IRC log for #asterisk on 20090506

00:00.10yo-mamaMeaw: if you HAVE to use g729 and you are REALLY cheezy, buy one license, get it working and then use cracked for the other channels
00:00.32pmhaddadMeaw, i would just avoid g729 - i see no real reason why you need it
00:00.44pmhaddadunless you know something about your setup i don't
00:00.45yo-mamaMeaw: how new is your install?
00:00.53pmhaddadwhich is totally possible
00:00.58yo-mamaMeaw: how good are you with Linux?
00:00.59[TK]D-Fenderyo-mama: Doesn't work
00:01.55yo-mama[TK]D-Fender: You used to be able to do that trick! I have a few of them scattered throughout Cali!
00:01.58yo-mama:P
00:02.37Meawyo-mama, im not the one who did the whole setup, we hired a guy to do it but im trying to finalize it.. we have a heavy traffic on the E1 thats why im trying to get this done at this time, after few hours i should unplug the E1 and put it on the original server
00:02.39yo-mamaMeaw: Just reinstall asterisk!
00:02.57pmhaddadi already suggested that
00:04.00yo-mamapmhaddad: all hes has to do is backup his /etc/asterisk/ folder and ./configure && make clean && make && make install
00:04.44*** join/#asterisk infernix (i=nix@unaffiliated/infernix)
00:06.16yo-mamaHey, who here is am sms guru?
00:15.07cvnetif incoming call from DID is SIP which hits the asterisk box, and your outbound provider is h323 would asterisk translate it by itself?
00:17.40[TK]D-Fendercvnet: Just like EVERY call.  2 channels bridged.  Not Raw-Cat Science you know.  Never called from a SIP phone and out a DAHDI/ZAP channel before?
00:24.32nkohhi've never even built dahdi =(
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00:45.28Meawallright, i have recompiled asterisk
00:45.38Meawnow im getting this error when i try to load module g729
00:45.40MeawError loading module 'codec_g729-ast16-gcc4-glibc-core2.so': /usr/lib/asterisk/modules/codec_g729-ast16-gcc4-glibc-core2.so: cannot restore segment prot after reloc: Permission denied
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01:58.09kc8pxy[TK]D-Fender:  "raw-cat science"???
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02:17.46[TK]D-Fenderkc8pxy: hukt on fonix werkt 4 u
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04:47.06joelsolankiHi all
04:47.18joelsolankii have installed asterisk 1.6 for testing purpose
04:47.51joelsolankiI need to test the g729 codec stuff. can anybody provide me link for free g729 codec which will work with asterisk 1.6.0
04:47.52joelsolanki?
04:50.06drmessanojoelsolanki: Doubt it.. Digium handles the licensing for legal use of G729, and using anything else is more or less "warez".
04:52.40joelsolankiyes i will buy licenses once this setup works
04:52.52florzthat would very much depend on the jurisdiction you are in, I suppose
04:52.53joelsolankii use asterisk 1.4 and have licenes for it.
04:53.14joelsolankibut i cant move the licenses until i test it sucessfully on asterisk 1.6
04:53.20trnzmetahttp://www.readytechnology.co.uk/open/ipp-codecs-g729-g723.1/
04:53.22joelsolankitherefore i need g729 for testing purpose.
04:53.24joelsolankiok
04:54.42drmessanoNice
04:54.57joelsolankihttp://asterisk.hosting.lv/#bin
04:55.06trnzmetayeah that's the one
04:55.09joelsolankii actually downloaded g729 from above location
04:55.22trnzmetasame same but different
04:55.22joelsolankii have asterisk 1.6.0 and core2duo server
04:55.39joelsolankiso which codec i should download ?
04:55.49joelsolankii see alot
04:56.02joelsolankicodec_g729-ast16-icc-glibc-core2.so  ?
04:56.59joelsolankitrnzmeta ?
04:57.26trnzmetajust read the info
04:57.33trnzmetacat /proc/cpuinfo
04:57.48trnzmetaand see which flags your processor supports and match equiv
04:59.12joelsolankii see following
04:59.13joelsolanki<PROTECTED>
05:00.20*** mode/#asterisk [+b trnzmeta!n=bleh@secure27.lnk.telstra.net] by Qwell
05:00.31*** kick/#asterisk [trnzmeta!i=north@pdpc/sponsor/digium/Qwell] by Qwell (google: contributory infringement.)
05:01.12joelsolankioh trnzmeta was kicked ? why ?
05:01.28drmessanogoogle: contributory infringement.
05:02.09joelsolankiok
05:02.34Qwelljoelsolanki: you are also lucky you're still here.
05:02.52QwellYou *cannot* use those codecs.  Period.
05:03.03ltd_wkjoelsolanki: g.729 is patent encumbered, Digium sells licenses commercially.  The above codec is infringing on that patent.  You do the math.
05:03.18joelsolankiI see.
05:03.33drmessanotried to warn earlier
05:03.42joelsolankiqwell: so how can i test g729 ?
05:03.49QwellBuy a license
05:03.53joelsolankiit will work in pass thru mode if codecs is not installed ?
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05:03.58Qwellyes
05:03.59florzyou people are aware that patents are a terrirorial matter?
05:04.09drmessanoSame way you can test Vista or AutoCAD?
05:04.10Qwellflorz: most countries respect these patents
05:04.39QwellAnyways, *I* and *this channel* will not take part.
05:05.11joelsolankiwell i tried the calls without installing g729 to use as pass thru mode. but it didnt worked. the case was i wanted transcoding from g711 to g729
05:05.23QwellThat isn't pass through...
05:05.24joelsolankii think pass thru g729 will not transcode right ?
05:05.34joelsolankiyes got it.
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05:05.37Qwellthat is what pass through means
05:05.55*** mode/#asterisk [-b trnzmeta!n=bleh@secure27.lnk.telstra.net] by Qwell
05:06.17*** mode/#asterisk [+b trnzmeta!n=bleh@secure27.lnk.telstra.net] by Qwell
05:06.22*** mode/#asterisk [+b *!*n=bleh@*.guard.com.au] by Qwell
05:06.22*** kick/#asterisk [supa_disko!i=north@pdpc/sponsor/digium/Qwell] by Qwell (ban evasion)
05:06.25joelsolankimeans what ever codec the client has it will be passed. if customer is using linksys ata with g729 and we dont have codec installed on asterisk then asterisk will just pass g729 codec right ?
05:06.29QwellI was going to remove it..
05:07.21joelsolankiqwell: is my understanding correct above ?
05:07.27Qwelljoelsolanki: unless you need to do things with the audio, yes
05:07.34joelsolankihmm ok
05:07.46*** join/#asterisk Techdeck (i=Techdeck@77.125.43.202)
05:07.51Techdeckhey guys
05:07.53ltd_wkjoel: It's only $10 USD per concurrent call channel.
05:08.04Techdeckcan you guys help me with G.729?
05:08.08joelsolankiit was nice if digium had provided 1 g729 license for free. just for testing prupose :)
05:08.20Qwelljoelsolanki: tell the patent owners.
05:08.26joelsolankihehe :
05:08.27joelsolanki:)
05:08.40TechdeckI'm getting a permission denied error when trying to load the .so file
05:08.44KyleKcan I get some basic info on g729? I don't get why its a bfd that its patented, or that people want to use it
05:09.00QwellKyleK: the size/quality are "good"
05:09.05Techdeckwere you guys just talking about it??
05:09.37ltd_wkI don't agree that the quality is good, but the size certainly is :>
05:09.58Techdeckanyone?
05:10.02Qwellltd_wk: sure, that's fair.  I was saying the ratio though
05:10.10Qwellare there better?  absolutely
05:10.43Techdeckare you ignoring me?
05:10.51Techdeckis it because I'm not a female?
05:10.53QwellTechdeck: no..  permission denied doing what?
05:11.02TechdeckQwell, loading the module
05:11.06Qwellhow?
05:11.13KyleKTechdeck: permission denied usually means cant read the file or its the wrong file
05:11.13Techdeckmodule load ..
05:11.24Techdeckthe file is wrong, and I chmod'd it to 777
05:11.27Techdeckerr, right*
05:11.30Qwellcheck selinux stuff
05:11.37Techdeckoh, good idea
05:11.39Techdecklet's see
05:11.51Qwellotherwise, I would suggest calling Digium support
05:12.18TechdeckI did..
05:12.46Techdeckthey told me "You are doing something wrong then"
05:13.57Techdeckwhy thank you for that valuable information
05:14.06TechdeckI hate tech support
05:14.40Techdeckhmm nope, selinux seems disabled
05:14.57Techdeckany other ideas?
05:16.06jbjulyjust compiled and installed the latest beta 1.6.2 package, but I keep getting Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exit?) -- and it exist.
05:16.15QwellTechdeck: Do you have other modules in that dir?
05:16.28Techdeckyeah, tons
05:16.42Qwellpastebin the the console output
05:16.56Techdeckgimme one sec, I think I found something
05:17.33Techdecknvm, I had a typo, my bad
05:17.34Techdeckthanks!
05:18.20jbjulyjust compiled and installed the latest beta 1.6.2 package, but I keep getting Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?)
05:18.33nkohhis asterisk running?
05:18.39nkohhps aux | grep astersisk
05:18.49jbjulyno
05:18.57nkohhwell then that would be a fairly severe problem
05:19.02nkohhwhat're you typing?
05:19.05Techdeckmaybe you should grep asterisk then :P
05:19.05nkohhwhat command
05:19.19nkohhTechdeck: you meanGNU grep doesn't figure out what i want to search for, too?!?!
05:19.20jbjulyasterisk -rvvvv
05:19.24nkohhjbjuly: type asterisk first
05:19.25nkohhjust asterisk
05:19.30nkohhand then press the return button (enter)
05:19.37nkohhthen try the -rvvvv again
05:19.48jbjulysame problem
05:19.51Techdecknkohh, they have it as a feature request
05:19.52Techdeck:)
05:20.10nkohhjbjuly: did it output anything when you typed asterisk?
05:20.42jbjulynone
05:20.55nkohhwell that is very interesting
05:21.08Qwelljbjuly: When you installed, did you clean out the modules dir like you're supposed to?
05:22.56jbjulyIt's a fresh 1.6.2.0 beta1 compile and install
05:23.09Qwelland you've never had Asterisk on this system?
05:23.16jbjulyyes
05:24.20nkohhjbjuly: sounds like you might be suffering from bidirectional checksum rejection
05:24.32nkohhdo you have any anti static wrist bands?
05:24.37nkohhyou're gonna need three or four
05:24.53jbjulyi did restarted several times, changing /var/*/asterisk owner/groups/permissions, still the same problem
05:26.18Qwellasterisk -c
05:26.22Qwellput the entire output on pastebin.com
05:32.59jbjulyhttp://pastebin.archlinux.fr/347023
05:33.46Qwelladd a few v's
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05:36.18jbjulyhttp://pastebin.archlinux.fr/347024
05:36.58QwellSo, I'm going to ask again without the possibility of a negated answer.
05:37.06QwellHave you had Asterisk installed on that system before this?
05:39.00seanbrightpickle
05:39.20nkohhQwell: no, he hasn't had it installed before this. he installed it himself.
05:39.29jbjulyno, i had asterisk 1.6.0 but uninstalled it
05:39.56QwellOkay, and did you clean out the modules when you installed, like you're supposed to?
05:40.16jbjulyhow do I clean the modules?
05:40.54QwellYou know the big warning message that gets output when the `make install` is finished?
05:40.54Qwellthat
05:41.34jbjulyoldmodcheck:
05:42.27seanbrightrm /usr/lib/asterisk/modules/*.so
05:42.29seanbrightgmake install
05:42.37seanbrightwho do i have to screw around here to get you to try that?
05:42.43seanbrightscrews himself
05:49.28lanningremember, "Righty, tighty." :)
05:50.43seanbrightway ahead of you
05:50.46seanbrightbut for serious
05:51.24drmessanoLEFT LOOSEY
05:51.32drmessanoErr shit
05:51.35drmessanoLEFTY LOOSEY
05:51.57drmessanoTwo wrongs dont make a right, but three left's do
05:56.36jbjulyTHANKS It works!
05:57.08seanbrightQwell: nice work
05:57.35Qwelldoesn't that warning say something like "YOU REALLY SHOULDN'T IGNORE THIS"? :(
05:57.50seanbrightQwell: people run stop signs all the time
05:58.35seanbrightsomeone buy me a rubik's cube
05:58.42seanbrightannnnnnd go
05:58.50Qwell3x3?
05:58.58timgws9x9
05:59.07timgwsmake it something that is more sudoku-ish
05:59.08timgws:D
05:59.10seanbright3x3 yes
05:59.12seanbrightthe classic
05:59.13timgwshi guys
05:59.22Qwellrubix dodecahedron?
