IRC log for #asterisk on 20090701

00:00.50*** join/#asterisk bryanfe2 (n=chatzill@wsip-72-215-161-251.sb.sd.cox.net)
00:01.14*** part/#asterisk bryanfe2 (n=chatzill@wsip-72-215-161-251.sb.sd.cox.net)
00:02.53leifmadsenDLNoah: odd -- I can't find any issues like that in the bug tracker right now
00:03.12leifmadsendoesn't mean the issue doesn't exist -- just that it hasn't been reported before :)
00:03.22DLNoahyeah, I hadn't seen anything, didn't know if it was a bug or a PEBKAC
00:03.57leifmadsenya... could be a layer 8 issue, but not 100% sure of that :)
00:04.14leifmadsenDLNoah: oh -- ast_streamfile
00:04.26leifmadsenDLNoah: this may be related then:  https://issues.asterisk.org/view.php?id=15224
00:06.20DLNoahwell, if I'm reading that report correctly, Playback() from within the dialplan is working for him.
00:06.23DLNoahit is not for me.
00:06.53DLNoahis there a config file somewhere for format_mp3.so that I'm missing?  I've never really had to burrow into it b/c someone else set this server up and MP3 playback has "Just worked" since I took over maintenance
00:08.50*** join/#asterisk vegbox (n=kevinle@adsl-64-173-83-182.dsl.lsan03.pacbell.net)
00:17.15vegboxHi, I have a NEC Electra phone system with a ghetto cd player attached to it that plays our music on hold.  It has a regular stereo cable coming out of the NEC system and in to the cd player.  Is it possible to replace this cd player with asterisk?
00:19.18telnettechanyboady know of any company in US looking for an Asterisk technician?
00:20.07*** join/#asterisk boch (n=fran@200.69.230.177)
00:21.03leifmadsenDLNoah: you're recompiling addons once you've upgraded asterisk right?
00:21.08DLNoahyes.
00:21.18leifmadsenDLNoah: and format_mp3.so is loading? (core show modules like mp3)
00:21.19*** join/#asterisk bbryant (n=brett@c-68-59-22-114.hsd1.sc.comcast.net)
00:21.20DLNoahasterisk refuses to load format_mp3.so if I don't
00:21.24leifmadsenright
00:21.25DLNoahand yes.
00:21.29leifmadsenhmmm
00:21.50leifmadsenI've never used it, so I'm not 100% sure
00:22.39DLNoahok, thanks for the attempt
00:23.03bochhow can an agent log out after AgentLogin() without hanging up the phone ?
00:23.33DLNoahI also kinda have a feature request / suggestion for Park(), though I suppose 8pm at night isn't the best time to catch whoever I'd need to talk to for that.
00:23.33leifmadsenboch: I don't think you can.... what are you actually trying to do?
00:23.52leifmadsenDLNoah: heh, not really -- asterisk-dev mailing list is probably your best bet. There are not really any devs in here.
00:24.05DLNoahok, thanks.
00:24.35bochleifmadsen, i want my agents to be able to log in to a queue, or to dial manually the customers by themselves
00:24.58leifmadsenboch: then use AddQueueAgent()
00:25.01leifmadsenerrr
00:25.03leifmadsenAddQueueMember()
00:25.24leifmadsenand RemoveQueuMember() to logout
00:25.37bochleifmadsen, already tried, but calls does not goes to agents
00:25.44leifmadsenthen you didn't set it up right
00:26.09leifmadsenruns off to try and get this PXE server setup so he can load his new server since the CDROM appears to be dead
00:26.12bochleifmadsen, CLI reports agent added to queue, and i see the inbound call running Queue() app
00:26.37bochleifmadsen, but i cant hear "beep", neither the customer
00:26.45leifmadsenboch: status probably is set to unknown or something -- use a real channel, not an Agent channel
00:27.02leifmadsenhear beep? the queue() would call you back with AddQueueMember()
00:27.07leifmadsenit'd be more like a ring
00:27.13bochleifmadsen, im using an IAX channel, but maybe the status is the problem, ill check that, thank you
00:27.17leifmadsenand anyways, you just need a 2nd line on your phone if you wanna make a call
00:27.24*** join/#asterisk MmixX (n=mix@61.14.191.137)
00:27.44leifmadsenAgentLogin() is designed specifically to stay on-line while you're logged in. Hanging up is how you logoff
00:31.21bochleifmadsen, i can see the problem: IAX/1901 (dynamic) (Invalid) has taken no calls yet
00:31.28leifmadsenright
00:31.38bochdo you know how to set the interface state in 1.4 ?
00:31.51leifmadsenboch: setup hints
00:32.12bochexcuse me ?
00:32.50*** part/#asterisk korihor (n=korihor@190.77.92.57)
00:33.44leifmadsenboch: I don't use IAX (just SIP), but typically you need to enable hints in the dialplan for presence
00:33.53leifmadsenexten => 9001,hint,IAX2/9001
00:34.11leifmadsen*CLI> core show hints
00:34.22bochi see, thanks again
00:36.30KavanScan someone give me an idea as to the term I'm looking for?
00:36.59KavanSI'm trying to setup a system where someone can sit down at a different extension/station each day and then when someone were to dial ex: 954, they would reach the same person each time
00:37.13KavanSwould that be an "agent" ?
00:37.20WindowsUserfollowme
00:37.25leifmadsenKavanS: yes -- typically hot-desking
00:37.35KavanSok, hot-desking...let me google a little
00:37.38leifmadsendon't call it an extension -- it is a device
00:37.42KavanSthanks for the wording
00:38.01leifmadsenextensions are numbers, assigned to users, who can dynamically be assigned to any device that they login to (hot-desking)
00:38.16leifmadsenthey are all mutually exclusive of one another
00:38.27KavanSnice, that's pro as shit
00:38.42leifmadseno.O
00:38.44KavanSok, looks liek I have a short term asterisk project tonight
00:38.50leifmadsenshort term? heh.
00:39.01KavanScool...I figured there was a defacto solution to said problem
00:39.07leifmadsentook me 3 days to build a simple hot-desking platform :)
00:39.20leifmadsenand I've done it more than once
00:39.35leifmadsenalthough honestly most of that time was spent with stupid sangoma hardware
00:39.57KavanSok...hrm, well found a post that looks as if it has the logic needed
00:40.06KavanSjust need to make the additions to my asterisk box...
00:40.07KavanShttp://www.757.org/~joat/wiki/index.php/Simple_hot-desking
00:41.17leifmadsenhttp://astbook.asteriskdocs.org has some hot-desking stuff too
00:41.34KavanSok cool
00:41.54DLNoahleifmadsen: what about if I had a hot-desking situation, but also a shop where each person had a DID assigned to them... the customer wants the incoming callers to ring the designated agent if available, or to hear a message if the agent is unavailable (busy, etc)
00:42.21DLNoahand if the agent is unavailable, the caller would be presented with an option to go to VM, wait on hold (if the agent was merely busy), or talk to a different agent.
00:42.37DLNoahwould that be just a simple hot-desk thing, or would some sort of queue setup be more appropriate?
00:42.49*** join/#asterisk rockhard1981 (n=snusse@78-82-50-224.tn.glocalnet.net)
00:43.57leifmadsenDLNoah: that is just an IVR basically that rings the device the agent has "logged in" to
00:44.06leifmadsenqueue would be unnecessary there
00:44.11rockhard1981tzafrir_laptop: yes, any stream of sound to a channel like SIP/trunk-idid
00:44.36DLNoahok.  and basically a Background or Read for the "your agent isn't available" message, then?
00:44.52WindowsUserDLNoah: you could queue for if they choose a different agent
00:45.44rockhard1981tzafrir_laptop: like a reverese mixmonitor, where you put audio to a channel, instead of grabbing it.
00:51.18rockhard1981anybody knows if manger interface has a limitation to connections?
00:56.24*** join/#asterisk jtodd (i=cac6d84c@ns.fox-den.com)
00:56.24*** mode/#asterisk [+o jtodd] by ChanServ
00:58.25*** join/#asterisk MmixX (n=mix@61.14.191.137)
00:58.27*** join/#asterisk WindMaker (n=WindMake@unaffiliated/windmaker)
00:58.32WindMaker:(
00:58.37WindMakeri have a problem :(
00:58.55WindMakerbut my english it's bad :(
00:59.54ManxPowerWindMaker: Try this if you need to: http://babelfish.yahoo.com/
01:00.06WindMakerthks
01:00.36*** part/#asterisk ManxPower (n=manxpowe@69.73.94.162)
01:04.31*** join/#asterisk dwery (n=dwery@nslu2-linux/dwery)
01:07.53*** join/#asterisk thansen (n=thansen@76.27.110.194)
01:19.38WindowsUserMusicOnHold doesn't react to dtmf does it?
01:27.35*** join/#asterisk mltlnx (n=mltlnx@cpe-24-193-11-14.nyc.res.rr.com)
01:29.07leifmadsenno
01:33.04*** join/#asterisk asternic (n=chatzill@66.60.27.162)
01:34.01*** join/#asterisk Kumbang (n=gibrig@rusnas.paume.itb.ac.id)
01:36.08*** join/#asterisk propellerhead (n=yogurt2u@host176.190-136-58.telecom.net.ar)
01:39.53*** join/#asterisk implicit (n=bayan@unaffiliated/implicit)
01:51.05Pan3DWindowsUser: if you think about it, that's a good thing. Otherwise, people who are supposed to be on hold could accidentally/intentionally bounce themselves to another extension while on hold.
