IRC log for #asterisk on 20070901

00:02.58*** part/#asterisk cirgal (n=robert@pool-71-102-137-33.snloca.dsl-w.verizon.net)
00:05.38tzafrir_homeCCFL_Man2, that could be it
00:05.58*** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2)
00:06.03tzafrir_homeCCFL_Man2, there should be backports of zaptel 1.4 in pkg-voip's buildserver, if you're lasy
00:06.14tzafrir_homeI think that a sparc builder is included there
00:06.35tzafrir_homehttp://buildserver.net/
00:08.07CCFL_Man2hmm.. yeah
00:08.29CCFL_Man2udevinfo: command not found
00:09.01CCFL_Man2you apparently know about that bug, where anyway to fix it?
00:09.02Star568anybody here knows howcome there is no ring tone while waitting for the call get answered. i tried to use manager to bridge two SIP calls. anybody knows why?
00:09.08tzafrir_homehttp://pkg-voip.buildserver.net/debian/pool/main/z/zaptel/ . Hmmm... I don't see Etch sparc builds
00:09.13tzafrir_homestrange
00:09.29CCFL_Man2there are no etch sparc versions yet
00:09.33*** join/#asterisk LukinoVoip (i=LukinoVo@151.82.4.215)
00:09.48tzafrir_homethe latest one was: http://pkg-voip.buildserver.net/debian/pool/main/z/zaptel/zaptel_1.4.4~dfsg-2~etch.3851_sparc.deb
00:09.57tzafrir_homeand the same of that version
00:10.20CCFL_Man2hmm.. ok
00:10.34CCFL_Man2i'll give that a try
00:12.22CCFL_Man2sarge seems to use zaptel 1.2, etch seems to use 1.4
00:12.53CCFL_Man2but that buold bug, it looks for udevinfo on sarge but the command isn't found, there anyway to fix that?
00:15.43*** join/#asterisk nighty^ (n=nighty@sushi.rural-networks.com)
00:18.39etfonhomey_Has anyone here setup * with a Dynamic T1 connection to the PSTN?
00:19.10*** join/#asterisk JunK-Y (n=junky@modemcable183.17-83-70.mc.videotron.ca)
00:20.44CCFL_Man2tzafrir_home: i can't build 1.4
00:20.52CCFL_Man2udevinfo: command not found
00:21.05tzafrir_homeWhat version is it?
00:21.11JunK-YCCFL_Man2: install udev?
00:21.14[TK]D-Fenderetfonhomey_, Zaptel doesn't do "dynamic"..... channels are fixed, you should know that...
00:21.26tzafrir_homemake DYNFS=yes
00:21.42*** part/#asterisk BSD_Tech (n=BSDTech@adsl-69-234-202-145.dsl.irvnca.pacbell.net)
00:21.59tzafrir_homeyeah, and install udev, if you have 2.6.8:  apt-get install udev
00:22.09CCFL_Man2ahh, ok
00:22.09tzafrir_homebetter than devfs anyway
00:22.22tzafrir_homebut the above is also a workaround
00:22.32etfonhomey_[TK]D-Fender, I've only used fixed POTS channels with zaptel.  So, never looked at the T1 setup.
00:23.42*** join/#asterisk |dennis| (n=dennis@200.32.236.18)
00:23.53*** join/#asterisk [LiFE] (n=rawrrawr@unaffiliated/life/x-0000003)
00:24.18[TK]D-Fenderetfonhomey_, For what you described earlier, the layout would look like 1 - 7 B, 8-23 Data, 24 D
00:24.20etfonhomey_Anyone here seen a dynamic T1 setup?  How do they hand off the voice?
00:24.32[LiFE]hi, I just got a SPA400 and want to hook it up to my asterisk box, I can make outboud calls fine, but when I get calls in, it just gets a very short unavailable tone, and then disconnects
00:24.40[TK]D-Fenderetfonhomey_, Again, there isn't anything "dynamic about it.
00:24.42[LiFE]anyone have experience with the SPA400?
00:26.07CCFL_Man2udev requires a kernel >= 2.6.15, not started.
00:26.41*** join/#asterisk justdave (n=dave@unaffiliated/justdave)
00:28.13etfonhomey_[TK]D-Fender, are we talking about the same thing?
00:28.44[TK]D-Fenderetfonhomey_, About that split pri/data thing from earlier, no?
00:28.52rodent|Sblackhat_2007_audio_torrent
00:28.54rodent|Shttp://www.stonedcoder.org/tt/details.php?info_hash=da957418bff04c17aa1e357979084665643ef7e0
00:28.58rodent|Sdownload, and seed. bitchez.
00:29.00rodent|Senjoy.
00:29.17etfonhomey_[TK]D-Fender, in the dynamic T1, when no voice calls are in progress, the whole T1 is used for data.
00:29.39Qwellrodent|S: please don't be spamming that here
00:29.40[TK]D-Fenderetfonhomey_, * can't do that to my understanding...
00:29.50rodent|SQwell: spam ? i think not. but thanks for the input.
00:30.20CCFL_Man2dammit i can't get this right!
00:31.40CCFL_Man2i knew this would fight me all the way
00:31.48CCFL_Man2compiling again
00:32.48etfonhomey_[TK]D-Fender, I'm curious how the voice is handed of in a dyn t.
00:33.32[TK]D-Fenderetfonhomey_, setup & teardown would be arranged by the dchan I'm sure
00:34.18etfonhomey_[TK]D-Fender, it's not over a channelled T1.  At least that's how it was explained to me.
00:34.21[TK]D-Fenderetfonhomey_, So a call comes in, d-chan  requests a B chan to carry and one is sequentially knoccked off the start.  On hangup it passes back to data]
00:34.37[TK]D-Fenderetfonhomey_, Indeed not channelized, it'd be PRI
00:35.41[TK]D-Fenderetfonhomey_, There are VPRI's as well where they just carry BW to your door and fractionalize the stream with a T1 loop LOCAL to your box.
00:35.45CCFL_Man2etfonhomey_: you might have to use a cisco router to provide voice to the asterisk box and data over youe network
00:35.53[TK]D-Fenderetfonhomey_, Transparently throttled.
00:36.20*** join/#asterisk RipeR81 (n=ircap8@190.53.33.3)
00:36.38etfonhomey_[TK]D-Fender, i imagine that would be provide as by the provider of the dyn t.
00:37.24RipeR81hello everyone. im a newbie on asterisk and i was wondering if any of you guys with experience can suggest a linux distro that works better with asterisk ?
00:37.42[TK]D-FenderRipeR81, just about any will do once you've install its dependencies
00:37.57[TK]D-FenderRipeR81, Basically whichever you are most comfortable administering.
00:38.20RipeR81[[TK]D-Fender] thanks for the info... im just waiting for the server to come from customs...
00:38.24[TK]D-FenderRipeR81, Debian, FC, RHEL, CentOS, Slackware, etc....
00:39.11RipeR81[[TK]D-Fender] thanks ...
00:39.11[TK]D-FenderRipeR81, Plenty of people running it on Ubuntu even.
00:39.38RipeR81ubuntu really ? well... ubuntu has being becoming very popular
00:39.54RipeR81maybe bcuz of his policies of mailing free cds worldwide
00:40.09tzafrir_homeA nice desktop. Still has a way to go to become a solid server
00:40.57tzafrir_homeAlso in ubuntu you should be careful about stuff that is in universe, and hence not officially as well-supported as the rest
00:41.44[TK]D-Fendertzafrir_home, I largely concur.  I like something using traditional inits, etc myself.  Straight Debian or RHEL would make me more comfortable for sure
00:41.46RipeR81[tzafrir_home] thank.
00:41.58LukinoVoipdebian
00:42.12LukinoVoip:-)
00:42.34*** join/#asterisk zapp-branigan (n=zapp-bra@9.218.216.87.static.jazztel.es)
00:42.46RipeR81i see everyone is into debian here.. hahaha
00:42.50RypPnanyone have asterisk running successfully on sparc?
00:42.56tzafrir_homeCCFL_Man2, what's up?
00:43.06RypPnI seem to be suffering kernel oops with zttranscode
00:43.12tzafrir_homelinux or Solaris?
00:43.16RypPnlinux
00:43.23tzafrir_homeask CCFL_Man2 :-)
00:43.33RipeR81to be certain... i've just used RHEL, suse and madriva... i remember i even tried corel linux when it first came out
00:43.35zapp-braniganhi, please who can i generare a md5 password ?
00:43.42zapp-braniganfor asterisk
00:44.05tzafrir_homeRypPn, a kernel oops is always bad. But do you actually need zttranscode?
00:44.29tzafrir_homeRypPn, an oops or a panic? Do you happen to have the trace?
00:44.34RypPntzafrir: only insofar as all sound is dead, internal calls, vmail etc
00:45.16tzafrir_homethe system itself remained running?
00:45.20RypPnI can easily fire the box up again, its at my feet
00:45.24RypPnyes
00:45.29CCFL_Man2tzafrir_home: same error again
00:45.43CCFL_Man2ioctl(ZT_LOADZONE) failed: Invalid argument
00:45.44CCFL_Man2line 233: Unable to register tone zone 'us'
00:45.44CCFL_Man2ZT_DEFAULTZONE failed: Invalid argument (22)
00:45.46tzafrir_homedo you have a long, ugly, trace in /var/log/kern.log (is this Debian?)
00:46.08tzafrir_homeCCFL_Man2, what do you see in the kernel logs?
00:46.09RypPnno, its gentoo, I'll have a look for logs
00:46.54tzafrir_homeRypPn, do you have any zaptel hardware, or just ztdummy?
00:47.05CCFL_Man2http://rafb.net/p/seKA0x79.html
00:47.06[LiFE]anyone here with experience of SPA400 and Asterisk?
00:47.13CCFL_Man2thats what i see
00:47.13[LiFE]can't seem to get it to work with inbound calls
00:47.21RypPntzafrir: just using the ztdummy module for timing
00:47.38[TK]D-Fender[LiFE], go check out the forums at www.volxilla.com
00:47.46[TK]D-Fenderwww.voxilla.com ***
00:47.52[LiFE]I did... I see pple complaining about it, no fixes. :(
00:48.03tzafrir_homeCCFL_Man2, hmm... are you sure yu're not mixing ztcfg from 1.2 with drivers from zaptel 1.4 or vice versa?
00:48.29CCFL_Man2tzafrir_home: nope, i uninstalled everything and compiled 1.4
00:48.37CCFL_Man2well, i am sure
00:49.03tzafrir_homeJust to be sure, run ./ztcfg    (from the build directory)
00:49.09CCFL_Man2ok
00:49.11RypPntzafrir_home: http://rafb.net/p/LzeXzr51.html thats what dmesg shows
00:49.32[TK]D-Fender[LiFE], perhaps you should pastebin an inbound call attempt with SIP debug enabled so we have something to see and show us your sip.conf entry for it as well.
00:49.33[LiFE][TK]D-Fender: I followed the guides from voxilla, I can make outbound calls, etc... but when inbound calls comes in, the spa400 picks it up, then hangs up
00:49.35[TK]D-Fender~pb
00:49.36jbotA Pastebin is a place to paste your stuff without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste
00:49.36[TK]D-Fender^^^^^^^^^^^^^^^^^^^^
00:49.44CCFL_Man2tzafrir_home: same error
00:49.49[LiFE]asterisk -rvvvvvvvvvv ??
00:50.12[LiFE]or just do a sip debug?
00:50.16[TK]D-Fender[LiFE], could be the SPA's configured quite wrong at source and its not even trying.  Could also be taht your * config is screwed up so you just going to have to show us everything
00:50.30[TK]D-Fender[LiFE], in * CLI : sip debug
00:50.33*** join/#asterisk geoff_K (n=geoff@host81-152-90-217.range81-152.btcentralplus.com)
00:50.54tzafrir_homeRypPn, well, an invalid ioctl. But no real issue
00:50.55[LiFE]yah, it could be the SPA400, coz I did a debug today, didn't even show anything coming in
00:51.09[LiFE]it's like the call just hang up on the SPA400 and it never did anything
00:51.27tzafrir_homeRypPn, can you give a more complete description of what you ran before the oops?
00:51.42[TK]D-Fender[LiFE], then I suggest you host up some screenshots of its config.
00:51.46tzafrir_homeIs that now after a reboot?
00:51.55[LiFE]doing a upgrade of firmware to 1.0.0.c
00:51.58CCFL_Man2ioctl32(ztcfg:15412): Unknown cmd fd(4) cmd(80704a19){00} arg(efffb958) on /dev/zap/ctl
00:52.01CCFL_Man2ioctl32(ztcfg:15412): Unknown cmd fd(4) cmd(80704a19){00} arg(efffb958) on /dev/zap/cioctl32(ztcfg:15412): Unknown cmd fd(3) cmd(80044a18){00} arg(000282f0) on /dev/zap/ctl
00:52.26[LiFE]will do a pastebin on the sip debug, then do a paste of the screenshot... u know where I can put those pics?
00:53.18CCFL_Man2tzafrir_home: now i'm not using any zaptel hardware at all, just ztdummy
00:54.33CCFL_Man2so do i need to worry about ztcfg or no?
00:55.19tzafrir_homeyou use 2.6.8 now?
00:55.29[TK]D-Fender[LiFE], I recall a place called "photobucket" or something.  And there's always flickr.
00:55.45CCFL_Man2tzafrir_home: yes, 2.6.8
00:55.56tzafrir_homeok. Let's make sure that the current modules are loaded
00:56.04tzafrir_homermmod ztdummy zaptel
00:56.12*** part/#asterisk LukinoVoip (i=LukinoVo@151.82.4.215)
00:56.13tzafrir_homeinsmod ./zaptel.ko
00:56.19CCFL_Man2ok
00:56.20tzafrir_home./ztcfg
00:56.24*** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla)
00:56.24*** mode/#asterisk [+o russellb] by ChanServ
00:57.18CCFL_Man2Zapata Telephony Interface Registered on major 196
00:57.18CCFL_Man2Zapata Telephony Interface Registered on major 19Zaptel Version: 1.4.5.1
00:57.18CCFL_Man2Zaptel Echo Canceller: MG2
00:57.45CCFL_Man2same error
00:58.48tzafrir_homeCCFL_Man2, and are you sure that there's nothing in the kernel logs? Just "invalid ioctl"?
00:59.07CCFL_Man24 unknowm command errors
00:59.23CCFL_Man2right, just ioctl32(ztcfg:15427): Unknown cmd fd(3) cmd(40a04a39){00} arg(effff8a8) on /dev/zap/ctl
00:59.29CCFL_Man2and 3 more
01:01.14tzafrir_homeAre those new ones?
01:02.49tzafrir_homewell, anyway, if you just have ztdummy, you don't need ztcfg. So let's not waste any more time on this one
01:03.05tzafrir_home(no channel to configure)
01:03.09RypPntzafrir_home: It looks like I gutted asterisk from it. I'll take some time out to rebuild it now. With any luck the problems will just disappear *cough*
01:03.33tzafrir_homeRypPn, you can disable codec_zaptel
01:04.12tzafrir_homeNo use for you if you don't have a zaptel transcoder card. Still there may be endiannes issues or whatever there
01:04.16[LiFE][TK]D-Fender: http://www.pastebin.ca/677457
01:04.45CCFL_Man2tzafrir_home: right, they are the ones in the build directory
01:05.01CCFL_Man2all i'm using is ztdummy, no tdmoe or hardware
01:05.05tzafrir_homeCCFL_Man2, or just remove the "loadzone" and "defaultzone" lines from zaptel.conf
01:05.25CCFL_Man2tzafrir_home: those aren't needed if i just use ztdummy?
01:05.34tzafrir_homeno
01:05.43tzafrir_homeYou have no channel to configure
01:06.23CCFL_Man2maybe thats why i'm getting the errors?
01:07.58[TK]D-FenderL#
01:07.58[TK]D-Fenderooking for spa400 in from-trunk (domain 192.168.1.200)
01:07.58[TK]D-Fender#
01:07.58[TK]D-FenderReliably Transmitting (NAT) to 192.168.1.201:5060:
01:07.58[TK]D-Fender#
01:07.59[TK]D-FenderSIP/2.0 484 Address Incomplete
01:08.19[LiFE]eh?
01:08.20[TK]D-Fender[LiFE], bad dialplan it seems
01:09.24CCFL_Man2tzafrir_home: there a way to test that everything is working correctly, maybr zttest or ztmonitor?
01:10.00[LiFE]dial plan?
01:10.02[LiFE]it's incoming
01:10.07[LiFE]not outbound
01:10.09tzafrir_homeyou just need ztdummy for timing, right? so zttest
01:10.19*** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga)
01:10.23*** part/#asterisk N0S3 (n=cristian@host168.190-136-201.telecom.net.ar)
01:10.25CCFL_Man2tzafrir_home: right
01:10.26tzafrir_homehmm, well also mixing meetme
01:10.27CCFL_Man2ok
01:11.22CCFL_Man2zttest reports Unable to open zap interface: No such device or address
01:11.40CCFL_Man2that normal too?
01:13.41[LiFE]hmm?
01:15.53tzafrir_homemodprobe ztdummy
01:16.38JT[LiFE]: so? incoming and outcoming have a dialplan
01:17.05[LiFE]?
01:17.29[LiFE]sorry, I keep assuming dial plan = dialing patterns, could you elaborate?
01:18.59JTextensions.conf
01:19.24JTetfonhomey_: umm, those "dynamic T1s" probably aren't doing voice in a T1 way at all
01:19.31JTprobably SIP over T1
01:19.47JTyou lose capacity due to protocol overhead
01:20.30*** join/#asterisk rvhi (n=chatzill@66.135.230.96)
01:20.40[TK]D-Fender[LiFE], dialplan = extensions.conf
01:21.03harryr"dynamic T1"...
