irclog2html for #asterisk on 20050414

00:00.18*** join/#asterisk Weezey (WeezeyD@206.210.109.233)
00:00.55*** join/#asterisk iceyp (~icepick@202.150.105.150)
00:01.01NuggetI guess I just prefer less mainstream films.
00:01.07PTG123ah
00:01.32iceypare budgetones still the cheapest phones out there
00:01.34*** join/#asterisk Sedorox (brandon@Neptune.client.wlmsprt.pa.sed6.net)
00:01.49shmaltzhow can I find out on a blind transfer who is dialing the number?
00:01.55shmaltzI need it for billing
00:02.00shmaltzusing SIP
00:02.14Nuggetthree great films I've watched in the past month which are not on your list: amelie, the station agent, and house of sand and fog.
00:02.25shmaltzor is there any way I could find out that it is a blind transfer
00:02.47PTG123never even heard of those
00:02.57*** join/#asterisk jhowardPA (~jhoward@12.25.177.120)
00:03.12jhowardPAHello people!
00:03.19NuggetHello jhowardPA!
00:04.02jhowardPAI've got a problem with some cisco 7940's - know anything about 'em?
00:05.48Nuggetwhat's the problem?
00:06.18jhowardPAWell, when I make a call from one to another, the call is muted until I hold and resume it.
00:07.41*** join/#asterisk likwid-- (likwid@nc-67-77-138-97.dyn.sprint-hsd.net)
00:08.13*** join/#asterisk JunK-Y (~junky@modemcable174.107-81-70.mc.videotron.ca)
00:08.39Nuggetstrange, I've never heard of that.  any suspicious messages in the asterisk console?
00:08.52jhowardPANothing obvious...  :\
00:09.05jhowardPAdamn, phone call  ;)
00:10.42PTG123anyone need any servers, got a couple of AMD 1700s and some other ones gonna sell for $200-250 a piece? :)
00:10.44PTG123rackmount
00:12.11*** join/#asterisk NormAst (~NormAst@toronto-HSE-ppp3959569.sympatico.ca)
00:12.20Sedoroxspecs?
00:12.37WeezeyIf anyone's looking for a nice yet cheap headset, the Plantronics M175 is gret.
00:12.38PTG123dual 60gig ide drives, 512 memory, 2u
00:12.40Weezeygreat too.
00:13.02PTG123got some dual 1u 1gz ones too.. even got a scsi based with a dpt V raid card 2u dual 1gz but i may keep that
00:13.04Sedoroxnot bad specs
00:13.27PTG123i got a stack of servers, just trying to get rid of them first come first serve basis :)
00:13.31Sedoroxtoo bad i dun have the money.. i could use one
00:13.31PTG123from my hosting days
00:13.53Sedoroxhehe
00:14.06PTG123hrmph
00:14.14PTG123go rape an pillage your neighbors
00:14.15PTG123:)
00:14.17robl^a little off topic -- but -- http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&category=1469&item=5570085385&rd=1&tc=photo#ebayphotohosting
00:14.20Sedoroxtake payment plans? :-p
00:14.25Hogieis there like a ringing sound, or something I could use for intercom notification inside the sounds dir?  I dont see anything
00:14.31PTG123hah afraid not :) too much work
00:14.54robl^Hogie, I just use the beep.gsm :)
00:15.18*** join/#asterisk PoWeRKiLL (~PoWeRKiLL@l03m-16-26.d4.club-internet.fr)
00:15.24shmaltzhow do I add a new page to the wiki?
00:15.26Hogierobl^: I was hoping for like a 3beep, lol
00:15.40robl^Hogie, play beep 3 times
00:15.41shmaltzI know how to edit, but I want to add a new page, how do I do that?
00:15.50Sedoroxhmmm
00:15.51HogieI can't with dial...
00:16.06Sedoroxtoo bad I just got a new cell... coulda been able to do ~200...
00:16.21HogieI want it played when the receiving phone (which is setup for auto answer on that instance) picks up
00:16.49PTG123well damn you :)
00:17.12HogiePTG123: i'd buy, but im putting $700/month into flight lessons, sorry:(
00:18.12Sedoroxhey.. if your willing to hold one I could pay you in about a month.. but that isn't a option :-p
00:18.14SedoroxI gues
00:21.04PTG123heh not really trying to clean out my garage/storage
00:21.40SedoroxActually...
00:21.41Sedorox*thinks*
00:21.53Sedoroxcan I PM you?
00:22.13*** part/#asterisk xai (~pasta@cpe-70-112-17-10.austin.res.rr.com)
00:22.38PoWeRKiLL!seen HellHound
00:23.09niZonPTG123: any free stuff? :P
00:23.20Sedoroxahah
00:27.09SedoroxPTG123: if you can do paypal... I wanna chat about it.. hehe
00:28.32L|NUXhow can i set md5 password in asterisk and how can i create md5 pass ?
00:29.10JunK-YL|NUX: md5 app?
00:29.25L|NUXmd5 app ?
00:29.30JunK-Yshow applications like MD5
00:30.09L|NUXok
00:30.44L|NUXNuFW*CLI> show applications like MD5
00:30.47L|NUX<PROTECTED>
00:31.07JunK-Ywhich version?
00:31.16L|NUX1.0.7
00:31.16JunK-Yuse HEAD
00:31.56JunK-Ygoto in bug tracker and install these apps in u need just these.
00:32.06JunK-Ybut i recommend head
00:34.03*** join/#asterisk locoast (~locovox@218-153-89-200.fibertel.com.ar)
00:34.23locoasthi, anyone using simpletelecom.com?
00:34.53niZonooo iax termination
00:35.31locoastI can call continental US but cant call intl. They telling me I have to use g729 (which i dont have) or g723.1
00:35.38Sedoroxseems kinda expensive...
00:35.49locoastwhen I configure the sip part to use g723.1
00:35.58locoastasterisk says it cant make it compatible with my zapata
00:36.02locoastregular phone
00:36.15*** join/#asterisk Hackett (~chatzilla@cuscon1882.tstt.net.tt)
00:36.25locoastApr 13 21:32:33 WARNING[12267]: channel.c:2170 ast_channel_make_compatible: No path to translate from Zap/1-1(68) to SIP/simpleconnect-sip-be4f(1)
00:36.25locoastApr 13 21:32:33 WARNING[12267]: app_dial.c:1260 dial_exec_full: Had to drop call because I couldn't make Zap/1-1 compatible with SIP/simpleconnect-sip-be4f
00:36.32harryvvsimple g729 is a per port licenced codec. It also sounds the best
00:36.57locoastI'm trying to use g723.
00:36.57malbechAnyone konws a SoftSwitch Solution for a low cost ?
00:37.15locoastso I configured the SIP connection to use g723
00:37.24locoasthowever the ZAP/1-1 is not working
00:37.35file[mac]G723.1 is not officially supported in asterisk
00:37.37harryvvget zap to work before anything else
00:37.39locoastwith g723 and i dont know how to make it work
00:37.43file[mac]there is no official transcoder
00:37.55file[mac]so therefore, if asterisk has to convert between two codecs (in your case signed linear and G723) then it won't work
00:37.57locoastzap works if i use ulaw with simpletelecom
00:38.14file[mac]however, if both sides are using G723.1 then asterisk can act as a passthru and allow the audio to traverse to each
00:38.22file[mac]Thank you for choosing Asterisk. Have a peachy day!
00:38.44locoasthow do i configure ZAP to use G723.1?
00:38.51file[mac]it can't
00:39.14locoastso I can't use G723 with simpletelecom...
00:39.20file[mac]correct
00:39.26*** join/#asterisk obsidianr (~obsidianr@pcp03266395pcs.waldlk01.mi.comcast.net)
00:39.38harryvvfirst you need to know what zap is. Locoast how long have you spent learning asterisk
00:40.17harryvvfile, by chance do you know if telcos use any known form of compression that is non propriatory?
00:40.29file[mac]define 'non proprietary'
00:40.34file[mac]like give me an example of a codec
00:40.44harryvvother then the ones we know of
00:40.51harryvv7xx and gsm
00:41.02harryvvilib I think was another
00:41.02locoastwhat are you talking about?
00:41.12file[mac]ilbc? nah...
00:41.13locoast711u 711a gsm?
00:41.16harryvvlocoast its a off subject
00:41.18file[mac]G729 and G723 are ths tandard
00:41.20file[mac]er the standard
00:41.26file[mac]G723 less because of the CPU needed for it
00:41.26harryvvfor telcos?
00:41.47file[mac]well, for TDM equipment
00:41.54harryvvI am talking non voip carrier grade voice compression.
00:42.55file[mac]harryvv: your question confuses me though
00:43.05JunK-Yfile: nice
00:43.09mmlj4cluecon?
00:43.22file[mac]I'll be speaking about something or other
00:43.28PTG123they need to make a daughterboard for g729 compression
00:43.38locoastany recommendation for the best/cheap SIP provider?
00:43.45JunK-Y~seen jkerdev
00:43.47jbotJunK-Y: i haven't seen 'jkerdev'
00:43.50blitzragelocoast: I can provide SIP termination
00:43.52JunK-Ywhat's his nick again?
00:44.00harryvvfile[mac]: Do telcos like singular varizon telus use voice compression in there standard every day Centeral offices
00:44.03JunK-Y~seen jakevdev
00:44.04jboti haven't seen 'jakevdev', JunK-Y
00:44.11mmlj4blitzrage: details?
00:44.17locoastblitzrage, details?
00:44.18file[mac]harryvv: I highly doubt it
00:44.21blitzragemmlj4: well, depends what you need :)
00:44.29locoastblitzrage: any web site?
00:44.31mmlj4that's fair
00:44.43locoastSIP Providers... any other options?
00:44.56niZonblitzrage, how about termination to 204 in manitoba canada? :P
00:45.28locoastblitzrage, hurry wife calling
00:45.45blitzragelocoast: lol, sorry, the website isn't up right now I guess (I just started working for these guys :))
00:45.58blitzragelocoast: drop me an email to leif@leifmadsen.com and I can get you details if you're interested
00:46.24blitzrageI just admin the network, not really doing sales :)
00:46.57*** join/#asterisk Faithful (~Faithful@202-6-145-116.ip.adam.com.au)
00:47.05niZonthen you can sneak in some free options ;)
00:47.17blitzragefile[mac]: hehe... remember my postgres problems yesterday? Well, they are resolved, but because I kept installing overtop of the DB, it thinks there's users that aren't there :)
00:47.29blitzrageniZon: well... "I" get free options :D
00:47.38niZonlol
00:47.46niZoncan you terminate to area code 204?
00:47.49blitzragehey, anyone have 905 DID's in Oakville?
00:48.06blitzrageI can't seem to find anyone. I can 905 in Hamilton, but that does me no good
00:48.17file[mac]no DID for you!
00:48.21blitzrage*gasp*
00:48.24file[mac]and silly blitzrage for mucking with the database
00:48.32blitzragefile[mac]: well, its a test box, so no hard done
00:48.34*** join/#asterisk lilneon (~tj_r3@cuscon12298.tstt.net.tt)
00:48.35blitzrageharm*
00:48.39lilneongood night all
00:48.40blitzrageLOL
00:48.40JunK-Yblitzrage: u fixed ur psql issues?
00:48.41file[mac]hardon? :p
00:48.45file[mac]muahahaha
00:48.52blitzragefile[mac]: I knew someone was going to catch that :D
00:48.56blitzrageJunK-Y: yep! thanks for the help last night
00:49.00file[mac]yeah yeah twisted minds think alike
00:49.09JunK-Yblitzrage: y owe me 1 beer.
00:49.13blitzrageJunK-Y: it was the version problem :)
00:49.16blitzrageJunK-Y: done and done
00:49.34blitzrageJunK-Y: that's cheap. 2 hours of service for 1 beer. Can I hire you? :)
00:49.49blitzragea 2-4 ever 48 hours worth of work!
00:49.51want561or772didme want DID
00:50.03JunK-Yim like a dealer, when ya gonna be addict, i'll increase my rate :)
00:50.12blitzrageJunK-Y: LOL!
00:50.14L|NUXlol
00:50.31want561or772didhere i am. rock you like a hurricane
00:50.31jhowardPAI still can't find any details on why my 7940's are muted when a call's connected, until I hold and resume it.  Any new ideas?
00:50.47blitzragejhowardPA: that sounds like a very odd problem
00:50.54jhowardPAIt is  ;)
00:51.03obsidianranyone setup a siemens optipoint 100 advanced before? i'm having trouble
00:51.04jhowardPAI wish I knew where to start...
00:51.40blitzragejhowardPA: hrmmmm... packet traces :)
00:52.11jhowardPAYeah, that was step #2 after I verified that it is, in fact, a very odd problem.  ;)
00:52.12blitzragejhowardPA: do you have it nailed down to whether its the phone or asterisk?
00:52.27jhowardPANo sir.  I'm new to Asterisk, but I'm a quick learner.
00:52.37blitzragejhowardPA: thats what they all say! :)
00:52.46blitzragejhowardPA: explain again what is happening?
00:53.20rikstacrack pipe
00:53.38jhowardPAWhen I initiate a call between extn 200 and 204 (or any others), it rings normally.  When I pick up, both ends are fully muted until one end holds and resumes the line.
00:53.52*** join/#asterisk zotz (~zotz@24.231.32.109)
00:54.40rikstajhowardPA: pastebin your sip.conf and your phone .cnfs
00:55.08jhowardPADoesn't matter which end does the hold->resume, either will suffice to unmute both.  That's datum number one that leads me to suspect the server.
00:55.16jhowardPAok
00:55.44blitzragejhowardPA: and the relevant sections of your dialplan (extensions.conf)
00:55.54rikstayeah was just typin that
00:55.59jhowardPA[general]
00:55.59jhowardPAport = 5060           ; Port to bind to (SIP is 5060)
00:55.59jhowardPAbindaddr = 0.0.0.0    ; Address to bind to (all addresses on machine)
00:55.59jhowardPAdisallow=all
00:55.59jhowardPAallow=ulaw
00:55.59jhowardPAallow=alaw
00:56.01jhowardPAcontext = from-sip-external ; Send unknown SIP callers to this context
00:56.03jhowardPAcallerid = Unknown
00:56.08jhowardPA#include sip_nat.conf
00:56.09jhowardPA#include sip_additional.conf
00:56.12jhowardPAThere's my sip.conf
00:56.32*** join/#asterisk TheEmperor (~user@203.121.47.165)
00:56.40jhowardPAhere's one entry from sip_additional.conf (from AMP):
00:56.43jhowardPA[200]
00:56.43jhowardPAusername=200
00:56.43jhowardPAtype=friend
00:56.43jhowardPAsecret=111
00:56.43jhowardPAqualify=no
00:56.44jhowardPAport=5060
00:56.46jhowardPApickupgroup=
00:56.48jhowardPAnat=never
00:56.50jhowardPAmailbox=200@default
00:56.52jhowardPAhost=dynamic
00:56.54jhowardPAdtmfmode=rfc2833
00:56.56jhowardPAdisallow=
00:56.58jhowardPAcontext=from-internal
00:57.00jhowardPAcanreinvite=no
00:57.02jhowardPAcallgroup=
00:57.03niZonjhowardPA: http://www.pastebin.ca
00:57.04jhowardPAcallerid="Jon Howard" <200>
00:57.06jhowardPAallow=
00:57.12jhowardPAThanks, sorry.
00:57.17blitzrageeek!
00:57.18Sedorox..
00:57.23L|NUXflood :D
00:57.29Sedoroxnow see what you've gone and done!
00:57.36jhowardPAI'm retarded.  Sorry!
00:57.42Sedoroxlol
00:57.46blitzragelol
00:57.47jhowardPAStress, lack of sleep, and too much coffee.
00:57.50harryvvyea
00:57.55JunK-YjhowardPA: stop flooding and use www.pastebin.ca
00:57.55blitzragejhowardPA: I've seen dumber :)
00:58.02SedoroxI'm Sofa King We Tar Did
00:58.29*** join/#asterisk zilas (~1@adsl-158-98-233.mia.bellsouth.net)
00:58.38zilashello all
00:59.53zilasone quick Q: what function you should use when you call lets say your voicemail and you dont hear the begining?
00:59.56jhowardPAhttp://pastebin.ca/9511
01:01.11jhowardPAFlood-free!  :D
01:01.41SedoroxThis time....
01:01.46*** join/#asterisk PBXtech (~nik@70-58-41-173.slkc.qwest.net)
01:02.00jhowardPAFirst hit's free, gotta pay me for more.
01:02.06Sedoroxlol
01:02.43PBXtechStupid T1 cards
01:03.08L|NUXlol
01:03.18JunK-YPBXtech: which card?
01:03.29PBXtechquad 3.3 thinger
01:03.58PBXtechdoesnt play to nicely with ESI telephone system for some reason
01:04.37L|NUXsupport@digium.com
01:05.01PBXtechyea going to call support at ESI first
01:06.00*** join/#asterisk ManxPower (~eric@adsl-35-236-60.msy.bellsouth.net)
01:06.10ManxPower~docs
01:06.11jbotDocumentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org
01:06.13ManxPower~mailinglist
01:06.15jboti heard mailinglist is Search Asterisk mailing lists by prepending site:lists.digium.com to your Google search.  Browse the mailing list archive at http://lists.digium.com/
01:06.42PBXtechjust buggin me had it on the v5 card and it was dying once per week, now its on a v3.3 card and its dying daily. time to call ESI :)
01:06.59SedoroxDoes FWD pass CID on?
01:07.03jhowardPAblitzrage: any ideas?
01:07.20file[laptop]Sedorox: yup
01:07.29Sedoroxhmmmm
01:07.33blitzragejhowardPA: looking
01:07.39rikstaback
01:07.44rikstalooking too
01:07.44jhowardPAthank you, sir
01:07.45*** join/#asterisk chaoscon (~ph33r@chaoscon.user)
01:07.59rikstapastbin.ca is so slow
01:08.05jhowardPAblitzrage: I was afraid my rampant foolery offended you off.
01:08.07jhowardPA;)
01:08.25*** join/#asterisk NewSole (~david@i216-58-44-245.avalonworks.net)
01:08.26rikstawtf
01:08.30rikstawhere is the rest of the data
01:09.02rikstajhowardPA: ?
01:09.23rikstadid you read what i asked for?
01:09.34jhowardPAriksta: which part do you need?
01:09.48slePPer.. 'slaps'
01:09.52slePPnot lapes. wtf lapes is, i don't know
01:10.08jhowardPAriksta: second phone, same as the first, with a diff set of extn numbers.
01:10.23blitzragejhowardPA: extensions.conf contexts and Asterisk console output would be handy
01:10.38jhowardPAlemme fetch that...  one sec.
01:10.51rikstajhowardPA: was asking for the sip suff and extention stuff
01:11.15rikstano such word slePP :)
01:11.20slePPdamn
01:11.25|Vulture|whats the best method to sync config files from a centralized server to individual * boxes?
01:11.32|Vulture|rsync? ssh scripting?
01:11.33rikstanfs?
01:11.50rikstaor is it remote
01:11.50blitzrageswitch =>
01:12.20|Vulture|riksta: I want to be able to push all the config files from my windows box down to all my linux * boxes
01:12.45rikstaew :) dunno, i'd probably say rsync, but ask others!
01:12.48SedoroxCID doesn't get passed through queues??
01:12.49blitzrageswitch =>
01:12.51rikstaslePP: thats just mean
01:12.55slePPi know :>
01:12.59slePPi'm bad that way
01:12.59rikstaand, i'm not jewish, fool
01:13.02slePPhow'd that project go?
01:13.05SedoroxslePP: :-p
01:13.06slePPskullcap being the top of yer head :P
01:13.08shmaltzanybody here that has app_valetparking that works with latest CVS HEAD?
01:13.09slePPSedorox: 'lo.
01:13.13Sedoroxhow be thee?
01:13.13slePPSedorox: the 15th
01:13.15blitzragepssst... I think switch => would work
01:13.16rikstai got it in yeah, hopefully shud get a good mark
01:13.19slePPnot bad. bit out of sorts today
01:13.33Sedoroxwhen you get the thingy installed?
01:13.37slePP15th
01:13.44Sedoroxnice
01:13.51slePPwe finally got confirmation today
01:13.56Sedoroxkewl
01:13.58rikstaoohh, that skullcap :)
01:14.10Sedoroxthe other place is finally working on the number... I'm thinking about just telling them to forget it and refund me.. hehe
01:14.18slePPthat'd be a good idea :>
01:14.21jhowardPAriksta: here's my extensions.conf http://pastebin.ca/9512
01:14.36slePPi should give a prize to the 10,000th pastebin poster
01:14.44slePPi should remember to ship the prize for the draw winner, too
01:14.53blitzrageslePP: need my address right?
01:14.56rikstayou'll be there forever slePP ..it loads too slow ;)))))
01:14.57Sedoroxahahah
01:15.08slePPriksta: 8ms for me ;>
01:15.14rikstabah
01:15.20slePPblitzrage: heh. you didn't win :P
01:15.30blitzrageslePP: well... I got the email that said I did
01:15.38rikstaping times are ok across the pond here
01:15.42slePPi didn't send an e-mail. i sent a message on MSN :>
01:15.43rikstabut your server responds so slowly
01:15.48rikstartt min/avg/max/mdev = 141.862/145.076/190.753/9.387 ms
01:15.49blitzrageslePP: it said, "YOU BIG WINNAH!"
01:15.53slePPsee, that's weird, riksta. i dunno why..
01:16.00slePPblitzrage: y'sure it wasn't for v14gr4?
01:16.06blitzrageslePP: lol
01:16.17jhowardPAriksta: the sip.conf is http://pastebin.ca/9513
01:16.19rikstaslePP: i might try without privoxy, just incase
01:16.25blitzrageslePP: I was impressed with the use of ASCII art in a spam message today
01:16.35slePPreally? pastebin it :>
01:16.40rikstajhowardPA: ill have a look
01:16.50PBXtechasterisk@home has a bug if you call a phone that isnt registered it goes into a loop
01:17.01slePPriksta: it may possibly be the javascript bit.. i could drop that and see if it changes anything
01:17.10rikstajhowardPA: are you using any macros currently
01:17.17rikstaslePP: yeah would you like to try?
01:17.19*** join/#asterisk rrk (~chatzilla@rrcs-67-53-9-175.west.biz.rr.com)
01:17.20slePPk
01:17.36slePPriksta: try now
01:17.39rikstafirefox just sits "Waiting for pastebin.ca" tho
01:17.43blitzrageslePP: damn... after I saw it marked as spam and its gone
01:17.48rikstasame thing
01:17.49jhowardPAI'm using Asterisk@Home, so it's pretty likely.  I'm just trying to get my hands dirty with it, then move to a clean setup on debian.
01:17.51slePPblitzrage: :<
01:17.58rikstalemme put your pastebin.ca in the list of non proxy hosts
01:18.01Sedoroxriksta: I've had that problem before with .ca
01:18.05jhowardPAriksta: sorry, that was meant for you
01:18.08slePPsome .ca lookups can be damn slow
01:18.11slePPfor some odd reason
01:18.12blitzragepastebin.ca slooooow for me
01:18.12rrklot of luck with the clean debian
01:18.17rikstasee
01:18.20riksta:)
01:18.21jhowardPArrk: whaddya mean?
01:18.25blitzrageslePP: yep, thats basically what I'm seeing
01:18.28slePPsomeone run a dnstrace on that
01:18.39slePPmy lookup is 100ms
01:18.40rrki tried for three days to move it to debian
01:18.53rrkwith no sucess
01:18.53rikstajhowardPA: i would try without the macros, first of all
01:19.07jhowardPArrk: was it an asterisk versioning problem, or a debian problem?
01:19.17rrki assume you are interested in keeping amp
01:19.29rikstaslePP: seems to be faster now
01:19.29jhowardPAriksta: you got it.  I'll see what happens.
01:19.34file[laptop]ooh la la
01:19.36jhowardPArrk: yeah, I'd like to.
01:19.36*** join/#asterisk luke-jr_ (~luke-jr@207.192.219.246)
01:19.38slePPriksta: i didn't change anything :>
01:19.40jhowardPArrk: won't build?
01:19.49rikstaslePP: i just put pastebin.ca in firefox's no proxy list
01:19.51rikstacuz i use privoxy
01:19.53rikstalemme take it out
01:19.55rikstaand try again
01:20.14rikstaslooow
01:20.15rrkproblems with spandsp patch for the fax and some of the config files
01:20.17rikstamust be privoxy man
01:20.31Sedoroxthe internet here at school has been sucking balls lately
01:20.31rikstastrange
01:20.48rikstayep it's privoxy, slePP , ill look at its log file
01:20.51slePPk
01:20.59slePPmaybe revdns?
01:21.07rikstaApr 14 02:18:02 Privoxy(-1235059792) Request: pastebin.ca/ads/freestyle-1.gif crunch!
01:21.26rikstaApr 14 02:14:16 Privoxy(-1209631824) Request: config.privoxy.org/send-stylesheet crunch!
01:21.28slePPwtf is 'crunch'?
01:21.46rikstait takes out your gay ads
01:21.51riksta:)
01:21.54slePP:>
01:21.57slePPthose ads keep the site alive
01:22.24rrkthe zaptem would not build right with the 2.4.27 stable but would with 2.6.10 and 11.7
01:22.25Sedoroxdoes anyone know if queue's don't pass on CID... seems when I call in.. and into the queue.. the agent doesn't get the cid info
01:22.46jhowardPAriksta: I can't see how to del those macros without basically moving to a fresh install of asterisk.  Is that the goal?
01:22.47zilashow can I eliminate this gap when I call voicemail from sip phone and before I hear something * server already starts talking so I dont hear like first 5 seconds
01:23.22zilasno maybe 2 seconds
01:23.22rikstajhowardPA: just get the incoming context to Dial() the phone
01:23.28rikstadirectly, for testing
01:23.49PTG123<PROTECTED>
01:23.58Sedoroxlol
01:23.58PTG123damn space :)
01:24.07jhowardPAriksta: I said I was quick, but I also said I was new to Asterisk.  Could you elaborate a bit for me?  ;)
01:24.13SedoroxPTG123: if you can take paypal for one server.. I'm willing to talk :-p
01:24.25jhowardPASorry, I don't mean to be a goon, but I've had insufficient study.
01:24.30PTG123hah paypal is prefered :)
01:24.36PTG123msg me :)
01:24.37Sedoroxcan I pm ya?
01:24.38Sedoroxaha
01:24.46rikstajhowardPA: one second let me read the config a bit more
01:24.47PTG123and you better hurry people are emailing me like mad on the list :)
01:24.48PTG123yah pm me
01:24.53Sedoroxahah
01:25.07jhowardPAriksta: Thanks  :)
01:25.59L|NUXslePP : can you tell me how can i create md5 password for *
01:26.12file[laptop]PTG123 is gonna be rich
01:26.16Sedoroxlol
01:26.34PTG123hah
01:26.43*** join/#asterisk ta[i]nted (~tainted@adsl-69-108-114-226.dsl.irvnca.pacbell.net)
01:27.11SedoroxI need a decent system to either colo (eventually...) or just to run for *
01:27.26file[laptop]I run asterisk on a wide variety of things really
01:27.36rikstaL|NUX: can you not just use perl's crypt() ?
01:27.47JunK-Ylinux: i told u to use the application.
01:27.48file[laptop]Celerons, Xeons, AMDs, Geodes
01:28.05L|NUXJunK-Y : but there is not listed MD5 :(
01:28.11L|NUXin asterisk application
01:28.35JunK-Yuse head
01:28.40L|NUXi used
01:28.43L|NUXbut same
01:28.48*** join/#asterisk bah (048830696@AC8C1316.ipt.aol.com)
01:28.49L|NUXrecompiled
01:28.51L|NUXbut same problem
01:28.51JunK-Yhuh?
01:28.54L|NUXreally
01:28.54L|NUXwait
01:29.03Sedoroxhehe.. right now I have it on a celery 333, which doesn't have a working CPU fan.. ahah
01:29.08L|NUXhttp://digium-cvs.netmonks.ca/viewcvs.cgi/asterisk/md5.c?rev=1.11&view=auto
01:29.13JunK-Ydebian*CLI> show applications like md5
01:29.13JunK-Y<PROTECTED>
01:29.13JunK-Y<PROTECTED>
01:29.13JunK-Y<PROTECTED>
01:29.13JunK-Y<PROTECTED>
01:29.18SedoroxI'm asking for a death sentance :-p
01:29.32JunK-Yis that what u want?
01:29.43niZonSedorox: do you smell burning yet? :P
01:30.00rikstaperl -e 'print crypt("password", "salt"),"\n"'      ?
01:30.07L|NUXwait
01:30.14rikstais that md5?
01:30.18Sedoroxnope :-p
01:30.24L|NUXJunK-Y : but not showing in my *
01:30.28L|NUXi recompiled
01:30.50rikstaahh ok, this is md5 http://sial.org/howto/perl/password-crypt/
01:30.57JunK-Yya've a app_md5.c in ur /usr/src/asterisk/apps/ ?
01:31.10drumkillait's only in cve head
01:31.12drumkillacvs*
01:31.24JunK-Yyes, i already told him that.
01:31.28L|NUXhmm
01:31.29L|NUXwait
01:31.37L|NUXcan you give me a link ?
01:31.41JunK-Yls -l /usr/src/asterisk/apps/app_md5.c
01:31.52JunK-Ytype that in ur shell
01:31.56L|NUXk
01:32.05L|NUX[root@NuFW asterisk]# ls -l apps/app_md5.c
01:32.05L|NUXls: apps/app_md5.c: No such file or directory
01:32.07L|NUXO_o
01:32.14L|NUXi was doing with wrong file :$
01:32.15L|NUXshit
01:32.15L|NUXwait
01:37.50harryvvyou would be better to just wait and not type anything that does not fill up the window..one ..line ...at ...a ..time.
01:38.17rikstaslePP: that privoxy crap is weirdddd
01:38.30*** join/#asterisk bkw_ (~brian@bkw.developer.and.friend.of.asterisk)
01:38.30*** mode/#asterisk [+o bkw_] by ChanServ
01:39.14rikstabeautiful hostname bkw_ ... lol
01:41.13|Vulture|is there a way to write an SSH script that will log into a server and execute commands, such as put and say restart *?
01:41.49JunK-Yrsh ?
01:42.02JunK-Ythat's why rsh (remote shell) is there.
01:42.16|Vulture|hmmm
01:42.32decor you can setup SSH authorized keys and just do 'ssh hostname sudo /etc/init.d/asterisk restart'
01:42.53|Vulture|never used that
01:43.14|Vulture|I want to run these scripts from within windows...
01:43.28|Vulture|I tried using putty's pscp
01:43.44|Vulture|it sends the files fine, but duno if it will let me execute commands
01:43.56malbechAnyone konws a SoftSwitch Solution for a low cost ?
01:44.09mmlj4pscp is putty's scp client, useful mainly for sending files, as noted
01:44.21*** join/#asterisk mog_home (~mog_home@146.229.178.196)
01:44.25mmlj4putty itself can let you log in and run commands
01:44.45mmlj4you might be able to do something from windows itself if you install cygwin
01:44.47|Vulture|mmlj4: but I am trying to automated it all so I just have to click a batch file etc. and it will all do it
01:45.00mmlj4yeah, cywin.com, go install that
01:45.18*** part/#asterisk jhowardPA (~jhoward@12.25.177.120)
01:45.42mmlj4your "batch files" are going to be regular shell scripts... so you can do 'ssh hostname sudo /etc/init.d/asterisk restart' by clicking an icon, for example
01:46.08|Vulture|okay
01:46.17mmlj4hint; do a default cygwin install, then run the setup again to add ssh, etc.
01:46.30mmlj4what country are you in, |Vulture| ?
01:46.38|Vulture|USA
01:46.57mmlj4ok... choose either the nasa or kernel.org ftp sites, to do the cygwin install
01:49.46|Vulture|okay all done going to try and get ssh now
01:51.08*** join/#asterisk Dovid (~hirisk@pool-138-89-170-224.mad.east.verizon.net)
01:51.19harryvvis iax.cc down?
01:51.26harryvvIt shows me as not registered
01:53.13niZonmake a call
01:54.17harryvvi did
01:54.36harryvvsaid == Everyone is busy/congested at this time (1:0/1/0)
01:54.42niZonweird
01:54.49harryvvdid a iax2 show registry and shows me as not registered
01:55.06harryvvthis is of course not the first time thay have had problems.
01:55.12Dovidhi
01:55.20niZonthey're slow at getting DIDs
01:55.33*** join/#asterisk tessier (~treed@203.210.216.187)
01:55.35harryvvIm just using there outbound pstn
01:55.48Dovidi have x-lite. what ports do i need to open on my firewall at my home to access it ?
01:56.20harryvvdoe! Apr 13 18:53:04 NOTICE[7098]: chan_iax2.c:7090 iax2_poke_noanswer: Peer 'sixtel' is now UNREACHABLE! Time: 2892
01:56.50niZonharryvv: drop them an IM
01:56.55harryvv:)
01:56.55L|NUXDovid : 5060, 9999-20001
01:57.11harryvvnoZon, what is there IM
01:57.16Hogiebleh blah
01:57.44niZonharryvv: sixtel9 on aim, and msn@sixtel.net for msn (I think)
01:57.59DovidL|nux: Anyway that i can set it work with other ports ?
01:58.28L|NUXhmm
01:58.37L|NUXthink so
01:58.38L|NUXyou can
01:58.50Dovidhow ?
01:58.57L|NUX<PROTECTED>
01:59.03L|NUXand sip.conf
01:59.50harryvv- Registered IAX2 to '205.234.133.203', who sees us as 24.81.64.126:4569
01:59.59harryvvlooks like thay are now up
02:00.23ta[i]ntedharryvv how is their service
02:00.46ta[i]ntedbetter or worse than BV
02:01.01harryvvactually im not registered
02:01.20DovidL|nux: Thanks
02:01.32L|NUXDovid : Welcome
02:04.55*** join/#asterisk tawker (tawker@d154-5-108-14.bchsia.telus.net)
02:04.59tawkerhi
02:05.23L|NUXhey
02:05.59*** join/#asterisk bugbot (~bugbot@d141-234-145.home.cgocable.net)
02:06.03tawkeri was looking in the doc project
02:06.05blitzragewb bugbot
02:06.16tawkerand I can't seem to find anything close to a voicemail map
02:07.31tawkeri've also gone through voip-info
02:09.52tawkerany ideas
02:11.13*** part/#asterisk lilneon (~tj_r3@cuscon12298.tstt.net.tt)
02:11.18tawkerhi
02:11.23*** part/#asterisk JunK-Y (~junky@modemcable174.107-81-70.mc.videotron.ca)
02:11.55tawkerok, maybe this is a bad time
02:11.59tawkeri'll try again later
02:12.00ta[i]ntedyea
02:12.07ta[i]ntedit's rather idle bout now
02:12.15tawkeryes, for the number of people in here
02:12.21tawkeri'd expect it to be a lot more lively
02:12.26ta[i]ntedit's weird
02:12.39ta[i]ntedi think most of them are pulling their hair out configuring stuff
02:13.18tawkerok, anyways, i'll scram for a bit
02:13.24tawkertalk to you later
02:13.31ta[i]ntedgood times
02:13.42QwellHave you guys ever seen a problem with an Avaya PBX (only when you call TO an 8xx DID) giving a congestion signal if you call FROM an 8xx DID?
02:14.24ta[i]ntedcarrier issue?
02:14.27QwellIf I change my CIDNum to a non 8xx number, it works great
02:14.40Qwellits kinda weird...
02:14.49ta[i]ntedu sure its pbx
02:15.01ta[i]ntedcould be some kind of call blocking
02:15.18Qwellyeah, I can call the direct number to people in there, and if I spoof CID, it works fine
02:15.25ta[i]ntedis ORIG 8xx PRI?
02:15.34Qwellits from nufone
02:15.40ta[i]ntedlol
02:15.43ta[i]ntedcase closed
02:15.50QwellIts not a nufone problem.
02:15.57QwellI can call other 8xx DIDs fine, spoofed or not
02:16.00*** join/#asterisk iq (~IQ@70-57-182-73.omah.qwest.net)
02:16.07iqhi
02:16.15ta[i]ntedprobably carrier issue
02:16.20ta[i]ntedhow are u spoofing CID
02:16.26QwellSetCIDNum, heh
02:16.33ta[i]ntedwhat kind of line
02:16.34ta[i]ntedPRI?
02:16.35harryvvlooks like one of my work places got a suspected anthrax letter that was opened up.
02:16.47Qwellta[i]nted: the number I'm calling?
02:16.51ta[i]ntedno
02:16.57ta[i]ntedthe line u are spoofing from
02:17.06Qwellta[i]nted: my nufone account
02:17.13Qwellmaybe "spoofing" is the wrong word?
02:17.16harryvv<PROTECTED>
02:17.16harryvvApr 13 19:13:55 NOTICE[7098]: chan_iax2.c:7090 iax2_poke_noanswer: Peer 'sixtel' is now UNREACHABLE! Time: 729
02:17.23harryvvtonights not the nite to use iax.cc
02:17.26ta[i]ntedu can set callerid to whatever u want with nufone?
02:17.29Qwellta[i]nted: yeah
02:17.36ta[i]ntedwhat a bunch of jackasses
02:17.57sivanaI don't think the accountcode entry works from iax.conf.. is that right?
02:18.00*** join/#asterisk alegh (~ag10@OL217-17.fibertel.com.ar)
02:18.07Qwellta[i]nted: never seen such an issue though?
02:18.13ta[i]ntedQwell i have.
02:18.22ta[i]ntedbut it was carrier issue (XO specifically)
02:18.46ta[i]ntedtell nufone u have this issue
02:18.49Qwellis it possible that Avaya would be sending back a fuckoff signal, just because of my CID?
02:18.52ta[i]ntedthey can look into their profider
02:18.56*** join/#asterisk Treemole (Treemole@evvlinlwt-nas-07-s126.cinergycom.net)
02:19.46ta[i]ntedtry spoofing the 8xx from you nufone line
02:19.53Qwellthe one I'm calling?
02:19.55ta[i]ntedand calling the avaya box
02:19.57Qwellhmm
02:20.10Qwelllike, SetCIDNum(${EXTEN}), ?
02:20.29sivanaI don't think Nufone accepts CIDNum for outbound toll-free
02:20.50sivanasorry.. toll-free CIDnum
02:20.53Qwellwell, it works if I change 800 to 951
02:21.05sivanayou can't have toll-free CIDNum
02:21.24QwellThen what am I gonna use?  heh
02:21.28ta[i]ntedQwell are other 8xx able to call the avaya pbx
02:21.46Qwellta[i]nted: not sure.  lemme try changing it to its own number
02:21.55ta[i]ntedi think u need to try all the possibilities
02:22.05niZoniax.cc let me set my cidnum to 1337 :P
02:22.33Qwellheh, I confused the remote provider, or something
02:22.39Qwell"We're sorry, your call could not go through."
02:23.01Qwelloops, actually...