01:55.45*** join/#asterisk WindMaker (n=WindMake@unaffiliated/windmaker)
01:55.56WindMakerwhy -> error 408 ?
01:56.05WindMaker?
01:57.21*** join/#asterisk VoipForces (n=kvirc@mail.net-forces.com)
01:58.00VoipForcesMemory hole... In my brain... was the s,n(label) introduced in 1.2 or in 1.4 ?
01:59.41*** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir)
02:00.27*** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-d7ecbd540df25231)
02:01.01VoipForces[TK]D-Fender: Memory hole... In my brain... was the s,n(label) introduced in 1.2 or in 1.4 ?
02:01.35leifmadsen1.2 I think
02:02.55VoipForcesleifmadsen: That is what i tought, was not sure.
02:03.21leifmadsenit's been years since I've used 1.2 :)
02:03.26VoipForcesleifmadsen: Writing code that must run on 1.2 and I'm so used under 1.4 now
02:03.34leifmadsenbut ya.... I still can't believe people use priority numbers
02:04.07VoipForcesleifmadsen: Indeed, feels like doing Basic under DOS LOL
02:04.57VoipForcesAnd for younger guys/gals, DOS meant Disk Operating System back then, not Denial Of Service
02:05.11leifmadsenheh
02:05.17leifmadsenDOS 5.0 ftw!
02:05.29VoipForcesAhh the memories of dos 3.3 on 1 5.25" floppy, using edlin and debug
02:05.38leifmadsenis tweaking his X00 fossile driver for Frontdoor
02:06.09leifmadsengoes to hang out with the g/f
02:06.42VoipForcesHere everyone is asleep, gf, 3 kids, even the cat
02:09.50eppigyno need for other entities
02:10.14*** join/#asterisk mchou (n=quassel@unaffiliated/mchou)
02:18.07WindowsUserPan3D: yea, i'm just probing around Wake-up calls
02:19.10nkohhleifmadsen!! long time no see man! how are you? what've you been up to?
02:19.52*** join/#asterisk CryWolf (n=freedomb@mn01.freedombi.com)
02:20.04*** join/#asterisk MmixX (n=mix@61.14.191.137)
02:20.25*** join/#asterisk Deeewayne (n=dwayne@76.29.245.9)
02:20.25*** mode/#asterisk [+o Deeewayne] by ChanServ
02:21.07*** join/#asterisk andresmujica (n=andresmu@ubuntu/member/andresmujica)
02:22.05CryWolfApologies for using a gui, but I'm trying to figure out why accountcode is not being set for outgoing calls.  And as near as I can see, that is (or should be) a simple asterisk configuration, just one line in the appropriate sip peer definition.
02:22.32*** join/#asterisk implicit- (n=bayan@unaffiliated/implicit)
02:23.22CryWolfWell, I can set it for an individual extension, but I'm trying to set it for the trunk.
02:27.16WindowsUserso accountcode=xxxxx doesn't work?
02:27.49CryWolfCorrect.  It remains blank in the cdr.
02:31.21VoipForcesAnyone knows if the syntax Goto(${last_question_context},s,1) is valid? Variable ${last_question_context} containing a valid context.
02:35.35VoipForcesCryWolf: What gui?
02:35.54CryWolfVoipForces: freepbx
02:36.21CryWolfI've removed the 'accountcode=' from each of the extensions, so it's only defined for the trunks now.
02:37.17VoipForcesCryWolf: I don't this it's valid to put it in the trunk definition.
02:38.35CryWolfIt doesn't seem to be well documented, other than to say that it is used to populate the "accountcode" field of the CDR.
02:39.23CryWolfIt works for the inbound, just not the outbound.  I guess the inbound isn't a trunk, as such.
02:39.25VoipForcesCryWolf: thing is that the variable neds to be set dialplan wise and if you put it in the trunk definition it will not be populated.
02:39.40VoipForcesCryWolf: Exact.
02:40.18CryWolfVoipForces: Okay, I was beginning to suspect as such.  So I just need to find the correct place to put Set(CDR(accountcode)=...)
02:41.26VoipForcesCryWolf: something like that, but might be somewhat difficult if you do not want to break freepbx.
02:42.06CryWolfI've already modified it somewhat.  What's one more mod?  :)
02:42.21VoipForcesCryWolf: True.
02:42.32VoipForcesCryWolf: Just remember when you do an update :-)
02:43.10CryWolfVoipForces: Trust me, it's all documented.
02:43.16*** join/#asterisk chendy (n=chatzill@58.251.101.21)
02:48.32*** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net)
02:51.08*** join/#asterisk coppice (n=chatzill@204.196.17.210.dyn.pacific.net.hk)
02:56.21*** join/#asterisk moy (n=moy@CPE001cdfec4cee-CM00080dab8485.cpe.net.cable.rogers.com)
02:58.42*** join/#asterisk Iamnach0 (i=Iamnacho@ip174-70-137-120.ks.ks.cox.net)
02:59.51*** join/#asterisk modex (n=shupej@74.196.28.220)
03:01.05modexDoes anybody have any experience getting the Line1 button to work on the Grandstream GXP2000?
03:07.34*** join/#asterisk Jumpie (n=zz@c-76-100-241-4.hsd1.md.comcast.net)
03:07.35Jumpiehey guys
03:07.43Jumpieim having a particularly problematic 'gremlin'
03:07.58Jumpiei have a bunch of aastra 55i that i was deploying, went smooth
03:08.09Jumpieim using the dnd.php script on the phone
03:08.52Jumpiei have copied the same line of code, path, etc to each phone and they all work fine, changing of course each ?user=xxxx for each person..but just one particular person i get 'authentication error, you are not authorized to use this application" on the phone
03:08.57JumpieNOTHING is different
03:09.07Jumpieshes registered, can do anything/everything everybody else can, the path is fine
03:09.25Jumpieif i take out the ?user= portion, then she can activate it..but can't deactivate it unless i reboot phone....any ideaS? drivin myself nuts here
03:30.42CryWolfWoohoo!  Success at last!
03:38.59WindowsUserand odd characters in her name?
03:39.31*** join/#asterisk CunningPike (n=CunningP@S01060014bf81366b.vc.shawcable.net)
03:41.46*** join/#asterisk xflavio (n=xflavio@189.82.11.211)
03:45.38xflaviolist
03:53.49*** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir)
03:56.06*** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee)
03:56.52*** join/#asterisk fakhir_ (n=fakhir@unaffiliated/fakhir)
04:10.44*** join/#asterisk taxilian (n=richard@216.83.134.36)
04:11.07taxilian'evening, all
04:14.14taxilianany experts here who can tell me if there is a way to create a script (possibly AGI?) that can be used for outgoing calls that will trigger an incoming call from another system and connect them?
04:14.52taxiliani.e., call a PHP script with an extension number that uses a 3rd-party interface to instruct my work phone system to call me and connect the call to the line that is already open
04:18.45*** join/#asterisk trnzmeta (n=bleh@secure27.lnk.telstra.net)
04:21.13taxilianis there a different channel that would be better for this question?
04:21.14*** join/#asterisk Linuturk (n=linuturk@fluxbuntu/developer/Linuturk)
04:21.36WindowsUser~idle
04:21.37infobotIdle is a doodie head
04:21.49WindowsUseraww i was hoping it'd have something useful
04:22.45WindowsUsertaxilian: people will only respond if they have something to say thats not "i don't have a clue" :)
04:23.04taxilian=]  yeah, that's why I was asking if there is somewhere better to go =]
04:23.11*** join/#asterisk Iamnacho (i=Iamnacho@ip174-70-137-120.ks.ks.cox.net)
04:23.12taxilianseems like there ought to be some way of doing it
04:23.16taxilianbut it's kinda a weird problem
04:23.43taxilianI seem to mess with a lot of those :-/
04:24.19WindowsUserwhy are you triggering a call from remote?
04:25.02taxilianbasically, I can use the remote phone system to make a call
04:25.09taxilianbut I can only control it via HTTP
04:25.20taxilianso I send an HTTP command with the number to call and the number to call from
04:25.31taxilianand then it calls me and when I answer, calls the remote
04:25.43taxiliansomewhat similar to google voice
04:25.46WindowsUserhorray for call files
04:26.28*** join/#asterisk qdk (n=qdk@0x573d8dd1.bynqu1.dynamic.dsl.tele.dk)
04:26.37WindowsUseri wonder if you can silently enter a conference
04:26.54taxilianbut since I have this nice asterisk system barely being used at my place, I thought it should be possible to make a script that asterisk runs when you call out
04:27.09taxilianyeah, I was wondering the same thing; then have the incoming call automatically routed to the same conference
04:27.19taxilianit'd be max one call at a time, but I don't really need more than that anyway =]
04:28.41WindowsUserwell you'd be limited to starting a call one at a time
04:30.02*** join/#asterisk ddickenson_ (n=ddickens@166.205.4.157)
04:30.16taxilianright
04:30.35taxilianI'm still relatively new to asterisk, so I haven't played with conferences much yet
04:30.48WindowsUserit might also be called meetme
04:30.50taxilianthe other problem would be getting it to hang up on the incoming call when I disconnected from the conference
04:31.37WindowsUserkill the conference from the h extension if possible from dial plan?
04:31.41taxilianhmm.  that is pulling up some useful search results
04:31.44jayteeyou can turn off the announcement for when new people enter a conference but I'm pretty sure it affects all conference members.
04:31.59taxilianthat would be fine; I don't need any announcements
04:33.50WindowsUsertaxilian: 1.4 or 1.6?
