29.48*** part/#asterisk Zeeek (n=Zeeek@pdpc/supporter/active/Zeeek)
11:33.55*** join/#asterisk DrukenLPY (n=jdumais@74.115.68.16)
11:49.12*** join/#asterisk eltech (i=G00Ds@ool-457c94a3.dyn.optonline.net)
11:50.58*** join/#asterisk tzafrir_laptop (n=tzafrir@62.90.10.53)
11:56.53*** join/#asterisk miltux (n=miltux@ppp146-245.adsl.forthnet.gr)
12:00.01*** join/#asterisk mosty (i=mostyn@60-241-198-194.static.tpgi.com.au)
12:01.58*** part/#asterisk clona (n=Atle@82.196.214.14)
12:03.56*** join/#asterisk DrukenLPY (n=jdumais@74.115.68.16)
12:06.19redaxhi,
12:06.33redaxI'm having serious problem with mISDN + asterisk latest.
12:07.32redaxgetting these lines in asterisk/full:  channel.c: Avoiding deadlock for 'mISDN/1-1'
12:07.50redaxmeanwhile the kernel log floods: mISDN_rdata: rport queue overflow 256/256 [addr:52010201 prim:120282 dinfo:ffffffff]
12:07.59DrukenLPYuhg... finally... anyone want an AP? i'm about ready to throw it out in the street infront of a passing dumptruck
12:08.24redaxno incoming calls at all, restarting asterisk fix the problem
12:09.01*** join/#asterisk zotz (n=zotz@24.244.163.157)
12:09.39redaxsame effect with hfcmulti (HFC4S card) and with 4x HFC-PCI (singleport) card
12:14.34*** join/#asterisk AnThOnYhO (n=AnThOnYh@218.104.248.92)
12:28.00*** join/#asterisk saftsack (n=saftsack@pd9e04468.dip.t-dialin.net)
12:33.12*** join/#asterisk marcan (n=marcanso@160.10.7.121)
12:34.15*** join/#asterisk Pagautas (i=admin@potencial.us)
12:34.31*** join/#asterisk etfonhomey (n=etfonhom@74-140-213-69.dhcp.insightbb.com)
12:36.33*** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25)
12:37.55*** join/#asterisk shido6 (n=shido6@d221-68-200.commercial.cgocable.net)
12:38.08*** join/#asterisk hermuli (n=Eladamri@a88-112-255-26.elisa-laajakaista.fi)
12:41.59*** join/#asterisk DrukenHME (n=jdumais@74.115.68.16)
12:43.02*** join/#asterisk Pagautas (i=bigman@potencial.us)
12:43.25*** join/#asterisk Formater (n=formater@cable-87-116-148-176.dynamic.sbb.co.yu)
12:43.27Formaterhi
12:45.56nasls_lsaI have a beronet BNS04  ISDN card  .. can I use it with zapata.conf or I have to install misdn ?
12:49.46*** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com)
12:53.20Formateri'm using asterisk's Read command to get PIN number...... when i call the number from one trunk, it works.. when i call it from another provider, I see that asterisk recevies the DTMFs but at the end it says, it is empty :( but before that it says Sending dtmf: 57 (9), at 87.116.148.176
12:56.11*** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage)
12:56.37*** join/#asterisk Wvirtual (n=marcos@euro.wvirtual.com.br)
13:05.55*** join/#asterisk naitram_ (n=chatzill@216.77.58.40)
13:06.21*** join/#asterisk Faithful (n=Faithful@ns.linuxterminal.com)
13:08.05naitram_anyone know of a sip command line client for windows ce, mobile 2003.
13:09.08naitram_help
13:09.12*** join/#asterisk AndrewGearhart (n=chatzill@h1.39.213.151.ip.alltel.net)
13:09.48*** part/#asterisk naitram_ (n=chatzill@216.77.58.40)
13:10.14*** join/#asterisk naitram_ (n=chatzill@216.77.58.40)
13:11.16*** part/#asterisk naitram_ (n=chatzill@216.77.58.40)
13:12.03*** join/#asterisk dacter (n=dlittrel@207.200.33.213)
13:13.18*** join/#asterisk mog (n=mog@c-71-207-200-130.hsd1.al.comcast.net)
13:13.18*** mode/#asterisk [+o mog] by ChanServ
13:18.08shido6whats the dtmfmode set to for the other provider?
13:18.25*** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca)
13:19.21*** join/#asterisk pavlicek (n=jpavlice@mail.genevakc.com)
13:23.30LeddyHMYAY!
13:23.30*** join/#asterisk ToyMan (n=Stuart@72.168.167.241)
13:23.43*** part/#asterisk AndrewGearhart (n=chatzill@h1.39.213.151.ip.alltel.net)
13:26.54*** join/#asterisk oej (n=olle@80.251.207.43)
13:41.22*** join/#asterisk Koshatul (n=evangeli@ppp188-33.static.internode.on.net)
13:42.06*** join/#asterisk ctaloi (n=ctaloi@nat-66-218-1-47.usadatanet.com)
13:44.49brad_msswis asterisk 1.4 stabilizing yet?
13:47.07irulegood morning vietnam!
13:47.18mostybrad_mssw, it's still quite buggy
13:47.50brad_msswif I'm on 1.4.2, 1.4.4 should at least be less buggy, right?
13:48.02mostymaybe, maybe not
13:48.04iruledoes that mean that a production system should not be 1.4?
13:48.19mostyirule, for now, i wouldn't use 1.4 in production
13:48.36brad_mssw(seem to get random issues where with 1.4.2, we get calls in where the caller can hear us, but we cannot hear them ... didn't start happening until we went from 1.2 -> 1.4)
13:48.37xhelioxIt really also depends on what you're doing.
13:49.02xhelioxbrad_mssw: it's certainly not going to hurt you to upgrade to 1.4.4 if you're already on 1.4.2.
13:49.19brad_mssw... that said, 1.4 seemed to resolve some other issues with sip for us, especially related to quality
13:49.30brad_mssw(over high-latency extensions)
13:49.42*** join/#asterisk ToyMan (n=Stuart@72.168.167.241)
13:49.51xhelioxI'm using 1.4 on servers that I can stay on top of and where my ass isn't on the line (too much).
13:50.01ctaloihey all - question; when using a virtual IP on an Asterisk server, I am having trouble getting sip user to register to the ViP.  They are able to register with the true IP, just not the virtual.  It seems Asterisk is rejecting the SIP invite.  I've added all the IP's to the domain listing in sip.conf and in the Asterisk console a "sip show domains" shows both the virtual and the physical IP.  Am I missing something?  I have Asterisk bound
13:50.35ctaloinote: 1.4.1
13:51.23*** join/#asterisk champster (n=asterisk@AH.tescogroup.com)
13:52.20*** join/#asterisk etfonhomey (n=etfonhom@74-140-213-69.dhcp.insightbb.com)
13:53.48*** join/#asterisk _Vile (n=vile@bc182112.bendcable.com)
13:54.10*** part/#asterisk _Vile (n=vile@bc182112.bendcable.com)
13:55.12irulexheliox is there really a big difference?
13:55.37*** join/#asterisk ToyMan (n=Stuart@72.168.167.241)
13:56.58*** join/#asterisk Mercestes (n=Merceste@rrcs-71-41-157-70.sw.biz.rr.com)
13:57.01xhelioxhttp://ftp.digium.com/pub/asterisk/ChangeLog-1.4.4 - lots of changes made between 1.4.2 and 1.4.4.  Though I have a bunch it wouldn't fix your issue.. .
13:57.01*** join/#asterisk ToyMan (n=Stuart@72.168.167.241)
13:57.30*** join/#asterisk `pariah (n=josh@unaffiliated/pariah)
13:58.37*** part/#asterisk ghenry (n=ghenry@212.159.59.85)
13:59.14*** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com)
13:59.14*** mode/#asterisk [+o anthm] by ChanServ
13:59.35*** join/#asterisk [Airwolf] (n=airwolf@martijn.lico.nl)
14:04.14*** join/#asterisk axisys (n=axisys@155.70.141.45)
14:04.26*** join/#asterisk yenno (i=dfgdfgdg@84-72-188-41.dclient.hispeed.ch)
14:04.40yennohi, where can i download a ulaw encoded file?
14:04.58shido6u can make one
14:05.01shido6with Asterisk
14:05.37yennook thanks, and do you know what command?
14:05.42shido6show application Record
14:05.49yennothx
14:10.22*** join/#asterisk agile (n=mike@63.98.55.146)
14:12.33*** join/#asterisk walhala (i=cisco@resonix.fr)
14:12.37walhalahi
14:13.20walhalai have a problem with sccp and 7960
14:13.42walhalai received digit on my server but nothing else
14:13.52shido61 digit?
14:13.59walhalayes
14:14.03*** join/#asterisk anonymouz666 (n=anonymou@re366.compuland.com.br)
14:14.05shido6:)
14:14.26shido6are you able to dial all the digits or does the 7960 complain after 1 digit?
14:14.55walhalai can dial more than 1 digit if i dial and hook on
14:15.38walhalado you want a log sample ?
14:16.10walhalahttp://pastebin.ca/469009
14:16.37shido6what is that from?
14:17.05shido6what is collecting digits?
14:17.12*** join/#asterisk Gpl_Source (n=The_natu@unaffiliated/gplsource)
14:17.37walhalai don't know i just get this with skinny debug
14:18.08walhalahere an other example : http://pastebin.ca/469010
14:18.08*** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25)
14:18.13shido6are you forced to stick with skinny? you can put a sip load on the 7960
14:18.31Qwellexcept cisco sip sucks
14:18.35walhalai must stay in sccp yes :(
14:18.50drakohow i can disable the "transfer" behavior from "flash" or pressing the hang up key on the phone?
14:19.12drakoand just hang up the phone call
14:19.18shido6zapata.conf and features.conf
14:19.27walhalain the last example i dialed 132 and just press the dial key on cisco
14:19.52drakoshido6, but i still want transfer with #
14:20.29walhalaif i call my cisco from a SIP phone there is no problem
14:21.07*** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com)
14:21.52*** join/#asterisk ToyMan (n=Stuart@72.168.167.241)
14:22.11walhalanobody see any solution ?
14:23.44shido6whats your dialplan look like?
14:24.59walhalashido6: very simple http://pastebin.ca/469017
14:26.36walhalashido6: nothing difficult isn't it ? :)
14:27.57*** join/#asterisk af_ (n=getsmart@81-174-46-10.f5.ngi.it)
14:33.17*** join/#asterisk [Airwolf] (n=airwolf@martijn.lico.nl)
14:33.49*** part/#asterisk Defraz (n=t0tal@24-116-152-177.cpe.cableone.net)
14:35.01CrazyTuxIf I want an incoming call to ring two devices connected to asterisk at the same time, would that be nesc a que, or just extensions.conf setup to dial twice?
14:36.43*** join/#asterisk brodiem (n=brodiem@67.110.68.66.ptr.us.xo.net)
14:37.57*** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir)
14:38.10penguinFunkCrazyTux: just dial twice like this: exten => s,2,Dial(SIP/101&SIP/102&SIP/103&SIP/104&SIP/105&SIP/106&SIP/107,20,tr)
14:41.39*** join/#asterisk hwt (n=hwt@curb.thorkildssen.com)
14:41.52hwthi, which module provides SIPAddHeader()?
14:41.59hwtor where can i find out which module provides what.
14:42.09*** join/#asterisk VJFROMGT (n=vjfromgt@pool-72-80-126-195.nycmny.east.verizon.net)
14:42.17hwtis SIPAddHeader() at all available in 1.0?
14:42.26VJFROMGTis there a way to rotate my trunks so they all get used evenly?
14:42.30hwtor is there another way to achieve the same in 1.0?
14:43.32Mercestes<PROTECTED>
14:43.58neverblue~thebook
14:44.00jboti guess thebook is a book called  Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11
14:44.35*** join/#asterisk reber (n=reber@cl-158.dub-01.ie.sixxs.net)
14:44.58MercestesUJFROMT:  Did you use a roll over cable?
14:45.05*** join/#asterisk jm|laptop (n=jm@sentry.flags.co.uk)
14:45.50MercestesUJFROMGT:  You have to get the trunk channels started rotating from the SmartJack to yoru PBX and inertia will take care of it from there.
14:46.25neverbluemorning
14:46.57hwtapparently chan_sip provides SIPAddHeader.
14:47.01hwthmphf.
14:47.06hwti guess my * version is too old, then
14:47.25hwtMercestes: i can't. the version is heavily patched.
14:48.03Mercestesmorning, Neverblue.  :)
14:48.10Mercesteshwt:  Then patch it some more.
14:48.11neverbluehow are you Mercestes ?
14:48.35Mercesteshwt:  If it's that heavily patched then you can backport SipAddheader()  :D  get a dev machine and download the latest source and diff it.
14:48.48Mercestesneverblue:  Ok, I guess.  Yourslef?
14:48.59neverblueMercestes, today is a new day :)
14:49.18MercestesVJFROMGT:  Dial(Zap/G1/#######) should accomplish what you wish to do if you setup your groups correctly in zapata.conf.
14:49.27MercestesIt is indeed.
14:50.28hwtMercestes: i might.
14:50.48hwtMercestes: but are you saying there were no way to add/del headers in asterisk <1.2-CVS2004something?
14:51.37*** join/#asterisk ZaVoid (n=zavoid@c-67-165-25-195.hsd1.ct.comcast.net)
14:51.37Mercesteshwt:  Is that when SipAddHeader() was added?
