irclog2html for #asterisk on 20050221

00:00.30Pkunk...
00:02.25*** join/#asterisk jdg (~jdg@CA03F87A.adsl.mana.pf)
00:13.04user1fncan asterisk work on a machine with no sound card?
00:13.11Darwin35yes
00:13.13techexpressyes it can
00:13.20user1fnanything special needed?
00:13.25techexpressno
00:13.37user1fnok... just checking
00:13.46Darwin35just put noload => alsa and oss channels in the modules.conf
00:14.49Darwin35or just uncomment them
00:14.58*** join/#asterisk libpcp (libpcp@210.16.20.5)
00:15.17user1fncool... thanks
00:15.40user1fnso has anybody heard of any great strides in getting faxing to work more consitently with *?
00:16.09techexpresscan sombody know about ulaw codec making my outgoing call to telco make a twitchy noise
00:16.29techexpressinternal calls is ok
00:21.35iceypanyone know how to implement cheapest based routing?
00:21.46fileit's called least cost routing
00:21.59fileand yes, I know how to do it
00:22.11iceypyeah, do you know where I can find some documentation?
00:22.29fileabout what specifically? you have to find a least cost routing module for asterisk, or write your own
00:22.35fileand then read it's documentation...
00:22.44iceypwhich one do you use?
00:22.50filecustom.
00:23.02iceypok.
00:24.34iceypis voip-info down?
00:24.50iceypevery page i click on from google for voip-info not showing :/
00:25.24*** join/#asterisk adker (~adker@70-97-138-2.dsl1.glv.ny.frontiernet.net)
00:25.26Silik0ntry the google cache then seeings its prolly down with the problems its been having lately
00:25.49iceypdamn no cache
00:31.41techexpressis there a howto for asterisk?
00:31.54inticonnethehehehe
00:33.37*** join/#asterisk tessier (~treed@wsip-68-224-172-77.sd.sd.cox.net)
00:33.43tessierWhut up bitches
00:33.53Darwin35not much my hoe
00:34.22Darwin35voip-info.org
00:34.42techexpressok thanks ;-)
00:34.51Darwin35its called the wiki pages
00:34.57Silik0nonly problem with that darwin is it aint working tonight
00:35.06Darwin35wow ok
00:35.09Chuji~docs
00:35.11jbotDocumentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org
00:35.11Silik0ndamned things down again
00:35.22Chujiread the asterdocs
00:35.45myconidI wish broadvoice would let me set my own passwd
00:39.03dsmousemyconid: why? what did they choose?
00:39.28*** join/#asterisk Hmm-work (matt@24-119-151-19.cpe.cableone.net)
00:40.11*** join/#asterisk doughecka (~Doug@doughecka.user)
00:40.36dougheckahmm, long birthday too
00:40.53Hmm-workit's my birthday tomorrow
00:40.55doughecka18
00:41.12Hmm-workcongrats on cigarettes and porn
00:41.16dougheckaright
00:41.29Hmm-workif you're in the US
00:41.37dougheckaindeed :)
00:42.29Hmm-workslow in here tonight
00:42.49Hmm-workI was hoping to find someone to pay me ungodly amounts of money for advice
00:43.03dougheckahm
00:43.09dougheckaI would advice not to do that
00:43.28Mavvieasterisk console should have the option of beeping when there are no more calls active.
00:43.38Hmm-worklol
00:44.02dougheckajust restart gracefully :P
00:44.15Hmm-workyate looks interesting
00:44.30Mavviedoughecka: I've done that once, it segfaulted the moment I pressed return.
00:44.34*** join/#asterisk trelane (trelane@adsl-68-78-10-169.dsl.ipltin.ameritech.net)
00:44.38Mavvienever been able to reproduce it.
00:45.08dougheckahahah
00:45.11dougheckame too
00:46.15trelaneanyone here using broadvoice with asterisk? I've been attempting to follow voip-info.org's setup instructions without luck.
00:46.16*** join/#asterisk syslod (~yurplsl@65.114.0.198)
00:46.27syslodHello.
00:47.17*** join/#asterisk SirPrize (~blah@83.146.62.181)
00:48.00NukemizerCan anyone point me in the direction that will help me test why I can get my TE110P to work in e&M mode but not PRI ?
00:48.20SirPrizeHi folks.  Quick question: Can I, from the *Nix command line, cause Asterisk to initiate a call and start executing a certain dialplan sequence ?
00:48.23syslodNukemizer: What probs u having with PRI?
00:48.45syslodSirPrize:  Use callfile
00:48.45SirPrizeSpecifically, I'd like to cause Asterisk to call me, when it is triggered by an e-mail, and then that I'd get an IVR
00:48.47Nukemizersync errors up the wazzo
00:49.01SirPrizesyslod: Brilliant, thanks.  I'll take a look at that
00:49.04Hmm-workSirPrize: that is a serious pain in the ass
00:49.05Nukemizeri would only be enableing 1-8 b channels
00:49.06syslodcheck you span line and see if your timing setup is ok.
00:49.22Hmm-workI wrote some scripts to do it based on form input though
00:49.23syslodNo its easy just generate a callfile.
00:49.31SirPrizeHmm-work: how so?  Btw - SIP incoming DTMF works via different provider now
00:49.42Hmm-workgood deal
00:49.56NukemizerI  should back up.. I am attempting to get the PBX to talk to * via PRI not actually using * PRI to connect to telco
00:50.06Hmm-workcallfiles are easy to use yes.... triggering one based upon receiving an email is a different story
00:50.30syslodHmm_work: umm postfix will do that easy.
00:50.40Hmm-workoh really?
00:50.45MavvieSirPrize: http://megaglobal.net/docs/asterisk/html/nagiosasterisk.html
00:50.50Hmm-worktime to google
00:51.09SirPrizeI'm thinking of using either Procmail or QMail - as long as there's some way that I can trigger Asterisk from a script, I think I shold be able to do something
00:51.15SirPrizecheers, thanks for that Mavvie
00:51.22syslodBoth of those will also do it.
00:51.49Hmm-workI use an agi script to trigger a callback event
00:52.31syslodAnyone else a telco?
00:52.59Hmm-work9 minutes until the simpsons is on... woot!
00:53.07tzangerheh
00:53.18tzangersyslod: werd
00:53.21dsmouseHmm-work: are you streeming it via asterisk?
00:53.28syslodtzanger: Huh?
00:53.32Hmm-workwhat?
00:53.36user1fnin case you guys were interested... the unresolved symbols were a symptom of the sarge kernel and the config file that came with it (they didn't match)
00:53.40Hmm-workno, lol
00:54.05user1fnthanks for all of the help, though!
00:54.28Hmm-workyet another reason to compile your own kernel with the same gcc version you use to compile asterisk
00:56.32SirPrizeMavvie: I see the page you suggested makes use of a mkqcall.pl file.  That isn't present in my asterisk source/install, and a quick google doesn't bring up anything on it either.  Might you know anything about that file ?
00:57.07MavvieSirPrize: euhm. no. it was the closest thing I could give you to something useful.
00:57.48SirPrizeMavvie: that's ok.  Good starting point though
00:58.29iceypis this the easyest way of doing LCR...
00:58.30iceypexten => _X.,3,Dial(IAX2/user:pass@localprovider.co.nz/${EXTEN},60,t)
00:58.30iceypexten => _X.,4,Dial(IAX2/user:pass@secondprovider.co.nz/${EXTEN},60,t)
00:58.40iceypso if option 1 failts it will go to option 2
00:58.52Nukemizersyslod: since I ask about using less than 23 bchannels to see if that might be a sync problem.. I get red and yellow alarms every 30 seconds or so when PRI is connected
00:59.23Nukemizerin between span resets i might be able to place a call so I know I am close
01:00.12tzangermeans hello, long time, how are things.  :-)
01:00.34tzangerNukemizer: if your PRI is going up and down like a bridge's nightie you have other issues
01:00.42tzangerquestion is -- do you have trouble with a loopback plug?
01:00.49tzangerdoes your provider see issues when you loop back the smartjack?
01:01.11Nukemizertanger: yes same trouble with loopback plug
01:01.11tzangerhave you compiled zttool and noted any irregularities there?
01:01.21tzangerNukemizer: problem is on your end then for sure
01:01.25tzangersharing interrupts?
01:01.38tzangeris * running on a renderfarm node?
01:01.53Nukemizerbut when i make that same card become a regular e&M wink circuit it works fine
01:02.03Nukemizerthat is what I do not understand
01:02.14syslodNukemizer: Pastebin your zap files.
01:02.17tzanger"works fine" as in you have clear audio?
01:02.28tzangeryou may just not be SEEING the issues
01:02.32Hmm-workLOL
01:02.36Nukemizeryes , clear audio --- 30 minute call with no drops
01:02.45tzangerhmm
01:02.45Nukemizervery ture
01:02.50tzangergot a shitty version of libpri?
01:02.52Nukemizerone sec pasting
01:02.54syslodK
01:03.02syslodAlso what PBX are u connected to?
01:03.05tzangeror compiled asterisk with an alternative libpri version than you're actually running?
01:03.24Hmm-workis there any other sip testing tool besides sips?
01:03.44Nukemizeri am trying to connect to a Toshiba PBX
01:04.09*** join/#asterisk PBXtech (~upirc@wirelessdata-167-246.mycingular.net)
01:04.12syslodWho is providing timing?
01:04.35Nukemizerthe * should be providing timming
01:04.38Nukemizerto the PBX
01:04.47syslodWhy frac PRI?
01:04.56tzangerNukemizer: so you have span=1,0,0,b8zs,esf kind of thing?
01:05.08tzangerI doubt it's that though
01:05.21*** part/#asterisk PBXtech (~upirc@wirelessdata-167-246.mycingular.net)
01:05.36tzangerif E&M works and PRI don't yo ueither have an invalid/incompatible switchtype or your PBX isn't actually sending what it's saying it is
01:05.59Nukemizercant find my pastebin link to paste to
01:06.08tzangerpastebin.ca
01:07.34Nukemizerhttp://pastebin.ca/6214
01:07.51planet_guruvoip-info appears to be dead.. are there any mirrors around?
01:08.21Nukemizerthe span is currently in e&m mode however
01:08.29SirPrizeWheeeeee!!  I just got callback working! :-)
01:08.31tzangerNukemizer: well first off you're not providing sync, you're syncing to the PBX
01:08.48tzangersync of '1' means whatever's on the other end of this span is my clock sync src
01:08.53Nukemizerfrc t1 because PBX does not have enough license to work with all 24 currently
01:08.57tzangerer rather my primary clock sync src
01:09.13Nukemizerso for me Dchannel=24 and b channels are 1-8
01:09.30tzangerinteresting
01:09.38tzangerare you sure its 1-8 and 24?  not 1-8 and 9?
01:09.46tzangerI have no idea how a frac t1 works with the toshiba pbx
01:09.52tzangerdoes the dhcan come up?
01:10.00Nukemizerthat may just be my problem then , perhps I ned to have all 24 enabled ? 23+1
01:10.16tzangeryou can always tell * to only use 1-8 for calls
01:10.17Nukemizeryes but the D does keep being reset
01:10.27tzangerwhat's pri debug span 1 say
01:10.37Nukemizeror i will get a message that "assuming D channel is 24"
01:10.44*** join/#asterisk puzzled (~patrick@puzzled.xs4all.nl)
01:11.47Nukemizerthe Tosh can be partitioned just like the * but the Tosh is very particular as to what switch it likes to talk to on the PRI
01:12.45Nukemizerdebug just goes nutz, currently not pri so i cn not get deug, but like you said with a loop back plug in, i should get no errors
01:13.20Nukemizerthe only error i get with loop backl in, is that "we think we are primary and they think they are"  over and over..
01:13.39tzangerNukemizer: well you might want to play with the channel settings
01:14.27Nukemizeri have tried moving D channel to 9 and other locations - rebooted dozens of times and moved slots in the box even tried 3 seperate boxes
01:14.55tzangerstop that
01:15.08tzangerit works in E&M, it's not hardware
01:15.18NukemizerHad digium send me a new card this weekend but no luck, so i finally gave up on PRI and tried e&M.. and then I got some desperatly needed success :)
01:15.21*** join/#asterisk posit (~reiko@client-82-2-122-51.brnt.adsl.virgin.net)
01:15.23tzangerthis is a software problem
01:15.34*** join/#asterisk puzzled_ (~patrick@puzzled.xs4all.nl)
01:15.42syslodWhat toshiba u have?
01:15.49NukemizerCTX
01:15.52Pkunkis it possible to increase the gain/volume on only a particular SIP channel ?
01:16.03syslodCTX should support NI so I doubt it incompatiable.
01:16.17tzangerPkunk: not that I'm aware of
01:16.35NukemizerI think i have been trying to get NI2 to work. perhaps that is my problem
01:17.28NukemizerI sure to appriciate both of your imput as I just never seem to see an end to my problem and this helps
01:17.34syslodNI2 will work
01:17.54syslodHave you fixed timing as tzanger suggested?
01:18.31syslodAlso what does zttool actually say when connected?
01:19.06Nukemizerit appears I have no zttol .. looking for now
01:20.46Nukemizernot sure what timming i would change .. do you mean the setting on the span ? span=1,1,0,esf,b8zs  ?
01:21.27tzangerNukemizer: you need libnewt to compile it
01:21.41tzangerspan=1,1,0,esf,b8zs ==> span=1,0,0,esf,b8zs
01:21.44tzangerand rerun ztcfg
01:22.44*** part/#asterisk SirPrize (~blah@83.146.62.181)
01:22.47Nukemizertzanger: yes I have tried both of those and even rebooted and powered off to make sure card was not locked for somereason
01:23.05tzangerthe correct is the latter, with timing set to '0'
01:24.55shido6back
01:25.29Nukemizercurrently i have timming to span=1,1,0,esf,b8zs with e&m config and PBX being the slave. which is my guess that the PRI should be the same way
01:25.48tzangerno
01:25.54Nukemizeri will need to remove the standard T1 card and install the pri and reconfigure the lines though before i change *
01:25.56tzangertiming=1 means the zaptel is gonna be a slave
01:26.11tzangertiming=0 means I am not trying to sync ot this span, I am considered the master clock
01:26.28tzangerit'll likely work but you'll get little buzzes now and again as the frames slip
01:26.32Nukemizerthen i read that wrong the first time..
01:27.31Pkunkwhat are the CPU requirements of g.729 ?
01:27.52Pkunklike on a celeron 850 mhz , how many lines can it support ?
01:27.58tzangertiming is a number... 0 = do not try and sync (i.e. be the master) 1 = the other side is my primary src.  2 = the other side is my secondary sync src (if the 1st is down), 3 = other side is my tertiary sync (if first dow are down), tec.
01:28.40*** join/#asterisk yxa (~void@203.118.40.42)
01:29.18Nukemizertzanger: thanks, have written down
01:29.21Pkunki tried the Intel codec , and while the quality was acceptable the volume was low . Is this going to be a problem with digium's codec too ?
01:32.31Pkunksigh
01:32.43Nukemizertzanger:  syslod: thank you both i have testing to do now..  renewed hope !
01:33.38*** join/#asterisk mrproper_ (~psynode@61.95.55.242)
01:33.53mrproper_hi all does anyone know if you can download the knopsterisk iso anywhere?
01:34.36trelaneanyone here using broadvoice with asterisk? I've been attempting to follow voip-info.org's setup instructions without luck.
01:34.49*** join/#asterisk Sedorox (brandon@Neptune.client.wlmsprt.pa.sed6.net)
01:36.03Pkunktrelane: you need to register
01:36.12Pkunktrelane: only then it will work
01:36.42trelaneright the problem is when I have the registration sytax to what I think is correct, asterisk never attempts to register
01:37.54trelanePkunk, register phonenumber:password@sip.broadvoice.com
01:37.55planet_guruGuys, I'm using the 'read' command in my dialplan.. I just want to read in a 4 digit number. I'm using exten => s,3,Read(PSFILE,4)  but the application dies with Invalid extension '1'..  when '1001' is entered. What's the obvious mistake here please?
01:38.19Pkunktrelane: thats wrong
01:38.37Pkunkread the asterisk docs on broadvoice support
01:38.51trelaneasterisk docs or voip-info docs?
01:38.56*** join/#asterisk qwerp (~abc@219.95.105.74)
01:39.01qwerpharlo..
01:40.19qwerpis there anyway i can pick up an transfered call?
01:41.24Nukemizertzanger: last question - zttool can that only be compiled with a PRI card working ? is that tool only for troubleshooting PRI ?
01:41.38tzangerno
01:41.41tzangerzttool is a zaptel tool
01:41.48tzangerjust shows you goodstuff about your zaptel interfaces
01:42.16tzangermissed interrupts, RBS signaling
01:42.18tzangerthat kind of stuff
01:43.01tzangeruseless for PRI except to show missed interrupts and stuff
01:43.02Nukemizerok.. just wont compile so that might mean there are more problems
01:43.30tzangerno
01:43.38tzangerjust means you don't have libnewt
01:44.02Nukemizeror perhaps it is not new enough
01:44.03tzangerwhich is normal
01:44.04tzangerheh
01:44.05tzangerno
01:44.13Nukemizerjust installed off distro disk .. checking.. thank you
01:50.54qwerpis there anyway i can pick up an transfered call?
01:52.13planet_guruCan somebody please tell me how I use the Read command if the only arguments I want to specify are the 'variable' and the 'maxdigits' in this prototype: Read(variable[|filename][|maxdigits][|option])..  I'm currently trying it with    exten => s,1,Read(USER,4)  but that breaks
01:52.23inticonnetIm finally getting somewhere thanks to harryvv - except when I call my voicemal extension CLI states "Unable to find/create chanel"
01:52.27inticonnetany thoughts?
01:54.44iceypPkunk broadvoice any good?
01:54.54Pkunkrawking
01:55.08iceypand u can signup from anywhere in the world?
01:55.44planet_gurusurely I don't have tp create a minature silent gsm to stick in as a second argument to Read?? There must be a simple solution to this?
01:55.54planet_gurus/tp/to/
01:59.02Pkunkiceyp: well i signed up frm Asia . but you need g.729 for doing anything useful
02:00.01iceypmmm, true, cant use gsm or ilbc?
02:00.06iceypthats what my budgetone does
02:00.27iceypdid you have to purchase any gear?
02:01.16iceypis the quality real bad, or you dont have enough bandwidth to do any better?
02:01.27Pkunki have bandwidth for ulaw
02:01.38Pkunkbut its too expensive in the long run
02:01.46iceypbut they do support it?
02:01.52Pkunkprobly cheaper to buy livences
02:02.00Pkunkyeah ulaw works great
02:02.14iceypthey support gsm or ilbc?
02:02.39Pkunkwhat could be the problem .. if when i press dtmf's very fast on a Zap line i suddenyl get hanged up ?
02:02.51Pkunkiceyp: no
02:03.05iceypdamn, so only 729 and ulaw?
02:03.28inticonnetok i sort of fixed it but i still cant call voicemail from extension 2001 yet there is an entry for it in voicemail.conf. CLI says "unable to find/create chanel"
02:03.35inticonnethelpsies? :)
02:04.10Pkunkdoesn't d/c when i up busycount .. but then it doesn't ever disconnect the phone too
02:04.26inticonnetmm actually i cant save changes to voicemail.conf so there is no entry and i cant change the exisiting one :S
02:05.25iceypwhats ur username pkunk? i'll say your refered me
02:05.30iceypreferred*
02:07.31*** join/#asterisk shmaltz (~chatzilla@69.28.255.210)
02:07.46shmaltzanybody remember the tMobile hack a few weeks ago?
02:07.59shmaltzwell look at this:
02:08.00shmaltzhttp://www.drudgereport.com/flash3ph.htm
02:08.37Pkunk@#$@# .. i get disconnected while typing dtmf's
02:10.05*** join/#asterisk md99 (~root@port-222-152-49-44.fastadsl.net.nz)
02:10.30md99Hi.
02:10.47Pkunkwasn't happening with the 1 year old CVS i had installed before
02:12.58md99Does anyone know of anyone who has NOT had echo problems with AVM Fritz PCI Passive ISDN Cards?
02:14.15*** join/#asterisk harryvv (~comming@S010600055d210201.vs.shawcable.net)
02:14.32Mocwhat up tonight ?
02:15.24Mocmd99, to my knowleage, if you get echo from a ISDN call, it not a card issue, it your provider that has the echo from a analog source
02:16.21harryvvMoc what type of termination points into small pbx boxes use ? isdn or other
02:16.59Mocharryvv, in Euro, I guess they use ISDN, in Canada/US, small pbx use standards Analog line
02:17.01Mocor PRI
02:17.10harryvvokay
02:17.18MocBRI for voice is rare in the Canada/US
02:17.24harryvvyea, talking to a guy in Australia and ISDN is common there
02:17.32Mocyep
02:17.54MocISDN was supose to replace those Analog line we have... sadly the US/Can didnt follow
02:18.42harryvvWithout subscribing to a PRI can I setup some kind of system to simular pri out of a PC or something for testing reasons?
02:18.54harryvvneed cards or equipment of course.
02:19.28Mocwell VoIP is a cheap way of doing stuff similar to PRI
02:19.34harryvvtrue
02:19.53MocI get DID from a local provider
02:20.48harryvvbut then again that depends on a VOIP provider :) I have iax.cc and dont have one but use it for outgoing calls. But will for another costomer that gets alot of calls from canada into washington state. There only line is so bussy.
02:21.41Mocharryvv, btw you can simulate a PRI within * I think
02:22.24Mocyou dont need PRI hardware... I never tryed it, but maybe there is info on the wiki
02:25.25md99thanks moc, my echo problem only occurs from a local sip phone via isdn bri to an analog number. (to a gsm number works perfectly)
02:25.55*** join/#asterisk Legend (~legend@24.244.142.133)
02:25.56*** join/#asterisk yashax (~yasha_x@c-24-98-23-73.atl.client2.attbi.com)
02:27.07inticonnetMan oh man. Im so proud - After the amount of swearing I subjected you guys too last night opposed to today. I know have a functioning internal * box. Now to connect FWD Mwahahahaha
02:27.17inticonnet*now
02:27.49Mocgood ;)
02:28.29JerJergood now go get me a cup of coffee
02:28.51Mochey JerJer how it going ?
02:28.59JerJersnowed in
02:29.19JerJerwe are under a Winter snow warning
02:29.36Mocthat cool
02:29.41JerJerand the county has closed all roads to non-essential travel
02:29.52Mocvery cool..
02:29.57JerJeryet all the wireless links are still rock solid
02:29.59Mocthat mean work from home day!
02:30.10Mochehe
02:30.20Mocsnow doesnt affect wireless link that much
02:30.28Mocrain affect it more from my experience
02:30.53tzangerit's all moisture
02:31.04tzangerwe've got a good snowstorm here
02:31.05tzangerblah
02:31.12tzangerit's the 2nd half of february
02:31.34Mocdidnt had that much snow over here
02:31.48MocI should move south to get more snow these day
02:34.14*** join/#asterisk docelm0 (~brian@66.238.251.141.ptr.us.xo.net)
02:34.59docelm0Anyone know anything about the Cisco 7912G phones configured with NAT?   How is this accomplished?   I have looked at the wiki and got nowhere
02:35.38docelm0can they be configured with STUN?
02:36.00JerJerstun is not the answer
02:36.08JerJerTFTP
02:36.10JerJerall you need
02:37.34docelm0ok TFTP is fine however what do I need to do?   The calls coming into the phone work but calls out to the * box dont work.  I am suspecting its sending the private IP in the SIP setup messages
02:38.01docelm0with the STUN I was getting along the lines of Nat Transversal
02:38.21inticonnetwww.pastebin.ca/6217 can u guys have a look at that for me and tell me why now my internal calls are being passed as bogan calls too?
02:39.06JerJerdocelm0:  regsiter
02:39.07JerJerregister
02:39.24JerJerand nat=yes in the appropriate place(s) in sip.conf
02:39.30JerJerand enable nat processing on the device
02:41.53md99ok - do you think it is possible for the echo cancellation/suppresion code to be integrated with chan_capi?
02:42.05md99being the zaptel echo suppression code
02:42.09docelm0Well the phone register's with * but it cant make calls.   I have NAT set to yes under the context.  I have noticed there is a setting for NAT proxy but there isnt a proxy on the network.  Just a linksys router
02:43.04JerJeryou are setting just the sip proxy NOT outbound proxy, correct?
02:43.37docelm0yes.  but the problem is I am not setting either.  There is no sip proxy on the local network
02:43.39inticonnetargh now my second extension has gone back to playing dead
02:44.50inticonnettwo extensions configured excatly the same with exception to the username. Yet the second one constantly dies for no apparent reasson
02:45.15*** join/#asterisk BrianR___ (brianr@h006067091a61.ne.client2.attbi.com)
02:45.18BrianR___hey folks.
02:45.19docelm0check hardware I had the same problem with a Sipura 2000
02:46.10docelm0So Jer should I set the sip proxy the same as the gateway IP of the router?
02:46.13BrianR___Anyone know if there's plans to add logic to asterisk to allow for REINVITE between two sets behind the same NAT or one NAT'd host and one non-NAT'd host, but to keep the behavior of canreinvite=no for the case of two sets behind different NAT?
02:47.38BrianR___Also, anyone know if it's possible to twiddle codec selection based on ping time to hosts, as a cheap way for automatically picking ulaw/alaw for on-lan calls and gsm or another lossy codec for calls which cross a WAN link?
02:47.55JerJerdocelm0:  it doesn't have to be local
02:47.58JerJerthe sip proxy is your asterisk box
02:48.05posithi, when starting asterisk, I'm told that chan_zap.so fails to load because of undefined symbol: pri_dump_info
02:48.10positdoes anyone know what could be wrong?
02:48.11JerJerput the ip address of your asterisk box in the sip proxy field
02:48.13JerJerof the phone
02:48.52JerJerposit: need updated version of libpri
02:49.05Dhp4When i try running make install, i get this error: "/bin/sh: restorecon: command not found". The command is located in /sbin/restorecon, i am running Fedora 3, any ideas how to get this to work?
02:49.07docelm0ahh ok.  I have never setup a Cisco phone.  I am more a Linksys ATA or Sipura person.  Soyo and grandstream...  I get around.   I do alot of R&D at my company
02:49.13positJerJer: thanks, I'll check that out
02:49.18terrapenyawn
02:49.28terrapenwhat would be fun to write?
02:49.32JerJernot agi
02:49.35terrapenheh
02:49.39goatmilkbesides iaxcomm what's another windows client
02:49.45terrapendiax
02:49.46dsmousewhat exactly is agi anyway?
02:49.50terrapenfirefly
02:49.56terrapen~agi
02:49.57jbotit has been said that agi is the Asterisk Gateway Interface...  similar to CGI for web applications AGI lets you script call control and access databases using your favorite language.  AGI wrappers are available for Python (pyst), Perl (astperl?) and other languages
02:49.59JerJerAgi is to Asterisk as CGI is to Apache
02:50.01docelm0I have coded a ton of PHPAGI
02:50.24dsmouseah
02:50.40terrapeni gotta find something fun to do
02:50.43JerJerBrianR___:  you could write an asterisk C language application to do that test and set a codec
02:51.15BrianR___JerJer: Interesting.. Now what if I want the calls to reinvite so they go phone-to-phone instead of phone->asterisk->phone? :)
02:51.29JerJerthe trouble is pulling something like that off is going to be tough
02:51.39inticonnet"Unable to create/find Chanel" - What sort of chanel would it be refering too?
02:51.45BrianR___JerJer: Do you know by chance if the codec negotiation for the reinvite is done by asterisk or the phones themselves?
02:51.47JerJerchannel not loaded?
02:52.00JerJertype=peer not registered, in the case of ip based channels
02:52.05PTG123BrianR___: asterisk
02:52.20terrapenmaybe i could work on my m0n0bsd mods
02:52.26inticonnetjerjer- chanel not loaded- was that for me?
02:52.29docelm0ok Jer I will mess with it tomorr
02:52.30docelm0ow
02:52.39MocJerJer, btw, I got my Unistim channel driver to actually work correctly hehe
02:52.42docelm0and see what I can do ..  Problem is my test network isnt the best.
02:52.58BrianR___PTG123: Aah.. So it would be doable then. Perhaps I'll bang out the code for it at some point.
02:53.04Mocit just need alot of cleanup, and someone to make the phone structural design, but basic call is working
02:53.26PTG123BrianR___: What are you trying to do?
02:53.34PTG123BrianR___: just came in on tail end of conversation
02:53.39terrapenanyone ever used an 1A2 phone systme?
02:53.40*** join/#asterisk brenda (~nnnnn@c-67-182-205-227.client.comcast.net)
02:53.44terrapeni think that's what they are called
02:53.49terrapenthey are hella old
02:53.51BrianR___with monitor turned on, asterisk already has the ping time to a given host..
02:54.08inticonnetI like old systems :)
02:54.21BrianR___PTG123: trying to cause asterisk to do some semi-intelligent automatic codec selection for clients by figuring out if they're on the same LAN or if there's a WAN path involved.
02:54.23terrapeninti, im wanting one of these 1A2 phones for my home
02:54.28terrapenand i want to interface it with Asterisk
02:54.33terrapenprobably using an IAXy
02:54.34inticonnetwtf :P
02:54.42terrapenbut im confused about what the 1A2 connects to.
02:54.47terrapeninti, i know, its crazy
02:54.48inticonnetWhy tho?
02:55.02terrapenbecause the phones are sweet
02:55.06PTG123BrianR___: ah, well here is a problem with going phone to phone on same lan.  Most firewalls don't react well when you try and access an internal device using the external ip from another internal device
02:55.27PTG123BrianR___: asterisk will however automatically try and go phone to phone if it can..  its pretty intelligent
02:55.28inticonnetI have heaps of old telstra s240's (Siemens Rebranded) - Oh how I love their little clicking noises :)
02:55.42BrianR___PTG123: Ie, pick alaw/ulaw for clients on the same lan as eachother or the asterisk box, pick gsm or another lossy codec for calls which cross the WAN.
02:56.00BrianR___PTG123: Does reinvite currently disqualify NAT'd hosts?
02:56.34inticonnetI have an alcatel 4400 which is supposidly an absloute beast but the HDD on the CPU is dead and trying to build a new image for it is sposidly going to cost me 1k upwards
02:56.56terrapeninti: this is what i want to use:
02:56.56terrapenhttp://home.att.net/~wd0giv/comercialphones.html
02:57.04terrapenlook at those bad boys
02:57.08terrapenone uses punch cards!
02:57.09BrianR___Perhaps I'll spend some time tomorrow and read the source..
02:57.11terrapenhow fucking cool is that!
02:57.31terrapenthis will be my new desk phone at home:
02:57.31terrapenhttp://home.att.net/~wd0giv/Phones/1466b.JPG
02:57.33PTG123BrianR___: no it will do a reinvite via NAT, and in theory it should work.   Ah you want to change the codec if its going outside the network, or keep it at ulaw if inside?
02:57.45terrapenor maybe this:   http://home.att.net/~wd0giv/Phones/rack1a2.jpg
02:57.59JerJerI see a WiAXy
02:58.01terrapentell me that phone doesn't rule :)
02:58.02inticonnetYou need like a rack cabinet just to have a few key stations :)
02:58.03BrianR___PTG123: Yep. If the calls traverses the WAN, I want it to do alaw/ulaw :)
02:58.07BrianR___err...
02:58.23terrapeninti, is that what they have to interface to?
02:58.26BrianR___PTG123: Yep. If the calls traverses the WAN, I want it to do GSM or something. I want it to do ulaw/alaw if it's inside.
02:58.34PTG123BrianR___: afraid of cpu usage on local hosts, why not just always use g829 quality is the same, bw usage is non existent
02:58.43PTG123g729 that is
02:58.43inticonnetI got no idea what they interface with
02:59.06terrapenthey have amphenol connectors
02:59.17terrapennot sure if they work as regular old POTS
02:59.18inticonnetI would assume your going to have problems :)
02:59.22JerJerBrianR___: there is a way to set the codec using a channel variable
02:59.25terrapen(excuse the redundancy)
02:59.38terrapeninti, these systems were pretty solid
02:59.51inticonnetId say so were the desks they sat on :P
02:59.53JerJerwhy not make a macro that sets ulaw for station-to-station calls and g.729 for egress calls
02:59.56BrianR___JerJer: Tell me more about how I might use this for biasing the codec based on whether or not it crosses a WAN pipe?
03:00.19terrapenjerjer, you ever play with 1A2 phones?
03:00.24*** join/#asterisk Sedorox (brandon@Neptune.client.wlmsprt.pa.sed6.net)
03:00.25BrianR___JerJer: Well.. I want to deal with the case where an employee has taken home a voip set.
03:00.33PTG123JerJer: ya that would work actually, it knows after all if its a local extension
03:00.50BrianR___obviously they won't have their own asterisk at home.
03:01.07inticonnetI need some fud. Bbl my * loving chatters
03:02.02terrapendamn, its hard to find info on really old phone tech
03:02.07BrianR___also, the case of a temporary field office, perhaps with a handful of phones on the same lan but no asterisk box.
03:02.57terrapenlook at this bad boy
03:03.00terrapenhttp://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&category=11908&item=5752813003&rd=1&ssPageName=WDVW
03:03.14BrianR___JerJer: If I enable g.729 but don't have a g.729 codec on the asterisk box, will it do the right thing and pick another codec when users try to call voicemail or avr stuff?
03:03.57PkunkBrianR___: no
03:04.04PkunkBrianR___: it will spamflood
03:04.10Pkunktelling invalid codec
03:04.22BrianR___D'oh. I wonder if that would be hard for me to fix... :(
03:04.28*** join/#asterisk Varanger (~salmenara@201.240.147.103)
03:04.31Varangerhi ppl
03:04.33Pkunkdisable g.729 in asterisk
03:04.50Pkunkas long as it isn't in asterisks allow list you're fine
03:04.54VarangerI need a single FXO card... which one do you advise me?
03:05.18PkunkVaranger: get a tdm400P with only one fxo slot if you want something you can expand
03:05.22BrianR___With g.729 disabled in asterisk, can a call between two g.729 capable sets with reinvite use g.729?
03:06.17BrianR___I got a bunch of X100P clones on ebay for $10/ea for my initial asterisk testing. Hoping to pick up a wildcard quad pri if things work out.
03:06.28inticonnetterr- that ITT501 on ebay is keen :D
03:06.38terrapeninti, i wish i knew how 1A2 works
03:06.59terrapenlike, do i hook the phone to that ITT box and connect the ITT box to POTS?
03:07.00inticonnetYou could just buy some stuff and hope for the best
03:07.04_Vilesssdflsdjkh
03:07.13terrapenits hard as fuck to find info on 1A2 :P
03:07.14BrianR___trying to integrate with an old norstar box where we've outgrown the number of stations allowed.
03:07.19VarangerPkuk: How much is this TDM400p?
03:07.25VarangerPkunk
03:08.10_Vilego channel bank, t100P call it a day
03:08.13_Vilestop complaining
03:08.14BrianR___The TDM400p is like  $350 with all four ports configured.
03:08.18terrapeni will mount this on the side of my desk, heh
03:08.19terrapenhttp://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&category=58339&item=5751732807&rd=1
03:08.43_Vileuse a newbridge 3624
03:08.46inticonnetOh god.....Thats just sad :)
03:08.47_Vileebay is cheap
03:08.51Varangeroops... I just need a single port FXO card
03:08.57terrapeninti, is that not the coolest thing ever...
03:08.58Varangerwhich one is good?
03:09.07_Vilenewbridge, adit
03:09.17_Vileadit 600
03:09.19BrianR___Actually, between $300 and $330 at the asterisk store depending on our combination of fxo/fxs..
03:09.21inticonnetIts hard to imagine people actually using it :)
03:09.35_VileI would never bother with the TDM cards
03:09.37BrianR___dsmouse: Heh. I bought a ton of those. $11/ea including shipping. Wastes a lot of pci cards though.
03:09.42_Vilenever have, never will
03:09.51BrianR___s/cards/slots/
03:10.00BrianR___good enough for my test setup though.
03:10.01*** join/#asterisk sysdef (~sysdef@pD9561D9F.dip.t-dialin.net)
03:10.04terrapenim gonna have to spend about 400$ just so i can have an old early 80s phone system that works with Asterisk
03:10.08terrapenbut dammit, its worth it
03:10.11_VileI played with the X100P's, and had so many echo issues, I said fuck it
03:10.35inticonnetTerr- U could just buy a huge metal case and thro asterisks box in there, then glue a few jeyboard keys to ur desk :P
03:10.41dsmouseBrianR___: I need some xfs cards soon anyway :(
03:11.14terrapeni'll build a rack for the 1A2 key system in the closet
03:11.14_VileI'm building 3x Dell 1850 w/ two TE410Ps each
03:11.16inticonnetNot to mention strapping hundreeds of kron panels to the back of the metal box :P
03:11.20terrapenmount some IAXys underneath it
03:11.26inticonnetYeah
03:11.43inticonnetIt will be keen to hear the story when you have done it
03:11.54terrapendunno if i want to runthat huge cable all over the apartment
03:12.07terrapenmabe there is a way to rewire the cables to work with cat5
03:12.27inticonnetdepends on the phone
03:12.46terrapenit looks like it is 10-conductor cable
03:12.50terrapenhttp://i20.ebayimg.com/01/i/03/6a/52/94_1_b.JPG
03:13.01terrapenbut im not totally sure
03:13.12Varangercan I use Asterisk with a modem?
03:13.19inticonnetThats keen :)
03:13.24inticonnetWhat do you mean with a modem?
03:13.39terrapenif it really uses amphenol plugs, i can find the proper plugs and wire them myself
03:13.41Varangerthose we used to use to connect to the Internet
03:13.48Varangerbefore broadband and cable
03:14.04SedoroxVaranger: there is only a certain intel chipset that works
03:14.13Varangerwhich is?
03:14.24inticonnetPlus latencey would be huge?!
03:14.24Sedoroxh/o
03:14.25bjohnsonlisted on the wiki
03:14.37VoIPMastaWhere can I find the Asterisk::AGI pm?
03:14.37dsmouseVaranger: do you mean to have asterisk to use the modem to control a phone line or use asterisk to route phone service to a modem?
03:14.39bjohnson~docs
03:14.40jbotDocumentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org
03:14.53BrianR___Varanger: Do you mean placing a call over an asterisk provided fxs channel?
03:15.13terrapenhttp://www.sundance-communications.com/forum/Forum1/HTML/000146.html
03:15.15SedoroxFeb 20 00:36:48 <neopher> You have to be careful which specific 537 chipset you get. Intel has a list at [http://www.intel.com/design/modems/linecard.htm]. The 537PU (or 537PG) is good, with a
03:15.15SedoroxMD3200 controller. The 537EPU or 537EPG is no good, as it has an unsup
03:15.25Varangerfor instance, someone calling through the telephone network
03:15.49inticonnetHrmmmm
03:15.49VarangerAsterisk answering and I would speak through my PocketPC
03:15.54Varangerusing xten
03:15.58Varangersomething like this
03:16.41Sedoroxjbot: modem is  Only Certain Intel 537 Chipsets work, mainly 537PU (or 537PG) with a MD3200 Controller, However, the 537EPU and 537EPG will _Not_ Work
03:16.42jbot...but modem is already something else...
03:16.45BrianR___Varanger: There's modem emulation you can use for calls coming into the asterisk box... Will work about as well as a data call through a conventional PBX unless there's jitter or any non ulaw/alaw audio paths in which case it won't work at all.
03:16.49Sedoroxjbot: modem
03:16.50jbot[modem] (Modulator/Demodulator) A device to turn digital signals to analog ones and back again, so they can be transmitted and translated back to digital at another modem without loss. Used for communication through means of audio, telephone, CB, etc.  Random disconnects? S10=255 sure to do the trick!
03:17.01Sedoroxjbot: modem? is  Only Certain Intel 537 Chipsets work, mainly 537PU (or 537PG) with a MD3200 Controller, However, the 537EPU and 537EPG will _Not_ Work
03:17.02jbot...but modem is already something else...
03:17.06Sedoroxhmmm
03:17.12BrianR___Varanger: It's popular for making asterisk based fax servers...
03:17.23Sedoroxjbot: inet-modem is  Only Certain Intel 537 Chipsets work, mainly 537PU (or 537PG) with a MD3200 Controller, However, the 537EPU and 537EPG will _Not_ Work
03:17.24jbotSedorox: okay
03:17.47inticonnetMy second extension wont work at all any more guys :S
03:18.04BrianR___Varanger: Also, all of the really fast modem moulations are patent encumbered. But that won't matter much anyway since you're unlikely to get high speed connects anyway.
03:18.26inticonnetI might recreate its details in sip as a different extension number
03:20.36VoIPMastaWhere can I find the Asterisk::AGI pm?
03:21.31mikegrbhttp://www.fuckinggoogleit.com/
03:21.51inticonneti call 2001 and get the operator saying extension 2001 is unavliable :S
03:22.23*** join/#asterisk syslod (~yurplsl@65.114.0.198)
03:22.38Sedoroxinticonnet: look on the console to see what is going on
03:22.44inticonnetHeheh silly me :P I forgot to change the number after U
03:23.00*** join/#asterisk MichaelVanD (~MichaelVa@CPE-24-208-88-245.neb.rr.com)
03:23.41inticonnetmm somethings broke :)
03:28.08*** join/#asterisk sandnigg0r (~trilluser@66-55-197-254.gwi.net)
03:28.19Varangerbye
03:28.24*** part/#asterisk Varanger (~salmenara@201.240.147.103)
03:28.38inticonnetIm still getting "unable to find/create channel"
03:29.12inticonnetonly when I call out of extension 2002, but I can call into it fine?!
03:29.28Sedoroxhuh?
03:30.02dsmouseinticonnet: I've had that before... If I remember correctly, it was a network issue... like nat and stuff
03:30.09dsmousecan't quite remember tho
03:30.43*** join/#asterisk j_vianna (vianna@node-40247a6a.ewr.onnet.us.uu.net)
03:30.56inticonnetWell I try to call 2000 from sip client 2002, CLI states - "Unable to find/create channel". But I can call Sip Client 2002 from sip client 2000
03:31.04jsolaresinticonnet: with every extension you try to call out, or just one?
