IRC log for #asterisk on 20070326

00:00.27bulleCrashHD: sweet, then perhaps anyone in here that gets on just fine can explain the pretty basic info, on how to setup asterisk to place certain sip calls over a certain outbound proxy ? that should be a fairly common setup
00:00.51mekongwhat is the best way to strip ${BLINDTRANSFER} so that I am let with technology/exten as opposed to technology/exten-3145 ?
00:01.04bulleJT: the book is great, i have been reading it, but sadly the sip chapter doesnt talk about how to use different oubound sip proxies
00:01.08CrashHDyour proxy has to be setup for it
00:01.20CrashHDyou would setup the proxy info in your sip.conf
00:01.32CrashHDthen SIP/${EXTEN}@proxy
00:01.39mekongthats left with not let with
00:02.13bulleCrashHD: SIP/extension@ekiga.net@sipproxy.com        ?
00:02.37bulleCrashHD: give that i have a sipproxy.com in my sip.conf
00:02.52CrashHDI don't believe asterisk is setup that way
00:03.29CrashHDunless your other proxy parsed the sip headers
00:03.38CrashHDto understand that
00:03.43CrashHDsip debug it
00:03.47CrashHDsee what is actually being sent
00:04.07bulleCrashHD: well, the sip.conf docs has a keyword "outboundproxy" so i think its supported
00:05.40bullejust cant find any documentation about how to use it, and how it relates to what input i feed to Dial
00:06.32*** join/#asterisk tzafrir (n=tzafrir@62.90.10.53)
00:07.11CrashHDhttp://www.sineapps.com/news.php?rssid=1677
00:07.34CrashHDthat took me all of 15 seconds to type asterisk sip.conf outboundproxy in google
00:08.28bulleCrashHD: oh yes, as i said, i alreay know the keyword is supported, but how do i specify the given outbound proxy for my Dial ?
00:08.31bulleCrashHD: that is the question
00:09.16CrashHDfrom what I gathered just glancing, asterisk will use it on the backend when outbound sip calls are made
00:09.34CrashHDso you would specify SIP/extension@ekiga.net
00:09.36bulleCrashHD: yes, but i only want to use the outbound proxy on certain calls, not all calls
00:09.51bulleas i said earlier
00:09.55CrashHDso you setup the outbound proxy for [system1] in your sip.conf
00:10.07CrashHDhmm
00:10.14CrashHDnot sure it will be that dynamic
00:10.23CrashHDyou can use it on a per sip.conf entry
00:10.32bulleSIP/extension@ekiga.net@system1  ?
00:10.49bulleSIP/system1/extension@ekiga.net
00:10.58CrashHDI don't think that is how it was written
00:11.03CrashHDwrite oje on the mailing list
00:11.05CrashHDand ask him
00:11.24bulleye, i think i will actualy mail olle and ask
00:11.37bulleolle is the man, most likely
00:14.33bullei suspect the last syntax is the correct one
00:22.44ManxPowergenerally it is SIP/destinationumber@sipconfentry
00:22.59ManxPowerif you need to specify a username you would use fromuser= in the sipconfentry
00:23.13ManxPowerbut since the outbound proxy stuff is so new....
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00:28.29jeffikAll: need help setting up remote access for SPA-942 thorugh linksys router
00:29.02CrashHD~jbot crickets
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00:46.35lokkju_wrkany of you use vonage with a motorola VT1005v?
00:46.54CrashHDuh oh...better not say "vonage"...verizon might come get you
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00:46.59bkruse_home!
00:47.01bkruse_homelol
00:47.10bkruse_homewhats a good voip provider that does flat rate and not pay per minute
00:47.12CrashHDwith their big enum patents
00:47.58ManxPowerbkruse_home: they all suck
00:47.59lokkju_wrkheh, I need a valid MAC number so I can pull down a firmware image to play with
00:48.18bkruse_homeManxPower: really? should i just pay per minute?
00:48.20bkruse_homefor home use, that is
00:48.23lokkju_wrkso I can whip up some instructions on unlocking it - permenently
00:48.37ManxPowerbkruse_home: ALL ITSPs suck, but per min is actually usually the better deal
00:48.45bkruse_homereally?
00:48.47CrashHDbkruse_home: I find most the time you over pay with flat rate
00:48.49bkruse_homei dont want to have to worry about it, though
00:49.00apturaManxPower is it the tisp or the carrier thay use?
00:49.03lokkju_wrkbkruse_home, forget flat rate - it is actually usually cheaper to go with ppm - try genericvoip, voipjet, and voipstream connect (sp?)
00:49.08bkruse_homei might juts use my pots line to see how many minutes i use now
00:49.20bkruse_homethanks guys :] i might give it a shot
00:49.30ManxPowerI'm a fan of Teliax, but I've not used them since I had to go to satellite internet
00:49.32bkruse_homesometimes the minutes dont seem like much, but im not sure how many realistic mintues i use per month anyways
00:49.33lokkju_wrkvoipjet == dynamically set CID :)
00:49.34CrashHDbkruse_home, I like vitelity personally
00:49.35bkruse_homeyay iax!
00:49.48bkruse_homelokkju_wrk: i likey that :]
00:49.58bkruse_homeima have to rip my phone number away from good ole bell ;[
00:50.02apturacrashev I use them. ocationally there is a issue with vitelity
00:50.12CrashHDaptura: what issues have you had?
00:50.14bkruse_homei dont think itll be that hard, though, since that last law was passed having to do with #'s i thought.....
00:50.28CrashHDwe have 60k minutes a month running over them with very few issues
00:50.43apturammm disconect or the occational delay in the media stream. but its uncommon now.
00:50.54*** part/#asterisk wwalker (n=wwalker@pdpc/supporter/sustaining/wwalker)
00:51.01bkruse_homeim thinking more like........400
00:51.12apturaCrashHD is that for your own company or you a reseller
00:51.24bkruse_homeim getting 1 meg up and down from this random provider in monrovia
00:51.27asteriskguyBladerunner, I would go into the console itself and check to see if you have zap channels loaded
00:51.38*** join/#asterisk jovannotti (n=jovannot@190.84.87.64)
00:51.43jovannottihi to all,
00:51.45CrashHDwe have lots of clients
00:51.51CrashHDbut directly used
00:51.55CrashHDbrb
00:52.01CrashHDvitelity works for 400 as well
00:52.05jovannottiI am looking for a tool to generate h323 calls , someone knows sb about ?
00:52.08aptura400 what?
00:52.38bullejovannotti: ekiga supports h323
00:53.06jovannottiI can generate simultaneous calls from ekiga ?
00:53.30bullejovannotti: simultaneous calls ?
00:53.48jovannottithe purpose is to generate at least 50 simultaneous calls in h323, to test my asterisk server
00:54.13bullejovannotti: he, guess you should have said that then
00:54.21bullejovannotti: then ekiga wont fit your needs afaik
00:54.21bkruse_home400 minutes
00:54.25bkruse_homehow many calls per account?
00:54.30bkruse_homeima rent some out to my aunt with her DID
00:55.10jovannottisorry bullet, do you know some another tool ? If I could generate them via linux comands, I can create a perl scrip to send 50 calls simulataneously
00:55.19bkruse_homeanyone ever transfered their DID from the nasty ole bell south?
00:58.19fetcheryou mean the Nasty New AT&T :)
00:58.27*** join/#asterisk Strom_M (n=pocketir@m040e36d0.tmodns.net)
00:58.34bkruse_homefetcher: yes
00:58.37bkruse_homeanyone/
00:58.49jovannottisomething could help me ? with the tool to generate at least 50 simultaneous call in h323
00:59.03fetcherI've transferred business numbers (ISP dialup pools) to CLECs from them, without any problems.  Nothing involving VoIP so far...
00:59.07bkruse_homecall files?
00:59.07bkruse_homelol
00:59.19bkruse_homefetcher: interesting, i dont think itll be a problem
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01:02.56Strom_Mhello
01:03.23JThi Strom_M
01:03.56Strom_Mhow goes it
01:04.05JTnot too bad
01:04.07JTyourself?
01:04.18Strom_Mdoing well
01:04.40JTgood to hear
01:05.32Strom_Mjust sitting down for dinner :)
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01:08.26JTStrom_M: irc dinners ;)
01:08.40jovannottisomething could help me ? with the tool to generate at least 50 simultaneous call in h323
01:08.49Qwelljovannotti: use asterisk to generate the calls
01:09.53jovannottiI have 1.4 with most of its tools, can you guide me with which command I could generate this calls ?
01:10.15bkruse_homejovannotti: cd /usr/src/asterisk ; vi sample.call
01:10.44jovannottithanks a lot bkruse
01:10.57jovannottiI'll try right now
01:11.33bkruse_homekk
01:13.39bkruse_homethen after you have a good one, woot=0 ; while [ "$woot" -le "50" ] ; do cp newfile.call /var/spool/asterisk/outgoing/call$woot.call ; done
01:13.42bkruse_homei think....
01:14.40jovannottiyes, thanks ! I am lookin the sintaxis to create the files
01:17.22jovannottiI think I'll can test my (*) server. It works fine, but if I receive more than 50 calls ->> seg fault :(
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01:35.08flendersJT: on a PRI, can I create a group of channels? for example sip/01, sip/02 and sip/03 can only access 3 lines? if those lines are busy, they won't try to use other 'lines', even though the other (let's say) 20 are available
01:35.19*** join/#asterisk kupsi (n=kupsi@210.213.101.34)
01:36.14JTyes
01:36.16JTzap groups
01:36.52flendersJT: so it works the same way as channels on a TDM400P?
01:37.17JTi guess
01:37.33JTexcept works better, you'll notice it over time
01:37.49flendersand on extensions.conf I would just use g1, g2, etc?
01:38.00JTyes
01:38.06flenderstoo easy
01:38.08JTeither g or G or r or R
01:38.19flendersg == r?
01:38.24JTno
01:38.30flenderswhat does r stand for?
01:38.36JTround robin != normal group ringing
01:38.43mmartinnarrrrrr
01:38.48JTthere's a wiki article that explains it real well
01:38.56mmartinn~extensions.conf
01:39.12jbothmm... extensions.conf is at http://voip-info.org/wiki-Asterisk+config+extensions.conf, or know as dialplan, or known as extensions, or known as exten
01:39.12kupsihello guys, i have successfuly built an asterisk box. I can configure 2 softphones to talk to each other. Now, my question is: What are the things that I need in order for me to make outgoing calls to our PSTN, also what devices are needed in order for my asterisk box to receive multiple calls from the PSTN? Sorry I'm a n00b.
01:39.14Strom_Cflenders: it would be a better idea to restrict a group of phones to using only a certain quantity of channels rather than dedicating those actual channels to those sets
01:39.46JTwell i think he was suggesting the use of groups
01:39.55flendersStrom_C: that was the plan, actually, just didn't know it was possible
01:40.11Strom_Cflenders: yes, it's totally doable with dialplan logic
01:40.37flendersJT: I thought about groups as that was I thought was possible
01:42.40flendersStrom_C: any idea how I should search for that?
01:42.59Strom_C*shrug*
01:43.10Strom_Cit's something you'd have to hand-cruft
01:44.02flendersah ok, so, I could, based on context, check how many channels are already in use?
01:44.03kupsiT_T
01:44.10*** part/#asterisk mog (n=mog@c-71-207-200-130.hsd1.al.comcast.net)
01:44.42JTflenders: why would you need to do that?
01:44.53JTflenders: why are zap groups insufficient?
01:45.20flendersJT: dunno
01:45.45flendersJT: they're probably the same
01:45.49Strom_CJT: he shouldnt use zap groups in this situation because the telco won't segment inbound calls on their end
01:46.09Strom_Cand also, the telco will always dictate which channel you are to use on PRI, even if you request a different one
01:46.15Strom_Cso zap groups in this case reek of kludgery
01:47.01flendersso, inbound calls could come in on a channel that is "assigned" to another group?
01:47.07*** join/#asterisk mog (n=mog@c-71-207-200-130.hsd1.al.comcast.net)
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01:47.07Strom_Cyes
01:47.11JTflenders: oh definately
01:47.17flendersmakes a lot of sense then
01:47.34JTi didn't know telcos dictated channels for outbound calls
01:48.00Strom_CJT: when you place a PRI call, you request a channel, but the telco can force you onto a different channel
01:48.04*** join/#asterisk Noodleman (n=tuckerm@ip68-0-112-170.tu.ok.cox.net)
01:48.19JThmm ok
01:50.35lokkju_wrkI'm looking for someone that has an active vonage motorola vt1005v ATA - I just need you to either run a tftp command, or to give me the MAC address of your unit - I am attempting to provide some unlock methods for it, and to do so I need to get a new formware image.  someone want to help me?
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02:14.58kupsihello guys, i have successfuly built an asterisk box. I can configure 2 softphones to talk to each other. Now, my question is: What are the things that I need in order for me to make outgoing calls to our PSTN, also what devices are needed in order for my asterisk box to receive multiple calls from the PSTN? Sorry I'm a n00b.
02:20.26gambolputtyfxo card
02:21.03gambolputtyhttp://www.digium.com/en/products/hardware/tdm800p.php
02:23.23Strom_Cor an itsp
02:24.06bkruse_homesuccessfully built an asterisk box??? sh configure ; make && make install
02:24.07bkruse_homelol
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02:26.17CunningPikekupsi: Your choices are an FXO port (something you can plug a regular 1B line into), an ITSP (a company that has an existing connection to the PSTN and to which you connect via the Internet), or a PRI (a multi-channel digital connection to the PSTN)
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02:45.46voltagexAll incoming IAX2 calls are being dropped.
02:45.54voltagexwith a message something like NO AUTH
02:46.07Strom_Cvoltagex: dude, I said I'd help you
02:46.13Strom_Cwill you hang on a few minutes?
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02:47.43Strom_Cok, this is weird....make install on zaptel 1.4 svn branch is dying with "build_tools/genudevrules: line 1: udevinfo: command not found" on a fairly ordinary debian sarge install
02:48.43*** join/#asterisk PakiPenguin (n=uppal@linuxpakistan/admin/pakipenguin)
02:48.51Corydon76-homeudevtools not installed?
02:49.21Strom_Cno such package
02:49.42voltagexStrom_C: I was just coming in here because this is the official asterisk channel
02:49.47Strom_Chow do I tell if I'm running udev or not?
02:49.56buggsyCould someone point me to some documentation that would help me setup SLA?  The doc/sla.txt is too vauge for my brain at this time of night
02:50.03voltagexStrom_C: udev is for >=kernel 2.6
02:50.12Strom_Cyes, I'm running 2.6
02:50.25voltagexwell then it's likely that you have udev
02:50.50Strom_Cwell, "likely" my ass - how do I tell definitively? :)
02:51.04voltagexps aux | grep udevd
02:51.12Strom_Cnope, it isnt running
02:51.25HockeyInJuneAnybody here mix their own music?  My buddy needs someone for a video project.  If your good with stuff like that, please gimme a PM, for some details.
02:51.25voltagexwell afaik it doesn't exist then
02:51.50*** join/#asterisk isamar (i=1000@202.95.221.156)
02:51.54isamarhi folks
02:52.46isamarn1 here ?
02:52.52Strom_Cno, we're all dead
02:52.54Strom_Csorry
02:53.03Strom_Cfuneral begins tomorrow at 11
02:53.04isamar:-) I see...
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02:53.16voltagexStrom_C: also, my machine with udev has /sbin/udevd
02:53.22isamarI felt the smell here...
02:53.29isamar:-)
02:53.36voltagexisamar: sorry, I farted.
02:53.40isamarheheh
02:53.56Strom_Cok, now I have it running
02:53.57isamarjust wondering with anybody is playing with Yate+Sangoma to gimme a hand...
02:54.01Strom_Clet's see if it works yet
02:54.03isamarwith=if
02:54.19isamarhere
02:55.39*** part/#asterisk isamar (i=1000@202.95.221.156)
02:57.20voltagexok, normality resumes
02:58.26Strom_Chey, what do you know, it works
02:58.29Strom_Cthanks voltagex :)
02:59.52voltagexI don't feel like I helped much, but hey, can't turn down thanks from a man named after a phone :P
02:59.57voltagexnp
03:00.00Strom_Chehe
03:00.59voltagexnow, to teh debug logs
03:01.56voltagexStrom_C: are you able to call me on my extension/ip directly?
03:02.25Strom_Ci can if you have a guest iax account configured correctly...
03:02.51voltagexerr, I am an asterisk noob, please explain
03:03.07Strom_Cwell, by default, you generally do
03:03.25Strom_Chave a look in iax.conf at the [guest] section
03:03.30Strom_Cwhich context does that have set?
03:03.57voltagexsorry, not asterisk, trixbox
03:04.01voltagexshite.
03:04.04Strom_C...
03:04.07Strom_Cno
03:04.15Strom_Cget rid of that shit and install asterisk
03:04.28voltagexoh come on, everything else works
03:04.38Strom_Cit's impossible to debuf
03:04.40Strom_Cer, debug
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03:05.21Strom_C~trixbox
03:05.32jboti heard trixbox is junk - avoid.  It is also unable to be supported here. Try joining #freepbx and asking there, or use the trixbox forums at http://www.trixbox.org/modules/newbb/
03:06.22voltagexok, try calling me at 0@124.254.123.181
03:06.30Strom_CSIP, or IAX?
03:06.34Qwellh323
03:06.35voltagexIAX
03:06.51Strom_Cok, i'll do so once asterisk finishes compiling
03:06.57QwellSo, I have a logistics question for you guys
03:07.05Qwelllet's pretend I'm talking about a server room
03:07.05Strom_Cqwell: the answer is cheese
03:07.13QwellAssume the A/C is broken
03:07.14JTi use chan_modem, Qwell
03:07.17JTis this ok??
03:07.18QwellHow do you get hot stale air out?
03:07.25Strom_Cqwell: fans
03:07.31Strom_Ci ran into this problem at ticketmaster
03:07.32Qwellinward or outward?
03:07.38Strom_Cwe had an AC unit that blew up
03:07.57JTfans, or portable AC
03:07.57Qwellfor some reason, my office is *hot*
03:08.07JThire a portable AC
03:08.11Strom_Cwe ended up blowing cool air into the room from outside and had every fan on the floor moved into the server room to circulate it
03:08.12Qwellcurrently I've got a fan blowing out the window, but it isn't helping a whole lot
03:08.20Strom_Cblow cool air into the room
03:08.22Qwellit
03:08.23Qwellerm
03:08.26Qwellit's helping some, I guess
03:08.32Qwellthere is no cool air :P
03:08.38_Vilefigure out an exit for the ac, figure out how to create a negative pressure..... fans, yes...
03:08.40mmartinnDo you have a hallway?
03:08.56Strom_Cblow air onto yourself then
03:08.56mmartinnOpen the door to the hallway too
03:08.56_Vileopen windows
03:08.56Qwellbut no, I tried having air coming in, but it wasn't helping
03:08.56voltagexQuell: run away?
03:08.57QwellStrom_C: that worked, but everywhere but my chair
03:09.08Qwellerm
03:09.16Qwellbut everywhere but my chair was still really hot
03:09.33voltagexQwell: get a new chair?
03:09.38Qwellchair was fine :D
03:09.58Strom_Cqwell: is this at home, or at digigraph
03:10.01mmartinnIf this is mission critical, you rent a portable AC unit until the regular stuff is fixed
03:10.02Qwellhome :p
03:10.19QwellA/C works fine for the rest of the house, but sucks in my office
03:10.26mmartinnSo this is long termZ?
03:10.28mmartinnerr term?
03:10.32Qwellmmartinn: pretty much
03:10.40mmartinnGet a second A/C unit or put more ducts in
03:10.52mmartinnI did the fan thing for a bit; Im' in FL and my office had crappy ventilation in the summer time
03:11.05Qwellthe main duct is right below me, with an outlet right here
03:11.34mmartinnGet the main duct fixed then, or get a seperate fan for outside to pump hot air out, maybe?
03:13.03Strom_Cqwell: I went to homo depot and got one of those giant metal fans
03:13.27mmartinnQwell: You could punch a whole in the door and mount a box fan
03:13.27QwellI think I'm just gonna get a window fan
03:13.34Qwellmmartinn: door is open
03:13.37mmartinnWindow AC unit could help
03:13.49voltagexStrom_C: I'm immature enough to find homo dept funny.
03:14.00Strom_C:)
03:14.11Qwellofftopic, but somebody could easily hack a home depot sign to say that
03:14.18Strom_Coh, totally
03:14.27Strom_Cor, even more amusingly, "The Homo Despot"
03:14.46Qwellwhere you gonna find an s?
03:14.53voltagexwait, physical signs as in wood and metal and plastic or something like an LED sign
03:15.12Strom_Cvoltagex: there's a home improvement warehouse chain in the states called "The Home Depot"
03:15.15Strom_Cwww.homedepot.com
03:15.18JTor neon, if you're a particularly skilled home craftsperson
03:15.23voltagexI know
03:15.49voltagexunlike you USians, I know what the rest of the world is (j/k)
03:15.56*** join/#asterisk DaveCanoe (n=Dave@m815f36d0.tmodns.net)
03:17.23Strom_Cok, finally, it's done building
03:18.31mmartinnsleep!
03:19.44*** join/#asterisk nuonguy (n=john@c-24-6-175-26.hsd1.ca.comcast.net)
03:20.43Strom_Cgah, wtf did you do to asterisk, qwell
03:20.50Qwellbroke it
03:21.13voltagexoh nice, now my provider tell me that IAX calling is broken.
03:21.28Strom_Cvoltagex: that's what "free" buys you
03:21.49voltagexStrom_C: no, see this is the trial account to make sure things work before I pay.
03:22.03Strom_Ci thought this was FWD you were trying to debug
03:22.18voltagexStrom_C: gave up on FWD
03:22.54Strom_Cnone of my damned channel drivers will load
03:23.11Qwellmake distclean
03:23.14Strom_Cand the verbose console output is completely useless
03:23.15Strom_Cah ok
03:23.30voltagexwell I am going now for a bit
03:23.50Strom_Cvoltagex: still want me to call you when i get my system up?
03:23.55*** join/#asterisk Kumbang (n=kumbang@router-wmi.paume.itb.ac.id)
03:24.11voltagexStrom_C: not just yet, sorry, gotta run an errand
03:24.15Strom_Cok
03:24.47voltagexfaktortel.com.au is my provider
03:25.04*** part/#asterisk jeffik (n=Jeff@c-24-7-242-120.hsd1.in.comcast.net)
03:25.04voltagexread that as f..kertel.com.au when I first read it
03:25.23JTyou read right ;)
03:25.32JTthey get angry if you "make too many calls"
03:26.12voltagexJT: I'm limiting myself to VOIP providers that are on the Aussie PSTN gateway
03:26.43voltagexbecause the DIDs I can get are all STD rates.
03:27.22JTvoltagex: "on the aussie pstn gateway"?
03:27.23voltagexI'll be back later, JT if you have any suggestions I'd love to hear them
03:27.24Qwellstd rates?
03:27.31voltagexas in timed calls to that number
03:27.32JTsubscriber trunk dialling
03:27.38voltagexlocal calls are 25c
03:27.44JTstd is a term used in .au to describe national calls
03:27.52JTnational/non local landline
03:28.00voltagexJT: 1300 558 592
03:28.15JTvoltagex: what's that number for?
03:28.24voltagexJT calling voip phones
03:28.34JToh right
03:28.35voltagexalthough I think it's under heavy load today
03:28.40JTwhy don't you just get a DID?
03:28.42voltagexbbl
03:28.51voltagexas I said, STD rates
03:29.46JTnot if you get a voip did
03:29.54JTyou can get a did in whatever state you wanr
03:29.55JTwant
03:30.02voltagexno the point is calling my computer from a landline.
03:30.58*** join/#asterisk aptura (n=jondoe@S010600a0c93f6f7e.vs.shawcable.net)
03:31.10Strom_Coh, hehehehehe, i made an uber nub mistake
03:31.22Strom_Ci turned off "loadable module support" in menuselect
03:31.28voltagexStrom_C: but you *are* an uber nub
03:31.31Qwellnice :p
03:31.32Strom_Cyes yes
03:31.40voltagexoh, ok I shouldn't say anything, I've done that before
03:31.46JTvoltagex: so you call the DID number?
03:31.47voltagex+ turning off filesystem support
03:31.55Strom_Cheheh
03:32.10voltagexJT: it needs to be a local call...
03:32.36voltagexhence the voip gateway, 1300 number is 25c...at least I hope it is.
03:32.45JTvoltagex: ok, maybe i'm missing something, why won't getting a DID in an area where it will be a local call work?
03:33.01voltagexJT: I live in a hole?
03:33.06voltagexJT: Narooma, NSW
03:33.12JTvoltagex: hrm
03:33.21JTvoltagex: have you checked if engin can do it?
03:33.23Strom_Cvoltagex: also, for testing things, calling direct from your IP phone will be cheaper than using your DID
03:33.31JTthey have some of the most PSTN gateways in .au
03:33.33voltagexwhat IP phone?
03:33.34Strom_Cand by cheaper I mean free
03:33.55Strom_Ca softphone maybe?
03:34.20voltagexJT: can I use asterisk with Engin though? I thought I had to use their proprietry stuff
03:34.39JTvoltagex: yes, as long as you are on a Voiper plan
03:34.51voltagexJT: I'll look at the rates later
03:35.07JTalright
03:35.10JTif not
03:35.16JTthere may be other options
03:35.22voltagexI'm not rich :/
03:35.30JTvoltagex: let me know your scenario when you get back
03:35.32JTneither am i
03:35.43JTi like saving money with toll bypass where possible :)
03:48.59*** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn)
03:49.47*** join/#asterisk dc3aes (n=matt@S01060001023fe8ca.no.shawcable.net)
03:50.22*** join/#asterisk bigred (n=ian@75-1-208-180.lightspeed.snantx.sbcglobal.net)
03:50.56bigredis there a way to tell how many active outbound connections i currently have?
03:51.05Strom_C"show channels"
03:51.08Strom_Cor, if on 1.4
03:51.11Strom_C"core show channels"
03:51.49bigredStrom_C: ok, what if they are all going out a single sip provider channel?
03:52.00Strom_Ceach call is a separate channel
03:52.04Strom_Ceven if it's the same sip provider
03:52.14bigredawesome.
03:52.17*** join/#asterisk bmg505 (n=leon@c1-50-2.rndf.isadsl.co.za)
03:52.55bigredis this call available in the manager api. i am not seeing it
03:53.09JTbigred: the "channels" philosphy doesn't quite mesh with sip
03:53.14JTit's more connection oriented
03:53.31*** join/#asterisk kgx (n=kgx@60.234.20.178)
03:53.31aydiosmioconcurrency is only limited by the software
03:53.46aydiosmioby default the switch will accpet any call regardless of its source
03:54.09bigredso i am going to be doing a lot of outbound calling, and i want to make sure that my server doesnt get bogged down, so i want to rate limit the outgoing calls
03:55.29aydiosmioyou can limit eh number of calls per trunk
03:55.32aydiosmiothe
03:55.50aydiosmioconcurrent calls per trunk
03:56.00aydiosmioJT: I mean asterisk trunk
03:56.12JTwith sip?
03:56.14aydiosmioyou know, what trixbox calls contexts
03:56.27aydiosmioSORRY
03:56.30JTi know trixbox sometimes uses the term trunks :)
03:56.48aydiosmioI use freebpx
03:56.56JThaha
03:57.03JTthere there, we have top notch counsellors
03:57.16bigredhrm. so any suggestions on how to rate limit?
03:57.21bigredi guess i could use call files
03:57.27bigredand count the number of files in the queue :-)
03:58.22aydiosmioyou could write a script to monitor the asterisk API for calls, no?
03:58.55aydiosmioor use the CDR (seriously hackey)
03:59.13aydiosmiobigred: why would you rate limit vs. limiting concurrency?
03:59.36bigredi guess i am using the terms interchangably
03:59.53bigredi just want to say "no more than 200 concurrent outbound calls at a time"
04:00.06aydiosmioJT: PEER! That's what asterisk calls them
04:00.17aydiosmiobigred: configure your sip peer with  call-limit = number : Number of simultaneous calls through this user/peer.
