irclog2html for #asterisk on 20060121

00:02.07X-Fileseh
00:03.25X-FilesDe_Mon jaike Qwell: check this too http://pastebin.ca/37714 Please
00:04.13Ariel_X-Files, do you have 2 different eyebean phones setup
00:05.21X-FilesAriel_: yes :) 1) Presence agent --> http://pastebin.ca/37698
00:05.34*** join/#asterisk doughecka (n=Miranda@unaffiliated/doughecka)
00:05.38X-FilesAriel_: 2) Peer-to-Peer --> http://pastebin.ca/37714
00:05.53X-FilesAriel_: and not work :(
00:07.13*** join/#asterisk _deg_ (n=deg@201.22.26.70.adsl.gvt.net.br)
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00:15.48X-Fileseh
00:16.16zuhy all
00:16.36*** join/#asterisk BladeRunner05 (n=feelme@adsl-222-217.37-151.net24.it)
00:16.47*** join/#asterisk colinm_ (n=colol@VDSL-130-13-10-116.PHNX.QWEST.NET)
00:16.52Dr-Linuxanybody familiar with AGI?
00:17.18zuyes
00:17.36zuand ael
00:17.52Dr-Linuxzu: can i pm you?
00:17.58*** join/#asterisk RoyK (n=roy@host-81-191-147-248.bluecom.no)
00:18.04zusure
00:18.11Dr-Linuxthanks
00:18.31*** join/#asterisk shido (n=shido@d221-68-216.commercial.cgocable.net)
00:20.01*** join/#asterisk EminEm (i=StreeT@62.162.14.79)
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00:26.57*** part/#asterisk Cresl1n (n=matt@gateway.digium.com)
00:27.10dijit0are features such as *67 things you gotta write yourself?
00:27.26*** join/#asterisk zu (n=raz@102-pool1.ras14.floca.alerondial.net)
00:27.27*** part/#asterisk jaike (n=a@203.131.137.76)
00:28.25zugot dissed
00:32.40*** join/#asterisk twisted[asteria] (n=twisted@asterisk/friend-and-developer/pdpc.professional.twisted)
00:32.41*** mode/#asterisk [+o twisted[asteria]] by ChanServ
00:33.31*** join/#asterisk _cleric_ (n=dacleric@p54828EF2.dip0.t-ipconnect.de)
00:33.41sivanasince * uses RTP ports 10000-20000, could it be possible for an answered call to disrupt a chat program that uses port 10003 ?
00:34.47BeHappy_if it's 10003 udp yes
00:35.51sivanait's a browser chat using port 10003 (ie: http://blah.com:10003)
00:35.55sivanawouldn't that be tcp?
00:36.03Peggerrhttp://www.hacktopia.net/
00:36.10*** join/#asterisk S (n=_DJ_@62.162.14.55)
00:36.20BeHappy_yes and :10003 it's the server port, your local opened port surely wont be 10003
00:36.41BeHappy_well, not surely but it would be a big coincidence
00:40.50zu~seen Corydon76
00:41.02jbotcorydon76 <three@pcp01812660pcs.nash01.tn.comcast.net> was last seen on IRC in channel #asterisk, 752d 20h 39m 55s ago, saying: 'Why don't you try it out with IAXtel?'.
00:43.11jbroomethat's a long damn time ago
00:46.24GrubsCan anyone tell me where to view debug messages logged by RxFax when using rxfax(${FAXFILE}|debug)
00:46.28*** join/#asterisk maskEd (n=masked@static-203-87-16-192.vic.chariot.net.au)
00:46.40maskEdcan anyone recommend a wifi voip phone?
00:47.11iDunnonone of them, batteries suck? :)
00:47.23Ariel_maskEd, I just ordered a UTStarcom F1000 should have it next week.
00:48.15rob0and the week after that it will be on ebay ;)
00:48.23maskEddo any do iax?
00:49.40Ariel_sip
00:49.54Ariel_rob0, no I have used them before there actually pretty good.
00:50.05maskEdyeah i understand it does sip, but do any do iax?
00:50.12Ariel_maskEd, no
00:50.27maskEdwhile on the topic of wifi phones, well kinda... does anyone use their pocket pc's as a sip phone?
00:50.31*** part/#asterisk Grubs (n=Miranda@c211-28-119-169.eburwd3.vic.optusnet.com.au)
00:50.37maskEdi have troubles with mine, its very very choppy
00:50.50Ariel_maskEd, you have a F1000
00:52.48*** part/#asterisk Utah_Dave (n=boucha@0-1pool138-109.nas28.salt-lake-city1.ut.us.da.qwest.net)
00:53.35maskEdAriel_ no, im talking pocket pc
00:53.50maskEdsjphone specifically
00:55.13*** join/#asterisk exism (n=jon@66.77.78.228)
00:55.53Ariel_ahh... I have not tried any softphone on a pocket pc
00:56.19exismhello
00:59.16*** join/#asterisk S (n=_SELEN_@62.162.14.55)
00:59.20X-FilesAriel_: what u use version eyebeam ?
00:59.50*** join/#asterisk DaRk_LoVe_[18f] (n=Fire_Sto@62.162.14.55)
00:59.59*** join/#asterisk xachen (i=justin@magnum.thisgeek.com)
01:00.06Ariel_X-Files, I don't I use hint with our polycom phones I told you yesterday I use xlite and don't use any video cams
01:00.27X-Files:(
01:01.59infinity1<PROTECTED>
01:02.35*** join/#asterisk AlexCTI (i=AlexCTI@221.sub-70-219-12.myvzw.com)
01:03.48AlexCTIhi... I'm looking for a dialer under linux, some one has a good one?
01:04.13AlexCTIand Asterisk of course..!
01:04.15*** join/#asterisk riddlebox (n=james@24-171-11-166.dhcp.stls.mo.charter.com)
01:04.58maskEdAriel_ how do you think the F1000 would stnad up next to the zyxel Prestige 2000W?
01:05.10maskEdAlexCTI iWar
01:05.44Ariel_maskEd, don't know but I have some people say it's harder to setup with asterisk. The new F1000 firmware 3.8 is made for asterisk setups.
01:06.06*** join/#asterisk exism (n=jon@66.77.78.228)
01:06.08Ariel_AlexCTI, look at vicidialer
01:08.00maskEdAriel_ ok.
01:08.08exismi'm trying to configure asterisks with the real time extension to receive calls from through SIP from a provider, would someone be interested in helping direct me on how to get things initially working? (i just started a new job and need to learn this system with no one to teach me)
01:09.09*** join/#asterisk philm (n=a@r43h15.res.gatech.edu)
01:09.25exismreguardless, should i be able to setup everything for the number in sip.conf?
01:11.48*** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net)
01:12.52Zodiacalanyone know how i can append a string to the CID on incoming calls for a specific trunk?
01:14.39sivanaZodiacal: Set(CALLERID(name)=A-Sales-${CALLERID(num)})
01:15.20Zodiacalsivana Thank You!
01:16.33*** join/#asterisk ke4qqq (n=chatzill@srv.fgp.com)
01:16.56Zodiacalthat will append to it, even if cid is unknown?
01:17.00Zodiacali hope i hope
01:17.12Zodiacalit should right.. i'll go try
01:17.14Zodiacalthanks again
01:18.43sivana:)
01:19.28sivanaSet(CALLERID(name)=${CALLERID(name)}ThisWillBeAppended)
01:20.00sivanaSet(CALLERID(name)=ThisWillBePrepended${CALLERID(name)})
01:20.03*** join/#asterisk Evanrude (n=david@ip68-107-162-212.lu.dl.cox.net)
01:21.51sivanawhy'
01:22.43exismi wish i could find some decent decumentation
01:24.01Peggerrhow come the first time I do iax2 reload I see udp packets go by but any consecutive time I run iax2 reload I see nothing?
01:27.14*** part/#asterisk mog_work (n=mogorman@gateway.digium.com)
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01:33.35*** join/#asterisk ady_ad (i=ady1mail@86.55.229.217)
01:35.25*** join/#asterisk Camisa (n=Camisa@c-67-186-94-173.hsd1.in.comcast.net)
01:35.59*** join/#asterisk pifiu-laptop (n=someone@c-65-34-166-146.hsd1.fl.comcast.net)
01:37.01*** join/#asterisk dijit0 (n=dijit0@adsl-68-127-138-64.dsl.pltn13.pacbell.net)
01:37.29dijit0is midnight commander the best console editor to edit the .conf files?
01:37.38*** join/#asterisk _deg_ (n=deg@201.22.26.70.adsl.gvt.net.br)
01:40.22Camisadijit0: I'm looking for a good conf editor too.
01:40.36dijit0mc seems to be cool
01:40.41*** join/#asterisk Evanrude (n=david@ip68-107-162-212.lu.dl.cox.net)
01:40.46dijit0i just wanna know if theres anything better
01:41.05ke4qqqdijit0: if you don't mind the learning curve vi
01:42.34dijit0i c... lol
01:43.11Camisadijito0: nano works great for beginners if you're looking for a fast editor.
01:43.50*** join/#asterisk shredthis (i=Lolita23@209.91.114.235)
01:44.11ke4qqqCamisa: I agree, if you don't have the time, or aren't already familiar with vi or emacs, nano is a good way to go, you don't have quite the same capabilities,but if you don't know about them anyway it doesn't do you any good.
01:44.43dijit0and nano is something i would need to download extra?
01:45.38Camisadijit0: should come stock with your operating syste.
01:45.57dijit0i typed nano and nothing happened, lol if thats the command to run it
01:46.00Camisake4qqq: are there any gtk based conf editors?
01:46.10Qwellgvim
01:46.12exismhow is [general] configured in sip for real time?
01:46.21Qwellexism: flat file
01:46.22Camisake4qqq: asterisk configuration tools... that are gtk based I mean.
01:46.38Qwellor you can do realtime static
01:47.00exismunder realtime static what table is it in?
01:47.16QwellYou tell it which table in extconfig.conf
01:47.56exismis it the same tables as sipusers?
01:48.00Qwellno
01:48.40exism; sip.conf => mysql, asterisk, sip.conf
01:48.42exismthat's it
01:48.44exism?
01:48.48dijit0what do you guys use to upload files to a linux box?
01:48.52dijit0scp?
01:49.27Camisadijit0: most of us use linux as our main box.
01:49.47Camisadijit0: my other people use FTP to get them into linux.
01:49.52dijit0heh, i wouldnt care either, but i only got one monitor lol
01:50.09dijit0and the other computers going to be placed across the house
01:50.21*** join/#asterisk shredthis (i=Lolita23@209.91.114.235)
01:50.36dijit0ftp... hmm, and that can be done in console right?
01:50.41De_Monwtf: exten => s,1,MixMonitor(/home/asterisk/$DEPT-${DATETIME:0:8}.gsm|a)
01:51.01De_MonIt's not using the variable's values!
01:51.59*** join/#asterisk rene- (n=root@dsl-201-133-90-176.prod-infinitum.com.mx)
01:53.09shredthismust go TODAY.  MESSAGE ME ONLY ON MSN AT MCSLTD2@HOTMAIL.COM, AIM AT OGD443 or YAHOO at MCSLTD2 IF INTERESTED! 1 alienware desktop computer price $550, one alienware area51-m 5700 notebook price $550.  prices include sameday shipping, case, wireless router.
01:53.38rob0I bid $0.01
01:53.46*** join/#asterisk tronix (n=dsf@mappy.catbert.org)
01:54.40De_Mongive me a break
01:54.53rene-hey all, i am faced to work with polycom phones, they are only three of them so i dont think tftp style deployment is justified, the phone in question is  a 301 and they wont even try to register (sip debug show nothing) they cant dial either, i have tried to configure them seeing steps for polys 501's, i once got a 301 working but i cant remember what i did and i dont have that phone at hand, help please?
01:56.32*** join/#asterisk phos-phoros (n=James@unaffiliated/phos-phoros)
01:56.50zustop spamming shredthis
01:57.00rene-spam blows
01:57.16Mark_Halversondo you need zaptel to play moh?
01:57.25rene-dont think so Mark
01:57.45Mark_Halversonjust loaded 1.2.2 on fedora 4 64bit and it's not playing moh
01:57.57rene-weird
01:58.17zuuse the nativ moh
01:58.20Mark_Halversoncalls go through...just no moh...i wonder if it's the native mp3 thing....fules are mp3
01:58.25zuinstead of mpg123
01:58.31rene-yeah
01:58.33Mark_Halversonhow?
01:58.40rene-for mp3 i think you might need an addon
01:58.50rene-you need to convert to gsm
01:59.02Mark_Halversonok...let me try convert
01:59.24De_Monsoo.. how do I get mixmonitor to use the value of my variables as the filename
01:59.29zuits in /etc/asterisk/musiconhold.conf
01:59.52*** join/#asterisk andr3www (i=andr3www@HSE-Sudbury-ppp330630.sympatico.ca)
01:59.54andr3wwwhello
01:59.55zu${varname}
02:00.03De_Monexten => s,1,MixMonitor(/home/asterisk/$DEPT-${DATETIME:0:8}.gsm|a)
02:00.09zuwrong
02:00.18zuneeds to be ${DEPT}
02:00.20*** join/#asterisk ham (n=HamYai@125.24.8.55)
02:00.38QwellBZZT
02:00.48zuyay shredthis got klined
02:01.01SplasPoodAsterisk 1.4 is scheduled to be released in the beginning of July, 2006.    Aww, for my birthday.. how sweet
02:01.28andr3wwwhey I was wondering if it was possible to use Asterisk to route a cellphone call through a house line
02:01.44SplasPoodcall forwarding? :)
02:01.47andr3wwwso I can get rid of my land line but use the existing wires to handle my cell calls
02:01.49rene-andrew you buy a thing like a cell socket and then put that in an fxo port
02:02.32andr3wwwword, once this semester of university is done and I move out I am so doin this!1
02:02.40De_Monzu should my file be named /home/asterisk/$DEPT-thedate.gsm then?  It's not.
02:02.45De_Monshouldn't
02:03.14SplasPoodandr3www: Not really, no it can't...   There are GSM<->SIP devices
02:03.24SplasPoodAnd i believe there maybe be CDMA ones as well
02:03.32SplasPoodHowever I don't think cheap is gonna be the word...
02:04.12andr3wwwWhat if i had a Treo
02:04.28SplasPoodDe_Mon: if you want the VALUE of ${DEPT} to be in the filename, then you need to refer to it as ${DEPT} :)
02:04.30rene-anybody that has installed a poly 301 with asterisk? i cant make the thing to register, i havent tried kicking it but it starts to seem reasonable
02:04.39SplasPoodrene: I have, but I'm tftp all the way.
02:04.43rene-damm
02:04.48rene-damn?
02:05.04andr3wwwIf I could program the Treo to possibly route the call through Bluetooth to the Asterisk server to run the land lane
02:05.05SplasPoodAlthough I think I originally configured a 600 via the web interface
02:05.21De_MonGrrrrrr the variable ${DATETIME:0:8} is correct. when the file saves, the name contains the characters ${DATETIME:0:8} INSTEAD of the date!
02:05.22andr3wwwI am new at this
02:05.39SplasPoodandr3www: Some people have investigated that option...  check the mailing list, search voip-info.org.. I've seen rumblings in the past
02:05.59SplasPoodDe_Mon: oh.. interesting..
02:06.30dijit0feature codes like *67  are   things that you gotta write yourself?
02:06.36rene-andrew there is a cell compatibility list for cell socket device look for your treo in those
02:06.46andr3wwwok thanks
02:07.00andr3wwwI am trying to htink of other ways to get rid of land line service but keep the land line for my cell
02:07.01SplasPoodoh yea there is the cellsocket..
02:07.11SplasPoodI didn't recall them having very broad support tho
02:07.32bkw_WTH is this I hear that AGI's must be GPL?
02:07.41SplasPoodAGIs?!
02:08.03bkw_Well we had this argument a few months back
02:08.19bkw_where the GPL can reach across a socket and bind you to the GPL
02:08.34andr3wwwthese Cell sockets cost money, my mission is to go free (as in beer)
02:08.36bkw_which isn't the case.
02:08.36SplasPoodI hear if you're not careful the GPL will steal your children in the night
02:08.49*** join/#asterisk _deg_ (n=deg@201.22.26.70.adsl.gvt.net.br)
02:08.56SplasPoodandr3www: bluetooth is your only option, and I don't know that it's much of an option at this point
02:08.57bkw_just like Woomera can talk to SS7BOX
02:09.04andr3wwwwhat do you mean?
02:09.10De_Monthe docs say ^ will be unescaped to $ for <command> lets see if it works for filename too
02:09.35rene-splaspood, it seems wasteful but i will try the mass provisioning route for my problem, what is that polycom uses? a tftp or ftp server?
02:09.50SplasPoodandr3www: if you want it to be free, you need software, and bluetooth is going to be your only connectivity option, and although I've heard people talk about it, I dunno if anything has been written
02:10.01SplasPoodrene: either, actually
02:10.08De_Monnope that didnt work
02:10.28De_Monsaving file: ^{DEPT}-^{DATETIME:0:8}.gsm
02:10.40De_Mon:( I don't wana recompile something
02:10.56andr3wwwahh I see
02:11.04andr3wwwI am down for the challenge in May
02:11.23rene-that is allright, i remember doing a tftp install for unidens, there was a general txt file and then specific files for each phone named after its mac address, my problem tho is that the dhcp server is not under my control, (is not the same asterisk box)
02:12.07*** join/#asterisk shido (n=shido@d221-68-216.commercial.cgocable.net)
02:12.42andr3wwwI will have my degree in software engineering, I can give it a shot
02:12.59andr3wwwafter spending $40,000 I should be able to :p
02:16.46rob0naw, after you get the degree you'll be too busy trying to learn what they couldn't teach you in school. :)
02:17.05De_Monexten worked.. hrmm I wonder
02:19.18andr3wwwAll i need is to add Wifi to my Treo and I'm set
02:23.38De_MonSplasPood a string that contains a $ without {}'s does not do any variable translation.
02:25.12exismif you're using real time, do you still need to use sip.conf to register sip extensions?
02:25.20*** join/#asterisk welles (n=welles@219.145.1.38)
02:25.23*** join/#asterisk wellng (n=welles@219.145.1.38)
02:25.53*** join/#asterisk GD_ (n=GD@ppp35-adsl-244.ath.forthnet.gr)
02:26.56GD_hello... has anybody managed getting an isdn cordless phone to work with hfc cards and asterisk?
02:27.33Peggerr<PROTECTED>
02:28.10*** join/#asterisk CoolAcid (n=jason@216.99.98.39)
02:30.15ke4qqqhey guys, working on integrating an asterisk box with a legacy pbx, unfortunately it's a really old pbx and doesn't support dtmf reception natively. I have tie lines setup,using pulse dialling and about 50% of the time the call gets routed to the correct place. The other 50% the call gets misrouted or the call doesn't get routed at all. Any thoughts on what can be done from the asterisk side...
02:30.17ke4qqq...to help things out?
02:31.17*** join/#asterisk Assid (n=assid@203.115.64.10)
02:37.34rene-the polycom config files named after the mac addy of the phones, do the names need the colons in them or not?
02:39.08X-FilesDe_Mon: u can say, what you use version eyebeam ?
02:41.42*** join/#asterisk CyberPony (n=CyberPon@cpe-069-132-017-022.carolina.res.rr.com)
02:43.25CamisaWhich softphone can I use to test my SIP connection? I don't know asterisk configured yet.
02:44.31blitzrageCamisa: x-lite is a decent softphone
02:45.48Camisablitzrage: x-lite's support department answered my my quickly too. except all they said was for me to hit the forums.
02:45.51PeggerrCamisa, iax is much better and easier
02:46.03PeggerrCamisa, try kiax out i really like it
02:46.18SplasPoodDe_Mon: yes, I know that..  Thats what I was telling you..
02:46.25CamisaPeggerr: blitzrage: I have a SIP connection I pay for, and I can make an outgoing call in x-lite and twinkle, but not receive any incoming calls.
02:46.26blitzragePeggerr: between iax and sip, there is no better... both have their uses, and IAX isn't perfect
02:46.39blitzrageCamisa: sounds like a configuration issue
02:47.05Peggerrblitzrage, well with iax you dont have to keep on poking holes in everyoens firewalls
02:47.36blitzrageI suppose... but I was just using X-Lite with SIP behind a firewall at a training centre all week with no problems
02:49.19*** join/#asterisk mrdigital (n=mrdigita@pool-68-236-41-109.phil.east.verizon.net)
02:49.39mrdigitali see some Linksys WRt54g router flashes to install asterisk on it what purpose does it serve/
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02:51.12maskEdwhat purpose does a stun server have?
02:52.14mrdigital?
02:53.30*** join/#asterisk Soul (n=Soul@87-196-32-88.net.novis.pt)
02:53.43CyberPonymaskEd, a stun server is used to help in situations where you are behind a NAT
02:54.41Dr-Linuxanybody ever experienced using QueueMetrics ?
02:56.23maskEdCyberPony ok.
02:57.02mrdigitalCyberPony: what about my question ? :)
02:57.49*** join/#asterisk Err (n=Err@masaka.cs.ohiou.edu)
02:57.52maskEdwell im on a lan and dont require one but sjphone is determined to resolve a stun server name even if the field is blank
02:58.43ke4qqqmrdigital: check the wiki http://www.voip-info.org/wiki-Asterisk+Linksys+WRT54G
02:58.54mrdigitaldoesnt really say much
02:59.27maskEdmrdigital i would see that it could be useful for a home situation
02:59.47mrdigitalits limited im assuming
02:59.48maskEdas a mini pbx/answering machine for a voip line
02:59.49mrdigitalno pstn ports
02:59.49SplasPoodcan someone point me to a list of country codes (dialing) in some format such as csv?
02:59.56ke4qqqsays that the author of the article hasn't gotten anything registered on it....that it doesn't do transcoding well....that it was originally designed for one extension then trunking to a central asterisk server via wireless...
03:00.03*** part/#asterisk rene- (n=root@dsl-201-133-90-176.prod-infinitum.com.mx)
03:00.09maskEdmrdigital its only capable of processing about 2 simultaneous calls
03:00.14ke4qqqso kinda a single user built in with wireless pbx
03:00.24mrdigitalok
03:01.12*** join/#asterisk NeonLevel (i=HydraIRC@cable06mcg.cybercable.net.mx)
03:01.18maskEdpersonally im going to try it as soon as i get a wifi voip phone to replace my current cordless
03:01.36NeonLevelhi good evening, anyone can show me how to route a call based on callerid?
03:01.58ke4qqqit does look pretty cool, tho the articles author's results are encouraging
03:03.34QwellSplasPood: easiest way is with cid with the pattern, like
03:03.45Qwell_NXXNXXXXXX/6265551212
03:04.20NeonLevelthanks Qwell
03:04.26NeonLevelchecking it out
03:04.43Qwellweird
03:04.58QwellWhat're the odds of two people asking that within 2 minutes of each other?
03:05.17QwellSplasPood: I was...
03:05.20*** join/#asterisk slePP (n=slepp@S0106000f663692da.ed.shawcable.net)
03:05.24QwellslePP: !
03:05.24SplasPoodNo you weren't...
03:05.31Qwellumm
03:05.39Qwellwow, don't eat and type
03:05.39SplasPood<
03:05.39SplasPood>
03:05.43Qwellyeah :p
03:05.45SplasPoodhehe
03:05.51*** join/#asterisk jef_ (i=fischer@p54845FC3.dip.t-dialin.net)
03:06.13QwellSplasPood: there is a simple to parse list on google
03:06.25Qwell~google country codes
03:06.56slePPoi
03:07.07QwellslePP: wtf have you been? :p
03:07.10SplasPoodya easy enough I suppose.. that was the best one I found
03:07.20Qwelllast I heard, you were off getting married or some such, heh
03:07.31tainted-why does STUN server insist on an alternate IP?
03:07.34slePPi've been idling here. :P
03:07.41slePPtainted-: determine nat type
03:07.48QwellslePP: My tunnel stopped working like...6 months ago. :P
03:07.51tainted-i only have one NIC
03:07.53slePPwho needs tunnels :
03:07.59slePPtainted-: doesn't mean you can't have more than one IP with one NIC
03:08.01tainted-and one server
03:08.10tainted-i only have one IP
03:08.15tainted-but it's external
03:08.16slePPah, see. the last one is the killer
03:08.17slePPyou need two
03:08.19tainted-shouldn't that be enough?
03:08.22slePPno
03:08.28slePPgo read about the types of nat, the four basic types
03:08.34slePPeach one behaves differently
03:08.41slePPso it needs two IPs to determine that behaviour
03:08.42tainted-so the two IPs have to be on the same server?
03:08.50slePPQwell: i have no uplink myself atm
03:08.54Qwellahh
03:09.00slePPQwell: but we'll get it sorted when i get my new transit back up
03:09.06slePPtainted-: yes
03:10.05*** join/#asterisk troyb (n=central@CPE00907f17e478-CM014300013422.cpe.net.cable.rogers.com)
03:10.09slePPQwell: the old tunnel i went through, their entire colo facility shut down on them
03:10.12slePPand so i'm SOL
03:10.24Qwellouch
03:11.15slePPyeh
03:11.17slePPit's bad
03:11.30slePPtainted-: you'll need to a) get another IP, b) find someone else with two IPs, c) use a public STUN server
03:12.12Juggiestun is useless
03:12.43Juggietainted, why do you need stun?
03:12.48NeonLevelis there anyway, to tell asterisk to keep with the dial plan even if the call has hangup?
03:13.00QwellNeonLevel: caller or callee?
03:13.19JuggieNeonLevel, yes. did you look @ the documentation for app dial?
03:13.30NeonLevelQwell caller
03:13.35Qwellh exten
03:13.54NeonLeveli see h exten!
03:14.00NeonLevellet me try that!
03:14.14NeonLeveland how about? the callee?
03:14.19NeonLevelis that possible too?
03:14.28slePPyes, stun is useless
03:14.32Qwellthere is an option to Dial() as Juggie said
03:14.40NeonLevelthank you both!
03:14.41Juggieneon, it depends on what you want to do
03:14.44slePPDial(1234,,g)
03:14.45Juggieyou should use the h extension
03:14.48*** join/#asterisk ravsi (n=ravsi@pool-71-108-178-182.lsanca.dsl-w.verizon.net)
03:14.52Juggieyou should not be using g
03:14.53NeonLeveli see...
03:15.00slePPwhat's wrong with 'g'?
03:15.08slePPit's non-global behaviour vs. global behaviour
03:15.24Juggienothing, but its much 'nicer' to trap a hangup in the hangup context
03:15.40Juggiethere are other reasons why the next step after the dial could be run
03:15.42Juggiethat wont be a hangup
03:15.55Juggieso rather then having to check conditions in there, it would be easier to run in the h exten
03:16.13X-Filesppls, have eyebeam version 3010z ?
03:16.27slePPJuggie: depends on purpose. g is often more useful
03:16.44slePPand it doesn't mangle your CDRs quite as easily
03:16.52Juggieslepp, what if the dial fails
03:17.01Juggieyou have to check the condition then
03:17.52slePPthat's the idea.
