IRC log for #asterisk on 20120114

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00:28.44Iskorptix_*problem with flower fixed by resetting config through the phone*
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00:38.58ChannelZsounds more like a weed
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01:50.04*** join/#asterisk cstk421 (~cstk424@99-20-229-143.lightspeed.brhmmi.sbcglobal.net)
01:50.11cstk421hello all
01:50.25cstk421anyone decent with Hylafax ?
01:50.33cstk421basic modem setup
01:57.47phixhey cstk421
01:57.52cstk421sup
01:58.02phixI have setup Hylafax, with aix-modem
01:58.33cstk421ok i have 2 types of modems conexant and trendnets  both internal pci
01:58.42cstk421cant get them to work for anything
01:59.28lanningif they are "winmodems" good luck...
01:59.56phixhmmm what chip is on it?
01:59.59phixlspci to find out
02:00.29cstk421k
02:00.36phixcstk421: Most manufactorers of "winmodems" do not create drives for operating system other than windows
02:00.45phixdrivers* even
02:01.04cstk421<PROTECTED>
02:01.04cstk421<PROTECTED>
02:01.05cstk421<PROTECTED>
02:01.05cstk421<PROTECTED>
02:01.05cstk421<PROTECTED>
02:01.11cstk421thats the trendnet
02:02.25phixok apparatnly there is support for the Agere systems lucent v.92, however only 32bit support, not 64 from what I can tell
02:03.24phixThere are some deb packages for it too
02:03.27phixwhat distro you using again?
02:03.35cstk421centos 5
02:03.45phixok there are RPMs for it too
02:03.53cstk4215.7 to be specific
02:03.55cstk421k
02:04.13phixgoogle it, first result is from a mailing list with links to the drivers
02:04.22phixyou can also try the manufactorers website
02:06.03cstk421manufacturer has only windows drivers as far as i see
02:06.31cstk421although its 251mb
02:06.46cstk421would that have linux drivers if the manufacturers site sais its supported you think ?
02:07.23phixnot sure
02:07.31phixis it a 251mb exe or zip file?
02:07.47cstk421zip
02:07.59cstk421pullling it anyway to see
02:07.59phixthen maybe
02:14.07cstk421nope
02:14.10cstk421all windows drivers
02:14.16cstk421where did you find the other one ?
02:21.54phixfrom google search
02:21.55phixhttp://linmodems.technion.ac.il/packages/ltmodem/11c11040/
02:22.05phixgoogle ltmodem linux
02:22.09phixand you should get the official site
02:24.10cstk421and what defines which package is needed ?
02:26.42phixwell you would want the rpm, probably the src so you can build it yourself for your system
02:30.35cstk421seems more complicated then i expected.  so based on what i am seeing there is no driver to just install and get this up and running /
02:30.36cstk421?
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02:48.01itbrokeack
02:48.26itbrokecan't load dahdi or sip after recompiling and installing asterisk... complained about not being able to find modules.conf
03:04.55phixcstk421: of course it is :) blame the modem manufactors for not creating a GUI for linux to install it like they did for windows
03:05.23phixitbroke: nice, does the file exist?
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04:10.27phixhey I want a 4 port TDM card, should I get AEX431 (PCI-E) or TDM431 (PCI)?
04:12.35ectospasmphix: they're both functionally equivalent
04:12.47ectospasmphix: does your target system have PCI or PCIe slots?
04:14.38phixyes both
04:14.52[TK]D-FenderWhat's your future system likely to have?
04:15.12phixboth
04:15.13[TK]D-FenderMy money's on PCIe
04:15.20phixok
04:15.24[TK]D-Fenderfor if you'll only have one
04:15.54phixmightput another card in the future, but atm 1 is all I need
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04:46.25bluregardgood evening all
05:02.25p3nguinwants to build a fire.
05:02.38p3nguin-12 degrees tonight
05:04.51ectospasmeventually PCI will be very hard (and very expensive) to find.
05:10.34[TK]D-FenderAnd servers are very hard to find WITH PCI these days.  Prepare your cards for the future
05:10.47[TK]D-FenderWhich is actually.. the NOW when you think about it
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06:14.24*** join/#asterisk Chorca (~chris@230.88.204.68.cfl.res.rr.com)
06:16.00Chorcaheyhey, having an odd issue.. wonder if anyone here's seen it before.. My asterisk default vm prompts play back awesome to internal phones, yet lock up randomly while playing back over an IAX channel. Audio is great all day long over the channel, but playing back the prompts will just stop randomly, no errors, just dead air.
06:18.06ectospasmChorca: what codec(s)?