05:59.29seanbrightbite your tongue
05:59.59timgwshttp://twistypuzzles.com/forum/download/file.php?id=3713&sid=b8af5db7c60d41c95ffe97d036b46f97
06:00.04timgwshttp://twistypuzzles.com/forum/download/file.php?id=3714&sid=b8af5db7c60d41c95ffe97d036b46f97
06:00.09timgws^ pictures of the 9x9 :D
06:01.03seanbrightthat's pretty hot
06:01.16timgwshttp://twistypuzzles.com/forum/viewtopic.php?t=6315 xD
06:02.07seanbrightsure.  ruin it for me.
06:04.08KyleKargh too many tabs
06:04.48seanbrightyou gotta order something if you want a tab
06:05.55KyleKi mean in google chrome
06:06.23seanbrighti mean in BttF
06:06.34seanbrightis tired of people not getting his movie references
06:06.37seanbrightgoes to sleep
06:06.47KyleKah, well at 36 tabs i lose the icons
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06:07.11KyleKeh i cant even remember the correct pronounciation of 1.21 jiggawatts
06:07.36KyleKapparently its one point two one, not one point twenty-one
06:08.16QwellKyleK: it's the latter
06:08.39KyleKoh so I was right
06:09.42Qwelland it's gigawatts
06:10.25KyleKwell hes got a bit of a jay sound in it
06:10.43Qwelldoesn't mean it's spelled that way :p
06:11.02Qwellhttp://www.moviewavs.com/0085412111/WAVS/Movies/Back_To_The_Future/greatscott.wav
06:11.33Qwellbed
06:11.44KyleKgood night
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06:14.33KyleKare the asterisk sounds available in g729?
06:14.46MaliutaLapI have a set somewhere
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06:14.58MaliutaLapor you can always transcode them
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07:04.44salzhcan ooh323 support video now?
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07:09.00henkmoin
07:10.23henki have a phone number from a sip provider (let's assume it's 12345), is it possible to have different extensions with that number, let's say 12345-1 and 12345-2? i tried dialling another digit after my own number, but the call is never seen on my asterisk.
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07:45.59tamielHello, sometimes after restarting asterisk, I have some pri channels stuck with "PRI Flags: Resetting". Any idea ?
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08:13.29joobiesup ladies
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08:47.36tzafrir_laptoptamiel, what versions of asterisk? libpri?
08:49.54tamieltzafrir_laptop: asterisk 1.4.24.1 and last libpri 1.4.9
08:50.55tamieltzafrir_laptop: and I have this warning : chan_dahdi.c: Unable to specify channel 1: Device or resource busy
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08:59.23KyleKwhat are people using to transcode files to g729?
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09:04.09KyleKnm "file convert"
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09:17.01amaacheHow can i know if my PSTN phone line is numeric
09:21.04KyleK?
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09:27.12amaachei have to connect my * TrixBox with a old analogic hicom PBX
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09:27.33amaachei have TDM4000
09:27.42amaachei have TDM400
09:30.41amaachecan i connect * to PBX hicom with E1/mic
09:31.11joseph__guys any one worked before with app_dial.c
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09:39.09helloriteshany recommendation for a SIP trunk provider for US and International calls. I need to setup immediately so online activation is required?
09:40.30helloriteshAlso, I have outbound calls through Manager's interface and I wanted to know if there is a way (simialr to Dial Command) to specify multiple carriers (trunks) and the first one to answer gets the call?
09:43.45*** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw)
10:07.36viraptorhow safe is it to call dial_exec() from the chan_sip.c ?
10:08.09viraptor(dial being a module and all that stuff)
10:11.05amaacheHi,can i connect * with mic/siemens Hicom?
10:12.06*** join/#asterisk shinao1 (n=shinao1@78.138.29.146)
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10:21.08*** join/#asterisk huye (n=huye@soho2.i-xanadu.com)
10:32.38Jimbo12hi - has anyone in here sucessfully managed to connect an Asterisk 1.6.1.0 box with Microsoft OCS 2007 R2?
10:46.34*** join/#asterisk torrikft (n=afraguas@m85-94-191-59.andorpac.ad)
10:47.17torrikftmorning all, im trying to integrate OCS2007 R2 with asterisk 1.6.9 has anybody experience with this setup?
10:50.47*** join/#asterisk SkywaIker (n=pirch@58.147.17.166)
10:51.56*** join/#asterisk bobsaccamano (i=cb7e888e@gateway/web/ajax/mibbit.com/x-30197eca311e1d9f)
10:52.26bobsaccamanohi..is there an option for enabling/disabling call forwarding in asterisk?
10:53.22bobsaccamanoI am hooking it up with a client device that has call forwarding support..so i want to separate server side and client side call forwarding
10:55.45*** join/#asterisk helloritesh (n=hellorit@116.197.178.83)
10:56.08helloriteshGuys, any recommendation for SIP trunk provider? I need to route some calls asap
10:56.09bobsaccamanoanybody there?
11:00.34*** join/#asterisk proxium (n=proxium@196.203.51.238)
11:02.12helloritesh.
11:02.24proxiumHi, everyone, Finally I can use my Vicidial to make manual oubound  call and my previous problem was the conference error with Meetme and the time synchronisation in Vicidial 2.0.5
11:03.48proxiumNow with thos issue Fixed I receive this error in CLI when I do predective Outbound Call: [May  6 11:58:47] ERROR[3085]: chan_sip.c:15919 sipsock_read: We could NOT get the channel lock for SIP/tofreepbx1-0a54b700!
11:03.48proxium[May  6 11:58:47] ERROR[3085]: chan_sip.c:15920 sipsock_read: SIP transaction failed: 74bd46ba1a032e7243c6bd9d3694e4fd@192.168.1.89
11:06.24proxiumAny idea about this ?
11:09.03*** join/#asterisk Great_Anta_Baka (n=tensai@196.33.159.83)
11:30.12*** join/#asterisk maxagaz (n=maxagaz@soho2.i-xanadu.com)
11:41.41*** join/#asterisk gpuk3 (n=me@ARennes-205-1-6-216.w80-14.abo.wanadoo.fr)
11:44.29gpuk3hi all. We're running asterisk1.6.0.6 and have a pure voip setup (i.e. no Digium hardware or FXS/FXO cards). We have a Digium TDM411B PCI card arriving tomorrow and I wanted to know if there is a way I can check that our build of asterisk will support it without a re-compile?
11:45.31gpuk3Presumably the kernel will need the correct modules to talk to the card?
11:46.23beekgpuk3: You'll need to compile DAHDI, then recompile Asterisk.
11:47.25*** join/#asterisk KermitTheFragger (n=KermitTh@118-197.bbned.dsl.internl.net)
11:49.19gpuk3beek: right. From memory, I think we compiled asterisk with zaptel rather than dahdi
11:50.09gpuk3yep, just checked my notes - zaptel 1.4
11:50.26andrebarbosaanyone using hpec with dahdi?
11:50.28gpuk3but i didn't record what zaptel modules etc. we selected at compile time
11:51.00beekgpuk3: You're using what version of Asterisk
11:51.02beek?
11:51.10gpuk31.6.1.0
11:51.17gpuk3we elected to stick with zaptel
11:51.19beekThen zaptel is a total waste of time.
11:51.21gpuk3rather than move to dahdi
11:51.27gpuk3cos at the time
11:51.31gpuk3all we needed was the zt_dummy
11:51.33gpuk3timer
11:51.45beek1.6 requires DAHDI.
11:51.53gpuk3we hadn't planned to use any hardware
11:52.07beekBut now that you do, 1.6 requires DAHDI
11:52.08gpuk3ok... looks like i need to get dahdi up and running then
11:52.18gpuk3thanks for the heads up
11:52.22beekIt's not all that different from zaptel.
11:52.30gpuk3good :)
11:52.38gpuk3hopefully wont be too much of a mission
11:52.45beekIt's virtually replace zap_xxxx with dahdi_xxxx
11:52.59gpuk3ahh kk
11:53.09gpuk3goes off to do it
11:53.11beekSlightly different configuration files.
11:53.16gpuk3thanks again
11:54.38joseph__<PROTECTED>
11:55.06*** join/#asterisk stope (n=nobody@69.60.247.142)
11:55.12joseph__please inform me if you have an idea on how to ,or just if you know a hint
11:56.09bobsaccamanohi..i need some pointers on how to write a dialplan for call forwarding in asterisk
11:58.04beekbobsaccamano: http://tinyurl.com/c9bnrd
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12:04.31bobsaccamanois there an asterisk function for detecting a busy line in sip?
12:09.06stopeyou could check the ${DIALSTATUS}
12:13.39*** join/#asterisk yang (i=yang@CAcert/Assurer/freenode.sponsor.yang)
12:16.45*** join/#asterisk leifmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage)
12:16.45*** mode/#asterisk [+o leifmadsen] by ChanServ
12:17.15*** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163)
12:18.10beekmorning [TK]D-Fender
12:18.26bobsaccamanostope: would DEVICE_STATE function work?
12:18.46[TK]D-Fenderbeek: I feel like shit-on-a-stick
12:19.05beek[TK]D-Fender: Did you have a wild evening or are you ill?
12:19.56[TK]D-Fenderbeek: ill.  Day 2.  Some kind of flu.  Barely runny nose, headache, moderate congetion and loss of energy
12:20.26stopebobsaccamano: http://pastebin.ca/1414157
12:20.34beek[TK]D-Fender: You should be in bed getting rest.
12:21.35*** join/#asterisk salzh (n=Administ@122.144.138.49)
12:22.21*** join/#asterisk malcolmd (n=malcolmd@pdpc/sponsor/digium/malcolmd)
12:22.24bobsaccamanostope: thanks!
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12:35.40[TK]D-Fenderstope: you've nested an expression unnecessarily.  remove the outer $[]
12:38.51*** join/#asterisk jtodd (n=jtodd@88.128.82.57)
12:38.51*** mode/#asterisk [+o jtodd] by ChanServ
12:40.06stopeyes, yes, correct, tx   :)
12:40.33stopeI think that snippet was grabbed from another one that had an 'or' in it
12:43.47*** join/#asterisk ariel_ (i=3fd6eca9@gateway/web/ajax/mibbit.com/x-a7d297cbeae3e137)
12:45.05leifmadsenyep... all you're doing is always returning true
12:45.14leifmadsensince it's just checking if there is data, and since there is data, it is true
12:45.23*** join/#asterisk jaytee (n=jforde@unaffiliated/jaytee)
12:48.00*** join/#asterisk Kobaz (n=kobaz@its.kobaz.net)
12:52.33leifmadsenI've got "joinempty=yes" and "leavewhenempty=no", but callers are still falling out of the queue when agents are paused. Am I missing an option?
12:54.53[TK]D-Fenderleifmadsen: Whats the exit var say?
12:56.37*** join/#asterisk brad_mssw (n=brad@shop.monetra.com)
12:57.28leifmadsen[TK]D-Fender: CANCEL
12:58.12leifmadsennevermind, that was the agent
12:58.20leifmadsenI have to add a line of debugging
12:58.52leifmadsenoh I see the problem I think anyways
13:02.59leifmadsenyep, I had an IF() function in the Queue() call that was putting in a timeout value from a global variable if no timeout value was not returned from the database.
13:03.16leifmadsenanother IF() function to check for this particular queue and to return null fixed it right up :)
13:06.54[TK]D-Fenderleifmadsen: NA NA NA NA NA!
13:06.55[TK]D-Fenderleifmadsen: Leif is Leif!
13:07.25*** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net)
13:07.42leifmadsen[TK]D-Fender: I don't want to meet your mom!!
13:07.51[TK]D-Fenderleifmadsen: I just want
13:07.52*** join/#asterisk juanIMP (n=Juancho@200.71.41.22)
13:13.42*** part/#asterisk Sam2002gs (n=Sam2002g@h-213.61.105.202.host.de.colt.net)
13:13.45*** join/#asterisk Sam2002gs (n=Sam2002g@h-213.61.105.202.host.de.colt.net)
13:16.14[TK]D-Fenderleifmadsen: ... tease :p
13:23.42*** join/#asterisk Victor_Yure (n=victor@unaffiliated/victoryure/x-837844)
13:26.31*** join/#asterisk anonymouz666 (n=anonymou@189.24.118.128)
13:27.13Kattyhummmm
13:28.32*** join/#asterisk lowlevel (n=Stuart@lowlevel.ca)
13:28.36[TK]D-FenderKatty: Mew.
