irclog2html for #asterisk on 20051222

00:00.10rob0and the 1.0.7 talks IAX to other remote servers
00:01.06benjkget rid of WiFi and disable the firewall, then test
00:01.51benjktesting with "but I have done X and it *should* not make a difference" are no good
00:02.00benjks/are/is
00:03.01rob0hmmm, I do have one Windows box on the wired segment that I can try.
00:04.33rob0as I said, another wired Linux box works fine with IAX softphones. I'll try the same clients on that Windows box and report back in a bit. And thanks again to all who looked at this.
00:05.05benjknot another box, use the exact same box with the exact same software (the same instance of that software)
00:05.51benjkyesterday someone told me a funny story about using WiFi
00:06.15benjkthey had been using WiFi for two months and all of a sudden it stopped working
00:06.51rob0hmmm, but it's a laptop, and I have no Ethernet NIC for it
00:07.58benjkafter some serious troubleshooting of what seemed to be an extremely weird scenario, they finally realised that they had been using a neighbours WiFi base station all the time and the neighbour apparently had switched it off while going on holidays
00:08.17rob0haha, not possible where I live :)
00:08.24justinuhah
00:08.36benjkI am not saying that this is your problem
00:09.02maarkenha!  that's great.
00:10.17benjkin any event, testing with similar setups is never going to allow you to eliminate the possible trouble spots
00:10.26benjkyou have to test with the real thing
00:11.10justinusometimes you just don't have any choice
00:11.17justinugotta try and make it as close as possible
00:11.48benjkwell, I would get an ethernet nic for that machine
00:12.07benjkI have got a USB ethernet adapter for just that sort of thing
00:12.12justinuyeah, shouldn't be hard to do
00:12.43benjknever use it -- its only for testing
00:12.49Primercan someone show me an example of the new DB syntax? Seem this old syntax: exten => s,5,DBDel(LASTCALL/${ARG1}) exten => s,6,DBPut(LASTCALL/${ARG1}=${CALLERIDNUM}) isn't supported in 1.2
00:13.02benjkand it has already proven a time saver
00:13.25benjkfor all you know the WiFi driver on your windoze box could have gone funny
00:13.51justinuyeah
00:15.35*** part/#asterisk spackle (n=spackle@209.234.83.19)
00:16.43*** join/#asterisk ManxPower (n=ewieling@dpc6745150107.direcpc.com)
00:19.13*** join/#asterisk chris-fn (n=chris@66.51.202.171)
00:20.39*** part/#asterisk Utah_Dave (n=boucha@0-1pool138-58.nas28.salt-lake-city1.ut.us.da.qwest.net)
00:21.14*** join/#asterisk mogorman (i=root@user-24-236-84-48.knology.net)
00:22.07rob0well we're in luck :) -- same problem on the wired Windows bax using the same idefisk. Can receive calls, no calling out. I really think my best bet is to give up and try upgrading.
00:22.47rob0the Sveasoft router is not between this host and the 1.0.7 * server, and no firewall either way.
00:24.09rob0s/bax/box/ (makes it look like I use Dvorak!)
00:24.31*** part/#asterisk mogorman (i=root@user-24-236-84-48.knology.net)
00:26.48JohnnyCIs it possible to do Fax over IAXy ?
00:27.04Nuggetonly when you're lucky.
00:28.06JohnnyChehe
00:28.19JohnnyCNugget: http://shop.beronet.com/product_info.php/cPath/27/products_id/57?osCsid=0f83f2060a3ddbc19a81d54055e8c79d
00:28.28JohnnyCAnd a Sipura ?
00:29.17maarkenyou have to be slightly luckier.
00:30.07Dandanhm, is * interfaceable with Nitsuko?
00:30.15Dandanshitsuko? :>
00:30.46ManxPowerIf hotels in New Orleans expect toursists to come back, maybe they should stop charging $150 - $200 / night.
00:30.46Nuggetthat url doesn't work
00:31.22*** join/#asterisk mogorman (i=root@user-24-236-84-48.knology.net)
00:31.42JohnnyCsorry
00:31.48JohnnyCits a Sipura 2100
00:31.59JohnnyCit says it can do fax over SIP
00:32.00*** join/#asterisk peted20 (n=chatzill@71.39.93.58)
00:32.05JohnnyCmaybe it works ! :)
00:32.41ManxPowerJohnnyC, FoVoIP is unreliable at best
00:33.08ManxPowerJohnnyC, Perhaps you are confusing Fax with T.38
00:34.04maarkenwhich the 2100 does support.
00:34.25mogormanhey can somone private msg me
00:34.30mogormani want to test something
00:34.40mogormanand tell me if they get my response
00:35.33ManxPowergo ahead/
00:35.40mog_worknevermind it works
00:35.51mogormanerr nevermind it works
00:36.14ManxPowerI prefer my /away message
00:37.05peted20anyone out there that might be able to help me troubleshoot static on a TDM04b?
00:38.27JohnnyCManxPower: whats T38 ?
00:38.47ManxPowerJohnnyC, Google is your friend, grasshopper.
00:38.55JohnnyCI want to install a FAX on a PTP line but It cannot connect directly
00:39.05JohnnyCoki doki
00:39.33JohnnyCFax over ip
00:39.56JohnnyCI see, so Is there any way to receive sending faxes with an Analog fax using asterisk ?
00:40.00JohnnyCa realiable way ?
00:40.06shmaltzanybody ever heard or used this company:
00:40.07shmaltzhttp://www.spiritdsp.com/index.html
00:40.10shmaltz~seen jerjer
00:40.27jbotjerjer <n=JerJer@pdpc/supporter/bronze/jerjer> was last seen on IRC in channel #debian, 40d 16h 42m 56s ago, saying: 'not really a debian specific question, but someone here should know - Can i merge partitions in Linux?  like my / was created way too small and i would like to blow away another partiton and start over, but one issue is I am currently not ...
00:41.20*** join/#asterisk zotz (n=zotz@24.231.47.168)
00:42.16*** part/#asterisk mogorman (i=root@user-24-236-84-48.knology.net)
00:42.31Primerthis is strange...asterisk has been crashing lately but it reports "killed" in the terminal it was running from
00:42.47Primeryet, I'm pretty sure nothing external is signaling it
00:43.07ManxPowerperhaps OOM
00:43.09Primerperhaps some module would do this? asterisk 1.2.1 here
00:43.21ManxPowerWhat signal does it claim to get.
00:43.23Primerdmesg shows no OOM
00:43.50PrimerDec 21 16:30:33 ERROR[13544]: sccp_device.c:170 sccp_session_send: Tried to send packet over DOWN device.
00:43.53PrimerKilled
00:44.09Primerthat message repeated a few times, then "Killed", then back to the terminal
00:44.11ManxPowerSounds like a bug in SCCP
00:44.14Primerperhaps
00:44.23Primerupgrades, even
00:45.24Primerahh I actually have a core file
00:46.13Primerstack is hosed
00:46.42ManxPowerSounds like a bug in SCCP
00:47.33*** join/#asterisk mogorman (i=root@user-24-236-84-48.knology.net)
00:49.36*** join/#asterisk swm_ (n=admin@digitaldatabits.net)
00:49.39*** join/#asterisk anthm (n=anthm@h46085a06.area4.spcsdns.net)
00:49.39*** mode/#asterisk [+o anthm] by ChanServ
00:49.50*** part/#asterisk swm_ (n=admin@digitaldatabits.net)
00:49.53*** join/#asterisk swm_ (n=admin@digitaldatabits.net)
00:54.21*** join/#asterisk swm_ (n=admin@digitaldatabits.net)
00:54.45*** part/#asterisk mogorman (i=root@user-24-236-84-48.knology.net)
00:54.57Kattymoo.
00:55.24Primerno way
00:55.32swm_Mooooooooo
00:55.36Kattythen chocolate chip cookies are /way/ too old.
00:55.48swm_Meow...
00:56.01justinuTCP is too old
00:57.23shmaltzjustinu, thats why asterisk runs on UDP
00:57.34justinuit should use SCTP
00:57.35*** join/#asterisk loud (n=ariel@cypher.punk.net)
00:57.42swm_SCTP?
00:57.55shmaltz~sctp
00:57.58loudis there anyone having incoming issues with voicepulse ?
00:58.10shmaltzloud, yes someone in this channel
00:58.17justinuhttp://en.wikipedia.org/wiki/SCTP
00:58.18shmaltzhis/her name is loud
00:59.03loudk
00:59.30swm_~sctp
01:00.00loudyou mean
01:00.03loud~stcp
01:00.07justinujbot, sctp is the stream transmissions control transmissions protocol
01:00.08jbotokay, justinu
01:00.09shmaltzjbot sctip is Stream Control Transmission Protocol, http://en.wikipedia.org/wiki/SCTP
01:00.11jbotshmaltz: okay
01:00.19shmaltzlol
01:00.23shmaltzI missed it
01:00.23loudeh
01:00.41justinu"SCTP was originally intended for the transport of telephony (SS7) protocols over IP, with the goal of duplicating some of the reliability attributes of the SS7 signaling network in IP. This IETF effort is known as SIGTRAN. In the meantime, other uses have been proposed, for example the DIAMETER protocol."
01:00.49shmaltzbot sctp is also http://en.wikipedia.org/wiki/SCTP
01:00.59shmaltzjbot sctp is also http://en.wikipedia.org/wiki/SCTP
01:01.02jbotshmaltz: okay
01:01.07shmaltz~sctp
01:01.09jbotfrom memory, sctp is the stream transmissions control transmissions protocol, or http://en.wikipedia.org/wiki/SCTP
01:01.35shmaltzgtg
01:01.38shmaltzc ya
01:01.56justinuheh
01:04.00swm_Anyone know why a meetme room would drop a call suddenly???
01:04.21anthmcos it's asterisk *high hat*
01:04.41swm_Yeah well I never had this problem before...
01:05.16*** join/#asterisk nagl (n=nagl@vie-086-059-104-148.dsl.sil.at)
01:05.34*** join/#asterisk mover (n=dlu@gw-dus-net.dus.de.ncore.net)
01:05.37moverhi
01:05.51moveranyone got odbcstorage for voicemail working?
01:05.54swm_I basically set up another asterisk box w/o a X100P card in it to do all my high process crap and have anothher box that answers my X100P inbound calls and sends them to the faster box... Was too lazy to run phone cable to the other one. Wondering if the non existant X100P has something to do with it.
01:06.21anthmare the inbound calls to meetme voip?
01:06.31swm_yep
01:07.53*** join/#asterisk Laibsch (n=Laibsch@G0a44.g.pppool.de)
01:10.07denonswm_: you should use a timing source (like the X100P) for meetme stuff
01:12.43swm_using ZTdummy.
01:12.57swm_denon: ZTDummy is in use.
01:13.14anthmwhat kernel version?
01:13.46Kattyanthm: hi.
01:13.46swm_Kernel version... Umm. Pretty recent. Linux Slackware 2.4.31
01:13.54denonswm_: Ive never been a fan of ZTDummy .. but you should be running 2.6 if you want to go that route
01:14.28swm_What does the dummy driver and 2.6 have?
01:14.50denon2.6 has a more accurate timing available
01:14.56swm_Odd.
01:15.00denonnot really
01:15.15anthmhi Katty
01:15.17denon10 seconds on the wiki would explain this a ll a lot better
01:15.20Katty(=
01:15.23swm_Wonder if SMP has something to do with my issue..
01:15.40denondoubt it, everything's been pretty SMP friendly for a long time
01:15.47anthmztdummy may as well pretend it can't compile on 2.4 cos it doesnt work right
01:16.20swm_So if I throw a X100P into my box and rid of the dummy driver everything could possibly go away for problems.
01:16.27denonswm_:anyway, what we're trying to say, is either throw a cheap x100p in, or move to 2.6 .. or your results will always be unreliable
01:16.39denonone solution is free but takes time, the other solution is like 10 bucks but is fast. :)
01:17.19*** join/#asterisk tengulre (n=tengulre@221.11.5.180)
01:17.31swm_Yup.  I have a spare X100P sitting around here somewhere... The crappiest one ever made... Echo problems "major" on it. :) ... A piece of crap solves the problem :)
01:17.41denonyou dont have to use it ..
01:17.44denonfor voice, I mean
01:17.48denonit just has to be in there
01:17.53denon(and zap compiled/etc)
01:18.06swm_Oh yes, I totally understand that. Just going to use it for timing basically. I wont even plug a phone line into it :)
01:18.35denonk, just making sure
01:18.45swm_Okay. Now how to do take a audio source from a sound card and bring it into a musiconhold channel?
01:19.11denonhttp://asterisk.paperwork.com
01:19.12denongo search. :)
01:20.12swm_Answered. Awesome...
01:21.12swm_Anyone know of a good softphone for pocketpc?
01:21.15anthmas soon as you do it you will be back bitching cos the timing will be messed up with ztdummy ;)
01:21.48swm_anthm: WHAT?
01:23.00swm_Do you wish to elaborate on that subject to make it more clear so when I encounter a problem I can fix it quickly without "bitching?"
01:23.47anthmyour same timing issue will exist using musiconhold that's all
01:24.08swm_w/ a X100P card in the box?
01:24.17heath__swm_: search for app_playfifo
01:25.23swm_heath__: Does not exist via Yahoo, MSN, Google & many others.
01:26.40heath__my bad... it's not a full blown app, but you could probably hack something together that would work
01:26.41anthmtry http://www.pbxfreeware.org I know the guy
01:27.21swm_Would that be BKW?
01:28.11swm_Anthm/Tony wrote the fifo program.
01:30.03heath__someday when i got some free time i'm gonna figure out how to use that thing to pipe audio to/from a skype client
01:30.25heath__just got chipmunk noise last time i tried
01:32.32swm_Skype sux
01:32.58rob0chipmunk noise means a nut is on the line
01:33.07swm_Tony what are you doing in the 888 Conference w/ the Audio tests.
01:33.33anthmworking
01:33.37swm_on wat?
01:34.23anthmcode
01:34.46rob0hmmm, maybe Alvin works for skype
01:34.47swm_Yeah 996 and 888 are bridged now. :)
01:35.32swm_* I swear anything over 996 will be herd in 888... and backwards * ... :)
01:36.44anthmwhat are you doing that gives you chipmunks better known as improper audio timing?
01:37.13tzangerhey anthm kill the +r for this channel while you've got the fat +o :-)
01:38.12*** join/#asterisk hans (n=fugalh@205.208.239.165)
01:38.30heath__i use that dsp stealer thingy someone wrote for skype and piped the audio to a fifo and tried to play that.. i think it was a skype issue
01:38.33hansis there a trick for building zaptel on 2.6.14?
01:38.56tzangerheath__: dsp stealer thingy?
01:39.19heath__yeah can't remember what it's called, but it allows you to change devs for skype
01:39.45heath__dsphijacker or something like that
01:40.19tzangerinteresting
01:40.26anthmgives you slinear?
01:41.09heath__good question
01:41.21heath__did i mention that i pretty much had no idea what i was doing? :)
01:42.06anthmyou can try by saving it all into a file and name it .raw and try playing it
01:42.50heath__i bet you mean via 'play' cuz it'll go by the file extension?
01:42.50Dr-Linuxtzanger: is there any way to see on CLI the connted phone lines to the fxo cards?
01:43.02*** join/#asterisk froguz (n=froguz@201.222.128.128)
01:43.03tzangerDr-Linux: zap show channels?
01:43.12anthmya
01:43.14tzangerzap show status perhaps?  I think that's the command
01:43.35Dr-Linuxtzanger: i shows all the card's installed port, not connected lines though
01:43.45tzangernot sure what you mean
01:43.59Dr-Linuxtzanger: zap show status shows connected card, nothing else
01:44.35heath__what would be more badass would be to play audio from mysql blob fields
01:44.55Dr-Linuxtzanger: i have 2 tdm fxo cards installed (4 ports each) and i have 5 phone lines conntected to first 5 ports
01:45.16Dr-Linuxso is there anyway i can see these 5 lines on the CLI?
01:46.59*** join/#asterisk doughecka (n=Miranda@unaffiliated/doughecka)
01:47.21a1facan anybody explain me this macro @ http://pastebin.ca/34527
01:48.47*** join/#asterisk tengulre (n=root@221.11.5.180)
01:48.58a1faah i get it
01:49.46a1fawhat is ARG2 tho?
01:51.10a1fathe number dialed.. too obvious
01:51.45a1fahello?
01:53.32SkramXHello.
01:54.08*** join/#asterisk anu (n=anu@www.dns.in.th)
01:55.15a1fasyo
01:55.43anthmevery value after the macro name in the Macro app will be available in the macro ${ARG1} ${ARG2}...
01:56.18a1facool
01:56.21a1fai figured it out
01:56.28a1fathis extension is very very clean
01:56.28a1fawow
01:57.14a1faStarting Asterisk PBX: Starting Asterisk PBX: Unable to open pid file '/var/run/asterisk.pid': Permission denied
01:57.17a1fagrrr
01:58.04anthmtry saying please
01:59.02a1faplese?
01:59.41fugitivoalfa: mkdir /var/run/asterisk && chown asterisk:asterisk /var/run/asterisk
02:00.05fugitivoalfa: astrundir => /var/run/asterisk in your /etc/asterisk/asterisk.conf
02:01.00a1fanice
02:01.13froguzi can't get my T100P clon doing outgoing calls. asterisk automatically hang up when called phone answer
02:01.15fugitivoalfa: did it work?
02:01.18a1fahey. i want to limit number of modules loaded
02:01.21a1fafugitivo: yup
02:01.26fugitivoalfa: ok, now read more
02:01.30fugitivo~docs
02:01.31jbotextra, extra, read all about it, docs is probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html
02:01.32a1fai just want to load sip and music on hold for right now
02:02.12*** join/#asterisk alephcom (n=alephcom@host75.net14.mcsnet.ca)
02:02.42anuIf I want just a linux pc for answering machine, should I use Asterisk?
02:02.50a1faDec 22 02:02:30 WARNING[17905] pbx.c: Unable to register extension 's-BUSY', priority 2 in 'macro-stdexten', already in use
02:02.58a1fawhat???
02:03.07fugitivoanu: for answering machine? lol, you don't need asterisk for that
02:03.27hansfugitivo: it works great as an answering machine
02:03.37anufugitivo: thank you
02:03.49fugitivohans: asterisk is a pbx, not an answering machine, i know it'll work great for that
02:03.49a1faanybody know what a hell?
02:04.01hansfugitivo: and nothing else does work well for that
02:04.09hansI did it with vgetty. not fun
02:04.20fugitivohans: a regular answering machine maybe?
02:04.29fugitivohow much does it cost? 10 dollars?
02:04.54hansan asterisk box is a much better answering machine than a 10-buck pos
02:05.02hansbut yes it's more expensive
02:05.06anuI want something smart, like checking an e-mail or my online accounts
02:05.35hansexactly. it can email you your message, pick up immediately at night,
02:05.38fugitivoanu: what do you mean by checking an email or online accounts?
02:05.39a1fai get this error.. what a hell?
02:05.39a1faUnable to register extension 's-BUSY'
02:05.44hansbehave differently based on CID
02:06.01fugitivoalfa: "already in use"
02:06.17hansuse really neat messages you craft using audacity or something
02:06.24hansall of these are things I do with asterisk for my home system
02:06.29a1fafugitivo: i dont understand allready in use
02:06.31anufugitivo: I want an answering machine thai I can script for checking an email for me, or do something else and reply me via the phone, for example.
02:06.34hansloads of fun. do that with a $10 answering machine from walmart
02:06.54hansgreat way to learn asterisk, too
02:06.59a1fahttp://pastebin.ca/34527
02:07.01a1faline 13
02:07.11a1fahow can it be allready in use
02:07.19fugitivoanu: you can do that with asterisk, but remember that asterisk is a complete pbx, not an answering machine
02:07.42hansi think he also knows his PC is a complete computer, not an answering machine
02:08.01anufugitivo: thank you for an idea
02:08.03fugitivoanu: it's like using only the stereo for music of a ferrari :)
02:08.25hansfugitivo: sometimes all you need is a stereo
02:08.36a1fafugitivo
02:08.38anufugitivo: lol, so should I code from scratch myself?
02:08.40fugitivohans: i don't get a ferrari for that
02:08.45a1fahow can S-BUSY allready be used
02:09.08hanswell that's the beauty of OSS. you don't have to pay for asterisk, so it doesn't hurt to use it for an underkill application
02:09.09fugitivoanu: no, i'm sure there're solutions for what you need, but asterisk will do that too, just don't use it only for answering machine :)
02:09.14anuI just dont want to reinvent the ferrari wheels
02:10.34fugitivoalfa: exten => s-BUSY,2,Voicemail(b${ARG2})
02:10.38fugitivoalfa: exten => s-BUSY,2,Hangup
02:10.43fugitivothat's the error
02:10.50fugitivotwo times s-BUSY,2
02:11.00anuanyone come from digium?
02:11.01fugitivoreplace the last one with 3
02:11.08a1faoh
02:11.09a1fashit
02:11.11a1fai missed that
02:11.13a1fathnaks
02:11.30*** join/#asterisk mogorman (i=root@user-24-236-84-48.knology.net)
02:12.27Qwellmogorman: !
02:12.43mogormanQWELL!
02:12.57a1faone more thing
02:12.58a1fares_musiconhold.c: The old musiconhold.conf syntax has been deprecated!  Please refer to the sample configuration for information on the new syntax.
02:13.08Qwella1fa: self-explanitory
02:13.12Qwellread the sample config
02:13.12a1fatru :P
02:13.16fugitivoalfa: make samples for new config samples
02:13.22a1fai did ;P
02:13.47malverian[work]mogorman = mog_work ?
02:14.08mogormanmogorman = mog_home
02:14.13Qwellfugitivo: That's no good
02:14.14malverian[work]Good.
02:14.17Qwellmog_work != mog_home
02:14.17mogormanyou ready malverian[work]
02:14.22malverian[work]Ready as I ever will be.
02:14.25fugitivoQwell: why not?
02:14.41*** join/#asterisk Twister (n=jason@216.30.232.106)
02:14.44fugitivoQwell: of course he should backup everything first :)
02:15.13mogormanokies
02:15.21Qwellfugitivo: because copying one file is FAR easier
02:15.37Qwell`make samples` == breaks all files
02:15.47fugitivoyou're right
02:15.50Qwell`cp musiconhold.conf.sample musiconhold.conf` == breaks one file
02:15.53mogormanno it doesnt
02:15.59mogormanit just moves em
02:16.02Qwellmogorman: is the default to not overwrite now?
02:16.12fugitivoi think it keeps a backup
02:16.20Qwellstill
02:16.21mogormanyes
02:16.24mogormanits backs em up
02:16.26mogorman<PROTECTED>
02:16.31QwellThen you have to go through each file...make changes...blah, blah, blah
02:16.38Qwelland, a second make samples will hose it :D
02:17.08mogormanfine ill make it diff it
02:17.20fugitivoyeah, that'll be better
02:17.37anu:|
02:17.54anudigium booth beside the ibms'
02:18.04mogorman?
02:18.07fugitivo?
02:18.11Qwell!
02:18.12Dr-Linuxoo Qwell is here
02:18.18anuhttp://www.asterisk.org/vonfall2005
02:18.20fugitivo!$%@#!
02:18.27Dr-LinuxQwell: can i go with my question?
02:18.29QwellDr-Linux: just your imagination
02:18.36Dr-Linux:)
02:19.31*** part/#asterisk FuriousGeorge (n=brian@ool-44c5a9b8.dyn.optonline.net)
02:19.32Dr-Linuxis there any command that i can verify, that on which FXO ports analog lines are connected?
02:19.49fugitivo?
02:20.01rob0/.
02:20.07Dr-Linuxi  have 2 fxo (2 port each),  and 5 phone lines are connected to first 5 ports
02:20.23Dr-Linux8 zap channel are configured.
02:20.25QwellWhat do you mean verify?
02:20.45fugitivo2x2 = 4 ?
02:20.53Dr-Linuxsorry
02:21.04QwellDr-Linux: zap show status
02:21.13Dr-Linux2x4= 8
02:21.16Qwellif you see any red alarms, there is a problem
02:21.23fugitivonow we're doing math :)
02:21.26a1fawhat is outbound proxy?
02:21.31Dr-LinuxQwell: zap show status  shows 8 channels
02:21.42moverdamn
02:21.56fugitivoDr-Linux: cat /proc/zaptel/1 cat /proc/zaptel/2
02:21.59Dr-LinuxQwell: i have not physically access to the server,
02:22.07moverwhy i cant connect to a mysqlserver 4.1.10 via odbc?
02:22.18Qwellmover: set it up wrong?
02:22.58Dr-Linuxfugitivo:
02:23.04moverQwel res_odbc is
02:23.11mover[mysql]
02:23.12moverenabled => yes
02:23.12moverdsn => mysodbc
02:23.12moverusername => asterisk
02:23.12moverpassword => ******
02:23.12moverpreconnect => yes
02:23.16fugitivononono
02:23.19fugitivoNOo
02:23.24Qwellmysodbc?
02:23.26fugitivoDr-Linux: now you're in my ignore list
02:23.37fugitivoDr-Linux: and i don't remember why to remove that from BitchX
02:23.54Dr-Linuxok, i just wanted to show you the output, i'm sorry
02:24.03Dr-Linuxi coudn't paste that over here
02:24.13fugitivopastebin
02:24.14a1faSIP is UDP->5060
02:24.22fugitivocat /proc/zaptel/1 && cat/proc/zaptel/2
02:24.27Dr-Linuxokey wait
02:24.45fugitivocat<space>
02:24.48moverQwell odbc.ini
02:24.50mover[mysodbc]
02:24.50moverDriver          = myodbcdriver
02:24.50moverDescription             = MySQL ODBC 2.50 Driver DSN
02:24.50moverSERVER          = 1.2.3.4
02:24.50moverPORT            = 3306
02:24.51moverUSER            =
02:24.52a1faand is sceret in sip.conf md5 or just plain text?
02:24.53moverPassword                =
02:24.55moverDatabase                =
02:24.57moverOPTION          = 3
02:24.59moverSOCKET          =
02:25.03Qwell~pb
02:25.05jbotextra, extra, read all about it, pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.com/
02:25.08fugitivoalfa: plain text
02:25.15a1fasucks
02:25.15fugitivoalfa: until you use md5secret
02:25.17moveroh
02:25.19moverok
02:25.21moversorry
02:25.22Qwellmover: how is it going to know what datbase to use?
02:25.27Qwellyou kinda need to set it
02:25.32Qwelland the server...
02:25.35Qwellget real
02:25.37fugitivoalfa: you can use md5secret instead of secret, check the wiki
02:26.00a1fafugitivo: will sipura send clear txt and then md5 it on the server?
02:26.02fugitivocool, a server at address 1.2.3.4
02:26.10moverqwell i have changed the server
02:26.11fugitivoalfa: no idea
02:26.17Qwellmover: from what, 1.1.1.1?
02:26.19moverits a real instead
02:26.25QwellSo fix all the other settings
02:26.33Dr-Linuxfugitivo: http://pastebin.com/474526
02:26.40Qwellunless you plan on connecting to a null database with a null user
02:26.48Qwellsilly, silly, silly
02:27.35fugitivoDr-Linux: what do you need to know? if the cable is plugged?
02:27.45QwellI already said how to tell
02:27.51Dr-Linuxfugitivo: yes
02:27.52Qwellbut fine, ignore me, after specifically asking me
02:28.01Qwellsee if I help in the future. :)
02:28.09moverQwell WARNING[13261]: app_voicemail.c:2005 messagecount: SQL Alloc Handle failed!
02:28.14Dr-LinuxQwell: sir you are talking to me ? :S
02:28.19QwellDr-Linux: not anymore
02:28.22Qwelld'oh
02:28.31Dr-Linux:(
02:28.37fugitivoDr-Linux: zap show status like Qwell said
02:28.42Qwellmover: Yes, did you set your odbc.ini up properly?
02:28.46Qwellor, at least...halfassed?
02:28.57moveri have changed all settings to fit in odbc.ini
02:29.07Qwellall settings?
02:29.10moverall
02:29.29Dr-LinuxQwell: i already answered, that "zap show status" doesn't show lines, i shows only 2 cards name
02:29.34fugitivomover: don't use mysql, it's evil
02:29.35fugitivomuehehe
02:29.46Dr-Linuxand zap show channels,  shows all the 8 channels info
02:30.01mogormanhey malverian[work] still around?
02:30.29Dr-LinuxQwell: i wish i could understand english and you guys .. :S
02:30.51fugitivoDr-Linux: zap show channel x (where x is the number of the channel)
02:30.52moverQwell should i pastebin all files (sligthly mofidied in pass ans server settings? :-)
02:30.58Qwell~pb
02:30.59jboti heard pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.com/
02:31.12Qwellsince when do we recommend the .com?
02:31.13malverian[work]Yeah
02:31.17malverian[work]mogorman, I've been privmsging you
02:31.26mogormanhmm im not getting em
02:31.37mogormani think my jabber servre is dropping em
02:31.41fugitivoDr-Linux: check InAlarm
02:31.47malverian[work]Hrm..
02:31.49mogormani guess my goal of using jabber for irc fails....
02:31.52mogormanoh well
02:31.57fugitivoDr-Linux: if it's 1, then no cable
02:31.57Qwelljabber irc?
02:32.02mogormandid you get my messages
02:32.03malverian[work]You got the login information though?
02:32.04mogormanjjigw
02:32.08mogormanits a jabber to irc
02:32.11mogormantransport
02:32.13Qwellneat
02:32.21mogormanwell you can message me when you get back
02:32.29mogormani will be on a real irc client then
02:32.39nesyshi folks, someone using amp? I don't know how to configure voicemailmain ...
02:32.39fugitivoBitchX!
02:32.42malverian[work]You mean when you get back?
02:32.44malverian[work]I'm already here :)
02:32.45mogormanit is neat qwell but the private msg doesnt always work
02:32.48fugitivonesys: i use asterisk
02:33.04mogormani thought you were going home malverian[work] to get the echo
02:33.13malverian[work]No, I can do it from here.
02:33.15Dr-Linuxi gonna pastebin the output of, zap show channel 3
02:33.20malverian[work]Did you get my messages listing the 3 issues?
02:33.20*** join/#asterisk TheCops (i=nobody@got.securebinary.com)
02:33.21mogormanokies
02:33.23nesysfugitivo ... yes, with asterisk I know how ... but with 'web interface' I don't know :)
02:33.24mogormanyes
02:33.35malverian[work]Okay, then no messages were lost :)
02:33.40fugitivonesys: why you need a web interface if you know asterisk?
02:33.46mogormanoh so it is still working
02:33.46Dr-Linuxonce someone told me here, that the FXO cards do not know itself if the line is connected to it or not
02:34.08*** join/#asterisk scud (n=scud@12.214.190.139)
02:34.12fugitivoDr-Linux: InAlarm
02:34.14nesysfugitivo because I'm not alone :)
02:34.14malverian[work]mogorman, => query
02:34.21fugitivoDr-Linux: if 1 then no cable
02:34.24mogorman?
02:35.01nesysfugitivo but maybe my query is OT here ... better #amportal
02:35.26*** part/#asterisk mogorman (i=root@user-24-236-84-48.knology.net)
02:35.29fugitivonesys: i know nothing about amp, sorry
02:36.11a1fah,
02:36.53moverQwell please take a look http://pastebin.com/474537
02:37.03Dr-Linuxfugitivo: http://pastebin.com/474536
02:37.11moveri am frustrated
02:37.28Dr-Linuxplease check the output, and please also tell me what things represents that line is connected/working or not?
02:37.31a1fagod
02:37.37a1faasterisk wont answer
02:37.40a1fagod damn him
02:37.57fugitivoDr-Linux: InAlarm: 0  that means that channel is ok
02:38.00a1famy default context is incoming
02:38.05a1faand my sip account context is incoming
02:38.11a1faso when i ring it.. it just rings there
02:38.14*** join/#asterisk mog_home (n=mogorman@user-24-236-84-48.knology.net)
02:38.20Qwellmover: 2.50 myodbc?
02:38.20a1fabut there is exten => s answer()
02:38.24QwellThat's like...old
02:38.25fugitivoalfa: be sure you have an s extension or you specify what extension to go
02:38.29moverno
02:38.32mog_homeokay
02:38.34mog_homeim back
02:38.34QwellWhat are you running, Rh6?
02:38.34moverits only the desc
02:38.44a1fafugitivo: the menu is all set up
02:38.47moverits suse 9.3
02:38.55moverwait a paste the rest of infos
02:39.04TheCopsCan you configure rxgain and txgain for each channel indivudualy ?
02:39.29a1fai cant even see whats happening
02:39.31a1fagrrr
02:39.34Qwellshouldn't "driver=" be a path to the .so?
02:39.42a1fai can make phone calls
02:39.46Dr-Linuxfugitivo: well, my it shows InAlarm: 0  for all of my 8 channels, i have only 5 phones though
02:39.54a1fabut i cant recieve phone calls
02:40.04Dr-Linuxi mentioned my problem at to top of my pastbin as well
02:40.08fugitivoDr-Linux: 5 phones or 5 phone lines?
02:40.10Dr-Linuxfugitivo: http://pastebin.com/474536
02:40.23malverian[work]mog_home, I've got the number set up.
02:40.25Dr-Linuxfugitivo: 5 phone lines
02:40.29mog_homegood
02:40.30Dr-Linuxtotal: 8 ports
02:40.32mog_homeyou can msg me now
02:40.33moverQwell http://pastebin.com/474539
02:40.34mog_homeim available
02:41.12QwellI'm pretty sure driver= needs to be a path
02:41.19fugitivoDr-Linux: if no alarm, then everything is connected
02:41.23Dr-Linuxfugitivo: if you read my actual problem in pastebin, then cal you tell me please, is it phone lines problem?
02:41.30moverQwell http://pastebin.com/474539
02:41.37Qwellmover: still not a path
02:41.40moverodbcinst.ini has it
02:41.54a1fa[incoming]
02:41.55moverand this point over mysqlodbc to it
02:42.01a1fa<PROTECTED>
02:42.01moveror i am totally wring?
02:42.06moverwrong even
02:42.07QwellThen why does odbc.ini have it?
02:42.16Dr-Linux:( i'm still on start
02:42.25a1famy general context = incoming              ; Default for incoming calls
02:42.29moveron other machines i unse exact the same with open office to connect to the same server
02:42.33a1fait still wont answert
02:42.40Qwellokay, I see
02:42.51Dr-Linuxwell, my 3 ports are empty, but it also show for them InAlarm: 0
02:42.58Dr-Linuxbut my problem is...
02:43.01fugitivoDr-Linux: ok, hold on
02:43.12moverok now tell me where i am wrong? :-(((((
02:43.17Dr-Linuxmy problem is that, my first 3 lines are working fine, i can dial out. But when i tried last 2 lines (line4 and line5)
02:43.17Dr-Linuxit says "connected" but nothing else, it means i goes to the channels, but i can't find the lines or what? please help.
02:43.17fugitivoDr-Linux: do you know the phone number of the 2 lines that aren't working?
02:43.41moveri will email you a couple of beers then :-)
02:44.05fugitivoDr-Linux: ?
02:44.07Dr-Linuxfugitivo: no i don't know, but i can check them with dialing out
02:44.13a1fapedantic=no
02:44.23fugitivoDr-Linux: i bet you, that you'll have 2 busy lines
02:44.37Qwellmover: Does that file exist?
02:44.41Qwell/usr/lib/unixODBC/libmyodbc.so
02:44.45fugitivoDr-Linux: try to call all your numbers
02:45.12Dr-Linuxfugitivo: its not setup and new lines how it could be busy, and only i'm registed with the asterisk yet
02:45.19moveryes at bottom i have paste a ls -l /usr/lib/unixODBC
02:45.39moveryou see?
02:45.40fugitivoDr-Linux: just do that test
02:46.17Dr-Linuxcan i try on? fugitivo do you have number with caller ID, if i try to call you
02:46.18moverthere are two versions
02:46.33Dr-Linuxfugitivo: i don't know anyone in US, i'm from Pakistan
02:46.36fugitivoDr-Linux: i'm in argentina
02:46.36Qwellmover: You don't see anything extra if you turn on verbose?
02:46.36a1faguys
02:46.45Dr-LinuxOpss
02:46.46a1fai can recieve phonecalls, but i cant make phonecalls? whats up with that?
02:46.46moverno
02:46.50a1faerr
02:46.53a1faotherway around
02:47.04mover14749 pts/2    Sl+    0:05 asterisk -gcvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvv
02:47.07mover:-)
02:47.10a1fai can make phone calls using my broadvoice sip account
02:47.19moverhehe
02:47.26a1fabut when i call in form other phone, the phone just rings
02:47.31Dr-Linuxfugitivo: moreoever i can dial through my 2nd line, but when i dial this line, it acts like its going on FAX, and then hangup
02:47.42Qwella1fa: You have to answer it, and it'll stop ringing
02:47.47QwellThen you can talk too
02:47.52a1fahm
02:47.59a1faQwell: i have a incoming dial plan..
02:48.06Dr-Linuxfugitivo: the number is 650 2270552
02:48.08QwellYes, and that is why the phone is ringing
02:48.11Qwelltry picking it up...
02:48.36Qwella1fa: or, try being more explicit about what happens
02:48.54a1faQwell: shit,, that sounded really bogus. i am sorry.. i have a incoing dial plan
02:49.11a1faQwell: auto-attendad should answer and ask you for an extension.. but the attendand never picksup (answer())
02:49.25Qwella1fa: What does it say on the CLI?
02:49.29fugitivoalfa: remove the space between the ( )
02:49.32a1faQwell: http://pastebin.ca/34558
02:49.40moverQwell ...
02:49.41a1fai am not using the cli
02:49.49Qwellwell...why not?
02:49.52fugitivoalfa: in your extensions.conf
02:49.58fugitivoalfa: you have answer( )
02:50.11Qwellfugitivo: shouldn't matter.  they aren't required anyhow
02:50.13a1fauyes
02:50.45moverQwell u stop helping me?
02:50.47Qwellmover: What does it say about it when debug is on?
02:51.00movernothing more
02:51.18a1fagod
02:51.18a1faDestroying call 'SD2o1lf01-d462da76de6428a1485e163f12195bdf-js11002'
02:51.21a1fawtf :(
02:51.29Dr-Linuxfugitivo: i just wanna verify, if its not pbx problem
02:51.40Dr-Linuxa1fa: are you behind the firewall?
02:51.41fugitivoDr-Linux: did you call all the numbers?
02:52.13a1faDr-Linux: all inbound traffic is blocked except port 5060
02:52.18Dr-Linuxfugitivo: i'm sorry i don't know the numbers, but i can try to dial someone, who have caller ID, so he can tell me my number
02:52.20*** join/#asterisk klictel (n=klictel@24.200.108.185)
02:52.23fugitivoalfa: udp 10000-20000
02:52.24Dr-Linuxbut i don't know anyone in US :(
02:52.33a1fafugitivo: god thats alot of ports
02:52.49Dr-Linuxa1fa: open 10k to 20k
02:52.53fugitivoalfa: that's rtp, you'll need that :)
02:53.00j_viannafugitivo: brazilian ?
02:53.11fugitivoj_vianna: no, argentinian
02:53.14Dr-Linuxa1fa: add range
02:53.27a1faok
02:53.40*** join/#asterisk [TK]D-Fender (n=joe@66.11.164.239)
02:53.44klictelalfa: you can always limit the range in rtp.conf
02:54.10[TK]D-Fendera1fa : Did you make it back in time to get my pastebin?
02:54.34fugitivoDr-Linux: where are the lines, in the us or pakistan?
02:54.50a1fa[TK]D-Fender: wonderful example
02:54.51Dr-Linuxfugitivo: US
02:54.57a1fa[TK]D-Fender: i am using it right now
02:55.03[TK]D-FenderGood..
02:55.06a1fafirewall is open.. still no answer
02:55.09fugitivoDr-Linux: can you call from one line to another line?
02:55.11moveronly the fucking Dec 22 03:54:27 WARNING[18438]: app_voicemail.c:2005 messagecount: SQL Alloc Handle failed!
02:55.28a1fawow
02:55.29a1fastill Destroying call 'SD2oc3c01-9a243bac9af840f905fe12c74ec6293b-js11002'
02:55.34fugitivoalfa: what's the problem? incomming calls?
02:55.36[TK]D-Fendera1fa : having problems letting people in through NAT?
02:55.40Qwella1fa: pastebin a sip debug
02:55.59Qwell$10 says it's a 404
02:56.09Dr-Linuxfugitivo: yes thats how i'm checking, i can call from first 3 lines, but when i try from line4 and line5 it says >> trying >>. connected >> same ....... and nothing
02:56.13fugitivoi second that
02:56.23a1fa[TK]D-Fender: internet facing
02:56.28a1faQwell: yup
02:56.32a1faSip2.0 404
02:56.41Qwellpastebin all of it...
02:56.42fugitivoDr-Linux: try to call from line 1 to line 4
02:56.44a1faSIP/2.0 404 Not Found
02:56.45a1faLO
02:56.56Qwelland somebody owes me $10 :P
02:57.16Dr-Linuxfugitivo: i don't know the line 4 number right now.
02:57.31a1faone moment
02:57.34fugitivoDr-Linux: isn't it a consecutive number of the first 3 lines?
02:58.15Dr-Linuxfugitivo: all are the different number, i know only first 2 numbers
02:59.00fugitivoDr-Linux: then i can't help you, if you get the numbers, call and if the line is busy, i'm sure the wiring is bad
02:59.01[TK]D-FenderDr-Linux : do you still have the scripts I wrote for you to specify which outgoing line to use?
02:59.12a1faah
02:59.24a1fahttp://pastebin.ca/34559
02:59.35[TK]D-Fendera1fa : So someone's trying to call you through SIP and you're behind NAT and not getting the call?
02:59.53fugitivoalfa: what does your register line say?
02:59.56a1fa[TK]D-Fender: it is internet facing server, no nat
03:00.03a1faContact: <sip:102@XX.XXX.XXX.XX>
03:00.10a1fai think this could be the problem
03:00.16fugitivoalfa: my question
03:00.30Dr-Linux[TK]D-Fender: yes, thats how i'm trying to dialout :)
03:00.33*** part/#asterisk hans (n=fugalh@205.208.239.165)
03:00.37a1fafugitivo: /102 at the end
03:00.45[TK]D-Fendera1fa : pastebin your extensions.conf and SIP.conf for me.
03:00.45*** join/#asterisk BhaalWK (i=bhaal@freenode/staff/bhaal)
03:00.47a1fai think i need to remove that
03:00.54fugitivoalfa: you don't have 302 in your incoming context
03:00.56fugitivoremove it
03:00.59fugitivoand it'll go to s
03:01.03a1faah
03:01.04a1fasweet
03:01.04a1fathanks
03:01.07a1falet me try that
03:01.15Dr-Linux[TK]D-Fender: 6 [line] [number]
03:01.23moverQwell u cant help me?
03:01.30Qwellmover: Not without a proper debug
03:01.45a1faits working
03:01.48a1fabut one small problem
03:01.54fugitivooh god
03:01.54a1fait answers immediatley :P
03:01.57[TK]D-FenderDr-Linux : that one, yes.  Call line 1 from line 4
03:02.06Dr-Linuxbut my 4 and 5 lines are not working
03:02.11moverthere is no debug. i have switched all on and the only message is the known
03:02.22fugitivoalfa: Set(TIMEOUT(digit)=3)
03:02.32fugitivoalfa: errr, i mean
03:02.33[TK]D-FenderDr-Linux : not for incoming or outgoing?
03:02.35fugitivoalfa: Wait,2
03:02.40Dr-Linux[TK]D-Fender: yes i'm trying, but it says >> trying >> then Connected >>>  and same , no tone etc nothing
03:02.51a1fahm
03:02.52fugitivoalfa: exten => s,1,Wait,2
03:02.53a1faone more thing
03:02.55*** join/#asterisk BhaalWK (i=bhaal@freenode/staff/bhaal)
03:03.23moverqwell is there a linux tool to test the local odbc setting with a testconnect?
03:03.40Qwellmover: dunno
03:03.41Dr-Linux[TK]D-Fender: currently i'm just verifying for outgoing, as you know i had 2 lines first, now i added 3 more, so totall 5 lines
03:03.47[TK]D-FenderDr-Linux : You need your boss to really test that line.
03:03.59a1faits unable to dial a sip user behind a nat now
03:04.08a1fait is unable to transfer a call to sip user
03:04.10Dr-Linux[TK]D-Fender: 3rd line is working fine, i can dial it
03:04.19fugitivoDr-Linux: when you know the phone #, try to call that line
03:04.24[TK]D-FenderWell have him test the other lines.
03:04.36Dr-Linuxokey :)
03:04.45[TK]D-FenderGet them to do the job right and give you the info before expecting you to debug it.
03:05.25Dr-Linux[TK]D-Fender: do you have number with Caller ID, if i call the number and you see my caller ID?
03:05.54[TK]D-Fenderthats why I wanted you to dial line 1 from line 4.. <------ so you could do it yourself.
03:06.39Dr-Linux[TK]D-Fender: well, i never see my caller ID,  when call comes in it shows "asrecieved"
03:06.57[TK]D-FenderWith your luck they cheaped out and didn't even get it.
03:07.11Dr-Linuxsorry? :S
03:07.24[TK]D-FenderCallerID as an option
03:08.55alephcomIs anybody here using telasip and running -current?
03:20.01*** join/#asterisk ne_89 (n=wis@cust-224-152.dsl.versateladsl.be)
03:20.51*** join/#asterisk pengyong (n=lala@222.188.128.71)
03:24.55mog_homeQwell, you around
03:25.02mog_homeoh wait nevermind
03:25.07QwellI am
03:25.18mog_homei no longer need you
03:25.22mog_homein that thirty seconds
03:25.22Qwell:(
03:25.26mog_homeyou were replaced
03:25.31Qwellwas more like 7
03:26.23mog_homeseconds are short in mog_home land
03:27.08*** join/#asterisk _Madar (n=tophe@219-84-129-163-adsl-tpe.static.so-net.net.tw)
03:27.14a1faguys
03:28.51file[laptop]Qwell Qwell Qwell
03:29.38Qwellfile[laptop] file[laptop] file[laptop]
03:29.44file[laptop]hi
03:32.44jake1932damn echo
03:33.09TheCopsdoes ztmonitor can be use with a PRI ?!
03:33.19jake1932how do you get echo out of  chan_irc?
03:33.53mog_homeyes TheCops
03:34.20TheCopsnice
03:34.29mog_homeany zap channel
03:34.34mog_homeotherwise you need chanspy
03:34.40TheCopsmog_home, this is the same way for rxgain configuration as an analog ?
03:34.56mog_homeyup
03:34.58mog_homedont do it
03:35.00mog_home^_^
03:35.06TheCops? :P
03:35.22mog_homeyou very rarely need to adjust gains with asterisk
03:35.29mog_homeat least in my experience
03:36.06TheCopsI did with a telco 1004hz test phone number, with 14500 in rx, and this is better then ever.
03:36.17TheCopss/did/called
03:37.49mog_homeim not saying therany always cases where you need it
03:37.54mog_homeit just seems rare to me
03:37.58TheCopsok
03:38.20TheCopsIn what case you are adjusting it ?
03:38.43TheCopsWhat can cause the problem
03:38.56mog_homevolume issues
03:40.02TheCopsthat's why it is more clear now hehe
03:44.00a1fawhere can i found all those mp3s for ivr
03:44.17mog_home/var/lib/asterisk/sounds
03:44.23mog_homeand there is sounds.txt
03:44.30mog_homethat tells you what they all say
03:44.48file[laptop]mog_home: Mattttttt
03:45.05a1fathanks
03:45.11mog_homefile[laptop] Jossssssssssssh
03:45.19file[laptop]mog_home: how was your day?
03:45.35mog_hometiring
03:45.42jake1932file - matt is my new hero
03:45.50mog_homeyours?
03:45.51file[laptop]jake1932: which one? lol
03:45.53jake1932you're on the sidelines now
03:46.05file[laptop]Matt Ogordude?
03:46.13mog_homematt ogordude lol
03:46.18jake1932ogordude
03:46.19*** part/#asterisk DaPrivateer (n=matt7229@gateway.teamfloco.com)
03:46.19jake1932:)
03:46.31*** join/#asterisk Inv_arp (n=junya@c-66-176-211-109.hsd1.fl.comcast.net)
03:46.40mog_homemog, maf, man, mar, mab, mas : The power of the 6 matts
03:46.47file[laptop]mog_home: strike a pose!
03:47.24jake1932anyone know what a jazzy fizzle is?
03:47.24*** join/#asterisk vinko (n=root@208.5.87.254)
03:47.42vinkohello everybody
03:47.45mog_homejazzy fizzle?
03:47.49mog_homehi vinko
03:48.04jake1932yes - it's in so many hip-hop songs now
03:50.43a1faguys, if i have call waiting on my external sip account, and one line is in use
03:50.48a1faand one more phone comes in
03:50.51a1faand its auto-answer
03:50.54a1fawhat happens?
03:51.08brettnemhey all
03:51.10jake1932ring?
03:51.13brettnemI have a pattern matching question..
03:51.16mog_homelol thats a good question
03:51.18a1fado i hear a ring
03:51.21a1fa:
03:51.24mog_homei imagine it rings
03:51.24a1faor it just rings
03:51.36brettnemcan I do something like _[1]5125551212   <== Optional 1+ ?
03:51.38*** join/#asterisk bmg505 (n=leon@dsl-146-63-43.telkomadsl.co.za)
03:51.56vinkomog_home: yesterday I think I asked you about MOH problems, and you asked if I had digium hardware ... Do you remeber?
03:52.12Qwellbrettnem: 15125551212,1
03:52.13Qwellbrettnem: 5125551212,1
03:52.17mog_homesure
03:52.19mog_homeno but
03:52.22mog_homei can play it off that way
03:52.29mog_homesorries i talk to a crap load of people
03:52.45brettnemQwell: I have hundreds of DIDs.. any other way you think?
03:52.53vinkoOk... Anyway.. I'm aving  MOH problems.. (sounds like Darth Vadar).. and I have read that it probably has somthing to do with the timer...
03:53.00vinkoAs I'm running 2.6 Kernel.
03:53.05a1fahow many lines can one sip account have, and where do you specify that?
03:53.05mog_homeokies
03:53.06brettnemwhat about _!5125551212  (exclaimation point)
03:53.08Qwellbrettnem: nope...I've thought about adding a regex-like "?" pattern, but...
03:53.12*** join/#asterisk asteriskmonkey (n=phil@69.158.149.213)
03:53.38vinkoQuestion is... Is there a solution to this .. I have tried running zaprtc.. but no worky
03:53.42jake1932match on_1NXXNXXXXXX
03:53.46brettnemQwell: Needs to be like: _[1]{1}5125551212
03:53.52jake1932use goto _NXXNXXXXXX
03:54.11vinkoAnd I don't think I'm running digium hardware ( I think its a clone)
03:54.16brettnemjake: but that'll catch non-dids too.. outbound PSTN calls, where the 1+ needs to be right
03:54.20Qwellbrettnem: I was thinking something like _1?5125551212
03:54.34*** join/#asterisk sandra78 (n=sandra@200.106.125.141)
03:54.55brettnemQwell: that's the same idea.. an optional 1+
03:54.59Qwellyeah
03:55.06Qwelldunno...pipedreams, I guess
03:55.21brettnemblah
03:55.56sandra78Hi Guys i need to know if i can use any cellular phone with bluetooth with the bluetooth asterisk channel
03:56.01sandra78?