03:17.52*** join/#asterisk FastJack_ (i=fastjack@p5091E26A.dip.t-dialin.net)
03:18.16slePPmost people do that in 'h' anyway
03:19.40[av]banihttp://www.nasa.gov/mission_pages/stardust/multimedia/jsc2006e01008.html
03:23.10*** join/#asterisk file (n=joshnet@mctnnbsa24w-142167051174.pppoe-dynamic.nb.aliant.net)
03:23.14ravsithis really isn't a technical question, but is there a list of vonage type providers that work with asterisk? I have googled a bit I can find one that works WITH *
03:23.40ravsiI am going to move a buisness over to VoIP
03:24.04ravsibut I can't find any one that works with somthing other than there own software
03:24.11*** join/#asterisk pifiu-laptop (n=someone@c-65-34-166-146.hsd1.fl.comcast.net)
03:25.55fileslePP!
03:26.03mrdigitalravsi: i can help you
03:26.51riddleboxwhen using the GET DATA feature in AGI does it send the dtmf as 1234?
03:27.01slePPfile!
03:27.27filewhat'cha been up to?
03:28.33CamisaHow do I know if I'm behind a full cone / restricted cone / port restricted cone or symmectric NAT?
03:28.46slePPfile: uhm. working like mad?
03:28.52slePPgot our new switch in production, crap like that
03:28.58slePPCamisa: use a stun client to find out
03:29.13ravsiI am amazed at how hard its been to find company that allows other than there own stuff
03:29.18CamisaslePP: I downloaded the one from sourceforge.  I ran make on it, and don't know howto use it.
03:29.26slePPravsi: there are a lot of reasons to not allow random things onto the network
03:29.38slePPCamisa: ./client stun.server.address
03:29.40slePPi think is about it
03:30.24xachenIntellodesk is still too much in beta :(
03:30.27ravsiare the majority of * users hooking up to analog T1's?
03:30.52slePPyes, or pstn, or to other smaller carriers
03:30.57CamisaslePP: thanks.  I wonder what my stun server address is.  I have the domain/realm IP address, and I have my "nat aware proxy address" on port 5060... how do I know what the stun server address is from my SIP provider?
03:31.11slePPvonage is big, they need a controlled network. other companies are small, and can afford the resources to maintain multiple client types
03:31.27slePPCamisa: do they even have one? a lot don't
03:31.42*** join/#asterisk MikeJ[Laptop] (n=vircuser@d14-69-8-30.try.wideopenwest.com)
03:31.57slePPCamisa: try stun.sip.netmonks.ca
03:31.58ravsiwell, isn't it just SIP?
03:32.06slePPravsi: in a perfect world..
03:32.12*** join/#asterisk katakefalos (i=katakefa@194.214.77.65.in-addr.arpa.ethernext.com)
03:32.22ravsislePP: lame :(
03:32.28slePPasterisk's sip is a bit of a mess, and so it can be harder to support
03:32.39slePPplus it's not controllable.. anything could happenw ith it and there are liability issues
03:32.51slePPso, at the end of the day, mass market voip is not targetting the home grown users
03:33.02slePPwhat services are you looking for from a carrier?
03:33.43CamisaslePP:  STUN client version 0.96.... it says "Return value is 0x000015"... Primary: Independedt Mapping, Address Dependent Filter, preserves port, no hairpin...
03:34.03ravsihypothetical... if I was a major fortune 500 and wanted to go to VoIP, what would be my options?
03:34.21ravsiI always hear about how great it is
03:34.21slePPCamisa: anything else?
03:34.29katakefaloshi all, i have a TDM400 and i just installed a 4th module is there anything i have to do because it does not seem to regognize it it just comes up with 3, i did modprobe wctdm and ztcfg...?
03:34.32ravsibut I have yet to see a real application
03:34.42slePPravsi: it depends on what you want to do. fortune 500 is probably better off getting their own switch for $200k and being done with it
03:35.25ravsiwhat is asterisk's standard protocol?
03:36.01slePPiax2 is asterisk's pride and joy
03:36.39xachenalthough it is buggy :P
03:36.56ravsiand that switch goes to t1's and the like
03:37.12ravsianalog t1's
03:38.07ravsiI mean, the big reason they hype VoIP is the cost savings
03:38.33ravsibut if at the end of the day your still dealing with the local telco whats the point?
03:38.36katakefalosanyone help on this one?:  hi all, i have a TDM400 and i just installed a 4th module is there anything i have to do because it does not seem to regognize it it just comes up with 3, i did modprobe wctdm and ztcfg...?
03:39.21slePPravsi: DS3s, if you're fortune 500, but yes.. TDM
03:39.40slePPravsi: simple math here... a company we have as a customer, has 4 branches
03:39.42slePPall across canada/US
03:39.44[TK]D-Fenderravenpi : Its a question about the price of PBX hardware <-  My company saved $25k because of * and got a more functional system out of it.
03:39.48*** join/#asterisk loud (n=ariel@cypher.punk.net)
03:39.49slePPthey have 25 combined phone lines from the local telco
03:39.53slePPit costs them $65/line
03:40.09slePPand eah branch only gets X number of lines to use, and some don't need as many as others most days, but some days need more than they have
03:40.15slePPso.. they invest in a simple, single location T1
03:40.33slePPfor about $3k, add on a T1 for about $600/month, and they already have private IP link sbetween all the branches on a managed backbone
03:40.54slePPso. instead of shelling out $65 x 25, they get $600 for 23/24 lines, and capacity "on tap" at each branch
03:40.57slePPplus inter-branch calling, etc.
03:41.18slePPthey also can now use all sorts of other features like mobile workers and so on
03:42.24ravsibtw, I realllllly apreciate you answering my questions
03:42.53slePPvoip isn't just asterisk & ivr's
03:43.01slePPit has a lot of real applications that have nothing to do with the things asterisk is capable of
03:43.19slePPjust a simple branch->branch trunk over tiny little SIP gateways can make hundreds to thousands of dollars/month of difference to a company
03:43.24slePPfor a very very small initial investment
03:43.41slePPwhen you get to be big, and have hundreds of phone lines. you get a DS3 and a DS3 switch
03:43.52slePPor you go with like Level(3) and they do all your TDM<->VoIP over private connections
03:44.00slePPfor a price, of course, but in theory, less than what you'd pay each local telco
03:44.05ravsiTDM?
03:44.11CamisaslePP:  this is what I get. http://pastebin.com/515620
03:44.38slePPuse pastebin.ca, it's neater (shameless plug)
03:44.49slePPwho uses pastebin.ca, btw?
03:44.51[TK]D-FenderSangoma A200's listed at www.voipsupply.com :D
03:45.03slePPravsi: uhm.. T1s, PRIs, E1s, DS3s, etc.. that's TDM.
03:45.07slePPit's digital telephone stuff
03:45.11pifiu-laptopsleep interesting info you're putting out
03:45.19[TK]D-FenderslePP : I do.  its neater in one way, but the .COM one auto-loads after updating
03:45.30pifiu-laptopi use pastebin
03:45.34slePPuhm. pastebin.ca also autoloads?
03:45.38slePPafter about 8 seconds
03:45.41slePPbut!
03:45.44slePPnew version :> in the works
03:45.50slePPit's. cooler
03:45.54slePPand about 15 times faster
03:46.19pifiu-laptopok good to hear
03:46.27fileslePP: your IPv6 address for pastebin.ca doesn't work btw
03:46.31slePPi know!
03:46.35slePPi lost my ipv6 uplink about 4 weeks ago
03:46.43slePPand they're .. screwing the pooch in getting me back up
03:47.26pifiu-laptopsleep can you explain the setup your company has again?
03:47.30slePPuhm
03:47.31ravsiok so if I am switching a smaller company the gains are going to be on the hardware and the features and less so on the long distance costs(outside of branch offices) and local call cost
03:47.33slePPit's a bit different, but..
03:47.39pifiu-laptop1 t1 for 3k?
03:47.41slePPwe have a big softswitch, which has some DS3s plugged into it
03:47.57slePPand then we have about 8 servers behind that with asterisk/ser/openser/custom apps/etc. to do services
03:48.15slePPravsi: again, it all depends on what they want to do
03:48.20*** part/#asterisk NeonLevel (i=HydraIRC@cable06mcg.cybercable.net.mx)
03:48.33slePPpifiu-laptop: well.. you get your T1 for whatever price you get your t1 for. from what i understand, in the states, it's _cheap_
03:48.41slePPup here in canada, it's about $600-$800/month for a single PRI
03:48.49dijit0as far as feature codes go, do i have to make my own? ex. like *67 and stuff??
03:48.57pifiu-laptopright but what was the initial 3k investment on?
03:49.03slePPbut then you get some sort of gateway to work with that. so it could be like a vega 400, or an asterisk box w/ a sangoma or digium T1 controller, or various other things
03:49.07slePPall of which cost about $3k to build
03:49.19pifiu-laptopoh gotcha
03:49.23pifiu-laptopand you said you have 4 branches?
03:49.26pifiu-laptopand each brand has a t1?
03:49.30slePPdijit0: some of them yes, some of them no. depends on your end-devices
03:49.35pifiu-laptopso 23 lines each location?
03:49.36slePPpap2's, for example, do almost all the *xx's internally
03:49.38slePPpolycom's don't
03:49.51*** join/#asterisk konfuzed (n=KonfuzeD@H135.C72.B0.tor.eicat.ca)
03:49.53slePPpifiu-laptop: PRI has 23, CAS/RBS T1 has 24..
03:50.00slePPfor reference, i'm a carrier, not the customer i was talking about :>
03:50.04dijit0ahh i c.. thx.. and these are a part of extensions.conf as well?
03:50.06slePPbut their setup is one T1 in one city
03:50.14pifiu-laptopwhats the difference?
03:50.16slePPso 23 lines in one location
03:50.21slePPall shared from 4 cities
03:50.27slePPvs PRI and CAS?
03:50.28ravsislepp: we are looking to replace a old office phone system and voice mail for 2 offices and about 25 lines
03:50.38slePPone has a d-channel, one doesn't. one supports more features than the other, at the loss of a channel
03:50.42pifiu-laptopright pri vs cas?
03:50.52[TK]D-FenderPRI > CAS
03:51.03pifiu-laptopwhat is a d channel?
03:51.08slePPravsi: then voip makes sense, because the intiial hardware investment is cheaper or the same as new hardware PBX/KSU, and at the same time, opens up the world to new possibilities of building a phone system the way _you_ want it
03:51.17*** join/#asterisk bmg505 (n=leon@c1-91-8.rndf.isadsl.co.za)
03:51.20slePPinstead of a $10,000 feature on a hardware PBX for music on hold/queueing, you can use asterisk for cheap
03:51.31slePPpifiu-laptop: on a PRI, the d-channel is what is used to setup/end calls
03:51.37ravsiright
03:51.39slePPit sends signals between your equipment and the telco to do it
03:51.48pifiu-laptopas oposed to?
03:51.52pifiu-laptopinteresting never messed with a t1
03:52.07ravsiI shouldn't really expect any real gains on the local phone bills though
03:52.07slePPwell, with a T1, you have 24 channels at 64kbps
03:52.21slePPwith a PRI, you get 23 clear-channels (B channels) which carry nothing but audio
03:52.28slePPand one channel that carries nothing but data
03:52.38slePPwith CAS, you get 24 channels that steal bits of the 64kbps to do signalling
03:52.40pifiu-laptop64kbps though
03:52.46slePPso when you do a call, it sends the signal over a specific channel, like channel 7
03:52.52slePPthen maybe channel 4. etc. whatever channel it's going to use
03:53.11slePPravsi: not usually, no
03:53.19ravsithis has all been very informative
03:53.24slePPravsi: but again, it may be cheaper if you can get away with simpler/cheaper lines because of some other features somewhere
03:53.34slePPlike i could sell a company 20 lines for about $300/month
03:53.36*** join/#asterisk Naturalblue (n=Kay@195.26.12.229)
03:53.38*** join/#asterisk X-Files (i=x-files@x-files.lv)
03:53.40slePPpure voip
03:53.56slePPwhich is much cheaper than the local telco, which would run them about $1200
03:54.05ravsiright
03:54.13ravsibut you don't provide for 626 numbers :)
03:54.21slePPnah
03:54.25slePPcheck out didx.org :>
03:54.35Qwellravenpi: Where you at?
03:54.39slePPthey're a 3rd party broker for DIDs
03:54.39ravsicali
03:54.46Qwellyes, duh
03:54.47X-FilesPlease, say me good soft where worked Message, Online Status Line in users and Voice . Please
03:54.49Qwell626...where?
03:54.54ravsiLos angeles
03:54.55[TK]D-FenderslePP : having just quoted a PRI +/- $600/$800 mth, thats hardly $1200 :)
03:55.04slePP[TK]D-Fender: analogs
03:55.05Qwellravsi: yes, WHERE? :P
03:55.08slePParen't the same as a PRI
03:55.11ravsiArcadia
03:55.16ravsiand Glendora
03:55.17slePPin this case, centrex analogs
03:55.35slePP:>
03:55.40[TK]D-FenderslePP : I gues business analog lines @ 35$ tops.  20 lines = $700.....
03:55.42slePPwe migrate loads of centrex people away from stuff
03:55.46ravsi$300 for 20 lines would represent a sweeeet deal over current prices
03:55.51slePP[TK]D-Fender: not here, no
03:55.52*** join/#asterisk College (n=ben@adsl-34-44.swiftdsl.com.au)
03:55.55slePPthey average $65 a pop
03:55.57[TK]D-FenderslePP : Where?
03:56.00slePPedmonton
03:56.17[TK]D-FenderYou're kidding.. that major metropolitain area...
03:56.27slePPwelcome to edmonton :>
03:56.28[TK]D-FenderWho runs that shit, telus?
03:56.33slePPcentrex w/ direct dial is expensive
03:56.34slePPyes, telus
03:56.39slePPpbx style hunt groups are not so bad
03:56.43[TK]D-FenderFuckers... I hear the suck out west...
03:56.46slePPthey're evil
03:56.50slePPthere's a reason we move so many
03:57.11pifiu-laptopqwell i got IAX2 working
03:57.14pifiu-laptopwohooooo
03:57.15[TK]D-FenderslePP : Then again your cost of living and tax rate are pretty nice...
03:57.29slePPcost of living here is cheap, except downtown core in buildings own by boardwalk (ew)
03:57.33slePPand yes, no PST is good
03:57.39slePPprovincial taxes in general aren't too bad
03:57.45slePPgas is cheap :>
03:57.53slePPbut, data & tdm aren't
03:58.02[TK]D-FenderOh well...
03:58.05slePPwe pay about $4/gb for bandwidth, or $250/sustained mbit
03:58.08slePPdepending on carrier
03:58.28[TK]D-FenderslePP : Whats the base included BW?
03:58.29slePPand the DS3s each cost about $12k/month + $2k/month for the SS7 trunks
03:58.36slePPbased included? are you crazy? :>
03:58.37ke4qqqhey guys, working on integrating an asterisk box with a legacy pbx, unfortunately it's a really old pbx and doesn't support dtmf reception natively. I have tie lines setup,using pulse dialling and about 50% of the time the call gets routed to the correct place. The other 50% the call gets misrouted or the call doesn't get routed at all. Any thoughts on what can be done from the asterisk side...
03:58.39ke4qqq...to help things out?
03:58.42konfuzedslePP: gas is cheap in edmonton - i suppose you mean relatively like 5cents less or something
03:58.54slePPkonfuzed: at times, 10s of cents less
03:59.03slePPwhen all the prices went to like $1.20 and stuff, it was just over a dollar here at the same time
03:59.08slePPright now, it's 79 or 82 or something
03:59.15ravsididx would be great if there was someone selling in my area
03:59.21slePPravsi: thus the problem, then
03:59.35[TK]D-Fenderkonfuzed : Try Quebec's tax-cut on that... as soon as you cross tot he 401 is drops 10%
03:59.46slePP[TK]D-Fender: we pay about $2/gb for our commited bandwidth, and a bit more for overage
04:00.02slePPbut only on the per-gb peer. the other upstream is $450 for the wire, and $250/mbit
04:00.06konfuzedah i dont pay fopr gas any way.
04:00.10slePPthen our fibre is about $1700/month
04:00.13ravsiso the idea of that is that somes sets up a server, plugs in a tonne of t1's or ds3's then resells them over the net
04:00.16konfuzedI was just in montreal last week and it was crazy
04:00.18[TK]D-FenderslePP : That blows.
04:00.19slePPfor a 2 mile layer 2
04:00.21slePPit does
04:00.40slePPravsi: of didx? or someone with 4 channels sells one number. whatever, basically
04:00.47slePPwe have 12 numbers for sale on didx, just for "fun"
04:00.59ravsiand they can offer lower costs becuase they bought in bulk?
04:01.04slePPbut we have a lot more numbres we could sell than that
04:01.11slePPwho, didx or the person selling it?
04:01.17slePPthe seller picks the price, didx is just a broker. like ebay :>
04:01.30*** join/#asterisk Naturalblue (n=Kay@195.26.12.229)
04:01.32slePPthey're a pure middle man
04:01.59[TK]D-FenderI'm abusing my work setup so I have PRI + 1 at home effectively :D
04:02.09slePPheh
04:02.19ravsiand your able to offer a lower cost becuase you buy in bulk
04:02.21slePPit's fun
04:02.24slePPravsi: yes
04:02.31slePPbecause our connections get cheaper the larger it goes
04:02.44*** part/#asterisk College (n=ben@adsl-34-44.swiftdsl.com.au)
04:02.44slePP28 T1s will cost, in theory, more than a single DS3
04:02.45[TK]D-Fenderif I didn't need my a voice phone-line for DSL I'd be dry-loop maybe direct off of the PRI or redirected.
04:02.46konfuzedif ya go to montreal - do not bother stopping at Shwart's Deli
04:02.55slePPbuying 1000 numbers at a time is a lot cheaper than 10 at a time
04:02.56[TK]D-FenderCan you really own a DID and point it to any service you want?
04:03.00X-FilesPlease say me , need soft phone for windows, where worked Message, Status Line users (busy/offline/online) and Voice ?
04:03.08[TK]D-Fenderkonfuzed : Smoke Meat Pete's <-
04:03.13slePP[TK]D-Fender: the CRTC is a bit "fuzzy" on that
04:03.14[TK]D-Fenderkonfuzed : Godly...
04:03.28konfuzed[TK]D-Fender: ahh thats what I needed
04:03.41[TK]D-FenderslePP : Yeah I've been following the coalitions efforts against it...
04:03.42slePPX-Files: x-ten pro or eyebeam
04:03.55slePPafaik, that coalition is a client
04:04.16[TK]D-FenderslePP : A bunch of ISP's
04:04.19X-FilesslePP: eyebeam not worked status users and message , i how tested
04:04.26slePPX-Files: not sure, then
04:04.35slePP[TK]D-Fender: which coalition?
04:04.44slePPthere's one that was started in edmonton
04:04.48[TK]D-FenderX-Files : So you added ext's as "buddies" and it won't show their status?
04:05.10slePP'MyBuddies'..
04:05.17X-Files[TK]D-Fender: offline
04:05.45X-Filesi use for test version eyebeam 3010n
04:06.33[TK]D-FenderslePP : http://www.quebecispcoalition.ca/pressrelease05.html
04:06.34slePPanyone know of a decent b2bua, btw?
04:06.39slePPah, not that one
04:06.48[TK]D-FenderslePP : What wrong with that?
04:07.01[TK]D-FenderX-Files : Pastebin your dialplan
04:07.12slePPno, not the coalition i was thinking about, i mean
04:07.16ravsiI gotta find someone in my area that has a underused ds3
04:07.18*** join/#asterisk JMcA (n=jmcadams@71.31.33.169)
04:07.28slePPtelus is giving out raw DSL now
04:07.31[TK]D-FenderslePP : same ruling I think will cover canada as a whole
04:07.37ravsi:)
04:07.39slePPbut we still can't get them to give us copper loop at regular copper loop pricing
04:07.42slePPthey want hundreds per loop
04:07.43[TK]D-FenderslePP : Then again given your rates... hmmm
04:07.45X-Fileswait
04:07.46konfuzedravsi whats your area
04:08.14slePPravsi: heh. those're the best. call up some big company, and go get their DS3s :>
04:08.26konfuzedslePP is that raw dsl like Naked DSL
04:08.42slePPsure, let's call it 'naked'
04:08.44ravsikonfuzed: 626 Los Angeles
04:08.54slePPbut dsl w/out dialtone, as it were
04:09.00slePPbut it's stupidly expensive
04:09.02konfuzedoh well here in toronto its known as Naked DSL
04:09.03QwellslePP: yeah, we call that naked dsl
04:09.06slePPfor like $10 more, you get dialtone
04:09.24BeHappy_uhm.. there's a way i can send an Unauthorized error at INVITE from a sip user that has not REGISTERed with my asterisk?
04:09.27[TK]D-FenderslePP : At Bells band-rate system yeah...
04:09.27slePPthat's probably because you're all weird easterners
04:09.30konfuzedI should be able to setup naked dsl oncopper without phone service, just about anywhere in canada
04:09.50X-Files[TK]D-Fender: http://pastebin.ca/37698 <<- sip.conf and extensions.conf
04:09.52slePPBeHappy_: if you don't have a 'guest' sort of sip user, it will do that by default
04:10.07slePPout here, we have a few customers we provide RADSL to
04:10.10konfuzedgood for voip and such ;^)
04:10.15slePPand it costs us about $15/month for the copper "alarm loops"
04:10.19slePPbut they won't sell us those anymore
04:10.22BeHappy_slePP, what do you mean with 'guest' sip user?
04:10.29*** join/#asterisk Mag1KaL (n=Mag1KaL@S010600112f0d62ac.wp.shawcable.net)
04:10.30BeHappy_slePP, a sip user with no secret?
04:10.30slePPBeHappy_: check sip.conf for a [guest] entry
04:10.33slePPyes
04:10.35slePPand no host
04:10.36slePPetc.
04:10.50slePPkonfuzed: most of our customers are pulling fiber to our data center. that's fun :>
04:10.59slePPbut, we don't have a lot of little customers either.
04:11.22konfuzedthats fun and probably over kill but hey
04:11.24[TK]D-FenderX-Files : You shouldn't need the seperate subscribe context, and bring the hints into the main.
04:11.27slePPnot overkill, no
04:11.32konfuzedmaybe there is no infrastructure in edmonton yet
04:11.33slePPbut it's fun
04:11.43ravsislepp and all: thanks agian you have been most helpfull
04:11.49BeHappy_i have one user with username but without secret
04:11.53slePPkonfuzed: that's probably correct
04:12.01slePPlike i said, a 2 mile fibre run is $1700/month or whatever
04:12.11slePPthough we have 18 strands of that crap. heh
04:12.20X-Files[TK]D-Fender: check line 46-48 not this /
04:12.21X-Files?
04:12.30konfuzedso your pulling howmuch bandwidth on that fiber?
04:12.30slePPBeHappy_: check what asterisk thinks the sip caller is doing..
04:12.32BeHappy_and one user with no username but host=ip
04:12.43slePPkonfuzed: anywhere between 1.5mbit to 100mbit, depending on peer
04:12.51slePPbut only 3 strands are lit right now
04:13.01konfuzedllllike burstable?
04:13.12slePPsustained 100mbit on two, 1.5 burstable to 10 on one
04:13.16slePPthen we have our internet uplinks
04:13.35konfuzedfiber for less than 5MB is ludicrous (unless it is last option cause no other infrastructure and wireless would not do
04:13.47slePPthat's pretty much exactly it
04:13.51konfuzednot that im picky about bang for your buck ;^)
04:13.56slePPwhen it has to work, and be solid, fibre it is
04:14.02slePPwe don't pay for it :>
04:14.04slePPjust our own core fibre
04:14.12slePPthe customer pays their own connectivity
04:14.17konfuzedoh well thats different can t   beat free firber
04:14.27slePPwell, it's not like we can use it for anyuthing :>
04:14.28slePPjust voip
04:14.29[TK]D-FenderX-Files : http://pastebin.ca/37753
04:14.32slePPand that's from them to us
04:15.12X-Files[TK]D-Fender: ok wait..
04:15.42konfuzed4 load balanced 3Mb adsl is better than a 10Mb fiber cause of redundant links
04:15.54konfuzedand less than half the price
04:16.01konfuzedjust when im havin fun that is ;^)
04:16.05Mag1KaLWhat's the best software phone right now?
04:16.07slePPthe internet is our redundant link
04:16.22[TK]D-FenderMag1KaL : eyeBeam
04:16.23slePPwe bgp to the peers and we're multihomed ourselves across two DC's
04:16.33slePPand most of the customers we have are ISPs or ISP related :> works out nicely
04:16.37dijit0is there a simple dial plan command that i can use to authenticate users from a voicemail pass?
04:16.42konfuzedok i recognize each customer plot unto its own conundrums and priorities ;^)
04:16.48troybslePP what is your ASN?
04:16.54slePP14595
04:16.59troyb*grin*
04:17.23slePPkonfuzed: customers are crazy
04:17.36konfuzedslePP: so how did you end up with internet from shaw cable then??
04:17.45slePPlook at our path :>
04:17.55troybslePP fixedorbit isnt picking it up, i'll have to open a session to a router
04:18.16slePProute-views.org
04:18.21slePP13911 and 15290 are direct upstreams
04:18.23X-Files[TK]D-Fender: i replace files from site, restart asterisk and 2 eyebeam , this same problem, i can't see status :(
04:18.30slePP13911 peers directly to 852 and 6327
04:18.52slePPand 15290 is a peering whore
04:19.01konfuzedslePP: got a site I can look at with your hobby projects ;^)
04:19.08troybslePP yeah that is one of my sources :)
04:19.17slePPkonfuzed: netmonks.ca, geeksanon.ca.. pretty much those
04:19.25slePPpastebin.ca is of course a pet project
04:19.39[TK]D-Fenderdijit0 : VMAuthenticate
04:19.46troybslePP i like to play with the Juniper box
04:19.50slePPwho doesn't :>
04:19.54troybslePP peering shouldnt result in an AS announcements
04:19.56[TK]D-Fenderdijit0 : Try checking the WIKI for things like that....
04:20.02slePPtroyb: it doesn't
04:20.06dijit0i checked asteriskguru.com
04:20.08troybthere should simply be a static route between the party(s)
04:20.11slePP15290 and 13911 are upstream transit
04:20.19[TK]D-Fenderdijit0 : WIKI <-
04:20.28slePPbut we bgp to our non-transit peers and advertise up to a /28 and accept up to a /28
04:20.34slePPwith some very agressive timers
04:20.40dijit0ok thanks
04:20.42slePPwith no-export, of course. :>
04:20.46Mag1KaLWhat's the best open source software phone right now? ;)
04:20.59troybslePP i didnt think most carriers were able to route less then a /24
04:21.02X-Files[TK]D-Fender: in 2 eyebeam configured Presence mode peer-to-peer
04:21.11troyba /28 is like 16 IP's
04:21.18slePPyes it is
04:21.19[TK]D-FenderX-Files : You did a reload and nothing?
