irclog2html for #asterisk on 20060420

00:00.03tehdelywhere i can guarantee control will go ---> when they finish
00:00.15tehdelyjust the exit, Dial(), agi in the dialplan?
00:01.16*** join/#asterisk Abydos313 (i=abydos31@adsl-71-129-60-235.dsl.irvnca.pacbell.net)
00:01.23Hmmhesaysanyone ever set more than one variable with a callfile?
00:01.35tehdelyor to put more cleanly: exit from the agi, have subsequent priority in dialplan do the dialing, next priority reexec the agi
00:01.38anthmfor scale and reliable factor doing calls from the regular dialplan is most safe
00:02.07tehdelyok i will keep that in mind
00:02.10anthmwhat does yor agi have to do
00:02.25tehdelyanthm: the basics of call processing; finding a person and ringing their phone, placing an outbound call , etc.
00:02.41tehdelyi'm building it in ruby w/ active record as the state, so i can easily build a web app to manage it
00:03.03Mikesomeone has problems with unicall reseting channels once in a while droping al active calls?
00:03.13tehdelywhat is your suggested way of saving state
00:03.29tehdelyif i SetVar a bunch of things and exit do they persist?
00:04.14*** join/#asterisk JunK-Y (n=junky@modemcable205.175-81-70.mc.videotron.ca)
00:04.39anthmya
00:04.43anthmthat is 1 way
00:04.51anthmprobably the easiest
00:05.00anthmyou can set vars into the CDR record of the channel to
00:05.17anthmone of my many additions *bow*
00:05.22tehdelyjar
00:05.24tehdely*har
00:05.26tehdely:)
00:05.32tehdelydoes this require using mysql backend for cdr
00:06.09tehdelyor is cdr always available/retrievable
00:06.18anthmcdr_custom
00:06.35anthmyou can define all the fields in the csv file to your own
00:11.49tehdelyit is funny, the code flow i am visualizing now
00:11.55tehdelywoudl be perfect use case for continuations
00:12.02tehdelybut ruby can not serialize a continuation between runs :|
00:12.41tehdelygah, there is no way for agi to suspend is there?  only to exit
00:12.52Hmmhesaysno way you want to use
00:13.06Hmmhesaysmost languages have a sleep();
00:13.23tehdelywon't do me any good if i need to jump back into the dialplan :/
00:13.40Hmmhesaysnope
00:13.55Hmmhesaysi suppose you could set some channel variables and access them again if you have to jump back into agi
00:14.03tehdelyHmmhesays: yeah that is best option proposed so far
00:14.09tehdelywell, cdr-custom maybe
00:14.19*** join/#asterisk bobman (n=bobman@cpe-72-224-209-140.maine.res.rr.com)
00:14.27tehdelybut lets back up for a sec... other good ways of dialing aside from returning to dialplan.  manager interface?  callfile?
00:14.31tehdelyany of these ok or do they have the same problem
00:14.51tehdelythis question is @ anthm
00:14.52Hmmhesayscallfiles are cool, they take care of retrying on a failed dial
00:14.57tehdelyok
00:15.04anthmnot really
00:15.08Hmmhesaysif that interests you
00:15.24*** join/#asterisk angom_h (n=angom@red-corp-200.38.17.87.telnor.net)
00:15.29anthmnot if you want to wait for it to be over
00:15.35anthmand regain control
00:16.01anthmyou can do it in your agi you just may have less luck when you have thousands of it at once
00:16.16anthmand you will probably find some zombies of your agi
00:16.35tehdelyi see
00:16.50Hmmhesayswhat exactly are you tryign to do?
00:17.10anthmyou could look at my res_js addon and make javascript ivrs =D
00:17.12tehdelyheh best way i could explain is i am reimplementing priorities in ruby
00:17.19tehdelyso that a web app can easily manage ring groups, etc.
00:17.22tehdely"ring me at office"
00:17.24tehdely"ring me at home"
00:17.24anthmwhen you exec from that it's native
00:17.38tehdelyso it needs to bea ble to place the call, wait for exit, try the next number if the first didn't answer
00:17.55tehdelyif the answerer on a Dial()ed call hangs up is the AGI supposed to die
00:18.14anthmyou can also use the script to generate a static dialplan that is the safest way
00:18.22Z_Godis it possible somehow with SIP to configure your server so anyone can call you at yourname@asteriskserver, or does SIP require you to specificly 'peer' with each server?
00:18.42*** join/#asterisk Lino` (n=Lino@i577BDDC0.versanet.de)
00:18.58Z_GodI've now configured it so it works with SIPPhone, but I though SIP was similar to Jabber, when it comes to server2server
00:19.08Z_Godit->asterisk
00:19.09Hmmhesaysyou would need to have control of the  "asteriskserver" domain
00:19.11tehdelyanthm: how to exec the generated dialplan
00:19.22Z_Godyes, that was just an example :)
00:19.24tehdelymanager interface?
00:19.37Hmmhesayssystem call
00:20.19anthmyou can use your program in a cron to write regular extension callflow to disk
00:20.23Z_Godis there any guide on setting up asterisk as such an SIP server?
00:20.32anthmwhich can look ugly and hard to read
00:20.44anthmcos it's only for the interpreter to follow
00:20.59Z_GodI've only seen guides for setting asterisk up as an sip client or to allow clients to connect to it
00:21.11tehdelyyikes!
00:21.13tehdelyheh
00:21.13*** join/#asterisk oej (n=oej@h2.ast.sipit.net)
00:21.21rhoweIs it not possible for * to read a dialplan from a database or something?
00:21.25Z_Godnot really one that allows asterisk to really 'federate' with others
00:21.25HmmhesaysZ_God: i dunno, set up a dialplan in your default context
00:21.49anthmyah you can i made that too it's called res_config
00:21.58luke-jr_How can I determine why Asterisk isn't doing a native transfer of a call?
00:22.08Z_Godk, I'll try that
00:22.08rhoweanthm: Does it just read it once, at startup, or does it query for every call?
00:22.44anthmboth if you count the realtime one
00:22.59anthmnone of it scales tho
00:23.30luke-jr_Z_God: I think you want to enable guest SIP calls
00:23.57Z_Godk
00:24.16rhoweanthm: hm.. I would've thought you could do something like this on an incoming call .. SELECT * FROM dialplan WHERE context = 'foo' AND extension = 's' and then proceed using the results
00:24.26anthmyes
00:24.36*** join/#asterisk jsaunders (i=jsaunder@S01060060971c5817.va.shawcable.net)
00:24.36Z_Godis that an option in sip.conf?
00:24.39rhoweanthm: Admittedly, for thousands of calls, you'd be talking thousands of queries, but databases can take that, can't they?
00:24.52NuggetSounds fun, execept "select *" is a bad habit.  :)
00:25.00anthmyou can use realtime switch for that
00:25.17anthmyou can use res_odbc to suck the whole dialplan out of the database into memory
00:25.18Nuggetand you forgot to order the results, which could yield unpredictable behavior
00:25.32anthmor you can use res_sqlite to make a cached exten from sqlite
00:25.39jsaundersWould I have any audio problems running asterisk (sip) and gnugk (h323) on the same box with default rtp settings?  I know signalling is on differeing ports but what would happen if the rtp ranges overlapped between the two.   Is it possible that asterisk would pick up an rtp chan destined for gnugk, and vice versa?
00:25.53alephcomIs it possible to amplify a sip or IAX channel?  I've been unable to find a way but maybe I'm just unlucky.
00:26.02Z_Godhmm allowguest seems to be the default
00:26.24rhoweanthm: Hm, interesting
00:26.51*** join/#asterisk shido6 (n=shido6@d38-45-81.commercial1.cgocable.net)
00:26.58rhoweWouldn't that be far easier than reimplementing the whole dialplan logic in ruby?
00:27.10HmmhesaysZ_God you an use names in your dialplan
00:27.26Hmmhesaysexten => Z_God,1,Dial(SIP/Z_God)
00:28.19*** join/#asterisk nitam (n=none@201.138.73.214)
00:28.22anthmof all the ways to do it i still pick generate the regular exten file to disk for maxium reliability
00:28.41anthmput it in your db any way you want and then generate a giant dialplan
00:28.49tehdelywhat is your favorite method of getting asterisk to hash the file
00:28.51tehdelyand exec it
00:28.54tehdelyafter your agi has created it
00:29.01Hmmhesayswon't you constantly be reloading the extensions file though?
00:29.19nitamhi
00:29.24anthmthat is the advanced version
00:29.27anthm=D
00:29.52rhowehm, can asterisk use dnotify to automatically reload changed config. files?
00:30.01*** join/#asterisk Libila (n=vye@ip68-6-130-118.sd.sd.cox.net)
00:30.02DrukenHmmhesays: wouldn't that be easier to use the database dialplan crap with ?
00:30.19Hmmhesayswas that actually directed at me or did you tab complete that
00:30.34Z_Godcool stuff, I've got it to work :)
00:30.36Z_Godthanks a lot
00:30.36Drukendirected to you
00:30.39nitamdoes anybody know if there is a way to intercept a call ? i mean ... hear conversations btw 2 or more peoples ?
00:30.48Hmmhesaysanthm has an answer for that
00:30.48Drukeninstead of reloading the dialplan contantly
00:30.50Z_Godindeed just needed to fix my extensions
00:31.17anthmdatabase will will be fine unless you plan to have thousands of users
00:31.28Hmmhesaysyou need to use chanspy if you want to hear your wife talking to her boyfriend
00:31.51Drukenwtf....
00:32.09JunK-YHmmhesays: shit, u gonna hear me talking with ur gf :P
00:32.09HmmhesaysJunK-Y!
00:32.10JunK-Ymatttt!
00:32.13JunK-Ywhats up dude?
00:32.23Hmmhesaysnot much, just hanging out in the chan for awhile
00:32.28JunK-Yyay
00:32.30Hmmhesaysplaying a little guitar
00:32.35anthmchanspy, another of my inventions i almost forgot how much asterisk stuff i made
00:32.35JunK-Ycluecon, this summer?
00:32.38nitamHmmhesays: lol ... its not that, im administraiting an asterisk with broadvoice, zaptel and a panasonic pbx, so i need to know everything possible
00:33.01HmmhesaysJunK-Y yeah I think so
00:33.01nitameverything that i can do
00:33.17JunK-YHmmhesays: yay, strip club again?
00:33.22rhoweanthm: Know how to beat overfunctional PBXes into submission? :)
00:33.29JunK-Ywhat was the name of that club, again?
00:33.32Hmmhesaysits not in that suburb again is it?
00:33.40JunK-Yno, downtown.
00:33.53Hmmhesaysthat one place in the middle of nowhere
00:33.54Hmmhesaysi don't remember
00:34.04rhoweanthm: I've linked asterisk up to the PBX at work via a BRI interface and an AVM Fritz card. It works fine - can call SIP addresses and the calls get put out onto the PBX
00:34.09JunK-Yya, the place, where we had to drink water :(
00:34.28JunK-Ythe place where i spent A LOT of 1$ bill ! :P
00:34.32Hmmhesayslol
00:34.34Hmmhesaysyeah
00:34.35rhoweanthm: However, when calling asterisk from the PBX, I can't persuade it to send DTMF.. when I hit keys, it thinks I'm entering an extension to transfer the call to
00:34.57rhoweanthm: Something tells me I'm going to have to define the BRI interface as a trunk line rather than an extension...
00:35.26anthmasterisk thinks that ?
00:35.33rhowethe PBX thinks that
00:35.50DelmarDoes the message waiting indication for messages in mailbox 1000 in context [blah] correspond to the context the SIP phone is in? ie. if the SIP seting is context=siphones and the voicemail context for that phones extenstion is [blah] the MWI wont be sent? how does it know?
00:36.01rhoweSo asterisk never hears/sees that I've been hitting the telephone keypad. The PBX has intercepted it, taking it as an extension to transfer the call to
00:36.11*** join/#asterisk fjean (n=fjean@201009190027.user.veloxzone.com.br)
00:36.31rhoweIf you call extension x, then dial extension y without hanging up, when you hang up, the call will transfer to extension y
00:36.54nitamHmmhesays: i dont have that cmd, do you know if is independent from asterisk as well ?
00:37.12Hmmhesayschanspy has been in there forever
00:37.17fjeanhello, anybody has already connected a utstarcom with asterisk ?
00:37.38Hmmhesayscli>show application chanspy
00:37.41nitamHmmhesays: you mean from asterisk console, right ?
00:37.47rhoweIn order to do that, the PBX must be intercepting the keypresses and not sending them out as DTMF, but it doesn't do that for external calls.. I'm guessing that it doesn't do it if it goes out over a trunk line, but I might need to sweet-talk the PBX maintenance company into telling me how it all fits together
00:38.58rhoweAll the asterisk stuff is working great.. no problems with that at all
00:39.13rhoweOnce I figured out how to get chan-capi to pick up incoming calls, that is
00:41.02Drukenisn't there a way to make monitor merge the two streams?
00:41.03*** join/#asterisk lokkju (n=lokkju@unaffiliated/lokkju)
00:41.05*** part/#asterisk fjean (n=fjean@201009190027.user.veloxzone.com.br)
00:41.11nitamHmmhesays: yeah, you are right, i get a few lines about Chanspy when i do show application chanspy, but chanspy doesn't work ... I get No such command 'chanspy' blah blah
00:41.21Drukenthat whole in and out is a pain in the ass
00:41.35marlok, i know this is someware in the docs but i cant find it :( how do u set an extension to run an AGI script when configuring AAH from AMP?
00:42.00Hmmhesaysi hope you're not trying to call it from the console
00:42.36*** part/#asterisk Z_God (n=Z_God@jabber.xs4all.nl)
00:43.22*** join/#asterisk mrbnet (n=sureal@CPE-72-135-45-43.mn.res.rr.com)
00:44.35luke-jr_How can I determine why Asterisk isn't doing a native transfer of a call?
00:44.46tehdelyanthm: do you want to know how i have decided to solve this
00:44.49tehdelyi am in for much hair pulling
00:44.59tehdelyruby script will save its continuation, initiate call through manager interface
00:45.04anthmsure
00:45.06tehdelyloop, and wait for event taht says it hung up or failed
00:45.12tehdelyand then call the continuation and process from there, with status info
00:45.14tehdelyheh
00:47.16*** join/#asterisk ManxPower (n=ewieling@dpc67142183150.direcpc.com)
00:48.21*** join/#asterisk Libila (n=vye@ip68-6-130-118.sd.sd.cox.net)
00:51.55*** join/#asterisk RoyK (n=roy@28.80-203-106.nextgentel.com)
00:52.39anthmgood luck
00:52.49anthmit's got a chance for a while
00:53.00anthmyou may get some races in heavy situations but
00:53.09anthmthere is always C ;)
00:53.14tehdelytrue ;)
00:53.28tehdelyi'll probably get away with this loony shit just because each of these instances only lasts the length of a call
00:53.38tehdelyso the likelihood of crap piling up is kept to a minimum
00:54.21Ariel_anyone know if there working on asterisk sip account issue with more outbound routes to same IP address with different user name/logins?
00:54.30*** join/#asterisk redondos (n=redondos@190.48.54.232)
00:54.50Ariel_asterisk alway uses the last one in the sip.conf with the same IP address.
00:54.56*** join/#asterisk zotz (n=zotz@24.231.32.85)
00:57.16Hmmhesayswhy do you have same sip accounts with different ip's?
00:57.56luke-jr_Hmmhesays: ... same IP, different accounts
00:58.10Ariel_no same IP different acccounts
00:58.13Hmmhesayswow i read that wrong
00:58.28*** part/#asterisk Libila (n=vye@ip68-6-130-118.sd.sd.cox.net)
00:58.31*** join/#asterisk CletusColeman (n=CletusCo@adsl-69-155-216-32.dsl.rcsntx.swbell.net)
00:58.45Ariel_different users having accounts with same provider
00:58.55Hmmhesayswhere is the problem?
00:59.09Ariel_I am able to get calls in fine but outbound it uses only the last one in the sip.conf that matches ip
00:59.30Ariel_does not matter that I use different names
00:59.43Hmmhesayshmm
00:59.53kink0I success loading wct4xxp , but zttest does nothing, can somebody help me to find what is wrong ?
01:00.09Ariel_or even do dial(sip/username:password/${EXTEN}@IPADDRESS)
01:00.43SkramXHi.
01:00.45kink0still stoped at: Opened pseudo zap interface, measuring accuracy...
01:00.55lokkjuah
01:01.00lokkjukink0, what distro?
01:01.09lokkjukink0, and kernel
01:01.20Ariel_Hmmhesays, in fact for inbound I had to add to them insecure=very to get it to work for inbound correctly.  argh
01:01.22*** join/#asterisk yxa (n=diablo@58.185.90.101)
01:01.39kink0lokkju, slack 2.6.14.5 , was running util I reset the machine, but now always zap show stat reports RED alarm on zap
01:01.40Ariel_iax2 works fine for multiple accounts
01:01.44SkramXI have an inbound DID (toll free) and when users use it and get in the conference, they sound very low (in volume).. any ideas?
01:01.53lokkjuand is your zaptel.conf correct?
01:01.57luke-jr_does a transfer/reinvite need to occur before the call is answered?
01:02.13kink0lokkju, yes, is the same that was before I reset, even I compare and restored from a backup
01:02.14SkramXlokkju: its voip not a zap card.
01:02.16yxasomeone is asking me what version of SIP is * compiliant?
01:02.20SkramXoh
01:02.23ManxPowerluke-jr_, I believe it happens AFTER the call is answered
01:02.58luke-jr_ManxPower: so if I Dial(...,,A(beep)), it should work?
01:03.13luke-jr_(A plays the sound to the called party before bridging)
01:04.32*** join/#asterisk Libila (n=vye@ip68-6-130-118.sd.sd.cox.net)
01:04.36yxarather which RFC does sIP on * conform to?
01:04.57luke-jr_yxa: is there multiple RFCs? I'd assume the current one
01:05.33yxaluke-jr_ there are. 2543 and 3261
01:05.41file3261
01:07.01LibilaI'm having some trouble with a cannot create channel error when trying to make outbound calls on my TDM04P. SIP works fine, I can call everyones extensions. here is the error output when trying to make an outbound call:
01:07.16ManxPowerluke-jr_, you can't do reinvites if asterisk has to insert or listen for audio
01:07.25*** join/#asterisk trbldwine (n=trbldwin@71.194.161.170)
01:07.36luke-jr_ManxPower: not prefixing audio?
01:07.45Libilasorry, wrong clipboard. One second.
01:07.49ManxPowerLibila, you installed asterisk before installing zaptel.  you need to reinstall asterisk
01:08.01DelmarLibila, if there is more than a few lines.. use pastebin
01:08.37LibilaDelmar: I understand. I'm not used to the X clipboard. This is what I wanted to paste: http://rafb.net/paste/results/kbD9fb21.html
01:08.37DelmarLibila, do ..  zap show channels
01:09.32ManxPowerLibila, that error message confirms what I told you.
01:09.47Libilaok, I will reinstall.
01:10.31luke-jr_ManxPower: it can't play audio prior to the transfer/reinvite?
01:10.36DelmarManxPower, sure.. and it couldnt possibly be anything do to with the zaptel.conf or zapata.conf setup
01:11.14ManxPowerluke-jr_, I doubt it.
01:11.24ManxPowerluke-jr_, a TRANSFER is not a REINVITE
01:12.01ManxPowerDelmar, If it was it should be a can't create channel of Zap, not that there is no such channel type as Zap
01:13.35luke-jr_ManxPower: IAX2's 'transfer=yes' isn't the same as a reinvite?
01:13.50ManxPowerluke-jr_, no.
01:14.24luke-jr_so how would I get a reinvite with IAX2?
01:14.45ManxPowerall a reinvite does is send the audio between the two endpoints, bypassing the server, but signalling still stays on the server.
01:15.17ManxPowerwith iax2 transfer both audio AND signaling go direct between the two end points and the middle server is totally out of the loop, and so CDRs are not correct.
01:15.24russellbluke-jr_: you can't do it currently, though it is something we will be adding sometime this year, probably
01:15.33ManxPowerluke-jr_, a reinvite is an RTP thing.  IAX2 does not use RTP.
01:15.45*** join/#asterisk bjohnson (n=bjohnson@i216-58-92-2.cybersurf.com)
01:15.47luke-jr_ok... so I need to get my stuff to use SIP for now I guess?
01:15.56ManxPowerluke-jr_, what are you trying to do?
01:16.17luke-jr_ManxPower: simply forward an 866 number to a regular POTS number
01:16.28luke-jr_but play a beep to the called party prior to connecting
01:16.28ManxPowerso what's the problem?
01:16.36luke-jr_and keep CDRs
01:16.46ManxPowerluke-jr_, not gonna happen with IAX2
01:16.48luke-jr_well, I don't want the audio to go through my server
01:16.58*** join/#asterisk Libila (n=vye@ip68-6-130-118.sd.sd.cox.net)
01:17.27ManxPowerI do not know if A() will prevent reinvites or not with SIP.  I'll bet it does.
01:17.53luke-jr_why? :\
01:18.21luke-jr_if reinvites occur after there's audio anyway, I'd expect it didn't matter if both ends are at that stage
01:18.37luke-jr_or if the called end is instead of the callee
01:18.59ManxPowerluke-jr_, because some parts of Asterisk's reinvite support sucks.
01:19.00*** join/#asterisk oej (n=oej@203.178.156.27)
01:19.21*** join/#asterisk Utah_Dave (n=boucha@c-67-166-71-219.hsd1.ut.comcast.net)
01:19.29ManxPowerbut since oej is here, ask him, he wrote chan_zap
01:19.32ManxPower..er... chan_sip
01:20.28Ariel_ManxPower, how are you doing tonight?
01:21.25ManxPowerAriel_, still alive, so that's good.
01:21.38ManxPowertornado touched down about 10 miles from here.
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01:27.33russellbno, A() should not cause any problems.  the announcement wouldbe played before the two channels are bridged
01:27.42russellband the reinvite does not occur until that time
01:29.22ManxPower*sigh*  The roof is still leaking.
01:29.57yxawhen i bridge 2 zap analog channels, users complain the volume is soft. how do i mitigate that
01:30.40ManxPoweryxa, TELL THEM TO TALK LOUDER!!!!
01:31.10ManxPoweryxa, you prolly need to increase either txgain or rxgain.
01:31.19ManxPowerbe careful of echo if you do that
01:33.28mmlj4hey ManxPower
01:34.22ManxPowerhello, mmlj4
01:34.23mmlj4from craigslist: "i don't know anything about VoIP, i can't even make asterisk work on my box here at home, but I want to start a company, please do it for me"  # go find someone else, please
01:34.42tehdelyoof, i'm an idiot
01:34.43mmlj4then the fool found out I (used to) do PHP, so he decided he wanted me to help manage indian PHP coders for him
01:34.49tehdelyi can't place a call with originate and hook it up to an existing channel
01:34.50tehdelycan i
01:35.06ManxPowertehdely, send both channels to a meetme
01:35.31tehdelyfdasf
01:35.37tehdelyis that the only thing i can do?
01:35.40kink0a last question... is needle to mount devfs in any mount point ? or is /dev enough ? ( I means to use TE4xx cards from digium and kernel 2.6.x )
01:35.44ManxPowertehdely, there may be other ways.  I don't know.
01:36.22justinu|laptoptehdaly: there's an "app_bridge" in mantis, not sure if it works
01:41.10anthmi have one but it needs a little tweak
01:41.13anthmhttp://www.freeswitch.org/asterisk_stuff/app_forwardcall.c
01:41.17*** join/#asterisk lunaphyte (n=lunaphyt@pool-71-120-131-176.gdrpmi.dsl-w.verizon.net)
01:41.31anthmbasic bridge can do multithreaded bridge
01:41.40anthmbut zap channels will make the call end on dtmf
01:41.56*** join/#asterisk Mw3 (i=mw3@national.t-error.hu)
01:41.59anthmit needs to make exception on certian softhangup
01:42.07anthmcos some of them dont really mean hangup
01:42.08anthmlol
01:43.06luke-jr_aha, L() causes problems w/ reinviting
01:44.23*** join/#asterisk WeeZyyy (n=liquidni@247-50.35-65.tampabay.res.rr.com)
01:44.41luke-jr_now here's the curious part: the callee hears everything as they should, and the called hears the A() sound, but once the call is reinvited, the called *cannot* hear the callee
01:46.05Snake-EyesWhen ever I put g729 in allow for trunk the other two codecs stop working and I get the frame type error for them, but g729 works fine. I've cleared general of codec info to be on safe side. If ulaw and alaw are the only ones allowed they work fine. Any ideas how I can all three codecs working in sip.conf?
01:49.45tehdelyhmm interesting idea
01:49.58tehdelywhat if i had it use app_conference, which is just a basic bridge (but with multiple participants)
01:50.06tehdelyand make that a feature of the app
01:50.09tehdelyeasily drop in on any call
01:50.14tehdelyturn any 2-way into a 5-way
01:50.14tehdelyheh
01:52.24*** part/#asterisk mrbnet (n=sureal@CPE-72-135-45-43.mn.res.rr.com)
01:53.01theorem_simple one on one sex insto 5 way gang bang !
01:53.01theorem_wooo
01:53.43luke-jr_any easy way to get the incoming channel's hostname/IP?
01:54.29theorem_it should be in the logs too, right ?
01:56.52luke-jr_...
01:56.59luke-jr_in the dialplan
01:57.07luke-jr_I want to forward the call via the same IP it came in on
01:57.19*** join/#asterisk pauldy (n=pauldy@24-155-86-154.ip.grandenetworks.net)
01:59.13theorem_does anyone know the difference between the :  Linksys PAP2-NA  and the Sipura SPA-2002 ?
02:03.57ManxPowertheorem_, as far as I know "nothing important".  Cisco licensed the PAP2 design from SIPura.
02:04.12*** join/#asterisk marv (n=marv@12-219-145-181.client.mchsi.com)
02:05.19luke-jr_Why would the called party be unable to hear the callee in a reinvited call? (everything else works fine)
02:05.46Drukencisco owns sipura
02:07.35rowterIf I buy asterisk business I can't be able to compile unicall, steve's module?
02:07.53rowtercoppice?
02:09.41ManxPowerluke-jr_, NAT
02:10.23*** join/#asterisk tengulre11 (n=tengulre@221.11.5.180)
02:11.23*** join/#asterisk hinckc (n=hinckc@ool-4354c627.dyn.optonline.net)
02:12.06*** join/#asterisk MacDome (n=eseidel@c-71-198-177-144.hsd1.ca.comcast.net)
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02:13.38*** join/#asterisk bjohnson (n=bjohnson@i216-58-92-2.cybersurf.com)
02:16.44*** part/#asterisk jsaunders (i=jsaunder@S01060060971c5817.va.shawcable.net)
02:18.35ManxPowerluke-jr_, README.variables
02:20.55*** join/#asterisk oej (n=oej@h2.ast.sipit.net)
02:21.31tehdelyhmm
02:21.40tehdelyapp_conference loading makes asterisk exit with code 024
02:21.52tehdelythat is octal
02:22.07tehdelyit does not even get into the module's load functio it looks like
02:22.15tehdelyperhaps doens't like something about the binary?
02:24.21linageeis it possible to make a simple asterisk app that pulls a value from a database and speaks it using festival or something?
02:24.41theorem_one would think perl
02:24.42theorem_yes
02:25.09linageetheorem_: is there already a code sample for doing something like this? (i wouldn't think it's a new idea)
02:25.11theorem_the DBI module and modules for whatever dbase you're using should do the trick
02:25.37theorem_linagee - there may be, but it sounds fairly trivial if you've got the dbase all setup
02:26.01theorem_the dbase is merely a repository to store text, much like any file would be.
02:27.00linageetheorem_: i want to web scrape this page where we use this internal app at work, throw the values that are relavent to me into a DB, and have it accessible from asterisk. :)
02:27.20theorem_ha !
02:27.20FuriousGeorgei wonder how much it would cost to have allison record some prompts for me.  could i the re-release under the BSD licence?
02:27.21linageetheorem_: so i can just call on my cell phone. :)
02:27.25theorem_indeed, nice
02:27.28theorem_but ..
02:27.33theorem_why scrape the page ?
02:27.36linageetheorem_: well, i don't even have the web scraping down yet
02:27.41theorem_where does the page get it's data ?
02:27.46linageetheorem_: silly work does not provide any sort of XML interface or anything!
02:27.56theorem_join the clu
02:28.01theorem_*club
02:28.09luke-jr_ManxPower: I don't think there's any NAT-- I'm doing a reinvite to connect the same host with itself
02:28.12linageetheorem_: and although they use microsoft SQL, you can't log in directly.
02:28.19linageetheorem_: you have to go through their app
02:28.24theorem_... can;t log in ... directly ?
02:28.32theorem_I don't think I follow.
02:28.35theorem_hardly
02:28.37linageetheorem_: there's like half a dozen people in all of this large company that can log in directly.
02:28.40theorem_if you can access it remotely
02:28.49theorem_you can log in directly
02:28.51linageetheorem_: through a web interface
02:28.55Nuggetin my head, that sounds like Captain Kirk saying it.
02:29.08linageetheorem_: can't just tie right into the microsoft SQL server. hah. that would be too easy! lol
02:29.11theorem_linagee - all you need if the user which the web interface uses
02:29.14theorem_then use that to login
02:29.15theorem_easy :)
02:29.34theorem_unless they're the same box ?
02:29.47linageetheorem_: they use ASP for scripting and some sort of weird session thing. that's why it's hard to scrape
02:29.51theorem_in which case that use may be restricted to localhost only
02:30.02linageetheorem_: i think they are. lol
02:30.11theorem_linagee - what about snagging the mysql login/pass that the asp uses to login ?
02:30.30linageetheorem_: any company that does not provide databases to it's employees through an XML portal is just dumb
02:30.37*** join/#asterisk Mw3 (i=mw3@national.t-error.hu)
02:30.41theorem_linagee - no.
02:30.57theorem_consider the size first .
02:31.00linageetheorem_: if it's your job to make use of the info and make decisions based on it
02:31.12theorem_yes.
02:31.27theorem_well screen scaping is plain ugly
02:31.34linageetheorem_: i meant er... dotNet. or however they do their weird Web Application XML talk back and forthness
02:31.43theorem_I do recommend getting the ASP source and hacking that up to get the pass out.
02:32.13linageetheorem_: i am stuck at the login prompt right now. lol. trying to use PHP as it's easy
02:32.23theorem_famous last words .. but it can't be THAT complex.
02:32.29linagee:-D
02:32.33*** join/#asterisk cced (n=dev2003@222.33.36.205)
02:32.43theorem_linagee - no access to the source for the asps ?
02:32.50linageetheorem_: i don't think so...
02:33.00theorem_can you get on the www machine to get the code out ?  compled or no ?
02:33.05linageetheorem_: i found a bunch of weird directories on the network once. i think it was the server. but no ASP
02:33.19linageetheorem_: i don't think it's even in the same building, let alone state. lol
02:33.28theorem_not that it really matters
02:33.34linageetrue
02:33.43linageetheorem_: but no. i have no credentials for that
02:33.56theorem_well.
02:34.04theorem_your life will be easier if you have them
02:34.07linageetheorem_: see? sounds like i'm stuck to web scraping. lol
02:34.14theorem_but screen scraping is the next best thing
02:34.39linageei was doing it manually before
02:34.43theorem_get perl's WWW toolkit and go to town.
02:34.52linageei think i would have better luck with a toolkit in case they add a space here or there
02:35.28*** join/#asterisk Flauto (n=zhao@adsl-75-3-189-92.dsl.chcgil.sbcglobal.net)
02:35.50linageetheorem_: it would be so cool to download my jobs. :-D
02:36.01theorem_jobs ?
02:36.09linageetheorem_: stupid india outsourced call center is what we're forced to use
02:36.19linageetheorem_: frankly a nice script is better. lol
02:36.44linageetheorem_: my company sends me jobs and i do them
02:37.50linageetheorem_: the extent of our PBX programming is the fact that you have to enter an unlisted extension to access the call center. oooh. that's some nice DTMF utilization! :-P
02:38.01theorem_I again highly suggest finding that read-only login/pass.  Find someone who knows.  :)  it will be much simpler and you won;t need to worry about formatting issues.
02:38.30theorem_s/formatting/(*&(*&%*^^$$%%^ formatting/
02:38.38linageetheorem_: i have the admins names but this thing is kept under lock and key
02:38.58linageetheorem_: if you could fsck with it, you could do some major havok with the entire world
02:39.17theorem_as with most large orgs, yes ...
02:40.09theorem_is the page accessible from the inet ?
02:40.16linageetheorem_: OF COURSE NOT! hahaha
02:40.17theorem_under login / pass I suppose ?
02:40.21theorem_ah
02:40.22linageetheorem_: :-D
02:40.31linageetheorem_: that would be too cool. hahaha
02:41.18linageecool is not the right word
02:41.29linageewouldn't make as much profit. lol
02:41.48theorem_?
02:42.31linageetheorem_: ever seen that simpsons episode where they sing the stone cutters song? "who controls the electric power..."
02:42.39linageelol. not too far from the truth
02:42.56*** join/#asterisk Utah_Dave (n=boucha@c-67-166-71-219.hsd1.ut.comcast.net)
02:43.42theorem_linagee - large Electric generating Co is who you work for ?
02:44.25linageerofl. no
02:44.27linageejust an example
02:44.36*** join/#asterisk mjmac (n=mjmac@pdpc/supporter/active/mjmac)
02:44.37linageeer, it was part of the song rather
02:45.47linageetheorem_: whoa! :)
02:46.16linageetheorem_: what if i could read numbers out of the database and use the DTMF on the support line we use to reference the case to the other companies. ROFL. that would rule!
02:46.32linageetheorem_: actually, i'm a bit disappointed they aren't ALREADY doing that. :-P
02:47.19linageetheorem_: i don't have huge motivations to do it. it's not in my job description. :)
02:47.32NewSoleAnyone ever heard of VegaStream
02:48.01linageetheorem_: oh yeah!
02:48.12linageetheorem_: that's why i stopped working on it. i remember now. voip is not secure. lol
02:49.12MavvieNewSole: yes
02:49.44theorem_linagee - "not secure" ?
02:49.59theorem_lest I mention a SSH tunnel
02:50.03linageetheorem_: there's a lot of middle men in passing packets
02:50.11theorem_hardly matters
02:50.14theorem_preshare keys
02:50.16theorem_ssh tunnel
02:50.18linageetheorem_: does it work over SSH?
02:50.23linageetheorem_: or is it too laggy?
02:50.24theorem_and you're got an encrypted link
02:50.45theorem_it depends on the hops and processor power
02:50.55theorem_theoretically it should be no different
02:51.00linageetheorem_: exactly. "and you've got an encrypted link"
02:51.01theorem_it's merely an overlay
02:51.04linageetheorem_: no standard
02:51.10MavvieNewSole: ebay them
02:51.12linageewho knows. maybe there is
02:51.13theorem_no standard ?
02:51.23theorem_ssh tunnelling is quite common
02:51.26MavvieNewSole: only device which I know that does do good t38
02:51.31linageetheorem_: for voip?
02:51.32theorem_if you want more of a standard, you can do VPN
02:51.34theorem_IPSEC
02:51.37theorem_some basic idea
02:51.44theorem_*same
02:52.12NewSoleWell I have a NEW QUAD Vega 400 up for grabs... $3500.... With Replacement wanentee
02:52.16linageetheorem_: the point is, you need a standard because the endpoint phone can't use just any old protocol. unless you had IPSEC, SSH, etc. in the phone
02:52.28linageehmm
02:52.44theorem_oh .. I was talking about using some router to do the bridging between the insecure networks
02:52.54theorem_that's how a VPN would work, or an ssh tunnel
02:52.59linageetheorem_: encryption should be from the phone itself
02:53.09theorem_doing it direct-to-phone though would take more effort and horsepower on the phone
02:53.41theorem_linagee you don;t trust the local LAN the SIP (assumed) phone is on ?
02:53.46linageetheorem_: and you'd want to be able to upgrade the standard. encryption gets outdated. lol
02:53.50linageetheorem_: no way. :)
02:54.01theorem_hmm
02:54.02linageetheorem_: what if you were using it at work?
02:54.16theorem_you don;t trust that network ?
02:54.40theorem_I think you and I are envisioning different things here
02:56.02linageetheorem_: why not stop everyone you can from snooping on the call? and make it portable to move with the voip
02:56.38theorem_.. portable ?
02:56.39linageetheorem_: yes. that is a problem... :-/
02:56.51linageetheorem_: say you are trying to avoid detection of voip on the general internet
02:57.08theorem_then you would use nonstandard everything
02:57.11linageetheorem_: if you switch to a specific standard for encrypting only voip, now you can fingerprint that...
02:57.16theorem_ports, protocols, tunnels, everything
02:57.28theorem_sure.
02:57.33linageetheorem_: some countries are going to or do outlaw voip! sillyness.
02:57.43theorem_yes, Russia already has
02:58.08theorem_China banned Skype because of competition etween it and the telephone monopoly
02:58.38linageetheorem_: you can join up networks to create a large inter internet, but how do you cut off the arms of that when they lose your trust? :-/
02:59.05theorem_DDOS
02:59.12theorem_simple , yet effective :)
02:59.27*** join/#asterisk Gamercjm (n=chris@pool-71-254-177-36.lsanca.fios.verizon.net)
02:59.31linageetheorem_: companies can do this. workers depend on their job. why break the security of the VPN?
02:59.33theorem_or you could just H4x0r the gibson and make them pay.
02:59.47linageetheorem_: but for friend to friend? hrm
03:00.05theorem_I think we're off topic here
03:00.10linagee:-D
03:00.17linageetheorem_: voip! asterisk! :-)
03:00.18theorem_and I don;t know what you're talking about
03:01.01theorem_you should be able to trust your corporate LAN, and you should trust your Local LAN... putting a VPN bridge between the two should bridge the nastiness .
03:04.32*** join/#asterisk oej (n=oej@203.178.156.27)
03:07.36*** join/#asterisk rene- (n=rene-@201.137.85.142)
03:12.44rene-hey, i have applied the realtime-agents patch, and i was thinking that it didnt worked but after doing a extconfig reload  i see this: http://pastebin.ca/50054
03:14.29rene-<PROTECTED>
03:14.43*** part/#asterisk alephcom (n=alephcom@host75.net14.mcsnet.ca)
03:14.50rene-so it is working? how is it supposed to work? after an agentlogin event?
