irclog2html for #asterisk on 20051208

00:00.04Qwellshmaltz: sounds like a perfect time to upgrade then
00:00.12Qwellby the time you're done compiling, he'll be off the phone
00:00.16drraymy bad
00:00.30shmaltzQwell, that aint an option for tonigh :(
00:00.45shmaltzI first have to redo my DP to work nicely with 1.2
00:00.59shmaltzanybody know of any solution that works like app_valetparking?
00:01.06*** join/#asterisk Ridgeback (n=Ridgebac@104.243.8.67.cfl.res.rr.com)
00:01.10Qwellapp_valetparking does
00:01.10drumkillashmaltz: oej is working on one
00:01.14shmaltzapp_valetparking doesn't compile on 1.2
00:01.17Goshenperfect, I changed the name of vm-intro.gsm so it couldn't find it, but it still makes a loud screech, is there any way to remove that?
00:01.22shmaltzdrumkilla, that is great news
00:01.39*** join/#asterisk Sedorox (i=brandon@smartserv/cna/Sedorox)
00:01.40drumkillahe has a branch in svn for it
00:01.48drrayisn't the screech ADSI?
00:01.51Goshen'BEEP' 'comedian mail'
00:01.51drumkillai have no clue how far along it is
00:01.55*** join/#asterisk santiago (n=santiago@208.195.215.160)
00:02.01Goshenadsi?
00:02.12alephcomGoshen: http://www.nathanpralle.com/software/ast_masterlist.html
00:02.29Ridgebackhello
00:02.48alephcomthey don't list that file.  Sorry.
00:03.03Goshenit doesn't list it as playing a file when it makes the beep
00:03.57shmaltzwow, that trucking a55hole is still on the phone
00:04.16drumkillashmaltz: restart when convenient is your friend
00:04.56drray4554013
00:05.02drumkillashmaltz: sounds like a plan!
00:05.17drumkillabarge and drop it into a meetme so we can all listen
00:05.28alephcomQuick question...  I have a friend who has slapped together a php interface that interfaces with the -realtime database.  It is a gui but is used mostly/only to edit the -realtime database.  Is there any interest in something like this?
00:05.34Ridgebackjust curious will my old pre 1.2 config files move right over to 1.2? I read the changes doc, but it didn't seem to say there would be much to change.
00:05.45shmaltzdrumkilla, after all he is calling south america, where lots of these so called suck operations exists (I took the name from the list) :)
00:05.56drumkillaRidgeback: they should, yes
00:05.56shmaltzdrumkilla, that is realy a good plan :)
00:05.59shmaltz:P
00:06.02drumkillaRidgeback: you may see some deprecation warnings, though
00:06.06Ridgebackdrumkilla thanks
00:06.08drumkillabut the same options should still work ...
00:06.25Ridgebackdrumkilla: ok cool ill give it a try on the weekend
00:06.28ManxPowerRidgeback, In theory they should work just fine.
00:06.31*** join/#asterisk dwmw2 (i=ctrlprox@baythorne.infradead.org)
00:06.40drumkilla*should*  :)
00:06.50shmaltzRidgeback, what out for the j option
00:06.54RidgebackManxPower: good deal. was just worried my dundi stuff would get all dorked up
00:06.54drumkillathere was way too much time in between 1.0 and 1.2
00:06.55ManxPowerIf you are using ENUMLookup and not prefixing the query with a + then that will break
00:06.58shmaltzthat is the only think you realy want to know about
00:07.00drumkillaso it's quite a massive difference in code
00:07.24ManxPowerdrumkilla, and yet, 1.2 still doesn't have a SIP jitterbuffer......
00:07.44Ridgebackhave you guys played with the IAX encryption yet?
00:07.49drumkillaI didn't know you could have jitter in SIP signalling
00:07.57drraySSL tunneling?
00:08.12*** join/#asterisk kokey (n=ubunture@201.153.63.79)
00:08.19*** join/#asterisk dwmw2 (i=ctrlprox@baythorne.infradead.org)
00:08.23Ridgebackdrrayi thought 1.2 was to have AES encryption on IAX:
00:08.32shmaltzthis guy is out of his mind, look at this:
00:08.33shmaltzhttp://pastebin.com/453291
00:08.56drumkillarobin_sz: ot dpes
00:08.59drumkillacrap
00:09.00GoshenI keep getting this after upgrading to 1.2.1 from 1.0.7
00:09.01GoshenDec  7 17:08:00 WARNING[28113]: chan_sip.c:9601 handle_response_register: Got 200 OK on REGISTER that isn't a register
00:09.02drumkillaRidgeback: it does
00:09.05drumkillaI totally messed that up
00:09.39Ridgebackdrumkilla: cool stuff cant wait to try it out
00:09.55ManxPowerGoshen, your extensive search of the mailinglist archives did not help?
00:10.05*** join/#asterisk dwmw2 (i=ctrlprox@baythorne.infradead.org)
00:10.08ManxPowerdrumkilla, Please step away from the beer. 8-)
00:10.15drumkillaManxPower: none this evening :)
00:10.35darwin_35the only issue I have had is this stupid  spandsp issue
00:10.36drumkillabut I was at school all day ... same thing
00:10.40drumkillaThanks MstlyHrmls !!!
00:10.42darwin_35and I will get it working
00:10.49MstlyHrmlsobviously  you need one :-)
00:10.49darwin_35grrr
00:10.53drumkillano kidding
00:11.08ManxPowerdarwin_35, Got the 1.2 version of rx_fax, tx_fax now?
00:11.08drumkillaoh wait!!
00:11.13drumkillaI have one left!
00:11.28ManxPowerdarwin_35, We all do, dude, we all so.
00:11.29drraywith a happy ending
00:11.29darwin_35yes
00:11.35drumkilladarwin_35: me too ... my back is *killing* me for some reason :(
00:11.37GoshenManxPower: Just that the message is coming back too early or late or something like that, but doesn't give a fix
00:11.48darwin_35mine is spazming
00:11.48*** join/#asterisk dwmw2 (i=ctrlprox@baythorne.infradead.org)
00:11.49GoshenManxPower: It didn't give me that message before the upgrade
00:11.57*** join/#asterisk Flauto (n=zhao@c-24-14-197-214.hsd1.il.comcast.net)
00:12.10ManxPowerGoshen, I don't think the older version cared.
00:12.28darwin_35I have to compile spandsp 0.0.2pre21c
00:12.34ManxPowerGoshen, you don't have something silly like pedantic=yes do you?
00:12.35darwin_35but I have to patch it first
00:12.50GoshenManxPower: in sip.conf?
00:12.56darwin_35doing it now
00:13.26ManxPowerdarwin_35, I can tar up my asterisk, asterisk-sounds, zaptel, libpri, spandsp, with all the patches to load the apps in asterisk.  Oh and NXFaxDetect is included.
00:13.31GoshenManxPower: I didn't put predantic=no in sip.conf because I read that the default is no
00:13.33*** join/#asterisk dwmw2 (i=ctrlprox@baythorne.infradead.org)
00:13.49ManxPowerGoshen, the default is no
00:14.12Goshenno, I don't have predantic=yes in my sip.conf
00:15.18*** join/#asterisk dwmw2 (i=ctrlprox@baythorne.infradead.org)
00:15.29darwin_35manx that would rock
00:16.00drumkillashmaltz: maybe he left it off hook just to piss you off
00:16.01darwin_35no need for zaptel or libpri
00:16.30test34Can someone control asterisk from the internet and have asterisk call him and another person with VOIP 3-way to save on long distances ?
00:16.39*** join/#asterisk SPAD (n=chatzill@specdl05.cul.columbia.edu)
00:16.55*** join/#asterisk upsite (n=upsite@wls.swh.uni-halle.de)
00:17.02*** join/#asterisk dwmw2 (i=ctrlprox@baythorne.infradead.org)
00:18.12ManxPowerdarwin_35, hold on
00:18.45*** join/#asterisk dwmw2 (i=ctrlprox@baythorne.infradead.org)
00:18.52Goshenany ideas for getting rid of the Dec  7 17:16:01 WARNING[28113]: chan_sip.c:9601 handle_response_register: Got 200 OK on REGISTER that isn't a register
00:19.07GoshenPredantic=yes is not set
00:20.13ManxPowerGoshen, file a bug, then someone can tell you why it's not a bug 8-)
00:20.19Goshenlol
00:20.31Qwelland lose 2 karma points
00:20.31ManxPowerI wasn't joking.
00:20.31*** join/#asterisk dwmw2 (i=ctrlprox@baythorne.infradead.org)
00:20.32*** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl)
00:20.36Qwelldwmw2: stop that
00:20.40puzzledevening all
00:20.45Goshenwell, it was misconfigured a couple hours ago
00:20.49Qwellor not
00:20.54Goshenperhaps it is stuff coming back from that...
00:20.59Goshensome kind of crazy time warp
00:21.12Goshenwho knows, I will deal with it again tomorrow if it keeps it up
00:21.17shmaltzDrumkilla, he hung up now, he didn't leave it off hook because the system charges him, 1 sec checking............................
00:21.59alephcomtest34:  Short answer, yes.  Long answer it might be a bit of work.
00:22.09ManxPowerdarwin_35, http://www.fnords.org/~eric/asterisk.tar.bz2
00:22.14*** join/#asterisk dwmw2 (i=ctrlprox@baythorne.infradead.org)
00:22.24ManxPowerThis is the beginning of the -BTEL fork of Asterisk
00:22.47mog_work?
00:22.48test34alephcom, do you know of anybody that did it and posted a howto on the web ? ;)
00:22.55mog_workanother....
00:23.21Qwelltest34: seems easy
00:23.35Qwellwrite a simple script to drop a call file...
00:23.54mog_workor use manager
00:23.59*** join/#asterisk dwmw2 (i=ctrlprox@baythorne.infradead.org)
00:24.05test34Qwell, the problem is I'm new to asterisk..
00:24.11tzangermog_work: any good at threading code?  I have a question
00:24.13Qwelltest34: nothing a bit of reading won't fix
00:24.19mog_workdecent
00:24.25Qwellmog_work: anybody around that can "fix" dwmw2 here?
00:24.26mog_worki dont always do it right thouhg....
00:24.31Qwelland -dev
00:24.42mog_worki dont have ops anymore....
00:24.48alephcomtest34:  have you looked in the wiki?  I know it's been done but....
00:24.51tzangermog_work: http://pastebin.ca/32897
00:25.06shmaltzDurmkilla, that phone call will cost him $37.80
00:25.12tzangermog_work: I'm getting alarms more or less every 100ms as expected, but every so often my alarms come every 15-20ms for a few, then back to 100ms
00:25.31tzangermog_work: now I can see it SKIPPING now and again if it couldn't get the lock... but it's not
00:25.47test34Qwell, I will look into dropping a call file then
00:25.52*** join/#asterisk dwmw2 (i=ctrlprox@baythorne.infradead.org)
00:25.54test34alephcom, didnt find anything yet
00:26.05alephcomtest34:  Call files are actually quite handy and easy to use.
00:26.15ManxPowermog_work, I want to take 1.2.3, put in RTP jitterbuffer, the self timing RTP patch, spandsp, the NV*Detect stuff.
00:26.28*** join/#asterisk Pete_Largo (n=PeteLarg@225-196.35-65.tampabay.res.rr.com)
00:26.47Pete_Largotzanger: you around?
00:26.56tzangeryeah
00:27.22Pete_Largoit's been working like a champ since that day.  Thanks again for all your help :)
00:27.29tzangerPete_Largo: no problem at all
00:27.32tzangerglad it's working :-)
00:27.37*** join/#asterisk dwmw2 (i=ctrlprox@baythorne.infradead.org)
00:27.45Pete_Largohehe me too!
00:28.04drumkillamog_work: "fix" ?
00:28.10Pete_Largothat threading problem was making all kinds of trouble
00:28.14Qwelldrumkilla: kb dwmw2, would ya? :p
00:28.27drumkillawill do when he joins again
00:28.32Qwellhe's been on and off about 15 times now, in the last 15 minutes
00:28.49*** join/#asterisk Zodiacal (i=hehehe@bdsl.66.14.242.199.gte.net)
00:29.07Qwelldrumkilla: -dev too...
00:29.10*** join/#asterisk anthm (n=anthm@000-435-220.area4.spcsdns.net)
00:29.10*** mode/#asterisk [+o anthm] by ChanServ
00:29.18docelm0whadup?
00:29.21*** join/#asterisk dwmw2 (i=ctrlprox@baythorne.infradead.org)
00:29.25ManxPowerTHERE HE IS!
00:29.25*** mode/#asterisk [+b *!*=ctrlprox@*.infradead.org] by drumkilla
00:29.26*** kick/#asterisk [dwmw2!n=russell@asterisk/developer-and-stable-maintainer/drumkilla] by drumkilla (drumkilla)
00:29.27Pete_Largodocelmo!  hey
00:29.31ZodiacalHi, can i use an asterisk box as just a voice mail system? i have an existing office Merdian pbx?
00:29.42drumkillagot him!
00:29.43Zodiacaland have the phones allready etc..
00:29.45darwin_35manx you there
00:29.46Qwell<3 <3
00:29.53shmaltzwell, after all this I'm happy to report:
00:29.55shmaltzjumping channel 69 (adit fxs) from span 3 to channel 73 of span 4 (adit fxo), confirmed that the new (for me used for someone else) fxo card I got for $200.00 realy works, with callerid and hangup detection :)
00:29.57Pete_Largogood job drumkilla!
00:30.12drumkillayay
00:30.40drumkillaI'm drinking a Grolsch right now, hehe
00:30.49Qwellwtf is that?
00:30.59bweschkegood beer!
00:31.04anthmshuddup?
00:31.05drumkillaDutch
00:31.12bweschkedrumkilla!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!1
00:31.12anthmyou dont know what grolsch is ?
00:31.21tzangert's an ok beer
00:31.34ManxPowerThere is no beer better than Guinness.
00:31.37Zodiacalany ideas?
00:31.46docelm0Whadup?!!?
00:31.47darwin_35yes there is
00:31.51anthmahh the only thing better than a db or os zealot is a beer one!!!
00:31.55puzzledManxPower: they really brainwashed you didn't they :)
00:31.57bweschkeagree... unless of course it's a black and tan w/ Bass and guinness. :)
00:31.58bweschkebrilliant!
00:32.03darwin_35HappyDog Brew from RockBottom Brewry
00:32.03Qwellanthm: you missed the former two earlier
00:32.16drumkillaQwell: isn't that a daily occurance?
00:32.16docelm0Supe Pete?
00:32.25Qwelldrumkilla: well...yeah
00:32.30ManxPowerpuzzled, I sampled random Dutch and Belgian beers when I was in Europe.  Guinness is better.
00:32.33drumkillathat's why I tend to ignore this channel :-p
00:32.34anthmit's ok another one will come around in an hour or 2
00:32.42drumkillabweschke: nice
00:32.46Qwellanthm: yeah...we all know postgresql sucks
00:32.50Pete_Largodocelmo, what's with the 0 ?
00:32.59puzzledManxPower: sampling is good. I do that too
00:33.04bweschkeqwell: oh no u didn't
00:33.06bweschke:)
00:33.21drumkillaI'll tell you what else sucks, Linux ... and *BSD
00:33.37anthmnow your on the right track
00:33.42bweschkelol
00:33.46ManxPowerUsers suck.
00:33.51drumkillatotally
00:33.54alephcomlol  Amen!
00:33.54bweschkeding! ding! ding!
00:33.57*** part/#asterisk SPAD (n=chatzill@specdl05.cul.columbia.edu)
00:34.00anthmit's computers in general and everything associated with them that sucks =D
00:34.09drumkillaI'm so against users, that I don't even use the software I work on
00:34.17docelm0Just to make it a bit different for all my locations..
00:34.40Pete_Largolol, docelm0, just so no-one can find you!
00:34.48shmaltzanthm, you got my /msg?
00:35.00docelm0Well msg both of me..  I will answer sooner or later
00:35.04anthmwhich?
00:35.08anthmwas it in an email ?
00:35.13shmaltzok, both
00:35.17shmaltzan email and msg
00:35.25shmaltzemail to your yahoo account
00:35.28Pete_Largodocelm0 - that reminds me of a poem...
00:35.44docelm0ohh lord..
00:35.46anthmi am on an cheezy irc that doesnt mak a big deal ouyt of pm i must have missed it
00:35.47Pete_Largoroses are red, violets are blue, I'm a scizophrenic, and so am I
00:35.52anthmi do see 1 that says "did you get my message"
00:36.08shmaltzanthm, exactly that /msg i meant
00:36.13shmaltzbut I'm asking about the email
00:36.16shmaltzdid you get it?
00:36.22shmaltzfrom shmaltz@gmail.com
00:36.22anthmlet me look
00:36.26shmaltzthanks
00:36.28*** join/#asterisk EriSan (n=erisan@81-174-42-85.f5.ngi.it)
00:36.40*** join/#asterisk Godsey (n=admin@pdpc/supporter/sustaining/Godsey)
00:37.00upsitehey is someone using call pickup via the quickdial-buttons on a snom (220)
00:37.14shmaltzupsite, I'm sure that yes
00:37.20upsitehmm
00:37.29upsitemine is'nt working
00:37.43darwin_35Manx I will chat with you tomarrow
00:37.44upsiteits sending a notify instead of an invite
00:37.46darwin_35going home
00:37.50*** part/#asterisk darwin_35 (n=darwin35@sta-208-139-193-162.rockynet.com)
00:37.52shmaltzupsite, do a sip debug in asterisk press that button and pastebin your cli output
00:38.21upsiteok
00:38.36QwellI should hack up the extra softkeys on a 7940/60/70 with chan_sccp
00:38.56shmaltzQell, you using sccp?
00:39.02Qwellshmaltz: yeah, I love it
00:39.03shmaltzand it works nicely?
00:39.08Qwellworks great
00:39.14shmaltzwhy is it better than sip?
00:39.20Qwellhe'll be adding my reload and realtime patches soon...
00:39.26Qwellshmaltz: sccp is just...sexy
00:39.38Qwelland runs on 7970s ;]
00:39.42shmaltzQwell, that will just not let me do any work :P
00:39.50anthmdoes it not work anymore?
00:39.53shmaltz7970 doesn't support sip?
00:39.56Qwellnope
00:39.58ManxPowerpuzzled, my version of "sampleing" is to tell the bartender, "give me one I've not had before"
00:39.59shmaltzanthm, nope not in 1.2
00:40.31anthmstarbucks is daft they want 6 bux an hr for net access good think i have a sprint card sheesh
00:40.45Qwell$6/h?  damn
00:40.47puzzledQwell: which chan_sccp? the one from http://chan-sccp.berlios.de ?
00:40.52Qwellpuzzled: yeah
00:41.03shmaltzanthm, it's only round $3 here (NJ)
00:41.04Qwellit really does work very well
00:41.16Qwellwe're gonna go production with it
00:41.19mog_workhey what happened to that fork up above
00:41.22mog_workthe btel one
00:41.26anthmis head from this week close enuf to 1.2 for you ?
00:41.45puzzledQwell: I heard that one is working really well. yesterday I made an rpm of the latest release so I'm prepared when I get a few 7961's
00:41.57Qwellrpm?  eww
00:42.06Qwellpuzzled: I'd repackage it in a few days..
00:42.18Qwellfun new stuff should be coming "Real Soon" now
00:42.27puzzledok will track it. thanks
00:42.37shmaltzanthm, the problem is that I cannot take a chance to run head on this machine, I have to run stable, if HEAD from this week will allow it to compile on 1.2 then it's good enough for me
00:42.51Qwellshmaltz: I think thats what he was asking
00:42.56anthmi mean compat wise
00:43.07shmaltzI have been running HEAD on it until now, and I had too many crashes
00:43.30anthmwhat doesnt work about it ?
00:43.44shmaltzI'll give you the output of astxs -install in a sec
00:44.08anthmmy output is a newline
00:44.14anthmso you probably want my latest copy
00:44.52shmaltzhere:
00:44.54shmaltzhttp://pastebin.com/453349
00:45.10shmaltzanthm, I guess so, I have the one from bpxfreeware
00:45.22QwellYou need to include stdio.h before file.h :p
00:45.25anthmyah when did you dl it ast
00:45.25anthmlast
00:45.27Qwellsimplest fix ever
00:45.39anthmid try it fresh
00:45.50anthmyou know you can astxs urls to .c files too
00:45.52shmaltztoday was when I downloaded it last
00:46.09shmaltzanthm, I know, someone Tony told me about it, a while ago ;)
00:46.51orlokhmm, wtf. my 7940's red light is staying on
00:46.53drumkillaanthm: really?  that's hot
00:47.01Qwellorlok: the voicemail light?
00:47.25orloki think of it as the ring light
00:47.29orlokbut yes, that would be it
00:47.44QwellSo...you have voicemail
00:47.45shmaltzanthm, just donloaded it again, and still the same problem, FYI I did:
00:47.46shmaltzwegt http://www.pbxfreeware.com/app_valetparking.c
00:47.58*** join/#asterisk hhoffman (n=hhoffman@tor/session/x-2209c9215ad3a296)
00:48.07hhoffmanhi
00:48.23shmaltzorlok, just change the sip setting mailbox to a mailbox that doesn't have any messages, leave a message in that mailbox, then erease it
00:48.28shmaltzhi hhoffman
00:48.29anthmaha
00:48.37anthmthe good one wasnt in the url
00:48.50shmaltzanthm, so whre is it????????????
00:49.01anthmtry again
00:49.39anthmit was a stale copy
00:50.10*** join/#asterisk YoMama (n=tchen@pcp02689850pcs.roylok01.mi.comcast.net)
00:50.20YoMamahey all
00:51.01YoMamaif i'm calling in thru a sip proxy...why would I not be able to hear the * server's AA, but the * can hear me?
00:51.11shmaltzanthm, ya da man :)
00:51.13YoMamawhich is opposite to what i thought the problem would be
00:51.16orlokahh, asterisk wasnt running
00:51.27Zodiacalanyone know if i can use asterisk as just a voice mail system with my existing merdian pbx and phones?
00:51.31YoMama'cause if it were a firewall problem....then i'd be able to hear *
00:51.42YoMamaZodiacal: yes...theoretically
00:52.13shmaltzthank you everyone, gtg now
00:52.15anthmYoMama I think the official name for it is RTP/NAT HELL
00:52.17shmaltzthank you anthm
00:52.18Zodiacalyomama or would it be even easier to replace the pbx as well as the voice mail system with an * box? but i would like to use my existing phones....
00:52.22anthmnp
00:52.24YoMamaZodiacal: look in asterisk-wiki...there's some examples of it in there
00:52.35Zodiacalyomama wikipedia?
00:52.40YoMamaanthm: i don't think it's a firewall problem...
00:52.46Zodiacalor does * have its own wiki
00:52.50Zodiacalgot the url off hand?
00:52.55YoMamaZodiacal: www.voip-info.org
00:53.02drumkillathere is a wiki built into asterisk!
00:53.12Zodiacalyomama Thank You!
00:53.15anthmsomeone cant hear someone on rtp it's a port mismatch or the wrong ip or something along those lines
00:53.40Zodiacalyomama i don't wanta use voip tho.. just existing verizon business plan
00:57.14upsiteshmaltz : http://pastebin.com/453361
00:57.43YoMamaZodiacal: u don't haveta
00:57.59upsitethats happning when i press the function key while it's blinking to show me that a call is ringing on the other phone
00:57.59YoMamaZodiacal: u could use asterisk only as a voicemail server...how many extensions u got?
00:58.12YoMamaZodiacal: or i should say..how many voicemail ports u think u need?
00:58.22hhoffmanwhat is the "_" for before a extension number? I'm trying to find it in the book but can't
00:58.33YoMamahhoffman: means it's a pattern
00:58.48YoMamahhoffman: www.asteriskdocs.org...read the TFOT book..it explains it all
00:59.04*** join/#asterisk ldnblk (n=Just@optbom1.uk.access.vodafone.net)
00:59.20Zodiacalyomama 9 stations
00:59.24Zodiacal6 phone lines
00:59.38YoMamaZodiacal: ah..u could get away with a TDM400 card
00:59.44YoMamamaybe 2-3 ports for voicemail
00:59.49hhoffmanYoMama: thanks... is the TFOT book the same one as the OReilly book?
00:59.50Zodiacalyomama at least one voice mail port... i think thats what we have now. only one can record voice mail at a time i think
00:59.53Zodiacalbut that systme is so old and its dieing
00:59.56YoMamaget some FXO ports on it
00:59.58Zodiacalsystme = system
01:00.05YoMamaZodiacal: then get 2 or 3
01:00.15YoMama2-3 FXO
01:00.23Zodiacalwill that work with my phones. my phones have lcd screens and line buttons and function buttons etc..
01:00.27Zodiacali.e. paging..
01:00.38YoMamathe problem isn't asterisk..it's setting up meridian to properly signal asterisk to use the VM system properly
01:00.45Zodiacalic..
01:00.50Zodiacalwelp i hate the merdian thing
01:00.55Zodiacalmaybe i should just replace it too
01:00.56Zodiacal?
01:01.00Zodiacalwill * work with my phones?
01:01.01YoMamaZodical: the handsets aren't the problem...it's the programming of your PBX
01:01.18Zodiacalwhat if i use * as the pbx
01:01.19YoMamano it won't..unless your system's phones are analog..but the chances of that are almost zero
01:01.29Zodiacali belive they are digital
01:01.39Zodiacalcuz i can't connect a regular phone to the jack in the wall :P
01:01.41YoMamanope then..those handsets only work wiht your PBX
01:01.49Zodiacalplus they have station names etc on the lcd
01:02.03YoMama9 stations..that's a small one
01:02.04Zodiacaldarn
01:02.30dokhenchzodiacal: you can use them if you put a sip gateway in front.. lemme find the link.
01:03.07YoMamaZodiacal: if u were to replace the whole thing...do u have more than one ethernet run per station?
01:03.16Zodiacalyes
01:03.50YoMamak..then u could replace the system with $150 handsets
01:03.55YoMama$150 x 9
01:03.58YoMamaso a little over a grand
01:04.07YoMamaa box that u were already going to run asterisk on
01:04.15*** join/#asterisk m160858 (n=jsaenz@200.89.12.46)
01:04.16YoMamahow many incoming lines do u have?  DID?
01:04.20Zodiacal6
01:04.37m160858hi, i want to connect one asterisk with other by iax
01:04.47YoMamaZodiacal: u could either get a T1 card and a channel bank..or two TDM400 cards with FXO ports on it
01:05.03YoMamaof course the TDM400's would be cheaper..but won't offer the same amount of expandability
01:05.07Qwellor a TDM2400P with 2 FXO modules
01:05.14YoMamaoh right
01:05.16Zodiacalyomama think it would be cheaper if i just got another startalk voice mail system..?
01:05.16YoMamayeah..those too
01:05.17QwellThat's what I'd recommend
01:05.24YoMamaQwell: i always forget about that card :)
01:05.27dokhenchZodiacal, you can use this to reuse the existing phones http://www.citel.com/products/handset_gateways/index.asp#CITELlink
01:05.59YoMamaZodical: then finally...make sure you do a gap analysis between how your current system is used and what * can offer...* does almost everything that a traditional PBX does, but the handsets make all the difference..some do intercom and paging..some don't..it all depends
01:06.05YoMamaanyway
01:06.05Zodiacaldokhench coolness. do you know how much those cost off hand?
01:06.15*** part/#asterisk m160858 (n=jsaenz@200.89.12.46)
01:06.29*** join/#asterisk nassy (n=nassy@207-38-197-201.c3-0.wsd-ubr1.qens-wsd.ny.cable.rcn.com)
01:06.37YoMamahave fun...i gotta go pay some attention to my fiancee before she gets wise and figures out i'm a lazyass dork
01:06.43YoMamabbl
01:06.58dokhenchZodiacal, no idea
01:08.28Zodiacaldokhench Thank You! i have some stuff to think about now
01:10.06*** join/#asterisk Zach^^ (n=Zachary@65.121.244.130)
01:10.33dokhenchZodiacal, $3120.00
01:10.33Zach^^hello i have so problems trying to get out bound calls to go using voipjet
01:10.53*** join/#asterisk jtodd (n=jtodd@ti.fox-den.com)
01:11.17Zodiacalyikes
01:11.26Pete_LargoZach^^ are you still using the old server?  they changed servers a few weeks ago. problem with their East Coast server I think...
01:11.47QwellZodiacal: look at the tdm2400p
01:11.47Zach^^hmm i just setup my accout yesterday
01:11.56QwellYou'd need 2 FXO modules (which would give you 8 ports)
01:11.58Pete_Largooh
01:12.05Pete_Largothen you are probably using the new server :)
01:12.13Zodiacalqwell will that work with my phones tho?
01:12.25QwellI actually wasn't listening to what you're gonna do
01:12.26Zodiacal:P
01:12.28QwellZodiacal: what phones?  and, phones use fxs anyhow
01:12.34Qwellif they're digital...no
01:12.42dokhenchQwell, not his. he has pbx phones.
01:12.58Qwellhow many phones, how much you looking to spend total?
01:13.20dokhenchif it is 15 or so phones, it'd be cheaper to just get new phones.
01:13.23Zodiacali was hoping i could spend around 1000, 1500.. like for a * box and some cards..
01:13.35Pete_Largoit does support the Nortel Norstar line
01:13.37*** join/#asterisk p1tst0p (n=admin@82-38-106-54.cable.ubr03.donc.blueyonder.co.uk)
01:13.41QwellZodiacal: how many phones?
01:13.49Zodiacal~9 stations
01:13.52Zodiacal6 lines
01:14.09p1tst0pwould it be possible, to display current channels in use through php, would this require use of the Manager API ?
01:14.11Qwellone tdm2400p with 2 fxo, and 9 decent IP phones...could put you at about ....1500-2k?
01:14.11Zach^^anyone able to reach sixtel support?
01:14.27Qwell(plus the server, of course)
01:14.51Zodiacalim getting the idea that its probably best just to replace my existing startalk voice mail. thats the thing thats failing..
01:15.01Qwellwhat
01:15.05Qwell'll it cost to replace that?
01:15.16Zodiacaldunno exactly
01:15.19Qwellit may make fiscal sense in the short run, but...
01:15.26QwellYou'd be stuck with a junk pbx.  heh
01:15.34Zodiacali just wanted to see what * had to offer in features compaired to price...
01:15.43QwellZodiacal: can't beat free
01:15.56Zodiacal:P but the hardware costs bux
01:16.09Pete_LargoZodiacal: what model is your voicemail?
01:16.23Zodiacal1 sec
01:16.50ZodiacalStartalk Mini (NOR-015) ?
01:16.53Zodiacalis that right?
01:16.56Zodiacali just have some manuals for it
01:17.03Zodiacalit doesn't say much on the box it self.. just startalk
01:17.15ZodiacalStartalk Flash
01:18.48Pete_Largoyou could probably get one refurbished for around 1500
01:18.55Pete_Largothat's a guesstimate
01:19.01dokhenchZodiacal, the digium hardware is inexpensive compared to some of the stuff out there. pbx can cost $$$.
01:19.15QwellZodiacal: at that price...you could get new hardware with *
01:20.26Zach^^i get all circuits are busy no when i try to call out
01:20.53*** join/#asterisk m160858 (n=jsaenz@200.89.12.46)
01:20.55p1tst0panyone know if if/how i can display current channels in use through a php interface, would this require use of the Manager API ?
01:20.59m160858hi
01:21.24m160858excuse, but how i can register one asterisk local with other in the WAN?
01:21.48m160858on iax
01:21.58*** join/#asterisk wundaboy (n=asdf@c-67-164-107-68.hsd1.or.comcast.net)
01:23.09m160858newly, all are sleeping? :)
01:23.55wundaboyso, this may sound dumb but when im editing extconfig.conf where do i put my mysql credentials?
01:24.07Qwellwundaboy: in the mysql config
01:24.35wundaboyQwell: which file is that?
01:24.37wundaboymysql.conf?
01:24.55wundaboyoh
01:24.58wundaboyres_mysql.conf
01:24.58QwellI'm just guessing here, but res_config_mysql?