04:21.41slePPsome of our stuff there is good reason for a more specific, due to MED and so on
04:21.55X-Files[TK]D-Fender: yes ;(
04:22.03slePPand since the private peers are supposed to be pure voip, we keep certain ranges full of web servers off the link
04:22.12JMcAI've never understood the obsession that most carriers have with prefix length
04:22.19[TK]D-FenderX-Files : Pastebing a "sip show hints" from CLI
04:22.19troybslePP with most peering xchanges they use internal IP
04:22.23slePPwe have a peering agreement to our peers to advertise/receive longer than /24's, but we only advertise /24's to the world (one /24, at that)
04:22.24X-Filesok
04:22.36slePPtroyb: the last thing i want is private IPs in backbone voip :>
04:22.39troybjeez a provider with a C class eh? :)
04:22.46X-Files[TK]D-Fender: mabye "show hints" ?
04:22.55slePPour other ranges are aggregate inside our upstreams
04:23.21X-Files[TK]D-Fender: http://pastebin.ca/37756
04:23.24troyboh i see so its not Direct Allocation?
04:23.32[TK]D-FenderX-Files : yes that
04:23.45konfuzedslePP: is there anyone providing DID from that edmonton datacenter
04:24.01[TK]D-FenderX-Files : And what do you have in eyebeam for buddies to watch?
04:24.04tronixslePP: btw if needed, telnet route-server.gblx.net (open to public)
04:24.18troybslePP mind if i route you a /32 172.16.0.0 ;)
04:25.18slePPtroyb: the /24 we advertise isn't from arin, no
04:25.30slePPi already have a /32 172.16.x.x :P why do i want more?
04:25.36X-Files[TK]D-Fender: one pc have contact 14@ip_asterisk and other pc have contact 13@ip_asterisk ...
04:25.41Qwella /32?
04:25.49slePPkonfuzed: anyone provided DID from that data center? hmm?
04:25.50troybslePP your a hard customer :P how about 10.0.0.0 :)
04:25.56[TK]D-FenderX-Files : get rid of the suffix, it should only be 13 & 14
04:26.03slePPi'll let you send me 10.8.3.48/32.. deal?
04:26.14troybslePP sounds like a plan :)
04:26.22QwellslePP: make him provide his own nm + brd :p
04:26.23X-Filesok wait
04:26.24slePPkonfuzed: are you asking if i can give you edmonton DIDs? :P
04:26.29[TK]D-FenderI'm considering getting a /29 or so for my home LAN....
04:26.31troybslePP what happens when i need 1 more iP?
04:26.42troyb*if
04:26.44slePPtroyb: i'll give you 192.168.89.44/32
04:26.46Qwelltroyb: better ask for a /29 :p
04:26.59troybQwell yeah, i need the spares for my BBS
04:27.01Qwell[TK]D-Fender: I had /29's on my LAN
04:27.05slePPi hate nat
04:27.09Qwelleach box was it's own NAT, heh
04:27.13troybslePP no kidding :)
04:27.23troybRogers gives me 1 IP but im sure i could 'get' more
04:27.27slePPso i'm very tempted to take our new class c allocation and give myself some over a tunnel :>
04:27.33slePPshaw gives me two
04:27.34JMcAnat is evil
04:27.36konfuzedslePP: not precisely - nothing I saw suggested you provided them yourself - so I would look into any DID provider out of that datacenter
04:27.55X-Files[TK]D-Fender: i remove @ip_asterisk and press save, but @ip_asterisk restored ;(
04:28.03konfuzedim rather disappointed at the lack of support from my current DID provider
04:28.09slePPkonfuzed: oh :> yes, we do. we provide to calgary, edmonton, vancouver and toronto right now
04:28.13JMcAI spent pretty much all day trying to get some of our customers to understand why I didn't want to route 1918 space over VPN's to/from them
04:28.15slePPwith 29 other centers coming on as needed
04:28.16troybslePP i wonder what would happen if you created a second advertisement for a /32
04:28.29[TK]D-FenderX-Files : Are you sure presence support is enabled in eyebeam?
04:28.30slePPtroyb: to the public? both upstreams would filter it immediately
04:28.41slePPbut my peers may not. i don't think they setup very good filters
04:28.42troybyeah it wouldnt have a fighting chance
04:28.45slePPthey sent me shitloads of bad routes early on
04:28.57X-Files[TK]D-Fender: Yes, i turn "Peer-to-Peer" .
04:29.00slePPnot to mention my own filters would drop the /32 unless i added it to the prefix lists.. :>
04:29.15[TK]D-FenderX-Files : Don't think that is the way...
04:29.24konfuzedslePP: then have you got a site I can look with info about DID availability and rates
04:29.35slePPwell, depends on how much info you want
04:29.36X-Files[TK]D-Fender: your version eyebeam ?
04:29.41slePPwww.thinktel.ca
04:30.46X-Files[TK]D-Fender: i can put to pastebin debug from asterisk , pc 1 eyebeam and pc 2 eyebeam
04:31.09troybslePP do you peer over private circuits?
04:31.41slePPyes
04:31.52troybinteresting :)
04:32.05troybwhat facility are you guys in?
04:32.32*** join/#asterisk EriSan (n=erisan@81-174-25-141.f5.ngi.it)
04:33.09slePPwhich stuff? :>
04:33.11slePPallstream tower & cn tower
04:33.49[TK]D-FenderX-Files : Not sure what to do from here...
04:33.55*** join/#asterisk bkw__ (n=brian@ppp-70-128-122-10.dsl.tulsok.swbell.net)
04:34.04Qwellbkw__: !
04:34.22troybslePP this is toronto?
04:34.28slePPno
04:34.29slePPedmonton
04:34.39troybCN tower is in edmonton?
04:34.40slePP90% of this can be seen from our AS :>
04:34.50JMcACN Tower is in Toronto
04:34.50slePPyes, head of west coast and US operations of CN is in edmonton
04:34.55slePPit is also in Edmonton
04:34.56[TK]D-FenderX-Files : Works for me....
04:35.01slePPand there's one in Calgary
04:35.02[TK]D-FenderX-Files : just tried it.
04:35.11troybapparently there is more then one cn tower ;)
04:35.15slePPof course
04:35.24slePPjust like there're more than one telus and bell tower
04:35.25X-Fileshm
04:35.26slePPit's just the name
04:35.31slePPthe one in toronto is "The CN Tower"
04:35.37troyboh :)
04:35.38JMcA"The" CN Tower is in Toronto
04:35.39slePPthe one i edmonton.. well, it's just cn's tower
04:35.41slePPso, it's cn tower
04:35.52JMcAyeah...that's where I was going with that
04:35.55[TK]D-FenderX-Files : What does your contact say int he list while you're on the phone?
04:37.09X-Files[TK]D-Fender offline
04:37.22[TK]D-FenderX-Files : I dunno.. restart your phone
04:37.34X-Filesrestarted :)
04:38.42X-Files[TK]D-Fender: this same problem ;( 0 online users
04:38.47troybslePP im trying to determine the equiv command on a zebra box for displaying bgp data :(
04:39.18slePPshow ip bgp
04:39.30slePPshow ip bgp paths XXXX
04:39.35troybyeah i just got it :)
04:39.37troybthanks
04:40.29troybslePP that doesnt work when you want to input an as#
04:40.39slePPuhm
04:41.21slePPthat's interesting..
04:41.41troybits damn straightforward when your connected to the router directly
04:41.52troybthen as you said you can just do show ip bgp
04:42.01slePPweird
04:42.05slePPi forget how to do that in quagga now
04:43.16troybare you 195.x ?
04:43.23troyberr 159
04:44.37troybi think i found you :)
04:45.10troyb<PROTECTED>
04:45.10troyb*  159.18.161.0/24  195.66.226.109                         0 15444 15290 14595 i
04:45.35troybeh slePP?
04:45.54slePPyessir
04:45.59slePPshort path
04:46.21troybare they tunneling your traffic from toronto?
04:46.30slePPno
04:46.35slePPwe have local links
04:46.38slePPis that the only path you see?
04:47.19troybi see a lot of links in here :)
04:47.44bkw__OMG its slePP
04:47.49slePPOMG IT IS
04:48.11troybyour weighting everything at 0, atleast your not penny pinching with another carrier
04:48.11bkw__troyb, you having bgp drama?
04:48.18slePPhe's looking me up
04:48.21troybbkw__ not anymore
04:48.29troybroute-views.linx.routeviews.org> show ip bgp regex 14595
04:48.34slePPtroyb: we do MED to peers, not transit
04:48.34troybthat seems to do it
04:48.37slePPno point in transit
04:48.59troybhow much traffic are you guys pushing?
04:49.41slePPaggregate is about 10-15mbit/s
04:49.47slePPpublic is about 7mbit peak
04:49.47*** join/#asterisk postel (n=jp@host86-139-209-144.range86-139.btcentralplus.com)
04:49.49slePP3mbit sustained
04:49.53troybOrgName:    ThinkTel Communications Ltd.
04:49.54troybOrgID:      TCL-93
04:50.06slePPie, fuck all
04:50.06slePP:>
04:50.08bkw__AS 36348
04:50.29troybslePP im surprised you went to the time and trouble with BGP, though 3Mbps of VoIP traffic is still substantial
04:50.48troybmost DMS phone switches have an OC3 interoffice connection so. :)
04:50.52bkw__troyb, you have your own ip allocation from ARIN?
04:50.58troybbkw_ i do not :)
04:51.06troybmaybe at some point but not at present
04:51.15slePPtroyb: we're about uhm... 25 meters away from the dms500 we pull DS3s from
04:51.19troybbkw_ i dont see a point in paying ARIN for use of IP
04:51.28slePPwe're full bgp and everything for a few reasons, not the least of which is failover/multihoming
04:51.33bkw__troyb I can tell a provider to fuck off without having to renumber
04:51.40slePPbut, we also have only put our switch into production around the last week of december
04:51.45slePPso, we went from little to very big :>
04:51.46troybbkw_ you gave me a good laugh :)
04:51.52slePPand now we're moving up. about 2 months ago, our sustained was about 2mbit
04:51.56bkw__I don't like to renumber 2000 boxes
04:52.05troybslePP im impressed :)
04:52.12Qwellbkw__: oh come on, it's fun!
04:52.20bkw__OH HELL NO
04:52.25troybwhat kind of switch are you running, Nortel i assume?
04:52.27slePPand we have three new private peers to bring online in the next month
04:52.28bkw__ARIN is some IP nazi bastards
04:52.36[av]banigodwin
04:52.37slePPwhy on earth would i run a nortel switch? :>
04:52.40Qwellmy work has pre-ARIN allocations
04:52.45slePPbkw__: do you have a /32 ipv6 allocation yet?
04:52.50troyblol
04:52.55bkw__slePP, they don't give you one that small
04:52.55Qwelllike a /16 worth
04:53.04bkw__I have a /22 right now
04:53.06bkw__but I need more
04:53.12bkw__so i'll have another /22 in 3 months
04:53.23bkw__but the ARIN nazi's won't give me what I need to even cover my current needs
04:53.24slePPno
04:53.28slePP<PROTECTED>
04:53.30slePPnot /32 ipv4 :P
04:53.32slePPthat'd be stupid
04:53.34bkw__fuck ipv6
04:53.35bkw__nobody uses
04:53.45slePPonly americans think that :>
04:53.51slePPeither way, it's free
04:53.53bkw__until I can get a native v6 connection then I will
04:53.54slePPso you should get one anyway
04:53.58slePPwhere're you at?
04:54.08bkw__slePP, a true allocation from ARIN might take me an arm and a leg to get
04:54.13troybbkw_ /22 is pretty substantial 1024
04:54.14bkw__they waive the fee's for it
04:54.21DarkFlibbleyou don't need native ip6
04:54.24slePPa /32 from arin is easy
04:54.28DarkFlibbleuse an adhoc yunnel for now
04:54.30bkw__slePP, I can get it
04:54.33slePPit's free if you have an ipv4 block or arin membership
04:54.33bkw__but I don't want to tunnel
04:54.37bkw__slePP, right
04:54.39slePPthat's why i asked where you are
04:54.40bkw__IF APPROVED
04:54.47slePPapproved. yes... heh
04:55.01slePPthey're giving it away like campaign buttons
04:55.08bkw__I'll get one then
04:55.12troybslePP are you doing VoIP only or providing copper services as well?
04:55.21slePPtroyb: voip only to the edges, at least
04:55.24Qwellbkw__: give away tunnels with asterlink service.  heh
04:55.25*** join/#asterisk angler_ (n=angler@pcp01540308pcs.huntsv01.al.comcast.net)
04:55.25troybDarkFlibble can i borrow an octet or two
04:55.30troyb*laughs*
04:55.30slePPand a /48 for the real stuff
04:55.32Qwell2c/meg
04:55.34bkw__Qwell, what do you think i'm going to do :P
04:55.36Qwellheh
04:55.45bkw__tiz why I wanted a /19
04:55.49troybslePP i call the the first 4 bits
04:55.51bkw__but dem bitches won't give me one
04:55.52Qwellbkw__: well...hook it up
04:56.04bkw__I already returned a /24 to our provider
04:56.04DarkFlibble65536 networks of 16billion ips for my house...
04:56.12bkw__I have to return 1 more /24 before I ask for more IP's
04:56.16slePPDarkFlibble: not enough!
04:56.16DarkFlibbleerr... billion billion even
04:56.26slePPdon't forget the link local's and site local's :>
04:56.40troybslePP what i really want is for Bell to let me buy a copper pair to do point to point SDSL
04:56.44DarkFlibbleslePP, site local is depreciated
04:56.51QwellDEPRECATED
04:56.54Qwellno i
04:57.00Qwell~depreciated
04:57.05jboti guess depreciated is "you can use it, but it's no longer the best way and will probably be unsupported soon"
04:57.05Qwell~deprecated
04:57.07jboti guess deprecated is a typo of depreciated, see depreciated, or not a typo according to the jargon file... oh well here it is "you can use it, but it's no longer the best way and will probably be unsupported soon" since when become typos new words
04:57.07*** join/#asterisk coppice (n=chatzill@204.206.17.210.dyn.pacific.net.hk)
04:57.15slePPtroyb: telus is being sticky out here for that.. again, we have 4 copper (well, 8 pairs, 4 runs) right now with RADSL
04:57.17Qwellmeh...lies
04:57.24slePPbut they refuse to sell us more without low-pass filtering
04:57.27slePPDarkFlibble: so?
04:57.34slePPDarkFlibble: that shouldn't stop you from abusing it :>
04:57.35Corydon76-home~dict depreciated
04:57.46Corydon76-home~dict deprecated
04:57.47DarkFlibbleso.. you have link local and then world routable...
04:57.55Qwellyes, evil
04:57.59DarkFlibbleno need for anything else
04:58.14Qwelllike SetCallerID
04:58.18slePPsilly
04:58.19Corydon76-homeThe dictionary is not something other people can mess with
04:58.30slePPthat's like saying you don't need 10.0.0.0/8 on a single segment
04:58.30slePPsheesh
04:58.31slePP:>
04:58.48troybgeez time to learn about OSPF areas
04:59.13troyb1sweet :D
05:00.10DarkFlibbleso...one area...
05:00.16DarkFlibbleand its like wtf
05:00.24Qwellbkw__: So, we gonna go out drinking one night at ETel?
05:00.49troyb1DarkFlibble area 0 is my fav
05:00.51angler_Corydon76-home, i finally have a pn shirt that isn't way to big!
05:01.04bkw__Qwell, YES
05:01.10Corydon76-homeangler_: fabulous.  Wanna buy another?
05:01.13DarkFlibblethey also had both outgoing connections on a single router...
05:01.22QwellI get the feeling I've already asked you that :p
05:01.26DarkFlibblenever did find out why
05:01.46Corydon76-homeSo when that shirt starts looking grayish, you'll still have a black PN9 shirt?
05:02.41angler_Corydon76-home, maybe for pn 5
05:03.01Corydon76-homeThose shirts are all sold out
05:03.23angler_Corydon76-home, yea johnny x said he might make some though... he said to keep bugging him about it though
05:03.43Corydon76-homeHeh
05:03.48slePPi like when ospf carries like.. 4 routes
05:03.51slePPnice and clear
05:04.11troyb1slePP what happens when you max out on areas.. jenga :)
05:04.23troyb1can you say.. packet storm
05:04.52bkw__you can reuse areas in OSPF can't you if they don't overlap?
05:05.30DarkFlibblebkw_, sometimes...
05:05.59bkw__thats what I thought
05:06.24troyb1bkw_ what happens when you need to send packets from point 1 to point 2?
05:07.00DarkFlibbleyou basicly are advertising routes... not areas but it depends on implementation...
05:07.13slePPit just determines intra/inter based on area
05:07.15troyb1fair enough
05:07.22slePPdunno why'd you'd do that, though..
05:07.28slePPhave 4 area 0's :>
05:07.36*** join/#asterisk NeonLevel (n=NeonLeve@dsl-200-78-104-84.prod-infinitum.com.mx)
05:07.38slePPospf over gre is fun
05:07.43troyb1slePP any chance you want to let me play with your phone switch :P
05:08.10slePPhell. no.
05:08.10slePP:>
05:08.13mrdigitalanyone do web design?
05:08.23coppiceadvertising routes? can you advertise special offers on them too? send two packets, get one free?
05:08.48NeonLevelgood evening guys, sorry to bother you again, what i need now is to keep my cel phone bill, so i was checking the callback.agi app, anyone has this working? and could please help me?
05:09.07DarkFlibblecoppice, no...but you can advertise it half price (cost)
05:10.09mrdigitalguess no one does
05:10.28konfuzedmrdigital: i know someone that does
05:10.34Peggerrdoes anyone here use * with chan_sccp2?
05:10.51konfuzedmrdigital: but I suspect that you have unrealistic expectations
05:11.09killer-chenough beer for today .. going to bed .. n8 all together
05:11.16QwellPeggerr: the one from berlios.de?
05:11.23NeonLeveldid anyone readme?
05:11.24mrdigitalkonfuzed: what do you mean?
05:13.45NeonLevelcallback.agi or app_callback ?? anyone???
05:14.01*** join/#asterisk pengyong (n=lala@218.93.119.110)
05:16.32konfuzedmrdigital: ive never met someone wanting help from a developer that did not have unrealistic expectations
05:17.04coppicekonfuzed: I have - another developer :-)
05:17.06slePPbut the grandparent container's background :>
05:17.12konfuzedslePP: <--- knows unrealistic Geek Gods so that doesnt count
05:17.35konfuzedcause he can have his unrealistic fantasies developed on a whim
05:17.56konfuzed;^)
05:18.14konfuzedslePP: im not a developer ;^)
05:18.35konfuzedI just translate from those who want development to those who do development
05:18.55slePPoh :>
05:19.01slePPfine then
05:19.32konfuzedslePP: but I'll see If I can get a JavaHelp geek to take a stab at it
05:19.56slePPi don't think it's possible
05:19.57DarkFlibblein answer to the number of possible areas in ospf... its at least 10... can't find an exact figure in my books... but I have examples in the cisco ccie course books with 10...
05:20.03slePPwith pure and un-hacked-up CSS
05:20.11PeggerrQwell, what is berlios.de
05:20.20slePPDarkFlibble: isn't the only limit memory/cpu/etc.?
05:20.27slePPthat and actual field size in the packets
05:20.49DarkFlibbleto be honest I don't know... since many implementations actually number areas
05:21.14DarkFlibbleso its likely to be 256 or 65536
05:21.43DarkFlibblenever need an OSPF AS of more than 6 in real life tho
05:22.20DarkFlibbleme goes back and biggles at whet he just wrote...
05:22.22konfuzedslePP: i know you sent a couple of paste bin addresses but do you have ont that explains the objective and problem/errors and such
05:22.29konfuzedlke a well formed request
05:22.39DarkFlibbleI have never needed an OSPF AS with more areas than 6 in real life tho
05:22.56slePPkonfuzed: uhm. of which?
05:23.15slePPand that's even overkill
05:23.20slePParea 51 is a useless stub
05:23.25konfuzedthe best translating I do for non techs is to explain the well formed support request gets well formed easy solutions
05:23.27slePPand area 2 is nearly pointless
05:23.34konfuzedslePP: for the new pastebin :^)
05:23.41slePPkonfuzed: oh!
05:23.43DarkFlibbleyeah... this was a massive corporate network... *cough* bank *cough*
05:23.48slePPyes. no, not really
05:23.51slePPi just want opinions at this time
05:23.56slePPi'll get to bugs when i get it done this week
05:24.01slePPDarkFlibble: heh
05:24.39konfuzedslePP: i cant ask a geek i know unless I can provide a clear plroblem scenario, right.
05:24.51konfuzedpart of my translating ;^)
05:25.50DarkFlibbleslePP, rfc 2329 says that vendors have attempted upto 7 areas in a domain
05:25.54DarkFlibbleirl
05:27.16*** join/#asterisk klictel (n=klictel@modemcable119.206-200-24.mc.videotron.ca)
05:27.17xachen<3 pastebin :)
05:27.42slePPkonfuzed: i did describe the one issue :> but that's not a big issue. i imagine its impossible
05:27.53slePPDarkFlibble: that's interesting
05:28.14DarkFlibbleno need for more than that imo...
05:28.45HamYaIDarkFlibble: ManxPower said earlier about letting * play sounds before actually asnwering the line, u know how to do that?
05:29.06DarkFlibblenope... but I can read the scroll back and google it...
05:29.15slePPearly media
05:29.19slePPPlayback(invalid,noanswer)
05:29.26slePPsends SDP in a 183
05:29.48slePPso the call never connects
05:29.53HamYaIslePP: only in 1.2?
05:29.55slePPbut it will time out during 18x progress eventually
05:30.03slePPno, i do that in cvs from january of last year and 1.0.5
05:30.07slePPand in 1.2
05:30.14slePPxachen: v3.pastebin.ca
05:30.47DarkFlibblehttp://www.voip-info.org//tiki-pagehistory.php?page=Asterisk+cmd+Playback&diff2=6 <-- its actually on the wiki...
05:31.33HamYaIDarkFlibble, slePP: k, thanks
05:33.50slePPi'm bored now
05:34.04DarkFlibbledo what I do...
05:34.10xachensleP: nice :)
05:34.16xachenYou cuold fix my intellodesk now :)
05:34.27DarkFlibblehave 7 channels open... and read the web at the same time
05:34.32slePPwhat's an intellodesk? :P
05:35.00xachenits the latest in Helpdesk software
05:35.02xachenstill beta though
05:35.04slePPoh dear
05:35.13xachenI got it for a $59 special when it will be retailing for $600~
05:35.28*** join/#asterisk Trazz (i=Trazz@c-67-163-92-37.hsd1.il.comcast.net)
05:35.28xachenYeah. But intellodesk has the potential to outgrow cerberus and kayako
05:35.47HamYaIwhy is the 24 ports TDM so expensive?
05:37.07slePPT1?
05:37.07QwellHamYaI: expensive?  not really
05:37.07QwellslePP: tdm2400p, 24 port analog
05:37.07slePPah
05:37.07HamYaIslePP: analog
05:37.08slePPnew hardware
05:37.08Qwellit's like what, $1500 for 24fxs?
05:37.08QwellslePP: fairly, yeah
05:37.08xachen24 port analog....
05:37.09HamYaIQwell: yeah, but the digital ones are cheaper
05:37.09slePPwhy not get a dual T1 controller
05:37.09Qwellamphenol connector, 6x4 slots
05:37.09slePPand a channel bank :>
05:37.09Qwelldigital?
05:37.09Qwellthat isn't analog
05:37.15Qwell6x4 modular, of course
05:37.46HamYaIslePP: to get an E1 connection here it's like $2,500 for the setup fee
05:38.10slePPvs how much for 30 analogs?
05:38.19slePPmonthly & setup
05:38.33slePPout here in canada, a T1 is cheaper than the equivalent analogs
05:38.49slePPof course, i was thinking you were going to use the tdm2400p for internal lines
05:38.56HamYaIslePP: it's the same actually but I still need less than 30
05:39.06slePPeither way.. an E1 channel bank -> E1 controller may be easier/more "expandable"
05:39.34slePPthen you just plug in as many as you need to the bank and can always upgrade to a full E1 with about 30 seconds downtime :>
05:39.58HamYaIslePP: yeah, thinking about it
05:40.56HamYaIslePP: I'm just building up test stuffs, that's why I wanna lower the cost
05:41.20De_MonSplasPood (several hours ago) Oh.. Leme recheck the wiki then. I don't remember that being said...
05:41.29X-FilesPlease help, http://pastebin.ca/37767
05:43.04*** join/#asterisk lilo (i=levin@freenode/staff/pdpc.levin)
05:44.47*** join/#asterisk shekhar (n=shekhar@221-128-139-3.exatt.net)
05:45.06DarkFlibblealso T1/E1 uses digital signalling... its a much cleaner solution...
05:45.40slePPfaster
05:45.42slePPvroom
05:46.01slePPHamYaI: pickup an E1 controller and get a fractional E1
05:46.08De_Mon<PROTECTED>
05:46.14DarkFlibblejust read the amiling list for all the problems with off-hook detection on analogue
05:46.40DarkFlibbleworth it just to not face that problem
05:47.45*** join/#asterisk zu (n=raz@1-pool1.ras14.floca.alerondial.net)
05:48.10DarkFlibblealmost 6am... gonna go and eat breakfast
05:49.58*** join/#asterisk Camisa (n=Camisa@c-67-186-94-173.hsd1.in.comcast.net)
05:52.48*** join/#asterisk dudes (n=dudes@12-215-33-205.client.mchsi.com)
05:54.13TrazzWhats the best version of linux to install asterisk on these days?
05:54.24DarkFlibblefc4 worked well for me...
05:54.33DarkFlibblebut other rave about openSuSE 10
05:54.59*** join/#asterisk FuriousGeorg1 (n=brian@ool-44c5a9b8.dyn.optonline.net)
05:55.01DarkFlibblecentos is a nice possibility tho...
05:55.02[av]baniuse whatever youre comfortable with
05:55.08[av]baniasterisk cares not
05:55.14DarkFlibblesince its a RHEL clone
05:55.36FuriousGeorg1did freenode upgrade or something, i didnt have to register
05:56.57troyb1i use CentOS
05:57.37HamYaIslePP: don't think fractional E1s are available here
05:57.44NeonLeveland what's the difference between CentOS vs WhiteBox?
05:57.44slePPah
05:57.46slePPthat'll be a problem
05:58.10DarkFlibbleonly reason I chose fc4 over centos was because I needed some of the dev stuff that isn't in centos and didn't feel like messing with dependacies
05:58.18DarkFlibbleNeonLevel, not much
05:58.23FuriousGeorg1anyway, since i upgraded to 1.2 "#-xfers"  are not working.  i added the lines to features.conf and everything
05:58.58FuriousGeorg1i cant figure out what im missing
05:59.47*** part/#asterisk FuriousGeorg1 (n=brian@ool-44c5a9b8.dyn.optonline.net)
05:59.52*** join/#asterisk FuriousGeorg1 (n=brian@ool-44c5a9b8.dyn.optonline.net)
06:01.21*** join/#asterisk alphaque (n=alphaque@218.111.24.41)
06:04.45Trazzso what does the zapatel 1.2.2 fix over 1.2.1 ?