06:18.26Chorcagsm and ulaw
06:18.58Chorcatried forcing the iax to both, i'm able to get clear, no-jitter audio on both of them just fine over the trunk for minutes at a time, but the prompts stall about 5 seconds in.
06:19.44Chorcathis is Asterisk 1.6.2.5-0ubuntu1.4
06:20.15Chorcanot sure how to get more debugging than it showing it playing the prompts, other than packet-level captures
06:23.17Chorcahttp://lists.digium.com/pipermail/asterisk-users/2010-September/254319.html got posted awhile ago, but no resolution :/
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06:26.42ectospasmChorca: no telling.  Asterisk 1.6.2.5 is WAAAAAAY out of date.
06:27.20ectospasmonly distro I know of that keeps things truly up to date is AsteriskNOW.  PBX in a Flash does OK, too
06:27.24ectospasmbut then again, I'm biased
06:27.57_omerHello, I need to install a PBX for my office for internal calls, conference, calls transfers, etc .... so which PBX is good ... (I have never used any asterisk based PBX but worked on Asterisk for couple of years) ...
06:32.33ChorcaHmm, alright.
06:32.38ChorcaI can bump it up from source, no biggie
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06:47.24_omertrixbox or freepbx?
06:47.52Chorcathey're both pretty good
06:48.08Chorcai haven't used either for awhile, but they are both great frontends
06:49.08_omerand they both have same features?
06:49.21_omerI think I should go for trixbox ....
06:49.28_omeror may be freepbx :s
06:50.34Chorcaheh, just install one and see how you like it, they're free, no reason not to try both
06:51.11Chorcai think trixbox has traditionally had a bit more development because it was a company selling hardware, but like I say, I haven't touched either for awhile, so FreePBX may have caught up
06:52.07min3rPBX In a Flash is excellent as welll
06:52.27min3rGV for outbound right out of the box
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07:24.51bluregarddoes asterisk have issues with sip softphones on the same machine asterisk is running on?
07:25.15_omerfreepbx or trixbox ?? which one is recommended ?
07:26.26bluregardnever used freepbx, but I really hate trixbox.
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07:26.32kaldemar_omer: you're on a channel were people would like to answer "none", but choose silence instead. ask in #freepbx.
07:26.48_omerok
07:26.57bluregard_omer: don't cheat, learn asterisk.
07:27.25_omerI cant develop a pbx in asterisk manually :s
07:27.40bluregardwhy not?
07:28.19_omerthats gonna take month or months and then QA also needed
07:28.42_omerbut I need a pbx for my client in a week
07:29.19bluregardhow big of a phone system is this going to be?
07:29.29_omer30 - 40
07:29.46_omerit will be a replace of CISCO CALL MANAGER
07:30.39bluregardgood deal
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07:31.29bluregardfor what it's worth, Asterisk really isn't as hard as it seems.  The first time I started playing with it I had a working system up in about 2 hours.
07:32.14_omerbut you know client also needs an interface to create extensions, IVR, recording etc .... I cant develop interface that quickly ....
07:32.23_omerif I do, then it's gonna cost him alot
07:33.01bluregardI understand.
07:33.46_omerso freepbx includes all the features ? like call conference, call transfer, recording , etc etc
07:34.33bluregardno idea.  never used that one.  I was strong-armed into using trixbox once and I'll never do that again.
07:35.21_omerok :)
07:35.32_omeranyhow, thanks alot
07:35.40bluregardsure thing
07:35.43bluregardgood luck
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07:55.14Shazzamy~book
07:55.14infobotAsterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook
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08:33.37ChannelZ*asterisk* contains these features.  FreePBX and the like are just obfuscation GUIs that sit on top
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09:59.30KNERDI  kicked the power off on my phone and now it cannot register again
09:59.52KNERDI keep seeing a SIP/2.0 401 Unauthorized message
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13:22.43CadeyHi guys, anyone in here got a couple of mins to explain how we can setup a customer with an analog fax machine up on asterisk with a FXO media gateway
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13:23.38Cadeywe are looking for more of a "fax pass though" than asteisk handeling the faxes I guess
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13:48.37ketashmm
13:48.59ketashow about doing dial-up over voip
13:49.26Cadeywell I guess im trying to find the simplest way tbh
13:49.54Cadeywe are not talking VOIP at the moment because we have PRI's on site not a SIP trunk
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14:25.12twanny796when I am in cli, and I do a typo I cannot backspace and the cli freezes??
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15:27.05EmleyMoorAny known problems with thi native SIP client on Galaxy Nexus and Asterisk?
15:27.30EmleyMoorthe, even...
15:27.55EmleyMoorHave been trying out all the VoIP clients I could find for Android...
