IRC log for #asterisk on 20080908

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00:15.49phl4kxhi all
00:16.14phl4kxI have a AEX 800P with 1 FXO port but the card never anserw the call
00:16.26phl4kxwho can help me please
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00:17.39mouahi
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01:06.28TJNII/dev/md1              2.1T  115G  2.0T   6% /mnt/mediaHD
01:06.35TJNIIgiggles gleefully
01:07.32*** join/#asterisk jeev (n=email@unaffiliated/jeev)
01:09.12mouaanyone ?
01:11.04mouai have a sip account, i wish to get a busy signal on my DID when someone call it, and it should trigger a webpage (i want to activate a php script by call)
01:11.13mouasomeone know how to do this ?
14:57.25*** join/#asterisk jbot (i=ibot@pdpc/supporter/active/TimRiker/bot/apt)
14:57.25*** topic/#asterisk is Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.6.0-rc4 (2008/09/03), 1.4.22-rc3 (2008/09/03), *-Addons 1.6.0-rc1 (2008/09/03), 1.4.7 (2008/06/04), dahdi-linux-complete 2.0.0-rc3+2.0.0-rc2 (2008/09/03), Zaptel 1.4.12 (2008/09/03), Libpri 1.4.7 (2008/08/05) -=- Related channels: #asterisknow #asterisk-gui #switchvox #freepbx #asterisk-commits #asterisk-bugs #asterisk-dev. AstriCon! #astricon
14:57.48ManxPowerabout time you got here, jbot!
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14:59.26MrTelephonemanx
14:59.29MrTelephonehelp me with the digest :P
14:59.41MrTelephonehave you ever seen clients calculate the digest retardedly?
15:00.06jayteehahaha, jbot overslept!
15:00.22MrTelephonejaytard
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15:10.06*** join/#asterisk jbot (i=ibot@pdpc/supporter/active/TimRiker/bot/apt)
15:10.06*** topic/#asterisk is Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.6.0-rc4 (2008/09/03), 1.4.22-rc3 (2008/09/03), *-Addons 1.6.0-rc1 (2008/09/03), 1.4.7 (2008/06/04), dahdi-linux-complete 2.0.0-rc3+2.0.0-rc2 (2008/09/03), Zaptel 1.4.12 (2008/09/03), Libpri 1.4.7 (2008/08/05) -=- Related channels: #asterisknow #asterisk-gui #switchvox #freepbx #asterisk-commits #asterisk-bugs #asterisk-dev. AstriCon! #astricon
15:10.14MrTelephoneit's crazay
15:10.29*** join/#asterisk Vec (n=Vec@host-87-74-7-50.dslgb.com)
15:10.37MrTelephoneI had to edit chan_sip.c to log notice asterisks calculation so I can compare it to the response
15:10.44ManxPowerMrTelephone: or maybe the port changes and it's not reregistering.  set the register or nat keepalive on the device to a low setting
15:10.58MrTelephonei'll give it a try
15:11.05MrTelephoneits not behind a nat though
15:11.30MrTelephonedoes anyone have a little digest calculator :P
15:15.30MrTelephonejust checking my logs now and there are some ACL issues
15:15.31MrTelephonecaca
15:15.48MrTelephone10.1.3.85 registers with permit acl 10.1.3.0/255.255.255.0 and it denied it
15:17.43ManxPowerDid it register or did it get denied?
15:19.15MrTelephonenah i took the user out of sip.conf and it gives me that message.. wierd.. it usually says authenticated failure instead
15:19.28MrTelephonei should upgrade to 1.4
15:22.40ManxPowerso basically you are just trying random things rather than actually trying to troubleshoot it
15:23.49bakermdAre there windows apps that run in the system tray and comminucate with Asterisk such that I can see phone numbers on the web highlighted, and when I click one it places the call through asterisk to my desk phone?
15:24.21[TK]D-Fenderbakermd: There are Firefox add-ins for this.  Go browse for them
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15:24.32bakermd[TK]D-Fender, Will Do - Thanks
15:26.58fogoI've got a SIP phone that will register, but when they try to make a call, it goes unreachable. Shouldn't it be able to stay reachable if it can register initially?
15:27.56[TK]D-Fenderfogo: No, and your description could be a lot fuller
15:28.24*** join/#asterisk dr_gogeta86 (n=dr@81-208-88-100.ip.fastwebnet.it)
15:28.28dr_gogeta86hi to alle
15:28.30dr_gogeta86*all
15:28.55dr_gogeta86who here use asterisk to recive faxes;
15:29.56fogo[TK]D-Fender: alright, so I have a remote user that has x-lite; I don't know fully what his setup is; most likely NAT'd. * has a public IP, with many other outside users registered w/o problems
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15:30.48fogo[TK]D-Fender: the user in question can register, but when he goes to make a call, his phone changes state to unreachable
15:31.08ManxPowerfogo: do you have qualify=yes ?
15:31.25[TK]D-Fenderprobably not...
15:31.30[TK]D-Fenderfogo: ...
15:31.40fogoI believe so.. checking for sure
15:31.41*** join/#asterisk yidiyuehan (n=yidiyueh@bb121-6-113-252.singnet.com.sg)
15:32.02ManxPowerI assume you have nat=yes too
15:32.15fogoyes, qualify=yes, nat=yes
15:32.24[TK]D-Fender[~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3  otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions
15:32.26yidiyuehanhi, guys, any one knows whether it's possible to play a pre-recorded file to the called party before he can talk?
15:32.44ManxPowerfogo: odd.
15:32.46[TK]D-Fenderyidiyuehan: "core show application dial" <-
15:33.02ManxPowerfogo: you are not portforwarding anything on the client router?
15:33.21yidiyuehanD-Fender, thanks for the guide, if you have some web reference, I will really appreciate it
15:33.40fogoManxPower: I don't think they have any ports forwarded currently.
15:33.59[TK]D-Fenderyidiyuehan: web reference?
15:34.12[TK]D-Fenderyidiyuehan: Go to * CLi and read Dial's instructions.
15:35.08yidiyuehansure, brother,
15:35.15yidiyuehani am looking at it.
15:35.38yidiyuehanbut before that, is it possible to implement what I want?
15:36.04ManxPoweryidiyuehan: It is already a feature of the Dial command.  That's why [TK]D-Fender told you to look there.
15:36.36jameswf-home~book | yidiyuehan
15:36.54yidiyuehanok, stupid I am, thanks to both of you man
15:37.20ManxPoweryidiyuehan: how well it works depends on your interface to the telco.
15:37.52The-BatHie
15:37.56The-BatHow can I execute a shell script without using System()
15:38.15ManxPowerThe-Bat: you could write an AGI.
15:38.29yidiyuehani will implement using simple analog telephone lines, once it's answered, one IP phone will pick up the call,and hear a pre-recorded audio file, after that it can start talking.
15:38.30ManxPowerbut the real question is "why don't you want to use System()
15:38.43The-Batit gives me a APPERROR
15:38.53ManxPoweryidiyuehan: Asterisk does not know when the far ends picks up on analog ports.
15:38.54The-Batm unable to debug it
15:39.03ManxPowerThe-Bat: then you are doing something wrong.
15:39.14ManxPowerThe-Bat: try starting asterisk as "asterisk -cvvv" for testing.
15:39.18ManxPowerthen try it.
15:39.22The-Batit executes properly wen i run it manually
15:39.29The-Batok let me chk that
15:39.43The-BatManxPower, thx
15:39.46ManxPowerthat runs asteirsk in the forground so you should not do that on a production system
15:40.01yidiyuehanmanxpower, asterisk will know once somebody call in the fxo ports am i right? the other end will be the IP phone
15:40.36[TK]D-Fenderyidiyuehan: * will process the call however you set it up to
15:40.39ManxPoweryidiyuehan: yes.
15:41.01ManxPowerit's just calls going OUT an FXO signalled port that has this issue.
15:41.29ManxPoweryidiyuehan: if you sick to IP phones as the destination it should not be a problem.
15:41.29yidiyuehanmanxpower, then it should be fine, as I just want to play to the IP Phone, one way only.
15:41.54yidiyuehanManxPower,that's great, i will go and study the Dial application
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15:46.12dr_gogeta86Anyone here play asterisk to income faxes
15:47.20fogoManxPower: back to the previous question - it looks like the soft phone in question has a publicly-addressable IP as well (bypassing their router (NAT)) and they still can register, but when they go to make a call, status changes to unreachable. Any guesses?
15:47.56ManxPowerfogo: many guesses all involving a problem with the client router.
15:48.17ManxPoweryou're not doing something stupid like using bindip= or port= in sip.conf?
15:49.06fogoManxPower: hmm... port=5060
15:49.35ManxPowerDONT'T DO THAT
15:49.35AssidManxPower: you know any provider like voicepulse BUT white labelled so i can port incoming there.. without really disclosing who the end provider is?
15:49.43ManxPowerAssid: I don't know of any provider that would be suitable for the requirements you have.
15:49.43Assidsomeone that handles number portabilityh and gives unlimited incoming but based on channels
15:49.55Assidhrmm crap
15:50.05ManxPowermaybe you should start contacting carriers rather than asking on this channel every day
15:50.22Assidim trying
15:51.09*** part/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net)
15:51.40Assidcant find a carrier with similar features
15:52.18ManxPowerAssid: I doubt one exists.
15:52.51Assidthats not good :|
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15:54.29fogoManxPower: I nixed the port=; still having the same problem
15:54.56VJFROMGTi did a yum update and lost my crontab ,, anyone know how i can get it back (backup has new version)
15:55.03ManxPowerfogo: put your sip.conf on pastbin.con masking ONLY the passwords
15:56.38fogoManxPower: http://pastebin.ca/1197300
15:57.13[TK]D-Fenderlol
15:57.17ManxPowerfogo: if that is your entire sip.conf no wonder you are hving problems
15:57.18[TK]D-FenderManxPower: run :)
15:57.26fogooops.. entire
15:57.38fogo:)
15:57.38ManxPower[TK]D-Fender: if he can't follow simple instructions I doubt he can be helped.
15:57.57[TK]D-FenderManxPower: You shouldn't need to ask for the full file.  You should already know what you're looking at.
15:58.25ManxPower[TK]D-Fender: I'll go ask Cleo
15:58.41[TK]D-FenderManxPower: Don't need to be psychic...
15:58.44fogowell, I can post the entire file, but it contains roughly 1000 sip entries in it
15:59.10ManxPowerfogo: nevermind.  You are using FreePBX or some other sort of spawn of Satan GUI, aren't you?
15:59.15[TK]D-Fenderfogo: What he wants to see is [general]
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15:59.47fogoManxPower: yes, I am using a spawn of satan GUI
15:59.54ManxPowerTherefore I will not help you.
