IRC log for #asterisk on 20071103

00:14.02Alpha232<PROTECTED>
00:15.09Iamnach0uh....
00:15.18Alpha232just finished eating
00:15.34Alpha232bosses 13yo daughter made home made pizza
00:21.37*** join/#asterisk katsuodo (n=musashi@ool-457cc6dc.dyn.optonline.net)
00:23.59katsuodotelephone connection question
00:25.13*** join/#asterisk katsuodo (n=musashi@ool-457cc6dc.dyn.optonline.net)
00:25.56katsuodoThere two telephone jacks and one telephone number
00:27.05katsuodophone at desk is plugged in one jack and a extended cord plugged into port 4 of tdm card into the other jack
00:27.51katsuodowith analog phone plugged into port 1
00:28.01katsuodocan I have two extension
00:28.16Alpha232interesting, asterisk can run as a php server
00:40.24MrTelephoneWhats with this when I try to dial a sip phone that has an ongoing call and callwaiting=yes? Got SIP response 488 "Not Acceptable Here" back from
00:42.04*** join/#asterisk BBHoss_Work (n=hoss@216.186.235.254)
00:42.23BBHoss_Workhow is old 1.4.13?  i heard it was buggy
00:42.42Alowishustopic says 10/10/07
00:43.34BBHoss_Workwhat?
00:44.04Alowishuswhat... what?
00:44.38BBHoss_Workwhat does 10/10/07 have to do with how buggy it is?
00:45.06Alowishusooh lol I guess I'm tired... I thought you wrote "how old is 1.4.13"
00:45.10*** join/#asterisk Toerkeium (i=oo@201.216.206.221)
00:45.22Alowishusbut haven't heard anything about it being particularly buggy
00:45.43BBHoss_Workhow is old 1.4.13, you probably thought i was a foreigner that didnt know good english
00:46.00Alowishuslol no I just can't read
00:46.11BBHoss_Workok
00:46.15BBHoss_Workanyone else?
00:47.14Alpha23220:42 < BBHoss_Work> how is old 1.4.13?  i heard it was buggy
00:47.20Alpha23220:45 < Alowishus> ooh lol I guess I'm tired... I thought you wrote "how old is 1.4.13"
00:47.24Alpha232he did...
00:48.46BBHoss_Workno, i said how is old <version>
00:49.01BBHoss_Workwhat i guess i REALLY meant to say is how is ol' <version>
00:49.23BBHoss_Worki live in the south, youll have to excuse my accent :)
00:49.25Alowishuswhat version are you coming from?  1.4.12?  or a 1.2.x?
00:49.42BBHoss_Workanything but stable :)
00:49.45BBHoss_Worktrashbox
00:50.12Alpha232bah fscking BRI
00:50.56agxAlpha232, uh? :)
00:51.29Alpha232BRI support for * sucks in the us
00:51.52agxAlpha232, BRI does exist in US? O.o
00:52.35Alpha232I have one at home
00:54.19Alpha232agx: do you have BRI in Italia?
00:57.08MrTelephoneif i set allow=g729 and disallow=all callwaiting doesn't work
00:57.12MrTelephonewhats with that I wonder
00:59.36Alpha232do you have a license for G729?
01:02.12Mw3whats the difference between the us bri and the eu bri?
01:02.40Alpha232the protocol
01:03.39Alpha232for BRI in the US you find NI1
01:03.54MrTelephoneyeah i have g729
01:04.00MrTelephonebut the other client wasn't allowed to use it
01:04.01Alpha232er or others
01:04.03*** join/#asterisk techie (n=techie@adsl-76-214-7-165.dsl.lsan03.sbcglobal.net)
01:04.04MrTelephoneso callwaiting didn't work
01:04.28MrTelephonei have to get rid of my mgcp clients they don't work worth the crap
01:04.33MrTelephonemgcp/ncs
01:04.52MrTelephonei forgot to compile with the patch and lost about 20 clients
01:05.01agxAlpha232, yes i'm full of BRI
01:05.35Alpha2322B1Q coding is the standard used in North America
01:06.44Alpha2324B3T is a standard used in Europe and elsewhere in the world
01:06.50agxAlpha232, have lots of people with bugged NT1+ that does not work with bristuff... luckly mISDN rocks on BRI TE mode
01:07.37*** join/#asterisk yannj_fr (n=yannj_fr@vpn.intelunix.fr)
01:08.09Alpha232agx: hmm i didn't even think about that
01:08.22Alpha232the NT1 talks the 2b1q
01:08.33Alpha232how much more is needed to get it to work in the US
01:09.26*** join/#asterisk CrazyTux[m] (n=CrazyTux@adsl-68-90-41-25.dsl.hstntx.swbell.net)
01:10.35Qwellhey, what's a good electronics store?  Something like radio shack, but...good...and...an electronics store..
01:11.21MrTelephonen/a qwell
01:11.22agxAlpha232, i don't have any clue about the 2 differences
01:11.37Qwellneed to find a 3.5mm A/B switch
01:11.45MrTelephonegood luck
01:11.50MrTelephonethat would be a tough one
01:11.57MrTelephonemusic store?
01:12.03MrTelephonefrys electronics?
01:12.06Qwellyeah, that's probably a good idea
01:12.10Qwellno Fry's around here ;/
01:12.20Qwellso what's a good music store? :p
01:12.41MrTelephonegoto radio shack and buy 2 female 3.5mms and hook them together using some rocker switches or something
01:12.53Qwellmeh
01:13.03Qwelland I'd need 3
01:13.06MrTelephonemake a nice little case for it and sell it as an asteirsk addon for 50 bucks
01:13.10Qwellheh
01:13.36MrTelephoneor if your really nerdy eprom your own chip to do the switching
01:13.40MrTelephonehaha
01:13.42Qwellno
01:14.35MrTelephoneqwell, obdc voicemail, who do you know storing it?
01:18.06*** join/#asterisk LakeSolon (n=blake@64-83-209-86.dhcp.stcd.mn.charter.com)
01:18.39Qwellhttp://www.newegg.com/Product/Product.aspx?Item=N82E16826265015
01:18.42Qwellnewegg FTW
01:19.31MrTelephoneas if you found it that easy
01:19.54*** join/#asterisk LakeSolon (n=blake@64-83-209-86.dhcp.stcd.mn.charter.com)
01:19.59Qwellno, it took forever actually, heh
01:20.16MrTelephonei know cuz i look for odd shit too sometimes and i can't find dick
01:20.25MrTelephoneone good example is t1 failover switches
01:20.47Qwellhttp://www.thinkgeek.com/computing/speakers/8054/  I like that one better, but they stopped selling it
01:21.25MrTelephonenot too shabby
01:22.43MrTelephonewhat are you trying to do
01:22.51MrTelephonemixin some rap music off hours?
01:22.55Qwellswitch stuff :p
01:24.49MrTelephoneheres my switch toy
01:24.51MrTelephonehttp://www.dataprobe.com/products/switch/aps/t-aps/index.html
01:26.58MacWinneris there any issue with asterisk processing DTMF tones slowly?  ie, if I type in the PIN code very quickly, it misses some of them?
01:27.18MrTelephonemacwinner happens to me too
01:27.28MrTelephonei tell people to dial sloowww
01:27.30MrTelephonehaha
01:33.11J4k3woooord
01:33.53J4k3xv6700 irc via ssh
01:34.17*** join/#asterisk ectospasm (n=ectospas@c-68-62-219-11.hsd1.al.comcast.net)
01:35.53*** join/#asterisk ghento (n=ghento@64.180.85.230)
01:43.28*** join/#asterisk jm|laptop (n=jm|home@dilbert.jamiem.com)
01:45.26*** join/#asterisk CVirus (n=GoD@196.205.192.166)
01:48.48MacWinnermrtelephone: is it a known issue/bug?
01:54.24*** join/#asterisk saftsack (n=saftsack@pD9E04C9A.dip.t-dialin.net)
01:55.54*** join/#asterisk saint_ (n=saint@c-69-242-118-124.hsd1.nj.comcast.net)
02:03.04MacWinnerlooks like a bug that may have been addressed in: http://bugs.digium.com/view.php?id=10535
02:09.49TJNIIWell, now my sip phone will register, send, and recieve calls when away from home, but no audio.
02:10.02TJNIITime to read up on STUN, I think.
02:14.23*** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org)
02:14.45*** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org)
02:22.00*** join/#asterisk ReDNeQ (i=ReDNeQ@cpe-66-68-37-190.austin.res.rr.com)
02:23.18*** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com)
02:23.18*** mode/#asterisk [+o anthm] by ChanServ
02:23.40orkidwhy is encryption not that big in voip? or is it?
02:24.19*** join/#asterisk |omni| (n=rob@c-67-185-56-106.hsd1.wa.comcast.net)
02:25.12TJNIISTUN is supported in 1.4 but not 1.2, correct?
02:30.36*** join/#asterisk ZaVoid (n=zavoid@c-67-165-25-195.hsd1.pa.comcast.net)
02:34.07TJNIIOh.  Stun requires 2 IPs....  I only have one.  Bah.
02:36.07*** part/#asterisk Alowishus (n=jpenix@fwsdo.projectdesign.com)
02:36.28TJNIIAnd I get audio if the client is not NATed....
02:39.02*** join/#asterisk ghento (n=ghento@64.180.85.230)
02:40.34*** join/#asterisk denon (n=denon@tooth.decay.org)
02:40.34*** mode/#asterisk [+o denon] by ChanServ
02:45.40TJNIIhmmmm.... I have canreinvite=no, but I still see "Attempting native bridge" on the console
02:46.02*** join/#asterisk ReDNeQ (i=ReDNeQ@cpe-66-68-37-190.austin.res.rr.com)
02:46.07*** part/#asterisk techie (n=techie@adsl-76-214-7-165.dsl.lsan03.sbcglobal.net)
02:47.33*** join/#asterisk [hC] (n=hardcore@S01060016b61c8983.vf.shawcable.net)
02:47.55*** join/#asterisk FireMac (n=firemac@CPE000d88ae88b9-CM0011ae8bb0ee.cpe.net.cable.rogers.com)
02:50.34*** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla)
02:50.34*** mode/#asterisk [+o russellb] by ChanServ
02:51.07*** join/#asterisk gerphimum (n=trekkie@cpe-67-9-102-186.satx.res.rr.com)
02:55.52*** join/#asterisk L2SHO_ (n=adam@67.132.43.8)
02:56.57L2SHO_ok, so I set up a peer definition for a proxy, I set the fromuser=s12345678, but the machine I'm placing the call through only sees the   s, not the 12345678
02:57.29L2SHO_any ideas?
02:57.50JerJerdrop the s
02:58.32L2SHO_thats not an option, I don't have control of the machine I'm placing the call through
03:00.54TJNIIHmmmm... With canreinvite=no should I see "Attempting native bridge" on the console?
03:01.03JerJerthen use host=
03:01.15JerJerwithout other auth
03:03.06L2SHO_I'm already using host=    username=   secret= and fromuser=
03:03.37L2SHO_if I take out the fromuser= the other machine will ignore me
03:08.08*** join/#asterisk blitzrage (n=Leif@asterisk/documenteur-extraordinaire/blitzrage)
03:08.08*** mode/#asterisk [+o blitzrage] by ChanServ
03:08.47[hC]hey blitzy
03:08.54blitzrage[hC]: zup
03:08.57MrTelephonetjni, native bridging is between any of the two of the same protocols
03:09.04MrTelephonepardon my english
03:09.14[hC]just about to head out i think.. been lounging around the house all day with a hang over... might go check out guitar hero 3 at a friends place.
03:09.16[hC]:)
03:09.18[hC]how about you?
03:09.41blitzragemeh... friend Jen was over watching TV and a movie, and now thinkin' about going to bed :)
03:09.47MrTelephonel2sho, what kind of machine are u connecting too?
03:11.30[hC]blitzrage: i cant say that i dont appreciate where your head's at.. :)
03:11.48blitzragefor sure
03:11.50blitzragelazy weekend
03:11.54*** join/#asterisk saftsack (n=saftsack@pD9E04C9A.dip.t-dialin.net)
03:11.54blitzrageprobably not gonna do much
03:12.16TJNIIMrTelephone: So that can be seen if * is simply acting as a RTP pass-through
03:12.22[hC]ya im gonna try to get some long overdue errands accomplished
03:12.29blitzragenice nice
03:12.58blitzragewelp, I'm outta here
03:13.02blitzragenight all
03:14.35L2SHO_MrTelephone: it's an asterisk machine
03:15.07L2SHO_MrTelephone: an asterisk machine to another asterisk machine
03:18.27L2SHO_it seems like when the format is fromuser=s#######, only the s gets sent for some reason
03:18.45L2SHO_or something like that, I'm not really sure tho
03:20.23*** join/#asterisk Teln12100 (i=hello123@bas2-toronto12-1168023548.dsl.bell.ca)
03:20.33MrTelephonei never had to use it from asterisk to asterisk
03:20.52MrTelephoneyou don't have access to the remote machine?
03:21.13MrTelephonesome providers require trustrpid=yes
03:21.18MrTelephonetry that?
