irclog2html for #asterisk on 20070112

00:00.06*** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net)
00:01.49De_Monif it still loops, serverA is calling serverB and just having a grand time
00:02.47Dr-Linux|homeDe_Mon:  i'd need to grab your brain for my problem :)
00:03.32Dr-Linux|homeDe_Mon: why i can't transfer the a call to agi() ?
00:04.15*** join/#asterisk Skarmeth (n=Skarmeth@201009082231.user.veloxzone.com.br)
00:05.17Dr-Linux|homeDe_Mon: nevermind :)
00:05.48Dr-Linux|homeheh, i don't if ever my this problem will resolved :)
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00:07.39*** join/#asterisk Teeli (n=tili@172.Red-88-0-147.dynamicIP.rima-tde.net)
00:08.57De_MonDr-Linux|home your doing a SIP transfer to an extension that calls agi() and the script doesnt work or what?
00:09.05*** join/#asterisk Ard0gx (n=adiaz0@200.87.243.211)
00:09.29Dr-Linux|homeDe_Mon: thanks for discussing this issue
00:09.41*** join/#asterisk riddlebox (n=james@75-132-205-166.dhcp.stls.mo.charter.com)
00:10.01Dr-Linux|homeDe_Mon: correct, in that case agi() crashes
00:10.28Dr-Linux|homeDe_Mon: but if directly dial agi() extension that works just fine
00:10.58De_Monwhy does it say it crashed?
00:11.03Dr-Linux|homeDe_Mon: i can transfer the call anywhere into the dialplan, but not to agi()
00:11.10*** join/#asterisk Flusher (i=flusher@filer.euroserv.com)
00:11.34awannabeDe_Mon, i dont have a user directive in my sip.conf
00:12.03Dr-Linux|homeDe_Mon: it simply reutruns complated 0 , same like when agi script is wrong
00:12.05Hmmhesayscan someone send me a fax?
00:12.33Dr-Linux|homeDe_Mon: i tried different agi's , i tried different version, different servers, but same problem
00:12.49De_Monawannabe awannabe any particular reason you're not using IAX for your innerasterisk communication?
00:13.27Dr-Linux|homeDe_Mon: it could be a channel reason maybe :S
00:14.01*** part/#asterisk markit (n=konversa@host119-245-static.72-81-b.business.telecomitalia.it)
00:14.14De_Monawannabe http://www.voip-info.org/wiki-Asterisk+-+dual+servers might help you out
00:14.29De_Monhttp://www.voip-info.org/wiki/view/Asterisk+Connect+2+servers
00:14.40De_Monthat's another page, same topic
00:14.55De_MonDr-Linux|home have you tried a simplier agi script?
00:15.33Dr-Linux|homeDe_Mon: yes, it never works
00:16.02Dr-Linux|homeDe_Mon: maybe it's a bug
00:16.28Dr-Linux|homeDe_Mon: anthm said, it's "channel masqu..." problem, but i don't understand that
00:17.38*** join/#asterisk TripleFFFF (n=TripleFF@145-27.mc.cite.net)
00:17.49TripleFFFFto have more then 1 email to voicmeail notif is pipe the delim ?
00:18.47*** join/#asterisk xheliox (n=jeff@pdpc/supporter/active/xheliox)
00:21.17Ard0gxHi, doing TOP on my linux I see more than one asterisk process, is that normal ?
00:21.38rudholmit is
00:22.48rene-hello
00:23.04rene-where should a call process jump after a dial to a peer who has exceeded its call limit?
00:24.33*** join/#asterisk j3g (i=nobody@189.6.32.159)
00:25.22awannabeDe_Mon, yeah ive been looking at that
00:25.45j3gi am trying to get my VOIP provider to work with my asterisk box... everything seems to be set up correctly... but when I dial, I get no sound at all.. but If I Press the HOLD button on the softphone, I can hear music on the regular phone line... the codecs are set to be the same... what could be wrong?
00:27.15De_MonTripleFFFF no, tell your sendmail app that allusers@foo is multiple recipients
00:27.45De_MonArd0gx yes
00:28.27De_MonDr-Linux|home bummer, bug anthm some more :]
00:28.48`Seaneww
00:28.51`SeanSnom phones are soo ugly
00:28.52De_Monmaybe it works better in openpbx ;)
00:28.58`Seanthere like those classic operator types
00:29.18De_Monzoom zoom zoom i'm gona go eat eat eat
00:29.52j3gand is there a easy way to get asterisk to transcode GSM(my softphone) to g729 (my voip providers best protocol) ?
00:34.26De_Monis g729 supported?
00:35.08Ard0gxDe_Mon: but have more than 10 of it with just 50 calls is normal as well ?
00:35.26Ritalin2if you have allow=g729 (and other's disallowed) won't it automatically do whatever transcoding it needs ?
00:36.15De_Monyeah, it will
00:36.37De_MonArd0gx that sentance did not make sense
00:37.23j3gDe_Mon: g729 is not supported by me (i didn't buy the codec) but my VOIP provider supports it
00:37.48Ard0gxDe_Mon: well, the thing I have 10 asterisk process doing TOP but, I only have 50 calls doing show channels
00:37.48De_Monyeah, um, asterisk needs to support it for you to send or receive
00:38.06De_Monthe number of asterisk processes is not related to the number of channels/calls
00:39.31Ard0gxDe_Mon: yeah, that I understand, but I have another asterisk box running and I have this 50 calls or more and I just have one asterisk process
00:39.43j3gDe_Mon: i have set my sip.conf to only allow ilbc and alaw for both my extension (softphone) and my ISP.. the connection seems to go well but no sound :(
00:40.21TripleFFFFde mon no
00:40.30Ritalin2i didn't realize you had to buy g729
00:40.36*** join/#asterisk sahafeez (n=sahafeez@67.109.14.227.ptr.us.xo.net)
00:40.48`SeanRitalin2 you dont..
00:40.52Ard0gxDe_Mon: so, I don't understand why appears these asterisk process. Doing 'stop now' the main asterisk process dead, but all other one noT
00:40.52`Seanthere is a free way search
00:40.52`Sean:P
00:41.02TripleFFFFi want ozzy on crack as MOH
00:41.29*** join/#asterisk sjobeck (n=sjobeck@static-70-104-254-218.ptldor.dsl-w.verizon.net)
00:41.58Ritalin2I wish I had VPN setup :-/
00:42.39*** part/#asterisk TripleFFFF (n=TripleFF@145-27.mc.cite.net)
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00:45.19De_MonArd0gx if they arnt dying from stop now they arnt threads
00:45.32De_Monwhat distribution are you using?
00:45.42De_Monor did you compile yourself
00:47.31Ard0gxDe_Mo: I'm using RHEL 4.0
00:48.16rene-are hints useful when your phones dont support subscriptions?
00:48.36Ard0gxwith 'asterisk-1.2.12.1' and  'zaptel-1.2.9.1'
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00:51.51rene-so basically asterisk publishes information of extensions tru hints and subscriptions capable devices blink leds to extensions subscribed to?
00:52.50j3gone thing I noticed now... If I dial from asterisk to a regular phone (using my voip provider) and I keep talking on the microphone on the PC I get a few (maybe 2) seconds of sound going to the regular phone line.. then all gets silent
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00:57.39perdanyone here use 7960 with sip? i have an easy and quick question about it
00:59.21retrogradeorbityes. 7960, 7941 et al with sip. shoot yr question
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01:17.56Drukenanyone have experince with via sata raids ?
01:20.53awannabezaptel can be a biatch!
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01:31.15awannabeanyone around to help me trouble shoot a dual server SIP issue
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01:39.04Ritalin2awannabe: what now
01:39.35awannabestill doesnt work, something with the context I guess
01:39.38Ritalin2what is it you are trying to accomplish
01:39.41awannabeit just does a loop
01:40.01awannabeplace calls from serverA to serverB, serverA has PRIs in it
01:40.07awannabeso serverB can use the dialtone acourse
01:40.07Ritalin2okay how do you have things setup physically?  where are the two server's etc
01:40.17awannabeacross a WAN link
01:40.26Ritalin2are they in different locations or something?
01:40.33*** join/#asterisk dorphalsig (n=dorphals@pcsp163-73.supercabletv.net.co)
01:40.42awannabeyeah
01:40.53Ritalin2i think what you need to do is set up a link between the two as a trunk
01:41.06awannabehrmm
01:41.06Ritalin2though i dont know this for sure since i've never tried to do it.  that's just my understanding of how things work
01:41.23awannabei mean IP works fine, so i dont think it needs a trunk
01:41.38awannabeor from what ive read anyways
01:41.49Ritalin2are they both running asterisk?
01:41.50dorphalsigUmmm hi, does anybody know if I need a gatekeeper in order to be able to place/recieve calls thru oH323?
01:42.27awannabeyeah both asterisk
01:42.34awannabesame version, same hardware, all the same
01:42.53dorphalsigI'm trying to make a Cisco speak with * 1.2.14
01:43.10dorphalsigis it a lost case?
01:43.17Ritalin2awannabe: i dont know how to pass an exten from one server to the other.  i think what you want to do is dial like 9 or something to connect to the other server and then configure an extension on the other server so that it automatically gives a dialtone when you connect... enter your number and then it uses that to dial
01:43.27Ritalin2i mean... that'd be one solution ... again i'm no expert
01:43.30dorphalsig(and excuse my ignorance but wtf does a gatekeeper do?)
01:43.54awannabeyeah ie seen it done several times before without all that garbage
01:44.42Ritalin2you setup so that the remote server is registered?
01:44.49awannabenope
01:45.47Ritalin2how are you connecting to the other server?
01:45.56ManxPower"Gatekeeper" is an H323 term.  Think of it as sort of an H323 "router" or "proxy" (neither term is really correct)
01:45.57[TK]D-Fenderawannabe : You set up 1 SIP account on each side just like you would a phone.  You'd create a pattern match that will then pass the call off that account to the other server and it will process it just like a normal call from a standard phone.
01:46.14awannabe[TK]D-Fender, thats what i thought, but i get a nasty loop
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01:46.39perdwith the sip firmware on the 7960 i cant just start dialing numbers on the dialpad like i could with the skinny firmware, is there a setting for this or is it something in the dialplan.xml ?
01:46.41[TK]D-Fenderawannabe : then you need to fix contexts & prefixes.
01:46.51awannabeyeah, context is the prob
01:46.57dorphalsigBut, do I really need it in order to use asterisk with H323?
01:47.02Ritalin2or you don't have the serverA->serverB sip set up?
01:47.04ManxPowerIf you get a loop, then you have a dialplan problem, not a SIP problem
01:47.07awannabewhat should the context for the sip entries be then?
01:47.14ManxPowerdorphalsig: I don't think so.
01:47.20awannabethats whats confushing me i guess
01:47.43ManxPowerdorphalsig: The only people that use H323 with Asterisk are 1) People that like pain and 2) People that have no other choice.
01:48.11[TK]D-Fenderawannabe : I'm thikning you're passing an exten that happens to be the pattern match for heading the OTHER way as well... make sure they DON'T match or that your inbound context processes what its given and don't jsut feed it back in as-in (like stip prefixes, etc.
01:49.05awannabe[TK]D-Fender, yeah it is...cause that pattern it suppsoe to send it out the PRI
01:49.06ManxPowerRemember, if you have two pattern matches that could match a number, any pattern match in an include =>'d context will match last and never match if there is something more specific.
01:49.35bokanyone know if there is a way to limit the calls to a sip phone to one, even if that phone has multiple lines registered. ie if someone is on the phone all lines are busy?
01:49.47ManxPower[TK]D-Fender: I'll bet he's doing the classic newbie of exten => _XXX,1,Dial(SIP/${EXTEN}) instead of an individual Dial() line for each extension
01:50.19ManxPowerbok: That happens by default on almost all phones if you set up a separate SIP account for each line.
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01:50.43bokManxPower: ahh ok, so its a phone setting then, cheers!
01:51.16perdManxPower: so it's bad to do like.. _88XX,1,Dial.... ?
01:51.25awannabeManxPower,  exten => _NXXNXXNXXX,1,DIAL(SIP/super/${EXTEN},200,tr) is bad?
01:51.26perdyou enter every extension in separately?
01:51.46ManxPowerperd: of course.
01:51.57perdi see, what is the reason for that?
01:52.07ManxPowerHow else would you allow users to do custom call routing, custom features, etc
01:52.14perdseems like entering in a billion extensions would be a bad idea, since you waste time
01:52.29ManxPowerHeck, I could not even have a fax->email gateway if I didn't do it that way
01:52.43ManxPowerperd: you only do it once for each extension when it is created or changed
01:52.48perdi have a fax->email gateway and i do it that way
01:53.12ManxPowerperd: how many users have the fax->email gateway?
01:53.17perd25
01:53.21perdall of 'em
01:53.23ManxPowerand how do you map and extension to an e-mail address?
01:53.23perdsmall office :)
01:53.36perdi use iaxmodem
01:53.40perdand hylafax
01:54.04perdthe recvdfax script checks iax.conf, each persons 'modem' devined there has a ;email= line
01:54.19perdi just parse the iax.conf, get the email, convert tiff to pdf and shoot it out
01:54.30perdi couldnt get spandsp working with 1.4
01:54.39perdwell, for app_txfax
01:54.47[TK]D-FenderManxPower : For internal extension, I wouldn't say its "bad".....
01:54.57[TK]D-FenderManxPower : Let them 404... who cares?
01:55.15perdwhen people dial a bad extension they get a message
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01:55.44perdi use voicemail to do that.. if no voicemailbox is found it dumps out of the macro and goes to _88XX,2, which is pbx-invalid
01:55.54ManxPowerpart of my dialplan (domain is fake, of course)  http://pastebin.ca/313754
01:56.26awannabeoh shit
01:56.55ManxPower[TK]D-Fender: I use pattern match for dialed extensions that do not match a valid extension
01:57.30ManxPowerAs you can see there is SOME duplication of extension call routing, but not much
01:57.31awannabeyeah
01:58.09ManxPowerAlso, we do not use the extension as the SIP device ID for two reasons 1) EXTENSIONS AND DEVICES ARE NOT THE SAME THING and 2) it is easier for us.
01:58.47perdmy extensions.conf is kind of fucked up
01:59.03perdheh, this is how i do my internal extensions http://pastebin.ca/313757
01:59.19perdi have been flipping between defining each individually and using the pattern matching
02:00.00ManxPowerdo it individually.  It was massivly more flexable
02:00.12perdyeh i probably will
02:00.12Nivex<PROTECTED>
02:00.50ManxPowermacro-std-exten basically has a set of defaults, then you can override the defaults with setting dialplan channel variables
02:01.24awannabethat was both of them
02:01.51ManxPowerAlso I can find any specific line appearance on any phone by having the user read me the numbers off the little white sticker on the bottom of the phone (the MAC)
02:02.15ManxPowerThat is good when 1 extension rings 4 different line appearances on 3 different phones
02:02.30perdwhat are line appearances? multiple extensions on one phone?
02:02.45perdmultiple phones ringing from one DID?
02:02.47ManxPowerline appearance == line on the phone
02:02.52perdoh
02:02.57ManxPoweri.e. a 3-line phone has 3 line appearances
02:03.01perdright
02:03.13ManxPowerbut since they are not actually lines.....
02:04.34*** join/#asterisk shepimport (n=shep@h194.189.31.71.ip.alltel.net)
02:05.12shepimportHey all anyone have a sec to work on a trunk registration issue?
02:05.43ManxPower~freepbx
02:05.46jbotfreepbx is, like, unable to be supported here. It's a complex piece of dialplan and GUI, and can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there
02:05.49[TK]D-FenderManxPower : lol
02:06.09[TK]D-FenderManxPower : Down boy!  Let him finish incriminating himself, THEN decapitate ;)
02:06.32ManxPowerhe said "trunk"
02:06.44[TK]D-FenderManxPower : There is a small-moderate chance against the "easy guess" of "trunk" :)
02:07.09[TK]D-FenderManxPower : the term DOES get thrown around a lot... not enough to execute on for me yet...
02:07.31shepimportwow... thats the community spirit... actually i am a senior sales engineer for a session border controller company and work with sip signalling daily... but.. maybe they are more polite in the freepbx room
02:07.49[TK]D-Fendershepimport : So you are indeed using FreePBX?
02:08.15shepimportno, just writing to conf files
02:08.53[TK]D-Fendershepimport : Well then go right ahead and pastebin up your errors, and your sip config's (mask only passwords).
02:08.57[TK]D-Fenderuse www.pastebin.ca
02:09.33shepimporti am in the cli... created a new trunk, which is actually an ext off a broadsoft server (through an acme SBC)
02:09.53shepimporti see it register aok... than it loses the authentication...
02:10.02shepimportone sec i will paste
02:12.42Hmmhesays[TK]D-Fender your incoming fax working yet?
02:13.03shepimporthttp://pastebin.ca/313769
02:13.37rene-it is funny i thought asterisk was the acme voice product
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02:14.26[TK]D-FenderHmmhesays : Didn't check...
02:15.40[TK]D-Fendershepimport : You're IP's don't match between your peer & register
02:16.32shepimportno they do... i was changing them since the SBC is not fully locked down... but i just missed a few
02:17.55[TK]D-Fendershepimport : Missinga few doesn't SOUND good to me...
02:18.03Hmmhesaysi just faxed you something
02:18.34shepimporti meant when i was chaning them... they all match in whats in the conf file
02:19.18[TK]D-Fendershepimport : Look at your pastebin.  the register & peer IP's simply don't match
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02:20.48perdhe changed them for obfuscations sake, fender
02:20.57ManxPowershepimport: Ah, that explains the use of "trunk".
02:20.58shepimportfender: check it now... they do... i pulled that from a file i did not save
02:21.44ManxPowershepimport: is either end of the connection behind NAT?
02:21.59shepimporteven off ethereal thecaptures just show a standard username/authentication mismatch... but i know it is entered correctly
02:22.02[TK]D-Fenderclean it all up, and test again.  Also, in no setup for BV I've every seen do you enter raw IP's like tha, esp for named domains....
02:22.14[TK]D-Fendershepimport : Feels dangerous
02:22.31retrogradeorbitperd, there is a regular expression syntax in the phones config that deals with numbers and delays to deal with call initialisation
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02:23.10retrogradeorbitie at what point the number is considered 'complete' and the SIP call begins
02:23.11shepimportBV? BW?
02:23.28ManxPowershepimport: diagram it.  i.e. Asterisk -> Internet -> NAT Router -> Remote SIP server or however it is.
02:23.33[TK]D-Fendershepimport : nevermind, I misread something earlier
02:23.56[TK]D-Fendershepimport : This is your BroadSoft server... It hough it was an ITSP...
02:24.58ManxPowerIt sounds like Asterisk (SIP) <-> ???? <-> Broadsoft SIP Box
02:25.04ManxPowerIS that correct, shepimport?
02:25.07shepimportit is behind a NAT router... but the sofswitch is behind a Session border controller... so that does the NAT transversal by creating a remote ALG on my firewall... so the externalnat entry is not required
02:25.40ManxPowershepimport: so you are SURE the issue is not with the NAT box closing the translations because of inactivity?
02:25.40shepimportAsterisk --> nat ---> Acme SBC --> Broadworks/sof
02:26.16shepimportthe sbc is set to send an options every 2 seconds...
02:26.28shepimportcan you use a qualify on a trunk?
02:26.57ManxPowershepimport: In Asterisk there is no such thing as a "SIP trunk"  They are all just devices
02:27.41ManxPowerThe only people that say "SIP trunk" are the ones using some sissy GUI that complicates the config files so much we all just want to die.
02:27.45shepimportso let me add a qualify and a external NAT entry... and get back to ya... thanks for the support
02:28.04ManxPowershepimport: qualify basically sends an OPTIONS every second or so
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02:28.31shepimportyeah... qualify does one every 2 seconds unless maunally set
02:29.07shepimportok thanks a lot.. i will get back to yall and let ya know the outcome
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02:33.52awannabe[TK]D-Fender, you have any docs on setting two server up to talk to each other? cause i am having a hell of a time!!
02:34.22km-Heh, I have a bit of a weird situation here, and I think I'm just doing something stupid.  I have an ATA device.  Lets say I have a call going on that ATA.  I flash over and dial 700, thinking I'm parking the call in progress.  Oddly enough, it parks the new call instead.
02:34.51ManxPowerkm-: you need to complete the transfer
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02:35.32ManxPowerIf you didn't have to do that then you would never hear the parking slot number (the person being parked would hear it)
02:35.54ManxPowerParking a call is just an attended transfer to the parking extension
02:35.54km-I think I found my error.
02:36.03km-If I hit #, it brings up allison and "Transfer!"
02:36.05[TK]D-Fenderawannabe : lookup "asterisk dual servers" on the WIKI, that'll give you about 3 ways to do it
02:36.10km-If I flash, however, it opens a second line.
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02:36.39ManxPowerkm-: what happens if you just hangup after you hear the parkinglot number.  The call on hold should then be transfered
02:36.51km-yep, if I hit # it transfers.
02:36.56km-It was a misunderstanding on my part
02:37.00km-I thought flash = transfer
02:37.00awannabe[TK]D-Fender, yeah ive tried that
02:37.01Strom_Coh hell, not # inbound transfer
02:37.02km-but it's # = transfer.
02:37.05ManxPowerkm-: DO you know how to do an attended transfer using FLASH on your ATA?
02:37.07CrashsysAnyone have problems with a Sangoma A200D being quiet on rx?
02:37.16km-ManxPower: nope, have no idea.
02:37.17ManxPowerkm-: no, FLASH would START a transfer.  You still have to complete it.
02:37.30km-how would I do an attended transfer on a sip1000?
02:37.32ManxPowerkm-: figure it out and you'll be able to park with FLASH
02:37.33km-err, spa1000
02:37.36km-hmm.
02:37.41ManxPowerkm-: I'm sure it's in the manual
02:38.07km-I will putter with it some more
02:38.13Strom_Ckm-: try this
02:38.13ManxPowerI thought it was FLASH + DIAL NUMBER + HANGUP
02:38.22Strom_Cflash, dial number, hang oh damn you
02:38.29ManxPowerkm-: it is working the way it is supposed to work
02:38.37km-I'm not insinuating it isn't
02:38.42*** join/#asterisk BSDTech (n=RNeese@ppp-69-238-75-25.dsl.irvnca.pacbell.net)
02:38.42Strom_Cthat'll teach you to type faster than me!
02:38.45km-I'm saying I'm a dumbass and need to play with it more :)
02:38.48ManxPowerouch!
02:38.55*** join/#asterisk BZBW (n=wlwzhang@ip67-153-142-110.z142-153-67.customer.algx.net)
02:39.04ManxPowerkm-: Attended Transfer is required to do parking
02:39.21QwellStrom_C: http://consumerist.com/consumer/sprint/cancel-sprint-account-by-writing-intelligent-wellcrafted-emails-to-the-ceo-228053.php
02:39.23km-the "#" starts an attended transfer, I believe.  Since dialing 700 will indeed give me the "7 0 1"
02:39.25ManxPowerRegardless of if it is a FLASH attended transfer or a # attended transfer
02:39.27BZBWstrange, no one complains about BLF not working in 1.4, other than me:(
02:39.47QwellBZBW: Did you try the latest branch 1.4?
02:39.50*** join/#asterisk shido6 (n=shido6@d221-68-200.commercial.cgocable.net)
02:39.52ManxPowerBZBW: Guess what version of Asterisk I'm running?
02:40.22BZBW1.2.10?
02:40.30QwellManxPower: 1.0.8?
02:40.52ManxPower1.2.x
02:40.58ManxPowerDo you know why I'm running 1.2.x?
02:41.03*** join/#asterisk mjh001 (n=mjh001@c-68-37-78-102.hsd1.nj.comcast.net)
02:41.04BZBWQwell: why latest branch, I download the official one, is there any different official 1.4?
02:41.15QwellBZBW: there are bug fixes and such
02:41.18ManxPowerBecause 1.4.x has not been out long enough to have the critical bugs and not so critical discovered and fixed
02:41.48BZBW1.4 has been in beta for more than 3 months:(:(
02:41.49ManxPowerI call this the "I'm tired of users bitching at me" plan for upgrades
02:42.45BZBWQwell, if I need the latest branch, I can have to use SVN, right?  WHat branch is it?
02:43.01Qwellhttp://svn.digium.com/svn/asterisk/branches/1.4
02:43.35BZBWI hope it doesn't introduce other serious bugs:?
02:43.41*** join/#asterisk shepimport (n=shep@h194.189.31.71.ip.alltel.net)
02:44.33shepimportHey guys that did not work (qualify or externip/localnet) will try again tommorow... thanks for the effort... night!!! need to get home before the mrs strangles me
02:46.01awannabe[TK]D-Fender, each phone doesnt have to be registed on the other server, right?
02:46.34CrashsysAre milliwatt test-tone numbers generally a default number or is it all specific to provider?
02:46.40km-OK.  I think I figured it out.
02:46.49QwellCrashsys: it's specific to a provider and area.  call them and ask for it
02:46.50[TK]D-Fenderawannabe : not at all.
02:46.56km-I flash.  The other call connects.  I flash back, both parties are connected.  I then hang up.  Then the transfer completes.
02:47.08[TK]D-Fenderawannabe : you set up 1 account on each side, and the PBX's call each other to bridge the calls.
02:47.13awannabe[TK]D-Fender, anything else i need to check, if both server have static IPs, i dont need a regsiter =>, correct?
02:47.18CrashsysQwell: heh, I tried calling the operator and "help desk" and they said "milli-what?"
02:47.27Crashsysmaybe i'll run a line-truck off the road and ask them
02:48.56infinity1i have a few polycom phones in the office and the pbx is outside the office so we're using NAT. Obviously we're having problems getting the polycoms to work 100%. Is there a solution?
02:49.26[TK]D-Fenderawannabe  Correct
02:50.33*** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal)
02:50.33*** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal)
02:52.01awannabe[TK]D-Fender, my dial() that is SIP/server, server has a entry [server] entry in sip.conf
02:56.54awannabe[TK]D-Fender,  Forbidden - wrong password on authentication for INVITE to '"Front Desk" <sip:306@192.168.166.10>;tag=as336bd92f' i get that, but the phone is registered!
02:59.07BZBWI have a SIP service provider, only allow one call at the time, is there a function for me to detect if that peer account is used so that I will route the call to another SIP service provider?
02:59.56BZBWChanIsAvail is not the one, that i know of.