14:51.40ZaVoidmorning all
14:51.45Mercestesmorning, Void.
14:52.15ZaVoidanyone ever see somthing where IAX2 call processing just kinda dies with no messages in any logs. and when i do a iax2 show channels it gives a huge list(that appears to grow)
14:52.19ZaVoidso Mercestes
14:52.24ZaVoider so =sup
14:53.26MercestesZaVoid:  I haven't.  What are you connecting with IAX2?
14:53.33*** join/#asterisk unixlike (n=spid3r@31.67.modemcable.oricom.ca)
14:54.09ZaVoida custom iax2 dialer
14:54.13ZaVoidits worked fine for 10 months
14:54.28ZaVoidthen we built a new asterisk box and replaced(same build, new hardware) and now it seems to be crashing
14:54.29ZaVoidits very strange
14:55.41MercestesWeird.
14:57.49*** join/#asterisk shido6 (n=shido6@d221-68-200.commercial.cgocable.net)
14:58.04unixlikeanyone can help me with a softphone issue ?
15:00.02ZaVoidyeah it doesn't really make any sense ya know?
15:00.03*** join/#asterisk ToyMan (n=Stuart@72.168.167.241)
15:00.26ZaVoidand  i didn't capture any debugs when i confirmed it was all buggered
15:03.05unixlikei am using mitel hardphone and ifedisk softphone both LAN, both same SIP config. mitel phones works great, but i get voice lost with ifedisk only when i dial outside through tdm400p digium cards, not when i dial other softphone in lan... any ideas ?
15:03.28*** join/#asterisk wunderkin (n=kev@dslstat-ppp-95.fastq.com)
15:03.33*** join/#asterisk emiquelito (n=evandro@200-170-201-155.core01.spo.ifx.net.br)
15:03.37*** part/#asterisk emiquelito (n=evandro@200-170-201-155.core01.spo.ifx.net.br)
15:07.07*** join/#asterisk prevail20 (n=prevail2@wan-hts-26.highertech.net)
15:08.19*** join/#asterisk jusMe (n=info@cuscon27948.tstt.net.tt)
15:08.54jusMeGood day to all from the Caribbean!
15:09.31agilebleh
15:09.59Strom_Mcatsex
15:10.01unixlikegood day to u from Quebec, Canada
15:11.09jusMeTHANK YOU!
15:11.25*** join/#asterisk ChkDigit (n=mrw@static24-72-71-175.regina.accesscomm.ca)
15:12.37MercestesGood day to you from texas
15:12.46MercestesThe United State of Texas.
15:13.30jusMeAnyone working with asterisknow?
15:14.01MercestesTry #asterisknow
15:14.44prevail20hello, I am trying to use the playback app in extensions.conf and when I call I get this error:  file.c:512 ast_openstream_full: File /playback/Test.mp3 does not exist in any format
15:14.44prevail20the file does exist and is located in that directory. Here is my playback line in extensions.conf: exten => s,2,Playback(/playback/Test.mp3)
15:14.44prevail20Any advice would be helpful
15:14.46jusMeneed some help with lighttpd
15:14.52jusMeAlready in there!
15:15.06*** join/#asterisk Fieldy (i=8dfErHGS@gentoo/contributor/Fieldy)
15:15.37Mercestesprevail20, output of ls -l of /playback/Test.mp3  please?
15:15.40*** join/#asterisk hansin321 (n=eric@c-71-196-138-170.hsd1.co.comcast.net)
15:16.14*** join/#asterisk explidous (n=explidou@rrcs-24-173-134-222.se.biz.rr.com)
15:16.51*** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290)
15:17.30brodiemprevail20 don't put the file extension on the name. Playback(playback/Test) assuming it was in /var/lib/asterisk/sounds/playback/Test.mp3. I didn't think Playback would support MP3 but I geuss I've never tried it..
15:17.33[TK]D-Fenderprevail20: You cannot specify the file extension in the Playback app...
15:17.47carrarprevail20, put Test.mp3 in /var/lib/asterisk/sounds, then just use Test.mp3?
15:17.48*** join/#asterisk snook3r (n=ariel@bzq-219-46-202.isdn.bezeqint.net)
15:18.03carrarah
15:18.05[TK]D-Fendercarrar: nope ;)  see above
15:18.10explidousHi
15:19.35wunderkinyou would need format_mp3
15:19.42wunderkinor use mp3playback
15:21.34prevail20Mercestes: -rwxrwxrwx  1 root root 6603772 May  3 09:55 Test.mp3
15:22.05Mercestesprevail20, I think [TK]D-fender got it.  remove the .mp3
15:22.14*** join/#asterisk |dennis| (n=dennis@200.32.233.82)
15:22.40jusMetrying to use php in lighttpd for asterisk now, already intalled PHP but can't get it to work.
15:23.00[TK]D-Fenderwunderkin: I'll let him take all the rope he wants..... 1 yard at a time ;)
15:23.20prevail20[TK]D-Fender: I removed the .mp3 from the extensions.conf file and reload extensions in asterisk but I still get the same error only now it says /playback/Test does not exist in any format
15:23.41[TK]D-FenderjusMe: Not an * question for sure, and this isn't a GUI or lighttpd support channel.
15:24.05[TK]D-Fenderprevail20: Ok, now you need to have format_mp3 installed, which is part of the asterisk-addons package
15:24.16jusMeI understand, just trying to do something with asterisknow, and not gething help in #asterisknow!!! no problem, thank you!
15:24.48prevail20[TK]D-Fender: Addons should be already installed. Where is the format_mp3 located?
15:24.49jusMeif anyone can direct me in the right direction! I apreciate it!
15:25.49[TK]D-Fenderprevail20: What do you mean "should"?
15:26.30prevail20We installed the Asterisk business edition and included the Asterisk-addons
15:27.25[TK]D-Fenderprevail20: look at "show modules like format" and see if its there
15:28.29prevail20[TK]D-Fender: That must be the problem, it is not. I only have format_tiff and format_wav
15:28.35*** join/#asterisk NOT_guru (n=NOT_wiza@24-241-103-142.static.stls.mo.charter.com)
15:29.18prevail20[TK]D-Fender: I will install the addons and verify that the format_mp3 is installed, hopefully that will solve the problem
15:29.29prevail20[TK]D-Fender: and Marcestes: Thank you
15:29.34prevail20Mercestes
15:29.57*** join/#asterisk saftsack (n=saftsack@pD9E04468.dip.t-dialin.net)
15:29.58MercestesI've answered to worse.  But your welcome.
15:30.23*** join/#asterisk queuetue (n=scott@70.54.254.134)
15:31.09*** join/#asterisk AnThOnYhO (i=AnThOnYh@218.104.248.250)
15:31.39ZaVoidanyone else play with the new digium tcb400 cards yet? mine don't work and digium cant' figure out why yet :(
15:31.39kuku5Has anyone lately compile rxfax or valetparking ?
15:31.53kuku5ZaVoid: the 8 por t?
15:32.10NOT_guruI am in a similar situation I think zavoid
15:32.16NOT_guruI have the tdm804p
15:32.18*** join/#asterisk generalhan (n=asd@ip67-90-64-2.z64-90-67.customer.algx.net)
15:32.21NOT_gurudoes odd stuff
15:32.27generalhanhey all
15:32.39NOT_gurukuku5  I have the 8 port
15:32.49queuetueI'm trying to connect a vonage d-link adapter into a tdm port (to do outgoing over vonage using zap).  When I connect the phone cord, the vonage adapter flashes the phone indicator - as though it's "phone" was offhook - and zap does not work over it.  Can anyone explain this?
15:33.05dlynes_laptop[TK]D-Fender: I found out the problem with the BLF light staying on, seems to be an asterisk problem
15:33.18Qwell[]queuetue: what type of port are you connecting it to?
15:33.22Qwell[]It would need to be an fxo
15:33.29*** join/#asterisk WeBRainstorm (n=ask@81-174-12-48.f5.ngi.it)
15:33.35generalhanso im still having this ridiculous error that i was posting yesterday:: http://generalhan.pastebin.ca/468038  and im just wondering if i could get all the users off the phone so i can reboot the machine if things could actually get worse !?
15:33.44dlynes_laptop[TK]D-Fender: The blf light stays on, the show hints says the phone's in use, but show channels doesn't show it in use
15:33.55*** join/#asterisk nasls_lsa (n=chatzill@athedsl-212159.home.otenet.gr)
15:33.59queuetueQwell:  It's a tdm port with fxs kewlstart signalling - the same I use for my POTS lines.
15:34.02*** join/#asterisk Putzz (n=me@CPE00155824933c-CM00111ae07ac2.cpe.net.cable.rogers.com)
15:34.24queuetue(In fact, the same port works as my POTS connection)
15:34.24dlynes_laptop[TK]D-Fender: I'm going to file a bug report on it
15:36.08WeBRainstormhi guys, I'm having a problem with a queue and ast 1.2.18: some members are not being called
15:36.21WeBRainstormit only calls a subset of the queue members
15:36.24kuku5I'm trying to compile rxfax for almost a week now. Tried different spandsp versions, tiff versions, asterisk version. This is the least amount of errors that I got so far:   http://pastebin.ca/469115     Thank you.
15:36.41WeBRainstormeven if the corresponding channels seems to be available
15:36.56queuetueQwell: Any clue why the tdm would appear any different than an extension phone to the vonage adapter?
15:37.06*** join/#asterisk rene- (n=rene@200.34.66.137)
15:37.27MercestesWeBRainstorm, Are you using the priorities?
15:37.39WeBRainstormMercestes, no, they are at 0
15:38.20MercestesWeBRainstorm, Always the same subset?
15:38.46WeBRainstormMercestes: no, it changes from call to call, I'm preparing a pastebin with some output
15:40.24*** join/#asterisk phobus (n=phobus@crlspr-69.65.75.232.myacc.net)
15:40.44generalhanMercestes: do you remember my "Unable to find our posistion" error that you were helping me with yesterday ?
15:41.00Qwell[]Mercestes: sign up for trial
15:42.24kuku5Qwell: Can you take a look at the errors I'm getting when compiling rxfax ?
15:42.28Qwell[]no
15:42.57MercestesQwell[]  A trial for what?
15:43.01Qwell[]WoW :p
15:43.06MercestesOh!
15:43.07Mercesteslol
15:43.11Qwell[]I sent you one, heh
15:43.17Mercestesgeneralhan, yes.
15:43.29MercestesI saw, thanks.
15:43.39MercestesI'll do it this weekend
15:44.04queuetueA TDM port with fxs signalling is an fxo port, isn't it?
15:44.16Qwell[]queuetue: is it a digium card?
15:44.20NOT_guruqueuetue: as I understand it yes
15:44.22queuetueQwell: Yes.
15:44.27Qwell[]what color is the module?
15:44.53queuetueQwell: I think red, will have to pop open the server to check.
15:44.53generalhanMercestes: could restarting the server make this issue any worse ? cause things are getting real bad now, after a call is answered it takes a good 10 -15 seconds for it to actually connect the callers. And i cant afford for this things to just stop working completely
15:44.53Qwell[]red is fxo
15:45.25*** join/#asterisk ToyMan (n=Stuart@72.168.167.241)
15:45.27NOT_guru<PROTECTED>
15:45.39generalhanso i would rather have everyone be unhappy with the phones while i work on getting a replacement, rather than shutting it down and not being able to get it back up.
15:46.03Qwell[]generalhan: having problems getting it back up?
15:46.44MercestesQwell[]  he hasn't tried yet, he's afraid he wont' be able to get it back up
15:46.51*** join/#asterisk ToyMan (n=Stuart@72.168.167.241)
15:47.05*** join/#asterisk santibiotico (n=santi@ip23498.bcn.altecom.net)
15:47.06santibioticohi
15:47.07queuetueYes, red daughter card.
15:47.13generalhanQwell[]: yeah, what Mercestes said !
15:47.34santibioticoi'd like to have n-way calling feature under asterisk 1.2.14
15:47.37queuetueSo, if the port is an FXO, then why does the vonage adapter see it as "off hook"?
15:47.41NOT_guruqueuetue: yes  red daughter card for a digium is an FXO card
15:47.46generalhanand i went home and looked all night for any documentation on this kind of error, but found nothing
15:47.47santibioticoright now i am using the meetme feature
15:47.56Mercestesgeneralhan, I'd say it'd be relatively safe.
15:47.57ZaVoidnot guru your using the card for g.723 transcodings?
15:48.00queuetueNOT_guru: Thanks - I thought it was, but good to be reassured. :)
15:48.01NOT_guruqueuetue that I can't answer sorry
15:48.11Mercestesgeneralhan, but, jbot thinks I'm a nub.
15:48.17NOT_guruqueuetue: I help where I can
15:48.24generalhan~Mercestes
15:48.35jbotsomebody said mercestes was almost a total nub
15:48.35MercestesI think more importantly you need to figure out why that error is happening.
15:48.37MercestesIt could be a Hdd failure.
15:49.00generalhanMercestes: hmmm
15:49.01ZaVoidNOT_guru:  the tcb400b your using?
15:49.02santibioticoi've seen a howto for having n-way calling, but it uses channelredirect
15:49.12santibioticowhich is not an available app under 1.2.14
15:49.15santibioticoany help?
15:49.37generalhanMercestes: im really torn because i told my boss (the owner) that i was seeing issues with the asterisk server 3 weeks ago. and i told him we need another server so that i can have redundant phone servers, and he didnt listen.
