irclog2html for #asterisk on 20050219

00:00.09`SauronThat still doesn't answer the question. The broadcast has to come from somewhere...
00:00.18mafkeesping -b
00:00.24*** join/#asterisk Ayano (~erik_leee@209.143.187.254)
00:00.33*** join/#asterisk amir (~amir@shield.guindehi.ch)
00:00.34ovidiu_25this is L3
00:00.35`Sauronthat's still layer3
00:00.57`SauronI need to send a packet to MAC FF:FF:FF:FF:FF:FF...
00:01.07Luhiwuwhat about arp query?
00:01.43mafkeesuse a bridging firewall with ebtables
00:02.00*** join/#asterisk bkw_ (~brian@bkw.developer.and.friend.of.asterisk)
00:02.00*** mode/#asterisk [+o bkw_] by ChanServ
00:02.07mafkeesreroute ur package to the mac u want
00:02.11*** join/#asterisk hermie (~nick@24.236.167.53.bay.mi.chartermi.net)
00:03.00ovidiu_25but this is no broadcast
00:03.18`Sauronovidiu's got the idea
00:03.36mafkeesyeah, I'm lost
00:03.45Ayanodoes anyone have any good links on setting up IVR?
00:03.46mafkeeswhat is it that you want to do ?
00:03.53sivanawhat kind of address is this:  Mansarover Commercial Complex.near Habibganj Rly. Station
00:03.57NuggetAyano: http://voip-info.org/
00:04.01Luhiwu`Sauron, try an arp query, it does a L2 broadcast...
00:04.24ovidiu_25but only in the broadcast domain
00:04.40ovidiu_25the router break the broadcast domain
00:04.46`SauronHumm.
00:04.55mafkeesof course
00:05.02`Sauronovidiu: That's alright. I'll never leave an L2 domain
00:05.06`Saurons/an/the
00:05.16mafkeesyou don't want to address tho whole internet
00:05.24harryvv142 is a little steep for a simple spa 1000 but thats the going rate for a reseller up here in BC canada
00:05.25`Sauronfor my application, there will ever only exist a local subnet
00:05.43`Sauronharryv: you can get spa1k1's for less than that off the internet
00:05.51mafkeeswhy stick to level 2 ?
00:05.58`SauronBecause I want to? :)
00:06.00mafkeeslol
00:06.03harryvvyes...but there is the issue with gst/pst taxes
00:06.18`Sauronmafkees: It's part of the design requirement, as weird as that sounds.
00:06.27mafkeesbut if ur net is limited, the broadcast address is known
00:06.32*** join/#asterisk sjaak538 (~sjaaknabu@d5c53145.dsl.concepts.nl)
00:06.46harryvvand anyway i dont want to wait. Wife is complaining about the IVR playing when she picks up the analog phone and cannot stop it.
00:06.49ovidiu_25only for l3
00:07.09mikegrbharryvv: so put asterisk in line with all the phones
00:07.16mikegrbone fxo, one fxs
00:07.22mikegrbasterisk wasn't designed to share the line
00:07.25`Sauronmafkees: I'm trying to broadcast data between wireless devices - and want to do as little configuration as possible
00:07.37harryvvmike, i know
00:07.45mafkeeswhat kind of data ?
00:07.52`Saurondoesn't matter
00:08.02ovidiu_25but wireless use diferent protocol
00:08.24harryvvto bad there was a dtmf code to shut off the ivr is my wife does not pickup the phone in time.
00:08.25mafkeeslayer 2 is the same, right ?
00:08.33`Sauronyes, it is
00:08.37`Saurononly layer1 is different
00:08.41mafkeesyeah
00:08.44mikegrbharryvv: then add the dtmf code
00:08.48mafkeesthat's what I thought too
00:09.04ovidiu_25yap
00:09.05harryvvmike, thats a idea
00:09.16mikegrbit's as simple as 9,1,hangup
00:09.20mikegrband then she dials 9
00:09.23harryvvanyway im commited to buy this thing :)
00:09.49Beirdomikegrb: I am having fun with this :)  got firefly installed on the work laptop now
00:09.56mikegrb:D
00:10.07mikegrbfirefly is great for a free softphone
00:10.10mikegrband it's iax!
00:10.15Beirdoprecisely
00:10.20Beirdoonly does one call instance though
00:10.28Beirdoand that kinda sucks
00:10.31mikegrbja :/
00:10.39ovidiu_25but the wireless devices all the time make the broadcast
00:10.45BeirdoX-lite has the 3 lines, which is useful, too bad it's SIP
00:11.27moonwickwhat's wrong with SIP?
00:11.39Beirdothree letters
00:11.41BeirdoNAT
00:11.45moonwickah
00:11.49Nuggetthat sounds like something wrong with NAT.
00:11.50ovidiu_25all the clients near an AP are MAC known
00:11.57moonwickseems okay for me, behind nat
00:12.03mafkeesever tried to set it up between a server behind NAT and a client behind nat ?
00:12.04moonwickbut my server's on a public IP
00:12.07Beirdono, SIP is a NAT-unfriendly protocol by nature
00:12.18moonwickservers do not belong behind NAT.
00:12.20Nuggetputting a server behind nat is just dumb.  that's not SIP's fault.
00:12.21Beirdoyou can hack it into behaving, but it wasn't designed for it
00:12.27`Saurondum di dum
00:12.30Nuggetnat makes all sorts of things break.
00:12.47Beirdonot many things I have problems with with NAT
00:12.48NuggetNAT is a protocol-unfriendly protocol.
00:13.00greg_worki thought there were problems even if just your client was behind nat?
00:13.04Beirdoand I'm not paying to get a second static IP on my DSL, thanks
00:13.05`Sauronmost/many udp-based things break with nat
00:13.07mafkeesNugget: unless you have a state matching module in your firewall
00:13.18moonwicknat isn't really a protocol.  :)
00:13.37greg_workmoonwick: s/really//
00:13.37*** part/#asterisk ovidiu_25 (~dd@80.96.223.40)
00:13.42Beirdothe only thing that breaks for me with nat are protocols that are stupid enough to send the IP address INSIDE the protocol
00:13.44Nuggetthe only thing that sucks more than nat is not having connectivity, so nat beats the alternative in many cases, but nat blows goats any way you look at it.
00:13.47Beirdolike SIP
00:14.07moonwicknugget's sitting pretty on his stable of IPs over there
00:14.11moonwick:P
00:14.15Nugget*shrug*
00:14.21NuggetI'm willing to pay to avoid nat suckage.
00:14.23greg_workNugget: there are benefits to the way nat works. nat is like a condom for windows machines
00:14.28mafkeeswe just need ip v6
00:14.30mafkees:)
00:14.34`Sauronhum di dum
00:14.36Nuggetnat is not a security tool.
00:14.44mafkeesnat sux
00:14.45`SauronI doubt arp does any network requests
00:15.03`Saurondum di dum
00:15.43greg_workNugget: sure it is. the popularity of nat routers for residential users means less vulnerable windows machines on the internet
00:15.59Nuggetonly by accident and not very effectively.
00:16.02Beirdosay what you want, Nugget: at this point, the ONLY thing I have NAT issues with is SIP.  everything else works fine
00:16.24Beirdoeven FTP (with a helper) and H323
00:16.29greg_workgranted, it is simply making up for the absolute lack of even basic security that windows and most windows applications have ..
00:17.01mafkeesnot only windows
00:18.27Beirdothere are valid reasons for using NAT, and valid reasons not to.
00:18.56mafkeesindeed
00:19.24*** join/#asterisk dsfr (~dsfr@216.207.244.183)
00:19.47Beirdobut as it *is* a fact of life that NAT will be around until IPv4 dies, designing protocols to be NAT-unfriendly is boneheaded.
00:20.00mafkeesas long as ipv6 is not supported by 99% of all the services Out There (tm) nat is the only way for home users to have mone then 1 networked device hooked up to the internet
00:20.29Beirdowell, without paying loads of money to their ISP
00:20.45SexyKenAsterisk can process PHP files?
00:21.08mafkeesSexyKen: yes, as agi scripts
00:21.47mafkeeswe use it here to lookup caller id info on ISDN BRI channels
00:22.39mafkeesexten => our_phone_nr_1,1,agi(lookup.agi)
00:22.53mafkeesthat lookup.agi as a cli php script
00:22.58greg_workwhat makes you guys think ipv6 is going to solve the NAT problem?
00:23.13`Saurondum di dum
00:23.30SexyKen•mafkees• So do I need actual PHP installed or no?
00:23.31Sedoroxyou don't need nat with IPv6.. every machine has a unique addy
00:23.33mafkeesgreg_work: cause every cable/dsl connection will have 64k routable ip addies
00:23.44mafkeesSexyKen: yes
00:23.46greg_workspecifically, what makes you think that ISPs will suddenly start giving out multiple addresses for free, when it's a fee-per-IP service now?
00:23.57Beirdogreg_work: IPv6 tunnelling
00:24.05SexyKen•mafkees• So once php is installed it'll work out of the box?
00:24.08Sedoroxgreendisease: fee's for ipv6? ahah
00:24.10greg_workBeirdo: to where?
00:24.16mafkeesno, tunneling is as bad as NAT
00:24.25Sedoroxno it isn't
00:24.27mafkeesSexyKen: yes
00:24.42greg_worki mean, its nice to think that because its technically possible and numerically feasable, they'd do it
00:24.46Sedoroxtunneling with IPv6 has just about, sometimes better, latency then native ipv4
00:24.52greg_workbut i highly doubt they'd give up a possible revenue stream
00:25.05greg_workwhy don't telco's stop charging for LD costs?
00:25.20Beirdowell, the reason you pay for extra IPs now is because they are a relatively rare commodity
00:25.21greg_workpratically, theres really not much difference anymore between a local and LD call on the PSTN
00:25.33greg_workyet, LD calls from ma bell on a POTS line are expensive
00:25.41*** join/#asterisk neopher (~crazy@mail.techhelpresources.com)
00:25.49mafkeesgreg_work: here in .nl they already have dsl connections that give you 1 ipv4 addy and a whole /64 ipv6 net
00:25.52Beirdoright, and we get around that by not getting service from them
00:26.17greg_workmafkees: ok, well, thats a selling point. once ipv6 is mainstream i doubt you'd see that happen as much.
00:26.20Beirdosame thing goes with the ISP, you can tunnel your IPv6 to somewhere that will route for you
00:26.29neopher<---- banging head against server
00:26.37greg_worki may be wrong. ISPs doing that now may mean everyone does it.. depends on what gets establish i guess
00:26.51Beirdotrue
00:26.55mafkeesuhhuh
00:27.01greg_worki'm just saying, most ISPs are used to charging more for extra IPs, I don't see any reason why they'd suddenly decide to give out extra IPs for free
00:27.10Beirdobecause they can
00:27.18Beirdoand because it will get them customers
00:27.23mafkeesbut you can already get ur ipv6 /64 net for free @ 6bone
00:27.25neopheranyone here get a cisco 30VIP to work with chan_sccp?
00:27.26Sedoroxthey only charge because it costs them
00:27.27greg_workif it will get them customers - sure
00:27.34greg_workbut "Because they can" is NOT a business case
00:27.39Nuggetheh
00:27.40BeirdoSedorox has hit it on the head
00:28.17NuggetWhen I moved froma  /29 to a /28 it was only a one-time charge.  I was amazed to hear that it wouldn't increase my monthly costs.
00:28.18mafkeesneopher: I had some bad experience with chan_sccp
00:28.29Beirdogreg_work: if a SINGLE ISP in your area offered free IPv6 net and the others didn't, do you not think that would give them a lot more business?
00:28.52greg_worki can get additional IPs at my colo provider for free too, as long as i justify them
00:28.56mafkeesI would go for the one with free ipv6 net for sure
00:28.57NuggetI'd love to be able to get legitimate ipv6 space, though.
00:29.08Beirdomafkees: so would I.  there is the business case
00:29.14greg_workBeirdo: right now? no, i don't care about ipv6
00:29.15mafkeesgreg_work: colo providers are different
00:29.16neopheryou still have to tunnel that IPv^ trough you isp and the isp must support it as well as your router
00:29.20Beirdoby giving them away, they can get more customers
00:29.22harryvvexten => 91,1,Hangup stops the IVR but does not hangup the zap
00:29.26mafkeesI was talking about house dsl lines
00:29.28*** join/#asterisk sezuan (sezuan@port-212-202-57-119.dynamic.qsc.de)
00:29.33harryvvohh wait i think i can fix that
00:29.40Beirdothen once one does, the others will follow suit so they don't lose customers
00:29.46greg_workmafkees: i know, i was intending that for Nugget
00:29.58mafkeesneopher: here in .nl there are providers that have native ipv6 stacks already
00:30.14greg_workBeirdo: yeah, but like i said, it depends on who gets in at what time
00:30.40greg_workBeirdo: if the aol's and whoever else is a big isp these days decides to start charging, other isps might follow suit
00:30.44Beirdotrue, but ultimately, IPv6 will likely be the only thing that can or will eradicate NAT
00:30.52Beirdonah
00:31.01dsmouseeven that won't
00:31.01greg_work"hey look, that isp is making money off something we give away for free"
00:31.10*** join/#asterisk yaboo (~jsirucka@220.245.131.131)
00:31.12Beirdothey will lose so many customers they will have no choice
00:31.35mafkeesindeed
00:31.37greg_workmaybe
00:31.37neopherwow, well your ahead of my isp, they do not distribut ipv6, have to get a block from HE (hurricain Electric) and then route it towards my ipv4 address
00:31.47dsmouseit'll solve it there, but a lot of companies do it for security too
00:31.47Beirdocustomers go where they get more service for less money
00:31.51dsmouseit's a bad idea, but...
00:31.54Nuggetwith ipv6 there's really only one good reason remaining for nat -- isp independence, it makes swapping providers less painful because it allows you to avoid internal renumbering.
00:32.00greg_workif you can only get free additional addresses from hte fly-by-night ISP that goes down all the time and only has 9/5 tech support ......
00:32.07Beirdosigh
00:32.12greg_worki'd be willing to pay the additional fee.
00:32.17Nuggetbut I'm confident that we'll be able to correct that.
00:32.31BeirdoNAT for security is *not* a bad idea
00:32.33*** join/#asterisk yurpls (~yurplsl@65.114.0.198)
00:32.35Nuggetyes it is.
00:32.39Beirdohow so?
00:32.45mafkeesit is
00:32.48greg_workNugget: you can still run an internal network duplicated over top
00:32.53Nuggetbecause it's based on the false premise that all attackers are "out there"
00:32.54Sedoroxits security through obsecurity.. which is a bad idea
00:33.06Nuggetwhich more often than not is incorrect.
00:33.07BeirdoSedorox: partially
00:33.15mafkeesand it's not even obscured
00:33.17Nuggetand because nat isn't really a security mechanism.
00:33.25greg_workNugget: its certainly not the be-all-end-all of security.. but it does protect against that thousands of attackers that are on the internet
00:33.30mafkeesjust have a peek at the tcp window scaling option
00:33.36Beirdoby port forwarding your incoming ports, you secure those ports on all the other machines
00:33.40mafkeesand you know what os is in there
00:34.03Nuggetfor sufficiently laughable values of "protect"
00:34.14Beirdothat's not by obscurity, that's by proper setup
00:34.17greg_workNugget: with windows, theres no good firewall. (i dunno, maybe with xp? i still use 2k, no compelling reasons to switch)
00:34.29Nuggetso run a firewall.
00:34.47greg_workoh, you mean ,like a NAT router that doubles as a firewall because that's the way NAT works?
00:34.54Nuggetno, I mean like a firewall.
00:34.56mafkeeswith windows, put a bridge between the internet routing hardware and your windows box
00:34.56BeirdoNugget: in case you were unaware, most people who do NAT do it on a firewall
00:34.58Nuggetthe nat is unrelated.
00:35.06welbyhave an actual firewall running as well
00:35.11Nuggetsure, there are devices that are firewalls and nat routers.
00:35.12SexyKenInstead of dialing one call at a time in a follow me sequence, how can I dial all phones at the same exact time?
00:35.19Nuggetbut that doesn't make nat a component of the firewall.
00:35.27welbythe firewall may be the same package / software (like in pf  or iptables)
00:35.31SedoroxSexyKen: queue...
00:35.33Sedoroxringall
00:35.33dsmouseSexyKen: and just have one pick up?
00:35.33welbybut its still a different thing
00:35.34Beirdoif you are dumb enough to use NAT as the only security measure, you deserve what you get
00:35.46SexyKen•dsmouse• Yes.
00:35.47greg_workNugget: find a firewall applicance near the price of a home/soho-targetted "router"
00:35.56Nuggetnat is not a security measure.  it's just a hacky routing trick.
00:35.57SexyKen•dsmouse• All would be cool as well.
00:35.58Beirdobut saying using NAT as part of your security is bad is just wrong
00:36.01welbygreg_work: easy, wrt54g
00:36.10dsmouseSexyKen: queue... ringall
00:36.11SexyKendsmouse, Here is current code: http://pastebin.ca/6095
00:36.19greg_workwelby: that is a nat/firewall/router .. whatever you want to call it.
00:36.21greg_workit does NAT
00:36.29welbyyes
00:36.32Sedoroxbah
00:36.33*** part/#asterisk Sedorox (brandon@Neptune.client.wlmsprt.pa.sed6.net)
00:36.38SexyKendsmouse, I'd have to set it up as a queue?
00:36.48*** join/#asterisk zotz (~zotz@24.231.32.191)
00:36.48dsmouseI *think* so
00:36.59SexyKenCan I have it ring all at once and have them all connect?
00:37.01mafkeesnat = Network Address Translation
00:37.02greg_workNugget: yes, i'll give you that .. it IS a hacky routing trick. but it adds some security
00:37.05mafkeesthat's all
00:37.17dsmouseSexyKen: not unless you interface it with a conference somehow
00:37.18welbyhowever, even with the default firmware (ie linksys one) it does proper firewalling and natting
00:37.30dsmouseI wonder how that could work
00:37.34Beirdoeither way, my whole point was that NAT is a necessity these days due to the way IPv4 works, and SIP sucks because they intentionally ignored that fact.
00:37.41greg_workhehe
00:37.45greg_workyeah that got a bit OT :)
00:37.58mafkeesBeirdo: and that is true ;)
00:38.10jetsDoes * use tcp or udp generally for sip
00:38.18BeirdoUDP
00:38.19greg_workwhy DID they ignore that, anyways?
00:38.19mafkeesudp
00:38.20dsmouseudp always
00:38.33Beirdogreg_work: no idea, but it is a PITA
00:38.38greg_workis there a reason, or is it "just the way it works(tm)" ?
00:38.42SexyKenHow doe sone make astersik play back a number
00:38.56Beirdothey put the IP addresses inside the SIP packets
00:39.00Beirdoheheh
00:39.14mafkeesthat is bad anyways
00:39.26mafkeesonly the IP header should have that
00:39.29Beirdonow IPSEC I can forgive for being intentionally NAT unfriendly
00:39.38Beirdoit's an encrypted tunnel
00:39.45BeirdoSIP - there's no excuse
00:39.48mafkeesIPSEC is NOT nat unfriendly
00:39.56tzangermafkees: actually yes it is
00:40.04tzangeryou need to use nat-t to get through it
00:40.05mafkeesdepends on the ipsec implementation
00:40.05Beirdomafkees: quite so
00:40.20tzangerI mean think about it -- AH ensures that the headers weren't fucked with and that's exactly what NAT does
00:40.35mafkeesit only needs 1 rdr rule here
00:40.48Nuggetthis whole issue just feels to me like the people that complain about licenses that are not compatible with the gpl.  Lots of stuff is hard to accomodate with nat, just as many licenses can't coexist with the gpl.  but in both cases it's the fault of nat and of the gpl that they're so unaccomodating.  :)
00:40.56mafkeesthe rest is just plane state related stuff
00:41.19Beirdohow the hell is NAT unaccomodating?
00:41.28mafkeesgpl is evil anyways
00:41.32Beirdoall it is is remapping IP addresses in the damned IP header
00:41.39Nuggetbecause it turns the bidirectional nature of the internet into a fudamentally client-server model.
00:41.43tzangerBeirdo: it doesn't fix any addresses in the data payload
00:41.47Nuggetanything that needs or wants to go both directions has problems.
00:42.07Beirdotzanger: there should *BE* no addresses in the data payload, that's what the headers are for
00:42.22mafkeesI agree Beirdo
00:42.34mafkeesData != routing info
00:42.47tzangermaksim: I agree but that is what makes NAT unaccomdating :-)
00:42.56BeirdoNugget: a good NAT implementation has no bidirectional issues
00:42.57tzangerBeirdo: agree 100%
00:43.03tzangerBeirdo: bullshit
00:43.16NuggetBeirdo: well then, if that's truly the case then it's absolutely not adding any security either.  :)
00:43.17tzangera packet comes in to your NAT box, port 8342
00:43.18SexyKenIs there anyway to make just a certain extension use a different hold file
00:43.22neopheri use nat and have no probs with sip, even remotely
00:43.23tzangernow which of your 100 "clients" is it for?
00:43.46Beirdotzanger: it is for the one that sent to that socket, or it gets bounced
00:43.47mafkeestzanger: there should be a state for it
00:43.49tzangerNAT by its very design has issues with bidirectional transfers
00:43.52Beirdoor you port forward it
00:43.55tzangerBeirdo: as in client-server
00:43.58tzangerI had to make the request first
00:44.14mafkeesof course
00:44.17Beirdonot if you port-forward the incoming ports you *want*
00:44.19Beirdoduh
00:44.21mafkeesbut that's not only with nat
00:44.27mafkeesthat's with all firewalling
00:44.29SexyKenIs there anyway to make just a certain extension use a different hold file
00:44.33AhewesGot a problem with far end disconnect on an adtran ta750
00:44.40mafkeesall protocols that use dynamic ports
00:45.01tzangerBeirdo: but now you're patching up your NAT
00:45.12Beirdotzanger: no I'm not
00:45.13tzangerBeirdo: and again, you can't have two clients share the same port
00:45.17tzangersure you are
00:45.19bkw_http://bkw.digiweb.com/conf.gs
00:45.22bkw_http://bkw.digiweb.com/conf.gsm
00:45.24tzangeryou're explicitly telling your NAT box what to do
00:45.26bkw_for the dev conf recording
00:45.28mafkeesif you close everything and only allow ftp-data, how will you handle passive ftp ?
00:45.29bkw_just in case anyone wants it
00:45.36tzangerthanks bkw_
00:45.37mafkeesever on your border firewall ??
00:45.38mafkeesnot
00:45.48bkw_cant' pasive ftp work on just one port?
00:45.53bkw_wasn't that the point of pasv ftp?
00:46.05*** join/#asterisk vooduhal (~christoph@pcp01069659pcs.andrsn01.tn.comcast.net)
00:46.18Beirdomafkees: passive FTP opens the port from the client to the server, that's how
00:46.27mafkeesbkw_: it supports multiple ftp clients shoring the same public ip
00:46.32Beirdoactive FTP opens the data port from the server to the client
00:46.40mafkees01:46 < Beirdo> mafkees: passive FTP opens the port from the client to the server, that's how
00:46.43mafkeesindeed
00:46.50mafkeesso if the server denies that
00:47.03Beirdowhy would it be denying it?
00:47.03mafkees(as it should, cause it's a high port)
00:47.08bkw_no isn't pasv ftp let the client tell you what port ot use?
00:47.16bkw_not the otherway around
00:47.19bkw_so you can TELL it what port to use
00:47.26Beirdoand who the hell runs an FTP server behind NAT?
00:47.35mafkeesa lot of ppl
00:47.36Luke-JrActive FTP -- Server connects to client on client-specified host and port
00:47.49Luke-JrPassive FTP -- Client connects to server on server-specified port
00:47.51bkw_doesn't make sense the other way
00:47.57neopherwow, when did they integrate ftp with VoIP, hehe
00:48.07bkw_either way FTP sucks
00:48.09sezuanWhen I receive a call through a SIP/PSTN gateway (register => login:pass@gw), do I have access to the called number?
00:48.10mafkeesneopher: tftp
00:48.11Luke-Jrneopher: no idea
00:48.14mafkees;)
00:48.16Luke-Jrbkw_: agreed. I hate FTP
00:48.39sezuanExcept SIPGetHeader
00:48.51mafkeesLuke-Jr: but what's the alternative ?
00:48.52Luke-Jrsezuan: some providers use it instead of 's' for a starting ext
00:48.54Nuggetsure, but ftp sucks for reasons which are unrelated to the difficulty of using ftp in a nat environment.  :)
00:48.55BeirdoFTP is another NAT-unfriendly protocol, but it was invented way before NAT was thought of
00:48.58Luke-Jrmafkees: SSH
00:49.06mafkeesyeah right
00:49.09bkw_Luke-Jr you have it backwards
00:49.14Nuggetmafkees: what's wrong with that?
00:49.14bkw_pasv ftp the client says what ports
00:49.25mafkeesoverhead ?
00:49.27dsmousemafkees: http?
00:49.28Nuggetssh is a perfectly usable and dramatically superior alternative to ftp.
00:49.29Luke-Jrbkw_: Nope
00:49.38bkw_yes
00:49.43bkw_PASV the client sets it up
00:49.43sezuanLuke-Jr: I should be able to use directly in the extensions.conf?
00:49.43BeirdoNugget: ain't that the truth
00:49.44Nuggetfor many reasons.
00:49.45Luke-Jrmafkees: What overhead?
00:49.47mafkeesNugget: uhhuh, if you have enuf bandwidth
00:49.54Nuggetbandwidth?
00:49.55Luke-Jrsezuan: sure... set a global var to the dialed #
00:49.58mafkeesLuke-Jr: all the encryption
00:50.04NuggetI think you have an imperfect grasp of the issues.
00:50.05bkw_Passive FTP (sometimes referred to as PASV FTP because it involves the FTP PASV command) is a more secure form of data transfer in which the flow of data is set up and initiated by the File Transfer Program (FTP) client rather than by the FTP server program.
00:50.10Luke-Jrmafkees: doesn't use much CPU, in my experience
00:50.19mafkeesnot much CPU
00:50.31mafkeesbut a LOT of tcp traffic
00:50.31bkw_pasv = client side
00:50.34bkw_NEXT!!!
00:50.40Luke-Jrbkw_: the SYN comes from the client
00:50.44sezuanLuke-Jr: How do I access the dialed number?
00:50.45Luke-Jrbkw_: not the port/host info
00:50.56bkw_dumb ass don't argue with me
00:51.05Qwellbkw_: You are SO wrong.
00:51.05Luke-Jrsezuan: ${EXTEN} if they use it for your exten
00:51.10QwellDon't ban me :P
00:51.11bkw_no i'm not
00:51.12*** join/#asterisk FaithX (~Faithful@202-6-145-116.ip.adam.com.au)
00:51.13bkw_i'm reading it
00:51.21Qwelldunno, I just got here
00:51.24Luke-Jrbkw_: whatever. that's what actually happens, in my experience
00:51.29mafkeesa ftp link (without encryption) is around 30 to 60% faster than scp
00:51.32bkw_riiight
00:51.37Beirdomafkees
00:51.39Luke-Jrmafkees: over what connection?
00:51.42SexyKenCan I choose the music on hold to use for a certain extension???
00:51.42Beirdoit shoudn't be
00:51.46mafkeesdsl 8 mbit
00:51.50bkw_Luke-Jr you're dyslexic
00:51.58Luke-Jrbkw_: ...
00:52.00QwellSexyKen: like MusicOnHold(1234)?
00:52.09Qwellmusiconhold.conf, or whatever.
00:52.16sezuanLuke-Jr: EXTEN is not the name/number in front of the @ in the To: header?
00:52.17bkw_luke you're backwards
00:52.20mafkeesSexyKen: yes: SetMusicOnHold
00:52.28brc_Luke-Jr, wrong
00:52.31SexyKen•mafkees• So how would I set it ?
00:52.33Silik0nsomeone doesnt understand pasv mode obviously
00:52.48Silik0nPASV is needed because NAT breaks active FTP
00:52.56Qwellrtfrfc?