03:31.05inticonnetwhich makes me think it cant be a nat problem cause its internal only
03:31.10inticonnetjust one
03:31.26jsolaresmaybe then exten => 2000,x,... is bad
03:31.41inticonnetIve tried changing the ext number. Still does it
03:31.42jsolaresare both sip clients in the same context in sip.conf ?
03:31.58jsolaresare both exten's in the same context?
03:32.11jsolaresif those are a yes, then i have no idea
03:32.42inticonnethttp://pastebin.ca/6217
03:32.53inticonnethave a look see for urself
03:33.02j_viannaHi gurus! I have 2 asterisk running as "type=friend". Now when I receive a call in one part I want to send this call to the second asterisk box. Should I just use the dial(IAX...) command or I have a command to send this call to another asterisk box ?
03:33.23Sedoroxwhat phone are you dialing from (the SIP user) and what are you dialing to? (sip user?)
03:34.01inticonnetany thoughts jsolar
03:34.12Sedoroxj_vianna: I just dial the other box...
03:34.18*** join/#asterisk SirPrize (~blah@83.146.62.181)
03:34.19jsolareshavent looked yet, let me see
03:34.23inticonnetlol ok
03:34.53j_viannaSedorox: Thanks, I thought asterisk have a way to send the call not dialing...
03:35.06SirPrizeHow would I go about setting up a user who would be contactible as username@mysipdomain.com ?
03:35.19j_viannaSedorox: Like a softswitch...
03:35.37jsolaresi dont see anything wrong for calling one sip to the other
03:35.41SedoroxI dunno about that.. I know when the other box picks it up it bridges it
03:35.56SirPrizeI've read up some about DNS SRV entries - am I on the right track ?
03:36.00j_viannaIt's not SIP, the call is IAX.
03:36.28Sedoroxj_vianna: yea.. it just bridges the calls...
03:37.13j_viannaSedorox: When I dial the call, I just transfered the call or it still consuming my bandwidth ?
03:37.42Sedoroxno.. I think it still passes through the box.. using bandwidth.. haven't really looked at that yet..
03:38.09j_viannaSedorox: Thanks...
03:38.15Sedoroxyup
03:38.30jsolaresif the phones are iax, i think it tries to have both parties talk to themselves after the first server dialed out the other
03:38.39j_viannaAnyone using colocation in telx ???
03:38.53jsolaresatleast with sip if you dont have canreinvite=no on the phones, it tells both phones to connect to each other
03:39.02jsolaresnot sure what happens with iax
03:39.18SirPrizeAm guessing that setting up a usrename@mydomain.com account would mean setting up a DNS SRV entry pointing to the SIP proxy for the mydomain, and that that points at my Asterisk server?
03:39.49jsolaresand if the call originated with a sip phone and then goes out with iax towards the other server... i have no idea what happens then :|
03:40.03j_viannahsolares: I see... when you have canreinvite the first box still bridging the call... I see...
03:40.39jsolaresif you have canreinvite=no
03:41.21jsolaresi had to set that up with a phone behind nat, since i had two phones behind nat, they couldnt connect to each other, so i had to set it up to canreinvite=no on both, so the call HAD to go thru the asterisk server wich wasnt behind nat
03:42.33j_viannajsolares: have you tried to configue the ports in your router manualy ?
03:42.53jsolareswell both phones were behind a nat that i didnt have access to configuring the router
03:43.07jsolaresso no
03:44.24jsolaresmishehu: there's even an ISP wich is also the biggest telco that seems to block sip ports on their routers, had to put an IAXy there
03:44.34jsolaresbah, this nick completor sucks
03:44.36jsolaressorry mishehu
03:45.12inticonnet"Request to schedule in the past" - Constant message in my CLI. Has something to do with music on hold
03:45.18inticonnetany help with that one?
03:45.27brc_~seen kpfleming
03:45.29jbotkpfleming <~chatzilla@ip68-3-230-141.ph.ph.cox.net> was last seen on IRC in channel #asterisk, 6d 7h 18m 8s ago, saying: 'anybody here done a firmware upgrade on a Snom 200?'.
03:45.30stepcutinticonnet: linux ?
03:45.47inticonnetyeah rh 8
03:45.59stepcutinticonnet: are you using ztdummy ?
03:46.03inticonnetnup
03:46.21SirPrizeSedorox: any comments on DNS SRV entries ?
03:46.49Sedoroxdunno.. still looking it up.. what do you need? maybe I'll come across it
03:47.13inticonnetU no how2 fix it stepcut?
03:47.23*** join/#asterisk aminorex (~tony@c66.191.69.132.dul.mn.charter.com)
03:47.32SirPrizeWell, I'm wondering how to actually set up a SIP address of the form username@somedomain.com - I think this needs to be done via DNS SRV entries, but am not sure yet
03:47.32stepcutinticonnet: nope. But I am pretty sure it is related to music on hold
03:47.35*** join/#asterisk mrproper_ (~psynode@61.95.55.242)
03:47.40*** join/#asterisk ManxPower (~eric@dsl-208-164-150-160.datasync.com)
03:47.42inticonnetYeah
03:47.42ManxPower~docs
03:47.43jbotDocumentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org
03:47.55mrproper_does anyone know if its possible to push sip video over the h323 gateway?
03:48.20JerJernot today
03:49.12inticonnetThis whole bogan caller thing is a pia....its time to chop it
03:50.22SirPrizeonly problem is, my Domain Name registrar (Enom) doesn't allow entries of type SRV into one's DNS. :-(
03:51.09SedoroxSilik0n: then get another dns provider.. or you SOL
03:51.10docelm0What does the DNS entry SRV do in the nameserver?
03:51.12Sedoroxyou're*
03:51.18Silik0n?
03:51.36Sedoroxdocelm0: apparently to do
03:51.41Sedoroxbrandon@smart-serv.net to call me
03:51.55Sedoroxinstead of IAX2/guest@smart-serv.net/2000
03:51.57Silik0nok that wasnt for me
03:51.59Sedoroxot etc..
03:52.03SedoroxSilik0n: no.. sorry
03:52.06Sedoroxthat was for SirPrize
03:52.07Silik0nhah
03:52.08SirPrizeby what I've understood, SIP in general uses DNS SRV entries to find the SIP proxy for a given domain
03:52.09docelm0Sed do what?
03:52.09Sedoroxstupid tab
03:52.14Silik0nheh
03:52.30SirPrizeSedorox: shall I call you ?
03:52.32Sedoroxdocelm0: where in your softphone you can dial <user>@domain.comf
03:52.33Silik0nits not there arent a ton of DNS providers out there
03:52.41Sedoroxlol.. that isn't setup yet. but eh
03:52.42SirPrizeSedorox: shall I call you ? - is that what you meant ?
03:52.52docelm0ahh. . I dont use soft phones but ok
03:53.04ManxPowerI don't suppose anyone knows how to solve this error: Feb 20 18:28:19 WARNING[5070]: chan_zap.c:1313 zt_set_hook: zt hook failed: Device or resource busy
03:53.06SedoroxSirPrize: no.. I don't have anything setup yet...
03:53.50Silik0nthis copy needs to hurry up and complete so I can load the car heh
03:53.59SirPrizeAsterisk already DOES do SRV lookups, if I understood correctly
03:54.09Sedoroxyea
03:54.36SirPrizemmmmm...... looks like I might end up hosting my own DNS too.... mmmmm....... not something I'd prefer to do. :-(
03:54.54*** join/#asterisk anto9us (~chatzilla@cpc3-ptal1-5-1-cust123.swan.cable.ntl.com)
03:55.00inticonnetWhy?! Its not that dificult
03:55.31syslodbind is easy
03:55.35SirPrizeTrue - but with my current provider, I get geo-distributed DNS servers which host my entries.  If I host it myself, it'll just be coming from one single machine
03:55.46SirPrizeand if my machine goes down, my DNS and MX entries go down with it
03:55.50Sedoroxyea.. but if your box goes down.. then you don't have a domainname while its down
03:55.59SirPrizeyup
03:56.08syslodIs a domain without a connection any good?
03:56.30syslodAnyone here know about EMI, SECABS, BAF/AMI?
03:56.34SirPrizethe MX entries are still useful while your server is dead - it can reroute your mail to backup servers, for example
03:56.45SirPrizemakes sure you don't lose your mail while your server is dead
03:56.46inticonnetOk Im getting pissed off, Now when I call sip:myexternalip the free world dialup client im calling from starts ringing. Why are none of the calls going to asterisks even tho I have statically assigned the ports in my router!
03:59.03inticonnetWhat ports should I be forwarding?
03:59.31mrproper_does anyone know if its possible to push sip video over the h323 gateway?
03:59.33file5060, 10000-20000
03:59.38fileUDP.
04:00.09VoIPMastaHas anyone here used astcc?
04:00.18anto9usHi everyone, I'm looking configuring up to 10 voip computer terminals on a 2Mb adsl line using very old (500 mhz) workstations and 2Ghz/1GB linux box, am I in the right place to get advice on it and if so, does it sound feasible?
04:01.04VoIPMastaanto9us: it's doable as long as you are really getting 2mbps out of your adsl line
04:01.10inticonnetFile: Udp + TCP or Just UDP?
04:01.17fileUDP, just UDP
04:01.22inticonnetThx
04:01.32anto9usVoIPMasta: 2mbps upload?
04:01.49anto9usI don't think it has that
04:02.21VoIPMastaanto9us: then you first have to check your upload capacity
04:02.39VoIPMastaanto9us: and it also depends on which codec you're using
04:02.40anto9uswill do
04:03.26VoIPMastaanto9us: if you have ~512kbps upload then you would have to use GSM or some other narrowband codec
04:03.26bjohnsonSirPrize: get a secondary dns server
04:03.26bjohnsonSedorox: ^^
04:03.53Sedoroxeh? it was SirPrize we were talking about
04:04.05SirPrizebjohnson: and point a subdomain at the secondary DNS server, you mean?  Unfortunately, I can't even enter new NS records :(
04:04.19anto9usVoIPMasta: does asterisk support that codec out of the box?
04:05.19JerJeryes
04:05.22JerJeryes it does
04:05.29*** part/#asterisk |neuro| (~neuro_[ru@212.176.51.231)
04:05.37inticonnetOk next problem (The list has 6 items on it now)- When I call sip:192.168.5.26 (* IP) CLI dosnt even acknowledge the call and it just times out.
04:05.47VoIPMastaanto9us: yes, asterisk supports GSM and iLBC as free narrowband codecs, G.729 as a commercial one
04:05.52inticonnetI thought CLI should recognise it even if my extensions are wrong no?
04:06.12*** join/#asterisk soundguy (~soundguy@zeus.soundguy.id.au)
04:06.13VoIPMastainticonnet: you should call user/ext@ip
04:06.42bjohnsonVoIPMasta: where is that wiki page you were working on?
04:06.48inticonnetbut in the real world people just go sip:extip ?
04:07.00SirPrizeMmmm...... www.voip-info.org seems to be yoyoing up and down this entire weekend
04:07.07*** join/#asterisk |neuro| (~|neuro|@212.176.51.231)
04:08.00inticonnetI think I might go back to my NEC Xen and tell my boss we are never getting Asterisk.
04:09.20VoIPMastadoes anyone know where can I find docs for astcc?
04:09.31ManxPowerinticonnet, Asterisk is not a SIP proxy.  Asterisk is a PBX.  Users don't dial by IP, they dial extensions and Asterisk figues out the rest.
04:10.30anto9usVoIPMasta: I have 250kbps upload speed, will any of the codecs support 10 lines on that bandwidth?
04:12.18Hmm-workanyone using iconnecthere? i'm looking for a cheap backup DID
04:12.39Hmm-work16 bucks a month for 800 minutes
04:13.10inticonnetok then well 2000/2000@192.168.5.26 dosnt do anything either...in fact fwd client tells me it aint a real address
04:13.53SirPrizeinticonnet: try sip:2000/2000@192.168.5.26 as the address
04:14.47JerJeris there some other command than 'init keys' to load new rsa keys?
04:14.52inticonnetstill nothing from cli
04:15.20VoIPMastaanto9us: nope
04:15.26ManxPowerYou never ever have a : in a dial statement.
04:15.41VoIPMastaanto9us: maybe it's doable but with "robotic" voice
04:15.49JerJerManxPower:  what about specifying a port?
04:15.50ManxPowerDial(SIP/2000@192.168.5.26)
04:15.54JerJerie not using a type=peer
04:16.04ManxPowerJerJer, Yes, then you could use a :.
04:16.37ManxPowerThe Wiki and the Asterisk mailing list archives have 2.4 billion sample Dial lines for SIP.
04:16.53ManxPowerOK. maybe a few less than that, but they still have a lot.
04:17.20anto9usVoIPMasta: No, it's a telemarketing application, need to pitch a sale and record it
04:17.44Sedoroxdamn
04:18.25ManxPowerJerJer, Any ideas on this problem: http://lists.digium.com/pipermail/asterisk-users/2005-February/090578.html
04:18.30anto9ushow many lines would 250kbps cope with?
04:18.44inticonnetManxPower- No go. Told me it was an invalid string So i changed it to sip:2000@ip which tried to call but nothing in cli
04:18.51ManxPoweranto9us, What codec?
04:19.12ManxPowerinticonnet, Well sip:2000@ip will never work in an Asterisk Dial(... command.
04:19.28anto9usManxPower: whichever a novice like me could set up and on a very tight budget
04:19.28*** part/#asterisk SirPrize (~blah@83.146.62.181)
04:19.46ManxPoweranto9us, Without knowing the codec we can't know how many calls.
04:20.16ManxPowerinticonnet, Does the remote side require a password?
04:20.33ManxPowerinticonnet, Are you calling a SIP phone or a SIP service provider?
04:20.43_VileManx, you fucking moron, tell him the right answer
04:20.48VoIPMastaanto9us: Maybe 6 simultaneous calls using GSM, considering that you won't be using your ADSL for anything else other than VoIP
04:20.55anto9usManxPower: the codec would be dictated by the chosen service provider I take it?
04:21.17ManxPoweranto9us, Most service providers support a couple of codecs.
04:21.17VoIPMastaanto9us: wrong, you choose the codec, regardless of the termination provider
04:21.34ManxPower_Vile, I'm just not in the mood to spend the 30 mins extracting the required information from him.
04:22.12ManxPowerVoicePulse, That's a load of horse shit.  You can only use the codecs that your provider supports.  Most providers support 2 or 3 codecs.
04:22.13anto9usCould anyone advise me of a good termination provider for making calls in the UK?
04:22.17VoIPMastaanto9us: You could squeeze 7 and maybe 8 calls but that would be too risky
04:22.26_Vilemanx, tell'em to read the docs then
04:22.35ManxPower~docs
04:22.36jbotDocumentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org
04:22.43_Viledone, have a good day
04:22.44anto9usVoIPMasta: that would possibly be enough, the Adsl has a voice line too
04:23.00*** join/#asterisk agave (phanop@216.81.43.75)
04:23.11Hmm-workno one is using iconnect huh?
04:23.18inticonnetmanx- Im using a sip client to call asterisks - trying to call via my internal IP.
04:23.35ManxPowerinticonnet, Oh!  I can't help you then.  All my SIP clients dial by number.
04:23.52ManxPowerI assumed you had an ASTERISK question.
04:24.01inticonnetI do - Im trying to call asterisks
04:24.10mishehuinticonnet: what is "call asterisk"
04:24.11mishehu?
04:24.19inticonnetArgh Dont worry.
04:24.23ManxPowerinticonnet, Um, if the call isn't even getting out of your SIP client, then it's not an asterisk issue.
04:24.24JerJeranyone know why gcc 2.96 does like like this line struct ast_ivr_option options[];  inside of another struct
04:24.26inticonnetIve confused myself
04:24.38JerJerinclude/asterisk/app.h:62: array size missing in `options'
04:24.43ManxPowerinticonnet, Use X-lite.
04:24.59*** join/#asterisk |neuro| (~|neuro|@212.176.51.231)
04:25.16mishehuugh.  I'm so sick of people telling me to look them up on skype.
04:25.22mishehuskype shyte.
04:25.50_Vilethen stfu and look it up on skype.
04:26.23ManxPowerinticonnet, Really, seriously, check the docs for your SIP client, or dial by number, not URL.
04:26.59_Vileinti, check the docs, at least read them for a couple of hours before coming here and bothering people
04:27.02*** join/#asterisk SeaForth (~SeaForth@c-24-1-126-202.client.comcast.net)
04:27.13_Vileif you don't do that, people know
04:27.15inticonnetArgh - Ok I will explain my situation to any1 whos willing to help- I have an astrisks Server which I can call internal extensions on using Xlite. I have now tried calling inward from FWD but Nothing answers. However for some Farked up reasson at times the client im calling from (Sjphone) answers its own call
04:27.15Sedorox~firefly
04:27.17jbotfirefly is, like, http://virbiage.com/firefly/download/firefly-thirdparty.exe
04:27.27mishehuyes, if you don't read docs or join mailing lists, ManxPower will forever hate you.  ;-)
04:27.28_Vileand people will be more hesitant to help
04:28.03_Vilemish, don't get yourself kicked :)
04:28.12mishehu_Vile: by whom?  ;-)
04:28.19_Vileby you
04:28.38mishehuI am not a masochist and wouldn't kick myself
04:28.51inticonnetArgh *4 the second time today Nick kicks the crap out of asterisks*
04:28.51ManxPowerinticonnet, Set up an exten => in Asterisk something like exten => 8NXXNXXXXXX,1,Dial(SIP/fwduser:fwdpass@fwdipaddress/${EXTEN:1})
04:28.57mishehu~theanswer inticonnet
04:29.00jbotinticonnet: 42
04:29.20_Vile42 is the answer to life and everything
04:29.27inticonnetgive me a sec
04:29.38ManxPowerThen dial via FWD by prepending 8 to the number.  Come to think of it I have no idea what NXXNXXXXXX would be since I don't know the length of FWD numbers.
04:29.45mishehuit is the ultimate answer of Life, The Universe, and Everything
04:29.47ManxPowerHow long are FWD numbers?
04:30.00Hmm-worknip/tuck is getting a little freaky
04:30.05inticonnet617504
04:30.09mishehuManxPower: I think fwd numbers are 6 digits
04:30.16inticonnetthey are
04:30.28ManxPowerinticonnet, Set up an exten => in Asterisk something like exten => 8XXXXXX,1,Dial(SIP/fwduser:fwdpass@fwdipaddress/${EXTEN:1})
04:30.39snewpythey're also less than 6 digits
04:31.00inticonnethttp://pastebin.ca/6217 Would what I already have not suffice?
04:31.07mishehusnewpy: only the "special" ones, no?
04:31.15snewpymishehu: nope, mine's 84488
04:31.28snewpylots of 5 digit ones, at least
04:31.50ManxPowerinticonnet, Now what happens when you dial 7617504?
04:32.14ManxPoweron the asterisk console, of course.
04:32.20inticonnet"Number does not exist"
04:32.38inticonnetmm nothing
04:33.25ManxPowerNow, what happens if you use something like Dial(SIP/${FWDUSERID}@fwd-out/${EXTEN:1})
04:33.43inticonnetWhere do you want that put?
04:33.49ManxPowerI assume unsername= the value of ${FWDUSERID} in sip.conf.
04:34.16ManxPowerexten => _7.,3,Dial(SIP/${FWDUSERID}@fwd-out/${EXTEN:1})
04:34.29*** join/#asterisk BoRiS (boris@S01060040ca1e5b54.wp.shawcable.net)
04:34.40inticonnetargh I think i have to setup ssh...writting these down then flicking thru my kvm and reentering is painful
04:35.25mrproper_anyone here pusshing sip video to MS live comms server?
04:37.08ManxPowerAh, and PASTE the CLI output for the Dial line.
04:37.55inticonnetso who do you want me to call now?
04:38.33ManxPowerHmm?  I just know about sip.conf and Dial line formats, not FWD telephone numbers.
04:38.54mishehuinticonnet: you can try calling 256430
04:38.59mishehuthat's my fwd
04:39.17inticonnetso i have to prefix 7 now dont i?
04:39.34mishehuyup
04:39.58ManxPowerinticonnet, um, that's the exact same pattern match as you had.
04:40.07ManxPowerYou hat do dial 7 according to your pastebin
04:40.12inticonnetxlite tells me "call 404 not found" and nothing in the cli
04:40.26inticonnetive never had this working so it dosnt matter really :P
04:40.33mishehuinticonnet: did you do "extensions reload" recently?
04:40.41ManxPowerinticonnet, You are dialing 7256430 in X-Lite?
04:41.14inticonnetyes
04:41.24inticonneti always reload afetr a change
04:41.31ManxPowerinticonnet, You are do not have an include => fwd-outgoing in [from-sip]
04:41.35Dhp4the docs for the AMP say to isntall the cdr_mysql, in the dir /usr/src/asterisk-addons/ but its not there, was it moved in newer versions or what?
04:41.44ManxPowerSo X-lite cannot see the fwd stuff.
04:41.50mishehuthere is a possibilty that the problem is on my end.  I've not checked my FWD config in so long
04:41.58inticonnetargh :S
04:42.14mishehuDhp4: cvs
04:42.17ManxPowerAll this silly SIP diagnostics.
04:42.26Dhp4what about the CVS
04:42.29Dhp4im useing the CVS
04:42.30mishehuDhp4: or the asterisk-addons package
04:42.38Dhp4oh its an extra folder?
04:42.43mishehucvs co asterisk-addons I believe
04:42.47ManxPowerYou have to CVS checkout asterisk-addons
04:42.49inticonnetso add include => fwd-out in from sip?
04:43.09Dhp4ManxPower + mishehu - thanks
04:43.09ManxPowerNo, include => fwd-outgoing
04:43.17ManxPowerYu don't have a fwd-out CONTECT
04:43.26ManxPowerCONTEXT, even
04:43.32inticonneti did b4 i think....im really confused :P
04:45.01ManxPowerThe lack of CLI output would normally indicate that the call was not even getting to Asterisk.  Obviously you have mulpiple problems.
04:45.17ManxPowerBut the call is getting SOMEWHERE if you are getting a SIP 404 back.
04:45.34inticonnetOk I think we got somewhere. The op said "Im sorry but thats not a valid extension try again"
04:45.58ManxPowerinticonnet, sounds like the exten => _7.,4,Playback(invalid)
04:46.06JerJer_7X.
04:46.12ManxPowerinticonnet, now put the console output on pastebin.
04:47.30*** join/#asterisk Othello (Othello@nusnet-156-21.dynip.nus.edu.sg)
04:47.53inticonnetwww.pastebin.ca/6221
04:48.11ManxPowerFeb 21 15:47:26 WARNING[2452]: chan_sip.c:1405 create_addr: No such host: fwd-out
04:48.45Othellooh, just a quick one: Will asterisk work better in kernel 2.4 or 2.6?
04:48.50ManxPowerYou have fwd-outgoing in sip.conf, not fwd-out
04:48.54mishehuOthello: yes.
04:48.57Dhp4ok when i run the make for the addons i get 3 eroros, ast_list_remove' undeclared (first use in this funtion) (each undeclared identifier is reported only once for each funtion it appears in) make: *** [app_addon_sql_mysql.o] error 1
04:49.25Dhp4make clean
04:49.27Dhp4opps
04:49.59Dhp4any ideas?
04:50.05mishehutonight must be newbie night
04:50.24JerJerno kidding
04:50.24inticonnetfixed the outgoing issue in sip but still getting invalid extension message from op.
04:50.36JerJernot even a full moon
04:50.38ManxPowerinticonnet, well, we need another pastebin then, don't we?
04:50.43inticonnetMishehu- Ive been at it for 24 hours not just 12 :P
04:50.52mishehuJerJer: damn, I can't howl at the moon yet then.
04:50.58mishehumust wait for the full moon.
04:51.02Beirdogah
04:51.10BeirdoI'm feeling extra stupid tonight
04:51.22inticonnetManx - 6222
04:51.23mishehuinticonnet: try using it for a year, then you'll have earned some status.  ;-)
04:51.25Beirdowhat's the command to create a new voicemail box?
04:51.27ManxPowerBeirdo, don't feel bad, it's apparent it's an epidemic tonight
04:51.42inticonnethehehe
04:51.47mishehuBeirdo: I just edit the voicemail.conf or the db table ;-)
04:51.51BeirdoI've looked everywhere I can think, and for the life of me I can't find the thing
04:52.02Beirdono I mean to make the directory structure
04:52.10Dhp4mishehu: any ideas for my problum?
04:52.19ManxPowerinticonnet, You either didn't reload or you didn't change fwd-out to fwd-outgoing
04:52.33mishehuDhp4: that's not the full error msg, so no, I can't tell you what the problem is.  it could be that it's not finding asterisk.h
04:52.40mishehuthat's my only guess
04:52.40ManxPowerBeirdo, Um, Asterisk creates it for you.
04:52.54BeirdoOK, no wonder I can't find it
04:53.03Beirdowhat the hell was I thinking?
04:53.06agaveheh heh
04:53.06Beirdothanks
04:53.18ManxPowerBeirdo, Asterisk USED to require an external command, but that has not been needed for at least a year, maybe 2 years
04:53.27Dhp4ok i ran make again theres only 5 lines outputted so here it goes....
04:53.29Beirdoahhh
04:53.34inticonneti just reloaded and its still doing it.
04:53.39inticonnetill check the file again
04:53.41mishehuManxPower: about 2 years, as I've been using * since september of 2003.
04:53.49ManxPowerinticonnet, Well then you live in another universe since The Dial is using fwd-out.
04:53.51Beirdomaybe I saw some old doc somewhere in my travels
04:53.51mishehuand I never needed to run an external command.
04:54.25BeirdoI need to create some IVR menu recordings :)
04:54.25mishehuit's a mirror world.
04:54.29inticonnetTHATS WHAT WE WANT IT TO USE!
04:54.33JerJerwith lots of smoke
04:54.41ManxPowerinticonnet, then change it in sip.conf.
04:54.41inticonnetu said change it from outgoing to out
04:54.49inticonnetthats where i changed it
04:55.02ManxPowerinticonnet, Whatever the hell you use it must be the same in sip.conf and extensions.conf.
04:55.17ManxPowerThis message Feb 21 15:50:15 WARNING[2455]: chan_sip.c:1405 create_addr: No such host: fwd-out
04:55.18mishehuJerJer: hehe, I was actually referring to star trek tos & a song by S.P.O.C.K.
04:55.39ManxPowermeans "I can't find the section [fwd-out] in sip.conf so I'm going to assume it's a hostname and try to do a DNS lookup on it.
04:55.55mishehuyeah, I mean, we don't care if the context is [i-hate-you-all-now-die] as long as it's the same in both files.
04:56.09ManxPowermishehu, I need that context.
04:56.24*** join/#asterisk DHuang (~DHuang@adsl-102-99.swiftdsl.com.au)
04:56.28DHuanghi!
04:56.33mishehuManxPower: as long as I don't have the honor of having an extension in that context ;-)(
04:56.47DHuanghow do I resolve this? Unable to find a path from ilbc to g729
04:56.55ManxPowermishehu, That would be the default context, of course
04:57.01mishehuDHuang: do you have a license for g729?
04:57.17Dhp4xx-fPIC -I../asterisk -D_FNU_SOURCE  -I/usr/include/mysql      -c -o app_asson_sql_mysql.o app_addon_sql_mysql.c
04:57.17Dhp4app_addon_sql_mysql.c:164:49: macro "AST_list_Remove" passred 4 arguments but takes just 3
04:57.17ManxPowerDHuang, Purchase the G729 licenses or stop allow=ing g279
04:57.17Dhp4app_addon_sql_mysql.c: In function 'del_identifier':
04:57.17Dhp4app_addon_sql_mysql.c:165: error: 'AST_LIST_REMOVE' undeclared (first use in this function)
04:57.28mishehu~pastebin
04:57.29jbotit has been said that pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca
04:57.35Dhp4mishehu: thats ontop of what i posred
04:57.36inticonnetok now i get "cannot find extension context 2000"
04:57.38DHuangmishehu: yes I do.. I can force to get 2 x g729 working.. but if 1 is ilibc and 1 is g729 then stop working?
04:57.53Dhp4ehh sorry
04:57.57ManxPowerinticonnet, repastebin your config files.
04:58.02inticonnetok
04:58.03mishehuDHuang: hmm...  don't really know, I don't have a license for g729, and don't use ilbc
04:58.26DHuangmishehu: it's the same for ilbc to gsm too..
04:58.39mishehuDhp4: I think I know what your problem is.  do you have an older build of asterisk installed right now?
04:58.54Dhp4just what i got from CVS
05:00.04Dhp4so whatever is up in CVS is what i have - keep in mind asterisk starts and runs fine but i removed it and now am isntalling it to AMP's docs so i can use that
05:00.18inticonnetManx- 6224
05:00.33Qwellinticonnet: a full link is the "proper" way to do it
05:00.37*** join/#asterisk roamer323 (~sing@67.71.60.238)
05:00.59mishehuDhp4: well, I've seen that before myself, but I read the source and found what paramters it was looking for and removed the one it wasn't.
05:01.20inticonnetwww.pastebin.ca/6224
05:01.23roamer323do I need ztdummy for the cmd Ringing()  ? thx
05:01.24inticonnetsorry :S
05:01.28mishehuthat is, unless that was the time I accidentally had an older release of asterisk installed and it couldn't build against it
05:01.42Dhp4well
05:01.45Dhp4what one would it be?
05:02.02mishehuDhp4: gawd knows, that was about 3 months ago or so.
05:02.14ManxPowerinticonnet, http://pastebin.ca/6225  Notice that [general] is the FIRST section of extensions.conf and notice the change in sip.conf
05:02.16Dhp4so what should i do
05:02.37ManxPowerthe change in sip.conf is what I've been TRYING to make you understand
05:02.50ManxPowerDhp4, ask AMP users.
05:03.34Dhp4could i just skip it?
05:04.06ManxPowerDhp4, ask AMP users.
05:04.19Dhp4where ar amp usesers
05:04.37*** join/#asterisk |Vulture| (~Vulture@109.238.204.68.cfl.res.rr.com)
05:04.43ManxPowerOne would generally assume that support methods would be on the software's web site.
05:06.01inticonnetManx- Why do I now have both Out And Outgoing?
05:06.05|Vulture|Anyone know a site that shows average costs for a PRI?
05:06.28ManxPowerinticonnet, becuase there is no relationship whatsoever between them.
05:06.41inticonnetok
05:06.44ManxPowerA [context] in extensions.conf has nothing to do with a [section name] in sip.conf.
05:07.14ManxPowerThe only relationship beween the two is the context= line in the [section name] in sip.conf must corrospond with a [context] in extensions.conf
05:07.37inticonnetwww.pastebin.ca\6226
05:07.41*** part/#asterisk DHuang (~DHuang@adsl-102-99.swiftdsl.com.au)
05:08.31ManxPowerinticonnet, repastebin your sip.conf
05:08.50ManxPowerYou know I usually require dinner and drinks before this kind of handholding.
05:09.27syslodVulture:  THey vary alot.  What geo you in?
05:09.38inticonnetwww.pastebin.ca/6227
05:09.40Beirdoyou know what would be nice?  If you could break out of MusicOnHold with a dialed digit, say to break out to an operator
05:09.40mishehuManxPower: got the flu?
05:09.41mishehuheh
05:09.57ManxPowermishehu, newbie overload
05:10.03mishehuBeirdo: I believe that there is a way to do that
05:10.09syslodBeirdo: Doesn't background do that?
05:10.11ManxPowerIt's like the invading hordes of barbarians
05:10.26mishehuManxPower: there goes the roman empire...
05:10.32Beirdosyslod: how do you background MOH?
05:10.45ManxPowerCanada is prolly felling something similar from all the Americans moving there.
05:10.47inticonnetManx- If ur ever my way Ill take u down to the club I work at and give u a day of free drinks and food :P
05:10.49Beirdobackground is for playback of gsm files last I looked
05:10.49syslodTake a look at background
05:11.18|Vulture|syslod: I am looking for it in Jacksonville, Florida
05:11.24ManxPowerBeirdo, Background is of playback of any support audio file format AND expect DTMF.  Playback does the same without expecting DTMF.
05:11.24mishehuBeirdo: background is for playback of more than just gsm files
05:11.36syslodYour average should be around $800.
05:11.42mishehuyeah, exactly as ManxPower said
05:12.00Beirdocan it play via a custom script similar to MOH?
05:12.03inticonnetmanx- There u go www.pastebin.ca/6227 (I even put the url in this time :) )
05:12.18|Vulture|syslod: and you can get them in not full 23 channels for a lot cheaper right?
05:12.32ManxPowerinticonnet, Stop rearranging things!
05:12.42syslodVulture: Not really most lecs don't even offer frac PRI.
05:12.45inticonnetI did what u told me to :P
05:12.59ManxPower[general] must always be first.  register must always be in general.
05:13.15|Vulture|syslod: so if I was looking for around 10-12 lines I might just want to go with a fract T1 and a T100P card?
05:13.22syslodVulture: What are you looking for?  They don't really make alot of sense usually unless you have at least 8 lines.  I like to see ppl that need 16 or so before looking a PRI>
05:13.51syslodDo you need PRI?
05:13.52agave$800 for a PRI
05:13.52ManxPowerhttp://pastebin.ca/6228
05:14.03syslodagave: Yea in cities.
05:14.13agaveno, I agree
05:14.15agavethat's what I sell them for
05:14.17agave<-- clec
05:14.20ManxPowerThis is turning from handholding into something that requires monetary exchange.
05:14.20denonPRI can make a lot of sense if you need to set your own CID, or need some of the advanced features of audio quality
05:14.23|Vulture|syslod: no I only need between 10 and 12 lines
05:14.32inticonnetIm keen :)
05:14.32agavevulture: then buy from a SIP or IAX based provider
05:14.39syslodagave: U know anything about EMI or AMA records?
05:14.52agavesyslod: my billing analyst is better than i, but I may know the answer..
05:14.57syslodIAX trunks would be much cheaper.
05:15.04mishehu$800 for a pri?  damn.
05:15.14mishehuI can get loop for $300...  and I'm not a lec.
05:15.16ManxPowermishehu, that's pretty average.
05:15.25agavemish: retail.  wholesale you can get them for $250 - $500 depending on whether MOUs are included
05:15.31syslodagave:  We are look at writing a EMI combiner for VOIP to work with billing and SECABS.
05:16.03agavemm.. you want to bill out CABs records?
05:16.12syslodCABS and end user
05:16.12mishehuagave: you have to refresh my memory on what is an MOU...   the one acronym I always remember is PCMCIA...  "people can't memorize computer industry acronyms"
05:16.15|Vulture|right now we have 6 lines and 384K on a Frac T1 but we want to go to just phone lines
05:16.19agavemish: minutes of use
05:16.37denons/computer/telephony/
05:16.39mishehuagave: does that $800 include MOU ?
05:16.50|Vulture|pay ~$500 for the 6 lines and 384k
05:16.50syslodWe are doing it now but its a combination of outsourced.  I'm tired of not controling my billing.
05:16.54agavesyslod: well, if you're going to bill cabs then you're probably going to have to pull those records right off the TDM switch... you can't normally bill CABS off a voip trunk unless you're doing something... special
05:16.56mishehudenon: computers control your phone calls, so no need to correct
05:17.04agavemish: yes.  usually if you pay retail MOU is included, outbound and inbound
05:17.19denonmishehu: its a telephony acronym, though :)
05:17.22agaveto bill CABs you're going to need the CIC code to know who to bill, for instance
05:17.28syslodagave: We are doing something "special" and pulling off the switch.
05:17.31denoneven if all you had was Mr Bell and his tin cans and string :)
05:17.39mishehudenon: they merged together though.
05:17.54syslodYea we've got all that. LPIC, PIC the whole nine yards.
05:18.09inticonnetTime for more handholding manx?
05:18.26mishehuinticonnet: I thought you guys were up to kissing on the cheek
05:18.38inticonnetHehehehehe
05:18.39|Vulture|syslod: would it work getting a frac T1 with 12 lines and then plug it into a T100P?
05:18.43syslodagave: Are you doing billing in-house?
05:18.47*** join/#asterisk PBXtech (~upirc@wirelessdata-167-248.mycingular.net)
05:18.50BeirdoI thought I detected some pawing going on
05:18.54inticonnetI got invalid extension again...give me a sec ill get u another pastebin
05:19.03syslodVulture: Yes * can handle just about anything.
05:19.05agavewell our billing analyst wrote our CABS billing, we used to use a company called Intec...  really all you need to do for CABS is make sure you add them up correctly and bill the correct IXC... and charge the correct rates, and have it in your tariff
05:19.23mishehucabs == client access billing system?
05:19.26syslodagave: u won't rent you analyst out would you?
05:19.32agavewe bill out about $200,000 in CABS
05:19.35agavemonthly
05:19.35agaveheh
05:19.39|Vulture|syslod: this is just new to me Ive done a few installs with TDMs with only 4 lines, but this is a bigger office.. just trying to figure what we need
05:19.39agavesyslod, sorry, no can do.
05:19.42inticonnetManx- www.pastebin.ca/6229
05:19.52agavemish: no, CABS is carrier access billing
05:20.04syslodWe use intec now.
05:20.05inticonnetif i ever start an * support company ill make u the ceo :P
05:20.20agavesyslod: oh, I was going to say, if you need somebody, use intec.  having your own analyst is sometimes more of a headache
05:20.21denonagave: 200k in cellular cab?
05:20.39syslodagave: U wireless?
05:20.47agaveno, we're Facil. based
05:20.53agavei don't touch wireless
05:21.04denonwhat, no love of ulcers?
05:21.08denonlive a little :)
05:21.10agavehehe heh
05:21.16ManxPowerinticonnet, That means it's working, but the number you called is not currently registered with FWD.  You should remove the allow=all and put disallow=all and allow= the codecs you want.
05:21.31*** join/#asterisk jonas (jonas@jonas.probe-networks.host.freenode)
05:21.31syslodProblem with intec is they aren't integrated.  Compared to what we technically do know CABS doesn't look that bad.
05:21.35ManxPowerinticonnet, That's what they all say.
05:21.38denonManxPower: bkw would argue that now allow=all is a Good Thing(tm)
05:21.44denonsince he "fixed" it
05:21.49ManxPowerdenon, bkw_ is a lunatic.
05:21.57ManxPowerI like him, but he's still a lunatic.
05:21.58denonim just telling you what he would say :)
05:21.59agavesyslod: it's not hard.  unbillables are a PITA, and you'll get IXCs who don't want to pay
05:22.02agaveATT is the worst
05:22.12inticonnetThank Goodness :) Thanks Manx :D and at risk of breaking something else im not disallowing anything yet
05:22.21denonagave: so turn em over to colletions :)
05:22.23ManxPowerinticonnet, allow=all will break things.
05:22.25denonput a lean on their properties <G>
05:22.33syslod:) Don't term there calls.  When they call up after a customer complains let them know what they owe. :)
05:22.34mishehuatt is being bought out by sbc
05:22.37mishehugood luck!
05:22.40inticonnetargh so what should I change it to :S
05:22.45agavewe make the most CABs on our originated traffic, heh
05:22.57ManxPowerinticonnet, your original allow and disallow looked good to me.
05:22.59syslodorig traffic?  Third party or LD?
05:23.03agaveld
05:23.06agave+ tfns
05:23.13inticonnet*Ouch* I stuck my finger in my pcmcia slot. Its hot in there
05:23.14mishehuQ: where does a 300 billion monkey toss his poo?
05:23.15inticonnetok thx
05:23.25*** join/#asterisk datareactor (datareacto@203.81.192.33)
05:23.26mishehuA: wherever he likes to, which is usually right at you.
05:23.45mishehuerr insert "pound" between billion and monkey
05:23.53Inv_arpis there a ${TRUNK} command in *?
05:23.55syslodagave: U doing end user billing IH?
05:24.04agavewe do everything in house, now.
05:24.08mishehuInv_arp: ${TRUNK} is a variable
05:24.23syslodYea thats where we are headed.  Trying to find others so we aren't doing it alone.
05:24.44agave-txlinksyslod, are you a CLEC?
05:24.48syslodYea.
05:24.53agave-txlinkwhat state?
05:24.54syslodWe serve southeast
05:24.58agave-txlinkbell south?
05:25.11ManxPowerBellSouth is not a CLEC.
05:25.12syslodMost of there territory.  Little VZ too
05:25.19Inv_arpmishehu: a global variable used by *?
05:25.21agave-txlinkeek VZ
05:25.23agave-txlinkmanx: no shit
05:25.23syslodFacilities based.  Enterprise customers.
05:25.30agave-txlinkwe're mainly SBC
05:25.35agave-txlinksome qwest
05:25.41denonhuh wha? syslod owns MCI? </rumors start=true>
05:26.06denonpity, that'd almost work if they were privately held :)
05:26.15ManxPowerdenon, Obviusly not or he'd be in jail.
05:26.25syslodI won't of gave them a dollar for MCI. VZ is just gonna hull that infrastucture and gobble up the customer.
05:26.29agave-txlinkmanx: all CLECs have to operate in some ILEC territory...
05:26.30denonhah
05:26.39ManxPoweragave-txlink, I know.
05:26.39agave-txlinkhah, SBC got the better deal
05:26.41agave-txlinkthose assholes
05:26.58agave-txlinkmanx: thus my bellsouth comment said he said southeast...
05:27.02mishehuFeb 20 23:26:35 NOTICE[11040]: chan_sip.c:7271 handle_request: Failed to authenticate user "asterisk" <sip:asterisk@192.168.5.26>;tag=as485edb38
05:27.08mishehuis somebody trying to call me?
05:27.35ManxPowerMCI can't help but suck.  They are a company made up of companies they bought.  Integrating all those voice and data backbones and billing systems is not possible.
05:27.37Sedoroxthats a lock address....
05:27.40Sedoroxlocal*
05:27.43syslodagave: U are doing IAX term/orig?
05:27.48agave-txlinksyslod: si
05:27.59ManxPowerTHAT'S why we won't use MCI.
05:28.18syslodKinda like alcatel :)
05:28.32BeirdoManxPower: it's possible, but will take time
05:28.37ManxPowerI seem to recall an article I read a year or two ago that said that MCI had 37 billing systems.
05:28.41agave-txlinkheh
05:28.41mishehuyeah, and now mci is going to verizon now no?