02:23.20Qwellyeah, fast busy still
02:23.22*** join/#asterisk jbAU (~johnblade@61.8.110.41)
02:23.46jbAUCan anyone recommend a USA IAX service for PSTN calls ?
02:23.57QwelljbAU: nufone is great
02:24.48*** join/#asterisk shido6 (~greg@d57-87-253.home.cgocable.net)
02:24.51shido6boink
02:25.00Qwellshido6: ;]
02:25.11Qwellshido6: Any idea about the above?
02:25.17jbAUQwell: hmm can't seem to find the url for nuphone
02:25.25QwelljbAU: Fone...nufone.net
02:25.37Qwellsays they aren't accepting orders...but talk to my man shido here
02:25.56Qwelloh, hmm, missed the join...right
02:26.01jbAUahh ok
02:26.16jbAUi wonder if i click the 'let me in' button :)
02:26.24QwelljbAU: Thats how customers login
02:26.56niZonshido6 = nufone employee?
02:27.28QwelljbAU: basically, 2c/minute outgoing, and you can either get a MI DID for $8-9/month(I always forget), or a us48 tollfree, for 2c/min incoming
02:27.43QwellniZon: something like that :p
02:27.58niZonok :P
02:28.06jbAUQwell: that's pretty damn schweet
02:28.12QwelljbAU: yeah, nufone is great
02:28.42jbAUQwell: excellent... now if only they were available for business. :)
02:29.05QwelljbAU: if shido here doesn't respond soon (that might've been sent automatically when he joined), shoot an email to greg@nufone.net
02:29.10QwellHe should be able to care of you
02:29.24Qwelltell him Qwell sent you...
02:29.25jbAUexcellent - i'll send him a message
02:29.44jbAUsounds fair. :)
02:30.18QwelljbAU: I think he said that if people can use paypal for the first order, he can start an account for them
02:30.34jbAUQwell: ahh ok - that shouldn't be a problem
02:30.41Qwellyeah...it rarely is
02:31.01jbAUQwell: i don't expect to have too much use at the moment, as i don't have -that- much business in the states
02:31.19QwelljbAU: well, nufone also has good international rates
02:31.27tzangeryeah nufone is not taking new customers at the moment but if you msg shido6 he may be bale to set you up manually if you've got a paypal acct
02:31.45*** join/#asterisk Dovid (~hirisk@pool-138-89-169-188.mad.east.verizon.net)
02:31.53jbAUshido6: can you help me out with a nufone account please ?
02:32.03jbAUshido6: qwell sent me :)
02:32.04Dovidanyone know any sip phone that u can specify what ports u want it to use ?
02:32.10QwelljbAU: might be better off sending him an email
02:32.16jbAUQwell: indeed
02:35.35decis there some sort of no-audio timeout on IAX? on an IAX connection i'm testing, there's only audio going one way... and the connection keeps dropping out
02:37.42*** join/#asterisk MrBelvedr (~tt@ip68-227-209-110.dc.dc.cox.net)
02:38.46*** join/#asterisk riksta (~rick@81-178-248-131.dsl.pipex.com)
02:39.05Dovidanyone know of a sip phone that i can set what ports it should use ?
02:39.55*** join/#asterisk brc_ (~brian@brc.base.supporter.pdpc)
02:41.38*** join/#asterisk shmaltz (~chatzilla@69.28.255.210)
02:42.01file[laptop]...achoo
02:42.23Qwellfile[laptop]: cover your mouth
02:42.29shmaltzanybody here has problems with quad span interrupts?
02:42.31file[laptop]but then you won't get infected
02:43.11Qwellfile[laptop]: Make my work fix their PBX, would ya?
02:43.26file[laptop]Qwell: what?
02:43.29*** part/#asterisk Pinhole (~nuxi@lin-dsl-static-206-222-194-115.inetnebr.com)
02:43.33Qwelldunno, its b0rked
02:43.39file[laptop]that made absolutely no sense
02:43.45QwellIf my CIDNum is 8xx, I get fast busy.  Otherwise, its fine
02:44.07Qwelland its only the 800 DIDs at my work...if I direct dial to somebody, it works fine
02:44.27file[laptop]oh depending on the provider and stuff if you set your caller id number to a toll-free number it may or may not work
02:44.39Qwellmeh...thats my only phone number.  heh
02:44.40file[laptop]the provider where the number you're calling...
02:44.57file[laptop]it's just the way it is, I've found it out through my own... testing and junk
02:45.13Qwellfile[laptop]: but...the npanxx numbers at my work are off the same provider
02:45.35file[laptop]usually when you call toll-free numbers that it happens too...
02:45.45Qwellahh
02:45.48Qwellwell, hell...heh
02:45.54Hogieis digium in SC?
02:46.00file[laptop]Hogie: Huntsville, Alabama
02:46.06*** join/#asterisk PBXtech (~nik@70-58-41-173.slkc.qwest.net)
02:46.12QwellI'm gonna have to spoof my CIDNum then...Simpsons 939 number, here I come
02:46.21file[laptop]Qwell: tee he he
02:47.23HogieIm trying to figure out where my vendor had my card drop shipped from
02:48.14*** join/#asterisk mountie (~mountie@CPEdeaddeaddead-CM000a739acaa4.cpe.net.cable.rogers.com)
02:49.18robl^Hogie, India ;)
02:49.38Hogielol, its coming from SC
02:49.38MrBelvedri am trying to test the performance on my system. Can somebody tell me a process that I can run that will consume all my ram and force the system to start swapping?
02:50.05MrBelvedri only have 64 mb of ram so it should not be hard
02:50.50robl^MrBelvedr, Windows :)
02:50.51harryvvanyone here make there extensons failover from sip/iax service to zap? I have tried it before and did not seem to want to advanced to the next zap priority and make the call. Anyone done this before? seems simple to setup but only if it works.
02:51.38harryvvsixtel is now up!
02:51.55tzangerharryvv: sure
02:52.10tzangerif you get CHANUNAVAIL or CONGESTION, dial out the zap
02:52.58Hogieharryvv: N+1 on the priority of the extension after a Dial has worked for me
02:53.39harryvvtzanger thats the idea
02:53.56tzangerit's not just the idea, it's how it's done :-)
02:54.26harryvv:) its more of a wife bitching thing :)
02:54.27harryvvheheh
02:55.48tzangerharryvv: http://pastebin.ca/9517
02:57.15PBXtechanyone have one of those pulver wifi phones?
02:57.15Hogietzanger: does that only work on cvs, not stable?  Ive not seen the -CANCEL stuff and such so far
02:58.01tzangeryou need to use extension priority numbers rather than n but it should work otherwise
02:58.19tzangerHogie: if you really needed to you could use GotoIf()s to work around it but I think it works just fine on standard
02:58.22tzangerPBXtech: yeah they suck
02:59.06Hogieany of the wifi phones good tzanger?  or is it too early?
02:59.19tzangerHogie: I've only tried the pulver one but the others look like ass
02:59.30Nuggetthe pulver/zyxel ones are horrid.
02:59.32tzangerI have heard of one by an egyptian company and another from mexico that look better
02:59.37tzangerand might work better
02:59.40Nuggetvaewyn has the hitachi one and seems to really like it
02:59.40docelmoAnyone in here from the NYC or 2 hour north area?
02:59.57tzangerI'm ~2.5hrs from NY but not NYC
02:59.58mishehu"talk like an egyptian..."
03:00.02tzangerhehe
03:00.07HogieIm trying to find one for work... Im never at my desk, and that would be nice with the new phone system
03:00.19PBXtechwhats a good wimax phone then?
03:00.38tzangeruh PBXtech we just said we don't really know of any
03:00.39*** join/#asterisk bugbot (~bugbot@d141-234-145.home.cgocable.net)
03:00.53PBXtechoh
03:01.24*** join/#asterisk ScythelX (Fleb@pc-24-181-176-181.sbi.ct.charter.com)
03:02.32*** join/#asterisk Poemius (~poemius@adsl-70-48-192-81.adsl.iam.net.ma)
03:02.38Poemiushowdy everyone :)
03:02.52ScythelXhello all - can someone take a look at this - im having a problem with incoming calls only - its very choppy but outgoing calls work fine - i have a cisco 79xx phone and using nufone as my upstream provider- i have a cable modem with an upload speed of 384 / http://pastebin.ca/9380
03:03.36docelmoScythelX, whats your network topology look like?
03:03.52PoemiusScythelX : not sure if it is an issue... on some configs, X causes choppiness, try turning off X
03:04.01ScythelXnot using X
03:04.16ScythelXthe * box is at my house where the cable modem is and the cisco phone is located at my friends house
03:04.20ScythelXwhom also has a cable modem
03:04.50Poemiusshouldn't be an issue, maybe latency
03:04.51docelmoSo your cable -> router -> *?
03:04.59*** join/#asterisk libpcp (libpcp@210.16.20.5)
03:05.31ScythelXoutgoing call -> cisco phone -> router -> my house ( * box ) -> nufone to pstn
03:05.43ScythelXthe * box is directly connected to the internet
03:05.45ScythelXno router
03:06.23ScythelXi meant incoming call above
03:06.34ScythelXoutgoing calls are crystal clear
03:07.06docelmoWhat codec, dial plan are you using?   Do you have a default conext setup/
03:07.24ScythelXdid you look at that pastebin
03:07.28docelmoyes
03:07.38ScythelXits supposed to be using gsm
03:07.45ScythelXbut then again the cisco phones dont support gsm
03:08.27Nuggetyeah, you might really want to consider buying a couple g729 licenses.
03:08.28docelmoChange your codecs to g729 or ULAW that may fix the problem.   Is it transcoding?
03:08.34Moonwickfor what they cost, you'd expect them to be able to display full motion video on those ginormous LCDs
03:09.39docelmotrue..
03:09.41Poemiusor maybe do the dishes automatically :)
03:10.06docelmo10 buks..  I just told my company I needed to order 100 and they about died when I told them the cost
03:11.46Poemiuswell it depends if it is intra... ilbc/ speex might do the trick
03:12.31Poemiusor you may not need 100, it is the number of active lines at one time
03:12.45docelmoI dunno..  My network is about to go live and I am doing g729 only well for the most part..  I have to terminate to my down stream provider at ULAW..
03:13.02docelmoI run at any given point in time about 500ish
03:13.15ScythelXwell right now hte phone is set to use g711ulaw
03:13.25docelmoSet * to do the same
03:13.31Poemius:) much bigger than my tiny weeny setup :)... but I guess it makes sense :)
03:13.40ScythelXhmm ok
03:13.50docelmowith NuFone
03:13.56ScythelXthe config in ast is g711ulaw
03:13.57harryvvjust tested my call its not failing over
03:13.57Nuggetyou're not allowed to call it a "tiny weeny" setup unless your asterisk server is a mac mini.  :)
03:14.01ScythelXor just g711
03:14.11docelmog711
03:14.13Poemiusor a linksys :)
03:14.14docelmoor ULAW
03:14.28docelmoWell Im running Dual 248 Opterons w/ 4GB ram
03:14.30ScythelXdoes it matter which one
03:14.34ScythelXg711 or ULAW
03:14.39docelmoThey are the same
03:14.41ScythelXokie
03:14.59PatrickDKthey are almost the same
03:15.04PatrickDKg711 can also be alaw
03:15.10docelmoThis is true
03:15.18docelmoWhich is a fricken Euro standard..  :(
03:15.39PatrickDKalaw is suppost o give better sound quality
03:15.44Poemiusbut compared to the price of a setup this size though, the price of the licences should not be that much
03:15.48PatrickDKpersonally though, I do perfer the metric sytem
03:15.50PatrickDKsystem
03:16.01docelmoI run ALAW from Australia and it doesnt sound any better than the ULAW I use up here
03:16.13Poemiusit's true a decimal system makes a lot of things easier
03:16.30PatrickDKdocelmo, suppost to sound better, in mathmatical terms it does
03:16.34docelmoAnyone know of any good A-Z providers?
03:16.39docelmoheheh :)
03:16.40PatrickDKin actually use, well, you never know how that works out
03:16.45Qwelldocelmo: A-Z?
03:16.53Poemiustermination
03:16.55docelmoall international countries
03:17.09Qwellnufone doesn't?
03:17.09docelmoI have domestic..  Just bought 2.5 DS3's..
03:17.16docelmoI dont like nufone
03:17.20Poemiusat my size voipjet seems to be the cheapest
03:17.27QwellWhats wrong with nufone?
03:17.32docelmoI pled the 5th
03:17.51PoemiusI haven't tried them, but voipjet looks cool
03:17.59Qwelldocelmo: nufone is great
03:18.01docelmoPoemius, gimme a month or so.. You will see things MUCH cheaper..
03:18.02Poemiusat least from a cost point of view
03:18.37Poemius:) docelmo, I am definitely interested :)
03:19.18docelmo.02c for Domestic 48?   Wicked expensive..  I was looking at charging around 1c or less depending on volume
03:19.38docelmoAnd I will have origination in 15 markets about 300 rate centers
03:19.40Poemiusdicelmo I think voipjet is at 1.4
03:20.02docelmoya.. I use them now until my network is finished
03:20.14docelmoIm building at 60 hudson right now
03:20.49Poemiussounds very interesting :)
03:21.02docelmoPoemius where do you live?
03:21.11Poemius:) Casablanca, Morocco
03:21.23PoemiusI reckon, it's a bit late here (3 am)
03:21.41docelmoohh
03:21.55docelmoNo DID's there YET..  But I have 750 Rate Centers
03:22.04docelmoIn the states
03:22.16Poemiuslol :) I do need a DID in CT
03:22.19mishehuif only I could afford a single ds3...
03:22.24PoemiusI currently use broadvoice
03:22.42niZondocelmo: get some in canada :D
03:22.44Poemiusbut it's not flexible enough in my own taste
03:23.34Poemiusnot really sure, but I get the impression it is not as crowded in canada
03:23.42Poemius(voip competition wise)
03:23.42docelmoClosest I have to CT is MASS
03:24.20docelmoniZon, CA is next on my list. Working some deals up there. Trying to find a provider that will do SS7 and not rape me
03:24.30niZonlol cool
03:24.51Poemius:) that's one of the downsides of no competition
03:25.16docelmoDID's are gonna run about 9.50 for unlimited..  Once I build up the network things will get MUCH cheaper..
03:25.20Poemiushere in morocco, no competition, they charge 30 cents a minute to europe or US
03:25.39ScythelXdocelmo: who are you leasing your ds3 from
03:25.47docelmoMe to know.. :)
03:25.50niZondocelmo: sounds good
03:25.57ScythelXor is it in a telco hotel
03:26.04docelmoHotel..  60 Hudson
03:26.05Poemiusdocelmo: 9.50 for incoming is about twice as most
03:26.10ScythelXfigured
03:26.18docelmoPeo, unlimited?
03:26.25docelmoNot metered
03:26.40docelmoVoice Pulse is 11 something
03:26.42Poemiusah ok... it's true it's more interesting for big guys
03:27.03docelmoAND I offer LNP
03:27.13docelmoAlong with Caller ID Name
03:27.17ScythelXxo communications is the cheapest i think in 60 hudson
03:27.26Poemiusdefinitely interesting
03:27.29ScythelXalthough they suck balls
03:27.42Poemiusalthough u should have an option later for standard users
03:27.51docelmothe company I work for now uses them.. I have no complaints..  They do what I want when I need it.
03:28.23docelmoPoe once I am established prices will change.. Right now I need to cover costs..
03:28.25Poemiusit definitely sounds like a big and great project
03:29.01docelmoAnd the 1st DID is gonne be around 9.50 the additional's will only be $2.50 more
03:29.07Poemiusagreed... it's a good idea... plus if you target big companies that yet don't have voip,
03:29.34Poemiusah ok, definitely more interesting with that scheme
03:29.43MrBelvedrwhenver I startup asterisk it sometimes gets to the CLI, sometimes it does not. but it always says 'Killed.' Why is this happening?
03:29.47ScythelXmost big companies would have their own data lines
03:30.06jbAUMrBelvedr: try starting up with asterisk -vvvgc to see if there's an error
03:30.13MrBelvedrk
03:30.26Poemius:) ScythelX, not talking about fortune 500
03:30.27ScythelXie: american eagle
03:30.32ScythelXuses all voip
03:30.57docelmoIm targeting larger VOIP carriers as clients for termination or origination and small med business along with "bring your own gear" type of accounts and of course the end users
03:31.53Poemiusdocelmo: definitely with multiple dids, your offer becomes much more interesting
03:33.00docelmoIm going to be competitive in the market..  Like I said..  I am targeting bigger clients like Million Minutes a month +
03:33.12docelmoBut for the smaller guys a good service also
03:33.24ScythelXdo you have any clients
03:33.28*** join/#asterisk jakepdev (~jakepdev@pool-68-236-58-19.phil.east.verizon.net)
03:33.34ScythelXor are you just shelling out money for your ds3
03:33.54docelmoI have quite a few.  I am not paying for anything yet.  Still waiting to move into my cage
03:34.38jakepdevhello everyone
03:35.08Poemius:) as we all know, there is a large potential :)... the key is targeting things right :)
03:35.47docelmoWell I am using * as my backend highly modified
03:35.54Poemiusand from what u say, it sounds like a very nicely thought plan
03:36.01docelmoso I will support IAX/SIP/H323
03:36.20docelmoI have been engineering this network for about 8 months and its just now coming to light
03:36.38*** part/#asterisk Treemole (Treemole@evvlinlwt-nas-07-s126.cinergycom.net)
03:37.10Poemiusit seems like you've done ur homework well, I wish you good luck, and maybe get bigger than vonage :)
03:37.24docelmoVonage is being bought
03:37.29docelmoso I am not worried abou tthem
03:37.32*** join/#asterisk lotku (~hadme@210.213.173.53)
03:37.52Poemiuswell if u get the size of vonage and get bought :)
03:37.54docelmobesides there is enough of a market for VOIP and I am not going to target the same clients as vonage
03:38.00PoemiusI wouldn't worry much :)
03:38.08Poemiusretire on a beach and stuff :)
03:38.21docelmoI live in Tampa FL..  Im already on the beach.. :)
03:38.39docelmoAnyone going to Astricon in Atlanta in September?
03:39.01Poemiuswave on the beach next time, I'm right across the ocean :)
03:39.42docelmoIm in the Gulf..
03:39.53docelmoIts in october now?   Geesh
03:40.12Poemiusah ok :) I thought u were in the atlantic
03:40.26PoemiusI was gonna say... you dive in the atlantic, go towards south east, avoid a couple of hungry sharks...and boom, you're in casablanca
03:40.34*** join/#asterisk odie_flocon_ (~chatzilla@S01060011953994ee.cg.shawcable.net)
03:40.44odie_flocon_hello all.
03:40.51Poemiushi odie :)
03:41.12odie_flocon_how goes it Poe
03:41.28*** join/#asterisk dash (washort@68.212.221.181)
03:42.48docelmoThere's a bootcamp in Tampa..  WOO HOO!   3000 buks tho?   sigh
03:43.27dashHi. I'm trying to add some remote users to my Asterisk setup at an office; I need to be able to transfer calls to them, and have them transfer calls, etc. If I set them up with an IAXy, will this be convenient?
03:43.50dashi don't know much about the user interface to the iaxy
03:46.17docelmoUm, just use Flash and type the extension
03:46.20docelmoThats about it.
03:46.28*** join/#asterisk Poemius_ (~poemius@adsl-70-48-192-81.adsl.iam.net.ma)
03:46.31dashokay :)
03:46.39Poemius_@^#~{@~^{#~@{~# battery on my laptop
03:46.45dashdocelmo: is that attended transfer?
03:46.59Poemius_(message censored for sensitivity purpose)
03:47.00Poemius_:)
03:47.14decis there a way to find out what version of asterisk is installed on a server without being able to connect to the console? :P
03:47.33dashdec: call someone who is able to connect to the console? ;)
03:47.41decno one can :)
03:47.50deci have ssh and full root access
03:47.50decbut
03:48.01decasterisk -cr won't connect because its not listening on 127.0.0.1
03:48.11docelmohehe..  Kill the binary
03:48.15Poemiusnot sure if there is a -v param
03:48.19docelmoor check the source directory
03:48.24Poemiusasterisk -v?
03:48.30docelmodash yes
03:48.32docelmoshould be
03:48.41dash-V actually
03:48.44dashdocelmo: Awesome.
03:48.45Mavvie[root@mercury root]# asterisk -V
03:48.46MavvieAsterisk CVS-HEAD-02/10/05-22:44:29
03:49.01Poemiusnyep, -V :)
03:49.30Poemiusmine is more recent than yours :)
03:49.36decAsterisk CVS-04/28/04-15:16:48
03:49.38decgot it, thanks
03:49.39dec:)
03:49.42Mavviebut for your information, asterisk -r looks for /var/run/asterisk.ctl, not for a TCP socket.
03:49.57MavviePoemius: that's my test machine, the real one runs 1.0.7
03:50.21decMavvie: oh okay. does it only look in /var/run ? I have the asterisk.ctl file, but its in another location...
03:50.30Poemiusyou're entitled to even use older versions too
03:50.31Mavviewhere do you have it?
03:50.41Poemiusas long as they have the moose penis sound :)
03:51.48decMavvie: /var/horizon/server1/logs/asterisk.ctl and /var/horizon/server2/logs/asterisk.ctl
03:51.57decMavvie: (I didn't set it up this way, blame someone else :P)
03:52.16docelmouhh ya.. someone didnt use the default..
03:52.26docelmoDec if you need access kill the server and restart it.
03:52.47Poemiuskill -9 the sucker :)
03:52.54decIt's done like that because there's two asterisk instances on the same box
03:52.55Poemiusyeah, hehe, killing is coool :)
03:52.59Poemiushehhe hehe
03:53.06decI don't need to kill it... i wanted to get console access to check the debug output :P
03:53.16decBut its okay, I think I've fixed the problem anyway
03:54.24Poemiustoo bad, no killing
03:54.28dechehe
03:54.55Poemiusbetter luck next time
03:55.33decoooh, I could symlink /var/run/asterisk.ctl to the asterisk.ctl from the one I want to connect to :)
03:58.18Poemiusyep, if you do that often, even make a script that does all for u
04:00.12docelmoLook at your physical log
04:00.34Poemiusgood idea
04:01.14docelmo:)  I try
04:01.23docelmoeven tho I am tired as hell..
04:01.32*** join/#asterisk cp5 (~samy@chcgil2-ar7-4-3-040-086.chcgil2.dsl-verizon.net)
04:01.34cp5hi
04:02.10cp5has anyone ran into zaptel/asterisk dropping ALL calls when on a span when the asterisk machine is the network, not the CPE? i have a bunch of different, strange error messages
04:02.13cp5it happens maybe twice a day
04:02.17odie_floconHmm now I got 2 tdm400p cards.
04:02.25*** join/#asterisk AlexCeli (~Alex@200.37.85.95)
04:02.45AlexCelihi
04:02.46odie_floconand a Hitachi wireless IP phone.
04:03.06odie_floconand a polycom ip600
04:03.31docelmoI have 500 PAP2-NA's and RT31P2-NA's..  :)
04:03.44Poemiusp o w e r :)
04:03.51odie_floconmust be nice.
04:05.11AlexCelii have a problem, my provider sent me the cisco firmware 7.4 for the 7960 cisco phone, but my default firmware is the 3.2 and I got a lot of "Disk Full Errors", i checked on digium list that I need another older firmware, exist a way to resolv it?
04:05.59docelmoDownload a different IOS
04:06.40AlexCelidocelmo: Which version i need?
04:06.45*** part/#asterisk dash (washort@68.212.221.181)
04:08.13docelmoWhat IOS does Digium say you need?
04:08.51*** join/#asterisk Poemius (~poemius@adsl-70-48-192-81.adsl.iam.net.ma)
04:09.43docelmoIm looking at the IOS's now
04:11.49*** join/#asterisk tylorflys (~tylorflys@ip68-104-178-155.ph.ph.cox.net)
04:11.51docelmoAlexCeli, I looked at the site says 7.x works well with *
04:12.11*** join/#asterisk NewSole (~david@i216-58-44-245.avalonworks.net)
04:12.33AlexCelidocelmo: but i'm upgrading from 3.2
04:12.57AlexCelii don't have 4.X, 5.X, only i have 6.X and 7.X
04:14.23docelmoUse 6.0 seems most stable of them all
04:15.06docelmoAlex can you download or do you have 6?
04:16.09*** join/#asterisk afrosheen (~afro@c-67-166-172-141.hsd1.tx.comcast.net)
04:16.22afrosheenhello again
04:20.27docelmonite all..  work calls..  well sleep then work
04:21.19afrosheenyeah
04:23.11harryvvwell, to test my failovr dial plan for calls into the states  my sixtel does not failover to zap when the ether cord is disconected to the cable modem.
04:23.16harryvvhi afrosheen
04:23.22afrosheenhey harryvv
04:23.37harryvvseems simple enough with the N+1
04:24.36harryvvwhat is the default timeout that asterisk kills a iax session where it looses connectivity?
04:25.01harryvvthen advances to the next priority?
04:28.21*** join/#asterisk jtodd (~jtodd@garthim.fox-den.com)
04:30.12*** join/#asterisk kimosabe (~nat@201.129.75.182)
04:30.33afrosheenman it's dead in here tonight
04:30.43afrosheenthere are flies zooming around below the vultures
04:30.45kimosabecan you run a sipura device with 8k with g711
04:31.13jakepdevwhich sipura device?
04:31.21kimosabe2000 model
04:32.02AlexCeliruns good..!!!
04:32.18jakepdevhttp://www.sipura.com/Documents/SPA-2000.pdf
04:32.23kimosabethe thing is i have several voip setups for interoffice comunications
04:32.40kimosabeworks great only some times it gets choppy
04:33.10jakepdevoh - you asked a trick question
04:33.21jakepdevyou already knew the answer
04:33.32AlexCeliI have another error with the Cisco 7969, the tftp looks for P0S3-07-4-00.sbn but i only have with the firmware the P003-07-4-00.sbn, it's the same file?
04:36.41*** join/#asterisk ptblank (~MURDER1@68-169-176-137.lmdaca.adelphia.net)
04:37.20AlexCeliupss 7960*
04:37.44*** join/#asterisk shmaltz (~chatzilla@69.28.255.210)
04:40.45afrosheenAlexCeli: welcome to the wonderful world of cisco
04:43.14*** join/#asterisk bimbo (~konversat@200.66.20.198)
04:43.26bimbohello, I'm very newb to this voip thing...
04:43.34afrosheenbimbo: hello
04:43.47bimboso I only have a few questions to ask you in order to clarify some things
04:43.57afrosheenbimbo: don't ask to ask, just go
04:44.16bimbothe first question is: is it possible to make a pc to phone call with asterisk?
04:44.23bimboif so, what do I need for this?
04:44.47afrosheenbimbo: pc to phone call? like use your computer to call a phone line?
04:44.55SedoroxYes, a softphone, a computer running asterisk, and either a VoIP PRovider. or a FXO card...
04:44.57SedoroxNEXT...
04:44.58Sedorox:-p
04:45.05jakepdev~docs
04:45.07jbotDocumentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org
04:45.07bugbotdocs is assigned nothing and reported nothing.
04:45.47bimbohmmm difference between a voip provider and a fxo card
04:46.16jakepdev~fxo
04:46.17jbotforeign exchange office - type of port you need to connect a POTS (Plain Old Telephone Service) line from your telco to a pbx http://www.digium.com/index.php?menu=fxsvfxo
04:46.17bugbotfxo is assigned nothing and reported nothing.
04:46.31AlexCeliafrosheen: lol, i'm understanding the beautifull world of Cisco.
04:46.48afrosheenAlexCeli: oh it's beautiful alright..if you don't have to touch it :)
04:46.56jakepdevhttp://www.voip-info.org/wiki-VOIP+Service+Providers
04:54.14*** join/#asterisk r0d3nt (~RatMan@wsip-24-234-241-84.lv.lv.cox.net)
04:54.41*** join/#asterisk three55ml (trilluser@cpe-66-25-89-157.satx.res.rr.com)
04:54.52three55mlHey everyone
04:56.15three55mlQuiet tonight?
04:56.32bimbook lets say I choose to use the fxo card... I install it on my computer and then, would I be able to do what I'm thinkinig? or do I ened something else?
04:56.41*** join/#asterisk BoRiS (boris@S01060040ca1e5b54.wp.shawcable.net)
04:56.56Sedoroxyou can use it then.. just gotta learn howto setup asterisk
04:57.52afrosheenbimbo: you can do it with or without the fxo card..or an iaxy adapter, or a ton of other ways
04:58.04jakepdevanyone know how to get the call back after this: http://www.voip-info.org/wiki-Asterisk+cmd+MusicOnHold
04:58.10afrosheenbimbo: at the cheapest you can install Xlite or other soft client and get subscribed with a voip provider.
04:58.27three55mljakepdev: Take them off hold :)
04:58.35jakepdevhow?
04:58.36three55mljakepdev: That just sets the MusicOnHold to use
04:58.39bimboafrosheen: for some reason I don't want a voip provider...
04:58.49jakepdevno - it plays the music on hold
04:58.54jakepdevdoesn't it?
04:58.59three55mlLet me look
04:59.06afrosheenbimbo: then I guess you wanna pay through the nose on long distance and world calling ;)
04:59.08*** part/#asterisk DaLion (~DaLion@Toronto-HSE-ppp3770773.sympatico.ca)
04:59.14*** join/#asterisk DaLion (~DaLion@Toronto-HSE-ppp3770773.sympatico.ca)
04:59.25bimboafrosheen: if I buy the fxo card... this problem is solved I guess
04:59.28*** join/#asterisk dca (~dca@c-67-166-37-218.hsd1.co.comcast.net)
04:59.53three55mljakepdev: You're right, sorry
04:59.57*** join/#asterisk DaLion (~DaLion@Toronto-HSE-ppp3770773.sympatico.ca)
05:00.07afrosheenbimbo: the fxo just gives you an interface for your soft phone to use..i.e. an expensive phone jack.
05:00.08DaLionsyle u there ?
05:00.19jakepdevnp - but Q still remains - there's gotta be a way
05:00.29DaLiontryiong to find how to compile or even how to get mysql-max
05:00.30three55mljakepdev: What exactly you trying to do?
05:00.48jakepdevpipe music down while doing FastAGI stuff
05:01.14jakepdevwhen I'm done all data operations, start speaking variables
05:01.15three55mlAh
05:01.16bimbohmmmm so if I want to do pc to phone calls I have to expend some money
05:01.47jakepdevit's doing several AGI calls, so what happens now is music , pause, music, pause, etc....
05:01.48afrosheenbimbo: yeah at some point you will
05:01.49three55mlI'm sure there's an AGI command to put the user on hold.
05:01.57three55mlOh
05:02.15jakepdevnot the end of the world, but would be nice if there's a way
05:03.20three55mlYeah, off the top of my head I can't think of anything easy.  I'm sure there's a few ways to do it though.
05:04.14jakepdevwish I could understand the C code enough to figure it out... oh well.
05:04.54DaLionanyone know how to fucking compile mysql-max ?
05:05.13jakepdevwow - someone had a rough day
05:05.19DaLionyes
05:05.47bimboafrosheen: can I PM you?
05:05.49DaLiondarn site is lame
05:05.56three55mlI had a worse one.  My flight got cancelled and I got a speeding ticket.
05:06.10DaLionnothing about it they all point my ass to downloads page..only binairies are for friggin freebsd 4.7
05:06.13DaLioni got 5.3
05:06.25three55mlHave you looked at freebsd-ports?
05:06.26DaLionso i say ok ill do from source
05:06.28jakepdevplane too slow - car too fast
05:06.36jakepdevshould equal out
05:06.43DaLionlol three55ml i NEVER USE PORTS...
05:06.48DaLiontoo many oold shit
05:07.02three55mljakepdev: Well the lady messed up last week on the outbound flight and gave me a boarding pass for someone else, so it was all her fault :)
05:07.04DaLionan d i need to customize a lazy makinstall isnt enought for me
05:07.25three55mlWhy you so intent on MySQL-MAX now?
05:07.43jakepdevthat's 2 mistakes she made then?
05:07.45three55mlLast night you were running 5.x.  I would just stick to 4.x in a production environment unless you're doing things it doesn't support.
05:08.03jakepdevone for you and someone else
05:08.08three55mljakepdevL: Well it cascaded from there.  She basically inadvertantly cancelled my return flight.
05:08.13jakepdevugh
05:08.27three55mlAnd I found out at 7AM :)  Luckily there were seats.
05:08.35three55mlI did not want to be stuck in Kansas a day longer.
05:08.38DaLion?
05:08.42DaLionnew box
05:08.42jakepdevhehe
05:08.52three55mlDaLion: Ah
05:08.52jakepdevguess you're not in Kansas anymore
05:08.58DaLionfreebsd 5.3 .. and im trying to figure how to friggin compile max
05:09.02jakepdevsorry - had to go there
05:09.02DaLionits same source ?
05:09.14DaLiondoes it make automaticaly 3 deamons ?
05:09.28DaLionthe fuck theyse assholes cant write a darn install like it should
05:09.29three55mlNo
05:09.33*** part/#asterisk Enigma8121 (~Enigma812@pcp02587377pcs.shlb1201.mi.comcast.net)
05:09.36odie_floconsure
05:09.40odie_floconwhy is me
05:09.42DaLionok one good answer in 3 days form any forum post etc i saw
05:09.42odie_floconsoft
05:09.49Nuggetwhat do you expect?  it's mysql-related, it's bound to be full of misguided and malintentioned crap.
05:09.54DaLionahha
05:09.58DaLionok humor me
05:10.05DaLionhow does one get max made then ?
05:10.08Nuggetmysql attracts people who mean well but have no clue what to do.
05:10.16Mavvie:-)
05:10.18odie_floconhey when I dial with an extention * doesn't do anything?
05:10.26DaLioni prolly have anough shit with * as it is without this mysql crap
05:10.40DaLionodie_flocon get used to * barfing on you
05:10.40jakepdevodie - sure it doesn't do anything?  did you run a debug?
05:10.44three55mlDaLion: If it was me I would use 4.x and optimize it correctly.
05:10.54three55mlDaLion: 4.x has a proven record in production.
05:11.03DaLionoh and funny thing they invented RPM's for fucker who cant make shit
05:11.20DaLionthree55ml i need clustered solution on 4 servers
05:11.23DaLionso i need -max
05:11.38DaLionall i can google gets RPM shit pages
05:11.52DaLion~jbot RPM
05:11.53jbotRed Hat's package management system. URL: http://www.rpm.org/
05:11.53bugbotjbot RPM is assigned nothing and reported nothing.
05:11.54jbot...but rpm is already something else...
05:12.05DaLionlol
05:12.10DaLion~jbot SPM
05:12.11bugbotjbot SPM is assigned nothing and reported nothing.
05:12.12jbotokay, bugbot
05:13.01jakepdevwhy are these two bots in a converversaion - looks like a bug
05:13.25jakepdev~SPM
05:13.26jbotspm is, like, assigned nothing and reported nothing.
05:13.26DaLionman
05:13.26bugbotSPM is assigned nothing and reported nothing.
05:13.30jakepdevhaha
05:13.34DaLionthis i s totla shit
05:14.36jakepdevisn't there an IRC channel for this MySQL Max?
05:14.54DaLionno
05:15.08*** join/#asterisk scardinal (~supreme@port816.ds1-suoe.adsl.cybercity.dk)
05:15.34DaLion>> WHO #mysql-max
05:15.34DaLion[01:15]> #mysql-max End of /WHO list.
05:16.43three55mlIt'll be the same channel as MySQL
05:16.53odie_floconwhat company is that?
05:16.53three55mlProbably on EFNet
05:17.20odie_floconInternet provider?
05:17.23Qwellumm
05:17.33QwellDaLion: Which version of MySQL did you download?
05:17.39MavvieI would try freenode.net myself first, but okay.
05:17.49odie_floconwhere do you want?
05:18.23odie_flocongt?
05:18.27odie_floconno that's not GT
05:18.32odie_floconthat's MTS.
05:18.32three55mlMavvie: Yeah, #mysql on FreeNode
05:18.52odie_floconis that IP to pot's termination?
05:18.56DaLion4.1.11
05:19.54QwellDaLion: and did you compile it?
05:20.08DaLionno
05:20.23Qwellthen how are you going to have a mysqld-max binary?
05:20.26DaLionim trying to ./cponfigure it for max or something but nothing in docs and i read EVERY line of htem
05:20.38DaLionalways talk about RPm shit
05:20.39QwellSo, you also ready INSTALL-SOURCES?
05:20.47DaLionALL of it yes
05:20.52DaLioneven the windozes SHIT
05:20.59Qwelland?
05:21.19DaLionnow you agree they should of splitted install.unix and install.windoz.. or are theyre that much lamers out htere /
05:21.21QwellYou didn't read that mysqld-max is PART of mysql?
05:21.33DaLionnope
05:21.55Qwellwell, it makes it fairly obvious
05:22.00*** join/#asterisk ubergoober (~ubergoobe@c-24-16-110-117.hsd1.ca.comcast.net)
05:22.09DaLionwell im to tired then to do this stuff
05:22.35QwellWhy not just get it from portage?
05:22.37DaLionwhere ( line) does it say the mysqld mysql-max etc will be compiled in same time
05:22.44DaLionportage ?
05:22.47QwellDaLion: Where does it say that it doesn't?
05:23.16QwellYou're using FreeBSD, right?
05:23.34DaLion./usr/ports/databases/mysql41-server
05:23.36DaLionits thetres
05:23.53QwellSo why not just use portage?
05:24.15DaLion4.1.5
05:24.21DaLionnot 4.1.11
05:24.27Qwellvalid reason
05:24.32Qwellso, compile away
05:24.36DaLionhehe
05:24.38DaLiondoing
05:24.44QwellGenerally, you're supposed to try something first, before complaining...
05:24.51Qwellat least, thats how we like it done
05:25.03DaLionwell i looked into my other 5 sebrers running mysql and no max binarie so
05:25.18DaLioni `assumed` that it needed a compile flag
05:25.27DaLionand still do btw
05:25.47Qwellwell, most of the packages have a -max package
05:25.55QwellI'm fairly certain the source does not
05:26.03DaLionbinaries builds yes..but only for bsd 4.XX
05:26.07DaLionnot bsd 5.XX
05:26.33DaLionsoi m fucked ?
05:26.44Qwellno, just compile it
05:26.48DaLionill see.. should finish compiling in 34 -35 days max
05:26.58DaLion;)
05:27.01Qwellmysql takes like 30 minutes, tops
05:28.40Nuggettrying to run asterisk on non-linux platforms is not for the meek.
05:28.52Nuggetit's a damn shame, but that's how it is.
05:28.54DaLionnope
05:29.03DaLionexpesially zap
05:29.07Nuggetyeah
05:29.23DaLioncant use only linux box we run if zap sevrers
05:31.01Nuggetcant understand what you just tried to say
05:31.05*** join/#asterisk remmo (~rem@smack.isp.net.au)
05:33.31DaLioncompiled
05:33.37DaLion4 minutes not bad
05:34.25DaLionnow where should it be lol
05:34.46NuggetDaLion: are you under the mistaken impression that the mysql server port is version 4.1.5?