00:53.07bkw_http://searchnetworking.techtarget.com/sDefinition/0,,sid7_gci512897,00.html
00:53.18dsmouseno, pasv is needed cause people still use ftp
00:53.19Nuggetmafkees: so turn off encryption if it bugs you so much.
00:53.28mafkeesSexyKen: on your asterisk console type: show application setmusiconhold
00:53.29Nuggetyou still get the other benefits of scp/sftp.
00:53.43vooduhalexit
00:54.02mafkeesNugget: no, it does't bug me, it's just the downside of using SSH
00:54.03Silik0nunless you have a active proxy that fixes the FTP protocol messages active mode does not work because the server will try to open aditioonal ports the nat box and the nat box will not know what to do with them
00:54.13Nuggetmafkees: how can it be a downside if it can be disabled?
00:54.18Nuggetit's just a factor.
00:54.19Silik0nthus the whole idea behind pasv where the client controls eeverything
00:54.33Beirdohttp://www.faqs.org/rfcs/rfc1579.html
00:54.40Luke-JrI can prove it :)
00:54.43Beirdomaybe we should all go read the RFC
00:54.45mafkeesNugget: speed is everything
00:54.49QwellBeirdo: Thats what I already said, heh
00:54.59QwellBeirdo: rtfrfc, ya know? ;]
00:55.08Nuggetmafkees: so turn on ssh's compression (big win over ftp) and disable encryption.
00:55.09Beirdoyup
00:55.10ManxPowerThis E&M Wink problem is killing me.
00:55.12Nuggetsounds like ssh is superior.
00:55.14BeirdoRTRFC1579
00:55.20Nuggetif you think that speed is the only thing that matters.
00:55.23dsmousestill, "mode i" therfore ftp sucks.
00:55.28Silik0nLuke-Jr what just cause the server says hey I can use ports 10000 to 11000 to the client doesnt mean the server controls it... all comms in PASV mode originate from the client NOT from the server
00:55.29bkw_haha
00:55.30mafkeesdisabling encryption ???????
00:55.32mafkeeswtf
00:55.36*** join/#asterisk florz (nobody@odnb-d9baa40a.pool.mediaWays.net)
00:55.52Nuggetyes.  you are aware that you can disable encryption for the scp transfer, right?
00:56.00Nuggetsince that's what you're complaining about.
00:56.00mafkeesuhhuh
00:56.02Luke-Jrhttp://pastebin.ca/6096
00:56.07Beirdoor use a less expensive one :)
00:56.11Luke-Jrftp.gnu.org specifies host and port
00:56.32mafkeesno, I'm saying it will lose from ftp when you look at the speed
00:56.36Luke-Jr199.232.41.7 on port (134 * 256) + 185
00:56.52Luke-Jrmafkees: if you want speed, use TFTP
00:57.12mafkeesLuke-Jr: or a dvd ;)
00:57.12dsmousewhat's quicker, ftp, gopher, or http
00:57.33Luke-Jrdsmouse: HTTP
00:57.39Beirdonone of the above
00:57.41mafkeesdsmouse: gopher
00:57.44florzdsmouse: what's louder, a speaker or a human?
00:57.48Beirdothey are all TCP protocols
00:57.56Silik0nflorz: a stupid human
00:57.59*** part/#asterisk eKo1 (~bernd@207.42.191.66)
00:58.00Luke-JrHTTP is faster only because the initial latency is lower
00:58.01florzSilik0n: =:-)
00:58.05Beirdothe ultimate speed will be determined by your TCP windowing
00:58.06*** join/#asterisk Gator (~krp@adsl-068-209-187-058.sip.gnv.bellsouth.net)
00:58.07Luke-Jrand QoS is generally highest
00:58.16Silik0nhah
00:58.17mafkeesBeirdo: no
00:58.23Beirdoinitial latency has nothing to do with it
00:58.30dsmouseSilik0n: itym a stupid human that just screwed up his computer 20 mintues before a presentation to his biggest client
00:58.37Beirdodownload a 1G file with all three :)
00:58.38Luke-JrBeirdo: they all transfer the data the same way, so initial latency is all that's left
00:58.51Silik0ndsmouse hah
00:59.01Beirdothe initial latency will be in the range on ms on a several hour transfer
00:59.08mafkeesLuke-Jr: no, you have to take the overhead in account
00:59.09Beirdoit's irrelevant
00:59.12Luke-JrBeirdo: in which case, you won't notice any difference
00:59.26Luke-Jrmafkees: there is no overhead diff for HTTP and FTP
00:59.32Luke-Jrthey send the data over TCP the same way
00:59.32mafkeesthere is
00:59.47Beirdothere can be if your TCP windowing is done differently in the apps
00:59.59bkw_ok guys lets move on
00:59.59Luke-JrBeirdo: that's the app then, not the protocol
01:00.01bkw_you're all stupid
01:00.02mafkeesmy tcp windowing is fixed
01:00.04bkw_NEXT!!!
01:00.14bkw_and I can say that because its my birthday
01:00.16mafkeesyeah
01:00.18bkw_:P
01:00.18mafkeesNEXT
01:00.19Beirdobkw_: yeah, enough of this special olympics :)
01:00.21Luke-Jrbkw_: did you look at my proof against you, at least?
01:00.32bkw_Luke-Jr yes dear I did
01:00.34mafkeesNEXT == my naked wife in our bed ;)
01:00.35bkw_but you still don't get it
01:00.37mafkeeslater all
01:00.47Luke-Jrbkw_: what don't I get?
01:00.53dsmouseGAH
01:00.58bkw_Luke-Jr shut up now please
01:01.03BeirdoLuke-Jr: just drop it will ya?
01:01.04Silik0nhah
01:01.10Luke-Jrmeh
01:01.15Beirdosorry we even got into this in the first place
01:01.17Luke-Jrwhy did FTP come up in the first place?
01:01.30bkw_no fuckin clue
01:01.34bkw_but lets solve other issues
01:01.37Beirdoamen
01:01.53Beirdohappy birthday to bkw_
01:02.10Luke-Jrbkw_: so how would I get started doing * devel?
01:02.22bkw_open up app_skel.c
01:03.15*** join/#asterisk dontmsgme (~none@adsl-68-121-22-193.dsl.irvnca.pacbell.net)
01:03.29dontmsgmeHow come with Nufone I bought a DID 3 weeks ago and whenever I call it it says "you are unreachable"
01:03.37Beirdowow, look at the time...
01:03.50greg_workugh, the only thing worse than spam, is spam from people who don't know how to set their computer's clock
01:03.58Beirdoheh
01:04.21greg_workjust got one from 1/18/2001
01:04.22|Vulture|dontmsgme: prolly didn't register it right
01:04.27dontmsgmein iax?
01:04.42|Vulture|dontmsgme: does it show on the * box when you call it at all?
01:04.44greg_worki suppose it IS the 18th today.. but even so
01:04.54dontmsgmeNo
01:05.07bkw_paris hiltons birthday was yesterday
01:05.09bkw_that whore
01:05.14dontmsgmeBut I put the correct Register syntax
01:05.36|Vulture|dontmsgme: do a "iax2 show registry"
01:06.08dontmsgmeIt says State: Registered
01:06.41dsmousedontmsgme: did you do a "iax2 debug" and call the number?
01:06.56|Vulture|strange, if its setup right you should see the # trying to hit your server but not going anywhere if extensions.conf isn't setup right
01:07.25|Vulture|dsmouse:I thought he said there was nothing.. but he might not have been in debug
01:09.27bjohnsonset verbose 3 should show it without the info overload of debug
01:13.06SexyKenWhere is a good place to get sounds for asterisk?
01:13.11SexyKen"HOld up while we connect yo call"
01:13.43SexyKenAlso can background play mp3s?
01:14.19ManxPowerI think a telco problem has finally beaten me after all these years.
01:14.27ManxPowertelco == telcom
01:14.42Qwellbkw_: So, you can say you come after Paris?
01:14.56ManxPowerSexyKen, Did you try /var/lib/asterisk/sounds and the asterisk-sounds CVS?
01:15.07SexyKenI'll check it out.
01:15.13SexyKenCan Playback() play mp3's?
01:15.31SexyKenSorry I mean Background
01:15.31dontmsgmeIt said rejected, no authority found
01:15.48ManxPower<PROTECTED>
01:15.48ManxPower<PROTECTED>
01:15.48ManxPowerFeb 18 19:15:41 WARNING[2844]: chan_zap.c:4723 ss_thread: getdtmf on channel 9: Operation now in progress
01:15.48ManxPower<PROTECTED>
01:16.04ManxPowerSexyKen, no
01:16.21SexyKenIs there a way to play an mp3's file out of no where?
01:16.43ManxPowerSexyKen, "show applications" is your friend.
01:16.45dsmouseMp3Player
01:17.26dontmsgmeFeb 18 17:09:49 NOTICE[-235656272]: chan_iax2.c:5183 socket_read: Rejected connect attempt from 198.22.67.70
01:17.26dontmsgmeTx-Frame Retry[000] -- OSeqno: 000 ISeqno: 001 Type: IAX     Subclass: REJECT
01:17.26dontmsgme<PROTECTED>
01:17.26dontmsgme<PROTECTED>
01:17.36*** join/#asterisk ScythelX (Fleb@pc-24-181-176-153.sbi.ct.charter.com)
01:18.26SexyKendsmouse, How would I answer the call and play an MP3 while they're being connected to the call?
01:18.44ScythelXhello all - there is a company near me thats going out of business that has a shit ton of cisco phones I wanted to take a few of their 7940's off their hands...I've never used the cisco series phones but I was wondering if someone knew if I would be able to use them...ie i dont know if you need a special license key to operate them
01:18.55ManxPowerSexyKen, That is called MusicOnHold.  Do I need to point you to "show applications" and the Wiki?
01:19.14QwellScythelX: How much they selling them for?  heh
01:19.26ScythelXlike 100 bucks with the power cubes
01:19.28SexyKenManx, NO! I just dont know how to have it play while dialing!
01:19.29Qwellwow
01:19.34ManxPowerScythelX, Cisco will make you pay for SIP firmware.
01:19.40QwellNo, you don't need anything special, really
01:19.48QwellSIP firmware isn't a requirement, afiak
01:19.52ManxPowerSexyKen, "show application dial"  Pay special attention to the "m" option.
01:20.11ScythelXi just want them for home use only gonna by 3, but I dont want to hook them up and not be able to use them
01:20.12ManxPowerQwell, You don't need a hemlet when racinh motorcycles either, but it's usually a good idea.
01:20.30QwellManxPower: I'm just commenting.  You're right though
01:20.54*** join/#asterisk IQ (~IQ@70-59-167-66.omah.qwest.net)
01:21.11IQHi, anyone using Avaya gateway with Asterisk?
01:21.27ScythelXManxPower: so if I wanted to upgrade the firmware of the phone I would just have to pay for that?
01:21.37ManxPowerScythelX, Yes.  Welcome to Cisco.
01:21.39Qwellits like $8 per phone or something, isn't it?
01:21.58Qwellat only $100 per phone, thats not bad at all
01:21.58ScythelXManxPower: ok thank you for your input
01:22.03ScythelXQwell: you as well
01:22.04ManxPowerA support contract is $8/yr.  I belive that gives you access to the firmware, but not the legal right to use it.
01:22.08Qwellhell, the phones may even HAVE the SIP images already
01:22.29dontmsgmeDoes vonage ring to your Xlite phone?
01:22.34_VileI saw on the forums today that CDW can hook you up with a contract
01:23.12_Vilelists, rather
01:23.13Qwell_Vile: I heard that people had a difficult time getting them to give you one
01:23.34_VileCDW? I've heard more of a hard time from people getting them from independant vendors
01:24.04Qwelldontmsgme: Vonage has a thing for like $13 a month, that lets you use a softphone.
01:24.08Qwellsomething like that
01:24.19*** join/#asterisk mindCrime (~mindCrime@rrcs-24-106-188-6.se.biz.rr.com)
01:25.01ManxPowerI'm getting "chan_zap.c:4723 ss_thread: getdtmf on channel 10: Operation now in progress" on incoming calls.  Anyone have suggestions on how to fix that?
01:25.02QwellManxPower: So what gives you the legal right to use the firmware?
01:25.07_Vilebkw u wanna tweak the code to have newest messages play first in voicemail rather than oldest?
01:25.13ManxPowerQwell, US$130 I think.
01:25.19Qwellone time, or yearly?
01:27.34ManxPowerQwell, one time.
01:27.40ManxPowerYou are buying the software.
01:27.49Qwellahh
01:27.53ManxPowerOK, I posted a $100 bounty/bribe to get my problem fixed.
01:27.56*** join/#asterisk jsolares (~jsolares@200.12.44.18)
01:27.58Qwell$130 each time you upgrade too, or?
01:28.28ManxPowerQwell, Yes, unless you have a $8/yr support contract.  Gives you the right to .point upgrasdes.
01:28.33ManxPowerCall Cisco and ask them
01:28.45QwellI don't even have an IP phone, heh
01:28.47Qwelljust curious
01:31.16ariel_ManxPower, that is a strange statement you have thre about the getdtmf.
01:31.28ariel_what were you trying to do?
01:32.46ManxPowerariel_, Trying to get incoming DID calls on T-1 E8M Wink channels
01:32.49ariel_ManxPower, reason is I saw that once when were setup a pulse dial phone on a channel bank.
01:33.14ManxPowerWe confirmed that the telco is sending us DTMF.  The think is that about %50 of the calls actually DO work.
01:34.30BeirdoManxPower: go for both :)
01:34.47ManxPowerBeirdo, I already know that is a bad combination
01:35.07Beirdoheh
01:35.22ariel_ManxPower, have you tried the old style e&M wink from adtrans featd
01:35.24BeirdoI guess that would depend on intoxicant, but likely so
01:36.05ManxPowerariel_, Yes.  Console said something like "got something that doesn't belong in FeatGroup D, assuming E&M Wink
01:36.20ariel_ah I have seen that one.
01:36.22ManxPowerA suprizingly usefull message, actually.
01:36.58rikstashit there's been another quake in indonesia
01:36.59ManxPowerariel_, Based on the symptoms I assume it's actually a wink or dtmf timeing problem
01:37.00ariel_Is this going directly to the telco or an older pbx?
01:37.04riksta6.9 on the richter
01:37.15ariel_actually I think it's also a timing issue
01:37.58ManxPowerariel_, Telco -> Asterisk -> channel bank -> (analog) -> PBX.  We are only having problems with the EM/W channels (there are a bunch of loopstart channels too that are non-DID and work just fine.
01:38.00ariel_I had a similar problem with acepting dtmf on a Mitel which we had to upgrade there T1 to a PRI to get it working.
01:38.14*** part/#asterisk jpablo (~jpablo@host-148-244-137-95.block.alestra.net.mx)
01:39.23ManxPowerariel_, I didn't really need to hear that. 8-)
01:39.36ariel_ManxPower, you know me I don't lie.
01:39.50ManxPowerThe goal was to not make any changes on the T-1 or channel bank or the PBX for at least 30 days after getting it working.
01:40.03ManxPowerThen we can look into doing interesting stuff.
01:40.19ManxPowerThe problem, of course, is that if I can't get it working by monday 6am we'll have to take asterisk out of the path.
01:40.56ariel_ah I see. So the problem is mainly between the asterisk and the c/b or to the telco?
01:41.25ManxPowerThe problem is ONLY between the Telco and Asterisk
01:41.40ariel_hummm which telco?
01:42.09SexyKenAnyway to convert wav to gsm?
01:42.17ManxPowerI=55 Telecom, the ILEC is BellSouth.  However, everything work fine if we take the T-1 and plug it directly into the channel bank
01:42.46ariel_change the timing to the c/b as 1 and the t1 to 2.
01:43.12ManxPowerariel_, Huh?
01:43.19ariel_B/S sometimes does not sync timing correctly.
01:43.52ariel_your c/b is connected to the asterisk via a t1 correct?
01:43.55ManxPowerariel_, All the other 44 channels on the T-1 (all non-E&M) work fine. 8-)
01:43.57ManxPowerYes.
01:44.06tzangerok
01:44.16tzangerhow the fuck do you "accidentally" step into the path of a freight train?
01:44.29Qwelltzanger: I've seen it
01:44.31ManxPowerspan=1,1,0,esf,b8zs
01:44.31ManxPowerspan=2,2,0,esf,b8zs
01:44.31ManxPowerspan=3,0,0,esf,b8zs
01:44.31ManxPowerspan=4,0,0,esf,b8zs
01:44.42tzangerhttp://toronto.cbc.ca/regional/servlet/View?filename=to-train20050218
01:44.42ManxPowerspan 1 and 2 are coming from the telco, span 3 and 4 are going to the channel bank
01:44.43Beirdotzanger: it's called a walkman and headphones
01:44.44tzangeryeah so have they
01:44.47`SauronMmm.
01:44.52`Sauron<3 signaling
01:45.27ManxPowerspan 1, channels 9 - 12 are the E&M channels
01:46.05ariel_change the timing on span 1 to 2 and span 2 to 1.
01:47.10ManxPowerariel_, done.  restarting
01:47.41ManxPowerariel_, no difference
01:47.57ManxPowerzttool shows syncing from span 2
01:48.10ManxPowerspan2 is also coming from the telco
01:48.19*** join/#asterisk jetscreamer (~jetscream@adsl-64-219-216-41.dsl.hstntx.swbell.net)
01:48.36ariel_can you get to the configuration on the c/b
01:48.51ariel_see what there set to for 9 - 12
01:48.58ManxPowerariel_, no.
01:49.48ChujiManxPower : my Bellsout t1 does wink with no problem
01:49.59ChujiManxPower : I didn't have to do anything special
01:49.59ManxPowerariel_, Since the channel bank can no longer be managed via FDL by the telco.....
01:50.18ManxPowerChuji, wanna swap T-1s? 8-)
01:50.36ChujiHah, not real bad
01:50.54Chujibut, fwiw, e&m wink is fine
01:51.14ChujiI saw your post earlier on -users and people were saying it couldn't be done
01:51.39ariel_ManxPower, what c/b are they?
01:51.49ManxPowerariel_, Adtran I believe.
01:52.05ariel_I guess your doing this remotely?
01:52.24ManxPowerariel_, Yes, but in theory I can be on-site at some point.
01:52.49ManxPowerI was onsite last week, but we had....communications...issues with the people that manage the PBX.
01:53.19ManxPowerMy mind says "850" but I don't know if that's correct.
01:54.02ManxPowerI would rather just have our CLEC convert all the channels we deal with to PRI 8-)
01:54.02ariel_750/850 to work correctly with asterisk newer builds needs firmware 36 or above.
01:54.19ManxPowerBut we don't have any available channels for a D-channel.
01:54.37ManxPowerariel_, yes, but the problem is telco -> Asterisk, not Asterisk -> channel bank.
01:54.44rikstaanyone here using ADM?
01:54.57ManxPowerI can make DID calls via the channel bank all day without problems as long as the call comes in via VoIP.
01:55.31ariel_next put one line directly to the c/b with the problem and rest on the asterisk...
01:55.59ManxPowerariel_, That's what we did to make things work last week.
01:56.07ManxPowerIt works fine without asterisk there.
01:56.14ariel_have you spoken with mark about this?
01:56.19ManxPowerariel_, nope.
01:56.34ManxPowerI can't PROVE it's a bug and I know that I get bitchy when people bother me.
01:58.16ManxPowerI can imagine mark gets a zillion requests a day for help.
01:58.30ChujiMark would help you though man
01:58.39ChujiYou give back to community more than most
01:58.55ManxPowerariel_, I just /msg'd him.  He's away.
01:58.57ariel_this is one that he just might help with. Or maybe get one of the other guys to talk to him.
01:59.14ariel_He loves these type of problems.
01:59.17rikstaout of interest, what is mark's nick?
01:59.23ariel_kram
01:59.25rikstakool
01:59.32rikstaobvious :)
01:59.37ManxPowerI'm becoming rapidsly less than sober at the moment, however.
01:59.39Chuji~kram
01:59.40jbotLooking for the elusive BishopChicken.
02:00.03ariel_ManxPower, Martin P. is pretty good with T1's as well.
02:00.05ChujiManxPower : Nobody around there has a T-Berd?
02:00.26ChujiCorydon too, if you can track him down
02:00.57SexyKenIs there anyway to change this: exten => _1XXXXXXXXXX,1,Dial(${CZ}/${EXTEN},60,H|g) so it automatically appends the one if it isn't entered
02:01.26ManxPowerariel_, exten => _XXXXXXXXXX,1,Goto(1${EXTEN},1)
02:01.33ManxPower..er..that was for SexyKen
02:04.05ariel_ManxPower, just a strange setup but have you tried the sf_w for inband signal for the ports your having the problem with?
02:04.08ManxPowerariel_, the problem MAY happen more often or less often depending on the digits they send us.
02:04.18ManxPowerariel_, I can try
02:05.25ariel_looking at my notes I had to use that on a nortel with strange dtmf only 50% would work.
02:05.52ChujiWe have to use feature group D with our Toshiba
02:06.13ChujiIt sends ANI and DNIS as *xxxxxxxx*xxxx*
02:06.25Chujibut it still winks
02:07.03*** join/#asterisk DHuang (~DHuang@203.49.132.48)
02:07.19DHuanghi
02:07.22ManxPowerariel_, Asterisk wants sf to be setup in /etc/zaptel.conf too and the config for that looks bizzare.
02:08.23DHuangI've put load => app_prepaid_auth_cid.so  in the modules.conf but it's not showing up in the CLI > show modules ???
02:08.30ManxPowerariel_, But the nortel accepts DID calls just fine.
02:08.41ManxPowerariel_, Thanks.
02:08.55SexyKenThanks Manx.
02:09.23DHuang:-)
02:09.42DHuangany ideas?
02:10.12ariel_DHuang, did you restart asterisk
02:10.25DHuangneed to restart or reload?
02:10.45ariel_if you did not load the model it will not work
02:11.28ariel_you should be able to load it at the cli
02:11.46DHuangariel: thanks.. checking now
02:12.42dontmsgmeIs there any kind of expensive computer monitor
02:12.49dontmsgmeon a laptop
02:12.55dontmsgmeWhere you wont get sick of looking at it
02:13.07ariel_rofl
02:13.13DHuangplasma?
02:14.09modulus_jbot g-g-g-g?
02:14.10jbotG-UNIT!
02:14.11dontmsgmeIs there a plasma computer monitor?
02:14.14modulus_jbot g-unit?
02:14.15jbotg-unit stands for "Guerilla Unit". It's members are Tony Yayo, Lloyd Banks, Young Buck, and the leader 50 Cent. Their official DJ is DJ Whoo Kid. Also see http://www.g-unitsoldier.com/
02:14.17DHuangariel: Thanks.. silly me, not restarting....
02:15.50ariel_DHuang, np
02:15.59*** join/#asterisk jetx (~jetx@adsl-64-219-216-41.dsl.hstntx.swbell.net)
02:16.08*** join/#asterisk cbachman (~cbachman@victory.ece.northwestern.edu)
02:16.45fearnorwww.50shekel.com
02:16.50fearnorword up to j-unit
02:17.48modulus_jew-unit
02:17.50modulus_lol
02:17.55fearnoryes jew-unit!
02:18.09fearnorand lil' jap
02:18.54rikstajbot: g-g-g-g
02:18.55jbotG-UNIT!
02:19.12riksta:)
02:21.14SexyKenexten => _XXXXXXXXXXX,1,Dial(${CZ}/1${EXTEN},60,H|g)
02:21.19brc_yo yo yo!
02:21.21SexyKenthis doesn't automatically append a one does it?
02:21.46QwellSexyKen: that works for me here
02:22.05QwellI send my calls through the pstn like that, prepending a 9
02:22.15SexyKenhm
02:22.30SexyKenSo that should work if I dial just 6507841022?
02:22.54jsolaresit dails ${CZ}/16507841022
02:23.00Qwellshould
02:23.19*** join/#asterisk Brixius (Brixius@c-24-118-4-197.mn.client2.attbi.com)
02:23.39BrixiusHello
02:24.41SexyKenIs there anyway to make it so that a #asterisk moves to the next step
02:26.15ariel_ManxPower, I don't have access to the old server. I just emailed there IS person to get me the settings. I got him the job there so he should send the info to me.
02:26.33IQHi, any windows SIP soft phone beside x-lite?
02:26.36*** join/#asterisk _Raptor_ (~RaptorBlu@pD9E5AAAB.dip.t-dialin.net)
02:26.38_Raptor_hi
02:27.02ManxPowerariel_, Thanks.
02:27.15ariel_IQ, yes but why
02:27.29_Raptor_another question: i only get as last message:
02:27.37_Raptor_<PROTECTED>
02:27.37_Raptor_<PROTECTED>
02:27.37_Raptor_[1]WrapH323EndPoint::WrapH323EndPoint: Compile-time libraries OpenH323 v1.12.2, PWlib v1.5.2
02:27.37_Raptor_[1]WrapGatekeeperServer::WrapGatekeeperServer: Creating new gatekeeper.
02:27.37_Raptor_PObject
02:27.43IQariel_: just looking for something with small footprint
02:27.58_Raptor_what have i done wrong now?
02:28.01ariel_small if you use asterisk get diax
02:28.27IQariel_: I am using asterisk but I need SIP based softphone
02:28.50IQariel_: which one do you know?
02:28.56DHuangIQ: try firefly...
02:29.03SexyKenShoulnd't this play music: exten => _XXXXXXXXXX,2,Dial(${CZ}/1${EXTEN},60,H|g|m)
02:29.09IQDHuang: thanks
02:29.19DHuangIQ: not a prob.. it also support G729a
02:30.33IQDHuang: just looking for a link to download it - is this the one: http://www.virbiage.com/firefly/download/index.php
02:30.53DHuangIQ: yeap..download the 3rd party one
02:31.03DHuangIQ: http://www.virbiage.com/firefly/download/firefly-thirdparty.exe
02:31.08ariel_IQ, firefly will work. But you asked for small footprint.
02:31.38DHuang:-)
02:31.48IQariel_: Yes, thats right. I dont know anything about firefly yet.
02:32.02IQariel_: you said you know something - could you tell me what is that?
02:32.44ariel_I use xlite due to it works with most setups. And it's fairly easy to use it via a stun server.
02:33.09IQariel_: Yes, I know xlite. but your answer to my question was "yes but why" . :)
02:33.29DHuangxlite doesn't support G729 right?
02:33.37dontmsgmeI want a laptop holster so i can walk and type, any URLs?
02:33.41ariel_why is all the others I have tried have many issues.
02:33.45ManxPowerSexyKen, No.  You don't put | between options .... ,66,Hgm)
02:33.52ariel_DHuang, your correct.
02:34.21ariel_DHuang, it supports gsm
02:34.24ManxPowerno free softphone can include G729
02:34.45Luke-JrManxPower: even if they're out of the US?
02:34.56DHuangManxPower: firefly can, if you get the g729.dll
02:34.58ariel_Luke-Jr, hehehe
02:35.05*** part/#asterisk Grooby (~Grooby@ip24-250-126-171.dc.dc.cox.net)
02:35.08ManxPowerLuke-Jr, That is up for debate, but since no free softphone currently includes G729......
02:35.16ManxPowerDHuang, Yes, but that's not "included"
02:35.21Luke-JrManxPower: I think MPlayer has the code for it :)
02:35.25ariel_actually there is another sip software that was ok from a co. called SIPP in EU
02:35.51DHuangManxPower: you are right, their www site gives you the LIBS + C code to compile it..
02:37.35Luke-Jrariel_: on principle
02:37.44*** join/#asterisk techie (gus@asterisk.horizonte.us)
02:38.02Luke-Jrariel_: also, if I pay $10 to Digium, do I get the source?
02:38.02*** join/#asterisk brc_ (~brian@brc.base.supporter.pdpc)
02:38.24ariel_Luke-Jr, it's not gpl
02:38.27DHuangLuke-Jr:  I wish....
02:38.47Luke-Jrariel_: that's why
02:38.56ariel_Luke-Jr, there is a free one that works for 6 months from intel some where. But don't count on it too much for real production.