05:28.44agave-txlinkMCI reps
05:28.46agave-txlinkused to gain double commissions
05:28.50agave-txlinkby bouncing orders from one system to another
05:28.51agave-txlinkhehe
05:28.55Beirdokeep counting, ManxPower, I think that's low
05:29.06inticonnetI setup another fwd account and sucsesfully called myself :)
05:29.07Hmm-workthis guy has issue's
05:29.22ManxPowerBeirdo, Remember that 2 years ago they had time to integrate some of their systems.
05:29.23Inv_arphmm anyone have an ex.. on how to do sip to hardphone  xfers? in extension.conf
05:29.27Beirdomishehu: might be going to Verizon...  if the regulatory people let it happen, and the MCI shareholders agree
05:29.53syslodAre there any carriers left for the carriers carrier?
05:29.56md99is anyone online using passive BRI ISDN Cards?
05:29.58Sedoroxummm
05:30.00SedoroxQuestion
05:30.02ManxPowersyslod, Level3?
05:30.10terrapendoes anybody have a copy of the Bell System Practices document set?
05:30.13BeirdoManxPower: remember that 2 years ago, they went into bankruptcy protection due to stupid past excesses, and had to lay off a lot of people
05:30.14mishehuBeirdo: with republican federal gov't, and sprintpcs's purchase of nextel, and sbc's purchase of att, what makes you think it's not a done deal?
05:30.16Sedoroxwhy would the Shareholders not want it to happen when they turned down a offer from Qwest for a higher amount
05:30.22terrapen(does anybody know what i'm talking about?)
05:30.24syslodHmm.  Does SBC or VZ use level3?
05:30.38syslodterrapen: Which ones?
05:30.46Beirdomishehu: I think both of those are pending approval too
05:31.00terrapensyslod, i'm looking for some that might cover 1A2 systems
05:31.04terrapenbut really any woudl be nice
05:31.10BeirdoSedorox: they didn't turn down the Qwest offer, the *board* did
05:31.14terrapenhttp://www.bellsystemmemorial.com/cds-documents.html
05:31.17mishehuBeirdo: sprintpcs is approved I believe.
05:31.17terrapenthat's what i want
05:31.25agave-txlinkheh
05:31.27agave-txlinklevel3 sucks
05:31.31agave-txlinkwe have an interconnect iwth them
05:31.32syslodI've got a older copy from the contel days somewhere.
05:31.38agave-txlinktheir termination rates are horrible and their origination is worse
05:31.42Beirdoyou are likely right, it will probably get shoved through, though
05:31.43terrapenreally?
05:31.45agave-txlinkand they have a $50K / mo minimum commit
05:31.47Sedoroxhmmm
05:31.56roamer323I hear ringback when calling ATA to softphone, and softphone to softphone, but not from softphone to ATA, and not from any incoming DID call - anyone knows what the problem may be?
05:32.18Beirdoat which point MCI will be doing more layoffs
05:32.48BeirdoI hope to be re-employed by then (I'm on contract to MCI Canada right now - internal systems UNIX admin)
05:33.50inticonnetWould it scare u all If I told u I was a network admin :P Good thing we use windows and not linux I guess hehehe
05:33.56ManxPowerOne would assume that "A carrier's carrier" would have high min monthly billings.
05:34.18agave-txlinkone would also assume that if you're spending $50K a month you'd get a good rate
05:34.29ManxPoweragave-txlink, Yes, you would also assume that.
05:34.38ManxPowerI guess they want to make a profit. 8-)
05:34.54agave-txlinkbah.. they're selling a TDM product in a VoIP world
05:35.08ManxPowerAre they reliable?
05:35.11agave-txlinkno
05:35.18ManxPowerCan you get tech support when you need it?
05:35.19agave-txlinki have 503's from them a lot
05:35.23agave-txlinktech support?  LOL
05:35.31agave-txlinkyou haven't dealt with lvl3 .. have you ? :)
05:35.36ManxPowerThey sound like every other VoIP terminatin provider then.
05:35.58level3-idiothello, I believe the problem is that your SIP gateway is being interferred with by the earth's ionsphere
05:35.59ManxPowerlevel3-idiot, I'm not a carrier.
05:36.17level3-idiotnow buy another GIG-E port from us and the problem will go away
05:36.30level3-idiotwe'll only charge you two times the going rate
05:36.42level3-idiotand we'll buy you lunch at a crappy resteraunt
05:37.50agave-txlink<ManxPower> They sound like every other VoIP terminatin provider then.
05:37.55agave-txlinkyou sound bitter, manx... have you had bad experiences?
05:38.18agave-txlink:)
05:38.23mishehuwho HASN'T had bad experiences in telcom?
05:38.24ManxPoweragave-txlink, No, based on reports on the mailing lists I avoid using ITSPs except as a backup to my PSTN lines.
05:38.45agave-txlinkheheh
05:39.03ManxPowerI use VoIP, I just terminate my own calls.
05:39.19agave-txlinkbut you still need a backup ?
05:39.40syslodI think the model is be your own provider, own the last mile, rent them a phone system, bundle the internet web and email, kill off the competition, wait for the ILEC to raise pricing then raise them yourself.
05:39.53agave-txlinksyslod: amen
05:39.57ManxPoweragave-txlink, Why not have a backup?  You never know if you will run out of PSTN channels on a busy day or you never know when the PRI will go down.  Doesn't happen often, but it does happen.
05:40.05agave-txlinkor you could be like icenet and give it away and then wonder why you have $10m in debt...
05:40.12syslod:)
05:40.13agave-txlinkmanx, i'm just giving you a hard time :)
05:40.23*** part/#asterisk agave-txlink (phanop@216.81.43.75)
05:40.28terrapeni got a response on my question about 1A2 phone systems, posted to a telephone tech forum:
05:40.32terrapen"Have fun wrapping your apartment with 25 pair. It's quite attractive."
05:40.34syslodI see pricing going up, cost going down.  Perfect time to be in telecom.
05:40.35terrapenheh
05:40.40*** join/#asterisk agave-txlink (phanop@216.81.43.75)
05:40.41agave-txlinkwhoops
05:40.47terrapen25 pair cable is a bit excessive
05:40.51*** join/#asterisk |neuro| (~|neuro|@212.176.51.231)
05:41.02Sedoroxhow about 100pair?
05:41.09terrapenmaybe i should just say fuck it and give up on 1A2
05:41.14Sedoroxhad some of that a while ago... was a bitch to work with... and to throw out... lol
05:41.14syslodalum cable?
05:41.17ManxPowerAt the ISP I used to work at we had 25 pair CAT 5.  Really weird looking stuff.
05:41.17terrapeni *really* like these phones, though
05:41.21inticonnetAre u still on about that terra :P
05:41.24agave-txlinksounds like ABAM cable...
05:41.26terrapenits just a matter of HOW MUCH do i really like them
05:41.28terrapenyes, inti :)
05:41.47terrapenif i were building a new house, i would do this
05:41.49syslodterrapen: just gut it and put something inside that works.
05:41.49inticonnetwell I paid $1500 for a dead system once just cause I thought I could fix it :P
05:41.50terrapenbut im in an apartment
05:42.10mishehuinticonnet: sounds like that windows system you use.
05:42.19ManxPowerI have cat 5 running all over my apartment.  It's ugly, but anyone that cares that much about looks doesn't get invited to my apartment.
05:42.25terrapenhttp://home.att.net/~wd0giv/Phones/bigbuttonphone.jpg
05:42.30terrapenthere's a phone for me
05:42.33agave-txlinksyslod: are you using * for your enterprise deployments?
05:42.35inticonnetBloody windows
05:42.39terrapenmy sister had that phone, in like 1983
05:42.50mishehuManxPower: that includes potential S.O.'s I imagine
05:42.55agave-txlinkmishehu, hehee
05:43.06syslodRight now we have traditional iron running enterprise.
05:43.07ManxPowermishehu, Yes.  And current ones.
05:43.21syslodR&D with * right now.
05:43.35terrapensyslod, where do you work>?
05:43.39agave-txlinksyslod: okay, just wondering how it was working out for you.
05:43.46terrapenhttp://home.att.net/~wd0giv/Phones/aligatophone.jpg
05:43.47terrapenOH WORD
05:43.50terrapenaligator skin phone
05:43.56agave-txlinkwe have our call center and both admin offices running on it
05:43.58terrapenpuff daddy would be proud
05:44.14syslodWell, after spending 6 months integrating it with OSS it works great.  Click click and you have a working high end PBX phone.
05:44.31ManxPowermishehu, I have computer guts scattered around the apartment.  The Cat 5 is not unusual.
05:44.45terrapeni'm wondering how much IVR stuff our customers will tolerate
05:44.54terrapenit would be so nice to automate a lot of the stuff with IVR
05:45.01terrapenbut i'm afraid people will get pissed and hang up
05:45.10agave-txlinkcustomers love web automation... not so many people like IVR automation
05:45.11inticonnetI want to change ours at work so when they push 8 to log a fault it hangs up on them
05:45.23ManxPowerterracon, Um, that's the GOAL of an IVR!
05:45.23terrapenagave: isn't that funny.
05:45.23Hmm-workcustomers get pissed, hang up and call back
05:45.35Sedoroxterrapen: I wouldn't have more then a 3 menu deep system
05:45.40*** join/#asterisk twisted (~twisted@twisted-professional-pdpc.developer.and.friend.of.asterisk)
05:45.40*** mode/#asterisk [+o twisted] by ChanServ
05:45.46agave-txlinkheh...  I accidently had asterisk voicemail programmed wrong once so if someone hit an ext. for voicemail it hung up on them
05:45.46agave-txlinkwhoops
05:45.50Hmm-workand you won't piss them off completely if you have a way to put them in queue with someone live
05:45.51Sedoroxours is only one.. right now.. lol
05:45.55agave-txlinki wondered why the voicemails were light for a few days.....
05:46.04terrapenwell, i was thinking about having customers enter their home phone number (which we use as a customer ID) when they call into the IVR
05:46.13syslodWe have 3 test customers.  Insurance, Lawyer, and local govt building.  All have there own speical twists they like.
05:46.25terrapeni get so pissed when i enter a number in an IVR and then the rep asks me for my number all again
05:46.35inticonnetMicrosoft ay :)
05:46.52inticonnetThem and their dodgy product activation
05:47.05syslodagave: U doing configs by hand?
05:47.16agave-txlinksyslod: yes :(
05:47.17|Vulture|hmm this online quote tool is quoting $630 for a T1 PRI with unlimited local calling... 1 year contract
05:47.22agave-txlinkhaven't had time to play with realtime yet
05:47.22inticonnet"No this is the only computer this copy of windows is on, and what was the activation key again"
05:47.28agave-txlinkwe have the provisioners do it
05:47.38Inv_arphmm anyone have an ex.. on how to do sip to hardphone  xfers? in extension.conf
05:47.39agave-txlinki'd really like to move to web based so we can integrate with OSS
05:47.53ManxPowerInv_arp, Uh, use the SIP device
05:47.58ManxPowerInv_arp, Uh, use the SIP device's TRANSFER button?
05:48.01terrapeni don't see the point of realtime
05:48.18ManxPowerterracon, RealTime is useful for people with larger deployments.
05:48.21terrapenseems to be that it adds a very critical point of failure
05:48.25Inv_arpManxPower: i meant sip to pstn  if no one picks up sip
05:48.25*** join/#asterisk scardinal (~supreme@port816.ds1-suoe.adsl.cybercity.dk)
05:48.32agave-txlinkyeah
05:48.39agave-txlinkone month we had enough problem with cdr_mysql
05:48.45agave-txlinkstill not sure what happened, but we lost billing records
05:48.48syslodI'll trade you OSS for *, DMS-100 and semiens for CABS analisyt :)
05:49.03agave-txlinkasterisk was writing them to the cSV files but seemed to be logging them to mysql, too...
05:49.18agave-txlinksyslod: sorry, no deal, we have OSS for Siemens EWSD / DCO + DMS 500 already :p
05:49.25ManxPowerWe may move to RealTime when it's released.
05:49.25agave-txlinkwhat siemens are you using ?
05:49.27terrapeni do not want to bet my goddamned phonesystem on mysql
05:49.41syslodSame
05:49.44ManxPowerterracon, uh, RealTime lets you use pretty much any database you wany.
05:49.52ManxPowerwant
05:50.03agave-txlinkyes, the old siemens CLEC special combo
05:50.06agave-txlinkadd hot sauce for $1.99 extra
05:50.34agave-txlinkis realtime going to be in 1.2 stable ?
05:50.38syslodWe've went back to using files only for both configs and CDR.  any DB seems to loose data, crash or otherwise be unrelaible.
05:50.42ManxPoweragave-txlink, I assume so.
05:50.48inticonnetUmm guys, sorry to ask this again but sip:2000@192.168.5.26 should just work shouldnt it?
05:50.51ManxPoweror whatever Mark calls the next stable release.
05:51.12agave-txlinksyslod: yeah if we have another bad billing month i'm going to CDR-text too...  it's easy to do an import into SQL and then do mediation before hand anyway...
05:51.23agave-txlinkthe main reason we have hte damn SQL is so customers can look at their CDRs in real time
05:51.24syslod:)
05:52.04syslodWe have it down to about a 6 second interval but we keep the raw files. Pipes can get you real time CDR without loosing records.
05:52.22agave-txlinkhrm..
05:52.30syslodI don't know of any switches that do anything in a relational db so I assume they know something.
05:52.42agave-txlinklol
05:52.59terrapensyslod, i agree totally
05:53.03terrapeni won't bet on a DB
05:53.25terrapenmaybe postgres running on freebsd
05:53.30terrapenbut really not even that
05:53.31syslodIt might work for a small PBX but in our testing with like a 300+ extension system it just is not smart.  Take trunking 1000+ accounts and it really craps out.
05:53.31anto9ussyslod: I've always found postgesql very reliable, I have a cron job to back up all databases to another server every 15 minutes, haven't had to restore in 2 years
05:53.42ManxPowerYou know there is a CDR Fork application, right?
05:53.53ManxPowerLets you log CDRs to more than one destination.
05:53.53agave-txlinkmanx: explain ?  i have heard of that
05:53.55syslodWhat does fork do?
05:53.57agave-txlinkmanx: oh, okay.
05:54.10agave-txlinkso why does my asterisk log to .csv AND mysql ?
05:54.15agave-txlinkand i'm not using CDR Fork ?
05:54.23ManxPoweragave-txlink, I only know about it from reading the asterisk-cvs mailing list.
05:54.28syslodI still think its easier to do it the old fashioned way.  Log to a file and process a few records at a time.
05:54.30*** join/#asterisk rett (~rett@c-67-171-236-169.client.comcast.net)
05:54.37ManxPoweragave-txlink, I think it logs to the text file by default.
05:54.48agave-txlinkit may just be the * version
05:54.53agave-txlinkwhen I was using -HEAD it would do one or the other
05:54.54agave-txlinknow i'm using -STABLE
05:55.15terrapenhow do you get -STABLE via cvs
05:55.17syslodThe way our switches do it to have a set length of time to close a file.  Like 6 minuites.  It works really well.
05:55.23agave-txlink-rv1-0
05:55.24inticonnetargh theres no audio comming back in i just realised :'(
05:55.24agave-txlinkor something like that
05:55.29agave-txlinkinstructinos are on asterisk.org under download
05:55.33terrapeni've always folled the installation instructions on the wiki but i never know if im getting -HEAD or -STABLE
05:55.34inticonnetI called 7612 which is fwd automated time
05:55.35ManxPoweragave-txlink, You realized that CVS-HEAD is called "the developement version of Asterisk" for a reason, huh?
05:55.40inticonnetBUT THERES NO TIME
05:55.43inticonnet:')
05:55.49agave-txlinkmanx... uh... yeah?
05:55.54ManxPowerunless you specify -r v1-0 on the cvs line you are getting CVS-HEAD.
05:56.07Qwellinticonnet: use 613m and do an echo test
05:56.18agave-txlinkwe used to get CVS-HEAD but lately have been doing -r v1-0
05:56.21Qwells/613m/613,/
05:56.24inticonnetHeres the funny thing..I cant find my microphone so I cant :P
05:56.41inticonnetIm not having a good day
05:57.21syslodMy personal opinion is that * should have all settings in memory and something seperate to edit realtime.  LDAP would be best storage.  I mean think about it you wouldn't get a core IP router attached to MYSQL for routing DB.
05:57.51syslodOr would you? :)
05:57.52anto9usLDAP is not very good for frequent updates
05:58.13agave-txlinkbah...
05:58.19syslodIn all my * installings it like 99.99999 reads and like .000000001 writes.
05:58.19anto9usit's optimised for querying
05:58.34syslodAnyways.  I like the edit and publish approach.
05:58.47inticonnetSo without the ability to do an echo test would we assume its a port forwarding problem?
05:58.56rettHas anyone in here run multiple instances of asterisk within Xen?
05:59.21syslodanto9us: I've heard it before.  LDAP is slower a writing than reading.  I'm talking about configs not CDR so thats not an issue.
05:59.31*** join/#asterisk naouri (bonoi@d142-59-238-42.abhsia.telus.net)
05:59.35agave-txlinkwell for me configs don't really need to be realtime anyway
05:59.39j_viannaagave-txlink: I need good international rates for Latin-America and Europe, do you know someone ?
05:59.44agave-txlinkexcept for the pbx part
05:59.53agave-txlinkj_vianna: 3U Telecom is pretty good.  I don't do any international myself
05:59.58syslodMe either I like to edit and publish rather than do things realtime on configs.
06:00.25agave-txlinkif i were creating some kind of pbx product i'd want to do realtime though.. customer self-provision is hot right now
06:00.31Pkunki have busycount=6 in zapata.conf
06:00.31agave-txlinkbut thank $DIETY I'm not in that business
06:00.42agave-txlinkDEITY
06:00.43agave-txlinkrather
06:00.48syslodIn our OSS application you can choose interactive and batch.  Its all stored in LDAP and upon batch or interactive update it just spits out a file and reloads.
06:00.53Pkunkand if i dial 7 dtmf's then my line gets disconnected
06:01.22Pkunki have to space out the dtmf's at with least one second gaps ..
06:01.35Pkunkso what is the problem ?
06:02.12syslodYea thats what I am talking about.   Realtime to me is the system is using that DB for operations.  Non-Realtime but still able to selfprovison is batch type operation.
06:02.31Pkunkthis problem wasn't there with 1 year old cvs install
06:02.48Pkunki upgraded just yesterday and this problem popped up out of nowhere
06:03.58syslodagave: We are doing self-provisioning using LDAP storage, a publish application, and OSS interface.  Takes liek 1 or 2 sec to add any account.  Even have XML for poly and grandstream built in.
06:04.55inticonnetwhen i dont use * and go straight out thru fwd I do get audio comming back in!?
06:04.58Pkunkso is my problem .. there with the zaptel driver ?
06:05.04inticonnetWhat does that mean
06:05.06inticonnet?
06:05.06Pkunkor with asterisk itself ?
06:06.49terrapeni've never gotten up the nerve to learn LDAP
06:07.02terrapenbeen doing *nix for 12 years and still have yet to mess with it
06:07.15syslodLDAP is good at storing config data like * has where you have contexts and stuff.
06:07.41terrapenwhat is the advantage of LDAP over a traditional SQL DB?
06:07.53ManxPowerI would just be happy for LDAP to store user speccific settings
06:08.37agave-txlinkwow.. no love for sixtel/iax.cc
06:08.38syslodIts object oriented. Its very fast a reading.  An the code, at least for the 12 developers here, is easier to write and maintance since the DB matches the data.
06:09.22terrapenholy shit
06:09.30terrapenHunter S. Thompson committed suicide
06:10.15agave-txlinkno wonder ManxPower won't buy from ITSPss
06:10.16agave-txlinkhehhheh
06:12.07*** join/#asterisk djin (~djin@gridfox.xs4all.nl)
06:12.30WildPikachuterrapen, its optimized for read queries
06:15.35ManxPoweragave-txlink, LOL!
06:17.05agave-txlinkseriously, i saw bitches about iax.cc/livevoip/voicepulse and voipjet all in about five days worth of archives
06:17.45ManxPoweragave-txlink, I won't use VoipJet based on some nasty comments they made about other ITSPs in on asterisk-biz
06:18.09Qwell~itsp
06:18.34agave-txlinkyeah I remember that thread
06:18.35Qwelloh, silly me
06:18.36ManxPowerjbot, ITSP is Internet Telephonny Service Provider.  An ITSP is a "VoIP Phone Company"
06:18.37jbotManxPower: okay
06:19.04QwellManxPower: got an extra n in telephony there
06:19.08agave-txlink~itsp
06:19.09jbotitsp is, like, Internet Telephonny Service Provider.  An ITSP is a "VoIP Phone Company"
06:19.13agave-txlinklol
06:19.14agave-txlinkcool
06:19.15ManxPowerjbot, ITSP is Internet Telephony Service Provider.  An ITSP is a "VoIP Phone Company"
06:19.16jbot...but itsp is already something else...
06:19.16agave-txlink~mou
06:19.21ManxPowerjbot, no ITSP is Internet Telephony Service Provider.  An ITSP is a "VoIP Phone Company"
06:19.22jbotManxPower: okay
06:19.33ManxPower~itsp
06:19.34jbotitsp is, like, Internet Telephony Service Provider.  An ITSP is a "VoIP Phone Company"
06:19.45agave-txlink~CLEC
06:19.47jbotit has been said that clec is Competitive Local Exchange Carrier. The OTHER phone company. ;)
06:20.07agave-txlink~TxLink
06:20.56*** join/#asterisk santiago (~santiago@63.245.86.121)
06:21.24terrapenagave, do you work for txlink
06:21.32agave-txlinkyes.
06:22.00ManxPowerWOW!  Bellcore was bought by SAIC and renamed Telcordia.
06:22.31agave-txlinkterrapen: yes, why?
06:22.37ManxPowerI always thought SAIC was the public company the hid USA govt secret research projects and used their consulting services to fund them.
06:22.47terrapenjust curious.
06:22.50agave-txlinktelcordia runs like a govnt. agency now
06:22.53agave-txlinki wouldn't be surprised
06:23.06agave-txlinksan antonio ?
06:23.09terrapenyep
06:23.19agave-txlinkah, we're based in dallas
06:23.23terrapenyeah
06:23.24agave-txlinkwe do have facilities in satx however
06:23.32terrapen100 Taylor?
06:23.41agave-txlinknot sure to be honest
06:23.55agave-txlinki deal mainly with dal, lax, and nyc
06:23.56terrapendo you terminate calls there?
06:24.01terrapen(SAT)?
06:24.02*** join/#asterisk B4 (~B4@202.69.48.245)
06:24.03agave-txlinkyes, we term and orig. from satx
06:24.05inticonnetI think Ive been quiet for too long :) RAAAA DAMN FWD..Im over it :)
06:24.17ManxPower~acd
06:24.18jboti heard acd is A specialized phone system that handles incoming calls or makes outgoing calls. An ACD can recognize and answer an incoming call, look in its database for instructions on what to do with that call, play a recorded message for the caller (based on instructions from the database), and send the caller to a live operator as soon as the operator is free ...
06:24.23B4~seen zx81
06:24.35jbotzx81 <matt@222-153-114-115.jetstream.xtra.co.nz> was last seen on IRC in channel #asterisk, 1d 17h 48m 23s ago, saying: 'ok brb~'.
06:24.35*** join/#asterisk eipi (~eipi@40-142-89-200.fibertel.com.ar)
06:24.50QwellThats one hell of a "brb"
06:24.58inticonnet:P
06:25.08B4lol
06:25.15terrapen~TDM
06:25.34agave-txlinkTime Division Multiplexing
06:25.36B4time domain multiplexing :)
06:25.42B4oops division right
06:26.08QwellThats just as foreign, heh
06:26.13ManxPower~clec
06:26.14jbotclec is probably Competitive Local Exchange Carrier. The OTHER phone company. ;)
06:26.22Inv_arphow would i set up hold for the HT 486 .. do i setup a  key sequence for putting someone on MOH?
06:26.31ManxPower~clec
06:26.32jbot[clec] Created by the Telecommunications Act of 1996, a CLEC is a service provider that is in direct competition with an incumbent service provider. CLEC is often used as a general term for any competitor, but the term actually has legal implications. To become a CLEC, a service provider must be granted "CLEC status" by a state's Public Utilities Commission. In ...
06:26.52QwellIn ... ?
06:27.01agave-txlinkran out of buffer it seems
06:27.19ManxPoweryeah.
06:28.03inticonnetGuys should my extension ring for sip:2000@externalip
06:28.12*** part/#asterisk rett (~rett@c-67-171-236-169.client.comcast.net)
06:28.54Qwell~ilec
06:29.38Qwelleither I'm lagging, or he doesn't know
06:29.46agave-txlinkinticonnect: depending on how you have your contexts set up, possibly
06:29.55ManxPower~ilec
06:29.56jboti heard ilec is Typically the carrier that was granted the right to provide service as a result of the breakup of AT&T. These providers are also referred to as RBOCs (Regional Bell Operating Companies) or Baby Bells.
06:30.10agave-txlinkwell, that definition can be wrong
06:30.13agave-txlinkcenturytel is an ilec
06:30.17agave-txlinkthey are not an RBOC nor a baby bell
06:30.25agave-txlinksee also : valor
06:30.29agave-txlinket. al
06:30.35inticonnetIm having huge problems here :(
06:30.36bkw_centrytel is not an ilec are they?
06:30.40bkw_I thought they were a clec
06:30.40agave-txlinkyes they sure are
06:30.40ManxPowerTYPICALLY
06:30.50agave-txlinkno, they are incumbent in arkansas, missouri, and others
06:31.00agave-txlinkcitizens is also an IL
06:31.01agave-txlinkEC
06:31.16agave-txlinkand it possible to be an ILEC and CLEC, such as SC Telcom in kansas
06:31.23bkw_bet they are
06:31.51QwellDoes anybody know of a GOOD explanation of what has happened with the Bells in the last x(20?) years?
06:31.58agave-txlinki could tell you
06:31.58agave-txlinkheh
06:32.01agave-txlinkthe cliff's notes
06:32.20inticonnetif i call inwards from outside of my network (Eg. sip:2000@220.233.68.118) my laptop rings. Which has nothing to do with asterisk
06:32.20Qwelllike, who bought who, etc...the long drawn out details
06:32.20bkw_qwell they ripped off alot of people
06:32.20bkw_and now are gonna get bigger
06:32.21agave-txlink1984 - divesture -- judge splits ATT into several regional carriers like southwestern bell, bell atlantic, nynex, mountain bell, etc.
06:32.22Qwellheh
06:32.22bkw_and more unstopable
06:32.25agave-txlinkcreates LATAs
06:32.31Qwellbkw_: Thats kinda what I figured
06:32.33agave-txlink1996 --telecom act creates clecs
06:32.42QwellSo, it was all called "AT&T"?
06:32.50agave-txlinkmergers start happening ---  ameritech + southwestern bell = sbc
06:32.53ManxPowerQwell, The govt broke up AT&T into many different companies, they all ran around confizzled for a few years, then the govt forced them to stop locking out competition, currently they are all i the process of merging back into 1 company.
06:32.56agave-txlinknynex + bell atlantic + verizon = verizon
06:32.56QwellBesides from the name Alexander Graham Bell, where did "Bell" come from?
06:33.13bkw_agave-txlink, its just gonna get worse
06:33.18bkw_we have ATT+SBC
06:33.18agave-txlinkyeah
06:33.22agave-txlinkVZ+MCI
06:33.25bkw_MCI+VZ
06:33.33agave-txlinksprint and level3 will be bought soon
06:33.35bkw_== BAD TIME
06:33.40QwellSo, let me get this straight...
06:33.47bkw_the bells are getting bigger
06:33.52Qwellevery damn "major" teleco I've ever heard of, are...
06:33.52Qwellnow Bell?  heh
06:33.55bkw_right under the nose of the regulators
06:34.08bkw_SBC is evil
06:34.10Qwellhow?
06:34.10inticonnetOne of ur largest data providers here in australia recently went into administration and was bought out....
06:34.15bkw_the most crooked company on this planet
06:34.22bkw_very anti conpetitive
06:34.26bkw_er com
06:34.31agave-txlinkbkw speaks da "troof"
06:34.31bkw_they are not right
06:34.37bkw_I tell ya EVIL
06:34.39bkw_to the CORE
06:34.43bkw_they lie
06:34.45bkw_they cheat
06:34.49QwellHow are they able to do all these mergers with each other?
06:34.50bkw_I have caught them
06:35.00bkw_Qwell nobody is paying attention
06:35.05Qwellif its obvious to ME...
06:35.12Qwellit should be damn obvious to them
06:35.13bkw_see
06:35.20bkw_everyone is busy fighting terror
06:35.21ManxPowerQwell, These days "Bell" is not a correct term, but people use it to mean "The ILEC"
06:35.31terrapencan you reliably run two TDM400P cards in a single system>?
06:35.40QwellManxPower: Where did the name "Bell" come from, if it was ATT?
06:35.44agave-txlinkour telecom system is shit
06:35.45agave-txlinkin the US
06:35.45bkw_shoudl be able to
06:35.49terrapenbkw, tell us how you really feel.  :)
06:35.55bkw_agave-txlink, so i sour health care system
06:36.06bkw_they hold your health hostage
06:36.06terrapen<bkw_> shoudl be able to
06:36.08inticonnetU guys are sort of lucky with ur multiple provders. We only have 2 country wide, one of which feeds off the others network. There are of corse smaller "resellers" but yeah. Only 1 real provider
06:36.10Qwellsee, this is why I want the long explanation, to see how everything is working
06:36.12terrapenwas that directed at me?
06:36.22ManxPowerQwell, Even though they were owned by AT&T the ILECs were still called Michigan Bell, Illinois Bell, etc.
06:36.27bkw_terrapen, yes
06:36.28Qwellthe AT&T split almost predates me, heh
06:36.29Qwellahh
06:36.32terrapenbkw: k, thx
06:36.39QwellManxPower: thank you
06:36.45bkw_who is terracon?
06:36.49bkw_is that you terrapen?
06:36.51terrapeni have no idea.
06:36.51QwellWere they always owned by AT&T?
06:36.51terrapenno
06:36.54bkw_ok
06:36.57terrapenpeople always call me terracon
06:36.59bkw_just anoys me
06:37.00agave-txlinkoh well it's still an interesting business
06:37.02terrapeni think its some nick complete script
06:37.08agave-txlinkalways a new challenge
06:37.08agave-txlinkheh
06:37.12Qwellterrapen: You're higher on the nick complete list
06:37.17terrapeni've just gotten used to it :)
06:37.19Qwellyou take two tabs ;]
06:37.22bkw_SBC MUST BE STOPED
06:37.28terrapenhahah
06:37.29bkw_VZ must be too
06:37.32bkw_EVIL EVIL EVIL
06:37.43terrapenmaybe i should be...
06:37.44terrapenterraben
06:37.46*** topic/#asterisk by bkw_ -> Asterisk: The Open Source PBX || Dev Conf 2PM CST FEB 24th -> IAX2/guest@66.250.68.194/996 || DOWN WITH SBC and VZ
06:37.55agave-txlinkBS sucks too
06:37.56ManxPowerQwell, A long time ago there were many, many phone companies, many cities had more than one phone comapny and they refised to talk to each other.  It was not uncommon for a household to have service from two phone companies in order to call their friends.
06:37.57Qwellterracon: scroll up a few days
06:38.09*** topic/#asterisk by bkw_ -> Asterisk: The Open Source PBX || Dev Conf 2PM CST FEB 24th -> IAX2/guest@66.250.68.194/996 || ILEC's can suck my ethernet cord!
06:38.10ManxPowerThen the govt stepped in and made AT&T out of all the little phone companies.
06:38.14Qwell2005-02-18 10:52:11 <Qwell>     terrapen: Change your nick to terraben, and you'll be first on the nick complete list. :p
06:38.23agave-txlinkuh... not entirely true
06:38.26terrapenhahah qwell
06:38.32bkw_not true boi
06:38.32agave-txlinkAT*T gobbled up the little guys by refusing to interconnect with them
06:38.34Qwellterracon: :P
06:38.37agave-txlinkforcing them into being bought
06:38.38ManxPoweragave-txlink, I'm simmarizing from memory.
06:38.47agave-txlinkyou're basically right
06:38.47terrapeni'm cool with terracon
06:38.52agave-txlinkbut really it was ATT not connecting iwth the small guys
06:38.53terrapeneven though it makes me look like a fraud
06:39.09QwellSo, where did AT&T come into play to begin with?
06:39.30ManxPowerhttp://www.scte.org/chapters/newengland/reference/Telephony/topic01.htm
06:39.34QwellI finally figured out where Verizon came from the other day, heh
06:39.40ManxPowerhttp://www.att.com/history/history3.html
06:39.58QwellManxPower: will read, thanks
06:40.12ManxPower"A year later, on July 9, 1877, the Bell Telephone Company was formed, and Alexander Graham Bell became the company's "electrician," at a salary of $3,000, and Watson became "superintendent" in charge of research and manufacturing. "
06:41.10terrapendammit, i just don't get this:
06:41.24terrapeni have a barely-used 100Mbit cnx to my provider's core switch
06:41.37terrapenping times used to be <1ms *always*
06:41.49terrapennow, for no reason at all, they jump up to 200+ms frequenty
06:41.53agave-txlinkterrapen: did you start using ip tableS?
06:41.58terrapennope :)
06:42.03terrapenusing OpenBSD
06:42.22agave-txlinkstrange
06:42.23agave-txlinkwell folks
06:42.25agave-txlinki am going to bed
06:42.26agave-txlinkthe wifey is pissed
06:42.29terrapennight man
06:42.31agave-txlinkthat i am still on the intarweb
06:42.34agave-txlinkat 12:42 AM
06:43.00terrapenhahahaha
06:43.14terrapenmy girlfriend hates the sound of the keyboard
06:43.21terrapen"WHO ARE YOU TALKING TO?"
06:43.34agave-txlinkheh
06:43.36terrapenshe knows when I IRC just by the change in typing noise
06:43.48terrapenand, for some reason, she HATES irc
06:43.53terrapenshe thinks im talking with another girl
06:44.20ManxPowerterracon, Just tell her you are talking geek talk.  Everyone knows there are no girl geeks
06:44.29terrapenhahah
06:44.44terrapenthen she does not understand why i would find this more interesting than her
06:44.47ManxPowerIMHO any girlfriend that jealous should be an ex-girlfriend, but that's just my opinion.
06:45.03terrapenwell...our situation is kind of strange right now
06:45.06terrapentechnically we are ex-
06:45.11terrapenbut we are still seeing each other
06:45.13ManxPowerterracon, She is apparently not being creative enough in getting your attention. 8-)
06:45.20terrapenhahaha
06:45.28QwellSo, how many RBOCs are left then?
06:45.38ManxPowerQwell, three?
06:45.42QwellThis says 4, but that was from 98
06:45.44ManxPowermaybe 4.
06:46.00ManxPowerVerizon, SBC, BellSouth.  Who else?
06:46.06QwellBell Atlantic, NYNEX, BellSouth, Ameritech, U S West, Pacific Telesis, and Southwestern Bell
06:46.12QwellThat was the original list of 7
06:46.40terrapenwhat happened to Mountain Bell
06:46.44terrapenwas it mountain bell?
06:46.46QwellBell Atlantic + GTE = Verizon?
06:46.48terrapenor was it western bell
06:47.01modulus_who just bought out MCI?
06:47.06ManxPowerQwell, Verizon also includes Nynex and parts of GTE
06:47.12Qwellahh
06:47.15terrapenqwell, yes
06:47.23ManxPowerterracon, Montian benn became Qwest
06:47.27terrapenah
06:47.57ManxPowerThey became the first ILEC to really try to fuck people using dry pairs for DSL.
06:48.50QwellGTE wasn't associated with Bell, was it?
06:49.04ManxPowerQwell, not that I know.
06:49.15ManxPowerThey were a sort of quasi-independent ILEC.
06:49.24ManxPowerI think they were owned by Sprint at one time.
06:50.05ManxPowerGTE's internet backbone was sold several times and I think it eventually became part of MCI's network backbone, but I could be wrong.
06:50.25Qwellsuch an odd history
06:50.43ManxPowerThe creation of Verizon included the merging of several companies, the breaking up of several companies and the merger of the resulting parts with several other companies.
06:50.56Qwell...hmm
06:51.02ManxPowerGTE was basically disected.
06:51.06modulus_was it verizon that bought out MCI?
06:51.07Qwellsomebody should do a timeline, similar to the UNIX timeline.  heh
06:51.19ManxPowermodulus_, trying to buy out MCI.
06:51.36*** part/#asterisk SuperMMan (~graphic@d209-89-191-155.abhsia.telus.net)
06:51.38Inv_arpis it a good idea to use *  format_mp3 for MOH?
06:51.46modulus_i thought the deal went through for like 6 and some odd bill?
06:51.50ManxPowerI think Sprint bought part of GTE as well (the areas Verizon didn't want)
06:52.10ManxPowermodulus_, Qwest is not out of the bidding picture yet.
06:52.34modulus_what's qwest's last bid?
06:52.42ManxPowermodulus_, go read news.com
06:52.48modulus_no
06:52.50modulus_i refuse
06:53.47ManxPowermodulus_, http://news.search.com/search?q=mci+verizon&x=0&y=0
06:54.39ManxPowerSprint is the only IXC that I know of that isn't a frakensfein of companies.  BellSouth is the same, but as the ILEC.
06:56.14modulus_manx, who do you think will come out with mCI?
06:56.16*** join/#asterisk neopher (~crazy@mail.techhelpresources.com)
06:56.30JerJeryour mom
06:56.44neopherhello everyone
06:57.02ManxPowermodulus_, No idea.
06:57.15ManxPowerBut I'm sure it will involve a bidding war and lawsuits.
06:57.24modulus_i didn't know qwest was that big
06:57.32modulus_mci is no small fry
06:57.44neopherwould anyone happen to have firmware for a cisco 30vip
06:57.57ManxPowermodulus_, Qwest covers a LARGE area of the USA.  Not a high population per sq/mile, but still.
06:58.05ManxPowerneopher, yes.
06:58.27*** join/#asterisk pascals (~248d34d6@ip503c8584.speed.planet.nl)
06:58.33modulus_hmm
06:58.56neophersweet, would you please send it, tring to get my 30vip to work with sccp
06:59.10JerJerahh Qwest, the whore of long distance
06:59.13neopherchan_scccp
06:59.35JerJercan't help you with chan_sccp, but I know my 30vip functions using chan_skinny
07:00.01ManxPowerneopher, No I will not send it.
07:00.11ManxPowerneopher, That's like asking for a MS Office license key.
07:00.14neophercool, can't get mine to work for eaither, i found out i have an old firmware image
07:00.20ManxPowerCisco wants to charge for firmware.
07:00.54neopherhmm, it's EOL and it is not on cisco's site anymore
07:01.12ManxPowerneopher, you have a CCO account that allows you to download firmware?
07:01.45neopherbut i understand, np, i'll have to email my rep there and see if they still have it
07:02.39neopheri know my cisco account allows me to get IOS for router, so i probobly could
07:02.55JerJerdownload a CCM executable and unpack it
07:03.04JerJermanually
07:03.12JerJerthen poke around for a 30vip firmware bin
07:03.48neopherhmm, call manager has it in there?
07:03.52JerJeryes
07:04.00JerJercall munger is why SCCP exists
07:04.25JerJeror find a friendly lamer that owns a CCM
07:04.34neopherdidn't know they had 30 vip firmware in there anymore, i'll go unpak
07:05.05JerJerlast i knew ccm still supported the 30vip's
07:05.09terrapendo you have to renew a CCO account every year?
07:05.09neophertnx again
07:05.11JerJerand 12sp+
07:05.14terrapenbecause i had one long ago
07:05.17terrapenmayeb it still works
07:08.52*** join/#asterisk ScythelX (Fleb@pc-24-181-176-10.sbi.ct.charter.com)
07:09.45ManxPower~rtp
07:09.46jbot[rtp] The Internet-standard protocol for the transport of real-time data, including audio and video. RTP is used in virtually all voice-over-IP architectures, for videoconferencing, media-on-demand, and other applications. A thin protocol, it supports content identification, timing reconstruction, and detection of lost packets.
07:09.48*** join/#asterisk DHuang (~DHuang@adsl-102-99.swiftdsl.com.au)
07:09.56PoincareGood morning
07:10.00DHuangmorning...
07:10.07neophergmorn
07:10.41DHuangjust wondeing if asterisk can convert difference codec on SIP connection?  ie. 1 SIP on iLIBC and 1 SIP on GSM?
07:11.18ManxPowerDHuang, Yes.  It does so by default.
07:12.26DHuangManxPower: I see. but I got this error msg.. channel.c:1734 ast_set_write_format: Unable to find a path from g729 to ilbc  (I have g729 license installed)
07:13.47ManxPowerDHuang, What is the output of "show g729"
07:14.45Inv_arpanyone use the * addon for MOH?
07:15.35DHuang<PROTECTED>
07:15.49ManxPowerDHuang, that is NOT the output of "show g729"
07:15.51*** part/#asterisk santiago (~santiago@63.245.86.121)
07:16.12DHuangManxPower: No such command 'show g729' (type 'help' for help)
07:16.18neopheri'm using music on hold
07:16.22ManxPowerDHuang, then you do NOT have the codec installed.
07:16.35ManxPowerEven if show codecs shows it.  It will show it without the codec being installed.
07:16.49DHuangOh.. :-( but I can get 2 x g729 SIP running..
07:17.04ScythelXprolly because your phones support it
07:17.05ManxPowerDHuang, that's because Asterisk is just passing thru the data.
07:17.17DHuangManxPower: how to install or make sure it's installed?
07:17.20ManxPowerDHuang, If this is a very old asterisk install maybe you have the old voiceage codec.
07:17.48ManxPowerDHuang, "show modules" should show a codec_g729.so or something similar.  format_g729.so does NOT mean you have a codec installed.
07:17.49DHuangManxPower: It's new install, from the CVS
07:18.27ManxPowervoip-1*CLI> show g729
07:18.27ManxPower0/0 encoders/decoders of 10 licensed channels are currently in use
07:18.34{zombie}DHuang: you need to purchase a g729 license then
07:18.49{zombie}it is restricted by patents, you have to pay royalties to use it
07:18.51DHuangManxPower: format_g729.so            Raw G729 data    only... Ok, I'll check the installation and check the .so file
07:18.56ManxPower"show codecs" will have
07:18.58ManxPowercodec_g729a.so            Annex A/B (floating point) G.729/PCM16 C 0
07:19.10DHuangzombie: Yes, I bought 10 license from digium
07:19.14ManxPowerDHuang, You are not using the pirate codec, are you?