02:59.59awannabeOMG!!! ok, it works!!
03:03.24[TK]D-Fenderawannabe : yay
03:03.42awannabehella weird
03:03.48awannabeusername on the phone, i switched it, it works
03:04.01awannabenow, i cant dial inward from PSTN to other side, sayin unknown hostname, so i gotta see what the heck that is
03:05.54CrashsysDoes a sangoma A200d have automatic gain control?
03:06.09ManxPowerCrashsys: any reason to think it might?
03:06.47CrashsysHoping it might :)
03:07.12CrashsysCause I have 16.0 set for my rxgain in zapata.conf and the people in the office say it's still to quiet...
03:07.27Crashsysseems like a lot
03:07.55Crashsysfor POTS
03:08.42[TK]D-FenderCrashsys : on an A200?
03:08.52CrashsysYeah
03:09.04CrashsysPolycom IP430's...
03:09.06[TK]D-FenderCrashsys : I'm guessing not... syour lines are whacked...
03:09.10danphmm
03:09.14[TK]D-FenderCrashsys : what firmware?
03:09.22CrashsysThe A200 is v07
03:09.31[TK]D-FenderCrashsys : the poly's
03:09.32danpi have mine set to 8.0
03:09.40CrashsysSip 1.6.7
03:10.04CrashsysTomorrow i'm going to walk around and make sure the volume is up on all the handsets...
03:10.14CrashsysSince the one computer/phone I sat at sounded plenty loud to me today
03:10.24[TK]D-FenderCrashsys : I think thats it... there were 2 releases of 1.6.7, and I found that the upgrade to 2.X helped a LOT
03:10.28CrashsysFor all I know is they turned the speakerphone up, then picked the handset up...
03:10.36danpmy tx was pretty hot so i had to turn it down to -5.0
03:10.45CrashsysMy TX is fine...
03:10.49Crashsysit's the RX that's lame
03:11.03Crashsysaccording to ztmonitor... but it looks like i'm getting good signal where it's at...
03:11.12Crashsysfender: you had low-audio issues with 1.6.7 firmware?
03:11.44CrashsysI'm trying to find Verizon Tampa's milliwatt test tone number...
03:12.28*** join/#asterisk TheCops (n=henri@got.securebinary.com)
03:12.41[TK]D-FenderCrashsys : yup
03:13.08in-pthi all
03:13.29in-ptcan anyone please tell me what is the syntax for dial string of skinny channel
03:13.40QwellDial(line@device)
03:13.43Qwellerm
03:13.47QwellDial(Skinny/line@device)
03:13.59Crashsysd-fender: Any reconfig necessary for upgrading?
03:14.03CrashsysOr just the same conf's you had?
03:14.14in-ptthnx Qwell
03:15.09BSDTechyour all fired
03:15.15Crashsysgood
03:15.17BSDTechwrong channel
03:15.18Crashsysnow I can go get drunk
03:15.23BSDTechlol
03:15.25Crashsysohh :(
03:15.36BSDTechits taking to long to fix the current issues
03:15.46BSDTechyou all must work extra hours
03:15.47wunderkinCrashsys, hey since you have an ip430, and are on 1.6.7 now and maybe upgrading (?) can you test something for me
03:15.54BSDTechfix it then your fired
03:16.09Crashsyswunderkin: I'm not at the site... but what ya got in mind?
03:16.20BSDTechip430 being a polycom why use the 1.6.7
03:16.28BSDTechwhy not move to 2.0.3
03:16.30in-ptwhat is the reson for this error on asterisk cli  Got SIP response 489 "Bad event" back from 192.168.0.80
03:16.39wunderkinyou don't have the phone where you're at? darn.. well how about someone else?
03:16.41CrashsysBSD: Cause I didn't think of it at the time?
03:16.57BSDTechok
03:17.12CrashsysDo I need to re-write the config's for 2.0 or will 1.6 configs run under a 2.0 sip without issues? That is my current mission...
03:17.22wunderkinCrashsys, redo
03:17.25BSDTech1.6 should work fine
03:17.29[TK]D-FenderCrashsys : Definately must rewrite
03:17.31BSDTechmine do
03:17.34Crashsys...
03:17.37[TK]D-Fenderbad risk...
03:17.43[TK]D-Fendermany parms changed.
03:17.44BSDTechthey are the same
03:17.48wunderkinso i guess i shouldnt ask BSDTech to test for me :D
03:17.51[TK]D-FenderI would NOT take chances with this
03:18.14BSDTechI will pass you a config from 2.0.3
03:18.19Crashsysblah... polycom needs a more sane config file layout..
03:18.33[TK]D-FenderCrashsys : Works rgeat for me....
03:18.33CrashsysFirst, I gotta figure out how to download 2.0.3 :D
03:18.37BSDTechit has 3 files now
03:19.11BSDTechsip.cfg mac.cfg and phonee####.cfg
03:19.40CrashsysMy 1.6.7 has that... <macaddr>.cfg, sip.cfg, and phone.cfg
03:19.52BSDTechthats it
03:20.20BSDTechthere is the mac-directory~xml
03:20.26BSDTechbut its not really needed
03:20.47*** join/#asterisk linlin (i=will@71.194.70.13)
03:21.12BSDTechI love mty 501 and soon to get a 601
03:21.26wunderkinwhile we are on the subject, can someone with an ip430 try making an outbound call (not answered), and call into the phone and answer the incoming call .. tell me if your phone stops responding and reboots..?
03:21.54TheCopsSomeone is using Callpickup ? I have difficulties to figure out how it is working. I only want to intercept an incoming call, but if I read the description, it intercept outbound call too.
03:22.12BSDTechwhat boot rom ver
03:22.32wunderkinme? the latest
03:22.46BSDTech3.2.2 and firmware 1.6.7
03:23.02wunderkinTheCops, no.. i dont think so..
03:23.54BSDTechwunderkin ?
03:24.00CrashsysAnyone got the Sip2.0.3 zip from polycom?
03:24.00TheCopswunderkin, So, If I add all SIP extensions in the group 1, I can use *8 to take an incomming call?
03:24.18wunderkinyeah
03:24.23BSDTechshhhhh where di you herar about 2.0.3
03:24.32BSDTechits a secrete
03:25.02wunderkinim using 2.0.3 now but i'm just trying to figure out if it is just me with the problem, other than polycom god
03:25.18BSDTechI know on the 2 we have at the office 2.0.3 and 3.2.2 works fine
03:25.28BSDTechnever went to 21.6.7
03:25.32wunderkini had this problem before 2.0.3 back to 1.6.x i think
03:26.04Crashsyswunderkin: I know that 1.6.7 will let me call the phone without rebooting...
03:26.26wunderkinyeah, that's a nice feature of a phone, but that wasn't my problem... lol..
03:26.46BSDTechwe have no issues with 3.2.2 bootrom and firmware 2.0.3 on the 430 we have
03:27.12BSDTechand I think I have 2.0.3 here some where
03:27.25danpi found it on google
03:27.35danpsearch for polycom 2.0.3
03:27.37BSDTechok
03:27.39danpthat is all i will say
03:28.12CrashsysCan you downgrade a phone from 2.0 to 1.6?
03:28.19BSDTechnope
03:28.23Crashsysdman
03:28.24wunderkinyeah you can
03:28.25Crashsyserr damn
03:28.29BSDTechnot if you have boot rom 3.2.2
03:28.34wunderkini have
03:28.46BSDTechand for the most part down grading can break a phone
03:29.32*** join/#asterisk j0 (n=dan@S01060016b6b541d2.va.shawcable.net)
03:29.34wunderkini don't know about that, i think mine were already broken, but you aren't supposted to use 1.6 configs on 2.0 apparantly ;P
03:29.55wunderkini believe it did say that somewhere in the docs
03:30.27BSDTechI just did a diff on 1.6 and 2.0 and there are a few minor change sin sip.cfg thats it
03:30.37wunderkinit would be nice to have a config validator and stuff, i really thought my problems were due to bad config... it just amazes me that this many phones are bad.....
03:30.58TheCopsI did an upgrade 1.6.7 to 2.0.3 and I keep the config of my 1.6.7
03:31.03[TK]D-FenderCrashsys : yes, you can downgrade SIP versions.
03:31.04TheCopsworking perfectly
03:31.18[TK]D-FenderCrashsys : its Bootrom's you're stuck with from 2.x to 3.x
03:31.34CrashsysI could care less about the bootrom... it boots... :)
03:31.38Crashsysit's 3.2.2 anyways
03:31.41[TK]D-FenderCrashsys : more on the 500&300
03:31.42BSDTechyeah the only changes are in the sip.cfg and it updates when you untar the file
03:32.04Crashsysyou mean overwrites?
03:33.06BSDTechyes it puts a new ver in its place
03:35.13*** join/#asterisk xheliox (n=jeff@pdpc/supporter/active/xheliox)
03:41.10*** join/#asterisk Globetrotter (n=eric@205.211.239.11)
03:41.39*** join/#asterisk BSDTech (n=RNeese@ppp-69-238-75-25.dsl.irvnca.pacbell.net)
03:41.50Globetrotterhi guys,,  how do i ftp to asterisk bussiness edition ???
03:43.49wunderkinyou sure are trying...
03:44.02*** join/#asterisk adrianqcs23 (n=Adrian@60.52.206.125)
03:44.08BSDTechyou dont
03:44.16adrianqcs23hi
03:44.20BSDTechyou have to buy it and they ship you a cd
03:44.49BSDTechand the manual
03:45.06Globetrotteri already have it,,  bnut i cannot get ftp access to it to work
03:45.21russellbhow does asterisk have anything to do with ftp access?
03:45.27adrianqcs23can someone help me with my call monitoring? i tend to monitor calls with the manager cli using telnet to record calls on the fly....but the sound file is empty i.e. 44bytes
03:45.27Nuggettelnet is eeeeeeevil!
03:45.37BSDTechssh
03:45.42BSDTechits your friend
03:45.52danpdoes the manager support ssh?
03:45.59rene-it doesnt
03:46.01danphuh.
03:46.03BSDTechwhat manager
03:46.06russellbit does support SSL in trunk :)
03:46.12BSDTechwtf you talking about
03:46.14rene-heh
03:46.25rene-BSDTech: you didnt seem you understood the issue
03:46.27Globetrotterbecause i am trying o get my monitored recordings of the server so that we can pull up calls at request
03:46.52adrianqcs23yes
03:47.07adrianqcs23i am using the monitor command
03:47.26adrianqcs23i.e. Action: Monitor
03:47.31BSDTechdog needs walking
03:47.33adrianqcs23in telnet
03:47.40BSDTechuse ssh
03:47.58adrianqcs23i can tel net
03:48.01BSDTechor get ari
03:48.14BSDTechthen your box is insecure
03:48.29BSDTechand a network secrity risk
03:48.31adrianqcs23i can execute orignate and monitor that channel
03:48.51adrianqcs23i can save the files
03:49.27*** join/#asterisk BitBandit (i=McDonald@68-187-56-58.dhcp.stgr.ut.charter.com)
03:49.34adrianqcs23but when monitoring a channel bridged, it saves 0 bytes of sound files
03:50.04adrianqcs23anybody an expert in asterisk?
03:50.08rene-adrianqcs23:  that is weird
03:50.09adrianqcs23hello
03:50.33*** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal)
03:50.33*** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal)
03:50.40rene-russellb:  how do i download branch asterisk?
03:51.08rene-BSDTech: it is recommended that manager access is disallowed outside of local net
03:51.26Globetrotterhi guys,,  me again..  i want to be able to share my monitor recordings to pull up on demand...  any ideas?
03:51.31rene-BSDTech: the asterisk creators recommend to firewall those boxes
03:51.43rene-Globetrotter: i would publish them over http with apache
03:51.49*** join/#asterisk sethwhit (n=SethWhit@70-56-234-164.clsp.qwest.net)
03:51.52rene-why bother with ftp
03:52.14Globetrotterbecasue i dont know any better :)
03:52.44rene-well i would set an alias for recordings to point to the location of asterisk in apache
03:52.47adrianqcs23Globelrotter: how do you monitor channels on the fly?
03:53.13rene-for specifics you might want to ask on #apache
03:53.23adrianqcs23Globletrotter: Without modifying the extensions.conf
03:53.38Globetrottersorry,,  but i am not too familiar with astrerisk yet..  still learning
03:53.43adrianqcs23I am trying telnet and accesing the manager
03:54.09adrianqcs23but have inconsistency in recording them.... some appear to be 44 bytes???
03:54.38TheCopsWhat's the relation with the callgroup and pickupgroup? I want a specific group can take call from another group, I don't understand how to configure it.
03:54.42rene-adrianqcs23: i dont know if this would help but you can use auto recording by dialing some feature code that is activated in features.ocnf
03:54.55rene-TheCops: that is tricky and i dont remember
03:54.57rene-heh
03:55.09TheCopsrene-, yup seem to be complicated
03:55.33rene-but configuration is not difficult
03:55.38wunderkincallgroup sets the group .. pickupgroup gets the groups... you set pickupgroup to the callgroups that you want to allow it to pickup
03:55.56rene-oh
03:56.06rene-that is clean
03:56.25TheCopsho
03:56.29*** join/#asterisk bhrobinson (n=brobinso@northtx1-static.telwestonline.com)
03:57.03TheCopswunderkin, so, if I want ext 200 to pickup call for 201,I put callgroup=1 to 201, and pickupgroup=1 to 200
03:57.47wunderkinyeah i guess
03:57.59TheCopsNice thank you
03:58.47bhrobinsonis anyone in here from iowa?
04:00.08adrianqcs23One-touch recording
04:00.11adrianqcs23<PROTECTED>
04:00.11adrianqcs23<PROTECTED>
04:00.11adrianqcs23IN and OUT audio will be split into two files, and will be available on your asterisk server in the following directory:
04:00.19adrianqcs23what does it mean by one-touch?
04:00.37rene-that you press some combination of keys to enable instant recording
04:00.51wunderkinkeys != 1
04:01.01rene-maybe it is not one touch since most key combinations are composed of multiple keys
04:01.25rene-but that is what is called
04:01.35adrianqcs23arghhh!
04:01.43rene-i am trying to do svn checkout http://svn.digium.com/svn/asterisk/branch asterisk but it doesnt work
04:01.47adrianqcs23i wan to record without the user to press anything
04:02.12rene-it should be possible but i have never done it
04:02.26rene-then why not enable recording of every call?
04:02.31rene-who decides when to record?
04:02.43wunderkinasterigod
04:02.56rene-wunder any clue on how to download branch?
04:03.49rene-it seems to be svn checkout http://svn.digium.com/svn/asterisk/branches asterisk
04:04.03rene-but that got me 1.2
04:04.06rene-:p
04:04.09adrianqcs23how do you enable recording of every calls?  do i need to go to every line extension and run the monitor command?
04:04.19adrianqcs23that would be tedious!
04:04.39wunderkinbranch?
04:04.47rene-wunder yes
04:04.48adrianqcs23by the way the extension is maintained by fonality... so i dun wan to mess up the extension.conf
04:05.07[TK]D-Fenderadrianqcs23 : GUI's are not suported here, sorry
04:05.09wunderkin... you mean trunk, but why?
04:05.13rene-i mean branch
04:05.17wunderkinguis are too pretty for [TK]D-Fender
04:05.27rene-i have branch installed by digium
04:05.38rene-but that has a broken manager interface
04:05.38adrianqcs23i need some way to 'record the channels' without messying with the configuration files
04:05.53rene-adrianqcs23: it will be harder that way
04:06.02*** join/#asterisk raina (n=raina@pdpc/supporter/active/ro3159)
04:06.29rene-i want a newer branch install
04:06.31adrianqcs23<[TK]D-Fender> yeah i know that.... i am trying to recording calls
04:06.44wunderkin~ GUIs are too sexy for [TK]D-Fender's shirt.. too sexy .. it hurts! ~
04:06.45jbotokay, wunderkin
04:06.51adrianqcs23<[TK]D-Fender>  without accessing the gui
04:06.53wunderkinlol ! jbot agrees
04:07.07adrianqcs23<[TK]D-Fender>  preferably by agi or telnet
04:07.23rene-agi is messing with the dialplan
04:08.09[TK]D-Fenderadrianqcs23 : in in tru fashion chances are any changes you do manually will get blown away then next time you apply any changes to the GUI info.
04:08.26[TK]D-Fender~lart jbot
04:08.29[TK]D-Fender:O
04:08.32wunderkin:D
04:08.37[TK]D-Fenderomg pwned
04:08.43[TK]D-Fender~jbot
04:08.44jbotextra, extra, read all about it, jbot is only marginally useful at best,  He got a C- on his Turing Test
04:08.44wunderkinpwned.
04:08.57[TK]D-Fenderjbot : suck THAT biotch!
04:08.58jbotACTION sucks THAT biotch!'s lips
04:09.04[TK]D-Fender:D
04:10.23*** join/#asterisk CunningPike (n=CunningP@S010600095b33697f.vc.shawcable.net)
04:11.12[TK]D-Fender~[TK]D-Fender
04:11.13jbothmm... [tk]d-fender is rockin' the casbah !!!
04:11.17[TK]D-Fenderhuzzah!
04:11.49wunderkinlock the cat box
04:12.04wunderkin~guis
04:12.05jbotguis are too sexy for [TK]D-Fender's shirt.. too sexy .. it hurts! ~
04:12.09wunderkinsweet
04:13.56rene-he
04:14.00rene-h
04:14.27*** join/#asterisk Splat (n=Splat@220-253-104-247.TAS.netspace.net.au)
04:15.00rene-well lads this has been largely futility but as always one keeps coming for the bot
04:15.42rene-jk
04:15.54*** join/#asterisk |dennis| (n=dennis@200.32.233.82)
04:16.25wunderkin~seen justinu
04:16.28jbotjustinu <n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net> was last seen on IRC in channel #asterisk, 31d 9h 17s ago, saying: 'yeah... they were like half price... so it was $500 instead of $1000'.
04:16.36wunderkinwhere's that boy been
04:17.40[TK]D-Fender~seen justinu_laptop
04:17.42jboti haven't seen 'justinu_laptop', [TK]D-Fender
04:17.53[TK]D-Fenderhe has another ID IIRC
04:18.02wunderkin~seen justinu[laptop]
04:18.03jboti haven't seen 'justinu[laptop]', wunderkin
04:18.09wunderkinshrug
04:18.35[TK]D-Fendersomehting like that
04:19.01[TK]D-Fender~seen justinu|laptop
04:19.03jbotjustinu|laptop <n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net> was last seen on IRC in channel #asterisk, 50d 1h 21m 7s ago, saying: 'bluregard: heh, same here'.
04:19.10[TK]D-Fendereek, even LONGER
04:19.36wunderkinyeah.. :~(
04:21.41rene-mmm
04:21.44rene-SVN-branch-1.4-r50468  still has a broken manager
04:21.47rene-why oh why
04:27.22Corydon76-homeDid you file a bug?
04:30.47CunningPikerene-: Ah - is that why FOP won't worl?
04:30.53CunningPikes/worl/work/
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04:34.18BSDTech?
04:34.25BSDTechI missed some one paging me
04:34.48BSDTechok
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04:54.36wunderkinCunningPike, don't you have polycom phones?
04:54.48CunningPikewunderkin: Indeed we do
04:54.58wunderkinhave an ip430? available now..?
04:55.13CunningPikewunderkin: 'Fraid not - we only have 501s and 601s
04:55.23wunderkinhmmm k
04:55.47wunderkinwish i had the $ to spend on getting a 501 or 601 for home too..
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05:00.47CunningPikewunderkin: Work for someone who uses them ;)
05:01.30wunderkinyeah well thats the only reason i had to spring for an ip430 finally, i had 2 bt101s :P
05:01.41CunningPikeUgh
05:04.52wunderkinbarbietone :D
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05:15.04Qwell~lart wunderkin
05:15.29russellbQwell: i hate moving :(
05:15.32Qwellrussellb: yeah...
05:15.35russellbQwell: but QuickOrbit rocks and is helping me :D
05:15.41Qwellsweet
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05:15.49Qwellsomebody buy that man a beer :D
05:16.00QuickOrbithe is paying me well
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05:16.48russellband by well, he means with nothing
05:17.07QuickOrbitjust wait till you get your credit card statement
05:17.12wunderkinspecial services
05:17.37russellbwunderkin: sssh!
05:18.12*** kick/#asterisk [QuickOrbit!n=russell@asterisk/developer-and-stable-maintainer/drumkilla] by russellb (russellb)
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05:18.51wunderkin;)
05:18.52QuickOrbitthanks bud
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05:19.18russellbbkruse_home is here, everyone stop talking about him
05:19.19Strom_Chi
05:19.26QwellStrom_C: see link above :p
05:19.36Strom_Cand there was that one time where bkruse told me about his---oh ok, whatever, russel
05:20.05Strom_Cthe sprint link?
05:20.08Qwellmmhmm
05:20.16Strom_Cyeah, i glaced at it before i went out
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05:20.36Strom_Chehe
05:20.39QwellQwell: 2, Sprint: 0
05:20.41cbullock81Hello all
05:20.42bkruse_homeok ive given you time to stop :D
05:20.43Strom_Cjust cant stay away
05:20.50Strom_Coh durh, that's you :D
05:20.54bkruse_homeQwell i was reading that in the redbull
05:21.00bkruse_homehi-larious
05:21.05bkruse_homerussellb: hello ^_^
05:21.36russellbbkruse_home: greetings
05:22.17cbullock81I have a question... a very noobish question...  I am trying to understand the differences between lines and call appearances.  I'm just having a hard time understanding everything related to that... anybody have any good resources I could read?
05:22.41bkruse_homecbullock81: asterisk.........the future of IP telephony.
05:22.48bkruse_homenot sure if that will help.......
05:22.49bkruse_home~thebook
05:22.53jbotthebook is probably a book called  Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11
05:22.53cbullock81i read it
05:22.54bkruse_home:P
05:22.58cbullock81i might need to hit it up again
05:23.16cbullock81i setup my 1st polycom phone today
05:23.22Strom_Ccbullock81: well, in a modern telephone system there really isn't any such thing as a "line"
05:23.26[TK]D-Fendercbullock81 : what model?
05:23.31cbullock81IP650
05:23.55bkruse_homerussellb: hows it going? havent talked to you in forever!
05:24.00cbullock81when i have 3 or 4 polycom phones setup... how can the other users tell if all the "lines" are in use
05:24.10russellbbkruse_home: moving ...
05:24.11cbullock81i have 2 wildcard x100p cards in my server
05:25.07[TK]D-Fendercbullock81 : ok you have (w/o expansion) 6 line keys.  each can be for a different registration (ID), and each can handle 1-8 calls depending on your setup
05:25.34cbullock81so its correct that that line key shows my extension?
05:25.52[TK]D-Fendercbullock81 : in what way?
05:26.15cbullock81well like my extension is "199", and thats what it says out to the right of the line button
05:29.34cbullock81am i totally not making any sense?
05:31.16[TK]D-FenderHey, I'm working on a 1.4 system and am getting this message that I can't understand the origin of : [Jan 12 04:27:34] WARNING[25488]: pbx.c:776 _extension_match_core: Wrong usage of [] in the extension
05:31.37[TK]D-FenderThis is after a dial has executed, but not immediately after
05:31.49[TK]D-Fender1.4 FTP release
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05:34.04russellb[TK]D-Fender: what does the pattern look like?
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05:34.22dorphalsigHi?
05:34.35TeneHi!
05:34.49dorphalsigCan anybody help me get an h323 box working with asterisk?
05:34.58dorphalsigI have already oh323
05:35.03dorphalsiginstalled and into asterisk
05:35.50dorphalsigI just cant get the damned box to hit the server
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05:39.00[TK]D-Fenderrussellb : Ahh.. I think I missed that
05:40.02russellb[TK]D-Fender: oh, so the message was right?  missed an ending ] ?
05:40.51russellbthe extension matching code was rewritten for 1.4
05:40.55russellbso just making usre it wasn't broken ...
05:41.03dorphalsighello?
05:41.41[TK]D-Fenderrussellb : I had _[3,4,6,9]11 . No ","s needed.
05:41.59russellbahh, gotcha
05:42.06[TK]D-Fenderrussellb : _[3469]11 should be correct, right?
05:42.16russellbyes
05:42.28[TK]D-Fenderrussellb : So what changed?
05:43.31russellbwell, mostly for performance and code quality in general
05:43.39russellbbut one thing was changed ...
05:43.45bkruse_homerussellb: YAY!
05:43.53bkruse_homerussellb: yay bluemen
05:43.55russellbpreviously, you could have two extensions in a context ...
05:44.03russellb12345 and _1XXXX
05:44.15russellbwhich one do you think would get matched first?  you would think 12345 since it is more exact ...
05:44.41russellbwell, it never actually worked that way, and people had to force sort order by splitting things into contexts and including them in clever order
05:44.49russellbbut now 12345 is actually matched first.
05:45.01bkruse_homewoot
05:45.30[TK]D-Fenderrussellb : Ok, so basically 2 successful matches are scored against each other for "Wildcardness"?
05:45.41Qwellwildcardosity
05:45.49[TK]D-FenderQwell : Duly noted
05:45.56bkruse_homeQwell: your going to court for killing jbot
05:46.11[TK]D-FenderQwell : theres a ..... truthiness about that ;)
05:46.17russellb[TK]D-Fender: sort of
05:46.39russellbthe way it works internally is that they're sorted based on ... wildcardnessosity
05:46.42bkruse_homejason you going to see the bluemen?
05:46.47russellband then to match, it just traverses them until it finds a match
05:47.24[TK]D-Fenderrussellb : so a fixed length match is higher ranked than one with "." for instance?
05:48.00russellbyes, i think so :)
05:48.09[TK]D-Fenderrussellb and a char series map higher than "x" per char?  :)
05:48.22[TK]D-Fenderrussellb : I could devise a sick compound scoring system for this :)
05:48.50russellbexact match should always be highest priority match
05:49.00[TK]D-Fenderrussellb : Very naturally.
05:49.03russellband then it starts matching patterns ... based on pattern type, and pattern length
05:49.52[TK]D-Fenderrussellb : Better than before... I can definately live with this.  to think we could have had this ages ago if we left out AEL ;)
05:50.29russellbheh
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05:56.10bkruse_homerussellb: when you gona be in meh city!