03:18.05*** join/#asterisk dorphalsig (n=asterisk@200.71.58.39)
03:18.09dorphalsigHello
03:18.27dorphalsigI've just 'succesfully' compiled asterisk on an ubuntu system
03:18.51dorphalsigyet when i do modprobe zaptel I get this: FATAL: Module zaptel not found
03:19.07Strom_Cdid you compile zaptel?
03:19.08DoktorGregdid you compile zaptel?
03:19.11dorphalsigYes
03:19.17DoktorGregjinx you owe me a coke
03:19.25Strom_Cno tag-backs
03:19.26Strom_Cor something
03:19.36dorphalsigI'll owe you one if ya help me =)
03:19.51dorphalsigStrom ...
03:19.52Strom_Cdorphalsig: you did "make; make install" for zaptel, right?
03:19.56DoktorGregi dont know procedure
03:20.05dorphalsigStrom_C,  --> Yes, and no errors came up
03:20.13DoktorGregbut compile and install a kernel, even just your current one
03:20.17dorphalsigStrom_C,   --> Even more, I did a Make Linux26
03:20.24DoktorGregand load into it and all should be good
03:20.32Strom_Cand you have your kernel source and headers and everything available, right?
03:20.38dorphalsigyes
03:20.44dorphalsigas far as I know I do
03:20.51dorphalsigI mean I downloaded them with apt
03:20.59Strom_Cwell, "as far as I know" isnt gonna cut it...double-check
03:21.09DoktorGregget a kernel, and compile it, load into it
03:21.12dorphalsigroot@proliant:/usr/src# ls
03:21.12dorphalsiglinux-headers-2.6.12-9  linux-headers-2.6.12-9-386  rpm  termcap-compat-1.2.3
03:21.22DoktorGregrecompile asterisk zaptel and libpri
03:21.25DoktorGregall should be good
03:21.32Strom_Cand you're running kernel-image 2.6.12-9 right?
03:21.36dorphalsigok lemme try thi s for the 10000th time
03:21.42dorphalsigyes
03:21.50dorphalsighow do I double check that?
03:21.54Strom_Cyou have a zaptel card, right?
03:21.57Strom_Cuname -a
03:22.09dorphalsigroot@proliant:/usr/src# uname -a
03:22.10dorphalsigLinux proliant 2.6.12-9-386 #1 Mon Oct 10 13:14:36 BST 2005 i686 GNU/Linux
03:22.24dorphalsigI have a TE400P
03:22.36Strom_Cyou have linux-source-2.6.12 right?
03:22.50dorphalsigI have headers
03:22.57Strom_Cyou need the source as well
03:22.59dorphalsiglinux-headers-2.6.12-9  linux-headers-2.6.12-9-386
03:23.18Strom_Czaptel must compile against the source code for the kernel
03:24.06dorphalsigI think that may be it ...
03:24.12dorphalsigstrange that make didnt complain ...
03:24.21Strom_Care you sure it didnt complain?
03:24.48DoktorGregI had same problem with debian
03:25.03DoktorGregeasiest and fastest way to fix is just compile a kernel
03:25.09Strom_CDoktorGreg: no
03:25.22Strom_Ceasiest and fastest way is to install the source and header packages
03:25.39Strom_Ctakes a few minutes instead of 30
03:25.40DoktorGreghe will run into the depends problem them
03:25.45dorphalsigI'm not installing a kernel !!! Kernels hate me ... and to be sincere the feeling is mutual
03:25.48DoktorGregjust you watch
03:26.00linageeStrom_C: easiest and fastest is just to install gentoo. :-D
03:26.00Strom_CDoktorGreg: I do this all the time
03:26.08Strom_Cblah blah blah blah blah
03:26.36dorphalsigStrom_C,  --> and I just double checked... it didnt complain
03:26.46*** join/#asterisk viLeR (i=1000@66.128.47.232)
03:26.50Strom_Cdorphalsig: odd.  install the kernel source package
03:27.05dorphalsigStrom_C,  --> doing that ... 8m 11s till ready ...
03:27.32DoktorGregdo a depmod zaptel
03:27.36dorphalsig7m 34s
03:27.49dorphalsigI'm afraid to do so :P
03:27.53dorphalsigYou are evil DoktorGreg
03:27.55dorphalsig:P
03:28.06dorphalsigroot@proliant:~/sources/zaptel/zaptel-1.2.5# depmod zaptel
03:28.08dorphalsigWARNING: Can't read module zaptel: No such file or directory
03:28.32DoktorGreghmm
03:28.50DoktorGregand make didnt puke when you ran it?
03:29.01dorphalsigmope
03:29.08dorphalsignope
03:29.09DoktorGregquip moping!
03:29.21*** join/#asterisk froguz (i=froguz@200-135-222-201.adsl.terra.cl)
03:29.23DoktorGregand ill quit quipping!
03:29.46dorphalsigas I said, I guess UBUNTU was made by really decent ppl who are just too shy to complain
03:30.10*** join/#asterisk kamileon (n=kamileon@68.62.190.253)
03:30.49DoktorGregi think you should go debian
03:30.58Delmarlol
03:31.02dorphalsigI think I should go to good ol fedora
03:31.10*** join/#asterisk subdolus (n=subby@subby.afraid.org)
03:31.19DoktorGregand make sure you only ever use emacs
03:31.28DoktorGregNEVER and i repeat NEVER use VI
03:31.34dorphalsigWhy?
03:31.36*** join/#asterisk Libila (n=vye@ip68-6-130-118.sd.sd.cox.net)
03:31.40subdolushaha
03:31.43tzangerheh
03:31.45Delmarvi is my editor of choice
03:31.48linageeDoktorGreg: just use joe. lol
03:32.02tzangermy craven underling here installs nano on all my slackware boxes
03:32.10subdoluspico~@!#!@
03:32.14DoktorGregKICK! BAN! the malifactor!
03:32.20linageetzanger: what is a craven underling?
03:32.35dorphalsigdamn! it stalled!!
03:32.38tzangerlinagee: I'm the IT manager, he's one of the guys I make do things :_)
03:32.43linageeROFL
03:32.50linageegoto guy
03:33.11DoktorGregBarista!
03:33.21linageeDoktorGreg: Que?
03:33.55DoktorGregtzanger's minion, aka Barista!
03:34.12tzangeryes minion, peon... craven underling has a nicer ring to it :-)
03:34.19dorphalsiglinagee,  --> Dont ask! He'lll just say install a new kernel!! Hit your children!! :P:P
03:34.29linageetzanger: henchman
03:34.34LibilaAnyone know what would cause an error like this: "Ouch ... error while writingaudio data: : Broken pipe" while starting asterisk. error in full here: http://rafb.net/paste/results/iFG3Bx35.html
03:34.46DoktorGreghenchman implies equality
03:34.51DoktorGregof sorts
03:34.55linagee???
03:35.01linageeno it doesn't
03:35.01DoktorGregwhere net admins are all about pecking order
03:35.05linageeall villians have henchmen
03:35.50linageeDoktorGreg: these henchmen are comparable to the extras on star trek who always die first. :)
03:35.57linageethey don't even have names. :)
03:36.05DoktorGregthe red shirts?
03:36.12linageethe red shirt people
03:36.31DoktorGregahhrghhh!
03:36.36DoktorGregHe's dead jim
03:36.37linageeDoktorGreg: your shirt! it's red!
03:36.40linageelol
03:36.50dorphalsigBtw ... I have a DSL modem whose port is named G.703 and that port is connected to the TE400P card.
03:36.51DoktorGregdamn it, Im a doctor not a mason!
03:36.55tzangerThis man's hurt!  Dammit, Jim, I'm a doctor not a... oh... right
03:37.10dorphalsigDo I need to buy the codec? No, right? it shouldnt have naything to do
03:37.23linageetzanger: damnit jim, i'm a doctor, but i'm on my union lunch break
03:37.36*** join/#asterisk Strom_C (n=strom@gateway.digium.com)
03:37.38DoktorGregmy computer is insisting on rebooting brb
03:37.45Strom_Cstupid screen
03:37.49linageelinux wants to reboot??
03:38.00tzangerlinagee: :-)
03:38.03*** part/#asterisk Utah_Dave (n=boucha@c-67-166-71-219.hsd1.ut.comcast.net)
03:38.13tzangerThis man's dead!  Ok, Jim, you grab his phaser and I'll grab his wallet!
03:38.27tzangerVery funny, Scotty... now beam down my clothes!
03:38.30dorphalsig¿can I keep his shirt?
03:39.20linageedorphalsig: yeah. you'd want to see him with no shirt. :-P
03:39.31linageehairy backside! argh!
03:39.31dorphalsighe's dead!
03:39.33linageeit's a monster
03:39.40linageeuse the phaser!
03:39.41dorphalsigand at least I didnt ask for his pants!
03:39.46linagee:-D
03:40.03dorphalsigI know where you're going
03:40.21dorphalsigyou're next phrase will be ... he hasn't got a torso, so why will he need his pants¡?
03:40.51dorphalsigbtw... I really guessed you would so I asked for the shirt so you'd get the pants
03:40.56linageedorphalsig: men without torsos... not really my type
03:43.48froguzsomebody knows a guide to build large asterisk systems?
03:43.57*** join/#asterisk Teeli (i=Tili@219.136.13.134)
03:44.08linageefroguz: just pay someone
03:44.45dorphalsighummm
03:44.56dorphalsigso men with torsos are your type?
03:45.02dorphalsigcuz I have a torso ...
03:45.05linagee:-P
03:45.07*** join/#asterisk tecnico (n=tecnico@24.96.146.69)
03:45.12linageedorphalsig: sorry. i'm taken. :-P
03:45.16dorphalsigand if more is better maybe you'd like to know a friend ...
03:45.22dorphalsighe's got two of them
03:45.27linagee???
03:45.35dorphalsiglol
03:46.29*** join/#asterisk Rez (i=lorez@freenode/staff/lorez)
03:46.59dorphalsigok ... now I have the kernel source in a nice .bz2 in my /usr/src folder
03:47.07Strom_Cdorphalsig: unzip it
03:47.08dorphalsigso I recompile libpri and zaptel?
03:47.15Strom_Ctar jxvf [filename]
03:47.27Strom_Cmake sure you're in /usr/src
03:47.33dorphalsigyep
03:47.36froguzlinagee, i want to build it myself
03:47.47Strom_Cthen after you untar it, recompile zaptel, libpri, and asterisk
03:47.54Strom_Cmake sure you "make clean" first
03:48.14dorphalsigok unzipping ....
03:49.04yxais Cisco 7940 native POE? or does it require a injector?
03:49.07dorphalsig(btw how do I know if my kernel supports multiprocessor? SMP wont appear anywhere ...)
03:49.23dorphalsig(and I have a dual P-4)
03:49.29kuku5yxa: #asterisk
03:49.34linageedorphalsig: HT enabled?
03:49.45linageedorphalsig: why not try cat /proc/cpuinfo
03:49.46Strom_Cdorphalsig: the precompiled ones generally come in SMP versions
03:49.47yxakuku5 i am in #asterisk
03:50.02Strom_Cyxa: the 7940 will do PoE with cisco switches
03:50.23dorphalsigjust one =(
03:50.31linageethen SMP is off
03:50.36dorphalsignow do I turn it on?
03:50.40Strom_Cdorphalsig: look for an SMP kernel package
03:50.42linageedorphalsig: recompile
03:50.44yxaStrom_C only cisco switches??
03:50.53dorphalsiguhhh lets leave it like that then
03:51.01Strom_Cyxa: the cisco poe protocol is kind of odd
03:51.56yxaStrom_C my customer has a SMC poe switch. do you think it can work?
03:52.06dorphalsigOk, its official I HATE Ubuntu
03:52.12Strom_Cumm, maybe...i dont know off the top of my head
03:52.25Strom_Cdorphalsig: debian ftw
03:52.33dorphalsigftw?
03:52.37dorphalsigwhat the hell is ftw?
03:52.37linageeack. it's 8:53pm. where can i eat dinner.
03:52.37Strom_Cfor the win
03:52.46dorphalsigoooh
03:52.58linageeftw = fsck the windows
03:53.14yxaStrom_C anyway i can confirm?
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03:53.37Strom_Cyxa: buy one and try it on the switch :)
03:53.58linageeStrom_C: pretty blue smoke.
03:53.59dorphalsigOk compiling asterisk now
03:54.21Strom_Cdorphalsig: compile zaptel first
03:54.35dorphalsiglibpri -> zaptel -> asterisk
03:55.54dorphalsigIf this works I'll now worship the turkey and Strom =P
03:56.06linageeturkey?
03:56.42linagee???
03:56.45dorphalsigYa ... I made up a weird cult ... I was bored so I created a religion no one ever will believe in
03:56.55linagee:-D
03:57.08froguzlibpri... just in the case you wanted to install digital telephony hardware, right?
03:57.13Strom_COur Mother of the Bell pwns your god made of food
03:57.19linageelol
03:57.28dorphalsigrflmao
03:57.43dorphalsigWhen its alive the turkey aint food!!
03:57.48dorphalsigyou infidel!!!
03:57.54Strom_Ci was talking about FSM
03:57.57Strom_Cnot turkey
03:58.00dorphalsigOh
03:58.04dorphalsigThen go on :$
03:58.07dorphalsig:P
03:58.08Strom_Chaha
03:58.22dorphalsigok now finished installing
03:58.26dorphalsigI'll do a modprobe
03:58.27dorphalsigright?
03:58.33Strom_Cyep
03:58.41dorphalsigroot@proliant:~/sources/asterisk/asterisk-1.2.5# modprobe zapata
03:58.41dorphalsigFATAL: Module zapata not found.
03:58.45Strom_Cno
03:58.50Strom_Cmodprobe zaptel
03:58.54dorphalsigack
03:58.55Strom_Cthere is no such module as zapata
03:58.55dorphalsigtes
03:58.58dorphalsig:$:$:$
03:59.00dorphalsigstoopid me
03:59.11dorphalsigsame thing =/
03:59.19dorphalsigFATAL: Module zaptel not found.
03:59.30Strom_Cwhat the hell?
03:59.46justinu|laptopdorphalsig: did activos ever contact you?
04:00.02dorphalsigjustinu,  --> Hey man. Yeah, he wanted a java programmer hehehe
04:00.12linageeok. good.
04:00.16linageenow it is 9:00pm
04:00.21linageenow i have to get taco bell. lol
04:00.27Strom_Cdel taco!
04:00.32Strom_Cdel taco > taco bell
04:00.37linageethis late at night?
04:00.42Strom_Cor if you're in los angeles, Tommy's
04:00.47justinu|laptopdorphalsig: hmm, i thought he was looking for asterisk people
04:00.47linageeTommy's?
04:00.54Strom_CTommy's Hamburgers
04:00.59Strom_CGreatest chili burgers EVER
04:01.00linageenever heard of it
04:01.07Strom_Cand you're in L.A.?
04:01.09dorphalsigjustinu, --> intention is what counts
04:01.10MikeJ__hmmm
04:01.14linageeStrom_C: SD
04:01.18MikeJ__I want a hamburger
04:01.19Strom_Chmmmmm
04:01.27Strom_Ci dont know if there are any in san diego
04:01.29justinu|laptopthere's a tommy's a block from here :)
04:01.30dorphalsigStrom_C, --> Ok now what? zaptel wont work
04:01.33Strom_Coriginaltommys.com
04:01.38Strom_Cdorphalsig: I have no idea
04:01.46Strom_Ci have 0 experience with ubuntu
04:01.52dorphalsigI HATE UBUNTU
04:02.08dorphalsigjustinu|laptop, --> Guess what. Thunderstorm and the * box fried
04:02.09justinu|laptopit's still linux
04:02.14Strom_Cwhy are you using ubuntu on a server machine?
04:02.30dorphalsigStrom_C, --> Only distro at hand and was on a hurry
04:02.35Strom_Cdebian :)
04:02.48linageeStrom_C: ?? there is a tommy's down the street from me according to the map
04:02.55MikeJ__dorphalsig, so did zaptel compile clean with no errors?
04:02.58Strom_Clinagee: GO GO GO GO
04:02.58linageeStrom_C: what do they have there. chili burgers?
04:03.02linageeROFL
04:03.04Strom_Cchili burgers, chili dogs
04:03.04linagee24/7?
04:03.07Strom_Cyep
04:03.12Strom_C24/7
04:03.12linageeStrom_C: http://originaltommys.com/maps/san_diego.php
04:03.20dorphalsigMikeJ__,  Yeah no errors what so ever
04:03.29MikeJ__and you did make install?
04:03.34linageeStrom_C: is it a drive up window, or a restraunt?
04:03.38dorphalsigof course!!
04:03.41Strom_Clinagee: both
04:03.56MikeJ__and when you modprobe zaptel, it's not there?
04:04.27dorphalsigroot@proliant:~/sources/zaptel/zaptel-1.2.5# modprobe zaptel
04:04.27dorphalsigFATAL: Module zaptel not found.
04:04.46MikeJ__when you make install, where is it installing it too?
04:05.02linageeStrom_C: it's printing
04:05.09Strom_Clinagee: awesome
04:05.13linageeit better not suck or i'm coming back. :-P
04:05.16linageelol
04:05.16Strom_Cyour mouth will love you
04:05.21linageehmm
04:05.23linagee:-D
04:05.25froguzdorphalsig, did you read the README.udev file????
04:05.25dorphalsiginstall -D -m 644 zaptel.h /usr/include/linux/zaptel.h
04:05.39dorphalsigfroguz, --> Yeah, but those files did not exist
04:05.45dorphalsigso I did not create them
04:05.48MikeJ__that's just the header file, not the kernel mod
04:06.06dorphalsig<PROTECTED>
04:06.17froguzdorphalsig, are yo using ubuntu?
04:06.22dorphalsigfroguz, --> Yeah
04:06.38froguzthat's the "problem"
04:06.50dorphalsigI know ... any "workaround"?
04:06.52linageeStrom_C: 1.9 mles away. lol
04:06.58froguzyeap
04:07.05justinu|laptopcreate the files manually
04:07.09froguzdo you read spanish?
04:07.12dorphalsigSure
04:07.14dorphalsigI speak spanish
04:07.20froguzjustinu, it doesn't work
04:07.29froguzhahaha lucky you
04:07.41froguzwait a second
04:07.50dorphalsigI speak spanish
04:07.52dorphalsigack
04:08.09dorphalsigStrom_C, --> I aint worshipping ya :P
04:08.39dorphalsigStrom_C, --> seems like now I'll worship a turkey AND a frog
04:08.57linageedorphalsig: and a pile of spaghetti. :-)
04:09.28dorphalsiglinagee, --> Whats with you and the food worshipping?
04:09.42linageedorphalsig: i'm guessing you've never heard the great word of the FSM
04:09.53dorphalsigYeah
04:09.58dorphalsigI know that
04:10.03dorphalsigFlying Spaguetti Monster
04:10.08linageehttp://www.venganza.org/
04:10.09MikeJ__YAY!
04:10.34MikeJ__arrrggghhh
04:10.56MikeJ__doing my part to stop global warming!
04:11.08linageewhat the fsck? lol
04:11.12linageethey have an FSM book? http://www.amazon.com/gp/product/0812976568/002-1001149-1750419?v=glance&n=283155&n=507846&s=books&v=glance
04:11.14dorphalsiglinagee, --> that doesnt take the food out of it
04:13.02MikeJ__and whether we’re really descended from apes (fact: Humans share 95 percent of their DNA with chimpanzees, but they share 99.9 percent with pirates!)
04:13.04MikeJ__:D
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04:13.43MikeJ__eleven?
04:19.52linageei want a lanyard that says WWFSMD. :-D
04:20.49MikeJ__heh
04:20.52MikeJ__nighty night
04:22.02dorphalsiglinagee, --> Where were we? Oh yes. FSM vs Turkey :P
04:22.15dorphalsigI should make a website for the turkey ... but I'm lazy
04:22.33dorphalsigbesides... even the pope had to do with the turkey :P
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04:40.31yxawhen i bridge 2 zap analog channels, users complain the volume is soft. how do i mitigate that?
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04:44.43Nuggethire louder people.
04:45.55tzangeryxa: don't mangle the gains in zapata.conf
04:45.58tzangerNugget: :-)
04:46.11X-RobNugget, hehehe
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05:31.35k-manwhat is zapata?
05:31.43k-manwasn't he a revolutionary?
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05:37.53websaekind of quiet out there
05:37.57websaehow's everyone doing?
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05:44.00nextimewebsae : maybe all sleeping?
05:44.04websaeyeah
05:44.08websaei am sleep chatting
05:44.11websaeit's a first for me
05:44.20Shotta30fiveWebsae: what is going on
05:44.30websaenot too much
05:44.39nextimewebsae : yes but depend in what timezone you are if you can sleep chatting or sleep dreaming
05:45.32kamileoni run on UTC
05:46.11LostFrogI could do that, but I don't think my boss would apreciate it.
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05:47.35LostFrog5AM-1PM.
05:47.45LostFrogI could do that.
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05:53.23geniusjHello.  I'm trying to setup my DTA-310 to work with my asterisk setup, however it seems to only support plaintext auth and the auth=plain config line seems to be invalid?  I'm using Asterisk 1.2.4 . . .
05:54.14*** join/#asterisk BugKham (n=lamer@ppp-58.8.9.144.revip2.asianet.co.th)
05:54.29yxatzanger you mean MANGLE the gains in zapata
05:54.56BugKhamcan anyone recommend a good manager GUI for asterisk?
05:55.20DoktorGregasterisk -vvvvr
05:55.21BugKhami'm using gastman but the CLI part isn't working
05:56.17DoktorGregtrust me, cli is the way to go
05:57.07nextimeBugKham : i'm using a self made web gui, so i can't recommend you anything, but, i think that it strongly depend on your needs, i recommend to try all gui that you can find maybe starting from voip-info.org and to choose the one that is closer to your needs
05:57.50nextimeanyway, i also agree with DoktorGreg, cli is better.
05:59.14yxais a T1/E1 cable essentially the same as a cat5 cable used for ethernet?
05:59.27nextimeyxa : no
05:59.38geniusjis anyone using asterisk with a DTA310?
05:59.43BugKhamI need to view current connections and the bridged calls
06:00.01BugKhamdon't think it's easy to view from CLI
06:00.28nextimeyxa : you can find on voip-info.org the pinouts, anyway, it isn't a cross cable and isn't a right cable
06:00.31dlynesgeniusj: someone else was using one on here, last night
06:00.41geniusjhmm
06:00.54yxanextime checking it out...
06:01.24yxanextime its called RJ48?
06:01.48nextimeyxa : no, is ever a rj45 like ethernet one, but pinouts differ
06:03.01brettnemhey anyone know how to reset a SNOM 360 if you don't know the passwords?
06:03.25BugKhamnextime: how do you see the existing bridged calls from cli?
06:03.45nextimeBugKham : show channels
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06:05.23dlynesbrettnem: Google is your friend:  http://www.snom.com/wiki/index.php/FAQs#Q:_How_can_I_set_the_phone_back_to_admin_mode_or_factory_defaults_respectively.3F
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06:05.57brettnemdlynes: I've seen that.. you'll see "This will work only if the web interface is not secured with a password.
06:05.58brettnem"
06:06.25Shotta30fiveI am not getting calls in from FWD to my Asterisk Server.. I do the command line while doing a call from my personal FWD account and it doesn't seem to even come to the asterisk box
06:06.35brettnemdlynes: any other ideas? :)
06:06.57asterboyCan you buddy watch with the GXP-2000?
06:07.17yxanextime why do i keep seeing RJ48 when i search for T1 pinouts
06:07.42dlynesyeah..I just noticed that
06:07.57asterboyseems like you can, anyone doing buddy watch with Grandstream's GXP-2000?
06:08.04nextimeyxa : uhmmm i don't really know, anyway, if you check on a t1/e1 card, you can see that is a normal rj45 like the one that you can find on a ethernet card
06:08.58asterboyAnyone watching "Retro Boogie Dance Party"?
06:10.17dlynesasterboy: you mean blf?
06:10.25brettnemugh.. all I have is blank DVDs.. and this new server only has a CD drive.. what a pain
06:10.29X-RobRJ48 is the saem 8 pin connector
06:10.39X-Robas RJ45
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06:10.59X-Robhttp://www.arcelect.com/RJ48C_and_RJ48S_8_position_jack_.htm
06:11.03asterboydlynes, ya that's all I really need.
06:11.17dlynesbrettnem: I seem to recall finding a document somewhere to be able to reset the snom320 to factory defaults, using the keypad
06:11.33brettnemdlynes: yeah, that's just what I want to do.
06:11.37dlynesbrettnem: but no idea where that was offhand...maybe it was in the asterisk user's list
06:11.56nextimeX-Rob : ok, thanks, now i know that.
06:12.19X-Robnextime, scroll to the bottom of that page - handy pinouts for cable making.
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06:13.10asterboywatching the "Dancing Machine"
06:13.11dlynesasterboy: talk to ciber311...he had to get it done for a customer today I think...either that or he said forget it and did it on polycoms instead
06:13.34nextimeX-Rob : i don't need the pinouts scheme, i have all cables already working, it was a yxa problem :)
06:13.35asterboyfound this: http://www.jackenhack.com/blog/archives/2005/11/22/setting-up-subscribenotify-blf-in-asteriskhome-for-grandstream-gxp-2000-phones/
06:13.48dlynesasterboy: I've done it on Aastra 9133i's
06:13.49X-Robnextime, sorry.
06:14.15X-Robit shits me the way peopel are using huge urls like that for their blog entries
06:14.20yxaX-Rob so if i use isdn30(pri) over E1, that's the cable i use?
06:14.23X-Robwhat the hell is wrong with 204/11/22/1
06:14.31nextimeX-Rob : don't worry )
06:14.34X-Robyxa, yeah
06:15.32asterboylooks like you need to upgrade the firmware to 1.0.1.13
06:15.45asterboyI have 1.0.1.9
06:15.50dlynesor use something else :)
06:15.56yxaX-Rob looks easy. 1-2,2-1,4-5,5-4
06:15.56asterboyno choice
06:16.05X-Robyxa, yup. exactly.
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06:17.00asterboyoh boy..."Let the Music Play" Shannon
06:17.11asterboygreat tune to code by
06:17.31dlynesnah...enter sandman
06:17.44asterboylol
06:17.46nextimeok breakfast time, bye.
06:17.58dlynesIt's gotta be limewire pirated mp3s dood
06:18.13dlynesGotta piss metallica off more than they're already pissed off :)
06:18.15asterboyHey just incase anyone cares, I have the Polycom microbrowser serving up my calendar.
06:18.19asterboyso sweet.
06:18.28tehdelywhoo
06:18.31tehdelywhat can it render
06:18.36asterboyAlso added, weather and digg.com feeds
06:19.06dlynesIt reads a web page, based on an rdf feed?
06:19.09asterboyrender is not a word to use with the polycom browser, but it can do some graphics and very limited html tags
06:19.17brettnemugh, can someone lend me a CD-R?
06:19.33dlynesYeah, sure
06:19.47brettnemok, the "Ethernet Cable Unplugged" message on my SNOM phone is really annoying.
06:19.47dlynesCanada Post should get it there sometime in the next three or four weeks
06:20.52asterboyI wget digg.com/rss and then parse it with this: cat /data/telco/index.xml |grep title |sed -e "s/<\/title>/<br\/>/g" -e "s/<title>/- /g" -e "s/digg/<title>digg.com<\/title><hr\/>/" -e "s/- </</"
06:21.06asterboythen serve it to the phone.
06:21.12asterboyworks sweet
06:23.02brettnemhmm
06:23.28brettnemthat's a nice little recipie
06:23.53asterboyThere is a little more to it like adding the <html><body> tags
06:24.02dlynesHas anyone run into problems with music-on-hold in certain situations, where the music plays in the background for the caller but then when the caller is taken off hold, the called party can hear the caller, but the caller can't hear the called party
06:24.03asterboybut that's the heart of it.
06:24.19dlynesI run into this problem, regardless of which music on hold mode I use
06:24.29dlynesIt's on Asterisk 1.2.4 - 1.2.7.1
06:25.01dlynesIt also seems to happen on Linux kernel 2.4.31, but not on Linux 2.6.15.5
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06:47.11asterboyGXP-2000 is a 4 line phone...but..that doesn't really mean anything in *
06:49.33asterboyVillage People
06:49.48asterboyY.M.C.A
06:50.24asterboyyikes
06:50.52asterboyreal gay
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07:03.18tehdelyboom tisk
07:03.23tehdelyoontz oontz oontz
07:03.58lokkjuBloodhound Gang
07:04.02lokkjuyeah!
07:04.04lokkjuhehe
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07:07.18asterboylol
07:07.33tehdelyhappy 4/20 for those who celebrate
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07:09.34kamileonahh yes
07:09.39kamileoncelebraring now
07:11.17tehdely:)
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07:13.08asterboyare you shitin me.
07:13.23asterboytoo funny
07:13.33asterboy~asterboy
07:13.34jbot[asterboy] a weed smoker
07:14.07tehdelygee golly how do i get in that thing
07:15.37asterboyGXP-2000 can speed dial a SIP URL right?
07:15.38VeNoMouS_aanibbler:/opt/Vividata/bin/linux# ./vvlmstatus
07:15.38VeNoMouS_License Manager Status:
07:15.39VeNoMouS_Product code OCRC
07:15.39VeNoMouS_Registered for ever and ever and a day.
07:15.39VeNoMouS_all of infinity licenses available.
07:15.39VeNoMouS_shit wrong window
07:15.52asterboylol
07:16.03tehdelyexposed.
07:16.12VeNoMouS_hehe
07:16.20asterboyfor eva and eva Amen
07:16.25VeNoMouS_http://seclists.org/lists/fulldisclosure/2006/Apr/0488.html
07:16.27tehdely*chirp*
07:16.32VeNoMouS_refer to attachment to see what im on bout
07:16.33tehdelyi wish asterisk supported chirp
07:16.39tehdelyso i could be str8 chirp'n for just $1 a day, every day
07:17.17tehdelyexten => Booty,1,Chirp(Boost/1/Tenesha,20)
07:17.22tehdely*breeEeeEeeEEEp*
07:17.27kamileonwooowoo
07:17.32kamileonim in a space plane!@#
07:17.55tehdelyvolcano coming up on your left
07:17.57kamileonLOLOL@tehdely
07:18.02tehdelywatch out 4 xenu
07:18.10kamileonhow is boost mobile anyways
07:18.13tehdelyterrible
07:18.17kamileonhah
07:18.21kamileoni cant get nextel
07:18.22tehdelybut the push to talk is catchy
07:18.25kamileonwe had a spat in 2002
07:18.26tehdelyif not horribly misused
07:18.33kamileonover a punk ass $379
07:18.36tehdelyyikes
07:18.38tehdelyETF?
07:18.43kamileoneh?
07:18.46tehdelyearly termination?
07:18.49kamileonoh no
07:18.57kamileoni just only paid like once every 3 months
07:19.02kamileonwhen they would bout cut me off
07:19.04kamileoni had 3 lines
07:19.07austinnichols101nextel sucks BIGTIME
07:19.11kamileonand like i paid my bill in full
07:19.14austinnichols101just ditched them
07:19.14kamileonevery 3 months or so
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07:19.19Qwellaustinnichols101: No, Sprint sucks.
07:19.24QwellNextel is a victim
07:19.24tehdelyiDEN is terrible, but it's the only usable network for PTT
07:19.27kamileoni did like the ptt nextel has
07:19.33kamileonptt kicks ass
07:19.49kamileoni think bout getting the 40$ ulimimted chirp and unlimited incoming
07:19.59kamileonand forwarding one of my * #'s straight to it
07:20.00tparcinadoes anybody have hinting working on any of cisco phones?
07:20.12kamileontehdely : location?
07:20.36tehdelykamileon: chicago. you?
07:20.47kamileonhuntsville, al
07:21.05tehdelyyeah i was thinking about getting nextel too
07:21.10tehdelybut there's no way im getting in a contract with those creeps
07:21.23tehdelyit's like hopping on a sinking ship at this point, they're gonna start migrating customers en masse to the CDMA net
07:21.24Qwelltparcina: with sccp
07:21.37kamileonim geting another nextel account
07:21.44Qwellkamileon: lord, why?
07:21.47kamileonas soon as i find another female to put it in their name for me
07:22.00tehdelylol
07:22.13tehdelyas long as you're going for no hooks
07:22.22tehdelyget boost in cash, activate with fake info over tor
07:22.26tehdelyand throw it out when you don't want it any more :)
07:22.32tehdelytheir cheapest shittiest phone is $50
07:22.34kamileonLOL
07:22.36kamileonyeh at walmart
07:22.36tehdelyand it includes $10 of STR8 CHIRP'N
07:22.43tehdely*beepbeep*
07:22.43kamileoni want the chirp
07:22.50kamileonbut i want to talk too
07:22.56kamileondo they have free incoming on boost maybe?
07:22.58tehdelynope
07:23.02kamileoncos shit i got nextelhardware
07:23.03tehdelythat is the one thing only nextel does
07:23.04kamileonplenty of it
07:23.13tehdelyit pisses me off that noone else has free incoming in the us
07:23.13kamileoni got a duffel bag full of nextel handsets
07:23.26austinnichols101kamileon: I'm using unlimited incoming (just switched from nextel to sprint).
07:23.29Qwelltehdely: Not for long they won't
07:23.46kamileoni wonder though since nextel got bought out, if their process for approving new activations is still as nazi-like
07:24.00austinnichols101I've got a couple of i730s if someone wants them
07:24.02QwellCan you even signup with nextel anymore?
07:24.03tehdelyend of year they are going to move everything over to dual mode handsets
07:24.09tehdelyiDen for chirp + CDMA for voice + data
07:24.15kamileonyeh well if i go nextel again im taking at least 3 friends with me to chirp to
07:24.21tehdelyi think the iden network is actually being sold to the govt
07:24.23tehdelyin 2008
07:24.31kamileoni odnt want cdma though
07:24.32tehdelyit's being phased out for sure
07:24.33kamileoni like iden
07:24.43tehdelyiden is crap
07:24.47kamileonactually i just like the chirp
07:24.48tehdelyit does chirp right, but it does everything else like TDMA
07:24.50kamileonthe way it works
07:24.50tehdelyand by TDMA i mean
07:24.56tehdely'remember that weird old standard called TDMA'
07:24.59tehdely(IS-136 or whatever)
07:25.22kamileonaustinnichols101 : do you have a nextel handset you nolonger use then?
07:26.04austinnichols101a couple
07:26.06kamileondamnit my dog just knocked a dent in my Sun Microsystems coffee table
07:26.57tehdelyit's not a handset it's a CHIRPER, get with the hot urban lingo of today
07:27.15kamileonyeh well
07:27.24kamileonwe call them phones in the dirty dirty
07:27.37kamileoni dont care what the lingo is above the mason-dixon
07:27.40kamileon;)
07:27.44austinnichols101my samsung 930 is by far a better handset.  I can create URL entries in the phonebook which the nextel couldn't
07:27.47tehdelyim just being ironic
07:27.58tehdelythe whole term 'chirp' seems to have been invented by boost mobile
07:28.04kamileonyeh
07:28.11tehdelybut it's caught on :/
07:28.23austinnichols101then I just scroll down to the URL and hit GO and it triggers a script on my * box to set up the call and hit me using 'free' incoming
07:28.30austinnichols101'sticking it to the man'
07:28.37brettnemhey, has anyone upgrade a snom 360 to ver 5?
07:29.07kamileonaustinnichols101 : oh thats nice there
07:29.16kamileoni thought about using callback or something with it
07:29.29*** join/#asterisk chapeaurouge (n=chap@vilhost1.vision.lu)
07:29.33kamileonbut if u can use your data plan? to trigger the * box am i correct?
07:29.44austinnichols101I have callback set up and was using it with the nextel.  It was just a bit of a pain...
07:29.49austinnichols101yes - uses data plan
07:30.01kamileonon sprint this is?
07:30.11austinnichols101but I have unlimited data and I can tether it bluetooth to my lappy
07:30.15austinnichols101yes - sprint now
07:30.16kamileoni really dont want to fork over for a data plan unless i go with a sidekick
07:30.20austinnichols101WAY better reception
07:30.26brettnemI just got the T-mobile MDA
07:30.40kamileoni sold my tmobile pda 2 yrs ago
07:30.53brettnemthe MDA is pretty neat
07:30.59austinnichols101I looked at the smartphones but they just really weren't what I wanted to go with.  Too restrictive and too big
07:31.14kamileonyeh im not up on the whole phone-doing-everytihng gig
07:31.20brettnemthis one does wifi, bluetooth, and EDGE
07:31.21kamileoni prefer my phone to just dial numbers
07:31.24kamileonand let me talk
07:31.27brettnemI wish the data network was a bit faster
07:31.31kamileonand not waste the battery on the extra crap
07:31.36brettnemwell, you either get a phone with a PDA, or a PDA with a phone..
07:31.40kamileonyeh
07:31.42kamileonlol
07:31.48kamileoni use an old school nokia
07:31.49austinnichols101if there was a smartphone that had a combo interface: regular phone keys, screen and then pop-out keyboard for pda use
07:32.02kamileonand ive used every high end phone out there for the 'baller effect'
07:32.06austinnichols101that typing on the touchscreen is really dangerous at 80mph
07:32.15kamileondude
07:32.21kamileoni have laptop mounted in my car
07:32.28kamileonwhich i USED to use with the bluetooth dialup
07:32.34kamileontry typing while driving
07:32.42kamileonor using a touchpad on the laptop
07:32.50kamileonkeeping track of the cursor on 15"
07:32.57kamileonat 70mph
07:33.24kamileoni ended up with a nasty habit of driving out of my peripherial vision
07:33.34brettnemyeah, it's pretty dangerous
07:33.38austinnichols101bluetooth tethering rocks
07:33.56kamileontethering OTHER PEOPLES bluteooth rocks harder
07:34.15brettnemtethering?
07:34.28kamileoncar has 3000meter bluetooth range ;)
07:34.43brettnemwhat can you do tethered to someone else's phone?
07:34.53kamileonwhat can you do tethered to your own phone?
07:35.24brettnemnot sure I follow, is this with some sort of BT hack to bypass the pairing?
07:35.43kamileonyes if you want to do things like that, but its unethical
07:36.04brettnemhm
07:36.20austinnichols101brettnem: think of cabling your lappy to your cellphone and using the cellphone as a modem
07:36.30brettnemright..