04:00.36ManxPowerbigred: there are several features for handling limiting stuff in asterisk  see GROUPCOUNT
04:01.23aydiosmiobigred: also incominglimit and outgoinglimit = Number : Limits for number of simultaneous active calls for a SIP client. Valid only for type=peer.
04:02.33bigredaydiosmio: the problem that i have seen with the call-limit feature is that if i am already at my limit, then the waiting calls will get a busy signal, so i will have to somehow guess the number of retrys
04:03.02ManxPowerbigred: that is why GROUP and GROUCOUNT is handy
04:03.26bigredManxPower: i am looking at that right now
04:03.32ManxPowerbigred: also Dial will give you a HANGUPCAUSE or DIALSTATUS for help you with retries
04:16.42*** join/#asterisk xo8ox (n=pride_32@wsip-66-210-250-2.ph.ph.cox.net)
04:17.04xo8oxanyone up ?
04:19.36*** join/#asterisk Iamnach0 (i=Iamnacho@ip68-103-153-140.ks.ok.cox.net)
04:19.56wunderkinyes..
04:22.27*** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290)
04:39.48*** join/#asterisk wundaboy (n=hixscrip@c-67-171-137-48.hsd1.or.comcast.net)
04:41.10vader--hey any of oyu uy familiar with wiring 66 blocks?
04:41.10*** join/#asterisk reber (n=reber@cl-158.dub-01.ie.sixxs.net)
04:41.25vader--i have a place that has 4 pots lines
04:41.28apturajust a punch down tool
04:41.41vader--they want to put in 10 locations that have access to all 4 lines
04:41.50vader--i am going to run cat 5 to each location
04:41.58vader--and use a pair per connector
04:42.22vader--a 66 punch down block can be setup to handle 2 phone lines right?
04:42.31vader--so i would need two 66 punch down blocks
04:42.57apturaI am upgrading and seems zaptel src is missing autoconfig.h when using make or is there another reason.
04:43.54apturavader you have to look at one but I think the blocks are grouped so more then one phone on the same circuit can ring.
04:44.32apturaI dont think it would be to hard to figure out.
04:44.39sevardvader--: you might want to google or wikipedia a 66 block if you are questioning its ability to handle two lines.
04:45.40vader--this is pots lines
04:45.47sevarduh huh
04:46.36apturasevard ever compile 1.2 and got some zaptel error of missing autoconfig.h
04:46.37aptura?
04:46.44*** part/#asterisk Noodleman (n=tuckerm@ip68-0-112-170.tu.ok.cox.net)
04:47.07bkruse_homenow to sleep ;[
04:47.12sevardhttp://www.wikipedia.org
04:47.25bkruse_homelata all
04:47.33sevardaptura: no
04:47.38*** part/#asterisk bkruse_home (n=kruz@69.73.127.92)
04:47.45sevardare you using a hardware device?
04:48.00apturayou mean a fxo card yes.
04:48.45Strom_Cwoot...I'm totally managing to nub up this skinny firmware upgrade on my 7960
04:48.59*** join/#asterisk brian (i=brian@unaffiliated/brian)
04:49.22sevardaptura: do you have the modules for your card and zaptel compiled before you attempted to compile asterisk?
04:50.04apturaI downloaded the zaptel source from digiums site. Seems the xorcom download failed misserably so downloading the needed source code and do it from scratch :)
04:50.13*** join/#asterisk Kumbang (n=kumbang@router-wmi.paume.itb.ac.id)
04:50.35sevardstart from square one.
04:50.37apturaI have 1.2.16 zap src
04:50.39apturahehe
04:50.59apturagreat
04:51.01aptura:)
04:51.08sevardStrom_C: you using skinny now?
04:51.12apturawell this is a file that is missing :)
04:51.15Strom_Cwell, trying to
04:51.30sevardwhy?
04:51.52JTasterisk was treating him too well
04:51.54Strom_Cwhy not?
04:51.57*** join/#asterisk noli_r (n=chatzill@destiny-mis.d-one.net)
04:51.58JThe needed a challenge
04:52.28sevardI'm just curious, i never hear of anyone using skinny anymore
04:53.26noli_rI need help in configuring E1 DID to asterisk
04:53.28lokkju_wrkI'm looking for someone that has an active vonage motorola vt1005v ATA - I just need you to either run a tftp command, or to give me the MAC address of your unit - I am attempting to provide some unlock methods for it, and to do so I need to get a new formware image.  someone want to help me?
04:53.33noli_rany tutorial available?
04:54.16apturagood luck!
04:55.34sevardskinny sounds pretty cool, central administration, faster for the cisco phones, etc
04:55.43*** join/#asterisk asteriskguy (n=learnast@cpe-75-80-111-113.socal.res.rr.com)
04:55.52Strom_Csevard: yeah, i've been promising qwell i'd use it for some time now
04:55.57JTnoli_r: do you mean an E1 PRI?
04:56.21noli_rEI, as Ive seen on Definity's config its CAS
04:56.43noli_rsorry but a have very little knowlege about E1
04:56.52JTi see
04:56.53Strom_Cbleh, "application upgrade failed"
04:57.01voltagexI'm baaaaaaaaaaaaaaaaaaaaack
04:57.03JTnoli_r: what will asterisk be connecting to over E1?
04:57.11Strom_Chi voltagex
04:57.41voltagexJT: can I /msg you?
04:57.47JTvoltagex: i suppose
05:01.18*** join/#asterisk noli_r (n=chatzill@destiny-mis.d-one.net)
05:01.38noli_r:(
05:01.47noli_rgot disconnected
05:03.02noli_rJT: can you please point me to some good tutorial?
05:04.21*** part/#asterisk mog (n=mog@c-71-207-200-130.hsd1.al.comcast.net)
05:05.43*** join/#asterisk neoalex (n=neoalex@user-0ccenic.cable.mindspring.com)
05:05.48*** join/#asterisk b0on (n=b0on@cpe-66-61-165-172.indy.res.rr.com)
05:06.14neoalexhey guys... why is it that my asterisk is running two identical mpg123 processes for my MOH
05:06.33neoalexI'm using an online radio as my MOH
05:07.53[TK]D-Fenderneoalex, Perhaps you left a MOH definition with an mpg123 mode still in existance yeet not attached to a device...
05:08.45neoalexI didn't I checked, plus the processes are identical... meaning they're playing the same URL
05:09.30b0oncould you maybe not have commented out a previous MOH?
05:09.57neoalexnope... I checked
05:10.03vader--tkd work much with 66 punch down blocks?
05:10.15Strom_Cvader--: i punch those down all the time
05:10.32b0ona whole main frames worth
05:10.51vader--i have a place that has 4 pots lines
05:10.52b0on:)
05:11.01vader--and they want to have 10 jacks that have all 4 lines available to them
05:11.15Strom_Cvader--: simple
05:11.16vader--if get a 66M block i can put two pots lines per right?
05:11.25Strom_Cvader--: get one pair of blocks
05:11.31Strom_Con one block, have the telco lines terminate
05:11.33vader--so i would need to 66 blocks
05:11.39Strom_Con the other block, have the station wire terminate
05:11.43vader--to =two
05:12.14Strom_Cthen run jumpers from the telco block to the station block, using the non-cutting side of the 66 blade to have the line show up on multiple station cables
05:12.25vader--ya
05:12.34vader--like a zig zag down the blocks
05:12.41Strom_Cwell, down the ONE block
05:12.42vader--on the innter channels
05:12.57Strom_Cyou should only have one jumper per line
05:13.09*** join/#asterisk Noodleman (n=tuckerm@ip68-0-112-170.tu.ok.cox.net)
05:13.24neoalexb0on: just to be on the safe side, I just copied only the class I wanted in musiconhold.conf, and reloaded and it still started two mpg123s
05:13.41b0onthats crazy
05:13.47b0oni've never seen that happen b4
05:14.00neoalexle'me just restart asterisk alltogeather
05:14.03b0on(not saying its not tho)
05:14.20Noodlemanquestion: do payphones still use dtmf to signal coins, and could asterisk/zap device/whatever pick those up?
05:14.32b0onpayphones don't no
05:14.37Strom_CNoodleman: the phones use 1700+2200
05:14.42Strom_Cnot "DTMF"
05:14.52Strom_Cand yes, in theory, one could hack the zap driver to recognize it
05:15.06b0onsetting up some SIP payphones? :)
05:15.10Noodlemanhrmzz
05:15.18Noodlemanb0on: i'm thinking about it :-)
05:15.21voltagexmore like an asterisk redbox
05:15.28*** join/#asterisk Beirdo (n=gjhurlbu@unaffiliated/beirdo)
05:16.10Noodleman*gears turn upstairs"
05:19.29neoalexb0on: http://paste.biz/paste-1239.html see if you think there's anything wrong with my musiconhold.conf
05:20.22neoalexreally weird
05:20.34b0onyou restarted *
05:20.37b0on?
05:20.43neoalexyeah
05:20.48neoalexstop gracefully
05:21.01neoalexkillall mpg123 (none left, but just to make sure)
05:21.09neoalexthen asterisk again
05:21.17neoalexstarted two mpg123s again
05:25.27*** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25)
05:26.05*** join/#asterisk voiper1 (n=luke@ozvoip.dsl.onthenet.net)
05:26.24b0onwow
05:26.30b0oni'm sorry man.. its beyond me
05:27.33andrew`i once had something like 75 mpg123s
05:28.05neoalexha... so I guess I'm a lightweight
05:28.10JTto correspond with a load average of 75? ;)
05:28.24b0onmine only opened up one with that conf
05:28.27b0onso... i know its not that
05:29.00neoalexweirder
05:29.08vader--what the price of cat5e plenum 1000ft going for?
05:29.22neoalexforget about it... it's not that big of a deal anyway
05:29.27neoalexle'me ask you something else
05:29.40neoalexjust noticed AsteriskNow exists
05:29.53neoalexcan I install the interface over my existing asterisk
05:30.08JTasterisknow is a distro that wipes everything
05:30.10*** join/#asterisk fx0 (n=ariel@cypher.punk.net)
05:30.15JTasterisk-gui is the interface
05:30.21b0onwhat distro u running?
05:30.45b0onoh. asterisknow? never used it
05:31.02b0oni have used freepbx and i've really really liked that
05:31.12vader--how is asterisk-gui?
05:31.54*** join/#asterisk `p4r14h (n=j0sh@69.92.145.178)
05:32.02neoalexI'm using it, and I wanted to put freepbx but the guys supporting it said, migrating the current configs I have would be a royal pain
05:32.18neoalexI'm using slackware I meant
05:32.55neoalexok... so JT, is asterisk-gui downloadable and instalable sepparately
05:33.52JTit is, but i don't use it
05:34.24b0onhey neoalex.. look in your usr/bin .. check out the symlinking for mpg123
05:35.29*** join/#asterisk ShadowHntr (i=sentinel@wikipedia/Shadowhntr)
05:35.32neoalexno symlinks, only the executable itself
05:36.02neoalexJT would I be able to use the current configs?
05:36.11JTno idea
05:36.33b0onhave u tried executing the program itself outside of asterisk, to see if it opens up the dual processes
05:39.14neoalexno... just a sec
05:40.24neoalexyes it does... ok then it's not an asterisk issue
05:41.22vader--what is a good 4 pots line phone?
05:41.32JTwhat
05:41.37JT4 pots line phone?
05:41.48vader--ya
05:41.53vader--analog phone lines
05:42.01vader--pots = plain old telephone systems
05:42.20JTi know what it stands for
05:42.28JTwhat do you mean 4 pots line phone?
05:42.46vader--a phone that can handle 4 lines
05:43.17JTthose sort of things are relics of the past
05:43.23JTand pretty rare these days
05:44.17voltagexebay it!
05:45.02JTyeah that's probably where he'll need to go
05:45.21vader--ya im not sure what to recommend to this company
05:45.29vader--they want to have all 4 lines on every desk
05:45.30b0onmight get lucky at a Salvation Army or goodwill
05:45.37vader--and they have to be pots
05:45.50b0onwhy not setup some channel banks
05:46.01vader--they don't want pbx or voip
05:46.03b0onand have them connect to said four lines thorugh each computer at the desk through a softphone
05:46.09b0onah
05:46.09JTvader--: can't you just say to them "you guys are crackheads!"
05:46.10b0onwell
05:46.35apturaodd 4 lines to every desk and no pbx?
05:46.42vader--it's only 4 desks
05:46.46apturaokay
05:46.48JTsometimes a client needs to be told when they're dead wrong
05:46.56JTthat's just a stupid setup
05:47.06b0onheh.. is the answering machine micro-cassette based too?
05:47.06vader--and they can't use voi because one of the line is used for programming some sort of hardware panel
05:47.11vader--and voip doesn't workwell with it
05:47.12apturai could have seen this as a possible wiring nightmare if it was big :)
05:47.23vader--and one of the other lines is a fax line
05:47.23JTwhat hardware panel?
05:47.32vader--security alarm
05:47.37JTah
05:47.48JTpeople should not be making calls on those lines anyway
05:47.51apturawhat kind of biz is this?
05:48.13apturadoes the alarm have a panic button?
05:48.19neoalexvader--: if they still want to be in the dark ages, then here: http://www.nextag.com/4_-_line-phone/search-html
05:48.22apturaprobebly not :)
05:48.27b0onya.. it just says .. "That was easy!"
05:48.31vader--monitoring of the alarms is done by another company
05:48.34JTyou can get 5 line analogue phones
05:48.35vader--they just do some sort of programing
05:48.37JTi have one at home
05:48.42JTmediatrix or something
05:48.43*** join/#asterisk flip123 (n=flip@ip70-162-47-138.ph.ph.cox.net)
05:48.50JTsitting in its box
05:49.05apturaI can see the fire alarm going off and sending a signal to the ununicator panel then trying to dial out when the line is in use. big problem.
05:49.07flip123sigh. Some days
05:49.27flip123upgraded to 1.2.17, because of the skinny/chan_sip vulns
05:49.38flip123now the audio goes all choppy after about 5 seconds
05:50.34flip123though, I did a kernel upgrade at the same time, I suppose I could blame that
05:50.40vader--ya i need 8 rj 45 connectors and 32 6 pin connectors
05:50.45vader--with wall plates
05:51.42flip123hmm
05:51.53flip123looks like there are some reports of choppy sound on 2.6.9-42
05:51.54*** join/#asterisk oej (n=olle@apollo.webway.se)
05:52.13flip123but its in reference to vmware, which I don't use
05:52.14vader--fucking cat5e is exspensive as shit
05:52.32b0oncopper in general
05:52.38vader--ya
05:52.55vader--1000ft plenum is close to 250-300$
05:53.00b0onholy crap
05:53.05JTlol, plenum
05:53.11JTthat's why
05:53.14vader--ya
05:53.18b0onreminds me of college
05:53.20vader--well it's in a ceiling
05:53.25JTworld copper prices are high though
05:53.46neoalexvader--: http://www.warehousecables.com/cgi-bin/shopper.cgi?keywords=%22cat%205e%22%20and%20solid%20and%20utp%20not%20plenum&search=action
05:53.55neoalex$108 for 1000 ft
05:54.03JTneoalex: he needs plenum
05:54.28neoalexoh... yeah that's 210 there too
05:55.18b0onok heres what you do.. you find a new some construction going on for new office buildings..
05:55.36b0onsneak in at night.. swipe a couple boxes of plenum
05:55.42flip123yarg, anyone else reported issues with 2.6.9-42 and asterisk?
05:56.48JTflip123: why not just upgrade?
05:57.14neoalexb0on: great idea
05:57.23flip123JT: asterisk or the kernel?
05:57.27JTkernel
05:57.27voltagexwhat's plenum?
05:57.31JTthat's a very old kernel
05:57.32flip123I'm trying to stick with the centos packaged kernels
05:57.36JTpfft
05:57.46voltagexhang on, trixbox uses centos
05:57.55flip123voltagex, lots of people use centos
05:58.05JTi don't :)
05:58.09b0onya.. not that great of an idea.. my mind is gone, and i'm up writing design doco's
05:58.12JTand screw pre-packaged kernels
05:58.14*** join/#asterisk MooingLemur (n=troy@unaffiliated/mooinglemur)
05:58.16flip123and lots of people dont :-)
05:58.17b0onand not watning to
05:58.17neoalex<PROTECTED>
05:58.18neoalex<PROTECTED>
05:58.29JTplenum refers to airspace
05:58.49voltagexoh, it's a brand?
05:58.51JTno
05:58.56JTit's a specification of cable
05:59.03*** join/#asterisk tengulre (n=tengulre@222.90.66.10)
05:59.07JTthat has a fire-resistant protective jacket
05:59.13JTrequired by a lot of building codes
05:59.31neoalexyoup
05:59.34voltagexI want some
05:59.37neoalexplenum space is the airspace
05:59.51neoalexplenum cable goes into the plenum space and must have that jacket
05:59.56voltagexthen if I have a pyromania outburst, at least my cable's ok
06:00.12Strom_Cit's not that the jacketing is fire-resistant
06:00.22neoalexit's not ok... it's just slightly better off then regular cable
06:00.25Strom_Cit's that the jacketing doesn't produce toxic smoke
06:00.36Strom_Cor at least, smoke that's as toxic as non-plenum cable
06:00.36rudholma plenum in a building is anything that has positive air pressure with respect to the rest of the building, in this context "plenum cable" means cable rated for use in building plenums.
06:00.41neoalexit's both Strom_C
06:00.53voltagexok, if I go pyromaniac, I won't breathe toxic smoke from my cables.
06:01.12rudholmit's not really for you, it's to protect firefighters mostly.
06:01.17neoalexyeah... that's why I used it throughout my apartment too
06:01.28neoalex:))
06:01.35voltagexoh ok, I don't matter, only the firefighters do.
06:01.41*** join/#asterisk zeeesh (i=zeeesh@202.38.55.125)
06:01.42zeeeshhi
06:01.53neoalexthey hav oxygen anyway
06:02.00rudholmwell, the idea being that in a commercial building, the occupants evacuate quickly, but the firefighters do not
06:02.13voltagexyeah.
06:02.13b0onheh .. start free-basing your plenum...
06:02.15rudholmbut yes, you don't count as much as a fireman :)
06:02.20voltagex:P
06:02.31tengulrehi,all
06:02.36neoalexaaaaw.... that's gotta hurt voltagex :D
06:02.43voltagexFirefighting outfit: +10 of worthiness
06:02.56tengulrewhat is different between codec g723 and g723A?
06:02.57voltagexneoalex: I've been in here for a while, I'm getting use to it
06:03.08neoalex=))
06:03.20flip123JT: yeah, generally I agree, prepackaged kernels suck.
06:03.46flip123but with remote boxes, sometimes its just the way to be
06:04.09flip123not that it helps any tonight
06:04.22voltagexflip123: I don't actually know what hardware is in this box so the latest ubuntu kernel suits me fine
06:05.00flendersroot@pbx:/etc/asterisk# uname -a
06:05.01flendersLinux pbx 2.6.8-3-686 #1 Tue Dec 5 21:26:38 UTC 2006 i686 GNU/Linux
06:05.05flenders:D
06:05.10flendersand it works just fine
06:05.29neoalexLinux pbx... what's that
06:05.38voltagexneoalex: hostname
06:05.43apturaI once worked at a In route ATC center and the fire/hvac system is interesting. The main ATC control room is supplied with positive air pressure. So the entire building except that room can be on fire and the controlers should still do there job. Only in a last second case do the controlers leave but then all inbound air traffic comming into Vancouver would be in limbo. 50 miles of wires in that one building ;)
06:05.44flendersneoalex: that's my hostname
06:05.55neoalexoh... sorry... /me dumb
06:06.00flip123flenders, except it has multiple locate root vulnerabilities
06:06.06flip123s/locate/local
06:06.19flip123trival ones
06:06.28flendersflip123: no one has access to it
06:06.38voltagexflip123: why are you using something that old?
06:07.01flip123voltagex, I'm using 2.6.9-42
06:07.18rudholmflip123: that's like, so 10 minuts ago
06:07.28voltagexah, it was flenders with the really old one
06:07.30flip123because its the latest kernel available for centos/rhel
06:07.33voltagexstill, yours is old too
06:07.35neoalexvoltagex: I use 2.4.33.3 if you think that's old
06:07.38neoalex:D
06:07.44flendersvoltagex: debian stable
06:07.45flip123yes, i know its old
06:07.59flip1232.6.9 is, supposedly, stable
06:07.59voltagexneoalex: isn't that the latest 2.4 stable?
06:08.01*** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk)
06:08.10voltagexflenders: aaah. makes sense now.
06:08.12*** join/#asterisk wundaboy (n=hixscrip@c-67-171-137-48.hsd1.or.comcast.net)
06:08.22voltagexflenders: that's cutting edge for debian.
06:08.23neoalexyes it is, but it's 2.4 not 2.6, therefore it's old
06:08.23flip123flenders, wow, debian stable has moved onto 2.6?
06:08.35flip123hasn't 2.6 only been out for 3 years or so?
06:08.41flenders:D
06:08.43voltagexmeh, 2.4 works better in some cases
06:09.03neoalex2.4.34.2 is actually the latest
06:09.26neoalexmine is whateva came with Slack 11.0
06:09.48flip123so I assume there's no work around for the alleged 2.6.9-42 issues?
06:09.51flendersit doesn't bother me... as I said, no one access it
06:10.43flip123flenders, nod, as long as you have no services exposed to the network, you should be fine
06:11.32flendersflip123: exactly... all you can reach on this box is SIP and ssh from my box
06:12.01vader--yo are you guys familiar with number porting?
06:12.02voltagexdo I need a DNS server on an Asterisk box?
06:12.29flip123voltagex, no
06:12.43*** part/#asterisk b0on (n=b0on@cpe-66-61-165-172.indy.res.rr.com)
06:12.44vader--there is a number this one person owns and this person wants to move it to another person's account
06:12.52vader--verizon says they can't port the number
06:13.26Strom_Cvader--: is it across rate center boundaries?
06:13.38vader--same city
06:13.48Strom_Cdoesnt matter
06:13.52Strom_Cis it across rate center boundaries?
06:13.58vader--i dunno
06:14.31flip123guess I'll try downgrading to 2.6.9-34
06:14.36flendersaren't numbers tied to exchanges?
06:14.40Strom_Cvader--: which city
06:14.51Strom_Cflenders: numbers are tied to rate centers
06:14.56vader--philadelphia, pa
06:15.33voltagexdoh, just went to make a call and realised asterisk isn't installed yet
06:15.38Strom_CPM me the area code and prefix of the number you're trying to port, and the number on the account that you're trying to port it to
06:16.01flendersStrom_C: had no idea what a rate center was
06:16.32voltagexquestion...what's involved in connecting a mobile phone up to asterisk?
06:17.14*** join/#asterisk Avochelm (n=damien__@gw-morphett.koalatelecom.com.au)
06:18.29neoalexvoltagex: a SIP GSM gateway
06:18.30neoalexlike this one:
06:18.30neoalexhttp://www.gsmsave.co.uk/VOIP_GSM_SIP_Gateway.htm
06:18.30voltagexneoalex: I was told it was possible using bluetooth
06:18.39JTflenders: rate center is an american concept
06:18.42*** join/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl)
06:18.51flendersJT: how do we call it here?
06:18.54neoalexhmmm... I'm not sure voltagex
06:18.58JTno idea
06:19.04JTvoltagex: chan_cellphone
06:19.07*** join/#asterisk k-man_ (n=jason@unaffiliated/k-man)
06:19.08JTand a bluetooth adapter
06:19.18k-man_anyone heard of the epygi voip pabx system?
06:19.24neoalexnow I am... here: http://www.voip-info.org/wiki-Asterisk+Bluetooth+channels
06:19.30JTheard of it
06:19.31JTit costs a bomb
06:19.41k-man_jt, does it?
06:19.49JTyes
06:19.55JTi don't see the point in it
06:19.59k-man_my wifes work is thinking of getting it - and im trying to convince them to go asterisk
06:20.09JTshrug
06:20.13JTup to them i guess
06:20.19voltagexJT: will that kind of thing work over USB1?
06:20.21k-man_the vendor says asterisk and epygi will have similar cost of install, but asterisk has a higher cost to maintain
06:20.28voltagexas always
06:20.34JTthe question i'd ask is who would be maintaining the asterisk
06:20.42k-man_jt, they would
06:20.45neoalexvoltagex: http://www.voip-info.org/wiki-Asterisk+Bluetooth+channels
06:20.53k-man_jt, either way, they would be maintaining it
06:21.08JTvoltagex: i don't see why not, bluetooth isn't high bandwidth
06:21.44voltagexok
06:21.54voltagexunfortunately only usb1 ports on this old box
06:22.31neoalexthe problem with bluetooth is you obviously have to be close to the box
06:22.31*** join/#asterisk Mahmoud (n=fake@unaffiliated/mahmoud)
06:22.32voltagex<PROTECTED>
06:22.51voltagexI'll just buy the cheapest bluetooth enabled phone I can find on ebay.
06:23.21neoalexoh... you want to use your cellphone service as a peer?
06:23.47voltagexneoalex: if you mean to receive SMS to asterisk, yes
06:23.52k-man_is it possible to run asterisk on an embeded machine with no hdd and no fan?
06:24.04k-man_or no moving parts hdd at any rate, flash is fine
06:24.24neoalexk-man_: yes
06:24.38Mahmoudany one tried this out? http://www.digium.com/en/products/hardware/asteriskappliance.php
06:24.45MahmoudAsterisk APpliance
06:24.45k-man_neoalex, is there any hardware you can point me to?
06:25.04*** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25)
06:25.29neoalexk-man_: Mahmoud just did, that one is a pretty good choice
06:25.44neoalexif you want to install it on one yourself then...
06:25.47neoalexwait a sec
06:26.15neoalexhttp://www.embeddedarm.com/epc/ts7400-spec-h.htm
06:26.15Mahmoudi'm actually looking at Sipura 3000
06:26.23*** join/#asterisk ftexcom (n=ftexcom@14.Red-80-26-4.staticIP.rima-tde.net)
06:26.33neoalexI meant the asterisk appliance
06:26.34k-man_thanks
06:26.37Mahmoudsipura is the best for my choice, one fxs/fxo port
06:27.08neoalexk-man_: openwrt also has an asterisk package I believe
06:27.19k-man_neoalex, ok, thanks
06:27.25Mahmoudneoalex, what's this?
06:27.37neoalexwhat's what?
06:27.48Mahmoudthe chip you posted earlier
06:28.24flip123oh thank goodness
06:28.38flip123a downgrade to 2.6.9-34.0.2 solved the problem
06:28.43flip123(choppy audio)
06:28.46neoalexit's an embedded device
06:28.56flip123which means tomorrow, when this customer goes to show me the issue
06:29.14flip123I can continue to blame his internet service provider
06:29.15neoalexhas an ethernet, 2 USB 2.0 and an SD reader
06:29.18k-man_flip123, please tell me about your choppy audio problem , i have been having the same problem
06:29.29k-man_flip123, was it the voice prompts that were choppy?
06:29.31Mahmoudneoalex, how is it related to asterisk appliance?
06:29.33flip123k-man_, i upgrade to kernel 2.6.9-42
06:29.36flip123k-man_, yes
06:29.50flip123k-man_, anyway, i had upgraded to kernel 2.6.9-42
06:30.14flip123and the audio got super choppy
06:30.19flip123and would degrade over time
06:30.19neoalexI was responding to k-man which was looking for a no fan no moving parts device for asterisk
06:30.19k-man_flip123, which distro?
06:30.19flip123k-man_, centos
06:30.21flip123I downgraded to 2.6.9-34.0.2
06:30.27flip123and the choppy audio is fixed
06:30.31*** join/#asterisk voltagex (n=voltagex@124-254-123-181-dsl.ispone.net.au)
06:30.35Mahmoudneoalex, do you use this embeded computer?
06:30.36k-man_neoalex, how would you hook the ts700 up to pstn lines?
06:31.18neoalexyou wouldn't, you'd have to go SIP only
06:31.39flip123now to think of anything else I should fix before I have to talk to this customer tomorrow
06:31.54k-man_neoalex, ah, ok
06:31.56Mahmoudk-man_, buy Sipura 3000, FXS/FXO PCI cards, or asterisk appliance
06:32.13Mahmoudk-man_, if you want one fxs and one fxo port, then Sipura 3000 is the best choice
06:32.31neoalexMahomoud the TS7400 deosn't have a PCI bus
06:32.51k-man_ok
06:33.03Mahmoudneoalex, i'm not talking about your embeded computer
06:33.21Mahmoudneoalex, TDM400 has pci bus
06:33.29flip123i could be a real bastard and drop all these test calls out of the billing system
06:33.31neoalex<k-man_>neoalex, how would you hook the ts700 up to pstn lines?