02:38.58DHuangJuke-Jr: you might be able to find source code for it.... as H232 is developed based on G729
02:39.02Luke-JrIf I were to pay for any software, it would need to be open source
02:39.14ManxPowerLuke-Jr, I heard a rumor that the patent holders refused to sell Digium a license unless the put copy protection in.
02:39.36ManxPowerHell, I have the source for the ITU G729 codec.  It's not that hard to get.
02:39.40Luke-JrManxPower: Then I will gladly go with a non-US-regulated codec
02:40.00mishehuspeex up, I can't here you
02:41.44brc_here?
02:44.54mishehuhear
02:44.56mishehuwhatever
02:45.10mishehuMr. Bell, you are ringing...
02:48.31DHuanganyone know why the asterisk is not detecting Budget Tone-100 key press?
02:48.59fearnorbudget
02:49.16DHuang:-p
02:49.54DHuangis it BT-100 setting or is it asterisk?
02:50.24ariel_DHuang, I think that BT use info and not inband for there dtmf
02:51.04DHuangariel: I see....I'll check the BT
02:52.06DHuangariel: Send DTMF:     in-audio      via RTP (RFC2833)      via SIP INFO   , which to pick SIP?
02:52.24*** join/#asterisk gabb0 (~gabb0@CPE0006258dff02-CM000a73661510.cpe.net.cable.rogers.com)
02:53.26gabb0hello all
02:53.49ariel_DHuang, your sip.conf for the bt need to have dtmfmode=info. But last time I used one was over 1 year ago.
02:54.15DHuangAriel: I see.. :-) .. THanks
02:55.04gabb0quick question, I recorded a call using Monitor and recorded it in pcm format.  How do I convert that to a wav or mp3?
02:55.17Strom_Cgabb0: lame?
02:55.18*** join/#asterisk agave (phanop@216.81.43.75)
02:55.31bjohnsonfor what? playing on your mp3 player?
02:55.31Strom_Calso, pcm == wav
02:55.32gabb0won't sox do it
02:55.51gabb0I know that but try playing it with itunes
02:55.53agavei want to have asterisk send a call to a new context if an incoming call doesn't have callerid, similar to privacy manager, how do I do that?
02:55.57gabb0doesn't sound pretty
02:56.12bjohnsonagave: gotoif()
02:56.31agaveI can test on ${CALLERIDNUM}
02:56.39agavedoes it come in as null if it is blocked or not available?
02:56.40bjohnsonexactly
02:56.44bjohnsonnull
02:57.03gabb0Strom_C, what's the syntax for lame to convert
02:57.04agaveokay that's what I thought... does asterisk have a NULL or should I just do if ${CALLERID} = ""
02:57.05bjohnsonjust do if = ""
02:57.18agavecool
02:57.19agavethanks
02:57.25DHuangariel: I've set the BT to DTMF visa SIP..and works now... :-)
02:57.27Strom_Cgabb0: I don't know.  read the manpage.
02:57.30bjohnsonthere's an example on the wiki ..
02:57.44bjohnsonthat prompts a person to enter their callerid if it comes in as null
02:58.14bjohnsonand prevents them from coming in otherwise .. not too customer friendly but might work in some cases
02:58.19brc_~seen atacomm
02:58.24jbotatacomm <~dan@69.54.45.98> was last seen on IRC in channel #asterisk, 16d 1h 1m 31s ago, saying: 'anyone want a IP 3000 conference phone?  looking to replace ours with a IP 4000 model.  Barely been used, in great condition.... looking for around $500'.
02:58.52mishehuthe ip3000 doesn't have sip firmware though (last time I checked)
02:59.08agavethe wiki si down :(
02:59.31agavedamn, I tried to trap on $CALLERIDNUM but since it's a SIP did it is coming in as the SIP URL so it doesn't see that it is blocked
03:00.03bjohnsondon't know how to help you on that one
03:00.32bjohnsonall my dids are iax so the format is easier I guess .. I just check on callerid
03:00.32agavewell, caller id name is set to asterisk...
03:00.35bjohnsonlike normal
03:00.40agavei guess i could trap on that
03:06.40SexyKenHey guys, so I have a 'call through' system setup...so I can call into asterisk and dial a number and it calls out...
03:06.55SexyKen....is there anyway to allow me to play music on the line if a button is pressed so both parties can hear?
03:07.08IQ????
03:07.42*** join/#asterisk brookshire (~nobody@pcp01541028pcs.huntsv01.al.comcast.net)
03:08.46SexyKen???????
03:08.53SexyKenWHat do you need exlplained more
03:10.56bjohnsonwrite a program to do it I guess
03:11.13bjohnsonmaybe some kind of 3 way call conferncing thing
03:11.16bjohnsonor using meetme?
03:11.44bjohnson* doesn't monitor for much after a call has been successfully answered
03:12.04bjohnsonand what it does monitor will usually take you out of the call
03:19.44Mochi everyone
03:21.43tzangerevenign moc
03:23.27_Raptor_anyone here who can explain this: http://rafb.net/paste/results/M5Alwm11.html ???
03:24.02brookshirebah.. h.323
03:29.55dougheckais voip-wiki down?
03:30.30brookshirelooks like it
03:31.04dougheckacrap
03:31.06doughecka:P
03:31.31FaithXvoip-wiki goes out a fair bit
03:32.16dougheckathey need a better host
03:32.25dougheckawhats the backend?
03:32.49brookshiredigium just needs to document stuff
03:32.50dougheckawould it be simple to mirror the site?
03:32.50brookshirehaha
03:33.11dougheckawell I need info for cisco phones
03:33.18dougheckalike how to get the external directory
03:33.23dougheckathing to work with asterisk
03:33.23brookshireoh
03:34.01brookshirelike is said.. digium needs to document stuff :)
03:34.17dougheckalol
03:34.37mtqhdoughecka: digium does not need to document stuff.....they charge for that
03:35.08brookshireheh.. are you complaining?
03:35.20dougheckadocument?
03:35.28dougheckathey have AWESOME documentation
03:35.29doughecka:)
03:41.26*** join/#asterisk PCadach (~paul@www.east.telecom.kz)
04:04.27ScythelXdoughecka: the external directory is an xml file located on the websever that the phone accesses
04:10.50shido6boink
04:10.59shido6anyone seen Constantine?
04:11.36*** join/#asterisk greendisease (~jack@greendisease.fedora)
04:12.09*** join/#asterisk coppice (~chatzilla@245.195.17.210.dyn.pacific.net.hk)
04:12.59*** join/#asterisk roamer323 (~sing@HSE-Toronto-ppp130885.sympatico.ca)
04:13.35BeirdoHmmm
04:13.46*** join/#asterisk TheEmperor (TheEmperor@218.111.51.3)
04:13.48Beirdohaving a hard time getting Festival to behave
04:14.58*** part/#asterisk SuperMMan (~graphic@d209-89-191-155.abhsia.telus.net)
04:16.43*** join/#asterisk BoRiS (boris@S01060040ca1e5b54.wp.shawcable.net)
04:17.29dougheckacan I do something with mpg123 to make the MOH play louder?
04:17.37*** join/#asterisk PCadach (~paul@www.east.telecom.kz)
04:20.17shido6festival is fun
04:20.30shido6heh, sorta
04:20.44Beirdoit doesn't seem to do squat
04:22.00coppiceof course it doesn't do squat. only real peopel do that :-)
04:22.03QwellBeirdo: the syntax is really weird
04:22.20Qwellfestival (SayText "test")
04:22.33coppiceits lisp
04:22.35Qwellprobably have to quote it, and escape the quotes
04:22.40shido6rerecord the mp3 louder
04:23.18dougheckashido6: btw the cisco phone works great :)
04:24.29BeirdoQwell: is that a line for in extensions.conf?  I'm looking for some online examples that use it
04:24.52QwellBeirdo: Not sure how to use it in asterisk, I just remember screwing with it for a bit from CLI
04:25.41shido6doughecka uhhh
04:25.45shido6did we talk on the phone?
04:25.49dougheckanah
04:25.52shido6k
04:25.55shido6dont drop ur cisco
04:25.58dougheckayou pointed me to voip-supply for the power supply
04:26.00shido6u CAN drop the budgetone tho
04:26.03shido6oh
04:26.05shido6heh
04:26.05*** join/#asterisk muesli (~muesli@mail.muehlhaeuser.de)
04:26.06dougheckahah
04:26.07shido6did u tell them i sent ya
04:26.08shido6?
04:26.24dougheckaactully I forgot, I ordered it when I woke up and wasnt exactly awake then
04:26.26shido6they get criscos , I mean cisco's at stupid low prices
04:26.36shido6in blocks
04:26.44dougheckaI could email them and say greg sent me
04:26.47dougheckaand make them all confused
04:26.48doughecka:P
04:26.53shido6big phatass toomuchcashtothrowaround blocks
04:27.01dougheckahahaha
04:36.55TheEmperorhow i call someone if their sip number is 12345@abc.com ?
04:37.05TheEmperori mean, how do I call :)
04:39.52loudhttp://slacker.com/~nugget/asterisk7.php
04:40.43TheEmperorloud: thanks :)
04:45.14ScythelXfinally got ser working correctly
04:45.46*** join/#asterisk neopher (~crazy@mail.techhelpresources.com)
04:46.13neopheranyone here have Firmware image for a cisco 30 vip?
04:46.32*** join/#asterisk yashax (~yasha_x@c-24-98-23-73.atl.client2.attbi.com)
04:46.46TheEmperorloud: so i have to configure my exten file? i thought there is a softphone you can just put the address into
04:48.06loudno, its not that simple, unless you have a 7960.
04:51.30wankelyeah, well it's not that easy to type that on a 12-key pad either :P
04:53.45BeirdoOK, almost got it :)
04:55.16BeirdoFeb 18 23:54:32 NOTICE[12811]: rtp.c:430 ast_rtp_read: RTP: Received packet with bad UDP checksum
04:55.20Beirdogrrr
05:00.56hmodeswheee
05:03.47*** join/#asterisk Defraz (~t0tal@65.103.222.4)
05:10.07stepcutHMBeirdo: Feb 18 18:11:23 NOTICE[65130]: rtp.c:452 ast_rtp_read: RTP: Received packet with bad UDP checksum
05:10.10stepcutHM:p
05:11.04*** join/#asterisk eipi (~eipi@40-142-89-200.fibertel.com.ar)
05:12.26*** join/#asterisk NormAst (HydraIRC@Ottawa-HSE-ppp4116580.sympatico.ca)
05:12.47NormAstHi all.
05:13.05BeirdostepcutHM: got any ideas how to fix that? :)
05:18.01*** join/#asterisk uberwolf (~djdjs@c-67-165-175-191.client.comcast.net)
05:18.05uberwolfhello all
05:18.12*** join/#asterisk guugmember (~nramos@200.6.221.64)
05:18.22NormAstHi.
05:18.25uberwolftrying to build my first *
05:18.30NormAstGREAT!
05:18.38guugmemberis there a way my teleco can close my IAX comunication from my home to my Asterisk in my office?
05:18.57uberwolfgot all the way through with at conf and I am looking at the console when you try to make a SIP to SIP call
05:19.06uberwolfand I get this on the screen
05:19.19uberwolfFeb 18 10:11:53 NOTICE[5454]: app_dial.c:746 dial_exec: Unable to create channel of type 'SIP'
05:19.19uberwolf<PROTECTED>
05:19.28JerJerthe type=peer is invalid
05:19.49uberwolfah
05:19.52uberwolfok
05:20.04yashaxGuys, how can I force the ifconfig command to set the IP address to be a permanent change (static). Everytime when I issue the command, it get's the dynamic address from DHCP server after a while...?
05:20.12uberwolfI was using the sample config in mahler's book
05:20.16*** join/#asterisk pawnbroker (~rstevensj@ca-santaanahead-cuda1-c5a-45.anhmca.adelphia.net)
05:20.21ManxPoweryashax, Step 1: /join #linux
05:20.51guugmemberwho is going to go to Miami to the TMC telephony show next tuesday?
05:20.51yashaxI have, have been there for 15 min with no help at all.. so thought to try it here.. I am here a lot, so....
05:21.02guugmemberjsolares, que onda chema
05:24.13pawnbrokeris the gang up for a newby question?
05:24.47sivanayashax: try ##linux
05:25.01sivana#linux is read-only :P
05:25.31ManxPowerThen try #linux-help
05:25.37ManxPoweror #linux-newbiw
05:25.41ManxPowers/w/e
05:25.54yashaxthank you....
05:27.32pawnbrokercan * simlt ring multiple remote extensions and a cell phone?
05:27.40neopherlooking for cisco 30 vip Firmware image, anyone gots?
05:27.52neopherpawn: yes
05:28.13ManxPowerneopher, There's no SIP firmware for that phone.
05:28.37neophermanx: not looking for sip, looking for the latest bootloader
05:29.02BeirdoI lost my phone line I was going to use to hook up the FXO
05:29.08Beirdoway to go, dumbass
05:29.32pawnbrokercool... I assume fist answer grabs the call. how well does it work with the ixay?
05:29.51neophermanx: when i hat *** the phone doesn't show firmware version, hence it is real old and i can not use it with chan_skinny
05:29.56Inv_arphey who offers cheap local iax did's, not 800 besides VP connect
05:30.01guugmemberhave anybody experience that the fxo card is making calls every 15 minutes?
05:30.33neopherguug: had that one, had to restart the * box
05:30.54guugmemberneopher, we have restarted, but also makes the call
05:31.21neopheris it just hooking or is it actually dialing
05:31.32guugmemberjust hooking
05:31.52neopherhmmm, line voltage variation
05:31.55neophermaybe
05:32.02guugmemberneopher, how can I change that?
05:32.03neopherhad that happen too
05:32.18guugmemberchange the pci slot?
05:32.18neopheryou can't
05:32.23*** join/#asterisk murangd (~nukaidc@pool-162-84-144-211.ny5030.east.verizon.net)
05:33.02neopherif it is a line voltage variation, it would be a prob at the CO which the telco would have to fix
05:33.23guugmemberneopher, what is CO?
05:33.41sivanacentral office
05:34.00neopherif the voltage fluxtuates (can't spell) then the modem/asterick thinks a call is coming in
05:34.21neophercould also be a modem the is starting to go, seen that b4
05:34.32neopherthanks siv
05:35.45neopheralso , check for line noise, sometimes if the line sounds staticky it may mean a wireing prob somewhere along the line that could cause the prob
05:37.00neopherhow many fxo card do you have, and what ones are you using
05:37.58guugmemberneopher, I have a TDM04B, 4fxo, and just using 1
05:38.25BeirdoI got one, a X100P clone, not hooked up as I can't find the f'ing phone cord I have for it
05:38.55neopherhmmm, you shouldn't have a prob with that card at all, digium makes good shizit.
05:39.44neopheri use 2 x100p clones, they work great, then again, there just an intel modem
05:40.37Beirdoif Digium still made the X100P, I might even feel guilty
05:40.45*** join/#asterisk TheEmperor (TheEmperor@218.111.51.3)
05:40.52TheEmperorloud: where can i get a cheap one of those? :)
05:41.13neopherthey stopped making them because the chipset (made by intel) was EOL
05:41.17Beirdohmm?
05:41.19guugmemberneopher, so it looks like a voltage fluctuation problem
05:41.39neopherguug: quite possibly
05:41.50guugmemberneopher, and is exactlly every 15 mins
05:41.54Beirdoyeah, well, maybe some thought should be put into making more current cheap single-line cards :)
05:42.05*** join/#asterisk sandnigg0r (~trilluser@66-55-197-254.gwi.net)
05:42.08neopherhmmm, thats a little weird
05:42.27neopher<PROTECTED>
05:42.28Beirdoif I had the time and the money, I'd be happy to make a DSP card that would do the trick.
05:43.02mishehuand what of the tigerjet chipset used on the tdm400's ?
05:43.23Beirdoisn't that the chipset used on the clones too?
05:43.33neopherbeirdo: no need for single line cards, just get broadvoice, they are cheap and support * sip connections
05:43.47Beirdoneopher: yes there is a need
05:43.56neopheryes that chipset is used on the clones to
05:44.03Beirdomany of us have one or two phone lines that we want to hook up
05:44.17mishehuif a deal goes thru, I'll be using a t110p with a pri
05:44.22guugmemberneopher, ok thnx a lot, will see the config files
05:44.23neopherthere are actually 2 diff. chipsets that act exatly alike, one with more option then the other
05:44.30Beirdomishehu: you are a bastard :)
05:44.41mishehuBeirdo: 2 phones, you might as well use a tdm02b
05:44.48mishehusave some interrupts
05:45.00Beirdomishehu: for about $250?  no thanks
05:45.10mishehuBeirdo: hey, I need to make sure the costs of the line are covered before the deal goes thru...
05:45.38neopherguug: if you catch BTW, ask him, he may have more insight with that card
05:46.02mishehubtw?  or bkw?
05:46.33Beirdomishehu: cool.  you are making me jealous
05:47.01*** join/#asterisk abatista (~Ariel@dsl-20-177.cofs.net)
05:47.11_Vilebkw u wanna tweak the code to have newest messages play first in voicemail rather than oldest?
05:47.16neopherhow about this, ethernet ata fxo's , no irq usage, use sip, less driver headack
05:47.29_Vileasdf
05:47.32neophermishehu: thanks, thats what i meant
05:47.55*** join/#asterisk yasha (~yasha_x@c-24-98-23-73.atl.client2.attbi.com)
05:48.01Beirdoneopher: that's a possibility, but they aren't as easy to find, and I don't have IRQ issues
05:48.04yashaGuys, what/where is the best way to terminate * via IP to get unlimited local/long if possible?
05:48.38ManxPoweryasha, pretty much none.  Providers that give unlimited require you to use their box
05:48.43neopheri got four letters EBAY, there all over the place, at least when i was looking last week
05:48.51guugmemberneopher, ok i will ask BTW
05:49.08neopheryasha: broadvoice
05:49.18yashak... where would you recommend the best/cheapest place would be?
05:49.21neopheryasha: if you want to connect via sip
05:49.44neopheryasha: broadvoice  www.broadvoice.com
05:49.59neopheryasha: i use them and love them
05:50.08yashaI can terminate * withit? How much is it?
05:50.22Beirdoneopher: fxo sip devices?  I haven't seen terribly many of them
05:50.45neopheryasha: yes you can connect * via sip to them
05:50.50Inv_arpyasha: 9.95 activation plus 9.95 a month
05:50.59guugmemberis there a way my teleco can close my IAX comunication from my home to my Asterisk in my office?
05:51.05neopheranywhere from $5 to $25 / month
05:51.33Inv_arpyasha: look at iax.cc, livevoip.com, voicepulse.com  other good alternatives
05:52.09yashaneopher: unlimited?
05:53.05Inv_arpguugmember: you mean your isp stopping the packet transmission or somethin?
05:53.36neopheryasha: yes, unlimited us,ca and 21 contries, $20/month
05:53.48neopheryasha: yes, unlimited in state 9/mo
05:53.59yashalocal DID?
05:54.33Inv_arpyasha: www.broadvoice.com explains all
05:54.34neopherif you isp is blocking VoIP packets, launch a complaint with the FCC as there are law that protect this
05:54.36guugmemberInv_arp, my ISP closing IAX port
05:54.47Inv_arpguugmember: use another port
05:55.15guugmemberInv_arp: like 80?
05:55.27Inv_arpguugmember: whatever your heart desires
05:55.35guugmemberInv_arp, great
05:55.57guugmemberInv_arp, so its almost impossible for your ISP to stop IAX packets
05:56.34Inv_arpguugmember: unless there running traffic analyzers... to see what type of data is floing thru ports
05:56.35guugmemberInv_arp, in my country you sing a contract saying that you are not allow to use the conection for VoIP
05:56.40*** join/#asterisk lilo (lilo@levin-pdpc.staff.freenode)
05:57.15Inv_arpguugmember: ahh sucks
05:57.29guugmemberInv_arp, totally
05:57.52guugmemberInv_arp, even thoug nobody respects that
05:57.59Inv_arpguugmember: heh
05:58.23Inv_arpi luv iax cause it just works
05:58.54brookshiresing a contract?
05:58.54brookshirethat's awesome
05:59.25guugmemberInv_arp, yeah it works great in my country too
05:59.31pawnbrokeris anyone here using the iaxy's? pro's con's?
05:59.52JerJerIAXy rules
06:00.00guugmemberpawnbroker, i only have pros on the IAXy
06:00.20guugmemberpawnbroker, a little con can be the price, close to US$100
06:00.31Inv_arpi knew i sould have gotten one
06:00.48pawnbrokerthanks
06:01.27*** join/#asterisk NormAst (HydraIRC@Ottawa-HSE-ppp4116580.sympatico.ca)
06:03.32pawnbrokerforgive me i'm new to *.. :) do you need to subscribe to a service to use iaxy remotely?
06:03.54*** join/#asterisk adker (~adker@70-97-138-2.dsl1.glv.ny.frontiernet.net)
06:04.28guugmemberhttp://fp1.a2zinc.net/clients/fpvon/spring2005/flash/fp.aspx
06:04.30guugmemberwow
06:04.58*** join/#asterisk kpfleming (~chatzilla@ip68-3-230-141.ph.ph.cox.net)
06:05.00guugmemberthey should be using Apache at least
06:05.10NormAstLink doesn't work.. Microsoft.
06:05.54guugmemberand its a von link
06:06.02jetscreamerServer Error in '/clients/fpVON/Spring2005/flash' Application.
06:06.20neopherlinux = servers running     microsoft = everyone running from the servers
06:06.49NormAstpawnbroker: http://www.digium.com/index.php?menu=iaxy
06:06.49jetscreamervb
06:07.02*** join/#asterisk Luhiwu (~marsosa@200.63.89.209)
06:07.17guugmemberNormAst, i just love that pic
06:07.18NormAstvb for *
06:07.47pawnbrokerNormast: thanks
06:08.04guugmemberpawnbroker, better price i found http://voipstore.pulver.com/product_info.php?products_id=52
06:08.49NormAstpawnbroker: so... IAXy is a little box that connects to your asterisk box.  If you have two of them you can talk to your self.... You can use the echo test and talk to yourself...
06:08.56neopheranyone notice that in that pic ip phone entns should be ip phone extns
06:10.15guugmemberneopher, entns is entrance?
06:10.21neopherNormAst: if i get three can i talk to the voices in my head? hehe
06:10.39neopherguug:ahhh
06:11.05guugmemberneopher, what do you mean by ip phone ents should be ip phone extns
06:11.09pawnbrokerguug: thanks
06:11.09NormAsthmm... I think so... I just hear voice... Never thought about talking to them.
06:11.25*** join/#asterisk sudoer (~sudoer@65.75.148.190)
06:11.53neopherguug:i thought they meant to say ip phone extensions
06:11.54sudoerif I have an ethernet cable, can I convert it to rj11 for analog phones?
06:12.11NormAstYou can.. Won't look nice.
06:12.23guugmemberneopher, ahh ok
06:12.41sudoerI have cabling setup already and I dont want to rerun analog lines
06:13.30neophersudoer: just use a pair in the center 2, no prob with that
06:13.33guugmemberany one recommends me a good channel bank, and not that expensive
06:14.36sudoerneopher: is there a little converter plug I can buy or will I have to rewire the ends?
06:15.36pawnbrokersmoking quells the voices...
06:15.49neophersudoer: don't know of any converter, but you could essentually make one with an rj45 keystone and an rj-11 keystone
06:16.41neopherpawnbroker: haven't been to the movies since the doctor told me to stop entertaining the voices
06:16.54mishehuhmm...  voip-info isn't responding to me.  I tend to get users/peers mixed up... an iax peer can receive calls from me, and an iax user can make calls to me, right?
06:17.36pawnbrokerneopher: I'm sitting here beside myself
06:17.51sudoerso neopher , you mean just cut the head off of rj45 cable, then replace it with rj-11 keystone?
06:18.13neopheryou could do that
06:18.38neopheror crimp a rj11 end on
06:19.00neopheryou ar only using 2 wire for analog telephone
06:19.17Beirdomishehu: I believe that is correct
06:19.33neopherand it is the center 2 conductors that are used on the connectors
06:19.40mishehuBeirdo: my mind gets kind of polluted from all the crap I deal with day-to-day...
06:19.44Beirdoand friend can do both (as it's a combination)
06:19.56mishehumaybe I should install a garbage collector routine in my head ;-)
06:20.04mishehuyeah, friend I know can.
06:20.21Beirdoheh, vodka's my garbage collector tonight
06:20.51neopherno fair, you got vodka, i got relatives
06:21.14Beirdohehe
06:21.41neopherthey decided they were going to stop by and spend the night
06:22.02Beirdoso get fit-shaced
06:22.45neophernaaa, going to try to get up early go snowboarding, thats my outlet
06:23.23pawnbrokerneopher: where do you board at?
06:24.10neopherin the poconos
06:24.30neopheri live like 15 min from the resort
06:24.41pawnbrokeri'm on the west coast Mammoth every other week to decomp
06:25.21neopheri wish i was on the west coast, this ice shizit is really pizzing me off
06:25.22pawnbrokerlucky you!
06:26.16pawnbrokerits great here now with all this h20, a 4 hr drive for me
06:26.33neopherbrb, crapper
06:27.05pawnbrokeryou have boilerplate to ski on and normally we get what we call sierra cement
06:27.26*** join/#asterisk elric (~kavit@ppp114-10.static.internode.on.net)
06:32.08pawnbrokerThanks gang for all the * info, I'll be back
06:32.12*** part/#asterisk pawnbroker (~rstevensj@ca-santaanahead-cuda1-c5a-45.anhmca.adelphia.net)
06:36.27*** join/#asterisk DaLion (~Miranda@Quebec-HSE-ppp225437.qc.sympatico.ca)
06:36.44DaLionHey all
06:36.44DaLionanyone tried  teliax.com?
06:37.23DaLionand implicit u still drunk ?
06:39.29*** join/#asterisk murangd (~nukaidc@pool-162-84-216-79.ny5030.east.verizon.net)
06:40.03murangdwhat's this IAX2/guest@66.250.68.194/996
06:40.22Inv_arpDaLion: they seem good but they dont have local DID's in my area ...  try also livevoip.com or iax.cc
06:41.50*** join/#asterisk pranav (sameer@202-149-48-198.broadband.isp.exatt.net)
06:45.26murangdInv_arp: livevoip.com charges like 1.2cents per minute for incoming calls to your did
06:45.41murangdis there like a provider of DID that doesn't chare any $$ for incoming calls to a did
06:47.21*** part/#asterisk pranav (sameer@202-149-48-198.broadband.isp.exatt.net)
06:47.24*** join/#asterisk pranav (sameer@202-149-48-198.broadband.isp.exatt.net)
06:47.54Inv_arpmurangd: livevoip has an unlimited $6.99 plan   iax.cc 10.49 unlimited .. think VP connect also
06:48.11murangdInv_arp: not for commerical use
06:48.33neophersoft cap?
06:48.51neopherhow can they call it unlimited if there is a soft cap
06:49.01murangda message from livevoip.com
06:49.02murangd1.1 cents per minutes everytime the number is called. We are forced to pay termination on DID's, everyone is.
06:49.12murangdguess they are not so unlimited
06:49.28Inv_arpmurangd: http://connect.voicepulse.com/   or broadvoice or iax.cc
06:49.32ManxPowerTalk to your locel CLEC.
06:49.43murangdManxPower: CLEC?
06:49.47ManxPowerWe have totally free calling within Mississippi and Louisiana
06:49.59neopheri use broadvoice which is unlimited, but you connect via sip
06:50.09ManxPowerCLEC = competitive Local Exchange Carrier.  i.e. a company that competes with Bell.
06:50.15ManxPowerWe like our CLEC.
06:50.36Inv_arpneopher: how much is it?
06:50.59EssobiI know a few CLEC. :)
06:51.14neopherwww.broadvoice.com
06:51.42neopher$20/mo for unlimited us canada and i beleive 21 other countries
06:51.51murangddoes anyone the know the CLEC's for new york city
06:51.57Inv_arpbah i dont need unlimited outgoing,  just unlim incoming
06:52.07murangdyeah me too
06:52.12murangdI just want unlimited incoming
06:52.13ManxPowerunlimited incoming is pretty common
06:52.14Inv_arpi use voipjet for outgoing
06:52.14neopherthen get the cheaper plan
06:52.15murangdforget about outgoing
06:52.28murangdInv_arp: what you use for incoming
06:52.50neopher5.95/mo unlimited incomming
06:52.53murangdI don't understand.. why do some companies provide unlimited INCOMING and others charge you for incoming
06:52.57murangdneopher: what provider?