16:00.05ManxPowerBest of luck and maybe you should ask on the correct channel.
16:00.14fogoalright. thanks :)
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16:16.55doolphx
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16:17.45mighty-dHi
16:17.50doolphhi
16:19.24mighty-dim learning all-i-can from asterisk, and im not sure about something, it is codec-related, there are a lot of card vendors in the market, is there any special consideration with codecs like 711, GSM, etc.. with the cards, i mean is the codec applied by the card or is this done by the OS ?
16:19.57[TK]D-Fendermighty-d: Codecs have nothing to do with cards for your server
16:20.15mighty-das i tought, thanks TK-Fender :)
16:23.03*** join/#asterisk FlyboySR22 (n=rsears@hq.fw.americanis.net)
16:29.05mighty-dWhat should i use to protect analog cards from electrical hazzards?
16:29.25jayteeline surge suppressors
16:30.10jayteeand if you have POTS lines from your telco they "should" be wired already with a PET (protected entry terminal)
16:30.49jayteebut alot of telcos are cheap and don't install them except for their larger customer accounts.
16:31.01doolphlol
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16:32.39jayteeyou can also get CAT5 suppressors too for ethernet connections. we use em here at the zoo because we get surges all the time when there are lightning strikes nearby.
16:34.39drmessanoAPC
16:34.49*** join/#asterisk qdk (n=qdk@0x573fe57e.bynqu2.static.dsl.tele.dk)
16:35.08jayteethat's what we use
16:35.15jayteeI swear by APC
16:35.33jayteeTripplite,.....not so much
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16:36.51*** join/#asterisk ManxPower (n=manxpowe@130.sub-75-249-91.myvzw.com)
16:37.54jayteehad a flood once in a basement server room due to a broken line in an upstairs dishwasher. Water was almost touching the motherboards of the 3 primary servers before we got in to kill the power. 4 computers plugged into an APC surge suppressor were still good because the APC gave up it's life to protect them. Called APC to tell em how good their stuff was and they said they'd send a replacement. I said, but it wasn't your products fault and it did
16:37.54jayteeit's job.
16:38.19jayteethey said it didn't matter, still covered under the warranty.
16:42.22drmessanonice
16:42.58drmessanoI have had two APC's almost catch on fire, yet I still use them because the others have saved my bacon more often than not
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16:43.44denonwe run a ton of APC, but I trust Liebert a lot more
16:47.35Juggieour system @ work is leibert
16:47.52Juggiewe have enough batteries to power an entire lab for 12hrs :)
16:48.02Juggieand thats only incase the generator doesnt start
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16:51.24tvirusI created 2 queues, http://rafb.net/p/KhcSQo42.html . My goal is to have it ring queue 901 if no one picks up on 900. This is inside extensions.conf http://rafb.net/p/MyRlX298.html . Queue 900 works fine but if I dial 901, I get a busy signal and if I dial 900 and wait it just rings and then hangs up after 30 seconds or so.
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16:53.50jayteeall these doommongerers going on about the LHC firing up wednesday. "oh, no! we're all gonna die!!!". I swear cranial anal embedding has reached epidemic proportions.
16:53.55*** join/#asterisk sah-work (n=Bawbatos@65-119-47-100.dia.static.qwest.net)
16:54.01mv2can someone explain AST-2008-003 ?
16:54.22tvirusI'd rather die from a black hole opening in the LHC than from a car accident.
16:54.29*** join/#asterisk Levonk (n=lk@76.237.13.156)
16:55.05jayteemv2, yeah it's 3 letters and a group of 4 numbers and then a group of three numbers all seperated by a dash. :-)
16:55.51mv2jaytee:thanks
16:56.24Kobazis there an easy way to play ringback (instead of music) when dialing into a queue, and periodicly play a message saying please hold blah blah
16:56.42Kobazi can pass an r to the queue for ringback, but then how would i play a message
16:59.07*** join/#asterisk AndyML (n=alauppe@pool-71-185-74-86.phlapa.fios.verizon.net)
17:00.29AndyMLhey there. I have a VoiceMailMain() channel stuck. any ideas on how to kill it short of restarting asterisk?
17:00.54QwellAndyML: soft hangup <channel> from the CLI
17:01.04x86omg I just a whole fleshlight
17:02.43JG7634My company has two offices both with nortel norstar digital pbx and digital phones (52/16 phone stations). The two offices are trunked with Citel Gateway 2 and Citel Extender 7000. We have a 5Mb ethernet between the offices. We would like to replace the trunking Citel equipment with a Asterisk Trunking Solution and keep using the existing pbx phone systems with extension dialing still...
17:02.45Qwell...what?
17:02.45JG7634...working. Is this possible with Asterisk? Thanks.
17:03.05QwellJG7634: how are they connected?  what protocol?
17:03.11ManxPowerKobaz: you can't have ringing and a message
17:03.19KobazManxPower: that's what i thought
17:03.33Kobazi can do a ring moh though
17:03.50ManxPowerthat's not ring, that moh that sounds like ringback
17:03.55[TK]D-Fendertvirus: It'd be real nice if your 901 exten had a priority 1 in there....
17:04.03JG7634qwell: rvp over ip
17:04.12*** join/#asterisk fskrotzki (n=fskrotzk@host.textwise.com)
17:04.19ManxPowerJG7634: Asterisk does not support RVP
17:04.22QwellJG7634: then no, Asterisk doesn't support that
17:04.47[TK]D-FenderAndyML: "soft hangup [channelwithoutbraces]"
17:04.56drmessanoRob Van .. oh nevermind
17:04.58Qwell[]=optional :p
17:05.12*** join/#asterisk c4t3l (n=root@74.95.210.124)
17:05.30x86http://forums.joerogan.net/showthread.php?t=74612
17:05.58JG7634qwell: if we are replacing the citel equipment we dont have to use the same protocol right? what would Asterisk use?
17:05.59jayteethe devs should build in a couple of settable parameters in queues.conf to allow for RAN announcements. Like ranmessage1= filename, i.e. "Your call is totally unimportant to us, please continue to be an idiot and hold" and a ranmessageinterval=30 for every thirty seconds.
17:06.13QwellJG7634: presumably SIP, if the other box supports it
17:06.24QwellJG7634: are you replacing both sides?  maybe I misunderstood
17:06.52jayteeI can do first and second level RAN announcements, interflow, overflow and nite service treatments for Nortel ACD queues but * is kinda "weak" in that area.
17:07.25JG7634qwell: keeping both offices digital pbx and phones. Just thinking to use Asterisk to create a link between the offices.
17:07.46Qwellso, 1 Asterisk box at each office, and a PRI link to the PBX or something?
17:08.20JG7634qwell: yes i think on those lines
17:08.43QwellSIP or IAX2 probably
17:08.54ManxPowermost PBXs are very limited in their ability to integrate with 3rd party stuff like Asterisk
17:09.05JG7634qwell: the citel equipment current links the two offices but the equipment fails often
17:09.41JG7634qwell: over point to point ethernet
17:09.42ManxPowerI thought Citel was a device to convert digital phones into SIP?
17:09.51c4t3lyes it is
17:10.11JG7634manxPower: not the gateway 2 and extender 7000 they are propriatery I thought
17:10.27tvirusThat fixed it from not working, thanks, TK.
17:10.31ManxPowerJG7634: Don't think.  Know or you will waste money
17:11.28ManxPowerJG7634: how is the Citel connecting to your PBX.
17:11.38c4t3li demo-ed the citel gateway equipment.  It seemed pretty cool.  But I've never seen it work in the real world
17:12.23JG7634manx: pri t1 card
17:12.41ManxPowerJG7634: and you are SURE it's a PRI, not a channelized T-1?
17:12.56ManxPowerbecause nortel charges like $3,000 extra to enable PRI support on their PBXs
17:14.01JG7634manx: its a large multipin plug about the size of a thumb, rj-21 cat 3 punchdown block
17:14.14ManxPowerthat would be analog
17:14.26ManxPowerPRI uses exactly 4 wires (2-pair)
17:14.26Qwelleww...
17:14.41*** part/#asterisk mv2 (n=mv@83.240.229.38)
17:14.41ManxPowerIt uses an RJ-45/RJ-48C
17:15.34ManxPowerSo you have PBX <- 25 pair AMP -> Citel?
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17:17.10ManxPowerJG7634: I don't see a "gateway 2" model from Citel
17:17.33JG7634manx: PBX <- 25 pair AMP -> Citel Gateway II-> 5Mb Ethernet <- Citel Extender 7000 <- 25 pair AMP <- PBX
17:17.51Qwellthat's nasty
17:18.03ManxPowerJG7634: looks analog to me.
17:18.08[TK]D-FenderYUP
17:18.14ManxPowerwhat EXACT model is the Citel Gateway II
17:18.14*** join/#asterisk Thorn (n=thorn@unaffiliated/thorn)
17:18.20AndyML[TK]D-Fender: soft hangup doesn't seem to work for this channel. if i showed you a CLI output do you think you could offer anything else?
17:18.24JG7634manx: ok yes it is Analog equipment with a digital line
17:18.36ManxPowerJG7634: it will never ever work with Asterisk
17:18.54[TK]D-FenderAndyML: CLI of your attempt & the channel dump would be nice.
17:19.00AndyMLwill do
17:19.14Qwellsure it would..  TDM2400 does AMP
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17:20.08ManxPoweryou don't want to use analog
17:20.22ManxPowerQwell: and since it's a Citel it's not analog.
17:20.33Qwelloh...
17:20.35ManxPowerLooks like they use PBX station ports or something like that.
17:20.39Qwellyuck
17:20.40[TK]D-FenderManxPower: makes you wonder WTF is going on actually...
17:20.47AndyML[TK]D-Fender: http://pastebin.ca/1197371
17:20.52[TK]D-FenderManxPower: Could indeed be 2-wire station ports
17:21.07JG7634manx: http://www.citel.com/Products/EXTender.asp ill get you model number
17:21.44tvirusWhat's the proper syntax (1.4.x) for a queue to go to the next queue if no one picks up in 25 seconds? exten => 900,1,Queue({$EXTEN}|t|||25)
17:21.47[TK]D-FenderAndyML: Try an AMI redirect to a dead-end
17:22.02[TK]D-Fendertvirus: "core show application queue"
17:22.08AndyML[TK]D-Fender: not sure what you mean.
17:22.10ManxPowerhttp://www.citel.com/Products/EXTender/Citel_PBX_Gateway_II.asp
17:22.28[TK]D-Fendertvirus: Whats the point of doing queue's for so short a period of time?
17:22.38tvirusCall center.
