irclog2html for #asterisk on 20051229

00:05.17*** join/#asterisk postel (n=jk@area41.OSPF.netmonks.net)
00:05.57*** join/#asterisk YaroMan (i=YaroMan@cpe-204-210-153-209.hvc.res.rr.com)
00:07.29*** part/#asterisk YaroMan (i=YaroMan@cpe-204-210-153-209.hvc.res.rr.com)
00:07.43*** join/#asterisk YaroMan (i=YaroMan@cpe-204-210-153-209.hvc.res.rr.com)
00:07.50YaroManHi
00:07.59YaroMani need some help with asterisk!
00:08.19mogormanheh
00:08.21mogormanyou and the world
00:08.22*** part/#asterisk eKo1 (n=bernd@63.245.57.70)
00:08.37YaroManI installed a server at datacenter with Asterisk @ Home 2.2
00:08.49YaroManbut my SIP phone cant conect to server
00:08.53YaroManwhat should I do
00:08.54*** join/#asterisk kram (n=mark@gateway.digium.com)
00:08.59YaroManmy ping is 4 MS to datacenter
00:09.01*** join/#asterisk monachoi (n=bonvoyag@cpe-24-174-162-34.satx.res.rr.com)
00:09.05mogormanuninstall asterisk@home
00:09.10*** mode/#asterisk [+o kram] by ChanServ
00:09.22YaroManwhy should I uninstall it?
00:09.39mogormanbecuase its horrid
00:09.40bweschkehey kram! how's paris?
00:09.57YaroManmaby i need to setup some special dns settings for it?
00:10.06mogormanprobably not
00:10.07trixterwoo just got my free stuff for  the sacramento asterisk users group -- for contest prizes..  5 handytone 286s 2 gxp2000s and a tdm410p.  hopefully that is incentive enough for people to actually compete :)
00:10.10kramit's okay
00:10.13krami'm pretty tired though
00:10.17mogormanmarko!
00:10.20mogormanyou back?
00:10.21trixterthanks to digium and thevoipconnection.com
00:10.27krami get back jan 2
00:10.46mogormanahh lame
00:10.48bweschkeu over on biz or pleasure?
00:10.48mogorman^_^
00:10.59mogormanwhats the difference bweschke
00:11.24bweschkemogorman: I would say sleep, but I know you all better than that. :)
00:11.48mogormanheh i have sleep in a can
00:11.51*** join/#asterisk SIPposed (i=Spaceb@h88n1fls309o838.telia.com)
00:12.07bweschkei'm sure u do.
00:12.39*** join/#asterisk ryan (n=ryan@londonderry-cuda2-68-171-162-161.lndnnh.adelphia.net)
00:13.10SIPposedAllow me to be the dumb and stupid one for some minutes, I have my reasons. has anyone had success making Asterisk work on Windows 2003 server?
00:13.19tzangermerry christmas, happy new year and all that, kram
00:13.39bweschkeSIPposed: about as much success as driving the reverse direction on a state highway
00:13.47SIPposedhhehe
00:13.54bweschkethere is a port for Cygwin, but it's not recommended for production I think
00:14.18SIPposedwell
00:14.24SIPposedI am running it
00:14.41bweschkecongrats.
00:14.44meredyddtzanger: hallo.
00:14.55tzangermeredydd: gah!  You're invading my channels again!
00:14.57meredyddtzanger: Would you perchance be the same guy from #vexi?
00:14.58SIPposedbut I think my NAT is catching it before it passes the proxy
00:14.59meredydd:D
00:15.06SIPposedhmm
00:15.17tzangerwhat's up what's new
00:15.32meredyddwell, I'm getting into IP telephony, as you may have gathered :)
00:15.58meredyddwhat are you doing these days?
00:16.01tzanger:-)  I've been here for about 2.5 years.  long enough that the regulars /ignore me now
00:16.14SIPposed:P
00:16.29SIPposednot after just 5 mins hehe
00:16.30_Sam--western digital sata drives ok?
00:16.34*** join/#asterisk Ateboy (n=ugob@modemcable002.152-81-70.mc.videotron.ca)
00:17.08tzangerI like WD
00:17.16tzangerhell I get by with maxtor, WD have never done me wrong
00:17.31_Sam--ive had a bunch of maxtors go on me in the past year
00:17.42_Sam--i was leaning towards seagate w/5yr warr.
00:17.45YaroManHelp!!! "sip.broadvoice.com:5060         8458671927@s       120 Request Sent"
00:17.52tzangeryeah seagate is always nice
00:17.52YaroMancant go no where
00:17.55meredyddtzanger: Cool. In that case, you might actually be able to answer my question :P
00:17.57AteboyHi, little question about call waiting and asterisk
00:18.10SIPposedtzanger, do you know of anyones in here, who has supposedly experiemnted with any winblows SIP proxies? like the Asterix Cygwin edition?
00:18.11tzangermeredydd: quite possibly, unless it has to do with SIP, I've avoided that particular protocol
00:18.19meredyddah, bugger.
00:18.22tzangerSIPposed: no...  I avoid SIP
00:18.30SIPposedyou do?
00:18.34meredyddYeah. I'm beginning to understand why.
00:18.39SIPposedheard it was the lastest shit
00:18.47meredyddIf this continues, it looks like I'm about to have to roll my own implementation.
00:18.54SIPposedmaybe it is just shit? =P hehe
00:19.00meredyddas there are NO decent libraries for sip+voip
00:19.03tzanger:-)
00:19.06tzangeractually there are
00:19.08AteboyIf I use my pstn line that features a call waiting option, and connect it to a fxo port, will * ever know if someone calls while I'm on the phone?
00:19.16tzangeropal is there, there is also another I can't for the life of me remember
00:19.19meredyddSIPposed: It's hideously complicated, the protocol stack makes your ears bleed, and debugging is a pain.
00:19.20tzangerthere are some stinkers though
00:19.41SIPposedyes meredydd, I have figured that part out
00:20.01meredyddI mean...
00:20.14meredyddwhat kind of brain-dead protocol sends its control signals as UDP??
00:20.24fileSIP can send it via TCP
00:20.31SIPposedwell
00:20.32meredydd(sometimes. Other people use TCP. So, guess what? You have to implement it both ways!)
00:20.46meredyddfile: Yep. That makes things even worse.
00:21.34SIPposedI got this nice IP adaptr from my telecompany, selected the biggest and most expensive one, in the hopes, taht it would work more easily. But I were wrong
00:21.41meredyddtzanger: I am dangerously close to proposing marriage to you.
00:21.42tzangerno SIP sends everything in english... that was the most braindead decision
00:21.52SIPposedof cause it is locked up, so I have no access to the inside
00:21.57tzangerthe only thing they could have done to make it worse was to encapsulate the RTP in XML
00:22.03meredyddtzanger: Heh. I actually like that. It gives you a cat in hell's chance of understanding what's going on
00:22.06tzangermeredydd: I am kind of lonely
00:22.15tzangermeredydd: yeah but fuck is it a pain
00:22.26tzangerlet's ENCODE all the data into english, transport it, and then DECODE it again
00:22.27tzangerbleh
00:22.29meredyddtzanger: See my previous comments :)
00:22.42tzanger>-)
00:22.44tzangerer :-)
00:22.57*** join/#asterisk J4k3 (i=j4k3@dhcp-12-197-128-55.intrastar.net)
00:23.00Ateboyanyone has a clue or a link to help me about * and call-waiting?
00:23.02SIPposedI have written SMTP, HTML1.1, XML, even WebDAV parsers, I am NOT going to try on SIP too
00:23.19meredyddSIPposed: Oh, well, there's osip for that.
00:23.31tzangerSIPposed: well you have all the ingredients for a hell of a SIP parser
00:23.32SIPposedosip for what?
00:23.40meredyddSIP parser/transaction manager.
00:23.44tzangerlord knows it'd probably implment all of the stack... something nobody's done to date :-)
00:23.46meredyddBut then you have to build the session manager on top of that
00:23.49SIPposedallright
00:23.55SIPposedhmmm
00:24.04meredyddand then you need to grab an RTP lib, and use that
00:24.17meredyddand stitch it together with the codec libs, which seem to be only available separately...
00:24.23SIPposedto accomodate the "secret" interests of my telecom
00:24.40meredydd(I just wish the world would use (open)h323. Life would be nice again.)
00:24.44tzangereep
00:25.04_DAWAteboy: You still using the SPA-3000?
00:25.09tzangeroh323 is only better than SIP because it tries to encapsulate telecom standards into internet formats
00:25.15tzangerIAX2 baby... finish off that protocol and you have a winner IMO
00:25.23Ateboyyes
00:25.37meredyddtzanger: Okay, fine. As long as there are good libraries, I don't care.
00:25.44Ateboy_DAW: I found a thread on the mailing list... reading
00:25.48SIPposedwell, I need a hint: IF the adapter is aware, of the proxy on my gateway, is it supposed to contact my isp directly on port 5060?
00:25.58_DAWthe spa handles the call waiting
00:26.04tzangermeredydd: well.. the protocol's not quite there so the library isn't quite there, and you can only talk to * and PA1688 phones
00:26.18meredyddBut I want to do voice plumbing. I do NOT want to have to implement most of the damn protocol stack before I can bridge between two systems.
00:26.33tzangeryeah
00:26.37tzangerjust use netcat then :-p
00:26.58meredyddtzanger: Oh, that it were so simple.
00:27.10SIPposedwell, thing is, that I thought being smart, invested in IP telephone, instead of the regular one
00:27.36*** join/#asterisk kazalt (i=kazalt@Quebec-HSE-ppp220225.qc.sympatico.ca)
00:28.07tzangerhahaha
00:28.08tzanger"While I'm all for porn and violence, let's not pretend that it somehow builds character and prepares you for life"
00:28.16SIPposedand it worked well, when only the modem was connected. The telecom pretty much pulled my nose "Sorry lad, our systems only allow us to give you one public IP"
00:28.16sansanhummm, turned off acpi, and it doesn't panic when loading zaptel, just freeze :)
00:28.26tzangersansan: ugh
00:28.27Ateboy_DAW: example, If I welcome people with a greeting like "for john, press 20, for johnny, press 21, for ateboy, press 22".  Someone calls in, it hits 22, so ateboy is talking with someone over the pstn line.  What would happen if another person calls on te PSTN line?
00:28.35tzangersounds like some kind of nasty PCI IRQ routing issue
00:28.57*** join/#asterisk zu (n=zu@14-pool1.ras14.floca.alerondial.net)
00:29.04tzangerdammit I downloaded hitch but it's on polish
00:29.13tzangerand other than a few cuss words I don't know polish
00:29.23zulol
00:29.32SIPposedthe movie?
00:29.44sansannow, it just says, "zaptel: no version for "struct_module" found: kernel tainted.", before freezing
00:29.59tzangersansan: get a NORMAL kernel and a NORMAL zaptel build
00:30.18meredyddtzanger: i liked that movie (and I'm not afraid to admit it)
00:30.24sansantzanger, only the kernel is diferent, zaptel came from asterisk.org
00:30.37tzangersansan: well get yourself a normal kenrel
00:30.51tzangerand don't use 1000Hz timer, it's useless for * and mucks up zaptel anyway
00:31.10tzangermeredydd: it was recommended to me but again... I can't understand Polish so I'm back to square one
00:31.15sansani'm afraid, it cause mess on  centos, but ok i'll try it
00:31.46zutzanger: umm asterisk runs just fine on centos 4x+
00:31.53tzangerzu: then help this guy out
00:32.07zuok
00:32.14SIPposedtzanger: Just a simple yes or no. My firewal is reporting my SIP adapter to connect directly to sip.telecompany.com:5060 ..not myproxy:5060 ..is there anything wrong with this?
00:32.16sansanredhat stuff (centos based on RHEL), can cause incompatibilities
00:32.25zusansan: have you done a yum -y update
00:32.37sansanzu, you have have zaptel running on centos 4.2?
00:32.41tzangerSIPposed: I honestly have no idea...  I avoid SIP for this exact reason
00:32.41zuyes
00:32.46sansanzu, yes centos 4.2  fully updated
00:32.50Corydon-wsansan: the other possibility is that you have a hardware failure of some sort... like bad memory or an overheating processor (or motherboard)
00:32.58zusansan what version of zaptel
00:33.04SIPposedanyone else who can answer this question?
00:33.10sansanzu, i had kernel panic when modprobinf zaptel, turned off acpi, and now it just freezes
00:33.22Corydon-wFreezes for zaptel are invariably caused by bad hardware
00:33.29sansanzu, i tried 1.2 from svn, and the release tarball 1.2.1
00:33.40zusansan what zaptel device are you using?
00:33.42sansanall the system freezes
00:33.53sansana tdm11b card
00:33.54Corydon-wsansan: how new is the system?
00:34.03sansanand a zaphfc too
00:34.09zutry pci=noacpi on the kernel line
00:34.39sansanwell, amd2000 with asus mobo, it worked fine before with asterisk and the tdm400 p card
00:34.45sansank
00:35.11Corydon-wsansan: have you done anything that stress tests the system, like building a kernel (or building gcc)?
00:35.29benjkSIPposed: if you use SIP, either use public IP addresses (not NAT) or build a tunnel
00:35.32*** join/#asterisk Ariel_ (n=Ariel@dsl-20-177.cofs.net)
00:35.38zusounds to me like a possible irq problem also check cat /proc/interupts
00:35.45Corydon-wWithout one of those activities, I'm not impressed by claims of "it was working fine"
00:35.56zudisable all the serial/parralell/usb ports
00:36.35zudude a successful emerge is more than a test for any gentoo system ;)
00:37.27Ateboysansan: is the * box connected to a good UPS?
00:37.47sansanCorydon-w, i was running *  and zaptel 1.0 before
00:38.01sansanAteboy, yes
00:38.34Corydon-wsansan: can you still load the zaptel 1.0 drivers?
00:38.37*** part/#asterisk jake1932 (n=jake1932@pool-68-236-10-151.phil.east.verizon.net)
00:39.08sansanoh wait! i see some IRQ routing conflicts on messages :)
00:39.10zuasterisk 1.2.1 with my patch to fix reinvite issues from 1.2x cvs is now working just fine for high latency sattellite connections :)
00:39.56zuI tested it on a 1000ms+ connection
00:40.26sansanyou can  simulate a high latency with the firewall
00:40.35sansanwell with ipfw on freebsd at least :)
00:40.44sansanpacket loss and everything
00:40.50zusansan you cant because sat connections break rfcs
00:41.04sansanah k
00:41.19*** part/#asterisk Primer (n=vi@sh.nu)
00:41.20zueven with tcp packets they send acks before they even have the packet :/
00:41.28sansanliars
00:41.41sansanjust like hard drives
00:42.02AteboyWhat protocol should I use if I need to do this:  phone -> firewall (nat) -> internet  -> firewall(nat) -> * ?
00:42.12zuiax
00:42.31zuits built with nat issues in mind
00:42.34AteboySIP won't do it?
00:42.35Ateboyok
00:42.43zuyea it will but its a pain in the ass
00:43.06Ateboyok, 'cause I thoulght I could use a sipura 841 for that purpose, but doesn't seem to support iax
00:43.11zuyou need to forward 5060, 10000-20000 ports + set a externip= in ast
00:43.24Ateboyzu : done that
00:43.34zuand use a stun server
00:44.00distortionand it still doesnt work w/some routers/firewalls :)
00:44.05zuyup
00:44.27zumostly you calls will proccess but there will be RTP issues
00:44.34zuthen no audio :/
00:44.48Ateboydistortion/zu : m0n0wall (freebsd) on both sides, for what I've tested
00:44.51benjkAteboy: build a tunnel
00:44.56zuuse sip debug ip address and rtp debug ip  address
00:45.00*** join/#asterisk mtnbkr (n=mtnbkr@c-67-165-9-234.hsd1.ct.comcast.net)
00:45.04Ateboypractical there, but not everywhere
00:45.20*** join/#asterisk nagl (n=nagl@vie-086-059-104-148.dsl.sil.at)
00:45.22Ateboythe device seems to not being able to register
00:45.36zuIm looking at getting one of the 941's.
00:45.44Ateboy941s support iax?
00:45.55*** join/#asterisk PBXtech (i=nik@71-38-252-47.slkc.qwest.net)
00:45.57zuAteboy: Thats a sip problem port 5060
00:46.05benjkAteboy: if you have FreeBSD on both ends, use KAME to setup an IPsec tunnel
00:46.24sansanor the easier openvpn
00:46.28zuIll buy a mac if I want ro use freebsd
00:46.32zu:P
00:46.42Ateboybenjk: I can build tunnels easily on m0n0wall... that is not exactly the point.
00:46.53benjkit is exactly the point
00:47.00zuget some cisco pix's
00:47.12benjkbuild a tunnel and you won't have any NAT traversal issues with SIP
00:47.28PBXtechis the chan_sccp stable enough for the 7971?
00:47.38*** join/#asterisk asteriskgeeks (n=SIPdawg@pbxtech.com)
00:47.38asteriskgeeks<PROTECTED>
00:47.54*** join/#asterisk tdonahue (n=tdonahue@208.51.101.201)
00:47.56Ateboyok, if the financial resources can not permit the construction of a VPN tunnel?
00:48.00*** join/#asterisk ManxPower (i=ewieling@6.sub-70-219-54.myvzw.com)
00:48.07benjkfinancial resources?
00:48.21sansanwhat's wrong with ipsec or openvpn?
00:48.29sansanfinancially :)
00:48.31Ateboyfinancial/time/whatever
00:49.05benjkSIP does require far more financial resources to be pursuaded to do double NAT
00:49.15Ateboyok
00:49.49*** part/#asterisk SIPposed (i=Spaceb@h88n1fls309o838.telia.com)
00:49.50Ateboyso my solution is to either get a phone that does IAX, build a VPN tunnel, or set an * on the other side?
00:49.53benjkso if you haven't got the "financial resources" to setup a tunnel, then you don't have the "financial resources" to use SIP in your environment either
00:50.07benjkyes
00:50.18benjkor an ATA that does IAX
00:50.53benjkthe tunnel is probably the easiest and cheapest way since you already got a Sipura ATA
00:51.12Ateboyyes, I got a spa 3000 and a 841 phone
00:51.31*** join/#asterisk ApEtc (i=apetc@ip68-3-225-51.ph.ph.cox.net)
00:51.40benjkwell, then to protect your investment into that equipment a tunnel would seem to be your best option
00:52.23Ateboyanyone has a recommendation for an ata that does IAX?
00:52.34benjkIAXy from Digium
00:52.40Ateboyok
00:53.01benjkAtcomm also make IAX ATAs
00:53.48AteboyI'll see what $ the person at the other end is willing to spend
00:53.54*** join/#asterisk Sedorox (i=brandon@smartserv/cna/Sedorox)
00:54.18benjkwhat type of router is at the other end?
00:54.34Ateboycheap netgear :(
00:54.42Ateboyrp614 I think
00:54.53Ateboyhence the reason of the need for $
00:54.54sansanwell to get openvpn working, you just need to port forward on one location
00:55.03benjkcheck if it supports IPsec. many SOHO routers not support at least 2 tunnels
00:55.07sansanon the other location, you set openvpn as client
00:55.19benjkoops
00:55.21*** join/#asterisk wundaboy (n=asdf@c-67-164-107-68.hsd1.or.comcast.net)
00:55.29benjkfreudian lapsus
00:55.47benjkI meant many do now support it
00:55.59Ateboyyes, but this one is fairly old...
00:56.33benjkdoes it at least support IPsec passthrough?
00:56.41Ateboysansan: were you talking to me, about openvpn?
00:57.13sansanAteboy, openvpn is easy to set up
00:57.20sansanpainless
00:57.39benjkif his router doesn't support IPsec, it wont support OpenVPN either
00:57.43benjkso this is moot
00:57.55sansanwhy not?
00:58.05sansanMTU problems?
00:58.08sansanchange it
00:58.36benjkbecause its a SOHO router at the remote end and it belongs to a customer/client/friend whatever
01:00.19ManxPowerI thought practically all cheap routers supported IPSec passthru these days.
01:01.13benjkManx: he said the router was fairly old
01:01.28Ateboywait... I'm discussing live with "the other side" :)
01:01.44ManxPowerbenjk, Ah.  Anyone that can't afford a cheap Linksys router should not be on the internet.
01:01.59benjkManx: :D
01:02.01Ateboyeheh, he's got a chep netgear router...
01:02.16ManxPowerI SO much hate netgear.
01:02.27Ateboythat's ok
01:02.30benjkAteboy: if it does IPsec passthrough, then you should be able to do it
01:02.30ManxPowerEvery single Netgear box we have bought in the past 5 years was crap.
01:02.37*** join/#asterisk N9URK (n=icechat5@user-0ce2dhc.cable.mindspring.com)
01:02.39ManxPowerOn the other hand some people think they are better than sex.
01:02.40Ateboyok, it works for many others...
01:02.51ManxPowerI call those people "crazy".
01:03.10benjkI have had good experiences with Netgear and lots of trouble with Linksys
01:03.47ManxPowerbenjk, older Linksys boxes did have significant issues with SIP
01:04.00benjknot with SIP, with IPsec
01:04.13ManxPowerbenjk, Doing passthru?
01:04.33benjkwhenever you set up a tunnel, they went stale after about two to three weeks
01:04.41*** join/#asterisk scolsuckz (n=scolsuck@202.58.252.15)
01:04.54benjkyou needed to power cycle them to reboot and it was ok for another 2-3 weeks
01:05.19ManxPowerbenjk, Ah.  We don't use Linksys for VPN anymore because we could find no way to keep the machines from bypassing the VPN and going directly to the internet
01:05.52benjkif there had been a way to automate this so they'd reboot every Sunday night or something like that it would still have been sort of ok, but you had to actually pull the plug and back in
01:06.04ManxPowerbenjk, Also we only have ever tried NetGear SWITCHES
01:06.19Ateboybenjk: if id does ipsec passthrough, then you should be able to do it... : what and how?
01:06.54AteboyI have a di-804HV at one of my clients'.  Tried once to build an IPSec tunnel and failed, but I must retry...
01:06.56benjkAteboy: you need to have some box behind the router to be the end point of your tunnel
01:07.13benjkeven a Windows machine can do that
01:07.16Ateboybenjk yes, I know, but that is extra $ and another point of failure
01:07.33AteboyI'd rather install a m0n0wall there as well
01:07.51*** join/#asterisk N9URK (n=icechat5@user-0ce2dhc.cable.mindspring.com)
01:08.11benjkwell, in that case its cheaper to buy a 50 USD Linksys router with support for 2 IPsec tunnels
01:08.17Ateboyyup
01:08.25PBXtechis the chan_sccp stable enough for the 7971?
01:08.40QwellPBXtech: I use the one from berlios.de, it works great
01:08.50Ateboybut I just tought of something: we're both on cable modem (dhcp), can I build an IPSec tunnel considering that?
01:09.17PBXtechTY
01:09.21benjkdoes your friend/client have an old PC they don't need anymore? like a Pentium first generation (50MHz/75MHz) ?
01:09.48AteboyI've done tunnels from dynamic -> static, and I must reset the tunenl if the dynamic ip changes, but what with 2 dynamic addresses?
01:10.18benjkthe dynamic addresses can be dealt with by using some service like DynDNS.org
01:10.20PBXtech[Qwell]: are you missing any major features?
01:10.31Ateboybenjk: well, just forget this avenue... it is either a change in protocol, either we change his router
01:10.37QwellPBXtech: not really
01:10.42PBXtechcool
01:10.49QwellPBXtech: name some, I'll tell you if they're supported
01:10.52Ateboybenjk :we both have a dyndns account...
01:11.01PBXtechwhats the diff in the 7970 and 7971?
01:11.16benjkAteboy: in that case the dynamic addresses are dealt with
01:11.17Ateboya DI804HV is 81$ CDN so that could do
01:11.43Ateboybenjk : ok
01:11.55benjkif he is happy to spend a little on a router that can be a VPN endpoint then that's a solution
01:11.55QwellPBXtech: The 7971 is technically 7979G-GE
01:12.01QwellPBXtech: which means gbit ethernet
01:12.06Qwell7971g-ge
01:12.11*** part/#asterisk Utah_Dave (n=boucha@0-1pool139-113.nas28.salt-lake-city1.ut.us.da.qwest.net)
01:12.37Ateboybenjk: ok, but I could get a Gnet ATA taht supports IAX for 59$CDN...
01:12.39PBXtechawe ok
01:12.40benjkotherwise, if he has got an old PC collecting dust, you could download Wolverine and turn it into a PIX compatible VPN firewall router
01:13.01Ateboybenjk: I'd rather work with m0n0wall, as I'm more familiar with it
01:13.27benjkfair enough
01:13.59AteboyI'll see if I can return the 814, I think I do
01:14.34benjkAteboy, sure you can use IAX but if he's got an SPA-841 he wants to use, then tunneling is a better option
01:14.56ManxPowerUm, do you really need a tunnel?
01:14.57benjkalso, you could run multiple SIP UAs on his end
01:15.11benjkManx: NAT on both ends
01:15.24Qwellnat on both ends isn't too much of an issue
01:15.25ManxPowerbenjk, Um, double NAT works just fine with Asterisk.
01:15.35ManxPowerI've done it many, many times before the Storm of Doom
01:15.35Qwelljust gotta know what you're doing
01:15.44benjkwell, it's a bad hack and doesn't always work
01:16.08benjkits not a solution, its a workaround that comes with compromises
01:16.14mogormanqwell!
01:16.18benjka tunnel is a solution
01:16.20Qwellmog!
01:16.21ManxPowerMy asterisk server at home was behind nat and I used my SIPura in hotels, and conferences.
01:16.33Qwellbenjk: a high overhead solution
01:16.37ManxPowerHell, I even used it when I was in madrid and that was double or tripple nat
01:16.43swm_~beat Qwell
01:16.44jbotACTION beats Qwell with a large stick.
01:16.47benjkQwell: not really
01:16.59Qwelludp>tcp>udp...
01:16.59swm_~beat everyone
01:17.00jbotACTION beats everyone with a large stick.
01:17.02Ateboymanx: is it very hare to configure to make SIP work through nat?
01:17.03Qwellthat's a bit of overhead
01:17.09benjkif you have just one or two SIP phones, it's negligible
01:17.16ManxPowerQwell, any decent Tunnel will use UDP or GRE
01:17.21QwellManxPower: oh
01:17.31QwellI was assuming he was saying stunnel or something :p
01:17.33ManxPowerAteboy, not if you know what NAT really is and how SIP works.
01:17.34AteboyI'd only have one or 2 tunnels
01:17.54Ateboysorry, not tunnels, 2 phones
01:18.04benjkAteboy: if you want a proper solution, a tunnel is what you want
01:18.12benjkanything else is duct tape
01:18.12*** join/#asterisk Sedorox (i=brandon@smartserv/cna/Sedorox)
01:18.15ManxPowerNow if you have some silly broadband connection with a dynamic IP address on each end, things get more complicated.  But they would for a tunnnel too.
01:18.37Qwelldyndns
01:18.50ManxPowercrappy DNS support in Asterisk
01:18.58Ateboymanx: the connexions are both cable modems woth dyn address.. but pretty stable (both the inet access + ip addresses)
01:19.08sansanis there a way to see for what kernel version was the zaptel module compiled?
01:19.15Qwellsansan: compiled, or installed?
01:19.21ManxPowerAteboy, Then configure the ports you want to use for audio in rtp.conf and on the SIPura.
01:19.22sansancompiled
01:19.33benjkManx: as long as the routers work fine with DynDNS you don't have to worry about that
01:19.34Qwellsansan: modinfo
01:19.42swm_dyndns? Isin't that a dynamic domain service that takes forever to update your information?
01:19.50ManxPowerThen portforward 5060/UDP and the ports you set up for audio (UDP)  (that's on the NAT for Asterisk)
01:19.53SedoroxMost cable companies... if you keep the router/router on... and it doesn't get reset often.. you end up keeping the same address.. as long as it can renew it
01:19.58sansanah yes thanks Qwell
01:20.25ManxPowerthen set localnet= and externip= in sip.conf on Asterisk.  put nat=yes in the peer's config section of sip.conf, tell the SIP devices to register
01:20.27swm_Comcast cycles ip addresses every 5 to 6 days
01:21.04Ateboymy ISP is good for that, been on the same IP for ~6 months
01:21.08ManxPowerOh, disable NAT support in your SIP devices.
01:21.44Ateboymanx: done all that, except "tell the sip devices to register", except if that means to power up the phone...
01:21.47ManxPowerand disable any SIP NAT stuff on the router if it does special stuff for SIP (like Ciscos do)
01:22.08swm_I killed a nat yesterday... not much blood such a small bug :)
01:22.10ManxPowerAteboy, you should set the port range for audio on the SIP device too.
01:22.17Qwellswm_: gnat
01:22.34Ateboymanx: my firewall isn't doing anything with SIP...
01:22.41ManxPowerAteboy, most don't.
01:22.47QwellManxPower: got an easy one for you.  Why does rtp use so many ports?  Does each channel need a unique port for rtp?
01:22.57ManxPowerbut the one time I don't mention it someone will have a router that does.
01:23.04Ateboymanx : eheh ;)
01:23.19benjkQwell: because the designers of SIP modeled a circuit switched telephone exchange
01:23.20ManxPowerQwell, each call uses 1 or 2 ports for RTP (I believe 2 ports(
01:23.25Qwellk
01:23.37benjkSIP is a circuit switched design on top of a packet switched transport
01:23.54implicitbenjk, not true
01:23.58ManxPowerI only portforwarded like 16384 - 16393 UDP.
01:24.01benjkit is true
01:24.05implicitnope
01:24.08ManxPowerI'll never send more than 5 calls thru it
01:24.11implicitincorrect
01:24.14benjkthe RTP ports represent the trunks
01:24.22benjkthe SIP port represents the signaling path
01:24.23Qwell10,000 ports seems extremely excessive, as the default
01:24.34benjkof a conventional telephone exchange
01:24.36ManxPowerbenjk, only if the endpoints of those trunks can magically move from location to location
01:24.40ManxPowerQwell, it is.
01:24.58benjkstill the design philosophy is based on the idea of a circuit switched exchange
01:25.04zuQwell: all 10000 ports transcoding :/
01:25.05ManxPowerAteboy, oh canreinvite=no in sip.conf as well.
01:25.07Ateboymanx: I' ve got some "NAT Support Parameters" in the 841 config... and RTP parameters (rtp port min, rtp port max)
01:25.18benjkhence the thousands of dynamically assigned RTP ports
01:25.20ManxPowerAteboy, turn off the NAT support parameters
01:25.20Ateboymanx: done for canreinvite
01:25.22QwellAteboy: You want to set the rtp min/max ports
01:25.32zuQwell with ser on a dual xeon you can do 100,000+
01:25.32Qwellor will * handle that entirely?
01:25.35ManxPowerI recommend using 16384 as your rtp port min.
01:25.41Ateboythey're 16384-16482
01:25.53ManxPowerand maybe 16393 for rtp port max
01:25.56implicitthe SIP part of it is not modled after circuit switched networks at all, if anything it is derived to some extent from the design of ss7 where it is a packet switched design used for the control of the circuit switched network
01:26.23Ateboyall the nat support parameters are to no, except NAT keep alive intvl : 15
01:26.24implicitif it was modeled after a circuit switched design you'd be looking at something like ccitt5 over IP :)
01:26.26benjkimplicit: what part of RTP ports represent trunks do you not get?
01:26.34implicitbenjk, you said SIP
01:26.37implicitdon't be a fucking dumbass
01:26.38ManxPowerAteboy, that's fine.
01:26.43benjkSIP/RTP
01:26.50implicit<benjk>SIP is a circuit switched design on top of a packet switched transport
01:26.51benjksince SIP alone won't do you anything
01:26.54ManxPowerI sould suggest you set the same port range in rtp.conf in /etc/asterisk
01:26.58implicitand RTP is not circuit switched at all
01:27.00implicitit goes point to point
01:27.07implicitin a SIP call
01:27.11AteboyI was wondering why the ATA and ip phones are not sold supporting more than one protocol...
01:27.34ManxPowerAteboy, Flash memory is not cheap.
01:27.34implicitnothing about the design is circuit switched, IAX trunking on the other hand, is
01:27.41benjkthe use of SIP and RTP for telephony is modeled after a circuit switched telephone exchange
01:27.56robl^Ateboy: not *universally* true.  some phones support multiple protocols
01:28.06benjkbecause it borrows that from H323, which is modeled after a circuit switched telephone exchange
01:28.06ManxPowerI'll bet the SIP RFC has information on what it's modeled on.
01:28.10implicitbenjk, the PSTN is not a strictly circuit switched network
01:28.13zuhmm,  im thinking about converting all the audio to g729 to avoid the transcoding overhead
01:28.19sansanzu, do you have the file gcc-version.sh on your system?
01:28.35implicitall the signalling and control on the modern PSTN is done over a packet switched network
01:28.35sansanfrom the kernel sources
01:28.38zuits at the office and theres no external access there
01:28.42implicitand the trunks are the only things that are circuit switched
01:28.46benjkManx: the SIP RFC doesn't say any such thing because SIP borrowed concepts from H323
01:28.59implicitsure, the RTP acts like individual channels through what you were saying about ports
01:29.01zurfc's plural
01:29.06implicitbut they are not switched as virtual circuits
01:29.18Ateboymanx: I see...
01:29.24implicitthey go point to point without caring about the underlying network structure at all
01:29.28benjkin thise case it would be singular as in original RFC
01:29.43benjkcause later ones wouldn't bother to mention any history/intro
01:30.11Ateboymanx: and then I must forward ports 16838-16482 to my * box ?
01:30.17implicitbenjk, admit that you are wrong
01:30.26implicitbenjk, admit it now, cause your ignorance is making us all sick
01:30.38implicitbenjk, and your pride and refusal to admit your dumbassedness is even worse
01:30.39zuInternet-Drafts are are working documents and have no standards status. They are valid for six months, and may be updated, replaced or obsoleted at any time
01:30.41benjkimplicit: the point is that the overall mechanism how all the parts work together are still following the general concept of a circuit switched exchange
01:31.02benjkwhich is a result of borrowing that from H323
01:31.07implicitbenjk, THEN SHOW ME AN EXAMPLE OF A SINGLE *VIRTUAL CIRCUIT* THAT YOU ARE DESCRIBING
01:31.11impliciteven in h323
01:31.14implicitit doesn't exist
01:31.17mogormanlol
01:31.23zuhttp://www1.cs.columbia.edu/sip/drafts.html is a good resource for sip rfc's
01:31.25mogormandarn
01:31.32Qwellmogorman: nope, you're just in time
01:31.35mogormansweet
01:31.37tzangerdamn I missed one too then
01:31.39mogormanboth of yall suck
01:31.43mogormani rule
01:31.43implicitmogorman, dont worry its still going on
01:31.44tzangeroh this zaptel thing again
01:31.48mogormanasterisk forever!
01:31.52benjkyou are obviously as ignorant of TCP/IP design principles as the designers of both H323 and SIP
01:31.53implicitbenjk is starting to realize his stupidity
01:32.03mogormanheh dont see that happening
01:32.15mogormanoh benjk i did get info on the lgpl thing
01:32.17mogormanyour wrong
01:32.23QwellPWNED
01:32.24Qwell:D
01:32.28mogormanand i will email whoever you want if you want facts checked
01:32.35implicitbenjk, lol, no examples?, if you are using B2BUAs, sure you can do some virtual circuit switching
01:32.47implicitbenjk, but hte protocol itself relies on NO circuit switched principals
01:32.53zuYou are using a konqueror3.5 browser. MICE can only be used with IE 5.0, IE 6.0, Netscape 7, Opera or Mozilla, not IE 4.0, Netscape 4.x or earlier browser lmfao the computer science at colubia university suck so much it can handle konq
01:32.54implicitif anything it tries to completely eliminate them
01:33.05zus/can/cant
01:33.11zuhttp://www.cs.columbia.edu/
01:33.37benjkthe point is that signaling and trunks are seen as separate channels that needed to be modeled separately
01:33.48Ateboymanx?
01:33.54benjkTCP/IP has layers for that
01:34.08zuyes but rtp is udp
01:34.47benjkso there is no reason to spread your data transmission out over multiple ports to keep them distinguishable from each other
01:34.47ManxPowerAteboy, Sorry, I was reading SIP and RTP RFCs
01:34.54Ateboymanx: lol
01:34.55ManxPoweryes, you want to forward the range of ports to your asterisk box
01:35.22zuManxPower: which one the link I posted?
01:35.41ManxPowerzu, google is my friend.
01:35.41Ateboymanx: but why is rtp involved even if I set canreinvite = no?
01:35.56ManxPowerAteboy, You'll understand someday, grasshopper.
01:36.03zuAteboy:  that hase nothing to do with the rtp stream for the audio
01:36.06*** join/#asterisk mmlj4 (n=looseduk@ip70-171-92-106.no.no.cox.net)
01:36.10ManxPowerfor now, just know that it is.
01:36.19Ateboygrasshopper?
01:36.24mogormanahh /me has to become mog_home till later
01:36.25mogormanpeace
01:36.25*** part/#asterisk mogorman (n=mogorman@gateway.digium.com)
01:36.44zureinvites are just saying hey you can connect directly to the device you are calling so it reinvites you to the end device
01:37.30ManxPowermmlj4, Satellite Internet SUCKS.
01:37.44Ateboyok, but rtp is used even w/o end-device-to-end-device communication?
01:37.52zuhttp://www.cs.columbia.edu/~hgs/rtp/ < links to rtp information
01:37.56QwellAteboy: rtp is where the audio is
01:38.00zuyup
01:38.05ManxPowerAteboy, RTP is what your AUDIO goes over.
01:38.14Qwellso, technically, it'll still work without rtp
01:38.18Qwellbut it'll be useless.
01:38.39ManxPowerQwell, SIMPLE, but I won't confuse the ussye.
01:38.41Ateboyok, thanks... if I had forwarded UDP/10000-20000, I should disable that now, right?
01:38.42Qwell:P
01:38.44ManxPowerissue, even
01:38.56QwellAteboy: just use the range you've set
01:38.59ManxPowerAteboy, you forward whatever you set in trp.conf on Asterisk.
01:39.25zuyup and with nat problems thats exactly what happends the phone will ring and if there is no rtp no audio :)
01:39.25ManxPowerAteboy, ..e.r.. rtp.conf
01:39.31Ateboymanx: done
01:39.41QwellManxPower: re asterisk-dev & ztdummy - well said
01:40.28ManxPowerAteboy, a FEW of the things I've told you to do are not technically required, but I recommend them
01:40.33ManxPowerQwell, thanks.
01:40.48Ateboywill * use the values for rtp in rtp.conf for both incoming + outgoing?
01:41.14ManxPowerAteboy, no.
01:41.17zuyea that and if you get a polycom phone get a good xml editor :)
01:41.51Ateboyok
01:41.56ManxPowerbut I RECOMMEND setting the ports on rtp.conf to be the same as the min and max ports on the SIPura.
01:43.02ManxPowerAteboy, btw, Cisco routers that are set to give audio a higher priority usually expect audio to be on UDP ports (I don't recall if even or odd) starting at 16384
01:44.11ManxPowerIt's a power of 2 so it's easy to remember.  If you ever deploy QoS on your network using Cisco stuff you won't have to change any of your asterisks or phones
01:44.50distortionanyone use voicemail with g729 passthrough?
01:46.56Qwelldistortion: you have you voicemail prompts recorded as g729?
01:47.13Qwellbecause unless you do, you aren't going to do passthrough
01:47.20distortionyes sir.
01:47.57ManxPowerdistortion, as long as you have everything in the same format, Asterisk should not have to transcode.
01:48.02QwellI think you'll still need a license, since it has to write the files
01:48.15distortionhere is the error i get: http://pastebin.ca/35106 lines: 252-255
01:48.17ManxPowerQwell, Well one license at least.
01:48.18*** join/#asterisk dcoulson (n=dcoulson@wilbur.geekcolony.net)
01:48.18Qwelland I don't know if g729 is a supported vm format
01:48.32distortionit seems like its allowed as a write format.
01:49.12Qwellit's recording the files as slin
01:49.20ManxPowerdistortion, Asterisk will ALWAYS prefer almost any other format over G729
01:49.43QwellIF it's supported, you'll need to be very explicit in voicemail.conf, about the format
01:50.05distortionyeah, i have format=g729 in voicemail.conf, and the sip.conf entry locked to g729
01:50.10Qwelland since the format is a global option, it will be a very large issue, if one of your phones don't support g729
01:50.48Qwellso, if you have zap users, forget it
01:50.56ManxPowerdistortion, ask on the mailing list, then file a bug.
01:51.31distortionManx: ok, thought id check first to see if there was something foolish i was missing.
01:51.37distortionthanky
01:51.45ManxPowerdistortion, It could be, but I'm missing it too then.
01:51.52QwellI have a feeling I know what the answer will be...
01:52.00Qwell"Just buy a few licenses"
01:53.31distortionI have a few licenses, the issue is more complicated... I am using g729 on the server as well, and g729 is broken on h323, all calls try to transcode to ulaw even if both endpoints are g729.
01:53.41Qwellneat
01:55.06Qwellahh...
01:55.13Qwelldistortion: I have a quick hack for you
01:55.23Qwellturn maxsilence off in voicemail.conf
01:55.35zuhmm I guess I am converting all the audio to g729 prompts on the sip only servers
01:55.38QwellIt needs SLIN for the silence detection
01:56.15distortionmaxsilence=0 or maxsilence=off?
01:56.19Qwell0 works
01:56.40zuSooo If I have a sip only server and I buy 1 g729 license and convert all the audio to g729 on it im in like flint
01:57.13Qwellzu: except for the aforementioned (major) drawback
01:57.24QwellIf you have any clients that don't support g729, voicemail will be a problem
01:57.34*** join/#asterisk zishanov (n=mail@d57-249-149.home.cgocable.net)
01:57.39zuyea and you know of any sip phones that dont do g729?
01:57.59zuThis is for a app that I dont care about the voicemail
01:58.08distortionqwell: heh, i think that worked :D
01:58.20[TK]D-Fenderzu : how many licenses do you have?