13:30.38*** join/#asterisk bgmarete (n=marebri_@196.201.210.130)
13:32.02defsworkHey did I update here with my problem the other week ? Turned out the Echo Cancellation chip was dead on my Sangoma A101
13:32.17jayteewow
13:32.34defsworkturn off echo cancellation - all worked
13:32.40beekmorning jaytee
13:32.44defsworknew card in now - works properly
13:32.48*** join/#asterisk BlargMaN00 (n=blarg@12.234.16.130)
13:32.52jayteemorning beek
13:33.02defsworkStressful though when on site and nothing works :)
13:33.29*** join/#asterisk tobias (n=tobias@user-0ce2hp1.cable.mindspring.com)
13:37.37leifmadsen[TK]D-Fender: ! ! !
13:37.45leifmadsentease, lol
13:38.07leifmadsensorry, I don't get notified unless I'm looking at the window
13:38.22stopeI'm trying to get SLA working on 1.4.23, am I wasting my time with it?
13:38.35leifmadsencan't say I've ever used it
13:38.57stopeit works with SIP trunks for outgoing calls, but incoming is where I'm having issues and my customer 'wants it'   :(
13:39.04*** join/#asterisk awk_r (n=awk_r@nat/digium/x-3c3a7eb32bf3ffe7)
13:39.18*** join/#asterisk UQlev (n=yuriy@nb11-125.static.cytanet.com.cy)
13:40.25bgmareteHello guys. Does SugarCRM support Asterisk through any free component? Anyone with experience on this here?
13:40.26[TK]D-Fenderstope: Only works for "lines", not "phones", and requires a lof of rpesence enabled speed-dials.
13:40.36[TK]D-Fenderstope: Maybe some day * will support real SIP-B
13:40.51stopeok, maybe there's a better approach, I need to place a call on hold and pick it up at another extension... should I just use call parking instead? (hosted pbx, many contexts..)
13:41.38[TK]D-Fenderstope: Yup
13:41.38stopek, tx
13:42.25*** join/#asterisk wilsonj (n=jeremy@unaffiliated/dethstar)
13:45.34proxiumHi again, I don't receive any feedback, so I post again (may be [TK]D-Fender has an idea)
13:45.50proxiumI receive this error in CLI when I do predective Outbound Call:
13:45.57proxium[May  6 11:58:47] ERROR[3085]: chan_sip.c:15919 sipsock_read: We could NOT get the channel lock for SIP/tofreepbx1-0a54b700!
13:46.07proxium[May  6 11:58:47] ERROR[3085]: chan_sip.c:15920 sipsock_read: SIP transaction failed: 74bd46ba1a032e7243c6bd9d3694e4fd@192.168.1.89
13:46.09[TK]D-Fenderproxium: What ver of *?
13:46.12*** join/#asterisk Victor_Yure (n=victor@unaffiliated/victoryure/x-837844)
13:46.19*** join/#asterisk tobias (n=tobias@user-0ce2hp1.cable.mindspring.com)
13:46.25proxium1.4.22
13:50.31[TK]D-Fenderproxium: Well thats alreadya  few versions behind
13:50.46*** join/#asterisk theHub (n=theHub@69.177.93.21)
13:51.40proxium[TK]D-Fender: plz can you explain ?
13:52.04*** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq)
13:52.30*** join/#asterisk [intra]lanman (n=lanman@freeswitch/developer/intralanman)
13:52.31stopeversion 1.4.24.1 > 1.4.22
13:52.45[TK]D-Fenderproxium: What's to explain?  We're up to 1.4.24.1
13:52.59[TK]D-Fenderproxium: Sorry, I don't teach mathematics.
13:53.33stope1.6.1.0 ready for production use?
13:54.47proxiumOk, I don't need mathematics courses but I don't speak English well that's all
13:56.29beekproxium: Any time you get strange errors and are not at the newest version of Asterisk you're going to want to upgrade first.
13:58.27proxiumbeek: ok, I'll try to upgrade but I should be sure about that error to be (version dependant) not a bad configuration or something else.
13:58.31[TK]D-Fenderstope: 2 words : bleeding edge
13:58.49stopeaye
13:58.53[TK]D-Fenderstope: I would always wait about 1-2 decimal increments for the big bugs to come out
13:59.10stope1.4.24 it is ....   :)
13:59.13[TK]D-Fenderstope: Or 2 months without any announcements
13:59.25[TK]D-Fenderstope: 1.6.0 is viable.
13:59.29beekstope: 1.6.0.9 has been rock-solid for me.
13:59.43[TK]D-Fenderstope: remember that 1.6.1 is a whole other branch with the new release cycle
14:00.06stopehmm, maybe I'll try 1.6.0.9 if it's been a good soldier so far
14:00.21stopeTLS would be good
14:00.33beekstope: I've had zero issues with the 1.6.0.x branch.
14:00.34*** join/#asterisk el_-- (n=el@asterisk.net.informatik.tu-muenchen.de)
14:00.42stopek, tx
14:01.53el_--Hi I'm using trixbox and I need some help with dialpatterns... The PBX should change the numbers that the international prefix is allways used...e.g. I dial 03 0301234455 it should dial 004930123...
14:02.12el_--the first 03 should choose the outbound route
14:03.37*** join/#asterisk mleino (n=mle@140-120.adsl.lpoy.dnainternet.fi)
14:03.59[TK]D-Fender~trixbox
14:04.00infobotrumour has it, trixbox is a full linux distro that includes FreePBX and other 3rd party add-ons. It is all this extra stuff which makes trixbox VERY difficult to support, and is not supported in #asterisk. Try asking in #trixbox or on their forums & wiki at http://www.trixbox.org/.  We do not recommend using it.
14:04.04[TK]D-FenderelNot supported here
14:04.07jerliqueI have a registration, which is set to infineltey regsiter ever 20 seconds, yet it stops try after a while. is this a bug>
14:05.09*** join/#asterisk desdesdesdes (i=desdesde@196.211.34.3)
14:05.28desdesdesdeshi there what does jitterbuffer=yes do?
14:05.35nkohhjerlique: regardless of whether or not there is something preventing you from reregistering every 20 seconds, it is a fantastically bad idea to register that frequently.
14:05.57nkohhdesdesdesdes: http://www.voiptroubleshooter.com/problems/jitterbuffer.html
14:06.23el_--how would a dialpattern look like for asterisk? withoud freepbx?
14:07.22jerliquenkohh: The main switch sets the minimum to 180, but asterisk is set low, so that if it d/c from the switch, it will reconnect quickly
14:07.39[TK]D-Fenderel_--: depends on a lot of things
14:07.51desdesdesdesthx nkohh
14:07.55[TK]D-Fenderel_--: and we cannot help you witht he way you have to code this in FreePBX.
14:08.12[TK]D-Fenderel_--: It is not supported here.  Please use their support channels for this
14:08.16el_--ok thanks alot anyway
14:12.23mleinoHi, I have a problem in fresh Asterisk + FreePBX installation (Asterisk + addons 1.6.1.0, dahdi 2.1.0.4, dahdi-tools 2.1.0.2, libpri 1.4.9, freepbx 2.5.1). I have set up a conference via freepbx, and when I try to join to conf, first user goes in nicely after entering pin, the second one doesn't, if enter correct pin, nothing happens, line stays muted. And when I disconnect both users from conf, asterisk goes to 99% of cpu, but no errors ca
14:12.57*** join/#asterisk KavanS (n=KavanS@static-71-117-242-28.ptldor.dsl-w.verizon.net)
14:13.58*** join/#asterisk oej (n=olle@93.82.216.154)
14:15.36*** join/#asterisk seanmh (n=johndoe@198.59.129.24)
14:16.34plundraHmm, isn't there any way to do a iftime in regular ael? :-[
14:17.37[TK]D-Fenderplundra: Its jsut a function, I don't see why not
14:17.44[TK]D-Fenderplundra: And what is "regular" AEL?
14:17.44plundraRegular as in not AEL2 :)
14:18.00[TK]D-Fenderplundra: Still on 1.2?
14:18.25plundra[TK]D-Fender: Asterisk? No 1.6
14:18.45[TK]D-Fenderplundra: Last I checked there was no AEL as of 1.4
14:18.50[TK]D-FenderAEL1*
14:19.55plundraHmm, ok? :) I've just assumed that AEL2 always was written with the 2.
14:23.42*** join/#asterisk DavidR2008 (n=chatzill@fw1.safedataisp.net)
14:24.10plundraGah! :-D It _does_ work. Had to use the |-syntax for the timespec.
14:24.16plundra[TK]D-Fender: Thanks 8-)
14:24.53desdesdesdeswill jitterbuffer=yes and forcejitterbuffer=yes  help in solving delays in speech from fxs extension to iax2 trunk?
14:26.00[TK]D-Fenderdesdesdesdes: Doesn't solve latency, only helps cover up PL & mis-ordering better
14:26.46anonymouz666and could increase delay
14:28.20desdesdesdesdo you add this line on the iax.conf or zapata.conf or both?
14:29.12desdesdesdesor clearer question can u use the jitterbuffer in iax.conf
14:29.17jayteei've heard of triple-des encryption but never quadruple-des encryption
14:30.47[TK]D-Fenderdesdesdesdes: .... jitter is an IP problem.  do the math,,,
14:31.32jayteenuthin goes into nuthin nuthin times, carry the nuthin.....hmmm, I got nuthin
14:37.09*** join/#asterisk jasonwoot (n=some@bookit-dev.com)
14:37.52jasonwootdoes anyone have an isymphony update mirror address?
14:39.28*** join/#asterisk rbd (n=rbd@rrcs-96-10-27-206.se.biz.rr.com)
14:39.57rbdhi guys, in a macro is it possible to modify the values of ARG1, ARG2, etc. .... for example: exten => s,n,Set(ARG1=${IF($[ "${CALLERID(name)}" = "ConfExpansion" ]?ltqj:${ARG1})
14:40.52nkohhI don't have two phones in my home office to test this, but does Asterisk play music-on-hold if you put a conference on hold?
14:41.07nkohhI'm not sure if it makes any effort to' differentiate' between the two...
14:41.11nkohhoops, 'differentiate'
14:48.11*** join/#asterisk therealcircut (i=circut@smoke.dope.org)
14:48.15therealcircuthey all
14:48.35nkohhhello
14:48.45therealcircuti have a bunch of polycom 601's that im trying to set the line keys to dial extensions
14:48.53jayteewhat a funny URL
14:49.10therealcircutext, line key1 == my extension, line key2 == dial 701, line key3 == dial 6000
14:49.11therealcircutetc..
14:49.41therealcircutis this possbile? I have the latest firmware / bootloader and have been combing the documentation but i cant seem to find the documentation
14:50.25therealcircutto accomplish this...
14:50.33[TK]D-Fenderrbd: Should be able to
14:50.54rbdok thanks
14:51.31[TK]D-Fendertherealcircut: set the # of line-keys on your reg to fewer than you have, and the remainder spill over from your phone's directory in SD order
14:52.05[TK]D-Fendertherealcircut: thats what that magical button labeled "Directories" is for ;)
14:52.21*** part/#asterisk gego (n=rick@b238085.customer.hansenet.de)
14:52.44*** join/#asterisk Meaw (n=dino@213.244.81.144)
14:52.50therealcircut[TK]D-Fender: nice, i was looking for it in the registry stuff
14:52.54therealcircutwill give the directory a go
14:53.39*** join/#asterisk plq (n=plq@88.250.169.4)
14:56.37*** join/#asterisk jmacz (n=jmacz@190.144.75.22)
14:58.20eppigynever drinking again
14:58.28*** join/#asterisk Ziaeon (n=ziaeon@75-149-177-2-Miami.hfc.comcastbusiness.net)
14:58.33Techdeckuntil tomorrow
14:58.35*** join/#asterisk BlargMaN00 (n=blarg@12.234.16.130)
14:58.38leifmadsenbtw: I really hate how X-Lite modifies the audio volume when it starts and shuts down
14:58.50leifmadsenreally annoying when you're listening to music with headphones and it turns the volume all the way up
14:59.04*** join/#asterisk gr0mit (n=tim@extrt.txrx.org.uk)
14:59.17eppigyi keep my volume all the way up
14:59.27eppigyi get angry because xlite lowers my volume
14:59.58leifmadsenvolume all the way up is too loud for these headphones, and taking it from nearly all the way down to all the way up hurts my ears
15:00.04leifmadsenit shouldn't modify the volume at all!
15:00.11nkohhwhenever my coffee grinder is near my computer the screens flicker :(
15:00.19leifmadsenI don't drink coffee
15:00.22nkohhyou're missing out.
15:00.26leifmadsennah
15:00.29nkohhyah.
15:00.36BlargMaN00i drink entirely too much coffee...
15:00.36leifmadsenif I drank coffee, I'd get addicted to it
15:00.40*** part/#asterisk Techdeck (i=Techdeck@77.125.43.202)
15:00.43Qwelland?