06:05.04JunK-YTrazz: see the changelog.
06:05.16TrazzJunk, anything notable ?
06:05.35DarkFlibbleI suppose I really should finish off my web interface for uk dids
06:05.43DarkFlibblebbl
06:06.00*** join/#asterisk thazza (n=thazza@229.9.233.220.exetel.com.au)
06:06.21X-Filesppls, please check http://pastebin.ca/37767
06:06.24Trazzhas any web based or pc based gui been created to to configuration?
06:06.39slePPTrazz: asterisk management portal
06:06.40slePPor something
06:07.47*** join/#asterisk wasim (n=wasim@pdpc/supporter/active/wasim)
06:08.54Trazznice..
06:08.55Trazz/bin/sh: -c: line 0: syntax error near unexpected token `;'
06:08.55Trazz/bin/sh: -c: line 0: `if [ -n "" ]; then  if [ -f  ]; then mv -f  .ba
06:09.09Trazzmake: *** [install] Error 2
06:09.09Trazz[root@voip zaptel-1.2.2]#
06:09.18Trazzcant make install the zaptel 1.2.2
06:19.18*** join/#asterisk zu (n=raz@12-pool1.ras14.floca.alerondial.net)
06:28.07*** join/#asterisk wasim (n=wasim@pdpc/supporter/active/wasim)
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06:43.06*** join/#asterisk shekhar (n=shekhar@221-128-138-206.exatt.net)
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06:49.58[av]baniwow packet8 quality is terrible
06:51.21Trazzwhat is the graphical call manager and where can i get it?
06:51.52FuriousGeorge~fop
06:51.53jbotAn XSL formatter written in Java that outputs PDF. URL: http://www.jtauber.com/fop/, or the Flash Operator Panel
06:52.20FuriousGeorgeFlash Operator Panel
06:52.37Trazzis it good?
06:52.56FuriousGeorgeanyone know why #-transfer stopped working when i upgraded to 1.2?
06:54.07*** join/#asterisk Husk (n=Husk__@202.55.153.169)
06:55.05*** join/#asterisk IronHelix (n=irc@ool-45785cfe.dyn.optonline.net)
07:04.14wasimFuriousGeorge: new DEA policy against using hash
07:06.27thazzawasim: Whats the new policy say?
07:06.38wasimdon't use #
07:08.06FuriousGeorgewasim: actually i use hash twice
07:08.22FuriousGeorgei smoke two joints in the morning, i smoke two joints at night
07:08.27FuriousGeorgeseriously though
07:08.32wasimFuriousGeorge: good stuff?
07:08.34thazzawasim: Any reason why so?
07:09.00[av]banihttp://video.google.com/videoplay?docid=4776825453418327083&q=shirt+fold
07:09.04wasimthazza: they seem to think that hash is bad for you some reason, as opposed to alcohol and tobacco
07:09.24FuriousGeorgei use ##, at least i want to, but since i upgrade neither # nor ## works *and yeah its a pretty good "beaster" as we call it)
07:09.27thazzawasim: oh i see.. It is a drugs joke..
07:10.08FuriousGeorgethazza: this just in:  The DEA doesnt care how you transfer calls
07:10.19wasimas long as you don't use hash!
07:10.40FuriousGeorgeas long as your not making money off of drugs (without kicking up to uncle sam) the DEA dont care
07:11.16thazzawasim: or the vars of $cocaine, or $whitestuff
07:11.25FuriousGeorge"i smoke two joints in the afternoon, makes me feel all-right"
07:11.26wasimnow those are bad for you
07:11.45FuriousGeorge$MR_BROWN_STONE
07:12.18FuriousGeorgeanyway, if no one knows about my transfer issue im going back to watching forensic files
07:12.26*** join/#asterisk IronHelixz (n=irc@ool-45785cfe.dyn.optonline.net)
07:12.27FuriousGeorge~seen mog_work
07:12.37jbotmog_work <n=mogorman@gateway.digium.com> was last seen on IRC in channel #asterisk, 8h 56m 28s ago, saying: 'wanna fight about it ^_^'.
07:12.37thazzalol @ FuriousGeorge
07:12.38FuriousGeorge~lastseen mog_work
07:12.42thazzaFuriousGeorge: Mog does work?
07:13.06FuriousGeorgeneeds to be doing some on astjab.org, i finally set aside a minute to install it
07:13.09FuriousGeorge:)
07:13.19FuriousGeorgewhoops its up
07:13.24thazzaFuriousGeorge: I think jbot is just showing you that i am correct and mog doesn't do anywork.
07:14.05FuriousGeorge:)
07:14.13FuriousGeorge~lastseen mog_home
07:14.25FuriousGeorge~seen mog_home
07:14.28jbotmog_home <n=mogorman@user-24-236-84-48.knology.net> was last seen on IRC in channel #asterisk, 1d 14h 8m 14s ago, saying: 'bye peoples'.
07:14.45FuriousGeorge~seen mog_drunk_and_gambling
07:14.47jbotFuriousGeorge: i haven't seen 'mog_drunk_and_gambling'
07:15.05FuriousGeorgejbot: consider yourself lucky
07:15.11thazzaLOL @ FuriousGeorge
07:15.35FuriousGeorge~FuriousGeorge
07:15.37jboti guess furiousgeorge is a knife-fighting monkey last seen with The Man with the Yellow Bat
07:15.47thazza~seen FuriousGeorge_In_A_Dress
07:15.49jboti haven't seen 'furiousgeorge_in_a_dress', thazza
07:16.00thazzajbot: Lucky for you.
07:16.03FuriousGeorgejbot: stop liein cuz ur friends are around
07:16.04jbotACTION leaps to his feet and stops liein cuz ur friends are around
07:16.16FuriousGeorgethats better
07:16.42hohumhey
07:16.49FuriousGeorgeho-oo
07:16.58hohumI can't for the life of me figure out why VoiceMailMain is not working on my dialplan
07:16.58hohum!
07:17.06hohumVoiceMailMain()
07:17.09Trazzjust installed 1.2.2 and Jan 21 01:29:10 WARNING[5295]: file.c:821 ast_streamfile: Unable to open press-7 (format ulaw): No such file or directory
07:17.14FuriousGeorgehohum: now your supposed to say hey again
07:17.22hohumno thanks though
07:18.01FuriousGeorgeexten => 5000,1,voicemailmain , right
07:18.07FuriousGeorgeor somethng like that
07:18.40hohumexten => 9000,3,VoicemailMain()
07:18.54FuriousGeorgeyou do have a 1 priority right
07:18.56FuriousGeorgeand a 2
07:19.08hohumI do
07:19.15hohumexten => 9000,1,Answer()
07:19.16hohumexten -> 9000,2,Wait(1)
07:19.27FuriousGeorgecli output
07:19.31hohum:w
07:19.37FuriousGeorge?
07:20.03hohum<PROTECTED>
07:20.03hohum<PROTECTED>
07:20.03hohum<PROTECTED>
07:20.04hohum<PROTECTED>
07:20.04hohum<PROTECTED>
07:20.17hohumand the Playback is being called from my Hangup
07:20.19hohumso
07:20.31hohumit's going from that exten to hangup
07:20.42FuriousGeorge== Auto fallthrough, channel 'SIP/3102-9bec' status is 'UNKNOWN'
07:20.46hohumskipping the VoicemailMain call
07:20.57Qwellwait
07:21.03FuriousGeorgei never got that
07:21.07QwellDid you paste the priority 2?
07:21.11Qwellor did you retype it?
07:21.14Qwellnote the ->
07:21.18hohumoh
07:21.21hohumgrrrr
07:21.33Qwellno...let me
07:21.40Qwell~lart hohum
07:21.46wasimno, no, please let jbot
07:22.14*** join/#asterisk Someone123 (n=m00p@pcp0011543623pcs.mainf01.in.comcast.net)
07:22.44wasimin @ pk, 2nd test, 1 day, first session is even stevens ... 120 for 2
07:24.37FuriousGeorgeawww, shake that thang 0-/-<
07:24.56FuriousGeorge(baby, shake that thang) 0-\-<
07:25.47*** join/#asterisk [1]JohnJacob (n=m00p@pcp0011543623pcs.mainf01.in.comcast.net)
07:28.16*** join/#asterisk Mark_Halverson (n=mhlvrs@67-139-119-152.dsl1.pco.ca.frontiernet.net)
07:29.12FuriousGeorgeQwell: i know that you know, that you know that i know, that you know why my #-xfer stopped working when i upgraded to 1.2
07:30.16QwellFuriousGeorge: new DEA rules
07:31.12wasimtee hee
07:31.21*** join/#asterisk cyber (n=kani@220.247.245.36)
07:31.38FuriousGeorgelol
07:31.50FuriousGeorgei like when we keep it going like that
07:31.58DarkFlibble~DarkFlib
07:32.00jbotsomebody said darkflib was great
07:32.00FuriousGeorgeforensic files it is
07:32.06DarkFlibbleyay!
07:32.21FuriousGeorge:|
07:32.23DarkFlibble~DarkFlibble
07:32.50DarkFlibblehmmm...
07:33.03DarkFlibblethat line must've been there for a while...
07:33.14FuriousGeorge~Darkfibble
07:33.38FuriousGeorgejbot: no, darkfibble enjoys the company of young male farm animals
07:33.58FuriousGeorgehmmm
07:34.11FuriousGeorgecan you believe they disabled my favorite feature of jbot
07:34.17DarkFlibblejbot: FuriousGeorge was last seen skulking around strange bars in red light districts
07:34.29FuriousGeorge:|
07:34.35DarkFlibble~FuriousGeorge
07:34.38jbotmethinks furiousgeorge is a knife-fighting monkey last seen with The Man with the Yellow Bat
07:34.38*** join/#asterisk dmz (n=dmz@209.133.52.162)
07:34.41FuriousGeorge~darkfibble
07:35.02dmzhey y'all is asterisk-users not in use anymore?
07:35.09DarkFlibblewho is this darkfibble? Can't be me since I have an L in my nick
07:35.10FuriousGeorgei wish i was in holland :)
07:36.02DarkFlibbleI wish I had a proper job
07:36.24cyberhey is any one using radius with asterisk
07:36.40DarkFlibblecyber, not yet... but maybe soon
07:36.53cyberit's not possible is it ?
07:37.00DarkFlibbleyes...
07:37.06*** join/#asterisk Qwell (n=north@unaffiliated/qwell)
07:37.10cyberOIC
07:37.25DarkFlibblemost methods are a little convoluted...
07:37.57cyberDarkFlibble what can i use for billing
07:38.45DarkFlibblecyber, what requirements do you have and what infrastructure already exists?
07:39.02Mark_Halversonany good places to learn abount dundi?
07:39.17cyberi have installed asterisk and h323 on a linux server
07:39.20DarkFlibbleMark_Halverson, try leifmadsen.com
07:39.24cyberall works gr8
07:39.31Mark_Halversonthanks
07:39.58DarkFlibblenp
07:40.41DarkFlibblebrb... door
07:42.09DarkFlibbleib
07:42.38cyberDarkflibble do u use php agi for billing ?
07:43.17*** part/#asterisk ManWithTheMetalB (n=brian@ool-44c5a9b8.dyn.optonline.net)
07:43.54DarkFlibblecurrently... but only because I haven't wrote any proper app for it
07:44.12DarkFlibblecauses a second delay in setup using agi...
07:44.12cyberi saw a2billing
07:44.29cyberwhich looks very good
07:44.35*** part/#asterisk dmz (n=dmz@209.133.52.162)
07:46.47DarkFlibblenot looked at alternatives to my own stuff since my requirements are fairly narrow
07:47.00cyber:)
07:47.25cyberbut i need a wholesale one :(
07:48.33DarkFlibbleI'm probably gonna have to write an extension to asterisk to do what I want or hook into the manager interface...
07:56.25*** join/#asterisk [Airwolf] (n=airwolf@82-171-75-4.dsl.ip.tiscali.nl)
07:57.05Mark_Halversonok let me ask this here...i dont understand dundi yet - I have a TDM DS3 and willing to offer the * community access to USA Toll-Free @ no charge - could this be made available using dundi?
07:57.18wasimyes
07:57.19DarkFlibbleyes
07:57.28Mark_Halversonok
07:57.57Mark_Halversonhow does dundi handel callerid  -  the only restriction is that i MUST pass a valid ANI/CallerID
07:58.36Mark_Halversoni sent leif an email -  but I am going to need some assistance setting this dundi thing up
07:58.39wasimyou can either choose to trust and vet inbound clid and if failed, set your own
07:58.46wasimleif == blitzrage
07:59.08DarkFlibbleleif co-wrote the oreilly book
07:59.09Kattyallo.
07:59.36Mark_Halversoni love oreilly - there in my home town...and soon to move back.....Sebastopol, CA
08:00.16Mark_Halversoni didn't know there was an oreilly book on dundi - let me surf and check
08:00.23Qwellnot dundi
08:00.26Qwellasterisk..
08:00.29Qwell~thebook
08:00.43jbotextra, extra, read all about it, thebook is Asterisk: The Future of Telephony, released under the Creative Commons license and available at http://www.asteriskdocs.org << Read the book online!
08:04.44*** join/#asterisk ^^Gu[L]Can (n=uvey-Kiz@85.108.151.16)
08:07.24Mark_Halversonok that will be some exciting reading tomarrow 376 pages!
08:07.44drumkillait's a great book  :)
08:08.05Mark_Halversonk
08:08.06Qwellindeed it is
08:08.57*** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk)
08:09.28zoahey ho drummer
08:09.42Qwellzoa: Gonna get some guys to test the new idefisk right now. :)
08:09.51zoasuper!
08:10.55Qwellsheesh, no ebuild? :p
08:11.11zoawhat is an ebuild ?
08:11.14Qwellgentoo
08:11.23Qwell"package"
08:11.35zoaaaah
08:11.49zoabut gentoo would want to compile it and we dont provide sources
08:11.55*** join/#asterisk argos73 (i=1000@jason.argos.org)
08:12.01Qwellit can do binary packages too...like sun j2re
08:12.08zoaaha
08:12.17zoai will try to make some packages for it
08:12.31zoa.deb .rpm and ebuild then :)
08:12.47QwellI'd say the first two at least.
08:12.50QwellI was mostly joking. :)
08:13.10zoawell, im adding new things to the todo list all the time :)
08:13.20iDunnono source?! insanity!
08:13.21zoait can always improve :)
08:13.53zoaiDunno: will you pay the programmers ? :P
08:14.10iDunnodo they work for a pint a week?
08:14.20Qwellactually...
08:14.32Qwellzoa: The cool thing about making an ebuild, would be automatic dependancy checking
08:14.37zoayeah i know
08:14.43Qwelllike this libexpat...I have no clue where it comes from, but I need it
08:14.43zoathat also why i was thinking about it
08:15.04zoaproblem is that i need yet another guy on making packages every day then :(
08:17.57argos73is there a "clean" way to check for the existence of a DB variable (sans DBget) ?   GotoIfThisDbVariableWasNotSet(1206) application???   want to be able to change timeout on a Dial() per extension, with default 20 secs if DB variable doesn't exist...
08:18.49drumkillaargos73: yes
08:18.55drumkillaargos73: show function DB_EXISTS
08:19.33argos73drumkilla: cool...  tnx.  (that new in 1.2?  just upgraded about an hour ago)
08:19.43drumkillayes, it is new
08:19.48argos73gotcha
08:20.13argos73that'll work
08:20.30drumkillaGotoIf(${DB_EXISTS(foo/bar)}?2:3)
08:20.45argos73perfect
08:20.53DarkFlibblequick question... I am getting the caller id in my agi... but not in the ${CALLERID} variables... why could this be happening?
08:20.56drumkilla:)
08:21.27DarkFlibbleits also known to be send (in the IAX debug...)
08:21.50DarkFlibblehmmm...
08:22.16DarkFlibbleis ${CALLERID} one of the variables that was converted to a function in 1.2?
08:23.07zoayes
08:23.09zoajust a sec
08:23.33zoahttp://www.asteriskguru.com/tutorials/dialplan_functions.html
08:23.35drumkillayes, but CALLERID wasn't removed ...
08:24.37DarkFlibblefound the changes.... its been almost a year since I left asterisk to work on other projects so I wasn't uptodate with the changes
08:24.50DarkFlibblehttp://www.voip-info.org/wiki/index.php?page=Asterisk+func+callerid is what I needed
08:26.37drumkillaor just 'show function CALLERID'  :)
08:28.07DarkFlibbleyeah... wasn't aware that the old method was long gone
08:28.17DarkFlibble:)
08:28.23drumkillait's not
08:28.28*** join/#asterisk power1 (i=daemon@dsl-146-51-208.telkomadsl.co.za)
08:28.29drumkillaor sholdn't be ......
08:28.29drumkilla:)
08:28.52*** join/#asterisk svenna_ (n=svenna@p548D399D.dip0.t-ipconnect.de)
08:30.22power1Any1 here using a tdm400p and asterisk @ home, I want to make only cpecific sip phones ring based on which fxo channel on the digium card the incoming call originates on...can any1 help?
08:30.53drumkillaDarkFlibble: yeah, they're still there ...
08:31.15DarkFlibbledidn't work on my box...
08:31.20DarkFlibblehmmm... no matter...
08:31.28DarkFlibblethe new method will be fine
08:32.23Qwellzoa: works good, except the whole oss thing
08:32.27Qwelloss vs alsa, that is
08:33.07zoaexplain ?
08:33.31Qwelldunno, oss is old school
08:34.01*** join/#asterisk spunz_ (n=spunz@h081217096096.dyn.cm.kabsi.at)
08:34.02zoait doesnt work with alsa ?
08:34.37QwellIt wants /dev/dsp
08:34.52Qwellalsa uses stuff in /dev/sound/
08:35.17zoaaha
08:35.28zoawe could probably fix it by using a newer portaudio
08:36.04*** join/#asterisk TonyM (n=TonyM@adsl-solo-80-168-225-147.claranet.co.uk)
08:36.40zoahow do you like it for the rest ?
08:36.46zoaeuh
08:36.47Qwelllooks great
08:36.49zoathat was not english
08:36.56zoaworking fine ?
08:37.10*** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com)
08:37.14Qwellwell, I haven't tested the audio...my soundcard is flaking out on me.  nothing will play audio right now
08:37.49zoaah damn
08:38.37Qwellbut, the others guys are...they had real problems with iaxcomm...they say this is working much better
08:38.42Qwelland a lot cleaner looking to boot
08:38.49zoacool
08:41.36power1<PROTECTED>
08:55.16*** join/#asterisk Mag1KaL (n=Mag1KaL@S010600112f0d62ac.wp.shawcable.net)
08:56.17Mag1KaLWhat are the requirments to be able to play mp3s?I want some on hold music but I'm just not getting any...
09:00.17NewSolehmm
09:02.04DarkFlibbleMag1KaL, it used to be one specific version of the mpg decoder... I'm not sure if that is still true
09:02.43zoampg123, version 0.59r is needed
09:07.38argos73drumkilla: after some drunken coding, the DB_EXISTS thing works nicely.  thanks
09:08.43*** join/#asterisk edwin_ (n=edwin@252-131-222-203.rev.techex.net.au)
09:11.09DarkFlibbleMag1KaL, if you can't find it I should have a staticly compiled version knocking around
09:20.54Mag1KaLNo I found it and installed it. Does Asterisk need a recompile though?
09:21.18DarkFlibblenope
09:21.55DarkFlibblehmmm... my moh breaks up a little unless I press a button... then its perfect...
09:22.10DarkFlibblenot sure of the cause of that
09:22.50Mag1KaLIt works ;P
09:23.43DarkFlibbleputtimg a wait in seems to work
09:23.53DarkFlibblestops the break up
09:24.28Mag1KaLHm, it only works when my SIP softphone asks my IAX one to hold...
09:27.39*** join/#asterisk RoyK (n=roy@10.80-203-106.nextgentel.com)
09:30.00RoyKmorning
09:30.10Mag1KaLHello
09:30.19Mag1KaLCrap it is morning.
09:37.06DarkFlibbleits 9:30... of course its morning
09:37.44*** join/#asterisk dmz (n=dmz@209.133.52.162)
09:38.01Mag1KaL4AM here
09:39.17dmzcan i ask a user question here, or should it goto #asterisk-users?
09:40.58power1I want to make only specific sip phones ring based on which fxo channel on the digium card the incoming call originates on...can any1 help?
09:42.12DarkFlibbledmz, not many people in #asterisk-userrs
09:42.39dmzyeah nooone there
09:42.54dmzthat's why i wanted to ask here :)
09:43.06dmzkinda wierd that noone is in asterisk-users
09:43.43DarkFlibblebrb... phone
09:46.43*** join/#asterisk fenlander (n=fenlande@82.152.81.57)
09:57.01*** join/#asterisk apardo (n=apardo@62.97.121.95)
09:57.54*** join/#asterisk pb__ (n=pb@cpc1-cmbg6-5-0-cust20.cmbg.cable.ntl.com)
10:02.26*** join/#asterisk abatista (n=Ariel@dsl-20-177.cofs.net)
10:06.55dmzok i'm just going to ask (since noone is @ asterisk-users)...i have asterisk going in my colo on a public IP, am playing with twinkle and know it works great with local net asterisk. i setup twinkle to use my coloip as the gateway, can dial voicemail, meetme, etc...but my dtmf isn't being picked up by the asterisk server..any suggestions?
10:11.28*** join/#asterisk benjk (n=benjamin@f8a01-0357.din.or.jp)
10:11.57*** join/#asterisk speedwagon (n=Ariel@dsl-20-177.cofs.net)
10:15.29*** join/#asterisk [Airwolf] (n=airwolf@82-171-75-4.dsl.ip.tiscali.nl)
10:19.07*** join/#asterisk tonico (n=tonico@chello213047065175.12.14.univie.teleweb.at)
10:21.15dmzdoh, ignore that. i had dtmf settings screwed up. now just gotta get my dial plans together :) yeah
10:21.21DarkFlibblelol
10:21.29DarkFlibblewas just reading your problme
10:21.39dmzi feel stupid
10:21.50dmzguess that's what i get for drinking while setting it up
10:21.56dmzok, time for some scotch to celebrate
10:22.03DarkFlibblewhats twinkle?
10:22.16DarkFlibbleahh... linux softphone
10:25.15*** join/#asterisk phos-phoros (n=James@unaffiliated/phos-phoros)
10:26.03dmzyeah
10:28.01dmzis there any easy way to debug the dialplan/extensions? i think i have fwd & iaxtel in ok, but don't know if the calls i'm making are making it to those contexts..
10:29.13Qwellzoa: still around?
10:29.20dmzahh i think i get it...hmm pulling off too many #s, ignore me i'm going to drink now
10:29.33DarkFlibbleadd NoOp(myextname) as the first pritority
10:30.20DarkFlibblecan't type...
10:30.26DarkFlibblekeep tpyoing
10:38.02*** join/#asterisk felipex (n=dsfdsf@85-18-250-142.ip.fastwebnet.it)
10:38.41dmz:( iax2 & fwd both keep saying circuit busy :( oh well something for tomorrow
10:40.42DarkFlibblechanges in FWDs IAX config (enbling/disabling) can take a while to kick in...
10:40.55DarkFlibbleuse SIP until they do...
10:41.11DarkFlibblesince you can have multiple registrations on FWD
10:42.41RoyKzoa: ping
10:42.45RoyK~seen zoa
10:42.53jbotzoa is currently on #asterisk (1d 2h 27m 46s). Has said a total of 140 messages. Is idling for 1h 40m 10s, last said: 'mpg123, version 0.59r is needed'.
10:42.56DarkFlibble~seen DarkFlib
10:42.57jbotdarkflib <darkflib@dialup357.ts002.bmt.esat.net> was last seen on IRC in channel #asterisk, 320d 12h 37m 40s ago, saying: 'depends if you have an answer before you dial the sip phone'.
10:43.22DarkFlibblea while ago...
10:50.09dmzah, any info on sip setup? and same for iax setup time?
10:53.47dmzi'll try again tomorrow, need sleep :) (i tried changing to SIP/...sip.fwdnet.net and it said still conjested)
10:54.35RoyKshit. 10 hours of asterisk debug logging == 2GB
10:54.57dmzouch
11:04.56*** join/#asterisk ToTo (n=ToTo@host221-49.pool870.interbusiness.it)
11:06.58*** join/#asterisk raybo (n=raybo@211-11-156-45.withe.ne.jp)
11:07.25*** join/#asterisk pepzi_ (i=pepzi@hd5e25643.gavlegardarna.gavle.to)
11:07.31raybohello
11:07.33*** join/#asterisk pa (n=Paolo@unaffiliated/pa)
11:07.53rayboanyone have any experience with Japanese J1 circuits?
11:09.17rayboI am trying to get a J1 PRI connected and working, but I seem to be having an issue with signaling
11:11.18pepzi_I'm trying to get rxfax working, when I dial (from my normal sipphone) an internal extension that is answering and executing rxfax, I hear fax tones.. but when I try to use rxfax in my from-pstn context and call my pstn-number, asterisk picks up the call, and executes rxfax, but no signals are heard.. oh, and I'm using alaw
11:12.17*** join/#asterisk CPC (n=cleyvers@201.29.70.152)
11:14.41CPCwhat should I install before install asterisk?
11:21.44*** join/#asterisk nextime (n=nextime@213-140-6-103.ip.fastwebnet.it)
11:21.45RoyKCPC: install linux
11:22.20RoyKlol
11:23.44*** join/#asterisk CPC (n=cleyvers@201.29.70.152)
11:24.31raybosorry pepzi_ never used rxfax before. Have you tried ulaw?
11:24.38CPCok but...what libs may I have to install?
11:24.55raybolibpri
11:25.10CPCjust it?
11:25.15pepzi_raybo: yes.. and what is most strange is that it worked a couple of months ago before I upgraded to * 1.2.1 and forgot about rxfax until now
11:25.35CPCwhat about ncurses, ncurses-devel, openssl, etc???
11:25.42raybowell there are three things; first libpri, second zaptel and last asterisk
11:26.02rayboall that was installed at the time I installed Slackware 10.2
11:26.09CPCwhat does libpri means?
11:26.23raybopepzi_, I may have an idea for you then
11:26.43raybolook at the asterisk.org web site it is on the right side
11:27.11raybopepzi_, I have had issues like yours when upgrading before
11:28.27CPCthx
11:29.46raybopepzi_, when i have had issues before I removed all the files in /usr/lib/asterisk/modules then did a make install in libpri, zaptel and asterisk
11:30.17rayboseems some of them do not get overwritten when you are upgrading
11:31.29pepzi_raybo: i'm only using SIP and IAX, but I might as well go through modules :) thanks
11:32.36raybojust a thought as I have hit that wall twice now
11:32.47pepzi_:)
11:36.10*** join/#asterisk o1y1a8r4l (i=x-files@x-files.lv)
11:38.34*** join/#asterisk lahaine (n=lahaine@21.68.119-80.rev.gaoland.net)
11:39.54*** part/#asterisk tonico (n=tonico@chello213047065175.12.14.univie.teleweb.at)
11:43.57*** join/#asterisk shekhar (n=shekhar@221-128-139-79.exatt.net)
11:44.26CPCshould I install anything special to conecct 2 * servers using IAX??