01:58.33Qwelldistortion: I accept paypal "thank you"'s. :P
01:58.52zu[TK]D-Fender:  I have lost count
01:58.59distortionhehe
01:59.20zukram, wake up and fix the god dam g729 online ordering system
01:59.32zulol
01:59.32lunkQwell: fees yo fees, twenty five cent to connect
01:59.57Qwellzu: email the person in charge of that
02:00.00*** join/#asterisk tengulre (n=root@221.11.5.180)
02:00.07tengulrehi,all
02:00.40zuQwell:  Why I have a iax connection to digium I can just dial there extension ;)
02:00.47Qwellzu: whatever works
02:01.41*** join/#asterisk Gimpy (n=knoppix@h24-207-33-168.dlt.dccnet.com)
02:03.47*** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn)
02:04.22*** join/#asterisk tengulre (n=tengulre@221.11.5.180)
02:04.25*** part/#asterisk Gimpy (n=knoppix@h24-207-33-168.dlt.dccnet.com)
02:10.08mrdigitalhttp://cgi.ebay.com/Asterisk-PBX-Installation-CD-WEB-GUI-Voice-Mail-SIP-CRM_W0QQitemZ5845649658QQcategoryZ11908QQrdZ1QQcmdZViewItem
02:10.22mrdigitalnot mine i just found it
02:11.11swm_I thought it's illegal to sell it?
02:11.15*** join/#asterisk zishanov_ (n=mail@d57-249-149.home.cgocable.net)
02:11.18MGSsancholol
02:11.21Qwellsell open source?
02:11.32swm_Yeah? You cant sell open source can you?
02:11.40Qwellso, how has RH made a business out of it?
02:11.51swm_LOL I dunno
02:12.02swm_Oh so I can sell open source and make a profit out of it?
02:12.10benjkthat's not true, the GPL doesn't forbid you to charge for open source
02:12.10Qwellyes
02:12.25swm_Well I'll see ya in a week as a millionare :) heh
02:12.32Nuggetsure, but it's a "wink wink, nudge nudge" sort of situation.
02:12.41swm_?!
02:12.44benjkthe only thing it requires is that you provide source code when you distribute
02:12.45Nuggetsince GPL'ing code reduces the market value of that code to $0, you can't sell it for very long.
02:13.03benjkand that you grant the same rights you have been granted under the GPL
02:13.04swm_Nugget: Elaborate
02:13.18benjkNugget, not necessarily
02:13.31swm_who has a definitive answer?
02:13.43benjkswm: the FSF
02:13.46junbugswm_: BSD'd you can sell w/o realsing source
02:13.50Ateboymysql sells open source
02:13.51swm_WTF is WSF?
02:13.58NuggetGPL'd code has an effective market value of $0.  The only way people have been successful "selling" GPL'd code have been companies that are really selling you service and support or companies that wrap you up in ugly EULAs
02:14.04Nuggetmysql does not sell GPL code.
02:14.10QwellAteboy: mysql sells mysql...which is not quite gpl
02:14.14Nuggetthey sell a commercially-licensed mysql which is not GPL.
02:14.17benjkFSF = Free Software Foundation, creators and guardians of the GPL
02:14.20QwellI mean...it's gpl, but...yeah, what Nugget said
02:14.22ManxPowerswm_, There are entire web sites devoted to legal GPL suff.  Unless you are a lawyer
02:14.33Ateboythey sell services for gpl code though
02:14.38benjkTake Yellow Dog Linux as an example
02:14.41NuggetI was not talking about services.
02:14.46benjkits sold on CD
02:14.47ManxPowerThe owner of the copyright gets special exemptions
02:14.49benjkwith sources
02:14.50Nuggetstrictly speaking of the code, the market value of GPL'd code is $0.
02:15.09QwellI think what RH is really selling, is the packaging, etc
02:15.12benjkand with a little delay it is available also as a free download
02:15.14Qwell(and of course, support)
02:15.22*** join/#asterisk postel (n=jk@area41.OSPF.netmonks.net)
02:15.23zuand updates
02:15.26benjkyet some people prefer to buy the CD pack
02:15.32NuggetRH also employs some nasty EULAs on their advanced server product, to build a moat around their GPL'd code.
02:15.40Qwellzu: no, you're paying for the right to use their service of updates
02:15.41Qwell:p
02:15.52zu:)
02:15.52Qwellzu: You're more than welcome to download the updates yourself
02:15.59swm_I thought you could only sell GPL code at the cost of making the CD & Shipping and nothing to make a profit margin?
02:16.05benjkand there is nothing in the GPL that says that Terrasoft can't sell their Linux distro as a CD pack
02:16.19Nuggetyou can sell GPL'd code for whatever the market will bear, but for all practical purposes that amount is $0.
02:16.27benjkswm: you could even charge for the right to download
02:16.30zuOr sell a server with gpl software and charge for support in the prices
02:16.52benjkwhat you cannot do is forbid anybody else to make it available for free after they donwloaded from you
02:16.56swm_zu: I like that idea heh
02:17.38swm_Just bought myself a cd printer/burner... $1,200 ...
02:17.47swm_Dual burns and prints! Woo hoo
02:17.53Nuggetredhat sells an embarassing number of copies of their advanced server product, and in just about every case the customer is buying it because they need to be able to say "we are running redhat advanced server" in order to receive support for their hardware or application software (like oracle)
02:18.00*** join/#asterisk _DAW-LAPTOP (n=DAW-LAPT@adsl-156-94-42.msy.bellsouth.net)
02:18.08Nuggetit's quite a nice scam for redhat, if you ask me
02:18.11swm_I would love a copy of redhat.. not fedora..
02:18.15SedoroxAnyone have Asterisk doing timing on a SuperMicro box w/o zaptel hardware?
02:18.16zuNugget or for some its CC cert
02:18.23Nuggetyeah
02:18.31swm_I guess redhat is more stable with 64 bit systems
02:18.44Nuggetnobody wants redhat, but lots of people need to be able to say they're using redhat.
02:18.47zuBut with xp gaining cc cert its basicly something you buy now
02:18.52swm_Sedorox: No timing devices suck
02:19.07zuyou should read the opensuse eula lol
02:19.10swm_Nugget: I prefer slackware heh
02:19.14zuyou cannot bundle or sell
02:19.22Sedoroxswm_: yea... and to top it off.. this supermicro.. is OCHI.... and ztdummy needs UCHI (or whichever way it is...)
02:19.31NuggetI use slackware too, but not on my oracle boxes (because I need oracle to answer the phone when I call)
02:19.38benjkOHCI and UHCI
02:19.40zuSedorox: run a newer version that uses rtc
02:19.53swm_Sedorox: So buy a cheap 19.99 X100P and use that for timing
02:19.56Sedoroxzu.. of zaptel?
02:20.05Sedoroxswm_: can't.. 1u.. only has 1 pci.. which has a SATA controller...
02:20.31swm_Sedorox: Change to a 2U??? heh
02:20.34zusince ztdummy  use kernel system tick timer if PC architecture RTC is avail
02:20.42Sedoroxlol.. can't :p
02:21.01Sedoroxzu:  know what version? I'm running 1.0.10.... maybe 1.2 supports it?
02:21.02swm_RTC is unreliable
02:21.11zuyea it works thouhg
02:21.15zu1.2x
02:21.20SedoroxI just need something for conference... and MoH
02:21.32swm_I built a Intel SE7500WV2 Dual 2.0 Xeon Core box and well... figured out why not to rely on the RTC
02:21.35zuI have servers with usb disabled and rtc used for meetme sip conf's
02:21.36SedoroxI didn't even realise to look about the usb when we bought this box
02:21.48zuthey are dual xeons too
02:21.57Sedoroxthis is just a Dual P3 1gig
02:22.11swm_1 Gig whadda waste
02:22.24Sedoroxwell... its our pbx...
02:22.38Sedoroxand also does backups/replicas for mysql
02:22.41swm_Dual P3 with 1 Gig of ram? WTF!
02:22.45Sedoroxnoooooo...
02:22.49Sedoroxproc's are 1gig
02:22.51Sedoroxonly 512 ram
02:23.03implicitSedorox, ok
02:23.04swm_Oh I think i need some nausea drugs
02:23.15Sedoroxlol
02:23.21Sedoroximplicit: eh?
02:24.52SedoroxI've been meaning to upgrade to 1.2.... just want to do it on my box here first.. work out thr quirks.. then put it on the production box...
02:25.29swm_Wonder if they could create a USB 2.0 Adapter that plugs into a 1.0/1.1 ? heh
02:25.34N9URKHas anyone here tried using "Using cepstral webbased voice synthesized demo to generate voice prompts for asterisk on the fly"
02:25.34implicitSedorox, don't worry about it
02:25.41N9URKfound at http://www.voip-info.org/tiki-index.php?page=Asterisk+text2cepstral+www+demo
02:25.53Sedoroxmmm ok........
02:27.58swm_I was reading somewhere if you stick a lamp cord on pins 1 and 26 of the usb control chip you can get some colorful effects @ 120V 5A :)
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02:35.08Sedoroxhmmm
02:41.26_Sam--can anyone point me in the right direction for piping asterisk console output to a front panel display on a case like this:   http://www.ahanix.com/ahanix_product.asp?pid=25
02:42.06*** part/#asterisk kazalt (i=kazalt@Quebec-HSE-ppp220225.qc.sympatico.ca)
02:44.53*** join/#asterisk postel (n=jk@area41.OSPF.netmonks.net)
02:47.11tzanger_Sam--: you could tail one of the logs
02:47.22_Sam--thats not a bad idea
02:47.29tzangeror do some System() commands in your dialplan at strategic points
02:48.16_Sam--im coming up at a loss though to find the linux/kernel type stuff to even make that thing work
02:48.26_Sam--what linux package would i look at?
02:48.44_Sam--there is lcdproc, but that is for a different type of front panel
02:49.20CoaxDlcdproc sux. i wrote a bsd licensed lib a while back that emulated lcdproc on a 20x4 display
02:49.31CoaxDwell, it didnt emulate. it was a completely different api and such
02:50.03tzanger_Sam--: there will be userspace drivers for it
02:50.09tzangercheck freshmeat.net
02:50.19tzangerahh lcdproc
02:50.22CoaxDeven did an auto-backlight-shutter-offer with pthreads.  (at the time, the matrixorbital lcd displays didnt have functional timer-shutter-offers)
02:50.26tzangerit's quite trivial to run that stuff
02:50.35tzanger"auto-backlight-shutter-offer" ?
02:50.39CoaxDtzanger; yeah
02:50.42tzangerwtf is timer-shutter-offer
02:50.53CoaxDtzanger; if ya set the backlight timer, it stayed on infinitely
02:51.00CoaxDtzanger: so i rewrote it in software
02:51.02tzangerahhhhhhhhhhh
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02:51.16tzangerI was thinking "offer" as in to grant access
02:51.18CoaxDtzanger; it'd spawn a thread and wait the default timeout period and then shut the light off
02:51.21CoaxDah. no
02:51.25CoaxDhehe
02:51.32cp5has anyone experienced asterisk 1.2.1 crashing intermittently when going into a Queue with Agents?
02:51.33tzangerand shutter as in what lets light through
02:51.42CoaxDtzanger: bwahahahaha
02:51.47tzangerso you can understand my confusion
02:51.57CoaxDfyi; band camp was gay.  but then again, so were the other movies
02:53.21CoaxDtzanger: ohyes, i understand
02:53.34tzangercp5: I don't use queues, sorry
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02:55.56cp5no prob
02:56.45Qwellcp5: I think there is something on the wiki
02:56.55Qwellcp5: And when are you releasing the new version of your stuff? :p
02:56.59Qwellerm, not wiki, bug tracker
02:57.02cp5what's up qwell :)
02:57.03QwellI keep doing that
02:57.05cp5i'll take a look
02:57.16cp5the nat2nat stuff?
02:57.19Qwellcp5: You promised me a release, months ago. ;)
02:57.37cp5i got consumed with a myspace worm i guess
02:57.48Qwellmyspace worm?
02:58.51*** part/#asterisk nswint (n=nswint@c-24-98-129-84.hsd1.ga.comcast.net)
02:58.53Qwellahahaha, that was you? :P
02:59.23cp5hah yeah
02:59.27cp5what have you been up to
02:59.52Qwellthis...
03:01.52Sedoroxanyone know what the zapnet and zapras flags do in gentoo/portage for zaptel?
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03:04.39SpaceBassd'oh
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03:24.15Qwellbkw__: y0
03:24.56postelcoppice: Hi, any idea where can i get networking and voip gear in HK?
03:25.12zuHonk Kong?
03:25.19postelyeap
03:25.28zumainland from shen zen
03:25.28junbugon any street corner i would presume
03:25.33posteland its HonG
03:25.34coppicenetworking gear is available just about anywhere. VoIP is really hard to find, for some reason
03:25.39Qwelljunbug: :P
03:25.56posteljunbug: heh, aint like that anymore
03:26.17zumainland china
03:26.37coppicedunno. I think i could probably find the odd hub or two in the local supermarket :-)
03:26.44postelzu: we're talking HK, mainland is faaaaar
03:27.03Qwellis HK part of China?
03:27.10coppiceif you are in sheung shui, the mainland is very close :-)
03:27.23postelQwell: since 1997
03:27.28coppiceQwell: I guess you are american
03:27.32Qwellcoppice: indeed
03:27.49Qwellcoppice: It'd be like you asking me if Toronto is part of Florida. :p
03:28.04coppicepart of china, and yet not part of china. you need a visa, and pass through immigration to get from HK to mainland china
03:28.09Qwellahh
03:28.10zuI live in florida and in the winter time it is part of toronto
03:28.21coppicepostel: what do you need?
03:28.39postelcoppice: routing gear, mostly junipers and ciscos
03:29.04postelcoppice: also some wireless cards and wifi routers
03:30.10coppiceoh, big stuff. there are several distributors for cisco, and at least one for juniper. check their distributor lists. simple stuff like wireless cards are cheaper when purchased in one of the several computer malls
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03:30.27gambolputtyhi
03:30.38TheCopsI'm services and reseller for cisco, if interesse postel.
03:30.47gambolputtyanyone know of a good way in * to generate a random number without using AGI?
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03:31.56postelcoppice: i think i found somebody to source my junies and he said he'll have a look for my ciscos, any idea of specific shops to source some cheap wifi pcmcias and a couple of WRT54 GS?
03:32.17postelTheCops: where are you located?
03:32.24TheCopspostel, Canada
03:32.27TheCopsQuebec
03:32.50coppicego in the malls in Mong Kok, Sham Shui Po, or Hennesey Road and there are lines of small shops selling linksys stuff
03:32.53postelTheCops: you're far away, my Tokyo rep is closer than you are
03:32.58TheCopslol
03:33.00junbuggambolputty: accessing $RANDOM env var?
03:33.08zuwww.ebay.ca
03:34.00gambolputtythats a bash or sh shell thing right?
03:34.35junbugyea
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03:36.06coppicepostel: going for a big deployment at mega-iadvantage?
03:37.03tengulreanybody using speakfreely under asterisk?
03:38.51postelcoppice: its an R&D project, lots of cash to kill, but lots of things that needed too, so im trying for best pricing
03:40.29postelcoppice: would the mongkok shops beat the pricing of Fortress and Broadway? any idea of the scale of difference?
03:40.47Kattyhi lads.
03:43.43mrdigitalKatty: how well do you know linux?
03:44.01Katty...
03:44.08Kattytwice as much as half.
03:44.08dudeswhat's linux???
03:44.43slappingtdoes asterisk do a good job keeping tele-marketers away?
03:44.44mrdigitalKatty: pm?
03:45.04Kattymrdigital: am.
03:45.20slappingtif you configure a VM menu system
03:48.03mrdigitalKatty: can i message you?
03:48.09coppicepostel: broadway and fortress are probably OK, but they don't have a wide range of stuff
03:48.32Kattymrdigital: please don't.
03:48.43Kattymrdigital: it's 10pm and my brain is off.
03:49.46tzangeryeah after 10pm I'm the only one who can message her :-0
03:49.48tzangerer :-)
03:50.21Sedoroxlol
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03:52.48coppicepostel: where are you?
03:52.57Kattytzanger: not /just/ you
03:53.02Kattytzanger: about three other people can too
03:53.09tzangerwhat?  why you hussy...  :-)
03:53.23Kattywhat's a hussy?
03:53.26tzangernot sure
03:53.28Kattydoes it have anything to do with hugs?
03:53.30tzangersounded right
03:53.31postelcoppice: in a abandoned dungeon under kai tak
03:53.47tzangerI'm watching a PBS show on tesla
03:53.50Kattytzanger: silly.
03:54.19coppicepostel: well, at least there's lots of good Thai food nearby
03:55.05posteli prefer sushi and sashimi, thai and chinese wont do it for me sadly
03:55.15coppicepostel: watch out of all the leaking oil, though
03:55.23postelheh
03:55.43tzangertesla's story is fascinating
03:55.52tzangerthere are a lot of conspiracy theories I know that surround him too
03:55.56coppicepostel: kowloon city is famous for thai food. you'll have to go at least an extra 500m for sushi :-)
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03:56.50Kattyhi mike.
03:57.33tzangerit's amazing... he tore up a ... shit what do you call it when you get paid so much per unit sold, etc.?  He tore up one that would give him $2/HP that was powered by AC
03:57.33postelcoppice: naah, just jump on a plane and pay Tokyo a visit, have a local itamae prepare some fugu sashimi for you, by the time you inish your sake you'll catch the flight back :>
03:57.33tzangerdo you have any idea what that would be worth these days??
03:57.47Kattytzanger: worth a hug?
03:58.05coppicepostel: that was probably more practical when kai tak was the airport
03:58.36Kattytzanger: i think a hug value would be more comprehendable right now.
03:58.45tzangerwell yes probably
03:59.11postelcoppice: kai tak was so cool, the plane almost touched rooftops before touching down, you were able to watch people through windows having diner
03:59.37coppicepostel: that airport was a lot safer than the new one, though
03:59.45tzangerthis reminds me of hedy lamarr
04:00.08coppicethe lady who got ripped off over CDMA
04:00.11tzangera hot chick *and* brilliant... god damn
04:00.15tzangeryup
04:01.00_Sam--do most linuxes support the mini-itx motherboards?
04:01.01coppicealthough i doubt anything practical could have been built before ay interesting patents ran out
04:01.07postelcoppice: the plane was coming parallel to the sea and then U-turning to reach the runway, people that were not flying HK often were scared to death, such a laugh
04:01.30tzangerbah I can only find her movies on ed2k
04:01.38tzangerno actual history on her technological achievements
04:02.05coppice_Sam--: I run FC3 on a mini-itx. some distributions screw up, though. FC4 would not install
04:02.28benjksomebody here from Canada?
04:02.33coppicepostel: it *seemed* like a U-turn, but was only something like 60 degrees
04:02.45SedoroxI know people in Canada
04:02.45Sedorox:p
04:02.45tzangeryep
04:02.47_Sam--thanks coppice
04:03.03TheCopsbenjk, me
04:03.13benjkcan people in Canada not call US 1-866 toll free numbers?
04:03.22tzangerbenjk: depends on what the 866 was provisioned for
04:03.34coppicepostel: I loved watching to 747s pass below me, while sitting on top of the kowloon hills
04:03.36benjkit was provisioned by NuFone
04:03.43tzangeryou can have them provisioned for US-only, Canada only, US/Canada, US/Canada/Mexico, you name it
04:04.01postelcoppice: 60 or 90 or 180 it was pretty amazing, all the people shouting and clapping hands in the cabin spoiled the moment
04:04.24benjkdo you happen to know what the default provisioning is NuFone uses?
04:04.32SedoroxI have a 866 thats US50/Canada....
04:04.33tzangerthat I don't
04:05.06tzangerhahaha tesla destroyed colorado springs' generator
04:05.07tzangerhahahaha
04:05.08tzanger"oops"
04:05.08benjkseen ~JerJer
04:05.17benjk~seen JerJer
04:05.23jbotjerjer <n=JerJer@pdpc/supporter/bronze/jerjer> was last seen on IRC in channel #debian, 1d 23h 7m 1s ago, saying: 'thank you'.
04:05.29benjk~seen shido6
04:05.30jbotshido6 is currently on #asterisk.  Has said a total of 5 messages.  Is idling for 9h 24m 28s
04:05.41coppicepostel: now they are spending a fortune cleaning up all the aviation fuel pollution so they can build housing on kai tak
04:06.25_Sam--coppice:  what kind of processor do you use in your mini-itx?
04:06.56postelcoppice: heh, cleaning up the pollution, thats a funny one, there are more minerals in HK's atmosphere than on the periodic table
04:07.27zuyea but at least they have lots of smoking resturants
04:07.33tzangergod damn
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04:07.43tzangercan you imagine being the guy responsible for bringing electricty to the world?
04:08.37coppice_Sam--: C3. I think the C7 is supposed to be OK with any Linux. the C3 needs to be treated as an i586. distributions keep screwing up and making one or two packages built for i686 only. in the case of FC4 it seems to be something in SELinux. Even if you choose not to install that, it still fouls up
04:09.07coppicetzanger: So you believe lightning was created by god? :-)
04:09.23zucentos works well on the via's
04:09.24tzangeryou *know* what I'm talking about :-)
04:10.09coppicepostel: when kai tak airport closed they expected to quickly redevelop it. then they found how much leaking from fuel tanks had occurred :-)
04:10.20_Sam--coppice:  thanks again for the info....do you run asterisk on it?
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04:10.57zuI run linux on the i586 vias it runs just fine
04:11.11coppice_Sam--: no. I use it as a nice quite domestic file/print/gateway/general jack of all trades box that runs 24x7
04:11.36_Sam--the c7 would be this one? :   "VIA Eden. EBGA 600 MHz fanless processor"
04:11.45_Sam--i see alot of the c3
04:11.52postelcoppice: if they do manage to clean the area up though it would have plenty of space for housing development
04:13.00coppicethe C3 is far more common. it has a couple of missing instructions that stop it being a proper i686. the c7 has those fixed. I'm not sure if the eden is the super low power version of the C3, or if its the C7
04:13.29_Sam--thank you...thats good info
04:13.33coppicepostel: and high value development too - good location and great views
04:13.59zuThe only problem on the c3s-c7s is lack of support for the watchdog timer
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04:14.26coppicewhat watchdog timer?
04:14.48_Sam--im just trying to wow some clients with different form factors and looks
04:14.55zua hardware watchdog time has a device node that must be written to evey minute, if it is not it reboots the machine
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04:15.09_Sam--those c3/c7 could handle a reasonable number of simultaneous sip clients/registrations?
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04:15.23coppicethe ITX boards are OK, but the boxes cost too much
04:16.08coppicezu: I didn't what a watchdog is. i asked which one you refered to. normal PCs have no watchdog, but some of the ITX stuff does
04:16.57coppice_Sam--: as long as the audio goes peer to peer, doing SIP is *very* lightweight
04:17.01postelcoppice: should i stay for new year's festivities or fly to jp? its the year of the dog coming up, right?
04:17.41_Sam--could it run a PRI?
04:17.49zuI prefer years of the dragon
04:18.07coppicepostel: nothing much happens at chinese new year in HK. its like christmas in europe or the US. everyone goes home to their families, and the SHOPS ACTUALLY CLOSE :-)
04:18.51zuThats one nice thing about christmas in the us you can still get chinese delivery :) yummy dumplings
04:18.57coppiceif you know where to look, there will be lion dances in various spots, but that's about it
04:20.09postelnot much then
04:21.14coppicewell, if your family is here it can be fun. for an outsider its really dull. much less colourful than new year in the chinatowns of big western cities
04:22.00postelcoppice: your english is far better than the average chinese i know, you're a gaijin yourself?
04:22.43coppicei'm one of those unusual tall, blond, blue eyed asians :-)
04:23.03postelheh, spot on
04:23.09benjkisn't that called gweilo in HK?
04:23.42postelmost likely, it seems i spent more time with japanese hookers than chinese ones
04:24.15benjkhaha
04:24.26coppicegweilo is correct - devil man
04:24.56postelyeah, they call people like yourself Ghosts
04:25.03benjksame in Japanese before the word 'gaijin' became popular - ohni = devil
04:25.38coppicewell, you can translate the gwei as ghost or devil. some kind of nasty spirit
04:26.19wasimour new years party theme is "angels and daemons" ...
04:26.41postelcoppice: so you must have been there for quite sometime then, you actually got to fly kai tak
04:27.06coppicei've lived here 15 years.
04:27.21postelwow! long time
04:27.35coppiceit used to be a great place for telecoms. now it seriously sucks
04:27.46coppicesame for electronics generally
04:28.29postelis the economy still as bad as it was? people were working overtime with no pay
04:29.35postela friend told me the manager was waiting by the door at the end of the shift taking notes by the door on people leaving
04:29.41coppiceworking overtime with no pay is the usual practice in asia. :-) the economy has improved a lot, but I can't work out how. somehow people seem to be making money from something. electronics is terminally ill, though
04:29.45postelnext dauy something like half of them were sacked
04:32.43coppiceit doesn't seem to be that much better in china. two or three years ago design centres were popping up everywhere in shenzhen. a lot seem to have closed lately
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04:35.56postelcoppice: mainland is and has been technology-work-dead, if HK doesnt give employment opportunities where do that thousands IT/technology graduates end each year?
04:36.55Qwellpostel: I know several that have come here
04:37.06zume 2
04:37.12postelQwell: #define here
04:37.25Qwell#define here localhost
04:37.56coppicethe mainland has a phenomenal output of electronics graduates. the standards are very variable, though. a few people, like Hua Wei and ZTE, do advanced stuff. Quite a few do original, but not particularly demanding stuff. huge numbers package kit solutions
04:39.14coppiceI can't imagine why anyone in HK would enter an engineering degree, other than civil, these days
04:40.22postelyeap, civil would help, you can be part of the construction team in kai tak ^_^
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04:54.06Kernel_corehi all
04:54.07livindedhas anyone else had problems sending their cid to cinguler numbers, whenever i set mine all i see is anonymous
04:54.16livindedcingular*
04:54.36Kernel_coreI have 10 SIP phone connected to asterisk , how do I enable call transfer on them ?
04:55.25Kernel_coreor conference ....
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05:00.43Qwelllivinded: name or number?
05:01.45wasimKernel_core: use features.conf or meetme.conf or the config on teh phones
05:02.08Qwelltransfer is often a feature of the phone.  What type of phone is it?
05:03.07livindedsamsung x427
05:03.12livindedit used to work fine
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05:03.33Qwelllivinded: not you :p
05:03.47Qwelllivinded: It's probably on Cingulars end...nothing you can do about it if that's the case
05:03.52livindedok
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05:10.41Kernel_coreQwell: I use Xten
05:11.32distortionhmm, so when will asterisk support sip via tcp? :)
05:13.15coppicewhen its good and ready :-)
05:13.36Qwellsooner, if somebody writes it. ;]
05:13.55distortion:( <-- has too many hung calls with canreinvite=yes
05:14.09Qwelldistortion: So don't reinvite...
05:14.50distortionwell contrary to my voicemail issue earlier, another server i have i do 6t1's of g711
05:15.02distortion== lots of bandwidth
05:17.48zuip? or tdm?
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05:18.27distortionip
05:18.33distortionwhich is 2x worse
05:19.56distortionseems to be like 20 megs of bandwidth
05:20.36zuya thats ulaw for ya
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05:21.19distortionwell, either sip+tcp to stop hung calls or g729+t38 for fax :D
05:29.51edison?
05:34.57*** join/#asterisk genmud (n=genmud@ip68-98-82-206.ph.ph.cox.net)
05:35.58*** join/#asterisk darkpadden (n=wes@adsl-066-156-076-204.sip.asm.bellsouth.net)
05:38.31*** join/#asterisk J4k3 (i=j4k3@dhcp-12-197-128-55.intrastar.net)
05:46.42linlinwhat config file would IVR related stuff tipicallt be located in
05:47.15Qwelllinlin: extensions.conf
05:48.52linlindo you know if AMP edits that file to do its IVR tasks also or does it do something strange
05:48.57Qwellmeh
05:49.00QwellAMP breaks things
05:49.07*** join/#asterisk tengulre (n=tengulre@61.185.224.66)
05:49.36linlinnot if you ONLY use amp
05:49.41linlinfrom what i hear
05:49.42Qwellyes, if you use only AMP
05:50.02QwellAMP is *awful*
05:50.09Nuggetyes it is.
05:50.17linlinive modified some stuff, added things to some other configuration files and i havnt noticed a problem, but i know what your saying, ive dealt with tools like amp before
05:50.24Qwellneed to find somebody to buy this hardware
05:51.28*** join/#asterisk jalsot (n=tamas@195.56.44.83)
05:51.49linlinanything cheap?
05:52.01linlincheap FXO card/box maybe?
05:52.09Qwelllinlin: It's just a few things.  mb+cpu+video card
05:52.18linlinoh ok
05:52.28linlinim afraid i have a rom full of that stuff :)
05:52.31Qwelland a tdm400p with an fxs if this other guy doesn't want it, heh
05:52.33linlinroom*
05:56.46*** join/#asterisk camonz (n=camonz@200.8.21.129)
05:59.51asterboyhow much for the fxs?
06:00.58Qwellasterboy: like $100.  I shouldn't even be answering that question in here...
06:01.53camonzhi
06:02.14camonzi'm having problems playing mp3s with asterisk on suse linux
06:02.27Qwellcamonz: what problems?
06:02.28*** join/#asterisk cricalixwood (n=rh@hillconsult.force9.co.uk)
06:02.48camonzfor some reason i have mpf321 installed and i cannot seem to remove that package
06:02.54camonzsorry, mpg321
06:03.27asterboyQwell, is that the only telecom hardware for sale?
06:03.32Qwellasterboy: yeah :p
06:03.57camonzalso when i do make mpg123 in the src/asterisk directory it says, it's compiling that versions, but in the end it only softlinks to mpg321 binary
06:04.22asterboynew is $140 so its a fair price...I'll inquire later if some of my bids fall through on eBay
06:04.58*** join/#asterisk jalsot_ (n=tamas@abacus.eworldcom.hu)
06:05.30postelcamonz: go to mpg123.de get 0.59r and build it
06:05.59Qwellpostel: nah
06:06.06Qwellmake mpg123 install
06:06.07camonzwouldn't building mpg123 on this system that already has mpg321 create inconsistencies?
06:06.58postelcamonz: no it wont, just simlink mpg321 to 0.59r and it would STFU
06:07.16camonztrying to remove mpg321 with rpm -e mpg312-xxx-xx gives me a segmentation fault
06:07.28postelthe joys of suse
06:07.40Qwellcamonz: What version of *?
06:07.41camonzthanks, letme try
06:07.45camonz1.2.0
06:07.50QwellJust use native MoH
06:07.56Qwellscrew mpg123
06:08.03*** join/#asterisk angom_h (n=angom@red-corp-200.76.251.120.telnor.net)
06:08.16camonzhow so?
06:08.45Qwellcamonz: look at the musiconhold.conf.sample
06:08.53camonzthanks :->
06:14.57camonzin the sample file it mentions the dir moh-native wich i don't have on var/lib/asterisk, is it ok, if i create the dir and add the files in there?
06:15.10Qwellcamonz: yep
06:15.39camonz:->
06:38.36*** join/#asterisk jalsot (n=tamas@abacus.eworldcom.hu)
06:43.52N9URKHI guys, I have a question that may be straightforward, but I can't figure it out.  I have a script generating a sound file when someone dials a particular extension and I would like to start the music on hold until the file is generated and starts playing.  WaitMusicOnHold(10) just delays the processing and MusicOnHold() seems to do nothing but play the hold music.  What command should I use?  THanks
06:55.37*** part/#asterisk N9URK (n=icechat5@user-0ce2dhc.cable.mindspring.com)
06:55.52*** join/#asterisk N9URK (n=icechat5@user-0ce2dhc.cable.mindspring.com)
06:56.18*** join/#asterisk [hC] (n=hardcore@S0106000e9b96114f.vf.shawcable.net)
06:56.38linlinwhat would a command be to just play a sound file to whoever is on the line
06:56.57N9URKPlayback(file_name)
06:57.20linlindo i need exten => before it?
06:57.35N9URKright
06:57.53N9URKexten =>99999,2,Playback(file_name)
06:58.01linlinwhat do the 9999 and 2 mean
06:58.35N9URKyou will need to learn how to configure extensions.conf
06:59.04linlinwhat would something like exten => s,9,Playback(custom/aa_1) do
06:59.05N9URKdo a google search for "asterisk tfot" and I think ch04 and ch05 talk about it
06:59.15N9URKthat would work
06:59.22*** join/#asterisk nagl (n=nagl@rih.zid-nw.wu-wien.ac.at)
07:01.38linlinim just having a problem setting up my ivr, its like, its trying to play the file, because the silence jasts for jsut about hte right amount of time, and i can can enter input to perform the tasks, it just wont play the file ive created for it to play
07:02.04N9URKwhat is the name and extension of the file?
07:02.30linlinname is aa_1.wav
07:02.46N9URKyou need to convert it to .gsm
07:03.04znoGbefore linlin asks his next question, the answer is "sox"
07:03.05linlini think it is in gsm its just named wav
07:03.20N9URKthen you need to rename it to .gsm, I think
07:03.46linlini used something like exten => 456,3,Record(/tmp/asterisk-recording:gsm) to record it
07:03.52linlinthat would encode it to gsm right?
07:03.57linlinit saved it as a wav though
07:04.21N9URKtry renaming it and seeing what happens
07:04.26linlinok
07:05.31tainted_where does asterisk store voicemail passwords?
07:05.38Qwelltainted_: voicemail.conf
07:05.42tainted_strange
07:05.51Qwellnot really
07:05.58camonzN9URK: i'm trying to play an mp3 file with asterisk, at first i got some problems with mpg123 and mpg321, but i now have the right version mpg123 .059r, the CLI console says it started the proccess of playing the mp3 file but i don't get any sound on the other end
07:06.00tainted_after user changed voicemail password asterisk removed the line from voicemail.conf context
07:06.12Qwelltainted_: let me guess...
07:06.18Qwelltainted_: last line in the conf file?
07:06.22tainted_yea!
07:06.34tainted_did i forget EOF or something
07:06.37Qwellyeah...
07:06.39N9URKhow are you dialing in?
07:06.40tainted_or to leave a line break
07:06.46QwellI thought that was fixed.  What version?
07:06.47camonzsip
07:06.51tainted_1.0.7
07:06.55N9URKwhat carrier?
07:06.55Qwellyeah, there you go
07:07.11N9URKSometimes the music doesn't go through sip connections, like Gizmo Project
07:07.12tainted_nice catch
07:07.14Qwelltainted_: always add a newline at the end of config files.  I know it affected vm, but I'm not sure what else
07:07.24camonzit's a direct connection in my home LAN
07:07.25Qwelland...
07:07.26tainted_i remember it affected something else too
07:07.26Qwellupgrade! :p
07:07.38linlinYES! Finally :)
07:07.49linlinthanlyou so very much
07:07.54N9URKGlad to help
07:08.03N9URKwhat did you do? change the name?
07:08.03Qwelllinlin: the new way to do a Record is filename.format, instead of filename:format
07:08.08camonzalthought, Playback() works all right
07:08.28linlinok, so thats why it recorded to .wav and not gsm like i thought i specified?
07:08.34Qwelllinlin: see above
07:08.37camonzi'm setting a reallty simple example of dialing into the server, and depending on the extensions play 3 different sound files
07:08.41QwellThat may help.
07:09.05*** join/#asterisk illljay (n=Illjay@c-67-171-86-175.hsd1.pa.comcast.net)
07:09.17N9URKcamonz, I think that you can only play mp3s for the music on hold
07:09.25N9URKplayback must be gsm
07:09.29tainted_Qwell how do i remove "press 3 for advanced options" from VoiceMailMain?
07:09.30QwellN9URK: not true
07:09.34N9URKok
07:09.36Qwelltainted_: app_voicemail.c
07:09.37N9URKthanks Qwell
07:09.38illljayHey .. I keep getting this error when I try and make outbound calls .. but incoming calls work just fine.. any ideas?:   Dec 29  WARNING[888]: chan_iax2.c:5566 socket_read:        -- Call rejected by 66.225.202.72: No authority found
07:09.46Qwellcamonz: If you want to play mp3's with Playback(), you need to install asterisk-addons
07:09.50illljayI have nufone.. anybody ever had this prob before?
07:09.51Qwellwhich includes format_mp3
07:10.00camonzi have it,
07:10.03dudesillljay - if it's not setup right
07:10.07znoGillljay: yes, are you trying to dial international?
07:10.12illljaynope
07:10.14loudor low credit.
07:10.25illljayjust national
07:10.48camonzQwell : but i'm trying to do it with MP3Player, i got the right ver of mpg123, and the CLI doesn't gives me any errors now, but i'm not getting any sound on the caller end
07:11.19N9URKCamonz > do you have the addons he mentioned?
07:11.25camonzyep
07:11.31N9URKhmmmm
07:11.32dudesMOH is cool
07:11.39tainted_Qwell there's not voicemail.conf setting?
07:11.40N9URKI am not sure
07:11.44Qwelltainted_: don't think so
07:11.51tainted_damn
07:12.41camonzi also seem to have an unexpected behaviour of the dialplan, when i set the dialing extension as s, the call doesn't get answered
07:13.02camonzi have to set it to _X. to be able to answer it, or set a ringing sound
07:13.27dudesthat may be cause the extension you're calling doesn't match ./w s
07:13.36dudeswhile _X. pattern matches
07:14.30N9URKHow can I play MusicOnHold while * is executing a php script to generate a voice file that takes about 10 seconds without having the music play for 10 seconds then then having there be silence while the file is generated?
07:15.00N9URKWaitMusicOnHold(10) just pushes everything back 10 seconds whilst it plays hold music
07:15.10camonzdudes: shouldn't the s extension be the extension that all calls begin with
07:15.20dudesno
07:15.24N9URKMusicOnHold() never seems to let the next steps happen
07:15.33*** join/#asterisk jalsot (n=tamas@abacus.eworldcom.hu)
07:15.44dudesif you call into a extension it has to be matches
07:16.03dudesif you want s,1 to be your execution point, you'll need to goto(context,s,1)
07:16.11QwellN9URK: startmusiconhold?
07:16.34*** join/#asterisk svenna_ (n=svenna@p548D3ED1.dip0.t-ipconnect.de)
07:17.48*** join/#asterisk tuxinator_linuxM (n=spabin@70-32-106-248.ontrca.adelphia.net)
07:17.50camonzdudes: i know, that's the default behaviour for a user defined extension, although from reading the examples on asteriskguru.com about extensions.conf the use the s extension as the startpoint of the dialplan
07:17.57N9URKQwell > I cannot find any docs on it
07:18.03N9URKdo you know where I can read more about startmusiconhold?
07:18.22dudescamonz - play with * for a bit and you'll understand what I said above
07:18.48camonzthanks :->
07:19.25dudesie, try just having your extension instead of _X. and it should go through
07:19.38dudesif you want that extension to goto s,1 do a goto
07:20.03*** join/#asterisk argos73 (i=1000@jason.argos.org)
07:20.40camonzactually what i want is that all incoming calls hear a ringing tone and then process them according to the extension dialed
07:20.55camonzthat's why i wanted to use the s extension thinking it would match all calls
07:22.27dudesnormally when I get a call I say have (several DIDs from wisc.) so I setup in my from-sip _262.,1,Goto(s,1) which then follows my menu
07:22.42QwellN9URK: show application startmusiconhold
07:23.03N9URKdebian*CLI> show application startmusiconhold
07:23.03N9URKYour application(s) is (are) not registered
07:23.17N9URKIs that the desired result?
07:23.53dudesNow if I want one of those DIDs goto my phone instead of menu I'd 262NXXNXXX,1,Dial(sip/mycontext)
07:24.10QwellN9URK: What version of asterisk?
07:24.16dudesI'm too tired to think right (that's two sentences I've f'ed up tonight)
07:24.29camonz:->
07:24.33N9URK1.0.7
07:24.38QwellN9URK: upgrade
07:24.49argos73yippee - my new pri works!  :)
07:24.56N9URKI am using what was in apt-get.   How difficult is it to upgrade?
07:25.12dudesyou compile it
07:25.14QwellN9URK: asterisk packages suck.  grab the stable tarball
07:25.15dudesit's pretty easy
07:25.40dudesit's just make && make install
07:25.44N9URKok
07:25.51dudesthen it's like meeting a really easy chick taking her home and yea
07:25.56N9URKwill it overwrite any files I need?
07:26.09QwellN9URK: nope, as long as you don't do `make samples`
07:26.28Qwellbackup /etc/asterisk/, and apt-get remove asterisk
07:26.46Qwellthen install from source, and copy it back
07:27.04dudesI haven't used stable is a very long time
07:27.13QwellI've never used stable, heh
07:27.27dudesI think early Gnudialer stuff was done on 1.0.7
07:27.36dudesCan't remeber (might have been 8)
07:27.38Qwellwell, maybe for a week when I started
07:27.40dudesThen we moved to head
07:28.08N9URKWhat is the stable version 1.2.1?
07:28.18QwellN9URK: that's the latest stable
07:28.28N9URKthanks
07:28.32dudeshead seems to meet our needs well enough, but Richard does some modding of the manager for stuff /w Puff.
07:28.49dudesPretty cool stuff though
07:29.48N9URKI am making now
07:29.58*** join/#asterisk mogorman (n=mogorman@user-24-236-84-48.knology.net)
07:30.12Qwellogorman!
07:30.19mogormanQwell!
07:30.27mogormanhows skinny
07:30.33Qwellgetting fat
07:30.39mogormanheh
07:30.40Qwelltoo much fast food
07:30.41dudesbeer will do that
07:31.08N9URKconfigure: error: termcap support not found
07:31.09N9URKmake: *** [editline/libedit.a] Error 1
07:31.14N9URKHow bad is this error here "configure: error: termcap support not found"
07:31.16QwellN9URK: You need curses
07:31.20mogormandnnanan install ncurses
07:31.35dudesN9URK - the wiki is your friend
07:31.38mogormani need a funny output if you dont have libiksemel
07:31.42dudesvoip-info.org asterisk debian
07:32.04Qwellmogorman: "Click _here_ for more information."
07:32.08dudesthat'll answer all your questions 10 times quick ...