15:00.53leifmadsenQwell: I don't need any more vices
15:00.54BlargMaN00i drink about 3-4 pots a day
15:00.58leifmadsenthat'd be me
15:01.01Qwellbut it's coffee
15:01.09leifmadsenand not good for your heart
15:01.13nkohhBlargMaN00: me too! one in the morning before i go to my office, then two at work. sometimes one before bed
15:01.17leifmadsenwhen you drink that much
15:01.26[TK]D-Fendergets very cranky when he gets too much blood in his caffeine stream
15:01.38nkohhyeah, that's why you've got to smoke cannabis. it offsets the jitter ;)
15:01.43leifmadsenI like being able to wake up and function without coffee
15:01.43BlargMaN00[TK]D-Fender: i'm right there with you...
15:01.49eppigycoffee has lots of antioxidants
15:01.54eppigyyou cannot drink too much
15:02.03leifmadsenyou can always have too much of anything
15:02.08eppigylies
15:02.13eppigyand deceit
15:02.21Qwellleifmadsen: I'm gonna help you with your addiction to water
15:02.28leifmadsenQwell: you can die from too much water!
15:02.36nkohhwater intoxication is serious business
15:02.38Qwellthat's why I'm going to help!
15:02.39leifmadsentotally
15:02.49stopejust like in the hold your wii contest a few years ago
15:02.56leifmadsenyep
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15:03.22nkohhwe had a guy die here. these people were supposed to not go to the bathroom for as long as possible and whoever won won a bedroom set or something
15:03.25nkohhthe last guy died
15:03.27nkohhso nobody got the set.
15:03.38nkohh100% true story.
15:03.39BlargMaN00that sucks
15:03.43eppigylol
15:03.48eppigyfor a damn bedroom set
15:03.50nkohhBlargMaN00: no, darwin at work
15:04.00leifmadsenlady died in a contest in N.A. too
15:04.07BlargMaN00or karma...  which ever you prefer...
15:04.11nkohheppigy: it probably had a tv or some such thing... it was quite a while back, I don't remember for sure. it was on local TV and radio and everything.. was a big spectacle...
15:05.00eppigybetter have been a 72'' money printing tv
15:05.07nkohhlol
15:06.00leifmadsenthat'd be sweet
15:07.10therealcircutwe had a similar incident here in chicago
15:07.19therealcircutwhere kids would drink shitloads of water
15:07.31therealcircutn effectively saturate their bodies / brains
15:09.38jjshoebusy day here I see ;)
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15:10.13BlargMaN00i love it when we all get so much work done...  8)~
15:10.47therealcircuti got a question for ya
15:10.49jayteechkdsk reports 53% complete but that nasty clicking noise is getting on my nerves
15:11.26therealcircutso on our asterisk server, we have an extension which effectively does: MusicOnHold(default), ParkAndAnnounce() x 4, then hangs up
15:11.55therealcircutit works, but the problem is when the next ParkAndAnnounce() call is hit, the music on hold, stops , then restarts
15:12.11therealcircutwere using asterisk 1.4.24.1
15:12.19BlargMaN00therealcircuit: that sounds like a pointless extension...
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15:12.53BlargMaN00therealcircuit: why would you park a call 4 times??
15:13.30therealcircutwell the calls come in to our recptionist, then she transfers to 5XX, where XX is the dest EXT
15:13.46therealcircutits here that they hit that ParkAndAnnounce() routine
15:14.11therealcircutand it runs 4 times because there are 4 people that it gets announced to
15:14.46therealcircutim fairly new to asterisk ;/ so i apologize.
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15:20.08therealcircut[TK]D-Fender: you the man
15:20.24therealcircutBlargMaN00: so i should Park() the call, then ring those groups?
15:21.01BlargMaN00therealcircuit: in theory (i have never tried it) you should be able to use one ParkAndAnnounce() to dial all four extensions...
15:21.21therealcircutBlargMaN00: yea that was the inital attempt
15:21.35therealcircutbut it didnt work, was complaing about resources being unknown
15:21.50therealcircutwhen i used: Local/1@office&Local/2@office....
15:22.11BlargMaN00try creating an extension that just dials those four extensions, and use that in the ParkAndAnnounce() app
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15:25.53[TK]D-Fendertherealcircut: How do you really want to announce it?
15:26.36therealcircutwell when the receptionist gets a call, the person says ' i want to talk to blah..' so she hits xfer, then that persons extension
15:26.55therealcircutthe thing is, it needs to announce it to 4 people, myself, the overhead speaker, and 2 other people
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15:27.30therealcircutammend that xfer thing, she hits transfer, then 5XX, where XX is the persons extension
15:28.29therealcircutso idealy, i would like the person to be put into a parking lot, then ring the people it needs to
15:28.42therealcircutand if the call isnt answered goto soem default vm
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15:30.18salzhhi, can i set sip calls in Asterisk to work in passthrough mode?
15:31.09[TK]D-Fendertherealcircut: if it hits a VM thats because thats what the exten you are transfering it to does.
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15:31.24[TK]D-Fendertherealcircut: Yuo need to make sure that you send it somewhere that will not do this.
15:31.55[TK]D-Fendertherealcircut: What you should do is call a local channel taht will PAGE them with auto-answer.
15:32.24[TK]D-Fendersalzh: "canreinvite=yes".  And only works where NAT is not involved.
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15:32.38[TK]D-Fendertherealcircut: "core show application page"
15:33.24[TK]D-Fendertherealcircut: And set the SIP header to have your Polycom's auto-answer on speakerphone.  You will clearly have to prepare this in your phone's provisioning.
15:34.23therealcircutyea, i have a paging macro setup
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15:47.28el_--hi... in which file do I have to place SipAddHeader
15:47.29el_--?
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15:47.54[TK]D-Fenderel_--: Extensions.conf
15:48.10[TK]D-Fenderel_--: Which of course... gets blown away by your GUI
15:48.43deadpigeoni know this is a voip channel, but out of curiousity anyone here familar with adtran TA1500 or perhaps a CACS Access Navigator GR303 concentrator?
15:52.04el_--ok lets assum I do not have the gui problem
15:52.31el_--i want to have the following for all extensions specified to SIP calls initiated using Dial(). Non-standard SIP headers should be preceded with an X- as in X-Asterisk-Accountcode:.
15:52.35el_--Should be used with caution as different SIP devices expect different headers and respond differently to them. May produce unexpected behavior.
15:52.38el_--Returns 0.
15:52.41el_--exten => 123,1,SIPAddHeader(X-Asterisk-Account: ${CDR(accountcode)})
15:52.43el_--exten => 123,n,ups
15:52.46el_--no
15:52.50el_--this: SIPAddHeader(Remote-Party-ID: <sip:${MYNUM}@ipadresse>\;party=calling\;screen=yes\;privacy=full)
15:53.14el_--how would I put that into extensions.conf... or do I have to add that for all extensions separately?
15:55.48andrebarbosaanyone know what asterisk version is used on switchvox 4.0?
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16:00.53ghentoHi all. I'm trying to wrap my head around the proper way to bridge two calls.  I am doing two outbound calls, and the goal is to bridge the calls to allow them to converse.  I know I can call out to one (callerA), and then use Dial() to call CallerB, however should I be using Transfer() instead to just connect the two calls and then not have asterisk involved? If I do the latter, would callerA be paying any long-di
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16:11.28ornI am trying to get the Diversion SIP header from a SIP message, but unable because Asterisk says it only applies to SIP channels. Call comes from PSTN, goes to SIP server, SIP server adds diversion, sends call back to PSTN
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16:11.48orneven though there is no active SIP channel, there still is a SIP message with the diversion header
16:12.14ornhttp://pastebin.com/m7594ed71
16:13.29ornThe reason I need it is that when there is more than one diversion, asterisk uses the oldest one to use as the RDNIS, instead of the newest one.
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16:16.12jerliqueWhy would * stop trying to regiser a sip registration entry. Its in state "No Authentication"
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17:58.57[TK]D-FenderLinuturk: And saying "same configs" doesn't mean they are GOOD for a 650
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18:02.35Linuturklol
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18:02.48Linuturkfirmware 1.5 vs 3 in current release
18:02.49Linuturkno wonder
18:03.50[TK]D-FenderLinuturk: SMRT
18:04.07[TK]D-Fendergoes to pout some more leaded gasoline into his VW Jetta TDI
18:04.11[TK]D-Fenderput*
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18:27.02leifmadsen[TK]D-Fender: lol
18:27.14leifmadsenI suggest diesel in a gas engine
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18:28.56empiricguys i have Dlink dvg 3004S FXO gateway i have make 4 sip users and configure in Dlink all 4 lines are registered
18:29.06empirichow i cal on PSTN line?
18:29.10empiricany idea?
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18:30.26seb-[TK]D-Fender: is a default * install secure?  if i just add sip stuff w/ a password is it secure? there isn't 100 other services open by default?
18:30.46bmoracapour + put = pout?
18:31.18seb-[TK]D-Fender: i tested appconference (MeetMe replacement)....i didn't get an error...it was quiet...perhaps you drop into a conf room and sit there until others arrive? are you familiar w/ appconf?
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18:33.17fun330hey i want to beable to display realtime call duration on a webpage what would be the best way to do that?
18:33.35fun330does anyone do that currently
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18:35.01[TK]D-Fenderseb-: nope
18:35.33[TK]D-Fenderseb-: disable channel drivers you don't need like H323, IAX2, MGCP, Skinny, etc
18:35.43seb-[TK]D-Fender: nope to which?
18:35.50seb-[TK]D-Fender: ah
18:35.57seb-[TK]D-Fender: how do that?
18:35.58[TK]D-Fenderseb-: Firewall AMI port.  That just about covers it
18:36.10[TK]D-Fenderseb-: and nope = never used app_conference
18:36.15seb-[TK]D-Fender: ok
18:36.30seb-[TK]D-Fender: how disable channel drivers?
18:37.00[TK]D-Fenderseb-: noload => chan_mgcp.so
18:37.19[TK]D-Fenderseb-: in modules.conf.  Repeat for other unneeded modules
18:37.29fun330why don't you want to load the channel drivers?
18:37.39fun330i only use sip should i disable the otheres?
18:37.47empiricguys any idea?
18:37.50seb-fun330: i think he thinks it is a security problem
18:37.57seb-fun330: i only use sip too
18:38.27seb-fun330: [TK]D-Fender should know...he lives this stuff
18:38.42fun330okay
18:38.46fun330even with a firewall?
18:39.01seb-fun330: ask him
18:40.44fun330TK D fender: even if i have a firewall should i disable those channel drivers?
18:42.19*** join/#asterisk paulius (n=paulius@unaffiliated/paulius)
18:42.33pauliusWhat's the best POTS to SIP converter out there?
18:43.04NuggetThe one where you cancel your analog lines and buy a PRI.
18:43.14pauliusFunny.
18:43.22pauliusI actually have that, but I'll have a second POTS line.
18:43.52henki have an account and one phone number from a sip provider. is it possible to 'expand' that number? so it's not just 12345, but also 12345-1 and 12345-2 with different extensions?
18:44.19pauliushenk: Really doubt it. What you're looking for are extensions.
18:44.26BlargMaN00paulius: what are you trying to accomplish??  just a POTS to SIP gateway??  or what?
18:44.31[TK]D-Fenderfun330: If someone breaks into your system with a weak account they can then attack * locally
18:44.44[TK]D-Fenderfun330: Security is a PROCESS, not a "solution"
18:44.48pauliusBlargMaN00: Just a way to make and receive POTS calls on my asterisk gateway.
18:44.50seb-[TK]D-Fender: dude! modules.conf doesn't load jack by default..the only non-comment, non-noloads
18:45.00seb-[TK]D-Fender: are [modules]
18:45.00seb-autoload=yes
18:45.00seb-load => res_musiconhold.so
18:45.06seb-[TK]D-Fender: that's pretty skimpy
18:45.07[TK]D-FenderAUTOLOAD <-
18:45.16seb-[TK]D-Fender: is autoload bad?
18:45.17pauliusNugget: You know these cheap unlocked Linksys SIP gateways that they have on ebay... That isn't what I'm looking for is it? Those are just to implement SIP on local POTS?
18:45.26BlargMaN00paulius: your best bet, is prolly gonna be to put an analog card into your * box...  how many POTS lines are we talkin??
18:45.30[TK]D-Fenderseb-: so selective EXCLUDE things you know you don't want
18:45.34pauliusBlargMaN00: One.
18:45.44pauliusBlargMaN00: Home setup, nothing fancy.
18:45.47seb-[TK]D-Fender: not sure what that means
18:45.50Nuggetyeah, those linksys pap boxes are a good solution.