01:24.59wundaboymybad
01:25.02p1tst0p<PROTECTED>
01:25.04p1tst0pyeh
01:28.51*** join/#asterisk Seldon1975 (n=someone@gatekeeper.radintl.com)
01:29.00Seldon1975hi all
01:29.03m160858anybody?
01:29.40Seldon1975I have my asterisk server up with two Polycom Soundpoint 501s on the network - can someone tell me how to assign them Extension numbers in * config
01:29.47Zach^^i get all circuits are busy no when i try to call out using voipjet... anyone help?
01:31.40m160858i want to know register one local asterisk with other asterisk in the WAN
01:31.47m160858it's possible?
01:32.20alephcomIt definitely should be.
01:32.30nassyZach^^: im new to asterisk and i get that also. i have a VoIP provider also. it happens to me when i dont dial the correct number. ie, i forget to dial a 1 infront of the number
01:32.48m160858how?
01:33.01alephcomWhat are the details of the setup?
01:33.11m160858yes
01:33.22Zach^^nassy who is your voip provider?
01:33.40nassytelasip
01:33.46m160858i want to connect my ATA's to my asterisk local with ullaw
01:34.05alephcomThat should not be a problem
01:34.13SkramXanyone connected asterisk with ventrilo
01:34.16m160858then, this local asterisk connect to the other asterisk in USA with g729
01:34.21SkramXI have a user who avidly wants this..
01:34.24alephcomright
01:34.42m160858and then that asterisk in USA call with gsm
01:34.51Zach^^nassy you have a inbound voip?
01:34.52nassySkramX: whats ventrilo, a VoIP provider?
01:35.03m160858but, i don't have idea
01:35.06SkramXa team-spreak time thing
01:35.17m160858you're have some examples
01:35.19nassyZach^^: yeah, they are my in and out to pstn
01:35.32Zach^^SkramX hey.. i signed up for voipjet and sixtel... and need a little help.. you think u can help?
01:35.36m160858i don't know to create in iax.conf
01:35.42SkramXZach^^: possibly
01:35.49m160858the configuration for this
01:35.50SkramXIm me or open a support ticket with my company.
01:35.53*** join/#asterisk christo (n=chris@brezhnev.spiration.co.uk)
01:35.56christoevening all
01:37.34hhoffmanhiya christo
01:37.40Zach^^SkramX sent u a msg
01:37.59*** join/#asterisk ctooley (n=ctooley@rrcs-24-227-212-162.sw.biz.rr.com)
01:38.22nassySkramX: just curious. is ventrilio useful with asterisk. i thought the audio on the phone was mono. is it useful with mono, or is the audio not mono
01:39.10hhoffmanif you are using "ext => X,n,... using n instead of a number how do you do a n + 101?
01:39.23Qwelln+101
01:39.41hhoffmanoh! it that it :-)
01:39.45QwellShouldn't use 101 jumping anymore though
01:39.46Zach^^SkramX what is your company?
01:39.47hhoffmans/is that/
01:39.56Qwelljust check the return code of the stuff when you need to
01:40.09hhoffmanQwell: ah, ok. thx
01:40.16Pete_Largono more +101???
01:40.20Pete_Largowhat the hell?
01:40.30QwellPete_Largo: jumping like that isn't recommended anymore
01:40.47Pete_Largojesus, next I know it's going to be calle #pound instead of #asterisk
01:40.47QwellCheck things like ${DIALSTATUS} and do stuff based on what that says, instead of blindly jumping elsewhere
01:41.10QwellIt just isn't useful anymore
01:41.12hhoffmanQwell: is this explained somewhere? I'm going off of the OReilly book and it's a bit different there
01:41.19Pete_Largosure thing, where can I read up on that?
01:41.23Qwellchangelog?
01:41.24Pete_Largoduh, "check the wiki"
01:41.34QwellIt's simple really
01:41.39Qwells,1,Dial(blah)
01:42.00Pete_Largodon't stop
01:42.06Pete_LargoI'm learning new stuff!
01:42.32Qwells,2,GotoIf($[${DIALSTATUS} = BUSY]?5:3)
01:42.35Qwellor some such
01:42.58Qwellor, as macro-stdexten does it
01:43.09Qwells,2,Goto(s-${DIALSTATUS})
01:43.15hhoffmanah, cool... I'll look at that
01:43.18Qwells-BUSY,1,do(something)
01:43.23Qwells-NOANSWER,1,do(something else)
01:43.36QwellFAR more useful than just going to 2,102
01:43.39Qwells,102*
01:44.19Pete_Largoer, that should, have been jump-no-more-academy
01:44.26hhoffmanyeah, that's awesome
01:44.43Pete_Largowell, time for me to go watch Veronica Mars in a few minutes :)
01:48.13Zach^^:s can anyone help with voipjet?
01:48.36tzangeryeah, try nufone, asterlink, unlimitel...
01:49.10Zach^^tzanger when i try to dial out i get an error all circuits are busy
01:50.20SkramXIm back
01:56.56*** join/#asterisk insomni (n=insomni@x1-6-00-e0-18-6f-34-ad.k455.webspeed.dk)
01:58.11*** part/#asterisk m160858 (n=jsaenz@200.89.12.46)
01:58.37*** join/#asterisk r0d3nt (i=nobody@wsip-24-234-241-145.lv.lv.cox.net)
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01:59.37Seldon1975hi - my polycom501s assume all phone numbers are 2 digits long - whats up with that?
01:59.50Qwellthe phones have a dialplan...change it
01:59.52Seldon1975after 2 digits it tries to dial the number
02:00.00MstlyHrmlssounds like your dialplan is pooched
02:00.06Seldon1975ok so its a phone-specific thing
02:00.11Seldon1975not asterisk
02:00.20MstlyHrmlsyeah
02:01.08Seldon1975anyone there using Polycom Soundpoint 501s?
02:01.31kuku5Don't know what to do if second ROSE component is of type 0x6   < what does this mean ?
02:02.18MstlyHrmlsSeldon1975: yup
02:02.21CinenDoes anyone know where some good resources are for Dundi configs. I tried #Dundi but nobody is home.
02:02.42Zach^^/dms 72.20.32.117
02:02.43CinenI set it up like the wiki said with no luck\
02:03.02Seldon1975MstlyHrmls: can you tell me how to change the dialplan?
02:03.08Seldon1975or where to get the guide
02:03.41MstlyHrmlssure, just give me a sec
02:03.48Seldon1975thx
02:06.01MstlyHrmlsSeldon1975: OK, probably the best thing to do is to root around here for the Administrator guide for the Polycoms: http://www.polycom.com/support/voip
02:06.14MstlyHrmlsI think you can only get it for 1.5.3, not the latest release though
02:06.39MstlyHrmlsin there, it says to refer to RFC 3435 section 2.1.5 for the dialplan reference
02:06.43Seldon1975ta
02:06.55MstlyHrmls(which is the MGCP RFC, but polycom uses that dialplan)
02:07.14MstlyHrmlsif you have any questions, I might be able to answer them
02:07.30MstlyHrmlsI'm no expert, but I've fiddled with them a bit :-)
02:07.59*** join/#asterisk PBXtech (i=nik@120.sub-70-218-79.myvzw.com)
02:08.31p1tst0pcan you playback an mp3, over a channel from the CLI ?
02:08.40p1tst0por a gsm or whatever
02:11.00PBXtechredirect the channel
02:11.13tzangeranyone here do much voicexml stuff?
02:13.03p1tst0pJunK-Y not really mate, just play a audio file, when someone is running out of credit say...
02:13.10Zach^^anyone here help me with voipjet?
02:13.27p1tst0pPBXtech, how does one redirect form CLI ?
02:13.44JunK-Ybut why from the CLI?
02:15.51*** join/#asterisk tuxinator_linuxM (n=spabin@70-32-106-248.ontrca.adelphia.net)
02:17.15Seldon1975mstlyhrmls: I've set up two extensions, 1000 and 1001 in my * sip.conf; do I need to tell the phones what their extensions are?
02:18.38MstlyHrmlsis this part of the dialplan still, or is this on to getting the phones to register?
02:21.48*** join/#asterisk outofjungle (n=outofjun@61.247.249.13)
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02:24.14*** mode/#asterisk [+o anthm] by ChanServ
02:24.31Zach^^WTF why cant i get past all circuits are busy now
02:27.23MstlyHrmlsSeldon1975: still working on the dialplan, or getting the phones to register?
02:29.14*** join/#asterisk brettnem (n=brettnem@72.29.102.158)
02:29.53Seldon1975MstlyHrmls registering the phones now
02:30.00Seldon1975MstlyHrmls any ideas
02:30.14Seldon1975do the phones need to be told?  or will they read the info from *
02:30.20MstlyHrmlsthen, yeah, you have to tell the phones what extensions to use :-)
02:30.24Seldon1975k
02:30.27Seldon1975how? :D
02:30.45MstlyHrmlsthe easiest way is by the web interface
02:31.03MstlyHrmlspoint Your Favourite browser at the IP of one of the phones
02:31.38Seldon1975ok I'm there
02:32.01Seldon1975Sip Conf?
02:32.03MstlyHrmlsyou'll get a "SoundPoint IP Configuration" page. click on the SIP menu
02:32.04MstlyHrmlsyup
02:32.28Seldon1975OK I assume it's 'Local SIP Port'?
02:32.38MstlyHrmlsno
02:33.09MstlyHrmlssorry, the Lines config is under "Lines", SIP is where you configure the asterisk address
02:35.27Seldon1975so from http://phone.ip I go to which of 'Home' 'Core Conf' 'SIP Conf' 'Registration' ?
02:35.43MstlyHrmls'Registration'
02:35.50*** join/#asterisk fcr (n=fran@r201-217-143-141.dialup.adsl.anteldata.net.uy)
02:35.51MstlyHrmlswhat SW are you running btw?
02:36.19*** join/#asterisk doughecka (n=Miranda@unaffiliated/doughecka)
02:37.32Seldon1975i think 1.6
02:37.56fcrHello. I need to record a few telephone lines without hunging up, and I would like to know if this is possible with asterisk and some hardware.
02:38.06Seldon1975do you see a 'Lines' section under 'Registration'
02:38.08Seldon1975because I dont
02:38.38MstlyHrmlsSeldon1975: I don't think you're using 1.6, because that's what I have and I'm seeing different things on the webconfig than you are :-)
02:38.46Seldon1975bootrom of phone is 2.6.1.0003
02:38.49Seldon1975ok
02:38.52Seldon1975damn
02:39.24MstlyHrmlsare you at the phones right now?
02:39.35SkramXfcr: it's possible
02:40.00Seldon1975yes
02:40.14Seldon1975the phone's App version is 1.4.1.004
02:40.18fcrSkramX, ok, so what hardware would i need
02:40.33MstlyHrmlsSeldon1975: yeah, I thought that might be it
02:40.43SkramXSo you want to be in the middle of a call and press a button and it records?
02:41.03MstlyHrmlsSeldon1975: unfortunately, I don't have a phone running 1.4 handy, and I don't remember it well enough to step you through from memory :-7
02:41.05SkramXfcr: any sip phone should work, just tell asterisk to start recording when you pickup
02:41.17fcrIn fact i need to record a line all the day.
02:41.22MstlyHrmlsSeldon1975: unless you wanted to set up a provisioning server for the phones, rather than just go through the webconfig
02:41.58Seldon1975its ok MstlyHrmls; Im about to upgrade the phones to 1.6
02:42.04Seldon1975then we'll be on the same page :D
02:42.09MstlyHrmls:-D
02:42.34Zach^^can you transfer users to the digital receptionist? or the queues?
02:43.35MstlyHrmlsSeldon1975: the coles notes version is: put the asterisk address in the "Address" box of the "Server 1" section on the SIP page
02:44.06Seldon1975ok Cinen is helping me modify my sip.cfg file which I think is the same setting
02:44.16MstlyHrmlsSeldon1975: then put the "1000" or "1001" in the "Address" box of the Line 1 section on the "Lines" page
02:44.23MstlyHrmlsyes, it is
02:44.38Kattyhi lads.
02:44.59MstlyHrmlsSeldon1975: are you creating <mac>.cfg files, or just using the 000000000000.cfg file?
02:47.26Seldon1975the former
02:48.42Seldon1975we are copying the file and renaming it to each phone's mac address
02:49.52MstlyHrmlscool
02:50.15MstlyHrmlsSeldon1975: while you're at it, make a copy of the phone1.cfg file for each phone
02:50.37MstlyHrmlsSeldon1975: rename it phone1000.cfg and phone1001.cfg or something
02:50.44Kattyway to not say hi.
02:50.49Kattybunch of anti social geeks.
02:51.03MstlyHrmlsthen you can program the extensions in there, and avoid using the web config entirely
02:51.24KattyQwell: k, grown up laddy.
02:51.32Qwellclose enough
02:52.05*** join/#asterisk bjohnson (n=bjohnson@i216-58-63-114.cybersurf.com)
02:52.31docelm0hmmmm
02:53.09MstlyHrmlsKatty: hi! if we weren't anti-social geeks, wouldn't we be out having lives instead of being here on IRC? :-)
02:53.57Seldon1975MstlyHrmls do you mean that the extension setting is in (mac)-phone.cfg?
02:54.22KattyManxPower: uhh.
02:54.32KattyManxPower: i'm a female, not a feline.
02:54.50KattyManxPower: also, i'm a vegan...so please remove all mouse related food items from general vicinity... kthx.
02:55.05MstlyHrmlsSeldon1975: if you use the web config, that's where the phone will store it; but you can put it into the phone-<extension>.cfg file
02:55.18mishehuKatty: what about the food I feed my Logitech mouse?
02:55.26Kattymishehu: yay, mishehu
02:55.35Kattymishehu: that doesn't count.
02:56.19mishehuKatty: I hate doing dumb coursework.  blah.  I have one more assignment for this dumb economics class to finish, and I'm working on it now.  :-/
02:56.27Kattymishehu: :<
02:56.52file[laptop]Katty: you're silly
02:57.03Kattyfile[laptop]: takes one to know one, dear.
02:57.03mishehuwhy not file[desktop]
02:57.18mishehuanyway, back to this dumb coursework...
02:57.25Kattymishehu: you could always take a little trip to the paper shredder :>
02:57.37mishehuKatty: I need the grade, it's a required course.
02:57.44Kattymishehu: :<
02:57.54Kattymishehu: you could give all related stuffs to birdy when class is done :>
02:59.16file[laptop]I _hate_ the seatbelts in the Honda Civic Coupe... hate with a passion
02:59.26Kattyfile[laptop]: did they try to strangle you?
02:59.48file[laptop]Katty: well yes, but that's not it - they aren't attached to the seat so you have to reach back far to get them... almost dislocated my shoulder!
03:00.03file[laptop]which is bad mmmkthxbi
03:01.33Kattycopykat.
03:01.42file[laptop]not I.
03:01.53Kattylittle red hen.
03:02.23mishehuKatty: I assure you I was already planning on lining hte birdcage with it.  ;-)
03:02.36Kattymishehu: whoo!
03:02.45Kattymishehu: post gifs when the shredding begins :>
03:02.57file[laptop]how was your day KittyKatty?
03:03.36Kattyi went on a service call an hour away to fix a symantec antivirus problem.
03:03.47mishehuKatty: unfortunately the shredding is never that exciting
03:03.51Kattyand then, when i came back, 10 people pounced me with zomghastobedoneNOWstuff.
03:04.03Kattyand then i studied mcse stuffs.
03:04.08Kattyand went home at 4
03:04.17hhoffmanso are macro-superdial and macro-stdexten exclusive?
03:04.18Kattyso here i've been, ever since, playing earthbound on snes.
03:04.49Kattyfile[laptop]: how has your day been?
03:05.31Kattyoh! and i put my christmas tree up :>
03:05.38Kattybut only one string of lights.
03:05.41Kattycause i'm lazy.
03:05.50Zach^^Katty you us astrisk?
03:05.55Seldon1975MstlyHrmls thanks for your help
03:06.06KattyZach^^: why else would i be in here?
03:06.10KattyZach^^: for the annoying questions?
03:06.30Zach^^Katty: yep :p you think you can help a newb
03:06.30KattyZach^^: yes, i use asterisk.
03:06.40KattyZach^^: oh come on, i'm playing earthbound!
03:06.55KattyZach^^: what's your issue?
03:07.19Zach^^i have voip jet for youtbound and sixtel for incoming and when i place a outbound call i get all circuits busy
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03:07.46KattyZach^^: can't help, sorry.
03:08.02Zach^^:s why?
03:08.34Kattythat's now what my setup is like.
03:08.34Kattys/now/not/
03:09.05MstlyHrmlsSeldon1975: no problem
03:09.08Zach^^can anyonehelp?
03:09.12Kattybut if you have issues with making sugar cookies turn out right, i'm totally the person to ask.
03:09.29jake1932yum
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03:09.39Zach^^yum make me some
03:09.42puzzledhmm cookies. reminds me I still have some stroopwafels
03:09.47justinumy fiance makes kick ass cookies
03:09.47Kattyi'm not a maid, sorry.
03:10.28Kattysounds painful.
03:10.34Zach^^how do i transefer users to a queue?
03:11.23kuku5Is there a way to pickup a ringing phone from a phone that is not ringing ?
03:11.32Kattykuku5: features.conf
03:11.37Kattykuku5: usually *8 or something
03:11.43kuku5wow
03:11.47kuku5so I can do taht /
03:11.54Kattykuku5: yup
03:12.05kuku5kickass
03:12.14Kattynonono
03:12.17Katty*8
03:12.19Kattyget it right.
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03:13.13kuku5*8 and then the extension ?
03:13.18Kattykuku5: no.
03:13.23Kattykuku5: it picks up a random ringing phone
03:13.32Kattykuku5: i think it's random, don't actually know.
03:13.47Kattykuku5: you can edit your features.conf to stipulate what you want it to be
03:14.00Kattykuku5: or 'show features' at the asterisk CLI
03:14.19kuku5ok
03:14.24kuku5i'll need to look into it
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03:17.05kuku5Builtin Feature           Default Current
03:17.05kuku5---------------           ------- -------
03:17.05kuku5Pickup                    *8      *8
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03:37.29sylewhy do employees act like kids, test you for how much they can get away with
03:38.54syleanswer: because its fun
03:38.55brookshirewhy do you always see how much you can get out of a customer?
03:39.12file[laptop]brookshire: why are you so hot?
03:39.13file[laptop]:P
03:39.14file[laptop]lol
03:39.17brookshireOMG!
03:39.19brookshirehehe
03:39.22brookshirehey file
03:39.26file[laptop]hi brookshire
03:39.41file[laptop]still hard at work?
03:39.49file[laptop]and don't read that with a nasty mind
03:39.56brookshiredrum and bass is so hot
03:39.58brookshire:)
03:40.10brookshireit's been a long day :/
03:40.17file[laptop]ah... poor you
03:40.29brookshirelong and hard day :(
03:40.39file[laptop]well tomorrow is another day!
03:41.04brookshirehopefully a short one
03:41.12file[laptop]hopefully
03:42.10hhoffmananyone do dynamic meetme extensions with dynamic pin numbers?
03:42.33file[laptop]brookshire: I suggest you go home and get some sleep though
03:43.25brookshirei need to finish my staging server first
03:43.38file[laptop]bah
03:43.47sylewtf is a staging server
03:43.51syledoes it dance?
03:44.03brookshireno.. you have a live server and a staging one :)
03:44.12brookshirestaging is what you do development on
03:44.14syleyou mean development box
03:44.18jake1932and a development one
03:44.31jake1932there could be 3
03:44.35Nuggeta staging server is a way for smart admins to accomodate software packages that have crappy or unreliable release cycles.
03:44.47Nuggetit's not the same as a development server
03:44.47*** join/#asterisk mog_home (n=mogorman@user-24-236-84-48.knology.net)
03:44.54brookshireyup..
03:44.56*** join/#asterisk perlmonkee (i=oginvu@c-24-22-125-15.hsd1.or.comcast.net)
03:45.06brookshirelike we're trying to safely upgrade to mysql 4.1 from 4.0
03:45.13sylegood luck
03:45.14perlmonkeeI'm having a problem with Asterisk and SipPhone
03:45.21brookshireand as quickly as possible with little downtime
03:45.32perlmonkeeI can place outgoing calls through SipPhone, and `sip show registry` says I've registered with SipPhone.com
03:45.41hhoffmanonly problem I had with that upgrade was CHARSETs
03:45.44sylei did that ages ago, moving to 5.x was less painful from 4.1
03:45.46perlmonkeeI've set up call routing for incoming calls to my extension
03:45.57brookshire4.0 to 4.1 is a bitch
03:46.00perlmonkeebut every time I call my SipPhone number (from another number), it jumps straight to SipPhone Voicemail.
03:46.13brookshirebut then again i hate everything about 3.x :(
03:46.18brookshireabout = above
03:46.20syleif i was you i would just dump all my data and go to 5.x like everyone else
03:46.23perlmonkeeIf I register directly to SipPhone with a softphone, everything works.
03:46.23hhoffmanbrookshire: what's making it so bad for you?
03:46.38syleover 1 million people downloaded 5.x in first 3 weeks it became stable are the stats
03:46.42brookshirehhoffman: when you go from 4.0 to 4.1 the authentication methods change
03:47.02brookshireso php stops working
03:47.07brookshireand perl stops working
03:47.09hhoffmanah, yeah... we upgraded all clients and perl/php
03:47.12brookshireand etc, etc etc
03:47.21brookshirewe've just been putting it off
03:47.31hhoffmanthere's a way around upgrading the clients
03:47.36brookshirebecause truthfully.. there is no advatage to goto 4.1
03:47.39syleno, you just have to recompile perl DBI again new mysql api and php never had a problem
03:47.54syles/again/against
03:48.07brookshiresyle: exactly :)
03:48.16perlmonkeeCan anoyone help with Asterisk+SipPhone.com problems?
03:48.17sylean extra 10 min not a big deal
03:48.22brookshirebut the problem is...
03:48.25hhoffmanwe experienced it in a DBI:: app but couldn't upgrade... there's a way in the FAQ to deal with that
03:48.43brookshirewe have to do all of our servers at the same time, because they depend on each other, lol
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03:49.08syleif you read the mysql docs you can tell it to be backwards compatible by setting password=old or whatever
03:49.27brookshiresyle: i haven't had time yet
03:49.29sylei prefer to move on with changes
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03:51.08brookshireanyways.. better to be safe than sorry
03:51.19brookshireand have asterisk.org down for a couple of days
03:51.24brookshirelol
03:51.52Cinenanyone here had any luck with Dundi? I asked in #Dundi but nobody is home
03:52.02sylei would just do it right brookshire in the first place
03:52.10syleexport everything go to 5.x
03:52.28sylethen you won;t have to worry about it for 2 years insteead of comming back to this in a year going to 5.x when 5.1 is now stable
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03:52.53brookshiresyle: actually.. i've found 4.1 kind of unstable, until the more recient releases
03:53.07brookshirereceint also
03:53.08brookshirelolalskdjfasdf
03:53.45brookshirerecent
03:53.47brookshireTHERE
03:53.56sylenot like asterisk.org is taking a million hits a day, don; think you have to worry to much :)
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03:55.08sylei wonder if NASA and sourceforge are using 4.1 or 5.x
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03:55.23brookshirewell.. more like 200,000 hits
03:55.26brookshirebut yeah
03:55.29brookshirenot a million
03:55.38syleno shit
03:55.40syleunique?
03:55.45brookshireoh... just hits
03:56.01brookshire4,000 visits
03:56.06brookshirei think
03:56.11syle4000 unique isn;t bad
03:56.35syledamn in a few years that will be quite amazing
03:56.48brookshirei know.. we're working on it :)
03:57.20brookshirethat doesn't include downloads, or svn
03:57.33brookshireviewcvs
03:57.40syleyou keep track of download stats?
03:57.49orlokHmm.. I seem to have lost my outbound call routing
03:57.54brookshireyup :)
03:57.59sylehehe nice
03:58.14brookshireclose to 2,000 a day
03:58.18brookshirejust asterisk
03:58.20syleno shit
03:58.33syledamn i would never have guess that
03:59.10syleare you sure 2000 a day?
03:59.16syleunique ip downloads?
03:59.24brookshirewell it's been that reciently because of 1.2
03:59.33syleowww that makes sense
03:59.41brookshirerecently
03:59.42sylehow about on average when no new releases?
03:59.44brookshirethere i go again, lol
04:00.00jake1932i before e except when no i
04:00.09brookshirethere is no i in recently
04:00.16file[laptop]brookshire: you can spell... right?
04:00.17tainted_lol
04:00.27brookshirei keep wanting to add it though
04:00.28brookshirelol
04:00.47tainted_jake1932 what english class did YOU attend
04:00.59jake1932none - got everything from IRC
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04:01.17tainted_sweet
04:01.39sylei think i was taught in university by programmers to tell people who had spelling mistake problems with you that you aren't an artsy fartsy, your a science man hehe
04:01.40jake1932(and read the wiki, of course)
04:03.00jake1932any of you use shorewall with asterisk?
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04:03.18docEya?
04:03.20sylealthough the wars between arts and sciences were always bad in my schools, science students use to wear shirts that said "friends don't let friends take ARTS"
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04:04.27jake1932i'm going to attempt setup tomorrow and just wanted to know that it does work on the same box as asterisk giving asterisk the public ip
04:06.16syleif you didn;t find that funny, you've never been to university :) well back to coding
04:06.18jake1932and btw - arts != english - I know plenty of musicians who can't spell worth a damn
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04:07.54syleyou must realize science people are only obligated to take first year english, and are usually passed even if they fail because their in sciences and need to move on
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04:08.42syleif your in arts, noone cares about you
04:09.29syleget a BA and work at a gas station, or become a lawyer hehe
04:09.48Nuggetscientists who fail that first year english are likely to fuck up the difference between "there", "their" and "they're"
04:09.58jake1932only in IRC
04:09.59Nuggetobviously disqualifying them from a technical life.
04:10.13*** join/#asterisk MikeJ[Laptop] (n=ircatjer@d14-69-8-30.try.wideopenwest.com)
04:10.35sylemaybe thats why you can barely even understand what doctors write on prescriptions hahaa
04:11.36jake1932don't quite understand why I can easily explain the difference between when to use there", "their" and "they're, but when I'm typing fast, it doesn't come out right
04:12.20syleyeah don't mistake can't spell between types to fast for sure
04:12.26NuggetI absolutely hate it when people use apostrophes in plurals, and yet from time to time my fingers do it without my involvement.
04:12.33jake1932right
04:12.37NuggetI'm always left staring at the little buggar wondering how the hell it happened.
04:12.49jake1932there must be something to it
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04:13.00sylemy problem is using ; instead of ' when typing to fast lol
04:13.00Pete_Largothat happens to me a lot too
04:13.02Nuggetand I'm totally ruined for typing the word "serve" without throwing an extra "r" on the end.
04:13.12Nuggetnot to mention "distribute" without a trailing "d"
04:13.13Pete_LargoI do that too
04:13.17Pete_Largoand that
04:13.35jake1932there must have been a study done on that
04:13.36Pete_Largoand the / and \ always get me too
04:13.45Pete_Largowhich goes where
04:13.55syleyes because of going between creating windows shares and working with unix directories
04:13.59jake1932Pete - that's a windows/linux thing
04:14.02Pete_Largoright
04:14.04sylealways think its / on windows at first lol
04:14.06Pete_Largojust like ls and dir
04:14.10Pete_Largodel and rm
04:14.14Nuggetit's a windows/UNIX thing.  :P
04:14.15Pete_Largohappens all the time
04:14.28Nuggetbecause it happens to me, too, and I never use linux.
04:14.49syleits a windows\UNIX thing, its a UNIX/windows thing :)
04:14.53Nuggetheh
04:14.55jake1932hah
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04:15.49jake1932night people
04:16.10sylei'm just waking up
04:17.21dokhenchhave you're self a few shot of jack, then right off to bed again. you'll do the sleep wrap around in no time. =)
04:17.28dokhenchyourself that is.
04:17.49*** join/#asterisk MikeJ[Laptop] (n=ircatjer@d14-69-8-30.try.wideopenwest.com)
04:17.52sylewhats the theory behind that
04:18.19dokhenchthe jack will drunk ya up, before you have a chance to fully wake up. =)
04:18.34sylehehe
04:18.46sylei should try that :)
04:19.14Nuggetall I have here is a fe bottles of wine and a fifth of wild turkey.
04:19.17Nuggets/fe/few/
04:19.48sylewhjen i have wine i;m drunk as a skunk
04:19.52syleworse than beer
04:20.09Nuggetwine drunk creeps up on you
04:20.11syleseem to pound back a wine bottle quite fast
04:20.24Nuggetyou feel fine and then get up to go pee and the room spins
04:20.31syleyeah its great
04:20.37sylethink i;ll have some for christmas
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04:21.23sylered wine that is, never got same effects with white
04:22.21NuggetI'm not fond of white wine.  I only drink it when I'm having cheese fondue or something like that.
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04:25.09sylefondue hmm haven;t had that since i was in montreal
04:25.12syleyou from canada?
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04:26.41Nuggetno
04:26.48sylei like sweet red wine's but all the damn restaurants only serve dry pisses me off
04:27.02syleso liquor store works out great
04:27.54sylei;ve considered making my own at home, but then i think would become an alcoholic lol
04:28.05NuggetI brew my own beer.  It's fun
04:28.12NuggetI don't know much about winemaking, though
04:28.44syletakes less time to sit
04:28.47syleabout a week
04:28.51Nuggetcool
04:28.56sylebeer can take a month to get some good alcohol content
04:29.12sylewell depends how much yeast you add hehee
04:29.13Nuggetright
04:29.26Nuggetit takes a month before it will taste good, for sure.
04:29.36Nuggetthe alcohol doesn't take that long to build
04:30.03syleno but you have 1 month cyles, so you have to know how much you drink in a whole month all the time :)
04:30.17Nuggetnah, if I overproduce I just invite people over.
04:30.21Nuggettotal non-problem
04:30.24sylegood man :)
04:30.33sylewhere you live again :)
04:31.14Nuggetaustin texas  :)
04:31.24syleoww the US kewl
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04:32.26sylei only had one experience with texas, passing through that airport back to canada from the carribean, customs can really go screw themselves on all their security bush created, i was waiting in line 2 bloody hours, never go through the US again
04:32.47syledirect flights only for now on :)
04:32.47Nuggetaustin is almost entirely unlike the whole rest of the state.
04:33.34sylethink i met a few people from austin in panama
04:33.49syleall into real estate
04:33.58NuggetI've lived a lot of places, and this is the one where I never want to leave.
04:35.25syleyeah i couldn;t handle your heat
04:35.28Jack_Stormsyle: WTF are you talking about? if customs fucked with you, you gave them reason. I came back from Russia to Atlanta, had customs there and here, and they had no problems when me, even thou I was smuggling
04:35.55Jack_Stormand here is New Orleans
04:35.57syledude there was no reason, that was the damn lineup
04:36.15sylei was stuck in line 2 hours with other connecting flights
04:36.26Nuggetflying through houston or dallas?
04:36.36fugitivoUS inmigration department sux
04:36.38sylehmmm i think it was houston
04:36.50Nuggethouston is really bad.  they have like six hundred lines for us citizens and three for non citizens
04:37.14justinuthat rules
04:37.47sylewell they always fuck with me leaving canada anyways, their reason: you brought a laptop: my mind saying: go fuckyourself everytime
04:38.06Nuggetevery time I would go on a business trip with Ivo at my last job (he's dutch) I'd clear customs in 5-10 minutes coming back and he'd take an hour.
04:38.25fugitivosyle: don't tell them you have linux on it, it's worse (you have linux, right?)
04:38.29syleif i wanted to hide something on my laptop i;d download off the net after i crossed, no reason to search it
04:38.42syleyes
04:38.45syledouble booting
04:39.06sylesorry laptop i take is dual booting XP and freebsd
04:39.11sylebut they only see the XP
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04:39.29sylei pretty much use to the routine now
04:39.31Jack_Stormsyle: did you have a .ca customs stamped document showing that the laptop was yours and in .ca prior to your departure?
04:39.33fugitivothey check your files?
04:39.43sylestart menu->find-> *.*
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04:39.51fugitivoreally?