07:32.27mogormanheh im thinking more like "refer to previous message" as the first error message
07:32.32Qwellheh
07:33.05Qwellpull ownership verification questions from old games, like Leisure Suit Larry
07:33.30mogormanlol
07:33.32mogormani loved those
07:33.35Qwellheh
07:33.38mogormani had a doctor game
07:33.44mogormanwhere you performed surgery
07:33.49mogormanbut every now and then you got paged
07:33.58*** join/#asterisk tartar (n=tartar@CPE0004e27b716e-CM014370001917.cpe.net.cable.rogers.com)
07:34.00mogormanand had to type in a number from this secret decoder ring
07:34.03Qwellawesome
07:34.06mogormanlost decoder ring after a month
07:34.08mogormanmany died
07:34.14dudesForget that! Playing doctor is for kids! try playing gynecologist.
07:34.56*** part/#asterisk tartar (n=tartar@CPE0004e27b716e-CM014370001917.cpe.net.cable.rogers.com)
07:35.03Qwellhaha
07:35.04dudesI wouldn't deal /w that either
07:35.16Qwellscared him off
07:35.16mogormani dont think you can be a gyno
07:35.21mogormanand turn those people away
07:35.21dudesBut I suppose I dont really know what that is either
07:35.33dudesI know, but I don't know
07:35.36Qwelldudes: be glad you don't
07:35.46dudesI'm sure I am
07:35.49mogormanameb
07:35.52mogormanerr amen
07:36.01mogormani imagine most days its as fun as a proctologist
07:36.06N9URKdo I need curses or ncurses?
07:36.12mogormanncurses
07:36.17dudesdude, wiki is your friend
07:36.33N9URKthe fucking wiki page didn't say, dude
07:36.34mogormanN9URK,  dont mind them
07:36.40N9URKoh well
07:36.42dudesbullshit
07:36.45mogormanwe or at least me is here to help
07:36.45N9URKit can't say everything
07:36.45dudesyou didn't f'n look then
07:36.58dudeshttp://www.google.com/search?hs=d5W&hl=en&lr=&safe=off&client=firefox&rls=org.mozilla%3Aen-US%3Aunofficial&q=asterisk+debian+site%3Avoip-info.org&btnG=Search
07:37.02dudesfirst result
07:37.06Corydon76-home"John Rocker, your proctologist called.  They just found your head."
07:37.22dudesso there, it does say everything
07:37.22N9URKhttp://www.voip-info.org/wiki-Asterisk+Linux+Debian
07:37.32N9URKI do see down there
07:37.35N9URKgetting tired I guess
07:37.59dudesI'm getting drunk and tired
07:38.01mogormanheh Corydon-w
07:38.17QwellCorydon76-home: why only the 76 at home?
07:38.19N9URKwhy didn't it emerge when I did apt-get install asterisk the other day?
07:38.28QwellN9URK: because debian is silly
07:38.35Corydon76-homemogorman: only the people who are from around here know who that is
07:38.47*** join/#asterisk [Airwolf] (n=airwolf@attilla.nl)
07:38.59mogormannah john rocker is pretty famous idiot
07:38.59Corydon76-homeQwell: it's a leftover from when I only IRCed on Undernet
07:39.01dudesI've always wanted to spoof CID and call people from a movie place and tell them "In her ass 22" is late or something
07:39.09Corydon76-homeQwell: 9 chars is the max len on Undernet
07:39.15QwellI see
07:39.20N9URKI already have ncurses (I thought so)
07:39.24mogormandudes you can do that anyway
07:39.27Corydon76-homeand when I registered my channels, it was at work
07:39.30dudesI can
07:39.31QwellN9URK: need the dev package too
07:39.35mogormani mean who is gonna know your number isnt blockbuster
07:39.35Corydon76-homeHence, I can't change the nick
07:39.36dudesJust haven't gotten around to it yet
07:40.04dudesN9URK - apt-get the list ont he wiki, cause it's your f'n friend
07:40.41Corydon76-homeI suppose I could change my nick and link in to the old one, but the nickname is coded into a bunch of scripts, and I don't think about changing them when I'm there
07:41.42Corydon76-homeBesides, the only reason I ever added the 76 was to differentiate myself from others who might choose the same moniker
07:42.08Corydon76-homeWell, that and to answer a third of the usual ASL question
07:42.21Qwell2/3, I'd say
07:42.32Qwellthe L is quite obvious also
07:42.42Corydon76-homeOh, really?
07:42.46Qwellhome?
07:42.52Corydon76-homeI'm not in Indiana
07:43.00Qwellwell, "home" is a location :p
07:43.13Corydon76-homeEverywhere is home to somebody
07:43.19Corydon76-homeDoesn't really answer it
07:43.22dudeshome is where you come home to a nagging .....
07:43.32N9URKare you from IN Corydon?
07:43.36Qwelldudes: sure, if you're married
07:43.36Corydon76-homeNo
07:43.49dudesQwell - you make a valid point
07:43.56Qwell...or living in your mothers basement
07:43.57Corydon76-homeQwell: I might as well be married, but he doesn't nag all that much
07:43.59N9URKok, I thougth that was what you might have been implying
07:44.05QwellCorydon76-home: lucky you
07:44.08dudesI don't live in my moms basement
07:44.21dudesI have no one living here (thankfully).
07:44.36Corydon76-homeQwell: I also own the house and pay the majority of the bills
07:45.10QwellCorydon76-home: wives...heh
07:45.17Qwell(or...whatever)
07:45.22Corydon76-homeHusbands
07:45.25Qwellclose enough
07:45.47dudesDo you come home to a hot meal?  if not I'd be pissed.
07:45.52Corydon76-homeonly for government work
07:46.04Corydon76-homeI used to
07:46.11Corydon76-homeand sometimes still do
07:46.39dudesI suppose it depends on their cooking skills
07:46.40Corydon76-homebut we're on different diets now.  He can take 6 ounces at a time, and I can eat whatever I want
07:46.55Qwellbypass?
07:47.04Corydon76-homeNo, banding
07:47.11Corydon76-homereversible
07:47.19Qwellhmm
07:47.31Qwellswm_: good afternoon
07:47.37Corydon76-homeAdjustable in the doctor's office, too
07:47.45Qwellneat..new procedure?
07:47.50swm_noon? Uhh it's pitch black .. sun went down 4 hours ago
07:47.59dudesit's 1:49AM here
07:48.07Corydon76-homeLapBand... it's been the standard for weight reduction in Europe for at least a decade
07:48.28Corydon76-homeAmerica favors the more drastic and more dangerous surgeries
07:48.29swm_It's 11:55 AM here
07:48.31QwellCorydon76-home: seem to work pretty well?
07:48.41dudesStaples works though
07:48.44dudessometimes
07:48.47N9URKYE$$$$$$$$$$$, they make the surgeon more money
07:48.58Corydon76-homeQwell: he's down 50 lbs so far (since September) and has another 70 to go...
07:49.11Qwellwow, so, yeah, similar to the other
07:49.42Corydon76-homeThe weight reduction between bypass and banding is similar overall... but banding gives a more gradual reduction
07:50.02dudesMy ex's mom was a big women and she got staples and for being 50 some ... she not bad
07:50.03Corydon76-homebypass gives a huge drop in the first 2 months, then it levels off
07:50.23QwellCorydon76-home: and more importantly, sticks with you forever...that's a huge downside
07:50.26Corydon76-homeplus banding is much less risky
07:50.57Corydon76-homePeople who get the bypass have a huge risk of malnourishment
07:51.17Corydon76-homebecause that's exactly how bypass works... limit the amount able to be absorbed
07:51.17N9URKA friend of mine had the bypass ca. 18 mos ago
07:51.34Corydon76-homebanding, otoh, makes you feel full with less food
07:51.38N9URKthey were very very carefull to guarantee she got enough protein
07:52.26coppiceloosing weight is the easy part. staying slim it the real test
07:52.30Corydon76-homeYep, same here... for his 6 oz, the doctor told him to prioritize on protein.... because after that amount, nothing more is going to stay down
07:53.41Corydon76-homeThe banding might as well be permanent... once he gets to his target weight, the doc extracts saline out of the adjustable sac, and the constriction is removed
07:54.00Corydon76-homeHe can have it readjusted to whatever size he needs to keep his weight down
07:54.13illljayAnybody know what in my config would cause every single call wether internal or external ALWAYS drop after 6 seconds of being connected could be?
07:54.16Qwellbut the device is still there forever?
07:54.21illljayeven just dialing a local extension.. =\
07:54.40Corydon76-homeThe device can stay put forever... but if complications ever arose, it could be removed
07:54.43dudesillljay - timeout
07:54.47Qwellahh
07:54.53coppicethis is kinda the theory, but fat people get fat again the moment they relax
07:55.10QwellThat seems to be ideal...wonder why the bypass is more popular here.
07:55.16illljayhmm what i thought
07:55.22illljaycant seem to find where though
07:55.24Corydon76-homePlus he's got more energy than ever before, and he's exercising on a regular basis
07:55.27illljaylike timeout on what part you know
07:55.31Qwellillljay: How are you dialing?
07:55.38coppicethat's why eating less, and going through the short term pain that causes is the answer. you adapt to the new you as you go
07:55.58dudescoppice - agreed
07:56.09dudesJust get off your ass and move and ...
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07:56.21Corydon76-homeQwell: once the insurance companies start learning that LapBand is cheaper, and can be done outpatient, same day as surgery, they're gradually beginning to favor it in the US
07:56.25Qwellcoppice, dudes: It's really not that easy for some
07:56.46Corydon76-homeWith bypass, you're never in the hospital shorter than a week... and those bills add up
07:56.53Qwellyeah, I bet
07:57.01coppiceif you ask the dietitians who aren't making healthy profits from slimming, they will tell you to just eat like a person who is the weight you want to be.
07:57.05QwellCorydon76-home: how soon until you can go back to work and such?
07:57.23Corydon76-homeHe was back to work with only a week off
07:57.31dudesIt's probably not, cause I know I like my whisky at night and my smokes.  But eating is something I do once or twice a day and it's normally healthy
07:57.31coppiceQwell: nothing worthwhile is easy. there is no easy solution that works long term
07:58.18Corydon76-homecoppice: true enough.  You can defeat bypass, just as you can defeat banding... they're just tools, to make the process easier
07:58.23coppiceQwell: at least with the short term pain approach you get the nastiest part out of the way first
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07:58.54coppiceCorydon76-home: and quite profitable too
07:59.05Qwellcoppice: I once cooked for a guy who was > 500lbs...  I'd LOVE to see somebody try to seriously tell him to eat better, and get more exercise.
07:59.06dudesI'm hungry all day long.  An hour after i finish eating I could go back and load up again.  But I just hate growing a belly.  So instead I limit myself regardless of my urge to wanna goto my kitchen and make something tasty.
07:59.40coppiceQwell: nothing will ever get such a person slim *and* keep him that way. its not their nature
07:59.44Corydon76-homecoppice: the doc who did his surgery is the only one in town who does that surgery.  He has a waiting list... He's doing multiple of these surgeries a day
07:59.53Qwellcoppice: he was thin all his life.
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08:00.31coppiceQwell: >500lb and thin?
08:00.33Corydon76-homeMy husband got to the hospital before 6 am to have his surgery...
08:00.45Qwellcoppice: well, obviously, thin before he got large.
08:01.11FuriousGeorge~lastspoke FuriousGeorge
08:01.14Corydon76-homeTarget weight is 250... still overweight, but not dangerously so
08:01.23coppiceQwell: so, he let himeself go a bit :-) if he spent a long time thin, there is hope for him
08:01.30FuriousGeorgehey all
08:01.40*** join/#asterisk linux2unix (n=linux2un@ip68-106-117-229.dc.dc.cox.net)
08:02.36coppiceMy sister weighs 900lb, and isn't fat at all
08:02.38coppiceShe isn't 1lb overweight
08:02.40coppiceBut tall, terribly tall
08:02.56dudes900!
08:02.57Qwelluhh
08:02.59N9URKQwell: make install here
08:03.07FuriousGeorge19'4" is average for that weight i believe
08:03.12*** part/#asterisk linux2unix (n=linux2un@ip68-106-117-229.dc.dc.cox.net)
08:03.18dudesthat's a huge bitch
08:03.45Corydon76-homeYour sister's an elephant?
08:04.06coppicedudes: duh! that's a noel cowerd lyric
08:04.07dudesI'm not trying to be mean ...
08:04.37dudesI heard that somewhere can't think where though
08:04.48FuriousGeorgebut you are creating a hostile environment for women in IT, or so i read :)
08:05.03coppicehungry women, hungry women, I feed them and weep!
08:05.14dudesBitch get off my leg!
08:05.21dudesThat's a bad bitch, it's my leg!
08:05.48Corydon76-homeIs it a hostile environment if you've boinked all the unmarried women in your company?
08:05.55dudesNo kitty it's my potpie!
08:05.58dudeshaha
08:06.16Qwellhostile for them maybe
08:06.23Qwellfor you...not so much
08:06.37dudesNo unmarried chicks in my company
08:06.38Corydon76-homeWell, considering there's only one unmarried woman in my company...
08:07.10FuriousGeorgeCorydon76-home: i guess the hostility is relative to the type of sex :)
08:07.10Corydon76-homedudes: ah, so you HAVE boinked all the unmarried chicks in your company...
08:07.22FuriousGeorgeis spanking hostile?
08:07.25FuriousGeorgeanyway
08:07.27Corydon76-homeFuriousGeorge: which time?
08:07.34dudesHow could I since there isn't any
08:07.36FuriousGeorgethe time you spanked her
08:07.49Qwelldudes: you've done none, you've done them all
08:07.57Corydon76-homedudes: all and none at the same time
08:08.14dudesit's all or nothing ... none isn't all since there is none to begin with
08:09.10Corydon76-homeThe real question is, is 0 times 0 zero?  Or is zero times infinity still zero?
08:09.38FuriousGeorge~PredicateLogic
08:09.43camonz0 by inf is undeterminate, strangely enough since everything by 0 is 0
08:10.07camonzis the same with 1^infinity
08:10.15dudesNo the real question is, since all is greater than none ...
08:10.30Corydon76-homecamonz: How about zero to the zeroth power?
08:10.34camonz1
08:10.37dudesis all greater*
08:10.38QwellBeing or representing the entire or total number, amount, or quantity
08:10.45Corydon76-homecamonz: is it really?
08:10.48camonzyep
08:11.00Corydon76-homecamonz: even though zero to the first power is 0?
08:11.05Qwelldudes: and, if you want to take an informal interpretation, all is more than one
08:11.09camonzyep
08:11.17QwellBeing more than one: "Who all came to the party?"
08:11.29Corydon76-homeI thought zero to the zeroth power was also undefined
08:11.35camonznope
08:11.46camonzany number to the zeroth power is 1
08:11.50N9URK^=
08:11.55camonzprobably except infinity
08:11.56Qwellyeah, that always bothered me
08:11.59Corydon76-homebut 1 to any power is 1...
08:11.59N9URK0^0=1
08:12.26Corydon76-homeso 1 to the infinite power should still be 1, but it isn't
08:12.29camonzexcept 1^infinity wich is undeterminate
08:12.38sbingnerthat's stupid
08:12.41sbingnerI can tell you it's 1
08:12.43camonzyep, i didn't understand it either when i saw that calculus class
08:12.44FuriousGeorgejbot: no, predicate logic is a formal representation of mathematical logic which attempts to account for existence (or lack there of) in its veriables.  For instance, "everything is such that" is the opposite of "no things are such that".
08:12.45jbotFuriousGeorge: okay
08:12.55N9URKwhat about 1^(infinity +1) ?
08:13.03sbingnerlol
08:13.03Corydon76-homeN9URK: same
08:13.06camonzinfinity + any number is still infinity :->
08:13.13FuriousGeorgeshoot
08:13.15Qwellthat can't be so
08:13.21Qwellbecause then you've defined infinity
08:13.24Qwellwhich makes it finite
08:13.28sbingnerbut what about infinity^infinity! lol
08:13.34FuriousGeorgei hope there wasnt something important filed under "predicate" i just overwrote
08:13.43Corydon76-homeThen we get into the interesting part of infinity squared or infinity to the infinite power
08:13.50dudesI figure .. if (noWomen < all) { std::cout << "There are no bitches to have casual sex with ..." << std::endl; }
08:14.05Corydon76-homeThen we have to have differing levels of infinity
08:14.05camonzyou want a better one, try integration on complex variable, all the results are 2 * PI * Zo
08:14.36N9URKIf a chicken and a half could lay an egg in a half in a day in a half how many chickens would it take to lay a dozen eggs in 6 days?
08:14.39FuriousGeorgedudes: if something is a woman then it is such that it is not a bitch or you can have casual sex with it
08:14.43sbingnerCorydon: they don't actually say that do they? that's stuff I'd have said when I was 6
08:14.43Corydon76-homedudes: but 0 !< 0
08:14.46camonzZo, is the function to be integrated evaluated at the indeterminate point inside the integration region
08:15.27Corydon76-homesbingner: it's a higher level of mathematics
08:15.32dudesCorydon76-home - 0 isn't inequal to zero and since noWomen is less than all (since all being more than 1)
08:15.33QwellN9URK: 2
08:15.45sbingneryea, infinity squared... that's higher lol
08:15.51Corydon76-homedudes: but all is defined as 0 in your case
08:16.02Corydon76-homeso none is NOT less than all
08:16.08Corydon76-homenone is EQUAL to all
08:16.20Qwelltherefore, none = all = 1
08:16.28Corydon76-homeQwell: no, zero
08:16.34dudesCorydon76-home - How is all 0?
08:16.39Qwellthen all = none = 0
08:16.51sbingnerall is however many women there are I presume
08:16.52QwellI always did hate math
08:16.53Corydon76-homedudes: no unmarried women, so the total of unmarried women is 0
08:17.22Corydon76-homeTherefore, you had sex with all unmarried women
08:17.24sbingnerwhy can't you have casual sex with a married woman?
08:17.36sbingnerI mean, it's be awkward to have non-casual sex with them
08:17.59camonzlol
08:18.14dudesCorydon76-home - in order to have sex I need to stick it in some chick
08:18.19QwellPI IS EXACTLY THREE!
08:18.28FuriousGeorgestop liein
08:18.49dudesCorydon76-home  - therefore, you're wrong and being completely unidealistist
08:18.56mogormanahhhh qwell
08:19.00mogormanmy brain hurts
08:19.03Corydon76-homedudes: not true.  How could you do something zero times, if you had to do something at least once?
08:19.05N9URKPi are square.  the famous phrase that made the hillbilly take his kid out of school as everyone knows that pie are round
08:19.08sbingnerpi is 4, what are you talking about?
08:19.19sbingnerN9URK: lol
08:19.19dudesCorydon76-home - simply by not doing it at all
08:19.24Corydon76-homeIt's an inherent contradiction
08:19.34mogormanonly for really large values of  pi sbingner
08:19.54Corydon76-homedudes: what you're essentially there is that you don't believe that zero is a real number
08:19.59N9URKhttp://3.141592653589793238462643383279502884197169399375105820974944592.com/
08:20.13sbingnerhahahaha that's cool
08:20.18Corydon76-homeYou only have to do something for values greater than 0
08:20.41dudestherefore, since there are no unmarried women ... I couldn't possibly do any therefore all the women in my company being 0 ... would evaluate to false
08:20.58Corydon76-home0 = 0 is false?
08:21.12camonzi wonder how did they discover the value of PI
08:21.17Corydon76-homeStop the presses, the C compiler is wrong!
08:21.29Qwellcamonz: easy
08:21.33dudesCorydon76-home - 0 (false) 1 (true) a f'n boolean
08:21.45sbingnercamonz: uh, http://science.howstuffworks.com/pi.htm
08:21.49dudeswhat is hard about this ... all can't be 0 cause it'd f'n be FALSE
08:21.59dudesDo you f'n know anything
08:22.00Corydon76-homedudes: yes, but the statement is a comparison of two values
08:22.06Qwell0 = false, 0 == 0 = true
08:22.17camonzactually 0 means true
08:22.18camonz1 is false
08:22.21sbingnerall can be zero if all is nothing
08:22.23Corydon76-homeby having sex with all unmarried women, you are comparing 0 with 0, which is true
08:22.27camonzat least in rs-232
08:22.33sbingnerhow many are all the 15-headed redheaded whales?
08:22.34N9URKif current leads voltage by 45 degrees and the inductive reactance is 100 ohms, what is the value of the resistor?
08:22.53dudescamonz - We evaluate 0 false in C++
08:23.14dudesor a negative
08:23.19sbingnerN9URK: pi
08:23.36Corydon76-homeNow, at the same time that you have boinked all the unmarried women, you still have all of the unmarried women left to boink in order to have done all of them
08:23.44camonzstrangely i'm used to doing it the opposite in assembly, on C i just use simple bool types, without worrying too much on it's numeric value
08:23.51Corydon76-homeIsn't logic great?
08:24.05Qwelldudes: how many of them HAVEN'T you slept with?
08:24.08dudesCorydon76-home - I can't boinked a bitch IF she's not f'n there
08:24.11QwellAll of them?
08:24.11dudestherefore, it's false
08:24.14dudesall is false
08:24.23QwellOr would you still argue none?
08:24.24Corydon76-homeBut the comparison isn't false
08:24.29sbingnerCorydon: no, if there are no unmarried women you have no more umarried women to do, even though you never did any unmarried women and have done all the unmarried women
08:24.31Corydon76-homethe comparison is always true
08:25.01Corydon76-homesbingner: you actually have all AND none left to do
08:25.06Corydon76-homeat the same time
08:25.19sbingnerI'd agree with you but that wouldn't be as much fun
08:25.36dudesCorydon76-home - you can't have two statements the same
08:25.45QwellYou most certainly can
08:26.03Corydon76-homedudes: I know it sounds like a contradiction, which is why it's so much fun
08:26.19dudesif you're trying to find true/false and both are true therefore it's a fucked up stupid function that's going to make a shitty program
08:26.31dudesCorydon76-home - you're being a stupid fuck
08:26.35N9URKWhy is formula for Circumference the first derivative of the formula for Area?
08:26.42Corydon76-homeNo, I'm being a logical prick
08:26.43dudesSomething can't be true and false
08:26.58Corydon76-homeIt's not both true and false... it's always true
08:27.11Corydon76-home0 == 0
08:27.17dudesthere can't be unmarried people if there are none to start
08:27.21dudestherefore it's false
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08:27.29sbingnerdudes: its not true and false... it's true that you've done all of them, and it's true that you've done none of them it's not false that you did all of them or false that you did none of them
08:27.33N9URKon a serious note: what does this mean? "Make: warning:  Clock skew detected.  Your build may be incomplete."
08:27.33dudesand if you try and evaluate true you'll have a mislogi
08:27.34FuriousGeorgeN9URK: because its like 1D vs 3D
08:27.38FuriousGeorgei mean 2D
08:27.42FuriousGeorge1 vs 2
08:27.50Corydon76-homedudes: actually, you're doing the equation zero divided by zero
08:27.58Corydon76-homewhich is quite naturally undefined
08:28.17FuriousGeorgeis the derivitive of area, volume?
08:28.17coppicethe IEEE define it as NAN
08:28.21sbingnerbut that would be the percentage of them that you did... not wether or not you did all of them
08:28.34Corydon76-homeSince you can plug in any number as the result
08:28.37FuriousGeorgeof a circle i mean
08:28.44*** join/#asterisk Kumbang (n=kumbang@167.205.24.5)
08:28.51dudessbingner - we want true logic not it's true I did something I didn't do in the first place ... therefore it's false to keep logic write.  And more importantly in the "real world' if you don't get your dick wet you didn't, thereforem, it's false that you did cause you didn't.
08:28.51Corydon76-homecoppice: IEEE 754
08:29.00camonzactually if he did 0/0 woman, that tends to infinity
08:29.04QwellYou actually know the IEEE number?
08:29.11sbingnerdudes: you did none of them right? if all of them is none of them, then you did all of them
08:29.11N9URKthe first derivative of Volume  would be surface area?
08:29.25coppicecamonz: i think he would tend to get tired
08:29.25sbingnerbecause none == all
08:29.27FuriousGeorgedudes: i told you, this is best addressed by predicate logic http://en.wikipedia.org/wiki/First_order_predicate_calculus
08:29.35camonz:->
08:29.35Corydon76-homeQwell: yeah, it's the earlier standard for floating point
08:29.43FuriousGeorgetake your premise, and you coclusion, prove or disprove
08:29.46Corydon76-homeQwell: also IEEE 854
08:30.12Corydon76-homethough 754 had the definition for 0 / 0 being NAN
08:30.30dudesif (unMarriedWomen > 1 ) { std::cout << "You might have screwed them, WTG!!" << std::endll; } else { std::cout << "haha, you didn't have no bitches" << std::endl;
08:30.32camonzN9URK: the volume of a sphere is 1/3 * A*R^3 * 2* PI
08:30.37Qwellvoid sleptwith(int unmarriedwomen) {int none, all; count = unmarriedwomen; all = count; none = 0; return;}
08:30.45FuriousGeorgeN9URK: i dont know, thats what i was wondering
08:30.51Corydon76-homedudes: relax, they're just word games
08:31.01Corydon76-homedudes: you don't have to take it personally
08:31.07coppiceIEEE754 is the NANny of floating point systems
08:31.21dudesWe could write a class to define unmarried women
08:31.25dudesthen evaluate it
08:31.26N9URKFG I will look in one minutes
08:31.40Qwelldudes: want me to?
08:31.52sbingnerdudes: if (screwdunmarried_women >= total_unmarried_women) print 'I screwd em all'; if (screwdunmarried_women >= 0) print 'I didnt screw any of them';
08:31.52N9URKAfter upgrading * I now am getting this error
08:31.53N9URKdebian:/var/run# asterisk
08:31.53N9URKdebian:/var/run# asterisk -r
08:31.53N9URKUnable to connect to remote asterisk (does /var/run/asterisk.ctl exist?)
08:31.58sbingnerer
08:32.02sbingnerdudes: if (screwdunmarried_women >= total_unmarried_women) print 'I screwd em all'; if (screwdunmarried_women <= 0) print 'I didnt screw any of them';
08:32.05QwellN9URK: asterisk -cvvvvv, see why it isn't loading
08:32.11FuriousGeorgeN9URK: readme.UDEV?
08:32.15N9URKthanks
08:32.20Corydon76-homeQwell: you don't know hell until you've coded multiplication in assembly using only shift, add, and subtract
08:32.28sbingnerthe two tests are independant
08:32.31Qwellheh
08:32.32sbingnerand could have been == yea yea yea
08:32.33Corydon76-homeOh, and division in assembly
08:32.38Kumbangduh, my * cannot connect to telco after replacing E100P with TE110P
08:32.53Corydon76-homedivision in assembly is actually more difficult with only shift and subtract
08:32.53coppiceCorydon76-home: software multiply is trivial
08:33.00N9URK[chan_modem.so]Oct 15 03:36:03 WARNING[17694]: loader.c:325 __load_resource: /usr/lib/asterisk/modules/chan_modem.so: undefined symbol: ast_pthread_create
08:33.00N9URKOct 15 03:36:03 WARNING[17694]: loader.c:499 load_modules: Loading module chan_modem.so failed!
08:33.01N9URK[chan_modem.so]Oct 15 03:36:03 WARNING[17694]: loader.c:325 __load_resource: /usr/lib/asterisk/modules/chan_modem.so: undefined symbol: ast_pthread_create
08:33.06dudessbingner - a class appending to a vector would cut down on ifs  =0
08:33.21sbingnerheh
08:33.26Corydon76-homecoppice: yeah, multiply isn't all that bad
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08:33.43Corydon76-homecoppice: division is annoying though, with all the tests and branches
08:34.04camonzdivision in multiples of 2 is easy though
08:34.10sbingnerN9URK: what does that have to do with screwing all of the no unmarried women?
08:34.12camonzjust right shift
08:34.15dudesI try to code /w real logic (meaning either it's true or false) ... and since a agent can't be waiting if he never was in the first place ...
08:34.20Corydon76-homeespecially when you do 32 bit division on an 8 bit processor
08:34.23FuriousGeorgeN9URK: comment needing in modules.conf
08:34.24N9URKIt won't let me screw anyone
08:34.25dudestherefore, he's not waiting so it's false
08:34.26FuriousGeorgeits obvious
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08:34.41FuriousGeorgeonce you open
08:35.03coppiceCorydon76-home: division is always bad for speed, but not terribly hard. once you can divide, you can also square root with minimal additional code, which is fun
08:35.07QwellN9URK: rm /usr/lib/asterisk/modules/chan_modem.so
08:35.07N9URKoh boy
08:35.20dudesN9URK - no
08:35.27sbingneroh yea chan_modem is depreciated isn't it
08:35.34Qwellsbingner: along with many others
08:35.34dudesjust noload => chan_modem.so in modules.so
08:35.35N9URKnow I am getting more errors
08:35.35Corydon76-homeDEPRECATED
08:35.43Corydon76-homedepreciation is something different
08:35.45QwellN9URK: I'd actually remove that whole dir, and run the make install again
08:35.50N9URKI commented the chan_modem.so line
08:35.55sbingnerlol
08:35.56N9URKis that good enough
08:36.03Qwelldepreciation is loss of value - deprecation is a strong suggestion against doing something
08:36.05N9URKremove what whole dir?
08:36.07QwellN9URK: not really
08:36.11Qwellyeah, the whole modules dir
08:36.17sbingnerCorydon: I actually never realized they were spelled differently... I suck
08:36.22Qwellthen make install again
08:36.28sbingnerthat's gonna be a bitch to fix in my head
08:36.33Corydon76-homesbingner: pronounced differently, too
08:36.43sbingnerI've never heard anybody say it :(
08:36.53sbingnerde-pre-cated?
08:36.53Qwellsbingner: It's pronounced like it's spelled
08:36.58Qwellpretty much
08:37.06Qwellwell, no
08:37.10Corydon76-homedeh PREE she eight ed... DEP reh kate ed
08:37.15Qwellyeah
08:37.18Qwellnot a strong "pre"
08:37.30N9URKcan i jsut make install or do I need to make as well?
08:37.41QwellN9URK: just make install
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08:37.50sbingnerI sleep now, thanks for the english lesson :)
08:37.54sbingnerheh
08:38.03N9URKmake installing now
08:38.05N9URKthanks
08:38.10Corydon76-homepronunciations always help
08:38.21dudesN9URK - if you get it again take my suggestion
08:38.35QwellI also absolutely love how "pronounce" and "pronunciation" is spelled differently
08:38.54QwellYou can't help but like the English language
08:38.54Corydon76-homeI'm also going to bed... so much for my husband's admonishment not to come to bed late
08:39.07sbingnerCorydon: my god you're a woman?
08:39.11sbingnerno offense
08:39.12N9URKFG  Surface area of a sphere is the first derivative of the volume of a sphere
08:39.18camonzQwell: being a non native english speaker i agree
08:39.20kippihi
08:39.22sbingnerbut like, wow heh
08:39.22Corydon76-homesbingner: No, I'm not... no offense...
08:39.28sbingneraah ok
08:39.28N9URKv=(4/3)pi ^3
08:39.35camonzbut actually the most interesting things of english come from old french
08:39.42camonzbureau by example
08:39.44N9URKs=4pi^r2
08:39.50Corydon76-homeAnd don't forget 355 / 113
08:40.03N9URKmake that v=(4/3)pi r ^3
08:40.19drumkillawow, volume of a sphere
08:40.24drumkillamy head hurts!
08:40.25coppicepi. pi. hey, there are mince pies in the lounge :-)
08:40.26drumkilla:-p
08:41.03kippiare Digium in this week?
08:41.08Corydon76-homecoppice: gotta love doing calculations with pi with only integer math... ;-)
08:41.20drumkillakippi: yes
08:41.40coppiceFP is only simulated through integer maths and some justification
08:42.40Corydon76-homeWell, there's also fixed point math
08:43.14*** join/#asterisk af_ (n=af@ip-142-84.sn1.eutelia.it)
08:43.40*** join/#asterisk lme (n=lme@gw-r-ifc.infoclip.fr)
08:44.08*** join/#asterisk |dennis| (i=dennis@200.32.215.84)
08:44.30Corydon76-homeThough yes, fixed point might as well be integers...
08:44.43coppicewhich is integer maths with human justification in advance
08:45.42Corydon76-homeAnyway, I'm off to bed now
08:45.48N9URKgood night corydon
08:45.57Corydon76-homeNight
08:47.03*** part/#asterisk lme (n=lme@gw-r-ifc.infoclip.fr)
08:51.00camonzi'm checking out as well
08:51.10camonznice talking to you all :->, cya tomorrow
08:51.11N9URKgood night
08:51.15N9URKadios
08:51.18camonzchao
08:51.20camonz:->
08:53.48N9URKmake[1]: warning:  Clock skew detected.  Your build may be incomplete.
08:53.50N9URKwhat does this mean? make[1]: warning:  Clock skew detected.  Your build may be incomplete.
08:54.32dudesclock skew
08:54.47coppiceit means some files have a date in the future
08:54.56N9URKoh ok
08:54.58dudesI was getting to that
08:55.13dudesWe compiled a * from the past before
08:55.20N9URKyes
08:55.31N9URKfrom the past that we have not entered yet
08:55.39dudesand my friend said, I didn't get any warnings on load ... and I told him it's cause they haven't been invented yet
08:55.44N9URK2006 is the past, we jsut don't know it yet
08:56.07dudessicne you can't be in 06' yet it is in the past
08:56.09coppicewe can predict it with high likelihood, though
08:56.38dudesHopefully 06' will be a good year (peaceful at least)
08:57.02coppicethat would be unique
08:57.07dudesBut I have a feeling there will be more violence in the middle east
08:57.12dudesand possibly asia
08:57.21N9URKYou know what is hard to believe though?
08:57.43N9URKShrub hasn't even served a whole year of his second term yet
08:58.10coppicei think that explains dudes comments
08:58.30*** join/#asterisk brettnem (n=brettnem@72.29.102.158)
08:58.37dudesI hope for peace
08:58.48dudesBut to be honest, I don't see why we can't all just get along
08:58.51N9URKIt seems like it has been ages since the elections of 04
08:58.58N9URKI hope it is peacefull as well
08:59.07dudesbut I do like Bush
08:59.28coppicethen you apparently don't hope for peace
09:00.04N9URKyou all of seen the signs that say "Know Jesus - Know peace"?
09:00.07*** join/#asterisk psk (n=psk@golia.caltanet.it)
09:00.11dudesthen you like dictators who torture your fellow man
09:00.15N9URK"KNow Bush - No Peace"
09:00.20*** join/#asterisk olle (n=olle@apollo.webway.se)
09:00.23N9URK"No Bush - Know Peace"
09:00.43dudesNo Bush == no peace sounds right too me
09:00.50dudesand where are you two from?
09:00.56*** join/#asterisk _Madar (n=tophe@219-84-129-163-adsl-tpe.static.so-net.net.tw)
09:01.08dudesI'm from minnesota
09:01.10dudesUS
09:01.17N9URKNC
09:01.30dudesA dumb demi, huh
09:01.59N9URKnope
09:02.05dudesIf you hate bush, you do have the right to decend
09:02.15N9URKI am not a Republicrat
09:02.43dudesYou do know our godfathers were Republicans right
09:02.56N9URKlooks like it has made ok this time
09:03.45dudesLet's say
09:03.49FuriousGeorgeN9URK: did you find that line in modules.conf?  are u up and running?
09:04.14N9URKits running now
09:04.23N9URKdid you see my message to you earlier FG?
09:04.25dudes"Bush was raping and torturing your family" ... You'd want Tony Blair to come and kick his ass"
09:04.32FuriousGeorgewhat about the derivative of the forumla for the area of a circle?
09:04.37N9URK03:38
09:05.05N9URKLook at the messages starting 03:38
09:05.05dudesI know I would ... But I suppose I have the right to decend if I please.  Something some don't have that we do.
09:06.04dudesSome don't have the right we do *
09:06.54dudesIt's not like we invated a country for profit ... cause if we did we would be filling our tanks /w a f'n five spot
09:07.16N9URKyou may not be profiting, but Halliburton sure is
09:07.36*** join/#asterisk TonyM (n=TonyM@adsl-solo-80-168-225-132.claranet.co.uk)
09:07.53dudeshaha
09:07.59dudesyou're an idiot
09:08.07dudeswhich explains why you can't compile *
09:08.09coppiceyou might like to ask the women in iraq about freedom now they've had practically all signs of sexual equality removed
09:08.15dudescause your're too stupid to read
09:08.50N9URKFurious George: are you still around?
09:08.56dudescoppice - and you're smoking what
09:09.18coppicedudes: you don't get around much, do you
09:09.37N9URKthose MN winters freeze the brain
09:10.59dudeshaha
09:11.08dudesthat's funny
09:12.17N9URKnow that it is up and running, I am not getting any of the sound files to play.  The sound files were in /usr/share/asterisk/sounds.  Do they need to move to /var/lib/asterisk/sounds?
09:12.37dudesit's cause you're not smart enought to read
09:12.44dudesand you want everyone todo everything for you
09:12.55dudesa typical democrat .. oh poor me and my f'n life
09:13.02N9URKI have a better idea
09:13.09dudesI'm a loser lowlife too lazy to get off my ass and do something
09:13.14dudesI know
09:13.14N9URKFixed that problem
09:13.31iDunnocould just symlink it :P
09:13.35*** join/#asterisk razu_ (n=razu@tln-kontor.norby.ee)
09:13.39N9URKok thanks
09:13.41dudesThen why say something 10 seconds prior to the contrary
09:13.44N9URKI love the iggy bin
09:14.07dudesOh, cause your a anti democracy clearly
09:14.15iDunnowtf?!
09:14.39N9URKthe iggy bin fixed my dudes problem
09:14.42*** join/#asterisk brettnem (n=brettnem@72.29.102.158)
09:14.44iDunnoSo, yesterday there was an argument about... what was it... milk. That was it.
09:14.47N9URKThank you for the help
09:15.01N9URKthat is great
09:15.02dudesget a life and move somewhere where you get tortured
09:15.11iDunnoand today we've got a democracy/anti-democracy thing going on.
09:15.29FuriousGeorgeN9URK: so apparently i guessed right
09:15.41dudesCause you do have the right to decend ... so do it
09:15.45N9URKyeah, you were right (if you said sa was v')
09:15.50N9URKI think that was what you said
09:15.58iDunnodecend? what? down the mountain?
09:15.59N9URKtoo lazy to look back ;)
09:16.21FuriousGeorgeN9URK: i was intending to extend upon what you said vis a vi the formuli for are and circumference
09:16.33N9URKyeah
09:16.37dudesdescend rather
09:16.39FuriousGeorgewhichever i said :)  i already forgot which way we were deriving
09:16.50dudesmeaning --- move elsewhere
09:16.51N9URKVolume = (4/3) pi r^3
09:17.03N9URKS.A. = 4 pi r ^2
09:17.08iDunnohmm - spherical.
09:17.16*** join/#asterisk znoG_ (n=gs@33-138-114-200.fibertel.com.ar)
09:17.19N9URKtherefore S.A. = V'
09:17.28iDunnoerm.
09:17.31FuriousGeorgeits been about 8 years and behvrl sci degree b/t my last real math class in high school, i shoulda done something math related
09:17.34FuriousGeorgeanyway
09:17.36iDunnoin lala land, maybe ;)
09:17.53N9URKmath does help
09:18.04dudesMy friend could own you /w math
09:18.11FuriousGeorgeN9URK: what do you think is the significance of the derivative of the volume?
09:19.05N9URKYou do remember what a derivative is?  (since you ref. your behvrl sci deg I assume you may not)
09:19.19FuriousGeorgei guess if a sphere grows at a constant parabolic rate you could use the derivative of volume to calculate size?
09:19.25N9URKyeah
09:19.38FuriousGeorgeN9URK: pulled that one out of my butt too
09:19.51FuriousGeorgewonder if its accurate at all. i shoulda done something with math
09:19.52N9URKI never got a real good answer from my calc teachers
09:20.06dudesgo figure, pussy democrat anti-freedom fags ... Pussies
09:20.23coppicei love a finely reasoned argument
09:20.28iDunnoit's too early for this shit.
09:20.38dudesIt's only 3:22AM
09:20.43*** join/#asterisk cjk (n=cjk@80.92.64.103)
09:20.44iDunnoespecially when it's entirely one sided, dudes against the world ;)
09:20.50N9URKif you have f(x) = x^2 the first derivative of that function is f'(x) = 2x
09:20.51FuriousGeorgedudes: you obviously have no ocean
09:21.00iDunnodudes: it's 9:20 here, and I've only been in work for 30 mins or so.
09:21.09N9URKif you have f(x) = x^3 the first derivative of that function is f'(x) = 3x^2
09:21.36dudesI'm at my work station.  It's just not 10AM yet for work
09:21.55dudesand I still got a few shots of BV left
09:22.25*** join/#asterisk BladeRunner05 (n=feelme@adsl-14-214.38-151.net24.it)
09:22.56N9URKI think I am up and going.  THanks for all of your help
09:23.07N9URKthis is only my 4th day of using *
09:23.13N9URKso your help is much much appreciated
09:23.27iDunnoN9URK: why are we doing calculus in #asterisk?
09:23.39dudesCause their ignoring me
09:23.47N9URKresults from a discussion from earlier and FG's question to me from earlier
09:23.51iDunnodudes: what hte hell is BV?
09:24.00dudesWhisky
09:24.12FuriousGeorgeN9URK: later on
09:24.24N9URKI'll be around a few more mins
09:24.46dudesthen you'll find the 9milli
09:25.02iDunnodudes: ahh - hmm. not a real whisky, I assume... not a nice single malt scotch...
09:25.11FuriousGeorgeN9URK: so will i but i wont be paying attn again, im always here and almost always idle.
09:25.15FuriousGeorge:)
09:25.16dudesit's Canadian
09:25.17FuriousGeorgelater
09:25.18FuriousGeorgeall
09:25.21dudesBut it's smooth
09:25.22N9URKGood night
09:25.43iDunnoyeah - that's not real, whisky is scottish and irish, dammit ;)
09:25.51iDunnothe rest of them are poor imitations ;)
09:25.53dudesI'd agree
09:26.06dudesBut they really don't have much decent shit around here
09:26.22dudesjack and bc are the best
09:26.27*** join/#asterisk uchman (i=pean4661@hamberg.it.uu.se)
09:26.29dudesthough I like the taste of Jack
09:26.37dudesI should get a liter tomorrow
09:27.09dudescan't dawg jack
09:27.12dudesjack is good
09:27.50*** join/#asterisk uchman (i=pean4661@hamberg.it.uu.se)
09:28.03dudesand BV is tasty ... Though I do enjoy R&R (and it's cheaper) I do like BV more.