06:53.07neopherbroadvioce
06:53.12Inv_arpiax.cc  for personal uses no business plan as of yet
06:53.15murangdconnect.voiceplus.com offers unlimited uncoming for $8 per did
06:53.23murangdbut $8 is to expensive
06:53.40murangdespically when services offer it for free
06:54.01murangdManxPower: why do you suspects some providers charge you for incoming DID
06:58.21Beirdohmmm.
06:58.34Inv_arpneopher: yea i might go with broad voice 5.95 plan .. u have good connection with them etc...?
06:58.41Beirdois there a way to tell a particular extension NOT to use MOH?
07:00.55ManxPowermurangd, Spend some time finding out pricing for all the IAX and SIP VoIP providers, you'll find a few that charge per min for DID.
07:01.17murangdManxPower: yeah that's no good to charge for DID
07:01.23murangdthey should give you free unlimited did's
07:01.43ManxPowerI don't mind a monthly charge for a DID, just a per min charge.
07:01.47murangdI am looking for DID for NJ/NY
07:01.59murangdManxPower: yeah me too, per min charges is what I was refering to
07:02.09neopherInv_arp: haven't had one prob, i love them
07:02.14murangdipkall.com = unlimited DID
07:02.39Inv_arpneopher: hmm that 5.95 plan is unlimited incoming? cant find where i says that
07:02.40ManxPowerI just wish the cost of DIDs would come down.
07:02.55ManxPowerWe pay $20/month for 100 DIDs from our phone company.
07:03.12*** join/#asterisk pfn (500@adsl-69-107-210-254.dsl.pltn13.pacbell.net)
07:03.27pfn, pfn\
07:03.28neopher<PROTECTED>
07:04.36Inv_arpneopher: nice  just hope its as easy to setup as iax since * is behind nat
07:05.06*** join/#asterisk abombss (~abombss@c-67-175-115-51.client.comcast.net)
07:06.29neopherInv_arp: broadvoice is not IAX, it is sip
07:06.39ScythelXbroadvoice is cheap
07:06.45neopherInv_arp: still very nice
07:06.56murangdManxPower: $20 is not a bad price
07:07.00murangdManxPower: you get free incoming?
07:07.29Inv_arpneopher: yea i know im hoping its as easy as my iax setup's   since my * is behind a nat
07:07.45neopherInv_arp: verry easy
07:08.27Inv_arpneopher: ahh k
07:09.34ManxPowermurangd, $20/month for 100 DIDs, free unlimited calling withing Louisiana and Mississippi and free 256K internet (that we don't really use)
07:09.42ManxPowerBut this is a local phone company.
07:11.01murangdManxPower: ah I see
07:11.56ManxPowerVoIP is nice, it's useful, but compared to local PRI lines it's horribly complicated.
07:14.05coppiceVoIP sucks, but its the future
07:16.08ManxPowerWe have a frame relay network with 0CIR so no VoIP over THAT.
07:16.48JerJerwholy fire that ISP
07:16.54JerJerbatman
07:26.46DaLiontrying to instal suse 9.2
07:26.55ScythelXhey all looking for suggestions - not sure if I should just get a t1 line or use a datacenters interconnect for termination
07:27.23DaLiongo for teliax.com
07:27.25DaLioni did
07:28.36DaLiontring to build a unix gateway
07:28.36DaLionbetween pci 802.11 and dsl
07:28.48ScythelXwell im looking to reduce the amount of hops
07:29.12ScythelXeither the t1 directly or host the box in the datacenter using their voice termination
07:31.53neopherwhat happened to voip-info.org
07:32.26brookshiredied
07:32.28brookshire:)
07:32.40Beirdofall down go boom
07:34.37DaLionethernet everywhere nc100 v2.1 is what
07:39.22neopheri figured out that the best mtu for voip traffic with dsl using pppoe is 1492, just an addon for everyones notes incase you run into it in the future
07:39.46jpaynethat may well depend on the dsl provider
07:40.03Daminneopher: That happens to be the best MTU for must PPPoE and PPPoA type services...
07:40.15Beirdoneopher: the best MTU for ANYTHING over PPOE is 1492
07:40.17neopheryes
07:40.29Daminneopher: Leaves just enough room for the overhead of the PPPoE packets. :)
07:40.30BeirdoPPPoE rather
07:40.48Qwellhttp://ipv4.uuoc.com/?id=803  Is that fairly normal, for only one instance of * running?
07:41.13neopherbut some people recommended 1428 which makes voip traffic scetchy
07:41.31*** join/#asterisk pranav (sameer@202.149.48.198)
07:43.00neopherQwell: going overboard with the v's ---> asterisk -vvvvvvvvvvvvc
07:43.09DaLionok so a network everywhere nc100 .. i cant find in modules list of suse install ..whats is it like..
07:43.12neopheri only use five, and thats plenty
07:43.13Qwellneopher: yeah, I hold it for a second or two :p
07:43.31sivanayes!! I found it
07:43.44QwellBut, is that normal?  Having it show up so many times?
07:43.45DaminQwell: Yeah.. that looks right..
07:43.51sivanaheh.. a bug in my app which would cause * to freeze up solid :P
07:44.38neopherqwell: I'm going to call you when i am ready for bed and you put me on hold so i can listen to you hold music
07:44.43DaminQwell: If you only want to have one MPG123 process, then comment out all the other classes in /etc/asterisk/musiconhold.conf and add a -z to the end..
07:44.46Qwellneopher: heh
07:44.56DaminQwell: Ala default => quietmp3nb:/var/lib/asterisk/mohmp3,-z
07:45.22DaminQwell: Then restart asterisk and kill all the mPG123 processes..
07:45.22QwellDamin: I was more worried about the multiple * processes
07:45.30Beirdois there a way to tell one particular extension not to use MOH?
07:45.32QwellI have 0 calls, and I've had 0 since I started it
07:45.53DaminQwell: It's multithreaded..
07:45.56Qwellok
07:45.59DaminQwell: That is good.. ;)
07:46.03*** part/#asterisk brookshire (~nobody@pcp01541028pcs.huntsv01.al.comcast.net)
07:47.23*** join/#asterisk lilo (lilo@levin-pdpc.staff.freenode)
07:48.09Beirdoand lilo's back :)
07:48.43QwellDamin: The only reason I ask, is I had some weird problems lately, I don't recall seeing that in the past
07:49.22ScythelXhttp://www.empirix.com/default.asp?action=article&ID=69
07:55.50*** join/#asterisk jonas (jonas@jonas.probe-networks.host.freenode)
07:57.28ScythelXcan someone shed some light on this question for me - you have a company like voipjet.com that hosts their equip at a datacenter - where do they get their dids from? are they doing TDMoE?
07:58.13JerJerlol
07:58.28JerJerthey buy them from some SIP provider
07:58.51ScythelXoh
07:59.16QwellIf you "buy" a bunch of DIDs, how long do you have them for?
07:59.21QwellIf you're a provider or something
07:59.31JerJerthey are not a real telco or anything...they simply bought space in some IP colo facility
07:59.31QwellDo you "own" them, or do you pay a yearly fee or something?
07:59.39ScythelXso there is still an extra network hop then
08:00.15ScythelXme --> voipjet.com --> their sip provider at another datacenter?
08:00.39JerJeryes
08:01.35ScythelXoh they make it sound deceiving
08:01.47ScythelXlike they're the actual teleco
08:01.52JerJerhell no
08:02.07ScythelXso thats the same with nufone as well
08:02.17JerJerno
08:02.24JerJerabsolutely not
08:02.50JerJerwe have our own TDM network into an SS7 switch
08:03.04JerJersoon two switches
08:03.31ManxPowerJerJer then get dids in a wide area of the country
08:03.52JerJerwe have DIDs in many states, NuFone simply does not offer them
08:03.54JerJeryet
08:04.23*** part/#asterisk DaLion (~Miranda@Quebec-HSE-ppp225437.qc.sympatico.ca)
08:04.30QwellJerJer: Do you personally own NuFone, or are you just somebody higher up in the chain?
08:04.40Qwell"just"...not that thats bad
08:04.43JerJeri own NuFone
08:04.57Qwells'what I thought, didn't want to sound like an idiot in the future though :p
08:05.03ScythelXI use nufone
08:05.49ScythelXso is your stuff onsite or do you have your equip in a datacenter
08:06.28JerJerdata center
08:06.30JerJermost certainly
08:06.48JerJerour own in Southfield, then we are in Equinix Chicago
08:07.09ScythelXcool
08:07.12JerJerSouthfield, MI
08:07.14Qwellout of curiousity, if you guys are in IL, why MI DIDs?
08:07.18Qwellnevermind, that answers that
08:09.24JerJerwe have IL DIDs as well, just don't offer them
08:09.34*** join/#asterisk djin (~djin@gridfox.xs4all.nl)
08:10.30Qwellany plans on a timeframe for offerings in other states?
08:10.45JerJernope
08:11.50Beirdowww.nufone.net, right?
08:14.52*** join/#asterisk Edgan (~edgan@okcforum.org)
08:16.24*** join/#asterisk pranav (sameer@202-149-48-198.broadband.isp.exatt.net)
08:16.39pranavhello everyone
08:17.04pranavi am still facing a problem with the fwd stuff
08:18.45pranavit still says that "its an invalid extension"
08:19.43pranavi have made some changes in the sip.conf and i have pasted the sip.conf and the extensions.conf in the pastebin.ca/6112
08:20.17pranavis there anyone else on the channel
08:20.55*** join/#asterisk mbranca_home (~matteo@host-84-222-20-161.cust-adsl.tiscali.it)
08:21.13pranavhi mbracana
08:23.07Qwellpranav: I can't get to that domain right now for some reason.  Mind repasting it at pastebin.com?
08:23.33pranavhello someone? there
08:23.51pranavya sure
08:24.02pranavjust a minute
08:24.09*** join/#asterisk DaBigMac (~JJ@203-173-48-1.dyn.iinet.net.au)
08:24.14DaBigMachello
08:28.09pranavya i have pasted it in pastebin.ca/6113
08:30.57*** join/#asterisk inspired (mikael@host-81-191-119-90.bluecom.no)
08:32.13*** join/#asterisk scardinal (~supreme@port816.ds1-suoe.adsl.cybercity.dk)
08:33.42*** join/#asterisk tecnico (~tecnico@user-24-236-123-31.knology.net)
08:37.18*** join/#asterisk dawda (pranav@202-149-48-198.broadband.isp.exatt.net)
08:37.30dawdahello
08:37.42dawdaqwell sorry i got disconnected
08:38.01dawdai am pranav-- registered with different name
08:38.49dawdadid u check in pastebin.ca/6113
08:42.30DaBigMacgents I have what I hope is a simple question answered a million times.............I have convered over to VoIP using an asterisk server.....what Id like to do is connect my previous landline to my asterisk box so I can receive calls on my old number via asterisk...that way I dont need 2 sets of phones in the house......whats the easiest cheapest card/way to do it
08:42.38JerJerBeirdo: yes that is our crappy website
08:42.48BeirdoHeh
08:42.57JerJerDaBigMac: TDM01B from Digium
08:43.02BeirdoOK, just wanted to make sure I was at the right site
08:43.25DaBigMacJerJer thanks for the info Ill go suss it out
08:44.16BeirdoOooh, and you take PayPal. :)  Excellent
08:45.12heragis the wiki down?
08:49.22*** part/#asterisk dawda (pranav@202-149-48-198.broadband.isp.exatt.net)
08:50.26ScythelXherag: yeah
08:54.55brc_JerJer!
08:54.57brc_howdy
08:59.14JerJermooo
09:03.10mikegrbooooom
09:03.43DaBigMacJerJer : Just checked the Digium website hardware products section and TDM01B isnt listed
09:04.05QwellDaBigMac: You have to go to the Yahoo store, and look at the tdm400p section
09:04.12QwellThen you'll see the different configurations
09:04.21DaBigMacthanks qwell
09:04.29*** part/#asterisk djin (~djin@gridfox.xs4all.nl)
09:04.48Qwellits probably 4-5 clicks from digium's site
09:07.56JerJerthe TDM01B is a specific bundle of the TDM400P chassis and TDM Modules
09:08.25QwellI would have said that, but was too lazy to read up a few lines to see if it was covered
09:08.38Qwellthe 01 implies 0 FXS modules, 1 FXO
09:16.17JerJeri was trying to be nice
09:16.36JerJersome people seem to think I am always angry and/or hateful to people in the channel
09:16.55DaBigMacso the telco line to my house is designated FXO
09:17.12*** join/#asterisk pranav (pranav@202-149-48-198.broadband.isp.exatt.net)
09:17.36JerJeryou are at the station end of the line
09:17.37pranavhello everyone
09:17.48JerJerso you have to use FXS signalling, which means you need an FXO device
09:17.54QwellJerJer: Who says that?
09:17.56JerJersince FXO uses FXS signalling
09:18.03JerJerand FXS uses FXO signalling
09:18.15pranavhi qwell , sorry i got disconnected
09:18.40JerJerwonderful circular dependency
09:18.53pranavi have pasted the extensions.conf and the sip.conf in the pastebin.ca/6113
09:19.15Qwellpranav: And I said I couldn't get to pastebin.ca ...anyhow, bed time
09:19.47pranavthats ok
09:21.04pranavcan anyone else check, i am able to call internally, mobile and to pstn, but i am not able to make fwd calls
09:21.33JerJer_. is EVIL
09:21.33DaBigMacok so the tdm01b is a full height PCI card? do they come in half height?
09:21.36JerJerPURE EVIL
09:21.45JerJerDaBigMac: nope
09:21.58DaBigMacdamn
09:22.05DaBigMacnew PC time :-)
09:22.25JerJerand _7. is technically invlaid
09:22.26JerJerinvalid
09:22.32JerJeruse _7X.
09:22.50JerJerDaBigMac: yes you want to make sure it is newish
09:22.56*** join/#asterisk jerlique (jerlique@lnk254.adl0.adsl.esc.net.au)
09:23.03JerJerthey use the new PCI standard thingy
09:23.28DaBigMac2.2
09:23.29pranavi get an error saying that got sip response 404 "not found " back from 69.90.155.70
09:23.31DaBigMacyup
09:24.22pranavand when i type http://69.90.155.70/ the fwd page opens
09:24.30TheEmperorJerJer: which softphone you recommend that has instant messenging?
09:25.14jerlique~docs
09:25.15jbotDocumentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org
09:25.18JerJerTheEmperor: none
09:25.20JerJer7960
09:25.36TheEmperorJerJer: 7960 has instant messenging?
09:26.04JerJerpranav: then whatever you are sending to that host is not finding anything that matches
09:26.25JerJerTheEmperor: not in the typical sense no
09:26.39JerJerbut I have written an instant messaging service for those damn phones, yes
09:26.58TheEmperorJerJer: those phones are expensive :)
09:27.19JerJeryes, yes they are
09:27.39TheEmperorJerJer: any other recommendations aside from 7960
09:28.11JerJer7970
09:28.34TheEmperorok...
09:30.05JerJerThat Hitatchi wifi phone looks promising
09:30.49DaBigMacthanks for the info guys, have a good evening/day/morning/night whatever :-)
09:31.07TheEmperorJerJer: what billing system do you use for NuFone? Mysql database?
09:31.18JerJeri wrote my own
09:31.21*** part/#asterisk DaBigMac (~JJ@203-173-48-1.dyn.iinet.net.au)
09:31.21JerJerand mysql, yes
09:32.20*** join/#asterisk soundguy (~soundguy@zeus.soundguy.id.au)
09:34.07ScythelXJerJer: how many t1s does nufone have
09:36.06brc_http://despair.com/achievement.html
09:37.17JerJerScythelX:  t-1s lol
09:37.21JerJerwe have three DS-3s
09:37.28ScythelXoh wow
09:37.34ScythelX$$
09:37.45JerJernot $$
09:37.55JerJerthe loop is all of maybe 40 feet long
09:37.57JerJer:P
09:38.22*** join/#asterisk SplasPood (~jwb@paravolve.net)
09:42.19JerJeri hope pranav doesn't try to run VoIP on the same connection he is IRC-ing from
09:42.43modulus_i hope he does
09:43.56jerliqueanyone help with agent logins/logouts?
09:49.12*** join/#asterisk GMsoft (~r0_ot@gmsoft.developer.gentoo)
09:49.14GMsofthi
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09:50.08sysdefhi
09:51.16*** join/#asterisk multrix (~chatzilla@ALyon-252-1-19-10.w82-122.abo.wanadoo.fr)
09:53.49JerJerhoe
09:54.27modulus_eoh
09:57.56*** join/#asterisk CleanerX (~nix@nat-ph3-wh.rz.uni-karlsruhe.de)
09:59.54jerliquewhere is the structure/programming/information which describes how queues handle functions like queue-youarenext
10:01.05GMsoftanyone have experience with chan_bluetooth ? I'm getting SCO connection reset by peer each time I try
10:06.15JerJeri cannot even make my bluetooth dongle work in linux at all
10:07.07JerJerperhaps you need to pay the SCO license fee
10:07.14GMsofthaha :)
10:09.01*** join/#asterisk Zeeek (~Zeeek@Zeeek.sustaining.supporter.pdpc)
10:20.17jerliqueis everyone in bed?  this channel is quiet as
10:24.27*** join/#asterisk darkpioneer (~Pioneer@spc1-hava1-4-0-cust101.cosh.broadband.ntl.com)
10:25.47Zeeeknight
10:26.04darkpioneeri have an 2 x100 cards in my asterisk server and when im makeing calls with them i get like "pip" noises on the line
10:26.22Zeeekremove the seeds
10:26.40darkpioneer?
10:26.41Zeeekdarkpio have you checked IRQ sharing?
10:26.52darkpioneerill have a look
10:26.56ZeeekI have 2 X100P
10:27.05Zeeekso I know of which I speaks
10:28.37darkpioneeralso get the pip noises on incomeing calls
10:30.43ZeeekIRQ?
10:32.23darkpioneeri have to admint, i havenot cheqed irq befor so i dont know what to type
10:33.25ZeeekEither I have to go look or you have to go look to find that info. Which is more logical? Make sure your cards are not sharing IRQ and check some docs to find out more
10:33.27ZeeekStarter tutorial:
10:33.27Zeeekhttp://www.onlamp.com/pub/a/onlamp/2003/07/03/asterisk.html
10:33.27Zeeekhttp://www.onlamp.com/pub/a/onlamp/2004/01/22/asterisk2.html
10:33.27Zeeekhttp://www.automated.it/guidetoasterisk.htm
10:33.27ZeeekTHE reference of the moment:
10:33.29Zeeekhttp://www.asteriskdocs.org
10:33.49darkpioneerroger
10:33.51darkpioneerthanks
10:34.04Zeeekok - if I knew I'd tell you, but I can never remember :)
10:34.14darkpioneerk
10:34.22ZeeekOS dependent anyway I guess
10:34.46*** join/#asterisk jofa (~jofa@a80-127-56-82.adsl.xs4all.nl)
10:34.52darkpioneerFC3
10:36.09ZeeekI use slackware and I think the info in in /proc/interrupts
10:37.11darkpioneer<PROTECTED>
10:37.36darkpioneerahh
10:37.39darkpioneersoundcard
10:37.40darkpioneerhmm
10:37.43Zeeekdump it!
10:37.52Zeeekand where's the second X100 ?
10:38.03darkpioneer<PROTECTED>
10:38.16darkpioneerhmm
10:38.20darkpioneerremoveing soundcard....
10:38.28ZeeekI'd pull the sound card to check, then if it is that, try moving it around or playing with BIOS
10:38.34darkpioneeryeah
10:38.44darkpioneeri dont need the soundcard anyway
10:38.52ZeeekI have a PIII-800 in an ASUS mobo and three digium cards
10:39.04Zeeekthe hardest to install was the TDM400P
10:39.13Zeeekbut that one finally settled down too
10:39.42ZeeekI disabled all interrupts like USB, etc. The box does nothing but asterisk - and I did pull the sound card as well
10:40.01Zeeekit still has an unneeded mouse which uses an IRQ but there are several free ones anyway
10:40.32darkpioneeri was trying to use the soundcard for a pa
10:40.50Zeeekyou can maybe move it or change IRQ in BIOS
10:40.55darkpioneeryeah
10:41.01darkpioneerill have to have a look
10:42.23darkpioneerdamit, forgot the soundcard was onbord. ill have to dissable it in the bios
10:42.39Zeeekif you have to go to bios, see if you can change the IRQ
10:43.00darkpioneeryeah
10:43.30darkpioneerbrb
10:44.00GMsoftI've got a voice modem. which chan should I use ? would chan_capi works ?
10:44.07Zeeekno idea
10:55.09Zeeekstill trying to figure out why no callerid on ONE phone
11:00.32zoachan_modem
11:00.35zoabut trust me
11:00.38zoait wont work :p
11:00.53GMsoftheh
11:01.42GMsoftI'm trying with chan modem but asterisk doesn't seems to like my modem :)
11:02.12zoathats normal
11:02.18zoayou need a wildcard
11:03.05GMsoftyou mean for the extention ? that's not the problem
11:03.23GMsoftchan_modem fails to configure the modem and then stop asterisk
11:06.40*** join/#asterisk HuangDi (TheEmperor@218.111.51.155)
11:08.10darkpioneerhow do i start zaptel?
11:08.20darkpioneerseems i doesnt like starting automaticly
11:09.57*** join/#asterisk rajo (~rajo@graphics.cs.uni-sb.de)
11:10.47*** join/#asterisk maik_ (~maik@scumm.cs.uni-sb.de)
11:11.15Zeeekagain
11:11.16ZeeekStarter tutorial:
11:11.16Zeeekhttp://www.onlamp.com/pub/a/onlamp/2003/07/03/asterisk.html
11:11.16Zeeekhttp://www.onlamp.com/pub/a/onlamp/2004/01/22/asterisk2.html
11:11.16Zeeekhttp://www.automated.it/guidetoasterisk.htm
11:11.16ZeeekTHE reference of the moment:
11:11.17Zeeekhttp://www.asteriskdocs.org
11:11.35Zeeeklook at automated - there is a complete guide from a-z for zaptel
11:11.39*** join/#asterisk Fanguin (~Fanguin@p50819BF0.dip0.t-ipconnect.de)
11:12.13*** join/#asterisk Tili (~Tili@202-133-65-239-dialup.sat.net.pk)
11:13.12darkpioneerright
11:20.53*** join/#asterisk lilo (lilo@levin-pdpc.staff.freenode)
11:23.10*** join/#asterisk pranav (pranav@202.149.48.198)
11:24.53pranavhello everyon
11:25.15Zeeekhi pranav - I'm afraid to ask...
11:28.44*** join/#asterisk pranav (pranav@202-149-48-198.broadband.isp.exatt.net)
11:28.59pranavsorry i got disconnected
11:30.00pranavi am still facing the same problem , the calls to fwd are not gong
11:30.11pranavgoing
11:30.22Zeeekhow far have you gotten?
11:30.28Zeeekeverything else works?
11:30.32Zeeekto SIP or IAX?
11:32.06pranavthrough sip
11:32.14pranavya everything else works
11:32.20Zeeekyou have other SIP providers working though?
11:32.50*** join/#asterisk sysdef (~sysdef@pD9560EB9.dip.t-dialin.net)
11:32.56pranavi can call to pstn, mobile and also internally
11:33.21Zeeeknone of that answers my question though which is: you never have made a SIP call work to a provider ?
11:34.05pranavya  have not made any call to sip provider
11:34.26Zeeekwhy don't you get another free account somewhere else and try it? It will only take 5 minutes
11:34.55Zeeekthere's like2phone, gossiptel...
11:35.14Zeeekor if you speak German or Italian... providers in those places
11:35.27pranavno i know only english
11:35.44Zeeekand what else?
11:35.45pranavso i'll create another fwd ccount
11:35.53Zeeekno that won't prove anything
11:35.58pranavand other indian languages
11:36.05Zeeekhindi ?
11:36.11pranavyes ofcourse
11:36.17Zeeekand another 14 dialects ? :)
11:36.28pranavare you from india aswell
11:36.45Zeeekfar from it! but I have worked with a few in Huntsvill, AL years ago
11:36.56pranavok
11:36.59Zeeekthat is ironically where asterisk is (digium)
11:37.18Zeeekanyway I suggest you try an account at a different provider first
11:37.45Zeeekin fact - you could get a test account at voipjet if they are still offering them
11:37.51Zeeeklet me see
11:38.18ZeeekYes, look here: https://www.voipjet.com/join.php
11:38.22shido6u dont like NuFones sip termination?
11:38.37Zeeektalking about FWD
11:39.32Zeeekvoipjet has a nice feature with that test account thing. You can get it working before paying :)
11:39.57pranavok let me try with voipjet
11:40.07Zeeekpranav or try one of those other free ones. Also IAX with iaxtel
11:40.16Zeeekin fact IIRC voipjet is IAX
11:40.40pranavok
11:41.17pranavok see i have a sip provider in uk , if i want to connect to that then what should i do?
11:41.26*** join/#asterisk lilo (lilo@levin-pdpc.staff.freenode)
11:41.37Zeeekwho are you talking about
11:42.03pranavsee we have our own server in uk , i want to connect to that
11:42.43Zeeekwhich one? Free?
11:43.07ZeeekI just noticed I have a DID in UK from gisspitel
11:43.13Zeeekfree. Gossiptel
11:43.25pranavok gossiptel is free
11:45.17*** join/#asterisk gambolputty (~gambolput@cblmdm65-221-51-80.buckeye-express.com)
11:45.34*** join/#asterisk lilo (lilo@levin-pdpc.staff.freenode)
11:48.35Zeeekheh I had to try about 80 emails before I could get my gossiptel password sent to me
11:52.07ZeeekI must have about 15 providers - gotta clean those files up some day
11:55.55Mw3hm
11:56.01Mw3i can't sign up to gossiptel
11:56.14Mw3it said there were too many credit card fraud from hungary :(
11:58.25Zeeekawwww geee
11:58.53Zeeekthere are about 90 others you may be able to sign up for
11:59.14Zeeekyou need a UK DID?
11:59.40Zeeekdoes everyone realize that these free DID in the UK are VERY expensive for the person calling?
12:02.52*** join/#asterisk Xenesis (~Xenesis@212.127.97-84.rev.gaoland.net)
12:04.25murangdZeeek: do you know any cheaper providers than voipjet for call termination for the carbriean islands
12:07.16Zeeekmurangd you have been asking the same two or three questions for several days here
12:10.07Mw3Zeeek: why so expensive ?
12:10.41Zeeekwhat uk did? Dunno - but nothing is ever free - guess they have to make money, right?
12:11.25*** join/#asterisk rkjpl (~rk@adsl-209-233-135-56.dsl.lsan03.pacbell.net)
12:11.40Mw3that's right
12:12.02Zeeekso the best thing is to first get free accts with FWD and IAXTEL
12:12.15Mw3by the way can i get somewhere DID from other countries than UK
12:12.22Mw3for free
12:12.25Zeeekmake sure your friends get on voIP and it'll be free
12:12.40ZeeekBagdad?
12:12.50Mw3no :), pragha
12:12.55sysdefgermany
12:13.07sysdefnikotel ?
12:13.12Zeeekah
12:13.33Mw3i've uk did and when my brother was in .uk he could call us for about nothing (local call price) ...
12:13.34sysdefhas also numbers from london
12:13.43Mw3he will go to pragha soon :)
12:13.48ZeeekMw3 where was that UK DID?
12:13.52murangdZeeek: no one have provided me with a good answer
12:13.58Mw3Zeeek: fwd
12:14.12ZeeekcallUK? Far from free
12:14.29Mw3cheaper than calling hungary from uk
12:14.35Mw3much cheaper
12:14.54Zeeekthere are very few good reasons to call hungry from anywhere that's why.