01:21.04rvhihi, i have checkmwi=10 in sip.conf, it means that it sends out mwi every 10 seconds to the phones, really annoying
01:21.05etfonhomey_JT, Are you in Australia?  If so, do they offer them there?
01:21.09rvhiis that right?
01:21.12CCFL_Man2tzafrir_home: silly me, here i thought it was loaded
01:21.13harryrthere is no such thing as a dynamic T1 - it's either a T1 or it's not
01:21.47[TK]D-Fenderharryr, dynamic between voice/data SPREAD across it.
01:22.05[TK]D-Fenderharryr, AKA need more B-chans?  reduce datat chans
01:22.11harryrah fair enough
01:22.16CCFL_Man2tzafrir_home: ok, well i think that works, i get an average of 99.97%, i suppose thats pretty accurate timing?
01:22.24etfonhomey_http://www.dslreports.com/nsearch?q=%22dynamic+t1%22
01:22.46harryrit's just I've heard sales people trying to sell "dynamic T1s" that are just sdsl connections capped at 1.5mbit
01:23.12tzafrir_homeyes
01:23.39[LiFE]exten => SPA400,n,Goto (from-did-direct, 311, 1)
01:24.52RypPntzafrir_home: here's an example log of what I'm talking about, when I dial *65 I hear nothing. http://rafb.net/p/DRyUAo83.html
01:26.40tzafrir_homeRypPn, what are the results from zttest ?
01:26.42RypPnI think I know what it is... let me test
01:26.53CCFL_Man2tzafrir_home: so i guess the next step is configuring asterisk :P
01:28.04*** join/#asterisk domger (n=domger@ip150.202-51-69.sogetel.net)
01:28.04tzafrir_homeCCFL_Man2, BTW: have you actually tried using asterisk from packages?
01:28.09tzafrir_homeas a baseline
01:29.00RypPntzafrir_home: how many passes would you like? It never drops below 99.98
01:33.58CCFL_Man2tzafrir_home: actually no, but i installed asterisk and supporting packages that it installed
01:34.27*** join/#asterisk ming_zym (n=ming_zym@124.254.52.236)
01:34.54CCFL_Man2never configured it yet
01:35.14*** join/#asterisk shadowtech (n=Jonathan@adsl-074-186-013-209.sip.mia.bellsouth.net)
01:35.34shadowtech<PROTECTED>
01:35.50CCFL_Man2shadowtech: wtf?
01:36.00*** part/#asterisk shadowtech (n=Jonathan@adsl-074-186-013-209.sip.mia.bellsouth.net)
01:38.02CCFL_Man2tzafrir_home: what i wanted was to interface asterisk to my channel bank via ip with a cisco voice concentrator
01:38.05CCFL_Man2mc3810
01:39.45JTetfonhomey_: yes, and no
01:40.29JTharryr: for what it's worth, most T1s and E1s are delivered over SHDSL these days
01:40.48[LiFE]i give up, redoing an installation, this time without freepbx
01:41.19CCFL_Man2JT: HSDL in the US i think
01:41.35JThsdl?
01:41.45CCFL_Man2HDSL, rather
01:41.53JTanyway, same difference
01:41.56JTover dsl
01:42.02CCFL_Man2right
01:42.46CCFL_Man2with adsl in the US, ATM is sent over it
01:43.09[TK]D-FenderJT : many quite possibly.  Look for CBR in the data spec if youwan to see that you're getting a "real" T1
01:43.09etfonhomey_JT, here's Nuvox's version of a Dynamic T1:  http://www.nuvox.com/ResourceCenter/Products/voxip_email_011707.pdf
01:43.12JTyes atm and adsl are good buddies
01:43.21*** join/#asterisk rodent|S (i=nobody@foster.stonedcoder.org)
01:43.59JT[TK]D-Fender: usually there's no problem with it being delivered over shdsl, because T1s can't go very far in their ntive format
01:45.26hmmhesaysok, there was some bug in fedora where I had to copy some config file somewhere to make the zaptel script start ztdummy
01:45.47JTetfonhomey_: no-one bothers with "dynamic T1s" in australia
01:45.52JTstopgap for stingey businesses
01:46.14etfonhomey_No stingy businesses in Australia?
01:46.31JTour E1 pricing structures are different
01:46.39JTand we also have BRI
01:46.47JTand a much wider variety of DSL choice
01:47.55JTno telcos offer this SIP over E1 bs as far as i'm aware
01:48.57etfonhomey_JT, What do you mean by "SIP over E1"?
01:49.15JTin the US "dynamic T1s" are usually SIP over T1
01:49.36JTrun voip over the T1 to dynamically change the amount of bandwidth available for data
01:50.23*** join/#asterisk fakhir (n=Fakhir@unaffiliated/fakhir)
01:50.41etfonhomey_JT, do you have any links explaining that?  I can't find anything explaining the technical side of a dynamic T1.
01:50.58JTit's simple you terminate it to a voip capable router
01:51.48etfonhomey_And then how does the router hand the customer voice and data?
01:51.55etfonhomey_separately?
01:52.18CCFL_Man2etfonhomey_: use a router and have the data on the ethernet and the voice on a CAS T1 or over voip
01:52.33CCFL_Man2a cisco router
01:52.41etfonhomey_Do you know a particular model?
01:52.43JTetfonhomey_: they don't, it's all one big data pipe.
01:52.57JTthere is nothing CAS or PRI about it usually
01:53.21etfonhomey_So, then how could you hook up the voice part of it to your PBX?
01:53.43JTthe router either gives you ethernet or T1 handoff
01:53.56CCFL_Man2etfonhomey_: mc3810 is a good modem, but for voip it needs a new prom, 64mb ram, updated ios, the MFT T1 card for the data/voice trunk and DVM-T1 card if you want voice over a CAS T1
01:54.02JTor FXS, worst case :P
01:54.14CCFL_Man2err modem = router
01:54.28*** join/#asterisk Netgeeks (n=chris@gw-hmb.netgeeks.net)
01:54.35CCFL_Man2but the MC3810 is EOL
01:55.31CCFL_Man2modern routers can do it
01:55.37CCFL_Man2modern cisco routers
01:55.44etfonhomey_JT or CCFL_Man2: So, if you're getting the T1 handoff, then a dynamic T1 should work with *?
01:55.54JTi guess so
01:56.17etfonhomey_CCFL_Man2?
01:56.24CCFL_Man2i see no reason why not, but you won't have access to data
01:56.31etfonhomey_?
01:56.39CCFL_Man2if you just have the T1 going into asterisk
01:56.43JTdata goes out a seperate interface
01:56.54CCFL_Man2you won't have access to data
01:56.54JTasterisk doesn't need the data
01:57.08JTthat's for the Internet side of things
01:57.19CCFL_Man2can that be configured in zaptel?
01:57.21russellbyou can set up part of the T1 for data in linux
01:57.41JTdepending how the router hands off the T1
01:57.42russellbsetting it up is even covered in free installation support
01:57.45*** join/#asterisk coppice (n=chatzill@140.196.17.210.dyn.pacific.net.hk)
01:57.45russellb(from digium i mean)
01:57.54russellbtrue ..
01:58.00JTthe router might have an ethernet port just for dat
01:58.02JTa
01:58.07JTwhich would make far more sense
01:58.10*** join/#asterisk domger (n=domger@ip150.202-51-69.Sogetel.net)
01:58.14etfonhomey_That's what I am hoping.
01:58.25hmmhesaysheh, the zaptel script does not start ztdummy on fc7
01:58.29etfonhomey_I haven't found anyone in here who has a Dynamic T1.
01:58.29CCFL_Man2but if he has a T1 router already, why not just use it to seperate voice and data
01:58.31domgerHi, Is possible to have multiples trunks between a Asterisk and a Softswith?
01:58.52hmmhesaysit exits after -f /etc/zaptel.conf
01:59.04JTetfonhomey_: if you come here at the right times, there are followers of the dynamic t1 religion
01:59.09JTbut i think it's arse
01:59.12etfonhomey_CCFL_Man2, the office in question will be a new install.
01:59.49etfonhomey_JT, my situation is a satellite office, that currently has 6 POTS lines and dual DSL.
01:59.49JTdomger: if you're talking about VoIP, there are no trunks
02:00.57domgerJT: in my ITSP softswitch they says trunk...
02:01.11JTwell it's incorrectly named
02:01.53CCFL_Man2JT: voice and data T1s are cheap
02:02.27hmmhesaysthat is just strange
02:02.33JTCCFL_Man2: it's still arse :)
02:02.35hmmhesayszaptel didn't install zaptel.conf
02:02.45CCFL_Man2etfonhomey_: a newer cisco router can seperate voice and data and send it to asterisk over voip
02:03.02CCFL_Man2JT: i know, but it vworks :P
02:03.05CCFL_Man2works
02:03.13JTso?
02:04.25domgerJT: in Asterisk world, how do you call a SIP connection without user?
02:04.46JTdomger: in the world in general, not just asterisk, SIP has no trunks
02:04.52JTit sets up connections on demand
02:05.16etfonhomey_CCFL_Man2, got any references for dynamic T1's?  I'd like to read a little more about the technical side of things.
02:06.01*** join/#asterisk VJFROMGT (n=vjfromgt@static-68-161-227-229.ny325.east.verizon.net)
02:06.11CCFL_Man2etfonhomey_: honestly no, but i'd suggest searching cisco for docs on the subject
02:06.37etfonhomey_domger, JT may be trying to say that trunk implies both incoming/outgoing calls going over the same "pipe".  With SIP, you're outgoing and incoming calls are different channels.
02:06.51etfonhomey_CCFL_Man2, Is dynamic T1 the right terminology?
02:07.33JTetfonhomey_: no, a trunk is a pipe that handles a certain amount of voice calls
02:07.37JTSIP does not do this
02:07.47etfonhomey_Gotcha.
02:08.26JTlots of morons are using terms like "sip trunking" but in reality, sip doesn't have anywhere near the level of reliablity to be called a trunk
02:08.30etfonhomey_Generally, when people say SIP trunk, they're talking about their SIP connection to their ITSP.
02:08.30JTit has no circuit
02:08.34JTno guarantee
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02:11.15VJFROMGTinbound calls work fine but outbound disco after about 15 seconds
02:11.17VJFROMGThttp://pastebin.ca/677489
02:11.31VJFROMGTeven outbound extension to extenson
02:14.04domgerBTW, my problem is that my ITSP is providing me 2 "connections" from the same softswitch.  One is associate with a number from city A and the other with a number from city B.  Both cities have different local calling area... So I want to "choose" my outgooing "connection" to fool his softswith and save long distance call...  Since both connection  are connected on the same Asterisk, my ITSP identify all my calls like calls coming from "connec
02:14.58JTdomger: avoid runon sentences, the ircd cuts them off
02:15.31VJFROMGTi have no idea what domger is saying but it sounds liek he is using the wrong isp for what he needs
02:16.05RypPnare there any management interfaces anyone would recommend?
02:16.13JTno
02:16.20RypPnother than freepbx, vi, nano etc
02:16.22JTwell, vi
02:16.27RypPn;)
02:17.11coppiceleave management alone, and just try to live a good clean life apart from them
02:18.18RypPnI think it was freepbx that killed the audio on my sparc, it may not have recognised any feature access codes before, but at least it told me so
02:19.00RypPntime to roll the sleeves up I guess
02:28.30CCFL_Man2JT: is pots a trunk?
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02:29.17CCFL_Man2i'd say sip, rtp, and the combination of that and the ip link and the physical links, i'd call it a trunk
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02:31.23coppiceVoIP trunks? I thought VoIP was based on hatchbacks and station wagons
02:32.14joatcoppice: if you try hard enough, you can get them up on the roof racks
02:33.08JTCCFL_Man2: you can say what you like, doesn't make it true.
02:33.39CCFL_Man2so there is no such think as a sip trunk?
02:34.50JTcorrect.
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02:35.22De_MonI've never really though of a 'trunk' referring to any sort of reliability. what do you perfer to call them instead jt?
02:35.26coppicemany telephony people only consider something a trunk when it bundles calls. there is a provisional spec for putting multiple calls in an RTP packet, to reduce the awful overheads, and its termed RTP trunking
02:36.31JTDe_Mon: calls, connections, whatever
02:36.38JTsip just randomly sets up calls over ip
02:36.41JThardly a trunk
02:37.23De_Monits the medium the calls and connections go 'through' though
02:37.52CCFL_Man2De_Mon's definition seems more correct
02:38.00JTCCFL_Man2: blah blah
02:38.05CCFL_Man2hehe
02:38.10JTyou're in no place to judge
02:38.27CCFL_Man2yeah, i'm new to telephony
02:38.46De_Monopinions are like assholes, most of them stink
02:39.34*** part/#asterisk kclaussen (n=kclausse@204.13.224.242)
02:39.52coppicethere is no real definition of trunk, unlike its in a cabin on a cruise ship, or an elephant is attached to it
02:39.59*** part/#asterisk joat (n=joat@ip70-160-147-169.hr.hr.cox.net)
02:40.04CCFL_Man2if you get your nose in there deep enough all of them smell, atleast girls anyway
02:40.08CCFL_Man2err..
02:40.19CCFL_Man2:P
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02:42.36coppicewell all know VoIP calls are carried in a more generic form of trunk, called a tube
02:44.41QwellDe_Mon: nice
02:47.02coppiceDe_Mon: opinions are more like farts. only other people's really stink badly
02:47.21CCFL_Man2ahahahah!
02:47.31CCFL_Man2i like that one
02:47.33De_Moneven better!
02:48.06De_Monfantastic 4: silver surfer really sucked. not even worth renting how sad
02:48.08JTDe_Mon: so why should sip be considered trunking?
02:49.55De_MonA trunk is a line or link designed to handle many signals simultaneously.. a sip trunking provider typically
02:50.37JTdo you understand how sip works?
02:50.40Qwellcoppice: and for the record - I think it was based on the fastbacks and notchbacks :p
02:50.42De_Monfor me, it speaks to the volume which doesn't seem to be as true in the telephony world
02:51.12JTthat made no sense
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02:54.41coppiceQwell: I think it was based on something from GM, because nobody seemed to give a damn about the enormous overheads
02:55.02De_Monyes, but its like looking at a drawing and saying its just a bunch of lines, some people see lines, others see a picture
02:55.32JTDe_Mon: no idea what you're going on about
02:56.17coppice"Some people can read War and Peace and see nothing more than an adventure story" - I was trying to prepare my 10 year old to read War and Peace last week, and found myself quoting Lex Luther from Superman II :-)
02:56.47De_Monyoure looking at 'sip trunking' and seeing lots of individual connections and I see a trunk
02:57.00JTDe_Mon: you imagine things
02:57.05JTthere is no dedicated circuit
02:57.51coppicetoo right. there's no bloody dedication at all amongst the jonny come lately internet providers
02:58.49JTheh
02:59.53[LiFE]anyone here knows whether we must have CID enabled in order to use SPA400
03:00.34De_Mon[LiFE] were you looking for a yes/no question?
03:00.37De_Monerr answer
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03:01.57[LiFE]yap!
03:02.25De_Monin that case the answer is most definately yes
03:02.59CCFL_Man2strange how you must use CID for an ATA to work
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03:04.32De_Monwhoever said there is no such thing as a stupid question hasn't sat on irc long enough
03:07.09CCFL_Man2personally i don't like atas
03:07.28CCFL_Man2i prefer a channel bank
03:08.09CCFL_Man2anyone here use an adit 600?
03:08.49JTsip gateways make a bunch of sense at a number of channels for quite a few applications
03:09.53CCFL_Man2honestly, the only ata i've used was a linkshit PAP2, maybe thats why i don't like them
03:10.51[LiFE]nevermind, found the answer
03:11.10[LiFE]evil thing requires the device name in DID inbound route
03:11.17[LiFE]which is crazy, I tot ANY covers ANY
03:11.25[LiFE]and apparrently not
03:11.43JTtot?
03:12.26[LiFE]hought
03:12.35[LiFE]ehh.. 1 finger typing.. lzy
03:13.38CCFL_Man2i tot i taw a puddy cat
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03:14.51JTi guess he expects us to magicall understand [LiFE] english
03:16.28CCFL_Man2so no one uses the adit 600?
03:16.42CCFL_Man2i need the camual and they won't give it to me
03:19.19CCFL_Man2fuckers
03:22.14harryrcamual?
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03:26.03CCFL_Man2manual
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05:02.52nclxIs there a way to make an extension global among contexts?  For example I have extension 7170.  This dials a SIP phone.  I have my extensions.conf broken into many contexts for managability, but I would like that if you dial '7170' from any context you will ring the SIP phone.  Must I define it in each context, or is there a way to make it global?
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05:06.42russellbyou can define a context called [global_extens] or something and then include it in all of the others
05:06.49russellbthen just add extens there that you want available toa ll
05:06.59nclxaha thanks that is simple and awesome
05:07.18russellbyou're welcome
05:07.23_x86_heh
05:07.39_x86_nclx: that's standard dialplan logic man :)
05:07.51nclxI'm new to this game
05:07.55_x86_ah cool
05:07.56russellbooh, it's a game?
05:07.59russellbam i winning?
05:08.02_x86_welcome to the circus ;)
05:08.04nclxyeah probably you are
05:08.10russellbawesome
05:08.14_x86_nclx: home use or business?
05:08.19nclxbusiness
05:08.25_x86_what kind of hardware?
05:09.19russellbvery hard hardware
05:09.25nclxAMD64 4600+ Dual Core, 4GB RAM, 300GB SATA, Ubuntu-Linux Server, TDM400 PCI card
05:09.37russellbnice
05:10.03russellbthat's quite a powerful box for a TDM400 :)  i assume you have a lot of VoIP usage, too?