09:28.49uchmanI have a ivr-meny, how do i configure the extension to timeout if there is no input for 10s and goto blah?
09:29.15dudesyou'd need to show application [app]
09:29.25dudesyou can normally set a timeout based on this app or that app
09:29.54dudesor just response timeout
09:30.17dudesIn the event you simply use background/playback or whatnot
09:31.09dudesexten => s,5,Set(TIMEOUT(response)=90) --- 90 being whatever you want
09:31.29uchmanexten =>   400,1,Answer
09:31.29uchmanexten =>   400,2,SetMusicOnHold(default)
09:31.29uchmanexten =>   400,3,DigitTimeout,5
09:31.29uchmanexten =>   400,4,ResponseTimeout,10
09:31.29uchmanexten =>   400,5,Background(new/ny)
09:31.53dudesexten => s,1,Set(TIMEOUT(digit)=10|TIMEOUT(response)=15)
09:32.25uchmanWhat's the difference?
09:33.03dudesdigit ... response
09:33.23dudesI hit a _____ ... and I can wait ________
09:34.00dudesso if I don't hit a digit in 10 ... it timesout cause there was no response
09:34.37iDunnoin 15.
09:34.46dudesWhatever you set it to
09:34.49iDunnooh, no, you're right ;)
09:34.55dudesI know =)
09:35.05iDunnosorry - read what you'd plonked wrong - eye's aren't quite caffienated yet :)
09:35.18dudesI know that feeling
09:35.20dudeshehe
09:35.35dudesI stay away from irc for half or more
09:35.43dudesgotten wait for the coffee to kick me
09:35.52dudesthen it's on like donkey kong
09:36.02dudesor I clean my house instead of working
09:36.55uchmandudes, So i can use 400,3,Set(TIMEOUT(digit)=10|TIMEOUT(response)=15)?
09:37.51iDunnoheh
09:38.36uchmanhaha. Nice. No Application.
09:38.38uchmanWhatever.
09:40.20uchmanDec 29 10:39:43 WARNING[5615]: pbx.c:1952 ast_pbx_run: Timeout, but no rule 't' in context 'bleh'
09:41.07dudesuchman - what * are you using
09:41.29uchmanAsterisk 1.0.8-BRIstuffed-0.2.0-RC8h
09:41.55dudesSet works /w head as I know so if it doesn't work
09:43.53N9URKHow can I get * to play Music on hold while it is performing a eten=> system(blah blah blah) ?
09:44.30*** join/#asterisk oej (n=oej@apollo.webway.se)
09:45.02iDunnothat's what Background is for.
09:45.20Kumbangwhat should be the problem with these
09:45.26KumbangDec 28 12:26:56 WARNING[25784] chan_zap.c: No D-channels available!  Using Prima
09:45.26Kumbangry on channel anyway 47!
09:45.26KumbangDec 28 12:26:56 WARNING[25784] chan_zap.c: PRI Error: We think we're the CPE, bu
09:45.39Kumbangt they think they're the CPE too.
09:45.51iDunnoyou've got no D-channels, must be bad.
09:46.07dudesconsidering they're not configured ...
09:46.12Kumbangbad card?
09:46.18dudesbad config
09:46.56Kumbangi think the configs should fine
09:47.08Kumbangsignalling=pri_cpe
09:47.14dudesif there is no dchan it's probably a config issue
09:47.34dudesand use pastebin.ca or your httpd instead of flooding chan
09:47.43N9URKIdunno, I am wanting it to play the files I have in the mohmp3 directory while it is performing the next commands, am I SOL on that one?
09:48.15Kumbangok which part of config
09:48.25Kumbangzaptel.conf , dchan=47
09:49.03Kumbangzapata.conf , signalling=pri_cpe
09:49.10dudesI actually have a hard time taking to serious /w your name
09:49.11Kumbangshould i set to pri_net
09:49.21Kumbanghahahhaha
09:49.47Kumbangannoying huh
09:50.21*** join/#asterisk jcwunder (n=chris@ppp-82-135-79-235.mnet-online.de)
09:51.29dudesimmature may be more inline
09:52.12*** join/#asterisk knight_ (i=[U2FsdGV@blackhole.phunc.com)
09:52.13*** join/#asterisk morale (i=russell@S010600111155e117.cg.shawcable.net)
09:52.40Kumbangi just dont get it, why the telco insists that they are the cpe too
09:53.05dudesperhaps you should ask them since they seem to kumbang
09:53.06*** join/#asterisk shido6 (n=shido6@d221-68-216.commercial.cgocable.net)
09:53.23iDunnowhy shouldn't they? :)
09:53.42Kumbangthat's what the log say
09:54.35dudesEither you have an issue with config or a f'ed up teleco
09:55.02Kumbangmaybe TE11XP dont fully compatible with the telco, it works fine with E100P
09:56.18dudesno
09:56.37Kumbangwhy only E100P, what happened with TE11XP
09:56.55dudesI've had a single box /w 5 TE410's talking to the teleco
09:57.04dudesIt's your config
09:57.16dudesor your teleco
09:57.55Kumbanglooks, i just changed card E100p with TE11XP
09:58.30Kumbangi think config doesnt matter, why should i change the configs
09:59.37dudesYour name troubles me ...
09:59.49dudescomeon, Kumbang (what is that)
10:00.09Kumbangdont make me talking dirty please...
10:00.31Kumbangim serious
10:00.43dudesGo figure
10:00.56dudesHey, how about I talk dirty cause I'm such a bad ass pimp
10:01.39Kumbangshould i turn to sangoma
10:02.06coppiceit has nothing to do with the card
10:02.18Kumbangso...
10:02.21dudesIf you want to turn to sangoma you should
10:02.25dudescause yea
10:02.50dudesBut Digium cards are good and if they don't work ... well you should've paid for support
10:03.18Kumbangfyi, the telco is lucent 5ess switch with E1 interface, i dont know that module they're using
10:04.16dudesand that's our issue Kumbang
10:04.16coppiceif they insist on being the CPE, then be the CO end and off you go
10:05.32trixterya know www.listyourself.net is pretty painless..  suprised that a free service like that would be so painless (I used a disposable email account, wonder if it will start to see more spam) ...  for a free service I recommend it!
10:05.49trixteralthough they seemed to have issues with my NCFA number in the UK
10:07.28trixterI am curious how they are able to do this for so many countries, the national registry in north america is fairly painless but I could see real issues in foreign countries
10:10.58*** join/#asterisk elisa (n=elisa@unaffiliated/elisa)
10:12.16*** join/#asterisk saitech (n=admin@85.235.237.14)
10:14.07FuriousGeorgeis figureitout a valid vulue for nat= in sip.conf?
10:14.31trixterheh
10:22.03FuriousGeorgeseriously though, it would be good
10:23.00dudesthey need a auto setting for nat
10:23.10dudesnot need (but it'd be nice)
10:23.13FuriousGeorgewhy should i have to have two accounts per user.  especially with the implications for how quirky sip clients handle the multiple accounts
10:23.16dudesfor a hosted situation
10:23.37FuriousGeorgeanyway, good night all
10:23.43ManxPowerUm, nat=yes really means figure it out.
10:24.15dudesthat makes sense'
10:24.17*** join/#asterisk ast_freak (n=jesse@68-112-134-195.dhcp.stcl.mn.charter.com)
10:25.13saitechi am having a problem with "chan_sip". It keeps looping messages like "Failed to grab lock... trying again" and when it does, it kindda hangs on the sip channel, until the loops stops after 5-10 secs. Need help badly.
10:25.51ManxPowersaitech, I've never heard of that problem.  What version of Asterisk are you using?
10:26.02dudesI was wonder that too
10:26.21saitechasterisk 1.2.1
10:26.58saitechit is vry difficult to find anything about in the mailing list. i have seen an old bug report with the problem, but it from 2003 january, so i cant really use it.
10:27.10saitechi am using gentoo with a 2.6.14 kernel.
10:28.37saitechi had the same problem on the same machine with centos 4.2
10:29.01saitechi am using asterisk 1.2 beta1 on a similar machine. Here i am not having the same problem.
10:30.31*** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl)
10:31.23puzzledmorning
10:31.24dudesI've never heard of tha tissue
10:31.45dudesperhaps looking in chan_sip for that error or turning on better debug and such
10:34.45saitechi have looked, but i dont fully understand the chan_sip code :/
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11:09.42oejsaitech: You need to provide more information about what is happening within your asterisk that causes these locks, provide debugging information.
11:10.49dudesand using pastebin
11:14.46ManxPowerUgh.  I am reminded yet again that I am not a math.geek.
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11:15.34antonios<PROTECTED>
11:16.23antonioshello, I upgraded my asterisk to 1.2.1 and I get  No D-channels available!  Using Primary channel 16 as D-channel anyway! every second or so, any ideas?
11:17.42dudescheck sourse or move back to prior
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11:30.39Mimmushi, does anyone know if there are suggested values for jitterbuffers in zapata.conf?
11:31.00*** join/#asterisk Vijay (i=Vijay@203.122.28.109)
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11:36.16Vijayanyone installed asterisk 1.2on gentoo 2005
11:37.17Vijayhas anyone installed asterisk 1.2 on gentoo 2005?, i am facing some issues in regards to the same, let me know if you have done the same then i would like to take few suggestion from you
11:40.34*** join/#asterisk A-Tuin (n=a-tuin@steves.ip.v4.me.uk)
11:42.47ManxPowerVijay, Why do you believe your problem is unique to Gentoo?
11:42.50*** join/#asterisk zishanov (n=mail@d57-249-149.home.cgocable.net)
11:43.31ManxPowerantonios, I cannot think of any reason for that.  Updgrading should not cause that problem.
11:44.12Vijaybecause when i installed it on fedora, it was working fine
11:44.27Vijayi have copied and pasted all the conf files
11:44.33ManxPowerVijay, What SPECIFIC problem are you having?
11:45.37Vijayi have configured asterisk 1.2, now when i send the calls from my voip device to asterisk server to dial any internal extension, it does not work
11:45.49Vijayand simply transfer this to the voice mail
11:46.20ManxPowerAny error message on the console?
11:46.55Vijayno error message
11:47.14ManxPowerput the CLI output of a failed call on pastebin.ca
11:47.34Vijayi just do it
11:47.40*** join/#asterisk wasim (n=wasim@pdpc/supporter/active/wasim)
11:47.55ManxPowerand paste the URL of the information to this channel.
11:49.01Vijayjust switching on my system
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12:07.38fulgashey
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12:16.21fugitivoho
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12:20.02MarkTVAny1 having experiences with vlines 120 ?
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12:21.33teleniekoHi ppl. On Debian, when compilling zaptel-source, how can I enable Bristuff ?¿ thanks :)
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12:24.30ManxPowertelenieko, BRIstuff is not part of the stock zaptel.
12:24.53*** join/#asterisk wasim (n=wasim@pdpc/supporter/active/wasim)
12:24.53ManxPowerCheck the Wiki for where you can download zaptel+BRIStuff
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12:27.45teleniekoManxPower I know, but on the debian's zaptel-source it is bundled, but it says "disabled" with no clues on how to re-enable ;(
12:27.58teleniekobristuff-0.3.0 I think
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12:28.06viperdudeexit
12:28.08ManxPowertelenieko, I can't help you with distro specific stuff
12:28.08viperdudequit
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12:28.18teleniekoManxPower thx anyway ;))
12:28.24moraleapt-get source zaptel && cd zaptel-* && make && make install
12:28.53ManxPower<PROTECTED>
12:28.57teleniekomorale I did: apt-get install zaptel-source; m-a a-i zaptel :)
12:29.01moralethat just builds it
12:29.14moralebri stuff? hold on
12:29.33ManxPowerremember, distro packages are always horribly out of date and buggy.
12:29.36*** join/#asterisk joelsolanki (i=joelsola@202.160.161.93)
12:29.36teleniekomorale if you look at zaptel-source changelog you'll see "Disabled bristuff" :o
12:30.00*** part/#asterisk JunK-Y (n=junky@67.71.110.21)
12:30.02teleniekoManxPower debian ones seem to work well except of that bristuff issue :)
12:30.03moralewhat version of zaptel source?
12:30.15ManxPowertelenieko, what version of zaptel does it have?
12:30.32morale1.2.1 says nothing about it
12:30.40moraleeven better download it from ftp.digium.com
12:31.23fugitivodistros are buggy and out of date!
12:31.26fugitivouse lfs!
12:31.30telenieko1.2.1 and bristuff 0.3.0
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12:42.11kippihey
12:42.41kippiIf I have [internal] exten => 1001,1,Macro(extensions,${EXTEN},${EXTEN}) to add another extension would I just add exten => 1020,1,Macro(extensions,${EXTEN},${EXTEN}) ?
12:43.58ManxPowerkippi, Well, that would depend on macro-extensions
12:44.06kippiwhere would I find that?
12:44.33ManxPowerHmm?  I have no idea.
12:44.51kippi[macro-extensions]
12:44.52kippiexten => s,1,Dial(Sip/${ARG1}|20)
12:44.52kippiexten => s,2,Voicemail(u${ARG2})
12:44.52kippiexten => s,102,Voicemail(b${ARG2})
12:44.54ManxPowerI don't think it's part of the sample config files
12:45.24ManxPowerSo what's the problem?
12:45.52kippiso to add another extension i would just add exten => 1020,1,Macro(extensions,${EXTEN},${EXTEN})
12:46.18ManxPowerNo.
12:46.54ManxPowerWhat do you want to add another extension TO?  You want to ring two extensions at the same time?
12:47.37fugitivoi think he wants to know how to use that macro
12:47.40kippiI just want another exenstion like all the other ones
12:47.53ManxPowerkippi, I do not understand your question.
12:48.08fugitivokippi: it's ok how you wrote it
12:48.10theNOTOHas anyone used both the Aastra 480i and the Polycom 501 and could give me a comparison of the two?
12:49.34tzangertheNOTO: I have heard that they are both very good phones.
12:50.03theNOTOtzanger: Yeah, I am trying to decide between the two
12:50.06tzangerso much so that those are the only two phones I'd think of using for office use, cisco notwithstanding (too expensive)
12:50.20ManxPowerI use Polycom
12:50.29theNOTOtzanger: I have heard the Aastra's are having some problems with recent firmware
12:50.32ManxPowerAastra is a pretty recent product.
12:50.39ManxPower(well Aastra SIP)
12:51.14tzangerwell aastra is also a very very good POTS phone (I use their PT390s all the time) so I know they have the phone bits down pat (weight, feel, etc.).  I've never tried either though, I always use POTS phones :-)
12:51.18theNOTOManxPower: which polycom model?
12:51.37ManxPowertheNOTO, IP 30x, 50x, 60x
12:52.07tzangerpolycom is world-renowned for their POTS and digital (KSU/PBX) speakerphones and video phones (we have their super-expensive VSX7000 at work)
12:52.10theNOTOManxPower: Is the 30x a big step down from the 50x series?
12:52.16ManxPowertheNOTO, yes.
12:52.29theNOTOManxPower: that's what i had heard
12:52.36fugitivoprice too
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13:01.51yxau cannot talk using 301 speakerfone
13:02.11tzanger301 does not have a speakerphone IIRC
13:02.15yxaand POE support is not native
13:02.31yxait does, the speakerphone is just one-way for listening only
13:02.53yxaother than that, i think polycom has the best sound quality
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13:05.30theNOTOI had heard Polycom's voice quality was the best
13:05.49theNOTOmaybe i should just stick with POTS phones for now
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13:10.40RoyKmorning
13:11.57coppiceRoyK: do you have much snow there? it seems further south had quite a bit these last 2 days
13:12.57Mimmushi, does anyone know if there are suggested values for jitterbuffers in zapata.conf?
13:12.58RoyKnot really a lot
13:13.01RoyKat least not here in oslo
13:14.36coppiceRoyK: I am abandoning work on T.38 for *. the initial passthrough stuff has languished in the bugs blackhole for nearly 4 months. I've lost interest
13:15.02RoyKseriously_
13:15.04RoyK?
13:15.06fugitivocoppice: why?
13:15.10RoyKwhat can be done to mend this?
13:15.30fugitivopeople? money? women?
13:15.33coppicebugs.digium.com only seems to exist to piss people off
13:16.27RoyKs/bugs\.(\w+)\.com/$1/ ?
13:16.38fugitivolol
13:16.40fugitivothat's true
13:17.12ManxPowercoppice, contribute it to OpenPBX
13:17.28*** join/#asterisk wasim (n=wasim@pdpc/supporter/active/wasim)
13:17.38coppiceits in openpbx already
13:17.44ManxPowercoppice, Ah.
13:18.08ManxPoweradd a note with that info on your bugs.digium.com entry 8-)
13:20.18*** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com)
13:22.48coppicemaybe digium aren't very interested in the whole setup, since it requires spandsp, which is GPL. you'd think someone might at least show a little enthusiam for passthrough, though
13:24.57ManxPowercoppice, passthru should not need spandsp, should it?
13:25.17coppiceno. the contributed code is all that is needed
13:25.35Mimmushi, if my interlocutors on analog phones hear my voice from VoIP phone with very bad quality, where can look for problems?
13:25.37RoyKdisclaimed?
13:25.52ManxPowerinterlocutors?
13:26.08MimmusManxPower: no? sorry for my bad english
13:26.14ManxPowerMimmus, using MeetMe?  What SIP phone?
13:26.26MimmusManxPower: no no, during a normal call
13:26.28fugitivoMimmus: network latency, codecs, microphone, network cable
13:26.33RoyK'where can i look for problems'
13:26.39ManxPowerMimmus, What VoIP device?
13:27.05MimmusManxPower: thanks, I have an Atcom AT-320 (made in China) VoIP phone
13:27.19fugitivoMimmus: THAT'S THE PROBLEM
13:27.20fugitivolol
13:27.25fugitivothat phone is crap
13:27.38ManxPowerMimmus, Many things can cause the problem you experience.
13:27.46fugitivoManxPower: it's the phone
13:27.52Mimmusfugitivo: i think so too! it seems that handset+microphone work very well
13:27.56ManxPowerfugitivo, Um, many people use those phones.
13:28.00fugitivoMimmus: are you using iax or sip?
13:28.05*** join/#asterisk littleball (n=littleba@cm240.epsilon174.maxonline.com.sg)
13:28.06Mimmusfugitivo: SIP
13:28.08fugitivoManxPower: i have one of those here in my desktop
13:28.23fugitivoMimmus: are you using qualify=yes ?
13:28.46Mimmusfugitivo: in sip.conf?
13:28.51ManxPowerMimmus, If there is an option to select Audio Packet Size, set it to 20ms.  Many phones use 30ms packet size.  30ms will cause audio problems.
13:28.52fugitivoyes
13:29.04Mimmusfugitivo: no, always qualify=no
13:29.10fugitivoset it to yes
13:29.17Mimmusfugitivo: what is this?
13:29.32fugitivoi want to know how many ms you have from asterisk to the phone
13:29.48MimmusManxPower: ok, I will check in a few seconds...
13:31.47zoa2coppice didnt you disclaim spandsp ?
13:31.47coppicenope
13:31.48coppiceyou may not use spandsp with *, if you are using * outside the GPL. that means no G.729 or openh323
13:31.48MimmusManxPower: I have 'audio frames=2', 'jitter size=0'
13:32.24*** join/#asterisk jimmy_deanPB (n=jhodapp@indianalifesciences.com)
13:32.51fugitivoMimmus: did you set qualify=yes?
13:32.58*** join/#asterisk pakipenguin (n=Junaid@linuxpakistan/admin/pakipenguin)
13:33.01pakipenguinevening
13:33.06fugitivomorning
13:33.13RoyKafternoon
13:33.23Mimmusfugitivo: no, what is this option? I have 2-2.5 ms from asterisk to phone
13:33.59fugitivoMimmus: it will show the status of the phone when doing sip show peers
13:34.24*** join/#asterisk Phrog123 (n=fmenard@cable-234-38.cgotr.infoteck.qc.ca)
13:34.40littleballhello, i have install asterisk 1.2.1, i try to use postgresql to store cdr. but i cannot find any cdr record in the cdr table. any idea?
13:34.45Phrog123folks,
13:34.51Mimmusfugitivo: ah, ok, it is for this reason I have 'Status: unmonitored'
13:34.57fugitivoMimmus: right
13:35.07Phrog123is there a way to have asterisk not pick-up a line if it has been picked up outside of asterisk.
13:35.22fugitivolittleball: did you setup cdr_pgsql.conf?
13:35.29littleballfugitivo, yues
13:35.34littleballi set it already
13:35.42Phrog123i.e. I have a couple of pots phone at home, I put wait(12) in case my wife picks up pots, but the darn wcfxo picks up the line after 4 rings
13:36.11fugitivolittleball: show modules like cdr_pgsql
13:36.16fugitivodo you have the module loaded?
13:36.35Phrog123Is there a way to have asterisk check for a ring tone just before answering and if it does not see a ring, it sleeps?
13:36.51littleballfugitivo, do you mean lsmod?
13:37.00fugitivolittleball: no, from the CLI
13:37.02Mimmusfugitivo: I set qualify for all sip extensions
13:37.09fugitivolittleball: run this "show modules like cdr_pgsql"
13:37.12fugitivowithout the "
13:38.27littleballno
13:38.37littleballit is not there
13:38.44fugitivothen it'll not work
13:38.55fugitivothe problem is that with the new version of asterisk, that module is obsolete
13:39.00fugitivoyou have to enable it at compilation time
13:39.05fugitivoor use odbc
13:39.08littleballbut i configure it already . then how to fix it?
13:39.17littleballok
13:39.28littleballodbc for postgresql?
13:39.30fugitivoi don't know why they did that
13:40.32littleballwhich configuration file i should change?
13:40.33*** join/#asterisk JaredBluestein (n=Jared@nwlnnhbas01-pool0-a209.nwlnnh.tds.net)
13:42.03MarkTVHi all. Just got a vlines120 (IAX2), Outbound calls is working fine, but inbound calls wont work...any ideas ?
13:42.29RoyKlittleball: you prolly need the postgresql developer package to have asterisk make that module
13:42.45littleballRoyK, i compile postgresql myself
13:42.50RoyKok
13:43.02Mimmusfugitivo: now I'm seeing an unreal high value (195 ms)
13:43.16RoyKdo a make clean in asterisk, make all &> logfile and look through the logfile for postgres stuff
13:43.58littleballRoyK, i can have a try
13:43.58RoyKmimmus: that qualify value is the time it takes before the client answers some SIP OPTION packages, IIRC
13:44.33*** join/#asterisk _Madar (n=tophe@219-84-129-163-adsl-tpe.static.so-net.net.tw)
13:44.50MimmusRoyK: other extensions have 1,2 ms!
13:45.10RoyKMimmus: whatever. some clients take their time
13:45.13*** join/#asterisk nfi|ermes (n=ermsewrk@217.220.121.62)
13:45.19PoWeRKiLLSomeone use phpagi with asterisk 1.2.1 ?
13:45.30*** join/#asterisk pengyong (n=lala@218.93.103.120)
13:45.48MimmusRoyK: ok, am I worryng without reasons?
13:45.59RoyKprobably, yes
13:46.00MarkTVPHPAGI works perfect with my 1.2.1
13:46.12RoyKMimmus: i usually set qualify to 5000
13:46.31MarkTVwhy ?
13:46.44MimmusRoyK: I'm trying to investigate some bad behaviour of my phones
13:46.53RoyKbecause a value of 2000 may cause clients on a figgin LAN to become 'unreachable'
13:47.06Mimmusand <fugitivo> suggested me to look at this value
13:47.08RoyKprolly something in the asterisk code that's quite bad
13:47.40PoWeRKiLLMarkTV I can't get DIALSTATUS variable anymore one * 1.2.1 and it was working on 1.2.0 any idea ?
13:47.56RoyKwtf?
13:47.57PoWeRKiLLMarkTV I get error Invalid or unknown command
13:47.58RoyKPoWeRKiLL: sure?
13:48.09fugitivoMimmus: i have the same problem with that phone
13:48.10PoWeRKiLLRoyK: yes
13:48.42fugitivoMimmus: if you use the iax firmware, you'll se that number lower
13:48.42Mimmusfugitivo: well, next time I will buy Grandstream phones
13:48.45*** join/#asterisk sigmounte_ (n=sigmount@www.sighq.net)
13:49.00fugitivoit seems the firmware of that phone isn't well developed
13:49.30Mimmusfugitivo: yes, in fact there was v.1.47 but now they come back to 1.43
13:50.02Mimmusfugitivo: and v.149 is coming
13:50.02fugitivoand if you use jitterbuffer with the lastest iax firmware, the phone won't pickup the calls
13:50.05RoyKPoWeRKiLL: how do you try it? what do you try to do?
13:50.23MarkTVPoWeRKiLL wait i'll take a look
13:50.34RoyKPoWeRKiLL: i got this when i dialed a dummy client
13:50.39RoyK.
13:50.43Mimmusfugitivo: can I set jitterbuffer=0 or there is some controindication?
13:51.06fugitivoMimmus: what do you have now/
13:51.07RoyK-- Executing NoOp("Zap/1-1", "CHANUNAVAIL") in new stack
13:51.16RoyKseems to me dialstatus works......
13:51.25Mimmusfugitivo: default, 4 I think in zapata.conf
13:51.36PoWeRKiLLtry to get via DIALSTATUS
13:51.37fugitivoMimmus: but that's for zaptel
13:51.48fugitivoMimmus: that's not the problem you're having
13:52.00PoWeRKiLLtry to get DIALSTATUS via phpagi
13:52.01Mimmusfugitivo: sorry, it was a mistake
13:52.15littleballfugitivo and RoyK, i think cdr.c give me some idea
13:52.16RoyKPoWeRKiLL: that was from a NoOp(${DIALSTATUS})
13:52.20littleball<PROTECTED>
13:52.20littleball<PROTECTED>
13:52.20littleball<PROTECTED>
13:52.20littleball<PROTECTED>
13:52.20littleball<PROTECTED>
13:52.23littleballze, batchtime);
13:52.24littleball<PROTECTED>
13:52.26littleball<PROTECTED>
13:52.27Mimmusfugitivo: I have default value
13:52.28littleball<PROTECTED>
13:52.29RoyK~pb
13:52.31jbotrumour has it, pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca
13:52.41littleballsorry, i should paster it there
13:52.43fugitivoMimmus: the problem is from atcom320 -> pstn ?
13:52.45PoWeRKiLLI get error Invalid or unknown command from res_agi when running GET VARIABLE DIALSTATUS
13:53.14fugitivolittleball: do you have postgresql installed?
13:53.16Mimmusfugitivo: yes, users at pstn hear bad quality voice
13:53.22littleballof course, fugitivo,
13:54.02littleballthe log message of /var/log/asterisk/messages show "CDR simple logging enabled"
13:54.14littleballi think i should enable batchmode
13:54.24littleballand set the batchtime to zero
13:54.45Mimmusfugitivo: and at atcom320 I hear some echo and lag
13:55.33littleballfugitivo, i think the only way to log this message is due to batchmode is false
13:55.44littleballlet me try now
13:58.00fugitivoMimmus: did you try another phone?
13:58.11fugitivoMimmus: a regular phone with an ata?
13:58.19Mimmusfugitivo: only softphones, and it seems that they are ok
13:58.49Mimmusfugitivo: next days, I will buy a Grandstream
13:59.32Mimmusfugitivo: in the meanwhile, I'm upgrading to * 1.2.1 and changing Dgium E1 card with a Sangom!
13:59.51fugitivowhy are you changing the digium e1?
14:00.23Mimmusfugitivo: to eliminate any doubt!
14:00.26PoWeRKiLLI get error Invalid or unknown command from res_agi when running GET VARIABLE DIALSTATUS from PHPAGI
14:02.38RoyKPoWeRKiLL: can you retreive other variables_
14:02.39RoyK?
14:02.50PoWeRKiLLno
14:03.15PoWeRKiLLI try to get other variable same error
14:03.17littleballfugitivo, still cannot
14:03.41littleballi just don't understand why asterisk always use csv format to store cdr
14:03.44ManxPowerPoWeRKiLL, what version of Asterisk?
14:03.50RoyKPoWeRKiLL: res_agi error?
14:05.39fugitivolittleball: type "cdr status" on the cli
14:05.56littleballok
14:06.18fugitivothe problem is that you don't have cdr_pgsql.so
14:06.29PoWeRKiLL1.2.1
14:06.34littleballi think the one you are interested is CDR registered backend: csv
14:06.37*** join/#asterisk saftsack (n=oliver@p54A7D1AA.dip.t-dialin.net)
14:06.40littleball"CDR registered backend: csv"
14:06.45saftsackhi
14:06.51PoWeRKiLLthis error came from GET VARIABLE variablename
14:06.58saftsacksomeone who has the grandstream budge tel 101?
14:06.59fugitivolittleball: did you compile from source?
14:07.04iCEBrkryo
14:07.04littleballfugitivo, yes
14:07.25littleballwhy?
14:07.50RoyKlittleball: did you look through the compile log_
14:07.51RoyK?
14:07.51littleballi download asterisk-1.2.1 from asterisk.org
14:08.07littleballlet me do now. RoyK
14:08.08fugitivolittleball: if you don't have that module, it didn't find your pgsql files for compiling
14:08.17ManxPowerPoWeRKiLL, Weird.
14:08.18RoyKlittleball: the problem is prolly there
14:08.33RoyKlittleball: usually asterisk just can't find the psql headers and skips that section
14:08.40mutilatoranyone know if there is a way to get date or something else to do date calcs?
14:08.40ManxPowerCan you GET VARIABLE variablenameinglobalsection ?
14:08.42mutilatorlike date -d now-1day +%F
14:08.52fugitivolittleball: check asterisk-1.2.1/cdr/Makefile to see where asterisk look for the postgresql files
14:08.59littleballRoyK, ok
14:09.09fugitivomutilator: date command?
14:09.11saftsackg711 is isdn, right, but what is g722?
14:09.22ManxPowersaftsack, no, G711 is not ISDN
14:09.30saftsackwhat is isdn?
14:09.33littleballfugitivo, no
14:09.40fugitivolittleball: no what?
14:09.42littleballit doesnt look for post*
14:09.42ManxPowersaftsack, ISDN is a signaling protocol.
14:09.45saftsackok
14:09.47RoyKsaftsack: g.711a and g.711u are the codecs used on isdn
14:09.53saftsackok :)
14:10.02saftsackbut my budgetel 101 doesnt provide g711
14:10.08littleballfutitivo, i search Makefile, no text match post
14:10.12saftsackalso tehre arent alaw and ulaw :(
14:10.15*** join/#asterisk newl (n=newlook@203-59-61-62.dyn.iinet.net.au)
14:10.19littleballpostgre
14:10.23fugitivolittleball: asterisk-1.2.1/cdr/Makefile??
14:10.28ManxPowerG711 is alaw (aka PCMA) or ulaw (PCMU).  This is the "codec" all normal PSTN calls use.
14:10.31fugitivolittleball: search pgsql
14:10.32saftsackyes
14:10.32RoyKsaftsack: g.711a == alaw. g.711u == ulaw
14:10.38littleballyes
14:10.39saftsackok :)
14:10.44littleballit is there
14:10.45saftsackwhat has better quality?
14:10.48saftsackalaw or ulaw?
14:10.57RoyKsame shit
14:11.00littleball/usr/include/postgresql
14:11.07ManxPowersaftsack, both are the same.  the PSTN in usa/canada uses ulaw, PSTN in most of the rest of the world uses alaw
14:11.07fugitivolittleball: ok, do you have those directories?
14:11.08RoyKsaftsack: but ulaw is used in north america
14:11.18littleballno
14:11.20saftsackok im living in europe ;)
14:11.24littleballthis could be the reason?
14:11.26fugitivolittleball: then you don't have postgresql
14:11.29RoyKsaftsack: then just stick to alaw
14:11.33saftsackok :)
14:11.48fugitivolittleball: or you installed postgresql on a non-standard directory
14:11.56littleballfutitivo, i have installed it but i think the header file is not in that place
14:12.05littleballyes
14:12.18littleball/usr/local/pgsql/include/
14:12.20RoyKfugitivo: he prolly installed it in the 'standard' dir, being /usr/local/postgresql or something
14:12.21RoyKyeah
14:12.22littleballis my directory
14:12.32RoyKlittleball: just symlink the header files to /usr/local/include
14:12.36fugitivolittleball: find / -name pg_config.h
14:12.37littleballok
14:12.59RoyKlittleball: or /usr/include
14:13.02fugitivooh, you have it
14:13.02littleballwait
14:13.03saftsackhowto control which codec is active?
14:13.04RoyKlittleball: just make sure asterisk finds them
14:13.08saftsacksip show peers shows nothing
14:13.12fugitivoRoyK: is that standard? /usr/local? :)
14:13.15RoyKsip show peer
14:13.22fugitivothat's redhat standard ;)
14:13.24RoyKfugitivo: standard from pgsql, yes
14:13.24littleballin Makefile, it seems that it also search for /usr/local/pgsql/include
14:13.31RoyKok
14:13.32saftsackRoyK, i did it ;)
14:13.36saftsackbut nothing
14:13.37ManxPowersaftsack, you control it in sip.conf for SIP devices.
14:13.48RoyKsaftsack: sip show peer xxxx shows what codecs xxxx is using
14:13.52saftsackok
14:14.06ManxPowersip show channels will show you the codec for each ACTIVE call.
14:14.26littleballfugitivo, for what purpose to find / -name pg_config.h?
14:15.03fugitivolittleball: none, you already know where your headers are
14:15.17saftsack<PROTECTED>
14:15.22littleball/usr/local/pgsql/include/pg_config.h
14:15.47*** join/#asterisk frenzy (n=frenzy@80.255.63.30)
14:15.51frenzyhey all
14:15.53fugitivowell, asterisk already look at that directory for compilation
14:16.05frenzyhow do I setup DNID ?
14:16.16frenzyI have calls coming in via IAX
14:16.18RoyKwtf is DNID?
14:16.24littleballfugitivo, so i don't think this is the reason, right? becuause it is there
14:16.50frenzythe number from which the call is coming through
14:17.03fugitivolittleball: ls -la /usr/lib/asterisk/modules/cdr_pgsql.so
14:17.06fugitivodo you have that file?
14:17.46ManxPowerfrenzy, you don't.  It's enabled by default.  If the carrier supports it, the call will be sent to the exten => line matching the DID or some part of the DID.
14:18.03ManxPowernot all carriers send the DNID info, in which case exten => s is matches
14:18.08frenzyI have it seupt in the same way
14:18.22frenzybut for some reason AGI says Unknown
14:18.25*** join/#asterisk brad_mssw (n=brad@shop.monetra.com)
14:18.25littleballno
14:18.33littleballfutitivo, no
14:19.09ManxPowerfrenzy, you didn't say you were using AGI.
14:19.18littleballi am recompile again. make clean;make all|tee /tmp/1
14:19.38frenzymy bad :(
14:19.54ManxPowerAGI should always have the EXTEN (DNID) available in the info sent to the AGI when it starts.
14:20.13littleballfugitivo, after recompile, still cannot find this file. do i need to enter cdr directory to make
14:20.14littleball?
14:20.51iCEBrkrlittleball: Do you have postgres-devel installed?
14:20.53frenzy<PROTECTED>
14:21.23littleballgot it already
14:21.34iCEBrkrlittleball: Then what's failing?
14:21.53littleballiCEBrkr, i got this file(stupid error when i recompile, i forget to use root to make install
14:22.01littleballlet me try now
14:22.06fugitivo...
14:24.04ManxPowerfrenzy, you need to talk to the people that wrote a2billing thebn
14:24.49littleballby the way, what is the benifit of RealTime ?
14:25.23fugitivorealtime changes without reloading
14:25.26*** join/#asterisk amir (n=amir@hacker-217-147.congress.ccc.de)
14:26.06cfhhas asterisk support for sip "replaces" ?
14:26.09De_MonI'm using a sip phone with a configured stun server, do I need to do something to point asterisk to the same stun server?
14:26.20fugitivobut it's a little benefit compared to the weight of using a database for your pbx
14:27.25De_Monhave a stun client running on the asterisk machine maybe? I'm not sure how to connect them
14:28.50littleballhi, futitivo, it works now
14:28.52littleballthanks very much
14:29.27*** part/#asterisk cfh (n=luca@82.193.23.6)
14:29.45Mimmustoday fugitivo is a great resource!
14:31.12littleballanyway, what is the best way to get the cdr data and send them to external program through AGI?
14:31.36*** join/#asterisk cfh (n=luca@82.193.23.6)
14:31.48frenzyhow do I add extensions to IAX Registrations?
14:31.59frenzylike on SIP simply add /exten
14:32.06littleballcurrently, i am using exten=>h and DeadAGI
14:33.08MimmusI configured native MusicOnHold but 'ps -ax' still shows mpg123 processes. How can I disable these?
14:35.20iCEBrkrMimmus: Sure those processes aren't left over from before you changed it?
14:36.08iCEBrkrlittleball: AGI doesn't scale and it's slow as shit
14:36.31Kattyi could use support this morning.
14:36.35Kattybut it has nothing to do with asterisk.
14:36.42iCEBrkrKatty: You need more than 'support' you need a support group.
14:36.47KattyiCEBrkr: i know.
14:36.58KattyiCEBrkr: but my batch script keeps imploding.
14:37.00iCEBrkrKatty: How about you stand up and introduce yourself to everyone?
14:37.09KattyiCEBrkr: i did that /ages/ ago.
14:37.17iCEBrkrMaybe you should try again?
14:37.23Kattystanding takes too much effort.
14:37.28iCEBrkrAgreed
14:37.57*** join/#asterisk P4C0 (n=paco@200.124.22.34)
14:38.55P4C0hello guys, me again... asterisk fails to re-registry with my sip provider: wrong password
14:39.00littleballiCEBrkr, AGI is not so slow. i am using AGI(agi://....
14:39.07*** join/#asterisk wasim (n=wasim@pdpc/supporter/active/wasim)
14:39.13iCEBrkrP4C0: Dump that shitty provider and get a real provider :)
14:39.18littleballi am using asterisk-java
14:39.21iCEBrkrlittleball: Trust me. It's slow.
14:39.28fugitivojava is slow
14:39.31littleballthen what is the best way?
14:39.41iCEBrkrlittleball: Depends on what you're trying to do.
14:39.42P4C0iCEBrkr, naah, it works well with a sip phone and they are using asterisk as well...
14:40.17P4C0iCEBrkr, i mean I can register and after a while it fails... and keep failing... is like if the re register don't sent the user / pass
14:40.22littleballfutitivo, iCEBrkr, i don't think so because AGI only send data to a local loop pipe (local TCP socket), that is all.
14:40.42iCEBrkrP4C0: Well, a lot of people use Asterisk and don't have that problem, so I'm thinking it's your provider :)
14:40.46fugitivolittleball: java is slow, agi is slow, so agi+java will be SLOW
14:40.54iCEBrkrfugitivo: thank you.
14:41.03ruzais there a way how can i delete multiple messages from voicemail ?
14:41.15fugitivoiCEBrkr: you're welcome
14:41.18P4C0iCEBrkr, maybe the configuration...?
14:41.37littleballand the TCP connection is persistently connectled. Yes, if without agi://..., it is big problem. but by using agi://, it works very well in my 1.09 box
14:41.37iCEBrkrlittleball: TRUST me on this, mmk?  I've been dealing with AGI for the past few months and it's dog slow when you're trying to scale your system.
14:41.47fugitivoruza: rm -rf /var/asterisk/spool/voicemail/extension/folder/*
14:41.54*** join/#asterisk saftsack (n=saftsack@p54A7E001.dip.t-dialin.net)
14:41.55saftsackhi
14:41.56littleballiCEBrkr, are you using asterisk-java?
14:42.02iCEBrkrlittleball: It doesn't MATTER
14:42.02saftsacki have a problem and its gross :(
14:42.10fugitivoerrrr
14:42.13iCEBrkrlittleball: The AGI() call itself is SLOW
14:42.21saftsackevery asterisk whcih is compiled says this while starting
14:42.24fugitivoi mean
14:42.32fugitivo<PROTECTED>
14:42.40littleballiCEBrkr, then how to solve such problem (sending channel data to external program)
14:42.46iCEBrkrlittleball: Not to mention, you're making a socket connection and then java?! You're nutso
14:42.49saftsackOuch ... error while writing audio data: : Broken pipe
14:42.49saftsackWarning, flexibel rate not heavily tested!
14:42.52fugitivoruza: /var/spool/asterisk/voicemail/
14:43.08iCEBrkrlittleball: As I said before, it really depends on what you're trying to do?
14:43.12P4C0iCEBrkr, when registering, how can I tell asterisk to use the public ip instead of the private? maybe in the reregistering he is using the private ip and my server blocks it...!?
14:43.30littleballiCEBrkr, i think you misunderstand
14:43.45fugitivolittleball: if you have a small scenario, AGI it's ok
14:44.21iCEBrkrlittleball: No, I think you misunderstand.
14:45.04ruzafugitivo: can voicemail user do it somehow ?
14:45.11iCEBrkrlittleball: I have a $25,000 project that when I started, I was using AGI thinking it'd be OK to use. It is not.
14:45.16littleballiCEBrkr, if what you said is that AGI () writing data to local socket (an existing socket) is proved to be slow, then i agree with you.
14:45.32littleballiCEBrkr, how do you try AGI?
14:45.52iCEBrkrlittleball: I had several 5 line PHP scripts that I called via AGI.
14:46.04iCEBrkrlittleball: Regardless of the scripts, AGI is slow
14:46.08fugitivoiCEBrkr: how many concurrent calls?
14:46.08iCEBrkrI'm not gonna say it again.
14:46.16*** join/#asterisk _Madar (n=tophe@219-84-129-163-adsl-tpe.static.so-net.net.tw)
14:46.17iCEBrkrfugitivo: I need up to 72
14:46.42iCEBrkrfugitivo: Doesn't matter anymore, as I've since switched to using Corydon's ODBC function stuff
14:46.46fugitivolittleball: iCEBrkr is saying that the function AGI() is slow, not what you execute with AGI
14:46.52P4C0can I put like sip debug for the registry only!?
14:46.54*** join/#asterisk af_ (n=af@ip-142-84.sn1.eutelia.it)
14:47.28fugitivoP4C0: sip debug peer xxx
14:47.42iCEBrkrlittleball: I was in your shoes at the begining, I really didn't think AGI() was all that slow and everyone here kept telling me the same thing....