15:49.59generalhanSo i dont know if i should care ! lol. other than having 50 pissed off employees at my door.
15:50.19queuetueDoes anyone know how to get the IP address from a Dlink VTA-CV?
15:50.29Mercestesgeneralhan, It may be time for an "I told you so" speech then.
15:50.45NOT_gurusorry  phone brb
15:51.16[TK]D-FenderMercestes: Yeah.... VERY smart...... thats an ECM (Ending Career Move)
15:52.01Mercestes[TK]D-Fender, Otherwise known as a "fire hazard"
15:52.28Mercestesgeneralhan, I'd check your Hdds.
15:52.44Mercestesgeneralhan, And isn't the choppy audio in your IVRs?
15:53.44*** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl)
15:54.16WeBRainstormMercestes: http://www.pastebin.ca/469137
15:54.52WeBRainstormMercestes: here is queues.conf - sip show peers - show queues - and a log of an incoming call
15:56.27generalhanMercestes: no its only in the voicemail system
15:56.28WeBRainstormas you can see only some queue members are called
15:56.41MercestesAye.  Why are some of these peers offline?
15:56.43generalhanmy recorded wavs that i use for the IVR, they all play perfectly
15:58.25Mercestesgeneralhan, I ran into a similiar problem with another customer.  On hold music sounded like *crap*
15:58.25lesouvageI'm working on a server and suddenly the sound quality is like ztdummy isn't properly working any more. Is there a way to check if zaptel and ztdummy is doing what it is supposed to do?
15:58.30Mercestesmp3s.  Sounded like it was playing uberfast but it was just skipping audio lik emad and playing every 5th frame or something.
15:58.49Mercestesgeneralhan, only in voicemail.  Gah.  Nice.
15:59.09MercestesWeBRainstorm, Are these all the same phones or are they different types?
15:59.10*** join/#asterisk FluxIRCd (i=Prif@cpe-069-132-040-093.carolina.res.rr.com)
15:59.24generalhanits funny though, for her to finish saying you have 10 new messages and 1 old message, takes about 45 seconds ! lol
15:59.42FluxIRCdso is there a windows client to connect to the server or?
15:59.46WeBRainstormMercestes: Xlite
16:00.20WeBRainstormMercestes: a tethereal shows that no net trafic is sent to peer 327 (as example)
16:00.32MercestesWeBRainstorm, My problem is that you have random ping times on all these phones, and some of them show an "unknown" status and others show an actual status, and some of them are randomly offline.
16:00.59Mercestess/problem/concern/
16:01.04Mercestes~botsnack
16:01.05jbotMercestes: :)
16:01.28WeBRainstorm:)
16:01.29MercestesWeBRainstorm, 327 shows "unknown" as a status, but it *has* taken calls.
16:02.08WeBRainstormMercestes: some of them are not logged, infact sip show peers says that, and that's ok
16:02.43MercestesYea, but I just checked and every phone that showed an actual status was notified.
16:02.52WeBRainstormMercestes: but 327 has a "OK (49 ms)" and in the queue is marked as unknown
16:03.08WeBRainstormMercestes: that's incoherent, isnt'it ??
16:03.15Mercestesnot at all.
16:03.21Mercestesunknown means it's state is unknown
16:03.25*** join/#asterisk [Airwolf] (n=airwolf@martijn.lico.nl)
16:03.32MercestesAsterisk doesn't know if it's inuse/available/offline, etc.
16:03.57Mercestesonly two unknown phones were noified in yoru queue, so I think that's your symptom.
16:04.06MercestesOr one of yoru symptoms.
16:04.13Mercestesare all the xlite phones the same version?
16:04.18FluxIRCdpoing
16:04.23FluxIRCds/poing/test
16:04.31FluxIRCdblah
16:04.54*** join/#asterisk rmayorga (i=rmyorg@unaffiliated/rmayorga)
16:05.02MercestesACK!
16:05.19WeBRainstormMercestes: you're right... it should try unknown members
16:05.27WeBRainstormMercestes: yes, xlite same version
16:05.36*** join/#asterisk Qwell[] (i=qwell@pdpc/sponsor/digium/Qwell)
16:05.36*** mode/#asterisk [+o Qwell[]] by ChanServ
16:05.44Mercestesit either should or should not, it bugs me that it only tries them sometimes/half the time.
16:06.08WeBRainstormMercestes: do you think that a queue with SIP and IAX members mixed could cause probles ?
16:06.25WeBRainstormMercestes: it should me deterministic... but it isn't.. :(
16:06.44MercestesNah, should be ok to mix sip/iax2
16:06.55MercestesAre ppl using DND in the queues?
16:07.07WeBRainstormyou mean on the xlite ???
16:07.10Mercestesyea
16:07.36MercestesDND + queues = random bad things.
16:07.47WeBRainstormthey say they are not... but shouldn't DND justt causing a BUSY to be sent back to asterisk ??
16:08.08MercestesWell, sorta.
16:08.39MercestesAsterisk checks the phone to see if it's able to sen da call to it and the phone answers that it can recieve a call.  So asterisk sends the call, then the phone goes "oops, sorry, DND."  and asterisk is all "WTF?" because asterisk already checked if tha tphone was in use
16:09.25MercestesIt's a little different on a call because asterisk tries the call first, then error handles, in a queue I think it error handles first, and then tries the call and gets pissy if there are errors after that.
16:09.34Mercestesbut that's a *guess* at this point.
16:09.55MercestesIf they say they're not using DND then I wouldn't blame DND at this point.
16:09.57*** join/#asterisk zm23 (n=chatzill@dyn-160-39-251-228.dyn.columbia.edu)
16:10.49NOT_guruZaVoid:  I am back now   sorry  my sister called me and went on a whine rant
16:11.03zm23hello all.  A quick question about asterisk voicemail app.  How can i delete my temp greeting using the phone access to voicemail
16:11.03ZaVoidno worries
16:11.13MercestesNOT_guru, is she hot?
16:11.13NOT_guruZaVoid : I am using a Wildcard TDM800P
16:11.14[TK]D-FenderNOT_guru: Its clearly genetic ;)
16:11.20ZaVoidahh different
16:11.23NOT_gurulol  she's 41 now
16:11.26ZaVoidthats not for g723 transcoding right?
16:11.29WeBRainstormMercestes: no DND and then i don't see network trafic to that host...
16:11.44NOT_gurunow now fender
16:11.53Mercestes...so asterisk is clearly refusing to send traffic to that device.  hrm.
16:11.57Mercestesatleast sometimes.
16:12.02NOT_guruand yes  my buddies all still wanna "be with" her
16:12.12MercestesI'm not interested in a 41 year old.
16:12.25NOT_guruoh and Fender  were you the one helping me yesterday with this card?
16:12.47NOT_guruand asked why I configured all 8 ports when its only got 1 daughter card in it
16:12.56NOT_guruwell  genzaptelconf did that
16:12.59WeBRainstormMercestes: yes, it's not considering it, let's get a look at full log
16:13.03MercestesWeBRainstorm, Well I would pose it to the smart guys and see if they have an answer and then bug report it.
16:13.56[TK]D-Fenderzm23: its in the menu in VoicemailMain
16:13.57MercestesI can't really say whether asterisk should or should not send traffic to an unknown device but I think it should pick one.  Could be bad network connections but, I'm giving the network the benefit of the doubt.
16:14.12[TK]D-FenderNOT_guru: Yes, I was
16:14.20WeBRainstormMercestes: here i am... May  3 18:52:21 ERROR[27150] chan_sip.c: Unable to build sip pvt data for '327' (Out of memory)
16:14.33oejThat's unusual
16:14.40oejsmall or large system?
16:14.57NOT_guruDFender: well  I am building a paste for when tzafrir gets back to me
16:15.05NOT_guruI will link it in a minute
16:15.43Mercestesew
16:15.46Mercestesgoogld that.
16:15.50Mercestess/googld/google/
16:16.31*** join/#asterisk kikoafonso (n=rafonso@cronopio.rits.org.br)
16:16.45*** join/#asterisk Zeeek (n=randulo@pdpc/supporter/active/Zeeek)
16:17.08WeBRainstormMercestes: and I also have a 'Too many open files'
16:17.15Mercestesoh!
16:17.17NOT_guruFender  here is the aste I have of information for tzafrir   http://generalhan.pastebin.ca/469164
16:17.19MercestesThere is a fix for that.
16:17.25Mercestesgoogle asterisk too many open files
16:17.37Mercestesor linux too many open file
16:18.19[TK]D-FenderNOT_guru: fix your channels in zapata.conf too...
16:18.29NOT_guruwell look now
16:18.34NOT_guruerr will look now
16:18.36NOT_guruthank you
16:18.58WvirtualHi, i have problems when i try to start my asterisk, after install and register the g729 codec. I´m using FreeBSD 6.2 Stable
16:19.00[TK]D-FenderNOT_guru: Its probably still trying to call channels that are no longer defined at all
16:19.01Wvirtuallog: WARNING[82799] loader.c: /usr/local/lib/asterisk/modules/codec_g729a.so: Undefined symbol "__errno_location"
16:19.40[TK]D-FenderGTG, back in a few
16:23.04WeBRainstormMercestes: I don't think it's a problem of increasing ulimt... the server has 3 active calls but it has 900 UDP open ports...
16:23.04queuetueHas anyone else ever had a problem trying to connect an FXO port to an ATA adapter?  This should work, shouldn't it?
16:23.04WeBRainstorm# lsof |grep asterisk |grep UDP |wc -l says 968
16:23.20WeBRainstormMercestes: asterisk   5654 asterisk 1007u     IPv4      91811                 UDP *:19311
16:24.16explidousqueuetue:  you always have to connect FXO to FXS
16:25.04queuetueexplidous: yes, i meant the FXS port of an ATA adapter.
16:25.10*** join/#asterisk CunningPike (n=CunningP@dhcp-10-234.district.north-van.bc.ca)
16:25.13explidousqueuetue:  However there might be electrical problems depending on the manufacturer
16:25.48queuetueexplidous: The FXO is a TDM card from digium, the FSX is a d-link VTA-CV from vonage...
16:25.49*** join/#asterisk crochat (n=crochat@84-74-150-141.dclient.hispeed.ch)
16:25.53crochatHello
16:26.01*** join/#asterisk qdk (n=qdk@213.150.62.33)
16:26.09crochatIn indications.conf, what is stutter used for ?
16:26.15explidousqueuetue:  I think I get the idea ;-)
16:26.20Mercestesgtg.  BBL, Sorry WeBRainstorm.
16:26.28WeBRainstormok... here I am
16:26.29WeBRainstormhttp://bugs.digium.com/view.php?id=9235
16:26.34WeBRainstormfixed today !!!
16:26.45crochatI mean, in which case you will hear the stutter tone ?
16:26.53explidousqueuetue:  should work, do you have problems?
16:27.57queuetueexplidous: yes, the vonage adapter flashes as though it is off-hook, and asterisk can not call out on it.  (console reports connection, but then a click, and then dead air forever.)
16:28.07queuetueNo call is actually made.
16:28.52explidousThere are a number of parameters for call signaling, however I think vonage does not let you change them...
16:29.21*** join/#asterisk aptura (n=jondoe@S010600a0c93f6f7e.vs.shawcable.net)
16:29.41Zeeektomorrow at this time
16:29.48explidousI wonder if one could sniff out the sip connection of their softphone since that is Eyebeam...
16:30.00*** join/#asterisk menfin (n=cray@AMontpellier-152-1-57-98.w83-197.abo.wanadoo.fr)
16:32.23*** join/#asterisk jtexter3 (n=jtexter3@COX-66-210-197-34-static.coxinet.net)
16:34.36*** part/#asterisk Wvirtual (n=marcos@euro.wvirtual.com.br)
16:36.05*** join/#asterisk DrCool (i=DrCool@drcool.hungamacable.com)
16:37.10*** join/#asterisk CrazyTux (n=CrazyTux@64.95.219.140)
16:37.59WeBRainstormMercestes: tnx for help
16:38.04*** join/#asterisk neverblue (n=profx@unaffiliated/neverblue)
16:39.04neverbluewhat do I need to install to play mp3 files ?
16:40.41*** join/#asterisk qdk_ (n=qdk@213.150.62.32)
16:40.51Putzzwhat do I need to make a sandwich?
16:40.53Putzzheh
16:43.05Qwell[]bread...  all others are optional
16:43.58anonymouz666tzanger: do you use chan_cellphone without problems?
16:44.51*** join/#asterisk dzlabing (n=dzlabing@wan-gw.wien.zlabinger.at)
16:51.54*** join/#asterisk pif (n=ldm@zenon.apartia.fr)
16:53.54pifhi, has the '.' atom changed in 1.4.x 's dialplan ?
16:54.14pifis it still an optional character?
16:54.56*** part/#asterisk Zeeek (n=randulo@pdpc/supporter/active/Zeeek)
16:57.10FuriousGeorgewhats the difference between a wanpipe and a crackpipe?
16:57.46FuriousGeorgeseriously, im trying to figure out what this wanpipe that sangoma keeps advertising is
16:58.03Corydon-wFuriousGeorge: you look less retarded when you're smoking crack
16:59.06FuriousGeorgeThis package installs configuration tools and firmware modules for the Sangoma S508 and S514 router cards. You may use this software to build a stable and flexible WAN router for frame-relay, PPP, or Cisco HDLC leased-line links based on these cards.