07:19.42pascalsGood morning
07:19.46DHuangManxPower: no not pirate code, ran the register from digium and everthing is fine...
07:19.56ManxPowerDHuang, Didn't I just say that format_g729.so does NOT indicate you have the codec installed?
07:20.20ManxPowersorry, "show modules" will have "codec_g729a.so            Annex A/B (floating point) G.729/PCM16 C 0"
07:20.22DHuangManxPower: Thanks.. now I think where to look for now.... ie. put the .so from digirum site to the modules
07:20.26inticonnetArgh comming from a pbx backgrpund setting up a queue and setting up moh is all good but Im now trying to make an emergancy call to our isp and it just plain sucks
07:21.13pascalsI think I have a codec problem: I can answer an ISDN call with an IAX2 softphone, but I have no audio when firefly->*->misdn calls out
07:23.02DHuangManxPower: Thanks... got ti working now.. :-)
07:23.10ManxPowerDHuang, what was the problem?
07:23.30DHuangManxPower: the .so file is corrupted... replace with http://www.digium.com/downloads/ftp/asterisk/g729/glibc_2_3/pentium4/codec_g729a.so
07:24.39pascalsThe odd thing, to me anyway, is that sound quality is superb for incomming calls, but absolutely nothing happens for outgoing calls...
07:25.10*** join/#asterisk ranliv (~ranliv@210.5.85.11)
07:26.58*** join/#asterisk odie_flocon (~Odiefloco@S01060011953994ee.cg.shawcable.net)
07:27.07*** join/#asterisk neopher (~crazy@mail.techhelpresources.com)
07:29.07ManxPowerI really hope I can take a nap in the morning.
07:30.15*** join/#asterisk neopher (~crazy@mail.techhelpresources.com)
07:30.16shido6no naps
07:31.05ManxPowerWhips are fun!
07:31.16*** join/#asterisk troniz (somebody@zappy.catbert.org)
07:31.39ManxPowercatbert.org?  Cool
07:31.42troniz:)
07:31.45Inv_arppascals: same provider for outgoing? any errors in console
07:31.48tronizalso have evilphb.org too
07:31.52tronizdilbert theme obviously ;)
07:31.56*** join/#asterisk rodizump_ (~chatzilla@dsl-213-023-227-121.arcor-ip.net)
07:32.08rodizump_hi everyone
07:32.27neopherhello
07:32.37rodizump_does anybody know how to restrict the total amount of calls asterisk box can accept ?
07:32.57tronizdecided to check out asterisk some more after reading a great article on it in SAGE's ;login: magazine for this month
07:33.33rodizump_i want to set a limit of say 60 incoming SIP channels max per box
07:34.47JerJerrodizump_: Group
07:35.31*** join/#asterisk outsidefactor (barf@203-173-32-225.dyn.iinet.net.au)
07:36.02JerJershow application SetGroup
07:36.40rodizump_after the channel hangs up, does the group counter decrement automatically ? when setgroup()/chekcgroup() is used in dialplan ?
07:37.55rodizump_did anybody successfully used checkgroup/setgroup with asterisk ?
07:40.33*** join/#asterisk eipi (~eipi@40-142-89-200.fibertel.com.ar)
07:42.10JerJeri see that Willie is talkin Bio-Diesel now... hell yeah  good stuff
07:46.09terrapenwilliw is talking bio-diesel?
07:46.13terrapenerr willie
07:47.00terrapeni wonder if the Honeysuckle Rose runs bio-diesel now
07:50.45JerJeris that is bus?   yes!
07:50.51JerJerhis
07:53.40terrapenyeah
07:53.46terrapenHoneysuckle Rose
07:53.52terrapenand Rooster drove/drives it
07:53.56terrapeni've met Rooster
07:54.05terrapenhe lives in Bandera, where I used to live
07:54.21JerJerCBS Evening Snews did a report on his quest the other week
07:55.06ManxPowerI believe Willie Nelson is the father of Melissa Ethridge.
07:55.16ManxPowerlet's try that again
07:55.16*** part/#asterisk djin (~djin@gridfox.xs4all.nl)
07:55.22ManxPowerI believe Willie Nelson is the father of Melissa Ethridge's CHILD.
07:55.25terrapenhttp://www.msnbc.msn.com/id/6826994/
07:55.48terrapenhahah, i've been to Carl's Corner Truckstop
07:56.21terrapenbest quote ever
07:56.28QwellManxPower: I think it was somebody else...I saw it on TV the other day
07:56.28terrapeni just wish i could remember who said it
07:56.50terrapen"Willie Nelson was busted in Laredo, TX last week for possession of a small amount of marijuana."
07:56.51ManxPowerQwell, I saw it on VH-1 so it must be true!
07:57.15terrapen"His lawyers are fighting the charge, contending that the police did not have probable cause to search his motel room."
07:57.20modulus_vh-1 is fact
07:57.23ManxPowerAny cop that busts Willie Nelson for posession of pot is just a plain old asshole.
07:57.33terrapen"Probably cause?!?!?  How about, "HE'S WILLIE NELSON""
07:57.42terrapenerr probable
07:57.51terrapenthat was david letterman or jay leno or someone
07:57.58modulus_new gaim
07:57.59modulus_112
07:58.02QwellManxPower: It was David Crosby, heh
07:58.03modulus_err 1.1.3
07:58.10terrapenWillie Nelson is just about as american as it gets
07:58.14ManxPowerQwell, HMM?  Are you sure?
07:58.15modulus_sundays are my weekly cvsup and portupgrade
07:58.18Qwellyeah
07:58.20terrapenit will be a very, very sad day when he dies
07:58.39QwellManxPower: http://music.yahoo.com/read/news/12040513
07:59.12terrapen“I got on the computer and punched in biodiesel and found out this could be the future,†said Nelson, who now uses the fuel for his cars and tour buses.
07:59.24ManxPowerQwell, Thanks!
07:59.40Qwellsaw it on...umm...VH1. :p
07:59.46QwellI love the 90s?
08:00.00QwellI forget
08:00.15ManxPowerQwell, The show was a show about gay/lesbian rock stars.
08:00.25Qwellsounds familiar
08:00.28ManxPowerI think I have it on tape somewhere.
08:00.44Qwelldunno, just saw it a few days/weeks ago
08:00.50Qwelldays, I think...
08:00.55terrapenhttp://www.wnbiodiesel.com/Willie%20Nelson.jpg
08:00.55Qwellanyhow, off to bed
08:00.57ManxPowersex and popular culture / sex and history facinates me.
08:01.28Qwells/ and .*int/ int/g
08:01.28terrapeni saw this band last night, Cooder Graw, and they had a great song with the chorus:
08:01.40Qwells/int/faci/g
08:01.45terrapen"Don't wanna be famous...or be a star...I just want my name on Willie's guitar"
08:06.30ManxPowerThe show "Sex at 24 Frames Per Second" is a very good one.
08:06.45ManxPowerTraces the media.
08:07.43pascalsI'm having trouble connecting to pstn phones using isdn, can anyone help?
08:08.02pascalsI can make the call, but no audio.
08:08.09*** join/#asterisk djin (~marius@62.58.40.196)
08:08.15pascalsIncomming calls work flawlessly.
08:09.18*** join/#asterisk tecnico (~tecnico@user-24-236-123-31.knology.net)
08:10.29trymrtp issues?
08:10.47trymor is there no sip/rtp involved?
08:10.55pascalsiax2 clients
08:11.27pascalsand sip alike, although with sip clients, * keeps complaining about rtp problems
08:11.56pascalsSo I suspect recode problems, or something.
08:12.43pascalsI've tried forcing everything to alaw, which I gather is what ISDN uses.
08:13.27ManxPowerpascals, ISDN in the CA/USA use ulaw, but most other places use alaw.
08:13.41pascalsI am in Euroland. Netherlands
08:15.07ManxPowerpascals, or as I like to call it "alaw land
08:15.22pascals:)
08:17.03pascalsCan it be that recoding isn't working?
08:18.16pascalsThe softphone itself is working, I can call another softphone, talk, run the * echotest, etc.
08:18.34ManxPowerThis is the time of night that I stop helping people and simply wax philosophically about various topics.
08:19.34ManxPower~docs
08:19.35jbotDocumentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org
08:19.50pascalsYes, I know all those
08:20.00ManxPowerlook for NAT related stuff, and codec related stuff.
08:20.49*** join/#asterisk DHuang (~DHuang@adsl-102-99.swiftdsl.com.au)
08:21.11*** join/#asterisk abatista (~Ariel@dsl-20-177.cofs.net)
08:21.41DHuangManxPower: Do you know how to check if the voicemail is connected to the SQL?
08:22.17ManxPowerDHuang, That's a CVS-HEAD only thing.  I won't know anything about it until I start using the next stable release of Asterisk
08:22.49DHuangManxPower: I see.. thanks.. ;-)
08:23.59visik7is there something to use * as a video-entryphone, I would like to replace my home PBX with * but my home pbx has 3 video entryphone and 2 external camera
08:26.51*** join/#asterisk Firestrm (F1r3570rm@S010600047577bccd.gv.shawcable.net)
08:27.02Firestrmhello..
08:28.16Firestrmis it me and my bad luck, or are sipura adaptors total crap?
08:28.41Firestrmi cant get the $#)@#@_#(! thing to work properly..
08:30.04JerJersipura's are pretty damn good for being SIP devices
08:30.23Firestrmthe dang PSTN portion of the thing echos's badly
08:30.32*** join/#asterisk scanna (~scannachi@81-174-16-211.f5.ngi.it)
08:31.00Firestrmthe fxs-> sip works great.. but the sip->PSTN protion blows
08:31.38Firestrmive tried everything i can think of.. still bad audio on pstn..
08:32.37Firestrmi was thinking of using them for a project to bring in remote 1B lines, but im glad i tested one first.. that would have been an embarrasing and expensive mistake..
08:33.41Firestrmand Sipura tech support had been less than useful.. they wont even answer my emails..
08:41.49*** join/#asterisk Guest^DJ (some@211.24.146.10)
08:42.14*** join/#asterisk microlab (~leichangs@203.88.33.179)
08:44.30*** join/#asterisk pashah (~pashah@relay.patentica.com)
08:48.12*** join/#asterisk eivindtr (~Eivind@193.91.146.34)
08:53.27*** join/#asterisk zoa (~zoa@pirus.securax.be)
08:56.22*** join/#asterisk DEVILoper (~x@202.5.145.50)
08:56.38DEVILoperHi All
08:56.52inticonnetwats with the sudden influx of people :P
08:57.24Firestrmnobody here but us lurkers ;)
08:57.33DEVILoperMy Zaptel Card unable to detect call hangups. any help ??
08:57.42JerJershow processlist;   in the mysql shell
08:57.59md99can someone tell me what txgain and rxgain in capi.conf mean - mine by default is 0.8 which is a unit of something?
08:57.59DEVILoperZAptel=FXO
08:58.19FirestrmDEVILoper, i have the same problem.. in my case, ive tracked it down to how my telco provider handles hangup notification
08:59.24*** part/#asterisk DHuang (~DHuang@adsl-102-99.swiftdsl.com.au)
08:59.25DEVILoperis there any way to check what signalling is provided by Telco (Kewl start,loop start or Ground start ??)
08:59.51*** join/#asterisk welby (~welby@80-192-119-210.cable.ubr04.dund.blueyonder.co.uk)
09:01.04FirestrmDEVILoper, Use Kewl, i have yet to see a case where loop or ground makes a difference.. some telco's will be nice and provide a polarity reverse at hangup, which zap will detect, but in most cases telco responds to hangup by removing loop power after 30 sec or so.. not good for us..
09:01.59FirestrmDEVILoper, its not even ZAP in this case, my P.O.S SPA-3000 reponds exactly the same way..
09:03.11*** join/#asterisk Verliba (~Miranda@250.1-14-84.ripe.coltfrance.com)
09:04.13JerJerFirestrm:  read   http://www.voip-info.org/wiki-Sipura+3000
09:04.50*** join/#asterisk Delvar (~irc@83.146.53.34)
09:06.05DEVILoperhttp://www.voip-info.org/tiki-pagehistory.php?page=Asterisk+Disconnect+Supervision&diff=3 right now readng this article
09:06.16DEVILoperhelpful i hope so
09:07.16FirestrmJerJer, thats the document i used to set up it up in the first place, its working, but the audio quality on incoming calls to the PSTN interface it terrable.. low level audio, echo, hard to understand..
09:07.22terrapenholy shit
09:07.31terrapensure is late
09:07.35modulus_yup
09:07.37modulus_1.07am here
09:07.44modulus_imma bout to hit the pool hall
09:07.44terrapen3:07
09:07.48terrapenand i have work in the morning
09:07.52modulus_free coffee at pool hall
09:07.53terrapenwhere r u mod
09:07.56modulus_LA
09:08.05terrapenah
09:08.08terrapenpool hall in LA?
09:08.17terrapennever saw much of that when i lived there
09:08.38*** join/#asterisk netsurfer (~bbjunkie@dreambox.myvnc.com)
09:08.55JerJerFirestrm:  I had the low audio problem until i brought the gains to 0
09:09.02JerJerthey defaulted to -3 db
09:09.04modulus_terrapen there's lots of pool halls
09:09.13terrapeni lived in pasadena
09:09.17terrapenwhich is probably my problem
09:09.20JerJerperhaps you have a backwards ring and tip?
09:09.31modulus_terrapen, nothing in pasadena is open after 6pm
09:09.36modulus_except gas stations
09:09.36terrapenheh
09:09.39terrapenso true
09:09.42modulus_manned by iraqis and other middle easterners
09:09.44terrapenit was *miserable*
09:09.54terrapeni really wanted to move back home to texas or to utah
09:09.57terrapenand ended up doing so
09:10.10FirestrmJerJer, tried that.. it helped, but still getting bizzare echo, not there all the time, just occasionally it punches through, like the echocanceller is not working properly or cant lock on.
09:10.11modulus_i'm downtown
09:10.26terrapenever go to Antone's (sp?)
09:10.32terrapenhome of the french dip sandwich?
09:11.32terrapenwow its humid outside
09:11.33terrapen95%
09:11.39FirestrmDEVILoper, good link... good general info on disconnect supervision.. unfotunatly my telco doesnt offer it on residential, im im stuffed on the home line..
09:11.43terrapenthick fog
09:12.19Firestrmwe have been getting california weather rather than big dumps of rain we usually get for winter weather..
09:12.26*** join/#asterisk pif (ldm@zenon.apartia.fr)
09:14.02modulus_** Listing the failed packages (*:skipped / !:failed)
09:14.02modulus_<PROTECTED>
09:14.18modulus_ugh
09:14.22modulus_stupid mplayer-skins
09:14.22Firestrmyikes..
09:14.30modulus_they never maintain the portstree
09:17.17modulus_i don't even use mplayer-skins
09:17.22modulus_pkg_deinstall -f
09:19.25Delvaris it morning?
09:19.32*** join/#asterisk Fabe_ (~spamhere@pD95B0BF7.dip0.t-ipconnect.de)
09:19.46Firestrmit is here..
09:19.55eivindtrWhen does morning end?
09:21.36*** join/#asterisk RoyK (~roy@80.239.107.80)
09:23.01*** join/#asterisk ckruetze (~ckruetze@i3ED61A56.versanet.de)
09:23.30pascalsI'm having trouble connecting to pstn phones using isdn, incomming calls have audio, outgoing calls do not.
09:23.52RoyKpascals: using what sort of hardware?
09:24.04pascalsberonet 4port HFC-4S card
09:24.23visik7pascals capi or zaphfc ?
09:24.28pascalsmisdn
09:25.00RoyKiirc misdn has several issues
09:25.16RoyKis that a 4port BRI?
09:25.18Poincarebetter try the zaphfc
09:25.22pascalszaphfc had problems too, with a passive card
09:25.30pascalsYes, 4 port bri
09:25.38RoyKpascals: zaphfc only work on passive BRIs
09:25.44pascalsDoes that work with the zaphfc driver?!?
09:25.47Poincareif you're using a beronet card better try the zaphfc
09:25.50RoyKit was originally designed for the HFC-PCI driver
09:26.31pascalsAh, didn't know that.
09:26.53pascalsI thought zaphfc was for passive cards only
09:27.09RoyK<pascals> zaphfc had problems too, with a passive card
09:27.10RoyK?
09:27.41pascalsI could not dial out with that card, it complained about not being connected or something
09:27.41RoyKpascals: try emailing the list, asking Klaus Peter Junghanns about it
09:28.37pascalsThe asterisk-users digium list?
09:29.45pascalsAh. bristuff eq zaphfc... I have the zaphfc driver installed...
09:29.54pascalsNot in use for this card, though.
09:32.29inticonnetSYSTEM RUNTIME (TASKAGENT9k)
09:32.43inticonnetargh sorry i ran out of mouse cable and had to drop what i was draging :P
09:39.04eipithere's anyway to redir musiconhold to uncompressed audio from /dev/radio?
09:51.11pifhi, can I identify a call coming in from, say, zap/13 in extensions.conf ? what variable should I look at?
09:52.31VoIPMastaDoes anyone have some experience with Areski-CC?
09:52.58JerJerlol
09:53.03RoyKnever heard of it :)
09:53.12JerJerits on the wiki
09:53.16*** join/#asterisk meppl (~mephisto@p3E9E2F75.dip.t-dialin.net)
09:53.21JerJerits a joke
09:53.27JerJeri'm just waiting for the punch line
09:53.29zoai agree with jj
09:53.34zoapunch me too
09:53.37VoIPMastawhat's a joke?
09:53.51JerJerAreski-CC
09:53.57VoIPMastamay I ask why?
09:54.11JerJerhorible implemenation
09:54.17JerJerhalf thought through
09:54.18*** join/#asterisk Zeeek (~icechat5@Zeeek.sustaining.supporter.pdpc)
09:54.24VoIPMastawhich prepaid calling card system would you recommend JerJer?
09:54.32JerJeri am biased
09:54.59VoIPMastaI was trying astcc, developed by Mark, however it doesn't detect hangups
09:55.13JerJerum no
09:55.19JerJeryour channel isn't detecting hangups then
09:55.29VoIPMastaany special params to consider?
09:56.32VoIPMastahttp://pastebin.ca/6231 <-- There's my zapata.conf
09:59.30JerJerdon't depend on analog to always detect a hangup
09:59.38JerJeresp if you are in some crazy country
09:59.44VoIPMastaI'm in Mx
09:59.51JerJermucho crazy
09:59.59VoIPMastaI know
10:00.05VoIPMastabut how can I detect hangups?
10:01.39JerJerdoes your telco line have disconnect supervision on it ?
10:02.14VoIPMastanot that I'm aware of
10:02.26JerJerthen asterisk will never hang that line up
10:02.30JerJernot just astcc
10:02.48JerJercomplain to your telco you need Disconnect supervision on your line
10:03.44VoIPMastaI'm not even entire sure if they offer it here in mx
10:04.00qwerpharlo..
10:04.22qwerpis there anyway i can block only 15 incoming line on a PRI line?
10:05.43eipican i integrate sox with musiconhold?
10:06.00VoIPMastaJerJer: I'm just doing some research and it seems like every line here in mexico has disconnect supervision enabled
10:07.10pifphone packed in opium cardboard?
10:11.19RoyKpif: :P
10:18.59*** join/#asterisk ezabi (~ezabi@82.201.233.198)
10:20.08ezabihi everbody, the usual question, any codec recommendations?
10:20.45zoagrrr
10:20.51zoathere is no best codec
10:21.00zoathere is just a best codec for your situation
10:21.22ezabii prefer gsm but was told that g723 is best for bandwidth consumption
10:22.25ezabithe point is i should standardize the codec all over the different sites
10:22.54zoayou cannot do g729
10:22.55zoaeuh
10:22.57zoag723
10:23.15zoag723 sounds robotic btw
10:23.55RoyKezabi: how much bandwidth can you use?
10:23.58eye69What does DID stand for?
10:24.11RoyK~did
10:24.12jbotdid is, like, Direct Inward Dialing
10:25.31ezabiRoyK: the lesser the better of course because there are some site on rural areas with low bandwidth, as low as 4k
10:25.51eye69Thanks.
10:31.47*** join/#asterisk zeedo (~zeedo@www.bsrf.org.uk)
10:34.27*** join/#asterisk Slothbag_ (nerf@203-206-241-47.dyn.iinet.net.au)
10:34.54*** join/#asterisk visik7 (~ciao@host38-39.pool80182.interbusiness.it)
10:36.20*** join/#asterisk CleanerX (~nix@nat-ph3-wh.rz.uni-karlsruhe.de)
10:36.29RoyKezabi: see http://www.newport-networks.com/pages/voip-bandwidth-calculator.html for bandwidth calculation
10:37.01RoyKezabi: you don't get any lower than around 20kbps full duplex with asterisk
10:38.05*** join/#asterisk r1 (~erwan@www.thiscow.com)
10:38.24*** join/#asterisk jofa (~jofa@a80-127-56-82.adsl.xs4all.nl)
10:43.58*** join/#asterisk tafazzi (~Dario@eporroj0-1.customer.nettuno.it)
10:45.20*** join/#asterisk christo (~chris@office.enovi.com)
10:46.13jofai'm using an old dial-plan with a newer asterisk. It now spontaniously jumps into the h(angup) extension right after a goto, did something change there recently?
10:46.25jofait does this on sip zap and capi channels :-/
10:48.34ezabiRoyK: thx
10:53.48*** join/#asterisk r0d3nt|m (~RatMan@wsip-24-234-241-84.lv.lv.cox.net)
10:54.02*** join/#asterisk goof (~goof@81.199.100.163)
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10:54.20[ro]nic3tryre all
10:55.21[ro]nic3tryhow do I set a maximum number of calls that asterisk will handle ?
10:56.02*** join/#asterisk sysdef (~sysdef@pD9561E44.dip.t-dialin.net)
10:56.38goofin etentions this used to work before: "exten => _096.,1,Dial,Zap/g1/75${EXTEN}|20" , but now it does not. Any idea
10:59.13*** join/#asterisk doolph (doolph@200.46.148.46)
10:59.15doolphhello
10:59.34doolphanyone know how to setup a E1 to terminate minutes
11:00.01tessierdoolph: That's like asking if anyone knows how to build a nuclear powered aircraft carrier
11:00.12tessierThe answer is yes, but it is not going to be explained to you today over IRC
11:00.31doolphheh
11:00.36doolphis that too complicated?
11:00.50tessierI assume you want to bill for these minutes?
11:00.53doolphi though it was buy some hardwards and set it ups
11:01.02jofadoes asterisk (cvs) build with gcc 2.95.4? or does it require a 3?
11:01.05doolphreally it is not necesary
11:01.20tessierAsterisk based business plan: 1. buy hardware 2. set it up 3. ??? 4. PROFIT!
11:01.21doolphbecause i will have an external billing server
11:01.37doolphwell tell me what do i need
11:01.41[ro]nic3tryhow do I set a maximum number of calls that asterisk will handle ?
11:02.07*** join/#asterisk Mike_TK (~Mike_TK@212.165.78.5)
11:02.08JerJershow application SetGroup
11:02.20JerJer1 Sell T1 Boards
11:02.26JerJer2 ????
11:02.32JerJer3 take over the world
11:02.33doolphJerJer?
11:02.41doolphto who
11:03.02Slothbag_can anyone help with a relatively easy public asterisk/nat'ed clients setup??
11:03.19JerJerusing IAX, it is painless
11:03.35Slothbag_clients are budgetone 100's
11:03.37Slothbag_:(
11:03.45ezabiIAX is best behind nat
11:03.49JerJerprepare for heartburn and hairloss
11:04.14jofaslothbag: just make sure * is not natted, it works fine here..
11:04.18ezabifor budgetone u usualy have to provide a public stun
11:04.31JerJerand in extreeme cases, diarrhea
11:04.40JerJerezabi: um no
11:04.43JerJerstun is a joke
11:04.54JerJertell the device to process nat
11:05.00Slothbag_my asterisk is not natted.. but the budgetone is behind a nasty unconfigurable NAT firewall
11:05.15JerJerset nat=yes in the approprate spot in sip.conf
11:05.20Slothbag_it can register etc, but no audio makes it
11:05.28JerJerand register
11:05.55ezabiJerJer: still even i configured the device for natting, i used the vovida public stun and it worked fine
11:06.10doolphwhat can i do with
11:06.11doolphDigium Wildcard E100P - Single E1 PCI card - SIP IAX H.323 Asterisk
11:06.12Mike_TKSlothbag_: Maybe you have some 'smart' sip aware box between?
11:06.29JerJerMike_TK:  no need
11:06.37JerJerjust a stateful router
11:06.52ezabiwith sip behind nat the problem is usualy with rtp
11:06.53JerJerwhich any even semi-current device can do
11:06.54*** join/#asterisk cjk (~cjk@80.92.75.91)
11:07.07JerJerezabi: nat=yes minimizes this problem
11:07.09Slothbag_i was thinking just install a SIP proxy like SER on the asterisk machine
11:07.25JerJerand registering keeps the udp path thru the NAT open
11:07.57Slothbag_similar to how FWD do it
11:07.58Mike_TKJerJer: No, I mean ofthen this nat box is 'very smart' and broke a SIP messages. I face with this problem sometimes.
11:07.59JerJerum no, that is not the answer
11:08.25tessierSER is not a bad idea but you don't want it on the asterisk machine
11:08.29JerJerfwd does not use asterisk and ser together, they are separate
11:09.11Slothbag_yeah, i think thats my problem.. i couldn't get it to work on the one machine
11:09.29JerJeryou dont want/need it on the same machine
11:09.29Mike_TKtessier: I had configuration with ser and asterisk on same box without any problems.
11:09.34tessierThey would both want to listen on 5060 for one thing
11:09.50tessiermikegrb: I'm sure did. That's not to say it isn't the best way to set up a voip network though.
11:10.06Slothbag_but i only have the one machine..
11:10.12JerJerinteresting... it just lightning and thundered here
11:10.24JerJeralmost scared the shit right out of me
11:10.26tessierThen you can probably do without SER and just use asterisk as your network isn't big enough to really ned it.
11:10.34tessierIt's raining pretty good here in San Diego.
11:10.43tessierWe're getting plenty of weather this year. It's a nice change from draught.
11:10.51doolphhey
11:10.52JerJerwe've gotten over a foot of snow since like noon yesterday
11:11.12Slothbag_but the client (over the net) cant get rtp over his NAT without a proxy
11:11.20JerJerand still comin down, hard
11:11.25doolphis that isdn/pri like the fx0 line that come from my isp, and i need to connect it to E100
11:11.30JerJerSlothbag_:  then upgrade your edge device
11:11.36JerJeryou do not need a proxy locally
11:12.05*** join/#asterisk Mike_TK (~Mike_TK@212.165.78.5)
11:12.12JerJerif your edge device has stateful inspection of the packets, it will just magically work
11:12.32Slothbag_yeah, that would be ideal.. but im trying to remove the complexities from the clients and make it easy for them to setup
11:12.35cjkhi, is the realtime module dead or is it just working great. im asking because i see no activity
11:12.39JerJerperhaps a simple flash upgrade is all you need
11:12.51JerJerSlothbag_:  SIP is not going to do that for you
11:12.57Slothbag_hehe
11:13.11Slothbag_when is someone gonna write a IAX firmware for the BT100 :))
11:13.33JerJerright after the iax firmware for the spa 3k comes out
11:13.38Delvarcjk: realtime seems to be working fine a the minute
11:13.59JerJerrealtime is so evil
11:14.24cjk2 different opinions. JerJer why?
11:14.44JerJerwhy force asterisk to depend on a database to operate?
11:15.11ezabino iax devices yet except the IAXy???
11:15.20cjkJerJer, scalability
11:15.52JerJerasterisk doesn't have to be forced to depend on the database
11:15.54JerJerto scale
11:15.59*** join/#asterisk Zeeek (~icechat5@Zeeek.sustaining.supporter.pdpc)
11:16.12cjkJerJer, ok except on the opinion on how asterisk should operate. do you know if its stable
11:16.15pascalsManageability is the correct term, I think
11:16.28cjkJerJer, yep i know it hasnt. but it makes things so much easyer
11:16.40JerJerum I use a database
11:16.40tessierEvery other phone system requires a database. Why not asterisk?
11:16.43*** join/#asterisk The_Ball (~alex@203.221.68.29)
11:17.09JerJernotice I did not say to NOT use a database
11:17.26The_BallHow do i know which channel is which on a TM100 card? Is there a scan utility?
11:17.28JerJeri said to force asterisk to DEPEND on the database
11:17.32cjkJerJer, you would prefer something like odbc?
11:17.37JerJerhell no
11:17.38tessierI think most systems depend on their db's too
11:17.54cjkJerJer, ok you mean if mysql is down *  is down
11:17.58JerJertessier: which is a major contributing factor why they suck
11:18.40tessierI believe realtime has a couple of modes of operation doesn't it?
11:18.51JerJerno idea
11:18.57tessierOne where it really is real time and another where it checks the db on reloads and restarts only?
11:19.05JerJeri was disgusted on the first cvs commit
11:19.12tessierNot sure but I seem to recall reading something like that on the realtime page in the wiki
11:19.27tessierI think it's good to have a standard schema for asterisk regardless
11:22.42ezabiok, so does anyone have any idea how the digium directory is done on the dial plan, i mean to dial the first three letters of the last name and it gets back to u
11:23.26The_Ballwhich module should i load for a TM100 card? zaptel?
11:24.00JerJerexten => 1234,1,Directory(contex_in_voicemail_dot_conf)
11:24.15JerJerThe_Ball:  wctdm after you have configured zaptel.conf
11:24.31The_Ballokey, thanks
11:24.42Zeeek.
11:24.59doolphtessier: with PRI and a E1 card can I terminate minutes right
11:25.17*** join/#asterisk brandao (~brandao@200-206-135-147.dsl.telesp.net.br)
11:26.30brandaoHi guys. Please, how can I make dial command at console work again?
11:26.32brandao*CLI> dial
11:26.32brandaoNo such command 'dial' (type 'help' for help)
11:26.51*** join/#asterisk UPMeduardo (~UPMeduard@tauro2.dit.upm.es)
11:26.51JerJeryou have to have a sound card channel driver loaded
11:26.53JerJerchan_oss
11:26.54JerJeror
11:26.56JerJerchan_alsa
11:27.03JerJerdepending on the sound subsystem you have going
11:31.24brandaotks!
11:32.17JerJerthank you, drive-thru
11:32.52JerJerdon't forget to beat your serving wench
11:39.26*** join/#asterisk gdh (foobar@213-2-2-26.uk.vianw.net)
11:40.21gdhmorn'
11:44.19eipijerjer, i think that i have resolved my weird radio system
11:44.37eipiwith shoutcast
11:45.09*** join/#asterisk jerlique (jerlique@lnk254.adl0.adsl.esc.net.au)
11:46.30*** join/#asterisk pranav (~dawda_pra@202.149.48.196)
11:50.12jerliqueanyone have anything to say about various channel banks?
11:50.14*** join/#asterisk visik7 (~ciao@host178-39.pool80182.interbusiness.it)
11:50.48*** join/#asterisk sambal (~sambal@gateway.office.flatbox.nl)
11:50.57*** join/#asterisk Mw3 (mw3@195.56.193.13)
11:51.38Delvarnice asterisk behavior, sip entity set to type=peer can make inbound (from client to asterisk) calles, but type=user gets droped into the default context, whats going on, i thought it was suposed to be the other way round? :)
11:52.01pranavhello everyone
11:52.40pranavif i want to connect to another server which is a sip, how to connect it from asterisk
11:53.12brandaothere is no sound card at the machine :(, so no dial command at console?
11:54.12Delvarbrandao: use a .call file
11:55.53pranavwhat to add in the extensions.conf
11:56.19[ro]nic3tryHELP ..i set my asterisk to use only G729, calls works fine, but sounds (like demo-info) doesnt work anymore ... why ?
11:56.43pranavdo we need to add something in sip.conf as well
11:57.13The_Ballis there a sample setup, or guide to setup a simple TDM100B card with asterisk?
11:57.24Delvar[ro]nic3try: do you have licances installed on that server?
11:57.36eipipranav: in extensions.conf exten => 222, 1, Dial(SIP/........,,r)
11:59.04[ro]nic3trynope.. just asterik
11:59.29pranavok i.e the ipadress i place of ....
11:59.47[ro]nic3tryshould i need one ? how do i get one ?
12:06.57Delvar[ro]nic3try: YES you need licances to run g729 on asterisk, look at asterisk.org or digium.com
12:08.09*** join/#asterisk brazil (~cleber@200.198.105.37)
12:08.26*** join/#asterisk oej (~oej@ua-213-115-215-100.cust.bredbandsbolaget.se)
12:08.39brazilgood morning all?
12:08.54oejGood afternoon from Stockholm
12:09.20gdhGood day :)
12:09.46*** join/#asterisk muesli (~muesli@mail.muehlhaeuser.de)
12:10.57brandaoDelvar: the .call solved my problems!
12:11.18brazil:)
12:11.42CMikeoej: Good Morning... ?   I thought you were awake :)
12:13.22*** join/#asterisk e3eli3h (~e3eli3h@static-np1-5.cytanet.com.cy)
12:16.53gooffolks, I had the following working some time back, now when I dial, * ignoes the "75" prefix
12:17.08goofexten => _096.,1,Dial,Zap/g1/75${EXTEN}|20" ,Any idea
12:18.06goofif i put the 75 at the beginning like _75096., it workd but I want save the user having to dial all that
12:31.18*** join/#asterisk meppl (~mephisto@p3E9E2F75.dip.t-dialin.net)
12:38.27The_Ballztcfg -vvv shows two channels 1 - FXO and 2 - FXS and no errors, where do I get asterisk to bind to the FXS and provide a dial tone?
12:39.20*** join/#asterisk Zeeek (~icechat5@Zeeek.sustaining.supporter.pdpc)
12:44.08Zeeekk
12:46.43Zeeek... .... .. _
12:47.53[ro]nic3tryhas anyone ever use Cisco Sip Proxy ?
12:49.01*** join/#asterisk ezabi (~ezabi@82.201.232.190)
12:50.35zoai have a copy of it here
12:50.38zoabut never tried it
12:51.25Zeeek.
12:51.48[ro]nic3try.. aha.. i'm having a litle problm with it.. i'm looking for help :)
12:51.56Zeeekcoffee anyone? I'm going to get one
12:52.19djincount me in, Zeeek.
12:52.20[ro]nic3tryoh.. but please :D
12:54.56Zeeekthe decaf is for who?
12:55.17gdhwrong room, try #pointless
12:55.37gdhbbl =)
12:56.13ZeeekI take it you don't want any gdh?
12:58.19ZeeekI keep losing the connex
13:07.56*** join/#asterisk pointer-gaim (~pointer@router.cathey.us)
13:10.27*** join/#asterisk TheEmperor (TheEmperor@218.111.48.121)
13:10.36Zeeekhi djin
13:10.44*** join/#asterisk jedirl (~fdsafasdf@213.162.200.226)
13:10.45jedirlHello
13:11.04jedirlWhich H.323 channel for asterisk is better ?
13:13.03brazilH.323 is the most used signaling for any device in the world however SIP is yang...
13:13.52*** join/#asterisk bjohnson (~bjohnson@jecinc.tor.istop.com)
13:14.08ezabibrazil: still iax is best
13:14.26jedirlI mean which one of the implementations available for asterisk is better
13:19.20braziltks ;)
13:20.25Zeeekheh
13:24.08*** join/#asterisk Manipura (~chatzilla@dsl-ep-209-115-250-i114-cgy.nucleus.com)
13:25.02*** join/#asterisk SeaForth (~SeaForth@c-24-1-126-202.client.comcast.net)
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13:26.20*** part/#asterisk e3eli3h (~e3eli3h@static-np1-5.cytanet.com.cy)
13:28.34*** join/#asterisk gambolputty (~gambolput@cblmdm65-221-51-80.buckeye-express.com)
13:31.08brazilPeople.. What kind of QoS implementation do you use toghether asterisk?
13:32.12jedirlpacketshaper :)
13:34.01The_Ballis channels FXS's only or is FXO's also channels?
13:35.20zoaWhiii
13:35.21zoa:p
13:35.49jedirlzoa: that involved any codec translation?
13:36.09zoaare you crazy?
13:36.32jedirli'm just asking
13:37.55abatistazoa, ok please explain about how you got 1600 simultaneous calls on a asterisk?
13:38.06ariel_morning everyone
13:40.32braziljedirl: expensive solution ;) !!
13:40.38Zeeekaout
13:40.49ezabizoa: can you please ellaborate
13:41.45ezabizoa: state hardware, protocol, codec, and cafeine type
13:42.11Zeeekespecially cafeine type
13:42.36MocI used apple juice to make my channel driver
13:42.51bjohnsonThe_Ball: channel is a term that generally refers to anyway into or out of asterisk
13:43.33ManipuraAnyone know where I can find more info on mysql realtime other than the wiki?
13:43.36bjohnsonThe_Ball: it can be a hardware device, a SIP account, an IAX account, or some other things
13:44.51The_Ballaha, you see im have a phone connected to the FXS and I get a dial tone, but I would like to play the demo which is in extentions.conf, but I haven't got my terms straight yet
13:44.59bjohnsonManipura: try again later.  The "experts" aren't in yet
13:45.08Manipuraah
13:45.28bjohnsonThe_Ball: try dialing 500.  I think that it is setup by default
13:45.58bjohnsonyou may have to config the fxs to talk to * though (and config * to accept the call)
13:46.06ariel_The_Ball, if you did make samples you can dial 500 it will try to call the digium site over iax2
13:46.18The_Ballit works!! yey
13:48.59The_Ballthat's amazing quality over this crappy dial up!!!
13:49.31brazilanyone had idea about the best QoS implementation for Asterisk? HTB, SFQ, etc?
13:49.38*** join/#asterisk planet_guru (~chris@office.enovi.com)
13:49.48ariel_The_Ball, dialup????? oh boy well the demo  is pre set for gsm
13:51.38ariel_brazil, I use asterisk mainly in small biz behind wrt54g and turn on the service on there web interface. But I have done a few via m0n0wall and it has options as well for ports that you give priority
13:52.25The_Ballariel_, i won't actually call someone will i? on the demo server?
13:52.53ariel_The_Ball, yes if you use 500 it actually goes out to digium's site
13:52.54Zeeekcall an 800 number thru IAXTEL
13:53.01*** join/#asterisk didz_ (didz_@200.218.192.52)
13:53.38*** join/#asterisk coppice (~chatzilla@245.195.17.210.dyn.pacific.net.hk)
13:53.42The_Ballariel_, yes, i understand it goes to their server, but is that just a test server, or if I select sales, will I actually call sales?
13:53.54ariel_yes
13:53.56Zeeektry calling support - no danger there :)
13:54.04ZeeekJOKE
13:55.17ezabigo into the directory and try calling mark, he never actually answers
13:55.17*** join/#asterisk e3eli3h (~e3eli3h@static-np1-5.cytanet.com.cy)
13:59.18*** join/#asterisk cervajs (~cervajs@cervajs.fpf.slu.cz)
13:59.45brazilAriel: Tks very much!
14:01.16*** join/#asterisk Nix (~Nix@dsl81-214-9283.adsl.ttnet.net.tr)
14:01.27bjohnsonbrazil: ipcop does QOS too (through standard linux tools)
14:01.56bjohnsonbrazil: higher end switches will also do QoS with VLANs
14:02.28brazilhmm, good... I looking for!! tks
14:03.51ariel_bjohnson, gsm is pretty good but it really depends on the b/w you have. ulaw being best and the only one you can use for faxes.
14:05.53bjohnsonwe don't do faxes over IP and currently I'm using ulaw .. but we only typically have one voip call at a time right now while in testing phase
14:08.14ariel_bjohnson, then go ahead and try it.  In most of the setups I have done we use gsm for most of our connections between asterisk boxes.
14:09.00ariel_I also use xlite and it's able to use gsm which if great for when your on the road.
14:09.37*** join/#asterisk zeedo (~zeedo@www.bsrf.org.uk)
14:09.39[ro]nic3tryhow do i install asterisk header files ?
14:09.54bjohnsonany idea what the gsm codec is listed as on a SPA 2000?
14:12.09bjohnsonit lists g711u (ulaw), g723, g726, and g729a
14:12.18ariel_bjohnson, it's not
14:13.48ariel_I guess you could try g726 it's about the same as gsm in size the one included with asterisk the -32
14:14.43greg_workanyone know if AlarmReceiver() would work over a voip line?
14:14.46bjohnsonok.  thnx.
14:14.52*** join/#asterisk nicox (~nicox@83-64-42-210.prater.xdsl-line.inode.at)
14:15.02nicoxhello !
14:15.16nicoxdoes anybody tested ser with asterisk?
14:15.25ManxPower~doc
14:15.29ManxPower~docs
14:15.30jbotDocumentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org
14:15.33bjohnsonI'm trying to figure out how I can do some testing on codecs without affecting the office lines or my wife :)
14:15.40ariel_nicox, some people do test it but most here just use asterisk along
14:16.08ariel_bjohnson, do you have a spare old pc around?
14:16.20bjohnsonyes, why?
14:17.02zoawhy would you want to do testing on codecs ?
14:17.05ariel_bjohnson, make your test asterisk box.
14:17.26ariel_zoa, you never told us about your 1600 calls?
14:17.35zoahey
14:17.38zoajust did that
14:17.43zoanow trying to go to 15000 :)
14:17.59bjohnsonzoa: I have a few issues that may be related to bandwidth.  Also, when out of testing, there will likely be more external voip calls so bandwidth will continue to be an issue (currently everything is ulaw)
14:18.03*** join/#asterisk scardinal (~supreme@port816.ds1-suoe.adsl.cybercity.dk)
14:18.05*** join/#asterisk yashax (~yasha_x@69.15.218.218)
14:19.02ariel_zoa, we still want to under stand how you can get 1600 on asterisk now your after 15,000 ????? oh boy.