05:35.01DaLionit is
05:35.03Nuggetno it is not
05:35.39DaLionbash-2.05b# cd /usr/ports/databases/mysql41-server
05:35.40DaLionbash-2.05b# make
05:35.46DaLion>> mysql-4.1.5-gamma.tar.gz doesn't seem to exist in /usr/ports/distfiles/.
05:35.58DaLionhmm that looks like 4.1.5 to me
05:35.59Nuggetso update your damn ports.
05:36.05DaLionlol
05:36.05Nuggetthat's your fault.
05:36.09Nuggetthe port is at 4.1.11
05:36.18DaLionyeah i compiled 4.1.11
05:36.26DaLionstill cant find where it makes mysql-max
05:36.28Nuggetbut you could have just used the port.
05:36.39QwellmysqlD-max
05:36.41Nuggetno wonder you complain that ports is all full of old stuff, you seem to not know how to update it.
05:36.42Qwelld, d, d
05:37.00three55mlI think I hear an echo in here, I remember saying to use ports 30 minutes ago :)
05:37.12Qwellthree55ml: yeah, so did I, about 10 minutes ago
05:37.21DaLionso ?
05:37.36DaLionwhat the diff ? i got 4.1.11 sources and compiled.. wont change anything would it ?
05:37.53three55mlDaLion: Not trying to be rude, but why are you using things you're not familiar with?  Last night you were in here asking how to fix MySQL because your max connections was 100, and now you're setting up a 5-server cluster all of a sudden?
05:38.02DaLionahah
05:38.04NuggetI see several patches in the mysql port.  no clue how important they are, but yeah, it does chance things.
05:38.34DaLionupdating ports then
05:38.47Nuggetnot to mention the added convenience of the startup/shutdown scripts being automatically set up by the port.
05:39.39Nuggetand what the hell did you change root's shell to bash for?  :)
05:39.41Qwelland the updating
05:39.45Nuggetthat's just evil
05:39.58Qwellbash with root is evil on fbsd?
05:40.00DaLiondidnt chagne root cshell just my own
05:42.00DaLion<PROTECTED>
05:42.03DaLion?
05:42.16Nuggetis that supposed to be a question?
05:42.18Qwellgonna be a long night...
05:42.23Qwelland a long rest of your life. ;/
05:42.26DaLionahahah
05:43.09Nuggetyou pasted a random command and then added a question mark.  that's not a question.
05:43.18DaLionmy bad
05:43.26NuggetI can only presume that the "?" means "is this what I want to do" but how the hell should we know -- what do you want to do?
05:43.54Nuggetand how on earth did we get roped into the obligation of teaching you how to use your operating system anyway?
05:44.01Nuggetisn't there a #freebsd somewhere?
05:44.09DaLionno it was wtf is this doing in a file called UPDATE when it has nothing to do with updating
05:44.14Nuggetthis has absolutely nothing to do with asterisk.
05:44.25DaLionheu well yes it updates the index of ports lol but not the ports
05:44.29Nuggetthat command has a great deal to do with updating.
05:44.43Qwellemerge sync <3
05:44.44DaLionyeah sorry bye
05:44.47*** part/#asterisk DaLion (~DaLion@Toronto-HSE-ppp3770773.sympatico.ca)
05:44.53Qwellmy god
05:45.05Nuggetdamn.
05:45.08QwellI've never even used freebsd, and I know more then him...
05:45.12Nuggetheh
05:45.20Nuggetfor the record, "portage" is gentoo.  "ports" is bsd.  :)
05:45.25Qwellwhatever, heh
05:45.51Qwellwill remember that though
05:46.53*** join/#asterisk heison (~heison@206.16.163.128)
05:48.48Dovidhello all
05:48.57Dovidanyone know of a good free pocker pc phone
05:48.57Dovid?
05:49.08denonsjphone?
05:49.25Doviddidnt work
05:49.29Dovidanything else ?
05:49.47denonit works
05:50.04marlowe~softphone
05:50.06jbotsomething that should be drug out into the street and shot
05:50.06bugbotsoftphone is assigned nothing and reported nothing.
05:50.18marlowe~x-ten
05:50.20bugbotx-ten is assigned nothing and reported nothing.
05:50.22*** join/#asterisk _Eagle_ (nick@wasteland.net)
05:50.23marloweok i give up
05:50.28marlowetry x-ten xlite
05:50.29Dovidx-ten charges for thiers
05:50.35Dovidi need a free one for testing
05:50.42marlowex-ten light is free
05:50.42Nuggetyou didn't ask for a free one.  :)
05:50.45Nuggetoh, you did.  :)
05:50.48Nuggetnevermind me
05:50.57marloweoh wait
05:51.01marloweyou said pocker pc
05:51.06marlowefirst of all, its pocket :)
05:51.12marlowesorrry nevermind then
05:51.25marlowex-ten on pocket pc sucks
05:51.25_Eagle_can anyone tell me how the Transfer() application works?  i cant seem to get it to do what i want
05:51.29marlowesjlite is a lot better
05:51.40marlowesjphone omg
05:51.46marloweim geting my terms mixed up
05:51.50Nuggetheh
05:51.56Nuggetno more beer for marlowe.
05:52.02marlowe:-/
05:52.08marloweim sick, and it sucks.
05:52.30marlowei had to postpone my lasik surgery :(
05:52.45Nuggetsjphone is useful but kinda klunky.  x-lite is pretty, but it's cumbersome to use and intentionally limited.  eyeBeam is bloated, but the only option if you don't want to use a handset or headset.
05:53.01Nuggetand I've never even touched a pocke[rt] pc, so I have no idea what is viable on the platform, sorry.
05:53.15marlowex-ten just sucks on a pocket pc
05:53.21marloweI dont know why - but it does
05:53.36marlowewhen I do use a softphone on my pc i personally use x-ten pro w/ g.7229
05:53.40marloweg.729 too
05:53.47Nuggetit's a shame they don't sell x-pro any more.
05:53.55marlowethey dont?
05:53.57marlowewhere was i?
05:53.58Nuggetnope.  just eyebeam
05:54.05marlowethat's dumb
05:54.08NuggetI agree.
05:54.15_Eagle_what exactly is "Transfer()" expecting as an argument?  every time i try to use an extension, it gives an error
05:54.21marlowenot everyone wants eyebeam
05:54.30Nugget_Eagle_: "show application transfer"
05:54.36QwellNugget++
05:54.41_Eagle_nugget:  dont you think i already tried that?
05:54.41Nuggetor google "site:voip-info.org transfer"
05:54.49Nuggetno, I don't think you already tried that.
05:54.50marlowewtf
05:54.51_Eagle_i tried both of those, nugget
05:54.57marloweThey want to outsource X-PRO development
05:55.00_Eagle_it doesnt help me at all
05:55.00marloweI hate that
05:55.20Nuggetnobody in here *ever* tries those things first.
05:55.20Nugget:)
05:55.34marloweNugget:I do :(
05:55.45QwellNugget: sadly, that includes the people who offer that as advice to others. :p
05:55.45marloweI kung foo asterisk
05:55.46_Eagle_nugget:  ive been using asterisk for over 2 years now..  i know how to RTFM :-)
05:55.54marloweWhats a manual?
05:56.02Nuggetit's what cool cars use.
05:56.11AlexCelithe P0S3-07-4-00.sbn is the same file with P003-07-4-00.sbn? i can rename it?
05:56.25_Eagle_the docs i found dont explain the problem i'm having
05:56.36marlowe_Eagle_: Why dont you try explaining the problem.
05:56.38_Eagle_they say Transfer(extension)
05:56.49_Eagle_but it doesnt like it when i give an extension
05:56.56marloweDoesnnt like it?
05:57.01marlowe* doesnt have a personality
05:57.03marloweUnfortunately
05:57.19Nuggetbkw_ has enough personality for all of us.
05:57.38_Eagle_the *only* time it liked any of the 300 things i tried was when i did  Transfer(SIP/1000@sipmachine) then it sent the call to a different machine
05:57.46Nuggetheh
05:57.49_Eagle_but i want to transfer to another extension on the same machine
05:58.20three55mlTry including the context
05:58.21NuggetI don't know the answer.
05:58.24three55mlLike 101@internal
05:58.27_Eagle_so i'm asking, what exactly does Transfer want?
05:58.40Qwellare the extensions you're trying included in the current context?
05:58.41Nuggetcheck the source, I guess.
05:58.43_Eagle_three:  ok... but shouldnt it use th current context if i dont specify one?
05:58.44marloweIt wants a valid extension
05:58.49marloweMake sure it's included in the context
05:58.58marloweWooops Qwell beat me
05:59.17marloweI keep sneezing on the screen - I gotta go to bed
05:59.18_Eagle_qwell:  i tried that, yes
05:59.38_Eagle_i created a new extension inside the current context just for testing that
05:59.38three55mlWell technically you need to include the type.  If you want to sent to an extension in that, or even a differnt context, use Goto(context, exten, 1)
05:59.57_Eagle_no... i need to use Transfer i think
06:00.04_Eagle_Goto wont do what i need
06:00.05Qwelland...what happens?
06:00.32_Eagle_qwell:  that was one out of a hundred tries.. i dont remember the exact error... it just didnt work
06:00.47_Eagle_ill try again, if it will help
06:01.22_Eagle_what i'm trying to accomplish is using Dial() with the M() option to run a macro....
06:01.41_Eagle_then during that Macro, send the other side to an extension, then continue the original call where Dial left off
06:01.55_Eagle_i need to "bring another person into the system" from the outside
06:02.15_Eagle_while continuing the original caller's call independently after the second person is online
06:03.25_Eagle_the only way ive seen to bring an outside into the system is with /var/spool/outgoing... but as far as i can tell, /var/spool/outgoing doesnt give any sort of call progress or info if the call fails/ is busy or no answer, etc
06:03.37Qwellwait, you want to run something AFTER a Dial(), while its still active?
06:04.22_Eagle_i want caller A to Dial() caller B...  send Caller B to extension 1, and caller A to extension 2, independent of eachother
06:04.33Qwellafter the Dial()?
06:04.41_Eagle_after, during, whatever
06:05.18_Eagle_i need to be able to bring a third party into the system... without them dialing in
06:05.52_Eagle_picture for example a conferencing system... and the moderator does a dialout to another phone number... and wants to add that person into the conference.....
06:06.02_Eagle_thats just one example of what this could do
06:06.11Dovidcan onyone help me set up sjphone for my pockt pc ?
06:06.17Dovidi am having a bit of trouble
06:06.27decthat will be great if you can get it to work _Eagle_ :)
06:06.50_Eagle_supposedly it would work if i could get Transfer() to work properly
06:07.03_Eagle_but maybe i'm misunderstanding transfer() completely
06:07.12Dovidanyone here use sjphone ?
06:07.28denon_Eagle_: use an agi to fire off a spool file :)
06:07.41_Eagle_denon:  spool files dont give call progress
06:07.47*** join/#asterisk psycodad (~obiwan@2001:4060:4419:b1:0:0:0:2)
06:07.48denonnor do they give headaches ;)
06:07.49_Eagle_how do i know if it failed?  busy?  no answer?
06:08.01Dovidanyone here use sjphone ?
06:08.24dec_Eagle_: thats not how Transfer() works, from my understanding. But I can't actually work out what Transfer is meant to do, and how...
06:09.05psycodadanybody know why I get 'CAPI[contr1/8]/0' ast extension from capi instead of '8', I can't match against incoming capi calls
06:09.07*** part/#asterisk AlexCeli (~Alex@200.37.85.95)
06:09.41_Eagle_well, is there any other way to bring a third party into the system then?
06:10.08_Eagle_what i need, is kinda like a Dial() that fork()'s into its own channel
06:11.40foobospsycodad, that is just the channel name. you can still get the extension if you have incomingmsn and context properly set in capi.conf
06:12.06*** join/#asterisk clive- (~pirch@rndf-146-44-91.telkomadsl.co.za)
06:12.24_Eagle_from your reactions, it looks like i'm gonna get stuck writing my own application to do this :(
06:13.05psycodadfoobos: I have msn=3,4,6,7,8 and incomingmsn=*
06:13.14psycodadI also tried incomingmsn=3,4,6,7,8
06:13.40foobospsycodad, your phonenumber can't be 8 can it?
06:13.53_Eagle_anyone know which source file the /var/spool/outgoing stuff is in?
06:14.44psycodadno, I am behind a pbx with s0 bus and I only get the last digit I guess... the internal number would be 43,44,46 etc..
06:15.11*** part/#asterisk darkskiez (~mhb@host-84-9-102-21.bulldogdsl.com)
06:15.44elrichas anyone had much success with app_machinedetect.c ?
06:16.03fooboswell then if you have lets say context=capi-in in the capi.conf the [capi-in] 8,1,Answer() should work in extensions.conf
06:16.10_Eagle_ahh.. found it.... pbx subdirectory
06:16.19_Eagle_thanks people... bbl
06:16.37*** join/#asterisk Koshatul (~evangelio@inf-203-132-65-157.bne.ipnetworks.net.au)
06:16.48psycodadfoobos: do you mean since I don't have it defined in the capi-context it falls back to default and s ?
06:17.04foobospsycodad, that's how it works
06:19.48*** join/#asterisk PTG123 (~PTG123@ip68-106-24-139.ph.ph.cox.net)
06:20.28psycodadfoobos: I removed the extensions from default and moved them to local-capibus which is my capi context and I get:
06:20.32psycodad<PROTECTED>
06:24.48*** join/#asterisk jalsot (~tamas@abacus.eworldcom.hu)
06:24.48*** join/#asterisk gres (~serg@81.222.48.242)
06:26.25gresHi all. Have anybody success whith reciving fax by asterisk?
06:26.46*** join/#asterisk ra|n_man (~mIRC@210.213.173.53)
06:27.25ra|n_manis it possible to use non-digium products with asterisk?
06:27.40gresSending fax from pstn fax: fax(pstn)<->asterisk(te100p)<->cisco ata 186 <-> fax work well.
06:27.41ra|n_manwe have here a quintum tenor asg 400
06:28.06gresMmm...
06:28.21remmogres: i have fax receiving working
06:28.33remmogres: but its not compat with all faxes
06:28.41deccan i receive fax via an IAX connection? :P
06:28.52remmono
06:28.56decdamn.
06:28.57decthanks
06:29.01remmowell depends on the codec
06:29.04remmoand latency
06:29.13ra|n_manis it possible to use non-digium products with asterisk?
06:29.16decgsm and 40ms pings
06:29.21remmobut no is a very safe answer
06:29.27decokay, thanks remmo.
06:29.29gresremmo: you receive fax by RxFax by Steave Underwood?
06:29.35remmogres: yes
06:30.05remmogres: but i'm only seeing 85% compatiability with fax machines, there are just some faxes that can not connect
06:30.12*** join/#asterisk |Vulture| (~Vulture@64.234.204.68.cfl.res.rr.com)
06:30.15ra|n_man???
06:30.22*** join/#asterisk afrosheen (~afro@c-67-166-172-141.hsd1.tx.comcast.net)
06:31.01remmora|n_man: yes
06:31.36gresremmo: What hardware do you use? is it Digium's hardware?
06:31.52remmoi have used an e100p, a clone x100p and nothing
06:31.53*** join/#asterisk mcnobody (~laaksola@server.kopteri.net)
06:32.14heison|Vulture|: did you find a fix for the voicemail password problem?
06:32.21ra|n_manremmo: we have here a quintum tenor asg 400 gateway. can asterisk be configured to use this?
06:32.42odie_floconDamn, why won't my FXS pickup my DTMF
06:32.48remmodont know if it supports standards like sip and h323 there should be no reason, just time and tweaking
06:33.13gresWhat seting in you zapata.conf? Is Echocancelation off?
06:33.26odie_floconlet me see
06:34.21odie_floconno
06:34.26*** join/#asterisk jalsot (~tamas@abacus.eworldcom.hu)
06:34.43ra|n_manremmo: Quintum supports sip & h323
06:35.09odie_flocongres no it is not
06:35.17ra|n_manremmo: the thing is the quintum also has its own configuration software that looks like asterisk
06:35.23remmora|n_man: then should be fine
06:35.39remmora|n_man: that product should do everything you need
06:35.56*** join/#asterisk PoWeRKiLL (~PoWeRKiLL@d01m-1-164.d4.club-internet.fr)
06:36.29|Vulture|heison: yes it was DTMF problems with 1.0.5-->1.0.7 not voicemail
06:37.05ra|n_manremmo: but can i tie up asterisk with the quintum?
06:37.06|Vulture|prolly just bad config on my side and they tightened it up on their side and it messed up my config
06:37.27gresodie_flocon, remmo: Ok. thks. I'll try...
06:37.28ra|n_manremmo: am using asteriskwin32
06:38.01odie_flocon????
06:38.18BoRiSodie: Have you tried a different phone?
06:38.21remmora|n_man: probably not
06:38.27odie_floconyes 2 different phones
06:39.49*** join/#asterisk oej (~oej@apollo.webway.se)
06:39.59afrosheenso who knows how to use disa from the ivr to dump an outside caller into a meetme room
06:40.00ra|n_manremmo: i was hoping that i could use the fxs of the quintum with asterisk
06:40.24ra|n_manremmo: thanks anyways
06:42.28*** part/#asterisk ra|n_man (~mIRC@210.213.173.53)
06:46.03*** join/#asterisk Mazda-MX5 (~root@220-130-142-43.HINET-IP.hinet.net)
06:46.19Mazda-MX5hi , all
06:46.33remmomazda 6's are better
06:46.41Mazda-MX5ha ~
06:46.53Mazda-MX5I love MX5
06:47.16remmoi used to love 6. mazdas just dont have much BALLS
06:47.18Mazda-MX5but , no money
06:48.41Mazda-MX5I have a question , why the asterisk report  "Unable to find a path from alaw to g729" when I accept a call ??
06:49.16Mazda-MX5caller is LP-201 , be caller is Cisco 7940
06:51.19Mazda-MX5remmo , I love drift racing
06:51.34ZgarbiI have mazda 323, old but works
06:52.06Zgarbiin real
06:52.15Mazda-MX5I agree
06:52.50*** join/#asterisk wiseguy_ (~chivilis@vadyba.vtu.lt)
06:52.55wiseguy_hellow
06:53.01remmoMazda-MX5: hard to drift race in a front wheel drive
06:53.02Mazda-MX5hi
06:53.04wiseguy_anybody has ata186 cisco sip ISO?
06:53.09wiseguy_IOS im
06:53.10wiseguy_:)
06:53.19remmothat would be illegal
06:53.35wiseguy_what?:)
06:53.40Mazda-MX5How body know Why the asterisk report  "Unable to find a path from alaw to g729" when I accept a call ??
06:53.43remmoi didnt see anything
06:53.57remmocause the call authed ?
06:54.15remmoyou can accept a call but where any audio is passed thats another question
06:55.37Mazda-MX5I accept a call , then asterisk report it , and I can not listen any voice
06:55.42wiseguy_anybody
06:55.48wiseguy_using cisco ata186?
06:55.49Zgarbican somebody provide me - where is it possible to buy linksys pap2, as I'm international buyer and want to buy via internet, as cheap as possible. at this case 10 units.
06:55.49wiseguy_:)
06:56.27ZgarbiMazda-MX5 did u in sip.conf did: canreinvite=no ?
06:56.28Mazda-MX5No , I never use ata186 , sorry
06:56.42Mazda-MX5wait , I check
06:57.12Mazda-MX5my setting is canreinvite=yes
06:57.21Zgarbichange to no
06:57.29Mazda-MX5the keypoint is canreinvite value ?
06:57.42*** part/#asterisk tylorflys (~tylorflys@ip68-104-178-155.ph.ph.cox.net)
06:57.53Mazda-MX5thank you , I try it , Now
06:59.16*** join/#asterisk CleanerX (~nix@nat-ph3-wh.rz.uni-karlsruhe.de)
07:00.19*** join/#asterisk pif (ldm@zenon.apartia.fr)
07:01.18Mazda-MX5Zgarbi , Thank you ,but it do not work , I still asterisk report
07:01.42Mazda-MX5still see asterisk report it
07:01.43|Vulture|anyone use ChanSpy?
07:02.51Zgarbiwht hardware do u have?
07:02.53Zgarbiwhat
07:03.34Mazda-MX5caller is a LP-201 , be caller is CISCO 7940
07:04.27Mazda-MX5I think CISCO only support g729...
07:06.02shepherdcisco should support more than just g729
07:06.07Zgarbiis your asterisk supports g729?
07:06.21shepherdasterisk doesn't have to support g729 for g729 to work
07:06.43Mazda-MX5asterisk do not support g729 ??!!
07:06.43shepherdas long as it doesn't have to transcode
07:07.00shepherdasterisk supports g729, but you have to buy a license
07:07.03Zgarbias I remembers with addons
07:07.25shepherdbut, if both phones support g729
07:07.29shepherdyou don't need a license
07:07.53Mazda-MX5my /usr/lib/asterisk/modules have format_g729.so , but not have codec_g729.so
07:07.55*** part/#asterisk oej (~oej@apollo.webway.se)
07:08.19Silik0nthats cuase you gotta buy g729 cause its patented
07:08.27remmoif you search hi and low on the web you will find such a beast but its not called codec_g729.so
07:08.36remmothe only codec i'm missing is speex
07:08.41Mazda-MX5so , I see, thank you ,Silik0n
07:08.44Silik0nremmosure it is
07:08.45*** join/#asterisk pbxjunkie (~Stormtroo@ppp14-adsl-159.ath.forthnet.gr)
07:08.50Silik0nthere are 2 different one
07:08.51Silik0ns
07:08.55Mazda-MX5thank you , Zgarbi
07:09.01shepherdhttp://www.digium.com/index.php?menu=asterisk_g729
07:09.11shepherdyou can get the codec, but no license
07:09.27Silik0nthat license is only$10
07:09.31pbxjunkiemornin' :) Has anybody got any idea WHY my asterisk fails to load chan_zap.so ? with error message: ast_load_resource: /usr/lib/asterisk/modules/chan_zap.so: undefined symbol: ast_retrieve_call_to_death
07:09.35shepherdtrue :)
07:09.35Mazda-MX5but , CISCO sould support more than g729 , such as ulaw , alaw
07:09.44shepherdg711
07:09.46shepherdgsm
07:09.57shepherdgsm #1
07:09.57shepherd:)
07:10.31Silik0npbx its prolly configured wrong
07:10.36Mazda-MX5so , If my 3 IP phone all support gsm , my sip.conf only need allow=gsm ?
07:10.43shepherdyes
07:10.52shepherddisallow=all
07:10.53Silik0nMazda-MX5 only if you just want to use GSM
07:10.54shepherdallow=gsm
07:10.55shepherd:)
07:11.06Silik0non a lan use g711
07:11.21*** join/#asterisk pooh_ (user78@cust.15.241.adsl.cistron.nl)
07:11.28Silik0nhell 711 works just fine over the internet too... (unless you're on dialup or something)
07:11.33Mazda-MX5Thnk you , all~~~ Thank you very much!!
07:11.54shepherdand you're not downloading pr0n
07:11.59Mazda-MX5I try now~
07:12.07pooh_What is the international access code when dialing FROM the US pls ?
07:12.12Silik0n1
07:12.15Silik0n011
07:12.15shepherd11
07:12.17Silik0nrather
07:12.31pooh_ok, thx
07:12.33Silik0n011+number
07:12.36pooh_thx
07:13.32Zgarbican somebody provide me - where is it possible to buy linksys pap2, as I'm international buyer and want to buy via internet, as cheap as possible. at this case 10 units.
07:13.43Mazda-MX5Now I get "chan_sip.c:2773 process_sdp: No compatible codecs!"
07:13.48shepherdit would have been funny if you gave him some 24 number number
07:13.58shepherd24 digit
07:13.59shepherdalso
07:13.59shepherdhehe
07:14.00Mazda-MX5my sip.conf only have allow=gsm
07:14.17Silik0nMazda-MX5 that happens when theres not a common codec or transcoder available
07:14.18pbxjunkieif I keep 'load => chan_zap.so' in my modules.conf then asterisk fails, if I remove it.. then it loads fine.. and it also 'sees' zap channels (i.e. give me access to zap show channels command)
07:14.26shepherdyou could try allow=all
07:14.45|Vulture|zapata.conf/zaptel.conf is configured wrong pbxjunkie
07:14.55Silik0npbxjunkie yu dont have to tell it to load it... asterisk will autoload it when needed by default
07:15.11Silik0nif you are using zap devices you probably have your zap configs wrong
07:15.28Silik0nztcfg -vvvv will tell you if you have the drivers configured right or not
07:15.43|Vulture|wow... its that time again boys and girls... its time to redesign my dialplan LOL
07:15.48Silik0n* will seg like bitch with bad zap configs
07:16.18wiseguy_hellow
07:16.26Silik0nmy dialplan is exten => _X.,1,Dial(Zap/g1/911)
07:16.27wiseguy_anybody using cisco ata186?
07:16.28wiseguy_:)
07:16.39|Vulture|lol
07:16.47|Vulture|nice Silik0n
07:16.53Silik0naight i go bed now
07:16.57Silik0npeice out
07:17.00|Vulture|lata
07:17.06|Vulture|haha lata....
07:17.10|Vulture|nvm :P
07:17.34Mazda-MX5I can not write "allow=all" that will become "Unable to find a path from g729 to ulaw"
07:17.54pbxjunkiecan somebody quickly take a peek at my zaptel / zapata files? http://pastebin.ca/9522
07:18.00shepherdallow=all
07:18.05shepherddisallow=g729
07:18.16|Vulture|pbxjunkie: post your ztcfg -vv please too
07:18.26|Vulture|oh wow
07:18.27Mazda-MX5thank you , I am trying
07:18.27|Vulture|you did
07:18.28|Vulture|hahaha
07:19.21|Vulture|pbxjunkie: thats quite the setup... looks correct to me
07:19.26*** join/#asterisk Delvar (~irc@83.146.53.34)
07:19.37|Vulture|pbxjunkie: Id recommend backing it off and trying to get it to run with just 1 card configured
07:20.43pbxjunkie|Vulture| i only have 1 card :)
07:21.14|Vulture|pbxjunkie: I didn't mean ripping the card out I meant just configure 1 module at a time
07:21.18|Vulture|the the TE400P?
07:21.25pbxjunkiequadBri
07:21.49|Vulture|most Ive used is the TE110P
07:21.53pbxjunkieoh so I should try configuring just 1 module at a time
07:22.03Mazda-MX5shepherd , Thank you , I have been not see "Unable to find a path from g729 to ulaw" , but , when Iaccepted a call , I can not listen any voice ~><~
07:22.12|Vulture|yea... issolate your problem
07:22.19|Vulture|but the config looked fine
07:22.29|Vulture|* may not be liking the config of one of them though
07:25.18*** join/#asterisk zoa (~zoa@pirus.securax.be)
07:28.29*** join/#asterisk brc__ (~brian@brc.base.supporter.pdpc)
07:28.41*** join/#asterisk RestLessGemini (~umairbari@202.142.189.86)
07:29.18Mazda-MX5Oh~ I will crazy
07:32.22Mazda-MX5..
07:33.35*** join/#asterisk r0d3nt (~RatMan@wsip-24-234-241-84.lv.lv.cox.net)
07:35.21*** join/#asterisk UPMeduardo (~UPMeduard@tauro2.dit.upm.es)
07:38.48*** join/#asterisk ta[i]nted (~tainted@adsl-69-108-114-226.dsl.irvnca.pacbell.net)
07:40.24Mazda-MX5..
07:40.58*** join/#asterisk Verliba (~Miranda@250.1-14-84.ripe.coltfrance.com)
07:41.01DelvarMazda-MX5: do you have g729 licances installed on your asterisk server?
07:42.00DelvarMazda-MX5: if not either 1. dont use g729 2. use clients that both suport g729 3. buy licances from digium
07:51.03*** join/#asterisk darkskiez (~darkskiez@usergc137.dsl.pipex.com)
07:51.14*** join/#asterisk oden (~oden@194-237-146-22.customer.telia.com)
07:52.21ta[i]ntedis it possible to get the asterisk-addons 1.0.6 package from cvs?
07:52.22|Vulture|I need to get an electric shcoker for peoples chairs for when they twist the phone cord...
07:52.43ta[i]nted|Vulture| why
07:52.53|Vulture|ta[i]nted: http://www.asterisk.org/html/downloads/asterisk-addons-1.0.6.tar.gz
07:52.54darkskiezis it normal for the milliwatt test to pop on a lan ?
07:53.19*** join/#asterisk scanna (~scannachi@81-174-16-211.f5.ngi.it)
07:53.28|Vulture|ta[i]nted: they keep twisting the chords on these IP500s into a coiled bunch on the table where you can't even life the handset
07:54.05ta[i]ntedget one of those cord untwisters
07:54.14ta[i]ntedit's like 0.99
07:54.31Delvarsupperglue the handset down... that will teach them!
07:54.50|Vulture|ta[i]nted: yea we are going to do that... but still... is it that hard?
07:54.59|Vulture|they have to be like sitting and spinning on the chair
07:55.23|Vulture|they got plantronics headsets now though...
07:55.25Delvaryou would be supprised what users actualy do instead of working...
07:55.30ta[i]ntedthose are nice
07:55.56|Vulture|Delvar: its pretty hard for them to not work.. we have a pretty tight lock on our offices
07:56.15Delvarlol
07:56.48*** join/#asterisk ra|n_man (~mirc@203.87.133.32)
07:56.59|Vulture|yea... we are pretty big brotherish
07:57.24ra|n_mancan an asterisk server be configured to be a gatekeeper?
07:57.59ra|n_mani already have a gateway that i need to register to a gatekeeper
07:59.22ra|n_man???
07:59.22|Vulture|gatekeeper and keyholder... nevermind I have had lack of sleep
07:59.22|Vulture|just ignore me
07:59.30Delvarwondered why it sounded familiar :)
07:59.41kajtzuwasn't it the gatekeeper and keymaster?
07:59.46Delvarwatched it not so long ago...
07:59.52|Vulture|keymaster thats it
08:00.03ra|n_manseriously guys
08:00.07*** join/#asterisk inspired (mikael@213.197.167.61)
08:00.18Delvarserius? omg this is #asterisk!
08:00.20ra|n_manis it possible to configure asterisk as a gatekeeper?
08:00.57Delvartbh i dont know, i have no idea what a gatekeeper is
08:01.02remmono
08:01.41|Vulture|damn its 4am...
08:01.47kajtzuDelvar: it's a thingy that keeps track of h.323 zones
08:01.53kajtzuand registrants within them
08:02.02darkskiezthe mozilla xul xml namespace is http://www.mozilla.org/keymaster/gatekeeper/there.is.only.xul
08:02.21darkskiezthat  has to be in every firefox plugin, it makes me laugh... god i'm sad.
08:03.52remmoxul rox
08:04.16remmoanyway home tmie
08:04.28ta[i]ntedshould i put the contents of the asterisk-addons into /usr/src/asterisk-1.0.6?
08:05.02*** join/#asterisk RES2 (~res-1@gateway1.nemox.net)
08:05.03RES2hi
08:05.04ta[i]ntedwhen i try to make install, i get 'asterisk.h: No such file or directory'
08:05.46RES2Can anyone help me with a callerID-problem (sorry for my bad english)?
08:07.06RES2I make a call-redirection in a SIP-phone. So the outgoing callerID ist the callerID of the caller. I want to send the callerID of the redirectin phone.
08:07.33*** join/#asterisk ra|n_man (~mirc@203.87.133.32)
08:07.39ta[i]ntedhep me hep me peas
08:07.53ra|n_manam back
08:08.14ra|n_manso? is it possible for asterisk to be a gatekeeper for my gateway?
08:09.07ta[i]ntedwhat is gatekeeper
08:09.26ta[i]nteddefine gatekeeper roles
08:13.06ra|n_mangatekeepers give access to gateways and border elements to interconnect and do voip
08:13.21*** join/#asterisk Moc_ (~Moc@modemcable165.109-70-69.mc.videotron.ca)
08:13.59ta[i]ntedlike a proxy?
08:14.16ra|n_mankinda like that
08:14.26ra|n_manlike a security guard of a building
08:15.01ta[i]ntedyea like a proxy
08:15.07ta[i]ntedu could use asterisk
08:15.10Delvarthen yes asterisk can do that
08:15.15ra|n_manbecause a gateway needs to register to it before it can have access
08:15.23ra|n_manhow is it done?
08:15.33*** join/#asterisk nrc (~username@zeus.eurotux.com)
08:15.34ta[i]ntedwhat kind of gateway is it
08:15.43Delvarwww.voip-info.org - look for sip.conf - register =>
08:15.50Delvarits prety simple
08:16.02Delvarhahaha
08:16.26*** join/#asterisk ra|n_man (~mirc@203.87.133.32)
08:16.46ra|n_manmy gateway is a quintum tenor asg 400
08:16.47RES2Delvar ... do you mean me?
08:18.21ra|n_manhow do i configure asterisk to be a gatekeeper
08:18.23ra|n_man???
08:21.17cypromisyou don't
08:21.19*** join/#asterisk basta (~kqj@62-101-126-233.fastres.net)
08:21.23cypromisasterisk has no gatekeeper functionality
08:21.56bastahallo, anyone using cisco 7912 and/or 7960 out there ?
08:22.02*** join/#asterisk tainted_ (~tainted@adsl-69-108-114-226.dsl.irvnca.pacbell.net)
08:23.12darkskiezloads of folks
08:23.13*** join/#asterisk terracon (~tc@CPE0050da608e99-CM0012254076d6.cpe.net.cable.rogers.com)
08:24.19bastaseems music on hold/transfer stopped working with 1.0.7, can anyone confirm this ?
08:28.22bastasorry, it was 1.0.6
08:28.22inspiredcan a member in queues.conf be a IAX2 friend?
08:28.37bastaand sip protocol
08:31.10*** join/#asterisk PoWeRKiLL (~PoWeRKiLL@host.161.115.68.195.rev.coltfrance.com)
08:35.30psycodadI am still fidling with CAPI channel: I get the correct MSN in ${DNID} but in ${EXTEN} I get the full channel string not just the MSN... I guess I can do it with gotoif based on ${DNID} but that should work with ${EXTEN} too, right ?
08:35.54*** join/#asterisk Fabe (~spamhere@pD95B0BF7.dip0.t-ipconnect.de)
08:38.17*** join/#asterisk vagwin (~vagwin@mk-ns500-1.uk.tiscali.com)
08:39.52jalsothi
08:40.36jalsotdoes anybody know how much consumes one recorded call in sln format? is it 2x64kbps=128kbit/s ?
08:41.29jalsot[without mixing]
08:42.40PoWeRKiLLwho use call fowarding ?
08:44.00*** join/#asterisk FaithX (~Faithful@202-6-145-116.ip.adam.com.au)
08:44.58RES2Can anyone help me with a callerID-problem (sorry for my bad english)?
08:45.00RES2I make a call-redirection in a SIP-phone. So the outgoing callerID ist the callerID of the caller. I want to send the callerID of the redirectin phone.
08:51.53*** part/#asterisk RES2 (~res-1@gateway1.nemox.net)
08:54.31*** join/#asterisk DrJolo (~chatzilla@cerber.ftj.agh.edu.pl)
09:01.06psycodadanybody working with CAPI channels at all ?
09:02.19pooh_I do
09:04.03pgpkeysgoddamn, this is one hell of an active channel. i like, i like :)
09:04.24pgpkeysbeen lurking for the past few days, good stuff.
09:07.15pooh_psycodad: Fiddle with ${EXTEN:x} , where x is the number of digits you want to cut off of the exten from the start. e.g. exten=1234, ${EXTEN:2} = 34
09:09.47*** join/#asterisk Betu| (~betul@62.244.193.101)
09:10.17*** part/#asterisk Betu| (~betul@62.244.193.101)
09:10.50*** join/#asterisk zoa (~zoa@pirus.securax.be)
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09:11.33*** join/#asterisk Betu| (~betul@62.244.193.101)
09:11.40foobospsycodad, i'm guessing your MSN is too short so chan_capi stuffs the whole channel in exten
09:11.55fooboscause i get the exten just fine
09:17.34*** join/#asterisk ddum (~spamfilte@argus.nwl.se)
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09:21.47sympad!list
09:23.31*** join/#asterisk Alexi1 (~alexis@www.trim.it)
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09:25.04psiderovanyone can help me how to forward codec capabilities of the UAC to the pstn gw, not the codecs set in sip.conf
09:27.39psiderovor is there such scenario at all ?
09:28.25zoayes there is
09:29.35psiderovthanks zoa, any idea how to do that ?
09:31.57psiderovor where shoul I read that / who I have to ask ?
09:32.06*** part/#asterisk Alexi1 (~alexis@www.trim.it)
09:33.04psiderovI'm trying to edit chan_sip.c but still with no success
09:35.27zoayou dont need to edit anything
09:35.37zoathere are some vars / patches for it normaly
09:35.41zoadont know them by head
09:35.48zoaand dont have the time to look for it
09:35.49zoasorry
09:36.14psiderovjust something that can push me on ... name or something ?
09:38.50*** join/#asterisk ikey1 (ikey@220.226.16.30)
09:39.03psiderovname of the patch ?
09:39.12zoacodec preference maybe
09:39.30psiderovok, thanks zoa :)
09:43.37*** join/#asterisk anonymous12345 (~anonymous@60.48.111.128)
09:45.41wiseguy_anybody using cisco ata186?
09:46.12psiderovI have few clients using it ?
09:46.17*** join/#asterisk teq- (~p@xdsla026.osnanet.de)
09:46.32newlWe don't know, do you?
09:46.33bastawiseguy, me
09:48.37*** join/#asterisk yaboo (~jsirucka@220.245.131.131)
09:48.54*** join/#asterisk Betu| (~betul@62.244.193.101)
09:49.12anonymous12345hi i m interested to do further development on asterisk... since i m new i know the place to start if from bug level - i m from win background
09:49.50anonymous12345but i found out the cvs thing is rather confusing... anyone mind to explain...
09:50.05*** join/#asterisk Donuil (~cini_lab@217.9.64.213)
09:50.45bastayes, there's a guy named google who can perfectly explain you how cvs works ;)
09:51.15anonymous12345wow tat good... but the tree and branches is the one confuse me...
09:51.17Donuilhi to all... can someone give me a suggestion about configuration of zaptel.conf and zapata.conf?
09:51.43anonymous12345there is thing called cvs-head and a lot of other version stable version...
09:54.50*** join/#asterisk Betu| (~betul@62.244.193.101)
09:55.21anonymous12345is there any tools (web) that can view the cvs tree??