03:56.07Qwellsandra78: sure, probably
03:56.29sandra78i want to play with the bluetooth channel
03:56.48sandra78but i would like to buy a bluettoth cellular phone
03:56.49jake1932brettnem: maybe i don't understand, is this for DIDs and outbound?
03:56.53mog_homeso what happened vinko
03:56.57sandra78nokia 6255
03:57.40brettnemjake: local customers can call DIDs on the box.. and PSTN numbers.. so both
03:57.53sandra78:S
03:57.56sandra78:(
03:58.02jake1932how do you know the difference?
03:58.13jake1932or should you know the difference?
03:58.16brettnemjake1932: I need to say, the dids on the box can be reached with or without a 1+.. if there isn't a matching DID in the context then dial it how it was orignally dialed..
03:58.33brettnemjake1932: yes, if the exten exists
03:58.45brettnemis there a function to test if an extension exists?
03:59.50file[laptop]be very very quiet, I'm doing network maintenance
03:59.54jake1932brettnem: if an extension exists? extensions are matched firest
03:59.56sandra78anybody has work with asterisk bluetooth here?
04:00.17vinkoNever could get zaprtc to compile..
04:00.19jake1932brettnem: first then wildcard stuff
04:01.34[TK]D-Fendersandra78 : I've just picked up a Bluetooth phone and am looking on getting into using chan_bluetooth as well...
04:01.36jake1932brettnem: are you doing something like _XXXX,1,Goto(${EXTEN})?
04:01.51Qwelljake1932: nice infinte loop
04:01.55jake1932,1
04:01.57brettnemjake1932: right.. so if I list every DID with the 1+ and non 1+ versions then it works just like I want.. but that's a lot of lines of code to add
04:02.12brettnemjake1932: heh.. no
04:02.34jake1932brettnem: no you don't need to list everyone twice
04:02.40Qwell_1NXXNXXXXXX,1,Goto(${EXTEN:1})
04:02.42jake1932brettnem: for instance, i use this for toll free
04:03.01jake1932so i don't have to rewrite everything
04:03.02sbingnerchan_bluetooth isn't in main * tree yet right?
04:03.14brettnemQwell: that doesn't work.. PSTN numbers (numbers that don't match local DIDs) need to be dialed the way the customer typed it (without without a 1+)
04:03.33Qwellwithout without?
04:03.44brettnemer with/without
04:03.50brettnem"as is"
04:04.17jake1932then once the matching is complete, dial out the way the customer typed it
04:04.28brettnem1 asterisk box serves multiple calling scopes.. but on-net cusotmers are all considered local.. so 1+ or non 1+ works just fine.. this is my idea..
04:04.45QwellWhy are your incoming and outgoing contexts together in the first place?
04:04.57jake1932ah - good question
04:05.19brettnemnono.. if I use a _1NXXNXXXXXX,1,Goto(${EXTEN:1})... then no calls will EVER go to the PSTN with a 1 in front.. and real LD calls need them.
04:05.37Qwellseparate your incoming and outgoing...
04:05.40brettnemQwell: They arn't actually.. but via includes they are
04:05.43brettnemhmm
04:05.54brettnemyeah still.. doesn't work.. think about it.
04:06.10Qwellis 5125551212 something internal people would dial?
04:06.15brettnemmy pstn gateway has a route like exten => _X.,1,Dial(SIP/ser/${EXTEN})
04:06.31brettnemso it captures the call just like the caller made it.
04:06.44*** join/#asterisk doughecka (n=Miranda@unaffiliated/doughecka)
04:06.52brettnemhowever, I need to check to see if the call is local before I do that because asterisk's dumb loop detection. ;)
04:07.01jake1932ca't see any danger with stripping off the 1, trying to match with local extens, then fallthrough to  VOIP out and add the 1
04:07.14jake1932ant't
04:07.16jake1932can't
04:07.17brettnemjake1932: yeah, I'd need to store that 1 somewhere.
04:07.21Qwellor, just always add a 1
04:07.34Qwell_NXXNXXXXXX,1,Goto(1${EXTEN})
04:07.37brettnemQwell: that's a thought.. but local numbers shouldn't be dialed with a 1+
04:07.45QwellBut they can be dialed with areacode?
04:07.50Qwellthat's just silly
04:08.00brettnemthey must be dialed with areacode
04:08.14brettnemit's pretty typical in the NFL cities
04:08.19jake1932how many local area codes you have?
04:08.21Qwelleh?
04:08.31jake1932should only be a few
04:08.35QwellAre you saying people who like football are too lazy/stupid to dial a 1? :)
04:08.36brettnemjake1932: heh.. about 10
04:08.38*** join/#asterisk FuriousGeorge (n=brian@ool-44c5a9b8.dyn.optonline.net)
04:08.47sbingnerbrettnem: tru EnumLookup
04:08.49sbingnertry
04:08.53jake1932add the ten NXXNXXXXXX
04:09.00sbingnerand set up dns stuffs for it
04:09.08jake1932then fallthrough to 1${EXTEN}
04:09.15brettnemQwell: no.. I'm saying that people who like football congregate so close together that we need lots of area codes to support their lavish telecommunications needs
04:09.17Qwelljake1932: agreed
04:09.20jake1932should a simple
04:09.25jake1932thing to do
04:09.26Qwellbrettnem: so, whats with not needing a 1?
04:09.30*** join/#asterisk stkn_ (i=nobody@gentoo/developer/pdpc.active.stkn)
04:09.50brettnemsbingner: yeah, I probably should.. but looks ugly to manage.. probably not as ugly as this, eh?
04:10.05sbingnerbrettnem: or you could use astdb
04:10.07brettnemQwell: Local calls don't use 1+.. LD calls require 1+
04:10.18jake1932for 10 area codes???  come on here
04:10.19Qwellare they all really "local"?
04:10.24brettnemsbingner: no, I can't use astdb.. heh.. really..
04:10.34brettnemQwell: depends who you are
04:10.42sbingnerwhy worry about it? if you try to dial the long distance call w/o the 1 it'll fail
04:10.46sbingnerer
04:10.47brettnemQwell: we serve many calling areas from a single box
04:10.49Qwellimo, anything not in your area code, should need a 1
04:11.06sbingnerQwell: that's incorrect though
04:11.09Qwelland, you should be able to dial your own area code, with 1+npa
04:11.16sbingnerQwell: anything that requires you to pay toll charges requires a 1
04:11.25brettnemQwell: "area codes" don't mean anything anymore.. but I tend to agree
04:11.27Qwellsbingner: 800 isn't valid.
04:11.30Qwell1-800 is
04:11.34sbingnerthat's a special case
04:11.35Qwell1-800 doesn't cost
04:11.44Qwellwell, there shouldn't be any special cases
04:11.46sbingnerit's long distance, just not to you
04:11.46brettnemcorrection: dosen't cost YOU
04:11.57*** join/#asterisk _tekati_ (n=captain@cpe-66-75-215-63.bak.res.rr.com)
04:12.13jake1932either way - there are a limited amount of local area codes, use 1 for everything else... done
04:12.16brettnemwe use SER to manage calling scopes.. it's too much work for asterisk
04:12.18sbingnerlett me rephrase "anything that incurrs toll charges requires a 1"
04:12.18Qwelleither 1+npa+nxxxxxx or nxxxxxx...nothing else should be valid
04:12.32sbingnerQwell: you've never been to washington DC have you?
04:12.44QwellI didn't say is...I said should be
04:12.54sbingnerheh
04:13.08brettnemsbingner: according to the ACTIS INS, you can 1) require 1+ for a toll indicator 2) require no 1+ ever 3) require 1+ always
04:13.27sbingnerbrettnem: but why do you need to figure that out for the caller?  why not just forward it on however they dialed it? sorry I missed the beginning
04:13.28brettnemQwell: Houston has 3 overlay codes
04:13.28Qwellbrettnem: and I'd agree with that
04:13.42Qwell2 and 3 at least
04:14.00jake1932should be check for local users, check for local area codes, dial 1 + EXTEN.  am i missing something?
04:14.00brettnemsbingner: to know if I should reject or allow the call. If I sell a local only service for example.. then the question is "define local"
04:14.18sbingneraah, your upstream would accept w/o the 1?
04:14.28sbingneron long distance?
04:14.29brettnemWell 1+ has always been misunderstood... does anyone know what it really was SUPPOSED to mean? :)
04:14.38brettnemsbingner: there is not an upstream
04:14.47jake1932your call will cost more money (cept toll free)
04:14.49brettnemsbingner: I interface direct to RBOC tandems
04:15.00brettnemso go tell them to make it work.. hah
04:15.24Qwelland while we're on the subject
04:15.25brettnemActually 1+ was derived as a way of saying "TEN digits will follow instead of 7"
04:15.38Qwelllet's give canada and jamaica their own country codes
04:15.44brettnemWahoo
04:15.52sbingnerlol
04:15.55justinuany good asterisk sites that makes a good business pitch?
04:15.57brettnem1+ was never meant to have anythign to do with toll
04:16.09brettnemjustinu: http://www.iptel.org/ser
04:16.19justinuso why do you have to dial 1 + 7 digits in some areas?
04:16.22asteriskmonkeyanyone got a script for recieving faxes with asterisk 1.2?
04:16.23brettnemoh wait.. they don't use asterisk
04:16.32brettnemjustinu: because people are stupid. :)
04:16.38justinuno, because it's a toll call :P
04:16.39jake1932justinu: rebels
04:17.02a1fagyts.. i am trying to forward a call on timeout
04:17.12sbingnerjustinu: that's legacy I assume, used to work that way where I lived in canada then they changed it so area codde was required with 1
04:17.29jake1932a1fa: what version of asterisk?
04:17.33a1fa1.2.1
04:17.36brettnemactually.. lets see.. originally you can tell (a digit reciever) can tell if 10 digits would follow based on the SECOND digit.. if it was a 0 or 1 then 10 digits would follow
04:17.39a1fado i also need to hangup after I dial
04:17.40brettnembut not anymore..
04:17.43justinuyeah, a few years ago, in states like nevada with only one NPA
04:17.44a1fa1 Dail ()
04:17.47a1fa2 Hangup ()
04:18.03jake1932a1fa: check dialstatus first in the next priority
04:18.04justinuthey wanted you to dial 1 + 7 digits to dial "long distance' numbers, even tho they were in the same NPA
04:18.10brettnemthis is going to make asterisk + ss7 pretty ugly
04:18.12Qwellbrettnem: second digit?  What would the first digit be?
04:18.20a1fawhat does dialstatus do?
04:18.30brettnemQwell: think of the original area codes.. always 1 or 0 in the SECOND digit
04:18.40jake1932a few things
04:18.41jake1932http://www.voip-info.org/wiki-Asterisk+variable+DIALSTATUS
04:18.42Qwellbrettnem: ahh, 808, 515, etc, gotcha
04:18.55a1fajake1932: it does allready :) [TK]D-Fender gave me his macro
04:18.58brettnemand back in the days exchanges couldn't look like that.. it'd wreck the arch for step switches
04:19.03justinuactually, the 0 in the middle denoted the state had only ONE area code
04:19.05brettnemQwell: exactly
04:19.13sbingnersomebody say hawaii?
04:19.13justinuif an area code had a 1 in the middle, it meant the state had multiple codes
04:19.20brettnemjustinu: you could look at it taht way
04:19.26justinuit's how it was designed
04:19.32brettnemwhat if it has an "x" in the middle
04:19.44justinuthey couldn't in the original number plan
04:19.46rob0I'm fine thanks, hawaau?
04:20.07sbingnersorry, 808=hawaii heh
04:20.09brettnemhawaau? lazy chatters.. haha
04:20.16justinunew york city got 212, because it was the shortest possible NPA on a rotary telephone
04:20.26sbingnerlol
04:20.29brettnemjustinu: that's a good one
04:20.35justinuso cal got 213
04:20.39justinusecond shortest :)
04:20.45rob0312=chicago
04:20.49sbingnerand hawaii got the shaft
04:20.52justinuyep, third
04:20.55Qwell989=nebraska
04:21.02brettnemwow 808.. that is a lot of pulses!
04:21.15Qwellor...is 0 more pulses?
04:21.19Qwell909...Riverside, CA
04:21.20sbingner0=10
04:21.21brettnemI think 0 is more
04:21.21justinu0 is ten
04:21.29Qwelltop that
04:21.36rob0900 :)
04:21.36Qwell(and no, 900 doesn't count)
04:21.38brettnemhave you guys seen how step switches work.. it's wild stuff
04:21.38sbingner900-toll
04:21.39sbingnerlol
04:21.46justinuthat's why in the old days, you could pick up a phone and press the hook a bunch of times and get the operator
04:21.46*** join/#asterisk tasat (n=tasat@c-67-180-181-221.hsd1.ca.comcast.net)
04:21.50justinuyou see people do that in movies
04:21.54sbingnerjustinu: you still can
04:22.00brettnemjustinu: a bunch like 10
04:22.03Qwelland not only can you reach just the operator...
04:22.04justinucrossbar switches are even more impressive
04:22.05Qwellyou can call any number
04:22.12sbingnerbut it's really hard
04:22.15Qwellasterisk recognizes pulse. :)
04:22.21Qwelltry it with your iaxy sometime
04:22.29a1faguys, tell me if i need to hangup as well
04:22.30sbingnereverything still recognizes pulse
04:22.32justinuever been inside an operating crossbar office?
04:22.32sbingnerin USA
04:22.34brettnemactually, in the movies, I think they are using a switchboard technology with loop current lighting up a dashboard
04:22.36tasatCould really use some help here: I'm getting IAX2/--- is circuit busy from two providers, every server
04:22.39a1fahttp://pastebin.ca/34565
04:22.45a1fado i need to use hangup on time out as well
04:22.49tasatHow can I debug this?  I't sgot to be me....
04:22.51justinuit's LOUD
04:23.02brettnemjustinu: step offices are loud too
04:23.08tasatEverything was working fine before I tried adding another provider
04:23.13brettnemjustinu: never had the pleasure of seeing an x-bar.. I bet that's neat
04:23.38justinuthey used to use these big rotating drum machines to generate the tones
04:23.50brettnemyou can watch a call.. in a second you'll SEE why they call it DROPPING a call.. in a step switch, the spindle in the middle drops to hang up
04:23.57a1fahttp://pastebin.ca/34565 LINE 25.. DO i need to hangup () also?
04:24.24tasatWhat are the casues of the Everyone is busy/congested -- foobar is circuit-busy, etc?
04:24.38brettnemtasat: perhaps a bad dial statement
04:24.40justinuthis is a good site if you're into phone phreak nostalgia: http://www.bellsystemmemorial.com/
04:24.41sbingnertasat: couldnt contact remote IAX
04:24.49sbingneror that
04:24.53brettnemjustinu: cool.. always looking for that stuff
04:25.16[TK]D-Fendera1fa : line 25 has nothing to do with "hangup"
04:25.33benjktasat: this message is used for a whole bunch of things, it usually means "any other problem"
04:25.58brettnemHAHAHA
04:26.01brettnemLong, long, time ago,
04:26.01brettnemI can still remember,
04:26.01brettnemWhen the local calls were "free".
04:26.02brettnemAnd I knew if I paid my bill,
04:26.03brettnemAnd never wished them any ill,
04:26.04brettnemThat the phone company would let me be...
04:26.18brettnemBut Uncle Sam said he knew better,
04:26.18brettnemSplit 'em up, for all and ever!
04:26.18brettnemWe'll foster competition:
04:26.18brettnemIt's good capital-ism!
04:26.32jake1932a1fa: this is timout for entering an extension
04:26.36jake1932timeout
04:26.38brettnemI can't remember if I cried,
04:26.38brettnemWhen my phone bill first tripled in size.
04:26.38brettnemBut something touched me deep inside,
04:26.39tasatexten => _1NXXNXXXXXX,2,Dial(IAX2/###@voxee/${EXTEN},60,gM(playmsg))
04:26.39brettnemThe day... Bell System... died.
04:26.41brettnem</flood>
04:26.46tasatthis used to work before...
04:26.53a1fa[TK]D-Fender: do i need to add hangup after line 25
04:26.53tasatNo changes
04:27.03jake1932tasat: provider error
04:27.08justinuat one time, bell system was the largest employer in the USA (!)
04:27.14jake1932i can hardly ever use them
04:27.30tasatjake1932: I've tried two providers ...all their servers
04:27.41[TK]D-Fendera1fa : Only if your "i" ot "t" extensions if you want to limit how many times they can screw up or just sit around.
04:27.41asteriskmonkeymmmm anyone know why a number that i have set a fax on would ring busy
04:27.53jake1932tasat: voxee only works sometimes for me
04:28.10tasatjake1932: how about voipjet?
04:28.10jake1932usually says circuit busy
04:28.18a1fanice
04:28.21jake1932tasat: don't use them
04:28.22a1fajus the way i want it
04:28.23[TK]D-Fender"t" gets calles after the guy just sits around, and "i" if they tried entering an invalid extension.
04:28.38jake1932i use asterlink and junction now (along with voxee)
04:28.44jake1932just hunt till one works
04:29.11tasatjake1932: so not likely on my end?  Anyway I can double check?
04:29.15jake1932usually asterlink will work
04:29.20asteriskmonkeyah .. spandsp is a module right?
04:29.23a1fa[TK]D-Fender: hm it never plays that it is trying to connect
04:29.24[TK]D-Fendera1fa : so if they don't know which extension you just forward them to ${DEFAULT}?  Where is ${DEFAULT} and why would there be no voicemail?
04:29.38brettnemheh: his is the story of the unwarranted and criminal dismantling of a company which offered the American people the best telephone service in the world.
04:29.42a1fai will setup voice mail later
04:30.12jake1932tasat: you can iax debug and check the messages
04:30.19[TK]D-Fendera1fa : just use the macro I gave you.
04:30.23a1fa[TK]D-Fender: I am
04:30.35a1fa[TK]D-Fender: instead of Dial, cna i use macro ont hat?
04:30.37[TK]D-Fendera1fa : in the "t" extn I mean..
04:30.40a1faok
04:30.47a1facool.. thats what i thought.. i wasnt for sure
04:30.50tasatjake1932: trying, but don't know what to look for... any tips?
04:30.50justinubrettnem: that site might be a tad biased :P
04:31.04brettnema TAD? haha no, this is GREAT
04:31.07[TK]D-Fenderyeah like exten => t,2,Macro(stdexten,${DEFAULT},NONE)
04:31.13jake1932tasat: what does it say?
04:31.16[TK]D-Fenderor witha  VM box when you decide to make one
04:31.19brettnemjustinu: I'm a CLEC.. so I'm a bit baised
04:31.21brettnem:)
04:31.55jake1932tasat: pastebin
04:31.57a1fanice man
04:31.58a1fathanks
04:32.09swm_!pb
04:32.11tasatjake1932: one sec
04:32.12swm_~pb
04:32.14jboti heard pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.com/
04:32.14a1fawuat is NONE?
04:32.18a1fawhat is none? no timeout?
04:32.58a1fa?
04:33.15[TK]D-Fendera1fa : look at the goto's in STDEXTEN.
04:33.20asteriskmonkeywhere can i find the span dsp modules to compile
04:33.26asteriskmonkeyare they in the asterisk-addons?
04:33.40[TK]D-FenderIt allows you to template an extension and have it so you don;t HAVE to have a VM box when dialing.
04:34.06brettnemhey, these are great: http://www.bellsystemmemorial.com/oldphotos.html
04:34.36a1faoh
04:34.38a1faso its ARGV2
04:35.02a1fabut if it is none
04:35.07[TK]D-FenderYes, its the mailbox itself.  if you put NONE instead of a mailbox it'll skip being able to leave one.
04:35.31a1faok
04:35.34a1fagreat man
04:35.40a1fathis is so awesome
04:35.47a1fai'm loving it.. que cheasy music
04:35.48brettnemheh.. and you thought YOUR cell phone was big: http://www.bellsystemmemorial.com/images/oldphotos/ba_48.jpg
04:35.49a1facheesy
04:35.59[TK]D-Fendera1fa : Believe me.... this is the LIGHT version of what I do...
04:36.18jake1932*yawn*
04:36.19a1fait is awesome
04:36.33Qwellhttp://www.bellsystemmemorial.com/images/oldphotos/ba_21.jpg  50c/min
04:36.37QwellThat's just sick
04:36.52a1falol
04:36.57a1fahey, one problem
04:37.00brettnemI love these phone poles.. I especially like the one that appears to be.. well.. mobile: http://www.bellsystemmemorial.com/images/oldphotos/ba_33.jpg
04:37.13a1fait never plays connecting
04:37.16a1fapriv-trying
04:37.46[TK]D-Fender?
04:37.54[TK]D-FenderWhere?
04:37.58a1fain timeout
04:38.04a1fai put Playback(priv-trying)
04:38.06a1fait never plays it
04:38.20[TK]D-FenderWhere do I even see that line?
04:38.28a1faits my new version
04:38.34asteriskmonkeywhere do i get SPANDSP!
04:38.40asteriskmonkeyand will the old one work from like 2004
04:38.45[TK]D-Fender...maybe you should SHOW us before asking why it doesn't work :)
04:39.21a1faand enter-ext-of-person
04:39.26a1fait jumps in the middle of the sentence
04:39.27[TK]D-FenderOh, and if you're not feeling to dizzy take a peek at my STDEXTEN :D http://pastebin.ca/34566
04:39.39a1fa"sion of the person you are trying to reach"
04:39.41[TK]D-Fendera1fa : PASTEBIN IT.
04:39.45tasatjake1932: found a server that works...
04:39.51benjkbrettnem: US phone service totally sucks
04:39.57jake1932cool
04:40.00brettnemyep
04:40.05benjk13 business days and still on 1-800 number activated
04:40.09benjkthird world
04:40.26jake1932you can keep voxee in there and try it first since it's cheap
04:40.27brettnembenjk: the incumbants make competition almost impossible
04:40.37benjkin any halfway technologically advanced country this would have been a matter of max 30 mins
04:40.38tasatRx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 001 Type: IAX     Subclass: ACK
04:40.38tasat<PROTECTED>
04:40.38tasatRx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 001 Type: IAX     Subclass: REJECT
04:40.38tasat<PROTECTED>
04:40.38tasat<PROTECTED>
04:40.58tasatWhat's that? bad password?
04:41.04benjktasat: check your context
04:41.19a1fahttp://pastebin.ca/34567
04:41.24a1fa[TK]D-Fender: my man, http://pastebin.ca/34567
04:41.24tasatbenjk: what about the context?
04:42.08benjkyou often get this message if you have specified a context that doesn't exist, ie misspelt or forgotten to create it in extensions.conf
04:42.08jake1932sorry - gotta go to sleep :) see ya
04:42.41benjkin any event, its either username wrong, password wrong or context wrong
04:42.58brettnemit's like something terry gilliam came up with: http://www.bellsystemmemorial.com/images/ross_hamilton/walhwc08-1945_.png
04:43.04brettnemreminds me of "brazil"
04:43.15[TK]D-Fendera1fa : Answer FIRST, then wait 2 seconds before playing back.
04:43.42a1faok
04:43.48a1faDec 22 04:43:05 WARNING[19520]: app_playback.c:132 playback_exec: ast_streamfile failed on SIP/2025171605-5b76 for priv-trying
04:43.50a1fa:P
04:43.58a1fafigured out why that is not working
04:44.24tasatbenjk: thanks, not sure how the context can be wrong, can you give me an example?
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04:45.35a1faok.. thats working
04:45.38a1faneed to found a working sound
04:45.52benjkwell, like context=flintstones in iax.conf and [flintst0nes] in extensions.conf
04:46.57[TK]D-Fendera1fa, Just make your own!
04:47.09*** join/#asterisk implicit (n=implicit@200.179.233.155)
04:47.47a1fadude, i have a really bad russian accent :(
04:47.58a1faand i speak so fast, no body will understand me
04:48.06a1fathey be like : wtf? repeat
04:48.06implicit:)
04:48.10[TK]D-Fendera1fa, I made 28 recordings to support my dial-plan.... just make your own message.  I have almost as many recordings as you have lines in your dialplan :)
04:48.17[TK]D-Fender:O
04:48.26[TK]D-FenderSorry, make that 38 recordings...
04:48.29[TK]D-Fenderand oh well!
04:49.50[TK]D-Fendera1fa : or try pbx-transfer
04:51.44a1fawill do
04:54.25a1fai need to play that monkey sound
04:54.30a1fahheeh
05:01.14`SauronFender: Hehn. I don't want to know how many lines are in my dialplan...
05:01.29`Sauron-= 284 extensions (441 priorities) in 99 contexts. =-
05:02.16fugitivonot using macros?
05:02.27`SauronKATnip
05:02.31`Sauronget it? (lame :)
05:02.59`Sauronfugitivo: that's with macros
05:03.00Katty...
05:03.09Kattysilly rabbit.
05:03.14Kattynaps are for kids.
05:03.17`Sauronwc -l shows me 442 lines
05:03.26`Sauronwhat about nips?
05:03.26Kattyor something like that
05:03.41Kattythose are for grownup girls.
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05:08.40[TK]D-Fender`Sauron : OMG
05:09.42[TK]D-Fender-= 101 extensions (373 priorities) in 27 contexts. =-
05:10.01[TK]D-FenderHmmm, not far off ;)  just different distribution :D
05:10.15fugitivolook at this game  http://www.addictinggames.com/capoeirafighter3.html
05:11.46Kattymister fender.
05:12.08[TK]D-Fenderaye :)
05:13.29`Sauronstupid flash cra
05:13.30`Sauronp
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05:14.48fugitivoit looks a nice flash coding to me
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05:17.50[TK]D-FenderKatty :  mew?
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05:34.53surfdueis ther a gui built in now
05:35.00surfdueor you still have to get one seporately?
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05:42.04TwisterI know this is not the approporiate place for this and im sorry however i didnt know where else to go.  has anyone ever worked with avaya/lucent partner acs systems that would be willing to answer a question for me
05:42.54alephcomsurfdue:  You have to get one seperately.
05:43.10alephcomMany of us, me included hope that it stays that way for a LONG time. :-)
05:43.25surfduehttp://www.voiceone.it/download/
05:43.31surfduethis looks better then ampp
05:43.48[TK]D-FenderGUI = SUCK
05:45.16a1fayeha dude
05:45.22a1fawhat is fun messing with gui
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05:45.37a1fawhen you can write that shit in the same time it takes you to download that gui package
05:45.39a1faand fix up
05:45.40a1fa:P
05:46.15a1fanight
05:46.59kusznir_hi all:  I'm having trouble configuring a D-Link 1120S to work with my * server (1.0.7 on openWRT).  I have a GrandStream gxp2000 already working.
05:47.23Qwell!
05:47.53kusznir_the * server is showing the extention as registered, and sip debug shows a conversation that appears to authenticate sucessfully.
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05:48.29kusznir_However, my phone on the ATA does not provide a dialtone (it does allow me to generate touch tones).
05:48.40grandyhello... does anyone recommend a good language for writing agi scripts in?  I have used php, but i'm looking something that handles exceptions better so that i can make the scripts more robust easily... any suggestions?
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05:49.01Qwellgrandy: c
05:49.12Qwellexcept they wouldn't be agi scripts
05:49.41MasterObi-WanKhello anyone here that provides services to configure asterisk PBX by SSH ?
05:49.41mog_homeboo agi
05:49.50grandyQwell: why do you like c for 'em?
05:49.52QwellMasterObi-WanK: what's needed?
05:49.56mog_homei have a box MasterObi-WanK
05:49.57Qwellgrandy: c=fast
05:50.03mog_homec = super fast
05:50.22Qwellvbscript = super fast
05:50.29QwellSo, I've decided...
05:50.31mog_homesuper stupid
05:50.32mog_homebut close
05:50.37QwellI'm gonna say one thing daily, that will get me flamed.
05:50.50MasterObi-WanKQwell, I need built from scratch pbx or asterisk at home, to do sip trunking and IAX trunking optimizer for bandwidth saving, and more customizations
05:50.50QwellThat was me meeting my quota
05:51.14mog_homelol qwell one thing....
05:51.16mog_homesure
05:51.18grandyi like vb... if it had a better name it would probably have been popular in a wider variety of circles... like Ruby, for example...
05:51.20Qwellmog_home: :P
05:51.27QwellMasterObi-WanK: msg me with what you need (exactly), how much you're offering, and a timeframe required
05:52.13Qwelldaddy needs a new processor...
05:52.53mog_homelol no grandy
05:53.02mog_homeruby python etc are slow
05:53.06mog_homethey will be slow
05:53.10Qwellruby is teh hotness though
05:53.17mog_homeSlow hotness
05:53.23mog_homeC old and true
05:53.25Qwellhotness nontheless
05:54.37mog_homeif i wanted something slow
05:54.41mog_homeid run java
05:54.43Qwellgod
05:55.13QwellI loath java
05:55.14mog_homeand a python interpreter inside that
05:55.16Qwellheh
05:55.20mog_homeand inside that ruby
05:55.20[TK]D-FenderI wrote a language that is likely slower taht just about anything else out there :)
05:55.25Qwellrunning in windows
05:55.30Qwellembedded
05:55.32mog_homerunning in a vm
05:56.55mog_homeman word to the wise
05:57.00mog_homedont mix redbull and milk
05:57.04Qwelleww
05:57.06mog_homeits not as good as it sounds
05:57.13Qwellthat bad?
05:57.13Qwellheh
05:57.16Qwellit sounds awful
05:57.21mog_homeit was calciumy
05:58.40kusznir_what does:  WARNING[425]: chan_sip.c:694 retrans_pkt: Maximum retries exceeded on call....for seqno 102 (Non-critical Request) mean?
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06:08.58infinity1redbul and milk? that does NOT sound good
06:12.43Qwellwtf
06:12.49Qwelllilo jumped ship :P
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06:13.02QwellI thought the captain was supposed to stay, and sink?
06:13.04mog_homelol
06:13.06QwellAhh, there he is! :P
06:13.11Qwelllilo: ;]
06:13.17mog_homelilo is fine the other guy
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06:13.22mog_homealways fakes me out
06:13.46Qwellmog_home: dm?
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06:17.52*** topic/#asterisk is Asterisk 1.2.1 has been released! -//- http://www.asterisk.org
06:18.07Qwellnope
06:18.07*** join/#asterisk MGSsancho (n=user@67.126.140.54) [NETSPLIT VICTIM]
06:18.08Qwellturbulance still
06:18.51grandyQwell: do you seriously write agi programs in c?
06:19.05kusznir_Anyone here use the D-Link DVG-1120?
06:19.06mog_homewhat do you need to do in agi that you cant do in dialplan
06:19.16grandymog_home: me?
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06:19.42mog_homesure
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06:19.48MGSsancho>_>
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06:20.29grandymog_home: i access a db, check if the caller id is already a user, prompt for pin, etc... if i could do the db queries in the dialplan then it would be much easier... is that possible?
06:20.37mog_homeyes
06:20.47mog_homefor mysql and postgres
06:21.03Qwelland odbc
06:21.03mog_homeoh really rock on
06:21.05grandymog_home: then maybe i shouldn't be using agi...  do you know what i'd searhc for to find docs on the postgres db interface...
06:21.07mog_homei didnt no qwell
06:21.19Qwellmog_home: Realtime and RealtimeUpdate
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06:21.32QwellI use those for my macro-agentlogonoff
06:21.39mog_homeapp_postgress
06:21.42mog_homenice
06:21.50grandyoh also i'm using deadagi to call a script that scps some files... can i do that in the dial plan too?
06:21.58mog_homeyeah probably
06:22.02Qwellgrandy: h exten
06:22.36grandyQwell: whaddya mean?
06:22.46Qwelluse the h exten to do all of that
06:23.00Qwelldeadagi would need to be called from there anyhow
06:23.04grandyQwell: need to read about that, i don't know what it is...
06:23.20mog_homei am not saying agi isnt great
06:23.28Qwellmog_home: I'll say it. :)
06:23.31Qwellagi isn't great
06:23.35mog_homejust 99% of the time it isnt needed
06:23.36grandymog_home: sounds like you guys don't like it
06:23.49Qwellgrandy: no point in using it, a lot of the time
06:23.57mog_homeael is pretty hot
06:24.06QwellI've not touched ael yet...
06:24.08grandythe only other concern i have is, what if a command fails, can i still set up the dialplan to at least log intelligently or recover gracefully?
06:24.17Qwellgrandy: sure
06:24.28Qwellmany commands set a status variable
06:24.58grandyok... well i'm glad i have an excuse to really learn the dialplan b/c i was going to have to do so in order to set up some other stuff in a few weeks
06:25.21grandyso h exten ... lemme read about that for a minute... btw, is ael worth looking into for any of this?
06:25.24Qwellmog_home: I need to give somebody at work a crash course in asterisk...
06:25.38mog_homefun fun fun
06:25.39Qwellgrandy: ael is experimental, but it works
06:25.45mog_homeill ready app_segfault for ya
06:25.51mog_homeits not experimental
06:25.57Qwellisn't it?
06:26.00mog_homeit converts it to data structures
06:26.05mog_homejust like dial plan
06:26.14mog_homenot all things work in it like normal dialplan
06:26.20grandyhmm... ok...
06:26.22grandycool
06:26.22mog_homelike hints are still broken i think
06:26.27Qwellaww
06:26.28mog_homebut most things work just fine
06:26.28grandycan you do db calls from it do you know?
06:26.34mog_homeyes
06:26.41mog_homeit can execute all the normal apps
06:26.43Qwellgrandy: all the same applications/functions are available
06:26.49mog_homeexcept hints
06:26.56Qwellyou'd think hints would work
06:26.58mog_homeand some other little things
06:27.01Qwellaren't they just pri -1?
06:27.04grandywould you recommend based on what i've mentioned that i stick with the regular dial plan or that i look into aes?
06:27.04mog_homebut if it doesnt work its obvious
06:27.22Qwellmog_home: is everything gonna go to ael eventually?
06:27.24mog_homedial plan is like basic
06:27.27mog_homeael is like perl
06:27.34Qwellael reminds me a lot of c
06:27.37mog_homeyou can do everything in both
06:27.44mog_homeone is easier for some things than the other
06:27.52mog_homei dont think we will ever get rid of normal dial plan
06:27.58mog_homeas its easier for non programmers
06:28.01Qwellyeah...
06:28.12mog_homewe wont get rid of it till we go webgui everywhere
06:28.14mog_homeaka nevre
06:28.19Qwellwebgui?!
06:28.33mog_homeit was a joke
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06:28.37Qwelloh :p
06:28.54grandyok... thanks much...
06:28.54Qwellooo, you know what would be hot?
06:28.59Qwelldoxygen docs in ael
06:29.04grandyi'm going to look into it asap
06:29.06grandythanks guys
06:29.11mog_homeno prob grandy
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06:36.02Primerdamn, finally got this 7920 working with WPA
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06:57.16Primerdammit, seems that the 7920's RTP isn't punching through my NAT
06:57.25Primerit goes out but doesn't come back in
06:57.37Primerit comes back, but the NAT drops it
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07:02.10kuku5How do I "call" the company directory
07:03.14swm_kuku5: Pess Pound then shit in a bag and have your mother eat it?
07:03.34kuku5fuck off ? :)
07:03.48swm_LOL kinky...
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07:04.12kuku5im sure there is a # option sowhere that I have to put in
07:04.48swm_Uhh... Dial 1-800-IM-QUEER ?
07:05.29kuku5you done?
07:06.32mog_home?
07:06.42swm_Just helpin ya out... Thought ya were sittin in your closet talkin in here... Tryin to get ya out...
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07:09.30mog_homeswm_ i thought you agreed to stop being a loser in here
07:09.31mog_home?
07:09.45swm_Yeah but I saw that you were here so I could not resist getting a rise out of ya
07:09.56mog_homewell you did
07:10.00mog_homerack up a point
07:10.02mog_homeand knock it off
07:10.19swm_but if you got a rise and i knocked it off you would be a-sexual? right?
07:10.53mog_homewow
07:10.56mog_homenever a dul moment
07:10.59mog_homebut seriously
07:11.01mog_homeshut it
07:11.42swm_Uhhhh ... Uhhhhh huhuh ... He He ... Hey beavis... Heheh huh huehe huh ... He said "Shut it..." huh ehhe huhuh hehehe huh hehe yeah .. huh shut it... hehe huh eheheu huhe heh uhuh
07:12.09Primerhehe
07:12.42swm_Hey butthead ... i think he likes us... huh ehehuh huh eheh huh
07:13.21mog_homedo you want to be /banned?
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07:13.42swm_Really? So I can use another ip address? Proxy myself and go from somewhere else around the world... ok
07:13.49Primerdo it! do it!
07:14.12mog_homewow you and any other script kiddy
07:14.17mog_homeand we can just i dont know
07:14.18swm_Advantages of having 5 ISP's. :)
07:14.19mog_homekeep banning you
07:14.31mog_homeim all up for playing whack a mole
07:14.42swm_Oh I guess I could be good ...
07:14.46swm_for those without a sense of humor.
07:14.54lilowheeee
07:15.04lilomuch better now
07:15.19mog_homeyay lilo
07:15.46lilomog_home: it's just the volume of spambots we're getting....needed to adjust the way we were handling them
07:15.51lilomog_home: it's a resource hog
07:15.59mog_homei bet
07:16.09mog_homehow do you guys pin em down, just traffic?
07:16.12swm_Oh I hate those things.
07:16.42Primerargh, this is pissing me off
07:16.59Primerwhy does NAT fuck everything
07:17.04swm_Whats pissin ya off primer?
07:17.13wasimPrimer: NAT doesn't do anything to IAX
07:17.34PrimerI have a 7920 that I've finally got connected to my AP with WPA, and my NAT won't NAT the RTP
07:17.36lilomog_home: well, there will be a patch committed at some point, but I'm not talking much about it for now 8)
07:17.43Primerwasim: unfortunately cisco isn't fond of IAX
07:17.49mog_homeokies
07:17.59lilomog_home: I'm enjoying time off from the dern things 8)
07:18.00mog_homewhen did you start hanging out here lilo
07:18.08lilomog_home: I've been here for pretty nearly forever
07:18.15lilomog_home: I'm just usually pretty quiet
07:18.20swm_Patch? For NAT? to do what?
07:18.27PrimerI have a sipura behind NAT and the NAT box gladly NATs the RTP for it without any special consideration
07:18.29mog_homethat must be hectic lilo
07:18.33mog_homehow do you get things done
07:18.42swm_I'
07:18.51swm_I'm on 11 channels and that's enough for me.
07:19.00lilomog_home: well, mostly I just log and then go look for problems if someone complains a channel is being harassed
07:19.08lilomog_home: and it lets me keep up with news and so on
07:19.22PrimerI must confess, I'm on more channels on this network than the other 4 networks I'm on
07:19.26lilomog_home: I usually check in everywhere sooner or later
07:19.34Primer7
07:19.35mog_homeahh
07:19.35liloPrimer: well, I hope we're useful to you :)
07:19.41mog_homeive never noticed you in here
07:19.44swm_lilo: Any warez channels on freenode
07:19.50liloswm_: not if we can help it 8)
07:19.59Primerlilo: only one is an "off-topic" channel
07:20.00liloswm_: they do tend to leave a big footprint
07:20.06swm_lilo: good... Would hate to find some pirated software and turn someone in :)
07:20.15liloswm_: you should feel free to
07:20.30swm_lilo: I try..
07:20.51liloswm_: if you see something like that, point us to it....warez does not help us do what we're here to do
07:21.10PrimerI distribute tons of software on freeload
07:21.18swm_lilo: Tracked someone from New Jersey who was attemping to upload over 37 Gig's of MP3's to my server... lol... my T-1's were pretty used up ...
07:21.22lilowarez, hax0r, proprietary media trading, are all pretty aggressive cultures
07:21.51lilothey're off-topic here both because they're unlawful and because they tend to screw up the network culture
07:22.26lilofwiw, I haven't run a proprietary OS at home in ages....it really just hasn't seemed necessary
07:22.30Primerlilo: so what's your stance on proprietary software channels?
07:22.34Primerlike, #oracle?
07:22.44*** join/#asterisk Bobacus_Bum (i=Bobacus@ip68-97-65-77.ok.ok.cox.net)
07:22.52liloPrimer: http://freenode.net/policy.shtml#channelnaming
07:23.02kuku5whats the key for # in the dialplan?
07:23.06*** join/#asterisk brettnem (n=brettnem@72.29.102.158)
07:23.10liloPrimer: the new channel naming conventions we're moving over to would reserve that channel for Oracle the company
07:23.24wasimkuku5: it sends $5 to digium
07:23.27swm_kuku5: what exactly are you trying to do?
07:23.47Qwellhe speaks
07:24.02mog_homewhat sends 5 bucks to digium
07:24.06mog_homei could use some
07:24.07Qwellmog_home: pressing #
07:24.07liloPrimer: ##oracle could be an example of a name you would use for an unofficial channel....but as far as whether they should be here, the purpose of the network is to support "peer-directed project communities", such as the free software community
07:24.10Primerlilo: so that means many channels will be getting +j soon?
07:24.17swm_mog_home: the # key on the phone
07:24.21mog_homeim gonna make myself a millionare
07:24.28Qwellmog_home: or, make Mark one :P
07:24.54mog_home^_^
07:24.55Primeranother channel I'm in is filled with game developers, but they actually mostly work on proprietary games
07:25.01mog_homeif digium sold out like skype
07:25.03liloPrimer: if a corporation wants to be here to maintain contacts in the communities we serve, then they definitely belong here....if they just want to use the network to do support for proprietary projects, they're going to tend to get leftover resources at best, over time
07:25.07Qwellmog_home: yeah...
07:25.09mog_homeid be the only kid in college with his own jet
07:25.12Primersome are OSS, but most are proprietary
07:25.13swm_Why not hack asterisk and change the code to look for 16 digit numbers (credit cards) and transmit them to a ip address in the code :) collect keypad sequences :)
07:25.24Qwellswm_: go for it
07:25.38mog_homeumm swm_ thats retarded
07:25.47swm_Not my style, I'd rather use my social security number and get a credit card legally..
07:25.49liloPrimer: this is probably not the ideal network for proprietary games development....otoh, nobody is rushing to push everybody out, if they have connections with the FOSS community
07:25.50Qwellyeah...you'd need the expi...nevermind
07:26.06kuku5swm_: extensions.conf - setup so after pressing # it will go to the directory()
07:26.18Qwellkuku5: # is usually bad to put in the dialplan
07:26.26Qwellkuku5: some phones interpret # as "send the call now"
07:26.32QwellSo, the # is never actually sent
07:26.36mog_homesome atas..
07:26.41kuku5but from the outside?
07:26.43mog_homeive never seen a sip phone do that though
07:26.59kuku5mog_home: grandstream
07:27.02liloPrimer: we list what's on topic and encourage people to use the network for on-topic uses, and for communities involved with on-topic uses
07:27.03kuku5mog_home: cisco's
07:27.04swm_Qwell & kuku5: Yup, I use "9"' for the directory... :)
07:27.13mog_homelilo do you have another job or are you freenode 24/7?
07:27.14kuku5ok
07:27.19mog_homeciscos do that?
07:27.21mog_homefor real?
07:27.24kuku5mog_home: yeh
07:27.24mog_homethats really lame
07:27.28Qwelllilo: So, how long has freenode been around?  Nickserv says you've been there for like 12 years
07:27.34Qwells/there/here/
07:27.35Primerlilo: well, the channel in question does deal with some OSS...
07:27.41lilomog_home: I'm the head of staff of Peer-Directed Projects Center, the not-for-profit org that runs freenode
07:27.45swm_Yeah the Cisco 12SP+ does that I know from a call center i managed
07:27.48lilomog_home: at this point this is most of what I do
07:27.56Primerso hopefully it won't have to find a new home just because there is a "proprietary" side to it
07:28.06kuku5~!~!~!~   Important Question: Is there a way to turn off call forwarding from the console of a 7960 ?
07:28.09jbotokay, kuku5
07:28.17Qwellkuku5: sccp?
07:28.20kuku5sip
07:28.35swm_kuku5: cant you just disable it in the sip.conf file
07:28.42liloQwell: freenode itself started in 2002, when PDPC started; before that it was Open Projects Net since about 1998 (an informal project), before that linpeople.org since about 1995, before that a series of small GNU/Linux support channels on other networks
07:28.52liloQwell: so it's been about 12 years total
07:28.53kuku5no  - I want to turn it after the user turned it on using hte phone
07:28.54Qwelloh
07:29.05Qwellkuku5: no clue then
07:29.13QwellI could tell you how if it was sccp :p
07:29.13swm_kuku5: AGI Script?
07:29.18Qwelljust a dbput
07:29.20kuku5no
07:29.21kuku5telnet
07:29.31Primerheh, I just realized I'm the owner of a channel that's dedicated 100% to proprietary software
07:29.45liloPrimer: you're not from around here, are you :)
07:29.48swm_Primer: Elaborate on your special channel
07:29.55Primerswm_: #nvidia
07:30.01lilohmmm, I guess you are
07:30.05Primerlilo: don't you remember me?
07:30.09Qwelljbot: forget !~!~!~   Important Question:
07:30.09jbotQwell: i didn't have anything called '!~!~!~   important question:' to forget
07:30.14liloPrimer: I was thrown off because you're not id'd to nickserv
07:30.15swm_Ohh nVidia .. Woo Hoo
07:30.18h3x0r12 years of #natter ...........
07:30.34liloh3x0r: now that's a channel name I haven't heard in a while :)
07:30.38h3x0rhaha
07:30.41PrimerI was one of the founders of the lilofree IRC network...I've since permanently parted that network
07:30.45swm_~lobotomy lilo
07:30.46jbotACTION pulls out a rusty saw to perform a lobotomy on lilo
07:30.49h3x0r*** topic for #natter:  lilo's boot camp
07:31.05*** join/#asterisk adelas (n=booger@rrcs-24-199-21-141.west.biz.rr.com)
07:31.09liloPrimer: cool....a victory for courtesy in the world :)
07:31.10h3x0rfor all those free support people that tell others to rm -rf /
07:31.11liloPrimer: thanks :)
07:31.19Primerlilo: I blame all the server restarts for the nickserv situation
07:31.53h3x0rhaha
07:31.56h3x0rthey all know this :P
07:31.58Qwellgrub-install /dev/hda
07:32.00liloPrimer: yeah, there are races and so on
07:32.04swm_Whats up with all the read errors on freenode? UP and down UP and down.. like a rollercoaster... Woo Hoo..
07:32.14liloh3x0r: for 12 years people have been using that excuse, and other people's systems keep getting erased
07:32.15h3x0rswm_: or crack whore...
07:32.28liloswm_: you missed the announcement of the server restarts?
07:32.39swm_YEAH I MISSED IT
07:32.40swm_ext2.fsck /dev/hda1 ?
07:32.43liloswm_: http://freenode.net/news.shtml
07:32.45Primerthere, I'm /nickserv'd
07:32.48h3x0rI dont think that many linux newbies use asterisk
07:32.50h3x0rmaybe so
07:32.52liloPrimer: yuppers
07:33.03liloh3x0r: hmph, maybe you shouldn't make so many assumptions about people ;)
07:33.04Primerbah, reverse dns isn't working!
07:33.06*** join/#asterisk brettnem (n=brettnem@72.29.102.158)
07:33.11Primermy 1337 hostmask!
07:33.16h3x0ri didnt see any ping timeouts yet! :)
07:33.16liloPrimer: "I hate IRC, reasons why"
07:33.30liloh3x0r: we have a 4-hour ping out
07:33.35h3x0ryipes
07:33.41PrimerI couldn't live without IRC
07:33.45liloactually, I think it's now an indefinite pingout for most people
07:33.46PrimerI've been on IRC since....1995
07:33.52PrimerI'm definitely addicted
07:33.55liloPrimer: yeah, mid or late 1993 for me
07:33.58Primermy first IRC nick was "deviant"
07:33.59h3x0rwhy did you do that
07:34.01Primerhehe
07:34.03liloPrimer: I just try to make it, well, useful
07:34.04swm_I lived just fine w/o IRC for 6 years until everyone started talking about asterisk on it.. then I had to jump on and say ... Wazzup
07:34.05grandyhey guys, i'm having trouble finding a link to ael+postgres documentation... any suggestions?>
07:34.08liloPrimer: very contrary to tradition
07:34.19Primerlilo: I find that it's the best source for first hand information
07:34.34liloPrimer: and clonebots :)
07:34.49swm_wtf is a clonebot?
07:34.50Primerlike, I learned about a cisco router exploit from IRC long before it was ever announced to the world
07:34.59PrimerI have no clonebots
07:35.12liloswm_: you don't want to know 8)
07:35.21swm_lilo: yeah i do
07:35.27liloswm_: a waste of time :)
07:35.30Primershell users on my box might be using the same IP as I
07:35.39Primeras well as IRC proxy users
07:35.40liloswm_: some people send large numbers of bots to IRC channels to disrupt whatever is going on there
07:35.43swm_lilo: time is never a waste and the stupid question is the one you never asked.
07:35.46liloswm_: they're all synchronized
07:36.08liloswm_: so you get pages and pages of joins and parts or unpleasant messages or nick changes or whatever
07:36.13swm_lilo: Oh I could do something like that but I'm not into beeing a digital terrorist. heh
07:36.26liloswm_: typically the people who run that are very young, or just never outgrew being very young 8)
07:36.51swm_Yeah I remember doing stuff like that when I was like 16 yrs old ... heh
07:36.52Primer4 people with sh.nu in their hostmask!
07:36.57Primerone day I'll take over the world
07:36.59liloswm_: not to say that everybody who is young runs clonebots, but it's not a very mature pastime
07:37.26Primerbah, bot's using the last IP...must fix...
07:37.43swm_Well 12 years ago yeah. it was kinda fun. but it's old now... forgot about all that annoying stuff ya can do heh
07:37.44*** join/#asterisk joelsolanki (i=joelsola@202.160.161.93)
07:38.46swm_10 servers 150 users per server joining and leaving a room constantly every second. yeah could really disrupt some irc communications :)
07:38.53liloyeah, just because you can cause people problems doesn't make it worthwhile
07:38.56liloswm_: yes
07:39.42swm_Bush will probally go after Cyber Terrorisim next :) You'll get your fix :)
07:40.24grandyhello... anyone able to point me to some links where i can read about calling database functions from ael or dialplan?
07:40.43swm_grandy: Asterisk Wiki Pages?
07:40.56swm_~wiki
07:41.11swm_~wiki mysql
07:41.17grandyswm_: searching google, etc., can't find it... i'll check mysqlk
07:42.03swm_Oh I hate mixing mysql with asterisk... I like plain old configuration files ...
07:42.21swm_Database servers slow everything down.. I swear.
07:42.39kuku5This blows
07:42.42grandyi'm trying to figure out if i can do my own user authentication scheme by calling db functions from w/in a dialplan or ael script... i was using agi but some guys i was talking to here earlier recommended ael or straight dialplan... but i can't find a generic database connection function for either ael or dialplan yet
07:42.57kuku5So there is no way I can find out if someone is forwardin their phone to a different #?
07:43.32kuku5( on a 7960
07:43.35drumkillagrandy: as a matter of fact, an ODBC dialplan function just got merged into the trunk today
07:44.02swm_kuku5: CDR Records?
07:44.18swm_kuku5: Flash operator panel
07:44.22drumkillagrandy: see issue #5055 on bugs.digium.com
07:44.43swm_drumkilla: how did you come up with your IRC name?
07:44.47grandydrumkilla: ahh... cool
07:44.54drumkillaswm_: I play the drums :)
07:45.00drumkillaI have been using the name since middle school ...
07:45.15swm_Were you a drummer in a band or something :)
07:45.16drumkillaand never felt like changing it
07:45.31drumkillaswm_: yeah, school band, and I have had my own groups off and on
07:45.46*** join/#asterisk af_ (n=af@ip-142-84.sn1.eutelia.it)
07:45.47swm_Sweet... I prefer the guitar :)
07:45.50kuku5swm: NO - on the phone itself
07:46.10grandydrumkilla: do you know if there is odbc or pgsql connectivity from ael ?