05:10.03nclxSnom 320 && Budgetone SIP clients, broadvoice business line
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05:10.17nclxyes, the card is only for the timing
05:10.22russellbi see ..
05:11.11nclxI was having a ton of problems with getting audio to play okay when calling from PSTN or VoIP and getting tons of errors in my syslog about timing and the kernel missing interrupts etc.
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05:11.43nclxso I got the TDM card because I thought I might have had a BIOS issue on this motherboard after trying many different kernel compilation options with RTC and MPET stuff
05:12.08nclxturns out though...... rearranging my PCI cards and removing one non-used winmodem seemed to solve the issue
05:12.13nclxwhat a pain that was
05:12.45russellbouch
05:12.48russellbstupid hardware
05:12.53nclxyeah no kidding
05:13.13nclxyou think to yourself, well linux isn't even configuring the device how could that be the problem
05:13.24nclxbut I had a similar issue last week
05:13.46nclxI was upgrading ram on a box to 2gb from 256mb, and I kept getting no video and bios beeping no matter what combination I did
05:13.54nclxI even put it back the orig way still no luck
05:14.06nclxfinally I unplugged this cheap HP mouse I brought to test the box with
05:14.11nclxand it worked fine with the new mem
05:14.17nclxblew my mind
05:14.46nclxI feel like having an office space moment in a field with this mouse and winmodem
05:15.43*** join/#asterisk JoelSolanki (i=Joel@202.160.161.94)
05:16.22CCFL_Man2yes! my openwrt usb router outputs my mp3 collection on one usb audio stick and my favorite internet radio station on another
05:16.35JoelSolankii want to download zaptel from cvs.
05:16.40*** join/#asterisk kn0x (n=pinochle@75.127.83.141)
05:16.47JoelSolankibut cvs.digium.com is unavailable.
05:16.53JoelSolankiis this right link ?
05:18.56JerJersvn
05:19.34russellbsvn co http://svn.digium.com/svn/zaptel/branches/1.4 zaptel-1.4
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05:27.10nclxI can honestly say I haven't played with anything as fun as asterisk in a long time
05:29.14EchinosI just installed it, and spend 30 minutes going thru the sample gsm files
05:29.21Echinosvery fun
05:29.29QwellEchinos: There are extra sounds too
05:29.38Echinosthose were the ones I meant
05:29.38Qwelltons more
05:29.41Qwelloh
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05:29.49EchinosI couldn't spend 30 minutes on the base ones
05:29.53Echinos;)
05:30.01QwellThat'd be pretty tedious.
05:30.06russellband boring
05:30.12QwellI'm sure most of the devs have done it though ;p
05:30.14EchinosI was looking for the comedic ones, and the ones dealing with telemarketer torture ;)
05:30.14russellbextras has some good ones
05:30.16JoelSolankirseelb: thanks
05:30.18JoelSolankigot it.
05:30.23russellbyou're welcome
05:30.29EchinosI like jedi-extension-trick
05:30.46Echinos"This is not the extension you are looking for...."
05:30.48QwellEchinos: there are a few that were too good (or bad?) to put in there...
05:30.57EchinosAwww
05:32.16EchinosI listened to a defcon presentation from Strom Carlson & Black ratchet on Asterisk (a couple years old) and one of them mentioned how someone tested the limits of Alison Smith
05:32.21*** join/#asterisk Beirdo (n=gjhurlbu@unaffiliated/beirdo)
05:32.27Echinossending her progressively nastier stuff to say
05:32.32QwellThere are a few words that she won't say.
05:32.40QwellI know of one in particular
05:32.51EchinosWhy, have you asked for it? ;)
05:33.04Qwellno, just heard it a bunch of times
05:33.08EchinosAh
05:33.14codefreezeEchinos: Have ya seen my telemarketer torture stuff?
05:33.15Echinosbut not from her, apparently ;)
05:33.21Qwellright, heh
05:33.25Echinosum, maybe ;)
05:33.34Echinosthat's one of the things I want to do
05:33.39Echinossend me a link?
05:34.12QwellEchinos: where'd you hear this presentation?
05:34.25codefreezeEchinos: voip-info
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05:34.34EchinosI'm a relative noob with *, I installed it a few years ago to play, but never went anywhere
05:34.43Echinosumm... defcon archives, sec....
05:35.04Echinoshttps://www.defcon.org/html/links/defcon-media-archives.html
05:35.10Echinossearch the page for asterisk
05:35.23Echinostheres mp3/mp4/pdf i think
05:35.35Echinosbut the sound isn't too good, there's some sort of interference
05:35.36Qwellmp4...video?
05:35.39Echinosya
05:35.50Echinosthey have a speed dialing competition for swag ;)
05:36.07Qwellspeed dialing a phone number?
05:36.20Echinoscodefreeze: where's your telemarketer stuff?
05:36.22Qwellrotary, dtmf, or box? ;)
05:36.29Echinoscodefreeze: sorry, missed ^^
05:36.33Echinosheh
05:36.51Echinosdtmf, but they said they'd have done hook-tapping if they could
05:36.53Echinosbrb
05:42.01_x86_hooker-tapping is fun ;)
05:44.13Echinosumm... I'll take your word for it
05:44.49Echinosso, I'm thinking of getting an unlimitel account
05:44.54Echinos(I'm in Canadia)
05:45.16Echinoswanna see how/if I can save some $$, make a sweet system
05:45.25Echinosetc.
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05:47.54Echinoscodefreeze: is yours the one that mentions "bad dialplan design"?
05:48.04codefreezebingo
05:48.22codefreezeonly the best for telemarketers....
05:48.29codefreezethe best of the worst
05:49.43Echinoshehe
05:50.11EchinosMy other idea was to make a recording that would sound like someone actually answered, and to record it and play back later for fun
05:50.37Echinosput in the odd "hmmm" and "go on..."
05:50.54Echinosjust to keep them talking, kinda like a tarpit for telemarketers
05:51.14Echinosmaybe throw in some farm animal noises etc.
05:52.03russellbit's quite fun to make people call each other
05:52.07Echinoscodefreeze: did you record all the sounds for that?
05:52.21Echinosrussellb: how do you mean, exactly?
05:52.30EchinosMake telemarketers call each other?
05:52.35russellbsure :)
05:52.37russellbor 2 friends ...
05:53.15russellbit's quite easy.  I usually make a script that takes 2 phone numbers and generates 2 call files.  each call file originates a call to one number with the callerid of the other number, and connects them both to a MeetMe
05:53.33russellbthen I connect listenonly into the meetme, and usually record it, too, in case it's funny.
05:53.45Echinoswow
05:53.49Echinosgenius
05:54.04Echinostoo bad most telemarketer numbers are no-incoming
05:54.37Qwellrussellb: ever tried 3 at once?
05:54.47russellbQwell: yeah.  i did about 20 at once in a class
05:54.48Qwellthat'd be confusing
05:54.51Qwellhaha
05:54.52Echinosare you talking about hooker-tapping again?
05:54.56Echinos;)
05:55.01russellbi collected the cell phone numbers of everyone in class, and had them all ring at once
05:55.04Qwellnice
05:55.08russellband the callerid on all the phones was the teacher's
05:55.09Echinossweet!
05:55.45Echinosdo a huge conference for all of them
05:56.03Echinosdid you let on that you did it?
05:56.13russellbyeah, it was part of a presentation
05:56.19EchinosAh, ok
05:56.28Qwelldid you tell them beforehand?
05:56.34russellbnope
05:56.43russellbwell, asked for the numbers
05:57.09russellbit failed miserably the first time ... the server i had set up to do it ran out of disk space the morning of ... :(
05:57.15russellbso i didn't know it was down until i tried to do it
05:57.20codefreezeEchinos: yep, Hmmm. lessee. http://www.parsetree.com/TeleTorturePrompts.tar.gz
05:57.30Echinosso, besides getting a low-cost voip provider, and a FWD number, what other sensible things are there to do with asterisk?
05:57.44Echinoscodefreeze: Really? wow
05:57.51Echinos*yoink*
05:58.24Echinostyvm
05:58.31russellbEchinos: set it up as extremely overkill voicemail?
05:58.50EchinosWell, yeah, that's been mentioned ;)
05:58.54russellbok ok ... um.
05:59.01Echinosbrb
05:59.08russellbset up all the phones in your house as agents in a queue
05:59.17codefreezeEchinos: The End of Telemarketing calls / Charity Seekers. Haven't had but maybe 3 or so over last 3 years.
05:59.43russellband have the queue use the random strategy ...
05:59.48russellbyou never know which one is going to ring!!
06:00.03codefreezelol, the wife'd just LOVE that!
06:00.17russellbooh, it's in the bedroom, run!
06:00.35denonhmm, any aussies awake? trying to find out if this Telstra Next-G stuff is any good
06:01.24codefreezeSounds like the phone up in the attic is ringing, better run!
06:01.41russellbwith a ringtimeout of 5 seconds
06:01.48denonsupposed to be faster than EVDO
06:01.51russellbyou'll be all over the house trying to get the call :)
06:01.52denonthough I'm skeptical of the latencies
06:02.33russellbg'night all
06:03.17codefreezedenon: my latencies are so low, I can say something in one extension, and just hear the last half of it after racing to the other extension at the other end of the house!
06:03.48*** join/#asterisk Chunkz (i=Chunks@ool-43536cf6.dyn.optonline.net)
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06:03.52denonhehe
06:04.19denonnot quite what I meant
06:04.22denonwhere's JT anyway
06:07.22ChunkzI got a really newbish question... I haven't touched * in months and years. I was wondering if AsteriskNOW will be just as up to date as the core Asterisk system? Is there any real reason to not use *NOW if I'm happy with a UI?
06:10.38JoelSolankihi.
06:10.52JoelSolankii want to compile new cvs zaptel
06:10.56JoelSolankihttp://svn.digium.com/view/zaptel?view=rev&revision=2986
06:11.11nclxwell I would think that if there are newer options in the development branch of 1.4 they might not be exposed in the *NOW UI, but that doesn't mean you can't drop a command line and asterisk -r and fire away
06:11.19JoelSolankii downloaded zaptel-base.c and zaptel.c and kept in /usr/src/zaptel
06:11.33JoelSolankiand started compiling but it is giving me error
06:11.35nclxof course *NOW is probably not distributed with the most bleeding edge development release of *]
06:11.55nclxdid you download all of zaptel or just those files
06:11.58nclx?
06:12.05JoelSolankijust those 2 files
06:12.11JoelSolankiis that ok ?
06:12.12nclxdownload the whole directory
06:12.30EchinosTelephone scavenger hunt
06:12.32nclxno it is not that is why it failed, you just download it and run 'make', 'make install'
06:12.36Echinossounds fun
06:13.25JoelSolankinclx: i m having sangoma t1 card so i m going wanpipe directory and giving command ./Setup Install
06:13.38JoelSolankiwhich will update and compile the new zaptel
06:13.47JoelSolankilet me find the cvs
06:15.21EchinosI'm thinking of getting a cell plan that lets me call home for cheap, and then route thru * for $$ savings
06:15.54JoelSolankinclx: how do i download the whole directory of CVS ?
06:15.58JerJerEchinos:  find a friendly DID provider and add that DID to your 'fav-5s' or any of the other variants
06:16.00JoelSolankii m not familiar with CVS or SVN
06:16.20EchinosI'm on call for work (IT), so I was thinking of having a number to call for clients that I could check and call back for cheap
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06:16.40nclxHow do I setup an invalid extension handler?
06:16.46EchinosJerJer: that's kinda what I mean
06:17.03JerJerword
06:17.22JerJerthere are pretty kick ass GSM gateways now that talk SIP - which is your other option
06:17.22EchinosI dunno how much cheaper it would be, but it would be worth a try
06:17.32JerJerdrop in a sim, configure the sip and rock on
06:17.54EchinosJerJer: so, your cell phone dials overGSM to SIP, and then voip?
06:18.16JerJeryeppers
06:18.36EchinosSo then you can just make your cell route to your * box, or call any SIP # for cheapness?
06:18.37JerJerthink ATA with a GSM radio
06:18.43Echinosneato
06:18.54JerJerinstead of a fxs port
06:18.56Echinosdo you have a canadian example? ;)
06:19.20JerJeri haven't tried any canadian providers
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06:19.46JerJeri've done mostly europe or africa craziness
06:20.21JerJerbut I did get to play with one here at the bat cave (in the US) for a month, which was pretty fun
06:20.45JerJeri was hoping that unit would 'fall off the truck' but alas, it didn't
06:21.10EchinosI wonder if it would be good to have calls go to the * box and then have it sms me on cell, then I can call back
06:21.24EchinosMaybe I just need a pager ;)
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06:26.36JoelSolankihey guys i want to download this zaptel from svn. how do i do this ? http://svn.digium.com/view/zaptel/branches/1.4/?pathrev=2986
06:27.04JoelSolankisvn checkout http://svn.digium.com/svn/zaptel/branches/1.4/ zaptel   ????
06:29.38JoelSolankiany body plz ?
06:31.25JT????????????????////
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06:35.12JoelSolankihey guys i want to download this zaptel from svn. how do i do this ? http://svn.digium.com/view/zaptel/branches/1.4/?pathrev=2986
06:35.17JoelSolankisvn checkout http://svn.digium.com/svn/zaptel/branches/1.4/ zaptel   ????
06:36.12JTJoelSolanki: stop motherfucking repeating
06:36.15JTit's rude as all hell
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06:39.19JoelSolankiok
06:39.44JoelSolankithanks
06:39.58JoelSolanki<JT>
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06:57.48NuitariHi there, is there a way to make a patch from the branch in bug #6119 against the main source for 1.4.11? (or other nearby svn/trunk)
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07:00.39neoalexhi guys I have a question about CID manipulation, incoming not outgoing
07:01.02neoalexI have a provider that sends me caller ID in 9 digit form
07:01.21Nuitari9 digit form?
07:01.24neoalexI want to add a 0 in front of that number before it gets dispayed on my phone
07:01.32neoalexyes 748123456
07:01.44Nuitariwhich country are you in?
07:01.45neoalexand when I dial out I use 0748123546
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07:02.06neoalexI'm in the US but that provider is in romania
07:02.20neoalexand I use it to call romanian phones and also get calls from romania
07:02.31Nuitariok
07:02.50neoalexand instead of using the country code every time I dial romania I just make the difference with the 0
07:02.53Nuitariwell I never gave any though on doing it, but I'd like to know so that I can prepend a 9 automatically
07:03.04neoalexall romanian numbers have 0 at the begining while us numbers don't
07:03.28neoalexexactly, I wanna do the same thing with a 0
07:03.43neoalexotherwise if I have a missed call on my phone I can't dial it directly
07:03.47FlatFootNuitari:  It used to be SetCallerId(0${CALLERID})
07:04.01FlatFootin your dialplan in extensions.conf
07:04.13neoalexon the incoming extension
07:04.15neoalexright?
07:04.18FlatFootyep
07:04.33neoalexpriority 1 then 2 is the dial
07:04.43neoalexok... le'me try
07:04.56Nuitariisn't setcallerid deprecated?
07:05.22FlatFootnot sure if it is , might now be Set(0${CALLERID})
07:05.45FlatFootSetVar has changed to Set , so it might be the same
07:05.46Nuitari<PROTECTED>
07:06.08FlatFooti'm still using old versions
07:06.25Nuitariyeah I noticed, deprecated in 1.2, removed in 1.4
07:07.12Nuitariso Set(CALLERID(num)=0${CALLERID(num)})
07:07.15NuitariI'll try that
07:07.20FlatFootsounds good
07:09.20Nuitarithat works
07:10.55FlatFootnice
07:11.26FlatFooti have to use that because my telco never sends the leading 0
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07:12.01Nuitarihere we still prepend a 9 for an outside line
07:12.52FlatFootyep i use a nine to get outside then strip it to leave my *
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07:13.10neoalexyoup that worked... thanks
07:13.51FlatFootnp
07:16.49neoalexcool... that's going to make life easier
07:17.18neoalexoh one more thing... can I have voicemail greetings in different languages?
07:17.23Nuitariit's nice to get a glimpse at upcoming polycom stuff in their firmware update info
07:17.24FlatFooti'll go for that anytime , could do with an easy life
07:18.03NuitariI know you can have different context for voicemail
07:19.25neoalexyeah but I want it going to the same mailbox
07:19.42Nuitarihum
07:19.55Nuitariyou could write an outside script to move the files around on a regular basis
07:20.02Nuitarithough there must be a less convulted way
07:20.27neoalexyeah... I'm not using the busy message for anything
07:20.36neoalexso I'll use as a greeting in the second language
07:20.45Nuitarithat could work too
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07:27.12neoalexwhat's the temporary greeting, the mailbox setup tells me about it but I don't see it in the wiki
07:27.40Nuitariit overrides both of the other greetings
07:27.55Nuitarieg if someone goes on vacation or something
07:29.09neoalexah ok
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08:54.32JoelSolankihi all
08:55.07JoelSolankii have installed asterisk 1.4 and installed asterisk gui but digital config (beta) is not working.
08:55.13JoelSolankii m using sangoma t1 card.
08:55.30JoelSolankiit is not identifying the sangoma card with gui ?
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09:27.13zapp-braniganhi in a direct call iax2/user:password@ip   there is any way to select the codec ?
09:33.15zapp-braniganand i can change the default codec in the asterisk ?
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09:59.55*** join/#asterisk TylerJGillies (n=tyler@cpe-66-91-11-212.hawaii.res.rr.com)
10:01.12TylerJGillieswhat is asterisk in a nutshell, i'm brand new and all the technical information is kind of boggling
10:01.23TylerJGilliesi've developed IVR applications before. is it like an IVR platform?