06:33.33neoalexhe was
06:33.36voltagexStrom_C: you still around?
06:33.42Strom_Cyes
06:33.47neoalexnevermind...
06:33.47flip123In fact, I am a real bastard, and I am going to do that
06:34.03voltagexany compile/configure options I should use for asterisk?
06:34.20voltagexshiny new ubuntu server install is ready
06:35.20Strom_Cvoltagex: go with defaults for now
06:35.27voltagexok
06:36.25voltagex17mb...I am going over my download quota this month
06:36.46flendersvoltagex: who are you with?
06:36.47voltagexjust installing build-essential then I'll compile.
06:36.59voltagexflenders: australian isp, Southern Phone
06:37.18flendersvoltagex: i'm in .au too
06:37.38voltagexflenders: ah, so we can bitch and whine about broadband here together!
06:37.46flenders:D
06:37.53JToh no
06:37.57JTnot another whirlpool
06:38.14voltagexJT: oh come on! we're justified in whinging
06:38.21flendersvoltagex: well, I don't have much to complain about... I guess adsl2 plans are good for what you pay for them
06:38.27JTmy broadband is fine :)
06:38.37*** join/#asterisk Kizmet (n=Kizmet@AeriaSolutionsPtyLtd.fe0-1.aes-brd-0.agl.cbr.as-ip.net.au)
06:38.43flendersJT: which one?
06:38.44voltagexyes, you lucky people who can get ADSL2
06:38.57flendersvoltagex: :D
06:39.22flendersI have 2 x internode and 1 x iinet here
06:39.27flendersand with iinet at home too
06:39.55tzafriraptura, here?
06:40.42Kizmet2x Internode Syncing at 23821/2282 bound using a Cisco 3620
06:40.47Kizmet*grin*
06:41.29voltagexKizmet: how the f.... did you get that high sync
06:41.40voltagexare you sleeping in internode's dslam?
06:41.54Kizmetvoltagex, If I stick my head out my windo I can hear the exchange humming
06:41.59voltagexeven APC didn't get that high (I don't think) in a test lab!
06:42.06voltagexKizmet: you lucky bastard
06:42.14Kizmetits over the back fence
06:42.35JTKizmet: you have 2 connections at home?
06:42.41KizmetJT, Yes.
06:42.51JTnice
06:42.55KizmetI also have a PRI/10 for fun
06:42.59JTcool
06:43.04*** join/#asterisk Kumbang (n=kumbang@router-wmi.paume.itb.ac.id)
06:43.04JTdoes it actually bind properly?
06:43.12JTlike it's supported by internode?
06:43.12voltagexI'll be sleeping outside of your house, stealing your wifi, Kizmet
06:43.19Kizmetyes, Its bound at the internode side of things too
06:43.30JTdid you need to pay them big bucks to do that?
06:43.39KizmetIts connected to my companys 'Internode Business Connect' network
06:43.55JTso not to the Internet?
06:44.00Kizmetyep approx $800 p/m for both the tails
06:44.04JTso i have to ask, did *that* cost a lot?
06:44.05zeeeshwant to configure mailbox .. for my .. peers .. if i have ... 2 .. 3 or 10 peers .. i want .. everybody .. shud press the .. same digit for listening voicemail.. could not found how to do it ... ??
06:44.06JTright
06:44.21JT$800 just on your house end?
06:44.29KizmetIt routes back through my companys Rack in the Datacenter
06:44.31Kizmetyep
06:44.46voltagexyou're rich too.
06:44.48JTi assume the PRI does something useful
06:44.53JTor you just like spending money ;)
06:45.12KizmetNot really. Just serves as a backup PRI for the company
06:45.25JTfair enough
06:45.32JTthrough hellstra?
06:45.33voltagexah, he may not have a house, there might be all kinds of cabling coming into a cardboard box in the ghetto.
06:45.39Kizmethehe, TransACT
06:45.43JTah
06:45.46Mahmoudwow found sipura 3000
06:45.51Mahmoudthey'll tell me the price later
06:45.57Kizmetlol
06:46.01JTspa-3102 is current model
06:48.04MahmoudJT, didn't find it in sipura.com :/
06:48.07Mahmoudnor linksys.com
06:48.20MahmoudJT, any advantages over 3000?
06:48.23JTyeah anyway, it exists
06:48.30JTwell the 3000 is end of line
06:49.43Kizmetbbl
06:50.32flip123welp, looks like I can sleep ok tonight
06:50.52flip123g'night all
06:51.27Mahmoudi got sipura at 300AED
06:51.38Mahmoudmeans 81USD
06:51.50Mahmoudhorribly cheap?
06:52.04Mahmoudpeople here look at it as a normal adapter :P
06:53.27Mahmoudsipura 3000
06:53.34Mahmoudthey didn't have sipura 3102 :/
06:57.45voltagex<PROTECTED>
06:57.45voltagex<PROTECTED>
06:57.53voltagexStrom_C: help now!
06:58.49Strom_Cum
06:58.55Strom_Cyou must ask me specific questions
06:59.05Mahmoudman Strom_C
06:59.12voltagexok, which config do I edit now
06:59.20MahmoudStrom_C is the man page
06:59.24voltagexyep
06:59.28Strom_Cvoltagex: did you "make samples"?
06:59.36voltagexhe was the one who made me remove trixbox
06:59.46voltagexso now he gets to help me!
06:59.50voltagexStrom_C: yes
06:59.53Strom_Cok
07:00.07Strom_Cif you're setting up your iax provider, give iax.conf and extensions.conf a go
07:00.30ParaNoirhey, anybody experienced with SwyxWare?
07:00.48Mahmoudi hate "make samples"
07:00.50voltagexStrom_C: well apparently IAX isn't working properly with my provider at the moment, but I'm not tethered to any one provider yet
07:00.55voltagex:/
07:01.06Strom_Chow about this
07:01.08Strom_Cset up a softphone
07:01.24Strom_Cthen I won't have to hear you complaining about how it costs 25c a call to use your box
07:01.26voltagexdone
07:01.30voltagexlol
07:01.42voltagexset up a softphone with my current provider
07:01.47voltagex?
07:01.58*** join/#asterisk FlatFoot (n=simon@80.88.192.83)
07:01.59Strom_Cno
07:02.00Strom_Cwith your box
07:02.06voltagexdone
07:02.10voltagexalready
07:02.44Strom_Cthat was quick
07:03.15*** join/#asterisk nuonguy (n=john@c-24-6-175-26.hsd1.ca.comcast.net)
07:03.22voltagexsame IP from the trixbox install
07:03.28voltagexso the softphone is set up
07:03.37Strom_Cand what about on the asterisk box side of things?
07:03.44Strom_Cdid you configure sip.conf?
07:04.07voltagexerr, just looking at iax.conf now because I will probably go with another provider on what I'm hearing about Faktortel - not good
07:05.08voltagexwoot, I have a free $10 mynetfone credit here
07:06.05voltagexeven better, they provide the asterisk configuration for me :P
07:06.12voltagexI win, Strom_C
07:06.32Strom_Cgod, you're irritating
07:06.34voltagexalthough, are the configurations listed up the top ok? http://www.mynetfone.com.au/support/
07:07.01Strom_Cyeah, but you need to pick the relevant bits out of the config
07:07.28Strom_Cand their extensions.conf example uses way deprecated syntax
07:07.52voltagexI'm not intentionally trying to be irritating
07:08.03voltagexoh, and extensions.conf doesn't look that scary.
07:09.34voltagexwhoa, mynetfone have lower prices than the others I've looked at
07:09.37ParaNoirhey, is it possible to use a bri card with asterisk 1.4?
07:10.59Strom_CParaNoir: yes
07:11.02Strom_Cb410p
07:12.24*** join/#asterisk tzafrir_laptop (n=tzafrir@62.90.10.53)
07:12.51ftexcomParaNoir never use a bri card. It's hell on earth
07:12.51rudholmStrom_C: what about with National ISDN BRI?  :-p
07:13.21Strom_Crudholm: please hold
07:13.28Strom_Cyour estimated wait time is
07:13.30Strom_Csix
07:13.31ParaNoirwhy?
07:13.31Strom_Cmonths
07:13.34ParaNoirworks like a charm ;)
07:13.55ftexcomParaNoir I always have issues configuring it...but maybe it's because I suck
07:14.01ParaNoir;)
07:14.16*** join/#asterisk pif (n=ldm@zenon.apartia.fr)
07:14.22ParaNoircould get quadbri and single bri working, not bad for a newbie right? ;)
07:14.36*** join/#asterisk plasmid (n=noway@c-24-127-166-193.hsd1.pa.comcast.net)
07:14.40rudholmStrom_C: six months?!?  damn, I thought I was done with that kind of service when I cancelled Charter.
07:14.41ftexcomI also had a b410p working...once
07:14.55rudholmftexcom: with what kind of BRI?
07:15.04*** join/#asterisk nasls_lsa (n=chatzill@athedsl-145799.otenet.gr)
07:15.06rudholmftexcom: Euro ISDN?
07:15.17ftexcomrudholm yeap
07:22.26*** join/#asterisk yonahw-work (n=yonahw-w@genie03-173-74.inter.net.il)
07:29.02voltagexok Strom, now that I have proper asterisk, would you be able to help me get FWD working properly?
07:29.33*** join/#asterisk putzz (n=me@CPE00155824933c-CM00111ae07ac2.cpe.net.cable.rogers.com)
07:30.10Strom_Cmaybe
07:30.17Strom_Cit's been literally forever since I screwed with FWD
07:30.51JTvoltagex: fwd's online help, and the book should be enough to get you up and running with FWD
07:31.12voltagexJT:  I also found a nice tutorial
07:31.23voltagexwill Asterisk tell me if I'm using deprecated syntax?
07:37.16*** join/#asterisk Pilosopas (n=103392DD@clt-84-32-39-71.ktv.lt)
07:38.01sevardvoltagex: asterisk will tell you if you're using a depressing syntax.
07:38.11voltagex:/
07:38.27*** join/#asterisk tsurko (n=tsurko@77.70.24.142)
07:38.55Mahmoud81 USD for sipura 3000, what do you say? heap? normal? expensive?
07:39.09JTok
07:39.32JTnot brilliant, but it's not that awful i guess
07:39.42MahmoudJT, talking to me?
07:39.46JTyes
07:39.58Mahmoudso, 81 USD is normal price?
07:40.03flendersI bought mine for 120AUD
07:40.19JTthey probably cost less than that in the US, considering they're end of line
07:40.29Mahmoudflenders, i see
07:40.41flendersI thought they would be like 50 bucks over there.
07:40.57MahmoudFlatFoot, aus?
07:41.10JTMahmoud: who cares, you've bought it already
07:41.16Mahmoudi didn't
07:41.18JTwhy the need to validate your purchase so badly?
07:41.22Mahmoudi just found a shop selling it at 81USD
07:41.26JTi see
07:41.42voltagexflenders: whoa, best price I can see is $120AUD
07:41.44voltagexoops
07:41.45voltagex130
07:41.54JTeither buy it or don't, i'm about sick of seeing "81USD" now :P
07:42.11Mahmoudi'm sick of seeing your nick
07:42.14flendershahaha
07:42.27Pilosopascan anyone help me with zaptel problem?
07:42.31Mahmoud81USD o/~
07:42.36JTMahmoud: your inane questions are what make people decide to take a break from #asterisk
07:42.46JTever notice everyone shuts up when you say stuff now?
07:42.58Mahmoudwhat question?
07:43.12Mahmoudbring one inane questoin that i asked here dude
07:43.13JTyour continuous repetative questions, beating dead horses :)
07:43.23Mahmoudbring _one_ example
07:43.30voltagexSipura SPA-3000 FXO/FXS VoIP Gateway SPA-3000 AU $1.00, $6.00 postage on eBay
07:43.39JTIS 81USD GOOD PRICE
07:43.42Mahmoudlol 6 USD
07:43.48JTI THINK SIP CAN WORK THROUGH HTTP PROXY
07:43.54voltagexMahmoud: is 81US okis 81US okis 81US okis 81US okis 81US okis 81US okis 81US okis 81US okis 81US okis 81US okis 81US okis 81US okis 81US okis 81US okis 81US okis 81US okis 81US okis 81US okis 81US okis 81US okis 81US okis 81US okis 81US okis 81US okis 81US okis 81US okis 81US okis 81US okis 81US okis 81US okis 81US okis 81US okis 81US okis 81US okis 81US okis 81US okis 81US okis 81US ok
07:44.00MahmoudJT, yeah dude, it can work
07:44.17JTMahmoud: yes, ok, prove it, whatever, it's been discussed plenty already
07:44.19Mahmoudit needs some heavy modification to the client + server
07:44.20JT:)
07:44.20voltagexno, really, is 81US ok?
07:44.31JTvoltagex: depends where you are
07:44.34MahmoudJT, study how TCP works, and how to feed it with RTP data
07:44.38voltagexreally really really really really
07:44.44JTvoltagex: i'd buy the SPA-3102 if possible :)
07:44.58voltagex*Mahmoud bounces up and down like a fucking jack russel*
07:45.01flendersbut not for 81USD
07:45.23Strom_Chey guys, you'll never believe this, but
07:45.25Strom_C81USD
07:45.37voltagex/kick Strom_C
07:45.49pifis 81 USD too cheap ?
07:45.50voltagex/obliterate Mahmoud
07:46.01Pilosopascan anyone help me with zaptel problem? I get phantom rings every 2-5 minutes on my CO line (crapy line noise) :( Can zaptel deiver after registering incoming call doublecheck if it is ringing after a period of time and only then pass it to asterisk?
07:46.02pifplease please please tell me
07:46.08voltagex/youtoo pif
07:46.24voltagexthere needs to be a /obliterate command
07:46.35Mahmoudidiot
07:46.47voltagexand proud of it
07:46.53Mahmoudthat's the issue
07:46.54pifterrorist
07:46.56voltagexI can play the troll game too.
07:46.58Mahmoudyou are also a praud tax payer
07:47.07voltagexwhoa, nasty Pif
07:47.28Pilosopas:/
07:47.53voltagexPilosopas: I'm sure someone can answer after this flamewar is done.
07:48.07Pilosopasthe world's idiocy has no boundries :(
07:48.17PilosopasI know
07:48.39Pilosopasgot my head full with this problem for 3 days
07:49.09Mahmoudchannel operators need to be more strik against off-topic discussions, just like ##c =]
07:49.39voltagexMahmoud: 80US is offtopic
07:49.54Strom_Cvoltagex: shut up already plzkthx
07:50.01Mahmoudnot really, its aobut voip equipments anyway
07:50.10*** part/#asterisk voltagex (n=voltagex@124-254-123-181-dsl.ispone.net.au)
07:50.57dc3aesanyone having a problem with nufone, 2 of us here cant dial out
07:51.51Pilosopasvoltagex: who can answer my question?
07:52.20PilosopasI believe I'm newer here then you....
07:52.46Strom_CPilosopas: hmm
07:53.05Strom_Cyou have bad enough line noise that it causes the phones to ring?
07:53.11Pilosopasyes
07:53.30PilosopasI get random line spikes every 2-5 minutes
07:53.36Strom_C...
07:53.38Strom_Couch
07:54.02Pilosopasshielding is bad enough somewhere...
07:54.29Strom_Cdo you currently have usecallerid=yes?
07:54.36Pilosopasyes
07:54.39Strom_Chm
07:55.01tengulrehow to building the asterisk cluster?
07:55.45Pilosopas:/ what does RING_DEBOUNCE_TIME do?
07:56.39*** join/#asterisk s-ndh-c (n=michi@85.93.11.18)
07:57.47*** join/#asterisk awk (i=nobody@dsl-242-92-152.telkomadsl.co.za)
07:57.51awk~pb
07:58.02jbotrumour has it, pb is a place to paste your stuff without flooding the channel - try  http://pastebin.ca, or http://channels.debian.net/paste
07:58.02awkhmm, what can I use for pastebin
07:59.03awkhttp://channels.debian.net/paste/5820
07:59.19awkplease can somebody look at that paste, and on a default install of version 1.4 I get all those warning messages
07:59.49Pilosopasso.... No ideas? anyone?
07:59.59Strom_Cawk: don't worry about it
08:00.07awkStrom_C: I dont want to see those warnings
08:00.18Strom_CPilosopas: try again in eight to twelve hours
08:00.23Strom_Cawk: boo hoo
08:00.26awkI want a clean startup
08:00.37Strom_Cis this an upgrade from a previous install?
08:00.40Pilosopasyup, just like windows
08:00.43awkStrom_C: what are they related to
08:00.44Strom_Cor is this on a totally fresh machine?
08:00.45Pilosopasno
08:00.47Pilosopasnot upgrade
08:00.53Pilosopasclean gentoo install
08:00.53Strom_Ci was asking awk
08:01.05Pilosopasups
08:01.08Strom_Cso let me restate
08:01.12Strom_Cawk: is this a clean install?
08:01.15awkwell it was an upgrade, but I removed the modules directory and /etc/asterisk/*
08:01.24awkso its kind of clean
08:01.43Strom_C/usr/lib/asterisk/modules?
08:01.50awkyes
08:03.03Strom_Cwell, it's not going to cause you any trouble
08:03.10Strom_Cso live with it, or don't watch asterisk start
08:03.53awkso you not sure how to get rid of it
08:04.20Strom_Cno, I'm not, but it's not a warning which is going to actually affect anything
08:04.32awki still dont like seeing it
08:04.42awkand its a warning because things are not working 100^%
08:04.54awknevr mind, ill fix it, just thought somebody had come across it
08:05.37Strom_Cif you do find the cause, please let us know what it is
08:06.59awksure
08:07.17s-ndh-chey guys
08:07.45awkbusy going through /usr/src/asterisk-1.4.2/main/translate.c
08:09.55s-ndh-ci have a my asterisk working with one softphone and a voip provider and would now like to connect asterisk to out existing pbx, i would like to be able to make outgoing calls from the softphone on asterisk via out existing pbx`s t1 interface and have the phones connected to the existing pbx make voip calls via asterisk. what are my options to connect both?
08:10.18*** join/#asterisk rocket007 (n=saim@ner-as14402.alshamil.net.ae)
08:10.41Strom_Cs-ndh-c: do you have an extra T1 port on the PBX?
08:11.04s-ndh-cStrom_C: no idea, will have to look up that
08:11.57Strom_Ci'm not completely sure I understand your situation though
08:12.03*** part/#asterisk rocket007 (n=saim@ner-as14402.alshamil.net.ae)
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08:12.12Strom_Cyou want asterisk and the existing pbx to both be able to place outbound calls over the T1?
08:12.40s-ndh-cyeah and have both make outbound calls via voip if thats possible
08:12.48Strom_Csure
08:12.50Strom_Ceasy enough
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08:13.07Strom_Cget a dual-span T1 card, and put asterisk on the T1 between the telco and the existing pbx
08:13.57s-ndh-chehe, but if the asterisk box dies the existing pbx is dead too right?
08:14.29Strom_Cyes
08:14.40Strom_Cbut usually that's not a problem if you get a decent server-grade machine
08:15.05Strom_Cyou'd be an idiot to run this on some frankenbox you cobbled together out of parts from the basement
08:16.30jubei_guys what's the best kernel to build for a 1.4.2 system?
08:16.30s-ndh-cStrom_C: i see, its just some p4 desktop machine atm i cant do something like that i guess
08:16.34s-ndh-c:)
08:17.00brianHAY GUISE
08:17.16Strom_Cs-ndh-c: well, i run my own personal box on a piece of shit machine i got for free, but i wouldnt run a business on it :)
08:17.22s-ndh-cso easiest way is have the asterisk conencted to the telco t1 line and connect the exeting pbx to the second port on the dualspan card
08:17.33Strom_Cyeah
08:17.46s-ndh-cok will talk to my boss about that later
08:22.04kupsihi
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08:43.04voltagexI need some help with something that doesn't seem to be in TFM - Auto fallthrough, channel 'SIP/phone1-081ddeb8' status is 'CONGESTION'
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09:03.55kupsihello guys, i'm currently choosing between Sangoma's a-104 and a-104d... the a-104d has echo cancellation but it is $1000 more expensive... is it really worth to have this echo cancellation? afaik, most IP phones comes with "echo cancellation"... please help... tia
09:05.18JTif you want to be sure, then yes, it is worth it
09:05.30JTphones don't echo cancel far end echo as a rulee
09:05.33JT-e
09:06.31voltagexJT: having problems here without my shiny GUI ;)
09:07.12kupsiJT: should i go for it?
09:09.16JTkupsi: yes, if you can afford a quad port card, you should go EC
09:10.30kupsithanks... i'm now doing the project costing
09:15.12voltagexJT: getting circuit-busy, what should I check
09:15.25voltagexNOTICE[22240]: chan_sip.c:11831 handle_response_invite: Failed to authenticate on INVITE to '"Adam" <sip:0@fwd.pulver.com>;tag=as6219ab19'
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09:29.58demlakhmm.. kannt find login and password for the trixbox 2.0 vmware image.. anyone got them? =)
09:30.16Turt|eHi, I have an linksys spa941 behind nat, and i cant get i to connect to asterisk, in asterisk i have out in the nat=yes for the client, the asterisk server is on a global ip and not behind a nat, but there is an pix in between. My phone is setup with nat enabled and my router is setup with port forwared to my phone: 5060-5065, 10000-20000 and 5461. What could posible be wrong here ?
09:35.28voltagexI'm getting DNS errors, this host resolves on the asterisk box though
09:35.36voltagexe.g. [Mar 26 19:35:23] WARNING[22400]: chan_sip.c:7242 transmit_register: Probably a DNS error for registration to 762697@fwd.pulver.com, trying REGISTER again (after 20 seconds)
09:40.42*** join/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl)
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09:50.17*** join/#asterisk frenzy (n=frenzy@unaffiliated/frenzy)
09:50.38frenzyis there a way to fake DNID on incoming IAX calls?
09:51.15awkofcourse, you can ever spoof you outgoing calls to say its coming from any 10 digit number u want
09:51.20awkever/even
09:51.41frenzyI know I can set caller id
09:51.58frenzybut i'm looking at setting dnid for inboud
09:52.17frenzywhat is the command/code for it?
09:52.51awkwhy would you want to spoof the dnid ?
09:53.15frenzyfor filtering
09:53.34awkso you want to filter certain did's to not route?
09:53.38awkand route certain did's ?
09:53.42frenzyi am able to do it via SIP
09:53.48frenzyactually more like billing filtering
09:53.58voltagex[Mar 26 19:53:18] NOTICE[22501]: chan_sip.c:11831 handle_response_invite: Failed to authenticate on INVITE to '"Adam" <sip:0@fwd.pulver.com>;tag=as2152124a' << why is this? I'm getting registered ok.
09:54.08frenzybetween toll-frees and local dids
09:54.50*** join/#asterisk tr2x (n=alvar@80-218-185-55.dclient.hispeed.ch)
09:55.59awkfrenzy: sorry things have just gone pete tong here
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09:56.05awk*gone*
09:56.40giasai68hello i get this error can you help me to understand howi can fix it?
09:56.55giasai68<PROTECTED>
09:56.55giasai68[Mar 26 11:54:18] WARNING[2911]: app_macro.c:174 _macro_exec: Context 'macro-stdexten' for macro 'stdexten' lacks 's' extension, priority 1
09:56.55giasai68<PROTECTED>
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10:01.27*** join/#asterisk nothinman (n=martin@ast.advance-internet.com)
10:01.36nothinmanHello (-:
10:04.58nothinmanguys, how can I get just the first character from a variable? (let's say ${ABC})
10:05.13nothinmanso when it's "blah" I will get "b"
10:06.57Strom_Cnothinman: ${ABC:0:1}
10:08.11*** join/#asterisk Ahrimanes (n=ma@81.7.159.2)
10:08.12*** join/#asterisk wundaboy (n=hixscrip@c-67-171-137-48.hsd1.or.comcast.net)
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10:11.43awkStrom_C: 1.4 is it ready for production ?
10:11.59Strom_Csorta
10:12.10Strom_Ci'd still use 1.2 if you aren't feeling adventurous
10:12.28awkany idea when comfort noise will be included in 1.4 ?
10:12.37*** join/#asterisk canapa (n=canapa@83-64-148-98.wolfsberg.xdsl-line.inode.at)
10:12.50awkerrr silence suppression
10:14.49giasai68hello i get this error can you help me to understand howi can fix it?
10:15.03giasai68Executing [832113137@default:1] Macro("SIP/832113137-08208838", "stdexten|832113137|SIP/832113137&Zap/1") in new stack
10:15.03giasai68[Mar 26 12:12:12] WARNING[3402]: app_macro.c:174 _macro_exec: Context 'macro-stdexten' for macro 'stdexten' lacks 's' extension, priority 1
10:15.03giasai68<PROTECTED>
10:16.57tzafrirgiasai68, Context 'macro-stdexten' for macro 'stdexten' lacks 's' extension, priority 1
10:17.12Strom_Cno, reading the error message would be too easy
10:17.13tzafrirshow dialplan macro-stdexten
10:17.49*** join/#asterisk zeeesh (i=zeeesh@202.38.55.125)
10:17.50zeeeshhi
10:17.56tzafrir(and please don't paste it here in the channel)
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10:21.22giasai68ok thank you
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10:28.19awksomebody should stick pasting in the topic
10:28.19awkor have a services join message to state to use a pastebin
10:28.31Strom_Cawk: that wouldnt help
10:28.56Strom_Ctrixbox is in the topic, yet every day we get people in here that say "HI OMG I ARE TEH NEETING HALP WITH TRIXBOX"
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10:31.06*** part/#asterisk frenzy (n=frenzy@unaffiliated/frenzy)
10:31.07sbingneromg nobody said anything that wasn't in the error msg.... lol
10:34.16awkStrom_C: heh
10:34.57Ch0HagIs anyone here who is familiar with the Asterisk source code?
10:35.08Ch0HagEspecially chan_sip.c
10:38.06awkCh0Hag: hmf, i've actually only started really looking at it today
10:38.25awkbusy playing with translate.c
10:38.49Ch0HagOh.
10:39.17*** join/#asterisk nighty^ (n=nighty@211.154.202.98)
10:39.27Ch0HagWell I want some way to put a string in a variable in the dialplan and then have that string used, if set, in inviteprep()
10:39.33Ch0HagI guess I will keep poking around.
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10:54.27VecHow do I view all zap channels, if I do a zap show channels, it seems to only show me incoming calls on zap ?
10:54.52*** join/#asterisk DaveCanoe (n=Dave@m815f36d0.tmodns.net)
10:55.20Ch0Hag'zap show channel <n>'?
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10:57.19awkhmm
10:57.21awk5004:5082
10:57.28awkcould SIP use this entire range?
10:58.01Ch0HagSurely it can use any port you like?
10:58.27awki'm talking about default
10:58.30Ch0HagI can immediately see peers with ports as low as 1024 or as high as 5060
10:58.32awki see some f/w rules on voip-info
10:58.48awkhmf!
10:58.56awkthat doesn't make sense
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11:03.31Ch0HagWhat doesn't?
11:03.47awkwell, it cant just decide to use what it wants
11:03.51Ch0HagPhone registers with *, says it's listening on port X.
11:03.51awkyou open a set of rules
11:03.58awkmost poeple just have 5060 open
11:04.12awkso how could it use port 1024 - 5060
11:04.22Ch0HagThere's an option to make * listen to a port (or ports?) other than 5060.
11:04.37Ch0HagNot sure if it can do a range or not.
11:04.43Ch0HagUnlikely.
11:04.57awk12:58 < Ch0Hag> I can immediately see peers with ports as low as 1024 or as high as 5060
11:05.20Ch0HagThat's the phone.
11:05.26Ch0HagNot *.