09:57.03*** join/#asterisk Yellow_Fuzzy (yellow@c211-31-41-9.wavrl1.nsw.optusnet.com.au)
09:57.06Yellow_Fuzzyhi
09:57.34Yellow_FuzzyIm trying to setup my account on a new SIP provider with Asterisk@home but am having a few problems
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10:03.42*** join/#asterisk emitrax (~emitrax@ingnatdyn33.unime.it)
10:03.45emitraxhi
10:04.50emitraxWARNING[16658]: chan_sip.c:8838 reload_config: Section 'xlite1' lacks type
10:05.32emitraxnever mind... I'm an idiot :)
10:06.35emitraxNOTICE[16658]: chan_sip.c:7681 handle_request: Registration from 'sip:line1@X.17.32.4' failed for 'X.17.129.167'
10:07.05emitraxI just changed the firmware on a 7940 from SCCP to SIP 7.0 and now Im trying to register it with asterisk
10:07.11emitraxbut I get that error
10:07.41emitraxI think I edited correctly either sip.conf and extensions.conf
10:08.07emitraxso I don't know how to debug it... can someone give me a hint please?
10:08.24*** join/#asterisk Donuil (~cini_lab@217.9.64.213)
10:09.41DonuilI try to call my zap phone but unsuccessfully... can someone help me with the zapta.conf file?
10:09.57*** join/#asterisk Jax (~tsche@adsl-62-167-77-76.adslplus.ch)
10:10.03Jaxhello :)
10:10.49Jaxi'm realyl new to VoIP.. not sure what kind of hardware i need, if i need to get an additional service @ my ISP, etc.. maybe somebody can fill me in?
10:12.36*** join/#asterisk pbxjunkie (~stormtroo@videocomputer.gr)
10:13.29*** join/#asterisk Betu| (~betul@62.244.193.101)
10:14.30*** part/#asterisk Betu| (~betul@62.244.193.101)
10:18.16*** join/#asterisk Donuil (~cini_lab@217.9.64.213)
10:18.43pbxjunkieall this time I've been having trouble making calls through my quadbri when only now I noticed that all 4 leds on the back are black
10:19.08pbxjunkieeeer.. red :D
10:20.24Donuilsorry I'm falling sometime... so someone have experience with the configuration of the zapata.conf file?
10:21.32RoyKhttp://www-128.ibm.com/developerworks/library/pa-nl3-marenostrum.html
10:22.09DonuilI try to call unsuccessful my zap phone... I sure that the card is correctly installed...
10:22.54RoyKwhy do you try to be unsuccessful?
10:23.55Donuilbecause it doesn't ring and start voicemail as by setting
10:25.28FaithXdo you have to run ztcfg with those cards/
10:25.37FaithX?
10:26.00DonuilI have a digium tdm400p installed in configuration 22b, with 2 fxs and 2 fxo
10:28.01Donuili configured my zaptel.conf with fxo=1-2 fxs=3-4 and when I launch ztcfg -vvv all is right
10:30.24sympadput smth. in extension.conf ?
10:31.03bastadonuil, since the voicemail starts, isn't maybe an extensions misconfiguration ?
10:31.05Donuilthe I configured my zapata.conf  'signaling=fxo_ks callerid="mycallerid" <1003> channel => to call my zap phone on Zap/3... is correct according to you?
10:32.07sympadwhere from do you try to place the call ?
10:33.24Donuilsympad from asterisk console... basta according to you is a card problem?however I have no error logs about it
10:33.43facek_can someone help me with PAP2 ?
10:33.57newlyour provider can. :)
10:36.26Donuilmaybe a signaling problem?It is correct my configuration in zapata.conf matching with zaptel.conf?
10:36.51sympad1-2 are FXS or FXO ports ?
10:38.40Donuilin my zaptel.conf fxo=1-2 and fxs=3-4... Does means it that 1-2 are fxs port?
10:46.35elricdoes anyone have an example / snip of extensions.conf with MachineDetect() used in it? theres naught on the wiki about it
10:49.47elricor take a look at this and tell me what  i am doing wrong
10:49.48elrichttp://pastebin.ca/9527
10:50.29Jaxso will anybody tell me what i need to do VoIP ?
10:50.58elricwait i pasted the wrong section
10:51.31*** join/#asterisk pbxjunkie (~stormtroo@213.5.44.113)
10:54.34malbechAnyone konws a SoftSwitch Solution for a low cost ?
10:54.50kajtzucost is relative
10:54.50kajtzu:)
10:55.05malbechof course ... :)
11:01.48pbxjunkieguys, if I want to connect the telco's S0 lines onto my PCI QuadBri card, which mode is that? NT right?
11:02.18pbxjunkieTE is if you want to connect ISDN telephones on the port, right?:)
11:03.05Jaxnobody wants to tell me what i need for VoIP ;(
11:04.29emitraxhow do I konw whether a phone is registered with asterisk or not?
11:04.37Mavviesip show peers
11:04.42Mavviewell, if it is a sip phone
11:04.46emitraxyeah
11:04.52emitraxbut the status is unmonitored
11:04.58emitraxwhat does it mean?
11:05.09Mavvieunmonitored means: not monitored (i.e. can be up, can be down, asterisk doesn't care)
11:05.22emitraxmay I paste 3 lines in the chan?
11:05.35Mavviemeans yuo haven't set the host= statement in the phones sip configuration on asterisk.
11:06.08emitraxi did
11:06.13emitraxhost=dynamic
11:06.31Mavviewell, not a real hostname then.
11:06.39cypromisTE is for connecting to the NTBA
11:06.45cypromisNT is for emulating an NTBA
11:06.54emitraxso? ... I'm sorry but I don't understand...
11:06.59cypromisthat is for pbxjunkie
11:07.04Mavvietry it with a real hostname
11:07.09cypromisegocentrism is useless in a irc channel
11:07.11cypromis:P
11:07.24pbxjunkiecypromis:D
11:07.27emitraxhow do I put the hostname if they are phone ?
11:07.57pbxjunkieaand.. NTBA is? :D
11:08.07cypromisNTBA is the silly box you get from the telco
11:08.54pbxjunkieso to connect my quadbri to the silly box ..I neeed.. TE mode for the ports? NOT NT like I have now?
11:09.08pbxjunkieis that why all the leds are red?:)
11:10.12pbxjunkiebrb , off to re-jumper my quadBRI :D
11:11.16*** join/#asterisk wildgoose (~edward@xunil.mail.wildgooses.com)
11:12.22*** join/#asterisk MikeJ[Jayden] (~jayden@pcp02795302pcs.roylok01.mi.comcast.net)
11:13.11*** join/#asterisk nextime (~nextime@213-140-22-64.fastres.net)
11:18.08*** part/#asterisk Donuil (~cini_lab@217.9.64.213)
11:23.16*** join/#asterisk gambolputty (~gambolput@cblmdm65-221-51-80.buckeye-express.com)
11:24.45*** join/#asterisk pbxjunkie (~stormtroo@213.5.44.113)
11:29.08*** join/#asterisk L|NUX (~linux@202.5.145.58)
11:40.21*** join/#asterisk julianjm (~julianjm@250.Red-80-59-67.pooles.rima-tde.net)
11:44.47*** join/#asterisk zeedo (~zeedo@www.bsrf.org.uk)
12:03.52*** join/#asterisk Jackthe (MyIdent@prac002.ewi.utwente.nl)
12:03.52*** join/#asterisk ccfiel (ccfiel@210.5.72.36)
12:03.52ccfielhello ppl..
12:06.05ccfielcan somebody help me with my problem.. a remote iax connection connect to my * box ...and communicate with a local iax connection.. before a few second at least 10 seconds there is a lag in commnunication...after a while lag will disapper...what's would be the problem..can somebody help me with this? :(
12:07.06tzangerccfiel: what version of asterisk
12:07.29Jacktheccfiel: can you make a packettrace of the setup of the call?
12:07.34*** join/#asterisk wildgoose (~edward@xunil.mail.wildgooses.com)
12:07.42Jackthethen look at the voicetimestamps
12:08.25JacktheI think there will be a jump in the timestamp somewhere in the 1 second
12:08.44tzangerJackthe: stevek and I hae been working on this problem
12:08.57ccfieljackthe: what would be the solution for this?
12:09.10tzangerccfiel: what version of asterisk?
12:09.26ccfielwait..let me see..
12:09.53Jackthetzanger: great
12:09.56Yellow_Fuzzyhi
12:10.02ccfielsorry the dump question..how can know the version of my asterik?
12:10.15ccfielasterik/asterisk
12:10.23tzangerccfiel: show version
12:10.31Yellow_Fuzzyany one here happen to have any experience with OzTell?
12:10.31ccfielok..thanks...wait..:)
12:11.09*** join/#asterisk tzafrir (~tzafrir@62.90.10.53)
12:11.20tzangerYellow_Fuzzy: no, they made me send back the ruby slippers
12:12.51pgpkeyshah!
12:13.10Yellow_Fuzzylol
12:13.47tzangeryou know, it amazes me just how well safety scissors cut paper and nothing else
12:14.03*** join/#asterisk _THEEND_ (~DrEaM@80.18.184.226)
12:14.13pgpkeysoh now that's bad
12:15.26*** join/#asterisk dizzydiffi (dizzydiffi@adsl-70-240-211-145.dsl.hstntx.swbell.net)
12:15.30dizzydiffihello
12:16.09dizzydiffihas anyone compiled Asterisk 1.0.7 with OH323
12:16.28dizzydiffianyone alive in here
12:16.43Mavvieno, we just have been killed by a band of ninja turtles.
12:16.54dizzydiffiwow
12:16.59dizzydiffiturtles huh
12:17.02pgpkeysroll 2D20 for Ninja Master save
12:17.07dizzydiffithats crazy
12:17.15ManxPowerI really hate mornings
12:17.18ManxPower~docs
12:17.26jbotDocumentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org
12:17.26tzangerninja master gets 2d20?
12:17.26bugbotdocs is assigned nothing and reported nothing.
12:17.26tzangerdamn
12:17.26ManxPower~mailinglist
12:17.27jbotmailinglist is probably Search Asterisk mailing lists by prepending site:lists.digium.com to your Google search.  Browse the mailing list archive at http://lists.digium.com/
12:17.27bugbotmailinglist is assigned nothing and reported nothing.
12:17.43tzangerdamn that reminds me of my AD&D days
12:18.03pgpkeystzanger: i haven't gotten to the Magic Rat Tail adjustment yet
12:18.09tzangerhahaha
12:18.16tzangerrat tails... what a fashion statement
12:18.22pgpkeyshehe
12:18.39tzangerrat tails were the pre-mullet mullet
12:18.57pgpkeysthe rodent version of afros?
12:19.09tzangerno that is what happens when you christmas tree a cat
12:19.18pgpkeyshjahahahahaha
12:19.32pgpkeysthank you very much. i almost spit up my meatballs when i read that
12:19.40tzangermeatballs?  hwere are you at
12:19.50tzangerit's 8:20am here (EDT)
12:19.55pgpkeysbuffalo, NY USA. my rendition of breakfast
12:19.59tzangerahh okay
12:20.01pgpkeyslast night's spagetti
12:20.03tzangeryou're about 2.5h from me
12:20.07Godseythis is OT but might anyone use Illustrator? :)
12:20.12tzangerGodsey: not me
12:20.21GodseyI can't remember how to change the starting point of text to path tool once text is on it
12:20.22pgpkeysrochester?
12:20.25tzangerI love making a huge pot of spaghetti and then eating it cold for breakfast
12:20.36tzangernah, Kitchener/Waterloo, ON
12:20.40*** join/#asterisk Mimmus (~viggiani@ext.pitagora.it)
12:21.15pgpkeysahh other side of the mud splatter
12:21.16Mimmushi, I need a double E1 card to introduce Asterisk in my company, located in Italy
12:21.17tzangerand buffalo is *not* 2.5h from rochester
12:21.39tzangerMimmus: I'd suggest a TE405P, not a pair of TE110Ps...  lowre interrupt load and room to expand
12:21.40pgpkeysit's about that depending on if you go scenic or not ;)
12:21.43MimmusI see that Digium offers only quad card. Is it only chance?
12:21.50tzangerMimmus: pretty much, yeah
12:22.05tzangersangoma offers a 2-port card but I've no experience with it
12:22.07Mimmustzanger: and are themy chaper?
12:22.42tzangerwell a 1-port Sangoma or Digium is $500.  A 2-port Sangoma I think is $1000, and a 4-port Digium or Sangoma is $1500
12:23.08tzangerbut as I said with the 4 port you ahve room to grow iwthout buying additional hardware
12:23.30Mimmustzanger: well, and now a more difficult question: if I use Digium ,will I have more advantages (support, etc)
12:23.59MimmusI mean help on this channel, on the mailing list, etc
12:24.16*** join/#asterisk Betu| (~betul@62.244.193.101)
12:24.21tzangerMimmus: well everyone here tries to help, and several of us are quite good at it.  Digium has official support included in the price of their cards, plus there's the warm fuzzy feeling of helping out the people who wrote Asterisk
12:24.22blitzrageMimmus: IRC and mailing list are not official Digium support channels
12:24.26tzangermorning Betu|
12:24.40blitzrageMimmus: just like tzanger said :)
12:24.44tzangerblitzrage: what'd your name go for?
12:24.49blitzragetzanger: $7.00!
12:24.53tzangerblitzrage: ha
12:24.55blitzragehehehe
12:24.57tzangerhardly worth the auction
12:24.59elricdoes anyone have an example / snip of extensions.conf with MachineDetect() used in it? theres naught on the wiki about it
12:25.04tzangerblitzrage:  you like my name this morning?
12:25.06Mimmusblitzrage: I know but a good 'open' support is better sometime :-)
12:25.07blitzragetzanger: the auction only cost like .45 or something
12:25.19tzangerelric: where's MachineDetect?
12:25.25blitzragetzanger: yah, I laughed when I saw it :)
12:25.32blitzragetzanger: but alas, I have no prune juice
12:25.48blitzragetzanger: I'll be biking all over Mississauga today... probably like 30 km's or osmething like that
12:25.53elrictzanger, http://www.thenetbrain.com/files/app_machinedetect.c
12:26.03tzangerblitzrage: nice.  I used to do that
12:26.10blitzragetzanger: yah, its for work too :)
12:26.11tzangerblitzrage: I biked from listowel to kitchener and back
12:26.22blitzragetzanger: yah, thats a good little trek :)
12:26.33elrici got the link off www.voip-info.org
12:26.38blitzrageanyways, gotta shower and get breakfast, then head down to the GO
12:26.40blitzragelates
12:26.51tzangerblitzrage: bike courier?
12:27.13tzangeranyway I should get into work too
12:27.14blitzragetzanger: working on the phone systems at several CIBC branches
12:27.18tzangertalk to y'all in a bit
12:27.22tzangerblitzrage: asterisk-related?
12:27.29blitzragetzanger: unfortunately not
12:27.32tzangerblitzrage: :-)
12:27.34blitzragejust alarm system's
12:27.38tzangerahh oaky
12:27.39tzangerttyl
12:27.41blitzragelates
12:27.50_THEEND_b
12:27.55elrichttp://pastebin.ca/9527  <--- is a snip of my extensions.conf
12:28.14pgpkeyscibc  central intelligent bueracracy channel?
12:30.09*** join/#asterisk zotz (~zotz@24.231.32.109)
12:31.15ccfieltzanger,jackthe: my * version is : localhost*CLI> Asterisk CVS-HEAD-12/16/04-07:57:25 built by root@localhost.localdomain on a i686 running Linux
12:31.28ccfielis this the one?
12:31.48ManxPowerI just announced 2 bounties
12:32.33MimmusI'd like to integrate Asterisk with an existing Alcatel PBX... I'm planning to put * in front of Alcatel. Is it a goo didea?
12:32.48MavvieMimmus: yes
12:32.55MavvieMimmus: that's what we have here, in front of an A4400
12:33.28MimmusMavvie: the schema should be:     PSTN --- PRI --- Asterisk --- PRI --- ALcatel
12:33.33Mavvieyes
12:33.57MimmusMavvie: analog phones will remain connected to Alcatel
12:34.07cypromisManxPower: announcing bounties is for sure more positive than bitching at newbies
12:34.10cypromis;)
12:34.46MimmusMavvie: will it be difficult to configure Asterisk to route all calls to *?
12:34.53ManxPowercypromis, but much less rewarding.
12:34.58cypromishehe
12:35.00cypromistrue
12:35.01cypromis;)
12:35.12ccfieljackthe: still there?
12:35.15ManxPowerBounty 1: generate an errror if you try loading BOTH digium card modules and ztdummy
12:35.21MavvieMimmus: exten => _.,1,Dial(Zap/${TRUNK_A4400}/${EXTEN})
12:35.38ManxPowerBounty 2: generare an error if Hangup is run in exten =>
12:35.54ManxPowerMaveric, Why do you want to dial all calls twice?  That's what _. will do.
12:36.04ccfielcan somebody help me with my problem.. a remote iax connection connect to my * box ...and communicate with a local iax connection.. before a few second at least 10 seconds there is a lag in commnunication...after a while lag will disapper...what's would be the problem..can somebody help me with this? :(
12:36.16MavvieManxPower: to forward it to a different PRI.
12:36.23elriccan someone look at http://pastebin.ca/9527 and tell me if I am doing something wrong?
12:36.42MimmusMavvie: ah, ok! And if I want to use a set of extensions (starting with 6, for istance) to dial SIP phones, directly connected to Asterisk?
12:36.44ManxPowerMaveric, So as soon as the call ends it will dial the exact same Dial line twice.
12:37.07ManxPowerDON'T USE _.!!!!!!!!!
12:37.11MavvieManxPower: I don't see that behaviour here.
12:37.39ManxPowerMavvie, Maybe you have an exten => h also.  That would cause it not to loop.
12:38.01Mavvietrue.
12:38.17ManxPowerIt's still a bad idea.
12:39.12ManxPowerAnyone want to comment on the two bounties?
12:39.52bjohnsonarhg .. we be hunting bounties today mate
12:42.57MimmusCan I use Asterisk to setup a dialup server?
12:43.04ManxPowerMimmus, no
12:43.23ManxPowerif by "dialup" you mean "analog modem dialup"
12:43.41MimmusManxPower: ok, it was only an idea
12:44.31cypromisyou could do an isdn dialup
12:44.34cypromisbut not a analog dialup
12:44.56azidvoicemail-question: if a caller hangup during the vm welcome-message i always end up with a 0-legth message in my VM, anyone know why? (PSTN call over SIP)
12:46.17ManxPowerazid, Report it as a bug
12:46.36bastaanyone using cisco 7912/7960 with sip ? music on hold/forward stopped working and I can't understand why
12:46.41ManxPowerThere should already be a bug report about this, but I don't know the bug #
12:47.00azidmanxpower, ok. i was hoping i did something wrong in my config-files :(
12:47.26ManxPowerbasta, I have seen MoH during a transfer not working, but not music on hold during a forward.
12:48.10*** join/#asterisk bmg505 (~leon@rndf-146-59-117.telkomadsl.co.za)
12:48.20bastayes, i mean forward
12:48.49*** join/#asterisk jbAU (~johnblade@c210-49-42-214.rochd2.qld.optusnet.com.au)
12:48.56ManxPowerthen I've not seen it.  What kind of forward?  Dialplan forward, or a 302 Moved SIP forward?
12:49.27bastano, sorry, i mean i mean a transfer (need a holiday)
12:49.42ManxPowerbasta, Yes, I've experienced that.
12:49.48bastaany idea ?
12:49.57ManxPowerbasta, no,  I never fixed it.
12:50.21bastax|
12:50.22ManxPowerMoH works when the call is on hold, but not when the calls has been transfered and is runing.
12:50.33ManxPowerringing
12:51.44bastaif you do a direct transfer transferred person hears ringing, which is good since she has feedback on the call beying transferred
12:52.23bastabut on a hold she hears nothing, and sometimes they hangup, thinking the connection went lost
12:52.37bastamanx, what version of * ?
12:53.07bastamaybe it stopped working after an upgrade to 1.0.6 for me
12:57.26jakepdevanyone working with * outcalling?
12:58.02jakepdevdoes * succesfully detect disconnected/busy/no answer, etc?
13:02.30*** join/#asterisk jeffik (jefik@69.158.30.24)
13:02.45*** join/#asterisk delYsid (~user@delYsid.developer.debian)
13:02.59delYsidyay, my asterisk setup now works!
13:03.47tzangerexcellent
13:03.56*** join/#asterisk ckruetze (ckruetze@cpc3-cmbg7-5-0-cust100.cmbg.cable.ntl.com)
13:04.54delYsidsip proxy directly calls in to my 6970g
13:05.06delYsidI wonder though, how can I implement remote mailbox access now?
13:05.37delYsider, 7960g
13:07.25*** join/#asterisk iq (~iq@70-59-161-91.omah.qwest.net)
13:08.10ManxPowerjakepdev, Analog ports do not support detecting disconnected/busy/no answer
13:08.29ManxPowerwell, it can have a timeout, of course.
13:10.52tzangerManxPower: well they can detect disconnect but not busy/no answer
13:11.50ManxPowertzanger, Huh?
13:12.01ManxPowerANALOG FXO?
13:12.06jakepdevhttp://www.voip-info.org/wiki-NVLineDetect ???
13:12.09ManxPowerNuh uh!
13:12.34ManxPowerjakepdev, That's a 3rd party module and not part of Asterisk.  I do not know how well it works.
13:12.46jakepdevk
13:13.05jakepdevtnx
13:14.11ManxPowerjust get a non-analog connection like VoIP or PRI
13:14.36jakepdevthat may be a possibility - it's new client
13:14.51jakepdevthrough PRI - * can detect intercept tones?
13:15.06jakepdevbusy, no answer etc?
13:15.19jakepdevwithout NVLineDetect?
13:16.28ManxPowerjakepdev, yes.
13:17.00jakepdevgreat!  I'll push er. guide them in that direction
13:18.13Gand_DJWhat's the best codec to use for doing voip over 28.8k dialup? My fiance still has dialup in her area and I want to setup a softphone on her pc to link to my * server.
13:18.18tzangerManxPower: yes
13:18.42tzangerManxPower: CPD is certainly available IF YOUR TELCO OFFERS IT.  Bell Canada does battery reversal, and the FXO modules can see it
13:18.43vaewynAFKGand_DJ: g.729 or ilbc or sometimes GSM
13:18.46ManxPowerGand_DJ, none
13:18.58vaewynGand_DJ: and don't expect it to work
13:19.15vaewyn56k you have a prayer... 28.8 good luck
13:19.20Gand_DJheh
13:19.32ManxPowertzanger, CPD is not the same as "disconnected number"
13:19.54tzangerManxPower: oh I thought you meant disconnected as in hung up
13:19.57tzangermy apologies
13:19.58Gand_DJdoesn't g729 use 8kbps? that's only 1/3 of 28.8k dialup.
13:20.09tzangerthose are all inband audio and the callprogress just doesn't cut it at this point
13:20.16ManxPowerGand_DJ, You then have UDP overhead.
13:20.27tzangerManxPower: why should you not call hangup from h exten?
13:20.33tzanger(looking at your bounty post)
13:20.42ManxPowertzanger, Mostly because it's stupid.
13:20.47tzangerManxPower: so?
13:20.48vaewynok... the Norhell is on crack...  hooked the TE405P into our 5300 that is configured to talk to the Norhell... and voila... I have a connection...  plug it into the Norhell... and nothing
13:20.52tzangerManxPower: there are lots of stupid things people do
13:21.13ManxPowerBut also it will help generate errors when you accidently have _. that calls hangup
13:21.21tzangerI think it's far more of an issue that asterisk doesn't generate errors if you say "FunkyConfigParamThatDoesntExist=10e-98"
13:21.21ManxPowertzanger, Notice it's only a $10 bounty
13:21.45ManxPowertzanger, post a bounty for that
13:21.56tzangerI think that a better bounty would be whenever _. is used it posts "ARE YOU FUCKING INSANE?"
13:22.33tzangeror better, whenever _. matches a "special" exten like s,h,i or t
13:22.37vaewyntzanger: amen!
13:22.38jakepdevbut has an override
13:22.42jakepdevlike windows
13:22.46jakepdev"Are you sure..."
13:22.51jakepdevAre you really sure???"
13:23.04vaewyn"You do realize you are stupid?!?"
13:23.05tzangeractually _. would match o and a too
13:23.36vaewynhm... what's 'a' hadn't heard of that one
13:23.36tzangerpersonally I like having Hangup in h
13:23.46tzanger'a' is for voicemail when you hit '*'
13:23.50vaewyn'o' I use way to much
13:23.51*** join/#asterisk PCadach (~paul@www.east.telecom.kz)
13:23.56vaewynahhhh
13:23.58vaewynasterisk
13:23.59vaewynhehehe
13:24.17tzangeryou use Hangup when you want to make deadly sure that that's as far as that extension gets.  And if I have post-hangup stuff to do, I still end it with Hangup
13:24.37*** join/#asterisk moy (~kvirc@201.135.105.124)
13:25.14*** join/#asterisk blop (~blop@2001:6f8:204:33:bbbb:bbbb:bbbb:bbbb)
13:25.16blophi :)
13:26.19Gand_DJNon * related question.. anyone know what kind of codec msn & yahoo messenger use for audio connections?
13:28.34*** join/#asterisk funxion (~chatzilla@mtnuser.icgws.com)
13:28.51funxion|vulture| are you here?
13:29.24funxionanyone
13:29.27funxionanyone
13:29.31funxionbueller?
13:29.31mistrali have astrisk running here connected with a sip provider i seem to be able to dail out to voice mail and a talking clock but cannot dail full numbers. Also when i get an incoming call astrisk says something about no compatible codecs anyone know why this is ?
13:30.14funxionwhat codec are they connecting with?
13:30.26mistralin which connection ?
13:30.31funxioninbound
13:30.32inspiredcan asterisk 1.0.7 play gsm files in musiconhold?
13:30.36inspiredif so, how?
13:30.38mistraldont know
13:30.57mistralits actually an incoming call from a land line
13:31.10funxionhow is it hitting asterisk
13:31.15funxionfxo?
13:31.38mistralno there is a internet sip provider that allows incoming calls to a land number
13:31.55mistralas in the internet sip provider gives your a land line number
13:32.13*** join/#asterisk phpboy (~sj@tpr-165-239-114.telkomadsl.co.za)
13:32.18phpboyhello my loves
13:32.39mistrali can get it to work if i get x-lite soft phone to talk directly to the sip provider and it uses codec g711u
13:33.06foobosmistral, well force it ulaw then in asterisk. it prolly tries g729 and you don't have licesnse
13:33.17funxiondo you have g711u allowed
13:33.17mistralhow do i force it ?
13:33.32funxionallow it in sip.conf
13:33.45funxionmake it a higherpriority then g729
13:33.49phpboyleast cost call routing
13:33.53foobosmistral, in sip.conf disallow=all allow=ulaw
13:33.56mistraldisallow=all
13:33.56mistralallow=ulaw
13:33.56mistralallow=alaw
13:33.57phpboyis this a possibility with asterisk
13:33.57phpboy?
13:33.59mistrali have that
13:34.19foobosmistral, well then you can try "sip debug" in the commandline
13:34.34Hmmhesaysphpboy: why wouldn't it be?
13:34.48Hmmhesayslooks as though you are familiar with php, write an agi
13:34.51funxionanyone in here have a te110p installed?
13:35.01phpboyHmmhesays: agi
13:35.02phpboy?
13:35.04jakepdevfunxion - 2 of them
13:35.10Hmmhesays~agi
13:35.12jbotit has been said that agi is the Asterisk Gateway Interface...  similar to CGI for web applications AGI lets you script call control and access databases using your favorite language.  AGI wrappers are available for Python (pyst), Perl (astperl?) and other languages
13:35.12bugbotagi is assigned nothing and reported nothing.
13:35.13funxiondid you have any problems installing them
13:35.24jakepdevactually 3
13:35.28jakepdevnope - no problems
13:35.31funxionhmm
13:35.33funxionwhat os?
13:35.33jakepdevfit right in
13:35.46jakepdevCentOS 4
13:35.49funxionhmm
13:35.57funxioni have centos 4.3
13:36.07jakepdevshouldbe fine
13:36.12jakepdevwhat errors?
13:36.12funxiongetting modprobe error when I try to modprobe wcte11xp
13:36.38funxionI get /lib/modules/2.4.21-27.0.1.ELsmp/misc/wcte11xp.o: init_module: No such device
13:36.46funxionfollowing by some other errors
13:36.59jakepdevno such device... hmm...
13:37.07jakepdevirq sharing?
13:37.11funxion|vulture| tried to help me fix it yesterday
13:37.15funxionnot that I know of
13:37.25jakepdevtry digium
13:37.28funxionI updated zaptel source
13:37.32jakepdevthey offer free hardware install support
13:37.39jakepdev877-linux-me
13:37.42funxionrecompiled then tried to load modules again
13:37.44funxionyeah
13:37.45funxionkewl
13:37.48funxiondidnt know about that
13:37.50funxionthnx
13:37.52jakepdevnp
13:38.10ManxPowerfunxion, does lspci show the card?
13:38.14funxionyes
13:38.45funxion03:01.0 Network controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface
13:40.05Gand_DJHaving trouble decide on a voip provider to handle my * outbound. I'd prefer someone who has access to cdn DID, but not needed (I have a pots line ppl can call me on)
13:40.15jakepdevGand - try a few
13:40.19jakepdevthey're cheap
13:40.26delYsidI have exten => 666,2,Voicemail(u666) in my extensions.conf and the mailbox works if no one answers, but how can I do remote mailbox access?
13:41.21jakepdevthey're is no one size fits all VOIP provider
13:41.27jakepdevtherer
13:41.31jakepdevah f it
13:41.34RestLessGeminidoes anyone knows about any modem other Intel Ambiant, which can be used as fxo/fxs
13:41.56vaewynno "modem" can be used as FXS
13:41.56jakepdevya cheap bastard :) Just get a real FXO/FXS
13:42.10jakepdevthe SPA 300 goes for $100
13:42.13jakepdev3000
13:43.10RestLessGeminijakepdev: i am sitting in pakistan and I'm using Intel Ambiant, and the bad thing is, they are short in the market :)
13:43.24jakepdevi hear ya
13:43.25ManxPowerSome of the other SPAs can be $65 or less
13:43.42ManxPowerRestLessGemini, The ambient chipset is no longer made.
13:43.59*** join/#asterisk ccfiel (ccfiel@210.5.72.36)
13:44.06ccfielhello ppl..
13:44.22ccfielcan somebody help me with my problem.. a remote iax connection connect to my * box ...and communicate with a local iax connection.. before a few second at least 10 seconds there is a lag in commnunication...after a while lag will disapper...what's would be the problem..can somebody help me with this? :(
13:44.29RestLessGeminiManxPower: Yes i know, this is why I asked if any of you knows about any other compatable card like IA
13:44.39ManxPowerRestLessGemini, No.  there are none.
13:44.51RestLessGeminiManxPower: thanks
13:45.48ccfielhello  can somebody help me.. :(
13:46.43ManxPowerccfiel, It sounds like a router or ISP problem.
13:48.26ccfielmanxpower..i thinks it's not...because when a iax remote connection connect to my * and call a sip local extension..there is no lag at all..
13:48.38Gand_DJThe main ones that have my attention are broadvoice, link2voip, and teliax.
13:48.55ManxPowerccfiel, Your problem is very strange and I've never heard of such a problem that was not caused by a network problem
13:49.31ccfielthis happen with iax remote connection --> * server ----> iax local extension
13:49.52ccfielbut when i do ... iax remote connection --> * server ----> sip local extension it works well..
13:50.46ManxPowerthen the only difference is the sip local connection
13:51.04ManxPowerSoftphones are well known for causing latency problems
13:51.21ManxPower~google site:lists.digium.com softphone latency
13:51.22bugbotgoogle site:lists.digium.com softphone latency is assigned nothing and reported nothing.
13:51.52jakepdevGand - I tried 4 before I came to a conclusion - maybe cost me abut $20
13:52.03Gand_DJWho you currently use?
13:52.06jakepdevbut was worth it
13:52.34jakepdevin my area, voicepulse provided the best results
13:52.43jakepdevbut it could be different for you
13:52.57jakepdevvoicepulse just happens to have a POP close to here
13:53.15Gand_DJI live in canada, so most will probably be similiar..lol
13:53.22Gand_DJI found one called voipforcanada.com
13:53.29Gand_DJbut the site layout makes me wonder about it
13:53.37tzangerI use nufone almost exclusively (I'm canadian)
13:53.44tzangerpush about 5000 min/mo through them
13:53.45ManxPowerI use Teliax now.
13:53.50jakepdeveasiest way to find out is to try it
13:54.00jakepdevmake a few test calls
13:54.18Gand_DJI've thought of looking at nufone but they are still doing upgrades.. heh
13:54.21*** join/#asterisk [TK]D-Fender (~joe@4.67.252.216.dsl1.colba.net)
13:54.34tzangerGand_DJ: msg shido6 and he can set you up
13:54.36jakepdevif you pay via PayPal, they should accept you as a new usr
13:54.44vaewynThey rock if you can stand the wait
13:55.00tzangerjakepdev: unfortunately a few test calls will only reveal if they're poor at that time... mind you the poor ones, if they are truly poor will be crap all the time
13:55.33jakepdevright - but if you space the test calls apart...
13:55.47jakepdevover a few days, different times, etc
13:55.48Gand_DJwell, before I signup with someone (such as nufone), I'd want to read their rates & fees... and none of that is on their site currently.
13:56.14*** join/#asterisk zoa (~zoa@pirus.securax.be)
13:56.21tzangerGand_DJ: their rates are available on the main site
13:56.28jakepdevhttp://www.nufone.net/rates.csv
13:56.35tzangerfees are just their rates; there is no signup fee, there is no strange shit
13:56.43vaewyn2cents/min USA and such
13:56.45tzangerpush as many concurrent calls through them as you can handle, it's all the same per-minute
13:56.58tzangervaewyn: yeah but cdn term through nufone is slightly cheaper :-)
13:57.13vaewynyep
13:58.01Gand_DJWhat about connection fees, taxes, etc. (read teliax charges 2c connect fee per pots call)
13:58.19[TK]D-Fenderhttp://www.voipforcanada.com < $.01 min CDN for outbound
13:58.26Gand_DJSeems some of the voip providers want to charge you state taxes, even though it's illegal.
13:58.37nvrsworkhas anyone successfully gotten asterisk 1.0.7 gentoo port compiled on 2005.0?
13:59.29Gand_DJ[TK]D-Fender, yeah I like that part.... but the site makes me wonder how trustworthy they are. alot of stuff doesn't work for links.
13:59.56[TK]D-FenderGand_DJ : I noticed a lack of finish as well, but for the price, looks worth it.
13:59.57Gand_DJI have a free / test acct through them... also with simpletelecom right now
14:00.03[TK]D-Fenderesp for a trial
14:00.42Gand_DJhaven't setup the v4c acct yet as they only emailed me softphone settings.. nothing for * setup
14:00.59*** join/#asterisk shmaltz (~chatzilla@69.28.255.210)
14:04.29Gand_DJhrm... what's the difference between getting a DID and leasing a DID?
14:04.37Gand_DJeither way you don't own it.
14:04.41Gand_DJI think
14:05.14*** join/#asterisk malbech (Phils@m199.net81-66-243.noos.fr)
14:06.07delYsidahh, 'a' was what I needed
14:09.41*** join/#asterisk zaptel (~just@216.194.173.2)
14:12.14*** join/#asterisk wazquis (~akv@lnxbx.dk)
14:12.28*** join/#asterisk roamer323 (~sing@toronto-HSE-ppp4074896.sympatico.ca)
14:13.47wazquishey, is it possible to send a text message from one asterisk to an other?
14:16.11moydoes anybody knows how can i make that IAX2 use other port than 4569??
14:16.29moyi tryied modifying iax.conf port parameter, but does not work
14:18.04*** join/#asterisk bannerman (~bannerman@209.216.176.42)
14:19.32*** join/#asterisk fugitivo (~ajf@201.255.105.220)
14:19.35fugitivohello
14:21.25*** join/#asterisk gmcinnes (~gmcinnes@Toronto-HSE-ppp3681363.sympatico.ca)
14:22.07ManxPowermoy, you have to modify the asterisk source to change the port from 4569
14:22.17bjohnsonGand_DJ: some voip providers allow you to transfer out your DID .. others do not
14:22.32bannermanshido6: Did you guys make some changes that allowed G.729 to start working? Because, this morning, it's working fine...
14:22.56Gand_DJbannerman, you using firefly for g729?
14:22.59bannermanno
14:23.10ManxPowerI didn't think Nufone supported G729
14:23.17bannermanThey do.
14:23.23ManxPowerofficially?
14:23.31Gand_DJThey might not support transcoding.. but maybe passthrough
14:23.45ManxPowerGand_DJ, You can't pass thru to the PSTN
14:24.52gmcinnesAnyone used asterisk with a Nortel Meridian Option 61-C switch?
14:24.55Gand_DJRight, but does nufone directly link to pstn, or they link to another backend voip that goes direct to pstn?
14:28.38*** join/#asterisk carlos-d-man (~carlos@201.135.87.60)
14:28.41malbechI search a softswitch for a good price but it's very diificult to find one ... no ?
14:28.43carlos-d-manhi guys
14:30.18*** join/#asterisk girabraz (~christian@200.121.129.178)
14:31.15*** join/#asterisk olivier_ (~olivier_@obs92-4-82-239-116-113.fbx.proxad.net)
14:32.07*** join/#asterisk Dovid (~hirisk@pool-138-89-169-188.mad.east.verizon.net)
14:33.01*** join/#asterisk twisted[work] (~twisted@twisted-professional-pdpc.developer.and.friend.of.asterisk)
14:33.01*** mode/#asterisk [+o twisted[work]] by ChanServ
14:33.11jalsothi
14:33.49jalsotdoes anybody know what can be the problem with IAX2->*->*->PSTN while SIP->*->*->PSTN works fine?
14:34.42*** join/#asterisk bkw_ (~bkw_@bkw.developer.and.friend.of.asterisk)
14:34.42*** mode/#asterisk [+o bkw_] by ChanServ
14:35.26bannermanI think they officially do. Jeremy and Greg both said that they have digium G.729 licenses.
14:36.01Dovidcan anyone tell me the diffrence in version 1.0.6 to 1.0.7 ?
14:37.23bannerman0.0.1
14:37.27bannerman;-)
14:38.05Dovidhehe
14:38.07Dovidseriousley
14:38.17bannermanGand_DJ: Pretty sure Nufone does their own termination.
14:38.41robl^I've used g729 with Nufon
14:38.54bannermanrobl:^: Have you been using it this last week?
14:38.55Gand_DJhrm.. ne1 here have asterisk@home working with fwd for incoming? I keep getting busy signal when calling into by * box.
14:39.02carlos-d-manhow may I connect dialx or any other sip phone to asterisk?
14:39.11Gand_DJI can finally call out using fwd.
14:39.44robl^bannerman, no.  I've switched to gsm for Nufone.  I only use g729 now for one remote extension on a low bandwidth connection
14:39.52inspiredis it safe to use mpg123 for music on hold?