14:47.57ruzafugitivo: i.e. vmail.cgi ?
14:47.58P4C0fugitivo, thanks :)
14:48.04iCEBrkrlittleball: So, you can do whatever you want.
14:48.11fugitivoruza: using a web client, yes
14:48.19littleballif you call script, it is definitely slow. in my system, AGI do nothing. It just write to socket and then the external java return immediately and using a seperate thread to process the socket data
14:48.25fulgasagi is quite slow...
14:48.30fulgasi'm also using it
14:48.35ruzafugitivo: great :)
14:48.40saftsackcan i start asterisk without the colors just with a white background in the cli?
14:48.52iCEBrkrlittleball: I give up on you..
14:49.03fugitivosaftsack: don't use -c
14:49.28iCEBrkrlittleball: You just don't understand.. THE ACTUALLY AGI CALL IS F'N SLOW
14:49.31Kattyblast you infernal machine with entirely too much ram! you are /not/ having insufficent memory problems!
14:49.50fugitivoKatty: lol
14:50.05littleballiCEBrkr, if AGI call itself within asterisk is slow, then i agree with you
14:50.24iCEBrkrI've been saying that for the past 20mins
14:50.30tRSSI have one asterisk(A) and two zultys (B and C) ip telephony exchanges. B is able to dial international numbers through C. A is able to dial any extension on B and C but A is not able to dial international numbers through C. Instead C returns a 410 - Gone error. I would appreciate some help on it?
14:50.34P4C0humm strange, the reregistration when ok
14:50.46littleballiCEBrkr, what is your solution then?
14:51.03iCEBrkrlittleball: I have no solution cuz you won't explain to me what you're trying to do.. Which I've asked 3 times now
14:51.29littleballiCEBrkr, i just want to pass channel parameters to my external program
14:51.37iCEBrkrOh, descriptive.
14:51.42littleballsome channel parameters are defined by myself.
14:51.47iCEBrkrWTF are you trying to do?
14:51.50iCEBrkrya know what
14:51.50iCEBrkrfuck it
14:52.07P4C0how can I put my public ip in here: Via: SIP/2.0/UDP 192.168.6.10:5060 !?
14:52.10littleballe.g., callerid, ...
14:52.39fugitivoiCEBrkr: hey, don't start being a jerk :)
14:53.06iCEBrkrI have high regard for intelligence and no tolorance for ignorance
14:53.12littleballagi://localhost/mobmeee.agi?seconds=foo${allocatedcallingseconds}&duration=foo${DIALEDTIME}&billsec=foo${
14:53.12littleballANSWEREDTIME}&accountcode=foo${ACCOUNTCODE}&src=foo${CALLERIDNUM}&dst=foo${legb})
14:53.15frenzyforsome reason IAX calls DNID's are not being collected by AGI
14:53.18littleballthis is example
14:53.18iCEBrkrDon't ask for help if you're not going to take it.
14:53.44littleballi am using agi to pass some parameters to external program
14:54.06iCEBrkrWTF does your external program do.. ( I asked again.. this is 5th time )
14:54.07frenzy?
14:54.41littleballiCEBrkr, it is web based application which trigger the call and then receive the parameters passed back from agi
14:54.52iCEBrkrSounds retarded to me.
14:54.57iCEBrkrBut without knowing the details.
14:55.03iCEBrkrI can't help you make it the right way
14:55.11iCEBrkrYou keep spewing vague descriptions
14:55.21xhelioxlol
14:55.33iCEBrkrIt's like talking to a Magic 8-Ball
14:55.33xhelioxThis conversation haven't been informative, but it's been amusing.
14:55.39xhelioxhasn't*
14:56.06littleballiCEBrkr, i think it is so clear. i need asterisk to pass some parameters to my external program which need using such parameters to process or bill users etc
14:56.14iCEBrkrNO JACKASS
14:56.19iCEBrkrWhat's your end-goal
14:56.26iCEBrkrWho cares about passing data
14:56.37littleballthat is all what i need
14:56.50iCEBrkrwhere's the ignore button on this thing?
14:57.02tRSSI have one asterisk(A) and two zultys (B and C) ip telephony exchanges. B is able to dial international numbers through C. A is able to dial any extension on B and C but A is not able to dial international numbers through C. Instead C returns a 410 - Gone error. I would appreciate some help on it? What causes a "410 - Gone" error?
14:57.20iCEBrkr(09:55)Ignoring ALL from *!*@*.sg
14:57.29fugitivolittleball: describe all the path, for example, a call triggers the AGi and blah blah
14:58.09P4C0guys hard question: I need a sip softphone that works on linux, with artsd supoprt anyone knows?
14:58.40fugitivoP4C0: did you try xlite?
14:58.40iCEBrkrP4C0: #linux is down the hall
14:58.47littleballok. an external program trigger the call through asterisk amnager api and once the call is connected or hangup, the asterisk server will use agi to notify the external program
14:58.56littleballs/amnager/manager
14:59.01P4C0fugitivo, no I haven't, I will
14:59.07P4C0iCEBrkr, I love u too
14:59.10iCEBrkr:)
14:59.12iCEBrkr:*
14:59.27P4C0fugitivo, thanks
14:59.37iCEBrkrI've become just as jaded as the rest of the #asterisk guys
14:59.49*** join/#asterisk JaredBluestein (n=Jared@nwlnnhbas01-pool0-a209.nwlnnh.tds.net)
14:59.51cfhwhere can i find indications for  snom programmable buttons with asterisk ?
15:00.45*** join/#asterisk [TK]D-Fender (n=joe@toronto-HSE-ppp4122655.sympatico.ca)
15:00.51P4C0Dec 29 09:49:51 NOTICE[6664]: res_musiconhold.c:507 monmp3thread: Request to schedule in the past?!?! <-- why I'm getting like 100x lines of this?
15:00.58littleballiCEBrkr, you said that AGI is slow, so i think you give up this solution. then i think change your system design to avoid using agi, right?
15:01.20P4C0iCEBrkr, that's ok :)
15:02.38*** join/#asterisk Cresl1n (n=matt@gateway.digium.com)
15:03.23*** part/#asterisk JaredBluestein (n=Jared@nwlnnhbas01-pool0-a209.nwlnnh.tds.net)
15:05.31Kattyahhhhhh hahahahahahaha.
15:05.33Kattystupid windows.
15:05.35*** join/#asterisk Utah_Dave (n=boucha@0-1pool138-70.nas28.salt-lake-city1.ut.us.da.qwest.net)
15:05.47Kattywith its lolzinsufficentmemory /really/ means this filename is too long! i dunno how to read it!
15:06.31iCEBrkr'more'
15:06.32iDunnoheh
15:06.57iDunnostupid java with it's stupid non-portability.
15:07.06*** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com)
15:07.06*** mode/#asterisk [+o anthm] by ChanServ
15:07.12iCEBrkriDunno: ?
15:07.21KattyiCEBrkr: at least the mysterious of the Insufficent Memory has been solved.
15:07.33Mimmus"!! Unknown IE 26 (cs6, Unknown Information Element)": any idea?
15:07.34KattyiCEBrkr: i mean mystery.
15:07.37iCEBrkrtype filename.txt | more
15:07.58KattyiCEBrkr: you clearly didn't understand.
15:08.02iCEBrkr:)
15:08.04KattyiCEBrkr: the file was created in linux, and moved to windows.
15:08.10iCEBrkrok?
15:08.14KattyiCEBrkr: being over 255 characters, windows garbled it.
15:08.15Cresl1nMimmus: yeah, it doesn't know how to parse whatever IE it's being sent (it being libpri I would assume)
15:08.21iCEBrkrKatty: Oh come'on
15:08.22*** join/#asterisk file (i=file@neutron.file-radio.com)
15:08.23KattyiCEBrkr: even when attempting to go into the insanely long named folder...
15:08.28KattyiCEBrkr: windows gave out errors.
15:08.38KattyiCEBrkr: unable to read folder because it was too long of a name.
15:08.38iCEBrkrKatty: Oh the filename is over 255 chars?
15:08.45KattyiCEBrkr: the folder, yes.
15:08.47iCEBrkrLOL
15:08.51*** join/#asterisk Poroto (i=raul@tesla.xmission.com)
15:08.59iDunnoiCEBrkr: I'm having fun building some java software that seems to (at random) decide that it doesn't like the JVM
15:09.05iCEBrkrKatty: I'm not gonna ask why the directory name is over 255 characters.
15:09.18KattyiCEBrkr: i wouldn't have an answer for you anyway.
15:09.18iCEBrkriDunno: Versions match?
15:09.25KattyiCEBrkr: i blame file completely.
15:09.48iCEBrkrKatty: Who uses names over 255 characters?  That's pretty retardo
15:09.50[TK]D-FenderKatty : omgplzshsrinkitkthxbibi :)
15:09.56iCEBrkr[TK]D-Fender: LOL
15:10.15[TK]D-FenderX.X
15:10.15Katty[TK]D-Fender: that would be my boss.
15:10.23KattyiCEBrkr: ^--
15:10.27[TK]D-FenderHis head can be shrunk too!
15:10.31iCEBrkrKatty: Fire him tomorrow.
15:10.32KattyiCEBrkr: and the owner of this company does whatever he wants.
15:10.36KattyiCEBrkr: i just clean it up ;)
15:10.43iCEBrkrKatty: Fire the owner while you're at it.
15:10.47KattyiCEBrkr: k
15:10.55*** join/#asterisk J4k3 (i=j4k3@dhcp-12-197-128-55.intrastar.net)
15:10.59iCEBrkrKatty: Take over and YOU run the place.  End of problems.
15:11.02*** join/#asterisk fugitivo (n=ajf@201.255.179.191)
15:11.37KattyiCEBrkr: i dont' want the job.
15:11.55iCEBrkrKatty: Ok, then continue to wrestle with gay filenames with batch files.
15:12.17KattyiCEBrkr: obviously.
15:12.21KattyiCEBrkr: oh, i get it.
15:12.26KattyiCEBrkr: you're playing mister fix it, right?
15:12.34De_Monlittleball i think what ice was trying to get across is that there may be a better approach to whatever you're trying to do. But since you haven't explained why you are doing what you are doing he can't help you find a better solution.  Quit trying to make your square box fit in your round hole and ask someone how to make the hole round?
15:12.54Kattyi keep forgeting this comes naturally to males.
15:12.54_Sam--im doing some testing with ASTlinux...is anyone familar with it? (using USB pen as Key Drive)
15:13.02iCEBrkrDe_Mon: Oh? he's still here?
15:13.06iDunnoiCEBrkr: same version, just doesn't work in one build environment and does in the other, same versions from same packages from same servers with same source from same svn, at that.
15:13.17iCEBrkriDunno: Whacked.
15:13.40De_MoniCEBrkr he's quit talking talkfully, but maybe he'll 'get it' this time
15:13.51iCEBrkrhehe
15:13.53De_Monshrug
15:14.03*** part/#asterisk frenzy (n=frenzy@80.255.63.30)
15:14.17iCEBrkrI'm all about helping someone, but if they're unwilling to listen, then F'm
15:14.26[TK]D-FenderKatty : Thats right.  The problem with men is we keep tring to figure out the core of the problem an FIX IT.  Whereas women really only want us to understand the emotional trauma caused by the problems itself and to tell them its all ok :)
15:14.35Katty[TK]D-Fender: yes.
15:14.40[TK]D-Fender:D
15:14.42Katty[TK]D-Fender: this batch script is causing me EMOTIONAL TRAUMA
15:14.46Katty[TK]D-Fender: hugpls.
15:14.47iCEBrkr[TK]D-Fender: haha wow, are you sure you're not a chick?
15:15.05*** join/#asterisk cianhughes (n=cian@87.192.36.98)
15:15.09littleballDe_Mon, i think this should be a common problem.(AGI is proved to be slow in project). So the suggestion/recommendation should be very simple like "Don't use AGI and change your design to avoid using AGI).
15:15.37[TK]D-FenderiCEBrkr : No, I just pay a high price for near omniscience :)
15:16.00KattyiCEBrkr: you really should read the women are from venus/men are from mars book.
15:16.11iCEBrkrKatty: Not all men seem to want to fix the cause of the problem.. Take are government for example,  it's not abortion clinics that are the problem, it's lack of sexual education and bringing our kids up to know better.
15:16.26iCEBrkrBut that's another argument for some other time.
15:16.34[TK]D-FenderiCEBrkr : Followed by Dave Barry's "Dave Barry is from Venus AND Mars"
15:16.38Kattywell maybe the parents should teach the children.
15:16.45iCEBrkrhaha
15:16.50Kattyand not rely on the government's educational system to do it for them.
15:16.50littleballi need to find out how iCEBrkr run php scripts thgouth agi, i want to compare what he did with mine.
15:17.02Kattyoh, taking responsiblity
15:17.03iCEBrkrKatty: Nonon, see, it's the abortion clinics that's the problem!!!
15:17.04Kattyi forgot
15:17.06Kattyparents don't like doing that.
15:17.10iCEBrkrand I blame Beavis and Butthead too.
15:17.44iCEBrkrI grew up on Wile E. Coyote and Roadrunner, but you don't see me dropping anvils on peoples heads.
15:17.47[TK]D-Fender"Life is a sexually transmitted disease.  If you can read this you'
15:17.54[TK]D-Fenderre already &^%ed"
15:18.05littleballhttp://www.asterisk-java.org/latest/tutorial.html
15:18.15iCEBrkrSo no. Not all men fix the core problem.
15:18.25Katty[TK]D-Fender: one of the reasons i'm a happy straight edge.
15:18.28iCEBrkrDuct-tape and band-aids.
15:18.45[TK]D-Fenderyeah iCEBrkr, all this talk about violence on TV reflecting on our daily lives is BS and I"ll kill anyone who says otherwise!!!!
15:18.54iCEBrkr[TK]D-Fender: LOL
15:19.11iCEBrkrKatty: you're severely twisted.
15:19.12*** join/#asterisk altrinidi (n=altrinid@151.9.212.56)
15:19.52iCEBrkrYa know, I should start making bumper stickers that picture an arrow with the word 'straight' overlayed on it.
15:19.55KattyiCEBrkr: also, you have /no/ idea.
15:20.15iCEBrkrBecause I want the world to know who I am and what I'm about
15:20.55[TK]D-FenderiCEBrkr : Yeah... straight out in "left field" ;)
15:20.57*** join/#asterisk riddlebox (n=james@24-171-11-166.dhcp.stls.mo.charter.com)
15:21.00iCEBrkrKatty: Not enough drama in your life? Become 'straight-edge' so you have something to bitch and preach about.. oh and to get upset over others actions cuz they're not like you.
15:21.12jbroomeand get bad tattoos
15:21.13iCEBrkr[TK]D-Fender: Hey, I kinda like it out there.
15:21.31Kattyyay, the batch script is still going!
15:21.38KattyiCEBrkr: haha.
15:21.43KattyiCEBrkr: i'm not about drama, dear.
15:22.01iCEBrkrKatty: Oh, then you're not truely straightedge
15:22.07KattyiCEBrkr: and if you knew /anything/ about me, you know i don't get upset about other people and what they choose to do (=
15:22.20KattyiCEBrkr: but that's ok, i don't really care what you think either.
15:22.24*** part/#asterisk altrinidi (n=altrinid@151.9.212.56)
15:22.30iCEBrkrlol
15:22.31KattyiCEBrkr: so have at it, cowboy!
15:22.35*** join/#asterisk altrinidi (n=altrinid@151.9.212.56)
15:22.46iCEBrkryeeeeeeeehawwww
15:23.09iCEBrkrKatty: Ok, fine fine, you're an exception to the 'rule'.
15:23.27riddleboxsweet the same two people talking again :p
15:23.29*** join/#asterisk bjohnson__ (n=bjohnson@jecinc.tor.istop.com)
15:24.02[TK]D-FenderiCEBrkr : Think of "Pirates of the Carribbean"'s saying "they're more like 'guidelines'" :D
15:24.09iCEBrkrha
15:24.28Katty[TK]D-Fender: you must be /ancient/
15:24.40[TK]D-FenderYeah... 30 .. *shudder*
15:24.43Katty[TK]D-Fender: zomg.
15:24.54iCEBrkr[TK]D-Fender: Your typical vegan are pretty whacked in the head.. Then all those straightedge kids.. omg.
15:25.02KattyiCEBrkr: yes.
15:25.07KattyiCEBrkr: we're Just Awful(tm)
15:25.12iCEBrkrBad taste I tell ya..
15:25.17Kattyor maybe smart.
15:25.18tRSSwhat does a 410 - Gone error mean? My asterisk gets this error when it routes an international call to another ip exchange
15:25.24Kattynot getting involved in all that complicated nightmare.
15:25.37iCEBrkrKatty: I guess, but can they please act a little normal?
15:25.39[TK]D-FenderI'm just the product of every bit of wisdom buried in the trash that I've watched, read, and listened in on.  Much like what I do at work, I turn ^#%$ into GOLD
15:25.48KattyiCEBrkr: normal is so last season.
15:25.50*** part/#asterisk Phrog123 (n=fmenard@cable-234-38.cgotr.infoteck.qc.ca)
15:25.54iCEBrkrShit
15:25.59iCEBrkrI'm always missing the trendy stuff
15:26.13Kattyyou clearly don't read enough Teen magazine.
15:26.19[TK]D-Fender"Normal is another word for 'boring as dirt'"
15:26.25Katty...that is a magazine, right?
15:26.31P4C0why I'm getting: NOTICE[6664]: res_musiconhold.c:507 monmp3thread: Request to schedule in the past?!?! ?
15:26.42[TK]D-FenderP4C0 : More fun with MPG123....
15:26.58[TK]D-FenderP4C0 : by and large ignore that if everything seems to work.
15:27.00P4C0[TK]D-Fender, I haven't set music on hold or anything like that
15:27.25[TK]D-FenderP4C0 : MPG123 gets loaded and jsut sits around trying to synch with *'s clock (or something like that).
15:27.41littleballiCEBrkr, when your agi call php scripts, i think a php interpreter (#!/usr/bin/php4) needed to be loaded and initialized, right?
15:27.42P4C0[TK]D-Fender, oks, thanks
15:28.06fugitivolittleball: right, it's slower than a compiled program
15:28.17*** join/#asterisk duckz (n=duckz@193.192.46.26)
15:28.21littleballthen i think this is not an issue for asterisk-java
15:28.42littleballif you try to invoke/start an external process, it just cannot work
15:29.12littleballthe time/resource consumed by the initialization of the external process/or connection is the killer
15:29.19fugitivolittleball: no
15:29.27P4C0I re-register like 10 times ok, and then wrong password...  this is driving me crazy
15:29.40fugitivolittleball: php will be slower, but AGI is still slow
15:30.30iDunno(something *will* work today, dammit)
15:31.03littleballfugitivo, i don't understand because AGI command itself is just a asterisk application. its(FastAGI) function is only write a few bytes data into socket.It should be much fast then decode/encoding etc
15:31.04KattyiDunno: i recommend hugging.
15:31.11KattyiDunno: it works wonders sometimes.
15:31.24*** join/#asterisk mhnoyes (n=mhnoyes@user-2ivfijq.dialup.mindspring.com)
15:32.23*** join/#asterisk Ariel_ (n=Ariel@adsl-068-157-125-248.sip.mia.bellsouth.net)
15:32.28littleballfugitivo, i think "AGI" in your mind is something in the external program. My "AGI" means the application name of asterisk (like Dial)
15:32.30iDunnoKatty: hmm - possibly, it'd help if there was at least some females in the office, though ;)
15:32.50KattyiDunno: you clearly need to get some then.
15:33.46iDunnoor go across the road to the pub ;)
15:34.05iCEBrkrfugitivo: give up man, he's not listening.
15:34.33KattyiDunno: all the good girls aren't in the pubs.
15:34.38_Sam--anyone fooled with ASTLinux using a usb pen drive as the key device?
15:34.48Katty_Sam--: someone has.
15:34.53Katty_Sam--: but i don't remember who anymore.
15:34.58iDunnoKatty: ahh - but I might be looking for one that's not so good...
15:35.06KattyiDunno: )=
15:35.13_Sam--its a cool setup, but it when it loads, it doesnt read the old config from the USB drive
15:35.15littleballthanks anyway, i will be careful
15:35.19KattyiDunno: gold diggers.
15:35.31KattyiDunno: and you shall be a tool and source of amusement for awhile.
15:35.48iCEBrkrKatty: That's a two-way street.
15:35.50KattyiDunno: perhaps not all of them though (=
15:36.02*** join/#asterisk blogz (n=fred@host-87-74-0-40.bulldogdsl.com)
15:36.13KattyiCEBrkr: awful.
15:36.27*** join/#asterisk gnosys (n=gnosys@griffin2.GnoSys.us)
15:36.29iCEBrkrIsn't it?
15:36.34_Sam--is there a way to log one asterisk CLI to another remote CLI?
15:36.45KattyiCEBrkr: quite.
15:37.12iDunnoKatty: seems unlikely that I'd attract any gold diggers - they generally go after people with money, don't they? :)
15:37.26blogzhi - can anyone help me? My suse 10 asterisk box was working perfectly until a power outage - now I can getting CAPI errors with an AVM c4 card: " > CAPI INFO 0x3302: Protocol error layer 2" when attempting to dial out. Can anyone help?
15:37.31KattyiDunno: unless you appear to have something else of worth.
15:37.32fugitivolittleball: when i talk about AGI(), i talk about the AGI() function itself, not the application run by AGI
15:37.49iCEBrkriDunno: just hang a $100 out of your back pocket.
15:37.59KattyiDunno: and i'm /not/ going there.
15:38.06iCEBrkriDunno: Or what Katty said. Stuff a sock in your pants :)
15:38.11iDunnohehehe
15:38.20littleballfutitivo, ok. i am trying to read the source code of asterisk to find out why AGI () is slow now
15:38.58KattyiDunno: post gifs of beard.
15:39.09gnosysIs there a way, from the * console, to show exactly which codec is in use for a bridged connection/conversation?
15:39.42Kattyi think sip show channels shows it
15:39.50*** join/#asterisk juanjoc (n=jcomella@222-32-235-201.fibertel.com.ar)
15:39.52Kattyunder the format column or something
15:39.57gnosysprobably for sip channels, but what about for IAX channels?
15:40.24iDunnoKatty: http://www.sommitrealweird.co.uk/photos/me/ <-- though it's slightly longer than that at the moment.
15:40.30Kattyiax2 show channels?
15:40.57iDunnodoesn't "show channels" also do it?
15:40.57KattyiDunno: gosh, that is a beard.
15:41.15gnosysshow channels doesn't, but iax2 show channels does.  Thanks, Katty. :-)
15:42.38gnosysIs "ulaw" a preferred codec for iax2 over a relatively high-bandwidth data connection, say for example, a T-1?
15:42.45*** join/#asterisk vmwarez (n=jjones@216.147.224.254)
15:42.56iCEBrkrgnosys: Depends on how many calls and bandwidth you wanna use :)
15:42.57fugitivognosys: you decide what you prefer
15:43.08iDunnoKatty: it's good for the cold weather - we get a fair amount of cold over here.
15:43.10iCEBrkrand of course the quality of sound
15:43.17loudyes, ulaw is the best for high bandwidth and quality.
15:43.17blogzis anyone else here running an AVM c4 card in the UK?
15:43.20KattyiDunno: i bet. my hair doubles as a jacket.
15:43.24gnosysI want high-quality audio in my phone-cons... with that criteria, would ulaw be a good choice?
15:43.29KattyiDunno: cause i have lots of it. (=
15:43.35iCEBrkrKatty: you have hippie hair?
15:43.40KattyiCEBrkr: mew?
15:43.42fugitivognosys: how many simultaneus calls?
15:43.52iCEBrkrKatty: down-to-your-ass hair?
15:43.57KattyiCEBrkr: hips.
15:44.00iCEBrkrgeeesh
15:44.03gnosysno more than 6 simultaneous calls
15:44.12iCEBrkrKatty: that's a pain to maintain
15:44.16KattyiCEBrkr: haven't you ever seen a picture of me?
15:44.21iDunnoKatty: :) is it dark?
15:44.23KattyiCEBrkr: not really.
15:44.26KattyiDunno: indeed.
15:44.39iCEBrkrKatty: I dated a girl that had hair down to her belt/waste
15:44.47KattyiCEBrkr: k
15:44.51iCEBrkrKatty: I can't recall seeing any pics of you
15:45.00KattyiCEBrkr: i see.
15:45.24iDunno\o/
15:45.33iCEBrkrha
15:45.58gnosysloud says that ulaw is best for high-bandwidth and trying to get high-quality audio.  Is that generally agreed upon here?
15:46.00iDunnoit's only taken 2 days - but it works now :)
15:46.16*** join/#asterisk wunderkin (i=kev@ip68-226-113-228.ph.ph.cox.net)
15:47.08fugitivognosys: yes, but it requires more bandwith, so be careful how many simultanous calls you get using ulaw
15:47.26*** join/#asterisk jcwunder (n=chris@ppp-82-135-64-130.mnet-online.de)
15:47.57*** join/#asterisk umay (n=chris@65-37-2-236.nrp2.roc.ny.frontiernet.net)
15:49.10[TK]D-Fendergnosys : ULAW is what the telco operates on typically.  Anything else is a loss in quality. If you can afford the bandwidth use it.
15:49.34gnosysthanks everyone, for the feedback on codecs.
15:50.11Kattypun intended.
15:50.14*** join/#asterisk caryon (n=caryon@p54A3C8E3.dip0.t-ipconnect.de)
15:50.42iCEBrkrHrrm, I've been using g726
15:51.44*** join/#asterisk lorinc (n=ang@caracas-3188.adsl.interware.hu)
15:52.01brad_msswg726 isn't bad, telcos use that for international calls usually, afaik ... to save bandwidth, etc
15:54.28gnosysSo Katty, for those of us who've not seen a picture of you, where would one look?
15:54.30iCEBrkrWorks for me.
15:54.39iCEBrkrgnosys: She looks like a gnome.
15:54.45gnosys:-)
15:54.50iCEBrkrComplete with a little red hat and all
15:54.51iCEBrkra/clear
15:54.51Kattygnosys: elsewhere.
15:54.54gnosysIsn't there a sone about that?
15:54.57gnosyssong
15:55.13gnosysLong-haired, over-fed, leaping gnome...?
15:55.16KattyiCEBrkr: i am /not/ short.
15:55.21iCEBrkr:)
15:55.42KattyiCEBrkr: just a little vertically challenged, kthx.
15:56.29blogzis anyone else here running an AVM c4 card in the UK?
15:56.37*** join/#asterisk trym (n=trym@062016209171.customer.alfanett.no)
15:56.37Kattygnosys: and no, i don't really look like a gnome.
15:56.51fugitivolike a kde?
15:56.57Kattyfugitivo: exactly.
15:57.00fugitivonice
15:57.28Cresl1nit's kind of different from a gnome
15:57.34fugitivokde is smarter
15:57.34meredyddtall, blonde, and European, aren't they?
15:57.37J4k3gnosys: Donovan!? haha
15:57.42Kattymeredydd: not quite.
15:58.02gnosyssong reference: http://www.stlyrics.com/lyrics/rememberthetitans/spillthewine.htm
15:58.14*** join/#asterisk popvoxdave (n=popvoxda@pcp0011694398pcs.longhl01.md.comcast.net)
15:58.19ManxPowerWOW!  Infrared IP video cameras are expensive.
15:58.35KattyiDunno: boggler.
15:58.42tRSScan someone help me with this problem: http://forums.digium.com/viewtopic.php?t=3439
15:59.05fugitivoManxPower: you can use a regular ip camera and an external infrared light
15:59.05iDunnoKatty: yup - it's useful, and better than being a letch (apparently)
15:59.09J4k3hah, wow, I always figured that was donovan.. goes to show how much I pay attention :P
15:59.20KattyiDunno: letch? mew?
15:59.30ManxPowerfugitivo, How?
15:59.43iCEBrkrManxPower: Most cameras can already see IR.
15:59.48ManxPoweriCEBrkr, Ah.
15:59.51iCEBrkrManxPower: Actually, all CCDs can see IR
15:59.52iDunnohmm - 4pm and I haven't eaten yet :/
15:59.54ManxPowerAny recommended models?
16:00.14littleballfugitivo, after simple going through a few source files, i can only see the latency could be due to agi() look for .agi files from hardisk. other things are quite normal (c function calls-pbx_exec() method). If using AGI over TCP, it is not needed to read the .agi file from disk.
16:00.20iCEBrkrManxPower: So all you need is a IR light source.  typically, it's a ring of IR leds around the outside of the lens
16:00.43iCEBrkr...and they make IR light-bulbs
16:00.44fugitivolittleball: and the tcp latency?
16:00.58iCEBrkrfugitivo: LOL
16:01.10fugitivo:)
16:01.15littleballagi only establish one TCP connection (only the first time)
16:01.25littleballit is socket and not http .
16:02.12littleballi am sure local tcp socket writing is very fast because the whole asterisk system using so many tcp/udp sockets.
16:02.31*** join/#asterisk stkn_ (n=stkn@2001:5c0:8d7e:2:250:bfff:fe55:8812)
16:02.35*** join/#asterisk ha1ock (n=ha1ock@gb.jb.104.119.revip.asianet.co.th)
16:03.06littleballmy experience is that the problem is due to the initialization of external process such java virtumal machine or php interpreter.....
16:03.26littleballthis is the reason of coming of AGI Over TCP
16:03.49fugitivolittleball: AGI is more complex than just calling an external script and passing parameters
16:03.51*** join/#asterisk classicx (n=classic_@gb.jb.104.119.revip.asianet.co.th)
16:04.06*** join/#asterisk kiwnix (n=egarcia@82.158.152.212)
16:04.26littleballmaybe. i need to read more...
16:04.30iCEBrkrfugitivo: just put him on ignore
16:04.38iCEBrkrIt's not worth your bandwidth
16:04.45littleballAGI over TCP never call external script directly
16:05.27*** join/#asterisk [B]aby18 (n=AcMeMeT@85.108.144.188)
16:05.43fugitivolittleball: are you really using AGI capabilities or just calling a script?
16:05.47littleballiCEBrkr, if you really have some idea why not just tell me where is the bottleneck from the source code. I think you have the soruce code right?
16:06.08littleballusing AGI over TCP.
16:06.21littleballno, i don't call script
16:07.06fugitivolittleball: the "thing" you call with AGI, is really using AGI capabilities?
16:08.07littleballsorry? called with?
16:08.10fugitivoit seems to me that you're not really using agi
16:08.35littleball_X.,3,AGI(agi://localhost/mobmeee.agi?connected=footrue&seconds=foo${allocatedcallingseconds}&accountcode=foo${ACCOUNTCODE}&src=foo${CALLERIDNUM}&dst=foo${legb})
16:09.00fugitivowhat does mobmeee.agi do?
16:09.01littleballthis is what i used
16:09.07*** join/#asterisk docelmo (n=docelmo@static-71-251-95-4.tampfl.fios.verizon.net)
16:09.27littleballonly read the data from TCP socket . that is all
16:09.49*** join/#asterisk Twister (n=Administ@216.30.232.106)
16:09.56littleballread means parse "&"
16:10.07littleballand parameter and value pairs
16:10.16littleballthat is all agi done
16:10.17fugitivodoes it do anything with asterisk? are you using any AGI lib inside that .agi?
16:10.18ManxPowerand you have localhost listed in /etc/hosts ?
16:10.27littleballno
16:10.32fugitivolittleball: no?
16:10.37fugitivolittleball: then why are you using agi?
16:10.54littleballsorry, _X.,1,AbsoluteTimeout(${allocatedcallingseconds})
16:10.56fugitivoit's just a call to an external program
16:11.02littleballSetVar(legb=${EXTEN})
16:11.06littleballthen call agi
16:11.55littleballwhen a call is coming, the system answer and then call agi
16:11.59fugitivolittleball: agi is used for interaction with asterisk, if your program doesn't interact with the call, then there's no need for agi
16:12.17ManxPowerfugitivo, it would reduce the overhead of fork.
16:12.22ManxPowerlittleball, how much delay is there?
16:12.30littleballi cannot find any delay
16:12.36littleballit is really fast.
16:12.43ManxPowerSo what is the problem?
16:13.00fugitivothere's no problem
16:13.01littleballbecause iCEBrkr say it is very slow
16:13.11littleballand agi cannot be used in big system.
16:13.16Kattythere is no problem - we just like the story!
16:13.20littleballso i want to confirm whether it is true
16:13.28ManxPowerlittleball, FastAGI does not have the overhead of normal AGI
16:13.44littleballYes, i am using FastAGI(AGI over TCP)
16:13.56ManxPowerNormal AGI, where the script is run each time AGI is called will be slow.  FastAGI was created to remove that overhead.
16:14.43ManxPowerRemember, AGI is still fast, it's just if you want to run the agi several times per second you should use FastAGI
16:15.14littleballManxPower, yes, this is what i did. but iCEBrkr said it cannot be used in big system. In my system i tested 5 agi per/second. it is normal.
16:15.15azziehow many simultenious calls * can handle if it does not do codec conversion? on a decent pc
16:15.23ManxPowerlittleball, there is nothing special about AGI .vs. running any application many times per second.
16:15.36gnosysWhen I use SET CALLERID <number> to set a number for an outbound call, does this allow me to also set the caller text that accompanies the number?  Or would that be done by setting a different variable?
16:15.43ManxPowerazzie, Excelent question!  Let us know what you find.
16:15.47*** join/#asterisk opus_ (n=opus@dahphish.org)
16:15.51opus_anyone here use junction networks?
16:15.54wasimazzie: '000s
16:16.15littleballManPower, yes. becuase normally people run java/php interpreter, so it is very slow. It cannot run 1 time within one second becuase the initialization overhead is huge
16:16.16azzieManxPower :) i'm sure there a people who know the answer :)
16:16.16ManxPowergnosys, generally telcos ignore the calleid name and sets it based on the telco callerid info for the number you send.
16:16.28ManxPowerazzie, No.
16:16.29*** part/#asterisk cfh (n=luca@82.193.23.6)
16:17.00ManxPowerSince it depends on the interface, NAT, Dial options, disk speed, amount of logging, AGIs called, etc.
16:17.07[TK]D-Fendergnosys : What version of * are you using?
16:17.13gnosys1.2.1
16:17.17ManxPowerDigium says you can do 8 T-1s ona Dual Xeon with no transcoding.
16:17.20*** join/#asterisk SERGEUS_ (n=s@195.112.98.13)
16:17.21azzieManxPower, strip all that. Just calls with noreinvite.
16:17.31[TK]D-Fenderthen "Set(CALLERID(name)=Your Name)
16:17.44azziecool
16:17.50ManxPowerazzie, Um, why not invites?
16:17.56littleballI just DIY one server at about US$1600.
16:18.09littleballit runs 1.2.1 very well
16:18.12iDunnohmmm - pasty :)
16:18.13littleball4E1
16:18.14azzieManxPower, billing and other stuff...
16:18.21gnosysSo if I wanted to set both the callerID number and callerIDName (my words), then I would do SET CALLERID <number> and what you just wrote, [TK]D-Fender?
16:18.25*** join/#asterisk Vijay (i=Vijay@203.122.28.109)
16:18.49opus_is junction down?
16:19.04ManxPowerdefine "other stuff".  Remember IAX2 cannot keep correct CDRs with "reinvites".  Other protocols can provide correct CDRs with reinvites enabled.
16:19.07*** join/#asterisk Jay (i=Jay@192.153.153.179)
16:21.11azzieManxPower, it's not the point. My concern is that with reinvites has to switch RTP (does it?) and a PC can't handle too many kernel/process context switches per second
16:21.38[TK]D-Fendergnosys : 2 steps.  Set(CALLERID(num)=1234567) and Set(CALLERID(name)=Your Name)
16:21.47ManxPowerazzie, Reinvites removes audio from the server.  With reinvites the two end points send their audio directly to each other.
16:22.03gnosysThanks [TK]D-Fender
16:22.17ManxPowersignalling, of course, still goes thru the server.
16:22.28ManxPowerThis applies to any protocol that uses RTP (i.e. NOT IAX2)
16:22.52azzieso if your super-pc is capable of doing 3-5K context switches, it's just few hundreds standing calls
16:23.05ManxPowerHuh?
16:23.12ManxPowerreinvites LOWER CPU usage
16:23.13*** join/#asterisk florz (n=florz@2001:1a50:503c:0:0:0:0:1)
16:23.13azziewith noreinvites
16:23.56ManxPowerazzie, Asterisk is multi-threaded and that reduces context switching, as opposed to FORKing
16:24.42azzieManxPower, kernel-to-userprocess context switches. Get UDP to kernel, switch processor to user mode for *, switch it back to kernel to send UDP out
16:25.00*** join/#asterisk kio (n=kio@ool-4577adba.dyn.optonline.net)
16:25.08ManxPowerazzie, Ah.  Yes.  The same issues with any networked application like a web server
16:26.06kippihey
16:26.18kippiis John from diguim around?
16:26.52azzienobody cares if a HTML or GIF sticks in kernel buffer for hundreds of milliseconds because CPU is "busy"
16:27.22azziewith RTP it really matters
16:28.07tRSScan someone help me with this problem: http://forums.digium.com/viewtopic.php?t=3439
16:29.13*** part/#asterisk cpm (n=Chip@pdpc/supporter/sustaining/cpm)
16:30.23gnosys[TK]D-Fender: it seems that ManxPower's comment about telcos ignoring the callerIDName (my word) is true.  Either that or my syntax for setting the callerIDName is not quite correct.  I did use exactly what you suggested.  Do you suppose that the problem lies with the telcos ignoring my setting?
16:30.29ManxPowertRSS, C is either in the wrong context in sip.conf or the internal dialplan for device C is not set correctly.
16:30.30*** join/#asterisk ast_freak (n=jesse@68-112-134-195.dhcp.stcl.mn.charter.com)
16:30.55ManxPowergnosys, I would be VERY VERY suprized if the telco passed your calleridname
16:31.08ManxPowerIt's normally the TERMINATING telco that does this.
16:31.15gnosysI can set the number to whatever, but it seems that the telcos just use that number to look up the person who subscribes to that number, and fills in the text with whatever they have associated with that number in their database.
16:31.18ManxPowerYour telco may or may not pass the name you set.  Most don't.
16:31.27tRSSManxPower: C's dialplan says: if recieved is XXXXXX. then send it to the PSTN connection
16:31.28ManxPowergnosys, Yes, that is the way it works.
16:31.39gnosysok.  Thanks, ManxPower.
16:31.41De_MoniCEBrkr I found abort73.com to be a pretty informative abortion website.
16:31.54tRSSManxpower: what do you mean that C is defined in the wrong context?
16:32.12iCEBrkrlol
16:32.13azziegnosys, and yes, it's pain in the ass to update thouse CNAM databases if you're not a big telco ;)
16:32.17tRSSManxPower: if you want, I can paste my sip.conf on pastebin?
16:32.17iCEBrkrDe_Mon: that's so 2hrs ago
16:32.27kippihas anyone got a url for a good setup for incoming calls?
16:32.33ManxPowertRSS, The sip.conf section for device C has a context= that points to a context in extensions.conf that is not allowed to dial international.
16:32.34iDunnoit's all working :)
16:32.44ManxPowertRSS, Sure.  I doh't guarntee I can look at it.
16:32.45gnosysazzie: not sure I follow?  What CNAM databases?
16:32.46*** part/#asterisk opus_ (n=opus@dahphish.org)
16:33.03*** join/#asterisk spunz_ (n=spunz@h081217096096.dyn.cm.kabsi.at)
16:33.15tRSSManxPower: Let me paste and see if you can help me. but thanks for the help so far! :)
16:33.15ManxPowergnosys, CNAM database is the telco database of name/number pairs (not technically correct, but close enough for this)
16:33.17azziegnosys, it's that database with caller names
16:33.25De_MoniCEBrkr I know :( had to scroll up to see if whats-his-face said anything inteligable and couldn't resist
16:33.46iCEBrkr:)
16:34.17gnosysSo... would I update that???  I'm confused?  For whom is it a pain in the ass?  The telco?
16:34.38ManxPowergnosys, if you own the number then call your telco to update the callerid name info
16:34.52gnosysOh.  I see.
16:34.59De_Monhearing Katty imply it was smart to 'not care' about the mass murder of babies deserved comment, the difference between blissful ignorance and being smart is just way too huge
16:35.01gnosysThanks fellas.
16:35.20*** join/#asterisk tecnico (n=tecnico@user-24-236-120-2.knology.net)
16:36.22ManxPowerDe_Mon, I think the key is that many people (myself included) don't believe and embryo is a baby.
16:36.33fugitivothat's retarded
16:36.42_Sam--i have one asterisk box talking to another one via IAX.....one box is a gateway.  when i call through the gateway from the other Asterisk IAX box i get:   -- IAX2/teliax-16393 answered IAX2/astlinux-2
16:36.42_Sam--<PROTECTED>
16:36.42_Sam--<PROTECTED>
16:36.42_Sam--<PROTECTED>
16:36.46*** join/#asterisk brad_mssw (n=brad@shop.monetra.com)
16:36.57_Sam--what happens there?
16:36.58*** join/#asterisk harryvv (n=none@S010600a0c93f6f7e.vs.shawcable.net)
16:37.07fugitivo_Sam--: happens that you need to use pastebin
16:37.12_Sam--heh sorry
16:37.18ManxPower_Sam--, I suspect either NAT or codec issues
16:37.26_Sam--they are both on the same LAN segment
16:37.39ManxPowerthen it's proly a NAT issue
16:37.42_Sam--but the non gateway box doesnt have an external (internet) ip
16:37.46_Sam--the gateway box does
16:38.05ManxPower"transfer" would mean "have teliax and astlinux talk directly to each other"
16:38.13tRSSManxPower: here is the pastebin URL: http://pastebin.com/482987
16:38.17_Sam--how do i prevent that?
16:38.23_Sam--i want the gatway asterisk in the middle
16:38.56ManxPower_Sam--, so you don't want IAX2 transfers?
16:39.13_Sam--not for what im doing now, no
16:39.33ManxPowertRSS, I only see 1 device set in sip.conf.
16:39.42De_MonManxPower correct, which is usualy (you man be an exception but) a belief rooted in utter stupidity and purposeful ignorance on the subject.
16:39.52ManxPower_Sam--, notransfer=yes in iax.conf
16:40.05_Sam--THANKS.