04:33.53taxilian1.4
04:34.03taxilianrunning on freebs
04:34.05taxilianfreebsd
04:34.30taxilianI'm using freepbx as an admin tool, but I'm roughly familiar with the config files as well
04:34.43taxilianstill learning advanced dialplans (trying, anyway =])
04:36.00WindowsUserin the h extension you could DeadAGI a script that uses the Asterisk Management Interface and kicks people out of the conference, probably very hacky, but the whole concept is ;)
04:36.45*** join/#asterisk Chris-NB (n=chris@home.fuerstaller.com)
04:36.51*** join/#asterisk NewtoPBX (n=shanker@c-75-70-225-138.hsd1.co.comcast.net)
04:37.14taxilianlol.  you have a point. this is probably a stupid question, but what is the h extension?
04:37.24carrarhangup
04:37.33taxilianahh
04:37.36carrar~book
04:37.37infobot[~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook
04:37.52taxilianthank you.  good resource
04:38.08taxilianI promise to read carefully before bringing too many more questions to the channel =]
04:38.20taxilianwell, that gives me some ideas; I sure appreciate the tips, gusy
04:38.28taxilianeven if I cant' tpye
04:42.28*** join/#asterisk errotan (n=errotan@5403E436.catv.pool.telekom.hu)
04:52.31taxilian'nite all
04:52.33*** part/#asterisk taxilian (n=richard@216.83.134.36)
04:52.34WindowsUserhuh asterisk not running future dated callfiles must be a new thing, I'm finding wakeup call scripts from 2004 that use cron
04:54.00*** join/#asterisk hi365 (n=hi365@94.159.178.144)
05:07.06*** join/#asterisk bbryant (n=brett@c-68-59-22-114.hsd1.sc.comcast.net)
05:08.30NewtoPBXso 1000 question time
05:08.32NewtoPBXhaha
05:08.35NewtoPBXanyone up for it ?
05:08.54[TK]D-FenderNewtoPBX: 1000?  No, try your luck and see how many you get
05:09.04NewtoPBXok
05:09.31NewtoPBXso i have installed asterisk to my linux machine
05:09.37NewtoPBXnow what do i do ?
05:09.43NewtoPBXi have no idea where to start
05:09.53NewtoPBXsince there is like 30 configure files
05:09.59NewtoPBXi have a sip provider
05:10.09[TK]D-Fender~book
05:10.10infobot[~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook
05:10.33NewtoPBXthat is the one i am reading
05:10.34NewtoPBXhaha
05:10.36WindowsUserwhat provider?
05:10.57NewtoPBXiptel.org
05:10.59[TK]D-FenderNewtoPBX: Here is a SAMPLE to use as inspiration :
05:11.00WindowsUsersome of them tell you what to put in sip.conf :)
05:11.01[TK]D-Fender~jerjerguide
05:11.02infobot[~jerjerguide ] Jeremy McNamara's quick sample guide to setting up * :  http://www.jeremy-mcnamara.com/2007/02/26/how-to-configure-asterisk-your-first-installation/
05:11.14NewtoPBXsweet
05:11.15NewtoPBXthanks
05:12.15NewtoPBXso here is a question to. can i just push this thing using tcp/udp using the nick on the pc or do i have to have the hard ware and phones ? or can i just call into it using the nic?
05:12.17*** join/#asterisk Iamnach0 (i=Iamnacho@ip174-70-137-120.ks.ks.cox.net)
05:12.43WindowsUser~softphone
05:12.44infobot[~softphone] A soft-phone is a program that lets your computer use a compatible audio interface (usually a speaker&mic, or headset), and optionally a webcam, to place VoIP calls with a given protocol (SIP/IAX/H.323/MGCP/etc). For links enter ~zoiper , ~xlite , ~twinkle , ~bria , ~eyebeam , ~ekiga
05:14.11NewtoPBXok so i guess the next question is..i am doing this for a project to see if i can make a conferance call server where you call in and ther is options like press 1 to talk to bla bla bal press 2 for bla bla. can asterisk do this ?
05:14.38[TK]D-FenderNewtoPBX: Yes
05:15.45NewtoPBXsweet
05:15.54NewtoPBXthen i will keep messing with it
05:16.12NewtoPBXinfobot that is a good link
05:16.13infobotNewtoPBX: I think you lost me on that one
05:16.14NewtoPBXthanks
05:16.43NewtoPBXlike.. say i want to talk to 10 people at once using asterisk
05:17.29WindowsUserthe bot is a bot btw
05:18.39NewtoPBXhaha
05:18.58WindowsUserNewtoPBX: they'd need headsets or handsets, 10 people all using the equiv to speakerphone would be really ugly under any circumstances
05:19.29NewtoPBXok
05:35.33*** join/#asterisk blkry (n=blkry@96.37.27.72)
05:36.49*** join/#asterisk blkry (n=blkry@96.37.27.72)
05:43.11*** part/#asterisk ramindia (n=balajibh@202.63.96.10)
05:45.08*** join/#asterisk phurl (n=mdupont@82.114.94.28)
05:48.15*** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir)
05:48.37[TK]D-FenderCheckout time, later all
05:49.48*** part/#asterisk phurl (n=mdupont@82.114.94.28)
05:50.58*** join/#asterisk leifmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage)
05:50.58*** mode/#asterisk [+o leifmadsen] by ChanServ
05:52.24*** join/#asterisk Heretic (n=BuRn@ZA1-securenode.echelon.co.za)
05:58.47*** join/#asterisk fakhir_ (n=fakhir@unaffiliated/fakhir)
06:17.20*** join/#asterisk callenb (n=callenb@host86-168-62-181.range86-168.btcentralplus.com)
06:17.50*** join/#asterisk Stdht (n=salfrede@212.98.187.129)
06:18.07StdhtHi! Could someone please help me. Asterik 1.4 under ubuntu. I try to connect mobigater via chan_celliax. I built chan-celliax.so from celliax_stuf.tgz from celliax.org
06:19.31*** join/#asterisk Chris-NB (n=chris@85-126-61-10.work.xdsl-line.inode.at)
06:20.58*** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun)
06:20.59StdhtI get segfault in app_userevent.so when incoming call
06:22.16*** part/#asterisk RIsa (n=Soap@84-75-148-82.dclient.hispeed.ch)
06:23.28*** join/#asterisk xrmx__ (n=rm@host129-254-dynamic.2-87-r.retail.telecomitalia.it)
06:26.54NewtoPBXsweet i got it working
06:26.56NewtoPBXthanks guys
06:29.26*** join/#asterisk serph (n=serph@CPE001f5b006e5e-CM001371144daa.cpe.net.cable.rogers.com)
06:36.28StdhtHow to ask for Help ?!!
06:37.20*** join/#asterisk botox93 (n=botox93@213.221.82.242)
06:38.46StdhtAnyone hears me??
06:43.25StdhtPLease HELP
06:43.51tzafrir_laptopinfobot, tell Stdht about ask
06:44.41tzafrir_laptopStdht, start by installing asterisk-dbg
06:44.48*** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290)
06:45.00tzafrir_laptopwith this you'll be able to make your core files meaningful
06:45.11tzafrir_laptop(assuming you use asterisk from packages)
06:45.42WindowsUserinfobot, tell WindowsUser about ask
06:45.58*** join/#asterisk fiddur (n=fiddur@c042.rit.se)
06:46.12WindowsUseri see, I'm used to doing those on channel
06:48.33tzafrir_laptopWindowsUser, you could simply use:  /msg infobot whatever
06:48.37*** join/#asterisk [T]ank (n=ckwall@c-71-199-25-239.hsd1.ut.comcast.net)
06:48.54tzafrir_laptopOr, alternativly, if you want this to show on the channel:
06:48.57tzafrir_laptop~whatever
06:48.58infobotsomebody said whatever was an expression of confusionambivalence, or something else
06:49.37[T]ankso what would cause a peer behind nat to register, but still show unavailable when I do a sip show peers?
06:51.00[T]ankI see what is going on... but not sure how to correct...
06:51.17[T]ankexternhost... looks like
06:53.08*** join/#asterisk puzzled (n=foobar@puzzled.xs4all.nl)
06:54.24Stdhtbefore I tried asterisk-dbg - I have asterisk1.4(from sources) under ubuntu8desktop. I have chan-celliax.so built from sources from celliax_stuff.tgz from celliax.org. I am testing asterisk with softphone on another PC. When I call extension which makes Dial(Celliax/mobi1:....) I receive "Unable to creare celliax channel cause-0 -Unknown" when I call from GSM to my celliax device (mobigater) - I get segfault in app_user_event.so
06:54.39*** join/#asterisk gr0mit (n=tim@2001:8b0:3d9:0:ecb8:f335:7b39:e2a6)
06:55.06Stdhtand asterisk crashes
06:56.53Stdhthow to install asterik_dbg
06:56.56tzafrir_laptopasterisk-dbg is indeed only relevant if you use packaged debs
06:56.58WindowsUsertzafrir_laptop: I was just testing if it'd work
06:57.22Stdhti can run asterisk -cvvvvvdddddddddd
06:57.36Stdhtbut there no any more specific info about error
06:58.02tzafrir_laptopStdht, can you get a core file?
06:58.17Stdhtwhat is it- core file ?
06:59.24Stdhtdump?
07:03.08*** join/#asterisk joako (n=joako@opensuse/member/joak0)
07:12.02*** join/#asterisk Stdht (n=salfrede@212.98.187.129)
07:12.05Stdhtsorry I was disconnected
07:12.20Stdhtwhich pid to use in  /usr/sbin/asterisk pid
07:13.51Stdhtor more generally hot to get core file
07:20.18tzafrir_laptopfor starters, run asterisk with -g
07:21.05Stdhtjr
07:21.08Stdhtok
07:21.10Stdhtstarted
07:21.36Stdhtand what to do now
07:24.11StdhtI tried asterisk -g ... it crashed as expected and what to do now?