05:56.21bkruse_homeso you can fix my linux + the asterisk kthx
05:56.49russellbsaturday
05:57.10bkruse_home!!!!!!!!!!!!
05:57.26bkruse_homestrait into a house, or apartment, or kp's?
05:57.53sethwhithow do I start Asterisk
05:58.00sethwhitfrom the command line
05:58.08russellbbkruse_home: kp's
05:58.10sethwhitI just need to get the server runnign
05:58.41Strom_Csethwhit: /usr/sbin/asterisk -cvvvvvvg to make sure it starts up cleanly, then "stop now" and then /usr/sbin/safe_asterisk
05:59.04sethwhitwhat if I want to start it at boot
05:59.13Strom_Cdepends on the OS
05:59.18Strom_Cwhat are you running?
05:59.18sethwhitCentOS
05:59.28Strom_C"make config" in the asterisk source directory
06:00.07russellbmake samples ?
06:00.19Qwellinit scripts
06:00.24russellboh
06:00.30sethwhithere is my problem
06:00.36sethwhitI have installed Asterisk on a VPS
06:00.45sethwhitand would like to use Freepbx to manage it
06:00.50bkruse_homerussellb: im going to come over and bring pizza
06:00.55bkruse_homemake samples is broke in trunk
06:01.06russellbbkruse_home: lol ... want to help me unload?  :)
06:01.08sethwhitwhen I try to start wit the amportal script I get
06:01.09sethwhitSTARTING ASTERISK
06:01.09sethwhitCannot find your TTY (9)
06:01.21Strom_Csethwhit: don't waste your time with freepbx
06:01.38sethwhitok
06:01.47Strom_Clearn config files :)
06:01.51sethwhitlol
06:01.52bkruse_homerussellb: ill come over and help you unpack, really
06:01.52sethwhityay
06:01.53sethwhit....
06:01.57bkruse_homeStrom_C: or use the gui
06:02.00russellbbkruse_home: sweet!  :)
06:02.02bkruse_home;]
06:02.11Strom_Cbkruse_home: i havent used the gui enough to know whether to recommend it yet
06:02.14sethwhitgui?
06:03.04russellbsethwhit: asterisknow.org
06:03.29Strom_CI should dick around with asterisknow again
06:03.36bkruse_homesaturday?
06:03.53russellbbkruse_home: yeah, i'm leaving here sometime in the morning
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06:04.57bkruse_homehow far does KP live from dig?
06:04.57bkruse_homethe dig*
06:04.57bkruse_homeStrom_C: i will not comment.
06:04.57bkruse_homelol
06:04.58bkruse_homeit will be really cool, and it looks awesome
06:04.58bkruse_home:P
06:05.03bkruse_homesethwhit: its cool.
06:05.03bkruse_homeStrom_C: eh.....wait for it to be non-beta
06:05.04bkruse_homeits kind of.......well just wait for the non-beta
06:05.06bkruse_homelol
06:05.08bkruse_homerussellb: how far does kpflem live from the atrium?
06:05.19Strom_Cbkruse_home: I cant wait
06:05.44russellbbkruse_home: um, a ways ...
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06:06.07Qwell800km
06:06.11Qwellobviously
06:06.14Strom_C:)
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06:06.19bkruse_homeouch
06:06.56dorphalsigHey, anybody has done anything with H323 here?
06:07.10dorphalsigI'm trying to get an Ericsson switch and asterisk 2 talk to each other
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06:07.14filekp is 800km from teh office
06:07.21filein both distance and time
06:07.21dorphalsigbut I dont understand a damned thing
06:08.14bkruse_home:X
06:08.59conver2in using odbcstorage, is the voicemail recording initially stored in file before being sent to db?
06:09.05Qwell1897110752113   10027110752113
06:09.07Qwellweeeeird
06:09.28QwellI dialed this number three times.  twice, it read those numbers off, and the third...somebody answered
06:10.07Qwellquite bizarre
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06:18.28[F]hey, has anyone here used: http://www.jajah.com/
06:18.43[F]and can tell me if its legit or not? (right now I have asterisk compiling on my freebsd box)
06:18.52Strom_Cdead hookers
06:19.17Tene[F]: a good thing to check is if google knows about anyone complaining about it.
06:20.18[F]hrm, unless I suck at googling, it seems to be something that works.
06:20.42[F]however, would jajah be completely useless (and/or redundant) if you have your own asterisk server?
06:22.48bkruse_home[F]: im checking it out, but yes, because they prolly use asterisk
06:23.02[F]bkruse_home: i just found the site out two seconds ago
06:23.21bkruse_home[F]: thats extremelly to do in asterisk
06:23.24bkruse_homeextremely *
06:23.55[F]if its something thats worth having, just setup an asterisk server so you could use your cell to call your house, and issue a script to launch the website and enter your cell number and the destination's number (which you'd feed to asterisk)
06:24.06[F]bkruse_home: what is extreme(ly) done in asterisk?
06:24.15bkruse_homelol
06:24.16bkruse_hometype
06:24.25bkruse_homes/extremelly/extremely easy/
06:24.37[F]ahhh, what, the service offered by jajah?
06:24.43bkruse_homeyes
06:24.49[F]oh cool.
06:24.55[F]i'm a complete asterisk noob  (its not even done compiling on my box)
06:25.11bkruse_homegl!
06:25.18[F]thanks!
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06:45.25cbullock81has anyone had any experience with SLA in Asterisk 1.4?
06:46.40*** join/#asterisk Aces1Up (n=rich@ip68-96-224-23.lv.lv.cox.net)
06:47.25Aces1Uphell i have been looking around in google the past 20min. anyone know of a good guide on how to set up a remote sip extension in asterisk to allow a remote softphone to connect to my asterisk server?
06:48.16Strom_C~thebook
06:48.19jbot[thebook] a book called  Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11
06:49.44x86Aces1Up: you may also want to check out voip-info.org
06:50.04x86Aces1Up: it's an invaluable reference to everything about asterisk (amoung many other things)
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07:29.42DrCroncan asterisk set up a channel just to a soundcard?
07:29.56naftali5chan_oss.so
07:30.30naftali5http://www.voip-info.org/wiki/view/Setting+up+paging+with+a+sound+card
07:31.11DrCronhmm, just thinking that it would be so nice to have a wider project to add an application to asterisk to do radio routing
07:31.41DrCronreplace this IRLP code disaster
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07:38.15DrCronasterisk doesnt have any support for video at this time, correct?
07:38.24perdpassthrough
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07:46.05naftali5yes
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07:46.49naftali51.4 has h264 passthrough
07:46.55naftali51.2.* can use http://www.asterisk-backports.org/wiki/index.php/Passthrough-h264
07:47.18naftali5may need to be manually applied, though
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08:12.38CunningPikeDrCron: http://www.zapatatelephony.org/app_rpt.html
08:13.02CunningPikeDrCron: What's the 'disaster' with IRLP?
08:13.13CunningPikeDrCron: VA7IRL, by the way
08:13.49DrCronKI6HFB
08:14.06DrCronwell, propriatary protocol, software, and hardware
08:14.10CunningPikeDrCron: Pleased to meet you :)
08:14.11DrCronclosed net
08:14.32DrCronsort of the oposite of asterisk
08:14.34CunningPikeDrCron: Ah, yes - a disaster in general then. I thought there was a particular crisis I hadn't heard about
08:15.01CunningPikeDrCron: Check out the link to Dude's work above - interesting stuff
08:15.04DrCroni was just trying to avoid foul language
08:15.25DrCronapp_rpt? yhea, I looked at that a bit, interesting
08:16.52CunningPikeDrCron: The other thing to check out is BCWARN - http://www.bcwarn.net/
08:17.45CunningPikeDrCron: Link looks busted atm
08:19.07DrCronthe radio interface card is just a bit expensive
08:19.22DrCroni dont see why its much better then a serial port and a sound card
08:19.48CunningPikeDrCron: Not familiar with it - anyway, bedtime here
08:19.53CunningPikeDrCron: Great to chat - 73
08:20.00DrCronhttp://qrvc.com/radiocards.html
08:20.02DrCron73
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08:32.09Dandrehello all,
08:33.03Dandreis there some documentation about the syntax used in http manager interface in asterisk 1.4?
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09:07.21x86new ShellShark site is live guys!
09:07.26x86http://www.shellshark.net/
09:10.28pifwhat's the diff between cdr-custom and cdr-csv ?
09:10.33pifthey look the same
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09:38.49xnonfriends
09:38.53xnongood morning
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09:38.59xnoni got a problem!
09:39.07xnonim try asterisk -vvvvvvvvvvvr
09:39.16xnonbut de console say:  Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?)
09:39.33xnonanybody can  help me
09:39.41james_try it without -r
09:39.48james_that tries to connect to existing process
09:39.53Mavviexnon: well, -r means: connect to a running asterisk server.
09:39.54james_which you obviously dont have running
09:40.14xnonok
09:40.34xnonok i did it, but say the same
09:40.36xnonUnable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?)
09:41.00xnonmy asteriisk are not running i think
09:41.27xnon:S the asterisk service not run i see it
09:41.34xnonwhat can i do?
09:41.47xnonmy asterisk service cant init
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09:43.27xnonummm its strange, because in /var/run/asterisk the file asterisk.ctl its ok!
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09:45.09naftali5tail /var/log/asterisk/full
09:47.51xnonok
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09:51.11xnonok the last line say Loading module chan_zap.so failed!
09:53.09Lokijihello, which codec is the best when i have some lost packet?
09:55.55naftali5xnon, your zapata.conf is probably not set properly or zaptel.conf
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10:13.57stimpiecould somebody tell my how to get the context from which a dialplan application was called?
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10:18.55naftali5http://www.voip-info.org/wiki-Asterisk+variables
10:23.27RoyKmorning
10:25.39stimpienaftali5, so: context = pbx_builtin_getvar_helper(chan, "CONTEXT"); should get the current context?
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10:26.33tparcinaasterisk 1.4, how do I define which processor I use? In asterisk 1.2 I could edit Makefile and specify i586 processor
10:31.10tparcinaanybody?
10:31.19Ahrimanestparcina, I would think autoconf figures it out
10:32.34e-ddieI wouldnt think
10:32.53Ahrimanese-ddie, I know.. you generally dont ;)
10:33.01e-ddieAhrimanes: exactly
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10:35.26tparcinaAhrimanes: it does figure it out - but it figures wrong :-(
10:36.08tparcinae-ddie: do you know how to define?
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10:37.18e-ddiegoogle usually works for me
10:37.33Ahrimanesyup
10:38.29tparcinai'm googling, but I can't find anything...
10:39.18Ahrimaneshttp://hobbes.bsd-dk.dk/~aron/funny_problem.png <- in this call, the caller doesnt get ringing, if i call out from a sip phone on the 1.2.7.1 customer server, i get ringing.. anyone seen this before?
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10:45.38tparcinaAhrimanes: do you have indications.conf?
10:46.22Ahrimanestparcina, hm yes, why?
10:47.44tparcinaAhrimanes: just checking, because that could be the reason...
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10:48.08Ahrimanestparcina, ok, well calls from sip phones to pstn and from pstn to sip phones work fine
10:48.33Ahrimanesbut seems when the 1.0.9 pstn gateway * is involved with both ends of the call, it screws up
10:48.54tparcinachannel, where to define processor on which I'm installing asterisk? on asterisk 1.2 i could modify Makefile and specify i586. what should I do at Asterisk 1.4?
10:49.50tparcinaAhrimanes: sorry, didn't have that kind of problem
10:50.14Ahrimanesok
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11:12.21Ahrimanesanyone here got good quality connection to somalia?
11:12.45e-ddiewhere i come from we have a place we call 'soma lia'
11:12.54e-ddieis that good enough=
11:12.55e-ddie?
11:12.59Ahrimaneshehe sure
11:13.07e-ddiegreat
11:17.05x86guys
11:17.14x86tell me what you think of my new site design
11:17.20x86http://www.shellshark.net/
11:21.09Ahrimanesx86, pretty nice
11:21.25Ahrimanesx86, but get a real ssl certificate?
11:21.36x86lol, it's on the way :)
11:22.16Ahrimanes:)
11:22.33x86layout and design looks nice though?
11:23.02Ahrimanesyep
11:23.43x86you should buy service from us ;)
11:28.28e-ddiehrm
11:29.03e-ddiehow come you have 'free us or canada phone number', and you dont get it with personal-sharkout?
11:29.09e-ddiedoesnt sound free to me :)
11:30.08*** join/#asterisk redax (n=redax@r6.hu)
11:30.09redaxhi
11:30.32redaxTRANSFER is not logged to queue_log,
11:30.43redaxwhat  do I messed here?
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11:36.49flendersdoes anyone have settings for linksys/sipura phones?
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11:49.22redaxflenders: what settings?
11:49.32redaxI've using SPA-942, and SPA-922
11:49.41redaxboth working well. + PAP2
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12:09.25x86e-ddie: it's free with certain plans :)
12:09.51x86e-ddie: SharkOUT is for outbound calling only, it can not recieve phone calls
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12:13.19flendersdamn laptop
12:13.49flendersbatteries could last 10 hours
12:14.41Ahrimaneshehe
12:15.41flendersif anyone replied to my previous question, could you please repost it?
12:16.06Ahrimanesflenders, i have some configs for SPA1001 and PAP2
12:20.31*** join/#asterisk voicetech (n=lotusscr@marnock.com)
12:23.53voicetechCan somebody please help with a caller id problem?  When there is a polarity change I don't get the caller id.  So in effect I only get every other caller id.  This is driving me insane!  I didn't have any problems with the x100p card.  But I do with the Digium 2 FXO Port TDM card.
12:28.41dlynes_laptopvoicetech: make sure your polarity isn't reversed
12:28.49dlynes_laptopvoicetech: after you've checked that, use fxotune to adjust the gains
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12:31.01voicetechI don't know if I can stop the polarity from being reversed.  Our telco can change it at any time.  at the moment its every other call
12:34.17voicetechCan you tell me how I make sure polarity isnt reversed?
12:45.38voicetechcan anyone tell me what this means and how to fix it please : chan_zap.c: Got event 17 (Polarity Reversal)...
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12:46.02voicetechand this: chan_zap.c: CID timed out waiting for ring. Exiting simple switch
12:46.35tparcinahow to define to Asterisk 1.4 that I have i586 processor? On Asterisk 1.2 I could edit Makefile.
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12:52.50delphiuk<PROTECTED>
12:53.07delphiuk1.4 before the GUI thinks you have ananlouge card installed? I have a x101 clone
12:53.27delphiuksorry for the disjointed lines
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13:05.45voicetechanyone help with uk callerid problem?
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13:48.09dhilldoes qualify= cause a lot of load with asterisk?
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13:51.18phatmonkey1.2.13: this is a SIP call
13:51.38phatmonkeywhoops
13:53.26penguinFunkvoicetech: what problem?
13:53.47phatmonkey1.2.13: this is an inbound SIP call, I cannot send audio from the phone to the server. the RTP traffic is getting through, i have checked with tcpdump, but even simple things like echo and voicemail do not receive any audio from the phone. voicemail complains of "Jan 12 13:53:26 WARNING[4932]: app.c:644 ast_play_and_record_full: No audio available on SIP/8201-0819c050??"
13:54.10penguinFunki have yet been able to get callerid working for calls orginating on pstn coming into asterisk out to sip users phones
13:54.27penguinFunk< also in uk
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14:13.35Kattymorning.
14:14.05IPmongergood morning
14:14.08phatmonkeyi've tried with two different clients now, a hard and softphone, with no success
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14:15.32Kattythat's not peachy.
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14:18.22phatmonkeysigh, my mistake, the traffic being received was from the other phone that i was using to test... no traffic gets through with the echo test. back to checking the NAT settings....
14:19.28*** join/#asterisk arcadies (i=arcadies@dsl-243-0-122.telkomadsl.co.za)
14:19.30vt3compared with commercial solutions, how well does asterisk perform conference calls? can it handle say a 100 users well?
14:19.58arcadiesis it possable to add some sort of "gate opening" thing to the pbx?
14:20.20arcadiesyou know, to open gates electronicaly
14:22.10luisjoseWho has set up 2 asterisk server in the same subnet behind a nat?
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14:24.04b11d|bblmorning all
14:24.47[TK]D-FenderKatty: Mew.
14:25.39*** join/#asterisk h0 (n=fakhir@unaffiliated/fakhir)
14:26.01[TK]D-Fenderphatmonkey: Where are your phone and * relative to each other
14:27.52xnondlynes_laptop, are you there friend?
14:28.07xnondr0ne, are you there !
14:31.43docelmoMEW!
14:32.17phearlesshello !
14:32.22arcadiesis it possable to add some sort of "gate opening" thing to the pbx?
14:32.25arcadiesyou know, to open gates electronicaly?
14:32.28*** join/#asterisk wunderkin (i=kev@ip72-208-3-221.ph.ph.cox.net)
14:32.29phearlessI got a question for the experts !
14:32.56phearlesshow can I send TWO commands, one after one, and get the output of each, to the Asterisk Manager ?
14:33.06phearlessit is for a PHP script
14:34.24SheriF_SpacEevening guys
14:34.33b11darcadies.. no doubt it is
14:34.40b11dif you wanted to do it via a serial port or something
14:34.56b11dyou'd need to write a gate controller app, and then just call it from an AGI script, or even system() within extensions.conf
14:35.15olivier__phearless> -->http://voip-info.org/wiki/index.php?page=Asterisk+manager+Example%3a+PHP
14:35.31arcadiesthx b11d ill try that
14:35.35b11dgood luck
14:35.47arcadiesill let you know
14:35.47arcadies:D
14:35.49b11dcool :)
14:35.52phearlessolivier__: I know this page
14:35.56*** join/#asterisk hohum (n=dcorbe@mercury.sunrocket.com)
14:35.59phearlessthey send only one command
14:36.11b11djust add a second then
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14:37.08ctooleyIs there a "!=" not equal in AEL2?
14:37.12olivier__use fputs and fget that'it
14:37.36Dandreis there some documentation about the syntax used in http manager interface in asterisk 1.4?
14:37.45ctooleyBecause I'm getting  "Jan 12 08:17:24 WARNING[20453]: ast_expr2.fl:321 ast_yyerror: ast_yyerror():  syntax error: syntax error, unexpected '!=', expecting $end; Input: != 0"
14:37.59phearlessb11d: it is weird
14:38.03phearlessI have to send : Action: Logoff
14:38.06phearlessto get the output
14:38.09b11dhrm
14:38.13phearlessand when I logoff, I can not send another command
14:38.20b11doh yeah I suppose not then
14:41.39wunderkinyou can send multiple... \r\n\r\n, then the next... if you are doing an originate you need async true
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14:49.27imesperI am trying to connect to ami but it always returns me error Missing action in request, I am using asterisk 1.4, in asterisk 1.2 it works fine
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14:50.48imesperCan someone give me a help?
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14:51.31voicetechne1 have callerid working in uk with tdm card?  care to share your config?
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14:52.53naitramneed dial command that lets me continue dial plan execution if either the caller or callee hang up
14:53.22naitramg option only works if called extension hangs up
14:53.57[TK]D-Fendernaitram: use the "h" standard extension.
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14:55.33naitram[TK]D-Fender: if i need specific execution for each extension, how would that work?
14:55.39*** join/#asterisk pif (n=ldm@zenon.apartia.fr)
14:55.56[TK]D-Fendernaitram: ..HUH?
14:56.20*** join/#asterisk mercestes (n=merceste@rrcs-71-41-157-70.sw.biz.rr.com)
14:56.22pifhi, what is the best ATA to connect to a doorphone (it needs to relay DTMF real good) ?
14:56.57pifI'm tryin with a sipura 3000 without success, won't relay dtmf tones
14:57.03[TK]D-Fenderpif: For high quality, go with MediaTrix
14:57.16pifoki, got a url?
14:57.41naitram[TK]D-Fender: here is what i need to do, i need to execute a php script before the connection is made and after the connection is made, each extension has to provide specific information in both scripts based on its extension, can i do this with the h extension?
14:57.49piffound
14:57.53ManxPowerSometimes I get tired of people blaming Asterisk for echo.
14:58.11[TK]D-Fendernaitram: Who are "each extension"?
14:58.28mercestesManxPower: Especially when they are on a data T1 using direct sip connections to you or another provider.
14:58.42*** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com)
14:58.55[TK]D-FenderManxPower: I sometimes get tired of people in #asterisk echoing .....  Thus there is balance ;)
14:59.19phatmonkey[TK]D-Fender, phone is behind NAT, server has public IP. it's quite odd though - I copied the configuration from a working system in an almost identical situation. i'm guessing it's something up with the firewall configuration on the server or something like that
15:00.00[TK]D-Fenderphatmonkey: Does your phones sip entry have "nat=yes"?
15:00.08*** join/#asterisk alexandrepos (n=alexandr@201.21.143.130)
15:00.15[TK]D-Fenderphatmonkey: And should also be "canreinvite=no"
15:00.22alexandreposanybody can help-me ?
15:00.32phatmonkeyyep, and qualify
15:00.34[TK]D-Fenderalexandrepos: If you actually just ask a question, MAYBE.
15:00.53phatmonkey[TK]D-Fender, canreinvite=no is in general
15:00.54[TK]D-Fenderphatmonkey: Ok, well that covers what you should need from what you described.  Start checking firewalls....
15:01.40alexandrepos[TK]D-Fender: i have two peers in sip, but only second peer receive a incomming call ! i i invert order another work
15:01.59alexandrepos[TK]D-Fender: u have ideia ?
15:02.52[TK]D-Fenderalexandrepos: No enough information yet.  pastebin your SIP configs for the phones, and the CLI output of a failed call.
15:02.54[TK]D-Fender~pb
15:02.57jbotsomebody said pb was a place to paste your stuff without flooding the channel - try  http://pastebin.ca, or http://channels.debian.net/paste
15:03.02ManxPowerCustomer: Sometimes e-mail to me bounces.  Me: Have the sender fax me a copy of the bounce message.  Me: (I get the fax) Tell the sender to spell your damn domain correctly.
15:03.34ManxPowerI am SO glad I'm not a first level support tech anymore.
15:03.44[TK]D-FenderManxPower: or remove spaces (hard to tell sometimes), or... or.....
15:04.18[TK]D-FenderManxPower: "Very few things in the Universe are unlimited.  Amongst them is human stupidity..."
15:04.35[TK]D-FenderKatty: Hugz
15:04.47ManxPower[TK]D-Fender: I suspect this was a 1-time issue.  The REAL issue might be still undiscovered.
15:04.58naitram[TK]D-Fender: each extension is the wrong semantics, I will be more specific, system(phpscriptSetBit 1 1) then Dial(sip/danslaptop)... connection is made, now regardless of who hung up system(phpscriptSetBit 1 0), now the arguments to phpscriptSetBit change based on who is being dialed. This works now, fine, if the called hangs up. But I cant get it to work if the caller hangs up.
15:06.48ManxPowernaitram: see the "g" option to dial.
15:07.10*** part/#asterisk delphiuk (n=richard@212.42.164.14)
15:07.29ManxPowerAsterisk does different things depending on who hangs up.  exten 'h' is for one side hangs up (I don't remember which one) and g option to dial is for when the other side hangs up
15:07.30naitramManxPower: the g option only continues if the called party hangs up, not if the caller hangs up. Right?
15:07.40pif[TK]D-Fender : you using Mediatrix ata's yourself?
15:08.03ManxPowernaitram: you need both "g" and exten "h" if you want to catch both types of hangups.  Yes, I think it is stupid too.
15:08.30coppiceMediatrix - where bugs are a way of life :-)
15:08.54naitramManxPower: if i use the h extension, how do i execute specific behavior for the extension that hung up?
15:10.51pifcoppice : oops, I almost clicked "order"
15:10.55phatmonkeyah, it probably isn't a firewall issue. RTP traffic can reach the server if another server with a public IP sends it, so it's probably some SIP addressing problem
15:11.21pifcoppice : what are your favorite ATA's (it's for a doorphone)
15:11.24ManxPowernaitram: you need to run your script BOTH in the "h" extension AND in the priority after the dial to catch both instances
15:12.24coppicepif: I don't have a favourite, but I have had to work around a number of stupid bug in Mediatrix ATAs. They only seem to work with other boxes because the other boxes have been made to tolerate their crap
15:12.57piflet's say the least problematic
15:13.05pifto your knowledge
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15:14.06naitramManxPower: yeah I got that, but when the h extension determines someone has hung up, is there information available that tells me who has hung up so that I may place the right value in my call to my cleanup script, like System(phpscript var1) where var1 is the extension that hung up
15:14.17phatmonkeydo i need to configure the clients with STUN servers or something?
15:14.24phatmonkeynot really sure what they do
15:15.51[TK]D-Fenderpif: Used a larger gateway once, and a small ATA once.
15:16.04[TK]D-Fenderpif: Somewhat complex, very powerful.
15:16.17ManxPowernaitram: you don't understand.  "h" will only be run if the caller hangs up, not of the callee hangs up
15:16.40ManxPowerso you know that if your agi is run out of exten "h" you KNOW the caller hanged up
15:16.54ManxPowerphatmonkey: I have never ever seen a real need for STUN with Asterisk.
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15:17.24phatmonkeyManxPower, yeah, i have the same setup on another server and I've had no problems
15:17.32phatmonkeyi've no idea why this is causing so much hassle
15:17.45[TK]D-FenderManxPower: Almost ass-backwards isn't it?  having to split your dialplan based on who killed the call.... figure it might eb better to set an env variable like we do for dialstatus + "g"
15:18.11ManxPower[TK]D-Fender: Yes, it is stupid.
15:18.37ManxPowernaitram: why do you need to know who hung up first?
15:18.42ManxPowerEventually both legs of the call will hang up
15:21.32naitramManxPower: I dont really care who hung up, i just need to run the scripts regardless of who hung up. But I need the information specific for the called extension
15:21.55naitramI clearly dont understand how the dial plan is executed
15:22.21naitramwhat is the pastbin used here
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15:23.10naitramill paste my extensions.conf and maybe you guys could tell me where I should put the h extension
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15:24.26ManxPowernaitram: http://pastebin.ca/314207
15:27.11naitramManxPower: here is what i have http://pastebin.ca/314211
15:28.03ManxPowernaitram: my VERY SIMPLE example should help you
15:28.29naitramIll read over it
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15:32.52Kattycan linux mount a tape as if it were mass storage?
15:34.04ManxPowerKatty: yes.  It cannot mount it as a random access device
15:34.29ManxPower"mass storage" is a generic term for ANY large amount of storage.