16:59.25FuriousGeorgeCorydon-w: i'd hit the wanpipe anyway
16:59.57FuriousGeorgei dont understand what a sangoma a200 with a few fxs has to do with my wan
17:00.07errrwhen I have iax2 debug turned on I see that when an incoming call has: CALLING NUMBER: 231   when I call it from my extension. How do I get this 231 value??
17:00.25*** part/#asterisk pavlicek (n=jpavlice@mail.genevakc.com)
17:01.38*** join/#asterisk qdk_ (n=qdk@213.150.62.32)
17:01.57errrbasicly I have another server running only voicemail and from the main pbx server you dial 4000 to trunk you to the VM server via iax2 and it goes straight into the main voicemail, but I would like for you to dial 4000 and take you straight to your box so all you enter is your vm password
17:05.53lesouvageshould ztdummy be loaded before or after zaptel?
17:07.42NOT_gurulesouvage as I understand it... ztdummy should be loaded after your physical cards
17:08.09NOT_guruI have 1 card  and its span1 and ztdummy is span 2..   but thats just my understanding.. I could be wrong
17:08.48*** join/#asterisk saftsack (n=saftsack@pD9E04468.dip.t-dialin.net)
17:09.32drakohow i can disable the "transfer" behavior from "flash" or pressing the hang up key on the phone?
17:10.18lesouvageNOT_guru: I don't have cards at all, I need it for the timing.
17:11.39NOT_gurulesouvage: lemme look at one of my systems for ref  one moment
17:12.43NOT_gurulesouvage I assume we are looking at /etc/zaptel.conf
17:13.51queuetueSince interfacing with the vonage adapter is off the table, can anyone recommend a decent internet-refillable online phone card merchant? :)
17:14.20DrCoolqueuetue: binfone.com
17:15.46queuetueDrCool: They are a VOIP carrier, aren't they?
17:17.42DrCoolyes.. you want to use pstn dialing?
17:17.42WeBRainstormFuriousGeorge: wanpipe is the unified sangoma driver
17:17.42queuetueDrCool: I think we're done trying to actually make VOIP work... every carrier has just been unusable call quality to date - at signup, or soon after.
17:17.43*** join/#asterisk fakhir (i=fakhir@unaffiliated/fakhir)
17:17.43queuetueI think we've got to go pstn from here on out.
17:17.43DrCoolqueuetue: :) I've been using binfone for the past 3 years and I've hardly (if ever) had probs with quality with them
17:18.03*** join/#asterisk shinao1 (n=shinao1@80.89.187.214)
17:18.07*** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca)
17:19.31dlynes_laptop[TK]D-Fender: btw...found out what the issue was with the blf, thanks to oej
17:19.42dlynes_laptop[TK]D-Fender: type=friend is incompatible with blf
17:20.15[TK]D-Fenderdlynes_laptop: Apprently peer works BOTH ways now...
17:20.34dlynes_laptop[TK]D-Fender: yeah, that's what oej was telling me
17:24.41lesouvageNOT_guru : Yes that was my guess also.
17:24.45*** join/#asterisk sysreq (n=sysreq@modemcable171.134-81-70.mc.videotron.ca)
17:26.20dlynes_laptop[TK]D-Fender: that was the whole reason I was using friend, because peer/user was very confusing
17:26.41*** join/#asterisk _Turbo_ (n=Turbo@firewall.turbolink.net)
17:26.42lesouvageNOT_guru: the only line uncommented is defaultzone=nl
17:27.04_Turbo_hey all
17:27.50_Turbo_does anybody have or know of any example code to join 2 inbound call legs together?
17:28.28_Turbo_there will only ever be at most 2, so conferencing will have too much overhead, I'd like to keep rtp unmolested if possible
17:30.02*** part/#asterisk jpablo (n=jpablo@linuxuanl.org)
17:30.20*** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-179-75-202.static.bezeqint.net)
17:32.17NOT_guruI am sorry  I was away hacking abuot with thsi tdm804p
17:32.49*** join/#asterisk shido6 (n=shido6@d221-68-200.commercial.cgocable.net)
17:38.58*** join/#asterisk clona (n=Atle@82.196.214.14)
17:39.12NOT_gurutzafrir:  you have a bit of time today or still busy?
17:39.25clonaHey, Does anybody know if app_mp4 has it's own "channel" ?
17:39.55Qwell[]why would mp4 be an app?
17:40.19clonaQwell[]: it saves down/plays h26x streams to mp4
17:40.26clonato/from :-p
17:40.34Qwell[]yeah, why would it be an app?
17:40.40clonaI dunno
17:40.43clonasome guys made it as a app
17:40.59clonahttp://sip.fontventa.com/contant/view/15/44
17:41.05clonathere you see :)
17:42.48*** join/#asterisk qdk_ (n=qdk@0x50c627be.bynxx11.adsl-dhcp.tele.dk)
17:48.49iruleI press 9111 and _91XX dials 111, is 9 treated in a special way? I would like to dial 8111 to have Dial dial 111
17:49.23VJFROMGTanyone here use polycom phones?
17:49.31MercestesVJFROMGT, yes
17:49.36*** join/#asterisk zm23 (n=chatzill@zaara.cuit.columbia.edu)
17:49.38Mercestesirule, Um, only if you program it that way
17:49.50VJFROMGTis there a way to configure the phone to use 2 different sip providers?
17:49.56VJFROMGT(multi line_)
17:50.59redaxgood evening,
17:51.09VJFROMGTyou still there mercester?
17:51.32*** join/#asterisk kFuQ (n=somedude@c-67-185-123-34.hsd1.wa.comcast.net)
17:52.00redaxif I dont have any ResetCDR / NoCDR in my dialplan, where, and when CDR record is dumped to csv/mysql or whatver ?
17:52.14redaxat hangup?
17:52.55MercestesVJFROMGT, yes.
17:53.24Mercestesredax, yes.
17:53.59VJFROMGThow do i configure
17:55.31codefreezeredax: usually;
17:55.49[TK]D-Fenderirule>I press 9111 and _91XX dials 111, is 9 treated in a special way? I would like to dial 8111 to have Dial dial 111
17:56.09[TK]D-Fenderirule: 9111 dial what you told it to.
17:56.28[TK]D-Fenderirule: 9111 could jsut as easily done nothing productive at all.
17:56.38hacimcan anyone point me to places to get DiDs for Sao Paolo?
17:57.20[TK]D-FenderVJFROMGT: Pretty mcuh every polycom supports 2 or more registrations.  Go read the admin guide to set it up.
17:57.51dacterhacim: doesn't the telco provide the dids?
17:58.36*** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir)
17:58.40redaxaha, so if I want to do anything with the actual CDR, I must put my stuff to the actual context's extension 'h'
17:58.45hacimdacter: well, I guess I need a voip gateway to the telco, because all I have right now is an asterisk server in a colo
17:59.26redaxis that correct if I do a `last chance' context with only extension `h' and I include this extension all over the contextes?
18:00.03redaxif the context has ext `h' then the the context's `h' executed, if it missing, then the included ext => h,...
18:00.08redaxis that correct?
18:00.40[TK]D-Fenderredax: Go read up on extensions sort/include priority on the WIKI
18:00.41irule[TK]D-Fender what I am interested in understanding is the treatment of 9 because a regular extesnion is 333 for local sip phones and 333 is dialed, not 33
18:01.30redax[TK]D-Fender: reading, just confused my mind...
18:01.32[TK]D-Fenderirule: "9" is not treated any different.  what you dial, and what it does have NOTHING to do with each other.
18:01.42redaxsorry for the dump Q.
18:02.07[TK]D-Fenderirule: exten => 88,1,Dial(SIP/5000)
18:02.23[TK]D-Fenderirule: See?  What does 88 have to do with 5000?  Nothing.
18:03.33[TK]D-Fenderirule: You should be looking at the actual CONTENT of what that extension is doing to see how the pattern matters.  it doesn't even HAVE to matter.
18:03.48*** join/#asterisk |dennis| (n=dennis@200.32.233.82)
18:04.18[TK]D-Fenderirule: exten => _91XX,1,Dial(SIP/5000) <- Again, 100 possible ways to ring a SINGLE SIP device.
18:08.04Mercestesboth 5000 and 88 are divisible by 2, 4, and 8 evenly.
18:08.18MercestesSee, it all makes perfect sense
18:09.08[TK]D-FenderMercestes: And we all just silently pray that you never multiply ;)
18:11.00Mercestes[TK]D-Fender, nah, only divide.
18:11.32[TK]D-FenderMercestes: My blades can divide VERY well ;)
18:11.54MercestesI heard you divide very well.
18:12.05[TK]D-FenderI should take a pic or two of the whole new mounting I've done...
18:12.26[TK]D-Fender</contextabuse> :D
18:12.47MercestesI don't want pictures of your mountings.
18:14.02MercestesI am sure they are masterful though.  :D
18:15.00iruleis it possible to create a group of trunks?
18:18.30Mercestesirule, yes.
18:18.54Mercestesirule, with group-
18:18.57Mercestesirule, with group= even.
18:19.32[TK]D-Fenderirule: Your term is dangerously vague.  "trunks" is not a word to use around here.
18:19.48[TK]D-Fenderirule: What kind of channels (and mix of them) are you referring to?
18:20.25iruleheh ok sorry [TK]D-Fender I ment I have a couple zap's, ie 2 x x100p
18:21.00irulewhat should I search for in voip-info?
18:21.15[TK]D-Fenderirule: "group=[number]" for those channels
18:21.40[TK]D-Fenderirule: and Zap/g[number]/somethinghere
18:22.49*** join/#asterisk lokkju_wrk (n=lokkju@unaffiliated/lokkju)
18:23.27PioneerVMI'm trying to run an AGI script that uses "say_alpha", but it just passed by it like it was never called -- say_number works but say_alpha gives a console error about "iax_read: i should never be called!"
18:23.48irulethanks
18:24.19PioneerVMactually now i dont get that error at all, but the system just passed it and never says anything
18:24.25PioneerVMwhy would say_number work but not say_alpha?
18:24.37PioneerVMscore($AGI->say_alpha('John Smith',1));
18:24.37PioneerVMscore($AGI->say_number('1'));
18:25.15*** join/#asterisk Polis_ttt (n=your@194-237-172-225-no48.business.telia.com)
18:27.06*** join/#asterisk l3jj (n=chatzill@dsl093-001-038.det1.dsl.speakeasy.net)
18:28.10*** join/#asterisk nasls_lsa (n=chatzill@athedsl-212159.home.otenet.gr)
18:28.22PioneerVManyone have any thoughts?
18:28.31l3jjHi everyone, I am looking for a recomendation on a voip provider, I need DID's at a low rate, (like .99 a month, and then a per min charge)  I want a provider, that will take a credit card number, and has a clue to what they are doing
18:28.40l3jjAm I dreaming?
18:30.02*** join/#asterisk crich1999 (n=crich@port-212-202-210-130.dynamic.qsc.de)
18:30.27*** join/#asterisk ToyMan (n=Stuart@12.23.30.130)
18:30.48iruleyou are referring to group= for my zap channels within zaptel.conf? this is not very informative http://www.voip-info.org/wiki/view/Channels+and+Groups
18:32.43*** join/#asterisk DocHolliday (i=RgRabbit@gateway/gpg-tor/key-0x0E4F6D6C)
18:33.12[TK]D-Fenderirule: On the contrary, that explains it PERFECTLY
18:33.37*** join/#asterisk beeew (n=chatzill@66-240-27-5.isp.comcastbusiness.net)
18:34.41*** join/#asterisk bkw_ (i=brian@adsl-70-143-36-78.dsl.tul2ok.sbcglobal.net)
18:34.56irulewhich file pleas?
18:35.19*** join/#asterisk saftsack (n=saftsack@pD9E04468.dip.t-dialin.net)
18:37.09[TK]D-Fenderirule: What do you mean which file?  You define the group in zapata.conf, and you USE it when choosing what channel to dial in extensions.conf
18:37.13*** join/#asterisk iEatBabies (n=DudeMan@204.26.87.226)
18:37.13dacterhm. regarding that "channels and groups" page, I think it proves that hackers can't write. Note that the three types of groups are never explicitly listed.
18:38.05irule[TK]D-Fender that is exactly what I needed to know and it is not on that doc, thanks a lot
18:39.05[TK]D-Fenderirule: It sort of implied.  You are talking about Zap channels, and it says that "group=" is a parameter to set in them.  You need to actually stretch those brain muscles of yours a bit more :)
18:39.39*** part/#asterisk iEatBabies (n=DudeMan@204.26.87.226)
18:40.05[TK]D-Fenderdacter: re-read the entire 2nd section.
18:44.05irule[TK]D-Fender I am, and I will NOT fail my job, I will become an * masta thanks to or despite you guys lol
18:45.16[TK]D-Fenderirule: Be like Bush... and just redefine "success" ;)
18:45.47irulelol
18:47.28drakohey can you use variables on sip.conf ?
18:47.41drakoso you don't have to repeat the same all the time for each entry
18:48.44pipwerkSet(VARNAME=value)
18:48.57[TK]D-Fenderdrako: No.
18:49.00pipwerkow, sip.conf, dri
18:49.22etfonhomey~wifisip
18:49.27jbotWi-Fi SIP phones suck.  All of them.  HARD.  Some only slightly less than others...
18:49.53drakohmm
18:52.01drakois a good idea store sip users on the database rather than on the plain sip.conf file?
18:52.11drakoif we have like near 100 extensions ?