17:22.38[TK]D-FenderAndyML: Go read up on AMI
17:22.39AndyML[TK]D-Fender: can you just grab a channel in AMI and redirect it?
17:22.46tvirusTo escalate a call if no one answers
17:22.50[TK]D-FenderAndyML: thats the idea
17:22.55AndyMLi'm familiar with AMI. just not with every feature. ok - will do.
17:23.09[TK]D-Fendertvirus: first should just be a Dial then
17:23.17linuxer_igorhelp, I have a problema with DAHDI
17:23.26*** join/#asterisk Ziaeon (n=Ziaeon@c-71-57-142-206.hsd1.fl.comcast.net)
17:23.29linuxer_igorupdate my server today for 1.0.6rc4 and the DAHDI don't work
17:23.40linuxer_igorstay up but do not register
17:24.03ZiaeonWhy does random play not work for me? It's all set up properly, random = yes, it just plays the same song over and over. The strangest thing is that it's not even the first song, alphabetically.
17:24.35Ziaeon(For MOH)
17:25.19ManxPowerhttp://metrocomminc.com/images/verso/extender_pbxgateway_ii.pdf
17:25.36ManxPowernotice the weird "telephony interface" on the picture of the back.
17:26.16JG7634Manx: yes
17:27.50JG7634manx: thats where the pbx connects in
17:29.01ManxPowerJG7634: Best of luck.
17:29.24[TK]D-FenderOh yeah, thats a Norstar Digital amphenol trunk device all right...
17:29.33JG7634manx: we want to get rid of that box though what are some equipment that can get us... analog pbx -> ? -> Asterisk -> 5mb ethernet
17:29.52[TK]D-FenderJG7634: If you don't have a real T1 port on that thing you're in for a real PITA job
17:30.09JG7634T1 port on the pbx?
17:30.13[TK]D-Fenderyes
17:30.15ManxPower[TK]D-Fender: no wonder it doesn't work well.
17:30.23[TK]D-FenderManxPower: Not an excuse for it
17:30.48ManxPower[TK]D-Fender: We are talking about Citel here.
17:30.55[TK]D-FenderJG7634: How many lines / stations at each site?
17:31.11JG7634fender: 56/16
17:31.24ManxPoweryou have 56 lines and only 16 stations?
17:31.43ManxPowermight want to get rid of some of those lines.
17:31.48[TK]D-FenderJG7634: You're almost better off ripping it all out...
17:31.50JG7634woops, 52 stations 13 lines
17:32.01ManxPowerJG7634: stop being inaccurate.
17:32.05JG7634other office 16 stations, dont know lines
17:32.45QwellJG7634: might as well rip everything out, and replace it all
17:32.46JG7634manx: i dont know everything, good luck on accuracy
17:32.46tvirus[TK]D-Fender: It's a queue because they dial in and get the voice menu and push 1 or 2 for Support/Sales and get into that queue.
17:33.04ManxPowerJG7634: then you are guaranteed to fail.
17:33.16[TK]D-Fendertvirus: Its a queues because YOU send them there.
17:33.26JG7634manx: this just feasability, professional will do lol
17:34.04ManxPowerJG7634: you can't find out feasibility if you don't know exactly how things are set up and what protocols are used.
17:34.29JG7634manx: we have to start somewhere
17:34.44ManxPowerDo you really want to find out it's not going to work when you blow up the asterisk card because you plugged it into a station line?
17:35.04ManxPoweror blow the line on the PBX?
17:35.04[TK]D-FenderJG7634: Go see if you have a real PRI port available on it.
17:35.30JG7634fender: ok checking for pri
17:35.31ManxPowerThis PRI port would look similar  a standard Ethernet port
17:35.38*** join/#asterisk linuxstb (n=linuxstb@rockbox/developer/linuxstb)
17:36.52*** join/#asterisk luca`gervasi (n=ashura@host76-170-dynamic.21-87-r.retail.telecomitalia.it)
17:36.55luca`gervasiHallo
17:37.29luca`gervasii need to test my line...can someone tell me a simple function to say "All right" ?
17:37.44[TK]D-Fenderluca`gervasi: test your line by USING it.
17:37.52ManxPower[TK]D-Fender: on a Nortel MICS you have to pay extra (thousands of dollars extra) for PRI and even MORE for support for "PRI Tie-line"
17:38.10[TK]D-FenderManxPower: Life sucks but rarely swallows
17:38.15luca`gervasiwell, when i try to dial a number, it says "no such extension found" :D
17:38.52[TK]D-Fenderluca`gervasi: then you should go look at what context its looking for that exten in and what exten it is looking for.  Then maybe you'll see why it isn't finding a match
17:40.07JG7634fender: our pbx has a T1 connected into a PRI
17:40.37Qwellyou could put Asterisk in front of it...
17:41.04QwellTelco > Asterisk > WAN > Asterisk > Telco AND, Telco > Asterisk > existing PBX > Digital phones (on both sides)
17:41.05JG7634qwell: so t1>Asterisk>PBX?
17:41.07[TK]D-FenderJG7634: I also said "available" <-
17:41.20Qwell[TK]D-Fender: depending on how he architects it, it would become "available"
17:41.36*** join/#asterisk gr0mit (n=tim@81.187.32.146)
17:41.37JG7634fender: looks like the only pri and its in use
17:41.46*** join/#asterisk bug2000 (n=bug@unaffiliated/bug2000)
17:41.51Qwellthen you could eventually phase out the existing PBX/phones
17:41.59Qwellwithout having to rearchitect it
17:42.06[TK]D-FenderQwell: Sure, he could spend a lot more for a dual port card and have to lose channels going to the other side.
17:42.18[TK]D-FenderJG7634: is this on BOTH sides?
17:42.23Qwellor he could spend a bajillion dollars on a T1 card for the other PBX
17:42.39[TK]D-FenderQwell: That's likely necessary as well :)
17:43.48JG7634fender: i dont know about the other site, i'll make a phone call
17:45.07*** part/#asterisk bug2000 (n=bug@unaffiliated/bug2000)
17:45.17*** join/#asterisk gbr_ (n=gbr@200.103.96.98)
17:46.06JG7634fender: looks like no pri in small office
17:46.25JG7634fender: I think boss would be willing to gut small office
17:48.32JG7634looks like we would have to spend more money than just tow asterisk boxes anyway
17:48.39JG7634*two
17:49.11*** join/#asterisk bbryant (n=brett@68.208.65.50)
17:49.25[TK]D-FenderJG7634: no, but more than clsoe enough to consider a complete replacement.
17:52.17JG7634Thanks for the help
17:53.32*** join/#asterisk ChkDigit (n=mike@static24-72-71-175.regina.accesscomm.ca)
17:56.05jayteemy system is setup like this   from telco -> PRI -> Nortel Option 11C inbound and any 4 digit extension on * exists as 4 digit extension on the Nortel that is externally callforwarded to the same extension with a 555 prefix which gets matched with _555XXXX and gets the first 3 of ${EXTEN} stripped out in the incoming context and then matched against my internal * VOIP numbers  from the outbound PRI from my Nortel pbx that connects to * and calls to
17:56.05jaytee<PROTECTED>
17:57.17jayteeworks like a charm and allowed me to avoid adding an additional T1 card to the Nortel pbx. Of course you need to be sure you have excess channel capacity to accomodate it.
18:01.20*** join/#asterisk arpu (n=arpu@chello084114025240.14.vie.surfer.at)
18:06.52*** join/#asterisk Levonk (n=lk@adsl-76-238-249-188.dsl.lsan03.sbcglobal.net)
18:09.52*** join/#asterisk trumee (n=trumee@cpc1-seve11-0-0-cust723.popl.cable.ntl.com)
18:10.22trumeeguys, i have lot of adresses in my kde addressbook. How can i import these into asterisk?
18:11.44ManxPowertrumee: Huh?
18:12.01ManxPowertrumee: you don't use address books in Asterisk
18:12.11trumeeManxPower, i want a call notification on the computer.
18:12.16ManxPoweryou might on thephone -- but that's not an Asterisk thin.
18:12.41ManxPowertrumee: then build something yourself or look on the Wiki to see if there's something like that already built
18:13.17ManxPowerSounds to me like you want "click to dial" type of thine.
18:13.17ManxPowerthing
18:13.26trumeeManxPower, ok lemme check
18:13.46ManxPowertrumee: it is NOT a feature that comes with Asterisk
18:14.03[TK]D-FenderAsterisk is not a phonebook and is not a phone.  It does not store any of this stuff...
18:14.38[TK]D-FenderManxPower: How do I set my bicycle on "puree"?
18:15.20jaytee[TK]D-Fender, when I press the Enter key, exactly what happens next? :-)
18:15.33[TK]D-Fenderjaytee: Where is the "Any" key?
18:15.43jayteeremember those Time-Life computer book commercials?
18:15.44WimpManOh, I've been doing it all the time. Lookup the number and broadcast number and name on the LAN to get some popups.
18:16.07*** join/#asterisk geoff2010 (n=geoffrey@75.150.14.50)
18:16.15*** join/#asterisk Bananaskin (n=Banana@93-97-226-229.zone5.bethere.co.uk)
18:16.19jayteeWimpman, what are you using to facilitate that?
18:19.41ManxPower[TK]D-Fender: TAPI is what I suspect he needs
18:20.07jayteebut TAPI only runs on Windows unless he runs it in (yuck) WINE
18:22.03[TK]D-FenderWhich clearly matches his description of "KDE"....
18:22.09jayteeI'm using a SIP TAPI plugin that let's me call people in my Outlook Contacts and in Internet Explorer for websites that support click to call.
18:22.17tvirus[TK]D-Fender: Is there a proper way to cascade queues? After the 15-25 second time out it just hangs up on the person. :\
18:22.45[TK]D-Fendertvirus: It doesn't.  Pastebin <-
18:23.06*** join/#asterisk reno_rr (n=reno@maillrar.vestcom.com)
18:24.17AndyML[TK]D-Fender: problems... http://pastebin.ca/1197421
18:24.21*** join/#asterisk angryuser (i=c392f452@gateway/web/ajax/mibbit.com/x-e3857a274a2fe382)
18:24.23jayteeI was wondering about that because if I want to recreate the functionality of my Nortel ACD queues when I move them to * I'm going to have to cascade them with a Playback message in between. Can an agent be logged into multiple queues at the same time?
18:25.12AndyML[TK]D-Fender:  you'll note there are 3 new locked up voicemail channels...
18:25.39WimpManjaytee: I've been using that system for ages. Was built for I4L originally. I wrote a little stand alone sender I called vial System() from *.