15:34.47pollerWhere "large" can be anything
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15:37.32sweeperisn't mass storage the generic windows driver/moniker?
15:38.46[TK]D-FenderManxPower: When you say "not random access" do you mean it can't SEEK live, or only that you can't WRITE to it (which is what I assumed already)>
15:39.13penguinFunktape is a FIFO device
15:39.23penguinFunkread it sequentially
15:39.25monstedsweeper: mass storage can be pretty much anything random access
15:39.27penguinFunknot random access
15:40.03ManxPower[TK]D-Fender: I mean that if you want to read the 100'th record on the tape you need to rewind to the beginning, then seek over the first 99 records
15:40.31ManxPoweralso, you want to INSERT anything yo have to rewrite everything after the inserted data
15:41.08penguinFunkcant you just append to the end?
15:41.16[TK]D-FenderManxPower: So just no tracking for "relative" movement then?  So if you have read up to 200, and need to go to 100, you have to back 200, then forward 100?
15:41.20ManxPowerTapes are horribly unreliable and I think they were invented by someone that hates people
15:41.39monstedManxPower: don't confuse shitty tape products with the real thing
15:41.41ManxPowerpenguinFunk: you can append to the end, just not insert into the middle
15:41.51penguinFunklol ManxPower
15:41.56penguinFunkyeh i thought so
15:41.59ManxPowermonsted: mainframe tapes were pretty reliabele
15:42.13monstedManxPower: anything that doesn't suck is very reliable :)
15:42.33ManxPowerI think think the Prime mini tapes (reel to reel) ever lost any of my data
15:42.39monstedManxPower: the storagetek products (9840 and 9940) and LTO are great products
15:43.12[TK]D-FenderMy Certance LTO3 seems to run just fine.... though too damned slow for some reason :/
15:43.32ManxPowermonsted: do the storagetek tape drives last more than WARRENTY + 1 Day?
15:43.42[TK]D-FenderIts 400/800gb size doesn't seem so impressive now that 1TB HD's are coming out :/
15:43.47monstedManxPower: unless they break, yeah :)
15:44.06monstedManxPower: you'd usually have a service contract to go with your $40,000 drive :)
15:44.16ManxPowerMy largest customer has something like 10 tape drives, not one of them works antmore
15:44.28monstedclever :)
15:44.42ManxPowerThey switched to rsync + big ass storage server
15:44.44monstedi think we've got 25 drives now, all of them in perfect working order
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15:45.31Kattyshould i be a draenei or a blood elf?
15:46.11monstedwell, blood elves associate with those smelly cows - i'd vote draenei
15:46.12ManxPowerKatty: carnivorous mushroom is not an option?  That would be cool.
15:46.12[TK]D-FenderKatty: Keebler :)
15:46.40[TK]D-Fender:D
15:46.50Kattymonsted: the draenei do have a pretty cute /train
15:47.30ManxPower8 mins to conference call
15:48.22monstedKatty: haven't seen it, i just can't be arsed to maintain a guild membership on both sides - i'll roll a draenei :)
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15:50.50ManxPowerLooks like I'm going to have to do a Customer Smackdown soon.
15:51.45*** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net)
15:53.40*** part/#asterisk alexandrepos (n=alexandr@201.21.143.130)
15:56.19KattyManxPower: so you can mount a tape like you would a flash drive, dump some files on it using cp in a script, and umount it?
15:56.34ManxPowerKatty: correct.
15:56.39Kattyhot.
15:56.42Kattywhy can't windows do that :<
15:56.48ManxPowerThat would, of course overwrite any files on the tape.
15:56.54Kattymost excellent.
15:56.57ManxPowerKatty: uh, windows can do that
15:57.03ManxPowerOh wait!
15:57.04Kattynot the way i want it to.
15:57.09ManxPowerNo, not MOUNT a tape drive
15:57.10monstedi did that on windows in '96 or so :)
15:57.38Kattyi want the tape to be accessible just like a hard drive would be.
15:57.45ManxPoweryou cp the files to /dev/tapedrivedevice
15:57.51ManxPowerKatty: never gonna happen
15:57.56Katty:<
15:57.57monstedKatty: that'll suck, but it's possible
15:58.01ManxPowerbecause tapes don't have a file system
15:58.48monstedyou just glue a file system on top of it - don't remember the software that does it, but it exists
15:59.05ManxPowermonsted: I would not consider that a workable solution
15:59.10Kattymeh
15:59.18Kattymaybe i'll just rip that tape drive out
15:59.21monstedif you don't mind having a hard drive involved, you could use a HSM solution
15:59.37monstedManxPower: as i said, it'll suck :)
16:01.03monstedtapes good, napster bad!
16:01.11monstedor something
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16:01.21Kattyyou're showing your age dear.
16:02.20monstedhierarical storage management is quite cool - you'll have your most often used/newest files on fast disks, older on slow disks and oldest on tape or optical
16:02.58b11dso, what do you kids think of this "music television" ?
16:03.05monstedand with recovery-friendly tape drives, it'll be quite usable
16:03.28b11di enjoy tape backup
16:03.34b11d400GB LTO's for me..
16:03.41b11dthose are probably small by now though
16:03.49monstedbackup to LTO is great, recovery not so much :)
16:04.00b11di dont use any kind of automated recovery.. so it works for me
16:04.21b11dautomated recovery from tape, that is.
16:04.33Kattyi just mount network drives, dump data, umount
16:04.48b11dwhatever works for you in your situation, thats whats best.
16:04.53cbullock81Hey. I have some confusion about SLA in 1.4.  I've seen the commads and config files in it, but i've read that it's not fully functional? Anybody have any info on this.
16:04.55monstedKatty: that might not work for our 2000ish servers ;)
16:04.56b11dtapes are nice and easy for me to take to the bank on my way home
16:05.03Kattymonsted: not my problem.
16:05.12Kattymonsted: that's your problem - you deal with it (=
16:05.17b11dhe is :)
16:05.26Kattymost excellent.
16:05.52monstedwe shoot it over fibre channel to another site about 5 miles away and onto storagetek 9840 tapes
16:06.21b11dhow much data do you backup?
16:06.22monsteddual-reel things that are incredibly fast at loading/recovering
16:06.39*** join/#asterisk CrashSys (n=kumba@bartleby.crashsys.com)
16:06.41*** join/#asterisk marv[work] (n=timr@24.214.206.254)
16:07.14b11dheh I like that -- storage life: 15-30 years
16:07.16b11dthats a big gap..
16:07.17b11d:)
16:08.34monstedb11d: i don't have any figures, but we've got about 12000 cartridges
16:08.42b11dimpressive
16:09.06b11di always wondered about enterprise level backup systems..  never had the chance to work on gear like that.
16:09.16monsteddual-reel tapes mean very low capacity, though
16:09.23b11dyeah like 20gb/ea or somethign eh
16:09.29b11di take it thats uncompressed too..
16:09.32cbullock81Hey. I have some confusion about SLA in 1.4.  I've seen the commads and config files in it, but i've read that it's not fully functional? Anybody have any info on this.
16:10.13b11dhttp://voip-info.org/wiki/view/Asterisk+SLA
16:10.17b11dthats about it.. i think..
16:10.26cbullock81yea... i read that... not much help tho
16:10.29b11dyeah..
16:10.34b11dyou should ask in #asterisk-dev maybe
16:10.38cbullock81k
16:10.40*** join/#asterisk UlbabraB (n=salama@host241-43-static.72-81-b.business.telecomitalia.it)
16:10.47b11dthose guys are probably going to hate me :)
16:10.55b11di doubt the devs want to be bothered :)
16:10.55b11dhaha
16:12.01monstedb11d: yeah, 20 gig each
16:12.21cbullock81heh... i hate to bother them, but im in the dark
16:12.47b11dwell it says that Asterisk doenst have the support yet.. so thats pretty much it.
16:12.50b11dend of question.
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16:14.58cbullock81they told me that it's being rewritten & to use the existing stuff @ my own risk and peril
16:15.20b11dfair enough
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16:18.55sweepermmm, getting some really awful distortion on the hold music
16:19.58b11dreally
16:19.58b11dKatty was yesterday
16:19.58b11doh no that was meetme..
16:20.06b11dhrm.. are you using mpg123 to play the MoH, or the one included with aster-addons?
16:23.05sweepermpg 123
16:23.07Dandreis there some documentation about the syntax used in http manager interface in asterisk 1.4?
16:23.36sweeperinterestingly, there are about 20 mpg123 processes running....
16:23.54CrashSysIs there an option in the polycom config files to make it remember volume settings? That way everytime someone picks up the handset they dont have to raise the volume?
16:24.24Kattyb11d: it was everything, actually
16:24.33Kattyb11d: but then mysteriously went away, until i registered another SIP phone
16:24.41Kattyb11d: and then it came back for awhile........then went away
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16:31.02jmlshey, I need some help with realtime queues - all the extconfig is set up, the tables have data, asterisk has no queues :(
16:31.13jmlsI was able to set up realtime voicemail with no problem
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16:32.11jmlsbugger. pressed the wrong button ;)
16:35.13[TK]D-FenderCrashSys: Yes, look for the "persist" tag in sip.cfg.  You can make any audio source "sticky" that way
16:35.31b11dweird
16:35.37b11dsweeper.. switch to the aster-addons mp3 player
16:35.45b11dit does a much better job
16:35.49b11di had the same issues with mpg123
16:35.56Qwell[]or just convert your moh to ulaw or something
16:36.02b11dthat didnt work for me
16:36.07b11di had to switch to formatmp3
16:37.08sweepermmm
16:37.13*** join/#asterisk ChicagoBud (n=Bud@adsl-70-228-35-78.dsl.chcgil.ameritech.net)
16:37.35b11d'twas the only way she cleared up..
16:38.09b11dKatty.. that is a bizarre problem.. i hate those sporadic ones.
16:38.39Hmmhesaysyeah sporadic problems suck
16:38.45Hmmhesaysb11d are you enxt to a fax machine?
16:38.48Hmmhesays*next
16:38.54b11di can be
16:39.04b11dright across the hall :p
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16:44.51ManxPowerFor some reason I thought in 1.2 Hangup() allowed you to specify a hangup cause, but "show application hangup" does not indicate this.
16:45.09rene-a gang-bang
16:45.12jmlsif you use realtime queues, show queues says "no queues", but you can use it .
16:45.19b11dhehe
16:45.46Qwell[]show queues only shows loaded queues
16:46.02rene-i think it will show queues after you have used it?
16:46.03Qwell[]you CAN do show queue blah, and it will load it (which means it'll be in show queues the next time you call it)
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16:46.05rene-them
16:46.11emiquelitohello asterisk hackers! When I type 'sip show channels' at the asterisk CLI I can see some channels which are not active anylonger. How can I avoid that or even drop these dead channels?
16:46.32alexandreposwhat a better phone sip or iax2 for kde ?
16:46.34rene-emiquelito: i think if you reload the sip module they will go away
16:47.08emiquelitorene- hmmm, but why it happens? Is there a parameter for this behaviour?
16:47.44wunderkinemiquelito, it will also show qualifies and registrations being sent since they are also calls
16:47.57rene-nice
16:48.14rene-wunderkin: can that behaviour affect call-limit ?
16:48.53wunderkinno
16:50.05rene-wunderkin: i have some dead channels to phones that have no call and then asterisk saying that  the phones are at their call-limit, the phones being members of a dynamic queue but the queue is empty at this time? is there any other reason for those dead channels?
16:50.30a1faDoes anybody know if it will be possible to get a sip client on Apple's iPhone?
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16:50.38Qwell[]a1fa: call them and ask :)
16:50.45Qwell[]ANYTHING we say would be speculation
16:50.52perda1fa just buy a laptop, it's cheaper
16:51.02wunderkinwow
16:51.20perdhey qwell, did you fix skinny yet? :P
16:51.27emiquelitowunderkin I understand. I'm developing a softphone right now and sometimes I have to take a look at asterisk to see how things are going on there. Using 'sip show channels' I found lots of INVITE lines for the same extension. is this correct? my softphone often tries to keep registered in the server... maybe that's the problem
16:51.31wunderkinrene-, i dont know,  you would have to show us what you are looking at
16:51.36Qwell[]perd: You didn't email me the 10 extra hours a day I need :P
16:51.43b11dhaha
16:51.47b11dyeah perd..
16:51.48perdaww i didnt attach that?
16:51.52perddamn me
16:51.52Qwell[]perd: guess not
16:52.01b11dchrist theres a bunch of people waiting for that
16:52.03a1faQwell[] : ring ring
16:52.05b11dand you cant attach 10 hours?
16:52.07b11dgeeze :)
16:52.13a1faperd : how much will that iphone cost ?
16:52.20a1fai've heard $400-$800
16:52.21perdi think it's 400+
16:52.22b11di understand the iPhone is a flop..
16:52.23sweeper500/600
16:52.23perdyeah
16:52.29sweeperfor 4gb/8gb
16:52.33monsteda1fa: well, jobs said it won't run custom apps and i very much doubt it'll have SIP by itself
16:52.47perdtouchscreen lcds suck
16:52.55perdhorrible idea
16:53.00sweepersnark
16:53.13coppiceoptical touchscreen LCDs don't suck
16:53.21sweeperthe lcd itself probably isn't touchscreen
16:53.26perdthey get all greasy
16:53.29monstedi think the iphone will sell well, despite it being teh sux
16:53.33perdyou have to constantly wipe it off
16:53.40perdyou cant feel the buttons so you have to look at the phone to dial it
16:53.56a1famonsted : jobs is a douche
16:53.58b11dyeah..
16:54.01sweepersome people have personal hygene, don't ahve horridly greasy hands, mr baconmitts
16:54.04b11dbut Steve Ballmber is a bigger douche
16:54.06b11dBallmer
16:54.11rene-what is a douche
16:54.11a1faBallsucker
16:54.16a1farene- : i dont know
16:54.17b11dreally?
16:54.18perd<PROTECTED>
16:54.20b11dhahaha
16:54.26rene-the iraqui used to
16:54.27b11da douche is what women use to flush out their vaginas after intercourse
16:54.32a1fai just like how it sounds
16:54.33perdfucking baconmitts hahah
16:54.39rene-jajaaj
16:54.43a1faskwak
16:54.48rene-then jobs is a douche
16:54.52b11dhttp://en.wikipedia.org/wiki/Douche
16:54.54a1falets write an email to jobs tell him he is an asshole
16:54.58a1fahe needs to let us run custom apps
16:55.00a1faWTF!
16:55.05a1fai am sure somebody will hack it
16:55.13b11dyeah you know it
16:55.16rene-well he hasnt opened ichat after so many years and nobody has hacked it
16:55.22b11dyueah well that sucked
16:55.45rene-otoh i think it does jabber now
16:55.50sweeperbecause ichat blows
16:55.58a1fawtf is ichat?
16:55.59a1faiDildo?
16:56.03b11dgarbage
16:56.07a1faiGoatSecX1
16:56.10b11dits what women use to flush out their ears after intercourse
16:56.10sweeperthing is, ipod linux is pretty unusable
16:56.11rene-well the sms thing in the appel phone is pretty much ichat
16:56.13De_Monsounds macish
16:56.41sweeperbut maybe the iphone will inspire better work
16:56.56a1fai like multi-touch idea
16:57.07sweeperor, they'll release some sort of api for the iphone. I mean, they said it supports widgets
16:57.17*** part/#asterisk jmls (n=asterisk@host86-135-41-172.range86-135.btcentralplus.com)
16:57.26a1fahm.. cool
16:57.29a1fawidget sip client
16:57.34rene-yeah but they didnt said third party widgets
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16:57.48a1fai am sure someone will hack it
16:57.51a1faipod was hacked
16:57.55heh_v_waterI doubt the iphone will be called iphone
16:57.59rene-i want someone to hack the apple tv
16:58.05sweeperheh_v_water: I saw that. lol cisco
16:58.11rene-could it be used as a desktop pc?
16:58.15rene-with like linux?
16:58.22sweeperwhy the hell would you?
16:58.25sweeperget a mini :/
16:58.31*** part/#asterisk Bazy (n=bazy@86.125.51.251)
16:58.39heh_v_wateryup mini's rock if you want mac
16:58.47a1famac < gay
16:59.01a1fatransvestite > mac < gay
16:59.07heh_v_waterlol
16:59.18a1fai got rage!
16:59.34rene-swepeer i have a mac
16:59.57rene-mini
17:00.05a1faheh
17:00.21rene-but if i could get a mac for what 250~
17:00.38[TK]D-FenderMac = lack of volume of good free sofware.  On WinXP I have TONS.  On Linux I have TONS.
17:01.07rene-D-Fender: most free software (open source) can be used and compiled in a mac
17:01.17[TK]D-FenderMac's Wifi is RETARDED and I can't get my stupid 40bit WEP working on a PowerBook G4 I'v got here...
17:01.26heh_v_water[TK]D-Fender, agreed BUT.. mac does have actual modern games that can be loaded on and played without too much heartache
17:01.33[TK]D-Fenderrene-: Problem is that not enough of it HAS yet.
17:01.38heh_v_waterI don't use macs myself
17:01.45rene-games are a problem
17:01.52rene-and ubuntu is so cool
17:02.07a1faD-Fender > *
17:02.35[TK]D-Fenderheh_v_water: Yes, different point though.  I'm talking about all the normal productivity stuff.  I haev PDF Creator, FileZilla, Scribus, InkScape, and so much more that runs very well natively on PC, just not for Mac.
17:02.40monsted[TK]D-Fender: you can compile almost anything on osx, just like on linux - what seems to be the problem?
17:03.00rene-pdf creator? just print to pdf,
17:03.05sweeperw00t, asterisk-addons ftw
17:03.08rene-inkscape have it installed
17:03.13CrashSysd-fender: You were saying to look for "Persist" to make the volume controls static?
17:03.14sweeperinkscape \o
17:03.23[TK]D-Fendermonsted: I'm not a coder and don't want to compile basic destop apps.  Sure for *, and a few other specialty servers, but not for joe-blow stuff
17:03.33wunderkinCrashSys, yeah... have you upgraded to 2.0.3 yet?
17:03.37[TK]D-FenderCrashSys: yes
17:03.44a1fain other news, Beckham is coming to U.S.
17:03.51monsted[TK]D-Fender: no different than linux
17:03.54CrashSysD-Fender: Gracias :)
17:03.57a1fagay > beckham < mac < gay
17:04.07monsted[TK]D-Fender: and there are tools to do it for you
17:04.12*** join/#asterisk Nukemizer (n=Nuke@67.137.28.165)
17:04.23[TK]D-Fendermonsted: Hate to say I'm expecting "Click&install).
17:04.34CrashSyswhoooo... gains...
17:04.58*** join/#asterisk bwlang (n=bwlang@bb-66-55-211-238.gwi.net)
17:05.17[TK]D-Fendermonsted: With APT on distro's like Ubuntu, installing is more than easy.  Windows is the usualOkFineSureMinusThatOneSetting standard install procesure.
17:05.22rene-inkscape was click and install for me i mean i cant apt-get it but average joe doesnt apt-get stuff
17:05.27a1faheh
17:05.31heh_v_wateryou would think by now there would be some sort of binary repository of open source software for mac
17:05.32[TK]D-FenderCrashSys: Don't mess with gains unless you HAVE to.
17:05.44a1fai prefer debian over ubuntu
17:05.52[TK]D-FenderCrashSys: Not to be mistake with VOLUMES.  You can cause echo and other shit if you screw up.
17:05.54a1faaka, the real thing
17:05.58rene-yes
17:06.04heh_v_watera1fa, amen
17:06.11rene-i am weary of using ubuntu on servers
17:06.13a1fagod bless you
17:06.14CrashSyswell those gains are actual DB right? not like zapata's gains right?
17:06.21rene-for laptops is teh thing
17:06.21PupenoRwould it cause any problem for queues to be named by numbers only ?
17:06.27[TK]D-Fendera1fa: Yeah I probably would too if they made an equally convenient installer disc.
17:06.43a1fa[TK]D-Fender : i use netinstall images
17:06.54a1fanetinstall in expert mode
17:07.03[TK]D-FenderCrashSys: I don't know that partuiculars, just set persist and adjust at source.  You should never have to mess with "raw" gains on SIP phones.
17:07.07*** join/#asterisk CunningPike (n=CunningP@dhcp-10-234.district.north-van.bc.ca)
17:07.20heh_v_water[TK]D-Fender, you may get your wish in the next few months.. when Debian Etch gets released there is going to be all kinds of new debian disks available including a live cd and such
17:07.26[TK]D-Fendera1fa: NO!  I want the basics (even old), to come off CD and from there let me choose what to do.
17:07.41a1faaha
17:07.44CrashSysI like my Slack
17:07.46CrashSys1-cd install
17:07.53CrashSysunless you want X or other stuff
17:07.55a1faCrashSys : i like my debian netinstall - 50mb
17:07.55[TK]D-Fendera1fa: I don't want to spend lots of time waiting for crap to download and install (or worse, compile like Gentoo..... friggen RICERS!)
17:08.01heh_v_waterI install debian with a 50MB business card cd.. :P
17:08.11[TK]D-FenderCrashSys: Agreed, but I don't like it of standard desktop use.
17:08.18a1fa[TK]D-Fender : i know what you mean.. gentoo is for S&M people
17:08.24a1fafucking masochists
17:08.30CrashSysI dont use linux as a desktop... it has a hard time playing Asheron's Call :D
17:08.58*** join/#asterisk h0 (n=fakhir@unaffiliated/fakhir)
17:09.14CrashSysPlus there's no real good source of video editing software for it, short of writing your own...
17:09.21heh_v_waterit took me years to finally stop using Windows as my primary desktop machine.. now I use debian almost exclusively.. when i use windows I yearn for ym debian
17:09.31penguinFunkcallers complain that they find it difficult to hear people in this office because the outgoing voice is a bit too quiet, people in the office can hear the callers perfectly fine tho. How can i increase the outgoing volume just a little bit ?
17:09.34a1faAMEN BROTHER!
17:09.36monstedit's too bad linux is so badly made
17:09.38BSDTechfreebsd
17:09.47Qwell[]badly made?
17:09.51BSDTecheasy fast clean and asterisk is in ports
17:09.52heh_v_waterlol
17:09.52a1famonsted : lol?
17:09.53sweeperpenguinFunk: rxgain/txgain in the channel conf
17:10.02a1fa~lart monsted
17:10.04CrashSysI like my OpenBSD!
17:10.05monstedif only people had thrown their support behind something that works nicely like freebsd, things would've been much better
17:10.09penguinFunkthanks sweeper
17:10.10CrashSysBut alas, no zaptel...
17:10.24CrashSysI'm hoping for a FreePBX/LibSangoma solution some time soon :)
17:10.27BSDTechand asterisk runs great
17:10.31sweeperOpenBSD is for hippy beer makers with attitude problem
17:10.33sweeper*s
17:10.35a1faCrashSys : i like openbsd as well.. reminds me of debian in early days
17:10.43sweeperI'M LOOKING AT YOU THEO
17:10.45CrashSysI make beer
17:11.01CrashSyswell, drink it...
17:11.03a1faDo you have any mexican in you CrashSys ?
17:11.09penguinFunkand by channel conf you mean zaptel.conf ?
17:11.12BSDTechmove asterisk to freebsd and you have a stable pbx
17:11.13CrashSysNo, but i've been in a mexican before...
17:11.15CrashSysdoes that count?
17:11.27a1faIn mexican accent, so would you like some, esse?
17:11.45BSDTechKpasa
17:11.45sweeperCrashSys: nobody wants to hear about your gay love affair with the delivery boy :v
17:12.01BSDTechcom-esta
17:12.02CrashSysawww, but he was so cute...
17:12.05sweeperbesides, we all know you're the catcher >.>
17:12.14CrashSysNow that's just rude...
17:12.16BSDTech).(
17:12.21BSDTechlol
17:12.28monstedQwell[]: please convince your digium colleagues to move primary development from linux to freebsd ;)
17:12.43Qwell[]freebsd is crap
17:12.45Qwell[]sorry
17:12.45fileha
17:12.52BSDTechI have been trying for 3 years to get them to move
17:12.55BSDTechthey wont
17:13.01[TK]D-FenderCrashSys: You looking for an automatic way to completely configure Sangoma's, or just get it working?  Latter should be aeasy
17:13.02BSDTechMark is to head strong
17:13.18CrashSysd-fender: eh?
17:13.34CrashSysbsd: You use sangoma for * on BSD?
17:13.34[TK]D-FenderAs Nugget usually says "Linux is poo"
17:13.40BSDTechif you use the sangoma rpb and its a a200 card there is a setup script
17:13.47BSDTechyes
17:13.56BSDTecha100 and a200
17:13.59sweeperQwell so mean D:
17:14.00rene-BSDTech: linux+asterisk doesnt crash for me, when you say stable you mean less line errors or less calls dropped for you?
17:14.29sweeperloooonix
17:14.32BSDTechall the above freebsd overall has been stable for me with sangoma cards and asterisk
17:14.41a1fa* 1.4 is most stable so far
17:14.48a1faDTMF works perfectly
17:14.50BSDTechnot
17:14.54rene-yeah sangoma
17:14.56CrashSys1.2.14 has been pretty stable for me so far
17:15.03CrashSysless "strange" things happening...
17:15.05BSDTechit has major transcoding issues
17:15.21rene-1.4 has broken transfers  for me
17:15.26BSDTechthat also
17:15.28rene-so broken it takes asterisk down
17:15.31sweeperI'd like to tune it to a streaming radio, but looks like a hassle :v
17:15.45BSDTechnope just install madplayer
17:16.10BSDTechand then setup madplayer in moh.conf andpoint it at the stream you want
17:16.16sweepermadplayer bitches about a lack of soundcard
17:16.27sweeperand then dies
17:16.43BSDTechnot on bsd
17:16.47rene-i have seen some roomba clones for like 30~ i wonder if those work well or what
17:16.47sweeperwell, lack of /dev/esd, to be specific
17:17.07wunderkinCrashSys, have you updated to 2.0.3?
17:17.08sweeperoops
17:17.10sweeper/dev/dsp
17:17.37sweeperfucking gentoo
17:17.37rene-BSDTect: not on bsd was you talkin to me?