10:12.29Blue_IceTylerJGillies: though question, asterisk is a pbx, with IVR, with conf call, with voicemail, with ...
10:13.25TylerJGilliesso IVR is built in?
10:13.42TylerJGilliesdoes it use standard VXML?
10:14.05Blue_Iceyeah, you have total control over the callflow, so you can use IVR routines in it
10:14.58Blue_IceTylerJGillies: I believe so (according to http://www.voip-info.org/wiki/view/Asterisk+cmd+Vxml , never played around with voicexml myself)
10:15.35TylerJGilliesi made custom IVR systems for all my friends with my voxeo developer account heh
10:16.25Blue_Icein a nutshell: Asterisk is anything you ever wanted for VoIP :-)
10:18.01TylerJGilliesawesome
10:18.40TylerJGilliesim deploying a co-located server and i'm in the process of deciding what platforms i'm going to support
10:19.36TylerJGilliesis it only for linux or for windows too? i was thinking of deploying a windows 2003 server because voxeo's prophecy platform runs that, but if i can get asterisk to do what i want i can save some money
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10:20.17Blue_Iceyou could use it on windows with cygwin, but not really recommended
10:23.10TylerJGilliesi'm not running cygwin on a devlopment server heh
10:25.22Blue_IceTylerJGillies: that's why it's not recommended :-)
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10:29.26JTwhy would you run a phone system on windows?
10:30.43JTand vxml being standardised is highly debatable
10:30.52MavvieJT: You should ask Cisco about that.
10:31.05JTheh
10:32.37TylerJGilliesthere is a standard people just choose not to be compatible
10:32.57JTuhuh
10:33.11TylerJGilliesjt: voxeo's system is pretty good, and it runs on windows
10:33.22JTwe don't care :)
10:44.16TylerJGilliesok
10:44.36TylerJGillies<JT> why would you run a phone system on windows?
10:44.38TylerJGilliesyou asked whyu
10:44.43TylerJGilliess/whyu/why/
10:45.16JTnot the pinacle of reliability
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10:48.46konqi_I can't dial out via my Digium TE420. Can anybody help?
10:48.54jsmith-awaykonqi_: Sure, I'll try
10:49.04jsmith(and a good morning to everyone!)
10:49.12konqi_good morning too
10:49.36jsmithkonqi_: OK... let's see what's going on.... first, give me a "pri show span X", where X is the span number (1, 2, 3, or 4)
10:50.13konqi_should i post that here?
10:50.35konqi_or maybe pastebin somewhere...
10:51.30konqi_jsmith: http://pastebin.com/m128f04d1
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10:53.09jsmithkonqi_: OK, good... both spans are up and active
10:53.26jsmithkonqi_: OK, can you paste the line from your dialplan that actually does the Dial()
10:53.41jsmithkonqi_: (and the output on the CLI when it tries to dial)?
10:54.17konqi_mh i don't know which macro it uses to dial actually
10:55.37jsmithAt the Asterisk CLI, type "core set verbose 3"
10:55.46jsmithThen dial, and see what pops up on the CLI
10:55.51konqi_okay
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10:56.44yannj_frhi all
10:56.45yannj_frl
10:56.54jsmithHello yannj_fr
10:57.27yannj_frI am looking for a way to declare phones in district, and set the codec to use between district, any idea?
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10:58.39konqi_i did "set verbose 3" with "core" it says "no such command 'core'"
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10:59.46TylerJGillieshow do i call asterisk demo? its extention 1000 but what sip number do i call?
11:00.15JunK-Yjsmith: 7am on a saturday? Goto(bed)
11:01.09konqi_jsmith: http://pastebin.com/md68da9a
11:01.55jsmithJunK-Y: Can't help it... I've been awake for two hours now
11:03.12jsmithkonqi_: OK, you're dialing an international number, right?
11:03.21konqi_right
11:03.37jsmithkonqi_: OK, you may need to adjust the "pridialplan" settings in zapata.conf
11:03.56jsmithkonqi_: As Asterisk is telling the remote swith that you're dialing a national number, and not an international number
11:04.35konqi_so i'll set pridialplan to international?
11:04.56jsmithkonqi_: Also look at the "internationalprefix" setting there... that tells Asterisk 'if the number starts with 00, it's an international number'
11:05.07TylerJGillieswhat softphones do you use with asterisk?
11:05.28jsmithkonqi_: Yes, set it to international to test, but then set it back to national and play with 'localprefix','nationalprefix', and 'internationalprefix'
11:05.54JTwe generally try and avoid softphones
11:05.54jsmithTylerJGillies: XLite, Twinkle, Idefisk/Zoiper, IAXPhone Pro
11:06.14jsmithTylerJGillies: I don't use any of them for production though, just for testing
11:06.33TylerJGillieshow do you dial an extension?
11:06.43JT~thebook
11:06.43jbotrumour has it, thebook is a book called  Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11
11:06.58TylerJGilliesis it a free book?
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11:08.06konqi_hmmm still says national number....
11:08.22tzafrir_homeTylerJGillies, CC NC
11:08.30TylerJGilliescc nc?
11:09.09tzafrir_homereleased under the Xreative Common atribution and no-commercial use
11:09.34JTCreative
11:09.39tzafrir_homeso yes, you can freely download it
11:09.48TylerJGilliescool
11:09.50TylerJGilliesthnx
11:09.56tzafrir_homeNot (yet?) version 2, I guess
11:10.04jsmithTylerJGillies: The second edition will be out shortly (I'm one of the authors)
11:10.15TylerJGilliesjsmith: awesome!
11:10.29jsmithtzafrir_home: I got my hard-copy version of the second edition on Thursday... PDF should be posted any day now
11:11.14tzafrir_homeCan't wait :-)
11:11.37jsmithkonqi_: You may have to restart Asterisk for those changes to take effect
11:12.01jsmithtzafrir_home: This should show how much better the second edition is: http://upload.zooomr.com/photos/jsmith/3087512/
11:12.19konqi_jsmith: i did a reload...
11:12.27yannj_frI have the PDF
11:12.48yannj_frbut it is written "safary propriety"
11:12.56jsmithkonqi_: I don't think a reload is enough...
11:13.05jsmithyannj_fr: Oh, of the second edition?
11:13.10yannj_fryes
11:13.16konqi_jsmith, okay - did a restart... testing now
11:13.33yannj_frbut I did a copy 'without disclaimer'
11:14.28tzafrir_homeIs the starfish included with the book, or only if you're the author?  :-)
11:14.29jsmithYeah, I keep forgetting that people can buy a copy... (which is fine with me, but it'll also be available for free)
11:14.36Davieyyannj_fr: I'm guessing it's not yet CC then?
11:14.54jsmithtzafrir_home: Sorry... I picked that up in Greece a couple of weeks ago.  I'd be happy to show you where you can get your own.
11:15.21jsmithDaviey: Yes, it's CC yet... I can show you a picture of the copyright page :-)
11:15.29TylerJGillieshttp://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 isn't connecting
11:15.36DavieyTylerJGillies: same here
11:15.44konqi_jsmith: looks like this now: http://pastebin.com/m41a7060b
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11:16.00Davieyyannj_fr: so can you redestribute it before it'sannounced, ie now?
11:17.37jsmithTylerJGillies: Yeah, the website has been up and down the past couple of days... I'd be happy to email the first-edition PDF to you.
11:17.52TylerJGilliesjsmith: please do. that would be much appreciated
11:17.54jsmithTylerJGillies: (and the second edition, once I have a copy)
11:18.00TylerJGilliesjsmith: tjgillies@gmail.com
11:18.05jsmithTylerJGillies: Assuming I can find it on my hard drive, that is...
11:18.53Davieysecond edition based on 1.4?
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11:19.31jsmithTylerJGillies: http://www.nufone.net/downloads/asteriskdocs/AsteriskTFOT.zip
11:19.34jsmithDaviey: Of course!
11:20.14Daviey\o/
11:20.25konqi_jsmith: added the pri debug span 1 ... cli-output here: http://pastebin.com/m17941210
11:20.35jsmithDaviey: Another reason for the delay is that O'Reilly decided to host the new PDF directly from their servers, so they could get a better sense of how free downloads might be affecting book sales
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11:21.24Davieyjsmith: ah.. If we want more CC books avaliable in PDF - we had better spread the word to grab it from mirrors then :D
11:22.09Davieytbh.. i usually buy useful paper copies; but like to get a feel - and PDF usage for 'in the field' use
11:22.42jsmithDaviey: No, the best thing you can do is tell everyone to get it directly from O'Reilly's website, but then *still* go out and buy the book :-)
11:22.58jsmithDaviey: To be honest, it's hard to curl up in an easy chair with a PDF
11:23.23Davieyespecially when lapppy's fan is broken and it burns legs
11:23.28jsmithkonqi_: You'll probably have to call the phone company and work with them on this problem... it seems we're doing everything we should, but they're not liking something
11:23.58jsmithDaviey: That laptop will come in handy when the next ice age hits us :-)
11:24.14Davieyheck, we use my laptop as a fireplace
11:25.20konqi_jsmith: i read somewhere about asterisk dialing out too soon or too fast for the telco - maybe there could be a possible solution
11:27.01jsmithkonqi_: Nope... that's never a problem on ISDN connections... that's only on analog
11:27.31konqi_jsmith: old pbx is still working on the pri... it's now connected to the second span - where we can dial out - but not it
11:31.54jsmithkonqi_: And the other pbx isn't somehow manipulating the digits before sending them to the telco?
11:31.56TylerJGilliesjsmith:                                                        As time went on, I was finally
11:31.56TylerJGilliesable to figure out enough to get Asterisk up and running.
11:32.02TylerJGilliesi love that quote
11:32.31jsmithTylerJGillies: You have no idea how long it took me to start to understand Asterisk... think *months*, not days...
11:32.43konqi_jsmith: now that's a good question - however - if we might get the old pbx to work on trunk 2 then maybe we could find that out
11:32.59TylerJGilliesjsmith: i don't feel so dumb now. i have no SIP VOIP experience besides writing VXML scripts heh
11:33.15jsmithTylerJGillies: When I started with Asterisk, the general consensus was "If you can't read the source code and figure out out from there, this isn't for you."
11:33.22jsmithTylerJGillies: Luckily, that's changed :-)
11:33.34TylerJGilliesjsmith: hardcore heh
11:33.36konqi_jsmith: for the old pbx it says:  -- Extension '' in context 'from-pbx' from '6778200297' does not exist.  Rejecting call on channel 0/31, span 2
11:33.49TylerJGillieselitism stops so much productivity
11:34.10TylerJGilliesK.I.S.S.
11:34.27jsmithTylerJGillies: This wasn't elitism... it's just that there were *so few* people using Asterisk, nobody had bothered to try to document it
11:34.57jsmithTylerJGillies: In December of 2001, there were only like twelve people on the Asterisk mailing list...
11:35.33jsmithkonqi_: Very strange... it's trying to dial *no* extension at all...
11:36.18TylerJGilliesjsmith: dang... good thing i'm starting now and not back then. but its due to the efforts of people like you that i'm even starting at all
11:36.51konqi_jsmith: well - that's what comes up when i just pick up the phone.... it's waiting for da dial tone i guess
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11:37.24jsmithkonqi_: You don't have "immediate=yes", do you?
11:37.50jsmithTylerJGillies: Well, I ain't all that and a bag of chips... but I try to help out the community :-)
11:38.10konqi_jsmith: it's immediate=no atm
11:38.12TylerJGilliesjsmith: you got more chips in your bag than i do.
11:39.23jsmithkonqi_: Then it shouldn't be dialing as soon as you take the phone off the hook
11:39.55jsmithBack in a little while...
11:40.26konqi_Good hunger anyways :-)
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11:57.57TylerJGilliesanyone know of a good hosted asterisk solution?
11:58.23*** join/#asterisk Koshatul (n=evangeli@ppp188-33.static.internode.on.net)
12:05.30*** join/#asterisk mitcheloc (n=mitchel@ppp-68-120-83-92.dsl.irvnca.pacbell.net)
12:08.40*** join/#asterisk Trentster (n=Trike@d58-104-33-175.dsl.vic.optusnet.com.au)
12:10.01TrentsterHey all, just a quickone, on my snom phone when I put someone on hold they hear the music on hold but from my side i Hear a long high ptched tone, until I take them off hold. What do I need to do in order for me to hear the music on hold as well, as the person being put on hold?
12:21.42WildPikachuhow does one comment the extensions.ael file?
12:21.52*** part/#asterisk TylerJGillies (n=tyler@cpe-66-91-11-212.hawaii.res.rr.com)
12:25.54WildPikachuhrmmmm
12:25.59WildPikachu; appears to generate errors
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12:26.30WildPikachuaha
12:39.20tzafrir_homeWildPikachu, // ?
12:39.28WildPikachuyea ... just found that, thanks
12:45.26*** join/#asterisk masus (n=tet@88.248.73.2)
12:45.29masus"ERROR[25340]: func_odbc.c:313 acf_odbc_read: Unable to load ODBC read class (check res_odbc.conf)"
12:45.35masusanybody an idea
12:45.45masusisql is working fine
12:45.49masuswith odbc
12:49.41*** join/#asterisk kFuQ (n=somedude@c-67-185-112-20.hsd1.wa.comcast.net)
12:51.37*** join/#asterisk centrex (n=mythtv@c-68-62-217-48.hsd1.al.comcast.net)
12:53.05VJFROMGTi need to record the ip for every incomming call, can someone tell me where to get help,, willing to pay
12:54.49centrexVJFROMGT, I would think cdr would record that information wouldn't it?
12:54.59VJFROMGTit does not
12:55.08VJFROMGTwhic is why i want it
12:55.59*** join/#asterisk kFuQ (n=somedude@c-67-185-112-20.hsd1.wa.comcast.net)
12:56.12masuscan u get the ip
12:56.36*** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25)
12:56.37VJFROMGTi am assuming it can be taken from sip packet or sip peer
12:57.14masusif u can get it u can store it vi func_odbc to db
12:57.58masusor with application MYSQL directly t mysql
12:58.09VJFROMGTi dont know how to ,, do u know who i can get to do it for me
12:58.12VJFROMGTwilling to pay
12:58.24masusi don't want money
12:58.30masusbut i want to help one mom
12:59.26*** join/#asterisk Ebola (n=Ebola@host86-138-6-48.range86-138.btcentralplus.com)
12:59.33masusSIP destination domain of an inbound call
12:59.37masusis this what u need
12:59.57masusone mom i'll test it
12:59.57VJFROMGTi am not clear what that means
13:00.18VJFROMGTi want to know what ip is making the call
13:00.35VJFROMGTor what ip an extension is on when it makes a call
13:01.05VJFROMGTmanus,can i pm u?
13:01.15centrexVJFROMGT, You could set the user caller ids to be their ips in sip.conf, and then record the callerid variables with a dialplan script, possibly?
13:01.15masusyesnp
13:01.28masusi have do it
13:01.37centrexnot the best solution but it's a kinda dirty hack I guess.
13:02.23VJFROMGTcentrex , it does not work  as my uses do not authenticate by username/passw
13:02.50centrexah, sorry
13:09.20*** join/#asterisk oej (n=olle@soll4-125.cust.blixtvik.net)
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13:24.19WildPikachua fxs card is fxo from the perspective of asterisk?
13:27.08blitzrageWildPikachu: it uses FXO signalling -- it still looks like an FXS port though
13:27.23blitzrageFXS ports use FXO signalling, and FXO ports use FXS signalling
13:36.43*** join/#asterisk domger (n=domger@ip150.202-51-69.sogetel.net)
13:38.11zimHi all i have a isdn2e line into my office with a block of 10 DDI numbers what card would i need to let my asterisk server us it ???
13:44.59WildPikachublitzrage, so its the opposite way around  :)
13:49.36blitzrageWildPikachu: it might appear that way
13:55.06Davieyzim: a BRI card :)
13:55.06*** part/#asterisk masus (n=tet@88.248.73.2)
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14:02.19So3krishello
14:02.35So3krisuse people pap2 with 1.4.11
14:05.29DavieySo3kris: people do use the PAP2 with * 1.4, yes
14:06.26So3kris1.4.11 ?
14:07.23So3krisI had upgade the 1.4.2 to 1.4.11 and my pap2 don't work
14:13.15zimDaviey Thanks
14:14.35zimDaviey you also helped me yesterday with the conf for sipgate ty for that still not working but a few steps forward
14:16.02zimps what does BRI stand for
14:16.17JTbasic rate interface
14:16.51zimty
14:18.19*** join/#asterisk famicon (n=pastry@c51447ddc.cable.wanadoo.nl)
14:18.49tzafrir_homeIs it BRI or PRI?
14:19.09tzafrir_home(2 Bchanns or much more?)
14:20.16JTit's bri that he's talking about
14:20.24tzafrir_home~wiki BRI
14:21.04zimwould any of the following work ???
14:21.04zimhttp://search.ebay.co.uk/search/search.dll?sofocus=bs&sbrftog=1&maxrecordsreturned=300&frpp=50&from=R10&satitle=bri+card&sacat=-1%26catref%3DC6&sargn=-1%26saslc%3D3&sadis=200&fpos=DT2+0PY&sabfmts=1&saobfmts=insif&ga10244=10425&ftrt=1&ftrv=1&saprclo=&saprchi=&fsop=1%26fsoo%3D1&coaction=compare&copagenum=1&coentrypage=search&fgtp=
14:21.17tzafrir_homezim, it can be handy to look things up in wikipedia , http://voip-info.org or whatever before asking here...
14:21.42zimit was just out of intrest
14:21.52zimcalm down
14:22.05JTi don't see anyone not being calm
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14:23.56zimis ok JT i was just pointing out that asking a little Q is not a bad thing
14:24.28JTall in your judgement i assume :)
14:24.58*** join/#asterisk adorah (n=Michael@87.69.57.73.cable.012.net.il)
14:25.12tzafrir_homezim, no offence :-) Just for future use
14:26.01adorahHi tzafrir_home whass'up?