11:45.19rayboIf they are on RFC1918 address space then no, but if they are on public then you should generate public keys and use those.
11:46.07CPCwhat is RFC1918?
11:46.21CPCthey are in the same net
11:46.34rayboit is the RFC that describes private ip address space
11:46.58raybosometimes called "fake" or "private" or "non-routable"
11:47.09CPCyeap
11:47.23CPCwe are using private IP
11:47.34CPCstatic IP
11:47.42Guggemandin other words, if they are reachable from the public net, use keys
11:47.59rayboyes
11:48.09CPCok
11:48.31CPCthean I dont have to do anythig special...Am I right?
11:48.55CPCi have just to configure iax.conf and extensions.conf...all roght?
11:50.14rayboyes
11:50.28raybomy iax.conf for one side looks like this:
11:50.31raybo[asterisk05]
11:50.31raybotype=user
11:50.31raybouser=asterisk01
11:50.31raybohost=192.168.202.6
11:50.31rayboauth=rsa
11:50.32rayboinkeys=asterisk-05
11:50.34raybooutkey=asterisk-01
11:50.36raybocontext=office5
11:50.38raybosendani=yes
11:50.40rayboqualify=yes
11:50.42raybocallerid="2024561414"
11:50.44raybotrunk=yes
11:50.46raybopermit=192.168.202.0/255.255.255.0
11:50.53raybothe keys are stored in /var/lib/asterisk/keys/
11:51.06RoyK~pb
11:51.12jbotmethinks pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca
11:51.32raybosorry
11:53.38CPCraybo, thanks
11:57.01CPChow to see what is send throght pb?
12:01.33*** join/#asterisk nibbler_ (n=nibbler@some.host.name)
12:01.42nibbler_hi
12:01.48DarkFlibblebasicly they paste  into pastebin and post a link to the entry...
12:01.58DarkFlibblethen people visit it and comment...
12:02.08DarkFlibblecuts down on the spam in channel...
12:02.13DarkFlibblenibbler_, hi
12:03.36nibbler_so... how do i tell zaptel (using a hfc card) what number to send?
12:03.40DarkFlibblenibbler_, where you based?
12:03.45nibbler_germany.
12:03.50nibbler_dtag.
12:05.54DarkFlibbleto be honest I have little experience with zaptel... completely ip here
12:08.51DarkFlibbleI worked out that for £9.99 (line rental on an unbundled analogue line) I could make an awful large amount of phone calls if I just used a equivilent DID and spend the rental on minutes instead
12:09.33nibbler_uhuhm ;) but in .de you have to subscribe for an isdn contract too if you want adsl
12:09.37DarkFlibblealso I have 3 mobile phones for emergency services
12:09.37nibbler_suckers...
12:10.05DarkFlibbleisdn and adsl?  its analogue lines and adsl in the UK
12:10.24DarkFlibblebut I'm on cable.... so no adsl required
12:10.58DarkFlibble90% of lines in the UK are analogue...
12:11.06*** join/#asterisk kio (n=kio@ool-4577adba.dyn.optonline.net)
12:11.35DarkFlibblegenerally the only people with ISDN are the people who needed fast internet before broadband... or businesses that can't get broadband...
12:11.35nibbler_cable here sucks - the tariff includes 5gb (they even call it 'fair-flatrate') for ~30eur/mon, the additional gb costs arround 10eur orso
12:12.14DarkFlibblehmmm... 3Mbit here with 75Gig/mo  but they don't complain if you go over
12:12.28DarkFlibblefor £35/mo
12:12.31DarkFlibbleiirc
12:14.12*** join/#asterisk [Airwolf] (n=airwolf@82-171-75-4.dsl.ip.tiscali.nl)
12:14.31DarkFlibbleI suppose that goes to show the difference between the EU and UK markets
12:14.33kioor people using teleconferencing alot
12:15.14*** join/#asterisk sulex (n=sulex@pdpc/supporter/active/sulex)
12:15.20nibbler_uhuhm. at least in germany isdn is massively pushed
12:15.25*** join/#asterisk X-Files (i=x-files@x-files.lv)
12:15.43DarkFlibbleit makes sense to get rid of analogue lines where possible....
12:15.55nibbler_they used to give you ~300eur off if you bought some smaller size pbx together with your isdn contract
12:15.59CPChow is the business to people who now asterisk?
12:16.00kiodon't remember but isdn is 128kb up / down with 1 voice?
12:16.08DarkFlibblebut unfortunately many of the cables in the UK are over 30 years old...
12:16.33DarkFlibblekio, ISDN == 64kbit/channel
12:16.52nibbler_kio: isdn is 3 channels, 2 channels of 64k (duplex) for voice or data (b channels), and one 9,6k channel (d-channel) for signalling data
12:17.02kioya
12:17.03DarkFlibbleBRI (domestic ISDN) is 2 B channels + a 16Kbit D channel
12:17.11nibbler_or 16k yup
12:17.17nibbler_9,6 is incorrect
12:17.48kiowe at the office still maintain 1 isdn for conferencing, weird
12:17.54DarkFlibblePRI is 23? B channels (US) or 30 B channels EU + a 64kbit D channel
12:17.57CPChow many active channel may i have with an high speed connection..like 600 Kbps??
12:18.06nibbler_dark: correct
12:18.25DarkFlibbleCPC, depends on the codecs you use...
12:18.32nibbler_CPC: there exists mlppp (multilink ppp) which allows you to bond $number of channels
12:18.41CPCwhat is the most commom used?
12:18.41DarkFlibbleCPC, also you need the same bandwidth in both directions
12:18.42nibbler_where $number is from 2-infinite ;)
12:18.48nibbler_CPC: two.
12:19.07CPCwow god, just 2??
12:19.07nibbler_since bri (basic rate interface) features 2 b channels.
12:19.44DarkFlibbleits rare that you bond more than 1 R
12:19.54DarkFlibble1 PRI worth of B channels
12:20.02nibbler_for more than two you would need pri (primary rate interfaces) which are usually more expensive than adsl/sdsl or classic e1 (where technicially pri is structured e1 methinks)
12:20.37DarkFlibbleISDN is dedicated bandwidth tho...
12:20.55DarkFlibbleADSL/Cable is normally shared between everyone and his dog in your area
12:21.10nibbler_that is incorrect (technicially)
12:21.25nibbler_cable is shared since it's one shared medium, yes
12:21.27DarkFlibbletechnically yes...
12:22.13DarkFlibblethere is normally contention at the switch for ADSL with each loop being dedicated...
12:22.34DarkFlibbleand cable is shared broadband signalled channel...
12:22.41DarkFlibblebut the effect is the same...
12:22.42nibbler_but adsl is a dedicated connection to a dslam and there's no reason why a telco shouldn't use the bandwith he needs to fulfill all customers at a dslam, stm-16 linecards are cheap
12:23.06Guggemandive never tried not getting my full bandwith on my adsl
12:23.47nibbler_and more than >2gbit/s on a single dslam is unusual ;)
12:24.50DarkFlibblebut I'm an IP guy... analogue makes me feel dirty...
12:24.53DarkFlibble:P
12:25.00nibbler_uhuhm ,)
12:25.25*** join/#asterisk Cleyverson (n=cleyvers@201.29.70.152)
12:25.29nibbler_connecting my parents' fax to the hicom pbx here ;)
12:25.57DarkFlibblenever really been exposed to that large an amount of telco equipment...
12:26.19DarkFlibblegive me IP switches and routers anyday
12:27.17*** join/#asterisk zotz (n=zotz@24.231.47.175)
12:28.36Cleyversonhey guys how is the market place to people who know linux and asterisk??
12:29.03DarkFlibblenot bad... I've got 3 things in the last 4 days from hanging out here...
12:29.04DarkFlibble:P
12:29.05zoanibbler_: do you know of any home built dslams?
12:29.15nibbler_zoa: look at ebay ;)
12:29.29zoai'm interested in making one from scratch :)
12:30.07nibbler_zoa: that's some work in deed. unless you want to do something like a sdsl to sdsl connection - that can be easily acomplished with 2 sdsl modems
12:30.26nibbler_Cleyverson: here's some job: earn $5 in cash for making my caller-id with zaptel work ;) *scnr*
12:30.59Cleyversonlol
12:31.34Cleyversonwow my jesus, i'm gonna be rich.. :)
12:31.58*** join/#asterisk bofh42 (n=bofh42@p5482B8D5.dip0.t-ipconnect.de)
12:32.02zoanibbler, how is that
12:32.09zoacan you do that on normal telephone lines ?
12:32.30nibbler_zoa: on an ordinary double-copper-wire line, yes.
12:32.42zoadoes the telco need to change something for that ?
12:32.47zoagot an url for such modems ?
12:32.54X-Filesppls,why EYEBEAM 1.1.3010n not work normal in ASTERISK ??? (not support Messager, Online Status Contacts)
12:33.16zoabecause asterisk doesnt support all that
12:33.19nibbler_the telco just needs to _directly_ connect 'your' 2 copper wires with the 2 copper wires of the person you want to exchange data with
12:33.27nibbler_that has nothing to do with asterisk
12:33.46zoa<zoa> because asterisk doesnt support all that -> was for x-files
12:33.49nibbler_asterisk is a pbx - what i'm talking about is two copper wires - no more, no less
12:33.57*** join/#asterisk apardo (n=apardo@62.97.121.95)
12:33.58zoai know i know
12:34.03nibbler_ok ;)
12:34.05zoajust said that for xfiles
12:34.11nibbler_ah, i see.
12:34.13zoai'd love to try thatr
12:34.53nibbler_i think the devices i've seen in such a setup were efficient networks speedstream sdsl modems
12:35.01*** join/#asterisk zeedo (n=zeedo@80-192-53-14.stb.ubr04.glen.blueyonder.co.uk)
12:35.26nibbler_if you get used ones from the german a/sdsl operator qsc the password is "pritt-stift" (in some variation i think)
12:35.46*** join/#asterisk wizard545 (n=wizard@tor/session/x-62ddc9b24cbcd14b)
12:35.57zoabut for sdsl you need a leased line ?
12:36.22nibbler_you need a directly connected copper wire.
12:36.25zoawhat are the chances the belgian telco would block us from using my own sdsl to sdsl connection ?
12:36.35nibbler_95%? ;)
12:36.42zoa:)
12:37.26nibbler_but as soon as you send some higher voltage (pulsing is a good idea) into the line their coils will go bananas and they'll eventually remove them ;)
12:37.38nibbler_that at least is the default procedure here in .de
12:38.12X-Fileszoa: http://www.voip-info.org/wiki/view/VOIP+Phones  , Xten eyebeam SIP video soft phone; with SUBSCRIBE / NOTIFY support (BLF)
12:38.23X-Fileszoa: i see supported !
12:39.11*** join/#asterisk pa (n=Paolo@unaffiliated/pa)
12:39.35*** join/#asterisk areski (n=areski@24.Red-83-55-100.dynamicIP.rima-tde.net)
12:39.46zoahey ho areski
12:39.57areskiHello Jo
12:40.04areskizoa, how r u doing ?
12:40.24zoafine, you ?
12:40.33areskiI wanna cure my hangover
12:40.47areskiv been drinking too much yesterday
12:42.33zoahaha
12:43.02GuggemandX-Files eyebeam supports video yes, but not when connected to asterisk
12:43.21zoai think it supports video to asterisk too
12:43.23zoakind of
12:43.25zoasome things
12:43.38Guggemandokay, not messenger and online status then :)
12:43.39areskizoa, do u know if there an other astricon in europe this year?
12:43.46X-FilesGuggemand: video supported in asterisk.
12:43.51zoain oktober i think
12:43.52X-Files:)
12:44.01wizard545any providers out there that have 8xx did's?
12:44.36areskiGuggemand, I ve been using eyebeam for awhile
12:44.47areskiand yes it s support video with asterisk
12:44.54h3xyou mean 8YY
12:45.06h3x8XX also includes area codes that start with 8 :)
12:45.06wizard545800 - 866 - 877
12:45.15X-Filesareski: messages and contacts status worked you ?
12:45.15wizard545yea..
12:45.40Guggemandareski, i kinda got that when the 2 other people told me too :P
12:45.46areskiI never need them
12:46.34wizard545areski are you part of the a2billing project?
12:46.38X-Files;<
12:46.57areskiwilymage, I am the only one working on this :D
12:47.26areskiwilymage, are u using it ?
12:47.37*** join/#asterisk oej (n=oej@83.210.106.6)
12:47.46wizard545you mean me?
12:47.50wizard545or wilymage
12:47.52areskiyes
12:48.00wizard545I am using a2billing
12:48.03wizard545just got it all setup
12:48.08wizard545you the developer?
12:48.11areskiyup
12:48.25wizard545nice software man
12:48.25wizard545really like it
12:48.26areskithx u :D
12:48.34areskinext release coming soon
12:48.41wizard545what new features?
12:48.58areskiit would be a surprise :)
12:49.09wizard545haha
12:49.12areskigot plain of funny stuff
12:49.57DarkFlibblewizard545, what DIDs in that area did you want?
12:50.03DarkFlibblejust toll-free?
12:50.07wizard545yea
12:50.31DarkFlibblenufone does them for $2.50/mo + 2c/min incoming...
12:50.41DarkFlibblebut there services isn't the best in the world
12:51.25DarkFlibblethe website has been 'in progress' for 2 years with little change... and their support can take a couple of days to get back to you...
12:51.49DarkFlibblebut still... if they are working they keep working...
12:51.54wizard545DarkFlibble, ok, that's not a bad cost though
12:52.01DarkFlibbleI know..
12:52.06*** join/#asterisk Shoragan (n=shoragan@d072.apm.etc.tu-bs.de)
12:52.12DarkFlibblegot 4 myself with them
12:52.33DarkFlibbleso noone accuse me of bias
12:52.35DarkFlibble:P
12:52.45nibbler_is there a capi support for asterisk 1.2.1 and later?
12:53.32wizard545DarkFlibble, so it's only 2/c nothing else? that's almost the same as a regular DID
12:53.56DarkFlibble2c incoming.. + monthly rental...
12:54.04DarkFlibblepretty similar to most other DIDs
12:54.17wizard545nice
12:54.21DarkFlibbleI can show my records if you need
12:54.33*** join/#asterisk tuxinator_linux (n=tuxinato@70-32-106-248.ontrca.adelphia.net)
12:54.38wizard545naw, but for a calling card business, it would be how much outgoing?
12:55.21DarkFlibbleoutgoing doesn't need to be the same as incoming...
12:55.40wizard545what kind of per minute can I get for outgoing?
12:55.47wizard545cheapest
12:56.03DarkFlibbledepends on destination
12:56.15wizard545USA 48 only
12:57.12DarkFlibblenufone are fairly competitive... but there are a *lot* of alternatives... it depends on the number of minutes you push to what pricing is doable
12:57.33wizard545ahh ok
12:58.28DarkFlibblepointless me quoting since I'm EU based
12:58.47wizard545ahh ok
13:01.30wizard545DarkFibble is nufone iax or sip
13:04.58*** join/#asterisk Naturalblue (n=Kay@195.26.12.229)
13:06.39*** join/#asterisk cyber (n=kani@220.247.250.71)
13:07.53cyberalo guys
13:12.25*** join/#asterisk ivanfm (n=ivanfm@201-27-21-115.dsl.telesp.net.br)
13:14.33*** join/#asterisk RoyK (n=roy@g-001.osl255.netcom.no)
13:25.08wizard545would broadvoice catch a high outgoing volume?
13:26.45cyberwizard it depends on the codec and the broadband speed
13:27.21wizard545which codec would be best
13:28.17cyber723 uses less bandwidth
13:28.24cyber729 has good quality
13:28.36*** join/#asterisk pigpen2 (n=mark@fw.seamans.cc)
13:28.38*** join/#asterisk bmg505 (n=leon@dsl-146-0-178.telkomadsl.co.za)
13:28.42wizard545gotcha
13:29.09*** join/#asterisk mmg1818 (i=mmg1818@86.55.238.46)
13:29.48DarkFlibblewizard545, they do iax
13:30.27DarkFlibblebroadvoice aren't good at detecting people abusing them... (at least from what I have seen and read)
13:30.55RoyKeer.r
13:30.56RoyK15 active channels
13:30.56RoyK49 active calls
13:33.06*** join/#asterisk Paolo (n=paolo@217.220.155.234)
13:33.14*** join/#asterisk Trazz (i=Trazz@c-67-163-92-37.hsd1.il.comcast.net)
13:34.45wizard545DarkFibble, time to signup a couple unlimited accounts
13:34.58wizard545anyone know how many concurrent calls they allow?
13:35.22*** part/#asterisk Paolo (n=paolo@217.220.155.234)
13:35.34cyberwith unlimited account only 1
13:35.54wizard545ahh ok
13:36.45RoyK29 packets transmitted, 20 packets received, 31% packet loss
13:36.45RoyKround-trip min/avg/max/stddev = 763.161/2128.823/6189.705/1375.203 ms
13:36.50RoyKnice for VoIP
13:38.37*** join/#asterisk riksta (n=rick@62.6.163.81)
13:42.10*** join/#asterisk zigman (i=zigman@irc.zigman.de)
13:43.25*** join/#asterisk af_ (n=af@ip-142-84.sn1.eutelia.it)
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13:49.03*** part/#asterisk Cleyverson (n=cleyvers@201.29.159.216)
13:52.26*** join/#asterisk apardo (n=apardo@87.218.44.151)
13:55.28*** join/#asterisk [TK]D-Fender (n=joe@66.11.164.239)
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14:04.19*** join/#asterisk razu (n=razu@80-235-90-19-dsl.prn.estpak.ee)
14:05.21*** join/#asterisk pepsis (n=pepsi@216.58.95.246)
14:07.10*** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl)
14:07.15*** join/#asterisk mountie (n=mountie@CPEdeaddeaddead-CM000a739acaa4.cpe.net.cable.rogers.com)
14:09.33lesouvageCan somebody point me to a list with the most commenly used voiceprompt or a list with the voiceprompt related to applications.
14:10.13De_Monwhats a voiceprompt?
14:11.27lesouvageDe_Mon:: a .gsm file played by asterisk, like "Something is terribly wrong, goodby"
14:11.38*** join/#asterisk nextime (n=nextime@213-140-6-103.ip.fastwebnet.it)
14:12.31lesouvageI wrote a routine to record them yourselve but adding almost a thousand voiceprompts makes it kind of unusable and worthless.
14:13.17[TK]D-Fenderlesouvage : You want a list of promts that other people might use that we don't already have?
14:14.28lesouvageD-Fender: No, I'm looking for a file of the most commonly used English voiceprompts so I can translate them and made myself, with the routine, a Dutch version..
14:14.57[TK]D-Fenderlesouvage : How about just taking the ones * already uses and doing them?
14:15.56zoalesouvage: a dutch version already exists
14:16.20lesouvageD-Fender: that's what I want but it would be handy to have them written out in a .txt or .doc file. I have a list but it has around a thousand voiceprompt and that is much more then usefull.
14:16.21zoahttp://www.asteriskguru.com/board/nederlandstalige-prompts-voor-asterisk-vt81.html
14:19.22lesouvagezoa: I kknow about the Tric vocepromts set, it's nice and of good quality  but it's not complete. Somethimes an English phrase shows up.And I think it's nice to have an easy way to make all the voiceprompt (standard and custimized) with the same voice. But thanks for the link..
14:20.12pepsisi have a * box with 3 X100P cards (not clones).  I can only get 1 PSTN line to work for outbound.  can someone point me to docs on how this should work?
14:21.40[TK]D-Fenderpepsis : Do you see the cards in in ztcfg?
14:21.50lesouvagepepsis: be sure that they all have a interrupt of there own. Your zapata.conf should have a line something like channels => 1 - 3.
14:21.54[TK]D-Fenderand in cat /proc/interrupts?
14:22.06X-Files[TK]D-Fender: hello
14:23.32pepsis<PROTECTED>
14:23.48pepsisthey all share 1 interrupt it appears
14:24.02wizard545can you setup asterisk to work wioth skype? so my asterisk can call skype people
14:24.17lesouvagepepsis: at the and of the [channel] part. "channels" is wrong this should be "channel"
14:25.04lesouvagepepsis: wth 3 x100p cards on 1 interrupt it is not going to work.
14:25.18pepsisis there a way to force them to use different interrupts?
14:25.20[TK]D-Fenderpepsis : Ok, sharing interrupts is bad, but lets see if its something else.  Show us your dial line for outbound zap calls, and your zapata.conf
14:25.23[TK]D-Fenderuse pastebin!
14:25.24[TK]D-Fender~pb
14:25.32jbotfrom memory, pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca
14:25.42[TK]D-Fenderpepsis : You need to try and do that in your motherboard BIOS
14:26.07pepsisi am using the "Asterisk@Home 2.2" distribution
14:26.15lesouvageD-finder: I just finished the bios line:)
14:26.52pepsisi think im gonna have to change the system its in then.  i was hoping to use a compact machine in a nice SFF case, but don't have t.
14:26.54pepsisdon't have to.
14:27.14*** join/#asterisk pigpen2 (n=mark@fw.seamans.cc)
14:27.15lesouvagepepsis: do you have other cards in your pc that you don't need for an asterisk box. You should remove this cards.
14:27.33h3xhow did you find a sff case with 3 pci slots
14:27.55pepsisim magical.  :)  (link coming9
14:28.17pepsishttp://www.andovercg.com/ebay/images/bor-0119.jpg
14:28.36h3xheh
14:28.42h3xare you trying to use this with a via epia mobo
14:28.44h3xasterisk that is
14:28.54pepsishttp://pastebin.ca/37811
14:29.09pepsisno, its a compaq d510 with an 815 chipset
14:29.10*** join/#asterisk QbY (n=QbY@adsl-068-209-210-253.sip.cha.bellsouth.net)
14:29.13h3xoh
14:29.22h3xno wonder. thats worse
14:29.22h3xheh
14:29.29h3xf'n compuke
14:29.37QbYare there any IAX2 hardphones that are worth recommending??
14:29.50h3xno
14:29.51pepsisnormally i agree...but its not the presario crap, its the deskpro en series.
14:30.02h3xdude.  its compaq.
14:30.03*** join/#asterisk A-jay (n=quirc@62.217.245.194)
14:30.12pepsish3x, use one.  :)
14:30.19zoaQbY: yes
14:30.20h3xduuuude.. its still compaq.
14:30.22zoathe st 302
14:30.24zoaquite good
14:30.30[TK]D-Fenderpepsis : Ok, using AMP.. *bad*.  We need the "included" flles in there too... and your DIAL line <-
14:30.33zoabetter than the gxp2000
14:30.48QbYzoa..  you have a link?
14:31.14zoahttp://www.voipsolutions.be/product_info.php/products_id/206
14:31.21zoai dont know the full name of the phone though
14:31.28zoasomething chinese probably
14:31.31pepsis[TK]D-Fender, weird thing happened, i had 2 of the x100p cards plugged into 2 local POTS lines.  when the phone rang, it hit my extension without any configuration and it passed along the Caller ID too.
14:31.47pepsisso something weird is going on.  i can't seperate the different x100p cards.  its showing them as one card.  :(
14:32.05zoapepsis, give up on AMP
14:32.14[TK]D-Fenderpepsis : Thats hard to say... you're running AMP and not showing us anything USEFUL.
14:32.20zoai tried amp yesterday
14:32.31pepsiswell then.  would i be further to wipe the box and stick a real distribution like sarge on it?
14:32.42pepsisi was hoping to be able to do it all from a browser.  :)
14:32.46zoais a gui really supposed to be "giving a hard way to configure something simple ?"
14:33.08*** join/#asterisk coppice (n=chatzill@204.206.17.210.dyn.pacific.net.hk)
14:33.57[TK]D-Fenderzoa : The ST 302 looks like jsut another craptastic PA168(8) chipset phone....
14:34.04h3xhaha craptastic
14:34.05h3xhahahaha
14:34.09zoait is the same chipset
14:34.13zoabut its not the same junk
14:34.15lesouvagepepsis: you can try xorcom. It's debian, ready for use and with usefull tools in case of problems with configuration.
14:34.15zoawe are quite happy with it
14:34.26h3xsnom rules.
14:34.28zoatrue
14:34.29pepsisgonna give xorcom a shot
14:34.31h3xatacomm sells the 360 for $199 now
14:34.32zoabut this phone is cheap
14:34.37h3x320 is like 179
14:34.39zoaand better than the gxp2000
14:34.45*** part/#asterisk Kryczek (i=kryczek@faked.name)
14:34.47pepsisi figure that once i'm setup and working with *, I will save about $150/month
14:34.51zoathats in the us, here they cost double
14:34.51zoa:)
14:34.54pepsisso its worth the time.  :)
14:35.10zoapepsis, pay me to configure it each month :p
14:36.03QbYzoa..  i can't seem to find a us reseller..  do you know of one?
14:36.15pepsiszoa, each month?  why not do it right the first time.  ;)
14:36.17zoano sorry
14:36.20zoahaha
14:36.26zoadepends maybe you want changes
14:36.48zoafirst step i do for free, which is deleting AMP :p
14:36.57h3xits people like zoa that cause the message "Please listen to the following menu options as our extensions have recently changed"
14:37.03h3xto happen
14:37.13pepsisyea...i hate those messages
14:37.17coppicesoa: who makes the ST320?
14:37.31pepsisshit, if i wasn't listening, how would i know what options were there?  why tell me to do what im doing already
14:37.33h3xwhy dosent anybody make a t.38 fax machine yet.
14:37.42zoahaha
14:37.49h3xethernet and power
14:37.55zoacoppice : i have no clue
14:37.58zoatried to find it
14:37.59*** join/#asterisk ivanfm_ (n=ivanfm@201-1-164-43.dsl.telesp.net.br)
14:38.04zoabut there is nothing written on the phones
14:38.08h3xoh i know why
14:38.08zoait might be siptronic or so
14:38.11h3xcoz nobody sells t.38 service
14:38.19coppiceh3x: I can't imagine why anyone would want to, really
14:38.21QbYST-302 is made by Siptronic
14:38.30h3xcoz
14:38.34zoabut i think siptronic might just be rebranded
14:38.35h3xthen it could be a printserver too for your LAN
14:38.42h3xmultifunction machine
14:39.10zigmanh3x there is no market to t.38 termination right now
14:39.18QbYi need to come up with a way to get two phones into an employees house--i'm looking at the IAXy, is that a good idea..  they'll need stuff like transfer, music on hold, etc..
14:39.22h3xspeaking of, what the dilly with t.38 in iaxphone library
14:39.25zigmanyou would still need the analog line
14:39.29h3xiaxclient i mean
14:39.43[TK]D-FenderQbY : SPA-2002.  $70 and thats it.