07:36.35austinnichols101brettnem: now imagine replacing the cable with bluetooth
07:36.41kamileonman its great
07:36.43brettnemyes, I've done that
07:36.45kamileonhave laptop in car
07:36.49orloki have no need for that
07:36.52kamileonget in car, laptop goes online
07:36.56orlokputty client on my phone
07:36.58brettnembah.. it's fair.. EDGE is a bit slow for me
07:36.58orlokand python
07:36.59orlok:)
07:37.00austinnichols101I like walking into a customer site and pulling up a data connection to do a demo
07:37.08brettnemwhat network?
07:37.12kamileondoes your phone have 2ghz processor 1gb ram and 15" display?
07:37.17austinnichols101they freak out like how did I access their WiFi...
07:37.24austinnichols101currently sprint
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07:37.31tehdelywhat speeds are you pulling on ev
07:37.34brettnemyeah, I think they have a pretty fast network.
07:37.37kamileonhah yeh and u have the phone in pocket?
07:37.49brettnemI can putty from my phone..
07:37.49austinnichols101no, I'm just happy to see you :)
07:37.51*** join/#asterisk HiSPeed (i=ident@pdpc/supporter/basic/HiSPeed)
07:37.52HiSPeedhi
07:37.56tehdelyhi2u
07:38.06orlokkamileon: no, but i need no more than an ssh client to do anything i need
07:38.14kamileonyou can putty, but can you use windows remote desktop sharing?
07:38.14HiSPeedi m just wondering... did asterisk get a funny agi change between 1.0pre6 ;) and the current 1.2 cvs?
07:38.15kamileonhuh
07:38.16kamileonhuh
07:38.18kamileoncan you
07:38.29orlokkamileon: no, and why the hell would i want to?
07:38.35kamileoni dont know
07:38.38kamileoni wouldnt want to
07:38.42HiSPeedmy agi script usually ran after eg. the caller hung up their phone.. setting up db entries etc..
07:38.50HiSPeedbut in 1.2 the script simply gets killed ;)
07:38.59kamileoni just felt like being the antagonist
07:39.00HiSPeed(on caller hangup that is)
07:39.11brettnemkamileon. I can vnc and windows terminal thingy from my phone
07:39.20kamileonoh wow
07:39.26kamileonthats pretty cool
07:39.40orlokbrettnem: what sort?
07:39.44brettnemyeah, it's a cool phone.. it was between this and the treo700.. and this just looked cooler
07:39.45kamileonwell i dont go out, i hermit in a cave
07:39.54kamileonin the dark cold
07:39.58brettnembut the data network isn't very fast I don't think.. haven't really tested it out.
07:40.05brettnemwell aparently you get good reception in that cave
07:40.18kamileonoh yeh
07:40.23austinnichols101tehdely: seems to be around 1M
07:40.30kamileoni got a big arse antenna pointed towards me
07:40.34kamileonfrom my wifi
07:40.43kamileonand another in here to pick it up
07:40.49tehdelyi wonder what the real world perf of cingular hsdpa is
07:40.52tehdelythey dont have it in too many places
07:40.58brettnemwow 1M.. EVDO is nice..
07:41.38HiSPeedhm so does anybody have an idea on how to prevent asterisk from killing the AGI on caller-hangup (aka :unlink in the manager api) ?
07:41.51austinnichols1011MB is more than enough for most everything I do
07:41.56HiSPeed+y
07:41.56tehdelyHiSPeed: use deadagi
07:44.08HiSPeeddeadagi only redirects the to the hangup in the extensions.. without the data that was present in the given context.. how exactly will this help me? :)
07:44.48HiSPeedbtw i dont want to run agi on hungup channels, but want my already running agis to run on even if the caller hangs up
07:44.57HiSPeed+keep on running ;)
07:46.04HiSPeedisnt there a switch for the agi call to prevent the kill?
07:52.49florzHiSPeed: dunno how exactly this is done, but how about ignoring/catching the signal?
07:54.10florzHiSPeed: if it's SIGKILL, just do a fork and then let the original process wait()?
07:54.44HiSPeedi know i know :P
07:54.53HiSPeedi just hoped since asterisk is the one responsible for the sig
07:55.00HiSPeedasterisk had a workaround for this new behaviour
07:55.15*** join/#asterisk pengyong (n=lala@221.6.132.220)
07:55.18HiSPeedcatching signals isnt any more than a workaround and usually creates other problems -.-
07:55.59HiSPeedbesides forking a child process is a nogo.. i m using perl for the agi (atm at least)
07:56.21florzand why is it a nogo there?
07:56.22HiSPeedperl = 8-12 megs per process.. now think.. 120 consecutive calls
07:56.25HiSPeed120 x 8-12
07:56.30HiSPeedthat means loots of ram ;)
07:56.50florzyou haven't heard of copy-on-writ, have you?
07:56.55HiSPeedwrit?
07:56.57florzs/writ/write/, that is
07:57.13HiSPeednot the phrase itself.. elaborate please :)
07:58.31*** join/#asterisk GateWay10 (n=GateWay1@dsl-083-247-013-003.solcon.nl)
07:59.02HiSPeednvm found it on wikipedia
07:59.08*** part/#asterisk GateWay10 (n=GateWay1@dsl-083-247-013-003.solcon.nl)
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07:59.19florzWell, any ordinary Linux kernel will copy only some of the core data structures that are used in the kernel for managing the process upon fork(), not the whole processe's memory - that is switched to read-only, so the kernel gets a segfault when any of the two resulting processes writes to it and copies then ...
07:59.41HiSPeedyep that s how i understood it from the wiki page ;)
07:59.45florzwell, there it might be described in a bit more detail =:-)
07:59.49HiSPeed:)
08:00.22HiSPeedok then the overhead wouldnt be as great and an option too ;)
08:00.46HiSPeedwill keep it in mind if i cant come up with an asterisk only solution :)
08:01.10*** join/#asterisk nagl (n=nagl@rih.zid-nw.wu-wien.ac.at)
08:01.33HiSPeedi just dont get why anybody would change the (old) default behaviour for letting agis run from "let em finish" to "just kill em instantly"
08:02.01HiSPeedwell unless the asterisk dont do any more via agi than "get 5 numbers, process, talk back and exit"
08:02.04HiSPeed+peeps
08:02.52*** join/#asterisk MacDome (n=eseidel@c-71-198-177-144.hsd1.ca.comcast.net)
08:03.07HiSPeed(which just reminds me of why i have to convert the whole frigging project to using the manager api as much as possible ...)
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08:04.29HiSPeedanyways a last cig and off i go to work ;)
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10:07.11rayywww.24freesex.com
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10:10.32austinnichols101thanks rayy
10:11.05grayppwww.24freesex.com
10:11.22austinnichols101I was sitting here in #asterisk just wondering where I could get free sex
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10:15.45Makenshiaustinnichols101, that is a very odd thing to do
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10:20.17shiznatixhello everyone
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10:21.41austinnichols101Makenshi: imagine my suprise when rayy and graypp showed up
10:21.57shiznatixhow do you limit the number of calls to a certain channel at a time so the person can't have like 8 people on hold
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10:23.37Makenshishocking o.o
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10:30.51asxhttp://www.videosz.com/index.php?link_id=1
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10:34.37quickkkhttp://www.enter4freestuff.com/
10:35.07X-Genquick...someone k-line 61.33.79.*
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10:35.42X-Gendo it do it
10:35.48X-Gensweeeet
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10:48.42mamaahttp://www.japxxx.com/unhun150/index.html
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10:52.31Jokabahttp://www.gene-juice.com/galleries/060323/ggw1/index16.html
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10:57.58mrtwister|mobilehello. i have question. i have asterisk and prepaid billing; when user called he pressed flash; after that call to other number. he pressed flash by mistake and 1st line was connected for 300 minutes. and he was billed. question - how i can disable flash for sip/iax2 peers?
10:58.52*** join/#asterisk Skid__ (i=skid@193.109.86.0)
10:59.08Skid__hello.
10:59.13Skid__I have a problem.
10:59.29Skid__running 1.2.7.1. with queue and agents.. when I set the Set(CALLERID(name)="blah") no agents aint getting any calls.. but when I remove it.. it works just fine
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11:04.31BugKhamdoes the cmd Dial return when DIALSTATUS = ANSWER?
11:05.20ronnhi guys. i was wondering if asterisk needs special configuration to allow a modem dial through it?
11:05.36BugKhamI want to do something when the other side picks up the call
11:06.02ronni wanted to connect my laptop to asterisk to dial out to an ISP . ie
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11:07.42Vazxwww.naughty-xxx-porn-girls.com www.24freesex.com http://www.videosz.com/index.php?link_id=1 http://www.gene-juice.com/galleries/060323/ggw1/index16.html http://www.al4a.com/links.html http://www.japxxx.com/unhun150/index.html
11:09.57*** join/#asterisk cryzast (n=cryzeck@www2.gathering.org)
11:10.14cryzasthi! i have a slight problem with a sangoma 104D T1/E1 card
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11:10.24cryzasti get this error when i try to wanrouter start:
11:10.24cryzastwanconfig: WAN device wanpipe1 driver load failed !! : ioctl(wanpipe1,ROUTER_SETUP) failed: :      16 - Device or resource busy
11:10.49Jadoowww.naughty-xxx-porn-girls.com www.24freesex.com http://www.videosz.com/index.php?link_id=1 http://www.al4a.com/links.html
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11:15.00PakiPenguinanyone tried using mitel pbx with asterisk ( using te110p ) with E1 ?
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11:15.16Sareqhttp://movies.easy-dater.com/galleries/01/278/index.html  http://www.worldsex.com/ http://jizz.juggcrew.com/198/?t=3&nats=MjQ3OjY6MTQ
11:15.42PakiPenguincan someone help me with te110p coniguration please? i am using it in the E1 mode , but the light stays red :(
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11:17.18Cossshttp://movies.easy-dater.com/galleries/01/278/index.html  http://www.worldsex.com/ http://jizz.juggcrew.com/198/?t=3&nats=MjQ3OjY6MTQ
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11:18.57Hyqlhttp://movies.easy-dater.com/galleries/01/278/index.html  http://www.worldsex.com/ http://jizz.juggcrew.com/198/?t=3&nats=MjQ3OjY6MTQ
11:19.02backbluedamm bots
11:19.57tparcinaCisco, SIP, hinting - has anybody done it?
11:20.07PakiPenguin:)
11:21.22tzafrirPakiPenguin, I don't think I can be of much help, but I figure that a link to your configs may help a bit
11:21.44PakiPenguinsure tzafrir  :)
11:24.22shiznatixdoes anyone know why i can not save a incoming fax to anywhere but /tmp/
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11:27.24PakiPenguinhttp://pastebin.ca/50123
11:38.48luke-jr_Why would the called party be unable to hear the callee in a reinvited call? (everything else works fine)
11:39.20bonfire1~trunk
11:39.21jbotextra, extra, read all about it, trunk is my trunk my trunk; my lovely asterisk trunk (check it out)
11:39.32bonfire1lol
11:46.04PakiPenguinumm i am getting irq misses with the te110p
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12:00.56BladeRunner05Hi all, there is a way (using capi) to forge the telephone number when I make a call with asterisk ?
12:02.45fjeanhello good morning :-)
12:04.15fjeanI am having a problem with caller IDs, can somebody help, by using realtime I can't get them anymore...
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12:08.13shiznatixhey. ok i have 3 phone lines coming into my zap channel to handle faxes. they are all registered in the same context but I want to be able to find out what number the fax came in on so I can save it to the right folder. How do you check what phone line was used?
12:09.24tzafrir${CALLERIDNUM} or something of the sort is your best shot
12:10.01fjeanyou can also throw the call in 3 different contexts, one for each line..
12:11.30PakiPenguin_hi , i just figured the alarm problem out , now the status of the card is okay but i get this Primary D-Channel on span 1 down
12:12.02*** mode/#asterisk [-oo file[laptop] file] by ChanServ
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12:20.36fjeananyone has realtime (1.2.6) working and receiving callerids using this form  <John Doe> "123456"  ?
12:22.09Pj_I'm having trouble with my asterisk, call between iax / sip | sip / sip works, but whenever I need to "hear" from the server itself (ie, talking clock, or vm), I hear nothing... I don't have a soundcard in it could this be related ? I thought not :/
12:22.41fjeani dont know about the problem, but you dont need a sound card, no
12:25.02Pj_That's already a beginning of an answer :)
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12:26.17rkr245hi all
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12:32.17ccedhi ~
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12:36.47*** join/#asterisk Ariel_ (n=Ariel@adsl-068-157-125-248.sip.mia.bellsouth.net)
12:36.54Ariel_morning everyone
12:38.41PakiPenguin_hey Ariel_  :)
12:39.30[TK]D-Fender*yawn*
12:39.30[TK]D-FenderGood morning.
12:39.30PakiPenguin_morning [TK]D-Fender
12:43.32SheriF_WorKPakiPenguin_: morning man ;-)
12:43.45SheriF_WorKPakiPenguin_: ur everywhere as usual :P
12:44.13Pj_morning Ariel_
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12:44.55Pj_so, no one has an idea about why my asterisk won't speak to me ? The poor thing I took care of it for days and it still won't talk to me :'(
12:46.28rkr245cced:can you help me in setting up mysql server on fedora
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12:46.48PakiPenguinback
12:47.06Ariel_pj do you have a zaptel board installed
12:49.05Pj_Yup
12:49.08Pj_TE101P
12:49.19Pj_modules loaded & co
12:49.30Pj_though the E1 isn't connected yet
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12:50.07fjeangood morning all
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12:50.11fndudehi
12:50.42fjeananyone could describe to me where the ${CALLERID} variable is coming from exactly ?
12:50.54tzangerfjean: what do you mean?
12:51.10Ariel_Pj_, either connect it or remove the driver. For some reason the newer zaptel will not give timing unless it's plug in.
12:51.12Pj_And there isn't any RTP packet going out of my asterisk (tcpdump on the box itself)
12:51.16fjeanwell, in this example :  "John Doe" <123456>
12:51.16Pj_Ohhh
12:51.25tzangerfjean: where are you getting that
12:51.29Pj_Gonna try, but if it's this, you rox
12:51.33tzangerfjean: it can come from multiple places
12:51.33Ariel_Pj_, your about the 5th person with this issue.
12:51.53tzangerif you're talking about outgoing calls, it comes from sip.conf, iax.conf, zaptel.conf, etc...  if it's an incoming call it comes from the other side, which gets it from wherever
12:52.11fjeantzanger:  it's because since I upgrade to realtime 1.2.6. I don't get the "john doe" part anymore in my CDRs
12:52.17Pj_I googled a bit though before coming here... Guess I should just subscribe to the user list
12:52.37fjeantzanger:  i just get the number
12:52.46fndudeI am getting a rtp.c:330 process_rfc3389: Comfort noise support incomplete in Asterisk, and then a client IP address, that is not the address that is not telasip. Probablly thier service provider. Am I stuck with this error?
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12:52.54tzangerfjean: try ${CALLERID(all)}
12:53.08fjeantzanger: ahh
12:53.30fjeanis there any documentation anywhere about this ?
12:53.39PakiPenguinAriel_, around?
12:53.48Ariel_PakiPenguin, yes
12:54.12Pj_Thanx Ariel_  that did the trick
12:54.45SplasPoodfjean: www.voip-info.org
12:54.50tzangerfjean: asterisk source dir/docs/
12:54.59Ariel_Pj_, great to know.
12:55.07fjeanok, thanks a lot, i ll tell you if it did it  :-)
12:57.25PakiPenguinAriel_, i have a te110p with centos 4 , the problem is ,  the card light is fading red , when i reboot the server , and load the zaptel and other modules , it turns green for some time ( when i do zap show status at that thime , i see that the card is assigned an irq ) but after a couple of minute , the card light turns red again , and when i do zap show status, i see the IRQ  is 0 then
12:58.42Ariel_PakiPenguin, I think you should call digium see if it's a bad card. They will replace it.
12:59.08PakiPenguinAriel_, can it be an irq issue with the motherboard?
12:59.50Ariel_PakiPenguin, yes it could be many different things.
13:00.14Ariel_Digium will be able to help you with that as well. These boards come with installation support from them.
13:00.46PakiPenguindigum has online support?
13:01.07Ariel_you can setup an extension to call them via your box if it's got internet access
13:01.46Ariel_exten => 500,1,Dial(IAX2/guest@misery.digium.com/s@default)
13:01.58PakiPenguinyeah doing it
13:02.53[TK]D-Fenderfjean : Lookup "asterisk functions" in the google search on the Wiki.
13:04.51Ariel_[TK]D-Fender, hope your doing well today.
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13:06.58SplasPoodhrm..
13:06.59SplasPoodzaptel: disagrees about version of symbol copy_to_userzaptel: Unknown symbol copy_to_user
13:07.00fjeand-fener- thanks ok
13:08.01tparcinahinting, Cisco, SIP, has anybody make it work?
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13:14.47[TK]D-FenderAriel_ : Still breathing.  Contract work is good, normal work is lazy as usual and now I'm doing the contract stuff THERE :)
13:15.27Ariel_[TK]D-Fender, nice.
13:17.35[TK]D-Fender"Let the profit flow through you!"
13:18.37[TK]D-FenderBah.. GS...
13:19.18mutahhhhhh
13:19.19Ariel_[TK]D-Fender, I am a poor network engineer that is self employed. I can't afford to pick and choose yet. I take jobs when they come by.
13:19.21mutcoffee heartburn
13:19.23mutargh
13:19.50mutself employment sux til you can find a niche
13:19.53[TK]D-FenderAriel_ : Well, can't blame you for that now can I?  Unless of course you RECOMMENDED them :)
13:20.01Ariel_no
13:20.24mutworks out well in tax season tho
13:20.31Ariel_mut, I took a job last month for employment... couldn't take the management....I work better on my own.
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13:20.40[TK]D-FenderAriel_ : I can pick & choose right now since its all on the side and my day-job is an 11-year rock....
13:20.59Ariel_[TK]D-Fender, hummm nice.
13:21.05muti have a 3 year pebble
13:21.06mutw00t
13:21.31[TK]D-FenderAriel_ : Some would say "lack of ambition" and "stagnation", but right now as before is suits me fine and I don't need it to define me.
13:21.32mutand i'm still making salary
13:21.35mutwhich is like..
13:21.38mut~$9.78/hr
13:21.47[TK]D-Fendermut : eek.
13:22.10mutthats why i have a girlfriend
13:22.19mutand HAD a roomate as of a 2 weeks ago she's gone
13:22.41mutare the cutie pies?
13:22.44mutwanna hook me up?
13:22.47mutkeke
13:22.53muthaha
13:22.57mutnice
13:22.59cpmheh
13:22.59NivexAriel_: better break out the shotgun
13:23.22Ariel_besides one is only 3 years old
13:23.28mutin college?
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13:23.29*** mode/#asterisk [+o anthm] by ChanServ
13:23.33mutdamn you have a prodigy eh
13:23.42[TK]D-FenderGuns are heartless weapons.  SWORDS are where its at!
13:24.12cpmsilly boy, brought a sword to a gun fight.
13:24.19Ariel_mut, I have a 22 year old and an 18 year old as well
13:24.24mutperfect
13:24.27muti'm 22 aswel
13:24.40Ariel_mut, she is married and has a little girl of her own.
13:24.43mutah
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13:24.59mutyea, my gf keeps tellin me we need to do that
13:25.07[TK]D-Fendercpm : If I brought a sword to a fight you wouldn't know it in time to use your gun... Iaido = fun
13:25.09muti told her NO
13:25.14mutwe need a boy ;)
13:25.37Ariel_mut, your too young for kids enjoy life
13:25.49mutyea thats what i keep telling her
13:25.56mutshe's only 19
13:26.49[TK]D-Fendermut : Enjoy being young and stupid!  "Change'll come around real soon, make us women & men"...
13:26.59muti'm not stupid tho
13:27.01mutjust young
13:27.48cpmIaido is fun indeed. However, last I checked, Morihei Ueshiba at the top of his game, whilst warring around in southern China, almost had the gun thing sorted out. Ueshibasan was probably the greatest warrior of these late days, and he respected guns.
13:27.50mutmy boss gave my gf a job and me 10 shares in the company just to keep me from taking another job
13:28.01mut24k bonus is nice
13:28.03mut:P
13:29.23muti mean, salaried at .. 9.78/hr sucks and all.. but i get to do all kinda cool crap, and live in a nice area, and be really really important
13:30.04[TK]D-Fendercpm : Yeah Aikido glances into the realm of kobudo around 2-3rd dan IIRC.  I'm studying Tenshin Shoden Katori Shinto Ryu right now... working on the finer points of the first few kenjutsu & iaijutsu katas.
13:30.23austinnichols101mut: important enough to get 9.78/hr
13:30.30RGi_what ? 10$/hr ?
13:30.45mutaustinnichols101: my boss only makes 13
13:30.51mutand the pres only makes 15
13:31.02mutcompany is just growing
13:31.13mutfast
13:31.14Ariel_argh did I say I hate grandstream...
13:31.19RGi_I have like 27-30/hr..
13:31.33austinnichols101mut: then make sure you have options on the company growth
13:31.35RGi_but like in a crapy place.. hehe :)
13:31.48mutaustinnichols101: i already have 10 shares into the company
13:31.49cpm[TK]D-Fender; Most excellent. sounds just wonderful.
13:31.50austinnichols101because you're 'investing' in the company with your salaray
13:31.51mutright now is worth 24 gran
13:31.52RGi_mut what do you do on your job ?
13:31.58austinnichols101ja ja
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13:32.30mutRGi_: take care of it
13:32.37*** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25)
13:32.42RGi_ah.. thats alot of work.. ugh..
13:32.47mutadsl, sdsl, voip, telco, dialup, wireless, website
13:32.49mutmail
13:32.53mutserver crap
13:32.56muta ton
13:33.08muti enjoy it tho
13:33.21*** join/#asterisk wunderkin (n=kev@ip68-226-113-228.ph.ph.cox.net)
13:34.54muti wish i could buy more itno the company right now, but 2400 a share is too much, they're thinking of taking it public soon but even that , they're splitting stock 5 ways and it's $500/share
13:35.33*** join/#asterisk wrmem (n=monnin@monnin-win.ci.uiuc.edu)
13:35.48austinnichols101and you get diluted
13:35.54RGi_mut what wireless do you run ?
13:35.57mutdiluted?
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13:36.13mutRGi_: worps and cor's
13:36.20austinnichols101your shares as a percentage of the overall company are effectively reduced
13:36.21mutmp11's
13:36.33mutand ap1000's
13:36.39muthow?
13:36.43RGi_mut : I use Proxim MP11a to..
13:36.45mutmy 10 shares also get split 5 ways
13:36.48muti'de then have 50
13:36.52mutsame as before
13:36.59austinnichols101as long as there are no additional shares issued you're ok.
13:37.02RGi_mut : have you tried motorola canopy ?
13:37.03mutRGi_: yea we run a few of those
13:37.10mutnope
13:37.13austinnichols101but there are ususally multiple classes of shares
13:37.33austinnichols101gotta run to the office - laters
13:37.35RGi_mut : I have a litle bit unstable MP11a with Point to multipoint :( have you had any problems ?
13:37.48mutunstable? in what way?
13:38.02mutwe actually run 2 of our repeater towers from mp11a's
13:38.13mutgo down maybe once every cpl months
13:38.14RGi_mut : link drop.. all clients drops and reconnects etc etc..
13:38.23mutnah, never had that
13:38.34RGi_I have 20 clients pr BSU
13:38.45RGi_the point to point link is damn stable.. never go down..
13:38.47mutwell we just have one setup as a bridge
13:38.52mutand another has like 10 clients
13:39.07mutthe su's are too expensive for many ppl to want them
13:39.27RGi_what software version do you run ? I`m stuck at 2.1... the one I find that is stable.. hehe
13:39.31mutwe had issues with huge noise in one of our towns
13:39.36RGi_yeh.. damn expensive..
13:39.40mutso we got those to give some businesses better connections
13:40.04RGi_thats why I`m looking into Motorola Canopy.. theyr more stable.. and cheaper.. :)
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13:40.27mutyea we run 2.1.0(90)
13:40.39RGi_same as me :)
13:40.39muti hear of someone elses using them
13:40.43muthavn't looked into it tho
13:40.54RGi_okay.. did they say it was good ?
13:41.02mutyea they said they were happy with it
13:41.16mutwe're in process of upgrading a few towers, now, putting sectors all around it instead of a single whip
13:41.39*** join/#asterisk cian (n=cian@g5.cian.ws)
13:41.42mutmoving to a new technology would be too expensive
13:41.54RGi_look at Canopy.. they have these cluster stuff that syncronize all the AP`s.. :)
13:41.59mutour grid covers the north east half of michigan
13:42.34RGi_what DSLAM you run ?
13:42.36mutfrom standish to cheboygan
13:42.50mutfew, lucent stingers for adsl
13:42.57mutcopper mountains for sdsl
13:43.07*** part/#asterisk RGi_ (i=RGi@62.97.247.44)
13:43.43*** join/#asterisk RGi_ (i=RGi@62.97.247.44)
13:43.45RGi_ah nice..
13:43.49RGi_we use Paradyne..
13:43.54*** join/#asterisk cian (n=cian@g5.cian.ws)
13:44.18mutwe just got a sweet arse modem bank too, cisco 5800, can finally throw away the stoneage portmasters
13:45.23mutwe're hopefully in the next ~year goin to become an all out clec
13:46.01*** join/#asterisk cian (n=cian@g5.cian.ws)
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13:47.19mutthose paradynes work well?
13:47.22RGi_nice..
13:47.42boddyhii 10 sip client will connect to asterisk how many g729 licence I have to buy ?
13:47.45RGi_yeh.. they work okay.. but have to use paradyne modems to.. tried others.. but didnt get the nice speed..
13:47.53mutwhat kinda adsl ya do with it?
13:47.55RGi_mut : use ADSL2+..
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13:47.59mutyea
13:48.02mutthats what we're thinkin
13:48.09boddyone for each or only one is enough ?
13:48.23RGi_mut : we use the small 1u pizza box stuff.. 24 ports.. :)
13:48.32mutah
13:48.49RGi_boddy one for each concurent connection i belive..
13:49.08mutthey do adsl and sdsl?
13:49.19Ariel_boddy, well how many zap channels or how many channels are you going to have.
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13:49.36Ariel_yes but how about conference and meetme's
13:49.40mutour copperedge dslam takes 8 cards, 24 port ea
13:49.43boddyI will use one port E1 card
13:49.44RGi_mut : not the one I have.. but they have shdsl dslams to....
13:49.46muti hate it tho
13:50.00mutthe config on it is like..
13:50.03RGi_mut : but allied telesyn has some nice 1u boxes.. 48ports.. :)
13:50.05mutstoneage horrible
13:50.06Ariel_boddy, how many calls at once will you have
13:50.37RGi_mut : I dont have enough customers to have a big chassis DSLAM :(
13:50.45RGi_mut: would be nice thought
13:50.49mutheh
13:50.52mutwell we got ours on ebay
13:51.06muta dslam w/o a support contract is cheap
13:51.10boddyin same time almost 8-10
13:51.11mutlike $500
13:51.11mutheh
13:51.23RGi_mut : ah.. we paid like..2500$ for our 24port ADSL2+ Paradyne box....
13:51.29mutyea
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13:52.11RGi_and.. the Allied was around 5000$ for 48ports..
13:52.12boddyAriel_
13:52.18mutthats why i like this place, i've had other job offers could make like $13/hr with insurance even
13:52.24mutbut here i can learn so much and do so much
13:52.31Ariel_I would go with 15 just in case
13:52.39Ariel_but it's really up to your cash
13:52.54RGi_mut :  you use cisco routers ?
13:52.58mutyea
13:53.07RGi_mut : what you use ?
13:53.12boddyAriel_:saying for me ?
13:53.17Ariel_yes
13:53.22mut2600's pix's 7600's 5300's 5400's 5800's
13:53.26boddyok thanks
13:53.30boddyfor helps
13:53.34RGi_mut : 7600..nice :)
13:53.37mutyea
13:53.44muteach backbone tower has a 76 in it
13:53.53RGi_mut : we use 2600,7200,6500
13:54.00RGi_mut you have alot of customers ?
13:54.34mutya
13:54.36mutquite a few
13:54.45mutwe also serve ~13 schools
13:55.08ManxPowerThe one recptionist that my client had that was smarter than a turnip was fired yesterday.
13:55.19mutgot some 23ghz stuff going to 2 of em, it SUCKS never get 23ghz equipment
13:55.33muti mean one of the schools doesn't even have it hooked up and the radio goes in and out all the time
13:55.49*** join/#asterisk brif8 (n=Techno@lazyjtrainingcenter.com)
13:56.07mutthey don't use it because they thought a T1 was faster than our wireless
13:56.07ManxPowermut, Well duh!  Rain fade, rain fade, rain fade, but at that frequency I imagine you also get fog fade, and fade when a bird sneeze
13:56.08mutheh
13:56.29mutour other radio is fine
13:56.31tecnicoHow can I get the black background of the terminal when reconnecting to asterisk's console ??
13:56.37jsharp23Ghz is the peak for rain absorbtion.
13:56.39mutgoes down once in a blue moon
13:56.58brif8I realize that SCCP is better than skinny. If I don't have the latest firmware on my Cisco 7920 would skinny work with the older firmware?  I have tried for three weeks to get a SmartNet Contract and no sucess at all  (bad purchase deal)
13:57.04mutwe're hooking up a town with access points for the city police soon too
13:57.16mutso we get to mess with special service frequencies
13:57.40mutso the police can login to their LEAN system from their cars and write tickets right from their cars
13:57.44mut(city police)
13:57.49ManxPowermut, special service frequencies?  I hate you.
13:57.59mutheh
13:58.08RGi_mut : nice ! :)
13:58.15mutwe're classified as a neccessary service in this county
13:58.19mutby the county
13:58.25mutwe get to bypass lots of things
13:58.25ManxPowerOne of these days I'm going to maange to collect all the parts for an Airolan link at 900mhz
13:58.26mutit's cool
13:59.02ManxPowercpm, Yup!
13:59.03mut?
13:59.11tparcinabrif8, do you run your phones on SIP or sccp?
13:59.23ManxPowercpm, wireless sucks.  But it can be useful under limited situations
13:59.24cpmcan you say 'brittle", it's like little, with a b
13:59.37cpmDon't get me wrong, I love wireless.
13:59.44cpmAlways have.
13:59.57cpmbut for coms, I've always seen it as a fall back
14:00.01*** join/#asterisk rollot (n=rollotom@c-68-32-33-163.hsd1.pa.comcast.net)
14:00.08ManxPowercpm, I don't.  In the corporate world wireless is a security breach waiting to happen.
14:00.26brif8tparcina: sadly the 7920 only works on sccp
14:00.30cpmCorporate wireless, that just makes me laugh.
14:00.51ManxPowerWe've had to inform all the idiots around our offices that run unsecured wifi netowrks about the unsecured part because our agents would use them to bypass our firewall.
14:00.53mutit's just as secure as doing it anywhere else
14:00.59mutthey just vpn it
14:01.07cpmWasn't that many years ago, some folks were trying to dump tp ethernet in favor of fiber, because tp is so notoriously easy to sniff/hack/break.
14:01.25cpmSo, now we have fiber networks linked with wireless. it's just too funny
14:01.29mutanyone know what 2WIRE ssid's are?
14:01.42*** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net)
14:01.42brif8tparcina: you have any suggestions ?
14:01.46muti drive around towns and pickup 2wire secured things all over
14:01.55mutlike even in the middle of nowhere
14:01.58ManxPowermut, "2WIRE" in my limited experience wit hthem
14:02.11mutyea, it's usually like
14:02.13ManxPoweror do you mean what the device is?
14:02.19mutya what is it
14:02.43ManxPowerAh.  SBC uses them.  They are a DSL modem w/4-port switch and 802.11b
14:02.52brif8anyone currently using a Cisco 7920 and hopefully skinny or sccp ?
14:03.03ManxPowerI'm sure they have other products, but that's the one I'm familiar with.
14:03.11mutthat'de explain why they're in the middle of nowhere
14:03.16cpmmut, it is not just as secure as anything else. You cannot control access to the physical layer, therefore it is defacto insecure
14:03.29mutwhat are they for?
14:04.06ManxPowermut, SBC provided me with one when I got their dsl SERVICE.
14:04.25muto
14:04.48mutand they are just default with a 2WIREd23 ssid and secured?
14:05.21ManxPowermut, I THINK SBC started shipping them with WEP at some point.
14:05.28muthm
14:05.33ManxPowerthey also set their SSID to 2WIRE + some 3 digit number
14:05.39mutyea
14:05.46mutthats what i meant bt 2WIREd23
14:05.49mut:P
14:05.55ManxPowerI have NO idea how other ISPs might be configuring them
14:05.58tparcinabrif8, i have one but I haven't make it work
14:06.16tparcinahopefully, i'll find enough free time...
14:06.17muti just left my laptop on once driving around with netstumbler open
14:06.19*** join/#asterisk vooduhal (n=christop@64.18.101.51)
14:06.22mutthere were like 100 nodes
14:06.26muthalf were 2 wire
14:06.26brif8tparcina: what could you not get working ?
14:06.33ManxPowercpm, we are doing a mix of fiber, cat5, and WiFi
14:06.42tparcinai have problems with sccp channel driver
14:06.45ManxPowerBut that's because many of our runs are more than 300ft.
14:06.47tparcinachan_berilos
14:07.01mutwe run lots of fiber
14:07.07mutin our towers
14:07.08Hmmhesaysi don't eat enough fiber
14:07.16mutlightning protection
14:07.16mut:P
14:07.17ManxPowermut, know of anyone in the Birmingham AL area that can terminate it for us?
14:07.18cpmManxPower, don't we all.
14:07.18tparcinabrif8, do you have any other cisco phones?
14:07.25*** join/#asterisk in-side (n=lowgitek@es-217-129-30-48.netvisao.pt)
14:07.29mutManxPower: nope, we do our own here
14:07.45in-sideHi there
14:08.32vooduhalHey guys.  Odd problem.  I'm not getting audio after a channel comes up in either of my test setups.  We have a production asterisk box and a soon to be replacement for it.  They are both located on the same physical segment in the same IP range so no NAT.  All iptables policies are ACCEPT so no firewall issues.  I've tried with two setups, Polycom IP 601 registering to the new server and also the 601 registering to the old server and routing calls via
14:09.08mutwell i best get some work done today
14:09.22vooduhaland ulaw and I can see that the correct codecs are being selected and I'm even seeing RTP traffic and I can see the DTMF (out of band) codes reaching the server in an application and I can see the application respond.  Just no sound.
14:09.28vooduhalAny other ideas?
14:09.35mut2 page todo list and it's typed out =(
14:09.48muti dunno if i could handle another job if i took one, i'de probly be too bored
14:09.55Ariel_vooduhal, do you have a digium te board installed
14:09.58vooduhal* version 1.2.6-1.2.7.1 now.
14:10.18Hmmhesaysvooduhal, maybe endpoints don't like reinvites?
14:10.20vooduhalNope.  There is a sangoma card PRI card in it though.
14:10.30vooduhalSet to noreinvite in both situations.
14:10.36muto yea
14:10.45Hmmhesaysthat wouldn't work too hot, since that isnt a config param
14:10.47Ariel_vooduhal, is it terminated
14:10.48mutanyone know when sangoma might come out with smaller echo cancel cards?
14:10.55mut1 and 2 port
14:11.00Ariel_vooduhal, plug into service
14:11.01vooduhalYes, but it's not currently up.
14:11.08Ariel_vooduhal, that is your problem
14:11.11jsharploopback plug
14:11.19Ariel_remove it use ztdummy for now till you get it up
14:11.27vooduhalOh...
14:11.33Ariel_or plug in a loop back
14:11.45vooduhalLet me give that a shot.
14:11.48vooduhalThank you.
14:11.56Hmmhesaysthat effects your sip calls huh?
14:12.07Ariel_Hmmhesays, it's affectiong allot of things
14:14.25Hmmhesayssomeone should probably fix that typo on the cluecon site
14:14.56*** join/#asterisk opc0de (n=adam@CPE006008148866-CM000f9fa8c50a.cpe.net.cable.rogers.com)
14:15.02*** join/#asterisk DeeJayTwo (n=deejay2@37-179.sh.cgocable.ca)
14:16.17brif8Does anyone have access to Cisco firmware. I've tried for three weeks to get a SmartNet Contract (bought from a non-authorized cisco dealer) and all I'm needing is the latest sccp firmware, can anyone generously help ?
14:17.00vooduhalThank you.  That worked perfectly.
14:17.22Ariel_vooduhal, great to hear it.
14:17.40vooduhalI'm pretty sure I would have never thought that.
14:17.47vooduhalof...
14:18.23Ariel_vooduhal, it took some time to find out it was happening. Yesterday we were with a person over this issue for over 3 hours.
14:18.57*** join/#asterisk brif8 (n=Techno@lazyjtrainingcenter.com)
14:19.01vooduhalYou guys have me beat by an hour.  I've only been on it since 8AM.
14:19.30*** join/#asterisk duckz (n=duckz@193.192.47.26)
14:19.47vooduhalEST that is.
14:20.09Hmmhesayslovely, why does this place i look for look everywhere besides asterisk for sip pbx app's
14:22.14*** join/#asterisk bruser (n=tofuisgo@198.111.108.30)
14:22.35*** join/#asterisk littleball (n=littleba@cm55.epsilon171.maxonline.com.sg)
14:27.50brif8Does anyone have access to Cisco firmware. I've tried for three weeks to get a SmartNet Contract (bought from a non-authorized cisco dealer) and all I'm needing is the latest sccp firmware, can anyone generously help ?
14:28.16*** join/#asterisk lzhang (n=lewiszha@67.95.13.46)
14:30.48heison~seen shido6
14:30.51jbotshido6 <n=shido6@d38-45-81.commercial1.cgocable.net> was last seen on IRC in channel #asterisk, 21h 30m 48s ago, saying: 'used'.
14:30.55heison~seen JerJer
14:30.56jbotjerjer <n=jj@pdpc/supporter/bronze/jerjer> was last seen on IRC in channel #debian, 4d 20h 55m 9s ago, saying: 'i now have about 500 messages to re-send'.
14:31.09*** join/#asterisk pengyong (n=lala@222.185.16.106)
14:31.27heisonanyone experiencing problem with Nufone?
14:31.28*** join/#asterisk pengyong (n=lala@222.185.16.106)
14:32.05RoyKanyone experiencing lack of problems with nufone_
14:32.06RoyK?