14:26.07zimI use wikipedia all the time anyway this is much apu about nothing
14:26.29JTzim: because you made it into drama :)
14:26.51zim" zim, it can be handy to look things up in wikipedia , http://voip-info.org or whatever before asking here..."
14:27.02JTyes that was helpful advice
14:27.03zimanyway I have move on
14:27.05JTyou say "thanks"
14:27.30zimmoving on
14:27.51JTlot's of moving
14:28.33tzafrir_homehi adorah
14:29.00zima Q that wiki* wont answer in the link I posted is a few cards what are the pro's/con/s/won't works with them ?
14:29.43JTactually the wiki talks all about bri chipsets and drivers
14:29.59*** join/#asterisk etix (n=etix@nala.l0cal.com)
14:30.35zimlets close down IRC
14:31.49JTyou should get a hfc card and bristuff
14:32.19zimTY
14:32.57JTjust don't make up bs about it not being on the voip-info wiki :)
14:35.28zimi never said that at all all i was saying is I can spend all afternoon trying to work out a prob (RFM) or ask in here and if someone like you or Daviey who are kind enough to give a quick answer it great and Y i love GNU
14:35.45*** join/#asterisk keulin (n=cray@AMontpellier-152-1-103-78.w86-211.abo.wanadoo.fr)
14:36.22JTthe reality is that bri is a pain in the arse in asterisk
14:36.33JTand all the different solutions have their shortcomings
14:36.35zimwhy ?
14:36.45JTbecause asterisk is from america
14:36.50zim:)
14:36.54JTand bri is rare there
14:36.59JTand there's no etsi bri there
14:37.54zimwhat about analogue lines
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14:38.07zimie bog standard bt line
14:38.36zimok let me fill u in on where i am at
14:38.46JTyes they work
14:38.47JTbut
14:38.50JTanalogue is evil
14:38.59blitzrageanalogue is practically useless
14:39.21blitzragebetter off using an ITSP in my opinion
14:39.33JTwell not for reliability reasons
14:39.33blitzragesome people have success with the Linksys ATA devices though
14:40.05blitzragemaybe I've just used the wrong hardware on analogue... but it always seems like such a pain in the ass, and not very reliable -- although I avoid it so I don't have the "tricks" some people might have
14:40.17JThangup etc is unreliable
14:40.24JTbut in terms of circuit availability
14:40.34JTVoIPoI is a gamgle
14:40.38zimin our office we have an isdn2e (bt versatility) digital for analog calls and a analogure line for internet (only in the UK)
14:40.38JTgamble
14:41.46zimoh yes and we live in the sticks so cant get 100% uptime
14:42.15zimso where i am aiming is bit of everything
14:42.48zimsipgate isdn2e landline loads of fallover
14:43.07tzafrir_homewell, bristuff doesn't have to be *that* painful
14:43.44JTit's fairly reliable with hfc
14:43.46JTstill some bugs
14:44.10zimalso i am just getting into voip so dont know whats good/bad/ok etc
14:47.59tzafrir_homeduh. Is there an easy way to go back to an earlier version on the voip-info wiki?
14:48.37tzafrir_homeJust managed to remove most other vendors from the "asterisk hardware" page :-(
14:48.56JTdiff
14:48.59JThistory
14:49.04JTdunno if it support reverting
14:50.06zimback to sipgate what ports will it want to have open back to my asterisknow server ???
14:51.24zimat the mo i have 5060,5061,4000,4001,4004,4005
14:51.35JTweird selection
14:51.40zimare there any others ???
14:51.41JTudp 5060, 10000-20000
14:51.50JTmost of those you listed aren't needed
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14:52.52zimim using a linksys pap2 to connect and x-lite all just playing trying to work it out (lots of fog that will all make sence oneday)
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14:53.51muh-die-kuhhi
14:56.18muh-die-kuhi just set up a asterisk box, connected to sipgate. with my gxp2000, i can call my normal isdn phone. otherway round doesnt work. if i call my isdn phone with my voip phone, i can hear everything said into the voip phone out of the isdn phone, but otherway round doesnt work. can someone give me some hints regarding this? i guess its something with my nat, i think i allready read sth. about this issue, but i cant figure out, where...
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15:06.59tzafrir_homeJT, does't really.
15:07.16tzafrir_homeI ended up copying from the "osurce" of an earlier revision
15:10.48*** join/#asterisk nighty^ (n=nighty@sushi.rural-networks.com)
15:11.39JTtzafrir_home: fair enough, they're not as anal as wikipedia about that sort of thing anyway
15:11.57JTi often wonder if voip-info is ever maintained by the hosts at all
15:12.30tzafrir_homeA wiki hsould not be maintained by the "hosts". It should be maintained by "a bunch of users"
15:13.06tzafrir_homeIf you won't wantch some pages, who will?
15:14.14Davieyzim: what's the problem with sipgate.co.uk & your settings?
15:14.22DavieyThat conf i provided should be dandy
15:15.32zimDaviey i think it was it did not carry the hold on will find it
15:16.07ziminsecure = port,invite
15:16.24Davieythat's what you need with sipgate & asterisk 1.4
15:16.25zimso i just edited the user.conf
15:16.55zimi added what you sent to providers.conf
15:17.12zimthen in the gui added sipgate
15:17.48Davieyand what happend?
15:17.48WildPikachuwould anything prevent me from using system()?  even a    echo hello > /tmp/file.log    gives me an error
15:17.50zimbut it did not carry across to the user.conf
15:18.01JTtzafrir_home: as in stopping the site from falling apart, blocking spammers
15:18.10Davieyzim: :s, it should!
15:18.21WildPikachu[Sep  1 17:18:02] WARNING[16199]: app_system.c:105 system_exec_helper: Unable to execute '"echo hi > /tmp/ast.log"'
15:18.28zimi would think so but it did not
15:18.43zimwill do it again and pastebin it to show you
15:18.49zimhold on#
15:19.54tzafrir_homeBut unlike wikipedia, they don't allow non-subscribers to fix content
15:20.11Qwelland they have ridiculous licensing
15:20.15Davieyzim: it should add http://pastebin.ca/677986
15:20.25Qwelltzafrir_home: ever read the voip-info terms?
15:20.32tzafrir_homehmmm...
15:20.36Qwellmakes me not want to contribute
15:20.43DavieyQwell: no.. what's the highlights
15:20.52Qwellthey're downright draconian
15:21.12zimthis in in my providers.conf http://pastebin.com/m7af290d3
15:21.53Davieyzim: looks good
15:22.15zimok will now add using the gui
15:24.10tzafrir_homeQwell, I understand you
15:24.12Davieyzim: Regarding BRI; you have 10 DDI's and only need two channels?
15:24.24zimhttp://pastebin.com/m23971b05
15:24.28tzafrir_homeDo you know of any alternative site with a good license?
15:24.43tzafrir_home(that is: no non-commercial clause)
15:25.04Qwelltzafrir_home: not that I know of..  isn't that horrible though?
15:25.44zimDaviey in the uk you buy them in blocks of 10
15:25.49WildPikachuman ... wish i could find out why my system() command is just not working
15:25.54JerJerQwell:  lets start a new one   :)
15:26.01zimif you want more than 1 you get 10
15:26.09JerJervoip-false-info.org
15:26.39zimDaviey you still about ?
15:26.44Qwellhow would it be different from voip-info? ;)
15:27.08Davieyzim: sorry.. had to go for a few mins - I am in the UK
15:27.15zimkewl
15:27.21zimyes is the answer
15:27.21Qwellthis is the line that gets me every time...
15:27.25Qwell"[You may not] - Reproduce, duplicate, copy, sell, resell, visit or use for other commercial purposes this site or any portion thereof."
15:27.28tzafrir_homeI never really understood system. Does it expect separate parameters for exec, or does it throw anything to system()
15:27.34QwellYou can't even *VISIT* the site for commercial use
15:28.01zimDaviey where in the uk
15:28.09DavieyQwell: "You may not ... vist" ?!!
15:28.15Daviey*visit*
15:28.34Davieyzim: Hants
15:28.39tzafrir_homewell, that's a problem. Because when you press the "search" box, you get to google. And google seems to abuse voip-info's page in violation of the license, right? :-)
15:28.42Qwellyes, as in, you are not allowed to go there if you are planning on making money by using what you learn in any way.
15:28.50zimHants?
15:28.58Davieyzim: Hampshire u?
15:28.58zimwhere is that
15:29.07zimok Dorset
15:29.26JerJercommpartners sucks
15:29.30DavieyQwell: bah.. i just assumed it was a CC licenced
15:30.11tzafrir_homeDaviey, "CC license" does not really mean anything. There are too many of them
15:30.52Davieytzafrir_home: Well any one of them is better than what they have
15:31.00*** join/#asterisk CrazyTux[m] (n=CrazyTux@c-98-195-23-75.hsd1.tx.comcast.net)
15:31.19Davieymay not *visit*.. how crazy is that?
15:31.32zimDaviey did you have a look at the second paistbin?
15:31.42Davieyno
15:31.54Davieyzim: missed the link
15:31.55zimhttp://pastebin.com/m23971b05
15:32.28Qwelloh, heh, you can't even modify the site without their permission either
15:32.35Davieyzim: something is wrong with asterisk-gui, it seems then
15:32.56zimasterisknow clean install
15:32.56Davieyreplace the contents with what I pastebined
15:33.01*** join/#asterisk af_ (n=getsmart@81-174-47-128.dynamic.ngi.it)
15:33.39Davieyzim: http://pastebin.ca/677986  ; use that
15:35.13*** join/#asterisk tsurko (n=tsurko@78.90.100.214)
15:35.52tzafrir_homeIt seems that they have been bitten by imposters, and want to make sure they can remove such content upright
15:36.02*** join/#asterisk saftsack (n=saftsack@p57A757D2.dip.t-dialin.net)
15:36.16tzafrir_homeApart from that, they basically reserve themselves the ability to relicense the contents
15:36.36Davieyzim: then backup your settings and drop asteriskNow in favour of a proper linux distro
15:36.48Daviey(just my opinion)
15:37.00zimno need to backup
15:37.05tzafrir_home(And from what I understand, it's more o a "he" than a "they")
15:37.12zimwas just using it to learn faster
15:37.22zimI am an ubuntu fan
15:37.29zimwill that do ???
15:37.32Davieyzim: yes
15:37.41zimwhat is the best
15:38.06Davieyzim: well if you /whois me ; You can see why i wouldn't suggest another distro <GRIN>
15:38.16tzafrir_homeIf you're comfortable with Ubuntu, use Ubuntu
15:38.51zimkewl
15:38.55Blue_Icejust use the distro which suits you best ...
15:39.01tzafrir_homeI would personally suggest Debian, but that's a different story
15:39.07QwellDaviey: When're you guys gonna fix gnome+nvidia+xinerama? :)
15:39.45DavieyQwell: erm i thought it was now working?
15:39.53Qwelldunno, I had to switch to debian
15:39.58Daviey(i could be wrong; /me doesn't use nvidia)
15:40.15Qwellit was like that for months, and nobody would acknowledge the problem
15:40.19Blue_Ice<PROTECTED>
15:40.37DavieyBlue_Ice: But i guess you do use a desktop of some sort? ;)
15:40.50zimDaviey would you use 6.10 or 7.04 server/Desktop
15:41.09Blue_IceI used to work with XFCE, but now only use linux for the servers, and wintendo / os X for clients
15:41.21Davieyzim: dapper 6.10 until, March imo
15:41.35DavieyBlue_Ice: you have my sympathy
15:41.53zimserver/desktop
15:42.10Davieyzim: Dapper 6.10 is supported, for security updates etc until 2011
15:42.26Davieyzim: 7.04 only until 2008 :`(
15:42.50tzafrir_homewell, I had more problems with 606
15:42.53Blue_IceDaviey: Lunar ftw, bye bye sympathy? ^_^
15:42.58zimyes but this is just a play box
15:43.03tzafrir_homeincluding a not-so-smooth installation of Zaptel
15:43.17Davieytzafrir_home: i have ztdummy working fine on dapper
15:43.31zimto learn the ropes with asterisk
15:44.05Blue_Icebtw, anybody using a TDM400 with a soekris net48xx?
15:44.11zimlife span aside what would you use
15:44.12Davieyzim: i would also suggest building from source, rather than packages.. When you are learning; the packages do too much for you - imo
15:44.16Blue_Icesucks that the pci-slot is inverted ...
15:45.01Davieyzim: Personally Dappper for a * dedicated server.. but tbh - it makes little difference
15:45.37zimkewl the one thing i do like about 7.04 is dash
15:45.52zimtab is my friend
15:46.03Davieyzim: careful with dash...  some scripts rely upon bashisms
15:46.19zimasterisk scripts ?
15:46.22tzafrir_homeSo those scripts should be fixed
15:46.32tzafrir_homeThat's the Debian policy
15:46.55tzafrir_home(and if it is "safe_asterisk" you refer to: you should not use ut anyway)
15:47.24tzafrir_homeA script that uses bashism should have an explicit /bin/bash
15:48.12tzafrir_homewell, the packages save you the dirty work, and allow you to focus on hte important stuff
15:48.33zimi still dont know y people still like vi compaired to vim
15:48.39zimbut they do
15:48.59tzafrir_homezim, which wi exactly? nvi? yuck
15:49.16tzafrir_homeever tried using it?
15:49.43zimsorry that was all lost on me
15:49.59zimis wi an editor
15:50.40tzafrir_homenvi is a reimplementation of the original vi. There are a bunch of other vi clones out there (elvis, viper (ahem), etc.)
15:50.58JTbut you said wi :)
15:51.00tzafrir_homeand there is vim :-)
15:51.17tzafrir_homeI mean vi
15:51.31tzafrir_homemeant
15:51.50zimok is nvi better than vim ?
15:52.07Qwelltzafrir_home: With freebsd not having a /bin/bash, how can people be expected to specify it?
15:52.15Qwelljust tell them to use a real OS? :p
15:52.55tzafrir_homeQwell, what is /bin/sh on your FreeBSD system, then?
15:53.15Qwell/bin/sh is sh, but bash is like...
15:53.25tzafrir_homewich sh exactly?
15:53.36heeliosjust sh...
15:53.39tzafrir_homewhich
15:53.54*** join/#asterisk coppice (n=chatzill@140.196.17.210.dyn.pacific.net.hk)
15:54.03zimjust ln -s /bin/bash /bin/sh
15:54.07Qwell/usr/local/bin/some/more/random/garbage/here/would/be/dumb/but/they/do/it/anyways/bash
15:54.07Qwellthere is no /bin/bash on freebsd
15:54.16tzafrir_homezim, no.
15:54.23heelioshttp://en.wikipedia.org/wiki/Bourne_shell
15:54.26tzafrir_homeThat is handled by the alternaitves
15:54.41tzafrir_home(or overrides? I don't remember)
15:54.43zimok
15:54.59tzafrir_homedpkg-reconfigure dash
15:55.12tzafrir_homebut you lose time at system startup ...
15:55.50QwellI'm not saying we should be using /bin/sh in stuff that requires bash (and I fixed a few instances of that recently)
15:55.52Qwellbut, what should be use when they want to use bashisms?
15:56.09Qwellsince /bin/bash is apparently OS specific
15:56.09tzafrir_homeQwell, but which shell exactly is /bin/sh ? pdksh? ksh?
15:56.17Qwellbourne shell
15:56.23*** join/#asterisk fx0 (n=fx0@commandline.terrorist.net)
15:56.49tzafrir_homeSo it should fail as well with safe_asterisk
15:56.59Qwellit's sh compliant, because that's what the spec came from ;)
15:57.07Qwellyeah, it should, I'm not saying it shouldn't..
15:57.22tzafrir_homedash is also posix sh compliant.
15:57.25QwellI mean, instead of #!/bin/sh, if we want to use bashisms, what should we use?  #!/bin/bash is invalid
15:57.59heeliosQwell: you could just install bash I guess.
15:58.09Nivex#!/usr/bin/env bash
15:58.09Nivex?
15:58.11Qwellon freebsd, bash is at /usr/local/bin/bash
15:58.41tzafrir_homeI fixed the init.d script by replacing &> with &2>  , because this is what was needed
15:58.43zimall i will say is Tab is my friend
15:59.05Qwell`/usr/bin/env bash` on linux for me returns nothing
15:59.06tzafrir_homeI really couldn't care less about safe_asterisk
15:59.24NivexQwell: it doesn't return anything because it invoked bash for you
16:00.04Qwellwell, that's silly
16:00.10mmlj4tzafrir_home: you really should... you might end up catching an asterisk virus otherwise
16:00.26tzafrir_homeI odn't like that form, because it doesn't easily allow adding parameters to the program (something perl people are fond of, and I also often use for -x)
16:00.59mmlj4or some other pbx disease, you know
16:01.06Qwellmmlj4: like ccm?
16:01.17mmlj4</joke>
16:02.29Qwellyou guys know much about valid cable internet signals?  specifically dBmV values?
16:02.48QwellReceive Power Level: -7.9 dBmV
16:02.48QwellTransmit Power Level: 42.5 dBmV
16:02.51QwellIs that bad?
16:03.01zimany way I am now installing ubuntu 7.10 as it has dash (hopefully all scripts have been updated) and i dont need LTS
16:03.11zimsorry 7.04
16:04.24*** join/#asterisk JunK-Y (n=junky@modemcable183.17-83-70.mc.videotron.ca)
16:04.43CCFL_Man2Qwell: i only know adsl signal properties
16:05.05zimwhat should i start by installing as I am a competent user and not a dev i want to use apt-get
16:05.06CCFL_Man2you have a cisco cable modem WIC?