18:26.05tvirus[TK]D-Fender: Extensions.conf http://rafb.net/p/wx5reo46.html | Queues.conf http://rafb.net/p/V6f4c941.html
18:26.37Ziaeonhow do you specify music on hold for outbound calls per extension
18:27.32ManxPowerzamba: in the Dial line
18:27.38*** join/#asterisk Trekk3r (n=Stargaze@c-67-163-135-32.hsd1.pa.comcast.net)
18:27.46ManxPowersorry, you mean which class, right?
18:27.57jayteeWimpMan, sounds cool. I'm just trying to figure out how to send MWI to my Polycoms from Exchange Unified Messaging without using a third party app like what Geomant offers.
18:27.58Ziaeonyeah
18:28.21ManxPowerzamba: "show applications like music" and channelvariables.txt
18:28.30ManxPowerI don't recall if it's a variable or an app.
18:29.15Ziaeonalright
18:29.48jayteethere's the MUSICCLASS function and a sip parameter you can define per phone in sip.conf also
18:30.13ManxPowerjaytee: that is for OUTGOING calls from the device
18:30.33ManxPowernot outgoing from Asterisk
18:30.39ManxPowerwhich would be incoming to the device.
18:31.59[TK]D-Fendertvirus: Wheres the failed call's CLI output?
18:32.09ZiaeonFrom outside -> asterisk -> myext, my MOH class works fine.   From myext -> asterisk -> mycoworkersext, default MOH is used. From myext -> asterisk -> outside number, default MOH is used.
18:32.56Ziaeonmy MOH class is defined in my inbound route and per exten
18:33.43Ziaeonbut it seems it wants to use the outbound routes MOH, which is inappropriate because that would be global and not per who is calling out
18:34.00*** join/#asterisk luckyaba (n=lucky@ip68-6-98-146.sb.sd.cox.net)
18:34.33ManxPoweruse the musicclass before the Dial
18:34.51Ziaeonim going to try that now
18:35.01tvirusOne moment.
18:36.39jayteeand musicclass has been deprecated in favor of CHANNEL(musicclass)
18:36.41tvirushttp://rafb.net/p/pa7HOn69.html
18:38.07Ziaeonalright, thanks, ill go from here
18:40.26[TK]D-Fendertvirus: Does exactly what its supposed to.  Drops them in a queue and noone answered so it quit and the ran out of things to do and hung up.
18:41.26tvirusHow can I make it go to the next queue :-\
18:41.43Trekk3rWould anyone happen to have a thought on why I wouldn't hear anything when I call into my asterisk even though asterisk says it is playing a file (weasels )?
18:41.58jayteeyour RTP configuration?
18:42.11[TK]D-Fendertvirus: By actually having more to do in that EXTENSION.
18:42.11jayteeor your firewall blocking the RTP ports
18:42.28*** join/#asterisk JenniferAkemi (n=akemi@206-248-161-97.dsl.teksavvy.com)
18:42.29Trekk3rall ports are open, no firewall.
18:42.35*** join/#asterisk [hC] (n=hardcore@S01060016b6b53c0c.vc.shawcable.net)
18:42.59jayteeTrekk3r, any errors or warnings on the CLI?
18:43.33Trekk3rno, everything *looks* fine
18:44.03tvirusDo I need to use a GotoIf() in there or something?
18:44.19[TK]D-Fendertvirus: No, you need to have more to do in your extension.
18:44.24jayteeTrekk3r, calling from an IP phone and if so does 2 way audio work when calling another IP phone?
18:44.49Trekk3rI'm calling from a land line into a SIP
18:45.36jayteeyou mean from a landline to an ITSP provided DID number that routes into your * server?
18:46.05Trekk3rsounds right? (i'm not the most knowledgeable on the lingo)
18:46.32jayteedo you have IP phones on the local net your * server is on?
18:46.33Trekk3rall I'm trying to get * to do is answer the phone and record an wav file, nothing else.
18:46.59jayteeyou said you couldn't hear it play ttweasels. Hearing and recording are two different things.
18:47.30Trekk3rwell it plays ttweasels before it records (which its not doing either). i figured i'd attack one problem at a time
18:47.46jayteeand for lingo what I'm trying to find out is HOW are calls getting to your * box from the outside? POTS or SIP?
18:47.47tvirus[TK]D-Fender: Something like exten = 900,n,Dial(SIP/901) ?
18:47.52Trekk3rSIP
18:47.59Trekk3rcallcentric.com account
18:48.00[TK]D-Fendertvirus: SOMETHING
18:48.13[TK]D-Fendertiveright now you go into 1 queue for 15 seconds and thats it.
18:48.47*** join/#asterisk thansen|laptop (n=thansen@19.243.sfcn.org)
18:49.01tvirus[TK]D-Fender: I don't know what that SOMETHING is. I have no idea, which is why I'm here. Give me a hint.
18:49.01Trekk3rjay, could i PM you a few lines of whats going on here? that might make what i'm doing more clear
18:49.43jaytee[TK]D-Fender, so in that scenario would Dial(queue#) then Playback(some message) then Dial(newqueue#) work?
18:49.55[TK]D-Fendertvirus: What part of "you only have 1 priority in your exten and if you wanted it to do something following leaving that queue maybe you should add MORE priorities" are you having trouble with?
18:50.01jayteeTrekk3r, use a pastebin link
18:50.15jaytee~pb
18:50.15jbot[~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste
18:51.18tvirusYou didn't say all of that... :|
18:52.11Trekk3rhttp://pastebin.com/d1649900e
18:52.29jayteescrolls backwards.........yep, he did say that.......twice.
18:53.06[TK]D-Fenderjaytee: nobody reads the BIG PRINT
18:53.32tvirusYay, now it works.
18:53.34Trekk3rthats what color is for :P
18:53.36tvirusThanks [TK]D-Fender :)
18:53.47*** part/#asterisk MACscr (n=Mark@c-98-214-92-79.hsd1.il.comcast.net)
18:53.48jaytee[TK]D-Fender, Oh? so now we're supposed to ignore the BIG PRINT and the fine print?
18:53.50tvirusI didn't think about adding another priority for it.
18:54.01tvirus[TK]D-Fender: What's your address so I can send you some home made cookies?
18:54.11[TK]D-Fenderjaytee: I'm going monospace-medium from now on
18:54.16jayteehahaha
18:54.33tvirusOr do you prefer brownies.
18:54.35tvirusYour call.
18:55.38jaytee[TK]D-Fender, just use Dingbats and say "screw 'em!" like Anthony Hopkins in Legends of the Fall
18:55.43[TK]D-Fendertvirus: FDA wouldn't let it cross the border, and both are against my diet
18:56.13jayteeotherwise I'd be FedEx'ing him some fresh roasted coffee
18:56.13*** join/#asterisk Alan_Hicks (n=alan@cardinal.lizella.net)
18:56.37Alan_HicksHowdy.
18:57.10tvirusI could always use sendthemcookies.com :D
18:57.32Trekk3rjay, did that link i paste mean anything to ya?
18:57.44Alan_HicksI upgraded from asterisk-1.4.10 (I think) to 1.4.21.2.  Somewhere along the line my Polycom phones stopped doing paging correctly.
18:58.06Alan_HicksThey no longer auto-answer.  sip.cfg and extensions.conf to follow.
18:58.16*** join/#asterisk n3hxs (n=HAMming@151.196.87.132)
18:58.56jayteeTrekk3r, that pastebin is chopped off on the right margin and missing parts of the lines or your console window is too narrow. You're getting a Warning about maximum retries but it's chopped.
19:00.02jayteeTrekk3r, and if you don't mind me asking for a second time. Do you have 2 IP hard or softphones on the same local net as your * box you could test AUDIO with?
19:00.16Trekk3rsorry, http://pastebin.com/d80cc811
19:00.26Trekk3rno i don't
19:00.33Alan_Hickshttp://lizella.net/asterisk/sip.cfg   http://lizella.net/asterisk/extensions.conf
19:00.39Trekk3rthis is not a home machine either, its a hosted VPS
19:00.50Alan_Hicks^^^^^ Pastebin was too much trouble. :-)
19:01.36Alan_HicksWhen extension 899 is dialed, the phones ring normally rather than auto-answer as they should.  Has something changed in asterisk between those two versions?
19:01.49Alan_HicksPerhaps the "SIPAddHeader" has changed somehow?
19:02.59jayteeTrekk3r, something is messed up with your connection to your ITSP. read and follow this guide using whatever configuration parameters callcentric.com has provided you for your account.
19:03.03jaytee~sipnat
19:03.03jbot[~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions
19:04.04*** join/#asterisk Vec (n=Vec@78-86-163-97.zone2.bethere.co.uk)
19:04.19Trekk3ri'll give it a shot, thanks
19:05.25Kobazwith a queue, how would i not play the "please hold.." message after 4 seconds of moh
19:05.36Kobazi want the full periodic-announce-frequency to pass before playing any tracks
19:05.41jayteeTrekk3r, sorry, I'd missed the line about your * box being a hosted VPS. It's looking like audio is the least of your problems :-)
19:07.04Trekk3rwell my home connection cannot handle the traffic i need to take so :P
19:07.59[TK]D-FenderAlan_Hicks: Short list : quotes areound Auto-Answer = bad, 2nd did you even set this up in your Polycom provisioning? 3rd Stop using priority jumping, its antiquated
19:08.23x86Alan_Hicks: usually auto-answer is phone dependant
19:08.32x86Alan_Hicks: check the settings on your phones
19:08.39*** join/#asterisk implicit (n=bayan@unaffiliated/implicit)
19:08.49x86Alan_Hicks: if they require a special header, there is an asterisk application called sipaddheader, iirc
19:08.51Alan_HicksThe settings on the phones haven't changed.
19:09.03x86show application SIPADDHEADER
19:09.08[TK]D-FenderAlan_Hicks: rest applies
19:09.08Alan_HicksAnd it was working before the asterisk upgrade with the same config files.
19:09.30Alan_HicksHowever, I did add the quotes in SIPAddHeader hoping it would fix it and apparently forgot to revert that change.
19:09.32implicitanyone know what the acronym COB means in a telecom context?
19:09.46DarKnesS_WolFx86: go sleep !
19:10.29x86DarKnesS_WolF: in the middle of the work day? :)
19:10.29DarKnesS_WolFx86: so what :P
19:10.29[TK]D-FenderAlan_Hicks: fix that, then show us the failed call with SIP debug
19:10.29x86DarKnesS_WolF: wish i could get paid for sleeping ;)
19:10.29jayteeimplicit, usually Class Of Business
19:10.35DarKnesS_WolFx86: nop :P ur fired :D
19:10.43x86;)
19:11.23implicitjaytee: haha, people using too many acronyms
19:11.26implicitthey meant close of business
19:11.27implicithahahahaah
19:11.31Alan_HicksJust a minute.  Big pastebin.