14:40.04inspiredI can't get the fake mpg123 script to work
14:40.58robl^inspired, I've used mpg123 for more than a year now.  I have little trouble with it.  Asterisk used to leave it running when you stopped, but not anymore
14:41.23inspiredok
14:41.34ManxPowerDovid, It's all listed in the asterisk-cvs mailing list (only messages with Tag 1-0 apply to 1.0.x) .  There is also a changelog included in the tarball
14:42.02ManxPower~mailinglist
14:42.03jbothmm... mailinglist is Search Asterisk mailing lists by prepending site:lists.digium.com to your Google search.  Browse the mailing list archive at http://lists.digium.com/
14:42.03bugbotmailinglist is assigned nothing and reported nothing.
14:42.14ManxPowerbugbot needs to be melted down.
14:42.14inspiredis it possible to convert a .raw file back to .mp3?
14:42.27Dovidis it stable ?
14:42.27ManxPowerinspired, That is not an Asterisk question.
14:42.36robl^inspired, the worst thing with mpg123 is that it uses a bit of memory and cpu.. but not bad unless you have a large number of calls on a server
14:42.41ManxPowerDovid, Is what stable?
14:43.05Dovid1.0.7
14:43.28ManxPowerDovid, There have been some reports of SIP issues with 1.0.7 but I've not seen any major issues with 1.0.7.
14:43.35johnnybIs there a timeline for releasing 1.1?
14:43.35ManxPowerDovid, You don't read the mailing lists, do you?
14:43.48Dovidi get them, dont have time to read em
14:43.51robl^Dovid, version 1.0.x is ALL considered stable.
14:43.55ManxPowerjohnnyb, sometime before the sun burns out and the universe implodes.
14:44.07johnnybManxPower: so, like, this year?
14:44.19ManxPowerDovid, but you have time to ask OTHER people to read them for you and then distil the information for you.
14:45.07*** join/#asterisk mhnoyes (~mhnoyes@user-2ivfj1c.dialup.mindspring.com)
14:45.26pgpkeysOH NO! THE UNIVERSE IS IMPLODING?? Man! now they tell me, and i just bought a new planet for the kids too.
14:45.28pgpkeysdamn it!
14:46.03ManxPowerI SUSPECT 1.1 will be released in 6 - 9 months, but I have no proof.
14:46.05*** join/#asterisk IronHelix (~irc@ool-182c3fe9.dyn.optonline.net)
14:46.07robl^pgpkeys, they have a 30 day return policy.  just a 15% restocking fee
14:46.23file[laptop]ManxPower: proof pfft, who needs that?!?
14:46.26pgpkeyswell then that's fine.
14:46.37inspiredrobl^: is there a way to do music on hold without using so much cpu?
14:46.40fugitivoanyone is using cisco ip phones?
14:46.43ManxPowerfile[laptop], Well, I have plenty of poof, but no proof.
14:46.45pgpkeysthat's a measley $1,500,000,000 for restock. i can deal with that
14:48.22robl^inspired, really, the CPU hit is not bad.  I have never had any performance problems.  one tip is to use only one  moh class.  if you have it playing multiple playlists concurrently, you use more CPU and memory
14:50.39*** join/#asterisk _THEEND_ (~DrEaM@80.18.184.226)
14:51.00_THEEND_which fax software is better with asterisk?
14:51.40pgpkeysthe one you write
14:51.53_THEEND_?
14:51.56_THEEND_sorry?
14:51.56vaewynhylafax or spandsp
14:52.07_THEEND_i'm looking for hylafax
14:52.10inspiredrobl^: ah, I see
14:52.18pgpkeys_THEEND_: my fault. I read taht as better THAN not better WITH
14:52.54_THEEND_:)
14:53.29_THEEND_but Hylafax interfacing with asterisk it's less documentated
14:55.34ManxPowerhylafax requires a modem
14:56.19_THEEND_also only for receving fax?
14:56.50*** join/#asterisk veto (~upirc@015-823-051.area5.spcsdns.net)
14:58.08*** part/#asterisk veto (~upirc@015-823-051.area5.spcsdns.net)
14:58.40*** join/#asterisk vooduhal (~christoph@67.19.25.178)
15:00.11vooduhalHey, can anyone answer a few res_perl questions?  Specifically, where are the channel status constants and what is the proper way of replacing ast_readstring since I can seem to find it in res_perl?
15:01.15_THEEND_hmm
15:01.38ManxPowerI want a nap.
15:01.49_THEEND_but if i have an hfc-s chip based isdn card for calling with isdn can i use that card as fax modem?
15:02.09vooduhalNM, the channel state problem I just found where they are defined.
15:05.19*** join/#asterisk eivindtr (~eivindtr@062016241059.customer.alfanett.no)
15:06.23gmcinnesHi, all you helpful little elves ;)
15:06.41gmcinnesdoes anyone know anything about Meridian 1 systems?
15:06.52gmcinnes*ouch
15:07.27gmcinnesI know its blasphemous to ask about such things on an asterisk channel :)
15:08.06*** join/#asterisk pointer (pointer@aj.catt.com)
15:08.07Moonwickonly if it's not in the context of "how do I connect asterisk to it in order for asterisk to eventually replace it" :)
15:08.23gmcinnesMoonwick: that's it exactly :)
15:09.51gmcinnesI need to know what kind of signalling it can do, and whether I can connect a TDM400P card to it without melting the card.
15:11.50gmcinnesah. the silence is deafening :)
15:12.49MoonwickPBX knowledge is hard to come by in here unless you're lucky
15:13.02Moonwickfwiw, I doubt you'd melt the card
15:13.17tzangermcnobody: what are you trying to do?
15:13.22Moonwickbut beyond that, I'm not sure how much cooperation you'll be able to coerce out of the two systems
15:13.30ManxPowerI don't give a flying rat about the brand of PBX.  If you tell me what port types you have available, I can tell you what might work.
15:13.34tzangeryou can't use a TDM card with a Norstar system unless you're going through an ATA
15:13.48jakepdevwhat's the easist way to do a simple lookup table in the * dialplan?  i.e.  if IN=1, set OUT =9
15:13.56ManxPowertzanger, unless you use FXS ports on the TDM card and connect them to the CO ports on the PBX
15:14.06tzangerManxPower: well yes you could do that too :-)
15:14.09ManxPowerjakepdev, gotoif
15:14.16*** part/#asterisk pointer (pointer@aj.catt.com)
15:14.27gmcinnestzanger: what's an ATA ?
15:14.47tzangernorstar ATA or ATA2
15:14.57ManxPowergmcinnes, it sort of converts the nortel phone ports into an analog port.  notice the "sort of"
15:14.57tzangerlets you connect a regular phone/fax as an extension
15:15.04*** part/#asterisk sympad (~Misha@195.138.127.98)
15:15.17ManxPowerI was never able to have nortel digital phones to send DTMF out an ATA
15:15.19tzangerI have asterisk between our Norstar MICS and the PRI
15:15.22tzangerworks well
15:15.33ManxPowertzanger, Yeah, well that's the RIGHT way to do it. 8-)
15:15.33jakepdevManxPower - that creates a ton of branches.  I'm looking for something I can do in a single line if possible
15:15.37tzangerManxPower: intersting; I've never had to try that.  it does work the opposite way :-)
15:15.42ManxPowerjakepdev, can't be done.
15:16.04ManxPowertzanger, yes, but the ata ALSO doesn't seem to provide CPD
15:16.10tzangerManxPower: no it doesn't
15:16.19gmcinnesI built an ivr on asterisk. It only needs to recieve incoming dtmf.  It was based on being connected to POTS lines, but now they want it on a DID from their Option 61 C switch
15:16.32tzangerI've heard rumour that you should be able to detect a 'D' tone on hangup but have not played with it
15:16.40ManxPowergmcinnes, use a PRI port on the Nortel if you can
15:17.07vooduhalSo anyone with ideas on the proper replacement for ast_readstring in res_perl?
15:17.14tzangerManxPower: will a dialogic card not work?  Ithought that was an option (an expensive one)
15:17.18gmcinnesManxPower: And something like a TE410 on the asterisk box?
15:17.27tzangergmcinnes: well a TE110P would work just fine
15:17.30ManxPowergmcinnes, yes.
15:17.56gmcinnesManxPower: but there's no easy way to connect a DID to a TDM400P ?
15:18.05ManxPowerThe best way is Telco(T-1/PRI)<-><T-1/PRI)PBX(T-1/PRI)<->(T-1/PRI)Asterisk
15:18.13tzangergmcinnes: with an ATA or ATA2, but you won't know if it works properly without testing
15:18.24gmcinnestzanger: Forgive my dense-ness.  This pbx stuff is all new to me.
15:18.38tzangerManxPower: well not necessarily
15:18.38ManxPowergmcinnes, yes, but you will have all sorts of problems unless you are a telco, asterisk, and Nortel expert.
15:18.42pgpkeyshehe i'm still learning to set mine up
15:18.45gmcinnesManxPower: I am none of the above.
15:18.46pgpkeysnever set up a pbx before
15:18.47ManxPowertzanger, Well for some values of "best"
15:18.51ManxPower8-0
15:19.01tzangerManxPower: the Norstar PRIs will not work as extensions without an expensive MCDN license key... and there is no support for MCDN or SL1 in libpri
15:19.28tzangeryou can create a 3-digit or 4-digit extension and try ot route it out (I do that now) but it's hacky
15:19.28ManxPowerAn alternative is Telco(T-1/PRI)<->(T-1/PRI)Asterisk(T-1/PRI)<->(T-1/PRI)PBX
15:19.36*** join/#asterisk UK_Mister (~fred@host81-137-167-89.in-addr.btopenworld.com)
15:19.45UK_Misterhello all
15:20.15ManxPowerAn alternative is Telco(T-1/PRI)<->(T-1/PRI)Asterisk(T-1/PRI)<->(T-1/PRI)ChannelBank(ANALOG)<->(ANALOG)PBX
15:20.21*** join/#asterisk Rick_Hunter (~rhunter@05-136.008.popsite.net)
15:21.03ManxPowerReworded: In a Perfect World he best way is Telco(T-1/PRI)<-><T-1/PRI)PBX(T-1/PRI)<->(T-1/PRI)Asterisk
15:21.06inspiredrobl^: what bitrate should I use?
15:21.12inspiredand sample rate?
15:21.40ManxPowerNortels do not exist in a Perfect World 8-)
15:21.44gmcinnestzanger: does it matter that I don't ever need to route a call out from the asterisk box?
15:21.44UK_Misteri've been playing around with asterisk for a few weeks, i'm about to install it at one of our small sites, anyone recommend some good SIP handsets?
15:22.11ManxPowerUK_Mister, Polycom IP 500 or SIPura SPA-841
15:22.27UK_Misterthnx ManxPower
15:22.44ManxPowerThe Polycom IP 300 is you MUST have PoE.  Otherwise use IP500's for people that need speakerphone, SPA-841 for people that don't need speakerphone.
15:23.18UK_Misterdo they all have programable soft keys?
15:23.32gmcinnesManxPower: no, they certainly don't.
15:23.44*** join/#asterisk dasuberdavid (~david@207.111.174.1)
15:24.28Gand_DJanyone here used Simpletelecom? so far 1 person doesn't like them..
15:24.50UK_Misteri'm looking for some cheap and cheerfull single line display and a couple of executive with soft keys etc
15:25.56ManxPowerUK_Mister, The polycoms do, but I never managed to make them work.  Didn't try all THAT hard.
15:26.19ManxPowerUK_Mister, buy one of each.  Test.
15:26.43*** join/#asterisk Nix (~Nix@dsl81-214-65337.adsl.ttnet.net.tr)
15:26.55UK_Misterwill do, thnx again
15:26.56malbechI search a softswitch for a good price but it's very diificult to find one ... no ?
15:27.57UK_Misteri've got a grandstream handset, it works perfectly, just a bit to simple
15:32.09ManxPowerHave y'all heard that lighters will be banned on USA airplanes soon?
15:32.14tzangerManxPower: I prefer Telco - PRI - * - PRI - PBX
15:32.59ManxPowerI wonder what they do with all the lighters that are taken.  I wonder if I could just go down to the airport anytihng I run out of lighers and get a bag of them from airport security.
15:33.09ManxPoweranything = anytime
15:34.49robl^ManxPower, they give them away for every used fingernail clipper you buy :)
15:35.04_THEEND_none has interfaced hylafax with asterisk?
15:35.11Moonwickit'd be a fun prank to run some sort of "lighter exchange" program
15:35.19Moonwickcoordinate with friends in a few cities
15:35.34Moonwickand have them set up shabby little stands in front of the entrance to security at a few airports
15:36.11Moonwickwhere people getting ready to go somewhere can trade lighters and  nail clippers for a coupon to receive same at their destination airport
15:36.18ManxPower_THEEND_, Sure!  Plug the modem Hylafax uses into an FXS port of Asterisk.  Done!
15:36.40ManxPowerrobl^, I think the "h" bounty has generated more messages than any other bounty I've ever posted.
15:36.42_THEEND_uhm...
15:37.06ManxPower_THEEND_, Other than that stop complaining and write support.  you'll find out it's VERY VERY tough.
15:37.34ManxPowerYou'll have to basically write a softmodem.  You could use the DSP from spamdsp.
15:37.46*** join/#asterisk bkw_ (~brian@bkw.developer.and.friend.of.asterisk)
15:37.46*** mode/#asterisk [+o bkw_] by ChanServ
15:37.48Gand_DJkewl.... voipforcanada supports G729
15:38.13ManxPowerIn fact you might only have to write an AT command set emulator and serial port emulator.
15:38.32tzangerManxPower: :-)
15:38.44tzangeractually I was going ot write a serila port telnet server
15:38.53ManxPowerThat should be EASY compared to coppiece's software DSP library
15:38.56tzangerbecause I have a Lucent Max that you can telnet to port 9000 and get a modem
15:39.01tzangerplug THAT in to HylaFax
15:39.24ManxPowertzafrir, there are a couple of things available to do that with Linux
15:39.52tzangerManxPower: I have only found the reverse
15:40.25tzangertelnet servers that connect to serial ports, not serial port emulators that really telnet
15:40.53ManxPowertzanger, Oh!  sorry.  wasn't reading carefully enough
15:41.05ManxPowertzafrir, serredir?
15:41.09ManxPowermodempoold
15:41.13ManxPowermpoold
15:41.15tzangerthat's the idea yeah
15:41.18ManxPowerthere's an RFC for it
15:41.19tzangerbut it wouldn't be a daemon
15:41.21*** join/#asterisk jcollie (~jcollie@lt16586.campus.dmacc.edu)
15:41.29tzangeror I guess it would be
15:41.40jcollie~seen anthm
15:41.41jbotanthm <~anthmct@CPE-69-76-83-52.wi.res.rr.com> was last seen on IRC in channel #asterisk, 8d 22h 28m 58s ago, saying: 'at cluecon!'.
15:41.42bugbotseen anthm is assigned nothing and reported nothing.
15:41.50tzangerbasically opening /dev/ttyT0->T23 would telnet to w.x.y.z port 9000
15:42.00ManxPower*nod*
15:42.27*** part/#asterisk jcollie (~jcollie@lt16586.campus.dmacc.edu)
15:42.32tzangerwell if nothing else, ManxPower, your h,1,Hangup bounty has created discussion :-)
15:42.33psycodadhow do i initiate a supervised call transfer (which keys) ?
15:42.44ManxPowertzanger, Yup!
15:42.53ManxPowerpsycodad, WHAT PHONE????
15:43.36psycodadManxPower: i.e. a sip phone, i can do #extension and get blind transfer, but how to talk to the target party first ?
15:44.50tzangerpsycodad: which phone
15:44.59tzangerspecifics man, speicifics
15:45.02ManxPowerpsycodad, I didn't ask you what type of phone.  What BRAND of phone.
15:45.03cypromishow do you know the target party wants to talk to you ?
15:45.09psycodad2$ analog phone on ATA
15:45.15ManxPowerpsycodad, #transfer is an ugly evil hack and you will go to hell for using it.
15:45.21ManxPowerpsycodad, FLASH
15:45.23psycodad;-)
15:45.28tzangerManxPower: hahaha
15:45.35ManxPowerpsycodad, or read the damn documentation for the ATA.  That's what it's there for.
15:45.40jlewisIf I have a couple of POTS ports on a TDM400P, * isn't smart enough to find a port (in a group) with working dialtone for making outgoing calls is it?
15:45.41tzangeryeah I am a huge fan of hookflash for zap...  but I odn't know how well that works on an ATA
15:45.59tzangerjlewis: Asterisk is a PBX, not an aswering machine
15:46.04ManxPowertzanger, Since all this stuff is handled totally by the ATA it really depends on the brand of ATA.
15:46.07tzangerput it in charge of the phone lines and it will do its job properly
15:46.26ManxPowerbut since psycodad seems incapable of telling us the brand of ATA I think he needs to be put on my /ignore list.
15:46.27jlewistzanger: don't think you understood the question...let me restate
15:46.36tzangerjlewis: I understand the question
15:46.37cypromisso now that you know that SIP ATA's suck
15:46.41cypromiswhat will you do ?
15:46.43cypromis*gg
15:46.48tzangeryou have a number of POTS lines that * is sharing with phones outside of asterisk
15:46.49ManxPowerjlewis, Asterisk cannot detect dialtone on a line.
15:47.10tzangerin effect, you want * to act as an aswering machine does, subservient to everything else on the line
15:47.13bjohnsonGand_DJ: I'd like to here how you find the voipforcanada service.  They seem to have started up recently (domain registered in Feb 2005)
15:47.17mrunixis ftp.asterisk.org being wonky?  it always times out on me from work and home
15:47.27Wonka.oO( mmh? )
15:47.30jlewisso if I have a Zap group and 1 line goes dead, outgoing calls using Zap/g1/number will break until the bad channel is unconfigured from the group?
15:47.35psycodadManxPower: I have a few Zyxel 2002, a Zap IF and Zyxel 2000W as well as some softphones
15:47.44tzangerjlewis: asteirsk is a PBX; it does not play nicely "in parallel" with other equipment on the FXO lines
15:47.46ManxPowertzanger, You go to the 17.25th level of hell.  That's where you are surounded by Polycom IP 600's, Cisco 7960G's and high end SNOM phones.  All JUST out of reach.
15:47.49tzangerjlewis: and in the case of a line going dead... no
15:47.52Gand_DJbjohnson, you can call toll-free # for free
15:48.02Gand_DJjust signup for a an acct...
15:48.04tzangerManxPower: hahaha wouldn't be bad for me...  I don't do SIP
15:48.08jlewisok...that's the way it appeared...just wanted to be sure I wasn't missing something
15:48.18ManxPowerpsycodad, The correct way to do a transfer is to read the documentation for the device you are using.
15:48.20cypromisManxPower: fine, I'll take my swissvoice with me
15:48.26Gand_DJI signed up & got my sip info.
15:48.31jlewisit'll just use the first available channel...whether there's a good line there or not
15:48.35Gand_DJalso a v4c extension #
15:48.37*** join/#asterisk RoyK (~roy@host-81-191-165-149.bluecom.no)
15:48.38RoyKv
15:48.50cypromisw
15:48.50tzangerjlewis: now "go dead" meaning "not plugged in" or "no battery voltage" -- * does see that
15:49.11psycodadManxPower: ok, I'll try that, thnx so far... but anyway, is there a way for supervised transfer via gets-me-to-hell-pound-hack ?
15:49.25cypromisdepends
15:49.27Gand_DJthey seem to accept Ulaw, Alaw, GSM, and G729
15:49.29cypromisthere where some apps for that
15:49.31cypromisand some patches
15:49.31ManxPowerpsycodad, only in CVS-HEAD, but I don't think it works (maybe its been fixed)
15:49.33Gand_DJnothing else
15:49.37tzangerpsycodad: park it, call the 2nd exten and tell them it's on 701. :-)
15:49.38cypromisand they  where allways broken 2-3 days later
15:49.39cypromis:)
15:49.39jlewisin the case I was testing, we only had one line plugged in, and it happened to be the last one...outdial via Zap/g1 would pick channels with no lines attached
15:49.45bjohnsontzanger: actually, my SPA running parallel to my answering machine copperates nicely .. but that is a feature of that ATA .. not asterisk
15:49.56tzangerjlewis: odd, what card?  TDM400P?  It should throw red alarm on nonconnected lines
15:49.57bjohnsonjlewis: ^^
15:50.03tzangerbjohnson: correct
15:50.13jlewisit is a TDM400P
15:50.18ManxPowertzanger, It should, but I don't think I've ever actually seen a TDM400P go into red alarm
15:50.18bjohnsonexactly
15:50.24tzangerManxPower: ahh...
15:50.43ManxPowerjlewis, Asterisk SHOULD detect if there's no line plugged in (i.e. no voltage), but I have never actually TESTED that.
15:51.04Gand_DJbjohnson, I tried calling a toll-free number.. and audio seems to cut out after 6 seconds..lol
15:51.06psycodadtzanger: okay, that sounds not too bad for the start ;-) Anyway the Userguide for Zyxel 2002 sucks big time, no mention of transfering anything, I guess they don't think you ever get a call on this crap ;-)
15:51.07Gand_DJtrying another one
15:51.32Gand_DJbjohnson, if you signup on them.. then we can try calling eachother for quality test
15:51.48psycodadtzanger: how do I park the call...sorry for sounding stupid ... I have a printed asterisk doc in front of me and spidered voip-info.org for the last 3 days ;-)
15:51.51jlewiszttool shows no alarms on the TDM400Ps (there are 2 4-port boards installed) even though only the last channel on second card has a line
15:51.58Dovidhi
15:51.59bjohnsonI've already got about 6 voip accounts .. I'm not signing up unless I think they offer good service
15:52.03Dovidi am new to asterisk
15:52.04Gand_DJhehehe
15:52.06Gand_DJok
15:52.15tzangerpsycodad: find yourself valetparking or supervaletparking, that's what you want
15:52.18Dovidwhat is fxo for and what is fxs for ?
15:52.23shido6good, Dovid
15:52.28tzangerDovid: FXO means you plug the Central Office into it
15:52.35tzangerFXS means you plug a telephone SET into it
15:52.35Dovidfxo = pstn
15:52.36Dovid?
15:52.40Dovidah
15:52.42bjohnson~fxofxs
15:52.43jbotmethinks fxofxs is An FXO port expects to receive dialtone and receive ring voltage.  An FXS port expects to provide dialtone and provide ring voltage.
15:52.43bugbotfxofxs is assigned nothing and reported nothing.
15:52.46tzangerFXO means goes to PSTN (O)ffice
15:52.52jlewishttp://www.digium.com/index.php?menu=fxsvfxo
15:52.53tzangerFXS means goes to tleephone (S)tation
15:52.54bjohnsonwell .. not always
15:52.56Dovidso fxs is to a fax machine or pts phonr
15:53.00bjohnsonwell .. not always
15:53.11tzangerbjohnson: for all intents and purposes, yes
15:53.25cypromismost itents and purposes
15:53.35bjohnsonhehe .. yes most
15:53.37cypromismy phones mostly go to NT mode BRI ports
15:53.43tzangerbjohnson: yes you can conjure up some oddball scenarios but don't confuse the poor guy, at least wait until he's discovering why exten _. is bad
15:53.48bjohnsongets confusing when intercting with another pbx
15:53.51*** join/#asterisk kant (~bernd@metrored-gw.tropicohn.com)
15:53.54psycodadtzanger: thnx found it...will try that way... and if anybody has Zyxel 2002 ATAs : How do you transfer calls (except with #)
15:54.02*** join/#asterisk bonez41 (~aint@c-67-166-77-14.hsd1.ut.comcast.net)
15:54.03*** join/#asterisk CoolAcid (~jk@216.99.98.39)
15:54.34*** join/#asterisk tessier (~treed@210.245.102.159)
15:54.56jakepdevis there a way in 1.0.7 stable to send DTMF digits?
15:55.04jakepdevto the same trunk
15:55.10tzangerjakepdev: SendDTMF()?
15:55.24jakepdevin HEAD only I think
15:55.45file[mac]nope it's in stable too...
15:55.47Gand_DJI appear to be the 26th person to signup for voipforcanada... based on the owners ex # being 8000 and I am 8026 :)
15:55.48tzangerjakepdev: ahh
15:56.07Gand_DJI checked their paypal log acct, and so far no payments were made to them
15:56.44ManxPowerjakepdev, HUH?  I do that ALL THE TIME.
15:56.53jakepdevi call flash - work
15:56.55jakepdevworks
15:57.06jakepdevthen call SendDTMF - and it doesn't come up
15:57.24ManxPowerjakepdev, stop being so lazy.  "send dtmf digits" can mean about 450 different things.  Make sure people understand what you mean.
15:57.34jakepdevwhat are you talking about
15:57.34jakepdev?
15:57.41jakepdevSendDTMF is what I'm looking for
15:58.00`SauronHehn
15:58.05`SauronFresh from NANOG:
15:58.11`Sauronyou wanna see bad second grade playground behavior, try the
15:58.12`Sauronasterisk mailing list.
15:58.27jakepdevjust isn't running the cmd
15:58.42ManxPowerI can send DTMF digits out a Zap FXO to dial a number.  I can send DTMF using D() in Dial.  I can SendDTMF to the caller on an inbound call that's in an IVR, I can send DTMF while on a call to interact with an outside IVR.
15:58.45ManxPowerShall I continue?
15:58.49jakepdevnope
15:59.02jakepdevcause I'm the one with the error :)
15:59.09ManxPowerOh, I can send dtmf to the callee using the M() option and a macro during a dial
15:59.24file[laptop]ManxPower: How sexy!
15:59.34Gand_DJhrm... ne1 using *@home and have fwd setup for incoming? I can call out (finally), but get busy signal for incoming.
15:59.37ManxPowerjakepdev, what makes you think SendDTMF is the correct way to do what you want?
16:00.42jakepdevjust want to Flash the Send didts to pass back to the host switch
16:00.42tzangerManxPower: well you could use playtones but that's pretty hardcore :-)
16:00.42ManxPowerOh!  Yes.  You can use Playtones to send dtmf too!
16:00.43*** join/#asterisk tzafrir_laptop (~tzafrir@bzq-179-40-134.cust.bezeqint.net)
16:00.43jakepdevwas hoping not to go that route
16:00.43*** join/#asterisk jf_ (~jeanfranc@HSE-Montreal-ppp332742.sympatico.ca)
16:00.43ManxPowerOK, maybe not 450 ways, but more ways than you can ever imagine, Grasshopper
16:00.57jakepdevtnx - 1 working way is enough
16:01.09jf_I've just installed  * on a new system, i get these error, who can help me
16:01.09jf_Apr 14 12:06:49 WARNING[9080]: chan_iax2.c:7478 load_module: Unable to open IAX timing interface: No such file or directory
16:01.10*** join/#asterisk Marlow (~martin@cerberus.bluetree.ie)
16:01.16jakepdevmaybe it's failing on Flash
16:01.34jf_<PROTECTED>
16:01.42tzangerjf_: read the zaptel README
16:01.55jf_read it already
16:02.03ManxPowerjakepdev, Explain what you want to ACCOMPLISH.
16:02.07*** join/#asterisk sean (~sean@iconoclast.caedmon.net)
16:02.27jakepdevjust want to Flash then Send DTMF to pass back the call to the host switch
16:02.31gmcinnesjf_:  what distro are you using?
16:02.38ManxPowerYou are asking "how do I chop down a tree for firewood?" when you really mean "I want to stay warm in winter.  What is the best way to do that?"
16:02.41jf_gentoo
16:02.56ManxPowerjf_, then read it again.
16:02.58*** join/#asterisk Blackthorn (blackthorn@ws-10.smyth.net)
16:03.16jakepdevi'mm spending too much time trying to decipher that last comment
16:03.30jakepdevthe dialplan is easier
16:03.35BuckRogersgood morning
16:03.41*** join/#asterisk RChadwell (~rob@rrcs-24-227-48-86.se.biz.rr.com)
16:03.58RChadwellAm I right that you can only register one sip connection?
16:04.06jakepdevrc - no
16:04.14jakepdevyou can register as many as you please
16:04.27RChadwellhow about for incoming calls
16:04.45tzangerjf_: if you read it, read it again
16:05.14L|NUXis there any Presence server for *
16:05.15tzangerjf_: you're not listening to it
16:05.21tzangerjf_: if it's a 2.6 kernel, read README.udev
16:05.31jf_ok
16:05.34tzangerjf_: FIRST RULE of open source:  what little documentation is provided, READ IT ALL  :-)
16:05.49tzangerjf_: it's a lot easier to read it than the ocde itself, which is also available
16:05.55cypromistzanger: ever met someone following that rule ?
16:05.56jf_thank i will in the future
16:06.05tzangerwhich is a major plus over closed source, where the documentation's equally shitty, but you can't see the code when you need it :-)
16:06.10RChadwellscenario I am lookging for is 2 numbers each ringing to the s extension for a different context - seems that the incoming calls all go to the 2nd sip registered - is it because of the /s on the end?
16:06.14tzangercypromis: I can try to change the world, can't I?  :-)
16:06.29ManxPowerThe second rule is:  TEST TEST TEST, PROTOTYPE PROTOTYPE PROTOTYPE
16:06.36tzangerRChadwell: don't put both SIP clients in the same context=
16:06.38tzangerin sip.ocnf
16:06.51ManxPowerAnd the 3rd rule: Users are lieing weasels and should never be believed.
16:06.52RChadwellthey aren't - that is the weird thing
16:07.01tzangerManxPower: that one is very true
16:07.08tzangerRChadwell: what is the context line for each
16:07.11RChadwellI think it is 1) Collect Underpants 2) ??? 3) PROFIT!
16:07.24*** join/#asterisk outtolunc (~me@adsl-66-218-53-170.dslextreme.com)
16:07.33tzangerRChadwell: no that business model's proven to be ineffective due to the sheer volume of underpants gnomes
16:07.35RChadwellcontext=cmgincoming for first ---- context=chadwellhomesincoming for second
16:07.43RChadwellHA
16:07.45ManxPowerFor the install I'm doing: User said: We need 4 fax machines and 1 modem.  What we discovered: They need 15 analog ports for analog devices
16:07.49BlackthornHow do you check the version of * that I am running. I did a show version but it just says head and then gives a date. I noticed on the * download it's at 1.07
16:08.03tzangerRChadwell: ok, now "show dialplan cmgincoming and show dialplan chadwellhomesincoming
16:08.06tzangerand make sure they're not the same
16:08.20tzangerin extensions.conf you may have included a context with an s exten
16:08.23tzangerand it's getting called
16:08.30*** join/#asterisk jdg (~jdg@CA03F897.adsl.mana.pf)
16:08.31jlewisthis is weird...X100P in the same system does give me an alarm / no alarm when the line is missing / plugged in...but the TDM400Ps are always happy
16:08.43tzangerjlewis: yeah the TDM400P doesn't seem to have that bit of code
16:08.56tzangerjlewis: if you're eager, it shouldn't be too terribly hard to add to wctdm
16:08.58RChadwellgotcha - I will check that - I read in the asterisk FAQ that it can't register more than one sip - I guess it was old
16:09.01jlewisthat's surprising since they're "higher end" cards
16:09.04*** join/#asterisk egon_ (~egon@pc-10-19-104-200.cm.vtr.net)
16:09.07L|NUXis there any Presence server for *
16:09.12tzangerL|NUX: no
16:09.22tzangerthere's a jabber server, you could probably integrate with that
16:09.31jlewismy C / driver writing foo would not likely cut it
16:09.41tzangerjlewis: there's no better way to learn
16:09.45vaewynthat @#$@#$@#$ putz... had me plugged into the wrong port on the Norhell....  plug it in the correct port and VOILA! I have a norhell <-> asterisk connection....  zero config
16:09.55tzangerwho doesn't want strong fo
16:09.55tzangerer foo
16:10.06vaewynI like a strong bar
16:10.06tzangervaewyn: :-)
16:10.07vaewyn:P
16:10.24jlewisis this really the way its supposed to be?...wondering if a post to the list is worth it
16:10.49tzangerjlewis: it just needs to be done is all
16:11.14ManxPowerjlewis, Post a bounty to the mailing lists.
16:11.23ManxPowerI'll bet you could get it added for less than $100
16:11.23*** join/#asterisk ikey1 (ikey@220.226.42.220)
16:11.29tzangeryeah the lists are hot for bounties right now
16:11.33tzangerwe just like btiching abotu the implementations
16:12.07*** join/#asterisk gnmraju (gnmraju@220.226.41.204)
16:12.17jf_tzanger: do i run modprobe aftter adding the lines
16:12.26L|NUXtzanger : hmm
16:12.38tzangeryeah remove and reinstall the wctdm and zaptel drivers
16:12.39tzangerit should just work then
16:13.18jf_remove it physically or the bin
16:14.05drumkillaManxPower: why do you want an error on exten => h,1,hangup ?
16:14.22drumkillawoah ... or I could just read all of the replies
16:14.25tzanger:-)
16:14.32tzangerjf_: no no just rmmod wctdm zaptel
16:14.36malbechI search a softswitch for a good price but it's very diificult to find one ... no ???
16:14.39tzanger(or whatever wcxxx card you're using)
16:14.46*** join/#asterisk cpatry (~grepmoo@65.39.228.5)
16:15.17jf_ok and then modprobe
16:15.52ManxPowerdrumkilla, because people call hangup from within _. and don't realize what's happening.
16:15.57ManxPowersee the discussions on -dev
16:16.31ManxPowerdrumkilla, For my next trick watch me start a riot on the mailing list! *grin*
16:16.47drumkillahaha, yeah, I'm reading now
16:16.56drumkillasorry, didn't think about stupid people ;)
16:17.09ManxPowerdrumkilla, you know that I help a lot of stupid people.
16:18.02drumkillayeah, and it's really cool that you have the patience to do so
16:18.13jf_got that now  WARNING[8898]: chan_zap.c:771 zt_open: Unable to specify channel 1: No such device or address
16:18.39JunK-Ujf_: cause ur /dev/zap/1 isnt there.
16:18.48ManxPowerjf_, maybe the module on your TDM400P is really on port 4 and not port 1
16:19.04jf_i only have a wcfxo
16:19.32tzangerjf_: them rmmod wcfxo zaptel and modprobe wcfxo
16:19.57ManxPowerdrumkilla, someday I'll go to geek heaven for this.
16:20.08ManxPowerbut I'd rather go to supermodel heaven
16:20.41bannermanManxPower: If a geek dies and doesn't see supermodels, he went to the other palce.
16:20.59gmcinnesManxPower:  You help me a lot, and I'm pretty thick :)
16:21.02jalsotis it possible to make a call transfer when the number is dialed through AGI?
16:21.08gmcinnesManxPower: Thank you.
16:21.20ManxPowergmcinnes, you can thank me by finding me a job in europe.
16:21.20gmcinnestzanger: You too!
16:21.39tzangertime for lunch
16:21.42ManxPowergmcinnes, you can thank tzanger by finding me a job in europe. 8-)
16:21.43tzangerno proble gmcinnes
16:21.45tzangerhahaha
16:21.48tzangeryeah pay it forward to manx
16:21.54gmcinnesManxPower:  europe == supermodel heaven? :)
16:22.01ManxPowergmcinnes, nope.
16:22.14eKo1wwwwhat?
16:24.34Marlowdrumkilla: wouldn't it be better to use that ticket for the flight, rather than letting the ticket fly on it's own ?
16:24.49Marlow:o)
16:24.59drumkillaheh, I'll still be using it
16:25.17*** join/#asterisk egon_b (~egon@pc-10-19-104-200.cm.vtr.net)
16:25.22drumkillavon europe
16:25.27vaewynok... that DS3 card is calling my name ;P
16:25.34ManxPowerdrumkilla, You going to Von Europe?
16:25.41ManxPowerI'll be good to meet you
16:25.46RoyKI'll try to go ther
16:25.47RoyKe
16:25.54*** join/#asterisk Gh0sty (~Ghosty@ip-81-11-201-68.dsl.scarlet.be)
16:25.54vaewyndrumkilla is a cool guy IRL
16:26.02RoyKto see stockholm during the  weekend and then the von
16:26.04ManxPowervaewyn, I'm an asshole IRL. 8-)
16:26.05vaewynhe can't say the same for me :P
16:26.06file[laptop]drumkilla rocks
16:26.21vaewynManxPower: that's different how?!?  ;P
16:26.23Marlowroyk: there isn't much you'll see in stockholm ..
16:26.26vaewynManxPower: had to... :P
16:26.30ManxPowervaewyn, LOL!
16:26.35ManxPowervaewyn, Someone had to.
16:26.42Marlowroyk: usually it's just getting drunk, and then you won't see anything after that .
16:26.44RoyKMarlow: what? I've been there several times - I like the city :)
16:26.49RoyKhehehe
16:27.02RoyKthat's part of the seeing the city :)
16:27.06Marlowroyk: i've been living there for over 1 1/2 years :)
16:27.13RoyKokk
16:27.16ManxPowerWhoo!  Whoo!  The TA 750 is "out for delivery"
16:27.21RoyKI just live in oslo .......
16:27.28RoyKta750?
16:27.31Marlowroyk: me galway
16:27.41ManxPowerRoyK, Adtran channel bank
16:27.46RoyKk
16:27.52drumkillaManxPower: yep, I'll be there :)
16:27.53RoyKmarlow where is galway?
16:28.03Marlowroyk: irish westcoast :)
16:28.19drumkillaManxPower: should I be scared to meet you? :p
16:28.20RoyKahki. north of limerick somewhere?
16:28.37Marlowroyk: yeah .. about 1 1/2 hours drive north of Limerick
16:28.44BuckRogersgood morning
16:28.49RoyK"stab city"
16:28.50RoyK:P
16:28.59RoyKmarlow talar du svenska?
16:29.19Marlowroyk: jeg er dansk, men jeg pratar också svenska, om det skal vära
16:29.27RoyKok
16:29.33vaewyndrumkilla: meeting anyone on here IRL is a crap shoot...  emphasis on crap   ;P
16:29.42Marlowroyk: det var jeg godt klar over :)
16:29.46RoyKahki
16:30.17RoyKMarlow: du får komme inn på asterisk-no ..
16:30.18RoyK<PROTECTED>
16:30.26*** join/#asterisk jtodd (~jtodd@dsl027-191-178.sfo1.dsl.speakeasy.net)
16:30.30vaewynactually I am surprised how well everyone on here gets along IRL...  usually better than on here :P
16:30.31Marlowroyk: bedre ?
16:30.36RoyKæøå
16:30.44Marlowroyk: æøå
16:30.47RoyKMarlow: ah
16:30.48RoyK:)
16:30.57Marlowroyk: default er UTF-8 i gaim
16:31.03Marlowroyk: havde ikke fikset det
16:31.14RoyKoki
16:33.08RoyKMarlow: er du fra København?