14:20.33*** join/#asterisk SeaForth (~SeaForth@c-24-1-126-202.client.comcast.net)
14:20.36Zeeekhe exceeded his test data
14:21.05[ro]nic3tryhow do i install asterisk header files ? pls
14:21.16*** join/#asterisk ToyMan (~stuq@204-8-82-238.webjogger.net)
14:22.05ariel_[ro]nic3try, what do you call header files?
14:22.36RoyK[ro]nic3try: make install :P
14:22.38[ro]nic3try*.h
14:23.02ariel_all the files if you get the cvs should be in /usr/src/asterisk
14:23.53[ro]nic3try.. i'm tring to install g729.. so i need these files ..
14:26.59*** join/#asterisk ArkyLady (ArkyLady@206.255.93.95)
14:27.50ArkyLadyanyone know of a good virtual server host? I want to be able to do my own DNS, etc
14:29.24ariel_ArkyLady, I am using www.race.com
14:29.33ArkyLadythanks, I'll check it out :)
14:30.16*** join/#asterisk montoya (montoya@200.195.87.230)
14:30.42ManxPower[ro]nic3try, I don't thibk anyone will help you install the non-digium unlicensed G729 codec.
14:32.12*** join/#asterisk Luhiwu (~marsosa@200.63.89.248)
14:32.24tzangerManxPower: got that right
14:32.47bjohnsondoes FWD offer voicemail?  I see that sipphone does
14:33.07coppiceG.729 warez edition
14:33.13*** join/#asterisk pif (ldm@zenon.apartia.fr)
14:33.40tzangerhahaha
14:34.10tzangercoppice: I am hoping to have some time to go back over the zaptel code and basically start a year ago and incrementally try zaptel drivers until the fax stops working again
14:34.16tzangerand see if the CPU load spikes occur then too
14:35.16coppicethat problem is causing pain for me too. I wonder nobody at Digium seems interested
14:35.16ezabizoa: still haven't told us about ur cafeine configuration
14:35.40zoahehe
14:35.41zoai wont
14:35.44zoacompany secret
14:35.45zoa:p
14:35.47zoano its not
14:35.53zoaits just nothing special to talk about
14:36.39*** join/#asterisk multrix (~chatzilla@ALyon-110-1-24-35.w81-51.abo.wanadoo.fr)
14:37.58ariel_zoa, 1600 calls on asterisk at the same time is special at least in my view it is.
14:39.19ezabizoa: well it is special, of course u r not using only one machine for this
14:39.30*** join/#asterisk brc_ (~brian@brc.base.supporter.pdpc)
14:40.31bjohnsonit could be used as a benchmark for everybody who asks "Can it handle x number of users?"
14:40.38Hmmhesaysis there a cheap easy way to ring all phones in a context?
14:40.39jedirlis there a warez-edition of G.729? hahaha
14:40.46Hmmhesaysyes there is
14:40.50Hmmhesaysg.723 too
14:40.53bjohnsonwith the consisitent follow up question "What hardware would I need?"
14:41.07*** join/#asterisk PoWeRKiLL (~PoWeRKiLL@host.161.115.68.195.rev.coltfrance.com)
14:41.15jedirlg.723 may have sense to exist in warez ... but g.729 from digium is quite cheap
14:41.17bjohnsonHmmhesays: use & in the dial command
14:41.25jedirlI don't understand people warez-ing it
14:41.33tzangerjedirl: absolutely..  $10/channel is peanuts -- you make up for it in bandwidth and LD charges
14:41.44Hmmhesaysbjohnson: that's what I thought
14:41.52Hmmhesayspeople warez it simply to prove that they can
14:41.56Hmmhesayswhat more reason do you need?
14:42.09jedirlbut it is digium's codec cracked?
14:42.19ManxPowercoppice, I don't suppose you, oh Asterisk guru whom I am but a guppy in your presence, have any comments on this: http://lists.digium.com/pipermail/asterisk-users/2005-February/090578.html
14:42.21jedirlor it's another one?
14:42.27Hmmhesaysno, you can compile the code from intel
14:42.47*** join/#asterisk Zeeek (~Zeeek@Zeeek.sustaining.supporter.pdpc)
14:45.38*** join/#asterisk LoRez (lorez@lorez.staff.freenode)
14:46.27*** join/#asterisk santiago (~santiago@63.245.86.121)
14:47.10ManxPowerjedirl, The pirate codecs use Intel's developement kit.
14:47.16*** join/#asterisk v_a_d_e_r (~root@82.147.138.26)
14:47.17jedirlahhh
14:47.29jedirl"pirate" in the sense you don't have a patent grant to use it, right?
14:47.41ManxPowerThere is also the G729 codec from the ITU as well, but won't compile for Asterisk out of the box.
14:47.49ManxPowerjedirl, Correct.
14:48.01coppiceor copyright
14:48.12*** join/#asterisk anthm (~anthmct@CPE-69-76-83-52.wi.rr.com)
14:48.13*** mode/#asterisk [+o anthm] by ChanServ
14:48.34Hmmhesaysno, but you can just download the patch
14:49.18Hmmhesaysfor testing purposes for 30 days of course you can use it legally
14:49.33*** join/#asterisk pif (ldm@zenon.apartia.fr)
14:49.38bjohnsonjedirl: pirate in the sense that you're required to wear an eye patch and say "arr" a lot
14:49.39jedirlI guess digium's performs much better, right?
14:49.52ManxPowerI just think that if G729 was legal in many countries then there would be many free implimentations of it.  Much like when crypto was illegal to export from the USA.
14:50.40Hmmhesayswhere it IS legal to compile your own g.729 or g.723 bandwidth is so damn expensive it's not worth it to put boxes there
14:50.47Delvarthere is a free distro of g729 that works with asterisk....
14:50.52*** join/#asterisk HitTop (~Miranda@host6614613596.biz.tor.fcibroadband.com)
14:50.58HitTophi, does anyone know wat res_crypto.so is for?
14:52.34coppiceDelvar: from from legality, you mean :-)
14:52.48HitTopim trying to slim asterisk as small as possible.. and i found that this one module takes up 500kb.. so i was thinking if i can take "res_crypto.so" out or not...
14:53.03Delvarwell im not sure if it is leagal...
14:53.07Delvarauther seems to think so
14:53.24*** join/#asterisk stickynomore (~jeff@nsc66.147.11-46.newsouth.net)
14:53.42Hmmhesaysthe rule of thumb is.... if your country does not have indoor plumbing, and the majority of the population lives in grass huts then it's legal for you to use g.729 there
14:53.45*** part/#asterisk sektor195 (~please@216.86.45.98)
14:54.28coppiceDelvar: the "author" authored about 50 lines of code. The rest is stolen
14:54.28*** join/#asterisk eKo1 (~bernd@207.42.191.66)
14:54.54*** join/#asterisk brc-tux (~cbrinz@pD9E9A4C3.dip0.t-ipconnect.de)
14:55.07Delvariv not realy looked into it farther than a glance at the readme
14:55.13*** join/#asterisk Chuji (Chuji@pcp09929633pcs.tulipgrove.tn.nash.comcast.net)
14:55.19Delvarwouldnt suprise me tho
14:55.25DEVILoperHave any tried GSM modem with *
14:55.55jedirlis it possible to use ulaw/alaw with NuFone's H.323?
14:57.46*** join/#asterisk Syncros (~sysop@noc.routermonkey.net)
14:58.04jedirlI guess I'm the only one here crazy enough to use h.323 :)
14:58.11tzangerindeed :-)
14:58.20tzangerjerjer's the wizard behind the h323 stuff
14:58.23tzangeryou mgiht want to try and corner him
14:58.29jedirlhehehe
14:59.03djinUnable to open IAX timing interface: No such device or address
14:59.04bjohnsonso .. I guess maybe someone has already evaluated these SPA units to determine which codec is good combo of quality / low bitrate.  Or do I need to do it myself?
14:59.19djinWhat does this IAX timing interface mean?
14:59.36*** part/#asterisk brc-tux (~cbrinz@pD9E9A4C3.dip0.t-ipconnect.de)
14:59.39tzangerdjin: that's a great question
14:59.43tzangerare you trying ot use trunking?
14:59.54*** join/#asterisk Tall-guy (tall-guy@hssxrg207-195-103-110.sasknet.sk.ca)
15:00.06jedirlI have a teles VoIP gateway using a GnuGK H.323 gatekeeper; I'm trying to make asterisk answer phonecalls from the VoIP to a concrete extension
15:00.22djintrunking? I'm not sure.
15:00.38tzangerdjin: do you have trunk=yes in the iax.conf anywhere
15:00.56tzangerwhat are you trying to do to get this message?
15:00.57jedirlI've done in h323.conf: type=h323, e164=myphonenumber, context=default
15:01.11jedirlam I doing something bad here?
15:01.18tzangerno idea jedirl
15:01.19Luhiwujedirl, i was unable to change the context in h323 based on the h323 peer, did you try something like that?
15:01.37djintzanger, no
15:01.44[ro]nic3tryhas anyone instaled G729 ?
15:01.45jedirlLuhiwu: nope
15:01.49djinI get this message at startup of asterisk
15:02.04*** join/#asterisk cbachman (~cbachman@victory.ece.northwestern.edu)
15:02.06djin[ro]nic3try, yes.
15:02.07Luhiwujedirl: ok, i also was unable to corner JerJer :)
15:02.18tzangerdjin: post your entire asterisk -vvvvvvvvvv to pastebin
15:02.21jedirlLuhiwu: I'm just trying to make asterisk listen on a H.323 phone number and perform an IVR
15:02.28djintzanger, ok
15:02.36jedirlwell in fact I'm trying to run ASTCC over it :)
15:02.46Luhiwujedirl: did you try changing context=default to context=inc-h323 and then use the s extension in inc-h323 to do the ivr?
15:02.53[ro]nic3tryi'm instaling the free version .. http://www.readytechnology.co.uk/open/g729/INSTALL-041103.txt
15:02.58*** join/#asterisk heison (~heison@dns.somanetworks.com)
15:03.05jedirlLuhiwu: not yet
15:03.18Luhiwujedirl: ok, it should work without problem, i have it working almost out of the box
15:03.38[ro]nic3tryand at step 5a .. i'm lost :  If you have icc 8.0 libimf, move intel_cc_80/lib/libimf.so to
15:03.38[ro]nic3trysomewhere out of the way.  This will allow you to link libimf.a
15:03.38[ro]nic3trystatically with codec_g729.so
15:04.03[ro]nic3trywhat to do ???
15:04.12jedirlLuhiwu: I'm going to try it :)
15:04.18*** join/#asterisk brc-tux (~brc-tux@pD9E9A4C3.dip0.t-ipconnect.de)
15:04.44*** join/#asterisk dsfr (~dsfr@216.207.244.183)
15:05.02djintzanger: http://pastebin.ca/6240
15:05.41*** part/#asterisk brc-tux (~brc-tux@pD9E9A4C3.dip0.t-ipconnect.de)
15:07.05ManxPower[ro]nic3try, don't ask that stuff here.
15:07.16*** part/#asterisk DEVILoper (~x@202.5.145.50)
15:07.21[ro]nic3tryok.. sorry
15:07.36ManxPowerWe do not help people running non-digium G729 codec.
15:07.45*** join/#asterisk ^Fenris (~mazurbul@d3-31.rb.ot.centurytel.net)
15:07.52Luhiwuany g729 expert here? don't run, i've seen you speaking about g729 just a few minutes ago :)
15:08.14*** join/#asterisk MichaelVanD (~MichaelVa@rrcs-24-123-121-190.central.biz.rr.com)
15:09.08LuhiwuManxPower: i have a digium 729 codec, how do i get some help? :)
15:09.19Luhiwui'm getting this error in the * console: "Dropping extra frame of G.729 since we already have a VAD frame at the end"
15:09.25Luhiwuand the sound gets choppy
15:09.51ManxPowerLuhiwu, I believe that is caused by one device trying to use VAD / Silence supression.
15:09.51mikegrbturn off vad
15:09.55montoyaasterisk exist how to doc ?
15:09.56eKo1Luhiwu: That's not an error. I get that all the time.
15:10.03ManxPower~docs
15:10.06jbotDocumentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org
15:10.14ManxPowereKo1, Choppy voice is an error. 8-)
15:10.18*** part/#asterisk [ro]nic3try (~iancu@81.181.199.39)
15:10.22eKo1I don't get choppy voice though.
15:10.27*** part/#asterisk ManxPower (~eric@dsl-208-164-150-160.datasync.com)
15:10.33eKo1Or atleast, nobody has told me about it.
15:10.58mikegrbeKo1: turn off vad, asterisk doesn't support it and it will make the voice choppy
15:10.59Luhiwui do get choppy voice, a lot of it when the message appears, that's why i think it is an error :)
15:11.01eKo1I do have silence suppression disabled on the client that is causing those messages.
15:11.19eKo1But VAD is a different story.
15:11.36Luhiwui can't turn off vad, the gateway is not under my control, and the carrier is not very friendly :(
15:12.03Luhiwuis there any way to force a sip client to unregister from the console?
15:12.38eKo1Not that I know of.
15:12.55*** join/#asterisk kpfleming (~chatzilla@ip68-3-230-141.ph.ph.cox.net)
15:13.40Luhiwui'm having some problems with Linksys PAP2, it appears as registered but when i call the extension, i get "No one is available to answer at this time"
15:14.02ionixlinksys PAP2 can't be used with anything else than Vonage ?
15:14.15Luhiwuionix, i have a unlocked one
15:14.16Connor-PAP2-NA can. normal PAP2 can't
15:14.24Luhiwui have a PAP2-NA
15:15.06*** join/#asterisk sandnigg0r (~trilluser@66-55-197-254.gwi.net)
15:15.23Connor-Yea. the -NA version is for other VoIP providers other than vonage. the non -NA versions are the ones you find in compusa, bestbuy etc..
15:16.02*** topic/#asterisk by bkw_ -> Asterisk: The Open Source PBX || Dev Conf 1PM CST FEB 24th -> IAX2/guest@66.250.68.194/996 || ILEC's can suck my ethernet cord!
15:17.24coppicetelcos sure can suck
15:17.46Zeeekso many things do these days
15:18.16PoincareIAX Subclass: INVAL => means that the extension (on the other side) is invalid?
15:18.31coppiceI bet a winCE vacuum cleaner wouldn't
15:19.37Tall-guyGuys, I have a relatively simple 4 Fxo / 4 fxs box, when I do FXS to sip calls, quality is great, when I come in from outside (ie: FXO) to sip I get what sounds like half duplex issues...can someone point me in a direction? (gently)
15:23.25eipianyone have working asterisk with a router wrt54g?
15:23.42kpflemingyou mean _on_ a wrt54g?
15:23.55*** join/#asterisk RoyK (~roy@host-81-191-147-248.bluecom.no)
15:24.17eipibox <=> wrt54g < => internet
15:24.38eipiunder wrt54g intranet, all works fine
15:25.07eipii have configured ports under application and gaming section
15:25.18ariel_Luhiwu, add qualify=120 to your sip setup for the linksys pap2-na this will help keep it up.
15:25.27*** join/#asterisk jsolares (~jsolares@200.30.141.85)
15:25.49eipibut when an external user tries to connect, my asterisk dont receive the register request
15:26.09eipikpfleming do you have wrt54g?
15:26.26ariel_eipi, it only allows one connection per port like one 5060 or one 5061 to an address. I use the wrt54g here.
15:26.34kpflemingyes, i have lots of them installed, they work fine
15:26.48kpflemingbut i don't have asterisk running behind them, my only * installs are on public IPs
15:27.18eipiyes i forwarded to 1 ip
15:27.22*** join/#asterisk TrevorSHarrison (~trevorsha@24.49.36.218)
15:27.46eipisip,5060,5060,udp, theonly*serverIP
15:28.09eipiiax2,4569,4569,udp, theonly*serverIP
15:28.24eipiiax,5036,5036,udp, theonly*serverIP
15:28.36kpflemingdon't need 5036, it's not being used any longer
15:28.48eipiok, but for this case is the same
15:28.53ariel_eipi, don't forget 10,000 to 20,000
15:28.59kpflemingSIP behind a NAT is a major pain in the butt, you need to use IAX2 or put * in the DMZ to make things easier
15:29.02eipiyes i have it
15:29.13eipirtp, 10000,20000,,udp, theonly*serverIP
15:29.21eipimgcp,2727,2727,,udp, theonly*serverIP
15:29.41dsmousekpfleming: /me hist SIP.
15:29.44dsmousehits
15:30.05eipikpfleming: but * never receives a sip request
15:30.17kpflemingthen something is badly wrong with your network :-(
15:30.51kpflemingyou have SIP Phones working locally on the LAN with that * server?
15:30.54eipiyes
15:31.38*** join/#asterisk user1fn (~Joe@64.90.195.240)
15:31.39kpfleminghmm
15:31.44ezabizoa: how many calls so far?
15:31.44user1fnhowdy, yall
15:32.07user1fngot a zap problem again... so here goes
15:32.09kpflemingyou need to put a box running tcpdump or ethereal on the LAN and see if the packet is being forwared by the WRT to _anywhere_ inside... if not, you have a NAT forwarding problem in the router
15:32.41user1fnfinally got it compile and install, but sometimes I can load the card driver (wcte11xp) and most of the time it says it can't find the hardware
15:32.42ariel_eipi, also make sure your running the lastest firmware on the linksys.
15:32.45eipiits weird i have another network ports open and working
15:33.45ariel_eipi, which device are you using to connect to your asterisk a sipphone/sipura?
15:35.30user1fnnobody in the mood for such a paltry dillemma?
15:35.33user1fn;)
15:36.33*** join/#asterisk ACiDV (~joel@69.156.197.246)
15:38.11ACiDVHi, I have a dial plan that do: exten => 611,1,Answer ... 2,SayDigits(1234) ... 3,Hangup ... and after I have exten => _.,1,Macro(dial-result) ... does it's normal that the dial-result macro is executed after the hangup ?
15:39.18shmaltzACiDV, nope
15:39.34shmaltzdo 3, ${DIALRESULT}
15:39.58shmaltzanyhow in your case you don't even have a dial command
15:41.33ACiDVNoOp(${DIALRESULT}  doesnt return any code
15:41.51shmaltzAciDV, b/c you didn't dial anything
15:41.59shmaltzyou first have to issue the dial command
15:43.19ariel_shmaltz, you should not use _., use _X., instead
15:43.29ACiDVI cannot use _. ?
15:43.33shmaltzariel_, what?
15:43.46*** join/#asterisk codebreaker (~codebreak@flexserv.de)
15:43.52ariel_you should not use it due to it will match everything not just numbers.
15:43.57codebreakerhello.
15:44.15shmaltzI SHOULD NOT USE IT, or YOU SHOULD GET YOUR GLASSES?
15:44.26ACiDVI have use the example on http://www.voip-info.org/tiki-index.php?page=Asterisk%20i%20extension  .... exten => _.,1,what_to_do_for_fat_fingers_always_misdialing
15:44.31codebreakeris ther a softwareclient like kphone etc.. available wich speaks the IAX protocol?
15:44.36ariel_so it's a very bady Idea to have _., it will match i s and everthing else so you can't debug correctly.
15:44.51shmaltzariel_ you still talking to me?
15:44.57*** join/#asterisk eipi (~polarisx@40-142-89-200.fibertel.com.ar)
15:45.02eipiim back
15:45.16ACiDVIts me that have a problem with _., :)
15:45.17eipii restarted wrt54g and now i receive this messags on console: Maximum retries exceeded on call d220ef75892fb87a for seqno 1 (Non-critical Response)
15:45.20ariel_shmaltz, anyone that is here that cars to listen.
15:45.22shmaltzACiDV, that is ther as a hypothtical example
15:45.38ACiDVok =) I will try with _X.
15:45.41eipiand Unable to create/find channel
15:45.48tzangerugh
15:45.49shmaltzACiDV, it will not work
15:45.53tzangerdon't use _X. unless you have no other choice
15:46.00tzangerdon't use '.' unless you have no other choice
15:46.14codebreakerah and i am looking still for a $frontend (web or else) to asterisk
15:46.19ACiDVnot more ? :) if it's the last exten of the list ? to catch all invalid extension ?
15:46.36tzangeruse 'i' to catch invalid extensions
15:46.50ariel_ACiDV, asterisk takes and does it's matching not in order of listed.
15:47.30tzangerariel_: huh?  it most certainly does
15:47.52shmaltztzanger, no it doesn't
15:47.56tzanger...??
15:48.02ariel_sorry baby on lap does not allow me to type correctly.  Asterisk not really if sorts the dialing rules unless you use includes.
15:48.07tzanger_1234567,1,dosomething
15:48.08shmaltztzanger, it maches accordig to context
15:48.11tzanger_1234XXX,1,dosomethingelse
15:48.15bjohnsonACiDV: the order in extensions.conf may not be the order that * processes the extensions
15:48.18tzangerwell yes within a context, that is a given
15:48.24ACiDVok
15:48.45tzangerit is 'first-match' within a dialplan
15:48.46bjohnsonACiDV: a _. or _X. could be the very first extension ever run .. and since it matches eveything .. nothing else in the dial plan would ever run
15:48.59shmaltztzanger, nope
15:49.08tzangershmaltz: that's most certainly how it works on my setup
15:49.11tzangerif I include a,b,c
15:49.16tzangerit will first-match
15:49.19*** part/#asterisk nicox (~nicox@83-64-42-210.prater.xdsl-line.inode.at)
15:49.27bjohnsonit is also how it is documented in the wiki
15:49.32shmaltzhttp://www.voip-info.org/tiki-print.php?page=Asterisk+config+extensions.conf+sorting
15:49.36tzanger[mycontext] includes a,b,c in that order
15:49.41shmaltztzanger, look it up
15:49.45tzangerit will match a number in A if hte same number is in C and the default context
15:49.46bjohnsonI have also personally seen the dial plan order change if I restart * and use show dialplan
15:49.52tzangerer not default but [mycontext]
15:50.02shmaltztzanger, yep after includes but not without includes
15:50.09tzangershmaltz: huh?
15:50.30tzangerif I don't include anything in [mycontext] it does first-match within [mycontext]
15:50.42bjohnsonAsterisk does not match against the extension patterns in the order you define them; the extension patterns are sorted first
15:50.43shmaltzin each context the order might not be what you expact, but if you use includes then you are enforcing the order
15:50.46tzangerif I have an exten that is 1234567 and one underneath that is 1234XXX it will match 1234567 first (if I dial 1234567)
15:50.47bjohnsonfrom the wiki
15:51.01bjohnsonif you do pattern matching .. you have no idea which is checked first
15:51.07tzangerbaloney
15:51.12tzangerthat's exactly how I do my DIDs
15:51.30tzangerI have the specific DIDs first, followed by _292XXXX doing "the number I have is ${EXTEN}" being played
15:51.34tzangerworks just fine
15:51.40tzangerand has for the past year, over numerous CVS HEAD upgrades
15:51.46ChujiUse includes if you want something to be matched first
15:51.53bjohnsonChuji: exactly
15:52.04Chujiinclude => specifics and then include => patterns
15:52.11tzangereither I'm misinterpreting or you're all on crack
15:52.14bjohnsonChuji: or avoid having duplicate pattern matches
15:52.30tzangernow if you have MULTIPLE pattern matches then you might be on to something
15:52.41tzangeri.e. a _292XXXX and a _.
15:52.47Chujitzanger : You may have just got lucky, but the dialplan has some wacky behavior if it could match multiple patterns
15:52.47bjohnsonand _. is by default, a MULTIPLE of something else
15:53.08tzangerI dont' use multiple patterns within a context
15:53.14bjohnsonhence the rule of thumb to never use _. or _X.
15:53.33bjohnsonor if you need to, put it in another context that gets "included"
15:53.36tzangermaybe I just use intelligent dialplans and have never run across it becaues of that.  :-)
15:54.00shmaltztzanger, of course it works, its a specific number and not a pattern
15:54.04bjohnsontzanger: likely .. but doesn't mean that newbies don't stick in _. in inappropriate locations
15:54.09ChujiI've hit it with exgirlfriend logic in DIDs
15:54.23bjohnsonariel_: <- a very wise person
15:54.45ChujiI have a whole range of DID's going to macro, but I want to pull one out
15:54.47tzangeras I said, don't use '.' unless there is no other way
15:55.18tzangerno wait I do use multiple patterns in some dialplans but they're always pretty specific
15:55.26tzangermy local # matching, 800# matching,e tc.
15:55.27bjohnsonso, I concure with ariel_'s first statement that ACiDV should not use _. in his dialplan in the way is is using it
15:55.29ariel_tzanger, I was trying to just explain to him orginally not to use _., and why this is just the way things happen.
15:55.31shmaltztzanger the reason 1234567 and 1234XXX is matched first against 1234567 is b/c that is not a pattern but a number
15:55.32tzangerbut no two patterns will match each other :-)
15:55.57tzangershmaltz: ok, but what in the case of _123XXXX and _1234XXX
15:56.06bjohnsonshmaltz: he does a pattern match on _292XXXX
15:56.13tzangerwill not 1234XXX match 'more' and thus be executed over _123XXXX if the user dials 1234567 ?
15:56.18shmaltzthen it migth or might not be in order
15:56.37shmaltztzanger not according to the wiki
15:56.38shmaltzhttp://www.voip-info.org/tiki-print.php?page=Asterisk+config+extensions.conf+sorting
15:57.08bjohnsontzanger: if _123XXXX and _1234XXX are separate extensions (each with a 1 priority) and in the same context .. you have no way of knowing which will be run first
15:57.25shmaltztzanger, in that example 918 will be sorted last
15:57.25bjohnsonorder can even change from one reload to the next
15:57.31tzangerbjohnson: interesting...  I've never had to run against that :-)
15:57.47tzangerinteresting what you learn
15:58.16bjohnsonI've run into dialplan order problems that generally confirm what that wiki page says
15:58.21shmaltztzanger, each day is a new day with new things, make the best of it today, for tomorrow something new is coming
15:58.23bjohnsonsounds like others here have too
15:58.52greg_workholy crap phone companies drive me nuts
15:59.08shmaltzgreg_work, drop them
15:59.15bjohnsonso .. I think both shmaltz AND ariel_ listed 2 different potential problems with ACiDV's original post
15:59.24shmaltztell your clients they can now do orders over IRC
15:59.38shmaltzbjohnson, exactly
16:01.37Beirdoariel_: only if you beat me to it :)
16:01.40*** join/#asterisk Darwin35 (~Darin@c-24-3-241-22.client.comcast.net)
16:02.09*** join/#asterisk km- (~km-@67.105.178.130)
16:02.48km-ok, whatcha guys think. 2.4 or 2.6 for an asterisk system with a T1 card?
16:03.03shmaltzariel_, enjoy that upgrade,
16:03.29shmaltzkm-, 2.6 has proven to be stable enough by now,
16:03.45km-I remember there being some issues with zaptel and 2.6 a few months ago
16:03.48km-thats why I'm asking
16:03.56km-I heard some people have problems getting a T1 card to behave in 2.6
16:04.41tzangernot I
16:04.42jedirlI can't make H.323 work with asterisk
16:04.57km-tzanger: oh, so maybe it's working now...
16:04.59jedirlanyone could take a look at what asterisk tells me when I make a phonecall?¿
16:05.01shmaltzonly if u use udev and you don't RTFM
16:05.01km-cool, I'll get 2.6.9
16:05.01Tall-guyariel: I'll trade Active Directory experience for future asterisk help :)
16:05.11tzangerI'm running with 2.6.10
16:05.47km-oh, heh, I didn't see it in the ls
16:06.32jedirl<PROTECTED>
16:06.33jedirl<PROTECTED>
16:06.33jedirl<PROTECTED>
16:06.41jedirlanyone has ever seen this ?
16:06.47codebreakeris there a softwareclient like kphone etc.. available wich speaks the IAX protocol? or a hardwarephone. but i didnt find any :(
16:06.51tzangerI imagine if you were able to figure out what cause 7 was it'd help
16:06.58jedirlcause 7 is when you already have a channel used
16:07.30km-hmodes: you awake? :)
16:07.42km-oh lord this box is slow
16:07.46*** part/#asterisk djin (~marius@62.58.40.196)
16:07.55km-I've got a frac t1 and I'm getting 5kb/sec download off this system
16:08.02ariel_Tall-guy, hummm I have been working with windows systems for many years.  But at the present time I am trying to change there old dns to work correctly.
16:08.34tzangerkm-: enable more than one ds0 :-)
16:08.38roamer323codebreaker - iaxcomm and dragonfly (software),  IAXy (hardware) - use google or voip-info.org to find links
16:08.56jedirlnoone knows what could be the reason?
16:09.11km-tzanger: haha
16:09.17km-tzanger: lemme just up the bw here.... <whomp>
16:09.49codebreakerroamer323: i have found iaxy. but i like to have something like an hardwarephone like amtel.. but thanks for dragonfly/iaxcomm. i will try
16:10.02*** join/#asterisk eipi (~eipi@40-142-89-200.fibertel.com.ar)
16:11.06*** join/#asterisk Othello (Othello@hssml0175.pc.nus.edu.sg)
16:11.37Luhiwucodebreaker: i did a iax softphone based on iaxclientocx, pm me if you want
16:13.45*** join/#asterisk RoyK (~roy@host-81-191-147-248.bluecom.no)
16:14.40MichaelVanDcodebreaker: iaxclient.sf.net/index.html lists a few, but I'm partial to iaxComm ;)
16:15.56coppiceI think the world needs a lot more softphones. All similar. All substandard. The important thing is merely that there should be a lot of them. I know many others agree :-)
16:16.42Tall-guycoppice :)
16:17.36kpflemingcoppice: and we need more of them that don't conform to _any_ of the standard UIs for the systems they run on, that's especially helpful
16:19.20ACiDVI have a a TE405 card, I plug a crossover cable between a Nortel BCM and port 1 of Digium Card and after a few seconds, I get a seg fault on asterisk... if I dont plug the T1 cable, the asterisk can work days and days..
16:19.33coppiceand interacting with other tools, like address information is a definite no-no. All available development resources should be focussed where it is most need - on skins
16:20.25zoanext thing to try
16:20.30LoganI have a problem.  The Sipura SPA-841 phone cannot do a real blind transfer.  However, using asterisk's mechanism (triggered by the # key) is problematic, because we often call out to IVRs that require the use of the # key.  Has anyone dealt with this problem before?
16:20.35zoasetup 20 asterisk machines in the test lab
16:20.36zoa:)
16:20.39tzangerACiDV: what's your system processor
16:21.00ACiDVtzanger ... Intel Xeon 2.8ghz
16:21.05tzangerok
16:21.14ACiDVrelated to SMP ?
16:21.14tzangerdoes it segfault if you use a loopback cable
16:21.18*** join/#asterisk Ayano (~erik_leee@209.143.187.254)
16:21.28doughecka_Logan: if you do a search, I seem to remember a patch that required 2 #'s to do a transfer
16:22.02Ayanois there anything special to hook up a ip500 to asterisk?
16:22.04greg_workLogan: if you hit xfer immedately it will be 'blind'
16:22.25shmaltzLogan, you can change it now from # in features.conf
16:22.39MichaelVanDNow I remember why I stopped lurking here.  I wrote iaxComm to fill a personal need (the only iax softphone available anywhere was Steve Kann's wx demo).  I turned it into a real application, and there are a number of users who do appreciate it.
16:22.58Logangreg_work: Sure, I can tell my users "Yes, these phones support blind transfer.  Just press this sequence of buttons erally really fast."  But that's not ideal.
16:23.15LoganI'd prefer to have a way to do a transfer that synchronizes the transfer with the dial of the third party.
16:23.16ACiDVtzanger I've not test with loopback cable yet...
16:23.22*** join/#asterisk jtodd (~jtodd@mccpool-11.ci.monterey.ca.us)
16:23.28tzangerACiDV: also when it segfaults load up the corefile in gdb -- where's it segfaulting
16:23.29Loganshmaltz: Even if I change it from #, there's no good chioce to change it to.
16:23.36Loganshmaltz: The use of a DTMF tone is problematic.
16:23.43coppiceyeah, just get 1.0 out the door so I can start commiting to CVS :-)
16:23.46MichaelVanDThe only feedback I've ever gotten from the irc community is snide childish comments like "suxxors" or vague comments like "unstable" or we don't need another softphone.
16:23.50LoganIs there a way for a user, during a call, to disable the special semantics of the # key?
16:23.52greg_workLogan: yeah, true
16:24.07Logangreg_work: In fact, that's what the Sipura manual says.
16:24.10LoganGod I hate these phones.
16:24.22greg_workLogan: get sipura to update their sofkeys so theres another button for blind transfer
16:24.47greg_workwhile they're at it maybe they can change it so the second menu is displayed during a call, instead of useless "redial" and "directory" buttons
16:24.47Ayanois there anything special to hook up a ip500 to asterisk?  A friend of mine said he set it up and it is not even trying to authenticate.
16:25.18ACiDVtzanger :) not gdb guru... I write: gdb (corefile) ... hmm some help on how to get some info from gdb ? :D
16:25.39MichaelVanDThe whole point of iaxComm is that it written to a crossplatform user interface library.  I don't see how it could be 100% compliant with UI guidelines for three different platforms.
16:25.55greg_workyikes
16:26.07coppiceisn't that what wxwidgets is supposed to achieve?
16:26.21doughecka_or wine
16:26.23doughecka_=D
16:26.25ACiDVok, found the Asterisk Debuging page in wiki
16:26.29greg_work"hi, calling from ma bell.. got a request to cancel the line xxx-xxxx" ... "no.. you're supposed to cancel the voicemail on that line, not the line itself" .. "oh, it just says the line here.. good thing i checked"
16:26.30tzangerACiDV: gdb -c corefile
16:26.33tzangerand then type 'bt'
16:26.36tzangerand pastbin it
16:26.48coppiceno. wine blurs the sense so you no longer care :-)
16:26.49doughecka_greg_work: dweeb =D
16:26.53LoganI think I'm going to have to modify asterisk.  Anyone know where the code that captures the # key is?
16:26.54doughecka_coppice: :P
16:27.50*** join/#asterisk jesse_132 (~chatzilla@207.246.72.150)
16:28.02ACiDVtzafrir_home : not very long ...  #0  0x00ea2c6a in q931_getcall (pri=0x88a87e0, cr=32768) at q931.c:2157
16:28.08ACiDVops... not tzafrir_home but tzanger
16:28.11Beirdogreg_work: must be Bell Canada
16:28.18Tall-guybeird: or sasktel :)
16:28.27jedirlI get "-- ClearCall: Request to clear call with token ip$213.162.200.83:20004/202, cause 7 " when I try to call from outside to an asterisk with h.323... anyone knows what may be happening?
16:28.49jesse_132I am trying to call between two non-NAT sip phones and get  chan_sip.c:683 retrans_pkt: Maximum retries exceeded on call 7c3442a971503d5b06fd47d22c6b2c08@192.168.1.119 for seqno 102 (Critical Request) ...  I can call out with either of them.  Anyone know what could be wrong?
16:28.52BeirdoTall-guy: same thing, they're both BCE :)
16:29.08Tall-guybeirdo: yup
16:29.35*** join/#asterisk MajestiK (~MajestiK@S01060800208687ec.ed.shawcable.net)
16:29.46BeirdoBell Canada is such a bunch of incompetent fools
16:29.59Beirdothey have NEVER gotten an order right for me, not once
16:30.06coppicehey, they're a telco :-)
16:30.11tzangerBeirdo: heheh
16:30.24Tall-guybeirdo: they got one right for me the other day.....honest.....everything was perfect.....I damn near died......
16:30.24MichaelVanDwxWidgets is supposed to provide you with a single API that you can write to so that your app will recompile and run on other platforms.  As I understand it, compliance with platform UI recommendations/norms is a secondary considration.
16:30.35Beirdowow
16:30.58Beirdothat's impressive, Tall-guy.  you sure it wasn't a dream sequence?
16:30.58tzangerMichaelVanD: use qt now that it is also win32 free
16:31.33coppicewhy? what would that give you?
16:31.38tzangerI never liked wxwindows
16:31.56MichaelVanDand what's my payoff for throwing out the existing code?
16:32.04coppiceall windowing systems are a PITA. wxwidgets is not so bad
16:32.08tzangerMichaelVanD: oh I didn't realize you had a ton of existing code
16:32.12tzangermy apologies
16:32.28coppicethe lack of a good screen painter is its biggest drawback
16:32.43MichaelVanDI didn't say I have a ton of code.  What's my advantage to throwing out 1.4 lbs of code?
16:32.55tzangercoppice: what, you want to get more artistic than with the white-out?
16:33.09tzangeryou have a pound and a half of code?
16:33.22coppiceon punch cards, yes :-)
16:33.24tzangerare you trying to be amusing or am I just missing the joke
16:33.28codebreakerare ther some webfrontends now available for asterisk? i only find closed source like switchfox or gofon?
16:33.36coppicetzanger: I mean a good dialog designer
16:33.37tzangerthere's amp too
16:33.46tzangercoppice: qtdesigner?
16:33.48tzangerwhat's it called
16:34.00coppicei was talking about wxwidgets
16:34.09codebreakerroamer323: iaxcomm is really good. it fit my needs
16:34.21MichaelVanDXRCed isn't elegant, but it works.  But, maybe switching to Qt would make my app no longer substandard
16:34.34tzangerdesginer is the binary name
16:35.01coppiceXRCed sucks, but it seems to be the best there is
16:36.14coppiceI think the biggest problem with wxwidgets is it has lost momentum
16:36.29coppiceit seems to be changing very slowly these days
16:36.36bjohnsontzanger: a 1.4 lbs of code approx. = 0.07% of a "ton" of code (since 1 ton = 2000 lbs)
16:37.00tzangerbjohnson: ha
16:37.06MichaelVanDAnd I really am open to constructive criticism.  It's just frustrating that too many people won't take the time to offer it, rather "substandard".
16:37.13tzangerwell I didn't say it was a metric ton or an imperial ton
16:37.15coppice1 ton = 2240 pounds, actually
16:37.24tzangerI usually measure things like this in metric buttloads, personally
16:37.28Tall-guyhow many hogsheads in a firkin?
16:37.32coppicemetric and imperial are <1% different
16:37.33bjohnsontzanger: no such thing as a metric ton
16:37.36bjohnsonit's a tonne
16:37.40tzangeris it a european or north american firkin?
16:37.41MichaelVanDthe metric buttload is 11% larger than the imperial
16:37.54Beirdohehe
16:38.11tzangerMichaelVanD: eh?  I'm not criticizing you
16:38.13bjohnsonTall-guy: does it make a firkin's difference?
16:38.29bjohnsonwe're on metric time here (20 hours a day)
16:38.30tzangerif you don't have a lot of code and wxwindoes is causing you sufficient pain, try something else was all I was saying
16:38.36bjohnsoncauses confusing with the yanks though
16:38.42tzangerbjohnson: hahaha
16:38.45coppiceMichaelVanD: just call it 1.0, so I can starting commiting to CVS
16:38.57Tall-guybjohnson: was just adding my 2 X 9/5 +32 cents :)
16:39.00tzangerthere was an april fool's on CBC a few years ago where they caused quite a stir saying that the govmn't was going to SI units of time
16:39.02MichaelVanDtzanger: I know that.  I'm saying: "What's the upside to me throwing out evn 1.5 lbs of code to switch to qt?"
16:39.21bjohnsonbut with 500 days a year, we get more stuff done in a year than non-metric countries
16:39.28tzangerMichaelVanD: much larger install base, active, flourishing development, commercial support if you need it, etc.
16:39.34Beirdotzanger: of course, our second IS actually SI, but it's more fun with the spoof
16:39.41tzangerBeirdo: yes
16:39.50MichaelVanDI don't have a ton of code, but it's all written to wxWidgets
16:39.52tzangerwhat is it, 32000 oscillations of a cesium atom or something?
16:40.08tzangerMichaelVanD: Qt is also C++ not sure if WxWindows is C/C++ or just C
16:40.12jesse_132Dial("IAX2/NuFone@xxx.xxx.xxx.xxx:4569/6", "SIP/2000|20|tr")   works but Dial("SIP/2001-ccc8", "SIP/2000|20|tr") doesn't ...  Any pointers?
16:40.15Beirdosomething lame like that, yeah
16:40.25Tall-guyceisum the day :)
16:40.26*** join/#asterisk jaiger (~jaiger@fire.innovationsw.com)
16:40.26coppiceMichaelVanD: It seems like wxwidgets is never gonna move past 2.4.2. It seems to have stagnated
16:41.00jedirlQt's C++ is far from standard
16:41.29coppiceNow Qt has gone GPL for windows I suspect it might get a lot more buyin, and become the standard platform
16:41.43MichaelVanDI'm willing to be convinced, but wxWidgets does what I need in 2.4.2, 2.5.3 *is* in active development with 2.6 slated for release (this summer?)
16:41.44coppicemost C++ is far from standard
16:41.46jedirlMichaelVanD: Have you seen FLTK or Fox?
16:41.47_m_Is there anyone working on CSTA support for *?
16:41.49tzangercoppice: :-)
16:41.58bjohnsonerr .. maybe the standard platform for people trying to achieve cross-compatibility
16:42.03coppice2.5.3 has been in the pipeline forever
16:42.08jedirlMichaelVanD: those are simple lightweight and quite powerful toolkits
16:42.24coppiceFLTK is good in many ways, but never looks native
16:42.32tzangerjedirl: yeah but you run against hte look
16:42.36tzangerand why the app looks/works "different"
16:42.36_m_fltk looks nice, indeed, and it is very lightweight.
16:42.38coppiceFLTK has a nice dialog designer
16:42.52jedirlI don't think an app really needs to look native
16:43.01jedirlin fact most common windows apps don't look "native"
16:43.06tzangerjedirl: when you're running native apps beside it it does become an issue
16:43.10tzangerin my experience
16:43.15coppiceCrap System for Telecoms Applications
16:43.17tzangerjedirl: only the fucked up skinned ones that we don't run anyway
16:43.21MichaelVanDAnd I could be way wrong on this, but I think that wxWidgets goes a long way to making the application have a more native look and feel that qt
16:43.23jedirlonlhy MacOS takes "native-look" serious
16:43.54denonmacos wastes too much time on stupid crap, and not enough time on taking their whole product seriously
16:44.08jedirlI think that an app that has a clean and lightweight interface is enough, but that's just my opinion
16:44.23coppiceor Computer Supported Telecommunications Applications
16:44.31coppicedepends on your mood
16:44.46jedirldoesn't have to have the buttons exactly the same size and shape than native platform ones
16:44.59*** join/#asterisk e3eli3h (~e3eli3h@83.168.2.150)
16:45.07_m_coppice: I'm leaning towards "crap system" right now.