19:11.36jayteethat too,
19:11.42*** join/#asterisk logicwrath (n=no@68.41.24.98)
19:12.10logicwrathhow do i restart the httpd daemon in asterisk now
19:12.23logicwrathits taking too long to respond and not showing me the web GUI
19:12.38[TK]D-Fenderlogicwrath: You ask in #asterisknow
19:12.45Trekk3rjay, i have a dedicated IP... this nat stuff still gonna apply for me?
19:12.55jayteeTrekk3r, nope
19:12.55logicwrathno one is ever in asterisknow
19:13.19[TK]D-Fenderlogicwrath: Still fails to be our problem.  This is a distro question.
19:13.19Alan_Hickshttp://lizella.net/asterisk/sip_debug.txt
19:13.21jayteeTrekk3r, don't know how your VSP provider has their stuff setup, you'd have to work with them
19:13.53Trekk3rfundamentally this is some sort of port problem you think then?
19:15.01[TK]D-FenderAlan_Hicks: And what happens on the actual phone?
19:15.27Alan_HicksIt rings normally and doesn't auto-answer.  In that debug attempt I manually answered, then hung up.
19:15.46ManxPowerAlan_Hicks: quotes are not part of anything
19:15.47[TK]D-FenderAlan_Hicks: The header is in there just fine.  Must be your phone config.
19:15.56x86logicwrath: unload http, load http?
19:16.04*** join/#asterisk gaetronik (n=gaetan@eficaz2.manquehue.net)
19:16.13Alan_HicksHmm....
19:16.30Alan_HicksI don't know what could have changed...  I didn't touch /home/polycom at all...
19:16.34gaetronikHi, i've a question about the L option of dial
19:16.42Alan_HicksThanks for your help.  I'll dig into the sip.cfg further and try to figure this out.
19:17.22x86Alan_Hicks: it's gotta be a phone issue at this point
19:17.33Alan_Hicksnods.
19:17.40jayteelogicwrath, doubt if it works CoronaryWard or whatever that dogshit distro *NOW uses but you could try service httpd restart
19:17.50gaetronikif i make a dial(SIP/EXTEN@ipaddr)
19:17.55logicwrathhttp show status showing funny port, i got to it by using :8088
19:18.22logicwrathfor some reason it did not redirect me to 8088 from 80 so, I will research on my own.  thanks
19:18.22gaetronikthe time of the l start from the dial not from the time where the communication start really between two human
19:18.50*** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com)
19:20.56ManxPowergaetronik: we have no idea what you are talking about.
19:21.45gaetronikManxPower, i will try to xplain it more clearly
19:22.01seanbrighthe is saying the time used is from the start of the dial, not the answer time
19:22.11gaetroniki use sip to cellphone gateway
19:22.16ManxPowerseanbright: but that's not what he said. 8-)
19:22.32gaetroniki want to limit call time to cell phone to 3 minutes
19:22.36seanbrightit is what he said... just not how he said it :)
19:22.37ManxPower"cellphone gateway" changes everything
19:22.43gaetronikyes
19:22.59seanbrightwhat about the S option?
19:23.20ManxPowercell phone gateways typically answer incoming SIP calls as soon as they arrive.  You can see this in the CLI of Asterisk
19:23.37gaetronikso i'm screwed?
19:23.54*** join/#asterisk lanning (n=lanning@66.151.128.195)
19:24.29ManxPowergaetronik: unless you can make the SIP gateway stop answering until the outbound leg answers
19:24.49*** join/#asterisk hi365_m (n=hi365@213.151.62.113)
19:25.14gaetronikok
19:25.23gaetroniki don't know if it's possible
19:26.10gaetronikplanet things
19:27.29*** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir)
19:29.38*** join/#asterisk xacatecas (n=jkroon@41.25.174.193)
19:29.55Alan_HicksThanks guys.  I don't know how that happened, but I've got it fixed now.
19:30.06Alan_HicksI had to add a second alertInfo section to sip.cfg.
19:30.12xacatecashi all, any ideas what could be wrong if the Monitor() application can record the individual streams perfectly, but MixMonitor() ends up with a totally garbled audio stream?
19:30.16Alan_Hicks<alertInfo voIpProt.SIP.alertInfo.2.value="Auto Answer" voIpProt.SIP.al
19:30.17Alan_HicksertInfo.2.class="3"/>
19:32.09[TK]D-FenderAlan_Hicks: Don't, you'll kill the first.
19:32.17[TK]D-FenderAlan_Hicks: you need to do both in the same tag
19:33.07Alan_HicksOH!
19:33.08Alan_HicksThanks.
19:33.26Alan_HicksI was bound to figure that out sooner or later. :-)
19:36.50ManxPowerxacatecas: mixmonitor uses sox, IIRC to mix the audio.....
19:37.02ManxPowermaybe your sox is bad?
19:37.27*** join/#asterisk vgster (n=vgster@93-96-221-240.zone4.bethere.co.uk)
19:39.33*** join/#asterisk Meaty (n=patate@office.abi.ca)
19:41.04gaetronikManxPower, fxo answer might be the setting?
19:43.13ManxPowergaetronik: Everything I know on the subject I have already said to you.
19:45.01Alan_HicksOdd, it works fine both ways.
19:45.19Alan_HicksI'm sure the single <alertInfo> line is correct though, so I'm using that.
19:45.35gaetronikthanks ManxPower
19:45.41gaetronikthe better way is trying
19:49.28*** join/#asterisk Great_Anta_Baka (i=c419fff6@gateway/web/ajax/mibbit.com/x-fff5b2528c0e7f3f)
19:53.59MeatyHi Everybody. Hi have trouble with t38 configuration.  I place t38pt_udptl=yes in general config section of sip.conf, and t38 is active for my sip device X, even if i place t38pt_udptl=no in my X config section.
19:54.05MeatyHow its possible ?
19:54.24*** join/#asterisk linuxer_igor (n=linuxer_@mlsrj200152096p202.mls.com.br)
19:56.10linuxer_igorplease anybody help me! (DAHDI)
19:56.24linuxer_igorPRI span 1/0: Provisioned, Down, Active
19:56.36linuxer_igordon't work
19:56.59linuxer_igortoday upgrade my asterisk 1.6.0rc4 + DAHDI
19:57.21linuxer_igorand dont register channels in my E1
19:57.31ManxPowerlinuxer_igor: you understand this is not even release code, right?
19:57.49[TK]D-FenderRC!
19:57.56linuxer_igoryep
19:58.01linuxer_igorI know
19:58.03*** join/#asterisk dieno (i=771e6e9b@gateway/web/ajax/mibbit.com/x-15a3356c7dba8d3a)
19:58.11ManxPowerlinuxer_igor: then your best bet is #asterisk-dev
19:58.17linuxer_igorbut I need use this
19:58.23*** join/#asterisk Phree_Beer (n=chatzill@fctnnbsc27w-142166248246.pppoe-dynamic.nb.aliant.net)
19:58.43dienocan any one tell me why my billsec goes 0 when i reset CDR from macro
19:58.48ManxPowerlinuxer_igor: most of us here run released codes.
19:59.04linuxer_igorthey see mi  "bugs.digium.com"
19:59.09ManxPowerdevelopment code should stay on #asterisk-dev
19:59.21ManxPowerlinuxer_igor: then that is where you must go.
19:59.58ManxPowerI suspect not a single person on this channel uses 1.6 prelease and DAHDI prerelease
20:00.16*** part/#asterisk geoff2010 (n=geoffrey@75.150.14.50)
20:00.24*** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl)
20:00.32[TK]D-FenderI suspect there is a single person and they are feeling very lonely.  Won't yous tay a while? ;)
20:00.46x86DAHDI?
20:01.21ManxPowerx86: the new name for Zaptel (zaptel is (tm) someone not-digium).  There have been several announcements on the mailing lists.
20:03.16x86ah, ok
20:03.23x86missed that one somehow
20:07.13ManxPowerdieno: does it not do that when you reset CDR not in a macro?
20:07.21*** part/#asterisk Alan_Hicks (n=alan@cardinal.lizella.net)
20:09.10*** join/#asterisk talntid (n=t@66.208.251.170)
20:09.13talntidBoo
20:09.45talntidAnyone wanna buy a Rhino R1T1-EC card for $550 shipped? A few months old. No longer need it. bought from voipsupply.
20:09.56seanbrighti'll give you $55
20:10.20russellbI'll give you - $1000
20:10.20mvanbaakManxPower:
20:10.20x86hahahaha nice, asterisk-users mailing list has an archive from 2016 ;)
20:10.22mvanbaakAsterisk SVN-branch-1.6.0-r139471-/trunk built by root @ threed002 on a i686 running Linux on 2008-08-22 17:28:21 UTC
20:10.24x86http://lists.digium.com/pipermail/asterisk-users/2016-May/thread.html
20:10.35*** join/#asterisk reno_rr (n=reno@maillrar.vestcom.com)
20:11.06talntidheh ;
20:11.06talntid)
20:11.11ManxPowerx86: You should get out more.  It's had than for at least a yeart
20:11.13seanbrighttalntid: you drive a hard bargin... $60... not a penny more
20:11.41ManxPowerMr Randal Law had his date wrong
20:11.53seanbrightMr. Edward Nigma
20:11.58seanbrightanyone?  anyone?
20:12.22gaetroniki'm quite lost with bridgng and so on
20:12.53ManxPowermvanbaak: Thanks for volunteering to help linuxer_igore, but he's already gone.
20:12.56gaetronikwhen i make a call using a cellphone gateway, the bridging log message appaears in asterisk only when i hangup
20:17.31x86ManxPower: I've not really played with 1.6 yet...
20:17.43x86ManxPower: I did, however, stay at a holliday inn express last night
20:17.45x86grins evilly
20:18.01x86ManxPower: I just sub'd to 3 of the asterisk mailing lists
20:18.11M1s3ryManxPower, it's odd that linuxer_igore was in here asking the same thing that he asked in #asterisknow, and I sent him several messages on...
20:18.49x86hmm
20:19.19x86I can see that asterisk-commits was not a good idea to subscribe to... been sub'd maybe 5 minutes now and I've got 6 emails already from it
20:20.17gaetronikthen answer of a sip gateway is the time of the bridging, right?
20:20.23seanbrightx86: you subscribed in the middle of an RC release
20:20.33seanbrightx86: though it is pretty chatty in general
20:21.00x86seanbright: yeah I seen them making the 1.4.22-RC tag
20:21.21x86hmm
20:21.59bipolarMaybe someone here can help me.... I have my Asterisk box behind NAT, and I want to connect to my sipphone account to use it as a SIP trunk. It works when the box has a public IP, but not when behind nat. What do I need to forward for this to work?