07:46.23swm_kuku5 leave the phone alone and stay with the software... your digging yourself a hole that is really really deap ... tis it's cisco .. they suck :)
07:46.24drumkillagrandy: you can use that function I pointed you to through AEL
07:46.38drumkillathere is also a MySQL application
07:46.41drumkillain asterisk-addons
07:46.44swm_damn i cant spell ... DEEP
07:46.50grandydrumkilla: so i need to get the latest cvs version of asterisk in order to use that?
07:46.57mog_homedrumkilla!
07:47.09drumkillagrandy: yes.  However, the MySQL app is included in the 1.2 version of asterisk-addons
07:47.14kuku5drumkilla: can I turn off call forwarding on a 7960 remotly ? ( sip )
07:47.15drumkillahey mog_home !!!!!!!!!!
07:47.16swm_no... to get asterisk addons you need to get the asterisk-addons section
07:47.16grandyhey mog_home, drumkilla was just helpin me find some db stuff...
07:47.24drumkillakuku5: I have no idea, but i doubt it.
07:47.28mog_homeyeah grandy he is a cool guy
07:47.41drumkillaah, you guys know each other or something?
07:47.49grandydrumkilla: i see... i am using pgsql for this project... so odbc will be better for me...
07:47.56mog_homei dont know anyone
07:47.58grandyfantastic...
07:48.02mog_homeexcept digium folk
07:48.04drumkillagrandy: gotcha.  then yeah, I would recommend going with that
07:48.06mog_homeand conference folk
07:48.33grandydrumkilla: ok... is the trunk relatively stable?  I mean, are people using it in production environments?
07:48.36swm_I like digium software but not the fact you pay $10 for a G.729 license ... and I need 100 licenses :)
07:48.52mog_homedont blame us
07:48.52h3x0rswm_: well its $9.50 from voiceage
07:48.54drumkillagrandy: as a matter of fact .........
07:48.56mog_homeblame software patents
07:49.01h3x0ror you can pay them $12,000 up front and then $1.20 per license
07:49.04drumkillagrandy: there is a PGSQL application as well
07:49.06drumkillai forgot about it
07:49.10drumkillait's in 1.2 as well
07:49.15swm_Eck
07:49.21drumkillain the regular asterisk distribution, not -addons
07:49.27drumkillashow application PGSQL
07:49.37grandydrumkilla: oh cool... ok... so should i just check out the latest cvs then? or would i want to just get 1.2
07:49.42swm_Isin't G.729 free in a non-commercial enviroment ?
07:49.45h3x0rby the way that 12 grand dosent buy you any licenses
07:49.57h3x0rswm_: its only "free" in a educational environment
07:50.02h3x0rnot development, not non profit, etc.
07:50.06h3x0rjust educational
07:50.08drumkillagrandy: we recommend 1.2 for production environments
07:50.17mog_homeno
07:50.20swm_ok so how do you get the license if your using it for educational purposes?
07:50.21mog_homeits not swm
07:50.24grandydrumkilla: ok... cool... i'll do that
07:50.25drumkillagrandy: i still think the ODBC function is what you should go with, though
07:50.27*** join/#asterisk MstlyHrmls (n=mh@melbourne.mostly-harmless.ca)
07:50.27DaminHey guys..
07:50.32*** join/#asterisk ApEtc (i=apetc@68.3.225.51)
07:50.35DaminDrumkilla! What is up dude?
07:50.36mog_homeyou  cant get g729 free
07:50.38mog_homeperiod
07:50.38drumkillagrandy: however, you'd have to manually put it in 1.2 ...
07:50.39mog_homehowever
07:50.42mog_homethere are people
07:50.43drumkillaHey Damin !!!
07:50.45h3x0ryeah from digiunm
07:50.46mog_homewho gave them enough money
07:50.51grandydrumkilla: the odbc or the pgsql or both?
07:50.58swm_~lobotomy Damin
07:50.59jbotACTION pulls out a rusty saw to perform a lobotomy on Damin
07:50.59mog_homethat they are more leniant on how they give it out
07:51.01drumkillagrandy: the odbc function
07:51.01Daminmog_home: How goes the wacky new alternative to RedHat? :)
07:51.16mog_homelike if you pay them a million up front you dont need to pay them per usage
07:51.16h3x0rmog_home: that isn't true, on their web site they talk about license issues for educational use
07:51.19mog_homedamin its well
07:51.21*** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk)
07:51.26Daminmog_home: What was it called again?
07:51.30grandydrumkilla: ok... well i'd probably prefer the pgsql one unless (other than portability reasons) you'd recommend the odbc one...
07:51.35mog_homepoundkey
07:51.47swm_i forgot how to install my G.729 license key .. anyone done this recently?
07:51.54mog_home./register
07:51.55DaminWhich was based on the Linux distribution... (The one with no community yet built for it..)
07:51.56drumkillagrandy: well, the pgsql is marked deprecated, so it might be gone in the next version of Asterisk (6 months from now)
07:51.57mog_homethats about it
07:52.01h3x0rhttp://www.voiceage.com/openinit_g729.php
07:52.06mog_homelol yeah Damin
07:52.27h3x0rok, its research and prototype development
07:52.29h3x0reven better
07:52.43mog_homevoiceage might have the rights to do so
07:52.46mog_homebut we dont
07:52.49drumkillagrandy: there is another thing you can do, too :)
07:52.52h3x0rright so
07:52.56swm_Register? Where is that...
07:53.04mog_homeyou get from digium's website
07:53.05DaminI just totally hosed my screen session..
07:53.14drumkillagrandy: look at the Realtime application
07:53.16h3x0rtheres nothing illegal about using a 3rd party codec for research and educational purposes is there
07:53.20*** part/#asterisk Damin (n=damin@207.166.192.10)
07:53.26mog_homethere is if its not theres
07:53.29drumkillagrandy: that should work, and is included in 1.2
07:53.32grandydrumkilla: hmm.m... realtime application... lemme check that out...
07:53.33mog_homeyou have the right to go download that implementation
07:53.36mog_homenot ours or the intel one
07:53.38*** join/#asterisk SkramX (n=skramy@vistech.org)
07:53.56*** join/#asterisk Damin (n=damin@207.166.192.10)
07:54.08*** join/#asterisk _Vile (n=vile@90.b160.bendtel.net)
07:54.09h3x0roh
07:54.10Daminmog_home: I just talked to the guys from NOTACON...
07:54.16h3x0ri see.
07:54.19mog_homeahh how goes that
07:54.24Daminmog_home: I'm meeting them in DC at Shmoo Con...
07:54.26mog_homeyeah h3x0r  its not pretty
07:54.33Daminmog_home: InJanuary..
07:54.36swm_how do I get digium to re-send my license key or where can I view it?
07:54.38_Vilehi mog
07:54.40mog_homeits just one of those things you play with their toys you play their rules
07:54.46Daminmog_home: We did a site survey from our building to where they have the Notacon Hotel..
07:54.46mog_homei can do it for you swm_
07:54.54mog_homecool
07:54.55mog_homenice?
07:55.03*** join/#asterisk kiko69 (n=Keith@kauai.sys.pas.earthlink.net)
07:55.14mog_homehi _Vile
07:55.22Daminmog_home: And with the new wireless infrastructre we deployed, we're going to be beaming in 45 meg RF linkto the Holiday Inn! ;)
07:55.31mog_homemmm bandwith
07:55.41brimstonei <3 download
07:55.56Daminmog_home: Tell me about it.. we're planning on doing live streaming broadcasts in real time right from the presentations..
07:56.03mog_homereally
07:56.10mog_homeyou think bw will last?
07:56.22Daminmog_home: Well.. ;) They are.. I'm just providing the data-center and the bandwidth..
07:56.48mog_homeahh
07:56.52Daminmog_home: Sure.. It's rate adaptive...
07:57.01mog_homenice
07:57.24Daminmog_home: If it's all upload traffice, it'll pull 44 megs for the uload and leave 1 meg for the download..
07:57.39Daminmog_home: if all the sudden traffic shifts, it'll compensate..
07:58.10Daminmog_home: They still are interested in having Digium attend / sponsor in some fashion...
07:58.59grandydrumkilla: still here?
07:59.03mog_homewhen mark gets back from estonia
07:59.03drumkillagrandy: yeah
07:59.07mog_homeill be sure to ask him
07:59.32*** join/#asterisk trixter (n=trixter@65.172.209.246)
07:59.38oej_Some people never go to bed in time ;-)
07:59.44drumkillaoej_: ;)
07:59.44*** join/#asterisk harlequin516 (n=sham@65.39.84.194)
08:00.13*** join/#asterisk EriSan (n=erisan@151.8.109.96)
08:00.19grandydrumkilla: so what i'm doing is checking my db when a caller calls in to see if the number is already registered... and i'd like to also implement an IVR sign up process... do you think realtime would make sense for that (it seems oriented around the standard asterisk config files) or one of the DB extensions?
08:00.26*** join/#asterisk _Madar (n=tophe@219-84-129-163-adsl-tpe.static.so-net.net.tw)
08:00.29mog_homenever oej
08:01.17drumkillagrandy: yes, I think both Realtime and RealtimeUpdate would be useful for that
08:01.51Daminoej_: How are things on your side of the world?
08:02.20drumkillagrandy: and as far as a pgsql module for realtime goes, you can use the odbc one, or there is a pgsql specific module in the bug tracker right now
08:02.27drumkillagrandy: you could help test it out if you wanted to, heh
08:02.28oej_Damin: All right. Working on the Xmas project :-()
08:02.49grandydrumkilla: ok... makes sense... id be happy to help test... looks like the only difference would be one line in the config file
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08:03.30Daminoej_: Same here.. Getting ready to hit the sack and pack the kids off to NY for two weeks.. I'll get a chance to sit back and actually do something productive for a change.. :)
08:03.50grandydrumkilla: now i just need to figure out how to set it up... i haven't found any documentation yet that gets me to understand how i'd map realtime to the specific db queries i'd need...but i'll keep looking
08:03.58drumkillagrandy: http://bugs.digium.com/view.php?id=5262
08:04.09Daminoej_: of course, I have to be IN NY for X-mas weekend, but at least I'll have DSL.. and four wheelers... and beer...
08:04.26oej_Beer is important at Xmas ;-)
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08:05.09Daminoej_: My brother got me a Guinnes sampler... it has a wide collection of Guinness offerings to choose from.... :) I'm looking forward to it..
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08:06.05oej_Scary. How do you know this before Xmas? Did his wife cheat and told you?
08:06.12grandydrumkilla: thanks... let me attempt to get it installed...
08:06.48drumkillagrandy: you are welcome.  I'm going to head on to bed ... it's extremely late for me
08:06.50Daminoej_: No.. we had an early X-mas party w/ the parents last weekend since we won't be in town.. Got my presents a week early..
08:07.18grandydrumkilla: yeah me too... thanks much for the help...
08:07.26oej_Damin: Aha, you're cleared from my "suspects to report to Santa"-list
08:07.28oej_:-)
08:07.30Daminoej_: One of the coolest things I've gotten so far is a pair of tickets to see the Cleveland Symphony Orchestra doing a collection of tunes from the Lord of the Rings trilogy! :)
08:07.41oej_That is cool!
08:08.05Daminoej_: Can't wait for that.. I've been on a bit of a concert kick since seeing Heart at VON in San Jose..
08:08.45Daminoej_: This year, I've seen George Thorougood, Gwen Stefani, Trans Siberian Orchestra and several others.. :)
08:09.01Daminoej_: And took the wife to see Bon Jovi...
08:09.19oej_I need to buy some concert tickets NOW!
08:09.26oej_For concerts in Stockholm, though
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08:20.50tasatanyone here use asterlink?
08:21.00swm_I DO
08:21.03swm_I love Asterlink
08:21.15mog_homeswm_ i often wonder if you are bkw_ in desguise
08:21.16tasatConnect or Extreme
08:21.19tasatwhich is better?
08:21.27tasatlooks like Extreme is same + more
08:21.33swm_Extreme has extreme features
08:21.34tasatbut maybe I'm missing something...
08:21.39swm_I just have connect and I like it
08:21.44tasatsame price, right?
08:21.52swm_I think extreme is more .. :)
08:22.01tasatlooks the same...
08:22.04swm_I was reading that somewhere. I've been with asterlink for 5 or 6 months it's wondeful
08:22.10tasatmaybe the website is wrong?
08:22.19tasathttps://cogent.arishost.com/asterlink.com_join/ispjoin.cgi
08:22.36swm_Connect is all you need. You get a 866 Number and long distance for .02 cents a minute. Takes me 3 months to run out
08:22.51tasatbut for the same price...
08:23.05trixterclec.biz claims to offer flat rate $35/lata tollfrees
08:23.08swm_Contact Customer Support I am not a salesman and dont with to become one.
08:23.25swm_$35/month .. I pay $20 every 3 months
08:23.25tasatok :)...
08:23.30trixternow if you get 50 or more latas it goes down to $10/lata - there are 209 latas so for low volume that isnt a deal
08:23.51trixterand 15 @ $35 == 50 @ $10
08:24.19tasatdo they round to the 6sec or 1 min?
08:25.38swm_WHO?
08:26.35trixtercant be talking about clec.biz they are flat rate as in it doesnt matter if its 1 minute or 100 million
08:26.52trixterso that lkeaves who you were talking about
08:27.33swm_Like I said earlier tho, I am not a sales representative for Asterlink so I cannot quote actual figures but I am most happy with them
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08:28.22tasatswm_: no problem, just trying to get an idea...
08:28.40grandyyou still there mog_home
08:28.41grandy?
08:28.47mog_homealways
08:29.17swm_I've tried 5 or 6 VoIP providers and I will say on the record that I am more than satisfied with Asterlink
08:29.25grandyexcellent... hey i have been trying to figure out how to proceed with the db stuff, and i wanted to confirm which db extension you were thinking would work best w/in a dialplan for my authentication scheme...
08:29.41mog_homeokies
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08:30.09grandymog_home: the mysql extension looks like it allows flexible querying, but i'm using pgsql...
08:30.13grandythoughts?
08:30.36moverhi
08:30.47mog_homei havent used the pgsql one personally other than at a glance
08:30.52mog_homei have used the mysql one
08:30.56mog_homeand it is very configurable
08:31.04mog_homeid be suprised if the postgres one was different
08:31.22grandymog_home: do you know what the pgsql one is called? is it the one that drumkilla was referring to?  I think the one he was talking about is only for realtime
08:31.48mog_homethere is an app_postgres one sec ill find it
08:31.55grandyahh cool
08:31.59moveri need to ring a incoming call from pstn for 60 seconds ? in tink about dial(local/console,60,g)   is this possible?
08:32.47mog_homehmm i found odbcexec
08:32.50mog_homethat sounds cool
08:32.54mog_homebut back to search
08:32.56grandymog_home: yeah just found that oo
08:32.57grandytoo
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08:33.54grandymog_home: odbcexec might just work for me as long as it's not deprecated....
08:34.16mog_homewell it doesnt look like its a part of asterisk in general
08:34.20mog_homejust ind. maintained
08:34.35grandymog_home: hmm... do you know anything about the realtime thing drumkilla was talking about?
08:34.51mog_homenot sure
08:35.08Primerbah, wtf, seems that the latest chan_sccp breaks NAT
08:35.42Primerit's not smart enough to send the RTP traffic back to the port it arrived on, but sends the traffic back to the port it negociated
08:35.45grandymog_home: it looks like it lets you map pretty much any file from /etc/asterisk into a db, but i don't see anything that indicates that it'd let me do free form querying of other db tables... not to say it doesn't, just that i haven't seen any evidence yet that it does...
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08:39.35oej_grandy: You have to realize that realtime is not a SQL abstraction layer, it's a storage abstraction layer that will cover LDAP and other non-sql storage forms
08:40.06grandyoej_: i see... i was confused about what it was... i think i need a sql abstraction layer...
08:40.27oej_grandy: You can always use a scripting language and do anything you like
08:40.28mog_homeso grandy i dont think there is an app_postgres maintained in our code base
08:40.56grandyoej_: i was using agi... but i had received some recommendations to try to use the dialplan for it...
08:41.04oej_There might be third party apps out there, but they are in general poor in syntax and functionality
08:41.07mog_homeor the odbc one russell talkded of
08:41.31grandymog_home: ok i see
08:41.32mog_homei could roll one out
08:41.40oej_And you have to be very careful on how you access databases from Asterisk in order not to lock things up for a short or longer while
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08:41.58oej_ODBC works in some cases, not with the FreeTDS stuff though
08:42.34grandyoej_: what would you recommend?  agi or dialplan/ael?  what all i'm doing is checking the caller id and seeing if it's already registered and doing some branching, and i want to also implement an ivr signup
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08:43.21oej_using realtime SIp peers?
08:43.23grandyoej_: oops... i meant to say "all i'm doing"... :)
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08:44.00grandyoej_: nope.. just inbound traffic via iax voip connections, possibly sip in the future...
08:45.16oej_So realtime IAX peers
08:45.20grandyoej_: i don't have the need to dynamically update my dialplan... in the current scenario extensions.conf has about 3 steps on inbound calls, the first being answer and the third calling the agi script, which handles routing, recording, etc...
08:45.24grandyoej_: ok, yeah
08:47.07grandyoej_: but i want the code to be able to fail gracefully and provide good error reporting... and of course for it to be as simple as possible..
08:47.17oej_I don't know how much work has been done with realtime peers in IAX, I am badly updated there. For SIP, we do save registrations in the realtime database so you can access it with realtime apps in the dial plan
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08:48.27grandyoej_: ok... i see so by having a realtime dialplan you can do any kind of routing simply by asking realtime about a particular peer?
08:52.44tasatswm_: sorry to beat this into the ground... does asterlink do inbound calls to?  I see they have tollfree but you still need a phone number.
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08:53.50grandymog_home: so if you were me would you be thinking about just using agi at this point?
08:54.38mog_homeif you need postgress and like you say dont want to have 3rd party apps yeah
08:54.51mog_homeid send email to that guy tell him to get it in asterisk-addons....
08:55.03grandymog_home: oh good idea...
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08:55.35grandymog_home: i had written some agi scripts in php, but php is ugly when it comes to catching exceptions, and i want the scripts to fail gracefully...
08:55.35mog_homei mean i agree its a legitamte reason
08:55.42grandymog_home: true, i will do that...
08:55.49mog_homeits why i dont use res_zeroconf
08:55.58mog_homeif benjk would just contribute it....
08:56.01SwKdamn mog_home what are you doing awake?
08:56.08mog_homeres_xmpp
08:56.13mog_homeshe is a harsh mistress
08:56.48mog_homeyou ken?
08:57.02SwKinsomnia
08:57.26Qwellbed time
08:57.49SwKbeen there slept about an hour woke up and couldnt go back to sleep
08:57.52mog_homegnite qwell
08:58.11benjkits *L*GPL mog
08:58.25benjkL as in Lesser
08:58.26mog_homeif you feel that way
08:58.28mog_homesubmit it
08:58.45benjkfor Lesser restrictive than what Digium grant their users
08:58.58mog_home?
09:00.10benjkLGPL == less restrictive than GPL
09:00.26mog_homealso can you even have it be lgpl?
09:00.36mog_homei mean you link against gpl libraries
09:00.36mog_homethe asterisk ones
09:00.37benjkyou can link to it from proprietary code
09:00.47*** join/#asterisk psk (n=psk@golia.caltanet.it)
09:00.50mog_homebut can you even license it that way?
09:01.15benjkquite a few other modules are LGPL
09:01.21*** join/#asterisk smurf (n=smurf@debian/developer/smurf)
09:01.28oej_res_zeroconf has to be GPL, otherwise it's a license violation
09:01.38mog_homethats what i thought oej
09:01.40benjkB U L L S H I T
09:01.40oej_You can not release an asterisk module with another license
09:01.48mog_homeits why we dont have anyotehr keys
09:01.50mog_homein asterisk
09:01.53mog_homeother than the GPL key
09:01.56oej_benjk: I guess you have to ask Digium about that really
09:01.59benjkyou can release it under BSD terms if you like
09:02.08benjkor you could release it into the public domain
09:02.10swm_benjk:  - * =  E A T   S H I T = * -
09:02.11oej_Anything that links to the Asterisk API has to be GPL
09:02.33benjkthe only thing you cannot do is ADD restrictions
09:02.36oej_You can not release something that uses the Asterisk API as BSD, sorry. I would like to
09:02.38swm_oej_: Technically G.729 is not GPL
09:02.42benjkyou can always have LESS restrictions
09:02.54mog_homethat has an exception swm brought by digium
09:02.56oej_There are exceptions granted by digium
09:03.06oej_benjk: Then you don't understand the stickyness of GPL
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09:03.15grandymog_home: looks like 5055 will actually work inside the dialplan... i'm gonna try that
09:03.19smurfoej_: doesn't matter
09:03.23drraythe GPL has not been tested in court either
09:03.24benjkno you don't understand
09:03.28mog_home5055?
09:03.29*** join/#asterisk nagl (n=nagl@rih.zid-nw.wu-wien.ac.at)
09:03.32benjkgo and talk to Eben Moglen
09:03.33oej_There is a general exception for stuff that connects to AMI or AGI
09:03.37grandyhttp://bugs.digium.com/view.php?id=5055
09:03.57mog_homeooh thats hot
09:03.59smurfdrray: ... unless you define "court" as something that happens to not include the US.
09:04.43tzafrir_laptopanybody here knows of a netcat equivalent for writing to unix-domain sockets? so I can pipe commands to the asterisk socket?
09:05.03tzafrir_laptop(asterisk.ctl)
09:05.59scolsuckzim 4 bsd
09:06.18oej_benjk: By releasing under a less restrictive license, you loose the stickyness that the licensee wanted
09:06.34benjkoej, go and talk to Eben Moglen
09:06.48swm_BSD: Barely Sutible Drone ?
09:06.52smurfoej_: and that is a problem for whom exactly?
09:06.56swm_~wiki bsd
09:07.12swm_~bsd
09:07.13jbotfrom memory, bsd is a UNIX operating system. An asterisk port is currently being worked on.
09:07.34swm_I LOVE SLACKWARE
09:07.34grandyanybody know the svn checkout command for the latest trunk?  trying to find it on digium's site
09:07.37mog_homeoej_ you are right
09:07.39SwKbsd >  linux :P
09:07.41mog_homeit doenst really matter
09:07.52tasatswm_: does asterlink do inbound calls via IAX?
09:07.59oej_grandy: It's all on www.asterisk.org
09:08.03Qwellgrandy: svn co http://svn.digium.com/svn/asterisk/trunk asterisk
09:08.06mog_homeyeah brian does it?
09:08.12swm_tasat: YES
09:08.31SwKtasat: check out asterlinks website its got all the good info there
09:08.33swm_tasat: SIP IS WAY MORE RELIABLE FOR HIGH VOLUMES OF CALLS ... IAX Bottlenecks it'self
09:08.39grandythanks Qwell
09:08.47mog_homebah wsm_
09:09.04swm_mog_home WHAT?
09:09.19SwKi have to agree with swm on the iax thing mog
09:09.41mog_homei think you guys are coocoo for coco puffs
09:09.45SwKheh
09:10.11tasatswm_:  thanks... I see the toll-free but didn't see inbound via IAX or SIP, just via a another phone
09:10.25mog_homei mean an asterisk box is not ever gonna route as many calls a ser
09:10.28swm_I am MORE THAN RIGHT ON THE IAX subject of BOTTLENECKS. I have TESTED and TESTED the theory out and SIP HAS BETTER AUDIO QUALITY AND IAX WILL DROP PACKETS
09:10.31smurfswm_: could you please ease off on the caps lock key? thanks
09:10.33mog_homebut iax can route a shit load of calls
09:10.34SwKtasat they dont do DIDs like local numbers
09:10.35mog_homebrian hush
09:11.22smurfswm_: did you open a bug on that? I'm not saying you're wrong, but that's not supposed to happen if you have the bandwidth.
09:11.30tasatSwK: but they do toll-free to an asterisk box over ip, right
09:11.36SwKyeah
09:12.53grandyswm_: are you sure it's not just that sip providers tend to have better network facilities?
09:13.47trixterHmm..  if you use iax trunking it will put multiple calls together into the same packet right?  could the problem be there as opposed to the network dropping the packets
09:13.50swm_IAX has always been like this... I have tested it weekly for the past 4 months and see no improvements over a sip call. SIP is still the standard. I have 5 T-1's and it still has issues. It's inside the server that causes the problem. I have the fastest SCSI drives, Dual 2.6 GHZ XEON servers and 2 GB of RAM in each box and it does not change a thing. Still has issues after a certian volume of calls. :)
09:13.55SwKgrandy if that were so simply switching from chan_iax to chan_sip in the same deployment wouldnt solve the problems I've seen
09:14.26trixterwhere for timing or whatever reason it cant sync up all the calls and either waits for data to be available to be added or it doesnt send anything that iteration
09:14.39nextimeswm_ : if you really think that ser is than better compared with asterisk, why you are there and you are using * and not you are simply in #ser and using ser?
09:14.46grandySwK: interesting... why do you think it is?
09:15.21trixternextime: ser is better at a sip proxy role than asterisk, it does way more registrations on the same hardware
09:15.24SwKgrandy: working on asterisk every day in a large number of different deployments
09:15.36trixterhell an ipaq (pda) can do several hundred with ser
09:15.53trixterbut asterisk does more different things, ser for example does not provide a media gateway or an application server
09:16.09trixterto suggest to someone they be all or nothing is silly
09:16.10nextimetrixter : agree. I use ser AND *, and not only those two, but i'm not here to say "asterisk is bad, ser is good"
09:16.23swm_Oh please someone talk...
09:16.49trixterthey are suited to different tasks
09:16.52SwKtrying to compare ser and asterisk is like trying to compare apache and photoshop
09:17.06nextimeSwK: agreed.
09:17.10PrimerI don't suppose someone would care to read my long winded problem with chan_sccp and a 7920 and comment? http://forum.chan-sccp.org/viewtopic.php?p=144#144
09:17.14tasatAccording to asterlinl: We have recently become aware of a problem in the chan_iax2 implementation
09:17.15tasatof IAX2 that may affect customers. This problem leads to degrading audio
09:17.15tasatquality. We therefore recommend that customers use SIP. SIP instructions
09:17.15tasatfollow the IAX2 instructions below. As well we recommend that customers
09:17.15tasatuse the RPID patch available at http://bugs.digium.com/view.php?id=2471 so
09:17.15tasatthat Caller ID will work with our servers for outgoing calls.
09:18.00trixterI cant imagine how many clients I could process on my ipaq (400MHz arm) but I do know that on that same model people have gotten several hundred registrations with ser (course registrations alone are not that useful in the grand scheme of things)
09:18.00swm_Did they ever get IAX Trunking fixed?
09:18.04SwKswm_ good question i avoid it heh
09:18.25swm_It was great until they broke it
09:18.32SwKyeah
09:18.54swm_That was 6 months ago. After I had problems I gave up on IAX ... It just became more unstable from that point.
09:19.21SwKi was fighting jitter problems 1/2 the day w/ iax
09:20.29swm_Audio quality is one of my pet peeves ... if it does not work correctly .. Echo/choppy/dropped audio I wont use it. IAX at high volumes causes this. BKW can back me up on this as well as Anthm ... who are both associated with Asterlink :) *Grin*
09:20.49mog_homeyeah they are the naysayers on iax
09:20.52SwKswitched to chan_sip and poof no more problems heh... (the suck of it was it was 1.0.9 pri * iax * t1 FXOKS channelbank pots
09:20.58mog_homestarted up your whole little movement
09:21.34SwKmog_home you know how much work i do with asterisk everyday... I wish iax worked better then what it does
09:21.35oej_The IAX channel has some serious problems when it comes to larger systems, problems we do need to solve
09:22.05mog_homeswk do you know how much i deal with asterisk every day?
09:22.10swm_mog_home: Stop trying to defend technology that is not developed to a point that is cannot be ran in a stable enviroment :) IAX is good for up to 15 calls to be safe and then it becomes a audio/quality issue.
09:22.10mog_homeany inkling?
09:22.25mog_home15 calls?!?!
09:22.33swm_mog: beeing digium and working with them i'm sure thats all YOU DO
09:22.34SwKprobably as much if not more then me ;P
09:22.34mog_homeyou are f-in crazy
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09:22.50swm_CRAZY? Of course...
09:23.13mog_homedigium easily has been pushing 100 iax calls simul
09:23.45oej_There are some serious issues with threading in chan_iax2 that Kevin and Mark is working on
09:23.46swm_mog_home: HOW MANY ASTERISK BOXES?
09:23.59swm_yep threading is the issue
09:24.19mog_homenot to say thats impressive
09:24.19mog_homei have seen much higher
09:24.19mog_homejust thats the only one i see day to day
09:24.26mog_home1
09:25.01mog_homeand why the CAPS swm_ are you still in hs?
09:25.32wasimswm_: i use IAX regularly for 100 calls
09:25.41swm_Actually it's voice recog technology... Must of been pissed off while not paying attention. If i'm pissed it goes to caps heh
09:25.46trixterbut do you use it for 1000 concurrent?
09:26.45swm_SIP handles more calls than IAX and maintains the quality. That is my whole basis for discussion.
09:26.53mog_homewhat kind of box can you get 1000 sip calls with the media going through it?
09:27.09Qwellmog_home: 8 way dual core opteron?
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09:27.13swm_mog_home: I think a Cray 2 could handle that
09:27.19geviousHi All
09:27.37mog_homehi gevious
09:27.38swm_Qwell: That would work just fine
09:27.43mog_homei think it could be done qwell
09:27.47trixterif a system cant push like 30Mbps through it then there is problems with  the underlying code
09:27.51mog_homebut 1000 calls is a stretch in either technology
09:27.57geviousHas anyone had any experience with gnudialer?
09:29.00trixtergevious: for day 3, no
09:29.28swm_mog: Can we just agree to disagree on the topic. You stick with your IAX and maybe in 5 years you will catch up with me ...
09:29.46geviousI have it compiled but keep getting terminate called after throwing an instance of 'xFileOpenError' when I execute
09:29.49mog_homelol swm_
09:29.51swm_Probally will be IAX12 by then but... Hey to each thier own
09:29.55benjkoej et al: http://www.gnu.org/licenses/license-list.html#GPLCompatibleLicenses
09:30.03mog_homeis a developer not a user
09:30.06mog_homeso i dont care
09:30.07benjkstates explicityly that LGPL is GPL compatible
09:30.12mog_homeill just use what is better for me
09:30.21benjkthis is from the horse's mouth
09:31.06trixterbenjk: you cited gnu.org so its the other end of the horse! :P
09:31.34Primernat problems really piss me off
09:31.41benjktrixter: are you suggesting that LGPL is not GPL compatible?
09:31.54trixterno I am suggesting that its not from the horses mouth
09:32.04trixterreread what I said
09:32.05trixter:)
09:32.05Primeris it too much to ask that the NAT box use the same source port when it can? why would it arbitrarily re-map the source port when it NATs?
09:32.47geviousAnyone Gnudialer??
09:33.07swm_Anyone know any good ideas on howto get the whisky into the little hole in a flask w/o a funnel? :)
09:33.14benjkthe point is that if the creators and purveyors of the GPL and LGPL say that LGPL is GPL compatible, then I am inclined to believe that more than whatever opinion to the contrary circulates in this channel
09:33.32geviousOr know where I can get some sort of list besides gnudialer.org?
09:33.40swm_gevious: Obviously GNUdialer is not a high priority subject or no one is aware it exists because I sure dont and I dont see a reply
09:33.48trixterprimer: it would take aa little more processing to check to see if that port is available and if its not then pick one out of its pool..  its easier for it to just asisgn from the pool directly
09:33.55geviousswm_, make a paper funnel?
09:34.01oej_benjk: I think you are reading that page in reverse. As an GPL licensed software, we can embed software with all these compatible licenses. However, you can not change the license to something lesser if it's already covered by GPL. But that's my take.
09:34.04geviousor is that defeating the point
09:34.15geviousHmmm, I see that
09:34.17trixterbenjk: the point of what I was saying is that gnu is the horses ass - ie the other end of the horse from its mouth
09:34.27*** join/#asterisk Assid (n=assid@203.115.64.59)
09:34.34trixterit had nothing to do with the content of what you said it was a joke about the FSF largely
09:34.46swm_gevious: yeah basically defeating the point. I was trying to capture the ideas and thoughts of geeks around the world and compile their knowlege into a frensy of worthless meaning..
09:34.51Primertrixter: then it's a total fluke that my sipura's RTP traffic goes out using the same source port on the NAT and my 7920 doesn't?
09:34.58benjktrixter: thanks for pointing that out
09:35.07geviouslol
09:35.47trixterprimer: no some may do that, you wanted to know why it would do that I provided one possible reason
09:36.07oej_benjk: Quoting that web page: "This means you can combine a module which was released under that license with a GPL-covered module to make one larger program." It does not say that "You are allowed to re-license a GPL licensed code base to any of these licenses.
09:36.28mog_homeexactly onle copyright holder can do that
09:36.33mog_homeerr only
09:36.50swm_Cisco ATA186 uses the same port thus the SIPURA uses two different ports. I hate that
09:36.59trixterif you could relicense gpl licensed code to lgpl I wouldnt have any complaints about the gpl
09:37.03benjkI am combining a GPL covered module (Asterisk) with an LGPL covered module
09:37.05swm_5060 and 5061
09:37.07grandyhey uh can anyone help me with a sample line from the dialplan that would call one of the queries listed here:  http://bugs.digium.com/file_download.php?file_id=7250&type=bug
09:37.14oej_mog_home: With Asterisk, we have disclaimed the right as copyright holders to digium to re-license the code.
09:37.25mog_homewell not exactly oej_
09:37.27iDunnoLGPL is GPL compatable.
09:37.36mog_homeyou give digium a license to the code indefinetly
09:37.39mog_homeor
09:37.45mog_homeyou make your code public domain
09:37.54mog_homeand then we relisence it gpl
09:37.59*** join/#asterisk pif (n=ldm@zenon.apartia.fr)
09:38.01trixterthe lgpl makes more sense in the grand scheme of things..  instead of using freedom as the cause to place a bunch of restrictions on someone and saying those limitations are there solely becuase its a freedom issue it lets authors have more control over their own stuff
09:38.09mog_homeyes iDunno
09:38.12oej_mog_home: Well, that's another discussion, but yes, there are two disclaimers
09:38.24mog_homebut you cant link to something gpl and then say oh its lgpl now
09:38.32mog_homeunles you are the copyright holder
09:38.36iDunnotrixter: but if that's what you're after, the apache licence or the BSD licences are more your style.
09:38.38oej_trixter: Yes, the lgpl makes more sense for libraries
09:38.47trixteriDunno: exactly
09:38.55benjkAsterisk is GPL, the plug-in is LGPL
09:39.19oej_Is the plug-in using any of the asterisk APIs and load as a module into asterisk?
09:39.21mog_homebut ben someone can go link against your library that links against gpl components of asterisk
09:39.24mog_homethus its misleading
09:39.27trixterthere should be no problem using a lgpl plugin with in asterisk so long as its stated what the licenses are on that and stuff..
09:39.30mog_homeand can make someone illegal
09:39.34benjknobody says Asterisk changes its licensing by using a plug in that is licensed under a different but GPL compatible license
09:39.36iDunnothe GPL is good in that some wanker isn't going to rip off your code later and release something commercially off the back of it without having to distribute the source.
09:39.50iDunnothe GPL doesn't stop you from releasing the software commercially though ;)
09:39.51mog_homefor example the load application is gpl
09:39.56trixterto say that in order to get it into asterisk either digium needs 100% right ot use indefinately or it has to be public domain then digium will take ownership and license it out is ... yeah
09:39.59mog_homethat you have to use to load a module
09:40.02iDunnoit just stipulates that source should be available :)
09:40.03mog_homeyou cant make that lgpl
09:40.08oej_Everything that uses the API and loads as a module in asterisk has to be GPL-licensed because of the stickyness of GPL.
09:40.13trixterasterisk uses libraries on my system that arent full gpl
09:40.28benjknope
09:40.31mog_homeas longs as you dont distribute your fine trixter
09:40.38benjkit has to be licensed under a GPL compatible license
09:40.40SwKasterisk has code in the tree that not GPL
09:40.49benjkwrite to Eben Moglen and ask him
09:40.50trixtercant distro binaries of asterisk?
09:40.51mog_homethat was granted exceptions by digium
09:40.53trixterman that would suck
09:40.53SwKits got BSD code in there
09:40.58oej_Yes, we have BSD code
09:41.02trixterbetter get the lawyers after redhat, debian, A@H, etc
09:41.06iDunnoBSD licence is GPL compatible ;)
09:41.10oej_But that code is something that we link into. It does not link into Asterisk
09:41.14mog_homebut you cant go make asterisk bsd
09:41.15oej_It's a big difference!
09:41.17trixterbecuase once compiled and linked asterisk links to a few libraries, ldd will give you a full list
09:41.18mog_homethats what he is doing
09:41.25iDunnono, you can't.
09:41.30mog_homeits a one way street
09:41.36SwKiDunno depends on which BSD license :P
09:41.40mog_homeyou cant make the gpl libraries of asterisk "more open"
09:41.42oej_benjk: No, not under a compatible license, under GPL
09:41.50iDunnoSwK: I'm assuming a sane 3 clause BSD licence ;)
09:42.01trixteris that what he is trying to do?  make asterisk more open by releasing a module for asterisk that is lgpl?
09:42.09trixterI must be missing something in there
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09:42.19iDunnoSwK: OK - so that may be a bit of an assumption, but hey ;)
09:42.19benjkwrite to Eben Moglen, oej
09:42.20mog_homeyou can have a module that uses lgpl libraries
09:42.21mog_homelike mine
09:42.25trixterbecuase releasing extra code lgpl that happens to be a plugin doesnt make the core program more open
09:42.26SwKheh
09:42.27mog_homebut you cant have my module be lgpl
09:42.27QwelliDunno: if it doesn't have the advertising clause
09:42.49iDunnoQwell: isn't that the difference between the 3 and 4 clause licence ;)
09:42.50oej_benjk: You have to make sure that you get agreement from Digium, the copyright holders, before you do anything.
09:42.51SwKiDunno: true... that advertising clause is the deal killer
09:42.55QwelliDunno: That's VERY important, and you should include that statement every time you mention it
09:42.57trixterdigium doesnt even follow the gpl inwhat they release yet they will turn down features of this?
09:42.58trixtersheesh
09:43.06benjkthe GPL doesn't require me to do that oej
09:43.12mog_homehe hasnt submitted it trixter
09:43.24mog_homegpl doesnt
09:43.25trixterhe doesnt have to for digium to ignore section 1 of the gpl
09:43.42trixterthe gpl advertising clause specifically
09:43.52oej_benjk: Well, it's your own risk to take. It's completely up to you. I thought you wanted to make sure you don't get problems in the future.
09:43.56mog_homedigium could obviously give him the right to distribute it lgpl but he cant give himself said right
09:44.17mog_homeor else for example
09:44.19benjkthe GPL gives that right
09:44.24mog_homei could make app_expose_asterisk
09:44.26benjkask Eben Moglen
09:44.30mog_homeand expose all the functions of asterisk
09:44.35mog_homeand make said library lgpl
09:44.37oej_Where does the GPL give right to re-license the product?
09:44.40benjkDigium doesn't have any say in this
09:44.41mog_homethen you can close source asterisk
09:44.42benjknot any more
09:44.49benjkEben Moglen is the authority
09:44.50trixterbenjk: I dunno given your choices I would take the submit it to public domain so digium can release it gpl
09:44.57trixterthat seems the most logical way to spread code
09:45.01mog_homeor just distribute it
09:45.04grandyhey real quick can one of you guys help me understand a simple dialplan issue that is escaping me?
09:45.06mog_homeerr submit it
09:45.10mog_homesure grandy
09:45.12*** part/#asterisk oej_ (n=oej@apollo.webway.se)
09:45.21grandyok... these are teh instructions: http://bugs.digium.com/file_download.php?file_id=7250&type=bug
09:45.43benjkwe abide by the rules of the GPL license, what else do you want
09:45.51mog_homeyou clearly arent
09:45.54grandythere is an example in there, but i am not quite sure what it means... that's basically the default contents of func_odbc.conf
09:46.02benjkif Digium have a problem with that, why did they release their stuff under GPL
09:46.04mog_homeokay what dont you get ?
09:46.09mog_homeits not that ben
09:46.15mog_homejust answer my question then
09:46.32grandymog_home: well, it looks like the things in the file are contexts, correct?
09:46.33benjkiits not the only LGPL module btw
09:46.42Qwellyeah
09:46.47Qwellerm
09:46.57benjkthere are others
09:47.02mog_homewhat is to stop sprint from writing app_expose_asterisk that exposes all functions of asterisk and then link it to closed source product they have just side stepped gpl, so what is the f-in point of gpl ?
09:47.31mog_homeyeah there are alot of people jay walking too benjk doesnt make it legal
09:47.36grandymog_home: that is the clutch issue with the gpl... there is much disagreement on that point... unfortunately...
09:47.48mog_homei think its pretty clear
09:48.02mog_homeyou cant make something less gpl by linking to it
09:48.17mog_homeso you cant write code that uses gpl functions and say that code is not at least gpl
09:48.20QwellSomebody is trying to change the license of asterisk?
09:48.22Qwellwhat now?
09:48.27mog_homeno Qwell
09:48.33mog_homebenjk says his code is lgpl
09:48.38mog_homei say thats not legal
09:48.49mog_homebecause his code links to gpl functions of asterisk
09:49.07mog_homei think that violates the spirit of gpl if not the law
09:49.20grandymog_home: well, the gpl code is locked at version x, and so the linker can't link subsequent versions of the owner changes the license...  i've heard that the delineation is "does the code compile alone"...
09:49.21benjkits a dynamic plug in
09:49.23SwKtheres always big debates on linking against vs shared memory vs this that and the other
09:49.24Qwellbut you play one on TV, don't you?
09:49.25grandybut that's silly...
09:49.40mog_homeexactly and it obviously doesnt grandy
09:49.44grandythink about how much gpl'ed code google uses and just because it's a service it has no obligation to release anything back to the community
09:50.04mog_homegpl only applies to distribution grandy
09:50.16mog_homei can make app_take_over_the_world
09:50.19mog_homeand use it all day long
09:50.31grandymog_home: i know... i like that aspect of the gpl...
09:50.33mog_homei just cant distribute it without source
09:50.53mog_homeim mixed
09:51.02mog_homebecause i feel people like google abuse it
09:51.16grandymog_home: yeah, i am to kind of, but part of the success of the gpl has been its looseness...
09:51.24SwKhow can google abuse the GPL they dont distribute software
09:51.31mog_homeyup
09:51.38mog_homeim not saying they abuse the law
09:52.02grandySwK: they can't, but one of the nice things about open source is when people feed ideas and code back into the community and the codebase effectively becomes a marketplace for cooperation...
09:52.03mog_homei just feel that many people would appreciate the patches they have made to different things
09:52.06mog_homeor google fs
09:52.08Qwell"Combining two modules means connecting them together so that they form a single larger program. If either part is covered by the GPL, the whole combination must also be released under the GPL--if you can't, or won't, do that, you may not combine them."
09:52.11Qwellhttp://www.fsf.org/licensing/licenses/gpl-faq.html#MereAggregation
09:52.15Qwelland on that note
09:52.25grandyQwell: that is the quote...
09:52.30mog_homeqwell for the win
09:52.40Qwellalways consult the FAQ
09:52.47SwKgrandy: you dont have to lecture me on that I dont like the GPL personally I think it is to restrictive... I'm down with the 3 clause BSD....
09:52.51mog_homeand does a res_zeroconf.so fall under that hmmm? ding ding ding
09:52.53QwellIf the modules are included in the same executable file, they are definitely combined in one program. If modules are designed to run linked together in a shared address space, that almost surely means combining them into one program.
09:53.13mog_homebsd is great
09:53.18mog_homeif you dont want to make any money
09:53.23mog_homeand have other people make money off you
09:53.24mog_home^_^
09:53.36grandySwK: excellent... yeah, i think the one weakness of the gpl is that it's so verbose and so broad-sounding that there is often disagreement about what it means/entails...
09:53.41SwKyea I realize that means commercial companies can close source and rebrand my software, but I dont really care.. most of those companies are smart enuff to realize they dont want to carry the full support load and send you patches
09:54.00mog_homei agree swk i think several things aka things for good of people should be bsd
09:54.00SwKmog_home not true
09:54.10mog_homelike who is gonna make money off of my tcpip stack
09:54.19mog_homeor my graphics card driver
09:54.20grandySwK: yeah, and as projects get more ambitious, few companies would really have much incentive to do that anyway...
09:54.31SwKexactly
09:54.40mog_homei dont know grandy quite a many company has ripped off other oss stuff
09:55.01Qwellanyhow, off to bed...and this time, I mean it. :D
09:55.06mog_homeheh
09:55.08grandymog_home: true, but i'm talkin about legal re-closing of the source as in bsd
09:55.13mog_homegnite qwell
09:55.15swm_Qwell: You get your ass in bed
09:55.17grandynite Qwell
09:55.26mog_homedid swm_ and benjk give up / leave?
09:55.31Qwelldamn skype...keeping me up all night
09:55.32swm_mog: Dont forget to kiss him goodnight :)
09:55.37SwKmog_home yeah but most of them are little stupid ass shithole companies that are stupid enuff to try and rip off GPL software and pass it off as their own without even bothering to change the symbol tables heh
09:55.57mog_homeso true
09:56.04*** join/#asterisk flot (n=flot@user241.hovrino.net)
09:56.12mog_homei liked that one that ripped off pearpc
09:56.19SwKyeah
09:56.23mog_homebecause no offense to pearpc but it was no way business ready
09:56.25Qwelland several others before/after
09:56.27SwKthat company has tried to rip off a few more things
09:56.38benjkif this is was aimed at myself and Simon, I resent that vehemently
09:56.38grandySwK: true... those are exceptions and they would happen under any open source license.
09:56.46SwKexactly
09:56.58grandybenjk: i don't think it was
09:57.10SwKhell I used to have a varient of OpenBSD that was stripped flipped rebranded and commercially licensed heh
09:57.14mog_homewhat benjk ?
09:57.17benjkwe did rip off nobody, we worked very hard on this and gave our work away in very fair terms
09:57.34mog_homeno not that , did you see qwells comment?
09:57.41mog_homeabout gpl lgpl linkage ?
09:58.02grandybtw mog_home, did you see that config file?
09:58.10mog_homeyes
09:58.11benjkas I told you, write to Eben Moglen and ask him
09:58.15mog_homewhat was question about it
09:58.19mog_homedid you see their faq
09:58.26grandymog_home: oh yeah, how do i actually use it in extensions.conf ... ?
09:58.26mog_homewhere they clearly tell you you are wrong
09:58.32mog_homewhy would i bother him?
09:58.40mog_homeoh
09:58.43benjkbecause he's the authority
09:58.44trixtertheir faq doesnt always convey things correctly
09:58.46mog_homeits easy grandy
09:58.54trixterthey misquote the underlying laws and in some cases terems in the contract
09:59.20SwKyou know the funny thing about Moglen... he actually reads and replys to email
09:59.24grandymog_home: i'm sure it is, but it's 5am... and i don't really know dialplans very well...
09:59.27mog_homegrandy
09:59.33trixterthere is the difference bewteen what the FSF wants (ownership of all GPL code) and what their license states
09:59.34benjkyes he does
09:59.41benjktypically within less than 2 days
09:59.57mog_homewell tell you what ill write him
10:00.03SwKbenjk that blew my mind when I asked him a GPL question a while back
10:00.04mog_homeand tell you what i get back
10:00.07grandytrixter: that is a very true point
10:00.09mog_homeand if he says im right
10:00.13mog_homewill you disclaime it
10:00.15mog_homeand if im wrong
10:00.22mog_homeill do something for you
10:00.29benjkI will not disclaim it
10:00.39benjkwe will rewrite it to run outside of Asterisk
10:00.44mog_homeugh
10:00.48mog_homesome people.....
10:01.00mog_home*_*
10:01.02trixterwell if assigning to digium is the answer why doesnt digium assign asterisk to the FSF?
10:01.04trixter:)
10:01.16mog_homewell because digium has asterisk-be
10:01.20benjkbecause I don't agree with Digium's policy to get exceptions from themselves without granting exceptions in kind
10:01.22*** join/#asterisk jluk (n=jon@80-235-135-92.cable.ubr07.nail.blueyonder.co.uk)
10:01.30benjkfor themselves
10:01.45benjkwhere I come from every contract is a two way street
10:01.52grandymog_home: uhm... any hints on that dialplan thing?
10:01.52trixterahh so its ok for digium to gte others to contribute code free of charge just so they can sell it
10:02.00benjkyou grant me an exception and in return I grant you an exception
10:02.06trixterwas was that comment about how BSD is a good license if you want to be poor and have others make money off your work?
10:02.10benjkthat's how things should be
10:02.11mog_homedid you not get my privatemsg?
10:02.15trixterI do remember something about that a couple minutes ago
10:02.17grandyoh apparently not
10:02.43mog_homeokay im spamming channel then ^_^
10:02.44mog_homeexten => 100,1,Set(something=${ODBC_SQL('SELECT COUNT(*) FROM cdr')})
10:03.00grandymog_home: i forgive you... :) thanks1
10:03.01grandy!
10:03.12SwKtrixter some comment that everyone will get rich of your code and you wont if you BSD it
10:03.22benjkhowever, if Digium will sign over their ownership in Asterisk to the FSF, then I will convince Simon that we should sign over our module to the FSF, too
10:03.41mog_homeheh sure
10:03.47benjkwhy not?
10:03.49trixterswk: ahh I thought that was how the gpl worked, you submit code to digium so they can sell it in ABE and you get nothing
10:04.05benjkwhat's wrong with signing over your code to somebody who will look after it for you?
10:04.18mog_homeabe
10:04.18benjkthat's what you suggest to me
10:04.23trixterthey want people to assign to them not them to anyone else :P
10:04.38grandymog_home: fantastic!  thanks much! this will be much simpler than the agi stuff i was messin around with
10:04.56mog_homeasterisk would be where bayonne is today without mark
10:05.05mog_homeand mattf
10:05.07mog_homeand kevin
10:05.12mog_homeand countless others
10:05.21mog_homeif you dont see digiums contribution to asterisk
10:05.25mog_homeand its good stewardship
10:05.28mog_homeyou have head in your ass
10:05.35mog_homeno offense
10:06.01benjktrixter: sure, but the point is that if mog asks me to sign over the code to Digium and he thinks we are somehow abusing community spirit then I why would there be anything wrong with me suggesting to them that they sign over their stuff to the FSF?
10:06.08swm_MOG: Are you referring to the disk space you allow for the project to continue?
10:06.23swm_MOG: Or the fee's for G.729?
10:06.36trixterbenjk: because they cant make money off your code if 1. you dont submit it to them free of charge and 2. they assign it to someone else
10:06.59trixterits all about making money off other people - the claim about the BSD license appears to be with the gpl
10:07.38mog_homehow about putting roofs over several coders heads
10:07.50trixterlike benjk?
10:07.55mog_homeor supporting several developers
10:08.03swm_LOL I code and a put a roof over my head... Without anyone's help :)
10:08.05trixteror is this another 'its ok for us to get roofs but not the people that submit code' comment
10:08.20mog_homewell some of us swm are poor college student
10:08.20mog_homes
10:08.52mog_homeand digium has allowed me to get much more stuff done
10:08.59swm_LOL not saying I am rich but I make enough to get by every month. I never went to college and I make 2x more than most people with a college degree
10:09.07mog_homewhere do you see sangoma, aculab or dialogic supporting oss community?