03:22.05L2SHO_I have remote access, but I'm not authorized to make any changes
03:22.49MrTelephonewhats the config look like
03:22.54L2SHO_thats bizzare, x-lite was working on the same machine a few min ago, now it's not working
03:23.01L2SHO_something strange is going on
03:23.09MrTelephonehow far is the machine away?
03:23.15MrTelephonesame network?
03:23.22L2SHO_no
03:23.26MrTelephonesip debug <peer>
03:31.29TJNIIHmmmmm.... So if my SIP client has a public IP address it works, but if is behind a NAT id doesn't at all.  I'm thinking it may be a lcient config issue....
03:33.43MrTelephonereinvite=no and nat=yes
03:34.03MrTelephoneand you should register every 30 seconds if the client is behind a nat
03:34.27MrTelephonesip show peers and see what ip address asterisk has for your client
03:35.43TJNIII have reinvite=no and nat=yes
03:36.07TJNIIThe server is NATed as well, I think those options are working
03:36.14MrTelephonethe server is nated?
03:36.27TJNIIUnfortunately.  So 2 NATS
03:37.03TJNIIThe server side router has a ststic IP and port forwarding enabled.
03:37.18TJNIII did forget to use sip show peers to check the IP, though
03:37.22MrTelephonebut u have to setup outside nat address in sip.conf for the server
03:37.25MrTelephoneand specify localnets
03:37.30TJNIIRight.  Did that
03:37.51MrTelephoneforward port 5060 but what about rtp?
03:38.06TJNIISet the rtp port range to 10K-15K and forwarded it
03:40.20MrTelephonesounds like you ocvered all the bases
03:40.22TJNIIThe 30 sec register is for keeping the RTP "hole" open, correct?  It's not for SIP signalling
03:40.54TJNIII found at least one mistake in the client config after getting back home that could cause this.
03:40.54MrTelephonei set my clients to 30 sec cuz home routers usually close the port when there is no activity
03:41.02MrTelephone30 sec register helps you receive calls
03:41.06MrTelephoneoutgoing calls always work
03:41.15TJNIIIt's the RTP port that needs to kept open?  Or 5060 for sip
03:41.17MrTelephoneincoming calls won't work if the router shuts the port down to the client
03:41.25MrTelephone5060
03:41.29TJNIIhmmmm
03:41.41TJNIIIt can send and recieve calls OK, just no audio
03:42.35MrTelephoneone way audio or none
03:42.45TJNIINone.  Which is perplexing
03:43.03TJNIIWith a client outside a nat (right on the intarnets) audio works both ways
03:43.19MrTelephonewith the server behind nat?
03:43.34TJNIIWith the settings and forwarding mentioned above, yes
03:45.00TJNIIToo many tees in my internet tube, I guess....
03:45.05MrTelephoneyour client side router sounds like its blocking the rtp
03:45.14*** join/#asterisk CVirus (n=GoD@196.205.192.166)
03:45.22TJNIIYea, that sounds right.
03:45.32MrTelephonewhat happens is that the rtp ports are in the sip transmissions
03:45.42MrTelephonebut some routers use different outside ports than inside ports
03:45.52TJNIII tried two locations, both no-go
03:45.59MrTelephoneso ur client says my rtp port is 1000 but the router decides to use 2000
03:46.16TJNIII didn't have sip debug on, though (Didn't know about it 'till about an hour ago)
03:46.35MrTelephonedo rtp debug and make sure asterisk is sending rtp to the right client ip
03:47.05TJNIIThe public IP and not the NATed IP, you mean?
03:47.10MrTelephoneyeah
03:47.17MrTelephonertp should be sent to the clients public ip
03:47.38TJNIILemme see if I can poke Joel into firing a client up again
03:48.45MrTelephoneyou have to force asterisk to send rtp to a certain port i guess
03:48.48MrTelephoneand open that up on the router
03:50.09TJNIII'm thinking the client IP address may be the problem
03:52.26*** join/#asterisk bmg505 (n=leon@196.209.183.44)
03:54.48*** part/#asterisk karleeto (n=karl@207.191.91.242)
03:57.34*** join/#asterisk Flauto (n=zhao@71.194.141.225)
03:58.40Flautoi use voipmich for toll free terminations but it stopped working, anyone knows what is going on?
03:59.10*** join/#asterisk denon (n=denon@tooth.decay.org)
03:59.10*** mode/#asterisk [+o denon] by ChanServ
03:59.16L2SHO_Via: SIP/2.0/UDP 192.168.1.200:5060;branch=z9hG4bK61a408d5;rport
03:59.28TJNIIThe IP address looks okay but I do see port 55515 not 5060....
03:59.41L2SHO_is it ok to see a private IP in the Via: field of a sip packet?
04:00.21Flautois there any other provider offers free toll free termination
04:00.49*** join/#asterisk mtaht4 (n=m@166-108-62-200.enitel.net.ni)
04:01.21jqlL2SHO: in any but the topmost header, yeah
04:02.07jqlin the topmost header, /sometimes/
04:04.47L2SHO_<sigh>  damn, I thought maybe that was my problem
04:05.11*** join/#asterisk MaliutaBris (n=nikolai@c210-49-69-241.rochd2.qld.optusnet.com.au)
04:05.26jqlit could be. after all, your bottom field is the phone's top field... :)
04:05.42jqldepends on your network
04:06.28*** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn)
04:06.46*** join/#asterisk remmo (n=junk@202.1.119.80)
04:07.01L2SHO_ok, so how can I make asterisk put the public IP in all the fields when I make a call?
04:07.52jqlthere's a public ip parameter in sip.conf
04:08.03jqlI could look it up, but that'd spoil the fun
04:08.31L2SHO_I've got externalhost=myhost/domain, but that didn't seem to do it for me
04:08.45MaliutaBrishost?
04:08.56MaliutaBrisyou mean externalip right?
04:09.28jqlyeah, externhost isn't the most helpful option, there
04:10.15*** join/#asterisk Mavvie (n=edwin@ppp121-44-127-146.lns10.syd6.internode.on.net)
04:10.25L2SHO_so externip than?
04:10.50*** join/#asterisk Mavvie (n=edwin@ppp121-44-127-146.lns10.syd6.internode.on.net)
04:11.13L2SHO_or theres nat=yes
04:12.15MaliutaBrisand then there's localnet=127.0.0.1/8
04:12.31MaliutaBristhey are not exclusive
04:13.23Alpha232moooo
04:14.01MaliutaBrisbaaaaa
04:14.41Alpha232hmm lambchops
04:15.10*** join/#asterisk hijacked (i=rUgb@66.255.220.17)
04:16.56MaliutaBrisI would go "quack", but I might have to roast my self for eating
04:17.02Alpha232lol this is sad, calling into my own system to listen to hold music
04:17.19MaliutaBrisand I don't know how to type a representation of the noise a roo makes
04:17.23*** join/#asterisk mindCrime (n=chatzill@cpe-065-190-188-124.nc.res.rr.com)
04:17.35MaliutaBrisI call my system to hear the monkeys scream
04:17.50Alpha232lol i don't have that recording
04:18.32MaliutaBrisit's in the standard addons package
04:19.45ectospasmis it in addons?  or extra-sounds?
04:20.00ectospasmtt-monkeys I thought was in the standard distribution...
04:20.27Alpha232-rw-r--r--  1 root root   4983 2006-06-20 15:30 the-monkeys-twice.gsm
04:20.28Alpha232ahh
04:20.34Alpha232-rw-r--r--  1 root root  26697 2006-06-20 15:30 lots-o-monkeys.gsm
04:20.48Alpha232but apparently there is no context that uses them
04:20.50Alpha232bummer
04:21.52MaliutaBrisAlpha232: set one up, that's what Play() is for
04:23.23MaliutaBrisTJNII: soft or hard?
04:23.45Alpha232i just did
04:23.58TJNIIHard
04:24.17MaliutaBrispeople are making IAX hardphones?
04:24.25TJNIISrting too
04:24.30TJNIIs/too/to/
04:24.49TJNIII have one that does IAX/IAX2/SIP but the DSP is crap
04:24.54*** join/#asterisk CunningPike (n=CunningP@S010600095b33697f.vc.shawcable.net)
04:26.13MaliutaBrislooks like I am going to have to start looking at usb handset/softphone option as part of my new job
04:26.39MaliutaBrisneed to provider road warriors with extensions
04:27.43Alpha232ugh god
04:28.26MaliutaBriswell it's to go with laptops and 3G cards
04:28.39MrTelephonemy coworker wants "down with the sickness" as his ring tone
04:28.55MaliutaBrisit's doable
04:28.57MaliutaBrisI would
04:29.04MaliutaBrisfor a select group of people
04:29.49Alpha232argh i need a meet me to work and i don't have the damn dummy module
04:29.49Alpha232ugh
04:30.03MrTelephonehow can you not have it
04:30.07MrTelephoneinsmod ztdummy
04:30.20MrTelephonei couldn't stop the damn modules from loading
04:30.25Alpha232apt-get install insmod: can't read 'ztdummy': No such file or directory
04:30.26Alpha232iios
04:30.27Alpha232oops
04:30.27Alpha232insmod: can't read 'ztdummy': No such file or directory
04:30.44Alpha232lol  i did an apt-get on ubuntu for asteisk
04:30.56*** join/#asterisk sid (i=unstable@tor/regular/sid)
04:31.13sidI have some mini-itx linux box...what do I need to use asterisk?
04:31.23MrTelephonehaha
04:31.33MrTelephonesid, some ram and a cpu
04:31.34Alpha232sid: the question is, to use it for what
04:31.42sidI have ram and cpu, and a disk
04:31.44sidand PSU
04:31.46sidetc
04:31.47MrTelephonehe wants to run a 300 person call center
04:31.55MrTelephonej/k
04:32.09sidAlpha232: I have an office, and I'm pretty much the only one who will use the phone. so I just wanted to hook a sip phone to it
04:32.19sidI have vonage
04:32.44sidbut there is this aastra speaker phone/head set I want..and those mother fuckers don't allow certain mac addresses to work with their system
04:32.48MaliutaBrissid: you may also require some technical skill to run asterisk
04:32.56sidand if I have to setup a linux box to spoof the mac address...
04:33.04sidI might as well go one step further and cancel vonage and setup asterisk
04:33.13sidbut I've never set it up, so I don't know what to get
04:33.19MrTelephoneyou have to pretty much be a genius to get asterisk working properly
04:33.32MrTelephoneplus you need to know c programming so you can make it work
04:33.32MrTelephoneheh
04:33.42sidWhat hardware do I need?
04:33.46sidsome card from digium? which one?
04:34.22MrTelephonesingle port fxo card
04:34.40*** join/#asterisk bungalow (n=yakkop@ip72-205-203-201.sb.sd.cox.net)
04:34.43sidMaliutaBris: I setup crappy crm systems, installing php, mysql, and setting all that up. Is it about the same level as that?
04:35.05bungalowHi -- anyone know of a way to manually force Asterisk to re-Invite?
04:35.23MaliutaBrissid: read the book first
04:35.25bungalow... i.e. through a manager or dial plan command
04:35.38MrTelephonebugalow, are you trying to do a handoff?
04:35.38sidMaliutaBris: "the book"?
04:35.46MaliutaBris~book
04:35.47jbotAsterisk: The Future of Telephony 2nd Edition (ISBN  0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf
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04:36.09sid15 megs?
04:36.11sidare you serious?
04:36.18bungalowMrTelephne: yes, but rather than Asterisk do it automaticaly after dial is answered, I'd like to do it from a dial plan app or agi
04:36.40bungalowMrTelephone: just want to control when Asterisk gets out of the media stream
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04:37.08sidMaliutaBris: over 600 pages need to be read to know how to setup asterisk?
04:37.29MaliutaBristhere is a little more to it than that
04:37.33TJNIIsid: Read the intro.  You'll figure out what you need from there
04:37.40sidMaliutaBris: mysql.com has thousands of pages of docs, but I never needed to read any of that to setup CRM websites, you just need two commands to add user and add database.
04:37.54jqlno, over 600 pages need to be read before being worthy of asking a challenging question. Not that simple questions won't be answered here. Just an FYI. :)
04:37.59TJNIIsid: Skimming it is a good idea.  Give it an hour or two.
04:38.17MaliutaBrissid: so have n technical skills
04:38.38bungalowMrTelephone: still there?
04:39.01MrTelephoneyeah im just doing some reaing on reinvites
04:39.16sidHow much should this "single port fxo card" cost in USD about?
04:39.30MrTelephoneasterisk will reinvite only if the channels/codecs are the same and if canreinvite=yes
04:39.37MrTelephonebut i don't see a command to hand it off after
04:39.40MrTelephonethats a cool idea though
04:40.38bungalowthanks for checking
04:40.39MrTelephoneI don't think there is a command to do it
04:41.05MrTelephoneif anything it would be in the dial command
04:41.12MrTelephonedon't take my word for it im probably wrong
04:41.42MrTelephonesid, 50 bucks for a good one at the most.. u want to plug into your vonage adaptor?
04:42.14sidMrTelephone: no, I want to cancel vonage, and use a GNU/Linux box connected to a linksys router
04:42.27sidand get some card and plug it into my GNU/Linux box via PCI
04:42.45MrTelephoneyou can pay for a voip provider too if you want to avoid hardware
04:42.59MrTelephonewhat do you have for phone service?