18:45.51jerliqueWhy would * stop trying to regiser a sip registration entry. Its in state "No Authentication"
18:45.55henkpaulius: what do you mean with "i'm looking for extensions"?
18:46.10seb-[TK]D-Fender: can i turn off autoload? will i then need to load => sip.so explicitly?
18:46.12BlargMaN00paulius: gotcha...  hold on a sec...  lemme find a link...
18:46.14[TK]D-Fender[14:36]<[TK]D-Fender>seb-: noload => chan_mgcp.so <-- I gave you a sample.
18:46.17NuggetI would advise against a card, an external FXO is going to be a lot less hassle
18:47.04seb-[TK]D-Fender: oh i think i misunderstood
18:47.06pauliusNugget: Well actually I can't even have a card. Plan is to run this on a Mac mini
18:47.16seb-[TK]D-Fender: if you don't specify a module in modules.conf it still gets loaded
18:47.29seb-[TK]D-Fender: you need to explicitly do a noload on what you don't use
18:47.53seb-[TK]D-Fender: is there a way to invert that so you don't load anything by default? is that what autoload=no would do?
18:48.52pauliusSo guys, you're saying that something like this should work: http://cgi.ebay.ca/Linksys-PAP2-NA-2-ports-Sip-Voip-gateway-ATA-UNLOCKED_W0QQitemZ370197821545QQcmdZViewItemQQptZAU_Mobile_Phones?hash=item370197821545&_trksid=p3286.c0.m14&_trkparms=72%3A1215%7C66%3A2%7C65%3A12%7C39%3A1%7C240%3A1318%7C301%3A0%7C293%3A1%7C294%3A50
18:48.57paulius(sorry for long link)
18:48.58[TK]D-Fenderseb-: Based on whats in the rest of you files... load NOTHING.
18:49.21[TK]D-Fenderpaulius: That is for 2 PHONES, not LINES.  Do not mistake this
18:49.26pauliusAh gotcha.
18:49.32Nuggetno, the PAP2 is an FXS device.  It pretends to be a POTS line so you can plug analog devices in to your asterisk server.
18:49.35Nuggetyou need an FXO device
18:49.42pauliusNugget: Gotcha.
18:49.51pauliusBut do they have any sort of 2 in 1 devices?
18:49.56Nuggetyes
18:50.05pauliusMy long-term plan is to obviously have the POTS phones go through the gateway.
18:50.08[TK]D-Fenderpaulius: Linksys SPA-3102 = 1 FXS, 1 FXO
18:50.12pauliusAny model recommendations?
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18:50.29Nuggethttp://www.ipphone-warehouse.com/ProductDetails.asp?ProductCode=spa3102
18:50.36pauliusah thanks
18:50.46pauliusI'm in Canada though, so it'll be finding one here.
18:51.12[TK]D-Fenderpaulius: Describe your expected connectivity requirements
18:51.21pauliusMeaning?
18:51.35[TK]D-Fenderpaulius: How many lines, how many analog phones, etc
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18:51.49pauliusOne line, have 2 phones on the POTS network here.
18:52.07pauliusWill have one vo-ip Cisco IP phone.
18:52.21pauliusI want the asterisk to connect to the POTS and also another SIP provider.
18:52.35pauliusI'll want the analog phones to use the local POTS and be able to chose on the IP Phone
18:52.54[TK]D-Fenderpaulius:  jsut FYI you won't really be able to use call-waiting on that line, and telco VM will kind suck for not having an indicator of use
18:53.15paulius[TK]D-Fender: I don't have call waiting on my pots line anyways
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18:53.32[TK]D-Fenderpaulius: thent he SPA should do fine
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18:56.52paulius[TK]D-Fender: Do you know any good places to buy it from?
18:57.14[TK]D-Fenderpaulius: http://www.canadianvoipstore.com/home.php
18:57.22pauliusheh wow
18:57.28pauliusAnd how reliable is that adapter?
18:57.43[TK]D-Fenderpaulius: http://www.voipdepot.ca/
18:57.45pauliusYou have to realize that I'm a guy who hates cheap consumer hardware. I'm running a proper Cisco router and switch at home.
18:57.51[TK]D-Fenderpaulius: Its OK normally
18:58.02pauliusWhat does normally mean?
18:58.10[TK]D-Fenderpaulius: And unless you already own that Cisco IP phone.... DON'T
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18:58.25pauliusI'm a sucker for Cisco devices.
18:58.35pauliusOriginal plan was to use CUCM with it.
18:58.36[TK]D-Fenderpaulius: Means if your line is especially shitty you might have quality issues on top from EC freak-out
18:58.56Nuggetthe cisco phones look sexy and are sure to impress geek visitors.
18:58.58Nuggetbut they suck to use.
18:59.25pauliusWell I'll see about that.
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18:59.34pauliusSuck and Cisco are words that I've never seen in the same sentence.
18:59.35Nuggetin excrutiating detail
18:59.48beekpaulius: Get over your Cisco fetish and buy a Polycom if you want a great VoIP phone.
18:59.49pauliusAnd no, I'm not talking about their consumer Linksys stuff
18:59.54NuggetI have 30 cisco phones.  Take my advice -- buy a polycom.
19:00.18Nuggetcisco phones are really only practical if you're running callmanager and pay for support
19:00.24KyleKhehehe "nobody ever got fired for choosing cisco"
19:00.35[TK]D-Fenderpaulius: their Linksys VoIP gear is BETTER <-
19:00.38Nuggettrying to make them talk to asterisk is a sisyphean task
19:00.41[TK]D-Fenderpaulius: Polycom > All
19:01.07QwellKyleK: really?  because I know a guy...
19:01.12[TK]D-Fenderlol
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19:10.22cp5hey guys, ulaw is the best format to use (least translation cost) for calls over a US PRI, right?
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19:10.50[TK]D-Fendercp5: Yes
19:11.07cp5[TK]D-Fender: cool, i assume it's really determined by the channel driver you're using, right?
19:11.17jayteeso I push the Cisco boulder up the hill but just as I get to the top I always lose my footing and the boulder rolls back down. At least I ain't chained to a Grandstream rock while vultures pick out my liver.
19:11.24cp5in which case is slin or slin16 ever used? just as an intermediate?
19:12.14[TK]D-Fendercp5: Every channel has its own parameters.
19:13.01cp5i see, cool
19:15.47Kattysneaks a hug to jaytee :>
19:16.17Kattyhugs on eppigy too :>
19:16.45*** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw)
19:17.33eppigy:D
19:17.40sHoZaIbhello there, when I call out from my sip device through asterisk, call gets establish and also got voice response from other end but for few seconds
19:17.42eppigyherro
19:18.06cp5[TK]D-Fender: do you see any reason to use the .sln16 audio files or have them around? are they sometimes used as an intermediary for transcoding?
19:18.59eppigyKatty: how are you? :]
19:20.04pauliusYeah you guys are gonna kill me but I'm about to buy another Cisco IP Phone
19:20.05pauliushiges
19:20.07paulius*hides
19:21.57beekWhatever floats your boat paulius.   You have a knack for ignoring advice from individuals who have already been there, done that.
19:22.30beekI guess that's one of the joys of being a teenager.
19:22.30KyleKdoes chan_sip.so not load if i screw up sip.conf?
19:22.56*** join/#asterisk hfb (n=hfb@pool-96-247-49-46.lsanca.dsl-w.verizon.net)
19:23.23KyleKbeek: if hes got a stack of them one or two more doesn't make much of a difference
19:24.17beekKyleK: It was Nugget who had the stack of them.
19:24.46[TK]D-Fendercp5: Not worth thinking about
19:24.49sHoZaIbhello there, when I call out from my sip device through asterisk, call gets establish and also got voice response from other end but for few seconds after that no voice from both side and call remain connect
19:25.06[TK]D-Fenderpaulius: just sad...
19:25.51[TK]D-FenderKyleK: Indeed, screw up your SIP.CONF bad enough and it won't load at all
19:26.37cp5[TK]D-Fender: alrighty, thanks
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19:50.47pauliusWhat was that phone brand that you were so fond of, [TK]D-Fender?
19:51.00[TK]D-FenderPolycom
19:51.07pauliusAny particular model?
19:51.30[TK]D-Fenderpaulius: All are good, which one I would suggest depends on the needs
19:52.56*** join/#asterisk mmlj4 (n=jkelly@ip70-171-94-246.no.no.cox.net)
19:53.24Anth8708paulius:  we're using the soundpoint 330s for analog replacements and the 560s to replace nortel 2616s
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19:54.34[TK]D-Fender560?  Who needs GB phones :)
19:56.01mmlj4all things being equal, should I answer a call and fork off to an AGI process, or handle them via extensions.conf only? is or can AGI be robust enough to handle a call from beginning to end, as opposed to native *?
19:56.23pauliusHmm.
19:56.30pauliusWell but doesn't Polycom make Cisco's phones?
19:57.50[TK]D-Fenderpaulius: Only their speakerphones.
19:58.10pauliusWell but they're much more expensive than the Cisco phones though
19:58.15[TK]D-Fenderpaulius: And its not the HARDWARE that sucks, its Cisco fIRMWARE and licensing BS
19:58.25paulius[TK]D-Fender: Yeah I know that.
19:58.31pauliusI don't agree with their corporate policies either.
19:58.39pauliusBut I like the fact that their hardware works.
19:58.52[TK]D-Fenderpaulius: Well their SIP stack sucks.  Licensing sucks, etc
19:59.17[TK]D-Fenderpaulius: yes, well lof of hardware works.  theirs jsut takes a complete 2nd rate approach to SIP
19:59.48[TK]D-Fenderpaulius: standards compliance isn't compatible with "vendor lock-in"
20:01.12Nuggetthe cisco sip firmware is neglected and poorly maintained
20:01.29Nuggeteach new release is just as likely to introduce new bugs as it is to clear up old ones
20:01.32mmlj4Anth8708: you do know that asterisk can talk to those nortel phones via chan_unistim or somesuch, right?
20:01.32pauliusIt still works, doesn't it.
20:01.41Nuggetbarely
20:01.48pauliusgood enough!
20:02.06Nuggetplus you need a support agreement that nobody will actually sell you in order to get access to the firmware
20:02.23Anth8708mmlj4: that's what I've been told (the voip phones, like i2004), but I've tried and not been able to get it functioning
20:02.42mmlj4it worked for me, first time out, using * 1.6
20:02.49Anth8708mmlj4: got registered and can receive and make calls from an i2004, but no audio
20:03.02Nuggetand, to top if off, the sip firmware is effectively undocumented, so you'll have a really fun time figuring out how to configure them.
20:03.29KyleKis that so people become cisco voip phone certified?
20:04.20Nuggetno, it's because cisco don't give a flying muffin about people who don't buy callmanager.
20:04.30Anth8708[TK]D-Fender: who needs GB phones?  me:).  We have a bare minimum wiring infrastructure, so as I upgrade phones to voip, I'm passing lan through the phones, just splitting out the traffic over vlans
20:04.35*** join/#asterisk Ast001 (n=uros@cable-89-216-155-28.dynamic.sbb.rs)
20:05.02Ast001Hello I've found following error in my /var/log/asterisk/messages
20:05.03Ast001Failed to open /dev/zap/transcode: No such file or directory
20:05.17Nuggeteverything that the community knows about configuring cisco phones for asterisk is simply by virtue of trying to deconstruct/trace from a callmanager installation
20:05.24Ast001and there is /dev/zap on system I wonder what is that error connected too ?
20:06.07KyleKAst001: so /dev/zap is a device? that means /dev/zap isn't a dir for a /dev/zap/transcode file :-/
20:06.12[TK]D-FenderAnth8708: that addas 60$ per phone assuming you even wanted the 5XX series base.  You saying a cable drop would cost you more?  Keeping in mind you're also losing flexibility as well
20:06.22Ast001no I meant /dev/zap/transcode
20:06.29Ast001I have dir /dev/zap
20:06.55Ast001lsmod | grep zaptel gave me only wctdm which I need for my digium card. Do I need zttranscode ?
20:07.01*** join/#asterisk manxpower (n=Administ@router.asteriasgi.com)
20:07.24Ast001Should I manualy load zttranscode to avoid the problem ?
20:08.01KyleKi personally have no clue, give it a shot?
20:08.07*** join/#asterisk BreezBl0k (n=BreezBl0@5acd71a1.bb.sky.com)
20:08.28manxpowerzttranscode is only for use with the Digium transcoder card, IIRC
20:08.57BreezBl0kany one got a asterisk box behind pfsense?
20:09.11NuggetBreezBl0k: I do, but not behind nat if that's where you were headed.