15:30.04fireman_biffEmleyMoor: I've used CSipSimple successfully on a tablet
15:30.09fireman_biffnote very much, but its worked
15:30.13fireman_biff*not
15:30.57fireman_biffit integrates with your contacts and pops up a picture when you get a call, which is nice
15:34.58EmleyMoorfireman_biff: CSipSimple is the most promising of the ones from Android Market that I've tried - though I did have it stick ringing once, and it won't dial with Bluetooth on
15:36.40fireman_biffEmleyMoor: hmm... I never tried it with the bluetooth on
15:37.04EmleyMoorAcknowledged bug - they're trying to fix it
15:37.45EmleyMoorIt seems that even the native client is working over 3G - the manual implies it's WiFi only
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16:15.18mateuI'm using postfix with asterisk (on same machine) and I'd like to have voicemail emails be sent out on port 587.  I have postfix listening on port 587 (via submission service).
16:16.00mateuHowever, I haven't figured out how to get the voicemail emails to use port 587
16:16.38Guggesetup a smarthost in postfix, using whatever smtp server you want to sent to on port 387
16:16.40Gugge587
16:16.45Guggeits a postfix thing
16:16.50Guggehas nothing to do with asterisk
16:17.33mateuI want the localhost to be the sender, but yeah seems like it's postfix specific question
16:18.11mateuGugge: thanks.  I'll dig deeper (every example I've come across doesn't force sending on 587)
16:18.45Guggenormally it sends from a random port, to port 25 on whatever mailserver the receiver uses
16:18.58Guggeunless you specify a smarthost, then it sends everything to the smarthost
16:19.21mateuis trying to deal with port 25 being blocked outbound for verizon fios
16:20.04Guggeand they dont provide a smarthost on port 25 for the customers?
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16:53.53laurishi, is asterisk 1.6.2.x configuration/dialplan compatible with 10.0.0 ?
16:56.12[TK]D-Fenderlauris: Pretty much
17:05.54laurisis it possible to disable Network configuration menu in Cisco/Linksys SPA phones?
17:06.26laurisor to limit access to it, setting admin password doesn't help
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18:06.23p3nguinlauris: You should only be able to see the config, not change it, if you don't supply the password.
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18:07.48laurisi was able to change web server setting without providing a password
18:07.52lauristhrough the menu
18:10.33DennisGhi everyone
18:10.51DennisGare here people with a large CDR set?
18:11.04DennisGin file or SQL format
18:11.37ChannelZI have my csv log going back to 2007 when I brought the system up for the first time
18:12.08laurisDennisG, define large
18:12.11DennisGoke nice ChannelZ, do you have also a backup of that csv log?
18:12.33DennisGlauris, like 1 mil+ records
18:12.49ChannelZyes
18:12.49laurisok, then I have one
18:12.49lauris|  2704589 |
18:13.06lauristhis is a record count of cdr table
18:13.07DennisGwhat's the best way to create proper backups of the CDR data?
18:13.16DennisGhaha nice lauris, good job
18:13.39ChannelZwell it depends, where is yours?  A file or a database?
18:13.48laurisi use binlog on the mysql side
18:13.49DennisGa MySQL database
18:14.00laurisand do incremental sql backups every weekend
18:14.14ChannelZyou can use mysqldump
18:14.26DennisGbut via SQL statements or do you just copy some files?
18:14.31laurisvia SQL
18:14.47lauristo be more precise - via mysqldump
18:15.09DennisGoke
18:15.18DennisGi'm checking the documentation of mysqldump right now
18:16.22*** join/#asterisk fisted (~fisted@unaffiliated/fisted)
18:16.31ChannelZmysqldump --complete-insert -u username database > /somewhere/mydatabase.sql
18:16.51DennisGthank you ChannelZ!
18:16.52lauriswhy would you need complete inserts?
18:17.05DennisGi have the same question hehe
18:17.07laurisit's waste of space
18:17.14ChannelZFine, do whatever you want. Sheesh.
18:17.22DennisGi believe that the data is more then enough
18:17.23ChannelZ(ever heard of compression?)
18:17.40DennisGand how can mysqldump create incremental backups?
18:18.12laurisit can't
18:18.41DennisGhmm.. but IF i have a dataset of 10M records then i have to wait for a long time...
18:19.25ChannelZThen you're stuck using binlogs in mysql which are basically transaction logs
18:19.42laurisDennisG, if you use binlog you don't have to do full backup very often
18:19.56laurisand you can also have slave mysql server to take load of primary server
18:20.08ChannelZYou just have to mind to make a backup of the database and clear the log which is like a starting point.. then the binlog can "replay" everything that happened to the database since the snapshot
18:20.28laurisas for incremental backups
18:20.33laurisI use rdiff-backup for this purpose
18:20.38lauristhis is kind of *diff* tool
18:20.54DennisGhmm... so the best way is like: scp (or something else) the binlogs, delete them and repeat this every week?