18:52.43[TK]D-Fenderdrako: One more thing to break/support.  depending on the number of entries, potentially more or less work.
18:52.44pipwerkI like * realtime
18:53.15pipwerkI know big configs that are entirely generated out of a database
18:53.48drakoin my house im using sip.conf since they are only like 17 extensions
18:53.51drakoerm
18:53.55drako10 extensions i mean.
18:54.07drakobut in work its growing near 80 so far
18:54.37drakoand is getting annoying with the sip.conf
18:54.46PioneerVMgo to a script
18:55.13PioneerVMseems like you dont have to worry about the priority # changes too
18:55.14drakoPioneerVM, i was thinking about that, but seem for me easier to script it if the config is stored on a database.
18:55.29PioneerVMyea i mean use a script that accesses the dbase
18:55.41PioneerVMstore info in dbase and use a agi script to access it and generate config on fly
18:55.47drakothats why im thinking about migrate the sip accounts to a db.
18:55.58drakoaight
18:56.00PioneerVManything over a handful it seems thats the way to go
18:56.03drakois it stable?
18:56.08PioneerVMno idea :)
18:56.17PioneerVMim new to all this but im a programmer and just starting to write a script now
18:56.25PioneerVMactually as i write you im working in another window
18:56.30drako[TK]D-Fender, what did you mean with "One more thing to break/support"
18:56.35PioneerVMi have to do dynamic stuff for potentially thousands of #'s and mailboxes
18:57.03drakoPioneerVM, what are you coding?
18:57.10PioneerVMnew biz model
18:57.18PioneerVMi will have customers setting up lines dynamicall
18:57.19PioneerVMy
18:57.28PioneerVMand also changing menu dynamically
18:57.35PioneerVMso i need software/dbase type interface
18:57.50PioneerVMplus it seems if you do more than a handful of things its easier in code
18:57.57PioneerVMas you dont have to worry about these priority #'s etc.
18:58.07PioneerVMif you forget or skip a priority # it seems to f' everything up from what ive seen so far
18:58.24PioneerVMand if you want to insert you renumber etc. and have to keep creating contexts, its a pain for larger stuff
18:58.42PioneerVMi installed the perl interface, came with tons of samples
18:58.48[TK]D-Fenderdrako: Think about it, more effort ot install, configure, SECURE, and integrate a DB rather than a flat file.  How often are you going to make changes?  How sweeping are those changes ever likely to be?
18:58.49PioneerVMsuper easy
18:59.00wunderkinuse the n priority, that is what it is for
18:59.06PioneerVMyea i saw that
18:59.19*** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org)
18:59.21PioneerVMbut i have dynamic stuff
18:59.27PioneerVMIE: users will be able to change things on a web page
18:59.33PioneerVMeach voice mail will work different, etc.
18:59.44PioneerVM(and i love perl)
19:02.20iruleI have one sort of confusion, I can group zap1 and zap2, BUT I dial out differently because 1 is a direct line and 2 is through a regular pbx to get to the phone line, so I am using _9XXXXXXX for 1 on the direct line and _99XXXXXXX to get to the phone line through the other PBX
19:03.11*** join/#asterisk Mercestes (n=Merceste@rrcs-71-41-157-70.sw.biz.rr.com)
19:03.24iruleis there a way to GoTo dial _99. if using Zap2 or GoTo _9. if on Zap1?
19:04.52[TK]D-Fenderirule: put your channels in different CONTEXTS
19:05.42irule[TK]D-Fender arent I missing on the possibility of doing a round robin search for available channels?
19:06.03*** join/#asterisk Dr-Linux (n=Nothing@202.125.139.198)
19:06.05[TK]D-Fenderirule: You are completely mixing yourself up.
19:06.10iruleindeed
19:06.21Dr-Linuxany perl guy around?
19:06.21[TK]D-Fenderirule: Let me make an example for you.
19:06.30[TK]D-FenderDr-Linux: try asking in #perl
19:06.32irulethanks a lot, I gfreatelly appreciate it
19:06.58Dr-Linux[TK]D-Fender: it's about astman
19:08.01*** join/#asterisk chefrs (n=joe@c-24-8-226-145.hsd1.co.comcast.net)
19:08.21*** join/#asterisk |dennis| (n=dennis@200.32.233.82)
19:08.24chefrsAny idea why when someone faxes me it starts to come through, gets a bit and then says "Receive Failed" or when it goes to start it immediately drops to "Receive Failed" ?
19:08.59[TK]D-Fenderhttp://www.pastebin.ca/469426
19:09.44[TK]D-Fenderirule: In there, we define your 2 zap channels.  Each one has a differen INBOUND context.  the Dial stament I show you in there says that it will dial out the first available line BETWEEN those 2.
19:10.28*** join/#asterisk Exhar (n=Roy@213-73-139-87.cable.quicknet.nl)
19:10.38[TK]D-Fenderchefrs: * doesn't do faxing, so be precise about what software / tech is involved.
19:10.41*** join/#asterisk _DAW (n=chatzill@adsl-157-55-195.msy.bellsouth.net)
19:10.47_DAWAnyone here a polycom guru that can tell me if it is possible to disable the annoying call waiting beep on soundpoint?
19:10.55chefrs[TK]D-Fender: T1 into box. FXS out to an analog fax machine.
19:11.25Corydon-w_DAW: even better, you can CHANGE the call waiting beep
19:11.31[TK]D-Fenderchefrs: could be gain/echo, zap timing, etc.
19:11.51chefrs[TK]D-Fender: Hmm. I tested and we have multiple fax machines. #1 can fax to #2 and visa versa perfectly.
19:11.54_DAWI would just like it to go away.
19:11.57chefrsAnd when they do, it goes out to the PSTN
19:11.59[TK]D-Fender_DAW: set the type to "silent" in sip.cfg
19:12.14Corydon-w<CALLWAITING se.pat.callProg.6.name="call waiting" se.pat.callProg.6.inst.1.type="silence" se.pat.callProg.6.inst.1.value="6"/>
19:12.19[TK]D-Fenderchefrs: could be the far side
19:12.31_DAWThanks.. will give it a try.
19:12.32chefrs[TK]D-Fender: Yeah, try convincing these people of that.
19:12.47chefrsIt's happened too many times though for it to be a conincedence, IMHO
19:13.02Corydon-wchefrs: TDM400 or TDM800?
19:13.07chefrsNeither.
19:13.13chefrsRhino Dual FXS Module
19:13.24Corydon-wOh, then it's a Rhino issue for them to deal with
19:13.51chefrsSo you believe there's nothing I could adjust in * to help? It's just outright Rhino's issue?
19:14.06Corydon-wCorrect
19:14.36Corydon-wThe T1 card works fine with a good channel bank
19:14.48Corydon-wespecially for faxing
19:15.56Corydon-wIt's the analog cards that have issues
19:15.57chefrsHmm. Could it be echo cancellation or such?
19:15.57Corydon-wHighly unlikely, unless your wiring is bad
19:15.58*** join/#asterisk red9012 (n=marc3234@206-248-160-251.dsl.teksavvy.com)
19:15.59chefrsThat's entirely possible.
19:16.00Corydon-wWiring between your Asterisk box and your fax machines is bad?
19:16.10chefrsI wouldn't doubt it.
19:16.17Corydon-wThat's the only possible cause of echo in your situation
19:16.17red9012while in privacy I get: press 1 to accept call. press 2....  but keypress sometime dont work. why?
19:18.00Corydon-wchefrs: besides, echocan is automatically turned off when a fax tone is detected
19:18.09chefrshmm
19:18.14chefrsBut it's immediately routed to that zap channel..
19:18.15[TK]D-Fenderchefrs: Grab an analog phone, slap it on the line and see how it feels...
19:18.20chefrsIt works fine.
19:18.38Corydon-wAny echo with an analog phone?
19:18.43chefrsNot really
19:18.50chefrsBut I was directly at the box, not where the fax machine is
19:18.52Corydon-wThen it's probably the analog card
19:18.59Corydon-wOh
19:19.17Corydon-wYeah, you could check that
19:20.03chefrsMaybe I should boost are Tx gain?
19:20.20Corydon-wGrabbing at straws, are we?
19:20.29chefrsAll I got are straws to grab
19:20.36Corydon-wAnything to avoid calling Rhino tech support?
19:20.45chefrsMeh, Im reading their forums first.
19:21.05Corydon-wI'd call them and ask for a fix
19:21.26chefrsI may here in a bit
19:21.58Corydon-wIn that exact situation, I've used a CAC-1 channel bank with a PRI line going out, and fax works perfectly
19:22.21Corydon-wlegacy wiring to the fax, and all
19:22.22chefrsChannel bank isn't an option.
19:24.32*** join/#asterisk tuan_modulis (n=chatzill@3-82-252-216-static.enter-net.com)
19:25.13tuan_modulisHi, I'm a little bit stumped... need to install NVFaxdetect, but i seem to find outdated instructions so far... such as http://www.aussievoip.com/wiki/Asterisk+With+NVFaxDetect
19:25.43chefrsMy install came withit
19:26.23*** join/#asterisk tuan_modulis (n=chatzill@3-82-252-216-static.enter-net.com)
19:27.27*** join/#asterisk mocker (n=mocker@198.247.173.227)
19:29.16*** join/#asterisk jmacz (n=jmacz@190.24.97.247)
19:31.06tuan_modulisanyone know if http://www.newmantelecom.com/asterisk/faxdetect/ is down indefinitely? OR is it just down?
19:31.19tuan_modulisthat's where I can download the fax detect app
19:32.57*** join/#asterisk hfb (n=hfb@pool-72-67-156-130.lsanca.dsl-w.verizon.net)
19:35.05*** join/#asterisk saftsack (n=saftsack@pD9E04468.dip.t-dialin.net)
19:35.16*** join/#asterisk oej (n=olle@cust-IP-12.data.tre.se)
19:35.35etfonhomeyJust sat through a Nortel/Windstream sales pitch and it was pretty laughable.
19:36.41tuan_moduliswhat did they sell?
19:36.58pipwerkor try to sell?
19:37.04tuan_modulisya
19:37.19Mercestespot
19:37.23etfonhomeyWell, it turns out our Meridian PBX is still running software from 1996.
19:38.03pipwerkso why not? 'never fuck with a running system'
19:38.06tuan_modulisi say, out with the old
19:38.09etfonhomeySo, our CFO signed a contract to update it and my manager thought it might be a good idea to see if VoIP might be a good idea and how it would fit in with the Meridian PBX.
19:38.35etfonhomeyI work at university and as you can tell, that's not where I do my * work.
19:39.09pipwerk:)
19:39.15*** join/#asterisk fakhir (i=fakhir@unaffiliated/fakhir)
19:39.23etfonhomey600 phones on campus.
19:39.29etfonhomeyEasily doable with *.
19:40.36pipwerk600? small uni
19:40.46etfonhomeyBut, they'd rather pay ridiculous $$ for the Meridian "IP" license and $1K per 8 IP phone licenses.
19:40.55etfonhomeyYes, 1200 students + 300 fac/staff
19:41.15sevardTell them that you bought the Meridian, put in *, pocket the difference.
19:41.19Qwell[]etfonhomey: what school is this?
19:41.21pipwerkmy boss is thinking of going all-voip
19:42.04etfonhomeyQwell[], http://www.transy.edu
19:42.19Qwell[]yikes, that's easy to misread
19:42.31pipwerkuhuh
19:42.45etfonhomeyQwell[], which is why we've purchased as many domains close to ours as possible...
19:43.04Qwell[]I'm afraid to check
19:43.26etfonhomeyQwll[], they're committed to the meridian PBX, but are entertaining VoIP solutions, especially for a new building.
19:43.33*** join/#asterisk Deeewayne (i=dwayne@nat/digium/x-7be1d10820e13a01)
19:43.50*** join/#asterisk zapp-branigan (n=zapp-bra@81.202.214.78.dyn.user.ono.com)
19:44.00etfonhomey* can be easily integrated with a Meridian switch, right?
19:44.08tuan_moduliswell... what was that other famous school that changed everything to * with cisco phones
19:44.13tuan_modulisPrinceton?
19:44.17tuan_modulisMIT?
19:44.21tuan_modulisi think MIT
19:44.41etfonhomeyCisco is coming next week to talk Call Manager...
19:44.58etfonhomeyCisco + Nortel Meridian switch...
19:45.02etfonhomey:)
19:46.12*** join/#asterisk NOT_guru (n=NOT_wiza@24-241-103-142.static.stls.mo.charter.com)
19:46.26NOT_guruah  much better
19:46.28pipwerkhmm, how much would a t1 for the meridian be?
19:46.38NOT_gurudunno what happened to my connection
19:46.53NOT_guruand I officially hate kudzu
19:48.07NOT_gurulesouvage you still around
19:48.30etfonhomeypipwerk, the meridian is already on a t1 or two.
19:49.05pipwerkso, you get one t1 to go into your * and some smart dailplan
19:49.39pipwerkas soon as everyting has walked over to * you tell the management... owww, that huge support bill, just don't pay :)
19:51.23etfonhomeyI can't believe how expensive Nortel equipment is, but I bet it's comparable to Cisco.
19:51.49pipwerkhmmm, cisco can be pushed to huge discounts
19:51.53etfonhomeyBase model IP phone they quoted $250 + cost of a license.  (And they use H323)
19:52.08*** join/#asterisk sysreq (n=sysreq@modemcable171.134-81-70.mc.videotron.ca)
19:52.30pipwerkfor $250 you can get everybody on campus a dect handset
19:52.52pipwerkand have spare change for a pabx :)
19:53.04etfonhomeyThat model was comparable to the Polycom 330 which retails for $129 or so.