12:17.31Mw3it's a good reason that i've been living there :)
12:17.44Zeeekeven so, I wouldn't :)
12:17.54Zeeekpay to call u
12:22.40*** join/#asterisk zotz (~zotz@24.231.32.191)
12:24.00*** join/#asterisk ckruetze (~ckruetze@i3ED61A1E.versanet.de)
12:29.52Zeeek.
12:33.14*** join/#asterisk ZX81 (matt@222-153-114-115.jetstream.xtra.co.nz)
12:33.40ZX81:)
12:33.57ZX81hello all
12:33.59RaYmAn-BxZeeek: sipgate does provide free proper UK DID...admittedly they do require you to sign up from a UK ip address but still.
12:34.22RaYmAn-Bxand obviously they can't change the price of geographic DIDs like they can with some of the 08XX numbers
12:34.37ZX81downloading message 114 of 582...
12:34.41ZX81omg 1 day!!!
12:34.43ZX81lol
12:35.15ZX81~ping
12:35.35jbotpong
12:35.45ZX81heh it works lol
12:36.07ZX81airpor (KL) -> home (NZ) -> irc
12:36.08ZX81:)
12:36.12ZX81ok brb~
12:40.44*** join/#asterisk postel (~jp@postel.user)
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12:47.32*** join/#asterisk zeedo (~notroot@www.bsrf.org.uk)
12:47.47Zeeekthose numbers cost a huge amount to call - I have sipgate, calluk and gossiptel they are all "national" no's
12:48.07Zeeekgo look at the rates - they are higher than calling a european num from a regular phone
12:48.38*** part/#asterisk zeedo (~notroot@www.bsrf.org.uk)
12:48.41*** join/#asterisk jhiver (~jhiver@ABoulogne-102-1-3-10.w193-253.abo.wanadoo.fr)
12:52.33RaYmAn-BxZeeek: outgoing or? They can't control the price on geographic DID's...
12:53.21Zeeekjust call one of those numbers and find out - my wife called me from London once - it was outrageously expensive
12:53.34Zeeeklike maybe 10c/min ?
12:53.42Zeeeklook up BT UK national rate
12:54.36RaYmAn-Bxthere's a difference between 08XX numbers are 01/02 numbers
12:55.03Zeeekgive me an example of a free DID in London and the rate to call it?
12:57.04ZeeekI would love to be wrong - it would be nice for anyone who needs a FREE DID
12:58.58RaYmAn-BxI have an oxford DID free from sipgate...
12:59.18RaYmAn-Bxit costs the same as it would cost to call any other oxford number
12:59.19Zeeekand what does it cost per mintue to call this thing from there?
12:59.25Zeeekwhich is?
12:59.45RaYmAn-Bxno idea. My point is that it cost EXACTLY the same as calling any other UK number...
13:00.00Zeeekmaybe BT is just way expensive
13:00.28RaYmAn-Bxthe price to call it depends on your phonecompany..with "BT Together Option 1" calls are 3p per minute for daytime calls
13:00.38RaYmAn-Bxthat is stupidly expensive but that's just the price there
13:00.47RaYmAn-BxI can call UK from denmark for around 2p/min any time of the day
13:00.53Zeeekby the way, nowadays there are several companies offering unlimited national dialing for insanely cheap rates
13:01.27RaYmAn-Bxbut my point is rather that the dids aren't any more expensive than any other UK number
13:01.37Zeeeklike $20/mo including 8Meg DSL
13:02.03Zeeekwhen I signed uop for a london number it was... I haven't needed one since but then it was only national rates
13:02.21ZeeekI still have a sipgate.de acct and maybe even calluk if they don't purge their files
13:02.46Zeeekthe national numbers are "bend over!"
13:03.13Zeeekcompared to say a nufone toll free at 2c/min
13:04.06*** join/#asterisk Koshatul (~evangelio@202.9.38.223)
13:04.06ZeeekDoes sipgate have an english lang site now? They didn't when I signed up. Maybe I pressed the "Bend over" button?
13:04.15RaYmAn-BxI have heard of special london numbers that are national from anywhere
13:04.33RaYmAn-Bxbut yeah, the 0870 and 0845 are generally extortionate prices
13:04.49RaYmAn-Bxbt even mentions them as premium rate on some parts of they website :>
13:05.07Zeeekthey're INCREDIBLY expensive - anyone point taken
13:05.32Zeeeksince I haven't needed the numbers. I wonder what voiptalk is offering these days? I have an acct there as well
13:05.52RaYmAn-Bxanything non 01xxx codes should be stayed clear off imho :>
13:06.00ZeeekI'l looking at voiptalk
13:06.03RaYmAn-Bxand yeah, sipgate has sipgate.co.uk now as well
13:06.28RaYmAn-Bxand they only give out proper DID's to people who connect from a UK ip address (And can give a UK address)
13:06.45ZeeekTelappliant can now provide you with an 0845 number, which can be configured to point to your IP phone or IP PBX, enabling callers to dial into your IP network via a conventional landline telephone number.
13:06.53Zeeeksupposedly "local UK"
13:06.58RaYmAn-Bxyeah
13:07.00RaYmAn-Bxthat's bullshit
13:07.03RaYmAn-Bxit's not nocal uk
13:07.03Zeeekcharged to the caller at the same rate as a local rate UK telephone call.
13:07.08RaYmAn-Bxit's Lo-Call
13:07.18RaYmAn-Bxthere's a big difference
13:07.35Zeeekone time 10£ charge for those
13:08.00RaYmAn-Bxdo they claim it's charged at the same rate as local rate?
13:08.08RaYmAn-Bxif so then that's a gigantic lie
13:08.19ZeeekI'm quoting the site above
13:08.49RaYmAn-Bx0870 is supposedly national rate as well...but generally it's a lot higher (and most companies are starting to offer same price for local and national as well...excluding 08XX)
13:08.50Zeeekfax to email. Nice
13:09.29Zeeekmy cust in England moved to Geneva. As a bonus, callerID usually works now too. It never did from UK
13:09.46Zeeekso I need no UK no's which is why I have three I never use
13:11.23Zeeek40£ /yr for a local UK no
13:11.50Zeeek01865
13:11.55Zeeekfor oxford
13:12.41*** join/#asterisk pranav (pranav@202.149.48.198)
13:12.44RaYmAn-Bxand sipgate gives that free.
13:13.05Zeeekcan you have as many as you want?
13:13.11RaYmAn-Bxno
13:13.15RaYmAn-Bx1 only I think
13:13.29*** part/#asterisk pranav (pranav@202.149.48.198)
13:13.36ZeeekWere gonna be in Switzerland at some point too
13:13.43ZeeekThey were
13:29.37jhiverarrrgh guyz I have something kinda strange
13:29.54jhiverwhen I place VoIP calls => no echo, but with X100P card => echo
13:30.20jhiver(well, X100P "clone" card in fairness...)
13:30.23Zeeeknot that strange
13:30.40jhiverso... u know how to troubleshoot this?
13:31.16Zeeekread the wiki first
13:31.28Zeeekvolumes  written on ech
13:32.29coppiceand a little of it is even accurate :-)
13:32.41Zeeeksturgeon's law
13:32.48Zeeek90% of everything is crap
13:32.59jhiveralso, how do I reload zaptel modules? do I need to reboot the box or is simple asterisk restart ok?
13:33.14Zeeekrestart should do it
13:33.24coppicethat's not true. at least 98% of everything is crap
13:33.31Zeeekheh
13:33.48Zeeekno that can't be because that self invalidates what you just said :)
13:34.02*** join/#asterisk ariel_ (~Ariel@dsl-20-177.cofs.net)
13:34.10coppicehow?
13:34.41Zeeekbecause you (or me) saying everyhting is crap invalidates 90% of what we are saying too
13:35.02Zeeekso only one in ten of my responses can ever be correct
13:35.30coppiceah, you bring this 90% measure right down the sentance level?
13:35.43Zeeekto the letter level and beyone!
13:35.48postelAm i the only one that cant get the damn BT line to Hangup on a X101P (NOT a clone)
13:35.55Zeeekthere is a difference between é and e
13:36.20Zeeekpostel usually it hangs up automatically right at the beginning of the call
13:36.34postelheh
13:36.59ZeeekI would take a gander at the zapata.conf sample file
13:37.20postelwell, been there, sone that, got the tshirt and everything
13:37.23Zeeekand perhaps the mailing list for BT woes
13:37.24postelno cigar.. :/
13:37.33Zeeekprogress and all that?
13:37.43postelyeapers
13:37.53posteli have 64386434834638468 google tabs
13:38.01Zeeekso the $90 saved in not buying digium....
13:38.08postelIT is digium
13:38.23Zeeekoh yeah that was jhiver that had the close
13:38.28Zeeekclone
13:38.43coppiceso the $90 wasted in buying Digium.......
13:38.52Zeeek90% 90%
13:39.03postelcoppice: hehe
13:39.20Zeeekcoppince out of nowhere, faxes are now being received! Mostly spam faxes unfortunately
13:39.34Zeeekbut technology doesn't read them
13:39.45Zeeekjust saves them and sends em on the me :)
13:41.09ZeeekI swear there are patches for UK polarity reverse shit or whatever it is?
13:41.45posteland for some stupid reason it Dials to SIP on incoming (as it should) but the damn analogs that are on the line go crazy, ringing like there's no tommorow, a constant pitching tone that blows your drums away
13:42.12Zeeekwhat line?
13:42.17postelBT
13:42.31Zeeekyou left phones on the asterisk lines?
13:42.44ZeeekI've done it but ist isn't good
13:43.08coppiceall telcos want to be like BT, but most  just can't seem to drag themselves down to that level
13:43.16postelwell, when * answers the call they should STFU right?
13:43.40Zeeekno you shouldn't have other shit on the X100 lines
13:44.18postelthats a new one to me
13:44.27posteldanka Zeeek
13:44.29Zeeekit's common wisdom AFAIK
13:44.44postelwell, aint
13:44.48Zeeekit should screw up the answering though, but who knows?
13:45.09ZeeekWhen I first began at the office we left the phones hooked up. Bad.
13:45.17RaYmAn-Bxdoes it actually answer before the SIP extension answers?
13:45.21Zeeekfirst they ring for several rings more after answered
13:45.37Zeeeksecond, asterisk doesn't knbow if someone is on the phone
13:46.32postelZeeek: you're in the uk?
13:46.42Zeeeknot if I can help it!
13:46.46postelheh
13:46.56postelthats my boy
13:46.56Zeeekweather is just as bad here, so what would the point be?
13:47.05ZeeekParis, FR
13:47.16postellet me tell ppl to remove all analogs
13:47.25postelyou're close enough
13:47.32Zeeektry it and see if it helps your problem
13:47.43postelim back in a sec
13:47.53ZeeekI have one phone that doesn't show callerid here
13:53.43*** join/#asterisk zotz (~zotz@24.231.32.191)
13:58.03*** join/#asterisk kiran (~kiran@202.62.88.140)
14:02.01kiranhi all
14:06.29ariel_morning all
14:06.44kiranmorning.........
14:18.48jhivermore like 'nite over here :)
14:18.56jhiverwell, getting late neway
14:19.02Zeeekcloudy?
14:19.21jhiveryup it's been pouring down for the last 3 weeks or so
14:19.34Zeeekdid you try wengo btw?
14:19.35jhiver'tropical depression' they call it over here... => lots and lots of rain
14:19.51jhiverdidn't no
14:19.59Zeeekyou have family or freinds here?
14:21.40*** join/#asterisk Kumbang (~ecvs@167.205.24.4)
14:22.15jhiverhere? where?
14:23.37ZeeekFR Métropole
14:23.43jhiveryeah
14:23.55jhivergot a few mates in bordeaux
14:24.01Zeeekwe've talked before - you have a "slow" DSL yes?
14:24.09jhivercall them occasionally tru voipjet, works fine
14:24.17jhiverI used to have ISDN
14:24.25Zeeekwengo is 6eu/mo unlimited
14:24.26jhiverbut yesterday finally got DSL
14:24.33Zeeekah congrats !
14:24.44jhiver512/128, better than 64/64 :)
14:24.53Zeeekabsolument
14:25.04jhiverbut it really more like 300/120...
14:25.11jhiverah well
14:25.19jhiverreally much better for shaping
14:25.23Zeeekstill... beteer than ISDN
14:25.37jhiverI have set up a linux box as ethernet bridge and I do shaping at that level
14:25.43jhiverso the whole LAN is shaped
14:26.34jhiverworks nicely... I have bittorrent, ssh and the family surfing like crazy and it's nice sound quality for VoIP
14:27.52jhiver6 EUR unlimited is pretty cool... I'll have to try that @ some point
14:29.21ZeeekYou have a geo DID "free" with it
14:29.42Zeeekalthough I'm not sure about DOM-TOM
14:29.44jhivernice
14:29.54Zeeekwouldn't be goe but 08 something
14:29.57*** join/#asterisk doughecka (~Doug@doughecka.user)
14:30.15Zeeekhttp://www.wengo.fr/assistance/forum/viewforum.php?f=9
14:30.20dougheckawoot, its still my birthday
14:30.32jhivertoo bad
14:30.43jhivertomorrow you'll be ok for about a year
14:30.48Zeeekhttp://www.wengo.fr/assistance/forum/viewtopic.php?t=1793
14:30.53ZeeekMartinique ^^^^
14:30.57dougheckahaha
14:31.30Zeeekbetter yet jhiver: http://www.wengo.fr/assistance/forum/viewtopic.php?t=1736
14:34.07*** join/#asterisk Mike_TK (~Mike_TK@212.165.78.5)
14:41.32Zeeek.
14:41.40file[laptop]..
14:41.51Zeeek...  ....  ..  __
14:41.59file[laptop]..?
14:42.04Zeeekmorse code
14:42.15Zeeek... S
14:42.16file[laptop]ah
14:42.18Zeeek.... H
14:42.23Zeeek.. I
14:42.26Zeeek___ T
14:42.35*** join/#asterisk eKo1 (~bernd@207.42.191.66)
14:43.04Zeeekeveryone should know morse code. What if you're lost in a cave?
14:43.38file[laptop]then I'll get out of the cave and yell, "LONELY UNGUARDED FEMALE SEEKING COMPANIONSHIP" and see who shows up
14:43.46eKo1Zeeek: If you're lost in a cave, morse code is not going to get you out of there.
14:44.13ZeeekWhat you never saw any Sagal or STallone movies?
14:44.24Zeeekor submarine ones?
14:44.28file[laptop]so if it's an old movie, too bad
14:44.32file[laptop]I don't watch many movies either
14:44.58Zeeekthose aren't that old - I didn't expect to discuss "It's a Wondeful Life" though
14:45.24Zeeekanyway in all these moves people are buried alive and tapping in morse
14:45.49eKo1Zeeek: Yeah, but that doesn't help if nobody else understands it.
14:46.05jhiverthis ART stating that it's illegal to redirect DIDs to elsewhere is entirely bollocks!
14:46.05Zeeekexactly! that's why it should be mandatory
14:46.11jhiverjesus
14:46.20Zeeekno one pays attention tot hose
14:46.26Zeeekto those
14:46.30eKo1People will just think the tapping is arbitrary and ignore it.
14:46.35file[laptop]oh, hrm, I had more money then I thought
14:46.40ZeeekNO! Not if you're any good at it!
14:47.07jhiverdi di di da da da di di di
14:47.14Zeeekthere's only one flaw. Everyone will have their iPods on so they won't hear
14:47.38Zeeekthose are dits, not di
14:47.55Zeeekdit dit dit, dah dah dah, dit dit dit
14:48.06Zeeekfaster than callerid and SMS
14:48.10eKo1What is that? Baby talk?
14:48.35Zeeek..__. .._ _._. _._
14:48.50eKo1Much better.
14:48.58Zeeeka very poor dialplan IMO
14:49.13eKo1But let's face it, if morse code was any good, then telegraphy would still be around.
14:49.14Zeeek..__. .._ _._. _._,1,Dial(SIP/2000)
14:49.22ZeeekIT IS AROUND!
14:49.53eKo1Zeeek: You need to get out of that 1890 time hole.
14:50.06Zeeekthink of the joy of discussing like we do here, but at the lightening peeds of 20 wpm!
14:50.30LuhiwueKo1, i know lot of examples where morse code is still in use here in Argentina
14:50.33Zeeektelegraphy is still a requirement for radio licenses even tho they have satellite
14:50.42Zeeekon ship
14:50.55jhivercq cq dx guys..
14:51.03jhiver'nouf morse
14:51.05ZeeekI know a ship's radio op that has seamen on her passport :)
14:51.20jhivernot enough speed with morse... sucks
14:51.23Zeeekbug laugh in every port
14:51.26jhiverI have a much better protocol
14:51.29eKo1They don't use morse on ships, they use telex.
14:51.33ZeeekRTTY
14:51.39jhivertap = 1, no tap = 0
14:51.40jhiver:)
14:52.00ZeeekI used to receive RTTY on my TRS80 thgru the casette port
14:52.03file[laptop]someone write app_morsecode!
14:52.19Zeeektoo easy
14:52.40Zeeekfor the deaf - they can hold the receiver and feel the message
14:53.10Zeeek. . . . __   . . __
14:53.14Zeeekthat is the answer
14:53.14eKo1They can also look at it from their cell phone.
14:53.42Zeeek. . . . __   . . __,1, Dial(ZAP/1/42)
14:54.05file[laptop]woot 42
14:54.59Zeeekwhen is digium gonna make a deal with motorola like skype has done?
14:55.20Zeeekmotorola is building skype into cellphones
14:57.24Zeeekso supposedly when you're near a wifi point, you can call via skype
14:57.47Zeeekwhich is of no interest except the fact that the app will be built in
14:57.54Zeeekto the phones
14:58.01GodseyI would bet the driving force would be vonage :)
14:58.16Zeeekif only they were using IAX
14:58.18LuhiwuZeeek, the application will be skype? i've heard about configurable voip, not exactly skype
14:58.31Zeeekthis is a deal done a few days ago
14:58.56Luhiwusomebody calls Nokia now! :)
14:59.32Zeeekhttp://www.technewsworld.com/story/wireless/motorola-skype-mobile-voip-40622.html
14:59.52Luhiwutnx for the link
14:59.53Zeeekin fact, http://news.google.com/news?q=motorola%20skype&hl=en
15:00.21coppiceI wonder how commited anyone is to UMA
15:00.44Luhiwui've made a iax softphone based on the iaxclientocx library, maybe i should convert it to java and upload it to some java based phone...
15:00.59ZeeekIt would be worthwhile trying that
15:01.22ZeeekThey're mostly commited to NBT
15:01.35ZeeekAll business is looking at NBT
15:02.03Luhiwuwhat is NBT?
15:02.05Zeeekthe NextBigThang (tm)
15:02.11Luhiwuthanks :)
15:02.31ZeeekWAP yesterday, GPRS today, UMA and then.... NBT!
15:03.11coppiceWiMAX? WiNOT!
15:03.52Luhiwumaybe we all should put some money in an asterisk bounty to finance a java based softphone for cells... i don't think i could port it to java alone, but i'd help if the proyect begins...
15:03.53ZeeekInteresting: Spokespersons at both Skype and Siemens said there are no immediate plans to market the connector and phones in North America.
15:04.05Zeeekreferring to the Siemens cordless USB handsets
15:04.55coppicepeople are moving on from the NBT to the one after next big thing. most 3G networks are not yet on the air, and some operators are already taking about a migration to flash-OFDM
15:05.19Zeeekyou know that 3G backfired into a major fiasco anyway
15:05.30Zeeekthey paid zillions of the licenses
15:05.46coppicewell, the phones are networks are starting to trickle out now
15:06.01coppicestill very troublesome, though
15:06.32ZeeekI've seen ads for video on cellphones, I spose that's using UTMS of whatever it is
15:06.40coppicethey get a rush of early adoptors, then sales dry up. then they need to discount to below GSM
15:07.05Zeeekall that cell stuff has been a license to print money over here in EU
15:07.14Zeeek8 year olds have 2 cells each
15:07.22coppiceyeah, but the video over UMTS is a joke. the only reason they can do that is there are few subs. with more subs they won't have the capacity
15:08.13Zeeekbad enough housewives are talking on the bloody things in the supermarket aisles
15:08.25coppicethey are signing up people for one year of unlimited video at a low price. this means they have no expectation of lots of subs within one year
15:08.34Zeeek90% of the planet has never seen a doctor even once
15:08.51coppicewow, that healthy, eh? :-)
15:08.59Zeeekya, amazing eh?
15:09.15Zeeekthey should tax "where are you?" cell calls about 500%
15:09.19coppice1 in 5 humans now carries a GSM phone.
15:09.27Zeeekgot one in my pocket right now
15:09.45Zeeekand I have almost no need for it
15:09.48Luhiwucoppice, i know a lot of not-so-humans that carries a GSM phone :)
15:10.05ZeeekI refuse to sign a sub though
15:10.57Zeeekkinda nice now that we have SMS working on asterisk - we can send each other SMS from the office asterisk on any browser
15:11.19coppicereally very poor people have them. the compulsion to have them seems really string
15:11.19Zeeekso for $15/mo it's like a fancy pager
15:11.59ZeeekAs I walked by him, a beggar sitting on the sidewalk pulled one out to see if he had any messages!
15:12.25coppicefor $15 we get 1500 minutes airtime, so few people send SMSes
15:12.33Zeeekthey're powered byt he kids over here - I'd hate to be a parent at the moment
15:12.59Zeeekwhen I'm out on business I can't answer my phone but I can read SMS
15:13.18coppicepeople here don't care :-)
15:13.19Zeeekand all incoming are unmimited
15:13.27Zeeekunlimited even
15:14.03coppicewe have the same deal. I can send and receive for free, but people still don't use them
15:14.30ZeeekI could send commands to asterisk from SMS
15:15.01Zeeekanother kind of DISA :)
15:15.34Zeeek"With over 68 million downloads of their client in the last 18 months, we believe Skype is a natural fit with our vision of simple and seamless connectivity for our consumer customers around the globe."
15:16.07ZeeekI wonder how many asterisk downloads there have been in the last 18 mo?
15:16.51Godseywe pay cingular something like $40/line and each user has 200 minutes :)
15:17.56Zeeekyou may as well pay a hooker and get 3 min for $100
15:18.29ZeeekBy making Voice over IP truly mobile and easily accessible, we can make
15:18.29Zeeekcommunications seamless for consumers as they travel throughout the
15:18.29Zeeekenvironments of their day - at work, at home, in the car, or out in the
15:18.30Zeeekworld,
15:18.31ZeeekBy making Voice over IP truly mobile and easily accessible, we can make
15:18.50ZeeekSo why not put a SIP or IAX client then instead of Skype!
15:19.07*** join/#asterisk sandnigg0r (~trilluser@66-55-197-254.gwi.net)
15:23.04jhiverthing is, skype sucks because it's proprietary but you have to admit, it just works...
15:23.41jhivertoo bad it's so hard to work with and do cool stuff with it
15:23.51Zeeekit does work well. Too bad about prioritary
15:23.54*** join/#asterisk mhnoyes (~mhnoyes@user-38lc08a.dialup.mindspring.com)
15:24.22eKo1Well, it really all comes down to social networking. The people at Skype have good contacts so...
15:24.29jhiverit'll be good when somebody reverse engineers the protocol and we start to see alternative clients :)
15:24.46Zeeekok, we all need this:
15:24.46file[laptop]yes, yes it wil
15:24.47Zeeekhttp://www.ubergizmo.com/15/archives/2005/02/clothes_that_bi.html
15:25.52jhiverlol
15:25.58Zeeekok wait:
15:25.59Zeeekhttp://www.worthersoriginal.com/index.php?id=25
15:26.22Zeeekubergizmo is pretty crazy site
15:26.29eKo1I don't need that. I just rub my feet on some carpet and ZAP!
15:26.45eKo1But then again, I also get zapped.
15:26.54Zeeekhere's a little more down to earth view: "Motorola and Skype hop in bed on devices and accessories"
15:28.33coppicejhiver: once you have reverse engineered it, you will hit patent problems
15:29.02eKo1You mean license problems.
15:29.24coppiceno, patents
15:30.06ZeeekEurope does not recognize software as patentable
15:30.11Zeeekdoes the USA?
15:30.13jhivertrue
15:30.24jhiverbut it will eventually
15:30.27coppiceoh, don't start that crap. codecs are patented everywhere
15:30.40Zeeekbut it isn't the codec that makes skype what it is
15:30.41jhiverthere's massive lobbies and they just won't let go
15:31.11jhiverI mean the OEB has been accepting patents and know they're working to get them enforceable
15:31.19coppicebut for compatibility you need to use the iLBC which is patented and not free. they don't use the free one, even for narrow band
15:31.23Zeeekthe latest ruling is negative
15:31.40coppicecodec patents are enforceable, and always have been
15:31.49jhivercompatibility isn't the issue
15:31.50Zeeeklicenses aren't that expensive for each unit are they?
15:32.25coppicethey only reason you reverse engineer is to achieve compatibility, isn't it?
15:32.29*** join/#asterisk RoyK (~roy@80.239.107.80)
15:32.32Zeeekafter all the BT100 does all those
15:32.41ZeeekiLBC, 729
15:33.01coppiceno it doesn't. that's the narrow band iLBC. skype doesn't use that
15:33.07jhiverWell not necessarily... if you could make something that works just as well as skype but that's open then it would kick ass
15:33.21jhiverof course asterisk works well but it's damn hard to set up
15:33.28jhivertoo powerful for m. joe sixpack
15:33.30Zeeekcoppice ? I didn't knwo there were several iLBC?
15:33.35coppicelots of things are open and work as well as skype.
15:33.44coppicelook on the GIPS site
15:34.08Zeeekthe skype phenomenon is NOT from codecs though, it's that you start it up and it works
15:34.17Zeeekany idiot can use it
15:34.17jhiverwell, the P2P telephony thing *is* a good idea and the fact that you can have 2 NATTed skype users call each other is pretty cool too
15:34.34Zeeekso they went the extra mile on that part
15:34.47jhiverI agree
15:35.04Zeeekthe rest, no one gives a f what codec the software is using
15:35.16coppiceyep, but you need to be compatible with it. the use of widebnd plays a strong part in people's perception that for the first time VoIP is actually better than PSTN calls
15:35.22Zeeekand now, if they do make deals right and left, the name has so much buzz value
15:35.44Zeeekthey may be able to make a niche for themselves
15:36.01Zeeekalso voIP will never be sold to the public as voIP - it's the ugly sister
15:36.06coppicethere's no room for niches in telecoms. its all or nothing
15:36.11Zeeekthe public isn't geeky - they want PnP
15:36.51Zeeekskype will likely be bought by a big op RealSoonNow, no?
15:36.52ariel_actually the skype appeal is due to the kids using it.  There the ones that use it most.
15:37.03coppicedoubt it.
15:37.05Zeeekthat is also true, they power the cell revolution too
15:37.16jhiverWell if skype manages to take over the world and all phone is free then I say why not. The problem is that if they do achieve monopoly then it won't stay free very long...
15:37.16Zeeekcoppice why not?
15:37.31Zeeekjhiver skype phoning isn't free except to other PC
15:37.40Godseyit isn't free
15:37.44Zeeekthey now sell minutes like everyone else
15:37.49coppiceif a telco buys it, all the other telcos will run away. then it is worthless
15:37.54Godseynot free as in monitarialy or freedom :)
15:37.54ariel_free I don't think that the bell's will allow that since they have there needs.
15:38.06jhiveryeah but I don't think their plan is to make money selling minutes
15:38.15Zeeekthey are selling thelm now
15:38.17jhiverI think it's more like:
15:38.24jhiver1. get everybody to use us
15:38.30Zeeek[and cool accessories]
15:38.42jhiver2. argue that too many natted clients => need specialised servers
15:38.42GodseyI'm just waiting for a SBC Verizon merger :)
15:38.52jhiver3. next version only connects to skype's servers
15:39.07jhiver4. charge small amounts of money for calls
15:39.25GodseyI think phones are in a space easier to adopt ipv6 :)
15:39.40jhiverbut maybe they really do want to make it free for everybody... don't know, i don't know them :)
15:39.42coppiceif they have sense they won't charge for calls. they will charge a simple subscription
15:39.55jhiveryeah
15:40.12jhiverand premium rates for businesses of course...