16:41.26zoa2hey ho sam
16:41.35tRSSManxPower: that is not the device. See if I have to make international calls, my dial would look like: exten => _011.,1,Dial(SIP/${EXTEN}@uszultys,20,rt). I have set the polycom device as another context
16:41.53_Sam--hey zoa!
16:42.07ManxPowertRSS, the polycom cannot dial international?
16:42.40tRSSManxPower: any extension on asterisk can not dial international. I can add more of sip.conf if you wish?
16:42.42ManxPowertRSS, BTW, for get EVERYTHING from anyone that tells you to use "r" option on Dial.
16:42.44_Sam--ManxPower:  now i get:  Attempting native bridge of IAX2/astlinux-3 and IAX2/teliax-16390    but it sounds like ass
16:42.48*** join/#asterisk florz (n=florz@2001:1a50:503c:0:0:0:0:1)
16:43.00_Sam--what should i be checking?
16:43.20*** join/#asterisk rikstah (n=rick@87.113.11.91.bbplus.pte-ag1.dyn.plus.net)
16:43.22tRSSManxPower: I will make a note of it
16:43.49*** join/#asterisk zishanov (n=mail@HSE-Toronto-ppp3489695.sympatico.ca)
16:44.08ManxPower_Sam--, the message prolly is talking about internal "native bridge", which I don't know much about, rather than a native transfer
16:44.21_Sam--zoa2:  when you getting me a new fisk? :)
16:44.30zoa2aha
16:44.33zoa2good question
16:44.38zoa2the 1.27 is out
16:44.39ManxPowertRSS, you need an entry in sip.conf for any SIP device that will be talking to Asterisk
16:44.52zoa2but dont have any pro's and the guy is on vacation till wednesday
16:45.46tRSSManxPower: I have updated the pastebin page: http://pastebin.com/482993
16:45.49P4C0is it normal that sip show peers shows all the peers with the same port?
16:46.09zishanovwhats the cause of error :line0: Unable to open master device '/dev/zap/ctl'
16:46.22zishanovP4C0, that is normal
16:46.34*** join/#asterisk JunK-Y (n=junky@67.71.110.21)
16:46.35zishanovP4C0: howz your voice mail now
16:46.46P4C0zishanov, but I need to change the port if I want to run a sip phone on the same machine right?
16:46.50P4C0zishanov, fine, thanks
16:47.01ManxPowertRSS, EVERY entry in sip.conf for type=user or type=friend should have a context=something which is the [something] section of extensions.conf
16:47.34zishanovP4C0: how did you fix it
16:47.34ManxPowerzishanov, zaptel drivers not being loaded.
16:47.58ManxPowerP4C0, you don't want to run a SIP phone on the same machine as Asterisk
16:48.11tRSSManxPower: i thought, if I don't, it will take the value from the default context
16:48.12zishanovManxPower: I did make clean, make linux 26, make install on zaptel. What else I need to do
16:48.13P4C0ManxPower, why? :(
16:48.33P4C0zishanov, changed the seccions from name to extensions in sip.conf
16:48.46ManxPowerP4C0, for one thing you should not run graphics applications on an Asterisk server
16:49.16P4C0ManxPower, cpu? no prob
16:49.55ManxPowersrt, yes, but you don't want that to happen for security reasons.  put context=INVALID in [general] then put the correct context= in each other sip.conf section
16:50.17ManxPowerP4C0, no, interrupt lacency.
16:50.25ManxPowerlatency, that is
16:50.29P4C0ManxPower, oks
16:50.42*** part/#asterisk JunK-Y (n=junky@67.71.110.21)
16:50.50shido6or create a default context with something referring to your contact number to sign up (wasting b/w or bringing in new sales?)
16:50.55zishanovP4C0, I have two of my sip phones working with names in their sections. But in this case you have to define in extensions.conf, in [global], JOHN=SIP/201 etc
16:51.16P4C0yes
16:51.43*** join/#asterisk bn-7bc (n=bjarne@87.252.64.25)
16:53.27bn-7bccan anyone telle me how to set up asterisk for use wit me account from 2phone (norwegian voip provider) ther costumer service seem to be ower worked atm?
16:55.11iCEBrkrGeesh, I can spend all day on digg.com
16:55.22wunderkinhmm im trying to figure out how to get cause codes off of my pri.. i specifically want to know if a number is disconnected.. i tried to dial a number that i know is disconnected and of course it doesnt answer the line.. after awhile it finally hangs up with cause 16.. do i need to change the inband/outofband indication thing or something?
16:55.47iCEBrkrwunderkin: There's a list of cause codes on the wiki somewhere
16:55.48zishanovManxPower, why would zaptel not being loaded, what have I missed
16:55.49iCEBrkrI've seen it
16:56.08wunderkiniCEBrkr: yeah, im looking at it but it is only returning 16 which is normal
16:56.11*** join/#asterisk mog_work (n=mogorman@gateway.digium.com)
16:56.13iCEBrkrhrrrm
16:56.34fileapparently it's my birthday on January 1st and I get free local calling all day...
16:56.38filewhich is odd, because it was on October 6th
16:56.55Kattywhich is the day before my birthday.
16:57.21filevery very strange
16:58.00ManxPowerzishanov, perhaps you forgot to set the system to load it on boot.  Asterisk doesn't have magical config gnomes, you have to set it yourself
16:58.27iCEBrkrManxPower: What about magical config KDEs?
16:58.44shido6if you build it it might run
16:58.55tRSShow can I remove the first digit off the number dialled? e.g. user dials 1011XXX but I want to remove the first digit and send 011XXX
16:59.06shido6${EXTEN:1}
16:59.13tRSSshido6: thanks!
16:59.33ManxPowertRSS, Dude, read the Asterisk book.  Also read README.variable and in fact read ALL the files in the docs directory
16:59.36iCEBrkrHey the answer to that can be found on the Wiki
16:59.41zishanovManxPower, I ran make config. When I see lsmod, it shows that zaptel is running
16:59.42ManxPowershido6, you sure are easy today.
17:00.03ManxPowerzishanov, the CARD or DEVICE driver
17:00.05iDunno*yay*! beer after work!
17:00.14iCEBrkriDunno: Mmmmbeeer
17:00.18ManxPowerzaptel doesn't do you any good if it can't talk to a card driver
17:00.37filegah
17:00.42shido6in the root dir of your asterisk dir there is a "doc" dir
17:00.43shido6have at it
17:00.57filewhy am I getting 0 rows...
17:01.05Kattyfile: because you need a cookie.
17:01.08Kattyfile: obviously.
17:01.08iCEBrkrfile: you suck at sql?
17:01.11fileI do ,I do
17:01.18fileiCEBrkr: haha
17:01.20iDunnofile: ahh - because the table has no data in it :)
17:01.29iCEBrkrKatty: ya know, if you keep handing out all these cookies, we're gonna get all big and fat.
17:01.34KattyiCEBrkr: k
17:01.39iDunno(it can't be an SQL error, no one would ever be that silly *grin*)
17:01.42iCEBrkrKatty: and it's gonna be ALL YOUR FAULT
17:01.43JonR800or your query is wrong :)
17:01.46KattyiCEBrkr: k
17:02.03iCEBrkrKatty: Don't you care?!
17:02.08KattyiCEBrkr: k
17:02.16iCEBrkrDAMNIT WOMAN!
17:02.21Kattywatch your tounge.
17:02.39iDunnoiCEBrkr: look, if you're worried about getting fat, STOP ACCEPTING THE DAMN COOKIES ;)
17:02.51iCEBrkriDunno: but but but... I can't help it.
17:02.54fugitivoor get another job
17:03.03iCEBrkr8()
17:03.07iDunnoiCEBrkr: ahhh - lack of self control? awwww ;)
17:03.22N9URKany milk discussions today?
17:03.30iCEBrkrN9URK: Nope, cookie time.
17:03.36De_MonManxPower the [general] context= doesnt apply if it's excluded from the entry?
17:03.41fugitivoyes, milk is evil, mysql is evil, and microsoft is evil
17:03.44iDunnoiCEBrkr: ahh - but with cookies, don't you need milk?
17:03.47N9URKI think I will have some Newman-Os
17:03.51filefugitivo: I'm evil too
17:03.53N9URKthey are yummy
17:03.56iDunnofugitivo: 2 out of 3 ain't bad ;)
17:03.56fugitivofile is evil
17:04.00*** join/#asterisk caryon_ (n=caryon@p54A3C8E3.dip0.t-ipconnect.de)
17:04.05De_Monnevermind...
17:04.11iCEBrkriDunno: Yeah, cuz cookies and beer just isn't right
17:04.22N9URKwhat about cookies and hot tea?
17:04.28fugitivoi like tea
17:04.32iDunnoN9URK: that's good, or with coffee.
17:04.45Kattyfile: mew?
17:04.45fugitivogreen tea
17:04.50iDunnothough, bestest thing in the world is strong coffee and a mint aero to dunk in it :)
17:05.03N9URKgreen tea is awesome
17:05.07fileKatty: your mewish level is low today
17:05.15Kattyfile: quite.
17:05.18fileoh cool, download is done
17:05.20fugitivored tea is nice too
17:05.28kippiwill exten => 9X.,1,Dial(Zap/g1/${EXTEN:1})
17:05.36N9URKI slept late today.  I am going to be in a rush to get my client's project done.  I think I am getting sick
17:05.49kippiwill this ring any nummber after the 9 ?
17:05.50fugitivoN9URK: drink tea
17:06.07N9URKGood advice, fugitivo
17:06.48fugitivokippi: correct syntax would be _9X, for only one digit after the 9 or _9. for every digit after the 9
17:06.54N9URKKatty, what kind of salt do you use?
17:06.59fileKatty: >.<
17:07.15fileKatty: feels like heaven!
17:07.45wunderkini tried priindication = outofband and inband but neither seem to change this.. i still only get cause 16 on disconnected numbers... hmm?
17:07.51iDunnofugitivo: _9. is EVIL! should be at least _9X.
17:08.33iDunno(my opinion only, of course, and my opinion is worth bugger all :)
17:08.37kippiWith _9X. I am getting a 404 error
17:09.10[TK]D-Fenderkippi : Pastebin your Zapata.conf & Extensions.conf.
17:09.11[TK]D-Fender~pb
17:09.15jbotsomebody said pb was a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca
17:09.26file[TK]D-Fender: !!!
17:09.30ManxPoweriDunno, No.  _. is evil.  _9. is does have not have the same problem.
17:09.33[TK]D-Fenderfile: !!!
17:09.58ManxPower_9. is silly and stupid, but isn't evil.
17:10.03[TK]D-FenderManxPower : 9. *is* evil.  On analog channels gives UNRESTRICTED dial-tone.
17:10.16[TK]D-FenderYeah I guess STUPID works ;)
17:10.29*** join/#asterisk schuyler_ (n=schuyler@gateway.digium.com)
17:10.34ManxPower[TK]D-Fender, Only if you use 9 for your outside line indicator.
17:10.43ManxPowerMost of the world uses 0 for that.
17:10.44fugitivoMYSQL AND FILE ARE THE ONLY EVIL HERE
17:10.51ManxPowerBut I see your point.
17:11.17ManxPower[TK]D-Fender, I sometimes forget about that.  Our LD company makes sure that no toll calls happen without an auth code.
17:11.37ManxPowerI let them deal with it 8-)
17:16.14kippihttp://pastebin.com/483032 and http://pastebin.com/483044
17:18.13*** part/#asterisk SwK[Work] (n=SwK@64.89.118.139)
17:18.19*** join/#asterisk SwK[Work] (n=SwK@64.89.118.139)
17:19.50harryvvWho can provide same day service if I buy DIDs?
17:20.24[TK]D-Fenderkippi : What context are your phones using?
17:20.52kippihow can I double check?
17:21.11[TK]D-FenderSIP.CONF
17:21.18[TK]D-Fenderlook at the context in each phone entry
17:21.42[TK]D-FenderOh, I found your problem....
17:22.05[TK]D-Fender505. include=outgoing <- should be "include => outgoing"
17:22.21[TK]D-Fendermissed a ">"
17:22.29[TK]D-Fenderfix, reload, test....
17:22.31kippiwhere is that?
17:22.42[TK]D-Fenderline 505 in your extensions.conf pastebin
17:23.12harryvvTK, seen a case of Putting in info of [Inbound] that it interfeared with outbound calls? I would get a disconected notice on my ip500. I soon deleted that DID configuration and it returned to normal.
17:23.45[TK]D-Fenderharryvv ... I did not follow that AT ALL :/
17:23.58harryvvMaking any local call would get a disconect busy tone on my ip500
17:24.10[TK]D-FenderThat makes no sense...
17:24.19harryvvI know
17:24.20fileprobably some configuration issue... sip debug tells you all
17:24.33[TK]D-FenderMaybe being forced to use 10 digit dialing?
17:24.54harryvvIt was comming from my asterisk box...so removed the [inbound] and the following DID information.
17:24.55[TK]D-Fenderor is it the phone itself that is generating the tone?
17:25.08[TK]D-Fenderharryvv : ~pb
17:25.26harryvvTK, my phone system works 100% without the configuration for the DID info.
17:25.32harryvvkinda odd
17:27.12[TK]D-Fenderpastebin the non-working version so I can get it straight.
17:27.21[TK]D-Fenderkippi : well?
17:28.30*** join/#asterisk TheCops (n=henri@206-248-136-187.dsl.teksavvy.com)
17:28.35TheCopshey [TK]D-Fender
17:28.38[TK]D-Fendery0
17:28.40TheCopsI just received the IP501 phone!
17:28.42TheCopswhouhou!
17:28.44[TK]D-FenderCool
17:28.46TheCopsSo cute!
17:28.50TheCopsand so clear!
17:28.52[TK]D-FenderIts pretty nice.
17:28.58[TK]D-FenderAll the Poly's are.
17:29.09*** join/#asterisk eKo1 (n=bernd@63.245.57.70)
17:29.14ManxPowerTheCops, I think in MA you are allowed to marry your Polycom.
17:29.41harryvvonly thing i dont like about the polycoms is no backlight
17:30.15*** join/#asterisk chapeaurouge (n=chap@85.201.81.201)
17:30.40TheCops[TK]D-Fender, hey, I've got MicroBrowser XML options in web administration
17:30.52[TK]D-FenderTheCops : You do?  Thats odd.....
17:31.05[TK]D-FenderCould be present but non-functional for your model
17:31.10ManxPowerTheCops, They might be fake.
17:31.14TheCopsIt's asking me Main Browser Home and proxy
17:31.22TheCops[TK]D-Fender ho ok
17:31.39[TK]D-Fenderharryvv : The only SIP phone I've seen with backlights are Grandstream's, and that is their only redeeming value!
17:31.50ManxPowerTheCops, don't expect it to work, but you can try it.  one of the reasons for the switch from 500 to 501 was to add more memory to add more features
17:31.56N9URKIs anyone here using *@home?
17:32.02ManxPower[TK]D-Fender, SIPura does too.
17:32.19TheCops[TK]D-Fender, where's is the text configuration
17:32.20[TK]D-FenderN9URK : Yes, but they are smart enough to remain silent about it;)  Try #amportal.
17:32.22TheCopsI hate web interface
17:32.26TheCopsand always lost
17:32.43ManxPowerTheCops, download and read the ADMIN GUIDE
17:32.54N9URKok, I'm not using just a little curious about it; I don't think I will use it
17:33.06[TK]D-FenderTheCops : http://www.freedomphones.net/polycom/files/
17:33.23[TK]D-FenderTheCops : and grab the version that matches whats loaded on the phone to start.
17:33.25*** join/#asterisk Assid (n=assid@203.115.64.59)
17:33.27ManxPowerWhat I did when I started with the polycoms was configure 1 phone the way I wanted it via the web interface, then allow the phone to upload it's config to the server, then used that as a basis for the central config stuff
17:33.48[TK]D-FenderTheCops : And yeah download the Admin guide for your SIP version and get reading.  Lots of fun stuff.
17:33.50kippiis there an echo test i can do?
17:34.13Assidpolys rock!
17:34.25[TK]D-FenderManxPower : I found the defaults pretty decent in the support pack and modded my way into it.  I didn't trust the phone generated one myself.
17:34.38[TK]D-Fenderkippi : Between what?
17:34.51kippito test the echo on the card and lines
17:35.12[TK]D-Fenderkippi : hmm.. just try calling someone I guess....
17:35.16ManxPower[TK]D-Fender, as I said, I used it as a BASIS, mostly to learn.
17:35.30ManxPowerkippi, call someone on an ANALOG phone.
17:35.38[TK]D-FenderManxPower : I suppose.  Thats the "reverse engineering" approach anyways.
17:35.39ManxPowernot a cell phone, not a PRI, not a VOIP line.
17:35.42kippiwhats a really quick script that will see that its an DID coming in and send that to an extension?
17:35.50wunderkinanyone know of any phone numbers that are disconnected? ;)
17:35.52[TK]D-Fenderkippi : What ManxPower said...
17:36.06[TK]D-Fenderkippi : You are running a PRI?
17:36.06ManxPowerkippi, exten => mydid,1,Dial(SIP/wanker)
17:36.30Ariel_http://blog.tmcnet.com/blog/tom-keating/videos/a-very-Cisco-Christmas.wmv
17:36.41Ariel_You just have to see this
17:36.54Ariel_great sell for voip....
17:37.10ManxPowerAriel_, how big is it?
17:37.36Ariel_ManxPower, don't know size. But it's a great spoof
17:37.42[TK]D-Fenderkippi : So I take it your outside calls are working now?
17:38.14De_MonAriel_ i'm not convenced
17:38.29Ariel_De_Mon, hehe
17:38.34*** part/#asterisk Enderson (n=enderson@smtp.gentoo.org)
17:39.18De_Monthe day a girl invites me to come over and play with her router...
17:39.47TheCops[TK]D-Fender, I'll see you in a week, I just printed the 166page of the Admin guide
17:39.50TheCops*g*
17:40.19[TK]D-FenderDe_Mon : Yeah I can picture you getting slapped asking to "open a socket and try some plug-ins"...:/
17:41.16eKo1hahaha
17:41.28[TK]D-FenderAriel_ : That video was just ...dumb...
17:41.43Ariel_Yes but it's so stupid that it's funny
17:41.54De_Monnah, it was just stupid
17:42.11ManxPowerAriel_, For some reason I can't view it.
17:42.53*** join/#asterisk chapeaurouge_ (n=chapeaur@85.201.81.201)
17:43.09[TK]D-Fender*boing*
17:43.15*** join/#asterisk chapeaurouge (n=chapeaur@85.201.81.201)
17:43.41[TK]D-Fender*boing*
17:43.41*** join/#asterisk saftsack (n=oliver@p54A7D1AA.dip.t-dialin.net)
17:54.42*** join/#asterisk licued (i=licucude@ool-44c784a0.dyn.optonline.net)
17:54.51*** join/#asterisk davidw (n=davidw@apache/committer/davidw)
17:54.54wunderkinheh..
17:55.11davidwhi all... any idea why asterisk would try calling me twice when using the call file spool?
17:55.25davidwMaxRetries is 0
17:56.07*** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net)
17:56.10harryvvthats interesting
17:57.47ManxPowerdavidw, Do you have an exten => _. pattern in extensions.conf?
17:58.14davidwexten => s,1,Wait(3)
17:58.14davidwexten => s,2,TxFax(${FAXFILE}|caller)
17:58.16davidwis what I've got
17:58.33davidwwell, then it hangs up, but you get the idea
17:58.49ManxPowerbecause exten => _. will cause everything to be run twice.
17:59.15davidwit's definitely hitting the TxFax
17:59.18ManxPowerone when you dial the digits and once when you hang up (extension "h" will be matched by _.
18:00.18davidwthere is no _ in the [sendfax] context
18:00.22davidwjust 's'
18:00.30ManxPowerweird
18:00.33davidwyeah
18:02.20Assidhrmm.. next up on my play list.. rxtxfax
18:02.38davidwtxfax seems a bit... mmmm it needs more hacking IMO
18:02.55*** join/#asterisk _DAW (n=bob@adsl-156-94-42.msy.bellsouth.net)
18:03.16davidwI wrote the author yesterday, but he hasn't responded yet
18:03.37Assidwell.. takes time
18:05.05davidwharryvv has a good testing idea: it only dials once when going to SIP
18:05.25*** join/#asterisk razu_ (n=razu@ip220.cab17.mus.starman.ee)
18:05.59davidwweird:-/
18:06.30wunderkinhmm how about this.. anyone with a pri here that could do a quick favor for me?
18:06.45wunderkin(LD PRI)
18:06.49riddleboxcan you access your voicemail from outside, like when you call in and hear your greeting, can you press * to get to the login prompt?
18:06.56*** join/#asterisk chapeaurouge (n=chapeaur@85.201.81.201)
18:07.26ManxPowerriddlebox, see "show application voicemail"  pay special attention to the "a" and "o" extens listed
18:07.32[TK]D-Fenderriddlebox : Its all in how you set up your dial-plan.
18:07.43[TK]D-FenderMake an exten in your menu that will allow you access to voicemailmain
18:07.53riddleboxok
18:08.07davidwok... outgoing voice call over CAPI gets done twice as well
18:08.09davidwvery weird
18:08.33[TK]D-Fenderriddlebox : And defiantely set up a STDEXTEN style macro and use the "a" exten for entering your VM box...
18:08.53ManxPowerdavidw, well put the CLI output on pastebin.ca
18:10.46*** join/#asterisk nmsclera (n=arthurh@71-33-40-22.albq.qwest.net)
18:10.55davidwhttp://pastebin.ca/35174
18:11.51davidwit calls up, waits(3), does saydigits(666) and then hangup()
18:12.22*** part/#asterisk Utah_Dave (n=boucha@0-1pool138-70.nas28.salt-lake-city1.ut.us.da.qwest.net)
18:12.32fugitivousername=87777
18:12.32fugitivouser=87777
18:12.32fugitivotype=peer
18:12.32fugitivosecret=87777
18:12.32fugitivoinsecure=very
18:12.32fugitivohost=sip.ia.com.ar
18:12.33*** join/#asterisk DarthClue (n=DarthClu@adsl-69-153-33-232.dsl.snantx.swbell.net)
18:12.34fugitivofromdomain=sip.ia.com.ar
18:12.36fugitivocontext=mainmenu
18:12.39fugitivocallerid=87777
18:12.40fugitivoouch
18:12.44nmscleraWhat would be the appropriate process to debug DTMF issues.  We're connecting to a SIP Provider via G729, tried different combinations of info and rfc2833 / relaxdtmf etc.. but for some reason inbound dtmf is not working at all..
18:12.44fugitivolol
18:13.05Kattyhmm.
18:13.15fugitivodamn middle mouse button
18:13.33iCEBrkrfugitivo: You're fired
18:13.49[TK]D-Fendernmsclera : pastebin your current sip.conf (masking PW's) and we'll take a look
18:13.57fugitivoiCEBrkr: shhhh
18:14.11[TK]D-Fender~pb
18:14.14jbotpb is probably a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca
18:14.30Kattyfugitivo: you're fired /and/ you must stay.
18:14.31davidwManxPower, can you make heads or tales of why it's running that twice?
18:15.00KattyDarthClue: hey you (=
18:15.24iCEBrkrha
18:16.19Katty:<
18:16.31ManxPower<PROTECTED>
18:16.41Kattyalso, my soy nog has gotten cold.
18:16.44Kattythis is just awful.
18:16.48DarthClueI had to move.  Went way South.  Almost to Mexico.
18:17.03QwellDarthClue: California?
18:17.05KattyDarthClue: why did you move?
18:17.05Qwellheh
18:17.10davidwManxPower, so... mmm why would that cause it to run twice?
18:17.14ManxPowerdavidw, also looks to me like Asterisk does not have permission to modify the .call file.l
18:17.35DarthClueTexas, an hour from civilization, 5 minutes from a Wal-Mart, 30 minutes from Mexico.
18:17.36davidwManxPower, yeah, but would that cause it to run twice?
18:17.42Qwellbbl
18:17.48ManxPowerdavidw, I can see how it might.
18:17.48*** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net)
18:18.05davidwmmmmm mm I guess if it keeps track of state by modifying it it might... eah
18:18.31davidwugh... now it just worked...
18:18.37SwK[Work]anyone know if there is something I can send to /dev/zap/ctl to force a span to go out of loop?
18:18.55ManxPowerSwK, zttool, tell it to unloop.
18:18.55DarthClueKatty, was time to do so.  Needed a change of environment.
18:19.05SwK[Work]ManxPower: hah yeah right
18:19.10ManxPowerUnless of course your TELCO has the loop set, in which case call the telco
18:19.28ManxPowerSwK, Um, why is zttool not acceptable?
18:19.31SwK[Work]ManxPower: telco sent a loop up command and we did, now its hung there and they cant even see the loop up
18:19.53ManxPowerSwK, pick the span from zttool, pick loop, then pick loop again.
18:20.00ManxPoweror just unplug the cable from the card
18:20.02*** join/#asterisk Nube101 (n=boanthra@pool-151-203-204-70.bos.east.verizon.net)
18:20.06ManxPoweror stop asterisk and unload the card driver
18:20.07*** join/#asterisk postel (n=jk@area41.OSPF.netmonks.net)
18:20.24wunderkinSwK[Work], i was told that its not possible to loop up the card.. but then im not sure what the loop option is for in zttool then
18:20.26SwK[Work]cant restart asterisk... gotta about 50 calls up on it already
18:20.28ManxPowerSwK, Get them to loop to the smartjack
18:20.54ManxPowerSwK, You have more than 1 span?
18:20.57SwK[Work]yes
18:21.16KattyDarthClue: k
18:21.18KattySwK[Work]: k
18:21.19Kattyi mean
18:21.20SwK[Work]not on the same T but the box as 3 active PRIs in it
18:21.21KattySwK[Work]: hi
18:21.24SwK[Work]hey katty
18:21.36*** join/#asterisk pakipenguin (n=Junaid@linuxpakistan/admin/pakipenguin)
18:21.46*** join/#asterisk J4k3 (i=j4k3@node218-157-88-65.1dial.com)
18:21.47ManxPowerSwK, have they looped from the CO to the smartjack?
18:21.51SwK[Work]as far as zttool... it just hangs as of late
18:21.59SwK[Work]yes they have
18:22.03*** part/#asterisk J4k3 (i=j4k3@node218-157-88-65.1dial.com)
18:22.09wunderkinoh, i guess i mean remote loopup.. but im not sure why it didnt work when i try to loop it.. shrug
18:22.10*** join/#asterisk L|NUX (i=linux@203.101.165.132)
18:22.24ManxPowerSwK, and they have seen no problems or errors after keeping the loop for 30 mins?
18:22.34SwK[Work]the funny thing is we both see good framing and timing and all the alarms clear ok... the D chan bounces on one box and doesnt do shit on the other... and the B's dont come up either
18:22.49SwK[Work]loop pattern test etc etc etc
18:23.01ManxPowersounds to me like a interrupt latency issue.
18:23.06*** join/#asterisk schuyler_ (n=schuyler@gateway.digium.com)
18:23.09SwK[Work]nope... 9 misses
18:23.15ManxPowerdo you get HLDC abort errors?
18:24.02SwK[Work]and if it was an interupt problem or a hardware issue it wouldnt give the same or very similar symptoms on a different box with a TE110 in it
18:24.03SwK[Work]nope
18:24.28ManxPowerSwK, only if that box wa a different motherboard/shipset
18:24.47nmsclera[TK]D-Fender: http://pastebin.com/483125
18:25.04SwK[Work]same and same
18:25.06davidwwill try again tomorrow
18:25.24nmsclera[TK]D-Fender: Sorry took so long, had to clean it up a bit, take out all kinds of commented out crap
18:25.31SwK[Work]we know the T1 has an issue on the Carrier side
18:25.39*** join/#asterisk J4k3 (i=j4k3@node218-157-88-65.1dial.com)
18:25.49*** part/#asterisk J4k3 (i=j4k3@node218-157-88-65.1dial.com)
18:26.20SwK[Work]my original question was is there a way without resetting the entire card is there something i can send to dev/za/ctl to restart just that 1 span
18:28.28*** join/#asterisk saftsack (n=saftsack@p54A7E001.dip.t-dialin.net)
18:28.29saftsackhi
18:28.33saftsackhas someone of you asterisk?
18:28.40[TK]D-Fendernmsclera : Ok, which is for your ITSP, and what standard are they supposed to be using for DTMF?
18:28.42saftsackaeeh wow didnt think
18:28.55saftsacki wanted to ask if someone has asterisk with misdn ^^
18:29.42nmsclera[TK]D-Fender: I'm not entirely sure, have a call in though -- are there any other options via g729 other than info and rfc2833?
18:30.25harryvvdoes a callfile need a contect in extentions.conf to send out a fax? somone in private chat asked me that and it did not sound right.
18:30.27[TK]D-Fendernmsclera : there is inband.  but you should be using what your provider TELLS you to.
18:30.40nmsclera[TK]D-Fender: ITSP is oneconnect (the only register addy)
18:30.42pakipenguin[TK]D-Fender, i have problems with g729 and dtmf doesnt seem to work
18:30.43pakipenguin:(
18:30.46iCEBrkrharryvv: It needs a context, sure
18:30.52[TK]D-FenderAnd DTMF doesn't have anything to do with the codec.
18:30.52harryvvokay created one.
18:30.55harryvvBut
18:30.58nmsclera[TK]D-Fender: g729 supports inband?
18:31.13harryvvWhat is caller mean in
18:31.15nmsclera[TK]D-Fender: I get a slew of messages on the console when I did try inband
18:31.15harryvv[callme]
18:31.15harryvvexten => s,1,Wait(3)
18:31.15harryvvexten => s,2,TxFax(${FAXFILE}|caller)
18:31.20[TK]D-Fendernmsclera : inband means * detects DTMF in the audio stream regardless of which codec.
18:31.25harryvv|caller
18:31.46[TK]D-Fendernmsclera : Make sure you are using what your provider suggests.
18:31.46iCEBrkrharryvv: huh?
18:31.51nmsclera[TK]D-Fender: I understand what inband means, but I understood that inband does not function properly with any non ulaw protocol
18:32.04nmscleraor uncompressed
18:32.08harryvviCEBrkr It is a example on http://www.voip-info.org/tiki-index.php?page=Asterisk+auto-dial+out
18:32.13[TK]D-Fendernmsclera : news to me, but ok......
18:32.37nmsclera[TK]D-Fender: Primiarly because the compression distorts the autio
18:32.43nmsclera[TK]D-Fender: Audio, even.
18:32.45[TK]D-FenderThen they should be providing it by INFO or RFC2833.You need to confirm with them.
18:33.08harryvvwait, sorry that contect in extentions.conf was given to me by another user.
18:33.16iCEBrkrharryvv: http://www.voip-info.org/wiki/view/app_rxfax+and+app_txfax
18:34.04nmscleraAnd another question for everyone, speaking of rxfax and txfax -- what are the overall results for faxing over uncompressed codecs (SIP Fax channels)?  They work well?
18:34.50harryvvIce, I was testing callfile
18:34.52*** join/#asterisk Sedorox (i=brandon@smartserv/cna/Sedorox)
18:34.55harryvvnot just sending a fax.
18:35.05iCEBrkrharryvv: ok, so one thing at a time.
18:35.40asterboy~fwd
18:35.42jboti heard fwd is Free World Dialup:  Brainchild of Jeff Pulver.  URL: http://www.pulver.com/fwd/
18:36.00harryvviCEBrkr we can talk here or in private..show some of my configs.
18:36.11iCEBrkrharryvv: I'm still confused as to what you're having problems with
18:36.13filethis server hates me... it has to
18:36.30saftsackare there more stable versions of spandsp now?
18:36.53ManxPower[TK]D-Fender, He is correct.  Only ulaw and alaw transport DTMF and tones correctly.
18:37.10ManxPowerthe reason there is such a thing as OOB (RFC2833) DTMF is because of this.
18:37.28harryvviCEBrkr baicly just want to make a callfile that will send a test fax of a tif image.
18:37.58iCEBrkrharryvv: ok, in your call file just set the Context: to your [fax] context
18:38.13harryvvie, directory/image.tif
18:38.34iCEBrkrno
18:38.45*** join/#asterisk Xen^ (i=linux@203.101.161.17)
18:39.22_Sam--hey manx:  as earlier, i have two asterisk boxes talking to eachother via IAX on the same LAN segment (192.168)...one is a gateway.  if i call the non-gateway box from an internal phone on the 192 network it sounds perfect...if i try to call out through the non-gatway box (which uses the other * box as the gateway) the calls sound terrible...what should i check?
18:39.23iCEBrkrharryvv: So which is it? faxing or callfile?
18:39.42ManxPowersaftsack, with recent verisons of SpanDSP the only time people have problems is when they try to send over VoIP
18:40.01ManxPower_Sam--, no idea
18:40.25harryvvice, send a fax for example one phone number with callfile. Then in the future, may want to send to 10 or more numbers.
18:40.33saftsackManxPower, sounds great :)
18:40.46iCEBrkrharryvv: Ok, so you can create a call file, right?
18:40.51harryvvsure
18:40.52*** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com)
18:41.04iCEBrkrharryvv: and you understand the Context: line within the call file?
18:41.14*** join/#asterisk privalodc (n=someone@Toronto-HSE-ppp3715118.sympatico.ca)
18:41.17harryvvmade one...just directing SetVar to the directory where the tif image is. I have read it.
18:41.30iCEBrkrharryvv: and you understand the Context: line within the call file?
18:41.34privalodcAnyone wants to help me get a Dchannel up on a PCI with a Sangoma A102?
18:41.55iCEBrkrprivalodc: You need to talk to your telco on that one
18:42.48privalodcAllstream was just here to test the loop...
18:42.57rikstahprivalodc, whats the prob
18:43.19privalodcrikstah: The dchannel does not get up...
18:43.36rikstahwhat errors
18:43.38iCEBrkrViagra
18:44.14saftsackManxPower, do you know what versions are recent?
18:44.15privalodcTimeout occured, restarting PRI
18:44.27privalodc<PROTECTED>
18:44.36rikstahno good
18:44.40harryvviCEBrkr so far no responce. Im sure spandsp is installed on @home it has fax test feature which I have tested.
18:44.49privalodcT200 counter expired, What to do...
18:45.07iCEBrkrharryvv: ok, have you created a callfile to dial your cellphone or some other number?
18:45.25harryvva fax number
18:45.41harryvvcould try to make it call my softphone extention.
18:45.45iCEBrkrdoes the fax ring?
18:45.47harryvvso at least i can hear it.
18:45.57harryvvfax is across town
18:46.11iCEBrkrharryvv: well, make a call file that dials a number you can answer
18:46.15iCEBrkrharryvv: or hear ring
18:46.23privalodcAny ideas?
18:46.26pakipenguinanyone knows if there is a neat/working implemenetation of web call back?
18:46.32harryvvgoing to have it call my polycom.
18:47.07*** part/#asterisk schuyler_ (n=schuyler@gateway.digium.com)
18:47.19*** join/#asterisk alephcom (n=alephcom@openbsd.hagenhomes.net)
18:47.36asterboyAnyone testing FWDin and FWDout?
18:48.18*** join/#asterisk RoyK (n=roy@host-81-191-147-248.bluecom.no)
18:48.42privalodcrikstah: those messages ring a bell?
18:49.32*** join/#asterisk Darkhalf (n=darkhalf@cpe-70-93-239-175.san.res.rr.com)
18:49.59asterboybenjk will like this: http://www.freeworlddialup.com/help/?p=knowledgebase&c=1&a=20
18:50.40[TK]D-Fenderprivalodc : JF?
18:51.06rikstahprivalodc, there isn't enough detail
18:51.26*** part/#asterisk alephcom (n=alephcom@openbsd.hagenhomes.net)
18:52.08[TK]D-Fenderprivalodc : What does "wanrouter status" say?
18:52.41harryvviCEBrkr in SetVar: I need to point to the file to be faxed? like SetVar: TMP=/PATHname/File.tif ?
18:53.02iCEBrkrharryvv: have you gotten it to dial?
18:53.13harryvvno im not at that stage yet
18:53.21iCEBrkrharryvv: then why are you worried about setvar?
18:53.29harryvvAm I right?
18:53.49iCEBrkrThis is why people can't figure shit out.. They're all over the map, working in a star pattern..
18:53.53iCEBrkrharryvv: you got ADD?
18:54.19[TK]D-FenderiCEBrkr : Multi-threading!
18:54.23iCEBrkrharryvv: using SetVar vs. Set depends on your version of Asterisk.
18:54.36iCEBrkr[TK]D-Fender: Someone can't keep track of the mutexes
18:56.37saftsackhow can i proof if my isdn card has a hardware contact to the ntba?
18:57.10pakipenguinsaftsack, send a call to ur isdn , you should see something in messages
18:57.36harryvvodd, I write the file testcall.call to /var/spool/asterisk/outgoing and it does not show up. Does asterisk delete this file after the call has been made?
18:58.18privalodcDevices currently active:
18:58.18privalodc<PROTECTED>
18:58.18privalodcWanpipe Config:
18:58.18privalodcDevice name | Protocol Map | Adapter  | IRQ | Slot/IO | If's | CLK | Baud rate |
18:58.18privalodcwanpipe1    | N/A          | A101/2   | 22  | 1       | 1    | EXT | 0         |
18:58.18privalodcWanrouter Status:
18:58.20privalodcDevice name | Protocol | Station | Status        |
18:58.22privalodcwanpipe1    | AFT HDLC | N/A     | Connected     |
18:58.47*** join/#asterisk KrayZK (n=ykhan3@mbl-99-57-76.dsl.net.pk)
18:58.50saftsackpakipenguin, hmm asterisk doesnt start because the card has no connect
18:59.05saftsackand i thought, that i can receive informations from a more base layer
18:59.05pakipenguinthat is strange
18:59.17saftsacki tested the second cable now
18:59.32iCEBrkrharryvv: huh yup
18:59.32saftsackmaybe the card is broken?
18:59.34KrayZKPlease help, voice quality on IAX is very bad
18:59.38pakipenguinnot really?
18:59.45saftsackpakipenguin, dunno
18:59.56harryvviCEBrkr mm well its not making the calls.
19:00.02Pegggerasterisk is supos to be runing on port udp 4569 but which i do a nmap -sU on the box which only nmaps for udp i dont get any  4569 only 111, 123, 668 and 671 what is going on here?
19:00.08[TK]D-Fenderprivalodc : You should set up your card to clock external (telco).  You'll need it for sync
19:00.24iCEBrkrharryvv: Watch the CLI after you move the file into outgoing
19:00.53privalodc[TK]D-Fender: Ok, will look at that
19:01.26KrayZKhow can I improve voice quality on IAX channels, I even have jitterbuffer enabled
19:01.34PegggerI need to figure out this port stuff so I can open it on my firewall
19:01.37iCEBrkrKrayZK: Get more bandwidth
19:01.50iCEBrkrKrayZK: and block eMule and other file sharing apps
19:01.51privalodc[TK]D-Fender: Whare is that set by the way?
19:02.03iCEBrkrPeggger: read the wiki.
19:02.14harryvvice, good point
19:03.49*** part/#asterisk genmud (n=genmud@ip68-98-82-206.ph.ph.cox.net)
19:04.09*** join/#asterisk genmud (n=genmud@ip68-98-82-206.ph.ph.cox.net)
19:04.10saftsackpakipenguin, can i take a normal ethernatcable?
19:04.24pakipenguinumm nope
19:04.27pakipenguini dont think so
19:04.37pakipenguinyou have this setup
19:04.39harryvviCEBrkr nothing happening on cli so far
19:04.47pakipenguintelco --> NTU ---> your card?
19:04.49saftsackk because i have an ethernetcable for testing
19:04.51ManxPowerA standard ethernet cable will work as a T-1 cable.
19:05.02saftsackbut my isdn cable doesnt work too :(
19:05.09ManxPowerHOWEVER, a ethernet crossover cable will NOT work as a T-1 crossover cable.
19:05.12iCEBrkrharryvv: odd, you should see it dial
19:05.43iCEBrkrharryvv: set verbose 4
19:05.44ManxPowerISDN is two wire and will work with most ANY regular telephone cable.
19:06.08[TK]D-Fenderprivalodc : thats in your wancfg.
19:06.29*** join/#asterisk bkw_ (n=bkw_@ppp-69-155-251-101.dsl.tulsok.swbell.net)
19:06.32pakipenguinyes
19:06.53harryvv[callme]
19:06.53harryvvexten => s,1,Wait(3)
19:06.53harryvvexten => s,2,TxFax(${FAXFILE}|caller)
19:06.53harryvvChannel: Zap/1/604xxxxxxx
19:06.54harryvv<PROTECTED>
19:06.56harryvv<PROTECTED>
19:06.58harryvv<PROTECTED>
19:07.00harryvv<PROTECTED>
19:07.02harryvv<PROTECTED>
19:07.04harryvv<PROTECTED>
19:07.06harryvvWhat is the extnetion part for?
19:07.06RoyK~pb
19:07.08jbothmm... pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca
19:07.12iCEBrkrharryvv: Dude, you know about pastbin man
19:07.20KrayZKI have more than enough I think only 5 channels and 512 kbps bandwidth
19:07.20harryvvyup
19:07.24iCEBrkruse it
19:07.42iCEBrkrKrayZK: yeah? and how many calls at once?
19:07.49docelmoYippie!
19:07.56KrayZKwhat is eMule and how do I start stop it, or even find out if it is installed
19:08.02iCEBrkrdocelmo: What are you all yippie about?
19:08.03KrayZKonly 5
19:08.13docelmodunno.. Im still on vacation? :)
19:08.21iCEBrkrdocelmo: screw you
19:08.22harryvvhttp://pastebin.ca/35181
19:08.26TheCops[TK]D-Fender, wow, polycom centralized phone config is kinda complicated for a normal computer guys, I gave the job to a employee to learn the phone...but when he arrived in the XML config section
19:08.30[TK]D-Fenderprivalodc : TE CLOCK MODE = normal in t1 advanced configuration optiosn.
19:08.31docelmohaha
19:08.31_DAWDoes anyone have an opinion on the reliability of realtime in 1.2?
19:08.53iCEBrkr_DAW: I personally wouldn't use it
19:09.24Vijayare there any issues in practical implementation of 1.2?
19:09.25saftsackpakipenguin, what do you think can be broken?
19:09.29_DAWI get that impression from much of what I have read.