04:43.27sidI have my local cable monopoly(cablevision), and vonage.
04:43.57jqlwait, you want to reinvite mid-call?
04:44.04sidthey're both digital to analog adapters, and you just plug plain analog phones into them and get a dialtone
04:44.59sidI could get the vonage straigt voip adapter, and plug some voip phone into vonage and use that. But vonage is anti-competitive, and they only let certain phones connect to their network.
04:46.33MrTelephonei'd setup an asterisk box, hook up a sip phone and pay voicepulse.com or something 20 bucks for a bundle of minutes
04:46.35jqlyeah, those analog boxes don't do it for me
04:46.50jqlI need my polycom fix
04:47.14sidWhat PCI card do I buy?
04:47.19MrTelephonethe only way to reinvite would be to dial another sip agent that is revinite=yes and the call will be handed off
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04:47.32sidI have Ubuntu on a box, I just need a PCI card
04:47.41MrTelephonesid, why not use another phone provider over the internet?
04:47.47MrTelephoneyou want to stay with vonage?
04:47.51MrTelephonesangoma A200
04:48.58sidMrTelephone: What phone provider?
04:49.20MrTelephonevoicepulse is one that comes to mind
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04:50.56sidevince sucks
04:51.01sidxpdf is so much better
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04:54.37MrTelephoneg729 to g729 doesn't do transcoding right
04:55.13orkidi agree with sid
04:55.52MrTelephoneevince?
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04:56.19orkidreader for gnome iirc
04:58.12orkidwhat does it take for a number provider (like bell) to be able to transfer a number to a voip provider?
04:58.52orkidit's been a WHILE since voip 'came out' into mainstream, and still in my area this is not available. sorry,this is more voip related, kind of ot for #asterisk, but perhaps someone will be kind and provide some answer.
05:00.32TrentCreeku have to ask the VOIP provider to do it
05:00.42TrentCreekand they will charge u
05:04.34tzangercitats: wow nice you've got svn commit access
05:04.56tzangerI ain't programmer enough to let them let me get near that stuff with a 10' pole :-)
05:06.08orkidyeah. ... but i check vonage, and it says my number is not avaible to be transfered... sooo... you're saying it is, but it'll cost me? TrentCreek ?
05:07.19MrTelephoneit means they don't have an agreement with the facility that owns the number
05:07.28MrTelephoneyou won't be able to get it i think
05:09.43Sweeperuh
05:09.51Sweeperdon't they like, legally have to port your number?
05:10.01Sweeperlike cellphone numbers....
05:16.25MrTelephonenot sure
05:17.41MrTelephonei thought rfc2833 was the best
05:17.51MrTelephonethis company that makes sip adapters called arris uses inband by default
05:17.51MrTelephonehmm
05:19.35BeeBuui have 2 FXO and 2 FXS ports,how can i call 2 FXS ports each other?
05:21.21MrTelephonewhat
05:21.33MrTelephoneof course
05:21.45MrTelephonering both at the same time?
05:21.57MrTelephoneor call fxs2 from fxs1?
05:23.19BeeBuucall fxs2 from fxs1
05:24.10MrTelephonefxs1 is ZAP/3 and fxs2 is ZAP/4?
05:24.21MrTelephonegive them extensions in extensions.conf
05:24.34MrTelephone333,1,Dial(ZAP/3) ; fxs1
05:24.45MrTelephone334,1,Dial(ZAP/4) ; fxs2
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05:25.06alpha232fscking power company
05:25.13MrTelephonefsck them
05:25.24MrTelephoneim livin off solar
05:25.26BeeBuuhow to set the 333 is ZAP/3?
05:25.35BeeBuuwhich conf file,please?
05:25.41alpha232MrTelephone: it's 1:25am, night time, not in the artic circle
05:25.46MrTelephonebeebuu, google extensions.com
05:25.48MrTelephoneoops
05:25.51MrTelephoneextensions.conf
05:26.01MrTelephoneits 1:25am here too
05:26.04BeeBuuO
05:26.05MrTelephoneits pitch black
05:26.13MrTelephonei canadian tire solar panel powers a radio
05:26.21BeeBuuthanks.
05:26.59MrTelephonebeebuu, looks like you have some research to do :(
05:27.33BeeBuui'm a newbie,sorry for bother
05:28.03MrTelephonethats ok im a newbie too
05:30.34BeeBuuanother question:how to ring ZAP/3?
05:31.03BeeBuui can dial zap/3,but the phone not ring at all..
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05:31.24alpha232not fun
05:32.01alpha232so much for my record uptimes ;(
05:33.35BeeBuuMrTelephone: are you still there?
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05:44.38MrTelephonealpha its time to get yourself a massive battery!!
05:44.49MrTelephonebeebuu almost bedtime
05:45.26BeeBuuoh,yeah,it's! battery low~~~~~
05:45.45BeeBuuMrTelephone: have a good dream,sir.
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05:54.40L2SHO_ok, looks like I've found the problem, but I'm not sure how to fix it.  When my machine is registering, in the sip packet it sets "Contact: <sip:s@192.168.1.102>"  when it should be "Contact: <sip:s1234578@192.168.1.102>"
06:00.45L2SHO_any ideas would be appreciated
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06:12.29[TK]D-FenderL2SHO_, fix your register.  You're missing something at the end....
06:17.46Tclpcan anyone recommend a voip provider for Asterisk that uses SIP for Canada ..
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06:25.43L2SHO_[TK]D-Fender: I think my register statement is good, it registers just fine
06:25.55L2SHO_[TK]D-Fender: but the the wrong name somehow?
06:26.46[TK]D-FenderL2SHO_, register => user:pass@host/extentodialcauseyoudontwantSnowdoyou?
06:28.55L2SHO_[TK]D-Fender: you are a freaking genius
06:29.38L2SHO_I wish it would have had that /extension stuff in my Asterisk: The Future of Telephony book
06:30.05L2SHO_thanks
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06:31.17[TK]D-Fendernp
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06:33.09[TK]D-Fenderok, bed time,later all
06:34.41nclxI'm trying to get voicemail to email me. I specified in my voicemail.conf serveremail=vm@mydomain.com and fromstring=vm@mydomain.com asterisk sits on pbx.mydomain.com, the mail server is mail.mydomain.com, but every time it emails it is rejected because it is sending as root@pbx.mydomain.com which is an internal DNS name, it should be sending as vm@mydomain.com, any ideas why it isn't? postfix is the MTA /usr/sbin/sendmail -t is the mailcmd
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06:55.54orkidsip demystified looks like an interesting, and thorough book.
06:56.04orkidanyone read it and can comment?
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07:32.08L2SHO_is there any way to find out why my box would be returning a 403 Forbidden when I try an incoming call?
08:02.43BeeBuuanyone still here?
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08:41.14Sweeperno
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10:03.49deeganHi, i just upgraded to asterisk 1.4.13 and now that i call various extensions that are to use Background to play files i hear nothing. What could be the problem for this, i dont really know where to start as asterisk config has not been altered from the last version.
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10:37.19deeganThe cmd MP3Player works just fine.
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11:08.32greybeardwisemanis this for applications on phones?
11:09.02greybeardwisemani was curious about installing something on mine
11:09.11greybeardwisemanthis is my phone http://www.imagehosting.com/show.php/1329108_model800.jpg.html
11:13.18greybeardwisemanthis one too http://www.imagehosting.com/show.php/1329115_img5597.jpg.html
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11:17.52Woifi1988how can i rebuild asterisk?
11:18.02Woifi1988only by "make clean" and "make install"??
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12:35.34ZeNNsomeone working with visdn in combination with asterisk and debian ?
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14:50.49littleballhello, from where i can get the PRI error codes? i got error code 16, and need to know what it means
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14:58.48jameswf-home16 is normal termination... whats the full error
15:03.44blitzragepatience is a virtue
15:04.28jameswf-homecandy is a stripper that works with patience (another stripper)
15:04.36blitzrage:D
15:05.26mvanbaakhhmm
15:05.36mvanbaaknow we have LUA dialplan language
15:05.42mvanbaakwe can also add php
15:05.43mvanbaak:)
15:06.22mvanbaakor python
15:09.23*** join/#asterisk Adjoin (n=rangers@88.246.31.237)
15:10.40Sweeperor use a decent language like ruby ;)
15:12.28jameswf-homelmao ruby people are like bsd people, think their shi... dont smell
15:15.37Kobazokay, so, i have the zaptel drivers all going all nicey nice, how do i get asterisk to go with it now
15:16.33jameswf-homeu make && make install
15:16.52Kobazumm
15:17.05Kobazthe zaptel drivers are installed
15:17.21jameswf-homeyes so now buid asterisk
15:17.28jameswf-homeload chan_zap.so
15:17.49Kobazwell asterisk is built, i have it up and going
15:17.51Kobazk
15:18.30jameswf-homeif you made asterisk b4 zaptel chan_zap may not have built
15:18.38Kobazoooh
15:22.33Kobazchecking for ZT_DIAL_OP_CANCEL in zaptel/zaptel.h... yes
15:22.34Kobazchecking for ZT_EVENT_REMOVED in zaptel/zaptel.h... yes
15:22.34Kobazchecking for ZT_TCOP_ALLOCATE in zaptel/zaptel.h... yes
15:22.34Kobazyay
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15:32.33Kobazhmm
15:32.38Kobazchan_zap didn't get built
15:32.40ZaVoidsup all
15:32.52ZaVoidanyone seen random sound files kinda "stutter" WHEN playing
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15:38.34BeeBuuhello,all
15:39.59IPetrovHi, anyone know why callback failing when attended transfer from queue (version trunk)?
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15:40.44BeeBuui want to record all calling in,can i just use: Monitor(wav,myfilename) ?
15:41.03IPetrovbetter use MixMonitor
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15:45.16BeeBuuso where the record file in?
15:45.50IPetrovuse path for example /home/asterisk/${UNIQUEID}.wav
15:46.05blitzrageotherwise, they go into /var/spool/asterisk/recordings I think
15:46.11blitzrageor something to that effect
15:46.12ManxPowerBeeBuu: I take it you never did "show application mixmonitor" in the Asterisk CLI.
15:46.20blitzrageobviously not -- that would be stupid
15:47.12BeeBuuhm...
15:47.20brookshireirc, from the iphone!
15:47.31blitzragegross
15:47.42blitzrageI'm surprised you don't have all sorts of typos :)
15:47.42ManxPowersick
15:47.46Kobazoh
15:47.54Kobazi see why chan_zap isn't being built
15:48.13KobazZapata Telephony
15:48.14KobazDepends on: res_smdi(M), zaptel_vldtmf(E), zaptel(E), tonezone(
15:48.14KobazCan use: pri(E)
15:48.36blitzragejust run make menuselect again and make sure chan_zap is selected
15:48.40blitzragethen it'll enable all the deps
15:48.48Kobaznow i just need to figure out what M and E mean, which one means missing and which one is found
15:48.57Kobazwell i can't select chan_zap since it's missing dependencies
15:49.05brookshirei know right
15:49.12IPetrovanyone know why callback failing when attended transfer from queue (version trunk)?
15:49.25ManxPowerKobaz: you installed Zaptel BEFORE you installed Asterisk, right?
15:49.36Kobazyeah it's in
15:49.41Kobazokay, so E means missing
15:49.54blitzrageonce you've installed zaptel, make sure you run ./configure again
15:50.00Kobazyeap
15:50.11blitzragebecause that'll tell asterisk to find the modules, then make menuselect should let you just select the channel
15:50.20blitzrageI've never had to do anything different from that
15:52.08ManxPowerblitzrage: it is pretty common for menuconfig to not find an installed zaptel.
15:52.30Kobazheh
15:52.40ManxPowerI *think* it only happens if you installed asterisk, installed zaptel, then tried ./configure && menuconifg
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15:53.02Kobazwell i built the zaptel stuff
15:53.09Kobazbut my zaptel card wasn't working at the time
15:53.13Kobazand then i built asterisk
15:53.22Kobazbut that shouldn't matter though
15:53.23ManxPowerI bet rm'ing .configure.cache would help.
15:53.33Kobazsince the modules were built fine
15:53.39ManxPowerKobaz: zaptel sometimes fails to install, especially on Debian
15:53.40Kobazi do a make dist-clean
15:54.08ManxPowersomething with having udev partially installed or something.
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15:54.52Kobazi'm not even using udev
15:55.12ManxPowerKobaz: neither was any of the people that had the problem.
15:55.17Kobazheh
15:55.31Kobazsome part of me never has the urge to select udev when building a kernel
15:56.18ManxPowerdo you have a /etc/udev directory?