18:21.21laurisDON'T DELETE IT MANUALLY
18:22.07ChannelZhave fun. I'm going gun shopping.
18:22.15DennisGoke, with the tool mysqlbinlog then?
18:22.31laurismysqldump can do this for you in a proper way
18:22.34laurisjust read the manual
18:23.31DennisGi reading this: http://dev.mysql.com/doc/refman/5.6/en/mysqlbinlog-backup.html
18:23.36DennisGi'm *
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18:25.29DennisGcreating incremental backups like: mysqlbinlog --read-from-remote-server --host=host_name --raw binlog.000130 binlog.000131 binlog.000132
18:25.40DennisGrestoring backups like: mysql --host=host_name -u root -p < dump_file
18:26.08DennisGit looks promising
18:26.55DennisGit's even possible to create backups via a slave SQL server to offload the master
18:27.19lauristhat's what i was telling
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18:27.42DennisGhaha oke sorry, but it's not the mysqldump program
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18:28.40ven0mQuestion - does asterisk support "on-the-fly" codec change? meaning - can it handle situations where client switch codec in the middle of coversation for example
18:29.33DennisGven0m i think not.. SIP use a triple handshake system for using a correct codec
18:29.40DennisGeg Asterisk uses that
18:30.40ven0mhmm
18:31.17ven0mso basically the codec which the client will use is determined once he connects to astersk (event before he makes teh call) ?
18:31.21DennisGlauris, thank you for everything!
18:31.48DennisGyup, if Asterisk and the other end use the same codec
18:32.04DennisGaka you can use multiple codecs in a specific order
18:32.19DennisGfirst codec have then the highest priority
18:33.03ven0mSo even though client/server list the same codecs, i cannot switch between them, even through both client and server support it?
18:34.39DennisGchange the order of the codecs
18:34.48DennisGbut it will work after a new call
18:35.16ven0mI see. what i wanted to do is build an iphone client, with abiliy to switch the codec it uses
18:35.22ven0mI guess its impossible
18:37.31DennisGi think it is with SIP
18:37.36DennisGbut i'm not a SIP-expert..
18:38.31ven0mI see
18:38.32ven0mthanks
18:45.22[TK]D-Fender[13:31]ven0mso basically the codec which the client will use is determined once he connects to astersk (event before he makes teh call) ? <- there is no "codec" before a call
18:45.31[TK]D-Fendercodec offerings are part of the call
18:45.50[TK]D-FenderCheck the SDP offerings in the INVITE
18:46.11ven0mSDP? (sorry i am a new in the space)
18:47.13ven0mIs there some guide you could recommend which describes how asterisk determines which codecs both parties will be using throughout the call?
18:50.33[TK]D-Fenderven0m: Enable SIP DEBUG at * CLI and watch the invite process
18:51.12ven0mThank you
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21:06.34Cadeywill Digium still release asteriskNow now that FreePBX have there Distro sorted?
21:08.26KNERDWhat could cause a phone to get a "SIP/2.0 401 Unauthorized" error while functioning just fine for 2 months, then power is acctidentally kicked out, now gets that error.
21:08.46Cadeypower out of the phone?
21:08.51Cadeyor out of the asterisk box?
21:08.57KNERDphone
21:09.24Cadeywas the extension/secret on the phone only in memory and not in its config file
21:09.34Cadeycheck the config file first is what I would do
21:09.53KNERDconfig is fine, passwors is still in phone
21:09.58KNERDnothing has been changed
21:10.33Cadeycan you run a sip trace on the box whiile the phone is trying to register
21:11.08KNERDI have never done this
21:11.44KNERDsip set debug on?
21:12.07Cadeysip set debug on <--- in console
21:12.25Cadeyyou can pipe all the output from verbose and debug into a text file also
21:12.25KNERDyeah I did this, and how I am telling what error I am getting
21:12.47Cadeywell can you see it authenticating
21:13.20Cadeywhat messages is the phone sending before asterisk returns the 401
21:13.23KNERDI can see it trying to register, but the system is refusing it
21:13.41Cadeycan you paste the trace?