16:45.25coppiceI've done a number of things with FLTK, and it works well.
16:45.46tzangerI am partial to Qt since I am also a KDE user
16:45.51tzangerit's nice to have apps work across both
16:45.59coppice_m_: CSTA is a pain to work with. not enough is tied down by the specs, and you end up with much pain and misery.
16:46.00tzangerand I also use Psi which uses Qt
16:46.24_m_coppice: I was afraid someone would tell me exactly that.
16:46.40jedirlwhat is CSTA?
16:46.48tzangerI am very much in favour of ONE cross-platform windowing toolkit that works
16:46.51jedirlis it a standard or something?
16:46.56tzangerinstead of a half dozen fighting for #1
16:47.02coppiceactually almost any modern toolkit except GTK works pretty well cross platform
16:47.07_m_jedirl: ECMA standard
16:47.07MichaelVanDI know that iaxComm looks like a native app on Win98, Win2000 and WinXP.  To my untrained eye, it "looks right: on RedHat9 and on OSX.  While I know that it doesn't exactly follow Apples guidelines for menu layout, that's to blame on iaxComm, not the toolkit
16:47.18jedirlI don't think GTK works that bad, in fact, GAIM runs smooth on windows
16:47.32tzangerI hate GTK
16:47.38coppicemost things using GTK are very troublesome on windows
16:47.40tzangeryou want nonstandard, THAT's one fucked-up toolkit
16:48.11jedirlanyone knows how to make H.323 work with *? :)
16:48.15coppiceI think GTK is great on Linux. because its in C it plays nicely with everyone.
16:48.19tzangerIMO GTK is more an excercise in how to make C behave like C++ because you don't like C++ rather than a decent toolkit platform
16:48.20ZeeekI think PAN uses GTK fwiw
16:48.34tzangerit's infuriating to use
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16:49.05coppiceits biggest problem is the lack of effort put into polishing the windows port.
16:49.06tzangeragain though, my opinion only
16:49.06*** join/#asterisk sysdef (~sysdef@pD9561E44.dip.t-dialin.net)
16:49.38jedirlI think the solution for a powerful windows GTK is cairo
16:50.09*** join/#asterisk DsrtZrzmra (~DsrtZrzmr@dsl-200-67-75-232.prod-empresarial.com.mx)
16:50.32jaigerhas anyone looked at the Mono gtk stuff?
16:50.34MichaelVanDAVG antivirus and Forte Agent are two commercial products that I use that are written using wxWidgets.  Audacity is a cross platform app written with wxWidgets
16:51.07coppiceaudacity might be a bad example. it tried to look non-native everywhere -)
16:51.09jedirljaiger: I've tried it, small "Hello world" apps , seems great
16:51.34MichaelVanDcoppice: point taken ;)
16:51.35DsrtZrzmraHas anybody played with GUI's for Asterisk?
16:52.35MichaelVanDcoppice: have you tried iaxComm?
16:53.01coppicecoppice == Steve Underwood
16:53.09DsrtZrzmraim trying to implement a GUI: ACTOS seems to be a nice one, but i really want to hear your words before installing a hundred packages and 5 GUIs.
16:53.43MichaelVanDOh, OK.
16:53.49jedirloff-topic: anyone knows how to make a teles VoIP gateway run SIP? I'm going crazy trying to make asterisk talk H.323 with this crap
16:55.57coppicewhat iaxcomm needs is not a new toolkit. it needs to move from useful to great, and that  means great audio and real useful features. the audio side should initially come from Steve Kann's work. If I have time, I will move it beyond that to state of the art. It needs to work in Unicode, but I have patches to do most of that waiting to commit after 1.0
16:56.31*** join/#asterisk ChulJin (~chuljin@adsl-68-121-94-237.dsl.irvnca.pacbell.net)
16:58.16AyanoDsrtZrzmra:  get a cheap pc, and try asterisk @ home.  The new version has a bunch of guis there.
16:58.18NuggetDsrtZrzmra: there are several good asterisk GUIs which are servicable for delegating or simplifying the more tedious aspects of asterisk management.  If you're hoping to ifnd a GUI which will allow you to avoid having to learn how to edit the config files, though, you will be very dissatisfied with the options.
16:58.37*** part/#asterisk codebreaker (~codebreak@flexserv.de)
16:58.43AyanoNugget:  So true.
16:58.49MichaelVanDAre you waiting for the library to be declared 1.0?
16:59.02Nuggetit is simply not possible (nor, imho, should it be) to deploy asterisk without developing an understanding of the various config files.
17:00.02coppiceNo. Any good library never reaches 1.0 :-) I am waiting for iaxcomm to reach 1.0 before commiting anything more than bugfixes.
17:00.38*** part/#asterisk km- (~km-@67.105.178.130)
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17:00.57*** part/#asterisk PTG123 (~PTG123@ip68-106-19-249.ph.ph.cox.net)
17:05.18MichaelVanDI have no bug reports for 1.0rc2 for windows.  I have a bug report that it sometimes hangs on exit under linux.  You and I have discussed the hang on exit probem, and I understand that it works for you now, right?
17:06.01coppicethat works for me now. things still sometimes go wrong in the conf menus, though. Seemd erratic
17:06.25MichaelVanDAnything more specific?
17:06.54coppicenothing more than I reported before
17:07.13ChulJinis anyone else not able to resolve cvs.digium.com? or am I just using 'bad' DNS servers?
17:07.41brendaIs that Xorcom any good?
17:09.42malcolmdChulJin: Bad storms took ous out.
17:09.58malcolmdous = us
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17:15.09zoawe will release a new iaxphone soon
17:15.12zoastill working on it
17:15.39*** join/#asterisk Fabe (~spamhere@pD95B0BF7.dip0.t-ipconnect.de)
17:15.46coppiceYAIAXP, maybe?
17:16.23ariel_brenda, I have not used it But it should get you started on a basic setup.
17:16.34ariel_looks like digium and the mailing list is down.
17:16.53stevekstevekhey Steves :)
17:16.59malcolmdyes, we're down right now
17:17.22coppicewhat is it about softphones that seems to make most people incapable of cooperating to make one of real worth
17:17.35stevekstevekcoppice: I dunno.
17:17.50*** join/#asterisk paulc (~paulc@S010600062586a0b4.vc.shawcable.net)
17:18.05stevekstevekEveryone wants to do their own thing..  At least now most people are using the same library..
17:18.10coppiceI think the main attraction of developing yet another softphone is the potential to make 133t skin for it :-)
17:18.29MichaelVanDcoppice: I didn't know that you still had your nose out of joint about me losing one diff over a year ago.  "Incapable of cooperating?"
17:18.31stevekstevek'could have made it GPL, which would have at least forced source to be out there.. but that wouldn't necessarily force cooperation.
17:19.06zoahey steves yes :)
17:19.08*** join/#asterisk PTG123 (~PTG123@ip68-106-19-249.ph.ph.cox.net)
17:19.10PTG123Feb 21 11:23:18 WARNING[784]: chan_sip.c:728 retrans_pkt: Maximum retries exceeded on call 289cd1e64ca8e27037efa6f717d2b003@66.55.69.242 for seqno 102 (Non-critical Request)
17:19.11coppiceMichaelVanD: I wasn't referring to that. I was refering to the proliferation of second rate crap phones
17:19.14PTG123anyone know what that is from?
17:19.18zoastevek, the lib is great
17:19.22user1fnanyone familiar with using cdr_odbc and mysql?
17:19.25zoaits for very fast development
17:19.31ChulJinmalcolm: ah OK thanks. :(
17:19.45malcolmdChulJin: yup :(
17:21.39ariel_PTG123, network issues
17:23.52PTG123ariel: what type of network issues?
17:24.51stevekstevekOK, I've finally caught up..
17:25.20ZeeekPTG like when a SIP phone can't be reached
17:25.21stevekstevekat some point there, I thought Coppice and Mvand were going to start throwing things at each other..
17:25.28Nuggetheh
17:25.50coppiceeh?
17:26.49*** join/#asterisk Lloydio (~sdfs@geton2.gotadsl.co.uk)
17:26.59stevekstevekjust seemed like you were offering (mostly valid) criticism, and Michael was starting to see it more personally.
17:27.12coppicea couple of years ago people joked about ghello projects on freshmeat. Its probably time for gphone jokes
17:27.30LloydioHi, i got a real strange problem with a x100p card is there anyone that knows abit about them?
17:27.42coppiceMichael's just a very sensitive guy. :-)
17:28.05Nuggetnothing will ever top FSF's hello world, which is self-modifying and includes a mail reader.
17:28.20*** join/#asterisk fishboy1669 (proxyuser@62.69.81.129)
17:28.26fishboy1669hi guys
17:28.27ZeeekLloydio say it
17:28.38Lloydiowell its quite long
17:28.50Zeeekthen don't
17:28.51stevekstevekcoppice: I did more research into the pareto stuff, and it seems like it's more expensive than just keeping the last 500 timestamps in a buffer, and actually going through them..
17:28.55Lloydioit answers the phone and after that it doesnt put down the phone, if i pick the phone up i get no dial tone and when i put the phone back down and pick it up again i get a dial tone
17:28.55Lloydio<Lloydio> but then its picks up the phone again after 60 seconds
17:28.57user1fnagian... has anyone got cdr_odbc working?
17:29.02Ayanois there anything special to hook up a ip500 to asterisk?  A friend of mine said he set it up and it is not even trying to authenticate.
17:29.04coppicephones aren't like that. phones don't expand until they can handle mail. they get half developed and then abandoned
17:29.32ZeeekLloydio what country are you in ?
17:29.37LloydioUK
17:29.41BrianR___I had to firmware upgrade my polycom ip600 to get sip registration that actually worked
17:29.50*** join/#asterisk user1fn (~Joe@64.90.195.240)
17:29.56coppicestevestevek: what's pareto?
17:30.14AyanoI gave him that suggestion already, I know they don't work out of the box for most polycomm.
17:30.23AyanoI think he is trying now.
17:30.28ZeeekLloydio you have your X100 connected to what? BT line?
17:30.31stevekstevekPareto distribution.
17:30.42Lloydioyes a BT line
17:30.43BrianR___the web site linked from the voip info wiki has working firmware and the user guide
17:30.46Lloydioanalogue one
17:31.06stevekstevekBasically, one of these JB papers talked about estimating the parameters of a Pareto distribution, in order to be able to estimate jb parameters
17:31.07Zeeekso it answers when the line rings and then never hangs up?
17:31.14stevekstevekbased on certain loss percentage.
17:31.17Lloydioyes thats correct
17:31.23BrianR___I do think I need to fix the g729 passthru in asterisk - or I have it misconfigured somewhere...
17:31.40fishboy1669has anyone here used cli on x100p in uk
17:31.59ZeeekLloydio have you scanned the mailing list and the wiki? BT is full of challenges for the X100P cards
17:32.10coppicestevestevek: I think that's the wrong approach
17:32.16stevekstevekbut, (a) the algorithm they used in their papers didn't seem to work for me [has problems with delays of zero or negative amounts, at least], and (b) seems more expensive than just doing it manually.
17:32.29Lloydioyes its been a nightmare i have been searching for a good day
17:32.57stevekstevekcoppice: I'd appreciate ideas you might have..
17:33.01Zeeekonly 1 day? That's nothing!
17:33.23Lloydiohehe :P
17:33.52ZeeekI see threads called Digium cards connecting to BT in UK
17:34.03junky[work]whats that exactly
17:34.03junky[work]Feb 20 11:16:07 WARNING[10091]: Unable to set linear mode on channel 69
17:34.06junky[work]?
17:34.30Lloydioi have seen alot of people have success with the cable companies and ISDN but none really clarifying that they are using a analogue line
17:34.46Lloydiook Zeek ill have alook at that
17:34.50LloydioThanks ;)
17:34.55ZeeekLloydio there are many posts from UK with problems with both X100P and TDM400 FXO
17:35.06ZeeekI see no solutions, only people with problems
17:35.26*** join/#asterisk Zebble (~Zebble@66.207.107.50)
17:35.33BrianR___I ordered a whole bunch of x100p knockoffs on eBay
17:35.55fishboy1669lloydio what is your issue with the x100p?
17:35.59Lloydiommmm
17:36.07fishboy1669anyone used x100p with bt cli?
17:36.25Zeeekfishboy you're in UK?
17:36.27Lloydioit answers the phone but then doesnt put the phone down
17:36.31|Vulture|there really is no reason to use a x100p other than testing... the TDMs are more practicle
17:36.51Zeeekpeople in UK rae having problems with TDM4xx FXO
17:36.53Zebblei've found that the x100p is more "forgiving" on some lines than the TDM FXO modules.
17:36.55fishboy1669is it an officeal x100p?
17:37.00Lloydioi pick up the phone and put it down and when i pick it up again the dial tone apears
17:37.03fishboy1669hey llydio where u based?
17:37.13|Vulture|Zebble: for echo?
17:37.35LloydioBournemouth , Dorset
17:37.37Zebble|Vulture|:  nah, for lines being left in an off-hook state.
17:38.09|Vulture|ah Ive only had a problem with 1 office, but thats on the telco side.. not sending hangup
17:38.26BrianR___even when terminating on a real pbx, analog trunks need a lot of tuning to make everything go echo-free
17:38.42ZebbleAsterisk isn't a real PBX?  :)
17:38.44*** join/#asterisk Shrink (~tgb@cpc1-cwma1-6-0-cust233.swan.cable.ntl.com)
17:38.56*** join/#asterisk Jackthe (~jesse@thewhitehouse.adsl.utwente.nl)
17:39.28BrianR___heh.. Certainly not on the scale of my previous employer's dms100 :)
17:39.35Zebbleis the digium.com domain down for anybody else?  cvs.digium.com isn't responding (DNS servers can't be reached)
17:39.56ChulJin'If it's not proprietary, it's not "real".' - Lucent marketing materials
17:39.57Shrinkhi, having a problem with the avm c4 card - whenever I make an outgoing call the console says everybody is busy
17:40.08Zebbleah.. there it goes.  Digium is responding again...
17:40.12ChulJinZebble: per malcolm, it was taken out by a storm
17:40.18ZebbleChulJin:  very true.
17:40.26Zebbleouch!  seems to be backup now.
17:40.34ShrinkI've followed the wiki entries on voip-info.org but no luck
17:41.13Zebbleor not.  DNS is responding, but no CVS.  oh well.
17:41.17*** join/#asterisk reval (~reval@83.149.40.131)
17:41.20BrianR___Asterisk is real enough for me... We're considering inflicting it on around 300 people...
17:41.31EssobiMAha
17:42.13coppicestevestevek: There are several people who have tried something interesting in the last year or two. I may have mentioned it to you. I was thinking along similar lines, but with a few differences. These all use a modified WSOLA scheme. I want to try a modified PICOLA, which should be more efficient. I also find different patterns of delay behaviour in the jitter from what most papers are...
17:42.15coppice...showing, and that makes we want to try some other differences.
17:42.16coppiceThis has some wave files to listen to: http://ivms.stanford.edu/~liang/research/sigproc2/
17:42.18coppiceand here are some papers on the topic (I can't find the paper with the most interesting practical results at the moment):
17:42.20coppicehttp://ivms.stanford.edu/~liang/ research/publications/icassp01.pdf
17:42.21coppicehttp://www.tsp.ece.mcgill.ca/Kabal/ papers/2003/ShallwaniC2003.pdf
17:42.23coppicehttp://netmedia.kjist.ac.kr/old_home/ jongwon/papers/2002pa-jinyong.pdf
17:42.31*** part/#asterisk PTG123 (~PTG123@ip68-106-19-249.ph.ph.cox.net)
17:42.46JacktheHello, I'm looking for the sourcefiles of the iaxclient for some testing, does someone know where I can download those in a tarball or something like that?
17:43.35Zebble~google iaxclient source
17:44.14jsolaresanyone know of a voip provider that has unlimited calls plan and byod and iax2?
17:44.40Zebblewww.spectravoice.com  - you have to ask specifically for IAX2
17:45.14jsolaresthanks, i'll look at their website
17:45.25greg_workjsolares: "unlimited calls" plans are not as good as you might think
17:45.30fishboy1669hi zeek yes
17:45.31Jackthejbot, thanks but I already tried that one
17:45.31jbotno worries, Jackthe
17:45.44jsolaresgreg_work: why?
17:45.49fishboy1669i hav just found the wiki details for a x100p cli patch
17:46.01greg_workjsolares: you can get DID's at a lot of places for a couple dollars a month, plus 1 - 2 cents/min usage ..
17:46.10greg_work(less if you're doing very high volumes)
17:46.16Jackthejbot, they show the sourceforge site but won't give me an actual download of the tarball
17:46.39greg_workmost of time you have to use around 3000minutes before the 'unlimited' part pays off
17:46.48stepcutjsolares: I believe voicepulse offers iax2
17:46.55techieheh 'unlimited'
17:46.56jsolareswell i dont need DID's, and they're going to be set up at a remote location, think a rural community in central america. and they *might* go for more than 3000 minutes
17:47.02fishboy1669i havent had any experience of the tdm cards
17:47.05jsolaresstepcut: thanks
17:47.07Jacktheso I was wondering if someone here has it, I need the C-code for some testing
17:47.36stepcutjsolares: I have no experience with them, but they do seem to be iax+unlimited+byod (they explicity mention asterisk)
17:48.09stepcutjsolares: they talk mostly about sip on their promo material, but if you search the web for iax2 and voicepulse..
17:48.35stepcutjsolares: or look at this page: http://www.iaxtel.com/sponsors.html
17:48.46jsolares:D thanks a bunch
17:48.54fishboy1669foobar anyone know where to get the uk bt cli patch for the x100p
17:49.11kpflemingjsolares: exactly what do you need? I can provide origination via IAX2 at a flat-rate
17:49.42jsolaresi need to be able to setup up IAXy's directly to the provider and make calls into the US
17:49.53jsolaresno need for DID's
17:50.19Zeeeknufone
17:50.21kpflemingso you want IAX2 termination, not orignation
17:50.28jsolaresyep
17:50.35kpfleminghow many ATAs would be invovled?
17:50.54jsolaresfrom 5 to 40, each could be a separate account
17:51.11kpflemingbut you'd rather they all be a single account sharing the pool of channels/minutes, i'm sure
17:51.41jsolaresi really dont know the volume to expect, which is why i'm looking for unlimited and not get burned if they use a bunch
17:51.58kpflemingyou want unlimited 1+ dialing? (US/Canada LD)
17:52.05jsolaresyes
17:52.12kpfleminghmm... you know that's not going to be cheap
17:52.30jsolaresoh i know, but i want to know how much is not cheap
17:52.32JerJerat this point in the game, there is no such thing as unlimited
17:52.40techieso true
17:52.45kpflemingnot truly unlimited, thats true
17:53.12kpflemingi would do it based on a number of simultaneous channels, each allowed a reasonable amount of LD per month
17:53.51jsolareswhat range per ATA per month in price are you thinking of?
17:54.08greg_workjsolares: how are you charging this out to people, or is it a free service you're providing?
17:54.11kpflemingit would be per channel, you can have as many ATAs sharing those chanenls as you want
17:54.31kpflemingper channel would probably be $30-$35 per month, something like that
17:54.48jsolaresgreg_work: it's still a plan on it's infancy, i need to know costs to set up the chargin price
17:55.12kpflemingthe advantage of "channels" is that you can add them when you need them, if your users are getting busy signals you just add more channels :-)
17:55.18greg_workjsolares: well, wouldn't getting a per-min plan be better than? you can offer it cheap, and just charge per min
17:55.38jsolaresgreg_work: the hassle of chargin per minute is what i'm trying to avoid
17:55.40greg_workand that way, if they use a ton of minutes and it costs you a lot, you get a lot of revenue as well so ti works out
17:55.48kpflemingrevenue is good :-)
17:56.27jsolareskpfleming: pm contact info so we can discuss this further :)
17:56.31greg_workjsolares: you can do it with accounts, to avoid the 'hassle'.
17:56.45greg_workput $10 on a calling card, then you can make calls until you use that up
17:56.53greg_work* has a calling card application
17:57.03*** join/#asterisk SeaForth (~SeaForth@c-24-1-126-202.client.comcast.net)
17:57.25jsolaresit's still a hassle for me and the customer, it is a good thing and i know it works, but it wouldnt for me, atleast not how the ata's would be set up
17:57.28greg_work(a "calling card" could just be an account number, you don't have to make up physical cards)
17:57.53Hmmhesaysyeah, but it's damn near easier just to whip up an agi cc app
18:02.45*** join/#asterisk jets (~jetsn@guardian.pmt.org)
18:03.10fishboy1669hi guys
18:03.12fishboy1669hows life
18:03.20fishboy1669anyone winning today?
18:03.23Delvarnight all!
18:03.24fishboy1669i was but not now
18:03.26fishboy1669lol
18:03.31Delvar:)
18:03.32fishboy1669delvar night
18:03.43fishboy1669i think hes scared of me?!
18:03.45fishboy1669lol
18:03.50Delvarjust a bit
18:03.56fishboy1669he he
18:03.57Delvaryou seem a bit too jolly
18:04.09fishboy1669its 6pm home time
18:04.17fishboy1669good enought reason to be jolly
18:04.31fishboy1669on top of getting my phones to show the time and dl config of server
18:04.41fishboy1669cant get my x100p to do cli though
18:04.42fishboy1669:(
18:04.56fishboy1669what i have read up till now is that the patch dont work no more
18:05.05fishboy1669and cant see any mention of a new one :(((((
18:05.08terrapenugh i hate html
18:05.25fishboy1669why whats wrong with it?
18:05.36fishboy1669thml
18:05.40fishboy1669lmht
18:05.41terrapentrying to do fancy things without javascript is hard
18:05.48fishboy1669aha i see
18:05.50fishboy1669yes
18:05.58fishboy1669but html is just for text really
18:06.02fishboy1669dhtml is what u need
18:06.09fishboy1669if u wanna be a record breaker
18:06.10fishboy1669lol
18:08.15*** join/#asterisk harryvv (~comming@S010600055d210201.vs.shawcable.net)
18:09.34fishboy1669christ i have scared everyone off!!!!
18:09.37fishboy1669booo hooo
18:10.14Zeeekyep you did it nopw!
18:10.42Zeeek"Concerto for Piano, Voice and 500 Screaming A**holes - DVD VIDEO"
18:12.03*** join/#asterisk jayden (~ircatjerr@65.170.43.34)
18:14.33harryvvSeems like everyone and there perents own some kind of phone releated domain name.
18:16.10fishboy1669lol
18:16.18greg_workyou know, theres really a steep learning curve in the voip and really telephone industry in general
18:16.28fishboy1669yes
18:16.30fishboy1669there is
18:16.38fishboy1669thats why were the unsung heros
18:16.43greg_worksomeone just came in my office and saw "Order 800 DID's" on a webpage on my screen, and asked "what are DIDs and why do you want 800 of them?"
18:16.45fishboy1669and i still dont get paid enough
18:16.46fishboy1669sob
18:16.48fishboy1669sob
18:16.50fishboy1669bob
18:16.58*** join/#asterisk denon (denon@synapse.subneural.net)
18:16.58*** mode/#asterisk [+o denon] by ChanServ
18:17.09harryvvgreg heheh
18:17.28fishboy1669lol greg tell them there digitally integrated dildos and u are going to resell them on ebay
18:17.32dsmousegreg_work: I have no idea why you would want 800 of them... unless of course you started resaleing thme
18:17.45dsmouseblah
18:17.49jesse_132Trying to figure out what is wrong... No NAT involved -- Working: Internal IAX->SIP, IAX->IAX, internal SIP->external IAX --  NOT working: Internal SIP->Internal SIP , Interal SIP->Interal IAX
18:17.52junky[work]~agi api
18:17.54jboti heard agi api is at http://home.cogeco.ca/~camstuff/agi.html
18:18.07fishboy1669~did
18:18.08jbotextra, extra, read all about it, did is Direct Inward Dialing
18:18.09*** join/#asterisk Conductor (~thomas@62.8.240.132)
18:18.11greg_worklol dsmouse it meant 800 as in "toll-free"
18:18.11*** join/#asterisk dahunter (~joe@lsanca1-ar8-4-60-068-194.lsanca1.dsl-verizon.net)
18:18.19dsmouseI know :)
18:18.28dsmouseBut that's not what he ASKED
18:18.34Hmmhesayshmmm could I get away with 8 fxo and 4fxs ports in a single p4 2.8ghz ?
18:18.46greg_workno but my point was more .. to someone who knows nothing about it, the webpages are really confusing
18:18.50Conductorwhat options do i have to set to my kernel 2.6 config to make it work with a Digium E100 Card + Asterisk*?
18:19.02JerJerHmmhesays:  using a TE405P and TA750, sure
18:19.11dahunterIs it possible to record every phone call?  You know do something like Play "This call may be recorded for quality purposes" and then do something like a Backgroundrecord(somefile:gsm)
18:19.38yashaxGuys, can someone please assist me in upgrading the firmware for Polycom IP500 from Altigen to SIP?  Thank you....
18:19.41harryvvdahunter yes. BTW, what are the laws concerning recording calls on a server?
18:19.45JerJerdahunter:  show application Monitor
18:20.31dahunterharryvv: Well, in California, you have to alert them that you are going to record them.
18:20.31greg_workoh btw, i was meaning to ask.. i attempted using fax with a SPA-2000, and although it kinda worked, and i could probably tweak it a bit, its just not worth the hassle to me .. if i was to get a TDM400P with an fxs port, would it work properly? is anyone doing faxing with fxs and fxo ports right in the * machine?
18:20.31dsmouseharryvv: vary state by state. In some states you have to have concent from all parties, some states only need one party to agree to it
18:20.32harryvvdahunter okay like " this call may be recorded for quality assurance" :)
18:20.38dsmouseconsent rather
18:20.43Hmmhesaysthanks, i didn't even want to attempt it if it were a pointless venture
18:20.50dahunterharryvv: Yes, but allow them to opt out if they object.
18:20.53Ayanoharryvv:  What kind of laws?
18:21.00jesse_132is asterisk.org down?
18:21.10dsmouseAyano: for recording phone calls
18:21.12Hmmhesaysit seems digiums site is down
18:21.22JerJerDNS is hosed
18:21.24dsmousedahunter: like to hang up?
18:21.31dahunterdsmouse: Sure ;)
18:21.36fishboy1669any one know if this is still a valid patch?
18:21.37jesse_132anyone know the ip to ftp.asterisk.org?
18:21.37fishboy1669http://www.mail-archive.com/asterisk-dev@lists.digium.com/msg04797.html
18:21.38Hmmhesaysmust be some sun spots
18:21.49dsmousehrms.
18:21.50Ayanodsmouse:  But what do they prohibit or whatever.
18:22.02jesse_132nevermind... voip-info has mirrors listed
18:22.36Hmmhesaysnow if someone had digium's ip that I could add to my host list
18:22.37dsmouseAyano: recording a phone call without consent of [one|all] of the party(s) involved?
18:22.39Hmmhesaysthat would be helpful
18:23.25JerJerHmmhesays: 69.16.138.164
18:23.35Ayanodsmouse:  Oh, I knew that, but as long as you are warned, or know that it could be recorded, it puts you in the clear though right?
18:23.41harryvvI think the police here dont say to the interviewed party thay are being recorded while in interagation then transcribe all there words on msword for legal reasons.
18:23.58yashaxGuys, anyone?
18:24.10AyanoCA doesn't either, and all highway patrol carry recorders.
18:24.10dsmouseharryvv: is that over the phone?
18:24.22harryvvdsmouse no thats at the police station.
18:24.35dsmousethen it's not recording a phone call :)
18:25.01Ayanodsmouse:  Oh, I knew that, but as long as you are warned, or know that it could be recorded, it puts you in the clear though right?
18:25.13dsmousesure
18:25.21dsmouseafaict
18:25.32dsmouseianal, btw
18:25.48Ayano?
18:26.06dsmousesure, as far as I can tell. I am not a lawyer, by the way.
18:26.48harryvvi would have no use for it though unless we got harrasment calls.
18:27.07AyanoI was just wondering.  You do have some sort of knowledge obviously.  Other wise I wouldn't ask.
18:27.18AyanoYou guys are great.
18:27.19harryvvof what
18:27.25*** join/#asterisk imagmo (~imagmo@c-24-20-249-117.client.comcast.net)
18:27.54harryvvI was hopping a package would arive today but its a holliday in the states and its hung in limbo.
18:27.54harryvv;)
18:27.55dsmouseAyano: I've read a few cases and stuff; nothing spectacular
18:27.56DJ-Pyrospeaking of which, is there an easy way to start recording the call if someone hits a key (say *) during the middle of a call?
18:27.57*** join/#asterisk visik7 (~ciao@host178-39.pool80182.interbusiness.it)
18:28.11*** join/#asterisk anto9us (~chatzilla@cpc3-ptal1-5-1-cust123.swan.cable.ntl.com)
18:28.19malcolmdbbiab...
18:28.33harryvvDJ you mean a hot key
18:28.34jesse_132DJ-Pyro: yep...
18:28.44jesse_132DJ-Pyro: but then it would be heard by the other party
18:28.58jesse_132DJ-Pyro: if you have a phone that supports hot-keys, you can make it silent...
18:29.21Ayanodsmouse:  That's more than I have.
18:29.37DJ-Pyrojesse_132: any guidance on how to begin something like that?
18:29.39harryvvbtw, asterisk does say anouncements in different languages?
18:29.46AyanoDJ-Pyro:  you can create a script to do it from a computer....
18:30.03jesse_132DJ-Pyro: you seen voip-info.org yet?
18:30.23DJ-Pyrojesse_132: yeah, I'm just looking for a keyword to search on
18:30.23Ayanoclick a button and it starts recording.
18:30.48*** join/#asterisk malcolmd (~malcolmd@malcolmd.digium.sponsor.pdpc)
18:30.55harryvvWe have a mail service carrier that gets alot of calls from BC and thay have a hard time understanding what thay are asking because of there accent. Mostly manadarin and punjabi are spoken.
18:31.01jesse_132it used to be done with a meetme hack...  Ayano is it just monitor now?
18:31.28dsmouseharryvv: yea...
18:31.28harryvvThay ask probebly simular questions about there mail or packages.
18:31.32*** join/#asterisk FuRR_ ([3N5VaQTxr@bko29.chapman.edu)
18:31.54harryvvdsmouse what options would that be
18:32.02bjohnsonAyano: I think you can record in Canada if the people are notified but that recording cannot be used in any kind of legal case (I'm not a lawyer either).  I'm pretty sure cops recording a witnesses statements are not valid either, they have to be transcribed and signed
18:32.14Ayanojesse_132:  I have only done it with a meetme hack,,,  but I from what I understand it is just as easy through monitor.
18:32.33AyanoI see.
18:32.34bjohnsonI think the only time recordings can be used as evidence is if it is done with a warrant
18:32.45dsmousebjohnson: no
18:32.45jesse_132so DJ-Pyro your keywords are meetme/monitor ;)
18:33.44dsmousebjohnson: it should be allowed as long as it's legal to make it and you[or someone] can testify as to it's origins.
18:33.49AyanoDJ-Pyro:  the setup to record a meetme is easy.  Just create a file with specific data and drop it to the spooler dir, and it starts the recording.
18:33.55dsmouseand that it hasn't been edited
18:34.03dahunterWhat's wrong with this: s,2,Monitor:wav:("/usr/pl/${EPOCH}.wav")
18:34.05Beirdogood luck proving that
18:34.21AyanoLooks right.
18:34.42mtqhno .wav?
18:34.57*** join/#asterisk sangee (ravi@209.250.129.135)
18:34.58dahunterDo I need to do something special to enable it, I keep getting: No application 'Monitor:wav:' for extension (incoming, s, 2)
18:35.01dsmouseBeirdo: if you heard it, and then you say "Yes, that sounds correct" it would probably be enough
18:35.05*** join/#asterisk Sedorox (brandon@Neptune.client.wlmsprt.pa.sed6.net)
18:35.18Beirdoheh
18:35.34bjohnsondsmouse: I'm not going to argue legal technicalities because frankly, I don't know anything about them
18:35.44mtqhdsmouse.... please to a show application montior
18:35.55Beirdo"Yeah, that sounds like what I remember from 2 years ago"...  that doesn't say it hasn't been modified, and I doubt you could testify to that
18:36.14Beirdodo you remember stuff like that perfectly after a long time?
18:36.16dsmousemtqh: erhm?
18:36.29Beirdolong time meaning >= 1 day
18:36.30jesse_132mtqh: do "show application monitor" in your console
18:36.45mtqhok so I can't spell
18:36.51dsmousebjohnson: true
18:36.59jesse_132mtqh: sorry, I meant to write that to dsmouse ;)
18:37.15dsmousebjohnson: this is just a friendly discussion, of course :)
18:37.18sangeeI want to send hard codeed ANI (always "9058049111") when i dial out how to do that?
18:37.37mtqhsangee  Show application setcallerid
18:39.50dsmouseanyway
18:42.15*** join/#asterisk malcolmd (~malcolmd@malcolmd.digium.sponsor.pdpc)
18:43.14*** join/#asterisk yogurt2ungue (~charlie@203-132-126-200.fibertel.com.ar)
18:44.24|Vulture|anyone else unable to resolve proxy.mia.boradvoice.com?
18:44.41kpfleming'mia' seems very appropriate
18:44.50Sedorox[13:44] Host proxy.mia.boradvoice.com not found: 3(NXDOMAIN)
18:44.56dsmouseboradvoice?
18:45.08Sedorox[13:44] Host proxy.mia.broadvoice.com not found: 3(NXDOMAIN)
18:45.11dsmousenot broadvoice?
18:45.11Sedoroxer
18:45.19SedoroxI just copied what he said
18:45.21dsmouseproxy.mia.broadvoice.com has address 147.135.4.128
18:45.24Sedoroxhe spelled it wrong..
18:45.32Sedoroxyea.. I get the same thing
18:45.34|Vulture|hahaha
18:45.36Sedoroxwhen spelled right :-p
18:45.36Zeeekanyone tried chanspy?
18:45.44|Vulture|they have it spelled wrong in their install info
18:45.50|Vulture|http://www.broadvoice.com/support_install_asterisk.html
18:45.54Sedoroxinteresting
18:46.03|Vulture|thanx for pointing that out... I just did a copy/paste
18:46.08Sedoroxajaha
18:46.09Sedoroxnice
18:46.12Hmmhesaysno, but you can write a script pretty easily to transfer 2 calls into a meetme room
18:46.35Sedoroxhehe, then they gget the little noise that they've joined
18:46.50Hmmhesaysturn the little noise off
18:46.58Hmmhesaysyou can transfer to meetme silently
18:47.27Sedoroxinteresting
18:47.38Hmmhesaysin fact if you want a gui based way to do it, check out FOP
18:47.47Hmmhesaysyou can barge in on a call easily
18:49.45Hmmhesaysif you want to do it with an extension... it's pretty easy with an agi, you can send a manager command to redirect to meetme
18:51.51*** join/#asterisk denon (denon@synapse.subneural.net)
18:51.51*** mode/#asterisk [+o denon] by ChanServ
18:51.59FuRR_Hmmhesays: is there anyway you can do a barge without MeetMe
18:52.06FuRR_say, you want to interupt a call in progress
18:52.36ManipuraAnyone know where I can find more info on mysql realtime other than the wiki?
18:52.45Hmmhesaysand have 3 way calling?
18:53.01Hmmhesays* 3 way conference
18:53.10*** join/#asterisk sysdebug (~jonasgoes@200.163.193.247)
18:53.30bkw_can anyone else get to bugs.digium.com?
18:53.34FuRR_Hmmhesays: no, more of an administrative operator thing
18:53.47Hmmhesaysyou want to barge in and listen to a call right?
18:53.50denonbkw: doesnt look promising
18:53.57FuRR_like with the pstn where an operator can barge to tell you that someone is trying to get through
18:54.01denonbkw: especially since it doesnt resolve
18:54.11Hmmhesaysuse meetme
18:54.12bkw_OH
18:54.14Hmmhesaysit's not hard
18:54.23ZeeekI'm getting bugs.digium.com
18:54.34denonZeeek: you must have it cached
18:54.43Zeeeknot me - but maybe provider
18:54.48denonthe dns, that is, not the content
18:54.54Hmmhesaysunless you want to brush up on your programming skills and make chanspy work
18:54.55Zeeekya
18:54.56denoncontent is probably just fine
18:55.02denonwhats the ip?
18:55.07ZeeekAPPLICATION ERROR #1100
18:55.07ZeeekBug 2379 not found.
18:55.08Zeeekactually.... "APPLICATION ERROR #1100"
18:55.37Zeeek[69.16.138.164]
18:55.52bkw_or you can pay anthm to install chanspy
18:56.04Zeeekdoes chanspy work?
18:56.07Hmmhesaysdoes chanspy work?
18:56.10bkw_yes
18:56.12HmmhesaysLOL
18:56.19Zeeekdoes it? well ? well?
18:56.36Zeeekdoes it make beds, clean floors
18:56.36bkw_yes
18:56.39bkw_no
18:56.42denonbkw: echo 69.16.138.164  bugs.digium.com >> /etc/hosts :)
18:57.12Zeeekone of the few times DNS cacheing helped ;)
18:57.42denonzeeek: yeah .. looks like all of digium's dns is borked
18:57.54bkw_and the fun part is I really do have /etc/hosts on my desktop machine now
18:57.56Zeeekshit happens
18:58.09jesse_132woot
18:58.38denonhmm
18:58.46denonit looks like he forgot to renew digium.com, until today
18:59.04denonso maybe not so much dns, as registrar
18:59.05Zeeekdid't this already happen once?
18:59.11denonthink that was asterisk.org
18:59.24Zeeekah yes - you geek-devel guys are sooooo lax on that shit :)
18:59.32denondont look at me
18:59.35Zeeekcan't afford $8 a year
18:59.37Sedoroxdigium is still register
18:59.47denonSedorox: they renewed today I think
18:59.52denonand you mean Registrant
18:59.55Sedoroxwell I got whois info back...
19:00.02Sedorox[13:59]       Expires on: 21-FEB-06
19:00.02Sedorox[13:59]       Last Updated on: 01-FEB-05
19:00.38redder86Why the *heck* are they only registering for a year at a time?!?
19:00.42denonhmm .. you know ..
19:00.46denonall the nameservers are borked ..
19:00.49sangeeit's working thanks (mtqh)
19:00.50denonit may actually be a real dns issue
19:00.55*** join/#asterisk Zaw (zaw@zaw.subneural.net)
19:01.17Sedoroxyea.. all 7 dns servers went down :-p
19:01.43denonactually, more likely the zone got messed up
19:01.51denonby the at-a-glance looks of things
19:02.07Zeeekmust be using the GUI to edit the zone
19:02.10Sedoroxhmmm
19:02.23malcolmdhi, yes, we're down.  we're doing what we can, but right now we're waiting on our CLEC
19:02.31Zeeekin the meantime there's always http://www.digium.net/
19:02.59Sedoroxweird
19:03.05Zeeekand http://www.digiumresearch.com/
19:03.24ManipuraWhat * hardware do I need when I get a pri?
19:04.18djinManipure, are you for real?
19:05.43djinYou might want to check Digium for the Wildcard TE110P, TE405P and TE410P.
19:07.01ManipuraThanks
19:07.04bjohnsonManipura: you might also want to consider what hardware you need for your phones
19:07.04*** join/#asterisk klasstek (~peracles@sta-206-168-231-55.rockynet.com)
19:07.15*** topic/#asterisk by denon -> Asterisk: The Open Source PBX || Dev Conf 1PM CST FEB 24th -> IAX2/guest@66.250.68.194/996 || Digium's having some outtages - major Internet outtage in their area, please be patient.
19:07.46malcolmddenon: thanks :)
19:07.47*** topic/#asterisk by denon -> Asterisk: The Open Source PBX || Dev Conf 1PM CST FEB 24th -> IAX2/guest@66.250.68.194/996 || Digium's having some voice and data outtages - major Internet problems in their area, please be patient.
19:07.55denonvoice and data, fyi
19:07.56bjohnsondenon: why not list IP if it's just a dns issue?
19:08.10Manipurabjohnson, softphones and DID's
19:08.12klasstekAnyone dealt with voipsupply.com for purchasing phones?
19:08.20bjohnsonManipura: ewww
19:08.20DJ-Pyroklasstek: yes
19:08.26malcolmddigium.com + asterisk.org = dns issue.  digium's voice circuits = waiting on clec to fix us up.
19:08.29*** topic/#asterisk by denon -> Asterisk: The Open Source PBX || Dev Conf 1PM CST FEB 24th -> IAX2/guest@66.250.68.194/996 || Digium's having some voice and data outtages - major Internet problems in their area, please be patient. (you can use http://69.16.138.164 temporarily)
19:08.31bjohnsonklasstek: damn near eneryone'
19:08.54klasstekgood, bad or indifferent?
19:08.56stevekstevekkinda sucks that CVS is unavailable... :(
19:09.08bjohnsonklasstek: good
19:09.09DJ-Pyrowe have a 7700USD order pending
19:09.11bkw_clec is on crack
19:09.13DJ-Pyrovery good klasstek
19:09.15jaigerklasstek, I've purchased polycom phones, worked for me
19:09.15malcolmdstevekstevek: ja, sorry. :(
19:09.19bjohnsonklasstek: gotta watch shipping charges though
19:09.27klasstekThanks
19:09.35malcolmdbkw_: yup, especially since they've handed the problem off to the rboc now
19:10.53klasstekjaiger: Any exp with the Polycom IP 4000 conference station?
19:11.08jaigerklasstek, no, I use IP300, IP500 and IP600
19:11.14jaigermostly IP500
19:11.34klasstekthx
19:11.48klasstekAnyone else tried the Polycom IP 4000?
19:12.00Sedoroxdamn
19:12.04jaigerklasstek, but I'm happy enough with polycom to stick with them
19:12.31Zeeekwhat power supplies do polycoms come with?