07:24.30yanghelloes tzafrir_laptop
07:24.42tzafrir_laptophi
07:25.03tzafrir_laptopdid it leave a file called "core" ?
07:25.19Stdhtwhere?
07:25.59Stdhtoh yeas
07:26.06Stdhtit left
07:26.12*** join/#asterisk tamiel (n=tamiel@213.30.183.226)
07:26.26Stdhtcore.1387
07:26.30*** join/#asterisk UQlev (n=yuriy@nb11-125.static.cytanet.com.cy)
07:27.38*** join/#asterisk NewCastleScott (n=Scott@76-235-245-36.lightspeed.cicril.sbcglobal.net)
07:27.59tzafrir_laptopnext, get yourself a gdb (aptitude install gdb #or whatever)
07:28.20StdhtI have gdb installed
07:28.32tzafrir_laptopand run:   gdb -c core.1387 /usr/sbin/asterisk
07:28.46tzafrir_laptopin the gdb prompt, get a backtrace:
07:28.49tzafrir_laptopbt
07:29.03tzafrir_laptopbt full #more details, more noise
07:30.09Stdhtok I typed
07:30.25Stdhtdgb -c -core ... then bt full
07:30.52tzafrir_laptopif the output does not make sense to you at this point, pastebin it
07:30.53tzafrir_laptop~pb
07:30.54infobot[~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 , or , http://bin.cakephp.org/
07:32.06tzafrir_laptopinfobot, pb is also apt-get install pastebinit
07:32.07infobottzafrir_laptop: okay
07:32.11tzafrir_laptop~pb
07:32.12infobot[~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 , or , http://bin.cakephp.org/, or apt-get install pastebinit
07:32.39*** part/#asterisk [T]ank (n=ckwall@c-71-199-25-239.hsd1.ut.comcast.net)
07:36.53Stdhtsomething with souund system .... one mometn
07:37.33Stdhthttp://pastebin.com/m4d5aed1f
07:40.24Stdhtdst_exten - is smothing that I can't understand
07:41.40*** join/#asterisk icyValk77 (n=icyValk7@gateway.ash.thebunker.net)
07:45.39tzafrir_laptopStdht, as you can see, the crash is somewhere inside chan_celliax.so . But there does not appear to be debug information for it
07:45.57tzafrir_laptopDid you get it as  aseparate binary? or build it on your own?
07:46.32*** join/#asterisk leifmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage)
07:46.32*** mode/#asterisk [+o leifmadsen] by ChanServ
07:46.48StdhtI got it from http://www.celliax.org/celliax_stuff.tgz
07:46.59Stdhtthan I maked it
07:47.23StdhtI changed in make file just the path to asterisk 1.4 include
07:50.17Stdhtwhat could I do next,
07:50.18Stdht?
07:54.14*** join/#asterisk grEvenX (n=even@apb99b.ip.ssc.net)
07:57.03*** join/#asterisk war9407 (i=war@liquidswords.org)
07:59.21Stdhtcelliax.org is down with all of its forums.....:((
08:00.24*** join/#asterisk obnauticus (n=obnautic@about/windows/regular/obnauticus)
08:06.23*** join/#asterisk sulex (n=sulex@pdpc/supporter/professional/sulex)
08:08.55*** join/#asterisk s0lid (n=s0lid@203.177.143.137)
08:17.12*** join/#asterisk KermitTheFragger (n=KermitTh@118-197.bbned.dsl.internl.net)
08:26.20*** join/#asterisk jerryeguru (n=Administ@41.222.2.65)
08:27.51jerryegururecently i fitted an old degium pci card, not installed asterisk or anyother related software and lspci does not show me the card!
08:28.16jerryegurucould that mean, without related software the card is not visible or it is damaged
08:28.34jerryeguruI would like to get information about this interface card
08:32.34*** join/#asterisk ik_5 (n=ik@95.211.21.45)
08:34.18*** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-9046704fcc777d9c)
08:36.30*** join/#asterisk kamanashisroy (n=kamanash@202.56.7.215)
08:51.47*** join/#asterisk frk2 (n=frk2@zivios/member/fkhan)
08:56.18WindowsUserjerryeguru: if its pci and its plugged in properly it should show up in lspci
09:05.53WindowsUsersleepytime
09:06.16WindowsUserall i can suggest is reseat it (while powered down!)
09:12.38*** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-d1a1dc9d959fa4f3)
09:25.29*** join/#asterisk jks (n=jks@193.189.93.254)
09:26.29*** join/#asterisk jmls (n=jmls@host217-36-208-155.in-addr.btopenworld.com)
09:26.42jmlsmorning all
09:26.46jmlsusing 1.4, is there anyway of lighting a BLF on cisco/aastra phones from the dialplan or AMI ?
09:26.56jmlsexcept for using voicemail
09:30.16*** join/#asterisk zeeesh (n=zeeesh@203.215.179.43)
09:31.27ZhadCan you do it using SIP Notify?
09:31.55jmlsI really don't know ;)
09:32.13jmlscan you construct your own SIP notify message ?
09:32.25jmlsor are you limited to the existing ones ?
09:32.36ZhadProbably, using Add-Header, I've never tried it.
09:33.53ZhadSorry, SIPAddHeader
09:34.51Zhadalthough that only adds a header to the next call you make afaik.
09:35.34*** join/#asterisk arpu (n=arpu@chello062178159144.10.14.univie.teleweb.at)
09:41.06*** join/#asterisk mattboll (n=mattboll@lns-bzn-51f-81-56-151-195.adsl.proxad.net)
09:41.10mattbollhi
09:42.18mattbollI've got the "ringing but no voice" problem, but it happens only when I redirect an extern call to an other extern call
09:42.47mattbollI've got voip phones, i can call extern phones like a cellular
09:43.03mattbollI can call from a cellular to my voip phone
09:43.33mattbollwhen I receive a phone from a cellular, I can "transfer" it
09:44.23mattbollbut when I received an external call with this property :
09:44.24mattbollexten => 0974534610-0952044141,3,Dial,SIP/0632882022@ovh-3612
09:44.41mattbollthen it rings but we don't hear each other
09:44.52mattboll(btw : sorry for my english)
09:45.53mattbollSo, it seems to be a firewall problem but I don't know what I should search for because everything else work
09:46.28mattbollfirewall/nat…
09:49.48Zhadhave you tried turning can reinvite off?
09:49.57Zhadie. making sure that you stay in the loop.
09:52.09ZhadI didn't know you could do exten => 0974534610-0952044141
09:54.18mattbollI'll try the reinvite off
09:54.29mattbollthe exten is called by Goto(filter,0974534610-${CallerIDString},1)
10:03.03*** join/#asterisk grEvenX (n=even@apb99b.ip.ssc.net)
10:07.42*** join/#asterisk tamiel (n=tamiel@213.30.183.226)
10:08.32mattbollI had tTr so canreinvite was off, so I tried to turn it on but it didn't work
10:16.28*** join/#asterisk arpu (n=arpu@chello062178159144.10.14.univie.teleweb.at)
10:32.33Zhadwhat's wrong with using exten => 0974534610/0952044141 ?
10:32.49Zhadunless 0952044141 is not the CallerID(num).
10:50.38*** part/#asterisk callenb (n=callenb@host86-168-62-181.range86-168.btcentralplus.com)
10:53.05*** join/#asterisk Faithful (n=Faithful@ns.linuxterminal.com)
11:04.34*** join/#asterisk cusco (n=tralala@2001:0:53aa:64c:1460:36b:b2c9:9de6)
11:07.26*** part/#asterisk Stdht (n=salfrede@212.98.187.129)
11:27.11*** join/#asterisk Stdht (n=salfrede@212.98.187.129)
11:33.21zeeeshsip to yahoo voice chat ... is there any posibility with asterisk ?
11:37.09*** join/#asterisk mltlnx (n=mltlnx@cpe-24-193-11-14.nyc.res.rr.com)
11:42.21*** join/#asterisk thansen (n=thansen@76.27.110.194)
11:44.17*** join/#asterisk pbxgeek (n=pbxk1064@203.171.196.78)
11:49.40*** join/#asterisk voipheroes (n=voiphero@AMontsouris-757-1-28-165.w90-46.abo.wanadoo.fr)
11:54.11mattbollI didn't know the 0974534610/0952044141 syntax, but it works so I won't change now
11:54.32mattbollwell it works unless I call an external number :/
11:57.06*** join/#asterisk AsteriskDom (n=amunoz@75.112.88.200.m.sta.codetel.net.do)
12:02.04*** join/#asterisk frk2 (n=frk2@zivios/member/fkhan)
12:02.39*** join/#asterisk l2trace99 (n=jr@rrcs-71-43-104-238.se.biz.rr.com)
12:06.58*** join/#asterisk l2trace99 (n=jr@rrcs-71-43-104-238.se.biz.rr.com)
12:09.14*** join/#asterisk angryuser (n=angryuse@LPuteaux-151-42-27-99.w193-251.abo.wanadoo.fr)
12:11.31*** join/#asterisk SuPrSluG (n=SuPrSluG@72.237.213.162)
12:16.23*** join/#asterisk ariel_ (n=chatzill@198.211.95.37)
12:16.26ariel_morning
12:20.57*** join/#asterisk michael-i (n=michael-@141.41.40.153)
12:23.07*** join/#asterisk fernandojdk (n=fernando@189.88.64.36)
12:25.15*** join/#asterisk qdk (n=qdk@81.7.168.130)
12:27.59*** join/#asterisk mohawk (n=ross@host217-40-110-153.in-addr.btopenworld.com)
12:32.02*** join/#asterisk DarkRift (n=dark@65.92.165.235)
12:36.08*** part/#asterisk icyValk77 (n=icyValk7@gateway.ash.thebunker.net)
12:36.27mattbollariel_: it's 2pm here >_<
12:37.14ariel_mattboll: sorry about that, but it's just 8:37 am and just starting....