20:09.28BreezBl0kafraid so Nugget :(
20:09.33Nuggetsorry  :)
20:09.40*** join/#asterisk neurosys (n=vinix@sheltercorp.net)
20:09.42BreezBl0kthanks any way
20:10.39mmlj4Ast001: if you don't HAVE a transcode device, you don't need to load the module for it
20:10.43*** part/#asterisk SparFux (n=raoul@e182023188.adsl.alicedsl.de)
20:10.54manxpowerforward port 5060/UDP and 10000/UDP - 20000/UDP (or whatever the range is in rtp.conf) on your NAT router.  Then set localnet= and externip= in sip.conf.  Presto!  Done!
20:10.56Ast001ok
20:11.23Ast001but why is this happening ? codec_dahdi.c: Failed to open /dev/zap/transcode: No such file or directory
20:11.42Ast001I can not record calls using digium licenced g729 codec
20:11.58BreezBl0kNugget any guide you used to configure pfsense without NAT for asterisk ive got a couple IP's free but idealy id like to set it up behind NAT but having no luck
20:12.01Ast001I bought 2 licences.
20:12.27manxpowerAst001: How many g729 channels does Asterisk show that you have available.
20:12.27QwellAst001: codec_dahdi isn't for the g729 module
20:12.48Ast001show g729 ?
20:12.53manxpowerrecording will take up 1 license,
20:12.59manxpowerAst001: depends on your asterisk version.
20:13.04Ast0011/0 encoders/decoders of 2 licensed channels are currently in use
20:13.25Ast001Asterisk 1.4.21.2
20:13.30QwellAst001: ignore that error unless you have transcoding *hardware*.
20:13.47Ast001ok but I can't ignore that I can't record calls I need that
20:14.06Ast001Recording is working fine from LAN with alaw but from WAN (g729 is not)
20:14.19*** join/#asterisk UQlev (n=yuriy@91.184.221.31)
20:14.19mmlj4hey manxpower
20:14.55Anth8708[TK]D-Fender: it's $50/phone (academic pricing:)) and that's definitely less than a new drop to the majority of my locations because of construction (cinder block, so everything is surface mount, which means changing raceway since the idiot before me used junk), not to mention new patch panels + labor (I don't have time to do it)
20:15.28[TK]D-FenderAnth8708: Question of balance and long-term value
20:15.29manxpowerAst001: I don't see any error messages
20:15.37Ast001[May  6 17:32:14] WARNING[7682] codec_g729a.c: out of G.729 decoder licenses
20:15.46Ast001how is that possible ?
20:15.48[TK]D-FenderAnth8708: And of course... Gigabit
20:15.50mmlj4manxpower: done anything with AGI?
20:16.00manxpowerAst001: and right after that message show your g729 channels and see what it says
20:16.03manxpowermmlj4: yes
20:16.26Anth8708[TK]D-Fender: I know.  Having gig to the desktop just feels like future proofing for the execs.  I will likely only have 10-15 of those and 250 of the 330s.
20:16.33Ast001[May  6 17:32:14] WARNING[7682] translate.c: g729tolin did not update samples 0
20:16.50Ast001then repeat and etc.... milions of times
20:17.07Anth8708quick question guys, i'm googling this and checking wikis, but
20:17.08manxpowerAst001: that doesn't look much like the number of encoders/decoders you have in use.
20:17.35Anth8708since we have activity right now, does * 1.6 support either shared line appearances or bridged line appearances?
20:17.48mmlj4manxpower: ok.. in your opinion, can AGI be robust enough to handle a call by itself, as opposed to straight extensions.conf routing?
20:17.50Anth8708just a yes or no will suffice, i can find the documentation eventually
20:17.53[TK]D-FenderAnth8708: No.
20:17.57manxpowermmlj4: yes
20:18.09mmlj4cool
20:18.18Ast001I dunno what's happening it said 1/0 encoders/decoders of 2 licensed channels are currently in use
20:18.31manxpowerI recommend FastAGI rather than AGI
20:18.42Anth8708[TK]D-Fender: recommendations on how to handle secretary situations?  They need to be able to answer the exec's line and see if they are on the phone.
20:18.54mmlj4I'm looking at perl's Asterisk::AGI
20:19.14manxpowerAnth8708: You can do that.
20:19.21[TK]D-FenderAnth8708: Seeing if they are on the phone is basic Presence and your 560 can do that.  Answering calls going to them can be managed differenly
20:19.31Ast001I also have this http://pastebin.com/m12b6ea7e
20:19.44[TK]D-FenderAnth8708: You can either setup another reg that will be rung at the same time, or use a call-pickup feature
20:20.03Ast001you said I need 2 g729 licences for 1 concurent call which is recorded and I bought second licence
20:20.53Anth8708[TK]D-Fender: gotcha.  I'll look up presence, thanks.
20:22.09Anth8708one more question and then i'll try to just answer.  anyone successfully using softkeys on polycom phones?  I've been trying to figure it out and it looks simple in the xml, but I'm stuck.  I never get anything but the default keys, even though I do have preceeding='1' and my custom xml loads before ( to the left) of the standard sip.conf
20:22.39manxpowerAst001: you should contact Digium then.  You bought the license, you get support for it.
20:22.50Ast001I made a ticket
20:22.57manxpowerAst001: then why are you here?
20:23.00*** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun)
20:23.08*** join/#asterisk juanIMP (n=Juancho@200.71.41.22)
20:23.19[TK]D-FenderAnth8708: what are you trying to do?
20:23.20Ast001They need few days to answer and I need system to work as soon as possible
20:23.40manxpowerAst001: best of luck with that.
20:25.03Anth8708[TK]D-Fender: The 560 doesn't show the "callers" key like the 330 does and I'd like it to.  I'd also like the 560 to show the Dir on the idle screen.  Also, when you have a VM, I'd like to have a key to call the vm number.
20:25.50Anth8708[TK]D-Fender: I think I understand the xml decently enough, it's fairly plain, but all I can get are what appear to be the "default" softkeys
20:25.53[TK]D-FenderAnth8708: No need, thats what the arrow keys are for.  IP 330 "Callers" key is vastly inferior <-
20:26.08*** join/#asterisk justdave (n=dave@unaffiliated/justdave)
20:26.18Anth8708[TK]D-Fender: wow.  didn't even know that works
20:26.21Anth8708*worked
20:26.43Anth8708[TK]D-Fender: Thanks.
20:26.54[TK]D-Fenderup= dir, Left = answered, down = missed, Right = dialed
20:28.10Nuggetup, down, left, up, power = punch caller in the face
20:28.25*** join/#asterisk jpcansa (n=jpbenavi@201.201.20.90)
20:29.02[TK]D-FenderFUDOKEN!!!!!
20:29.07[TK]D-Fendercheckout time, later all
20:29.31Nuggetcheers
20:29.42jpcansadoes any body knows what the "requested special control" line means here: http://pastebin.com/
20:30.10manxpowerjpcansa: try giving us the correct URL
20:30.22*** join/#asterisk thehar (i=thehar@thehar.xmission.com)
20:30.25theharrussellb: ?
20:30.26jpcansahttp://pastebin.com/d4c1f50dc
20:30.28jpcansasorry
20:31.03jpcansa8th line
20:31.34*** part/#asterisk manxpower (n=Administ@router.asteriasgi.com)
20:32.08pmhaddad-workanyone know an easy way to raise outbound call priority?
20:32.27pmhaddad-worklike i want 911 calls to be sent out at a higher priority level than anything else
20:33.05pmhaddad-workshould i make the 911 stuff priorty 1 and everything else a higher number?
20:34.51stopeNugget - I like the 'punch caller in the face' part....  Nugget++
20:35.09*** join/#asterisk Led-Hed (n=Led-Hed@66-189-167-116.dhcp.trlk.ca.charter.com)
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20:41.31Led-HedDo IP Phones need to be MultiLine to transfer calls to other IP Phones, or can Asterisk handle that?
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20:42.39*** part/#asterisk mclugh (n=mpearson@67.214.244.42)
20:43.47Nuggetmultiline not necessary
20:43.50pmhaddad-workLed-Hed, no, asterisk handles that
20:44.27Led-Hedpmhaddad-work, ok so what benefit does a multiline IP phone have
20:44.49kb3ienwhats the minimum connection time for a fax of 1 page. i'd like to consider any calls shorter than that as broken?
20:44.56Led-Hedsorry, I'm new to asterisk and VIOP in general
20:44.59pmhaddad-workyou can be talking to multiple exts concurrently basically
20:45.18*** join/#asterisk macros73 (n=cs_@dsl093-063-232.pit1.dsl.speakeasy.net)
20:45.28pmhaddad-worki mean ya, if you wanted to be on the phone using it on a call and take another call and transfer that to another ext then you need a multiline phone
20:45.41Led-Hedpmhaddad-work,  what do I need multiline Phone to put someone on hold?
20:46.03pmhaddad-workonly if you want to be on another line talking to someone else at the same time
20:46.19*** join/#asterisk jeffgus (n=jeffgus@green.zimage.com)
20:46.23Led-Hedok, I need multiLine then .  Thanks
20:46.28pmhaddad-workk :)
20:49.57*** join/#asterisk _Sam-- (n=sam@unaffiliated/sam--/x-573746)
20:50.13_Sam--hey what happens if when a caller who is blacklisted this way calls?   database put blacklist 9408989740 1
20:50.24_Sam--they just get a busy signal, or what happens?
20:50.51jayteenothing, unless you build a lookup to the blacklist in the incoming call handling
20:51.03_Sam--i see.
20:51.14jayteethere's some stuff on the WIKI about it
20:51.35_Sam--do those database puts stay after restarting asterisk?
20:51.50jayteeyes, they should
20:51.50Led-Hedok, just to clarify,  If while talking on the phone, I receive another incoming call and I want to place the currnet call on hold and answer the incoming call I need a MultiLine Phone.  (Sorry, just dont want to buy the wrong phones)
20:52.03jayteeLed-Hed, yep
20:52.08Led-Hedperfect.  Thanks
20:52.24Led-Hedany phone Recommendations?
20:52.32pmhaddad-workLed-Hed, polycom
20:52.57pmhaddad-work320, 330, and up should all do what you need
20:53.12pauliusCisco IP Phones, lol
20:53.15pmhaddad-worklol
20:53.24_Sam--thanks, J.
20:53.25Led-Hedwhy the LOL?
20:53.33Led-Hedare Cisco Phones crap?
20:53.35pauliusBecause people here hate Cisco.
20:53.38pmhaddad-workbecause its a major PITA with asterisk
20:53.39pmhaddad-workand that
20:53.40Led-Hedahh
20:53.41pauliusBecause Cisco is proprietary.
20:53.49pauliusMeh, I'm about to use a Cisco phone with Asterisk.
20:53.53pauliusI like Cisco's hardware.
20:54.07Led-HedI dont like them either.  And from what I saw, the Phones are built by Linksys.
20:54.41pauliusCisco bought Linksys and the Linksys branding is their consumer products.
20:54.46*** join/#asterisk delta_16 (n=delta_16@84.26.9.136)
20:54.56delta_16hey guy's
20:55.04delta_16go a couple of questions
20:55.05Qwellhi delta_16 is.
20:55.24beek"Meh, I'm about to use a Cisco phone with Asterisk." -- let us know how that works out for you paulius
20:55.32Corydon76-digOdd, I had a defense contractor INSIST on Cisco phones.  Six months in, they replaced almost every phone with Polycom
20:55.36pauliusbeek: Hehe I will.
20:55.46beekJust don't slit your wrists.
20:55.48delta_16im looking for a way to get german phone numbers ... any idea how i can do that ?
20:56.22Led-Hedwhen the phones have 2 Ethernet Ports, do you connect them In-Line between the wall and the PC?
20:56.30beekLed-Hed: Cisco phones are just fine -- if you're using Cisco call manager.    If you're using Asterisk they're a royal PITA.  Just get Polycom.
20:56.36beekLed-Hed: ys
20:56.38beekyes
20:56.45Corydon76-digpaulius: the only phones they didn't replace where executives, where looks were more important than functionality
20:56.45_Sam--jaytee :  basically all i need to do is add the lookupblacklist function to my dialplan?
20:57.11jaytee_Sam--, basically
20:57.15Led-Hedand is it reliable to run them "In-Line"  or is it likely to cause networking issues?
20:57.59_Sam--thanks, again.
20:58.12delta_16i have a registerd adress in germany but i would like to get a phonenumber.
20:58.20*** join/#asterisk xpot (n=james@70.91.210.233)
20:58.22beekLed-Hed: they give priority to the voice packets.   They'll be fine, but they're 100M (unless you get the newer 1G models)
20:58.46delta_16my phone line in germany is not activate couse i would like to you my asterisk server
20:58.48Led-Hedbeek, only need 100M
20:58.50Led-Hedthanks
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20:59.44jayteeLed-Hed, I have a Polycom 330 on my desk, it has 2 lines and both are setup as a single "extension" so if someone dials my number Line 1 rings, if I'm on Line 1 and someone else dials my number Line 2 rings and I can say "Please hold" to the person on Line1 and then simply press Line2 and the phone will put Line 1 on hold automatically.