16:05.32zimDaviey any ideas
16:07.10Davieyzim: bbl
16:07.26zimbbl ???
16:07.49*** join/#asterisk Strom_M (n=strom@netblock-208-127-172-112.dslextreme.com)
16:07.56QwellStrom_M: you
16:09.17Davieybe back later
16:10.11tzafrir_homeAny mandriva users here?
16:10.47tzafrir_homeI'm a bit lost with the reply I got in asterisk-users regarding the guy who had problems  building zaptel on his Mandriva
16:11.05tzafrir_homeI wonder where that "custom" came from
16:11.12*** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2)
16:12.42Strom_MQwell: me?
16:12.47Qwellyou
16:12.57Strom_Mok
16:14.07CCFL_Man2tzafrir_home: again, thanks for your help
16:15.09zimDaviey you still around
16:15.27QwellI'd like to kick somebody at Comcast...
16:15.39QwellI'm pretty sure my packets are being shaped out of existence
16:16.00CCFL_Man2Qwell: cumcast sucks
16:16.08Qwellfrom 9pm to 11am, for the last like 3 nights, I've been getting about 30% packet loss...
16:16.15CCFL_Man2Qwell: they shape traffic like crazy
16:16.21Davieyzim: bit tied up atm
16:17.06CCFL_Man2Qwell: you in charge of the premis wiring?
16:17.14Qwellyeah
16:17.53*** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga)
16:17.54CCFL_Man2Qwell: if everything is ok on that front then call cumcast and bitch at their shitty tech support
16:18.07Qwelldid, and I got the worst support person *ever*
16:18.11zimno prob just a bit of feed back to ubuntu on the 7.04 installer if you want a static ip and you are running a DHCP server on the network you have to unplug the cable so it fails before saying set it manualy
16:18.13Qwelland I've dealt with adelphia and sprint
16:18.23Qwellso that's really saying something
16:18.45CCFL_Man2Qwell: cumcast is a pathetic company, i'd suggest adsl
16:18.57Qwell(adelphia, the company who had a rep try to tell me that there are 1028 bytes in a bit (yes, seriously), is better than the idiot I talked to)
16:19.06CCFL_Man2though verizon aint much better
16:19.07QwellI don't do Bell
16:19.08CCFL_Man2lol
16:19.28CCFL_Man2oh, you don't care for your RBOC?
16:19.43Qwellno such thing as RBOCs anymore, heh, it's all AT&T
16:19.56Strom_MQwell: you don't have to get ADSL from AT&T
16:20.07coppiceIts still a Really Bad Operating Company, though
16:20.16QwellI've done the third-party DSL thing before.  I'm NOT doing that again.  EVER
16:20.20*** join/#asterisk TedNJ38 (n=HungLad@ool-4573adc7.dyn.optonline.net)
16:20.27Strom_MQwell: por que?
16:20.30QwellEarthlink
16:20.33QwellThey need to die.
16:20.42Strom_Mwell yeah, I meant a REAL DSL company
16:20.43TedNJ38Hay aqui alguien que me pueda ayudar en Espanol?
16:20.46Strom_Mnot earthlink
16:21.02Qwellwell, we don't have like dslextreme out here, heh
16:21.07coppiceQwell: They just laid off half their staff, so they are making progress toward your goal
16:21.13Qwellcoppice: good
16:21.21QwellI hope they die.  Really.  Dead.
16:21.26Qwelland I don't mean the company
16:21.29CCFL_Man2Strom_M: in my local exchange there is only the RBOC, i can't get pots from any provider except the RBOC, so i'm stuck with the RBOC's adsl
16:21.48CCFL_Man2TedNJ38: engrish only plz
16:21.59Strom_MCCFL_Man2: the CLEC doesn't have to have facilities in your specific CO
16:22.13CCFL_Man2Strom_M: nope, none in my area do
16:22.17Strom_Mthey can backhaul it to their switch
16:22.35Qwellbesides, I don't have a phone line anyways
16:22.46Strom_Mmost facilities-based CLECs only have one or two switches for a major metro area
16:22.47Qwellso, unless t-mobile wants to run dsl to my cell...heh
16:22.58CCFL_Man2Strom_M: but then i'd get crappy adsl performance because the COs that they are in are fairly far away
16:23.06TedNJ38I need help please.  I am relatively new to Asterisk.  I have  network of phones all over the US which have been scattered among my relatives.  I want to purchase a landline in Peru so whenever any of us wants to talk to them we could dial 9 + phone number and get connected to them.  And whenever any of our relatives in Peru want to call here, they dial a number and they get prompted to select who they want to talk to.  Is that possible?
16:23.14Strom_MCCFL_Man2: no, that's not how ADSL works
16:23.15QwellCCFL_Man2: from the dslam, it's all fiber
16:23.30CCFL_Man2i know that
16:23.31Qwellyou just connect to the dslam at the co, and from there it goes to your ISP
16:23.36CCFL_Man2fiber feeds the dslam
16:23.52Strom_MCCFL_Man2: and the DSLAM is *always* in your end office
16:24.01CCFL_Man2i doubt my local CO has any dslams other than their own
16:24.28QwellCCFL_Man2: your isp doesn't need a dslam at the co
16:24.34CCFL_Man2oh, the dslams are shared?
16:24.46Qwellwell, the telco owns it
16:24.51*** join/#asterisk darkfires (n=darkfire@xplr-ts-t11-208-114-158-128.barrettxplore.com)
16:24.53CCFL_Man2og
16:25.05CCFL_Man2h, but they lease circuits
16:25.14Qwellafaik
16:25.26darkfireswhen you use exten => s,n ... can you use gotoif to specify line numbers that are automatically incremented by n ?
16:25.31Strom_MCCFL_Man2: for example, my ADSL is from a company called DSLExtreme.  The DSL transport between by prem and the CO is AT&T, but then AT&T hands it off to my ISP, who backhauls it to their router in downtown los angeles
16:26.03QwellStrom_M: how do you like dslextreme?  I was looking at them when I was in covina
16:26.05CCFL_Man2Strom_M: you think verizon is that nice?
16:26.20QwellCCFL_Man2: I'm pretty sure they're forced to..
16:26.23Strom_MQwell: I've been an extremely happy customer for three years.
16:26.57Strom_MCCFL_Man2: where are you, specificaly
16:27.52coppiceQwell: the only thing a telco is forced to do is suck
16:27.55CCFL_Man2Qwell: ahh, wasn't sure
16:28.07Qwellcoppice: and wiretap
16:28.08CCFL_Man2Strom_M: northeast PA
16:28.18Strom_Mman, you people are excessively pessimistic
16:28.29coppiceQwell: only without a court order, though
16:29.02WildPikachuwhy do some config options have =>  and some  =  ?
16:29.14TedNJ38I need help please.  I am relatively new to Asterisk.  I have a network of phones all over the US which have been scattered among my relatives.  I want to purchase a landline in Peru so whenever any of us wants to talk to them (our releatives in Peru) we could dial 9 + phone number and get connected to them as a local call.  And whenever any of our relatives in Peru want to call one of us here, they dial a local number and they get promp
16:29.28CCFL_Man2Strom_M: can i get pots from another provider too using the same pair that connects to my exchange?
16:29.53Strom_MCCFL_Man2: yes; usually the PUC/PSC/FCC force the telcos to allow that
16:30.02QwellPSC?
16:30.12coppiceStrom_M: pessimism and realism aren't always easy to tell apart with telcos
16:30.16tzafrir_homezim, if you want a static IP set it up after the installation is over
16:30.17Strom_Mpublic service commission
16:30.21Qwellfigured
16:30.22CCFL_Man2coppice: true
16:30.25Qwellwhat do they do?
16:30.32tzafrir_homeDon't unnecessarily complicate the installation process...
16:30.35Strom_MQwell: equivalent of a PUC
16:30.37Qwelloh
16:30.43Qwelldo some states not have a PUC?
16:30.47Strom_Mit's just called a PSC in some states (new york, for example)
16:30.51Qwellweird
16:31.04Strom_Mbut the function is the same
16:31.17Qwelltake complaints, and never follow up on them? :)
16:31.34zimi can/will do it but why can't they ask "Would you like to automatical conf you network or would you like to set it up your self"
16:31.47CCFL_Man2Strom_M: so my pair can be connected to my dslam, but other companies have access to it?
16:31.54Strom_MCCFL_Man2: yes
16:32.09QwellI had to report some stuff to the PUC when I was going through the whole Verizon/Earthlink fiasco
16:32.27zim<PROTECTED>
16:32.34Strom_MQwell: the CPUC is actually unusually competent and subscriber-oriented
16:32.49QwellI never heard back from them ;/
16:32.54tzafrir_homezim, you don't have to configure everything at install time
16:33.11zimit just makes life easy
16:33.35tzafrir_homethere are a bunch of other questions they could have asked. Use a custom install
16:33.38Strom_MQwell: *shrug*
16:33.38zimyou dont have to build everything form source
16:33.47zimyou can if you want
16:33.58CCFL_Man2Strom_M: so my line's performance will be the same because it's using verizon's dslam, just going through a different pipe to a 3rd party adsl provider?
16:34.03Strom_MCCFL_Man2: yes
16:34.05Qwellthe nice thing about huntsville, actually, is I have a choice of cable providers here
16:34.06zimGNU is about choosing
16:34.22CCFL_Man2Strom_M: the question is, why am i still with verizon...
16:34.34QwellCCFL_Man2: most people aren't aware there are alternatives
16:34.38CCFL_Man2i hear speakeasy and dslxtreme kick ass
16:34.55QwellI think dslextreme is a southern CA thing
16:34.59QwellI could be wrong though
16:35.09codejunkyhi, I have the following line in my extensions.conf: exten => _0.,1,Set(CALLERID (num)=1234567). If I dial asterisk shows the following error: ERROR[7098]: pbx.c:1437 ast_func_write: Function CALLERID  not registered. Any ideas why it does not work?
16:35.21Strom_MI have eight static IPs, no port filtering, 6 megabit service...my ISP doesn't give a hoot what I do.
16:35.34tzafrir_homezim, in fact a server with address from DHCP is not uncommon
16:35.35CCFL_Man2Qwell: i was under the assumption that unless my rboc let a 3rd party dslam in to their CO, there were no alternatives
16:35.47CCFL_Man2ahh
16:35.57tzafrir_homeYou can reserve IP address if you want
16:35.58CCFL_Man2Strom_M: you use dslxtreme?
16:36.02tzafrir_homeeasier to manage
16:36.14Strom_MCCFL_Man2: dslextreme, yes
16:36.29CCFL_Man2Strom_M: kick ass, just how it should be
16:36.32Strom_Mbut I think they may be california-only
16:36.32zimok let me re ask the Q how many ubuntu servers have static IPs
16:36.42CCFL_Man2ahh, unfortunately
16:36.54Wonkacodejunky: remove the space between CALLERID and (
16:36.55zimanyway not that important
16:37.03Strom_Mbut there should be some equivalent in pennsylvania
16:37.13Wonkacodejunky: there is no function "CALLERID ", but only "CALLERID"
16:37.14CCFL_Man2Strom_M: epix, as far as i know
16:37.28*** part/#asterisk pkunkra (n=chris@cpe-74-73-28-89.nyc.res.rr.com)
16:37.30*** join/#asterisk pkunkra (n=chris@cpe-74-73-28-89.nyc.res.rr.com)
16:38.24QwellCCFL_Man2: dslr.com
16:38.26Qwellerm
16:38.33Qwelldslreports.com
16:39.58zimanyway I am about to install asterisk on my new box should i just use the default asterisk or add somthing to my source.list ???
16:40.25codejunkyWonka: Thanks. Now I am rid of the error. Is it possible that misdn does not support to set the caller id? On my mobile phone the number I set is not shown as the caller.
16:41.00Wonkacodejunky: maybe your telco doesn't allow you to set the callerid
16:41.07Wonkacodejunky: i don't know if misdn does
16:43.13codejunkyWonka: The problem is that we have 4 numbers (german isdn) and I want to have my telephone number shown when dialing out and not the main one.
16:43.24*** join/#asterisk apardo (n=apardo@bonemachine.ipv6.theprimusproject.com)
16:44.27*** join/#asterisk elixer (n=sbright@m645f36d0.tmodns.net)
16:44.46elixerQwell: you around?
16:45.02Qwellyep, saw the patch
16:45.07elixercool
16:45.14elixerhappy saturday, btw.
16:45.27elixermmmm
16:45.29elixerheh
16:45.45QwellI need to go to starbucks in the near future...
16:46.14elixeri'm on black-eye #2 for the day already
16:46.20elixerdon't have the shakes yet
16:46.21Qwelldouble add-shot frapacino...oh yeah...
16:46.51elixerfrapacino?  don't forget your skirt! ;-)
16:46.56Qwellyeah, yeah ;)
16:47.15QwellI don't see the point of hot coffee, really
16:48.10*** join/#asterisk wunderkin (n=wunderki@ip72-223-86-126.ph.ph.cox.net)
16:48.25heeliosQwell: s'better than cold coffee. <_<
16:49.50elixerwell what is the point of a frapacino?
16:50.34QwellI can drink it without burning my tongue :p
16:50.59elixerheh
16:51.02elixertouche
16:51.26Qwelland if anybody says they've never burned themselves with coffee - they are lying
16:51.33elixeroh i've done it
16:51.41elixerabout 3 weeks ago
16:51.47Qwellevery 3 weeks? :)
16:52.07elixerwent through the starbucks drive-thru, the girl handed me my large coffee and she hadn't properly secured the lid
16:52.14*** join/#asterisk CrazyTux[m] (n=CrazyTux@c-98-195-23-75.hsd1.tx.comcast.net)
16:52.20Qwellugh
16:52.29elixerthrow in some blinding pain and briefly blacking out
16:52.43elixerand now my passenger seat floor is a lovely shade of brown
16:52.44elixerheh
16:52.52Qwellsue!
16:52.57elixernah
16:53.01elixeri'm not one of those people
16:54.49_x86_anyone use Polycom IP301 phones?
16:55.19_x86_When I put people on speakerphone, I can hear them great, but they can't hear anything
16:55.55CCFL_Man2Qwell: i have one cable option and two adsl options, both adsl options suck
16:55.56_x86_is there a trick to turning on the mic? supposedly it supports speakerphone
16:57.48_x86_CCFL_Man2: ADSL always sucks... cable rocks
16:58.09_x86_SDSL is cool if you can find someone to give you some decent speeds on it :)
16:59.50_x86_elixer: give me your receipt and sell me your car at FMV... then watch how _I_ handle it ;)
17:00.02_x86_chaaaaaaa-ching!
17:00.08elixerheh
17:00.14elixer_x86_: god bless america
17:00.18_x86_*nod*
17:00.37_x86_home of the... eh... money grubbing bitches ;)
17:00.48elixeri'm with adam corolla.  each american gets one lawsuit per lifetime.
17:00.52elixerso make it good.
17:01.42_x86_i've never been in a court room except when i made the terrible mistake of marrying my wife, and they should have found me guilty of being a retard and thrown me in jail for it...
17:01.52_x86_;)~
17:01.53elixer_x86_: still married?
17:01.54elixerheh
17:01.58_x86_*nod*
17:02.12elixerwell.  no one is perfect.
17:02.16_x86_hah
17:02.29_x86_i'm a glutton for punishment i guess
17:02.45elixerso how do i return my local svn copy to match whats on the server?
17:02.49elixerjust remove and re-checkout?
17:02.53elixeror is that an easier way?
17:03.01elixers/that/there/
17:03.14_x86_svn update, or just svn up
17:03.18_x86_:)
17:03.33elixerbut that won't blow away all my local changes
17:03.38elixeri want to blow away my local changes
17:04.05_x86_it should blow away all local changes, it's an update from the server, not a merge
17:04.28_x86_alternatively, rm -rf working-copy-dir, svn co ;)
17:04.31elixerheh
17:04.31elixerright
17:04.35elixeri think i'll go that route
17:06.27elixerahhhh
17:06.27elixermuch better.
17:09.25yannj_frIs it possible to create area and define codec to use between them?
17:10.35*** join/#asterisk zeedo (n=zeedo@reboot-robot.net)
17:11.56_x86_yannj_fr: you can specify which codec(s) you want per individual peer
17:12.58yannj_fryes but I would like to define a group  and use differents codec for inside and outside the groups
17:13.07_x86_group of what?
17:13.25CCFL_Man2x86: you have cumcast?
17:13.26yannj_frof user
17:13.42yannj_frmy context : I have a central server for several agencies
17:13.56yannj_frI would like user in the same agency (LAN) use G711
17:14.08yannj_frand between agencies G729
17:14.40TedNJ38I need help please.  I am relatively new to Asterisk.  I have a network of phones all over the US which have been scattered among my relatives.  I want to purchase a landline in Peru so whenever any of us wants to talk to them (our releatives in Peru) we could dial 9 + phone number and get connected to them as a local call.  And whenever any of our relatives in Peru want to call one of us here, they dial a local number and they get promp
17:14.46_x86_CCFL_Man2: nope... Insight
17:14.58_x86_CCFL_Man2: i've heard nothing but bad about cumcast
17:15.03*** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog)
17:15.20_x86_yannj_fr: best to do that in the peer definitions
17:15.58_x86_TedNJ38: sounds easy enough... what hardware do you have?
17:16.18TedNJ38I have Sip Phones and I have a trixbox server.
17:16.29TedNJ38I just don't know what to look for if I want to purchase the landline in Peru.
17:16.32TedNJ38I don't know what they are called.
17:16.41yannj_fr_x86_ : but in peers definition you cannot divide in local or remote cases
17:16.57TedNJ38Should I google "Trixbox Landline Peru"?