12:42.38*** join/#asterisk jaytee (n=jforde@unaffiliated/jaytee)
12:50.38*** join/#asterisk [intra]lanman (n=lanman@freeswitch/developer/intralanman)
12:50.56Pan3Dexten => _1NXXNXXXXXX,1,Dial(goodmorning@you,300,r)
12:51.05*** join/#asterisk davevg-btwtech (n=davevg-b@67.76.177.147)
12:54.18mattbollgeek ! >_<
13:00.28*** join/#asterisk jicksta (n=jicksta@c-67-169-183-106.hsd1.ca.comcast.net)
13:11.39*** part/#asterisk Stdht (n=salfrede@212.98.187.129)
13:12.50*** join/#asterisk iksik (n=xk@livedata.pl)
13:13.19*** join/#asterisk hi365 (n=hi365@94.159.178.170)
13:17.18*** join/#asterisk leifmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage)
13:17.18*** mode/#asterisk [+o leifmadsen] by ChanServ
13:17.25*** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net)
13:19.38*** join/#asterisk Whitor (n=Whitor@64.128.237.124)
13:21.01*** join/#asterisk brad_mssw (n=brad@216.155.101.90)
13:22.29*** join/#asterisk qdk (n=qdk@81.7.168.130)
13:25.10*** join/#asterisk Stdht (n=salfrede@212.98.187.129)
13:26.08StdhtHi. Does asterisk passes dtmf through by default... For instalce I call from sip softphone to asterisk and asterisk calls external line... how could I pass dtmf to external ivr
13:27.09*** join/#asterisk jmacz (n=jmacz@190.144.75.22)
13:32.07*** join/#asterisk tobias (n=tobias@cpe-071-070-219-040.nc.res.rr.com)
13:32.10*** join/#asterisk juanIMP (n=juan@200.71.41.254)
13:32.12*** join/#asterisk UQlev (n=yuriy@nb11-125.static.cytanet.com.cy)
13:32.35*** join/#asterisk minotaur01 (n=test@dhcp-0-13-46-41-d9-29.cpe.mountaincable.net)
13:33.51minotaur01hello everyone
13:34.57AiatekStdht thats call disa
13:36.42Stdhtyes Disa I've found that asterisk reproduces  dtmf
13:36.50Stdhtto other side
13:37.47*** join/#asterisk l2trace99 (n=jr@205.245.6.162)
13:38.12minotaur01im having a problem with audio playback (IVR/Directory) over IAX and im wondering if anyone can help me
13:39.33Aiatekminotaur01 describe your problem and maybe someone here can help you
13:40.36*** join/#asterisk Great_Anta_Baka (n=tensai@196.33.159.83)
13:40.49minotaur01when i use a SIP connection the audio plays fine but over IAX it's choppy and it's as if it's in slow motion
13:40.54Great_Anta_Bakasup all!!!
13:41.05*** part/#asterisk Stdht (n=salfrede@212.98.187.129)
13:41.08*** join/#asterisk Stdht (n=salfrede@212.98.187.129)
13:41.49SuPrSluGminotaur01:do you have jitter buffer enabled?
13:41.56Great_Anta_Bakaplease tell me if i am laughing for nothing.. this company has come into this business park which has 19 offices and is running an asterisk server in each of the 19 virtual machines!!!
13:42.41*** join/#asterisk brah (n=asdfaf@190.224.131.138)
13:43.04minotaur01SuPrSluG: it's a clean install let me check
13:43.21*** join/#asterisk Iamnacho (i=Iamnacho@ip174-70-137-120.ks.ks.cox.net)
13:43.25*** join/#asterisk fish-bulb (n=cstewart@nat/digium/x-66708dca34377bb5)
13:46.11*** join/#asterisk ingenius (n=alektro@host147.190-229-174.telecom.net.ar)
13:52.03minotaur01SuPrSluG: i belive it's not on i checked the iax conf files there is no jitter buffer setting
13:52.35*** join/#asterisk DarkRift (n=dark@65.92.165.235)
13:54.34leifmadsenminotaur01: see https://issues.asterisk.org/view.php?id=15337
13:59.13*** join/#asterisk uTx (n=unix@modemcable195.1-23-96.mc.videotron.ca)
14:03.59*** join/#asterisk maddog01 (n=test@24.213.70.158)
14:10.16Kattypeeks in
14:12.48*** join/#asterisk davevg-btwtech (n=davevg-b@67.76.177.147)
14:13.30*** join/#asterisk Amorsen (n=Amorsen@94.127.50.7)
14:14.05*** join/#asterisk KavanS (n=KavanS@static-71-117-242-28.ptldor.dsl-w.verizon.net)
14:14.07AmorsenIs there a way to detect whether a card is TE410P (3.3V) or TE405P (5V) without driving to the server room and opening the case?
14:14.45*** join/#asterisk galeras (n=galeras@186.80.186.118)
14:14.48*** join/#asterisk davevg (n=davevg-b@nj-67-76-177-147.sta.embarqhsd.net)
14:15.41galerasSemi * question: what mail client do you suggest to implement a mail to fax solution, i f mail server is exchange?
14:16.05mogAmorsen, its probably easier to figure out what kind of server it is in
14:16.12mogas most servers have 3.5 or 5 v slots
14:16.29Amorsenmog: Dual voltage, unfortunately
14:16.41mogaww
14:16.51AmorsenHandle 0x0018, DMI type 9, 13 bytes
14:16.56mogthere is no way to tell that i know of
14:16.56Amorsen<PROTECTED>
14:16.56Amorsen<PROTECTED>
14:16.56Amorsen<PROTECTED>
14:17.06AmorsenThat's dmidecode
14:17.35mogAmorsen, digium might have started giving them different sub ids recently, but as long as i have known they look the same
14:17.37AmorsenThere's no identifier anywhere on the card?
14:17.40AmorsenOk
14:18.29AmorsenHmm, maybe iLO knows.
14:19.27AmorsenNope.
14:20.58*** join/#asterisk Vec (n=Vector@host-87-74-7-50.dslgb.com)
14:21.29AmorsenThe other card is easy:
14:21.31Amorsen05:01.0 Communication controller: Digium, Inc. Wildcard TE410P quad-span T1/E1/J1 card 3.3V (rev 02)
14:21.45AmorsenSo that probably means card 1 is 5V
14:25.31*** join/#asterisk [TK]D-Fender (n=joe@64.235.218.194)
14:27.37VecI am trying this: but its not working, any ideas:
14:27.38Vecexten => s,n,ExecIf($[${LEN(ARG5)} = 0]|Set|TIME=4)
14:27.49VecI would like to keep it to 1 line
14:28.52*** join/#asterisk jplank (n=GBove@98.24.169.236)
14:29.43*** join/#asterisk theHub (n=theHub@69.177.93.21)
14:30.08beekVec:  more like:   ExecIf($[${LEN(ARG5)}=0]?Set,TIME=4)
14:30.20beekVec: which version of asterisk?
14:30.45Vecbeek : 1.4 latest
14:33.04*** join/#asterisk telnettech (n=telnette@cpe-71-74-91-116.insight.res.rr.com)
14:33.44Vecbeek : not working, the variable is not being set although ${LEN(ARG5)} does = 0
14:34.09*** join/#asterisk Deeewayne (n=dwayne@75.76.254.162)
14:34.09*** mode/#asterisk [+o Deeewayne] by ChanServ
14:34.25*** join/#asterisk grEvenX (n=even@cB78D5AC1.dhcp.bluecom.no)
14:37.24Vecbeek : found the prob: I was looking at the length of ARG4, for some reason even though it is not SET it still has a length of 4, for some reason it contains 4 spaces.
14:38.17wdoekesVec: isn't LEN(ARG4)=4 ?
14:38.30wdoekesor does it expand $ARG4 ?
14:38.37wdoekes*${ARG4}
14:39.03[TK]D-Fender^^
14:39.15[TK]D-FenderYou don't pass it a var nam, you pass it TEXT
14:39.22beekVec: Missed that... it IS LEN(${ARG4})
14:40.39leifmadsen[TK]D-Fender: I can understand that confusion since there are several functions (like CUT()) that don't want the text -- they want the variable name
14:41.56[TK]D-Fenderleifmadsen: Yeah... but they both have instructions....
14:42.04[TK]D-Fenderleifmadsen: But seriously... who reads those? ;)
14:42.28leifmadsenno effin' idea
14:44.05beekIt's not a lack of RTFM, it's more that your eyes start to go buggy looking at that melange of ({[]}) characters.   The dialplan instruction looks more like transmission line noise than code.
14:45.19[TK]D-Fenderbeek: /me squelches beek
14:45.23[TK]D-Fender:O
14:47.00brahwhat is this i don-t even
14:47.02brahchannel.c:2401 set_format: Unable to find a codec translation path from 0x4 (ulaw) to unknown
14:50.17*** join/#asterisk [T]ank (n=ckwall@132.sub-75-216-112.myvzw.com)
14:51.46[T]ankso I have a server I set up last night which is remotely hosted. actual phone calls work perfectly. But any playback of files, for example Playback(tt-monkeys) doesn't play back very well. The server shows it working... and every once in a while you can hear the file clip in and out... the are playing at normal speed and such, you just cant hear it. any ideas on what could cause that?