19:53.33pipwerkthe linksys phones are cool, so are the snom units
19:53.58etfonhomeyPolycom is goooooood.
19:54.32sevardetfonhomey: So $55 dollar 79xx models aren't considered base models (and are SIP)?
19:55.01pipwerkSIP is to easy and open
19:55.06etfonhomeysevard:  I have no idea what you're talking about.
19:55.08pipwerknot cisco
19:55.13sevardetfonhomey: http://www.horizondatacom.com/eCart/browse/productinfo.php?manpartnum=CP-7910&flash=0&dram=0&price=55&condition=Refurb&voltage=#overview
19:55.33sevardetfonhomey: base cisco phone, $55, not H323
19:55.37LeddyHMI relaly like our polycom's
19:55.40etfonhomeyrefurbished without a license?
19:55.56sevardI believe so
19:56.33Qwell[]7910 doesn't run SIP...
19:56.38etfonhomeyThat doesn't really compare to a new Polycom 330 for $129 that plugs in and works.
19:57.38*** join/#asterisk shido6 (n=shido6@d221-68-200.commercial.cgocable.net)
19:57.46etfonhomeyI haven't dealt with Cisco.  How much is a license for one of their phones to run on their system?
19:57.55sevarda license is what, 7 freaking dollars?
19:58.14*** join/#asterisk IPmonger (n=ipmonger@63.139.176.1)
19:58.42sevardQwell[]: I thought there were SCCP and SIP images for those phones
19:58.43Qwell[]sevard: license?  $100+
19:58.45pipwerka license is free if they want you bad enough
19:58.49Qwell[]I don't think so - not for the 7910
19:59.00*** join/#asterisk adorah (n=Michael@87.69.57.73.cable.012.net.il)
19:59.37etfonhomeyGotta jet.
19:59.41etfonhomeyexit
19:59.48sevardlast time I got a license for one phone I swear it was $7.00
20:00.24sevardand the license didn't give me any valueable tools that I needed
20:00.37Qwell[]license != support contract
20:01.00Qwell[]and there is only skinny firmware for 7910
20:01.11Qwell[]7911 can do sip
20:01.58ZaVoidno one should buy the tcb400 card if they have asterisk 1.4.2 apparently
20:02.13Qwell[]ZaVoid: why is that?
20:02.18ZaVoidbec ause it doesn't work
20:02.22ZaVoidand digium for 3 days can't find out why
20:02.36Qwell[]well, have you tried upgrading?
20:02.41ZaVoidwhenver aq g.723 call is made it sounds metallicy
20:02.42Qwell[]1.4.4 is the latest...
20:02.46ZaVoidthey have upgarded my kernel
20:02.50ZaVoidupgarted to 1.4.4
20:02.53ZaVoiddowngraded to 1.2.x
20:03.16Qwell[]and did it work on 1.4.4?
20:03.19ZaVoidg.729 works fine.. just not g.723
20:03.20ZaVoidno
20:03.27ZaVoidif it worked on 1.4.4 i woulda kept it there )
20:06.12ZaVoidi should just get a real media gw i guess
20:06.30sevardreturn it and get one from digium?
20:06.54*** join/#asterisk |dennis| (n=dennis@200.32.233.82)
20:07.14sevardoh, heh, i'm mis reading again
20:07.55*** join/#asterisk [hC] (n=hardcore@66.119.167.162)
20:12.16*** join/#asterisk neolynx (n=u@165-123-204-62-pool.dsl.fcom.ch)
20:12.30neolynxhello
20:12.42neolynxsome swiss people around ?
20:12.46neolynxich have a weird problem...
20:13.03neolynxi'm afraid swiss DSL providers start to block IAX traffic
20:13.19neolynxhad it first with cablecom.ch, now with green.ch
20:13.27neolynxsucks bigtime
20:15.33*** join/#asterisk KuJaX (n=kuj@customtrading.dsl.xmission.com)
20:16.17neolynx305 users and noone says hello
20:16.24neolynx:-)
20:16.30KuJaXhi hi
20:16.36Corydon-w~ask
20:16.45jbotQuestions 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.
20:16.45neolynx:)
20:16.47filetelnet
20:16.47Nuggettelnet is eeeeeeevil!
20:17.26neolynxCorydon-w: tell me something I don't know :)
20:18.39Strom_Mwhat Corydon-w was going to say was "---------- --- --------- - - ----------- - ----------"
20:19.42neolynxok :)
20:19.55neolynxhoping it was somethign about *** and DSL providers
20:20.01Corydon-wThere's undoubtedly quite a bit of that, but I don't talk of such things during work hours.
20:20.16*** join/#asterisk AndrewGearhart (n=chatzill@h1.39.213.151.ip.alltel.net)
20:20.18pipwerkneolynx: so ask your provider
20:20.19Corydon-win this channel, anyway
20:20.22neolynx10pm here... should stop working...
20:20.26*** part/#asterisk goozbach (n=goozbach@brooks.netradius.com)
20:20.42Corydon-w15:20 here
20:21.16Mercestes15:21 here
20:21.19AndrewGearhartwhat's the difference between point to point T1 and T1 for internet use?
20:21.21*** join/#asterisk Aphelion (n=lk@unaffiliated/lv)
20:21.38Corydon-wAndrewGearhart: nothing
20:22.07neolynxAndrewGearhart: 2mbit link to your provider, or to your buddy
20:22.09Corydon-wthe distinction is on what the other side is connected to, not anything about the T1 itself
20:23.09*** join/#asterisk pfn (n=pfnguyen@netblock-66-245-252-239.dslextreme.com)
20:23.10AndrewGearhartneolynx: so, essentially, I'm choosing who the T1 will be connected to... the T1 provider... or some other ISP?
20:23.24Corydon-wor not an ISP at all, but simply another customer premise
20:23.32neolynxAndrewGearhart: at least in europe with our E1 it's like that
20:23.54AndrewGearhartneolynx & Corydon-w : that make sense
20:23.54neolynxAndrewGearhart: T1 are 2 copper lines to *somewhere*
20:24.06AndrewGearharthehe
20:24.09neolynxAndrewGearhart: put some modem on both sides, and do what you want
20:25.04neolynxAndrewGearhart: use a HSDL modem for telefony, and you'll get a G.703 primary rate interface
20:25.28AndrewGearhartthanks. :)
20:25.39AndrewGearhartJust got my quote back on PRI pricing
20:25.55neolynxAndrewGearhart: I can sell you some PCI hardware to use it with the asterisk server I'll sell you afterwards
20:25.56neolynx:->
20:26.30Strom_MG.703 doesn't actually specify ISDN PRI
20:26.45Strom_Mit's merely the T1 or E1 circuit itself
20:27.07explidousneolynx how do you block IAX? you can use any port and TCP you should be able to find a way around that...
20:27.28Qwell[]Strom_M: what's the official time you have to wait between pressing the hookswitch and releasing, before it's considered a "new" call?
20:27.35neolynxI *dont* want to block it, my internet provider is blocking it
20:27.37neolynx!
20:27.47Strom_MQwell: I believe it's two seconds
20:27.53neolynxStrom_M: ah, an expert :)
20:27.55Qwell[]http://bugs.digium.com/view.php?id=9660
20:28.06Qwell[]I'm kinda mehish about that
20:28.26Aphelioni've just been asked to create a rails interface to "an asterisk install using mysql". that is seriously -all- i have to go on atm... if i understand correctly, is the info on asterisk realtime architecture what i need to understand in order to accomplish this?
20:28.28neolynxStrom_M: you don't get G.703 directely out of your copper lines
20:28.30Strom_MQwell: it should be two seconds IIRC
20:28.30AndrewGearhartQwell: for POTS... it's 9 seconds
20:28.37Qwell[]AndrewGearhart: eh?
20:28.39Strom_MAndrewGearhart: uh, no
20:29.10neolynxAndrewGearhart: yes, but only if you receive the call
20:29.31Strom_MQwell: if any services which require the use
20:29.32Strom_Mer
20:29.35AndrewGearhartneolynx: ah, sorry... were we talking about a dialed call?
20:29.37Qwell[]hmm, that is true...  it could be different on incoming vs outgoing
20:29.59Qwell[]he didn't specify I don't think
20:30.16Qwell[]oh, yes he did..  outgoing
20:30.18Strom_Mif all services which need a recall dial tone to operate are turned off, then in theory you can merely bump the hookswitch and get a new dial tone
20:30.38Strom_Motherwise, two seconds
20:30.46Qwell[]this is gonna suck to fix
20:30.57Strom_Mpor que
20:31.23*** join/#asterisk phocus (n=phocus@67.32.20.66)
20:31.58phocushey guys , do those cheep vonage usb phones at walmart work
20:31.59phocus?
20:32.03Qwell[]no
20:32.14Strom_Mno
20:32.20AndrewGearhartvonage doesn't use sip... does it?
20:32.34*** part/#asterisk NOT_guru (n=NOT_wiza@24-241-103-142.static.stls.mo.charter.com)
20:32.57AndrewGearhartwell... to rephrase my level of uncertainty... does vonage use sip?
20:33.23*** join/#asterisk vykarian (n=stefano@200.212.169.2)
20:33.30Qwell[]sure it does
20:33.56AndrewGearhartam I thinking of Skype then? (that doesn't use sip)
20:34.01Corydon-wCorrect
20:34.02Strom_MQwell's answer could also apply to the question "does vonage blow dead yaks?"
20:34.02vykarianhi all
20:34.10vykarianhas someone already seen that?
20:34.11vykarianMay  3 17:28:45 ERROR[7246] chan_unicall.c: Unable to open channel 1: Success here = 0, tmp->channel = 0, channel = 1
20:34.18Qwell[]Strom_M: the "no", or "sure it does"?
20:34.21Qwell[]hopefully the latter
20:34.24Strom_Mthe latter
20:34.31Qwell[]good
20:35.04phocusso the usb skype phones wont work with asterisk
20:35.08Qwell[]no
20:35.27Strom_Mphocus: just shell out the money for a real phone or a real line interface card :)
20:35.39explidousneolynx: sorry for the missunderstanding, it is very difficult blocking IAX, how does your provider block it, did you try to connect via udp or tcp?
20:36.31*** join/#asterisk csaba (i=HydraIRC@adsl5-204.ptt.yu)
20:37.47explidousphocus: I suppose they wont, as most of the vonage boxes don't as well... beside of the ones that can be cracked...
20:38.12csabaHello, I have talked to a person from this channel about a week ago regarding a solution, but I forgot your user name. If you remember me could you message me?
20:38.52Sweepergood cell phones with a SIP client available?
20:39.23*** join/#asterisk zotz (n=zotz@24.244.163.157)
20:39.40pipwerkSweeper: nokia e60 etc
20:39.58KuJaXHello.  I recently created a temporary vacation message.  I cannot get my normal voicemail away message to come up and there doesn't seem to be an option when dialing *98 to use the other message over the vacation message.  Any suggestions?
20:40.53AndrewGearhartcan somebody help me understand what DID service is? My understanding was that it was phone numbers to go with your service (VoIP ITSP/PRI) ... I'm beginning to think I was wrong
20:41.45pipwerkhttp://en.wikipedia.org/wiki/Direct_Inward_Dialing
20:41.50pipwerki guess
20:42.06*** join/#asterisk Maxxed (i=foobar@65.59.245.122)
20:42.14Maxxedheya fellas, i gota quickie..
20:42.23Maxxedi have a asterisk box with a t1 line card
20:42.45Maxxedwhen a did is dialed, asterisk is only getting the last 4 digits
20:42.55Maxxedlike the telco is only sending the last 4 of the did
20:43.03Sweeperpipwerk: mmm
20:43.03Qwell[]Maxxed: yeah, tell them to send all of them
20:43.07Sweepere60 is massive
20:43.12Qwell[]Maxxed: that's precisely what's happening
20:43.16SweeperI'm wanting something like the e70
20:43.17Maxxedand im trying to send, say for example 000-000-1234 to exten 200 for example
20:43.18Sweeperactually
20:43.24SweeperI just want a freaking e70-2
20:43.28Sweeperunlocked D:
20:43.31Maxxedqwell, tell them?
20:43.36Qwell[]your telco
20:43.48Qwell[]you usually have a choice of how many digits you receive
20:43.49Maxxedcall them and say, hey bitches, send the whole number?
20:43.52Qwell[]yes
20:43.53Maxxedah!
20:43.54Strom_Mwell, unless your telephone company is Cox, in which case you'll ask for DNIS remapping and they'll go "What's 'genus'?"
20:44.01Qwell[]tell them you want 10 digits
20:44.07Maxxedcool cool
20:44.11Maxxedwell hey thanks :)
20:44.17Qwell[]~thanks
20:44.17jbotde rien, Qwell[]
20:44.38Qwell[]jbot: no, thanks is <reply> Just send money...
20:44.40jbotokay, Qwell[]
20:44.50Strom_M~thanks
20:44.50jbotStrom_M: de nada
20:45.05Mercestes~mercestes
20:45.07jbotyou are probably almost a total nub
20:45.12Strom_M~strom
20:45.15jbotrumour has it, strom is Southern California's only residential ISDN BRI customer
20:45.23Qwell[]nice
20:45.26serotonin|worklol
20:45.32Qwell[]how's that working out, anyhow?