17:17.08*** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla)
17:17.08*** mode/#asterisk [+o russellb] by ChanServ
17:18.53_x86_yannj_fr: why not? you don't know if the peers are local or remote?
17:19.02_x86_that doesn't make sense
17:19.18_x86_TedNJ38: you have to have hardware to connect to the phone line
17:19.25_x86_TedNJ38: you have a server in Peru?
17:19.34TedNJ38No, the server is here.
17:19.48_x86_ok, then you will not be able to get a phone line in Peru
17:19.54TedNJ38But I want to know if there is any company in Peru that would sell a local phone number that would be easy to link to a VOIP Server here.
17:20.00_x86_but, you can get a Peru phone number, transported over SIP
17:20.06_x86_sure
17:20.27_x86_http://www.voip-info.org/wiki/view/DID+Service+Providers
17:20.38TedNJ38Thanks
17:22.43_x86_to call Peru, you'll also need a provider that has a PSTN route there
17:22.52yannj_fr_x86_: but I can just force them to use specified codec, I cannot say : G711 to call peer2 and G729 to peer3, isnt it?
17:23.28_x86_yannj_fr: yes, you can
17:23.45yannj_frhow?
17:23.56yannj_fr'sorry, I feel stupid now'
17:24.10_x86_yannj_fr: [peer1] disallow=all allow=ulaw [peer2] disallow=all allow=g729
17:24.22yannj_frhum
17:24.32_x86_works with both sip.conf and iax.conf
17:25.15*** part/#asterisk TedNJ38 (n=HungLad@ool-4573adc7.dyn.optonline.net)
17:25.41yannj_fryeah, but I cant say for exemple : Peer1=> Peer2 : G711 Peer1=>Peer3 : G729, Peer3=>Peer2 : G729
17:25.46yannj_frfor exaple
17:26.01_x86_it works the same way
17:26.20*** join/#asterisk spark_elite (n=god@c-76-21-213-240.hsd1.dc.comcast.net)
17:26.39_x86_setup each server to only talk whatever codec you want with the peers it is connecting to
17:26.53_x86_you can even have different codecs for inbound or outbound calls
17:27.15yannj_frwell, this is the problem, I only have 1 server
17:27.34spark_elitehey, i've just setup asterisk and am working on outbound calls. I can hear my roommate perfectly on his cellphone but I'm choppy to him, we've tried a couple of codecs
17:27.36*** join/#asterisk Sweeper (i=sweeper@softcheese.net)
17:27.39Sweeperanyone want some t1 routers?
17:27.42SweeperI've got 3 spannet 2500, 4 serial cards (v.35, I think), and 3 t1 cards
17:30.45spark_eliteis there any debugging I can do to see why the outgoing data is choppy?
17:30.59MihiNomenEstso, my voip provider says that asterisk isn't clearing the lines fast enough.  is there some sort of timeout I can set?
17:35.08spark_elite_x86_, any idea on debugging outgoing choppyness?
17:35.49*** join/#asterisk obnauticus (n=obnautic@c-76-115-29-47.hsd1.wa.comcast.net)
17:39.49Strom_Mspark_elite: what kind of connection are you on?
17:40.10spark_elite8MB down and 1 MB up
17:40.19*** join/#asterisk kkn088 (n=kikoun@84.4.60.70)
17:40.22Strom_Mmegabytes?   or megabits?
17:40.27spark_elitebits
17:40.38Strom_Mok, if it's megabits, that's Mb, not MB
17:40.47spark_eliteright
17:41.04Strom_Mwhich ITSP are you using?
17:41.15spark_elitecall with us
17:41.25Strom_Mthat's the name?  "call with us"?
17:41.55spark_eliteyea - http://www.callwithus.com/
17:42.17*** join/#asterisk bintut (n=bintut@cm161.gamma176.maxonline.com.sg)
17:42.27Strom_Muh, okie dokie...
17:42.39Strom_Mwhere is their PoP in relation to you?
17:43.05spark_elitethey have 2 servers, east and west coast, we are in DC on the east coast
17:43.11NuggetHoP on PoP.
17:43.21Strom_Mright, but "east coast" is pretty vague.  Where /specifically/ is it
17:43.43spark_elitechecking...lansing michigan
17:43.53Strom_Myeah, that's not the coast :)
17:43.57Strom_Mhow many ms away is it?
17:44.21spark_elitewe are getting pings of 130ish
17:44.27Strom_M130?!
17:44.32Strom_Mthat's ridiculous
17:44.51Strom_MI can go from los angeles to new york in 80-90
17:45.16Strom_Mlet's try this...
17:45.42spark_elitei just pinged google and am getting 120 avg
17:46.03Strom_Mping the server I just sent you in PM and tell me what kind of latency you get
17:52.08*** join/#asterisk WilliamK (n=noc@static-71-170-144-3.dllstx.fios.verizon.net)
18:06.04*** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25)
18:06.22spark_eliteStrom_M, what was that service again, trying to find it on TEH google
18:06.52Strom_Mteliax.com
18:13.28*** join/#asterisk ManxPower (n=manxpowe@178.sub-70-216-22.myvzw.com)
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18:28.34jameswf-homegreetings
18:29.37Strom_Msgniteerg
18:30.52CCFL_Man2anyone here use an adit 600?
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18:57.42Sweeperwhat's the best way to unload some telco gear? not very popular on ebay, but I shrink from throwing them away, they still work and are useful :v
18:58.02heeliosSweeper: what kind of gear?
18:58.22Sweeper3 spannet 2500 t1/v.35 routers
18:58.41Sweeper4 t1 cards, and 4 v.35 cards
18:58.52Sweeperthey've got 10mb ethernet interfaces
18:59.36heeliosi see. nothing i have a use for. :s
18:59.37*** join/#asterisk CrazyTux[m] (n=CrazyTux@c-98-195-23-75.hsd1.tx.comcast.net)
18:59.41Sweeperand 2 4port serial matrices
18:59.49Strom_MSweeper: craigslist perhaps?
19:09.38SweeperStrom_M: eh, local CL is pretty much dead
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19:27.42WildPikachuhow can I escape - in AEL?
19:27.52WildPikachuMONITOR_FILENAME=${EXTEN:1}\-${TIMESTAMP};
19:27.55WildPikachui have that
19:30.54daakui want to setup a `pass-through` of sorts to call from a US phone -> SIP line with US number in India -> regular Indian phone line - can someone nudge me in the the right direction? i'm not even sure of what to google for
19:36.05codefreezeWildPikachu: Set(MONITOR_FILENAME=${EXTEN:1}-${TIMESTAMP});
19:36.13WildPikachufanks man
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19:38.13WildPikachuhrmmm, can I setup monitoring before sending a call to the queue to record it when a person picks it out of the queue?
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19:52.12ChunkzAre all features that are in * available in *NOW? *NOW just has a pretty gui interface, and a minimal linux installation?
19:53.05ChunkzI'm pretty much a complete newbie at this point :)
19:53.29DavieyChunkz: pretty much
19:53.39mvanbaak*NOW is a complete distro including asterisk-gui
19:53.54mvanbaakit's running plain asterisk, so everything is there
19:54.01Davieythe gui interface can be limiting, but for newbiew use; it'll get you started :)
19:55.45Chunkzawesome. as long as I can go "behind the scenes" to tinker a bit, I think it'll be perfect for me. Was deciding between that and a ubuntu install + *
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19:59.20DavieyChunkz: asterisk + asterisk-gui + ubuntu # = same end result
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20:04.33*** join/#asterisk Teln1100A (i=hello123@bas2-toronto12-1168023849.dsl.bell.ca)
20:09.23matt_hello, i'm using a linksys pap2 and i'm getting feedback and i sound quite to the remote person
20:09.41matt_the feedback isn't delayed so it seems to be an issue with this device
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20:38.47WildPikachushew ... renice -10 should put asterisk above most IO
20:40.24mvanbaakWildPikachu: yeah
20:40.32mvanbaakyou can also use the -p flag
20:40.35WildPikachumust just shove in my fxo's
20:40.38WildPikachuah
20:40.41*** join/#asterisk Koshatul (n=evangeli@ppp188-33.static.internode.on.net)
20:41.03WildPikachudarn, i'm using safe_asterisk
20:43.09mvanbaakwhy ?
20:43.20matt_i have a question
20:43.24matt_if i answer a line
20:43.35matt_and play a ring tone down it, does it cost the other person money ?
20:43.43mvanbaakyes
20:43.47matt_ok
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20:50.34Strom_Mmatt_: moral of the story is don't supervise until you have to
20:51.52matt_huh?
20:52.44Strom_Manswer supervision
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21:01.14linageehas anyone used the IAXy S101I? does it work well?
21:01.20linagee(IAX2 ATA)
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21:20.35heeliosanyone knows a good voip termination provider in europe?
21:20.43zapp-braniganhi , who can i select the codec to call when i call usind Dial(IAX2/user:password@ip/0${exten},120) the default password is ulaw
21:20.55mvanbaakheelios: 12connect, speakup
21:21.08ManxPowerzapp-branigan: You can't.
21:21.13ManxPowerDial by iax.conf entry
21:21.16heeliosthanks.
21:21.35ManxPowerwell, the codecs in [general] will be used if you dial by ip.host
21:21.43*** join/#asterisk xtr (i=94752345@216.19.191.111.novuscom.net)
21:21.47zapp-braniganno i male a dial in the dial plan
21:22.12zapp-braniganok
21:22.27zapp-branigangeneral del iax.cong is disable all
21:22.35zapp-braniganallow g729
21:22.43zapp-braniganbut not work
21:22.57ManxPowerdisallow=all and allow=g729  You would prolly need a G729 license for most uses of G729
21:22.59zapp-braniganin the extensions i must to write this too?
21:23.09zapp-branigani have licenses
21:23.55ManxPowerWell, what is the ERROR MESSAGE
21:23.57zapp-braniganbut in default codec in the source the codecs are ulaw alaw y linear
21:24.36zapp-braniganand use the defaults codecs
21:24.55ManxPowerthen you did not disallow and allow correctly.
21:25.06ManxPowerAgain, I ask.  What is the error message you are receiving?
21:25.14zapp-braniganonly in the iax.conf ?
21:25.22ManxPowerONLY IN iax.conf
21:25.45ManxPowerBut I cannot help you further until you tell us what the error message on the asterisk console is.
21:26.16*** join/#asterisk famicon (n=pastry@c51447ddc.cable.wanadoo.nl)
21:26.19zapp-braniganunable to create transtor path from unknown to ulaw
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21:26.45ManxPowerput a copy of your iax.conf on pastebin.ca
21:26.50ManxPower~pastebin
21:26.50jbotsomebody said pastebin was a place to paste your stuff without flooding the channel - try  http://pastebin.ca, or http://channels.debian.net/paste, or http://rafb.net/paste/, or http://pastebin.com is usually painfully too slow and unresponsive to use, use one of the other pastebin sites, or dpaste.com is a very nice pastebin as well
21:27.56zapp-branigani have the two lines
21:28.08zapp-branigandisallow=all
21:28.13zapp-braniganallow=g729
21:28.44ManxPowerEither do what I tell you or find someone else to help you with your problem.
21:28.47zapp-branigancodecpriority=host
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21:50.40Shadowfire_Hello people
21:51.51*** join/#asterisk l3jj (n=chatzill@dsl093-001-038.det1.dsl.speakeasy.net)
21:54.09KDanwhat people
21:54.12KDanthere are no people here
21:54.17KDanonly long-running shell scripts
21:55.01ManxPowerI'm a perl script, thankyouverymuch!
21:55.14KDanthat's what YOU think!
21:55.14*** join/#asterisk apardo (n=apardo@bonemachine.ipv6.theprimusproject.com)
21:55.33ManxPowerOMG!  I've been rewritten!
21:55.56KDanI spoke to your parents, and they said you were from a long line of tcsh scripts
21:56.40Nuggetheh
22:01.25wunderkinmy godaddy vps is fucked up for the 3rd time within a couple weeks, most of the time i can ssh in, 1st time i couldnt, cant ping localhost or get to the default gw, this time i can ssh, ftp, imap in though...
22:01.31*** join/#asterisk xtr (n=94752345@216.19.191.191.novuscom.net)
22:03.43mvanbaakthat's why I only run on my own stuff
22:04.28wunderkini wish that i could but i'm only using it for personal email and asterisk right now so even paying $30/mo for the vps is a lot
22:05.07KDan$30/mo??
22:05.13wunderkinit has been fine for like a year or so i think, at least several months
22:05.22KDanwtf... about $10/m for vps-link
22:05.25*** join/#asterisk johann8384_home (n=johann83@71-81-221-188.dhcp.stls.mo.charter.com)
22:05.42wunderkinprobably shitty too
22:06.02KDanfor sure, but at least it's dirt cheap
22:06.43KDanhttp://www.vpslink.com/vps-hosting/
22:08.10JTgodaddy being shit, how shocking ;)
22:08.20JTlinode.com
22:08.37JTalmost no-one has a bad word to say about their VPSes
22:11.10mvanbaakhttp://www.easyspeedy.com
22:11.38mvanbaakVPS == trouble
22:11.43mvanbaakthat's my experience
22:12.14wunderkinnice, they are resetting it.. nice fix
22:12.32JTa VPS is fine for a lot of things
22:12.49KDan"Congratulations! We have reset your VPS. You now have a brand spanking new Debian install, fresh and clean."
22:13.05wunderkinno, just a 'shutdown -r now' not a reprovision...
22:13.44wunderkini wonder why i would be the only person there having that problem
22:14.43asdxwhats better than a vps
22:14.49KDana dedicated server?
22:14.50JTi co-lo server
22:14.55KDannah, colo sucks
22:14.56JTa
22:15.02JTKDan: ?
22:15.13KDanhave to actually physically deal with the server
22:15.18JToh gno!
22:15.23KDanbetter to get a dedicated server on a monthly tariff
22:15.27heeliosKDan: colo is the most flexible option you have.
22:15.31KDancheaper in the slong run
22:15.31JTlike if you were running a real business or something!
22:15.33KDan*long
22:15.35heeliosapart from building your own datacenter. <_<
22:15.41JThaha if you think so
22:15.42JTheelios: ?
22:15.43KDanJT: I am running a real business
22:15.57KDanand the last thing I want to do is have to leg it to the data centre to change a hard drive
22:16.06heeliosAND it gets cheaper in the long run.
22:16.23JTnot a big problem if the datacentre is in the same city
22:16.31wunderkinheh
22:16.38KDaneven in the same city
22:16.44KDanon top of it, colo is way more expensive
22:16.50wunderkinwell linode.com is in he.net fremont so thats cool, wonder how i know which one i will be setup on though
22:16.50JTnot really
22:16.56KDanin london it is
22:17.07KDanunless you have lots of servers to begin with
22:17.12JTwunderkin: you can lodge a support ticket to have it moved to fremont
22:17.27JTmaybe you're just getting the wrong prices
22:17.53JTbut yeah, i don't want to use the crappy dedicated servers that hosts get for their customers if i can avoid it
22:18.04KDannah. 1U was a ripoff (price of a cheap dedicated server basically) and quarter-racks were very expensive
22:18.15wunderkinJT, yeah thanks, i will take a look at them, i just saw their irc support on their webpage now..
22:18.41JTwunderkin: everything provisiong automatically/via their webpage, it's pretty neat
22:18.44KDanJT: serverbeach seem to work fine for us. cheap, good, etc.
22:18.45JTKDan: how much?
22:18.59JTfor 1RU
22:19.00KDancan't remember now, but in the region of £40 iirc
22:19.07KDan/month
22:20.17JTfor how much data
22:21.54KDandata was not even included :-~
22:21.57KDan:-|
22:22.17JTit could still be economical depending on the price of data
22:22.26JTobviously rack space gets a lot cheaper the more you get
22:22.43KDanwell, on ServerBeach we got a dedicated server with 2000 GB of bandwidth for $120/month
22:22.57*** join/#asterisk justdave (n=dave@unaffiliated/justdave)
22:22.58JTyeah oversold to all hell
22:23.27heeliosi have to say that sounds shady. <_<
22:24.16JTi don't think i'll take hosting advice from KDan, with recommendations like that ;)
22:24.21KDanhave not had any bandwidth issues on SB
22:24.33JTnot yet
22:24.38JTbut it's a possibility, for sure
22:24.42KDanbeen with them for 3 months
22:24.47KDanso far so good
22:25.38JTyou've got to realise that 2TB is bs
22:25.56JTif everyone downloads/uploads 2TB it will probably catch fire
22:25.57KDanof course i do. All I care about is "it won't run out and it's fast"
22:26.01*** join/#asterisk fakhir (n=Fakhir@unaffiliated/fakhir)
22:26.12KDanoverselling is part of business
22:26.24JTsome business.
22:26.25KDani have no issues with it so long as the quality remains good
22:26.40JTand you don't know how crappy the machine is
22:26.46JTif it has any redundancy at all
22:26.47KDani'd say it's bad business not to sell the same thing twice if you can!
22:27.06JTdepends if you're after quality or a cheap buck
22:27.10KDanit has more redundancy than it would have if i had had to scrape it together myself!
22:27.45KDanconsidering they also host youtube and are owned by Peer1 (fairly large peering network), I think they offer both quality and good prices
22:27.55KDananyway, enough with this discussion
22:27.56KDangot work to do
22:28.00JTenough?