15:56.43Kobazi do now :P
15:56.46Kobazapt-get install udev
15:57.02ManxPowerI assume you are using the LATEST Zaptel, libPRI, and Asterisk?  Many, many bug fixes go into each release of 1.4.x
15:57.03Kobazi never noticed before that zaptel's make install was wanting udevinfo, which i lacked
15:57.10Kobazyeah all the newest
15:57.25ManxPowerKobaz: it should NOT require it.  I guess the bug was not fixed/
15:57.40Kobazwell it did ALOT more stuff now that udev is in
15:57.54Kobazso i assume asterisk was wanting all that extra stuff
15:57.59Kobazyay
15:58.01Kobazchan_zap
15:58.04Kobazyaaaaay
15:58.14Kobazyaaaaaaaaaaaaaaaaaaaaaaaay
15:59.14Kobazit would be nice to have a faster box to compile this one
15:59.15Kobazon
15:59.31Kobazbut why waste a fancy new processor on a little router
15:59.48*** part/#asterisk ozus (n=ozus@ip72-205-206-86.sb.sd.cox.net)
16:00.01Kobaznext step is getting h323 working
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16:01.16Kobaz<PROTECTED>
16:01.18Kobazyaaaay
16:01.22mockerWoo.
16:01.37mocker(oh, how punny)
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16:07.18BeeBuuexit
16:09.18KobazSCHWEET
16:09.32Kobazzap has channels
16:09.34Kobazyaaay
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16:10.05ToTohi all
16:10.25Kobazand it works, yay
16:10.39Kobazi think i need to switch codecs
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16:12.34Kobazcall quality is kinda crappy, but that's what you get with a 20 dollar fxo
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16:37.36Kobazhmm
16:37.43Kobazwhat's an easy way to convert from wav to gsm
16:39.01coppicesox
16:39.27Kobazyeah i just found it
16:39.27Kobazsox foo.wav -r 8000 foo.gsm resample -ql
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16:43.45Kobazhow do i show the current call status
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16:47.34ManxPowerKobaz: "show channels"  In fact you will find much of what you need in the CLI.
16:47.49Flautohaha
16:47.55Flautoright, kobaz
16:48.03Flautotype help under cli
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16:51.38Flautohi dlynes
16:51.42Flautohow are you doing
16:52.15Kobazaah
16:52.17Kobazokay
16:52.24Kobazi was trying everything else other than zhow channels
16:52.55Kobazokay so
16:53.05Kobazany idea why playing tracks would be really super staticy
16:53.10*** join/#asterisk implicit_ (n=implicit@wsip-70-167-153-251.oc.oc.cox.net)
16:53.12Kobazbut regular calls are fine
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16:53.36mrtelephoneis there a command similar to show g729 for ulaw?
16:53.51dlynes_laptopHi flauto
16:54.03dlynes_laptopGreat
16:54.05dlynes_laptopJust going through telephone hell right now, though:)
16:54.07dlynes_laptopGot a new install where every single phone rings in the whole shop on every incoming call
16:54.09dlynes_laptopAnyone run into this issue before?  [Nov  3 09:50:40] WARNING[13359]: channel.c:2317 __ast_read: Exception flag set on 'SIP/223-083a62a8', but no exception handler
16:54.12dlynes_laptopWhat's an exception handler in asterisk, and is there any documentation on how to implement one?
16:54.14dlynes_laptopKobaz: perhaps you didn't do a proper conversion to the format you're using?
16:54.16dlynes_laptopmrtelephone: no, because ulaw doesn't need to be licensed
16:54.49Kobazdlynes_laptop: k
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16:55.32Kobazhow do i know what format it's trying to use
16:55.33dlynes_laptopmrtelephone: all show g729 tells you is how many licensed channels are in use
16:56.03dlynes_laptopKobaz: generally, it'll try to use the same format as the channel that's being connected to it, is
16:56.24dlynes_laptopKobaz: so, if you connect with ulaw, it'll try to find a sound file that's in ulaw...otherwise asterisk will try to convert it, inline
16:56.25Kobazi'm playing a gsm to a ZAP
16:56.38dlynes_laptopKobaz: it's better to have it as a ulaw then, not gsm
16:56.42Kobazah okay
16:56.57Flautodlynes, i was talking to my wife last night after talking to you on the phone, my place is big enough to host you and your gf but it is not as nice as a hotel room, though, you are welcome to stay if you want to save a buck or two
16:57.01dlynes_laptopKobaz: zap channels use ulaw internally
16:57.07Kobazk
16:57.26dlynes_laptopFlauto: ok...i'll talk to julia about it...see what she says
16:57.33Flautookay
16:57.50Flautojulia is the cantonese girl?
16:57.53dlynes_laptopnod
16:58.54mrtelephoneflauto where do you live?
16:59.20mrtelephonedlynes, i just wanted to know of the call reverted to ulaw even though g729 is set priority on both sip devices
17:00.37dlynes_laptopmrtelephone: it could, depending on what codecs you've got defined, and what bandwidth setting you've decided on
17:01.00dlynes_laptopmrtelephone: bandwidth=high, would put a preference on ulaw; bandwidth=low would put a preference on g729
17:01.12mrtelephonei don't remember the bandwidth option
17:01.38dlynes_laptopmrtelephone: to see if the call reverted to ulaw, you can also do show channel <hit tab for completion>, and then select the channel you want to view
17:01.46dlynes_laptopmrtelephone: it'll tell you what codec is currently in use
17:01.52mrtelephonecool
17:02.03dlynes_laptopFlauto: Yeah, she's originally from Guangzhou
17:02.13dlynes_laptopFlauto: But people often think she's Filipino
17:02.27dlynes_laptopFlauto: she speaks Mandarin as well
17:02.33Kobazdlynes_laptop: what extension should the ulaw file be
17:02.41dlynes_laptopKobaz: .ulaw
17:02.53Kobazhmm, now it's complete static
17:02.58Kobazi did name it as .ulaw
17:03.00mrtelephoneyeah its using g729
17:03.01mrtelephonenice
17:03.02dlynes_laptopKobaz: ummm
17:03.11Kobazi converted it to u-law 2:1
17:03.12dlynes_laptopKobaz: renaming a gsm file to ulaw is not fixing the problem
17:03.15Kobazno no
17:03.17Kobazi converted it
17:03.28dlynes_laptopKobaz: converting a gsm file to ulaw is not fixing the problem, either
17:03.30Kobaz8000 hz mono
17:03.35dlynes_laptopKobaz: save it as a ulaw file, to begin with
17:03.36Kobazno no, i converted the wav to ulaw
17:03.39dlynes_laptopKobaz: ah
17:03.58dlynes_laptopKobaz: take a look at it in audacity, or something similar
17:04.08dlynes_laptopKobaz: see if perhaps your volume is peaking over the rms
17:04.19dlynes_laptopKobaz: if it is, that could be a good reason why it sounds like crap
17:04.40dlynes_laptopKobaz: http://audacity.sf.net/
17:05.06dlynes_laptopthere's versions for Windows and Linux
17:05.33Kobazyeah i already apt-got it
17:05.39Kobazoh
17:05.39Kobazhaha
17:05.44Kobazyeah
17:05.47Kobazit's waaaaay over
17:07.08Kobazokay so
17:07.12dlynes_laptopKobaz: so it probably sounded like crap as gsm then, too
17:07.16Flautonice
17:07.19Kobazsaving the file from audacity, what header should i use
17:07.23Flautoshe can talk to my wife in cantonese
17:07.34dlynes_laptopFlauto: ah...your wife is cantonese?
17:07.39Flautoyes
17:07.43dlynes_laptopFlauto: ah
17:07.48Flautohehe
17:07.53dlynes_laptopFlauto: does she speak a dialect, too?
17:08.13Flautoshe is from zhuhai, guangdong
17:08.28Flautoi dont' know if she does speak another dialect or just cantonese
17:08.30dlynes_laptopYeah...don't think that's where Julia's from
17:08.46Flautothe point is that i dont' understand any of them
17:08.48dlynes_laptopJulia's from what used to be the countryside...it's now right by the new Guangzhou airport
17:08.50Kobazdlynes_laptop: okay now i hear frogs
17:09.02mrtelephoneribbit
17:09.13Flautookay
17:09.13Kobazno no like, a deep groaning frog
17:09.17Flautoi know where it is
17:12.05dlynes_laptopKobaz: you mean the voice is kinda warbling?
17:12.18dlynes_laptopKobaz: i.e. it's still distinguishable that it's human voice, but it sounds like crap?
17:12.38Kobazno i just hear like a groan
17:12.53dlynes_laptopKobaz: oh, sorry...that was me...let me put the old sound file back....
17:13.03Kobazheh
17:13.11Kobazwhat header should i use?
17:13.12Kobazraw?
17:13.24dlynes_laptopKobaz: for wav?
17:13.37Kobazfor the ulaw
17:13.46dlynes_laptopKobaz:  Yeah, probably
17:13.52dlynes_laptopKobaz: sox will work, too
17:14.16dlynes_laptopKobaz: normally i just save it as a wav file in audacity, and then convert it using sox
17:14.31dlynes_laptopKobaz: keep in mind that the regular microsoft wav file format is wav49, not wav
17:14.41Kobazyeah
17:14.49dlynes_laptopwell, in digium speak
17:14.55dlynes_laptopin normal language, it's a riff wav file
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17:20.17dlynes_laptopFlauto: she'll talk to me about when I get home
17:20.26dlynes_laptopFlauto: but we already have a hotel booked for the first three days
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17:20.33dlynes_laptopCunningPike: good morning, anthony
17:20.49CunningPikeMorning!
17:23.30ManxPower1.4 has a built in utility to transcode files.
17:24.17Flautothat is okay
17:24.39Flautoi can hang out with you guys for sure
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17:28.25dlynes_laptopFlauto: yeah...for the most part, we'll probably be going to check out some apartment buildings
17:28.51dlynes_laptopFlauto: we're wanting to put in an offer on at least one building while we're there
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17:29.06Kobazdlynes_laptop: yay.... it was the sample rate
17:29.12Kobazdlynes_laptop: i forgot to set it to 8000
17:29.17dlynes_laptopKobaz: ah...haha
17:30.26Kobazsexy
17:32.44Flautoyou are buying a building?
17:33.01dlynes_laptopFlauto: hoping to get some income properties, yeah
17:33.11Flautookay
17:33.12dlynes_laptopFlauto: no such thing as an income producing property in Vancouver
17:33.22dlynes_laptopFlauto: It's almost as bad as california
17:33.38dlynes_laptopFlauto: the cap rate here is less than the cost of inflation
17:33.40Flautoi dont know if chicago is a good place either
17:34.06dlynes_laptopFlauto: a lot of the buildings there are 9% or higher cap rates
17:34.16dlynes_laptopFlauto: here, you're lucky to see 3%
17:34.23Flautoreally
17:34.29Flautogood to know
17:34.34dlynes_laptopFlauto: yeah...most are around 1 to 2%
17:34.38Flautoyou are buying or your gf is buying
17:34.49dlynes_laptopFlauto: and california has a lot of negative cap rates
17:34.58dlynes_laptopFlauto: we're doing it together
17:35.03Flautonice
17:35.18dlynes_laptopFlauto: she's got the good credit rating, but I'm able to do the research and that kinda thing
17:35.37dlynes_laptopFlauto: currently building a database for it, too
17:35.45Flautohehe
17:36.09dlynes_laptopFlauto: otherwise i'm not going to know how long the properties have been sitting on the market for
17:36.17Flautowhen you get here, you can teach me more about linxu
17:36.38dlynes_laptopok
17:36.42Flautothanks
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17:36.52Flautoi will talk to you later
17:36.57Flautoi have your phone number
17:36.58Flautoand email
17:37.07dlynes_laptopok
17:37.12dlynes_laptophave a good weekend
17:37.23dlynes_laptopremember to set your clock back an hour tonight
17:37.51dlynes_laptopchicago's on CST, right?
17:38.22Kobaztonight... back... oh no
17:41.00TJNIIBah.... I can get my sip traffic going around one nat, 2 not so much.
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17:48.18Kobaz<PROTECTED>
17:48.19Kobazhmm
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17:48.45cspotdlynes_laptop: yes
17:49.34*** join/#asterisk WindBack (n=jorge@200.117.115.188)
17:50.08phacehi all... I need one information. I have followed several guides about the Cisco phones and Asterisk. Now after changing lets say the username from 3000 to 3001 it doesnt read the changes. I can see that it downloads the new config file from the TFTP server but it seems like it doesnt apply it.
17:50.39WindBackI have to buy 10 ATA. Can you recomendme a mark
17:50.51WindBackwho work well with *
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17:51.06linxroutedd
17:51.36phaceWindBack: Try linksys, they have good models :)
17:52.04WindBackphace, for example spa2002
17:52.14WindBack(sipura)
17:52.59phaceWindBack: yes
17:53.13WindBackphace, thank
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17:53.48WindBackphace, Do you know the Grandstream mark?
17:54.11phaceWindBack, well I have only used Cisco and Linksys (ATA devices).
17:54.30WindBackphace, ok
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18:00.06linxroute.
18:21.23mrtelephonei bought one of those linksys 8 port jobs but i havn't had time to try it yet
18:24.55dlynes_laptopWindBack: pap2's work just fine
18:25.09dlynes_laptopWindBack: grandstream's also known as grandsuck
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18:25.23WindBackdlynes_laptop, thank, I supose it
18:26.04dlynes_laptopWindBack: grandstreams in general are quite bad...I've only used the grandstream budgetone 102's, but a few people in here have used the ata486(?) as well
18:26.12dlynes_laptopWindBack: i'm always hearing how bad they suck
18:26.40ManxPower~gs
18:26.41jbotGrandSuck phones & gateways are cheap junk which should be avoided with extreme prejudice.