19:12.53*** join/#asterisk twisted[work] (~twisted@twisted-professional-pdpc.developer.and.friend.of.asterisk)
19:12.53*** mode/#asterisk [+o twisted[work]] by ChanServ
19:12.57twisted[work]digium fall down go boom?
19:13.00jaigervoipsupply gives a choice, some have PoE and some use wall warts
19:13.09DJ-Pyrotwisted[work]: topic
19:13.11Sedoroxread topic
19:13.17twisted[work]DJ-Pyro, no shit sherlock.
19:13.21ZeeekI'm asking because I'd need a wallwart and one that works with 220v
19:13.38jaigerthe PoE ones use wall warts to inject into the PoE cable
19:13.41twisted[work]I'm just trying to figure out what happened
19:13.44twisted[work]I already knew they were out
19:13.45doughecka_bwuhahaha
19:13.56doughecka_SBC is having "major troubles"
19:14.00doughecka_no outgoing calls
19:14.02malcolmdtwisted[work]: ahoy
19:14.02twisted[work]they're not SBC
19:14.08twisted[work]hey malcolmd, whassabi
19:14.12doughecka_fast busy, but incoming calls come in
19:14.40doughecka_no, this is me
19:15.00doughecka_new albany/Jeffersonville, IN
19:16.52*** join/#asterisk implicit (~implicit@ip68-5-148-1.oc.oc.cox.net)
19:17.12twisted[work]ahh.
19:17.19twisted[work]these storms are rocking our boat
19:17.20jaigerZeeek, my polycom wall warts are 120VAC to 12VDC
19:17.32Zeeekok thx
19:18.28Zeeekanayone here dealt with atacomm ?
19:18.50denonyep
19:18.58Zeeekand? any good?
19:19.03denonyeah, nice folks
19:19.07Zeeekare you in MN?
19:19.16denonyep
19:19.19denonnot close to them, though
19:19.35ZeeekI was born in Mpls - coming there to visit and noticed they are located in Maple Grove
19:19.39*** join/#asterisk _tekati_ (~captain@cpe-66-75-215-63.bak.rr.com)
19:19.43denonah, yep
19:19.46Zeeekso my wife will never see what hit her :)
19:19.53denonhehe
19:19.56denonnice area
19:20.00Zeeek"I think I'll take a quick ride over to Maple Grove"
19:20.14mikegrbZeeek:
19:20.16mikegrber
19:20.19mikegrbZeeek: I so would
19:20.21ZeeekMy step bro lives there. I'll park her there and disappear with the checkbook
19:20.35denondont think he has a retail location..
19:20.39mikegrbZeeek: wife may tell me I have to walk home but at least I'd have goodies
19:21.03Beirdotwisted[work]: where are these storms?
19:21.06ZeeekWell the address is published, I'd guess pickup may be possible - Minnesota nice and all
19:21.16denonyep, see your msgs
19:21.56Zeeekdenon was that to me?
19:22.00denonyeah
19:22.03LoganAll the prompts that asterisk plays are way too loud, compared to the level we hear when actually talking to someone on a bridged phone.
19:22.09ZeeekI have all queries turned off
19:22.14twisted[work]Beirdo, midsouth
19:22.17LoganDespite trying to force the wave files we're using to have a lower volume, they remain way too loud.
19:22.33denonah .. well, I just mentioned to give Dan a call, and tell him that denon told him to let you in :)
19:22.40Beirdoahh.  so my friend in Tennessee is likely having fun then
19:22.40*** join/#asterisk thefallen (PolarBear@thefallen.user)
19:22.41Zeeekhaha
19:22.52Zeeekwhere are you denon? Not TC?
19:22.55*** join/#asterisk abernathy (~abernathy@c-24-98-249-157.atl.client2.attbi.com)
19:22.59denonbit south
19:23.07ZeeekStCloud?
19:23.10denonI get up to tc fairly often though
19:23.11Zeeekno that's north...
19:23.17denonI get to st cloud often too
19:23.17denonhehe
19:23.24Zeeeklemmies see
19:23.27Zeeekummmm
19:23.42denondont worry ..
19:23.45Zeeeklet me guess
19:24.02Zeeekfarmington?
19:24.25*** part/#asterisk Edgan (~edgan@okcforum.org)
19:24.31denonnah
19:24.38Zeeekcloser?
19:24.44Beirdoof course if this were the world of 24 or something, they could track you to a precise location in your building just from your IP address :)
19:25.01abernathyCan anybody help me figure out why audio on incoming calls doesn't seem to work? (When I do an echo test, or call vm, nothing ever gets sent to the other end)
19:25.01denonBeirdo: yeah .. except my proxy server is in texas
19:25.14Beirdohehe, that's nothing for the CTU, of course :)
19:25.17denonoh, no, its in pennsylvania now
19:25.31Zeeekabernathy SIP? NAT?
19:25.39abernathyyes and yes
19:25.43Beirdooh oh, moving your proxy might be considered terrorist activity
19:26.02Zeeeknat=yes canreinvite=no ?
19:26.13Zeeekwhat phone?
19:26.41abernathyxten xlite, and cisco 7960, and yes on both of those
19:26.52Zeeekdid you set Transmit SIlence to YES?
19:26.59Zeeekon X-Lite?
19:27.12abernathyno, it works with other voip (fwd)
19:27.30Zeeek*and set the RTP ports to the * range?
19:27.50Zeeekfwd will be using STUN though, no?
19:29.01Zeeekabernathy better to stay public - with luck someone who knows what they're talking about will take over :)
19:29.15ZeeekI'm getting near the rim of my knowledge galaxy
19:29.26abernathyoops :-)
19:29.43ZeeekBut I have used X-Lite with * behind NAT on both ends and Xmit silence had to be YES
19:29.44abernathynot sure about the RTP thing
19:29.54Zeeekset the X-Lite to use 10000
19:30.07Zeeek(or change asterisk if you want)
19:30.09abernathyok, one sec
19:30.28Zeeekdenon, mankato, altbert lea... ?
19:30.40Zeeekwasn't there an asterisk meeting in the TC recently?
19:31.42denonprobably... Ive been too busy to even think about going
19:31.43*** join/#asterisk Frantic (~ab@24-193-46-85.nyc.rr.com)
19:32.02ZeeekI make Mark himself come over here and buy us wine in Paris
19:32.12Zeeekmuch more productive meetings
19:32.44denonyou're in paris?
19:32.51BeirdoMmmm.  wine
19:32.53Zeeekyes sir
19:33.01Zeeekthere is no lack of wine here
19:33.02denoncourse I bet he'd be more reluctant to do so, if he didnt have family there
19:33.12Zeeekindubitably
19:33.21Zeeekbut there is a good group here
19:33.24denonI havent been to paris in a long time
19:33.31Zeeekfierce open-source guys
19:33.36BeirdoI've never been
19:33.37Zeeekmilitant even
19:33.56Beirdoguh.  that kind of person tends to annoy me after a few hours
19:33.59denonyou know.. the one admin I do know in Paris, is a real stuck-up bast..er, bastille-lover
19:34.06harryvvAnyone know of a price compedative closed case wall mount atx case without paying commercial rates?
19:34.21harryvvI have seen some with lockable front covers.
19:34.52Zeeekheh, well we mustn't generalize - there's good and bad everything - even cops
19:34.54abernathyOk, I think I have some other issues with this... might not be my server. might be the network I'm currently on
19:35.03denonZeeek: agreed
19:35.08denonwell, dunno bout cops .
19:35.08*** join/#asterisk phantam (~phantam@63.210.60.199)
19:35.12denonbut that we shouldn't generalize
19:35.12phantamhey guys
19:35.21Zeeekgood luck abernathy - I need to pretend I haven't been online for the last three hours
19:35.21phantamasterisk been working great
19:35.24denonthats nice
19:35.25denonNEXT
19:35.31Zeeekheh
19:35.37phantami have a hidden extension playing --- southpark-kyles mom is a big fat *****
19:35.37phantam:)
19:35.55Zeeek'night all - thanks for the info denon - I'll be there in May and I'll "take a ride"
19:36.05phantamhowever
19:36.07phantami keep getting
19:36.09denonZeeek: sounds like fun, drop me a msg before ya leave
19:36.10phantamwrapendpoint.cxx:716: error: 'class H323AudioCodec' has no member named 'IsDescendant'
19:36.16denonand I'll see what my sched is
19:36.18*** join/#asterisk ZX81_laptop (~chatzilla@81-208-60-207.fastres.net)
19:36.20*** part/#asterisk Zeeek (~Zeeek@Zeeek.sustaining.supporter.pdpc)
19:36.21phantamwhen trying to compile oh323 for some reason
19:36.32*** join/#asterisk Zeeek (~Zeeek@Zeeek.sustaining.supporter.pdpc)
19:36.44denonback so soon?
19:36.46ZeeekOops slipped - denon I read that last - ok :) later
19:36.50*** part/#asterisk Zeeek (~Zeeek@Zeeek.sustaining.supporter.pdpc)
19:37.52phantamdenon any idea what that could be?
19:38.24denonnope, I never use h.323
19:38.30denonmake sure you're running current everything
19:38.36denonand try make-cleaning everything
19:38.37denonetc
19:38.38denonthe usual
19:38.43phantamhmmm
19:38.51phantamdamn portage why cant it have the up to date stuff
19:39.14Hmmhesaysphantam: did you follow the instructions for which versions to use?
19:39.34phantamwhich instructions are you refering to
19:39.41phantami kinda had to fool asterisk into installing
19:39.48phantambecause i installed from portage
19:39.54phantamand then was getting adsi errors for some reason
19:40.05Hmmhesaysthe ones in the oh323 readme
19:40.12phantamso i coppied all the modules and conf files from a working distro of asterisk
19:40.18phantambut... it didnt have h323 or iax
19:40.23Hmmhesaysoh323 is extremely easy to install
19:40.28Hmmhesaysif you follow the instructions
19:40.29phantamand when i install oh323 from portage
19:40.32phantamemerge asterisk-oh323
19:40.35phantami get
19:40.38phantamwrapendpoint.cxx: In member function `virtual BOOL WrapH323EndPoint::OpenAudioChannel(H323Connection&, BOOL, unsigned int, H323AudioCodec&)':
19:40.42phantamwrapendpoint.cxx:716: error: 'class H323AudioCodec' has no member named 'IsDescendant'
19:40.44phantamwrapendpoint.cxx:717: error: 'class H323AudioCodec' has no member named 'IsDescendant'
19:41.06mikegrbthat would be because portage sucks
19:41.13phantamasterisk first program i run into that has such bad implementation in portage
19:41.18phantamlol never failed me before
19:41.34Hmmhesaysyeah well 50 dollar hookers never failed me either.. till last time
19:41.35mikegrbit fails all the time
19:41.40mikegrband is slooooooooow
19:41.50phantamhow is it slow?
19:41.58mikegrbfor an "optomized" distribution, why is portage so slow
19:42.07mikegrbever do an emerge rsync?
19:42.16phantamu mean emerge sync
19:42.17*** join/#asterisk HiB (~jsgehris@ip68-9-56-18.ri.ri.cox.net)
19:42.29phantamtakes about 1.5 mins on my server
19:42.43mikegrbI'm not talking about the transer
19:42.47phantamstill doesnt tell me how to get iax and h323 to install
19:43.04phantami know what u mean im talking the whole thing from command prompt to command prompt
19:43.07mikegrbI'm talking about the cpu time it takes to "recalculate dependencies" and what not
19:43.07phantam1.5 mins
19:43.16mikegrbI doubt it
19:43.22mikegrbat any rate, it is crap
19:43.28phantamlol
19:43.31mikegrbit's a 50,000 line python script
19:43.35phantamso how should i install h323
19:43.43mikegrbpython because they can't learn to write real code
19:43.56mikegrbyou shouldn't use h323, it's almost as crap as gentoo
19:44.27|Vulture|is $430 good for a 12line PRI?
19:44.52mikegrb|Vulture|: really depends on the location
19:45.05mikegrb|Vulture|: for some places that is quite good, other places possibly cheaper but not bad
19:45.13phantammikegrb: weather it is crap or not is not my business it is what the person thats owns the server needs so im doing it
19:45.21mikegrbweather?
19:45.27mikegrbit's sunny out and I have the AC on
19:45.31mikegrbhow's the weather there?
19:45.35*** join/#asterisk HiB (~jsgehris@ip68-9-56-18.ri.ri.cox.net)
19:45.40|Vulture|mikegrb: thanx
19:45.43doughecka_mikegrb: crap
19:45.45mikegrb|Vulture|: $35/line is pretty good
19:45.54mikegrbdoughecka_: :<
19:46.11mikegrb|Vulture|: what's the location and provider, out of curiousity
19:46.22|Vulture|mikegrb: and a full pri was quoted at $650 but we don't need all 23 lines
19:46.32|Vulture|Jacksonville, FL; Nuvox
19:47.01mikegrbahh, I'm in pensacola
19:47.07|Vulture|we have had a Fract T1 Data/Voice for the past 3 years
19:47.32mikegrb35/line is definitly cheaper then analog lines
19:47.54|Vulture|but with * going in there I want to put a PRI, and they are suppose to quote me on 786K + Frac PRI
19:48.19|Vulture|yea plus it includes 1000mins of LD
19:48.31mikegrbahh
19:48.40mikegrbwell sounds like a pretty good deal
19:48.44|Vulture|yea
19:48.50mikegrbI like jacksonville
19:49.00mikegrbespeacially the people mover downtown
19:49.07mikegrbwent there just to ride it
19:49.08|Vulture|I actually go to school down in Orlando
19:49.12|Vulture|hahaha
19:49.34|Vulture|the city looks really nice right now because of the superbowl
19:49.42mikegrbhttp://thegrebs.com/~michael/pictures/jax/jax.html
19:49.44|Vulture|they had all the lights on
19:50.16mikegrbhttp://thegrebs.com/~michael/pictures/jax/jax-Pages/Image10.html <-- I like this one
19:50.18|Vulture|mikegrb!! D70
19:50.26mikegrbhave a 20x30 print of i  on the wall
19:50.27*** join/#asterisk MichaelVanD (~MichaelVa@rrcs-24-123-121-190.central.biz.rr.com)
19:50.31|Vulture|http://www.the-vulture.com/gallery
19:50.31mikegrbs/i/it/
19:50.51|Vulture|that is a nice shot
19:51.01|Vulture|you using the kit lens?
19:51.11phantamwhats a good webinterface for asterisk
19:51.12phantam?
19:51.22phantamvoxbox looks aight but shitload of requirements
19:51.29mikegrbMythbusters Q&A!
19:51.40|Vulture|:)
19:51.42mikegrbyeah, have the kit lens and a 70-300mm nikor
19:52.06|Vulture|ah, I have the 70-200VR, kit, 85 and 50
19:52.18|Vulture|the 85 is one of my fav lenses
19:52.40bjohnsonjust for line cost comparisions, we're paying $33 CAD / line for analog lines
19:53.21|Vulture|I didn't take those Mythbuster shots, they were done by my friend... hence the crappy pics :(
19:54.57mikegrbhttp://www.the-vulture.com/gallery/displayimage.php?album=lastup&cat=12&pos=0 <-- details? use filters on that one?
19:55.26mikegrb1/60 sec exposure so I assume a flash, built in or a speed light?
19:55.34mikegrbI've been damn happy with the SB-800
19:55.58mikegrbI cracked the lcd when I was in jax though so it is a pain in the ass to switch back in forth between remote slave and TTL
19:56.31*** join/#asterisk ACiDV (~joel@69.156.197.246)
19:57.34|Vulture|mikegrb: kit lens, circular polar filter
19:57.44|Vulture|oh yea the SB-800 is a god
19:58.05ACiDVHmm I have made a CVS (zaptel, libpri, asterisk, ...) update, make clean/install this morning... I load my TE405 drivers, no problem, I check status with zttool and all channel are now OK... no link connected. I must see a red alarm true ?
19:58.25mikegrbahh, polarizer's are great, at sunset I assume?
19:58.31*** join/#asterisk SeaForth (~SeaForth@c-24-1-126-202.client.comcast.net)
19:58.54heison|Vulture|: what camera do u have? D70?
19:59.20|Vulture|mikegrb: yes and great at eliminating overexposure due to reflections
19:59.30mikegrbindeed
19:59.35|Vulture|heison: yes mikegrb has one as well we were just discussing
19:59.47heisoni have it too ;)
19:59.56|Vulture|haha such a great camera
19:59.57heison18-70, 28-70, 70-200VR
20:00.12|Vulture|heison: you have a film camera too?
20:00.13heisoni want to ditch my 18-70 and get 17-35
20:00.24heisonno.. i have d70
20:00.44mikegrbI want some VR lenses
20:00.46|Vulture|ah why the 28-70 and a 18-70?
20:00.50mikegrbbut had a kid so... bleh
20:01.00heisonwith a new shutter ;) Nikon Canada replaced mine -- it's worned out
20:01.00|Vulture|I have the 70-200 as well... its a work of art
20:01.21|Vulture|heison: how many shots did you put through your d70?
20:01.25mikegrbhttp://thegrebs.com/~michael/pictures/hunter/hunter.html
20:01.45Beirdoand of what calibre bullet?
20:01.58|Vulture|mikegrb: did you use the sb-800 for those?
20:02.08mikegrb|Vulture|: ja
20:02.27heison24k shot when it was replaced
20:02.32mikegrbhttp://thegrebs.com/~michael/pictures/fireworks/fireworks-Pages/Image8.html <-- I like the fireworks pictures too but they weren't with the D70
20:02.32|Vulture|that omni bounce is great for making pictures look real to life
20:02.49mikegrbindeed
20:03.29heisoni now have close to 26000 shots on it total
20:03.35|Vulture|hahah nice croc shot
20:03.44|Vulture|heison: how much was the replacement?
20:03.47mikegrbhttp://thegrebs.com/~michael/pictures/remote/remote.html <-- I'm proud of these nice night shots too
20:03.54mikegrbthe croc shots were for the paper
20:03.57heisonit was 'free', 2 yrs warranty
20:04.14heisoni intend to bring it back before the 2 years for another replacement
20:04.25|Vulture|mikegrb: I like that #1 night shot
20:04.34|Vulture|no blooming... good work
20:04.44mikegrb|Vulture|: those were with the canon powershot
20:04.58mikegrball my night shots I take my 12" ibook with me and use it to trigger :D
20:05.09|Vulture|I have a S410 as my backup/party cam
20:05.11BrianR___Funny.. This channel suddenly became #photography.. I thought my irc client was broken or something..
20:05.20|Vulture|lol
20:05.40BrianR___I use a d70 also. Love that camera.
20:05.46mikegrb:O
20:05.50Beirdohey mikegrb: why is there a customs agent with the croc?  was it smuggling drugs?
20:05.50|Vulture|wow thats #4
20:05.55mikegrbthis is the #d70 channel!
20:06.05|Vulture|Asterisk users love D70s!
20:06.22heisonchan_d70
20:06.40heisontime to write one ;)
20:06.54|Vulture|lol
20:07.17|Vulture|all my hurricane pics were on my S410
20:07.52mikegrbmy son was born just after the huricane with flashlights!
20:08.14heisonsome of my D70 pics... http://photos.zealnetworks.com/Clara_and_Heison/Gallery/
20:08.21|Vulture|wow...
20:09.06mikegrbI need to upload my mardi gras pictures
20:09.07*** join/#asterisk mbranca_home (~matteo@host-84-222-6-8.cust-adsl.tiscali.it)
20:09.08|Vulture|heison: http://photos.zealnetworks.com/Clara_and_Heison/Gallery/index.php?image=20040603-113541.jpg&d=d.html where is that?
20:09.11BeirdoI still take more film pics
20:09.41heison|Vulture|: Santorini, Greece
20:09.42Beirdohttp://pics.beirdo.ca/gallery/
20:09.56|Vulture|I have a N80 and use Velvia 50 on it... great for when you really want a pic
20:10.02Beirdoone of these days I want a good Nikon digital to use my F65's lenses on :)
20:10.28|Vulture|Beirdo: F65?
20:10.52Beirdoyes, I believe that's the model
20:11.08mikegrbBeirdo: you cane buy me a nikon D2H and have my d70
20:11.20Beirdoheh
20:11.25heisoni want D2X
20:11.28|Vulture|Beirdo: is it a large metal camera about 7 years old?
20:11.33Beirdono
20:11.41Beirdoit's a film SLR
20:11.44Beirdoabout 4 years old
20:11.53|Vulture|oh I guess there is a F65...
20:12.01|Vulture|I was thinking of the F5 and F6
20:12.06Beirdo:)
20:12.28Beirdohttp://pics.beirdo.ca/gallery/photo.php?photo=1624&exhibition=29
20:12.33Beirdommmm, I wanna go back
20:12.56|Vulture|Beirdo: what film do you shoot?
20:13.11BeirdoFuji 35mm, usually 200ASA
20:13.37Beirdofor some of the pics at the Toronto Molson Indy one year, I used Fuji 1600ASA :)
20:13.41|Vulture|yea I like Fuji too
20:13.54Beirdopricey stuff, but MAN did it do a good job
20:15.01mikegrbBeirdo: nice pics
20:15.08BeirdoThanks.  :)
20:15.09|Vulture|dell is taking their sweet time to build my new server... :(
20:15.14mikegrbBeirdo: film is harder, don'g get instant feedback to tweak ;)
20:15.17Beirdoit's hard to screw up Colorado though
20:15.28Beirdoyeah, you have to get used to framing the shots
20:15.29|Vulture|you guys ever shoot medium format?
20:16.20*** join/#asterisk randu (~randu@pool-141-151-118-76.scr.east.verizon.net)
20:16.30randuHello Oeveryone
20:16.34randuEveryone
20:16.34*** join/#asterisk angler_ (~angler@207.111.168.75)
20:16.43angler_grr
20:16.45*** join/#asterisk ckruetze (~ckruetze@i3ED6843F.versanet.de)
20:16.45|Vulture|hey
20:16.53randuI am getting this when using parked calls, trying to: == Spawn extension (vi, 710, 3) exited non-zero on 'Parked/SIP/147.135.0.129-0855f070<ZOMBIE>'
20:17.10randuany ideas
20:18.11*** join/#asterisk dsfr (~dsfr@207.111.168.75)
20:19.02*** join/#asterisk nextime (~nextime@ns0.nexlab.net)
20:21.12harryvvwhat does a u1 chasis case run these days?
20:22.11*** join/#asterisk dca[laptop] (~dca[lapto@sta-207-174-139-178.rockynet.com)
20:24.34PyroSteveman, i iax.conf is confusing for me
20:24.40*** part/#asterisk sysdef (~sysdef@pD9561E44.dip.t-dialin.net)
20:24.51*** join/#asterisk SeaForth (~SeaForth@c-24-1-126-202.client.comcast.net)
20:26.22PyroSteveif I am registering with a server via iax
20:26.40PyroStevedo i need to define a section for the host as well ?
20:28.11Sedoroxno.. just the remote server
20:30.24*** join/#asterisk drmac (~drmac@64.72.107.1)
20:30.46*** join/#asterisk phantam (~phantam@63.210.60.199)
20:30.55drmacwhere can i get a polo shirt with the Asterisk logo on it?
20:31.44phantamshido u there?
20:34.02dsmouseERROR[13760]: cli.c:50 ast_cli: Out of memory
20:34.21phantamwhere did shido go
20:34.49dsmouse'help show config handles' was the command, btw
20:34.53bjohnsonBeirdo: engineer?
20:35.05Beirdoyup
20:35.10BeirdoElectrical... why?
20:35.15bjohnsonOttawaU civil
20:35.28*** join/#asterisk chaoscon (~ph33r@chaoscon.user)
20:35.29BeirdoUWaterloo Electrical.
20:35.34Beirdo:)
20:35.45heisonBeirdo: UW CS, Ryerson Electrical
20:35.53bjohnsonI have 1 borther UWaterloo Electrical and 2 Uwaterloo computer
20:36.13bjohnson4 boys.  All geeks
20:36.25Beirdohehe, geek is fun
20:36.36Beirdobjohnson any of them '97?
20:36.57heison97 ID or 93 ID?
20:37.12dsmousein school there were ads for fraternities that said "go greek", one of my friends removed the r
20:37.14bjohnsonI think Brendon in computer was 97
20:37.15Beirdo'97 grad year
20:37.23LoganCan anyone tell me how to make Playback and Background playback wave files more quietly?
20:37.25BeirdoBrendon's your brother.
20:37.27BeirdoOh jeez
20:37.30bjohnsonyes
20:37.33Beirdoheh
20:37.51Beirdocraziness
20:38.05bjohnsonwith a little charm
20:38.48shmaltzLogan, you got the # changed for transfers?
20:38.57BeirdoI definitely remember him...  can't place a face at the exact second, but that was almost 8 years ago now
20:39.16bjohnsonspeaking of telecom .. he works for Qualcomm in SD
20:39.17phantamhmmm
20:39.18*** join/#asterisk mrempire (~user1@h71032.upc-h.chello.nl)
20:39.19phantamhe left?
20:39.30Beirdostill?  didn't he go there right after grad?
20:39.38Beirdoalong with several of his classmates
20:40.42Beirdowhat a small world it is.
20:41.20bjohnsonyeah
20:43.15*** join/#asterisk file (~file@mctn1-1987.nb.aliant.net)
20:43.28SexyKenHey guys.
20:43.38Beirdoif you met many of his friends, you likely know quite a few people I know :)
20:44.02SexyKenI have a problem I need to fix right away. When people call into my Asterisk box and get entered into a queue, they go to voicemail if the agent is taking a call instead of holding them in the queue.
20:44.09SexyKenI just dont know why this is happening.
20:44.10Poincarehow can i check how many licenses are in use for g729 or what codec a channel is using?
20:44.29Loganshmaltz: I changed it so '##' just sends a '#' tone.
20:45.35bjohnsonI think show channels will show the codec
20:46.46*** part/#asterisk djin (~djin@gridfox.xs4all.nl)
20:46.48SexyKenAnyone know why this would happen?
20:47.27shmaltzI'm having problems with call parking? when doing an attended transfer.
20:48.58malcolmdokay, I think we're back...
20:50.51JerJeryay
20:51.03mutilatorhttp://ned.ucam.org/~sdh31/misc/destroy.html
20:51.05mutilator:P
20:52.10*** join/#asterisk scrubb (~scrubb@OCI-19-41.onecall.net)
20:52.31JerJerPoincare:   g.729 show licenses i think
20:52.38JerJerjust type g.729 and press tab
20:53.03jetsg729unlock
20:53.08Poincaretab does nothing :-(
20:53.18drmac"show g729"
20:53.18SexyKenI have a problem I need to fix right away. When people call into my Asterisk box and get entered into a queue, they go to voicemail if the agent is taking a call instead of holding them in the queue.
20:53.21SexyKenAnyone know why this would happen?
20:53.27Poincareah ok :-)
20:53.36Poincareshow g729, thanks JerJer
20:53.44drmac??
20:53.52Poincaredrmac: you too :-)
20:53.56drmac:)
20:54.28JerJeri new g.729 was in there somewhere
20:54.32JerJer+k
20:56.07HiBexit
20:56.13HiBexit
20:56.21HiBexit
20:58.13yashaxGuys, what is the menu command to reboot Polycom IP500?
20:59.49snewpyyashax: vol+, vol-, hold and messages, iirc
20:59.57PoincareJerJer: is it normal that SIP/RTP doesn't work when I have 2 IP's on a interface?
21:00.59*** join/#asterisk PoWeRKiLL (~PoWeRKiLL@l02m-29-240.d4.club-internet.fr)
21:01.11thieumShas anyone experience with Digium TE405P ?
21:01.28thieumSand 2.6
21:04.04yashaxsnewpy: Awesome, thank you!!!
21:04.23*** join/#asterisk DonX (don@tool.sparkhosting.net)
21:04.33DonXHow can I find out what timer asterisk is using?
21:04.45Sedoroxin the console
21:04.46Sedoroxshow versioj
21:04.48Sedoroxversion
21:05.27DonXAsterisk CVS-HEAD-02/21/05-14:51:31 built by root@pbx-svr1 on a i686 running Linux
21:05.29DonX?
21:06.04*** join/#asterisk Uajal (~icechat5@ool-182e86f3.dyn.optonline.net)
21:08.10Sedoroxyour running CVS version of asterisk
21:08.17Sedoroxbuilt today
21:08.27DonXyes, I just CVSup'ed
21:08.41DonXI'm trying to chase down an issue and I'm trying everything
21:08.43UajalI can call to SIP phone (SIP/2001, 20, Tr). How should I call to external phone, to the cell e.g. 646-3948? SIP/?????????????
21:09.05jetsUhm zap/1/646-3945
21:09.15bjohnsonif you have a zap device
21:09.18jetscorrect
21:09.18UajalI have no ZAp card
21:09.26jetsThen you need an iax/sip provider
21:09.28bjohnsonUajal: what device are you expecting to use?
21:09.33DonXhow do you get your PSTN access?
21:09.36bjohnsonjets: or a sip fxo
21:09.37*** part/#asterisk didz_ (didz_@200.218.192.52)
21:09.42jetsyup
21:09.54UajalI have SIP provider. Asterisk is connected to it It is broadvoice
21:10.02bjohnsonI guess could be a iax fxo but I haven't seen those yet
21:10.18bjohnsonUajal: so follow their instruction
21:10.48bjohnsonshould be something like dial(sip/6463948@broadvoice)
21:11.11bjohnsonmight need your username and password there depending on how you set up your sip.con
21:11.14bjohnsonmight need your username and password there depending on how you set up your sip.conf
21:11.17UajalAsterisk works with them. I can press "4" and my SIP phone connected to LAN calls. I want that my cell will call instead now.
21:11.59UajalI mean Asterisk works with them. I can press "4" and my SIP phone connected to LAN rings. I want that my cell will ring instead
21:12.04bjohnsonI don't understand
21:12.41UajalAsterisk is connected to broadvoice OK
21:13.23UajalNow I want to make such picture that by pressing "4" in main menu my cellular phone will call
21:13.54harryvvnetsurfer been around at all?
21:14.16SedoroxUajal: all you want to do is set it where when you press four.. it dials out broadvoice with your cell number
21:15.23UajalYes that Agent will be not on SIP phone in office but on cell phone
21:16.30*** join/#asterisk wizhippo (~wizhippo@Quebec-HSE-ppp233869.qc.sympatico.ca)
21:16.33Sedorox...
21:16.56UajalI cannot understand format of dial(SIP/6463948, 20, Tr) or it is wrong?
21:17.55Sedoroxyou want to make it the format of
21:18.10Sedoroxdial(SIP/Broadvoicesetup/yournumber,25,tT)
21:18.48UajalI had exten 4 => dial(SIP/6463948, 20, Tr) . It worked. Now I want 4 => dial(SIP/ MY_CELL_PHONE_NUMBER, ...) It doesn't work
21:18.54fileeven that's a little wrong, it's proper to do SIP/number@broadvoicesetup
21:18.57*** join/#asterisk tekati (~captain@cpe-66-75-215-63.bak.rr.com)
21:19.00Uajalexcuse me
21:19.26*** join/#asterisk sneak (~sneak@64.220.234.21.ptr.us.xo.net)
21:19.44UajalI had exten 4 => dial(SIP/2001, 20, Tr) . It worked (2001 is ectention of SIP phone). Now I want 4 => dial(SIP/ MY_CELL_PHONE_NUMBER, ...) It doesn't work
21:19.55Sedoroxyes
21:19.56Sedoroxwe get that
21:19.58Sedoroxbut
21:20.00Sedoroxin sip.conf
21:20.06Sedoroxwhat do you have broadvoice as?
21:20.08Sedoroxin the []?
21:20.42*** part/#asterisk yogurt2ungue (~charlie@203-132-126-200.fibertel.com.ar)
21:21.00Uajal[sip.broadvoice.com]
21:21.05jetsUajal you'll have to specify your sip peer
21:21.17jetsSIP/4347146@sip.broadvoice.com
21:21.24jetsactually it would be a full 10 digits with broadvoice i think
21:22.01UajalI tried only 9. I'll try 10 now
21:25.45tzangerNorstarMICS [PRI1] TE405 [IAX2] TE405 [PRI2] Telco
21:25.49tzangerif I ztmonitor the PRI ot the telco, I do not hear the echo
21:25.51tzangerif I ztmonitor the PRI to the MICS, I hear the person on the MICS echoing
21:25.55tzangerwould that not indicate that the echo is on the Norstar MICS PRI?
21:29.45*** join/#asterisk bjohnson (~bjohnson@jecinc.tor.istop.com)
21:30.35*** join/#asterisk Pauljohnhull (~Paul@81-86-141-177.dsl.pipex.com)
21:31.03*** join/#asterisk MajestiK (~MajestiK@S01060800208687ec.ed.shawcable.net)
21:33.36SedoroxI hate echo
21:35.43*** join/#asterisk Trionnis (buffy@12-203-113-15.client.insightBB.com)
21:36.09bjohnsonI hate echo too
21:36.12bjohnsonI hate echo too
21:36.17bjohnsonI hate echo too
21:36.53UajalCalled 6465153948@sip.broadvoice.com     Got SIP response 404 "Not Found" back from 147.135.0.128
21:37.42Sedoroxfor some reason.. I get echo even on SIP-IAX2-IAX2-SIP
21:38.24UajalFor SIP phone I have special record in sip.conf beginning with [2001] type=friend ... Should I have smth similar for cell  phone?
21:40.15bjohnsonhttp://www.broadvoice.com/support_install_asterisk.html
21:40.23bjohnsonlook what google turned up ^^
21:41.53bjohnsonSedorox: I'm told echo is usually gain related
21:42.02Sedoroxthese are BT100's
21:42.18Sedoroxconnected via SIP on two *'s.. and they are linked via IAX2
21:43.31UajalTo <bjohnson> I made these settings of broadvoice besides 6 because it is not outbound call. I want to connect inbound call with cell by pressing "4"
21:44.12SedoroxUajal: we explained to you how to do it
21:44.22bjohnsonanswer, dial?
21:44.39*** join/#asterisk tih (tih@athene.hamartun.priv.no)
21:44.42bjohnsonUajal: look at the authbycid macro on the wiki
21:45.13*** join/#asterisk jufineath (jufineath@stan.othius.com)
21:45.16*** join/#asterisk nextime (~nextime@ns0.nexlab.net)
21:45.26bjohnsonerr .. I mean look at the user authentication page (link from the tips and tricks page)
21:45.37ariel_afternoon everyone
21:46.20Uajalwhere is authbycid? Search on Wiki doesn't show it
21:47.46UajalCannot find user authentication page on Wiki
21:48.30dsmouse~google authbycid
21:48.49bjohnsonthat is a really neat tool !!
21:48.56bjohnsonwhat do they call it?   google?
21:50.26*** join/#asterisk Moc____ (~mochouina@64.235.210.66)
21:50.57*** join/#asterisk SeaForth (~SeaForth@c-24-1-126-202.client.comcast.net)
21:51.16Moc____Does anyone have Document that show the benifit of Asterisk over Cisco/Avaya/Nortel/3Com ???
21:51.56eKo1Moc____: Eh, it's free.
21:51.59JerJerDocument:  Open-Source
21:52.14JerJerfree is a very abused word
21:52.31eKo1Fine, it's GPLed software.
21:52.40JerJerasterisk is absolutely NOT without cost
21:52.51ariel_Moc____, frist one on the list is 1) Free 2) does Voip to PSTN transcoding. 3) Free
21:52.53jaigerthere is no free lunch
21:53.07JerJerbut you have freedoms that proprietary solutions simply cannot provide
21:53.34ariel_Well at least you don't have to pay for the software. But all the rest you do.
21:53.41dsmouseMoc____: you get the source code with it, and you can use a off-the-shelf pc (plus card, if you need it), which means you can replace it with off-the-shelf parts
21:54.01wizhippoAsterisk simply rocks
21:54.14UajalDidn't find relation of this authentification to my problem. I suppose that there is some simple mistake with the dialing external numbers with SIP. May be there should be some specific settings in sip.conf or other conf that solves the problem and make command dial(SIP/NUMBER@sip.broadvoice.com working. Should in conf files be smth specific for this dial?
21:54.16dsmousewizhippo: it complexly rocks too
21:54.17jufineathit starts with an a, so it's first in the phone book, which means it must be better.
21:54.19*** join/#asterisk qiu (~andrei@home-073519.b.astral.ro)
21:54.23Moc____JerJer:  Im talking abotu a document to sell Asterisk to the biggest Lawfirm in canada that Asterisk can do better more than the other systems...
21:54.47JerJerdon't give them a document then
21:54.59JerJerhand them a working system and let them utilize it for a week
21:55.00bjohnsonhehe .. proof
21:55.05tzangerthat's what I always do
21:55.09tzangerbut people want sheets to read
21:55.13tzangerI have the same problem
21:55.16bjohnsonto me, the biggest advantage is flexibility
21:55.18Moc____JerJer: I wish I could do that...
21:55.21tzangerMoc____: what's your email, we'll write something up
21:55.47dsmousecat ~/irc.log | mail Moc___
21:55.52ariel_tzanger, just put it on the wiki for all of us to use.
21:56.01denonJerJer: thats a pretty big job ... lots of vmail to config, lots of weird queues and stuff
21:56.10denonif the office really is that large
21:56.26tzangerariel_: will do once it's done
21:56.45Moc____they want to go with a Nortel IP PBX !!!
21:57.06Moc____and I hate to see it happen if I didnt try alittle ..
21:57.08bjohnsonI'd say breeze over the basics of what they would expect .. and then hit them with voip specific solutions.  Cheaper LD, off-site workers, emailed voicemail, cheapper cell phone LD, more concurrent lines
21:57.45ariel_The 3 main reasons we picked Asterisk over Nortel 3 years ago for use were. 1) Able to use normal Analog phones 2) Able to do Voip 3) Able to be installed on normal 1U PC.
21:58.16bjohnsonie .. flexibility
21:58.20ariel_2nd reasons were price, Voicemail, Queues (needed for support department).
21:58.49ariel_Last was it could use the analog phones we had already.
21:59.45ariel_Moc____, Nortel IP PBX will only work well with there phones and gateways.
21:59.53denon[their]
21:59.53Moc____I wish a FXS channel bank with PRI could offer a analog line for 56k dialup...
22:00.02Moc____ariel_: , my brother firm got it with about 20 phone
22:00.14wizhippofinding a good voip provider in canada, that will be your biggest challenge. at least thats what i'm finding.
22:00.23greg_workmine are 1) voip (multiple locations),   2) flexibility   3) price - we didnt really have the budget to get a huge system, if we didnt use * we wouldn't be able to have a lot of the features at all
22:00.24bjohnsonthey might like ability to look up info over phone from remote locations and have it read back.  Might not be something they want to pay for up front but ability to do in the future could be a hook
22:00.32UajalIs there in Internet live examples of dialing external numbers (not extentions) with SIP?
22:00.39ariel_We setup the support department to use modems for support calls via asterisk analog c/b to pri without any problems.
22:00.45Moc____wizhippo: we dont need to do VoIP for in/out, we have already 4 PRI
22:01.15wizhippoI envy you
22:01.16greg_workMoc____: what happens if you get rid of some PRI's?
22:01.19ariel_funny thing is that there still to this date not using a voip service for there LD
22:01.27Moc____greg_work: they wont
22:01.31greg_worki mean, if you can replace them with voip (depending on your needs)
22:01.32bjohnsonUajal: dialing local sip extensions is no different than dialing external sip extensions
22:01.49Moc____voip aint stable ennuf.  Or I should say, Internet aint stable ennuf
22:01.53bjohnsonUajal: use the & in the dial command.  Read the dial command wiki page
22:01.59greg_worksaving say $1k / month wouldn't hurt the feature list of using * :)
22:02.04*** join/#asterisk multrix (~chatzilla@ALyon-252-1-23-71.w82-122.abo.wanadoo.fr)
22:03.05bjohnsonor even .. "possible" saving of $1k/mo
22:03.13bjohnsonlet then choose it or not
22:03.24bjohnsonshow them the flexibility to change at any time is there
22:03.30Moc____they just moved our LD from Allstream (5cent/min) to TelUs (3cent/min)
22:03.38denonyeah .. "after the first few days of lost business, you can always switch back"
22:03.53bjohnsonbtw .. Allstream just went to $0.04/minute
22:04.17greg_workprimus did as well, apparently
22:04.18Moc____i know, people doing those decisions, doesnt know what things is going with the world
22:04.23bjohnsondenon: phased switching .. also a feature showing flexibility
22:04.27Moc____anyway
22:04.35Moc____tzanger:  you got my email ?
22:04.48Moc____ok got the email ;)
22:05.02greg_workallstream is my favourite telco to deal with though
22:05.13greg_workwait, i worded  that wrong. "least hated"
22:05.44*** join/#asterisk xachen (justin@toto.citelnetworks.com)
22:05.50bjohnsonsounds like they are sold on the voip idea for internal .. so you should concentrate on comparing Nortel vs * for just the internal system
22:05.59bjohnsonthen hit them with some external voip uses
22:06.20bjohnsoneven if they don't switch the entire firm over, there is likely some uses which they would like
22:06.35UajalBjohnson: I didn't find the dial at wikipedia
22:06.42bjohnson~docs
22:06.43jbotDocumentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org
22:06.43Moc____All the time I called support, the first person who answered, is the person who have access to all the diagnostic... except going on site, but the guy will call the exact same guy
22:06.55Moc____I like Allstream support for that
22:06.55bjohnsonUajal: http://www.voip-info.org/wiki-Asterisk keep looking
22:07.10Moc____and they dont try to send the problem to someone else until they are certain
22:07.47xachenVoip info has all you rneeds :)
22:10.33UajalBjohnson: If you mean http://www.voip-info.org/wiki-Asterisk+cmd+Dial I read it several times. Still I didn't find any example of dialinf external number with SIP. It is not the same as extention. Because extention should be preconfigured in sip.conf e.g. [2001] tpe=friend ... But external number is not preconfigured so it is different.
22:11.09shmaltzI am faced with the following problem:
22:11.10shmaltzI need to do virtual PBX hosting, and I need a way of doing call parking without interfering with each other, how can this be done?
22:13.27UajalBjohnson: that is what I ask: should external number be preconfigured in sip.conf as extention (e.g. [6465153498] type=peer ....)?
22:13.47shido6y?