04:39.52sylelook at all my files
04:39.54syleok you can go now
04:40.07syleyes man
04:40.16fugitivothey shouldn't look at your files
04:40.16fugitivothat
04:40.19fugitivothat's private
04:40.28syleits customs they can do anything they want
04:40.31MstlyHrmlsI've never had them do that to me
04:41.01Jack_Stormsyle: I came back from Russia, they didn't want to look at anything, it's all on how you present your self.
04:41.02MstlyHrmlsI've never really had much flack at all
04:41.21Jack_StormMstlyHrmls: nod same here
04:41.29MstlyHrmlsperhaps the pre-clearance centres are a little more lax
04:41.47syleits only certain airports
04:41.57MstlyHrmlsbut as long as I'm only going down for meetings, they don't seem to care
04:42.02Jack_Stormsyle: no nothing to do with Bush then?
04:42.05sylewell 1 airport
04:42.10Jack_Storms/no/so/
04:42.44syleyou kidding?
04:42.53sylehow about the iris and fingerprint scanners
04:43.58fugitivous inmigration department is a joke
04:44.04syleconvenient to pass through the iris scanner very quick, but thats quite an invasion of privacy
04:44.14Jack_Stormsyle: he wouldn't do that to only certain airports
04:44.23fugitivothey check your files, that's invasion of privacy
04:44.23syleonly time i want to voluntary give out my fingerprints to any country is if i;m being arrested thankyou
04:44.44Jack_Stormsyle: by your def, your passport is an invasion of privacy.
04:44.56sylehow so?
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04:45.15delox99hi all
04:45.27Jack_Stormsyle: does it not contain a likeness of you? your sig? your address and where you have been?
04:45.45syleits a picture with a my signature, not much different than my drivers license
04:46.13sylewell where you have been is just you know your from here
04:46.56syledoes it really matter?
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04:47.05Jack_Stormsyle: and your personal info on your drivers lic is ok, but an eye scan to pass you quickly is an invasion of privacy?
04:47.10syleif they have your SIN or SSN they got everything they need anyways
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04:47.33delox99in zapata.conf what s the difference between group, callgroup, pickupgroup, and group for channels?
04:47.41sylejack yes
04:48.28sylelets say i figure out my fingerprints to some asswipe at some countries border, he decides to frame me for a crime, he has my fingerprints, totally screw me over, like i said invasion of privacy
04:48.52syles/figure/gave
04:49.13Jack_Stormsyle: I have your sig off your passport, forge it to a check that was used to pay someone to kill someone, same diff.
04:50.00syleyes but DNA data is unargueable in court
04:50.20sylei could argue the signature thing
04:51.09Jack_Stormsyle: and I could argue the iris scan, and the MS backed filesystem and DB cluster.
04:51.21syleand we all know americans get ahold of all the customs records who pay for it
04:51.31syleand they get sold on the black market
04:51.41syleas i said less info the better
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04:52.18syleyou gonna make an arguement that you don;t have corrupt politicians good luck
04:52.31Jack_Stormsyle: and thats why I can buy a HDD in Moscow filled with .ca personal info and photos of people with out passports?
04:52.39pigpen2Could someone answer a question regarding the Polycom 601?
04:53.19syleyes
04:53.21MstlyHrmlspigpen2: maybe
04:53.50pigpen2Ok..with the exten presence (ie: hint) it will monitor 7 exten's....
04:53.58syleyou know enough wealthy people in american you can get any info you want
04:54.03pigpen2is this per phone, or per expansion module?
04:54.14MstlyHrmlsper phone
04:54.35pigpen2that is what I thought...thanks..you saved me time of testing it.
04:54.43MstlyHrmls:-)
04:54.47pigpen2any news on it getting expanded?
04:54.56pigpen27 just doesn't cut it.
04:55.22MstlyHrmls"in a future release"
04:55.33pigpen2so there is rumor...
04:55.48MstlyHrmlsand, no, 7 doesn't cut it
04:55.54pigpen2I figured it was 7 due to the fact the 600 did not have the expansion module.
04:56.21asterboyhow much did you pickup your 601 for?
04:56.24pigpen2ok..next question (sorry, not polycom related)
04:56.33pigpen2601:  $239 US
04:56.48asterboygood price
04:57.09pigpen2Has anyone used the "metermaid" patch for monitoring parked exten's?
04:57.55pigpen2ie:  http://bugs.digium.com/view.php?id=5779
04:58.37*** join/#asterisk `Sauron (i=sauron@h-69-3-12-50.hstqtx02.covad.net)
05:00.32pigpen2well, seems pretty cool...we are compiling it in now.
05:00.49*** join/#asterisk wundaboy (n=asdf@c-67-164-107-68.hsd1.or.comcast.net)
05:03.14*** part/#asterisk tholo (n=tholo@nat.sigmasoft.com)
05:05.13*** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net)
05:06.07*** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net)
05:06.35denonyou guys seen this? http://denon.cx/christmas
05:06.39denonsomeone's got way too much time on their hands
05:06.56*** join/#asterisk rajiv (n=irc@gentoo/developer/rajiv)
05:08.31syledamn
05:08.45sylethat makes me want to drink beer on that lawn
05:09.14denonu huh
05:09.27sylethis would be great for an outside bar
05:10.12sylewell if you know anything about bars you know most popular ones are the ones  that invested alot of money in lighting shows
05:10.39*** join/#asterisk mog_home (n=mogorman@user-24-236-84-48.knology.net)
05:11.00syleone owner here retired in 4 years
05:11.02fugitivobars or discos?
05:11.09sylepaid 1 million for his lighting in his bar
05:11.28sylemade that back in 1 year, made another 5-10 mill, retired
05:11.36sylemoved back to europe
05:12.30sylebut we;re talking about a 1 million dollar laser system show
05:12.47sylemirror galore etc
05:13.44sylesmart guy, and i beleive the move to europe was for a tax break on his last year he made money
05:14.31syleas you know canada, US and australia are the top taxing countries, if you make alot of money in one year its best to move out of those countries for abit if possible
05:16.33syleand thats why security is bad, gov't is using it as a way to track your offshore bank accounts
05:17.04SkramXHi All
05:19.31orloksyle: i must say, i have never, ever been struck by lighting shows as being something a bar needs
05:19.34orlok1. beer
05:19.41orlok2. something to rest the beer on
05:19.51orlokand if you are lucky, 3. somewher to place your butt
05:19.55syleyour wrong its all about atmosphere
05:19.57orlokwalls and a roof are optional
05:20.17orlokyeah, and its a pub, not a rave or a disco!
05:20.33sylewell depends if you want to make 50k a year or a million i guess
05:20.51orlokwhich country you from?
05:20.54sylewell thats not accurate
05:20.56orlokthe states?
05:21.03syleall bars make a few hundred k a year
05:21.15orlokcos whereever it is, they have a warped view of pubs compared to the ones here in australia
05:22.18syleyou have to realize there is no outdoors in winter here
05:22.33orlokwhich country?
05:22.40sylecanada
05:22.47orlokahh
05:22.51orlokyeah, i been there in winter
05:22.58orlokin early 1989
05:23.07orlok-26.. my face froze up i swear
05:23.22syleonly -26 damn come in january :)
05:23.39syle-40 at times with windchill :)
05:24.14syleliterally my eyelids can freeze in about 20 sec
05:24.16syle:)
05:24.19orlokyeah, it was jan.
05:24.27orlokon the way back home after xmas in england
05:24.43*** join/#asterisk Smi|k (n=smilk@adsl-66-159-200-157.dslextreme.com)
05:24.48orlokhmm, i hooked my bosses voip phone up to asterisk
05:24.56orlokand i've slightly buggered up inbound and outbound calls
05:25.05sylei stayed in phoenix arizona for awhile when i was working, i couldn;t handle that heat
05:26.05sylehaha and your boss wants to kick your ass to fix it or put it back the way it was right hehe
05:27.27mog_homeanyone ever set up jabber2 lately, i switched from ldap to pam and cant seem to get it to let me log in
05:27.40ManxPower~docw
05:27.45ManxPower~docs
05:27.46jboti 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
05:27.48ManxPower~mailinglist
05:27.49jbotextra, 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
05:27.54mog_home?
05:28.35syleorlok you have extensive experience with asterisk and mysql?
05:29.03*** part/#asterisk fcr (n=fran@r201-217-143-141.dialup.adsl.anteldata.net.uy)
05:29.15sylei'm looking for beta testers today, releasing beta version of module i have been working on for 2 months
05:29.19kuku5jbot: put it on the topic
05:29.23kuku5what module
05:29.43syleSunsaturn
05:29.55kuku5eh
05:30.25Smi|kanyone mind looking over a new page design and giving me a little feedback?
05:30.51sylesure
05:31.03sylei;m great at critisizing designs
05:31.19Smi|kwww.mp3yourcar.com vs. http://mp3yourcar.com/default3g.asp
05:32.19*** join/#asterisk MikeJ__ (n=ircatjer@d14-69-8-30.try.wideopenwest.com)
05:32.44sylejust google Sunsaturn asterisk if you want more info
05:33.20asterboydam both are good designs
05:33.38syleyou designed that?
05:33.55Smi|kI edited the first into the 2nd
05:34.09*** join/#asterisk MikeJ[Laptop] (n=ircatjer@d14-69-8-30.try.wideopenwest.com)
05:34.27sylehttp://mp3yourcar.com/default3g.asp is better
05:34.34Smi|k25% of people who hit the opening page of the first one leave right away
05:34.47Smi|kdont click anything else, dont use the dropdown menus etc..etc..
05:34.54Smi|k2nd one I dont know if it will be better or not
05:35.05syleof course it is
05:35.09orloksyle: no, def. not extensive :)
05:35.19syleyou finally put that damn search stuff out of the way on the right
05:35.25syleinstead of in main body
05:35.39Smi|kyou cant actually buy ANYTHING until you search for your vehicle
05:35.56syleyes but second design i can see products on the page without scrolling to
05:36.01Smi|kso the idea behind clicking one of the ipods is the next step you click your car make
05:36.06sylethats where my attention goes
05:36.07Smi|kthen you click your car model, then you click your car year
05:36.26Smi|kwe dont actually sell ipods, it saying to select your ipod
05:36.32Smi|kthats just to have poeple click something
05:37.09syleright but design 2 is better cause then i can see what i can get for my car first
05:37.26Smi|kits not showing anything you can get for your car unless you select your car
05:37.43Smi|kwe sell parts that connect the ipod (which you already have) to your car (which you already have)
05:38.17sylethen your not taking advantage of marketing
05:38.30syleyou should open reseller account for the actual ipods, and sell the parts
05:39.02Smi|kassuming I want to sell car adapters
05:39.35sylemore you sell more you make
05:39.36sylesimple
05:40.53sylei can;t say i like the design then if its for just the car audio hookup
05:40.55orlokheh, nope :)
05:41.15syleat least show some pictures of cars, people happy with their new sound system etc
05:42.08syleor keep your currrent design and take advantage of being an ipod reseller as well
05:43.10*** join/#asterisk tuxinator_linuxM (n=spabin@70-32-106-248.ontrca.adelphia.net)
05:44.11sylei;ve always been an expert when it comes to design and marketing, just suck with dealing with sales(hate talking with people) but i can get them all there :)
05:45.01sylelast 3 companies i worked for are millionares now :)
05:45.45Smi|khrm
05:45.52Smi|kso how do I sell ipod car connections
05:46.10sylefirst you have to consider whats the first thing they see when they come to the page
05:46.43syleand control the flow from there, i don;t have enough time to give that many tips but gl
05:46.44Smi|kshould be the header sentence
05:46.59Smi|k"  
05:46.59Smi|kFinally, an iPod Car Kit That Doesn't Force You to Settle for Poor Audio Quality
05:46.59Smi|k"
05:47.26asterboyyou have to sell the right market.
05:47.46sylemarket is different all together
05:47.50Smi|kthe market is people who tried the itrip and hate the fm quality
05:47.52sylehes talking about just design
05:48.04Qwellsmall market
05:48.16Smi|koh, I'm not changing design because I'm an artist, I want more people to buy car ipod adapters
05:48.22asterboyque in the 18 year old girl that I will get because I'm listening to "your" ipod equipment.
05:48.23tuxinator_linuxMwhat's an Ipod?
05:48.35tuxinator_linuxMjust kidding
05:48.40tuxinator_linuxMI don't have one
05:48.41Qwell"If you buy this, more girls will sleep with you."
05:48.43syleyes asterboy makes a good point
05:48.50sylewhat is your target audience?
05:48.58tuxinator_linuxMpeople with money
05:49.03Smi|k21-35 male
05:49.07asterboyyoung geeks
05:49.09syleyour design should be influenced around that age/group
05:49.09Qwell"...or boys...we don't judge."
05:49.51Smi|kproducts differ based on their vehicle
05:49.59tuxinator_linuxMI like the "don't suck" idea
05:50.00syleso 21-35 male, likes: fast cars, kewl gadgets, easy on their pocket books, do some research and tailor
05:50.01Smi|kso before I can show them products I must find out what kind of car they have
05:50.14Smi|kor else they see 10 products with 10 diff pics/descriptions and only one of the 10 works for their car
05:50.30*** join/#asterisk alvariux (n=alvaro@201.155.166.186)
05:50.33alvariuxhello
05:50.36*** join/#asterisk r0d3nt|m (i=nobody@wsip-24-234-241-145.lv.lv.cox.net)
05:50.55sylehere is a very big tip: They want to know how it will change their life
05:51.05Smi|ksyle, the ipod is a culture of its own now, so instead of focusing on fast cars I can sell to a massive number of fast car drivers AND toyota prius owners because they can identify with ipod culture
05:51.34alvariuxim buildg asterisk from svn buillds ok but when i try to start it shows me this: loader.c:414 __load_resource: app_db.so: load_module failed, returning 3
05:51.41sylewill i be kewler having one? will i be popular with the girls?
05:51.52alvariuxsomebody knows what is wrong?
05:51.52Smi|kI tried to hit the changing their life aspect #1 by writing "you dont have to settle for poor quality music"
05:52.09Smi|ksyle, you will enjoy your ipod MORE.
05:52.13asterboyhow about reversing the image?
05:52.14sylesettling is not changing
05:52.45Smi|k"It is time to enjoy your iPod in the car"
05:52.46Smi|khow about that?
05:52.58alvariuxcan somebody help me
05:53.01syleyes but what can the IPOD to for me?
05:53.03syledo
05:53.14asterboyyou show a young geek male with a rust bucket...but his clean ipod gives him the illusion of driving a porche.
05:53.15Smi|kthey already paid $300 for an ipod and loaded it with music
05:53.28Smi|keveryone coming to my site has an ipod already
05:53.41syleyour thinking wrong, maybe they will buy an ipod and your hookup
05:53.45QwellSmi|k: I don't.  Otherwise I'd buy one now
05:54.01Smi|kapple doesnt have online resellers like that
05:54.01syleyou can make profit twice
05:54.06Smi|kand ipods dont have margins
05:54.48Smi|kif market research says 99% of the people who come to my site have an ipod already I should base the site around the assumption that the visitor has an ipod already correct?
05:54.52*** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk)
05:55.15sylesmi|k truely all this shit is trial and error and takes experience, you setup your programming scripts to track stats based on each new design you put out and watch the popularity between designs
05:55.24Smi|konce they enter the vehicle search I find out if they have a 1995 Nissan Ultima or a 2006 Mercedies S500
05:55.33Smi|kbefore they do that I only know they have an ipod
05:56.15syleonly stupid people use the same design all the time
05:56.23Smi|khrm, who usually does this, because I am totally lost... you are helping a lot right now, but I mean in the long run I need to do it right
05:57.22sylei use to run a marketing operation for a company and made about 12 thousand a day
05:57.31Smi|kwebsite design contest signup link on the right hand side?
05:58.26sylehehehe
05:58.27Smi|kevery week give away one car adapter to the top front-page-design winner, and monthly the best front-page-designer gets $xxxx to design the rest of the site
05:58.31Smi|kor does stuff like that not work
05:58.39sylejust try new designs
05:58.43syleand go from there
05:58.58Smi|kits the whole chicken-egg problem here... sales from design, money from sales, money for design, design for sales
05:58.59syleyou;ll need to learn alot more about marketing to sell
05:59.31syleunderstand basic marketing lingo first
05:59.45syleCPM CPC branding, co-branding
06:00.17Smi|kthe new logo at the top is trying to create a "brand" for the store, I was told that would help shoppers be more comfortable vs. no brand as before
06:00.52syleare you the owner or you some guy working on this website?
06:01.02Smi|kunfortunantely both
06:01.09Smi|kchicken-egg :)
06:01.49*** join/#asterisk rezEdit (n=rezEdit@zapdos.omnigroup.com)
06:02.02syledepends how far you want to go with this
06:02.13Smi|kto the end, I started it in the beginning
06:03.06Smi|kmy passion, my life, its all been based around it for almost 10 years now
06:03.49sylei don't understand, ipods haven't been out that long
06:03.55Smi|khttp://wired-vig.wired.com/news/print/0,1294,17738,00.html
06:04.35Smi|kclick the name Jeremy, then click "order now" on the upper right link of the page it loads
06:06.13sylehttp://www.ipodhacks.com/article.php?sid=185
06:06.17syletop google search
06:06.35Smi|kfor what?
06:06.42syleipod car adapter
06:06.46Smi|ktry ipod car kit
06:06.51Smi|kor ipod car
06:07.40Smi|kor a page I really dont understand as I no longer understand google, try "car kit"
06:07.40sylehmm nice
06:07.50sylehow many unique hits a day?
06:08.10rezEdithey everyone.... I am having a bit of trouble with zaptel.conf and was wondering if anyone had any hints....  I started with a TDM04B card (4 FXO modules) and that was all good.  Now I have added a TDM20B (another card with 2 more FXO modules and 2 empty spots) and am not sure how to get it to properly handle this config....
06:08.26Smi|knot too sure, marketings never been my thing, I just like mp3's in my car a lot and always have
06:08.31Smi|keverything else is just there
06:08.38sylehehe
06:08.43asterboycan I get a car kit to turn my ipod into a wifi enabled handsfree sip phone?
06:08.45rezEditno matter what I add to zaptel.conf, it can not work with more than 4 channels
06:08.59Smi|kasterboy, you will need a new toaster oven for that
06:09.04asterboylol
06:09.11fugitivorezEdit: did you edit zapata.conf also?
06:09.33asterboysure your edititing the right zaptel.conf file?
06:09.56rezEditfugitivo: not yet, but right now I am just running ztcfg so does zapata.conf matter?
06:10.05Smi|knevermind on weird result, damn desktop search
06:10.16fugitivorezEdit: pastebin your zaptel.conf
06:10.32rezEditasterboy: yeah, the output from ztfg changes as I make edits
06:10.44Smi|kwait, no its not, I dont get it, any experience?
06:10.57asterboyok good.
06:11.22rezEditfugitivo: it's just basically fxsks=1-4
06:11.22rezEditfxsks=5,6
06:11.39rezEditI have also tried 7,8 instead of 5,6
06:11.49asterboypastebin time
06:11.53rezEditand 1,2 with 3-6
06:11.57rezEditok
06:12.50asterboyinclude /proc/interrupts and modules
06:13.02rezEdithttp://pastebin.ca/32931
06:13.07fugitivorezEdit: fxoks=1,2 for your fxs
06:13.21fugitivoerr
06:13.27fugitivofxoks=5,6
06:14.25rezEditoh damn
06:14.37fugitivook, does it work?
06:14.45rezEditTDM20B - that's FXS modules
06:14.58fugitivoyes
06:15.06fugitivoyou should use fxoks, not fxsks
06:15.25rezEdityeah, what I meant to have was a TDM02B
06:15.30rezEditmy bad, sorry
06:15.56fugitivooh
06:16.03fugitivothen
06:16.20rezEditI already have a TDM04B and wanted 2 more FXO modules, but I guess outr operations person ordered the wrong card.
06:16.44fugitivoyou want fxo or fxs?
06:16.46fugitivoi'm confused
06:17.13asterboyDo you want to sell those FXO modules?
06:17.17rezEditheh.... I need to bring in 6 analog phone lines so I need 6 FXO modules
06:17.28fugitivook
06:17.37rezEditbut now I have 1 card with 4 FXO's and 1 card with 2 FXS
06:17.59drraydigium sells the modules seperate
06:18.00fugitivodo you have a TDM02B or TDM20B? :)
06:18.11asterboyOh, you meant to say they miss ordered FXS...you WANT FXO
06:18.28rezEdityeah
06:19.12rezEditcrap.  I was hoping to go live with this tomorrorw.
06:19.21fugitivosleep time
06:19.23fugitivobye
06:19.29asterboynight
06:19.32rezEditfugitivo: g'nite.
06:19.47asterboyIf you burn out R13 and R14, you turn the FXS into an FXO
06:20.32*** join/#asterisk doushanes (n=doushane@c-67-173-1-227.hsd1.il.comcast.net)
06:25.44asterboy:P
06:26.05drrayasterboy?
06:26.36asterboyyes
06:26.57drraythat's the only difference between an FXS and FXO module for the TDM400p?
06:27.21asterboyno...I was just seeing if anyone was paying attention.
06:27.33asterboywould be nice though.
06:28.52*** join/#asterisk kimosabe (n=kimosabe@201.135.10.173)
06:29.09kimosabehow can i make my asterisk box grafical
06:29.25drrayyou can use FOP
06:29.26ptiggerdineamp
06:29.32drrayflash operator panel
06:29.49drraybut I don't like amp or fop really as solutions
06:30.06asterboyThe X100P clone cards are actually intel modems...those have the R13 and R19 resistors that can be removed to make the card show as a "genuine
06:30.12drrayyeah
06:30.13asterboyversion
06:30.22drrayor you can modify the zaptel source
06:30.42kimosabewhat is the site for amp
06:31.16drraylet me qualify what I said about amp and fop, I personally don't care for them..
06:31.22rezEditasterboy: heh, you almost had me.  I'll have to call viopsupply in the morning.  I just checked the order and we did ask for a TDM02B but they sent a TDM20B
06:31.34rezEditbastards!
06:31.55asterboythey killed kenny!
06:32.05drrayI think digium was too clever for their own good in naming those cards that
06:32.11mog_homewhat you say
06:32.16mog_hometo clever....
06:32.18mog_homeindeeeeed
06:33.04*** join/#asterisk testmachine (n=assink@ip237-239-58-62.adsl.versatel.nl)
06:33.06mog_homehehe i have my new server up and running
06:33.11rezEditagreed
06:33.15mog_homeall off my little raq box
06:33.33asterboywhat is the html front end for * ??
06:33.43ptiggerdine* = ?
06:33.52drray* = asterisk
06:33.59mog_homenone
06:34.08ptiggerdineAMP
06:34.14mog_homeamp is the closest thing to it
06:34.14asterboy~*
06:34.16jbotit has been said that * is asterisk
06:34.19ptiggerdineasterisk management portal
06:34.34mog_homeconfig files arent hard
06:34.36mog_homeenjoy em
06:34.38ptiggerdine~anyhtinguseful
06:34.54asterboy~beer
06:34.56jbotextra, extra, read all about it, beer is not just for breakfast anymore
06:34.57ptiggerdine~AMP
06:34.58jbotamp is probably a web based interface for configuring Asterisk. See http://amp.coalescentsystems.ca/
06:35.00drrayI don't think AMP makes asterisk easier
06:35.07mog_homeno it doesnt
06:35.09ptiggerdinedrray, agree
06:35.09mog_homejust harder
06:35.31doushanesspeaking of html front ends for asterisk, anyone try signates asterisk manager?
06:35.44ptiggerdineand the permission issues sux
06:35.49asterboyya, I can't leave my CLI
06:36.09ptiggerdinecomerical propiety shit
06:36.55ptiggerdineFLOSS!!!
06:37.52asterboyFor those that want AMP up and running fast. Coalescent Systems will install AMP for a small setup fee of $249 USD
06:37.55rezEditI started knowing NOTHING about this stuff and am glad I never tried AMP or anything else.
06:38.01rezEditdig into the configs
06:38.08rezEdityou learn more and have more control.
06:38.11asterboyso where is the download if I don't want them to install it?
06:38.38drrayI see why people want AMP
06:38.48asterboywhy
06:38.49drrayAMP just does not do it for me
06:39.12rezEditI wish there was something nice for a front end so that mere mortals could update configs as needed.
06:39.18*** join/#asterisk |dennis| (i=dennis@200.32.215.84)
06:39.31*** join/#asterisk MikeJ__ (n=ircatjer@d14-69-8-30.try.wideopenwest.com)
06:39.48drrayasterboy - because non * people want a way to manage extensions, my boss is already beholden to teh telco guys on the mitel.. I'm not sure he wants to be beholden to a unix guy too
06:40.04ptiggerdinethere is a OSX interface to asterisk that I like ( or looks a lot better than AMP)
06:40.11rezEditWe have been using a 3com NBX 100 system which has a half-decent web front end.... something anyone on our staff can use when need be
06:41.01rezEditptiggerdin: OS X interface?  you mean the sunrise people?
06:41.22drrayI just have not found a GUI that has grabbed me, I've dinked around with making a gkrellm tool for asterisk, and with a ncurses tool, but vi and asterisk -r work best for me
06:41.37asterboyI have used simple text forms to be sent via email to change many things in linux.
06:42.36asterboyzsh?!?
06:42.51rezEditLOL  It's what I learnt on
06:42.57asterboyhaven't heard that word since SCO
06:43.08rezEdithah!
06:43.16asterboylol
06:43.54rezEditit's the standard here at work, and we have a lot of sheel scripts that depend on it so.... I don't have much choice.
06:44.00rezEditer shell.
06:44.32asterboythere are substantial differences.
06:45.11drraycan't you just run the zsh scripts from bash?
06:45.12ptiggerdinerezEdit, how well does asterisk intergration into the 3com NBX 100/
06:45.20rezEdityeah, as I have had opportunity to discover.... I guess I don't use the shell for enough stuff that it matters
06:45.48rezEditptiggerdin: No integration here.  Completely replacing the NBX 100 which is about to die.
06:45.55ptiggerdineoh ok
06:46.15drrayI'm waiting for my last two mitels to die
06:46.16ptiggerdinebugger eh, would have being keen to do that.
06:46.24asterboyI guess you could zsh from bash.
06:46.29asterboyvisa versa
06:46.33bsdfreakheh
06:46.37rezEditasterboy: yup
06:46.39bsdfreak=]
06:46.50ptiggerdinesomething about support for legacy with asterisk makes it fun.
06:46.57drrayyes
06:47.04drrayor at least interesting for an afternoon
06:47.18asterboyI'm glad I skipped the whole PRI thing.
06:47.23rezEditour NBX 100 is in BAAAAAAD shape.  We can still make and recieve calls, but while customers can leave voicemail, we can't retrieve it.... it just hangs up on us.  The IMAP interface to voicemail has stopped working, and we have not been able to access the web interface for weeks.
06:47.28asterboyIP phones all the way baby!
06:47.35*** join/#asterisk MikeJ[Laptop] (n=ircatjer@d14-69-8-30.try.wideopenwest.com)
06:47.39*** join/#asterisk AFK1 (n=itsme@203.81.233.76)
06:47.43drrayI hate IP phones
06:48.14rezEditThe hard disk is dying a slow death, me thinks.
06:48.22ptiggerdineah ok
06:48.26asterboywhy so?
06:48.38drrayrezedit - have you thought of imaging the hard drive and doing a transplant?
06:49.17rezEditdrray: no.  The system is like 5 years old, and we wanted to go VoIP anyways.
06:50.09rezEditdrray: Plus I have no idea how to work with that thing.  There is a serial interface, but we have very few PCs here, and I wouldn't know what to do once I got a connection.  There is very little documentation out there, since 3com wants to charge for all support.
06:50.16drrayI have a serious hard on for zaptel
06:50.31asterboylol
06:51.26asterboyI just love the concept that you can take your IP office phone with you anywhere in the world and stay connected as though you were right at the office.
06:51.36drrayI also don't have a building setup where voip works
06:51.42drrayI have to use copper pairs
06:51.50asterboyright...zaptel it is!
06:52.00rezEditasterboy: that's another plus.  We have a few people off-site and this will save big on long-distance.
06:52.05asterboy(reference to brain candy
06:52.19asterboytotally
06:52.20drrayI like how zaptel, has a longer range.  is line powered..
06:52.32asterboywait till the wifi phones take off.
06:52.34drrayand just needs a pair
06:52.34*** join/#asterisk mog_home (n=mogorman@user-24-236-84-48.knology.net)
06:52.52asterboyI do like that too.
06:52.54rezEditasterboy: I already have them logging in to the system using X-Lite until I can send them their IP 501's.
06:52.57drrayour building has metal door frams, metal siding and flourescent lighting.. wifi will not work here
06:53.06asterboynice.
06:53.08drrayhotel faraday
06:53.14asterboyhow much did you get the 501s for?
06:53.25asterboylol, hotel faraday!
06:53.31tainted_omg atacomm is such a ripoff when it comes to shipping
06:53.38asterboyone big capacitor
06:53.40rezEditasterboy: not sure....  from voipsupply, but they had a black friday sale.
06:53.43AFK1hii all
06:53.54AFK1need some information about a server required
06:53.59drraythe building was built in 1993, so they went out of their way to not use cat3 or better cable in the walls
06:54.00rezEditasterboy: we had an order in but cancelled it in time and reordered under the sale.
06:54.00AFK1for putting up Digium 2400 model cards
06:54.30asterboydidn't know about the black friday sale...is that every friday?
06:54.37AFK1we want to put 24 FXS and 24 FXO in one server
06:54.50rezEditasterboy: no, just the Friday after Thanksgiving Thursday
06:54.50AFK1can any one suggest what server specs are required
06:54.50AFK1???
06:54.51asterboyno problem
06:55.26AFK1NEED HELPPPPPPPPPPPPPP guys
06:55.27asterboydarn, missed that...no big deal...got my IP 500s for $95 delivered.
06:55.37mog_home1 ghz
06:55.44mog_homeany machine you can buy these days
06:55.44asterboyAFK1, 80386
06:55.45rezEditasterboy: Yikes!  from where!?!?!?
06:55.46mog_homego to walmart
06:55.49mog_homeget an emachine
06:55.54mog_homeif you want to be a cheapo
06:55.56asterboyebay
06:55.56AFK1asterboy :)
06:55.58mog_homeif you want to spend it
06:56.05rezEditasterboy: I am sure we paid like $200 a pop.
06:56.06mog_homeget a dell or some server
06:56.07drrayalmost any PC will work with with one zaptel card
06:56.13mog_homeexactly
06:56.29asterboyAFK1, seriously...you can use any new machine...plenty of jam.
06:57.03asterboyIf you can afford the 2400s you can afford a dual core P4
06:57.06AFK1asterboy: we want to terminate 24 - live PSTN in 24 FXO card and then forward these out of the 24 - FXS card in the same server, in the process we want to record all ongoing calls, basiclaly we r trying to use ASterisk for recording services
06:57.10drrayI need to find some chumps to buy my TDM4xxp cards so I can get a new TDM2400 cared
06:57.34asterboyAFK1, buy 2 servers exactly the same.
06:57.36j4m3swhat revs?
06:57.39asterboyshare load them.
06:57.47rezEditAFK1: They don't lie.  I have asterisk running with 1, 4-port card on old Power Mac hardware and it never even begins to get anywhere near coughing while serving 30 phones.
06:58.13j4m3sdrray, what rev tdm?
06:58.24asterboyya, the digium cards your buying, preferrably with hardware echo, will do most of the work.
06:58.28*** join/#asterisk krisguy (n=krisguy@h216-170-039-057.adsl.navix.net)
06:58.31AFK1two servers, the solution we r putting is for like 48 ports , that means 4 servers in all??
06:58.40AFK1rezEdit :) i m sure they dont
06:58.42drrayI'd need to look, but it was from the dev kit back in the day.. when it included x100p
06:58.54j4m3sdo you still have the x100ps?
06:58.55AFK1we will definitely go for the echo cancellation cards,
06:58.58rezEditasterboy: Seriously, where did you get your IP501's?
06:59.04drrayj4m3s - yes
06:59.19j4m3swould you bundle them together?
06:59.20asterboyseriously...ebay
06:59.34rezEditasterboy: ahhh.  Ok.