16:33.17Marlowroyk: nej .. sønderjylland
16:33.23RoyKok
16:33.48Marlowroyk: men jeg har boet lidt af hvert sted både indenfor og udenfor DK
16:34.04RoyKskjønte det.
16:34.06Marlowroyk: er lige flyttet fra Dublin til Galway
16:34.16RoyKhvor stort er galway?
16:34.18file[laptop]vaewyn: indeed, we're just a great big bunch of geeks IRL
16:34.29Marlowroyk: 66k i byen, 150k i countien
16:34.39RoyKfile[laptop]: correct
16:34.42*** join/#asterisk [TK]D-Fender (~joe@4.67.252.216.dsl1.colba.net)
16:34.48ManxPowerdrumkilla, Prolly no need to be scared.
16:34.54ManxPowerI'm MUCH more scary online than in person
16:34.55RoyKMarlow: ok, som en middels stor by i .no, med andre ord :)
16:35.02RoyKs/middles/ganske/
16:35.07file[laptop]in person he's a pussy cat I bet :p
16:35.12Marlowroyk: og områdemæssigt en af de største counties i irland
16:35.20RoyKok
16:35.20Marlowroyk: + den mest venlige :)
16:35.24RoyKfint :)
16:35.30RoyKikke som limerick?
16:35.35RoyKaka stab city?
16:35.37Marlowroyk: venlige = social
16:35.46Marlowroyk: limerick = dell .. that's it ..
16:35.49RoyKvet
16:35.52Marlowroyk: der er ikke andet i Limerick
16:36.07Marlowroyk: Galway er en by, hvor man lever
16:36.14bannermanHow do the pro's do QoS? By owning the router on both sides?
16:36.15RoyKjeg har vært der én gang, og det var på fabrikken til dell for å teste ut en SAN-løsning som sugde noe helt jævlig
16:36.33ScythelXnorsk?
16:36.33Marlowroyk: eh .. som kunde ?
16:36.37RoyKja
16:36.41RoyKjobba i fast.no på den tida
16:36.50MarlowscytheIX: royk = norsk, marlow=dansk
16:37.15Marlowroyk: jeg har arbejdet for Dell indtil sidste uge
16:37.18RoyKetter å ha påpekt det for dell at løsninga deres ikke holdt mål fikk jeg masse pepper fra sjefen om at jeg var "uprofesjonell" mot våre leverandører......
16:37.21ScythelXhehe i understand very little i have a lot of friends from denmark and norway
16:37.23Marlowroyk: gad ikke mere
16:37.29vaewynvaewyn = stupidamerican
16:37.31vaewyn:}
16:37.34RoyKdet her var før dell begynte å brande SMC
16:37.42RoyKeh. ikke smc, men .. ?
16:37.51MarlowEMC
16:37.52RoyKScythelX: where from?
16:37.57RoyKMarlow: yeah
16:38.10ScythelXI live in the US now, but I used to live in Nice, France
16:38.20Marlowroyk: 650F er ok, 660F = wouldn't touch it ..
16:38.22jeroand me in marseilles :)
16:38.35RoyKde hadde en hjemmesnekra løsning før det uten LUN masking eller noen ting. alt som fantes av LUN masking måtte gjøres i software
16:38.37BlackthornFollowing the * web site instructions on how to login to cvs and update your * source files, as it was scrolling across the screendoing the update i saw "xxx file no longer in repository" does that it mean that it deletes it from the current source directory? Since i guess it's no longer needed?
16:38.54vaewynBlackthorn: yep
16:38.59RoyK...og når vi da skulle ha opp freebsd og linux uten annet enn vanlig fibrechannel-drivere, gikk det til helvete...
16:39.00Blackthornkewl
16:39.03Marlowroyk: jup ... men der er ikke mange af dem mere ...
16:39.08RoyKdet er godt
16:39.11RoyKveldig godt
16:39.31jeroobenstrü
16:39.33RChadwellI think the problem is in my sip.conf file - mind if I put a portion here?
16:39.34Marlowroyk: de bliver ikke solgt mere ... i flere år
16:39.47eKo1RChadwell: use pastebin
16:39.51Blackthornnow that i've updated my * source I guess i just enter each directory and do the make clean;make install correct?
16:39.57*** join/#asterisk didz_ (didz_@200.218.192.52)
16:40.03file[laptop]I'm afraid
16:40.07file[laptop]my step-dad knows what bluetooth is
16:40.21denonfile[laptop]: he just got back from the dentist?
16:40.22eKo1Blackthorn: eh, 'make update && make clean && make install'
16:40.28RChadwellNever heard of pastebin - any instrux online?
16:40.35eKo1~pastebin
16:40.36jbotpastebin is probably a place to paste your stuff without flooding the channel - try http://pastebin.ca
16:40.36bugbotpastebin is assigned nothing and reported nothing.
16:40.36file[laptop]he was unpacking a plantronics wireless headset
16:40.36[TK]D-Fenderwww.pastebin.ca
16:40.45file[laptop]denon: but anyway, what did you want me for?
16:40.52RoyKmarlow jeg vet, de tok til vettet :)
16:41.01denonnothin .. had a Q last night, figured it out
16:41.06file[laptop]now now otl
16:41.46ManxPowerjbot, in my experience rope works better.
16:42.06didz_anyone knows how to pause a queue member from console?
16:42.09RChadwellhttp://pastebin.ca/9545
16:42.10RoyKyou'll need all those hooks in the floor
16:42.15RChadwellbtw - that is very cool
16:42.25ManxPowerRoyK, You don't already have hooks in the floor???
16:42.35RoyKManxPower: no, sorry
16:42.36vaewynkinky
16:42.38RoyK:)
16:42.50denonManxPower: bedposts
16:43.07vaewynack... jbot goes metric on us  ;P
16:43.12RoyK:)
16:43.34jf_is there any to know the ring length to set distinctive ring
16:43.38vaewynjbot: 1 mile per hour in fulrongs per fortnight
16:43.45vaewynack... bad spelling
16:43.52RoyKfulrong..
16:43.53vaewynjbot: 1 mile per hour in furlongs per fortnight
16:44.13RoyKwtf is a furlong? I've heard the word, but....
16:44.24vaewyndistance measurement
16:44.28RoyKyeah
16:44.31RoyKi knopw
16:44.32RoyKknow
16:44.35RoyKbut how long?
16:44.43robl^its longer than a cubit
16:44.48RoyK.no and .se has something called Mil, being 10km :P
16:44.49RChadwellThe problem is that every incoming call uses the [sip2.broadvoice.com] registration
16:45.09vaewyn1 furlong = 660 feet
16:45.20RoyKI should have thought of that...
16:45.22RChadwellany ideas?
16:45.26RoyKpretty obvious
16:45.36Gand_DJbjohnson, I just noticed on voipforcanada website they added "We are currently in Beta stages of development". lol
16:45.59Gand_DJExplains why audio cuts out after 5 seconds.. which sucks. emailed them on it
16:47.29RoyKhm. so one furlong is ten chains, one mile eight furlongs..
16:47.34didz_anyone knows how to pause a queue member from console?
16:47.38vaewynIs there a way to get callerid name passed over the PRI to my Norhell?  I seem to remember that isn't an option but...
16:47.38RoyKstupid english measurement
16:48.28Marlowroyk: 1 scandinavian mile is 10km .. the only ones that aren't using that anymore are us danes :)
16:48.28jero:)
16:48.39vaewynI'm surpised the metric nazis havn't come up with a way to make time divisible by 10
16:48.45ManxPowerGand_DJ, When you researched voipforcanada by asking on the mailinglists nobody mentioned problems?
16:48.47RoyKMarlow: poor buggers
16:48.58Marlowroyk: take ireland .. that's the complete mess ..
16:49.11ManxPowervaewyn, It's called Internet Time.  I think Swatch has info on their web site.
16:49.12RoyKMarlow: it used to be slightly more than 10km, being the distance you could walk before having to rest
16:49.16Marlowroyk: when i came over here, speeds where in mph, but distances in km
16:49.30vaewynManxPower: yeah... but even scientists think iut is crap :P
16:49.33BuckRogersin canada voip service providers have 90 days to comply with 911 calling regulations or shut down
16:49.33RoyKhttp://en.wikipedia.org/wiki/Mil
16:49.34*** join/#asterisk boch (~as24@200.59.172.98)
16:49.36Gand_DJManxPower, I haven't done any google searchs for mailing lists that would have voipforcanada in it
16:50.05ManxPowerhttp://www.timeanddate.com/time/internettime.html
16:50.06RoyKspeed measurement should perhaps be chains per teabreak
16:50.07RoyK:P
16:50.10Marlowroyk: now it's "go metric", meaning speeds were also changed from mph to km/h, but you have to be careful, if there is a "km/h" in the bottom of the speed-limitation or not ..
16:50.19jf_is there any to know the ring length to set distinctive ring
16:50.23Marlowroyk: you can from time to time still get some, that are in mph
16:50.55RoyKwell wtf. just treat all signs as km/h and hope they're mph and no cops around :P
16:50.55Marlowroyk: same as that you still find signs on the country-side, that show the distance in miles, instead of kms
16:51.02ManxPowerjf_, where would you set distinctive ring?
16:51.21|Vulture|is there a way to set a variable to the reciever of a current call?
16:51.28RoyKcan someone please tell me wtf "distinctive ring" really is?
16:51.28jf_on my line: there is 2 number, with to different ringtone, i want * to handle only 1 number
16:51.29Marlowroyk: cops aren't really a problem around here, and irish people can't drive .. it's worse than stockholm or paris
16:51.53Marlowroyk: you can have different ringtones for different numbers on the same pstn  line
16:51.53|Vulture|say x101 picks up a call that rings into 101/102/103, and you want to set userfield to 101
16:52.08ManxPowerlook in zapata.conf.sample
16:52.12ManxPowerI'm outta here
16:52.23vaewynRoyK: is a good way to make a phone sound sick ;P
16:52.29RoyKhehe
16:52.37RoyKis it that fast ringing tone?
16:52.48RoyKas in 500ms on 500ms off or something?
16:52.52vaewynthere are 4 patterns
16:53.00RoyKok
16:55.11*** join/#asterisk Ridgeback (~Ridgeback@ppp130-78.lns1.adl2.internode.on.net)
16:55.20Ridgebackgood morning!
16:55.21jf_manxpower: i knew that i cant just find which length, can it be 900,0,0
16:55.25RChadwellhttp://pastebin.ca/9545 - Would someone be interested in taking a quick peek at the sip.conf file to tell me why all calls are going to the second context?
16:55.59*** join/#asterisk florz (nobody@2001:1a50:503c:0:0:0:0:1)
16:56.12RChadwellCan you register a sip connection within a context or does it have to be in the general context?
16:56.32Gand_DJDumb question.. if you can use iax & sip for for asterisk to a voip provider... would iax be the better choice in general?
16:56.48vaewynYes
16:56.51RChadwellless overhead with iax2
16:56.55vaewynyou can trunk then
16:57.03Gand_DJk
16:57.04RidgebackRChadwell, does your second register command need to be sip2.broadvoice.com???
16:57.13Gand_DJSigning up at link2voip.com
16:57.22RChadwellnot really - just wanted a way to differentiate between the 2
16:57.26Ridgebackwoohoo got Dundi working between 4 switches!!!!!!!
16:57.36Gand_DJne1 tried them? They seem to have ok rates, and are from canada, but don't mention if fees are cdn or urd
16:57.37Gand_DJusd
16:57.51RChadwelloh - sorry, misunderstood - broadvoice requires that register command
16:58.09Ridgebackhey when is nuphone going to accept new customers again? I need a 1800 did :)
16:58.27RChadwellregister - then create a context for it that drives to an extensions.conf context - right?
16:58.43Wazbwhen patteren will asterisk receive after establishment of call from Regular phone in order to select other option
16:59.59RChadwellI mean, it shouldn't be hard to take two phone numbers (both sip) and route them to 2 different extensions.conf contexts...
17:00.14RidgebackRChadwell, yes the register statement will have a matching block with a context labeled. in this context one would then put thier exten statements
17:02.45gnmrajuis there any sppech processing module for Asterisk available anywhere
17:03.10Ridgebackgnmraju, festival can talk, sphinx can listen
17:04.11robl^..and Alison can "mooo"
17:04.22*** part/#asterisk Marlow (~martin@cerberus.bluetree.ie)
17:04.34Ridgebackyeah she can :)
17:04.47denonhas anyone implemented  sphinx  with asterisk though?
17:04.54Gand_DJappears link4voip is USD... says so after you logged into the site
17:05.33*** join/#asterisk myrkraverk (~user@myrkraverk.user)
17:05.44myrkraverkhello
17:06.02gnmrajui heard about Sphinx, but Ridge do u know any guy who can integrate everything for us on digium quad wild card
17:06.22myrkraverkdoes it make sense to use asterisk for voip backend in a real-time collaboration tool?
17:06.37RoyKgm
17:06.39Ridgebackgnmraju, not native to asterisk, it would all be a custom setup you desing
17:06.39RoyKhmmmm
17:06.42myrkraverkor am I researching the wrong tool?
17:06.48Ridgeback*design
17:06.58RoyKrunning 1.0.6 it seems asterisk leaks quite a lot of memory
17:07.01Ridgebackmyrkraverk, it would woork well
17:07.11RidgebackRoyK, uses the latest CVS, much better
17:07.27myrkraverkRidgeback: k, then I'll continue researching ;)
17:07.31RoyKRidgeback: what do you mean? cvs head????
17:07.49RidgebackRoyK, yes the latest works great with my shiny new Polycom IP600  :)
17:08.02gnmrajuyes Ridge i know, but do u know any person providing paid consultency on this or any person who can integrate, like some guys who configure vc dial etc
17:08.23Ridgebackmyrkraverk, yes meetme conferences are great for collarboration. easy to setup too
17:08.50RoyKRidgeback: you don't want to use cvs head for production!
17:08.55Ridgebackgnmraju, consultants? hmmm dont know any one who does, but the http://www.voip-info.org has lists to Asterisk consultants
17:09.00*** part/#asterisk JunK-U (~grepmoo@65.39.228.5)
17:09.03myrkraverkRidgeback: cool stuff
17:09.27bkw_where the hell is manx
17:09.33RidgebackRoyK, i agree, but I've used the latest CVS head for a year now on 4 switches. they all work fine
17:10.02vaewynbkw_: left about 45 mins ago
17:10.13RidgebackRoyK, maybe valgrind memory leak detecto could help determine which module asterisk is leaking out of?
17:10.17RoyKRidgeback: also, this system uses sipfriends, the old stuff from 1.0
17:10.26robl^bkw_, his internet went down
17:10.31RidgebackRoyK, good grief!
17:10.42RoyKRidgeback: what else is there that is 'stable'?
17:10.48*** join/#asterisk jf_ (~jeanfranc@HSE-Montreal-ppp332742.sympatico.ca)
17:10.50Ridgebackrob, oh the humanity!
17:11.17RidgebackRoyK, hmmm  the latest stable version of asterisk is probably umm stable   ;)
17:11.31RoyKthis is the latest 'stable' asterisk
17:11.33RidgebackRoyK, sorry didnt mean to sound like ajerk  ;)
17:11.34RoyKor pretty close
17:11.38RoyKyou don't
17:11.43jf_any way to call on zap channel without have to press 9 before, for now it trunk the forst umber of the telephone number
17:11.44RoyKbut asterisk jerks
17:11.52RidgebackRoyK, lol
17:11.56*** part/#asterisk Nix (~Nix@dsl81-214-65337.adsl.ttnet.net.tr)
17:12.19vaewynjf_: remove the :1 from ${EXTEN:1}
17:12.20Wazbwhat patteren will asterisk receive if option is selected after establishment of call from Regular phone?
17:12.51jf_k
17:12.51RidgebackRoyK, one thing you could try is one of those new asterisk on CD. i jsut read an article and it said this one particular variant was the most stable he has ever seen
17:13.10bkw_robl^, haha bet he did exten => h,1,Hangup
17:13.20RoyKwhere are the test results? what tests have been run? where are the patches that makes the diff from 1.0.7?
17:13.55RidgebackRoyK, dont know. I dont use the stable versions...
17:14.15*** part/#asterisk myrkraverk (~user@myrkraverk.user)
17:16.01jf_any know the ring lenght of bell canada for long ring
17:16.02didz_anyone knows how to pause a queue member from console?
17:16.04robl^bkw_, of course!  he  needed to hang up after hanging-up!
17:16.30bkw_haha
17:17.26Ridgebackanyone here use dundi yet?
17:17.32robl^why does "sudo rm -fr /*" delete everything?   gotta love wild cards
17:20.25foobosrob, * is not required there even.. since -r implies recursive
17:20.48Ridgebackhey is there a way to use the SendText() application to send text to another users phone?
17:20.53*** join/#asterisk Holos (~asdf@207.164.188.10)
17:21.56Ridgebackall SendText() offers is SendText(text) no extension number to send text toward (other than the calling extension)
17:22.14robl^foobos, shh!!  I was making a point. you gone and messed it all up. :)
17:22.28HolosCan anyone give me some extra tips on reducing echo on SIP to PSTN with TDM400? I have tweaked my RXgain to 5,10, and 12 and TXgain to 0,-5,10 and I am not noticing much difference, but there is no echo on sip-sip or on far side sip to pstn.
17:23.28robl^Holos, sometimes echo comes from cheap phone / speaker phone on the pstn side.
17:23.48Holosrobl^: I get echo while listening to ringtone.
17:24.42HolosMy Sip client is x-lite and is using a headphone/mic combo
17:24.43robl^Holos, I haven't seen/heard that one before.
17:24.54*** join/#asterisk [Outcast] (~bill@c-24-218-94-11.hsd1.ma.comcast.net)
17:25.35HolosWhat are most peoples RX/TX gains set at for TM400 cards?
17:26.22RidgebackHolos, im not sure gain has much to do with echo. thats usually a latency issue with your ISP
17:27.08robl^right.  gain is more for sound level
17:27.59jakepdevManx - that SendDTMF turned out to be a bug in the Flash cmd
17:28.14Ridgebacktime for bed for me... later guys!
17:28.22zoacheers
17:28.35zoabbl
17:28.49*** join/#asterisk agent_sx (~agent_@12-220-171-226.client.insightBB.com)
17:28.56jakepdevif you attempt Flash on an IAX trunk, instead of just returning -1, it quits
17:31.08*** join/#asterisk zoa (~zoa@pirus.securax.be)
17:33.26*** join/#asterisk illek (~Mike@ip68-13-238-168.ok.ok.cox.net)
17:33.44*** join/#asterisk tekati (~captain@cpe-66-75-215-63.bak.res.rr.com)
17:34.25*** join/#asterisk Sedorox (brandon@Neptune.client.wlmsprt.pa.sed6.net)
17:35.33*** join/#asterisk Bentley (~Bentley@S01060080c8135e6a.cg.shawcable.net)
17:37.17Holosrobl^: In this case there is less then 1ms to my server, and then it goes over copper pstn.
17:39.08gnmrajudoes voicexml work with asterisk
17:39.25file[laptop]no, NEXT!!!
17:39.35cypromisbut you are welcome to add voicexml support
17:39.45Bentleyhi all, in queues.conf, setting "monitor-format=wav" will save the recordings using Monitor() defaults.  Is there a way to also pass options (like basename & flags)? (v1-0)
17:39.53*** part/#asterisk illek (~Mike@ip68-13-238-168.ok.ok.cox.net)
17:40.58*** join/#asterisk jets (~brian@guardian.pmt.org)
17:41.32slePPwhat joy :>
17:41.57Sedoroxhmmmmm
17:41.57jf_why * does not transfert when i press # key
17:41.58file[laptop]what'cha up to?
17:42.06tzangerjf_: show application dial()
17:42.29jf_tzanger: what you wanna see
17:42.41foobosjf_, you need to give certain flag to Dial() to enable #-transfer
17:42.48jf_oh ok
17:43.01tzangerjf_: it's not what I want to see, it's what you need to see.  :-)
17:43.39jf_so i just create a context or put dial() what in the context i want
17:44.19slePPwhat'm i up to?
17:44.21tzangerA 911 dispatcher was reprimanded for responding to a mother's plea for help with an unruly child by saying: "OK. Do you want us to come over to shoot her?"
17:44.27slePPmaking SER go zoom
17:44.36foobosjf_, no.. you put t or T flag to dial-string.. but i've forgotten which one is it
17:44.40tzangerjf_: you need to read the asterisk handbook and get a good handle on the basics
17:45.00file[laptop]slePP: Yay
17:45.07*** join/#asterisk juiceib269 (~juiceib26@24.236.130.31.bay.mi.chartermi.net)
17:45.28robl^slePP, zoom?  or BOOM!!  ?
17:45.34gnmrajudoes voicexml work with asterisk
17:46.11Sedorox[13:39] <gnmraju> does voicexml work with asterisk
17:46.12Sedorox[13:39] <file[laptop]> no, NEXT!!!
17:46.26Sedoroxso to repeat.. No.. it doesn't
17:46.54fugitivowill it work? :)
17:47.33robl^Sedorox [13:39] <gnmraju> does voicexml work with asterisk
17:47.34robl^Sedorox [13:39] <file[laptop]> no, NEXT!!!
17:47.40*** join/#asterisk rene- (~root@200.106.49.195)
17:47.59file[laptop]maybe this is better
17:48.03file[laptop]gnmraju: no, NEXT!!!
17:48.10Sedoroxlol
17:48.31Wazbonce call established with Asterisk , after greeting if user press 1 then what format of extension will asterisk get
17:48.51file[laptop]Wazb: what?
17:50.10rene-hey
17:50.57Wazbi need to know when person call to asterisk server through DID , after hearing greeting , if he press 1 then will comes to * ?
17:51.23file[laptop]Wazb: DTMF tones go to asterisk yes, there is an application called Background which will allow you to make an IVR (a digit based phone menu)...
17:51.34tzangerWazb look at the example dialplan, there's an entire IVR in there.
17:51.39slePProbl^: zooooooooooooom
17:51.44*** part/#asterisk nitram (nitram@superblob.com)
17:52.37*** join/#asterisk Bola_King (~john@62.175.14.244)
17:53.18Sedoroxhmmmm
17:53.39Holosrobl^: SOLVED: You need to define your channels after the echocancel=yes otherwise it just sets it to no. zap show channel 1 reported that there was no echo canceling enabled.
17:54.01Bola_KingI'm new to this forum, is it ok to post a question related to td400 ?
17:54.23rene-i have a tdm400, and im seeing the trunks blocked every now and then, zap destroy channel does not seem to free the trunk, only restarting asterisk solves the problem, what could be causing my trunks getting blocked?
17:55.28[Outcast]has anyone seen anthem?
17:55.34cypromishe is in away
17:56.01[Outcast]has there been a fix for res_perl for 1.0.7?
17:56.37RChadwellHas anyone got two phone numbers to work with Broadvoice? I am still working on it and all calls go to the second sip registration - which sucks.
17:57.21*** join/#asterisk BoRiS (boris@S01060040ca1e5b54.wp.shawcable.net)
17:57.57RChadwellIt is fun to search the web for comments on Broadvoice from like 2003 - I was hoping someone here has actually setup 2 sip connections
17:58.04RChadwell:)
17:59.01robl^RChadwell, I have 2 BV accounts.  works without a problem
17:59.16RChadwellwill you post your sip.conf to pastebin?
17:59.31RChadwellor just the relevant register and initial contexts?
17:59.42robl^RChadwell, just a sec
17:59.45rene-i also have another question, im running voice over a vtun vpn with IAX, sometimes voice just stops but the iax channel keeps itself up, the wan links are not heavily used but i understand that the data is travelling over the internet, and that data loss is likely, i was loooking at the drop count parameter, ihave set it up at 2 and well im seeing less of this issue, what is a good value for this param?
17:59.48RChadwellThanks - lifesaver
18:03.20Wazbtzanger , where i can find that exaples
18:03.27tzanger/etc/extensions.conf
18:03.31tzangerer /etc/asterisk/extensions.conf
18:04.25*** join/#asterisk ikey1 (ikey@220.226.29.74)
18:05.14Wazbtzanger , i am pointing my DID to * and it gives me greeting
18:06.23file[laptop]dejavu
18:06.58rene-dejavu you say
18:07.16rene-that only happens when they change something
18:08.00Wazbtzanger , but when i press 1 then it wont execute extern => 1,Dial(sip/1....)
18:08.05file[laptop]well they certainly didn't increase the average IQ of people in here
18:08.25Sedoroxlol
18:08.30rene-i can see that
18:08.38Sedoroxno.. but jbot turned into a agent...
18:08.42rene-nor did they improve response times
18:08.47file[laptop]indeed
18:08.51vaewynjbot always was an agent :P
18:08.54Sedoroxtrue...
18:09.30robl^RChadwell, http://pastebin.ca/9556  that's the relevent bits
18:09.38Wazbtzanger , but when i press 1 then it wont execute extern => 1,Dial(sip/1....)
18:10.20tzangerWazb: you need to read the handbook and work through some examples
18:10.25tzanger~handbook
18:10.26jboti guess handbook is http://www.digium.com/handbook-draft.pdf
18:10.26bugbothandbook is assigned nothing and reported nothing.
18:11.08*** join/#asterisk egon_l (~egon@pc-10-19-104-200.cm.vtr.net)
18:11.33*** join/#asterisk Aze` (~aze@host229-162.pool80105.interbusiness.it)
18:11.36vaewynwe don't need 2 $#@%#$% bots
18:12.15robl^me need 3 bots!
18:12.30robl^there are always plenty of bots! :)
18:13.58RChadwellrobl^, At the end of your register line you have a 10-digit extension. Does that mean that you route directly to that extension in extensions.conf?
18:14.34Wazbtzanger , this is what i have used in my extension file
18:14.35Wazb[test]
18:14.35Wazbexten => 416123456,1,Goto(test,s,1)
18:14.35Wazbexten => s,1,Answer
18:14.35Wazbexten => s,2,Ringing
18:14.35Wazbexten => s,3,Background(thnaks)
18:14.36Wazbexten => s,4,Wait(2)
18:14.38Wazbexten => 4455,1,Dial(SIP/4455,20,rt)
18:14.41tzangerWazb: do not flood
18:14.51Wazbsorry
18:14.53robl^RChadwell, all incoming SIP calls go to a single context for me.  incoming-sip.  then that context looks at the extension and directs it to a different context
18:15.30tzangerWazb: don't use Wait() it doesn't listen for digits
18:15.38tzangerBackground(silence/2) or Read()
18:15.54tzangerWazb: have you gone through the handbook?  have you gone through the default extensions.conf?
18:16.39Wazbyes i did
18:17.38tzangerWazb: there are specific examples in the handbook IIRC
18:18.33zoahey ho
18:21.16Wazbtzanger , in Asterisk Handbbok Version 2 , right?
18:21.32tzangercorrect
18:21.33tzanger~handbook
18:21.34jbothandbook is, like, http://www.digium.com/handbook-draft.pdf
18:21.45bugbothandbook is assigned nothing and reported nothing.
18:25.22Wazbthanks tzanger
18:29.18RChadwellThanks robl^, your solution made my life a lot simpler. That is awesome!
18:31.30tzangerWazb: chapter 4 specifically deals with the dialplan
18:31.39*** part/#asterisk Gand_DJ (~gandalf@ptr-207-54-104-24.ptr.terago.ca)
18:33.30*** join/#asterisk IPmonger (~ipmonger@pcp0010577106pcs.coatsv01.pa.comcast.net)
18:33.42*** join/#asterisk afrosheen (~afro@txprotoa22.august.net)
18:35.05*** join/#asterisk shmaltz (~chatzilla@69.28.255.210)
18:35.14afrosheenso what's going on these days
18:35.17*** join/#asterisk NewSole (~david@i216-58-44-245.avalonworks.net)
18:35.18afrosheenchannel = slow
18:35.27*** join/#asterisk NK123 (~p645@cpe-024-163-078-012.nc.res.rr.com)
18:36.07Wazbthanks again tzanger
18:36.15NewSolequestion... anyone here have a PRI with DID's
18:36.31*** join/#asterisk _Brian (brian@unix01.voicenet.com)
18:36.47afrosheenhas anyone done a robust meetme interface, like having people about to join a meetme record their name, then * announces their arrival in the meetme room?
18:37.46_Briandoes anyone know if * has any time of audio detection?  I have an application that needs to Flashhook a call to put them on hold and then dial another extension utilizing SendDTMF.  The problem I am having is that * will continue to the next step even before the remote party answers.  If I utilize a Dial string, then i use another channel......
18:37.54_Brianum..time should be type :)
18:38.29_Briana Wait would not be applicable either, since it could be several rings before the remote party answers
18:38.59Juggiehow do i fix, chan_iax2.c:5067 socket_read: meta trunk cmd 1 received, I only understand 0 (perhaps the remote side is sending trunk timestamps?)
18:39.42afrosheen_Brian: a wait would work, because you'll want a finite time before the dial gives up and finishes your dial string
18:40.53_Brianafrosheen: that is fine, but if the person answers the call on the first ring, then they will be stuck waiting for the next prompt to be played out
18:41.17_Brianafrosheen: if they answer it after the 5th ring, then it might play fine, but the early answerer's would be penalized
18:41.22afrosheen_Brian: yeah true :(
18:41.55_Briana Dial would work great for this, but I am forced to flash hook the extension and then send the DTMF digits...
18:42.11_Brianat that time, i am connected........so it is not typically a dial, since I am utilizing the same channel
18:42.30*** join/#asterisk DrJolo (~chatzilla@217.153.194.10)
18:43.02NewSoleWhen a call comes in off PRI group is there a wat to know what DID number they called
18:43.29bjohnsonfor anyone who cares .. just got asterisk running on a wrt54g (as reported by other users).  Haven't had time to test quality or capacity yet
18:43.34tzangerNewSole: ${EXTEN} or ${DNIS}
18:43.41_Brianbjohnson: cool...
18:43.46tzangeror if you want the number that redirected them to the DID, ${RDNIS}
18:43.53Sedoroxbjohnson: I thought you've always had one running? or was that someone else?
18:43.55tzangerbjohnson: nice
18:43.59tzangerSedorox: that was jerjer I think
18:44.04Sedoroxah ok
18:44.05Sedoroxmaybe
18:44.06bjohnsonbunch of others
18:44.06tzangerbjohnson: any major hurdles?
18:44.19bjohnsonI was trying to get it to run but kept banging my head against space limitations
18:44.29_Brianafrosheen: any other ideas?
18:44.29afrosheenbjohnson: cool
18:44.30Sedoroxlol
18:44.35bjohnsonno major hurdles if you do it right the first time :)
18:44.37afrosheen_Brian: naw, I'm not that smart yet
18:44.45_Brianafrosheen: rofl
18:44.48*** join/#asterisk Bonbon (~bonbon@83.146.53.34)
18:45.10Bonboni haven't tried flash operator panel, but does anyone know if you can transfer calls with it?
18:45.15bjohnsonI still have to set up a nfs share to use sound files and store voicemail messages
18:45.22afrosheenBonbon: yeah you can
18:45.26_Briananyone else?
18:45.42bjohnsonbut the working part of the system is stored on the flash of the wrt itself
18:47.15SedoroxBonbon: I tried it.. I actually didn't like it too much... but thats just me...
18:47.58The_Apehmm, i have a question..  when i type "sip show peers" in the CLI, in the name/username field everyone shows up as 1000/1000 and 1001/1001 and so on .. extept one that shows up as 1004/s. everyone has the same config. Any ideas?
18:50.20afrosheenThe_Ape: monkey with the database command, see if you can catch a bad entry in it for that extension
18:51.19_Briani guess that would be a no ....oh well..i will keep plugging away..
18:52.04HogieIm running Whitebox Linux 4... I just installed a TDM03 (3fxo), I have in rc.local to /sbin/modprobe wctdm then to /sbin/ztcfg -vvv, and when I do boot up, I get line 0: Unable to open master device '/dev/zap/ctl', but it works when I login as root.  What can I do to fix it?
18:52.06*** join/#asterisk riksta (~rick@81-178-248-131.dsl.pipex.com)
18:52.52*** join/#asterisk brainchil (~daver@bigbrother.kdsi.net)
18:53.00cypromisHogie: check README.udev in the zaptel source dir
18:53.39brainchilcould some one point me to a faq/info/article on setting up a button on a polycom phone to do unattended transfers like a tdm pbx?
18:53.40The_Apeafrosheen: sip/registry/1004 IP-address:5060:120:s:sip:s@IP-address. The s is there.. where is he doing it wrong? he's still getting registerd?
18:53.44brainchilor tell me if it's possible
18:54.09Hogiecypromis: i did that, this is before login that it fails
18:54.49brainchilI know they can do it with software but I want to be able to have a transfer -> dial extension from the phone itself
18:55.03brainchiland searching google and voip-info I came up short
18:55.33afrosheenThe_Ape: have a look at your sip.config and your extensions.conf, it's getting it from somewhere
18:55.43afrosheenThe_Ape: woops I meant sip.conf
18:58.26The_Apeafrosheen: no "s" in either file. :/
18:58.28brainchilanyone?
19:01.20johnnybhehehehe -- one of my users just plugged BOTH the PC and LAN connection of the phone into the same hub
19:01.31johnnybNow THAT will bring down a network.
19:04.08*** join/#asterisk syle (~blah@wnpgmb02dc1-176-155.dynamic.mts.net)
19:05.04The_Apeafrosheen: The s was the remote asterisk's local incomming extensionname for my box. Thanks for helping :)
19:05.14BlackthornFor those that might be interested I am running an 802.11b network around my town. Currently 3 towers, 43 users with 27 of them using sipura phones connected to * here in my office. So far been working great.
19:07.20vaewynBlackthorn: my home runs off a 6.1 mile 802.11b link  :P  SIP and IAX2 calls work great :P
19:08.51bkw_haha
19:08.59bkw_why do people really fail to understand what _. can do
19:09.20Hogiedoes digium still do free installation support?
19:09.26Hogieof their hardware?
19:09.32bkw_you need help?
19:09.59Hogieyes, 1 stupid problem with a tdm card
19:10.22funxionyeah me too
19:10.53funxionI finally got the modules for my te110p to load
19:11.35funxioncan someone help me with zapata.conf
19:11.50*** join/#asterisk hohum (corbe@snoop.burghcom.com)
19:12.12hohumanyone who knows the asterisk codebase well enough point me in the right direction?
19:12.18hohumI want to disable loop detection
19:12.26bkw_loop detection on what?
19:12.26hohumnot sure what file(s) deal with that
19:12.34hohumbkw: SIP channels
19:12.39bkw_chan_sip.c
19:12.47file[laptop]hahaha....
19:12.51bkw_and It shoudln't do the loop detected thing anymore
19:12.58hohumno?
19:13.02bkw_let me look at the code
19:13.03hohumwhy?
19:13.29Nuggetno!  we won't let you!
19:13.29bkw_well I recall this
19:13.40bkw_you shouldn't call yourself
19:13.45file[laptop]it'll do a Loop Detected if a literal loop occurs... ie: asterisk sends out an invite, and it goes back to itself
19:13.47bkw_you deal with local stuff in the local dialplan
19:13.47bkw_duh
19:13.47hohumI'm not calling myself
19:14.08bkw_you're calling another extension in your asterisk box
19:14.11hohumI'm calling a SIP UA that's registered to my SER box and has call forwarding enabled
19:14.12bkw_its still a loop
19:14.24bkw_show me CLI output
19:14.28hohumso it starts another phone call with the same call ID
19:14.41bkw_OH
19:14.43file[laptop]what?
19:14.47file[laptop]what odd behavior
19:14.47bkw_munge the callid in ser
19:15.07hohumaccording to my interpretation of the RFC loop detection is optional anyways
19:15.32bkw_then look in chan_sip.c and comment out that part
19:16.21hohumguess I'll find out tonight when I go to upgrade this asterisk box :)
19:16.40hohumI'm going to move it from 1.0.7 to CVS-HEAD
19:17.00Sedoroxwhich is newer, head or release?
19:17.04*** join/#asterisk shido6 (~greg@d57-87-253.home.cgocable.net)
19:17.05hohumhead
19:17.11Sedoroxwhats more stable? lol
19:17.13shido6head
19:17.15*** part/#asterisk didz_ (didz_@200.218.192.52)
19:17.15shido6and mo head
19:17.20Sedoroxhmmmm
19:17.31Sedoroxmaybe I'll try out head...
19:17.40_Brianafro: looks like you can use BackgroungDetect, but it appears that does not work correctly :(
19:17.48hohumrelease would be more stable
19:18.05hohumI only want head</pun>
19:18.24hohumcos some changes were made to the way SIP Codecs are handled
19:19.12MajestiKI'm having some troubles with my x100p card, it doesn't seem to drop the line when someone hangs up, and I'm getting "hang up your phone now" voicemails
19:19.26shido6hehee
19:19.27vaewynSo... how long until head becomes 1.1?  ;P
19:19.31bkw_hohum, those changes are in cvs-stable
19:19.32bkw_btw
19:19.39hohumoh?
19:19.40hohumneat
19:19.42bkw_yes
19:19.46bkw_backported because it was a major issue
19:19.49bkw_that needed to be fixed
19:19.51funxionI just set up a te110p t1 card and got to the point of configuring it in zapata.conf and zaptel.conf when I do zap show channels it shows me 23 channels for PRI but the card still has red light
19:20.01funxioncan anyone point me in the right direction
19:20.13bkw_funxion, show me your spanline
19:20.51tzangerbkw_: that's personal
19:21.11_Briandoes anyone know if * has any time of audio detection?  I have an application that needs to Flashhook a call to put them on hold and then dial another extension utilizing SendDTMF.  The problem I am having is that * will continue to the next step even before the remote party answers.  If I utilize a Dial string, then i use another channel......
19:21.58juiceib269funxion have you trieds the command ztcfg?
19:22.49juiceib269opps is it a red light or is it blinking red?
19:24.02vaewynHmm... why is my span saying 'internal timing' when it should be getting it fromthe norhell?
19:24.19vaewynIt is working ok... so I won't argue but...  seems odd
19:27.16*** join/#asterisk PoWeRKiLL (~PoWeRKiLL@d01m-1-164.d4.club-internet.fr)
19:27.59hohumbkw: thanks
19:28.56Bonbonafrosheen: so we can use a touchscreen tft monitor and use flash operator panel with no need to use a mouse
19:28.56Bonbon?
19:29.25rrkquestion about did's?
19:29.47vaewynFOP is cool... but wth flash and not just dhtml?
19:29.57vaewynbut anyways :P
19:30.50funxionshido6 you still there?
19:30.54*** join/#asterisk dmccollum (~dmccollum@69-164-245-72.atlaga.adelphia.net)
19:31.05Bonbonvaewyn: would my idea to use with a tft touchscreen mean that the operator doesn't have to touch a mouse?
19:31.14nestArhrmm. monitor-join doesn't seem to wrok.