14:51.53[T]ankI am not getting any errors.
14:52.01[T]ankjust not hearing more than about 1% of the total file
14:52.10*** join/#asterisk ManxPower (n=manxpowe@69.73.94.162)
14:52.13[T]ankcalled from multiple cell carriers to test
14:52.19[T]anktried multiple files.
14:52.36maddog01[T]ank: what version are you using
14:52.37[T]ankthe voicemail messages are the same quality as playback
14:52.55ManxPower[T]ank: What is the problem?
14:53.12[T]ankmaddog01: 1.6.1.1
14:53.20maddog01[T]ank: and are you using iax
14:53.27[T]ankyes
14:53.30[T]ankwell...
14:53.37[T]ankfrom the carrier it is iax.
14:53.37*** join/#asterisk jpeeler (n=jpeeler@asterisk/digium-software-dev/jpeeler)
14:53.39maddog01same problem i just fixed
14:53.52[T]ankcarrier to the server... then from the server to the phone it is sip
14:53.52maddog01down grade to 1.6.0.1
14:54.01[T]ankreally?!!
14:54.02maddog01all will be fixed
14:54.04[T]ankjust a bug?
14:54.08maddog01it a bug
14:54.12[T]ankok
14:54.13maddog01wait one sec
14:54.14[T]ankgood to know.
14:54.28maddog01https://issues.asterisk.org/view.php?id=15337
14:54.37maddog01there thats the info
14:54.57Vecwdoekes, [TK]D-Fender, beek : for some reason Arguments that are passed into a Macro even when they are empty eg, ${ARG9} seem to consist of 4 spaces "    ", so LEN returns 4, is this normal ?
14:55.21[T]anklooks a little different... mine you just cant hear... that looks more like the issue that the files are played back super slow.
14:55.26[T]ankam I wrong?
14:55.30[TK]D-FenderVec: not AFAIK, show me
14:55.39[T]ankmy issue is that I cannot hear the file even though it is being played.
14:55.51[T]ankeasy fix for me... just change to sip :-D
14:55.53maddog01i had the same problem
14:56.01[T]ankI will try sip first.
14:56.08[T]ankthanks so much for the reply
14:56.11ManxPowerI like SIP for inter-asterisk stuff.
14:56.11maddog01i couldn't hear anything from the outside
14:56.34maddog01[T]ank: i couldn't hear anything from the outside either
14:57.03[T]ankthanks... now I know how to fix. two options. wanted to move to sip anyway
14:57.07[T]ankthanks a bunch
14:57.25maddog01np
14:57.49ManxPowerVec: Paste the exact Macro(.... line you are using
14:57.57Vec[TK]D-Fender : not sure how to show u, but here is the code and the output: http://pastebin.com/d672439a2
14:58.47[TK]D-FenderVec: I don't see you executing the macro <-----
14:58.49ManxPowerthat's not the macro(... line.
14:59.29[TK]D-Fenderexten => s,n,Verbose(1|The Length of ARG5 is ${LEN(ARG9)})  <----  SAYS 5 BUT YOU'RE LOOK AT ***9***
14:59.45[TK]D-FenderAnd without reverenceing the car
14:59.55[TK]D-FenderAnd without referencing the variable
14:59.56Vecyes I know 9 is not in use, the same thing happens with 5!!!
15:00.01[TK]D-FenderVec: We just went through this!@
15:00.09leifmadsen${LEN(${ARG9})}
15:00.12VecI will get the macro line
15:00.15ManxPowerVec: less talk, more pasting of the actual calling of the macro
15:00.18leifmadsenthe length for ${LEN(ARG9)} is 4
15:00.21[TK]D-FenderVec: NO
15:00.22Vecok ok 1 sec
15:00.28[TK]D-FenderVec: You pass LEN **test**, not a var name!
15:00.31[TK]D-Fendertext
15:00.44ManxPower[TK]D-Fender: good call!
15:01.09ManxPowerARG9 is 4 chars.  Of course ${ARG9} would be however many chars the value is.
15:01.16*** join/#asterisk comradeb14ck (n=comradeb@72.37.252.50)
15:01.30Vecoh oops
15:01.35comradeb14ckhi all
15:01.52VecLOL that is dumb
15:01.53Vecok thanks
15:01.54Vecget it
15:03.17*** join/#asterisk marv[work] (n=timr@router.asteriasgi.com)
15:05.52*** join/#asterisk errotan (n=errotan@5403E710.catv.pool.telekom.hu)
15:07.10*** join/#asterisk boch (n=fran@200.61.191.9)
15:08.47bochis it possible to continuing executing dialplan after Park() is called? I am taking the call with another call but when the second one ends, it hangs up the first one too
15:10.22[TK]D-Fenderboch: Show us
15:20.44*** join/#asterisk s14ck (n=s14ck@ccscliente156.ifxnetworks.net.ve)
15:21.09*** part/#asterisk [T]ank (n=ckwall@132.sub-75-216-112.myvzw.com)
15:21.13*** join/#asterisk Defraz (n=T0tal@fw-poky.fuzecore.com)
15:21.44*** join/#asterisk DLNoah (n=chatzill@72.11.29.130)
15:21.59boch[TK]D-Fender, http://pastebin.com/m5286df7b
15:23.50*** join/#asterisk coppice (n=chatzill@204.196.17.210.dyn.pacific.net.hk)
15:27.12*** join/#asterisk neurosys (n=neurosys@173-9-159-182-miami.txt.hfc.comcastbusiness.net)
15:27.54[TK]D-Fenderboch: How are you transferring the call?
15:27.54*** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw)
15:27.58[TK]D-Fenderboch: And next time, show the ENTIRE call
15:28.14*** join/#asterisk amazinzay (n=amazinza@67.108.187.186.ptr.us.xo.net)
15:28.20*** join/#asterisk ajohnson (n=ajohnson@63.147.46.186)
15:28.25*** join/#asterisk awk_r (n=awk@nat/digium/x-ed3c56d7c25d73ba)
15:28.52boch[TK]D-Fender, with Park() app
15:28.58amazinzayIs there any way to disconnecte manager connections from the CLI?
15:29.18[TK]D-Fenderboch: You don't transfer with park, you transfer with your PHONE
15:29.58boch[TK]D-Fender, so im not transfering the call, im running Park() in dialplan
15:30.37[TK]D-Fenderboch: You do an attended transfer of a call you're on to Par(),  You don't jsut start a new call and hang up
15:32.12korcancan I use open source Asterisk with a PRI card?  OpenVox D410E ?
15:32.35*** join/#asterisk [intra]lanman (n=lanman@freeswitch/developer/intralanman)
15:32.50korcanSomeone told me I have to pay for a business license to use a PRI card...
15:32.57russellbkorcan: that is not true.
15:33.03russellbthat person is full of lies.  :-)
15:33.39korcanThanks :)  thats what I needed to know!!!
15:33.45*** join/#asterisk jameswf (n=james@unaffiliated/jameswf-home)
15:33.54korcanrussellb, any experience with OpenVox cards?
15:34.19*** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-179-75-202.static.bezeqint.net)
15:34.21russellbI have no experience with them.  I also do not recommend them.  I recommend the use of Digium cards (I work for Digium).
15:34.33russellbI do not support cheap chinese clones :-p
15:34.41comradeb14ckkorcan, the older models of openvox cards (like the d410e) is an exact clone of digium
15:34.49Joeldon't use a clone
15:34.55Joelanyone who says a clone is ok is retarded
15:35.03comradeb14ckthe newer openvox models, the 800 and 1200 are not clones (they have their own drivers)
15:35.32*** join/#asterisk intralanman (n=lanman@173-132-223-176.pools.spcsdns.net)
15:35.56comradeb14ckIf you want something stable that will last you a while, I recommend digium/sangoma/rhino cards. + those companies all have English tech support. OpenVOX has no english tech support.
15:36.05comradeb14ckSo you'll be kinda screwed if you need assistance with it.
15:36.13amazinzayDoes anyone know how to disconnect a manager from the CLI?
15:36.19*** join/#asterisk IBC_jkenney (n=jkenney@99.23.50.73)
15:36.23comradeb14ckrestart now? =p
15:36.29[TK]D-Fenderkorcan: The OpeVox clones are the crappy old PCI design, and their support sucks, so if you have problem, you're asking for trouble
15:36.50ManxPowerDigium moved away from their original design for a reason.  PCI issues.
15:36.52jameswfkorcan: And they hate kittens...
15:37.06jameswfjust sayin
15:37.23amazinzayanything that doesn't take the phone sytem down?
15:37.32korcanI need a 4 port PRI Card, and a 12 Port FXS Card
15:37.36denonManxPower: it's a pity USB was designed so poorly
15:37.51jameswfUSB 3.0 <3
15:38.01coppiceif they hate kittens they can't be all bad
15:38.01korcanany recommendations?
15:38.26korcanI must have missed something....  What do kittens have to do with anything?
15:38.27denonjameswf: I was thinking something more like ePCIe :)  or Infiniband on commodity PCs
15:38.43korcanPCI Express
15:40.00jameswfkorcan: you havent been on the internet verry long kittens have everything to do with everything
15:40.05jameswfhttp://icanhascheezburger.com/
15:40.58IBC_jkenneyI am also confused on the kittens
15:41.10russellb<3 kittens
15:41.56*** part/#asterisk jerryeguru (n=Administ@41.222.2.65)
15:42.05korcanso any PCI-E card recommendations?