06:59.39AFK1asterboy: Do you think the way we r trying to use asterisk for recording calls is good enough?
06:59.41*** join/#asterisk magic_1 (n=maig_1@dsl-165-149-85.telkomadsl.co.za)
06:59.44drrayhell yeah, I only have 1 x100p
06:59.46rezEditasterboy: Thanks.
06:59.47asterboyyes
06:59.54asterboyno prob
07:00.06magic_1lo all
07:00.19drrayAFK1 - that will work
07:00.29AFK1i was thinking of putting up 2 x 2400 in server class machine with 1 x Xeom , 1 GB RAM, for recording 24 ports
07:00.36rezEditTime for me to head home folks.  Have a good night, all (or day as the case may be).
07:00.39asterboyAFK1, totaly...you only need 1 server...but 2 is nice in case one goes offline...you can transfer cards to the remaining working one.
07:01.41drrayI don't know that you need that much hardware either (cpu wise)
07:02.01asterboyI concentrate on redundancy.
07:02.26asterboyWhen phones go down....everyone gets excitted...when computers go down...they go for a coffee break.
07:02.51magic_1hi all -AFK1-would u possibly know how i would integrate the AMP with my existing asterisk box
07:04.08asterboyfrom what I have heard here... "rm -rf AMP*" and "asterisk -c" seem to be popular.
07:04.43drrayif he wants to run AMP, let him
07:04.59drrayhell, he could be the one to make it suck less
07:05.04asterboylol
07:05.08magic_1hehehhe
07:05.28drrayand I'm not saying AMP sucks, just personal preference
07:05.55drraythe orrielly book is the best GUI for asterisk
07:06.08asterboynot tried it...but after reading the consensus here, I'll be happy with a shell prompt.
07:06.28drrayasterboy - it's like using a cruise missle to remove a stump
07:06.36asterboylol
07:06.40drrayit tries to be everything
07:06.51magic_1its just that i am rightfully a newbie and,asterisk is not the easiest to get going ,i heard that  AMP might be the eassiest to start of with
07:07.08drrayasterisk by itself is the easiest
07:07.16magic_1i wanted to use asterisk and not asterisk@home cause i want to do this properly
07:07.18asterboyya gotta agree there.
07:07.19magic_1heheheheh
07:07.27drrayAMP,FOP and anything else won't work if your base asterisk install is befuct
07:07.36magic_1heheheheh]
07:08.06magic_1nah my asterisk is working ,just i dont know how to work it
07:08.13asterboyno...the proper way would be to buy an Asterisk book a developer kit and play around.
07:08.19magic_1and i would really like to learn
07:08.23drraymy big beef with FOP is that he uses flash, and flash plus firefox has killed every OS I've ever run it under
07:08.29magic_1that is true
07:08.45brookshireat least it's not java
07:08.49magic_1i have heard similar things
07:08.52drrayhey, I like java
07:08.57magic_1heheheh
07:08.58brookshirei'm sorry
07:08.58brookshire:)
07:09.14brookshirei like programming in java.. but i hate running it
07:09.15brookshirelol
07:09.25drraywell, I don't like java, but I like being able to sell applications to linux/osx/windows/bsd/solaris
07:09.39magic_1my thing is i am having trouble (as stupid as it is )setting up a simple ext
07:09.42magic_1heheheheheh
07:09.48magic_1hehehehehe
07:10.02drraymagic - what kind of hardware do you ahve?
07:10.47magic_1AMD 64 3500+ 1gig corsair 80 serial ATA and GIGabyte motherboard
07:10.56drrayI'm sorry, what kind of asterisk hardware?
07:11.12magic_1also running fedora core 3 64bit with asterisk 1.2
07:11.19*** join/#asterisk MikeJ[Laptop] (n=ircatjer@d14-69-8-30.try.wideopenwest.com)
07:11.27drrayso you want to setup sip extensions?
07:11.34magic_1true
07:11.52magic_1cause i dont have zatel hardware i use ztdummy
07:12.01magic_1as i am sure that u know
07:12.10drraysip softphones?
07:12.24magic_1no grandstram phone
07:12.39magic_1i meant grandstream
07:12.40magic_1heheh
07:12.42magic_1sorry
07:12.55drraybudgetone was part of my "dev kit"
07:13.01magic_1nice
07:13.08asterboyare they any good?
07:13.09magic_1i quite like this phone
07:13.20magic_1very have had no hassle to thus far
07:13.29asterboyprice sure is good
07:13.39drraythey are like a $50 gun, they are better than nothing in a pinch, and should be used to get you a better gun
07:13.52drrayI've since upgraded to cisco 7960's
07:13.59magic_1my colleuge is running a *@home solution and is having no hassle with it
07:14.15magic_1cisco is very nice but here in SA quite expensive
07:14.32drrayyeah, it's expensive here too
07:14.36magic_1hehehehe
07:15.32asterboypolycom seems to balance cisco quality with price.
07:15.32*** join/#asterisk MikeJ[Laptop] (n=ircatjer@d14-69-8-30.try.wideopenwest.com)
07:15.32magic_1yeah but it is worth it
07:18.25magic_1my problem mainly is ,is that i am a network admin and engineer,i have never worked with asterisk before like this,only the hardware setup ,an have now dicided to get more acquanted with it so there it is all quite new to me
07:18.26drrayhttp://www.voip-info.org/wiki-Budgetone
07:18.26trixteris there emphasis on the budget part of budgetone?
07:18.27magic_1hehehe
07:18.32drraythe budgetone is not a bad phone
07:18.39drrayfor what it is
07:18.50magic_1true
07:18.58magic_1i quite like it
07:19.06drraymine has been on the shelf unplugged for a year or so
07:19.10orlokmagic_1: hey, have you heard of Obsidian?
07:19.15magic_1yeah
07:19.31magic_1use it u wont be sorry
07:19.53magic_1just make sure u do all the firmware upgrades
07:20.02orlokmagic_1: mate of mine went over there a few years ago to do some stuff with them
07:20.10magic_1nice
07:20.20orlokyeah, nandos stuff
07:20.27orlokapparently thye are cool guys
07:20.30magic_1hehehehehe
07:20.44orloknandos, portugese chicken from south africa :)
07:21.01magic_1heheheheheh
07:21.10magic_1hows that for u
07:21.28orloklove nandos, we had some for lunch today :)
07:21.57magic_1nandos rocks
07:25.25drrayYate for windows looks interesting
07:26.03mog_home?
07:26.59*** join/#asterisk Assid (n=assid@203.115.64.62)
07:27.19*** join/#asterisk ORiON2012 (n=orion@cpe-70-117-0-232.satx.res.rr.com)
07:27.26magic_1so any one got an idea how i can get am installed in asterisk
07:28.28sylequestions like that will get you very hated in this channel
07:28.43magic_1heheheheheh thanks for the heads up
07:29.16drrayahve you looked on the wiki?
07:29.19drrayhttp://www.voip-info.org/wiki-Asterisk+Management+Portal
07:29.28ORiON2012Quick question from a noob. I have voip service from Time Warner.  Their system plugs into a POTS jack in my home and "energizes" the other jacks to enable use of my regular phone. Is something like that possible with asterisk?
07:29.35magic_1its just i cant get a ext running on my asterisk 1.2 and from what it seems it looks so straight forward
07:29.48QwellORiON2012: get a/some FXS card(s)
07:29.54Qwellcard/port
07:30.04mog_homeQWELL!
07:30.08QwellMOG!
07:30.20Qwelltip: noload => chan_skinny.so
07:30.24Qwellit helps ;]
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07:31.17mog_homeman i need to make asterisk run with less ram
07:31.18ORiON2012Qwell: do I have to hook the phones up directly to the card, or can I use the existing wiring in my home?
07:31.26mog_homeits running pretty fat on my little server
07:32.50brookshiremog: goto sleep :)
07:33.31trixter21M on mine with 9M resident
07:33.32trixter1.2
07:33.49drrayI don't even look at ram
07:33.51trixterseems high for what its doing, right now there arent even any calls
07:40.22mog_homenaver
07:40.30mog_homeim goin in early tommorrow
07:42.02*** join/#asterisk MikeJ[Laptop] (n=ircatjer@d14-69-8-30.try.wideopenwest.com)
07:44.51*** join/#asterisk MikeJ__ (n=ircatjer@d14-69-8-30.try.wideopenwest.com)
07:47.40QwellORiON2012: sometimes you can use the existing wiring.  not always
07:47.50Qwelldepends on how it comes out of the box
07:48.38mog_homeman i need more ram for this box...
07:49.20Qwellram is always good
07:49.40mog_homei have 64mb
07:49.48mog_homewhich is good for a 250mhz machine
07:49.56mog_homejust no dma on the box....
07:50.07mog_homemeh its good enough for what i need it to do...
07:50.36*** join/#asterisk SERGEUS (n=s@195.112.98.13)
07:51.24*** join/#asterisk MikeJ[Laptop] (n=ircatjer@d14-69-8-30.try.wideopenwest.com)
07:53.00asterboyya I have a 450 with 64Mb ram for my asterisk...doesn't skip a beat.
07:53.19Qwellman...my 110mhz router has 4x that much..
07:53.25asterboylol
07:54.02asterboyI'd like to get my hands on one of those PowerMac 9600s
07:55.52asterboyshould put a wanted ad on Google Base
07:58.32drrayok, what is google base?
07:58.37drrayurl?
07:58.48drraym,
07:58.50drrayer,nm
08:01.33*** join/#asterisk MikeJ[Laptop] (n=ircatjer@d14-69-8-30.try.wideopenwest.com)
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08:01.51asterboyback
08:02.01asterboyjust checking out google base.
08:02.17asterboyhttp://base.google.com/base/default
08:03.53*** join/#asterisk Math[laptop] (n=math@modemcable148.4-81-70.mc.videotron.ca)
08:04.39Math[laptop]I setted up dundi between 2 servers but, calls are failing with "Call rejected by 10.0.0.5: No authority found"
08:04.53Math[laptop](using iax2 as tech)
08:04.54asterboylol "setted up"
08:05.08Math[laptop]yeah well... you know.. :P
08:05.16asterboy:P
08:05.36Math[laptop]this is without mentionning the fact that asterisk advertised itself as 127.0.0.1 on the dundi net
08:07.04asterboyhave not done dundi yet...sounds like a great directory tool if you can get it working.
08:07.24Math[laptop]its just the part of dundi that handles authentication didnt register itself into the iax2 user list
08:07.46Math[laptop]I can always failover by creating a "dundi" user in iax2 with no password, but its kind of a security flaw
08:08.11Math[laptop]especially when pstn routes are announced
08:08.30asterboyI'll have to get into that.
08:08.44asterboynight for now.
08:08.46Math[laptop]nite
08:09.02*** join/#asterisk MikeJ[Laptop] (n=ircatjer@d14-69-8-30.try.wideopenwest.com)
08:09.25*** join/#asterisk Gunnar (n=gunnar@bkkb-gw.bitcon.no)
08:09.37Math[laptop]hey MikeJ[Laptop], did u play with dundi on head?
08:11.16perlmonkeeI'm having an Asterisk+SipPhone.com problem.  I can place calls out through SipPhone.com, and `sip show registry` says I have registered, but when I (or others) try to call my SipPhone number, it goes straight to SipPhone voicemail, as though I weren't registered.  If I have my soft phone register directly with SipPhone.com, I can recieve calls no problem.
08:11.23perlmonkeeCan anyone shed any light on this?
08:11.38perlmonkeeIs there something I have to do other than reguster => user:passwd@proxy01.sipphone.com ?
08:11.47perlmonkeeregister*
08:11.58Math[laptop]uhm yeah
08:12.03Math[laptop]register => user:pass@host/exten
08:12.12Math[laptop]context=the_context_you_want
08:12.19Math[laptop]and define the exten
08:12.24Math[laptop]oh ok
08:12.32perlmonkeeI've followed the examples on Voip-info.org
08:13.00Math[laptop]well... whats you dialplan for the incoming call?
08:13.43perlmonkeejust a moment.
08:14.28perlmonkeeexten => ${SIPPHONEUSERID},1,Macro(stdexten,100,SIP/perlmonkee)
08:15.19Math[laptop]can u pastebin your extensions.conf
08:15.52Math[laptop]because I think thats not the way to call stdexten, but Im not sure til I use ael with home-made-stuff(tm)
08:16.33perlmonkeewell, here is an example of a working stdexten call:
08:16.36perlmonkeeexten => 100,1,Macro(stdexten,100,SIP/perlmonkee)
08:16.49Math[laptop]ah
08:16.56perlmonkeethat one works no problem.
08:17.18Math[laptop]and, when you run asterisk and someone calls you, what do you see on the CLI
08:17.23Math[laptop](with a lot of -vvvvvvvvvv)
08:17.36perlmonkeeIf someone tries to call my SipPhone number - I see nothing.
08:17.40drrayset verbose 10
08:18.00Math[laptop]then try again
08:18.00perlmonkeeokay, set to 10 - trying again.
08:18.03*** join/#asterisk lorinc (n=ang@caracas-0554.adsl.interware.hu)
08:18.07perlmonkee(it was at 4)
08:18.29Math[laptop]k
08:19.08*** join/#asterisk kks (n=kks@202.73.11.205)
08:19.25perlmonkeeI see my outgoing call being setup and then sipphone.com answering it.
08:19.38perlmonkeeand sipphone.com gives me the "This user is currently offline" message.
08:20.17*** join/#asterisk MikeJ[Laptop] (n=ircatjer@d14-69-8-30.try.wideopenwest.com)
08:20.18perlmonkeenothing about any incoming calls.
08:22.28*** join/#asterisk MikeJ[Laptop] (n=ircatjer@d14-69-8-30.try.wideopenwest.com)
08:22.33Math[laptop]perlmonkee, whats your number
08:22.44Math[laptop]maybe your not allowed to receive and send a call at the same time
08:22.59Math[laptop]lemme call you
08:23.34perlmonkee17476278960
08:23.52delox99Math[laptop]: are you behind a router or firewall?
08:26.25perlmonkeemy asterisk box is not.
08:26.26Math[laptop]working?
08:26.26Math[laptop]ringing here
08:26.26perlmonkeeI am not recieving any call.
08:26.26Math[laptop]1-747-627-8960
08:26.26perlmonkeenothing on the console =/
08:26.26perlmonkeeyup
08:26.26Math[laptop]what about sip debug peer [peername]
08:26.26perlmonkeeokay, just a moment.
08:26.29perlmonkeeenabled
08:26.29perlmonkeeDestroying call '3d40662a245d05342bc05ddc59af4042@192.168.1.1'
08:26.40*** join/#asterisk chapeaurouge (n=chap@85.201.80.52)
08:26.42perlmonkeethats all I've seen since enabling it.
08:26.49*** part/#asterisk doushanes (n=doushane@c-67-173-1-227.hsd1.il.comcast.net)
08:27.00Math[laptop]k calling
08:27.45perlmonkeewoah - okay - a bunch of crap.
08:28.19Math[laptop]didnt call ye tlol
08:28.29trixtergee yahoo is doing voip now to fight google..  wonder if its like sonys 'voip plus' wherei ts only on net calls and doesnt in any way allow a regular phone call (that is all I have seen from yahoo) to further confuse people..  a HIGH percentage of people that are voip aware dont know you can call PSTN  numbers with any voip
08:28.45perlmonkeeit was all registration crap
08:28.55Math[laptop]damn the cat5 wire of my ATA broke
08:28.56Math[laptop]brb
08:28.58*** join/#asterisk MikeJ[Laptop] (n=ircatjer@d14-69-8-30.try.wideopenwest.com)
08:31.52Math[laptop]dialing
08:32.44*** join/#asterisk spunz_ (n=spunz@h081217096096.dyn.cm.kabsi.at)
08:33.35Math[laptop]got anything
08:33.55Math[laptop]answer+hangup?
08:35.18perlmonkeenothing
08:35.52perlmonkeejust registration crap.
08:36.02perlmonkee>_<
08:36.18perlmonkeeokay, I'm going to start all over with sipphone - erase everything I have and write it all again.
08:37.04perlmonkeesip.conf relevant stuff:
08:37.28perlmonkeeregester => number:passwd@proxy01.sipphone.com/perlmonkee
08:37.47perlmonkee[sipphone]
08:37.50perlmonkeetype=peer
08:38.05perlmonkeehost=proxy01.sipphone.com
08:38.14perlmonkeecanreinvite=no
08:38.18Math[laptop]~pastebin
08:38.19jbotmethinks pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca/
08:38.23perlmonkee=(
08:38.23perlmonkeesorry.
08:38.29Math[laptop]:)
08:39.02Math[laptop]er... register => user:pass@host/extension
08:39.09Math[laptop]the exten must be a digit
08:39.13Math[laptop]or a number
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08:42.38magic_1at the host part can i put the host ip instead is that possible
08:42.51perlmonkeehttp://pastebin.com/453746
08:43.54magic_1perlmonkee,may i ask u a question
08:44.13perlmonkeeYou may.
08:44.19perlmonkee(how polite)
08:45.29perlmonkeeokay... so I made that change - verbosity is set to 10, debugging for peer sipphone is enabled.
08:45.41perlmonkeeif anybody wants to try a call - I'd appreciate it.
08:45.59magic_1thankz
08:48.31perlmonkeeI'm not too great with this myself just yet - but setting up an extension is pretty easy.  I'm sure if you ask a more specific question I, or any number of more experienced people here would be happy to try and answer it.
08:49.03magic_1thanks i appreciate
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09:01.30niZonsomeone tell me why x-lite uses more ram than windows explorer
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09:03.31pifhi, with what directive can I test an IAX link for latency?
09:06.42drraypif, i've dialed in to my asterisk box with the milliwatt() app and listened for defects
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09:09.48manypif: qualify=yes ; iax2 show peers
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09:13.05kkssorry to jump topic, Is anyone get MYSQL cmd work asterisk realtime? i'm working with Asterisk 1.2.1 and Asterisk-addon 1.2.1
09:15.51pifmany : thanks, qualify=yes checks for peers every 60s, right?
09:16.12pifbut not the latency to the peer
09:19.51manyit does check the the latency then
09:20.34pifso qualify=1000 will skip a host over 1000ms latency?
09:21.08manyi think it declares a host lagged when > 1s and unreach when > 2s
09:21.38pifI'm googling but can't find authoritative info on that
09:21.45manytx-rico          172.21.253.254  (S)  255.255.255.255  4569          OK (34 ms)
09:21.54manyUTSL :)
09:22.18pif?
09:22.27manyuse the source luke
09:22.32pifah :)
09:22.39manythe only authoritive information you will ever find is in the source.
09:22.50manyyou know, the config files whose extension is .c
09:22.59many:-P
09:23.12pifdude
09:23.34manyYea? :)
09:23.53*** part/#asterisk ORiON2012 (n=orion@cpe-70-117-0-232.satx.res.rr.com)
09:23.57pifhow do you test for lagged status before Dial'ing ?
09:24.14manyoh. i dont.
09:24.32pifor does Dial return a failure?
09:24.51manyno idea, honestly. i didnt ever had calls with high lat.
09:25.14manythe 30ms you see above is my sucky dsl over a vpn
09:25.21manyor rather a vpn over my sucky dsl
09:26.22manyor rather: iax2 over tcp over vpn over pppoe (1000/128kbit)
09:26.30manyerr, i give up, you get the idea. :)
09:26.38Qwellkludge
09:26.53pifbetter to let iax2 packets on the wild internet
09:27.23manythe whole fucking IT is a single sucking kludg.e
09:27.35manysorry. should watch my word.
09:27.36manys
09:27.40piflayers of encapsulations mess with error correction of voip protocols
09:28.13manyerror correction? in a tcp based protocol? uh. well..
09:29.13pifyou vpn packets are tcp based (will wait and retry) when the aim of UDP/voip is immediate delivery
09:29.23manyyoure wrong
09:29.26manymy vpn runs over udp
09:29.40manyand iirc iax2 isnt udp, but tcp.
09:29.44Qwelludp
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09:29.47pifno sir
09:29.53pifQwell : yes
09:30.02manyokay, so why did i think that iax2 did not suck, then?
09:30.05manymhh.
09:30.14pifudp is good for voice
09:30.14Qwellsccp is tcp ;]
09:30.17mog_homeiax is the coolest....
09:30.26Qwellmog_home: go to bed
09:30.31mog_homenever
09:30.32manypif: yeah, basically yea.
09:30.51QwellI got stuck watching a police chase by my house...
09:30.54manybut udp is a good shoot-in-the-foot with alot of soho routers and stuff.
09:31.06Qwellwent on so long though, that the helicopter ran out of gas, and had to take off.  heh
09:31.23mog_homelol thats funny
09:32.07pif"out of gas" + "had to take off" = core dumped
09:33.25iDunnoQwell: surely it had already taken off, hence it running out of gas... ;)
09:33.58iDunno9.30am and I'm already slapped, interesting.
09:34.11pifok, notice to all west-coasters : herbal tea, jerk off and to bed
09:34.51pifattaboy
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09:36.38manydig chicks instead of just jerking off
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09:38.38pifchicks, you mean all that flesh around the pussy?
09:39.09manyYah.
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09:42.13mog_homeanyone good with iptables
09:42.23mog_homei have  a really simple firewall i want to set up
09:42.40mog_homeeh never mind
09:42.45mog_homefound info i needed
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10:15.24perlmonkeeDoes anyone know of a service that you can have call you to test your incoming call routing?
10:15.34perlmonkeeI know I used one about a week ago, but I can't remember where it was.
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10:18.41shido6what do you mean perlmonkee ?
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10:20.09DandreHello all,
10:20.30DandreI have a problem with fop and asterisk 1.2
10:21.08RoyKfop?
10:21.13RoyKFoIP?
10:23.22drrayflash
10:23.26mog_homeflash operator panel
10:23.27drrayoperator
10:23.31drraypanel
10:23.55perlmonkeeshido6: I'm looking for a form I can put my SIP address in and then some server that processes this form will initiate a call to me.
10:24.00perlmonkeethus testing my incoming call routing.
10:27.10perlmonkeeI'd still like to find a service like I mentioned though (especially since I used one not very long ago)
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10:34.45Dandresorry guys I have been disconnected
10:35.23Dandredid you seen my previoux post about FOP?
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10:42.20DandreIs there any one here?
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11:08.01KriS83Hi... could someone have a look at this, and tell me why ${foo} is empty? or what I am doing wrong?
11:08.05KriS83http://pastebin.ca/32944
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11:09.34MassiveBluehi there
11:09.47KriS83Hi
11:10.03iDunnoAh ha! That's what I was looking for, the s option to ChanIsAvail :)
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11:22.40dario77hi
11:23.37dario77i have a problem concerning asterisk & rtcp, can you help me?
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11:35.13netwindhmm
11:35.30netwindjerjer doesn't appears here ?
11:35.50netwindwich openh323 version are recommended for now?
11:36.13netwindi can't build char_h323 in new asterisk 1.2.1
11:36.42netwind1.0.10 build is mailfunctional
11:40.19shido6chan_h323 ?
11:40.45netwindof course chan_h323
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12:03.32shido6is there a nagios check plugin?
12:03.46shido6for mysql
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12:08.38zoashido there is
12:08.41zoaive seen one before
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12:35.41MassiveBlueDec 7 18:56:14 WARNING[5135] chan_iax2.c: Firmware file '/var/lib/asterisk/firmware/iax/iaxy.bin' fails checksu
12:35.41MassiveBluem
12:35.44MassiveBlueoops
12:36.51MassiveBluei get warnings "Firmware file '/var/lib/asterisk/firmware/iax/iaxy.bin' fails checksum" - anybody knows something about this warning?
12:37.12KriS83Hi... could someone have a look at this, and tell me why ${foo} is empty? or what I am doing wrong?
12:37.14KriS83http://pastebin.ca/32944
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12:44.44nextimeanyone using astbill?
12:46.23mrtwisternextime: i use
12:46.33iDunnohmmm
12:46.41mrtwisternextime: and areski and sometimes mcc :)
12:47.31iDunnoor rather, how it works.
12:47.52nextimemrtwister : i'm trying to test it, but i have some trubble creating the database tables, 20 asv* tables are "in use" from mysql using the default .sql to install, is it a common error or so on?
12:48.29iDunno(hmm, not as I want it to, damn)
12:48.31mrtwisternextime: you installed mysql 5?
12:48.36nextimemrtwister : of course yes
12:48.38mrtwisternextime: and what os you have?
12:48.46nextimedebian sarge, mysql 5.0.16
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12:48.58mrtwisternextime: hm, i have no issues on debian
12:49.25mrtwisternextime: on fedora core got error once, but later corrected... do full reinstall of mysql
12:50.09nextimemrtwister : mysql is working great, i don't think that is the problem ( i have a 18 gigs complex db on the db server.. )
12:50.10mrtwisternextime: btw i still not know, imho areski at present is better, and with astbill you cannot use AMP portal
12:50.36nextimemrtwister : i hate AMP, it don't reflect my need, areski the same
12:50.40mrtwisternextime: week ago i installed astbill, no problems at all.
12:51.01mrtwisternextime: try mcc also :) there is no AGI
12:51.16mrtwisternextime: compiled c program. www.paskambink.lt/mcc/
12:51.53nextimei will look mcc
12:51.59mrtwisternextime: astbill i instaleld and test and maybe will use. good that it integrated to CMS drupal, where i can use other modules and do e-commerce and other things
12:52.36nextimemcc need postgres at the momenty
12:53.12nextimei don't want to install another db server while i have a mysql cluster in production
12:53.46mrtwisternextime: you have to use simple machine for gaming and tests :)
12:53.51mrtwisternextime: or vmware
12:54.21mrtwisternextime: anyway, i had no problems with astbill, but there at forum lot of topics, seems product is not finished until end.
12:54.53nextimemrtwister : yes, but i must test something that i can use in the production environment :)
12:56.11RoyKzoa: ping
12:56.21nextimei think that i will return to use my simple mod_python based web administration, i'm tired to test something that don't work at 100% or isn't good for my use, anyway, thanks for your opinions.
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13:07.06onasmrtwister :)
13:07.20mrtwisteronas: ?
13:07.41onasmrtwister entusiastic page www.paskambink
13:08.58sivanaYahoo Messenger adding computer-to-phone capabilities
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13:24.00gnosysanyone here ever connected a Cisco 7960 phone to the Free World Dialup service?  Could you offer some pointers?  I just got through connecting to FWD using X-Lite from CounterPath, so my next step is seeing if I can do the same with my hardware SIP client (the Cisco 7960... loaded with SIP firmware, version 7.1).
13:24.48asteriskmonkeygnosys: you shoul have no problem if you if you use the same settings
13:26.36gnosysthe thing that's confusing me is the differently-named settings in the Cisco.  I have the 7960 working with my DHCP and TFTP servers, but when it boots up, it's telling me that it's unprovisioned.  In the phone status message center (LCD screen), it's complaining about W351 and W350 (unprovisioned proxy_emergency and unprovisioned proxy_backup, respectively)
13:27.55gnosysAnd I don't get a dial tone when I pick up the handset...
13:28.19bintutwhat is the best digium card with 4 FXOs you can recommend? i find confusing the digium website.
13:30.13asteriskmonkeybintut : get a tdm400 with 4 fxo's on
13:30.26asteriskmonkeygnosys: have you provisioned it yet usiung the tftp?
13:30.48asteriskmonkeygnosys: open a log window to see if it grabs the config from the tftp
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13:31.12DandreHello,
13:31.25RoyK~seen zoa
13:32.57jbotzoa is currently on #asterisk (1h 46m 26s).  Has said a total of 2 messages.  Is idling for 1h 24m 16s
13:32.58gnosysopen a log window... by telnetting into the phone?  or is that accessible from the lcd?
13:32.58asteriskmonkeybintut: http://www.digium.com/index.php?menu=product_detail&category=hardware&product=TDM400P
13:32.58zoaim here royko
13:32.58RoyKoki
13:32.58asteriskmonkeygnosys: where is your tftp program? run the monitor there to see if the phone connects and grabs the file from it
13:32.58RoyKzoa: i just tried the latest version of the jb
13:32.59DandreI am trying to use ALERT_INFO with 1.2. It seems to be different in 1.2 as in 1.0.7 but I can't find the docs on voipinfo :-(
13:32.59zoapatch3 ?
13:32.59gnosysdone that already.  it does load the SIPDefault.cnf and SIPmacaddress.cnf.  I guess that maybe I've set something incorrectly in one of those two files...
13:32.59zoawhat happened ?
13:32.59RoyKzoa: yeah
13:32.59RoyKzoa: sound is terrible
13:33.04RoyKzoa: that is, incoming audio on the asterisk server
13:33.24RoyKzoa: can i have your email address once more? i can email you a monitored file
13:33.30zoacan you post a message on mantis with the flow ? ( i mean what protocol to what protocol + what options so that slav can have a look at it ?
13:33.34zoajoachim@securax.be
13:33.51asteriskmonkeygnosys: most likley a spelling mistake in your config file :)
13:34.07RoyKzoa: sent
13:34.31zoathnx
13:34.49zoabe sure to post the message on mantis
13:34.54bintutasteriskmonkey: thanks for the link
13:35.07gnosysI guess that could be asteriskmonkey...  i thought I reviewed it pretty carefully for that.  any chance i could get you to look it over for me?  maybe send you a file or something?
13:35.18RoyKzoa: left channel is me talking from a sip ata (that black one) using a 1024/200 adsl line. the other side is a gsm phone
13:35.53zoais the quality only bad with patch3 ? or also with patch 2 and 1 ?
13:36.16asteriskmonkeybintut: no prob, if your in canada let me know i can give you the number for this distributor :)
13:36.18RoyKiirc same with the older ones
13:36.23zoaaha
13:37.30gnosysthe other thing, asteriskmonkey, is that I'm using STUN in X-Lite to go through my natted firewall by linksys to the fwd proxy, and I don't see settings in the 7960 for doing that...
13:37.50zoaRoyK, can you also try it with the other 2 models ?
13:39.14asteriskmonkeygnosys: you are behind a firewall or nat i take it then?
13:40.04DandreI have found, there must be an underscore before ALERT_NFO!
13:40.29MassiveBluei get warnings "Firmware file '/var/lib/asterisk/firmware/iax/iaxy.bin' fails checksum" - who knows something about this warning?
13:42.09asteriskmonkeysounds like you have a corrupt file
13:42.20asteriskmonkeydownload it from cvs
13:42.55asteriskmonkeyasterisk automatically upgrads the iaxy firmwares as soon as iaxys connect to that asterisk server :) sweet
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13:44.46MassiveBlueasteriskmonkey: i downloaded this file from CVS on 16 Nov 2005 17:46:59 -0000
13:44.54kletter-matze<PROTECTED>
13:45.02kletter-matzeif I dial out, I can see at the cli "Executing SetCallerID("SIP/sysadmin-105c", "+49711123440")
13:45.11kletter-matzeis there anything else I have to do?
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13:46.03bintutasteriskmonkey: i'm in south east asia..
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13:49.07tengulrehi,all
13:50.03tengulreanybody active?
13:51.10zoaim not here
13:52.35RoyKzoa: will do
13:53.17Cinenanyone here had any luck with Dundi? I asked in #Dundi but nobody is home
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13:56.50RoyKwtf?
13:56.55RoyKsame noisy sound without the jb
13:57.00tengulreanybody building CALL CENTER with asterisk ?
13:57.58asteriskmonkeymassiveblue: try downloading from head
13:58.11mrtwistertengulre: asterisk@home is small call center :)
13:58.27mrtwistertengulre: it is possible in other words... but need lot of work
13:58.38asteriskmonkeytengulre: call centers are easy to build with asterisk :D
13:58.53gnosysasteriskmonkey: yes, behind a linksys natted firewall.
13:59.08MassiveBlueasteriskmonkey: okay I'll try
13:59.11mrtwisterhow to compile zaptel on debian. i have installed kernel sources, but have to do some operations before starting compilation. where i can find faq?
13:59.17gnosysshould I not expect the 7960 to be able to handle that with FWD then?
13:59.56asteriskmonkeygnosys: open the corresponding porst then on the router then and make sure nat is mapping the right ports to that device
14:00.39gnosysSo I guess that implies that the 7960 cannot do STUN like the X-Lite client does?