18:27.33WindBackdlynes_laptop, I heard something that pap2 only work with vonage
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18:28.16ManxPowermany people say "PAP2" when they mean "PAP2NA", which is the non-locked version
18:28.28ManxPowervonage moved to using motorola boxes a while ago.
18:29.15WindBackManxPower, ahhh, ok
18:30.31WindBackManxPower, are there something like PAP2NA with more FXS ports ?
18:30.32coppiceI think PAP2NA means North America
18:30.48orkidyeah. there is also EU
18:31.10coppiceand SG, though SG is also sold in HK :-\
18:34.09ManxPowerWindBack: once you go above 2-ports, the cost per port tends to go up, not down as those tend to me more business oriented products where the 2-port ones are more consumer products.
18:35.30phacehi all... how to create a trunk between two asterisknow gateways ?
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18:36.28WindBackManxPower, ok, thank for your advice
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18:38.05WindBackManxPower, I
18:38.26WindBackManxPower, is very usefull for me this recomendation
18:38.29dlynes_laptopWindBack: that's the pap2 regular (and it works with whatever provider linksys has struck a deal with)...but the pap2-na is unlocked
18:38.53WindBackdlynes_laptop, yea
18:39.04dlynes_laptopWindBack: it's exactly the same device, though
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18:39.42dlynes_laptopWindBack: are you wanting cheap, or good?
18:39.56WindBackdlynes_laptop, good
18:40.06WindBackdlynes_laptop, good and cheap ;)
18:40.55dlynes_laptopWindBack: try going with a 4 port audiocodes ata, or a 4 port sangoma a102u or a102d, or a 4 port digium 400p
18:41.17dlynes_laptopWindBack: the sangoma and digium cards are your best bets, unless you want to remain open to trying other telephony platforms
18:41.26dlynes_laptopWindBack: then you might want to consider the audiocodes instead
18:41.43dlynes_laptopWindBack: also with the audiocodes, you can go after an embedded system without a pci bus, too
18:42.17dlynes_laptopWindBack: you can also try the xorcom box, too...it's a usb breakout box for fxo, fxs, door switch, ...
18:42.30coppicesomeone was complaining heavily about audiocodes boxes here a few days ago
18:42.56coppicei've never used their boxes, but the audiocodes silicon and software in other people's boxes is very good
18:43.43WindBackdlynes_laptop, Digium 400p is the tdm 400p?
18:43.48dlynes_laptopWindBack: yes
18:44.51WindBackdlynes_laptop, yea, but in my server I already have a tdm400p with 4 fxo and I heard that isn't good to have two
18:44.53ManxPowermost complaints about Audiocodes that I have heard was about it being hard to configure.
18:45.29coppicethis guy was complaining about quirky audio
18:45.43mrtelephonehow come there is a company that can do asterisk clustering and yet we can't with the open source? what is the company doing?
18:45.52dlynes_laptopWindBack: oh...no idea...try going with an a200d then, with multiple remora daughterboards, instead
18:45.57dlynes_laptopWindBack: or go with a tdm2400p
18:46.15dlynes_laptopmrtelephone: you can do clustering with the opensource version too
18:46.24mrtelephonehow do you centralize your voicemail?
18:46.38dlynes_laptopmrtelephone: imap?
18:46.59mrtelephonenot supported in asterisk?
18:47.01dlynes_laptopmrtelephone: odbc?
18:47.12coppiceNFS
18:47.17dlynes_laptopmrtelephone: it's all supported...just a matter of how buggy you want it
18:47.40mrtelephoneI want to do odbc but I don't see much posts about it
18:47.42dlynes_laptopnfs, coda, smbfs, afs are all options, too
18:47.55mrtelephonetrue
18:48.15mrtelephonedoes asterisk still crash if you write a voicemail to a broken nfs
18:48.30dlynes_laptopin recent versions of the 2.4 kernel and in the 2.6 kernel, the smbfs support is pretty stable now, too
18:48.48mrtelephoneI always had trouble relinking nfs after a crash
18:48.53mrtelephonenever really looked into it
18:49.03mrtelephonesmbfs is supported in the kernel now? thats cool
18:49.18coppiceit probably crashes to exactly the same extent as the commercial clustered offering :-)
18:49.18dlynes_laptopmrtelephone: it's been supported in the kernel since what?  linux 2.0?
18:49.38mrtelephoneI forgot about filesystem mounts
18:49.42dlynes_laptopmrtelephone: smbfs has been in the kernel for as long as I can remember
18:49.57dlynes_laptopmrtelephone: it just wasn't stable until later versions of linux 2.4
18:50.02TJNIISo is STUN only used in initial IP discovery, or is it actively used during the call as well?
18:50.03mrtelephoneI really like that odbc and then do mysql replication between the two machines
18:50.12mrtelephoneTJNII, initial
18:50.14coppicesince my memory fads after a few days, I'd definitely agree with that
18:50.15dlynes_laptopmrtelephone: the mounts would mysteriously hang or go hidden for no reason before
18:50.25WindBackdlynes_laptop, Why you toldme that I need audiocodecs
18:50.26WindBack?
18:50.30mrtelephoneevery call should STUN itself
18:50.47dlynes_laptopWindBack: i gave you options...it's up to you to pick the option that's best for yourself
18:50.47TJNIImrtelephone: Okay.  So really theres no good reason why I shouldn't point my phone at, say, stun.ekiga.net
18:50.48mrtelephoneim using 2.6.22
18:51.10mrtelephonetjnii, use any stun server you want
18:51.23mrtelephonebut if someone doesn't want you using their stun they might filter your ip one day
18:51.27dlynes_laptopWindBack: also, coppice said he heard one person complaining up and down about audiocodes the other day...he didn't say he heard everyone complaining about it
18:51.34mrtelephonethey'd be like, who the hell is stunning me and not even using my proxies :(
18:51.53mrtelephonestick with polycom cisco or linksys for voip products
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18:52.35mrtelephonecisco doesn't use sip as much as it should though
18:52.43coppiceI've used various bits of kit with the audiocodes silicon and software in it, and they are usually amongst the least troublesome units
18:52.45mrtelephoneI wish their wirless handheld would use it
18:52.58TJNIIWell, I'd start my own STUN server but it requires two IPs and I only have one.
18:53.21mrtelephoneTJNII, use someone elses
18:53.39coppicethere are so many public STUN servers out there, I see no reason to add more. the load you cause by using one is soooo light
18:53.50TJNIIRight, hence my first question
18:54.00mrtelephoneit just sends a packet to a remote host and the remote host returns a packet saying your ip is whatever so that the sip client sends it out in the messages
18:55.03coppiceyou say to the stun server "who do I appear to be in the outside world" and it tells you. end of transaction
18:56.13dlynes_laptopAnyone happen to know what exception flags are in asterisk, or how to avoid them?
18:57.02ManxPower"exception flags"????
18:57.09TJNIIcoppice, mrtelephone: Okay.  That's what I gleaned off of many web pages, I just wanted to be sure.  Thanks.
18:57.46dlynes_laptopManxPower: [Nov  3 11:24:19] WARNING[13359] channel.c: Exception flag set on 'SIP/229-b6831670', but no exception handler
18:57.53ManxPowerSince Asterisk's nat=yes basically determines the public IP and port number on it's own, you generally don't need configure STUN on SIP clients connecting to Asterisk.
18:58.09ManxPowerdlynes_laptop: never heard of it.
18:58.30dlynes_laptopManxPower: the number in square brackets after the 'WARNING' text is the line number in channel.c, right?
18:58.43ManxPowerdlynes_laptop: should be.
18:59.02ManxPowerbut I guess it could also be the PID of the process.
18:59.02dlynes_laptopManxPower: ok...i'm getting this coming up from five different line numbers, then
18:59.15ManxPowerdlynes_laptop: must be 1.4 specific as I've never seen that in 1.2
19:00.15dlynes_laptopManxPower: must be the pid, or the tid, or something...it's not the line number, anyways
19:01.17coppiceManxPower: if you want peers to talk to each other, and they are NATed, * can't help you. STUN can.
19:01.41TJNIIManxPower: I'm running into issues due to two nats.  I'm going to try STUN, otherwise I need to reconfigure my network.
19:01.59dlynes_laptopManxPower: well, it was in 1.2.0, too
19:02.33mrtelephoneasterisk acts like a stun doesn't it when y ou specify nat=yes?
19:03.33*** join/#asterisk asdx (n=diego@adsl-159-70.click.com.py)
19:03.41asdxhi
19:03.48asdx~wiki
19:03.59mrtelephonetjnii, what kind of router do you have?
19:04.11asdxis the wiki a good place to start?
19:04.18mrtelephoneyeah
19:04.20coppicemrtelephone: nat=yes is nothing to do with STUN
19:04.34mrtelephonevery good retard
19:04.43mrtelephonebut it uses the outside ip of the client
19:04.44TJNIIOne NAT is an actiontek, the other is going to be a netgear.
19:04.46mrtelephoneto return messages
19:05.59coppicebut it doesn't help with you want to let two peers behind NAT talk to each other. it only lets a NATed client talk to a public server
19:06.46mrtelephonei din't know you had a choice
19:06.54*** join/#asterisk CunningPike (n=CunningP@S010600095b33697f.vc.shawcable.net)
19:07.01mrtelephonetwo clients behind the same nat still have to go through the asterisk sever
19:07.04mrtelephoneserver
19:07.12mrtelephoneor not?
19:07.41coppicetwo clients behind different NATs is a more interesting case. STUN will support that
19:08.57*** join/#asterisk Cherebrum (n=jgarland@pdpc/supporter/base/Cherebrum)
19:10.02mrtelephonewith reinvites?
19:10.03*** join/#asterisk badcfe (i=christia@alltid.dritings.no)
19:10.30coppiceor with initial invitation to go directly peer to peer
19:11.02coppicein the general case there is no reason for the server to ever be involved in the media
19:11.40mrtelephonea peer to peer call you would have to contact a STUN server
19:11.56mrtelephonedoes the STUN server reply with the outside router port as well?
19:12.21Cherebrumew.. yea. a re-INVITE at the start of the call is UGLY. You almost always get a little hiccup in the audio and when the person answers the call you hear "lo?" instead of "Hello?"
19:12.54coppicemrtelephone: that is all it reports back
19:13.37mrtelephonecherebrum, thanks for informing me. I always wondered why first words were getting cut off
19:13.43*** join/#asterisk nighty^ (n=nighty@sushi.rural-networks.com)
19:13.55mrtelephonebut how do you fix it...
19:14.11mrtelephonecoppice, so it reports the port and ip?
19:14.40mrtelephonethen the client sends that in the sip message but if the client is set to receive on the inside port what happens then
19:14.45mrtelephoneits complicated
19:15.00coppiceThe STUN server gets a UDP request from some IP/port combination, and sends back to the combination what it is.
19:15.13Cherebrummrtelephone: you can not re-invite the audio, which sucks. Or you can convince someone to fix Asterisk so it doesn't touch the audio. Or you can use different software. ;)
19:16.42mrtelephonei thought the rtp gets sent once the clients are configured
19:16.47mrtelephonedoesn't that make sense?
19:19.13mrtelephonecherebum, don't you find that intermitten?
19:19.28coppicewith reinvites they are configured, and then reconfigured. if the reconfigure isn't fast enough the audio does odd things when people are listening
19:20.02mrtelephoneits the consumer ata's that get chopped all the time
19:20.06mrtelephoneI guess it depends on the latency
19:20.37mrtelephonebut there are no reinvites allowed on the remote atas
19:21.01mrtelephonethe rtp is being received before the ata is invited to the call
19:21.03mrtelephoneis that what happens?
19:22.51asdxi just got the asterisk 1.4.13 source, is that enought for starting?
19:23.00asdxim compiling on slackware now
19:23.14asdxenough*
19:23.20mrtelephoneyeah it should be
19:23.28asdxthanks
19:23.32mrtelephonelibpri
19:23.35mrtelephoneand zaptel
19:23.41mrtelephoneyou should compile those first
19:24.16asdxok
19:24.20asdxwhat are those?
19:24.30Kobazthings you may need
19:25.54asdxok
19:26.29Kobazif you're using it purely for voip, you don't need either, but if you have hardware you want to plug into asterisk, then you do
19:26.42asdxKobaz: i want to play with pure voip first
19:26.50Kobazthe just monkey with asterisk
19:26.57Kobazs/the/then
19:26.58asdxKobaz: ok
19:27.54Kobazyou might as well just install a distro package, since it's most likely gonna have all you'll need
19:27.56CherebrumPCI T1/E1 cards are lame anyways. Just use a voip gateway with proper hardware echo cancellers and DSP chips in it like an Audiocodes gateway or something.
19:27.59Kobazif you dont need hardware support
19:32.40mrtelephoneaudiocodes t1 gateway.. is that a media server?
19:32.44mrtelephoneI don't see their gateways
19:35.00asdxwhat is a good solution for voip <-> pstn without having access to hardware?