11:05.49VecI just had my asterisk box get stuck, on     -- Executing [s@macro-dialBasicExten:3] Dial("Zap/0-2", "SIP/2128|20") in new stack
11:05.49Vec<PROTECTED>
11:05.55Vecoops
11:06.25VecAfter a number was dialed it would say Called 2128, but then it did not ring and then after a few seconds it would go engaged
11:06.26awkCh0Hag: so for your phones to work what range did you need open
11:06.39Ch0HagThe phones only conect to port 5060.
11:07.10awkthen?
11:07.23Ch0HagThen they tell * what port they are listening on.
11:07.43Ch0HagWhether that's explicit or part of IP I don't know.
11:07.45Ch0HagProbably the latter.
11:07.46awkok, but what happens if everything is closed, except your rtp range
11:07.57awkand sip for registration
11:08.07awkhow would it comunicate on port 1024 as you said earlier
11:08.17Ch0HagIf the firewall was blocking it, it wouldn't.
11:08.46awkso thats what im getting at
11:08.58awkwhat solution would you do, to not allow such a large range open
11:09.14Ch0HagTell the phone somehow what port range it can listen on.
11:09.26Assidawk: set your rtp range on your own
11:09.29awkeg: 5060 only
11:09.36awkAssid: hey ?
11:09.41awki have my rtp range set
11:09.48Assidon the server?
11:09.53awkyes
11:09.59awk10000 10100
11:10.02Assidokay so.. set the same on yuor firewall
11:10.13awki dont have f/w issues
11:10.19awkim trying to understand what Ch0Hag was saying
11:10.28Assidoh ok
11:10.34Assidsorry.. me goes away again
11:11.33Ch0HagI don't know whether the phones are listening on (eg.) 1024 for SIP *and* RTP.
11:11.56Ch0HagI have no need to restrict outgoing traffic through the firewall at all so it has yet to be a problem.
11:11.58awkoh, I see
11:12.31Ch0HagI think that 10000-10100 setting is for * listening for RTP.
11:12.39Ch0HagNot for a phone listening.
11:13.04Ch0HagOn this Snom that is set with 'Dynamic RTP port start/stop:' to 49152-65534
11:13.07Ch0HagFor whatever reason.
11:13.23awkoh, intresting
11:13.40Ch0HagAh 49152 == 0xC000
11:13.51*** join/#asterisk zotz (n=zotz@24.244.163.157)
11:15.19awkmaybe 1 of you can help. snmp
11:15.48awkdoes mrtg use udp 161 or tcp 199
11:15.56awkto query your snmp daemon ?
11:16.01Ch0HagDont know snmp.
11:22.21nothinmanStrom_C: thank you. it does seem to work :-)
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11:32.45*** join/#asterisk AdamCrowe (n=newbie@host-87-74-57-8.bulldogdsl.com)
11:33.24AdamCroweHello
11:33.43AdamCroweDoes anybody have any experience with ISDN cards?
11:34.38tzafrirsome people have some experince some of the time
11:35.40AdamCroweI have a B410P which I managed to install and I can do incoming and outgoing calls with...
11:35.44*** join/#asterisk stimpie (n=michiel@ip565faf27.direct-adsl.nl)
11:35.48mkl1525Hi, does anybody know how to write sms/text on snom displays? tried sipsak but get a "destination unreachable" although I don't have problems to access the web interface.
11:36.29AdamCroweBUT - the L2Link keeps going down, and unless an incoming calls "waked it up" or I manually type "misdn port up 1", I cannot make any outgoing calls on that port.
11:36.36Ch0Hagmkl1525: http://www.snom.com/wiki/index.php/Xmlobjects
11:37.53mkl1525Ch0Hag, thanks for the hint, but we've got snom300 too that don't have the xml browser so I think I'll need some simple text message
11:37.56*** join/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl)
11:38.24Ch0HagI don't know about that.
11:38.34Ch0HagProbably just replace it with a newer one :)
11:39.01Ch0HagWhat are ANI, DNID and RDNIS?
11:39.40zeeeshi registered 2 peers 100 and 200 .. at my asterisk server .. 100 can make call at 200 but 200 unable to make call 100 ... extensions are (from 100   exten => 200,1,Dial(SIP/200)   ) (from 200   exten => 100,1,Dial(SIP/100)   ERROR IS .. "5 NOTICE[3633]: app_dial.c:1056 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination)
11:39.41zeeesh<PROTECTED>
11:39.41zeeesh<PROTECTED>
11:40.20Ch0Hag100 may need to reregister.
11:40.37zeeesh100 is registered ..
11:40.59*** join/#asterisk heliosj (n=jeff@pdpc/supporter/active/xheliox)
11:43.14AdamCroweAnybody???
11:44.37Ch0HagNot me. I know next to nothing about ISDN.
11:46.50zeeeshNEHLEEEOOOOOOO ...
11:47.55Ch0HagI think nobody is really here.
11:48.20Ch0HagExcept me, and I'm by no means an Asterisk guru.
12:41.30*** join/#asterisk jbot (i=ibot@pdpc/supporter/active/TimRiker/bot/apt)
12:41.30*** topic/#asterisk is Asterisk: The Open Source PBX -=- Asterisk 1.4.2 (Mar. 19, 2007), Asterisk 1.2.17 (Mar. 19, 2007), Zaptel 1.2.16 (Mar. 19, 2007), Zaptel 1.4.1 (Mar. 23, 2007) -=- Other fun channels: #asterisk-gui, #asterisknow, #asterisk-commits -=- Join #freepbx for freepbx/trixbox support.
12:43.10*** join/#asterisk myiagy (n=myiagy@200.175.61.250.static.gvt.net.br)
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12:44.58m-00kiehello.
12:45.22m-00kiedoes asterisk's oh323 support conferencing?
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12:56.37puzzledm-00kie: why not? just send the call coming in via h323 to a meetme
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13:02.06zaffaAnyone had problems with zaptel 1.2.14? I Have strange problem with te4xxp - once or twice a month PRI line 'locks' that no calls can be made nor received.
13:02.37zaffa'No D-channels available! Using...' and then 'Unable to create channel of type 'ZAP' (cause 34 - Circuit/channel congestion)'. zap show status shows no alarms.
13:02.44necromcrhi, i'm having some "goto" problems (i cant seem to make it work to dial 0 and then match following numbers in other context using misdn line). can please anyone take a moment?
13:02.57zaffareloading zaptel modules and asterisk solves the problem. Any ideas what can be the reason?
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13:11.59unicehi all
13:12.04*** join/#asterisk angryuser (n=Miranda@i03v-213-44-169-43.d4.club-internet.fr)
13:12.34angryuseri have some problems with my isdn ports, any idea what is it? http://www.pastebin.ca/403464 asterisk 1.41
13:12.47angryuseri am using B410P misdn
13:13.48angryuserthey are configured in TE mode now, but still same problems
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13:27.38mkl1525Hi, it's possible to change the idle screen of snoms but is there also a way to change the screen when I'm talking to somebody?
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13:33.40f_akmalhi all, i have problems compiling app_voicechangedial.c, can anyone help?
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13:36.58Kattymew.
13:37.54Ch0HagWhere is struct varshead defined?
13:38.21Ch0HagI can't find it anywhere, except in use.
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13:48.01Ch0HagWell that doesn't seem to be what I want anyway, so if I Set() a variable in the dialplan, how can I access it in the source?
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13:51.47sashionCh0Hag, I believe it might be under pbx.c
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14:00.21achucan we redirect asterisk's /var/log/full to stdout ?
14:00.38Gido-Ewhy not?
14:01.08achucan you pls tell me which file I have edit ?
14:04.03necromcrachu: logger.conf?
14:05.32achuits onlys shows full => notice,warning,error,debug,verbose , can I remove all that option and add /dev/stdout ?
14:05.49achusorry if I am wrong
14:05.50necromcrstdout?
14:05.56necromcrwhy would you want to do that?
14:05.57e-ddieachu: can you please read the manual?
14:06.07necromcrlog to a file and do tail -F <that file>
14:07.37e-ddiesounds like a better idea would be for him to reinstallwindows
14:08.50achunecromcr : with my small knowledge I understand that the full message specified in logger.conf goes to /var/log/asterisk/full , I would like to use it for daemontools for multilog
14:09.06achuwhen I configured it is not rotating logs
14:09.32achuso I want to redirect the asterisk full log to /dev/stdout
14:10.50necromcrachu: i'm not quite familliar with multilog but.. cant you use metalog? it's got filtering options as well
14:11.06achuk
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14:11.33necromcrawsome.. i just made infinite loop in my context :P
14:12.36anonymouz666infinite loop?
14:12.39anonymouz666welcome to the club
14:12.41anonymouz666lol
14:12.54Dr-Linuxcisco 7920 is a WIFI phone?
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14:13.40tdonahue-laptopanyone have a link to the changelog for 1.4.2
14:13.44tdonahue-laptop?
14:14.57achunecromcr : When I enabled syslog.local0 in logger.conf it starts showing the messages in /var/log/messages
14:14.58heisoni'd like to get Asterisk 1.4.2 via svn, and i'm unsure if this will actually get me 1.4.2... svn checkout http://svn.digium.com/svn/asterisk/tags/1.4.2 asterisk    how is that different from http://svn.digium.com/svn/asterisk/trunk ? if i'm getting the latest release, should they be the same?
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14:18.19necromcrachu: you need to log it into some other file?
14:18.27russellbheison: that is correct for getting 1.4.2 from svn
14:18.35*** part/#asterisk sergee (i=kvirc@195.94.224.197)
14:18.38russellbheart: trunk is the development tree, where new things are getting added and broken
14:18.53*** join/#asterisk AdamCrowe (n=newbie@80-47-100-100.lond-hex.dynamic.dial.as9105.com)
14:19.08AdamCroweAnybody here with ISDN experience?
14:19.44heisonrussellb: thx!
14:19.58coppiceI developed the world's first ISDN mux. does that count? :-)
14:20.04heisondoes anyone know if rx_fax and tx_fax are working with 1.4.2?
14:20.33sashionwhat would cause asterisk to send a CANCEL to a phone after 1 ring?
14:20.54necromcrdamn.. one would really need a documentation in a style of chm or javadoc :-|
14:21.18AdamCroweI'm from the UK... Layer 2 keeps going down (as it suppose to do in most Europe), but while it's down, I cannot make outgoing calls unless I manually take it up via "misdn port up 1" or through an incoming call.
14:21.37russellbcoppice: oh yeah?  well I had the de-mux before you had the mux.
14:21.50coppiceprobably not
14:21.58achunecromcr : I don't want asterisk to right it's logs to file
14:22.34AdamCroweHow can I keep layer 2 up or at least make it go up when making an outgoing call?
14:22.47Ch0HagDoes anyone know the * source?
14:22.55necromcrachu: eee..
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14:23.05coppicerussellb: i suspect you might have been a bit young at that time
14:23.10Ch0HagBecause either I don't get something or a huge chunk is broken and I'd like to work out which it is.
14:23.17russellbcoppice: probably :)
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14:25.11AdamCrowecoppice: any ideas?
14:25.11coppiceI think that must have been about 1986
14:25.12coppiceAdamCrowe: I said I *developed* the first mux. I didn't say I ever used one :-\
14:25.12russellbha ..
14:25.12russellbcoppice: yeah, i was only 2 years old.
14:25.12necromcrachu: sorry, too much for me
14:25.13AdamCroweO_o
14:25.46achunecromcr : ok , I am sorry
14:26.51necromcrwhat's the idle policy on this channel?
14:27.00russellbidling is encouraged
14:27.08russellbis that a policy?
14:27.27adam_vollrathThere probably isn't a policy, per se.
14:27.47fileanyone who fails to idle may be randomly kicked ...
14:27.50file(not really)
14:28.12russellbwell, talking does encourage your chances of getting kicked
14:28.47russellbnooo
14:28.49adam_vollrathAnyone who idles will not be kicked.  I think that's certain.
14:28.59russellbunless your nick alone is trolling
14:29.10russellblike ... /nick asterisk_is_so_laME
14:29.11adam_vollrathI think the bar is pretty high for that.
14:29.35russellbpwned by the length limit
14:30.16*** kick/#asterisk [asterisk_is_lame!i=russell@asterisk/developer-and-stable-maintainer/drumkilla] by russellb (russellb)
14:30.38russellbi have teh power!
14:31.00necromcrok.. idle it is
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14:31.55*** kick/#asterisk [russellb!n=file@asterisk/developer-and-muffin-lover/file] by file (file)
14:31.57file!!!
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14:32.05*** mode/#asterisk [+o russellb] by ChanServ
14:32.16russellber, not Fieldy !
14:32.17russellbfile.
14:32.25russellbstupid tab completion :(
14:33.42Ch0Hagvoip-info says 'the variable ${VXML_URL} can be used to add additional items to the To: header'. This is done bi setting p->options->vxml_url in a AST_LIST_TRAVERSE loop, however that loop is never entered because there are no variables to loop over.
14:36.33*** join/#asterisk infernix (i=nix@unaffiliated/infernix)
14:39.51putzz9000000000000=[-p999990-\=-[][
14:39.52putzz=;=']
14:39.57putzz\--0p-;['
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14:59.18Bladerunner05using gui in the installation procedure (browser based) in the Local Extension Settings the next button is disabled, could be a permission problem in /etc/asterisk directory ?
14:59.53MercestesBladerunner05, Could definately be a #freepbx or #asterisknow question.
15:00.11Mercesteswe don't do GUI here.  We're text only.
15:01.29*** join/#asterisk qdk (n=qdk@0x50c627be.bynxx11.adsl-dhcp.tele.dk)
15:02.17necromcryup, text only (CTCP VERSION reply from necromcr: irssi v0.8.10 - running on Linux i686)
15:04.59Mercestesnecromcr, are you a bot?
15:05.34MercestesWe're thinking necromcr is a bot.
15:07.44Mercestesnecromcr, what is your prime directive?
15:09.16adam_vollrathAbout that GUI thing.  We're using a 3Com NBX100 system now, and we like its web-based administration (even if it is ancient), because we change our dial plan frequently.  Is there a GUI to ease Asterisk dial plan maintenance?
15:09.30russellbadam_vollrath: check out asterisknow.org
15:09.40adam_vollrathressellb, kthx
15:10.14*** join/#asterisk `p4r14h`work (n=josh@24-119-48-78.cpe.cableone.net)
15:10.34Ch0HagIs SipAddHeader() supposed to do absolutely nothing?
15:10.38adam_vollrathCan I use the AsteriskGUI without using this Asterkisknow Linux distro?  We'd planned on using Fedora.
15:10.48Qwell[]Ch0Hag: clearly it isn't
15:10.52russellbadam_vollrath: yeah, you can
15:10.58Ch0HagWell it does.
15:10.58russellbadam_vollrath: it's super easy to install, too
15:11.01Ch0HagOr doesn't...
15:11.07russellbadam_vollrath: the instructions are in the topic of #asterisk-gui
15:11.18Qwell[]what header are you setting, and do you know that your provider/device supports said header?
15:12.08*** join/#asterisk aptura (n=jondoe@S010600a0c93f6f7e.vs.shawcable.net)
15:12.18russellbQwell[]: /me drools at the weather report ... high of 84
15:12.47russellbbah
15:12.51Qwell[]heh
15:12.53Ch0HagThat is neither here nor there.
15:12.53Ch0HagI have tried both SipAddHeader(Bro=ken) and SipAddHeader(Bro: ken), and the headers remain unaffected.
15:12.57Qwell[]How was SJC?
15:12.58apturasounds like LA
15:12.59Qwell[]weatherwise
15:13.02Ch0HagUnless for whatever random reason the headers 'sip debug' shows are not the headers that are actually sent.
15:13.16apturaOur area has been beaten with tons of snow and rian this year.
15:13.18russellbQwell[]: nice ... 60's 70's ...
15:13.33Qwell[]russellb: we should move Digium there :P
15:13.49Qwell[]AL summer is gonna suck - I know it
15:14.43apturaQwell, the city managers and planners did a poor job of building the city in the first place. Turned it into a huge oven with not enough trees or white roofs to absorb or reflect the heat.
15:14.55Qwell[]what city?
15:15.08apturayou are in Alabama right?
15:15.11Qwell[]yeah
15:15.14russellbAL summer is going to r0x0r
15:16.12apturaI read a report it was Birmingham
15:16.32Qwell[]good thing we aren't in Birmingham then ;)
15:16.42apturawhere are you then
15:16.46Qwell[]Huntsville
15:16.55apturaOkay wait it was Atlanata
15:17.06Dr-Linuxcisco 7920 is a WIFI phone?
15:17.11Dr-LinuxQwell[] :)
15:17.13Qwell[]Dr-Linux: skinny, yes
15:17.16*** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman)
15:17.17Qwell[]it isn't great
15:17.17apturaHere in the summer it is cool
15:17.43apturawe get the fresh pacific air but on the downside the tons of moisture it brings in the winter.
15:17.55Qwell[]Seattle? :P
15:18.04Dr-LinuxQwell[]: will 7920 will work for me, or still i'll have issues like other phone?
15:18.10Qwell[]Dr-Linux: I don't really recommend it
15:18.12Dr-LinuxQwell[]: and what protocol should i use?
15:18.13*** join/#asterisk Damin (n=damin@nucleus.nacs.net)
15:18.19Qwell[]You can only use skinny wth the 7920
15:18.36Qwell[]there are better wifi phones
15:18.44Dr-LinuxQwell[]: you mean the asterisk skinny will work for me?
15:18.58Dr-LinuxQwell[]: i'd agree but my boss has one, and he asked me to configure it
15:19.03apturaQwell, wait till the ice caps melt it regultes the tempature of the atlantic ocean. And with it the eastern seaboard.
15:19.10Dr-Linuxeven the phone is in USA and i'm here
15:19.27Qwell[]Dr-Linux: it'll work with chan_skinny in 1.4
15:19.58*** join/#asterisk codefreeze (n=steve_mu@216.166.159.235)
15:20.09Dr-LinuxQwell: what about 1.2.0?
15:20.12Qwell[]no
15:20.23mafkeeschan_sccp
15:20.54Dr-Linuxmafkees: chan_sccp sucks :@
15:21.05mafkeesno kidding
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15:21.30Dr-LinuxQwell[]: the thing is that, i can't use 7935 on production system
15:21.39Dr-Linuxthat patch is only for svn version
15:21.53*** join/#asterisk sahafeez (n=sahafeez@67.109.14.227.ptr.us.xo.net)
15:22.22Dr-LinuxQwell[]: can you please give me the link for that patch again? the for cisco 7935
15:22.36Qwell[]I don't know the bug number, you'll have to search for it
15:22.42Qwell[]look for bugs by "slimey"
15:22.57Dr-Linuxok
15:24.03*** join/#asterisk __freedom__lover (n=eduardo@clipper.provale.com.br)
15:24.06__freedom__loverhi all
15:24.22*** join/#asterisk `p4r14h`work (n=josh@24-119-48-78.cpe.cableone.net)
15:24.40__freedom__lovercan anyone tell me what mean blue/yellow/red alarm in zap status?
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15:31.55__freedom__loveralow?!
15:32.12tzafrirred: "nothing connected" (or: no layer 1 connection)
15:32.34*** join/#asterisk tuan_modulis (n=chatzill@3-82-252-216-static.enter-net.com)
15:33.00tzafriryellow: IIRC it means bad CRC. Not really sure
15:33.15__freedom__loverhum, thanks
15:33.16tzafrirAs for blue: I don't know.
15:33.24tuan_modulishello everyone, im looking for a recommendation for an IAX2 softphone
15:33.34tzafrirkiax
15:33.38tuan_modulisgotcha!
15:33.52__freedom__lovertzafrir: i've looked for that information, but nothing good explanation
15:35.07*** join/#asterisk Slingky (n=na@modemcable199.182-200-24.mc.videotron.ca)
15:35.18ManxPower__freedom__lover: What issue are you having?
15:35.30Slingkyhi! could somebody tell me how to change sounds ?
15:35.36Slingkyi download french-gsm
15:35.45Slingkybut do i just need to replace a folder or what ?
15:35.53tuan_modulismake a french folder
15:35.58tuan_modulisput them all in there
15:36.01tzafrirSlingky, it is done automatically when you set the language
15:36.14tuan_moduliswhen you set the language, set to name of folder
15:37.14Slingkytzafrir, i think it point currently to /var/lib/asterisk/sounds , no ?
15:37.29tuan_modulissee thewiki for SetLanguage
15:37.34tuan_modulis~thewiki
15:37.36jbotsomebody said thewiki was at http://www.voip-info.org/wiki-Asterisk
15:38.07Slingkythanks a lot guys!
15:38.55*** join/#asterisk sahafeez (n=sahafeez@67.109.14.227.ptr.us.xo.net)
15:38.58Ch0HagI think I have a bug here.
15:39.24Ch0Hagsip_addheader() is called with struct ast_channel set to one value, then sip_call is called with it set to another.
15:40.08Ch0HagEither another ast_channel is erroneously created or one is mis-copied to the other.
15:40.54__freedom__loverManxPower: hi, i've configured my te110p card, but it is changing its status, among blue, yellow and red
15:41.22ManxPower__freedom__lover: That is usually a telco issue.
15:41.27ManxPowerAssuming you have your card configured
15:41.35ManxPower__freedom__lover: what country are you in?
15:41.41ManxPowerand what is your span= line?
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15:44.37__freedom__loverManxPower: i'm from brazil
15:44.53ManxPower(10:41:35) ManxPower: and what is your span= line?
15:45.36__freedom__loverManxPower: my span is 1
15:46.39ManxPowerThat is wrong.  It should be something similar to span=1,1,0,ccs,hdb3,crc4
15:46.56__freedom__loveroh yeah
15:47.09__freedom__lovermy span is 'span=1,0,0,cas,hdb
15:47.14ManxPowerWhat is your EXACT span= line.  copy and paste it.
15:47.14__freedom__loversorry
15:47.27__freedom__loverspan=1,0,0,cas,hdb3,crc4
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15:47.58ManxPower__freedom__lover: Do you know for sure that your line is cas, hdb3 and crc3?  Only your telco can tell you.
15:48.05ManxPower..er..crc4
15:48.37ManxPoweryou should have a 1 instead of a 0 as the 2nd field or your timing will be off.
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15:54.30Dr-LinuxManxPower: what you would recommend upgrade to latest 1.2.x? or 1.4.x?
15:56.40aydiosmioI had no idea the original telephone systems had on clock source (like a cluster of cesium clocks)
15:56.45aydiosmioone*
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15:59.03Dr-LinuxQwell[]: what you recommend?
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16:01.02ManxPowerDr-Linux: I have never used 1.4 because it has not been out long enough and there seems to be too many outstanding issues.
16:01.35Dr-Linuxi see
16:01.37MercestesDr-Linux:  CCM.  *runs and hides*
16:01.50ManxPoweraydiosmio: T-1 timing has NOTHING TO DO WITH TIME
16:02.01Ch0HagHence the name.
16:02.21Hmmhesaysack i'm out of cdrs
16:02.23Dr-LinuxCisco Call Manger?
16:02.35MercestesDr-Linux, yea, but I'm kidding.  I've never used it.
16:02.41aydiosmioManxPower: not for tick tock time
16:02.44aydiosmiobut bit timing
16:03.01aydiosmiohttp://www.oreilly.com/catalog/t1survival/chapter/ch05.html
16:03.04ManxPowerA more correct term would be "sync" not timing
16:03.12aydiosmiowhat I'm reading
16:03.12Dr-LinuxMercestes: if i'd have it handly i would do something, but i never seen
16:03.28Dr-Linuxhandy*
16:03.32MercestesDr-Linux, It's like the opposite of free.
16:03.58*** join/#asterisk deeperror (n=deeperro@mail.banctel.com)
16:04.00Dr-Linuxsorry , but i didn't understand :S
16:04.21MercestesDr-Linux, I was just teasing/trolling.
16:04.30Dr-Linuxlol
16:04.32Dr-Linuxgood good
16:04.33Dr-Linuxdo it
16:04.38Dr-Linuxi wont mind :D
16:06.06*** join/#asterisk nasls_lsa (n=chatzill@87.203.116.128)
16:06.12deeperrordoes anyone know of some good quality cordless sip phones?
16:07.00ealdfun stuff, I have an asterisk box that grows in cup usage by their own, each week has 10% more CPU usage than the previous one, but the machine is rebooted every night and numbers of calls is around the same
16:07.08*** join/#asterisk SwK[Work] (n=SwK@24.214.206.254)
16:07.52*** part/#asterisk oQPa (n=uawename@84.Red-83-40-182.dynamicIP.rima-tde.net)
16:08.54k31thbit off topic i know but can anyone recomend a decent UK VoIP provider?
16:09.07Ch0HagDefine 'decent'.
16:11.17*** join/#asterisk musse- (n=kallekas@static-212.214.40.123.addr.tdcsong.se)
16:12.07*** join/#asterisk Gunnar (n=gunnar@nat.sigmasoft.com)
16:17.15ManxPowerAll VoIP providers suck
16:17.38mogheh
16:18.02*** join/#asterisk fakhir (i=fakhir@unaffiliated/fakhir)
16:20.44ManxPowerFor some bizarre reason when I added a 4th site to my network OSPF started losing routes.
16:21.47*** join/#asterisk harleya (n=harleya@207.108.166.2)
16:22.07__freedom__loverManxPower: thanks
16:22.13*** join/#asterisk koel (n=ParaNoir@84-53-96-51.fiber.unet.nl)
16:22.33koelHey, does anybody know which SIP client displays contact status information? (online/offline)
16:22.45*** part/#asterisk harleya (n=harleya@207.108.166.2)
16:23.19ManxPowerkoel: what makes you think SIP even supports that feature?
16:25.58koel:) the wiki says the latest dev of asterisk has a presence feature ;)
16:26.13koelhttp://www.voip-info.org/wiki/view/Asterisk+presence
16:26.39Gido-Ekoel, asterisk doesn't only support sip.
16:27.57ManxPowermost of those features are for call status.  hint is specifically for call status (in use, not in use)
16:28.06ManxPowernot "ping able, not pingable)
16:29.24ManxPowerthe only way to really know of a device is reachable is to send data to it.
16:31.12*** join/#asterisk syzygyBSD (n=chatzill@66.226.228.204.cpe.speedyquick.net)
16:31.22necromcris there any way to push an extensio on to stack manually?
16:31.26syzygyBSDhi
16:31.36*** join/#asterisk russellb (i=russellb@asterisk/developer-and-stable-maintainer/drumkilla)
16:31.36*** mode/#asterisk [+o russellb] by ChanServ
16:31.40necromcrsay, i have an extension number in a variable and want to dial it
16:31.42syzygyBSDnecromcr: do you mean through the console?
16:31.47syzygyBSDoh...
16:31.56syzygyBSDgoto(context,extension,priority)
16:32.13aydiosmiostack? hehe
16:32.38necromcrsyzygyBSD: no.. um..
16:33.01syzygyBSDumm explain yourself better
16:33.27necromcrexten => _80!,1,Set(ORIG_EXTEN=${EXTEN:2}) exten => _80!,n,Goto(notranji-klic,s,1)
16:33.33necromcr(trying to :)
16:34.02necromcri pick up isdn phone, dial 80 and get notranji-klic. there i handle the next 3 digit extension
16:34.11necromcr(so full number i dial is say 80123)
16:34.52necromcr:-\
16:35.12necromcrwell, if i enter 80123 and then dial.. 80 get's handlede and 123 doesnt
16:35.16necromcrhandled
16:35.20syzygyBSDoh.. so in the end you want either a exten=>_XXX,1,goto(newcontext,${ORIG_EXTEN}${EXTEN},1)?
16:35.22aydiosmiowhy not goto(notranji-klic,${EXTEN},1)?
16:35.49syzygyBSDanother good solution...
16:36.00aydiosmiootherwise ${EXTEN} in the new context is "s" not "80123"
16:36.37ManxPoweruh, why not just Goto(newcontext,${EXTEN},1)
16:36.51ManxPowerthe only time you would have "s" is in macros or if you Goto(s,1)
16:37.22aydiosmioanyone else want to give him the solution?
16:37.24ManxPowerand in a macro you can do an exten => s,1,Goto(${MACRO_EXTEN},1) and having a matching exten line in the macro
16:38.45necromcrdarn
16:38.50necromcri bow to you all
16:39.47necromcri must have been really sleepy when i was testing that option as i was sure, goto(..,EXTENSION...) wouldnt work if i had pattern matching extensions..