14:00.43asteriskmonkeymrtwister: use google for compiling files on linux
14:00.56asteriskmonkeygnosys: nope :)
14:01.11gnosysOK.  I'll try your suggestion then.  Thank you!  :-)
14:01.21asteriskmonkeythe 7960 is a much older phone.. i havnt seen there latest software though
14:01.23asteriskmonkeyno prob
14:01.30tengulreasteriskmonkey: how many lines does asterisk support?
14:01.33mrtwisterasteriskmonkey: not found proper doc.. still looking :)
14:01.56mrtwistertengulre: no limits, depends of 'style' of usage and cpu
14:02.03asteriskmonkeymrtwister: do you have gcc and all the other things installed required to build things properly?
14:02.16mrtwisterasteriskmonkey: yes.
14:02.20asteriskmonkeytengulre: as many as you can build a system to handle
14:03.25asteriskmonkeymrtwister: go to the directory you have the make/source files in and type make clean; make install;
14:03.47asteriskmonkeymrtwister: or on bsd style boxes one like :) make clean install
14:03.53tengulreasteriskmonkey: I want to building a 30 analog lines and 10 clients?
14:04.15[TK]D-Fendertengulre : very easy
14:04.28[TK]D-Fendertengulre : You mean 30 phone lines (from telco)?
14:04.29asteriskmonkeytengulre : get an t1/e1 card
14:04.31tengulremrtwister: are you use digium cards?
14:04.46asteriskmonkeyget something like the t11op
14:04.54tengulre[TK]D-Fender: yes!
14:05.00asteriskmonkeyso you want your users to have 30 lines 3 lines each :) nice
14:05.04iCEBrkrWerd up!
14:05.12tengulreasteriskmonkey: does it support chinese telcom?
14:05.14mrtwisterasteriskmonkey: i know it. but i have to do some thinks with kernel source to prepare it for usage
14:05.16[TK]D-Fendertengulre : where are you located?
14:05.30tengulre[TK]D-Fender: china!
14:05.32asteriskmonkeytengulre: what the hell is chinese telecom ? :P
14:05.33mrtwistertengulre: i like sangoma, but digium also ok, works :)
14:05.36tengulreXi'an of china
14:05.45asteriskmonkeyah... should do fine
14:05.54asteriskmonkeydo you use t1 or e1 there/
14:06.10[TK]D-FenderGet off of analog lines and get a T1/E1/J1 (whichever is appropriate to your area) digital link and you're set.
14:06.26tengulreasteriskmonkey: I m worried it's different chinese
14:06.52iCEBrkrmrtwister: How's the config/setup for the Sangoma cards?
14:06.55[TK]D-Fendertengulre : I'm sure your area uses one of the 3 major standards.  Do you use normal POTS phones on those lines?
14:07.04asteriskmonkeytungulre : t1 and e1 are standards ask which your telco provides
14:07.13[TK]D-FenderiCEBrkr : WANCFG = Creemy goodness :)
14:07.29mrtwisteriCEBrkr: ? it is well described in docs, lot of examples online...
14:07.44iCEBrkrI'm thinking of getting one.
14:07.53iCEBrkrBut I'm familiar with Digium's cards.
14:07.54tengulreasteriskmonkey: really?
14:08.22iCEBrkrIt's unfortunate that I don't have much time to muck around.
14:08.43asteriskmonkeytengulre: yep
14:08.46mrtwisteriCEBrkr: almost same sangoma
14:08.54coppicetenguler: why do you want to use analogue lines? are E1s hard to get in your area?
14:09.07mrtwisteriCEBrkr: only need to install wanpipe, all is on CD, no errors all works :)
14:09.09asteriskmonkeythink he just dosnt know telecom stuff
14:10.03[TK]D-FenderiCEBrkr : very easy to setup and a very solid experience.
14:10.14tengulreanybody know who are seller in this channel
14:10.34[TK]D-Fendertengulre : Not for telephone service in our are I would think.
14:10.39[TK]D-Fenderyou*
14:10.42[TK]D-Fenderyou*
14:10.44[TK]D-Fenderyour*
14:10.48[TK]D-Fenderdamn... can't type today!
14:11.27*** join/#asterisk eye69 (i=magnus@upcore.net)
14:11.39docelmoits cause your dialplan is 20k
14:12.07asteriskmonkeytengular: i work for the distributor of digium in canada
14:12.25[TK]D-Fenderdocelm0 : Ah the bitter face fo jealousy.....
14:12.33[TK]D-Fender:D
14:12.34tengulreasteriskmonkey: are you programmer?
14:12.36docelmofo.. I dont think so..
14:12.38Assid[TK]D-Fender!!!
14:12.52asteriskmonkeytengulre: yes i program manny things for asterisk and other things too
14:12.56[TK]D-Fenderhey Assid sorry I didnt get a chance to forward that config file...
14:13.05tengulreasterikmonkey: cool!! :)
14:13.06[TK]D-Fender1.6.2 right?
14:13.16Assidhehe.. np.. i actually came across some weird issue yday.. but solved it
14:13.18Assidyeah 1.6.2
14:13.23iCEBrkrmrtwister: haha, I suppose if I got zaptel crap compiled, configured and working, anything else can't be much more complicated :)
14:13.47Katty[TK]D-Fender: thx.
14:13.53Katty[TK]D-Fender: just what i always wanted.
14:13.59tengulreasteriskmonkey: my boss let me to select a call center in linux platform, so I want to select asterisk? but I 'm a begginer for it!
14:14.01[TK]D-FenderAssid : is it working now or would you still like my file?
14:14.01mrtwisteriCEBrkr: yes. in general, sangoma will use same settings
14:14.13iCEBrkrSo what's a Sangoma A104 cost?
14:14.16[TK]D-FenderKatty : "If thine eye offends thee"...
14:14.17iCEBrkrballpark
14:14.23Assidoh that bug i had was for something else.. like using only line 3 for another user
14:14.23tengulreasteriskmonkey: I want to got many documents of asterisk!
14:14.26[TK]D-FenderiCEBrkr : Par with Digium
14:14.29asteriskmonkeytengulre: its not hard to learn
14:14.31iCEBrkr[TK]D-Fender: Right on
14:14.34asteriskmonkeylots of help online
14:14.35Assidapparently i had to register it under line 2
14:14.39mrtwistertengulre: i think, only asterisk
14:14.48mrtwistertengulre: for callcenter
14:14.49[TK]D-FenderAssid : definately no need for that.
14:14.58[TK]D-Fenderlet me package it up for you
14:15.01Katty[TK]D-Fender: mew?
14:15.12mrtwistertengulre: what you want to do
14:15.17[TK]D-Fenderthere there.....
14:15.42Assidwell h thats what happened .. i had line 1-2 on 1 user.. and line 3 for another user..  but in the web control.. i jhad to register line 3 under line 2 to get it working
14:15.43Katty[TK]D-Fender: you don't pet normal girls, do you?
14:15.49Assiddoes katty bite?
14:15.58[TK]D-Fender:O
14:16.01docelmoyes
14:16.12KattyAssid: why don't you ask her and find out.
14:16.18docelmoor so I have heard..
14:16.34docelmohehe
14:16.37AssidKatty: you bite?
14:16.42KattyAssid: yes.
14:16.47Assiddamn
14:16.48tengulremrtwister: I want to building a call center platform, about 30 analog lines and 10 clients! client application use Microsoft windows , when a caller incoming then auto pop a menu on desktop... ..
14:17.10[TK]D-FenderAssid : how many differnt regs do you really what the phone to have?
14:17.36Assidonly that unit.. needed 2 registrations.. others.. single registration..- 3 lines
14:17.39[TK]D-FenderAssid : all of mine have 1 reg with multiple line keys attributed to that reg.  Subsequent calls just use the next available line key
14:17.50[TK]D-FenderWhy 2 reg?
14:18.05Assidhave a seperate number for that extension..
14:18.09mrtwistertengulre: tell more. what you said is present in asterisk and in related applications
14:18.17Assiddirect number
14:18.42[TK]D-FenderAssid : you mean DID directed to your phone?
14:19.00Assidyep.. asterisk gets it.. forawrds call to that phone
14:19.11Assidno ivr.. nothing
14:19.24[TK]D-FenderAssid : definately no need.  All of my workers have DID's and 1 reg.  You could clean things up a LOT....
14:19.27Assidoh damn,..i was supposed to ask shido to pickup something
14:19.36*** join/#asterisk Feral_Kid (n=Feral@red-corp-200.56.96.178.telnor.net)
14:19.47Assid[TK]D-Fender: HuH ?
14:20.56[TK]D-FenderAssid : My phones use 1 reg using 2-6 line keys (non-lines are speed dials, buddy watch, etc).  They have an internal ext (for dialing and callerid purposes) but have DID pointed to them from the outside.  You don't need a 2nd reg for that.
14:21.07SkramXHi all.
14:21.15Assidright..
14:21.42[TK]D-FenderAssid : would allow you to trim a lot of unnecessary stuff from sip.conf and trim your dialplan.
14:21.43Assidi have a IVR thing.. for all internal extensions..
14:28.53Feral_KidAny of the NuFone folks around?
14:29.25mutilatorDruken: radio is dead!
14:29.44zoanufone just left
14:41.18*** join/#asterisk javar (n=javar@69.79.133.185)
14:42.19*** join/#asterisk wundaboy (n=asdf@c-67-164-107-68.hsd1.or.comcast.net)
14:43.19wundaboycan somone refresh me on the differentce between a friend,user,peer?
14:46.57zoawundaboy
14:46.57mrtwisterwundaboy: peer = asterisk --> somewhere; user = client -> asterisk, friend = asterisk <---> gateway
14:47.36wundaboyso if im setting up my voip provider (voip-pstn), i should set it up as a 'user'?
14:48.12mrtwisterwundaboy: peer or friend
14:48.17wundaboyoops, i meant to say peer
14:48.28iCEBrkrbeer?
14:48.29wundaboyif i just send calls, it should be a peer right?
14:48.35mrtwisterwundaboy: but friend ok too, look also to context=
14:48.46wundaboyand if i send and recieve it should be friend?
14:49.22wundaboyalso, in the iax.conf what does the 'context' directive do?
14:51.00*** join/#asterisk Ariel_ (n=Ariel@adsl-068-157-125-248.sip.mia.bellsouth.net)
14:51.13Ariel_Morning everyone
14:52.16[TK]D-Fenderwundaboy : For clients its the context which controls what they can dial, for providers if the context that will receive incoming connections.
14:54.37Kattyhewwo Ariel_
14:54.45*** join/#asterisk negatendo (n=negatend@c-67-172-149-125.hsd1.co.comcast.net)
14:54.52Ariel_Katty, hello, good morning
14:55.09KattyAriel_: not good morning
14:55.12KattyAriel_: bad morning
14:55.16KattyAriel_: temp is dropping. :<
14:55.26KattyAriel_: and it's snowing :<
14:55.27Ariel_Katty, sorry to hear it.
14:55.55Ariel_I am having 2 of the worst weeks that I can remember. But lets not go there.  I hope your day gets better.
14:56.28KattyAriel_: :>>>
14:56.39KattyAriel_: hope your weeks getting better too (=
14:57.04Ariel_as soon as we finally do our closing on the house it will.
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15:04.15*** part/#asterisk SERGEUS (n=s@195.112.98.13)
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15:09.24zoaanybody from sineapps / freedam here ?
15:13.44*** join/#asterisk vaewyn (i=freeman@mail.parrishmachine.com)
15:14.09vaewynBoooyahhh!   (or in other words... long time no see... and good morning :} )
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15:15.19*** join/#asterisk ping1 (n=DLBaker@mail.sheffieldsteel.com)
15:15.24vaewynso... anyone know if you can prevent native bridging on ZAP channels?   especially on a per call basis?
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15:24.14blopvaewyn, why would u do so ?
15:27.11vaewynblop: our nortel doesn't have the option to allow forwarding of off campus calls to off campus...   and when it sees that bridge attempt to take place it slaps it down... and for some reason I would like to avoid the 4500$ for the option :P
15:27.57vaewynblop: it works fine for locally generated calls... just not external
15:28.01*** join/#asterisk Seldon1975 (n=someone@gatekeeper.radintl.com)
15:28.17blop:)
15:28.39vaewynand so far I don't see a canreinvite type deal for zapata  :}
15:28.53vaewynlet alone a per call thing
15:32.06*** join/#asterisk shmaltz (n=chatzill@mail.dmaven.com)
15:32.50Seldon1975hi.. no matter what i put for 'fxsks' and 'fxoks' in zaptel.conf, I get "ZT_CHANCONFIG failed on channel 1: No such device or address (6)"
15:33.30ManxPowerSeldon1975, and lsmod shows the modules loaded?
15:34.31Seldon1975manx - you're right; it was because the driver wasnt loaded
15:34.31vaewynlong time no see
15:34.42ManxPowerhello vaewyn
15:35.23*** join/#asterisk Tili (n=Tili@202-133-67-86-dialup.sat.net.pk)
15:35.39vaewynHey ManxPower... you don't by any chance know if there is a way to control native zap bridging do you?
15:35.44Seldon1975soz'
15:35.46Tilianybody has any URL for a symbian VoIP client. I have heard of Buzz2Talk but never found it
15:36.26ManxPowervaewyn, I can't imagine why you would do that.  The only difference between a native zap brige and a zap non-bridge is the format of the audio internally to asterisk, as I understand it.
15:38.26vaewynManxPower: Well...  When * attempts to native bridge a call that is   off campus -> nortel -> * -> nortel -> off campus   The nortel slaps it down...  where it doesn't slap down a    on-campus -> nortel -> * -> nortel -> off-campus..   We are almost positive it is because we don't have the option on the nortel for forwarding external to external
15:38.59ManxPowervaewyn, Perhaps you are confused about what a Zap native bridge is.
15:39.01vaewynSo I was trying to find a way for it to not inform nortel that these are the same call
15:39.20ManxPowervaewyn, Asterisk should not inform the nortel anything.
15:39.34ManxPowervaewyn, what interface?  CT1/CE1 or PRI?
15:39.36*** join/#asterisk Assid (n=assid@203.115.64.62)
15:39.54vaewynPRI - esf,b8zs
15:40.08ManxPowervaewyn, What makes you think Asterisk is infoming the nortel?
15:40.16bintutis the Digium TDM04B PCI card works on a standard 32bit PCI slots of the ordinary athlon/p4 motherboards?
15:40.30*** join/#asterisk |dennis| (i=dennis@200.32.215.84)
15:40.31ManxPowerbintut, it should
15:40.56bintutManxPower: so, it doesn't need to have a pci-x or pci-express slots, right?
15:41.10vaewynManxPower: because I can call both directions... and nortel phone sourced calls it works...  outside sourced calls the nortel requests a hangup the second I see the native bridge line
15:41.12ManxPowerbintut, The part number you quoted does not.
15:41.28ManxPowervaewyn, You have some other problem
15:41.57ManxPowervaewyn, you might have a native brigde if you put something like "t" as an option on the dial line.
15:42.26ManxPower..er...it might not native bridge if you put something like "t" as an option on the dial line.  But that's not the problem.
15:42.31vaewynManxPower: well... as far as * is concerned the calls should be identicle...  both are coming from the PRI and are going back out the same PRI
15:42.56vaewynNortel is the only one that "knows" the sources
15:42.59ManxPowervaewyn, no, each call is unique.
15:43.11ManxPowerYou have two calls, the call into asterisk and the call out of asterisk
15:43.29bintutManxPower: i just want to confirm to you.. the Digium TDM04B PCI card is good enough for 4 FXOs and all other FXS are VoIP phones.. that card is good voice quality, right?
15:43.30vaewynWell... lets just say... they both follow an identicly dialplan path... and the nortel requests a hangup
15:44.09ManxPowerbintut, that should work, but a T-1/E-1 would me more reliable.
15:44.50ManxPowervaewyn, what happens when you put "t" on the Dial line.
15:45.00ManxPower(t should prevent a zap native bridge)
15:45.00bintutManxPower: we only have 3 phone lines from a local PSTN.. the other one is reserve.. do i need a t1-e1 for this?
15:45.06vaewynattempting that now
15:45.18[TK]D-Fenderbintut : For that few lines a TDM400 card would be fine
15:45.25ManxPowerbintut, A T-1/E-1 will always be more reliable than analog lines.  But analog lines will work.
15:45.30[TK]D-Fenderbintut : Definately no need for T1/E1
15:45.30bintutManxPower: but if you recommend it because of best voice quality output, i should consider it
15:45.41ManxPowerUnfortunatly, in my experience the TDM400Ps crash about once a month.
15:45.43[TK]D-Fenderbintut : Would be expensive....
15:45.49bintutyeah
15:45.53bintutthanks guys..
15:45.59[TK]D-FenderYou could always use SPA-3000's for your lines...
15:46.31*** join/#asterisk taec (n=phil@eventhorizon.hosting365.ie)
15:47.00*** join/#asterisk marc32422 (n=marc3234@206-248-133-186.dsl.teksavvy.com)
15:47.26bintut[TK]D-Fender: i'm thinking of buying Sipura SPA-2100 Analog Telephone Adapter instead
15:47.40taecI've got Asterisk setup with SIP based IP phones and a ZAP module connected to PRI .... if I'm at the asterisk CLI, is there an easy to way to see the CID's of the incoming calls and which extensions/queues they're currently in?
15:47.54ManxPowertaec, no.
15:48.04*** join/#asterisk af_ (n=af@ip-142-84.sn1.eutelia.it)
15:48.17ManxPowertaec, you can see the callerid number on the console "Accepted call from '1234567890'" etc
15:48.42taecOk, is there a hard way? I've noticed that AMP's flash panel can do it fairly easy. I know it interacts with the Asterisk Call Manager. Can I interact with that to get the information I'm looking for simply enough?
15:49.04*** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com)
15:49.04*** mode/#asterisk [+o anthm] by ChanServ
15:49.06*** join/#asterisk brimstone (n=brimston@pdpc/sponsor/digium/brimstone)
15:49.28*** join/#asterisk brettnem (n=brettnem@72.29.102.158)
15:50.22taecAMP's flash panel seems to be able to grab the external CID of the caller fine and even associates it with an extension (e.g. if they're currently on the phone to them) ... so I presume it's possible without a lot of hassle. I simply presumed it could be done through the CLI as well. How would I go about seeing the CLI of callers who are currently active? no way?
15:50.44*** join/#asterisk jjuhlh (n=Irix@nat.kollegienet.dk)
15:51.25ManxPowertaec, you are wrong.
15:51.34bintutdo you think this mobo <http://tinyurl.com/8hd25> is good enough for a simple PBX with a Digium TDM04B PCI card?
15:51.40ManxPowerAMP prolly gets the info via the manager interface.
15:51.58taecManxPower: So it is a lot of hassle? or it's not possible?
15:52.06ManxPowertaec, it's a lot of hassle.
15:52.22taecManxPower: yes, it does get it via the manager interface ... but does that not provide the same interface as the CLI?
15:52.30ManxPowerYou can always put a Noop in the dialplan so you can see the callerid when the call hits the Noop
15:52.42ManxPowertaec, no, the manager interface is different from the CLI.
15:53.10taecOK, I should probably look into that a little bit more so
15:53.19ManxPowertaec, why do you want to know the callerid info?
15:53.57taecSmall project I thought of, was that an agent here could login to a page, with an extension number and password and if a customer dials in off a known number, their details could be displayed on-screen
15:54.06anthmHowever, I like to make cli commands in anticipation of using it from manager which is possible and is flexable because you can use it from the command line and from a remote program
15:54.40taecantha: I was under the impression that it was only CLI commands you could use through the Manager, I must have been mistaken!
15:54.44marc32422ne1 can recommend a switch for connecting asteirsk servers together?
15:54.53[TK]D-Fenderbintut : I'd pick a better board if I were you.  An ASUS or GigaByte.... something with a better bios so you can guarantee it getting a seperate IRQ
15:55.19ManxPowertaec, there is a readme or .txt file documenting manager as past of the Asterisk source.
15:55.32anthmin reality it's the only one you need manager has a way to install a clumsy manager action thing but the Command action is more than enough
15:55.33ManxPowerpast == part
15:55.41*** join/#asterisk asterisk99 (n=dunoowhy@modemcable169.194-130-66.mc.videotron.ca)
15:56.00coppiceIts sad. there was a time when ECS made *the* premium motherboards
15:56.27anthmand when USR made nice modems =D
15:56.46asterisk99anyone familiar with installing zaptel on Gentoo? My ztcfg will not run after reboot :(
15:56.53coppiceI remember when an ECS 486 motherboard cost US$3000 :-)
15:57.11*** join/#asterisk citats (n=james@bgp925576bgs.brghtn01.mi.comcast.net)
15:57.19ManxPowertaec, think of it this way: The manager interface gives you access to all CLI commands, but it also has additional information it can provide to the app that uses the manager interface.
15:57.44[TK]D-Fendercoppice : And when swooping pteradactyl's were the greatest treat to humanity!
15:57.48[TK]D-Fenderthreat*
15:58.10iCEBrkrasterisk99: I know what your problem is..
15:58.33coppicei guess you've been to that creationist musuem :-)
15:58.34taecManxPower, and what you're telling me is that some of the additional information it provides is CLI information?
15:58.38ManxPoweriCEBrkr, the fact that he doesn't understand the Gentoo boot process?
15:58.38iCEBrkrasterisk99: You're running Genpoo.
15:58.42taecapologies, CID information
15:58.47ManxPowertaec, Yes.
15:58.52iCEBrkrManxPower: :P
15:59.00taecMuch appreciated ManxPower, thanks for your help!
15:59.04asterisk99iCEBrkr: hahaha
16:00.04docelmoYippie!
16:00.24docelmoCentOS is the only way to go..  Except Dovecot..  That program F*!?ing SUCKS!
16:02.31iDunnoCentOS is the only way to go if you're clinically insane.
16:02.43iDunno(or really really like rpm dependency hell)
16:04.01vaewynManxPower: t option doesn't change it...  still hangs up
16:04.02docelmoiDunno, its not that bad..  works very well for me..
16:04.17*** join/#asterisk DrDeke (i=dekemar@deculator.engin.umich.edu)
16:04.41vaewyndebian rocks
16:04.45docelmoThe biggest problem I had with 4.2 was Dovecot and getting it to work..  I ended up chucking it and going with washington.edu's ipop3 and imapd
16:04.48vaewynIMH(BC)O
16:05.09docelmoI like gentoo..  but its a pain in the ass to install
16:05.33iDunnoyou had problems with Dovecot?! how?! is simple!
16:06.24asterisk99docelmo: do you have asterisk running on gentoo?
16:09.12fugitivoasterisk99: i do
16:09.22fugitivodocelmo: it's not a pain in the ass
16:09.32docelmonope..
16:09.53*** join/#asterisk kokey (n=ubunture@201.153.63.79)
16:09.58asterisk99fugitivo: have any problem getting ztcfg to run after reboot?
16:09.59docelmoiDunno, dunno..  I installed it and everything configured it and couldnt get my email clients to work with it.
16:10.06fugitivoasterisk99: no
16:10.23docelmoI would if I had the patients to install it.
16:13.52coppiceif you were a doctor you might be able to get your patients to install it
16:14.18iDunnoweird.
16:15.19*** join/#asterisk paryl (n=paryl@216-201-177-82.res.logixcom.net)
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16:17.13paryli can dial 800 numbers from asterisk, but i can't dial a specific one that a user needs.  if i dial it with a cell phone, it works.  any idea what could cause behavior like that?
16:17.45parylit comes back (after about 6-7 seconds) with "== No one is available to answer at this time (1:0/0/0)"
16:18.39SkramXparyl: is your dial plan correctly configures?
16:19.36*** join/#asterisk sysdebug (n=sysdebug@200.163.193.247)
16:20.26ManxPowerparyl, what is the value of HANGUPCAUSE after the call ends?
16:21.00parylmanxpower... i don't know how to get that
16:21.16parylSkramX, it's dialed like every other 800 number, and they work
16:22.01ManxPowerparyl, the priority after Dial would be Noop(HANGUPCAUSE=${HANGUPCAUSE{)
16:22.08ManxPowerwell, with the correct braces, of cours.
16:23.29vaewynManxPower: figured it out...  nortel is catching on via the RDNIS somehow...  need to alter that
16:23.59ManxPowervaewyn, I DID say it was not a native bridge issue 8-)
16:24.21parylmanxpower hangupcause=34
16:24.37vaewynManxPower: Reason I thought it was native bridging is because it dies right on that spot
16:25.04parylwhich is normal congestion... but a cell phone will ring it without issues
16:25.04DrDekeHey guys; there is a bug in Zaptel/wctdm.c that makes it not decode pulse-dialed digits on FXS ports correctly. It is very simple to fix, you just have to change one value, but it seems like this should really just be fixed by the project administrators. Would someone please tell me the correct way to report this that would get it added in 1.2.2 or whatever?
16:25.04ManxPowerparyl, 34 is "no circuit or channel available".
16:25.13ManxPowervaewyn, MANY things happen at that spot 8-O
16:25.29ManxPowerDrDeke, bugs.digium.com
16:25.33parylthe wiki says "#define AST_CAUSE_NORMAL_CIRCUIT_CONGESTION 34 "
16:25.55vaewynmeans it is full or unavailable
16:26.22vaewyncan also mean number blocks or such
16:26.30jjuhlhquestion :) when a PBX is able to handle 250 calls simultaneous, how many many custommers would you expect to have to reach this limit ?
16:26.33vaewynkindof generic :}
16:26.59ManxPowerjjuhlh, What is your customer/line ratio?
16:27.20vaewynjjuhlh: we run 10/1 ... but universities are not normal situations
16:27.22ManxPowerjjuhlh, There is NO general answer to your question.  Each company/industry has different requirements
16:28.34ManxPowerWe run closer to 2 agents for every 1 line at some of our offices.  In other offices we run 3 employes / for every 1 line
16:28.46ManxPowerOur people spend their life on the phone.
16:29.44parylthis makes no sense... i dial it on a cell phone and it answers immediately, i dial through asterisk and it gives code 34
16:30.09jjuhlhyes I know... I'm trying to make a budget and have to like explain how many customers I would expect to have after a year, so I also have think about how many PBX I have to run .. if you for example have 5000 customers after one year... just a estimate
16:30.50paryljjuhlh: we have ~3500 active customers and service them on one t1 with 4 regular agents
16:31.01*** join/#asterisk Seldon1975 (n=someone@gatekeeper.radintl.com)
16:31.09DrDekejjulh: It completely depends what you are planning to do with this phone system, what business your customers are in, and so on.
16:31.15parylof course, we have rollover queues for busy times, but that's normally enough
16:31.28jjuhlhokay :)
16:32.33paryli have the txgain bumped down... could by some chance that cause the connection error?
16:32.54jjuhlhparyl, 4 regular agents.. ?
16:33.43jjuhlhwe are talkink about 4 PBX ?
16:33.47jjuhlhtalking
16:34.04paryljjuhlh: no, 4 people answering the main line.  that's all
16:34.33jjuhlhokay... and this is only one PBX ?
16:34.38parylyes
16:34.45jjuhlhinterresting
16:34.52paryl35 stations, most are outgoing calls
16:35.04parylcustomers call into one number, and get routed to queues
16:35.05jjuhlhyes I could imagine.. :)
16:35.06*** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-179-75-202.cust.bezeqint.net)
16:35.12docelmoAnyone know of any decient GSM Gateways compatible with asterisk?
16:35.18parylbut only 4 are logged on to the main queue at any given time
16:35.40fugitivodocelmo: www.2n.cz
16:35.53docelmothanks..
16:36.00docelmoI am looking to do some playing
16:36.50fugitivoit's like 800 euros
16:37.02fugitivobut it's the only voip gsm gateway i've seen
16:37.15docelmoWhats 800 euros in USD?
16:37.24DrDekeoff the top of my head, about 1200
16:37.26DrDekeerr
16:37.28DrDekemaybe more like 1100
16:37.33docelmocrap..
16:37.39DrDekewait no it's only $938
16:37.39parylhaha
16:37.44docelmoI just wanted like a single line to play with.
16:38.00fugitivodocelmo: you could get a regular gsm gateway, those are cheap :)
16:38.03fugitivonot voip
16:38.07DrDekeYou can do it much more cheaply, over chan_bluetooth or something for instance.
16:38.12*** join/#asterisk KranZ (n=user@sme.bestline.net)
16:38.12docelmoMy wife would kill me for 1200 buks..
16:38.22*** part/#asterisk KranZ (n=user@sme.bestline.net)
16:38.26*** join/#asterisk KranZ (n=user@sme.bestline.net)
16:38.26zoadocelmo there is easier stuff
16:38.26docelmoWell what I need is a gsm gateway that will do SMS
16:38.32zoaaha
16:38.32DrDekeoh
16:38.36zoause a gsm gateway :p
16:38.40zoaeuh
16:38.41DrDekeI would DEFINITELY do that with bluetooth.
16:38.43zoaan sms gateway
16:38.45DrDekeOr USB
16:38.47DrDekeOr anything of that nature
16:38.50fugitivouh
16:38.51DrDeke(If you wanted it on the cheap)
16:38.52fugitivoonly sms?
16:39.12fugitivouse a webpage :)
16:39.15docelmosms/voce
16:39.18docelmoerr voice
16:39.34*** join/#asterisk Dibbler_ (n=Dibbler@snaddy.plus.com)
16:40.50docelmoBut mainly SMS..  Website is fine..  But that kinda defeats the purpose of being able to do SMS..  :)
16:42.27zoabehave mr!
16:42.31vaewynhehehe..
16:42.32*** join/#asterisk Igbothom_III (n=HiltonT@static-84.217.240.220.dsl.comindico.com.au)
16:42.38vaewynlong time no see (on my part :{ )
16:42.52zoayes
16:43.18vaewynman... I was gone so long you guys actually got head stable enough to release as something
16:43.25zoa:)
16:44.16*** join/#asterisk znoG (n=gs@OL101-122.fibertel.com.ar)
16:46.14jjuhlhanyone have a good suggestion to a Switch/Router capable of redudant...?
16:47.02docelmoredudant what?
16:47.25asteriskmonkeycisco catalysy 10000 is nice
16:47.44docelmowell took my idea..  :(
16:47.54jjuhlhsorry redundancy.... I'm danish hehe
16:48.07*** join/#asterisk Kokey (n=Kokey@201.153.63.79)
16:48.31[TK]D-Fenderjjuhlh : redundant what?
16:48.46fugitivois any way to see for example, what modules has a card without having a complete /etc/zaptel.conf?
16:49.51*** join/#asterisk Samoied (n=Samoied@popeye.opens.com.br)
16:50.53jjuhlh[TK]D-Fender, was that a joke..hehe ;)
16:51.23vaewynHow on earth does this Nortel know it is a forwarded call when I wipe the RDNIS and PRIREDIRECTREASON before I send the call over the PRI?
16:51.30vaewyngrrr
16:51.35ManxPowerfugitivo, ztcfg only shows what you configured, not what modules are ACTUALLY on the card.
16:51.39vaewynNORHELL SUCKS!
16:51.41[TK]D-Fenderjjuhlh : no, redundant wht?  power?  wan conenction? Some other proxy service?  BGP?
16:51.49ManxPowerwhen you modprobe the card driver, it should print the info into syslog or dmesg
16:52.46jjuhlh[TK]D-Fender, it was just about a Switch/Router
16:53.05fugitivoManxPower: if you modprobe with no config in zaptel.conf, will it display the info i want?
16:53.43ManxPowerfugitivo, with or without a config, the info will still be in syslog/dmesg
16:54.23fugitivoManxPower: i know it'll display that it found a card, but i'm not sure about the number of channels, i don't have a card right now to test it
16:54.41*** join/#asterisk veepster_ (n=veepster@vbn.0012297.lodgenet.net)
16:54.42ManxPowerfugitivo, then ask again when you have a card to test.
16:54.49veepster_<veepster_> to test asterisk, do I need any additional hardware or will a basic linux server do?
16:54.57ManxPowerfugitivo, it will display the channel number for each module
16:55.12fugitivook, thanks
16:55.29vaewynveepster_: for softphones or network (SIP) phones you just need the server
16:55.35postelHow can i configure ringing a specific number when the handset is lifted in *? in cisco is a single command, cisco calls it PLAR (Private Line Auto Ringdown)
16:56.07postel(think emergency phone) although my scenario is different
16:56.24ManxPowerpostel, on a zap card?