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21:00.18Led-Hedjaytee, perfect.  Thats exactly what I'm looking for
21:00.56Led-Hedjaytee, and how would you transfer a call to another person/phone in the office.  (From Reception to Sales)?
21:00.57jayteeprogramming the phone that way is an option, it can also be programmed to be 2 different lines.
21:01.30Led-Hedjaytee, with 2 different PHone Numbers?
21:02.13jayteeLed-Hed, when on a call there are 3 "softkeys" on the display, one of them is End Call, the next is Transfer and the last Conf.
21:02.32jayteeand yes the phone can be setup to handle TWO seperate numbers.
21:02.50[TK]D-Fenderjaytee: ... kinda lazy answer :)
21:03.14[TK]D-Fenderjaytee: 2 REGISTRATIONS.  "Numbers" is kinda generic
21:03.30Led-Hedare there any steps I should take when configuring my router for VOIP/Asterisk?
21:03.46KyleKforward some ports for sip and rtp?
21:03.51[TK]D-Fenderjaytee: What you look like dialing out direct from the phone, then through * are 2 different matters.. and inbound... well.. we know that story too :)
21:03.57jaytee[TK]D-Fender, that's much more specific, thanks! and where were you two minutes ago? :-)
21:04.05*** part/#asterisk joseph__ (i=CK@93.185.225.225)
21:04.07[TK]D-Fenderjaytee: In transit home :)
21:04.13[TK]D-Fender~sipnat
21:04.14infobot[~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions
21:04.19[TK]D-FenderLed-Hed: ^^^^^^^^^
21:04.43Led-Hed[TK]D-Fender, thanks
21:04.48bmoracaLed-Hed: depends on which router you're talking about and which direction the phone connections are going and what kind of router you have.
21:05.06bmoracaLed-Hed: the question you asked is kinda like "is there anything I should watch out for when driving?"
21:05.06jayteeok, it's quittin time for me too, be back later
21:05.17Led-Hedbmoraca, pfSense or Smoothwall
21:05.27Led-Hed(FreeBSD or Linux)
21:06.01Led-Hedbmoraca, ya I know it was kinda an open ended question
21:06.09bmoracauhg.  i'll refrain in saying my opinion of PC-based routers right now.  not in the mood for that right now.
21:06.09jpcansadoes any body knows what the "requested special control" line means here: http://pastebin.com/d4c1f50dc
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21:11.16bmoracadoes anyone have experience using Adtran TA900 series IADs with Asterisk?
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21:19.04mmlj4I want to place a call, and instead of getting hung up on my end when the remote end of the call terminates, be given a dialtone as if I'd hung up and started to place another call... how, please?
21:21.09[TK]D-Fendermmlj4: "core show application dial" <-
21:22.35mmlj4danke
21:23.11Led-HedI'm looking for VOIP Providers, a lot of them offer "Auto Attendant" features, will this interfere with Asterisk or can they typicaly be turned off?
21:24.45[TK]D-FenderLed-Hed: that implies they are offering you hosted-pbx plans
21:25.09[TK]D-FenderLed-Hed: And it depends what you buy and who you buy it from.  You want simple termination & origination
21:25.28Led-Hed[TK]D-Fender, ok, maybe I'm looking for the wrong service.
21:26.09Led-Hed[TK]D-Fender, any recommendations for VOIP Providers?
21:26.10[TK]D-FenderLed-Hed: Evidently
21:26.16[TK]D-Fender~itsplist-us
21:26.17infobot[~itsplist-us] Here are some popular ITSPs (USA) starting with the more respected ones : http://www.teliax.com , http://connect.voicepulse.com , http://www.nufone.net , http://www.broadvoice.com, http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net
21:26.28Led-Hedgreat!  thanks
21:26.57pmhaddadnufone is dead i thought
21:27.12pmhaddadbandwidth.com is good, i use broadvoice a lot too
21:27.55_Sam--im having a problem with this line thinking every call is in the database blacklist.... any help appreciated.    exten => 877294XXXX,1,GotoIf($[{DB_EXISTS(blacklist/${CALLERID(num)})]?blacklisted,s,1)
21:38.40[TK]D-Fender_Sam--: Count your braces
21:38.48_Sam--i added one, but it still didnt fix it.
21:38.53_Sam--right before the ]
21:39.02[TK]D-Fender_Sam--: And the only thing I see is a clearly broken line
21:39.05*** join/#asterisk ber_ (i=brad@66.94.69.34)
21:39.20_Sam--[TK]D-Fender:  im not surprised.  that is why im here asking for expert help!
21:39.20*** part/#asterisk Cresl1n_ (n=matt@asterisk/libpri-and-libss7-expert/Cresl1n)
21:39.21[TK]D-Fender_Sam--: IO more ways than one.
21:39.23[TK]D-FenderIn*
21:39.37ber_hi, I am trying to run an IVR after the calling party has hung up for the called party to complete
21:39.39[TK]D-Fender_Sam--: Also missing "_"
21:39.44ber_i see this command in the dial g: When the called party hangs up, exit to execute more commands in the current context.
21:39.49generalhananyone in here use Queuemetrics ? I seem to be having issues with the real-time data, and i cant find anything about what steps i need to take to make sure that i get agent logon/logoff information -- anyone have reference material they can direct me to ?
21:39.50ber_i want the exact opposite
21:39.54_ShrikEAnyone here using asterisk with virtual iron?
21:40.03ber_does anyone know if that is possible without running a conference?
21:41.11BreezBl0kany SIP behind NAT gurus here? im having one way audio problems with Elastix but with that same configuration Trixbox works but i desperatly dont want to use use Trixbox you can see my sip.conf here: http://pastebin.com/d44dbca66
21:42.27jameswftook 45min for my free kfc :(
21:42.28_Sam--[TK]D-Fender :  would this work:   GotoIf($[{DB_EXISTS(blacklist/${CALLERID(num)})} = 1 ]
21:42.47pmhaddadjameswf, oooh i should get that for dinner tonight
21:42.55_Sam--er...sorry bad formatting, again.
21:43.26[TK]D-Fender_Sam--: Doesn't need to be an expression
21:44.07*** join/#asterisk BadHAL (n=nn@66.194.174.11)
21:44.14_Sam--k.  trying to figure out where the _ would go...i only put XXXX to block the number for IRC
21:44.16jameswfthis is an interesting question http://trixbox.org/forums/trixbox-forums/help/play-beep-every-15-seconds-while-call-progress
21:44.47KyleKBreezBl0k: so the ports specified in rtp.conf are forwarded to the asterisk box?
21:44.52jameswfmaybe use a whisper page...
21:44.56pmhaddadjameswf, i had an issue very similar a while ago, was my fxo card
21:45.11pmhaddadoh
21:45.15pmhaddadi totally misread that
21:45.49BreezBl0k<KyleK> works fine for trixbox but if i setup Elastix with same settings oneway audio
21:46.18BreezBl0k<KyleK> yeh RTP ports and SIP port forwarded
21:49.05_Sam--[TK]D-Fender :  thanks for your patience.  i got it.
21:49.05_Sam--thanks again.
21:50.11*** join/#asterisk putnopvut (n=putnopvu@asterisk/master-of-queues/mmichelson)
21:50.11*** mode/#asterisk [+o putnopvut] by ChanServ
21:51.11[TK]D-Fenderjameswf: "core show application dial" <---
21:51.23KyleKBreezBl0k: the RTP stuff works by saying, hey send data to 1.1.1.1 port 1 so check the sip packets on the side that gets no audio?
21:52.04KyleKBreezBl0k: also I turned on canreinvite=no
21:52.19[TK]D-Fender^^^^^^^
21:52.25[TK]D-FenderCanreinvite is the problem
21:52.38jameswfgoogle found it Asterisk+cmd+Monitor ... putting it in trixbox another story..
21:52.45*** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee)
21:53.02BreezBl0k2secs
21:53.21ber_does anyone know how to keep a called party on the line and have them navigate an IVR once the calling party hangs up?
21:54.24BreezBl0k:( canreinvite=no has not fixed it
21:54.49KyleKcheck the conf for other canreinvite lines?
21:56.10*** join/#asterisk DarkRift (n=dark@65.92.166.246)
21:56.30[TK]D-FenderBreezBl0k: Confirm that you are directed to the proper internal IP, and repeat EXACTLY which ports are forwarded
21:57.25BreezBl0k<[TK]D-Fender> i made it the same IP as the trixbox that was working and switched the trixbox off
21:58.03BreezBl0k<[TK]D-Fender> so the port forwards were deffinetly working before and ive checked them a few times
21:58.22BreezBl0ki think im doooomed :(
21:58.23[TK]D-FenderBreezBl0k: pastebin CLI output for a failed call with SIP DEBUG enabled
21:58.34BreezBl0krgr
21:59.21[TK]D-FenderBreezBl0k: Also, any ITSP peers should have "nat=no" <-
21:59.34[TK]D-FenderBreezBl0k: that usually kills things as well
22:00.10BreezBl0k<[TK]D-Fender> ITSP would sipgate for example?
22:00.49[TK]D-FenderBreezBl0k: Yes
22:03.09*** join/#asterisk lowlevel (n=Stuart@lowlevel.ca)
22:03.56BreezBl0k<[TK]D-Fender> http://pastebin.com/d73f20b9b debug output here
22:04.58[TK]D-FenderBreezBl0k: I do not see a CALL ATTEMPT anywhere in there
22:05.37*** join/#asterisk colinm_ (n=colinm@97-124-108-180.phnx.qwest.net)
22:08.40BreezBl0k<[TK]D-Fender> sorry ill redo it http://pastebin.com/d3f72f3fb
22:10.43[TK]D-FenderBreezBl0k: New configs please
22:11.21BreezBl0kwhich ones?
22:11.34*** join/#asterisk patrick-- (n=patrick@eos.openroot.de)
22:11.40patrick--Hey all im having a Problem
22:11.50patrick--even though pbx_spool is loaded call files are not processed
22:12.02patrick--when copied into /var/spool/asterisk/outgoing
22:12.06patrick--permissions are correct
22:12.11patrick--call file syntax is correct
22:12.15patrick--no output on the CLI
22:12.53BreezBl0k<[TK]D-Fender> what do you want me to do
22:13.18[TK]D-FenderBreezBl0k: SIP clearly
22:13.53hardwireSIP Fresca.
22:14.00hardwiremmm.. now I want a soda.
22:14.11beekhardwire: Do they still make that stuff?
22:14.21hardwireyou bet your sweet patookie they do.
22:14.26hardwirehaven't you seen the amazing fresca girls?
22:15.09hardwirewhat is the voip users conference?
22:15.30beek#voup-users-conference
22:15.32*** join/#asterisk delta_16 (n=delta_16@84.26.9.136)
22:15.47BreezBl0k<[TK]D-Fender> http://pastebin.com/d1e73efcb
22:16.59*** join/#asterisk Torrieri (n=Torrieri@nelug/crew/torrieri)
22:17.26*** join/#asterisk ecret (n=ecret@CPE001e9002348e-CM001225d8ab30.cpe.net.cable.rogers.com)
22:19.24*** join/#asterisk Micc (n=Michael@c-76-121-255-52.hsd1.wa.comcast.net)
22:20.11[TK]D-FenderBreezBl0k: "canvite=no" <-- spelling is horribly off.  "canreinvite=no"
22:20.51Qwell[TK]D-Fender: chan_sip doesn't parse the middle few chars
22:21.15BreezBl0k<[TK]D-Fender> good shout
22:21.34[TK]D-FenderQwell: Yeah... they're silent... like the "p" in swimming
22:21.58Led-Heddo you think Voice mail and logging for 4 users would destroy a CF card in a short period of time?
22:22.36*** part/#asterisk delta_16 (n=delta_16@84.26.9.136)
22:22.38QwellLed-Hed: no, just don't use a journaled FS
22:22.39Ziaeonmy meetme stopped working on one of my asterisk boxes. Randomly. I put in the pin and it says that is an invalid pin number and hangs up on me (which is a different error than if i intentionally mess up the pin, where it doesnt hang up on me). Still works on my other asterisk boxes. What the feck.
22:22.44BreezBl0k<[TK]D-Fender> reloaded sip still no audio :(
22:23.11Led-HedQwell, thanks
22:23.35[TK]D-FenderBreezBl0k: I do not see good configs, including your ITSP specific peer.....
22:23.53MiccAnyone have an idea why when I call a ulaw phone from my polycom on g722 it sounds all garbled? Its fine when I dialout, but when I call one of our customers and go through their queue, I'm g722 while in their IVR system, then when they answer it sounds like mr roboto.