17:17.38BSDTechmadplay works fin on my server and it has no sound card
17:17.52sweeperstop giving bsd a bad name :/
17:18.24BSDTechgentoo-bsd
17:18.26BSDTechlol
17:18.53luisjoseBSDTech, my asterisk crash when a linphone client try to log in
17:18.56luisjoseBSDTech, seg fault
17:19.19sweeperluisjose: awsome
17:19.25BSDTechwow
17:19.32BSDTechare you on linux or bsd
17:19.37luisjoseBSD
17:19.44BSDTechwhat ver of asterisk
17:20.01cbullock81i got a newbie question. how do you guys deal with letting the phones (or users) know that all outbound lines are busy before they try a new call?
17:20.14sweepersee, I realized why you * zealots like to compile from source...very few dependancies :P
17:20.19*** join/#asterisk Assid (i=assid@59.183.52.135)
17:20.26BSDTechyoucansetup a roll over
17:20.40BSDTechand have it tell the user that the lines are busy
17:20.56BSDTechluis what ver of asterisk
17:21.19cbullock81could you expound on that a little... i am totally new
17:21.42BSDTechis this for sip or zaptel setup
17:21.46cbullock81zap
17:21.53[TK]D-Fendercbullock81: clarify the "BEFORE they try a NEW call" part please...
17:22.56cbullock81like if i have 2 zap channels outbound & both are busy. When i pickup and try to dial an outbound line, is there a way to notify me that the lines are busy
17:23.46BSDTechhe wants a check stataus
17:23.50BSDTechsetup
17:24.26BSDTechwhere he dials the number if line 1 and line 2 are busy it then says all outbound circuts are busy please try again ltr
17:24.27luisjosesweeper, I did compile asterisk from source.
17:24.40BSDTechlouis you did not answer me
17:24.44BSDTechwhat ver of asterisk
17:24.55luisjoseBSDTech, oh sorry
17:24.56cbullock811.4
17:25.02[TK]D-Fendercbullock81: Yes, but not BEFORE they dial....
17:25.07luisjoseAsterisk 1.2.13
17:25.35cbullock81well, i would love for the phones to indicate the lines are busy before even trying an outbound call, but i dont know if asterisk supports that
17:25.37BSDTechdo a update on your ports tree I believe it should be 1.2.14 now
17:25.46[TK]D-Fendercbullock81: They dial, * reports back "CONGENTION" in DIALSTATUS and then you play a sound file saying "try again later please".  Thats how
17:25.48BSDTechand I believe there is some fixes
17:26.19wunderkinwell, anyone here with an ip430?
17:27.03a1faWhen is the new version planed for release?
17:27.12[TK]D-Fenderwunderkin: You mean to do more tests than we've already gone through?
17:27.35luisjoseBSDTech, oh ok, wondering when 1.4 is going to be on ports
17:27.50BSDTech1.4 is a bug release it was ment to bug the hell out of you
17:27.57BSDTechwhen it is fixed
17:28.11a1falike i said, i had no problems, except pitch changer wont compile
17:28.17luisjoseoh i thought it was released already.
17:28.18wunderkinyeah, i just wanted to get a couple other people to test it, either its this batch of hardware in the phones or my config, but they say to rma but i dont know if that was from polycom or not
17:28.21a1fajustin is releasing new patch next week
17:28.33BSDTechtry transcoding g729 to ulaw or gsm
17:28.38BSDTech<PROTECTED>
17:28.47a1fanah :)
17:28.50a1fai keep it simple
17:28.53BSDTechtry g726 to ulaw it crashes
17:28.55a1faulaw across the board
17:29.03Qwell[]BSDTech: try upgrading to 1.4 svn
17:29.13BSDTechok will do
17:29.20luisjoseHey fellas, for example, which hardware you guys will use to set up lets say an hotel with 200 extensions?
17:29.27luisjoseFXS hardware.
17:29.31Qwell[]luisjose: analog?
17:29.39Qwell[]some quad T1 cards and channel banks, probably
17:29.55wunderkinanyone that tells you to use sip phones in the rooms are freaking high
17:30.05luisjoseQwell[], yes.
17:30.10a1fawunderkin : unless its a high class hotel
17:30.17a1fai've been in hotel rooms with cisco phones
17:30.26BSDTechwhy you can get grandstream 200's for 48 bucks each
17:30.37BSDTechand they work
17:30.39wunderkinive never been in a 'high class' hotel before, 2 star for me baby
17:30.48a1fawunderkin : cheap ass
17:30.50BSDTechbut soon hotel rooms wont have phones
17:31.03a1faI dont stay in a hotel unless its 5 * or higher
17:31.15luisjosea1fa, bastard
17:31.18a1faBSDTech : comparing to $5 for ordinary hotel phones
17:31.23*** join/#asterisk [hC] (n=hardcore@S0106000fb51cc225.vf.shawcable.net)
17:31.23a1faluisjose : i was kidding
17:31.28BSDTechhotels are over rated impose on friends its more fun
17:31.31luisjoseOk sorry then.
17:31.34a1fayeah
17:31.38a1fai impose on my friends most of the time
17:31.39Qwell[]5 star hotels are for nubs
17:31.42Qwell[]I stay in 7 star hotels
17:31.47a1faI pay in beer
17:31.50Qwell[]of course, there is only...1, but what the hell
17:31.57luisjoseOk so, how are these channel bank?
17:32.01a1fa24 pack of Patt's Blue Ribbon
17:32.05luisjosenever seen one on sites or anywhere
17:32.07BSDTechlol
17:32.08Qwell[]luisjose: They connect to a T1 card, and split out the signalling
17:32.11BSDTechand pizza
17:32.13luisjosehow many fxs ports they have?
17:32.14Qwell[]to fxs/fxo ports
17:32.19Qwell[]24 or 32, depending
17:32.21a1fa24
17:32.21BSDTechand the guest room is always open
17:32.26Qwell[]24 or 31?  dunno
17:32.33BSDTechok 200 rooms
17:32.35wunderkin42
17:32.36a1fai thought T1 had 21 channels
17:32.41Qwell[]24 channels
17:32.49a1fayeah.. 24 channels
17:32.50a1fasorry
17:32.56luisjoseso if i need 200 extensions?
17:32.57BSDTech23 voice and 1 data
17:33.03Qwell[]luisjose: you just get a bunch of them
17:33.07Qwell[]or a higher density channelbank
17:33.07wunderkinBSDTech, thats for pri
17:33.08BSDTech23b and 1 d
17:33.09danpthat's PRI
17:33.12BSDTechyes
17:33.25Qwell[]I think there are dual and maybe quad T1 channelbanks out there
17:33.27luisjosea t1 card for each 24 ext?
17:33.31Qwell[]with 48 and 96 channels
17:33.35Qwell[]luisjose: correct
17:33.45luisjosethats a waste, they wont be using all ext at same time.
17:33.47BSDTecha101 from sangoma
17:33.53BSDTechis a quad card
17:33.54Qwell[]luisjose: but you must split them out
17:34.02wunderkinuse monkeys
17:34.07Qwell[]otherwise, when two people pick up a phone sharing a port...they can hear each other
17:34.08danpjust have rooms share lines...that should be fine
17:34.14Qwell[]and if somebody calls them...BOTH rooms ring
17:34.16Qwell[]that is also bad
17:34.32BSDTechI do hotels
17:34.34Qwell[]you don't actually need T1s to the telco for as many rooms as you have
17:34.40luisjoseQwell[], in what kind of computer im gonna hook 8 t1 cards?
17:34.40BSDTechI have done 24 dyas in hotels
17:34.51BSDTechdays inn that iis
17:34.53monstedluisjose: if you're using channel banks, each T1 channel corresponds to an analogue phone - you can still oversubscribe the outgoing T1 line
17:34.54Qwell[]luisjose: 2 quad span cards in 1 machine isn't unreasonable
17:35.04a1fayeah
17:35.05BSDTechand we use a channel bank
17:35.07a1fayou can run 8 fine
17:35.10[TK]D-FenderBSDTech: You mean A104.....
17:35.13monstedhell, some servers actually have 8 pci slots :)
17:35.14a1fadual opteron
17:35.17BSDTechsorry yes
17:35.19BSDTechthe 104
17:35.21a1fa16 GB of ram
17:35.23Qwell[][TK]D-Fender: you mean TE410p ;)
17:35.27BSDTechbrain a little fried
17:35.36BSDTechbeen up for 36 straight
17:35.39luisjoseOk wait wait
17:35.51[TK]D-FenderQwell[]: Of all Digium models, that is the LAST one I'd choose :)
17:35.58luisjose<monsted> luisjose: if you're using channel banks, each T1 channel corresponds to an analogue phone - you can still oversubscribe the outgoing T1 line <- i dont get it
17:35.59BSDTechget a rhino channel bank
17:36.00Qwell[]okay, TE412p
17:36.09Qwell[]but, for a channelbank...really, an echo can?  nah
17:36.43Qwell[]luisjose: For 200 users, you'd have something like maybe 2-3 T1s to the telco
17:36.45tzangerQwell[]: you need 'em
17:36.55tzangerQwell[]: or rather CAN need 'em
17:36.57Qwell[]tzanger: I stand corrected then.  I've never done fxs :)
17:37.01tzangerhell I need an echocan for my PRI
17:37.03Qwell[]not like that anyhow
17:37.09BSDTechdepends on how many outbound /inbound lines the hotel needs/wants
17:37.39tzangerechocan for fxs?  not really needed for most installs I've seen
17:37.39tzangermy mistake
17:37.48tzangerFXO needs echocan for sure (usually sw works fine but really the TE407 is amazing, I'm VERY happy)
17:38.02wunderkinluisjose, think of the channel bank as a switch i guess
17:38.05tzangerFXS needs for ecohcan are much less stiff, but I could forsee it
17:38.13luisjoseOk I chosed a bad example. no hotel, corporation with a lot of inside traffict not outgoing calls
17:38.20luisjosejust on internal extensions
17:38.29luisjosecorporation
17:38.32BSDTechthen Polycom
17:38.40luisjosebut with fxs
17:38.44luisjoseanalog
17:38.46BSDTechdepends on how many line buttons you want
17:38.58luisjosedamnit.
17:39.00BSDTechthen a channelbank
17:39.05luisjosechannel bank
17:39.18BSDTechlook at rhino
17:39.19luisjoseok so how many analog phone can I hook to a channel bank?
17:39.33danp24, same as in a hotel :P
17:39.38BSDTechdepends on the channel bank you get
17:39.45Qwell[]luisjose: 24 if it's a single span channelbank
17:39.53Qwell[]I'm fairly certain that dual exists, and likely quad
17:40.05monstedhmm
17:40.06luisjoseso i must die with a line per channel.
17:40.09Qwell[]and hell, you could probably find an 8 span if you look hard enough
17:40.15rene-asterisk seems
17:40.20Qwell[]luisjose: for the phones, yes, BUT, NOT for the "outgoing" lines
17:40.23monstedany reason not to buy a european channel bank and get 30 channels for the price of 23? :)
17:40.24Qwell[]ie; the T1's to the telco
17:40.41luisjoseit does not make sense to me, what about i get ip phones and connect it to my lan
17:40.49luisjoseon a giga ethernet
17:40.49Qwell[]you only really need a 1/4 or 1/3 ratio or something.  That's something you'd have to just estimate at first, then expand later
17:40.52rene-i downloaded trunk and manager seems still broken to me
17:40.55Qwell[]You could, sure, BUT...
17:41.00BSDTechthen get polycoms
17:41.02rene-not able to login
17:41.05Qwell[]IP phones are much easier/more valuable to steal
17:41.10[TK]D-Fendertzanger: You said Sangoma's have failed for you, have you tried Digiums new Otasic-powered lineup?
17:41.17rene-yeah but generally
17:41.23Qwell[]luisjose: If you aren't worried about theft, then absolutely, go with IP phones
17:41.24luisjoseyes but why you can apply that concept to fxs lines
17:41.30BSDTechI have yet to have a sangoma fail me
17:41.38Qwell[]nobody is gonna steal a $20 analog phone, heh
17:41.42BSDTechI have had 4 digium cards go south on me
17:41.48Qwell[]but, if a polycom is sitting in there...yeah, it's more likely
17:42.09[TK]D-FenderBSDTech: I know, tzanger is the only case I've heard, and he's knowledgeable and has work with Sangoma support for a LONG time working on it.
17:42.34tzanger[TK]D-Fender: as I said, TE407 kicks serious ass
17:42.36luisjoseOk but i think you guys dont get my point
17:42.50Qwell[]luisjose: maybe not - explain it
17:42.51luisjosewhy i dont need to use a 64kbps per channel
17:42.52[TK]D-FenderQwell[]: Thats why I bought 2 x Uniden UIP-200's for high damage/theft risk areas :)  Phones that can take a beating, and I could afford to lose
17:42.55luisjosei mean
17:43.02BSDTechhmmmm
17:43.05luisjosei can hook as many phone as my bw allow me
17:43.05BSDTechweird
17:43.05Qwell[][TK]D-Fender: those SIP or something?
17:43.08BSDTechwhat card
17:43.15[TK]D-Fendertzanger: The good way, right?  So why do you need add'l echo-can?
17:43.17tzanger[TK]D-Fender: Sangoma I am not 100% failed me.  It took fucking FOREVER to get it replaced, and then I specifically asked for a failure report, to know what exactly went wrong, and they coughed and replied that they didn't test it, they just sent me a new one
17:43.27luisjosemaybe if i have low traffic i can hook 500 ip phones
17:43.28[TK]D-FenderQwell[]: Yup.. They're CRAPTASTIC!
17:43.33tzanger[TK]D-Fender: TE407 has the octasic echo can, that's what I'm talking about
17:43.40BSDTechyou can also get the cheap grandstream 200's and no one will steal them
17:43.47BSDTechand they are single line phones
17:43.49tzanger[TK]D-Fender: I *have* echo can, in the form of the octasic echo can in the TE407
17:43.51luisjosein other setup ill be only able to use 50 phones but!
17:44.06luisjosewhy i have to waste 2 T1 on a setup with a low traffic on FXS line.
17:44.09[TK]D-Fendertzanger: Thought you just asked about getting one for PRI though... whats that about then?
17:44.11Qwell[]luisjose: Those are the tradeoffs
17:44.14BSDTechwell wait
17:44.18luisjosetrying to get stuff cheap.
17:44.19*** join/#asterisk shepimport (n=shep@h194.189.31.71.ip.alltel.net)
17:44.23Qwell[]luisjose: like I said, one of the most important factors (in a hotel env) is theft
17:44.29BSDTechthen go grandstream
17:44.38Qwell[]If you can afford to lose a $100 IP phone every couple days...then go for it
17:44.38luisjosegah
17:44.45tzangerI had TE405 on this PRI before, and had echo problems.  I had TE406 on this PRI before, again with echo issues (the 406 is TRASH).  A104d worked well but had audio artifacting they could NOT solve, and I haven't tried the replacement A104d yet, since the TE407 is working 100%
17:44.47[TK]D-FenderBSDTech: like Andrew Dice Clay said "Some people play hard to get.... I play hard to WANT! heeeyyyyaaa!"
17:44.51b11dsweeper.. did you switch to format_mp3?
17:44.51luisjoseyou can get a POTS for $9 :P
17:44.54monstedQwell[]: you'd bill the customer if the phone has gone missing
17:44.57BSDTechlol
17:45.10Qwell[]monsted: good luck A) proving it, B) getting the money (in a timely matter)
17:45.11monstedQwell[]: and as a bonus, you can check logs to see when the phone went offline
17:45.12BSDTechTK your to much
17:45.15BSDTechlol
17:45.22[TK]D-Fendertzanger: Not any clearer.... whats this about a CURRENT need for a PRI EC?
17:45.23BSDTechLOUIS
17:45.23Qwell[]A is easiest, but...yeah
17:45.27BSDTechget a grip
17:45.28luisjoseso as conclusion, I need a channel bank and a t1 every 24 analog phones?
17:45.30BSDTechand listen
17:45.40Qwell[]luisjose: If you use analog phones, yes
17:45.43tzanger[TK]D-Fender: heh.  I have a PRI and it needs EC.  The TE407 is my EC for that PRI that needs it.
17:45.51monstedluisjose: you're building from scratch?
17:45.58luisjosemonsted, yes
17:45.58shepimporthey all... anyone have asterisk w/ a session border controller experience?
17:46.07[TK]D-Fendertzanger: As in you'd like to take the TE407P *out* of the equation?
17:46.09luisjosemonsted, well looking forward to.
17:46.10monstedluisjose: it'd be insane to buy analogue stuff then :)
17:46.28luisjoseanalog phone goes from $9...
17:46.33BSDTechtzangert sangoma card did you have
17:46.36luisjoseip phones at least $40
17:46.46monstedchannel banks and T1 cards are far from free
17:47.18a1fajust use ip phones
17:47.24a1fasee how many get stolen
17:47.28tzanger[TK]D-Fender: no no no no no
17:47.30monsteddefinitely go for IP phones
17:47.31[TK]D-Fenderluisjose: tzanger: Perhaps I misinterpreted your conversations starting line : <tzanger>hell I need an echocan for my PRI
17:47.34a1fain frist week of operation
17:47.48a1famake sure you use budgetcrap phone
17:48.00a1faand dont take cash for rooms.. take visa/mc
17:48.08a1faso you can charge them if they steal your sip phone
17:48.10tzanger[TK]D-Fender: I do not want to take the TE407 out.  I love it, it is working 100%
17:48.19*** join/#asterisk alamantia (i=Anthony@nat/digium/x-588291bb6c886e47)
17:48.21tzanger[TK]D-Fender: I was just saying "I have a PRI that needs echo cancellation"
17:48.29`Seana1fa
17:48.32a1faYou can get a big discount if you buy 200 IP phones at once
17:48.35`Seanthen they can just dispute the charge
17:48.40[TK]D-Fenderluisjose: Channel banks mean you spend a lot of money on a T1 card, the channel bank itsel, phones, power bars to support the power bricks for the phones, and then the sucky interfacs that is Zaptel FXS :(
17:48.41a1fafor like $20 per phone
17:48.57rene-look for hospitality ip phones
17:49.01luisjoseill just embed  a pc into the wall and use softphones
17:49.05rene-they are expoensive but they can take a beating
17:49.11[TK]D-Fendertzanger: Yeah, that to me says I have a need... no that you FILLED it :)  Clear now!
17:49.12monstedi'm hoping we land a deal that involves us getting 35000 phones :)
17:49.20a1faluisjose : i can get you 200 ip phones for $20+ or so
17:49.22BSDTechcall grandstream and strike up a deal
17:49.23rene-35k
17:49.26rene-wheeew
17:49.42a1faluisjose : i will cut you a good deal on ip phones
17:49.45[TK]D-FenderGrandSuck.... *shudder*
17:49.47sweeperb11d: yes I did, works awsomely now~
17:49.48BSDTechthe issue you will have is a operator console
17:49.51a1fai take paypal ;)
17:49.55rene-a1fa
17:49.58rene-what model
17:49.59rene-make
17:49.59luisjosea1fa, good to know.
17:50.06luisjosea1fa, I'm in Venezuela.
17:50.07rene-have pictures and specs?
17:50.12a1faok no free shipping
17:50.16a1famama vuevo
17:50.17luisjoselol
17:50.25luisjose8==D
17:50.28a1faBT102
17:50.28BSDTechI take paypal/visa/amex/discover/diners/left nut/first born
17:50.33sweeperit's huevo
17:50.33*** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal)
17:50.33*** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal)
17:50.40luisjoseactualy is guevo :P
17:50.45luisjoseactually*
17:50.50sweepermmm
17:50.59sweeperI'm pretty sure it's huevo
17:51.03luisjosesweeper, on swearing context
17:51.20sweeperecodeMP3: Junk at the beginning of frame e108bf2a
17:51.34a1faok 200 phones for $14k?
17:51.34sweeperwell, you're still saying "egg sucker"
17:51.37a1fasounds good?
17:51.45sweeperoops, damned paste buffer
17:51.48a1fafree shipping
17:52.49a1fa200 BT101 for $14k + additional $1k for a configuration TFTP server
17:52.49luisjosesweeper, yap but you want to saw "di** sucker"
17:52.54luisjosesay&**
17:53.01BSDTechok the best setup for a hotel is the existing punchdown block and the sangoma a400
17:53.11BSDTechyou can have upto 4 in a systemn
17:53.30`Seani want a 7985 :P
17:53.31`Seanfrom cisco
17:53.32`Seanlol
17:53.34`Sean:D
17:53.36sweeperwtf
17:53.41Qwell[]`Sean: buy me one, and I'll get it working with chan_skinny
17:53.46BSDTechsangoma a400
17:53.51`Seanlol qwell
17:53.54[TK]D-FenderBSDTech: EW.....
17:53.54sweeperweird venezuelan slang :P
17:53.58`Seanwhy doesn't digium buy them for you :P?
17:54.01Qwell[]`Sean: That...wasn't a joke
17:54.03ChicagoBudBDSTech: are the grandstream 200's decent office phones?  How is the speakerphone?
17:54.06`Seandigium will never cut support, for cisco
17:54.26sweeperwoohoo, 66black
17:54.26[TK]D-FenderBSDTech: thats a horrid setup... SIP gateways!
17:54.28sweeper*black
17:54.30sweeper*BLOCK
17:54.41`SeanQwell any idea how much they are?
17:54.44ManxPowera1fa: 200 BT101s don't actually cose $14k.  They would cost you your job.
17:54.49BSDTechhttp://sangoma.com/datasheets/p_a400-specs
17:54.50Qwell[]`Sean: a lot :P
17:54.53Qwell[]> $2k
17:54.55`Seanlol
17:54.56shepimporthey--- anyone interested in helping me with some signalling issues?
17:55.02luisjoseBSDTech, punchdown?
17:55.02sweeperwell, depends. if you've got decent phones already, no point in replacing everything
17:55.03ManxPowerYou know that the BT101s can't even display non-numbers, right?
17:55.05Qwell[]MUCH >
17:55.05`Seanya i see it
17:55.08[TK]D-FenderManxPower: LOL.... don't get "smart" here.... its over most of their heads :)
17:55.27*** join/#asterisk sahafeez (n=sahafeez@67.109.14.227.ptr.us.xo.net)
17:55.31sweeperif you're building new, then yea, run at least two lines of c5 to every room
17:55.36BSDTechits where the lines from the rooms meetup in the room with the pbx
17:55.39sweepergigabit switch, the whole 9 yards :)
17:55.44ManxPowerYou have to press SEND or wait for the dialing to timeout on the BT101s, no dial pattern matching
17:55.50Qwell[]eww
17:55.53Qwell[]ManxPower: really?
17:55.53[TK]D-FenderBSDTech: See my not on Zaptel FXS.  Poor deployment methodology.
17:55.55`SeanQwell i dont think they have good encoding for the video, because it'd use alot of bandwidth
17:56.06Qwell[]`Sean: probably h263 or whatever
17:56.15a1faManxPower ?
17:56.17ManxPowerQwell[]: yes, really
17:56.18`Seanhrmp yea probably
17:56.23[TK]D-Fendersweeper: That'd have to be Cat5e then.
17:56.28Qwell[]cat6
17:56.32Qwell[]ftw
17:56.42Qwell[]Run your TV over it also
17:56.52Qwell[]If you've got money to burn, might as well go all out
17:56.58ManxPowerwe will be putting in our first Gig Fiber link soon
17:57.09sweeper[TK]D-Fender: hard to find non-e c5 these days :P
17:57.12Qwell[]put a media system in every room, streaming on-demand video
17:57.27sweeperapple tv :D
17:57.32BSDTechok
17:57.39Qwell[]get one of those stupid N64 controller remotes that nobody ever uses
17:57.39BSDTechTK looking
17:57.46sweeperhahaha
17:57.50luisjoseBSDTech, where do you hook the pots on that card?
17:57.54Qwell[]seriously, who uses those?
17:58.02ManxPowerinstalling Grandstream stuff is a good way to get fired.
17:58.12sweeperQwell[]: I have, when company was footing the bill, and I was bored
17:58.39BSDTechthere is a wireing diagram for it
17:58.46BSDTechthat shows how
17:59.06sweeperchaaaaaannel bank
17:59.09Qwell[]is that the stupid one that uses a VGA connector or whatever?
17:59.18sweeperdb13?
17:59.20luisjoseBSDTech, on the PDF?
17:59.26Qwell[]sweeper: 15
17:59.33sweeperorite
17:59.43sweepersounds like a PITA
17:59.50sweepert1 card + chan bank for me
18:00.10sweeperyay for hyooge oil services companies that don't mind spending bux to get the good stuff :)
18:00.45[TK]D-Fendersweeper: Meditrix 1124 + Amphenol punchdown block.
18:00.48monstedmmm, cisco gear ;)
18:00.59Qwell[]monsted: he said good stuff
18:01.02sweeperwe use Adit600's
18:01.06Qwell[]not pointlessly expensive
18:01.10luisjoselol
18:01.21monstedQwell[]: i quite like my catalyst 6500s :)
18:01.25sweepermonsted: cisco <> good stuff
18:01.25BSDTechyes
18:01.29BSDTechlooking fo rit
18:03.02sweeperbah, mediatrix thing has no expandability :/
18:03.09sweeperI can just add cards to an adit chassis
18:03.44Qwell[]sweeper: That's why the mediatrix is less expensive
18:04.06[TK]D-Fendersweeper: Instead you gain redundancy, mounting ease, easyier deployment, lower server load, and no need for a HUGE case to support those monster cards.  Plus better call handling.
18:04.54ManxPowerWe use Adtran Total Access 750s
18:05.13sweepermonster cards? wha?
18:05.31[TK]D-Fendersweeper: "huge".
18:05.44sweeperhave you SEEN an adit 600?
18:05.51[TK]D-Fendersweeper: You want a sime rack mount server?  Hope it isn't 1-2U....
18:06.00[TK]D-Fendersimple/slim
18:06.03luisjosewhat is a punch block to wire analog phones?
18:06.10sweeperluisjose: yea, aka 66block
18:06.22Qwell[]luisjose: It's where the wires "terminate"
18:06.25[TK]D-Fenderluisjose: Sounds like you need to learn a lot about phones in GENERAL
18:06.31[TK]D-FenderStrom_C: Link him!
18:06.36Qwell[]they go from the room to the punchdown block, then from there to...whereever
18:06.38luisjose[TK]D-Fender, yes, Im just starting.