16:56.47postelManxPower: nope, sip client
16:56.57ManxPowerpostel, it's configured in the SIP client then.
16:57.12ManxPowerSIP devices collect digits and THEN send the call to their server.
16:57.29ManxPowerpostel, Cisco and SIPura support PLAR-like features.
16:57.30iDunnoas * doesn't know anything about the phone being picked up other than what the phone tells it.
16:58.32posteli see, so its not possible, since * would get nothing until i DIAL something out
16:59.22[TK]D-Fenderpostel : its the phone's joib to do the dial though.  So if your phone has an option for an "immediate" call to a specific # it would dail that as soon as it goes off-hook
17:00.21veepster_vaewyn, can you point me towards a link where I can learn more? I want to test asterisk, Im not entirely sure what that entails yet
17:02.33postel[TK]D-Fender: the feature set of the phone is not in question here, the phone may be able to PLAR without even * being there, i was thinking if the status of the handset on/off could be monitored and trigger an action. It seems thats only possible in ZAPtel and not SIP
17:02.50*** join/#asterisk SpaceBass (n=sp@static-71-251-230-2.rcmdva.fios.verizon.net)
17:03.12ManxPowerpostel, what specific device do you have?
17:03.43[TK]D-FenderNo, not in SIP.  The phone would have to send some sort of signal...  In SIP the phone is "kink", maybe MGCP would be able to do something like that...
17:03.55[TK]D-Fenderking*
17:04.54postelManxPower: DECT Panasonic phones on ATAs, some cisco VICs, some 7960s and some analogs on TDMs with FXS modules, why?
17:04.59ManxPowerYou could also buy a SIP device that has PLAR functionality
17:05.08ManxPowerpostel, WHAT ATAs?
17:05.16postelManxPower: 186 ciscos
17:05.18file[desk]darn so cold
17:05.34ManxPowerOh!  EASY.  The docs for the Cisco ATAs talk about PLAR stuff.
17:05.41postelthey do?
17:05.48ManxPowerfile[desk], You are in CANADA, of course it's COLD.
17:05.49postelthe Admin Guide?
17:05.59ManxPowerpostel, That is correct, grasshopper.
17:06.15ManxPowerpostel, I used to use Cisco ATAs until SIPura came out with devices that are SO much better.
17:06.32postelI've been over the Admin Guide, that was sometime ago though
17:06.43file[desk]the Sipura stuff is beautiful, same for their SIP implementation
17:06.51postelwhen i had trouble with the SS bits
17:07.22ManxPowerpostel, I even used the PLAR (they may not use that term) with the Cisco ATAs, but that was like 2 years ago.
17:09.08postela quick google search didnt give me much, the good news is i have access to gold partners so if the feature is there they should know bout it.
17:09.47postelthanks anyways, i'll dig a bit more
17:10.35KranZ[TK]D-Fender: are the polycom 301s decent phones?
17:16.47ManxPowerpostel, I think Cisco calls it "hotline" and "warmline"
17:17.50postelyeah, thanks, i found it
17:18.00postelits available from release 2.14
17:29.09*** join/#asterisk jpablo (n=jpablo@200.94.130.194)
17:29.14*** join/#asterisk Mother (n=mother@93.Red-80-32-127.staticIP.rima-tde.net)
17:29.55jpablohi, is auth=plaintext still in asterisk 1.2 ?
17:34.29Corydon-wjpablo: for what protocol?
17:34.37[TK]D-FenderKranZ : Poly 301 = wase of time.
17:34.41[TK]D-Fenderwaste.
17:34.56[TK]D-FenderToo low-end for the money.  501+ = good
17:35.11[TK]D-Fenderotherwise look at the SPA-941 or so.
17:35.30Corydon-wThe 301 certainly is a good phone... but I wouldn't recommend that you mix phones at an install.  People get phone envy
17:35.55vaewynYeah... spend the couple extra bucks for the 500+
17:35.58[TK]D-FenderGood yes, worth the money and setup for the features?  no
17:36.31fugitivoanyone using polycom 501?
17:36.34Corydon-wThe setup for either the 301 or the 501 is the same
17:36.54Corydon-wSame config files, same FTP method
17:37.01[TK]D-Fender2 line phone?  ick.  SPA-941 gives 4 calls, SPEAKERPHONE< and more.
17:37.13Corydon-wand if you learn to automate the process, it's even easier
17:37.30[TK]D-Fenderfugitivo : I've used 501's, but run 600's and 601's here.  I run all Sipura at home.
17:38.00fugitivo[TK]D-Fender: how much is the 601?
17:38.10[TK]D-FenderCorydon-w : if you're deploying other polycom models I'd say sure, but if the 30x is as high as you go, I wouldn't bother.
17:38.12jpabloCorydon-w, for sip, sorry for the delay
17:38.24fugitivo[TK]D-Fender: can you run xml apps on it?
17:38.24vaewynWill say.. 600 only real advantage I like is the microbrowser...  really wish the 500s had that...  although they might by now... havn't checked firmware in 6+ months
17:38.25jpabloCorydon-w, im trying to setup grandstream early dial
17:39.08Corydon-wjpablo: uh, why?
17:39.32[TK]D-Fenderfugitivo : 250$USD and I use the microbrowser on mine for all sorts of stuff...
17:39.53fugitivo[TK]D-Fender: that's the 601, the 501 doesn't have microbrowser, right?
17:40.10vaewynthe microbrowser rocks...  weather tickers... stock tickers...  al sorts of fun you can do with it...  ours we made into timeclocks :}
17:40.16[TK]D-Fendervaewyn : I also like having 6 line keys.  I use 3 for 1 reg normally, and the others for buddy-watch (presence) and speed-dials
17:40.18jpabloCorydon-w, grandstream dosn't like md5 auth
17:40.33[TK]D-Fenderfugitivo : 60x = MB, 50x = nope
17:40.47jpabloCorydon-w, i need plaintext auth
17:40.54Corydon-wShould work fine
17:40.56[TK]D-Fendervaewyn : timeclocks?  as in to punch-in in the morning?
17:41.04vaewyn[TK]D-Fender: That's the other thing...  does presence work in asterisk now?   I have been gone a while
17:41.17Corydon-wjpablo: are you just asking, or do you actually have an issue?
17:41.25[TK]D-Fendervaewyn : works real nice on my 600's and 601's (with attendand modules :D)
17:41.26vaewyn[TK]D-Fender: yep :}  it drops them on a page and they fill in their ID/pass
17:41.34*** join/#asterisk sahafeez (n=sahafeez@67.109.14.227.ptr.us.xo.net)
17:41.36[TK]D-Fendervaewyn : neat-o
17:41.37*** join/#asterisk zeedo (n=zeedo@reboot-robot.net)
17:41.52[TK]D-Fendervaewyn : I made one that does a presence detect for the phone company :D
17:41.55vaewyn[TK]D-Fender: * handles the presence?  or external program?
17:42.01vaewynbwahaha
17:42.02[TK]D-Fenders/phone/whole
17:42.04vaewynnice
17:42.05[TK]D-Fender*
17:42.25vaewyncool
17:42.29[TK]D-Fenderso I get "phones in use" and a line by line list of everyone, name & number
17:42.43vaewynsweeeet
17:42.46[TK]D-Fenderand I use the MB in dle mode to show live queue stats for my CSR's :)
17:43.01[TK]D-Fenderilde* on 10 sec interval
17:43.04vaewynI really need to get back into the * swing...  being tasked with other crud has been a real drag
17:43.14jpabloCorydon-w, i actually have an issue, and in the wiki it saids to use plaintext auth, but it is giving me a warning in asterisk 1.2
17:43.15[TK]D-Fenderincluding our company logo
17:43.46vaewyn[TK]D-Fender: yeah... we use idle mode for current weather and MOTD type stuff
17:44.06Corydon-wWhat's the warning?
17:44.12*** join/#asterisk ctooley (n=ctooley@rrcs-24-227-212-162.sw.biz.rr.com)
17:44.34[TK]D-Fendervaewyn : I'm going to make a MB page so they can subscribe to Idle services :)  And maybe use that to implement "messaging"
17:44.43jpabloCorydon-w, Dec  8 11:40:56 WARNING[27542]: chan_sip.c:11749 add_realm_authentication: Format for authentication entry is user[:secret]@realm at line 1020
17:44.55jpabloCorydon-w, it looks like auth= changed it's meaning
17:44.58vaewyn[TK]D-Fender: heh... not a bad idea :}
17:45.13[TK]D-Fenderabstraction is the key to success....
17:45.18ctooleyjust sent out details of our Asterisk Bounty Pool fund raiser to the asterisk-user mailing list.  I'll be putting up a site soon with a running total of the money that's been donated and details on which bounties have been offered and how to claim them.
17:46.06Corydon-wjpablo: plaintext is the default
17:46.08*** join/#asterisk xheliox (n=jeff@user-0c6s3h2.cable.mindspring.com)
17:46.19vaewynok... how the heck does the norhell know I am fowarding the call when I wipe RDNIS and PRIREDIRECTREASON prior to making the outgoing call
17:46.49[TK]D-Fendervaewyn : You doing a "Dial" to do your redirect?
17:47.31jpabloCorydon-w, i don t think so, im seeing the packages and i see auth=md5 in the sip packages
17:47.56Corydon-wjpablo: how about you just comment it out?
17:48.04vaewyn[TK]D-Fender: What we have is that when people call in... if their phone is unavailable it attempts to dial a fallback number... the norhell sees this outgoing call and smacks it down because it is originating originally from off campus...  and somehow it is figuring that out
17:48.23jpabloCorydon-w, i get the same
17:48.35Corydon-wjpablo: did you restart?
17:48.39vaewyn[TK]D-Fender: Nothing like Norhell and their "options"
17:48.59jpabloCorydon-w, grandstream getting confused with the 407 messages
17:49.06jpabloCorydon-w, yes, asterisk & the phone
17:49.25[TK]D-Fendervaewyn : Cheap hack - Make 2 SIP accounts on your * and have your * register to itself.  then place a sip call from 1 account to the other (effectively internal redirect) and it should count the incoming channel as SIP.  that way RDNIS = HISTORY.
17:49.40Corydon-wjpablo: well, someone with a bit more experience than you is going to need remote access to look at the machine
17:50.01jpabloCorydon-w, what do you know about my experience ?
17:50.13[TK]D-Fendervaewyn : And because it'll never actually leave the network interface you get NO overhead and G.711u leaves you with no transcoding.
17:50.15Corydon-wjpablo: Grandstream phones work fine with 1.2
17:50.17vaewyn[TK]D-Fender: bwahaha... I might fall to that idea if I can't figure it out another way...  Just wish this Norhell didn't think it was so smart
17:50.27jpabloCorydon-w, including the Early Dial feature?
17:50.37[TK]D-Fendervaewyn : Nortel = &*^#@$ stupid appliances...
17:50.41Corydon-wjpablo: why are you using early dial?
17:50.51[TK]D-FenderSOOO glad I got rid of my 8x24...
17:50.54jpabloCorydon-w, cause that's a nice feature that i want ...
17:50.59Corydon-wBecause you think it's a whiz-bang cool feature?
17:51.02Corydon-wTurn it off
17:51.06[TK]D-FenderI am the king of cheap hacks :)
17:51.20[TK]D-FenderJust never ask me to do it the "right" way :D
17:51.21vaewyn[TK]D-Fender: I would ditch ours...  but is Option 11C running a @#$@$load of digital phones
17:51.26jpabloCorydon-w, i know the phones work without early dial, im trying to make early dial work, that's the point of my questions ...
17:51.32[TK]D-Fender11C?
17:51.38jpabloCorydon-w, try to make early dial work
17:51.40Corydon-wjpablo: ask someone else
17:51.58vaewyn[TK]D-Fender: Up to 10000 TNs
17:52.04[TK]D-Fender?
17:52.11Corydon-wI can help make it work, but if you insist on using something that it doesn't work with, I'm not going to be any help
17:52.11vaewyn[TK]D-Fender: currently using 4800 or so
17:52.13[TK]D-Fenderbreif explin plz...
17:52.21[TK]D-FenderTN's (digital sets?
17:52.28vaewyn[TK]D-Fender: basically
17:52.33[TK]D-FenderOMG.  What for?
17:52.41vaewynUniversity
17:52.52[TK]D-Fender4800 phones?!?!?!?!
17:53.05vaewynhandles combo of analog and digital units...  1200 digital... rest are analog
17:53.10vaewynYep
17:53.14[TK]D-FenderOMG.
17:53.15ctooleyvaewyn, which University?
17:53.21[TK]D-FenderOk maybe too much to convert all to *
17:53.21jpabloCorydon-w, my question was simple, how to enable auth=plaintext, that's it
17:53.35jpabloCorydon-w, i know what's the problem, the asterisk 407 messages
17:53.36vaewynctooley: Andrews University... private one in SW Michigan
17:53.41[TK]D-FenderAnd I'm sure no budget to do so either...
17:54.03ctooleyUTexas uses a Nortel as well, I'm pretty sure they're around 10 or 15 K phones
17:54.16vaewyn[TK]D-Fender: actually... the Norhell is out of service plan next year... and they want 2.5Mil to get it back on with current gear so... :}
17:54.22Corydon-wplaintext auth is the default
17:54.35[TK]D-Fendervaewyn : Citel <----
17:54.44[TK]D-FenderSER & *
17:54.53ctooleyvaewyn, 2.5 Mil might not buy you enough phones and network to replace it though.
17:54.58*** part/#asterisk Dandre (n=Dandre@was59-3-82-236-48-30.fbx.proxad.net)
17:55.03vaewyn[TK]D-Fender: If I can prove * can handle it then we are going to go off plan and use the Norhell as a channel bank :P
17:55.47ctooleyah. Slowly migrate people to newer devices when the old ones finally die and keep the channel back for all those analog lines.
17:55.48tzangervaewyn: already in the works :-)
17:55.52vaewynctooley: If we stick to analog with channel banks... and only use SIP for replacing the digitals we can come in under that
17:55.58jpabloCorydon-w, ok, thanks.
17:56.08tzangercitel has a norstar phone driver ... turns them all into sip phones
17:56.36vaewyntzanger: Umm... so how do they break out that many ISDN lines?
17:56.49vaewyntzanger: or are you talking the nortel network digitals?
17:56.59tzangervaewyn: ?  it's a D50 to the desk sets
17:57.03tzangerthen ethernet for SIP
17:57.22vaewynhmm... don't know my norhell... D50?
17:57.33tzangerAMP D50 (25 pairs)
17:57.41[TK]D-Fender1200 digital = 50 Citel ($2700) = $135,000.  the other 3600 = 150 Audiocodes ($2300) = $345,000 ------ $480,000to convert the sets
17:58.04vaewyntzanger: ahh... 25 pair... hehe...
17:58.05tzanger[TK]D-Fender: as opposed to what
17:58.21vaewyntzanger: so it drives the fake ISDN crud?
17:58.25ctooleyputting that many extra devices on the network though would likely make the network guys scream upgrade
17:58.26tzangervaewyn: correct
17:58.34[TK]D-Fendertzanger : as opposed to the 2.5M they were asking
17:58.42vaewynnetwork is already sized for this and more
17:58.44tzangerctooley: you don't put it on the data network
17:58.50tzangeryou put the 50 citel on their own switches
17:58.51vaewyn8 pair fiber to all buildings
17:58.56[TK]D-FenderPRIVATE GB LAN <-
17:59.06tzangernice
17:59.18vaewynOnly real problem we have is power...  We need more generators and UPS
17:59.18tzangerI have a telebridge hooked up to my norstar
17:59.20tzangerit's kind of neat
17:59.40vaewynmain IT building is the only 24/7/365 power so
17:59.49ctooleyyeah, those aren't free either
17:59.51vaewynrest are at the whim of nature
18:00.09ctooleyenvironmental costs are probably going to be higher than replacing the handsets
18:00.18vaewynluckily they are looking at that anyway for security systems and such so :P
18:00.33[TK]D-FenderClearly you would use SER to handle the SIP accounts and * as an app server.
18:00.59vaewynYeah...  we sized the * boxes and phones and channel banks... and about 1.5Mil...  Generators and UPS another 1 Mil+
18:01.15vaewynhell no...  * it all :}
18:01.15ctooleyBetween design, implementation, parts and configuration, 2.5 mil would be a tight budget
18:01.32[TK]D-Fendervaewyn : My idea came out to $480k and lets you keep your phones :)
18:01.36ctooleyMight be possible, but it would be tight.
18:02.01[TK]D-Fenderyet still completely eliminates Nortel (except for the digiat sets.
18:02.04ctooley[TK]D-Fender, the Citel's use network for interconnect or the phone lines?
18:02.14vaewynBut... the nice thing is...  we know that in 4 -7 years we will have to pay for another frame to move the norhell to VoiP anyways...   and that they are already quoting us another 7+mil...  so 2.5+ now... and save 7 later is looking very nice to the VP types
18:02.44[TK]D-FenderCitels offer Norstar Digital on Amphenol and SIP out.
18:02.46vaewyn[TK]D-Fender: citel does seem tempting... will have to look into that
18:02.58*** join/#asterisk brettnem (n=brettnem@72.29.102.158)
18:03.25vaewynwonder what I can get for a bulk buy on those :P
18:03.26[TK]D-Fendervaewyn : but oh boy are you going to need a good redundant SIP router and PRI gateway.
18:03.39[TK]D-Fendervaewyn : for 50 I'm sure you could :)
18:03.39SkramXI cant find any of those free TTS websites.. anyone know?
18:03.42ctooley<PROTECTED>
18:04.05[TK]D-Fenderctooley : depending on what you consider cause for replacement (defect or desire)
18:04.21vaewynctooley: I am liking the quality on the Poly 500/600... Is comparable with the norhells
18:04.26[TK]D-Fenderctooley : but the advantage is no WIRING change. and no change of the handesets themselves.
18:04.44tzanger[TK]D-Fender: if you go the citel route throw a couple extra in there I will buy them off you
18:04.48[TK]D-Fendervaewyn : Polycom quality >>> Nortel
18:04.51vaewyn[TK]D-Fender: any idea how far those will power a line?  We have a couple 2000+ft runs
18:04.52ctooleyvaewyn, yes, but the Polycom's are certainly not as sturdily built as the old Nortels
18:05.12[TK]D-Fendertzanger : I'm not going the Citel route, I'm suggesting it to vaewyn
18:05.30[TK]D-Fenderctooley : I suppose in the abuse-taking scenario, yeah....
18:05.44vaewynThe poly's I have are what I would consider on par with the norhell's in abuse
18:05.47[TK]D-Fendervaewyn : got to their site and read up.
18:06.01jpablofucking early dial, that's a great feature
18:06.08vaewyn[TK]D-Fender: am there now :P
18:06.13[TK]D-Fendervaewyn : I'm not really sure... I investigated it as an option for us here.  We're an all-Polycom outfit now :D
18:06.16jpablowhy it doesn't work
18:06.22ctooley[TK]D-Fender, the problem with the Poly's is that display.  It lasts quite well until something is spilled on the phone, and then cleaning is impossible.
18:06.27asteriskmonkeygo with that aastra phones there sick :)
18:06.31asteriskmonkeythey have poe aswell
18:06.45Cinenanyone here had any luck with Dundi? I asked in #Dundi but nobody is home
18:07.08[TK]D-Fenderctooley : Yeah thats the price you pay with anything witha  nice LCD...
18:07.37*** join/#asterisk Meaty (n=cp_simbu@office.abi.ca)
18:07.58ctooley[TK]D-Fender, yeah, and the display does nothing but confuse my users.  The very idea of soft keys confuses them.
18:08.15asteriskmonkeylol
18:08.23MstlyHrmls...
18:08.25[TK]D-Fenderctooley : font forget the Norstar's use soft-keys too, and they are not as easily defined (borders, etrc)
18:08.38[TK]D-Fenderand fewer characters.
18:09.24[TK]D-FenderFUGLY <-  Anybody who can't figure out the display by just loking at it for a sec is pretty technologically challenged IMO....
18:10.03ctooleyMost people are pretty technologically challenged
18:10.24[TK]D-Fenderadd features = add complexity.  more descriptive interface = less complexity.  Theres a balance to be struck in there somewhere.
18:10.38[TK]D-Fenderctooley : I agree.... we should cull the herd a bit ;)
18:10.56ctooleyOne thing that is really irritating about the Polycoms is that the soft keys move
18:11.11ctooleyFor instance the "Send" option moves depending on the status of the handset.
18:11.31[TK]D-Fenderctooley : yeah there are some softkey's that when you go from option to option don't overlay in the most logical order.  but there are worse.
18:12.09ctooleyIf the handset is down and you just start dialing "Dial" is the second one, when you pick up the handset and start dialing "Send" is the first one.  Not only are they named differently, they're in different places.
18:12.13[TK]D-FenderWhat a thing it would be to help design a "superior" phone....
18:12.36ctooleyOr just find a phone manufacturer that had a real UI designer working for them.
18:12.37[TK]D-Fenderctooley : right on the money...
18:12.59asteriskmonkeylets make a touch screen phones nothing but a big lcd and handset
18:13.20ctooleyI'm definitely not a UI expert, but even I can tell moving button locations an renaming them for the same functionality is a bad idea,
18:13.21*** join/#asterisk destructure (n=irish@rrcs-24-173-126-174.se.biz.rr.com)
18:13.23[TK]D-Fenderasteriskmonkey : umm.... NO.  Too fragile.
18:13.30ctooleyasterboy, that sounds like a disaster in the making
18:13.34vaewynHeck yeah...  who needs these @#$#@$ hard buttons... give me a point-of-sale type interface :P
18:13.49*** join/#asterisk heison (n=heison@ns.somanetworks.com)
18:14.17[TK]D-FenderI like Poly's use of soft keys.  Not exactly their location at all times, but the fact and reasons behind it.  it just needs a little tweaking...
18:14.42asteriskmonkeydude a touch screen lcd phone would rock you could make so many cool looking button combos and dial pads that way
18:14.52asteriskmonkeyand have em play like movies when your bored
18:14.53fenlanderasteriskmonkey: like the AT&T broadband phone from their research lab?
18:15.06[TK]D-FenderMeans you can make a "generic" ohone whose abilities are variable.  only the keypad and direct audio controls (volume, mute, spkr, etc) should be "hard"
18:15.09asteriskmonkeyfenlander: never seen that one ? have a link?
18:15.20*** join/#asterisk IOscanner (n=IOscanne@c-24-0-183-125.hsd1.tx.comcast.net)
18:15.21vaewynHeck... a desktop unit like my iPaq PDA running iaxcomm :}
18:15.45[TK]D-Fendervaewyn : My cell is toast and I refuse to pay for anything but a hybrid VoIP one :)
18:15.54IOscanneranhyone have problems with openvox 4 port fxo cards?
18:16.05fenlanderhttp://www.cl.cam.ac.uk/Research/DTG/attarchive/bphone/
18:16.09vaewyn[TK]D-Fender: heh :}  can get them from Japan...  but good luck with any US vendor :{
18:16.37vaewynfenlander: exactly!   I had forgotten about that one
18:16.38[TK]D-Fendervaewyn : wah!  I might be willing to import if they're in english
18:16.41vaewynbut that is the right idea
18:17.03vaewyn[TK]D-Fender: I have seen them in at least 'engrish' mode
18:17.10[TK]D-FenderEEK
18:17.19[TK]D-FenderI'm willingt to pay for a nice one...
18:17.52vaewynone of my co-workers from japan had it... was quite cool...
18:18.53[TK]D-Fender1h20m = CentOS 4.2 download :) 2h30m +/- = Helix 1.7
18:19.20jpablostupid granstream, why they don't have a dialplan in the phone like sipura
18:19.31vaewyn7min = Debian install from local mirror + 1 min asterisk install
18:19.32vaewynhehehe
18:20.01SpaceBassI have 2 of the iPicasso touch screen phones... but they don't seem to have firmware
18:20.03[TK]D-Fendervaewyn : CH34T4|2!
18:20.08vaewynhehehe
18:20.35vaewyn[TK]D-Fender: I have to cheat...  50+ machines to keep in sync :P
18:20.35[TK]D-FenderI need to learn Debian and RH...
18:21.02vaewyndpkg --get-selections | ssh newmachine dpkg --set-selections
18:21.03vaewyn:}
18:23.15*** join/#asterisk Mitsch (n=mh@u-121-144.adsl.univie.ac.at)
18:23.36KriS83Is it possible to set any CIDNumber? I mean can CIDNumber be anything?
18:23.50SkramXKriS83: It cant be a letter.
18:23.54SkramXLOL
18:23.57KriS83ok
18:24.03KriS83cos it's a number :)
18:24.14KriS83No but I mean can I set any number?
18:24.14SkramXCIDName can be letters...
18:24.19SkramXYes..
18:24.22vaewynKriS83: anything numeric... and yes... all the pranks have been done...  like 8675309
18:24.28SpaceBassi wish broadvoice allowed setting the CID
18:24.37SkramXSpaceBass: I heard they did.
18:24.44vaewynNuFone does
18:24.47SpaceBassSkramX reaaaaalllly?
18:24.47KriS83hmm
18:24.47SkramXI have a client who also works at BV, I will ask him
18:25.02SpaceBass[TK]D-Fender lol
18:25.18vaewyn[TK]D-Fender: hehehe.... I have even better...  1-700-867-5309  is mine :}
18:25.20[TK]D-Fendersut
18:25.28[TK]D-FenderSLUT
18:25.29vaewynI need to hook it up to something :P
18:25.36*** part/#asterisk Mitsch (n=mh@u-121-144.adsl.univie.ac.at)
18:25.43SkramXSpaceBass: I just text messaged the guy who works there.. have you asked their Customer Support/
18:25.53*** join/#asterisk tdonahue (n=tdonahue@208.51.101.201)
18:26.03SpaceBassSkramX I have not asked lately
18:26.19*** join/#asterisk SplasPood (i=jwb@ludicrous.paravolve.net)
18:26.21KriS83The thing is I have a number lets say 0694483817 this is my ISDN Phonenumber... but then I also have a so called speical number 01803XXXXX which is forwarded to my 069XXX number.. I can't set the 01803 number as CIDNumber :(
18:27.16SkramXSpaceBass: I was mistaken.
18:27.28SpaceBassSkramX just confirmed the same with a quick google
18:27.34SpaceBassbut NuFone allows setcallerid?
18:27.45SkramXTHey do not allow it. They set it in the background and do an ANI fail, so if they let poeople spoof charged numbers.. hehe
18:27.53SkramXSpaceBass: Yes, so does Iax.cc
18:28.18SkramXI wonder if viatalk allows it
18:28.20SpaceBassdont get me wrong... I'm purely going to prank potential here!
18:28.39SpaceBassactually it would save me the porting process with BV too... but thats a more practical reason
18:28.53zoai called my friends from callerid 2004 on new years eve 2003
18:29.05zoaand gave em a prank call with the poke speaking
18:29.26zoathen made a mistake and had 120 channels call all my friends all night
18:29.40vaewyn"mistake"
18:29.41vaewynhehehe
18:29.46manyhaha
18:29.58zoaif they hung up before it was over (and the recording was 2 minutes)
18:30.00zoait retried
18:30.08zoathat was fun
18:30.11*** join/#asterisk Stealthmethod (n=123@72.242.62.209)
18:30.20*** join/#asterisk SplasPood (i=jwb@ludicrous.paravolve.net)
18:30.23SpaceBassanyone else use iax.cc ? how reliable are they?
18:30.27zoaalso it called people 20 times at the same time
18:30.40zoaleaving 19 messages on the voicemail at a time
18:31.12KriS83Ok I have another question then:     -- Accepting overlap voice call from '06941903117' to '<unspecified>' on channel 0/2, span 7 <- the number shown, which "$var" is this? Cos thats the number I want to be forwarding.
18:31.43*** join/#asterisk Sobakai (n=jmwoodga@45.e6.d12c.cidr.airmail.net)
18:32.16bsdfreakiax.cc is decent, but their support sucks.
18:32.21bsdfreakit's practically non-existent
18:32.22tzangerno, iax.cc is NOT decent
18:32.28bsdfreakheh
18:32.30tzangerstay FAR AWAY from sixtel/iax.cc
18:32.36SpaceBassreally?
18:32.39tzangertheir service does work to an extent yes
18:32.57tzangerbut getting anything updated/changed/troubleshot is a royal pain
18:33.01tzangeras is getting service terminated
18:33.04bsdfreakthat's what i just said.
18:33.05tzangerSTAY FAR AWAY
18:33.05bsdfreakheh
18:33.07bsdfreaksupport is teh sux
18:33.16SpaceBassim happy with BV
18:33.16SkramXI use iax.cc for personal stuff, not Business.
18:33.28tzangerfuck that they get none of my business
18:33.29bsdfreaki'm happy with asterlink
18:33.34bsdfreakvery good support
18:33.35SkramXI got a 25 dollar credit to my account with iax.cc
18:33.41SkramXYes, AsterLink is great.
18:33.47bsdfreaknufone was nice until they had a 3 day outage and wouldn't even tell me wtf was happening
18:33.50bsdfreakclaimed everything was fine
18:33.53tzangerI'd be happy with asterlink if I could figure out why qualify is so shitty with them
18:33.54bsdfreakand NOONE was getting incoming calls
18:33.57justinuif only asterlink had DIDs
18:34.00*** join/#asterisk bkw__ (n=brian@68.32.112.142)
18:34.03tzangernufone is rock-solid but it's good to have backups
18:34.05bsdfreakasterlink does have DIDs
18:34.06SkramXjustinu: I agree.
18:34.08justinuyes, they were talking about you
18:34.12justinulocal DIDs
18:34.13tzangeryes asterlink does have dids and as I siad, they work quite well
18:34.14bsdfreaktz: nufone is hardly rock solid LOL
18:34.17SkramXWell, just 800's
18:34.18bsdfreakthey go out at least once a month
18:34.21tzangerbsdfreak: for termination or origination?
18:34.34bsdfreakfrom what i've heard, both.
18:34.43bsdfreakand from what i've experienced, termination.
18:34.50tzangerbsdfreak: I've been using them as my primary termination for 2.5 years and they ahve not gone offline for termination ONCE
18:34.54vaewynbsdfreak: I have never had them go out...  I have had a router between here and there go south... but once BGP fixed that we wre good
18:35.00SkramXI would like Asterlink to do LOCAL DID's (origination)
18:35.09tzangerI don't use them for origination(I don't use anyone ofr origination)
18:35.10justinuasterlink is the only prepaid origination i tried that doesn't sound like crap
18:35.22bkw__let me get our network 100 times more redundant and renumbered into our own space from ARIN
18:35.30bkw__then more things will come down the pipe
18:35.37bsdfreaksup bkw :)
18:35.38SkramXbkw__: good deal.
18:35.41tzangeryup as I said my *only* "problem" with asterlink is that my iax2 qualify to any of their switches bounces around at least 10 times a day
18:35.45SkramXBut I bet it will take a while.
18:35.50tzangerwhen calls go through they go through perfectly
18:36.01justinui've been using SIP w/ astelink
18:36.04jpablogrrr, stupid GS, they fixed the early dial issue GXP-2000 but no in BTs :(
18:36.07vaewynhey bkw... how's it been?
18:36.12bkw__tzanger, quality has bugs in it yo
18:36.21bkw__vaewyn, yo yo yo
18:36.24bkw__doh
18:36.25bkw__qualify
18:36.26bkw__damn keyboard
18:36.27bsdfreakhehe
18:36.42vaewynquality has bugs in it also these days :P
18:36.43*** join/#asterisk Cresl1n (n=matt@gateway.digium.com)
18:36.46tzangerbkw__: I would tend to agree if it wasn't *just* asterlink I was having trouble with :-)
18:37.10tzangeractually I think I am in the process of uncovering an insidious little IAX2 call progress bug
18:37.59tzangerOffice* --iax2--> Colo* --iax2--> termination  (asterlink, nufone)
18:37.59vaewynOk... there has got to be a way to make this call clean of any "forwarding" or "redirect" telltales before I send it to the norhell
18:38.16tzangeroffice*/colo* use RFC1918 over private link
18:38.27tzangeroffice* CDRs show every call, whether it answered or not
18:38.36tzangercolo* CDRs only show calls that went through
18:38.53tzangerand any calls that did NOT go through seem to have 1-way audio (I can hear them but they can't hear me)
18:38.59tzangercall back it may work, it may not
18:39.05tzangerI have verified that IAX2 packets are flowing
18:39.24tzangerit's like the far side is eithe rnot sending an IAX2 call completion IE or I'm not getting it
18:39.28*** join/#asterisk toddf (n=toddf@ns0.fries.net)
18:39.37bkw__tzanger, we haven't updated our side to the code that doesn't have that bug.. I suspect thats why
18:39.48bkw__it goes away for 10 seconds and comes back
18:39.55tzangerbkw__: what's that??