20:22.04x86I seem to remember someone saying 1.2.x is no longer supported, yet there are (2) 1.2.x releases on the announcements on the front page of asterisk.org... are they just bug fixes / security updates?
20:22.14seanbright~sipnat
20:22.15jbot[~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions
20:22.20seanbrightbipolar: ^^^
20:22.46seanbrightx86: 1.2.29?
20:22.51seanbrightand .30?
20:23.02bipolarseanbright: thanks. I'll look at those pages. Everything I found so far has assumed connecting sip clients to asterisk behind nat, not the other way around.... :P
20:23.09x86seanbright: *nod*
20:23.16seanbrightyes, those were security releases
20:23.31x86also, why is there no libpri-announce and zaptel-announce mailing lists?
20:23.43seanbrightthose get announced on the asterisk lists
20:23.47x86ah ok
20:23.51x86as does dahdi?
20:23.59seanbrightyep
20:24.03x86cool
20:24.21x86when will 1.6 be ready for prime time? :)
20:24.41x86russell was linking me on some very cool stuff he was planning for 1.6 a long time ago
20:25.13x86like the ability to interrupt on a monitor session (via ChanSpy and/or ZapBarge)
20:25.30seanbrightx86: that is already in trunk
20:25.34gaetronikno one to help me understand bridge logic
20:25.39x86I'd really like that functionality with 1.4, as we're a call center
20:25.41seanbrightx86: ChanSpy barge mode
20:25.50x86that's probably the most requested feature by my managers
20:25.59x86seanbright: will it get back-ported to 1.4?
20:26.07seanbrightx86: no sir, not by the project anyway
20:26.22x86:(
20:26.26seanbrightx86: shouldn't be *that* hard though.  1.4 had audiohooks backported to it and chanspy uses those.
20:26.39seanbrightx86: for the right price, i could do it.
20:26.39seanbright;)
20:26.47x86I'm a far cry from a C developer ;)
20:26.51*** join/#asterisk kotique (n=picachu@host-static-89-41-72-147.moldtelecom.md)
20:26.53kotiqueyo.
20:27.05x86nah, we've waiting this long, we can continue to wait until 1.6 is stable enough
20:27.07kotiqueDial("SIP/11101-b5b04330", "IAX2/11100/${DID}||tr") - bad :/
20:27.15kotiquewho's familiar with dialparties.cgi ?
20:27.21x86wtf is that?
20:27.28seanbrightconsole output
20:27.29x86cgi? you mean agi?
20:27.35x86seanbright: no kidding ;)
20:27.40seanbrightwell you asked.
20:27.43x86seanbright: i was referring to the specific cgi ;)
20:27.48kotiquewhat means dial= config option in iax2.conf /
20:27.57seanbrightdialparties = AMP/FreePBX
20:27.59kotiquesearching all internets - no luck
20:28.01seanbrightiirc
20:28.11x86you searched ALL the internets?
20:28.14kotiqueyep
20:28.17kotiqueALL
20:28.20x86oh man, that's impressive
20:28.27x86I just a whole fleshlight too
20:28.38Kobazkotique: did you try teh interwebs too
20:29.00seanbrightkotique: you're using freepbx?
20:29.02x86seanbright: do you forsee a timeline as to when 1.6 will be production ready?
20:29.18seanbrightx86: i'll probably run it in production right after release
20:29.27Kobazseanbright: haha
20:29.38x86seanbright: we've got 7 offices, each with a single Asterisk box and  anywhere from 2-6 T1 ports per server
20:29.39seanbrightisn't joking
20:29.53xacatecasManxPower, so it records the two streams separately and then uses sox to mix afterwards?
20:29.53x86seanbright: when will it be released?
20:30.06seanbrightx86: once the RCs are fleshed out and such
20:30.14seanbrightx86: another RC is being rolled right now
20:30.38x86so the answer is, no there is not an approximate timeline? :)
20:30.39Kobazis there an overview somewhere of all the new cool stuff in 1.6
20:31.01seanbrightx86: i'd say "soon"
20:31.02kotiquehey guys. what does "dial" option mean ?
20:31.09x86Kobaz: I know of at least a couple new badass features that have me salivating already
20:31.11kotiquein iax2.conf 's peer configuration
20:31.25Kobazx86: like what?
20:31.29seanbrightx86: but i don't make those decisions, so...
20:31.33seanbrightshrugs
20:31.40x86Kobaz: like the new ChanSpy barge-in ability :)
20:31.43Kobazheh
20:32.03Kobazanything else?
20:32.12seanbrightapp_fax
20:32.22x86there was something else russell was telling me about (or perhaps linked me to on his website)... can't remember what specifically, but I was all I CAN HAZ?
20:32.43kotiqueso russel is that development guy?
20:32.56x86meh, I've had such bad luck with faxing and asterisk, I just outsourced my faxing to hylafax and an external 56k modem ;)
20:33.05Kobazhehe
20:33.19kotiqueHAI GAIS. CAN I HAS VARIABLE dial in IAX conf explain PLIZZ ???
20:33.20seanbrightall the new stuff is in the UPGRADE.txt file
20:33.34kotiqueok
20:33.43seanbrighthttp://svn.digium.com/view/asterisk/branches/1.6.0/UPGRADE.txt?revision=137631&view=markup
20:33.44x86i've tried spandsp (app_rxfax, app_txfax), as well as iaxmodem + hylafax, and nothing was stable at all...
20:34.01x86so I just put a modem on another box and have it setup with hylafax, works like a champ
20:34.06kotiquei'm using iaxmodem + hyla, working great.
20:34.27x86kotique: yeah i had it working, but it was far from reliable
20:34.38x86I got a good 60% reliability out of it
20:34.50kotiquewell, go g711 and be lucky
20:34.59seanbrightand CHANGES -> http://svn.digium.com/view/asterisk/branches/1.6.0/CHANGES?revision=134126&view=markup
20:35.03x86err, I would never even try anything else but g711 ;)
20:35.13x86and actually, the call came in on a zap channel anyway ;)
20:35.19kotiquewhy not ? t.38 pass-through
20:35.23kotiqueor even termination
20:35.35x86telco --> POTS --> channel bank --> asterisk == iaxmodem fail
20:35.59x86t.38 only applies when you're using IP somewhere, no?
20:36.03kotiqueI still don't get why people dont' use email
20:36.06x86(which I am not)
20:36.19kotiqueyea, t.38 is ip thingie
20:36.38x86yeah I had low reliability with a POTS line on a local zap channel
20:36.44x86no IP involved at all
20:37.08*** join/#asterisk StooJ (n=stooj@johnston37.plus.com)
20:37.10x86of course we get about 50 faxes a day, I'd venture to guess
20:41.04x86out of the 50, we'd have about 35 or so that would come through ok... the rest would be trauncated, garbled, or just complete failure negotiating the fax handshake
20:41.28x86we've got a 100% success rate with hylafax and an external modem though
20:43.02wiscadosQQ. can use asterisk to be a sip provider
20:43.19*** join/#asterisk Ditegen (n=ditegen@mafet.ru)
20:43.48drmessanoAsterisk provides SIP
20:44.21Ditegenyeah?
20:45.35drmessanoWell, it is a server that makes SIP connections available client.. Like Apache makes HTTP connections available to clients, so yet.
20:45.39drmessanoyes*
20:46.22wiscadosok, so I can set up a asterisk server at example.com and procide you@example.com and me@example.com
20:53.14Ditegenhow about redundancy? have you been lucky with it?
20:55.20*** join/#asterisk [TK]D-Fender (n=chatzill@64.235.218.194)
20:57.35*** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk)
21:05.30*** join/#asterisk Paige (n=Paige@208.89.241.31)
21:05.51jayteequittin time, back later
21:05.51Paigehi. which addons will work with 1.6.0-rc4?
21:07.35x86ugh, DAHDI channels have to be invoked by DAHDI/channum, instead of Zap/channum... that's going to mess up my flash operator panel, and most of my AMI stuffs :(
21:07.46[TK]D-FenderPaige: Probably the one listed right along side it in the topic...
21:10.09AndyML[TK]D-Fender: did you happen to see my last message a couple hours ago?
21:10.29[TK]D-FenderAndyML: not that I recall
21:11.58x86ugh, and Macro() is deprecated now?
21:12.25mchouwhat??
21:14.48Knightfal<PROTECTED>
21:16.26WimpManWell, never having used Macros befor I had the hope, they wouldn't change the context written to the cdrs, but as they do, I don't see any use anyway.
21:16.39gaetronikx86, are you kidding?
21:18.07Kobazx86: what replaces Macro?
21:18.10Paige[TK]D-Fender, http://pastebin.com/d608752
21:18.22Paigei get errors trying to build it
21:18.35gaetronikyes since it was very usefull for calling various Gateway
21:18.43gaetronikand do thing between it
21:18.52WimpManNothing. You have to Goto or Gosub.
21:18.59Kobazthat's kinda ghetto
21:19.02*** part/#asterisk JenniferAkemi (n=akemi@206-248-161-97.dsl.teksavvy.com)
21:19.03tzafrir_laptopx86, what's the big deal? rewrite some config files
21:19.35WimpManHmm. Actually there IS a point in Macro: Passing arguments.
21:20.02Kobazyeah, then you have to ghetto it up with using delimiters in the exten and then using CUT
21:20.50*** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose)
21:21.02KobazGoSub(mysub,arg1^arg2^arg3^arg4,1)
21:21.16Kobazor whatever
21:21.18WimpManOr Set() before. That would be more elegant anyway, as the parameters have sensible variable names then, but that makes two lines (Set, Goto) instead of one.
21:21.22Kobazi guess you could set a channel variable
21:21.28Kobazyeah
21:21.53*** join/#asterisk Superbaloo (i=FN@energeek.net)
21:21.58Superbaloohi there
21:22.07Kobazwhere can we petition to not have macro removed
21:22.15x86Kobaz: GoSub / Return
21:23.10x86Kobaz: it wont be removed yet, it's just deprecated
21:23.20x86Kobaz: due to "performance reasons"
21:23.26x86<PROTECTED>
21:23.26Kobazdeprecated = removed in the future
21:23.31*** join/#asterisk [hC] (n=hardcore@S01060016b6b53c0c.vc.shawcable.net)
21:23.43x86Kobaz: it explicitly states it will NOT be removed, but it is deprecated
21:23.53Superbaloohowto bind sip on two interfaces ?
21:24.02Kobazwhy deprecate something if you're not planning on removing it
21:24.25Superbaloowhat is the syntax of the sip.conf ?
21:24.25WimpManSuperbaloo: By not restrictiong it to only one?