18:06.39sweeperluisjose: http://www.wildtracks.cihost.com/homewire/phoneblk.jpg
18:06.57BSDTechin the us they have a plastic block with metel clips called a punch down block
18:07.01[TK]D-Fendersweeper: That'll do in the mean time.
18:07.02Qwell[]real cablers use punchdown blocks for cat3 for ethernet
18:07.13Qwell[]we had that at our old office when I worked for WF :D
18:07.17[TK]D-FenderQwell[]: EW.
18:07.21cpm66 or 110?
18:07.23Qwell[]"How come my jack doesn't work anymore?"
18:07.23sweeperit burns
18:07.28Qwell[]"It never worked...it's cat3"
18:07.33Qwell[]"uhh...no, it worked.  fix it!"
18:07.49Qwell[]"okay, but it's cat3 and it terminates at a punchdown block"
18:07.51[TK]D-FenderQwell[]: I got to completely GUT this new building I'm in and go Gigabit dual-LAN :)
18:07.55BSDTechI have done 4 a
18:07.58BSDTechsorry
18:08.10luisjoseOk i see is pretty much like a patch pannel but how it is connected to a DB25 port?
18:08.12BSDTechI have used 2 a400 with 4 doughtervoadrs
18:08.21Qwell[]luisjose: no, no, no
18:08.26rene-Are analog interfaces (channel bank) more reliable for office extensions than sip phones?
18:08.32Qwell[]a punchdown block is basically a patch panel, but it doesn't connect to a DB25...
18:08.33BSDTechI have used 2 a400 with 4 doughter boadrs  giving 96 lines
18:08.42Qwell[]UNLESS you're doing something stupid, like using that ridiculous sangoma card
18:08.48BSDTechsome do
18:08.53BSDTechsome dont
18:08.55Qwell[]luisjose: sometimes they go to amphenol
18:09.05Qwell[]which is kinda like DB25, but it's not
18:09.13luisjoseQwell[], lol ok so you punch down the cables from the analog lines and it end as?
18:09.21Qwell[]it's a standard telcom jack (unlike the video card cable you need for the sangoma...*cough*)
18:09.29BSDTech?
18:09.32sweeperluisjose: there is something called a "connectorized 66 block"
18:09.34BSDTechnot for the a400
18:09.46sweeperwhich has an amphenol connector, all wired into one side of the block
18:09.48Qwell[]BSDTech: maybe it's the a800 then...  I don't honestly know sangoma models
18:09.50BSDTechbut I wired my own jack
18:10.04BSDTechthe a800 isfor ds3
18:10.07Qwell[]I just know they have stupid connectors on some of their cards, heh
18:10.20Qwell[]BSDTech: their model numbers are just confusing
18:10.25Qwell[]which one is the 8 port analog?
18:10.26sweeperthat amphenol connector is used to connect to a channel bank or other analog phone system
18:10.40[TK]D-FenderAmphenol is a longer Centronics connector.
18:10.42Qwell[]the TDM2400 uses amphenol
18:10.43BSDTecha400 uses 25 pin serisl/parallel type port
18:10.59Qwell[]BSDTech: Is that the 4 port analog, or quad T1?
18:11.03[TK]D-FenderBSDTech: ... no, not DB25... AMPHENOL....
18:11.04cpmRJ-50
18:11.19luisjoseoh well i thought it was more simple, my home system is built with 2 SIPURA 2002 and 1 SIPURA 3000
18:11.19sweeperhttp://www.voipsupply.com/product_info.php?products_id=569 <-- the adit 600 is cheeper
18:11.24BSDTechno its the 12 line ver of the a200
18:11.32ManxPowerQwell[]: no more confusing that Digium's part numbers
18:11.32Qwell[]and the a200 is...?
18:11.39[TK]D-FenderOMG.. I'm mistaken!
18:11.40luisjosesweeper, let me see what is a 66block
18:11.41ManxPower405, 415, 425?
18:11.46BSDTecha200 is the 4 port card
18:11.47Qwell[]ManxPower: come on now..  we have fairly standard stuff, and it actually makes sense
18:11.51Qwell[]BSDTech: 4 port what?
18:12.02BSDTechhttp://sangoma.com/datasheets/p_a400-specs
18:12.06Qwell[]see, their 4 port cards have a freaking 2 in the model number
18:12.09BSDTechthats the a400
18:12.12[TK]D-Fender"Each 12 port A400 REMORA card is connected by means of a standard 12 line color coded telephone cable terminating at the card in a robust DB25 connector, and ready for hard wiring into a punch block at for the PSTN connection."
18:12.21BSDTecha200 is a 4port tdm fxs/fxo card
18:12.21Qwell[]and a 12 port card has a freaking 4
18:12.23[TK]D-Fenderz0mg!  Retards!
18:12.37ManxPowerQwell[]: that is because (I think) the 4-port Sangoma T-1/E-1 cards are REALY a 2-port card with a 2-port daughter card
18:12.41Qwell[]ManxPower: surely, ours make at least a little sense :p
18:12.43BSDTechno the 12 port has a 25 pin connector
18:12.45sweeperluisjose: http://www.alliancesystems.com/Products/Images/Cables/L-CAB0550.jpg
18:13.22sweeperthat black connector goes to your channel bank
18:13.33sweeperand then you punch down the phone line pairs on the other side
18:13.42sweeperand use bridge clips to connect/disconnect
18:15.11[TK]D-FenderSangoma A400 = 12 port huge card w/o amphenol.  FUGLY!
18:15.15Qwell[]ManxPower: Kenny explained the Sangoma model numbering scheme very well
18:15.35Qwell[]<kenny> Qwell: I think you have to adjust for canadian currency
18:15.55[TK]D-FenderQwell[]: With GWB...you mean PAR ;)
18:16.14Qwell[][TK]D-Fender: huh?
18:16.20Qwell[]~par
18:16.30jbotParagraph formatter for plain text. URL: http://www.cs.berkeley.edu/~amc/Par/
18:16.54BSDTechso I like the a400 its a nice card has worked well
18:16.54[TK]D-FenderQwell[]: Dubbaya sinking your currency's value to the point where the IS no conversion between currencies :)
18:17.01Qwell[]ahh :p
18:17.07Qwell[]figured PAR was an acronym
18:17.12BSDTechhave had no issues
18:17.31[TK]D-FenderQwell[]: No, I have no "bold" in this IRC client (that I've found), so I have to place emphasis with caps :)
18:17.43Qwell[]/par/
18:17.46Qwell[]*par*
18:17.55Qwell[]<i>par</i>
18:17.57[TK]D-FenderQwell[]: doesn't appear as such on my client...
18:18.07Qwell[]it's not supposed to ;)
18:18.07[TK]D-FenderQwell[]: none of those are bolded here...
18:18.13[TK]D-FenderQwell[]: :(
18:18.19Qwell[]but better than a non-TLA TLA
18:18.21[TK]D-FenderI miss mIRC sometimes...
18:18.26Qwell[]~lart [TK]D-Fender
18:18.32Qwell[]~lart [TK]D-Fender with mIRC
18:18.42Qwell[]That is an acceptable lart
18:19.07[TK]D-FenderQwell[]: that should read "strips", not "stripes" :)
18:19.21Qwell[][TK]D-Fender: red and flesh stripes
18:19.22[TK]D-Fenderjbot: self-fornicate!
18:19.43Qwell[]but...
18:19.46Qwell[]~lart [TK]D-Fender with mIRC
18:20.03[TK]D-Fender~jbot
18:20.06jbotmethinks jbot is only marginally useful at best,  He got a C- on his Turing Test
18:20.07[TK]D-Fender^^^^^
18:21.31tzangerhaha
18:21.37tzangerC- on his Turing test
18:21.47*** part/#asterisk pythos (i=lanebob@unaffiliated/pythos)
18:23.38BSDTechto test on bsd
18:23.43*** join/#asterisk naftali5 (n=naftali5@ool-44c05121.dyn.optonline.net)
18:25.05luisjosesweeper, and you hook it to a channel bank?
18:25.10CrashSysD-Fender: You, sir, are TEH MANG!!!
18:25.22BSDTechluisjose here look at this
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18:25.25BSDTechhttp://www.888voipstore.com/rhino-equipment-mid-21.html
18:25.26luisjosesweeper, and its identified by the order or pounching right?
18:25.31[TK]D-Fender~[TK]D-Fender
18:25.33jbot[[tk]d-fender] rockin' the casbah !!!
18:25.33BSDTechthose are channel banks
18:25.35luisjoseBSDTech, checking/
18:26.06perdboo yah
18:26.35Hmmhesays[TK]D-Fender: you ever used a macro to create an ivr menu?
18:26.48CrashSysRhino's stuff on par with Sangoma?
18:26.58[TK]D-FenderHmmhesays: Not a full IVR, but using Read, sorta.  You shouldn't do that normally.
18:27.16[TK]D-FenderCrashSys: AVOID, and no... its super-sup-par
18:27.19BSDTechhttp://www.888voipstore.com/rhino-channel-bank-cb24-fxs-110v-pr-16391.html
18:27.31CrashSyssup or sub?
18:27.34BSDTechthats the one you whould use to put phones in the rooms
18:27.39Hmmhesaysyeah I have a unique scenario here, I need an ivr based on entries in voicemail.conf and app directory
18:28.19[TK]D-FenderBSDTech: http://voipstore.atacomm.com/Shops/ViewItem.aspx/27934028032-29013176064.htm
18:28.44[TK]D-FenderBSDTech: Same price, don't need a T1 card, lower system load.  No special cards.  REDUNDANT.
18:28.48rene-there is the openvox12 port analog card
18:28.51rene-has anyone used it
18:28.57rene-openvoice sp?
18:29.02[TK]D-FenderChannel bank = only if you're cheap and got it as a GIFT
18:29.09*** join/#asterisk ShadowHntr (i=sentinel@wikipedia/Shadowhntr)
18:29.12[TK]D-Fenderrene-: More stuff tro avoid....
18:29.27b11d.
18:29.42[TK]D-FenderHmmhesays: Gimme some recordings!
18:30.05rene-D-Fender: what do you use: polycom?
18:30.16Hmmhesays[TK]D-Fender: recordings of what?
18:30.26ManxPowerChannel banks are totally awesome!
18:30.27[TK]D-FenderHmmhesays: Your stuff....
18:30.41BSDTechok cool
18:30.46BSDTechnice
18:30.50ManxPowerANYTIME I need analog, I go with a channel bank
18:30.50[TK]D-FenderHmmhesays: Now that I'm feeling kinda funk leaning sweeps, soloing is just so much more fun :)
18:31.27rene-ManxPower on a green field pbx install do you go for analog or polys?
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18:32.02luisjose[TK]D-Fender, thats like a big fxs sipura?
18:32.21[TK]D-Fenderluisjose: Yes, thats a close enough description.
18:32.37luisjose[TK]D-Fender, i like that solution
18:32.40[TK]D-Fenderluisjose: 24 ports, uses a telecom standard Rj21 (amphenol) connector.
18:32.40BSDTechhttp://www.888voipstore.com/grandstream-gxp-2000ext-pr-17149.html would make the cheapest best front desk terminal for a hotel
18:32.58b11dgrandstream really?
18:33.04b11di didnt think they had ANY good use
18:33.12BSDTechbut they need to make it with a shift button
18:33.13[TK]D-Fenderb11d: KINDLING
18:33.17danpheh
18:33.21b11dhaha
18:33.32[TK]D-Fenderb11d: Get a good Charcoal filter first though...
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18:33.37BSDTechthats the only one I have seen with that many lines you can monitor
18:33.40CrashSysChannel Banks = The Bomb for hotels/call centers/etc...
18:33.46BSDTechyes
18:33.46b11dnah I like the plastic in the air
18:33.50CrashSysSuck for an office tho :)
18:33.53BSDTechbut for the front desk
18:33.56perdyou can get a nice amphenol -> 24 port patch panel from graybar
18:34.08b11dyeah i've got a few of those
18:34.23BSDTechbut you need a front desk terminal
18:34.35b11dhave one phone per line on the front desk
18:34.37luisjose[TK]D-Fender, and the other side it goes RJ45 to a switch/NIC right?
18:34.37b11dthat'd be the best
18:34.38b11d:)
18:34.47CrashSysI use 66-blocks
18:34.53CrashSysI'm cheap :(
18:35.03BSDTechlol
18:35.04luisjoseCrashSys, 66-blocks connected to?
18:35.04b11di like the 66 over the 110 still
18:35.08CrashSyschannel bank
18:35.11AstaWerksDotComaastra has a new phone comming out in feb that has a side car for  20 extensions  that would be good for a front desk
18:35.17CrashSysAmp 24-pair --> 66-block
18:35.17luisjosegod damnt!
18:35.19[TK]D-Fenderluisjose: Exactly.
18:35.20luisjoseim a bit lost
18:35.26[TK]D-Fenderluisjose: Decent web-admin on it.
18:35.37CrashSys110 is all there is for Cat5 :(
18:35.40BSDTechbut the grandstream has 56 on each
18:35.44BSDTechside cad
18:35.47CrashSys110 seems flimsy
18:35.52b11dyou *can* do cat5 on 66 blocks..
18:35.55b11di do it..
18:35.56luisjoseamphenol isnt 66-block?
18:35.57b11dit sucks
18:36.01CrashSysb11d: it sucks
18:36.02CrashSyserr
18:36.04b11di know
18:36.04b11dhehe
18:36.13b11d110.. yeah flimsy is right..
18:36.17luisjose[TK]D-Fender, so you have used it?
18:36.20BSDTech112
18:36.22AstaWerksDotComweb interface on grand sterams fail always
18:36.23CrashSysAmphenol 24 is a type of connector...
18:36.35[TK]D-Fenderluisjose: Yup, works well.
18:36.42CrashSysI think it's actually a 25-pair connector
18:36.44luisjoseBSDTech, lol that such an ugly phone.
18:36.58luisjose[TK]D-Fender, on normal or low load?
18:37.00[TK]D-FenderCrashSys: 25 pair, only 24 used.
18:37.01BSDTechsorry .
18:37.09perdhttp://pastebin.ca/314379   anyone see a problem with that configuration for a 7902 with chan_skinny ?
18:37.13BSDTechnot every phone can look like cisco
18:37.15CrashSysI'd use a shitstream GXP2000 for a hotel/receptionist phone in a heartbeat... cheap, big, and cheap...
18:37.23perdfor some reason it doesnt work and it causes my phone to act very strange
18:37.27[TK]D-Fenderluisjose: normal.
18:37.49[TK]D-Fenderluisjose: its industrial gear.  It works, and works well.
18:37.53BSDTechthe rev2's work great
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18:38.15luisjoseOk so im getting it slowly at least
18:38.30[TK]D-Fenderluisjose: Good to hear.  Progress = good
18:38.51BSDTechand the hotels I have done we used the gxp2000 as a desk phone with 2 side cars
18:39.03luisjosestill dont get the link between the 66-block and channel banks.
18:39.05BSDTechbeing the hotels where only 100 rooms
18:39.11BSDTech<PROTECTED>
18:39.14*** join/#asterisk endikos (n=endikos@gtlgateway.nmsu.edu)
18:39.15*** join/#asterisk spatulamaan (n=ggilmore@207.188.8.252)
18:39.17[TK]D-FenderBSDTech: That store is kinda pricey
18:39.22BSDTechso it monitors them all
18:39.44[TK]D-FenderBSDTech: You know if you just want to monitor, use FOP or the like....
18:39.46CrashSysLuisjose: Most phone companies will terminate POTS lines on a 66-block... it's a punchdown connector that you put raw wires into...
18:40.28CrashSyshttp://en.wikipedia.org/wiki/66_block
18:40.28b11di had a bunch of 66 blocks actually "fail" on me once..
18:40.35b11dall the metal rapidly oxidized for some reason
18:40.40CrashSysb11d: too many punches? salt water?
18:40.50b11dno.. no salt water.. and not too many punches..
18:40.51BSDTechfop has issues I like hud
18:40.54b11dthey just stopped working.
18:41.00b11dit didnt make ANY sense
18:41.06CrashSysHere in florida we get salt water issues a lot..
18:41.10Hmmhesaysmaking an ivr system with different entries per user just sucks
18:41.12b11dyeah I dont doubt that..
18:41.12BSDTechand am now setting up hotels with a terminal running hudpro
18:41.12[TK]D-FenderBSDTech: Sure, whatever...
18:41.13CrashSysspecially near the beach...
18:41.49CrashSysb11d: I've used a die-electric aerosol spray and actually sprayed the blocks down before if it looks like humidity is an issue...
18:41.59b11dhrm, thats cool..
18:42.00CrashSysSure the block is a lil sticky, but it wont effect nothing...
18:42.03b11dyeah
18:42.14CrashSysEasier then swapping a block out...
18:42.24b11dyeah it was a huge pain in the ass
18:42.35BSDTechand I thoght FOP had issues with more then 40 exten
18:43.29BSDTechok 1.4 is comppiling
18:43.33BSDTechkon freebsd
18:43.46CrashSysdrumroll
18:44.36BSDTechIstill think we wll have to do some patching
18:44.41BSDTechlike we did on 1.2
18:44.45BSDTechallthe time
18:45.04BSDTechthen I can get freepbx to work
18:45.12BSDTechand Iwill have a rocking pbx
18:45.33BSDTechmight look at the asterisk gui when the get it working
18:46.01BSDTechok its installed
18:46.10BSDTechnow to test it with g729
18:46.29BSDTechI love my dual p3 1.2 ghz
18:46.35BSDTech1 gig ram
18:46.49b11dheh.
18:46.54b11di have no issues with asterisk on FreeBSD at all
18:46.58b11di heart it
18:47.04BSDTechok looks like transcoding is fixed
18:47.19BSDTechso 1,4 svn works
18:47.29BSDTechcool
18:47.46ChicagoBud<luisjose> you terminate the lines from each room to a 66-block.  Then you conecct the channel bank to the 66-block via a cable with the AMP connectors
18:48.35BSDTechok I am happy
18:48.50BSDTechnow I am create a 1.4 port
18:48.52b11dwell this is a cause for felicitations!
18:48.59BSDTechbut zaptel is still in the works
18:49.09b11dwtf is with zap?  it works fine for me
18:49.11b11don 6.2-PRE
18:49.21ChicagoBud<luisjose> LAN < 1 - 1> CB <1 - 1> 66 <1 - 24>
18:49.31BSDTech?
18:49.43BSDTechwe are porting 1.4 zatel
18:49.53BSDTechright nwo only the 1.2 is in ports
18:49.58b11doh yeah I dont use the ports..
18:50.04b11dcool, i see what you mean
18:50.09BSDTechbut libpri and freebpx should work now
18:50.26b11dcool
18:50.26BSDTechhow did you get zaptel 1.4 to compile
18:50.32b11di just compiled it?
18:50.33*** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal)
18:50.33*** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal)
18:50.34BSDTechits ment for linux
18:50.43b11di used zaptel-bsd-trunk
18:50.53BSDTechthats whats in ports
18:50.54b11dfrom svn
18:51.14BSDTechfrom digium or the asterisk-bsd dev group
18:51.21b11dfrom the svn.pbxpress.com
18:51.33ChicagoBud<BSDTech> Is the Grandstream Budgetone 200 a decent office phone for a 10 user system?  How is the speaker phone on them?
18:51.40ManxPowercan anyone recommend a SIMPLE and free .WAV file editor for Windows?
18:51.43*** join/#asterisk zotz (n=zotz@24.244.163.157)
18:51.58b11dAudacity is the best
18:51.58b11dimho
18:51.58CrashSysI've used the BT-200
18:52.01Qwell[]ManxPower: sndrecorder
18:52.09CrashSysIf money is an issue it will suffice...
18:52.09Qwell[]comes with windows since like 3.1 :p
18:52.11b11dsndrec32
18:52.14Qwell[]whatever
18:52.30BSDTechb11d svn sting pls
18:52.40b11d?
18:52.44AstaWerksDotComgoldwave.com
18:52.56ManxPowerChicagoBud: Grandstream has a history (years) or terrible firmware on somewhat crappy hardware.  I would not give one to my ex-wife.  I would give my ex-wife cyanide, but not a grandstream product.
18:52.57AstaWerksDotComfree sound player called goldwave there
18:53.03AstaWerksDotComeditor
18:53.12BSDTechthe zaptel-bsd-trunk
18:53.14ChicagoBud<ManxPower>Cool Edit used to be really good but Adobe owns it now -- not sure if it is free anymore
18:53.20BSDTechIdont find the link
18:53.22ManxPowerQwell[]: sndrecorder is what comes with Windows?
18:53.27b11dhang on
18:53.35Qwell[]sndrec32
18:53.39CrashSysJust write a record macro in *
18:53.40CrashSyscall it
18:53.41CrashSysdone :D
18:53.47[TK]D-FenderManxPower: Audacity rocks...
18:53.56[TK]D-FenderManxPower: http://audacity.sourceforge.net/
18:54.04ChicagoBudIs there a decent under $100 phone that is generally recommeded for small offices?
18:54.19AstaWerksDotComaastra 9112 borders  100 its a great phone
18:54.33CrashSysChicago: You need speakerphone?
18:54.35[TK]D-FenderChicagoBud: None.
18:54.44CrashSysPolycom IP301's for $115
18:54.46ChicagoBudyes on the speakerphone
18:54.48CrashSysno speakerphone tho
18:54.56AstaWerksDotCom9112 has speaker
18:55.00ChicagoBud$125?
18:55.01[TK]D-FenderChicagoBud: Do you have or are planning on getting PoE?
18:55.16ManxPowerQwell[]: Thanks!  sndrec32 was just what I needed.
18:55.19ManxPowerjust barely
18:55.22Qwell[]heh
18:55.28ChicagoBudDo not have POE but would consider it but
18:55.29AstaWerksDotCom$107 on my website but can get it down to $100 for you
18:55.30Qwell[]it's simple, but it works
18:55.32b11dbastard
18:55.36b11dI said sndrec32
18:55.36b11d:)
18:55.53b11dhttp://www.pbxpress.com/~gonzo/zaptel-bsd-trunk.tar.gz
18:55.57b11dthere it is
18:56.02*** join/#asterisk olinux (n=olinux@starbucks.wellspublishing.net)
18:56.03b11dfrom here:   i
18:56.04b11ddoh
18:56.07*** join/#asterisk sbingner (n=sam@pdpc/supporter/sustaining/sbingner)
18:56.10b11dhttp://www.voip-info.org/tiki-index.php?page=FreeBSD+zaptel
18:56.25b11dgot the tarball, svn'd myself an update, and that was it
18:56.57*** join/#asterisk Dr-Linux (n=Nothing@202.125.139.198)
18:57.04luisjoseChicagoBud, so is a standar connector that use 66block and channel banks
18:57.19Dr-Linuxwhy i'm facing this problem queue module? >> ivr1*CLI> reload app_queue.so
18:57.19Dr-LinuxThe previous reload command didn't finish yet
18:57.44BSDTechok thats from the asterisk-bsd dev group I am apart of and thats 1.2.4 zaptel
18:57.49BSDTechit needs updating
18:58.28ChicagoBud<luisjose> most channelbanks send out the 24 analog lines via an AMP connector (25-pair cable)
18:58.29BSDTechand we are working to move it to 1.2.12 I think is the current zap and 1.4 for zap
18:58.29Qwell[]Dr-Linux: how long ago did you reload app_queue.so previously?
18:58.31BSDTech2 ports
18:58.42BSDTechthere will be 2 branches of each
18:58.46b11dohh
18:58.47b11dok
18:58.48BSDTechin the ports tree
18:58.55Dr-LinuxQwell[]: 2 minutes ago
18:58.57ChicagoBud<AstaWerksDotCom>looking now
18:58.58BSDTechshower time brb
18:58.59[TK]D-FenderChicagoBud: If you really need to save $, then Polycom IP 430.  Supports PoE, and has all the features your normal users should need @ $150/ea
18:59.00b11di've got to run..
18:59.00b11dttyl
18:59.08[TK]D-FenderChicagoBud: Possibly less
18:59.09Dr-LinuxQwell[]: bcoz queue application was not answering :S
18:59.23CrashSysIP430 = Good shiznit
18:59.25Dr-LinuxQwell[]: i can't unload the module as well
18:59.30CrashSysnot a receptionists phone but good stuff
18:59.34Dr-Linuxivr1*CLI> unload app_queue.so
18:59.34Dr-LinuxUnable to unload resource app_queue.so
18:59.34Dr-LinuxJan 12 10:13:23 WARNING[5616]: loader.c:135 ast_unload_resource: Soft unload failed, 'app_queue.so' has use count 9
18:59.47ChicagoBud<[TK]D-Fender>I'll take a look. Thanks.
18:59.57[TK]D-FenderChicagoBud: www.telephonydepot.com
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19:00.28Dr-LinuxQwell[]: any idea why?
19:00.45Dr-Linuxor suggestions
19:01.55Dr-Linuxshould i restart the asterisk?
19:02.06*** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290)
19:02.29perdis chan_skinny just broken or am i doing something wrong? it doesnt appear to be working in 1.4 at all
19:02.37L|NUXhello
19:02.41perdand that seems odd, since im pretty sure people use it :)
19:02.48Qwell[]perd: it's your phone
19:02.54CrashSysPeople should just avoid cisco
19:03.00Dr-LinuxQwell[]: no clue? :)
19:03.04perdqwell they all do it though, the 7902, 7912 and 7960
19:03.10Qwell[]7960 should work
19:03.18perdbut the other two shouldnt?
19:03.19L|NUXi am getting following error on astersik 1.2.10 launch_script: Unable to create toast pipe: Too many open files
19:03.20CrashSysIt's like the sony of the networking world... it's everywhere, but only plays nice with other cisco stuff... )
19:03.22CrashSys:)
19:03.23*** join/#asterisk WAudette (n=WAudette@c-71-237-146-239.hsd1.or.comcast.net)
19:03.23Dr-Linuxhow can i kill the app_queue.so module
19:03.26Qwell[]perd: the other two are completely untested
19:03.29L|NUXcan some one tell me how can i fix this issue ?
19:03.30perdah
19:03.30Qwell[](by me)
19:03.44Qwell[]Dr-Linux: Do you have any queues running?
19:03.49perdwell, if you want more dumps let me know, anything i can do to help you make it work :)
19:04.05luisjoseBSDTech, did you test the gui on freebsd?