16:39.58syzygyBSDit is hard when you think something is a lot harder than it is...
16:40.08necromcrsyzygyBSD: yea :)
16:40.11penguinFunkit's easy when you know how
16:40.25penguinFunksame with anything
16:40.41syzygyBSDna, some things are never easy
16:41.24penguinFunklike dealing with girlfriend on pmt
16:41.27penguinFunk?
16:42.16syzygyBSDI was thinking comming up with an answer to "does this make me look (fat|bloated|like a cow)"
16:42.39aydiosmiothe answer is "honey, you look amazing"
16:42.41*** join/#asterisk mmartinn (n=martins@128.227.123.22)
16:43.02penguinFunk;)
16:43.04*** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290)
16:43.04penguinFunkeasy
16:43.14syzygyBSDhmm, guess I didn't know how...
16:44.01mmartinnhey hey... question for the gurus... Gentoo wants to give me a new /etc/hotplug/usb/xpp_fxloader.usermap... will this affect Zaptel?
16:44.59syzygyBSDmaybe.. just recompile afterwards...
16:45.08bulleManxPower: just save the old file just in case ?
16:45.09ManxPowermmartinn: do you have an XPP channel bank?
16:45.15mmartinnManxPower: Negative
16:45.25ManxPowermmartinn: then it should not be a problem
16:45.28mmartinnI suppose I'll save the old one and see what happens :)
16:45.34mmartinnthanks guys :)
16:45.46necromcrhotplug?
16:45.56necromcrisnt' hotplug out of the date with udev?
16:46.04bullenecromcr: yes and no afaik
16:46.07necromcr(remove the)
16:46.15bullenecromcr: the hotplug functinality is now mainly handled by udev
16:46.26mmartinnI believe it was built in, but the names of some scripts and files are still hotplug
16:46.34bullenecromcr: but some distros ( dont know about gentoo ) still keep the name as hotplug
16:46.37bulleManxPower: ye
16:46.53mmartinnYeah... Gentoo does AFAIK; udev now handles it but there's still stuff called it.
16:46.59*** join/#asterisk nextime (n=nextime@unaffiliated/nextime)
16:47.22mmartinnThanks guys... I'll be back in later, when not at work 8-)
16:47.28necromcrgood luck mmartinn
16:47.36nextimehello. In * 1.4.x ther's a "stun" command in cli. Is it a stun support for sip channels? how to can i use it?
16:47.44*** part/#asterisk mmartinn (n=martins@128.227.123.22)
16:47.46*** join/#asterisk Fieldy (i=eKsWJ8KX@gentoo/contributor/Fieldy)
16:50.01*** join/#asterisk shido6 (n=shido6@d221-68-200.commercial.cgocable.net)
16:50.19Mercestesnextime, yes.  You take an FXS module and run two leads off of it, wet someone' skin a little, place the leads on their wet skin, and then call a "ringing" to that interface.  viola, stunned
16:50.47aydiosmioLadies and gentlemen, set your faces to "Stunned"
16:51.04nextime"ah. ah. ah."
16:51.31necromcrrofl :)
16:51.55Mercestesnextime, If you mean a "stun server" I've only really seen that used in ATAs and I'm not sure how much Asterisk is involved in that functionality.
16:52.08Mercestesmaybe google aterisk wiki stun
16:52.20ManxPowerI've never actually seen a situation where you need STUN
16:52.22Mercestess/aterisk/asterisk/
16:52.35MercestesManxPower, Might explain why I've never used it.  lol
16:52.56*** join/#asterisk jm|laptop (n=jm|home@dilbert.jamiem.com)
16:53.02ManxPowerMercestes: STUN is one of MANY ways to deal with NAT + SIP
16:53.21syzygyBSDanother way is not to use NAT
16:53.46necromcrsyzygyBSD: some are forced to use nat
16:54.05necromcrsyzygyBSD: here in .si umts lines are all nat-ed
16:54.07ManxPoweror use the Asterisk SIP+NAT features
16:54.14nextimeManxPower : for example where you are using asterisk as a "client", you are inside a nat, and you can't do a port forward, and you are a dummy user that want to set stun 'cause your voip isp tell to use it?
16:54.36syzygyBSDwhere is .si
16:54.47ManxPowerI guess the only time STUN might be useful is if you are a cheap bastard and have your Asterisk server behind a dynamic IP address that is natted, but honestly, if that is the case SIP+NAT is the least of your problems
16:55.13ManxPowernextime: in that case you also have much more serious issues than SIP+NAT
16:55.27nextimeManxPower : i don't need stun for myself. I'm working on a http manager based gui (something like the asterisk-gui), so, if the stun cli command exists, i want to know why exists and eventually how can i use it
16:55.28syzygyBSDWe have used stun for phones behind nat
16:55.30bulleManxPower: its not a choice for everyone, to get a public static ip adress
16:55.45penguinFunkslovenia
16:55.54ManxPowerbulle: if you want to run a server you should have a public static IP address
16:56.14bulleManxPower: yeah, but what do you do, when you cant get one ?
16:56.25ManxPowerbulle: you don't run a server.
16:56.42ManxPower"How can I drive to work if I don't have a car?"  "You get a car."
16:56.44bulleManxPower: sounds like a pretty crappy solution to the problem imho
16:56.59penguinFunkManxPower: or cycle ?
16:57.07bulleManxPower: or just take the buss, or train, or walk ?
16:57.23nextimeManxPower : what about running a "local" server for clients in lan, and use it as a "concentrator" to connect all clients on a voip isp in a small office where you can't have a pubblic ip? ( just to say an example )
16:57.33ManxPowerbulle: correct.  So co-locate a server somewhere with a static IP.
16:57.43nextimeservers != public servers
16:57.48ManxPowernextime: you don't need a static ip for that
16:57.53bullenextime: indeed
16:58.15*** join/#asterisk raisendman (n=masterra@203.87.200.78)
16:58.20ManxPoweryou need a static IP if you want clients to connect to the asterisk server from outside the NAT domain/network
16:58.32penguinFunkdyndns ?
16:58.38ManxPoweryou do not need it if asterisk is acting as a sip client
16:58.59ManxPowerpenguinFunk: that would be a solution if asterisk did not stop working after any DNS failure.
16:59.09MercestesManxPower, don't you love how people come in here askign questions and then tell you your answers are wrong?  lol
16:59.22ManxPowerMercestes: you get used to it after a while.
16:59.24cpmMercestes, you R rong!
16:59.33Mercestescpm:  I'm sorry!  I tried.
16:59.51ManxPowerMercestes: I've been using asterisk for a long time.
16:59.53nextimeManxPower : right. so, if asterisk is running, from the "public" view, only as a client, and your voip isp want that you use stun, and "maybe" stun is now supported, why don't use it?
16:59.55Ch0HagIt's not quite as much fun as coming in to ask a question to be told that your question is wrong.
17:00.10ManxPowernextime: because STUN IS NOT NEEDED in that situation
17:00.10Qwell[]Ch0Hag: questions are very often wrong
17:00.11adam_vollrathFun?
17:00.27cpmwould be nice if there was a bar one could set, that forced folks to actually have a basic * system working before coming here with all their wierd stuff
17:00.29Qwell[]usually because the person doesn't understand asterisk enough to ask what they want
17:00.42ManxPowerheck you don't even need to port forward if asterisk is acting as a client in that case
17:00.49syzygyBSDIt takes a true genius like Qwell to know what people mean without them saying it
17:00.52*** join/#asterisk mattchis (n=mattchis@216.54.143.246)
17:00.58*** join/#asterisk outlier (n=tom@70.141.147.180)
17:00.59Qwell[]syzygyBSD: indeed
17:01.00Ch0HagQuite.
17:01.01*** join/#asterisk anthony] (n=anthony@175.21.188.72.cfl.res.rr.com)
17:01.08nextimeManxPower : you know. I know. but not all people know. Expecially where voip isp explicitally say "if you are under nat, use stun", and expecially when you can't put your hand on the router to do a port forward.
17:01.27ManxPowernextime: then you should read the asterisk docs and not your isps docs.
17:01.43Ch0HagAsterisk docs?
17:01.44ManxPowernextime: you don't need to port forward in that situation either
17:01.48Ch0HagWhich ones would those be then?
17:01.58Ch0HagIncomplete set A, or contradictory set B?
17:01.58ManxPowerCh0Hag: The Book
17:02.05nextimeManxPower : I know. You know. not all people know.
17:02.06ManxPower~osmosis
17:02.17jbotosmosis is probably the act of beating yourself on the head repeatedly with THE BOOK, until some measure of absorption has occured ...  or at least until your unconsciousness restores peace to the channel ...
17:02.17nextimeManxPower : anyway
17:02.17syzygyBSD~book
17:02.20jbotmethinks book is a book called  Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11
17:02.20nextimemy question was simple
17:02.20syzygyBSD~docs
17:02.22jbotit has been said that docs is Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com
17:02.24nextimewith asterisk 1.4 a "stun" cli command appear.
17:02.29Ch0HagOr the incomplete *and* contradictory source, maybe?
17:02.29mattchisCan anyone point me to where I can set long distance authorization codes in asterisk?
17:02.42nextimeso, what's this fucking stun command?
17:02.42ManxPowermattchis: you don't.  you write your dialplan for that feature.
17:02.54ManxPowernextime: "help stun" does not help?
17:02.59nextimeManxPower : no.
17:03.01*** join/#asterisk ToyMan (n=Stuart@72.168.167.241)
17:03.05ManxPowerodd
17:03.19nextime<PROTECTED>
17:03.19nextime<PROTECTED>
17:03.29nextimebut stun debug on what?
17:03.32ManxPowernextime: that sounds pretty simple.
17:03.42ManxPowerum, debug information for stun stuff.
17:03.52ManxPowerlike sip debug is debug info for sip stuff
17:04.02nextimeManxPower : yes, it's simple to debug. But how can i use it? ( if i don't use it, what sense as debugging it? )
17:04.16Ch0HagAnd iax debug is debug info for the bits of iax you don't need to debug.
17:04.22ManxPowernextime: if you don't use it then I would expect you would not get any debug output
17:04.32nextimeManxPower : ta-da!
17:04.32*** join/#asterisk inv_arp[work] (n=junya@c-67-191-12-203.hsd1.fl.comcast.net)
17:04.39nextimeManxPower : so, how can i use it?
17:04.56aydiosmiojust give up
17:05.01ManxPoweraydiosmio: me too
17:05.24MercestesCh0Mag:  might I suggest Cisco Call Manager then so you can berid yourself of asterisk's presence?
17:05.55Ch0HagUm.
17:05.56Ch0HagOK?
17:05.57ManxPowergreat.  Now I have people /msg'ing me asking questions.
17:06.07ManxPowerLike I'm their own personal tech support bitch
17:06.13Mercesteslol
17:06.16Qwell[]ManxPower: aren't you?
17:06.17Mercestesyour popular, Manx.  :)
17:06.19aydiosmio<PROTECTED>
17:06.20Qwell[];)
17:06.35*** join/#asterisk oej (n=olle@apollo.webway.se)
17:06.53Mercestessee.  If you were a complete ass like me, you wouldn't have ppl messaging you (like me).
17:06.56ManxPowerIf you want personal private help from me the first thing we need to do is decide how I am going to bill you for it.
17:07.09nextimeManxPower : setup an * server connected to telephony world by a PRI, take a premium number and redirect people to your phone numbers, so, you get some money :)
17:07.12mattchisHows it going Mercestes!!
17:07.24Qwell[]m/sg ManxPower hi, can you help me be the next vonage?  I can paypal you $20
17:07.26Mercestesmattchis!  Sup man?  What are you doing on this side of freenode?
17:07.37Mercestesyou already have private support.  :P
17:07.44ManxPowerm/sg My rate is $78/hr
17:07.53Qwell[]ManxPower: 78?
17:08.03ManxPowerQwell: Yes, I am a cheap whore.
17:08.21ManxPowerQwell: the rate does go up if you are an idiot.
17:08.32Qwell[]to 79.42?
17:08.40ManxPower$2,000/day
17:08.42MercestesManxPower:  I was wondering why you charged me twice that.
17:09.02lokkju_wrkwtf is this vonage patent on VOIP stuff all about?
17:09.09Qwell[]Ch0Hag: and btw, if you don't like the current documentation situation...  feel free to contribute
17:09.10ManxPowerMercestes: you didn't use any lube, that's why
17:09.18*** join/#asterisk canapa (n=canapa@83-64-148-98.wolfsberg.xdsl-line.inode.at)
17:09.28Ch0HagWell yes, but unfortunately I need to actually *understand* the spaghetti code first.
17:09.34*** join/#asterisk CunningPike (n=CunningP@204.239.8.149)
17:09.40Qwell[]such as?
17:09.47MercestesManxPower, Oh.  I thought there was an *extra* charge for lube, not a storage and handling fee for nto using it.  =/
17:09.56Qwell[]because a lot of people understand it just fine
17:10.02Ch0HagThe 13 thousand lines in chan_sip.c, for example.
17:10.21MercestesCh0Mag:  Might I suggest Plan9 then?  YOu can code the entire thing in 13 lines.
17:10.23canapawhat kind of port forewarding would i have to make in my firewall for asterisk to work proper ?
17:10.24aydiosmiolokkju_wrk: I'm sure Google News can help you there
17:10.25Qwell[]well, ask Cisco for their Call Manager code then so you can modify that
17:10.27*** join/#asterisk Y0da^ (n=jwilson@70.159.118.70)
17:10.29Qwell[]...oh, right
17:10.45Ch0HagMercestes: gcc has this wonderful feature called 'linking'.
17:10.53Ch0HagPerhaps you've heard of it?
17:11.06MercestesCh0Hag:  Sorry?  I'm not entirely fluent in troll.
17:11.10*** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982)
17:11.33Qwell[]Mercestes: he's saying we should break up chan_sip.c so that in order to change things, you need to edit 10 files
17:11.46MercestesQwell[]:  Oh!!!  brilliant!
17:11.57Ch0HagI must say, this Digium-recommended discussion forum hasn't exactly got me enamoured of any paid-for support they may be offering.
17:12.17MercestesQwell[]:  And instead of doing it himself (since we gave him the source and all) he wishes to incessantly bitch in here (us being responsible for the source and all) until it magically fixes itself, right?
17:12.36MercestesCh0hag:  There is a major poitn your missing.  This isn't paid-for support.
17:13.01MercestesCh0Hag:  If you wish to *pay* us to bitch then I will be more than happy to personaly listen to you and reciprocate for as long as you feel it' snecessary.
17:13.13MercestesCh0Hag:  I am certain Digium feels the same way.
17:14.48*** join/#asterisk sevard (i=chuck-th@adsl-71-129-115-242.dsl.irvnca.pacbell.net)
17:15.12aydiosmiois it too early to tell Ch0Hag to GFY?
17:15.20Ch0HagPerhaps you should look up the definition of 'enamoured'.
17:15.24heisonanyone here using blacklist() in Asterisk 1.4?
17:15.34MercestesCh0Hag:  Perhaps you should look up the definition of troll.
17:15.44Qwell[]"No definitions were found for enamoured."
17:16.10Ch0HagAmericans...
17:16.12Ch0HagEnamored thes.
17:16.13*** join/#asterisk Dovid (n=Dovid@l192-117-114-1.broadband.actcom.net.il)
17:16.14Ch0Hagthen
17:16.23heisoni used to have exten => s,n,LookupBlacklist(j) and i wonder how the new function should perform the jump
17:16.25giasai68hello i have some problem with fax t.38 on my aterisk 1.4.1 : Unsupported SDP media type in offer:
17:16.26MercestesThis isn't digium.  This is #freenode.  Digium merely suggested #asterisk was helpful for being free.  If you want digium...go buy something.
17:16.30giasai68can you help me?
17:16.40Dovidhi guys. i am trying to install g729 on my asterisk. does asterisk only have g729a or can i get vanilla g729 ?
17:16.47Qwell[]Dovid: only g729a
17:16.49MercestesWe already gave you the book and voip-info.org .  Everyone else figured it out with that.
17:16.50Dovid:(
17:16.53Dovidthanx
17:16.59Qwell[]but isn't it supposed to be compat or something?
17:17.19Qwell[]Mercestes: the people who wrote the book figured it out with far less..
17:17.24giasai68chan_sip.c:4586 process_sdp: Unsupported SDP media type in offer: image 10912 udptl t38
17:17.26Ch0HagAs Digium explicitely recommend this channel, it must be considered an advert for them, and by my experience, and watching the experience of others, it is not a very good advert at all.
17:17.46denonthis channel isn't an advertisement for anyone
17:17.51Gido-ECh0Hag go wank your self.
17:17.51denonwe're just here to talk about asterisk
17:17.55MercestesCh0Hag:   Sorry to loose you as a customer then.  please feel free to report me specifically.  bye.
17:18.08Ch0HagI am not, and have never been, expecting free support, but some manners wouldn't go amiss.
17:18.17ManxPowerDovid: almost nothing out there supports plain g729
17:18.32adam_vollrathManners?  Welcome to IRC.
17:18.33aydiosmioamiss?
17:18.40tuan_modulisthere are lots of unix users here.... we're all egoists by nature
17:18.49Gido-EI am VERRY VERRY VERRY happy with Asterisk and Digium ofcourse.
17:18.54Ch0HagBut then I guess I can't really expect very much on that score from Americans.
17:19.07denonCh0Hag: stop trolling. now.
17:19.14aydiosmioya rly
17:19.27denonif you have a legitimate question - someone may try to help. Not everyone knows everything, and not everyone has time to help.
17:19.41Ch0HagI don't expect them to.
17:19.49Ch0HagHell, I don't even *expect* them to be polite.
17:19.53denonthis is peer to peer - and a very international committee
17:20.01denoner community
17:20.13Ch0HagBut as the public face of a growing company, it's at least a bloody good idea.
17:20.34aydiosmiopretty sure this channel predates Digium.
17:20.40aydiosmiowink wink
17:20.48denonCh0Hag: are the people you're talking about even involved with digium?
17:20.53Mercestes#asterisk is not digium, public face or otherwise.  We're users of the product, asterisk with little or no affiliation to digium.
17:20.58denonany random troll can walk into the channel and start beging rude
17:21.06Ch0HagThat is irrelevant.
17:21.08Mercestesyea, look at me.
17:21.13denonpoint :)
17:21.48denonCh0Hag: it's a community, not a digium support channel
17:22.35Ch0HagDigium recommend #Asterisk, there is even a chance that some of its staff are present. In that case any company with half a clue would do its level best to ensure the company and product is always looked upon in a good light.
17:22.38tuan_modulisI join the channel mostly to leech off circumstantial advice
17:22.56Qwell[]tuan_modulis: that's basically how I learned..
17:22.58denonCh0Hag: that would involve us silencing everyone but digium staff
17:23.04denonwhich is impossible in a peer to peer community
17:23.10denonyou take the good with the bad on irc
17:23.13Qwell[]at that point, it's no longer a community
17:23.18denonexactly
17:23.30denonCh0Hag: you may find the mailing list to be more to your liking, if you don't like the casual atmosophere of irc
17:23.32Ch0HagMy point, from way back when, is that either Digium don't care about their public appearance or would probably be wise to no longer recommend this channel in any form.
17:23.39MercestesCh0Hag:  Is there a specific issue/problem you have?  now would be a good time to bring it up.  You have our attention now.
17:23.56denonCh0Hag: I believe it's not a matter of them not caring about their appearance, just that they were trying to give you some community options
17:24.06aydiosmioCh0Hag: please, just leave
17:24.21Mercestesyea, judging Digium based on an IRC channel is like judging microsoft based on the Dixie Chicks.
17:24.45denonCh0Hag: you'll have to excuse me, I'm late for a meeting - but I do think you'll find a mix of good and bad on irc, but sift through it all and you'll find an extensive amount of knowledge
17:24.56MercestesI mean, all that awful noise did cmoe through media player, right?  If Microsoft cared about their image they'd censure some of that instead of allowing their trademark to be defamed.
17:24.59Ch0Hag"Many of these people join *our live IRC Asterisk chat channel*..."
17:25.14aydiosmioCh0Hag: just. go.
17:25.18adam_vollrathThey should probably change that.
17:25.26Gido-ECh0Hag bye!
17:25.42Ch0HagThat would be a good idea. It demonstrates a sense of ownership/control which is not there.
17:25.45aydiosmioif we're wasting so much of your precious time not answering your query, perhaps it's not wise to waste it arguing with the wind
17:26.01Mercesteshe doesn't even have a query...
17:26.42Ch0HagAll the egotism and bad manners I've seen could easily be ignored if Digium made it explicit that they only affiliated with the channel in an anciliary manner.
17:26.50aydiosmiothanks, noted
17:26.53aydiosmiosee ya later
17:27.01adam_vollrathSend them an email.  Seriously.
17:27.12Ch0HagAs they state the opposite, I assumed that they had at the very least some presence in here.
17:27.22Qwell[]of course they have a presence in here
17:28.06Ch0HagPersonally, I couldn't give a toss, I'm in a large family and I've been a geek all my life, I can give and take internet crap with the best of them.
17:28.23CunningPikeCh0Hag: If you want the complete and undivided attention of Digium staff, an ABE license may be right for you
17:28.24aydiosmio]okay see ya then
17:28.25wunderkin...
17:28.28aydiosmiobuh bye
17:28.33Ch0HagBut Digium's public image is almost certainly suffering.
17:28.45techienah
17:28.49Gido-ECh0Hag no, you let us suffering.
17:28.51aydiosmiono, but thanks for your concern
17:29.01Qwell[]Ch0Hag: Do you have a specific complaint?
17:29.04Corydon-wCh0Hag: that's enough.  You've berated the channel for the past hour.  Ask a question now or leave.
17:29.12Ch0HagWell, no, Digium's public image *has* suffered.
17:29.15DovidQwell: can you have a look at this ?
17:29.16Dovidhttp://www.pastebin.ca/410774
17:29.38Qwell[]Dovid: You're going to have to call Digium support
17:29.48Dovidis it a server issue or an issue with me ?
17:29.53Gido-E:-)
17:29.58Dovidur guess
17:29.58Qwell[]well, it looks like a proxy problem on your end perhaps
17:29.59Dovid?
17:30.06Dovidthanks
17:30.28Ch0HagQwell[]: No. A suggestion. That Digium don't condone the channel or give explicit, clear indication that it has nothing (formally) to do with them.
17:30.38*** mode/#asterisk [+b %Ch0Hag!*@*] by Corydon-w
17:30.52Gido-Ethanx Corydon-w
17:31.22DovidThanks Corydon
17:31.32DovidQwell: do u guys work on UDP or TCP for the server ?
17:31.34CunningPikeCh0Hag: You're absolutely correct - sales@digium.com
17:31.56Corydon-wCunningPike: please don't feed the troll.
17:31.56Dovidi tried to get to it from a diffrent server in a diffrent data center thru telent and conneciton refused
17:32.05*** join/#asterisk rudholm (i=rudholmm@nat/yahoo/x-29f6201b3a2f8b78)
17:32.24Corydon-wCunningPike: he can't respond in-channel anyway
17:32.42Qwell[]Dovid: unfortunately, nobody in here is going to be able to debug that issue at all..  about the best I can recommend is checking your firewall settings
17:33.00DovidQwell: even from 2 diffrent server in 2 seperate locations ?
17:33.03Dovidok. i will call em
17:33.11Qwell[]Dovid: anything's possible :)
17:33.13Qwell[]and that would be best
17:33.38Dovidthx. in Israel ATM - on evdo so i cant use voip @ 15cents a minute....
17:33.58aydiosmiotalk faster.
17:34.03Dovidlol
17:34.06*** join/#asterisk dhill (i=dhill@fog.mindcry.org)
17:34.11Dovidlike the cell commercials
17:34.18dhillhello
17:34.26dhillany chan_sip developers here?
17:34.43aydiosmiodhill: tried #asterisk-dev?
17:34.47dhillahh, ok
17:34.49*** part/#asterisk dhill (i=dhill@fog.mindcry.org)
17:35.13russellbthere are no chan_sip developers
17:35.40aydiosmioin soviet russia, chan_sip develops you!
17:36.24russellbi was looking at some code in there last week
17:36.27russellbit did something to me ...
17:36.40Mercestesnothing....unnatural I hope.
17:37.05russellbmommy, chan_sip touched me :(
17:37.07DovidQwell: may I PM ?
17:37.52*** part/#asterisk nextime (n=nextime@unaffiliated/nextime)
17:37.57Dovidor russelb: may I PM ?
17:38.20russellbno
17:38.30Dovidok. something that i dont wana say in channel ;)
17:39.06MercestesSo!  I'm installing asterisk onto a 256k flash memory card using Soentoo with a Tei410P mounted in an embedded Soekris box.  Anyone familiar?
17:39.23MercestesDovid:  We already know.  You have a crush on russelb.
17:39.25Qwell[]256k?
17:39.38Qwell[]seriously?
17:39.41MercestesQwell[]:  yea, my boss is cheap.  Plz!  help!  I'll lose my job.
17:39.48Qwell[]ahh, smartass
17:39.50Qwell[]:p
17:39.51MercestesQwell[]:  256mb actually
17:39.55Qwell[]You had me for a second there ;)
17:40.01Mercesteslol
17:40.04Mercestesactually, I really am....
17:40.11Mercestesbut I'm not asking for help (yet.)
17:40.30Mercestesactually, th ecard says 256(sigma symbol)   no clue what that means.
17:40.40Mercestesoh.  *pulls card out*  it's mb sideways.
17:40.59Dovidhehe
17:41.29DovidQwell: turns out my register program is from the last time i bought codecs which was 3 years ago
17:41.38Qwell[]nice
17:41.41*** join/#asterisk Fieldy (i=ntes4WZj@gentoo/contributor/Fieldy)
17:42.56Mercestesnow I just need to know if I can fit a kernel/asterisk/mysql/apache/Fop all on a 256mb card.  =/
17:43.07Qwell[]Mercestes: 256mb is *huge*
17:43.23Mercestesyea, for linux I suppose.
17:43.29MercestesI plan to remap all the /var stuff onto a lappy Hdd.
17:43.39MercestesFor CDrs and recordings.
17:43.50Dovidi like the new one much better ;)
17:44.37MercestesI haven't even installed an OS yet.  I dragged out a PC, put a gentoo boot cd in, downloaded stage 3 and source, chrooted into gentoo, downloaded soentoo source, chrooted into that...
17:46.20MercestesSomeone wanted to knwo how to setup phones through 2 nats and a firewall.  I told them OpenVPN.
17:50.27*** join/#asterisk oej (n=olle@apollo.webway.se)
17:50.50*** join/#asterisk r0d3nt (n=RatMan@punk.valuetel.net)
17:53.58MercestesHrm.  I should write a wiki on this Soentoo build.
17:55.17*** join/#asterisk DarKnesS_WolF (n=sherif@unaffiliated/sherif)
18:02.11ManxPowerMercestes: I've done Asterisk <-> NAT Router <-> Internet <-> NAT Router <-> NAT Router before without any issues
18:02.49ManxPowerASTERISK and the NAT Router for Asterisk had to be set up correctly, but that was all
18:03.04syzygyBSDhow do I view the debug log.. stupid question I know
18:03.44ManxPowersyzygyBSD: do you have logging of debug mesages enabled?
18:03.56MercestesManxPower, Your not even from this solar system tho.  We're talking about humans setting it up here.
18:04.07syzygyBSDManxPower: yes
18:04.18ManxPowersyzygyBSD: /var/log/asterisk
18:04.46MercestesBut they are gonig gateway -> some unnamed firewall -> asterisk (with a public ip of 192.169??  huh?) with their own iptables magic setup.  so, yea, not setup right.
18:05.15MercestessyzygyBSD, You can also add debug to the console in logger.conf if you wish'
18:05.36syzygyBSDhmm, I checked that and I dind't think it had the debug messages...
18:05.36syzygyBSDahh, thanks
18:05.38toombaloombaquick question, when I have something like exten => 1111,1,Agi(agi://<server_ip>/cnam.agi?number=${CALLERID}) what kind of connection is being made from my asterisk box to this server at the IP, is it HTTP?
18:05.39ManxPowerMercestes: that is like watching a retarded kid take the SATs.  It would be funny if it was not so sad.