21:24.32Kobazi would think that there's a high liklyness that it will go away at some point if it's deprecated
21:24.42WimpManKobaz: Indeed. That doesn't make much sense to me, either.
21:26.11SuperbalooWimpMan, hum that's an interesting proposition but the 0.0.0.0 option is not really one for me (i'm running asterisk in vserver) and i need to have sip running on two interface (wifi and network)
21:26.29x86oh man
21:26.50x86asterisk 1.6 supports using X11 as a local video source for video over chan_oss
21:26.56WimpManSuperbaloo: And why would you need less than all then?
21:27.31SuperbalooWimpMan, cause on vserver it bind on all of my ip (including ip of others vservers)
21:28.28WimpManIsn't vserver supposed to get that straight?
21:28.42Superbalooit should yes
21:29.12Superbaloook so just forget that ...
21:29.21Superbalooi would do it with iptable
21:29.44WimpManOr fix vserver
21:29.50Superbaloothanks anyway
21:29.56SuperbalooWimpMan, hum
21:30.04Superbalooiptables would be fine :D
21:31.25*** topic/#asterisk by russellb -> Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.6.0-rc5 (2008/09/08), 1.4.22-rc4 (2008/09/08), *-Addons 1.6.0-rc1 (2008/09/03), 1.4.7 (2008/06/04), dahdi-linux-complete 2.0.0-rc4+2.0.0-rc2 (2008/09/08), Zaptel 1.4.12 (2008/09/03), Libpri 1.4.7 (2008/08/05) -=- Related channels: #asterisknow #asterisk-gui #switchvox #freepbx #asterisk-commits #asterisk-bugs #asterisk-dev. AstriCon! #a
21:31.46Qwellrussellb: pwnt
21:31.47WimpManwas under the impression that this used to worl with vserver. But it's quite some time I looked into it.
21:32.33russellb\topic Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.6.0-rc5, 1.4.22-rc4 (2008/09/08), *-Addons 1.6.0-rc1 (2008/09/03), 1.4.7 (2008/06/04), dahdi-linux-complete 2.0.0-rc4+2.0.0-rc2 (2008/09/08), Zaptel 1.4.12 (2008/09/03), Libpri 1.4.7 (2008/08/05) -=- Related channels: irc://chat.freenode.net/#asterisknow irc://chat.freenode.net/#asterisk-gui irc://chat.freenode.net/#switchvox
21:32.53russellbmajor ownage
21:33.04QwellO.o
21:33.06russellblol
21:33.25*** topic/#asterisk by russellb -> Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.6.0-rc5, 1.4.22-rc4 (2008/09/08), *-Addons 1.6.0-rc1 (2008/09/03), 1.4.7 (2008/06/04), dahdi-linux-complete 2.0.0-rc4+2.0.0-rc2 (2008/09/08), Zaptel 1.4.12 (2008/09/03), Libpri 1.4.7 (2008/08/05) -=- Related channels: #asterisknow #asterisk-gui #switchvox #freepbx #asterisk-commits #asterisk-bugs #asterisk-dev. AstriCon! #astricon
21:34.05Paigeanyone have any ideas why it enild?
21:34.32*** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl)
21:34.45*** join/#asterisk mvanbaak_ (n=michiel@asterisk/contributor-and-bug-marshal/mvanbaak)
21:34.59WimpManAMI bridge sound cool.
21:40.04*** join/#asterisk samad (n=samad@116.71.178.25)
21:40.16samadhi everyone
21:40.36samadanybody knows about nortal programming
21:40.38samad?
21:42.32[TK]D-Fendersamad: try #nortel , you're in the wrong channel
21:43.37samadok
21:44.26samadis there any other room you can suggest
21:46.30samad?
21:47.28*** join/#asterisk Titanous (n=titanous@unaffiliated/titanous)
21:47.36Titanousis there any way to do a SIPAddHeader with a manager originate?
21:48.55[TK]D-FenderTitanous: Use a Local channel as your "Channel:"
21:50.12*** join/#asterisk EI5GTB-macbook (n=EI5GTB@78.16.75.126)
21:50.13Titanous[TK]D-Fender: I'm not sure I follow, I am using a local "Channel:"
21:50.27TitanousI'd like to add a SIP header to the call to that channel
21:50.59[TK]D-FenderTitanous: Show us a sample of your "channel" line
21:51.16TitanousChannel: SIP/200
21:51.20WimpManTitanous: You want to put it in your dialplan and call that via local/exten.
21:51.25[TK]D-FenderTitanous: that is NOT a Local channel
21:51.37[TK]D-FenderTitanous: Go read up and Asterisk's channel types
21:51.49Titanousah, ok, now I understand, thanks
21:54.27outtolunche should also be able to use Variable: SIPADDHEADER=yadda
21:55.56russellbplease don't do that ...
21:56.15russellbthe name of the vars that store the headers is an internal implementation detail
21:56.24russellbhowever, you _should_ be able to use the dialplan function that way.
21:56.56russellbby adding a line like ... Set: SIP_HEADER(X-My-Header)=foo
21:57.25outtolunche was wanting to do it from an originate
21:57.30outtoluncanyways
21:57.38russellbah, same thing, in a Variable: header or whatever
21:57.53lmadsenthe point is to use the dialplan function to set the header :)
21:58.01russellbright.
21:58.28lmadsenembrace the dialplan function; love the dialplan function
21:59.24russellb:)
22:00.35[TK]D-Fendergoes to make some more Kool-Aid
22:00.47lmadsenI prefer Crystal Light
22:01.02outtolunci was gonna say, isn't kool-aid deprecated
22:01.09outtolunchides
22:01.15lmadsenheh
22:01.20lmadsenoh no
22:01.22lmadsenoh no
22:01.26lmadsenoh ya!
22:01.31[TK]D-Fenderlmadsen: I prefer the "Presidents's Choice" alternative brand and the 50% it leaves in my bank account :)
22:01.42lmadsen[TK]D-Fender: I have not seen such a thing :)
22:01.51lmadsenor I would consider buying it
22:01.59*** join/#asterisk jjshoe_ (n=jjshoe@72.37.252.50)
22:02.22jjshoe_what's the best way to test for file existance like a voicemail greeting? readfile?
22:02.39lmadsensure
22:02.50outtoluncstat?
22:02.57[TK]D-Fenderlmadsen: it is 1/2 the price... goes by the title "Cool Delight"
22:03.00russellbthere is a STAT function somewhere ...
22:03.04russellbmight be 1.6 only though
22:03.08lmadsenI think only 1.6
22:03.10russellbah.
22:03.13[TK]D-Fenderlmadsen: I buy them in droves :)
22:03.15lmadsenhowever, I did copy it to 1.4 and it "just worked"
22:03.21russellbnice.
22:03.44lmadsen[TK]D-Fender: crystal light certainly is tasty -- I like the idea of a cheaper version
22:03.49outtoluncSTAT() is in 1.4.19 also
22:04.13[TK]D-Fenderlmadsen: Same thing. Look around.  We've got it at Maxi, Loblaws, etc....
22:04.16lmadsenreally? hrmmm... something I was using was 1.6 only...
22:04.29lmadsen[TK]D-Fender: hrmmm... I tend to shop at Dominion
22:04.37[TK]D-Fenderlmadsen: Avoid aisle kiosk bis... thats where they keep pimping the brand name items
22:04.56[TK]D-Fenderbins*
22:07.03*** join/#asterisk bkruse (n=bkruse@nat/digium/x-e5c46459dd29b9a5)
22:07.03*** mode/#asterisk [+o bkruse] by ChanServ
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22:11.19YournameDoes anyone know of an online interface much like "ARI from Little John Consulting" that would let users check voicemail/call forward, etc? I can't seem to find the ARI thing either.
22:16.48[TK]D-FenderYourname: what "ARI thing"?
22:17.20Yourname[TK]D-Fender: It's a web interface basically, the only way I know what it is is at the bottom of the page where it sas "ARI From little john consulting", so I thought someone would know
22:17.26tzafrir_laptopYourname, the most up-to-date version is probably included in FreePBX
22:17.50Yournametzafrir_laptop: Probably mixed in with almost everything in freepbx :S
22:18.06tzafrir_laptopI don't think that the original is being developed anymore
22:18.33[TK]D-FenderI see references to it, and links online...
22:20.04outtolunche probably got tired of answering the thousands of 'trixbox users' questions and not seeing a damn dime <G>
22:21.18Yournamelol
22:21.25YournameHis name was Dan.
22:21.28Yournameor probably still is.
22:21.48YournameDoes anyone know of anything else that's like it?
22:22.43*** join/#asterisk AndyML (n=alauppe@pool-71-185-74-86.phlapa.fios.verizon.net)
22:24.33AndyMLI have an asterisk-gui system running in a small call center. I've configured it for dynamic login/logout via the dialplan but watching the Asterisk CLI makes me think I've done something incorrectly...
22:25.59AndyMLexamples - http://pastebin.ca/1197591
22:26.06jameswf-home~asterisk-gui
22:26.07jbot[~asterisk-gui] The Asterisk-GUI is an open source project lead by Digium. It's client-driven, and its only dependency is Asterisk itself. Check out the new version from svn here: $ svn co http://svn.digium.com/svn/asterisk-gui/branches/2.0 asterisk-gui-2.0
22:26.38jameswf-homeshouldnt that say goto #asterisk-gui
22:27.17AndyMLit shouldn't say goto asterisk-gui because asterisk-gui is part of a standard asterisk distribution.
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22:27.25QwellNo it isn't
22:27.35jameswf-home~asterisk-gui is also For support go to  #asterisk-gui
22:27.35jbotjameswf-home: okay
22:28.35AndyMLpretend for a second that I didn't say i was using asterisk-gui. I built most of the logic directly in extensions.conf and queues.conf. if anyone here knows anything about asterisk queue engine, they can probably help me.
22:28.47jameswf-home~ask
22:28.48jbotask is probably Questions in the channel should be specific, informative, complete, concise, and on-topic.  Don't ask if you can ask a question first.  Don't ask if a person is there; just ask what you intended to ask them.  Better questions more frequently yield better answers.  We are all here voluntarily or against our will.
22:29.20jameswf-homepastebin your dial plan and give up the issues
22:29.52AndyMLjameswf-home: see above - I have pastebin'd my issue already.
22:30.27AndyMLi'll dig up the dialplan as you look at that. If you're willing to help, i would appreciate it greatly.
22:30.51jameswf-homewell i cant be a condecending a-hole and read geesh... looking
22:31.24AndyMLi'm confused - have I offended you?
22:32.22jameswf-homeheh no not an easy task.....