19:04.06Qwell[]perd: eventually I'd like to try to get "feature dumps"
19:04.18Qwell[]ie; a dump of it putting a call on hold, and doing conferencing, stuff like that
19:04.22perdsure
19:04.30Dr-LinuxQwell[]: ofcos i'm using queues for our callcenter
19:04.48Qwell[]Dr-Linux: then you can't unload it if queues are running
19:05.06Dr-LinuxQwell[]: ?
19:05.17Dr-LinuxQwell[]: "show channel" shows there is nothing
19:05.38CrashSyskill -9 will unload it... :)
19:06.06CrashSysThey should let me drink at work... i'd be so much more productive...
19:08.45BSDTechnot yet
19:08.49BSDTechits in the plans
19:09.00BSDTechbut I know it will take some patching
19:09.15BSDTechmostlikly
19:09.29perdqwell for feature dumps do you want two of the same type of phones or does it matter?
19:09.44Qwell[]huh?
19:09.47perdnot matter that is
19:11.01perdnm
19:11.08BSDTechI like the freepbx interface but it is database driven
19:11.37BSDTechand i like the 3rd lane interface
19:11.44BSDTechits more robust
19:11.52BSDTechbut they all have issues
19:11.55BSDTechon bsd
19:11.56*** part/#asterisk reber (i=reber@gateway/tor/x-a6716cbe69f64a14)
19:12.36Dr-LinuxQwell[]: i restarted the asterisk, and everything is fine now
19:13.04Dr-LinuxQwell[]: but not sure what was the problem with app_queue.so module
19:13.07*** part/#asterisk WAudette (n=WAudette@c-71-237-146-239.hsd1.or.comcast.net)
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19:13.12*** part/#asterisk WAudette (n=WAudette@c-71-237-146-239.hsd1.or.comcast.net)
19:13.18Dr-Linuxand how can i avoid this problem in future
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19:29.38McGeeHi, what is the status of call rerouting on T-Com PTP ISDN? Is it possible or is there still codeing work to be done?
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19:47.33Marty-OTTI just received a Dul port T1 card from Sangoma - going to put it in my Asterisk box... :P
19:47.43Marty-OTTIt's MY FIRST TIME.. I;m a VIRGIN at this
19:47.53*** join/#asterisk bprice20 (n=brandon@216.120.224.199)
19:48.02*** join/#asterisk shido6 (n=shido6@d221-68-200.commercial.cgocable.net)
19:48.23Marty-OTTAnyone has any hints about when my FreeBSD box boots back up with the Sangoma card in it?
19:48.53pifsangoma is not supported on freebsd
19:50.01mercestespif:  Wow, what a mood kill
19:50.04luisjoselol
19:50.17bprice20Has anyone gotten res_snmp under 1.4.0 to toss out number of registered sip users
19:50.33*** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal)
19:50.34pifgood cherry popping for a virgin, hey? :)
19:50.35*** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal)
19:50.39bprice20I can get other statistics out of it like number of sip channels
19:51.10bprice20just not number of registrations; I think I need to right snmp OID
19:51.12mercestespif:  He didn't even get to take his clothes off.  It's like, taking yoru girlfriend to a hotel room, and she whispers "i'm a virgin."  and you get in there...and *all* the rooms are chaparoned.
19:51.24mercestesand their's no privacy anywhere.
19:51.33pifdude
19:51.58mercestesyea
19:52.16mercestesworst nightmare I ever had.
19:52.21ManxPowerIs it "newbies answer questions they know nothing about day" on asterisk-users?
19:52.38mercestesManxPower:  Must be, you're here.
19:52.48mercestes:P
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19:54.03[TK]D-Fender:O
19:54.17pifdunno, just reading "my freebsd box" made my debian juices flow
19:54.32ManxPowerWheres a gun toting Republican like JerJer when you need one?
19:54.41tzangerhahaha
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19:55.57Marty-OTTpif:  As far as I;m reading here - yes it is
19:56.10Marty-OTTI checked this out with the Sangoma themselves before buying
19:56.31*** join/#asterisk cmdln (n=cmdln@64-126-105-14.static.everestkc.net)
19:56.33Marty-OTTI'm reading the installation instructions under FreeBSD
19:56.38*** part/#asterisk cmdln (n=cmdln@64-126-105-14.static.everestkc.net)
19:56.42pifsure, I just wanted to give a friday night adrenaline rush
19:56.49pifto a virgin
19:56.54Marty-OTT:P ... well done... lol
19:57.09Marty-OTTwas concerned someone had issues with Sangoma card(s)
19:57.26pifyou should install linux nevertheless
19:57.39pifget rid of that crap *bsd
19:58.15ManxPowerMarty-OTT: so few people use Asterisk on anything except linux just assume you are on your own with *MSD
19:58.57ManxPowerJust think of yourself as so "special" nobody will be able to help you
20:00.07CrashSysEven if you were to use Linux so few people would want to help you that you would still feel special :)
20:00.20piflol
20:00.24CrashSysMost people will just quote you their hourly rate
20:01.09[TK]D-FenderQuick stuff I offer free.  Full setups I charge cheaply for :)
20:01.22cpmbtw
20:01.23ManxPowerI have several 2-port Sangoma cards on Linux with Asterisk
20:01.24cpmsip sucks
20:01.34[TK]D-Fenderjust yesterday did a 7-phone Polycom / Sangoma setup from scratch.
20:01.36ManxPowercpm: heretic
20:02.06*** join/#asterisk h0 (n=fakhir@unaffiliated/fakhir)
20:02.15pifI've been more lucky with sip to trunk my asterisks than with iax2
20:02.29pif(random UNREACHABLE crap)
20:02.40cpmHow to use sip in the *real world* 1, set up asterisk, set up a sip device, go on the road, realize you need another pocket asterisk box of some kind running iax to tunnel back to your main pbx, because, , , SIP SUCKS
20:02.45ManxPowerpif: me too.
20:03.28ManxPowercpm: Um, Before Katrina I roamed between variaous nat and non-nat networks just fine and connected back to my Asterisk server, also behind nat.
20:03.35ManxPowerno config changes needed at all
20:03.46ManxPowerUsing a SIPura
20:03.49cpmcool, glad it works well for you.
20:04.00cpmAhh, Sipura, I hear good things there
20:04.03ManxPowerAnyone that can't roam with a SIP device doesn't have things set up correctly.
20:04.17cpmtell that to Polycom
20:04.34ManxPowercpm: Polycoms should do it just fine too.
20:04.43ManxPowerI had NO NAT config stuff on the SIPura
20:04.54cpmprolly 70 percent of the time
20:04.57*** join/#asterisk Strom_M (n=strom@netblock-66-159-243-60.dslextreme.com)
20:05.17cpmend up fireing that hoplessly awful kiax, and bingo! I can connect.
20:05.43*** join/#asterisk UlbabraB (n=salama@host241-43-static.72-81-b.business.telecomitalia.it)
20:05.46cpmnot that the kiax folks are doing anything wrong, I'm grateful for that softphone. Because it pretty m uch always works, when my polycom won't.
20:06.11*** join/#asterisk Chris-NB (n=chris@argos.campus-sbg.at)
20:06.40cpmI should get some sipura stuff.
20:06.48ManxPowerIt sucks to be you
20:06.50luisjosesipura is ok for a house
20:06.59cpmNot really.
20:07.06robl^new polycom firmware has NAT-friendly features.. even NAT keepalive.
20:07.20pifmy spa-3000 won't relay dtfm tones to a fucking door phone
20:07.33luisjosepif, how come?
20:07.44pifand it's configured "inband" all the way
20:07.45cpmgood thing that relaying to door knobs isn't often needed
20:07.53luisjosedoor phone?
20:07.54ManxPowerpif: How do you know that?
20:07.59[TK]D-Fenderpif: use "INFO"
20:08.03ManxPowerpif: put a phone on it and see if you hear the tones
20:08.12pifgood thought
20:08.20ManxPowerpif: it should have been your FIRST thought.
20:08.26pifit's a remote install
20:08.32*** join/#asterisk s1gny|wrk (n=s1gny@p54914E21.dip.t-dialin.net)
20:08.36pifmust use an employee to test
20:08.53pifbeen bugging these guys all day :)
20:09.01*** part/#asterisk s1gny|wrk (n=s1gny@p54914E21.dip.t-dialin.net)
20:10.02[TK]D-Fenderpif: Then it should come as no surprise to them ;)
20:10.57pif[TK]D-Fender : why info?
20:11.04*** part/#asterisk bprice20 (n=brandon@216.120.224.199)
20:11.21pifI want the sound to goto the door phone in order to unlock the door
20:11.25[TK]D-Fenderpif: Will sed DTMF between the ATA & * OOB, but reconstruct clean on each side.
20:11.35pifhmmm
20:11.44[TK]D-Fenderpif: I'm betting its distorting
20:12.03ManxPower[TK]D-Fender: I say it's a volume issue.  The tones are not loud enough.
20:12.28[TK]D-FenderManxPower: Could very well be, and going OOB will stop gain from being a normal issue hopefully.
20:12.40[TK]D-FenderManxPower: Gain of the standard voice channel anyways
20:12.49pifif INFO == rfc2833 ?
20:12.53pifs/if/is
20:12.55ManxPower[TK]D-Fender: yes, but OOB would make the tones pretty short, perhaps too short for the door device to work
20:12.58ManxPowerpif: NO!
20:13.08ManxPowerINFO == INFO.  RFC2833 == RFC2833
20:13.24[TK]D-Fender1 + 1 = 3!
20:13.38pifthere's no INFO on the spa-3000
20:13.43pifscrap that
20:13.52ManxPowerpif: they call it something else I'm sure.
20:13.56pifthere's no rfc2833 on the spa-3000
20:14.01ManxPowerAVT is what they call RFC2833
20:14.07pifyes!
20:14.18pifthanks
20:14.34pifwhat does avt stand for?
20:14.46ManxPowerask SIPura.
20:14.55pifsure ;)
20:15.09ManxPowerI always set mine to AVT in the SIPura and set it to RFC2833 in Asterisk and NEVER EVER had a problem
20:15.29pifhave tried that, will do, thanks again
20:15.38pifs/have/haven't
20:16.07*** join/#asterisk shidan (n=chatzill@CPE0013107d30c4-CM001371871af0.cpe.net.cable.rogers.com)
20:16.22naitramis there a way to concat two strings like System(1ststring . ${VAR1}), does that work, the dot is a php concat
20:16.51ManxPowerpif: you are wasting time.  Get the box into a a test enviroment and play with the settings
20:17.14ManxPowernaitram: in what l;anguage / enviroment"
20:17.31ManxPowerSetVar(FNORD=${BOB}${DOBBS})
20:17.38pifI'm fucking stupid, I let go my last unit, and now I must rely on strangers to test
20:18.36naitramManxPower: in the dial plan scripts exten => 101,1, System(/var/www/myscript.php 1stargument 2nd 3rd . ${MYDIALPLANVARIABLE})
20:19.36ManxPower<PROTECTED>
20:19.53ManxPowerremember variables are substituted first thing
20:20.25naitramManxPower: oh, ok. Thanks for the help. New to asterisk.
20:20.39ManxPowerin your example you have 5 args, where "." is the 4th arg
20:22.05naitramManxPower: Gotcha, thanks
20:22.49pifhmm, I see "DTMF Playback Level: -16" could that be the cluprit?
20:23.13pifand "DTMF Playback Length: .1"
20:23.15*** join/#asterisk Defraz (n=t0tal@24-116-159-197.cpe.cableone.net)
20:24.11ManxPowerSince we don't know what volume level or tone length your "door phone" requires, you'll just have to try it.
20:24.31ManxPoweryour EAR is the best thing to test with, so get your ass over to the client and try it.
20:25.29pifyep
20:26.15sweeperhmm
20:28.01sweepertime to find out if 1ghz/512mb can handle 13 channels sans transcoding
20:28.18sweeper*23
20:28.34Strom_Mprobably
20:28.56[TK]D-Fendersweeper: Should have no issue
20:29.03sweeperyay
20:30.05ManxPowersweeper: I have a 1.8Ghz machine that handles 96 channels with only a small amount of transcoding
20:30.42sweepercool \o
20:31.10ManxPowerfor various reasons, assume only 24 channels are ever in use at the same time
20:31.44*** join/#asterisk s1gny|wrk (n=s1gny@p54914E21.dip.t-dialin.net)
20:32.05*** part/#asterisk s1gny|wrk (n=s1gny@p54914E21.dip.t-dialin.net)
20:32.32*** join/#asterisk Qwell[] (i=qwell@unaffiliated/qwell)
20:32.32*** mode/#asterisk [+o Qwell[]] by ChanServ
20:33.00CrashSysI'm glad I haven't had to worry about transcoding yet :)
20:34.55ManxPowerCrashSys: we don't do a lot of VOIP outside the LAN
20:36.39cbullock81hey D-Fender
20:36.53cbullock81can i hit you up one more time about polycom phones?
20:38.17cbullock81I'm trying to setup Busy Lamp Field for the polycom ip650 phones, and i must be overlooking something. anyone have any experience with this?
20:39.37wunderkinon text:ip650:/ignore $nick
20:39.38wunderkin;p
20:42.56cbullock81what was that for?
20:43.14Qwell[]he's jealous :P
20:43.18cbullock81heh
20:43.19CrashSysEnvy
20:43.33wunderkinenV
20:43.51cbullock81have any of you used BLF with polycom before?
20:44.22danpfor messages or "buddies"?
20:45.18cbullock81i guess for buddies... basically i just want it to show if the extension is in use
20:45.23danpyeah
20:45.33BSDTechno but I have on the gxp2000
20:45.42BSDTechthats simple
20:45.47*** join/#asterisk denon (i=denon@synapse.subneural.net)
20:45.47*** mode/#asterisk [+o denon] by ChanServ
20:45.59danpyou need to turn messaging on in your phone config, add hints in asterisk and add the things you want to watch to the directory on the phone with buddy watching turned on
20:46.57cbullock81about adding hints... i did something like this  "exten => 101,hint,SIP/101"  is that correct?
20:47.14danpyep
20:47.33cbullock81ok. then how do you tell the phone to subscribe to that hint (or however you word it)
20:47.53danpon the phone itself, hit the directories button and then go into the contact directory
20:48.13danpthe contact field will be your hint name (101). be sure to turn buddy watching on near the bottom
20:48.28*** join/#asterisk |dennis| (n=dennis@200.32.233.82)
20:48.33*** join/#asterisk linlin (i=will@c-71-194-70-13.hsd1.il.comcast.net)
20:48.38danpbut that only shows up if you've turned on messaging...which you can do via the XML configs. not sure if you can do that via the web interface
20:48.47naitramtyring this; exten => 101,1,Set(MEMAREA=6800 101) ; getting this error No application 'Set' for extension (sip, 101, 1).
20:49.01BSDTechok I am building my new freebsd/asterisk server now
20:49.12BSDTechand I will use the thirdlane interface
20:49.14naitramis this not how to set a variable?
20:49.23BSDTechwich I thinkis what digium should use
20:49.41cbullock81danp: i'm about to enable messaging and see what i can come up with. i appreciate the help.  i might have to hit you up again in a bit if thats ok :)
20:50.01ManxPowernaitram: what version of Asterisk?
20:50.16danpcbullock81: sure! it's pretty easy once you get messaging turned on
20:50.33*** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal)
20:50.33*** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal)
20:50.33CrashSys2.0!
20:50.44naitramManxPower: 1.0.7-BRI
20:50.52Strom_M*blink*
20:50.57Strom_M1.0.7
20:50.58ManxPowernaitram: set is not valid in 1.x, use SetVat
20:51.05ManxPowersetvar
20:51.09BSDTechthe phone does the blf
20:51.20*** part/#asterisk shidan (n=chatzill@CPE0013107d30c4-CM001371871af0.cpe.net.cable.rogers.com)
20:51.22BSDTechyou just need to put the name and exten number
20:51.30BSDTechand it should work
20:51.33naitramManxPower: ok, what is the latest stable version of Ast
20:51.45BSDTechit does in the grandstreams
20:52.07danpBSDTech: it's basically the same with the polycoms but there's a little extra setup to do
20:52.18BSDTechahh ok
20:52.27CrashSyspif: Good luck
20:52.30CrashSysWifi = Suck
20:52.32[TK]D-Fenderpif: Try 2010.
20:52.37pifi know
20:52.42pif2010?
20:52.54BSDTechI wish the cisco wifi phone was sip but its not
20:53.01BSDTechits skinny sccp based
20:53.22CrashSysOther then Aastra are there any good cordless (not wifi) phones?
20:53.24pifthe 7920 ?
20:53.30[TK]D-Fenderpif: Yeah, I figure maybe another 3 years or so they'll get it right :D
20:53.45pifit blows chunks (sold mine on ebay :)
20:54.06De_MonSweet! bleach 110 is out
20:54.17De_Monoops, wrong channel/network
20:54.34sweeperjust a bit
20:54.38CrashSyslol
20:54.40pifDe_Mon : /join #teenporn
20:55.12CrashSysThe sad part is I bet atleast 25% of this channel knows what he is talking about :)
20:56.52BSDTechok I like the new thirdlane interface
20:56.57De_Mon? teenporn since when was bleach considered teen porn?
20:57.06BSDTechits much better then the asterisk gui
20:57.53De_Monthe wifi cisco phone has a better gui than the asterisk?
20:57.57*** join/#asterisk |dennis| (n=dennis@200.32.233.82)
20:58.20BSDTechno
20:58.26danpwhy does the thirdlane pbx manager have a link to google in the menu
20:58.37CrashSys$$$?
20:59.03sweeperok
20:59.04*** join/#asterisk [hC] (n=hardcore@206.108.27.93)
20:59.13sweeperwhy the FUCK does * use awk in the build process?
20:59.35*** join/#asterisk airjump (n=airjump@p508AD998.dip.t-dialin.net)
20:59.39mogwhy not ?
20:59.56De_Monbecause awk is better than sed
20:59.57De_Mon:P
21:00.05sweeperalso, why does it have a HARD LINK to awk
21:00.25sweeperthat's just fucking stupid
21:00.26*** part/#asterisk mtaht4 (n=m@c-71-198-23-124.hsd1.ca.comcast.net)
21:00.45CrashSysThirdlane is a webmin module... sounds interesting...
21:00.49De_Monuh, okay.
21:00.58mogsubmit a bug sweeper
21:01.04De_Monya I think i've looked at 3rdlanes pbx manager before
21:01.07De_Monand a patch
21:01.12Marty-OTTso...  anyone on here who's installed a Sangoma T1 card on FreeBSD here?
21:01.38*** join/#asterisk oQPa (n=roque@15.Red-83-40-197.dynamicIP.rima-tde.net)
21:01.41cbullock81danp: ok. it now shows if an extension is registered on asterisk, but when that extension is busy, the status does not change... any ideas?
21:01.41*** join/#asterisk codefreeze (n=steve_mu@216.166.159.235)
21:02.12ManxPowerMarty-OTT: You are on your own, masochist.
21:02.46danpcbullock81: on your phone, does the entry show up as a silhouette or as a number pad icon?
21:02.58Marty-OTTI'm not alone in this boat.. the card's been detected but the Sangoma docs are unclear.
21:03.11Marty-OTTI already sent a support e-mail
21:03.29pifand you visa card cryptogram ?
21:03.37cbullock81danp: silhouette
21:03.55sweepermog: I'll put it on the todo list :v
21:05.18danpcbullock81: hmm...does 'show hints' say you have a watcher?
21:06.14cbullock81danp: yea it does
21:06.51danpif you get that device on a call and check 'show hints' again, does it show its status as InUse?
21:07.01cbullock81lemme see
21:07.09Marty-OTTfound the answer..
21:07.47cbullock81danp: yea it still shows idle
21:08.10danpcbullock81: hmm, you're actually in a call, right? not just off hook?
21:08.33*** join/#asterisk tsurko (n=tsurko@vpn-pppoe-213-240-242-237.megalan.bg)
21:08.38cbullock81yea. i placed a call
21:08.53danpodd
21:10.09danpon my console, i have my verbosity set to 10 and i see messages such as:
21:10.14danp<PROTECTED>
21:10.25danpdo you see anything like that?
21:12.34*** part/#asterisk CrashSys (n=kumba@bartleby.crashsys.com)
21:12.37*** join/#asterisk `Sean (i=Un1x@CPE000c148d127c-CM00140458831c.cpe.net.cable.rogers.com)
21:13.33ManxPowerYet another noob on the mailing list that think a phone has to be registered in order to make calls
21:13.57[TK]D-FenderManxPower: *sigh*
21:14.22ManxPowerand one that things "sip show registry" will show you devices registered to Astertisk
21:15.17ManxPowerOh and one that thinks there is no echo on the PSTN and that it is caused by VoIP
21:17.39pifif .fr our condoms are called ManX
21:17.51ManxPowerpif: I can live with that
21:17.53pifcoincidence?
21:18.06ManxPoweras long as they are the extra large version
21:18.34pifjust bought a pack in anticipation for sunday night
21:18.56ManxPowerI usually just mail order mine in bulk
21:18.58Qwell[]"if .fr"?
21:19.19pifmet a chick on meetic.fr
21:22.15pifviagra is always handy on first dates
21:23.29pifwhere pressure to perform can kill your member
21:23.45Juggiewhat does this have to do with asterisk
21:24.43Strom_Mbonarpillz
21:24.45pifjust meant to say that * users are not all faggots
21:24.53piflike its developpers
21:24.56Corydon-wHey, now, watch it.
21:25.08[TK]D-Fender....
21:25.21*** mode/#asterisk [+b *!*n=ldm@*.apartia.fr] by Qwell[]
21:25.21*** kick/#asterisk [pif!i=qwell@unaffiliated/qwell] by Qwell[] (Qwell[])
21:25.25Qwell[]watch nothing
21:25.38Qwell[]That was inexcusable
21:26.04[TK]D-FenderQwell[]: Yup.  WAY over the line.
21:26.09ManxPowerHmm?  He should be much more offensive than that to be banned.
21:26.22[TK]D-FenderManxPower: Depends on the term...
21:26.54ManxPowerMaybe in France the term is not offensive.
21:27.02Corydon-wWouldn't that be "bundle of sticks"?
21:27.10Qwell[]ManxPower: "<pif> like its developpers"
21:27.35Qwell[]That comment absolutely shows that it was meant to be offensive
21:28.23Juggiethat type of conversation has no place in here.
21:28.57[TK]D-FenderCorydon : extra double-entenre of "piles" ;)
21:29.06[TK]D-FenderCorydon : Recursive humour ;)
21:30.15[TK]D-FenderQwell[]:  http://dictionary.reference.com/browse/piles
21:30.44Qwell[]gotcha
21:30.58[TK]D-Fender:D
21:31.11Qwell[]That bit of info is unfortunately not something one can "unlearn"
21:31.32[TK]D-FenderQwell[]: I will rest well knowing I've made an impact :)
21:31.47[TK]D-Fenderbbiab
21:31.52[TK]D-Fenderheading home...
21:45.29*** join/#asterisk Netgeeks (n=root@pbx5.netgeeks.net)
21:46.01NetgeeksAnyone see anything wrong with this dialplan command:  GotoIf($[${X} >= ${DIGLEN}]?end-loop:start-loop)
21:46.44ManxPowerNetgeeks: what are you trying to ACCOMPLISH?
21:47.26NetgeeksI need to cycle through a variable lenght string of digits and break them out one at a time to playback the reading of the digits
21:47.47Netgeeksso I get the length, and loop over the variable for each digit
21:47.59Netgeeksbut, that GotoIf is failing
21:48.43Netgeekshrm, I think I figured out why
21:50.33*** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal)
21:50.33*** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal)
21:52.39*** join/#asterisk [TK]D-Fender (n=joe@64.235.216.2)
21:55.24in-pthello all
21:55.37b11dgo away
21:55.39b11dwe're closed
21:55.39BSDTechback
21:55.42b11dhehe
21:55.48in-pti installed asterisk-1.4.0, but dont see skinny on asterisk cli
21:55.53in-ptwhat could be the problem
21:55.55b11dsigh
21:56.00b11d33 more minutes..
21:56.01b11dlets go
21:56.26BSDTechThe technical channel for wich you have reached is currently closed please logout and try back ltr. Beeeeeepppppp
22:00.35perddoes option p (screening mode) for dial in 1.2 ?
22:00.42perderr, does option p work
22:02.08perdgoddamn cisco phones forcing me to downgrade
22:03.11BSDTechok kids whats going on here
22:03.15BSDTechThe technical channel for wich you have reached is currently closed please logout and try back ltr. Beeeeeepppppp
22:04.59b11dahhh
22:05.26cbullock81when you are using hints in extensions.conf, are there any additional settings in sip.conf that must be configured (asterisk 1.4)
22:05.28b11ddoes Vista continually attempt to re-validate itself against MS or what..
22:05.37b11dcbullock81.. not in my experience
22:05.50danphmm
22:05.54b11di had to make changes in <MAC>-sip.cfg for Polycom's to get hints working for Buddy lists.
22:05.55cbullock81the status will not change
22:06.24cbullock81my buddy list shows the phones that are turned on properly & has the ones that are not connected showing as away
22:06.32b11dis it a polycom?
22:06.35cbullock81but when a call is made, the status does not change
22:06.37cbullock81yea
22:06.39cbullock81ip650
22:06.50b11dare you using a provisioning server?
22:06.52cbullock81yea
22:07.02b11dhrm.. did you edit the <MAC>-sip.cfg appropriately?
22:07.27cbullock81i guess... i must have misseed something
22:07.44b11dyeah.. i removed my 'hints' shit because i never needed it..
22:07.52b11dbut it took a little work to get going.. once it worked, it worked..
22:07.59b11dcant speak to the 650..
22:08.02b11dthese were on 501s
22:08.05danpyeah, i can't think of anything else i had to do. i have 601's
22:08.08*** part/#asterisk naitram (n=danny@216.77.58.40)
22:08.24cbullock81hmmm... you didnt use hints in *?
22:08.32danpi think this is all i had to add to the phone config: <feature feature.1.name="presence" feature.1.enabled="1" />
22:08.33b11dyes
22:08.35b11di did
22:08.43b11dbut I disabled it, because i had no real use for it
22:08.48cbullock81o
22:08.49b11di dont just roll shit out because "its neat"
22:09.07b11ddanp is right.. thats what i had to change
22:09.18cbullock81k... i did that, but im going to try again
22:09.23b11dthen i had to have hints in the extensions.conf..  on both extensions that were to be monitored
22:09.41b11dlike, say 1234A wanted to monitor 1234B -- both extensions had to have HINTS
22:09.52danpboth do? hmm
22:09.55*** join/#asterisk ta^3 (n=tacvbo@189.146.191.134)
22:09.58cbullock81would that setting need to be in my local-settings.cfg?