15:40.14eKo1Yeah, the charge per minute model is getting old.
15:40.23ariel_subscriptions is the way that most will go just like the new napster.
15:40.34eKo1It's old and complicates my programming substantially.
15:40.58coppice90% of the code in a switch is doing accounting
15:41.05eKo1Tell me about it.
15:41.22file[laptop]MOOOOOOOOOOOOOOOOO
15:41.25*** join/#asterisk numBone (~numBone@c-24-129-204-233.se.client2.attbi.com)
15:41.32Godseyaccounting records ware what I spend the most time dealing w/ as an isp too :)
15:42.01Zaware there any TDM cards that are compatible with freebsd and asterisk?
15:42.15Godseytho I like the flexability of charging for usage or flat for the customer
15:42.27Godseypeople like both ways, and since it's $$ I do to
15:42.28Godsey:)
15:42.31Godseytoo
15:42.42dan2anybody have a sipura 2100 and know what the sipura code to turn on the wan side webserver is?
15:42.47ariel_I wish that the colo were flat rates.
15:42.51eKo1Yeah, but using a flat fee business model simplifies everything.
15:42.55coppiceas soon as you break even a little bit from the subscription model you get 100% of the problems
15:43.21*** join/#asterisk polymath (~jeffg@dsl027-163-129.atl1.dsl.speakeasy.net)
15:43.28eKo1dan2: I don't think there is a wan side to it.
15:43.32coppicepeople will always go for subscription over pay as you go
15:43.34jhiveron the other hands you have to deal with abusers with flat rate model...
15:43.36ariel_dan2, I know it's a web setting I did not know there was one for the ivr for the 2100
15:43.46jhiveror special cases like "i have lots of family in india"
15:43.48eKo1jhiver: yeah.
15:43.57jhiverso ATM it's kinda hard
15:44.05plappythats why ya just buy an uncapped link. :) thats flat rate.
15:44.28ariel_how can you abuse flat rate you charge for unlimited it should mean unlimited.
15:44.40coppiceit makes no diffence to skype. they only do the call switching
15:44.54jhiversure but termination isn't flat... you have to work out what customers are gonna cost you on average and if you underestimate your averages (or say your currency goes down) u r screwed
15:45.30polymathany chance somebody could help me troubleshoot a wcfxo problem?
15:45.31jhiverI'm talking from the business point of view, not the consumers'
15:45.31eKo1jhiver: That's why you overcharge the customer so your ass is covered.
15:45.33ariel_jhiver, that is due to the cash cows telco's
15:46.00ariel_polymath, state your problem
15:46.03jhiveryeah but _practically speaking_, doing flat rate voip service at the moment is hard
15:46.19jhiverhave you seen the prices on some mobile destinations?
15:46.23ariel_jhiver, it's a mixed bag right now.
15:46.23coppiceoh, for termination you have no choice but to work within the local framework. if local calls are charged you have no choice but to charge. you need to base a business model on increasing IP-to-IP and decreasing termination, though
15:46.24eKo1jhiver: Yeah
15:46.50polymathi'm getting a red alarm when i plug in a real rboc line to my wcfxo, but if i plug in the tel port of my iaxy the alarm clears and * takes calls
15:46.53eKo1But once those mobile destinations start using VoIP, it will all be cheaperl.
15:46.58*** join/#asterisk otiske (~otiske@kauai.sys.pas.earthlink.net)
15:46.59ZeeekI go away for two minutes and people are discussing how to make money. SHameful!
15:47.01ariel_that is why if something like dundi takes off would help us allot.
15:47.36coppiceskype can only be an interim step. the long term has no need for a middle man
15:47.40otiskehas anyone built asterisk with ICD on FreeBSD 5.3?
15:47.50ariel_polymath, what is an rboc
15:47.54polymathto be up-front, the wcfxo is a generic (winmodem).  i know that's a touchy topic, but i only need it for a month and will be buying more digium hardware
15:48.06polymathrboc == regional bell operating co  (bellsouth in this case)
15:48.19bjohnsonwhat is dundi anyway?
15:48.25bjohnsonis it like e164.org?
15:48.29polymathbjohnson: http://dundi.info
15:48.31ariel_polymath, I have mine here plugged into the bellsouth line works fine.
15:48.53coppicesoon to be tonboc = the one nation bell operating co :-)
15:48.59polymathariel_, is yours a generic or a supported wcfxo
15:49.08ariel_the real thing
15:49.19polymathcoppice, heh... i'm starting an office pool on how many years it takes to go from divestiture to 100% reconsolidation
15:50.52eKo1The trick is, you only make it look like a divestiture so you never have to reconsolidate.
15:51.05bjohnsonis dundi already in use in a public system?
15:51.10polymathariel_, when i plug in the bell line, even with wcfxo and zaptel unloaded, it's as if the line loops up -- call it and it's busy
15:51.41polymathbjohnson, http://dundi.info/members.html
15:52.04polymathbjohnson, bellster^H^H^H^H^H^H^H^Hfwdout uses dundi in a limited capacity
15:52.36ariel_polymath, use http://pastebin.ca and post your zaptel and zapata.conf files
15:53.20coppicepolymath: it won't be reconcildation. it will be building a co strong enough to compete in the new deregulated global marketplace...... which just happens to have a total monopoly whereever it operates, as it owns all the coax, pairs and fibre
15:53.39*** join/#asterisk loick (~loick@ATuileries-151-1-27-239.w82-123.abo.wanadoo.fr)
15:54.00bjohnsonpolymath: anyone else?  I guess what I'm getting at is .. how usable is it currently?
15:57.21*** join/#asterisk polymath (~jeffg@dsl027-163-129.atl1.dsl.speakeasy.net)
15:57.38*** join/#asterisk RoyK (~roy@80.239.107.80)
15:57.52polymathariel_, sorry, had an xkill mishap
15:58.57ariel_fxs_ks I use this instead of fxs_ls
15:59.26ariel_bjohnson, dundi is still in testing
15:59.52polymathariel_, i tried earlier with fxs_ks, but let me give it a shot again
16:00.03ariel_bjohnson, but I have used it for an enterprise system. Which can be great for internal asterisks systems.
16:00.52GMsoftwhat's the status of asterisk on bug endian box ? is it working ?
16:01.02GMsofterr s/bug/big/
16:02.08polymathGMsoft, * allegedly works on powerpc, which is normally big-endian
16:02.33GMsoftok. I'm compiling on parisc right now
16:02.54polymathGMsoft, linux or hpux?
16:03.09GMsoftgento linux :)
16:03.13GMsoft+o
16:03.20polymathGMsoft, nice
16:03.28GMsoftmy kbd is not cooperative today :)
16:03.33Zeeekpolymath - a while back I had a power failure. The x100P shorted the phone line even when power was restored
16:03.53Zeeekbut having removed power for 24hrs, they work again
16:04.13Zeeekit was depressing to shut down the asterisk box and plug in phones :(
16:04.36ariel_Zeeek, he says it works off the iaxy as the a source.
16:04.54polymathariel_, nod
16:05.09Zeeekok, just though I'd chime in with that useless bit of info
16:05.18polymathZeeek, thanks though
16:05.23Zeeekwhile I'm download 400 megs of music
16:05.56ariel_polymath, here is my files. http://pastebin.ca/6129
16:06.06Zeeekthe Internet is a wonderful invention for instant satisfaction of many kinds :)
16:06.47*** join/#asterisk ariel_ (~Ariel@dsl-20-177.cofs.net)
16:07.02Zeeekanother example of "if you build it they will COME"
16:07.15polymathariel_, about to try your configs now, thanks
16:07.19ariel_don't you just hate it when you press the wrong key and you exit all the program instead of others.
16:07.25*** join/#asterisk trym (~trym@linux.debian.us)
16:07.55polymathariel_, yes -- i use evilwm, which puts "kill window" right next to "switch to desktop 1" =]
16:08.51*** join/#asterisk NormAst (HydraIRC@Ottawa-HSE-ppp4116580.sympatico.ca)
16:09.16NormAstHi all.
16:10.00polymathariel_, about to have to rmmod -f zaptel (accidentally tried to rmmod it while * was running) so may have to reboot...
16:11.23ariel_try service zaptel restart
16:11.39dan2hmm... Wonder how much longer I'm going to have to stay on hold...
16:12.07ariel_dan2, is the moh good?
16:12.09polymathariel_, no luck.  reboot is in order i think
16:12.13polymathbrb
16:12.21dan2ariel_: sounds like elevator music, so no
16:12.34dan2ariel_: its quire ironic I'm calling my own companies support line
16:12.43ariel_heheeh
16:12.44Silik0nelevator music is the best
16:12.58dan2Silik0n: it sucks compare to 32 channels of xm
16:13.08Silik0nhah
16:13.13ariel_funny I dial my moh on my system allot just to listen to it.
16:13.24Silik0nsame here
16:13.44Silik0nbut I have all kindsa of crazyness on my moh
16:14.24Silik0nmost of which I doubt any person in their right mind would put on a corporate MoH tho
16:14.46Silik0ndan2 headset
16:14.56dan2Silik0n: don't have one of those either
16:17.42*** join/#asterisk polymath (~jeffg@dsl027-163-129.atl1.dsl.speakeasy.net)
16:17.51inspiredhmm, is it possible to convert a doc file to tif on linux?
16:17.59inspiredor doc to pdf to tif
16:18.16inspiredI know pdf to tif works, I just have to understand how to convert a word document to an image
16:20.18*** part/#asterisk Kumbang (~ecvs@167.205.24.4)
16:20.45RoyKinspired: openoffice can do that
16:20.47RoyKhm
16:20.55RoyKwhy tif? fax?
16:20.57inspiredopenoffice requires X :p
16:20.59inspiredyes, fax
16:21.30polymathariel_, still same symptoms with your config (adapted to my environment)...
16:21.35inspiredour users are probably too stupid to understand that they have to convert from doc to pdf on their own computer, so our machine has to do it for them
16:22.23polymathariel_, got to run, thanks for your help!
16:22.35ariel_polymath, sorry could not help more.
16:23.45Zeeekinspired sometimes a printer driver is the best solution
16:23.54Zeeekthey may understand that
16:24.22Zeeekhow are they getting their document to you?
16:24.27inspiredour customers are not going to use any fax machines. we are doing web to fax
16:24.30inspiredand fax to email
16:24.43coppiceLots of people are used to print to FAX on windows
16:24.48Zeeekwhy not Word to "print as fax"?
16:25.02ZeeekI think there are even free drivers
16:25.03inspiredhow will that help me?
16:25.12Zeeekprint as fax and mail
16:25.28Zeeekor upload
16:25.35inspireduhm, print as fax = creates a file?
16:25.40Zeeekof course
16:25.42Zeeeka TIF
16:25.42inspiredah
16:25.47inspirednice
16:26.12Zeeeksee somehow the average user will get that where as a convert or save as is seen as an extra step
16:26.44Zeeekfor a quick idea try j2.com - I think you can download a free driver - or efax
16:27.09Zeeekor maybe there's open source stuff out there
16:27.36inspiredok
16:28.15Zeeekhttp://wwwi.efax.com/fr/efax/twa/page/download
16:28.30Zeeekoops they outsmarted themselves detecting Fren,ch
16:28.52Zeeekhttp://www.efax.com/en/efax/twa/page/download
16:28.59inspiredI don't want our customers to use efax
16:29.02inspiredI want them to use us ;)
16:29.04*** join/#asterisk cybercron (~test@208-216-127-234.cust.gti.net)
16:29.47Zeeekyeah but the software is free -
16:29.52Zeeeknot the faxing
16:29.53ManxPowerDoes eFax offer branded services?
16:30.00Zeeekthey might?
16:30.19inspiredwell, seems that print to fax is the best idea
16:30.30inspiredit will work well with our product
16:30.56Zeeekyou can prolly find a driver out there somewhere - in fact maybe built in to Windoze
16:31.05ZeeekXP?
16:31.32Zeeekhehe, then there's Exchange :)
16:31.38inspiredprint to fax is not standard in word?
16:31.39`SauronIs X-Lite the most popular softphone for winblows?
16:32.10Zeeekinspired I don't think so but XP is different it has a lot of that built in
16:32.38|Vulture|anyone ever get outbound fax working sucessfully with * via fax--fxs gateway--asterisk--(voip provider or POTS)
16:32.55eKo1|Vulture|: not me.
16:32.59`Sauronhum, and voip-info seems to be down, bummer.
16:33.12|Vulture|yea I tried it awhile ago never got it to work
16:33.30eKo1The problem is that your provider may not be using ulaw so...
16:33.43|Vulture|true.. but POTS shouldn't matter
16:34.12eKo1Yeah, I was able to send faxes no problems but I could never receive any.
16:34.21inspired"Microsoft Office Document Writer" is default in MS Word 2003
16:35.19Godseyvulture: yes we send faxes all the time
16:35.36Godseyyou just have to disallow=all, allow=ulaw for the device and sip provider
16:36.29ariel_|Vulture|, I have fax working via an iax provider.
16:36.44GodseyI wish winfax pro had sip support :)
16:37.48ariel_|Vulture|, in fact I have fax working via sipura-2000 and sipura-2100. What problems are you getting?
16:38.03|Vulture|I think it might be our fax machines
16:38.06|Vulture|they are POSes
16:38.09Godseywe use linksys pap2 devices
16:38.19Godseynot sure what it is rebranded from :)
16:38.26|Vulture|ariel_: you use it over VPC?
16:38.32ariel_Godsey, there sipura
16:38.49ariel_|Vulture|, no actually i use it via race.com
16:39.04ariel_voipjet is not good for it either.
16:39.16|Vulture|ariel_: do you know what fax machine you use? we have some all in wonder I was working with
16:39.18|Vulture|it sucked
16:39.19ariel_And vpc I can get them inbound but only 50% of them out bound.
16:39.41ariel_well here I am using hp officejets and internal modems.
16:39.43|Vulture|hmm race.com not found
16:39.50ariel_www.race.com
16:40.04|Vulture|strange... nothing
16:40.13ariel_There network is directly to tdm switches
16:40.44ariel_yes your right. hummm I know why.
16:41.03*** join/#asterisk drumkilla (~russell@12.21.241.80)
16:41.03*** mode/#asterisk [+o drumkilla] by ChanServ
16:41.19ariel_There moving this weekend to a new location for there colo. There going to offer service to the public in a week or so.
16:41.41|Vulture|oh oky, I think I am going to buy a nice fax to test with
16:46.36`SauronIs there a simple SIP softphone where you can just put in username@domain.com for dialing someone who's got SRV records and al that set up properly?
16:46.43*** join/#asterisk oej (~oej@apollo.webway.se)
16:48.29*** join/#asterisk BoRiS (boris@S01060040ca1e5b54.wp.shawcable.net)
16:48.31*** join/#asterisk djin (~djin@gridfox.xs4all.nl)
16:50.12BoRiS!seen paulc
16:50.23BoRiS~seen paulc
16:50.25jbotpaulc <~paulc@S010600062586a0b4.vc.shawcable.net> was last seen on IRC in channel #asterisk, 7d 21h 36m 33s ago, saying: 'Firestrm: I gotta head out for lunch, got an appointment, I'll leave you in the capable hands of Dr B Johnson :-)'.
16:53.14inspired`Sauron: most/all should support that
16:53.32*** join/#asterisk visik7 (~ciao@host11-39.pool80182.interbusiness.it)
16:53.51visik7what's the * pastebin site ?
16:53.56visik7that I forgot :)
16:53.56Nivex~pastebin
16:53.58jbotextra, extra, read all about it, pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca
16:54.03visik7grazie
16:54.09visik7sorry
16:54.10visik7I mean
16:54.12visik7Thank you
16:54.17`Sauroninspired: Hum. Maybe I didn't look enough at the config, then.
16:54.36Nivexvisik7: You are welcome.
16:55.13inspiredif the SRV records for domain.com are set up and pointed to a SIP server with the user "username", it will work
16:55.28inspiredjust add exten => username to the standard context on your sip server
16:56.22inspiredi.e. [default]
16:56.46inspiredexten => username,1,Goto(users,0001,1)
17:02.26Zeeekfunny, I thought that worked without SRV
17:04.01`Sauronhum, I see.
17:04.07RoyKfsck
17:04.13RoyKvoip-info.org is down again
17:04.14Zeeektsk, tsk
17:05.02Beirdoit's making RTFM'ing harder :)
17:05.05`Sauroninspired: If I have [default] with include => from-sip
17:05.25`Sauronand [from-sip] does a bunch of exten => s,1,.... etcetc
17:05.26Silik0ndont you mean RTFW'ing harder ;)
17:05.27`Sauronthat should work
17:05.32ManxPowerThis is funny: http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&category=28048&item=5752360720&rd=1&ssPageName=WDVW
17:05.50BeirdoSilik0n: yeah, I kinda consider the W part of the M :)
17:05.57ManxPowerNotice the half naked women in the listing pics, no aparent reason for her to be there, just to make make geeks look at the ad.
17:06.41Silik0nManxPower yeah but i bet it does wonder for their sales
17:06.42RoyKManxPower: hehe
17:06.52RoyKManxPower: press 'play' below and you get the shots
17:07.10*** join/#asterisk salmandr (~salmandr@66-188-101-214.mad.wi.charter.com)
17:07.15Beirdomaybe she comes with the PBX?
17:07.17Silik0nand do you really wanna buy a MICS anyway?
17:07.34ManxPowerSilik0n, No, I want a PRI card for the existing MICS
17:07.47Zeeekhttp://web.archive.org/web/20040220020259/www.voipinfo.org/tiki-index.php
17:08.21RoyKZeeek: what's that?
17:08.41RoyKjust a copy?
17:08.46Zeeekcache
17:08.57RoyKk
17:09.04RoyKwho's running the wiki?
17:09.14Zeeekthe wikimasters
17:09.17RoyKs/s r/s supposed to be r/
17:09.36ZeeekI think they had to shut down to add another 16megs of RAM
17:10.02RoyKlol
17:10.04Zeeekthat chaches version is from Feb 11
17:10.08Zeeekcache
17:10.14Silik0ntheir 486 dying under the load?
17:10.29ZeeekI mean it's not like NEW! H323 plugins
17:10.47*** part/#asterisk maksim (~max@213.142.207.2)
17:11.01*** join/#asterisk kpfleming (~chatzilla@ip68-3-230-141.ph.ph.cox.net)
17:12.44Silik0nok anyone know anything about porting drivers to OSX?
17:12.44Zeeek1:36 to go on 400megs
17:12.57Silik0nZeeek whi
17:13.07Silik0nwhat movie you downloading?
17:13.15Zeeeksaxophone lessaons
17:13.20Silik0nheh
17:13.21Zeeekand spelling lessons
17:13.33Silik0nsend me the spelling lessons
17:13.34RoyKhttp://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&item=5752356642&ssPageName=MERC_VI_RSCC_Pr4_PcN__Stores
17:13.46RoyKseems to come with something similar as the other
17:13.58Zeeekif you build it they will come
17:14.05Zeeek8 seconds
17:14.27ZeeekBAD CHECKSUM PLEASE BEGIN DOWNLOAD AGAIN!!!!
17:14.30salmandri'll buy if that girl will deliver it :)
17:15.11`SauronGrf.
17:15.25`SauronSo I set myself up in the address book on x-lite, with user@domain.com
17:15.34`Sauronand when I go to dial, it wants me to configure a default SIP proxy
17:15.43`Sauronbut the point is not to have to set one up. Grf.
17:16.30Beirdosalmandr: she's likely even higher maintenence than the PBX
17:16.41Zeeekheh
17:22.03*** join/#asterisk jetscreamer (~jetscream@adsl-64-219-216-41.dsl.hstntx.swbell.net)
17:23.24*** join/#asterisk fafnir (~hello@tdds-gw.Moscow.gldn.net)
17:23.39`SauronX-Lite blows
17:27.48*** join/#asterisk shayne (~shaynebat@ip68-100-97-241.dc.dc.cox.net)
17:27.59shayne?
17:28.35shaynehas anyone tried configuring asterisk on os x ?
17:29.27Silik0nyou can build it on OSX and it runs just fine...
17:29.35Silik0nhowever there are no TDM drivers for OSX at this time
17:29.44Nuggetyeah, I use asterisk on my powerbook for when I travel.
17:29.47*** join/#asterisk dudewhere (~ashly@adsl-68-72-128-234.dsl.chcgil.ameritech.net)
17:30.00Nuggetsince sip is a pain in the ass through hotel nat hell, I use x-lite to a local asterisk and then iax to my main server.
17:30.03*** join/#asterisk abombss (~abombss@c-67-175-115-51.client.comcast.net)
17:30.04Silik0ni have it compiled on my G4
17:30.37shayneThanks...I'm a complete newbie to this...have setup asterisk from sunrise-tel.com and wanting to make it work with a sipura 3000
17:30.48shaynewith FWD
17:30.50Nuggetthat should work just fine in os x.
17:31.10dudewhereAnyone know where I can find a list of packages need to install * FC2 I want the OS to bare bones, but I know I need NCurses, SSH, Bison and I'm not sure what else.
17:31.18shaynecant seem to find much documentation on configuring since I'm not a linux techo
17:33.08*** join/#asterisk o-m-a-o-m-a (unknown@80.81.19.75)
17:33.41o-m-a-o-m-aGood evening
17:34.18trymgood morning
17:34.58o-m-a-o-m-aI need some kind of hint. What means "chan_zap.c:7411 zt_pri_error: PRI: !! Got S-frame while link down
17:35.06*** join/#asterisk RoyK (~roy@80.239.107.80)
17:35.14RoyKinspired: ping
17:36.28RoyK~seen inspired
17:36.30jbotinspired is currently on #asterisk (9h 5m 33s).  Has said a total of 24 messages.  Is idling for 39m 44s
17:37.19Mw3~seen P-Chan
17:37.20jbotp-chan <~pchan@68.142.66.200> was last seen on IRC in channel #asterisk, 6d 52m 8s ago, saying: 'oh'.
17:44.21*** join/#asterisk santiago (~santiago@63.245.86.121)
17:48.24*** join/#asterisk sysdef (~sysdef@pD9560EB9.dip.t-dialin.net)
17:48.41*** join/#asterisk lilo (lilo@levin-pdpc.staff.freenode)
17:51.28Mw3hm, mohmp3s are gone from asterisk debian packages in sid :(
17:54.21RoyKdon't use packages
17:55.14*** join/#asterisk Sedorox (brandon@Neptune.client.wlmsprt.pa.sed6.net)
17:57.06*** join/#asterisk lyroy (~lyroy@modemcable117.123-202-24.mc.videotron.ca)
17:57.56*** part/#asterisk santiago (~santiago@63.245.86.121)
18:02.39*** join/#asterisk bkw_ (~brian@bkw.developer.and.friend.of.asterisk)
18:02.39*** mode/#asterisk [+o bkw_] by ChanServ
18:04.05dan2could someone verify for me if there are issues on broadvoice dca?
18:05.26*** join/#asterisk lilo (lilo@levin-pdpc.staff.freenode)
18:08.27bjohnsondudewhere: there is some fedora info on the wiki
18:08.31bjohnson~docs
18:08.32jbotDocumentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org
18:08.43bjohnsonalways a good place to start looking for info
18:09.04bjohnsonunfortunately, down today
18:09.27Silik0nyeah someone needs to whoever runs is and say "y0 WTF the wiki is down"
18:10.13RoyKwho /is/ running the wiki?
18:12.25eKo1Is the wiki down?
18:12.45eKo1I can't seem access it.
18:12.50Silik0nwhat ever gave you that idea
18:12.59RoyK:)
18:18.07*** join/#asterisk jhiver (~jhiver@ABoulogne-102-1-3-10.w193-253.abo.wanadoo.fr)
18:18.14jhiverhi lads
18:18.23bkw_HEY HEY HEY
18:18.28jhiversilly question: how can you debug a Perl AGI script?
18:18.38bkw_print to STDERR
18:18.43bkw_or exec verbose
18:18.52jhiverOK so I can just use warn
18:19.02jhiverwill it appear on the CLI?
18:19.10bkw_use verbose then
18:19.16bkw_but printing to STDERR is better
18:19.28jhiverok
18:19.43*** join/#asterisk ranliv (~ranliv@203.172.11.239)
18:19.45jhiverI'm tying together the stupidest script ever but it should do the work...
18:20.11jhiverbasically using non-crap demo voice synthetisers rather than festival... horrible horrible stuff
18:20.27jhiverbut it'll be better than mr. robocop talking :)
18:21.00Mw3has anyone managed to get this work with capi driver: "Eicon Networks Corporation Diva 2.01 S/T PCI" ?
18:21.21jhiverit's not a server card
18:21.25jhiverI have the same at home
18:21.33jhiverwon't work me thinks
18:21.55Mw3ah :(
18:21.56Mw3damn it
18:22.05jhivertoo bad... cheap card...
18:22.23Mw3works with i4l modem driver but that's not the best channel driver :)
18:23.38ranlivhello guys, I need help!  My asterisk box is located behind a traditional pbx and I need to dial 9 first before dialing the destination number.  how can i  dial 9, put a 1 sec delay and dial the destination number from my Zap channel?
18:24.27*** join/#asterisk lattice (~lattice@S010600045ad57bb6.vc.shawcable.net)
18:24.42jhiverDon't know about the 1 second delay but do you really need that delay?
18:25.01ranlivhow do I do this?
18:25.13jhiverOtherwise it would be like Dial(9${EXTEN})
18:25.34bkw_is it Analog?
18:25.49bkw_Dial(Zap/g1/9w${EXTEN})
18:25.58o-m-a-o-m-asome hardware PBX need some time to catch the free line sign
18:26.29o-m-a-o-m-aw like in the good old modem times :-)
18:26.56ranlivwhat do w in 9w stands for wait?
18:27.06bkw_dials 9 then waits
18:27.06bkw_duh
18:28.36*** join/#asterisk Rick_Hunter (~rhunter@04-158.008.popsite.net)
18:29.25RoyK~seen inspired
18:29.26jbotinspired is currently on #asterisk (9h 58m 29s).  Has said a total of 24 messages.  Is idling for 1h 32m 40s
18:29.54RoyKFeb 19 19:29:42 WARNING[24545]: channel.c:1555 ast_prod: Prodding channel 'SIP/1001749-49a0' failed
18:29.58RoyKwtf does that mean?
18:30.10bkw_prodding channel failed
18:30.18bkw_look in channel.c line 1555 to see what it was doing
18:30.30bkw_it prints line numbers for a reason :P
18:33.02hermiehow do you send ps/ali over a PRI? Is there some part of the IE that has the unique station identifier?
18:39.29ranlivguys thank you very much! It worked already
18:48.15*** join/#asterisk [cc]smart (~smart@gw.ptr-62-65-149-158.customer.ch.netstream.com)
18:48.57[cc]smarti'm using ztdummy on kernel 2.6, but having stutter in MOH et al...
18:49.11[cc]smartOpened pseudo zap interface, measuring accuracy...
18:49.11[cc]smart97.558594% 97.460938% 97.460938% ...
18:49.30[cc]smartsomebody has an idea what to do ?
18:49.57[cc]smartSMP system BTW
18:50.55*** join/#asterisk Nivex (kjotte@user-0ce2jqe.cable.mindspring.com)
18:51.17ManxPower[cc]smart, Find a way to get this numbers to 99.7 or better.
18:51.42ManxPowerSearch the mailing list archives for HDLC Abort and you'll find suggestions for improving interrupt latency.
18:51.45*** join/#asterisk atmel (~vlad@ip68-4-101-199.oc.oc.cox.net)
18:54.12*** join/#asterisk delchi (delchi@amanda.dorsai.org)
18:54.24delchimornin'
18:54.39delchi( well in this part of the world at least )
18:55.48delchican anyone point me in the direction of information as to wether or not the el cheapo Linksys / other VOIP adapters are compadible with * ?