21:13.45Cadeypastebin.com or somthing
21:14.31Cadeyyou can pipe the verbose+debugger output to a file from the /etc/asterisk/logger.conf config
21:14.50Cadeywith the -> /var/log/my/file/path/file.txt
21:15.16Cadeyits been a while since I did it so that may be incorrect syntax but its as simple as that :)
21:17.21KNERDhttp://pastebin.com/xKShBipm
21:17.28KNERD1002 is the one
21:19.20Cadeyhum
21:20.35KNERDexactly..nothign given
21:20.46Cadeycan you check the extensions.conf
21:21.30KNERDcheck it for what?
21:23.45Cadeyits registering with out the extension
21:23.54Cadey51.REGISTER sip:localhost
21:25.06KNERDacutally I chnaged that in the paste because I dont want it to be part of Google history.. it's a domain name in there
21:25.08Cadeyoh no sorry my bad
21:25.17Cadey:)
21:25.55KNERDyeah this setup had been working fine for 2 months, then all of a sudden it's not working?
21:26.46Cadeywell a 401 is because the credentials are wrong normaly
21:27.09Cadeyso triple check the secret in the config file of the phone and the secret against 1002 in the extensions.conf
21:27.17KNERDif they were wrong, my fail2ban would have banned the IP
21:27.39Cadeycheck anyway
21:27.42KNERDi did
21:27.45KNERDonly 6 times
21:28.04Cadeystupid question but have you tried rebooting asterisk
21:28.16KNERD"amportal restart"
21:28.33Cadeyasteirsk-CLi> Core restart now ?
21:28.35KNERDbut not a complete system reboot
21:28.42KNERDyeah I did that too
21:28.49Cadeydo a sip show peers
21:28.52Cadeydo you see 1002
21:29.25KNERDyes
21:29.35Cadeyok, try this
21:29.48Cadeylog out 1001, change its config to 1002 and see if that phone work
21:30.03KNERDok
21:30.34Cadeyalso, are you authing with scretMD5 or normal secret
21:30.35KNERDmayb eyou mean 1000
21:31.14Cadeybecause the 401 is telling the phone to use MD5 for auth
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21:32.03KNERDhmm...let me recheck the phone
21:33.38KNERDi dont even see a whay to see on th ephone..see no MD5
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21:38.12Cadeywell
21:38.29Cadeycheck in extension.conf a sec and see if secretMD5 is set for 1002
21:38.37Cadeyor is it just Secret
21:39.45KNERDit seems the phone does have the ability
21:41.39Cadeywell I may be wrong so check the extension.conf and see if its asking for SecretMD5
21:42.03Cadeyas the phone may simply have reset its self to open text instead of MD5 :)
21:44.02KNERDlooking
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21:48.25Cadeyany good?
21:53.16KNERDno
21:56.31wdoekes2KNERD: looks like your phone doesn't get the answer from your asterisk
21:56.38wdoekes2iptables?
21:57.46wdoekes2you're seeing retransmissions of the original register.. no amount of password setting will fix that
21:59.17KNERDas I mentioned.been functioning fien for 2 months then I kicked the power out by accident
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21:59.36KNERDthat should be no reason to stop functioning properly
21:59.39wdoekes2hm.. the problem is probably with the nat router on 99.27.165.224
21:59.57KNERDi can try to power cycle it
22:00.19wdoekes2yea.. or switch local port in the linksys to something other than 5061
22:02.43KNERDwhy would it be on 5061?
22:03.47KNERDwell..power cycle time...brb
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22:05.28KNERDand it worked
22:05.42KNERDwdoekes2: thankgs for poitning that out
22:05.58wdoekes2np
22:06.13KNERDCadey: and thanks too
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23:34.11blognewbhey guys
23:35.08blognewbi hope this is not a stupid question: is there a way to use an android phone without being signed up to a network/carrier? just use google voice and obi?
23:36.12[TK]D-Fenderblognewb: over what?
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23:37.00blognewbhey tk sorry i know its off topic, I forgot the name of the pbx channel
23:37.04raggithe Global Satellite Meganet, duh!
23:37.24[TK]D-Fenderblognewb: That's fine. So wover what?
23:37.27[TK]D-Fenderover*
23:38.33blognewbi dont even know what im talking about lol i guess i need to connect a network still, i just find everyone else expensive except for PlatinumTel but they don't carry the galaxy s2
23:39.17[TK]D-Fenderthen buy an unlocked phone
23:40.03[TK]D-FenderOr a locked one and unlock it
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23:46.28brainiacI am running Asterisk v. 1.8.7 and am trying to use the MeetMeCount app.  When dialing a conference room ext, the system complains that there is no such application.  Can anyone provide some insight?
23:53.31p3nguinShow us.
23:53.33p3nguin~pb
23:53.33infobot[~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com , http://pastebin.us , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ ; or install pastebinit with yum or aptitude.
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