18:05.40syzygyBSDthat is what I was looking for Mercestes
18:05.52MercestessyzygyBSD, Your welcome.
18:05.56MercestesManxPower, I agree.  Lol
18:06.03ManxPowertoombaloomba: IP
18:06.28*** join/#asterisk lucifr (n=chatzill@66.6.221.64)
18:07.00MercestessyzygyBSD, don't forget logger reload or core reload logger or whatever it is .
18:07.13lucifrHi everybody..
18:07.15toombaloombaManxPower im asking because this is for a CNAM service but I dont want asterisk to call directly, I want to make a HTTP call to it from another system, and then my asterisk will get the CNAM from here
18:07.29toombaloombaManxPower so im wonder if I do http://IP/cnam.agi?number=12354 what will I get
18:07.31ManxPowertoombaloomba: it sucks to be you
18:07.53Mercesteshi, luci.
18:07.54ManxPowertoombaloomba: What in the world makes you think asterisk will do an HTTP connection
18:07.59ManxPoweror even support it.
18:08.02toombaloombanot from asterisk
18:08.06lucifrHi there
18:08.13toombaloombaManxPower not from asterisk itself
18:08.19toombaloombaManxPower and yes it can do HTTP, u heard of CURL?
18:08.22ManxPowertoombaloomba: if it's not from Asterisk then why are you running exten => 111,1,AGI(whatever)
18:08.29*** join/#asterisk Dovid (n=Dovid@l192-117-114-1.broadband.actcom.net.il)
18:08.30lucifrI'm new to Asterisk (AsteriskNOW) and I've been trying to get it to work with my VoIP provider but no luck. Now I'm trying it for the second time, but I think I'll need some help.
18:08.32ManxPowertoombaloomba: Curl != AGI
18:08.34giasai68i have this problem:chan_sip.c:4586 process_sdp: Unsupported SDP media type in offer: image 10912 udptl t38
18:08.35Dovidhi guys
18:08.46giasai68can you help me?
18:08.50*** join/#asterisk logan|w (n=sadfasdf@station441.dallasix.net)
18:08.55DovidMar 26 14:07:54 WARNING[2416]: channel.c:2603 ast_request: No channel type registered for 'h323'
18:08.55DovidMar 26 14:07:54 NOTICE[2416]: app_dial.c:1059 dial_exec_full: Unable to create channel of type 'h323' (cause 66 - Channel not implemented)
18:08.58ManxPowergiasai68: tell your SIP client to stop sending unsupported stuff.
18:09.11ManxPowerDovid: That is a pretty obvious error message.
18:09.17Dovidthis means that i didnt install the h323 driver for asterisk ?
18:09.26ManxPowerDovid: Yup!
18:09.36Dovidwierd. cause i did it
18:09.37lucifrIf I install AsteriskNOW can I avoid the GUI for now and configure the configuration file manually?
18:09.38Dovidlet me try again
18:09.50ManxPoweractually it means "none of the channel drivers told asterisk they support Dial(h323/whatever)
18:09.50giasai68how can i fix? i'm tring to send a fax trought atserisk but i'm not able to do this
18:10.10ManxPowergiasai68: Since asterisk does not support fax over ip....
18:10.16Dovidah
18:10.23Dovidthe readme says to do make opt
18:10.24ManxPowergiasai68: what verison of Asterisk
18:10.40giasai68version 1.4.1
18:10.42Dovidbut when doing make opt i get
18:10.42Dovidmake: *** No rule to make target `opt'.  Stop.
18:10.54ManxPowerDovid: I can't help yoiu with H323, very few people can.
18:11.32ManxPowergiasai68: you are trying to do T.38.  Check the Wiki, as I don't know anyone that has gotten T.38 to work with Asterisk, but rumor is that 1.4 is supposed to support it in a limited way.
18:11.40Mercesteslucifr, #asterisknow can help you with asterisknow.  I woul dsuggest just using the config files tho.
18:11.46Mercesteslucifr, I can't answer your question tho.
18:12.02ManxPowergiasai68: by "limited way" I mean asterisk cannot terminate a T.38 call, it can only pass it thru to the final destination
18:13.12Dovidanyone here know h323 ?
18:13.42ManxPowerDovid: there are at least FOUR different H323 channel drivers for asterisk.  you'll need to ask if people are using the one you are using
18:14.18Dovidvanilla h323
18:14.18ManxPowerDovid: no such thing
18:14.19Dovidtrying to compile /asterisk/channels/h323
18:14.19*** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman)
18:14.19ManxPowerDovid: that would be the "Nufone H323"
18:14.22Dovidah.
18:14.27Dovidthat is made for nufone ?
18:14.51*** join/#asterisk mark_coastal (n=chatzill@rrcs-67-78-216-114.se.biz.rr.com)
18:14.51ManxPoweras opposed to the Objective Systems H323, Woomera H323, and the GPL one in asterisk-addons
18:15.00DovidManxPower: where are the other ones ? and i want a list of other ones
18:15.01giasai68yes now i'm doing this i'm using asterisk as pass it thru... i'm snding fat with fax machine on adapter phone via ip, this equipment sipport t.38, i'm using asterisk as sip proxy and asterisk machine send this to a gateway connected to pri
18:15.04ManxPowerDovid: no it is made BY nufone.
18:15.08Dovidok
18:15.53mark_coastalhello, looking for a manageable way to set oubound callerid for extensions (or groups of extensions) - preferably using realtimedb. Using vanilla asterisk.
18:16.25*** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com)
18:17.00ManxPowerSince the GPL one in asterisk-addons was comissioned by Digium, I would look at that one.
18:18.02*** part/#asterisk deeperror (n=deeperro@mail.banctel.com)
18:18.02*** join/#asterisk xo8ox (n=pride_32@wsip-66-210-250-2.ph.ph.cox.net)
18:18.07xo8oxguys when dialing an extention I get this warning:
18:18.09xo8oxWARNING[16593]: app_dial.c:1081 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination)
18:18.26Mercestesxo8ox:  That means the peer is offline.
18:18.37xo8oxthe phone u mean ?
18:18.44Mercesteswhatever it is.
18:18.50Mercestescould be a sip oxen for all I care.
18:18.56xo8oxits a polycom phone I'm trying to dial to
18:18.57xo8oxlol
18:19.01DovidManxPower: whats the diffrence between h323, oh323 and ooh323 ?
18:19.03MercestesIt is offline then.
18:19.04xo8oxI hate these polycom phones
18:19.07Doviddiffrent protocols ?
18:19.09Mercestespolycom ftw!
18:19.10ManxPowerit could also mean that the device never registered or that you screwed up your sip.conf entry
18:19.15Mercestespolycom > world
18:19.22ManxPowerDovid: they are all written by different people
18:19.35*** mode/#asterisk [-b %Ch0Hag!*@*] by Corydon-w
18:19.41ManxPowerxo8ox: "sip show peers" will tell you if it is registered or not
18:19.44lucifrMercestes, do yo know where I can find the configuration files for AsteriskNOW>
18:19.46lucifr?
18:19.48Dovidbut they call act the same ?
18:19.55xo8oxaha ok thanks
18:19.56Mercesteslucifr, in #asterisknow
18:20.00ManxPowerxo8ox: we drown people on this channel that dis polycoms
18:20.05lucifrok, thanks
18:20.07Qwell[]polycom sucks
18:20.07ManxPowerDovid: I highly doubyt it.
18:20.08Dovidjust compiled it from asterisk add-ons and still getting errors - this is fun
18:20.10Qwell[]kidding!
18:20.17Qwell[]don't tar/feather me please :p
18:20.18*** join/#asterisk techie (n=gus@voip.routedsystems.com)
18:20.26*** join/#asterisk jm|laptop (n=jm|home@dilbert.jamiem.com)
18:20.31*** join/#asterisk billytwowilly (n=chris@S01060016b649355d.ed.shawcable.net)
18:20.41ManxPowerDovid: h323 is one of the hardest, most poorly documented, most complicated things you can do with asterisk.  NOBODY uses h323 if they have any way around it.
18:21.05DovidManxPower: Yup. I agree. But I must use it. carrier only supports it :(
18:21.12MercestesDovid:  get a new carrier
18:21.17Dovidi may have to get a cisco box - which i dont wana so
18:21.23ManxPowerDovid: expect to spend a week getting it to work the way you want.
18:21.26Qwell[]mog: :(
18:21.28DovidMercestes: no one can beat thier rates :(
18:21.32billytwowillyspeaking of new carriers, which one should I use in Canada for the cheapest/best service? The voip wiki is.. confusing.
18:21.45mogQwell, ?
18:21.53Qwell[]<---
18:21.55MercestesDovid:  yea, your bill will continue to be $0.  Now if yo ucan get it to actually *work* every now and then, they may begin charging you.
18:22.02Qwell[]relative to your current location
18:22.14Qwell[]or maybe --->?  whichever
18:22.31Ch0HagWhat's the state of h323 generally?
18:22.40Mercestes*sighs*  Any chance for inherent SS7 support in Asterisk anytime in the near future?  in time for 1.6 maybe?
18:22.54Ch0HagIf this carrier supports only it, that implies that [some] people aren't doing everything they can to avoid it.
18:23.10MercestesCh0Hag:  I think manxpower covered it earlier with:  h323 is one of the hardest, most poorly documented, most complicated things you can do with asterisk.  NOBODY uses h323 if they have any way around it.
18:23.22Ch0HagMercestes: Except Dovid's carrier.
18:23.33mogMercestes, it already works
18:23.35moglibss7
18:23.37Dovidhehe
18:23.38MercestesCh0Hag, didn't you just get unbanned?
18:23.43Ch0HagIs it something it's worth adding to [the bottom of] my list of things to learn?
18:24.53mogbut i dont know anyone that would reccomend it for production
18:24.53Mercestesmog:  zomg!  what ver?  1.4?
18:24.53MercestesCh0Hag:  no.  Honestly, no.  Unless you wanna maintain/fix it.
18:24.53mogits a seperate library i think it only works with trunk
18:24.54Mercestesmog:  hrm.  Yea, I avoid trunk.
18:24.54ManxPowerMercestes: where are you located?
18:24.54Ch0HagRight.
18:24.54mogbeggers cant be choosers Mercestes
18:25.06Mercestesmog:  Aye.  Maybe I can patch it into 1.2.17 or something.  =/
18:25.07Dovidmog: if i get ABE will i get h323 support ?
18:25.14MercestesManxPower, H-town.
18:25.23xo8oxguys in linux how do I add a perm route ?
18:25.27ManxPowerMercestes: you must be a carrier.
18:25.42ManxPowersince no telco that I know will provide an SS7 line to a non-carrier
18:25.49MercestesManxPower, Used to be.  this is to plug into sprint and make my PBX part of their network for free cell phone calls.
18:26.08DovidQwell: if i get ABE will i get h323 support ?
18:26.19Qwell[]Dovid: You'll need to call sales and ask :)
18:26.23Dovidthanx
18:26.35Mercesteslol
18:27.13Mercestesmog:  ....maybe I can come up with a trunk box and iax it on over as a ss7 gateway. =/  hrm.  *ponders evil*
18:27.31*** join/#asterisk `p4r14h`work (n=josh@24-119-48-78.cpe.cableone.net)
18:28.31ManxPowerMercestes: you'll give me a guest account on the box, right?
18:28.45*** join/#asterisk spatulamaan (n=ggilmore@207.188.8.252)
18:28.52MercestesManxPower, I unno, maybe I can hook something up.
18:29.31MercestesManxPower, Or I can document the setup and point you at the (rather cheap) sprint plan that allows you to do it.  :)
18:29.55*** join/#asterisk jm|laptop (n=jm|home@dilbert.jamiem.com)
18:30.17ManxPowerMercestes: how are you linking to spring?
18:30.20ManxPowersprint, even
18:30.24jm|laptop:(
18:30.58jm|laptopI'm having problems with my zap channel - when I try to place a call through the zap chan 1 or if I pickup an incoming call there is just loud hum and a high pitched squeak
18:31.09jm|laptopthe POTS line is fine, I checked with an analogue phone
18:31.27ManxPowerjm|laptop:  could it be ECFO?
18:31.30ManxPower~ecfo
18:31.41jbotEcho Canceler Freak Out, this happens when the rxgain is too high and the echo canceler freaks out.  Some users describe it as "screeching", "feedback", "static", or other useless terms.  If users report "static" on a system where there cannot be static (all digital, PRI, SIP, etc), you might be experiencing ECFO. what happens when the echo canceller suddenly ...
18:31.54jm|laptophmm
18:32.00jm|laptophow might I check?
18:32.03ManxPowerusually it happens in the middle of a call.
18:32.11ManxPowerjm|laptop: set your rxgain to -6
18:32.18*** join/#asterisk Deeewayne (i=dwayne@nat/digium/x-a35a3dbd23c050d7)
18:32.27ManxPoweror better yet, lower it in increments of 2
18:32.48jm|laptopthis is way before a call
18:32.50MercestesManxPower:  http://www.sprint.com/business/products/products/pcsIntegratedOffice_small_tabA.html
18:33.56ManxPowerMercestes: Uh, yeah.  Let me know how it works out for you.
18:34.09Qwell[]eww, sprint
18:34.13Strom_Chi
18:34.17jm|laptopmaybe it's the card
18:34.19jm|laptopbrb
18:34.23ManxPowerjm|laptop: it can't hurt to try it.  also confirm you do not have irq issues
18:34.58MercestesManxPower, Sure.  I'll document for you.  Could be a good sale.
18:35.09ManxPowerMercestes: Did they forget to put in the part where it cures baldness, is a solution to world hunger and gives great backrubs?
18:35.34ManxPowerMercestes: I'm skeptical, but if it works for you, let me know
18:35.53MercestesManxPower, they forgot to put in the part where it requirse SS7 signalling and actually getting it setup requires you to jump through about 90 hoops just to talk to the guy who knows what it is
18:36.02*** join/#asterisk TAMIKAMI01 (n=radamant@200.34.113.90)
18:36.11ManxPowerMercestes: that is typical
18:36.18ManxPowerMercestes: how much does it cost?
18:36.20Mercestesyea.  ss7 is my only hangup righ tnow.
18:36.23Mercestesabout 10 bucks a phone.
18:37.54*** join/#asterisk groogs[h] (n=gregm@cbl-66-102-80-229.wtccommunications.ca)
18:38.03ManxPowerIt would he a hard sell to my users
18:38.23Mercestesgives you 4 digit extension dialing into the office.
18:38.33ManxPowerthat is what would confuse them
18:38.40Mercestestrue.
18:38.41Qwell[]ss7 phones?
18:38.54MercestesQwell[]  Nah, I just need ss7 in *.
18:39.00Qwell[]oh
18:39.02MercestesMog says it's in trunk as libss7 but...it's trunk only
18:39.18Mercestesstupid zaptel blowing up on stupid embedded gentoo system on my stupid dell.
18:39.25ManxPowerI'll have to wait for 1.6 to be released themn
18:39.28Mercestesstuipd ebuilds.
18:39.28mogjust take trunk chan_zap and run it in 1.4
18:39.57Qwell[]frankenversion?
18:40.18ManxPowerAny time there is a phone glitch the head of accounting goes to the president of the company demanding he fire the entire IT department
18:40.20TAMIKAMI01hi
18:40.42Strom_CQwell[]: i got my 7960 working with skinny firmware
18:40.46Qwell[]Strom_C: w00t
18:40.49Qwell[]1.4?
18:40.56Strom_Cis it possible to do transfers?  the key doesnt seem to do anything
18:40.59Qwell[]no :D
18:41.00Strom_Cyeah, 1.4 svn branch
18:41.06Strom_Cthat's...disappointing
18:41.19Qwell[]eventually it will
18:41.52Mercestesyay, frakenversion.
18:41.59Mercestesmog:  svn checkout zaptel?
18:42.01Mercesteshrm.
18:42.09Mercestesso half ebuild, half source, half trunk.  ...I like it
18:42.10tzafrirwhere can I find some asterisk-based games? (dialplan)
18:42.12TAMIKAMI01i have and TDM2400P with (S400M)X4 AND (X400M)X1 and my cuestion i how i be sure i have this modules presents for zaptel?
18:42.25Qwell[]tzafrir: menuselect :p
18:42.28TAMIKAMI01exist any command to checkout this?
18:42.32tzafriranybody with useful keywords?
18:42.49tzafrirQwell[], have you integrated tetris into it yet? or minesweeper?
18:42.56Qwell[]something like that
18:43.02Qwell[]tzafrir: hit 'i' ;)
18:43.20Mercestestzafrir:  I have some instructions on some dialplans using system() to destroy linux distros
18:43.23Qwell[]russellb: gmenuselect ^^
18:43.43tzafrirno, somethis that can be done through the phone
18:43.53*** join/#asterisk jm|laptop (n=jm|home@dilbert.jamiem.com)
18:43.58Qwell[]yeah, twisted and russellb wrote something in dialplan once
18:44.01jm|laptopffs
18:44.01MercestesThis can be done through the phone.
18:44.03Qwell[](I think)
18:44.07adam_vollrathI guess all user input should be validated, even in a Dialplan.
18:44.09TAMIKAMI01i have and TDM2400P with (S400M)X4 AND (X400M)X1 how i configure them?
18:44.14Mercestestwisted doesn't like me. :(
18:44.20tzafrirTAMIKAMI01, at the load of wctdm2400xxp
18:44.25Qwell[]Mercestes: s/me/a lot of people/ :p
18:44.30Mercesteslol
18:44.33Qwell[]twisted rocks though :D
18:44.35tzafriror using either zapscan (.bin) or genzaptelconf
18:44.43TAMIKAMI01zaptel                211364  12 wcusb,wctdm,wcfxo,wcte11xp,wct1xxp,tor2,wctdm24xxp,wct4xxp
18:44.45Mercesteseh, he's never been of any real use to me.
18:44.56Qwell[]Mercestes: get him drunk
18:44.59tzafrirTAMIKAMI01, no need to re-ask the same question over and over again
18:45.17tzafrirTAMIKAMI01, which distro do you use?
18:45.27TAMIKAMI01i'm using fedora core 6
18:45.41TAMIKAMI01with linux kernel 2.6.18-1.2798.fc6xen
18:45.48Mercesteseh.
18:45.49jm|laptopwhen I invoke ztmonitor 1 I get a loud tone from my soundcard, too
18:46.01Qwell[]jm|laptop: that's totally a shared inq
18:46.02tzafrirlook at /var/log/messages . Look for the part where the module has loaded.
18:46.03Qwell[]irq*
18:46.11MercestesI'd rather get him roofied and drop him off at a few bars I know of in the more festive side of town.
18:46.12jm|laptoperk
18:46.15tzafrirso: games, anybody
18:46.21jm|laptopit sort of changes when I dial out via zaptel
18:46.35russellbyou weren't supposed to just tell people.
18:46.35russellbnub
18:46.37jm|laptopbut my tones chirrup rather than ... well ... tone
18:46.38Qwell[]:(
18:46.40TAMIKAMI01tzafrir, kernel: Found a Wildcard TDM: Wildcard TDM2400P (24 modules
18:46.42Qwell[]my point is still valid!
18:46.49russellbheh, yeah
18:47.00russellbbut i'm too lazy
18:47.03Qwell[]heh
18:47.29jm|laptophmm
18:47.47jm|laptopso sip to sip works and when someone dials IN they can here me but I just get a zuzzing noise
18:48.56jm|laptops/here/hear/
18:49.10jm|laptoper: yeah.  Thanks jbot
18:49.21*** join/#asterisk ToyMan (n=Stuart@user-12lcqol.cable.mindspring.com)
18:50.37Mercestesjm|laptop, something is severely wrong. :P
18:50.52Mercestesjm|laptop:  what card is it?
18:51.09jm|laptopOK              Wildcard X100P Board 1
18:51.35*** join/#asterisk bkuhn (n=bkuhn@fsf/member/bkuhn/bkuhn)
18:51.49TAMIKAMI01why i when i use make linux26 for zaptel ztdummy, zttool doesn't compiled?
18:51.57jm|laptopZaptel Configuration ====================== Channel map: Channel 01: FXS Kewlstart (Default) (Slaves: 01) 1 channels configured.
18:52.29TAMIKAMI01why i when i use make linux26 for zaptel ztdummy, zttool doesn't compiled... make: *** [zttool.o] Error 1 ??
18:52.37jm|laptopit was working until a few days ago, incidentally
18:53.53*** join/#asterisk sysreq (n=sysreq@41-198-0-72-ppp.3menatwork.com)
18:54.17TAMIKAMI01why i when i use make linux26 for zaptel ztdummy, zttool doesn't compiled... make: *** [zttool.o] Error 1 ??
18:54.28wunderkinheh z0mg
18:55.15aydiosmioTAMIKAMI01: google PEBKAC
18:55.16Corydon-wTAMIKAMI01: that doesn't tell us the error, only that there was one.  Please use http://pastebin.ca to paste the entire set of error messages
18:56.25TAMIKAMI01ERROR: Module wcfxs does not exist in /proc/modules ????
18:56.30TAMIKAMI01ERROR: Module wcfxs does not exist in /proc/modules ????
18:56.34wunderkinheh z0mg
18:56.41wunderkin:D
18:57.15TAMIKAMI01CAS signalling on span 3 conflicts with Clear channel on channel 64???
18:57.17TAMIKAMI01ERROR: Module wcfxs does not exist in /proc/modules ????
18:57.26Mercesteswhat part of "pastebin.ca to paste the entire set of error messages" did you not understand???
18:57.53wunderkinall of it apparantly
18:58.06MercestesJust a random guess here, but I think your problem is that you didn't follow directions.
18:58.13wunderkinha
19:02.12syzygyBSDshould asterisk auto create the tables in mysql it needs?
19:02.26syzygyBSDassuming it has permission
19:02.29Qwell[]syzygyBSD: no
19:02.42gambolputtyI don't think * can auto create tables
19:02.58*** join/#asterisk pfn (n=pfnguyen@netblock-66-245-252-239.dslextreme.com)
19:03.04syzygyBSDahh, well, does anyone have a dump of the create statments I need?
19:03.19Qwell[]syzygyBSD: they should be in the doc/ directory
19:03.26syzygyBSDthanks
19:03.32MercestessyzygyBSD, They also show up under asterisk wiki RTA google searches
19:03.53syzygyBSDMercestes: doing one now...
19:03.57MercestessyzygyBSD, Possibly not 100% correct but pretty close, and core debug on with debug to console will reveal the SQl statements that are failing if any are failing.
19:04.09syzygyBSDsadly no...
19:04.11Mercestes.
19:04.25syzygyBSDbut I am using an old version trying to debug someone elses
19:05.35*** join/#asterisk bkuhn (n=bkuhn@fsf/member/bkuhn/bkuhn)
19:10.20*** join/#asterisk markit (n=konversa@host119-245-static.72-81-b.business.telecomitalia.it)
19:11.05markithi, how can I "reject" an incoming call with mISDN? without answering it, I mean (I think is possible)
19:11.28*** join/#asterisk CrashHD (n=crashhd@c-67-166-155-233.hsd1.ca.comcast.net)
19:12.37necromcrmarkit: hangup doesnt work?
19:13.15markitnecromcr: I suppose I have to answer first... I want the caller find "busy" without waste a call
19:13.44markitnecromcr: I think ISDN is like GSM, that you can "reject" incoming call without the need of answering it first
19:15.20necromcrmarkit: well.. work for me ..
19:15.41necromcrmarkit: do you want a busy signal or just a simple rejection?
19:18.22anonymouz666r TOK_LP or TOKEN; Input:
19:18.23anonymouz666<PROTECTED>
19:18.32anonymouz666WTF?
19:19.27*** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-88-155-97-204.red.bezeqint.net)
19:19.59aydiosmiooh my lord. verizon biz just sent me 150 pages of test procedures for certification
19:21.16*** join/#asterisk Gunnar (n=gunnar@nat.sigmasoft.com)
19:22.03*** join/#asterisk |dennis| (n=dennis@200.32.233.82)
19:22.13sahafeezquestion, and yes i googled but it is a common term. if i have a CRM like in this case Act! and Asterisk with Polycom IP phones, is there anyway to get the phones to dial via the Act! applicaiton?
19:23.00sahafeezi see the TAPI stuff but i am trying to figure how it dials the phone on the desk
19:24.17Mercestessahafeez, google asttabi
19:24.23Mercestess/asttabi/asttapi/
19:24.45aydiosmioyou have the option of dropping a call file into asterisk via some api or netowrk file system and putting the two lines into a conference or connect them in a sepearate context
19:25.04markitnecromcr: mmm what is the difference? I would like the caller to think is busy
19:25.04TAMIKAMI01somebody speak spanish????
19:25.16Mercestesmarkit, match the callerID and feed them Congestion() with no Answer()
19:25.30aydiosmioMercestes: this looks neat, I may have to integrate this into our SRM
19:25.31aydiosmioCRM
19:25.44necromcrmarkit: http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Busy ?
19:25.51MACscrcan someone remind me of the name of the package that includes debian, asterisk, and freepbx?
19:26.04sahafeezthanks. reading now
19:26.29*** join/#asterisk Moobius (i=Moobius@www2.techcavalry.com)
19:26.46MercestesTAMIKAMI01, Hablo un poco de spainish pero el pequeño muchacho de I habla el asno roto muy pequeño que viola el sintaxis bueno con la lengua no.
19:26.48aydiosmioMACscr: you mean CentOS, asterisk and freepbx, trixbox?
19:27.04markitMercestes:, necromcr thanks a lot, I will try/have a look
19:27.06Mercestesaydiosmio, pretty sweet.
19:27.13MACscrno, its actually debian and asterisk, it might include freepbx
19:27.17MACscri think it starts with an x
19:27.38aydiosmiotoo bad our office phones are POTS with some propreitary PBX, or I could use it to ring our extentions here
19:28.24MercestesTAMIKAMI01, Estaría alegre comer lo más humildemente posible a tus niños asados en un mirador con las flores en asterisco.
19:28.29markitI'm trying to replicate the behaviour of my old isdn pbx... when someone calls, 3 phone ring, but if the "main" phone is busy, then no other phones rings and the caller gets a "busy" signal
19:28.38aydiosmioXorcom Rapid is a Debian/Asterisk distribution program that features an auto-install for
19:28.45aydiosmioMACscr: that it?
19:28.45*** join/#asterisk fbffff (n=fbffff@r74-193-79-81.pfvlcmta01.grtntx.tl.dh.suddenlink.net)
19:29.00Mercestesmarkit:  but..  Asterisk can do so much better than that.
19:29.03MACscraydiosmio: yep, taht was it, thanks
19:29.09markitMercestes: like?
19:29.30Mercestesmarkit:  voicemail.  Queues.  Messages that go "please continue to hold, or press * to leave a message and we will return your call."
19:29.43*** join/#asterisk frenzy (n=frenzy@unaffiliated/frenzy)
19:29.43Mercestesmarkit:  Call forwarding.  Telepathy.  etc.
19:29.55*** join/#asterisk brodiem (n=brodiem@67.110.68.66.ptr.us.xo.net)
19:29.56frenzyHi
19:29.56aydiosmiololIVR
19:30.00markitMercestes: LOL :) queques don't need "agents" and registration?
19:30.22frenzyfor somereason i cannot register from one * to another *
19:30.23frenzyusing IAX
19:30.23frenzy<PROTECTED>
19:30.23Mercestesmarkit:  only if you want them to
19:30.23frenzy<PROTECTED>
19:30.23frenzy<PROTECTED>
19:30.29frenzyTx-Frame Retry[001] -- OSeqno: 000 ISeqno: 001 Type: IAX     Subclass: REGAUTH
19:31.00tzafrir_laptopMACscr, looking for something?
19:31.03drakohey
19:31.12drakocan I use zaptel 1.4 with asterisk 1.2 ?
19:31.22tzafrir_laptopdrako, yes
19:31.37markitMercestes: in any case, I think that the simple "busy" works very good for my business, since seems hard to replicate... I've a isdn bri, 1 port, so a "line" is available even if the "main phone" is used, and the caller does not find busy
19:31.52MACscrtzafrir_laptop: it was xorcom, i couldnt think of the name, but aydiosmio helped me
19:31.58drakotzafrir, i still have the problem with freezing the linux with the echocancelation
19:32.23tzafrir_laptopanybody here actually managed to get a working gtalk connection?