14:39.50zigmanallthough a t.38 softfax modem for windows would be great
14:39.54coppicebuy why t.38? t.37 makes more sense. there are a number of vaguely t.37 fax machines, although they don't seem to follow the standard properly
14:40.10h3xwell
14:40.11*** join/#asterisk newbieasterisk (n=junk@59.93.67.211)
14:40.13coppicezigman openh323 has one
14:40.13zoawhy not use email :)
14:40.20h3xi like t.37 better too but
14:40.25coppicethat is what t.37 does
14:40.28zigmancoppice isn't that the same with t.30 ( following the standards) ;)
14:40.37zigmancoppice ?
14:40.43zigmanopenh323 has one for linux
14:40.46zigmanbut not for windows
14:40.54zigmanafaik
14:40.55h3xthere is a softphone for windows that does t.38
14:40.59coppiceyou can't actually follow t.30. its full of holes
14:41.11zigmanh3x which one ?
14:41.38QbY[TK]D-Fender : So are you saying SIP would be better than IAX?  And this device would support two phones behind a router?  I thought IAX would be better because it doesn't require the port forwarding, etc..
14:41.52h3xkapanga
14:41.59h3xkapanga.net
14:42.15h3xerm
14:42.28pepsiscan asterisk do call-backs?  like if i'm in another city, call my *, have it call me back at a specified number?
14:42.33zoawe didnt do any t.38 development in the last 2 months
14:42.39zoapepsis: yes it can
14:42.45coppiceI see a picture of the ST302. I've seen that case before with other things on it
14:42.58[TK]D-FenderQbY : You don't actually need port forwarding.  the SIP keep-alives will keep the link up.  how do you think Vonage used thousands of these type boxes to run their company otherwise?
14:43.00pepsissweet.  make a $0.25 call and have it call me back using vonage.  ;)
14:43.05zoayeah i think they have a zillion names
14:43.10zoapepsis, or even better
14:43.17zoahave it recognize your callerid
14:43.20zoabut not pickup the line
14:43.24zoaand have it call you back
14:43.32pepsisthat would be better
14:43.56QbY[TK]D-Fender - True.  I've just always had problems it seems like with SIP (softphones) behind a router.
14:44.23[TK]D-FenderQbY : You need to set things properly on them, but I do it all the time
14:44.28zigmanh3x thx
14:44.57QbYOne last thing.  What Analog phone would you recommend with this SPA2002--with hold buttons (that have use the moh) and transfer?
14:45.00pepsiszoa: no msg here.  not registered.
14:45.58coppiceits hard to tell who makes a lot of voip stuff. you look at some and thing another company made it, then you find they both make the chipset vendor's reference board without changes
14:46.14[TK]D-FenderQbY : Analog phones don't have those capabilities themselves.
14:46.17zoayes
14:46.18[TK]D-FenderThats the ATA's job
14:46.59coppiceyeah. you tell the ATA what you want by typing morse on the DTMF keys :-)
14:47.14[TK]D-FenderQbY : Sipura's implement transfers / hold / etc through use of hook-flash and * codes.
14:47.43QbYhehe..  but there is no way i can buy a phone and program the "hold" and "Transfer" key?
14:48.33[TK]D-FenderQbY : Possible, but at that rate just put a real IP phone there.
14:49.52QbY[TK]D-Fender -- Excuse the dumb questions.  But if I grab a few SIP phones and a Linksys router, both of those phones could be used simultaneously behind the router to our Asterisk server in another location?
14:50.47[TK]D-FenderQbY : Sure.  Just let the phones and * know they're behind NAT, set them to different ports for signalling, and that should be it.
14:50.59QbYawesome
14:52.16QbYi wish they sold this stuff at office depot or best buy.
14:53.05h3xif they did it would cost $599 for a spa-941 phone
14:53.11[TK]D-Fender:O
14:53.30QbYhehe..  yeah..  i just want to get one and play today..
14:54.19[TK]D-Fenderfrom where I live I could walk to Gentek (who is a big distributer here) and stock up :)
14:54.29QbYeverything seems bundled with CallVantage or Vonage
14:54.46h3xyou can hack the vonage pap2 box
14:55.07QbYh3x.. you have any docs?
14:55.18h3xvoip-info.org has plenty
14:55.27h3xyou must do it with a virgin pap2
14:55.27QbYi'm off to office depot
14:55.37QbYvirgin meaning i just bought it from the store
14:55.38h3xdont plug it in the public internet til you hack it
14:56.02QbYk
14:57.11[TK]D-FenderQbY : Just buy from a normal reseller!
14:57.20[TK]D-FenderQbY : Save yourself the frustration.
14:57.31h3xactually
14:57.43h3xbest buy and friends some times have instant rebates or gift certificates
14:57.52QbYtime is of the essence.. i have to get them running by monday morning
14:57.54h3xi just bought one from best buy coz i have a business rewards card
14:58.03h3xand i got a $25 gift cert back on it
15:00.20[TK]D-FenderQbY : use a softphone and don't leave things to the last minute.
15:00.37QbYk.
15:01.14[TK]D-FenderQbY : Then monday morning place an order for some real phones.
15:05.57|vinsik|what is the best WLAN phone in your experience?
15:06.09zoanot the zyxel
15:06.10zoaBrrr
15:06.12|vinsik|i have tested: Saneo, Suncom, zyxel
15:06.13zoazyxel
15:06.24zoazyxel is something to warm your ears
15:06.34zoaand talk 5 minutes then have an empty battery
15:06.37|vinsik|yeah i got that opinion too.
15:06.41zoaor just leave it on standby for an hour
15:06.45zoaand have an empty battery
15:06.58|vinsik|one day max on my zyxel
15:07.02lesouvagesoundquality of zyxel is great but battery live is kind of short.
15:07.17zoahow is saneo and suncom ?
15:07.34|vinsik|suncom a copy of zyxel but with less efficient processor
15:07.40|vinsik|cheaper :)
15:07.47|vinsik|sanoe is pretty good
15:07.50|vinsik|saneo
15:08.01zoazyxel is good quality if you disable encryption
15:08.02|vinsik|battery life is good.. maximum online was 5 days
15:08.06zoait just cant do the encryption
15:08.06|vinsik|without recharge
15:08.09[TK]D-FenderThe only one I haven't heard was "crap" was the Hitachi WIP500
15:08.10zoagot a url for the saneo ?
15:08.23|vinsik|uhhh
15:08.24zoathe wip500 is better, no ?
15:08.27|vinsik|ill check
15:09.32|vinsik|nope dont have it.
15:09.38|vinsik|it cost about 200e
15:10.15|vinsik|d-fender: u mean to say its good?
15:11.15[TK]D-Fender|vinsik| : No I mean to say that numerous others have NOT said its "crap".
15:11.31|vinsik|ok
15:11.31[TK]D-FenderI don't think there is one out now that should be considered "good"
15:11.33zoaaaah
15:11.46zoagot an url for the wip500 too ?
15:11.51[TK]D-FenderIts just "degress of meaning well"
15:12.02|vinsik|saneo's sound quality was good, and battery life was outstanding...
15:12.10[TK]D-Fenderzoa : www.voipsupply.com
15:12.29|vinsik|but it looks kinda cheap ..
15:12.35*** join/#asterisk Gunnar (n=gunnar@34.80-203-209.nextgentel.com)
15:13.17zoaits a wip5000 i think
15:14.55lesouvageHas anybody tried the zultys wifi phone?
15:20.45tronixany recommendations on a couple of decent SIP or IAX2 soft phones for Linux? i've got xtensoftphone and twinkle; any others?
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15:24.55X-Filesneed eyebeam 3010z
15:25.00tronixheh. slow day, I guess. I'll keep poking 'round. thanks.
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15:45.35NewSoleanyone know why keep getting chan_sip.c:1208 retrans_pkt: Maximum retries exceeded on transmission
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16:33.24tronixI've got xtensoftphone and twinkle. they can call each other ok, but xten is doing gsm and twinkle is doing ulaw
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16:33.38tronixI presume that's a problem? How do I make asterisk tell
16:33.44tronixboth to do gsm or do ulaw?
16:34.14JMcAdisallow=all
16:34.18JMcAallow=ulaw
16:34.41JMcAthat limits you to *only* using ulaw
16:34.52h3xor just configure your client right
16:34.55sm7xabHi! Anyone except me who is having problem with sound playback from asterisk on Gentoo? messages from the voicemail application are so bad that you can't hear what's beeing said...
16:34.55JMcAthat too
16:35.14sm7xabI'm running Asterisk 1.0.10.
16:35.54sm7xabHave the same problem on two machines. Don?t think it's hardware related...
16:36.04tronixcool, thanks jmca/h3x
16:36.27tronixsm7xab: i'm running * on Gentoo, too, but unfortunately, i'm deaf so i'm useless with sound issues. :) sorry. :(
16:36.52sm7xabtronix: Oh! Too bad!
16:37.04tronixheh I'm just trying to get my TDD working with *
16:37.12tronix(45.5 bps baudot, pulse dialing, etc)
16:37.33tronixit's an interesting challenge.
16:37.41brookshiretronix: sarahem does a lot of work on that
16:37.45X-Filesppls,why EYEBEAM 1.1.3010n not work normal in ASTERISK ??? (not support Messager, Online Status Contacts)
16:37.58brookshirei wonder if i have her contact
16:38.16JMcAI suspect the biggest difficulty is the use of baudot vs. ascii
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16:38.18tronixjmca: they could. a friend wrote mac and windows programs to do TDD calling; main prereq was a modem w/dsp that could do a/d conversions -- not many of these modems around
16:38.19DarkFlibblesm7xab, I saw a similar issue on a machine yesterday... putting a wait(1) before voicemailmain seemed to cure it
16:38.27tronixjmca: but a winmodem, being a dsp, should be able to do the job
16:38.40sm7xabDarkhalf: I'll try that!
16:38.50tronixbrookshir: oh? intereseting. thanks!!
16:38.52JMcAtronix: right...the old lucent winmodems had dsps...they shoulda been able to do it...but noone ever wrote the code to do it  :/
16:39.10tronixjmca: indeed. :( ah, well, that'd make for a good weekend project too ;)
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16:40.01sm7xabDarkFlibble: Can't be the problem here. have a wait 2 before VoicemailMain()
16:40.16DarkFlibblehmmm...
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16:41.02DarkFlibblewhats the bandwidth like? also is the harddrive running at full throttle?
16:42.10sm7xabDarkFlibble: BW = 100Mbit FullDup. HD running att 100% on a Promise card. Same card worked wonders on old machine running * without any problem.
16:42.42sm7xabDarkFlibble: * running at "nice -15" with -p option.
16:42.50DarkFlibblehmmm...
16:43.03DarkFlibblehow are you connecting to asterisk?
16:43.24*** join/#asterisk cpm (n=Chip@pdpc/supporter/sustaining/cpm)
16:43.24*** join/#asterisk cpm_ (n=Chip@dns1.eruditium.org)
16:43.35DarkFlibbleand if you grab the messages via email or direct from the file system are they recorded properly?
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16:44.04sm7xabDarkhalf: Using a Vood VTA 111 PAP box. Also using Linksys PAP2 bax for connection. Normal speak phone ot phone work without a hitch.
16:44.21*** part/#asterisk cpm (n=Chip@pdpc/supporter/sustaining/cpm)
16:44.31DarkFlibblehmmm... speaker phone.. have you tried it on a normal handset?
16:44.47sm7xabDarkFlibble: Problem is that noone can record a message because the "please leave a message after the beep" is not decipherable by the llistener.
16:45.23DarkFlibblewhat about quality of other prompts and such?
16:45.34DarkFlibbleis it limited to voice mail?
16:46.02sm7xabDarkFlibble: No prompts are decodable by ear. Same goes for Music on hold.
16:46.37DarkFlibbleand normal calls, with and without reinvites?
16:46.43NewSoleanyone know why keep getting chan_sip.c:1208 retrans_pkt: Maximum retries exceeded on transmission
16:46.59NewSoleon every call that is
16:47.08sm7xabDarkFlibble: Normal calls work OK. Reinvite is something I don't know.
16:47.41sm7xabDarkFlibble: sip debug on, set verbose 4 = no indication of problem. Logfile is nice and empty.
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16:48.33DarkFlibblemake a call and see if the codecs are the same... if not then there must be transcoding on the asterisk box... if they are then they *may* have the audio via the asterisk box or they may reinvite and send data direct...
16:49.04wizard545anyone use sip broker?
16:49.19DarkFlibblewizard545, I was looking at it earlier... but not yet
16:49.44wizard545DarkFlibble, correct me if i'm wrong, it allows free incoming?
16:49.53wizard545People who have a PSTN phone (or a DID) can map that phone number to a SIP URI (for free at e164.org)
16:49.54sm7xabDarkFlibble: The only thing the two machines have in common is the processor type. P3.
16:50.19DarkFlibblesm7xab, I'm just trying to work out the boundry conditions of the problem...
16:50.31DarkFlibblewe can narrow it down then...
16:51.05DarkFlibblewizard545, it is more of a way to switch between many different networks... a central peering point... at least at a cursory glance
16:51.07sm7xabDarkFlibble: I understand. That's why I add some information. I feel a bit frustrated about the problem...
16:51.37DarkFlibblesm7xab, I have had p3s running asterisk fine...
16:51.53DarkFlibbleso the processor is not likely to be a big problme
16:51.59wizard545oh i get it...
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16:53.40DarkFlibblewizard545, I need to maintain links to 5 seperate networks at present due to peering... a single point (of failure) might be an alternative...
16:53.42sm7xabDarkFlibble: I've set up several of them and all were working OK. These two last are stupid :-(
16:53.44DarkFlibble:P
16:53.54DarkFlibblesm7xab, what distro?
16:54.08sm7xabDarkFlibble: Gentoo.
16:54.12DarkFlibblewizard545, dundi is more tempting atm tho
16:54.21DarkFlibblesm7xab, hmmm...
16:54.27wizard545yea..
16:54.37DarkFlibblesm7xab, I don't know to be honest...
16:55.29sm7xabDarkFlibble: How stable is * 1.2.0 by now? Thinking of trying that one instead. Also thinking of recompiling my kernel to become preemptive.
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16:55.58DarkFlibblesm7xab, I haven't had any problems in the last few days, besides pebkac issues... :P
16:57.17sm7xabDarkFlibble: Last working version I had was 1.0.9. Think I'll tryout 1.2.0 instead. Thank's for the help!
16:57.27DarkFlibblenp
16:57.46DarkFlibbleits hard to debug when you can't test stuff yourself
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17:04.24*** join/#asterisk QbY (n=QbY@adsl-068-209-210-253.sip.cha.bellsouth.net)
17:05.09QbYSo I went to Office Depot, got a PAP2, Unlocked it.  Excellent.  One problem: My phone doesn't ring.  The caller ID shows the call coming in, and I can answer it, but it doesn't Ring...  Anyone ever experienced this?
17:05.57QbYhehe
17:06.18tronixwhich's a better codec -- ulaw or gsm?
17:06.31tronixI understand that g729 is better than both but licensed.
17:06.32*** part/#asterisk JaredBluestein (n=Jared@nwlnnhbas01-pool3-a18.nwlnnh.tds.net)
17:06.45tronixulaw seems more popular, tho?
17:06.50tronix(ulaw vs gsm)
17:11.30eKo1I don't recommend ulaw if you have a bad connection
17:11.30tronixfor fax traffic, i wouldn't use lossy like gsm but curious which folks likes better for voice traffic
17:11.39tronixinteresting. thanks!
17:11.51eKo1i'd use ulaw for everything except that i have bandwidth constraints
17:11.57tronixahh makes sense.
17:12.29tronixi've got decent broadband (currently a T-1 but soon to go with an upload-limited cable modem)
17:13.00JMcAI've got 256kbps upload, and use ulaw
17:13.23tronixhmmm. think mine would be 40kbps upload or round there. guess I should look at a
17:13.30tronixbusiness cable modem plan to get 256. :)
17:14.06JMcA40kbps?  that's modem speed
17:14.12tronixheh aye
17:14.12JMcAballpark
17:14.16JMcAthat bites
17:14.18tronixnot much competition in town
17:14.27JMcAgood lord
17:14.32tronixonly two places. dsl by local ILEC telco or roadrunner cable modem
17:14.36wizard545tronix, you should host your asterisk box at our datacenter
17:14.51tronixand the ILEC is fibbing when they say I don't qualify for DSL... I live down the road from the CO
17:14.58tronix(and already have existing T-1..geez..)
17:15.11tronixwizard545: hmmm. where is it?
17:15.18wizard545USA ohio
17:15.21tronixnot bad
17:15.22wizard545100meg connection
17:15.44tronixwhich part of ohio? akron/cleveland or closer to cincy? columbus?
17:15.47JMcAtronix: just 'cause you're close to the CO doesn't mean you qualify
17:15.53wizard545columbus
17:15.57tronixjmca: that's true
17:16.06wizard545downtown columbus
17:16.20tronixI can't imagine why I wouldn't qualify, 700 ft away, tho
17:16.27tronixor given that I have existing T-1 service.
17:17.01tronixthe irony is that I used to work for this telco :) and knew the switch techs and could go into the switch room if I needed something. not anymore, tho.
17:17.07[TK]D-Fendertronix : ULAY is better than pretty much everything except wide-band.  GSm is MUCH worse, and 729 is a bit better
17:17.20tronixwizard545: thanks -- got an url for colo'ing info?
17:17.55tronixthanks!
17:18.09tronix[TK]D-Fen: interesting. *takes notes*
17:18.13wizard545no problem
17:18.40tronixjmca: what annoys me is that only a few months ago, they were telling us we qualified. I wonder if they oversubbed or something.
17:19.45tronixoh well, not a huge deal. roadrunner's ok, and they do have that business plan.
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17:27.00wizard545nufone seems scaru
17:27.02wizard545scary
17:27.03wizard545haha
17:27.06wizard545anyone use them?
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17:28.57tronixi do but haven't set it up yet
17:29.04tronix(just signed up)
17:29.14tronixan existing * user I know in Canada highly recommended them
17:29.22tronixhe said no-frills but techy people
17:29.50tronixthey were pretty quick with signups and DID provisioning (on the spot)
17:29.58tronixand few mins turnaround on support queries
17:30.23tronixthey also provide sample * configs to plug in, specific to your provisioned setup
17:30.27RoyKwizard545: jerjer is scary...
17:31.03tronixrate's not bad. 2 cents/min for U.S. domestic calls; a little more to Canada and something like 10-12 cents/min to places like Russia? (vague recollection)
17:31.16[TK]D-Fendertronix : Where are you located?
17:31.20tronixwestern NY
17:31.32Math`tronix: use voipjet for termination
17:31.44tronixhmmm. i'll have to check that out, thanks.
17:33.14wizard545tronix 2/c a min was 800 inbound right?
17:33.37tronixbelieve so
17:33.54wizard545not bad for 800
17:34.59JMcAso...not that I have any need to do this yet, but can you do  SetVar( ${_foo} = ${foo} )  and have it do what's expected?
17:35.25*** join/#asterisk ptblank (n=MURDER1@68-169-161-61.lmdaca.adelphia.net)
17:36.24X-FilesNeed help configure asterisk, to worked Messages and Status Users in eyebeam 1.11.3010
17:36.36outtolunclocate README.variables
17:38.51*** join/#asterisk psk (n=psk@golia.caltanet.it)
17:40.46JMcAactually...I guess my question really is for    SetVar( _foo = ${foo} )   and seeing the description, in README.variables, I'm thinking it does work, but still not sure
17:41.26outtoluncwell depending on the age of your asterisk, you should be asking about SET, not SETVAR
17:41.46JMcAI've got one 1.0.9 and one 1.2.1
17:41.53outtoluncand it tells you about inherited vars
17:42.16drumkillaJMcA: That's probably not what you want
17:42.33drumkillaThat sets the variable " _foo " to the value of " ${foo} "
17:42.37drumkillaincluding the spaces
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17:42.53JMcAok...good point...i was including the spaces for readability
17:43.28outtoluncalso, why just _ and not __
17:43.33JMcAso, SetVar(_foo=${foo})
17:44.09JMcAouttolunc: what I'm getting at is, can I take a variable that is not set to be inherited, and make it inheritable without jumping through the extra hoop of another temporary variable?
17:44.44JMcAI'm thinking for a potential situation where the variable has already been set at some previous point, but now you want to make it inheritable
17:45.01*** join/#asterisk zotz (n=zotz@24.231.47.175)
17:45.08JMcAand not __ because I believe in keeping namespace clean and tidy and not exporting crap all over the place if its not needed
17:45.11outtoluncright, and _ and __ is how deep it inherits
17:45.43outtolunck
17:46.20JMcAright, my point though is that I may set a variable for the dialplan...but in one particular spot, I need it to inherit, but don't want to set it as inheritable all over the place
17:46.31outtoluncSet(__FOO=bar) ; Sets an inherited version of "FOO" variable
17:46.31outtoluncSet(FOO=bar)   ; Removes the inherited version and sets a local
17:46.32outtolunc<PROTECTED>
17:46.53outtoluncsays it all which was why i suggested reading README.variables in the first place
17:46.53JMcAok...so it sounds like that would work
17:47.15JMcAyeah...of course, I'm asking for the reverse of that, but that gives me some confidence that it will work
17:47.20outtoluncnods
17:47.31JMcAand I'm currently reading through that file
17:47.38outtoluncit's the first one, just level 1 inherit
17:47.40JMcAjust got to that spot, actually
17:48.43JMcAouttolunc: still not exactly what I was asking about, but, again, gives me confidence that it will work
17:48.58outtoluncbar can be anything
17:49.05JMcAand besides...this is all pretty hypothetical...don't hvae any need for anything like this at this point, was just a question out of curiousity
17:49.08outtolunc"bar" ${bar}
17:49.10outtoluncwhatever
17:49.27JMcAyeah...since it gets substituted...that makes sense
17:50.07outtoluncimagine this Set(${bar}=_{bar})
17:50.11JMcAso did SetVar become Set in the transition from 1.0 to 1.2?  or is it more complicated than that?
17:50.13outtoluncer vice versa
17:50.31outtoluncnods somewhere in there
17:51.48*** join/#asterisk oceanlan (n=irc@cpe-69-133-109-130.woh.res.rr.com)
17:54.21oceanlanso whats up in the voip world today?
17:54.24oceanlanany news?
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17:55.21*** part/#asterisk epoch (n=epoch@octane.breakbeats.org)
17:55.55JMcAuhm...we determined earlier...may have been late last night after I got in...that NAT is evil
17:56.08JMcAslow news day
17:57.08[TK]D-FenderJMcA : Yes.  Setvar is depricated.
17:57.28[TK]D-FenderJMcA : NAT is only semi-evil.
17:58.04JMcAI firmly believe that NAT is fully evil...but I tend to be pretty opinionated on that subject
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18:00.47Jammy.
18:02.00znoGskype released under OSS eh?
18:02.03oceanlanHAA! JMcA: I agree..nat is evil!
18:02.07znoGerr LGPL
18:02.16oceanlani hope next gen firewalls have more support for voip traffic
18:04.36JMcAskype lgpl?  I missed that
18:05.57*** join/#asterisk Cleyverson (n=cleyvers@201.29.71.11)
18:06.47JMcAand I'm still missing it
18:07.35outtolunchttp://share.skype.com/directory/open_source_development_library/view/
18:07.41znoG++skype library only
18:07.44znoGnot skype itself
18:07.46tronixahh.
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18:08.10drumkillaand i believe it only supports a single channel
18:08.13znoGhrm, so does dundi rely on the fact that you list all the extensions in each server under a context?
18:08.14drumkilla... not very useful
18:08.24JMcAah...ok
18:08.30znoGfor example, I don't list any of my extensions in the dialplan, I have an AGI script to dial the right user
18:11.41QbYanyone have any experience with the linksys pap2--speciifically a phone not ringing when connected to it (caller id shows call coming in, and you can answer it, but it doesn't ring)
18:12.25brookshireanyone have an old 1 u server they want to sell? can be any speed...
18:13.06h3xi always wanted to know this
18:13.44h3xQbY: Why use a negative charge (-48 volts) for Ring instead of a positive
18:13.45h3x<PROTECTED>
18:13.57h3xThe reason for doing this is galvanic corrosion protection. A conductor
18:13.57h3x<PROTECTED>
18:13.57h3x<PROTECTED>
18:13.58h3x<PROTECTED>
18:16.36JMcAh3x: of course, they'll be attracted on the other end of the line
18:16.49eKo1the next question would be: Why do we care about Cl ions?
18:17.07h3xno it wont coz its the equivalent of earth ground
18:17.09JMcAeKo1: you don't want the connector where you plug your phone in corroding
18:17.26[TK]D-Fenderbrookshire : Just go to tigerdirect.com
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18:17.40JMcAh3x: huh?
18:17.49h3xtip is like "ground"
18:17.50QbYh3x..  i am confused as hell..  what do charges have to do with my phone ringing?
18:17.53eKo1the plug on my phone is more likely to break than to corrode
18:17.56h3xring is a negative voltage reference to tip
18:18.01brookshire[tk]: i need old server.. do they sell used ones?
18:18.04h3xburied cable
18:18.08h3xoutdoor termination
18:18.09h3xetc
18:18.24JMcAh3x: oh, ok...
18:18.29JMcAI was thinking of a different issue
18:18.36h3xelectricity flows from - to +
18:18.42h3xits the opposite of what most people think
18:18.46oceanlanbrookshire: i am sure you looked on ebay?
18:18.53QbYso..  h3x.. are you suggesting there is a setting for - or + in the pap2?
18:19.01JMcAt1's do alternating voltages for marks to avoid oxidation from dissimilar conductors and connection points
18:19.10oceanlanh3x: you are correct.. - to + is accurate
18:19.11JMcAI was thinking along those lines
18:19.28h3xyeah but a POTS line is DC current except when its ringing
18:19.30*** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl)
18:19.33JMcAh3x: right
18:19.51[TK]D-Fenderbrookshire : I'm sure there are a few, and a bunch of low end.  Do you jsut need the rack format?
18:19.51JMcAlike I said...I was thinking about t1 type stuff...I was in the wrong context
18:19.54oceanlanits 9 volts DC when it rings a 5 volts standing isnt it?
18:20.13JMcAringing voltage is a minimum of 40 volts
18:20.41brookshireyeah.. need a cheap server to put in a rack
18:20.44oceanlanwow, i always thought it was 9volts...i have an electronics bakground, not telecom =(
18:20.47brookshirebut we only have 1u
18:21.17JMcAyeah...typically up in the 50s...amperage is pretty low, though...still stings if you get hit by it
18:21.21JMcA(which I have)
18:21.27oceanlansame here!
18:21.36oceanlanit felt like 9 volts to me!! hehe
18:21.58JMcAI got nailed across a finger and had trouble moving it for an hour or so
18:22.18oceanlanits amazing how often it can happed in a multi-tennant building with 17 idf's when you are tracing down new connections!!