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22:32.45outtolunctakes jameswf-home's eggo
22:32.49jameswf-homemy network doesnt like pastebin.ca hmmm
22:32.59jameswf-homeleggo my eggo
22:33.01AndyMLwhere else would you like me to put it?
22:33.05jjshoe_Yourname ari? asterisk remote interface?
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22:33.21jameswf-home~pb
22:33.22jbot[~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste
22:33.29AndyML.com - ok.
22:33.30jameswf-homelet me see what loads
22:33.46jameswf-home.com is good
22:33.59[TK]D-FenderAndyML>it shouldn't say goto asterisk-gui because asterisk-gui is part of a standard asterisk distribution. <- no such thing as an "Asterisk distribution".
22:34.25AndyMLyou're right - i used the wrong noun. "installation"
22:34.41jameswf-homeisnt installation a verb
22:34.47[TK]D-FenderAndyML: And its not part of any of MY standard * installs
22:35.02[TK]D-FenderAndyML: 0 for 2, care to make it a nice round 3?
22:35.25AndyMLlook - i love community supported software, but does ridicule have to be a part of it?
22:35.45jameswf-homeAndyML: ummm yeah
22:35.47AndyMLnobody likes being made a fool of.
22:36.13jameswf-homeAndyML: its all free without ridicule of newbs we would see no return
22:36.39[TK]D-FenderAndyML: Believe me, I haven't actually started any kind of ridicule yet.  That was a minor jab at best.
22:36.47AndyMLI've been supporting Asterisk professionally for more than 3 years. Can you save it for the newbies at least?
22:36.53jameswf-homethe only place the god complexes get worse is in -dev
22:37.05[TK]D-FenderAndyML: Now what you seem to show in your PB (what little there is to see), seems to indicate you keep calling a busy device.
22:37.07jjshoe_AndyML it's more prevelant in certain channels
22:37.19jjshoe_AndyML it's strange to see it so strong on freenode
22:37.35[TK]D-FenderAndyML: a better PB would have included SIP debug and the call from beginning to end.
22:37.39jameswf-homeheh heh we are 1337 :))
22:37.41jameswf-homesorry
22:37.49AndyMLi'm not new to IRC either, or the god complex. I guess you just get sick of it after a while.
22:38.01[TK]D-Fenderjameswf-home is teh shizit y0!!!! r3sepkt!
22:38.34[TK]D-FenderBest part about my God complex : no peer pressure :)
22:39.00*** join/#asterisk rasterix (n=IceChat7@host86-154-172-124.range86-154.btcentralplus.com)
22:39.04jameswf-homehmmm thinktel.ca < get caught up at.... gueass thats the great firewall of canada
22:39.36[TK]D-FenderAndyML: So I've already given you 2 better ideas on providing more complete info to examine your problem...
22:39.54AndyMLI can only process the shame and copy/paste so fast.
22:40.55[TK]D-FenderAndyML: And you fear you didn't set your dialplan up properly but showed us only the smallest amount of debug info, and not even the dialplan you are doubting.
22:41.15[TK]D-Fendersits back and sits his post-fondue broth
22:41.22[TK]D-Fendersips*
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22:41.23rasterixugh dont mention firewalls
22:41.34AndyMLoh, even the great [TK]D-Fender mistypes sometimes.
22:42.02Yournamejjshoe_: Probably? I'm not sure.. but looks good, and want it lol
22:42.03AndyMLi was pasting it all at pastebin.ca when jameswf-home said he couldn't get to pastebin.ca.
22:42.21[TK]D-FenderAndyML: I cut my thumb open with a sword by accident this weekend.  Excuse enough :)
22:42.29jameswf-homecanada rejected my passport
22:42.40[TK]D-Fenderjameswf-home: TERRIST!
22:43.10jjshoe_terrist?
22:43.11jjshoe_sounds vicious
22:44.03jameswf-homeheh heh terror www.youtube.com/watch?v=1uwOL4rB-go
22:44.15rasterixfender only mistypes to make himself appear human
22:44.15rasterixpfft canada rejected my country
22:44.15rasterixfender is impersonating a moron that cant type... dont be fooled its a cunning disguise
22:45.48jameswf-homehttp://www.100factsabout.com/ircuser/%5BTK%5DD-Fender
22:47.06jameswf-homeheh # Give a man a fish, and you will feed him for a day. Give a man anything that is better than a fish, and Ircuser [TK]D-Fender will beat his ass and take it.
22:47.40jameswf-homebetter: # Ircuser [TK]D-Fender can eat a rubix cube and crap it out solved.
22:47.46jameswf-homeneat
22:48.21[TK]D-Fenderjameswf-home: Yeah I saved the link to that page you posted a few weeks ago :)
22:48.33[TK]D-Fender<- pornographic memory
22:48.35rasterixwhat worries me is how he peels all the labels off and sticks them back on b4 he craps it ><
22:48.53scooby2Any idea why idle agents would not ring when there are calls waiting in the queue? Ring-all strategy. Or is this to force you to ditch agents with 1.4.x?
22:49.28rasterixscooby it cud be bcoz u have agents that do not appear busy when they are
22:49.47AndyMLhttp://pastebin.com/m51964c1e
22:50.05AndyMLexcerpts from extensions.conf, queues.conf, the CLI and a SIP debug.
22:50.43scooby2In what sense? We have zero wrapuptime. Asterisk will usually show a message if they turned DnD on
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22:52.14rasterixexample a sip phone with multiple lines... the agent is on one but asterisk passes the call to the second line
22:52.14rasterixbut that could have been down to logging devices rather than agents on to the queues
22:52.47Yournamejjshoe_: Any help on that? :)
22:52.53[TK]D-FenderAndyML: So far looks like a phone on DND...
22:53.10scooby2rasterix: if the agent logs out and logs back in then they get a call. It is strange.
22:53.12[TK]D-FenderAndyML: And your extensions.conf does not play into this
22:53.58AndyMLlet me log that phone out of the queue.
22:54.18rasterixsounded similar to what u have
22:54.18rasterixwhat phones are you using scooby?
22:54.41AndyMLactually - that phone is just taking a call. they're an active member of the queue
22:55.01scooby2rasterix: SPA941-NA
22:55.17scooby2call waiting turned off
22:55.36*** join/#asterisk jm|laptop (n=jm|home@zen.jamiem.com)
22:56.11AndyMLcall waiting is off because the queue engine would send it more than one call at a time if it were on
22:56.38rasterixoooh
22:56.38rasterixlinksys
22:56.38rasterixsounds even more familiar
22:56.38rasterixtheres a setting
22:56.38rasterixi had to set on our spa942
22:56.38rasterixto get queues working right
22:56.58*** join/#asterisk jksM (i=jks@193.189.93.254)
22:57.45AndyMLthese are polycom phones
22:57.49[TK]D-FenderAndyML: What ver of *?
22:58.09AndyML1.4.19.1, as per pastebin entry ( <grin> )
22:58.25[TK]D-FenderAndyML: Go follow UPGRADE.TXT and it won't send calls to busy members..
22:58.58rasterixscooby u will prolly find it if u google spa942 queues and asterisk
22:58.58rasterixi forget what it was but basically there is a setting to disable multiple lines
22:58.58rasterixandy im talking to scooby
22:59.29scooby2ok
22:59.52AndyMLrasterix: its ok - i got that.
23:00.36AndyMLI'll look at UPGRADE.TXT again.
23:02.22jksManyone know a simple call generator for SIP/RTP that can be used to stress test asterisk? (preferable one that is free or cheap)
23:02.43rasterixits definitely solvable scooby
23:02.43rasterixi fixed it
23:02.43rasterixand im an idiot
23:02.43rasterixask anyone here
23:02.52scooby2lol
23:02.56[TK]D-Fenderrasterix: with m4d l33t sk1llz like this : http://www.youtube.com/watch?v=eJEmtLxkEoI @ 4:06
23:05.22rasterixscooby if u cant find it
23:05.56*** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee)
23:06.09rasterixmsg me
23:06.10rasterixill log into work i have the solution logged in our "bible" somewhere
23:06.24scooby2ok thanks. not finding it so far
23:06.39scooby2but i'm looking still
23:06.59rasterixif i log into work ill get kicked off of here
23:07.19rasterixdumb vpn
23:08.02scooby2lol
23:10.03scooby2the only good thing about these phones is that they were cheap. Other than that, I wish we would have gotten Polycom
23:10.03rasterixno scooby
23:10.20rasterixthey r great phones
23:10.20rasterixdont get mad over this
23:11.07[TK]D-FenderSPA's ar a decent runner up, but indeed behind in the race...
23:11.08scooby2is it lines or queues that I should be searching for?
23:11.46rasterixdisable call waiting
23:12.09[TK]D-FenderFix your SIP configs <-
23:12.33scooby2[TK]D-Fender: to send it only one call at a time?
23:13.16rasterixrawr i cant believe i cant remember this
23:13.16[TK]D-Fenderscooby2: Set them up properly and you can leve CW on and * queues won't send calls to busy agents
23:13.20rasterixi might log into work
23:13.39scooby2[TK]D-Fender: that would be a nice feature
23:14.11[TK]D-Fenderscooby2: UPGRADE.TXT <-
23:15.15rasterixah ok my fix is for 1.2
23:15.15rasterixmaybe there is a better solution for 1.4?
23:15.51scooby2i just upgraded from 1.2.x to 1.4.21.2 last Wednesday
23:17.27rasterixand u had the problem since u upgraded?
23:17.32scooby2yeah
23:17.45scooby2in 1.2 we just turned off call waiting on the phones
23:17.49scooby2hack but it worked
23:18.40rasterixthats what we did
23:18.40rasterixand no problems when we upgraded to 1.4 (about 2 months ago)
23:21.59rasterixscooby have u upgraded all ur spas to the latest firmware?
23:21.59rasterixcoz id hate to struggle on this and find ur running 1980s firmware :P
23:22.18AndyMLrasterix: as you've seen - queues in 1.4 isn't totally awesome out of the box.
23:23.10AndyMLerr scooby2 ... looks like rasterix has been able to make it work
23:23.16scooby2not the latest latest but close. All the latest firmware does is allow four lines instead of the default two. Previously they charged for the upgrade.
23:23.18rasterixneither was queues in 1.2
23:23.18rasterixi only have a month or 2 on 1.4
23:24.32AndyMLwell i pretty much hate queues...
23:25.29rasterixscooby on the 942 the same time as the upgrade from two to four lines they added a new option for queues
23:25.29rasterixto fix problems like this
23:25.29rasterixur really going to make me log into work
23:25.30rasterix*Sigh*
23:25.30rasterixill be back
23:25.30rasterixbye
23:26.28scooby2thanks
23:26.33AndyMLthank you rasterix ...
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