19:09.30iCEBrkrharryvv: Extension is the extension in your [context]
19:09.38pakipenguinumm saftsack either the cable
19:09.40pakipenguinor your card
19:09.43KrayZKiCEBrkr: any ideas?
19:09.46pakipenguinis your ntu synced with the exchange?
19:09.47harryvvyes?
19:09.47iCEBrkrVijay: Yea, know what the hell you're doing first
19:09.48saftsackthe card is 2 weeks old :(
19:10.07saftsackim going to test the cable now with a telephone on my normal tk
19:10.15[TK]D-FenderTheCops : Poly's XML isn't that hard.... shouldn't take long.
19:10.23*** join/#asterisk gdsaf3 (n=root@telmo.ifxnw.com.ar)
19:10.31*** join/#asterisk jeffik (n=Jeff@CPE0050babf4cd5-CM014350000760.cpe.net.cable.rogers.com)
19:10.34iCEBrkrsaftsack: if you can test that cable with a normal phone, you got the wrong cable. :)
19:10.49TheCops[TK]D-Fender for a guys who never seen an HTML or any programming code, this is scared for him lol
19:10.55TheCopsI'll be obligate to learn it alone
19:10.56TheCopserrr
19:11.18[TK]D-FenderTheCops : I may just pass you a working sample later.
19:11.37_DAWTheCops: Give them a copy of XMLexplorer.  The gui makes it less intimidating.
19:12.07TheCops[TK]D-Fender, I found something on the voip info wiki, some sample...
19:12.12TheCopsI'll work from that files
19:12.14iCEBrkrharryvv: and you're Priority: is wrong too
19:12.20[TK]D-FenderCAREFUL ON THAT
19:12.21TheCopslol
19:12.39[TK]D-FenderPoly XML configs are version specific.  What SIP image did your's come loaded with?
19:12.42*** join/#asterisk MikeJ[Laptop] (n=ircatjer@d14-69-8-30.try.wideopenwest.com)
19:12.53TheCops1.6.2 If I remember
19:13.15TheCopsI'll have to verify but I'm kind sure
19:13.28[TK]D-FenderMake sure, then download the appropriate firmware/ sample pack and base your's off of that.
19:13.30KrayZKiCEBrkr: I'm using 5 systems dialing, Xlite to Asterisk aLaw and Asterisk to VoIP Provider GSM
19:13.54KrayZKand I don't have eMule running or installed
19:13.55saftsackiCEBrkr, ???
19:14.11iCEBrkr< saftsack> im going to test the cable now with a telephone on my normal
19:14.11iCEBrkr<PROTECTED>
19:14.13saftsackon a ntport i can connect a one to one cable
19:14.25saftsackand on a telephone to
19:14.40saftsacki need just a crossed cable if i configure my card to provide a nt port
19:14.49privalodc[TK]D-Fender: I got TE_CLOCK        = NORMAL and TE_REF_CLOCK    = 1 now.
19:15.03saftsackiCEBrkr, do you agree?
19:15.08iCEBrkrsaftsack: no.
19:15.21TheCops[TK]D-Fender, XML sample are in the firmware files from Polycom website ?
19:15.33saftsackhmm i have many cards which are working on a normal cable
19:15.40saftsackiCEBrkr, what for a cable do i need?
19:15.51*** join/#asterisk razu_ (n=razu@217-159-240-134-dsl.est.estpak.ee)
19:15.51pakipenguinsaftsack, a 4 wire cable
19:15.58harryvvice, so what you are saying is drop the extention => s,1,Wait (3) and put 800 in the place of s ?
19:16.12saftsackand 6 wire cable doesnt work?
19:16.23KrayZKplease help me, I am being beaten over the head repeatedly with a large club, because agents can't hear the customer or the customer can't hear agent
19:16.26iCEBrkrharryvv: Do you understand the concept of contexts, extensions and priorities??
19:16.29harryvvsure
19:16.31saftsackbecause avm ships the cards with a 6 wire cable
19:16.43harryvvJust some guy here who showed me his...obviosly does not.
19:16.44harryvv:)
19:16.47iCEBrkrharryvv: If you did, all of this would make sense :)
19:16.55saftsackdo the avm cards need other cables than the hfc cards?
19:17.04iCEBrkrharryvv: hold up a sec
19:17.15harryvvyea..he did not know what he was doing. I just never messed with call files thats all.
19:17.27iCEBrkrIt's all one for one.
19:17.38iCEBrkrSet the context.. Set the extension, set the priority
19:17.40saftsackiCEBrkr, do you have a link where i can read everything about isdn cables?
19:17.48KrayZK* now bleeding profusely, but still can't manage to improve voice quality
19:17.54iCEBrkrsaftsack: Yeah, it's called www.google.com
19:18.24saftsackim looking there already thanks
19:18.52[TK]D-FenderTheCops : http://www.freedomphones.net/polycom/files/
19:19.23KrayZKIAX and bad voice, anyone....going once...going twice...
19:19.33iCEBrkrharryvv: http://pastebin.ca/35185
19:19.59[TK]D-Fenderprivalodc : So kill *, do a "wanrouter restart", then "wanrouter status" until its up 100%, thenr estart *
19:20.39*** join/#asterisk kio (n=kio@ool-4577adba.dyn.optonline.net)
19:20.41KrayZKI can't seem to pinpoint the voice quality issue on IAX channels
19:20.51iCEBrkrKrayZK: how do you know it's the IAX leg?
19:20.58*** join/#asterisk oej (n=oej@apollo.webway.se)
19:21.07saftsackiCEBrkr, i have a 4 wired cable here
19:21.07fugitivoKrayZK: what devices are you using?
19:21.22KrayZKbecause calls between extensions are just fine
19:21.30saftsackits not crossed and it doesnt work on my hfc card in te but on a telephone
19:21.37saftsackso my isdn card is broken, right?
19:21.52iCEBrkrKrayZK: How do you know it's not a problem between the second Asterisk box ==> SIP
19:22.18RoyKsaftsack: get another hfc card and wire a loop
19:22.24KrayZKthe second asterisk box is using IAX
19:22.30RoyKbtw isdn is 4 wire, not two
19:22.52RoyKthe S0 bus, to which the hfc card is supposed to be attached, is a 4-wire bus
19:22.52saftsacki didnt say 2 wire
19:22.54iCEBrkrKrayZK: ok, so what's the layout then??
19:22.58saftsacki have a 4 wire cable here
19:23.07saftsackand that cable works on my isdn telephone
19:23.12RoyKsomeone said 2-wire...
19:23.19saftsacksomeone said crap ^^
19:23.34RoyKsaftsack: what driver? visdn or bristuff?
19:23.41saftsacka loop? do you mean from card one to card two?
19:23.45saftsackRoyK, no, misdn
19:23.51RoyKouch
19:23.56KrayZKa few things i tried are: placing calls only on alaw to reduce transcoding; enabled jitterbuffer, tried trunking which gets rejected from the provider
19:23.58RoyKsaftsack: yes, from card to card
19:24.00saftsackwhat ouch? ^^
19:24.07RoyKsaftsack: one in TE mode and one in NT mode
19:24.09*** join/#asterisk BugKham (i=BugKham@61.47.108.39)
19:24.10saftsackyes ok
19:24.16RoyKsaftsack: try bristuff, works for me (tm)
19:24.22saftsackbut i need a crossed cable then or?
19:24.42saftsackRoyK, it worked already with the actual config, so i think, that the card is broken
19:24.44KrayZKI have 5 SIP Xlite softphones connecting using SIP to my asterisk box and an IAX connection with the provider
19:25.03BugKhamanyone knows where to find compilation information about nist-sip?
19:25.05saftsackso im going to implement the second card now
19:25.09iCEBrkrKrayZK: I still think you have bandwidth issues
19:25.16*** part/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net)
19:25.35[TK]D-FenderiCEBrkr : Hey, your dumb-ass telco do their job yet?
19:25.46iCEBrkr[TK]D-Fender: I haven't even cared at this moment.
19:25.51[TK]D-Fender:/
19:26.02KrayZKeven with 512 kbps for 5 systems only, when I run iftop, my outgoing connections show only 25 kbps used/channels
19:26.04iCEBrkr[TK]D-Fender: I got my Miami stuff working and tested.. I just got a new server in yesterday.  So I'll be building that.
19:26.05*** join/#asterisk chasej (n=chasej@X.gouldacademy.org)
19:26.13RoyKsaftsack: i beleive bristuff currently works better than misdn
19:26.18RoyKalthough i do not know for sure
19:26.23iCEBrkr[TK]D-Fender: I can't me leaving the PRIs in alarm as they'll just flip them into loop-back
19:26.27iCEBrkrs/me/be
19:26.59harryvviCEBrkr is the call file execution instantanius when the file is moved or saved there?
19:27.10iCEBrkrharryvv: pretty much
19:27.23fugitivocopy, never move
19:27.24KrayZKI have the same problem even when one agent is dialing, b ut it develops slowly over time
19:27.32iCEBrkrfugitivo: Stoner?
19:27.34harryvvfug, okay :)
19:27.37iCEBrkrfugitivo: you move the file.
19:28.02ManxPowerfugitivo, no, for .call files you MOVE not COPY.
19:28.05malverian[work]Anyone here using spandsp/app_rxfax ?
19:28.12fugitivoManxPower: it depends
19:28.16iCEBrkrManxPower: He's stoned.
19:28.20iCEBrkrfugitivo: depends?
19:28.22KrayZKiCEBrkr: infact the problem usually gets worse over time
19:28.24_Sam--im trying to setup broadvoice as a backup for our termination....i can register and receive calls in through the broadvoice number, but i cant make outgoing:   Dec 29 14:27:30 WARNING[30018]: chan_sip.c:1966 create_addr: No such host: sip.broadvoice.com/13029833466
19:28.24ManxPowermove is an atomic operation so there is no change the file is 1/2 written when Asterisk reads it.
19:28.25fugitivoManxPower: if the .call file has errors, you'll lose the file
19:28.26malverian[work]Sorry, more specifically, is anyone using spandsp-0.0.2_pre20 with app_rxfax
19:28.33fugitivoManxPower: and you'll need to rewrite it
19:28.36ManxPoweriCEBrkr, A little early in the day for that.
19:28.44malverian[work]And if so, does it work best with a specific version of the rxfax module?
19:28.45iCEBrkrManxPower: Tell him that.
19:28.47iCEBrkrfugitivo: you're stoned.
19:29.07ManxPowerfugitivo, you can always copy into another filename, then move THAT file.
19:29.10_Sam--the sip.broadvoice.com has a context in my sip.conf...and it points to the right host
19:29.11iCEBrkrfugitivo: You MOVE the file. Because there's a chance Asterisk will find the file and read it without the copy being completed.
19:29.25fugitivonever happened to me
19:29.26*** join/#asterisk _Madar (n=tophe@219-84-129-163-adsl-tpe.static.so-net.net.tw)
19:29.40KrayZKI can't seem to pinpoint the voice quality issue on IAX channels
19:29.44iCEBrkrfugitivo: The possibility exists.
19:29.45ManxPowerOr you can create the file, then set the time on the file to be way in the future (keeps asterisk from trying to process it), then write the file, close the file, then set the time on the file to the current time.
19:29.51KrayZKIAX and bad voice, anyone....going once...going twice...
19:30.19*** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-179-75-202.cust.bezeqint.net)
19:32.08harryvviCEBrkr here is what I have again. http://pastebin.ca/35190
19:32.32iCEBrkrharryvv: I edited yours and updated it.. did you see it?
19:32.37harryvvsorry no
19:32.43ManxPowerKrayZK, do an "iax2 show channels" when there is an active call.  notice the lag and jitter
19:32.51iCEBrkrharryvv: http://pastebin.ca/35185
19:33.23*** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net)
19:33.33*** join/#asterisk Dr-Linux (n=Nothing@202.125.141.6)
19:33.50harryvvyour changes are below the == line?
19:33.57iCEBrkrabove.
19:34.12harryvvyea sorry yea see that
19:34.20*** join/#asterisk saftsack_ (n=saftsack@p54A7F5C2.dip.t-dialin.net)
19:35.14harryvvas far as the file the call file grabs is SetVar: tmp=/pathtofile/filename.tif correct ?
19:35.37Dr-Linuxanybdoy experience with SPA-2100?  i have some problems, analalog phones connected to spa-2100 rings only 2 times, and after dialing it takes too long to esteblish the conntion? spa is behind the NAT
19:35.39iCEBrkrharryvv: dude, you still havent' had it dial why are you worried about setting variables?!?!
19:36.46harryvvyea :)
19:37.03iCEBrkrharryvv: I'm refuse to help you with that until you get it to dial.
19:37.51saftsack_so now i test visdn
19:37.53fugitivoi
19:39.14*** join/#asterisk backblue (n=moo@82.102.1.42)
19:40.04KrayZKMost calls have no real jitter or lag but some calls suddenly get a high jitter of about 700 or so and sometimes followed by a [WARNING]: chan_iax2.c: network thread: ast_sched_runq ran XXX tasks all at once
19:40.10backblueif i try to call to another server with asterisk2asterisk, if the user in this side, its the same in another asterisk, it will not work, why?
19:40.14backblueit asks for auth
19:40.21KrayZKhow can i find out what these tasks might be
19:41.41*** join/#asterisk _fan_ (n=allan@static-64-222-189-188.man.east.verizon.net)
19:41.46TheCops[TK]D-Fender there again ?!
19:42.47[TK]D-Fender?
19:42.55*** join/#asterisk kio (n=kio@ool-4577adba.dyn.optonline.net)
19:43.09TheCops[TK]D-Fender you can't download current software on the polycom website ?!
19:43.22ManxPowerTheCops, No.
19:43.27TheCopsduh
19:43.39TheCopshow can I get sample for this version ?!
19:43.46saftsack_chan_visdn.c:3982: warning: passing arg 1 of `ast_channel_unregister' from incompatible pointer type
19:43.48saftsack_:(
19:43.49ManxPowerTheCops, Polycom does not want to support end users.  It's one of the few annoying things about it.  freedomphones web site has new firmware within a few weeks of release
19:44.01ManxPowerPolycom wants their resellers to support users.
19:44.03Dr-Linuxanybdoy experience with SPA-2100?  i have some problems, analalog phones connected to spa-2100 rings only 2 times, and after dialing it takes too long to esteblish the conntion? spa is behind the NAT
19:44.13TheCopsduh
19:44.15TheCopsI hate that
19:44.22ManxPowerTheCops, You get used to it.
19:44.46ManxPowerTheCops, with Cisco you have to buy the phone, then a power supply for it, then the SIP firmware for it.
19:44.54saftsack_ooooh f***
19:44.55TheCopsManxPower, I'm cisco reseller :P
19:44.59TheCopsit's more easy for me
19:45.05TheCopsI have all ios or stuff like that on the website
19:45.07TheCopshehe
19:45.07saftsack_i have debian and debian has no udev. can i install visdn on debian?
19:45.09[TK]D-Fenderand THEN kiss their ass to download support stuff off their site ;)
19:45.15ManxPowerTheCops, become a polycom reseller then you get access to the polycom firmware
19:45.20TheCopshrmm
19:45.20harryvvAll of Vancouver internationals phone system was updates with the cisco IP phones. What a way to advertise.
19:45.22TheCopscan be great
19:45.26TheCopsand start selling on the net
19:45.28TheCopsat good price
19:45.32TheCopscan be nice..
19:49.24TheCopsManxPower or [TK]D-Fender, do you have 1.6.3 firmware or 1.6.2 ?
19:49.37ManxPowerTheCops, it's not on freedomphones?
19:49.55ManxPowerTheCops, BTW, all the release notes and docs are available from the polycom web site for free.
19:49.56TheCopsManxPower, freedomphones, what's that ? a model ?
19:50.08ManxPowerTheCops, the wiki.  someone put the link up for you.
19:50.14ManxPower~polycom
19:50.17TheCopshehe
19:50.22*** join/#asterisk kiwnix (n=egarcia@82.158.152.212)
19:50.42ManxPowerTheCops, search the wiki for freedomphone or freedomphones
19:50.50*** join/#asterisk shido6 (n=shido6@d221-68-216.commercial.cgocable.net)
19:50.53[TK]D-FenderTheCops : I Have many versions. 1.4.2,1.5.2,1.6.1,1.6.2, 1.6.3
19:50.56TheCopsho my god
19:50.57TheCopsyeah
19:51.02*** join/#asterisk fulgas (n=fulgas@a81-84-117-84.cpe.netcabo.pt)
19:51.04TheCopsI just found it
19:51.21Beirdo[TK]D-Fender: that almost looked like an SNMP OID :)
19:51.27TheCops[TK]D-Fender, can you send me 1.6.3 ?
19:51.33TheCopsIt dont have it on the freedomphones
19:51.34[TK]D-FenderBeirdo : uhhh ok!
19:51.48Beirdobeen doing too much cacti lately, I gues
19:51.52[TK]D-FenderTheCops : just use the one for 1.6.2 for now get it working, THEN upgrade
19:52.00KrayZKAny asterisk users from Pakistan in here?
19:52.09TheCops[TK]D-Fender, okidoki
19:52.12TheCops:)
19:54.41saftsack_my visdn is searching for channel_pvt.h
19:56.39harryvviCEBrkr My asterisk box attempted to send the fax but the remote end user said there fax did not pickup. Here is the pastebin. http://pastebin.ca/35192
19:57.29*** join/#asterisk jrsharp (n=jrsharp@pcp0010016319pcs.galitn01.tn.comcast.net)
19:57.30iCEBrkryay
19:58.00jrsharphey all
19:58.46jrsharpI recently set up a clone x100p card in my asterisk box and have successfully gotten it working to accept incoming calls as well as route outgoing calls to PSTN...
19:59.02harryvviCEBrkr going to try it on there main phone even though thay said thay did not hear the fax machine ring.
19:59.08iCEBrkrharryvv: What version of Asterisk you have?
19:59.13jrsharpI was wondering, is it possible to "share" the analog line in my home with an analog telephone that is NOT an extension in *
19:59.14jrsharp?
19:59.25iCEBrkrjrsharp: Not really.
19:59.50jrsharphmm
19:59.52[TK]D-Fenderjrsharp : Clarify "share"
19:59.54iCEBrkrjrsharp: Tho someone claims you can set your x100p not to answer on the 1st ring
20:00.08iCEBrkr[TK]D-Fender: He means so asterisk isn't always answering.. well, I assume so anyhow
20:00.10harryvv1.0.9
20:00.26jrsharpwell, I was thinking that I could have asterisk decide whether or not to pick up a call based on CID, perhaps?
20:00.43jrsharpyeah
20:00.49iCEBrkrharryvv: ok, in your callfile SetVar: Faxfile=filename.tif
20:01.00fugitivojrsharp: ex-girlfriend
20:01.17jrsharphehe... yeah, something like that ;)
20:01.20iCEBrkrjrsharp: That's kinda hoakey, but ok.
20:01.50fugitivojrsharp: i mean, check the exgirlfriend functionality
20:01.51iCEBrkrjrsharp: You can do all that logic in your dialplan. So your extensions don't have to ring
20:02.10*** join/#asterisk andye (n=endocran@oriongw.orion-design.com)
20:02.18jrsharpwell, here's the deal, I'm still experimenting with *, and I know my wife doesn't want * picking up the phone for most calls...
20:02.28harryvvokay thanks
20:02.34jrsharpwe would want most calls to continue going to the answering machine
20:02.35iCEBrkrjrsharp: I'd say it's all or nothing :)
20:02.43jrsharphmm... ok
20:03.06jrsharpwell, let me ask another, related question...
20:03.06iCEBrkrjrsharp: I had that issue at first too.  I didn't have confidence in my configuration.  But then I said 'screw it' I'm gonna make it work
20:03.17iCEBrkr...and I did.
20:03.18jrsharpahh... ok... will do
20:03.21jrsharpawesome
20:03.26fugitivoi remember that guy that installed asterisk on a company, but the boss wanted to use a regular answering machine, so he was looking for atas for that
20:03.27fugitivolol
20:03.32iCEBrkrNow, I don't even have PSTN and use VoIP 100%
20:03.49jrsharpis it possible to "ring" that analog handset from *?
20:04.02iCEBrkrjrsharp: You mean without any adapters?  No
20:04.17jrsharpit's an FXO/FXS thing, right?
20:04.18eKo1* is software
20:04.20saftsack_someone of you compiled the new visdn devel version?
20:04.30iCEBrkreKo1: very good..
20:04.42iCEBrkrjrsharp: So you'll need like a SIP phone or ATA
20:04.51iCEBrkrjrsharp: I recommend Sipura
20:05.04jrsharpyeah... I've got a SPA 1001
20:05.10iCEBrkrcool
20:05.27jrsharpok... well, like you said, I guess I've just got to get confident in my setup
20:05.29iCEBrkrjrsharp: I know what you're getting at tho.. It's a pain in the ass to ATA all the phones in your house.
20:06.06*** join/#asterisk ohad (n=ohad@19-231-13-72.cosmoweb.net)
20:06.07jrsharpyeah, I just thought there might be a way to do like a party line or something and make the house phones ring at once...
20:06.17ohadhi, how do i change the hours of operations of my queue?
20:06.19jrsharpseems like I used to be able to do something like that as a kid
20:06.30iCEBrkrjrsharp: My solution for that-- I was gonna put my Asterisk box in my basement next to where the phone lines come in, get a SPA and wire the house lines into the SPA.
20:06.38ohadi looked into extensions.conf and queue.conf but couldn't really find anything.. ideas?
20:06.54jrsharpiCEBrkr... yeah, I did that once before
20:06.59jrsharpit worked quite well, actually
20:07.23iCEBrkrjrsharp: Yea, I didn't wanna have to run ethernet and have to buy 4 Sipuras :)
20:07.26jrsharpbut... the first VoIP experiment did not garner favor with my wife who decided the quality was bad and it was unreliable
20:07.43jrsharpI was not using asterisk yet
20:07.46iCEBrkrjrsharp: So keep the PSTN.. You don't HAVE to use VoIP
20:07.54iCEBrkroh? Vonage or something?
20:08.01jrsharpI was just connected straight to a VoIP provider
20:08.03jrsharpyeah
20:08.09jrsharpit was Broadvoice
20:08.20iCEBrkrAhhh, I haven't heard anything too good about them
20:08.39harryvvnifty, asterisk just crashed.
20:08.40iCEBrkrI have a friend who's been using Vonage for about 2-3yrs now, no complaints
20:08.49jrsharpyeah, it was a horrible experience, really, and I hate that it's affected my ability to play with VoIP at home now
20:08.58ohadi am trying to set my queue to be between xtime-ytime.. how do i defien that in the queue?
20:08.59jrsharpyeah, Vonage seems to be pretty good
20:09.00[av]baniits poisoned your wife
20:09.01[av]bani<3
20:09.03loudthey are excellent, your girlfriend can stay up to 7 hours talking to another country and not bothering you.
20:09.04jrsharpyeah
20:09.23harryvvloud, you mean in north america right?
20:09.31loudbroadvoice i mean
20:09.33jrsharpwell, I don't mind sticking with PSTN, but I want my cake and eat it, too
20:09.49[av]banipstn is so old hat
20:09.55harryvvdepends how dependable you want your connection to be.
20:09.56saftsack_^^
20:09.58saftsack_thats true
20:10.07harryvvpstn is also 99.9% reliable
20:10.08iCEBrkrjrsharp: So thats why you dial 9 before you dial.
20:10.16[av]banii dont see why telcos dont just sell voip trunks over ethernet
20:10.20jrsharpeh?
20:10.21eKo1more like 99.999%
20:10.33[av]banicheaper and easier for them
20:10.35iCEBrkrjrsharp: I had mine setup when I dialed 9+phone number it dialed via PSTN
20:10.39[av]banibut they like to do things the hard way
20:10.44iCEBrkrjrsharp: if I didn't dial 9, it went out VoIP
20:10.48jrsharpyeah, that's what I've got now
20:10.52fugitivoi'm having a problem with a sip provider, i get calls, but when i pickup the phone and say "hello" the call is ended, i get 503 Server Error
20:10.54jrsharpoh, I see
20:10.55jrsharpyeah
20:11.07saftsack_chan_visdn.c:41:34: asterisk/channel_pvt.h: No such file or directory
20:11.11saftsack_why? :(
20:11.13jrsharpI don't have a VoIP provider, really, though... I'm just connected to FWD at the moment
20:11.14iCEBrkrjrsharp: That way I still had the best of both worlds...
20:11.37andyeI have a quick one (hopefully) i have one dtdm13b and one dtdm21b - three fxo four fxs - the fxs lines make some loud clicking prior to * inicating pickup
20:11.39iCEBrkroh, 8+number was FWD :)
20:12.10jrsharpwhat VoIP provider do you use for LD?
20:12.14iCEBrkrjrsharp: Voicepulse.
20:12.24jrsharpIAX?
20:12.28iCEBrkrjrsharp: yeah
20:14.21jrsharpwell, that's good to know
20:14.24jrsharpthanks
20:15.23jrsharpI guess that if I just set up * and put that ATA on the house line, and route all calls over PSTN by default, I shouldn't really have any quality issues... right?
20:15.35jrsharpor will the x100p clone cause me quality issues
20:15.42harryvviCEBrkr I copy the callfile.call to the outgoing and it crashes asterisk.
20:15.52fugitivojrsharp: echo and problems detecting hangup
20:16.04iCEBrkrjrsharp: Yea, it acts as a normal phone
20:16.08jrsharphmm... that might be a problem
20:16.10iCEBrkrfugitivo: You're smoke'n still.
20:16.15iCEBrkrfugitivo: It works flawlessly.
20:16.18jrsharpyeah?
20:16.21fugitivoiCEBrkr: known problems with x100p clone
20:16.35iCEBrkrfugitivo: I've used my x100p for over a year.  zero problems.
20:16.44harryvvWhat would cause a call file to crash asterisk?
20:16.57jrsharpdid you have to do any echo training or anything?
20:17.02fugitivoiCEBrkr: still it can happen ;)
20:17.03iCEBrkrjrsharp: I didn't do anything special.
20:17.15jrsharphmm... that's hopeful
20:17.20fugitivothere're known problems with that hardware
20:17.26fugitivoyou can have the problems, or not
20:17.38saftsack_fugitivo, do you have isdn?
20:17.43fugitivono
20:17.54*** part/#asterisk _fan_ (n=allan@static-64-222-189-188.man.east.verizon.net)
20:18.02saftsack_oohh why has no one isdn :(
20:18.02*** join/#asterisk Umaro (n=umaro@68.142.142.105)
20:18.27fugitivosaftsack_: why you have isdn? :)
20:18.33iCEBrkrfugitivo: Puff-Puff pass man, you're mess'n up the rotation
20:18.41Umarohey guys.. I'm getting this "Bad FCS (6) on span 2" error on my TE405P now. ML archives say it's timing or interrupts, but it's not
20:18.42saftsack_because im living in germany :)
20:19.02eKo1isdn is standard in germany
20:19.28fugitivoiCEBrkr, jrsharp: http://www.voip-info.org/tiki-index.php?page=X100P+clone
20:19.48iCEBrkrfugitivo: Hey, I'm just telling you what I experienced. And I didn't have any problems with it.
20:20.04iCEBrkrfugitivo: it's problably Digium propaganda so you buy their card instead
20:20.14fugitivoiCEBrkr: I'm just telling what is public knowledge and my experience :)
20:20.43*** join/#asterisk ikey (i=ikey@220.224.14.31)
20:21.00fugitivoi have 2 x100p clone cards with that problems, i don't have those problems with the tdm400, so the problems are real and depends on your line
20:21.09iCEBrkrfugitivo: you work for digium, don't you?
20:21.18iCEBrkrMy lines were shitty
20:21.36iCEBrkrThe only problem I had was sometimes people said I sounded far away/quiet
20:21.45iCEBrkrSo I just tweaked the tx/rxgain settings.
20:21.45fugitivothat's a known problem :)
20:22.11fugitivosometimes, no matter how you tweak your tx/rx settings, it'll still sound far away
20:22.13iCEBrkrYou sure you got the correct card?
20:22.28iCEBrkrCuz theres and E and a EP and one without any letters.
20:22.38iCEBrkrThe boxes look the same
20:22.47fugitivoboxes?
20:22.49fugitivolol
20:23.05privalodcWell, still not able to the the D-Channel up....
20:23.22saftsack_eKo1, do you have isdn?
20:23.43fugitivo"Many users have reported echo or delay problems using a clone. The x100p does not have a programable chipset for impedance matching. wcfxo with option opermode=1 parameter was intended for the TDM card. There seems to be different DAA's used in the clones."
20:23.51iCEBrkrblah blah blah
20:24.15iCEBrkrfugitivo: How many people come in here bitching about their X100P clone having echo?
20:24.18saftsack_iCEBrkr, do you use bristuff or what?
20:24.35*** join/#asterisk J4k3 (i=j4k3@dhcp-12-197-128-55.intrastar.net)
20:24.38privalodcIs there a way to change the override the "automatic" (away/busy) voicemail message selection. Meaning, bu dialing say *80, the user selects that the away message should always be used...
20:24.38iCEBrkrsaftsack_: I *JUST* started doing PRIstuff :D har har har har
20:24.38fugitivoiCEBrkr: not much people buy that card lately
20:24.45andyeAny reason the tdm400 would be clicking (click, pause, click, click)  before Ring Begin and Ring/Answered (digium cards)
20:24.47saftsack_iCEBrkr, WHAT? :)
20:24.51eKo1saftsack_: err, no
20:24.59fugitivoiCEBrkr: some time ago more people came here with that problems :)
20:25.00*** join/#asterisk caryon__ (n=caryon@p54A3C8E3.dip0.t-ipconnect.de)
20:25.03saftsack_but i saw, that a singleport pri card costs just 500euro
20:25.24iCEBrkrfugitivo: I've been in here since some time ago.. I don't recall much talk about it
20:26.55fugitivoiCEBrkr: well, i'll not discuss about this
20:27.17iCEBrkrfugitivo: Whatever man, you're telling people to copy files instead of move them :P~~~~
20:27.53fugitivowo?
20:27.54fugitivoso?
20:29.15trixteryes always mv /etc/passwd not cp :P
20:29.17Dandanwie?
20:31.11[TK]D-Fenderprivalodc : Is the PRI confirmed functional with the telco?
20:32.05privalodc[TK]D-Fender: Yep, a tech was here this morning and did a test.
20:32.27trixterdid the tech pass the test?
20:32.43trixterwas it multiple choice or essay?  what kind of test?
20:32.45[TK]D-Fenderdowned wanrouter, restarted it, and does it say connected?
20:33.44privalodc[TK]D-Fender: you mead wanrouter stop; wanrouter start ?
20:33.55[TK]D-Fenderprivalodc : yeah, and what does your status say after?
20:34.36*** join/#asterisk saftsack__ (n=saftsack@p54A7DF90.dip.t-dialin.net)
20:35.19privalodc[TK]D-Fender: Status is Connected, let me look at the d-channel
20:35.57*** join/#asterisk Tili (i=Tili@202-133-67-128-dialup.sat.net.pk)
20:37.09[TK]D-Fenderstatus from "pri show span 1" <-
20:38.12privalodcStatus: Provisioned, Down, Active
20:38.30*** join/#asterisk kio (n=kio@ool-4577adba.dyn.optonline.net)
20:38.46[TK]D-Fenderprivalodc : make sure your startup scripts are ok, then do a cold shutdown and restart and see if it comes up.  Mine seems to do that sometimes.
20:38.52privalodcThere was 8x "Primary D-Channel on span 1 up" and then "Primary D-Channel on span 1 down"
20:39.26privalodcI don't like the "do that sometimes" :-)
20:39.28*** join/#asterisk insomni (n=insomni@x1-6-00-e0-18-6f-34-ad.k455.webspeed.dk)
20:39.29harryvvlooks like my call file Almost was able to fax a document but got a error. The rest is on http://pastebin.ca/35198 if anyone is interested in looking at it.
20:39.46[TK]D-FenderBefore you do, feel free to pastebin your wanpipe1.cfg, zapte.conf, and zapata.conf
20:40.14[TK]D-Fenderprivalodc : I'm not sure the origins of my problems yet, but its infrequent.
20:40.36iCEBrkrharryvv: doh
20:41.11*** join/#asterisk r0d3nt|m (i=r0d3nt@tinfoilhat.net)
20:41.13harryvvice, what?
20:41.16iCEBrkrharryvv: Welp, I can't help ya any further, I haven't messed with faxing before.  But ya got it dialing *golf clap*
20:41.18harryvv:)
20:41.25harryvv:)
20:41.40harryvvyea googling around for the answer.
20:41.53harryvvI should try a sound file.
20:41.58iCEBrkrhaha
20:42.01iCEBrkrtext file
20:42.19harryvvthats another option
20:42.22privalodc[TK]D-Fender: config at http://pastebin.ca/35199
20:42.46harryvvI will try a strait text file and see if goes though.
20:43.29iCEBrkrharryvv: Back when I was trying to use QModem:Pro and my fax-modem.  I was only able to sent ascii files.
20:43.41iCEBrkrI'm not sure if this has the same limitations
20:44.05Dr-Linuxanybody using spa-2100 behind the NAT ?
20:44.20iCEBrkrDr-Linux: I'm using a SPA2k behind nat
20:45.01rayvd"the" nat??  the one and only nat? =-o
20:45.15harryvvthat must have been a long time ago.
20:45.28iCEBrkr"The NAT" tonight at 11 on MTV:Oddities
20:45.36iCEBrkrharryvv: very!
20:45.44Dr-LinuxiCEBrkr: you may help me out
20:45.53iCEBrkrDr-Linux: what's not working?
20:46.08harryvvI first heard about the internet in 1991. suprisingly, the whitehouse had a site in ip format.
20:46.32harryvvNeighbor told me about it and gave him a strange look..odd concept :)
20:46.51iCEBrkrhaha
20:46.59harryvvDNS did not come around what untill 1992?
20:47.08Dr-LinuxiCEBrkr: i have problem, 1st my spa connected phones got dropped/unregistered after some time. 2nd when i dial from sama analog phones it takes too long to esteblish
20:47.33*** join/#asterisk andye (n=endocran@oriongw.orion-design.com)
20:47.41Dr-LinuxiCEBrkr: and last problem is, when i call to those phones, i rings only 2 times
20:48.44twisted[asteria]didn't you guys know
20:48.49twisted[asteria]the internet is a big NAT
20:49.11*** join/#asterisk klictel (n=klictel@modemcable185.108-200-24.mc.videotron.ca)
20:49.14znoGwhats the best way to set up a dialplan so that if a call needs to go out to 911, it will check if a channel is available and disconnect a random one if there isn't to make room for the call?
20:49.25[TK]D-Fenderprivalodc : span=1,0,2,esf,b8zs should be span=1,1,0,esf,b8zs
20:49.45rayvdznoG: there's a solution for that on voip-info.org
20:49.45twisted[asteria]znoG, i usually assign a channel for 911 to use
20:49.49znoGchanisavail looks like a good option
20:49.54twisted[asteria]and just hang it up before we dial on it
20:50.06znoGtwisted[asteria]: yea, unfortunately there's not enough lines to assign one for 911
20:50.10twisted[asteria]nono
20:50.14twisted[asteria]i don't mean ONLY for 911
20:50.18znoGoh
20:50.19twisted[asteria]i mean you tell 911 a specific channel to use
20:50.23znoGoh, right
20:50.33twisted[asteria]regardless if it's in use or not, i hang it up, wait 2 seconds, then dial 911 on it
20:50.36znoGso you soft hangup the specific channel
20:50.44znoGbefore dialing
20:50.44rayvdwhat does it do if someone makes a 911 call, is in the call and then someone else makes a 911 call?
20:50.52twisted[asteria]yeah, then if dial fails on it for chanunavail, i just loop until i get through
20:50.54privalodc[TK]D-Fender: tried that also. I was playing with the LBO and timing to seee if it changed something...
20:51.30rayvdhttp://www.voip-info.org/wiki-Asterisk+tips+911
20:51.34Dr-LinuxiCEBrkr: any idea?
20:51.36znoGactually never really thought about this, but from a AGI script using Asterisk::AGI, how do I get the return code for a Dial command? i guess i can do a get_variable CHANSTATUS or whatever it is
20:51.40Umarohey guys.. i'm trying to bring a NFAS PRI (x2) up, and asterisk shows it's sending SABME, and then it receives a DM frame
20:51.47twisted[asteria]znoG, DIALSTATUS
20:52.13[TK]D-Fenderprivalodc : your channel line in zapata is also wrong.  You are missing your ">"
20:52.26znoGtwisted[asteria]: thats the one
20:52.29saftsack__has anyone a zaptel.conf for one hfc card in te mode? ^^
20:53.46*** join/#asterisk jijgeh (n=luken@static-66-182-95-76.bbsc.net)
20:56.01harryvvnasty, call file again crashed asterisk. Well, I know now that thats not going to be a part of any regular asterisk install
20:56.02harryvv:)
20:56.04saftsack__modprobe hfcpci protocol=0x2 layermask=0xf
20:56.05Dr-Linux[TK]D-Fender: i got 3 DID numbers from voip provider on my IP and port, i can get call via them, but i wanna dialout using those number, how can i?
20:56.12saftsack__this is for temode, right?
20:56.25privalodc[TK]D-Fender: Status: Provisioned, Down, Active after reboot
20:57.17*** join/#asterisk ryan (n=ryan@londonderry-cuda2-68-171-162-161.lndnnh.adelphia.net)
20:57.33*** join/#asterisk Bakermd (n=bakermd@exchange.i2telecom.com)
20:58.08rayvdDr-Linux: set your dial plan to set the caller id for you
20:58.09*** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl)
20:58.19rayvdif your voip provider will honor it
20:58.45*** join/#asterisk loud (n=ariel@cypher.punk.net)
20:58.56privalodc[TK]D-Fender: Never had problem using channel=... also zap show channels shows all my 23 channels
20:59.45*** join/#asterisk file (i=file@2001:5c0:8d80:0:2cc9:55bf:839a:96e7)
21:01.21*** join/#asterisk swm_ (n=admin@digitaldatabits.net)
21:01.51*** join/#asterisk RoyK (n=roy@host-81-191-147-248.bluecom.no)
21:01.57lesouvageI'm looking for a routine to enter and validate a future date and time using a phone. Is there a routine (agi or context) that can do this?
21:02.58*** join/#asterisk schuylerdigium (n=schuyler@gateway.digium.com)
21:03.01Dr-Linuxrayvd: what do you mean, i didn't understand?
21:03.55saftsack__what is torisa?
21:04.48*** join/#asterisk _fan_ (n=allan@static-64-222-189-188.man.east.verizon.net)
21:04.52iCEBrkrDr-Linux: callerid= in your sip.conf
21:04.59*** join/#asterisk zishanov (n=mail@HSE-Toronto-ppp3489695.sympatico.ca)
21:05.05*** join/#asterisk Hmmhesays (n=Neg@72.24.227.83)
21:05.48*** join/#asterisk zishanov (n=mail@HSE-Toronto-ppp3489695.sympatico.ca)
21:07.04*** join/#asterisk zishanov (n=mail@HSE-Toronto-ppp3489695.sympatico.ca)
21:07.34zishanovsorry, I got disconneted. I was asking how to check my modules are on which channels on my digium card
21:07.56zishanovI have 4 channels but only two FXO modules installed
21:09.43_fan_Is anyone using * 1.2.1 successfully with QuantumvVoice
21:10.31harryvviCEBrkr anyone here that has experiance with call files?
21:10.55iCEBrkrclsIVRCallFile_inc.php
21:11.01iCEBrkrcreate_callfile.php
21:11.10iCEBrkrI think I have some experience with creating call files :)
21:11.26harryvvhhe
21:11.44harryvvyea web interface is always a better idea.
21:12.03iCEBrkrWho said anything about web interfaces?
21:12.14iCEBrkrJust cuz they're PHP files doesn't mean they're web pages.
21:12.34vmwarezhey, anyone know how to make moh play when transferring someone rather than them hearing rings?
21:12.40harryvvI always associated php with web sites. learn something new every day.
21:12.41harryvv:)
21:12.48azziecan I register a Polycom 501 with two different SIP providers ?
21:12.53iCEBrkrharryvv: you can run PHP scripts at the command line
21:13.12harryvvI have not really worked with php.
21:15.14*** join/#asterisk asdf__ (n=asdf__@c9110469.rjo.virtua.com.br)
21:15.14iCEBrkrharryvv: So what's your deal with callfiles?
21:15.56*** join/#asterisk asdf__ (n=asdf__@c9110469.rjo.virtua.com.br)
21:17.17harryvviCEBrkr I dont usally give up on something. It may become handy for reason such as, calling a group of homes in a area where a known child abduction may have just occured.
21:17.48harryvvI want to see how far it it capable of going under a load.
21:18.07iCEBrkrharryvv: A call is a call regardless how it was initiated
21:18.18*** join/#asterisk seele_ (n=seele@200.124.172.72)
21:18.19iCEBrkrTho, I suspect there might be a little more overhead with faxing
21:18.25dudesharryvv - like a auto dialer that warns about this or that
21:18.30harryvvyes
21:18.36seele_I need help with the password for the panel in AMP Asterisk At home
21:18.37dudesthat's easy to do
21:18.41Umaroguys? what does it mean when the remote PRI switch is sending asterisk a "DM (disconnect mode)" frame?
21:18.44seele_Which is the dafault password??
21:18.46LostFrogOr a way to win radio contests automatically?
21:18.59iCEBrkrLostFrog: lol
21:19.08harryvvdudes, you have made it work as a auto dialer?
21:19.19iCEBrkrLostFrog: I remember doing that with my modem and terminal program
21:19.31LostFrogHow many modems, iCEBrkr?
21:19.34LostFroglol
21:19.46iCEBrkrLostFrog: haha only one.. but the AT command to 'speed' dial was set pretty hi
21:19.48iCEBrkrhigh
21:19.48dudesharryvv - yea
21:19.57harryvvdudes, thats my next stage of testing callfiles with with a recorded voice. Seems that faxing is just not passing right now.
21:19.59iCEBrkrharryvv: I've been working on an automated 'dialer' for the past month.
21:20.06seele_please anyone who knows which is the password for the amp "panel"
21:20.12dudesBut I don't use callfiles
21:20.17iCEBrkrseele_: don't use AMP
21:20.19seele_Flash operator Panel
21:20.21iCEBrkrdudes: why not call files?