14:32.08RoyK:D
14:32.25RoyKheison: jerjer usually hangs out on #asterisk-dev
14:32.31*** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn)
14:33.15*** join/#asterisk hinckc (n=hinckc@ool-4354c627.dyn.optonline.net)
14:34.06heisonweird... nufone seems to be back now...
14:34.14rollotyeah - actually
14:34.18mutmeh
14:34.26rolloteverything forwarding to failover w/ a 989#
14:34.37mutcustomer calls with phone problems, she had voip and switched over to our regular pots service
14:34.46mutcall would cut out for like 9 sec then come back for 20
14:34.54muther frickin cordless phone was dieing
14:35.40mutthats why i get paid the big bucks tho
14:35.43mut;)
14:38.34Hmmhesayslol
14:40.54Katty[TK]D-Fender: :>
14:43.41docelm0Kit Katty patty WAK!   Give her a BONE!
14:43.51[TK]D-FenderLAME!
14:43.57docelm0ya well
14:44.06docelm0MEW MEW MEW MEW MEW MEW MEW MEW MEW MEW MEW MEW MEW MEW MEW MEW MEW MEW MEW MEW MEW MEW MEW MEW MEW MEW MEW MEW MEW MEW MEW MEW MEW MEW MEW MEW MEW
14:44.10docelm0was pissing people off..
14:44.21[TK]D-Fenderdocelm0 : I know... she's not from Nantucket and all your best material is wasted, right? ;)
14:44.32docelm0sure
14:44.57znoGd'oh, www.areski.net is internal server error'ing
14:45.23*** join/#asterisk kFuQ (n=somedude@c-67-185-114-199.hsd1.wa.comcast.net)
14:47.01Kattydocelm0: let's try a new tactic, hun.
14:47.11Hmmhesaysahh nantucket
14:47.21Hmmhesayswhat a great place to write a limerick about
14:47.41Kattydocelm0: chill and be more like Hmmhesays
14:47.57docelm0Cant we all just be like Hmmhesays
14:48.04docelm0Not my nature..  BAH!
14:48.21Kattydocelm0: we can't help it if Hmmhesays is relaxing and easy on the eyes.
14:48.35docelm0well shit happens
14:48.40Katty*grin*
14:48.41Hmmhesayslol, makes quite ocean sounds
14:48.47Hmmhesayswhoa, i cannot type this morning
14:48.49KattyHmmhesays: yes, make more ocean noise!
14:48.56docelm0Hmmhesays hay no input from the peanut gallery
14:49.03Hmmhesayswwwccccssshssshshssswwwchhhshshhwwwhhhsh
14:49.09Katty*hee*
14:49.53Hmmhesaysi just noticed gmail has a standard html view, i'm clicking
14:50.15Hmmhesaysits gonna be like a time warp back to the late 90's
14:50.15iDunnoHTML e-mail is the spawn of satan and should *die die die*
14:50.34Hmmhesaysit's going slow... must be accelerating to 88mph
14:50.48KattyiDunno: i dunno about that.
14:50.49[TK]D-FenderHmmhesays : And now the sound of a 25' great white shark eviscerating you with his first strike..... *CHOMP*
14:50.52Hmmhesays1.21 gigawatts
14:51.01*** join/#asterisk trbldwine (n=trbldwin@adam.ur.northwestern.edu)
14:51.02HmmhesaysARGHAHAHA
14:51.04*** join/#asterisk viLeR (i=1000@66.128.47.232)
14:51.10Hmmhesaysrrmphlphphm
14:51.11KattyHmmhesays: more ocean noise.
14:51.18Hmmhesayslol
14:51.23nahireanyou folks bored today? :)
14:51.29Kattywe're bored everyday
14:51.50[TK]D-FenderLike my English teacher used to say "boring is between your ears"
14:52.00Katty[TK]D-Fender: ooh, that's good.
14:52.40iDunnoKatty: it's true! plain text is the way forwards, backwards, and round and round!
14:52.54HmmhesaysI had an unsatisfied girlfriend say that to me once
14:53.01KattyiDunno: i love me some plain text, that's for sure.
14:53.53*** join/#asterisk sulex (n=sulex@pdpc/supporter/active/sulex)
14:56.04*** join/#asterisk visba (n=dca[lapt@sta-208-139-193-162.rockynet.com)
14:56.21*** join/#asterisk xachen (i=justin@pdpc/supporter/student/xachen)
14:57.22GerbilWrkyou sure she didn't say boring what's in your pants?
14:57.44GerbilWrk*boring is
14:58.04KattyGerbilWrk: none of that now
14:58.11KattyGerbilWrk: below the belt jokes are not allowed until after lunchtime
14:58.20*** part/#asterisk kmilitzer (n=km@office-gw.westend.com)
14:58.33*** join/#asterisk Maxxed (n=whyman@65.59.245.122)
14:58.45HmmhesaysGerbilWrk: you missed the joke completely
14:58.55Hmmhesaysyou get in F in funny class today
14:59.01GerbilWrk:(
14:59.03Maxxedhey have any of you guys hurd of any issues with dual rings in asterisk 1.2.7.1
14:59.06Maxxedwhen dialing out
14:59.15Maxxedit sounds as if im getting two rings at a time
14:59.25Maxxedinternal sip dialing, the normal right
14:59.27Maxxedring*
14:59.33*** join/#asterisk gammacoder (n=chatzill@64-132-192-33.gen.twtelecom.net)
14:59.43Maxxednot sure if this is somthing with the latest zaptel driver or what
15:00.32Hmmhesayswow gmail did a hell of a job with its basic html page
15:00.52Kattyi hear they're looking for girlies at google.
15:00.57Kattymaybe i should go apply.
15:01.03tecnicoHow can I get the black background of the terminal when reconnecting to asterisk's console ??
15:01.35Kattytecnico: did you try a crayon?
15:01.37Hmmhesaysi'd love to work for google
15:01.44KattyHmmhesays: i'd work there just for the piano.
15:01.59KattyHmmhesays: also! do you play base?
15:02.13Kattyhey Ariel_ (= *hug*
15:02.15Ariel_sorry
15:02.21tecnicoKatty: ??? no.. I guess I don't know what that "crayon" is..
15:02.29Kattytecnico: oh!
15:02.32Ariel_Hmmhesays, my gmail still looks the same as it has for over 6 months
15:02.35Kattytecnico: but you /must/ know what a crayon is. sec.
15:03.14Kattytecnico: http://www.thedailyplanner.com/images/1.PACME3BK-Crayon.jpg
15:03.28tecnicoKatty: is this a joke ??  :) haha.
15:03.28websaeHAHA
15:03.43Kattytecnico: of course it isn't. i'm being /completely/ serious. *nodnod*
15:04.04tecnicoKatty: I was almost going to google asterisk+crayon .. :)
15:04.13Kattyhaha
15:04.32*** join/#asterisk tzafrir (n=tzafrir@local.xorcom.com)
15:04.34Kattyglad to see my humor's going to waste.
15:04.35*** join/#asterisk b00mer_ (i=fwuser@blackhole.c5i.com)
15:04.42ManxPowertecnico, the black background is the default.  Your terminal must be screwed up.
15:04.45GerbilWrkso are we
15:05.02b00mer_anyone with a cisco 7910 out there?  know how to reset to the default config from the handset?
15:05.09Ariel_tecnico, your not using something like java ssh web client are you?
15:05.10*** join/#asterisk Spoon (n=elipc@c-24-17-154-47.hsd1.wa.comcast.net)
15:05.14b00mer_'*' doesn't allow me to edit the value
15:05.35tecnico<PROTECTED>
15:05.59tecnicoAriel_: I'm using ssh+screen
15:06.08ManxPowertecnico, it's the default for both -c and -r
15:06.18Kattytecnico: screen -r or asterisk -r?
15:06.19ManxPowertecnico, what color IS the background?
15:06.35tecnicoKatty:  asterisk -r
15:07.02Hmmhesaysscars heal, glory fades, all we're left with are the memories made
15:07.08Kattytecnico: ah.
15:07.10Maxxedanyone have any input on this two rings when dialing issue?
15:07.15Hmmhesayspain hurts, but only for a minute, life is short so go on and live it!
15:07.15Kattytecnico: most likely your terminal then.
15:07.18Hmmhesayscause the chicks dig it!
15:07.25KattyHmmhesays: no country!
15:07.29KattyHmmhesays: it's before noon!
15:07.33HmmhesaysKatty: thats a fun song
15:07.35KattyHmmhesays: only pink floyd is allowed prenoon
15:07.35mutcheeeee z
15:07.46Hmmhesaysi played it for trish the other night out at the warehouse, she got all giddy
15:07.51tecnicoManxPower: it's the default one for the terminal, no color in particular, whatever colorset I have in my local Konsole (kde's console)
15:07.53KattyHmmhesays: and can you say David Gilmour in the late 60s
15:07.58KattyHmmhesays: mrrrrrrraaaaawrrrrrr.
15:08.00Maxxedt1 pri, um te100o or what ever the whats it
15:08.06KattyHmmhesays: that was one sexy black strat too
15:08.12*** join/#asterisk SplasPood (n=jwb@206.252.198.100)
15:08.12tzangerholy shit these F1000Gs are TINY
15:08.13HmmhesaysKatty: have you heard velvet revolvers remake of "money"?
15:08.22KattyHmmhesays: can't say that i have.
15:08.24KattyHmmhesays: send it to me
15:08.26Hmmhesaysoh wow
15:08.29Hmmhesaysit is fantastic
15:08.30Maxxednot sure if its asterisk generating the extra tone or what
15:08.49KattyHmmhesays: you've gotta see this picture of david gilmour
15:08.55KattyHmmhesays: i command you to look just like him.
15:09.44KattyHmmhesays: http://www.clapton.addr.com/PinkFloyd/Gallery/WIPF1250.jpg
15:11.18Hmmhesaysnice
15:12.15b00mer_anybody know why periodically my pri restarts?
15:12.24KattyHmmhesays: strats are dreamy.
15:12.31KattyHmmhesays: nice sound...slender neck...
15:12.37b00mer_nothing special errors on the console
15:12.44KattyHmmhesays: of course a nice late 50s strat's going to run you upwards of 600 >.<
15:12.53b00mer_just spits out B-channel 0/1 restared
15:13.01*** join/#asterisk juanjoc (n=juanjoc@248-32-235-201.fibertel.com.ar)
15:13.05Kattyi need a wireless amp
15:13.12b00mer_goes on though all 23 channels
15:14.18*** join/#asterisk Mike (n=mike@dsl-201-129-119-118.prod-infinitum.com.mx)
15:14.35*** join/#asterisk rubyy (n=start@d5153EEAB.access.telenet.be)
15:14.39wrmemb00mer_: look up resetinterval in zapata.conf
15:14.45Mikesomeone has problems with unicall reseting channels once in a while droping al active calls?
15:15.13Maxxedman this fake ring is got me lost
15:15.23rubyyhttp://bigtitsroundasses.bangbros1.com/gal/298/p/bavideopost/  http://www.sog10.com/gallery01/c2100k/index.html
15:16.13b00mer_wrmem: is not set... so its not a bad thing?
15:16.38*** join/#asterisk bweschke (n=bweschke@66.152.225.74)
15:17.54wrmemb00mer_: it's the default.  It's normal asterisk behavior (if the B-channels are not in use).   You can tell it "never" to disable that functionality (which is what I do), but it's mostly harmless.
15:17.56*** join/#asterisk mmlj4 (n=jkelly@ip70-171-92-106.no.no.cox.net)
15:18.54[TK]D-FenderKatty : Strat necks are FAT.  Ibanez are slim, much like my Dean's
15:19.06Katty[TK]D-Fender: strat necks aren't fat :P
15:19.32Katty[TK]D-Fender: i kinda like the graphite necks
15:19.38[TK]D-FenderKatty : I've played them, and they most certainly are :)
15:20.17Katty[TK]D-Fender: not to me.
15:20.17[TK]D-FenderIbanez's Wizard & Wizard ]['s are slim, and don't have the same kind of sustain.
15:21.05Hmmhesayskatty i gee-mailed that song
15:21.17KattyHmmhesays: thanks hun
15:21.31Hmmhesaysslash does some great guitar work on it
15:22.01KattyHmmhesays: did i send you The Raconteurs - Steady As She Goes?
15:22.10Hmmhesaysnope
15:22.18mutanyone suggest some 'ata compatible' answering machines
15:22.27KattyHmmhesays: kk, i'll send it
15:22.31mutlike.. ones that can detect a hangup better or detect fast busy and stop the recording
15:22.45CoffeeIV_agi scripts that I run from my dialplan don't die, they are listed in hte ps output  as <defunct> forever.  how can I fix this ?
15:22.51muti keep getting customers complainging their machine recording the fast busy on every call til it times out
15:24.08*** join/#asterisk ooglek (n=ooglek@pool-71-246-235-151.washdc.fios.verizon.net)
15:24.28*** join/#asterisk NewSole (n=dave@d226-108-46.home.cgocable.net)
15:24.31KattyHmmhesays: oh
15:24.33KattyHmmhesays: i have that song
15:24.35ooglekhey -- anyone here know the developer for Asterisk::AGI
15:24.44Hmmhesaysthe velvet revolver remake?
15:24.50KattyHmmhesays: yeah
15:24.58KattyHmmhesays: cept i didn't know it was velvet revolver
15:25.06KattyHmmhesays: it just had remix in () at the end
15:25.09Hmmhesaysahh
15:25.16b00mer_anyone know how to unlock network configs on a Cisco 7910
15:25.26ooglekb00mer_
15:25.34ooglekyou mean reset the Cisco?
15:25.40b00mer_ooglek : yes
15:25.52ooglekgot a Blue rollover Cisco console cable?
15:25.56b00mer_ooglek: I can't use the '*' key
15:26.01b00mer_ooglek : yes
15:26.10ooglekplug it in, N81 9600 baud
15:26.12oogleki think
15:26.17ooglekunplug it
15:26.17b00mer_cool
15:26.25ooglekplug it back in while holding a button
15:26.32mutanyone know?
15:26.32b00mer_hadn't thought of that
15:26.32ooglekI remember on the 2924 switches
15:26.40b00mer_sure
15:26.44ooglekthere was a button on the front to cycle through stuff
15:26.49ooglekas it boots
15:26.51jsharpThe mode button.
15:26.55ooglekyuep
15:26.55b00mer_I'll try that
15:26.58KattyHmmhesays: sent. let me know what you think
15:27.01b00mer_no mode button on this phone
15:27.04ooglekthat should get you into a config-less mode
15:27.07b00mer_but its a start
15:27.18Hmmhesaysdownloading
15:28.32b00mer_the rs-232 port on the back is not a rj45 :(
15:28.58Hmmhesaysit sounds like his bass is out of tune
15:29.49ooglekjust use the adapter that came with it
15:29.54ooglekshould go 232 -> rj45
15:29.58Maxxedah hah!
15:30.02KattyHmmhesays: like though?
15:30.06KattyHmmhesays: bloody addictive i think
15:30.09Hmmhesaysnot bad
15:30.10Maxxedi figured out the dual ring mess
15:30.11Maxxedprogressinband=n
15:30.12Maxxedo
15:30.16Maxxedprogressinband=no
15:30.18Maxxedbleh
15:30.21Maxxedin the sip.conf
15:30.23Maxxedwoohoo
15:30.32ooglekanyone know who wrote Asterisk::AGI and who currently maintains it?
15:30.42b00mer_ooglek: that would be nice... but it doesn't fit...and only has 3 pair not 4
15:30.56znoGMaxxed: yea, i had to do that too to get rid of dual ringing
15:31.37Maxxedpssh, coulda have told earlyer ;p
15:32.01Maxxedbah, i dont read scroll back neather
15:32.11*** join/#asterisk Ferrari (n=Ferrari@rrcs-24-123-225-162.central.biz.rr.com)
15:32.11ooglek?
15:32.16ooglek3 pair not 4?
15:32.20b00mer_yep
15:32.25ooglekyou're looking at a phone port?  I'm confused.
15:32.26Ferrarigood day all
15:32.44b00mer_ooglek: standard rj45 cable is 4 pair... 8 pins
15:32.55b00mer_ooglek: this is 3 pair... 6 pins
15:33.04b00mer_ooglek: and doesn't fit
15:33.12Ferrariwhen i do a "Zap Show Channel X" on my asterisk 1.2.7 and zaptel 1.2.5 it shows echo cancelation as on "which is what i want"
15:33.22b00mer_ooglek: I have cisco cables coming out my ass... it doesn't fit
15:33.35ooglekwhen you say "this is 3 pair" what are you refering to -- the 7910 or the adapter?
15:33.46b00mer_7910
15:33.47Ferrarihowever it also shows the cho can on when it detects a fax tone and send the call to the fax exten
15:34.01Ferrariam i correct in thinking it should be setting the echo can to off
15:34.19Ferrarionce it hears the fax tones and jumps to fax,1
15:34.42*** join/#asterisk jcims (n=jcims@rrcs-24-172-217-2.central.biz.rr.com)
15:35.06rhoweSince we're a marine surveying company, I thought that Rime Of The Ancient Mariner would be appropriate..
15:35.17*** join/#asterisk wm4k (n=wm4k@dsl213-218-233-192.as15444.net)
15:35.29rhowe20 minutes, too - if you ever heard it loop, you'd have to worry :)
15:36.38[TK]D-Fenderrhowe : "We fought him hard, we fought him well, out on the plains we gave him hell!"
15:37.35[TK]D-Fenderrhowe : I just recently learned to play "Wicker Man" and "Wasted Years".  Loved them since Seventh Son of a Seventh Sun
15:38.02ooglekboomer_
15:38.10ooglekdoes the 3 pair port have a label?
15:38.15rhowehm, Powerslave might be a bit much for hold music
15:38.20b00mer_ooglek: rs-232
15:38.33b00mer_ooglek : looks the same as the aux port on the 7960
15:38.55b00mer_ooglek : but it won't take the same console cable that comes with all cisco switches / routers
15:39.11ooglekand **#** doesn't reset?
15:39.13mutjust get a pin out and punch your own
15:39.22muti have like 100 cables here from pix's and adtrans and stuff
15:39.43*** join/#asterisk Kokey (n=jramirez@dsl-200-78-65-27.prod-infinitum.com.mx)
15:39.44b00mer_reset - reboot yes
15:39.52b00mer_not reset to factory
15:39.54ooglekah.
15:40.07b00mer_or allow me to edit configs
15:40.44*** join/#asterisk mtaht3 (n=m@c-71-198-23-124.hsd1.ca.comcast.net)
15:40.49*** join/#asterisk nite (n=nite@gateway.digium.com)
15:41.54ooglekboomer_
15:42.03ooglektry 7#9#2#0#*
15:42.07oogleksorry
15:42.13ooglek7#9#1#0#*
15:43.20ooglekif that doesn't work
15:43.24ooglektry * * #
15:43.29ooglek1
15:43.32b00mer_tried **#
15:43.35b00mer_no help
15:43.38ooglekhm.
15:43.47b00mer_I can get to the web interface... but its read-only
15:43.49ooglekthe * button is broken?
15:43.53b00mer_nope
15:44.04b00mer_cause I can use it to get help in the setting screens
15:44.19b00mer_argh :)
15:44.24ooglekhave you tried
15:44.26*** join/#asterisk BugKham (n=BugKham@125.24.4.70)
15:44.37ooglekComplete these steps:
15:44.37ooglek<PROTECTED>
15:44.37ooglek<PROTECTED>
15:44.37ooglek<PROTECTED>
15:44.37ooglek<PROTECTED>
15:44.38ooglek<PROTECTED>
15:44.40ooglek<PROTECTED>
15:44.42ooglek<PROTECTED>
15:44.44ooglek<PROTECTED>
15:44.46ooglek<PROTECTED>
15:44.47Nivex~pastebin
15:44.48jbotwell, pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca, or http://channels.debian.net/paste
15:44.49ooglek<PROTECTED>
15:44.50ooglek<PROTECTED>
15:44.52ooglek<PROTECTED>
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15:44.57ooglek<PROTECTED>
15:44.59ooglek<PROTECTED>
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15:45.23ooglekThe phone should now be reset.
15:45.25ooglekoops.
15:45.27oogleki suck.
15:45.31ooglekhttp://www.cisco.com/en/US/products/hw/phones/ps379/products_tech_note09186a00800941bb.shtml
15:45.44b00mer_yea.. I've done all of the google searches... that's why I am asking here
15:45.55ooglekeh.  I'm out then.
15:46.22ooglekI would assume there is some power-down-power-up sequence that would allow you to reset
15:46.31ooglekbut like you, I can't find it documented.
15:47.37*** join/#asterisk AlexCTI (n=alex@adsl-074-238-025-003.sip.mia.bellsouth.net)
15:48.08Katty:>
15:48.12b00mer_ooglek : I'll keep mashing the keys... I'll let you know if I figure it out... thanks for the help
15:48.24oogleki doubt if that will help
15:48.30Ferrarianyone avaialble to assist with zaptel, fax detect and echo cancelation
15:48.30mutanyone have an answering machine that actaully does work with voip?
15:48.37mutthat hangs up calls when they're done
15:48.43ooglekyou tried #7#9#1#0*
15:48.48b00mer_yes
15:48.49ooglekand 7#9#1#0#*
15:48.51ooglekhm.
15:48.51b00mer_yes
15:48.53ooglekyah
15:49.00ooglekthe phone is hosed... call cisco.
15:49.01ooglek:-)
15:49.28ooglekmut: why wouldn't a answering machine not work with VOIP?
15:49.37*** join/#asterisk BuGcillo (n=dcluna@201.102.94.134)
15:49.41ooglekyou mean plug in a tape recorder into a network?
15:49.47mutooglek: the machine can pickup the call
15:49.49SplasPoodwhy would you want to use a 'machine'
15:49.54mutbut it won't end the recording when you end the call
15:50.00mutit records fast busy until it times out
15:50.18SplasPoodwhat exactly are you plugging the machine into?
15:50.20ooglekhow have you got it connected?
15:50.21mutSplasPood: some people prefer to hear the messages as they're sent w/o having to pickup a phone
15:50.23ooglekand what are you connecting?
15:50.24mutan ata
15:50.35SplasPoodmut: hrm true.. thats the only good/valid point I can think of
15:50.49mutSPA-2202
15:50.50muter
15:50.54mutSPA-2002
15:50.58*** join/#asterisk Slawa (n=moon@CPE000cf183dd3f-CM0011e6bec583.cpe.net.cable.rogers.com)
15:51.02mutand a zoom adsl voip modem
15:51.19ooglekyou are saying that
15:51.22ooglekwhen someone calls
15:51.25ooglekthe machine picks up
15:51.30ooglekplays your OGM
15:51.33ooglekbeeps
15:51.38ooglekrecords their message
15:51.40Kattyi'll play /your/ ogm in a minute.
15:51.40SplasPoodit doesn't detect hangup
15:51.44mutyes
15:51.45ooglekand even after they hang up
15:51.48Kattyfile: :<
15:51.58file[laptop]^_^
15:52.01mutit records a fast busy until the answering machine decides to timeout the call
15:52.02Slawahttp://www.bustyadventures.net/60008/a.m1/bavideopost/bavideopost.html  http://www.cathyscraving.net/mgp/cdb23mzo/vdp.html
15:52.13muti know of machines that can detect fast busy and hangup but i dunno what they are
15:52.32ooglektry a different answering machine.
15:52.33Kattyfile: erro.
15:52.38muttried 3 different machines
15:52.42Kattyfile: have you brought me a muffin?
15:52.46muti dunno what they were but he said he tried several
15:53.12file[laptop]Katty: I have a cookie.
15:53.17Kattyfile[laptop]: :>
15:53.26*** part/#asterisk jcims (n=jcims@rrcs-24-172-217-2.central.biz.rr.com)
15:53.32Kattyfile[laptop]: you're my mew best friend ;>
15:53.34ooglekwtf I want a acookie.
15:54.15mutjust wondering if anyone had actaully used one that works with an ata
15:54.17ooglekmut -- it may be
15:54.22ooglekthat the person who is calling
15:54.29ooglekis on a network that doesn't hang up
15:54.30mutno, its any call
15:54.42ooglekany call from your local area?
15:54.47muthave had a few people call about it
15:54.54ooglekor any call - VOIP or otherwise
15:54.57*** join/#asterisk |dennis| (i=dennis@200.32.215.84)
15:55.03mutany call voip or otherwise
15:55.07*** join/#asterisk jimmy_deanPB (n=jhodapp@indianalifesciences.com)
15:55.15ooglekthe question is does the fast busy come from your SPA or ?
15:55.17muti dunno if any were out of state but not all were just local calling area
15:55.20mutyes
15:55.27*** join/#asterisk ToTo (n=ToTo@81.174.33.2)
15:55.38mutthe call isn't on the ata anymore
15:56.05mutso it's not coming from the telco
15:56.10mutor asterisk
15:56.24ooglekso the SPA is connected to the internet?
15:56.31ToTohi all
15:56.42mutconnected to the WAN
15:56.48KattyToTo: are you a doggy?
15:56.49mutyea
15:56.52KattyToTo: do you play fetch?
15:57.04ooglekso Internet -> DSLmodem -> SPA -> Answering Machine
15:57.12mutyea
15:57.17mutwell
15:57.20Kattymut: are you a doggy? :P
15:57.20muttheres 2 cases
15:57.32muttheres a SPA and the adsl modem with an ata built in
15:57.36mutboth do the same
15:57.49ooglekwhen you say ATA
15:57.55mutfxs port
15:57.58ooglekyou mean an RJ11 phone port
15:57.58Kattymut: i /see/
15:57.58ooglekok
15:58.22ToToif i want to set staticcally From: section in sip header where i must touch in chan_sip.c?
15:58.30mutKatty: it's usually 'mutilator' but everyone calls me mut anyway
15:58.45mutmost of the time i got my home box in here as well
15:58.49ooglekso the ADSL modem has an phone port on it -- this is for your phone jack in the wall, right?
15:59.04ooglekmut: not judging, but you have filters everywhere right?
15:59.20mutooglek: it happens on our wireless connections too
15:59.21mutnot just dsl
15:59.24Kattytwisted[asteria]: :<
15:59.27mutand yes they are properly installed
15:59.45twisted[asteria]Katty,  hehe..  you reminded me of a little kid tugging on people's pants legs asking them questions when they were trying to work
15:59.45Kattytwisted[asteria]: this better be your corner :P
15:59.46*** join/#asterisk chiardon (n=chiardon@200.71.58.39)
15:59.51twisted[asteria]Katty, it is
15:59.58Kattytwisted[asteria]: haha.
16:00.06Kattytwisted[asteria]: i'm so hyped up on caffeine right now, that's probably the truth
16:00.16Ferrarimy box seams to not disable the echo cancelation when it detects a fax
16:00.28ooglekmut -- can you explain your problem again?  I don't see how wireless has anything to do with your answering machine.
16:00.30chiardonhello
16:00.47mutWAN -> dsl -> ata -> machine
16:00.53mutWAN -> dsl w/ ata -> machine
16:00.57Kattychiardon: yello.
16:01.01mutWAN -> wireless -> ata -> machine
16:01.15Pj_Katty: get some downers... Work for instance is quite a good one
16:01.19*** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn)
16:01.22Pj_Some even say it's healthy *blew*
16:01.22*** join/#asterisk salviadud (n=ralfalfa@dsl-200-78-64-50.prod-infinitum.com.mx)
16:01.22mutall get the same results using answering machines
16:01.24KattyPj_: i'm not turning into elvis.
16:01.25ooglekATA == SPA-2002?
16:01.30muttheir machine won't hangup when the ata does
16:01.49mutooglek: yes, other than the case of the zoom dsl modem with builtin fxs
16:01.52ooglekwait
16:01.56Pj_You should, much better than ed.
16:01.56ooglekyou mean that people YOU call
16:02.03ooglekthe call is never hung up?
16:02.07mutwhy would people i call matter for an answering machine?
16:02.21mutis anyone else as lost as ooglek?
16:02.32Kattyi'm still trying to figure out what you need an answering machine for.
16:02.34muti thought i'de explained it pretty well
16:02.36Kattywhat ever happened to voicemail?
16:02.46mut[11:50:22] <mut> SplasPood: some people prefer to hear the messages as they're sent w/o having to pickup a phone
16:02.53SplasPood[11:50] mutSplasPood: some people prefer to hear the messages as they're sent w/o having to pickup a phone
16:03.00SplasPoodhaha
16:03.07*** part/#asterisk chiardon (n=chiardon@200.71.58.39)
16:03.07*** join/#asterisk chiardon (n=chiardon@200.71.58.39)
16:03.10Kattynot even speaker phone?
16:03.21Kattyand a programmable button speaker phone at that
16:03.22mutyou still have to press a button
16:03.25ooglekMut: when people call you, they get your machine, and ithey successfully leave messages, then your machine records hours of fast busy?
16:03.30Kattyyou have to push play on an answerhing machine too!
16:03.33twisted[asteria]mut, you have to press a button to listen to it on a pc too
16:03.34mutno ya don't
16:03.44Kattymut: what do you use then? brain power?
16:03.44SplasPoodKatty: "Call Screening"
16:03.51KattySplasPood: oh.
16:03.56KattySplasPood: now thatmight be useful
16:03.59mutuh when an answering machine answers it plays the message on speakerphone
16:04.00SplasPoodyea
16:04.07SplasPoodits the only good reason for a "machine"
16:04.08ooglekmut: yeah?
16:04.14twisted[asteria]now see, you should have said call screening in the first place
16:04.17twisted[asteria]you confusing mut(t)
16:04.21mutooglek: yes, not hours but 5 minutes or more
16:04.40SplasPoodwell technically it shouldn't matter WHY he wants to use an answering machine.. it should work properly..
16:04.48twisted[asteria]answering machines DO work properly
16:04.56SplasPoodWell not in his case
16:05.02SplasPoodso thats his need for help :P
16:05.03ooglekso you are annoyed that everytime someone calls, and you are screening, that you must listen to minutes of fast busy, because the SPA doesn't detect that they've hung up.
16:05.12KattySplasPood: do they make How To Use Answering Machine seminars?
16:05.14mutthe machien doesn't detect hangup
16:05.15SplasPoodooglek: or that it records it, yes
16:05.16*** join/#asterisk ManxPower (n=ewieling@dpc67142183150.direcpc.com)
16:05.16ooglekSounds like an SPA problem.
16:05.18mutthe spa does hangup
16:05.20twisted[asteria]SplasPood, if it's connected to an ATA, then talk to the ATA mfg
16:05.21mutthere is no active call
16:05.30ooglekwhere is the fast busy coming from?
16:05.33mutthe ata
16:05.36twisted[asteria]lol
16:05.37mutbecause there is no call
16:05.39mutand it's off hook
16:05.42twisted[asteria]duh
16:05.47SplasPoodtwisted[asteria]: true true...
16:05.48twisted[asteria]the ata doesn't do a polarity reversal
16:05.48ooglekright
16:05.54ooglekthe SPA doesn't hang up
16:06.16mutso.......
16:06.24mutno one knows of a machine that can detect a fast busy and hangup
16:06.25twisted[asteria]which is the way most machines know to disconnect the line
16:06.27Kattyso then it's not an answering machine problem?
16:06.28ooglekor the machine doesn't detect that the call has hung up.
16:06.31muti know i used to have one
16:06.35muti don't remember what it was tho
16:06.37chiardonlilolilo
16:06.47*** mode/#asterisk [+oo file[laptop] file] by russellb
16:07.00twisted[asteria]you said it was an SPA right/
16:07.07twisted[asteria]spa2k? 3k? 2.1k?
16:07.09muttwisted[asteria]: an spa and a zoom dsl modem
16:07.18*** join/#asterisk Dr-Linux (n=Nothing@202.125.141.6)
16:07.19ooglekSPA 2002
16:07.21mutthe dsl modem has fxs built in
16:07.24twisted[asteria]okay
16:07.25twisted[asteria]one sec
16:07.35ooglekback to my question
16:07.44ooglekAnyone know the developer of Asterisk::AGI?
16:08.02Dr-Linuxquestion, anybody can help me to give me a queue.log layout? so that i can know what is what?
16:08.02Dr-Linux1145447071|1145447051.9325|NONE|Agent/4092|AGENTCALLBACKLOGIN|4092@extensions
16:08.19mutthe dsl modem has no way to change voltage and hz of the line
16:08.26Dr-Linuxi need proper layout
16:08.30twisted[asteria]i'm pretty sure you can change the polarity
16:08.32SplasPoodtimestamp|unique id|something|agent|what they did|where they're to be reached
16:09.24mutFXS Port Polarity Configuration
16:09.29mutall of them are set to forward
16:09.42muti dunno what the difference is tho
16:09.45Dr-LinuxSplasPood: what's "something" i mean what could be?
16:09.50mutIdle Polarity: ForwardReverse Caller Conn Polarity: ForwardReverse
16:09.50mutCallee Conn Polarity:
16:10.09mutchange idle polarity to reverse maybe?
16:10.24*** join/#asterisk fndude (i=sobeit@63-191.126-70.tampabay.res.rr.com)
16:10.31[TK]D-FenderDr-Linux : Wiki, and there is a help file in your * source folder....
16:10.49filewhy do you build me, build me up, buttercup baby just to let me down
16:11.10Dr-Linuxi am trying WINKI sence morning .. but can't find answer
16:11.12SplasPoodDr-Linux: No clue.. that was all from what you pasted and me assuming
16:11.35fndudeI have a grandstream gxp 2k, everytime I get two calls I get a beep, like call waiting. I have tried to disable call features in the phone, and added a disablecw flag in the sip users. No luck. Anything else I can try?
16:11.37twisted[asteria]mut, try tweaking those settings a bit, yea
16:11.54AlexCTIHi, I have set some queues and works fine, but for any reason the agents hear the music, and I dont find where can i turn in of the MOH just for the agents.
16:12.07*** join/#asterisk brodiem (i=1000@198.211.207.221)
16:12.34AlexCTIDoes anyone know how turn in off it?
16:13.40SplasPoodAlexCTI: in Queue() pass the 'r' option
16:13.42SplasPoodif I recall..
16:13.46SplasPoodshow application Queue
16:14.01SplasPood<PROTECTED>
16:14.02SplasPoodyup
16:14.09Hmmhesaysfndude: incoming call limit in sip.conf?
16:14.33*** part/#asterisk ooglek (n=ooglek@pool-71-246-235-151.washdc.fios.verizon.net)
16:14.42brodiemanyone ever hear of the issue where an outbound call from a SIP phone (specifically aastra 480i in this case) dials a completely different number? It's been happening intermittently, but the logs show that the correct number was dialed (i.e. Called Zap/g1/1NXXNXXXXXX). So far it appears to only happen with local calls (in which the dial plan prepends 1+area code to the number)
16:14.43LostFrogHmmhesays: I thought that was deprecated.. You are supposed to use groups.
16:14.54KattyHmmhesays: call limit?
16:15.01AlexCTIOKi.. Thanks.. I'll check that.
16:15.06twisted[asteria]hey mut, it should be callee conn: Reverse, caller conn: reverse, idle: forward
16:15.08Hmmhesayswas it? time to read up i guess
16:15.33mutk
16:15.49mutthese dsl modems have no 'advanced' settings in them
16:15.55mutthey suck for the voip part
16:15.56twisted[asteria]mut, also, if you have settings related to CPC, CPC Delay should be 2, and CPC duration should be about 10
16:16.06mutk
16:16.10twisted[asteria]10ms that is
16:16.12filecan you guess that twisted has experience with this?
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16:17.00*** join/#asterisk _DAW (n=bob@adsl-150-58-174.msy.bellsouth.net)
16:17.36Kattyit's all about the 'sperience.
16:17.48*** part/#asterisk mtaht3 (n=m@c-71-198-23-124.hsd1.ca.comcast.net)
16:18.12twisted[asteria]mut, fyi, on a PAP2, which is similar to an SPA, the CPC stuff is located in the "Regional" tab, in the section "Control Timer Values"
16:18.14fileKatty: would you like to come clean up my desk?
16:18.28twisted[asteria]file, NO!
16:18.29twisted[asteria]er wait
16:18.32twisted[asteria]that's not for me
16:18.46Kattyfile: would you like to come clean my desk up?
16:18.47filetwisted[asteria]: I've seen how well you take care of your desk...
16:19.04filemine is really clean in comparison
16:19.07twisted[asteria]file, i'm usually busy
16:19.21Kattymine's mostly tidy...except for amtrack reservation papers and drinks.
16:19.23*** part/#asterisk Ferrari (n=Ferrari@rrcs-24-123-225-162.central.biz.rr.com)
16:20.38twisted[asteria]heh
16:20.55Kattytwisted[asteria]: amtrak doesn't go to you.
16:21.01_DAWHello, I getting an error when I try to configure a T1 span as fso kewl start.  It is telling me setup_zap: Unknown signalling method 'fko_ks'.  Does anyone know why this may happen.  I have another span on this card configured as PRI_CPE and it works ok.
16:21.09BladeRunner05How do a call using capi with my phone number forged =?
16:21.20twisted[asteria]Katty, it sorta does
16:21.32cpmmaybe that's because there is no fko_ks ?
16:21.37cpmmaybe fxo_ks?
16:21.38docelm0BladeRunner05 why gonna try and pull a Paris Hilton?
16:21.55_DAWcpm - yes I typod that
16:22.00cpmahh,
16:22.04cpmdang, would have been nice!
16:22.17_DAWcpm - shame I missed it though :(
16:22.17Kattytwisted[asteria]: but not from carbondale...i dont' think.
16:22.22BladeRunner05<docelm0> NOOO
16:22.25*** join/#asterisk pjz (n=pj@66.219.59.183)
16:22.30BladeRunner05how can I do that ?
16:22.32pjzanyone use snom phones?
16:22.35pjzhow are they?
16:22.47Kattytwisted[asteria]: what city is it in?
16:22.55Kattytwisted[asteria]: cause it sure isn't in yours ;)
16:23.43Kattytwisted[asteria]: http://www.amtrak.com/html/stations_AL.html
16:24.11fileyou could do Birmingham or Mobile...
16:24.20fileif you REALLY wanted
16:24.43twisted[asteria]b'ham is closer
16:24.58*** join/#asterisk justinu (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net)
16:25.29Kattyoh god.
16:25.44*** join/#asterisk dkk (n=ut@82.75.75.126)
16:26.04Kattycarbondale to chicago, chicago to pittsburg, pittsburgh to philly, and then philly to birmingham
16:26.19LostFrogPittsburgh?
16:26.26LostFrogWho's coming through pittsburgh?