18:39.56bkw__I just turn that crap off :P
18:40.06bkw__their was a timing bug in those qualify packets or something
18:40.10tzangerbkw__: ahh
18:40.13bkw__it was a few months ago it was fixed
18:40.27tzangerinteresting
18:40.28anthmin exchange for breaking 10 other things =D
18:40.46anthmso alas we cannot inherit the fix
18:41.02bkw__ya we have to plan plan plan upgrades so we don't break stuffs
18:41.13bkw__I usually roll out one drone with the upgrade... let it sit
18:41.17bkw__and see if anything breaks
18:41.27*** join/#asterisk earlt (n=earlt@200.62.22.11)
18:41.33tzangeranthm: :-)
18:42.16anthmtzanger, how's your select hell comming along?
18:42.37ctooleybkw_, is that what happened yesterday?
18:42.45vaewynok... anyone know how to make a call coming in loose all the RDNIS/PRIREDIRECTRESON...etc.. stuff?
18:42.55vaewynI want it virgin when it hits the PRI again
18:43.06tzangeranthm: poorly
18:43.18tzangeranthm: every time I think I get it working well something ocmes up and bites me
18:43.20vaewynnorhell is repressing me
18:43.33[TK]D-Fendervaewyn : Submit to my hack! ;)
18:43.38*** join/#asterisk MikeJ[Laptop] (n=ircatjer@adsl-69-208-116-100.dsl.sfldmi.ameritech.net)
18:43.50tzangercoppice says not to use sched_yield unless you *need* to give up CPU when you normally wouldn't.  poll/cond_wait/sleep all yield just fine and far more effectively
18:44.08tzangerso I took that out, but now I'm having other odd little issues I'm debugging
18:44.09vaewyn[TK]D-Fender: heh...  I might have to
18:44.11tzangervaewyn: what's your specific issues?
18:44.59vaewyntzanger: norhell sees that it is a call that it generated from off campus... and then says "dude! I don't have th option installed to let offsite call offsite"  *smack* call dies
18:45.15tzangervaewyn: well get the option then :-)
18:45.19vaewynevery other call in any manner is fine...
18:45.35vaewyngrrr... that's 4500$ + 120$/month I don't want to spend
18:45.38tzangeryeah
18:45.45tzangeryou don't have DISA?
18:46.01vaewynNope...  no need for it so...
18:46.12tzangercall in on a specific DISA DID ... at second dialtone dial out...
18:46.12*** join/#asterisk Darwin35 (n=Darwin@sta-208-139-193-162.rockynet.com)
18:46.20*** join/#asterisk xianlp (n=xian_1@M1116P028.adsl.highway.telekom.at)
18:46.21Darwin35good morning
18:46.29jpablowhy most of the docs you find about asterisk are from years ago and mention stuff that is no longer relevant :(
18:46.33vaewynand no matter what I set RDNI and such to ... it still sees it as a forwarded call...  has to be a way to scrub that
18:47.04tzangervaewyn: you can't run custom libpri that just strips out the RDNIS IE?
18:47.42vaewyntzanger: only reason we are running into this is we are having the * box send calls to failed phones to a fallback number... and when it sees offcampus->norhell->*->norhell->offcampus it goes "BAD MAN!"   *smack*
18:47.52tzanger[main] loop took 14 ms
18:47.52tzanger[main] loop took 10 ms
18:47.52tzanger[main] loop took 0 ms
18:47.54tzanger[main] loop took 0 ms
18:48.08tzangeranthm: that's my big problem there ... why the blue fuck is my main loop taking 0 ms without poll() getting hit?!
18:48.11tzangerit's literally
18:48.27vaewyntzanger: might have to try that...  should be a way to tell * to wipe that data though
18:48.46tzangerdo { poll() if(poll_result) { } gettimeofday() if(time > alarm1) { } if(time > alarm2) { } } while (!done)
18:48.53anthmis it suppoerd to be a timer ?
18:48.57anthmsame every time?
18:49.04tzangervaewyn: no there is no *-endorsed way to mangle IEs
18:49.33vaewyntzanger: YOu would think though it would let me mangle the RDNIS in * and send the new value
18:49.40tzangeranthm: I just pasted the loop...  it calls poll(wait up to 7ms) every iteration, then compares the time against a bunch of alarms
18:49.45vaewynkindof odd though
18:49.45tzangervaewyn: nope
18:50.00tzangervaewyn: since RDNIS is a PRI term and * has no specific PRI channel
18:50.02vaewynshould make that a forced read only variable then
18:50.14anthmso when nothing is happening it should be 7 every time
18:50.15tzangervaewyn: I'd just mangle libpri at this point to see if that fixes your problem
18:50.27tzangerit should be AT LEAST 7 yes
18:50.44bkw__mangle or strangle?
18:50.45tzangerbut add in the if()s and the gettimeofday() and you're usually up around 9-10ms
18:50.50tzangerbkw__: :-)
18:51.32vaewynhmm... also I am using Dial to send this call out... why does it treat it as a forward and not as a new clall?
18:51.35vaewyncall even
18:52.00Cresl1ntzanger: gettimeofday probably hits it pretty big.  The kernel usually uses that opportunity to schedule()
18:52.03vaewynthat has to be a * thing... cause the PRI has no clue those 2 channels are talking to each other
18:52.16tzangervaewyn: I woudl try and figure out why/what the IEs are getting set to
18:52.31anthmare you calcing your own ms elapsed ?
18:52.32tzangerCresl1n: yep, and that's fine... but why is my main loop not waiting at least 7ms due to the poll?
18:52.37vaewynHmm... I know how to fix this...
18:52.39tzangeranthm: I'm using your awesome code fragment
18:52.54file[desk]brookshire: poke
18:53.02*** join/#asterisk sofh (n=ok@203.101.182.121)
18:53.15sofhhello all
18:53.34sofhcan somebody give me idea , how to integerate a usb CDMA phone with asterisk ?
18:53.42anthmmaybe it's not awesome enuf if it doesnt work lol
18:53.57marcus2so whats this about meetme coming onto the line and saying something every 10 minutes?
18:54.35parylwill ztcfg kill all calls in progress?
18:55.08sofhhey Guys...any idea abuot some USB phone ? like CDMA ?
18:55.12paryli need to initialize a channel on my channel bank
18:56.02tzafrir_laptopsofh, do you know of any USB phone with some sort of Linux interface? Why USB?
18:56.24sofhits configured fine on linux as a modem
18:56.25tzafrir_laptopUSB CDMA phone, of course
18:56.26sofhi can do
18:56.33sofhi can do ATD comm via minicom
18:56.42*** join/#asterisk MikeJ[Laptop] (n=ircatjer@adsl-69-209-166-207.dsl.sfldmi.ameritech.net)
18:56.43tzangerhttp://pastebin.ca/32998
18:56.47sofhso now should configure it via modem.conf or what ?
18:56.48tzangerthat's the mainloop code
18:56.56tzangerhow the fuck it's getting 0ms every now and again is beyond me
18:57.20tzafrir_laptopparyl, ztcfg generally shouldn't if there are no relevant config changes
18:57.50sofhtzafrir_laptop! can you help me a little ?
18:57.51tzafrir_laptopan ATA modem is not something asterisk can use
18:58.01sofhit is not an ATA modem...
18:58.13sofhi just tried to access it jst like an ISDN adaptor
18:58.25sofhbasicaly its a wireless CDMA phone
18:58.39Seldon1975hmm I have two Polycom 301 SIP phones on the network with my * server.  When I dial extension 1000 from 1001 the call pops up in the * console but the phone at 1000 doesnt ring
18:58.43parylactually... i added a channel in zapata.conf, but haven't changed zaptel.conf, so i guess i don't even need to run ztcfg
18:58.56parylbut after adding it, i try to dial the channel and i get "Unable to create channel of type 'Zap'"
18:58.59tzafrir_laptopsofh, blutooth headphone or something?
18:59.01tzangeranthm: any ideas?
18:59.09sofhno its not blue tooth..
18:59.11Seldon1975I have configures [1000] and [1001] in sip.conf
18:59.27sofhits CDMA phone and using CDMA technology...bluetooth isn't that device
18:59.33tzafrir_laptopztcfg applies configuration from zaptel.conf
18:59.38sofhit works over wll
18:59.44tzangersofh: you have a CDMA USB phone?
18:59.52sofhyes tzanger
18:59.58tzangersofh: got a link?
19:00.11sofhtzanger:  link for ?
19:00.14tzangerthe phone :-)
19:00.15*** join/#asterisk Gunnar (n=gunnar@34.80-203-209.nextgentel.com)
19:00.31sofhmy phone is connected to * box via usb port
19:00.40tzangerohhhhhhhhhhh  sorry I understand now
19:00.44vaewynIf I remember right... over the USB link you can only use those as modems... it doesn't have audio paths
19:00.47sofhnow just not getting to play it in modem.conf or where
19:00.59tzangervaewyn: correct, it's just a modem (data only) device.
19:01.14tzangerat least anything I've found
19:01.25sofhmeans there is no way to use a CDMA phone with * ?
19:01.32tzangersofh: not that I'm aware of, no
19:01.32tzafrir_laptop"usb" is as good as "pci". It is a nice pipe to pass data through, but won't give you much more without a more specific driver
19:01.39tzangernot even as an SMS portal
19:02.01sofhhmm....though the correct hardware is installed in linux
19:02.11sofhlike i can dial via minicom
19:02.35vaewynYou can use it like a data modem...  but it is not a "voice" modem
19:02.40tzangersofh: again, you're using it as a data device there, not as a phone
19:02.53sofh:(
19:03.03sofhi wish i could use it just like X100P
19:03.11tzangersofh: unfortunately not
19:03.16paryli added a channel to zapata.conf... shouldn't reload in the console make the changes?
19:03.17tzangersofh: I know what you're up to and it would rock
19:03.20anthmyes
19:03.38sofhyeah tzanger..but not getting a true way..
19:03.43anthmwhen the poll has an event
19:03.50tzafrir_laptopan X100P still has a driver that knows very speicfic details about its controller's registers
19:03.50anthmyou must clear it
19:03.54tzangeranthm: I do
19:03.56anthmor it will return 0
19:03.58sofhwhen asterisk is dialing even via bluetooth to a cell phone..then why don't a CDMA directly connected to itself ?
19:04.06tzangerpollfds[p_dn].revents &= ~(POLLIN | POLLPRI);
19:04.27sofhsuppose we assume it as ISDN BIR , still it can't be use ?
19:04.51tzafrir_laptopsofh, does it provide an interface of an ISDN card?
19:05.01sofhoffcourse not :(
19:06.08sofhby ISDN i mean just to configure it in modem.conf
19:06.09parylso i changed zapata.conf, did a reload, but "zap show channels" doesn't show the channel i added
19:06.33parylhow can i get it to reload?  (without taking asterisk down
19:07.16tzafrir_laptopztcfg applies changes to zap channels (kernel). zapata.conf is the configuration of asterisk's chan_zap.so module.
19:07.33[TK]D-Fenderparyl : gotta take it down... its the zaptel driver that needs to be reloaded and * has to be out of the way for that...
19:07.44*** join/#asterisk brad_mssw (n=brad@shop.monetra.com)
19:07.44tzafrir_laptopon asterisk 1.2 a reload should reload chan_zap
19:08.07tzangeranthm: I do clear it...
19:08.13tzafrir_laptopthough that code is rather untested
19:08.29sofhtzanger: if i'd an ISDN bri adaptor..internal or external what ever..then how to configure * to use it for outgoing channel ?
19:08.37anthmyou read from the fd
19:08.40anthmas much as you can
19:09.20tzangeranthm: I do
19:09.34tzangerI flag condition in the dn thread actions bitmap
19:09.36tzangerand it does that
19:09.47tzangeroh I think I see what you are saying
19:10.00tzangerif that thread doesn't wake up right away I could spin in the main loop with poll() returning right away
19:10.03tzangerer no
19:10.06tzangerer yes
19:10.07tzangerheh
19:10.54tzangersince I didn't read yet but I cleared the event, it'll come right back
19:11.29tzangerlet me remove the dn thread entirely to see if that makes the spinning go away
19:11.37tzangerthe only thing left will be the stdin poll() and the timer stuff
19:11.41*** join/#asterisk shido6 (n=shido6@d221-68-216.commercial.cgocable.net)
19:11.43anthmthe only thing that takes away a poll event
19:11.52anthmis reading the fd dry
19:12.01paryl[TK]D-Fender: d'oh, and, thanks.
19:12.34tzangeranthm: yes I realize that :-)  I understand why poll() may be returning immediately now though..  I'm removing the dn stuff entirely to see if that makes the spinning go away
19:12.35anthmhttp://www.sofaswitch.org/eg/lame.c
19:13.41tzangeryup that does make it go away
19:16.26tzangerwow thanks ofr the insight :-)
19:16.52*** join/#asterisk elg (n=fugalh@falcon.fugal.net)
19:17.12tzangerholy fuck my four-layer board's here already
19:18.01elgany idea if anyone's working on V.150.1 (MoIP)?
19:18.41zoawhat is MOIP ?
19:18.54brad_msswmodem over ip
19:18.58zoaaha
19:19.05zoathat thing works ?
19:19.09elgdemodulate, ip, remodulate
19:19.09brad_msswand yes, it's tied in with the T.38 support that's being worked on
19:19.18brad_msswdon't expect it for a little while though
19:19.28zoawe are working on t.38
19:19.43zoanot the same stack as coppice's
19:20.43brad_msswthink v.150.1 might work via pass-thru with these patches : http://bugs.digium.com/view.php?id=5090    not sure though
19:21.03brad_msswthough I don't know of any ata that supports it :/
19:22.33*** join/#asterisk heison (n=heison@ns.somanetworks.com)
19:22.49SkramXhow nice.. nufone web management is down
19:23.02*** join/#asterisk r0d3nt (n=RatMan@tinfoilhat.net)
19:26.27parylcan Background play wav files?
19:26.31tzangersweeeeeeeeeeet: http://www.mixdown.ca/~andrew/dump/ipg1.jpg
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19:29.55bzbwhi, is there a way to have * to send the voicemail via email but not saving it in the server for a particular extension? Wiki does not seem to have such info
19:32.48*** join/#asterisk bwzb (i=bwz@ip67-153-142-109.z142-153-67.customer.algx.net)
19:33.38bwzbhi, anyone knows whether I can enable the voicemail to be sent via EMAIL ONLY for a select extension? WIKI does not seem to have such info
19:33.52sivanabwzb: yes
19:33.53SkramXwhat do you mean ONLY
19:33.58SkramXlike its not saved on the server?
19:34.37bwzbI have a catch all extension that have way TOO many voicemails, and it is taking the server space quickly
19:34.38Cinenit must store it on the server for a few minutes. It can delete right after it emails it out
19:34.59bwzbThanks Cinen.  How do you set up that way?
19:35.05Cinenwill delete=yes in voicemail.conf noit work for you
19:35.16sivanabwzb: mailbox => password,Catch-All,someemail@mycompany.com,,|attach=yes|delete=no
19:35.30sivanaexcept.. make it delete=yes :)
19:35.47Cinenyea like ivana said
19:35.48bwzbCinen: I did not try delete=yes, I will try that now!
19:35.59bwzbthanks you all!
19:36.01Cinennp
19:36.04Seldon1975are the samples that * creates when you do 'make samples' available thru SVN?
19:37.45*** join/#asterisk santiago (n=santiago@208.195.215.160)
19:37.52Seldon1975does anyone kinow?
19:38.04*** join/#asterisk MikeJ[Laptop] (n=ircatjer@adsl-69-209-166-207.dsl.sfldmi.ameritech.net)
19:38.44bwzbCinen: thanks, you know the link?
19:39.11CinenI do not off the top of my head. Just google it
19:39.50CinenI can send you the file if you want
19:40.41bwzbthx!
19:41.55*** join/#asterisk Micc (n=dotirc@c-24-16-228-130.hsd1.wa.comcast.net)
19:42.03Seldon1975the sample extensions.conf has extension 500 set up
19:42.10Seldon1975what should I hear when I dial this?
19:42.20MiccWhy is it that I can't use G729a for both lines on my linksys pap2-na?
19:42.37MiccI have 5 licenses.
19:43.21sivanaMicc: only 1 at a time
19:43.26Miccsivana, why?
19:43.29sivanaMicc: by  design
19:43.37MiccIs it the linksys that is the problem?
19:43.56sivanaby design on Linksys' side... they didn't put a big enough processor to handle 2
19:43.59MiccIs there another adapter that works both lines?
19:44.25sivanano idea
19:44.30Miccoh thats lame. I need a better adapter then.
19:44.40justinuthe spa's can probably do it
19:44.43iCEBrkrZzzzzzzzzzzz
19:44.44Miccwhats the next best codec to use that it can handle?
19:45.03sivanaI like ulaw myself
19:47.28Micculaw has jitter for me on a cable modem.
19:47.59*** join/#asterisk raptorrat (n=ucs_rat@ab1-1-26.shsu.edu)
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19:48.42*** join/#asterisk waddy (n=waddy@83.218.4.231)
19:49.49waddywhen i try to compile asterisk-addons - i get this error
19:49.50waddy../asterisk: Not a directory
19:49.50waddymake: *** [app_saycountpl.o] Error 1
19:49.56waddyanyone can help?
19:50.33*** join/#asterisk arguile (i=user224@66.38.201.234)
19:50.50*** join/#asterisk seant (n=seant@67.105.255.132.ptr.us.xo.net)
19:51.04denonwaddy: did you get asterisk's souce, and put it in the same directory as asterisk-addons?
19:51.05denon(no)
19:51.13parylis there a way to respond to dtmf tones while musiconhold is playing?  (like a "if you know your party's ext, you may dial at any time" type thing)
19:51.38waddyahh ok ill try that thanks
19:51.47denonfollow the instructions on the download page
19:51.53denonas you're probably missing numerous other steps :)
19:52.03*** join/#asterisk trig_hm (i=jason@home.monkeypr0n.org)
19:53.46seantQuick Question:  Does asterisk support the ability to designate multiple recipients for a voicemail on the fly?  ex.  Dial multiple extensions who should receive the following voicemail.
19:55.13*** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com)
19:57.46KranZseant: yes
19:58.56KranZseant: Voicemail(1234&1235)
19:59.29*** join/#asterisk greekman (n=alex@host254.209.113.199.conversent.net)
20:00.37greekmancan anyone help with manipulating queue entries in the cdr?
20:04.11*** join/#asterisk vexorg (n=vexorg@CPE000021ded913-CM001225419164.cpe.net.cable.rogers.com)
20:11.36[TK]D-Fendergreekman : what do you have in mind?
20:12.49[TK]D-Fenderparyl : when is this MOH playing?
20:13.03*** join/#asterisk eKo1 (n=bernd@metrored-gw.tropicohn.com)
20:13.25*** join/#asterisk MikeJ[Laptop] (n=ircatjer@d14-69-8-30.try.wideopenwest.com)
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20:18.53*** join/#asterisk paljas (n=paljas@tuxtown.xs4all.nl)
20:23.29paryl[TK]D-Fender: i play a background recording and go right to moh
20:23.31parylas soon as it's done playing, you can't dial any digits
20:26.06Seldon1975ive set up extension=>1111 to playback tt-monkeys and when I dial is the Asterisk console says "Playing 'tt-monkeys'" but I dont hear any monkeys!
20:26.20Seldon1975where should this sound file be on the asterisk server?
20:27.03*** join/#asterisk paryl (n=paryl@216-201-177-82.res.logixcom.net)
20:27.21parylhrmm... lost my connection.  Fender: did you respond?
20:27.56greekmanD-Fender: i have a unique situation where I build call files based on an incoming click-to-call. I send the call first to a queue, when an agent picks up, the other leg to the requesting party gets done. The problem is when I look at my cdr, 2 entries are made... the first is the que entry that shows the agent it connects the call to in dstchannel, the second is the actual call with no agent info init, but it has the actual duration of
20:28.34[TK]D-Fendergreekman : ok, out of my league but maybe someone else can pick up on that.
20:28.41greekmanlol
20:28.47greekmanthanks anyway
20:29.02[TK]D-Fenderparyl : WHY are they getting MOH and what kind of options are they supposed to be able to enter?
20:29.07*** join/#asterisk kimosabe (n=kimosabe@201.135.10.173)
20:29.26[TK]D-Fenderparyl : You mean like you want background music while they make up their mind?
20:29.42ctooleygreekman, which click to call provider are you with?
20:29.47[TK]D-Fendergreekman : Good description though, just wait around a bit for an answer on that...
20:30.28ctooleygreekman, I write my own cdrs for this reason.  Doing an Originate through the Manager interface instead of using call files might work.
20:32.41kimosabei am trying 2 use asterisk as sip client i have 2 sipuras and on my asterisk box i configured one icoinnect acount the servers says its conected but when i dial it says everyone is busy
20:33.29kimosabei want the sipuras to connect 2 asterisk box and dial out threw icconect
20:34.18kimosabecan some one help me out a bit
20:36.02parylD-Fender: actually, i WANT them to be thrown into a queue and allow them to dial an extension if they have to hold... but i don't know if that's possible
20:36.13Seldon1975<PROTECTED>
20:36.19Seldon1975any ideas
20:36.48[TK]D-FenderSeldon1975 : You need to "Answer" first. <-
20:37.10Seldon1975oh
20:37.12bwzbhi, after building 1.2.1, I can not restart my *, anyone know what seems to be the error? I remember someone mentioned about changing a header or something
20:37.18Seldon1975how do I do that Fender?
20:37.21[TK]D-Fenderparyl : ok, thats already easy to do (a dial-out menu while in queue)
20:37.33bwzbthe error is like this:  [cdr_addon_mysql.so]Dec  8 12:29:22 WARNING[25951]: loader.c:325 __load_resource: /usr/lib/asterisk/modules/cdr_addon_mysql.so: undefined symbol: ast_load
20:37.34bwzbDec  8 12:29:22 WARNING[25951]: loader.c:554 load_modules: Loading module cdr_addon_mysql.so failed!
20:37.34[TK]D-FenderSeldon1975 : Do an Answer before your playback!
20:37.48Seldon1975ok thanks
20:38.11bwzbthe last error: Ouch ... error while writing audio data: : Broken pipe
20:40.45bwzbmaybe 1.2.1 change something in dynamically linking the modules: __load_resource: /usr/lib/asterisk/modules/cdr_addon_mysql.so: undefined symbol: ast_load
20:43.19bwzbanyone here can help me???
20:45.05Seldon1975Fender, i put the Anser in - still no monkeys :(
20:45.11*** join/#asterisk QbY (n=QbY@adsl-068-209-210-253.sip.cha.bellsouth.net)
20:45.20Seldon1975the * console shows the Answer and the PLayback
20:45.24QbY<PROTECTED>
20:46.06ctooleybwzb, you didn't build the asterisk-addons package for version 1.2.1
20:46.09*** part/#asterisk elg (n=fugalh@falcon.fugal.net)
20:47.39parylD-Fender: how do you do a dialout menu?  i know how to do '0' for an operator, but i thought that was just built-in
20:48.17*** join/#asterisk The_LightSide (n=lightsid@wbs-196-2-122-41.wbs.co.za)
20:48.24parylbut i don't know how to control the scripting of 'play this message, then music, then this message, etc'
20:49.06The_LightSidehi all, where could i find info on how to setup AAH2.1 with the built in bristuff?
20:51.01The_LightSidehello?
20:51.29greekmanctooley: I made my own click-to call
20:51.33*** join/#asterisk kubejm (i=kubek@AMontpellier-151-1-10-135.w83-205.abo.wanadoo.fr)
20:51.48*** part/#asterisk kubejm (i=kubek@AMontpellier-151-1-10-135.w83-205.abo.wanadoo.fr)
20:53.11parylD-Fender: the bigest thing i'd like to do that i can't seem to figure out, is i'd like the queue to dial any available agents immediately, and only play a 'all reps are busy' if the customer has to be thrown into MOH
20:54.17Seldon1975when I dial an outside line I get:
20:54.36bwzbctooley: do I need to build  asterisk-addons?
20:54.55Seldon1975NOTICE[3292]: pbx.c: 1731 pbx_extension_helper: Cannot find extension context 'default'
20:55.02Seldon1975anyone know what this means?
20:55.09ctooleyyou do if you want cdr_ADDON_mysql.so
20:55.32*** join/#asterisk klictel (n=klictel@207.107.208.137)
20:56.56The_LightSidewhere could i find info on how to setup AAH2.1 with the built in bristuff?
20:57.30ctooleyThe_LightSide, repeating your self will not get you answered.  If someone knew, they'd answer you
20:57.32*** part/#asterisk Samoied (n=Samoied@popeye.opens.com.br)
20:57.36KranZSeldon1975: means you dont have a default context
20:57.57The_LightSideok....
20:58.02The_LightSideta
20:58.24KranZ@home blah
20:58.32*** join/#asterisk peter_l (n=ploeppky@store-fw.porchlight.ca)
20:58.37KranZthey should drop that
20:59.02*** join/#asterisk mountie (n=mountie@CPEdeaddeaddead-CM000a739acaa4.cpe.net.cable.rogers.com)
21:01.00mutilatorDOH!
21:01.02mutilatorFrSIRT.com RSS Feeds:
21:01.02mutilator2005-12-08 This XML feed has been disabled. FrSIRT security advisories are available on FrSIRT.COM
21:01.27[TK]D-Fenderparyl : Sorry, they get MOH until the call is answered...
21:03.16KranZmutilator: they prolly need AD revenue
21:05.15bsdfreakpoop dawg
21:05.36*** join/#asterisk Stealthmethod (n=123@72.242.62.209)
21:05.43*** part/#asterisk Stealthmethod (n=123@72.242.62.209)
21:07.26parylok, how about this... when a call in a queue starts ringing to an extension, will it continue to ring that extension if a timeout occurs?
21:07.40paryli.e. could i just set the first queue to timeout in 5 seconds
21:07.40KranZif you want
21:07.42paryl?
21:08.12zoahey hp
21:08.13zoaho
21:08.37file[desk]zoa: zoa zoa zoa
21:09.36*** join/#asterisk flashbac1 (i=flashbac@68-235-251-168.atlsfl.adelphia.net)
21:09.43flashbac1hello!
21:09.47*** join/#asterisk supaigtr (n=yurplsl@152.53.16.10)
21:09.50flashbac1any realtime gurus here?
21:10.08supaigtrAnyone seen the adtran Netvanta 7100?
21:10.26zoafile file file
21:11.32zoago poke yourself you dirty bastard!
21:11.41file[desk]:p
21:12.06*** join/#asterisk saftsack (n=saftsack@p54A7FE77.dip.t-dialin.net)
21:12.11*** join/#asterisk nagl (n=nagl@213.235.241.6)
21:12.12[TK]D-Fenderparyl : sort of against the idea of a queue....
21:12.29saftsackhi
21:12.29flashbac1hey guys, i keep getting this error with realtime:
21:12.44flashbac1Dec  8 16:04:37 WARNING[1817]: res_odbc.c:171 odbc_smart_execute: SQL Execute returned an error -1: 42000: [Sybase][ODBC Driver][Adaptive Server Enterprise]Implicit conversion from datatype 'CHAR' to 'INT' is not allowed.  Use the CONVERT function to run this query.
21:13.07flashbac1i think this might be a bug...
21:13.15*** part/#asterisk QbY (n=QbY@adsl-068-209-210-253.sip.cha.bellsouth.net)
21:13.17zoait looks like a bug
21:13.23saftsacki defined in my extensions.conf that if someone called the number xyz asterisk answers. and then the caller can call after hearing a voicemessage a special telephone. but now theres a problem
21:13.24zoaor you fucked up your table
21:13.25bsdfreakbega
21:13.49saftsackif the caller hangs up, the other telephone will ring and ring and ring ...
21:13.50*** join/#asterisk n4y (n=tmalkut@fw.orasoft.net.pl)
21:13.54saftsackit wont stop ringing
21:14.01saftsacksomeone can help?
21:14.23KranZsupaigtr: that thing is a monster
21:14.30*** join/#asterisk Little-L (n=daniel@0x50a471db.svgnxx1.adsl-dhcp.tele.dk)
21:17.28supaigtrMonster???
21:17.39Jessteranyone successful making custom ring tones for Cisco 7940/7960? i've followed what i've read and still no luck.
21:17.40KranZyes
21:17.51KranZit does about everything
21:18.01supaigtrMonster as in penis or breasts or monster as in problem?
21:18.05Darwin35what is everything
21:18.13*** join/#asterisk A-Tuin (n=a-tuin@steves.ip.v4.me.uk)
21:18.14Darwin35do you have call back if busy
21:18.16paryl[TK]D-Fender: my thought is, if they come into the queue and the call isn't answered immediately, they will be bumped out to an 'all reps are busy' message, and then put back into the queue
21:18.16supaigtrKranZ: Supports polycom?
21:18.21Darwin35do you have fallow me working
21:18.35KranZsupaigtr: it supports sip
21:18.50Darwin35do you  have it call oyu vibrator on tech calls
21:19.05*** join/#asterisk dudes (n=dudes@12-215-32-62.client.mchsi.com)
21:19.07supaigtrKranZ: Any provisioning support for phones?
21:19.16supaigtrKranZ: is it asterisk based or other?
21:19.20[TK]D-Fenderparyl : there is already a "thanks for holding" message that you can configure, including its frequency.  I might suggest playing a message BEFORE the call enters the queue so they know they might be holding for a while
21:19.21Darwin35I still need call back wen busy
21:19.25Darwin35grr
21:21.56*** join/#asterisk Assid (i=assid@59.183.2.83)
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21:24.03DrDekeHey guys, is there any way in an AGI script, to receive DTMF digits without the # meaning "stop receiving" ??
21:24.17DrDeke(I would like to receive the # into my application just like any other digit)
21:24.21[TK]D-FenderDrDeke : Tons
21:24.50[TK]D-FenderLook at the AGI command list.  Should be pretty obvious....
21:25.11bwzbanyone know what is the issue with these warning message:
21:25.12bwzbDec  8 13:24:42 WARNING[32300]: format_wav_gsm.c:243 update_header: Unable to find our position
21:25.12bwzbDec  8 13:24:42 WARNING[32300]: format_wav.c:247 update_header: Unable to find our position
21:25.32*** join/#asterisk devel (n=devel@wiggum.digitalcoven.com)
21:26.36*** join/#asterisk darby_t (i=darby_t@dkr19.neoplus.adsl.tpnet.pl)
21:26.38bwzblooks like everyone is on vacation now
21:27.20*** join/#asterisk dtev001 (n=mikeh@cpe-24-168-18-30.si.res.rr.com)
21:27.43dtev001hey there... when you create a .call file, what context does asterisk use to dial out the call
21:28.03DrDekeTKD: I am looking at the AGI command list, but get_data appears to be #-terminated and wait_for_digit only takes one digit...
21:28.14DrDeke(actually it doesn't appear to be #-terminated, I tried it and it is)
21:28.50*** join/#asterisk _ozstealth_ (n=ozstealt@64.141.48.2)
21:28.59*** join/#asterisk jpablo (n=jpablo@200.94.130.194)
21:29.19jpablohi, has anyone upgraded to the 1.0.7.11 (beta) firmware in a budgetone ?
21:29.38jpabloit is trying to download bt-100.bin that isn't included in the firmware zip
21:29.53_ozstealth_Hi There,
21:30.22DrDekeTKD: Any other ideas?
21:30.25_ozstealth_I have a question relating to making outbound calls with a recorded (or tts) message - is this the right place to be asking?