19:32.24markitMercestes: in short, I would like to have the possibility of 2 calls outgoing, or 2 coming, but have to be answered only if the "main" phone (my secretary) is free
19:32.25MACscri really dont like rpath at all, so im looking into new options
19:32.29Mercestesmarkit:  congestion() should work then. I think we also have a busy().
19:32.40MACscrman, i cant stand rpath
19:32.48Mercestesmarkit:  And a chanisavail( as well
19:33.00TAMIKAMI01jajajaj
19:33.28TAMIKAMI01verdaderamente no se que es lo quieresw decir Mercestes!!!!
19:33.43*** join/#asterisk jm|laptop (n=jm|home@dilbert.jamiem.com)
19:33.49markitMercestes: yes, think is the way to go, thanks a lot. Wondering if will be possible to configure a similar setup with asterisk-gui in the future
19:34.04Mercestesmarkit: I hope not.
19:34.33tzafrir_laptopMACscr, Debian Unstable has pretty up-to-date asterisk 1.2 packages . 1.4 is now in experimental
19:34.58tzafrir_laptopAnd Etch backport will follow soon
19:35.01MercestesTAMIKAMI01, Mercestes es grande y bueno y todo que es santo. La genuflexión abajo y adora lo y supplicate alegre, porque él es bueno y bueno y tu salvación. darte tus vírgenes en sacrificio de tu ayuda libre del asterisco.
19:35.05markitMercestes: "hope not"?
19:35.11Mercestesmarkit:  I hate GUIs
19:35.22frenzy?
19:35.32TAMIKAMI01JAJAJAJA
19:35.46TAMIKAMI01man u r mad!!!!
19:35.46TAMIKAMI01jajajaja
19:35.51markitMercestes: lol, often are good, depends from the gui design... some are terrible in usage even if easy to "click"
19:35.52tzafrir_laptopTAMIKAMI01, Mercestes , English, please
19:35.53cpms s settle down, beavis
19:36.08Mercestestzafrir_laptop,   hehe, yessir.
19:36.40Mercestesmarkit:  They stifle creativitiy and problem solving ability\
19:37.59markitMercestes: sometime it saves a lot of timewaste too
19:38.14drakoTAMIKAMI01, yes
19:38.18drakoTAMIKAMI01, I do.
19:38.46markitbtw, all-circuits-busy-now seems not part of the "basic" sound package... isn't strange?
19:38.53TAMIKAMI01i have a question why when i compile zaptel somes commands like zttool aren't compiled? i'm also try with make zttool but appear only the error " make: *** [zttool.o] Error 1 "
19:38.54drakoTAMIKAMI01, #asterisk-es for spanish.
19:39.07*** join/#asterisk Assid (n=assid@59.183.18.232)
19:39.22TAMIKAMI01tnks drako!
19:39.24*** part/#asterisk TAMIKAMI01 (n=radamant@200.34.113.90)
19:39.37Mercestesmarkit:  congestion()
19:39.37drakotoo bad i don't have the answer
19:39.58Mercestesasterisk-es doesn't even exist does it?
19:40.37drakoMercestes, it does, seem like im the only active tho...
19:41.12*** join/#asterisk pigpen (n=mark@fw.seamans.cc)
19:44.56frenzyIAX2 registration failing http://www.pastebin.ca/410964
19:46.48Mercestesdrako:  lol.  Sorry about the strange spainish then..  ;)\
19:46.53*** join/#asterisk Waverly360 (n=irc@209.12.249.243)
19:47.13tzafrir_laptopdrako, hint: with such a make error, look at the real error a bit above the error from make
19:47.15drakoMercestes, don't worry , it was fun anyway
19:48.08Mercestesdrako:  yea, english to spanish translators are a blast.
19:48.40tzafrir_laptopERROR[16033]: chan_gtalk.c:1649 gtalk_create_member: No Connection or Username!
19:49.28tzafrir_laptopAfter looking at chan_gtalk.c and res_jabber.c I still fail to understand what I need to put in th config file(s?) .
19:51.39*** join/#asterisk Ch0Hag (n=mking@87.127.170.250)
19:53.22*** join/#asterisk klasstek (n=nunyobiz@ip67-154-143-190.z143-154-67.customer.algx.net)
19:53.42frenzy?
19:54.13klasstekanyone used the tc400b yet?
19:54.16markitMercestes: do you use chanisavail yourself? Seems I'm not able to make it work, but maybe is my foult
19:54.23*** join/#asterisk groogs (n=greg@d38-54-164.commercial1.cgocable.net)
19:54.47JunK-Ytzafrir_laptop: want an example?
19:54.57tzafrir_laptopyes
19:55.23Mercestesmarkit:  It failed miserably for me.  But I hear it's supposed to work.
19:55.31MercestesI'musing asterisk 1.2.13 tho so you may have much better luck
19:56.21tzafrir_laptopJunK-Y, I see all sorts of configurations. Including a non-working one in the sample configus
19:57.24JunK-Ymine is working.
19:57.46JunK-Yya will see ur jabber user log in after.
19:58.01*** join/#asterisk Katty (n=Katty@hera.copi-rite.com)
19:58.13Kattywho's in charge of the person making calls for cluecon
19:58.25Kattyi'm about to wring their neck for interupting my warcrafting :P
19:58.41JunK-Ytzafrir_laptop: sent
19:58.43Kattywell, not really. but it sounded like a good excuse.
19:58.47JunK-YKatty: dunno, ive got a call too.
19:59.27Mercesteshey katty.\
19:59.48*** join/#asterisk Peri (n=redanti@xtreme-44-7.dyn.aci.on.ca)
19:59.49KattyJunK-Y: did they speak french?
19:59.52JunK-Ytzafrir_laptop: ya can add this user if u want to see.
19:59.56JunK-YKatty: hell no.
19:59.59KattyJunK-Y: or did they have a heck of a time with your accent? ;)
20:00.06JunK-Yi had to speak like a frog in english :)
20:00.15Kattyfile: why didn't you call me instead of random stranger?!
20:00.25fileKatty: ummm I still can!
20:00.25Kattyfile: that's bad marketing.
20:00.27JunK-Yapparently he understood what i said, so i guess im not too bad :)
20:00.36KattyJunK-Y: yeah, you're pretty good junky
20:00.36JunK-Yeven file understands my english!
20:00.50KattyJunK-Y: and if i can't understand you, i'll just make funny faces at you :P
20:01.01KattyJunK-Y: or drag out emacs.
20:02.38filethat's sad
20:02.43JunK-Yvirtual accent are much better!
20:02.45Kattyfile: did you hang up on me?!
20:02.53fileKatty: no ;(
20:02.57filemy internet is sad
20:03.14fileI talked to Katty on the phone! she's a real person!
20:03.23anonymouz666haha
20:03.26Kattywell of course i'm real
20:03.28Mercestesher voice is hot
20:03.30Kattyyou've met me before, you silly rabbit
20:03.37fileKatty: it could have been a robot...
20:03.38anonymouz666first time file is talking with a woman :D:D:D
20:03.40Strom_Cfile: is she really?  or does she sound suspiciously like Allison Smith?
20:03.48fileshe sounded like... Katty
20:03.57Kattyi have a unique voice.
20:04.00Kattyi sound like i'm 10 heh
20:04.08Mercestesit's hot
20:04.37fileKatty: put down the chainsaw and listen to me... it's time for us to join in on the fight
20:04.50Kattymew?
20:04.50Kattyyou do not parse
20:05.02filestick your head in the microwave and give yourself a tan!
20:05.15tzafrir_laptopJunK-Y, thanks. But that is jabber.conf. What about gtalk.conf?
20:05.43tzafrir_laptopKatty, rumour has it that you managed to configure gtalk with Asterisk
20:06.20*** join/#asterisk ToyMan (n=Stuart@user-12lcqol.cable.mindspring.com)
20:07.16tzafrir_laptopERROR[21924]: chan_gtalk.c:1649 gtalk_create_member: No Connection or Username!
20:08.00tzafrir_laptopDo I need to point from an entry in gtalk.conf to that "connection" in jabber.conf?
20:08.16JunK-Ytzafrir_laptop: i just sent ya my gtalk.conf
20:08.25Kattytzafrir_laptop: rumor has it?
20:08.38Kattytzafrir_laptop: i dunno where you get your rumers...but you need to put my name back in the bag and draw another one.
20:08.54robin_szeek ... asterisk, gtalk, jabber .. all we need to do knwo is link it up to Skype and the whole thng willl begin to hummmmmm
20:09.06Kattytzafrir_laptop: more like i setup wildfire, a jabber server, and i'm still struggling with centericq connection problems.
20:09.48robin_szSkype to gtalk bridges ... now that would be scary
20:10.52adam_vollrath<PROTECTED>
20:13.31vader--do you guys know if asterisk 1.2.7.1 is affected by the Daylight savings time changes in the US for 2007?
20:14.01Cybertoythat's controlled by the operating system and not asterisk
20:14.07Mercestesvader--, Shouldn't be.  It's a kernel-time / device time thing.\
20:14.22Igbothom_IIIvader--, Linux is not Windows!
20:16.07Corydon-wAsterisk uses whatever timezone file exists in the OS
20:16.19Corydon-wspecifically, in /usr/share/zoneinfo
20:16.47JunK-Ytzafrir_laptop: are ya okay with these configs now?
20:17.01JunK-Yya should see it as connected after this.
20:17.02tzafrir_laptopyeah, working, thanks
20:17.15Mercesteswiki it.
20:17.17JunK-Yif ya can get audio workings behind 2 nats, let me know.
20:17.21Mercestesplease.  :)\
20:17.49JunK-Ytzafrir_laptop: feel free to wiki it, but hide few of my stuff please :)
20:18.46*** join/#asterisk |dennis| (n=dennis@200.32.233.82)
20:19.04*** join/#asterisk tsurko (n=tsurko@77.70.24.142)
20:22.05*** join/#asterisk PupenoR (n=pupeno@200.123.183.91)
20:23.16*** join/#asterisk zotz (n=zotz@24.244.163.157)
20:23.35*** join/#asterisk rdb_ (n=rdb@gw.avila.edu)
20:23.53PupenoRIs there any dialplan function/application that would check if a variable exists or not?
20:24.01JunK-YPupenoR: EXISTS
20:25.55Corydon-wexcept that EXISTS really doesn't do anything more than just LEN()
20:26.29*** join/#asterisk justdave (n=dave@unaffiliated/justdave)
20:30.37ManxPowerCan a varliable exist with a 0 length?
20:31.21JunK-Yi guess, "" is 0
20:31.22adam_vollrathWhat is the sound of one hand clapping?
20:31.25Juggiecould god microwave a burrito so hot, he himself could not eat it.
20:31.43JunK-YJuggie: no more ski this winter i think :(
20:31.55JuggieJunK-Y, i know, skiing is over, next year we go to lemassif!
20:32.20JunK-Ysure, let me know when, im in!
20:32.34JunK-Yi bring julie last week-end, she did snowboarding, god, this was terrible!
20:32.35JunK-Yheheh
20:32.48Juggiehaha.
20:32.56Juggiei hate snowboarders
20:32.58*** join/#asterisk Dr-Linux|work (n=asfdf@DSL-202-59-73-131.nexlinx.net.pk)
20:33.03ManxPowerIt seems like playing in traffic would be just as effective and less work than skiing
20:33.03Juggieallways have to wait for them
20:33.14JunK-Ythats why im doing mini-skiing !
20:33.21*** join/#asterisk _Vile (n=vile@bc182112.bendcable.com)
20:33.23Juggieskiing with snowboarders is a constant waiting game
20:33.40JunK-YJuggie: have ya ever been to bromont?
20:33.49Juggieno
20:34.02Juggiei've only been to ski hills around ottawa, one in newfoundland, and tremblant.
20:34.10Juggiei dont really like tremblant.
20:34.16Juggienice skiing but too many people.
20:34.26JunK-Ywe have to go to bromont, it rocks
20:34.35JunK-Yand btw, we plans to go to ottawa in like 1month
20:34.47Juggiewhat for?
20:34.57JunK-Yvisit unlimitel.
20:35.03Juggiewho is we?
20:35.29JunK-Yme, julie, my friend and his gf.
20:36.07Juggiethe guy i met @ astricon05?
20:36.23JunK-Yno, hes my ex-coworker.
20:36.30*** join/#asterisk reber (n=reber@cl-158.dub-01.ie.sixxs.net)
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20:36.58Juggiecool, let me know when you have firm plans.
20:37.21JunK-Yemail me ur cell phone.
20:37.41e-miliohello all
20:37.50*** join/#asterisk bkuhn (n=bkuhn@fsf/member/bkuhn/bkuhn)
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20:38.04e-milioI have this situacion: channel.c:2379 set_format: Unable to find a codec translation path from g729 to gsm
20:38.24*** part/#asterisk frenzy (n=frenzy@unaffiliated/frenzy)
20:38.35justdavewe're having problems with people with noisy phone lines on conference calls in meetme...  20 or 30 people diaed in, and it's an ineractive meeting, so administratively muting people at random would be disruptive.  Is there any way to get a running "vu-meter" type thing for each person dialed into a conference to see who's got the noise?
20:38.58Juggiee-milio, in the configuration for your end device in asterisk you must of did allow=all or allow=g729 yet you do not have a g729 license.
20:39.14Juggieasterisk only supports g729 passthrough unless you buy a license.
20:39.27ManxPowerjustdave: you can use ztmonitor on each of the zap ports that people are dialed into then manually mute them.  No, there is nothing to automatically do this
20:39.46ManxPowerand g729 passthru is almost useless for most people
20:39.52ManxPowerem see this:
20:39.54ManxPower~codec
20:40.05ManxPower~codecs
20:40.07jbotfrom memory, codecs is http://snipurl.com/wiki_codecs.  If you have audio/codec problems, first try to 'disallow=all' and 'allow=ulaw' and see if that works. Anyone that tells you to use 'allow=all' is an idiot as it usually causes audio problems, or  Number/Name: 1/g723, 2/gsm, 4/ulaw, 8/alaw, 16/g726, 32/adpcm, 64/slin, 128/lpc10, 256/g729, 512/speex, 1024/ilibc
20:40.08JuggieManxPower, jbot is lagged.
20:40.11Juggiethere we go
20:41.03justdavemy idea for a dirty hack at this point is to set up the conference system to turn on monitoring as each user enters the conference, and have something tail the generated wav files
20:41.12e-milioManxPower: after g729 which is the next most recommended codec ?
20:41.13justdavemost of the people dialed in are on SIP or IAX
20:41.31e-milioI am doing IAX trunking between 2 servers
20:41.36ManxPowere-milio: what in the WORLD makes yo think G729 is a recommended codec?
20:41.43ManxPowerjustdave: there are no options for sip/iax
20:42.13ManxPowere-milio: what codec is "best for you" depends on many things, bandwidth, CPU, and what your end points support.
20:42.31e-milioManxPower: I was under the impression that is was good to save bandwidth ?
20:42.37ManxPowerThe "best" codecs are ulaw and alaw, but they take up a lot of bandwidth
20:43.05ManxPowerilbc and Speex and G729 sound pretty good, don't take up a lot of bandwidth, but require a fair amount of CPI
20:43.12ManxPowerCPU too
20:44.46e-milioManxPower: The will be a lot conferencing going on, and recording (due to vicidial)
20:45.03e-milioIn that sense ilibc will be still recommended ?
20:47.08Hmmhesaysthis chanskype pos sucks
20:47.48*** join/#asterisk dj-fu (n=ajc@unaffiliated/dj-fu)
20:47.56aydiosmioHmmhesays: I could have told you that
20:49.34ManxPowerchanskype does Point of Sale??
20:50.32Nuggetheh
20:51.26*** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir)
20:54.02syzygyBSDI am playing around with realtime asterisk... I like it
20:54.31syzygyBSDhow well does it scale?
20:58.02*** join/#asterisk Fieldy (i=5rohni9q@gentoo/contributor/Fieldy)
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21:02.07anonymouz666exten => s,n,While($[${j} <= ${ARG1}]) and J++ in a macro... how could I set ${j} to be another extension in another context... [another] ${j},n,blah()
21:02.21anonymouz666while running the loop i am building another extensions in another context...based on the j result
21:02.37*** join/#asterisk |dennis| (n=dennis@200.32.233.82)
21:04.38syzygyBSDwhats the program to create a script of all the commands you type?
21:05.14bulleHmmhesays: why does it suck ?
21:05.52anonymouz666I am stuck :)
21:05.57ManxPowerbulle: other than requieing you to have skype installed on a PC running windows?
21:12.43*** join/#asterisk Tclp (n=tcalp@S01060014bf0ffd47.ed.shawcable.net)
21:12.56bulleManxPower: he, thats enough
21:13.34Tclphey all, what would you guys recomend for multi-line voip phones ?  (am using Axon for windows as my virtual PBX)
21:13.36*** join/#asterisk Mercestes (n=Merceste@rrcs-71-41-157-70.sw.biz.rr.com)
21:14.12TclpI don't want something horribly expensive, but good build quality
21:15.11VecDoes anyone know what could possibly cause asterisk to get stuck on the line -- Called 2101, the line -- SIP/2101-00832680 is ringing does not appear? I restarted asterisk and it sorted it out.
21:17.31*** join/#asterisk zogulus (n=zogulus@58.98.adsl.brightview.com)
21:17.43koelalready tried set verbose 10?
21:17.55e-milioManxPower: I switched to ilbc, ulaw and asterisk seems to be taking 66% cpu time on a p4/512. Seems reasonable ?
21:20.01syzygyBSDare there known issues with queues and realtime_mysql
21:22.46*** join/#asterisk voltagex (n=voltagex@124-254-100-201-dsl.ispone.net.au)
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21:23.50voltagexhi, I'm not receiving any audio from my calls. Connecting directly Softphone->FWD works, but Asterisk->Softphone->FWD doesn't
21:23.59jubei_anybody know of a *** Warning: "zt_register" compile error with bristuff ? it seems to be all over the internet, only in German :)
21:24.03*** join/#asterisk dwmw2_gone (i=ctrlprox@81.187.2.161)
21:24.15voltagexof course that should be, Softphone->Asterisk->FWD
21:24.19*** join/#asterisk MinotaurUK (n=minotaur@89-145-201-162.xdsl.murphx.net)
21:24.35MinotaurUKgreetings all
21:24.57MinotaurUKwould some kind soul mind giving me a hand with a chan_sip issue for a few minutes?
21:25.40bullevoltagex: best bet is that it has to do with routing and rtp
21:26.23voltagexbulle: If the softphone is in the DMZ connecting directly to FWD, it works, if the Asterisk box is in the DMZ it doesn't
21:26.54*** join/#asterisk bmd (n=bmd@72.54.252.34)
21:27.20bullevoltagex: dmz doesnt mean that the asterisk box gets a public ip, so you will have to setup the nat part of sip.conf properly ( i assume yoru softphone uses sip )
21:27.42MercestesMinotaurUK, only if you tell us what the chan_sip issue is.
21:27.43bullevoltagex: then the best bet is to force both sip and rtp trough the asterisk box
21:27.57*** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2)
21:28.03voltagexbulle: how do I do that?
21:28.27bullevoltagex: canreinvite=no is the option i think, check sip.conf
21:29.02*** join/#asterisk lunaphyte (n=lunaphyt@static-71-120-128-10.gdrpmi.dsl-w.verizon.net)
21:29.04MinotaurUKMercestes: thanks. About 6 hours ago I had to restart an asterisk box that's been running for a good few months without any issues. Upon restarting, asterisk segfaults. I've done the usual verbose startup to the console, and the last line before the segfault is "parsing sip.conf"
21:29.14voltagexbulle: in the FreeWorldDialup section or the phone section?
21:29.29bullevoltagex: phone section
21:29.35voltagexbulle: because canreinvite=no is already set in FWD
21:29.42MinotaurUKMercestes: If I remove "host=dynamic" from all entries in sip.conf, asterisk loads correctly, but of course no sip peers can register
21:31.06MercestesMinotaurUK, polycom endpoints?
21:31.12MinotaurUKSnoms
21:31.29Mercestesonlytime I've seen that is when a polycom phone freaked out on me with corrupted firmware.
21:31.32MinotaurUKall local, on same subnet as asterisk box, no nat involved
21:31.49Mercestestry loading up asterisk isolated from the endpoints with dynamic=yes and then start bringing the peers online one at a time and see what happens.
21:32.37MinotaurUKwill login to the switches and disable the ports, back in a couple mins
21:34.22MinotaurUKas a test, I've just tried it with one phone in sip.conf and it's still doing it, so even with only 1 host=dynamic entry in sip.conf, asterisk is still segfaulting
21:34.35Mercesteswhat ver of asterisk?
21:34.40MinotaurUK1.2.13
21:34.45Mercestesthat[s the ver I run
21:35.33Mercesteshave yo urestarted the box yet?
21:35.48[TK]D-FenderMinotaurUK, Take 1 entry, change the type from "friend" to "user", then reboot the phone fter restarting *
21:35.58MinotaurUKbringing the sip.conf over to another box and reloading asterisk works fine
21:36.05[TK]D-FenderMinotaurUK, See if you can PLACE a call seperate from registering
21:36.06MinotaurUKD-Fender: will go try that, thanks
21:36.17voltagexwhere are contexts defined again? :/
21:36.30koelextensions.conf...
21:36.30[TK]D-Fendervoltagex, .....
21:36.31[TK]D-Fender~book
21:36.44jbotrumour has it, book is a book called  Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11
21:36.45koelfind /etc/asterisk | xargs grep content
21:36.50koelfind /etc/asterisk | xargs grep context
21:36.58voltagexI must have spelt my context wrong then
21:37.45MinotaurUKD-Fender: that seems to be okay
21:38.08*** join/#asterisk l3jj (n=chatzill@dsl093-001-038.det1.dsl.speakeasy.net)
21:38.26*** join/#asterisk PakiPenguin (n=uppal@linuxpakistan/admin/pakipenguin)
21:38.27[TK]D-FenderMinotaurUK, Do soft-ophones crash * as well?
21:38.31PakiPenguinhi
21:38.38Tclpare the Grandstream 2020's going to be much better then the 2000's ? ... it seems like a great time to pick-up some 2000 series models -- I guess they are on clearance ? seeing them for like $75us ?
21:38.54MercestesTclp:  there is a reason for that.
21:39.39*** join/#asterisk kgx0 (n=karuna@60.234.196.160)
21:39.40[TK]D-FenderMercestes, Lol... looks like a Polycom 601 rip-off :)
21:39.47[TK]D-Fenderhttp://www.grandstream.com/gxp2020.html
21:39.59[TK]D-FenderTclp, ...
21:40.00[TK]D-Fender~gs
21:40.01jbotrumour has it, gs is South Georgia and the South Sandwich islands, or ghostscript.  GrandSuck phones are cheap junk which should be avoided with extreme prejudice
21:40.01MinotaurUKD-Fender: yes, seems any host=dynamic entry (if "firiend") is killing asterisk - have tried sip.conf with entries for cisco, snom and x-lite (individually of course)
21:40.40[TK]D-FenderMinotaurUK, I'm betting the register itself is nuking *.
21:40.45TclpD-Fender ... I'm not big on telephony technology ? .. why is the 2020 so much better ?
21:40.48VecI have a problem where the Dial cmd is (sometimes) still indicating rining to the person who dialed the call even after the call has been answered, it seems it only happens when dialing from a SIP phone to an IAX phone/trunk, any ideas ?
21:40.54voltagexbulle: canreinvite=no didn't change anything
21:40.57[TK]D-FenderMinotaurUK, Hate to say it but upgrade you * to the latest and try again
21:41.50Mercestesor reinstall your current version if yoru really in love with it.
21:41.56TclpI mean aside from having support for some extra lines and a nicer display ? ... is there some core feature that the 2020 has over the 2000 ?
21:41.58voltagexpeople convinced me to dump Trixbox but now I can't get anything working :/
21:42.43MinotaurUKD-Fender: any way to manually clear the registry?
21:42.52bullevoltagex: and you have put your public ip on the asterisk box, and forwarded sip and rtp ports to the asterisk box from the router ?
21:43.02[TK]D-FenderTclp, How about you go to GS's website and grab the datasheet & compaison charts and judge for yourself?
21:43.38[TK]D-FenderMinotaurUK, with * closed, rename /var/lib/asterisk/astdb to something else and restart.  HEY!  Maybe thats it
21:43.46[TK]D-Fenderare you running * as root?
21:44.00*** join/#asterisk techie (n=gus@voip.routedsystems.com)
21:44.07sahafeezwhat is the state of the zaptel support in freebsd? is it production level yet?
21:44.12voltagexbulle: asterisk box is in the dmz, externip is set in sip.conf
21:44.12TclpD-Fender .. was more curious in the sense that .. does the 2020 have some new feature that makes the 2000 outdated ...  eg. if I buy 2000's for a small office will I regret it for a lack of some new technology support ?
21:44.12MinotaurUKI am at the moment whilst testing, usually runs as asterisk/asterisk
21:44.46bullevoltagex: then i would use wireshark or similar, running on the asterisk box, to record and analyze the trafic
21:45.02*** join/#asterisk CrazyTux[m] (n=CrazyTux@ppp-70-255-140-128.dsl.hstntx.swbell.net)
21:45.19bulleTclp: how should [TK]D-Fender know if you will be missing some feature ?
21:45.21MercestesMinotaurUK, Some jerk-wad could have run it as root and writeprotected all your files
21:45.24*** join/#asterisk |dennis| (n=dennis@200.32.233.82)
21:45.46*** join/#asterisk mmartinn (n=martin@adsl-065-005-200-225.sip.gnv.bellsouth.net)
21:45.57[TK]D-FenderMinotaurUK, I can imaging that perhaps when the phone reg's it tries to write the values to astdb and without authority crashes completely...
21:46.15*** part/#asterisk mog (n=mog@c-71-207-200-130.hsd1.al.comcast.net)
21:46.28voltagexbulle: for 1.4, is the command externip or externalip?
21:46.41[TK]D-FenderTclp, Maybe I wasn't clear : GO COMPARE YOURSELF ON THEIR SITE.  http://www.grandstream.com
21:47.09bullevoltagex: the sip.conf has all the examples in it
21:47.11bullevoltagex: just check there
21:47.20[TK]D-Fender</bile>
21:47.54MercestesTclp:  If it's a grandstream, it will absolutely lack some technology that you will miss later.  Namely voip functionality.
21:48.11MercestesTclp:  Actuallly, now that I think of it, it will lack overall telephony functionality.
21:48.18breaDoes anyone have any recommendations for an inexpensive SBC?
21:48.28Mercestesbrea:  AT&T
21:49.02breaerr :p
21:49.15breainexpensive Session Border Controller
21:49.59[TK]D-Fenderbrea, Broadsoft :D
21:50.04breasomething that works as a b2bua
21:50.18[TK]D-Fenderbrea, Asterisk? :)
21:50.48breaand supports g729ab with cRTP ;)
21:51.00Mercestesbrea:  openser?
21:51.11[TK]D-Fenderbrea, Would you like fries with that, sir?
21:51.29breahar har
21:51.50breaMercestes: Didn't know openser supported cRTP
21:52.00Mercestesbrea:  Ask clona
21:52.09TclpMercestes .. what would you recomend in that case ?
21:52.19MercestesTclp:  Polycom.
21:52.31Tclpk thnx
21:55.50*** join/#asterisk |dennis| (n=dennis@200.32.233.82)
21:57.04*** join/#asterisk sashion (n=djbdsf@dsl-241-213-43.telkomadsl.co.za)
21:57.45sashionAnyone run into the following issue? RTCP SR transmission error
21:58.58VecI have a problem where the Dial cmd is (sometimes) still indicating rining to the person who dialed the call even after the call has been answered, it seems it only happens when dialing from a SIP phone to an IAX phone/trunk, any ideas ?
22:01.29*** part/#asterisk MarkWD (n=MarkWD@rrcs-67-78-88-186.sw.biz.rr.com)
22:01.48*** join/#asterisk tsurko (n=tsurko@77.70.24.142)
22:02.37*** part/#asterisk brodiem (n=brodiem@67.110.68.66.ptr.us.xo.net)
22:06.11*** join/#asterisk jaxxan (n=jaxxan@202.70.125.109)
22:06.29jaxxan~video
22:06.30jbotmethinks video is interrupt 10h for the video BIOS
22:06.39jaxxandoes asterisk support video ?