19:31.27Bonbonvaewyn: in order to transfer a call
19:31.48_Brianfunxion: i think he disappeared
19:31.55funxionkewl
19:32.04fearnor<Bonbon> vaewyn: in order to transfer a call
19:32.05fearnorerr
19:32.53*** join/#asterisk P-Chan (~jpfingstm@68.142.66.200)
19:33.23P-ChanHello.  I'm using AMP and in my log I have ast_yyerror(): syntax error: syntax error; Input:
19:33.24P-Chan<PROTECTED>
19:33.30P-ChanI think it comes from:
19:33.39*** join/#asterisk n4y (~tmalkut@fire2.orasoft.net.pl)
19:33.41P-Chanextensions_additional.conf:exten => s,1,GotoIf($[${DIALSTATUS} = ANSWER]?4)     ;
19:33.51agent_sxdaper dan god damit
19:33.53agent_sxlol
19:34.08vaewynBonbon: correct
19:34.09agent_sxgreat moviee
19:34.21P-ChanAny idea what I should do about this?  I hate "ignoring" errors - I can't seem to see it causing a problem tho.
19:38.02*** join/#asterisk easimon (~easimon@baghira.kawo2.RWTH-Aachen.DE)
19:39.39sylecan asterisk tell difference between an answering machine and a real person?
19:39.56tzangersyle: no
19:42.14vaewyncan tell a fax or modem...  but not answering machine
19:42.32tzangerthere is an app_machine() but it's not part of the standard set and I have no idea how well it actually works
19:43.09SedoroxI think we should make a app_whops that runs whenever something isn't configured right.. and output something like "Check your config, stupid"
19:44.13tzangerSedorox: nah I am a huge fan of having everything default to the default context, and the default context be FORCED to have nothing but exten => _.,1,Wait(1), 2,Playback(please-configure-asterisk) 3,Goto(s,1)
19:44.17hohumanswering machine detection is very difficult to do
19:44.36hohumthe old school train of thought is to measure how much background noise there is on the call
19:44.51hohumbecause answering machines typically generate alot of white noise
19:44.53Sedoroxahaha
19:44.56hohumbut that doesn't work too well anymore
19:45.04*** join/#asterisk dca[laptop] (~dca[lapto@sta-206-168-218-206.rockynet.com)
19:45.05hohumCell phones do the same thing
19:45.08vaewyntzanger: but then more people know about _.  and that is evil   ;P
19:45.14tzangervaewyn: hmm this si true.
19:45.16tzangerfine then
19:45.27Sedoroxhmmm
19:45.37tzangers,1,all that and _X.,1,Goto(s,1)
19:45.37*** part/#asterisk IPmonger (~ipmonger@pcp0010577106pcs.coatsv01.pa.comcast.net)
19:46.54hohumand I come from a telemarketing background :) so
19:48.57rene-what type of hardware could possibly service the DS3 card from digium?
19:50.16vaewynquad opeteron :P
19:50.19vaewynopteron even
19:50.36vaewynjust enough to split the traffic up for other servers :P
19:50.51vaewyndon't even think of landing G.729 on it  ;P
19:51.12*** join/#asterisk eKo1 (~bernd@207.42.191.67)
19:51.44*** join/#asterisk WGFreewill (~chatzilla@24-75-221-174.miamfl.adelphia.net)
19:52.32hohumasterisk TDM :(
19:52.49P-ChanIs it possibly to forward a call from an asterisk server with a PRI to another via IAX2 trunking and have spandsp on the otherside of the trunk do fax detection?
19:53.22*** join/#asterisk mgth (~chatzilla@dsl093-001-038.det1.dsl.speakeasy.net)
19:54.53dca[laptop]anyone ever seen this error using res_config/realtime:
19:54.54dca[laptop]Apr 14 13:40:38 WARNING[8281]: res_config_odbc.c:277 realtime_multi_odbc: SQL Fetch error!
19:54.54dca[laptop][SELECT * FROM extensions WHERE exten LIKE ? AND context = ? AND priority = ? ORDER BY exten]
19:55.16dca[laptop]it's like something snaps and this error just spews
19:55.58*** join/#asterisk n4y (~tmalkut@fire2.orasoft.net.pl)
19:56.10slePPdoes the extensions table exist and have fields 'exten', 'context' and 'priority'?
19:56.14slePPdoes the user accessing it have permission?
19:56.28dca[laptop]yes, and i think so
19:56.39slePPtest the same query from isql
19:56.40outtoluncwhat version of mysql?
19:56.45WGFreewilliax jitterbuffer broken in CVS?
19:56.53dca[laptop]it works like a charm for a ton of calls, then, for not apparant reason, whamo!
19:57.03slePPwhich db backend is it using?
19:57.13*** join/#asterisk darby_t (mua@dno210.neoplus.adsl.tpnet.pl)
19:57.27dca[laptop]slePP: res_odbc with mysql 5
19:57.38slePPis mysql allowing enough connections from the host?
19:57.48dca[laptop]how can check?
19:57.53slePPdunno :>
19:57.56dca[laptop]hehe
19:58.13file[laptop]muahahahaha
19:58.33dca[laptop]slePP: and, lets say it isn't, would would that cause a bottle neck? or to break, like it currently is?
19:58.39slePPfor a bill for 130.75 hours. they're arguing 124.25 hours of it
19:58.57slePPdca[laptop]: i could see if it tried to open another connection, and it was refused, it'd blow up
19:59.06dca[laptop]hmm
19:59.19dca[laptop]time to /join #mysql
19:59.19Wazbtzanger , i looked at handbook and exaples but i am confuse in on thing.
19:59.29tzangerslePP: that is why I almost universally prefer to quote for a specific block of work and get a P.O.
20:00.27rene-vaewyn: nice, so any codec translation or agi should be done on other machines, right? would you use TDMoE to forward the calls to the other machines?
20:00.28file[laptop]slePP: How are they arguing?
20:00.51Wazbtzanger , in SIp.config at top i changed context = demo , i got nothing when i dial my DID no.
20:01.18tzangerWazb: because that is not how it works with PRI
20:01.30*** join/#asterisk FuriousGeorge (~brian@ool-43516ebb.dyn.optonline.net)
20:01.37vaewynrene-: TDMoE?  just IAX2 it off :P
20:02.30vaewynrene-: and yes... do EVERYTHING you can elsewhere
20:02.30FuriousGeorgehi all
20:02.30*** join/#asterisk jcollie (~jcollie@lt16586.campus.dmacc.edu)
20:02.32Wazbtzanger , so i which config file i need to configure
20:02.36rene-that is simpler and nicer
20:02.40jcollie~seen bkw_
20:02.52jbotbkw_ is currently on #asterisk (4h 25m 6s).  Has said a total of 26 messages.  Is idling for 42m 39s
20:02.52bugbotseen bkw_ is assigned nothing and reported nothing.
20:03.01tzangerWazb: you have a PRI.  Quick quiz.  What interface do PRIs come in on?
20:03.26FuriousGeorgewhats preferable for a small * install.  using a residential gateway "router" device, or using the asterisk box for the NAT, dhcp server and firewall?
20:03.56WGFreewillAnybody know how to troubleshoot DTMF across H323
20:04.18FuriousGeorgei ask because i hear QoS to the outside world is mostly a myth, but i wonder about incomming QoS for voice streams
20:04.20Wazbtzanger , actually i have Cisco in which PRI is terminating and from there through i am forwarding call to * usign SIP
20:05.34*** join/#asterisk stevej (~stevej@67.97.36.243)
20:06.05rene-what do you people think is the  recommended sub 200 business phone?
20:06.16jcollierene-: SPA-841
20:06.26nvrsworkif I have a X100P clone do I need to have ztdummy in the kernel?
20:06.36hohumclone?
20:06.38nvrsworkyes
20:06.42tzangerWazb: oh so you're really a glutton for punishment
20:06.43hohumpeople are cloning digium cards now?
20:06.54jcollienvrswork: no, you shouldn't
20:06.59nvrsworkthanks
20:07.15hohumwhat kind of horse shit is that?
20:08.53brainchilhow do you do an unattended transfer from a sip phone on asterisk?
20:08.57Wazbtzanger , anything wrong?
20:09.05TomL~seen manxpower
20:09.11jbotmanxpower <~eric@adsl-35-236-60.msy.bellsouth.net> was last seen on IRC in channel #asterisk, 3h 16m 59s ago, saying: 'I'm outta here'.
20:09.11bugbotseen manxpower is assigned nothing and reported nothing.
20:09.11brainchilor rather is it possible
20:09.15dca[laptop]any mysql guru's here? need to know if there is a bottleneck in mysql my.cnf that is causing asterisk to spew errors...
20:09.18hohumI wish someone would make a card that does hardware TDM and works with PRIs in Asterisk
20:09.26tzangerWazb: my first guess is that * is not seeing DTMF from the Cisco
20:09.34FuriousGeorgelet me ask this way:  does anyone use their * server as their NAT and firewawll and pppoe gw etc
20:09.43FuriousGeorgeis that a bad idea?
20:09.44tzangerdon't use inband unless you can't help it, and if you can't help it, you must only use ulaw between the cisco and asterisk
20:09.50rene-brainchil: sometimes the phone implements those things for you
20:10.03brainchilrene-: How and what phones?
20:10.20rene-siemens optipoint 400s can do it
20:10.29rene-but you could also do it in asterisk
20:10.29P-ChanIs it possibly to forward a call from an asterisk server with a PRI to another via IAX2 trunking and have spandsp on the otherside of the trunk do fax detection?
20:10.35rene-with a parking extension
20:10.46*** join/#asterisk ManxPower (~eric@stirprop-S0-0-0-26.ndcr2.datasync.net)
20:10.59tzangerP-Chan: yes but you must be very careful...  spandsp on the far side (i.e. an IAX2 hop away from the PRI) will be spotty
20:11.00rene-but you should only do # EXTEN
20:11.08rene-and it should be taken care of
20:11.11ManxPowerOK eveyone I'm calling in some of my Good Asterisk Karma
20:11.29tzangerManxPower: after that exten -> h bounty you have no karma
20:11.35tzangeror rather it's dangerously low
20:11.37*** join/#asterisk ritesh (~ritesh@natint3.juniper.net)
20:11.45hohumhey
20:11.49brainchilrene-:I'll look at that .. is there a faq/howto someplace that you could point me ... I have been unable to locate one
20:11.51hohuma juniper employee :)
20:11.56hohumcan I get a free M20?
20:12.14ManxPowerI'm having a problem with Call Parking.  Call comes into an extension which runs a macro.  The user transfers to the call parking extension.  If nobody picks up the call goes to extension "s", not the correct extension that parked it.
20:12.19ManxPowertzanger, Ha!
20:12.25P-Chantzanger: so it would be better to have the asterisk server @ the PRI receive the faxes (perhaps on a dedicated DID?
20:12.33tzangerManxPower: without the macro does it work?
20:12.34*** part/#asterisk ritesh (~ritesh@natint3.juniper.net)
20:12.37*** part/#asterisk jcollie (~jcollie@lt16586.campus.dmacc.edu)
20:12.41tzangerP-Chan: very much so, yes
20:12.43tzangerbut be careful
20:12.48hohumI guess that was a big fat "no"
20:12.49hohum:(
20:12.51tzangerI have had spandsp crash out asterisk the odd time
20:13.31P-Chantzanger:  Oh...so it might be a problem where the asterisk server is mission critical.  I may not want to do it like that... hmm...
20:13.42rene-how does one go about blocked analog zap channels
20:13.56rene-i was having issues with spandsp and rxfax
20:14.09ManxPowertzanger, you DID have to ask a useful question, huh?  I assume it does work, but I'll check
20:14.15rene-i got rid of all of it (rxfax)
20:14.26rene-but some calls still block my outbound zap trunks
20:14.32tzangerManxPower: :-)
20:14.38tzangerP-Chan: correct.
20:15.12P-Chantzanger:  Thanks.
20:15.36ManxPowertzanger, checking now
20:15.37rene-whats the deal with zap destroy channel. i t doesnt really do anything for me
20:15.54gnmrajuexit
20:18.10n4yDoes polycom ip 300 phone work with Asterisk?
20:18.25ManxPowertzanger, Nope!  Doesn't work!
20:18.30ManxPowerCVS 1.0.x
20:18.40Wazbtzanger , i addedd dtmfmode = inbad in SIP.conf file and it works
20:19.14FengShuiis there any easy way to signal a flash to a zap channel that's briged to a cisco 7960?
20:19.16tzangerrene-: what do you expect it to do, shoot smoke out the TDM card?
20:19.22FengShuiI want to flash the zap channel from the 7960l.
20:19.24rene-heh
20:19.36*** join/#asterisk bah (048830696@ACA1C854.ipt.aol.com)
20:19.41tzangerWazb: make sure you do not use anything other than ulaw in that case
20:20.11rene-tzanger: well unblocking the trunks would be nice
20:20.12*** join/#asterisk riksta (~rick@81-178-248-131.dsl.pipex.com)
20:20.19fugitivoone question, I need asterisk to answer a call (zap channel), and play the IVR, then, I need to redirect the call to another pbx, i was trying to use flash and senddtmf, but that way i lose the control of the call, is any way to do that?
20:20.39rene-you do dial fugitivo
20:20.47rene-like dial(C_Party)
20:20.51fugitivocan't do that
20:20.54rene-asterisk will bridge the call
20:20.56ManxPowertzanger, I find it hard to believe that call parking is really broken in 1.0.x.  Let me investigate more
20:20.57fugitivothe zap channel is in use
20:21.00_Brianrene- if you use dial, you will connect two channels
20:21.09rene-he said redirect
20:21.11_Brianrene- he wants to use one channel for this.....
20:21.16rene-my mistake
20:21.18_Briansounds like my problems..somewhat
20:21.26_Briandoes anyone know if * has any time of audio detection?  I have an application that needs to Flashhook a call to put them on hold and then dial another extension utilizing SendDTMF.  The problem I am having is that * will continue to the next step even before the remote party answers.  If I utilize a Dial string, then i use another channel......
20:21.37bjohnsonwhat is res_crypto.so used for?
20:22.02Wazbthnaks tzanger
20:22.15fugitivo_Brian: it's a similar problem
20:22.20Corydon-wbjohnson: for rsa authentication
20:22.35_Brianfugitivo: yup..the problem i am seeing is that the * system does not handle call progress when doing a FlashHook...
20:23.11_Brianfugitivo: all call progress is built into the Dial Command, but this utilizes another channel.....
20:23.48fugitivoyes, i don't want to use another channel
20:24.00_Brianfugitivo: yup..i hear ya...
20:24.24_Brianwhat do you want to do with the call once you flashhook to send to the other pbx
20:24.36ManxPowertzanger, http://pastebin.ca/9566
20:24.55*** join/#asterisk Tili (~Tili@202-133-65-206-dialup.sat.net.pk)
20:25.07fugitivoi only want to detect if it's busy, and return to the ivr
20:25.18fugitivoif the call is answered, then hungup the channel
20:25.36bjohnsonI need to delete some files on this wrt54g to make room for the nfs package so I can use the fileserver
20:25.53ManxPower<PROTECTED>
20:25.55bjohnsonhow about res_agi.so? is that needed?
20:25.59ManxPowerread that carefully
20:26.06FuriousGeorgewhats preferable for a small * install.  using a residential gateway "router" device, or using the asterisk box for the NAT, dhcp server and firewall?
20:26.16_Brianfugitivo: sounds like you are looking for the same as I am...call progress detection with a Flashook
20:26.20FuriousGeorgei ask because i hear QoS to the outside world is mostly a myth, but i wonder about incomming QoS for voice streams
20:26.27rene-pointers on dealing with blocked analog zap trunks?
20:26.35tzangerManxPower:  005   == Parked SIP/0004f200cf85-a-7187 on 3516. Will timeout back to toll-access,s,1 in 30 seconds
20:26.36ManxPoweryou can't do QoS on incoming packets, only transmitted packets
20:26.41tzangerit looks like it's specifically doing that
20:26.50fugitivo_Brian: it seems there's no solution...
20:26.50ManxPowertzanger, Yeah.  why? 8-)
20:27.00tzangerPark says it's being called with empty parameters
20:27.04dca[laptop]anyone ever seen this error:
20:27.05tzangerdoes it default to s?
20:27.06dca[laptop]Apr 14 13:40:38 WARNING[8281]: res_config_odbc.c:277 realtime_multi_odbc: SQL Fetch error!
20:27.06dca[laptop][SELECT * FROM extensions WHERE exten LIKE ? AND context = ? AND priority = ? ORDER BY exten]
20:27.10bjohnsonwhat about app_adsiprog.so?  what does that do?
20:27.25[TK]D-FenderFuriousGeorge : VoIP through NAT is a PITA.  Save yourself some suffering and use your * box as your router
20:27.29FuriousGeorgemanxpower, so i could use a residential "router" gw and there is no reason it wouldnt work as well as using a nix machine, or the  server itself to do the NAT, etc
20:27.33_Brianfugitivo: there is always a solution......just gotta find it :)
20:27.49*** join/#asterisk shmaltz (~chatzilla@69.28.255.210)
20:28.10ManxPowerFuriousGeorge, There are some ways to help with packet priority, but it's complex
20:28.13FuriousGeorge[TK]D-Fender:  going to wikipedia to see what PITA is.  you would do it on the same box as your * server (small install) or is that generally a bad idae
20:28.23[TK]D-FenderPainInTheAss
20:28.33FuriousGeorgegotcha
20:28.35fugitivoanyone knows the solution? :)
20:29.02*** join/#asterisk Delvar (~irc@83.146.53.34)
20:29.07[TK]D-FenderFuriousGeorge : Yeah, use your * box as a router (NAT on iptables).  I do it and everything works great with 3 shell lines to set it up.
20:29.32*** join/#asterisk bkw_ (~bkw_@bkw.developer.and.friend.of.asterisk)
20:29.32*** mode/#asterisk [+o bkw_] by ChanServ
20:29.36FuriousGeorgeinteresting b/c linux people always say that a server should be specialized
20:29.55FuriousGeorgeone of my friends said "your firewall takes all the crap" so you dont want to put critical stuff on it
20:29.56[TK]D-FenderoF COURSE i COULD PROBABLY BE syn'D OUT OF EXISTANCE, BUT i LIKE TO LIVE ON THE EDGE :d
20:30.10_Brianfugitivo: if i find it...i will let you know :)
20:30.14stevejD-Fender: Can you pass on the shell lines?
20:30.21shmaltzfugitivo, what you trying to do?
20:30.29fugitivoFuriousGeorge: it's true, but if the crap affects the firewall, then the crap will affect all you'r network ;)
20:30.32[TK]D-FenderFuriousGeorge : I use mine as my HTPC server too as well as X-10 :D  Don't worry about eggs in a basket
20:30.55_Brianshmaltz: he is looking for call supervision on a Zap channel when a Hookflash is sent...
20:30.55[TK]D-Fenderstevej : Nope, I'm outta here unfortunately.... I'm here often though.
20:30.58rene-FuriousGeorge: we use it as vpn and router also
20:31.06[TK]D-FenderLater people!
20:31.27shmaltzlike chanspy?
20:31.38FuriousGeorgelol, im a big fan of the one box does everything approach.  i love hearing that you use it as an htpc too
20:31.51fugitivoshmaltz: asterisk is answering the calls, playing the IVR, and then it transfers the call to another PBX, i'm using flash and senddtmf for that, but the problem with that, is the busy signal, once i transfer the call, i lost the control of it, so i can't do anything if the extension of the other pbx is busy
20:32.13FuriousGeorge[TK]D-Fender:  X-10, as in security cameras
20:32.14fugitivoshmaltz: i want to check if the other pbx is busy, if it's busy, then play again the IVR
20:32.51shmaltzwhat is the make of the other PBX?
20:33.32shmaltzb/c on some of them you could have asterisk set up as a foreign PBX, or VM system, and program it to always use DTMF, even for busy
20:33.40fugitivoshmaltz: it's just like that, i can't use asterisk for everything yet
20:33.59shmaltzfugitivo, that I understand, by question however is what PBX is it?
20:34.17shmaltzcan you program the PBX to use an external VoiceMail system?
20:34.41fugitivono :)
20:34.43ManxPowertzanger: This is a little simplier and clearer: http://pastebin.ca/9568
20:34.55fugitivoits a panasonic 616
20:35.10shmaltzfugitivo, you could with pana
20:35.20shmaltzjust let me hang up on this phone call
20:35.25fugitivookey
20:35.49tzanger023   == Starting SIP/0004f201e463-a-7650 at toll-access,s,1 failed so falling back to exten 's'
20:35.52tzanger<PROTECTED>
20:35.55tzangerI don't like that
20:36.01tzangerand Park is still saying "I *will* go to s"
20:36.04tzangerso that is where it's broken
20:36.17*** join/#asterisk jcollie (~jcollie@lt16586.campus.dmacc.edu)
20:36.39elrichttp://pastebin.ca/9569   <---- can someone please take a look at this and tell me if I am doing anything wrong?
20:37.52blitzragetzanger: get back to work
20:38.22ManxPowertzanger, Yeah.  Looks like it actually is broken.  This is bad.
20:38.45elricbasically the macro in that link I posted never gets executed once the call is connected
20:39.06*** join/#asterisk Juxt (~Juxt@adsl-068-213-216-087.sip.bct.bellsouth.net)
20:39.12Juxtgood afternoon
20:39.17*** join/#asterisk swente (faOsXUs6pl@hal.infinitumb.de)
20:39.26swente'lo
20:39.32Juxtcan someone suggest a linux or bsd based firewall distro?
20:39.49elricJuxt yes www.m0n0.ch
20:39.53shmaltzelric, whats that machinedetect command
20:40.29elricshmaltz, http://www.thenetbrain.com/files/app_machinedetect.c <-- that is an answering machine detect
20:40.35elricion app
20:41.20fugitivoJuxt: openbsd
20:41.51swentebefore going any deeper into voip-matter, i'd like to know what bandwith a voip-connection needs for say .. isdn telephony quality, and what ip-latency is considered as the minimal requirement.
20:41.56Juxt<PROTECTED>
20:42.00shmaltzelric, why u using transfer and not dial?
20:42.20fugitivoJuxt: then what are you asking for? livecd?
20:42.33Juxtmonowall probably will work just fine
20:42.42dca[laptop]anyone from digium around?
20:42.46Juxti tried smoothwall but it doesn't have traffic shaping
20:42.47Juxtmonowall does
20:42.49Juxtso i think i am set
20:42.51dca[laptop]or from voipsupply?
20:42.52*** join/#asterisk n4y (~tmalkut@fire2.orasoft.net.pl)
20:43.15elricshmaltz, thats an outgoing call made from the manager interface, if a machine is detected the call is to be transferred to an extension
20:43.36elricbut that will be eventually
20:43.51Poincarewhere do I define the sounds dir?
20:44.02elricright now i just want machine detect to woek and it doesnt. the macro never gets executed.
20:44.59shmaltzwhat do you mean never gets executed, you dont see it doing step 1 ?
20:45.41elricnope shmaltz , it is supoosed to display a message on the CLI, it never does.
20:46.11elricif i just do machinedetect it shows output on the CLI
20:46.11shmaltzthen you are not executing it right from the manager interface, try using a direct exten
20:46.32elricthat is a direct exten shmaltz
20:47.24elricif it works here, i will use it in my manager script.
20:47.27shmaltzelric you have something like this:
20:47.29shmaltzexten => 1234,1,Macro(detect)
20:50.16*** join/#asterisk Gh0sty (~Ghosty@ip-81-11-199-204.dsl.scarlet.be)
20:50.17elrichrm, MachineDetect actually needs to work on a connected outgoing call but I guess i could make that work somehow
20:50.17_Briandont you just love when someone says "Yeah, I can help you with your issue, just email me"....you email the specifics, and then you get a email back telling how the rates and how much they want to charge you for the help the offered for free....
20:50.37shido6I never offered help for free, _Brian
20:50.40_Brianhow sad..
20:50.46shido6it is
20:51.29_Brianyou never indicated that you were looking to consult either..
20:52.17_Briani am glad i included some configs...for your later use....bleh
20:52.46tzanger_Brian: relax, shido6's a regular here
20:53.04_Briantzanger: oh i am relaxed...:)
20:53.13tzangerI very seriously doubt he's interested in stealing your tech
20:53.18_Briani just hate misrepresentations
20:53.33Sedoroxhrm
20:53.37_Brianwhatever..
20:54.10*** part/#asterisk Juxt (~Juxt@adsl-068-213-216-087.sip.bct.bellsouth.net)
20:54.38smiley-is there any problems with softphones and #NN# extensions?    I just can't get them to work...
20:55.17shido6assumption is the mother of all fuckups
20:55.21elricah I wish techie were around
20:55.43smiley-with my regular phone connected to a SIP-box there is no problem..   but from all the softclient I have tested it fails..  Sjphone, X-lite and SIPS
20:55.44_Briani will have to quote you on that one shido6
20:55.54Sedoroxhrm
20:56.25_Brianrofl..now i will get a email saying I have to royalty fees for quoting...
20:57.31seanhello, all. I'm definitely a newbie when it comes to asterisk, but can generally find my way around. I'm trying to connect xlite to my asterisk. I get "Apr 14 16:57:01 NOTICE[6150]: chan_sip.c:7532 handle_request: Registration from 'sean <sip:sean@caedmon.net>' failed for '10.20.30.100'" Can someone please point me in the right direction?
20:57.33elric_Brian, i fail to see how someone could charge you royalty for colloquial forms of speech?
20:58.00_Brianelric: it was a joke.... :)
20:58.15elric_Brian, ah
20:58.16Nuggetsean: make sure sip.conf is well-formed and make sure you're getting the password right.
20:58.21_Brianmaybe not a good one :)
20:58.23syleApr 14 13:58:04 WARNING[23628]: res_musiconhold.c:565 moh_register: Unable to open pseudo channel for timing...  Sound may be choppy.
20:58.26sylewhat is this
20:58.29*** join/#asterisk izo (~izo@izo.warpl.ipxxi.pl)
20:58.33shido6sean, are you using a type=friend in your sip.conf?
20:58.35shmaltzsean, you are doing something wrong with the registration
20:58.36shido6pastebin.ca your config
20:58.45shido6and reply here with the pastebin link they provide you
20:58.51seansec
20:59.06*** join/#asterisk techie (gus@asterisk.horizonte.us)
20:59.26seanyes, type=friend for [xlite]
20:59.30seanI'll paste
20:59.32elricah techie i was looking for you, if you have a minute or two to spare?
20:59.33Nuggetfor [xlite]?
20:59.38Nuggetyou're trying to log in as [sean]
20:59.38shido6break it out to a peer and user for the xlite user
20:59.43techieelric: sure.
20:59.45shido6users dont need hosts and peers dont need contexts
20:59.57shido6and reload chan_sip.so at the CLI
21:00.27syleanyone running fedora core 3 with no problems with asterisk?
21:00.55RChadwellI am
21:01.00Nuggetit's impossible to run linux with no problems.  :)
21:01.04seanok, I'll try that. Thanks
21:01.54|Vulture|fc3 is fine
21:02.06|Vulture|fc3+te110p == problems but I worked around them
21:02.23|Vulture|but its also because its new hardware from digium
21:02.53*** part/#asterisk RChadwell (~rob@rrcs-24-227-48-86.se.biz.rr.com)
21:03.03bannermanI don't like the hardware that I'm using for my asterisk box, and another server became available, and I want to load a new system from scratch. Any advice for Linux distro?
21:03.27SedoroxGentoo....
21:03.41bannermanSedorox: seriously?
21:03.48SedoroxI love gentoo
21:04.09SedoroxGentoo.. Debian...
21:04.13Nuggetasterisk doesn't care, so use whatever pleases you.  anything beyond that is a religious decision.
21:04.14fugitivoGentoo!
21:04.15Sedoroxmy top choices
21:04.19Sedoroxor FBsd
21:04.31elrici use FreeBSD with *
21:04.34funxionhey |Vulture|
21:04.40Sedoroxsame here...
21:04.49Sedoroxmy * boxes are actually fbsd
21:04.51*** join/#asterisk RoyK (~roy@143.80-202-166.nextgentel.com)
21:04.53RoyKhm
21:05.00funxiongot past the module problem
21:05.05funxiongot another problem now
21:05.17RoyKseems to me a single xeon 3.0 can easily run two te410p cards
21:05.29funxiontrying to configure PRI to connect to meridian opt 81c
21:05.35*** join/#asterisk mhnoyes (~mhnoyes@user-2ivfnji.dialup.mindspring.com)
21:05.39funxionanyone ever done this before
21:07.25sivanais Digium going to have a single span card with built-in echo can?
21:07.42RoyKis jesus going to return as a muslim?
21:07.57seanpotentially stupid question: how do I decrease verbosity at the CLI
21:08.12drumkillaset verbose 0
21:08.21seanthanks.. sorry for the silly question
21:08.24RoyK:)
21:08.26drumkillano problem, it's not silly
21:08.38RoyKstupid is as stupid does
21:08.39RoyK:P
21:09.16drumkilla<PROTECTED>
21:09.17drumkillaoops!
21:09.19drumkilla:p
21:09.28sylethat ztdummy.o doesn;t even compile by default for rh9
21:09.43elricok i am beyond my wits end
21:09.49*** join/#asterisk brc_ (~brian@brc.base.supporter.pdpc)
21:10.04drumkillasyle: it doesn't compile by default, period
21:10.29sivanadoes any mfg sell a single T1 echo can?
21:10.36*** join/#asterisk Blackvel (~blackvel@dsl-082-083-172-135.arcor-ip.net)
21:11.21Blackvello, have no connected my ata (zap pbx crashed) to * and changed extensions.conf for using now a macro. anyone wants to give me a test call? fwd, sipgate or nikotel is fine
21:12.03dca[laptop]most boxes are 3.3v, yes?
21:12.36shido6next
21:12.47shido6brb
21:14.13*** join/#asterisk Syncros (~sysop@noc.routermonkey.net)
21:14.23*** join/#asterisk dave_mwi_ (~dave_mwi@adsl-11-102-74.mia.bellsouth.net)
21:14.46dave_mwi_anyone know off-hand how many auto atendants asterisk can handle at once?
21:15.36*** join/#asterisk Fabrizioxxx (1002@ip-138-106.sn1.eutelia.it)
21:15.49drumkilla021980293840234823!!!!!!
21:15.50*** join/#asterisk jf_ (~jeanfranc@modemcable077.187-80-70.mc.videotron.ca)
21:15.54Blackveloh cool
21:15.56*** part/#asterisk Fabrizioxxx (1002@ip-138-106.sn1.eutelia.it)
21:16.05Blackvelcalled my pstn voip number and it seems to work :)
21:16.16*** join/#asterisk harryvv (~leonardo@S010600055d210201.vs.shawcable.net)
21:16.39|Vulture|Blackvel: thats just the beggining
21:16.42harryvvHas anyone sucessfully tested astrisk on a ide flash card?
21:16.44Blackvelnope
21:16.44*** part/#asterisk swente (faOsXUs6pl@hal.infinitumb.de)
21:16.48Blackvelmy pbx crashed
21:16.55Blackvelhad to reconfigure my complete *
21:17.02P-ChanOk, instead of trying to figure ways to do it and then ask questions, let just throw this out there:  1 asterisk server w/ 2 PRIs, IAX2 Trunk to other remote asterisk servers, we want to have fax capabilities @ the sites with the remote asterisk servers, what's the best way to do this?
21:17.11harryvvdid not backup your config files?
21:17.17jf_Someone know why i can transfert from sip to sip or iax to sip but i can't transfert to another sip while talking on zap channel (pstn)
21:17.19Blackvel|Vulture|: i am no newbie :)
21:17.34Blackvelnot my asterisk pbx, but my telco pbx
21:17.35fearnorsivana: not that i know of
21:17.47fearnorbut i wish asterisk had a non-gay echo canceller so it wouldn't be necessary
21:17.49*** join/#asterisk Gunnar (~gunnar@34.80-203-209.nextgentel.com)
21:17.50Poincarewhere do I define the sounds dir?
21:17.51fearnorg.168 uber alles
21:18.08sivanafearnor: yes, I agree
21:18.09Blackveli can not use my phones anymore to call into * (call thru). too bad ;)
21:18.16harryvvfearnor, there are third party echo cancelers on the market
21:18.22drumkillaPoincare: don't think it's configurable ... /var/lib/asterisk/sounds/
21:18.37Blackvelokies, cu tomorrow
21:18.42|Vulture|Blackvel: oh nice... how much did that run you?
21:18.52Poincaredrumkilla: it's located elsewhere in the debian packagers
21:18.57sivanaharryvv: is there one that does a single t1
21:19.14Poincareis the location available as a predefined variable?
21:19.18harryvvsil, I think so ask paulc I think he told me one one once.
21:19.30fearnorharry: those are usually pretty big things
21:19.30*** part/#asterisk jcollie (~jcollie@lt16586.campus.dmacc.edu)
21:19.38fearnorand cancel 20 t1s at a time ;)
21:19.44|Vulture|fearnor: option AGGRESSIVE_SUPPRESSOR solved most my problems
21:19.45drumkillaPoincare: oh.  :)
21:20.03fearnorasterisk echo canceller is *broken*, period end of story.
21:20.16fearnoraggressive or not
21:20.21jf_Someone know why i can transfert from sip to sip or iax to sip but i can't transfert to another sip while talking on zap channel (pstn)
21:20.31|Vulture|fearnor: broken implies it worked at one time
21:20.39harryvvfearnor, my echo is totally gone on my x100p
21:20.44harryvvnone zero.
21:20.45*** join/#asterisk tessier_ (~treed@210.245.99.64)
21:21.57P-ChanAnyone know how to do "SIP Trunking"?  I can't find any info in voip-info.org wiki, maybe its called something else??
21:22.06*** join/#asterisk smurfix (~smurf@smurfix.developer.debian)
21:22.39syleApr 14 13:58:04 WARNING[23628]: res_musiconhold.c:565 moh_register: Unable to open pseudo channel for timing...  Sound may be choppy.
21:22.42sylethis a bad error?
21:22.59file[mac]it's a warning
21:23.04fearnorsyle: i don't know. do you mind choppy sound?
21:23.07sylethis server is just used for sip/aix only stuff no cards
21:23.23Corydon-w~ztdummy
21:23.24jbotztdummy is probably zaptel timing source which uses a usb-ohci compatible usb controller as source. (part of zaptel cvs)
21:23.25bugbotztdummy is assigned nothing and reported nothing.
21:23.38sylei tried loading it and got problems
21:23.44Corydon-wOoops... that should be usb-uhci
21:24.21harryvvprobebly? it is
21:24.32sylegoing to paste 4 lines sorry if it is not allowed
21:24.34syleice:~/voip/zaptel# insmod ./ztdummy.o
21:24.34syle./ztdummy.o: unresolved symbol zt_unregister
21:24.34syle./ztdummy.o: unresolved symbol zt_transmit
21:24.34syle./ztdummy.o: unresolved symbol zt_receive
21:24.34syle./ztdummy.o: unresolved symbol zt_register
21:24.35syleice:~/voip/zaptel#
21:24.39Sedorox~pastebin
21:24.40jbotpastebin is, like, a place to paste your stuff without flooding the channel - try http://pastebin.ca
21:24.44*** join/#asterisk Rick_Hunter (~rhunter@01-098.008.popsite.net)
21:24.45bugbotpastebin is assigned nothing and reported nothing.
21:25.10Sedoroxhehe
21:25.23SedoroxI _think_ you might need another module loaded
21:25.25|Vulture|isnt he suppose to be in -dev?
21:25.43file[mac]syle: try modprobe ztdummy
21:26.23sylemodprobe: Can't locate module ztdummy
21:26.23Sedoroxmodprobe ./ztdummy
21:26.41sylesame thing
21:26.49file[mac]syle: did you, install the modules?
21:27.00bkw_depmod -a
21:27.04bkw_modprobe ztdummy
21:27.27sylei thought the make install would have
21:27.29sylebut it didn;t
21:27.31|Vulture|bkw_: will there be a release of ChanSpy for v1-0?
21:27.44sylebad enough i had to do a make ztdummy just to get the .o file
21:28.13Sedorox....
21:28.22bkw_|Vulture|, NO
21:28.34syleso can i manually copy it somewhere maybe
21:28.39bkw_ztdummy builds by default on 2.6
21:28.48sylenot on redhat 9
21:28.59|Vulture|bkw_: hehee okay :P
21:29.24dca[laptop]bkw_: got a sec?
21:29.31dca[laptop]having realtime w/ odbc errors
21:29.38dca[laptop]wondering if my mysql5 is too blame
21:29.49*** join/#asterisk PTG123 (~PTG123@66.213.239.122)
21:29.57bkw_no time
21:29.58sylehmmm
21:30.07syleapparentely it install no modules anywhere on make install
21:30.18*** join/#asterisk Egonis (~chultay@69.194.211.129)
21:30.19jf_to transfert a call i any situation do i have to put T in all dial()
21:31.47syleoriginally i wanted to go with fedora core 2 when i first installed this but it didn;t support the DELL 2850
21:31.57syleso i went redhat 9.0
21:32.19sylei bet fedora core 3 would work now but that means a whole reinstall remotely blah
21:32.26brainchilredhat 9 is EOL almost a year ago
21:32.42brainchilno security updates unless you do it yourself
21:33.31sylewell security updates don;t bug me i compile everything by hand anyways, but updated package releases is a must
21:33.49sylelike cvs etc, standard crap
21:34.00syleso yeah i got to throw out rh9 eventually i guess
21:34.22brainchilUse centos if you want a redhat like OS
21:34.27*** join/#asterisk _mountie (~mountie@CPEdeaddeaddead-CM000a739acaa4.cpe.net.cable.rogers.com)
21:34.39sylei read up on centos and whitelinux or whatever
21:34.41brainchilit's a binary compile of the redhat enterprise distro
21:34.47sylewhitelinux doesn;t even want to do it anymore
21:34.51brainchilcentos is also what asterisk@home uses
21:34.55sylefedora there is a core team of developers
21:35.23brainchilyes but it's as unstable as charles manson by design
21:35.23sylei heard asterisk is now using windows as their devel environment
21:35.32brainchilit's meant to be bleading edge
21:35.36*** join/#asterisk zotz (~zotz@24.231.32.109)
21:36.23*** join/#asterisk topping (~topping@cpe-24-210-82-196.columbus.res.rr.com)
21:36.36sylei don;t know about unstable, i ran fedora core 2 on a couple other servers and it worked out fine
21:36.50*** part/#asterisk dave_mwi_ (~dave_mwi@adsl-11-102-74.mia.bellsouth.net)
21:37.15sylei know its redhat's development branch for their enterprise releases but still pretty stable
21:37.20brainchilI honestly gave up on redhat between 8-9 when they imported new posix threading into 2.4 initially and broke a bunch of stuff
21:37.36brainchilI don't even think Fedora is stable enough for desktop use
21:38.13bannermanbrainchil: my asterisk box ran for several weeks, at the time I had gnome and was using it as my workstation as well
21:38.14syleidk i don;t use fedora for desktop use, i use it as a server and xp with securecrt as a client
21:38.15Bentleyhello all.  I've got a system running fxos on 2 tdm400ps.  On occasion, an outbound call and an inbound call will get answered by the same Zap channel (ie: dial out with sip client and immediately find an unrelated caller on the horn with you).  Anyone else experience this?