22:09.59b11dthats how it worked for me
22:10.07b11ddunno..  i dont use a 'local-settings.cfg'
22:10.29sweeperok, so when configuring a 4-port t1 card, I do my channels like so: bchan=1-23,25-47 dchan=24,48?
22:10.45b11dcheck out the zaptel.conf.sample or something
22:10.45sweeperwell, assuming I'm using 2 t1s
22:11.11Marty-OTTBROADSOFT
22:11.13Marty-OTTVS.
22:11.13b11dwouldnt you define seperate spans, and then define b & d chans per span?
22:11.15Marty-OTTASTERISK
22:11.20Marty-OTT???
22:11.26perdOMG VS WTF
22:11.27b11dMarty.. lets drink Royal Reserve tonight..
22:11.28Strom_MMarty-OTT, that's irritating
22:11.37Marty-OTTsorry
22:11.47sweeperok, I needs a bettar examplu
22:11.55Marty-OTTLooking at Broadsoft solutions.. just sent an e-mail.. if I don't have to reinvent the wheel.. I won't
22:12.10Strom_Mwhat does broadsoft sell?
22:12.12perdthe wheel is obsolete
22:12.12sweeperthe example zaptel.conf doesn't really explain syntax for multiple spans
22:12.17b11dhrm...
22:12.26sweeperperd: so is your mom. doesn't stop us from using her every night \o
22:12.34Marty-OTTSoftPBX solution - pretty much a leader in the industry for paid products
22:12.53b11dhttp://www.voip-info.org/wiki/index.php?page=Asterisk+config+zaptel.conf
22:13.04sweeperI was just heading there ;)
22:13.09Strom_MMarty-OTT, you honestly don't have to reinvent the wheel with asterisk - it's easy stuff
22:13.14sweeperbetween voip-info and *guru, it's all good :)
22:13.16*** join/#asterisk simplton (n=root@c-24-99-119-243.hsd1.ga.comcast.net)
22:13.22james_Marty-OTT: Broadworks is about 100000000000000x more complete than asterisk
22:13.31*** part/#asterisk simplton (n=root@c-24-99-119-243.hsd1.ga.comcast.net)
22:13.37Marty-OTTStrom_M:  You know what - I agree - so far it hasn't been to hard...
22:13.39Marty-OTTJames???
22:13.50Marty-OTThow's that?
22:14.05Marty-OTTI haven't talk to anyone at Broadsoft and probably won't until Monday
22:14.11eGovMaybe he means complex?
22:14.16james_Marty-OTT: it's a complete platform, all administered via guis
22:14.35rene-like mitel
22:14.36*** join/#asterisk ivanfm (n=ivanfm@c93481ec.virtua.com.br)
22:14.37james_it's more complex, it's more complete, it has way more features, it's easier, it's more expensive
22:14.42Marty-OTTJAnes:  Can someone go in and turn on/off features on a per phone basis?
22:15.01james_on a per user basis, yeah
22:15.02rene-nothing you cant emulate with some mad SER & YATE & ASTERISK skillz
22:15.03Marty-OTTJames: Interesting.. you have any idea how expensive it is?  (I mean, I'll know on Monday/Tueasday anyways)
22:15.03*** join/#asterisk simplton (n=root@c-24-99-119-243.hsd1.ga.comcast.net)
22:15.19james_rene-: and maybe 5 years of time
22:15.34rene-of course wont be as polished
22:15.35Marty-OTTThat's the thing..
22:15.48Strom_MMarty-OTT, I sent you a PM
22:15.55james_Marty-OTT: no idea how much, my guess is quite expensive... i help support our countrys biggest telcos voip network, which is broadsoft
22:16.13Marty-OTTyeesh..
22:16.21rene-broadsoft is big iron voice session border controller stuff 100k usd and up
22:16.24*** join/#asterisk bkw__ (n=brian@ip68-0-120-100.tu.ok.cox.net)
22:16.29rene-and waaay up
22:16.30Marty-OTTyow!!!
22:16.54Marty-OTTWell, yeah, I don't want to spend more than $10,000
22:16.57b11dwhat country is .st?
22:17.03james_oh, i imagine it would be out of your price range
22:17.09james_b11d: haha, ignore that... i'm in australia
22:17.12b11doh
22:17.14b11d:)
22:17.20rene-yup, maybe some used cisco or mitel gear
22:17.22b11di'd move back to australia in a second..
22:17.26rene-mitel is user friendly
22:17.34rene-dunno about cisco
22:17.51rene-i wouldnt think it was user friendly
22:17.54Marty-OTTrene:  funny you say that about Mitel...  I taught a lot of CIsco courses for them and they tried to get me to take a free course on the 3300
22:18.01james_what cisco would offer for <$10k would be a joke
22:18.02Marty-OTTjust never got around to it.
22:18.52rene-well the stability problems of 1.4 in my shop is making managers want to revert back to mitel,  and me taking a course to learn about the gear.. sigh...
22:19.08rene-from what i have seen it seems quite simple
22:19.22b11dwhat kind of stability issues?
22:19.27b11ddeadlocks?
22:19.31rene-yes
22:19.31*** join/#asterisk dasenjo (n=dasenjo@208.195.215.51)
22:19.34b11dstill eh..
22:19.35b11dthat sucks
22:19.42*** join/#asterisk syzygyBSD (n=chatzill@66.226.228.204.cpe.speedyquick.net)
22:19.52Strom_Mrene-, this is why my clients are still on 1.2.x; i'm waiting until two major bugfix releases of 1.4 to put it in production
22:19.52*** part/#asterisk endikos (n=endikos@gtlgateway.nmsu.edu)
22:19.59Marty-OTTrene: 1.4 for what?
22:20.10eGovI assume *
22:20.16b11deGov would be right
22:20.23b11deGov, you better start piping up more often :)
22:20.27BSDTecho1.4  is good in svn head
22:20.28eGovhehe
22:20.30syzygyBSDI want to upgrade to 1.4 if only for the imap voicemail
22:20.32Marty-OTToh asterisk ..
22:21.01BSDTechwwhats asterisk and what can it do for me. how will it make me rich
22:21.04Marty-OTTRene:  did you simply use the 3300 as a Sip server?  The Mitel sets are expensive
22:21.05cbullock81b11d or danp: what options would you configure for the directory.xml  anything special
22:21.40b11di dont use it..
22:21.47b11dnot yet anyway.. the directory on polycom's sucks
22:21.53b11despecially when it gets over like 15 entries
22:21.55*** part/#asterisk ctooley (n=ctooley@rrcs-71-42-115-242.sw.biz.rr.com)
22:22.23syzygyBSDBSDTech: asterisk will search for little, lets call them "widgets" and will sell them on the black market, by doing this it will slowly amass a fortune so it can enter the drug trade,  shortly thereafter you will be rich!
22:22.27cbullock81well, that is where i added the extensions that i want to monitor to the soft keys
22:22.35b11din directory.xml ??
22:22.38b11dwtf
22:22.41b11dohh yeah thats right
22:22.42BSDTechlol
22:22.44b11dI forgot about that
22:22.59b11dman I need to not pipe up when im not doing somethign anymore :)
22:23.10cbullock81heh... i appreciate your input
22:23.11rene-1.4 for call center
22:23.20cbullock81i need any help i can round up
22:23.23b11dyeah i hear that..
22:23.24rene-Strom_M: yes
22:23.29b11done sec..i'll see if i still have anything
22:23.33rene-but the way queues work is a bit slow
22:23.36cbullock81that would be great
22:24.01b11dgot it
22:24.42BSDTechok all the bsd users to the left of the room and all the linux users to the right
22:24.59BSDTechbsd users raise your left hand
22:25.09BSDTechlinux users raise your right hand
22:25.20Marty-OTTRene:  So, what were using from Mitel?
22:25.21BSDTechand if you use both grab a seat on thefloor
22:25.36Marty-OTTI'm sitting... but I'm Freebsd
22:25.53BSDTechwell I have freepbx 2.2 almost working on bsd
22:26.05syzygyBSDya, our last bsd box died a month ago, we installed linux to replace it
22:26.14b11dweak
22:26.14BSDTechbumbs
22:26.24BSDTechmove back to bsd
22:26.28b11dyeah
22:26.47BSDTechthey should move the asterisk project to bsd only
22:26.48syzygyBSDdon't have the free time
22:26.57BSDTechfree time 45 min
22:26.58BSDTechok
22:27.08rene-Marty-OTT: Mitel is being used for queues
22:27.18rene-in some of the call center
22:27.20Marty-OTTyes, but shich box?
22:27.22Marty-OTTwhich box?
22:27.22BSDTechfor a basse install and apache
22:27.25rene-i think we have a 330
22:27.26rene-3300
22:27.29syzygyBSDfree time configuring 15 servers to work with BSD... and all the applications we need, custom scripts, etc
22:27.39BSDTechahh ok
22:27.49BSDTechbuild 1 and clone it
22:28.02rene-but everything is super expensive
22:28.02BSDTechthats what gmirror is for
22:28.06syzygyBSDlol, no chance, all our servers have different things running on them
22:28.29BSDTechno you clone the base install
22:28.36Marty-OTTzyaugyBSD: Hardward of software crash?  I've had FreeBSD running for 2 years - Apache, SSL, GNU RAdius, Postgres, Courier Mail, PureFTPD - never a single crash
22:28.37BSDTechthen add the needed pkgs
22:28.44syzygyBSDor, we could just stick with what is working...
22:28.53ManxPowerMost distros let you build an autoinstall
22:28.58syzygyBSDMarty-OTT: hardware
22:29.11BSDTechthen you could have just moved the drive
22:29.12syzygyBSDit was runnign for 4 years without a single problem though
22:29.17BSDTechunless the drive failed
22:29.32BSDTechand you should have had the drive raid 1
22:29.33syzygyBSDbackplane died, and we didn't have a duplicate box
22:29.43Marty-OTTWell... that's not a BSD issue then..
22:29.52syzygyBSDI didn't say it was...
22:29.59BSDTechthen just move the drive
22:30.03BSDTechit will boot
22:30.59Marty-OTTWell, so long as it's the same motherboard chipset..
22:31.04syzygyBSDwe did to pull the info off, but we didn't have a spare 2U server that would support 6 drives
22:31.23b11dbut this goes up to 11..
22:31.51syzygyBSDhad to move to a 5 disk configuration
22:31.55Marty-OTToh..
22:32.26rene-mitel works well but it is not nearly as flexible as asterisk
22:32.30syzygyBSDya, but setting up a server with 4 databases running on it in 2 hours is kinda fun
22:32.31rene-and it is oh so expensive
22:33.44Marty-OTTrene: cool..
22:33.49*** join/#asterisk robin_sz (n=robin@rapid2.gotadsl.co.uk)
22:34.23*** join/#asterisk Aughey (n=jha@64.219.54.121)
22:34.49robin_szhi .. using chan_mISDN, from at least a few landlines it doesnt seem to convert dtmf tones to dialled digits ... how can I make it more sensitive?
22:35.03Augheyhow do I change what is displayed on the display of the phone when its extension is dialed?
22:35.24*** join/#asterisk shido6 (n=shido6@d221-68-200.commercial.cgocable.net)
22:35.25syzygyBSDrobin_sz: change the DTMF mode
22:35.39robin_szright
22:35.49robin_szumm
22:36.00robin_szerr
22:36.02syzygyBSDdtmfmode=inband
22:36.05syzygyBSDor something...
22:36.09b11dAughey.. why dont you get a little more specific eh
22:36.26robin_szsyzygyBSD, in where exactly?
22:36.27b11doh wait no.. let me just get out my copy of "Manual for Every Kind of Phone in The World"
22:36.29*** join/#asterisk [hC] (n=hardcore@mail.metrobridge.com)
22:36.33syzygyBSDI don't do anything with mISDN, but that is normally the solution to your problem
22:36.46syzygyBSDrobin_sz: the top of the configuration file for mISDN?
22:37.16AugheyI have SIP telephones and they have either a 2 or 4 line display.  When you dial an extension, it displays "Call from XXXXX".  I want it to say something else (for extension sharing)
22:37.20robin_szsyzygyBSD, you are nto confusing this with the dtmfmode=rfc1183/inband problem are you?
22:37.37syzygyBSDsounds like that is the problem to me...
22:37.41b11dAughey.. do you think its going to be the same procedure for every kind of SIP phone available?
22:37.47b11dor do you maybe want to tell us what kind of phones you have?
22:38.05syzygyBSDis it just a gain issue?
22:38.21AugheyI have Grandstream and snom phones
22:38.24b11dugh
22:38.25b11dumm
22:38.40b11di have no further contributions :P
22:39.03syzygyBSDpoor people who have grandstreams, so shuned
22:39.17b11dyeah.. im an asshole..
22:39.25b11dits actually just that i have no experience with those..
22:39.55robin_szsyzygyBSD, err ...  sureley dtmfmode is only an option on SIP channels?
22:40.33robin_szI have a grandstream GXP2000 .. its great! I would recommend it to anyone
22:40.53syzygyBSDuhh, I only deal with SIP and PRI, so sorry if I am going in the wrong direction
22:41.04*** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca)
22:41.08robin_szits wonderful for stopping cars rolling on a slope
22:41.17*** join/#asterisk Dr-Linux|home (n=Dreamer@DSL-202-59-73-131.nexlinx.net.pk)
22:41.45robin_szor for putting under your pillow to provide a better back angle when relaxing reading a book in bed
22:41.50*** join/#asterisk Qwell[] (i=qwell@unaffiliated/qwell)
22:41.50*** mode/#asterisk [+o Qwell[]] by ChanServ
22:41.57Dr-Linux|homehi
22:42.34robin_szI did try using it as a phone once, but that was disastorous, I dont think it was designed for that purpose
22:43.16b11dhahaha
22:43.27b11drobin_sz.. i just imagine you at a conference saying that stuff, all straight faced..
22:43.51robin_sznah
22:44.11syzygyBSDnot the conference going type?
22:44.19robin_szid be out in the car park, making money  ... "grab a hammer, smash a grandstream, only $2 a time"
22:45.01syzygyBSDany tech conference could make a ton off that
22:45.13robin_szunless anyone can think of a way of getting reid of GXP2000's for more than £2 a pop
22:45.23[hC]Has anyone experienced an issue with analog lines in asterisk where someone tries to place a call out, "seemingly" when a call is trying to come in at the same time, and instead of being connected to the number that the person was trying to call OUT to, they are connected to the new incoming call?
22:45.29*** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com)
22:45.38[hC]Maybe an issue where a line does not hang up quick enough before asterisk tries to sue that channel to call out on again?
22:45.46[hC]sue=use
22:47.39Strom_M[hC], that's a problem called glare
22:47.53Qwell[]~glare
22:47.54jbotACTION glares at qwell[]
22:48.02Strom_Msilly qwell
22:48.03Qwell[]okay then
22:48.13Strom_M[hC], how many analog trunks do you have?
22:48.34b11dit wont
22:48.37b11dits already fried
22:48.40robin_szok, there seems to be a "dtmf threshold" value in the misdn module config
22:48.48robin_szbut its in milliseconds
22:48.52b11dyou hooked it up to the power supply again didnt you?
22:49.00robin_szhows that work then?
22:49.10robin_szI think this is a low gain sort of problem
22:49.17b11dmessing with gain sucks
22:49.20robin_szsending digits might be a bit quiet
22:49.23sweeperit did
22:49.26sweeperyay \o
22:49.49b11dexcellent
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22:52.54syzygyBSDrobin_sz: are you sure they are quiet?
22:53.09syzygyBSDie, have you recorded a call having problems?
22:53.24*** join/#asterisk jvalenzuela (n=jvalenzu@wingnut.dspfl.com)
22:53.42x86what do you guys think of my new web design: http://www.shellshark.net/
22:54.17b11ddoenst look bad
22:54.17syzygyBSDwow, a redirection page
22:54.17sweeperis nice, but redirects and bad ssl certs are bad
22:54.29sweeperjsut use mod-rewrite
22:54.39syzygyBSDloads kinda slow
22:54.45syzygyBSDbut looks good
22:55.03BSDTechok man I hate oh323
22:55.08sweeperyour "concurrent calls" thing could use some cleaning up
22:55.19BSDTechI have to kill it in the atserisk build for 1.4
22:55.52syzygyBSDx86: I would recommend you reorder the number of concurrent calls and place them next to eachother
22:56.13robin_szsyzygyBSD, no, im not sure, but certainly some analogue lines seem to have trouble and from what i can see, mobiles don't
22:56.24x86syzygyBSD: yeah, that's on the todo list already :)
22:57.45syzygyBSDhmm, the phone numbers seem a bit wrong too
22:58.46*** join/#asterisk justdave (n=dave@unaffiliated/justdave)
22:58.59*** join/#asterisk alamantia (i=Anthony@nat/digium/x-cb1f27dbaed7fec0)
22:59.34syzygyBSDI'd suggest a print link on terms of service too.
22:59.46syzygyBSDwish I could do graphic design
22:59.58[hC]Strom_M: 7.
23:00.06robin_szot looks nice enough
23:00.08robin_szit
23:00.32robin_szalthough I suspect its all hard-coded in some Perl scripts
23:01.16*** join/#asterisk xheliox (n=jeff@pdpc/supporter/active/xheliox)
23:01.30x86syzygyBSD: there is a print link, down at the bottom
23:02.06x86robin_sz: already using a template system, no HTML is hard-coded in perl ;)
23:02.15x86well, XHTML even ;)
23:02.18robin_szx86, mason?
23:02.25x86home built
23:02.39robin_szummm ... why?
23:02.40syzygyBSDx86, where is the print link?
23:02.53x86syzygyBSD: down by the copyright in the bottom footer
23:03.17x86robin_sz: for reasons I can not disclose here ;)
23:03.20syzygyBSDahh, k, I was viewing the version off the menu
23:03.43x86robin_sz: the CMS engine ties in with everything else we have, basically
23:04.08x86robin_sz: so we decided to build one from the ground-up to get complete integration, while maintaining maintainability ;-)
23:05.07robin_szx86, so the template system can read the same templates as use dby your CMS?
23:06.40x86robin_sz: what you are seeing is the CMS in action
23:07.08robin_szright, time to go and play with my robot
23:07.25sweeperbut perl D:
23:07.40sweeperruby on rail for me prz
23:07.43*** join/#asterisk sbingner (n=sam@pdpc/supporter/sustaining/sbingner)
23:10.50*** join/#asterisk WAudette (n=WAudette@c-71-237-146-239.hsd1.or.comcast.net)
23:12.30Strom_M[hC], are inbound and outbound calls hunting from different ends of the group?
23:14.13WAudetteIs anyone here familiar with Paging through the Sound Card?  It is working but, through chan_oss.  Are there any switches that I can set in (console/dsp) to tell an extention to not autoanswer in effect making it Ring?  Then I could have another extention page like normal with just (console/dsp)?
23:14.53Strom_MWAudette, huh?
23:15.13Strom_Mi'm not clear on what you're trying to do
23:15.17WAudetteIn oss.conf if I set autoanswer=no it rings but I want to set it per exention and not for the whole channel.
23:15.26Strom_Mwhy?
23:16.07WAudetteOverhead Paging in a warehouse... They want it to produce a ringing sound when a call comes in over the PA System.  They want to Intercom Page via the same PA System.
23:16.38Strom_Mwhy would calls come in via the PA system?
23:16.41WAudetteI can do one or the other so far, but can't figure out how to fanagle it to do either or.
23:16.45Strom_Mthat makes no sense
23:16.57WAudetteThey woldn't actuall come in... Just produce the ring sound.
23:17.07Strom_Moh, ok, your grammar is screwy
23:17.16Strom_Mso you want to use the PA as a ringer
23:17.22WAudetteYeah... Train of thought typing... sorry.
23:17.42WAudetteYes, and as an intercom.
23:17.55nortexWAudette, What paging system are you using?
23:18.07Strom_Mwell, paging moreso than intercom; intercom implies two-way communication
23:18.17WAudetteA legacy Valcom PA from the system I am replacing w/ *
23:18.31Strom_MWAudette, i'm trying to think of what you can do...
23:18.36WAudetteIt has a closed input... meaning it is simple and alway off hook.
23:18.42*** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca)
23:19.07Strom_Myou could generate a .call file when the call comes in, and have that .call file call console/dsp with a ringing sound or a message saying "call from [whoever]"
23:19.22WAudetteStrom_M: Good point on the terminology.... Paging is more specific and what I meant to ask.  Sorry.  :)
23:20.49nortexWAudette, To get a ring on the sound card you can turn off autoanswer on the chann_oss, then dial it with the incoming call.
23:20.55Strom_Mnortex, but that's not going to work in his situation
23:20.59Strom_Mhe wants to use it as a PA as well
23:21.03WAudetteOk.  There aren't any switches that you can think of like (console/dsp, ,A(custom/bosun)  but one that would tell chan_oss not to answer?
23:21.03Strom_Mso better to use the .call file
23:21.11Strom_Mnot AFAIK
23:21.46nortexSo you want to use the sound card to page and be the loud ringer?
23:21.53WAudettenortex:  Yes, that does work... but sets it up to no answer for all calls to (console/dsp)  and I still need the Paging system output too.
23:22.17Strom_MWAudette, see my solution
23:22.34WAudettenortex: Correct  Page and loud ringer... Excactly.
23:23.18nortexOk, That is different then mine, I use ZAP to page and the sound card to ring though a music input.
23:24.01*** join/#asterisk linlin (i=techpeps@71.194.70.13)
23:24.06nortexI mainly use a Viking system, but I have used a valcom with FXO and FXS inputs.
23:24.07WAudetteStrom_M:  Yes, to make a .call file.
23:24.08mitchelocyou could get a voip overhead speaker
23:24.23WAudetteThinking about it.
23:25.07mitchelocprobably not as cheap though
23:25.28WAudetteBut then I have to find a VOIP overhead speaker that does the same thing... Paging + Ringign.
23:25.32justdaveI'm trying to set up an inbound IAX trunk.  It registers successfully, but inbound calls generate an error in asterisk "Rejected connect attempt from <iax provider ip>, who was trying to reach '<our phone number>@'
23:25.33sweeperor get a cheep pc and put a softphone set to autoanswer on it
23:25.34nortexmitcheloc, I looked at those and was suprised at the cost, at least for small systems. The Viking one I've been using is $160 with the system and 1 8-ohm horn.
23:25.53*** join/#asterisk RoyK (n=roy@217-175-39.100710.adsl.tele2.no)
23:25.53BSDTechwaudette ok hire me
23:25.56justdaveanyone know what I should look for?  been looking at iax docs for a while and not seeing what I'm doing wrong
23:26.04WAudetteStrom_M:  Do you have an example .call file that I could work with.  I have never made one.
23:26.24WAudetteBSDTech:  Ok, I can do that.  <grin>
23:26.30Strom_Mno; check voip-info
23:26.35nortexjustdave, Does the number your calling exist in the context the IAX peer is in?
23:26.46WAudetteStrom_M:  Thanks for your input!
23:27.17BSDTechI will move up there in a heart beat
23:27.23justdavethere's an extension defined with the DID number, if that's what you're asking
23:27.42justdavein the from-outside context, which is the context defined in iax.conf
23:32.06nortexjustdave, can you pastebin the relevant iax.conf and extensions.conf sections?
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23:37.24justdavenortex: http://pastebin.mozilla.org/2796 http://pastebin.mozilla.org/2797
23:37.39justdavethe xxxxxs are actually the phone number of course
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23:39.10justdavethe extension stuff is identical to the from-outside-1650xxxxxxx-tl-allhours context is set up identical to one for a different inbound line (except that one comes in via SIP)
23:43.03justdavemy first thought was to add the @ on the end of the extension number since it was including it there on the inbound, but it didn't like that
23:43.08nortexSo the second pastebin is the context from-outside
23:43.17justdavecorrect
23:45.09justdavehmm, actually, that's in from-outside-redir
23:45.14justdavebut from-outside has this:
23:45.14justdaveexten => _X.,1,Goto(from-outside-redir,${EXTEN},1)
23:45.14justdaveexten => s,1,Goto(from-outside-redir,${EXTEN},1)
23:45.55nortexAhh, try commenting out the s
23:47.42riddleboxdoes anyone know why in fedora I cannot get my music on hold working?
23:48.09riddleboxall I get is some little static
23:48.20b11dcan you play the mp3 manually with mpg123 ?
23:48.25b11dand hear it over your speakers?
23:48.56nortexjustdave, That may not solve it, you might also try setting iax.conf context to from-outside-redir
23:48.57*** join/#asterisk Globetrotter (n=eric@205.211.214.167)
23:49.01justdavethe s is the default inbound isn't it?
23:49.24riddleboxblld, the problem is I dont have anything hooked up to the box, it is a stand alone machine plus on fedora it is mpg321 but there is a link to mpg123
23:49.29nortexyeah, s should be used if nothing matches, if I remeber right.
23:50.00justdavewe have other phone numbers so I'd prefer not to screw with that :)  I'll try changing the context on the iax.conf
23:50.15justdavenope, that didn't work
23:50.22GlobetrotterHi Guys,,  finally got my MOH towork.. how to i make it play the files at random..  mode=files, random=yes..  but no good
23:50.25justdavesame error
23:50.32Globetrotterplays the same files
23:50.33*** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal)
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23:50.49Globetrotterand i have mulitles files in my folder
23:51.33nortexjustdave, Does the number match exactly or is it like a DID with 10 digits or 4 instead of all 11?
23:51.54riddleboxblld, I guess I could hook up a headset and see, hold on
23:52.00justdaveit's 11 digits
23:52.08nortexjustdave, It is really odd to me that the number@context in the error has a blank context.
23:52.29justdavedoes IAX need to know the context on both ends?
23:52.40justdavemaybe I need to set it somewhere on the provider's control panel
23:53.46nortexNo your just telling it where to put the calls as they come to your system.
23:55.25nortexjustdave, Sorry I didn't help much.
23:56.09Dr-Linux|homeanybody is using agent/queue system?
23:59.39WAudetteStrom_M:  Do you have any hints on what to search for?  .call seems to be a wildcard of sorts catch almost everyting in voip-info.org.
23:59.54Strom_M"call file"
23:59.55Qwell[]WAudette: "call file"

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