19:35.12mrtelephonevoip provider
19:35.18mrtelephonevoicepulse or something
19:35.37asdxbut i can use asterisk with that?
19:36.51*** part/#asterisk Cherebrum (n=jgarland@pdpc/supporter/base/Cherebrum)
19:37.45Kobazasdx: yeap
19:37.51mrtelephonehow can you get asterisk to hand off a call to a voip gateway
19:38.03Kobazasdx: it's like half a cent a minute roundabouts for voicepulse
19:38.07mrtelephoneor would you just pass the traffic through?
19:38.11asdxKobaz: nice
19:38.38Kobazmrtelephone: you need to make a trunk and a route to that trunk
19:39.45*** join/#asterisk macros73 (n=cs@c-24-3-246-27.hsd1.mn.comcast.net)
19:39.59mrtelephonebut the sip traffic will still go through asterisk?
19:40.06mrtelephoneand rtp
19:40.24mrtelephonemy sangoma pci card does a really good job though
19:40.48mrtelephoneI don't see the point in having another piece of hardware just for the pstn gateway
19:41.04Kobazmrtelephone: it tends to be more flexible and easier to get going
19:41.25Kobazit's like external modems vs internal modems
19:41.31Kobazthey both do the same stuff
19:42.11mrtelephonedoes cisco have a small router with t1 and sip support?
19:42.16mrtelephoneI guess you'd call it sip express
19:42.20Kobazcrisco
19:42.27mrtelephonethen you need voice modules
19:42.31mrtelephoneand dsp modules
19:42.59mrtelephonecisco with 1 t1, dsp modules, sip express.. probably pretty expensive?
19:43.17Kobazanything from crisco is going to be expensive
19:44.16mrtelephonei just don't like how everything has to work with cisco unity
19:46.34J4k3I'll make cisco work with rick james unity
19:46.41J4k3unity!
19:47.12*** join/#asterisk ghento (n=ghento@64.180.85.230)
19:48.40*** join/#asterisk DarkRift (i=dark@bas10-montreal02-1177584490.dsl.bell.ca)
19:49.27ManxPower~trunk
19:49.28jbotextra, extra, read all about it, trunk is a word with varying definitions.  In Asterisk, a trunk is a "stream of UDP packets containing IAX2 frames from more than 1 call"; in telecom, a trunk is a "single voice channel between two pieces of switching equipment."  There is no such thing as a "SIP Trunk" -- Don't use the term.
19:49.46ManxPower<PROTECTED>
19:50.03mrtelephoneI want an AS5350
19:50.31mrtelephonehow can you dis invent the term sip trunk?
19:50.57badcfedoes cisco SIP gateways support the obsoleted BYE "Also:" header?
19:50.59ManxPowerI can't, but I can try to keep people from using the term because it is wrong.
19:51.16ManxPowerYou don't correct people when they say "web" when they mean "internet"?
19:51.36mrtelephoneI see the definition and it makes sense
19:51.41ManxPowerI don't.
19:52.25mrtelephonebefore I read it i was thinking.. the asterisk developers, because they invented iax to be a "trunking" system that any other trunking protocol would be taboo
19:52.44ManxPowerwhat other trunking protocols are there?
19:52.48badcfeanyone knows wether the cisco SIP gateways support the obsoleted BYE "Also:" header?
19:53.18mrtelephoneI always though of a trunk as being a channel of information or an elephants nose
19:53.45mrtelephonebadcfe, you'd have to check ciscos website because some versions may use it and some not
19:53.45ManxPowerSIP does not qualify.  From Asterisk's point of view there is no difference between a SIP endpoint that can handle 1 call and looks like a phone, and a SIP provider that provides a gateway to the PSTN.
19:54.16ManxPowermrtelephone: so by your definition a phone would qualify as a "sip trunk"
19:55.08mrtelephoneI guess you could say that.. isn't an analog copper pair a trunk?
19:55.23ManxPower"trunk" was coined by the people that do Asterisk GUIs because they thought their users were too stupid to understand "peer"
19:55.35ManxPower(at least as it applies to Asterisk)
19:55.38badcfemrtelephone: but do cisco tend to purge their firware from functions as soon as its defined obsolete  in an RFC or does they tend to keep stuff?
19:55.49mrtelephoneyeah because most people associate trunk with access to the pstn
19:56.02ManxPowerbadcfe: The current Cisco IOS still supports SNA and Token Ring.  You do the math. 8-)
19:56.08badcfemrtelephone: cause i really want this "Also:" to stay forever actually
19:56.16mrtelephonethe extensions.conf have trunk variables at the top and if your trunk is SIP/123913 then thats your trunk
19:56.23*** join/#asterisk Hadi- (n=Hadi@CPE001310492769-CM001225e00576.cpe.net.cable.rogers.com)
19:56.26ManxPowerum, what extensions.conf?
19:56.27Hadi-hello everyone
19:56.31ManxPowerMine doesn't.
19:56.47mrtelephoneit was in the default config
19:57.26ManxPowerUm, TRUNK=Zap/g2
19:57.34mrtelephoneyeah
19:57.35ManxPowerno sip in there at all.
19:57.40mrtelephonewell it could be
19:57.41mrtelephonehahah
19:57.44ManxPowerand if you want to call a telco connection a trunk, I'm not going to complain
19:57.48*** join/#asterisk adorah (n=Michael@87.69.57.246.cable.012.net.il)
19:57.53mrtelephoneim not trying to argue with you..
19:58.04asdxok, seems like i can start asterisk already
19:58.05mrtelephoneI see your point I understand
19:58.17badcfei have "  ; trunketytrynk trynk TRUNK" in my extensions.conf
19:58.25mrtelephonebut like you said It's like correcting someone saying web instead of internet
19:58.46ManxPowerIf more people did that, there would be fewer people walking around sounding like internet morons.
19:58.47mrtelephonesome guy gave a channel a lecture once because people were referring to internet speed as "bandwidth"
19:59.13mrtelephonewhere bandwidth is the frequency range of rf transmissions amoung other things above my pay grade
19:59.27ManxPowerMy usual response to a report of "The internet is down!" is "Even in China?"
19:59.38mrtelephonehahaha
19:59.54mrtelephoneyeah well just cause we have an upper hand in technology doesn't make us better or right
19:59.56ManxPower"The internet" has never gone down.
20:00.01mrtelephoneone day the itnernet will crash and we will be stupid
20:00.07Kobazi broke the internet once
20:00.09Kobazbut then i fixed it
20:00.10mrtelephoneeveryone else will know how to write on paper
20:00.16mrtelephonebut we won't :(
20:00.19badcfeManxPower: maybe someone thinks that when the internet is down then its down in china too
20:00.20ManxPowermrtelephone: but we should correct obvious tech wrongness when we can.
20:00.28mrtelephonei'll try manxpower
20:00.47ManxPowerbadcfe: not a single person has as far as I can tell.
20:00.49badcfeManxPower: hey you may ask is the internet down or the Internet down?
20:00.58mrtelephonemanxpower.. we should make tshirts that have some kind of phrase.. correct the tech wrongness
20:01.25mrtelephoneyeah!
20:01.29mrtelephoneanyways i gotta go out to eat
20:01.30badcfeManxPower: internet refering to your local offices inernal web of cables.  haha i sais web
20:01.40coppiceI've seen the internet almost go down in China, when any .com stopped resolving for an afternoon
20:01.43mrtelephonesorry for callin you a retard coppice.. i was a little offended
20:02.02asdxwhat is a good cross-platform open source free softphone?
20:02.07*** join/#asterisk implicit (n=implicit@c-67-191-24-188.hsd1.fl.comcast.net)
20:02.11ManxPowerHa!  No, the local area network is called the LAN, not the internet.
20:02.15coppicewhat could offend an asshole like you?
20:02.15badcfeInternet IS down in china.  because on Internet you can access any other peer with any other proto!
20:02.18Kobazasdx: kiax
20:02.21*** part/#asterisk mrtelephone (n=MrTeleph@h697179-171.picriverisp.net)
20:02.27asdxKobaz: thanks
20:02.31*** join/#asterisk xheliox (n=jeff@pdpc/supporter/active/xheliox)
20:02.38ManxPowerkiax is not KDE specific?
20:02.41Kobaznope
20:02.47Kobazit actually doesn't even use kde
20:02.58badcfein china they have this chineese wall wich means they dont have Internet at all
20:02.59ManxPowerit runs on win32, mac, and *nix?
20:03.03Kobazit does
20:03.21coppicebadcfe: by that measure the internet is always down, as there always routes which don't work, often on purpose
20:03.37Kobazi just wish it used alsa instead of oss
20:05.06ManxPowerI can be pretty confident my users have no idea as to the state of the Chinese part of the Internet at any one point in time.
20:05.38badcfeInternet is serving peers with IP and letting they fiddle with any other peer and implementing any protocoll in the end-user hosts.  The value of IP is its stupidness as opposed to X.25 and so.  No one -- even the most MORON of burocrat -- will be able to take the freedom of Internet away.
20:06.29badcfethe Internet is the last place on earth wich is free
20:06.32coppicedunno. for example, a large part of the internet can never access that site in europe where chan_ss7 is hosted.
20:06.45ManxPowercoppice: why not?
20:07.02coppicerouting blocks a large part of the planet from it
20:07.33badcfecoppice: hello. i remember you from xbpnepo (reversed)
20:07.35ManxPowermaybe that will change when people in china start to secure their systems against being zombies.
20:08.03coppicewho mentioned china?
20:08.14ManxPowercoppice: *blink*  I assumed it.
20:08.30coppicenope. a lot of the world cannot access that site
20:08.41ManxPowerdoesn't sound very internetish to me. 8-)
20:09.06coppiceI can't access it from HK, but I downloaded the chan_ss7 code from a site in china, once
20:09.13badcfecoppice: well then its not on the Internet
20:09.46ManxPowerWhat really annoys me is sites with admins that thinking blocking all ICMP is a good idea.
20:10.13coppicethere is a lot of blacklisting around the world, must of probably from sheer incompetance
20:10.57coppicewell, blocking ICMP is probably one of the reasons so much stuff doesn't work, as it makes things so hard to figure out
20:11.58ManxPower*nod*  ICMP is required for MTU path discovery.  With all this PPPoE happening, MTU path discovery is pretty important.
20:12.00badcfewhy would people block ICMP?  wouldnt any scanner use some SYN on 80 if it was really evil?
20:12.21ManxPowerbadcfe: admins that are not very experienced with networking.
20:12.45badcfeclock ICMP is like saying "my thingy here is not secure so i hope you dont see it"
20:12.50orkidPPPoE sucks :P I don't like it.
20:12.56coppiceI find ICMP seldom works these days
20:12.59ManxPowerorkid: me neither.
20:13.10badcfe<PROTECTED>
20:13.54badcfeicmp echo is swallowed?  i didnt know this was the new trend
20:14.02tzangercoppice: I found and fixed the source of that odd audio problem I asked you about a couple of weeks ago
20:14.13ManxPowerSELECTIVELY blocking ICMP is fine.  Blocking things like ICMP Packet Too Big is bad.
20:14.35ManxPowertzanger: what was the issue?
20:14.44ManxPowerwell, what was the fix.
20:14.50coppicetzanger: so, what was it?
20:14.59badcfeManxPower: and icmp redirect is maybe not good to block?
20:15.15ManxPowerbadcfe: that would be a good assumption
20:15.29tzangerManxPower: I'd implemented elastic buffers to decouple the TDM clock from the TDMoE clock... but then I was an idiot and tied the "kick a new TDM transfer" operation to occur whenever I received my TDMoE packets.  :-)
20:15.48tzangerI also had my DMA buffer ping-poinging backwards... it was pong-pongings
20:15.50tzangerer pong-ponging
20:16.09tzangerinstead of processing data from one buffer and transfering to the other, it was doing both to the same buffer
20:16.49*** part/#asterisk WindBack (n=jorge@200.117.115.188)
20:17.04badcfecoppice: are you still in jp
20:17.31coppiceyou have the wrong person
20:17.56badcfeoh okay.  but i remember you from open pbx
20:18.10badcfeare you guilty?
20:18.43*** join/#asterisk linxroute (n=dfsf@117.0.26.118)
20:19.01coppicewe all have a lot of guilt to bear, deep down inside
20:19.11*** part/#asterisk Strom_M (n=strom@208.127.172.112)
20:19.32badcfei guess so
20:22.56*** join/#asterisk Strom_M (n=strom@208.127.172.112)
20:27.06[TK]D-Fendertzanger, that'd be pong-pinging :)
20:27.18tzangerno, ping-poing and pong-ping would both work
20:27.23tzangerping-ping or pong-pong just won't do
20:27.48[TK]D-FenderMy new blade has been commissioned :)
20:28.05tzangerwhat's that?
20:28.10asdxok i got kiax
20:28.16[TK]D-Fendertzanger, New custom katana
20:28.18tzangernew blade?  as in shaving?
20:28.19coppicehe's turning Japanese
20:28.19tzangerahh
20:28.21tzangerholy crap
20:28.37asdxis there a way to see from asterisk
20:28.40tzangerspeaking of which
20:28.42tzangerI need to shave
20:28.42asdxwhen a client is attempting to connect?