20:45.40Strom_Mive been in England for two weeks
20:45.44Strom_Mso, it's still here
20:46.01Qwell[]BRI?  In Europe?  Are you mad?
20:46.07Strom_Mno, i'm in los angeles
20:46.15Strom_Mi /was/ in england
20:46.16Qwell[]England, LA?
20:46.17Strom_Mgot back tuesday
20:46.20Qwell[]oh
20:46.58Strom_Mwant to see the best of my vacation snaps?
20:47.17Mercestesno, jbot, mercestes is the dark overlord of #asterisk.
20:47.26Mercestes..>.<
20:47.28Qwell[]you fail
20:47.29Maxxedso when i call the telco, what should i say?
20:47.32MercestesI fail
20:47.40Maxxedsend all the crap?
20:47.45Qwell[]jbot: no, Mercestes is a total nub
20:47.47jbotQwell[]: okay
20:47.47Strom_MMaxxed: "I want 10 digit DNIS"
20:47.47MercestesMaxxed, "Screw you AT&T!  I went to Vonage!"
20:47.52Maxxedhah
20:47.58Maxxedok ok, cool cool :)
20:48.12Strom_Mand remember, it's pronounced "DEE-nis"
20:48.21AndrewGearhartwell folks... I'm outta here... catch ya tomorrow.
20:48.25Qwell[]~dnis
20:48.26jboti guess dnis is Dialed Number Identification Service, wherein the telephone company delivers the called number as part of the call setup information.  This service is useful when multiple numbers terminate on the same trunkgroup.
20:48.26MercestesDEE IN EYE ESS
20:48.47Qwell[]I should make a Preferred Number Identification Service
20:48.53Strom_Mbahahahhaha
20:48.58Mercestes....
20:48.59Mercesteslol
20:49.03fileStrom_M: vacation snaps? where!
20:49.19Strom_Mhttp://www.stromcarlson.com/photos/london-processed/
20:49.23Strom_Mthese are the best fourteen
20:49.34Maxxedoh! heres one
20:49.34*** join/#asterisk DrukenLPY (n=jdumais@CPE000e08cb2a29-CM00137189cb0c.cpe.net.cable.rogers.com)
20:49.41Maxxedsay for example if the caller
20:49.47Maxxedwell, its in san antonio
20:49.57Maxxedand they dont have to dial the area code when calling localy
20:50.03Maxxed*i thik*
20:50.06Strom_Mit should still send 210
20:50.12Maxxedok cool :)
20:50.12Strom_M210-555-2368
20:50.26*** join/#asterisk sysreq (n=sysreq@modemcable171.134-81-70.mc.videotron.ca)
20:50.31Strom_Mif dialing seven digits, the originating switch assumes 210
20:52.00*** join/#asterisk syzygyBSD (n=chatzill@66.226.228.204.cpe.speedyquick.net)
20:52.34syzygyBSDcan someone suggest a fax software to use on 1.2.17?
20:53.59[hC]Qwell:  I recently implemented a Basic Analog Line Level Syncronizer on my analog lines, it works great
20:54.31Qwell[]nice
20:54.48Strom_Mfor digital circuits, you're going to need a Circuit Order Controller Kit
20:54.53Qwell[]is it the Automatic Conductor type?
20:54.56explidoushC: that sounds interesting...
20:56.04[hC]Oh yeah, its fantastic. I even wrote support for it into my Command Line Interface Tools
20:56.34Qwell[]you scared him off
20:56.55Strom_Mwhat about the Automatic Noise Uncertainty Smoother?
20:58.04Strom_Mfile: do you like my photos?
20:59.59fileyessssss
21:00.23Strom_Myay
21:02.22*** join/#asterisk alexpe (n=alex@cev75-1-81-57-14-91.fbx.proxad.net)
21:05.07Hmmhesaysheh
21:05.09Hmmhesaystheres file
21:05.50Hmmhesayshaha
21:05.52Hmmhesayshow you
21:06.11filedo you mean... how ARE you?
21:06.29Hmmhesayswhat are you the freaking grammar police?
21:06.29*** join/#asterisk he11e (n=h@p549dbf51.dip0.t-ipconnect.de)
21:06.47fileI will neither confirm or deny that
21:07.45Hmmhesaysheh
21:07.58Hmmhesaysi actually bought a netgear wireless router
21:07.59Hmmhesayscrazy
21:08.31he11ehi all. on my ISDN TE line (mISDN) i get such "errors" -> P[ 2] * RELEASING CHANNEL pid:3 ctx:from-pstn dad:222}�� oad:43100812p�912. the extensions are totaly broken. can someone give me an hint what i can do to solve this?
21:09.46shido6...
21:09.53shido6...  (..lint)
21:11.34*** join/#asterisk Gpl_Source (n=The_natu@unaffiliated/gplsource)
21:13.21*** join/#asterisk digus (n=digus@206.222.110.30)
21:17.22shido6what kind of headset does bill buchanan wear in 24?
21:17.52Strom_Mdo you really want to take telephone advice from a TV show that uses an old AT&T Merlin ringer on a Cisco telephone?
21:17.54*** join/#asterisk toot (n=toot@84.19.255.123)
21:18.12Qwell[]yes, yes I do
21:18.21Mercesteslol
21:23.32shido6:)
21:24.26shido6kinda looks like the nokia BH-900 but with a thinner boom and its almost invisible
21:24.41Qwell[]it's a prop
21:24.53*** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2)
21:24.55shido6dont ruin my dreams
21:24.57Strom_Mno! it's TV!  therefore it MUSt BE REAL
21:25.18*** join/#asterisk dasenjo_ (n=dasenjo@190.24.176.78)
21:25.23Qwell[]if it were real, it would have a big logo
21:25.55mogheh
21:25.58shido6looks the jawbone is going to win
21:26.01mogi liked the giant cisco add in the last one
21:26.02vykariancan somebody help me with unicall??
21:26.07shido6eww
21:26.09shido6unicall
21:26.14shido6chan unicall
21:26.19shido6e1?
21:26.32vykarianr2
21:26.36shido6good lord
21:26.45shido6mexico? or middle east? where ar?
21:26.52shido6where at?
21:26.55vykarianbrazil
21:27.12shido6whats the prob?
21:27.30vykarianchan_unicall.c: Unable to open channel 1: Success
21:27.36vykarianhere = 0, tmp->channel = 0, channel = 1
21:27.39vykarianweird error msg
21:27.58vykarianthats when start asterisk, right after, it exits
21:28.09vykarianMay  3 18:23:35 ERROR[11777] chan_unicall.c: Unable to register channel '1-15'
21:28.55vykarianalready recompiled the zaptel driver and unicall patches =\
21:28.57vykariannothing
21:29.30vykariansearched at unicall.c source for that error message and nothing
21:33.14*** join/#asterisk Cinen (n=Cinen@208.70.20.33)
21:34.12shido6ok......
21:36.53*** join/#asterisk nasls_lsa (n=chatzill@athedsl-133936.home.otenet.gr)
21:37.55*** join/#asterisk NOT_guru (n=chatzill@209.145.181.55)
21:43.13*** join/#asterisk KuJaX (n=kuj@customtrading.dsl.xmission.com)
21:43.27KuJaXWhere are the pre-recorded voicemail box messages stored?
21:44.01shido6:)
21:44.53shido6<PROTECTED>
21:45.12shido6looking to change the sounds?
21:45.26KuJaXyes, right now i have a vacation sound and can't get my normal "away" message to prompt people.
21:45.28*** join/#asterisk umay (n=chris@71-208-167-161.hlrn.qwest.net)
21:49.11*** join/#asterisk erousse (n=chatzill@207.253.203.34)
21:51.24eroussehello, anyone here with some experience with VoiceGenie and Asterisk ?
21:51.52Corydon-wNever heard of it
21:53.36erousseIt's basically a IVR system, but it also support SIP. But I'm having so much problems trying to integrate it with Asterisk...
21:55.50Corydon-wAnd you're doing this, because Asterisk doesn't have an IVR?
21:56.06shido6so you're the guy from QC
21:56.11shido6boucherville ?
21:57.11shido6why are y ou scewing with the supheaders?
21:57.16shido6sipheaders
21:57.21erousseI'm doing this because we bought licenses for VoiceGenie... before even thinking about the IVR within Asterisk...
21:57.21*** join/#asterisk angler (i=angler@pdpc/sponsor/digium/angler)
21:57.21*** mode/#asterisk [+o angler] by ChanServ
21:57.22SweeperCorydon-w: and because Asterisk doesnt support SIP
21:57.41erousseshido6, yep!: P you must be the co-woker of Francis ? heh
21:57.56shido6no :)
21:58.06eroussehahah
21:58.23shido6incoming call from your lady
21:58.39*** join/#asterisk Mavvie (n=edwin@ppp175-226.lns4.syd6.internode.on.net)
21:58.53shido6Greg from NuFone
21:59.38shido6the music on hold is putting me to sleep
22:00.19*** join/#asterisk pfn_cIc (n=pfnguyen@64.235.249.50)
22:12.10*** join/#asterisk mitcheloc (n=mitchelo@titaniumsoft.net)
22:12.10*** join/#asterisk Cresl1n (i=matt@nat/digium/x-6d95380d69f3c62a)
22:12.10*** mode/#asterisk [+o Cresl1n] by ChanServ
22:16.31*** join/#asterisk ToyMan (n=Stuart@cpe-24-161-103-133.hvc.res.rr.com)
22:17.29*** join/#asterisk angler (i=angler@pdpc/sponsor/digium/angler)
22:17.29*** mode/#asterisk [+o angler] by ChanServ
22:17.41anglergrrr.....
22:22.38*** join/#asterisk Gregabyte (i=wintermu@nat/digium/x-cf37045d62d442f0)
22:22.53*** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca)
22:24.43*** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com)
22:24.49*** part/#asterisk brodiem (n=brodiem@67.110.68.66.ptr.us.xo.net)
22:25.07*** join/#asterisk rmayorga (i=rmyorg@unaffiliated/rmayorga)
22:25.22*** join/#asterisk NOT_guru (n=chatzill@209.145.181.55)
22:28.52*** join/#asterisk wunderkin- (n=kev@dslstat-ppp-95.fastq.com)
22:31.43*** join/#asterisk n4ycw (i=cspot@ip68-109-8-207.pn.at.cox.net)
22:33.29*** join/#asterisk Inode (n=Inode@modemcable114.59-70-69.static.videotron.ca)
22:41.54*** join/#asterisk kieranmullen2 (n=kieranmu@71.245.97.59)
22:43.20kieranmullen2hello all, I was wondering if I could  setup an ivr as an extension that can be called via a sip uri form the outside? if so how?
22:48.41NOT_guruGregabyte thanks again
22:49.43Gregabyteno problem
22:50.08kieranmullen2what did you have done?>
22:50.36*** part/#asterisk mog (n=mog@c-71-207-200-130.hsd1.al.comcast.net)
22:50.44NOT_gurugot some sense smacked into me
22:51.23NOT_gurunah  greg was awsome
22:51.43NOT_guruI was having troubles with my brand new tdm804p
22:51.50NOT_guruand greg aced it
22:52.13NOT_gurunothing better than strong support people
22:52.15filewhat was the issue?
22:52.36Corydon-wGreg's a very nice guy
22:52.37NOT_guruheh  1st  please don't spit on me.... I use trixbox 2.0
22:52.59NOT_gurubut I had already updated the zap drivers
22:53.11NOT_guruto 1.2.17.1
22:53.12Corydon-wangler, otoh... ;-)
22:53.21NOT_guruand I just had some conf issues and
22:53.25fileah
22:53.25NOT_guruyou know
22:53.36kieranmullen2did you end up dumping trixbox?
22:53.38NOT_gurutrixbox tricks that needed tricking
22:53.46NOT_guruoh  I plan on it
22:53.48kieranmullen2I am using freepbx & asterisk...
22:53.54kieranmullen2no trix
22:53.56NOT_gurubut I am not much in the linux world
22:53.58NOT_guruI am a BSD guy
22:54.07NOT_guruand the zaptel drivers for bsd are lagging
22:54.24kieranmullen2centos
22:54.27NOT_guruso I will eaither learn more about this "linux" thing
22:54.39NOT_guruyes centos as thats what I am exposed to now
22:54.56kieranmullen2are you the computer guy for your office or something?  (reason why you got that line card)
22:55.03NOT_guruactually ... in all honesty   I have another centos box I run VMserver on
22:55.10NOT_guruyes I am
22:55.11anglercentos still has the spin lock issue i think
22:55.23NOT_guruI patched that as well prior to the call
22:55.28NOT_gurubut thankyou for the heads up
22:55.44kieranmullen2Get spare box and put cent freepbx or trix  on it or *now
22:55.50JTkieranmullen2: trixbox uses freepbx
22:55.56JTkieranmullen2: don't recommend that in here :P
22:56.00JT~trixbox
22:56.02jbotTrixbox is a full linux distro that includes , FreePBX, and other 3rd party add-ons. It is these things on top of which make it seriously painful to support and hence you will find little help here for it. Try asking in #trixbox , or their forums & WIKI at http://www.trixbox.org
22:56.05NOT_gurucorrect
22:56.14NOT_guruis freepbx taboo here?
22:56.25JTwe don't support it
22:56.29NOT_guruoh I know that
22:56.37hardwireJT: liar
22:56.41NOT_guruI just also understand trixbox is frowned upon
22:56.43kieranmullen2join the #freepbx irc channel actually I am on both :-)
22:56.48JThardwire: ?