21:20.22harryvvWhat do you use?
21:20.24dudesI just set maxlines and it goes nuts
21:20.44seele_iCEBrkr, but i need to acces there
21:20.59seele_Is there another channel for my question??
21:21.02iCEBrkrseele_: Sorry, my personal opinion is AMP sucks.
21:21.04dudesiCEBrkr - not why not, why
21:21.19seele_iCEBrkr, Which one is the best then?
21:21.25Kattymy wrist needs support.
21:21.27Kattyit /hurts/
21:21.28iCEBrkrdudes: Cuz the mechanism for dialing is already there.  Dump the call file and let it go.
21:21.31Dr-LinuxiCEBrkr: why callerid is involved with my issue?
21:21.36dudes<PROTECTED>
21:21.58iCEBrkr< Katty> it /hurts/
21:22.02iCEBrkrThey all say that
21:22.18iCEBrkrDr-Linux: You said you wanted to dial out of your 3 DID with the correct CallerID
21:22.32Dr-LinuxiCEBrkr: no, i didn't said that
21:22.33iCEBrkrseele_: I edit the .conf files in /etc/asterisk.
21:22.56iCEBrkr(03:54)< Dr-Linux> [TK]D-Fender: i got 3 DID numbers from voip provider on my
21:22.56iCEBrkr<PROTECTED>
21:22.56iCEBrkr<PROTECTED>
21:23.05Dr-LinuxiCEBrkr: i said, i wanna dialout on those DID number, i can't dialout ...
21:23.25iCEBrkrDr-Linux: You're not understanding.
21:23.37Dr-Linuxi'm sorry :(
21:23.56*** part/#asterisk popvoxdave (n=popvoxda@pcp0011694398pcs.longhl01.md.comcast.net)
21:24.04iCEBrkrDr-Linux: You dialout via VoIP.. DID's are just numbers, not phone lines.
21:24.23*** join/#asterisk [TK]D-Fender (n=joe@toronto-HSE-ppp4122655.sympatico.ca)
21:24.41Dr-LinuxiCEBrkr: DIDs are just number, nothing else
21:24.47iCEBrkrYeah
21:24.49iCEBrkrwell
21:25.00iCEBrkrThere's more to it than just that, but for you on your end, they're just inbound numbers
21:25.21Dr-LinuxiCEBrkr: i can't dialout via them?
21:26.15[TK]D-FenderDr-Linux : DIDs "land" on a PRI.  You dial out on a PRI circuit, rigging the caller ID if you CHOOSE to. but the DID is just an INBOUND number
21:26.33iCEBrkr[TK]D-Fender: Thanks I couldnt' think of any other way to explain it :)
21:27.02[TK]D-FenderIf you want to place a call with the DID as the callerID, then set it that way in your dial-plan.
21:27.21[TK]D-FenderMine at my desk is set that way, and I even rig my office to look like my home when I trunk calls out from there.
21:27.29[TK]D-FenderiCEBrkr : np :)
21:27.37*** join/#asterisk diclophis (n=diclophi@adsl-69-238-124-226.dsl.pltn13.pacbell.net)
21:27.40*** join/#asterisk P4C0 (n=paco@200.124.22.34)
21:27.41diclophishello all
21:27.43P4C0hello guys!!
21:27.54Cresl1nhello!
21:27.56Cresl1n:-D
21:28.03Dr-Linuxoo ic i understand now
21:28.11privalodc[TK]D-Fender: Never had problem using channel=... also zap show channels shows all my 23 channels. I'm waiting for a call from Sangoma...
21:28.18*** join/#asterisk BugKham (i=BugKham@61.47.108.39)
21:28.21diclophisso.. how do i configure how many digits get sent to my asterisk machine?
21:28.21[TK]D-FenderHello gentlemen!  You have no chance to survive!  Make your time!
21:28.38diclophisright now i have several 8xx numbers coming in... but it is looking for a 7 digit exten
21:28.39*** join/#asterisk gnosys (n=gnosys@griffin2.GnoSys.us)
21:28.39P4C0have anyone tried asterisk 1.2.1 music on hold out of the box?
21:28.47diclophisand i need to make sure it looks for a 10 digit exten
21:29.02Kattymister fender...
21:29.05[TK]D-Fenderprivalodc : Ask for Nenad Korvic specifically if possible.  The guy wrote the docs for them and is going to fix things up for you FAST
21:29.08Katty[TK]D-Fender: make my wrist stop hurting.
21:29.11BugKhamis cmd Mysql working in 1.2.1?
21:29.32*** part/#asterisk schuylerdigium (n=schuyler@gateway.digium.com)
21:29.51BugKhamI had the error "app_addon_sql_mysql.c:115 find_identifier: Identifier 0, identifier_type 2 not found in identifier list"
21:30.11_fan_P4C0: yes
21:30.28privalodc[TK]D-Fender: Right now I'm with David Yat Sin and David Mandelstam
21:30.51BugKhamNever had this problem in 1.0.9 though
21:30.58lesouvagediclopsis: you mean you want to enter a phone number with 10 digits from a phone and asterisks do something with it as soon as you entered 10 numbers?
21:31.10P4C0_fan_, does it works? I'm getting heavy distortion
21:31.34_fan_it worked great - no issues at all with it
21:31.39iCEBrkrHey, can anyone help me get this Asterisk OS loaded on my machine.
21:31.55mrdigitaliCEBrkr: ill help
21:31.56jbroomeasterisk isn't an os
21:32.01iCEBrkrLOL
21:32.04iCEBrkrI'm kidding guys
21:32.11privalodc[TK]D-Fender: What are your tought abour using the other span? Is it possible I just hit a bad one?
21:32.24mrdigitaland by AOS i was assuming AAH :)
21:32.24diclophisno it has to do with the DID
21:32.30_Sam--loading it onto your machine is so 2005!   im thrilled with running it from CF card and USB pen drive
21:32.31diclophisi am only getting 7 digits  from my provider
21:32.33[TK]D-Fenderprivalodc : doubt it, but I've gotta jet for now, bbiab
21:32.38diclophisexcept they say they are sending 10
21:32.39iCEBrkrHow the hell did most of you guys find Asterisk???
21:32.44_Sam--i can take the cf card and thumb drive to any pc and plug it in and have my asterisk
21:32.48iCEBrkrLike, how do you not know about the WIKI?
21:33.01privalodc[TK]D-Fender: Thanks for the help and support
21:33.06diclophisiCEBrkr, i poked around in the dark until i found it
21:33.11P4C0_fan_, do I need to have to have the dsp free?
21:33.12iCEBrkr_Sam--: That's so last year.. We run Asterisk on our Linksys routers
21:33.14ManxPowerI don't suppose anyone has a DirecTiVo they want to sell?
21:33.21lesouvagediclophis: you have a problem with number id?
21:33.40BugKhamanyone using Mysql with Asterisk 1.2.1?
21:33.41iCEBrkrKatty: Problem solved.
21:33.41_fan_P4C0: I believe so
21:33.46diclophiswell.. i think it is related to the DID... i need a 10 digit did (because i am using 8xx numbers)
21:33.50_Sam--iCEBrkr:  what do you for your key drive?
21:33.51ManxPowerdiclophis, it's pretty easy to confirm that.
21:33.54diclophisbut my provider is only sending 7 digit DIDs
21:33.56ManxPowerpri debug span X
21:33.57_Sam--or you store all the config in nvram?
21:33.57diclophis... how so?
21:34.01iCEBrkr_Sam--: It's embeded yo.
21:34.18KattyiCEBrkr: meanie.
21:34.21iCEBrkrBugKham: I use it all the time.
21:34.21diclophiswhoa, thats alot of debug
21:34.28KattyiCEBrkr: why do you pick on me so much?
21:34.32iCEBrkrKatty: Restating the obvious?
21:34.43KattyiCEBrkr: what did i ever do to you?
21:34.46iCEBrkrKatty: I dunno, cuz you crave attention, so I figured I'd give it to ya.
21:34.51_Sam--im trying to sell a real estate office a 5,000 dollar phone system...dont want to walk in with a linksys router...HERE
21:34.55KattyiCEBrkr: is that what you think?
21:35.07iCEBrkr_Sam--: Or a USB thumb drive.....
21:35.15_Sam--thats why i biult a swanky looking machine
21:35.22_Sam--based on a HTPC case, even though it has no moving parts
21:35.40_Sam--that ahanix mce601 case with a cool lcd display that says stupid things
21:35.50iCEBrkr_Sam--: If I found out my phone system ran off a USB thumb drive, I'd hunt you down and beat you with the machine
21:35.52diclophisManxPower what do you think about this?
21:35.54diclophishttp://pastebin.com/483387
21:35.56P4C0_fan_, I'll try to release my dsp device
21:36.06_Sam--no, it runs off the CF card!
21:36.09diclophisi think it is the providers fault I am not getting 10 digits
21:36.36ManxPowerdiclophis, Where are you located?
21:36.45diclophisHayward, CA
21:37.20_Sam--iCEBrkr:  how much rom is there available on the router to work with
21:37.27_Sam--like how much space do you load your asterisk into
21:37.30ManxPowerdiclophis, is this the number that's calling? 9163840441
21:37.39diclophisyes
21:37.46ManxPowerthen the provider is sending it.
21:37.48iCEBrkr_Sam--: It's not a viable solution to a full blown phone system.  There's not enough horse power in that little thing
21:37.52*** join/#asterisk oej (n=oej@apollo.webway.se)
21:37.58ManxPowerdiclophis, put your zapata.conf on pastebin.ca
21:38.05_Sam--i been looking at mini itx for small but powerful
21:38.06diclophisthe number i am trying to all is 866 570 8001
21:38.18iCEBrkr_Sam--: Same here..
21:38.23_Sam--you dont think you could run a small office with 10 SIP clients off the linksys?
21:38.34fugitivo10?
21:38.35saftsack__my asterisk wants to load zaptel and torisa by starting
21:38.37iCEBrkr_Sam--: nope.. Maybe the mini-itx tho
21:38.41saftsack__but i have a noload chan_zap
21:38.53fugitivo_Sam--: the linksys will handle 5 sip clients without transcoding
21:39.07_Sam--good info...thank you
21:39.13*** part/#asterisk _fan_ (n=allan@static-64-222-189-188.man.east.verizon.net)
21:39.14diclophisManxPower http://pastebin.com/483391
21:39.40_Sam--has anyone ever outputted CLI messages to a front panel display?  using something like lcdproc for linux
21:39.47_Sam--i have my front panel work, and it can display system stuff
21:40.07_Sam--but i havent displayed any asterisk info on it
21:40.50ManxPowerdiclophis, tell your provider you require a tech with a T-BERD to come out to diagnose the problem
21:40.50ManxPower< Called Number (len=10) [ Ext: 1  TON: Subscriber Number (4)  NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) '2228001' ]
21:40.50iCEBrkr_Sam--: What's there to output?
21:40.50ManxPowerThat indicated that the number called is 2228001
21:40.50iCEBrkr_Sam--: A bunch of junk that no one but you could understand?
21:40.50_Sam--i would like it basically display the entire CLI console
21:40.50diclophiswhat is a T-BERD ?
21:40.55_Sam--exactly...it is meant to confuzzle potential customers
21:41.03_Sam--"Ohh...i dont know what that means..i shouldnt touch it"
21:41.12iCEBrkr_Sam--: you'd be better off writing a 'status' type script that used the manage port and display certain info.
21:41.14ManxPowerdiclophis, A T-1 and PRI test/diagnostics box.  They can see the DID that the the telco is handing you.
21:41.16fugitivoit'll look c00l for customers
21:41.24iCEBrkrfugitivo: ie. gay
21:41.27_Sam--thats an interesting way to think of it
21:41.31ManxPowerThe tech will say "We are only sending 7-digits", call it in and it will get fixed.
21:41.34_Sam--someone else suggested i tail certain logs to it
21:41.43harryvvis asterisk sms capable?
21:41.49fugitivo_Sam--: i suggest a porno animation
21:41.51diclophisso this is definatly not a problem on my side then?
21:42.31P4C0I have read that asterisk have their own mp3 player? but it keeps calling mpg123... !?
21:42.31Kattyproblems are never on your side. they're always against you.
21:42.54diclophisKatty, ha ...
21:43.03fugitivoP4C0: not mp3 player, native moh
21:43.17gnosysTo ring a Cisco79XX phone with a special ring pattern, I see some good docs at: http://www.voip-info.org/wiki/index.php?page=Asterisk+phone+cisco+79xx
21:43.45gnosysbut there seems to be at least two diff ways of doing it: SetVar(ALERT_INFO=<Bellcore-dr1>) and SetVar(_ALERT_INFO=something).
21:44.05gnosysIn 1.2.1, which works?  and are the <> necessary or there for illusttration?
21:45.05P4C0fugitivo, :) so files in .gsm or such
21:47.32*** join/#asterisk Utah_Dave (n=boucha@0-1pool138-70.nas28.salt-lake-city1.ut.us.da.qwest.net)
21:48.11Tiliwhich country has code 72 or 720
21:48.23*** join/#asterisk circuslila (n=circusli@p549F3727.dip0.t-ipconnect.de)
21:48.28ohadi am trying to set my queue to be between xtime-ytime.. how do i define  that in the queue?
21:50.01*** join/#asterisk darkskiez (n=darkskie@bb-195-172-50-165.ukonline.co.uk)
21:50.46Strom_COW
21:50.48LostFrogTili: Does google not work?
21:51.04Strom_Cit looks like someone removed the protector caps from the T1's appearance on this demarc
21:51.10Dr-LinuxiCEBrkr: my spa problem are fixed
21:51.12Strom_CI touched it with my finger.  ow ow ow.
21:51.20iCEBrkrDr-Linux: cool
21:52.29Tiliwell i thought to ask gurus at asterisk as this is sometimes faster.
21:53.01*** join/#asterisk beefsprocket (n=beefspro@DSL-207.35.14.190.csolve.net)
21:54.30iCEBrkrwelp, I'm going home
21:54.31beefsprockethas anyone who has used iaxcomm with ubuntu feel like giving me a hand?
21:54.36LostFrogTili: 72/720 doesn't look like it is a valid country code.
21:55.42Tiliyeah i thought so too and found so on google and so thougth to ask here
21:56.07P4C0how can I see the call of asterisk to mpg123? (command)
21:56.13gnosysany tips?  anyone?  on getting SIP handsets to ring with distinctive ring patterns?  I know the Cisco79XX has at least Chirp1 and Chirp2 available.. I've heard them with my phones in choosing a ring pattern from the phone config.  But how can I have asterisk ring the phone with one ring pattern or another (to distinguish, say, between internal calls and external calls)?
21:56.43*** join/#asterisk TheCops (i=nobody@got.securebinary.com)
21:57.22*** join/#asterisk _Madar (n=tophe@219-84-129-163-adsl-tpe.static.so-net.net.tw)
21:58.33ManxPowergnosys, what SIP phone?
21:58.44gnosysCisco79XX
21:58.54ManxPowerinfo shoul be in the Wiki
21:59.06gnosysAlso, Polycom... I forget which 601?
21:59.17ManxPowerwell each phone does it differently
21:59.52gnosysright.  Even just getting one to ring with a different pattern would be nice tho'.  I think the polycom's are 501s
22:00.12*** join/#asterisk rculp (n=rculp@66.173.240.20)
22:00.15gnosysI found this: http://www.voip-info.org/wiki/index.php?page=Asterisk+phone+cisco+79xx
22:00.31gnosysBut it's got two sets of guidance, and I can't seem to make either one work.
22:02.33saftsack__is libpir neccessary for asterisk to compile?
22:02.52gnosysno, but libpri prolly is...
22:03.07saftsack__ok
22:05.44tzafrir_laptoplibpri is not necessary if you don't need PRI upport, IIRC
22:06.42rculpI have a quick question for you guys. I noticed that in the asterisk.conf file it has a variable for astspooldir that is defined as /var/spool/asterisk which is all well and good
22:06.50rculpbut I want to be able to split out functions
22:06.57rculpwithout using a symlink
22:07.05rculpso that I can copy the contents
22:07.14rculpof the voicemail subdir
22:07.18rculpto an ntfs share
22:07.23rculpon a different box
22:07.51rculpis it possible?
22:08.30ManxPowerrculp, I don't think so
22:08.33diclophisis there a way to set a phone to 'auto dial' upon picking up of the handset?
22:08.43_Sam--couldnt you just tell the voicemail to be stored on the ntfs share?
22:08.44diclophisin particular a SIP phone
22:09.07rculpI didn't see a variable that set the voicemail path other than the astspooldir variable
22:09.19_Sam--i store my voicemail on a ubs key drive
22:09.23gnosysManxPower, have you had success in altering any SIP handset's ring pattern from Asterisk?
22:09.26_Sam--i think it is a symlink
22:09.33rculpwhich contains dictate, etc...
22:09.41rculpI figured I could do a symlink
22:09.44rculpbut wasn't sure
22:09.46ManxPowergnosys, yes.  The polycom.
22:09.54rculpif there was the ability to just split out the config
22:09.58rculpsymlink it is
22:10.07ManxPowerrculp, you should be able to symlink or mount at the correct point
22:10.14gnosyswould you mind sharing your notes?  I don't even find anything on the polycom phones as I did with the cisco phones (on the wiki)
22:10.56rculpmanx and sam: yeah, I'll just have to do a symlink
22:11.45bn-7bchmm meetme does not seem to work the consle gives me the message  "No application 'Meetme'"   installed asterisk and the demo config yesterday can anyone help?
22:13.06bn-7bcforhet it fount it on a forum
22:13.19fugitivobn-7bc: maybe you don't have zaptel installed, so maybe asterisk didn't compile meetme, so maybe the module app_meetme isn't loaded
22:14.15ManxPowerbn-7bc, Meetme won't be installed if zaptel isn't installed when you build Asterisk
22:16.48gnosysManxPower, mind sharing your notes on distinctive rings with polycom?
22:17.13ManxPowergnosys, I would have to go thru and learn how to do it again.
22:17.40ManxPowerAll the info I used was on the mailing lists or Wiki
22:18.31ManxPowerI am SO annoyed.  I got the new DirecTV DVR, not the TiVo version
22:18.47gnosyswell, i wouldn't want you to do that, but what are you doing in extensions.conf?  Or is it more elaborate than that?  I don't see anything specific to polycom on the wiki for distinctive rings.  I'll keep googling I guess.
22:18.53_Sam--mmmm directivo is the shizzit
22:19.05_Sam--i have a hacked one, can download all the recordings..it runs http, etc
22:19.10_Sam--its just a linux box
22:19.14fugitivoManxPower: why? use linux
22:19.28ManxPowerfugitivo, Huh?
22:19.29fugitivoManxPower: linux+mythtv
22:19.31_Sam--fugitivo:  its harder because its tied directly to directv
22:19.35ManxPowerI don't want any mythtv crap.
22:19.45_Sam--i guess you could still get the program guides somehow
22:19.47_Sam--but not as easily
22:19.53ManxPowerIf I wanted to build my own car out of parts I would have done so.
22:20.02ManxPowerI want a TIVO
22:20.07_Sam--should have bought one on ebay
22:20.09gnosysManxPower, isn't that what you did with your phone system?
22:20.10_Sam--directivo
22:20.14ManxPowerI've had a standalone TiVo for 4 years.
22:20.29gnosysManxPower, I have a TiVo, and I tell you, it SUCKS!
22:20.30fugitivoManxPower: why you use asterisk then? buy a pbx :)
22:20.34BakermdHey all - Question: I have 1 user out of 1000 that has a VoiceMail problem: The user is set to forward messages to an email address and then delete, but when I call his number and VMail picks up, it tells me that the mailbox is full...
22:20.38_Sam--what parts about the directv dvr sucks?
22:20.51_Sam--im just curious
22:20.56ManxPower_Sam--, I've yet to figure out Wishlists
22:21.10ManxPowerAnd I LIKE the TiVo interface.
22:21.17BakermdDec 29 16:24:23 WARNING[6106]: app_voicemail.c:2229 leave_voicemail: No more messages possible
22:21.29trixterhmm CDR billing gets overwritten with FAILED if you call multiple hosts and any one of them is unavailable even if one is avialable and answers the call.  at least a bug is filed on bugs.digium.com about it it should get fixed sometime
22:21.35ManxPowerBakermd, how many messages does the user have?
22:21.40Bakermd0
22:21.51_Sam--trixter:  i noticed that in my cdr logs also
22:21.53fugitivoBakermd: check /var/spool/asterisk/voicemail/context/vmnumber
22:21.59ManxPowerBakermd, check the permissios on their vm directory
22:22.00_Sam--i started making sure my calls were working
22:22.14Bakermdokay - will try those suggestions (Thank you)
22:22.27_Sam--trixter:  it only happens is the status is unavail?  not busy, etc?
22:22.46saftsack__everytime when i start asterisk it tries to load torisa and zaptel
22:22.51saftsack__is that a normal behaviour?
22:23.01ManxPowersaftsack__, no, it's normal for loading zaptel.
22:23.05ManxPowerbut not Asterisk
22:23.13*** join/#asterisk loud (n=ariel@cypher.punk.net)
22:23.24ManxPowerand you can tell it not to try to load all drivers
22:24.07trixter_Sam--: according to the bug report the person had a host unavailable (no route to host) error on theirs too
22:24.37Bakermdwow
22:24.38saftsack__ManxPower, i dont want asterisk to load zaptel. i did a noload => chan_zap.so in modules.conf but it doesnt help
22:24.42trixterI have only notiuced it if I dial multilple targets at the same time and any of  them arent registered (host is dynamic not registered means no ip means no route to host) it does failed
22:25.01BakermdThere were 99 messages in the folder - I was misinformed! Thanks all
22:25.03trixterI dont know if the problem is larger or not, only that I have the same conditions, even if not observed by the person that posted the bug
22:25.08ManxPowersaftsack__, What is the SPECIFIC message?
22:25.23trixterit means that people doing billing based on CDR can have problems actually collecting for calls that are made
22:25.47saftsack__ManxPower, the specific message? i got mass errormsg.
22:25.51saftsack__mom i paste them in pastebin
22:26.23saftsack__ManxPower,
22:26.24saftsack__http://pastebin.com/483441
22:26.30trixterI wonder if a no route to host error would cause a call that was in progress to be marked as failed well after the call was completed ...  that might be interesting - if true (and I dont know that it is) then people could intentionally abuse this to drop calls and not be charged for them
22:26.39ManxPowerso you type "service asterisk start" and you get massive error messages?
22:26.52saftsack__asterisk -vvvvvvvvc
22:27.00saftsack__no this errormessages are from dmesg
22:27.18ManxPowerthose error messages ARE FROM KERNEL DRIVERS
22:27.24trixterit wouldnt be that hard to write a firewall script to do that on a per call basis at least for testing
22:28.23ManxPowersaftsack__, What distro are you using?
22:28.27saftsack__debian
22:28.55ManxPowercan  someone tell this guy how to make the zaptel drivers not load on debian.
22:29.09saftsack__:)
22:29.21saftsack__maybe i should delete them ;)
22:29.21trixterHmm..  wonder if it could be forced ...  a provider that allows multiple registrations to their asterisk box may have that problem as well if one is set up specifically to firewall traffic to the RTP ports and the o ther is set up to actually answer the calls
22:29.26*** join/#asterisk nmsclera (i=nmsclera@188.rigozsaurus.com)
22:30.46vmwarezanyone out there know how to get music on hold to play while transferring someone instead of them hearing rings?
22:30.50*** join/#asterisk tartar (n=tartar@CPE0004e27b716e-CM014370001917.cpe.net.cable.rogers.com)
22:30.52nmscleraAnyone working with a Tekelect T6000 and getting trunking to work?
22:31.12nmscleraTekelec, that is.
22:31.59saftsack__ManxPower, i cant find the drivers in the kernel menu. should i delete the modules by hand?
22:32.18ManxPowersaftsack__, I cannot help you with debian specific stuff.
22:32.49ManxPowervmlinuz, "show application dial"  Pay special attention to the "m" option.
22:32.55saftsack__ok
22:33.02saftsack__thanks
22:33.19saftsack__ManxPower, on what for a distribution is your asterisk running?
22:33.40ManxPowersaftsack__, Mandrake, but most any RPM based distro acts the same way.
22:34.00ManxPower"chkconfig zaptel off" and "service zaptel stop"
22:34.26blitzragethat works the same way in CentOS / Fedora / RHEL too
22:34.36*** part/#asterisk fulgas (n=fulgas@a81-84-117-84.cpe.netcabo.pt)
22:34.38saftsack__yes, rpm distros ;)
22:34.41saftsack__but i dont like them
22:34.51*** join/#asterisk mogorman (i=root@user-24-236-84-48.knology.net)
22:34.59blitzragemogorman: zup zup
22:35.14*** part/#asterisk tartar (n=tartar@CPE0004e27b716e-CM014370001917.cpe.net.cable.rogers.com)
22:36.07*** join/#asterisk Neter66 (n=neter66@CPE0012170db24c-CM0012c9db4f34.cpe.net.cable.rogers.com)
22:36.52saftsack__ManxPower, do you know why modprobe knows what modules i have?
22:36.56Neter66Has anyone seen any issues with Ring Groups in 1.2.1?
22:37.38Neter66If i set a ring group (#1) to have 2 extensions in it, it will automatically go to vm.  (returns from dialparties with no parties to call)
22:37.56Neter66if I set the inbound to go directly to an extension, it works fine
22:38.48Neter66In verbose mode i see:   Executing Macro("IAX2/647722xxxx-1", "rg-group|ringall|20||2000-2001") in new stack
22:39.01PoWeRKiLLI can't get DIALSTATUS variable anymore one * 1.2.1 and it was working on 1.2.0 any other person that got this problem ?
22:39.01slappingtdoes ne1 here use the Sipura 3000?
22:39.10ManxPowersaftsack__, I cannot help you further
22:39.11Neter66so it sees the ringgroup, sees it as a ringall, for 20 seconds, and the two extens.  but right to vm
22:39.14saftsack__ManxPower, ok
22:39.27saftsack__ill test myself and then ill tell you how it worked :)
22:39.27blitzrageNeter66: that line doesn't tell us much
22:39.40blitzrageNeter66: try pasting the macro into a pastebin
22:40.11gnosysCan anyone point me to a *searchable* copy of the * mailing list archives?  All I see at digium is a browseable one...
22:40.23blitzragegnosys: use google
22:40.29_DAWDoes anyone have an opinion on the reliability of realtime in 1.2?
22:40.30blitzragegnosys: site:lists.digium.com
22:40.52blitzrage_DAW: as reliable as the database availability
22:41.13gnosysOk.  Thanks blitzrage.  But is the entire archive stored on that server?
22:41.44Neter66blitzrage: did you want the "IAX2/647...." macro?
22:41.44blitzragegnosys: should be (you mean lists.digium.com right?)
22:41.55gnosysright.
22:42.06*** part/#asterisk mogorman (i=root@user-24-236-84-48.knology.net)
22:42.23*** join/#asterisk NDT (n=me@cpe-24-195-216-41.nycap.res.rr.com)
22:42.58blitzrageNeter66: I don't... but if you want someone to help, they'll need to see the macro since it actually performs the action
22:43.06*** join/#asterisk Katty (n=angela@68.112.15.110)
22:43.18NDTHey guys...Digium 405 cards...is there anyway to tell that Ts are plugged into them remotely or at CLI prompt?
22:43.52blitzrageNDT: zttool?
22:43.54diclophiszttool ?
22:44.38NDTahh..thanks.. 8) sorry been mainly doing straight IP stuff...now trying to move into this realm of asterisk 8)
22:45.34Neter66blitzrage: it's there
22:45.51blitzrageNeter66: you'll have to paste a link here if you want someone to look at it :)
22:46.22twisted[asteria]kattay!
22:46.30Kattythere you are.
22:46.34Neter66http://pastebin.com/483463
22:46.40twisted[asteria]i am
22:47.32MikeJ[Laptop]nope
22:49.28*** join/#asterisk `lyme (n=Lyme@manufacturerstransportation.com)
22:49.45gnosysPolycom 501, ipmid.cfg, http://lists.digium.com/pipermail/asterisk-users/2003-December/031787.html is this file (ipmid.cfg) still used by Polycom phones (as described in the 2-year-old message at this URL)?
22:50.07gnosysThis is about distinctive rings for the polycom
22:52.03`lymeanyone have any idea why parties in a meetme conversion sound like their under water?
22:52.56nmscleraIs anyone doing ANY SIP trunking at an * termination point?
22:53.28Neter66anyone have any clues on the ring-group issue?
22:54.52*** join/#asterisk vexorg (n=vexorg@CPE000021ded913-CM001225419164.cpe.net.cable.rogers.com)
22:55.45*** part/#asterisk mhnoyes (n=mhnoyes@user-2ivfijq.dialup.mindspring.com)
22:55.46*** join/#asterisk mogorman (i=root@user-24-236-84-48.knology.net)
22:56.36*** part/#asterisk mogorman (i=root@user-24-236-84-48.knology.net)
22:57.35mrdigitalhey Katty: you awake enough to help me?
23:00.18*** part/#asterisk vmwarez (n=jjones@216.147.224.254)
23:00.34*** join/#asterisk roulduke_ (i=5x0z5i7k@p508D326A.dip0.t-ipconnect.de)
23:00.37hugo-v6hiho
23:01.29gnosysseems everyone's asleep, huh, hugo-v6?
23:01.39hugo-v6gnosys: seems so ;)
23:01.43*** join/#asterisk Johnnie (n=jdlewis@pdpc/supporter/active/Johnnie)
23:02.10gnosyshugo, have you done anything with distinctive rings?
23:02.17hugo-v6or at work or wtf ever ;)
23:02.32hugo-v6dunno what distinctive rings are ;)
23:03.35gnosysa call hits your asterisk box from a Zap channel, the internal phones ring with one pattern.  an internal call (from one ext to another) hits a phone, and it rings with a different pattern.  Audio clue that this call may be important, should not be ignored, etc.
23:03.46*** join/#asterisk ApEtc (i=apetc@ip68-3-225-51.ph.ph.cox.net)
23:04.38hugo-v6hmmm dunno how that should work
23:05.40hugo-v6i configured my snom 190 with 3 lines to differ 3 phone numbers
23:05.51*** join/#asterisk sigmounte (n=sigmount@www.sighq.net)
23:06.10hugo-v6which sucks hard
23:08.26*** join/#asterisk smirl (n=owned@68-169-204-147.agstme.adelphia.net)
23:09.20smirlwhen i get a call from a queue my phone shows 2 lines ringing for 1 call... anyone experienced this before?
23:09.38`lymehow do you unload the drivers for the x100p cards?
23:09.45smirlformat c: /q
23:09.52*** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage)
23:09.55smirlno jk
23:10.04*** join/#asterisk Defraz (n=t0tal@72.24.26.215)
23:10.05SwK[Work]where's the libpri/zaptel experts at?
23:10.05smirldunno
23:10.28smirli'm not an asterisk expert but i know all about PRI's
23:10.48*** join/#asterisk Johnnie (n=jdlewis@pdpc/supporter/active/Johnnie)
23:10.55blitzrageSwK[Work]: I thought that was you? :)
23:12.24smirlso, could anyone tell me why asterisk is delivering duplicate calls to each agent in the queue?
23:12.55smirli should google first probably
23:13.33blitzragesmirl: sounds like a config error... or maybe you have 2 agents trying to be delivered ot the same locatoin or something?
23:15.23hugo-v6./me is back but still looks bored
23:15.37hugo-v6hmz. gonna configure samba now :/
23:16.28smirlall the agents are static, there should be no login/logout
23:16.53blitzragehrmmm...
23:17.05blitzrageI haven't done much with queues
23:20.03saftsack__MISDN free_device: entitylist not empty
23:20.05saftsack__:(
23:21.37hugo-v6saftsack__: misdn is evil.
23:21.47hugo-v6i know it. i use it too
23:21.50*** join/#asterisk classicx (n=classic_@gb.jb.101.190.revip.asianet.co.th)
23:23.03saftsack__but why?
23:23.28saftsack__i cant get it. i tested the whole evening and it seems, that it wont running in the future
23:23.58*** join/#asterisk MikeJ[Laptop] (n=ircatjer@d14-69-8-30.try.wideopenwest.com)
23:24.03hugo-v6saftsack__: wokrs fine here.
23:24.14hugo-v6at least with * 1.0.10
23:24.15saftsack__hugo-v6, do you have a simple hfc a card?
23:24.23saftsack__and with * 1.2.1?
23:24.25hugo-v61x hfc 1x avmfritz
23:24.28hugo-v6no.
23:24.33hugo-v61.0.10
23:24.36saftsack__the hfc as nt, or?
23:24.48hugo-v6jep hfc as nt
23:25.11saftsack__german?
23:25.13hugo-v6i read that misdn from jolly in version 3.1 will work with *
23:25.16hugo-v6jep
23:26.02hugo-v6with install_misdn from beronet wit wont work with 1.2.x
23:26.10hugo-v6s/wit/it/
23:26.26saftsack__yes i used the beronet install_misdn
23:26.34hugo-v6wont work
23:26.49saftsack__what is the normal errormsg?
23:27.03hugo-v6dunno what u mean
23:27.33saftsack__i have a * 1.2.1 server running with install_misdn in my buero
23:27.35*** join/#asterisk jimmy_deanPB (n=jhodapp@cpe-65-26-179-224.indy.res.rr.com)
23:27.36*** join/#asterisk postel_ (n=jk@area41.OSPF.netmonks.net)
23:27.41*** join/#asterisk MikeJ[Laptop] (n=ircatjer@d14-69-8-30.try.wideopenwest.com)
23:27.54hugo-v6does everything compile fine? no errors? then modules loaded without errors? then chan_misdn loaded in * without erros?
23:28.04hugo-v6then look after misdn show stacks
23:28.06saftsack__yes
23:28.20saftsack__in my buero all runs finde
23:28.21saftsack__fine
23:28.28*** join/#asterisk gnosys (n=gnosys@griffin2.GnoSys.us)
23:28.36gnosyshiho
23:28.37saftsack__but here at home it compiles fine, but it doesnt find any connectivity
23:28.41hugo-v6re gn
23:28.45hugo-v6gnosys
23:28.52gnosys:-)
23:29.20saftsack__hugo-v6, wie aeussert sich es, dass misdn nicht geht mit * 1.2.1?
23:29.26hugo-v6saftsack__: well, configured everything? one card as te another as nt?
23:29.36saftsack__just one card as te here
23:29.47hugo-v6saftsack__: i get errors dur9ing the load of chan_misdn.so
23:29.54saftsack__No Connect port:1
23:30.04saftsack__thats my error. everything is connected fine
23:30.13saftsack__but sometimes i get broken pipe
23:30.16*** join/#asterisk joat (n=joat@ip70-160-150-20.hr.hr.cox.net)
23:30.19saftsack__u 2?
23:30.21hugo-v6misconfigured msidn.conf?
23:30.35hugo-v6not that i remember
23:30.35saftsack__hugo-v6, there isnt something for misconfigure, or? ^^
23:30.55saftsack__;)
23:30.58hugo-v6what do u mean?
23:31.09saftsack__i can paste my misdn.conf on nopaste
23:31.38hugo-v6or pastebin.ca what u like. ill look on it
23:31.50saftsack__http://nopaste.php-q.net/181297
23:32.58hugo-v6looks good
23:33.46saftsack__http://nopaste.php-q.net/181298
23:33.51saftsack__and this here to i think
23:33.57saftsack__back in two seconds
23:34.24gnosysbenjk: you here?  awake?
23:34.27saftsack__telephone is ringing
23:34.35*** join/#asterisk Johnsie (n=jdlewis@pdpc/supporter/active/Johnnie)
23:34.45hugo-v6saftsack__: u use the hfc as te?
23:35.16hugo-v6hmmz. i wanted to configure my samba nor?
23:35.19*** part/#asterisk Johnsie (n=jdlewis@pdpc/supporter/active/Johnnie)
23:35.32hugo-v6and my dog wnats to go gassi.
23:35.33*** join/#asterisk Johnsie (n=jdlewis@pdpc/supporter/active/Johnnie)
23:35.39*** join/#asterisk _cleric_ (n=dacleric@p548290E7.dip0.t-ipconnect.de)
23:36.07smirlok i fgound my queues problem... looks like a bug in AMP, in queues_additional.conf it writes all the settings once, then writes them all AGAIN in reverse order, so all the members are listed twice (along with all the other settgings...\
23:36.41*** join/#asterisk anonymouz666 (n=lynx@ns2.redetaho.com.br)
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23:37.41hugo-v6i dont understand why someone wants to use amp? that "tool" doesnt look userfriendly or helpful
23:38.00*** part/#asterisk Johnsie (n=jdlewis@pdpc/supporter/active/Johnnie)
23:38.10mrtwisterhugo-v6 - amp is good for novice and providing nice gui
23:38.13anonymouz666When everybody will be celebrating the new year I will be asterisking... heh
23:38.16mrtwisteroffer better :)
23:38.21saftsack__hugo-v6, yes
23:38.34hugo-v6mrtwister: dunno. was/are a novice and dont use it
23:39.23*** join/#asterisk Johnsie (n=jdlewis@pdpc/supporter/active/Johnnie)
23:39.29hugo-v6saftsack__: well.. paste the lines where chan_misdn.so is loaded. (start with a lot of -vvv's)
23:39.44smirlhugo-v6, i agree, a customer likes to be able to be able to change some basic settings on their system without having to call the vendor for support. things like names on extensions, voicemail passwords, etc, all change very frequently. and it needs to be accisible to the business manager who isn't going to want to learn using linux, not to mention vi for edting confs
23:39.51*** join/#asterisk [Airwolf] (n=airwolf@attilla.nl)
23:40.25mrtwistergui and usability is most important
23:40.26smirlsorry for bad spelling, i'm using a tablet pc and my handwriting sucks
23:40.36hugo-v6smirl: well... thanks good that i have customers which dont wnat to do that stuff for therself
23:40.53mrtwistersmirl congrats :) how fast your screen will die? :)
23:41.25saftsack__hugo-v6, ok
23:41.26smirlmrtwister, actually, this tablet is really old as it is. i bought it on ebay for $150.
23:41.51saftsack__hugo-v6, http://nopaste.php-q.net/181300
23:42.02saftsack__i tested the cable and i tested another card
23:42.04smirlthe stats suck terrible, it's a 233mhz pentium mmx with 32 megs ram running win98. but i use terminal services to my server, so using it is like using an uber fast server pc...
23:42.09saftsack__the same error everytime
23:42.14hugo-v6smirl: does it use a font like palm?
23:42.21smirldon't use the tablet for anything other than sending and recieving screen info from the server
23:42.27smirland it's wifi, so ...
23:42.35smirli can use my desktop pc from anywhere in the wifi range
23:43.04smirlhugo, yea similar
23:43.20smirlit's actually a bit different than the palm script
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23:43.53*** join/#asterisk Johnnie (n=jdlewis@pdpc/supporter/active/Johnnie)
23:43.58smirlbut for $150 it was WELL worth not having to sit at myh desk all day...
23:44.00*** part/#asterisk Johnnie (n=jdlewis@pdpc/supporter/active/Johnnie)
23:44.04smirland it has an available docking station
23:44.11smirland even with out that, it has USB ports
23:44.19smirlso i can still use a mouse and keyboard if i wanted
23:44.48smirlthe only thing i run on this besides remote desktop is x-lite
23:44.55smirland it actually runs ok
23:45.07smirlthe tablet has a sound card, and i have a headet for it
23:45.29smirlso it's basically like a wireless SIP phone with a huge touchscreen display :-)
23:45.33saftsack__hugo-v6, so why does it run crap?
23:46.25hugo-v6saftsack__: dunno. googled after "No Connect port:1"
23:46.41hugo-v6no hints but "use gcc 3.x"
23:46.51hugo-v6saftsack__: sorry. no idea
23:47.02hugo-v6but its not a config problem as i guess
23:47.26hugo-v6its a compiling/compatibility issue
23:47.48saftsack__yes i think so too
23:47.57saftsack__so i tried visdn but that doesnt compile
23:48.10saftsack__it searches for channel_vhp or something like that
23:48.14hugo-v6i would try 1.0.10 with install_misdn. if that works then go ahead.
23:48.28*** join/#asterisk aless (n=fruribe@pc-100-230-83-200.cm.vtr.net)
23:48.42hugo-v6never got time to test visdn
23:48.53saftsack__ok
23:48.54alesshi. how do i connect two asterisk servers
23:48.57saftsack__but visdn rools
23:48.58*** join/#asterisk RoyK (n=roy@55.80-202-161.nextgentel.com)
23:49.08hugo-v6and as slong as misdn works (and since a few customers have beronet bn4so-cards) i use misdn
23:49.43hugo-v6aless: i.e. via iax. in the demo u see how to connect one of them. ive never done that so read a lil bit ;)
23:50.21alesswhat demo?
23:50.32saftsack__my bn4so card runs well with misdn
23:50.34hugo-v6the sample configs.
23:50.34saftsack__te and nt mode
23:50.38saftsack__but my simple card not
23:50.42hugo-v6saftsack__: same here
23:50.53hugo-v6but still with 1.0.9
23:51.21saftsack__no 1.2.1 works fine too :)
23:51.31hugo-v6well gonna get my pants on and go with the dog
23:51.50saftsack__ok
23:51.51saftsack__have fun
23:52.00hugo-v6saftsack__: i meant it runs here stable with 1.0.9 didnt gave a test with 1.2.1
23:52.01*** join/#asterisk kink0 (n=k@62.37.205.161)
23:52.10hugo-v6ill have ... brb
23:53.10alessthanks :D
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23:56.04mrtwisterLEAVE
23:56.05saftsack__cYa
23:56.08*** join/#asterisk sigmounte_ (n=sigmount@www.sighq.net)
23:56.09*** part/#asterisk mrtwister (n=andrius@cable-10-68.cgates.lt)
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23:56.10*** part/#asterisk mrtwister (n=andrius@cable-10-68.cgates.lt)
23:56.12saftsack__has someone visdn?
23:57.10saitechcan anyone help me, with a problem lige this. Im getting some errors in the debug logger. It keeps spamming "chan_sip.c: failed to grab lock... trying again" its not contant, but it seems like it loops for some time now or then. When it spams me with these errors, it seems like asterisk is hanging a bit, and i cant getting any calls through.
23:59.56kink0what I need to write from an AGI to the stdio to do a Dial(Console/dsp5) ?

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