10:09.07swm_furthermore I am a high school dropout
10:09.29mog_homeoh they dont
10:10.07trixterdoesnt sangnoma support a large gpl pbx application?
10:10.12benjkswm: then you have a fair chance to become wealthy and famous
10:10.14trixterdoesnt odints as well?
10:10.14benjk;_0
10:10.16benjk:-)
10:10.27SwKsome of those people have released their drivers as GPL havent they?
10:10.42mog_homesome of us have gpl hardware
10:10.43trixterwell I think sangnoma supports a large asterisk like pbx application that is gpl
10:10.46trixterbut I could be wrong
10:10.58mog_homesangoma sends people to digium with their asterisk problems
10:11.04SwKmog_home you dont make the TOR-II's anymore ;P
10:11.12mog_homeno but gpl design is out there
10:11.15trixterand supporting or not you say you want a roof over your head what about benjk?  you suggested he release public domain so digium could sell his code.. what about his roof?
10:11.19mog_home<PROTECTED>
10:11.39mog_homedigium does many things for the community at large as well
10:11.47mog_homeand digium is always hiring ^_^
10:12.28mog_homeand how much code has benjk contributed to asterisk tree?
10:12.31SwKand if you dont wanna work for digium but are good with asterisk and need a job let me know (sorry mog :P )
10:12.34swm_I may have a fair chance but, What does college do for someone? Just another fee you'll pay off in 30 years. Jobs in Oregon do not require a Degree from college. Most places dont. If you can do the work, You can do it. What does a little paper from a college say? All it says is you sat and watched someone present and lecture you on specific topic's... Question is can you apply the knowlege in the real world... I've seen that fail 75% of the time. so 25%
10:12.46mog_homelol swk
10:12.51mog_homecan i come work for asteria
10:12.58SwKmog_home no
10:13.02mog_homeaww
10:13.04mog_homewhy not?
10:13.14mog_homeill come fix your iax problems
10:13.15SwKmog_home the same reason I cant come work at Digium
10:13.32mog_homeheh im sure christian would let me come over
10:13.33benjkSwK: you're at Asteria?
10:13.41SwKyeah
10:14.00benjkcan you ping somebody for me?
10:14.06SwKwho
10:14.15mog_homeand trixter digium has helped creat business such as asteria asterisklink etc to put roofs over peoples heads
10:14.30swm_I can ping someone to death if you would like :)
10:15.15swm_But that's just childish :)
10:15.43*** join/#asterisk zoa (n=kkk@pirus.securax.be)
10:16.12mog_homeZOA!
10:16.30zoalooks to the right
10:16.33zoano zoa there
10:16.33zoa:p
10:16.34swm_11010011 10110100 00101011 01001110
10:16.51mog_homelol
10:17.43swm_ok someone calculate the binary
10:17.56swm_without using a program to translate it
10:18.44benjkits not like those companies who are doing Asterisk support/integration/consulting aren't doing honest work to earn an honest living
10:19.00swm_211 180 43 78
10:19.09trixterahh so as long as digium creates jobs for some its ok to get people to give their code without compenstation for others
10:19.10trixterI see
10:19.20mog_homeno one is losing rights
10:19.22mog_homeyour code is gpl
10:19.29mog_homethrough and through
10:19.38trixterto even suggest that benjk release public domain so digium can release gpl is a bit sketchy ...
10:19.45mog_homei mean be is not as scary as some people would want to believe
10:19.51trixteryou said that he should release it public domain so digium can release it
10:19.54mog_homehe doesnt need to trixter
10:19.55trixterthat is losing rights
10:19.57swm_basically no-one w/ GPL has rights then. Once it's GPL you have no copyright to yourself? It is public domain?
10:20.02mog_homehe can just give digium right to reliacne
10:20.08mog_homeno
10:20.25mog_homeyou just give digium rights to do with perpetually you can change it all day long
10:20.36trixterswm: no mog suggested that a specific application be released public domain to digium its not that if you release gpl its public domain becuase that isnt the case
10:20.53mog_homethere are 2 disclaimers
10:21.06swm_trixter: ok throw me some variables here to work with
10:21.13mog_homeone makes code public domain,  the other just grants digium a licsense
10:21.21trixterswm: such as what specifically?
10:22.23swm_All I am seeing here is Digium benefits and the public looses. Why not Digium grant funds for code to the public. If you code it, You get DigiCredits or something for it. :)
10:22.39trixterthat would seem equitable
10:22.58mog_homeheh swm_ not a bad idea
10:23.01*** join/#asterisk fenlander (n=fenlande@82.152.81.57)
10:23.02benjkswm: precisely my point
10:23.03mog_homei code for redbull myself
10:23.06swm_I think it would be fair. If I coded something I would at least expect a T-Shirt :)
10:23.29mog_homei do find it funny though
10:23.35mog_homethat the people who complain the most
10:23.39mog_homecontribute the least
10:23.45mog_homeand the people who dont car
10:23.45mog_homee
10:23.51mog_homecontribute the most
10:23.59trixterya know when I got swag from digium I actually expected a tshirt so that it would be advertising, instead I goit mouse pads, pens and screwdrivers, not that I am complaining nor has anyone that received those items so far, but I like tshirts and stuff :P  although the tdm410 was nice
10:24.03trixteras a prize for contests
10:24.25mog_homeyeah we give a way hw all time
10:24.34mog_hometrick is being around mark at the party
10:24.52trixterits all in how you go about getting compensated.  however that doesnt address the public domain comment nor the one  that the BSD license lets somenoe sell your code and make money and the coder doesnt get anything ...  which is what it seems when it comes to digium and the gpl
10:25.16trixtermog_home: acutally I didnt get any of this from mark directly, I contacted others at digium though, took a while but I got it :)
10:25.24mog_homeyeah
10:25.32trixtergot thevoipconnection.com to give 5 ATAs and 3 gxp2000s as prizes too
10:25.39mog_homenice
10:25.44trixtertoday no startch press contacted me (without solicitation) about how they can donate
10:25.52mog_homeahh
10:25.55benjkSunrise maintains a TIT-FOR-TAT licensing policy
10:25.57mog_homei want to write a book for them
10:25.58trixteroreilly is donating, hell oreilly is giving me $9000 in etel tickets for a really good price
10:26.02mog_homei love no starch press
10:26.06trixterso the group can have a little bit of funds
10:26.26swm_I do care for some aspects for asterisk and some aspects I dont. If it affects my position in VoIP technology I will raise a point if it does not apply to my functions within the uses of asterisk I still care but I try to keep my nose out of unfamilure territory. Tho I have used 90% of asterisk's applications and features ... I dont speak until I am sure I know what I am talking about :) ... They should really invoke some kind of DigiCredit system for c
10:26.38trixtergot a did from didx.org (have yet to actually test it) got bandwidth and hosting from trxtel (my company) and doeshosting.com (friends company) for  the sac AUG as well
10:26.41benjkhowever, res_zeroconf is a community project, not Sunrise, so this different yet
10:26.46benjker still
10:26.55SwKspeaking of OReilly anyone going to that OReilly VoIP conference next month?
10:27.09benjkyes
10:27.09trixterI am a speaker there
10:27.10mog_homedoesnt think i won the pool though
10:27.14trixterI have tickets for etel for $200 each
10:27.18trixterif anyone wants to get in cheap
10:27.18swm_SwK: SWM IS GOING ...
10:27.18mog_homereally
10:27.23trixterthanks to oreilly for that one :)
10:27.23SwKi wanna go
10:27.27mog_homewhere is it again trixter
10:27.34trixterburlingame (san francisco)
10:27.41trixterhttp://conferences.oreilly.com/etel
10:27.44SwKSurj sent me tix
10:27.52trixtermark will be at etel
10:27.54mog_homei know
10:27.57mog_homewe are going i think
10:27.59mog_homeim just not
10:28.10trixterahh I heard from someone at digium that most digium people wont be there
10:28.16swm_Anyone in Oregon, Washington or California can be picked up via my plane too for a $25 fee :)
10:28.20trixterdavid indicated that it was really only mark
10:28.22mog_homeyeah we only have 50 people
10:28.32mog_homeso we cant go sending everyone everywhere
10:28.34benjkhow much for picking me up in Tokyo?
10:28.54benjk:-)
10:29.01SwKbenjk: maybe you can hitchhike
10:29.06swm_I dont think I have a big enough tank of gas to make it over the pacific ocean unless I want to fly from alaska to russia and down :)
10:29.15trixtera friend is trying to swing hotel space at the mariott too, his mom has worked for mariott for like 25 years and he is gonna try to get her to give me and one other free rooms there in support of the sac AUG :D
10:29.27mog_homefreaking awesome
10:29.29trixterswm: what type of plane?
10:29.37benjkyeah, I go to the marina in Yokohama and hitch a ride on a nice yacht
10:29.45trixteranyway etel tickets are like $1800 if anyone wants to get in for $200 lemme know
10:29.49benjkbut I'd have to leave soon
10:29.58benjkto make it in time for the conference
10:30.10SwKaight I go bed now
10:30.20trixterswm_: what type of plane?
10:30.31swm_Cessna Encore
10:30.39trixterI have only flown a cessena 182 (although I have been in many that is the only one I was at hte controls of)
10:30.40SwKits 430AM and i'm supposed to be at work at like 9 or something
10:30.43trixterahh
10:30.54trixterI did the 182 when I was in civil air patrol, the air force bought it for us
10:31.02SwKswm_ whats the range on that thing?
10:31.05trixterso classes were $10/hr in the iar and free ground school
10:31.11swm_Range?
10:31.15trixterI think the encore maxes at 1500 or less
10:31.26benjkthat's not good enough to even get to Hawaii
10:31.37*** join/#asterisk coppice (n=chatzill@247.198.17.210.dyn.pacific.net.hk)
10:31.39benjkneed to fly over Alasaka then
10:31.40SwKyeah you know how far can you fly on 1 fuel load
10:32.21trixtermost small planes can go further than 1500 miles..  I think the gulfstream (jets not props) are the only ones that could go across the ocean and then the smaller ones you are running on fumes typically when you get there - not always a good place to be incase of headwinds/weather
10:32.35swm_792/km hour / 5,375lbs of fuel / IFR Range is about 3000 km
10:32.50*** join/#asterisk [Airwolf] (n=airwolf@attilla.nl)
10:32.51trixterahh so my 1500 guess was close :)
10:32.59swm_not bad
10:33.04SwKtrixter our CAP unit had a T38 for a while until some jerkoff got caught flying people that couldnt pass a flight physical and the AF yanked that thing with the quickness
10:33.09trixterer half
10:33.16swm_I re-fuel every state border for saftey purposes :)
10:33.18trixtera nautical mile is longer than a statute mile
10:33.44benjkdifficult to refuel over the Pacific though
10:33.47trixter6023 feet approx vs 5280 feet for a statute
10:33.52coppiceT38 - the flying FAX protocol :-)
10:33.57benjkunless you have connection at the USAF
10:34.12SwKcoppice yeah... with 2 turbo fan engines
10:34.26*** join/#asterisk gaupe (i=rmo@slogen.sunnmore.net)
10:34.27swm_LOL I can do voip from my plane ... lags a little bit but satellite does that :)
10:34.28trixterbenjk: well even then if you arent equipped for in air fueling or carrier landings you cant relaly get anythingfrom them
10:34.34trixtereven if you have a good relationship
10:34.40coppiceis it a twin? I thought it was too small for that
10:35.02SwKcoppice its a 2 seater F105 basically
10:35.09benjktrixter: but in the movies they always do that sort of thing ... somehow ... :-)
10:35.36swm_SwK: which plane are you talking about
10:35.40coppicei remember them mostly from chasing shuttles, and other NASAy activities
10:35.41SwKcoppice NASA uses them for chase planes cause the can do MACH1+
10:35.48SwKswm_ T38
10:35.58swm_Oh yeah those are interesting little things :)
10:36.27SwKswm_ yeah got a ride in one back in the late 80s when i was doing the Civil Air Patrol thing...
10:36.39SwKsee my comments about it and the air force about heh
10:37.08SwKs/about/above/
10:37.15swm_Sweet, I got my license flying in the national guard. Started out flying choppers (black hawks) and got my own plane when my relative died
10:37.27SwKkool
10:37.40SwKi never did get my license (wish i did)
10:38.28swm_Yeah nothing like flying a chopper and having RPG's flying at you while your dropping 6 guys into a drug operation :) ... South america sucks ass :)
10:38.33*** join/#asterisk _Madar (n=tophe@219-84-129-163-adsl-tpe.static.so-net.net.tw)
10:38.56SwKhave a few hours of time tho... couple in sesna 172 about an hour in the T38 and about 2 hours in a A4 Skyhawk
10:39.09swm_sesna? CESSNA
10:39.18SwKhave about 1000hrs in the A4 Simulator tho heh
10:39.21SwKyeah cessna
10:39.49SwKi cant speel and I'm pissed cause I've only had about an hour of sleep in the past 24 hours and insomnia is kicking my ass
10:39.59*** join/#asterisk Aze` (n=aze@85-18-136-114.ip.fastwebnet.it)
10:40.07swm_I dont like simulators dont feel anything real. I like the vibrations, crosswinds & pressure changes... More real to me... :)
10:40.15SwKyeah
10:40.45swm_Oh flying a copper you get updraft and crossdrafts when your flying around mountians ... thats really fun... If you had a big lunch it wont be so big when you get out :)
10:40.53SwKwhen I was in the military tho I was in the ATC unit and the Simulator Officer was a friend of mine so I got time there whenever the pilots weren't around
10:41.40swm_SwK: That's hard to do now of days ...
10:42.02swm_9/11 Created alot of paperwork to become a pilot
10:42.14SwKswm_ I have a little time in a kiawa (OH-58)
10:42.17wasimespecially if you're brown and bearded
10:42.18SwKyeah i know
10:42.24SwKhahaha
10:42.25swm_lol
10:42.39SwKDONT LET THE BROWN MAN FLY!!!
10:42.39swm_wasim: Exactly putting the pushpin on the dot
10:42.50coppicei used to work with lots of ex fighter pilots. those guys have amazing reaction times :-)
10:43.02SwKyeah most of them are nuts
10:43.18SwKrotary wing pilots arent exactly sane tho
10:43.46swm_I agree with that
10:44.03SwKdamned Cobra pilots we had around when I was in the Marines were just completely insane
10:44.48SwKmy little brother was with a Kiawa scout unit... those guys have some balls... going after tanks and stuff with just a skid mounted .50
10:44.59SwKand a laser pointer heh
10:45.06*** join/#asterisk felipex (n=dsfdsf@85-18-250-142.ip.fastwebnet.it)
10:45.17swm_lol we had a guy in the guard that performed a reverse -5 1/6 dive in rollover and pulled out of it and I about shit watching it ... 240 feet elevation ... Just thought he was a dead man... but thats why people die... trying stupid shit
10:45.38SwKswm_ you know what CAX is ?
10:45.54swm_I know about 4 different acronym's for CAX
10:46.04SwKCombines Arms eXercise
10:46.14swm_That would be 1 of them heh
10:47.04swm_96-Hour test :)
10:47.29SwKthe Marines have CAX out on the west coast (usually 29stumps) about once a year... in 92 some batshit Cobra pilot was comming in yanking and banking and walked his main rotor down the sand a couple hundred meters
10:48.46swm_SwK: LOL ... Oh .. I would love to see that ... That would be like "Americas Stupidest pilots" ... Do you know what AIT iz? :)
10:49.01SwKAdvInfantry Training?
10:49.17swm_Advanced Individual Training
10:49.21SwKheh
10:49.33SwKnot familiar w/ that one
10:49.40SwKmust be a army thing
10:50.29swm_Training I undertook in the guard. Very limited people get exposed to this training. You have to be fast, quick, always on your toes looking for the most smallest threat. but they teach you things that other pilots never will see done.
10:50.48SwKaltho i must say CAX is the most awsome live fire exersize to witness live
10:51.20SwKkool
10:51.25coppiceswm_: sounds like living in some of the neighbourhoods i try to avoid :-)
10:51.39swm_Yeah I've taken out runway's in live firefights with drug lords in south america... Stuff that can keep a man up for months thinking about it
10:52.14*** join/#asterisk mutilator (n=animenod@65.111.201.79)
10:52.28swm_50mm shells, rpg's, surface to air weapons. 3 choppers moving in and they just unload on you...
10:52.30SwKswm_: alto I must say... when it comes to military officers the Navy and Marines got it right.... only part of the US military that puts officers in large piles of metal and throws them at the enemy
10:53.03SwK(re: cat lauch)
10:53.06SwKlaunch
10:53.14SwKdamn i cant tipe or speel tonight
10:53.45swm_I do prefer the Hades bomb for taking out large areas... the explosion is just way intense :)
10:54.00trixterI saw iron eagle
10:54.37swm_Iron Eagle? Isin't that some movie where a guy is flying some kind of jet powered military plane?
10:54.48SwKI prefer the  BLU-82
10:55.23SwKif you want to really screw up some real estate that is
10:55.26swm_Oh yeah the Daisy Cutter :)
10:55.49SwKbut the most fun is the MK19
10:55.52*** join/#asterisk heka (n=heka@80.80.174.140)
10:56.15SwKif you've never had the pleasure of letting a MK19 open up you havent had real fun
10:56.22swm_You normally drop that from a C-130 Transport ... That'll clear a 300 foot landing path for choppers :)
10:57.03SwKwell a BLU-82 is like a 15K lb bomb
10:57.18swm_For sure
10:57.32SwKits not like you can mount that one a wing pylon
10:58.38*** join/#asterisk lo_tech (n=lo_tech@209.36.181.24)
10:58.40swm_Did you know it was designed for clearing out the jungle for choppers?
10:58.45SwKyeah
10:59.06SwKEAF used them in 'Nam for forward refuel/rearm points
10:59.15swm_It's also designed to detonate 1 to 6 feet above ground to minimize the crater effect a more smooth surface
10:59.49SwKif it goes bang i've probably tried to get my grubby little hands on it ;)
10:59.56swm_Personally just drop a Hydrogen bomb and let that clear out the forest :) ..
11:00.40swm_How the hell did I get to talking about weapons and destruction in a PBX conference :) everyone must of fallen asleep :)
11:00.42mutilatoryea, then just go diving into radiation!
11:00.59swm_Radiation? Hydrogen is not a radioactive substance ...
11:01.13swm_Nuclear would be the word your searching to slam on
11:01.19SwKwhen it comes to small arms there not too much in the current military arsenal i havent fired.... (save stuff that was deployed after the mid 90s)
11:01.58swm_LOL they dont upgrade quickly ... they like to stay with proven technology.
11:02.09SwKyeah
11:02.24SwKi know... but they have a few new things on the horizon
11:02.27swm_LOL i used to live 20 miles from a Chemical Depot where they destroy gas weapons :)
11:02.42SwKi live right off Redstone ;)
11:02.56SwKand I was a jarhead heh
11:03.26*** join/#asterisk znoG (n=gs@33-138-114-200.fibertel.com.ar)
11:03.40swm_Marine ?
11:03.50swm_USMC
11:03.50SwKwhen I was in the Corps, i was FullTime staff for a reserve unit... the Armourer (who was also full time) got to be a good friend of mine...
11:03.51SwKyeah
11:04.12swm_Ok I am getting war hungry... Time to go make some pipe bombs :) heh
11:04.16SwKwe didnt have much to do when the reservist were not around so we went to the rang heh
11:04.21znoGguys, is it "bad" to use an AGI script to dial any number in the system? its too hard to do what I want to do with the standard dialplan, doing it in Perl is much easier/quicker/neater
11:04.43trixterI am slightly bored now..  I think I need something to do
11:05.08swm_Come over here trixter .. I'll show you how to make a hydrogen bomb out of Tin foil, water and liquid lye
11:05.12mutilatoruh
11:05.22mutilatorhydrogen bombs still produce radiation
11:05.25mutilatorthey are nuclear weapons
11:05.37SwKznog: if you are doing something int he AGI to manipulate the number or destination route go for it,,, just DO NOT CALL DIAL IN THE AGI... set a variable, exit the AGI then in the next priority in your dialplan dial the variable
11:05.57swm_mutilator ... Your computer screen creates more radiation than a hydrogen bomb lol
11:06.02mutilatorum no
11:06.06mutilatorit doesn't
11:06.10SwKmutilator I think he's refering to the conventional hydrogen
11:06.20swm_oh wow ...
11:06.23swm_yeah that might
11:06.34SwKhydrogren + O2 == big fireball lots o' heat
11:06.56coppiceit can also put a man on the moon
11:06.56swm_the whole purpose in the hydrogen bomb now of days (a little catchup course) is to minimize the radiation and have a huge explosion.
11:07.15coppicewhich was a waste of time, cos he didn't bring back any green cheese
11:07.26trixterswm_: neighbor when I was in high school used HCl (muratic acid specifically) and foil in a plastic soda bottle
11:07.56SwKcoppice he didnt bring back green cheese cause they didn't really go... they were on a sound stage at one of the studios in hollywood
11:07.57swm_trixter: How did you keep the bottle from melting from the chemical exchange?
11:08.08*** join/#asterisk steff (n=steff@80.125.254.220)
11:08.11steffhi all
11:08.16trixterany time you mix an acid and metal a salt is formed and hydrogen is released, if you get enough hydrogen (and oxygen) with enough heat it will explode
11:08.16znoGSwK: yea, actually, that was part of the plan, calling dial in the end... how come its best to set the variable and call dial from asterisk?
11:08.27trixterthat is the principle, not a nuclear weapon by any means
11:08.36SwKznog dial blocks and you AGI will stay running...
11:08.39trixter2 litre soda bottles worked fine
11:08.49trixterespecially when they were shoved in fiberglass mailboxes
11:08.54znoGSwK: for the duration of the call? or forever?
11:08.56swm_Trixter: Ok I was picturing a small little 20oz bottle
11:08.58SwKznog that means its eating system resources while you are running your mouth...
11:09.25trixteryou dont fill it full cause you need oxygen and when you increase pressure you create heat as well, which makes it handy for a detonator
11:09.55trixtercourse you could always just get calcium carbonate and water, when mixed they release a gas like acyletaine
11:10.06trixterer acyteline
11:10.11SwKfor the duration of the call... if its a busy system thats dozens of perl interpreters or PHP interpreters or whatever running when you can just set a variable exit the AGI Dial(${SOME_VARIABLE}) then on exten => h do a deadagi for clean up
11:10.28swm_Hydrochloric acid and Magnesium :)
11:10.36SwKif you even need to do the clean up
11:10.51flotWho is used SIPNET.ru ?
11:11.22swm_anyone from russia in a american chatroom? Ok I thought not... Will someone call border patrol...
11:11.23trixternow if you want a really big boom, home made fireworks out of N4 are trivial, amonia and chemical fertilizer (need indsutrial amonia the stuff in stores is 2% normally) which when mixed can make amonium nitrate, then add diesel fuel to make it N4..  as seein in the 1993 world trade center attack as well as oklahoma city
11:11.27znoGSwK: is the deadagi needed?
11:11.47SwKtrixter its called AmFo
11:11.54trixterswm_: what column is Mg in on the periodic table?  cause the one that Al is in works best..
11:12.10trixterSwK: its also called N4
11:12.12SwKznog: only if you need the AGI to process (or keep processing) after the call is hungup
11:12.50znoGSwK: ah okay.. can't see a need to do that just yet, but i'll keep it in mind. I'll probably do what you said. Thanks!
11:12.53swm_trixter: Not sure would have to look it up :) was not that big on chemistry but I did alot of research on making an explosion :)
11:12.54SwKtrixter and its easy to make... just get fertilizer at the farmers co-op... its got plent of AmmoniumNitrate in it ;)
11:13.09trixterthermite is also a fun toy..  50/50 ratio of aluminum powder and ferric oxide (iron rust) mixed evenly together..  light with something hot like magnesium and you have a few thousand degree incendary (WWII was about when that was discovered)
11:13.16*** join/#asterisk pa (n=Paolo@unaffiliated/pa)
11:13.25SwKRDX is easy to make too if you can get the old formulat fuel tables
11:13.28znoGSwK: the AGI is mainly to determine whether a pin should be asked for (ie. local extensions don't have a pin asked for, but local/mobile calls do)
11:13.28SwKerr tabs
11:13.43trixterSwK: some use that however some dont..  if you get chemical fertilizer at the local garden shop you want at least 70% nitrogen
11:13.47znoGSwK: it was just much easier doing it with AGI than using the * dialplan
11:14.08swm_OK enough military/weapons/bombs/explosions/chemical ratio's/etc ... I'm getting tired :)
11:14.22trixterthe amonia is harder to get but not impossible, its used as a refrigerant in many systems, especially portable ones like on boats and motorhomes
11:14.39SwKznog: i wrote 1000s of calls a day on 1 box with an AGI
11:14.41swm_52 hours of beeing awake and trying to think all this stuff up is just too much to think about :)
11:14.56znoGSwK: nice, always exiting the AGI to dial, right?
11:14.58SwKit works great for complex stuff
11:15.01SwKyeah
11:15.09znoGSwK: what sort of stuff do you do to "clean up" the AGI?
11:15.16swm_LOL wonder how Digium would stand if I flew a plane right into the building :)
11:15.18SwKrating a call
11:15.24znoGSwK: ahh
11:15.28SwKsetting states etx
11:15.35znoGSwK: when you call DEADAGI, it tells you how long the call went for etc?
11:15.47SwKtheres variables for that
11:16.00trixterI have to see how well this box will handle a load..  customer wants to test out 600k minutes a month (30 channels 11 hours per day) and grow to 1000 concurrent channels for 13 hours per day.  hopefully all goes well and they are happy cause that means I am happy especially since I will be making money off the voip service too :D
11:16.14swm_znoG: Your confusing yourself. Digging for and answer that would be normally right in front of your face
11:16.17SwKcheck out the variables that are passed by default and then there are some you have to query for
11:16.30trixterthat type of load testing though is gonna be a bit rough..  at least the way I wrote the custom code for it its easy to distribute everything
11:16.42SwKcheck out the wiki theres a good bit of AGI info there
11:16.49znoGSwK: yeah i printed all the keys and values for the standard ReadParse.. i just thought calling DEADAGI might include some other vars as the call would have been completed
11:17.13trixterswm_: how do you play zork on the phone then?
11:17.15znoGswm_: what alternative is there for complex stuff though?
11:17.16trixterzasterisk!
11:17.30mutilatoror et your local weather
11:17.32mutilatorget*(
11:17.34mutilatorget*
11:17.38SwKznog the alternative is to write app_whatever_you_need
11:17.53trixteryou can technically get the weather and do zasterisk via dbget or whatever
11:18.04trixterbut that is a pain to use channel variables in the dialplan and do it that way
11:18.05znoGSwK: heh, yeah, don't think i'm gonna go that far
11:18.10trixterthen you still need backend code to populate the DB
11:18.12SwKi do AGI/FastAGI cause its faster/easier in most cases
11:18.21swm_*** I DONT PLAY ZORK ON A PHONE ***
11:18.32trixterswm_: loser
11:18.38swm_I have better things to do with my time then tinker with games :)
11:18.42SwKok i really go sleep now
11:18.52znoGthanks for the info SwK :)
11:19.05znoGyou have saved me from lots of running processes running wild and free
11:19.06mutilatorif you had a database to dbget from?
11:19.29mutilatorcronjob a script to add it every hour or what?
11:19.34trixterI actually havent played it, but that is one example of taking a data file and without really converting it to something else being able to parse data out and use it in a telephony environment..  which can be a legit problem -- how would you be able to do that without an agi?
11:19.36swm_LOL I grew up and figured out by playing games it does not get you anywhere in life... Reality is the best way to live ... RPG's and Fantasy games or whatever are just stupid. Tic-Tac-Toe is the only game with logic :)
11:19.44*** join/#asterisk amir_ (n=amir@gentoo/developer/amir)
11:20.13mutilatorGRRR!
11:20.21mutilatorthey took down 2 of my modem banks in saginaw last night
11:20.32mutilatormrtg spit me out 70 billion errors in my email
11:20.41swm_mutilator: Did they mutilate them?
11:20.52trixterI played computer games in the 70s by the 80s I was only interested in cracking them ...  I generally dont play games now however zasterisk does show that you can parse arbitrary data from custom formats and interact with a caller based on that information
11:20.57mutilatorchanged ips or unplugged i dunno
11:21.09swm_mutilator: Maybe they had a hydrogen bomb go off near them and it created some electromagnetic pulses and knocked it out :)
11:21.10trixterwhich aside from the fact that its a game it does show a certain problem off in a specific way
11:21.11mutilatorthey're nice tc1000's tho i'de hope they didn;t mutilate em
11:22.18swm_Oh wait doesn't a bomb have to be nuclear to create EMP? Hmm.... Unless it's a highly overloaded energy field... **
11:23.20trixterno it doesnt
11:23.24trixterlook at flux compression generators
11:23.37trixtera charged coil (usually by capacitors) is compressed when a high explosive is detonated
11:23.38swm_trixter: Are you going to stick that in a bomb?
11:23.41trixterthat results in a large EMP
11:24.05mutilatorand they hooked up another pm4
11:24.06mutilatorGRR
11:24.34swm_LOL I think i'm gonna go down to my sub station and de-activate all the voltage regulation equipment and overload all the transformers in town :) ... Sounds like fun :)
11:27.41mutilatori wondered why that server was runnin so slow
11:27.49mutilatorlogrotate decided to kick in
11:28.03mutilator<PROTECTED>
11:28.26mutilatorand you're goin to bed
11:29.07lo_techthat's a silly amount of load just for logrotate
11:30.09mutilatorwell
11:30.13mutilatorit's our mail filter server too
11:30.24mutilatorload normally runs ~4
11:30.29mutilatoracross the board
11:31.06lo_techhave db servers with 5k+ connections that dont run that hard
11:31.41mutilatorit's comin back down now
11:31.47mutilator06:31:34 up 22 days, 21:42,  3 users,  load average: 9.49, 10.91, 8.35
11:32.00mutilatorjust does it when logrotate hits
11:32.10mutilatorcause it slows the server down moving files... it's only ide
11:32.38lo_techyou mean eide, right :P
11:32.58mutilatormeh
11:33.20mutilatorconcurrency on the mail runs ~60 msgs
11:33.31mutilatorand i think i have 9 spamd child threads for scanning it
11:33.49mutilatorand virus scanned at smtp
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11:37.28mutilatorhmm
11:37.31mutilatork maybe it's not logrotate
11:37.32mutilatorwtf
11:38.02mutilatorspamd runnin like mad
11:38.08Assidreduce the concurrency
11:38.10mutilatormust be what it is
11:38.16mutilatorAssid: then messages get lost..
11:38.22Assidnot really
11:38.26Assidthey go into queue
11:38.31Assidtakes 3 seconds more
11:38.40Assidon a 1.3 celeron
11:38.41mutilatorer.. concurrency = how many connections i take in at once
11:38.52Assidhrmm
11:38.59Assidi thinki have that at 100
11:39.11Assidi avg 1 email per 2 seconds
11:39.11mutilatormines set at 120 but i only run ~60
11:39.31Assidruns spamd and clamd
11:39.33mutilatori get a few more than that..
11:39.50Assidwell.. you have more than just a 1.3 celeron right?
11:40.21Assidi run a webserver.. database.. imap.. pop3.. email.. and few php cli apps.. and some other stuff too
11:40.21mutilatorya, dual 2.4 xeons with HT
11:40.27mutilatoractually ht isn't runnin on that
11:40.47*** join/#asterisk saftsack (n=saftsack@IP-213188106101.dialin.heagmedianet.de)
11:40.50saftsackhi
11:40.54mutilatorbox runs radius and asterisk too
11:40.54Assidwith database logging for every email in and out..
11:40.55*** join/#asterisk JohnnyC (n=JoaoCorr@195-23-115-68.net.novis.pt)
11:41.29mutilatorright now in 2 seconds i got 34 messages
11:41.48saftsackasterisk voicemail is the same like an answerphone on a normal telephone, right?
11:41.49Assidhrmm
11:42.16mutilatormail.freesprung.com:38.112.164.2::40590
11:42.18mutilatorAND
11:42.26mutilatormail.esmartloan.com:63.150.226.151::1685
11:42.29mutilatorare spamminating me
11:42.38Assidblock em
11:42.50mutilatoryeh
11:43.12Assidyou using qmail?
11:43.13Assidset them either in your iptables.. or tcp.smtp with a message to go away
11:43.36Assidyou may wanna subscribe to rbl too
11:43.43mutilatorya
11:43.51mutilatori got smapcop and spamhaus smtp level blocks
11:43.55mutilatorspamcop
11:44.08Assidsame here
11:44.16mutilatorthen the rest of the rbl stuff is spamd point based
11:45.30Assidi wish people moved onto spf
11:45.36Assidwould make life easier
11:46.42mutilatori thought about using it but i figured it's too new right now, and it'de fsck up mail and people would complain
11:46.47mutilatorand i just didn't wanna deal with that
11:47.14Assidi have it on softfail
11:47.26mutilatorwhat i have in place now has an awesome success rate so far
11:47.37Assidso.. i give points to those using it.
11:48.14Assidrather the other way around
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12:09.17*** join/#asterisk Gunnar (n=gunnar@bkkb-gw.bitcon.no)
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12:18.13zoahey areski
12:21.16areskiHello Zoa
12:21.29areskizoa, how u doing '
12:21.34zoaim fine
12:21.36zoayou?
12:21.40zoagoing to belgium for xmas ?
12:23.08areskizoa, not this time, My gf win and I am going with her in Italy
12:23.39areskiwell she is Italian! I will be back end of January probably
12:23.40coppicenot going to belgium sounds like a win :-)
12:23.50zoahehe
12:24.24areskiwell I have to spent xas with her familly
12:24.30areskithat s why she win
12:24.49areskione year each, i guess it s fair
12:25.00areskizoa, u r coming back
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12:27.46coppiceNick Serv has amnesia?
12:27.48zoanah
12:27.55zoabut im not going with my gf either
12:28.02zoaso now im in a big fight :)
12:29.07zoabut if i dont even like spending xmas with my own family, im certainly not going to spend half a week with someone else's family
12:29.14zoaespecially if they speak a different language
12:29.52*** join/#asterisk koperniqs (n=koperniq@129.187.15.40)
12:30.38areskisure you should have it harder ! Italian is kind of easy to learn like Spanish
12:31.13zoaespecially if you are french speaking :)
12:31.17trixtermoral question ...  what would you do if you saw a company that sold a commercial non-free license of software that used directly code that is owned by the FSF, netbsd and the regents of california (BSD stuff)
12:31.50zoayou can sell software with a bsd license, no ?
12:32.00trixterthere is a clause that requires :
12:32.05trixter<PROTECTED>
12:32.05trixter<PROTECTED>
12:32.05trixter<PROTECTED>
12:32.05trixter<PROTECTED>
12:32.12trixtersame for the regents just different attirbution
12:32.21trixterand the company doesnt include that in  their advertising materials
12:32.43trixterI am just curious what everyone would do if they discoverd that
12:32.58zoadepending on the case i would either not care
12:33.11zoaif i'd care a little i'd send them an email
12:33.21zoaif i care a lot i'd send a mail to the people from the project
12:33.43trixtercause its not just netbsd stuff, its also the regents of CA and the FSF that have code in there
12:33.44zoaif i'd care a whole lot i'd sent em to my gf's family for xmas
12:34.17zoanot that i dont like the people btw, i just dont like xmas
12:34.55trixterahh
12:35.24trixterI was thinking about submitting them to gplviolators.org and other places becuase this company makes it appear ot many that they are solely responsible for the software with their copyright and 'written by' notices
12:35.32drrayI thought the only condition of the BSD license was that you acknowldege the code
12:35.32trixterand they dont give credit where credit is due, and in some cases required
12:35.37areskisame for me, I guess... I dont have so much the christmas spirit :)
12:35.42trixterdrray: there are different BSD licenses
12:35.56areskiI still forget this year about the xmas tree
12:36.05trixterthe ones in question specifically require mention in advertising materials
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12:48.28*** join/#asterisk loop1979 (n=loop1979@187.212.76.83.cust.bluewin.ch)
12:48.32loop1979hi everybody
12:49.29loop1979can sombody help me? i installed asterisk@home 2.1 and now i can't use DTMF anymore...(in the 1.3 version it worked fine...)
12:49.37loop1979i use cisco's 7960
12:51.25*** join/#asterisk fulgas (n=fulgas@213.58.130.46)
12:51.41loop1979can sombody help me? i installed asterisk@home 2.1 and now i can't use DTMF anymore...(in the 1.3 version it worked fine...)
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12:52.33*** part/#asterisk oej_ (n=oej@apollo.webway.se)
12:52.36loop1979nobody here?
12:52.57loop1979oder kann man hier deutsch schreiben?
12:53.07*** join/#asterisk _Madar (n=tophe@219-84-129-163-adsl-tpe.static.so-net.net.tw)
12:54.23tzafrir_laptoploop1979, #asterisk-de ?
12:54.31loop1979english is also ok
12:54.54loop1979i need help with DTMF..
12:55.10loop1979i don't see what's wrong and why i can't use it anymore with the cisco's and other phones..
12:55.25tzafrir_laptoptry to give more details about phones and such. I figure that this is a SIP phone and its dtmf type has changed
12:55.38loop1979i use cisco 7960
12:55.42loop1979and asterisk@home 2.1
12:55.52loop1979i didn't change the phones ..
12:56.04loop1979but i updated from a@h 1.3 to a@h 2.1
12:56.10loop1979so the phones are still the same
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12:56.28tzafrir_laptopthe upgrade procedures of AMP are, well, strange
12:57.14loop1979in the extension setting i always have rfc2833 dtmfmode
12:57.20loop1979i think that's standart
12:57.51loop1979and in the cisco config i use inband dtmf = enabled  and outbadn dtmf = avt
12:57.56loop1979and payload = 101
12:58.04loop1979i think thats also standard..
12:59.00loop1979i can dial with every phone, no problem, but i can't use dtmf for example voicemail password...
13:00.31loop1979any ideas?
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13:03.17mog_homemwahahah
13:03.23mog_homenew source in asterisk-xmpp
13:03.23loop1979?
13:03.28mog_homefor those intrested
13:03.53loop1979hm can somebody help me with my dtmf problem?
13:04.07mog_homewhats wrong loop1979
13:04.13loop1979i don't know
13:04.16mog_homelol
13:04.17loop1979i updated from
13:04.17mog_homegood answer
13:04.23loop19791.3 to 2.1 a@h
13:04.33loop1979and everything work as it should...
13:04.39loop1979but the dtmf recon..
13:04.51loop1979whenn i use voicemail or something didn't recognice it
13:05.04loop1979so i can't give with dtmf my passowrd..
13:05.18loop1979asterisk ignore what i do with the dtmf..
13:05.39loop1979so it seems it didn't recognize dtmf ..
13:06.01loop1979on the extension i have rfc2833
13:06.15loop1979so it should work as normal...but it doesnt
13:06.20loop1979do you have any ideas
13:06.20loop1979=
13:06.22loop1979?
13:06.24tzafrir_laptopany chance I could use asterisk-xmpp with a jabber server that runs using a free JVM?
13:06.39tzafrir_laptop(free as in Debian/main)
13:06.57mog_homeasterisk@home
13:07.04mog_homejvm?
13:07.04loop1979yes
13:07.19mog_homeit will connect to jabberd1, and jabberd2 servers
13:07.20tzafrir_laptop~jvm
13:07.22jbot[jvm] Java Virtual Machine. deb ftp://ftp.tux.org/pub/java/debian woody non-free
13:07.28mog_homei know what jvm is
13:07.32mog_homeis it jive i take it?
13:07.42mog_homei think that only runs on java 1.5
13:07.51mog_homebut anyways yeah it will probably work there
13:07.58mog_home<PROTECTED>
13:08.03mog_homewhere as before it sucked
13:08.15loop1979mog_home: any ideas about my dtmf problem?
13:08.36mog_homei imagine you arent really in rfc2833 and are doing inband in a codec other than ulaw
13:08.42*** join/#asterisk pa (n=Paolo@unaffiliated/pa)
13:08.56loop1979i think the ciscos connect throug ulaw
13:09.02loop1979as far as i know ...
13:09.17*** join/#asterisk oej (n=olle@apollo.webway.se)
13:09.24mog_homeoej!
13:09.30mog_homei have updated asterisk-xmpp
13:09.32mog_homejust for you
13:09.37mog_homeit makes sense now!
13:09.42trixtermog_home: does ABE have h.323?
13:09.51mog_homei dont have abe
13:09.52oejmog_home: Have you been up all night?
13:09.55mog_homeyeah
13:09.56trixterahh
13:09.57mog_homei was working
13:10.00trixterwonder which one..
13:10.08mog_homewanted to get it published today
13:10.11mog_homebefore i go home
13:10.19oejmog_home: It has happened to me as well. I will take a look at it. I guess jingle bells go  a new meaning for you too...
13:10.21mog_homebecause im not sure if i will have net there
13:10.28mog_homelol
13:10.30mog_homejust wait
13:10.37mog_homei do have a little jingle in my step
13:10.41mog_homeits just not visible yet
13:10.41oejA channel is really needed
13:10.46mog_homessh
13:10.50mog_homeyou take away all the fun
13:10.56oejsorry
13:13.02trixtermog_home: do you have a url or osmething that lists thge features of ABE?  I thought it was basically the same cept for the license and stuff
13:13.24mog_homedigium site i assume ^_^
13:13.29mog_homei can look it up for ya
13:13.34mog_homeif its not included
13:13.39mog_homeyou can do oh323
13:13.42mog_homeas we include headers
13:14.11trixteryou include the headers in ABE?
13:14.20oejToday ABE is very much the same, trixter. In the future, a lot of licensed stuff will be added by third parties.
13:14.22mog_homeyes
13:14.27trixterooh323 is gpl how can you do that without violating the gpl?
13:14.33oejThe Asterisk core is the same as 1.2
13:14.57mog_homeyou dont distribute it trixter
13:14.57oejH.323 is not included in ABE
13:14.57mog_homei cant give you ooh323
13:14.57trixterbut you said that you did distribute it
13:14.57*** join/#asterisk sandos (n=sandos@tor/session/x-45decaf6d6f46475)
13:14.57trixtermog_home you can do oh323
13:14.58trixtermog_home as we include headers
13:15.01oejNeither is MGCP I believe
13:15.03mog_homewe include headers for asterisk
13:15.07sandosI have a problem with asterisk.. it wont start, and the logs dont tell me much..  a few warnings though
13:15.07mog_homeso you can build things against it
13:15.13mog_homeasterisk -vvvvc
13:15.14trixterahh I see what you are saying
13:15.16mog_homesandros
13:15.19mog_homeso you can build it
13:15.25mog_homebut you cant legally distribute it
13:15.29mog_homeis my understanding
13:15.31sandosaha
13:15.44trixteris the dialplan parser the same in ABE as GPL?
13:15.45sandosthis is always the last line: Dec 22 14:15:35 WARNING[22512]: loader.c:440 load_modules: Loading module chan_zap.so failed!
13:15.50sandosis this supposed to be fatal?
13:15.50mog_homeyes
13:15.54mog_homeeverything is the same
13:15.56trixterthat is GPL how can you release that?
13:16.00mog_homeyou probably have a bug in zapata.conf
13:16.04trixteraccording to the source code its based off bison
13:16.05sandosaha ok. will check.
13:16.07trixterowned by the FSF
13:16.18mog_homeits slightly different
13:16.19trixteror at least includes code from bison
13:16.23oejtrixter, but we do not link bison
13:16.27mog_homebecause you run bison to create the documents
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13:16.29trixterahh so its just a drived work from FSF owned GPLed software
13:16.32trixterso that makes it ok I see
13:16.34mog_homeits a direvitive work
13:16.37sandoshow would I disable everything related to zapata? I dont have any of that stuff.
13:16.43trixternah code from bison is in the .c file
13:16.58oejsandos: add "noload chan_zap.so" in modules.conf
13:17.01trixterso linking isnt the issue its copying gpled code into a commercial product not licensed under the gpl
13:17.06sandosthanks!
13:17.25oejtrixter: Yes, bison is a tool we use, like gcc, to build asterisk. But we do not link to either
13:17.30mog_homeits like it was built with gcc
13:17.30mog_homethings built with gcc are not ness. gpl
13:17.30mog_homewe dont link against any gpl binaries trixter
13:17.39trixteroej: lkook inside the .c file
13:17.46oejHmmm. Will do.
13:17.47trixterthe dialplan parser uses code from bison directly
13:17.59trixterlet me get the exact file
13:18.14trixterbecuase you guys now dont want to accept that I actually know how to compile or read comments in code
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13:18.19saftsackhi
13:18.26sandosah, this works. nice. I wonder why this happened though, I havent changed any config lately. maybe a debian update
13:18.40oejtrixter: I read the file and found this: As a special exception, when this file is copied by Bison into a
13:18.49oej<PROTECTED>
13:18.49trixterast_expr2.c
13:18.49oej<PROTECTED>
13:18.49trixterspecifically
13:18.49oej<PROTECTED>
13:19.03mog_homeother than libpri
13:19.09trixterfirst line specifically
13:19.10mog_homewe use bison to generate some files
13:19.16trixter<PROTECTED>
13:19.18oejtrixter: So the FSF gave us an exception
13:19.36trixterthey did?  suprising that they gave so much code
13:19.38trixtercause its more than that
13:19.53oejtrixter: Read a few lines down, the text I quoted. You are right, but there is an exception since people use it for development of commercial products.
13:19.59trixterI wouldnt have guessed that the FSF would give exceptions to people to use the FSF owned code in commercial products
13:20.07*** part/#asterisk sandos (n=sandos@tor/session/x-45decaf6d6f46475)
13:20.07mog_homewoah got a back log
13:20.19mog_homeyeah that too
13:20.39oejNo, that's very uncommon. But on the other hand, a lot of libraries have LGPL to enable build of commercial products with the GNU toolset, so it makes sense
13:20.59mog_homelike libiksemel the library i use for jabber stuff
13:21.06mog_homeor libcurl
13:21.14mog_homeetc
13:21.24oejFor every tool or library we add to asterisk, we do a license review
13:21.48mog_homeand do we need to add a dependency review....
13:21.55oejMark and Kevin are pretty strict about that
13:22.00zoathey have to be
13:22.03zoa:)
13:22.11zoaim also very strict about that
13:22.12oejYes
13:22.19zoafor our code i mean
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13:22.42oejMe too. That's why I got a bit upset this morning in the discussion with Benjk
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13:22.51mog_homeyou see that oej?
13:22.54mog_homehow do you fall
13:23.06mog_homelgpl module allowed or not?
13:23.06benjkoej, there is plenty of lgpl code in asterisk
13:23.17mog_homeben you dont get it
13:23.19benjkyou couldn't use CDRs for example
13:23.21trixteroej: yeah asterisk takes some lgpl software and passes it off as gpl
13:23.27benjkthe CDR stuff is LGPL
13:23.34mog_homeyou can use lgpl you cant make asterisk lgpl
13:23.45trixtercan asterisk make lgpl stuff gpl?
13:23.51trixteras it claims it does in its license
13:23.53mog_homeno
13:23.55trixtercan it include it in ABE?
13:23.55mog_homeit doesnt
13:24.03mog_homelgpl says you can include in commerical work
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13:24.13mog_homeas long as you dont change said lgpl library
13:24.15zoayou can include it in abe
13:24.24zoaas long as it dynamically linked
13:24.33trixterthe cdr stuff in asterisk is dynamically linked?
13:24.34zoayou are also allowed to change it, but need to give sources anyhow
13:24.35oejbenjk: You can not create a module based on Asterisk API and use any other license than GPL. However, Asterisk can link to other modules that are compatible to either GPL or the Digium commercial license, depending on what we want to do (asterisk or asterisk-addons)
13:24.38trixterhmm I will have to look at that
13:24.59mog_homeoej once again for the win
13:25.27mog_homeyou can make things more free but not less
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13:25.30benjkthe faq you quoted even says differently
13:25.42mog_homenot the one qwell quotes
13:25.53mog_homeand yours
13:25.56mog_homeyou just misread
13:26.13trixterahh so including lesser gpl code in the cdr stuff for asterisk makes it more free?
13:26.26mog_homeno
13:26.27mog_hometrixter
13:26.30zoano
13:26.37mog_homegpl software can use lgpl software
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13:26.38trixter<PROTECTED>
13:26.38trixter<PROTECTED>
13:26.42tzafrir_laptopasterisk has any issue using LGPL stuff? as in GNU TLS?
13:26.42trixtercdr_tds.c
13:26.43zoagpl is oke with lgpl
13:26.59trixterits not linked its included in a file copyright digium
13:27.02mog_homehowever you cant make gpl software lgpl
13:27.04oejThe LICENSE file in Asterisk says: If you obtained Asterisk under the GPL, then the GPL
13:27.08oejapplies to all loadable Asterisk modules used on your system as well,
13:27.10oejexcept as defined below. The GPL (version 2) is included in this
13:27.13oejsource tree in the file COPYING.
13:27.18trixteryeah
13:27.20*** part/#asterisk mog_home (n=mogorman@user-24-236-84-48.knology.net)
13:27.20trixterthat is lgpl code
13:27.22trixterso the gpl applies
13:27.25trixterwhich is more strict
13:27.36trixterkeep in mind that is part of not linked into part of asterisk
13:27.46trixter<PROTECTED>
13:27.47trixter<PROTECTED>
13:27.47trixter<PROTECTED>
13:27.49trixtertop of the same file
13:28.07trixterin essence digium took lgpl code and made it gpl which is more restrictive
13:28.31trixterand then aparently provides that same stuff under different lciense terms according to the license info
13:28.35trixterie ABE
13:28.45oejtrixter: Sorry, I missed which code you are talking about now
13:28.56trixtercdr code
13:29.46oejbenjk: Section 2b of the GNU GPL version 2 sdoes not give you right to license with a compatible license, as you claimed
13:30.02trixterfurther the advertising material I have for asterisk violates other licenses held by at least netbsd and the regents of california
13:30.21oejbenjk: It strictly says that you must license under the same License.
13:30.28mutilatordamn nextel callphones
13:30.33mutilatorwon't ring in on my voip lines
13:30.35trixterbecuase it doesnt give credit where credit is due, hell half the people that have copyrights on asterisk code arent even listed in the credits file but that isnt a license requirement so ...
13:30.37mutilatorthey hear no ringing at al
13:30.38mutilatorl
13:30.38oejtrixter: Which of the CDR drivers are LGPL?
13:30.50*** join/#asterisk nitram (i=foo@superblob.com)
13:30.54trixterpart of cdr_tds.c has lgpl code in it
13:31.12trixterat the top it states that its copyright digium but down a bit it states clearly that part of it is from libc and its lgpl
13:31.18oejHmmm. And it is rumoured to be a bad driver as well. That's interesting. Wondered who disclaimed it...
13:31.19trixterand copyright FSF
13:31.39trixterit appears to have been ganked from freetds.org
13:31.42trixteror was that com
13:31.55trixternope its org
13:32.08oejWhat in there is taken from LGPL code?
13:32.15coppiceonly microsoft could name a protocol TDS :-)
13:32.33oejIgm STRISTR
13:32.56trixterhmm indications.h is lgpl too
13:33.06*** join/#asterisk brettnem (n=brettnem@72.29.102.158)
13:33.21oejGuess someone missed this. Thanks for telling us. I will check this.
13:33.58oejThe author (Stephen R. van den Berg) might have given us rights to use it, but I do not know.