18:56.08delchi( they coem bundled with vonage and/or other services, I was just wondering if I could hack one togeter to work with * )
18:56.24ManxPower~docs
18:56.25jbotDocumentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org
18:56.31Sedoroxthey are locked to Vondage
18:56.34Sedoroxvonage
18:56.36delchithats what I needed to know
18:56.38delchithanks
18:56.45ManxPowerdelchi, Those people lock their stuff up pretty tight.
18:56.51delchiyeah I suspected as much
18:56.53ManxPowerVonage will UNLOCK their box, but there are restrictions.
18:56.59Sedoroxthe linksys's you can't even buy w/o being a VOIP Provider
18:57.01ManxPowerand it's not free.
18:57.11delchiHm
18:57.15JerJergrrr
18:57.21JerJerthey are not locked to vonage, just pre-configured
18:57.22delchiIve seen a pile of carious VOIP > POTS adapters on the shelves
18:57.33ManxPowerThe PAP-NA is pretty much the same as the SIPura SPA-2k
18:57.34file[laptop]they become lock when they get personal with vonage
18:57.36SedoroxJerJer: I thought they were locked... hmmm
18:57.43file[laptop]they 'get jiggy with it'
18:57.50*** join/#asterisk lilo (lilo@levin-pdpc.staff.freenode)
18:57.51JerJerManxPower: they 'are' a 2k
18:57.56JerJerjust with linksys plastic
18:58.01JerJerand blue LEDs
18:58.03Sedoroxlol
18:58.04ManxPowersoundguy, "pretty much"
18:58.07delchiso they are not locked
18:58.08delchiHm
18:58.11*** join/#asterisk Derkommissar (~Derkkommi@66.100.55.66)
18:58.13DerkommissarHello
18:58.13ManxPower..er... So  "pretty mcuh"
18:58.18Derkommissari have a small question,
18:58.21file[laptop]oh, that reminds me
18:58.25Derkommissarwhy doesnt asterisk sends a=fmtp:18 annexb=no
18:58.35Derkommissarat the end of an invite when using g729a
18:58.45*** join/#asterisk Darwin35 (~Darin@c-24-3-241-22.client.comcast.net)
18:58.57ManxPowerDerkommissar, I THINK annexb is silence supressions / Vad
18:59.10Darwin35hey manx
18:59.22ManxPowerHello, Darwin35
18:59.27Darwin35so what did I miss
18:59.29delchibasically what I need is something like the S100I
18:59.43DerkommissarManxPower yes anexb is silence supprecion
18:59.45ManxPowerdelchi, Just buy a SIPura.
18:59.48*** join/#asterisk WizzKid (~apryer@cpc3-lutn5-3-0-cust169.lutn.cable.ntl.com)
19:00.03ManxPowerDarwin35, Too many nose rings?
19:00.13Derkommissarbut without that line, the other party belives that the codec been used is g729ab istead of g729a
19:00.14Darwin35no birth defect
19:00.16Derkommissar:-/
19:00.23Darwin35my rings are else where
19:00.25delchiManxPower : perfect.
19:00.32ManxPowerDerkommissar, So?  They are functionally compatable
19:00.37delchiThats pretty much what Im looking for.
19:00.49DerkommissarYes
19:00.51delchiI was just hoping I coudl zip dowen to the store and buy one today, and not have to snail mail order it
19:01.11delchiOh that sounds like fun
19:01.16Derkommissarbut since the extra frames of Silence suprecion of the RTP is dropped, the quality comes out to be choppy
19:01.21delchiat least use a good car
19:01.53ManxPowerDerkommissar, Then the far side is not honouring the request for no VAD
19:02.16DerkommissarCorrect
19:02.54Derkommissarand its because they base themself to do audio based on the invite
19:03.55Derkommissarthe rfc3555 says that its a standart to put wheather annexb= yes or no
19:06.37Derkommissarare we disconected ?
19:06.43delchidamn the spa-2000 is cheaper than the iaxy
19:07.52Derkommissarcan we use g729ab with asterisk ?
19:14.51RoyKBREW pot HTCPCP/1.0
19:21.12EssobiMehe
19:21.29EssobiQuit talking to your coffee pot
19:23.16Sedoroxerrrr
19:25.59dan2who is using broadvoice proxy dca here?
19:27.45*** join/#asterisk file[laptop] (~file_lapt@mctn1-142166197096.nb.aliant.net)
19:28.04Sedoroxok.. all of the sudden.. when I dial 10-200, to login to a queue on a remote box.. and I press my password and press #, the local box wants to do a transfer... yet if I dial 8500, which is voicemail on another remote box, and I press #, it works fine... any clues?
19:28.45JerJerthere is a T or t dial modifier
19:28.49JerJerin use
19:28.51bjohnsonManxPower: do you have spa 3ks?  I have 2 probs I just can't nail down.
19:28.59Sedoroxyes.. both..
19:28.59JerJerbjohnson:  i have spa 3k
19:29.05JerJerSedorox: that is your problem
19:29.08Sedoroxhmmm
19:29.10bjohnsonI keep getting echo
19:29.22JerJerbjohnson:  if caller*id isn't present it won't make a SIP call
19:29.27JerJerno problems with echo
19:29.45JerJerhowever my audio volume is a little low
19:29.46SedoroxI didn't have a problem with it till yesterday it seems... but hmmm
19:29.56bjohnsonnot every time, but often, a pstn call in the fxo that then goes to the fxs in the same unit or the fxs on another SPA .. get echo
19:30.22JerJerrunning the newest firmware?
19:30.52*** join/#asterisk w0w0 (~apardo@80.26.166.71)
19:31.00bjohnsonclose to newest .. 2.0.11(GWg)
19:31.15*** join/#asterisk bobx (~bobx@206.124.165.14)
19:31.22JerJeri would go up to the newest
19:31.30bjohnsonit was newest about 3 weeks ago but I see a newer one there now
19:31.39CoaxDJerJer: Um, why would firmware version on spa-3000 matter for echo?
19:31.48*** join/#asterisk tekati (~captain@cpe-66-75-215-63.bak.rr.com)
19:31.58JerJerecho can tuned better
19:31.59CoaxDJerJer: I have spa-2000. they say not to upgrade to latest firmware unless there's a fix in the new one that specifically resolves the issue
19:32.00bjohnsonperhaps a better echo cancellation routine
19:32.00*** join/#asterisk welby (~welby@solas.plus.com)
19:32.05CoaxDjerjer: Ahhh.
19:32.15CoaxDbjohnson: Does your caller hear your echo or just you
19:32.26bjohnsonjust me
19:32.44JerJerthen its near-side echo
19:32.48bjohnson<PROTECTED>
19:32.52CoaxDbjohnson: Pretty much a guarantee that your rxgain on your x100p is set too high
19:33.01JerJerits not an X100P
19:33.02CoaxD(or whatever your fxo is)
19:33.07bjohnsonJerJer: yes .. but just on pstn calls.  Calls coming voip are ok
19:33.21bjohnsonit's a spa 3k
19:33.35bjohnsonSPA to PSTN and PSTN to SPA gains were set to 0
19:33.41CoaxDbjohnson: Yes, but your fxo doesnt connect to your spa, right?
19:33.49CoaxD(I forget what the difference between spa2k and spa3k is.)
19:33.54bjohnsonit's happening on two of 3 spa 3ks I have
19:33.57JerJerFXO
19:33.59Qwell3k has the fxo port, right?
19:34.02CoaxDah
19:34.04bjohnsonyes
19:34.09CoaxDprolly need to drop the rxgain somehow on fxo
19:34.27JerJeryou can do it in the advanced config mode
19:34.48CoaxDjerjer: yeah i figured. prolly needs the new firmware to finely tune it, too
19:34.48JerJerunder pstn tab, i think
19:34.57CoaxDi only have sp-2k's
19:35.00Qwellthe spa3k has what, 2fxs, 1fxo, and an ethernet port?
19:35.05CoaxDno need for fxo on pstn thru sip adapter
19:35.19bjohnsonyeah .. I dropped to -2 .. didn't want to go too low.  Found some discussion on the web about line impedence being a possible echo source but it was all about UK lines
19:35.22JerJernow only if the spa 3ks talked IAX
19:35.27bjohnsonQwell: one of each
19:35.29Qwellahh
19:35.36QwellWhat do those generally run?
19:35.38CoaxDJerJer: Dont give up yet :)
19:35.43JerJerbjohnson:  ahh yes!!!  there is an option to change that
19:35.49bjohnsonQwell: $100 at voxilla
19:36.03QwellI should have gotten one of those instead, heh
19:36.09bjohnsonyes .. but all info I could find just listed 2 possibles for N.A.
19:36.17CoaxDbjohnson: Would you beleive that on one of my incoming POTS lines, on an X100P, i have to drop the gain to -7.5?
19:36.26CoaxDbjohnson: I am 200 feet from the telco.
19:36.33CoaxDbjohnson: And cabling to them that is less than 10 years old
19:36.47QwellCoaxD: increase the loop length, heh
19:37.01Qwelljust add like 1500 feet of cable in your walls :p
19:37.04bjohnson600 or 900 are supposed to be NA standard and I have it set to factory default of 600
19:37.05CoaxDqwell: Just think about the 10 miles of wrapped up telco wire in the server room. *g*
19:37.11dan2I've got 'em all, sipura 1000,1001,2000,2100,3000
19:37.26CoaxDdan2: Their ip phones dont look half bad, either
19:37.26bjohnsonanyway I can check line impedance or should I just try the 900 option?
19:37.47dan2CoaxD: heh, I get my voip stuff for free
19:37.50CoaxDbjohnson: There are ways to check line impedance. all requires the competent phone tech to test both ends. :P
19:37.59Qwelldan2: hook me up with a 7940 :p
19:38.08CoaxDbjohnson: Rest assured, if you can put a regular phone on it and it dont echo, your line is gonna be fine. :P
19:38.14dan2Qwell: it sits on my desk
19:38.26dan2Qwell: oh, this is 7960
19:38.50CoaxDbjohnson: You can be almost 100% sure that your line is fine, and it is just your rxgain set too high
19:38.56QwellWell, you get free hardware, right?  Get a 7940, and send it on over. ;]
19:39.01CoaxDbjohnson: You might even notice callerid dont work right either
19:39.15CoaxDbjohnson: (or maybe only a portion of the time)
19:39.16dan2Qwell: I'm waiting for the panasonic and uniden cordless phones to arrive
19:42.13bjohnsoncallerid is only a problem for the one that sits behind a fax/data/phone auto switch
19:42.30bjohnsonso I guess I play with line impedence and gain until I get it to work
19:42.39CoaxDbjohn: yeah. just keep subtracting
19:42.44CoaxDbjohn: Your echo will go away.
19:43.18bjohnsonif I switch line impedence to 900 and that isn't the correct one .. should I have consistant problems?
19:43.31CoaxDquit mucking with line impedance!
19:43.38CoaxDI dont know why you think its a line impedance problem
19:43.43CoaxDit has zero to do with line impedance :P
19:43.47bjohnsonCoaxD: gain on the SPA to PSTN setting or the PSTN to SPA?  I think SPA to PSTN.
19:44.04CoaxDbjohnson: pstn to spa, of course :P
19:44.05bjohnsonCoaxD: just from googling other people trying to solve echo problems
19:44.15CoaxDbjohn: This is the difference between rxgain and txgain
19:44.20CoaxDrxgain would be 'pstn to spa'
19:44.29CoaxDtxgain would be 'spa to pstn'
19:44.38CoaxDyou dont need to transmit softer. you need to receive softer. :)
19:46.01CoaxDand if it gets too quiet, you can increase the txgain to your headset
19:46.09CoaxDi.e. your fxs
19:46.23CoaxD(Those are two different sets of settings.)
19:46.50bjohnsonI don't think these units have fxs gain control
19:49.09bjohnsonthe other problem I have, is that the fxs ports are connected to the line in connectors to a Nortel CICS (I don't think this has anything to do with the echo).  When a Nortel handset is hung up from a pstn call .. the line often rings back and noone is there.  From the logs I can see it is actually the fxo port calling back into the system (ie it isn't hanging up fast enough).  Any ideas on this one?
19:49.36PatrickDKthey have gain control, it's default is -3db
19:50.12CoaxDbjohnson: I have nothing to offer on that issue. :/
19:51.03*** join/#asterisk vs_ (~vs@host-175.voip-gw.dial-pool.macomnet.net)
19:52.00hermiefearnor: nice drumroll on -dev :)
19:52.17vs_howdy
19:52.37bjohnsonbtw, any technical or licensing reason why sipura couldn't make these support iax?
19:53.08Qwelllazyness, heh
19:53.16hermiebjohnson: they're fine as long as they do a clean-room implementation
19:53.36hermiebjohnson: which is hard because IAX is a moving target and not very well documented in writing
19:53.51hermiebjohnson: which we're trying to change over in ADP-land
19:53.52bjohnsonwell, I imagine they won't do it unless they think it will make enough increase in sales to justify the dev cost
19:53.57ManxPowerIAX isn't really the problem, lack of docs for writing firmware for SIPuras is the problem.
19:54.13o-m-a-o-m-aI need some kind of hint. What means "chan_zap.c:7411 zt_pri_error: PRI: !! Got S-frame while link down" on HFC-S / ISDN?
19:54.28hermiewhat't with asking the mailing list if something's down?
19:55.20hermielike the wiki
19:55.22hermieand broadvoice
19:57.09CoaxDkinda lame, yeah, hermie
19:57.24PatrickDKbjohnson, you have to remember, only asterisk supports iax, and lots of people support sip
19:57.53PatrickDKand to produce a product with limited people that would be interested in it, taks time and money
19:58.28*** join/#asterisk rodizump (~chatzilla@dsl-213-023-226-078.arcor-ip.net)
19:58.46*** join/#asterisk Frantic (~ab@24-193-46-85.nyc.rr.com)
19:58.52*** join/#asterisk Nukemizer (~Nuke@65.103.231.133)
19:58.56*** join/#asterisk cjk (~cjk@80.92.75.91)
19:59.08elricdo you reckon IAX will ever be made an IETF standard?
19:59.10PatrickDKif you could prove to sipura, that 50% of sipura users have asterisk, then they will probably consider supporting iax
19:59.19PatrickDKbut otherwise you will probably have a tough time of it
19:59.24rodizumphi everyone, is the www.voip-info.org really down or my ISP can't route there ? can anybody confirm ? please
19:59.38cjkrodizump, i confirm and i really need to access the site
19:59.47Frantic<rodizump> confirmed
19:59.57shido6i couldnt get to them lastnight either
20:00.06PatrickDKwhy can't people really believe that public access, none supported sites go down sometimes
20:01.23Sedoroxok...
20:01.33Sedoroxwhat would cause Asterisk to lock up three totaly different boxes on stop now?
20:01.51shido6same broken code on all three?
20:02.06Sedoroxtwo run 1.0.3 one runs 10.0.5
20:02.08Sedorox1.0.5
20:04.07Sedoroxreverse that...
20:04.13*** join/#asterisk darby_t (~tom@dol122.neoplus.adsl.tpnet.pl)
20:05.11Sedoroxwaiting on pastebin
20:05.58Sedoroxhttp://www.pastebin.com/243834
20:06.05bjohnsonPatrickDK: where is the gain control on a spa 3k fxs?  I can't find it
20:06.10eKo1Has anyone had a problem with certain SIP entries in sip.conf having an accountcode not being logged properly in the CDRs?
20:07.11*** join/#asterisk scardinal (~supreme@port816.ds1-suoe.adsl.cybercity.dk)
20:10.32Sedoroxanyone have any clues?
20:10.43NukemizerI am looking for a Digium T1 card MASTERto hire .. second card card from digium still will not load  properly and still get alarms . Any takers ?
20:11.16*** join/#asterisk PoWeRKiLL (~PoWeRKiLL@l01m-21-45.d4.club-internet.fr)
20:15.59PoWeRKiLLhi :)
20:18.07bjohnsonwell .. this is a nice tip about UPS shipments to Canada: Enjoy free customs clearance on UPS Express and UPS Expedited shipments
20:20.32PatrickDKls
20:21.34BoRiSI wish Canada Post would ship on saturday
20:21.56`SauronAnyone here with a working softphone?
20:22.00Sedoroxnayone know what would cause theses lock ups? http://www.pastebin.com/243834
20:24.13ManxPowerSedorox, You should be able to ignore those since you are shutting down Asterisk
20:24.50ManxPower"Yuck! Error in buffer handling...: Broken pipe " is an error from mpg123
20:25.23*** join/#asterisk kippi (chrisfrog@cpc4-hatf3-6-0-cust243.lutn.cable.ntl.com)
20:26.07kippihey is voip-info.org down?
20:26.12vs_hehe
20:26.13vs_ye
20:26.18kippidamm
20:26.36kippianyone used asterisk@home ?
20:26.48ManxPowerLots of people, but we don't talk to them.
20:27.04kippihow comes? it not worth using?
20:27.05*** join/#asterisk NetworkStorm (~afernande@adsl-69-108-160-246.dsl.skt2ca.pacbell.net)
20:27.13NetworkStormHello
20:28.32jetscreameris asterisk@home different from asterisk?
20:28.43jetscreameror is that just where you use it
20:28.44bjohnsonno
20:28.47bjohnsonjust uses amp
20:28.48jetscreamero
20:28.54SedoroxManxPower: the thing is.. after all that.. it crashed all the boxes
20:28.55bjohnsonas a gui
20:29.10bjohnsonand auto-installs asterisk and linux from cd
20:29.22jetscreamerah the livecd
20:29.24kippican you use linux from the bash?
20:29.35jetscreameryes...?
20:29.36NetworkStormI am having problems with asterisk not recognizing my TDM400P card.
20:29.42NetworkStormAnymore
20:29.48bjohnsonI think bkw_ has a different version of the same concept .. a prepared distro for asterisk server setup
20:29.50NetworkStormCan someone help me?
20:30.08shido6uhh
20:30.09shido6yeah
20:30.13shido6I can help ewe
20:30.17NetworkStormCool
20:30.18bjohnsonjetscreamer: I don't think it will run from cd .. just auto install.  but i haven't used it so might be wrong
20:30.21kippiso would poeple say to keep away from it?
20:30.30NetworkStormI worked last night with no problems
20:30.41jetscreamerlinux from bash. yes. learn concepts though.
20:30.46NetworkStormI turned it off, and this morning asterisk would not start
20:31.00bjohnsonkippi: don't worry about it.  it's just hard for use to help you since it makes a very complicated setup to start
20:31.14jetscreamersaying yes to that question is like fingernails on a blackboard
20:31.49Nugget"can you use linux from the bash"   <-- wow.  I'm beyond words.  This is so wrong on so many levels.
20:32.02jetscreamerbut the answer he seeks is yes.
20:32.07jetscreamerjsut bad q.
20:32.15kippiso would you say I would be better of installing slackware and then installing asterisk?
20:32.19bjohnsonI think he means access the asterisk cli from a bash prompt
20:32.35bjohnsonkippi: define "better off"
20:33.12kippiis the only con to installing asterisk@home is that it uses the gui ?
20:33.18bjohnsonif you can install ast@home and it does everything you want right away, then that might be best for you
20:33.43bjohnsonif it doesn't do what you want and you have to troubleshoot .. maybe format disk and start with a typical install
20:33.53kippihmm
20:34.21bjohnsonno sense troubleshooting their system .. usually easier to start fresh in that case
20:34.23rodizumpcan anyone tell how to make asterisk send BYE after RTP timeout, ie. remote side hangs up and asterisk sends bye to originator ? anyone ?
20:34.27jetscreamerdoes this @home install on a preexisting linux install or does it need the 'distro' it comes with
20:34.35eKo1Man...VoIP over 802.11b sucks orangutan nipples.
20:34.45bjohnsonread it's faq .. I am not it's developer nor user
20:34.54jetscreamerk thx
20:35.05bjohnsoneKo1: depends who you ask .. vaewynAFK loves it
20:35.21eKo1Well, for short distances it should be no problem.
20:35.41eKo1But after 5 miles, well...it stinks.
20:36.47eKo1Then the customers complain about bad voice quality and shitty web surfing and blah, blah, blah....
20:36.57PatrickDKhmm, voip over 802.11b works good for me
20:37.15PatrickDKI have only run it up to 1mile, with a voip phone
20:37.30eKo1I guess there is too much interference.
20:37.59*** join/#asterisk yashax (~yasha_x@69.15.218.218)
20:38.02kippiso then you are better of with a clean install and build the system up ur self
20:38.04kippi?
20:38.10rodizumpDoes anyone know how to set RTP timeout in asterisk ?
20:39.19NetworkStorm'
20:39.24NetworkStormHow can I verify that WBEL see my TDM400P card?
20:39.52MocWBEL ?
20:40.01`SauronHum, so that worked, at least partially.
20:40.16NetworkStormWhitebox linux
20:40.26NetworkStormIts basically RHEL
20:41.41NetworkStormbe wctdm
20:42.14vs_dropped calls with chan_oh323
20:42.20*** part/#asterisk eKo1 (~bernd@207.42.191.66)
20:42.25vs_and cisco
20:43.39MocI use TaoLinux
20:47.04shido6Tao
20:47.14shido6how is TaoLinux?
20:50.57*** join/#asterisk file[laptop] (~file_lapt@mctn1-142166197096.nb.aliant.net)
20:51.30ariel_NetworkStorm, zttool
20:51.59*** part/#asterisk NetworkStorm (~afernande@adsl-69-108-160-246.dsl.skt2ca.pacbell.net)
20:53.52GMsoftmhh does fxs/fxo signaling is able to send the called and caller id ?
20:55.18shido6yes
20:55.39*** join/#asterisk jsolares (~jsolares@200.12.44.18)
20:56.07GMsoftso I could ask to have more than one number routed to my fxo line and match the called id and the route the call correctly ?
20:56.28*** join/#asterisk zyke (~zakforeve@84.45.132.117)
20:56.29*** join/#asterisk ryguillian (~ryguillia@c-24-12-96-52.client.comcast.net)
20:58.54ariel_GMsoft, yes but you should check to see what the ${DNIS} says to make sure your provider is sending you info you can use.
20:59.33GMsoftariel_: ok thanks. I'll ask them before subscribing :)
21:01.06*** join/#asterisk zimdog (~zimdog@c-67-164-190-201.client.comcast.net)
21:02.28*** join/#asterisk miguellinux (~miguellin@200.47.223.190)
21:04.05BeirdoI'll try again today :)
21:04.24Beirdoanyone know how I can tell a particular extension not to use music on hold?
21:06.37Beirdoooh, the wiki's back.  I'll try RTFW again :)
21:06.42ariel_Beirdo, I don't understand your question?
21:07.12BeirdoI want to make it so if I call extension 502, then put it on hold that it doesn't play music on hold
21:07.46Beirdoas 502 goes to a meetme elsewhere, and if I put it on hold to answer another call, I don't need to be treating others on the conference to music :)
21:07.51ariel_make different context for them set the variable in that context not to have moh.
21:08.59Beirdonow I'm the one not understanding :)
21:09.10Beirdobut I'll look around on the wiki a bit
21:10.46`Sauronis the wiki back up?
21:11.09Beirdoseems to be
21:11.21*** join/#asterisk defian (ircuser@shakotay.alphanet.ch)
21:12.33*** join/#asterisk zimdog (~zimdog@c-67-164-190-201.client.comcast.net)
21:12.37defianhello, I am quite new here. Anyone fond of chan_zap & ISDN issues ? :)
21:13.31defian(asterisk 1.0.5; one call is OK, a second call is OK; a third call comes in and is rejected (good); however after that point no second call will get answered)
21:14.16roamer323is there any "production quality" iax2 softphones out there? similar to xten phones for sip?
21:15.05defianroamer323: I only know about the IAXy (analog-to-IAX2 adapter)
21:15.12defianah sorry
21:15.19defianroamer323: IAXcomm works very well here
21:15.31djinfirefly
21:15.33defianroamer323: both on GNU/Linux, Microsoft Windows and thrice on Mac OS X
21:16.09roamer323djin - can firefly be configured for any IAX2 provider in and out? or is it hardwire to theirs?
21:16.09djinhttp://www.virbiage.com/firefly/
21:16.22defianroamer323: with IAXcomm disable all automatic gain/filters and so on and use A-LAW or u-law and you get very good quality
21:16.37djinNo, it'
21:16.44defian(only used it locally on Ethernet though)
21:16.44djinNo, it's open
21:17.39roamer323djin & defian - thanks, I'll try both of them out... the xten UI is really slick and responsive; too bad they're not talking IAX (yet)
21:18.32empire667djin and all the others thanks for all the help, my asterisk box works great
21:18.45empire667My compliments to you all
21:21.47*** join/#asterisk jdg (~jdg@CA03F308.adsl.mana.pf)
21:26.28*** join/#asterisk outsidefactor (barf@203-173-32-225.dyn.iinet.net.au)
21:26.34*** join/#asterisk zotz (~zotz@24.231.32.191)
21:28.08bjohnsoncouple of errors in the example here because the wiki removes the square brackets: http://www.voip-info.org/wiki-Asterisk+user+authentication
21:28.18bjohnsonany idea how to show square brackets in the wiki?
21:29.06`Sauronumm
21:29.07`Sauronyeah
21:29.13`SauronI think ~[ and ~]
21:29.15`Sauronor something
21:29.43GMsofthehe asterisk compiled on my parisc box. it needs a little patch tho :)
21:30.18*** join/#asterisk mooboi (~selfsck@silenceisdefeat.org)
21:30.30mooboiany asterisk at home Aficionado around ?
21:30.40mooboii just finished installing an x100p fxo works fine, caller id work great too, i was just wondering what can beacomplished  next, voice mail ?
21:30.54defiancall transfer
21:30.56defian:)
21:30.57GMsoftconference room
21:31.18*** join/#asterisk Rick_Hunter (~rhunter@06-166.008.popsite.net)
21:31.28mooboiconf room would be great feat, can it be acomplished with only one line ?
21:31.46mooboii tought call trnasfert needed 2 pots lines
21:32.06defianno
21:32.25defianit depends :)
21:32.30mooboidefined in extensions.conf ?
21:32.57defiane.g.:   call -> zap/1; then zap/1 can transfer to another extension (zap/2 or whatever) using Flash key
21:33.36`Sauronmooboi: A digium x100p, or a clone card?
21:33.46mooboiclone card, im poor ; /
21:33.49defian(in my case it's a TDM card)
21:33.58`SauronHum, nice.
21:34.12mooboii would have gone with a tdm 1fxo 1fxs if it wanst for the 100$
21:34.16`SauronI've been thinking 'bout picking up a clone card, but people report mixed success with them
21:34.16defiananalog works great, I have a few issues with ISDNs
21:34.34mooboii guessed it like the loterry ... crossed with ebay
21:34.57hermiebjohnson: __~np~[whatever]~/np~__
21:35.02mooboimine seem to be fine so far , all for 12$ delivered to my door
21:35.12defianugh
21:35.20`Sauronmooboi: That's nice.
21:35.22mooboix100p//ebay
21:36.00mooboinow i neeed to figure out what else to do with it beside cid, i am looking for a more focused resource than voip-info.org
21:36.17`Sauronvoip-info is very useful
21:36.25`Sauronjust depends on what you're trying to do
21:38.14`Sauronset up a vru, set up meetme, set up voicemail
21:38.16`Sauronhave fun
21:38.23`Sauronconnect to FWD while you're at it
21:38.24`Sauronetc etc
21:38.37mooboito FWD ?
21:39.10`Sauronwww.freeworlddialup.com
21:39.17mooboioh ok ok
21:39.22tzangerwow this beer is awful yeasty today
21:39.32tzangerIt tastes like I'm eating sourdough
21:39.33mooboii was thinking about buying terminaition to pstn from iax.cc later on
21:39.42mooboitough luck
21:39.49tzanger... which I don't particularly mind
21:39.59rikstatzanger: don't make homebrew then ;)
21:40.03tzangerthis isn't
21:40.06tzangerit's Molson Ex
21:40.12tzangermy favourite brew
21:40.38defianBTW do you know if there is a way to interconnect with Skype?  (just curious)
21:40.43tzangerdefian: no
21:40.48rikstanop
21:40.57defiantzanger: it's still deadly proprietary?