15:43.35Qwellkorcan: Digium, naturally
15:43.55QwellI think all of our current hardware has PCIe models
15:44.14russellbkorcan: http://www.digium.com/en/products/digital/te420.php
15:44.25IBC_jkenneyI heard a rumor that if you have a Digium card you need to use a business license of asterisk
15:44.27IBC_jkenneyis that true
15:44.43QwellIBC_jkenney: no..
15:45.00IBC_jkenneyso that is just an ugly rumor?
15:45.09QwellIBC_jkenney: one I've never heard before
15:45.10russellbkorcan: http://www.digium.com/en/products/analog/aex2400.php ... it's a 24 port analog card, the next smaller is 8
15:45.14denonprobably heard it from you sangoma rep?
15:45.21russellbIBC_jkenney: where did you hear that?  that has never been the case
15:45.33denonyou/your
15:46.32jameswfadds rumor to sales schpeel
15:46.38jameswf:)
15:47.11Qwelljameswf: write me a script to automatically connect to all my VMs to build new packages when CentOS releases a new kernel update, k?
15:48.10Joelhaha
15:48.10[TK]D-FenderQwell: Outsourcing to the competition now? ;)
15:48.10[TK]D-Fender*ouch*
15:48.10Joelkeep dreaming Qwell
15:48.10jameswfholy netsplit batman
15:48.10Qwell[TK]D-Fender: it's in his best interest!
15:48.12denonQwell: PXE boot all your VMs, and NFS all your configs to one place :)
15:48.12JoelQwell just got told about cobbler, would have been very handy a few days ago :\
15:48.49*** join/#asterisk Corydon76-dig (i=three@pdpc/supporter/bronze/Corydon76-home)
15:48.49*** join/#asterisk coppice (n=chatzill@204.196.17.210.dyn.pacific.net.hk) [NETSPLIT VICTIM]
15:48.49*** join/#asterisk jpeeler (n=jpeeler@asterisk/digium-software-dev/jpeeler)
15:48.49*** join/#asterisk grEvenX (n=even@cB78D5AC1.dhcp.bluecom.no) [NETSPLIT VICTIM]
15:48.49*** join/#asterisk theHub (n=theHub@69.177.93.21) [NETSPLIT VICTIM]
15:48.49*** join/#asterisk Vec (n=Vector@host-87-74-7-50.dslgb.com) [NETSPLIT VICTIM]
15:48.49*** join/#asterisk fish-bulb (n=cstewart@nat/digium/x-66708dca34377bb5) [NETSPLIT VICTIM]
15:48.49*** join/#asterisk juanIMP (n=juan@200.71.41.254) [NETSPLIT VICTIM]
15:48.49*** join/#asterisk iksik (n=xk@livedata.pl) [NETSPLIT VICTIM]
15:48.49*** join/#asterisk mohawk (n=ross@host217-40-110-153.in-addr.btopenworld.com) [NETSPLIT VICTIM]
15:48.49*** join/#asterisk voipheroes (n=voiphero@AMontsouris-757-1-28-165.w90-46.abo.wanadoo.fr) [NETSPLIT VICTIM]
15:48.49*** join/#asterisk kamanashisroy (n=kamanash@202.56.7.215)
15:48.49*** join/#asterisk bbryant (n=brett@c-68-59-22-114.hsd1.sc.comcast.net)
15:48.51*** join/#asterisk korcan (n=korcan@99.23.50.73) [NETSPLIT VICTIM]
15:48.51*** join/#asterisk oilinki (n=oil@ip-208-109-22-97.ip.secureserver.net)
15:48.51*** join/#asterisk viq (n=viq@unaffiliated/viq)
15:48.51*** join/#asterisk malcolmd (n=malcolmd@pdpc/sponsor/digium/malcolmd)
15:48.51*** join/#asterisk MT`AwAy (n=MagicalT@shigoto.ookoo.org) [NETSPLIT VICTIM]
15:48.51*** join/#asterisk AndyML (n=AndyML@pool-173-49-143-205.phlapa.fios.verizon.net) [NETSPLIT VICTIM]
15:48.51*** join/#asterisk timeshell_atwork (n=chatzill@gw.lusi.on.ca) [NETSPLIT VICTIM]
15:48.51*** join/#asterisk smurf (n=smurf@debian/developer/smurf)
15:48.51*** join/#asterisk ltd (n=z@pat.transact.net.au) [NETSPLIT VICTIM]
15:48.51*** join/#asterisk creativx (n=creadure@226.62-97-205.bkkb.no) [NETSPLIT VICTIM]
15:48.51*** join/#asterisk Poincare (n=jefffnod@amp89.ampersant.be) [NETSPLIT VICTIM]
15:48.51*** join/#asterisk mmlj4 (n=jkelly@ip70-171-94-246.no.no.cox.net) [NETSPLIT VICTIM]
15:48.51*** join/#asterisk ISO9001 (i=blank@slu.ms) [NETSPLIT VICTIM]
15:48.51*** join/#asterisk DarkLogik (n=darklogi@darklogik.com) [NETSPLIT VICTIM]
15:48.51*** join/#asterisk wimt (i=wimt@freenode/staff/wikipedia.wimt) [NETSPLIT VICTIM]
15:48.51*** join/#asterisk NoxIn- (n=noxin@zorlit.org) [NETSPLIT VICTIM]
15:48.51*** join/#asterisk mnicholson (n=mnichols@nat/digium/x-7f5e8449999e8403)
15:48.51*** join/#asterisk MatBoy (n=MatBoy@wiljewelwetenhe.xs4all.nl) [NETSPLIT VICTIM]
15:48.51*** join/#asterisk Talkradio (i=talkradi@linuxgeneration.ca) [NETSPLIT VICTIM]
15:48.51*** mode/#asterisk [+o Corydon76-dig] by irc.freenode.net
15:48.51QwellJoel: meh
15:48.51[TK]D-FenderQwell: He doesn't charge any, damn sub-prime loans!
15:48.51Qwelldenon: persistent VMs + svn :D
15:48.51jameswfQwell: you know you don't actualy have to run a kernel to build against it...
15:48.51Qwelljameswf: I know
15:48.51Qwellbut I have to install it, and build the packages
15:49.01*** part/#asterisk Amorsen (n=Amorsen@94.127.50.7)
15:49.01jameswfI has a robot that syncs to the repos and spits out RPM's he is a neat little fella
15:49.22denonQwell: I find it's best to let your kernels age .. like a fine wine .. few new security advisories every few months gives them flavor
15:49.50*** join/#asterisk Iamnach0 (i=Iamnacho@174.70.137.120)
15:50.28jameswfdenon: Security is over rated... chmod -R denons_mom
15:50.46jameswfcrap
15:50.58jameswfchmod -R 777 denons_mom
15:51.14denonyeah chmod requires some args .. it's not funny once you screw up the insult :)
15:51.20jameswflol
15:52.06IBC_jkenney<russellb> a co worker brought it up plus i read it on a few lists
15:52.18russellbweird.
15:53.02jameswfProbably an old troll steve post
15:53.17[TK]D-Fender[11:32]<korcan>Someone told me I have to pay for a business license to use a PRI card...
15:53.18jameswfwhat happened to steve the sangoma troll
15:53.21[TK]D-FenderAnd this poor chap :)
15:53.38beekCan anyone here give me the definitive answer to the question:  Does ADA (Asterisk Desktop Assistant) have a plugin that works with Internet Explorer like it does with Firefox?
15:53.42IBC_jkenneysee i am not the only one
15:54.18denon[TK]D-Fender: I heard that digium cards all call home to activate, and digium is going to start charging ongoing license fees ... </rumor type='foolish'>
15:55.05JoelI heard Qwell was going to propose to me
15:55.52jameswfI heard digium is being bought by Microsoft
15:56.19Qwelljameswf: old rumor
15:56.26russellbwe're rewriting Asterisk in bash
15:56.26IBC_jkenneyI heard Microsoft is buying everyone
15:56.38JoelI heard a rumor asterisk was getting more stable
15:56.40Joel</burn>
15:56.41Joel;)
15:56.43QwellJoel: FALSE
15:56.45Qwellerr, wait
15:56.45*** join/#asterisk _mwoodj_ (n=mwoodj@pdpc/sponsor/digium/hyper-eye)
15:56.46jameswfExcept freeswitch that was bought by baracuda
15:57.24denonMORE stable? you imply that there's something more stable than perfectly stable?
15:57.31russellbslaps Qwell
15:57.32Joelspeaking of bs, global warming, it was 60F for the high yesterday, week before it was 85 out
15:57.40Qwellrussellb: what? :(
15:57.51russellbQwell: you said Asterisk wasn't getting more stable!
15:58.09QwellI misread!
15:58.29jameswfokay lets try this on
15:58.48jameswfDigium is reported to be in talks to buy Fonality for $50
15:58.53russellblol
15:58.53Qwellpfft
15:58.56Qwellclearly false
15:59.02Qwellwho would pay $50 for that?
15:59.03denonI doubt fonality would bring 50 bucks
15:59.04leifmadsenwhat a rip off that'd be
15:59.31denonunless the sale included some cheeeezburgers for kitty
16:00.31JoelI love how everyone knocks fonality, work environment wise it's probably the most fun I've ever had
16:00.47Joelread: awesome when not dealing with chris 'cry baby' lyman
16:00.52*** join/#asterisk ManxPower (n=manxpowe@69.73.94.162)
16:02.09leifmadsenhmmm... putting a DROP rule in iptables just before the 2 rules you want to allow traffic with is a sure fire way to make them not work :)
16:02.19Joelleifmadsen :P
16:02.26jameswfoh crap Fonality troll in here...