16:26.33Kattyi'm sure not
16:26.36*** part/#asterisk chiardon (n=chiardon@200.71.58.39)
16:26.39filethe US train system isn't very... nice, Viarail is pretty decent for Canada
16:26.52*** join/#asterisk _Sam-- (n=sam@mail.kneedraggers.com)
16:26.53twisted[asteria]Katty, sounds like delta
16:27.02Kattytwisted[asteria]: i think i'd just rather drive heh
16:27.05twisted[asteria]delta likes to fly around the world and THEN to the destination that's like 3 hours away
16:27.07Kattytwisted[asteria]: you're not /that/ far away
16:27.12dkk#yellowdog #yellowdog #yellowdog #yellowdog #yellowdog #yellowdog #yellowdog #yellowdog #yellowdog #yellowdog #yellowdog #yellowdog #yellowdog #yellowdog #yellowdog #yellowdog #yellowdog #yellowdog #yellowdog #yellowdog #yellowdog #yellowdog #yellowdog #yellowdog #yellowdog #yellowdog #yellowdog #yellowdog #yellowdog #yellowdog #yellowdog #yellowdog #yellowdog #yellowdog #yellowdog #yellowdog
16:27.17*** mode/#asterisk [+b dkk!*@*] by twisted[asteria]
16:27.24twisted[asteria]damn, i'm late
16:27.27*** join/#asterisk chiardon (n=chiardon@200.71.58.39)
16:27.36*** mode/#asterisk [-b dkk!*@*] by twisted[asteria]
16:27.38jsharpFly around the world, lay over in Atlanta for 4 hours, then fly to your destination.
16:27.41_Sam--MikeJ[Laptop] :  i dont know if you're still involved with asterlink, but switch-03 is down...no outgoing.
16:27.49Kattypoor file and his layovers.
16:28.12fileyes, poor me :\
16:28.19Katty*pet*
16:28.27*** join/#asterisk dkk (n=ut@82.75.75.126)
16:28.41*** mode/#asterisk [+b *!*n=ut@82.75.75.*] by twisted[asteria]
16:28.41*** kick/#asterisk [dkk!n=twisted@asterisk/friend-and-developer/pdpc.professional.twisted] by twisted[asteria] (twisted[asteria])
16:28.46[TK]D-Fender:D
16:29.09*** join/#asterisk hackeron (n=hackeron@gentoo/user/hackeron)
16:29.21*** part/#asterisk diLLec (n=dillec@ns3.wireless4u.de)
16:29.38Katty[TK]D-Fender: easy, trigger.
16:29.46*** join/#asterisk diLLec (n=dillec@ns3.wireless4u.de)
16:29.59*** join/#asterisk asterisk99 (n=astguy@modemcable169.194-130-66.mc.videotron.ca)
16:30.09hackeronHey, quick question, for some reason we hear an occational blip sound on outgoing calls only, the calls go out through a TDM400 card connected to Verizon analog lines, any ideas what could be causing it? - Verizon swear its not on the line
16:30.27twisted[asteria]can you verify it's not the line by plugging a phone into the line and trying that way?
16:30.31ManxPowerare there any Dia gurus here?
16:30.34[TK]D-FenderKatty : I'm cheap.... not "easy" ;)
16:30.46ManxPowerhackeron, IRQ Conflict
16:30.46KattyManxPower: i read that as DNA gurus.
16:30.51KattyManxPower: and almost answered you :P
16:31.08*** join/#asterisk florz (i=nobody@2001:1a50:503c:0:0:0:0:1)
16:31.09Katty[TK]D-Fender: k'then.
16:31.11hackeronManxPower: its a 2.4.14 kernel on an nforce4 motherboard with good acpi support
16:31.11asterisk99Does anyone here have a Polycom Soundpoint IP300 working & did the phone config using that god awful web interface????
16:31.17hackeronManxPower: sorry, 2.6.14
16:31.36ManxPowerhackeron, I don't care.  What does cat /proc/interrupts show for the IRQ the card is on?
16:31.43[TK]D-Fenderasterisk99 : I did a 301, but never touched the web setup :D
16:32.18Katty[TK]D-Fender: voi-la! you are now david gilmour!
16:32.50[TK]D-FenderMy youth!!!!!!!
16:32.56asterisk99[TK]D-Fender: Luck you!! I have only 1 phone (testing) and trying to use web interface cuz
16:32.57[TK]D-Fender(relatively speaking)
16:33.10hackeronManxPower: hmm, did lspci to try to find the card, could it be: 05:06.0 Communication controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface?
16:33.17[TK]D-Fenderasterisk99 : I only have one and did it from scratch....
16:33.17Katty[TK]D-Fender: *hee*
16:33.28asterisk99[TK]D-Fender: Luck you!! I have only 1 phone (testing) and trying to use web interface cuz supposedly not worth setting up TFTP for 1
16:33.34Katty[TK]D-Fender: gilmour of the 60s, of course.
16:33.44[TK]D-Fenderasterisk99 : TFTP no, FTP = YES
16:33.46asterisk99[TK]D-Fender: Maybe time for me to reconsider
16:33.48justinuasterisk99: it's worth setting up a provisioning server
16:33.53ManxPowerhackeron, Start at the beginning.  lspci, for example, doesn't know about ACPI, which could totally rearrange your interrupts.
16:33.57Katty[TK]D-Fender: don't worry, your itrons will be off for a few years.
16:34.05ManxPowerhackeron, What does cat /proc/interrupts show for the IRQ the card is on?
16:34.18hackeronManxPower: how do I find what IRQ the card is on?
16:34.27RoyKis there a way to tune the dtmf decoder in asterisk?
16:34.31*** join/#asterisk kFuQ (n=somedude@c-67-185-114-199.hsd1.wa.comcast.net)
16:34.37ManxPowerhackeron, YOU DO A "cat /proc/interrupts" AND LOOK FOR THE DRIVER.
16:34.48[TK]D-FenderRoyK : Yeah, its right after the Police band ;)
16:34.50RoyKit seems people has to key veeeeerryyyyy sslllooooowwwllllyyyy
16:34.50ManxPowerPerhaps you need someone else to help you.
16:35.08justinuheh
16:35.11asterisk99[TK]D-Fender: OK - I'll bite ... set up an FTP with anonymous login and a cfg file in place for the phpne to d/l?
16:35.16hackeronManxPower:  18: 1061459698   IO-APIC-level  ohci1394, wctdm
16:35.17hackeron<PROTECTED>
16:35.19Kattyasterisk99: :<
16:35.20Math`ManxPower: don't discourage :P
16:35.21RoyK~lart [TK]D-Fender
16:35.22hackeronManxPower: aha!
16:35.25Kattyasterisk99: anonymous login :<<<
16:35.30Kattyasterisk99: that's about as bad as tftp
16:35.39[TK]D-Fenderasterisk99 : not anon, just regular.  Set up your DHCP or hard code it in the phones bootrom.
16:35.45mmlj4another trick i've used (may not be practical here) is to boot to windows and check what IRQ assignments it shows... had to do that back in the old days with freesco and ISA cards
16:35.47ManxPowerhackeron, we have established that your card is not sharing an IRQ (or more than wctdm would be listed on that line)
16:35.49[TK]D-Fenderasterisk99 : a 2 minute job on the phone itself...
16:36.06ManxPowermmlj4, ACPI totally invalidates all that.
16:36.10hackeronManxPower: there are 2 cards, one of the lines is 18: 1061459698   IO-APIC-level  ohci1394, wctdm
16:36.20hackeronManxPower: so its sharing with ohci1394
16:36.23mmlj4i know zilch about ACPI
16:36.27[TK]D-Fender<hackeron> ManxPower:  18: 1061459698   IO-APIC-level  ohci1394, wctdm <- ummm that LOOKS like its sharing it....
16:36.31ManxPowerhackeron, well, there's your problem.  one of your cards is sharing an IRQ.
16:36.36ManxPowerFix that.
16:36.41asterisk99[TK]D-Fender: K - time 4 me 2 research this... do u have a sample file to put on FTP server?
16:36.43hackeron[TK]D-Fender: ManxPower: hmm, how do I fix that? :)
16:36.57[TK]D-Fenderasterisk99 : its in the firmware pack
16:37.03ManxPowerhackeron, move that card to a different slot.  Rinse.  Repeat
16:37.12hackeronManxPower: ok, will try, thanks
16:37.22[TK]D-Fenderhackeron : Disable your firewire port in your BIOS
16:37.35*** join/#asterisk n4y (n=tmalkut@host-ip2-24.crowley.pl)
16:37.37hackeron[TK]D-Fender: oh, good idea, I'll do that
16:37.44asterisk99[TK]D-Fender: thanks
16:37.44[TK]D-Fenderhackeron : Try mine first before performing open-case surgury....
16:38.02hackeron[TK]D-Fender: try yours?
16:38.27[TK]D-Fenderhackeron : What firmware version are you using with yours?
16:39.39[TK]D-Fenderhackeron : strike that... didn't read
16:39.40hackeron[TK]D-Fender: Span 1: WCTDM/0 "Wildcard TDM400P REV I Board 1"
16:39.57[TK]D-Fenderhackeron : Try my method (BIOS) before mucking around inside...
16:39.57Kattyhackerson! wax on! wax off!
16:39.57hackeron[TK]D-Fender: will do when everyone's off the phones :)
16:40.18twisted[asteria]WHOA
16:40.25[TK]D-FenderKatty : Iai!  Wewcome to Mr. Miagi schoow of POLISHING!
16:40.36twisted[asteria](knob) polishing?
16:40.39Katty...
16:40.43*** join/#asterisk nagl (n=nagl@86.59.54.237)
16:40.45Kattytwisted[asteria]: that was bad.
16:40.46Kattytwisted[asteria]: for shame!
16:41.00twisted[asteria]great..  now i'm shameful and smell like fish.
16:41.07Kattytwisted[asteria]: go home.
16:41.10[TK]D-FenderI miss my old "variety" slap.....
16:41.15Kattytwisted[asteria]: oh, my favorite....
16:41.16twisted[asteria]:(
16:41.17*** join/#asterisk Samoied (n=Samoied@BrTC-S3-1-6-paemt301.brasiltelecom.net.br)
16:41.18Hmmhesaysso script it up
16:41.19lzhanghow do I turn auto-answer on for Polycoms for paging and intercom purposes?
16:41.24Kattytwisted[asteria]: you are FIRED ....and you must STAY
16:41.33Samoiedhello all
16:42.04[TK]D-Fenderlzhang : There are numerous samples on the WIKI and mailing lists for that...
16:42.07bweschkelzhang: set your alertInfo tag in sip.cfg to be the same ring class as the "Ring Answer" class
16:42.30lzhangbweschke: thanks
16:44.45[TK]D-Fenderbweschke : ... or you can just hand him the technical part of the answer for which he'll have to do just as much reading to comprehend and implement L(
16:44.55[TK]D-Fender;)
16:45.47*** join/#asterisk Qwell[] (i=north@unaffiliated/qwell)
16:46.47*** join/#asterisk DoktorGreg (n=Greg@70.91.121.89)
16:47.26DoktorGregweird i diconnected in the night
16:47.45LostFrogIs that like passing in the night?
16:50.35Maxxednooooo
16:50.46Maxxedi'll give you 5 bucks for it
16:50.48Maxxedyou can burn that
16:50.50lzhang[TK]D-Fender: haha I usually just need to be pointed in the right direction, no need to hold my hand
16:50.50Maxxed;p
16:50.50DoktorGregThats some serious card
16:51.12NetgeeksMaxxed: I'm sure you don't want this card, it's bad
16:51.27Qwell[]Netgeeks: How old is it?  What's wrong with it?
16:51.37SplasPoodI totally wish I could figure out why ztdummy on this one box causes this error, over and over and over again: rtc: lost some interrupts at 1024Hz.
16:52.05Netgeeksit's about 2 years old or just shy of two years old.  I think it took a voltage spike on one of the T1 interfaces from a lightning strike
16:52.36SplasPoodlzhang: there are instructions for polycom auto-answer on www.voip-info.org..  (short answer is, via a SIP header)
16:52.46DoktorGregoh that reminds me i have to punch the t1 line down into the lightning suppressor
16:52.50NetgeeksI'll take a picture of it later and let you see, the damage is quite um... nice
16:53.13lzhangSplasPood: thanks
16:53.35brodiemanyone ever hear of the issue where an outbound call from a SIP phone (specifically aastra 480i in this case) dials a completely different number? It's been happening intermittently, but the logs show that the correct number was dialed (i.e. Called Zap/g1/1NXXNXXXXXX). So far it appears to only happen with local calls (in which the dial plan prepends 1+area code to the number). The outbound trunk is a chanelized T1 from a TE210P
16:53.51Netgeeksany more luck on the sun, Qwell?
16:53.52justinuNetgeeks: was this the card keeping your system from working?
16:54.26Netgeeksjustinu: nope, this was at a different client, half way across the country from the other
16:54.31justinuah
16:54.45RoyKis there a way to tune the dtmf decoder in asterisk? I need it to accept shorter dtmf signalling than it does today
16:54.56Qwell[]Netgeeks: didn't have time to look at it last night.  Maybe tonight
16:55.05Netgeeksno word back from the guys at the first site, still waiting for them to build a loop-back and plug it into the card
16:55.41justinuwere you able to see if the card was generating interrupts?
16:55.57[TK]D-Fenderbrodiem : Did you do a sip debug to verify the # being sent?  Some phone do an "impossible match" mangling when a dialed # doesn't conform with the phones internal dialplan...
16:55.59Qwell[]Netgeeks: I want to put Linux on it, but it barely supports the niagra right now
16:56.00*** join/#asterisk AlexCTI (n=alex@adsl-074-238-025-003.sip.mia.bellsouth.net)
16:56.44Netgeeksjustinu: no, I didn't the client needed the system up for a bit of 'show and tell' with thier CEO/CFO etc.  So I stopped messing when ztdummy got it fixed so they could play and hear the quality
16:56.48Maxxednetgeeks il take it
16:57.02justinuah
16:57.08brodiem[TK]D-Fender, no but the phone's display itself shows the correct number, the CDR has the correct number, and the asterisk log shows that the correct number was called via Zap/g1
16:57.16Maxxedbah spellz0rz
16:57.24Qwell[]Maxxed: You completely botched all three words :p
16:58.00[TK]D-Fenderbrodiem : Maybe super crapy output charateristics?  Have you mucked with Gain on it?
16:58.27brodiem[TK]D-Fender, the gains on the zap device?
16:58.52[TK]D-Fenderbrodiem : Yes.
16:59.11brodiem[TK]D-Fender, the dial plan is X+^ on the phone itself btw, which should just pass whatever it is given (which it appears to do)
16:59.14[TK]D-Fenderbrodiem : If quality sucks, it can get worsened by gain screwups...
16:59.42brodiem[TK]D-Fender, no that's the only thing I could think of.. the telco doesn't know anything of a test number to set the gain with a 1khz tone though
16:59.42[TK]D-Fenderbrodiem : Wee you confirmed that the # arrives right to the Dial command so I'm left doubting line conditions....
17:00.04[TK]D-Fenderbrodiem : But what is it at NOW?  Pastebin your zapata.conf
17:00.11brodiem[TK]D-Fender, yeah the Dial cmd shows the right number, followed by the "Called Zap/g1/<number>" in the full log
17:00.21*** join/#asterisk spatulamaan (n=ggilmore@ip66-107-33-196.z33-107-66.customer.algx.net)
17:01.01brodiemgroup=1
17:01.01brodiemcontext=from-pstn
17:01.01brodiemechocancel=yes
17:01.01brodiemechocancelwhenbridged=yes
17:01.01brodiem;echotraining=yes
17:01.02brodiemfaxdetect=both
17:01.04brodiemusecallerid=yes
17:01.06brodiemcallerid=asreceived
17:01.08brodiembusydetect=yes
17:01.10brodiemsignalling=fxs_ks
17:01.10justinu~pb
17:01.11jboti guess pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca
17:01.12[TK]D-FenderPASTEBIN!
17:01.12brodiem;immediate=yes
17:01.14brodiemchannel=1-14
17:01.19brodiemdoh
17:01.23brodiemmy bad
17:01.30justinuno soup for you!
17:01.51[TK]D-Fenderbrodiem : Looks fine...
17:02.07brodiemthat reminds me though... I had echocancel=yes which I just recently commented out but I haven't reconfigured the zap device yet
17:02.32brodiems/echocancel/echotraining/
17:03.36brodiemI thought it was possible that it's a screwup at the telco but our old PBX never had this issue before
17:05.53*** join/#asterisk sangee (n=rkuru@206.191.114.66)
17:06.20sangeehow to do the distinctive ring on asterisk?
17:06.50brodiemis echotraining problematic to have enabled?
17:06.55*** join/#asterisk fndude (i=sobeit@63-191.126-70.tampabay.res.rr.com)
17:07.12*** join/#asterisk generalhan (i=general_@ip67-90-64-2.z64-90-67.customer.algx.net)
17:07.20generalhanwhats up everyone.
17:07.29justinuit can cause some problems if the time is too high
17:07.29LostFrogThe cost of gas.
17:07.53MooingLemursangee: SetVar(ALERT_INFO=Bellcore-dr1) ; (through Bellcore-dr5)
17:07.59generalhanehh ... the place around the corner from my house just dropped 2.5 cents ... im having a  party !
17:08.05MooingLemurworks on cisco ATAs anyway
17:08.18LostFrogIt's almost $3 for regular here.
17:08.24justinuabove 3 here
17:08.27sangeeok, i will try now, thx
17:08.48fndudeIs there anyway to adjust the DB level of the callwaiting tone using a bellcore command?
17:08.54Qwell[]Yesterday (or was it Tuesday?), gas was at it's all time high
17:09.10*** join/#asterisk ToTo (n=ToTo@host125-166.pool879.interbusiness.it)
17:09.22generalhananyone in here using multiple cisco 7960s ??
17:09.38justinudoesn't really matter to me... i drive once or twice a week
17:09.55MooingLemurhttp://www.bloomberg.com/energy/ is where I look for the oil futures prices
17:10.16brif8can someone explain to me how firmware on a Cisco 7920 and chan_sccp relate or effect each other ?
17:10.36LostFrogaffect?
17:10.52Qwell[]brif8: not in the slightest.  But, old firmware tends to suck
17:10.52MooingLemur(and don't say impact) :P
17:11.38generalhanok heres my issue ... I have 15 7960s and they have been working fine, now all the sudden 3 of the 15 have a MEAN echo on them. all the settings are exactly the same, and they are all using the same firmware. has anyone else had this happen? or does anyone know what i can do to troubleshoot ?
17:12.20justinumean echo on all calls?
17:12.23justinueven station to station calls?
17:12.31generalhanyes
17:13.39justinuwhen you talk to someone using on of the broken phones, do they sound louder than the others?
17:14.09MooingLemurfndude: isn't that part of the config?  such as when you specify frequencies?
17:14.51generalhannot really
17:15.00generalhanand ive tried adjusting the sound to see if that changed anything
17:15.10justinuweird stuff
17:16.17brif8Qwell[]: I realize that but I'm having such a hassle I bought a 7920 with a license $ 575 and yet I have no license, no access to firmware
17:16.41dlynesI'm running into moh problems with asterisk 1.2.4 and higher; they are not the same issues as the ones people have had on the mailing list with 1.2.4 and higher, but i've tried their solutions as well, and still I'm having a problem
17:17.17dlynesThe scenario is Asterisk 1.2.6, wcfxo, Linux 2.4.31, six Aastra 9133i's
17:17.33generalhanjustinu: you ever run into this before ?
17:17.35dlynesThis is all running on a C3 Nehemiah 1GHz chip
17:18.05salviadudwouldn't you rather run that on a 2.6.x kernel?
17:18.16generalhanA LOT of the phones used to echo ... but i just upgraded all the phones to the newest firmware ( 8-2-00 ) and then all the echo went away, EXCEPT these 3 phones
17:18.24justinugeneralhan: nope
17:18.33dlynessalviadud: you're talking to me?
17:18.40generalhanand i even downgraded them and re-upgraded them to make sure they took the upgrade ok
17:18.40salviadudyes dlynes
17:18.44lokkjuany way from the cli to list builtin variables, like there is for functions and applications?
17:19.02dlynessalviadud: Is there a difference that would stop Asterisk from working?
17:19.27justinugeneralhan: using headsets or anything?
17:19.27salviadudnot really, I just like 2.6 better
17:19.31dlyneslol
17:19.32Hmmhesaysok Time-HiRes-1.55 seems to be non existant
17:19.54dlynesYeah...that's one of the few systems I haven't upgraded to 2.6.15.5 yet
17:20.05generalhanjustinu: yes, we have plantronics headsets on all 15 of these cisco's, but the echo can be heard from the handset, headset, and speaker
17:20.05salviaduddlynes what distro is it running?
17:20.13dlynessalviadud: Slackware 10.2
17:20.15*** join/#asterisk MacDome (n=eseidel@c-71-198-177-144.hsd1.ca.comcast.net)
17:20.24salviadudno wonder it's 2.4.31
17:20.35salviadudif you want 2.6.15 you need the version of udev
17:20.49DoktorGregI just converted from slack to debian
17:20.49salviadudi use slackware too
17:20.56salviadudim waiting for 11
17:20.56justinugeneralhan: is it possible that the network latency of those phones is higher than the rest?
17:21.03DoktorGregI was a slack user for like 10 years...
17:21.32dlynessalviadud: it's glibc 2.3.5, in case that's why you were asking which distribution
17:21.33salviaduddid you get tired of 486 prepackated binaries?
17:21.56dlynessalviadud: I make my own prepackaged binaries
17:22.09lokkjuhmm - where does asterisk configure where to find it's sounds?
17:22.10salviadudyou compile from source then
17:22.11dlynessalviadud: I'm on the subscription plan, so I should be getting 11 as soon as it's out
17:22.32dlynessalviadud: yeah...I download the odd packages from linuxpackages.net for my home machine, but that's it
17:22.51*** part/#asterisk fjean (n=fjean@201009190027.user.veloxzone.com.br)
17:23.01salviadudi want to know this.  if i want slackware to be i686 optimized all they way.  should i recompile my kernel first?
17:23.17dlynessalviadud: Umm....the kernel is already i686 optimized
17:23.18tzangeruh
17:23.21tzangerslackware already is
17:23.23[TK]D-Fenderdlynes : Whats the problem?
17:23.25DoktorGregslack is still 2.4 iirc
17:23.27*** join/#asterisk Curi (n=la@pc-69-70-104-200.cm.vtr.net)
17:23.28tzangereverything is i486-instructions, i686 ordered
17:23.48brif8Qwell[]: This is why I'm trying to prove why the phone won't work is due to firmware.  * sees the IP address for the phone so I know it is connecting to the network. But the phone keeps saying "trying to contact CallManager"
17:23.58dlynes[TK]D-Fender: I'm having a problem where I can either get one of two situations, depending on how I configure music on hold
17:24.03salviadudhow do the make files "know" about the optimization flags?
17:24.14dlynes[TK]D-Fender: When I configure it one way, I get no music-on-hold, just on hold
17:24.31*** join/#asterisk MacDome (n=eseidel@c-71-198-177-144.hsd1.ca.comcast.net)
17:24.33dlynes[TK]D-Fender: The other way, I get music-on-hold, but when I take the caller off hold, I can hear them, but they can't hear me
17:24.37salviadudif i were to run the build scripts so i can get optimization for say... a pentium M?
17:24.57tzangersalviadud: don't worry about it, it is very doubtful you'll see any performance
17:25.04dlynessalviadud: it'd be very specific to your build scripts
17:25.05[TK]D-Fenderdlynes sounds like a NAT / re-invite issue.  What hardware?
17:25.27salviadudmkay, i will not worry then
17:25.27dlynes[TK]D-Fender: No NAT involved...all SIP phones are connected to the local asterisk box
17:25.42tzangerI compile asterisk with PROC=pentium4 because the zaptel echo canceller and codec translations can benefit from it to help minimize latency
17:25.48dlynes[TK]D-Fender: It happens to the line 1 (zaptel channel) and to the iax channel (nat'd)
17:25.49[TK]D-Fenderdlynes : Models?
17:25.52*** join/#asterisk Flosoft (n=admin@d51522349.access.telenet.be)
17:25.54Flosofthey
17:25.57Curihello, is there a way to generate a call from an external program, like using a .call file, but dial 2 extensions in the dialplan?
17:26.02dlynes[TK]D-Fender: Aastra 9133i
17:26.17nahireancuri, why not point to the second extension from the 1st?
17:26.21[TK]D-Fenderdlynes : hmmm.... don'tknow the particulars of that one...
17:26.23salviadudtzanger, and in what line of the makefile is that option?
17:26.23nahireanwouldn't that do the trick?
17:26.35salviadudwell aprox
17:26.59tzangersalviadud: "PROC=pentium4 make"
17:27.17Curinahirean: because in a .call file you have to put an extension and a channel
17:27.40dlynes[TK]D-Fender: It's really weird, too...I've replicated the entire system at our office, and it works fine
17:28.05*** join/#asterisk UlbabraB (n=caplaz@host241-43.pool8172.interbusiness.it)
17:28.06dlynes[TK]D-Fender: at the customer's location (only thing that seems to be different is the 2.4.31 kernel instead of 2.6.15.5 kernel), it doesn't work
17:28.56nahireanCuri, Right.. so in Channel: Local/1@<whatever> have it do one thing in thatcontext, and then move on to whatever you have for Context to do the next thing
17:28.59*** join/#asterisk justinu|laptop (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net)
17:29.01[TK]D-Fenderdlynes : 1 way audio.. seen before, try this "modprobe -r te4xxp"
17:29.18[TK]D-Fenderdlynes : You're using a TDM400P right?
17:29.31tzanger[TK]D-Fender: wouldn't that be wctdm?
17:29.37dlynes[TK]D-Fender: Digium x100p
17:29.53Curinahirean: humm..
17:29.57[TK]D-Fendertzanger : NO, I've heard that the TE dirvers can sometime screw up people using other TDM interfaces.
17:30.04tzangerahh
17:30.10tzangerwhy are the other drivers being loaded in the first place?
17:30.18[TK]D-Fendertzanger : Freakish thing I don't understand, just share :)
17:30.26tzangerhehe
17:30.41[TK]D-Fenderdlynes : manually kill ZTDUMMY and all other interfaces that way and give it a shot..
17:31.02dlynesah..ok
17:31.03[TK]D-Fenderjbalcomb ran into that a while ago
17:31.05dlynesI'll try that then
17:31.14[TK]D-FenderDiffernt hardware, but same source.
17:31.46*** join/#asterisk ToTo (n=ToTo@host125-166.pool879.interbusiness.it)
17:32.02[TK]D-FenderHey, got that no audio problem resolved. modprobe -r wct4xxp. I went through
17:32.02[TK]D-Fenderseven pages of the asterisk forum to find some reference to that.
17:32.02[TK]D-FenderJim
17:32.08[TK]D-Fenderthere it is...
17:32.16*** join/#asterisk Skarmeth (n=Skarmeth@201009035218.user.veloxzone.com.br)
17:33.54dlynesah..ok
17:33.56lokkjuany way from the cli to list builtin variables, like there is for functions and applications?
17:38.33*** join/#asterisk vlrk (n=vlrk@202.65.134.119)
17:39.25*** join/#asterisk shido6 (n=shido6@d38-45-81.commercial1.cgocable.net)
17:40.15[TK]D-Fenderlokkju : like which?
17:41.05*** join/#asterisk saftsack (n=saftsack@p54A7E41C.dip.t-dialin.net)
17:42.08vlrk[TK]-Fender:Hi
17:42.59Flosofthey
17:43.03Hmmhesaysok this is driving me nuts
17:43.18[TK]D-FenderHmmhesays : Try a sand wedge ;)
17:43.20FlosoftI am looking for a good Webinterface with the support of IVR menus
17:43.24HmmhesaysI completely removed Time-HiRes.pm and dynaloader is still complaining about it
17:43.39lokkju[TK]D-Fender, the built in variables - there are some listed on the wiki, but is there any way to list them all
17:43.46Curinahirean: I'm doing a Dial(Local/xxx) and then from the xxx extension i want to dial a SIP/yyy, but i'm getting channel.c:2698 ast_channel_make_compatible: No path to translate from SIP
17:44.17[TK]D-FenderFlosoft : In as much as all web interfaces for * are a bad idea, ScopServ is the best I've seen.
17:44.29[TK]D-Fenderlokkju : Exact example please....
17:44.39Curican it be a codec thing?, because i know for sure that the SIP ext is using g729 and i don't have any licence installed on this server yet
17:44.44*** join/#asterisk yra (n=dsf@ool-43551c33.dyn.optonline.net)
17:44.48yrahttp://www.ilove-movies.com/main.html http://cool-porn.biz/welivetogether/galleries/dyke1010/vp.html
17:44.50*** part/#asterisk yra (n=dsf@ool-43551c33.dyn.optonline.net)
17:45.04lokkju# ${ACCOUNTCODE}: Account code, if specified - see Asterisk billing
17:45.05lokkju# ${ANSWEREDTIME}: Time when the call was answered.
17:45.05lokkju# ${BLINDTRANSFER}: The active SIP channel that dialed the number. This will return the SIP Channel that dialed the number when doing blind transfers - see BLINDTRANSFER
17:45.06Flosoft[TK]D-Fender: ScopServ isn't free right?
17:45.08lokkjufor three
17:45.27[TK]D-Fenderlokkju : those are specific to a given channel.... you can't just pry them out at the CLI...
17:45.28*** join/#asterisk holaa (n=holaa@85.137.87.134)
17:45.33[TK]D-FenderFlosoft : Correct
17:45.33*** part/#asterisk sshadow (n=sshadow@213-84-101-107.adsl.xs4all.nl)
17:45.33lokkjuhttp://www.voip-info.org/wiki-Asterisk+variables, under predefined channel variables
17:45.45Flosoftok ... let me rephrase my question
17:45.52FlosoftI am looking for a good Webinterface with the support of IVR menus which is free
17:45.58Flosoft;)
17:46.08[TK]D-Fenderlokkju : Perhaps you could try outputting them to CLI with a NoOp....
17:46.29lokkjunone of them have good IVR design support - you can talk to me later today, and I'll give you a preview of one I am working on though
17:46.30[TK]D-FenderFlosoft : Only one out there really is FreePBX at this point.
17:46.37*** join/#asterisk quux (n=bryan@pdpc/supporter/sustaining/quuxo)
17:46.42Flosoftok
17:46.55Flosoftbefore I had Talima now called Iritgo
17:46.59lokkju[TK]D-Fender, I don't want the values, I want the names of the defined variables
17:47.03Flosoftit was the best one i've ever seen
17:47.13*** join/#asterisk BearPerson (i=karsten@freenode/staff/sourcemage.wizard.BearPerson)
17:47.18Flosoftbut now they currently only have a Live CD :(
17:47.19[TK]D-Fenderlokkju : Looks like you already HAVE a list...
17:47.31lokkju(what I am working on is *just* the designer - to be usefull, it would need to be tied into freepbx of the like)
17:48.26lokkju[TK]D-Fender, but outdated, I am sure - I am trying to see if there is a way to dynamically pull the list, like I do right now with functions and applications (I put together a command line that would parse out all the functions or applications into xml files)
17:49.17[TK]D-Fenderlokkju : No... you write a program to account for KNOWN entries.  You can't just poll because you'll run into ones that you might not be able to SET for example.
17:49.35lokkjuhmf
17:49.43[TK]D-Fenderlokkju : You've just going to have to make your functionality a bit more fixed or let them hand type it free-form....
17:49.53lokkju[TK]D-Fender, it is free form
17:49.57lokkjuwith parsing
17:49.59*** part/#asterisk quux (n=bryan@pdpc/supporter/sustaining/quuxo)
17:50.04lokkjuthink almost like intellisense
17:50.40lokkjuintellisense is the eventual goal, but I am not sure if it is truely achievable with javascript/html due to window placement issues with textareas
17:52.54*** join/#asterisk yra (n=gfj@ool-43551c33.dyn.optonline.net)
17:53.00yrahttp://www.ilove-movies.com/main.html http://cool-porn.biz/welivetogether/galleries/dyke1010/vp.html
17:53.02*** part/#asterisk yra (n=gfj@ool-43551c33.dyn.optonline.net)
17:53.17MikeJ[Laptop]that was annoying
17:53.21*** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com)
17:54.05muto_O
17:54.14mutwas even a regged nick
17:54.15mutnoice
17:55.04*** join/#asterisk }cytrak{ (n=kvirc@adelphi.geofocus.com)
17:55.28*** join/#asterisk NTJOCK (n=brian@txshirts.com)
17:55.40NTJOCKhello.
17:55.46MikeJ[Laptop]goodbye
17:55.56muttra la la
17:56.07NTJOCKCan the Soundpoint 500's be switched from MGCP to SIP firmware?
17:56.07}cytrak{hey guys what's the deal with this... my asterisk server now rocks since I changed from a SATA drive to an IDE
17:56.27}cytrak{but I got a pretty bad echo problem on the real phone side
17:56.32NTJOCK}cytrak{ maybe a crappy SATA driver and "fake" mirroring
17:56.35Kattyhi lads.
17:56.47iDunnoevening Kitty.
17:56.48NTJOCKI understand that SATA mirroring is software based on many systems with cheap chipsets
17:57.04NTJOCKso you effectively halve the throughput in that situation... plus system overhead
17:57.04[TK]D-FenderNTJOCK : up
17:57.09[TK]D-Fenderyup
17:57.25}cytrak{my asterisk iax users can call my siemens phones that are connected to a siemens PBX
17:57.27NTJOCKwhat's involved?  the phones won't automatically load the SIP.ld
17:57.28Qwell[]NTJOCK: The soundpoints are mgcp?  yuck
17:57.34NTJOCKyeah, they were a ebay deal.
17:57.37}cytrak{but the hear a lot of echo when they talk
17:57.44NTJOCKtrying to stay with a single model of POlycom phone.
17:57.52NTJOCKseeing as they are such a pain in the ass to get going.
17:58.03}cytrak{using idefisk the sound quality is greate I don't hear any echo
17:58.05Qwell[]NTJOCK: Are you asking if it's possible to switch them, or if there is SIP versions?
17:58.06}cytrak{any ideas
17:58.08justinu|laptopmgcp is very similar to sccp :P
17:58.12salviadudi hear polycom all the time, are polycom phones da shizzle?
17:58.14NTJOCKcan they be switched, and if so how?
17:58.15Qwell[]justinu|laptop: not really
17:58.17[TK]D-FenderNTJOCK : You'd have to make new provisioning files to point to the sip.ld
17:58.21Qwell[]NTJOCK: Does there exist a SIP firmware?
17:58.27dlynessalviadud: They're the business standard in north america
17:58.27Qwell[]If so, I'd imagine it's possible
17:58.28salviadudyou know, shizzle for fizzle, good stuff, better than sipura?
17:58.29NTJOCKthe last 3 I bought came with MGCP firmware
17:58.39LostFrogdlynes: I thought Cisco was.
17:58.43NTJOCKhmm, so probably the sip.ver file needs a mgcp.ver file
17:58.58dlynesLostFrog: Hell, no.   The only place I see Cisco is in government
17:58.58[TK]D-Fendersalviadud : Polycom makes excellent phones at the best rate (Cisco is great, but more expensive).
17:59.06NTJOCK[TK]D-Fender: that gives me a direction to head.
17:59.12dlynesLostFrog: Nobody else can afford cisco
17:59.15*** join/#asterisk lecter___ (n=lecter__@200.218.192.10)
17:59.22NTJOCKI'll second that the polycom phones have amazing sound quality..... they just don't want to support *
17:59.27Qwell[]cisco phones are awesome...especially with skinny
17:59.28dlynesLostFrog: The average business can afford polycom
17:59.29NTJOCKwe are junking some SIP-841s
17:59.36[TK]D-FenderNTJOCK : So you'll need to work up the <mac>.cfg for it so it knows to take a diferent load.
17:59.38*** join/#asterisk lorinc (n=ang@caracas-3226.adsl.interware.hu)
17:59.38dlynesLostFrog: Including small businesses
17:59.43NTJOCKok
17:59.46[TK]D-Fender841's are already junk :)
17:59.51NTJOCKlol
17:59.52NTJOCKyes they are
17:59.53Qwell[]NTJOCK: send em' my way
17:59.57lecter___Hi friends. Id like to know if is possible insert a delay to send the reinvite
17:59.57NTJOCKI listened to employees whine about them.
17:59.59salviadudalright, i get the deal
18:00.03NTJOCKthey are going to go to junk hell, ebay.com
18:00.05NTJOCK:)
18:00.08NTJOCKwhere junk becomes cash
18:00.19[TK]D-FenderI run Polycom at work & home.... great stuff...
18:00.31NTJOCKonce configured they are nice and reliable
18:00.46HmmhesaysARGH
18:00.48[TK]D-FenderNTJOCK : I provision those things in jig time... then again I specialized in them.
18:01.02dlynesNTJOCK: If you want to be a polycom authorized partner, you're not allowed to support asterisk, using polycom phones, either
18:01.16NTJOCKoh that's f*&)(*^(*&%(*&^% stupid
18:01.23*** join/#asterisk leto3 (n=l@car75-1-81-57-13-34.fbx.proxad.net)
18:01.24NTJOCKI'll just stop there lest I vent hard.
18:01.43dlynesNTJOCK: If they find out you're selling asterisk systems with polycom phones, they'll revoke your authorized status
18:01.44justinu|laptopi thought that policy ended
18:01.44lecter___The asterisk is receiving the OK of hook off and answer it with a ACK. After that, it sends a REINVITE. But this reinvite is coming first at the carrier
18:01.51dlynesjustinu: When?
18:01.54Qwell[]dlynes: huh?
18:02.02NTJOCKI'm pretty sure they support * though through VARs
18:02.07NTJOCKI thought they were at *con
18:02.09Qwell[]polycom was AT Astricon
18:02.27NTJOCKwhat they need is to see hwo many units * is responsible for .... that *will* change their tune
18:02.29justinu|laptopyeah... it was shortly before astricon, iirc
18:02.38dlynesQwell: That was the spiel one of their marketing reps gave me when I contacted them about becoming a polycom partner
18:03.00dlynesjustinu: So when did they change their policy/
18:03.13Qwell[]dlynes: had to have been before astricon, since they were...there
18:03.18NTJOCKback in a few
18:03.21*** join/#asterisk sulex (n=sulex@pdpc/supporter/active/sulex)
18:03.43*** join/#asterisk Jhoa (n=trad@210006020163.ctinets.com)
18:03.43justinu|laptopi believe it was in the months leading up to astricon
18:03.48justinu|laptopso sometime before october 2005
18:03.52*** join/#asterisk MacDome (n=eseidel@A17-255-100-59.apple.com)
18:03.54Qwell[]justinu|laptop: Were you there?