13:34.06trixtermake sure that the advertising materials are also compliant with the netbsd and the regents of CA (2 different licenses largely identical just attribution must be different)
13:34.36trixterand I think MIT too requires advertising stuff have attribution to them but I could be wrong on that one, I know that marketing materials for asterisk must include netbsd and regents credits on it
13:35.07oejWhere do we have that code?
13:35.33trixterI will have to find it again
13:35.44trixterbut its there
13:35.55oejHmm. Indications.h is from zapata telephony.
13:36.08trixterand I am sure that the aparent licensing problems will be fixed by the next version one way or the other
13:36.11oejWe have some BSD stuff in strings.h I believe.
13:36.21oejtrixter: yes, but we always have to watch out
13:36.50trixter./asterisk-1.2/editline/readline.c
13:36.50trixter<PROTECTED>
13:36.50trixter<PROTECTED>
13:36.50trixter<PROTECTED>
13:36.51trixter<PROTECTED>
13:37.17trixterreadline is a feature of the cli is it not?
13:37.51*** join/#asterisk [TK]D-Fender (n=joe@toronto-HSE-ppp4122655.sympatico.ca)
13:38.17trixterthat is just one example, there are a few places where code was used from other projects that have different non gpl licenses
13:38.33trixterI think there are at least 3 different lgpl things in asterisk
13:41.45benjksome plug-ins are LGPL
13:42.17steffhi, anyone have capiCD working ?
13:42.38trixterzaptel is lgpl
13:43.08trixterit kinda has to be since some include files and such as well as some C files are lgpl and it would be hard to make part lgpl and others gpl only
13:44.00oejAh, readline. Yes, it's the CLI.
13:44.02zoa<oej> Hmm. Indications.h is from zapata telephony. -> im pretty sure that this was disclaimed
13:44.04zoai think
13:44.09*** join/#asterisk Seba_soy (n=s@64.76.126.29)
13:44.21oejOh, zoa *thinks* - what a nice surprice ;-)
13:44.22zoaor could be disclaimed as that guy is very close with mark
13:44.30oej(My apologies to zoa)
13:44.34zoaso lets focus on the other files first :)
13:44.37saftsackis it possible, that i can turn on and turn off the voicemail from my telephone?
13:45.17oejI think the cdr_tds is a mistake. I found the bug report (1859) and it does not mention this code. The author of the driver disclaims all of the code, but I guess he does not have the right to this function.
13:45.29*** join/#asterisk iCEBrkr (n=icebrkr@65.32.244.169)
13:45.36iCEBrkrWHUT UP!! WHUT UP!!
13:45.38oejAnd yes, I mailed Kevin and Mark about it, to bring it to their attention. They are the licensing and code masters
13:45.41iCEBrkr^5 ^5
13:45.44*** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com)
13:45.46trixterzoa: there are problems if its disclaimed though because the credits dont state all the authors and if someone contributes to a lgpl product its hard to make that anything but that license without everyone signing off on it
13:45.57[TK]D-Fendersaftsack : as in toggle the voicemail for an extension on your server?  Can be done in dial-plan raterh easily
13:46.18saftsacki thought, that a variable will be written, if someone does type some number
13:46.21oejtrixter: There is a reason why we save everything in the bug tracker to be able to track things
13:46.23saftsackor is that the false way?
13:46.34zoatrixter: yes
13:46.47zoadepends if it has only 1 author
13:46.49[TK]D-Fendersaftsack : pretty close.  You need something persistant like ASTDB to stor it.  I wouldn't trusta global var to it.
13:46.55trixterthey list primary not sole
13:47.06trixterthey clearly dont say there was only one and leave room to believe that there were others
13:47.18trixtermark is the primary for tonezone.c/h and um one other in  the same dir
13:47.20saftsack[TK]D-Fender, asterisk 1.1.0 hasnt astdb, right?
13:47.25oejI have suggested that commit messages should list developer as well. Someone on asterisk-users mistook them and claimed that "only Digium employees develop", since he had checked the changelog
13:47.28*** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com)
13:47.29trixterhowever if mark can submit lgpl code to asterisk why cant benjk?
13:48.00oejtrixter: It's very different
13:48.10[TK]D-Fendersaftsack : All versions did.  look up the DB() function on the WIKI
13:48.17iCEBrkrtrixter: this isn't some pollitical b.s. is it??
13:48.20saftsackok, thank you
13:48.20trixteraccording to mog its a violation to submit lgpl to asterisk
13:48.31oejtrixter: Benjk writes a module that links to Asterisk and thus is derivative work and has to be GPL
13:48.41trixteriCEBrkr: nah just pointing out the many license violations in asterisk code and hardware
13:48.41benjkin any event, we are calling Apple's library which is BSD licensed
13:48.42[TK]D-FenderiCEBrkr : So the dimwits give an update on your PRI's?  And has the brass been satisfied with your co-lo test?
13:48.44saftsack[TK]D-Fender, where to find asterisk wiki? do you mean the wiki on voip-info.org?
13:48.51oejtrixter: referring to the list of compatible licenses that we can link to
13:48.52benjkso even less restrictive than LGPL
13:48.52[TK]D-Fendersaftsack : yes
13:48.53trixteroh wait I forgot about the hardware that digium sells is gpl derrived and thus should be open
13:48.57trixterthat is a different story
13:49.00iCEBrkrtrixter: ahh
13:49.10iCEBrkr[TK]D-Fender: The miami colo is working as intended!
13:49.23oejbenjk: Yes, we can link to BSD, LGPL and a lot of different licensed libraries that will not enforce any change in Asterisk's license
13:49.26[TK]D-FenderBut was only for froof-of-concept rught?
13:49.29iCEBrkr[TK]D-Fender: As far as the local PRIs, not worried about it
13:49.44iCEBrkr[TK]D-Fender: Either way it would have been a proof of concept test.
13:49.45benjkour module doesn't force any change in the Asterisk license
13:49.47saftsack[TK]D-Fender, ODBC connection?
13:49.49oejbenjk: However, we do not want to link to a GPL licensed library, since that has serious impact of the Digium commercial license that Digium does not want
13:49.50[TK]D-FenderI would be given the cost of the card :/
13:49.52*** part/#asterisk KaZeR (n=kazer@81.80.32.245)
13:49.58iCEBrkr[TK]D-Fender: The whole reason for the miami colo was it was supposedly quicker to get provisioned for testing.
13:50.01[TK]D-Fendersaftsack : No, ASTDB is an internal DB1 database.
13:50.07saftsackok
13:50.11oejbenjk: No, but if you write a module that links to Asterisk, it will be GPL and GPL *only*
13:50.17[TK]D-FenderiCEBrkr : So your ass is out of the grinder :)
13:50.21iCEBrkr[TK]D-Fender: So far.
13:50.22oejbenjk: A module that uses the Asterisk API
13:50.24trixterso lemme understand
13:50.25zoaoej, unless you pay to digium
13:50.33tzafrir_laptopNote, however, that you can use a GPL-only asterisj. Basically.
13:50.33zoawhich is possible even without abe
13:50.34oejzoa: Right
13:50.35trixterasterisk can link to whatever and anyone else can only link if they are gpl
13:50.42saftsack[TK]D-Fender, there are thousands of entries :( but no is a documentation
13:50.50trixterbut doesnt that mean that the code that asterisk links to has to be gpl too since this linking is a two way street
13:50.59zoaasterisk cannot link to whatever
13:51.01oejtzafrir: Right, you can write code that links to GPL and not ask digium to include it in Asterisk. It will still be GPL
13:51.05zoathe gpl version of asterisk can do that
13:51.08oejtrixter: No
13:51.10zoathe abe cant
13:51.21iCEBrkr[TK]D-Fender: But the rules of the game changed.  My boss was going to be really kind to my team.  He was gonna split $25k between the 3 of us. :D  Since we were going to have to spend $25k on new hardware + licenses on our current IVR solution.
13:51.23oejtrixter: It depends on the license of the library we link to
13:51.33*** join/#asterisk Enderson (n=enderson@smtp.gentoo.org)
13:51.34iCEBrkr[TK]D-Fender: But now, it appears we don't need that extra hardware + licenses to get the job done. :(
13:51.42zoaice, bring some to digium :)
13:51.48trixterahh so if you link to lgpl code used inside your stuff that is ok but for benjk to release his software as lgpl that is bad
13:51.49trixterI see
13:51.50iCEBrkrzoa: I'm trying! I'm trying!
13:51.56trixterI think I am finally understanding
13:51.57trixterthanks
13:52.00oejtrixter: Right!
13:52.05trixterits a one way street
13:52.13trixterdigium can do whatever and everyone else must do what digium wants
13:52.19trixterat last I see what you mean
13:52.19zoanot true
13:52.23oejAnything that links to Asterisk has to follow the license for Asterisk
13:52.25[TK]D-Fendersaftsack : exten => _*11,1,Set(DB(disblevmfor/${CALLERID(num)})=YES)
13:52.27zoadigium can do that only for the gpl version
13:52.34[TK]D-Fendersaftsack : exten => _*12,1,Set(DB(disblevmfor/${CALLERID(num)})=NO)
13:52.35oejAnd if Asterisk links to something, we have to follow that license
13:52.37zoafor the abe digium cant do anything
13:52.46benjkit has to be compatible with GPL and that's it
13:52.52[TK]D-Fendersaftsack : theres your 2 functions then you check for them in your dialing sections.
13:52.55zoawhich means abe is asterisk stripped of all non digium owned gpl
13:52.58trixteroej just said that digium can link to non gpl code, bsd, lgpl etc..  and even include it within parts of their other code.  but if a 3rd party writes lgpl software asterisk cant link to it without violating
13:52.59iCEBrkrtrixter: Hey? You're just not understanding that Digium wears the pants??
13:53.04*** join/#asterisk brettnem (n=brettnem@72.29.102.158)
13:53.07*** join/#asterisk file[laptop] (n=jcolp@mctnnbsa24w-142167049176.nb.aliant.net)
13:53.10trixterthat is what I said..  its digiums way for whatever and everyone else has to do what they want
13:53.34iCEBrkrDigium's way or the highway
13:53.35trixterso if anything that links to asterisk has to follow the license why doesnt asterisk have to follow the lcienses of whatever it uses?
13:53.36benjkbesides if anyone is lax on licensing then it is Digium
13:53.37zoano, asterisk can only link equally unrestricted or even more unrestricted licenses than theirs
13:53.39oejtrixter: I don't think we can include LGPL code or GPL code within Asterisk, I did not say that. We can link to libraries with LGPL.
13:53.41drrayas opposed to dealing with cisco
13:53.51iCEBrkr*Whistles*
13:53.53trixterbut you do include lgpl code and other licenses without following them
13:53.56*** join/#asterisk zeronature (n=ws@60.48.199.177)
13:53.56oejBut the inclusion of code that you found is something that I believe is wrong
13:53.57benjknot giving credits is about the worst thing you can do
13:54.00[TK]D-FenderSet(isvmdisabled=${DB(disablevmfor/${exten})})
13:54.01zoaso, if bsd includes gpl code, than it becomes gpl licensed
13:54.01trixteryou convert lgpl licenses into gpl licenses at will
13:54.06[TK]D-Fenderand gotoif on that.
13:54.14trixteryou dont follow the netbsd and regents of CA license as stated in the code
13:54.26zoaif some gpl licensed code includes bsd license, than it stays gpl
13:54.26oejzoa: That is why OpenBSD and FreeBSD people write their own implementations of everything
13:54.27iCEBrkrDamnit, I gotta change the idle timeout on this ssh session
13:54.27drrayGPL has not been tested in court, I'd be weary of what people in IRC or otherwise tell you about it
13:54.47oejtrixter: I don't know if the ABE does that or not, that's something that Digium have to take care of.
13:54.52trixterzoa: but that doesnt work if 1. you violate the bsd code license (there are a few licenses you need to read the specific one) and 2. if you take lgpl code you cant just magically say its gpl now
13:54.54saftsack[TK]D-Fender, thank you :)
13:54.58zoatrig: true
13:55.08zoaoh that you can
13:55.21zoabut if you take only that code, its again lgpl
13:55.24oejtrixter: There is a difference between linking to LGPL code and actually copying the source
13:55.38trixtergpl is more restrictivve you cant add extra conditions to that code
13:55.42oejtrixter: We can't re-license the code, only the author and license holder can
13:55.57trixteryeah and you guys copy in the code as in cut and paste
13:56.13trixterand dont follow the license terms of that code
13:56.19oejtrixter: Which guys?
13:56.34trixterwell digium is the one distroing it
13:56.40oejtrixter: Someone submitted that code and disclaimed it all, without bothering to check the license. We have to review that part again.
13:56.44trixterwith claims that they can at will change the license for anything in there
13:57.02oejtrixter: And I don't really know if the TDS support is included in ABE or not, so I can't say anything about it.
13:57.09*** part/#asterisk SlackUser_ (n=acabi@200.194.102.234)
13:57.16zeronatureanyone know the error of Dec 22 19:47:07 NOTICE[14388]: chan_sip.c:10817 handle_request_register: Registration from 'Zaky <sip:zeronature@68.0.0.196>' failed for '68.0.0.127' - Username/auth name mismatch
13:57.29trixterwell there is always the bsd code that has advertising restrictions, specifically that any marketing materials give credit
13:57.32oejtrixter: I don't believe Digium claims they can change any license for any code at all.
13:57.35iCEBrkrzeronature: um, just as it says
13:57.57trixteroej: they say they have an umbrella license and can grant other licenses even other open source licenses for asterisk
13:57.57oejtrixter: Is it really "any marketing material"?
13:58.14trixterI believe its in advertising
13:58.24trixterwhich certainly includes the stack of flyers I have
13:58.27oejYes, digium has a license to or copyright of all of asterisk in the repository and can license that any way they want
13:58.41trixteroej: no they dont
13:58.48oejExcluding those libraries that is included and has a separate license
13:58.48benjknot any way they want
13:58.55benjkthey have to abide by the license terms
13:58.58trixterand I gave you several examples to show that digium is violating a few different open source licenses with asterisk
13:59.03drrayor not include it
13:59.08*** join/#asterisk jake1932 (n=jake1932@pool-68-236-10-151.phil.east.verizon.net)
13:59.17trixternot including it is part of following the license
13:59.32iCEBrkrYou guys are making me lose my hardon
13:59.33oejtrixter: Let's discuss that with Mark and Kevin and see what their response is.
14:00.00drraythey'll tell you that what you are suggesting is bunk
14:00.05*** join/#asterisk bjohnson_ (n=bjohnson@jecinc.tor.istop.com)
14:00.05trixtermy guessi s that the code will be 'rewritten' (ie variable names changed not much else) and the license notices removed
14:00.26iCEBrkrre-indent too
14:00.27trixterhey its not my fault they use code with at least 5 different not all gpl licenses
14:00.37trixterits not my fault that they dont follow the license terms
14:00.38*** join/#asterisk littleball (n=littleba@cm179.epsilon174.maxonline.com.sg)
14:00.53trixtereven their hardware is based off gpl designes so technically their newer cards that are based off that need to be open too
14:00.54*** join/#asterisk pengyong (n=lala@218.93.146.221)
14:01.13oejYou are allowed to include BSD code, but you have to follow the license as you say
14:01.31trixtercorrect
14:01.36zoadoesnt have to trixter, if they bought it from whoever had the copyright on it, they could still do as they want, except revoke the gpl license
14:01.50trixterbut they didnt
14:01.51*** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com)
14:01.56zoayou dont know that
14:01.58trixternor did they claim that
14:02.00zoaneither do i
14:02.05trixterheh
14:02.11zoaim not sure its based on the original card either
14:02.13zoaand i dont think it is
14:02.15trixterits nice to know what I know isnt it?
14:02.21zoaas the design is completely different
14:02.21zoa:)
14:02.27littleballhello, how does call forward works in asterisk?
14:03.00iCEBrkrIt works a lot like call fowarding
14:03.01oejThe license to editline says that "All advertising materials mentioning features or use of this software must display the following..:"
14:03.10trixterI bet that asterisk 1.2.3 will be out within a few days mostly just changes in code comments and variable names to avoid any lciense issues
14:03.16[TK]D-Fenderlittleball : depends on your phone and your dial-plan (VoIP hard phones typically have a hard transfer button, others rely on DTMF)
14:03.20oejI don't think any marketing material mentions editline
14:03.23zoaoej, that doesnt seem to be a problem
14:03.24zoatrue
14:03.27zoai think the same
14:03.29trixteroej: wow that looks like an abbreviated version of what I pasted
14:03.32zoaor even features of editline
14:03.45trixterthere are other places than that
14:03.50trixterthat was just one example
14:04.09oejastDB also include DB version 1 with BSD license
14:04.56iCEBrkrtrixter: Haven't you contributed a boat-load of asterisk code?
14:04.59benjkand why would they not just give credit
14:05.03oejThe same license as editline
14:05.07benjkit doesn't cost them anything
14:05.17benjkits really childish not to give credit
14:05.18littleballi have E1 line channels, when asterisk receive a call from PSTN networks(because some PSTN user call one of my E1 line number), i would like to "forward" this call to other place so that other PSTN users still can call the same E1 line number...
14:05.21*** join/#asterisk docelmo (n=docelmo@static-71-251-95-4.tampfl.fios.verizon.net)
14:05.26iCEBrkrdocelmo: werd, homie
14:05.31oejThe credit file has been poorly maintained
14:05.48littleballjust like server socket
14:05.49*** join/#asterisk dsfr (n=dsfr@pdpc/sponsor/digium/dsfr)
14:05.57trixteryeah why do half the copyright holders not exist in the credits file?
14:05.59jake1932littleball: other place?
14:06.02zeronatureanyone know what's my problem is??
14:06.03trixterwhy try to keep the authors more secret like that
14:06.06*** join/#asterisk JaredBluestein (n=Jared@nwlnnhbas01-pool3-a32.nwlnnh.tds.net)
14:06.12oejBut blaim all of the Asterisk developer community, not just Digium
14:06.13zeronaturei'm already post it at forum asterisk
14:06.14*** join/#asterisk af_ (n=af@ip-142-84.sn1.eutelia.it)
14:06.20trixterespecially in violation of even the editline license
14:06.33trixterthey arent listed in the credits file, which I think its sec 1 maybe 2 of their license
14:06.35jake1932littleball: you mean take call call totally off of your E1?
14:06.40iCEBrkrtrixter: Cuz Mark wrote it all.. Geesh man..
14:06.41oejWe do not try to keep authors secret, if people submit patches to CREDIT that we agree to, we do submit them.
14:06.41iCEBrkrduh
14:06.42littleballjake1932, what i want is to make one of my E1 line number like TCP server socket. For TCP server socket, it receive a connection request
14:07.03littleballit create a socket and then continue to listen
14:07.05jake1932littleball: I think what you're referring to is B2B xfer
14:07.21littleballjake1932, maybe what is B2B xfer?
14:07.22zeronaturei'm using xlite software phone running on winxp
14:07.38zeronatureand using asterisk 1.2 at fedora core 4
14:08.14littleballjake1932, how does B2B xfer works?
14:08.21*** join/#asterisk robl^ (n=robl@dsl093-025-118.hou1.dsl.speakeasy.net)
14:08.28zeronatureDec 22 19:47:07 NOTICE[14388]: chan_sip.c:10817 handle_request_register: Registration from 'Zaky <sip:zeronature@68.0.0.196>' failed for '68.0.0.127' - Username/auth name mismatch
14:08.37benjkok, time to go to the pub and have a pint of Guinness
14:08.38zeronaturehave to edit sip.conf??
14:08.39jake1932littleball: i think you need a 5ESS switch at the CO
14:08.47iCEBrkrbenjk: Have one for me too
14:08.52[TK]D-Fenderzeronature : Its saying your softphone isn't using the right username or password.
14:08.54benjkhehe
14:09.04trixterwash your mouth out first
14:09.07trixterhe doesnt want your germs
14:09.08jake1932littleball: http://www.voip-info.org/wiki-Asterisk+bounty+PRI+2B+channel+transfer
14:09.10saftsack[TK]D-Fender, why is there a 11?
14:09.11[TK]D-Fenderzeronature : its as obvious a message as it sounds
14:09.17[TK]D-Fendersaftsack : where?
14:09.17littleballthanks jake1932
14:09.21saftsack_*11,1,Set
14:09.21iCEBrkr[TK]D-Fender: yea, really
14:09.32zeronatureso what i need to do?
14:09.33iCEBrkrzeronature: it's telling you exactly what the problem is.
14:09.39[TK]D-Fendersaftsack : Its just a sample dammit!  Make the activate/deactivate codes whateve you want.
14:09.39iCEBrkrzeronature: you need to read the wiki
14:09.41iCEBrkr~wiki
14:09.43iCEBrkr~docs
14:09.45jbot[docs] probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html
14:10.16zeronaturei'm really excited to find out
14:10.17*** join/#asterisk [wiebel] (i=wiebel@cp883187-a.tilbu1.nb.home.nl)
14:10.22[wiebel]re
14:10.31saftsack[TK]D-Fender, is there something like if for the extensions.conf?
14:10.50[TK]D-Fendersaftsack : Maybe you haven't been paying attention....
14:11.01[TK]D-Fender[08:54] <[TK]D-Fender> and gotoif on that.
14:11.11[TK]D-Fender> GotoIf <
14:11.32[TK]D-FenderWIKIwikiWIKIwikiWIKIwikiWIKIwikiWIKIwiki
14:11.33[TK]D-Fender:D
14:11.43iCEBrkrhehe
14:11.51zeronaturewhere to find wiki??
14:12.01[TK]D-FenderObey the jbot!
14:12.02zeronaturesorry to ask..coz i'm really new user
14:12.13iCEBrkr[TK]D-Fender: I don't understand why people can't figure this shit out.  I'm not all that smart and I leared how to configure my setup from the Wiki.
14:12.13[TK]D-Fenderlook up at [docs]
14:12.39[TK]D-FenderiCEBrkr: Yeah, I work well with programmers references.  Gimme the commands and let ME choose where to use them.
14:12.49[TK]D-FenderExtensions really IS that simple.
14:13.14iCEBrkr[TK]D-Fender: I actually taught myself how to program flipping through the Turbo Pascal 5.5 language reference manual.
14:13.26iCEBrkrSo I do the same for everything else
14:13.40*** join/#asterisk Katty (n=angela@68.112.15.110)
14:13.45iCEBrkrAsterisk - documentation of application commands
14:13.45[TK]D-FenderiCEBrkr : I started much earlier with BASIC and continued with TP 5.5 and learned that straight from the help menu :)
14:13.47Kattymorning lads.
14:13.49iCEBrkrThat's my favorite page
14:13.53iCEBrkrKatty: ^5
14:14.07[TK]D-FenderKatty : mew?
14:14.10iCEBrkr[TK]D-Fender: I can't remember how I learned BASIC
14:14.22Katty[TK]D-Fender: mew?
14:14.40[TK]D-Fender:)
14:14.59Kattyit's like babysitting.
14:15.05Kattydon't touch that! don't download that!
14:15.13Kattydon't put that in your mouth, you don't know where it's been! etc.
14:15.21[TK]D-FenderYou're reloading... you ARE God to them :)
14:15.34littleballjake1932, it seems that asterisk cannot do such job, right?
14:15.36*** join/#asterisk pingywon (n=mike@pcp0010034410pcs.reding01.pa.comcast.net)
14:15.43[TK]D-FenderKatty: get your foot out of the gutter.... you're standing on my head :D
14:15.48Katty[TK]D-Fender: the bossman's laptop kept bsodding.
14:15.58jake1932littleball: with 5ESS it's supposed to be able to do it
14:16.22[TK]D-FenderKatty : Thats why I Ghost all of my systems here... just to save me the effort later....
14:16.24littleballjake1932, just assumeing it is 5ESS switch, how to do?
14:16.32Katty[TK]D-Fender: ghosting is good.
14:16.46Katty[TK]D-Fender: but still have to do windows updates and other things.
14:17.05[TK]D-FenderKatty : true, but thts a quick batch compared to the barrage of apps I install.
14:17.15Katty[TK]D-Fender: bossman doesn't have much, really.
14:17.18[TK]D-FenderI have a largely OSS system setup CD for my network.
14:17.29Katty[TK]D-Fender: office xp, abbyyfine reader...
14:17.38Katty[TK]D-Fender: some adobe pro stuff.
14:17.45iCEBrkrEEP!
14:17.46jake1932littleball: i'd have to review the docs
14:17.47iCEBrkrHELP!
14:17.59[TK]D-FenderI give OOo, PDF writer, Acrobat, tons of CODEC's, NVu, GIMP, 7-ZIp, and TONS of other stuff
14:18.01iCEBrkrI started writing a PHP loop as
14:18.09iCEBrkrfor x = 1 to 100
14:18.11jake1932littleball: maybe Transfer
14:18.12[TK]D-Fenderlol
14:18.21Katty[TK]D-Fender: ya, he doesn't use all that.
14:18.32*** join/#asterisk klictel (n=klictel@207.107.208.137)
14:18.36*** join/#asterisk Dibbler (n=Dibbler@snaddy.plus.com)
14:18.43littleballjake1932, let me see
14:18.44[TK]D-FenderKatty : Thats never the point with me.  Its a principals thing...
14:18.45*** join/#asterisk fryfrog (n=fryfrog@gallery/fryfrog)
14:18.48Katty[TK]D-Fender: i probably don't use as much as you either....
14:18.55Katty[TK]D-Fender: but i use a good bit
14:18.57iCEBrkrIT's about the PRINCIPALITIES
14:19.12KattyiCEBrkr: weirdo.
14:19.20iCEBrkrKatty: I never claimed to be anything else.
14:19.27KattyiCEBrkr: k
14:19.28fryfrogis "Asterisk@Home" a reasonalbe way to explore Asterisk as a total PBX (but not linux) newb?
14:19.35iCEBrkrfryfrog: nope
14:19.58iCEBrkrfryfrog: I think Asterisk@home is more detremental to your learning of things :)
14:20.09iCEBrkr...then again, I like to get my hands dirty.
14:20.10fryfrogi see
14:20.26fryfrogi agree with getting hands dirty, but i don't know *anythign* about corporate phone systems :)
14:20.31Kattyit's like 98 compared to xp pro, fryfrog
14:20.34fryfrogdo you suggest anything specific to read about?
14:20.36Kattyit's horror.
14:20.36iCEBrkrfryfrog: I didn't either.
14:20.44iCEBrkrfryfrog: I'm not even really a 'telco' kinda guy..
14:20.59Kattyfryfrog: i didn't have a clue either...in fact, i had to go read about voIP first :P
14:21.07iCEBrkrfryfrog: Ok, so I used to phreak back when I was 15, so I had a little understanding about phone systems and telco stuff.. But that doesnt' mean anything :)
14:21.09fryfrogyes, that is the stage i'm at :)
14:21.15fryfrogi know what voip *is* but thats about it
14:21.20Kattyfryfrog: excellent.
14:21.26oejtrixter: Thank you for the discussion and the feedback :-)
14:21.26*** join/#asterisk mog_work (n=mogorman@gateway.digium.com)
14:21.31Kattyfryfrog: next step is to get yourself framilier with linux.
14:21.39fryfroglinux isn't a problme
14:21.40robl^A@H is ok for a VERY VERY VERY basic setup, but the more complext the configuration, the more the more the AMP interface becomes a hinderence rather than an aid
14:21.43*** join/#asterisk jontow (i=jontow@secure-bsd.be)
14:21.45Kattyfryfrog: wonderful.
14:21.49Kattyfryfrog: then you're all set!
14:21.55mog_workamen robl^ speak the truth
14:22.03Kattyfryfrog: just go download/install asterisk and poke around at it
14:22.27fryfrogi guess what i'd like to read about is ...
14:22.29fryfrogbah, phone
14:22.34iCEBrkrfryfrog: Yea, Asterisk is editing a bunch of files and reading up on how to make it do karthweels. ;)
14:22.44robl^I use A@H for quick and dirty installs.. then remove all A@H config files and add my own.  :)
14:22.46*** join/#asterisk klictel_ (n=klictel@207.107.208.137)
14:22.47iCEBrkrfryfrog: http://www.voip-info.org
14:22.59iCEBrkrrobl^: That's pretty dirty
14:23.57robl^iCEBrkr: its for my test box that gets re-installed 20x a week :)  production is different..
14:24.21Kattygod bless my caffinated soda.
14:24.23iCEBrkrI guess.
14:24.26Kattyi am /social/ this morning!
14:24.30Kattyhow scary is that?
14:24.32lo_techthe 5ess can do a 'Transfer-Connect', but only on 800 numbers and then only if they have the feature associated with that 800#... and it's a fee each time it's used.
14:24.33saftsack[TK]D-Fender,  exten => 157925,2GotoIf($[${DB(office/anrufbeantworter)} = 1]?anrufbeantworter
14:24.43saftsackdoes it now jumps to the context anrufbeantworter?
14:24.51iCEBrkrKatty: I just think it's one of those 3 days you're not PMSing
14:24.54saftsackif anrufbeantworter=1 ???
14:24.55iCEBrkr>: )
14:25.04KattyiCEBrkr: i PMS everyday, dear.
14:25.09KattyiCEBrkr: just to annoy you.
14:25.12iCEBrkrKatty: i think that's what I was saying.
14:25.20[TK]D-Fendersaftsack : no, that'd look for a PRIORITY named that.
14:25.32KattyiCEBrkr: if it's any consolation to you, i got my neck out.
14:25.35saftsackok
14:25.37KattyiCEBrkr: and it hurts.
14:25.47iCEBrkrI've been suffering from PMS since July 3rd.
14:25.48[TK]D-Fenderexten => 157925,2GotoIf($[${DB(office/anrufbeantworter)}=1]?anrufbeantworter|s|1)
14:25.54saftsackthank you :)
14:26.00KattyiCEBrkr: aww, poor baby.
14:26.03[TK]D-Fendersaftsack : that would jump to s,1 in that context
14:26.17iCEBrkrKatty: What'd you do?  Your neck? huh?
14:26.28KattyiCEBrkr: i pulled a muscle in my neck.
14:26.33KattyiCEBrkr: hurts to rotate, etc.
14:26.34iCEBrkrOw
14:27.04iCEBrkrKatty: I've been down that road.  Can't turn your neck for shit.  Makes it painful to drive cuz you can't check over your shoulder without turning at your torso
14:27.29Kattythank god i don't have kids.
14:27.42lo_techgo faster than everyone else and you wont have to look :P
14:27.48Kattylo_tech: silly.
14:27.50*** join/#asterisk hypa7ia (i=hypatia@wsip-24-234-241-145.lv.lv.cox.net)
14:28.01iCEBrkrlo_tech: Exactly
14:28.20iCEBrkrSeriously, it's pretty dangerous to drive when you can't turn your head
14:29.18trixterwhen my father had cancer the 2nd time it was dangerous for him to be in a car for 6 months period
14:29.43Kattytrixter: silly rabbit.
14:29.52trixtercause it was the 2nd vertebrae under his skull where the cancer was and he had to have a pin put in cause too much of the bone was gone, one accident even a slow speed one could have caused him to break his neck
14:30.02trixterkatty: =^.^=
14:30.10iCEBrkrtrixter: Geesh
14:30.21*** join/#asterisk brad_mssw (n=brad@shop.monetra.com)
14:30.21trixterhe had to wear a neck brace anytime he got into a car
14:30.39trixterbasically a cervical collar, which meant he couldnt drive cause he couldnt turn his head at all
14:30.42Kattytrixter: ))))=
14:30.53saftsack[TK]D-Fender, i copied this from a page, but that doesnt work :(
14:30.54iCEBrkrtrixter: My friend had a two tumors in his lower back. So now it's not good for him to play basketball/football with us cuz there's no bone there anymore and any hits to that area could paralyze him.
14:30.55saftsack<PROTECTED>
14:30.56trixterKatty: you dont like my kitty drawing?
14:31.08Kattytrixter: that's not very catlike.
14:31.18Kattytrixter: =^@_@^= is more kittylike.
14:31.18trixterwell after 6 or so months with the pin the bone grew around it, he is more or less fine now
14:31.30trixterdepends on the cat :P
14:31.31saftsackno application 'Set' fpr extension (raus, 7, 1)
14:31.34iCEBrkrKatty: That cat looks like it saw a dog
14:31.42trixter<PROTECTED>
14:31.44KattyiCEBrkr: it saw /you/
14:31.50[TK]D-Fendersaftsack : pastebin it.  I need deatils
14:31.52iCEBrkrKatty: Oh come'on, I'm not that scary looking
14:31.53trixtersaftsack: what version?
14:32.00saftsackasterisk 1.1.0
14:32.00KattyiCEBrkr: well someone has to pick on you.
14:32.08KattyiCEBrkr: if i'm pestering you, i'm leaving fender alone ;)
14:32.11trixterset/setvar are for 1.2 and 1.0 so you might be using the wrong one for the version you are running
14:32.17iCEBrkrKatty: ...On second thought maybe I am.  I scared the crap outta a co-worker this morning.
14:32.22saftsackok i paste my extensions.conf
14:32.31[TK]D-Fendersaftsack : Are you on 1.2.x?
14:32.34saftsackno
14:32.36saftsackolder asteirsk
14:32.36iCEBrkrKatty: You're only giving me crap cuz he's not here
14:32.38iCEBrkrerr
14:32.39Kattyhaha, boss's laptop is being spammed.
14:32.44iCEBrkrKatty: there he is
14:32.44trixtersaftsack: try setvar
14:32.47saftsackok
14:32.47[TK]D-Fendersaftsack : my sample was for 1.2.x
14:32.52[TK]D-Fenderit won't work.
14:32.52Kattyi have asterisk send a message with smbclient to computers.
14:33.03Kattyand i forgot that i renamed a computer...and then gave boss's laptop that computer name
14:33.05saftsackyes but i think, that set should work for 1.1.0 too
14:33.06[TK]D-Fendertrixter : he's working with ASTDB, setvar isn't it....
14:33.10iCEBrkrKatty: LOL
14:33.10Kattyso asterisk is now spamming boss's laptop with smbclient
14:33.11DaminiCEBrkr: WHAT IS UP DUDE?
14:33.12trixterahh
14:33.14[TK]D-Fendersaftsack : NO
14:33.16iCEBrkrDamin: Hey man!
14:33.17trixterI only saw the one line he posted so ...
14:33.23jake19321.1???
14:33.23saftsackok ^^
14:33.26Kattyi should leave it on just to annoy the crap out of him ;)
14:33.33DaminiCEBrkr: How goes it in FLA?
14:33.35iCEBrkrDamin: bastard not calling me back!  That's ok, I got it all under control...
14:33.38trixterjake1932: 1.odd is dev versions
14:33.43jake1932ok
14:33.49KattyDamin: hi.
14:33.49iCEBrkrDamin: It's a balmy 47 degrees in Florida :-/
14:33.51[TK]D-Fendersaftsack : UPGRADE
14:34.06saftsackupgrade or setvar?
14:34.13jake1932upgrade
14:34.13DaminKatty: Hey there...
14:34.17saftsack<PROTECTED>
14:34.18saftsackok
14:34.19[TK]D-Fendersaftsack : Upgrade your Asterisk install!
14:34.19iCEBrkrupgrade
14:34.27saftsackoh ^^
14:34.28saftsackwhy?
14:34.35jake1932haa
14:34.38KattyDamin: staying out of trouble?
14:34.41saftsackits a production environment
14:34.46[TK]D-Fendersaftsack : because the DB cfunction doesn't even exist in your version!
14:34.55saftsackohh hahahaha
14:34.56saftsack^^
14:35.00Kattyhorror.
14:35.03saftsacki asked that before
14:35.04iCEBrkrDamin: Got my TE100P working. Been doing some proof of concept tests and all the answering machine detection code seems to be working as intended!
14:35.12[TK]D-FenderYou'd need to use DBGet / DBPut.  Go read those on the WIKI.
14:35.27saftsackin 1.1.0?
14:35.29iCEBrkrKatty: Damin's married with kids, (err kid) he doesn't have time to get into trouble.
14:35.48[TK]D-Fendersaftsack : yes.  And what is 1.1.0 ?  Not a version I've ever heard of....
14:35.56*** join/#asterisk tmccrary (n=tmccrary@68.78.185.254)
14:35.57KattyiCEBrkr: pfft.
14:36.07KattyiCEBrkr: everyone has time to cause trouble.
14:36.20iCEBrkrPLUG IT IN THE VCR! DUMBASS
14:36.28iCEBrkrThose! Where the days of 'trouble'
14:36.38*** join/#asterisk fugitivo (n=ajf@209.13.240.236)
14:36.38iCEBrkrKatty: :)
14:37.04Kattyok, i have a dumb question.
14:37.07iCEBrkrDooobee-dooobee-doooo
14:37.12iCEBrkrKatty: No such thing.
14:37.19Kattywhen using FOP, i can tell that asterisk is holding a line open for some reason
14:37.37Kattylike like 1 is always busy......yet no one is using it (analog)
14:37.44Kattys/like/line/
14:37.53saftsack[TK]D-Fender, the older stableversion
14:38.06Kattyusually, i stop asterisk and start it up again...and everything is all better.
14:38.17Kattybut i'd /love/ to not stop asterisk to Fix It(tm)
14:38.29*** join/#asterisk tdonahue (n=tdonahue@208.51.101.201)
14:38.32Kattyand destroying zap channels is not good :<
14:39.52saftsack[TK]D-Fender, so it isnt possbile to handle db's in the old asterisk?
14:40.08iCEBrkrKatty: are you sure the line is 'locked'?
14:40.16KattyiCEBrkr: we can't dial out on it.
14:40.20KattyiCEBrkr: and nothing comes in on it
14:40.31KattyiCEBrkr: i view that as a line being held open.
14:40.42KattyiCEBrkr: like it was hungup properly or something
14:40.43iCEBrkrKatty: zap show channel <x>
14:40.46iCEBrkrhave you done that?
14:40.49KattyiCEBrkr: yes.
14:41.30KattyiCEBrkr: i recall the last line, about hookstate, said offhook
14:42.07iCEBrkrwhacked
14:42.13KattyEcho Cancellation: 64 taps, currently OFF <- wow, that sorta bothers me.
14:42.28iCEBrkrEcho Cancellation: 128 taps, currently OFF
14:42.31iCEBrkr:)
14:42.38Kattyoh, off doesn't mean like...off off
14:42.41Kattyjust currently off
14:42.46Kattyor something.
14:42.54fugitivoKatty: off is when channel is not being used
14:43.22Kattyfugitivo: but what about echo cancellation currently off?
14:43.34fugitivoKatty: use the channel and it'll be ON
14:43.43Kattyfugitivo: oh ah. kthen
14:43.47fryfrogIs there some place that might have typical usage cases for Asterisk?
14:44.16saftsack[TK]D-Fender, is it possible to do db things without 1.2.x?
14:44.20Kattyfryfrog: ours is a little office. 8 lines and 10 phones.
14:44.33Kattyfryfrog: well, more like 13 phones.
14:44.58fugitivofryfrog: voip-info.org
14:45.34Katty43 windows updates :<
14:45.51Katty37 of which are High Priority and Critical Updates
14:46.04*** part/#asterisk JaredBluestein (n=Jared@nwlnnhbas01-pool3-a32.nwlnnh.tds.net)
14:46.10robl^1 OS X update, 1 of which is minor
14:46.23Kattyhorror.
14:46.29Kattyand all this rebooting.
14:47.56*** join/#asterisk MikeJ[Laptop] (n=ircatjer@mi.origenfinancial.com)
14:48.01Kattyhi mike.
14:48.14*** join/#asterisk klictel (n=klictel@207.107.208.137)
14:49.13*** join/#asterisk backblue (n=moo@82.102.1.42)
14:49.18iCEBrkr<PROTECTED>
14:49.19backblueanyone using Tiger3XX cards?
14:49.24iCEBrkrYay! I get to deal with this again!
14:50.30*** join/#asterisk h4mm3r` (n=h4mm3r@host214-51.pool8253.interbusiness.it)
14:50.50robl^ODBC is evil!  even MS has pretty much tried to  kill it :)
14:51.02fugitivoyes
14:51.04fugitivoand mysql is evil too
14:51.13*** part/#asterisk h4mm3r` (n=h4mm3r@host214-51.pool8253.interbusiness.it)
14:51.18iCEBrkrWell, Corydon made a sweet ODBC function module.
14:51.33iCEBrkrTo keep any and all MySQL/SQL calls native to the dialplan.
14:51.36ManxPowerDo you know the Muffin Man, the Muffin Man, the Muffin Man?
14:51.37iCEBrkrSince AGI is too damn slow
14:51.48coppicesweet and odbc do not belong together
14:51.57robl^he module isn't bad.. it just dealing with the ODBC layer that is a pain in the bum
14:52.06fugitivowell, odbc is know to be not the best performance solution...
14:52.10coppiceManxPower: who lives on drury lane?
14:52.23fugitivoknown
14:52.37iCEBrkrfugitivo: Well, it's still about 4x faster than AGI calls :)
14:52.39backblueanyone using tiger 3xx card?
14:52.48fugitivoiCEBrkr: i like text files :)
14:53.02iCEBrkrfugitivo: um, dude, I'm dealing with a database with over a million records.
14:53.20iCEBrkrand I could me writing up to 15000 rows.
14:53.22*** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com)
14:53.22*** mode/#asterisk [+o anthm] by ChanServ
14:53.36iCEBrkrAGI won't scale.
14:53.37[TK]D-Fendersaftsack : Again, watch what I'm saying -> [09:35] <[TK]D-Fender> You'd need to use DBGet / DBPut.  Go read those on the WIKI.
14:53.40*** join/#asterisk tzanger (n=tzanger@mixdown.ca)
14:53.40iCEBrkrTextfiles aren't DB friendly.
14:53.41jake1932backblue: tiger 3xx? i have a TE110P that shows up as Tiger 3xx.  are you talking about something different?
14:53.52saftsack[TK]D-Fender, ok
14:53.54robl^iCEBrkr: use a custom app_* and do native DB calls :)
14:54.00saftsackbut why are normal variables bad?
14:54.03fugitivoiCEBrkr: a million records for the dialplan??
14:54.08backbluejake1932: hum let me see.
14:54.12iCEBrkrrobl^: What do you think Corydon's code does?
14:54.18tzangeranthm, drumkilla, twisted ... SOMEONE remove that stupid +r
14:54.25iCEBrkrfugitivo: No, for customer info
14:54.28backbluejake1932: md3200 in my case.
14:54.31robl^iCEBrkr: uses ODBC.  :)
14:54.34fugitivoiCEBrkr: robl^ means using native mysql call, not odbc
14:54.35*** mode/#asterisk [-r] by anthm
14:54.38lo_techanyone have any experience with 'nsf=megacom' via PRI?
14:54.38iCEBrkrF that
14:54.48iCEBrkrCorydon code gets the job done
14:55.03*** join/#asterisk Defraz (n=t0tal@72.24.26.215)
14:55.14iCEBrkrPlus, I'm only using MySQL for testing.
14:55.19Corydon76-homeEh?
14:55.24iCEBrkrThis place is a MS-SQL shop :(
14:55.32robl^it is all Corydon's fault
14:55.34*** join/#asterisk Gunnar (n=gunnar@bkkb-gw.bitcon.no)
14:55.55jake1932backblue: a faxmodem?
14:55.56iCEBrkrI dunno, I actually like the ODBC Function stuff.
14:56.00*** join/#asterisk malcolmd (n=malcolmd@pdpc/sponsor/digium/malcolmd)
14:56.07iCEBrkrTemplating and all
14:56.28fugitivoiCEBrkr: postgresql!
14:56.29iCEBrkrSo I don't have SQL scattered throughout my dialplan.
14:56.49backbluejake1932: i dunno, it is?
14:56.57fryfrogwow, lotta info at voip-info.org :)
14:57.02jake1932backblue: what are you trying to do?
14:57.05iCEBrkrfugitivo: Can't man.
14:57.07fryfrogcan asterisk record phone calls?
14:57.11fugitivofryfrog: yes, but don't believe all you read :)
14:57.12iCEBrkrfryfrog: yeah
14:57.24iCEBrkrI read it on the internet, it must be true
14:57.32fugitivofryfrog: you can record and spy
14:57.45fryfrogman, that'd be nice...
14:57.48Corydon76-homeThe big advantage of templates is that you have to worry about syntax errors a whole lot less
14:57.54backbluejake1932: puting asterisk to work with this, i think its a modem. its Ambient md3200, its detected as a tiger 3xx card.
14:58.05fryfrogi would *love* to call communist cast and say "hey, here is a *recording* of us speaking with your tech about this issue 3 months ago"
14:58.07fugitivofryfrog: asterisk is a complete pbx
14:58.08fryfrogSMACK!
14:58.20Corydon76-homeFix them one place, and everywhere else you reference the template
14:58.26iCEBrkrCorydon76-home: amen
14:58.38fugitivofryfrog: remember that recording is illegal in some countries
14:58.51iCEBrkrCorydon76-home: the only draw-back is having to get this damn ODBC shit setup right.
14:58.52fryfrogfor someone setting up asterisk in their home, would one typically use a few ATAs or buy some decent IP phones?
14:59.02Corydon76-homeTrue enough
14:59.06*** join/#asterisk Equinox (n=secret@pool-71-251-73-183.tampfl.fios.verizon.net)
14:59.06fryfrogfugitivo: I would preface the answering system with "Calls may be monitored or recorded"
14:59.06fugitivofryfrog: what's your budget?
14:59.10jake1932but if you're a US president, you can record whoever you want
14:59.21iCEBrkrfryfrog: I prefer ATAs with a nice cordless phone
14:59.25lo_techjake1932: /smack
14:59.28Corydon76-homebut it's the only way to do this portably, to an acceptable number of databases
14:59.29fryfrogfugitivo: I am merely reading, so i'm pretty flexible
14:59.31EquinoxDepending on the state only 1 party must be aware it's being recorded.
14:59.42iCEBrkrCorydon76-home: I hear ya.  I'm not complaining. (much)
14:59.43robl^Corydon:  I think its a good idea.. I just have a loathing of ODBC..  I wouldn't mind a native version of calls using templating.. MySQL or PostgreSQL based :)  Its just the middle ODBC layer that I loathe
14:59.47jake1932backblue: http://www.voip-info.org/wiki-Asterisk+hardware
14:59.49fryfrogI don't want to *secretly* record the calls
14:59.57iCEBrkrrobl^: It's not my favorite either
14:59.58*** join/#asterisk implicit (n=implicit@mlsrj200152100p048.mls.com.br)
15:00.02[TK]D-Fenderfryfrog : ATA's area a great cheap way to get set up.  Both for bringing in your analog line and for extensions.
15:00.10fryfrogI just want to by default record all calls, perhaps being able to turn it off with a *70 type thing
15:00.23Equinox[TK]D-Fender - How much are they running now?
15:00.28[TK]D-FenderFor those I suggest only Sipura's line (unless its a larger install that warrant a greater expense)
15:00.36fugitivofryfrog: if you have low budget, an x100p clone (for pstn $20) and a pap2-na (ata 2 fxs $60-$70) will be enough
15:00.40Corydon76-homerobl^: and what about those of us who have to use SQL Server?
15:00.41lo_techfryfrog: how many concurrent recordings?
15:00.52iCEBrkrOhio law states only 1 of the parties on the phone call needs to know the call is recorded.
15:00.53fryfroglo_tech: probably just 1
15:00.54[TK]D-FenderEquinox : SPA-2002 (2 fxs)  80$, SPA-3000 (1fxs, 1 FXO) $100
15:01.09Equinox[TK]D-Fender - Oh yeah.. I havea a SPA-3000. Great box.
15:01.10[TK]D-Fenderer, $70 for teh SPA-2002
15:01.10fryfrogi think i need to read what fxo and fxs are :)
15:01.12[TK]D-Fendermy bad
15:01.17fugitivoiCEBrkr: really? that's silly
15:01.20Equinox[TK]D-Fender, I though u were maybe talking about some $20 thingy ;)
15:01.28fugitivo~fxsfxo
15:01.30jbotfxsfxo is, like, An FXO port expects to receive dialtone and receive ring voltage.  An FXS port expects to provide dialtone and provide ring voltage.
15:01.31lo_techfryfrog: o, np... we're doing up to 480 concurrents using zap, so it'd do-able
15:01.35[TK]D-FenderiCEBrkr : I though it was both parties in the US...
15:01.42fugitivofryfrog: read what jbot said
15:01.45EquinoxIF anyone gets SPA-3000.. Upgrade the firmware.. The newer firmware is so much better.
15:01.47fryfrogI *think* I would want each "phone" in the house to be an extension, but probably only 2-3 phones
15:01.50iCEBrkrfugitivo: I belivee that's how it works.
15:01.56iCEBrkr[TK]D-Fender: It's by state.
15:02.00jake1932Equinox: as of?
15:02.00[TK]D-Fenderah
15:02.09EquinoxHmm.. I got mine 4 months ago
15:02.20jake1932Equinox: no echo?
15:02.22EquinoxThe firmware it came with didn't treat FXO/FXS as completely independent
15:02.23*** join/#asterisk enota_ (i=dimka@freelsd.net)
15:02.27EquinoxThe newer one did
15:02.28robl^and what happens when caller 1 is in Ohio and Caller 2 is in another state?
15:02.38EquinoxI don't get echo
15:02.43fugitivorobl^: good question
15:02.50iCEBrkrrobl^: I believe it matters where the call is being recorded :)
15:02.57iCEBrkrrobl^: At least that was my 'out' if I ever got busted.
15:03.02[TK]D-FenderEquinox : rare echo on my side.  the SPA-3000 is an amazing little box.  practically a PBX in its won right...
15:03.04iCEBrkrThe PBX is located in Ohio.
15:03.04fugitivorobl^: what happens when caller 1 is in the US and caller 2 in another country? :)
15:03.07saftsackhowto check, that there is now active call on my line that can lead to immense costs?
15:03.11[TK]D-Fenders/won/own
15:03.12fryfrogwhere do i find those "SPA-???" in voip-info.org?
15:03.22[TK]D-Fenderfryfrog : depends where you live.
15:03.23Equinox[TK]D-Fender, Yes I'm very impressed with it.
15:03.28[TK]D-Fenderfryfrog : online stores.
15:03.29jake1932<PROTECTED>
15:03.30saftsackso i mean howto determine, that i have a SAFE config?
15:03.32Corydon76-homerobl^: in fact, I built the templated SQL for SQL Server, because I didn't want to have to be constantly altering a C program and recompiling... ;-)
15:03.36[TK]D-Fenderwww.voipsupply.com
15:03.37fryfrogno no, i mean to read about what they are thats all
15:03.45robl^iCEBrkr: and are they "phone calls" if they are VoIP?  the FCC seems to still be debating that one :)
15:03.51[TK]D-Fenderfryfrog : yeah, the WIKI has lots of info about them
15:03.51jake1932<PROTECTED>
15:03.59fugitivo~docs
15:04.01jboti heard docs is probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html
15:04.01[TK]D-Fenderwww.sipura.com
15:04.01fryfrogbut what section are they under?
15:04.02iCEBrkrrobl^: haha
15:04.10fugitivofryfrog: there you are
15:04.13[TK]D-Fenderfryfrog : just doa  search for Sipura ATA
15:04.16jake1932<PROTECTED>
15:04.17[TK]D-Fenderon the wiki
15:04.22fugitivoon the docs
15:04.26fryfrogk
15:04.26iCEBrkrrobl^: I guess it only matters in the end--- what you're gonna do with those recorded calls.
15:04.39*** join/#asterisk IronHelix (n=irc@ool-45785cfe.dyn.optonline.net)
15:04.42fryfrogactually, just knowing they are ATAs is enough i think
15:04.58iCEBrkrrobl^: My mind is so burned out, I forget a lot of things.  So when I make 'important' calls, It's nice to record them so I have a reference as to what I said and what the other person claimed.