21:30.35*** join/#asterisk Oryn (i=oryn@falcore.fsck.tv)
21:30.36DrDekeoz: Yeah, pretty much.
21:30.49_ozstealth_DrD
21:31.00_ozstealth_can you provide any advice?
21:31.18DrDekeWhat you probably want to look into is a "callfile"
21:31.18DrDekej/sec
21:31.25_ozstealth_k
21:31.37DrDekehttp://www.voip-info.org/tiki-index.php?page=Asterisk+auto-dial+out
21:31.39Orynanyone running the latest bristuff ? I cant get * to reload
21:32.35Jessteranyone successful making custom ring tones for Cisco 7940/7960? i've followed what i've read and still no luck.
21:33.03*** join/#asterisk Stealthmethod (n=123@72.242.62.209)
21:34.10*** join/#asterisk [Airwolf] (n=airwolf@attilla.nl)
21:36.45*** join/#asterisk tuxinator_linuxM (n=spabin@70-32-106-248.ontrca.adelphia.net)
21:36.46*** join/#asterisk Micc (n=dotirc@c-24-16-228-130.hsd1.wa.comcast.net)
21:37.20Miccwhy does my callerid show me the number instead of the name when I do set CALLERID(name)=something
21:41.59*** join/#asterisk southtel (n=slester@c-24-30-2-230.hsd1.ga.comcast.net)
21:44.52KranZMicc: paste your code for that
21:45.01*** join/#asterisk Flixor- (n=Flixor@ip5457002f.direct-adsl.nl)
21:45.03KranZthe line
21:45.06Flixor-hi everybody
21:45.34_ozstealth_DrD: Thanks for that - that page and one that links from it look awesome. Thanks for your help
21:45.39DrDekeCool, good luck.
21:45.41Miccexten => s,1,Set(CALLERID(name)=GlycoSystem)
21:45.45DrDekeFeel free to ask in here if you have any more questions.
21:46.45southtelAnyone know of any reason why a DeadAGI+phpagi app would end immediately when the user hangs up?
21:46.56KranZMicc: http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Set
21:46.58MiccI'm guessing that the PRI provider doesn't allow us to change the name.
21:47.22Flixor-guys is it possible to call on linux via the voipbuster network
21:47.35KranZyour telco does a "DIP" in the National CNAM database for caller name information
21:47.37Flixor-with a client
21:48.13KranZMicc: so to change the name, you need access to the cnam db, which you wont get unless your a telco
21:48.21KranZMicc: so ask the telco to change the name
21:49.53MiccKranZ, maybe the name was dropped when we transfered the number.
21:50.02azzieguys... 16bit PCM for SoX is .ul or .uw ?
21:51.11flashbac1can someone help me with a realtime problem?
21:51.20KranZMicc: yeah, ask whoever you ported over to to update the cnam info
21:51.44Beirdoazzie: ul is unsigned long (32 bit), uw is unsigned word (16 bit), ub is unsigned byte (8 bit) if my memory serves
21:51.54Flixor-KranZ, do you know i could use my voipbuster account on linux
21:52.05Beirdoand you can use sl, sw, sb too for signed
21:52.11Beirdothat's my memory of it
21:52.20KranZMicc: i think its silly that they have to lookup caller name info and it cant just get passed along with the ani
21:52.37KranZFlixor-: voipbuster?
21:52.46Flixor-yes
21:52.53KranZto do what
21:52.58Flixor-to call
21:53.07Flixor-check out www.voipbuster.com
21:53.13Flixor-they have only a windows client
21:53.30Flixor-but then somebody said to me that you could call under linux allso because its using the aix protocol
21:53.42KranZyou man iax?
21:53.44KranZer mean
21:53.48Flixor-ehm sorry
21:53.50Flixor-yes iax
21:54.09KranZim sure there's a softiax client out there somewhere
21:54.09*** join/#asterisk Assid (i=assid@59.183.2.83)
21:54.21DrDekeyeah i don't think too manypeople run asterisk on AIX machines ;)
21:54.52KranZor you can setup * to route to voipbuster and use a linux sip client to connect to *
21:55.04*** part/#asterisk southtel (n=slester@c-24-30-2-230.hsd1.ga.comcast.net)
21:55.27KranZ[linux sip client] <-sip-> * <-iax-> voipbuster
21:55.38*** join/#asterisk g__ (n=g@itd01fw-fibre.itdepartment.com)
21:56.06Flixor-ehm
21:56.10waddyneed help with an error when compiling chan capi
21:56.11waddymake: *** [chan_capi.o] Error 1
21:56.18MiccKranz, when I set the caller id number should it include the 1 and can it contain dashes?
21:56.25synthetiqhow can u limit the number of concurrent calls a line can have?
21:56.25Flixor-well i have kiax installed KranZ
21:56.40KranZMicc: only 10 digits
21:56.49KranZFlixor-: have you tried it?
21:56.54*** join/#asterisk pryk (n=tmalkut@fw.orasoft.net.pl)
21:57.07*** part/#asterisk _ozstealth_ (n=ozstealt@64.141.48.2)
21:57.13Flixor-ehm yes i seems that i can login to the server
21:57.14*** join/#asterisk ]expic (i=xuy@h160-60.uni.net.ua)
21:57.22Flixor-but then again when i call to myself it doesnt work
21:57.25KranZMicc: literally 10 digits
21:57.36]expicdoes anybody know how to get calling SIP Ip address in AGI variable?
21:57.43KranZdoes the windows version work?
21:58.04DrDekeI find that some mobile phone operators in the USA prepend a + to their incoming callerID values, which of course forms an invalid number unless you include the "1"
21:58.17]expicdoes asterisk set SIP client IP to some special variable?
21:58.19DrDeke(On the other hand, I have not yet tried including the "1" and seeing if that works on a landline callerid box)
21:59.06Flixor-yes the windows version does work
21:59.49KranZFlixor-: dunno, you'll have to start debugging your linux setup
22:00.12]expici need to track client IP addresses in PHP AGI script
22:00.12g__Hi, I'm having a problem where Agents answering calls from a queue can't transfer those calls to another extension.  around 50% of the time they get dropped.  I've searched the mailing lists, and I've found several people mentioning a similar problem, but I haven't found any solutions.  Any ideas?
22:00.14]expicplz help me
22:00.51KranZg__: do you have the latest * version?
22:01.12g__The latest old stable (1.0.9)
22:01.31KranZg__: do they know how to transfer?
22:01.33g__The agent code is a bit newer though.. are there reported problems with the 1.0.x branch?
22:01.54g__Yes: we just tried 5 times in a row.. on the 5th one it failed.
22:01.55*** join/#asterisk southtel (n=slester@c-24-30-2-230.hsd1.ga.comcast.net)
22:02.06KranZdid it spit out an error?
22:02.20g__No, there's nothing on the console.
22:02.34g__The caller remains "off hook" but without hold music.
22:02.34KranZwho hungup 1st
22:03.18*** join/#asterisk nagl (n=nagl@213.235.241.6)
22:03.18g__Neither.. it was an attended transfer: A joins the queue, B answers, B talks to C brief, then hits 'transfer'.
22:03.26KranZare you using flash to transfer?
22:03.42g__These are hardware Polycom Sip phones.
22:04.20KranZhmm...
22:04.40KranZg__: [TK]D-Fender knows alot about polycom phones but he's usually on in the morning
22:04.47KranZg__: you should try asking him
22:04.50DrDekeTKD was in here about 30 minutes ago
22:04.52Drukeng__: i had this problem a while ago, i don't think i ever fixed it... (don't remember) make sure your contexts are correct :)
22:05.04Drukenremember you can't transfer to an extension that doesn't exsist
22:05.05g__[transfer] extention [send] .. etc.
22:05.29g__KranZ: I'll ask him first thing tomorrow.. thanks for the suggestion.
22:05.35KranZnp
22:06.20g__Druken: the contexts look ok.  The 'attended transfers' work well: B gets to talk to C without any problems.
22:06.50g__It's just the Zap/1-1 line that gets "lost" in the system as soon as the agent hits "transfer".
22:07.08waddyanyone can help me with chan_capi 0.3.5 ?
22:07.18g__Still, I appreciate you checking the obvious..
22:07.33waddyi get an error on make
22:07.43waddychan_capi.c:4920: warning: implicit declaration of function `capi20_release'
22:07.43waddychan_capi.c:4933: dereferencing pointer to incomplete type
22:07.43waddychan_capi.c:4936: dereferencing pointer to incomplete type
22:07.43waddymake: *** [chan_capi.o] Error 1
22:07.54waddyIts on RHEL3
22:08.29southtelCould someone help me clear up some quiestions about AGI?
22:08.32waddyits doing my headin ive tried everything
22:08.55southtelSpecifically, I'd like to know just what DeadAGI is used for.
22:09.19g__Anyways, thanks for the suggestion Druken and KranZ!
22:10.25]expicdoes asterisk set SIP client IP to some special variable?
22:11.46*** join/#asterisk toddf (n=toddf@ns0.fries.net)
22:12.56KranZg__: np, good luck
22:12.57southtelexpic: I'm pretty sure that there isn't.
22:13.26g__Thanks KranZ.  What do you use Asterisk for?
22:13.41]expicsouthtel: how it's easy to get client IP from AGI script if this IP is dynamic?
22:14.41*** join/#asterisk Dead-Bum (n=Satan@tor/session/x-6ac3df61d4807d5e)
22:14.41southtelexpic: I'd guess it's possible, but it may take some fenagling.
22:15.32southtelexpic: These SIP users, are they calling in?  What are you trying to do?
22:15.57]expicsouthtel: they are calling in
22:16.04]expicsouthtel: i do prepaid service for them
22:16.20]expicsouthtel: now i want to create additional field with client IP in CDRS
22:16.33]expicsouthtel: but cannot find any predefined channel variable with client IP
22:17.00*** join/#asterisk Dead-Bum (n=Satan@tor/session/x-7433e025c63f1248)
22:17.15Dead-BumI can't seem to find a clear guide on getting asterisk to register through openSER (or regular ser for that matter) anyone got a good link?
22:17.21]expicsouthtel: regarding deadagi, it's used when the channel is hanged up
22:17.51KranZg__: pstn gateway, ivr, call-forwarding, directory, emergency voicemail, time of day routing
22:18.29KranZg__: i use SER as my SIP registrant/proxy
22:18.41southtelexpic: They're calling in over SIP or over the PSTN?
22:18.54g__How's SER treating you?  I've been considering SER for a while now..
22:19.12KranZg__: good, it's quite stable
22:19.16*** join/#asterisk RoyK (n=roy@55.80-202-161.nextgentel.com)
22:19.28KranZg__: i like the dialplan configuration better b/c its more programming oriented
22:19.29southtelexpic: I thought that DeadAGI also allowed the agi to continue on a channel once it's hung up.  Is that not so?
22:19.39KranZg__: and you can restart it w/o dropping calls
22:19.43g__Does it take a lot of time to keep the configuration in sync in asterisk?
22:20.00*** part/#asterisk ctooley (n=ctooley@rrcs-24-227-212-162.sw.biz.rr.com)
22:20.05KranZg__: my SER conf is pretty static
22:20.10g__(I understand Asterisk still needs to handle voicemail if you're using SER.)
22:20.16KranZyeah
22:20.31]expicsouthtel: don't think so, because it starts on h extension, when call is hanged up
22:20.37g__Does it have to change every time you add or modify a sip client?
22:21.07KranZg__: nope, u just add them to the database
22:21.17KranZ(db on SER box)
22:21.35KranZ...and of course make a voicemail box on *
22:21.37g__Berkley DB or mysql/postrgres?
22:21.42KranZmysql
22:21.50g__Intersting..
22:22.08KranZyou should start learning it now, the sooner the better
22:22.10g__thanks for the rundown, I'll certainly have to look at it.
22:22.15southtelexpic: if your users are connecting via SIP to initiate calls, then you could look at the channel.
22:22.22g__It improved the stability of your system?
22:22.33KranZit takes a load off *
22:22.52southtelexpic: the channel will have the SIP id, which you probably could use to then do a lookup.
22:22.55KranZand allows * to be on a private network
22:23.14g__We have several PBXes at many sites, so SER has been on my list of things to look at for a while.
22:23.20KranZDrDeke: what lang agi?
22:23.30g__Nugget: is that SER-related tong mutilation?
22:23.33KranZso you do office * setups?
22:23.39bsdfreaktao
22:23.51DrDekeKranZ: Perl. Actually, though, the AGI portion of the system works fine; I am just having trouble with the perl script I am using to process the output of the AGI script :)
22:24.12Nuggetno, mysql-induced.
22:24.16DrDekeWhat I am working on is not that hard, but I keep screwing it up ;). I really should just do it tomorrow, but I want to finish it bfore I go home for the day.
22:24.16DrDekebefore*
22:24.25g__I have perl scripts to handle individual sites, but inter-site configuration is still all manual.
22:24.25KranZDrDeke: i need to get into that, you do any db backend stuff with agi (non * db)?
22:24.37*** join/#asterisk Lurr (n=pr0ph3t@m615e36d0.tmodns.net)
22:24.48*** part/#asterisk Lurr (n=pr0ph3t@m615e36d0.tmodns.net)
22:24.52g__Nugget: do you have a choice with SER?  (I heard rumour * doesn't support postgre.)
22:24.53DrDekeKranZ: Not yet, but most programming/scripting languages (particularly perl) have very easy interfaces to various databases.
22:24.58Nuggetdunno
22:25.01]expicsouthern: yes i have sip id
22:25.16Nuggetyou can use postgresql for realtime in asterisk.  at least, some people do.
22:25.27NuggetI dunno how solid the support is.  I've never looked at realtime.
22:25.43*** join/#asterisk dalabera (n=dalabera@146.82.190.164)
22:25.50dalaberahello everyone
22:26.39dalaberainstalling 1.2.1... Is make linux26 removed from the install process?
22:26.43southtelexpic: what flavor of agi are you using?
22:26.57SwK[Work]damn it batman
22:27.56KranZdalabera: are you in linux 2.6?
22:27.56]expicsouthtel:  http://apollo.bcwireless.net/~matthewa/phpagi/
22:28.17dalaberayes
22:28.23g__Mysql might suck as a relational database.. but maybe it's not so bad as a directory service.
22:28.34*** join/#asterisk vexorg (n=vexorg@CPE000021ded913-CM001225419164.cpe.net.cable.rogers.com)
22:28.37*** join/#asterisk kokey (n=ubunture@201.153.63.79)
22:28.41KranZdalabera: so why not just use "make"
22:28.53southtelexpic: in that case, you can probably use phpagi-asmanager
22:28.53waddyanyone can help me with chan_capi 0.3.5 ?
22:28.54Orynis anyone using sipdiscount.com? I cant get any audio via my sip phones, but my isdn phones work fine
22:28.58waddyi get an error on make
22:29.00waddymake: *** [chan_capi.o] Error 1
22:29.24southtelexpic: You'll want to do a SIPshowpeer, and then parse the results.
22:31.19dalaberaKranz How long you being using Asterisk?
22:31.26SwK[Work]anyone know a good bug in chan_sip where on a reload chan_sip.so chan sip just hangs with no indication as to why
22:31.50KranZdalabera: almost 2 years
22:32.23dalaberagood
22:32.42KranZi wish it were more
22:33.05]expic<southtel> yes thank you
22:33.14dalaberathen you should knew that there used to be a make linux26 especifically for linux 2.6 kernel
22:33.48*** join/#asterisk Koshatul (n=evangeli@ip157-65-132.cust.bit.net.au)
22:34.04KranZdalabera: i might have run across it, but never needed it
22:35.02]expicsouthtel: actually i can easyly do that executing asterisk -r -x and parsing results, don't think i need to use this library just for getting IP
22:35.07southtelexpic: no problem.  That's not ideal, but it works.
22:35.33*** join/#asterisk toddf (n=toddf@ns0.fries.net)
22:35.35southtelexpic:  Much better!  Much less overhead as well.
22:36.23DrDekeshit.
22:36.49DrDekeWell, I think it's officially time to quit working for the day, as I just overwrote some unbackedup work of mine while making a mistake with I/O redirection :X :X :X
22:37.32*** join/#asterisk tainted- (n=somewher@mail.k2usa.com)
22:38.47DrDekeYeah, I know, but I only wrote this a few hours ago
22:39.11KranZheh
22:39.19DrDekeIt won't be particularly hard to replace, it's just annoying when you do something that stupid.
22:39.58*** join/#asterisk MikeJ[Laptop] (n=ircatjer@d14-69-8-30.try.wideopenwest.com)
22:40.14test34What do I need to connect a VoIP adapter to the computer so I can use asterisk ?
22:40.27sivanaheh
22:40.31ManxPowertest34, money.
22:40.34KranZa usb cable perhaps
22:40.35DrDekeHA!
22:40.36sivanaVoIP adapter to a computer
22:40.47ManxPowertest34, Buy a SIPura box.
22:40.51sivanatest34: why do you want to connect it to your computer?
22:40.53DrDekeyeah. money talks, especially when it comes to telephone systems ;)
22:41.02KranZno pun intended
22:41.24test34so I can use it as FXS ?
22:41.33KranZwww.sipura.com
22:41.34KranZyes
22:41.50DrDekeOnly some sipura units have an FXS port though.
22:41.55DrDekeYou would need to make sure you get one that does.
22:41.58sivanatest34: you have a computer with asterisk AND an adapter?
22:42.03*** join/#asterisk MikeJ[Laptop] (n=ircatjer@d14-69-8-30.try.wideopenwest.com)
22:42.09DrDekeSorry, it's only some that have fxO ports
22:42.19KranZthe spa2100 is what i recommend
22:42.36test34sivana, I only have asterisk right now, but I will get an adapter with the cable company when they come install it
22:42.45KranZheh
22:42.48KranZroadrunner
22:42.50sivanawhat kind of adapter?
22:42.51KranZ?
22:43.28synthetiqhow can u limit the number of concurrent calls a line can have?
22:43.33KranZtest34: most likely you wont be able to do anything with it except plug a coax cable into it, your ethernet and your phone
22:43.36sivanasynthetiq: groups
22:43.47test34sivana, I'm not sure
22:44.02KranZtest34: what cable company?
22:44.11test34KranZ, brighthouse networks
22:44.20*** join/#asterisk MikeJ__ (n=ircatjer@d14-69-8-30.try.wideopenwest.com)
22:44.33synthetiqis that a coman sivana
22:44.36sivanasynthetiq: look on the wiki... it's with group count or something
22:45.06*** part/#asterisk g__ (n=g@itd01fw-fibre.itdepartment.com)
22:45.09test34they will plug my phone line to the adapter so I can use the regular phone plugs
22:45.15*** join/#asterisk saftsack (n=saftsack@p54A7FFDA.dip.t-dialin.net)
22:45.19saftsackhi
22:45.38*** join/#asterisk Meaty (n=cp_simbu@office.abi.ca)
22:45.38KranZsynthetiq: you could set and increment a global variable, then de-increment it when the call hangs up
22:45.47KranZsynthetiq: add some gotoifs
22:45.49sivanaKranZ: it's even easier than that
22:45.52sivanaI'm looking it up
22:46.06KranZsivana: there's a line in sip.conf that'll do it for sip
22:46.35sivanaSetGroup
22:46.48sivanathe sip parameters are deprecated
22:46.48*** part/#asterisk dca_ (n=dca@sta-208-139-193-162.rockynet.com)
22:46.49sivanahttp://www.voip-info.org/wiki/view/Asterisk+cmd+SetGroup
22:47.18sivanasynthetiq: look there.. it even has an example
22:47.35KranZsivana: good find, i hadnt messed with that (nor needed).
22:50.30ManxPowerKranZ, he should be aware of race conditions, however.
22:51.36*** join/#asterisk InfideNino (i=InfideNi@164-233.surfsnel.dsl.internl.net)
22:52.18*** join/#asterisk MikeJ[Laptop] (n=ircatjer@d14-69-8-30.try.wideopenwest.com)
22:52.18*** join/#asterisk drray (i=drray@dsl254-011-243.sea1.dsl.speakeasy.net)
22:52.28InfideNinohi all
22:52.33InfideNinocan somebody help me?
22:52.45InfideNinoi need cisco firmware for a 7920...
22:53.01ManxPowerInfideNino, Call Cisco.  What firmware do you need?
22:53.05sivanaInfideNino: now that would be illegal
22:53.10ManxPowerSIP, SCCP/Skinny, MGCP, or H323?
22:53.24InfideNinoyes i know you need a contract, but i'm just a home user (poor student...)
22:53.35InfideNinowell if you have sip that would be amazing
22:53.41InfideNinobut i was looking for sccp
22:53.42sivanaeveryone's a poor student
22:53.55*** join/#asterisk mog_work (n=mogorman@gateway.digium.com)
22:54.13InfideNinohow much is a cisco contract anyway?
22:54.21sivanaso you didn't even call :p
22:54.21drray88 is waht I paid
22:54.36drray8 for the phone, and 80 for the general contract
22:54.45sivanaheh
22:54.55*** part/#asterisk [Airwolf] (n=airwolf@attilla.nl)
22:54.55InfideNinowell, how can i call them if my phone isn't working with the old firmware that it was shipped with ;-)
22:54.57test34sivana, I will be getting a digital phone/modem combo
22:55.02sivanathe firmware is free.. it's 88 million for the CD it came on
22:55.09*** join/#asterisk MikeJ[Laptop] (n=ircatjer@d14-69-8-30.try.wideopenwest.com)
22:55.14ManxPowerInfideNino, A support contract ($10/year) MIGHT get you access to the firmware, but it would not technically be a legal download
22:55.20*** part/#asterisk DrDeke (i=dekemar@deculator.engin.umich.edu)
22:55.22test34err.. digital phone (voip) adapter
22:55.44InfideNinoso if it's not a legal download anyway, one of you guys might as well just share it with me
22:56.02ManxPowertest34, Which specific make/model?
22:56.19InfideNinoi guess we have a cisco contract at work, but no voip; could i use that?
22:56.20test34I'm on hold, I should know soon
22:56.31test34(on the phone with them)
22:56.31ManxPowertest34, companies like AT&T, Vonage, etc LOCK the adapter to the service so you can't use it with anything else.
22:56.40ManxPowertest34, well come back when you know.
22:56.45sivanaInfideNino: if you can access TAC, you can download it
22:56.47test34ok
22:56.57InfideNinowhat's TAC?
22:56.59KranZi need to start thinking of new year's plans
22:57.03ManxPowertest34, You are not getting it via a service provider, are you?
22:57.12sivanaInfideNino: cisco's download area
22:57.29test34ManxPower, I get it from brighthouse networks.. I shouldnt ?
22:57.50ManxPowertest34, never heard of them, but as I said most companies LOCK their device to their service.
22:57.59KranZtest34: whatever they give you will be locked and only usable with their system
22:58.18ManxPowerReally - JUST BUY A SIPURA
22:58.20test34it is free anyways
22:58.35KranZManxPower: but that will only get him to his * box
22:58.37test34I might ask for a sipura for christmas
22:58.43ManxPowertest34, if it's free then it will be locked and you'll have to sign a contract (1 year is common)
22:58.46KranZManxPower: which prolly goes nowhere
22:59.09ManxPowerKranZ, Unless the ATA has an FXO port it's not going to do him any good anyway.
22:59.11test34there  is no contract.. but they might want it back when I discontinue the service
22:59.18KranZtrue
22:59.24*** join/#asterisk Darwin35 (n=Darwin@sta-208-139-193-162.rockynet.com)
22:59.37ManxPowertest34, You won't be able to get any help if you use a device nobody else has.
22:59.57*** join/#asterisk MikeJ__ (n=ircatjer@d14-69-8-30.try.wideopenwest.com)
23:00.55ManxPowerA SIPura is like $60 or so.
23:01.05KranZfyi: brighthouse networks resells roadrunner
23:01.22ManxPowerIf you cant afford $60 then you should be using a SoftPhone (ALL SOFTPHONES SUCK!) and not a hardware device.
23:03.32ManxPowerCisco needs to start putting a USB interface on their routers.  Then you could plug in a flashdrive for the OS
23:03.44test34KranZ, yeah I have roadrunner for internet cable
23:04.27test34they cant tell me what modem/adapter I will get , they say they have many brands/models.......
23:06.05ManxPowertest34, Whatever they send you it won't work with Asterisk.
23:07.10test34ManxPower, ok thanks
23:09.00KranZanyone tested the SPA-941?
23:09.20znoGdoes anyone know how to make asterisk wait about 3 seconds before detecting the distinctive ring pattern?
23:09.29ManxPowerKranZ, several have.  They talk about their experiences on the mailing lists.
23:09.48ManxPower~mailinglist
23:09.50jbotwell, 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
23:10.41KranZim gonna try building an office pbx setup with a mini-itx mobo acting as *box and internet gateway.... looking into ip phones for the testbed
23:11.21ManxPowerHas everyone here read the SECURITY file in the Asterisk source?  If not, you should do so right now.
23:14.30ManxPowerKran I recommend the following for a testbed:  Polycom Soundpoint IP 50x, SIPura SPA-941, SIPura SPA-2100, Digium TDM11B
23:15.05ManxPowerIf you have the money get a TE20xP and an Adtran TA750 or TS850 off of eBay
23:15.07KranZi'd like to try the polycom and spa941
23:15.22KranZi'd build the box to connect to SER
23:15.59ManxPowerThe list, in my opinion, gives you the most exposure to decent products without spending a fortune testing every product out there.
23:16.35ManxPowerWe wasted money on GrandStream, Cisco, Zultsys, etc.
23:18.40drrayproblem is phone tastes are subjective
23:19.02KranZtrue, i have a few spa-2100's deployed and only need t1 interfaces
23:19.10KranZhavent messed with any ip phones tho
23:19.13drrayyou'll not get my ciscos from me
23:19.39KranZcisco is everywhere which makes the hw cheap, but then you get stuck with licensing which sucks ass
23:19.50NuggetI like my ciscos well enough, but I don't think they're worth what they cost.
23:20.12KranZim sure they're sweet when you have em runnin with all the features
23:20.23*** join/#asterisk PakiPenguin (n=pingu@linuxpakistan/admin/pakipenguin)
23:20.24KranZof course, you'll be short an arm and a leg
23:20.41PakiPenguinhello everyone
23:20.44PakiPenguini have a fake ring problem
23:20.45PakiPenguini mean when i call some number , after dialing the number , while the call is still being made ( in progress of being sent to the voip company ) , i get this ring ring which is basically my system generated , i want the tone to start when the call actually starts ringing at the number that i called
23:20.45waddyanyone can recommend a pre-built asterisk system to buy?
23:20.55KranZcut-n-paste
23:21.09KranZwaddy: now why would you go and do that?
23:21.29PakiPenguinthis wasnt in 1.0.9 , i started getting it in 1.2 just now , as i upgraded , can anyone help me?
23:21.31waddyi want a nice web interface, stable and features
23:21.56KranZeveryone wants a nice web interface
23:22.01drraynot me
23:22.10KranZi'd like to get one goin for end users
23:22.13KranZa dashboard
23:22.16waddyyer
23:22.24waddysipX looks ok
23:22.54KranZcheck VMs, change features, check call records...
23:23.24KranZi hate web design
23:23.50drrayproblem is like the phones, a web interface would be subjective as well
23:25.54drrayI've written some simple tools for managing asterisk, mainly for bossman so I can keephim out of vi
23:26.56*** join/#asterisk implicit (n=implicit@ip70-181-114-97.oc.oc.cox.net)
23:27.19implicitanyone have any grandstream equipment on firmware 1.0.7.11?
23:27.20]expicdrray: did you try AMP?
23:27.37drrayI did, and I don't like AMP
23:27.53]expicdrray: rather good interface
23:28.28KranZNugget: for a dashboard?
23:28.32KranZdammit
23:28.35Nuggetnope.
23:28.37KranZnow how will i know
23:28.39KranZoh
23:28.44NuggetI'm happy with vi to manage my asterisk.  :)
23:28.51file[laptop]fatality!
23:29.12drrayI'm happy too, it's just when bossman gets ideas
23:29.18file[laptop]Nugget: do you come with sweet and sour sauce?
23:29.31nvrswell who knows best
23:29.32file[laptop]and yes, I'm running out of stuff to do with your nick
23:29.35nvrsyou or your boss
23:29.35Nuggetnope, honey mustard.
23:29.42nvrsyou got to shut him down
23:30.35KranZthe nugget was $5.50
23:30.52file[laptop]whyfor must you bill me
23:31.01PakiPenguinhmmm.. anyone?
23:31.07*** join/#asterisk tessier (n=treed@rrcs-67-53-110-66.west.biz.rr.com)
23:31.19*** join/#asterisk De_Mon (n=de_mon@fl-69-69-147-198.dyn.sprint-hsd.net)
23:31.49KranZPakiPenguin: i get that sometimes on a sipura on a cvs head from 2 months ago
23:31.53KranZdouble ring
23:33.13PakiPenguinKranZ: it never happened on 1.0.9 stable
23:33.56*** join/#asterisk zemmad (n=root@208.0.230.116)
23:34.40zemmadhow well does asterisk work with eyeBeam??
23:36.49*** join/#asterisk heison (n=heison@CPE000a01d49e6f-CM00111a59bec2.cpe.net.cable.rogers.com)
23:38.02KranZPakiPenguin: it probably did
23:38.38*** join/#asterisk Rez (i=lorez@freenode/staff/lorez)
23:39.52PakiPenguinKranZ: thanks to implicit , if you set
23:40.01PakiPenguinprogressinband=no , it wont happen
23:40.02PakiPenguin:)
23:41.27p1tst0pif it want to turn on one touch recording for inbound calls.. where would i put the Ww options ?
23:42.35brettnemcheck out phonecall to manage asterisk
23:42.39brettnem~phonecall
23:43.36brettnemjbot, phonecall is a web gui for Asterisk management and can be found at http://www.vecsector.com/phonecall
23:43.38jbotbrettnem: okay
23:43.43brettnem~phonecall
23:43.44jboti guess phonecall is a web gui for Asterisk management and can be found at http://www.vecsector.com/phonecall
23:49.42*** join/#asterisk harryvv (n=none@S010600a0c93f6f7e.vs.shawcable.net)
23:49.57*** join/#asterisk zmauve (n=mikael@c-9131e253.57-1-64736c10.cust.bredbandsbolaget.se)
23:50.20zmauveHi, I just got my asterisk system up and running :) it works great!
23:50.32mog_workyay
23:50.56zmauveI will propose we run it at work on monday :)
23:51.22zmauve(we have no telephony solution at work)
23:51.54harryvvyou mean no pbx
23:52.03zmauveyep, we have no pbx
23:53.13zmauvebtw, in my dialplan I now have: [incoming] exten => s,1,Answer() exten => s,n,Playback(demo-congrats) exten => s,n,Hangup()
23:53.16docelm0ya buddy!
23:53.32Corydon-wQwell: please try latest 1.2
23:54.25zmauveI tried to add exten => 10,1,Answer() and more so that I could dial my own phone number with an added 10 to it, but I still always get to the "s" extension; am I doing something impossible?
23:54.36nextimeanyone using mcc on SVN-trunk-r7230 ?
23:54.36harryvvI wonder if newer digium pstn cards have eliminiated that click when the pbx picks up
23:54.57zmauveharryvv, I hear no clicks, and I got my new card yesterday :)
23:55.11harryvvzmauve, what card
23:55.25zmauvetdm400p with on fxs and one fxo port
23:55.36zmauvelspci says TigerJet something
23:55.44harryvvyea popular card. I have the much older x100p card.
23:55.47znoGanyone here use distinctive ring with asterisk?
23:55.57harryvvznog I have
23:56.19zmauveharryvv, you don't happen to know If what I am describing above is impossible?
23:57.01harryvvs,2 s,3 not s,n
23:57.12KranZ6pm, time to go home
23:57.20znoGharryvv: are you able to show me your /etc/asterisk/zapata.conf?
23:57.23KranZbbt
23:58.00znoGharryvv: i just want to check whether I've configured my zapata.conf correctly
23:58.02zmauveharryvv, the asterisk book says you can use s,1 + s,n with asterisk 1.2.x; am I misunderstanding something?
23:59.46harryvvbest is to look at voip-info.org

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