21:41.11tzangerdefian: more or less, yes
21:41.29defiantzanger: ok :)
21:42.05*** join/#asterisk jtodd (~jtodd@h-67-103-42-29.snfccasy.covad.net)
21:42.11*** join/#asterisk Legend (~legend@24.244.142.133)
21:43.09*** join/#asterisk kFuQ (~somedude@c-24-17-224-78.client.comcast.net)
21:46.28cjkis it possible to connect with netcat to the asterisk manager so i can test a few things before coding them?
21:47.10tzangercjk: why wouldn't it be?
21:48.03cjktzanger, because i dont get any response when connecting to the asterisk manager and trying to do the login. im quite sure the username and password are working becaus an app not coded by me connects with this login
21:48.41tzangercjk: hmm
21:48.48tzangeruse some tcpdumpage and see what's really happenning
21:48.49MavvieI have found out that asterisk doesn't work if the ISDN setup packet contain the exclusive dchannel bit.
21:49.07Mavviethat has caused a backout of the transfer of 800 numbers yesterday :-P
21:49.30defianMavvie: great :)
21:49.36Sedoroxhmmm
21:49.41Mavviethese two are related :-)
21:49.48tzangerMavvie: that sounds bad
21:50.05cjktzanger, i think im doing something wrong with nc when entering the commands, maybe you have a short example?
21:50.53tzangerecho "blah" | nc asterisk.box 5038
21:51.08defianMavvie: if you have experience with ISDN, did you also experience problems with call waiting (a BRI has 2 B channels used; a third call comes in; in Asterisk 1.0.5 it gets correctly ignored; however then no more than 1 call at a time are answered)
21:51.45GMsoftyay asterisk works on my hppa :)
21:52.20cjktzanger, oh yes, but that wont give me a result
21:52.24tzangercjk: well no ;-)
21:52.50tzangernc is one-way
21:53.09`SauronNope
21:53.10Mavvietzanger: if should write the output to stdout.
21:53.12`Sauronnc is two-way
21:53.31GMsoftmhh anyone have doc to nat correctly IAX calls ?
21:53.39*** part/#asterisk djin (~djin@gridfox.xs4all.nl)
21:53.43tzanger`Sauron: not when used with |
21:53.49cjkyes nc normaly prints what it gets back
21:53.56`Sauroneh
21:54.00`SauronI think you're on crack
21:54.05Mavvie[~] edwin@k7>echo hi | nc 0 22
21:54.05MavvieSSH-1.99-OpenSSH_3.8p1
21:54.05MavvieProtocol mismatch.
21:54.45cjkok here is what i do: nc localhost 5038 [ENTER] Action: login[ENTER]Username: cjk[ENTER] Secret: **[ENTER]
21:55.09Mavviecjk: you should use telnet for that.
21:55.21`Sauron:)
21:55.38tzanger`Sauron: well yeah but that's coming back to stdout
21:55.46tzangeryou could write a perl script or something that worked with it I'm sure
21:56.12cjkMavvie, ok i will try
21:56.19*** join/#asterisk Nix (~Nix@81.213.125.220)
21:56.51cjkMavvie, thanks its working
21:57.09cjkMavvie, what is telnet sending different than netcat?
21:57.42Nixcjk: Telnet sends a buch of control chars on conntect...
21:58.32Mavviecjk: telnet is a terminal application, netcat just sets up a tcp connection.
21:58.51Mavviehttp://www.rhyshaden.com/voice.htm <- very interesting read for newbies.
21:59.30cjkMavvie, yeah, but now when I will setup a php script, it will do only a tcp connection. nothing more
21:59.45Mavviecjk: that's all what's needed.
22:00.50SedoroxQuestion.. why doesn't IAX have ipv6 support yet? or are they working on it?
22:02.23ManxPowerSedorox, Because nobody cares enough to add it.
22:02.35Sedoroxhmm
22:02.39SedoroxI may look into it then...
22:02.56MavvieSedorox: several "I'll take it" have been shouted in the -dev mailinglist.
22:03.02Sedoroxhmmm
22:03.16Sedoroxbut nothing has come as a result huh?
22:03.46vs_better to have t.38 :)
22:04.09Sedoroxlol
22:05.46eaperezhhi there, i have 128kbps adsl user that will connect to my 128kbps adsl asterisk, for fun....clients are IAX, should i use gsm or speex?
22:06.29Zaware there any TDM cards that are compatible with freebsd and asterisk?
22:07.00eaperezhZaw: sangoma if i recall correctly
22:07.14Zaweaperezh: thanks
22:08.10elrichi how is the support for sangoma t1/e1 interfaces? i am thinking of using an embedded solution with soekris x86 boards.
22:11.41kippihmm, been  trying out asterisk@home and i see what you mean by the config is complex
22:13.06mikegrboverly complex
22:13.11mikegrbneedlessly complex
22:13.18mikegrbjust install asterisk like normal
22:13.24mikegrbasterisk@home breaks stuff
22:13.42kippiwas trying to get the hold music to work and it was having none of it
22:14.04mikegrband voicemail users can't change thier passcode
22:14.10mikegrband probably loads of other stuff
22:14.31mooboiodd, all my call gets answered after 15sec and blank , nothing
22:15.05kippimine where getting the voicemail and then the voicemail was sending me a email which was cool
22:15.38cjkMavvie, ok then something like this should work? fputs($socket,"
22:15.39cjkAction: login
22:15.39cjkUsername: cjk
22:15.39cjkSecret: cjk111
22:15.39cjk");
22:16.13Mavvieif you google for asterisk nagios, you will find a script which connects to the manager interface.
22:16.16Mavvieperl script.
22:16.44*** join/#asterisk gambolputty (~gambolput@cblmdm65-221-51-80.buckeye-express.com)
22:17.54mikegrbkippi: another fine example of asterisk@home
22:17.55cjkMavvie, ok got it werking
22:18.37sivana~google asterisk nagios
22:18.54Mavviecjk: http://megaglobal.net/docs/asterisk/monitor_pbx.pl
22:19.18Mavvieit's not the best style of programming, but that's something you can overcome easily :-)
22:20.01sivana~google 35c to F
22:20.13sivanaheh
22:29.09*** join/#asterisk djMax (~djMax@artsalliancelabs.com)
22:30.34djMaxis there a way to tell if your wcfxs mo0dule is loaded with lowpower=1?
22:31.32*** join/#asterisk mooboi (~selfsck@silenceisdefeat.org) [NETSPLIT VICTIM]
22:31.32*** join/#asterisk sd-tux (user2267@emasq.stusta.mhn.de) [NETSPLIT VICTIM]
22:33.22*** part/#asterisk sudoer (~sudoer@65.75.148.190)
22:33.59djMaxanother try, why would modprobe say it can't find wcfxs, but lsmod show it?
22:35.29Corydon76-homeMight it actually say "No such device"?
22:35.39*** join/#asterisk luke-jr_ (~luke-jr@207.192.221.172)
22:36.01djMaxlooks like the name is now wctdm
22:36.04mooboiWARNING[1367]: Channel 'Zap/1-1' sent into invalid extension 's' in context 'group-all', but no invalid handler
22:36.10mooboiwhat am i missing here ^
22:36.21defian[group-all]
22:36.33Corydon76-homeYou're missing an s extension, perhaps?
22:36.38*** join/#asterisk chetan (freetibet@24-193-188-21.nyc.rr.com)
22:36.40defianextension => s,1,Dial(demo,s,1)
22:36.40mooboiindeed
22:36.42defianfor example :)
22:36.52|Vulture|defian: under [group-all] ?
22:36.53defians/DIal/Goto/
22:37.05mooboidamn, dinertime
22:37.10mooboibbl
22:37.22defian|Vulture|: I don't know what is group-all
22:37.39defian|Vulture|: do you?
22:37.40|Vulture|seems to be the context he is feeding zaptel to
22:38.02defian|Vulture|: if those are internal lines, demo could be useful for testing
22:38.10defian|Vulture|: if those are external lines, well, it depends
22:38.50defian|Vulture|: can be unsafe
22:38.50|Vulture|looks like hes trying to call in, and he doesnt have a s,1, or he doesnt even have a [group-all] context
22:39.14|Vulture|yea but hes just testing so it shouldn't be a problem
22:39.36defian|Vulture|: yes. But remember that test config have the usual bad habits to stay
22:39.55defian|Vulture|: I was dialing at a friend's PBX from outside and I hit 7 instead of 8 for some reason and I got the demo :)
22:42.07djMaxideas? WARNING[731]: loader.c:509 load_modules: Loading module chan_h323.so failed!
22:44.06djMaxor, alternatively, how can I disable this load attempt from h323
22:44.18|Vulture|haha nice
22:44.31|Vulture|modules?
22:44.35defiandjMax: noload => chan_h323.so in modules.conf
22:45.09|Vulture|defian: I breakup everything via contexts and have an admin context that allows me to use all the tests/demos
22:45.42defiangood concept
22:45.43djMaxwhew.  thanks, back up now.
22:46.53|Vulture|h323 is evil lol
22:48.00vs_no shit
22:48.04*** join/#asterisk PoWeRKiLL (~PoWeRKiLL@l01m-21-45.d4.club-internet.fr)
22:48.05zimdoganyone setup NuFone with AMP?
22:48.21vs_getting Payload type mismatch: expected PCMA, got CiscoCN. Ignoring packet.
22:49.14*** join/#asterisk MajestiK (~MajestiK@S01060800208687ec.ed.shawcable.net)
22:50.11*** join/#asterisk techie (gus@asterisk.horizonte.us)
22:52.00*** join/#asterisk SuperMMan (~graphic@d209-89-191-155.abhsia.telus.net)
22:52.30SuperMManquestion i`m trying to build a module for asterisk and i keep getting a core dump, can anyone recommend a way to find out what the core dump?
22:52.51defianyou get the core dump when compiling the module?
22:53.05SuperMMandefian:  no its when i go to use the module.
22:53.12SuperMManand make  a call
22:53.18defianahem
22:53.49SuperMManall i`m getting is a core dump, I would like to figure out why the core dump, but at this point i`m not getting enough information
22:54.41defianI have never debugged asterisk yet
22:54.58defianhowever gdb asterisk core might give some info (where command)
22:54.59PoWeRKiLLSuperMMan use gdb
22:55.08file[laptop]get a backtrace
22:55.14file[laptop]it'll tell you where/what is crashing it
22:55.44SuperMManfile[laptop]:  thank you.
22:56.57PoWeRKiLLanyone have a good date for getting a ~stable CVS version I can't use stable cause I need voicemail feature database
22:58.07djMaxwhat do you have to do to get rxgain to take effect, just restart asterisk?
22:58.32|Vulture|djMax: that will work
22:58.47djMaxput the rxgain up to 6 but ztmonitor still not really budging
22:59.20|Vulture|do you have "echotraining=yes"
22:59.24djMaxyes
22:59.29|Vulture|hmm strange
22:59.41djMaxI can set rxgain per channel right?
22:59.45|Vulture|yup
22:59.53|Vulture|just have multiple entries
23:00.03djMaxok, 20dB here goes.
23:00.44djMaxwow, now that's some serious echo. :)
23:00.48SuperMManfile[laptop]:  do you know where i can get a copy of backtrace from?
23:01.14file[laptop]what you do is...
23:01.18file[laptop]well, use gdb
23:01.23file[laptop]to open the core file... then type "bt"
23:01.27file[laptop]and voila, you shall get what you seek
23:01.42SuperMManfile[laptop]:  ya  i don`t have bt at all And i can`t seem to find it
23:01.52file[laptop]it's a command in gdb
23:02.19SuperMManoh ok
23:02.57defiangood night :)
23:04.59*** join/#asterisk abombss (~abombss@c-67-175-115-51.client.comcast.net)
23:05.20|Vulture|Anyone use a 7960? I have 7 lines registered and whenever a call comes in over any lines, it looks like it is coming in over line 1... any ideas?
23:06.29NuggetI have a 7960.
23:06.50NuggetI've never figured out how it handles that sort of thing.  I just use three lines.
23:06.57*** join/#asterisk SirPrize (~blah@83.146.62.181)
23:08.59|Vulture|Nugget: yea its kinda annoying because I can't see who is trying to call me
23:10.58Frantic<|Vulture|> had the same issues: I finally modified the caller id to show who it goes to.
23:11.33djMax6dB gain a reasonable number or does that indicate something horribly wrong?
23:12.41|Vulture|wonder if the IP600 suffers from this infliction
23:12.56hmmhesayswell my callback scripts are working nicely... i'll be the first to say... I rock
23:12.59zimdoganyone setup NuFone with AMP?
23:13.05|Vulture|Ive gone from 7940/60s to IP500s
23:13.13*** part/#asterisk sysdef (~sysdef@pD9560EB9.dip.t-dialin.net)
23:14.07SirPrizeMy Asterisk server accepts incoming PSTN calls via SIP.  If I call Asterisk SIP-to-SIP(Asterisk), it accepts DTMF tones (Using X-lite).  PSTN-to-SIP(asterisk) DTMF tones don't work for the menus using normal UK phones. Any idea what I can do ?
23:14.53SirPrizeI've registered incoming SIP line via register => sipinfo.  I've seen dtmfmode modifiers, but am unaware how to make them work for "register"ed SIP channels
23:15.23*** join/#asterisk abombss (~abombss@c-67-175-115-51.client.comcast.net)
23:16.10hmmhesayswhat kind of gateware are you using SirPrize
23:16.17*** join/#asterisk [TK]D-Fender (~joe@4.67.252.216.dsl1.colba.net)
23:17.04hmmhesays*gateway, or are you using a tdm card
23:17.34SirPrizehmmhesays: My Asterisk configuration only has SIP inbounds and SIP registered clients.  I use SipGate.co.UK who provide the PSTN number, which is registered to a SIP address, which I've registered into Asterisk, so not using a card
23:17.56[TK]D-FenderAnyone her have experience making gigabit patch cables?  I'm going nust here trying to get them to work.
23:19.04*** join/#asterisk hajekd (~hajekd@21.208.65.212.contactel.net)
23:19.54hmmhesaysset your dtmfmode to something different for outgoing to them
23:20.05hmmhesayssuch as rfc2833
23:20.44hmmhesayswhat codec are you sending them?
23:20.45SirPrizethat's my problem. I know how to set the dtmfmode for the registered clients/peers, but don't know how to set the dtfmmode coming in via the registered SIP channels. :-(
23:21.16SirPrizehmmhesays: I haven't disallowed any codecs - all the codecs are currently enabled
23:21.16hmmhesaysshow me your registration line out of sip.conf you can paste it at pastebin
23:21.19*** part/#asterisk WizzKid (~apryer@cpc3-lutn5-3-0-cust169.lutn.cable.ntl.com)
23:21.33SirPrizewhat's the complete address for pastebin? pastebin.com ?
23:21.37hmmhesaysyeah
23:22.24SirPrizeIs it visible here: http://www.pastebin.com/243885 ?
23:22.47hmmhesaysyeah
23:23.22SirPrizeI saw an example which specifies the SIP domain as a user, and thought I could specify the dtmfmode in that way.  That's also included in the paste if it helps
23:23.40hmmhesaysdtmfmode=inband? that'll only work for g.711, so fi they are not sending in g.711 that won't work
23:23.55*** join/#asterisk sudhir492 (~sudhir@4.7.59.232)
23:23.58sudhir492hi all
23:24.06sudhir492Anyone from Pakistan here?
23:24.06SirPrizeah - I see..... what dtmfmode can I use for standard PSTN ?
23:24.17hmmhesayswell first check what they are sending into you
23:24.19`SauronHum.
23:24.24SirPrizedoes the specification that I've done like this work at all ?
23:24.26`Sauronwhat linux command returns a sockaddr?
23:24.28`Sauronerr
23:24.30`SauronC function
23:24.31`Sauronblah
23:24.34hmmhesayscall in and sip show channels
23:25.07SirPrizeusing "alaw" format
23:25.12*** join/#asterisk verge (~jfargen@56-116.26-24.tampabay.res.rr.com)
23:25.17hmmhesaysthey are sending you alaw?
23:25.23SirPrizeyeah
23:25.51hmmhesaysand you are using alaw in xten?
23:26.18vergeI am new to * and I am looking for some guidance.
23:26.28Sedorox~docs
23:26.30jbotDocumentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org
23:26.38SirPrizeI checked just now - X-lite when it connects uses "gsm".  Explains why that works and PSTN doesn't.  I see now
23:26.58hmmhesayswell
23:27.11hmmhesaysin your sip.conf set the dtmfmode for xten to something other than inband
23:27.17hmmhesaysthen it will work
23:27.22hmmhesaysand you can still use gsm
23:27.40SirPrizeit's actually x-ten that DOES work, and PSTN incoming alaw lines that didn't work on the menu
23:27.55SirPrizeah - I see what you mean
23:28.18hmmhesaysok... so you call in from the pstn and asterisk picks up and gives you ivr?
23:28.24SirPrizeyes
23:28.27hmmhesaysand the buttons aren't working on the ivr menu's?
23:28.36vergeOk, I have * running and have looked at the docs. My questions are rather specific in regards to connecting to the PSTN through livevoip using IAX.
23:28.40SirPrizeexactly - from PSTN doens't work.  From XTen works
23:29.55vergeI am not sure how to map my DID's to my extensions.
23:30.04vergeCan anyone help me with this question?
23:30.06hmmhesays~docs
23:30.08jbotDocumentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org
23:30.23*** join/#asterisk file[laptop] (~file_lapt@mctn1-142166197096.nb.aliant.net)
23:30.38hmmhesaysSirprize set an extension to call a sip endpoint, press buttons on the calling phone
23:30.41hmmhesayssee if you can hear them
23:30.58hmmhesaysverge that is a very vague question
23:31.12SirPrizemmmm.... I changed the dtmfmode in the [sipgate.co.uk] section to rfc2833, but that didn't make a difference.  Is this the right way I am trying, to set the dtmfmode for incoming SIP registered channels?
23:31.38*** join/#asterisk abatista (~Ariel@dsl-20-177.cofs.net)
23:31.41SirPrizehmmhesays: I did that before to check, yes, I hear the button presses.  I've tried calling in from both mobile and land phone, and neither of them can access the IVR
23:31.58*** part/#asterisk numBone (~numBone@c-24-129-204-233.se.client2.attbi.com)
23:32.00hmmhesaysSirprize dtmfmode has to be inband for alaw
23:32.15hmmhesaysand the ivr menu's work using xten
23:32.31SirPrizehmmhesays: yes, menu works with xten, doesn't work with mobile or landphone
23:32.38Sedoroxverge: I think what you want is to have exten => <didnumber>,1,stuff-here in your context that your line is set for
23:32.38hmmhesayshrm
23:32.53vergehmmhesays: I am not sure what I should use for context= in iax.conf.
23:33.10hmmhesayscontext is the context in your extensions.conf
23:33.47hmmhesaysSirPrize that's a good one
23:34.06SirPrize?
23:34.47hmmhesayshow new is your build sirprize?
23:35.44SirPrizeam using asterisk-1.0.5 built from source the day before
23:35.50SirPrizestandard stable package
23:36.05vergesirprize: this is what I currently have as my exten "exten => 2000,1,Dial(SIP/2000,20)"
23:36.12vergeshould I change 2000 to my DID?
23:36.29hmmhesaysregister => 1433188:XXXXXXXF@sipgate.co.uk where does the call go when it comes in?
23:36.34hmmhesaysyou don't have an extension specified
23:36.56SirPrizehmmhesays: It goes to the 's' extension in the 'incoming' context.  I can hear the IVR activate
23:37.50SirPrizeverge: Sorry, I haven't been following your thread.  Let me check if I know what the
23:38.00hmmhesaysverge is just talking to anyone who is talking
23:38.01hmmhesayslol
23:38.04Sedoroxverge: what you wanna do is on the context that you have for the incoming.. you want to set exten => <did>,1,Dial(SIP/2000,20) to have it dial that phone
23:38.48Mavvie~q931
23:39.00Mavviejbot: q.931 ?
23:39.03*** join/#asterisk eye69 (magnus@ipv6.upcore.net)
23:39.35SirPrizehmmhesays: could you please confirm whether the [sipgate.co.uk] entry I have does in fact affect the registered incoming SIP channel?  Is there any way I can check what the dtmfmode on an incoming line is set to by default?
23:39.51Mavvieworthless bot.
23:40.13hmmhesaysSirPrize ok, try this.... register => 1433188:XXXXXXXF@sipgate.co.uk/12345 in sip.conf and  exten => 12345,1,SIPDtmfMode(inband) exten => 12345,2,goto(s,1)
23:40.20hmmhesaysin extensions.conf and test
23:40.30SirPrizeok, let me try that now
23:40.51eaperezhhi people
23:41.26eaperezhone quick question. is there any ready available script/plugin for * to do voicemail to email?
23:41.43Mavvieeaperezh: it comes with the package.
23:41.49ManxPower~docs
23:41.50jbotDocumentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org
23:41.50Sedoroxits already part of voicemail
23:41.52hmmhesaysset up sendmail and you're set
23:41.53Sedoroxlook in voicemail.conf
23:42.41Mavviehmmm... somebody here with a copy of the Q931 specification?
23:42.55hmmhesaysit's on voip-info isn't it?
23:43.06Mavviehmmhesays: only links to the ITU websites
23:43.18hmmhesaysis that not what you want?
23:44.40eaperezhMavvie: sorry but can you point me? i cant seem to find it
23:45.05hmmhesaysvi /etc/asterisk/voicemail.conf
23:45.06Mavvieeaperezh: in voicemail.conf, there is a field for the email address of the voicemailbox
23:45.21Mavviehmmhesays: oh, I can download three specs for free!
23:45.45hmmhesaysas for setting up sendmail, grab one of the eleventy billion guides you can find on your favorite friend and mine... GOOGLE!
23:46.06hmmhesayscould probably grab it off peer to peer also Mavvie
23:46.50eaperezhMavvie: will the voicemail remain as voicemail or will it be deleted and only remain in the user's mbox?
23:46.59Mavvieit will also be there.
23:47.10hmmhesayseaperezh:rtfm man
23:47.11eaperezhMavvie: nice
23:47.34eaperezhMavvie: will check on that right away.....many thanks
23:47.52Mavvieeaperezh: but there is nothing new I told you which you could find in the voicemail configuration file.
23:48.29Mavviewhich makes me wonder if you actually are skilled enough to get it all up and running.
23:48.29SirPrizehmmhesays: mmmmm......... am trying to implement that change, but now when I try to call in, Asterisk logs this message "Channel 'SIP/sip.gossiptel.com-081471a8' sent into invalid extension 's' in context 'incoming', but no invalid handler", and I get a 403 Forbidden ?!
23:48.35eaperezhhmmhesays: im new to *, im not a programmer and have u take a look at the manuals? explanation is below average
23:48.44`SauronAnyone on a linux box, run the following for me: grep '_len' /usr/include/bits/sockaddr.h
23:49.06`SauronHurmph.
23:49.11hmmhesays'Sauron: what is that going to do to my linux box?
23:49.13Mavvieeaperezh: ignorance is not an excuse.
23:49.13SirPrize`Sauron: No hit
23:49.15`SauronNothing
23:49.21`SauronSirPrize: Thanks. Grr.
23:49.24hmmhesayseaperezh: most of us in here are not programmers
23:49.42eaperezhMavvie: i do have i up and running with 4 iax phones and 2 port fxo card
23:49.48hmmhesaysin fact, i'm an insurance salesman from idaho
23:49.50`SauronAccording to UNPv1, sockaddr->sa_len should exists, but it's not created in sockaddr.h
23:49.56eaperezhMavvie: im implementing new functions as i learn
23:50.10Mavvieeaperezh: how did you do that? not by just guessing I take it?
23:50.14*** join/#asterisk Vulture- (~Vulture@109.238.204.68.cfl.res.rr.com)
23:50.16hmmhesaysSirPrize: did that work?
23:50.46SirPrizehmmhesays: I'm trying to implement that, but Asterisk now gives me an error message of "Channel 'SIP/sip.gossiptel.com-081471a8' sent into invalid extension 's' in context 'incoming', but no invalid handler".  Trying to figure out what's happening
23:50.49eaperezhMavvie: well (gasp) by readin the .conf files but i was not clear about the voicemal thing
23:51.04Mavvieeaperezh: so you did read the voicemail configuration file?
23:51.15Vulture-SirPrize: you need a context [incoming] and s,1,(command)
23:51.27Vulture-in extensions.conf
23:51.53eaperezhMavvie: kind of but i was thinking that the voicemail was going to be removed and placed only on the mbox
23:51.55SirPrizeVulture-: I do have both of those in my extensions.conf already. :-S
23:51.58`SauronSirPrize: You don't have an extension s defined in [incoming]
23:52.15Vulture-SirPrize: pastebin your extensions.conf
23:52.31hmmhesaysyeah I might have given you a bad command, or you didn't type it right
23:52.31`Sauronalso, make sure you do an extensions reload, just in case. :)
23:52.34eaperezhMavvie: thats the original question...but thanks for clarifying it for me
23:52.34Chujior just 'show dialplan incoming'
23:52.40Vulture-`Sauron: nice call
23:52.41Mavvieeaperezh: maybe you should read the configuration file *again*, and look for something like "from the server"
23:53.12eaperezhMavvie: will sure do.
23:53.13SirPrize`Sauron: Ok, I corrected it.  I had a duplicate line in there.
23:53.21Vulture-;)
23:53.22hmmhesayseaperezh: http://www.voip-info.org/wiki-Asterisk+config+voicemail.conf
23:53.40hmmhesaysthere's some good reading
23:54.33mooboiin order, what files should be configured first ?
23:54.46SirPrizehmmhesays: Now when I call in via XTen, it says "Inband DTMF is not supported on codec gsm. Use RFC2833", and PSTN phone still doesn't work.  Will try switching to rfc2833 and see what happens
23:55.11hmmhesaysSirPrize, we're just testing something, call in from the pstn
23:55.18hmmhesaysto your IVR menu
23:55.29*** join/#asterisk kFuQ (~somedude@c-24-17-224-78.client.comcast.net)
23:55.29SirPrizehmmhesays: it still didn't work even with inband, calling in from pstn
23:55.42hmmhesaysit should have
23:55.47hmmhesaysare you sure it didn't work?
23:55.54hmmhesaysbecause it should have
23:56.03eaperezhtomorrow i will learn how to put all this stuff in to a DB.......thanks all for you kind help and hmmhesays for the links
23:56.27`Sauroneaperezh: for the DB stuff, google for ast_data
23:56.36hmmhesaysmooboi: whatever your heart desires
23:56.43chipigast_data++
23:56.55SirPrizehmmhesays: yeah, unfortunately it didn't work.  :-( let me paste some things into pastebin
23:57.06hmmhesaysok
23:57.15mooboibut which confis executed as soon as the line rings ?
23:57.17`SauronI need to reconfigure ast_data here
23:57.26terrapenbrb
23:57.38SirPrizehmmhesays: http://www.pastebin.com/243897
23:57.39*** join/#asterisk BoRiS (boris@S01060040ca1e5b54.wp.shawcable.net)
23:57.52Chujimooboi : What kind of 'line'?
23:57.58SirPrizeLine 2 & 3 are for the incoming XTen call.  Everything below is for the PSTN incoming call
23:57.59hmmhesayshaha I really have no life, I should be out snogging some hot chick right now
23:58.04hmmhesaysinstead i'm doing tech support
23:58.07hmmhesaysk
23:58.10*** join/#asterisk terrapen (~cjs@fw-01.satx.bikeworld.net)
23:58.24SirPrizehmmhesays: with the help you're giving all of us, you sure do deserve the hottest chick :)
23:58.31hmmhesaysLOL
23:58.34hmmhesayswahoo!
23:58.50SirPrize*grin*
23:58.50hmmhesaysSirprize you got ssh access to your box?
23:59.08terrapenanybody here deployed a unified Asterisk PBX at multiple locations?
23:59.10SirPrizehmmhesays: yes, I do
23:59.21hmmhesaysif you give me access i'll take a look
23:59.30hmmhesaysi promise not to fuck it up
23:59.45SirPrizelet me set up an account ...
23:59.47hmmhesaysk
23:59.53hmmhesaysterrapen: yes

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