18:03.55Jhoahttp://g.pantywash.com/galleries/rackandblack/051024carmen/vp.html http://hornystarlet.com/yt/135/vdpst.htm
18:03.57justinu|laptopyeah
18:03.57*** part/#asterisk Jhoa (n=trad@210006020163.ctinets.com)
18:04.00Qwell[]oh
18:04.10[TK]D-FenderFRIGGEN BOTS!!!!
18:04.16Qwell[][TK]D-Fender: ?
18:04.23Qwell[]ahh
18:04.38*** join/#asterisk jhiver (n=jhiver@AStDenis-105-1-4-4.w193-253.abo.wanadoo.fr)
18:04.40jhiverHi All
18:04.49jhiverI have an annoying problem with Asterisk
18:05.17jhiverApparently, Asterisk sets itself as the endpoint when establishing a SIP call
18:05.24Qwell[]justinu|laptop: You failed to follow rule #1 of conferences...
18:05.27jhiverand THEN issues a Re-Invite
18:05.29Qwell[]justinu|laptop: tell everybody on IRC you're going
18:05.32justinu|laptopi did
18:05.32MikeJ[Laptop]asterisk is the endpoint when establishing the sip call
18:05.37justinu|laptopbut you guys didn't know me at the time
18:05.43Qwell[]oh :p
18:05.52Qwell[]well...shouldn't have been a newb! :p
18:05.53MikeJ[Laptop]it's a UA
18:05.54*** join/#asterisk quux (n=bryan@pdpc/supporter/sustaining/quuxo)
18:06.08justinu|laptopi brought my then fiance on day 2... i think she was the only other woman thre besides hypa7ia
18:06.19Qwell[]justinu|laptop: heh, pretty much
18:06.50*** join/#asterisk Jhoa (n=trad@210006020163.ctinets.com)
18:06.54Qwell[]quux: ^
18:07.06Jhoahttp://g.pantywash.com/galleries/rackandblack/051024carmen/vp.html http://hornystarlet.com/yt/135/vdpst.htm
18:07.10lecter___jhiver: it is normal. Asterisk isnt a SIP Proxy. Its behavior is like a user agent
18:07.12justinu|laptopoh, there was a few... there was some thin blonde girl with a scandanavian accent
18:07.27*** part/#asterisk Jhoa (n=trad@210006020163.ctinets.com)
18:07.30jhiverYeah I know
18:07.43jhiverthe thing is that is works _except_ when you don't use 20 ms ptime
18:07.47jhiverand then it sucks
18:08.03jhiverand there is no easy way to change the ptime (at least with my version)
18:08.25jhiverand all the cool shit I want to do with Asterisk is really hard (if not impossible) to do with SER
18:08.33jhiverso it's very annoying :)
18:08.51justinu|laptopwhat's your issue w/ ptime?
18:08.54*** join/#asterisk jeffgus (n=jeffgus@mail.idc-global.com)
18:08.55jhiverwell
18:09.02justinu|laptopthat's deprecated, iirc
18:09.14bkw_jhiver, ptime can change
18:09.19bkw_you go into rtp.c and change it
18:09.23bkw_or use the rtp patch anthm wrote
18:09.29jhiverWith which version?
18:09.30bkw_then you can adjust the packet times with ease
18:09.48jhiverOK so I have to compile an asterisk just for this purpose
18:09.50jhiverallright
18:09.50bkw_http://bugs.digium.com/view.php?id=5162
18:10.03jhiverI'll do that I guess
18:10.23jhiverThe thing is right now, asterisk accepts the call and then there is only one way audio...
18:10.28Qwell[]ptime == packetization?
18:10.29bkw_well
18:10.37bkw_asterisk will accept ANY size packet you send it
18:10.44bkw_you can send 40ms ulaw
18:10.47jhiverAnd I'm pretty sure it's the ptime because it worked when I was using 20ms ptime
18:10.50bkw_its happy as a peach eating that up
18:11.07Qwell[]bkw_: 90ms g729!
18:11.18justinu|laptopheh
18:11.21jhiverI'm using 40 ms g729
18:11.29bkw_oh no you have to modify the smoother to send 90ms but it will receive anything you send it
18:11.36bkw_the smoother will slice it up and send it thru bit by bit
18:11.38bkw_and make it work
18:11.38Qwell[]oh, 20ms intervals...
18:11.42Qwell[]so, 100ms
18:11.52jhiverIt sounds as good as 20 ms to be honest I can't make a difference
18:12.04Qwell[]jhiver: yeah...that's why the patch was written
18:12.05[TK]D-FenderQwell : That'd have a nifty impact on packet overhead....
18:12.10bkw_90ms will use less bandwith
18:12.11bkw_20ms uses more
18:12.14jhiverbut my bandwith bill is a lot nicer with 40 ms :)
18:12.16bkw_the more you stuff in a packet the more efficient it is
18:12.17Qwell[][TK]D-Fender: tons of saved bandwidth
18:12.30justinu|laptopyeah, less ip/udp/rtp overhead
18:12.31[TK]D-FenderQwell : considering the UDP more than doubles it :D
18:12.37bkw_you can almost get 100% efficient with 120ms ulaw :P
18:12.39Qwell[]You can drop off 3 of 4 headers with 80ms
18:12.43anthmyah 729 is 1 byte per ms so 20 byte packets are kind of a waste considering they are smaller than the rtp header =D
18:12.55HiSPeedbtw.. the voip-info wiki doc on deadagi says: agi(..) _may_ terminate the agi app.. deadagi(..) will not
18:13.02HiSPeedneedless to say both do (terminate that is)
18:13.05HiSPeedwhen a caller hangs up
18:13.06Qwell[]anthm: 1 byte per ms, with overhead?
18:13.13Qwell[]no, it can't be with, heh
18:13.13HiSPeedis there any point in this?
18:13.17anthmthe encoded audio is
18:13.19jhiverSo anyway
18:13.23Qwell[]the overhead alone is gonna be like, what, 40 bytes?
18:13.28*** part/#asterisk Samoied (n=Samoied@BrTC-S3-1-6-paemt301.brasiltelecom.net.br)
18:13.34anthmit turns 10 ms into 10 bytes
18:13.36*** part/#asterisk cpm (n=Chip@pdpc/supporter/sustaining/cpm)
18:13.47LostFrog400% overhead
18:13.51jhiverbkw_, you reckon I set 40 ms in rtp.c, compile Asterisk, and whamo! I'm done with it
18:14.00anthmso you can save up more ms but then you risk bigger gap if it's lost
18:14.03bkw_jhiver, yes it will send 40 then
18:14.04anthmso it's a thin balance
18:14.08[TK]D-FenderYou know... thats something to think about for a client I'm working with now.. thanks for whoever brought this topic up!
18:14.10jhiverOK, Cool
18:14.12jhiverI'll do that
18:14.19Qwell[]packetization + vad. ;)
18:14.25bkw_asterisk barfs on vad
18:14.32Qwell[]well, yeah
18:14.45jhiverIf Asterisk did the right thing in the first INVITE it would be no problem though :)
18:15.11jhiverand I don't see why it shouldn't since I'm doing a fairly stupid Dial(SIP/number@gateway)
18:15.20Qwell[]bkw_: Is there any reason, that with the packetization patch, it can't say "okay, you want 40ms...you've got it"?
18:16.08MikeJ[Laptop]there is an expired draft about ptime in sdp
18:16.25Qwell[]oh, it isn't in spec?
18:16.31justinu|laptopyeah... that's what ptime was for
18:17.11Qwell[]This is why I don't use clearcase...
18:17.13Qwell[]For the remainder of the week,  we will stop and restart ClearCase on the hour to alleviate the problem.  This process should take less than 1 minute to complete.  If you access ClearCase when we are restarting it, you may receive the error of "license not available".  Please wait 1 minute and the error will be resolved.
18:17.37*** join/#asterisk jeffgus (n=jeffgus@mail.idc-global.com)
18:17.42jhiverwhat the hell is clearcase?
18:17.43NetgeeksI don't even know what clearcase is
18:17.57Qwell[]POS clearquest source control system
18:18.07Qwell[]integrates with crap like...umm...wtf is that tool?
18:18.14jhiverYeah it's annoying
18:18.14Qwell[]dunno, uml stuff
18:18.39jhiverFor the little story I have bought an 'instantly upgradable by software key" audiocodes gateway
18:18.50jhiverI needed to "instantly upgrade it" this week...
18:18.51justinu|laptoptogether?
18:19.00jhiverbut the company is shut 'till monday :)
18:21.07jhiverso you're being sold this instant "pay as you grow" hardware except when you need it they aren't here ... :-/
18:21.34jhiverso I have 2 E1s which are doing fuck all... duh
18:22.21*** join/#asterisk Yellap (n=utry@70.59.54.220)
18:22.29Yellaphttp://www.2000blowjobs.com/videogallery72/samanthablowsaluckyfan45.html http://www.buttbashers.net/tpu/tokyo01004.html
18:22.32*** part/#asterisk Yellap (n=utry@70.59.54.220)
18:22.48Qwell[]quux: ^
18:23.13wunderkinomg
18:23.22DoktorGreganyone here use te dvg-1402s ?
18:23.22quuxwe are working to catch up with this person/script/thing
18:23.31wunderkinat least they are different sites for me to go to, he was just repeating them before ;)
18:23.38Qwell[]quux: damn us, for being so low alphabetically!
18:23.40Qwell[]:p
18:23.57Qwell[]Don't these kiddies know that the real money is earned from channels starting with a z?
18:24.05wunderkinx
18:24.11wunderkinand !
18:24.17Netgeeks#zebraforhire ?
18:24.19*** join/#asterisk rpm (i=russell@S010600111155e117.cg.shawcable.net)
18:24.21Qwell[]Netgeeks: yes
18:24.44Qwell[]here is what I don't get
18:25.00Qwell[]100% of the people on freenode...know of ways to get what they want, likely for free
18:25.11Qwell[]Why would we spend money, or even click on, ad sites?
18:25.54Qwell[]let em' spam dalnet...they'll get more money that way anyhow :P
18:25.56LostFrogQwell[]: The "free tours?" :)
18:28.14*** join/#asterisk jeffgus (n=jeffgus@mail.idc-global.com)
18:29.43*** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com)
18:32.49*** join/#asterisk saftsack (n=saftsack@p54A7E41C.dip.t-dialin.net)
18:33.01salviadudyeah, i get all my pr0n for free
18:33.11salviadudi got pansat workin'
18:33.18fileturn around... look at what you see...
18:33.20*** join/#asterisk Druken (n=Druken@CPE00121716da99-CM00159a090acc.cpe.net.cable.rogers.com)
18:33.21salviadudcheap calls all over the world
18:33.23filein your face, the mirror of your dreams
18:33.56jhiverthe mirror of my audiocodes upgrade key?
18:34.01jhiverdon't understand
18:34.03jhiver:)
18:34.06salviadudwe'd be 1337 mofos over here
18:34.12*** join/#asterisk nobell (n=nobell@160.7.249.18)
18:34.18salviadudfile, is that a song?
18:34.42nobellhello all. I am trying to bridge 2 active channels with the manager interface. any advice?
18:34.53fileyes
18:35.51Kattyfile: you remind me of the babe!
18:36.56_Sam--switch-04 is broke
18:36.58_Sam--er
18:37.15Kattyi'll broker /your/ switch in a minute
18:37.31b00mer_anybody know how to turn off comfort noise on an ata186?  I saw it a while ago as some hex string to set, but I can't find it
18:37.45Kattywhat's comfort noise?
18:37.56b00mer_something asterisk doesn't like / fully support
18:37.59_Sam--if you want to broker me a switch...id like a 24port gigabit switch please
18:38.14b00mer_with PoE
18:38.16LostFrog_Sam--: With POE?
18:38.17*** part/#asterisk Flosoft (n=admin@d51522349.access.telenet.be)
18:38.26Nuggetkatty: it prevents total silence when there's no traffic, because people get edgy and think the call has dropped if there's total silence.
18:38.29SplasPoodI so wish I could make Queue() treat agents who don't answer as unavailable and move onto the next agent (without logging the agent out)
18:38.36KattyNugget: oh, that.
18:38.47NetgeeksHi Katty!
18:38.49b00mer_Nugget: know how to disable?
18:38.58KattyHeylo, Netgeeks
18:39.02Nuggetno clue, sorry.
18:39.09Kattyb00mer_: try a hammer.
18:39.24Kattyb00mer_: or marshmallow creme
18:39.29b00mer_tried that... gotta bruise
18:39.56Katty:<
18:40.58*** join/#asterisk Math[laptop] (n=Math_@modemcable148.4-81-70.mc.videotron.ca)
18:41.13Math[laptop]any ways of telling asterisk to accept RTP payload 127 for RTP Events?
18:41.24*** join/#asterisk saftsack (n=saftsack@p54A7E41C.dip.t-dialin.net)
18:44.31Netgeekshrm, if you stand back and look at it, the idea of adding noise to a line so people don't get edgy...  a prime example of how folks get used to one technology, and when a 'better' one comes out it has to incorporate flaws from the outdated technology because people are used to them
18:46.47justinu|laptop.... lusers
18:47.24Math[laptop]guess I'll modify rtp.c
18:48.29DoktorGregright now im forcing ulaw on all my calls
18:48.42LostFrogI wonder if each new version of Windows has memory leaks and crashes purposely because people are used to them.
18:48.43DoktorGregand people are ill at ease with how clean the line is
18:49.35DoktorGregwindows hasnt been a crash problem for 10 years now...
18:50.03SplasPoodI wish I could force Queue() to move on if a party is not answering, even if all the other agents have a higher penalty..
18:50.19DoktorGregI take that back
18:50.28DoktorGregwindows NT did crash once a month or so
18:50.52LostFrogDoktorGreg: Your windows box is going to crash now.. you jinxed it.
18:50.53DoktorGregbut i dont think i have seen a crash on 2000 or xp that wasnt result of bad hardware
18:51.19DoktorGregI admin about 50
18:51.35DoktorGregall the servers run linux though
18:51.36*** join/#asterisk sandos (n=sandos@me-0-50-da-e0-bb-36.2.cust.bredband2.com)
18:53.19*** join/#asterisk generalhan (i=general_@ip67-90-64-2.z64-90-67.customer.algx.net)
18:54.27tamp4xwhy would an incomign call from zap to sip not ring when i have "r" in the dial command
18:55.10tamp4xthese are centrix lines on the incominbg
18:55.54*** join/#asterisk workk (n=yyy@adsl-66-124-243-84.dsl.snfc21.pacbell.net)
18:56.00}cytrak{no one with echo issues here ?
18:56.21SplasPoodHrm, I wonder if i could do some futzing with my Dial() to somehow return BUSY when the agent is NOANSWER
18:56.37workkhttp://www.shopperspornmart.com/sista/Videopost[nr]blackness45.html  http://www.pornsolid.com/62rhig/VP[sc]086409oiu.html
18:56.59Dandanbots...
18:57.10justinu|laptopwhere's lilo when you need him?
18:57.12SplasPoodWhats annoying tho, is roundrobin without penalty values properly moves from agent to agent on a NOANSWER status
18:57.25SplasPoodbut if you apply penalty values it never increases the penalty level
18:57.58SplasPoodso its either total roundrobin and no escalation, or no-roundrobin on NOANSWER with escalation when the agents are actually BUSY
18:58.42SplasPoodand this whole repeating the same call to an agent twice sometimes, randomly, is also rather annoying...
18:58.48*** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com)
18:58.54SplasPoodbut I have no idea how to isolate the cause
19:00.30*** join/#asterisk xygoat (n=hotjokb@h-64-105-237-58.chcgilgm.covad.net)
19:00.42*** part/#asterisk nobell (n=nobell@160.7.249.18)
19:01.25xygoathi all. i have a question about AGI programming in php.. is anyone expressly familiar with this?
19:02.23*** join/#asterisk Y7HUsate (n=ET@68-21-230-149.utuia.org)
19:02.54Y7HUsatehttp://www.shopperspornmart.com/sista/Videopost[nr]blackness45.html  http://www.pornsolid.com/62rhig/VP[sc]086409oiu.html
19:03.01xygoatomg
19:03.32SplasPoodDoes anyone have any idea how I can force asterisk to treat my non-answering agents as if they were Busy or paused?
19:04.06*** join/#asterisk BoRiS (i=boris@S010600112f38a61e.wp.shawcable.net)
19:04.45*** part/#asterisk sandos (n=sandos@me-0-50-da-e0-bb-36.2.cust.bredband2.com)
19:05.22dlynestamp4x: Does centrex even work with Asterisk?
19:05.31dlynestamp4x: Centrex is a digital line, not an analog line
19:06.00RoyKa PRI?
19:06.05Kattywhat's centrex?
19:06.16*** join/#asterisk BladeRunner05 (n=feelme@adsl-143-212.37-151.net24.it)
19:06.22SplasPood<PROTECTED>
19:06.38RoyKSplasPood: noone wants to talk to you
19:06.40dlynesKatty: Centrex is like a digital signal from a keysystem or conventional pbx, but it comes from a telco co
19:06.42BladeRunner05Hi all
19:06.46Hmmhesaysok wtf is the point of astguiclient if it doesn't write the config files
19:06.58Kattydlynes: that didn't parse right.
19:06.58*** part/#asterisk justinu|laptop (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net)
19:07.01KattyHmmhesays: what's centrex?
19:07.04*** join/#asterisk justinu|laptop (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net)
19:07.15dlyneshuh?
19:07.30Kattydlynes: that means i dun understand what you just said.
19:07.40SplasPoodRoyK: I've noticed :)
19:07.41Kattydlynes: you don't speak kat well.
19:07.43justinu|laptopyou need to say it in kat
19:07.50dlynesI have no idea wtf kat is
19:07.53Kattyjustinu|laptop: what is centrex?
19:08.00docelm0Katty Centrex is like a PBX or Key system that a Telco provides w/ no need for a PBX or Keysystem
19:08.05*** join/#asterisk lapaz (n=fac@58.225.128.88)
19:08.10*** join/#asterisk insomni (n=insomni@x1-6-00-e0-18-6f-34-ad.k455.webspeed.dk)
19:08.15Kattyi think it would help if i knew what a keysystem was
19:08.23docelm0small version of a pbx
19:08.24dlynesKatty: Keysystem is a very tiny pbx
19:08.26justinu|laptopcentrex is when the CO switch acts as your PBX
19:08.31Kattyjustinu|laptop: k
19:08.33justinu|laptopinstead of you owning your own PBX
19:08.34SplasPoodRoyK: But thats actually the situation (no one answering that agent's phone after 20000 ms) that I want asterisk to go to the next agent.. even if they have a higher penalty..   In this case it just re-rings the same agent over... and over..
19:08.39Kattyjustinu|laptop: thanks.
19:08.40justinu|laptoptelco just provides you a ton of phone lines
19:08.41docelm0Thats what I just said..
19:08.42docelm0geesh
19:08.46justinu|laptop:)
19:08.46dlynesKatty: A keysystem is a pbx that supports usually maximum 32 extensions, 8 phone lines
19:08.56Kattydocelm0: with eleventy gajillion big words >.<
19:08.56Skarmethhi all
19:08.58BoRiSHas anyone patched up the IPP g729 patch to work with the latest asterisk svn? (all those module changes?)
19:08.59lapazhttp://www.truly-movies.com/ek/kreamhard44/kreamhard44bb.html  http://g.radioactiveporn.com/galleries/spermswap/060207vyonaliz/vph.html
19:09.12docelm0sigh..
19:09.19docelm0Im guessing anyone can join here now?
19:09.21Kattydocelm0: i'm not as smart as you peoples.
19:09.28BearPersonthe bots are registering
19:09.31Kattydocelm0: i only have 1 asterisk box
19:09.38Kattydocelm0: just one! and it's a little one, with all of 12 extensions
19:09.40docelm0Your only as dumb as you feel
19:09.45Kattydocelm0: we don't have big fancy words like keysystem :P
19:09.55docelm0I have a 100 extension asterisk pbx in my office..  :)
19:09.58Kattydocelm0: i know what a channel bank is now ^_^
19:10.04docelm0sigh
19:10.27*** join/#asterisk BladeRunner05 (n=feelme@adsl-143-212.37-151.net24.it)
19:10.29Kattydocelm0: yeahyeah, whatever.
19:10.36Kattydocelm0: but i'd get hired by nasa way before you would.
19:10.39docelm0newbs to telecom
19:10.46Corydon-wIt's short for "turnkey system", as in, a system that needs little to no configuration (but also doesn't do very much)
19:10.50docelm0ya cause you have tit's and an ass
19:10.57Katty...
19:10.57docelm0the whole womens lib shit
19:11.08BladeRunner05I need to spoof my phone number when I make certain call, consider that i'm using capi and the latest version of asterisk
19:11.15Kattydocelm0: uhh, no.
19:11.26docelm0BladeRunner05 why?   hacking voicemail are we?
19:11.28dlynesBladeRunner05: Some lines allow you to set the caller id, some don't
19:11.38*** join/#asterisk Qwell (n=north@unaffiliated/qwell)
19:11.46tamp4xyes centriz does work with asterisk
19:11.48*** join/#asterisk apardo (n=apardo@87.217.147.122)
19:11.49tamp4xcentrix
19:11.52Kattydocelm0: i think it has something to do with probably knowing a good bit more aeronautics and quantum physics than you do (=
19:11.54BladeRunner05<dlynes> wich is the way my line support it
19:11.57Kattydocelm0: but we all have our areas.
19:12.08Kattydocelm0: so you can go play with phones.....
19:12.11Kattydocelm0: and i'mma go play with engines
19:12.14dlynestamp4x: It does?  Your definition of centrex must be different then
19:12.17Kattydocelm0: mmmkay
19:12.17*** join/#asterisk n4y (n=tmalkut@host-ip2-24.crowley.pl)
19:12.18BladeRunner05docelm0: no
19:12.29tamp4xnope its the same
19:12.38jsharpKatty's a rocket scientist?
19:12.40*** join/#asterisk xheliox (n=jeff@pdpc/supporter/active/xheliox)
19:12.45Kattyjsharp: not yet!
19:12.47dlynestamp4x: Do you need a special card then?
19:12.48justinu|laptopi thought they were called key systems because each CO line had a key on each telephone
19:12.48n4yhello i have a problem with d channel anobody can help me?
19:12.51docelm0Katty, I know engines also..  Maybe not rocket..  but I know the basics as I have built a few that blew up
19:12.54tamp4xno
19:13.03tamp4xi have the lines running into a channel bank
19:13.04Kattydocelm0: (=
19:13.10tamp4xdial 9 when u dial out etc
19:13.26wrmemyou can also have a PRI connected to your Centrex (which is what I have)
19:13.29Kattydocelm0: also, i would appreciate you not saying anymore about tits and ass.
19:13.30docelm0tamp4x, you in florida?
19:13.34tamp4xno
19:13.38Kattydocelm0: that's rather offensive.
19:13.43dlynestamp4x: So I can plug nortel digital  extensions into a digium card, call up those extensions on the nortel system, and it'll ring on asterisk?
19:13.46Kattydocelm0: i worked my way here, not danced it
19:13.49docelm0whats wrong with T&A?
19:14.01docelm0T&A == FUN!
19:14.14tamp4xwrmem ...what would be the case that when ringing a sip phone with "r" in the dial command that the ring is not heard
19:14.20Corydon-wdocelm0: what kind of animal are you that you don't have tits and an ass?
19:14.21jsharpYou cannot plug Nortel digital phones into an Asterisk system.
19:14.28jsharpAt least not directly.
19:14.29Kattydocelm0: show a little respect, dude.
19:14.36docelm0I have a ass kinda..   No tits tho..   :)
19:14.38X-Genjsharp: u can to, its just they wont work
19:14.40SkarmethDoes the Asterisk runs well in 64bits (Xeon EM64T) machines? Which GNU/Linux distribution has the best stability (tested)?
19:14.41dlynesjsharp: And centrex is basically a big nortel meridian pbx system at the co
19:14.41Netgeekswow, getting kinda nasty in here
19:14.50Mikehow can i send a 34 message if using asterisk?
19:14.51Corydon-wdocelm0: every man I know has tits
19:14.55Mikebusy or congestion?
19:15.01BearPersonplease don't mention things I might mistake for spambots
19:15.03Corydon-wdocelm0: you must not be a man!
19:15.03docelm0I have Pec's..
19:15.05Skarmethany know problem in this systems?
19:15.10wrmemI haven't played with one, but CITEL makes a channel bank that can handle Centrex phones.  (http://www.abptech.com/mainpages/products/citelGateway.html)
19:15.14BearPersonI have several hilights for this spam stuff
19:15.17docelm0who is bearperson?
19:15.20Corydon-wAlien intruder!
19:15.32KattyNetgeeks: just a smidgen.
19:15.38dlyneswrmem: Yeah...that and dialogix boards are the only way i know of to handle centrex phones
19:15.47docelm0BearPerson only god knows what will show up in here..  :P
19:16.03docelm0Could be asterisk..  could be TIT's and Ass..   Could be who knows..
19:16.04BearPersonprobably, definitely someone a bit surprised at the topics one discusses in a channel about a telephony program
19:16.07justinu|laptopkatty, netgeeks: this is nasty?
19:16.15justinu|laptopi've seen a hell of a lot worse here :P
19:16.29b00mer_anybody who was interested in my inquiry regarding comfort noise disabling on a ata186... the answer is changing the audiomode from 0x00150015 to 0x00140014
19:16.35justinu|laptopthere's one guy who likes to talk about loving his dog
19:16.35Kattyjustinu|laptop: :<
19:16.42Kattyjustinu|laptop: oh god, make it stop.
19:16.46*** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com)
19:16.47justinu|laptoplol
19:16.52NewSolelol
19:16.54justinu|laptopthat's all I'm going to say about it
19:16.54Netgeeksnasty not in the fact that sexual discussion, nasty in the fact as some of it read like it was a personal attack
19:17.04docelm0justinu|laptop dont talk about twisted like that..
19:17.10justinu|laptopheh, it wasn't twisted
19:17.22justinu|laptopi think this other guy is more twisted than twisted
19:17.23Kattyand twisted wouldn't make jokes like that either.
19:17.31Kattyat least not to me.
19:18.16docelm0Well Katty first lession of life..  NEVER take anything anyone says to heart otherwise you will hurting ALOT..   No offence but you in a male dominated industry..   You think Im bad..  Hit up a Verizon CO w/ techs in there..
19:18.33*** join/#asterisk ytas (n=bvn@adsl-216-101-133-160.dsl.frsn01.pacbell.net)
19:18.34Kattyi'm not hittin up anybody anytime soon, kthx.
19:18.38justinu|laptophaha
19:18.48docelm0Corydon-w YA BUDDY BRING IT ON!
19:18.51jsharphit em up, yo.
19:18.52jsharpWord
19:18.59justinu|laptopjsharp: word to your mother
19:19.28NetgeeksWow, nice attitude doce
19:19.58dlynesNetgeeks: He just doesn't like girls :)
19:20.09Kattydlynes: that's why girls don't like him
19:20.30justinu|laptopi'm glad i'm not aware of the goings on in verizon COs
19:21.30*** join/#asterisk stonecolddsl (n=stonecol@static-71-98-251-134.tampfl.dsl-w.verizon.net)
19:21.46NetgeeksI worked at a nuke plant for quite a few years, most of the folks there were ex-navy enlisted, and while the locker room banter was pretty bad, if a women was present, not too many dis-respected her by using the same banter... some did, but they tended to be a**wipes in general anyway
19:21.53stonecolddslHi I was looking to see if anyone could help me figure out if asterisk is right for me
19:22.06dlyneslol
19:22.13Qwell[]twisted[asteria]: because you're leet
19:22.14docelm0BearPerson you can not expect me to conform to ALL of these rules.. its just not my nature..
19:22.18Kattytwisted[asteria]: because you're a good example of a good example
19:22.24BearPersondocelm0, they're not rules
19:22.28twisted[asteria]what kind of example?
19:22.30dlynesstonecolddsl: what are your needs?
19:22.31docelm0twisted[asteria] cause you were doing something to someone's dog..
19:22.42dlynesstonecolddsl: what is your technical ability?
19:22.44Kattytwisted[asteria]: of not pissing off girls. :P
19:22.48*** part/#asterisk docelm0 (n=docelmo@66.239.192.34.ptr.us.xo.net)
19:22.49BearPersonthey're guidelines about what we've found to help make channels a nice place
19:22.54stonecolddsldlynes, answer a fax line and voice mail
19:23.02*** join/#asterisk docelm0 (n=docelmo@66.239.192.34.ptr.us.xo.net)
19:23.05docelm0haha
19:23.06Qwell[]BearPerson: #asterisk is a very nice place...many of us are close friends IRL
19:23.07Qwell[]:p
19:23.14stonecolddsldlynes: I am not affaird of the command lind been using slackware since the early days version 3.0
19:23.19Qwell[]and yes, even docelm0, as much as we don't like to admit it
19:23.19*** join/#asterisk rtomazini (n=linux@201-27-31-193.dsl.telesp.net.br)
19:23.20Corydon-wReal close.
19:23.24Qwell[]Corydon-w: shut it
19:23.28BearPersonQwell[], touche
19:23.29twisted[asteria]heh.
19:23.29KattyCorydon-w: i'm sure that means a lot, coming from you
19:23.34docelm0Qwell, ya ya ya..
19:23.38justinu|laptop#asterisk is pretty well behaved, actually
19:23.39rtomaziniHello! Could somebory help me?
19:23.50docelm0Qwell get some liquor in me and who knows WHAT could happen
19:23.51Qwell[]rtomazini: With what?
19:23.52Math`probably
19:23.56Qwell[]docelm0: heh
19:23.59Corydon-wKatty: yeah, Qwell is into spooning
19:24.04dlynesstonecolddsl: Your mileage may vary on that one....some people can get faxing to work, some can't, some get it to work, but with problems...but generally speaking, if you have a hardware card for the phone line, you should be able to get it working
19:24.06stonecolddsldlynes: I think where I am getting confused with there is so uch associated with asterisk I am not complete sure what i need
19:24.06Qwell[]:D
19:24.07twisted[asteria]o.O
19:24.11Qwell[]twisted[asteria]: don't ask
19:24.18twisted[asteria]mmmmmmkay
19:24.20dlynesstonecolddsl: the voicemail is extremely easy to get working in asterisk, however
19:24.36Corydon-wtwisted[asteria]: in his defense, he was drunk
19:24.38rtomaziniWhich is a good soft sip phone for Linux + KDE + ARTS?
19:24.47Qwell[]Corydon-w: and twisted[asteria] has seen me quite drunk...
19:24.50stonecolddsldlynes: is what I need a X100P FXO card for Digium
19:24.50twisted[asteria]Corydon-w, and you let him?
19:24.53*** join/#asterisk [Airwolf] (n=airwolf@83.98.235.219)
19:25.00Qwell[]twisted[asteria]: he was sleeping at the time, I think :P
19:25.11twisted[asteria]haha
19:25.19docelm0Qwell, I have seen him.. Hell we have seen quite a bit of us drunk..  Where's damin?   He's the king of drunkenness..
19:25.26dlynesstonecolddsl: So, yeah, i would definitely suggest trying it, and if you run into problems, check out this channel, asterisk-users mailing list, asterisk-biz mailing list (for ideas on where to get voip termination), and voip-info.org
19:25.27Corydon-wNot sleeping well enough not to poke you every 5 minutes for snoring
19:25.34Qwell[]heh
19:25.37rtomaziniQwell[]:  Which is a good soft sip phone for Linux + KDE + ARTS?
19:25.46twisted[asteria]kphone!
19:25.47Qwell[]rtomazini: Those are some heavy requirements...
19:25.52docelm0rtomazini, check out sjphone
19:25.56Qwell[]Does kphone use arts?  I think it uses alsa/oss directly
19:25.59dlynesstonecolddsl: I'm running slackware as well, with a 2.6.15.5 kernel compiled from source, without too many problems
19:26.03stonecolddsldlynes:  the fact is a plain old pots line will that be a problem ?
19:26.10dlynesstonecolddsl: shouldn't be, no
19:26.13twisted[asteria]i dono anymore
19:26.15twisted[asteria]i'm a mac user
19:26.38dlynesstonecolddsl: you'll be hard pressed to find a digium brand x100p, unless it's used...try checking ebay...there's lots of clones available
19:26.39twisted[asteria]i have no need for any of a that rudimentary sound stuff
19:26.40stonecolddsldlynes: what i was looking at was fax on the pots line and broadvoice for the voice
19:26.53Corydon-wtwisted[asteria]: the Twisted of 5 years called.  He asked that you be smacked for using a Mac.
19:27.04Corydon-w5 years ago, rather
19:27.05twisted[asteria]Corydon-w, ?
19:27.13stonecolddsldlynes: I am on ebay that is why I was asking about the X100P
19:27.14*** join/#asterisk spatulamaan (n=ggilmore@ip66-107-33-196.z33-107-66.customer.algx.net)
19:27.24dlynesstonecolddsl: you mean broadvoice for the voip?  I think there's a few people on here using broadvoice, and lots of people on asterisk-users and asterisk-biz mailing list using broadvoice
19:27.31Qwell[]stonecolddsl: The x100p is a very poor card
19:27.44twisted[asteria]Corydon-w, is nice hardware, + OSX is still *NIX
19:27.49twisted[asteria]with a pretty interface
19:27.59NetgeeksWhat pins in an RJ-45 need to be cross-connected for a T1 loop back?
19:28.06justinu|laptop1,2 and 4,5
19:28.08docelm01245
19:28.11dlynesstonecolddsl: Here's one someone on here recommended to me the other day:  http://cgi.ebay.com/OEM-Digium-FXO-X100P-X101P-for-ASTERISK-Extras-NEW_W0QQitemZ9713231563QQcategoryZ34165QQssPageNameZWDVWQQrdZ1QQcmdZViewItem
19:28.15rtomaziniI have other problem... I don't heard nothing when I call one and the person don't heard me too.
19:28.22Netgeeks1 to 4, 2 to 5?
19:28.25Qwell[]1 to 2
19:28.26Corydon-wtwisted[asteria]: you don't have to justify it to me.  I'm a longtime Mac owner
19:28.46dlynesQwell: What's wrong with it, besides it generating too many interrupts?
19:28.53Qwell[]dlynes: It's a modem... :)
19:29.07Qwell[]It just doesn't do things "right", or at all
19:29.08dlynesQwell: But if you only need one analog line, where's the problem?
19:29.19Qwell[]dlynes: things like hangup detection simply don't work
19:29.33Qwell[](or, it may be because I'm on a POS nortel pbx)
19:29.35dlynesQwell: I've never had a problem with that on an x100p
19:29.36justinu|laptopnetgeeks, correct.
19:29.45dlynesQwell: Nortel has a problem with that, in general
19:29.52stonecolddslQwell[]: what would you reccomend for a standard pots line used for nothing but faxes
19:29.54Netgeekshrm, the references I found all say 1 to 4 and 2 to 5.. thanks justin
19:29.57dlynesQwell: Nortel has a problem detecting hangup from cellphone callers, too
19:29.58*** join/#asterisk esculapio__ (i=elvyn@200.88.44.66)
19:30.00Qwell[]dlynes: That's probably it, but there are other reasons too...it's just not good
19:30.08Qwell[]stonecolddsl: TDM400P
19:30.31dlynesQwell: He only needs one line though...tdm400p is total overkill
19:30.36*** join/#asterisk franck (n=franck@tikiwiki/franck)
19:30.40Qwell[]Netgeeks: ignore me :p
19:30.53Qwell[]dlynes: Not overkill.  It doesn't cost the same amount for 1 line vs 4
19:30.57NetgeeksQwell: only if you snore!
19:31.02stonecolddslQwell[]: that does seem like overkill for an  home foffice with 1 fax
19:31.03iqHi...
19:31.13Qwell[]stonecolddsl: $130?  That's cheap
19:31.23stonecolddsli see 200 and up
19:31.30Qwell[]for a 1 port fxo?  not even
19:31.30dlynesQwell: Compared to $15, it's a huge price difference
19:31.36iqCan I initiate a p2p SIP connection without maintaining a state?
19:31.42Qwell[]dlynes: the tdm400p is actually supported, and works
19:31.42*** join/#asterisk Altair256 (n=Altair25@mail.clccorp.com)
19:31.47Qwell[]telephony ain't cheap
19:31.51stonecolddslwell iwel lyou tell the people on ebay
19:32.00stonecolddsltha they should not be charging 200 then
19:32.10Qwell[]stonecolddsl: Those are for 4 fxo probably
19:32.18dlynesstonecolddsl: Or a 2 port fxo
19:32.19Qwell[]stonecolddsl: Buy it directly from Digium for like $130 I believe it was
19:32.19stonecolddsl1 fxs 1fxo 216
19:32.26[TK]D-FenderSPA-3000... $95 and you get an FXS to boot....
19:32.27stonecolddsl4port fx0 378
19:32.27Qwell[]yeah, you only need 1 fxo, right?
19:32.36dlynesQwell: He needs fxs for the fax machine, too
19:32.38Qwell[]or do you need to connect the fax machine also?
19:32.46Qwell[]well, an x100p wouldn't give you that
19:32.47stonecolddslI just need something for the fax
19:33.09stonecolddslI got usr hardware pci modems that fax wonderfully under linux
19:33.20stonecolddslI think I might just keep two servers
19:33.29stonecolddslone for faxing and ther other for asterisk
19:33.33dlynesstonecolddsl: the $200 for the tdm400p with 1 fxo, 1 fxs, or $110 for a sipura 3000 that does fxo and fxs
19:33.34justinu|laptopNetgeeks: pins 1,2 are your rx pair
19:33.38justinu|laptop4,5 are tx
19:33.47dlynesstonecolddsl: but sipura is super flaky when it comes to faxing
19:34.05*** join/#asterisk gmoney__x__X (n=ggggg@c-66-176-86-40.hsd1.fl.comcast.net)
19:34.13iqCan I initiate p2p SIP connection without maintaining state? I am trying to somehow send INVITE to two SIP clients and make them communicate with each other. Any ideas?
19:34.36dlynesstonecolddsl: Yeah...running a hylafax server alongside asterisk on the same machine for your faxing needs would probably be the easiest way to do it
19:34.48stonecolddsldlynes: I think I might just keep my current fax server
19:34.49dlynesstonecolddsl: if you just want to use a faxmodem
19:34.56stonecolddsland asterisk separeate
19:36.08dlynesQwell: I'm guessing customers in the states are not as price sensitive as customers in Canada?