22:14.03shido6you can
22:14.08shido6so u can see the number in your softphone
22:14.10bjohnsonUajal: exten=_1NXXNXXXXXX, 1, dial(SIP/${EXTEN}@sip.broadvoice.com,30)
22:14.16shido6but thats what displayname is for
22:15.02bjohnsonexten=>_4,1,dial(SIP/6465153498@sip.broadvoice.com,30)
22:15.08bjohnsongeez
22:15.20Hmmhesayshave any of you seen a network setup where the default gateway is on a different network than the IP assigned to the machine?
22:15.30shmaltzI was thinking about using valet parking and have in each context (company):
22:15.32shmaltzexten => _7XXX,1,ValetParkcall(${EXTEN:1}|mylot|${CALLERIDNUM}|1|${CONTEXT})
22:15.33shmaltzexten => _8XXX,1,ValetUnparkcall(${EXTEN:1}|mylot)
22:15.35shmaltzand have XXX match only extensions allowed to be dialed for that compony, or have a parking lot for them, has anybody implemented this? does it make snese?
22:15.48shmaltzHmmhesays, yes if you use PPP
22:15.49BeirdoHmmhesays: that won't work
22:15.55thieumSis this true there are some jumpers on TE410P ?
22:16.03anthmlol
22:16.08Beirdounless it's point-to-point of course
22:16.18bjohnsonyes ..we've sent out the fire department to talk them down
22:16.26Hmmhesaysahhh yes, good call
22:16.31HmmhesaysI was stumped for a second
22:16.41UajalBjohnson: May be I don't understand but I didn't ask about outbound calls
22:17.05UajalThis patterns (as I understood) are for outbound calls
22:17.45anthmshmaltz, why is mylot static? the whole point of valetparking is that the lot name gives you and entire namespace of exten per unique lot name.
22:17.51bjohnsonUajal: notice the difference between:
22:17.53bjohnsonexten=>_4,1,dial(SIP/6465153498@sip.broadvoice.com,30)
22:17.54bjohnsonand
22:17.59bjohnsonexten=>_4,1,dial(SIP/2201,30)
22:18.12bjohnsonpretty much the same format correct?
22:18.16anthmperhaps you'd like to come to cluecon
22:18.21shmaltzanthm, thanks for this, I didnt think about this. thanks :)
22:18.28anthm=D
22:18.50bjohnsonUajal: or even exten=>_4,1,dial(SIP/6465153498@sip.broadvoice.com&SIP/2201,30)
22:19.29shmaltzso I can realy use:
22:19.29shmaltzexten => _7XXX,1,ValetParkcall(${EXTEN:1}|${CONTEXT}|${CALLERIDNUM}|1|${CONTEXT})
22:19.29*** join/#asterisk Frantic__ (~ab@24-193-46-85.nyc.rr.com)
22:19.29shmaltzexten => _8XXX,1,ValetUnparkcall(${EXTEN:1}|${CONTEXT}t)
22:19.32bkw_take the last "t" out
22:19.55shmaltzyep, thanks, bkw_, it was a typo while pasting
22:20.27UajalIn this example extention 2201 should be preconfigured in sip.conf as [2201] ... Should the phone number be preconfigured also or not?
22:20.49bjohnsonno .. just sip.broadvoice.com
22:21.16bjohnsonotherwise you would have to define a sip.conf entry for each phone number you would ever like to dial
22:21.37Sedoroxhmmmmm
22:21.46UajalWhat should I check if I receive: Called 6465153948@sip.broadvoice.com     Got SIP response 404 "Not Found" back from 147.135.0.128
22:22.13shmaltzhow can I have the recptionist(operator)  park a call for someone that is restrcited to a specific context?
22:22.15*** join/#asterisk chetan (freetibet@24-193-188-21.nyc.rr.com)
22:22.34JerJerUajal:  do they want an 011 prefixed?
22:22.49bjohnsonhttp://www.broadvoice.com/support_install_asterisk.html
22:23.20bjohnsonlooks like they expect typical NA style 11 digit dialing
22:23.57bjohnsonso should be exten=>_4,1,dial(SIP/16465153498@sip.broadvoice.com&SIP/2201,30)
22:24.19anthmshmaltz,  where are you getting valetparking from anyway i'm sure it must be out of date.
22:24.45UajalI will check again with 1646...
22:24.47shmaltzwhy should it be out of date?
22:24.53shmaltzanthm
22:25.39*** part/#asterisk wizhippo (~wizhippo@Quebec-HSE-ppp233869.qc.sympatico.ca)
22:26.08shmaltzI got it from the wiki
22:26.09shmaltzhttp://www.voip-info.org/wiki-Asterisk+addons
22:26.11shmaltzhttp://www.loligo.com/asterisk/misc/apps/app_valetparking.c
22:26.16shmaltzanthm, you any other solution?
22:26.22anthmyah old as a mofo
22:26.33terrapensome of these comments on the wiki are so retarded
22:26.38anthmunofficial release =D
22:26.40shmaltzbkw_, are you aware of anything as good, or better better?
22:26.56bjohnsonshmaltz: http://lists.digium.com/pipermail/asterisk-users/2004-October/067189.html
22:26.56terrapenhere's a guy bitching about nufone...and he's never even fucking used them!
22:27.46shmaltzbjohnson, that doens't help much at the moment
22:28.06shmaltzregular parking gives me too much trouble from my cisco xfer and blindxfer
22:30.15*** join/#asterisk neopher (~crazy@mail.techhelpresources.com)
22:30.55shmaltzanthm, stop it. I'm trying to work
22:31.23*** join/#asterisk verge (~jfargen@56-116.26-24.tampabay.res.rr.com)
22:31.35neopheranyone know how to unpack a windows installer .msi file, tring to get 30 vip fireware from CCM
22:31.42UajalBjohnson: The same error. Here is my [sip.broadvoice.com]
22:31.42Uajaltype=peer
22:31.42Uajalhost=proxy.dca.broadvoice.com
22:31.42Uajalfromdomain=sip.broadvoice.com
22:31.42Uajalfromuser=MYNUMBER
22:31.42Uajalsecret=MYSECRET
22:31.44Uajalcontext=from-broadvoice
22:31.46Uajalinsecure=very
22:31.51SedoroxUajal:
22:31.56Sedoroxnever... paste in the channel
22:31.58doughecka_neopher: maybe run it, and then check the temp folder?
22:31.58Sedorox~pastebin
22:31.59jbotmethinks pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca
22:32.12UajalOK. I didn't know this rule
22:32.34UajalI will not
22:32.56Sedoroxhehe
22:33.01Sedoroxjust note it for the future :-p
22:34.15neopherdoughecka_: tried that, the installer freeks out because i tried to install it on non cisco approved hardward
22:34.39doughecka_ah
22:35.07shmaltzanthm, no not that, but fighting in the public
22:35.26anthmfighting?
22:36.49terrapenthe interweb is down!
22:38.31mooboidoes anyone has a simple extension.conf file with a single x100p configured ? ive google my way around without any luck from voip-info.... i need a bare* one
22:39.17malcolmdneopher: http://protools.reverse-engineering.net/files/decompilers/fmsiu.zip
22:39.18Sedoroxummm
22:39.46Sedoroxexten => _9NXXNXXXXXX,1,Dial(Zap/1/${EXTEN:1})
22:40.12mooboithas plain basic ; )
22:40.26terrapenhow do you test 911?
22:40.32terrapenis there some procedure to follow?
22:40.36mooboiasteriskdoc is great, but still in its pre-teen
22:40.40zigmanyou don't ;)
22:40.49BrianR___tzanger: I'm about to do a Norstar MICS <-> asterisk intergation..
22:41.00dsmouseterracon: over pots or voip outbound?
22:41.03mooboi911 and voip dont mix so far, 911 is so tied to the land line for adress intervention ...
22:41.06Sedoroxterrapen: my poppop.. whenever he got a new phone.. he would dial 911.. and tell them that he is just calling to make sure it works
22:41.09Beirdoterrapen: you wait for a real emergency...  then dial
22:41.10dsmouseterracon: I ment terrapen again
22:41.12Sedoroxand they actually liked him calling
22:41.16terrapenhaha
22:41.17*** join/#asterisk dsfr (~dsfr@216.207.244.184)
22:41.23terrapenthat's solid
22:41.26Beirdomake sure you have a real phone nearby though :)
22:41.33dsmouseterrapen:  over pots or voip outbound?
22:41.37terrapenone of our stores was robbed once
22:41.40terrapends: voipo
22:41.42terrapenerr voip
22:41.45mooboiany good doc ressources beside asteriskdoc and voipinfo ?
22:41.50BrianR___Should be interesting..;
22:41.55terrapeni guess i could just make 911 go to 210.828.3321
22:41.58terrapenwhich is local PD
22:42.01terrapennon-emergency, tho
22:42.06anthmcome to cluecon and learn everything!
22:42.13mooboiurl ?
22:42.16mooboioh
22:42.20bkw_shmaltz, anthm wrote valetparking... When I took the parking idea to anthm I had a call parking switch.. which half ass worked.. he really made valetparking and made it more kick ass
22:42.20mooboicon...
22:42.24dsmouseterrapen: I suspect how they'll react is diffrent depending on the locality... call the non-emergancy number and ask if they mind?
22:42.27BrianR___A lot of local PD's don't offer non-911 emergency numbers... Very odd.
22:42.46terrapenhow do i get on the converence call?  is it as simple as setting up an extenstion that does:
22:42.49neopherours does
22:42.54terrapenDial(IAX2/guest@66.250.68.194/996)
22:42.54Beirdo"no, I'm sorry, you have the wrong number...  this is 912"
22:42.55bkw_shmaltz, and if anthm and I were fighting.. EVERONE here would know it!
22:42.58BrianR___The town I live in and the surrounding towns have 10 digit emergency numbers for historical reasons..
22:43.22dsmouseterrapen: or call 911 to report a road hazard
22:43.34terrapenmouse: hahaha
22:43.34SedoroxHomer: "Hello operater. give the number to 911!!!!"
22:43.39terrapenA DEER BIT ME
22:43.40BrianR___When i was a kid, 911 rang through to the LEC's operator.
22:43.44terrapenI NEED A BAMBULANCE
22:44.02yashaxGuys, can anyone recommend a good/reliable SNTP server? (is it the same as NTP?)
22:44.07dsmouseI ment more like obstruction in the road or something
22:44.14bkw_yashax, you have to ask?
22:44.25terrapenyashax, time.apple.com
22:44.31bkw_yes you can use ntp in place of sntp
22:44.35bkw_or time.windows.com
22:44.35bkw_haha
22:44.39BrianR___yashax: most ntp servers provide both the sntp (quickly set time) and the ntp protocol for sub-millisecond synchronization.
22:44.43bkw_or tick.usno.navy.mil
22:44.43yashaxthought so, but wanted to make sure....
22:44.44Sedoroxntp.nist.org
22:44.46SedoroxI think
22:44.47terrapeni used to use clock.home.net
22:44.47bkw_or tock.usno.navy.mil
22:44.51terrapenbut they went out of business
22:44.55denonbkw_: I thought one or both of those are gone now
22:44.59bkw_no
22:45.00drumkillatick.mit.edu
22:45.01bkw_they are there
22:45.03terrapeni'm thinking about setting up clock.gpstools.com
22:45.04denoner, closed
22:45.05BrianR___Use pool.ntp.org
22:45.18bkw_I don't trust that
22:45.20denonI dont like the idea of those round robin ntps
22:45.23Sedoroxthats what I ment
22:45.24denontoo much reliance on them sticking around
22:45.30bkw_denon, i'm with ya on that one
22:45.30JerJerbuy a GPS reciever
22:45.34bkw_or CDMA phone
22:45.36JerJerjack it into a linux box
22:45.37denonand too much reliance on them not screwing you, wheras most 1st and 2nd stratums are safe
22:45.44terrapenJerJer, we are a garmin dealer
22:45.50terrapeni guess i should hook it up
22:45.55bkw_getting time from GPS is da bomb
22:45.59denonJerJer: ya, but then you gotta have 3 or 4 of em for redundancy .. or one in each and every server
22:46.15*** join/#asterisk jdg (~jdg@CA03F960.adsl.mana.pf)
22:46.15terrapeni wonder if i can get OpenNTPD to work with a GPS receive3r
22:46.18BrianR___A large number of NTP servers would have to collude before your time could windup off...
22:46.24BeirdoOpenNTPD is ass
22:46.32denonBrianR___: or they could just point them all to their own bogus one
22:46.35terrapenbeirdo: por que?
22:46.41Beirdothey ripped out all the useful stuff and dumbed it down
22:46.47Beirdoit doesn't even track drift
22:46.52Beirdowhen I looked at it
22:46.54Beirdouse xntpd
22:46.56*** join/#asterisk pr0m (~pr0metheu@ip-wv-68-187-250-031.charterwv.net)
22:47.14yashaxStrange... Even though I input the right NTp server in Polycom IP500, it is still showing incorrect time... any ideas? (already tried 3 different servers)
22:47.17pr0mis asterisk available in fedora core 2 repositories?
22:47.25terrapenyashax, time zone
22:47.32DJ-Pyroyeah, what he said
22:47.34denonyashax: using the ip or the name? if the name, make sure you have dns set
22:47.56yashaxip
22:48.04Sedoroxyashax: make sure you have the currect timezone setup
22:48.07Sedoroxlike mine is -5:00
22:48.10Sedoroxfor EST...
22:48.12yashaxtime zone is set correct as well, but good call..
22:48.16Sedoroxif you don't.. it sets to what you have
22:48.20Sedoroxkk
22:48.25BrianR___denon: The Network Time Project folks could also release a backdoored version of their NTP server if they wanted to screw up time for people.
22:48.27denonok .. one more person tell him to set the timezone please..
22:48.40bkw_-5:00 is CST6CDT or something too
22:48.43Sedoroxdenon: I didn't see it asked before
22:48.46Beirdoand terrapen: I trust the collective experience of the ntp.org types over Theo *any* day
22:48.52denonBrianR___: yeah .. im just saying I can trust that stratum 1 servers and most stratum 2s wont screw with me
22:49.10BeirdoEST5EDT?
22:49.44denonusing the pool is probably safe enough .. Id just still prefer to have 2 other sources, if even due to a catestrophic dns failure
22:49.47denonor a domain hijacking
22:49.56terrapenyashax, its done in ipmid.cnf in the <SNTP> section
22:50.02terrapenand it uses a offset from GMT
22:50.03terrapenin seconds
22:50.06BrianR___Well... One shouldn't rely on a single pool..
22:50.08mikegrbtheo the rat
22:50.11terrapenie -21600
22:50.39Beirdomikegrb: that's the one :)  I'll live with OpenBSD for it's usefulness, but OpenNTPD can eat me
22:50.54terrapeni trust OpenBSD code over others
22:50.56yashaxterrapen: Can I not set it right on the phone? I am doing this now....
22:51.00terrapenit's never done me wrong
22:51.11denonits cheap enough .. just make a bios option to disable it for the privacy freaks
22:51.14terrapenyashax, try it and see if it works
22:51.19moonwicksounds like an expensive, silly feature
22:51.29yashaxdoing it now...
22:51.35Beirdodenon: no thanks, that's silly, you'd need good antennas, etc.
22:51.38Beirdowaste of money
22:52.19denonBeirdo: just an external connector
22:52.22terrapenone more component to break
22:52.22shmaltzhttp://story.news.yahoo.com/news?tmpl=story&ncid=1211&e=1&u=/nm/20050221/tc_nm/tech_security_dc&sid=95573372
22:52.26terrapenmore complexity
22:52.28terrapen<PROTECTED>
22:52.31denonBeirdo: it wasnt long ago, people thought a standard nic was a waste of money
22:52.34Beirdothe extra cost ain't worth it
22:52.37denonand before that, a modem
22:52.42terrapenUSB GPSes are cheap
22:52.46denonwhat extra cost, it could be built right into a chipset
22:52.54denonvery little cost
22:53.00yashaxrebooting
22:53.27BeirdoGPS has no useful purpose in most computers
22:53.30denonoh, pool.ntp.org is just a collection of whoever wants to add their server?
22:53.37denongood grief, thats lame..
22:53.49Beirdoyes, pool.ntp.org is almost all of the stratum2
22:53.52denonBeirdo: an accurate timing source could be valuable for lots of stuff
22:54.01terrapengpsdrive roxx tho
22:54.09shmaltzanthm, is this different than the other one?
22:54.19shmaltzanthm, thanks
22:54.19Beirdothe vast majority of computers are used for Winblows, playing games, etc.
22:54.31anthmyep
22:54.33BrianR___The round-robin DNS for pool.ntp.org is built programaticly based on reliability and accuracy measurement...
22:54.40anthmthis one has chan_valet on it too
22:54.43Beirdoit's only useful for servers, and only if you want a local NTP source on said computer
22:55.03shmaltzany doces or commands on how to use it? anthm,
22:55.22denonBeirdo: oh I dunno, it'd be nice if all home users knew their computers were perfectly in sync
22:55.31denonand most laptop users would want gps maps at some point or another
22:55.44Beirdonot worth the extra $50 cost per motherboard
22:55.48denon50? no way
22:55.50yashaxterrapen: What would be the offset for -5 (default: tcpIpApp.sntp.gmtOffset="-28800")
22:55.59Sedoroxyou could argue that the GPS built in is a invasion of privacy
22:56.02denonprobably more like 2-5 with a wide-spread chipset, and the connectors
22:56.07shmaltzanthm, how can I use the cannel in the dial plan?
22:56.08Beirdonot even worth an extra $2 per board
22:56.10terrapen-21600 + 3600
22:56.12anthmexten => *7,1,Dial(Valet/fifo:mylot)
22:56.18denonSedorox: yeah, like I said an hour ago .. give a bios or jumper option to disable it
22:56.25terrapenjbot: -21600 + 3600
22:56.26jbot-18000
22:56.30BeirdoSedorox: how so, it doesn't transmit :)
22:56.30terrapenthere ya go
22:56.41denonjbot: -e+1 * 5
22:56.42jbot2.281718171541
22:56.45terrapenhaha
22:56.46anthmwill unpark the longest waiting parked call when you dial that channel
22:56.47doughecka_hah
22:56.50terrapenjbot: 1/0
22:56.52jbot[1/0] undefined
22:56.53Sedoroxdenon: true...
22:56.56Beirdothat's like saying a TV tuner is a privacy invasion
22:57.01doughecka_~convert 1 year into fortnights
22:57.04terrapenjbot: sqrt(1/0)
22:57.05BrianR___standard gps on pc motherboards won't help your problem much anyway...
22:57.12SedoroxBeirdo: well yea.. but people may also want to locate their laptop if stolen
22:57.14BrianR___Most users won't bother to hook up the antenna
22:57.16terrapenruh roh
22:57.18SedoroxI personally would want that...
22:57.25BrianR___or if they do, they won't bother to make sure it has a clear view of the sky
22:57.26terrapendid i kill hiim?
22:57.29yashaxCan you force IP500 to re-read the config without having to reboot?
22:57.32denonBrianR___: a good sensative receiver with a loaded antenna in the case could probably do ok
22:57.34terrapenjbot: sqrt(1/0)
22:57.41Sedoroxwith being on a college campus.. if I mistakenly leave my $3k laptop somewhere.. I wanna know where it is!
22:57.43BeirdoSedorox: ahhh, like an on-star type of thing
22:57.49terrapenjbot has gone wonky
22:57.50doughecka_~httpdtype digium.com
22:57.50Sedoroxkinda of
22:57.51Beirdoor lo-jack
22:57.55doughecka_sweet
22:57.55Sedorox:-p
22:57.57shmaltzanthem, what is the /n for ?
22:57.59doughecka_~httpdtype msn.com
22:58.01Beirdoyeah, that could be useful
22:58.08denon~httpdtype localhost
22:58.10terrapen~httpdtype http://chrissnell.com:17411
22:58.15Sedoroxwe've had people just walk into dorms here.. the second week of school.. and stole to desktops...
22:58.21stevekstevekhmm, -e+1 * 5 = 2.281718171541?  (nevermind, doh! * has higher precedence.)
22:58.25terrapenwhy is he ignnoring me?
22:58.34TrevorSHarrisonyashax: re EST offset... give me a yell if you get the IP500 to correctly use that when configuring manually... my IP500's just ignore me.
22:58.35terrapen~httpdtype bikeworld.com
22:58.37doughecka_~httpdtype monkey.com
22:58.46denon~httpdtype 0.0.0.0
22:58.50doughecka_~httpdtype sco.com
22:58.50BrianR___denon: Most of my testing has shown that GPS is pretty much useless inside any concrete / metal building and mostly useless inside wood fram structures...
22:58.59Sedorox~httpdtype 127.0.0.1
22:59.01jetsSedorox: Write a mini application that is a heartbeat, every 5 minutes it connects to a url... runs in the background, etc.
22:59.03Beirdojeez
22:59.04terrapen~httpdtype saba.island.nu
22:59.06denonBrianR___: so how about a backup of radio :)
22:59.10terrapenHAHA
22:59.11jetsaka you will be able to track it by ip if they plug it in to the internet.
22:59.12shmaltzanthm, you mean valetparking?
22:59.15Sedoroxjets: yea
22:59.15terrapenI LOVE YOU JBOT
22:59.22Beirdo~google: dumbass
22:59.25anthmyah ??
22:59.26BrianR___We had to run put an antenna on the roof for our NTP time source...
22:59.31terrapen~httpdtype lamberttriebel.com
22:59.36stevekstevek~httpdtype 127.0.0.1:5038
22:59.45stevekstevekheh
22:59.54denon~httpdtype 127.0.0.1\:22
22:59.59Beirdo~google asterisk rules
23:00.00BrianR___couldn't even get the signal through the windows - some sort of tinting on them was attenuating it too much.
23:00.01Sedorox~httpdtype neltia.net
23:00.07terrapenbrian, what software are you using?
23:00.11yashaxterrapen: THANK YOU!!!!!!!!!!!  It worked... strange... does not work by manually entering the info into the phone...
23:00.11shmaltzis valetparking the command for using the channel?
23:00.14doughecka_~wtf
23:00.26doughecka_~wtf iirc
23:00.28terrapenyashax: no problem
23:00.34*** join/#asterisk syslod (~yurplsl@65.114.0.198)
23:00.45terrapenjbot: sqrt(0)
23:00.50terrapenhe doesn't like that
23:00.50bkw_terrapen, what softwarE?
23:00.53TrevorSHarrisonyashax: thanks... I ran into the same thing, just haven't taken the time to setup the offset in the dhcp options yets
23:00.55bkw_gotta narrow that down a bit
23:01.02terrapenbkw, im wondering what he's using to sync gps to ntp
23:01.05terrapengpsd?
23:01.14terrapenerr sync ntp to gps
23:01.15BrianR___terrapen: some sort of time source appliance.. Forget the brand name. It provides a 10mhz reference oscillator bus, TDM synchronization bus, and ntp over ethenret.
23:01.17bkw_you can use linux for that
23:01.24denonreal men write shell scripts to parse the serial data
23:01.25bkw_it shows up as a kernel clock src or something
23:01.27bkw_google for it
23:01.30shmaltzanthm, how can I get access to the valetpakring channel? is it thru the valetparking cmd?
23:01.38bkw_Dial
23:01.39bkw_:P
23:01.45anthmi pasted it, no ?
23:01.48anthmexten => *7,1,Dial(Valet/fifo:mylot)
23:02.12terrapenhttp://www.gpstools.com/components/catalog/product.html?pid=518&cat=376
23:02.15BrianR___We use it for T1 clocking crap too.
23:02.15jayden~asterlink
23:02.16jbothmm... asterlink is "<bkw_> http://www.asterlink.com we do sip and iax also boi"
23:02.16terrapenthats what i need to set up
23:02.23anthmthats optional , the rest works the same
23:02.42stevekstevekchan_boi  cool..
23:02.52bkw_haha
23:02.55bkw_smartass...
23:02.57bkw_haha
23:03.14syslodasterlink is down from here.
23:03.14jaydenso, who is in MI?
23:03.18bkw_stevekstevek, you wanna work and lets get your app_conf updated and for cvs-head?
23:03.31*** join/#asterisk Zaw (zaw@zaw.subneural.net)
23:03.42bkw_syslod, what country?
23:03.43stevekstevekbkw_: soon..
23:03.58stevekstevekcan we get the new jb in there first :)
23:04.09syslodUS.  I'm in the qwest pop right now.
23:04.11jaydenanthm- is that heading for CVS?
23:04.13bkw_lets ride mark like zoro for that
23:04.24bkw_syslod, asterlink.com loads here
23:04.26jaydenride like zoro... nice
23:04.33anthmit doesnt depend on core so it loads just fine from that url
23:04.41*** join/#asterisk booleahn (~buleeahn@66-141-61-2.ded.swbell.net)
23:04.51stevekstevekit needs testers..  And maybe someone to code up a better way to have it auto-disable when you're bridged to a VoIP channel..
23:04.57terrapenasterlink fine here
23:05.04syslodI believe you.  Looks like some kinda DNS issue.
23:05.08bkw_odd
23:05.13bkw_syslod, who is your ISP?
23:05.15jaydengotta run... have fun kids.
23:05.22terrapensyslod whats your NS
23:05.37syslodQWEST
23:05.41terrapeni'll tell you what does seem down again....sourceforge
23:05.53terrapenis it my network or is sourceforge always down
23:05.54anthmthose w head can issue /usr/src/asterisk/contrib/scripts/astxs -autoload -install http://www.pbxclue.com/asterisk_apps/app_valetparking.c and go from 0 to valet w/o even restarting *
23:05.58denon~httpdtype sourceforge.net
23:06.05denonlooks up to me
23:06.21terrapenits just....slow
23:06.44denonyeah .. its running on linux.
23:06.48terrapenhaha
23:07.01denondont worry, they'll move it to FreeBSD soon
23:07.10syslodbkw: Craps out in DC.  cogentco?
23:07.37terrapendenon: suuuure
23:07.48terrapenthey are zelots
23:07.50terrapenerrr
23:07.51terrapenzealots
23:08.03bkw_syslod, whats the last ip?
23:08.10doughecka_~httpdtype bkw.org
23:08.34bkw_telnet port 80 on it
23:08.36bkw_see if you get it
23:08.43terrapen~httpdtype acme.com
23:08.48terrapenword.
23:08.51terrapenthttpd++
23:08.58terrapen~httpdtype cr.yp.to
23:09.05bkw_haha
23:09.07bkw_lame
23:09.14terrapenpublicfile == weird
23:09.14doughecka_lol
23:09.16stevekstevekastxs seems cool..
23:09.24Sedorox~httpdtype microsoft.com
23:09.25syslod<PROTECTED>
23:09.25syslod.250.8.206]
23:09.25bkw_seems?
23:09.29Sedoroxdamn...
23:09.29bkw_is that all you can say SEEMS?
23:09.29denon~httpdtype /etc/passwd
23:09.34Sedoroxwas hoping to see apache
23:09.43stevekstevekI haven't actually _tried_ it yet :)
23:09.50doughecka_~httpdtype riker
23:09.50RGi_yo
23:09.50*** kick/#asterisk [Sedorox!~brian@bkw.developer.and.friend.of.asterisk] by bkw_ (NO DADDY NO!!!)
23:09.51*** join/#asterisk Sedorox (brandon@Neptune.client.wlmsprt.pa.sed6.net)
23:09.55stevekstevekI figured the cool people never actually try stuff :)
23:09.59doughecka_lol
23:10.01bkw_haha
23:10.06RGi_waz up
23:10.15Sedoroxhmmm
23:10.16syslodbkw: no telnet to port 80
23:10.18RGi_bala me?
23:10.31terrapen~httpdtype devnull.homeunix.com
23:10.32Uajalin outbound calls dtmf doesn't work (no reaction on pressing buttons). In inbound calls it works properly. What can be the reason. I tried all 3 settings: dtmfmode=inbound,info,rfc2833
23:11.42bkw_Uajal, this on SIP?
23:11.50Uajalyes
23:11.55bkw_between what and what?
23:11.59terrapenuhh
23:12.00terrapenits inband
23:12.02terrapennot inbound
23:12.03TrevorSHarrisonl8rs
23:12.10bkw_doesn't matter
23:12.18terrapenok
23:12.19bkw_the dtmf is getting negociated wrong
23:12.22bkw_on rfc2833
23:12.29bkw_I suspect one end or the other is in violation of the RFC
23:12.37bkw_namely the timestamps on the DTMF packets
23:12.44bkw_rtp debug can show you
23:12.49UajalInband works OK
23:12.54bkw_if you see the timestamps increase when you dial
23:12.59bkw_then that end is WRONG WRONG WRONG
23:13.05bkw_the timestamps on dtmf never increase
23:13.06terrapenuajal: so now it works?
23:13.26UajalNo
23:13.36bkw_rtp debug and get a sip debug
23:13.42bkw_and i'll show you exactly where its failing
23:13.51UajalHow can I see timestamps?
23:14.02syslodAnyone here today interested in SECABS or CABS BOS?  I hate writing things by myself.
23:14.05bkw_you'll need to dial digits on both ends while doing rtp debug
23:14.17bkw_Uajal, "rtp debug" "sip debug"
23:14.18bkw_duh
23:16.32*** join/#asterisk R3DB0x (nobody@66.142.28.36)
23:16.39Uajalrtp debug is essential or it can be done with sip debug?
23:16.39*** join/#asterisk bjohnson (~bjohnson@ip226-181.tor.istop.com)
23:18.42*** join/#asterisk ACiDV (Joel29@66.103.213.54)
23:19.41UajalI tried sip debug. It seems that nothing appear on the screen when I press buttons of my cell during this call (while inband my cell works OK)
23:20.08bkw_it won't
23:20.15bkw_rtp debug you'll see when you press digits
23:20.35shmaltzbkw_, is ther anyway I can park a call using Dial(valet/auto:mylot)? I only succeeded in unparking a call this way.
23:20.37UajalI have not rtp debug.
23:20.57Uajalin Asterisk help there is no such command
23:21.10bkw_shmaltz, nope
23:21.18bkw_Uajal, what version of asterisk are you using?
23:21.21bkw_latest stable or head?
23:21.48shmaltzbkw_, there is no autosensing in any of the commands?
23:22.54bkw_shmaltz, no really
23:22.58UajalCVS-v1-0-02/15/05
23:23.19UajalI suppose it is stable
23:23.51*** part/#asterisk paulc (~paulc@S010600062586a0b4.vc.shawcable.net)
23:24.07bkw_Uajal, I would have to bill you to even look at this
23:24.10bkw_I know whats going on
23:24.21bkw_but I can't relay to you it seems on how to collect the info needed
23:24.31bkw_rtp debug while you're dialing in rfc2833 s what I wanna see
23:24.38bkw_s/s/is/
23:25.00Sedoroxhmmm
23:25.21bkw_in and outbound dialing
23:25.29bkw_so call someone play marry had a little lamb or something
23:25.36bkw_:P
23:25.54*** join/#asterisk paulc (~paulc@S010600062586a0b4.vc.shawcable.net)
23:25.58bkw_thats what I did when I fixed dtmf last time it was broken
23:26.14bkw_asterisk was increasing the timestamps on the dtmf packets
23:26.18bkw_which is a no no
23:26.22Uajalbkw: didn't catch an idea behind
23:26.46bkw_brb
23:26.54bkw_Sedorox, why?
23:26.55*** join/#asterisk Cresl1n (~matt@216.207.245.23)
23:26.59bkw_speak up
23:27.04Sedoroxno...
23:27.06shmaltzbkw_, what am I doing wrong:
23:27.08shmaltzwhen nobody is parked and I do: valteparking(${EXTEN}|mylot|360|${EXTEN}|1|${CONTEXT}) I get sorry there is nobody ...
23:27.08Cresl1nyes
23:27.09Cresl1n:-)
23:27.10shmaltzwhen somebody is parked and I do: valteparking(${EXTEN}|mylot|360|${EXTEN}|1|${CONTEXT}) I get reorder
23:27.10Cresl1nok
23:27.11shmaltzwhat am I doing wrong?
23:27.12greg_workjbot nat
23:27.13jboti heard nat is Network Address Translation  Usable in Asterisk sip.conf file with externip, localnet, and localmask setup properly.  See docs.
23:27.44greg_workthats not what i was looking for
23:29.20*** join/#asterisk iceyp (~icepick@firewall.unix.co.nz)
23:31.12UajalThank you, bkw. Enough Asterisk depth for today for me. I am going to eat my lamb
23:31.13drumkilladoes anyone know how to automatically provide a pass-phrase to init the keys?
23:31.20drumkillaso it doesn't have to be typed in?
23:31.44syslodssh?
23:32.01drumkillano, asterisk keys
23:32.03greg_worki want to take one of my SIP phones home ... my home network is NAT'd. my * box has a direct ip.. what do I need to do? put nat=yes in sip.conf, and open up ports in iptables?
23:32.10syslodoh.
23:32.22drumkillaasterisk -i ...
23:32.27drumkillabut then you type in a passphrase for your private keys
23:32.38drumkillabut I need to have it done automatically ...
23:32.42Beirdogreg_work: and pray, I think
23:32.51greg_workBeirdo: lol
23:33.24*** join/#asterisk ManxPwr (~eric@dsl-208-164-150-160.datasync.com)
23:33.32ManxPwr~docs
23:33.33jbotDocumentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org
23:33.34ManxPwr~clec
23:33.35jbot[clec] Created by the Telecommunications Act of 1996, a CLEC is a service provider that is in direct competition with an incumbent service provider. CLEC is often used as a general term for any competitor, but the term actually has legal implications. To become a CLEC, a service provider must be granted "CLEC status" by a state's Public Utilities Commission. In ...
23:37.36greg_workhm, well it seems to work
23:38.03greg_work(from behind my NAT here, anyways.. hopefully its not just sneaking around and using the local ip's)
23:38.28Sedoroxgreg_work: when I took my voip phone home over a weekend.. I didn't need to open any ports or anything.. just worked (tm)
23:38.37greg_worki meant locally
23:38.45jsolaresgreg_work: if the phone supports it use qualify=x
23:38.54greg_worki have udp/5060 and 10000-20000 open
23:39.03greg_workand nat=yes and qualify=yes
23:39.10jsolaresthen you should be set to go
23:39.26jsolaresif you plan on calling other natted phones you have to use, canreinvite=no
23:39.45*** join/#asterisk eye69 (magnus@ipv6.upcore.net)
23:40.03greg_workseems to work with nat=never as well?
23:40.19jsolaresdunno
23:40.19denonprobably nat=anything
23:40.28denonexcept no
23:40.31greg_worki have nat=never and its working ;)
23:40.38terrapenParis Hilton's T-Mobile Sidekick getting stolen was probably a great thing for T-Mobile
23:40.46terrapenerr the info
23:40.52greg_worksip show peers shows it using the external ip of my NAT router
23:40.56terrapenimagine how many they have sold now
23:41.24shmaltzbkw_, is there a way to valetpark and not annouce the parking spot?
23:42.00yashaxI am working getting MWI to work on Polycom. Reading the how-to and trying get clarification on "You may have to add @context to the mailbox entry. This seems to fix things for many users."  Any pointers?
23:42.21terrapenmine worked using the supplied configs
23:42.23snewpyyashax: ignore that piece of advice :)
23:42.41terrapenand i used vmid@context
23:43.11yashaxhmm...ok... I went through config and no luck.  I know that I AM doing something wring, but...???
23:43.34terrapenit seemed pretty straightforward to me, man
23:43.45yashaxmost likely it is something pretty stupid....
23:43.56yashaxYeah, it is... but... still luck
23:44.08greg_worki tried connecting my fax to an SPA-2000.. somewhat worked, but lots of problems with the fax (some lines came out messed up, failed altogether receiving 1 in 4 times, failed to send at all).
23:44.08yashaxDid this: ipdmid.cfg:
23:44.08yashaxup.oneTouchVoiceMail="1"
23:44.08yashaxphone1.cfg:
23:44.08yashax<msg msg.bypassInstantMessage="1">
23:44.09yashax<PROTECTED>
23:44.09yashax</msg>
23:44.41greg_workdoes anyone have faxing setup using an fxs (and fxo) on a TDM400p or whatever, and does it work better?
23:44.41denongreg_work: fax over sip is always going to be hit and miss without T.38
23:44.48snewpyyashax: that should do it without any further config
23:44.48yashaxstupid question:  Is ext number same as voicemail ext?
23:44.54ariel_greg_work, it works on my system.
23:44.59terrapenyashax, not necessarily
23:45.03yashaxhmm...
23:45.04denongreg_work: you're using ulaw?
23:45.07terrapenyou have to set it up in voicemail.conf
23:45.07greg_workwell, is it worth getting an fxs card?
23:45.09greg_workdenon: yes
23:45.09snewpyyashax: assuming you have mailbox=something in sip.conf, and a correctly configured voicemail.conf
23:45.09yashaxhow can I tell?
23:45.19terrapenand i chose to use identical numbers for clarity's sake
23:45.21ariel_yashax, no it can be different it's up to you.
23:45.31greg_workI wouldn't mind being able to use that line, but if it has to be 100% fax-only, then .. whatever..
23:45.31*** join/#asterisk h3x (Justino@ip68-108-176-196.lv.lv.cox.net)
23:45.47greg_workariel_: do you have any problems at all?
23:46.00h3xcan somebody recommend a good screen pop program for windoze
23:46.01ariel_no it works just fine.
23:46.03h3xwell hell
23:46.12h3xdo any of the voip softphones have url push in iax2 yet
23:46.17*** join/#asterisk cbachman (~cbachman@129.105.7.250)
23:46.18greg_worki'm curious  because i never see that mentioned as a solution to SIP issues (SIP and IP not being involved would lead me to believe it would be fine)
23:46.22ariel_My sipura 2100 is set for ulaw on line 2 and has the fax enabled on it.
23:46.39greg_workariel_: oh, you mean you use it on a spa-2100
23:46.40yashaxI used Asterisk@home.. Everything works... but SIP.CONF does not have that info, but it is rather in SIP_ADDITIONAL.CONF... So I don't know what it used for VM ext.. looking at voicemail.conf
23:46.45syslodh3x: The best thing I've found is use manager proxy and write a small C# task app.
23:46.58ariel_greg_work, but there on the same network. I get my faxes via my pots line and asterisk detects it sends it to my siprua
23:47.08greg_workthats what I was trying to do
23:47.12shmaltzbkw_, I think I have a but for the valetparking app
23:47.15h3xhmm
23:47.19yashaxYeah, it is same as my ext.....
23:47.36h3xsyslod: I would but this is a very simple application
23:47.37terrapenyashax, read the wiki on voicemail.conf
23:47.39greg_workariel_: like i said though, ocasionally some lines would be a bit messed up. and i couldn't get it to send at all
23:47.46terrapenyashax, this stuff is really easy
23:47.52shmaltzwhen a calls b, and b blind xfers (using cisco blindxfer), the parked spot gets announced to the a
23:47.54terrapeni found it to be one of the easiest parts of setting up *
23:47.59h3xim gonan check the iax2 soft phones on the wiki
23:48.01*** join/#asterisk RoyK (~roy@83.80-203-29.nextgentel.com)
23:48.13yashaxYeah... the only thing that I did not do from how-to was:  "You may have to add @context to the mailbox entry. This seems to fix things for many users. "
23:48.30greg_worki'm tempted to buy a pci card with an fxs port, because i would think it would work with no problems (no SIP or IP involved) .. but i wanted to hear from someone else with that setup.. i've never heard of it being done before
23:48.41ManxPwryashax, mailbox=mailboxnumber@contextinvoicemailconf
23:49.14*** join/#asterisk iCEBrkr (icebrkr@chrome.cyberdyne.org)
23:49.34ariel_greg_work, I have a TDM11b in my system the fxo is connected to the pots line and the fxs is connected to my hp fax.
23:50.13ariel_HP fax don't work well with voip for some reason. But I also have my fax/modem on my laptop on the sipura 2100 port 2 and it gets it's faxes that way.
23:50.41shmaltzanthm, when a calls b, and b blind xfers (using cisco blindxfer), the parked spot gets announced to the a
23:50.48greg_workariel_:  how is voip involved?
23:50.53h3xbingo
23:50.57h3x- accept URLs during a call and open that page in the default browser when the call is answered;
23:50.59h3xdiax phone does it now
23:50.59h3xheh
23:51.00ariel_I am only able to send faxes via the voip to 2 providers I have so far. VPC goes through 80% of the time and race.com 95 % of the time. voipjet does not work
23:51.13ariel_I send them via voip
23:51.24ariel_and I also get them via my vpc number
23:51.27yashaxSo if my EXT is 100, it would be: mailbox=100@context (just like this)? Is that right?
23:51.50ariel_yashax, if that is the way your want it setup yes.
23:52.02yashaxor?
23:52.04*** join/#asterisk hardwire (~hardwire@209.112.194.45)
23:52.34ariel_yashax, you only need @context if it's something other then default.
23:52.48greg_workariel_: oh ok. i probably wouldn't bother with that (at least not yet). i'm more just interested in using one of my 4 POTS lines as a dedicated incoming fax line, and any as an outgoing fax or voice line (as opposed to having 3 voice POTS and 1 fax-only POTS line)
23:52.49*** join/#asterisk MichaelVanD (~MichaelVa@CPE-24-208-88-245.neb.rr.com)
23:53.01yashaxk.. trying...
23:53.15ariel_greg_work, then your set get the tdm400b
23:53.18greg_worksince * has no dialtone detection (though, someone did email me a week ago, i put up a bounty for it)
23:54.20*** join/#asterisk rett (~rett@c-67-171-236-169.client.comcast.net)
23:55.11*** join/#asterisk bkw_ (~brian@bkw.developer.and.friend.of.asterisk)
23:55.11*** mode/#asterisk [+o bkw_] by ChanServ
23:55.18*** join/#asterisk buddah (~hnic@208.179.86.5)
23:55.34buddahanyone have any idea what this error means?
23:55.35buddahFeb 21 15:55:21 WARNING[17500]: codec_g729.c:196 g729tolin_framein: Invalid data (4 bytes at the end)
23:55.41bkw_vad
23:55.45bkw_you might have vad on
23:56.11buddahhow do i check that?
23:56.33*** part/#asterisk rett (~rett@c-67-171-236-169.client.comcast.net)
23:57.23yashaxYep, that was it. I was missing the VM extension numbers.  THANK YOU GUYS!!!

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