21:38.21brainchilall of my servers are running slackware or debian now becasue I was tired of redhat breaking things
21:38.27*** join/#asterisk _mountie (~mountie@CPEdeaddeaddead-CM000a739acaa4.cpe.net.cable.rogers.com)
21:38.30bannermanbrainchil: using FC3 was an improvement from WinXP for sure
21:38.52bannermanBentley: I haven't, but it sounds hilarius :)
21:38.59brainchilusing anythins is an improvement from XP
21:39.24sylethey did break alot of things in redhat9 but from what i have noticed everybody is going that way, so if you don;t patch your programs they won;t work on anything new regardless
21:39.28Bentleybannerman: heh - not really
21:39.33brainchil8 new critical security updates this month alone :-)
21:39.49*** join/#asterisk PTG12345 (PTG123@66.213.239.122)
21:39.54harryvvbrainchil, it is. i use fc3x86_64 with kde
21:39.57PTG12345anyone know the command to include another conf file in a conf file? :)
21:39.59bannermanBentley: yeah, sorry, I do empathize with you. Still..
21:40.13file[mac]#include "blah.conf"
21:40.14brainchilmy desktops are all ubuntu and I PREFER slackware on the servers
21:40.29blitzragefile[mac]: without the quotes right?
21:40.30syleharryvv how is that working out for you?
21:40.32brainchilbecause it's so simplistic
21:41.22harryvvsyle, I have some gripes about x86_64 since its more orianted to the server market there is alot of apps that are not compiled for it. cannot see flash plugins for firefox is a example.
21:41.24sylei use to use slackware, then so many companies i worked for over last 10 years always used redhat, so i guess i got use to it
21:41.30bannermanbrainchil: I can't stand slackware because after all these years their installer is still horrid and frustrating. You mess up on one option, have to go through 5 minutes worth of crap, go go back, redo it, hope you don't mess up again...
21:41.41CoaxDwhat is the place i should be going to - to do my taxes online?
21:41.48brainchilmy only 64 gripe is that it's not really that much faster at much of anything :-)
21:41.53harryvvso, I should not have install x86_64 if it would have been a graphics work station.
21:42.01harryvvbrain you are probebly right ;)
21:42.20Sedoroxhmmmmm
21:42.21sylepersonally i run redhat for work crap, but i still have my freebsd at home :)
21:42.29brainchilbannerman: actually all of my servers are identical I do one slackware install and mirror
21:42.31blitzragesyle: I like CentOS
21:42.57file[mac]blitzrage: you're soooo cute my dear!
21:43.00harryvvbrainchil, is there a wiki on mirroring and how fast does it take to mirror to another system?
21:43.14brainchilheheh ... freebsd is fun if you have a good ups and generator if the power outage exceeds that .. otherwise 5+_ hour fsck ... no thank you
21:43.14bannermanbrainchil: there's just no reason they shouldn't take the time to make an installer that isn't designed to be frustrating.
21:43.24harryvvI keep hearing things about CentOS what makes it that much greater over Fedora?
21:43.28harryvvor redhat?
21:43.54sylebrainshild can;t say i ever had that experience, i;ve always ran latest 4.x stable branch with no problems
21:44.15brainchilbannerman: the point is that they have an installer that is very simple ... not frustrating ... but it was meant to be unixlike and simple
21:44.27brainchilsimple doesn't mean easy ... just simple
21:44.37harryvvor clean
21:44.50bannermanbrainchil: you can do simple without making it hard.
21:45.06sylei fell in love with freebsd a long time ago when i did make install mrtg and it installed all dependancies with it for me lol
21:45.06brainchilsyle: got a disk bigger than 40Gs? start a heavy write-read operation and hit the power button or unplug it
21:45.28brainchilit's meant to actually teach you linux/unix
21:45.29*** join/#asterisk mark_wales (~Mark@cpc3-swan1-4-0-cust224.swan.cable.ntl.com)
21:45.44brainchilredhat doesn't teach you linux it teaches you redhat
21:45.48*** join/#asterisk ManxPower (~eric@stirprop-S0-0-0-26.ndcr2.datasync.net)
21:45.52harryvvyup
21:45.58sylenaw freebsd is meant to be for more experienced unix users
21:46.04brainchilbecause like many other distros they have their own F%$ked up way of doing a lot of things
21:46.06syleafter your done with linux
21:46.11sylemove on to freebsd
21:46.14sylethen solaris etc
21:46.14harryvvI have done freebsd
21:46.24brainchilno freebsd is meant for people that are into S and M
21:46.39brainchilI like freebsd ... the filesystem is just a piece of shit
21:46.43fugitivo"after your done with linux" ?
21:46.48sylewell
21:46.50syledepends
21:46.51fugitivowhen are you done with linux?
21:46.54brainchiljust like openbsd
21:46.57syledepends on what you are doing
21:47.07sylei use freebsd on single cpu machines
21:47.08brainchiland don't even start talking about soft updates and background fsck
21:47.09elricbackground fsck
21:47.10elric:(
21:47.12*** join/#asterisk MatsK (~matsk@107.80-202-57.nextgentel.com)
21:47.18elriconly peev with freebsd
21:47.21sylei use linux on dual cpu machines as SMP sucks on bsd
21:47.22brainchilthat's a half ass solution to the problem
21:47.53brainchilever use a machine with a newer reiser/jfs/xfs?
21:47.55elricI like Solaris 10, performs alright on x86,
21:48.00mark_walesi wonder if anyone could help with a config/hw prov I have.... I am unable to make outbound calls from a SIP client but I am getting 'No Channel type registered for 'Zap' and 'unable to open /dev/dsp: No such device - i have used modprobe zaptel, modprobe wcfxo and ztcfg before launching asterisk.  I am using an X100P card connected to a standard PSTN line.  Any help would be greatfully appreciated
21:48.01Nuggethere comes the religion.
21:48.07fugitivoelric: isn't it slow?
21:48.12sylenaw, with ext3 out i stopped caring about installing reiser
21:48.12brainchilhard down to up and recovered with a 400 gig array in 10 minutes
21:48.26P-ChanSo, SIP trunking for ulaw support for fax transmission?  No FAQ on sip trunking on voip-info.org - is it even possible with Asterisk?  (Or am I only asking because I don't fully understand asterisk and need to read between the lines?)
21:48.39brainchilext3 is just ext2+journal ... it's really not very advanced
21:48.45elricfugitivo, a bit. But generally better than all previous releases of Solaris on x86
21:48.56harryvvmark, have asterisk running on it?
21:48.56brainchilit's just what redhat stuck with because it was an easy migration bath for ext2
21:49.03file[laptop]P-Chan: SIP, trunking, no
21:49.08fugitivoelric: did you try asterisk on that? :)
21:49.13syleare yopu running latest 2.6.x kernels on your slackware boxes?
21:49.18*** join/#asterisk riksta (~rick@81-178-248-131.dsl.pipex.com)
21:49.34harryvvmark_wales, is asterisk running on that solaris system
21:49.41elricfugitivo,  not yet, but I will soon. Right now I am smitten with FreeBSD
21:49.46P-Chanfile[laptop] - bah.... so for fax signals to "sanely" traverse a Asterisk to Asterisk trunk, what do I use?
21:49.58sylesolaris sucks on x86 you really want sparcs
21:49.58file[laptop]a regular ULAW channel?
21:50.13brainchilsolaris is faster on x86
21:50.15elricsyle, have you tried the new Solaris 10?
21:50.18mark_walesno SUSE9.1
21:50.24mark_wales2.6 kernel
21:50.33brainchilyes some 2.6.x kernels
21:50.38P-Chanfile[laptop] - ?  You mean don't use trunking, just a standard sip forwarding (for lack of a better term)?
21:50.39elricplus OpenSolaris should be interesting
21:50.45sylenaw i gave up on solaris after i stopped working for this company, who knows when they will start charging for it again
21:50.57brainchilbut the most open solaris will still be encumbered
21:51.08brainchilthough they at least do journaling :-)
21:51.23brainchilsorry ... filesystem logging
21:51.26brainchil:-)
21:51.45file[laptop]P-Chan: SIP trunking doesn't exist in asterisk
21:51.48syleif a company can afford sparcs and solaris and veritas, oracle, fiber channel arrays then sure go for it, otherwise a nice load balancer and multiple linux boxes is a much cheaper solution
21:52.02brainchilamen
21:52.06elricmy old work had a 14 processor Ultra Sparc3 machine
21:52.11elricit was a beast
21:52.18file[laptop]P-Chan: what I'm saying is just have it as a regular ULAW phone call..
21:52.18P-Chanfile[laptop] - Ok, I gathered that much from your first statement, but you say a "ULAW Channel" - can you clarify that a bit?
21:52.35brc_MARVIN[laptop]!
21:53.04P-Chanfileplaptop] - Ok, so add a sip client to sip.conf on the originating server and authenticate the receiving server that way and use "ulaw" as the compression?
21:53.24*** join/#asterisk MatsK (~matsk@107.80-202-57.nextgentel.com)
21:53.25file[laptop]P-Chan: sure... there's nothing special about the fax, it's just another phone call passed through uncompressed
21:53.43syleholy crap elric
21:53.52syleidon;t even want to ask how much that thing cost lol
21:54.00*** join/#asterisk R3DB0x (nobody@66.142.28.36)
21:54.12P-Chanfile[laptop] Ok, thanks.  This helps get me on the right path. ;)  I'm still kinda wet behind the ears - learning from an existing system as I need to. /sigh
21:54.16EgonisI just did a successful install and a sip phone is now connected... the demo worked, but dialing '#' or '2' do not cause any response..
21:54.45juiceib269sssssssssssssssssssssssssssssssssssssssssssssssssssssssssssssssssssssssssssssssssssssssssssssssssssssssssssssssssssssssssssss, .vjdfii`sssssssssssssssssssssssssssssssssssssssssssssssssssssssssssssssssssssssssssssssxljdsj
21:54.54blitzrageEgonis: sounds like a dtmf problem. Try dtmfmode=rfc2833 in sip.conf
21:55.18elriclol syle, a shitload i tell you
21:55.34Egonisblitzrage: will try that, ty!
21:55.51elricit had more ram than my hdd had gbs
21:58.14Egonisblitzrage: doesn't seem to have helped, I tried dtmfmode=info as well
21:58.35blitzrageEgonis: and you did a reload chan_sip.so ?
21:58.40blitzrageafter making the changes
21:58.50Egonisblitzrage: I did /etc/init.d/asterisk restart
21:59.11jf_when transfering a call, it seem to only take 1 digit, how can i set the length
21:59.14blitzrageEgonis: paste the dialplan and your phone definition in sip.conf
21:59.24blitzrageEgonis: to pastebin.com
21:59.33tzanger~handbook
21:59.34jbothmm... handbook is http://www.digium.com/handbook-draft.pdf
21:59.34bugbothandbook is assigned nothing and reported nothing.
22:01.00file[laptop]blitzrage: what's up?
22:02.26blitzragefile[laptop]: not too much... trying to install gstreamer
22:02.44blitzragefile[laptop]: keep getting a PKG_CONFIG_PATH error :)
22:03.04file[laptop]bad blitzrage bad
22:03.14blitzragethe handbook made sense after I understood Asterisk
22:03.25Egonisblitzrage: http://www.pastebin.com/271379
22:04.31pgpkeysi haven't even started a setup yet. i just printed the manual, installed asterisk via ports, looked through a couple *.conf files to see what everything looked like, read the first couple sections of the book,and waiting til i feel slightly better before diving in
22:04.33Egonisblitzrage: dialplan?
22:04.55pgpkeysdamn cold is kickin my ass. not much energy for my usual 24h+ runs i usually do to learn something
22:07.42*** join/#asterisk nvrs (RUR@toronto-HSE-ppp4255113.sympatico.ca)
22:09.30blitzragedialplan = extensions.conf
22:11.17syleanyone use osoft
22:11.50sylei bought that asterisk ebook and that is the reader for it, won;t get me copy anything to paste
22:11.55tzangersounds like toilet paper
22:12.19tzangeryou just HAVE to try this new toilet paper!  What's it called?  Osoft!
22:12.31syleosoft thout reader
22:12.38Egonisblitzrage: http://www.pastebin.com/271385
22:12.59syleanyone in here actually do development on asterisk?
22:13.25syleyou do the c coding for it?
22:13.29tzangerbkw, blitzrage, manxpower, mikej, kpfleming, pcadach, anthm, lots of people
22:13.39tzangerI have submitted a few patches
22:13.55blitzragewow, I'm honored to be included in that list :)  I'm no developer, I'm a solutions guy :)
22:14.03blitzrageI "get shit done".
22:14.14tzangeryeah me too but that sometimes involves coding
22:14.25Egonisblitzrage: fyi: I haven't touched extensions.conf
22:14.37sylehow do you feel about the windows environment they are working on now
22:14.59blitzragetzanger: true... I don't know C unfortunately. One of these days when I have some money I'm going to take a fast paced C course to get myself some fundamentals
22:15.10tzangerblitzrage: don't do that
22:15.15blitzrageI know of phone systems running on NT4 and they suck
22:15.19tzangerI can teach you the fundamentals of C easily
22:15.35tzangerI even have a fun doubly linked list course
22:15.40syleyou don;t learn c to learn fundamentals you learn it to code it everyday hehe
22:15.41blitzragetzanger: really? I need a crash course for a few hours
22:15.50EgonisMy server has no sound card... what do I need to config for alsa?
22:16.03tzangerEgonis: ... a sound card!
22:16.03blitzrageI know how to script, just don't know C syntax and pointers :)
22:16.12Egonistzanger: Do I *NEED* one for asterisk?
22:16.14*** join/#asterisk riksta (~rick@81-178-248-131.dsl.pipex.com)
22:16.20sylesocket programming is the most fun
22:16.33blitzragesound cards are NOT needed except for CONSOLE channel
22:16.52Egonisblitzrage: So do I leave alsa alone? because in the 'messages' log, it says /dev/dsp not found
22:16.59blitzrageif no sound card, don't load the modules. In modules.conf set: noload => chan_alsa.so
22:17.02sylebut i don;t consider myself a leet programmer by any means since i have never coded a device driver in c or asm lol
22:17.18tzangerEgonis: no, you need one for alsa
22:17.27tzangeryou asked how to config for alsa
22:17.30tzangerand you need a sound card for that
22:17.37blitzragesyle:  I just want to be able to read and understand C a bit better and be able to implement ideas
22:17.39Egonisblitzrage: that is already set
22:18.12blitzragetzanger: I'm serious, I want to learn C. Let me know if you are serious about teaching some fundamentals (you know my IM :))
22:18.26tzangerI am serious
22:18.29sylelots of free tutorials online
22:18.40blitzragesyle: thats not going to work. I don't have the time to learn it on my own
22:18.43sylefor basic variable, arrays , pointers stuff
22:18.51*** join/#asterisk Rez (lorez@lorez.staff.freenode)
22:19.14blitzragesyle: in terms of all the reading etc. A crash course would do me a world of good, then I can go and refine it all on my own time.
22:19.29blitzrageI hate being limited in Asterisk simply because I don't know C
22:19.35blitzrageits the only reason :)
22:20.35*** join/#asterisk Tuplink (~dsfsk@68-232-92-239.chvlva.adelphia.net)
22:21.16Tuplinkim having a problum...
22:21.31syleif it makes you feel any better i don;t have time for c either, i just code perl applications to run onto of base c things usually
22:21.32marloweTuplink - Ohreallllly?
22:21.53Tuplinki have FWD set to forward all calls to my PBX.... asterisk recives them but not user input afrer that
22:22.06Tuplinklike i hear it...
22:22.16Tuplinkbut nothing works the other way.
22:22.32Tuplinkcould it have somthing to do with NAT?
22:22.32blitzragesyle: I have weak programming unfortunatly... and if I'm going to spend the time learning one particular language well, by damn its going to be C :)
22:22.48*** join/#asterisk jf_ (~jeanfranc@modemcable077.187-80-70.mc.videotron.ca)
22:23.00syleif you learn c you can learn any language easily , just sytax differences
22:23.00jf_can someone help me out with call transert
22:23.21harryvvblitzrage, same here. I would start to learn it but dont see the end of the tunnel for something I would start to work on.
22:23.22sylei prefer perl usually since i had to in 2 lines of perl what it would take me 10 lines or more in c to do
22:23.31Hydroxidesyle: I think scheme is a bit more different than that, not that it's relevant for asterisk purposes
22:23.46drumkillathere is no end of the tunner :)
22:23.50drumkillatunnel*
22:23.57harryvvtypo
22:24.09syleasterisk has to stay c though for speed , i like fact its all c
22:24.12tzangerI *love* the way that bbq'd hot dogs grow to like 3x their original size
22:24.23drumkillaI love C.  :)
22:24.28*** join/#asterisk CoolCat_ (~god@200.162.252.66.user.ajato.com.br)
22:24.34harryvvtzanger,  try putting one in a microwave for 40 min :)
22:24.45CoolCat_hi again! =o)
22:24.57tzangerharryvv: forty minutes?? JESUS man I put them in for like 1
22:25.02harryvvhehehe
22:25.17harryvvit was a mistake once. It looks like what a dog would leave behind
22:25.21CoolCat_...more doubts emerge from my studies! =o/
22:25.25Tuplinkwhat ports need to be routed for IAX to work corectly?
22:25.28drumkilla4569
22:25.49CoolCat_does anyone here knows the hw zoom v3?
22:25.49Tuplinkforwarded threw a router?
22:25.53Tuplinkor just able to pass
22:26.03Hogiewhy doesn't a tdm03B detect red alarms on the line like my x101p does?  When I unplug the co line from the ports, it still tries to dial out of them....
22:26.42blitzragedrumkilla: damn those un-ending tunnels
22:26.49sylewell i am stil unsure of that ztdummy problem, the unresolved symbols usually mean its looking for dynamically linked libraries, any solutions from you guru c programmers much appreciated
22:26.50drumkillait's so much fun, though
22:26.53jf_can someone help me out with call transert
22:27.13drumkillatransert isn't even a word :)
22:27.25jf_what is the word then :)
22:27.52blitzragetransfer?
22:28.00blitzragenahh.... too easy
22:28.01file[laptop]Transfer in progress.
22:28.06file[laptop]Please hold two centuries while it is completed.
22:28.09sylemight be an idea to compile these zaptel modules statically not sure
22:28.15file[laptop]Thank you for choosing dial-up internet service.
22:28.25drumkillai'm about to work on a pretty silly bug
22:28.31file[laptop]silly you say?
22:28.33drumkillayep
22:28.37drumkillalook at 4020
22:28.39file[laptop]as silly as... YOU?!?
22:28.43CoolCat_http://www.zoom.com/products/voip_products.html  ... how good asterisk would be with it? can i gatewat global village SIP <-> skype using asterisk?
22:28.43jf_ya if i want to send the call to another extensoion like call parking
22:28.43drumkillanot quite :)
22:28.50file[laptop]otay
22:28.57file[laptop]oh that one
22:29.10blitzragedrumkilla: 4020 you say?
22:29.12file[laptop]yeah, it's uh interesting
22:29.17drumkillablitzrage: I know, haha ...
22:29.21blitzragedrumkilla: lol
22:30.52pigpenAm I correct that I can force the outgoing caller id info by:
22:30.53pigpenexten => _7NXXNXXXXXX,1,SetCallerID(5555555)
22:30.53pigpenexten => _7NXXNXXXXXX,2,SetCIDName()
22:31.02pigpenin my outgoing plan...
22:31.31pigpenEverytime I change it...it seems my line provider overrides it...
22:31.37Egonisblitzrage: so what should I do about my dtmf?
22:32.39*** join/#asterisk eKo1 (~bernd@metrored-gw.tropicohn.com)
22:33.02smiley-pigpen: sounds logical..   so you can't spoof your number
22:33.55easimonon analog lines you can't set it at all
22:34.11pigpenI really want the ability to not send caller id info at all...
22:34.16pigpenwe have a pri.
22:34.28smiley-ah
22:34.32Tuplink<--- needs help.... i can make a call from asterisk to a FWD user and all goes fine.... i can recive a call from FWD can hear me but i cant hear them whats the prob?
22:34.41smiley-in .se you can dial #31# as prefix to hide the number...
22:35.00pigpencan I have * prepend this automatically?
22:35.03easimonpigpen: did you try to clear it?
22:35.13pigpentry to clear it?
22:35.28easimonpigpen: instead of giving a bogus number.
22:35.31*** join/#asterisk Rez (lorez@lorez.staff.freenode)
22:35.42pigpenlike make it blank....yes.
22:35.51pigpenstill sends the real number.
22:36.00Tuplink<--- needs help.... i can make a call from asterisk to a FWD user and all goes fine.... i can recive a call from FWD can hear me but i cant hear them whats the prob?
22:36.24pigpenhmm...I will try one of my other numbers...
22:36.30pigpenI have 100...why not.
22:36.51easimonpigpen: setting any other *valid* number should work at least...
22:37.09pigpenyeah..that is what I was thinking...
22:37.25easimonpigpen: not sure about how to send no id...
22:38.05pigpenyeah..that worked..
22:38.09pigpenI will call the telco then...
22:38.15pigpenthanks...
22:38.37PatrickDKpigpen, have you tried *70,number
22:38.53PatrickDKor does that not work on pri
22:39.50Tuplinkany one here have a FWD station i can try to test my FWD stuff?
22:40.36PatrickDKtuplink, no, fwd has their own test numbers for you to use
22:40.49Tuplinkyes... what would that be?
22:41.07Tuplinkits jsut that i like a real person for trouble shooting
22:41.12PatrickDKI don't know, ask fwd
22:41.13CoolCat_someone saw my zoomtel v3 ask?
22:41.55jf_can someone tell me why i am able to park any call from my cell phone and im not from sip
22:42.12PatrickDKjf_, cause you supplied T or t
22:42.19harryvvTuplink, is FWD sip or iax?
22:42.26TuplinkIAX
22:42.28PatrickDKharryvv, both
22:42.34FengShuianyone here conversant with the channel cloning in ast_do_masquerade?
22:42.35harryvvwhich one is he using
22:42.50jf_Patrick: i set both
22:42.58PatrickDKjf, that is the problem
22:43.13PatrickDKone is for allowing the person making the call to, the other is for the person recieving the call to
22:43.29harryvvTuplink, wakup
22:43.51jf_i know
22:44.12PatrickDKwell, if you specify both, then everyone can do anything to the call
22:44.23jf_but for the one making the call (sip) i cant, i can from receiving the call
22:44.33jf_i know im just testing
22:45.32jf_can i allow both
22:45.41PatrickDKjf, sounds like you don't have dtmf signalling set right
22:45.51PatrickDKinband or outofband, or sip info
22:45.52jf_on sip side
22:46.00PatrickDKon sip phone and asterisk
22:46.04PatrickDKthey have to match
22:46.06jf_ok
22:47.04jf_maybe it's why
22:47.11PatrickDKwell, options are rfc2833, inband, and info
22:47.25PatrickDKI use rfc2833 always
22:47.30jf_ok
22:47.30PatrickDKgrandstream likes info
22:47.43jf_what do i use in xlite
22:47.55PatrickDKrfc2833, and tell xlite outofband
22:48.03PatrickDKatleast that is how I had mine setup
22:48.21shmaltzfinally my goverment is doing something usefull with the money I sent in yesterday to the Treasury department
22:48.23shmaltzhttp://news.yahoo.com/news?tmpl=story&e=3&u=/ap/20050414/ap_on_go_ca_st_pe/fugitive_roundup&sid=84439559
22:49.25*** join/#asterisk malcolmd (~malcolmd@malcolmd.digium.sponsor.pdpc)
22:49.35*** join/#asterisk ManxPower (~eric@stirprop-S0-0-0-26.ndcr2.datasync.net)
22:49.36harryvvshmaltz, and how many of those are illegal mexians entering our borders?
22:49.42harryvvmexicans ;)
22:50.00shmaltzwho knows, the article doesn't mention that
22:50.20shmaltzsince the DHS didn't take part in this, they might not know for another week or so
22:50.40harryvvI served in albuquerque nm while in the service and a collage girl was being followed by somone, she did a uturn and thay shot and killed her..the fugitives drove back into mexico.
22:50.54jf_Patrick: can u transfert
22:51.01PatrickDKya
22:51.17PatrickDKworst case, use inband, and use ulaw
22:51.19jf_what u suggest for dmtf
22:51.31harryvvTerrorist can easly slip into the states from mexico or canada. So many open holes.
22:51.33shmaltzthere was another story last weekend about some mexican that just crossed the border after killing an official in CA
22:51.42PatrickDKany other codec, you need rfc2833, and outofband
22:51.50shmaltzcanada is actualy better closed off then ppl think
22:51.52harryvvCalifornia?
22:51.52jf_k
22:52.07shmaltzthe only known hole is thru the inidian reservations
22:52.09shmaltzyep
22:52.13shmaltzcalifornia
22:52.40harryvvshmaltz, there are actually 50 open holes on the us/canada border people can drive into the states unimpeeded.
22:52.59shmaltzoh really, with out any border patrol?
22:53.03harryvvyup
22:53.11shmaltzlook at this one:
22:53.13shmaltzhttp://story.news.yahoo.com/news?tmpl=story&ncid=1212&e=1&u=/ap/20050414/ap_on_hi_te/comcast_internet_problems&sid=95573501
22:53.15newlor more if you've got a boat. :)
22:53.18shmaltzlooks like a DOS on their DNS
22:53.36harryvvyea
22:53.38harryvvmabey
22:53.42shmaltzwhats the official answer the that the DOT has for this?
22:53.49jf_patrick: im inband both side, does not work
22:54.00PatrickDKyou using ulaw right?
22:54.00harryvvI dont have that much confidence in voip if things like that happen.
22:54.19harryvvon a local network thats fine.
22:54.26jf_ya
22:54.29shmaltzharryvv, you are right about that, but if you really think about it this is what it comes down to
22:54.49shmaltzDOS on PSTN is easier than you think
22:54.56harryvvthe only way around it is route the calls though a router with two wan links
22:55.05harryvvwith two seperate providers
22:55.16harryvvtwo seperate backbones
22:55.28shmaltzyou could just take 2 VoIPs and make 50 simultaneous phone calls to a PSTN and that person is DOSed
22:55.44harryvvtrue
22:55.58jf_Patrick: should i use something else
22:56.00jf_alaw
22:56.07*** join/#asterisk Rez (lorez@lorez.staff.freenode)
22:56.08PatrickDKdunno, you sure it's using ulaw?
22:56.13jf_ya
22:56.15shmaltzusing what you are saying we actualy have better protection (if you use BGP you are even better protected) for VoIP than for PSTN
22:56.16PatrickDKit highlights ulaw when you make a call?
22:56.52harryvvI started to learn Border gateway protocol in cisco. thats a pretty high level protocol to learn.
22:57.16harryvvI think the major backbones based cisco routers use that right?
22:57.16newlout of 50 calls, only one is going to go through though.  The remaining 49 will receive BUSY, get diverted to voicemail, or perhaps a second call will ring on CW until it isn't answered and the possibly diverted to voicemail on no anwer. :)
22:57.43jf_can someone tell me what is that ARNING[12481]: chan_sip.c:694 retrans_pkt: Maximum retries exceeded on call 86B65EE2-AD38-11D9-9881-000D93C516B2@
22:57.45jf_....
22:57.56shmaltzso it comes down to when you have a DOS (most providers nowadays block spoofed IPs, so DDOS is realy not so common anymore) attack you might not have any connectivity, but the difference in price will make the decision for ppl
22:58.16shmaltzof course they *all* use BGP
22:58.19harryvvyea
22:58.21shmaltzread this:
22:58.25shmaltz~dos
22:58.27jbotfrom memory, dos is (Disk Operating System) This OS is what got it all started for PCs.  Denial of Service...
22:58.28bugbotdos is assigned nothing and reported nothing.
22:58.32shmaltz~ddos
22:58.33jboti guess ddos is Distributed Denial of Service, or http://grc.com/dos/grcdos.htm, or see slashdot.org
22:58.33bugbotddos is assigned nothing and reported nothing.
22:58.38CoolCat_what is the easy way of gateway skype and sip?
22:58.49shmaltzfollow that link, and read the one about drdos
22:59.08harryvvi already know what ddos is :)
22:59.11*** part/#asterisk Rez (lorez@lorez.staff.freenode)
22:59.18shmaltzthe DrDOS attack he had was exploiting the fact that most core routers use BGP
22:59.38shmaltzbut that link has something about DrDOS
22:59.52shmaltzDistributed Reflected Denial Of Service
23:00.01shmaltzthe reflected part is new
23:00.19shmaltzI think he was the only one that was a victim of this attack
23:00.33shmaltzI never heard anybody else that was a victim of this attack
23:04.53ManxPower*whine*  Nobody responded to my message
23:05.02marloweeh?
23:05.58harryvvwhat message
23:06.18shmaltzManxPower, looks like you posted the message to a different channel :/
23:06.27harryvvyea
23:06.35*** join/#asterisk bjohnson (~bjohnson@66.11.188.213)
23:06.36harryvvI looked back on this channel :)
23:08.17*** join/#asterisk Micc (~mic@c-24-18-35-120.hsd1.wa.comcast.net)
23:08.35Miccquick question.
23:08.42MiccDoes asterisk work with broadvoice?
23:08.59shmaltzMicc, why not
23:09.01MikeJ[Jayden]micc, yees
23:09.11shmaltzstupid article:
23:09.12shmaltzhttp://story.news.yahoo.com/news?tmpl=story&ncid=1212&e=10&u=/ap/20050414/ap_on_hi_te/voip_regulations&sid=95573501
23:09.21MikeJ[Jayden]there are some known qwirks with multiple broavoice accounts
23:09.41MiccOk, so I've built and installed asterisk on my linux box. Is there some docs on how to set up asterisk with broadvoice or other sip?
23:10.32shmaltznice move from bush:
23:10.33shmaltzhttp://story.news.yahoo.com/news?tmpl=story&ncid=1211&e=4&u=/nm/20050414/tc_nm/tech_government_amd_dc&sid=95573372
23:10.54shmaltzthis means asterisk can be installed in the white house ;)
23:11.46*** join/#asterisk PBXtech (~nik@67.107.241.3.ptr.us.xo.net)
23:11.47ManxPowershmaltz, no my mailing list message
23:11.57shmaltzoh,
23:12.02shmaltzwhats the question about?
23:12.14shmaltzsubject or otherwise
23:12.17*** join/#asterisk dmccollum (~dmccollum@69-164-245-72.atlaga.adelphia.net)
23:12.32PBXtechhow do you get over putting in the 'username' in the DIAL string for IAX2?   (AMP not likey)
23:12.39harryvvwow
23:12.53harryvvthats good new for amd
23:12.54harryvv;)
23:13.07harryvvAnd relief for smaller companies.
23:13.23shido6:)
23:13.56Tuplinkwhat is Congestion in extentions
23:14.05Tuplinkbussytone
23:14.12Tuplink?
23:14.51shido6yeah
23:14.57CoolCat_is there any free sip provider out there?
23:15.11shido6right
23:15.12pgpkeyswell free for the number but you still pay for calls.
23:15.23reallost1How do I unset a variable?
23:15.25pgpkeysnufone charges 0.02 a minute in the US for US48 and CAn calls
23:15.40shido6same for inbound 8xx
23:15.42shmaltzharryvv, yep
23:15.49pgpkeysright
23:15.52shmaltzManxPower, what is the subject line?
23:15.55pgpkeys0.02 regardless of in or out
23:16.02pgpkeysyou make it, they make it.. 0.02/m
23:16.10pgpkeys0.08 to alaska and hawaii
23:16.13TuplinkFWD has free 1800
23:16.24harryvvWhats the smallest motherboard that can take two pci slots and a embeded linux flash card?
23:16.48pgpkeysnufone gives a free toll free. 800, 866, etc etc. though their site is now saying that they have no numbers left. must have used up their allocation.
23:17.20reallost1ipkall has free incoming numbers.
23:17.25ManxPower[Asterisk-Users] Call Parking timming out to the wrong extension
23:18.05harryvvI want to get * off my work station and into something small.
23:18.26*** join/#asterisk fugitivo (~ajf@201.255.103.99)
23:19.55CoolCat_this sound free, but im not sure, some terms talk about charges! =o/ http://www.earthlink.net/membercenter/benefits/onlinecalling/
23:20.59harryvvyea..its a highly compeditive biz
23:22.50shmaltzManxPower, what app you using for parking?
23:23.19ManxPowershmaltz, you don't have to use an app.
23:23.33shmaltzI know
23:23.37shmaltzbut thats what I'm asking
23:23.40ManxPowerBTW, any telco geek out there.  Amphenol connector's gender is confusing.  Can anyone help?
23:23.43shmaltzyou just using xfer?
23:23.46CoolCat_fwd sounds free!
23:23.46ManxPowershmaltz, features.conf
23:23.50shmaltzok
23:24.05ManxPoweryes, transfer to 3515 which is defined in features.conf.
23:24.15shmaltzAmphenol that has the 3 mm strip sticking out and 25 lines of copper on each side is male
23:24.18MikeJ[Jayden]parking:http://www.pbxclue.com/asterisk_apps/
23:24.20ManxPowerSee http://bugs.digium.com/bug_view_page.php?bug_id=0004036
23:24.34ManxPowershmaltz, You are SURE?
23:24.50MikeJ[Jayden]stivking out is male, going in is female :)
23:25.10*** join/#asterisk darwin35 (~darwin35@24.3.226.147)
23:25.29shmaltzyep
23:25.38shmaltzusualy the 66 block comes with the male one
23:25.52harryvvManx, I have worked with amp connectors what are you working with
23:25.53shmaltzand the equipment with the female connector
23:26.27harryvvOhh telco mmm  amp has diagrams and specs on there connectors.
23:26.50ManxPowershmaltz, Do I need AMP or Avaya style?
23:26.50tzangerAmphenol male has the connectors in a row in the middle
23:26.52tzangerAmphenol female has the connectors around the inside with nothing in the middle
23:26.56ManxPowerSo I need a female connector to plug into the Adtran Channel Bank
23:27.14harryvvmanx whats the part number
23:27.25shmaltzI don't know about the Adrans but Adit needs a male
23:28.12ManxPowerThe adtran sticks out
23:28.44PBXtechhow do you get over putting in the 'username' in the DIAL string for IAX2?   (AMP not likey)
23:29.08harryvvManx, generally what is that connector called?
23:30.10ManxPowerharryvv, Amphenol
23:30.23ManxPowergotta run
23:30.25harryvvI know is the infinity style of connector
23:31.09*** join/#asterisk Sedorox (brandon@Neptune.client.wlmsprt.pa.sed6.net)
23:33.18harryvvAmphenol T-1 for the Total Access 750 Chassis T1 Channel bank
23:33.22harryvvMy guess
23:35.08*** join/#asterisk jf_ (~jeanfranc@modemcable077.187-80-70.mc.videotron.ca)
23:35.24harryvvtzanger, was he working with the 50 pin connector
23:35.55jf_someone can tell me why each time i reboot my linux kernel 2.6 i have to rerun make install on zaptel and then modprobe zaptel wcfxo otherwise it does not load module
23:36.00*** join/#asterisk Gunnar (~gunnar@34.80-203-209.nextgentel.com)
23:36.16tzangeryou asked for the amp connector on a channel bank
23:36.58shido6thats sounds terrible, jf
23:37.19shido6make config makes the init scripts you need
23:43.26*** join/#asterisk hermie (~nick@24.236.167.53.bay.mi.chartermi.net)
23:45.06harryvvBasicly COs mostly use channel banks for home users.
23:45.15harryvvAm I correct on this
23:49.18*** join/#asterisk logarno (~logarno@80.125.208.234)
23:49.44*** join/#asterisk JunK-Y (~junky@modemcable174.107-81-70.mc.videotron.ca)
23:50.13elrichow would you tell a macro to wait until the call has actually been answered by the party being called?
23:51.55jf_shido6: what do u suggest
23:52.01*** join/#asterisk shmaltz (~chatzilla@69.28.255.210)
23:53.05shmaltzcan I plug in an rj45 into a jack that is made for an rj48?
23:53.14Tuplinkis their some program to make the womans voice say what you want?
23:53.29tzangerI never knew the difference between RJ45 and RJ48 aside from that 'tit' out the side
23:53.46harryvv:)
23:54.49PatrickDKtzanger, you mean DEC style
23:54.51harryvvtzanger, Manx is setting up a channel bank for a local business?
23:55.03Tuplinkhow do i make the asterisk womans voice say what i want?
23:55.12tzangerharryvv: don't know what he's doing
23:55.23elricis there a variable that tells you if the number you were calling has answered the phone?
23:55.45tzangerelric: you mean after the Dial()'s complete?
23:56.38elrictzanger, i am doing Dial(${EXTEN}|60|M(macro))
23:56.47harryvvokay. I can understand a CO using channel banks to convert there high speed digital network into analog and push that to the residential homes in the area but what is the purpous in the astrisk realm? Why would a bussiness want to use analog lines instead of digital for its phones? or would it be used for fax?
23:56.53elricand the macro executes as soon as the number is dialed
23:56.56tzangerelric: add a g option and afterward look at HANGUPCAUSE and DIALSTATUS
23:57.01*** join/#asterisk Legend (~Legend@24.244.142.134)
23:57.09elrici want it to wait till the call is answered
23:57.18tzangerharryvv: I use channel banks
23:57.18elricthanks tzanger
23:57.28harryvvfor what application
23:57.35tzangerhigh quality hybrids, "high" port density
23:57.39Tuplinkhow do people make all of theyr menu sounds?
23:57.42tzangera T100P+CB can handle 24 channels with only one zap card
23:57.51tzangerTuplink: I use Record()
23:57.51harryvvbut only for anlog phones or say fax machines
23:57.56tzangerharryvv: depends
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23:58.20harryvvcannot work with digital phones that would normally use a pbx then.
23:58.26tzangerharryvv: sure it can
23:58.28Tuplinkwill Record make it that womans voice
23:58.37harryvvokay
23:58.37bkw_haha
23:58.38tzangerTuplink: no that is alison
23:58.55Tuplinkalison how do i get her to say what i want?
23:58.58MichaelCatDoes anyone want to help me try to fix caller ID inbound to my X100P clone
23:58.59tzangerharryvv: before I did a PRI connection to the Norstar MICS I did it with analog trunks and FXS CB modules
23:59.11harryvvokay
23:59.23bkw_Tuplink, you pay her.. thevoice.digium.com
23:59.33bkw_beware.. she will not say the word "cunt"
23:59.38tzangerbkw_: hahaha
23:59.38harryvvso say then the pbx was off site you would have the digital phones plug into the channel bank route the t-1 digital traffic to the remore asterisk system then.
23:59.46harryvvone application?
23:59.48tzangerI still think you get her to say "call hunt group" and edit it

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