22:28.07*** join/#asterisk d-tech (n=d-dtech@72.245.233.107)
22:28.11JTi don't think one single provider hosts youtube
22:28.26KDanthese guys sure claim to do so
22:28.39JTand i will stick with proper co-location for important stuff
22:28.48JTbeing able to access my servers is essential too
22:28.50KDanoutsourcing is the future
22:28.55KDanget on with it -- or die
22:28.56JTas well as having full control for my network
22:29.01*** join/#asterisk bminish (n=bminish@brenbox.westnet.ie)
22:29.02JTi don't just do web pages
22:29.09KDanme neither
22:29.15JTstop talking like a retard with brainwash braindamage
22:29.25JToutsource, ONLY if it makes sense
22:29.26WilliamK11 bgp peers for youtube.com
22:31.21JTdedicated servers make no sense for anything more advanced than a couple of web pages
22:31.24d-techI'm feelin pretty stupid here ... I'm attekpting to add a channel driver to asterisk ... the driver install package says it can't locate the 'kernel source directory' and is asking ME for it ... can anyone give me a clue? I'll pass it on to the stupid install script <vbg>
22:31.46heeliosd-tech: what distro?
22:31.56JTd-tech: make sure the kernel sources and header files are installed
22:32.01zapp-braniganhi
22:32.10d-techCentOS ... if i'm reading your question right?
22:32.36*** join/#asterisk CrazyTux[m] (n=CrazyTux@c-98-195-23-75.hsd1.tx.comcast.net)
22:33.01heeliosd-tech: you are.
22:33.17heeliosd-tech: im not familiar with it, but try # up2date --get-source kernel
22:33.33WilliamKheelios, it's the same as Redhat Enterprise 4
22:33.44d-techthanks JT ... your probably 100% correct ... but you might as well be speaking chinese
22:34.19*** join/#asterisk CrazyTux[m] (n=CrazyTux@c-98-195-23-75.hsd1.tx.comcast.net)
22:34.33d-techYes, I believe WilliamK is correct
22:34.53WilliamKd-tech: type this....  rpm -qa | grep kernel
22:35.00JTit's based off RHEL, and is not supported by redhat
22:35.03WilliamKmake sure your kernel sources are installed
22:35.24harryranybody tried Skype business edition?
22:36.11*** part/#asterisk bminish (n=bminish@brenbox.westnet.ie)
22:36.21JTwrong channel for skype ;)
22:36.37heeliosharryr: i dont think i ever will. <_<
22:36.38WilliamKd-tech, you should have 5 lines that get printed to your screen
22:36.59harryrheelios: I mean the concept
22:37.14d-tech<WilliamK> make sure your kernel sources are installed ---- I would LOVE TO ... but I haven't a clue WHAT that means, much less, how to??
22:37.23harryrI was going to knockup something very similar - absolute minimal business orientated VoIp
22:37.26WilliamKd-tech, can't copy and paste?
22:37.28harryrbut based on asterisk or yate
22:37.34WilliamKthat's the exact command
22:37.44JTharryr: you'd probably want a sip proxy too
22:38.04harryrJT: for chat?
22:38.16JTto load balance
22:38.31harryrJT: we're talking 2-50 users on a LAN
22:38.32d-techoh ... missed the command ...lemme try ..brb
22:38.40JToh ok
22:38.43harryrJT: with chat, local calling and outbound calling
22:38.54harryrand an uber simple control panel
22:38.58*** join/#asterisk zotz (n=zotz@24.244.163.157)
22:40.59d-techtwo items kernel-xxx & kernel-utils-xxx
22:41.13JTit says xxx?
22:41.26d-techno ... a long version number
22:41.28harryrhardcore kernel?
22:41.31harryroh
22:41.34wunderkinROFL, look at this response about my godaddy vps problem: Thank you for contacting hosting support. We had to reboot the server in order to restore external internet access. It is likely the issue you are having is a direct result of the Asterisk software on your server. Typically, our Virtual Dedicated servers will not reliably support VoIP. Dedicated servers can be configured to support VoIP, but we are unable to assist with any trou
22:42.03JTwunderkin: most VPSes are useless for VoIP especially asterisk
22:42.08famiconwunderkin
22:42.10harryrwunderkin: aha, ok that's godaddy blacklisted
22:42.13famiconwww.easyvpshost.com
22:42.15famiconuse that
22:42.20JTUML especially
22:42.22JTugh
22:42.23harryrJT: we've got something like 500+ VPS instances running Asterisk
22:42.27famiconyeah
22:42.37harryrJT: and no problems at all
22:42.43JTif you wanst to do asterisk, use real hardware or Xen or KVM
22:42.43famiconwunderkin if you sign up there ill give you 2 months off
22:42.45JTnot UML
22:42.53wunderkini understand that part, but it has been working fine for almost a year, asterisk is not causing the problem, i understand how a vps works and voip, i got that part, but asterisk does not cause the fucking vps to not even be able to ping localhost
22:42.53harryrJT: we're using Linux VServer
22:42.56famicongodaddy is bullshit anyways
22:43.18wunderkinand no im not using a firewall
22:43.23JTharryr: whatever that is :)
22:43.25d-techI'm guessing the first one is the one your looking for --- kernel-2.6.9-34.0.2.EL
22:43.33famiconyeah the moment you cant ping localhost
22:43.37famiconis when you got a problem
22:43.38famiconwunderkin
22:43.41harryrJT: it's like KVM, but more mature
22:43.46famiconcheck if you can upgrade the kernel
22:43.47famiconif not
22:43.50famiconask for a refund
22:43.52JTmore proprietary?
22:43.56*** join/#asterisk kiscokid (n=Ron_Laut@adsl-66-122-34-11.dsl.sntc01.pacbell.net)
22:43.56famiconand tell then that you wanted a VPS
22:43.58famiconnot a chroot
22:44.00harryrJT: no, completely opensource
22:44.06JTah ok
22:44.23harryrit's not in the kernel trunk though, but is well maintained and up to date
22:44.43wunderkinfamicon, yeah.. no kidding.. it is on FC4.. but has been fine for a long time, i'm trying to decide if i want to ditch vps altogether and just use a WRT with asterisk at home, shrug
22:44.44DavieyAnybody here using ztdummy in a VPS?
22:44.55famiconwunderkin eh
22:45.01harryrDaviey: not.. in a VPS, but yeah
22:45.06famiconjust go for my services ;)
22:45.10famiconi got fc 6
22:45.16famiconand centos 5
22:45.17wunderkini would have to figure out something else for my mail and stuff though :/
22:45.26JTdebian pls kthx
22:45.32harryrgentoo kthx
22:45.33Davieyfamicon: eh?
22:45.33wunderkinkplzthx!
22:45.39famiconJT i got etch
22:45.41famiconand sarge
22:45.42Davieyfamicon: are you using ztdummy in a vps?
22:45.45famicongentoo 2006 too
22:46.04JTfamicon: what virtualisation scheme?
22:46.06famiconDaviey havent tried
22:46.09famiconjt Xen
22:46.15Davieyfamicon: try.. it's fun
22:46.23famiconDaviey whats the problem?
22:46.34wunderkinfamicon, you aren't even in the us :D
22:46.39JTfamicon: is it that hard to put a : after peoples' names? ;)
22:46.42famiconyeah I'm in amserdam
22:46.43Davieyfamicon: it's a PITA to compile, then won't modprobe
22:46.59famiconDaviey never had that problem
22:47.08Davieyfamicon: but you've never tried ffs
22:47.09famiconyou can modprobe with xen you know
22:47.14harryrDaviey: in zen/vmware etc.?
22:47.22Davieyfamicon: I know..
22:47.23famiconDaviey I have had problems with that before
22:47.28harryror in something like vserver, openvz etc.
22:47.30Davieyfamicon: but you haven't tried!
22:47.45Daviey22:46 < famicon> Daviey havent tried
22:47.56famiconDaviey yeah buti know you can load kernel modules
22:47.57JTi hope your version string is fake, famicon
22:48.05Davieyfamicon: try it.. trust me
22:48.17famiconDaviey i bet it works
22:48.22Davieyfamicon: go on
22:48.31JTwhat a stupid bet to make when you haven't even tried
22:48.33famicon2 bucks it does
22:48.37Davieyfamicon: deal
22:48.42JTwhat a professional host...
22:49.07Davieyfamicon: i'm talking about, "Just Works"TM -- with no access to Dom0
22:49.17famiconDaviey oh.....
22:49.26Davieybut heck.. try it anyway
22:49.37Daviey$2 is worth the giggle
22:49.40famiconright now im kinda busy rewriting an xml parser
22:49.58JTbut never too busy to spam irc
22:50.00DavieySurely it's an easy two buck <grin>
22:50.15famiconjt yeah, cause im waiting for something to be done compilin
22:50.25JTJT:
22:50.35JTis it really that hard to attention remarks properly??
22:50.47DavieyJT clearly ;)
22:50.48famiconis it that hard not to be a complete douche?
22:50.58JTfamicon: fuck off
22:51.05JTyou're the one spamming your crap here
22:51.07Corydon76-homeSettle down
22:51.10JTmaking stupid untested claims
22:51.17famicondude, it was toungue in cheeck
22:51.19JTand not understanding how to use irc correctly
22:51.24JTsure it was
22:51.42famicondude i get like 0 sales off irc
22:51.49JTi can see why
22:51.52Davieyso why bother spamming?
22:52.14Corydon76-homeDaviey: It's not spamming if he's a regular here and isn't doing it repeatedly
22:52.31DavieyCorydon76-home: well miss-selling is spam IMO
22:52.37famicondude, i said, i have hosting that does support it, here's extra discount
22:52.44JTi don't think he's much of a regular
22:52.49famiconi wasnt going GET CH34P V14GR4
22:52.50Corydon76-homefamicon: let it go, please
22:53.03JTyou hadn't tested if compiling ztdummy even worked
22:53.07famiconCorydon76-home  yeah, good call
22:53.13famiconthis isnt worth arguing
22:53.18Davieybut happy to sell your services saying it does
22:53.52famiconi dont see why it wouldnt compile
22:54.04Corydon76-homefamicon: because it's a kernel module
22:54.18Daviey*cough* http://bugs.digium.com/view.php?id=9592
22:54.20famiconyeah, but i have compiled kernel modules before so i dont see why this one wouldnt work
22:54.22Corydon76-homeand you can't load kernel modules on virtual machines
22:54.33JTCorydon76-home: you can on some
22:54.39Davieyfamicon: Does xen support hardware kernel modules? :D
22:54.39famiconah
22:54.54famiconDaviey if theyre enabled on the dom0 yes
22:54.58Corydon76-homeJT: only for very specific purposes
22:55.08DavieyCorydon76-home: kernel modules that don't need hw access can load
22:55.25Corydon76-homeDaviey: right, well, ztdummy does access hardware...
22:55.27famiconand if it wouldnt work i would just make sure it gets installed on one dom0
22:55.37DavieyCorydon76-home: hence why i raised it
22:57.42wunderkini wonder if my vps problem could be from some kind of asterisk exploit, i'm running on old 1.2... this only started a month or so ago, i'm running asterisk as root (yeah, i know), maybe it is just causing different problems from the exploit because it is running in the vps..?
22:58.03famiconwunderkin aye
22:58.06famiconthats not too smart
22:58.12famiconwhat distro are you using
22:58.22Corydon76-homewunderkin: well, we have had a few vulnerabilities...
22:58.23harryrit's sort of "im driving my car drunk with a blindfold on.. WTF IS WRONG!"
22:58.30harryr;)
22:58.41wunderkinpeekeyboo
22:58.42kiscokidwhat is vps?
22:58.51harryrkiscokid: virtual private server
22:59.01WilliamKharryr, better hope the road is straight and your steering wheel doesn't move :)
22:59.07harryrkiscokid: a virtualized environment to allow multiple instances of linux to run on the same server
22:59.17wunderkinthe vps host (?) is fc4, that should be upgraded too... they only offer up to fc6
22:59.17WilliamKoh and your tires are aligned properly
22:59.18WilliamK:)
22:59.28harryrWilliamK: or you're sationary and everything else is moving around you
22:59.32harryr:)
22:59.42kiscokidharryr:  thanks.  where do you get it?
23:00.06WilliamKassuming you want to live, sure!
23:00.07WilliamK:)
23:00.15harryrkiscokid: vmware, linux vserver, openvz, xen.. there are lots of solutions out there which differ slightly which all provide the same sort of thing
23:00.24harryrWilliamK: or you're just hallucinating it all
23:00.24wunderkini'm not too keen on updating the os since they will nuke all of my stuff
23:00.30kiscokidharryr: thanks
23:00.47*** join/#asterisk yannj_fr (n=yannj_fr@vpn.intelunix.fr)
23:00.59WilliamKd-tech, if you're still there.. you're missing some files... what ver of CentOs are you running?
23:01.02harryrWilliamK: I once went for a drive in a broken car that was sitting in a barn
23:01.18WilliamKharryr, sounds very interesting
23:01.30WilliamKbut not something I have done nor plan to do :)
23:01.33harryrWilliamK: 6 people in the car beatboxing the jungle brothers, all extremely high
23:01.44harryryou probably shouldn't plan to either :)
23:02.24VJFROMGThello all, I am looking for someone to write a little script for my cdr to log IP from the call data (I need to know where every extension is calling from)
23:02.29VJFROMGTwilling to pay
23:03.04yannj_frVJFROMGT : you me caller ip?
23:03.07yannj_frmeant*
23:03.16VJFROMGTyes
23:03.28VJFROMGTcallers (extensions) call out via my asterisk
23:03.39yannj_frcould it bea solution to record it as account code?
23:04.02VJFROMGTyannj,, i am not clear
23:04.40harryrVJFROMGT: what do you have in-place so far?
23:04.42VJFROMGTjst one thing,, extension do not have username or passw . so they athenticate solely by ip
23:04.51yannj_frI meant if the ip is recorded in account code is it ok for you?
23:04.58VJFROMGTharry, the Areski CDR is in place
23:05.03harryrVJFROMGT: you can filter ${SIP_HEADER(FROM)} and get the IP from there
23:05.28VJFROMGTharry,, i know that can be done, i have zero programming skills, i want to pay someone to do it
23:05.37harryrah ok
23:05.46harryris it ok if the programmer is drunk and you have paypal?
23:05.46VJFROMGTup for the chanlenge?
23:05.49harryrif so I'm your man
23:05.56harryr:)
23:06.04VJFROMGTas long as he can get the job done, pm me
23:06.05*** join/#asterisk newsmafia (n=newsmafi@wsip-70-167-122-167.sd.sd.cox.net)
23:06.09harryrah
23:07.07yannj_frexten => s,1,Set(CDR(accountcode)=8675309)
23:07.43yannj_frVJFROMGT, the thing you need is to add a field with caller ip address, am I wrong?
23:08.25*** join/#asterisk Igbothom_III (n=Hilton@office.quarkgroup.com.au)
23:12.52mvanbaakI'm off
23:12.55mvanbaaklatero all
23:13.44*** part/#asterisk kiscokid (n=Ron_Laut@adsl-66-122-34-11.dsl.sntc01.pacbell.net)
23:14.47yannj_frbyebye
23:21.21harryrI'm suprised Asterisk doesn't log the from ip address to begin with
23:21.37*** join/#asterisk rlama (n=rlama@193.252.229.22)
23:21.40harryractually, mind you we don't do that either :\
23:23.46*** join/#asterisk riddlebox (n=james@ppp-70-225-67-135.dsl.covlil.ameritech.net)
23:27.30ManxPowerharryr: because many calls asterisk handles are not IP calls.
23:27.41ManxPowerstop being voip-centric.  Asterisk is a multi protocol PBX
23:27.54*** part/#asterisk rlama (n=rlama@193.252.229.22)
23:29.04harryrManxPower: ah fair enough
23:29.21*** join/#asterisk rlama (n=rlama@193.252.229.22)
23:29.54ManxPowerexten => s is never matched for a voip call, btw.
23:30.06ManxPowerexcept in very specific and unusual situations
23:30.14harryrit's not..!
23:30.32harryrare you sure? because I've used it many times with VoIP calls
23:30.45ManxPower"s" is matched when Asterisk does not know the dialed number.
23:31.04harryrI thought s was matched whenever the context was entered, regardless of the dialed number
23:31.08ManxPowerIf in VoIP calls, asterisk normally knows the dialed number.
23:31.13ManxPowerharryr: NO THAT IS NOT TRUE!!!!!!
23:31.18ManxPowerI wish people would stop thinking that.
23:31.30harryroh right, I see now
23:31.36ManxPowerwhere did you get that horrible mis information?
23:31.53harryruh, I inherited it from the company I work at
23:32.08harryryeah I see now reading he docs
23:32.13ManxPowerGenerally Asterisk does not receive the dialed number on analog FXO ports and T-1 channels that are FXO signalled.
23:32.19*** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com)
23:32.26ManxPowerand ONLY in those situations.
23:32.32ManxPowerOh, and macros
23:33.34harryrwell that's yet another asterisk myth crushed
23:33.54WildPikachui thought fxo's supported caller id?
23:34.00harryrwhy isn't it called "u" (unknown) instead of "s" (which I presumed, "start")
23:34.00ManxPowerharryr: no, it comes back like bad one-night stand.
23:34.13ManxPowerWildPikachu: callerid IS NOT the dialed number.
23:34.17ManxPowerit is the CALLING number.
23:34.19harryrI've always been too drunk to remember :\
23:34.22WildPikachuargh, my bad
23:34.30ManxPowerharryr: I think it should be named "none"
23:34.36harryrah
23:34.40WildPikachusorry, i must go to bed now  ... heh, can't even keep awake, now i'm making silly mistakes
23:36.01*** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga)
23:40.14*** join/#asterisk _santiago_ (n=santiago@debian/developer/santiago)
23:40.38Davieyfamicon: When you need my paypal details.. just ask ;)
23:44.59*** join/#asterisk elixer (n=seanbrig@c-68-55-114-113.hsd1.md.comcast.net)
23:56.12*** join/#asterisk gankhuu (n=luken@ns2.digis.net)

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