18:22.51oceanlani got my elbo once, it felt like someone punched me!
18:23.16tronixbrookshir: there's often sun netra t1 ac105 going for couple hundred bucks off ebay. cheap and has good remote management, if you don't mind fact it's SPARC-based :-)
18:23.19JMcAI used to work for an ISP in the old school analog phone line days...you could pretty much count on being hit if you were touching the exposed conductors at an ISP
18:23.50oceanlanyea...at least its not 110v ac ]P
18:23.58oceanlan:p **
18:24.18tronixheh good ole days. one employer (an early ISP) had two banks of modems... about 150 usr courier v.everything and 20 some cheap no-name modem.
18:24.26tronixguess which ones never failed? and which ones always failed?
18:24.47[TK]D-Fenderbrookshire : $400 - http://www.tigerdirect.com/applications/SearchTools/item-details.asp?EdpNo=1260771&CatId=0
18:24.57oceanlanthe cheap no-names!!
18:25.14tronixindeed. the CFO thought he was being smart by cutting costs...
18:25.14JMcAI'm gonna guess you're gonna say the courier's failed a lot...but my experience was exactly the opposite of that....
18:25.17tronixnope, nosiree.
18:25.19tronixhahaha
18:25.31tronixnah, the couriers were bulletproof.
18:25.34JMcAok
18:25.36brookshirecool.. thanks :D
18:25.38JMcAthat was my experience
18:25.49tronixi think we had only one incident of failure with the courier during the 2 years I was there
18:25.51JMcAwe had the USR Total Control racks
18:25.53oceanlanit was a 50/50 chance
18:25.54tronixthe no-names failed almost daily.
18:26.00tronix(all of them.)
18:26.14JMcAthat ISP is still running modems that are 8 or 9 years old
18:26.17tronixthink lot of it was due to quality of firmware.
18:26.46oceanlanthe modems you are talking about are at the ISP side?
18:26.59tronixthe couriers were very aggressive in holding on to even marginal calls
18:27.03JMcAthe management functionality on those Total Control racks was absolutely amazing....the connection statistics you could pull from those modems were seriously deep
18:27.34tronixand to this day, I still pat my own personally-bought courier at home. :-)
18:28.40JMcAthey had down and dirty details down to the analog signal details that the modem was using...carrier signals, gain, SNR, you name it
18:29.11oceanlanwow, sounds like our wireless gear
18:29.14JMcAtrellis-coding in use, what signal constellation was being used, even a frequency response graph of the line
18:29.18tronixoh yeah. the total control stuff was great.
18:29.22tronixwe had them 10 years ago
18:29.24tronixsolid.
18:29.36tronixlater migrated to Ciscos.. AS5200 then AS5300s
18:29.39JMcAyup, they steadily declined in quality...but still good
18:29.50oceanlanmodems are really similar to wireless radios...do you guys agree? i dont know too much about modems but that is what I have heard
18:30.18JMcAoceanlan: there are a lot of very similar technologies, concepts, and algorithms in use, yes
18:30.57tronixour primary modem administrator was in hog heaven with the TCH stats :-)
18:31.04oceanlanJMcA: thought so, just wanted to dispel any myths that i might have heard
18:31.11JMcAwhen it comes down to it, you're encoding digital data onto an analog medium...whether that's physical copper wire, or air, the concepts are the same
18:31.25tronixit was extra cool because at the time, we were part of an ILEC (telco) so we'd use them to get the switch people to reconfig paths...
18:31.59JMcAtronix: it was great...that was my role at the ISP...combine that with the ability to read hex dumps of PPP connections almost like reading a book and I *really* freaked some of our first level tech support folks out sometimes  :)
18:32.07tronixhahaha
18:32.25oceanlanbrb AFK
18:32.29tronixalaska was a tough place for that.
18:32.42JMcAand win95's inability to dump as anything other than hex....had to do it the hard way
18:33.25tronixhahaha *wince*
18:33.39tronixsometimes I'm suddenly reminded that the rosy-colored 'good old days' sometimes weren't *such* so much fun. ;)
18:34.52wizard545guys, is there anyway to stop the inbound/outbound fees when doing a call forward?
18:35.07*** join/#asterisk h4mm3r` (n=h4mm3r@85-18-14-10.fastres.net)
18:35.08wizard545just making it one way, so i don
18:35.17wizard545don't pay incoming 2/c and outgoing 2/c
18:37.38oceanlanis it your telco charging you the c/f fees?
18:38.41wizard545well, i'm taking the call for 2/c incoming and forwarding that to another number
18:39.02wizard545can i cut one side out of it? so i only get charged the one fee, without dropping the call
18:39.08*** join/#asterisk Evanrude (n=david@ip68-107-162-212.lu.dl.cox.net)
18:39.46oceanlani assume you can't just forward from point A to point C without stopping at B ? right? there is a reason you are hopping through a middle location?
18:40.00wizard545i am the middle location
18:40.18oceanlanright, why cant the A go strait to C?
18:40.23wizard545exactly
18:40.40wizard545well, it calls my asterisk box for verification
18:41.07oceanlanis this for calling cards?
18:41.08tainted-there's no way around it
18:41.20wizard545damn.
18:42.13wizard545thanks anyway
18:43.18*** join/#asterisk chewbacca (n=billg@ppp-70-243-153-206.dsl.stlsmo.swbell.net)
18:43.41*** join/#asterisk cpm (n=Chip@pdpc/supporter/sustaining/cpm)
18:44.34X-FilesOuch ... error while writing audio data: : Broken pipe
18:44.34X-FilesKilled
18:44.40X-Filesppls what this ?
18:44.58oceanlanis that for vmail or a custom audio file?
18:45.22X-Filescustom
18:46.11oceanlanwhat format are you trying to record with?
18:46.46X-Filesi not use vmail
18:47.04*** join/#asterisk DaPrivateer (i=Privatee@CRIMSON.OFF-HOURS.COM)
18:47.44ke4qqqhey guys, I am working on an asterisk integration with a very legacy pbx....one that doesn't support dtmf reception. Long story short, I have tie lines set up, and dial in with DP, however, it's routing calls successfully only about 50% of the time. The other 50% of the time the call is routed incorrectly (usually a lower numbered extension, ie, if I dial 235, I often get 232) Any ideas on...
18:47.46ke4qqq...improving *'s sending of pulses?
18:47.50oceanlanyea, but is it trying to record in .wav or .mp3?
18:47.58ke4qqqX-Files: Are you getting that error when you are trying to start asterisk?
18:47.59[TK]D-FenderLinksys SPA-942 is up on www.voipsupply.com .  2 x RJ45 w/ PoE, but (10mbit ONLY! UGH!) <-
18:48.11Qwellstill only 10mbit?  lame!
18:48.14X-Fileske4qqq: no
18:48.20tuxinator_linux[TK]D-Fender: need more than 10mbit?
18:48.23Qwell[TK]D-Fender: So, what did they improve?
18:48.32Qwelltuxinator_linux: yes, 10mbit on a switch port is stupid, at best
18:48.49Qwelljust look at the new cisco 79x1g-ge phones
18:48.50*** join/#asterisk kink0 (n=k@62.37.205.161)
18:48.51QwellThey're gbit
18:48.57Qwellvery very useful
18:49.05kink0hi, anyone know any channel here about movile GSM ?
18:49.09[TK]D-Fendertuxinator_linux : YES.  if you're going to plug a computer behind it, which is the entire point for 2 eth ports on a phone...
18:49.23tuxinator_linux[TK]D-Fender: true, good point
18:49.26Qwell[TK]D-Fender: would have cost them an extra...what...$.10 per port?
18:49.56[TK]D-FenderQwell : This mean that I could only suggest Linksys for HOME use now.  and that means even lower model than the 941.
18:50.01JMcAyeah, gig to the desktop isn't exactly a common thing, yet, but phones need to support it because it definitely is coming
18:50.21X-FilesCore was generated by `./asterisk -c'.
18:50.22X-FilesProgram terminated with signal 11, Segmentation fault.
18:50.22X-Fileswarning: current_sos: Can't read pathname for load map: Input/output error
18:50.23X-Fileshmm
18:50.25X-Filesinteresing
18:50.44[TK]D-FenderX-Files : Oh yeah... eyebeam presence has been known to crash certain versions of * <-
18:51.05[TK]D-Fenderok, bbiab
18:51.10X-Files8-E~
18:51.25X-Filesmazafaka :(
18:53.17*** join/#asterisk RoyK (n=roy@87.80-202-9.nextgentel.com)
18:54.04*** part/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net)
18:54.16*** join/#asterisk gnosys (n=gnosys@griffin2.GnoSys.us)
18:54.29*** join/#asterisk troy (n=central@CPE00907f17e478-CM014300013422.cpe.net.cable.rogers.com)
18:55.43gnosyssimple question: I run asterisk -r to reconnect to a console of a running asterisk process.  How do I disconnect from that console without killing the running asterisk process?
19:02.12[TK]D-Fendergnosys : "exit"
19:03.44*** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net)
19:03.58*** join/#asterisk rend (n=rend@unaffiliated/rend/x-000000001)
19:04.52gnosys<sheepishly> thanks, [TK]D-Fender.  I don't see 'exit' in the list of commands that one gets by using "help" at the CLI console.  Makes sense, but it just never occurred to me.  Thanks.
19:05.55rendhello. i have been using broadvoice and the service just keeps getting worse.
19:06.06tuxinator_linuxrend, I have them also
19:06.09rendi was wondering what is a good service to use with asterisk?
19:06.30tuxinator_linuxrend, what problems are you having?
19:06.57rendtuxinator_linux: dropped calls, sometimes i can hear the other party but they cant hear me. or the other way around
19:07.19tuxinator_linuxrend: I have the same problems
19:07.34tuxinator_linuxrend: happens infrequently, but it happens
19:07.47rendtuxinator_linux : figures. they try to blame my connection and ask for a tcpdump
19:08.02rendto me it happens all too often and has been getting worse
19:08.03troyb1!seen _Vile
19:08.06tuxinator_linuxrend: that is more than they do for me
19:08.13troyb1no chance :|
19:08.15rendlets say 75% of the time they work
19:08.24tuxinator_linuxrend: they make same changes everytime time I call in, and it gets better each time
19:08.34rendbut if im gonna drop my landline i need at least 95%
19:09.02tuxinator_linuxrend: I don't have a landline, just the broadvoice
19:09.15rendtuxinator_linux : i have even been using their system directly where i dont even use the net and their stuff is still crap
19:09.37rendits been almost 6 months and its worse. im gone
19:10.00rendim just looking for recommendations for another service
19:10.16tuxinator_linuxrend: you and me both
19:10.25gnosysFrom reading the docs, it's clear that there are many parameters to adjust and tweaks to make in trying to eliminate echos.  Can someone suggest an order of priority for these?  With zap fxo ports, would fxotune be a reasonable first thing to try?  Or should I start by trying to fiddle with the number of taps in echocancel (32-256).  It's at the default now (yes).
19:10.47rendtuxinator_linux : ahh glad you are not content on theri bad service
19:11.28tuxinator_linuxrend: nope, rather dissapointed, price is nice, but service is too bad, will switch as soon as I find a better option
19:13.21*** join/#asterisk Naturalblue (n=Kay@195.26.12.229)
19:14.21*** part/#asterisk Naturalblue (n=Kay@195.26.12.229)
19:14.56oceanlanHas anyone here ever ready the AsteriskTFOT? the book about asterisk?
19:14.58gnosysrend and tuxinator_linux: I have the same problems with VoicePulse.  In one case, I had 4 dropped calls in one day.  If anyone knows a service that has a very low dropped-call rate, I'd really like to know about it.
19:15.30tuxinator_linuxoceanlan: yep, I read it
19:15.45gnosysme 2
19:15.49*** join/#asterisk VJ (n=vijay@203.122.28.98)
19:15.54VJHello
19:16.09troyb1greetings.
19:16.09VJi need to install ztdummy on my slackware system
19:16.29VJcan you get me an idea about it
19:16.30oceanlani am on page 10...its very interest
19:16.32oceanlaning*
19:17.03troyb1does #0,0 still work here :P
19:17.48oceanlanI am gonna keep reading...dude is right though..."Not many people get excited about telephones...but those who do get REALLY excited"
19:17.48troyb1apparently not
19:18.18tuxinator_linuxtroyb1: #0,0 ?
19:18.38kink0about genext, I use it with an user registered as friend, and I set a number like 6969 to this user, but I am unable to dial 6969, and claims there not any extension with this number.
19:18.54kink0is that normal ? or I would be able to dial 6969 from console ?
19:19.11troyb1tuxinator_linux you used to be able to put commas in a channel name for exiting purposes
19:19.18tuxinator_linuxoceanlan: everyone should read the book and the wiki before they even ask a question on here
19:19.30rendhmm. someone told me that they never have dropped calls or delay with vonage
19:19.51RoyKerm
19:20.15tuxinator_linuxvonage doesn't play with *, if I remember correctly
19:20.23RoyKhow can I take the following, *81*number# and strip both *81* and #?
19:20.41RoyKStripAcid
19:20.57*** join/#asterisk [TK]D-Fender (n=joe@66.11.164.239)
19:20.59Qwell~striplastdigit
19:21.07jbotfrom memory, striplastdigit is ${EXTEN:0:$[${LEN(${EXTEN})} - 1]} , will remove the last digit from EXTEN, making 5551212 become 555121.  Change the "1" to remove more digits.
19:22.53rendanyone here used axvoice?
19:23.20VJCan anyone tell me how to install ztdummy on the slackware system with kernel 2.6.15
19:24.05[TK]D-FenderVJ : you shouldn't need ztdummy with 2.6
19:24.33[TK]D-FenderVJ : But it should be installed normally if you compile zaptel before asterisk
19:25.36tronixoceanlan: i've read asteriskTFOT. and author's right. :) i was reading a website last night of someone
19:25.46VJi have compiled zaptel before asterisk
19:25.48tronixwhom wired up his own strowger switch at home and integrated
19:25.49tronixwith asterisk
19:26.05tronixhe also had photos of various phones -- WE 500, etc
19:26.16VJbut now when i do modprobe ztdummy, it gives me an error message
19:26.28tronix(which I remember seeing at my grandparents' in my extreme youth -- the WE500 phone, that is.)
19:27.50*** join/#asterisk svenna_ (n=svenna@p548D2AD1.dip0.t-ipconnect.de)
19:28.10sivanawhat's the best tool to convert a wav to mp3?
19:28.21wizard545lame
19:28.26tronixocealan: if you want to see what I mean, just check out: http://home1.gte.net/dalderdi/phones/sxs1.htm
19:29.26VJhttp://pastebin.com/516458
19:29.36VJthis is the error message i am getting http://pastebin.com/516458
19:30.19VJHello [TK]D-Fender
19:30.41VJ<PROTECTED>
19:30.56VJor  Can anyone tell me how to install ztdummy on the slackware system with kernel 2.6.15
19:34.42QbYis there supposed to be hold music on a parked call?
19:35.07Math`yeah
19:35.10*** join/#asterisk moistbat (n=no@host86-130-139-1.range86-130.btcentralplus.com)
19:35.15moistbatmonkey rape
19:35.44tuxinator_linuxtronix: that stuff is older than me
19:35.54tuxinator_linuxlunch time
19:36.11*** part/#asterisk moistbat (n=no@host86-130-139-1.range86-130.btcentralplus.com)
19:36.30QbYhow do you place a call on hold from an analog phone with a ata?
19:36.36QbYbecause i'm not getting music
19:36.55*** join/#asterisk ToTo (n=ToTo@host226-162.pool875.interbusiness.it)
19:37.29Math`usually you press flash
19:37.41kink0would be dialable from console a registered user as friend with a genext ?
19:38.27VJ<PROTECTED>
19:39.40rendwhats an open source linux sip phone?
19:39.54rendsoftware phone..
19:40.21kink0rend:linphone may be
19:40.25*** join/#asterisk Sniper00X (n=sniper00@ool-44c061a7.dyn.optonline.net)
19:43.49tronixtuxinator: hahaha
19:44.30tronixrend: hmm... twinkle
19:44.47tronixseems decent.
19:46.22VJ<PROTECTED>
19:46.55QwellVJ: with 2.6, ztdummy is compiled by default
19:47.12Math`is it? I always de-comment it into the Makefile
19:48.12*** join/#asterisk shido6 (n=shido@d221-68-216.commercial.cgocable.net)
19:49.18*** join/#asterisk Cresl1n (n=matt@gateway.digium.com)
19:50.38*** join/#asterisk pifiu (n=myassisb@208.205.181.170)
19:51.08kink0I set a line like genext 6969 to an user, when is registered, I can not do a console dial 6969, is that normal ?
19:51.27Math`genext?
19:51.44kink0Math`, yes... let me check for synthax
19:52.08kink0regexten=6969
19:52.14kink0sorry... regexten
19:52.32kink0I put a regexten to a registered type=friend
19:52.49kink0but if i do a console dial 6969, I got that extension does not exist
19:52.51X-Filesppls, what this it ? http://pastebin.ca/37870
19:53.01rob0rend: kiax (IAX), kphone is SIP, there are many others, see freshmeat.
19:53.11kink0when I was supposing that exten would be assigned to the registered user.
19:54.15Math`kink0: in which context did you regexten it in?
19:54.29kink0Math`, local
19:55.00kink0context=local , but No such extension '6969' in context 'local'
19:55.07VJbut when i am doing zydummy, it gives me an error
19:55.26VJ<PROTECTED>
19:55.36VJHello Qwell
19:55.54VJ<Qwell> i am getting this error message " http://pastebin.com/516458"
19:56.27Qwellmeans you broke it
19:56.59VJhow?
19:57.22Qwelldunno
19:57.36wizard545anyone use voipbuster?
19:57.48VJwhats the solution?
19:57.50VJany idea?
19:58.14rob0VJ: you *did* what it suggested, "see dmesg"?
19:58.17kink0is normal that does not found that extension ? or must asign the extension to the registered user ?
19:58.27troyb1>pathping www.uranus.com
19:58.59VJyes
19:59.26Math`kink0: TRY dial 6969@local
19:59.49pifiuhey qwell wasup
19:59.59VJhttp://pastebin.com/516493
20:00.15kink0dial 6969@local
20:00.15VJhere is the status of dmesg "http://pastebin.com/516493"
20:00.15kink0No such extension '6969' in context 'local'
20:00.18kink0the same
20:01.21QwellVJ: You need the crc_ccitt stuff in the kernel
20:01.26rob0missing crc_ccitt I would say. Try "zgrep CCITT /proc/config.gz"
20:03.05VJshould i paste this command on my system
20:03.25VJit says gzip: /proc/config.gz: No such file or directory
20:03.32pifiuqwell did i tell you i got iax2 to work? =P
20:03.37Qwellno
20:03.41Qwellbut I can guess how
20:03.42pifiuwell i did
20:03.43pifiulol
20:03.46kink0Math`, but supposely is when I dial the extension I set as regnexten in sip.conf for an user, I would be able to dial that extension, right ? or is another the purpose of regexten ?
20:03.49pifiuoh yeah?
20:04.08rend"The cellphone industry has taught us that consumers really like nifty handsets that are fun and feature-rich,"
20:04.18rendthats bullshit since i have a very basic cellphone
20:04.33troyb1blame me i have a treo
20:04.34rob0VJ: try your kernel .config file
20:04.54tuxinator_linuxrend: mine is 5 years old, good ol nokia
20:04.56rendbut i do know idiots with blackberrys who cant figure out a computer
20:05.03*** join/#asterisk Mike (n=mike@201.135.48.190)
20:05.07rendtuxinator_linux : i just got a new samsung but it was $30
20:05.50Mikehey guys, im having alot of trouble with rawplayer, seems very unstable anyone has a patch or something ?
20:05.50kink0rend... about blackberrys and asterisk... and mobiles... do you know any movile terminal that supports at+fclass=8 ?
20:05.52VJrob0: where would this file be located
20:07.07rendkink0: no
20:08.08rob0VJ, how did you manage to get a 2.6.15 kernel without knowing where you configured and compiled it?
20:08.28*** join/#asterisk asterboy (n=Snake@S01060204ee2b6007.ed.shawcable.net)
20:09.00rendi just singned up for a new axvoice account but it looks like they dont activate accounts on the weekend :(
20:10.37asterboybunch of dope smokers
20:11.02asterboytell them to get back to work.
20:11.17rob0Get back to work, dope smokers.
20:11.21rob0done
20:11.22asterboylol
20:11.36asterboyanyone here have the Polycom IP 300?
20:11.54asterboy~polycom
20:12.02jbotextra, extra, read all about it, polycom is the manufacturer of one of the best IP phones in the market. http://polycom.com - Note: Here is where you can get some downloads: http://www.polycom.com/resource_center/0,,pw-6812-12612,00.html
20:12.02Math`I installed 301s
20:13.14*** join/#asterisk ThomasJ (i=thomas@535A853C.flatrate.dk)
20:14.00[TK]D-Fenderasterboy : what about the IP 300?
20:14.14*** join/#asterisk angler- (n=angler@24.214.255.222)
20:14.36ThomasJHello, can anyone help me with a voicemail isssue?
20:14.38asterboyJust setting one up for the first time...do I use the same SIP and bootROM from the IP 500?
20:14.53X-Filesppls, why i see warning http://pastebin.ca/37870 ? Please answer.
20:15.00*** part/#asterisk rend (n=rend@unaffiliated/rend/x-000000001)
20:15.10[TK]D-Fenderasterboy : Depends which version.  What does it have now?
20:15.25asterboychecking...
20:15.46asterboy2.5.0 for bootROM
20:16.26asterboynot sure for sip, does not show in ABOUT menu
20:16.44ThomasJPlease, i need help! :)
20:17.28[TK]D-Fenderasterboy : Its there...
20:17.34*** join/#asterisk teg (n=ter@217.164.220.197)
20:17.47*** part/#asterisk teg (n=ter@217.164.220.197)
20:17.50Corydon76-homeasterboy: yes, you use the same ROMs for the 300 as for the 500
20:17.52*** join/#asterisk nextime (n=nextime@213-140-6-103.ip.fastwebnet.it)
20:18.05asterboyok thx
20:18.05RoyKThomasJ: if you DO ask WHAT you wonder about then MAYBE someone MIGHTT help you. whining about that you need generic help is useless
20:18.09asterboytrying...
20:18.11Corydon76-homebut not nearly the same as the 301 and 501
20:18.20[TK]D-Fenderasterboy : I'd suggest 2.6.(1/2) BR, and 1.5.(2/3) SIP. and yes it uses the same firmware, just different parts
20:18.34Corydon76-homeThe latest ROMs from Polycom need more memory than the 300 and 500 have
20:18.34asterboyya, I like the 2.5
20:18.36[TK]D-Fender(internally)
20:18.37asterboyerr...2.6
20:18.48[TK]D-Fender2.6 is safe.
20:19.00[TK]D-Fenderits 3.x BR that you should avoid
20:19.04asterboyAnyone get these phones to work with FWD yet?
20:19.13[TK]D-FenderSIP is safe all around
20:19.16asterboyya the 3.x is a one way street.
20:19.33[TK]D-Fenderasterboy : SIP is SIP. its all the same...
20:19.45asterboyok, good to hear
20:19.55[TK]D-Fenderasterboy : besides aren't you going to use it through *?
20:20.15Qwellkram: morning
20:20.20krammoring qwell
20:20.30Corydon76-homeG'dafternoon
20:21.38[TK]D-FenderI brought an IP 600 home from work, but the wrong power adapter :(
20:21.45asterboyFender, eventually, right now I'm connecting direct to VOIP terminator.
20:21.56Mikeanyone knows how to make rawplayer stable enough for a production enviorment?
20:22.16angler_I think I've had it with maxtor hd's. I've had to many fail now
20:22.36asterboyI still can't get the Polycom to register with FWD
20:22.44asterboynot sure what I'm missing in config
20:22.45Qwellangler_: hopefully seagate won't make the maxtor line worse
20:22.55Qwellor, introduce the maxtor issues in the seagates, heh
20:23.09Math`asterboy: how come its not registering?
20:23.19Math`Qwell: hopefully
20:23.23angler_Qwell, what do you mean
20:23.32Qwellangler_: seagate owns maxtor now
20:23.40Math`maxtors have the reputation of... how can I say... failing easily
20:23.42angler_Qwell, oh!
20:23.55angler_Qwell, how good are seagates?
20:24.02Qwellvery, usually
20:24.14Math`5 years warranty on OEM
20:24.35angler_Qwell, i've only had maxtor and WD, my one WD failed also
20:24.45angler_stuck with maxtor and just had my 4th die
20:24.49Math`I had both, both failed
20:24.55Math`my seagate is still alive tho :)
20:25.17QwellWD is crap, heh
20:25.22angler_i'm glad this last time I made a raid, still got my data on one of the maxtors
20:25.47Errit's been my experience that batches of drives die, but not necessarily brands in general - and it's also been my experience that if a drive lasts 3mo, it will last 200 years :-)
20:25.56rob0<== planning to buy Seagates from now on
20:26.10angler_iv'e had these maxtors for about 3 years now
20:26.16Qwellimo, the maxtor line can only get better now
20:26.24Qwellespecially if seagate still offers the uber-warranty
20:26.37Qwelldid anybody notice that maxtor dropped the 3 year warranty a year ago or so?
20:26.54Errthat's doubtful - short warranties are one of the ways that hard drive companies can sell their cheap drives cheaper
20:26.57angler_well im out of here... my $100 compusa giftcard is coming soon so i'll use it... they better carry seagate too
20:27.14Qwellangler_: compusa...eh...expensive
20:27.23ErrMaxtor and WD have both started making *lines* of their drives that are 1yr warranted, instead of the traditional 3yr - and the drives are $20-30 cheaper
20:27.29angler_Qwell, yes i saw they dropped it. I remember when they said they raised it to 3 and i was excited... then they dropped it
20:27.52angler_well im off...
20:27.53Qwellspeaking of dropped...heh
20:28.06Qwellyell at the dumb bitch at the checkout...she WILL be rough with the drive
20:28.48*** join/#asterisk razu (n=razu@217-159-187-162-dsl.prn.estpak.ee)
20:29.16asterboyI have seen more maxtor failures then any other drive...I stay far away from them now.
20:29.24Kattymew.
20:29.41Qwellasterboy: even IBM deathstars?
20:30.03asterboyMath', not sure why my polycom won't register with FWD...I'd like to see someone else with a working config.
20:30.25asterboyIBM deathstars...hmmm..not heard of em.
20:30.26[TK]D-Fenderasterboy : is your phone behind NAT?
20:30.35asterboyyes
20:30.37Qwellasterboy: deskstar
20:30.39asterboysymetrically
20:30.48Qwellhad like a 50% failure rate or something :p
20:30.49[TK]D-Fenderasterboy : I believe there is a setting in sip.cfg you'll need for that then.
20:31.01[TK]D-Fenderok, gtg, later all
20:31.15asterboydo tell!