15:05.05robl^iCEBrkr: blackmail a rich personality with the call to a 900 "adult service"?
15:05.07fryfrogone would need *one* ATA device per phone correct? (if you wanted to do each phone as an extension?)
15:05.23jake1932depends on the num of ports
15:05.28jake1932per ATA
15:05.29Equinoxfryfrog- The Sipura 2xFXS would handle two phones?
15:05.30[TK]D-Fenderfryfrog : FXS ATA's are for pluggin in phones to use as extensions, FXO ATA's (or gateways) let you plung in LINES so that your PBX can use your analog phone lines.
15:05.31iCEBrkrrobl^: Yea, see, that's when the legallity of recording phone call becomes a problem.
15:05.36fryfrogAhhhh, i see
15:05.43*** join/#asterisk ard (n=ard@2001:7b8:32d:0:20c:6eff:fe18:d11f)
15:05.45fryfrogah, that makes sense
15:05.55fryfrogso a FXO/FXS would have 1 port of each
15:05.57[TK]D-Fenderfryfrog : you could put your whole home's phones on the same ATA, its jsut that they'd all be the same exztension and ring at once.
15:06.02fryfrogso it could handle a line in *and* 1 phone
15:06.10fryfrogahh, i see
15:06.17[TK]D-Fenderfryfrog : correct.  The SPA-3000 takes in your home line, and also gives you a seperate extension.
15:06.19*** join/#asterisk synthetiq (n=roger@64.201.13.50)
15:06.22Equinoxfryfrog, Yes I use my FXO/FXS both to 1. Power my phone (line 1) and 2. To route outgoing calls from asterisk if I'm feeling fancy (via sip)
15:06.32jake1932well 1 phone or group of phones with the same extension
15:06.35[TK]D-Fenderfryfrog : it also acts as a failover in case of a power failure.
15:06.49iCEBrkrWarning, flexibel rate not heavily tested!
15:06.50iCEBrkrhuh?
15:06.58Equinoxfryfrog, And if you're fancier than me you can put in a dialplan to control when you do FXS->FXO direct and when you hit asterisk
15:07.00fryfrogi see, so one would probably want an SPA 3000 near your point of entry
15:07.06jake1932<PROTECTED>
15:07.20Equinoxfryfrog, Such as 911 direct to the FXO port.
15:07.24[TK]D-FenderEquinox : Thats ugly though :)  Let * do its job :)
15:07.25iCEBrkrIt was in the middle of the ODBC function connection phase.
15:07.25fryfrogahhh
15:07.36[TK]D-FenderEquinox : ok, MAYBE for that reason :)
15:07.36*** join/#asterisk devel (n=devel@wiggum.digitalcoven.com)
15:07.41jake1932<PROTECTED>
15:07.45fryfrogare there any 4-8 port FXS?
15:07.47Equinox[TK]D-Fender, I just have a 2xline phone.. If I want a call out on the local 'fxo' I hit line 2.
15:07.56fryfroger, lemme read :)
15:07.58EquinoxIf I want VOIP I hit line 1.
15:08.00jake1932<PROTECTED>
15:08.10[TK]D-FenderEquinox : nifty idea if you had the phone already....
15:08.17EquinoxPhone was $40 or so
15:08.20[TK]D-Fendercool.
15:08.23EquinoxAnd I had it already :)
15:08.27iCEBrkrThat's weird tho showing up right when the ODBC DSN's go to connect
15:08.30EquinoxWas the only one with a headset they had.
15:08.41jake1932<PROTECTED>
15:08.45iCEBrkrmaybe
15:09.07*** join/#asterisk Seldon1975 (n=someone@CPE0013105d0913-CM0014e8b6162c.cpe.net.cable.rogers.com)
15:09.21jake1932<PROTECTED>
15:09.47iCEBrkrjake1932: I only noticed it just now when starting asterisk -cvvvvv
15:09.48*** join/#asterisk tux4pres (n=bsdvstux@24-155-81-66.ip.grandenetworks.net)
15:10.12tux4presCan anybody help me with an inbound routing schedule question?
15:10.26jake1932<PROTECTED>
15:10.44*** join/#asterisk olivier_ (n=olivier_@obs92-4-82-239-116-113.fbx.proxad.net)
15:10.58iCEBrkrjake1932: Not too worried about it. I won't be using it.
15:11.34*** join/#asterisk nesys (n=nesys@2001:1418:1a6:0:20d:93ff:fe28:3ef8) [NETSPLIT VICTIM]
15:11.56*** join/#asterisk ard (n=ard@2001:7b8:32d:0:20c:6eff:fe18:d11f) [NETSPLIT VICTIM]
15:12.19[TK]D-Fendertux4pres : just ask and we'll see what we can do
15:12.46tux4presI am using A@H 2.x...I wrote a bash AGI script that checks the current day/time in order for my [custom-schedule] context to determine whether we are open or not and play the appropriate message or route to a ring group....
15:13.02*** part/#asterisk oej (n=olle@apollo.webway.se)
15:13.07tux4presmy question is, how do I add this [custom-context] to all incoming calls without breaking the AMP GUI?
15:13.16jake1932you wrote an AGI to do that?
15:13.25jake1932i think there is a dialplan cmd
15:13.32tux4presGoToIfTime()
15:14.02tux4preshow do I add taht and bypass the AMP incoming calls section?
15:14.20fugitivotux4pres: don't use asterisk@home and you'll know how to do that
15:14.23iCEBrkrDon't use Asterisk*Home nor AMP :)
15:14.28fugitivolol
15:14.29iCEBrkrfugitivo: ^5
15:14.30fryfrogSo if one wanted to go with IP phones instead of ATA devices, what are decent options?  In theory, a "phone" could be in another location and configured to connect to asterisk correct?
15:14.34tux4preshe he...true true
15:14.48EquinoxWhat is asterisk*home anyway?  I always used asterisk
15:14.58fryfrogits a linux distro on cd with asterisk
15:14.59tux4presfugitivo,iCEBrker: I am using A@H for a friend who wants a GUI
15:15.03jake1932fryfrog: cisco 79xx, polycom makes some
15:15.03tux4presit works for them
15:15.18fryfroglike knoppmyth
15:15.20jake1932fryfrog: there is a 941 i've heard about
15:15.20tux4presI don't want to have to ngo in and change/add extensions for them all the time
15:15.29iCEBrkrCurrently, every GUI for Asterisk sucks is just plain WRONG
15:15.35Kattyi'm rich!
15:15.36iCEBrkrYou don't use a webpage for an operator panel
15:15.39[TK]D-Fenderfryfrog : Cisco = overpriced.  I suggest either SPA-941, or any Polycom.
15:15.43KattyiCEBrkr: i do.
15:15.45KattyiCEBrkr: it's flash.
15:15.46fugitivoasterisk@home is a try of doing a complex system into a userfriendly system, obviusly, without success, because people using asterisk@home, have more problems that people not using it
15:15.47robl^Equinox: A@H is A custom Linux distro with Asterisk, a bunh of add-ons, and a web interface to configure it.  Installs all in one go
15:15.48iCEBrkrKatty: You just turned a few tricks?
15:15.51fryfrogSPA-941, is that siphura?
15:15.52EquinoxI love my polycom IP500
15:15.55lo_techfryfrog, yes... a phone may be at a differerent location than the * server... but watch out for NAT, jitter, and rtt
15:15.57KattyiCEBrkr: it's FOP.
15:16.01KattyiCEBrkr: it's dreamy when it works.
15:16.04EquinoxKind of a learning curve(2 hours) for the first one.. But subsequent ones are quick.
15:16.04iCEBrkrKatty: FOP is gay
15:16.08KattyiCEBrkr: i like gay.
15:16.11fryfrogjitter and ftt?
15:16.11iCEBrkrKatty: oh yeah
15:16.12[TK]D-Fenderfryfrog : Linksys/Sipura, yes. (they got bought-out
15:16.12KattyiCEBrkr: i practically am gay
15:16.14iCEBrkroops
15:16.15iCEBrkrsorry
15:16.21iCEBrkrafk
15:16.36Kattysilly.
15:16.39tux4presiCEBrker: I under A@H sucks...I just need a way for them to be ablet to add/remove extensions withotu calling me..so basically, you are not aware of anyway of doing a custom from-pstn dialplan without letting the AMP GUI override it?
15:16.45EquinoxActually my IP500 is on it's way to Japan with a friend of mine. :)
15:16.57Kattyjapan scares me.
15:17.04EquinoxWhy?
15:17.08Kattywith it's insanely huge cities.
15:17.10Kattyand high techness.
15:17.16fugitivotux4pres: the mayority of us know nothing about amp or asterisk@home
15:17.21EquinoxYou're in a VoIP channel and high tech scares you?
15:17.28Kattyobviously.
15:17.35fryfroghaha
15:17.36tux4presfugitivo,iCEBrker: than you both, I apologize for the stupid question
15:17.47EquinoxThey have good sushi . .
15:17.49*** part/#asterisk brettnem (n=brettnem@72.29.102.158)
15:17.52EquinoxSushi scare you?
15:17.52[TK]D-Fendertux4pres : hardly stupid....
15:17.55Kattytux4pres: twisted[asteria] knows a bit about asterisk@home
15:17.59fugitivotux4pres: it's not a stupid question, maybe you're in the wrong channel, try #amportal
15:18.02Kattytux4pres: however, he hates it ;)
15:18.10tux4preshe he
15:18.12robl^japan is the only place that  has smart chips and artificial inteligence built into the toilet issue :)
15:18.17KattyEquinox: i don't eat sushi
15:18.20tux4presyeah, I tried the ampportal channel but no one will respond...he he
15:18.28*** join/#asterisk brettnem (n=brettnem@72.29.102.158)
15:18.30EquinoxBut does it scare you?
15:18.52KattyEquinox: it's not vegan.
15:18.52[TK]D-Fenderrobl^ : Gives new meaning to "shit for brains"
15:18.52KattyEquinox: sushi makes me sad.
15:18.52brettnemhey guys
15:18.52jake1932no question - you'll always get a response here when asking about @home or AMP
15:19.02EquinoxKatty- Sushi literally translated means 'vinegar rice'.  So it can be vegan.
15:19.13EquinoxKatty- As I'm pretty sure rice, and vinegar are vegan friendly.
15:19.19KattyEquinox: they are.
15:19.21[TK]D-FenderEquinox : indeed.  I just made 5 rolls last night for my company pot-luck lunch :)
15:19.28iCEBrkrtux4pres: When I get off my lazy butt and finish my Asterisk control panel.... No one will have to worry about stupid webpages and flash interfaces
15:19.28KattyEquinox: every sushi i've seen down here has raw fish in it.
15:19.29fugitivotux4pres: well, i don't know about amp, but if it's a gui, and doesn't have the option you need, you'll need to modify the gui
15:19.32fryfrogKnow the diff between SPA-841 and 941?
15:19.33Equinox[TK]D-Fender, Cool :)  It's hard to make them pretty.
15:19.35fryfrogabout 100? :)
15:19.38robl^[TK]D-Fender: *beep* Don't forget to wipe front to back.  *beep* remember to wash your hands..  *beep* you have used 4 squares thank you!
15:19.39jake1932http://voipspeak.net/index.php?option=com_content&task=view&id=41&Itemid=27
15:19.44[TK]D-FenderEquinox : Nah!
15:19.46EquinoxKatty- I doubt it.  Many have cooked fish but people assume it's raw.
15:19.46fugitivoiCEBrkr: i only use web applications
15:19.51Seldon1975D-Fender are you by any chance an Australian?
15:19.56jake1932fryrog - for you
15:19.59EquinoxKatty- The vegan stuff is less common but every restaurant I go to has it.
15:20.07tux4presiCEBrker: that is cool you are writing an interface.   what ar eyou writing it in?
15:20.27iCEBrkrfugitivo: I 'guess' it could be a webpage.. But webpages are limited.
15:20.35brettnemhey if I change rxgain and txgain settings in zapata.conf.. what do I need to do to reload those settings? Reload the kernel module, chan_zap, ?? any ideas? help
15:20.41iCEBrkrBut for 'professional grade' stuff. Web apps are retarded.
15:20.46robl^sushi is NOT raw fish..  :)  sushi has to do with the rice and being rolled up :)
15:20.53iCEBrkrSince everyone wants a webpage that acts like a windows app
15:20.53fugitivoiCEBrkr: ajax is great for that "limited" stuff
15:20.58Equinoxrobl- And vinegar in the rice. ;)
15:21.01EquinoxJust a bit.
15:21.04Seldon1975Sashimi is raw fish. mmmmmmmmm
15:21.05fugitivobrettnem: restart
15:21.08EquinoxYep.
15:21.09iCEBrkrfugitivo: Why bother with a hack when you can just do it the right way??
15:21.13brettnemfugitivo: restart what?
15:21.25brettnemfugitivo: so I don't need to reload the kernel module? just chan_zap?
15:21.33fugitivoiCEBrkr: why ajax is a hack?
15:21.40iCEBrkrfugitivo: I've been doing this web application dev for the better part of 8yrs.  and it just sucks. Everyone wants their web pages to act, look and feel like a windows app, so why not just make it a windows app to being with?!
15:21.47*** join/#asterisk thome (i=tm@mebes.info)
15:21.49thomehello
15:21.55tux4presiCEBrkr: I wanted to write an asterisk control app with the Wi.Ser framework....with that middleware you could deploy it as a fat client or a swing thin client without any code modification
15:22.00fryfrogI assume pretty low end hardware is a-okay?  Something in the 500mhz, 256mb, 10-50G HD range?  Asterisk only?
15:22.01iCEBrkrfugitivo: Because AJAX is a hack.
15:22.02thomei have a error:Dec 22 16:20:02 ERROR[19096]: chan_misdn.c:3467 load_module: Unable to initialize mISDN
15:22.05thomeDec 22 16:20:02 WARNING[19096]: loader.c:414 __load_resource: chan_misdn.so: load_module failed, returning -1
15:22.08thomeDec 22 16:20:02 WARNING[19096]: loader.c:499 load_modules: Loading module chan_misdn.so failed!
15:22.11thomeany ideas for me?
15:22.15fugitivoiCEBrkr: a hack with great success :)
15:22.18jake1932fryfrog:  yes
15:22.21Equinoxfryfrog, My understanding is only transcoding will boost your hardware requirements.
15:22.23iCEBrkrfugitivo: It only took 4yrs
15:22.35fugitivoiCEBrkr: because also microsoft is moving to web apps, look at the new office
15:22.38iCEBrkrXMLHTTP was an IE only thing way back in the day
15:22.38jake1932fryfrog: for a couple of phones, you should be fine
15:22.43fryfrogfavorite linux distro to use with it?
15:22.48Kattyuh oh
15:22.50EquinoxI like debian.
15:22.52fugitivofryfrog: the one you like most
15:22.54EquinoxAnd ubuntu
15:22.55fryfrogi'd probably go with gentoo, but i'm flexible
15:22.55iCEBrkrfugitivo: Becuase M$ is doing it?? LOL peer pressure
15:22.58EquinoxActually I like ubuntu more now.
15:23.07Seldon1975iCebreaker: "so why not just make it a windows app to being with?!" people dont want to install stuff if they can avoid it
15:23.23Seldon1975for one
15:23.28EquinoxiCEBreaker: 3 letters: DLL
15:23.32backbluei need zapata module for using one x100p clone?
15:23.33iCEBrkrSeldon1975: Umm. Stay in context
15:23.40iCEBrkrEquinox: .NET doesn't care about that crap anymore.
15:23.43fugitivoiCEBrkr: the true is that in a near future, we'll all be using diskless terminals with web apps, maybe not at home, but yes at companies
15:23.49fryfrogdebian sounds like a good idea, are there any X requirements for *?
15:23.52iCEBrkrSeldon1975: I'm talking about integrating apps with Asterisk
15:23.56fugitivofryfrog: no X
15:24.03tux4presSeldon1975: I agree with iCEBrkr on the web-only revolution..to minimize install just make it easier...take a look at WebStart's jnlps for example
15:24.03fugitivofryfrog: don't install X in your asterisk machine
15:24.04EquinoxiCEBrkr - How well does .NET run under linux?(mono)
15:24.04iCEBrkrfugitivo: ain't happening
15:24.16iCEBrkrEquinox: Who cares about Linux
15:24.19Seldon1975iCE: I dont see my comment as out of context
15:24.20fugitivoiCEBrkr: some companies are using that schema right now
15:24.21EquinoxMe.
15:24.24iCEBrkrEquinox: I can't really say that, since I'm a linux junky
15:24.25EquinoxI do server side stuff a lot.
15:24.29fryfrogand debian has debs?  stable, unstable or testing of deb?
15:24.42jake1932<PROTECTED>
15:24.44EquinoxAnd one of the devs I work with does .NET stuff I'm wondering about it running on the linux server
15:24.56fugitivofryfrog: don't use packages, use the sources and compile :)
15:24.58iCEBrkrfugitivo: are you saying that the PBX your company is using ( if it's not Asterisk ) doens't have a DOS or Windows frontend?
15:24.59jake1932<PROTECTED>
15:25.04*** join/#asterisk marv[work] (n=timr@64.89.118.139)
15:25.07KattyGot SIP response 500 "Internal Server Error" back from 192.etc
15:25.11fryfrogfugitivo: humm, sounds like a good reason to use gentoo then ;)
15:25.12Katty^- what's that mean?
15:25.12jake1932<PROTECTED>
15:25.23brettnemso zapata.conf changes require a chan_zap.so reload?
15:25.25fugitivofryfrog: gentoo is good
15:25.30jake1932Katty: look in the server logs
15:25.31fugitivofryfrog: or lfs ;)
15:25.35ManxPower~docs
15:25.36jbotfrom memory, docs is probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html
15:25.36jake1932Katty it's generic
15:25.38Kattyjake1932: that takes too much effort
15:25.38fugitivoiCEBrkr: web interface
15:25.39ManxPower~mailinglist
15:25.41jbotit has been said that mailinglist is Search Asterisk mailing lists by prepending site:lists.digium.com to your Google search.  Browse the mailing list archive at http://lists.digium.com/ or search through it at http://www.asteriskguru.com/archives/search.php, or and there is also the Macintosh Asterisk mailing list at http://www.astmasters.net/maml.html
15:25.41Kattyjake1932: k
15:25.48iCEBrkrfugitivo: Sure it does.
15:25.56Beirdo~beer
15:25.58jbotextra, extra, read all about it, beer is ummm, ummm good!, or good for you!... not just for breakfast anymore
15:26.02iCEBrkrHell, most PBX's still run OS/2
15:26.05tzafrir_laptopbrettnem, actually they may require a total restart of Asterisk. Maybe a reload will do on 1.2, I'm not sure
15:26.09fryfroghttp://fryfrog.com/wordpress/v/Miscellaneous/DespairLinux/
15:26.10ManxPowerKatty, Using Polycoms?  They give that error, but seem to work gine.
15:26.10fugitivoiCEBrkr: i don't like the idea of intalling software, that's all
15:26.14fryfrogsomeone should make one for LFS :)
15:26.16iCEBrkrfugitivo: Me either.
15:26.18KattyManxPower: (=
15:26.33fugitivofryfrog: i did
15:26.34iCEBrkrfugitivo: But if it's a 'net appliance' such as an Asterisk box for a PBX.  What's to install??
15:26.45fryfrogshow me!
15:26.54iCEBrkrThere's a much bigger picture here with Asterisk.  I'm not quite sure why everyone doesn't think bg.
15:26.54fugitivofryfrog: not ready for the public yet :)
15:26.57iCEBrkrerr big
15:26.57Kattyupdate 26 of 43!
15:26.58fryfrogbah!
15:27.02ManxPowerKatty, We have seen NO problems.   I suspect it's some issue where when the call ends Asterisk sends something to the phone after the user has hung up
15:27.06fryfrogi <3 the gentoo one :)
15:27.18KattyiCEBrkr: get off.
15:27.36ManxPoweri.e. The user hangs up the phone, then the far end hangs up, asterisk sends a hangup message to the phone and the phone complains because there is no active call
15:27.37iCEBrkrKatty: Ummm, you're not 'get off' material thanks.
15:27.48jake1932oooh
15:27.57Beirdoheheh
15:28.01fugitivoiCEBrkr: i still like the idea of webapps more and more
15:28.14iCEBrkrfugitivo: I guess it really depends on what you're trying to do.
15:28.23lo_techbrettnem: for changes to take effect in 1.2, we have to do a 'service asterisk stop; service zaptel restart; service asterisk start'
15:28.25KattyiCEBrkr: as much as i appreciate that comment, i hope it's not an insult ;)
15:28.50iCEBrkrKatty: I just throw'm out there.. YOu do what you want with them.
15:28.56KattyiCEBrkr: kthx.
15:29.00saftsackFile digits/1F does not exist in any format
15:29.00brettnemlo_tech: That would suggest the kernal module needs a restart.. argh.. just trying to figure out how it is.. don't want to drop all my T1s..
15:29.04iCEBrkr:)
15:29.07saftsackdoes anyone know this errormessage?
15:29.10Kattytwisted[asteria]: mew?
15:29.22fugitivobrettnem: i told you, you need to restart
15:29.31*** join/#asterisk TheCops (n=henri@206-248-136-187.dsl.teksavvy.com)
15:29.35ManxPowerWHAT?????????
15:29.39fugitivosaftsack: you don't have the file
15:29.41Beirdohappy birthday to me...  and for my birthday, I want this headache to fuck right off
15:29.50*** join/#asterisk __a (i=user@85.100.123.10)
15:29.55ManxPowerIn 1.2 all you have to do is reload chan_zap.so to apply MOST changes to /etc/asterisk/zaptel.conf
15:29.58lo_techbrettnem: that's affirm... cant change the configs for the T1 interfaces and have it take effect without mod'ing the devices
15:30.04Beirdoheh
15:30.04saftsackfugitivo, i know it ;)
15:30.10__aguys, is the jitterbuffer implementation in asterisk is adaptive?
15:30.13saftsackfugitivo, do you know a package for this?
15:30.13ManxPowerif you are fiddleing with /etc/zapata.conf then run ztcfg, but that will drop your calls
15:30.16fugitivoManxPower: zaptel.conf or zapata.conf?
15:30.18BeirdoI took some excedrin extra strength already, iCEBrkr
15:30.20Beirdo:)
15:30.21saftsackor an adress where i can download this file?
15:30.25fugitivosaftsack: so?
15:30.28iCEBrkrline 0: Unable to open master device '/dev/zap/ctl'
15:30.31__ai.e. if I set jitterbuffer=10, does it expect the jitter to be constantly around 200ms?
15:30.36fryfrogKRIKEY!
15:30.37iCEBrkrWTF did I change?!
15:30.39saftsackso that i can get the file ,)
15:30.42KattyBeirdo: hippo birdie two ewe
15:30.43ManxPowerfugitivo, I don't remember, for me it's /etc/asterisk/zap<tab>
15:30.45backbluewtf its zapata? where is that package?
15:30.46KattyBeirdo: hippo birdie two ewe
15:30.50KattyBeirdo: hippo birdie deer ewe
15:30.50backblue?? zapata
15:30.52fugitivoManxPower: :)
15:30.55KattyBeirdo: hippo birdie two ewe
15:30.57Beirdohehe ;)
15:31.03iCEBrkrKatty: You're stoned.
15:31.04Beirdothank ye, Katty :)
15:31.10__aguys
15:31.12__ajitterbuffer
15:31.14__aadaptive
15:31.16iCEBrkrlol
15:31.17KattyiCEBrkr: amn't.
15:31.17__ais it?
15:31.20KattyiCEBrkr: i'm a straightedge.
15:31.29ManxPower__a, If you read the sample config files you would knowl.
15:31.31ManxPowerknow
15:31.35iCEBrkrKatty: Straightedge, vegan.. THAT explains a lot.
15:31.41Katty^_^
15:31.54fugitivoi like vegan people
15:31.57iCEBrkrKatty: no offense, but your kinda people-- they're always COO-COO
15:32.01Beirdohey, be nice to the brave woman who will hang out with geeks
15:32.08KattyBeirdo: i am a geek.
15:32.14Beirdomy mother-in-law-to-be is vegan
15:32.17Beirdoyou be nice :)
15:32.17fugitivoKatty: are you a woman?
15:32.24Kattyfugitivo: hahahahaha.
15:32.27__aManxPower: there's just this
15:32.28__a; Configure jitter buffers in zapata (each one is 20ms, default is 4)
15:32.28__a;
15:32.28__a;jitterbuffers=4
15:32.35*** join/#asterisk bweschke (n=bweschke@pcp09754274pcs.narlington.nj.comcast.net)
15:32.35BeirdoKatty: I would hope you're a geek if you are gonna put up with us :)
15:32.40fugitivoKatty: no?
15:32.43KattyBeirdo: geeks are teh bestest.
15:32.44mutilatoranyone know if there is a way to set the tcp TIME_WAIT time to clear a socket faster in windows?
15:32.45iCEBrkrA few years ago.  There were about 30 of us who went camping for a long weekend.  I spent those days with about 8 vegans.. They're all LOONEY
15:32.47__aManxPower: I really can't figure it from that :(
15:32.48Kattyfugitivo: i'm quite female.
15:32.57fryfrogIRC:  where the MEN are MEN... and so are most of the WOMEN... also, the 13 year old girls are FBI agents...
15:33.00BeirdoI agree.  Geeks rule.
15:33.04ManxPower__a, Wrong sample config file.  that's for Zap.  Zap does not have an adaptive jitterbuffer because it doesn't need one.
15:33.05mutilatorand/or CLOSE_WAIT
15:33.08Beirdoof course, I'm biased
15:33.10iCEBrkrfryfrog: heheh
15:33.10coppicethere are no jitter buffers in zapata
15:33.11ManxPowerI assumed you meant for IAX
15:33.24fugitivoKatty: cool, and you know linux?
15:33.31Kattyfugitivo: ...
15:33.37Kattyfugitivo: no, i'm just here for the free torment.
15:33.38__aManxPower: what if I dial SIP via Zap?
15:33.41ManxPowercoppice, Well, there is a 4ms "jitterbuffer" but nobody ever fiddles with it.
15:33.54fugitivoKatty: how many programming languages do you know?
15:33.56Kattyfugitivo: next you'll be asking me if i even run asterisk, right?
15:34.04ManxPower__a, The VOIP part is what will have the jitter, not the zap part.
15:34.05fugitivoKatty: i know you run asterisk
15:34.08fryfrogstupid question:  what is zaptel in relation to asterisk?
15:34.10iCEBrkr"How to flirt with a geek girl" by fugitivo
15:34.13BeirdoKatty: just put on some nice pointy boots and start kicking
15:34.17ManxPowerIf your zap part has jitter you're REALLY SCREWED.
15:34.19KattyiCEBrkr: he sucks at it.
15:34.19Beirdoit will do wonders
15:34.21fugitivoiCEBrkr: nah, just curious
15:34.24iCEBrkrKatty: LOL
15:34.28KattyBeirdo: butbut, but
15:34.36KattyBeirdo: stompy boots are so much more better.
15:34.39Beirdooooh
15:34.40__acoppice: it actually helped me today, setting it to 10 made a noticable difference on a link with 200ms delay
15:34.40iCEBrkrKatty: I figured you'd get all misty when he asked how many programming lanuages you knew
15:34.47BeirdoOK, that will do too.
15:34.52fugitivoiCEBrkr: lol
15:34.52KattyiCEBrkr: more like annoyed.
15:34.55iCEBrkrhaha
15:34.57Beirdoand are more comfortable too.
15:35.05Kattylolzruaprogrammer????!!!!!!!1oneone
15:35.05fugitivoKatty: you can't code?
15:35.17lo_techmutilator: yes via reghack.. at least there is for win NT4.0+... ymmv for win9x/ME... found it about a year ago on goole
15:35.17ManxPower__a, Asterisk does not have a jitterbuffer for SIP, only for IAX2
15:35.37iCEBrkr"Hey baby, what's your sign" he asked.
15:35.38ManxPowerRemember jitter buffering happens on the INCOMING VoIP stream
15:35.50iCEBrkrShe replied "Octagonal, like a STOP sign, now beat it"
15:35.56ManxPowerSo Asterisk -> SIP device : the SIP device does the jitter buffering.
15:36.04KattyiCEBrkr: teehee!
15:36.18ManxPowerSIP device -> Asterisk : Asterisk would do the jitter buffering if Asterisk actually had a SIP jitter buffer.
15:36.23saftsackhowto upgrade asterisk?
15:36.23__aManxPower: so when an incoming VoIP stream is being passed to a zap channel, setting a JB for zap makes difference.
15:36.33saftsacki have 1.0.10 compiled from source on my computer
15:36.45ManxPower__a, no, it would not make any differece
15:36.46coppice__a: there is no jitter buffer for zap
15:37.06fugitivoiCEBrkr: i just wanted to know the geekness of Katty :)
15:37.14ManxPower__a, Zap jitter buffer would buffer jitter on the ZAP part and you should NEVER EVER get jitter on a zap port.
15:37.27__acoppice/ManxPower: it just did, believe me.
15:37.32iCEBrkrfugitivo: She's a pseudo-geek. Can't ya tell?
15:37.33__awhat's the point of the above setting then?
15:37.41*** join/#asterisk Chuji (i=Chuji@pcp09180602pcs.nash01.tn.comcast.net)
15:37.54ManxPower__a, The point is if your telco has really bad equipment that causes jitter on your T-1 line.
15:38.02*** join/#asterisk sivana (n=sivana@mixdown.ca)
15:38.12ManxPowerAnd I mean REALLY REALLY bad, like 1960's type equipment.
15:38.28iCEBrkrIt still clicks?
15:38.30lo_techhey! im 1960's issue equipment!
15:38.36coppiceManxPower: you *cannot* have jitter on a T1, no matter how bad it is
15:38.41__a:)
15:38.44iCEBrkrlo_tech: You have jitter?
15:38.55ManxPowercoppice, what's the jitter setting in zap for then?
15:39.03Dandanlo_tech: /nick hi_tech, check if that helps
15:39.03lo_techiCEBrkr: does jiggy count?
15:39.07iCEBrkrlo_tech: lol
15:39.20__amy theory is that Zap just buffers whatever is incoming, whether it's from telco or VoIP
15:39.35ManxPower__a, you are wrong.
15:39.39*** join/#asterisk nfi|ermes (n=nfi_erme@217.220.121.62)
15:39.43__atherefore increasing it for a 200ms delay line made a difference
15:39.43nfi|ermeshttp://lists.digium.com/pipermail/asterisk-dev/2005-June/013602.html
15:39.46saftsackis it necessary to update libpri with asterisk?
15:39.59nfi|ermesi have autofallthrough problem
15:40.05ManxPower__a, You are wrong.  I'm not going to argue with you.
15:40.12nfi|ermesis there any knowkn issue ?
15:40.22coppiceManxPower: nope. he's right. zap (the driver, not chan_zap) just buffers whatever passes through
15:40.28jake1932nfi|ermes: can you be more specific?
15:40.35__aManxPower: not according to sources.
15:40.36ManxPowercoppice, in EITHER direction?
15:40.42nfi|ermesmy dialplan never goes to t (timout)
15:40.51nfi|ermesit gives autofallthrough
15:41.08jake1932nfi|ermes: what version?
15:41.20nfi|ermesCVS
15:41.22saftsackcan i install the new asterisk over the old one?
15:41.23nfi|ermes<PROTECTED>
15:41.23nfi|ermes<PROTECTED>
15:41.23nfi|ermes<PROTECTED>
15:41.23nfi|ermes<PROTECTED>
15:41.23nfi|ermes<PROTECTED>
15:41.26jake1932wooo
15:41.30ManxPowernfi|ermes, exten => t is only for IVR apps that use playback, background, etc.
15:41.37[wiebel]anyone tried to compile zapata 1.2.1 under freebsd?
15:41.40saftsackjake1932, are you experienced with updating asterisk?
15:41.41[wiebel]and sucseeded
15:41.42fugitivo"Procaccia miseria"
15:41.55ManxPowernfi|ermes, turn off autofall thru
15:41.58*** join/#asterisk CoffeeIV_ (n=CoffeeIV@mail.airlinksystems.com)
15:41.58jake1932saftsack: somewhat
15:42.00[wiebel]s/zapata/zaptel/
15:42.00coppiceI dunno why they changed the name of the buffer control in chan_zap to jitterbuffers. it doesn't do jitter buffering at all. that just selects the number of 20ms chunks of buffering are in the zaptel kernel driver.
15:42.07saftsackcan i install over the old versions?
15:42.20nfi|ermes<PROTECTED>
15:42.28jake1932saftsack: d/l the source - make clean - make - make install
15:42.34saftsackok
15:42.37*** join/#asterisk Damin (n=damin@nucleus.nacs.net)
15:42.43saftsackshould i save the configs before?
15:42.53fugitivosaftsack: yes, just remove your /usr/lib/asterisk/modules before make install
15:42.57ManxPowerUgh.  I really hate being 1000ms from the nearest server
15:43.07backbluehow do i see this from my card, to know if i need to do this patch { 0xe159, 0x0001, 0x8085, PCI_ANY_ID, 0, 0, (unsigned long) &wcx101p },
15:43.09saftsackfugitivo, ok
15:43.10backblue?
15:43.13fugitivosaftsack: backup your config files if you'll do make samples
15:43.13ManxPowerfugitivo, unless he has custom modules
15:43.22saftsackok
15:43.27saftsackyes i did a backup
15:43.29saftsackfor the worst case
15:43.34*** join/#asterisk trixter (n=trixter@65.172.209.246)
15:44.10*** join/#asterisk eKo1 (n=bernd@metrored-gw.tropicohn.com)
15:44.15fugitivoManxPower: he'll need to build the new custom modules against the new asterisk, or not?
15:44.28ManxPowerfugitivo, You mean like G729?
15:44.40backbluejake1932: can you help me, how do i see that? with cat /proc/pci i have 0xe159 and 0x0001, but how do i see the 0x8085 field in my card?
15:44.43ManxPowerfugitivo, and that would depend on the module, of course.
15:44.56jake1932backblue: i don't have a X100p
15:45.44ManxPowerbackblue, The wcfxo kernel module does not recognize your X100P clone?
15:46.19eKo1i have two asterisk boxen: A and B. B is connected to the PSTN via a PRI. A passes calls destined to the PSTN to B and B places the call. I'm noticing though that, in the CDRs, the dial-status for attempted calls is always ANSWERED. Is is this normal?
15:46.36backblueManxPower: i dont know, in manual from voip-info, they speek about patchs and not speak about any wcfxo.
15:46.43nfi|ermesManxPower, have a loook to my context
15:46.48nfi|ermeshttp://pastebin.com/475153
15:46.49ManxPowereKo1, only if the PSTN lines are analog
15:46.50backbluejake1932: ok
15:46.57nfi|ermesit should go to timout extension
15:47.03nfi|ermesi use background
15:47.04ManxPowerbackblue, ignore that unless you have a problem
15:47.13eKo1ManxPower: well, B has a PRI
15:47.26eKo1so it shouldn't be that way then
15:47.30saftsackjake1932, so i hope it will be work without problems :)
15:47.35TheCopsdoes asterisk have a bug with enum and e164.org ?
15:47.43ManxPowereKo1, the CDRs on A or B?
15:47.47eKo1on both
15:47.50*** part/#asterisk __a (i=user@85.100.123.10)
15:47.58TheCopsI've got parse_naptr: NAPTR Regex match failed like error.
15:48.08eKo1it's so weird
15:48.17ManxPowereKo1, weird.  do you have any playbacks, backgrounds, or answers that the calls might hit?
15:48.34lo_techeKo1: or if box A has exten => _X.,1,Answer / exten => _X.,n,Dial(BoxB/XXX), etc... if you answer the call on A then dial through to B, you're going to see an answer on each
15:48.35ManxPowerTheCops, Your need to prefix a + on your ENUM lookups
15:48.39backblueManxPower: weird.
15:49.07CoffeeIV_I'm scriptifying the sending of a fax though * . . . so I can do it from a web page, I'm not a spammer . . . I am making a context for outgoing faxes, and a call file that will set a variable to the outgoing tiff file before calling -- anyone know of any examples of something similar, for eitther the dialplan or call file ?  I search voip-info and google
15:49.17eKo1ManxPower: nope, I don't have an Answer() anywhere
15:49.20TheCopsManxPower, so, +4503713039 (eg)
15:49.22ManxPowerLooks like I am NOT going to Lafayette next week.  YAY!
15:49.29ManxPowereKo1, many asterisk apps answer the line.
15:49.35ManxPowerTheCops, yes
15:49.42*** join/#asterisk duckz (n=duckz@193.192.46.26)
15:49.58backblueManxPower: they speek about wsfxo.c, i dont even have that file in the sources.
15:50.14ManxPowerbackblue, you need the zaptel sources
15:51.02backblueManxPower: i have it
15:51.03ManxPowereKo1, you are not doing something stupid like using busydetect or callprogress set to yes, are you?
15:51.04*** join/#asterisk popvoxdave (n=popvoxda@69.143.206.112)
15:51.06backbluei have compiled it
15:51.09*** part/#asterisk popvoxdave (n=popvoxda@69.143.206.112)
15:51.09backblueand it compiled ok
15:51.10TheCopsManxPower, hrmm, weird, I added + and when I did again the lookup, I've got the same error, and the + is not in the lookup (via the console)
15:51.18backbluei dont know if it will detect my card.
15:51.27ManxPowerTheCops, it should be
15:51.43*** join/#asterisk Lurr (i=user@adsl-067-034-122-207.sip.mia.bellsouth.net)
15:51.43ManxPowerbackblue, almost nobody has to patch zaptel anymore
15:51.47eKo1ManxPower: nope
15:51.48eKo1nothingç
15:51.50ManxPowerTheCops, you did a reload?
15:51.52*** part/#asterisk Lurr (i=user@adsl-067-034-122-207.sip.mia.bellsouth.net)
15:51.56eKo1just a straight Dial()
15:52.06TheCopsyeah
15:52.07TheCopsrestarted
15:52.14backblueManxPower: ok, tks i will try it.
15:52.18ManxPowereKo1, paste the console output to pastebin.ca for the machine that the phone device is connected to.
15:52.45eKo1ManxPower: ok
15:53.02backblueManxPower: i have to patch asterisk, because asterisk does not assume logins with ",
15:53.23ManxPowerbackblue, You don't know enough to patch asterisk yet.
15:53.39backblueManxPower: but i allready patched.
15:53.40*** join/#asterisk brettnem (n=brettnem@72.29.102.158)
15:53.47*** part/#asterisk brettnem (n=brettnem@72.29.102.158)
15:53.48backblueManxPower: and i need it.
15:53.50*** join/#asterisk Gronker__ (n=Gronker2@70.152.186.67)
15:53.54*** part/#asterisk Gronker__ (n=Gronker2@70.152.186.67)
15:54.04backblueManxPower: asterisk takes "." from the cids, and i need it.
15:54.14TheCopsManxPower, Dec 22 10:52:49 WARNING[8164]: app_enumlookup.c:99 enumlookup_exec: The application EnumLookup is deprecated.  Please use the ENUMLOOKUP() function instead.
15:54.15ManxPowerbackblue, if you say so.
15:54.20TheCopsWeird, this is the same command
15:54.24ManxPowerTheCops, ignore that for now.
15:54.32TheCopsok
15:54.57TheCopsManxPower, + before the numer dont work at all, very weird...hehe
15:55.19ManxPowerTheCops, It's required for anything 1.2RC2 or later
15:55.21backblueManxPower: do you use "." in cids?
15:55.32TheCopsManxPower, I have 1.2.1 isntalled
15:55.47ManxPowerbackblue, a period.  No, but I don't see why it would be a problem.  I use commas in CID all the time.
15:55.48backblueits a big problem in my point of view, asterisk dont suport virtual hosts, and things like that.
15:56.05backblueManxPower: commas? give me a example please.
15:56.07ManxPowerbackblue, it does something similar, called CONTEXTS
15:56.12ManxPowerWieling, Eric
15:56.15ManxPowerThere's an example
15:56.52ManxPowerUsing contexts you can have different companies (customers) use the same extensions and not have a problem.
15:57.15ManxPowerUnless you are doing something stupid like setting your SIP usernames to be the same as the extension for that device.
15:57.24ManxPowerThat's just ASKING for trouble.
15:57.46TheCopsManxPower, duh! I'm a stupid guys who have more server that I can remember. I was working in another server with similar asterisk installation lol
15:57.50*** join/#asterisk wunderkin (i=kev@ip68-226-113-228.ph.ph.cox.net)
15:58.09TheCopsThis is working now, hehe
15:58.09ManxPowerTheCops, does it work now?
15:58.12TheCopsyeah for sure
15:58.13TheCopsthanks
16:00.01backblueManxPower: so, you have 2 companys, xpto.com and xpte.com, and you have your asterisk getting calls form this 2 companys, and you have in company xpto.com the user john, ans in xpte.com the user john, how do you have them both?
16:00.03eKo1ManxPower: http://pastebin.ca/34612
16:00.32eKo1ManxPower: It seems the call is being answered...but why and where?
16:00.40*** join/#asterisk shido6 (n=shido6@d221-68-216.commercial.cgocable.net)
16:00.47ManxPowerbackblue, you don't.  Since the user will never actually SEE their SIP username, set it to something unique, like the MAC address of the Voip device.
16:01.02ManxPowerYou have extensions.
16:01.07ManxPowerusers see extensions
16:01.31ManxPoweror you can name the SIP username john-company-a and john-company-b
16:01.39shido6...
16:02.15ManxPowereKo1, what is this SIP/pstn thing?
16:02.23backblueManxPower: with mac addr how do you pretend to make calls?
16:02.25*** join/#asterisk xachen (i=justin@magnum.thisgeek.com)
16:02.27steffsome trouble with avm C2 in p2p mode, anyone can help me?
16:02.30saftsackhowto print a message on the telephones display? (misdn)
16:02.37backbluecall "ff:ff:ff:ff:.." ?
16:02.56eKo1SIP/pstn is box B (with the PRI)
16:03.01ManxPowerbackblue, in sip.conf make the userid for the device be the MAC address like deadbeef then Dial(SIP/deadbeef)
16:03.15jake1932deadbeef?
16:03.15Beirdommmm, shortbread :)
16:03.24ManxPowereKo1, Are you one of those lunatics that want to use SIP instead of IAX2 for inter asterisk communicatuions
16:03.33eKo1hehe, yeah
16:03.50*** join/#asterisk pjz (n=pj@zachs.place.org)
16:03.57lo_techdecafbad > deadbeef
16:04.00pjzhow do I set up call logging in asterisk?
16:04.04eKo1since i tried with iax2 and it didn't work at all for me
16:04.07Beirdo00coffee
16:04.11iCEBrkrpjz: CDR
16:04.11ManxPowereKo1, now pastebin the CLI output from server B
16:04.18nfi|ermesanyone has never seen this ??:
16:04.19nfi|ermes<PROTECTED>
16:04.19nfi|ermesDec 22 16:53:07 WARNING[2136]: chan_sip.c:1594 create_addr: No such host: 39-81
16:04.19nfi|ermesDec 22 16:53:07 NOTICE[2136]: app_dial.c:983 dial_exec_full: Unable to create ch
16:04.19nfi|ermesannel of type 'SIP' (cause 3)
16:04.21fugitivoi love "switch"
16:04.22iCEBrkrpjz: /var/log/asterisk/
16:04.37fugitivonfi|ermes: what the hell is 39-81?
16:04.47nfi|ermestwo extensions
16:04.52pjziCEBrkr: ah! thanks!
16:04.58lo_techpjz: by call logging are you talking call detail records, or the English mongrelization of the the term 'Call Recording"?
16:04.58nfi|ermesa group
16:04.59file[desk]umm, you want SIP/39&SIP/81
16:05.00fugitivothen you need to use SIP/39&SIP/81
16:05.01ManxPowernfi|ermes, you do not have a [39-81] section in sip.conf
16:05.20nfi|ermesno
16:06.00ManxPowernfi|ermes, you want to ring two devices at the same time?
16:06.03eKo1ManxPower: http://pastebin.ca/34613
16:06.11nfi|ermesyes
16:06.16fugitivonfi|ermes: read
16:06.27ManxPowerthen you Dial(SIP/device1&SIP/device2)
16:06.27rob039-81 is the negative of life, the universe, and everything
16:06.35nfi|ermesexten => s,2,Macro(rg-operatore,,,39-81)
16:06.47iCEBrkr<PROTECTED>
16:06.49iCEBrkrNOW WHAT?!
16:06.50iCEBrkrgrrrrrr
16:07.00jake1932comprimise
16:07.00ManxPoweriCEBrkr, one side has to be pri_net
16:07.09fugitivonfi|ermes: <file[desk]> umm, you want SIP/39&SIP/81  && <fugitivo> then you need to use SIP/39&SIP/81
16:07.17jake1932comPRImise
16:07.28iCEBrkrManxPower: if I set it to pri_net it says the samething but only claiming they think they're network too
16:07.29ManxPowernfi|ermes, We don't care about your macro.  If you want to dial multiple devices than you use the corect format or it will not work.
16:07.43ManxPoweriCEBrkr, Your telco has a loopback on your PRI
16:08.04ManxPower<PROTECTED>
16:08.11iCEBrkrThey better not
16:08.25iCEBrkr<PROTECTED>
16:08.40ManxPoweriCEBrkr, they do,  A loopback is the onlything that would cause that message that I can think of.  It could be a telco loopback or a loopback cable
16:08.48*** part/#asterisk pjz (n=pj@zachs.place.org)
16:08.54file[desk]iCEBrkr: commence stabbing telco
16:09.11iCEBrkrdual-wield!
16:09.14zoaiCEBrkr: check zapata.conf vs zaptel.conf
16:09.24iCEBrkreh?
16:09.26iCEBrkrhow's that matter?
16:09.47*** join/#asterisk azzie_ (n=az@66.193.84.130)
16:10.23zoaif you define net in one and cpe on the other, it wont be good
16:10.30zoaand you will get those errors
16:10.31zoaiirc
16:10.40ManxPoweriCEBrkr, Asterisk is sending out "Hi!  I'm PRI CPE" then that message is being sent back to Asterisk by something on the line.
16:10.52iCEBrkrWhere in /etc/zaptel.conf do you specifiy cpe or net? :D
16:10.56Seldon1975I have a TDM2422E with 8 FXS lines and 8 FXO lines on a network with 8 SIP phones.  I have near-side echo on the SIP phones only.  Can anyone suggest what might be causing this?
16:10.58iCEBrkrzoa: toker
16:11.01ManxPowerzoa, um NET and CPE s only done in /etc/asterisk/zapata.conf
16:11.07iCEBrkrManxPower: Thank you
16:11.15zoawell, i dont remember that the cards could see it
16:11.18iCEBrkrlol
16:11.18zoathats why i ask
16:11.19zoa:)
16:11.26zoai mean physically on the line
16:11.35ManxPowerSeldon1975, The people you are calling are using analog lines.  That's what causes it.
16:11.41zoa<PROTECTED>
16:12.03Seldon1975Manx: can you elaborate?
16:12.08ManxPowerRead the wiki and the mailing list archives.  Getting rid of echo is one of the hardest things to do in voip.
16:12.10ManxPower~docs
16:12.12jbotmethinks docs is probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html
16:12.16ManxPower~mailinglist
16:12.17jbotextra, extra, read all about it, mailinglist is Search Asterisk mailing lists by prepending site:lists.digium.com to your Google search.  Browse the mailing list archive at http://lists.digium.com/ or search through it at http://www.asteriskguru.com/archives/search.php, or and there is also the Macintosh Asterisk mailing list at http://www.astmasters.net/maml.html
16:12.19file[desk]KILL THE ECHO!
16:12.19fugitivolol, 15 free ubuntu cds arrived
16:12.28lo_techwho's the resident 'nsf=' guru again?
16:12.42JohnnyCWhen can I find Policies from DIGIUM about use of Asterisk Logo ?
16:12.49Seldon1975Manx: near-side echo for SIP phones only is a fairly specific subset of 'echo' issues; can you direct me to a specific resource
16:12.50*** join/#asterisk msw (n=msw@24.172.59.42)
16:12.56zoaJohnnyC: email them
16:12.57ManxPowerlo_tech, Check with the college people.  They are usually familiar with Non-Sufficient Funds
16:13.06zoai know their legal team doesnt like it
16:13.13ManxPowerSeldon1975, Um, actually near-side echo is the most common echo.
16:13.27Seldon1975is it usually restricted to the SIP phones though
16:13.34fugitivowhere the hell get the money to give free cds to everybody?
16:13.34Seldon1975the Analog phones are no problem
16:13.36JohnnyCzoa thanks
16:13.45ManxPowerit's not actually near-side echo, BTW.  Your voice is being sent out and being echoed back by the 2-wire analog line on the far end.
16:13.49*** join/#asterisk brettnem (n=brettnem@72.29.102.158)
16:13.54brettnemhey all
16:13.58Seldon1975what Im getting at is; is it likely a problem wiuth the SIP hardware or * or zaptel
16:14.12eKo1hmm...maybe i should start reading about using switch =>
16:14.13brettnemhey anyone know why on a T1 PRI call that voice would clip at the begining and ends of words?
16:14.15Seldon1975Manx: ok that's interesting
16:14.23eKo1never used it before. i wonder if it would solve my problems
16:14.31ManxPowerSeldon1975, WRONG!  Your analog phones have echo too, but because the audio path on analog has such a low latency you near the echo as a sidetone and not as an echo.
16:14.32brettnemer audio clips at the begining and ending of words
16:14.33jake1932brettnem: echo settings
16:14.34jake1932?
16:14.42Seldon1975Manx: my card has hardware echo cancelling - does that hardware typically not work?
16:14.46ManxPowerIn voip the audio path latency is high enough you hear the echo as echo.
16:14.52brettnemjake1932: it's not echo.. it's choppy audio when going from silence to speech
16:14.53TheCopsbrettnem, echo issue or a codec issue maybe
16:14.55ManxPowerSeldon1975, call digium.
16:15.03TheCopsbrettnem, a silence supressor ?!
16:15.09ManxPowerSeldon1975, if you have the hardware echocan and you are getting echo then you need to call Digium
16:15.11Seldon1975Manx: yeah, they have an engineer calling me back in an hour (hopefully)
16:15.11lo_techManxPower: aaaaaaaaah, ha?
16:15.15brettnemTheCops: sounds like it.. but this is PRI to ULAW SIP
16:15.25TheCopsbrettnem, show translation
16:15.25brettnemasterisk to asterisk.. no silence suppression
16:15.31TheCopswatch out the ms between translation
16:15.35TheCopsI guess it will be 1
16:15.50TheCopsbut just to verify
16:15.50brettnemTheCops: it's ulaw to ulaw!
16:15.55TheCopsyou told me ULAW to PRI
16:16.03ManxPowerbrettnem, turn off your damn onboard GigE
16:16.04brettnemwhat is PRI? a codec?
16:16.10iCEBrkrThis is fucked up.. I had this PRI working just fine yesterday.  Rebooted the box for some testing... and now the PRI is all jacked up
16:16.20brettnemManxPower: you really think that's it?
16:16.23Seldon1975Manx: have you heard of people having issues with crosstalk?
16:16.32ManxPowerbrettnem, It's a pretty common issue.
16:16.33TheCopsbrettnem, a PRI is a 23 numerics lines from a telco provider
16:16.39ManxPowerSeldon1975, No.
16:16.40synthetiqhow do u remove the max fowards in the sip 200 ok message
16:16.41ManxPoweronly echo
16:16.42synthetiq?
16:16.45Seldon1975ManxPower: we seem to have crosstalk - should that be handled by the TDM card too?