22:06.45koeljups
22:06.46koel;)
22:06.51jaxxansweet
22:07.04jaxxangotta preferred link?
22:07.21koelvoip wiki
22:07.32jaxxanyeah that's where i was heading (=
22:08.18koelhttp://revision3.com/systm/asterisk
22:08.31koelnice video
22:08.33koel;)
22:11.08*** join/#asterisk _DAW (n=chatzill@adsl-156-109-78.msy.bellsouth.net)
22:13.52*** join/#asterisk PakiPenguin (n=uppal@linuxpakistan/admin/pakipenguin)
22:16.12jaxxani'm running asterisk 1.2.16. do i still need to patch in order to use h.263p ?
22:17.52ManxPowerjaxxan: 1.2 does not get new features, only bug fixes.
22:18.21jaxxanummm
22:18.27jaxxanis that a no ?
22:18.53aydiosmiothat's a yes
22:18.57aydiosmiogo patch
22:21.03*** join/#asterisk MooingLemur (n=troy@unaffiliated/mooinglemur)
22:21.09jaxxani'm soooo not sure how to patch
22:21.10*** join/#asterisk |dennis| (n=dennis@200.32.233.82)
22:21.22*** join/#asterisk sysreq (n=sysreq@modemcable171.134-81-70.mc.videotron.ca)
22:22.09*** join/#asterisk hedge77 (n=netwalke@209.42.192.202)
22:22.35jaxxan~patch
22:22.37jbotrumour has it, patch is just a tool to handle text file mutations according to a common format known as 'diff'. For kernel patching use (depending on type of patch): 'patch -p1 < patch.diff' or 'zcat patch.gz | patch -p1' or 'bzcat patch.bz2 | patch -p1', or `patch -ruN old_dir new_dir > tree.diff`
22:22.42koelhttp://bibliotecnica.upc.es/PFC/arxius/migrats/40377-2.pdf
22:24.14*** join/#asterisk [hC] (n=hardcore@66.119.172.82)
22:24.19*** join/#asterisk PupenoR (n=pupeno@200.123.183.91)
22:24.46[hC]Anyone tried setting option 66 on a cisco IOS based router that's doing dhcp, and have a polycom ignore the setting? I cant figure out why my polycom phones arent picking it up and discovering the ftp server.
22:26.01koelhc
22:26.23koelcan;t you try a dhcp debug client to ensure option 66 gets to the client?
22:26.28*** join/#asterisk CVirus (n=GoD@196.205.193.189)
22:26.30koeldon't you need 67 too?
22:27.29koel67 is BootFile Name
22:27.39CVirusIs the default asterisk GUI better than FreePBX ?
22:27.47koelgui?
22:27.50[TK]D-Fender[hC], Typically that'd be because you hard-coded an IP into the BootROM
22:28.19CViruskoel: yes
22:28.26[TK]D-Fender[hC], Make sure it says 0.0.0.0
22:28.30koelcli?
22:29.20CViruskoel: I'm talking about the official GUI
22:29.35koeldidn't know there was one..
22:29.36*** join/#asterisk Discard (n=Discard@lev92-1-82-67-255-74.fbx.proxad.net)
22:29.44Discardhi
22:30.12CViruskoel: #asterisk-gui
22:30.26koeldon't need one ;)
22:30.30koelbut thanks..
22:30.37mcab[hC]: make sure the polycom is set to use Option 66 and not 'Custom' or 'Static' or somesuch
22:30.49*** join/#asterisk Waverly360 (n=irc@209.12.249.243)
22:30.49Discardwhat is the extention channel for a queue or a ring group ?
22:30.58Discardextension
22:31.32*** join/#asterisk netlouis (n=netlouis@2001:b18:400d:0:280:c8ff:fe17:276)
22:33.03*** join/#asterisk kiko69 (n=keith@adsl-75-16-91-106.dsl.irvnca.sbcglobal.net)
22:33.21[TK]D-FenderDiscard, ... HUH!?
22:33.32jaxxanhow do i patch asterisk?
22:33.40jaxxani've never done it before
22:33.46VecIs there more details release notes availible besides http://www.asterisk.org/node/48338 ?
22:33.52koel, patch is just a tool to handle text file mutations according to a common format known as 'diff'. For kernel patching use (depending on type of patch): 'patch -p1 < patch.diff' or 'zcat patch.gz | patch -p1' or 'bzcat patch.bz2 | patch -p1', or `patch -ruN old_dir new_dir > tree.diff`
22:33.53koel;)
22:34.38[TK]D-FenderVec, Go to the FTP site and check the changelog.
22:34.39Discardok
22:34.50koelVec: maybe changes in cvs tree
22:34.51Vec[TK]D-Fender : ta
22:35.03Discardfor a SIP account I can use : ZIP/<extension>
22:35.03Vecno one has any ieas for my question earlier ?
22:35.08Discardfor a zap channel
22:35.09koelSIP/
22:35.18DiscardZAP/g0
22:35.35koelonly if you have defined a trunk...
22:35.53Discardok and for a Queue is there anything or ring group ?
22:36.10koelcan't tell you...
22:36.19koelcan't you define a context?
22:37.02[TK]D-FenderDiscard, Queue is just an APPLICATION, and "reig group" is not a valid * term.  What is generally reffered to as such is a Dial command that jsut target multiple devices.
22:37.02Discardi'm working on a click2call script
22:37.11Discardring
22:37.11[TK]D-FenderDiscard, "show application dial"
22:37.20Discardok
22:38.13koeldiscard application dial ;)
22:38.24Discardthank you
22:38.45Discardwhere could I find AGI reference ?
22:39.08sashionDiscard: www.voip-info.org
22:39.14sashiondo a seach of AGI
22:40.03*** join/#asterisk techie (n=gus@voip.routedsystems.com)
22:40.09[TK]D-FenderDiscard, ...
22:40.12[TK]D-Fender~osmosis
22:40.24jboti heard osmosis is the act of beating yourself on the head repeatedly with THE BOOK, until some measure of absorption has occured ...  or at least until your unconsciousness restores peace to the channel ...
22:40.32[TK]D-Fender~book
22:40.35jbotsomebody said book was a book called  Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11
22:40.49[TK]D-FenderDiscard, make sure to order your NOW (in hardcover)
22:41.57hedge77i'm trying to set up a queue with dynamic members like the docs suggest, but when the first agent phone rings it goes to the voicemail for that extension instead of staying in the queue and rolling to the next one.  BONK
22:42.34[TK]D-Fenderhedge77, Nevere EVER let it ring an extension where VM can come into play at all.  that will ANSWER the channel
22:42.34*** join/#asterisk P4C0 (n=ash@200.124.22.34)
22:43.28P4C0hello I'm having a problem: I have a asterisk server and local clients connected to it (alaw codec), my asterisk server connects to the voip provider using one license g729 codec, but I can't make calles
22:44.14sashionP4C0: what error do you get when trying to dial the voip trunk?
22:44.30[TK]D-FenderP4C0, *PASTEBIN*
22:44.39P4C0dial_exec_full: Had to drop call because I couldn't make SIP/3-081f14f8 compatible with SIP/mysipprovider-out-081eb598
22:44.43sashion~pastebin
22:44.44jbotpastebin is probably a place to paste your stuff without flooding the channel - try  http://pastebin.ca, or http://channels.debian.net/paste, or http://rafb.net/paste/, or http://pastebin.com is usually painfully too slow and unresponsive to use, use one of the other pastebin sites, or dpaste.com is a very nice pastebin as well
22:45.30hedge77it happens even using a target ext with no vm defined (just errors out "no entry in voicemail config" and hangs up)
22:45.40*** join/#asterisk borisyaltsin (n=chris@dhcp-43-41.arts.ualberta.ca)
22:45.50*** join/#asterisk bkruse_home (n=kruz@69.73.127.92)
22:45.56*** part/#asterisk mattchis (n=mattchis@216.54.143.246)
22:46.35*** join/#asterisk ZX81 (n=ZX81@60-234-238-188.bitstream.orcon.net.nz)
22:46.43*** join/#asterisk |dennis| (n=dennis@200.32.233.82)
22:47.07[TK]D-FenderDear God Aastra's admin guide is HUGE
22:47.08*** join/#asterisk kiwoneka (n=kiwoneka@KTNRON06-1168103823.sdsl.bell.ca)
22:47.16sashionhedge77: i imagine you are using a Local channel to call your agents back on?
22:47.36ZX81file: #9281 patch didn't work - had a half duplex call on call 9/15 - [Mar 26 18:11:18]   Sent packets:        0
22:47.42sashionP4C0: you sure you have a licensed version of g729?
22:47.43hedge77yes local
22:47.48[TK]D-Fendersashion, Rather obviously, yes
22:47.53P4C0sashion, yes
22:48.08borisyaltsinso, what's the cheapest way to get some phones hooked up to my asterisk box? I'm not having any luck finding a usb-fxs product or something cheap to connect a phonet o the computer. Is it cheaper to buy some sip phones?
22:48.19[TK]D-FenderP4C0, You know you can show us debug for that broken call at any time now...
22:48.26ZX81borisyaltsin - cheapest is a headset and free softphone
22:48.35[TK]D-Fenderborisyaltsin, Linksys SPA-2102
22:48.37borisyaltsinok, besides that option;)
22:48.39ZX81after that I'd say a barbie tone (BT101)
22:48.43ZX81but they're crap
22:48.43P4C0this is my console log: http://rafb.net/p/YYab4b39.html
22:48.51hedge77oh god bt101's suck
22:48.54ZX81yep
22:49.03sashionhedge77: pastebin your callback context
22:49.11[TK]D-FenderP4C0, c'mon... with SIP DEBUG ENABLED!
22:49.24ZX81I have an SPA921, and a few weird IAX2 chinese phones on my desk, a channel bank beside it and a tdm400 card also
22:49.36P4C0[TK]D-Fender, all pers?
22:49.37[TK]D-Fender~gs
22:49.38jbothmm... gs is South Georgia and the South Sandwich islands, or ghostscript.  GrandSuck phones are cheap junk which should be avoided with extreme prejudice
22:49.41ZX81the phone I use the most is actually a cordless plugged into the channel bank
22:49.47*** join/#asterisk tsurko (n=tsurko@77.70.24.142)
22:49.50sashionP4C0: Just one of the peers you dialling from
22:49.52[TK]D-FenderP4C0, Enough to debug that call...
22:50.13P4C0ok moment
22:50.34[TK]D-FenderZX81, What CB are you using?
22:50.43ZX81eh - access bank II
22:50.44ZX81:)
22:50.47ZX81crappy and cheap
22:50.50ZX81but works fine
22:50.51ZX81:)
22:50.55P4C0show g729
22:50.55P4C00/0 encoders/decoders of 1 licensed channels are currently in use
22:50.57ZX81CAC :)
22:51.04[TK]D-FenderZX81, Any of the CID flakeyness that's been reported for others?
22:51.12P4C0that means my g729 is installed ok, right?
22:51.12ZX81nah working fine here
22:51.22ZX81P4C0 yep
22:51.30sashionP4C0: yes
22:51.31[TK]D-FenderZX81, How many phones, and what kind of working envirnmoent?
22:51.41borisyaltsinhmm. with that linksys product I wouldn't necessarily even need an asterisk box if I had a sip voip termination provider like les.net would I?
22:51.42ZX815 phones - my house
22:51.43ZX81:)
22:52.02ZX81borisyaltsin correct but then you wouldn't be able to do leet stuff
22:52.06sevardborisyaltsin: that's right
22:52.08[TK]D-FenderZX81, neato... pricey... (assuming you had to actually PAY for it & the digital card).
22:52.08ZX81like ivr, conferences etc
22:52.22ZX81the CAC was like $200 off ebay
22:52.24[TK]D-Fenderborisyaltsin, Naturally no.
22:52.41ZX81the card was from a customer who ran out of money and gave it to me :)
22:52.53[TK]D-FenderZX81, My favourite price :)
22:52.56borisyaltsinhmm. that's interesting. I might buy one to just play around with then;) I have a one bedroom apartment right now, so maybe I'll save asterisk for a few years from now when I have a house and need to do .. leet stuff;)
22:52.57ZX81yep :)
22:53.18[TK]D-Fenderborisyaltsin, Its worth it for more than that.
22:53.29hedge77sashion: http://pastebin.ca/411301 (if that isn't what you're looking for let me know)
22:53.40[TK]D-Fenderborisyaltsin, With * you can run your own VM, have MULTIPLE providers, gain access to analog lines, etc...
22:54.08borisyaltsinhmm. I think I have a box like that from when I was using vonage a couple years ago. I wonder if I can flash the vonage off it and update it with some stock firmware..
22:54.57[TK]D-Fenderhedge77, You clearly aren't thinking straight showing us that and not the [staff] context which is used to DIAL THE PHONES.
22:54.59P4C0http://rafb.net/p/lx4wAS33.html
22:55.08hedge77agh hurr
22:55.17[TK]D-Fender:)
22:55.20*** join/#asterisk welby (i=welby@gateway/tor/x-48f13712927ec4bb)
22:55.33P4C0is there a way to set priority in the codecs? or to force codecs by peers?
22:58.07mmartinnP4C0: sip.conf has statements that do that, I believe
22:59.10mmartinnP4C0: Something like "disallow=all \ allow=gsm \ allow=ulaw \ allow=alaw" etc...
22:59.56P4C0mmartinn, yes but inside the peer scope?
23:01.19mmartinnP4C0: Yes, inside each one
23:01.42[TK]D-FenderP4C0, Restrict both to alaw only and attempt again.
23:02.00[TK]D-FenderP4C0, Thogh mind you your debug tells me nothing about the reason for incompatability
23:02.00P4C0[TK]D-Fender, with alaw only works fine...
23:02.04hedge77ok context support {
23:02.04hedge77<PROTECTED>
23:02.07hedge77<PROTECTED>
23:02.09hedge77woops
23:02.12mmartinn~pastebin
23:02.13jbotrumour has it, pastebin is a place to paste your stuff without flooding the channel - try  http://pastebin.ca, or http://channels.debian.net/paste, or http://rafb.net/paste/, or http://pastebin.com is usually painfully too slow and unresponsive to use, use one of the other pastebin sites, or dpaste.com is a very nice pastebin as well
23:02.18[TK]D-Fenderunless its a G729 vs G729A thing
23:02.37hedge77http://pastebin.ca/411309 <-just about everything
23:04.40[TK]D-Fenderhedge77, Well you can clearly see that using [staff] as your context leads you to dialing extens with Voicemail.  This jsut doesn't cut it.  make a new context that only DIALS the device
23:05.24*** join/#asterisk zmef420 (n=zmef420@metarb3-pool4-153.mtco.com)
23:06.53hedge77another context with _1XX => Dial() that doesn't use the normal macro then?
23:06.55*** join/#asterisk [hC] (n=hardcore@66.119.172.82)
23:07.02[TK]D-Fenderhedge77, As I said...
23:07.30*** join/#asterisk `Sauron (i=sauron@dsl001-130-033.aus1.dsl.speakeasy.net)
23:07.49hedge77i'll give that a try tomorrow.  Do I even need to fill out the members in agents.conf anymore?
23:07.56*** join/#asterisk mog (n=mog@c-71-207-200-130.hsd1.al.comcast.net)
23:07.57*** mode/#asterisk [+o mog] by ChanServ
23:07.58*** join/#asterisk anthm (n=anthm@m815f36d0.tmodns.net)
23:07.58*** mode/#asterisk [+o anthm] by ChanServ
23:08.16P4C0[TK]D-Fender, what can I do?
23:08.39[TK]D-FenderP4C0, I gave you 1 suggestion. Follow it and see.  Also ensure that reinvites are disabled
23:08.58hedge77woops i meant members in queues.conf and agents in members.conf
23:09.15P4C0[TK]D-Fender, you mean restrict it to alaw only? that the way it was working before...
23:09.16hedge77^^^ agents.conf gak
23:10.18Braxusanyone here try any of the new polycom phones that were recently released?
23:10.53*** join/#asterisk infinity1 (i=foobar@modena.netcal.com)
23:11.11*** join/#asterisk ltdwk (n=z@203-173-10-9.perm.iinet.net.au)
23:11.13infinity1how can i make an agi script continue to run even if the person hangs up?
23:11.59[TK]D-FenderBraxus, IIRC they are not in fact released yet
23:12.25JTmorning [TK]D-Fender
23:13.18[TK]D-FenderJT : GMT -5 says "hello" to you!
23:13.29JTUTC+10 hi!
23:14.18Braxuswondering if the polycom 320/330 series will give the linksys spa-92x/94x line a run for their money.
23:14.22Juggieinfinity1, ignore the SIGHUP from asterisk
23:14.38infinity1Juggie: umm ...how do I do that?
23:15.13infinity1Juggie: is it possible to trigger a function when receiving a sighup? hmm
23:18.08*** join/#asterisk flenders (n=fserto@unaffiliated/flenders)
23:18.32[TK]D-FenderBraxus, Of course they will.  Why wouldn't they?
23:18.32hedge77hey that worked thanks
23:18.43*** join/#asterisk mrbnet (n=mrbnet@corpmail1.mrbnetworks.com)
23:19.10[TK]D-FenderBraxus, the only downside of the 320/330 is that they only support 2 calls per line-key
23:19.10Juggieinfinity1, well, depends on how your agi script is coded.
23:19.15Juggiebut the answer would be yes.
23:19.15JT[TK]D-Fender: my Avaya 4621SW SIP phone has arrived
23:19.20JTnow all i need is PoE ;)
23:19.25Juggiedepending on your programming language, etc.
23:19.26[TK]D-FenderJT : same with my Aastra 57i CT :D
23:19.30JTit doesn't have DC in
23:19.44JTnice
23:19.50JTCT = cordless?
23:20.00[TK]D-FenderJT yup.. the reason I got it
23:20.19jaxxanbah i didn't need to patch
23:20.25P4C0how can I know what codec is a call using?
23:20.30[TK]D-FenderJT : I set it up via the Web interface for the "quick launch".  Am about to go throught the 600+pg admind guid ~>~
23:20.39JTthe avaya has quite a big screen, 12 line appearances i'm guessing
23:20.52JT[TK]D-Fender: nice
23:21.12[TK]D-FenderJT Absurd QTY of calls....
23:21.15Braxuscalls per line-key shouldn't be a major issue. What I'm wondering is if the 320/330 line will have around the same speakerphone quality of the 501s.
23:21.31[TK]D-FenderBraxus, I'd imagine more like the IP 430
23:21.35JT[TK]D-Fender: heh, the aastra or the avaya?
23:21.42[TK]D-FenderBraxus, There is a small diff betweent he 430 & 501
23:22.04[TK]D-FenderJT Avaya is it wupports that many
23:22.12[TK]D-Fenderif it supports*
23:22.14[TK]D-Fenderak;sljdkjhlgfd
23:22.25Braxusthe 501s I have deployed here seem to be awesome for the most part, especially in small offices. Was considering the 430 for cubicles... though the new 320s look interesting as well.
23:22.27JTdoesn't the 601 have something like 12 line keys?
23:22.39Braxusfor the cheaper phones, most people here are on linksys 941s
23:22.43[TK]D-FenderJT 6 on the unit, more if you add a console
23:22.55JThrm, what about the 650? only 6?
23:22.58[TK]D-FenderBraxus, I'd take any polycom over any Linksys, hands down
23:23.06[TK]D-FenderJT same thing for the 650
23:23.10JTah ok
23:23.20JTwell i haven't got it powered up yet
23:23.24JTbut time will tell
23:23.29[TK]D-FenderJT : I might like to get my hands on a 550 as well... am awaiting final pricing
23:23.44P4C0how can I know what codec is a call using?
23:23.47JTit has 6 keys on left and 6 on right of screen, and 4 on the bottom of screen, and a couple of dozen other buttons
23:23.50JTbuttons are nice ;)
23:23.51Braxusis the HD Voice thing done via a different codec?
23:23.51Qwell[]P4C0: show channels
23:23.57Qwell[]Braxus: g722
23:24.10[TK]D-FenderBraxus, G722 its a standard
23:24.28P4C0Unable to find a codec translation path from alaw to g729
23:24.36JT[TK]D-Fender: oh and the vaya cost AUD$130, about USD$110 :D
23:24.39JTavaya
23:24.44P4C0but some times it works
23:24.51voltagexJT: Have you had any experience with KoalaVOIP?
23:24.53*** join/#asterisk coppice (n=chatzill@90.203.17.210.dyn.pacific.net.hk)
23:24.55flendersJT: where did you get it?
23:25.00JTflenders: ebay :)
23:25.02[TK]D-FenderJT : excellent sounding deal... if it works, and works well :)
23:25.04flendersJT: us?
23:25.07JTau
23:25.20P4C0show channels doesn't give me the codecs
23:25.29JTvoltagex: not first hand, but everything i've heard says "avoid"
23:25.34P4C0show channels
23:25.34P4C0Channel              Location             State   Application(Data)
23:25.34P4C0SIP/3-081f1a98       (None)               Up      Bridged Call(SIP/3800735-081ec
23:25.34P4C0SIP/3800735-081ecd28 s@commercial:1       Up      Dial(SIP/2&SIP/3&SIP/1|30)
23:25.34P4C02 active channels
23:25.35P4C01 active call
23:25.38[TK]D-FenderP4C0, "sip show channels"
23:25.52voltagexJT: yeah, but I have $10 free credit
23:26.02JTvoltagex: i guess you could give it a go
23:26.03voltagexit's just not registering :/
23:26.11[TK]D-Fendervoltagex, I'm suspecting it'll cost you :)
23:26.20Qwell[]sorry, show channel <channel>
23:26.25JTvoltagex: check ozvoipstatus to see if it's up
23:26.33voltagex[TK]D-Fender: in time only, they don't have any details
23:26.40P4C0does this makes any sense: 192.168.6.5      (None)      0F7C823D-ED  00101/00074  unkn  No       Rx: OPTIONS ? (in sip show channels? i have that line... aditional to the call from my server to phone and server to voip provider)
23:27.11*** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-88-155-97-204.red.bezeqint.net)
23:27.24JTvoltagex: also check the whirlpool voip wiki, it will have details on how to setup asterisk for it
23:28.39voltagexkoala's up, my config matches whirlpool's
23:29.01voltagexiax2 show registry is empty
23:29.49*** join/#asterisk lerat (n=dnormand@70.55.203.236)
23:29.50ZX81P4C0: will be a registration or SIP Options packet
23:29.56fall0utHas anybody done anything with ss7box?
23:29.57flendersJT: do you have a poe switch?
23:30.02JTflenders: no
23:30.11leratHi all
23:30.23JTi should see if PoE modules are available for my ProCurve
23:30.26JTi doubt it
23:30.32P4C0maybe it's because i'm not registered to my voip provider...
23:31.04flendersyou have a procurve at home?
23:31.33ZX81P4C0: I doubt your provider will be 192.168.x.x :)
23:31.42ZX81unless you are your provider
23:31.51leratI have already an asterisk ver. 1.4.1 with druid and i want to install an other GUI... how can i remove my old GUI without having to reinstall asterisk ?
23:31.55P4C0ZX81, no, it's not, I mean for the codec problem
23:31.57JTflenders: yeah, only 48 ports fitted i think...
23:32.16ZX81P4C0: when you do sip show channels, what codec does it say you are using?
23:33.08P4C0ZX81, the correct ones... alaw to the local phone, g729 to the provider... but that's when ringing...
23:33.28[TK]D-FenderP4C0, what phone?
23:33.47P4C0[TK]D-Fender, local sip phone... sjphone
23:34.14[TK]D-FenderP4C0, You have both with a #1 priority of G729.  IIRC it doesn't SUPPORT G729
23:34.24leratDoes anybody can help me ?????
23:34.27[TK]D-FenderP4C0, If you set the soft-phone for ALAW only, does it work?
23:35.05[TK]D-Fenderlerat, This is not a GUI support channel.  Check the one appropriate to the one you're using
23:35.18P4C0[TK]D-Fender, disallow=g729 inside the sip.conf local phone?
23:35.18leratsorry
23:35.25leratmy mistake
23:35.31[TK]D-FenderP4C0, disallow=all, allow=alaw
23:35.39*** part/#asterisk lerat (n=dnormand@70.55.203.236)
23:35.49P4C0[TK]D-Fender, let me check... i did only with disallow=g729 moment
23:35.49*** join/#asterisk Fieldy (i=sK5kdpjm@gentoo/contributor/Fieldy)
23:35.50*** join/#asterisk tcastleman (n=chatzill@pool-81-187-17-249.b3-it.com)
23:35.51*** join/#asterisk ToyMan (n=Stuart@user-12lcqol.cable.mindspring.com)
23:36.08P4C0[TK]D-Fender,  what about in global scope?
23:36.24Fieldyhm.. is there a certain type of MP3 that works best for music on hold? i put some up and some come over solid, but some come over extremely broken up. set verbose 10 isn't complaining
23:38.32P4C0http://rafb.net/p/7Q3FPS41.html
23:39.17[TK]D-FenderFieldy, 128kbit non VBR
23:39.19P4C0this is fustrating...
23:39.44[TK]D-FenderP4C0, make sure canreinvite=no for both.
23:39.51[TK]D-Fenderok, gotta go for now, back in a few hours
23:39.53P4C0[TK]D-Fender, it is
23:40.16Fieldy[TK]D-Fender: ok thanks
23:40.26P4C0I'll go as well
23:42.05tcastlemanHi there. We have a strange problem with asterisk's sounds suddenly not being audible. It is intermittent. Once it works it works. But intermittently on starting zaptel, misdn and amportal it occurs. Calls between SIP extensions are unaffected. It's just the sounds played by asterisk itself.
23:42.12tcastlemanOn the console all seems normal and asterisk 'says' its playing the sound, the nothing is audible on the handset
23:42.16tcastlemanwe are using asterisk 1.2.17, zaptel 1.2.17 with misdn for a TDM880P and a B410P
23:42.27tcastlemanon a Dell Poweredge 840
23:42.52*** join/#asterisk Stridernzl (n=neville@125-239-175-26.jetstream.xtra.co.nz)
23:44.13tcastlemanonce this problem is being experienced asterisk sounds are inaudible whether it be on a SIP extension, ZAP extension or via an misdn or SIP trunk. Nothing.
23:44.22tcastlemana series of starts and stops can fix it
23:44.31tcastlemanbut it seems to be completely random
23:45.02tcastlemanand the asterisk console shows no errors, it looks like all is operating normally..
23:45.08Dr-Linux|workany perl expert around? :P
23:45.21tcastlemanpermissions on the asterisk sounds is correct
23:45.31flendersJT: just about to buy that sangoma 101
23:45.45JTflenders: cool
23:45.47ealddoes the order of extra keys in AMI Actions, matters? for example in Action: Login if I send secret before login?
23:45.59JTtcastleman: ok, try this tesT:
23:46.23JTtcastleman: add a Wait(2) line before the audio is played back
23:46.49JTtcastleman: tell me if it does anything
23:47.04teneDr-Linux|work: a little bit.  what's up?
23:47.39Dr-Linux|workdon't worry, it's my 2nd day looking perl
23:47.45Dr-Linux|worktene: can i /msg you? :)
23:47.56teneDr-Linux|work: sure.
23:47.59tcastlemanJT, it's working now.. let me try to break it and test
23:48.05Dr-Linux|workthanks
23:48.09JTtcastleman: ok
23:48.53tcastlemanJT, I'm using free PBX. Where did you want me to put the Wait(2)?
23:49.23JTtcastleman: eh, well i expect you to know what you're doing with the dialplan if you come here with a freepbx problem
23:50.58tcastlemanJT, I do know what I am doing with the dialplan to a point. I have several custom contexts I could text with. Did you want me to test with one of those?
23:51.19JTwell yeah, somewhere that you are having a problem
23:51.31tcastlemanJT, ok cool no problem. give me a sec
23:53.41*** join/#asterisk Dovid (n=Dovid@l192-117-114-1.broadband.actcom.net.il)
23:58.05tcastlemanJT, typical I can't break it now.
23:58.22tcastlemanJT, the annoying think is that it seems to be completely random.
23:58.49JThmm

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