20:28.47ManxPowerJust wait until he gets into annime porn.
20:29.04asdxa log or something
20:29.08tzangernow that that noise issue's done I can push that out to 1200 zap channels
20:29.20ManxPowerasdx: sip debug | sip debug peer X | sip debug ip Y
20:29.35ManxPowersilly me.
20:29.39ManxPoweriax2 debug, of course
20:29.49[TK]D-Fendertzanger, modeled somewhat after this one : http://www.roninswords.com/sakura_kure.htm
20:29.53asdxManxPower: this client only says IAX Server, Username, Password, etc
20:30.05ManxPowerasdx: then I guess that is all you need.
20:30.13*** join/#asterisk syneus (n=syneus@host209-95-dynamic.1-79-r.retail.telecomitalia.it)
20:30.17ManxPowerI don't know what IAX2 debugging options there are.  I don't use IAX2 anymore.
20:30.31tzanger[TK]D-Fender: very ornate
20:30.41asdxManxPower: so i have to get a client that does support SIP?
20:30.46dlynes_laptopasdx: attempting to connect?
20:30.53asdxdlynes_laptop: yeah
20:30.54ManxPowerasdx: Huh?
20:31.05dlynes_laptopasdx: like iax?
20:31.20dlynes_laptopasdx: have you tried iax2 debug?
20:31.44asdxdlynes_laptop: not yet
20:31.47*** join/#asterisk ReDNeQ (i=ReDNeQ@cpe-66-68-37-190.austin.res.rr.com)
20:32.17dlynes_laptopasdx: have you tried putting noop()'s in your dialplan to show dialplan attempts?
20:32.19[TK]D-Fendertzanger, Using the same tsuba (handguard), different fittings, some custom paint work on the saya (scabbard), and so mods to the blade itself.
20:32.32dlynes_laptopasdx: have you tried any sort of debugging procedure, at all?
20:32.34asdxdlynes_laptop: no, this is the first time im using asterisk
20:32.39dlynes_laptopasdx: ah
20:32.58dlynes_laptopasdx: another thing you can try too, if you have a firewall set up, and you know how to configure it
20:32.59coppicetzanger: so you have a 1200 channel zaptel analogue card now? :-\
20:33.15asdxdlynes_laptop: ok
20:33.23dlynes_laptopasdx: is to set up logging whenever someone tries to connect to port 4569 udp
20:33.32dlynes_laptopasdx: don't deny the connection though...just log it
20:33.35tzangercoppice: no, I have four PBX chassis feeding 288 TDMoE channels each into a single PC
20:33.42asdxi only got the sip program/command, not iax2, i guess i should specify iax2 in compile time?
20:33.53tzangerit's a waste of a BF537 DSP, but that's what they're using
20:33.54asdxdlynes_laptop: ok, thanks :-)
20:33.57dlynes_laptopasdx: load chan_iax2.so
20:34.09dlynes_laptopasdx: or load => chan_iax2.so in your modules.conf file
20:34.16tzangercoppice: is echo can on an integer-only DSP that much more difficult than a floating-point one?
20:34.29coppiceblackfins are cheap. its hard to waste one
20:34.33tzangerhaha
20:34.47coppiceecho can is usually implemented in fixed point
20:34.58Corydon76-digI beg to differ.  I bet I can waste one with a 9mm pistol
20:35.10J4k3me too
20:35.49coppicetzanger: even the pentium software ones for fixed point, so they sit in the kernel well
20:35.54blitzragepffft.... Colt .45 represent
20:35.59tzangerthat is true
20:36.10tzangernever thought of that
20:36.23coppicepeople are actually designing MCUs into future 9mm bullets
20:36.27tzangerHPEC isn't a kernel module though, I don't know if I'd really trust the free zaptel ones
20:36.53*** join/#asterisk Dovid (n=Dovid@bzq-79-180-59-23.red.bezeqint.net)
20:36.56[TK]D-Fender.454 Cassull <- winner
20:37.00coppiceits not in the kernel? that's weird
20:37.13J4k3coppice: I make my own ammo.
20:37.15J4k3!!
20:37.22coppiceI think the octasic one is in the kernel
20:37.30J4k3I mean, its really *not* hard to cast up lead bullets.
20:37.32tzangerJ4k3: pretty soon you're gonna need a chip programmer :-)
20:37.37*** join/#asterisk agx (n=badpengu@81-174-45-186.dynamic.ngi.it)
20:37.40J4k3I should get an ROHS compliant sticker for the side of my luger
20:37.48J4k3tzanger: prolly not
20:37.50J4k3;)
20:37.57tzangerhahaha
20:37.59tzangerROHS
20:38.35coppicewith the stupidity of how RoHS is handled, I bet smart bullets with require lead free construction for the electronics
20:38.36asdxi just loaded chan_iax2.so but got tons of warnings messages
20:38.49J4k3coppice: nah, they'll insist on spent uranium.
20:38.50J4k3;)
20:38.54J4k3the gift that keeps on giving.
20:38.58asdx[Nov  3 17:37:58] WARNING[28896]: manager.c:2425 ast_manager_register_struct: Manager: Action 'IAXpeers' already registered
20:39.04J4k3heere in east texas its beer^H^H^H^Hdeer season
20:39.14J4k3which means we have drunk fuckers with eleplant guns
20:39.18J4k3err elephant
20:39.37J4k3one of which is my neighbor.
20:39.50coppicewhat kind of guns do elephants like to use>
20:42.37tzangercoppice: big ones is my guess
20:42.44tzangerthat reminds me of a joke
20:42.55tzangerhow do you know there are elephants fucking in your back yard?
20:43.01coppicetzanger: well, wherever HPEC sits I expect it is fixed point
20:43.12tzangerall your trash can bags are missing
20:43.28Dovidhahaha
20:43.32coppicewe have no back yard, so I know we have no elephants there
20:43.41tzangercoppice: pandas?
20:44.21coppiceits a bit warm here for pandas to be outside. the ones in Ocean Park are cooled the entire year
20:45.13badcfecoppice: you have fucking elephants in your back yard.  but their not there -- just as your yard
20:48.25*** join/#asterisk macros73 (n=cs@c-24-3-246-27.hsd1.pa.comcast.net)
20:58.06asdxdo i have to add SIP/IAX2 users in the asterisk server?
20:58.12asdxso my softphone will be able to connect
20:58.17*** join/#asterisk blq (n=Bl@dslb-088-067-019-189.pools.arcor-ip.net)
21:00.56*** join/#asterisk CunningPike_ (n=CunningP@S010600095b33697f.vc.shawcable.net)
21:01.32*** join/#asterisk ZaVoid (n=zavoid@c-67-165-25-195.hsd1.ct.comcast.net)
21:03.35[TK]D-Fenderasdx, if you want * to DO anything with them you'll kindof HAVE TO.
21:03.51orakleheh
21:04.22[TK]D-Fenderasdx, or ask the "Cooperation Faerie" for some magic dust
21:05.20asdx[TK]D-Fender: ok, i know i have to
21:05.33asdx[TK]D-Fender: i just don't know how to do everything yet
21:06.35linxroutehave anyone here tried T.38 passthrou with * 1.4 ?
21:14.43*** join/#asterisk TedNJ38 (n=HungLad@ool-4573adc7.dyn.optonline.net)
21:14.49TedNJ38I need help please...  I have a problem.  I am getting an error and I can't start my box...  loader.c:  /usr/lib/asterisk/module/format_mp3.so:  undefined symbol:  ast_module_register  How can I fix that?
21:16.53tzafrir_homeTedNJ38, you probably rebuilt asterisk and have not rebuilt format_mp3 (or asterisk-addons)
21:17.42tzafrir_hometo cure the symptom: add the line "unload => format_mp3.so" to /etc/asterisk/modules.conf
21:18.15tzafrir_homewhere do you have asterisk installed from? and asterisk-addons ?
21:18.40TedNJ38I just typed yum -y update and then my box stopped working.
21:18.56tzafrir_homeTedNJ38, trixbox?
21:19.19TedNJ38Yes.
21:19.21tzafrir_homedo you happen to have asterisk installed from source as well?
21:19.41TedNJ38No.
21:19.56TedNJ38Oh brother, now I got another module res_clioriginate.so
21:20.05tzafrir_homethat's easy to check:
21:20.30tzafrir_homerpm -qf /usr/lib/asterisk/moudles/format_mp3.so
21:20.56TedNJ38I get asterisk-addons-1.2.7_1.2.21.1-4
21:22.05tzafrir_homerpm -q asterisk
21:22.43linxroutesend DTMF during sip call
21:22.53TedNJ38asterisk -1.2.24-43.79171
21:23.01linxrouteis there something like senddtmf=yes
21:23.05linxroutein sip.conf ?
21:26.00[TK]D-Fender~trixbox
21:26.01jbotit has been said that trixbox is a full linux distro that includes FreePBX, and other 3rd party add-ons. It is all this extra stuff which makes trixbox seriously painful to support and hence you will find little help here for it.  Try asking in #trixbox or on their forums & WIKI at http://www.trixbox.org
21:26.41blitzragethat description is much nicer than I remember it :)
21:27.39[TK]D-Fenderblitzrage, You're welcome :)
21:27.48blitzrage:D
21:28.26[TK]D-Fenderblitzrage, The majority of these things are now either written, or re-written by me :)
21:28.49blitzrageyou tend to use them a lot, so that makes sense :)
21:29.57[TK]D-Fenderblitzrage, Who wants to spew out the same long spiel verbatim to every new schmuck who comes in here? :)
21:30.07blitzrageoh I agree
21:36.25CunningPike~wglwat
21:36.25jbotwell, wglwat is well, good luck with all that
21:36.35CunningPikeOne of mine :)
21:36.59*** join/#asterisk yannj_fr (n=yannj_fr@vpn.intelunix.fr)
21:37.04[TK]D-FenderCunningPike, It needed no enhancement.  Good one :)
21:37.08CunningPikelol
21:37.34CunningPikeI had something in there describing AAH as the Microsoft BOB of PBXs - I think it's gone though
21:37.41CunningPikeOne of my finer moments ;)
21:40.41linxroutesend dtmf during sip call, anyone ?
21:41.31asdxhow can i add users?
21:42.29CunningPikelinxroute: Push the keys
21:42.46linxroutewell
21:43.06linxrouteseems like it's does not work during the call
21:43.25*** join/#asterisk MrTelephone (n=test@S010600195b946fa3.ls.shawcable.net)
21:43.51linxroutei saw somewhere they had like senddtmf or something in sip.conf
21:43.59linxroutenot sure if there's such a thing
21:45.59MrTelephonedtmfmode
21:46.15MrTelephoneif you want to send dtmf then look at the dial cmd in extensions.conf
21:47.02MrTelephoneor senddigit maybe
21:47.02linxrouteyes i know
21:47.08linxroutebut during a call
21:47.28linxrouteyou want to do like transfer or other feartures
21:48.08linxroutepressing for exp 1* to do a transfer
21:48.50MrTelephonedon't you do it form the phone?
21:49.10linxroutewell with an ata -> analog phone
21:50.07MrTelephoneso asterisk is not detecting the dtmf from the analog phone?
21:50.33linxroutethe IP phone does not
21:50.40linxroutethe analog does
21:50.44linxroutethat's funny
21:50.49MrTelephonewhat kind of analog phone?
21:50.51MrTelephonei mean
21:50.54MrTelephonewhat kind of sip phone
21:51.03MrTelephoneset dtmfmode=inband in sip.conf
21:51.13MrTelephoneif that doesn't work set it to rfc2833
21:51.17MrTelephoneif that doesn't work wtf
21:51.39linxroutei used  rfc2833
21:51.47linxroutethe ATA is linksys pap2
21:52.02MrTelephonedo sip debug and find out if the sip phone is sending telephone-event in its sip message
21:52.17MrTelephoneswitch it to inband for the sip phone
21:52.52[TK]D-Fenderlinxroute, you shouldn't be using DTMF for transfer,etc on thata ATA.  It has its own means of doing those
21:53.11linxroutewell :)
21:53.23linxroutejust trying to get to understand more about asterisk
21:53.34[TK]D-Fenderasdx, vi, vim, emacs, gedit, kwrite, OOo Writer, mc, nano, pico, etc....
21:54.14[TK]D-Fenderlinxroute, You need to use certain channel variables & Dial options to allow DTMF transfers and naturally I don't trust for a second that you have done so properly.
21:55.24linxroutewell
21:55.24linxroutethat's why i'm trying
21:55.40linxroutewe soon will have a project that bring telephone to a remote area
21:55.48linxrouteusing wimax
21:55.48[TK]D-Fenderlinxroute, next, you want to know how somethign works, or why it ISN'T working : PASTEBIN IT <-
21:56.04asdx[TK]D-Fender: yeah, i personally like vim :-)
21:56.20[TK]D-Fenderasdx, More power to you then.  go add some users.