16:16.51Seldon1975oh
16:16.56brettnemTheCops: right not a codec.. the PRI uses PCM encoding
16:16.57ManxPowerBut digium's hardware echo can is so new.....
16:17.06iCEBrkrcrosstalk? that was a terminal program back in the late 80's
16:17.08iCEBrkr:)
16:17.20Seldon1975iCeBrkr: i wish
16:17.24ManxPowerSeldon1975, no idea.  In my three years of using asterisk and being on this channel and on the mailing lists I have never heard of anyone having a crosstalk problem
16:17.26Kattyunder the crosstalk!
16:17.29Kattydo do do
16:17.37iCEBrkrKatty: That was bad.
16:17.38*** join/#asterisk Assid (n=assid@203.115.64.59)
16:17.40ManxPowerbrettnem, Are you in alaw land or ulaw land?
16:17.53brettnemManxPower: the eth card doesn't share INTs with the wct4xxp
16:17.54KattyiCEBrkr: that's what i specialize in.
16:17.55brettnemulaw
16:17.55Seldon1975ManxPower: yeah - I don't understand how it can be happening; but it is
16:18.05ManxPowerbrettnem, doesn't matter.
16:18.19backblueManxPower: i still cant have 2 users with the same name, in the same server, with diferent domains.
16:18.22brettnemManxPower: argh.. this box is 200 miles from any hands
16:18.26ManxPowerbrettnem, The problem with onboard GigE or RAID is IRQ latency
16:18.27TheCopsManxPower, you are not kidding with onboard gigabyte ethernet card ? Why the hells it cause the problem
16:18.30coppiceManxPower people had a lot of crosstalk with the TDM400 early on. I think that was fixed
16:18.38ManxPowerbackblue, correct, but that is not a problem.
16:19.02Seldon1975coppice: thats interesting
16:19.05ManxPowerbrettnem, not ALL GigE causes problems.
16:19.13backblueManxPower: yes it is, ppl in companys doesnt want to have user-companyA or user-companyB
16:19.13brettnem03:0a.0 Ethernet controller: Intel Corp. 82541GI/PI Gigabit Ethernet Controller
16:19.18Seldon1975coppice: do you know if it was a hardware issue?
16:19.29brettnem01:01.0 Ethernet controller: Intel Corp. 82547GI Gigabit Ethernet Controller
16:19.36ManxPowerbackblue, Why don't they want that?  They will NEVER EVER KNOW if you have it set up correctly.
16:19.42backblueManxPower: they wanto user@companyA.com
16:19.56backblueManxPower: they will never know?
16:20.01ManxPowerbackblue, how do your users dial letters on the telephone keypad?
16:20.01synthetiqmy old provider cant support my sip 200 Ok messages because of the max fowards message with in it, how can i modify this with out hacking the code
16:20.09coppicepeople were getting weird noises in the analogue ports. I think it was a decoupling issue, and they had to add caps
16:20.12backblueso you can do like prepends?
16:20.20backblueManxPower: they use software phones.
16:20.29backbluethey can use letters to dial
16:20.37backblueone of the ideals of voip
16:20.39ManxPowerbackblue, There is your problem.  Don't do that.
16:20.49backblueits NOT to use numbers for dialing, and use your company email addr
16:20.55ManxPowerbackblue, Asterisk is extension centric.
16:21.13ManxPowerWhat you want is a real SIP router like SER.
16:21.23backblueManxPower: what do you mean by that?
16:21.23TheCopsManxPower, with a PCIx RAID controller, it can cause problem ?!
16:21.37ManxPowermy users access vouicemaiol by dialing 3509 or xxx819-3509
16:21.39backblueManxPower: SER can distribute calls?
16:21.44ManxPowerTheCops, tes.
16:21.46backblueby domains?
16:21.47lo_tech'one of the ideals of VoIP' is dial by email addy? since wtf?
16:21.48ManxPowerbackblue, yes.
16:21.52ManxPowerbackblue, I don't know.
16:22.14backblueso why are you saying, that i need SER? i dont understand.
16:22.19ManxPowerlo_tech, the SIP cheerleaders like saying you don't have to dial by number.
16:22.20*** join/#asterisk chaoscon (n=ph33r@smartserv/ceo/chaoscon)
16:22.30TheCopsManxPower, did you tried to use a IBM server as asterisk PBX ?!
16:22.38ManxPowerbackblue, ASTERISK IS A PBX.  PBXs route calls by extension.
16:23.00beebzhttp://64.233.167.104/search?q=cache:Rmfyp0mpk94J:www.voip-forum.com/%3Fp%3D153%26more%3D1+sip+dial+by+domain&hl=en&client=firefox-a << dial by domain
16:23.03lo_techManxPower: yep, but that's SIP, and SIP's not the only dog in the race
16:23.07*** join/#asterisk _Sam-- (n=sam@phone2.kneedraggers.com)
16:23.09brettnemlo_tech: duh.. but no one actually does it.
16:23.14ManxPowerTheCops, Why would I?  I have a standard server configuration that works for me.  I call up my vendor, tell them or order the "Stirling Asterisk Server" and a week later I have my box.
16:23.37_Sam--hey, i havent done any work with analog lines...what would be the best method to connect 4 POTs lines to asterisk?
16:23.43ManxPowerbackblue, if you really want to force your model on asterisk then you CAN do it.
16:23.51backblueManxPower: but its a pbx ip oriented. it should route only not from extensions.
16:23.56iCEBrkrDude.. VoIP must mean you dial by EMail address cuz it uses the INNERNET!
16:24.03TheCopsManxPower, was just a question, I'll use a IBM Server with a RAID controller as PCIx with a T1 card..I dont want to get problem hehe
16:24.07ManxPowerTheir SIP userid and all is their MAC address, then you point the device to a specific contect in extensions.conf
16:24.27iCEBrkrbackblue: http://www.voip-info.org, come back when you know what VoIP and a PBX is.
16:24.43ManxPowerbackblue, your problem is that you think SIP User ID = what you dial.
16:24.45ManxPowerThat is NOT true.
16:25.03*** join/#asterisk fugitivo (n=ajf@209.13.240.236)
16:25.12iCEBrkrManxPower: But sometimes it's easier to actually configure it that :D
16:25.16saftsackFile digits/1F does not exist in any format
16:25.22backbluewhy not?
16:25.24saftsackI cant find this file anywhere :(
16:25.29lo_techbut wtf does email addy come into it? ip is a routing protocol, email may be built on it, and SIP/IAX may run on it... but VoIP <> every app that runs opn IP
16:25.42iCEBrkrlo_tech: Don't you know. Email is the internet!
16:25.51ManxPowerI can have exten => john,1,Dial(SIP/0004f2011935-a)
16:25.57fryfrog<PROTECTED>
16:26.03saftsack[TK]D-Fender|AFK, do you know where to find this files?
16:26.04rob0the internet is the web :)
16:26.04iCEBrkrfryfrog: haha
16:26.05lo_techAOL 4tw!
16:26.15*** join/#asterisk Nico_Bdav (n=nico@shm67-2-82-227-191-194.fbx.proxad.net)
16:26.24backblueManxPower: i want john@company.com,1,Dial(SIP/WHATEVER)
16:26.26ManxPowerlo_tech, it's not an e-mail address, it just looks like an e-mail address.
16:26.26fugitivoi want to surf the internet
16:26.30fryfrogI get the AOL Internet cd in the mail every two days!  that way i keep up to date on websites!
16:26.41ManxPowerbackblue, you can't do that.
16:26.42backbluethat's what i need, and i dont understand why asterisk dont do it.
16:26.45iCEBrkrfryfrog: I'm trying to figure out how they put the WHOLE internet on 1 CD
16:26.56fryfrogits zipped
16:26.59iCEBrkrbackblue: you're smoking crack!
16:27.05*** join/#asterisk dnc (n=duncan@213-244-225-42.wireless.org.yu)
16:27.07iCEBrkrfryfrog: crazy!@
16:27.09backblueiCEBrkr: hehe, i dont even smoke
16:27.12ManxPowerbackblue, Asterisk is a MILTIPROTOCOL PBX and not all protocols support dialing by URI
16:27.22*** join/#asterisk _T3_ (n=rposada@200.63.231.210)
16:27.29ManxPoweriCEBrkr, no, he's been brainwashed by the SIP cheerleaders.
16:27.30fugitivobackblue: why you want an extension like that?
16:27.32fryfrogAOL: You're so easy to use, no wonder we're #1!
16:27.40dncokay, any way of seeing what my zaptel cards are jumper set to?   im getting "Dec 22 17:30:32 ERROR[1861]: chan_zap.c:6985 mkintf: Channel 24 is reserved for D-channel.
16:27.40dnc" when it should be connected to an E1
16:27.47iCEBrkrManxPower: Apparently
16:27.56backbluefugitivo: because its the future, and you could have virtualhosts.
16:27.58ManxPowerdnc, your card is in T-1 mode.
16:28.03fugitivodnc: it looks a t1
16:28.08iCEBrkrbackblue: Seriously, what planet are you from?
16:28.10fugitivobackblue: nonono
16:28.13ManxPowerbackblue, it's not the future ubnless the PSTN can talk to it.
16:28.15fugitivobackblue: you have extensions
16:28.24fugitivobackblue: extension@yourasteriskbox
16:28.28backblueManxPower: the asterisk does that.
16:28.32ManxPowerbackblue, Asteirsk is not a SIP PBX.
16:28.40iCEBrkr"Please enter the first 15 letters of the persons email address that you'd like to call"
16:28.59ManxPoweriCEBrkr, That would only be for callings coming from the PSTN.
16:29.01iCEBrkr30mins later.
16:29.04ManxPowerOr for calls from phones.
16:29.12iCEBrkr"I'm sorry, I cannot find anyone by that email address"
16:29.17ManxPowerbackblue, the way you want to do it ONLY works for softphones.
16:29.22backblueiCEBrkr: ?
16:29.32Nuggetthe future of telephony is mobile, and the future of mobile doesn't rule out alphanumeric targets.
16:29.35ManxPowerUsing number extensions works for any phone, soft phones, hardphones, the PSTN, etc.
16:29.46iCEBrkrManxPower: T9 baby!
16:29.51Nuggetbut you can't get there from here, that's for sure.
16:29.52backblueiCEBrkr: john@company.com, it checks dns for sip server @company.com and delivers there.
16:29.57backbluejust like email...
16:30.02iCEBrkrbackblue: You're stoned.
16:30.14ManxPowerbackblue, How do I call that person from my cell phone or my landline?
16:30.18Beirdohehe
16:30.23Beirdovoice recognition, of course
16:30.26file[desk]you don't, IT'S PSYCHIC!
16:30.30backblueManxPower: you will have voip numbers
16:30.31Nuggetenum lookups are a good compromise for the time being.  register your numbers with e164.org and start doing enum lookups on placed calls.
16:30.39backbluelike you have geografic and mobile
16:30.40iCEBrkrbackblue: WTF is a VoIP number?
16:30.46backblueyou will have voip ones
16:31.02backblueiCEBrkr: its just like another one.
16:31.04ManxPowerbackblue, What you want to do is not how we do things with Asterisk.  If you want to do it that way you won't get much help here.
16:31.07iCEBrkrI think backblue is trolling.
16:31.21backblueor if you talk SIP, you can use sip url, and dial directly to your email
16:31.32iCEBrkrOMFG
16:31.37ManxPowerbackblue, correct, but that only works for SIP.
16:31.54backblueyes, you have to use sip capable phones
16:32.01backblueoffcourse your mobile phone dont do that
16:32.09Beirdothank GOD
16:32.10iCEBrkrbackblue: So again, you're stoned.
16:32.15ManxPowerbackblue, No, you need to use SIP SOFTPHONES.
16:32.25backbluebut for that, your isp will get you a DDI, and forward to your sip device
16:32.26Nuggetthat's not true, Manx.
16:32.34ManxPowera sip hardphone will still require you to dial 50 zillion digits to enter the SIP URI
16:32.35Nuggetthe cisco phones can dial sip targets, for instance.
16:32.47Nuggetfrom an xml address book it's transparent.
16:32.51backblueyes
16:32.59*** part/#asterisk dnc (n=duncan@213-244-225-42.wireless.org.yu)
16:33.01NuggetI think you guys are just enjoying piling on too much and you're being too hard on backblue.
16:33.08iCEBrkrbackblue: puff-puff pass man, you're fucking up the rotation
16:33.09Nuggetit's not like the concept is crazy.
16:33.11ManxPowerNugget, Dialing a number extension is transparent then too.
16:33.15backbluebut thats its just products shit... just think in the implementation.
16:33.19Nuggetsure, but dialing a number is less useful.
16:33.32fryfrogi call myself by email all the time, it rules
16:33.48ManxPowerI
16:34.02backbluebut ser does that? i still dont get it!
16:34.02iCEBrkrfryfrog: Don't your fingers hurt from dialing all those numbers?
16:34.14ManxPowerI've been using Asterisk for 3 years, deploying it in production for 2 years and I have never had to dial anything other than numbers to call anywhere.
16:34.22NuggetThat's great for you.
16:34.27backblueiCEBrkr: you are in the past, come back to the present, and thinks in the future
16:34.34Nuggetbut it's not fair to act like we're all crazy for wanting to move in that direction.
16:34.34ManxPowerbackblue, SER is a SIP only call routing software.
16:34.50ManxPowerMy users would KILL me if I made them dial letters.
16:34.55iCEBrkrNugget: Please pass the bong.
16:35.01fryfrogdialing numbers?
16:35.02lo_technot crazy to be moving there... but dont ask us how to get the config working today!
16:35.03backbluewith wimacs you will have your sip phone, everywhere.
16:35.07*** join/#asterisk Sedorox (i=brandon@smartserv/cna/Sedorox)
16:35.08fryfrogi just put the address into my aol!
16:35.24ManxPowerNugget, The URI model of dialing prettty much excludes any device without a keyboard.  I consider that crazy and have from day 1
16:35.25lo_tech'why cant I' for items in the future is just aggravating
16:35.28Nugget90% of the calls I make are not dialed, they're from my address book.
16:35.35Nuggetso it makes no difference to me.
16:35.53backblueManxPower: i think you are wrong, diling letters will be just nicer that numbers.
16:35.55NuggetNobody is disagreeing with you on that point.  it's ridiculous to type letters on a numeric keypad.
16:35.56iCEBrkrNugget: I agree with you there.. But that's on my SidekickII
16:36.01*** join/#asterisk easel (n=erik@interlink-gw1.ilsw.com)
16:36.02Sedoroxin 1.0.10... ztdummy.. choppy audio... this due to having a OHCI USB controller and/or SMP system?
16:36.12ManxPowerbackblue, not if you have to do it from a DTMF keypad.
16:36.14Nuggetbut there's already plenty of methods to get around that, and going forward there will surely be more.
16:36.15*** join/#asterisk MSchroeder (n=root@216.64.133.250)
16:36.36ManxPowerNugget, then you have to maintain TWO dialplans.  One for URIs and one for digits.
16:36.40backblueManxPower: so the problem its not my idea, its your phones...
16:36.44ManxPowerSorry, but I have enough work already.
16:36.51backbluechange your phones, and your users will give you a kiss
16:37.00Nuggetwell, no, not really, but you and I have already been around the ringer on my fucked-up dialplan so let's not go there.  :)
16:37.02ManxPowerbackblue, It's not your idea.  It's the idea of the people that invented SIP.
16:37.11[TK]D-Fender|AFKnugget : Use Sphinx and say they URI......
16:37.14[TK]D-Fender|AFKthe*
16:37.18ManxPowerbackblue, Change my phones to softphones?  Never in a billion years.
16:37.28backbluehoo nice, i want to ask for a pizza, lets check my pizza restaurant web page, and see the email from them.
16:37.34Nuggetit's possible to make asterisk do exactly what backblue wants, but I don't suggest it.
16:37.49backblueManxPower: change your phones, to cisco phones
16:37.50iCEBrkrNugget: Think about the future man!!
16:38.00iCEBrkrTHE FUTURE IS OURS!
16:38.06[TK]D-Fender|AFKbackblue :  pass the crack please? :)
16:38.06Nuggetbecause asterisk isn't designed to do it, so you'll constantly be fighting the current, and two because of people in here who don't share that vision for asterisk's future development.
16:38.09kuku5Nugget: what address book do you use?
16:38.09backbluehave a centralized addr book, and you are up to a near future.
16:38.17ManxPowerbackblue, Cisco will not allow me to dial letters without lots of hassle.
16:38.17Nuggetkuku5: I wrote my own
16:38.24ManxPowerunless it's in the address book, of course.
16:38.31kuku5Nugget: web based?
16:38.39ManxPowerbut %95 of the places my users call are NOT in their address book.
16:38.45Nuggetos x address book with an xml component for my cisco phones.
16:38.50NuggetI can place a call from either place
16:38.57backblueManxPower: yes, that can be a problem.
16:39.01NuggetManxPower: yes, but we run asterisk, not manxpowerisk.
16:39.07Nuggetyour needs aren't really relevant to me
16:39.26ManxPowerNugget, I'm saying that YOUR way is not compatable with the way most people do business.
16:39.33kuku5Nugget: so you use the manager?
16:39.33NuggetWhy should I care about that?
16:40.07NuggetOr, more precisely, why should I let that deter me from doing what I want to do?
16:40.30ManxPowerbackblue, As Nugget says, you can torture Asterisk into doing what you want, but Asterisk was NOT designed to operate in the way you want.
16:40.31kuku5free * free * free *!
16:40.33kuku5<PROTECTED>
16:40.42eKo1dag nabbit...even with iax2 and the 'switch =>' statement, the dial-status is always ANSWERED...
16:40.51ManxPowerAsterisk is designed to have extensions be numbers.
16:41.15MSchroederExcuse me, but could someone with SCCP experience and some time please msg me? Thanks.
16:41.20ManxPowereKo1, I think your carrier is answering the call
16:41.30eKo1yeah, that could be. those fucks
16:41.36kuku5:)
16:41.43eKo1but why would they answer some and not others
16:41.48ManxPowereKo1, As you can see on Server B Asterisk thinks the destination answered.
16:41.56kuku5"quality assurance"
16:42.39*** join/#asterisk UlbabraB (n=caplaz@host23-25.pool82106.interbusiness.it)
16:42.39eKo1hmm...
16:42.53ManxPowereKo1, pri debug span x
16:42.59ManxPowerwill tell you more.
16:43.03ManxPowerdon't ask me to help you read it.
16:44.01eKo1let's see...
16:48.38*** join/#asterisk shido6 (n=shido6@d221-68-216.commercial.cgocable.net)
16:49.43eKo1you're right. the call is being answered by my carrier. wtf?!
16:50.10*** join/#asterisk apardo (n=_apardo@155.Red-83-44-177.dynamicIP.rima-tde.net)
16:50.46*** join/#asterisk L|NUX (n=linux@202.63.195.147)
16:51.16backblueManxPower: yes, i understand, tks, nice talk.
16:51.50*** join/#asterisk syzygybsd (n=chatzill@66.226.228.204.cpe.speedyquick.net)
16:54.42*** join/#asterisk riddlebox (n=james@24-217-15-91.dhcp.stls.mo.charter.com)
16:57.31*** join/#asterisk mishehu (i=mishehu@cshells.shavedgoats.net)
16:58.01*** join/#asterisk konfuzed (n=KonfuzeD@H135.C72.B0.tor.eicat.ca)
17:01.03Delvarhas cut been depricated?
17:01.12*** join/#asterisk tidify (n=tidify@24-182-200-159.dhcp.ftwo.tx.charter.com)
17:01.12lo_tech.
17:01.23*** join/#asterisk user1002 (n=telefoni@ACB2EB49.ipt.aol.com)
17:01.46*** join/#asterisk SwK (n=SwK@12-219-151-128.client.mchsi.com)
17:02.34fryfroghttp://www.bustedtees.com/shirts/speakercity  <-- i don't get it :/
17:03.36user1002Hi, I have a total of 6 Digium TE 405P Cards for sale, used 2 months. My asking-price is EUR 4800,--. You get an invoice, too.
17:03.54SkramXinvoice?! I love invoices, haha
17:04.58fugitivohow much without invoice?
17:05.03DelvarNo application 'CUT' for extension ... no app_cut.so :(
17:05.05Bobacus_Bumfryfrog: did you ever see old school
17:05.22kuku5how do I specify in the dial plan ( extensions.conf )    exten => _01148[NOT 6].
17:05.25user1002I have to sell with invoice.
17:05.42fryfrogBobacus_Bum: no :(
17:06.02Bobacus_Bumwell thats what that tshirt is from, one of the main charectors worked there
17:06.18fryfrogah
17:09.45Bobacus_Bumhttp://www.bustedtees.com/shirts/condiments <-- almost retarded
17:10.41Qwellkuku5: [123457890]
17:10.47TonyMSIP question: how can I make Asterisk send an UNREGISTER to a SIP server with which it has registered?
17:11.44TonyMI want to change to a different box, but the new box is getting a "403 Login limit exceeded", even though the original box is no longer registering
17:15.50kuku5thanks
17:17.12CoffeeIV_Hi guys -- I am sending a fax from a call file and I am getting empty faxes on the other end.  I've messed up some step of it -- dial plan, converting pdf to tiff/g3, or the call file -- please check out my pastebin and I'll appreciate any advice: http://pastebin.ca/34617
17:22.10eKo1hmm...pri debug span 1 shows the following: Progress Description: Call is not end-to-end ISDN; further call progress information may be available inband. (1)
17:22.28fryfrogI think i'd like one of almost each of those shirts
17:22.32fryfrogthey are quite funny
17:28.49ManxPowereKo1, I think that's common
17:30.54Delvarhmm Function CUT not registered
17:31.10Delvarhow do i use CUT on cvs?
17:32.02eKo1ManxPower: oh good. but then i'm not getting any call progress info. which means it isn't available inband
17:34.38Delvararggg its doing my brain in
17:35.25*** join/#asterisk CunningPike (n=arodgers@S010600095b33697f.vc.shawcable.net)
17:36.18CunningPikeGreetings all: Does anyone know if #include works in voicemail.conf? If so, how does * handle password changes if the voicemail entry is in an included file?
17:36.52*** join/#asterisk Jammy (n=jammy@24.244.182.192)
17:37.24mutilatoranyone used webSTONE before?
17:38.12jlukJust been reading that dial by 'email like' discussion.
17:38.12tzafrir_laptopCunningPike, #include generally works. Unless you use vmail.cgi
17:38.16_Sam--is there any FXO hardware that could fit in a rack mount chassis?
17:38.19jlukWhat's the problem - it works.
17:38.23*** join/#asterisk ast-newbie (n=newbie@80.93.236.106)
17:38.32jlukyou just map names to extensions
17:38.40file[desk]ManxPower: how is thy intarweb connexion?
17:38.42CunningPiketzafrir_laptop: Thanks - I'll try it out.......
17:38.44*** join/#asterisk stkn (n=stkn@gentoo/developer/pdpc.active.stkn)
17:38.57ast-newbieneed help with ztdummy don't want to work
17:39.11tzafrir_laptopCunningPike, if it doesn't work: let me know of your asterisk version
17:39.14ast-newbielinux kernel 2.6 usb-uhci loaded
17:39.19ManxPowerfile[desk], HIGH LATENCY BUT FAST
17:39.21CunningPikeOK - we're on 1.2.1
17:39.33CunningPikeI'll give it a try on our test server and see how it goes
17:39.38TonyMast-newbie: ztdummy doesn't use usb under 2.6 kernel
17:39.38tzafrir_laptopyou don't need usb for ztdummy in 2.6
17:39.52file[desk]ManxPower: nifty, sorta
17:40.01ManxPowerSORTA
17:40.01tzafrir_laptopast-newbie, kernel? distro? zaptel version?
17:40.09CunningPiketzafrir_laptop: Are you the tzafrir of Xirxom fame?
17:40.11ast-newbie@<tzafrir_laptop> i know but i wont work
17:40.25tzafrir_laptopwhat error do you get from 'modprobe zaptel' ?
17:40.32tzafrir_laptopCunningPike, yes, from Xorcom
17:40.36ast-newbie@<tzafrir_laptop> i got just funny noices when i load ztdummy
17:40.46CunningPiketzafrir_laptop: Apologies for the typo - oops :)
17:40.50ast-newbie@<tzafrir_laptop>none errors
17:40.55*** join/#asterisk Kynetix (n=syro@ip-143-80.sn1.eutelia.it)
17:41.10tzafrir_laptopast-newbie, so how can you tell that there is an error?
17:41.27tzafrir_laptopCunningPike, do you have a problem with such includes?
17:41.35CunningPikeNot yet ;)
17:41.48CunningPikeI'll give it a try and see if it works for me
17:42.23Kynetixgood day at all, i'm a newbie of asterisk
17:42.25ast-newbie@<tzafrir_laptop> i can't but as said before when i load ztdummy
17:42.41ast-newbiemeetme won't work
17:42.50Kynetixsomeone know the hardware needing for a little call-center completly voip?
17:43.19tzafrir_laptopast-newbie, head -c 0 /dev/zap/pseudo
17:43.26jlukkaldemar: depends on what you call a 'little call center'
17:43.27tzafrir_laptopdoes it give any error?
17:43.53jlukkaldemar: sorry meant for Kynetix
17:43.55tzafrir_laptop(in the linux CLI)
17:44.00jlukKy: depends on what you call a 'little call center'
17:44.22Kynetixi work in a little society, and there is a telemarketing work before the contact with the clients
17:44.30ast-newbie@<tzafrir_laptop> no
17:44.57tzafrir_laptopif this gives you no errors, then you have a zaptel timing source.
17:44.59Kynetixmy boss and me wants know the hardware for reduce all the call-center ina a little group af 6-10 voip phone
17:45.16TonyMast-newbie: look in ztdummy.c and find the line #define USE_RTC
17:45.18Kynetixfor calling mobile and home number
17:45.26TonyMis it surrounded by #if 0 ??
17:45.29ManxPowerfile[desk], as soon as I can deal with the cableing I'll route SSH via dialup, which will be lower latency
17:45.38jlukKynetix: what sort of incoming phone line - PRI ?
17:45.46file[desk]ManxPower: good good
17:45.47tzafrir_laptopast-newbie, use zttest to check if it actually works
17:46.09tzafrir_laptopBut even if it is lousy, meetme should work and not generate an error
17:46.26harlequin516Can I pass variables using setVar Application and Data when orginiating a call?
17:46.40Kynetixincoming dial pass into the adsl line of society and the same for outgoing dial
17:46.53Kynetixall the sistem based on viop phone
17:46.57tzafrir_laptopast-newbie, what error do you get from meetme? Did you try restarting asterisk?
17:47.11Kynetixfor calling everykind of harware telephone
17:47.29jlukKynetix: so voip all the way to termination provider.
17:47.34harlequin516Or is SetVar only for Context Extension Priority  Originates?
17:47.43Kynetixyes
17:47.50ast-newbie@<tzafrir_laptop> i am not that fast wait for sec
17:48.13jlukKynetix: so all you need is the servers + phones and away you go.
17:48.31jlukKynetix: Assuming you've enough bandwidth to handle the calls.
17:48.48Kynetixfor bandwidth is not a problem
17:48.54harlequin516Anyone expert here know call originate?
17:49.08Kynetixi need a powerfull server or i can use a old pc?
17:49.20ManxPowerKynetix, Yes.
17:49.39Kynetixpowerfull, like a P4 architecture?
17:50.05jlukKynetix: If you're only handling 10 extensions then most modern PCs will do.
17:50.05Kynetixexample: P4 1GB ram and 80GB HD is sufficient?
17:50.12jlukKynetix: yes.
17:50.17Kynetixok
17:50.26Kynetixexist ethernet voip phone?
17:50.53*** join/#asterisk A-Tuin (n=a-tuin@steves.ip.v4.me.uk)
17:52.09jlukKynetix: http://www.voip-info.org/wiki/view/Asterisk+phones
17:52.12*** join/#asterisk JunK-Y (n=junky@67.68.74.139)
17:52.15Dandananyone using PA168S based phone?
17:52.37CunningPiketzafrir_laptop: OK - I renamed my existing voicemail.conf to voicemail.default and then included it in a new voicemail.conf file. I then changed my password - funny thing is, the file doesn't show the password change, but it's changed when I log in. Does * cache voicemail.conf?
17:53.16benjkfor 10 extensions a 500MHz Celeron is even good enough
17:53.23*** join/#asterisk Ariel_ (n=Ariel@adsl-068-157-125-248.sip.mia.bellsouth.net)
17:53.53ast-newbie@tonyM there isn't any line containing USE_RTC
17:54.24Kynetixthanks a lot
17:54.44CunningPiketzafrir_laptop: When I restart *, it reverts to the old password in the file, so there must be some caching going on. However, I think when  the entries are directly in voicemail.conf, they get written right away....
17:54.57TonyMast-newbie: did you get zaptel from CVS or FTP?
17:56.22ast-newbiei am not sure my friend hwo is linux administrator helped me with installation and he brought source files
17:56.34TonyMsounds like you have old ones
17:56.40TonyMwhat version is your kernel?
17:56.44ast-newbiemaybe
17:56.49[TK]D-Fender|AFKPA168 = cheap junk.
17:57.13ast-newbie2.6
17:57.16tzafrir_laptopCunningPike, I needed to use a script to change the passwords
17:57.30TonyM2.6.what?
17:57.32TonyMuname -r
17:57.52CunningPiketzafrir_laptop: OK - that confirms what I thought, then - it doesn't write through to included files
17:58.07ast-newbie2.6.13-15-smp
17:58.09CunningPiketzafrir_laptop: Many thanks for your help - and best of luck with your products - they look great
17:58.43TonyMok, with 2.6.13 onwards you need the ztdummy from zaptel 1.2 - older ones won't work cos the kernel timing is different
17:58.58TonyMyou need one that has USE_RTC in the source code
17:59.06TonyMfor ztdummy.c
17:59.11ast-newbiebut i have asterisk 1.2
17:59.25TonyMbut you don't appear to have zaptel 1.2
17:59.46tzafrir_laptopNote that zaptel 1.2 should generally work with asterisk 1.0
17:59.55tzafrir_laptopnm
18:00.04TonyMyes, but ast-newbie appears to be the other way round
18:00.11ast-newbieyeah its 1.0.9.2 thanks a lot i am going to download it rright away
18:00.21TonyMok, that should fix it
18:04.04tzangertonym == anthm ?
18:04.53[TK]D-Fender|AFKtzanger : not unless he moves across the ocean...
18:05.58easelanybody here using 802.11 voip phones?
18:07.58loudsued to.
18:07.59tzangerheh
18:08.00loudused*
18:08.08easelhah. i'm guessing it sucked?
18:08.39loudsort of, used to have a zyxel prestige 2k, i gave it away
18:08.55[TK]D-Fender|AFKCrappy battery-life, interfaces, range, etc.  So far it looks like "just get an ATA"
18:09.06loudbut thats me, there ar elots of new wireless voip phones that i havent tried yet.
18:09.13easelhmm. we're standing here debating exactly that =P
18:10.00easelwe've got a bunch of snom 190's... and they are pretty much crap compared to the cisco 7960's we just go in... i don't want to take leap back by adding a wifi phone to the mix
18:10.32[TK]D-Fender|AFK7960's.... ouch.
18:10.44loud<3 7960's
18:10.44easelnot a  fan?
18:10.54*** join/#asterisk hugo-v6 (n=hugo@ns1.bundesunixminister.de)
18:10.56[TK]D-Fender|AFKJust pricey.
18:11.02hugo-v6hiho
18:11.03[TK]D-Fender|AFKVery nice, don't get me wrong
18:11.07easelthey're expensive but worth it in my book.... dealing with cisco on the phone to get firmware etc is total ass
18:11.15loudthey are expensive outside the us.
18:11.30*** join/#asterisk malverian[work_] (n=pawalls@pawalls.teamgleim.com)
18:11.34easelthe call quality etc is way better than snom thats for sure, and the 'feel' is right
18:11.58[TK]D-Fender|AFKSnom just looks awkward to me, and I really dislike the LCD'sa
18:12.31azzie_Snom's LCD usability is close to absolute zero
18:12.40azzie_makes you wonder why they did it at all :)
18:12.44CunningPikeWe're using a UTStarcom Wifi phone - so far so good
18:12.59hugo-v6.o(i heared snom?)
18:13.05easelyeah, its a waste... somebody needs to make an open source snom firmware, maybe it could get to the point of making the hardware usable
18:13.31hugo-v6got someone echoes on snom 190's?
18:13.32easelcunningpike: hows the battery life? will it run 8 hours between charges?
18:13.42loudhah
18:13.46*** join/#asterisk vmlinuz (n=nabudoco@ns1.ensenada.gob.mx)
18:14.08CunningPikeNot sure - we keep ours on the charger most of the time - help desk folks use it when they all need to  leave their desks
18:14.19CunningPikeBut only for short periods at a time
18:14.37easelahh, ok. call quality is generally ok? do you have multiple lines registered on it?
18:14.43easelis it the model F1000?
18:15.01CunningPikeOne line - quality is actually quite good. Yes, the FT1000
18:15.08CunningPikeOr whatever it is
18:15.26easelhmm cool. maybe i'll try that one instead of the zyxel
18:15.35CunningPikeThe call drops temporarily if you switch from one AP to another, but it picks it up again
18:15.39iCEBrkrIs there any way for zttools to take a PRI OUT of loopback?
18:15.57easelcool. this one will be used on a single AP so that won't be an issue for us
18:16.01[TK]D-Fender|AFKiCEBrkr : Nope, all telco side...
18:16.04iCEBrkrok
18:16.13loudtell your telco to stop playikng with your stuff
18:16.51easelok so since i'm chatting... anybody know how I can configure a zap channel and a zip channel to act like an old fashioned analog phone directly plugged into the line?
18:16.57CunningPikeIt's a little fiddly to set up, but works fine
18:17.06iCEBrkr[TK]D-Fender: Can zttools put the PRI in loop back?
18:17.16mog_workhas anyone heard of someone doing iax2 p2p without an asterisk server
18:17.22mog_worklike softphone directly to softphone
18:17.41[TK]D-FenderiCEBrkr : not afaik
18:17.45mog_worki realize it is possible
18:17.56mog_worki am just wondering if there is an implementation besides asterisk
18:17.57*** join/#asterisk BladeRunner05 (n=feelme@adsl-185-91.38-151.net24.it)
18:18.01[TK]D-FenderiCEBrkr : you can do it yourself, just make a cross-over dongle.
18:18.01Ariel_mog_work, you should be able to dial to IP address directly
18:18.52*** join/#asterisk mtnbkr (n=mtnbkr@c-67-165-9-234.hsd1.ct.comcast.net)
18:18.55mog_workyes i realize but is any client set to recieve calls on gues
18:18.55mog_workt
18:19.01mog_worklike i see how it could work
18:19.09*** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com)
18:19.11mog_worki just am wondering if their exists said implementation
18:19.12iCEBrkr[TK]D-Fender: Well the PRI was in loopback
18:23.00[TK]D-Fender:/
18:23.28TheCopshiya [TK]D-Fender
18:23.29TheCops:)|
18:23.32[TK]D-Fenderhey
18:24.24*** join/#asterisk EriSan (n=erisan@81-174-22-54.f5.ngi.it)
18:25.59*** join/#asterisk Seldon1975 (n=someone@199.243.101.131)
18:26.02Seldon1975yaay
18:29.19*** join/#asterisk razu_ (n=razu@213-35-170-55-dsl.trt.estpak.ee)
18:32.39Kynetixwhere can i find a book for dummies for asterisk???:D
18:37.04mog_work~thebook
18:37.06jbothmm... thebook is Asterisk: The Future of Telephony, released under the Creative Commons license and available at http://www.asteriskdocs.org << Read the book online!
18:37.12*** join/#asterisk pa (n=Paolo@unaffiliated/pa)
18:37.59*** join/#asterisk maggit (n=maggit@customer-200-36-59-130.uninet.net.mx)
18:39.08*** join/#asterisk florz (n=florz@2001:1a50:503c:0:0:0:0:1)
18:39.23*** join/#asterisk djMax (n=chatzill@artsalliancelabs.com)
18:40.15Kynetixthanks;)
18:40.59iCEBrkryay, back up and running
18:41.00djMaxso I just upgraded to 1.2 from some ancient thing from April.  Things startup fine from the console, but not with /etc/init.d/asterist start.  Any quick thoughts?
18:41.23iCEBrkr[TK]D-Fender: Get this.  Their NOC doesn't like to see things in alarm, so after awhile they disable the circuits or someshit. Which sounds retarded to me
18:42.12djMaxmagically, it works now, never mind.
18:42.29djMaxseparate note, has anybody implemented "call pickup from VM"?
18:44.40[TK]D-FenderiCEBrkr : These guys amazingly find new ways to lower my opinion of them....
18:45.20*** join/#asterisk alephcom (n=alephcom@host75.net14.mcsnet.ca)
18:45.26alephcomHello Everyone
18:46.36iCEBrkr[TK]D-Fender: Same here.. But I still think the zaptel stuff put this thing in loopback
18:46.57*** join/#asterisk JaredBluestein (n=Jared@nwlnnhbas01-pool3-a32.nwlnnh.tds.net)
18:47.15[TK]D-FenderI don't buy that...
18:47.25[TK]D-FenderYou said their side claimed to be CPE, no?
18:47.35iCEBrkr[TK]D-Fender: repeatedly :P
18:47.54[TK]D-FenderNo, thats not zaptel... they are complete and utter dumb-asses.....
18:48.07iCEBrkrWell something put it in loop back..
18:48.14*** join/#asterisk insomni (n=insomni@x1-6-00-e0-18-6f-34-ad.k455.webspeed.dk)
18:48.31djMaxThis Steal() application, will it do VM pickup for single channel *?
18:48.43iCEBrkr9 out of 13 emails in my bulk folder are about the XBox 360
18:48.43iCEBrkrlol
18:49.01file[desk]omg hi all
18:49.14iCEBrkrhello omg
18:49.23file[desk]that's like -so- hot
18:49.25fryfrogomgwtfbbq!
18:49.34[TK]D-FenderIf they say they're CPE (CUSTOMER  Premises Equipment), thats nothing to do with ZAptel.  They are schitzoid.
18:49.57iCEBrkr[TK]D-Fender: Asterisk was repeating "We say we're CPE but they say they're CPE too"
18:50.02iCEBrkrwhich is a loopback apparently
18:50.23[TK]D-FenderGuess so....
18:50.24iCEBrkrIf I setup zapata to pri_net, it'd repeat the same message, but with Network instead of CPE
18:50.29_T3_anyone knows how to connect asterisk to pstn network using safari c3
18:50.38[TK]D-FenderConfuse them by acting as the other side :D
18:51.10iCEBrkrThe guy took it out of loopback and someone else reset the line and all of a sudden everything works.
18:51.28iCEBrkrThese NOC guys were good about it.
18:51.43iCEBrkrIt's my rep that pisses me off.. She never calls me back
18:51.53iCEBrkrSo I've been collecting the numbers of tech guys so I can bypass her
18:52.03_T3_is possible use setcallerpres with sip??
18:52.31iCEBrkrDude, this rocks.  My answering machine detection code now works with my cellphone vmail
18:52.55*** join/#asterisk pifiu (n=myassisb@208.205.181.170)
18:53.58*** join/#asterisk insomni_ (n=insomni@x1-6-00-e0-18-6f-34-ad.k455.webspeed.dk)
18:54.07fryfrogAnswering machine detection code?
18:54.12iCEBrkrwell, not code, but logic
18:54.26fryfrogwhat use does it serve?
18:54.30fryfroger, purpose?
18:54.43*** join/#asterisk malverian[work_] (n=pawalls@pawalls.teamgleim.com)
18:54.49iCEBrkrfryfrog: Well you wouldn't want to leave some IVR menu on someones answering machine.
18:55.02iCEBrkrSo it determines it's a machine and hangs up
18:55.17fryfrogWooosh!
18:55.24fryfrogthats the sound of that going *right* over my head :)
18:55.29iCEBrkrhaha
18:55.40fryfrogwhat is an IVR menu?
18:55.55[wiebel]I = no clue
18:55.55iCEBrkrfryfrog: I'm using Asterisk like a predictive dial, but not so much predictive...
18:56.00[wiebel]VR is voirce responce
18:56.09iCEBrkrInteractive voice response
18:56.30iCEBrkrI'm not using it for telemarketing so just get that outta your heads right now
18:56.31fryfrogoidc
18:56.31iCEBrkr:)
18:56.44fryfrog== oh i don't cee :)
18:56.55fryfroglike, you call a person and say "press 1 for free monkeys"
18:57.08fryfrogor er, "say "free monkeys" for free monkeys"?
18:57.10iCEBrkrBut it does dial out and it has to make sure it's not an answering machine.. The idea is to be transparent and unannoying as possible.
18:57.28iCEBrkrfryfrog: Press 1 for... Is IVR, yes.
18:57.34easelanybody got any an idea how i can make a sip channel connect directly to a zap channel whenever i pick up the phone? as in simulate an old fashioned analog handset on the line...
18:57.48*** join/#asterisk pa (n=Paolo@unaffiliated/pa)
18:57.50fryfrogapplication might be... if you have a VM, it would call you and say "press1 to hear your messages"?
18:57.53iCEBrkreasel: A 'red-phone'?
18:58.07easelhrm. perhaps, i'm unfamiliar with that term
18:58.09iCEBrkrfryfrog: That too
18:58.11fryfroga "batphone"
18:58.17fryfrogbut i don't think that is what he is asking
18:58.21*** part/#asterisk JaredBluestein (n=Jared@nwlnnhbas01-pool3-a32.nwlnnh.tds.net)
18:58.22*** join/#asterisk ast-newbie (n=newbie@80.93.236.106)
18:58.27fryfrogyou mean, pick up phone and get dial tone?
18:58.38easelyes. pick up phone, connect to analog line on my channel bank immediately
18:58.39fryfrogNOT pick up phone, mom's phone is auto-dialed
18:58.41iCEBrkreasel: Red-phones are the phones you pick up and dial somewhere without actually dialing or doing anything
18:58.49easelah, no, i don't want that...
18:58.55iCEBrkroh
18:59.14easeli've got analog phones on desks... and in the transition to voip process, i want to be able to drop the current analog lines into the voip phones as one of the 'lines'
18:59.15iCEBrkrIf that's what you want, then why is asterisk in the loop?
18:59.17GXTidont most sip phones have a builtin dialtone?
18:59.25fugitivoeasel: what phone are you using?
18:59.33easelcisco 7960's
18:59.39fryfrogdo you just want people to *hear* the dialtone?
18:59.43fugitivoeasel: check if you have the "hotline" option
18:59.46easeli hve some snom 190's but i'm tryin gto get rid of them
18:59.50ast-newbiewhen i compiled zaptel i got following error or warning "clock skew detected. your build may be incomplite."
18:59.50fugitivoeasel: that'll dial when you pickup the phone
18:59.55ast-newbieany help
19:00.06easelfugi: ok ,i'll check for that
19:00.07fugitivoast-newbie: check the time of your machine
19:00.14iCEBrkrast-newbie: Your clock isn't set correctly on that machine
19:00.24iCEBrkrdate
19:00.30fugitivoeasel: you can setup a number to dial automaticaly when you pickup the phone
19:00.38iCEBrkrfugitivo: I don't think he wants that.
19:00.49fugitivoeasel: isn't that what you want?
19:00.53easelok, so i can dial a number. but then i need that number to connect to the zap channel
19:01.12easelwell, i want it to appear as if asterisk isn't involved on that particular line
19:01.13fugitivoeasel: yes
19:01.18iCEBrkreasel: I'm actually still confused as to what you're wanting to do.
19:01.21easelpick up handset, connect directly to the zap channel
19:01.33iCEBrkrWhy's it matter?
19:01.35easelcurrently, the analog phone is plugged into the line directly
19:01.35fugitivoeasel: what do you mean by "connect directly to the zap channel"?
19:01.46easeli'm going to plug the line into the channel bank
19:01.50fugitivoeasel: ok ok
19:01.53fugitivoi understand
19:01.58fugitivoyou don't want the pbx tone
19:02.02easel(right)
19:02.02fugitivoyou want the phone line tone directly
19:02.03fugitivoright?
19:02.16easeli want to give users the option to use the pbx tone and make voip calls
19:02.27easelor to choose a line on their phone, and just get the old fashioned line tone
19:02.34easelso they can transfer calls using centrex, etc, etc
19:02.58easelif it's not possible i'll survive, but i was thinking it might make transition a bit easier for people
19:03.10fugitivodo you plan to use an ivr for that?
19:03.24iCEBrkrfugitivo: Why not just use a dialplan.. If they dial 8 it'll do VoIP
19:03.51ast-newbiedo i need to configure somthing if i want to use ztdummy with asterisk
19:03.55fugitivoeasel: iCEBrkr is right, why not just use 9 or 8 to go to the pstn?
19:04.04easeli've already got that set up
19:04.16easeldoing that doesn't enable them to use call transfer etc
19:04.29fugitivoeasel: what call transfer?
19:04.30iCEBrkrI did that at home when I has PSTN.  I'd have to dial 9 to get a local line.. otherwise it went out VoIP
19:04.33easeli gotta run, i'll bbiab and hassle you all
19:04.38*** part/#asterisk CunningPike (n=arodgers@S010600095b33697f.vc.shawcable.net)
19:04.38easelcentrex call transfer
19:04.47fugitivoi don't know what is that
19:04.53easelas in the call transfer provided by the analog legacy pbx
19:04.54iCEBrkrThey should be able to transfer, it's just Asterisk would stil be in the loop
19:04.58easelhmm
19:05.01easeli'll have to check it
19:05.14fugitivoeasel: asterisk can do everything
19:05.31iCEBrkrlol
19:06.58tzafrir_laptopast-newbie, nothing to configure for ztdummy
19:07.10*** join/#asterisk PupenoL (n=pupeno@200.123.183.89)
19:07.28tzafrir_laptopast-newbie, is that module loaded? lsmod | grep ^zaptel
19:07.48iCEBrkrzaptel.init is kinda fuxored
19:08.41*** join/#asterisk zishanov (n=mail@d57-249-149.home.cgocable.net)
19:09.25*** join/#asterisk backblue (n=moo@82.102.1.42)
19:10.06tzafrir_laptoponce again I ask: short of writing my own program in C/perl: any simple way to pipe command to the asterisk ctl socket?
19:10.22iCEBrkrmanage port?
19:10.34backbluehi, i pluged in my fxo card x100p clone, its working with zaptel, now i need some tut to get it working with asterisk, does anyone can show me a good howto? i cant fund none with fxo, only with fxs.
19:10.36iCEBrkrasterisk -r -x <command>
19:11.10tzafrir_laptopthat requires a setup of user-name and password in advance. You also can't easily control the permissions of sockets (short of complex selinux setup)
19:11.15iCEBrkrbackblue: what's not working?
19:11.30iCEBrkrtzafrir_laptop: eh, thems the brakes
19:11.34tzafrir_laptopiCEBrkr, you can't pipe a stream of commands to asterisk -r
19:11.47iCEBrkrtzafrir_laptop: WTF are you trying to do?
19:12.01ast-newbieyes it is thanks for help
19:12.04tzafrir_laptopand starting asterisk for every now line is a pain
19:12.16iCEBrkrhuh?
19:12.44tzafrir_laptopone thing is to partially re-populate the asterisk db after asterisk has started.
19:12.53alephcomtazfir_laptop:  What are you trying to do.  It's not that hard to do it in perl
19:13.20iCEBrkrtzafrir_laptop: No offense, but I often wonder what people are thinking when they go down roads like this.
19:13.21backblueiCEBrkr: i need to intergrate with asterisk, how does asterisk knows from where to get the channels? i configured in zaptel.conf like this "fxsks=1-2
19:13.27iCEBrkrtzafrir_laptop: like really what are you trying to do..
19:13.32iCEBrkrend goal. big picture
19:13.34backbluebut how does it knows each card each channel is?
19:13.51iCEBrkrbackblue: Cuz that's what zaptel does?
19:13.55*** join/#asterisk chapeaurouge (n=chap@85.201.81.201)
19:13.56iCEBrkrbackblue: /etc/asterisk/zapata.conf
19:14.12iCEBrkrbackblue: you tell it how many channels in there too
19:14.13tzafrir_laptopbasically: restore configuration for an AMP-based setup.
19:14.26*** join/#asterisk ReD-MaN (i=redman@dhcp-0-2-b3-9a-4a-5b.cpe.quickclic.net)
19:14.28iCEBrkroh AMP.. Now I understand everything :P
19:14.37tzafrir_laptoponly AMP has a very intense usage of the asterisk database
19:14.45tzafrir_laptop(AMP was never my choice)
19:14.50iCEBrkrAMP == lame
19:15.20fugitivoamp is not lame, just brings more problems :)
19:15.38iCEBrkrIf you're trying to sound more positive. Sure
19:15.38tzafrir_laptopfugitivo, it is overly-complex and badly written
19:15.42backblueiCEBrkr: in zapata.conf i have
19:15.56iCEBrkrbackblue: ok, then Dial(Zap/1/${EXTEN})
19:16.29*** join/#asterisk RoyK (n=roy@80.239.107.70)
19:16.33fugitivobackblue: cat /proc/zaptel/x where x is the number of the channels recognized by zaptel
19:16.45backbluefugitivo: hoo that nice, tks
19:16.51tzafrir_laptopiCEBrkr, still, I need to modify values in the asterisk database. And I can't seem to find any db_* util to manipulate that file .
19:16.52backblue[channels]
19:16.52backbluesignalling=fxo_ks
19:16.52backbluelanguage=en
19:16.52backbluecontext=incoming
19:16.52backbluechannel => 1
19:16.55backblueups
19:16.58backbluesorry
19:17.01tzafrir_laptop~pb
19:17.02jbotrumour has it, pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.com/
19:17.11iCEBrkrtzafrir_laptop: You were screwed the second you used AMP
19:17.19backbluesignaling=fxo_ks ?
19:17.26backblueor fxs_ks?
19:17.31backbluemy card its fxo
19:17.40tzafrir_laptopfor an FXO card: fkx_ks
19:17.57backbluefkx_ks?
19:18.03*** join/#asterisk insomni (n=insomni@x1-6-00-e0-18-6f-34-ad.k455.webspeed.dk)
19:18.10Kynetixfugitivo: are you italian???
19:18.10tzafrir_laptopiCEBrkr, I asked a specific technical question, actually
19:18.14tzafrir_laptopfor an FXO card: fks_ks
19:18.27fugitivotzafrir_laptop: a gui for asterisk database would be nice
19:18.31fugitivoKynetix: no
19:18.35_T3_i have a cisco 7912 can i use it with asterisk???
19:18.46Kynetixok nothing...;)
19:18.57iCEBrkr_T3_: Does it do SIP?
19:18.57tzafrir_laptopfugitivo, that is *not* what I'm after. I need to do things in a script
19:19.07_T3_i dont know how to check it
19:19.12Kynetixsomeone is italina on this channel???
19:19.13iCEBrkr_T3_: Google?
19:19.16_T3_i just have the phone
19:19.20_T3_ohhhhhhh
19:19.26backbluetzafrir_laptop: where does that says?
19:19.30backblueman what?
19:19.35_T3_in google says that this thing can be setup for sip
19:19.47_T3_but i dont have callmanager
19:20.01_T3_neither the sip software
19:20.32fugitivobackblue: if you have fxo modules, you need fxs signalling, if you have fxs modules, you need fxo signalling
19:20.32tzafrir_laptopbackblue, you need to set the signalling to "fxo" in zapata.conf and in zaptel.conf
19:20.56alephcomfugitivo: What do you want it to do?