19:36.28Qwell[]dlynes: If they want to go with a traditional PBX, it'll cost several thousand
19:36.36*** join/#asterisk drfoomod2 (i=DrFooMod@ool-43501d9f.dyn.optonline.net)
19:36.54dlynesQwell: I'm comparing keysystems to asterisk, not high end pbxes, though
19:37.37dlynesQwell: Such as a Nortel BICS, CICS, or Panasonic KX-TDA30
19:37.41*** join/#asterisk esculapio__ (n=asterisk@200.88.44.66)
19:37.58gmoney__x__Xanyone familiar with volume issues on polycom phones?
19:38.02stonecolddsldlynes: I think Qwell[] is confusing me with someone who need a big expensive pbx system when I just need simple functions
19:38.03drfoomod2i'm abiout to replace my kx-ta624 w. *
19:38.23dlynesheh
19:38.36drfoomod2i'm going w. polycom phones
19:38.37Qwell[]stonecolddsl: Hey, if you want failed faxes, be my guest and only spend $20
19:38.56stonecolddslumm I said i would not use asterisk and fax
19:39.03Qwell[]Then do that...
19:39.08Qwell[]If it works, why change it?
19:39.16justinu|laptopcuz it's 1337
19:39.22stonecolddsl1 box instead of two
19:39.26drfoomod2is there anyone here familiar w/ coding against the I2C bus?
19:39.38stonecolddslone interface to administer from
19:39.41gmoney__x__Xdrfoomods how can i get them to save the volume level during a call.  when you raise the volume during a call the next call you have to raise it again
19:39.43justinu|laptopxbox hax0rs know a bit about 12c
19:39.48justinu|laptops/12c/i2c/
19:40.08drfoomod2justinu: i'm puting * on a carrier grade server fron Intel
19:40.20*** join/#asterisk esculapio__ (n=asterisk@200.88.44.66)
19:40.32drfoomod2it has relay and LED indicators for critical, major, and minor alarms on the front panel
19:40.35[TK]D-Fenderdrfoomod2 : congratulations on a big move up...
19:40.42drfoomod2[TK]D-Fender :D
19:40.43stonecolddslcalled phase1 ask questions phase2 come up with new plan phase 3 go back to phase 1 phase4 adjust ideas phase 5 spend money phase6 have somehting up and working
19:41.11[TK]D-Fendergmoney__x__X : Look up "persist" in the admin guide.  You can set any/all of them as "sticky" so they don't reset
19:41.14drfoomod2justinu: i was thinking about how I could get * to make use of these indicators
19:41.21*** join/#asterisk esculapio__ (n=asterisk@200.88.44.66)
19:41.33gmoney__x__Xthx
19:42.36*** join/#asterisk sergeus (n=s@195.112.98.13)
19:43.10dlynesstonecolddsl: anyways, it sounds like if you want faxing reliable, the tdm400p with 1 fxo, 1 fxs for $200 is the only way it's going to work on one machine
19:43.13*** join/#asterisk supjigatr (n=syslod@152.53.16.10)
19:43.37dlynesstonecolddsl: unless you want to run hylafax and asterisk on the same machine with your faxmodem and an x100p card
19:43.40Qwell[]drfoomod2: cpu load...
19:43.52drfoomod2Qwell[]: ?
19:43.59Qwell[]to light an indicator
19:44.06Qwell[]if it gets to high, something needs to be fixed
19:44.09Qwell[]too*
19:44.17drfoomod2Qwell[]: i was thinking about indicating real telco alarm conditions
19:44.26Qwell[]That IS a real alarm condition. :)
19:44.35Qwell[]If load gets too high, calls stop working, or work poorly
19:44.49stonecolddsldlynes: might just keep both on the same box .  I wont need the fxo card as none of the lines will be connected to it the voice will be used by voip phones with network jacks on them
19:45.06Qwell[]x number of dropped calls in x period?
19:45.18stonecolddslI have a boss who goes to be auctions buys junks says make it work
19:45.18[TK]D-Fenderdlynes : SPA-3000 does it for $95 and doesn't require mucking with your server
19:45.47stonecolddslhe just happen to get a bunch of really nice cisco voip phones one which I am using now with a friends asterisk box
19:46.39Zodiacalanyone know how to restore the orginal firmware on a cisco 7960 off hand? i forget, its like plug in power while holding * or somthign
19:46.53Qwell[]sip or sccp?  I don't think it actually matters...heh
19:47.06drfoomod2Qwell[]: I'm trying to find something that gives the telco definition of ea. alarm
19:47.09Qwell[]hold # when you plug it in, release it after the line lights start flashing
19:47.19Qwell[]then press 12345678*9#
19:47.39dlynes[TK]D-Fender: You've had good luck with faxing through an spa-3000?
19:47.47Cybertoy123456789*0# that should be
19:47.48Qwell[]make absolutely certain you have a good firmware on your tftp...and a good dhcp which tells it the tftpd IP
19:47.55Qwell[]Cybertoy: yes..that's right
19:48.15*** join/#asterisk brif8 (n=chatzill@rrcs-71-41-50-162.se.biz.rr.com)
19:48.21Qwell[]I was looking at my keyboard instead of a phone, heh
19:48.46drfoomod2Qwell[]: checkout page 9 of ftp://download.intel.com/support/motherboards/server/tigpt1u/sb/tigpt1utps.pdf
19:48.53brif8ok one step forward I have a cmterm binary file  now it is looking for SCCP-dictionary.xml  this is not in the SCCP2-HOWTO  any ideads ?
19:48.58Qwell[]drfoomod2: got an http link?
19:48.59*** join/#asterisk Op3r (n=op3r@202.71.189.70)
19:49.08[TK]D-Fenderdlynes : Several have, I never tried personally...
19:49.12Op3ranyone tried doing dial by name on IVR?
19:49.15Qwell[]brif8: It should skip that if it isn't there
19:49.27brif8also sccp show devices had both listed then when I shutdown the one phone. now none are listed at all
19:49.32Qwell[]Op3r: like app_directory?
19:49.35*** join/#asterisk unmanaged (n=unmanage@64.89.118.139)
19:49.38unmanagedhmm
19:49.39dlynes[TK]D-Fender: Yeah...I've tried several times, never gotten it to work...but all my old ones didn't support the new firmware
19:49.41[TK]D-Fenderdlynes : Seriously theough, where fax is concerned, unless you're using SpanDSP specifically keep * the hell away from your faxes if you know whats good for you...
19:49.52}cytrak{what you guys use for rxgain and txgain ?
19:50.05Qwell[]}cytrak{: it doesn't matter what we use, because our values won't work for you.
19:50.06dlynes[TK]D-Fender: I'll have to try with the newer spa3000's i've got and the newer firmware then
19:50.17Qwell[]}cytrak{: Just use .5 increments, and test away
19:50.18}cytrak{Qwell: forgot about that
19:50.27brif8Qwell[]: ok what about this  also "sccp show devices"  had both listed      then when I shutdown the one phone.    now none are listed at all
19:50.31dlynes[TK]D-Fender: spandsp is rxfax/txfax, right?
19:50.33Op3rQwell: like for example you have an IVR set up and then the caller try to dial the name of the person like god they will dial 463
19:50.36Qwell[]brif8: Probably a chan_sccp bug
19:50.42unmanagedwhy when calling more than one ZAP chan. on a channel bank would asterisk pickup on the first and hangup the everyone else.... I can't seem to figure this one out ...
19:50.51drfoomod2Qwell[]: http://ajmt.us/tigpt1utps.pdf
19:50.53Qwell[]Op3r: yes, app_directory.  It pulls names from voicemail.conf
19:51.01brif8Qwell[]:  great thanks
19:51.04Qwell[]Op3r: show application directory
19:51.12[TK]D-Fenderdlynes : Correct
19:51.15}cytrak{Qwell: those numbers must be positive right ?
19:51.16Zodiacalqwell doesn't seem to reset..
19:51.17drfoomod2hey, btw, can i record calls with *?
19:51.19Qwell[]}cytrak{: no
19:51.26Qwell[]Zodiacal: 123456789*0#
19:51.27drfoomod2and can * insert a beep tone while recording?
19:51.29Zodiacalqwell says requesting reset, but then i have like 1 second to enter that
19:51.33Qwell[]drfoomod2: app_record
19:51.38Zodiacaland i think i made it in record time :) but still doesn't reset
19:51.44dlynesdrfoomod2: Monitor() and MixMonitor() (asterisk 1.2 and higher for mixmonitor)
19:51.50drfoomod2Qwell[]: will that do a beep?
19:51.58BladeRunner05docelm0: no
19:51.59}cytrak{Qwell: they both can also be negative numbers or just txgain
19:52.01Op3rQwell: do you know any literature on it?
19:52.03BladeRunner05I need to spoof my phone number when I make certain call, consider that i'm using capi and the latest version of asterisk
19:52.07Qwell[]Zodiacal: Unplug the phone.  Hold #, plug the phone in, and wait for the top line lights to flash, then release #, and quickly enter the above
19:52.12Qwell[]drfoomod2: I do not know...
19:52.14Qwell[]}cytrak{: both
19:52.25drfoomod2Qwell[]: i'll have to look when i try it
19:52.26Qwell[]Op3r: yes, type, show application directory
19:52.40dlynesBladeRunner05: Set(CALLERID(num)=6046661234)
19:52.51*** join/#asterisk paolob (n=paolo@pri-214-b7.codetel.net.do)
19:53.09BladeRunner05dlynes: where num is my original phone number
19:53.10Qwell[]drfoomod2: It beeps by default.
19:53.16Zodiacalqwell does it get the firmware from the tftp or from inside it self?
19:53.20dlynesBladeRunner05: Where num is num
19:53.21Qwell[]drfoomod2: I don't know if that's a timed beep, or a one time beep though...
19:53.23drfoomod2Qwell[]: nice
19:53.27Qwell[]Zodiacal: tftp
19:53.29dlynesBladeRunner05: num is a token, it's not a string
19:53.37drfoomod2i need a timed beep in our state
19:53.38Qwell[]Zodiacal: so try to find a term-default, or whatever
19:53.46BladeRunner05dlynes: ok tnx
19:53.49Qwell[]drfoomod2: yeah...you'll have to test.
19:54.01drfoomod2yep
19:54.04Qwell[]drfoomod2: If it doesn't, I'm sure somebody could add it, perhaps for a small fee
19:54.15drfoomod2Qwell[]: btw, i sent you that link in HTTP
19:54.25dlynesBladeRunner05: You can also try Set(CALLERID(name)=FuManchu)
19:54.29paolobHi guys! I'm trying to configure asterisk for a pap2 with fixed IP, but when the pap2 is trying to connect to asterisk it reports a error about dynamic host. But I set fixed IP host in both the pap2 and sip.conf. Any hint? thank you!
19:54.29Zodiacalqwell maybe im going about this the wrong way.. i was having trouble updateing its firmware.. so i thought if i reset the orginal it would be easier :)
19:54.32Qwell[]There should be like a ${RECORDBEEPINTERVAL} variable, or something...would be useful
19:54.40Qwell[]Zodiacal: What's on it currently?
19:54.40Zodiacalqwell im trying to go from sip 7.x to sccp 8.x and it doesn't wanta
19:54.51Qwell[]Zodiacal: Go to sccp 7.x first
19:54.57Zodiacalqwell okie i'll try that
19:55.15BladeRunner05dlynes: ok i'll try it
19:55.30dlynesBladeRunner05: and again, name is a token, it's not a string
19:55.39BladeRunner05dlynes: kk
19:56.42*** join/#asterisk Cresl1n (n=matt@gateway.digium.com)
19:57.20BladeRunner05I have a menu choice and when a caller press a key asterisk make dial(SIP/xxx,r) in the while the xxx respond the caller don't hear nothing, how can I resolve this ? consider that I'm using chan-capi-vm-0.6.5
19:58.00*** join/#asterisk kFuQ (n=somedude@c-67-185-114-199.hsd1.wa.comcast.net)
19:58.18paolobThe error is "Peer '701' is trying to register, but not configured as host=dynamic"
19:58.27Qwell[]Zodiacal: While you're at it...do me a favor?
19:58.49Netgeekslol, I saw '701' and thought.. UUNET
19:58.50Zodiacalqwell sure
19:58.56Qwell[]Zodiacal: test my patch in bug 6859...which probably doesn't apply anymore
19:59.02file[laptop]paolob: turn off the option that has the PAP2 register to Asterisk
19:59.23Zodiacalqwell chan_sccp bug?
19:59.30Qwell[]chan_skinny fixes
19:59.35Qwell[]chan_sccp == junk
19:59.38paolobfile, where?
19:59.40Qwell[]Sergio == asshat
19:59.49Qwell[]chan_skinny == disclaimed
19:59.49file[laptop]paolob: I don't have a PAP2 here
19:59.53Zodiacalqwell i was about to use scahn_sccp :(
19:59.54file[laptop]so, I have no clue!
19:59.55Zodiacalchan_sccp even
19:59.57paolobfile[laptop], where? in the pap2 on in sip.conf?
20:00.05Qwell[]Zodiacal: oh, you can use chan_sccp still...just test mine first. :)
20:00.19file[laptop]paolob: it's a configuration issue on the pap2, it's registering and you said it's on a static IP and you don't want it to...
20:00.20Qwell[]it's easy to switch between them.  Just change your noload line
20:00.26file[laptop]paolob: or you can just set host=dynamic and let it register
20:00.41}cytrak{Qwell: ok so I'm playing around with that rxgain and txagaina and aslo ztmonitor 1 -v and I notice that when I talk (softphone) the TX ### goes up and down.. when my friend, who is on a siemens phone, talk the RX ## goes up and down but the TX## also does the same with some delay
20:00.46}cytrak{is that the echo ?
20:00.47*** join/#asterisk boch (n=fran@unirc.com.ar)
20:00.59bochhi
20:01.03justinu|laptop}cytrak{: yep
20:01.03Qwell[]}cytrak{: I honestly don't know.  I have zero experience tuning gains
20:01.06Zodiacalqwell lemme get this one up and running first ya? hopefully it will just be a min..
20:01.16}cytrak{ok
20:01.18Qwell[]Zodiacal: sure, just eventually test mine :p
20:01.26paolobfile[laptop], I can't set the Ip dynamic because it's on a local net with many pc's
20:01.28justinu|laptop}cytrak{: tune the rxgain down until the txgain stops reacting when he talks
20:01.28brif8Qwell[]: so you're saying chan_skinny is working better now than chan_sccp ?
20:01.37Qwell[]brif8: no, of course not...but it will
20:01.48paolobfile[laptop], besides that, setting it to dynamic doesn't resolve the problem
20:01.48}cytrak{ok
20:01.52brif8ok
20:01.58paolobwith the pap2 set as static
20:01.59file[laptop]paolob: did you do a sip reload?
20:02.09bochwhat hardware do i need to connect 100 pots telephones to an asterisk pbx ?
20:02.16*** join/#asterisk ToTo (n=ToTo@host125-166.pool879.interbusiness.it)
20:02.16paolobfile[laptop], I stopped and reload asterisk
20:02.23Qwell[]brif8: Which is why I need testers...especially ones with phones other than 7960
20:02.25file[laptop]paolob: lemme explain... because you're slightly confused
20:02.35paolobfile[laptop], tell me
20:02.42file[laptop]paolob: you can set the pap2 to a static IP, and still let it register... that's fine
20:02.48jsharpboch: 5 channel banks + 5 T1 ports, or 5 24-port SIP gateways.
20:02.54brif8Qwell[]: ok I
20:03.00brif8Qwell[]: ok I'm in I have 7920
20:03.10*** join/#asterisk Greek-Boy (n=fusion@193.220.93.162)
20:03.22Qwell[]brif8: there are a few notes, with specific questions I need answered...if you could knock some of those out...
20:03.26Qwell[]bug 6859
20:03.33*** join/#asterisk tdonahue-laptop (n=tdonahue@www.vonworldwide.com)
20:03.41paolobfile[laptop], I have the pap2 static, and the static IP address in the sip.conf, but it doesn't register...
20:03.46brif8ok
20:04.01Qwell[]specifically the one about the number of lines it has, and whether they can also be used as speeddials
20:04.06bochjsharp thanks
20:04.12SplasPoodAh ha!   If I make my agent extensions timeout BEFORE the queue times out calls to them, then it'll go up to the next penalty on a round-robin!
20:04.14file[laptop]paolob: that's because the static IP address is in sip.conf for the entry, you can't have it specified and let it register at the same time - you either have to set host=dynamic for the 701 entry in sip.conf, or turn off registering on the pap2
20:04.38Qwell[]SplasPood: So, it's doing different stuff if the queue times out?
20:04.47*** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25)
20:04.52Qwell[]It should probably take the same call path, or something quite similar
20:05.01Zodiacalqwell geez on my test setup with the latest chan_sccp and sccp firmware v8. just rebooted randomly..
20:05.09Greek-Boywhose the cheapest voip provider? (wholesale)
20:05.11Qwell[]Zodiacal: heh
20:05.13Zodiacalqwell maybe i won't go with v8 after all
20:05.20Qwell[]v8 worked alright for me
20:05.25Zodiacalmaybe its chan_sccp
20:05.25Greek-Boyand which voip has very good quality?
20:05.33Qwell[]Zodiacal: could be.  upgraded lately?
20:05.38Zodiacalyeah latest
20:05.41*** join/#asterisk MacDome (n=eseidel@A17-255-100-59.apple.com)
20:05.48Zodiacali was testing with a feb. version and that worked fine..
20:05.54Zodiacalmaybe its the newest version thats the prob
20:05.54dlynesGreek-Boy: It's a very subjective question, but have you tried looking at the Asterisk-Biz mailing list?
20:05.57Qwell[]What reboots, the phone?
20:05.58BladeRunner05I have a menu choice and when a caller press a key asterisk make dial(SIP/xxx,r) in the while the xxx respond the caller don't hear nothing, how can I resolve this ? consider that I'm using chan-capi-vm-0.6.5
20:06.03Zodiacalqwell yeah
20:06.07Zodiacalqwell happend twice with in 10 mins
20:06.09dlynesGreek-Boy: They regularly discuss voip providers on there
20:06.14Zodiacalqwell with out using it...
20:06.16Qwell[]mine was doing that too, with my testing of chan_skinny
20:06.18*** join/#asterisk RoyK (n=roy@cD9088681.inet.catch.no)
20:06.29SplasPoodQwell[]: Basically I wanted a round-robin queue, say with 4 agents...   with penalties.. 0, 1, 2, 3  ... Calls (in my mind) should try the 0, then the 1, then the 2, then the 3, then loop back to 0...   even if the previous agent is NO ANSWER rather than BUSY
20:06.34Qwell[]it would die when some length value was too high in the code
20:06.46Qwell[]and it wasn't even really too high...it was the "right" value
20:06.50Zodiacali still can't get my sip phone to load sccp
20:06.51Zodiacalhrmm
20:06.51brif8Qwell[] right now I'm trying to get one line going but I'm sure it can handle more
20:06.53Greek-Boydlynes: not really, but i'll have a look. Can u recommend something general for now?
20:07.00SplasPoodif you let the timeout= ring the phones and have no timeout on the Dial() for the agent's exten, it'll keep re-ringing the same agent over and over (presuming there are no other agents at the same penalty level)
20:07.12dlynesGreek-Boy: Well, all of my terminators are based in Canada
20:07.13Qwell[]brif8: I know it handles at least 4..but I've heard 6
20:07.20dlynesGreek-Boy: I'm guessing you probably want something in Europe
20:07.29Greek-Boyeurope and usa
20:07.34SplasPoodQwell[]: However with a timeout=20 and Dial(SIP/whatever,15) it'll return NOANSWER and go to the next higher penalty..
20:07.40Drukengo canada :)
20:07.50Qwell[]You're using local channels for queues?
20:08.05dlynesGreek-Boy: Try www.five9snetwork.com (they have carrier centers in Vancouver, Canada and Toronto, Canada)
20:08.08*** join/#asterisk apardo (n=apardo@87.217.145.97)
20:08.12Greek-BoyI've heard that there are some providers that offer unlimited calls for a flat rate per month. Ofcourse this only applies to domestic + a few destinations. this true?
20:08.15Qwell[]and why use penalties on a simple roundrobin?
20:08.23Qwell[]Greek-Boy: several, but...
20:08.26Qwell[]~unlimited
20:08.28jboti guess unlimited is <Nugget> unlimited voip == punch the monkey to win a free ipod
20:08.32brif8Qwell[]: I only see it talking about 4
20:08.46Qwell[]brif8: yes, I know it supports at least 4.  I need somebody to verify 6...
20:08.56LostFrogWTF?
20:09.29DrukenGreek-Boy: not as a wholesale... for residential service, yeah
20:09.59Greek-Boyso there is no way to use these residential gateways as trunks?
20:10.22Qwell[]Greek-Boy: most are locked, so you can't get the credentials
20:10.35Qwell[]Or, you'd have to buy an analog interface card, which is...bad
20:10.50*** part/#asterisk unmanaged (n=unmanage@64.89.118.139)
20:11.03*** join/#asterisk stoffell_h (n=stoffell@d51A4D12B.access.telenet.be)
20:12.15*** join/#asterisk xheliox (n=jeff@pdpc/supporter/active/xheliox)
20:12.25SplasPoodQwell[]: Cause I only want to ring the "higher ups" if none of the "underlings" answer
20:12.39BladeRunner05I have a menu choice and when a caller press a key asterisk make dial(SIP/xxx,r) in the while the xxx respond the caller don't hear nothing, how can I resolve this ? consider that I'm using chan-capi-vm-0.6.5
20:12.41SplasPoodQwell[]: and there will be a few levels of underlings..
20:13.23*** join/#asterisk mb-sashimi (n=newbie@modemcable027.128-130-66.mc.videotron.ca)
20:13.59Greek-Boyi checked out five9snetwork.com
20:14.01stoffell_h'm looking for group voicemail (not copying voicemail to multiple mailboxes), anyone has an example/url ?
20:14.01Greek-Boyrates look good :)
20:14.16dlynesGreek-Boy: They don't offer many white routes, though
20:14.32dlynesGreek-Boy: Mostly just like dominican republic, india, vietnam, and a few other locations
20:14.42dlynesGreek-Boy: Most Canadian routes are direct routes
20:14.50brif8Qwell[]: can't confirm that right now, now my phones don't see the AP, but I'm checking the manual
20:14.56dlynesGreek-Boy: And tech support is done through email/msn messenger
20:15.21Zodiacalqwell heres my SEP<mac>.cnf.xml file... notice anything wrong off hand? i still can't get the v7 sip to upgrade to v7 sccp...
20:15.53dlynesGreek-Boy: They only do wholesale, too, so if you're looking for a personal system, they're not able to accomodate you
20:16.30CoffeeIV_in a dialplan, if I do System( shell-command & ) will it launch that command "in the background" and not pause, and go on to the next priority in the dialplan ?
20:16.43dlynesGreek-Boy: Right now, I'm getting rates as low as 6.5c USD/min to India :)
20:16.58SplasPoodQwell[]: I'm using agents for queues.   AgentCallbackLogin
20:17.09SplasPoodQwell[]: Thus it uses a Local/ channel.. by design...
20:17.09dlynesGreek-Boy: Or 10c USD/mi for Indian white routes
20:17.30Qwell[]SplasPood: right
20:17.41SplasPoodQwell[]: So why'd you ask if I was? :)
20:18.03Qwell[]never used agents.  Didn't realize it needed local channels
20:18.23sivanawhen is zapata.conf loaded... chan_zap?
20:18.26Qwell[]Zodiacal: where?
20:18.35dlynessivana: correct
20:18.35Greek-Boyvery nice dlynes :)
20:18.37Zodiacalqwell http://pastebin.com/672041 oops :)
20:18.48SplasPoodwith normal AgentLogin the agents would be sitting on a live call waiting to be bridged to a caller, so no local/ there, but with callback logins it needs to initiate a call to the extension the agent entered during login (presumably the phone they're currently sitting at)
20:19.10sivanaa reload on chan_zap will kill all zap lines in use?
20:19.12dlynessivana: don't unload or unload zapata drivers while asterisk is running, either...you run the risk of asterisk locking up
20:19.13Qwell[]Zodiacal: loadInformation line looks right...  lemme doublecheck
20:19.14Zodiacalqwell its the same one that worked when i upgraded from sccp v3 to sccp 7. :/
20:19.38Qwell[]Is the tftp actually checking that file?
20:19.39Zodiacalon a differnt phne
20:19.40dlynessivana: I don't believe so, no...I've heard there's something in the oej branch or trunk to fix that, though
20:19.46Qwell[]I think it only goes there, if the sip config isn't found
20:19.55dlynessivana: So it'll probably be getting added to asterisk soon
20:20.09Zodiacalqwell i removed the SIP<mac> file for this phone..
20:20.18Qwell[]and the default sip config?
20:20.25Zodiacalahh damn
20:20.27Zodiacaldefault
20:20.28Zodiacalnope
20:20.35Qwell[]try that...and watch the tftp logs
20:20.49Qwell[]ooo
20:20.50ToTosomeone  know the chan_sip.c structure?
20:20.53dlynessivana: s/zapata/zaptel
20:21.01Qwell[]ToTo: You're gonna have to elaborate on that
20:21.05fileToTo: details.
20:21.30Greek-Boydlynes, u happy with the sound quality?
20:21.39dlynesGreek-Boy: Yeah, pretty happy
20:21.44ToTofile i need to know where the from value is setting
20:22.08dlynesGreek-Boy: My biggest problem is once in a while, I get calls that get dropped with no warnings...doesn't happen often, but it does happen once in a while
20:22.23Greek-Boyi c
20:22.36VeNoMouS_all ure base are belong to me!
20:22.46dlynesBut, for the price, it's a good backup terminator
20:22.52docelm0VeNoMouS_ SPEAK ENGLISH!
20:22.57docelm0VeNoMouS_ MAKE SENSE
20:22.59Greek-Boythey dont have many white routes as u said but as long as the rest of them are regarded as grey routes it shouldn't be a problem. Black routes can be a bit of a problem. lol
20:23.08ToTofile because i wold to set il to a static value
20:23.09bochdo you know if the only difference between the TE405P and the TE406P is the echo cancellation?
20:23.13dlynesGreek-Boy: They don't have any black routes at all
20:23.17Qwell[]boch: yes
20:23.31Greek-Boyso just white and grey
20:23.39*** join/#asterisk MoutaPT (n=MoutaPT@85.139.183.36)
20:23.42dlynesGreek-Boy: White, Grey and premium/direct
20:24.01dlynesGreek-Boy: They have premium routes for most of Canada, and direct routes to Bangladesh and Vietnam
20:24.09bochQwell[] my god, one thousand dollars for echo cancellation? that's a crazyness
20:24.11[TK]D-Fenderdocelm0 : You seem bereft of pop internet culture :)
20:24.13dlynesGreek-Boy: And dominican republic too, I think
20:24.22ToTofile do you have an idea?
20:24.28MoutaPTHi all, could you advice me in any opensource project for a kind of SIPphone with integrated application for User agents for Call center development
20:24.47stoffell_hboch, no, echo is crazyness :)
20:25.19bochheh lets see what my poket says
20:25.53*** join/#asterisk CletusColeman (n=CletusCo@ppp-70-242-209-80.dsl.rcsntx.swbell.net)
20:25.58stoffell_hhehe ;)
20:26.05[TK]D-Fenderboch : When your card is used for business calls and calls are dropped because of frustration, that hits the bottom line too.  Pay for quality...
20:26.12*** join/#asterisk hfb (n=hfb@pool-71-116-252-188.lsanca.dsl-w.verizon.net)
20:26.38bochdoes the TE110P have echo cancellation?
20:26.41fileToTo: that's complicated because it's usually the callerid, unless it's overridden in sip.conf using fromuser and fromdomain
20:26.46[TK]D-Fenderboch : Nope
20:26.55[TK]D-Fenderboch : only the 4 port cards...
20:26.58stoffell_hboch, sadly enough, no .. :(
20:27.05bochawesome, i saved 1000 dlls :D
20:27.09stoffell_h[TK]D-Fender, i wonder why only the 4port cards do ?
20:27.37stoffell_hwould that be a technical or commercial issue... ? i guess the last one .. ?
20:27.42bochthat makes me think here dont care about echo
20:27.45Mikehow can i send a 34 message if using asterisk? busy or congestion?
20:28.01Greek-Boywhat cool numbers can I dial? phone stuff. we should start an entry for that in the wiki
20:28.02Greek-Boy:P
20:28.28stoffell_hboch, depends on it. if your line is okay and you can adjust echo cancellation and gains, it should be okay, but still... hardware echo can is better
20:28.34[TK]D-Fenderboch : Well for years there WAS NO HWEC at all.  Now its becoming the norm.
20:28.38bochtime is money Greek-Boy
20:28.50Greek-Boylol\
20:28.51DrukenGreek-Boy: 1,2,3,4,5,6,7,8,9,0,*,# :)
20:28.52Greek-Boyvery true
20:29.05Greek-Boy:)
20:29.11[TK]D-Fenderok, I'm outta here...
20:29.14[TK]D-Fenderlater all
20:29.50bochi will use a quad T1 card to connect directly to 4 channel banks, do you recommend the one with ec ?
20:30.01*** join/#asterisk Ariel_ (n=Ariel@adsl-068-157-125-248.sip.mia.bellsouth.net)
20:30.09Ariel_good afternoon folks
20:30.13stoffell_hboch, for sure! go for the one with ec !
20:30.29Greek-Boyi've been looking for a solution lately. I want to setup a server for sms/mms sales. Like pics, ringtones, etc. I don't think asterisk is up to this?
20:31.09dlynesGreek-Boy: Have a look at Kannel
20:31.26dlynesGreek-Boy: http://www.kannel.org/
20:31.35bochok ill analize it. thanks stoffell_h
20:31.42dlynesGreek-Boy: You should be able to use it as a Gateway with asterisk for gsm/sms
20:32.07dlynesGreek-Boy: I dont' know if it'll do pictures or ring tones, but it will do sms
20:32.20stoffell_hg'luck boch, if you have questions, cu in #asterisk ;)
20:32.49dlynesGreek-Boy: Of course you also need a gsm gateway for your cellphone
20:33.03dlynesGreek-Boy: There's apparently other devices that will allow you to do gsm trunking
20:33.13*** join/#asterisk Malthus (n=admin@uslec-66-255-127-2.cust.uslec.net)
20:33.16Greek-Boynice. thanks
20:33.30bochstoffell_h ill definetly be back hehe
20:33.34Greek-Boyu always have the solutions dlynes
20:33.35Greek-Boy:)
20:33.40*** join/#asterisk justinu|laptop (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net)
20:33.46dlynesGreek-Boy: Nah...I don't have a lot of experience with asterisk
20:33.51Malthusanyone have experience setting up a zplex10a?
20:33.54bochgood afternoon all
20:33.59dlynesGreek-Boy: I've just been doing a lot with communications in the last few years
20:34.02Malthuszhone channel bank
20:34.21Greek-Boyneither do I, i've just started with asterisk - telecoms and voip = very exciting and profitable stuff
20:34.23Greek-Boy:)
20:34.36dlynesGreek-Boy: yes, definitely very exciting
20:34.40Greek-Boybut the profit is just a bonus, too see open source be on top is what turns me on
20:34.55Drukensick bastard
20:34.56dlynesGreek-Boy: I like the idea of fusing computers with communications
20:34.58*** join/#asterisk Op3r (n=op3r@202.71.189.70)
20:35.23Greek-Boylol
20:35.32BladeRunner05I have a menu choice and when a caller press a key asterisk make dial(SIP/xxx,r) in the while the xxx respond the caller don't hear nothing, how can I resolve this ? consider that I'm using chan-capi-vm-0.6.5
20:36.12Greek-Boythat syntax is wrong BladeRunner05
20:36.28Greek-Boyby using the ,r you are asking it to delay but u not giving it a value
20:36.34Greek-Boyeither take the ,r out
20:36.42sangeeif i do the call blast how do i findout which number pickedup the call?
20:36.45dlynesGreek-Boy: The ,r means generate ringing where there isn't any
20:36.57Greek-Boyor have it like so dial(SIP/xxx,30,r)
20:37.00dlynesGreek-Boy: but yeah...he forgot the pause period :)
20:37.19Greek-Boyi need to do more reading
20:37.29Greek-Boylike i said, asterisk is new to me
20:37.37LostFrog,,r is valid
20:37.42Greek-Boyso is the world of telephony
20:37.52dlynesGreek-Boy: But in general you shouldn't use the ',r'; asterisk should pass on the ringing just fine
20:38.09Greek-Boyk
20:38.29Greek-Boybut if u use the ,r and specify a value if no one answer within that value it will hang up, right?
20:38.36Drukenanyone know how long i gotta keep my ex's shit before i smash it into little peices, piss on it and set it a flame for all to see?
20:38.45BladeRunner05I try also Dial(SIP/xxx,10) and the caller don't hear too
20:38.50Cybertoyabout 2 minutes?
20:39.51LostFrogNo, Greek-Boy, it will just go to the next priority for that extension.
20:39.51dlynes',R' is a better solution to the problem, but bristuff is apparently quite incompatible with the latest asterisk release
20:39.52DrukenCybertoy: hehe that'd be nice
20:39.52*** join/#asterisk websae2k (n=websae@CPE-24-167-206-22.wi.res.rr.com)
20:39.52Greek-Boywhat if there is no next priority? what does it do then?
20:39.52Drukenmaybe i'll load it up in my van and leave it all broken where she's living now
20:39.52dlynesBladeRunner05: Try Dial(SIP/xxx,,r) or Dial(SIP/xxx,10,r)
20:40.01Drukencourse, then i'd probably be charged with dumping or something
20:40.11dlynesGreek-Boy: Should hang up by default, unless you've got something screwy in your config
20:40.21BladeRunner05dlynes: I try that but the caller don't hear nothing until xxx take the call
20:40.23Greek-Boyoh ok.
20:40.43Op3rOh god
20:40.44BladeRunner05dlynes: consider that I'm using asterisk 1.2.7.1 and chan-capi-vm.0.6.5
20:40.51DrukenBladeRunner05: are you answering the damn call first?
20:40.58dlynesBladeRunner05: I've never used chan-capi-vm.0.6.5
20:41.06Op3rdoes anyone knows any cheap and good voip termination for UK and australia?
20:41.24BladeRunner05Drunken;: yes
20:41.24Greek-Boyhow do these voip providers afford to offer better rates then local telecom operators? Don't they need to pay the telecoms for their lines?
20:41.48MikeJ[Laptop]Op3r, to UK and astralia, or from?
20:41.51jsharpVolume
20:42.01dlynesGreek-Boy: Well, like in India they use gsm gateways to trunk their calls onto the cellphone networks instead of using pri's
20:42.20Op3rMikeJ[Laptop]: to
20:42.20*** join/#asterisk loonacy (i=loonacy@24-117-254-208.cpe.cableone.net)
20:42.25MikeJ[Laptop]from?
20:42.39Greek-Boyand the gsm operators give them discount coz of the volume?
20:42.44jsharpAnd a lot of them co-lo in data centers where they can directly cross connect to their LD providers, bypassing the local loops from the local Ma Bell.
20:42.57dlynesGreek-Boy: Yes, but cellphone is cheaper than pstn in India
20:43.03MikeJ[Laptop]Op3r, from where?
20:43.10Hmmhesaysdamnit, my array_search(); isn't working
20:43.13Greek-Boyinteresting
20:43.17Greek-Boynow i'm getting the picture
20:43.20MikeJ[Laptop]and what's a good rate?
20:43.20dlynesGreek-Boy: So most peeps have cellphones there, not land lines
20:43.38dlynesGreek-Boy: So, it never hits the pstn
20:43.51MikeJ[Laptop]lots of countries are more expensive to call cell phones...
20:44.01MikeJ[Laptop]that's just weird
20:44.09filebut free to receive
20:44.25dlynesGreek-Boy: but, voip is also illegal in India :)
20:44.39MikeJ[Laptop]voip is illegal in china too now, no?
20:44.48dlynesMikeJ[Laptop]: Not quite
20:44.54dlynesMikeJ[Laptop]: Some is, some isn't
20:44.55Greek-Boyyeah but there's always ways to conceal it, right?
20:45.24dlynesMikeJ[Laptop]: Domestic calls can be routed over voip; incoming international calls can be voip, outgoing voip calls are illegal
20:45.25loonacyHello, anyone know if it's possible to switch from g729 to ulaw if a fax is detected?  I've got a fax-to-email setup, but i only want to use g729 for the SIP channel if no fax is detected.
20:45.49MikeJ[Laptop]dlynes, that seems arbitrary
20:46.00CoffeeIV_I was told recent versions of asterisk will do 2B channel transfer if you have a "DMS100" switch.  I asked my T1 provider's tech guy, and he said "the protocol for our DMS500 is NI1 and NI2".  Can * do 2B channel transfer on that ?
20:46.30MikeJ[Laptop]I think it works for NI2 now as well.. it does for 5ess for sure
20:46.34dlynesMikeJ[Laptop]: No idea, but that's what I heard from a friend of mine that's a senior manager with Shanghai Bell Alcatel
20:46.44*** join/#asterisk mtgh (n=chatzill@dsl093-001-038.det1.dsl.speakeasy.net)
20:46.57LostFrogThey are supposedly looking the other way on that now, dlynes.
20:47.05LostFrogAccording to my office manager in India.
20:47.06*** join/#asterisk apardo (n=apardo@87.217.145.97)
20:47.26dlynesLostFrog: The law's still there, but if nobody rats you out, you won't get busted, yeah
20:47.40dlynesLostFrog: But if you do get busted, you may as well kiss your ass goodbye
20:47.43LostFrogRight.. it is like the sodomy laws in most states in the US.
20:48.03dlynesLostFrog: You're going to get ratted out once your competition finds out about you
20:48.19*** part/#asterisk stonecolddsl (n=stonecol@static-71-98-251-134.tampfl.dsl-w.verizon.net)
20:48.20*** part/#asterisk spanglesontoast (n=edd@eddland.plus.com)
20:48.21justinu|laptopCoffeeIV_: 2BCT is in the NI2 spec as well
20:48.28LostFrogdlynes: I have no competitors.
20:48.34Greek-Boyare faxes billed on the per minute rate or differently?
20:48.36dlynesLostFrog: If it's for interbranch offices, it's usually  not a problem, because there's no competitors
20:48.49LostFrogdlynes: exactly.