21:56.28asdx[TK]D-Fender: ok
21:57.10linxrouteethnic people will be using voip :) as wimax dont need line of sign
22:00.13MrTelephoneif your going to have more than 1000 customers don't use asterisk
22:00.15*** join/#asterisk L2SHO_ (i=adam@static-host-24-149-138-156.patmedia.net)
22:00.17MrTelephoneor even more than 500
22:00.58linxroutethey are not really customer
22:01.02L2SHO_ok, so I've got outgoing calls working, but when I try to make and inbound call to my box, my provider is sending me the call, but my box is replying with 403 Forbidden
22:01.10[TK]D-Fenderlinxroute, USERS <-
22:01.28linxroutewe just try to bring telephone to those ethnic peoples
22:01.39linxroutesince they can get anymean of communication
22:01.45linxroutein such remote are
22:02.00linxrouteand they will be aroud 20 to 50 max
22:02.14[TK]D-Fenderlinxroute, thats nice, just that * should not be the main proxy for this
22:02.53L2SHO_but if I use X-Lite, it works perfectly fine
22:03.16*** join/#asterisk ReDNeQ (i=ReDNeQ@cpe-66-68-37-190.austin.res.rr.com)
22:03.28linxrouteyeap, hope one day it will be able to
22:04.03linxrouteNTT from japan have deploy a very large number of user
22:04.23linxroutethey must have made some changes
22:04.35*** part/#asterisk MrTelephone (n=test@S010600195b946fa3.ls.shawcable.net)
22:08.00linxroutebb all
22:10.24*** join/#asterisk watchy (n=watchy@c-68-51-54-72.hsd1.ar.comcast.net)
22:10.37watchyif my T1 card isn't pluged in can it still be a timing master?
22:10.59watchyor does it need to be pluged in to get timing?
22:25.09J4k3so...  my next project for around the house is a truely portable voip phone
22:25.20J4k3the prototype will use a gs bt101 based ;)
22:25.37J4k3err, will be gs bt101 based
22:25.47*** join/#asterisk IPetrov (i=IPetrov@ppp91-76-143-226.pppoe.mtu-net.ru)
22:27.52J4k3prolly just build a small slab to velcro/screw to the bottom of the bt101
22:28.19J4k3containing a routerboard 133, a minipci card, a battery and whatever battery circuit I'm going to get stuck using.
22:28.22J4k3:|
22:31.47tzafrir_homewatchy, it's technically not the T1 card, but rather a span in the card. If you have a single-port card ignore tha...
22:33.06tzafrir_homewatchy, when it gets sync it will try to take mastership. when it loses sync it will give up mastership
22:33.23tzafrir_homebut before it was ever synced: I think it will be just like any analog card
22:33.26tzafrir_homenot sure, though
22:34.36watchytzafrir: so untill its pluged in its not synced anyways
22:34.37watchycorrect?
22:34.56watchyit has to be up and functioning to be synced and pluged into the PRI
22:36.02asdxwhen i add users and stuff, is there a way to validate that, or try from asterisk directly without a softphone, so i can say "yeah, it works"
22:36.22[TK]D-Fenderasdx, once you've done one, you may as well have done 100
22:37.01asdxok
22:39.13asdx[TK]D-Fender: i like to edit configuration files, i just wanted to know if there is a way to connect/validate from cli.
22:39.31*** join/#asterisk Lann (n=spam@c-71-198-197-49.hsd1.ca.comcast.net)
22:39.48asdxerr, authenticate*
22:39.49Lannhey...where can I find a good softphone to test my asterisk with?
22:39.53*** join/#asterisk Dovid (n=Dovid@bzq-79-180-59-23.red.bezeqint.net)
22:40.01watchytk: does a T1 card to be pluged in to be synced?
22:40.02asdxLann: i saw a few in the wiki
22:40.11Lannwhich one works well?
22:40.28Lanni'm just wanting to test my dial-in diaplan, this isnt for like an office or anything
22:40.33L2SHO_Lann: I have not problems with X-Lite
22:40.48L2SHO_no*
22:41.10asdxs/validate/authenticate/g
22:43.43*** join/#asterisk Siya (n=djerk@217-195-248-251.dsl.easynet.nl)
22:43.47Siyaello
22:44.20Siyaanyone here who can point me to a simple tool to visiulise * cdr data?
22:44.58Siyatrying to set up cdrtool but I'm not getting very far and unsure if it will actually work on data from asterisk alone...
22:45.52Lanni'm so confused about how to get x-lite to call my asterisk box, my asterisk box is on the lan on a certain IP...i just want to call that IP and go to an extension
22:46.13*** join/#asterisk DarkRift (i=dark@bas10-montreal02-1177584436.dsl.bell.ca)
22:47.51Siyaanyone here using cdrtool?
22:59.33*** join/#asterisk Dovid[Laptop] (n=Dovid@bzq-79-180-59-23.red.bezeqint.net)
23:13.36*** join/#asterisk JT (n=j@unaffiliated/jt)
23:16.19*** join/#asterisk felix_da_catz (n=felix@c-76-31-59-80.hsd1.tx.comcast.net)
23:22.58*** join/#asterisk codeshah (n=codeshah@S01060011092d0063.ed.shawcable.net)
23:26.19*** join/#asterisk iphonecan (n=adsa@S0106001346face5f.ed.shawcable.net)
23:26.28*** join/#asterisk dseeb_ (n=dcb@CPE-124-187-252-4.vic.bigpond.net.au)
23:26.35iphonecananybody up
23:26.59J4k3iphonecan: you got a working sip client for the iphone?
23:27.18iphonecanthat would be nice
23:28.05*** join/#asterisk sakic (n=sakic@cpe-071-075-175-140.carolina.res.rr.com)
23:28.13iphonecanI have a question i have 4 office around north america , i would like to know what would be the idea situation. Have multiple asterisk boxes in each site
23:28.18iphonecanthere is only like 5 -10 users
23:28.39J4k3depends
23:28.40Dovidy would u want multiple at each site ?
23:28.44iphonecanin each office, or do i have one main server and have gateways on each line cause i would like local numbers
23:28.46J4k3do your users need to be able to call each other at the same site?
23:28.51J4k3under downtime conditions?
23:28.52iphonecanyea
23:28.57J4k3then you need asterisk boxes at each site
23:29.06sakicswitchbox simple and easy?
23:29.22iphonecanusers should be available to call to each other and be available to make long distance for free
23:29.32iphonecanwhat about trixbox
23:29.35J4k3sakic: bleh, just another trixbox mess, its just this one digium owns so its magically supported in here *eye roll*
23:29.54iphonecan=)
23:30.02J4k3I've had a decent experience with trixbox, but it has a lot of funkyness.
23:30.13J4k3for a simple office its plenty powerful enough and easy to configure, thats my opinion on it.
23:30.14iphonecanso i should look at switchbox
23:30.20J4k3but, if you want more than it can do easily, go for it.
23:30.23sakicI got someone to install my asterisk, but they did it w/ files and I have no clue how to modify it
23:30.25sakicneed a gui
23:30.35Lanncan someone help me configure x-lite to actually begin executing my [default] dialplan in asterisk? so i can test some things...
23:30.41J4k3I'd take a look at switchvox, trixbox, and there are a couple more
23:30.57iphonecanI would like the hardword applience too
23:31.17J4k3iphonecan: sounds like you have money to burn :)
23:31.21sakicswitchvox install include the os?
23:31.39iphonecanand the hardware
23:32.35iphonecanAsterisk Appliance what about that
23:34.56tzafrir_homeyes, but from the description there it looks like a cripleware
23:35.05sakicI am going to try it tomorrow, wipe out my current
23:35.16iphonecanJ4k3: So what should i look at switchvox
23:35.59J4k3iphonecan: look at everything and decide what sucks least
23:36.01J4k3for your situation
23:36.16sakiclol
23:37.08iphonecanwell i know but i am new to this
23:37.16iphonecani want to know from experience
23:37.37J4k3from experience I'd say don't use any of these and learn it all from scratch
23:37.47J4k3if you have any unix admin experience at all
23:37.57*** join/#asterisk dseeb_ (n=dcb@CPE-124-187-252-4.vic.bigpond.net.au)
23:38.09J4k3but, that can be a hard learning curve because you learn how sip works at the same time, and that can make for a very annoying experience.
23:38.26*** join/#asterisk yannj_fr (n=yannj@vpn.intelunix.fr)
23:39.06iphonecanI don have linux admin, I have done asterisk from scratch but i dont want to do it, i want to run the business
23:39.07iphonecan=)
23:39.34J4k3haha yeah
23:40.06J4k3if I was doing it over I'd check out switchvox first, then trixbox, asterisknow is another option...
23:40.18iphonecanthank you
23:41.44tzafrir_homeswitchvox is not free software. Anything with licensing overhead is a pain in the long run. At least that is my experince
23:42.35tzafrir_homeAnd speaking about that, there are now two completely different distributions called "trixbox":
23:42.49watchytzafrir: we are gonna take our stuff to the PRI tommorow to test
23:42.58tzafrir_homethe original one is now called Trixbox CE (Community Edition)
23:43.00watchybut everything from what I can tell is running
23:43.14watchyits starting and shutting down fine
23:43.15tzafrir_homeThe other one is some Fonality stuff, and seems to be called Trixbox Pro
23:43.18watchytrixbox is a POS
23:43.51J4k3watchy: its saved me about $1500 so far, and cost me nothing except maybe 3-4 hours of screwing with it
23:44.45watchyyea but if you knew what u was doing you could do the same thing
23:44.52watchywith just * a pc and zaptel drivers
23:44.55watchyand a book
23:44.59watchyguis make you stupid
23:45.11watchylook at all these linux techs that run GUied linux
23:45.14watchythey are all retarded
23:45.15J4k3watchy: correct, but a lot of people don't want to screw with it (I personally didn't have the time to learn...)
23:45.21watchyconsole COWBOY FOR LIFE BITCH
23:45.45watchyreal men use cli
23:45.53J4k3real men make money
23:45.59J4k3and lay the pipe
23:46.03J4k3beyond that its all gravy
23:46.07J4k3:)
23:46.58*** join/#asterisk mvanbaak (n=mafkees@vanbaak.xs4all.nl)
23:46.59J4k3like the guy said... hes done it before, he doesn't want to dick with it...  understandable
23:47.10J4k3if all you want is a very simple setup, trixbox and the like do the job fine
23:47.19J4k3I mean shit, its hard to deny that people want shit to just work
23:47.34J4k3Microsoft's being filthy rich proves this
23:47.47J4k3not that windows ever 'just works' but its less intimidating than say, gentoo.
23:48.13J4k3nor would I want to support 83 year old grandmas trying to run gentoo.
23:48.18watchytrix box is fine
23:48.24J4k3for me to poop on
23:48.24J4k3:D
23:48.25watchytill they quit giving it out for free
23:48.28watchythen your screwed
23:48.37watchyits the way crack dealers work
23:48.39J4k3yeah, same could be said for asterisk itself.
23:48.41watchyhere some free crack
23:48.45watchyget hooked
23:48.52watchyoh this crack aint free no more
23:48.55watchypay up hoe
23:49.30watchyman i've been in LR for the past 4 or 5 days
23:49.35J4k3LR?
23:49.36J4k3little rock?
23:49.36watchyi'm going home to camden guys
23:49.38watchylittle rock
23:49.43J4k3my girlfriend is in little rock
23:49.46watchynew company I work for is here
23:49.49J4k3I'ma be up there on thursday
23:49.49watchywhats her cell
23:49.52watchyimma go visit her
23:49.55J4k3501-your-mom
23:50.01watchythats my moms #
23:50.06J4k3yeah, she's good.
23:50.08J4k3:D
23:50.11watchyhaha
23:50.22watchyif i'm up here thursday you should buy me a steak
23:50.45J4k3might be friday, hard to say... I gotta buy some tires and get an inspection sticker and shit
23:50.51J4k3and a cv joint
23:50.59watchywhy you coming to LR?
23:51.31J4k3because my girlfriend won't drive down here again til I drive up there? :)
23:51.31watchyoh
23:51.36J4k3she's moving down sometime after christmas
23:51.36watchywell i'll keep her company if shes nic
23:51.37watchye
23:51.47watchyshe got a slutty sister or anyhting?
23:52.01J4k3kinda, but she's 4.  her sister points at everything and yells "gina!"
23:52.18J4k3its like "omg she's calling everything in the store a vagina... why?"
23:52.28watchyhaha
23:52.32watchy4 thats kinda young
23:52.37watchycall me in 3 years
23:52.39J4k3HAHAHA
23:52.48J4k3isn't the AOC in Arkansas about 7 anyways?
23:52.55J4k3or is that only if you're closely related?
23:52.59J4k3arkansas scares me.
23:53.20J4k3if it ain't imbred, its hellaghettothug
23:54.13watchyhaha
23:54.20J4k3hell, watch out
23:54.26J4k3LR is up to like murder #42 for the year
23:54.49watchycrazy
23:54.58J4k3quite a good showing considering its what, 1/10th the size of houston?
23:55.01watchyi'm probably responsilble for that
23:55.06J4k3houston is up to like 300ish
23:55.12J4k3stop busting caps
23:55.18J4k3and stickin hoes
23:56.35watchyhaha
23:56.44watchyman i'm headin out i'm starving and i got a 2 hour drive
23:56.56J4k3good lucjk
23:56.58J4k3er luck
23:56.59watchylater jake
23:57.10watchylater tzafrir

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