22:24.32denonMicc: is their queue in India?
22:24.33Micclet me clarify. dialout through an ITSP.
22:24.44Miccdenon, no, just down the street.
22:24.54MiccThey are on our asterisk server, only 45ms away.
22:24.56denonhttp://instantrimshot.com
22:25.14denonerm, nevermind :)
22:25.28BreezBl0k<[TK]D-Fender> :( they are the settings other users are using with sipgate and they have been working fine on trixbox thats the bit i cant get my head round
22:26.19[TK]D-FenderBreezBl0k: Says absolutely nothing to me.
22:28.19*** join/#asterisk tamiel (n=tamiel@ip-139.net-81-220-93.rev.numericable.fr)
22:29.18*** join/#asterisk jpeeler (n=jpeeler@asterisk/digium-software-dev/jpeeler)
22:31.00BreezBl0k<[TK]D-Fender> here are my ITSP settings
22:31.05BreezBl0k<[TK]D-Fender> http://pastebin.com/d567c8532
22:32.44[TK]D-FenderBreezBl0k: .. and [general]
22:33.12generalhani need a little agent help ... i need to define my agents in extensions.conf for the AddQueueMember, typically i would do that as AddQueueMember(Queue1|SIP/7001@internal) but this script wants it in the format - AddQueueMember(Queue1|Local/AGENT_NUMBER@internal). so how do i define my agent 1050 in [internal] to make this work properlly? any suggestions ?
22:33.58BreezBl0k<[TK]D-Fender> http://pastebin.com/d85d8b70
22:34.56*** join/#asterisk jpeeler (n=jpeeler@asterisk/digium-software-dev/jpeeler)
22:35.07[TK]D-Fendergeneralhan: thats DIALPLAN.
22:35.48generalhan[TK]D-Fender: yes. i dont understand how to set this up properlly in the dialplan.
22:35.55[TK]D-FenderBreezBl0k: [GENERAL] < should be lower case, and please confirm exactly what you have forwarded
22:36.29[TK]D-Fendergeneralhan: "exten => 1234,1,NoOp(FFS do something)"
22:41.49generalhan[TK]D-Fender: lol. i am VERY familiar with how to work MOST parts of my dialplan just fine.... in my agents context i used to say something like exten => 1050,1,Dial(SIP/${AGENT_SIP}) which was filled by a DB entry that i create when the agent logs in. however, now that im using Queuemetrics, the logon works differently, so i cant use my DB method anymore. so i dont know how to define my agents in my 'DIALPLAN' so that i can sync it up wi
22:42.39[TK]D-Fendergeneralhan: Well where DO you have a mapping of an agent to a device?
22:42.57*** join/#asterisk fatnasty1 (n=chatzill@cpe-72-190-76-209.tx.res.rr.com)
22:43.44generalhanexten => 20,3,AgentCallBackLogin(${AGENTCODE}||${AGENT_EXT}@extension-dial)
22:43.58*** join/#asterisk fatnasty1 (n=chatzill@cpe-72-190-76-209.tx.res.rr.com)
22:44.02fatnasty1I am using the meetme appication, when a participant enters the wrong digits for a a conference, the meetme app plays audio advising to re-enter the meeting info, then it hangs up on them without letting them enter anything. anyone seen this before?
22:44.17generalhan[TK]D-Fender: its a seperat extensions.conf file that is '#include'ed along with my actual extensions.conf file
22:45.06[TK]D-Fendergeneralhan: please show something useful...
22:45.58generalhan[TK]D-Fender: haha. i showed you EXACTLY what you asked for. no?
22:46.58[TK]D-Fendergeneralhan: Where do I see your extensions.conf?
22:46.58generalhani am mapping AGENT_CODE to AGENT_EXT@extension-dial
22:47.46generalhan[TK]D-Fender: the mapping itself doesnt take place in extensions.conf ... anymore
22:47.47[TK]D-Fendergeneralhan: You are adding a LOCAL channel to your AQM.  You should be adding and AGENT channel instead
22:47.53[TK]D-Fenderan*
22:47.57generalhanhmm
22:48.13generalhani am just using the dialplan as provided to me by Queuemetrics
22:48.27BreezBl0k<[TK]D-Fender> TCP/UDP 5004 - 5082 and TCP/UDP 10001 - 20000 to 172.16.0.69
22:48.43generalhanrunning that command that i just posted makes 'show agents' report my agent as being logged in to the extension i had supplied
22:49.01[TK]D-Fendergeneralhan: And they told you to use AQM like that exactly as well as ACBLI?
22:49.28[TK]D-Fendergeneralhan: because your AQM has absolutely nothing to dow ith that from what you've shown
22:49.38generalhan[TK]D-Fender: they didnt tell me anything. i click 'logon' on the web interface, and that is the dialplan peice that is run
22:50.13[TK]D-Fendergeneralhan: well so far those 2 pieces don't add up
22:50.26generalhanlet me peice this all this together in a pastebin ...
22:52.08*** join/#asterisk Firass-z0r (n=asadf@c-67-201-205-34.reshall.wwu.edu)
22:54.03Led-Hedwill a 500mhz CPU support 4 users?
22:54.35denonLed-Hed: that's a pretty broad question ..
22:54.47Led-HedI know.
22:54.49denon500mhz doing transcoding? sip to pstn?
22:54.59denoneither way, you can get something much faster for $20 or out of a dumpster ..
22:55.17Led-HedI was hoping to install Asterisk on an Alix2, which uses a 500mhz Geode CPU.
22:55.56denonthose boards have been tested with 15 sip calls (max)
22:56.13denon"(Nov 07) I have successfully compiled and installed Asterisk on an Alix board (AMD Geode 500 Mhz + 256 Mb RAM) on top of Voyage linux (a Debian variant)"
22:56.28denon"I wondered how much it could be loaded, so I tested it with pbx-test: I could place up to 15 simultaneous SIP calls before it got no more responsive."
22:57.02Led-Heddenon, great!
22:57.11Led-Hedso 4 users should be no problem
22:57.20denonI'm guessing you're safe with 4 ulaw to ulaw users
22:57.57Led-Hedsorry I dont know what a ulaw user is
22:58.06generalhan[TK]D-Fender: http://pastebin.com/d709fcf3c
22:58.11mmlj4those boards are light on RAM
22:58.13Led-Hed<--- = Total n00b
22:58.17denonerm, you probably need to do a bit more reading before you start
22:58.30Led-Heddenon, I know.
22:58.43Led-HedI'm just trying to see if its possible.
22:58.59denonprobably is -- not wise necessarily, but probably possible
23:00.10*** join/#asterisk freakazoid0223 (n=matt@pool-71-246-17-74.phlapa.fios.verizon.net)
23:00.13*** join/#asterisk shinao1 (n=shinao1@78.138.29.146)
23:01.28Led-Heddenon, will IAX or SIP make a significant difference on low powered CPU's?
23:01.52denonwell, I assume your phones will be sip right?
23:02.03denonand likely the itsp or whatever you're using will allow sip
23:02.10Led-Hednot sure. was looking at the Polycom 320
23:02.14denondoes it seem wise to want to get in the middle of that media? :)
23:02.19denonyes, sip
23:02.40beekLed-Hed: get the 330 -- only $20 more and it has a built-in switch.
23:02.45[TK]D-Fendergeneralhan: their sample if thats what this is, is broken
23:02.50denondo a little reading before you get started, these questions are a little like asking if the sky is blue .. :)
23:02.59Led-Hedbeek, ok thanks
23:03.21generalhan[TK]D-Fender: how so ?
23:04.45[TK]D-Fendergeneralhan: Using agents and AgentCallbackLogin, you have to tel AQM to CALL and AGENT.  not a LOCAL CHANNEL
23:05.12generalhanso if i just change local/ to agent/ it would fix this issue ?
23:07.42[TK]D-Fendergeneralhan: well it would sure help in using the agents you are loggin in
23:08.00*** join/#asterisk qdk (n=qdk@0x573d8dd1.bynqu1.dynamic.dsl.tele.dk)
23:10.10*** join/#asterisk _mwoodj_ (n=mwoodj@pdpc/sponsor/digium/hyper-eye)
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23:10.27*** part/#asterisk f0ner00t (i=f0ner00t@c-67-181-66-251.hsd1.ca.comcast.net)
23:14.46Led-Heddenon, I guess an Intel Atom 330 would be a better choice.
23:14.59denonatom's a nice cpu
23:15.01*** join/#asterisk paulius (n=paulius@unaffiliated/paulius)
23:15.04pauliusWould anyone be able to recommend me a good vo-ip unit which has an FXO and FXS port? People recommended the Linksys SPA3102 but it's also a router. I want something basic, stable, and that works well. I've had bad experience with consumer class devices in the past.
23:15.19Led-Hedthe Alix have a 9 sec PDD which is a long time for a business to wait for the phone to ring
23:16.17beekpaulius: The SPA3102 is works well.  I've used one for over a year and have had zero issues.
23:16.34beekYou can spend some more $$$ and get AudioCodes devices.
23:16.49pauliushttp://www.voipdepot.ca/index.php?main_page=product_info&cPath=1&products_id=116
23:16.52pauliusVoicedepo has that one.
23:16.57pauliusI wonder if it's better than the linksys
23:17.06KyleKpaulius: you dont have to use an spa3102 as a router
23:17.14pauliusKyleK: Well duh, I know that.
23:17.29pauliusBut I don't like the fact that it has it. It makes the device more mediocre.
23:17.35pauliusRather than being good at one thing.
23:17.35beekI've never used the Grandstream device.  Their phones aren't well thought of.
23:18.01pauliusbeek: So there isn't any even linksys device which just has fxs/fxo and no router?
23:18.13KyleKyea, i game in here and asked about the same HT-503 device and someone was all grandsuck
23:18.24pauliushehe
23:18.55beekI can't figure out what the big deal is.  The thing works extremely well.  Who gives a shit if it happens to have a feature that you don't want to use?
23:19.25KyleKpaulius: the router wont crash the rest of the unit if its not routing btw
23:19.46beekIt's all done with software.  If you don't run the routines, they won't hurt anything.
23:19.57pauliusKyleK: I know, but it does mean that Linksys used even cheaper components because they had to pay up to put in a processor which is capable of doing more than just voice.
23:20.38KyleKi think the version without router functions would use the same parts
23:20.41beekpaulius: So instead of taking all of the positive recommendations you're using that kind of rhetoric?
23:20.44beekWTF?
23:20.54KyleKbeek: thats kinda normal for here isn't it?
23:20.59pauliusI'm probably getting it, but I know what consumer gear means.
23:21.07pauliusAnyhow, it's only about $100.
23:21.18pauliusI've talked with a few people and a Cisco solution would cost roughly half a grand.
23:21.22pauliusFor FXS and FXO.
23:21.24beekKyleK: this is slightly different.
23:21.27pauliusSo we'll see.
23:21.44beekpaulius: It would cost $500 and not give you a damned thing more for your application.
23:22.58KyleKpaulius: are you west coast canada?
23:23.04pauliusKyleK: East.
23:23.17pauliusI'm probably ordering from SwiftGamers
23:23.53*** join/#asterisk lowlevel (n=Stuart@lowlevel.ca)
23:24.16KyleKI actually went to a store to buy mine since im close to ncix
23:24.23pauliusah cool
23:24.41pauliusoh bOOYAH
23:24.44pauliusthanks for reminding
23:24.48pauliuslocal computer store has it
23:25.23KyleKhttp://ncix.com/products/?sku=21919&vpn=SPA3102-NA&manufacture=CISCO 78.54cad
23:27.42bmoraca78.54 CAD?  that's like $15 in real money, right?  :P
23:29.49paulius...
23:29.53pauliusthat's like $70 USD
23:29.54KyleKi wish
23:30.02pauliusIt's $80 at local store.
23:30.05bmoracai was joking
23:30.10pauliusPlus tax. But same as $70 plus $10 shipping.
23:30.15pauliusPlus instant gratification
23:30.20KyleKhehe
23:30.32KyleKbmoraca: bison bucks aren't real money btw
23:31.19*** join/#asterisk juanIMP (n=Juancho@200.26.152.222)
23:41.18KyleKcan an agi see what codec a call is?
23:42.25*** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee)
23:42.27jameswfAmerican Idol upset by way of #asterisk... oh yeah!
23:42.49paulius??
23:44.14generalhanjameswf: lol, i have thought about doing that a million times with any of those shows
23:47.24*** join/#asterisk [gquit]bombadil (n=dana@CPE-72-128-66-243.wi.res.rr.com)
23:54.09*** part/#asterisk generalhan (n=asd@about/windows/staff/generalhan)

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