20:35.14tuxinator_linuxall but one of my deathstar drives lasted 3 or more years
20:36.03tuxinator_linuxone of my deathstars died last week, after 5 years of abuse
20:37.22tuxinator_linuxthe problem with that batch of deskstars, from what I hear, is that the drives were placed to close together, and the heads would get stuck
20:37.44Math`thats bad
20:41.05*** join/#asterisk joat (n=joat@ip70-160-150-20.hr.hr.cox.net)
20:44.51QbYDoes anyone know how a phone rings?  Why a phone would ring on a regular PSTN but not from an ATA?
20:45.17Errphones ring from a high-voltage sine wave being placed on the line
20:45.29Errit's possible that your ATA doesn't produce sufficient voltage to ring whatever phone you're using
20:45.44Err(if it's a mechanical bell phone - if it's an electronic phone, maybe the ATA isn't producing a clean enough signal...)
20:46.00QbYI have two phones here.  One without an AC ADapter, one With.  The one With the AC ADapter only shows the Caller ID--it never rings.  The other rings as normal..
20:46.03QbYBoth work when you pick it up
20:46.18Erris it possible that the ringer is turned off on the phone?
20:46.23QbYchecked and rechecked
20:46.33Errare you sure the ringer works?
20:46.37QbYplugged it into the regular phone outlet, and it works fine..
20:46.42QbYmove back to the ATA and it won't ring..
20:46.55Errsince it's handling caller ID, it must notice that the phone is ringing (since caller ID is only sent between ring pulses, IIRC)
20:47.09QbYexactly..  that's what is weird.
20:47.33joattoo many phones on the same interface/line?
20:47.40QbYone phone.
20:47.45joathmm
20:47.56QbYthere are more phones on the house (pstn) line..
20:48.43Erryes, but ma bell guarantees a REN of 10 - I'm sure your ATA doesn't have a REN of 10
20:48.45joatsame as the phone that won't ring?
20:48.53Errspeaking of, what is the REN on your ATA, and on your phone?
20:49.11QbYI have a PAP2..
20:49.24asterboyok, Polycom IP 300 operational.
20:49.39asterboyThat was easy...just copied my xml files from IP 500.
20:49.42QbYREN 0.0A 0.0B
20:49.45joatgoogle says pap2 has a REN of 5
20:49.49asterboyIP 300 of course ignores the 3rd line.
20:49.58*** join/#asterisk konfuzed (n=KonfuzeD@H135.C72.B0.tor.eicat.ca)
20:50.04Errwell, I can't imagine the phone has a REN > 1, so that shouldn't be an issue
20:50.25QbYREN is Ringer Equivalence right?
20:50.34Qwell~ren
20:50.36asterboyAnyone know what needs to be configured on the Polycom phones to get them to register with FWD?
20:50.42QbYthe bottom of the phone says, Ringer Equivalence 0.0A 0.0B
20:50.46joatringer equivalent number
20:50.48Erryes - it's the count of the number of mechanical-bell phones that can be on a given line
20:51.08asterboyAnyone have a Polycom configured with FWD?
20:51.20QbYi just for the life of me cannot figure out why it would ring any other phone, and not this one..  but if i plug this one in somewhere else, it works..
20:51.26Math`asterboy: you don't have an * box at home?
20:51.50asterboyI do, but I was hoping to configure it directly.
20:52.14Math`your phone service isn't voip?
20:52.49asterboyit is voip on one of the lines.
20:53.03asterboyWanted to get the other line to connect to FWD.
20:53.15asterboyHave all the settings in place, however, it won't register.
20:53.27Math`well use asterisk, make your phone register with asterisk and make your dialplan forward to fwd when you dial a prefix
20:53.28asterboySays wrong user name/pass in logs.
20:53.33Math`I dial **03[fwdnumber] to call out
20:53.45Math`then check your user pass?
20:53.50asterboyya that is most likely what I'll end up doing.
20:54.03asterboyI know the pass and name are correct.
20:54.21asterboyFender said there may be a setting that needs tweaking in sip.cfg.
20:54.26asterboyNot sure what it is though.
20:54.32Math`he said for NAT
20:54.37asterboyyes
20:55.05asterboyMy other line direct to VOIP provider works.
20:58.57QbYdoes anyone have an example of how to program a multi line phone..  ie. caller dials 123 if 123 is busy it tries 321 if 321 is busy, or is unanswered it goes to 123 voicemail
20:58.57asterboyFrom the log Registration Failed...Error Code:403
21:00.45*** join/#asterisk Ariel_ (n=Ariel@dsl-20-177.cofs.net)
21:01.52wasimQby: XX,1,Dial(IAX/123)\nXX,2,Dial(IAX/321)\nXX,3,Voicemail(${EXTEN})
21:02.14QbYthanks
21:02.47De_Moncan i unregister a sip line without commenting out the register => directive in sip.conf and reloading?
21:03.57Math`wasim: _XX
21:06.42*** join/#asterisk oceanlan (n=irc@cpe-69-133-109-130.woh.res.rr.com)
21:07.08*** join/#asterisk msw_ (n=msw@rdu-nat.rpath.com)
21:07.39*** join/#asterisk gvag11 (n=g@ppp18-adsl-195.ath.forthnet.gr)
21:07.43gvag11hi all
21:07.56*** join/#asterisk areski (n=areski@245.Red-83-60-89.dynamicIP.rima-tde.net)
21:10.37QbYPhone rings..  It was the voltage
21:11.19Erra lot of ATAs (including commercial PBXs) cheat on the output voltage
21:12.03QbYnow, i've gotta go buy the rest of the PAP2's at Office Depot
21:12.12QwellQbY: They on sale again?
21:12.19QbYit was $59
21:12.26Math`QbY: do you have a linksys reseller account?
21:12.27Errwithout getting Vonage service?
21:12.46QbYbut, since I've got it to work, I can send them out to everyone who needs them at our company instead of using softphones from home
21:12.49QbYMath.. No.
21:12.55Math`QbY: a company I work for is probably gonna buy 1000 of them so I'm gonna get some discounts :)
21:13.10QbYErr.  I followed the unlocking procedure, which worked well..
21:13.17ErrQbY: I meant about the price, actually
21:13.19Math`hehe
21:13.39QbYErr.  Yeah, without Vonage, $59..  They also gave me a $100 rebate
21:13.43QbYbut I have to sign up with Vongage
21:14.24Erryeah, I've seen huge rebates - I thought they were $160 with a $100 rebate; that's cool
21:17.02*** join/#asterisk Triple1243 (n=Triple12@modemcable171.79-70-69.mc.videotron.ca)
21:17.33*** join/#asterisk gambolputty (n=gambolpu@cblmdm72-240-116-131.buckeyecom.net)
21:17.37trixter'free' devices that come with a rebate that requires you to sign up and pay a monthly fee arent really free, and I think that advertising them as free should be banned by the FTC for false advertising
21:17.38trixter:P
21:17.55Math`hehe
21:18.21Triple1243anyone having realtime VM probs with 1.2.1
21:18.22Triple1243?
21:18.38QwellTriple1243: upgrade to 1.2.2
21:19.30Coccyxthere's an unlock procedure to unlock motorola ATAs from vonage?
21:19.47Triple1243hmm Coccyx if you knwo it tell me howlol
21:19.52Triple1243i heard sniffing could work
21:20.07Coccyxin the scrollback I thought QbY was talking about unlocking vonage ATAs
21:20.17Coccyxnot to connect to their service but to reuse on asterisk
21:20.35trixterafaik I am the only one that has worked on unlocking the motorola vt1000 from vonage
21:20.39*** join/#asterisk alephcom (n=alephcom@66.244.235.117)
21:20.42alephcomhello everyone
21:20.51Coccyxtrixter: any success?
21:21.04trixterI am not finished with it, although I have gotten almost all the way..  I have tools to make the configuration file that they use, maybe in feburary I will have time to finish that
21:21.11trixterI have gotten it partially unlocked
21:21.21trixterthere is an article about it on my webpage http://www.0xdecafbad.com
21:21.31alephcomI'm configuring a TDM400P FXO card.  when I try calling the number that hits it.  As soon as I hang up then asterisk recognizes that something happend and picks up.  Any comments?
21:22.33trixterI have more information on them than what is on my page as well, the tech manual I have (which I cant post on my page) has additional commands that arent standard to vxworks
21:29.16*** join/#asterisk Damin (n=damin@nucleus.nacs.net)
21:30.11*** join/#asterisk clive- (n=pirch@dsl-165-136-148.telkomadsl.co.za)
21:33.52*** join/#asterisk jhiver (n=jhiver@AStDenis-105-1-4-4.w193-253.abo.wanadoo.fr)
21:33.52jhiverhi all
21:33.52jhiverdoes anybody have this issue with VoIPjet where you dial, it hangs forever before rejecting the call
21:33.52jhiverand you see this on the asterisk CLI
21:33.52jhiverchan_iax2.c:1480 attempt_transmit: Max retries exceeded to host 216.118.117.46 on IAX2/voipjet/2
21:34.46filecouldn't send the packet to voipjet...
21:34.49fileno soup for you
21:35.17JMcAor perhaps it sent it, but the return packet(s) couldn't get back
21:35.18jhiverI wonder if it's because of me or if voipjet is down for some reason
21:35.39jhiverI have no such issue with NuFone or phonext
21:35.49filejhiver: your soup access is now, TERMINATED!
21:36.16jhiverfile what's you talking about :)
21:36.36JMcAeh...SOUP has a lot of overhead, if you can get away with just XML POST, you're better off  ;)
21:37.02jhiver...
21:37.19fileI'm Josh
21:37.20filenice to meet you
21:37.34fileYou may remember me from such Asterisk applications as app_directed_pickup, and chan_sip fixes.
21:37.40fileBut today I'm here to talk to you about something far more serious
21:37.43filePacket loss.
21:38.13Errjhiver: is this over a LAN?
21:38.16tuxinator_linuxoh no !!! packet loss !!!
21:38.27jhiverErr: no
21:38.46Erris there a firewall between the two boxes?
21:38.53jhiverit's My SIP ATA (public IP) -> SER (public IP) -> Asterisk (public IP) -> VoIPJet
21:39.12jhiverthere is a firewall but it works with NuFone as far as I can tell
21:39.13clive-file, what packet loss do you ahve
21:39.24fileI have no packet loss.
21:39.43*** join/#asterisk kilobit2001 (n=locid@206-248-159-174.dsl.teksavvy.com)
21:39.58clive-maybe I picked up the end of a conversation out of context:)
21:40.06*** join/#asterisk X-Files (i=x-files@x-files.lv)
21:40.09oceanlanjhiver: i see you are using SER? are you using this to combat NAT problems with phones that do not support STUN??
21:40.47tuxinator_linuxfile doesn't loose packets, he just misplaces them
21:40.56jhiveroceanlan, no I don't
21:41.00fileindeed
21:41.09oceanlanI am looking for a way to bypass NAT on networks that use phones that have no STUN support...any ideas?
21:41.09jhiverI use it to handle all the SIP registers
21:41.12filefor I am a corrupted file!
21:41.21jhiveroceanlan, SER does look good
21:41.26filenat=yes canreinvite=no in chan_sip works wonders you know...
21:41.28jhiveryou have a very good paper on onsip.org
21:41.45Triple1243you cant bind 2 ips to aasteisk ?
21:41.47kilobit2001hello,
21:41.54Triple1243then should i tunnel
21:41.58gambolputtyanyone use the REGEX function yet?
21:42.39oceanlanjhiver: what is your purpose for not having Asterisk do the SIP registration?
21:42.54oceanlanfile: was that directed towards me?
21:43.02fileit was directed to everyone
21:43.07fileor you
21:43.09fileyou choose.
21:43.10jhiverWell, I need SER to handle routing between a few boxes and some customers
21:43.13Triple1243anyone hknow how to bind multiiple ips to asterisk ?
21:43.25oceanlanok, i have all those settings and I still have problems on certain phones..
21:43.25jhiverand so I use it to register my SIP devices as well since it does it very well
21:43.44kilobit2001is it possible to specify that cdr only records specific actions into database, instead of all.
21:43.57jhiverthen I use Asterisk as a 'PSTN handoff' gateway
21:44.01dmzhmm still noone in #asterisk-users
21:44.10*** join/#asterisk NewSole (n=dave@d38-53-48.commercial1.cgocable.net)
21:44.15oceanlanjhiver: are there any performance issues and how does the RTP stream know to goto the asterisk box? is it determined in SER?
21:44.21oceanlanahhh
21:44.29jhiverRTP and SIP are separate things
21:44.45jhiverSIP is the signaling, it doesn't care about RTP so much
21:44.58oceanlanright, you answered my question basically by telling me that you are using Ast* as a pstn handoff
21:45.17jhiverok :)
21:45.23jhiverAnybody using plainvoip?
21:45.28jhiverare they any good?
21:45.28oceanlanright, i was just wondering when a call is in session, how did the asterisk no to handle it and not the SER..
21:45.45oceanlani have been looking into a SER box..
21:45.50jhiveroceanlan, not sure to understand what you mean
21:45.56kilobit2001i have cdr mysql running./but get one row for every keypress of callers. is this how it should work?
21:46.27oceanlanhehe...not sure if I do either...i have never tried to register a device to a different box than the one that is handeling the stream!
21:46.45jhiverwell ser just forwards the SIP call to asterisk and that's it
21:46.58jhiverit adds a Via: header to stay in the SIP signaling path
21:47.00oceanlanahhh..understood.
21:47.04jhiverand that's all there is to it
21:47.08oceanlanThat was my thought but I wasnt sure
21:47.36fileif you have record routing in your ser.cfg that is...
21:47.45jhiverfile, yes :)
21:47.50fileif not you can have it drop out... and act as a nifty load balancer
21:47.59jhiverotherwise you loose acks and byes and such ;)
21:48.11jhiverfile ?
21:48.46jhiverhow would you handle the load balancing with SER?
21:49.00fileclient sends INVITE to SER, SER load balances across multiple Asterisk boxes, SER does not record route, subsequent packets go between client and Asterisk
21:49.01Triple1243SER cant do IAX
21:49.12fileTriple1243: of course not, that's why it's called SIP Express Router
21:49.17Triple1243yeah
21:49.23Triple1243so doesnt really load balance asteriks
21:49.26Triple1243load balances SIP
21:49.26jhiveryeah sure but config wise how does this work?
21:49.34jhiverat the ser level
21:49.35*** join/#asterisk dudes (n=dudes@12-215-33-205.client.mchsi.com)
21:49.39filejhiver: custom module
21:49.43Triple1243complicated
21:49.45jhiveraaah
21:49.46oceanlanthat is good info to know...
21:49.46Triple1243for noob
21:49.54Triple1243but search voip-info i guess
21:49.55fileI have other... SER... stuff
21:49.56Triple1243never did it
21:49.56kilobit2001file-- sip users register with ser and not asterisk?
21:50.00filethat's cool ;)
21:50.02oceanlanthat might come in handy for redundancy and load balancing across boxes.
21:50.04filekilobit2001: depends on your setup
21:50.23Triple1243yeah usually you put SER on outside your firwalled net
21:50.29Triple1243and it pushes inside etc
21:50.38*** join/#asterisk Mike (n=mike@201.145.69.249)
21:50.54oceanlanSO this my be an effective way to get around NAT?
21:51.10Triple1243y
21:51.12Triple1243i assume
21:51.17filethe NAT approaches for both SER and Asterisk are very similar
21:51.22filenat handling approaches
21:51.28oceanlanPut the SER on the DMZ or something and have routes between it and the * boxes?
21:51.28kilobit2001file: you would do sip registeration with  ser, or asterisk?
21:51.44filekilobit2001: see above, depends on your setup
21:52.15wizard545anyone kn ow anything about the FCC fees involved with a 800 number being dialed from a payphone?
21:52.16filelook what I've started.
21:52.23oceanlanthis is very interesting...b/c i still have NAT problems on phones that do not support STUN
21:52.24Triple1243anyone kn ow anything about the FCC fees involved with a 800 number being dialed from a payphone? ?
21:52.25filewizard545: it's to reimburse the payphone owner
21:52.29Triple1243that passed to priovider
21:52.31Triple1243EX you
21:52.45wizard545am i always charged a fee?
21:52.56Triple1243so .. you dial from payphone to a voip provider.. they charge voip provier
21:53.02Triple1243yes
21:53.03wizard545does it show up on my bill from my ogrinator?
21:53.11filedepends on the provider whether they have the capacity or not to know whether to bill you it or not...
21:53.14kilobit2001what is the right way of forwarding calls back to ser.  is dial the command to use?
21:53.15oceanlanhmm...SER is on the Wiki right?
21:53.20oceanlanI need to do some research
21:53.46wizard545file, ok, so it's a 50/50 i might not get charged, but do you know the top i can be charged per call?
21:54.00fileask your provider
21:54.03Erryou should talk to your provider
21:54.06wizard545and can asterisk tell if a call is coming from a payphone?
21:54.18filethat's not up to asterisk, it's up to your provider
21:54.21Errmost providers don't pass along that info
21:54.40fileI work at Asterlink... and what we get on calls is ANI2 which tells us whether it's a payphone or not, and we bill on that
21:54.47filewe also have the capacity to pass it on in the dialed number
21:54.49wizard545this is a calling card business, so i would need to know, whether or not to surcharge the card
21:54.59filewizard545: then you have to talk to your provider
21:55.04wizard545ok
21:55.10trixterasterisk can tel if a call is coming from a payphone if you have caller id and LIDB access
21:55.17trixterodds are you dont have lidb though
21:55.33filetrixter: or have ANI2 if you have a direct PRI, or have ANI2 prefixed to the end of the dialed number
21:55.39Errthat's the round-about way to determine whether or not it's a payphone :-)
21:56.17trixterfile: if you have a provider that even gets that, many dont :/
21:56.31filetrue
21:56.35filewe had to fight for it...
21:56.39fileand right now I'm fighting for RDNIS
21:56.40trixterlidb requires ss7 typically (there are some gateways that dont but ultimately its an ss7 databse) so that isnt something a home user would do anyway
21:57.15wizard545.. hmm.. stuck.
21:57.22Errheh, if you have an SS7 connection surely you can get the ANI2 data as well :-)
21:57.27wizard545nufone is gonna take a week to get back to me
21:58.35trixterit depends on how you interconnect and what you have opted into for ss7 access
21:58.48trixteryou dont always get full access just becuase you have ss7 between two providers
21:59.03trixterwizard545: yeah that is normal
21:59.21trixterat least they will get back to you, many have reported they dont get back at all just ignore people that have problems
21:59.59tuxinator_linux~ss7
22:00.08jbotsomebody said ss7 was can be used in conjunction with ss7box.com - see the website.
22:02.56*** join/#asterisk RoyK (n=roy@87.80-202-9.nextgentel.com)
22:07.49X-FilesPpls help please, why crash asterisk ??? gdb from core : http://pastebin.ca/37897
22:14.14pifiuwhich file do i have to edit in order to have asterisk start at boot time on fedora core 4?
22:15.06Ariel_pifiu, just do in the /usr/src/asterisk directory make config it will setup the startup file
22:16.06pifiujust make config in that directory?
22:16.07pifiureally?
22:16.52wizard545anoyne know any good voip providers that do 8YY origination at around 2/c a minute except for nufone?
22:19.10inv_Arpwizard545: I hear asterlink is ok
22:19.39asterboyAnyone selling broken Polycom IP 500s for parts?
22:20.55asterboy~selling
22:21.08asterboy~jbot
22:21.11jboti guess jbot is only marginally useful at best, or a silly little bugger
22:21.11wizard545asterlink is 2/c AWESOME
22:21.16wizard545nufone was killing me
22:21.54wizard545file are you around?
22:25.41*** join/#asterisk lalito (n=erg@201.137.152.226)
22:27.08*** join/#asterisk troyb1 (n=central@CPE00907f17e478-CM014300013422.cpe.net.cable.rogers.com)
22:27.57pifiui could have sworn i had to edit some other file to do that
22:28.52troyb1hey slePP
22:30.07*** join/#asterisk Tozaz2 (n=tozaz@m116.net85-168-60.noos.fr)
22:30.18Ariel_wizard545, nufone was killing you?  are you talking about incoming to you calls or your outbound?
22:30.41wizard545incoming... no support
22:31.01Ariel_pifiu, the make config puts the script for autoboot for you on RH type of systems
22:31.08wizard545dropped a couple calls... but that happens.. but takes them a week to get back to me
22:31.56Ariel_wizard545, ok for incoming I have been using voicepulse connections side for 11 dollars unlimited inbound. They have been great for me over 2 years now.
22:32.42wizard545Ariel_ how many concurrent calls? can they give me a 800 DID?
22:34.04*** join/#asterisk bjohnson (n=bjohnson@i216-58-43-124.cybersurf.com)
22:34.04Ariel_wizard545, I don't have there 800 numbers but I have been able to get 4 calls inbound at the same time from them.
22:34.34wizard545.. i need a 8YY DI
22:34.35wizard545DID
22:35.18Ariel_800 service humm I have not check with them for it.  But I like that there iax2 connections.
22:37.48SkalTurauh
22:37.51SkalTurathis is demanding
22:38.09SkalTurai'm going to build my first asterisk box soon, infact first any kind of PBX or something like that
22:38.28SkalTuraand already we've got a client which is one of the largest phone service companies in this country!
22:38.55SkalTuraso basicly, i must provide straight from the beginning very high quality service
22:39.31*** part/#asterisk shido6 (n=shido@d221-68-216.commercial.cgocable.net)
22:43.20*** join/#asterisk nesys (n=nesys@ALICE-WHACKER.MIT.EDU)
22:44.30nesysHi folks, I've a problem with asterisk 1.0.9 and debian system ... when I try to make the first call after reboot, I receive an error message, then the safe_mode restart asterisk, and I never see an error
22:45.09nesysthe message is that: http://pastebin.com/516699
22:45.16nesysany advice will be appreciated :)
22:46.58SkalTurawith my limited knowledge about asterisk
22:47.07SkalTurai would say that is a definate hardware problem
22:47.26SkalTurait makes in illegitimate call and thus crashes
22:47.59nesysmmm ... memtest86? ;)
22:49.37SkalTurareally good idea
22:50.19nesysthanks SkalTura :)
22:51.00tronixcould be marginal memory or a cooling problem internally
22:51.37nesyscooling problem, I don't think so
22:51.49nesysTemp.= 26.0, 46.0,
22:52.03tronixguess that looks reasonable
22:52.05nesysfor a sempron 2.2 is good 46, I think
22:52.24nesysmaybe memory problem
22:52.36nesys(damn ... always mem problem :( )
22:53.06tronixdoes it always exit at same place?
22:55.01nesystronix yes always
22:55.42nesystronic not the same, sry
22:56.14nesysmemory problem ;)
22:57.04tronix:)
23:01.29*** join/#asterisk zotz (n=zotz@24.231.47.175)
23:02.20*** join/#asterisk dudes (n=dudes@12-215-33-205.client.mchsi.com)
23:03.43*** part/#asterisk Roofus04 (i=wassabi@ip24-170-193-101.ga.at.cox.net)
23:05.15dmzanyone have a moment to help with fwd+asterisk? I keep getting a "No Authority Found" error :(
23:07.45*** join/#asterisk MatsK (n=mk@c213-100-73-227.swipnet.se)
23:09.35*** join/#asterisk cpm (n=Chip@pdpc/supporter/sustaining/cpm)
23:10.34cpm<PROTECTED>
23:10.49cpmanyone clueful about this? lend us a hand for a bit?
23:11.20*** part/#asterisk clive- (n=pirch@dsl-165-136-148.telkomadsl.co.za)
23:15.35*** join/#asterisk lesouvage (n=lesouvag@82.74.11.143)
23:18.06lesouvageIs the asterisk community in any way involved in the work of voipsa ( http://www.voipsa.org/ )?  voipsa is an abbriviation voip security alliance.
23:21.31*** join/#asterisk BladeRunner05 (n=feelme@adsl-210-91.38-151.net24.it)
23:21.31*** join/#asterisk pigpen2 (n=mark@fw.seamans.cc)
23:27.04*** join/#asterisk SpaceBass (n=SpaceBas@static-71-251-230-2.rcmdva.fios.verizon.net)
23:27.11SpaceBasshey folks
23:29.16SpaceBassi recently upgraded my AAH install to the latest version and new hardware and I'm having bad jitter and some other prblems
23:29.48*** join/#asterisk cjk_ (n=cjk@11.121.9.213.dsl.getacom.de)
23:30.01*** join/#asterisk _cleric_ (n=dacleric@p5482856A.dip0.t-ipconnect.de)
23:32.03SpaceBassmy old box was a PII 300mhz with 256mb ram and worked fine for my few phones and trunks
23:32.22pifiuhow do i start the cdr on asterisk? so it can start logging?
23:32.44SpaceBassbut I was worried about it dieing so I moved to a 1.25ghz box with 512mb ram
23:33.29SpaceBassnow I have BAD jitter on my zap tel lines and my broadvoice is basically broken... when I call out on my BV trunk(s) it rings and as soon as the other party picks up we both just have dead air... no audio on either end
23:33.33SpaceBassanyone have any ideas?
23:35.34SpaceBassanyone awake?
23:35.48*** join/#asterisk Math[laptop] (n=Math_@modemcable148.4-81-70.mc.videotron.ca)
23:40.21SpaceBassit looks like it has something to do with the bridge to broadvoice
23:40.25SpaceBasshow can I keep it from bridging?
23:41.16VJHi guys, can anyone guide how to setup ztdummy on slackware with kernel 2.6.15
23:44.54*** join/#asterisk Synapes (i=S@bzq-206-125.red.bezeqint.net)
23:45.42SpaceBassanyone know why I get no audio on outbound calls using Broadvoice but incoming works fine?
23:46.05*** join/#asterisk JaredBluestein (n=Jared@nwlnnhbas01-pool3-a18.nwlnnh.tds.net)
23:46.23Synapesafter creating extension and creating a digital receptionist for it, when trying to call from another extension i get into an error: "486 Busy here" and the digital recpctionst won't answer, any ideas?
23:47.17VJ<PROTECTED>
23:47.31VJi need to install ztdummy instaed of any digium card
23:47.48VJany one knowing about the configuration how to do it?????
23:48.05ErrVJ: have you read the documentation?
23:51.56lesouvageI'm looking for a way to view my cdr.db file.  Is there a viewer or a tool I can install?
23:52.02Synapesanyone? please?
23:53.51*** part/#asterisk JaredBluestein (n=Jared@nwlnnhbas01-pool3-a18.nwlnnh.tds.net)
23:55.08srtlesouvage: db_dump
23:55.16lesouvageVJ: I'm not sure but i think "modprobe zaptel" and "modprobe ztdummy" will do the trick. Enter this on the linux prompt. With lsmod you can check  what modules are loaded.
23:55.53SpaceBassanyone else having problems with no audio on Broadboice?
23:55.58SpaceBassbroadvoice even
23:56.22*** part/#asterisk Cresl1n (n=matt@gateway.digium.com)
23:56.38lesouvagesrt: that's the name of the tool I need to install?
23:57.36srtyou probably already have it
23:58.28*** join/#asterisk Simon- (n=byte@proxima.arlott.org.uk)

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