22:56.58hardwireJT: you know what I'm talking about, I have pictures.
22:57.12kieranmullen2for home use those little 1 line analog line cards are $15 on ebay :-)
22:57.20JThardwire: err ok, you must be talking about someone else
22:57.56hardwireJT: you just keep thinking that
22:58.08JThardwire: ...
22:58.14NOT_gurumind if I pop a link in here?  I want to know if this howto for a real asterisk box on centos is.... acceptible to you all
22:58.16kieranmullen2in asterisk how could I setup an extension to go straight to ivr?  I have the ivr setup already...  This would be for sip direct uri calling ext@somdina.com]
22:58.18hardwireJT: ...
22:58.28JThardwire: dude, what's up? seriously
22:58.39kieranmullen2appreciate any information
22:58.56hardwireJT: absolutely nothing, just being a punk at random.
22:59.02JTkieranmullen2: those $15 cards are really poor quality
22:59.03hardwiredude, you've been punked.
22:59.18JTthe funny thing is, no-one's laughing
22:59.20anglerkieranmullen2, exten => 100,1,Goto(mainmenu,s,1) ?
22:59.21kieranmullen2jt - yes hence the words "play" and "home" use :-)
22:59.24JTlive in your deluded mtv world ;)
22:59.40hardwireJT: I'm not laughing either..
22:59.53JTkieranmullen2: my idea of "play and home use" is T1 channel bank + polycom sip phones
23:00.02hardwireit actually kind of sucked, but since you kept responding I had to go on and on.
23:00.11hardwireyadda yadda yadda.. here we are.
23:00.14kieranmullen2you have a t1 channel bank in your home?
23:00.18JTkieranmullen2: yes
23:00.32kieranmullen2running a business out of it I would guess and it is not for play
23:00.39JTit is for play
23:00.50JTi avoid consumer rubbish IT/telephony gear when i can
23:00.54JTbecause it's trash :)
23:00.58kieranmullen2is sagoma a naughty word?
23:01.06JTmaybe you mean sangoma
23:01.14kieranmullen2what is the deal the prices area about the same for normal ones
23:01.25kieranmullen2normal= digi
23:01.50JTyes they're competing :P
23:02.15kieranmullen2= no clear advantage to me
23:02.22*** join/#asterisk wunderkin (n=kev@dslstat-ppp-95.fastq.com)
23:03.02*** part/#asterisk Inode (n=Inode@modemcable114.59-70-69.static.videotron.ca)
23:03.55*** part/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage)
23:06.36*** join/#asterisk etfonhomey (n=etfonhom@74-140-213-69.dhcp.insightbb.com)
23:07.22*** join/#asterisk votre (n=jordan@69-178-156-154.static-ip.telepacific.net)
23:08.02votrei have recently started working with a small business and am trying to convince our head of IT to switch to an open source telephony server (I am not very familiar with the technology myself) and am wondering if Asterisk is the right product to replace our current product (altigen) with
23:08.25votreany thoughts?
23:08.51JTwhat do you need it to do?
23:09.50kuku5Can I park calls onto different lots ?
23:10.30votrewell we have a T1 line (half for internet and half for phone lines). We will need it to be able to provide voicemail, pass through caller ID from external calls, be able to direct external calls to the right line either by direct lines and/or extensions, and provide auto attendants
23:10.41votrei know for sure those are our most used and needed functions
23:11.23JTvotre: yes they can be done
23:14.34votrefantastic
23:15.05votreis there a prebuilt server I can purchase anywhere with all the compatible hardware or what is the most convenient way to go about that?
23:16.20kieranmullen2you could put together a decent white box server, put an os on it and hire someone to telnet in and configure it for you
23:17.04votrewell i am pretty linux savvy, i am more just concerned about hardware compatibility when it comes to telephony as I am not familiar with the technology much
23:17.25kieranmullen2buy digium cards they like asterisk :-)
23:17.35NOT_gurupretty much
23:17.39NOT_guruand great support
23:17.55votrehaha okay great, I will look into that. Thank you so much guys for your help!
23:18.07kieranmullen2sip phones?  should be cross compatible... people like the grandstream product line, polycomms
23:18.09NOT_guruseriously... I was having problems with my new tdm804p ( I am NOT linux savy )
23:18.13kieranmullen2I know nothing
23:18.39NOT_gurucalled support  they asked if they could get into the box, of course I said sure
23:18.49NOT_guruhe had it going in about 10 minutes
23:18.51votrewell all I know is we are currently using h.323 (i think thats what it is?) and our IT is very interested in switching to SIP
23:18.53NOT_guruMAYBE 15
23:19.36NOT_guruI like they cisco phones we had from buying a company
23:19.43NOT_gurubut the are pricey
23:19.45JTkieranmullen2: dude
23:19.49JTDaveCanoe: are you on crack?
23:19.52JTerr
23:19.54JTkieranmullen2: i meant
23:20.03JTno-one here recommends granstream
23:20.08JTlet alone likes them
23:20.10JT:P
23:20.24kieranmullen2sorry I like them.. they work for me  and I put a disclaimer in there already
23:20.31NOT_guruI wanted to try a grandstream 2000  just had these cisco's already
23:20.49kieranmullen2I know nothing
23:20.50NOT_gurunow I am leaning aastra to play with
23:21.04JTpolycom and aastra are good
23:21.06JT~phones
23:21.08jbotsomebody said phones was http://bani.anime.net/phones/.  SIP Hardphones in order of quality/suggestability : Polycom (any), Aastra 480i, Cisco 7940(+), Linksys SPA-9XX, Snom
23:21.26NOT_guruI have 7940's and 7960's
23:21.31NOT_gurugood tough phones
23:21.51NOT_guruas long as you don't bash the screen
23:22.01NOT_gurubut thats for any phone I guess
23:22.35JTbad company though
23:22.44NOT_guruwho  cisco?
23:22.51shido6crisco
23:22.56JTcisco are shocking with support if you don't spend zillions a year with them
23:22.57NOT_guru<PROTECTED>
23:23.05NOT_guruthis is true
23:23.07JTreal tight with their firmware upgrades
23:23.12JTstupid charging for firmware
23:23.15NOT_guruyou have to feel your away around cisco stuff
23:23.21NOT_guruthat too
23:23.23JTwith a hammer
23:23.33votreI know that are phones are currently just plugged in with regular phone cords into your generic phone jacks. We have plain old phones that can be bought from any Best Buy or Wal-Mart or anything, will these run the risk of not working with what Asterisk provides?
23:23.38NOT_gurubut really  I think you can get the firware account for like $7
23:24.10NOT_guruyou would want FXS ports on your card for the phones
23:24.32NOT_guruand FXO ports if your lines going into the phone system are analog
23:24.49NOT_guruor a t1 / e1 card if the phone lines are digi
23:24.50NovceGuruAnybody using http://voicelift.com/ ? I had a pleasant conversation with their "support manager" today, just looking for some opinions
23:25.20votreoh wow. This is starting to get over my head. Maybe I will just report to our head IT what you guys have told me and see if I can persuade him
23:25.22NOT_guruvotre how many extensions now?
23:25.36JTvotre:
23:25.38JT~thebook
23:25.43jbotextra, 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
23:25.51votrewell we have quite a few extensions but 12 dedicated lines
23:26.03JTvotre: are the lines analogue or digital?
23:26.19votreI am not sure. Is there a way to find out?
23:26.35votreI know it comes in as T1
23:26.58NOT_gurudo you have 12 phone lines plugging into your phone system
23:26.59votreBut again, I am pretty darn clueless when it comes to this technology
23:27.02JTT1s aren't analogue
23:27.09votreNOT_guru: yes
23:27.19NOT_gurubut it could be broken out into analog lines
23:27.29JTNOT_guru: well that'd be pretty stupid
23:27.32NOT_guruthats what we have done at work from previous system
23:27.38JTsounds like something "only in america"
23:27.39NOT_guruits lagacy
23:27.44NOT_guruyes it is
23:28.04NOT_gurubut anyways
23:28.20votreyes, that is what I am not sure about. I know there are software clients for the phone system that run on our Windows comps and that you can call from those clients if you have a headset in your comp
23:28.27NOT_guruif your t-1 get broken down into analog lines and then plugged into your phone system
23:29.40*** join/#asterisk pdx5k (n=casey@64.81.142.112)
23:30.03pdx5ki installed 1.4.4 and have it working fine except for meetme .. i can't get it to work
23:30.19pdx5kwhen i call the extension for the conference it says there's no meetme application for that extension
23:30.22shido6got ztdummy?
23:30.28pdx5kyes
23:30.37shido61/2 way tere
23:30.59NOT_guruprobably votre
23:31.17NOT_guruas JT mentioned  the book is probably a good place to see functions
23:31.33votreOK, well I will investigate further
23:31.42NOT_guru~thebook
23:31.43jboti heard thebook is a book called  Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11
23:32.14pdx5kwhat do you have to do besides install ztdummy?
23:32.21PioneerVManyone here use the AGI library in perl?
23:32.24NOT_gurudinner is calling my name  have a good night all
23:32.31shido6me, too
23:33.26*** join/#asterisk JT_ (n=jon@unaffiliated/jt)
23:34.47PioneerVManyone here familiar with "SAY ALPHA" command
23:35.03PioneerVMi'm having problems calling it -- it does not say anyting but the SAY NUMBER works
23:37.01pdx5kpbx_extension_helper: No application 'MeetMe' for extension (inbound, 7, 2)
23:37.17pdx5kztdummy and zaptel are loaded .. so why is it not working?
23:38.25JTmaybe meetme is not loaded
23:38.50pdx5khow do i check on that?
23:39.04JTsee if it loads when asterisk starts
23:39.09kieranmullen2I am having an odd problem with 2 ivrs playing at the same time.. I want an ext to go to ivr
23:39.09kieranmullen2extensions_additional.conf
23:39.09kieranmullen2[ext-local]
23:39.09kieranmullen2include => ext-local-custom
23:39.10kieranmullen2exten => 1,1,Goto(ivr-2|s|1)
23:39.10JTyou may need to use logging
23:39.12kieranmullen2exten => 2,1,Goto(ivr-13|s|1)  So the ext picks up but so does "im sorry that is not a valid extension"
23:40.19*** join/#asterisk ManxPower (n=manxpowe@64.246.207.186)
23:40.42kieranmullen2jt - me ?
23:40.43kieranmullen2http://clip.drlinky.com/20
23:40.50kieranmullen2I pasted it there
23:41.00kieranmullen2I am calling from fwd
23:41.10JTno, pdx5k
23:41.16kieranmullen2sorry
23:41.17pdx5kwhen i type 'core show applications' there's no meetme in there
23:41.22*** join/#asterisk sbingner (n=sam@pdpc/supporter/sustaining/sbingner)
23:44.11pdx5khow do i get meetme to load?
23:44.26JTcheck modules.conf
23:44.54JTand if there's any errors upon asterisk starting
23:45.23pdx5kmeetme isn't in the modules.conf file
23:45.38JTare modules set to autoload?
23:45.41pdx5kyes
23:47.14pdx5ki don't see any errors when it starts
23:48.18pdx5kwhat is the module name for meetme .. maybe i can load it via the console
23:49.33pdx5kwhen i do 'module show' meetme isn't there
23:51.41*** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25)
23:52.12*** join/#asterisk fab5freddy (n=vmware@bas1-montreal19-1177818910.dsl.bell.ca)
23:52.29fab5freddyWhat is a good soft client to use that can record phone calls?
23:52.29pdx5kwhy it's not there .. i have no idea .. app_meetme.c is in the apps/ dir .. so it should have built it
23:53.09kieranmullen2i dont know about good  but gizmoproject.com client also has a connect to asterisk feature and it records
23:53.12*** join/#asterisk ctaloi (n=ctaloi@nat-66-218-1-160.usadatanet.com)
23:53.32fab5freddykieranmullen2: is it in the debian repositories?
23:53.45pdx5kwhy would meetme not be compiled?
23:53.48kieranmullen2they have a linux client on their site
23:53.48JTfab5freddy: asterisk can do recording
23:54.01ctaloipdx5k - you need zaptel for meetme
23:54.06JTpdx5k: is the object file in the modules directory?
23:54.14kieranmullen2Linux Packages Avaliable
23:54.14kieranmullen2Linspire 5.0+ CNR
23:54.14kieranmullen2Binary Tarball
23:54.14kieranmullen2RPM Install Package
23:54.14kieranmullen2DEB Install Package
23:54.18ctaloianyone using multiple interfaces/ips and Asterisk ?
23:54.25shido6debian repositories.... hrmm compile man
23:54.29pdx5kctaloi: i have zaptel installed
23:54.30shido6dont be scared to compile
23:54.37shido6compile from source :)
23:54.42kieranmullen2http://www.gizmoproject.com/download-linux.html
23:55.39fab5freddykieranmullen2: thanks man
23:56.06*** join/#asterisk n4ycw (i=cspot@ip68-1-63-100.pn.at.cox.net)
23:56.16JTfab5freddy: why client side recording?
23:56.20pdx5kJT .. the object's not there
23:56.24kuku5How can I do multi lot parking ?
23:57.05fab5freddyjt: i want to reach a call centre who is recording my call so i want to record the call too
23:58.07kieranmullen2asterisk does a good job of recording on its own anyway right?
23:59.01fab5freddykieranmullen2: i am still not an expert with asterisk yet.. having some troubles getting it setup.. so i am using my soft client direct to the provider for the moment

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