19:20.59tzafrir_laptopoops, confused
19:21.07tzafrir_laptopbackblue, you need to set the signalling to "fxs" in zapata.conf and in zaptel.conf
19:21.08fugitivoalephcom: me?
19:21.20alephcomfugitivo: I work with somebody that has a little something put together in php for -realtime but...
19:21.25alephcomyes, you. :-)
19:21.37tzafrir_laptopsignalling=fxo_ks in zapata.conf, fxsks=NUM in zaptel.conf
19:21.39fugitivoalephcom: i don't use realtime
19:21.50iCEBrkrPeople are stoners!
19:22.11fugitivotzafrir_laptop: no! signalling=fxs_ks!
19:22.18*** join/#asterisk darby_t (i=darby_t@dka253.neoplus.adsl.tpnet.pl)
19:23.17tzafrir_laptopoops :-(
19:23.22fugitivo:)
19:23.29backbluetzafrir_laptop: hum ok
19:23.47backbluei need chan_local.so or chan_zap.so ?
19:24.14fugitivobackblue: if your modules are fxo, then signalling=fxs_ks in zapata.conf and fxsks=x in zaptel.conf
19:25.34backbluei need one of this .so modules?
19:25.51fugitivoyes
19:26.09backbluewhich one?
19:26.50fugitivoyou want both modules
19:27.49backbluewhy?
19:28.18*** join/#asterisk implicit (n=implicit@mlsrj200152100p048.mls.com.br)
19:28.28fugitivobackblue: show modules like zap
19:28.29fugitivobackblue: show modules like local
19:29.31*** join/#asterisk ceph (n=amit@adsl-156-151-237.mia.bellsouth.net)
19:29.41backblue*CLI> show modules like zap
19:29.41backblueModule                         Description                              Use Count
19:29.44backblue0 modules loaded
19:29.50iCEBrkrbackblue: you understand al this is docuemented on the wiki?
19:29.52backblueit does not have any
19:30.07backblueiCEBrkr: yes, i asked for the link, because i cant find it.
19:30.18iCEBrkrhttp://www.voip-info.org?? or you need more?
19:30.22backblueyes
19:30.23*** join/#asterisk AgiNamu (n=AgiNamu@8.7.80.197)
19:30.31backbluei must be looking for the wrong words
19:30.34iCEBrkrhttp://www.voip-info.org/wiki/view/Asterisk+config+files
19:30.36iCEBrkrThat's part of it
19:30.39backblueprobably my mistake
19:30.54iCEBrkryour zaptel.conf and zapata.conf stuff
19:31.28fugitivobackblue: load the modules first
19:32.53backbluewell
19:32.57backblueits not loading the modules
19:33.01backbluetomorow i will see this
19:33.02*** join/#asterisk JaredBluestein (n=Jared@nwlnnhbas01-pool3-a32.nwlnnh.tds.net)
19:33.02backbluetks folks
19:34.15*** join/#asterisk PupenoL (n=pupeno@200.123.183.89)
19:36.16*** join/#asterisk mikeyb_work (n=michael@66-193-82-211.gen.twtelecom.net)
19:36.51mikeyb_workcan a winmodem that works with hylafax under the ltmodem package be used with asterisk?
19:37.35*** join/#asterisk seele_ (n=seele@200.124.172.72)
19:38.17seele_I need to configure asterisk@home with a TDM431
19:38.21seele_some help please
19:38.46seele_my estencions works fine
19:38.54seele_but my trunks no
19:39.30*** join/#asterisk irc (n=jay@spcxn.got.net)
19:39.30seele_I dont know how to confiure the trunk line
19:39.36zishanovIn Asterisk@Home I am trying to make a Sales Queue with 3 Agents for the sake of learning, but it is not working. Can anybody help on this
19:39.49tzafrir_laptopalephcom, thank for the suggestion. converting the C code to perl slashed most of it.
19:40.02seele_example ... from ext1 mark 0 ... and have a local PSTN tone
19:40.25iCEBrkr<PROTECTED>
19:40.35iCEBrkrIs that supposed to happen?
19:40.39iCEBrkrLike randomly?
19:41.02*** join/#asterisk loick (n=loick@APuteaux-151-1-67-141.w81-249.abo.wanadoo.fr)
19:42.16loudall the time ?
19:42.20loudor hourly, etc.
19:42.32iCEBrkrhrrm, it's been about 45mins since we reset the PRI
19:43.04loudis acpid enabled ?
19:43.19loudkernel level.
19:43.27iCEBrkrshouldn't be.
19:43.50iCEBrkrI haven't recompiled my kernel and it's not in lsmod
19:43.54AgiNamuWhen you call Busy/Congestion , does Asterisk actually create those tones?
19:44.40AgiNamuiCEBrkr, yes this is documented in the zaptel or zapata config :)
19:44.55AgiNamuyou can change it if you dont want it restarting the PRIs.
19:45.35zishanovAnybody knows how to setup queues in AAH, I couldn't find any help on the Internet
19:45.59fugitivozishanov: #amportal
19:46.10zishanovok, I'll try that
19:46.29zishanovthx
19:46.42fugitivomayority of questions regarding asterisk@home are gui related, not asterisk
19:48.00*** join/#asterisk Renacor (n=kvirc@66.238.64.21)
19:48.15Renacoranybody have a good howto setup soundstation 4000 phones with asterisk?
19:48.53*** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net)
19:50.19[TK]D-FenderPolycom IP conference phones?
19:51.26Renacoryeah
19:51.50[TK]D-FenderThey set up pretty much like any other Sounpoint IP phone.  Have you provisioned one before?
19:52.04*** join/#asterisk EriSan (n=erisan@81-174-22-54.f5.ngi.it)
19:54.01*** join/#asterisk Gunnar (n=gunnar@34.80-203-209.nextgentel.com)
19:54.03Renacoryeah I did 300's
19:54.06*** join/#asterisk rculp (n=rculp@66.173.240.20)
19:54.08*** join/#asterisk brookshire (n=nubb@216.207.245.1)
19:54.11Renacorit uses the same bootrom?
19:54.17*** join/#asterisk jimmy_deanPB (n=jhodapp@indianalifesciences.com)
19:54.24[TK]D-FenderDepends, check the SIP changelog.
19:54.57Renacorwhere do I find that?
19:54.59rculpI have a compile question if anyone can helped. Tried wiki'ing it first and I may have missed something but
19:55.08rculptrying to change the default pid file location
19:55.16rculpwhen compiling
19:55.31rculpand it always defaults to /var/run/asterisk.pid and /var/run/asterisk.ctl
19:55.54rculpI want to do it as /var/run/asterisk/asterisk.pid and /var/run/asterisk/asterisk.ctl
19:56.20rculpI edited the defaults.h file, but it reverts back as soon as I do a make install
19:56.23rculpany ideas? :)
19:56.50MstlyHrmlsRenacor: as long as you're using a 2.6.x BootROM (or higher), it'll work fine on the IP4000
19:58.56*** join/#asterisk joe (n=nnnnnnnn@ip66-107-33-195.z33-107-66.customer.algx.net)
19:59.49RenacorMstlyHrmls: Stupid question, but how can I check what version Im using?
20:00.09rculpn/m, I think I found it
20:00.31MstlyHrmlsRenacor: coupole of ways, do you have th ephone booted and handy right now?
20:01.12*** join/#asterisk Lurr (n=pr0ph3t@host-63-69-20-3.compusource.net)
20:01.18*** part/#asterisk Lurr (n=pr0ph3t@host-63-69-20-3.compusource.net)
20:01.38tzafrir_laptoprculp, that file i auto-generated. the defaults are in the makefile
20:01.57rculptzafrir: I just realized I missed one variable in the makefile
20:02.05rculptraced the variables back
20:02.12rculpand grepped for ASTVARRUNDIR
20:02.18rculpand then found it in makefile
20:02.23rculpI think I have it this time
20:02.28rculpafter make is done
20:03.34*** join/#asterisk A-Tuin (n=a-tuin@steves.ip.v4.me.uk)
20:03.38*** join/#asterisk stevek (n=stevek@slim-eth0.horizonlive.net)
20:03.58RoyKzoa: ping
20:04.07*** join/#asterisk n4y (n=tmalkut@fw.orasoft.net.pl)
20:04.22RoyK~seen zoa
20:04.28jbotzoa is currently on #asterisk (7h 59m 58s).  Has said a total of 70 messages.  Is idling for 3h 51m 22s
20:04.37RoyKargh
20:06.16ceph.
20:08.41upsitesomeone using the snom 360 and its buttons with the hint-system &call pickup without patching chan_sip ?
20:09.22upsiteor is it possible without patching chan_sip ? ;)
20:09.49rculpsetting  ASTVARRUNDIR=$(INSTALL_PREFIX)/var/run/asterisk doesn't seem to fix it. the binary now has /var/run/asterisk/asterisk.pid in it instead of /var/run/asterisk.pid but it still decides to create it under /var/run when started manually
20:12.04RenacorMstlyHrmls: yes I do
20:12.43FuriousGeorgeso i use the ',t' otpion with dial() for legacy pbx stations we use.  the problem is some other IVRs want me to, say, "enter my pin then hit pund"
20:12.47FuriousGeorge*pound
20:12.50*** join/#asterisk Utah_Dav1 (n=boucha@0-1pool138-61.nas28.salt-lake-city1.ut.us.da.qwest.net)
20:13.01iCEBrkrFuriousGeorge: Yea, I hate that.
20:13.02MstlyHrmlsRenacor: ok, if you're running SIP 1.5 and above, you can go Menu->Status->Platform->Phone; then at the bottom of that screen will be the BootROM version
20:13.39*** join/#asterisk lo_tech (n=lo_tech@209.36.181.24)
20:13.47FuriousGeorgeiCEBrkr: any way to change it to ## or something
20:13.54iCEBrkrDon't think so
20:14.24Renacork the sip.ver on my tftp server says 1.4.1.0040
20:14.26iCEBrkrActually, I'm pretty sure it's hard coded in dial
20:14.59RenacorMstlyHrmls: where can I get a later version?
20:15.13FuriousGeorgeiCEBrkr: nuts
20:16.43MstlyHrmlsRenacor: either through "your authorized polycom reseller" (the official line), or poking around a bit on google
20:17.00Renacorheh thanks
20:17.07Corydon-wAnybody ever heard of a monitor blanking when zaptel is loaded?
20:17.36iCEBrkrCorydon-w: That's pretty cool stuff
20:17.38MstlyHrmlsRenacor: no problem :-)
20:20.07*** join/#asterisk Money5ack (i=moneysac@wer.will.spontanficken.de)
20:21.00*** join/#asterisk denon (i=denon@synapse.subneural.net)
20:21.00*** mode/#asterisk [+o denon] by ChanServ
20:21.39Assidanyone seen a way that i can have a followme or something setup ? using php
20:21.58pifiuyooo wasup
20:22.07*** join/#asterisk Tili (n=Tili@202-133-65-79-dialup.sat.net.pk)
20:22.07[TK]D-FenderAssid : any special reason to use PHP for that?
20:22.26Assidwanna put it into a database
20:22.38[TK]D-FenderThat's ASTDB worthy....
20:22.49Assidastdb?
20:23.05[TK]D-FenderAsterisk Database.  Just the Built-in DB! database
20:23.11[TK]D-FenderDB1*
20:23.23Assidhrmm.. but i want it to have a front end
20:23.24[TK]D-FenderI use if to handle my forwarding / follow-me
20:23.29Assidso these guys can change it
20:23.35[TK]D-FenderI use an IVR front end w/ prompts :)
20:23.49Assidyeah.. but how do i update the follow me number
20:23.54iCEBrkrYou do have a front end.. *80 *81
20:23.55[TK]D-FenderThrough IVR.
20:24.09RoyKhm
20:24.14[TK]D-FenderI jsut made a whack of contexts to prompt to enable/disable, where to send the call, etc
20:24.17iCEBrkr*80, "Please enter your follow-me phone number."
20:24.20RoyKasterisk 1.0.x seems to do something funny
20:24.36iCEBrkrRoyK: Nuhh uhhh.
20:24.51Assiddoesnt find that number
20:24.52RoyKtop reports asterisk is using 99.9% cpu
20:25.02iCEBrkrAssid: Well, you'd have to program it in the dialplan
20:25.06RoyKsar reports some 25% per cpu with dual xeon with ht
20:25.14RoyKso wtf is really happening?
20:25.27iCEBrkrRoyK: is your top SMP aware? :P
20:25.42Assidcan someone show me their dialplan for this?
20:25.48RoyKiCEBrkr: sure
20:26.04RoyKiCEBrkr: but that really doesn't change anything
20:26.08[TK]D-FenderAssid : You asked for it :D muahahaha
20:26.10[TK]D-Fender*cough*
20:26.14RoyKit seems asterisk is using exactly one logical cpu
20:26.28mutilatorthere any reason * lags out users?
20:26.33mutilatorbecause//
20:26.48iCEBrkrAssid: All you're doing is setting a FollowMe flag for whatever extension.  So anytime a call goes to that extension make it check the FollowMe flag. If it's set, DB(FollowMe/Number) to redirect the call
20:26.50[TK]D-Fenderhttp://pastebin.ca/index.php
20:26.59iCEBrkrindex.php?
20:27.09mutilatori have qualify on this user
20:27.10[TK]D-Fenderhttp://pastebin.ca/34630
20:27.13[TK]D-Fenderbetter :)
20:27.15mutilatorand a ping window open on their ata
20:27.31mutilatoravg 32ping, 3000 pings, highest ping 130
20:27.37mutilatorbut yet i still get
20:27.37mutilatorDec 22 15:11:20 NOTICE[13332]: chan_sip.c:8481 handle_response_peerpoke: Peer '9893332471' is now TOO LAGGED! (2133ms / 1000ms)
20:27.58iCEBrkrmutilator: I think I had a problem with my SPA doing that once.
20:27.59RoyKmutilator: increase the qualify= value
20:28.22RoyKmutilator: qualify doesn't have anything to do with ping. it takes the time the peer uses to answer an OPTION packet iirc
20:28.24mutilatori get like 40 users doing it at once
20:28.30mutilatorthen 10 sec later when it retried
20:28.36Assidokay lemme try that
20:28.36RoyKthen perhaps asterisk is stupid
20:28.39mutilatorthey all have 30ms or so
20:28.39RoyKqualify=5000
20:28.44Assidthanks [TK]D-Fender
20:28.46RoyKmutilator: read what i wrote above
20:29.07mutilatorwondering if it might be cause by another problem tho?
20:29.13mutilatordevice is being retarded
20:29.17mutilatoror asterish being retarded
20:29.26iCEBrkrmutilator: In my case the SPA was borked.
20:29.42mutilatorthese are all spa's i have on here
20:29.59mutilatorfor the most part all on a wifi WAN too
20:30.08mutilatorso it might be lost packets not being retransmitted?
20:30.14iCEBrkrmutilator: Oh, THAT could be your issue
20:30.47mutilatorya
20:30.51iCEBrkrmutilator: If someone copies a file across that WiFi it could saturate it instantly.
20:30.52mutilatorpossibly
20:31.08mutilatorit's not access point technology
20:31.12mutilatorit's COR and WORP
20:31.13Assidgimme a sec.. just gotta vnc these users then will play with this dialplan
20:31.37iCEBrkrmutilator: So you still think that link can't get saturated?
20:31.55mutilatorsure it can
20:32.14mutilatorbut it's not as easy as that
20:32.20iCEBrkrtrue
20:33.04mutilatori'll make my qualify longer
20:33.08mutilator3seconds or something
20:33.13mutilatornever seen any hit above that
20:33.17mutilatorless they died
20:33.47*** join/#asterisk Raul (i=raul@tesla.xmission.com)
20:33.55mutilatorif i set quality=3000 in the general section of the sip.conf... it applies to all right?
20:33.59mutilatordon't need to do individually
20:35.05mutilatorah well
20:35.26*** part/#asterisk Utah_Dav1 (n=boucha@0-1pool138-61.nas28.salt-lake-city1.ut.us.da.qwest.net)
20:36.49Assidokay now.. back..
20:36.56Assidi have NO CLUE. what exactly that does
20:38.00*** join/#asterisk insomni (n=insomni@x1-6-00-e0-18-6f-34-ad.k455.webspeed.dk)
20:38.51Assid[TK]D-Fender: what exactly does that do?
20:40.50*** join/#asterisk Gerriall (n=NonYa@209.42.198.18)
20:42.44*** join/#asterisk LostFrog (n=reallyno@dsl093-100-093.wdc2.dsl.speakeasy.net)
20:42.55*** part/#asterisk JaredBluestein (n=Jared@nwlnnhbas01-pool3-a32.nwlnnh.tds.net)
20:44.36LostFrogTell me something good, people.
20:44.37LostFrog:)
20:44.37*** join/#asterisk flot (n=flot@87.251.134.73)
20:44.54upsite!seen xylome
20:45.01upsitedamn :P
20:45.27jlukLostFrog: if you put enough detergent in a pond the ducks will sink.
20:45.44jlukLostFrog: there you go - that's something.
20:46.24Money5ackhey guys
20:46.44[TK]D-FenderLostFrog : If you mix equal parts of soap and gasoline you get napalm.
20:47.13Money5acki see currently on an asterisk 1.0.6 BRIStuffed
20:47.37jluk[TK]D-Fender: I thought napalm was parafin & gasoline ?
20:47.38Money5acknow for about 2 hours there isn go any calls over this asterisk
20:48.04Money5ackbut there are over 900 SIP-Channels open
20:48.10Money5ackis there any reason for ?
20:48.18Money5ackthe asterisk isn'
20:48.24Money5ackisn't behind nat
20:48.36[TK]D-Fenderjluk : Somewhat interchangeable :)
20:48.43jluk:)
20:49.05jlukwonder what he actually wanted to be told.
20:50.57SkramX[TK]D-Fender: haha
20:52.12*** join/#asterisk r0d3nt (i=nobody@wsip-24-234-241-145.lv.lv.cox.net)
20:52.16file[desk]FOOD
20:55.29*** join/#asterisk trixter (n=trixter@65.172.209.246)
20:56.30*** join/#asterisk oej_ (n=oej@213.204.186.40)
20:56.48*** join/#asterisk alephcom (n=alephcom@host75.net14.mcsnet.ca)
20:57.50*** join/#asterisk VxJasonxV (n=jason@unaffiliated/VxJasonxV)
20:58.35alephcom[TK]D-Fender:  I better remember that.  Yesterday I said something about a gun and this ladys like "You must be a terrorist"  I'm like lol
20:59.05alephcomTerrorists should know how to make napalm. :-)
20:59.33alephcomThat would be a good thing to experiment with at our family gathering.
21:00.21ast-newbiethanks to all of those who helped me i finally made ztdummy working
21:00.47jlukast-newbie: so what was the final fix ?
21:01.35ast-newbiewell i had old sources for zaptel
21:03.39*** part/#asterisk mikeyb_work (n=michael@66-193-82-211.gen.twtelecom.net)
21:04.51[TK]D-Fenderalephcom : i USED TO WORK FOR A FIREARMS IMPORTER / EXPORTER.  pEOPLE LOVE OVERHEARING MY CONVERSATIONS ABOUT WHICH EXCEEDING ACCESSABLE OLD HANDGUNS CAN CUT THROUGH CLASS 3 KEVLAR LIKE RICEPAPER :)
21:05.07[TK]D-Fenderoops./
21:05.10[TK]D-Fenderstupid caps
21:05.33[TK]D-FenderAl-aquaba jhyad! ... I mean..... pass the mayo please? :)
21:05.46tzanger[TK]D-Fender: heh
21:07.01RenacorAnybody here use soundstation 4000 phones?
21:07.31[TK]D-FenderRenacor : is it just the SIP config you need to get running?
21:08.18Renacor[TK]D-Fender: It's very strange, I have it configured to grab a ip from the dhcp server by mac address which works fine, but then I keep getting "Could not contact boot server, using default config"
21:08.28RenacorIm using tftp for the bootserver
21:12.16*** join/#asterisk areski (n=areski@32.Red-83-55-103.dynamicIP.rima-tde.net)
21:14.42[TK]D-FenderRenacor : Did you specify the "option tftp-server-name" section in dhcpd.conf and set up the service?
21:15.05[TK]D-FenderAnd for Polycom I highly suggest FTP for provisioning
21:15.39alephcomTK]D-Fender:   Lol.  Up here in Canada it's pretty funny.  They have "gun control" but it only applies to the honest people.  I got rid of my guns a few years ago because I did not want to register them but the whole thing is a big, expensive joke
21:16.28[TK]D-FenderShould have kept them.  Pre-reg guns become impossible to buy later.
21:16.33[TK]D-FenderAnd I'm Canadian :)
21:17.03[TK]D-FenderI was working for Canada Post when that bill was passed and got to DELIVER the notices door-door :(
21:17.10alephcomCool!   Yeah, I probably should have but..
21:17.46Renacor[TK]D-Fender: Yeah like this http://pastebin.com/475653
21:17.57alephcomDid you guys get shot at?  We had a guy in Stettler, AB who roughed up one of the door-to-door gun control advocates but that's to be expected
21:18.01alephcom:-)
21:18.12[TK]D-FenderRenacor : you use a FQDN?
21:18.34[TK]D-FenderNo, its just flyer distribution.... ad-mail
21:18.42Renacorfor the phone?
21:18.53Renacorit was set to ip address
21:18.57Renacorim trying string now
21:19.26Renacorugh same thing
21:19.56[TK]D-Fenderand a tcpdump on the server doesn't show the attempt?
21:20.30*** join/#asterisk Uther_P (n=uther_p@66.180.120.82)
21:20.40Renacorthen it tries to load it's build in sip.ld but dies, saying "Bootrom has changed, error is 0x0"
21:20.58[TK]D-FenderEEK
21:21.09Renacorwhat port does tftp use?
21:21.17[TK]D-FenderSwitch to a better BR / SIP revision, set it to load from FTP and make your life a lot easier
21:21.23[TK]D-Fendernot sure really...
21:21.30[TK]D-FenderI only use TFTP for my Uniden's
21:22.04[TK]D-FenderOnly thing they were good for was vandalism targeted phones....
21:22.20Renacorheh
21:22.57[TK]D-FenderSo switch to BR 2.6.1 & SIP 1.5.2 and life should be better
21:23.16[TK]D-FenderMy IP 600's started with 1.4.x and upgraded on first boot.
21:23.26*** join/#asterisk r0d3nt|m (i=r0d3nt@tinfoilhat.net)
21:27.56*** join/#asterisk harlequin516 (n=sham@65.39.84.194)
21:28.55[TK]D-Fenderok, I'm outta here, BBIAB
21:34.24AgiNamutftp is port 69 udp
21:34.30Jammyhey guys, im having a prob with a TDM400P on debian... whenever i try modprobe wctdm or wcfxs its telling me modules are not there... any idea what i maybe doing wrong?
21:34.30AgiNamuoh... im lagged :)
21:35.08jlukJammy: do you actually have to modules installed
21:35.40Jammyjluk: i'm assuming they are, zaptel and asterisk installed fine without any errors
21:35.54mishehugah.  I never remember how to convert wav49->wav with sox...
21:36.55jlukJammy: ok, if you look in the lib/modules/`uname -r`/ somewhere you should see the modules
21:37.01jlukcheck that they are there.
21:38.18*** join/#asterisk Seldon1975 (n=someone@199.243.101.131)
21:38.20*** join/#asterisk nextime (n=nextime@213-140-6-103.ip.fastwebnet.it)
21:39.10Seldon1975I can't get to the Wiki page, can someone please confirm the syntax for allowing users to transfer calls using the # key; is it: exten=>t,1,Dial(Zap/5|Ttr)
21:39.43AgiNamuusing google cache :)
21:39.58AgiNamut: Allow the called user to transfer the call by hitting #
21:39.58AgiNamuT: Allow the calling user to transfer the call by hitting #
21:40.07AgiNamur generates a ringing tone
21:40.19Seldon1975thanks
21:40.32Seldon1975so that syntax with the | is correct?
21:40.40AgiNamuno
21:40.41Seldon1975how do I add a timeout to that ?
21:40.44AgiNamuyou need timeout
21:40.48Seldon1975ok
21:40.49*** join/#asterisk grimse (n=grimse@p5481CB7E.dip.t-dialin.net)
21:40.50AgiNamuits Dial(tech,timeout,options)
21:41.00AgiNamuDial(type1/identifier1&type2/identifier2&type3/identifier3...,timeout,options,URL)
21:41.26Seldon1975so: exten=>t,1,Dial(Zap/5|Ttr,25) ?
21:41.33AgiNamuno, timeout is first
21:41.38Seldon1975ok
21:41.39AgiNamudial(bla,25,Ttr)
21:41.45Seldon1975right thanks!
21:41.56AgiNamuim assumging ZAP/5 is a FXS port
21:42.47*** join/#asterisk nextime (n=nextime@213-140-6-103.ip.fastwebnet.it)
21:43.00Seldon1975yes
21:43.02Seldon1975ta
21:43.15SwK[Work]exten => *,1,Dial(ZAP/${GO_TO_GOOGLE}/${FOR_ANSWERS}|${TIMEOUT}|${OPTIONS})
21:43.39Seldon1975SWK: did I mention I couldnt get to the wiki page?
21:43.42Seldon1975yes, i think i did!
21:43.48SwK[Work]google cache y0
21:43.55*** join/#asterisk dejan2 (n=root@213.137.102.178)
21:44.01SwK[Work]google has a good cache of the wiki
21:44.08Seldon1975CANT EVEN GET THERE
21:44.09dejan2hello ppl.
21:44.11Seldon1975oops
21:44.13Seldon1975sorry caps
21:44.15*** join/#asterisk brad_mssw (n=brad@shop.monetra.com)
21:44.36SwK[Work]hah
21:45.16SwK[Work]hahaha
21:45.30twisted[asteria]server load on www.voip-info.org is 70.29 as of 3 minutes ago
21:45.36AgiNamuyea, I said google cache too :P
21:45.48AgiNamutwisted, you work for Asteria?
21:46.05SwK[Work]AgiNamu: we fired him but he refuses to remove that tag from his name
21:46.06twisted[asteria]ye
21:46.11twisted[asteria]hah
21:46.16SwK[Work]heh
21:46.30*** part/#asterisk SwK[Work] (n=SwK@64.89.118.139)
21:46.36twisted[asteria]anywho
21:46.38twisted[asteria]yes, I do :)
21:47.08*** join/#asterisk SwK[Work] (n=SwK@64.89.118.139)
21:47.10SwK[Work]hah
21:47.12SwK[Work]asshat
21:47.14AgiNamuOh. We've been workign with  a few people over there :)
21:47.17AgiNamuTim, Ken
21:47.18twisted[asteria]:)
21:47.23AgiNamuand xin
21:47.34SwK[Work]AgiNamu: who are you?
21:47.44SwK[Work]Dash right
21:47.46AgiNamuBingo
21:47.49SwK[Work]werd
21:47.50twisted[asteria]ah hah
21:48.00brad_msswit seems as though DTMF codes are interpretted wrong on some cell phones ... is this normal ??
21:48.07AgiNamuAlthough, I've been here long before dash ;)
21:48.11brad_mssw(if you call in to asterisk from a cell phone)
21:48.30brad_msswhappens on both Zap lines here, and over teliax
21:48.35twisted[asteria]brad_mssw, it's not asterisk... it's the cell provider
21:48.56brad_msswwould have thought cell providers would provide dtmf out of band
21:48.58twisted[asteria]sometimes we just don't get the tones
21:49.35twisted[asteria]on PRI it *SHOULD* be
21:49.48twisted[asteria]but that's assuming the cellular provider is sending them properly
21:49.57twisted[asteria]and the switch can interpret them, and so on and so forth
21:50.00brad_msswyeah
21:50.15twisted[asteria]my cell phone, for instance, works fine, whereas the guys next to me doesn't
21:50.43brad_msswit seems consistent that some cell phones don't work
21:50.46twisted[asteria]but if you look, we don't seem to be getting the dtmf properly from his calls, but we do mine
21:50.55brad_msswbut those same cell phones will work on other pbx systems :/
21:51.09Renacorthis damn soundstation 4000 never even attempts to talk to the tftp server
21:51.16brad_msswdunno, not blaming asterisk by any means
21:51.45*** join/#asterisk jcwunder (n=chris@ppp-82-135-79-159.mnet-online.de)
21:51.52twisted[asteria]i'm just speaking from experience...  it could be something missing in the q931 stack for PRI in asterisk
21:52.02twisted[asteria]then again, it could be the provider not signalling it properly
21:52.15twisted[asteria]i just don't see them the same in the debug
21:52.39brad_msswland lines always seem proper though
21:52.44brad_msswwhich is good at least
21:52.47twisted[asteria]*nod*
21:53.16Uther_Pyea, screw those preppy, self-important, cell phone-carrying bastards :D
21:53.16twisted[asteria]anywho..  anyone in here know of a good fax detection that can be used when calling OUT to a receiving machine?
21:53.30twisted[asteria]i can get cng just fine, but answer tones i can't ever get a lock on
21:53.57Uther_Pefax works for me
21:54.12twisted[asteria]Uther_P, that's not the kind of solution i'm looking for here ;)
21:54.18Uther_Pheh
21:54.18SwK[Work]twisted[asteria]: hey I wonder if that Kevin Rose show that went wide band today about asterisk is causing the wiki to get buttraped
21:54.21twisted[asteria]i want to know if i've called a fax machien or not
21:54.33twisted[asteria]SwK, dunno
21:54.51twisted[asteria]but my outbound calls need to be filtered before I get screaming in my ear
21:55.17twisted[asteria]so i tell the system to call this number.  The system calls the number and determines if it is actually a fax number before trying to send a fax
21:55.32brad_msswwould think nvfaxdetect would be able to jump to the fax context if found, even if calling out, instead of incoming
21:55.42twisted[asteria]see,I thought so too
21:55.49brad_msswnever tried though
21:56.04twisted[asteria]but apparently it's only set up to listen for the outbound tones
21:56.10twisted[asteria]answer tones don't seem to trigger it
21:57.17*** join/#asterisk nitestarr (n=knightst@cpe-24-33-15-250.midsouth.res.rr.com)
21:58.27*** join/#asterisk n4y (n=tmalkut@fw.orasoft.net.pl)
21:58.44brad_msswtwisted[asteria]: hmm, zap lines can do it with  faxdetect=outgoing ... though nvfaxdetect just used this same code, except applied it to sip/iax channels
21:59.45twisted[asteria]brad_mssw, no, faxdetect=outgoing listens on the outgoing audio for fax tones
22:00.11twisted[asteria]ie, i'm sending a fax, route it somewhere else
22:00.24*** join/#asterisk zotz (n=zotz@24.231.47.168)
22:00.27brad_msswtwisted[asteria]: ahh, oops
22:00.31*** join/#asterisk Cheetah (n=Snak@62.217.48.111)
22:00.36twisted[asteria]faxdetect=incoming listenson the incoming audio; ie, i'm receiving a fax, route it somewhere else
22:00.46*** join/#asterisk MikeJ[Laptop] (n=ircatjer@d14-69-8-30.try.wideopenwest.com)
22:01.05twisted[asteria]what i need is something like "i've called a fax machine, route it somewhere"
22:01.14rculppart #asterisk
22:01.39Uther_Pum... /part?
22:02.04rculpaye
22:02.06rculpI typer gud
22:02.19*** part/#asterisk rculp (n=rculp@66.173.240.20)
22:04.20*** join/#asterisk adker (n=adker@170-100-233-205.dsl1.glv.ny.frontiernet.net)
22:08.16drraytwisted - you aer being paged in asterisk-dev :)
22:08.56Seldon1975All your base
22:08.57Seldon1975base
22:08.59Seldon1975All your base
22:09.02Seldon1975are belkong to us
22:09.04Seldon1975oops
22:10.30*** join/#asterisk fulgas (n=fulgas@a81-84-117-84.cpe.netcabo.pt)
22:10.47mishehuhas anybody had any problems with Monitor() ?  It was formerly working but now just records silence.
22:10.54mishehu(in 1.2.1)
22:13.59xhelioxhttp://heliosj.iddings.us/images/endisnear-cnn.jpg - omg, the end is near!
22:14.09*** join/#asterisk TheCops (i=nobody@got.securebinary.com)
22:15.15*** join/#asterisk darby_t (i=darby_t@djw207.neoplus.adsl.tpnet.pl)
22:15.21Uther_Poh shit, the cops are here!
22:15.24TheCops:0
22:15.24*** part/#asterisk Uther_P (n=uther_p@66.180.120.82)
22:15.25TheCops:)
22:15.29*** join/#asterisk Uther_P (n=uther_p@66.180.120.82)
22:15.34*** part/#asterisk n4y (n=tmalkut@fw.orasoft.net.pl)
22:16.12syzygybsdmishehu: you upgraded from 1.0.9 or something right?
22:17.23*** join/#asterisk MikeJ[Laptop] (n=ircatjer@d14-69-8-30.try.wideopenwest.com)
22:20.52Jammyjluk: weird... checked and reinstalled zaptel and asterisk, and still no go... Is there any change at how the new asterisk loads modules?
22:20.58_Sam--can someone point me in the right direction on where to look for limiting concurrent calls based on sip registration?  ie so that a registered SIP client can only make one call at a time
22:23.04jlukJammy: dunno - I'm just building 1.2 (or trying to). Last time I upgraded was back in Dec 2004
22:23.20brad_mssw_Sam--: GROUP_COUNT() and GROUP()  (formerly CheckGroup() and SetGroup())
22:23.39_Sam--ty
22:23.55jlukJammy: do you still need to run 'depmod -ae' after installing new modules.
22:24.36Jammywell i ran them when i got the error that modules.dep was out of date... but still no go, I'll try the stable 1.2...was using SVN
22:25.24*** join/#asterisk JFPMartin (n=JFPMarti@adsl-04-85.abel.net.uk)
22:25.34*** join/#asterisk nagl (n=nagl@vie-086-059-104-148.dsl.sil.at)
22:25.47jlukI'm finally getting somewhere. Bloody debian install didn't have a /lib/modules/kernel/build. So I had to build a damn kernel before I could build ztdummy
22:26.17jluknow I can find out what's changed in the last year :)
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22:26.31Jammyhaha
22:26.32Jammynice
22:26.38Jammywish i could get that far
22:27.12Jammyjluk: what hardware are you using with *
22:28.29jlukThis box is a dual PIII thingy - no zap hardware, will be used purely for sip/iax traffic
22:29.13jlukJammy: ok, modules seem to install in /lib/modules/kernel/extra/
22:29.20Seldon1975I cant find this in the Wiki - is it possible to specify more than one email address for a single voicemail box?
22:29.36jlukJammy: where kernel is the number of your running kernel
22:30.12DandanSeldon1975: yes
22:30.50Seldon1975Dandan, cool can you show an example?
22:30.54jlukJammy: Got same problem as you now :)
22:31.16_Sam--check voicemail.conf
22:31.23_Sam--there is example w/ more than 1 email in there
22:31.58jlukJammy: problem is that the modules are installed into /lib/modules/2.6.8/extra but my kernel is 2.6.8-2-686-smp so they should be in /lib/modules/2.6.8-2-686-smp/extra
22:33.05jlukfer f***s sake. WTF are the new modules doing in there.
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22:33.15Seldon1975_Sam: theres example with two; but are 3 or 4 possible?
22:34.01_Sam--i think you'll be about to figure it out.
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22:34.36Seldon1975_Sam--: no, I cant.  is the example you're poining to this: ;4200 => 9855,Mark Spencer,markster@linux-support.net,mypager@digium.com,attach=no|serveremail=myaddy@digium.com|tz=central|maxmsg=10
22:34.49_Sam--dont you see the delimiting?
22:35.08_Sam--and did you actually TRY anything?
22:35.28_Sam--sorry for being an ass...but i think its pretty simple
22:35.30SkramXHey Sistahz
22:35.35SkramXhaha
22:35.40Jammyjluk: ahhh will check that out
22:36.10jlukJammy: I still can't insmod the f;in things though.
22:36.18Seldon1975_Sam--: I don't thinhk theres any need to be hostile; sure I could make assumptions about the syntax but if someone here knows it Id rather ask them than spend an hour pissing about with it
22:36.23JFPMartinTwisted: are you listening?
22:36.38Jammyi dont even see that dir
22:37.24jlukJammy: do you see your kernel listed in /lib/modules/
22:37.28Jammyyup
22:38.43jlukJammy: so is there a /lib/modules/kernel/extra ?
22:39.08*** join/#asterisk saftsack (n=saftsack@p54A7CCC6.dip.t-dialin.net)
22:39.15_Sam--Seldon1975:  E-mail to multiple addresses: Sometimes it's quite handy to have a message go to multiple mailboxes but not so convenient to have to create additional entries in extensions.conf and voicemail.conf. This works great for "phone coverage" for when someone out of the office for a vacation. You can use the /etc/aliases or /etc/mail/aliases file to handle this through your MTA.
22:39.17saftsackhi
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22:39.26Jammyjluk: theres no 'extra'
22:39.35_Sam--taken from:  http://www.voip-info.org/wiki-Asterisk+config+voicemail.conf
22:39.58jontowanyone here familiar with H.323 (specifically H.245 negotiation..) .. need to know if Master-Slave determination has to happen before Terminal Capability Negotiation is exchanged..
22:40.08saftsackis it possible to create a new voicemailmessage directly from the telephone?
22:40.27Seldon1975_Sam--: thanks - so its not configured in the mailbox declaraion itself
22:40.31jontowsaftsack; sure.. call into your voicemail.. 'option 5' under top-level option 3?
22:40.44jontowit'll ask you for a mailbox number to leave it in, then prompt you for a message as though you'd called them
22:40.55saftsackcool :)
22:41.03saftsackso every caller will hear that message then?
22:41.16jontowno
22:41.20saftsackhmm
22:41.21jontowthat will leave someone else a message ..
22:41.28jontowyou mean record your unavailable and/or busy greeting?
22:41.29saftsackoh but that is that what i want ;)
22:41.34saftsackYES :)
22:41.55jontowok, also in your voicemail options.. i believe option 0(?) and it'll prompt you to change your PIN (password), or record unavailable or busy greetings
22:42.15jlukJammy: odd. if you run 'locate wcxfo' do you see a mention of it somewhere within /lib/modules ?
22:42.22eKo1what's the difference between 'sip/1234 is making progress passing it to ...' and 'sip/1234 is ringing' with calls in the cli?
22:42.57JFPMartinjontow: I think masterslave and capablity exchange happen async with each other. I looked through a number of traces and they happen in different orders depending on terminal.
22:43.15saftsackjontow, yes it was 0
22:43.17saftsackok thanks :)
22:44.09jontowdamn
22:44.37saftsackdamn?
22:44.58jontowJFPMartin: currently trying to debug this.. i have the ooh323c stack running and exchanging calls.. the remote end sends it's capabilities (i see it on the wire in tcpdump/ethereal perfectly) and master/slave determination happens perfectly (also see that)
22:45.03Jammyjluk: thats why its soo weird... i dont even see those anywhere
22:45.08jontowbut i don't see where ooh323c from asterisk's side goes across
22:45.19jontowand the remote side dies violently with "no capabilities received"
22:45.34jlukJammy: is your locate database upto date ? - try running 'updatedb'
22:45.34jontowthe logs from ooh323c seem to state that its happened, though
22:45.34saftsacki have another important question. howto determine, that i have a "safe" dialplan, also that there arent immense costs because theres an error?
22:45.58Jammyjluk: did that as well ...lol
22:46.03jontowsaftsack; monitor your server for a while.. flowchart your dialplan by hand so you know what paths can be taken, do a monthly audit until you're comfortable.....etc
22:46.12jlukJammy: what distro you using ?
22:46.42Jammydebian ... sarge me things
22:46.44saftsackjontow, can i call the telephone provider and asking if theres an open connection?
22:46.59jontowi .. suspect so, but that definitely depends on how nice your provider is to you..
22:47.10jontowbest use 'show channels' and watch your CDR closely.
22:47.19jlukJammy: ok, that's exactly what I've just built on :)
22:47.26JFPMartinjontow: sorry not using ooh323. I've been doing my work with ast_h323 so I can handle the RTP.
22:47.32Jammy:) great
22:47.34saftsackmy biggest fear is, that if someone stops a call and hangs up and asterisk didnt get it and the call is never ended
22:47.45jontowJFPMartin: ast_h323 ?  the one builtin to the asterisk SVN tree?
22:47.58JFPMartinjontow: yep
22:48.18Jammyjluk: this isnt the first time ive installed a* well maybe first time on debian, but this is weird that everything works fine, but the modules never show up
22:48.25Jammythink i should try 1.2 ?
22:48.36jlukJammy: I installed the newest 2.6 kernel (with apt-get) then svn fetched the newest 1.2 asterisk & zaptel stuff
22:49.03Jammyim running 2.4
22:49.07trixterin ael is there a way to include other ael files so you dont have to have one massive file?
22:49.13saftsackjontow, do you have an idea?
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22:49.33jlukJammy: ahh, I've not installed with 2.4 for a long long time.
22:49.42Jammyill try the 1.2's
22:49.48Jammysee if that makes a difference
22:49.51jontowsaftsack: could always try autofallthrough, though i don't much like it..
22:50.14saftsackwhat is autofallthrough?
22:50.14jlukJammy: worth trying - it's not as if zap takes a long time to build.
22:50.31eKo1i don't understand why all my calls through zap never show that they are ringing when indeed they are
22:51.22jlukJammy: mine is working now - well at least the module has loaded. Now to see if I can remember how to build a dial plan - I haven't had to touch mine in 7 months or more.
22:51.51Jammyhaha...good luck... ill see if i can make progress...then we can fiddle through these dialplans together...lol
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22:52.17saftsackjontow, :( ^^
22:52.23JunK-Ytrixter: #include?
22:52.33jontowsaftsack: check the example (included) extensions.conf -- it explains it at the top
22:53.06JFPMartinjontow: why'd you decide to use ooh323?
22:53.09jlukJammy: dial plans used to be pretty straight forward - I'm just wondering what might have changed. This box is only going to be an iax link people who work for me so it should be pretty easy
22:54.08saftsackjontow, where is an example? :( *gg*
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22:54.31trixterJunK-Y: does that work with ael?
22:54.40Jammyjluk: yeah it should be... i had a massive dialplan a year ago. then that machine crashed. and all my knowledge went up in smoke
22:55.04JunK-Ytrixter: polly
22:55.07jlukJammy: ahh, backups :D - my dial plan is simple.
22:55.08JunK-Yprolly
22:55.17RenacorAnybody know of any problems with the soundstation 4000 and getting configs and bootroms over tftp?
22:55.35trixterI am guessing no but I will try it
22:55.58jontowbecause it was written in C and i can understand the code ... and it would compile
22:56.10jlukJammy: calls an AGI which looks up everything in enum and away it goes. To add a new extension all I add is the extension to my enum tree
22:56.39JunK-Ytrixter: let me know how it goes.
22:57.13JFPMartinjontow: :-) I've been adding video support to the ast_h323 and the c/c++ can get to you. Does * get access to the rtp in ooh323? I was under the impression it didn't.
22:58.12trixternope it doesnt work
22:58.19Jammyjluk: nice, i've haven't worked with asterisk constantly in a while, in 2003 i was capable of doing something like that. now i have to start from scratch
22:58.31trixterit appears that with ael you have to make one large file and cant create others to be included for easier management
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22:59.43JunK-Yu use includes keyword ?
23:00.15jlukJammy: join the club - the AGI was written ages ago, if I have to alter it I have to start from scratch and work out wtf I did.
23:00.32trixterthat is for other contexts
23:00.34trixternot files
23:01.16JunK-Ylet me grab some food and i'll give a quick test.
23:01.59saftsackjontow, ok autofallthrough is an option to set, but i havent asterisk here. can you copy me the explaining entry to pastebin? :>
23:02.01trixterbtw I have a couple tickets to etel for $200 if anyone wants to go cheaper than the stock price of $1800..  http://conferences.oreilly.com/etel
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23:04.00jontowsaftsack; can't checkout a copy from CVS ?  or look on the wiki?
23:04.09saftsackyes or so
23:04.22saftsackbut i have to do make samples or do you know where they are exactly?
23:04.23jontowJFPMartin: honestly, i'm not fully sure at this point -- i'm rather new to H.323, having been thrown into it this week
23:04.31jontowasterisk/configs/extensions......
23:04.31jontow:)
23:04.48jontowJFPMartin: have learned an awful lot about an awful large number of protocols.. its just another one, i think ;)
23:04.50saftsackthanks :)
23:06.25JFPMartinjontow: H.323 is a minefield. Asterisk's support for it is v.simple. We've been shipping H.323 videophones for a few years now and we're still fixing interop issues with wannabees.
23:07.03jontowheheh
23:07.22drumkillaJFPMartin: hey there
23:07.32JFPMartinHi Drumkilla
23:07.47drumkillayou are the one that reported the videosupport bug, right?
23:07.52JFPMartinYep
23:07.53jontowdon't know if you're familiar with it -- but my project is to interface asterisk+somerandomH.323stack to our proprietary enterprise call center system.. CosmoCall Universe
23:08.14drumkillaJFPMartin: I saw later that you said you had code to implement videosupport per-peer?
23:08.19jontowi've tried woomera's stuff, ooh323c, and ast_h323
23:08.19drumkillais that without the global setting?
23:08.23saftsackjontow, autofallthrough RULES ;)
23:08.27jontowall with similar but not quite successful results
23:08.32JFPMartinJontow: Don't know about CosmoCall but I've been playing with 323 for a while.
23:08.36jontowsaftsack: in certain cases, i'll agree ;)  also makes you lazy, heheh
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23:09.00saftsackjontow, yes but its better so than having a worse telephone bill
23:09.03jontowJFPMartin: they've been around for a while.. one of the big ones, really.. but their platform is windows based, and so i don't much like it.. :)
23:09.31jontowbut, the problem is.. its VoIP only, and it needs some means to terminate from and originate to the PSTN..
23:09.36JFPMartinDrumkilla: yes I have the code fixes but you rather shut the door on my discussion with Twisted :-( I was hoping to find him here to talk it through.
23:09.47*** join/#asterisk RoyK (n=roy@55.80-202-161.nextgentel.com)
23:10.04drumkillaJFPMartin: I appologize, that's why I was glad I caught you
23:10.16drumkillaI didn't want to end any discussion, I just wanted to seperate the issues ...
23:10.26jontowcurrently we've got a proprietary (flaky) Arelnet PSTN<-->VoIP gateway, not preferred... ;)  another choice is a pair of cisco 3640's with PRI cards, they handle 3 PRIs each physically, and 11-15 calls simultaneously each last test we ran, we have 48 channels to terminate at a minimum.. :P  they'll not do very well
23:10.36drumkillafixing the videosupport is one thing, and then you had some other stuff as well
23:10.54katakefalosHello all, does anyone know how to raise the incoming volume on remote sip extensions somthing like rxgain and txgain in zap?
23:10.56drumkillado you think you could submit 2 patches, one for fixing videosupport, and one with your other changes?
23:11.07drumkillaI'd be happy to review the video support stuff
23:11.13JFPMartinJontow: I played around with H.323 and gnugk interfaced to AAH. * can be a good solution.
23:11.14*** join/#asterisk A-Tuin (n=a-tuin@steves.ip.v4.me.uk)
23:11.18jontowbut our best choice is asterisk, really.. quad cards are beautiful things ;)  and since i'm well versed with a local CLEC's PRIs .. i'm pretty confident if i can hook the two together, it'll be a winning match
23:12.23JFPMartinDrumkilla: thanks for the offer. I'm finding it tough to find people that'll get interested in video. And I appreciate your intension to separate the issues.
23:12.52drumkillaJFPMartin: we just have a ton of issues on there, so i'm trying to help get things the way that will get them processed the most quickly
23:13.05drumkillaJFPMartin: I would love to do more work on video, I just have no access to resources to test anything
23:13.14JFPMartinDrumkilla: I was finding it difficult to keep up with the SIP changes and the thought of doing another diff to only sort out the videosupport issues 5 mins before leaving for hols was too much :)
23:13.54drumkillaJFPMartin: ok, so would you like to revisit it after the holidays?
23:14.34JFPMartinDrumkilla: I think that was twisted's problem. He went through all the tests for peer caps but without trying to make a video call you couldn't see that the rtp wasn't up.
23:14.46drumkillaJFPMartin: exactly
23:15.00drumkillaand looking at the code, there wasn't an easy fix to make it work as expected
23:15.08drumkillaso I just reverted it for now ...
23:16.05JFPMartinDrumkilla: I guess I'll have to get back into it in the NewYear. There's a simple fix to always create the rtp for video but I'm worried about voice systems that are already creaking and me adding extra resource problems with the always approach.
23:16.29drumkillayeah, i'd rather not have to do that ...
23:17.09drumkillalike you said, the copy could have been moved up, but that requires it being set globally
23:17.17drumkillaand still doesn't even look at the peer settings
23:17.26drumkillathat function doesn't look at peer settings at all
23:17.27JFPMartinDrumkilla: The right solution is to only create the vrtp when we really know the cap of the peer/user. Though this is a big enough change that I want to test it for a few days.
23:17.37drumkillaJFPMartin: agreed
23:18.05drumkillaJFPMartin: I'll reopen the original bug for this so we can continue discussion there
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23:18.18JFPMartinDrumkilla: thanks.
23:20.21drumkillaJFPMartin: 5427 is now re-opened
23:20.51JFPMartinDrumkilla: I see the bug open again - thanks. I'll get to thinking about it over Turkey and let Mantis know what the bird says :-)
23:21.12drumkillaJFPMartin: haha, ok.  I appreciate your help
23:23.16*** join/#asterisk A-Tuin (n=a-tuin@steves.ip.v4.me.uk)
23:23.31*** join/#asterisk Strom_C (n=strom@198.172.114.2)
23:24.38Strom_Cthis is the last time I let a client specify that they want AMP on their goddamned asterisk box
23:24.45Strom_Cthe fucker won't install.
23:26.29Strom_Chave any of you ever actually gotten it to successfully install?
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23:33.10jontowg'night all :)
23:37.17xhelioxSure is quiet around here.
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23:38.57easelis there an asterisk command to 'barge into' a zap channel, regardless of if anyone else is connected to it?
23:42.15ManxPowereasel, no.
23:42.36ManxPowerthere is a command to LISTEN to a zap channel.
23:42.42easelyeah, i just found zapbarge
23:42.45easelbummer
23:43.00easelso theres no way to duplicate having two extensions on the same analog line then...
23:44.04ManxPowereasel, Well, you can physically connect two analog phones to an analog zap port.
23:44.36easelhmm yeah i'm trying to do it the other way though... i want to use a zap port as one of two analog phones connected to a line
23:44.59easeli've got the ringing in part working ok... but i just can't figure out how to 'pick up' the line if it isn't ringing
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23:47.21gcdtechREGISTER
23:48.01guest_must go today!  2 alienware laptops price $500 for one, 750 for two. message me on aim at mikcomputing, msn at heymikeeh@hotmail.com or yahoo at mperkelay if interested and wanting to buy only!!
23:48.10file[laptop]eh?
23:48.16rob0is that a spam?
23:48.46rob0sure looks like it
23:49.08jlukdid someone remove the requirement to register ?
23:51.26SkramX:)
23:52.06file[laptop]he he he...
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23:55.28tasatanyone here with experience configuring asterlink?
23:56.42rob0haha I bet there is
23:57.04rob0maybe even someone who WORKS for asterlink
23:57.29tasatthen I've come to the right place...
23:57.46*** join/#asterisk backblue (n=moo@87-196-12-38.net.novis.pt)
23:58.23tasatTrying to get incoming calls working... I call my number and just hear a three tone sequence...
23:58.26loudactualy no, try #asterlink maybe.
23:58.27rob0but anyway, the Wiki has anything you need, in general
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