20:48.57justinu|laptopCoffeeIV_: however, your providers DMS500 should be able to talk "DMS Custom" to your asterisk box, as well as NI2
20:49.09*** join/#asterisk MoutaPT (n=MoutaPT@85.139.183.36)
20:49.13*** join/#asterisk k-man_ (n=jason@unaffiliated/k-man)
20:49.16MikeJ[Laptop]justinu|laptop, the issue is what libpri supports I beleive
20:49.19LostFrogGreek-Boy: usually one page is considered one minute and it is billed by time.
20:49.23dlynesMikeJ[Laptop]: Btw, it's only enforced through law, not through the great firewall of china
20:49.31MikeJ[Laptop]heh
20:49.48LostFrogdlynes: I run over VPN anyways.
20:49.56justinu|laptopMikeJ[Laptop]: yeah, libpri might only support it for DMS custom, but his providers DMS should be able to offer either NI2 or DMS variants
20:50.06*** join/#asterisk ctooley (n=ctooley@rrcs-24-227-212-163.sw.biz.rr.com)
20:50.53dlynescoppice might know more about it though...he's in hong kong...don't know if he does much in the mainland, though
20:51.03ctooleyHas anyone come up with a good CDR reporting tool for things other than, split by account code?  I'd like statistical analysis, like concurrent call volumes, geographical distribution of calls, etc.
20:51.22Greek-BoyLostFrog; hope it doesn't take one minute to transfer a page
20:51.23Greek-Boylol
20:51.25Greek-Boyjust jokes
20:51.32dlynesi'm trying to get my buddy in hangzhou to do termination for me, but he's too afraid of getting busted
20:52.33Greek-BoyI can do termination in africa fo ru
20:52.38MoutaPTis there any sip application phone for user agents (callcenter) ?
20:53.08Greek-Boyhow does asterisk handle multiple users on the same hard sip phone?
20:53.17Greek-Boybut each use has different permissions (contexts if u like)
20:53.22Greek-Boyone user can phone local
20:53.25Greek-Boyone national
20:53.28Greek-Boyone international
20:53.29Greek-Boyetc
20:53.37dlynesGreek-Boy: btw...you can also try checking out www.calltermination.com
20:53.42sangeecan anyone tell me how to identify which extension picked during the call blast
20:54.03NuggetGreek-Boy: asterisk handles it however you program it to handle it.
20:54.08dlynesGreek-Boy: Authenticate()
20:54.46*** join/#asterisk ManxPower (n=ewieling@dpc67142183150.direcpc.com)
20:54.50dlynesGreek-Boy: You can supply a filename to authenticate if you want to use multiple authentication codes
20:55.35*** join/#asterisk zoa (n=kkk@pirus.securax.be)
20:55.36zoahey ho
20:55.50Greek-Boybut can a user roam with his extension?
20:56.01jsharpDefine roam.
20:56.16jsharpTake the phone from place to place or take the extension from phone to phone?
20:56.24Greek-Boylol
20:56.28Greek-Boyextensions from phone to phone
20:56.30dlynesGreek-Boy: With his authentication code?  Yes, if you allow all phones access to that context in which the authentication app is called
20:56.32Greek-Boyextension i mean
20:56.39*** join/#asterisk tier_1 (n=tier_1@c-24-9-75-234.hsd1.co.comcast.net)
20:56.48Greek-Boyoh ok
20:56.49dlynesGreek-Boy: If you want set relocation, asterisk does that automatically
20:56.57Greek-Boynice
20:57.06Greek-Boylike i said earlier, I need to do more reading
20:57.06*** join/#asterisk nothinman (i=shakey@83.11.237.245)
20:57.08dlynesGreek-Boy: If you want the opposite of set relocation, afaik, asterisk doesn't do that
20:57.11nothinmanhello...
20:57.17*** join/#asterisk ToTo (n=ToTo@host125-166.pool879.interbusiness.it)
20:57.45Greek-Boyset relocation means the user has to input his location?
20:57.47dlynesGreek-Boy: To be able to do the opposite of set relocation, someone would probably have to write some code to allow an extension to be assigned to a specific mac address
20:58.07dlynesGreek-Boy: Set relocation means no matter where that phone gets plugged in, it's always got the same extension
20:58.17Greek-Boyoh ok
20:58.18Greek-Boythats not bad
20:58.31Greek-Boybut the user has to forward to another extension, right?
20:58.34dlynesGreek-Boy: Set location disabled means whatever phone gets plugged into that specific jack in the wall gets that specific extension
20:58.54dlynesGreek-Boy: Not with set relocation, no
20:59.16dlynesGreek-Boy: All the extension configuration information is stored in the phone
20:59.29*** join/#asterisk aquanaut (n=jvargas@static-64-83-10-246.dsl.cavtel.net)
20:59.52dlynesGreek-Boy: With that feature disabled on traditional keysystems, if i take a phone out of a jack and i move it to a different jack, it simply will not work
21:00.06nothinmanquick question, as usual. If I call asterisk via Zap/1-1 and then application executes Dial("Zap/1-1", "SIP/myvoipprovider/telephonenumber||r") ...is it normal, that it hungs up on Zap/1-1 when the call on SIP/myvopiprovider is answered?
21:00.13Greek-Boyoh ok
21:00.16dlynesGreek-Boy: Asterisk doesn't have that feature...they only have the better feature where you can move phones around at will
21:00.35Greek-Boyso how does a user move around without having to take their phone with them?
21:00.45Greek-Boylike a temp forward
21:00.46dlynesThey don't
21:00.48dlynesOh
21:00.54dlynesYou can do a forward in the dial plan
21:00.55justinu|laptopauth codes, or agents
21:01.01justinu|laptopagent logins
21:01.02dlynesOr authentication codes, yeah
21:01.15Greek-Boyi c
21:01.32dlynesGreek-Boy: For people to better answer you, it might be helpful to know how many extensions you're planning to handle
21:01.51dlynesGreek-Boy: And if you want pickup groups, and all that other kind of stuff
21:02.44tzangerwow
21:02.49dlynesman
21:02.49Greek-Boyi'll take that into consideration and prepare myself
21:02.56dlynesthese nokia 6670's really suck sometimes
21:02.59dlynesunstable as hell
21:03.01*** join/#asterisk Faithful (n=Faithful@202-6-145-116.ip.adam.com.au)
21:03.02*** join/#asterisk file[laptop] (n=jcolp@mctnnbsa24w-142167060049.pppoe-dynamic.nb.aliant.net)
21:03.18tzangerstay away from voipsupply.  great stock, great prices, friendly people.  But they believe they have the right to tell me when I am big enough to use provisioning software.
21:04.02dlyneshuh?  what does selling you product have to do with recommending provisioning software?
21:04.03Greek-Boyany of u played with voip software on mobile phones? i take it the most common protocol on them is SIP?
21:04.32dlynesGreek-Boy: yeah...it's the most common...the 6670 has a sip phone for the symbian os
21:04.35tzangerdlynes: I cannot get the UTStarComm F1000G provisioning software because I did not buy at least 25 phones.  But don't worry, I don't have to buy all 25 at once, and until then I can pay them to write the provisioning files for me
21:04.52Greek-Boythe problem is I only have gprs right now, no 3G
21:04.56SplasPooddlynes: thats rather standard..
21:05.00dlynestzanger: ah...stupid
21:05.01Greek-Boyi dont think voip would work too good over gprs
21:05.05SplasPooderm
21:05.12SplasPoodtzanger even
21:05.23Nuggetvoip is challenging enough when the network isn't actively fighting you.
21:05.39SplasPoodDOWN SIP!  DOWN!   KNOW YOUR PLACE!
21:05.54dlynesGreek-Boy: not familiar with 3G...I think we only have GPRS and EDGE here
21:05.57tzangerI can totally understand not getting at the updates or software if you are not a customer...  but having a minimum buy?  forget it
21:06.05Greek-Boydlynes where u from?
21:06.10dlynesGreek-Boy: Canada
21:06.14Greek-Boynice
21:06.23dlynesYeah...Vancouver is the shizzit :)
21:06.25Greek-Boyi'm surprised 3G not in canada yet
21:06.33dlynesIt might be...I'm just not familiar with it
21:06.42dlynesI only got a camera phone about a month ago :)
21:06.49justinu|laptopyou canucks
21:06.59justinu|laptopwith your 1337 techmology
21:07.00Greek-Boybtw, what happens when u call services that charge u through your phone bill? i mean if u call it via voip?
21:07.22Cybertoyusually you can't
21:07.40dlynesYou mean like 1-900-GIME-SEX?
21:07.43NuggetI see that Greek-Boy's ISP charges by the letter.
21:08.19*** join/#asterisk x86 (n=x86@p3m/member/x86)
21:08.20Greek-Boyyeah, that type of thing
21:08.21Greek-Boylol
21:09.10dlynesYou would probably need to find a voip terminator that terminates 1-900, if such an animal even exists
21:09.16*** join/#asterisk RoyK (n=roy@28.80-203-106.nextgentel.com)
21:09.27justinu|laptopi think 1-900 numbers are all but gone
21:09.37dlynesNah...lots of them still around
21:09.37justinu|laptopever since the LECs started blocking them by default
21:09.46dlynes1-900, 1-876, 1-976, ...
21:09.56justinu|laptopyou never see them advertised anymore
21:10.00dlynesand now they've got dominican republic phone numbers and stuff like that, too
21:10.01Greek-Boywhat are LECs?
21:10.04justinu|laptopit's all toll free, credit card required
21:10.08dlynesLocal Exchange Carriers
21:10.10Druken876 is just the north
21:10.12Drukenhehe
21:10.18justinu|laptopwell... i know about all the caribbean island scams
21:10.23dlynesNah...i've seen 876/976 stateside, too
21:10.34dlynesDruken: We don't have it in Vancouver though...only Ontario
21:11.07Druken976 numbers? hehe never seen one
21:11.28dlynesYeah..876/976 both of those were quite popular in southern ontario when I lived there
21:11.41Drukenbut 1-876 is northern canada, nwt, yukon, nunavit
21:11.43dlynes976 was the first one to come out; 876 came out later
21:11.57dlynesDruken: 876/976 is a co, not an NPA
21:12.02justinu|laptop976 was an office code here
21:12.05justinu|laptopnot an NPA
21:12.12Druken876 is also a NPA
21:12.14Druken:)
21:12.33justinu|laptopcanadia is a bit different than US, i guess
21:12.40dlynes867, not 876
21:13.24dlynesjustinu: Nah....Druken's just very drunken
21:13.26Drukenoh shit, i'm thinking 867
21:13.32justinu|laptopheh
21:13.35Drukenmy bad :)
21:13.43Drukenseems 876 is jamaica?
21:13.50dlynesHe's been drinking too much Labatt's Crystal :)
21:14.04justinu|laptopmmmm.... jamaica
21:14.14justinu|laptopon a similar note
21:14.16justinu|laptoptoday is 4/20
21:14.25dlynesYeah...Jamaica is 876
21:14.27dlynesnot NWT
21:14.55Drukenso i had a dylexic moment
21:15.01justinu|laptopfor a while, there was only one NPA for carribean islands
21:15.04Drukeni've had a bad day.. piss off :)
21:15.13justinu|laptopand the office code determined what country you were calling
21:15.23SplasPoodugh why.. Solve one problem create/find another...   Now my queue rings the next agent, but only once for like a second then reports "timeout after 20000ms" as if rang for the full 20
21:15.43justinu|laptopSplasPood: cruel world, isn't it?
21:16.01*** join/#asterisk naturalblue (n=Administ@87.192.100.109)
21:16.05*** join/#asterisk sssk (n=sssk@s55935276.adsl.wanadoo.nl)
21:16.16dlynesDruken: You don't have Crystal in Ontario anymore?
21:16.34Drukenwtf is crystal?
21:16.42dlynesDruken: Labatt's Crystal
21:16.49tzangerCrystal is a disgusting beer
21:16.51Drukenwouldn't know... don't drink
21:16.53dlyneshehehe
21:16.58tzangerand yes, it's still around
21:17.16dlynesYeah...I remember when I used to live there, only the natives drank it :)
21:17.34tzangerit's the beer for cheap bastards.  :-)
21:17.37dlynesWell, and a lot of the northerners
21:17.42Drukenthat's because they'll drink anything
21:17.46justinu|laptopis it like bud ice?
21:17.52*** join/#asterisk Qwell (n=north@unaffiliated/qwell)
21:17.58dlynesNah...more like the Canadian version of Schlitz
21:17.59generalhanlike Natty ice ! haha
21:18.01tzangerno bud is like filtered monkey piss
21:18.18tzangerand coors light is bud watered down with battery acid
21:18.18justinu|laptopi generally don't drink much beer
21:18.29tzangerme either
21:18.30generalhanlol
21:18.31justinu|laptopbut when I do, I like a good hefeweizen
21:18.32dlynesCoors light is beer flavored soda pop
21:18.36generalhani only drink New Castle !
21:18.36*** part/#asterisk franck (n=franck@tikiwiki/franck)
21:18.53tzangerI'm an old fart though, I go for Export as a first choice, but Waterloo Dark, Richard's Red if I'm feeling for a headache...  but never bud or coor's
21:19.05justinu|laptopand I only like draught beer
21:19.17Op3rI only drink San Miguel Pale Pilsen and Light
21:20.26aquanautHello, Can anyone share any opinions or experiences with Fonality's PBXtra?
21:20.42*** part/#asterisk quux (n=bryan@pdpc/supporter/sustaining/quuxo)
21:21.25*** part/#asterisk paolob (n=paolo@pri-214-b7.codetel.net.do)
21:25.05*** join/#asterisk file (n=jcolp@mctnnbsa24w-142167058071.pppoe-dynamic.nb.aliant.net)
21:25.54aquanautHumm, none? Any recommendations for an of the shelf IP-PBX, or GUI for Asterisk?
21:28.27*** join/#asterisk Lizard-Ki (n=lizard@68-114-68-096.dhcp.plt.ny.charter.com)
21:29.10*** join/#asterisk paolob_ (n=donpaolo@pri-214-b7.codetel.net.do)
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21:31.23jovanhi
21:31.35paolob_hi
21:32.27paolob_I´ve got an issue configuring the PAP2 with Asterisk
21:32.44paolob_do you know who can help me?
21:33.07jovanno :(
21:34.59SplasPoodjustinu: asterisk is cruel, yes :P
21:35.00nahireanwhats the issue?
21:35.01*** join/#asterisk dwmw2_gone (n=dwmw2@baythorne.infradead.org)
21:35.52dlynespaolob_: what's the problem?
21:36.27lzhangdoes changing zaptel.conf rxgain and txgain have anything to do with VoIP (SIP) calls?
21:36.51justinu|laptopit does if the calls are bridged to zap channels
21:37.09*** join/#asterisk paolob-parroquia (n=paolob-p@pri-214-b7.codetel.net.do)
21:37.17paolob_we're connecting a couple of PAP2 Linksys to Asterisk
21:37.32*** join/#asterisk Dorphalsig (n=asterisk@200.71.58.39)
21:37.34paolob_we set the static IP
21:37.36DorphalsigHello
21:37.55DorphalsigI am trying to make an inbound call to my System (I have an E1)
21:38.16DorphalsigThe number is 6446800, but looking at the CLI I get the following message
21:38.18dlynespaolob-parroquia: go on
21:38.40paolob_but we're not able to complete the registration on Asterisk and to ear the tone from the phone
21:38.42*** join/#asterisk fjean (n=fjean@201009190027.user.veloxzone.com.br)
21:38.55fjeanhi all
21:39.07dlynespaolob_: is it going over the internet or a LAN?
21:39.22Dorphalsig-- Extension '6800' in context 'entrada' from '15922999' does not exist.  Rejecting call on channel 0/1, span 2
21:39.24fjeanquick one, is there any way to get the "sessionid" or somthing similar into the CDR ?
21:39.28b00mer_any cisco phone users here?
21:39.37paolob_LAN
21:39.49b00mer_trying to figure out how to conference more than 2 lines
21:39.50dlynespaolob-parroquia: can you pastebin your sip.conf file?
21:40.04justinu|laptopb00mer_: meetme conferneces
21:40.06DorphalsigHello?
21:40.13DorphalsigYo Justin!! =)
21:40.24*** join/#asterisk Whisk (n=whisk@whisk.gotadsl.co.uk)
21:40.29paolob_so I presume the Asterisk must act as SIP server..
21:40.39dlynesYes; the pap2 is a sip device
21:40.49justinu|laptophey Dorphalsig
21:41.19dlynespaolob-parroquia: so i take it you don't have chan_sip loaded?
21:42.12FuriousGeorgeb00mer_: whats wrong with meetme
21:43.01paolob_so fiirst of all I  have to configure the Asterisk as SIP server.. any suggestion?
21:43.09justinu|laptopeverytime I see b00mer, i think of battlestar galactica
21:43.10paolob-parroquiapaolob_, a pastebin is at http://extraball.sunsite.dk/notepad.php
21:43.21Nuggetpaolob: I suggest the documentation.
21:43.48justinu|laptophah
21:43.57justinu|laptopin other words, RTFM
21:44.25justinu|laptopif you have a specific question, someone here will most likely help you out
21:44.29DorphalsigI'm trying to test my new E1. I'm calling myself and my cli reports the following: -- Extension '6800' in context 'entrada' from '15922999' does not exist.  Rejecting call on channel 0/7, span 2
21:44.35FuriousGeorgethere's a battlestar galactiva manual?
21:44.47dlyneswtf?  I thought paolab_ and paolab-parroquia were the same guy?  they're both coming from the same host...
21:45.00justinu|laptopDorphalsig: then that extension must not exist in that context.
21:45.01SplasPoodOk... if I dial my Callback agents /w a timeout on the Dial() it does progress to the next highest penalty number.. but it only rings it a couple times since it appears to sill be waiting for the queue timeout= ... ANyone have any suggestions?
21:45.18SplasPood(progresses to the next highest assuming the first agent does not answer)
21:45.38*** part/#asterisk wrmem (n=monnin@monnin-win.ci.uiuc.edu)
21:47.29SplasPoodthis is driving me absolutely nuts..  I don't know if I'm doing something wrong, or asterisk is..
21:48.29Op3rdoes anyone tried call barging on asterisk?
21:48.43paolob-parroquiadlynes, http://extraball.sunsite.dk/notepad.php?ID=17127 is the sip.conf
21:48.50DorphalsigI'm trying to test my new E1. I'm calling myself and my cli reports the following: -- Extension '6800' in context 'entrada' from '15922999' does not exist.  Rejecting call on channel 0/7, span 2
21:49.07FuriousGeorgeOp3r: you can sorta do it with meetme
21:49.44Op3rFuriousGeorge: err, you have to be on mute to not be able to be detected right?
21:50.31FuriousGeorgeyeah there's an option you can use to join mute
21:51.07FuriousGeorgeOp3r: you can use page, and page someone but you wont be muted
21:52.18paolob-parroquiadlynes, did you see the sip.conf? I have chan_sip commented out... is that the reason of the fail?
21:52.45Dorphalsigjustinu|laptop, --> man. My pŕoblem is I am not trying to call any extension, I just programmed my extensions.conf to redirect anything entering _64468[09] to the autoattend
21:53.03Op3rFuriousGeorge: even for an inbound call?
21:54.34b00mer_sorry ...
21:54.39b00mer_I was a away...
21:54.57b00mer_I would want to be able to conference a couple of people... ad hoc
21:55.08b00mer_not haing to have people call in to a conference line
21:55.12b00mer_is that the only way?
21:55.30SplasPoodhrm... timeoutrestart = yes seems to be the solution to my problem...
21:55.42*** join/#asterisk ManxPower (n=ewieling@dpc67142183150.direcpc.com)
21:55.47ManxPowerWell THAT was fun.
21:55.51FuriousGeorge<PROTECTED>
21:55.56b00mer_I call one person... he asks me to conference is another ... finally one more, but Ican't seem to do that
21:56.08FuriousGeorgeb00mer_: meetme
21:56.19SplasPoodb00mer_: generally Conference on the phone is only 3-way
21:56.23Op3rFuriousGeorge: how about chanspy?
21:56.30ManxPowerdime to golf ball sized hail, wind, the roof started leaking, and the satallite went down to 1% signal for a short time.
21:56.43FuriousGeorgeOp3r: dunno
21:56.54b00mer_FuriousGeorge : is there a way to do it without having to have people call in?
21:56.57justinu|laptopleaking roofs suck
21:57.04*** part/#asterisk BuGcillo (n=dcluna@201.102.94.134)
21:57.28b00mer_I am thinking there can be an elegant solution, but I am too new to be doing anything too elegant
21:57.37FuriousGeorgeb00mer_: make a conference, join it call someone transfer him to it
21:58.19dlynespaolob-parroquia: was on hte phone
21:58.28b00mer_do people have coded up dialplans that create conferences on the fly?
21:59.01FuriousGeorgeb00mer_: meetme does that
21:59.08FuriousGeorged option i think
21:59.27b00mer_ok... I'll focus my research on that
21:59.33b00mer_thanks FG
21:59.33FuriousGeorgegood idea
21:59.35FuriousGeorgenp
21:59.39FuriousGeorgelater all
21:59.58paolob-parroquiadlynes, :-)
22:00.02Op3rhmmm
22:00.12Op3rI need to find a way to do call barging
22:00.12justinu|laptopdoes parroquia mean laptop or something?
22:00.28Op3rI tried chanspy and the audio quality suck balls
22:01.34*** join/#asterisk franck (n=franck@tikiwiki/franck)
22:01.46dlynespaolob-parroquia: Why are you trying to register to a sipura ata?
22:02.39paolob-parroquiadlynes, I must sip-connect asterisk with a pap2
22:02.47paolob-parroquiaand I can't...
22:02.51asterboyAt the beginning of a ZAP to SIP call, I'm getting some major echo and then it seems to die down...what fix exists for that?
22:03.16paolob-parroquiaasterisk is to drive a tel central, where the stations are connected by pap2
22:03.22paolob-parroquiadlynes, asterisk is to drive a tel central, where the stations are connected by pap2
22:03.58dlynespaolob-parroquia: but you don't need to register to the sip device
22:04.09justinu|laptopasterboy: how is 4/20 treating you?
22:04.14*** join/#asterisk bjohnson (n=bjohnson@i216-58-92-2.cybersurf.com)
22:04.21dlynespaolob-parroquia: Try the following page for configuration the pap2 on the pap2 side:  http://voxilla.com/ataconfig.php
22:04.29dlynespaolob-parroquia: And get rid of the register line
22:04.36dlynespaolob-parroquia: And then you should be fine
22:04.54paolob-parroquiadlynes, the register line in sip.conf?
22:04.58paolob-parroquiaor where?
22:04.59dlynespaolob-parroquia: correct
22:05.01*** part/#asterisk sulex (n=sulex@pdpc/supporter/active/sulex)
22:05.23dlynesAnyways...I've gotta run nwo
22:07.09paolob-parroquiadlynes, I haven't a register line in sip.conf
22:07.18*** join/#asterisk imperfect- (n=tbw@c-68-57-228-81.hsd1.in.comcast.net)
22:07.20imperfect-Howdy
22:07.22paolob-parroquiadlynes, do you refer to the host line?
22:07.44imperfect-How do I record a greeting for my voicemail? i.e. if I have 2 mailboxes, how to I record a greeting that says "1 for whatever and 2 for whoever"
22:08.55*** join/#asterisk MacDome (n=eseidel@A17-203-14-213.apple.com)
22:10.34*** join/#asterisk jeffik (n=Jeff@Crimson-111.085.ADSL.NetSurf.Net)
22:10.40*** join/#asterisk bkw__ (n=brian@adsl-70-142-62-199.dsl.tul2ok.sbcglobal.net)
22:10.53lzhanghow do I get chan_alsa.so?
22:10.56*** join/#asterisk Alric (n=nbowyer@masq.hyperusa.com)
22:11.32jovanbye guys ... i'm going to sleep :)
22:12.01*** join/#asterisk stkn (n=stkn@gentoo/developer/pdpc.active.stkn)
22:13.02luke-jr_What context does # transfer via?
22:13.32*** part/#asterisk fjean (n=fjean@201009190027.user.veloxzone.com.br)
22:14.47Alriccontext of the transferring channel in zapata.conf?
22:16.08luke-jr_I don't have a zapata.conf
22:16.34luke-jr_and its not using the context of either channel in sip.conf
22:18.18asterboy*cough* what?
22:18.26justinu|laptop4/20
22:18.35Hmmhesaysis there any good way to rewrite the from field in openser?
22:18.40asterboy*cough* *cough*, great
22:19.32franckI have some issues with asterisk negotiating dynamic codecs
22:20.25franckit seems that asterisk offer iLBC of type 97 but ekiga answer iLBC of type 115. Apparently this is authorised by the RFC, but asterisk choke on this...
22:20.43justinu|laptopthere's a bug report about that on bugs.digium.com franck
22:21.19*** join/#asterisk dsfr (n=dsfr@pdpc/sponsor/digium/dsfr)
22:21.49generalhanabout 4/20 ? lol
22:22.15Cresl1nit was 4:20 today
22:22.23generalhanstill is !
22:22.58CoffeeIV_it's 5:20 guys -- get with the problem
22:23.09generalhani AM getting with the PROBLEM !
22:23.13*** join/#asterisk lecter___ (n=lecter__@200.218.192.10)
22:23.19CoffeeIV_program, sorry
22:23.23generalhanlol
22:23.42lecter___hi all. id like to know if is possible to insert a delay before reinvite
22:24.26justinu|laptopit's 3:24
22:24.47justinu|laptoplecter___: not without modifying the chan_sip code, afaik
22:25.07lecter___:/
22:26.07*** join/#asterisk [hC] (n=hardcore@mail.metrobridge.com)
22:26.41Zodiacalanyone know why my cisco 7960 hardphone can see the status (hint) of my cisco ip communicator softphone and vise versa. but my hardphone can't see the status of another hardphone..
22:27.43Zodiacalchan_sccp
22:27.44*** join/#asterisk file[laptop] (n=jcolp@mctnnbsa24w-142167058071.pppoe-dynamic.nb.aliant.net)
22:27.49lecter___asterisk sends the ACK and, after, send the reinvite. The sipproxy openser receives the ACK first, but it prefers to send the INVITE first..
22:27.52Zodiacalsccp v7.2.4
22:27.54*** join/#asterisk alexis101 (n=alexis@70.54.204.92)
22:27.58lecter___but ok
22:28.00lecter___:)
22:29.19alexis101Hello , Is there a way in the dial cmd or anything else to mute the dtmf to the called user ?
22:29.50alexis101i mean if the calling user press *1 to monitor the call i dont want the called user to know it !
22:30.44*** part/#asterisk lecter___ (n=lecter__@200.218.192.10)
22:32.00justinu|laptopjust tell them your cat stepped on the keypad
22:32.03franckjustinu|laptop: you have a bug ID?
22:33.04justinu|laptopno, but it had something to do with speex
22:33.06justinu|laptopnot iLBC
22:33.09lzhangexten => *51,1,Dial(console/dsp)    <= does this work? I am trying to send out sound over alsa
22:33.10justinu|laptopbut it was the same issue
22:33.48lzhangI am getting app_dial.c: Unable to create channel of type 'console' (cause 66 - Channel not implemented)
22:34.40SplasPoodZodiacal: It depends on where the two parties are located
22:34.44SplasPooddifferent laws everywhere
22:35.01Zodiacalsplaspood usa for example
22:36.51franckjustinu|laptop: yes speex is a dynamic codec too
22:37.52alexis101well except the cat explication there is no other solution :P
22:39.18alexis101if you say that the call can be record the user dont have to know that the call is recorded
22:39.54alexis101not sure if you understant the difference my english is not perfect !
22:40.00alexis101understand **
22:40.55franckjustinu|laptop: I cannot find the bug report....
22:42.34justinu|laptophttp://bugs.digium.com/view.php?id=6568
22:44.24Zodiacalqwell u around? know off hand why why my cisco 7960 hardphone can see the status (hint) of my cisco ip communicator softphone and vise versa. but my hardphone can't see the status of another hardphone?
22:46.24franckjustinu|laptop: I think the bug is misreported... I will put a comment...
22:47.06SplasPoodZodiacal: it varies, state by state.. there's some info on www.voip-info.org
22:47.27Zodiacali'll take a look at that in a few.. intresting tho
22:48.03*** join/#asterisk Gunnar (n=gunnar@101.Red-80-24-171.staticIP.rima-tde.net)
22:48.09justinu|laptopfranck... there was another one on there, but I'm not able to find it
22:48.11SplasPoodZodiacal: And I'm not even sure how it works if your state says only 1 party needs to be aware, but the other party resides in a state that says both parties need to be aware
22:48.39Zodiacalprobably defaults to federal laws
22:48.40justinu|laptopfranck: try searching the dev lists
22:49.56DoktorGreg<PROTECTED>
22:50.15franckjustinu|laptop: http://bugs.digium.com/view.php?id=2293 is a good description of what is happening
22:50.34justinu|laptopah, that was the one
22:50.42*** join/#asterisk heison (n=heison@CPE000a01d49e6f-CM00111a59bec2.cpe.net.cable.rogers.com)
22:50.54franckjustinu|laptop: but still the issue is not fixed
22:51.00justinu|laptopheh
22:51.31generalhanalright everyone .... have fun in here !!! i gotta get outta work while its still 4:20
22:51.37saftsackhi
22:51.39justinu|laptopheh
22:51.41justinu|laptopbye han
22:51.52saftsackhow are the sangoma cards in comparison with the digium cards? (fxo/ fxs)
22:52.06generalhanwhen i come back tomrrow ill have a BIG sipura issue for you all ! lol
22:52.07generalhanhasta !
23:02.43franckjustinu|laptop: http://bugs.digium.com/view.php?id=7005
23:03.09justinu|laptopfranck: good luck
23:04.20*** part/#asterisk SkramX (n=mark@admins.sentiensystems.net)
23:05.36franckit is rather a blocker this issue...
23:06.14saftsackhas someone of you good experiences with sangoma fxs/ fxo cards?
23:06.54DoktorGregwhy do so many use sangoma instead of digium?
23:07.19saftsacki am on sangomas hp the first time
23:07.29saftsackand now i want to know why all people take them
23:07.44saftsackmaybe they are cheaper than the digium ones?
23:07.53*** join/#asterisk chiardon (n=asterisk@200.71.58.39)
23:08.09chiardonHello
23:08.17saftsackHi
23:09.37saftsackchecking whether to build Xgl DDX... checking for XGLMODULES... yes
23:09.38saftsackyes
23:09.43saftsackdoes it look good now? :)
23:09.53chiardontring to compile asterik . . seddenly this error taht Ive never seen . . . Apr 20 18:03:39 ERROR[10036]: chan_phone.c:1305 load_module: Unable to load config phone.conf
23:09.54chiardon<PROTECTED>
23:09.54saftsackoh sry
23:09.57saftsackfalse thread
23:11.08*** join/#asterisk darkskiez (n=darkskie@bb-87-81-62-203.ukonline.co.uk)
23:11.15chiardonwhere I can look for it???
23:12.00chiardonor what I'm missing?
23:12.51chiardonthis is the error: Apr 20 18:03:39 ERROR[10036]: chan_phone.c:1305 load_module: Unable to load config phone.conf
23:13.15*** part/#asterisk _Sam-- (n=sam@mail.kneedraggers.com)
23:13.54chiardonbuying ideas!!!!
23:14.34justinu|laptopwhat's chan_phone?
23:14.35chiardonall the massege: Apr 20 18:03:39 ERROR[10036]: chan_phone.c:1305 load_module: Unable to load config phone.conf
23:15.02chiardonsorry! Apr 20 18:03:39 ERROR[10036]: chan_phone.c:1305 load_module: Unable to load config phone.conf
23:15.03chiardonApr 20 18:03:39 WARNING[10036]: loader.c:414 __load_resource: chan_phone.so: load_module failed, returning -1
23:15.03chiardonApr 20 18:03:39 WARNING[10036]: loader.c:554 load_modules: Loading module chan_phone.so failed!
23:15.53chiardonerror 10036  . . .what's the meaning?
23:16.22justinu|laptopthat would be a process id, i believe
23:16.27chiardonHi justinu . . the coffe are comming!!! I soare!!
23:16.45justinu|laptophaha
23:16.48justinu|laptopcool :)
23:17.03chiardonjust till next week!!!!
23:17.19justinu|laptopit's a good thing it wasn't life saving medication :P
23:17.51chiardonsome hel in relation with those error that doesn't let us to finish the asterisk compile?
23:18.31*** join/#asterisk nservsucks (i=yashy@mail.yashy.com)
23:18.50justinu|laptopwell, i'm not familiar with chan_phone, but if you don't need it... try adding a noload chan_phone.so to modules.conf
23:18.55nservsucksWhat's the option I should look up to filter based on callerid #?
23:19.41chiardonbut  . . .you have some time to make some chooses (intense coffe(plenty of AA) . . . medium . . .or light but no saving medication!!)
23:19.54justinu|laptop:)
23:20.02justinu|laptopmedium is good
23:20.10chiardonthe compilation don't continue because this error!
23:21.09chiardonmedium?? . . .IMHO if you are looking for the best of the tropical taste . . .intense must make!!
23:21.21franckbbl
23:21.23chiardonhahahaha
23:21.23*** part/#asterisk franck (n=franck@tikiwiki/franck)
23:22.57justinu|laptopchiardon: i'll take whatever you recommend then
23:23.36chiardonOK . . apleasure!!
23:24.10justinu|laptopchiardon: i think you mean startup, not compilation
23:25.39*** join/#asterisk NewSole (n=dave@d226-108-46.home.cgocable.net)
23:26.20chiardonyepppp
23:26.52Netgeeksjustinu: so that te410p card that caused the apps not to work.... it's not generating any interrupts, and shows Yellow/NOP for all ports regardless of what is plugged into them
23:27.01Netgeekssounds like a bad card to me
23:27.35justinu|laptopyeah
23:27.44*** part/#asterisk xygoat (n=hotjokb@h-64-105-237-58.chcgilgm.covad.net)
23:27.48chiardonjustinu . . .how I override this?. . . because dorphalsig is onthe university now
23:27.54justinu|laptopchiardon: are you ready for the solution?
23:28.00justinu|laptopopen your modules.conf file
23:28.37chiardonwait . . .
23:29.06justinu|laptopadd a line that says: noload => chan_phone.so
23:29.11justinu|laptophopefully that'll get you past that
23:31.09chiardonI'll be tyring . . .TIA
23:31.15*** join/#asterisk Ariel_ (n=Ariel@dsl-20-177.cofs.net)
23:31.30*** join/#asterisk oej (n=oej@h2.ast.sipit.net)
23:32.03*** join/#asterisk alephcom (n=alephcom@host75.net14.mcsnet.ca)
23:33.03nservsucksWhat's the option I should look up to filter based on callerid #?
23:33.24marcus2is anyone here using CAC AB-IIs as FXS ports?
23:33.56justinu|laptopnservsucks: it's like this: exten => _X./8185551212
23:34.10justinu|laptopwould accept any extension, but caller ID has to match 8185551212
23:34.12*** join/#asterisk wm4k (n=wm4k@dsl213-218-233-192.as15444.net)
23:34.55tier_1whats the best ver of spandsp for 1.2.7 ?
23:35.44*** part/#asterisk naturalblue (n=Administ@87.192.100.109)
23:36.05*** join/#asterisk linlin (n=linlin@c-67-184-230-198.hsd1.il.comcast.net)
23:36.58nservsucksjustinu: I'd like a whitelist of callerids, and so if the callerid inbound is not in that list, it goes to Voicemail()
23:37.23tier_1whats the best ver of spandsp for 1.2.7 ?
23:37.51linlinlooking for some help with getting DTMF to work right on my outbound calls through SIPDiscount
23:38.13linlinit works if i bashe the number 20-30 times, which obviously doesnt work for multiple number prompts like passwords etc
23:38.46linlindtmfmode=rfc2833 is what im usin...what else might i try?
23:39.03tier_1inband
23:39.13justinu|laptopnservsucks: you can do all that
23:39.22justinu|laptopnservsucks: everyhting you need to know is on the wiki
23:39.29linlintier_1 was that directed towards me?
23:41.40Az_auanyone here familiar with snom 360's?
23:42.05linlinsnom?
23:42.14Az_auit's a brand
23:42.17Az_auwww.snom.com
23:42.19linlinoh
23:42.24Az_auhardphone
23:42.30linlini thought you were refering to the 360 area code that ipkall provides
23:42.37*** join/#asterisk CletusColeman (n=CletusCo@ppp-70-242-209-80.dsl.rcsntx.swbell.net)
23:42.37Az_aunah :D
23:43.14linlinlooks like a nice phone
23:43.22*** part/#asterisk aquanaut (n=jvargas@static-64-83-10-246.dsl.cavtel.net)
23:43.29Az_auyea they aren't bad.. got the mass deployment setup going which is nice... plug and play for the end users
23:43.50Az_aujust got a small problem where every now and then they ask for the account password on the phone.. you press ok and it goes away but still annoying none the less
23:45.15linlinhah i laugh at the "voip box"
23:46.09Az_auhaha yea i wouldn't go for one of those
23:47.01Az_aumight be ok for your small office with no tech skills tho.. mind you i haven't seen anyone actually selling them :P
23:47.17nservsucksjustinu: perfect.. can you just point me to where in the wiki? I came here as I can't find it :)
23:47.37linlinyeah i suppose
23:47.51linlinits so small, that would just aggrevate me
23:48.11linlindoesnt sit well on a table with all those have cables shooting out :)
23:48.14Az_auyou could loose it underneath your cableslaw in your patch panel :P
23:48.25linlinyeah that too
23:49.21*** join/#asterisk X-Rob (n=rob-x@dsl-202-173-151-24.qld.westnet.com.au)
23:51.15justinu|laptopnservsucks: search for privacy manager
23:51.50*** join/#asterisk towels (n=el_terri@67.58.10.46)
23:52.01justinu|laptopyou're a towel
23:52.07towelsno youre a towel
23:52.09justinu|laptopno! you're a towel!
23:52.24towelsim so high, i have no idea whats going on
23:53.05nservsucksperfect, thanks
23:54.45towelswet towel fight!!!
23:55.55towelsok lets get serious
23:55.59towelsi have a musiconhold question, how can i make it start at a random location, using rawplayer? instead of starting and stopping each time from previous location
23:59.12Alricrawplayer or native?
23:59.53towelswell my rawplayer is a script that does 'for name in $@; do cat $name ; done'
23:59.57towelsso i dunno what that translates to

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