16:02.36leifmadsenis an asterisk troll
16:02.39leifmadsenso it balances out
16:02.44leifmadsens/troll/advocate
16:03.00jameswf!!!!!!!!
16:03.08leifmadsenI don't want to meet your mom.....
16:03.11jameswfs/!/poo /g
16:04.00leifmadsennice :)
16:04.07Joeljameswf how's rhino's deal with fonality going?
16:04.17jameswf?
16:05.03Joeloh oops, nm.
16:10.50jameswfwants a Jalipeno cheese bagel...
16:11.04theharmmmm me 3
16:11.25ariel_oh boy, it's hot pockets for lunch today....;-(
16:11.36Joelariel_ barf pockets?
16:11.51*** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun)
16:11.53ariel_well yes but it's cheap and it's in the vending machine
16:12.33*** join/#asterisk Boingo (n=malachi@mail.techglia.com)
16:13.31*** join/#asterisk DelphiWorld (n=Miranda@41.201.82.28)
16:13.34DelphiWorldhello
16:13.36BoingoHello everyone, I have been looking for hard drive sizing suggestions for Asterisk.  I found CPU/RAM/etc but not much for hard drive.  I intend to save voicemails for < 10 users, only a few voicemails a day.
16:14.06DelphiWorldQwel: what is the included asterisk version in asterisknow 1.5?
16:14.37QwellBoingo: assume each of those 10 users have 100 VMs saved..  thats 1GB *at most*
16:14.52ManxPowerBoingo: What format will your voicemail be in and how many mins of messages do you want to store (not number of messages, number of mins)
16:15.51leifmadsen10 users? any modern HD is going to be fine
16:16.05BoingoDont really care which format.  Figure 2/3 minutes per message?  3 per day? So 9ish minutes per day * < 10 users.  90 minutes per day?
16:16.25ManxPowerBoingo: I suggest using WAV49
16:16.29leifmadsen120GB will be more than enough
16:16.33BoingoThat is way on the high side of what will actually happen.. but it should be good for an estimate.
16:16.39ManxPowerAs leifmadsen said, any modern drive will be fine.
16:16.41leifmadsenit's not gonna be like 1MB a minute or anything :)
16:16.47leifmadsenand even then... :)
16:16.55ManxPowerleifmadsen: WAV would be like 1 mb/min wouldn't it?
16:16.58BoingoWhy Wav49?
16:17.15leifmadsenManxPower: yes, unless you use wav49 like others use :)
16:17.27ManxPowerBoingo: because if at some time in the future you want the messages e-mailed to the user, WAV49 is the only small format WMP knows about without extra software.
16:17.41BoingoOh.
16:17.48BoingoI do want to email the voicemails.
16:17.52ManxPowerand you do NOT want to change formats on your voicemail store in the middle.
16:17.57DelphiWorldqwel: tel me please
16:18.08BoingoWMP doesnt know about mp3?
16:18.10DelphiWorldqwel: what is the included asterisk version in asterisk now 1.5?
16:18.15BoingoOr is Wav49 smaller?
16:18.28BoingoOr is MP3 generally frowned upon for some other reason?
16:18.30ManxPowerBoingo: What makes you think Asterisk can (out of the box) save voicemails in MP3 format?
16:18.41BoingoNo particualr reason.
16:18.57ManxPowerMP3 is patent encumbered.  Any MP3 support would have to be added from an outside package/source
16:19.03BoingoI guess somewhere in the back of my head, that was what I was aiming for.
16:19.08BoingoAh, ok.
16:19.12BoingoNo biggie.
16:19.15JoelQwell ok for the record, you don't need a comps.xml in the yum repo. so yum repo solves the issue for me
16:19.15BoingoWhatever works.
16:19.20leifmadsenBoingo: wav49 is basically GSM with a WAV wrapper on it so Windows users can play it.
16:19.36QwellJoel: you do for groups
16:19.38leifmadsenBoingo: i.e. it is a compressed format
16:19.41JoelQwell correct
16:19.42BoingoI assume Mac/Linux as well?
16:19.51JoelQwell but who needs those in a headless install?
16:19.53*** join/#asterisk phurl (n=mdupont@82.114.94.9)
16:19.58BoingoI have users on all OSes.
16:20.01leifmadsenBoingo: correct
16:20.04BoingoCool.
16:20.08BoingoSounds like a winner.
16:20.54BoingoIs there anything else that is hard drive consuming besides voicemail?
16:21.02ManxPowerBoingo: logs
16:22.06BoingoAny ideas on sizing for that?
16:22.20BoingoI am assuming I don't need to keep logs until the end of time.
16:22.34BoingoRotate them out after a certain amount of time.
16:23.05ManxPowerBoingo: it all depends on how much logging you enable and how many calls you get.
16:23.23ManxPowerOn some systems I've installed logs take up like 100K/day.  On other systems it's like 5GB/day
16:23.42BoingoWell, I am probably a lot closer to the bottom ned.
16:23.44Boingo*end
16:23.49ManxPowerDo you really think a modern HD would not meet your needs?
16:24.01BoingoYes, I think it will.
16:24.06BoingoJust trying to get an idea.
16:24.50BoingoDoes Asterisk play nice in a VM?  All VoIP, no cards.
16:25.03BoingoMainly my reason for asking about the HD I guess.
16:25.06ManxPowerBoingo: no.
16:25.10BoingoNo?
16:25.16BoingoI am surprised.
16:25.21ManxPowerBoingo: Asterisk is a pesudo realtime system.
16:25.28BoingoLooking at the CPU/RAM specs.... I figured.
16:25.54ManxPowerBoingo: will you be using any conferencing?
16:26.01BoingoDoubtful.
16:26.14ManxPowertry it and see, just don't expect it to work well under load.
16:26.29BoingoWhere would the bottleneck be?
16:26.37ManxPowerBoingo: latency and jitter
16:27.23BoingoSo I am better off dropping it on an old 486 than I am putting it in a VM with 10x the horsepoweR?
16:27.49ManxPowerBoingo: a 486 may not have enough cpu power to do much better.
16:28.21BoingoI was reading a page that had all sorts of really old computers listed that were running it.  Or was I missing something?
16:28.35ManxPowerRemember Asterisk is a SOFTWARE PBX Toolkit.  It takes all the stuff that used to be done on expensive telecom cards' and does it in software on the PC.
16:29.07ManxPowerBoingo: I once ran Asterisk on a low end Pentium.  It could handle 1 call at a time and the codec has to be ulaw.  otherwise there was not enough CPU to keep up
16:29.09theharIs anyone familiar with Queuemetrics?
16:29.41ChainsawBoingo: Virtual machines fall down on one specific thing.
16:29.46ChainsawBoingo: Timing.
16:30.10ManxPowerChainsaw: But applications should not care about timing!  *grin*
16:30.27ChainsawBoingo: When handling voice packets that are so latency-sensitive that even TCP is regarded as overhead, you can not afford the non-deterministic timing that a VM will give you.
16:31.15BoingoSo, essentially, older harderware still trumps my much better (spec wise) VM?
16:31.33ManxPowerBoingo: for the most part yes.
16:31.45BoingoInteresting.
16:32.06ManxPowerJust like older hardware would be better at realtime data capture than the same software running in a VM.
16:32.24BoingoIs there any harm in trying it in a VM?  Or is it a waste of time?
16:32.24ManxPowerwhen you go realtime or pseudo-realtime all the rules change.
16:32.42ManxPowerBoingo: I expect it will work in a VM with low usage.
16:32.50ManxPoweronce you ramp up is another story.
16:32.56BoingoI mean, is it the sort of thing where %100 of everyone thinks it sucks and goes back?
16:33.03*** join/#asterisk rene- (n=renemend@200.34.66.137)
16:33.09BoingoOr is it the sort of thing where it has worked for some people?
16:33.29BoingoIf I ramp up, then I can figure out moving it later.
16:33.39ManxPowerBoingo: and some people win the lottery.
16:33.44[TK]D-FenderBoingo: Has worked for quite a few.  YMMV, don't get your hopes to high and be prepared for great frustration and the possibility you'll never get it wuite right
16:33.50BoingoNot what I ment.  :-)
16:34.18BoingoIt cant be that hard to move it to a different box later.
16:34.24BoingoRight?
16:35.58*** join/#asterisk xpot-mobile (n=james@66-182-83-11.static.slcl007.digis.net)
16:36.14*** join/#asterisk brad_mssw (n=brad@shop.monetra.com)
16:37.49[TK]D-FenderBoingo: Move what?  * configs?  jsut a bunch of text files...
16:38.02BoingoThats what I figured.
16:38.24BoingoInstall it on the other end, copy over the configs (and possibly VMs) and start it up.
16:38.48BoingoChange a few IPs/DNS and should be all good.
16:39.08Joelrecordings, prompts, call records
16:39.12BoingoBut then again, I have never used Asterisk so maybe I don't know what I am talking about.
16:39.20jameswf~nowwhat
16:39.21infobotSo you just installed Asterisk now what? http://www.youtube.com/watch?v=FJ3oHpup-pk
16:43.45*** join/#asterisk rudeboy_xix (n=rudeboy@ded-139-109.eglobalreach.net)
16:44.36*** join/#asterisk hfb (n=hfb@pool-98-112-239-34.lsanca.dsl-w.verizon.net)
16:44.44[TK]D-FenderTALKIN' BOUT FLEA MARKET!
16:45.30*** part/#asterisk rudeboy_xix (n=rudeboy@ded-139-109.eglobalreach.net)
16:46.37russellblol
16:48.20*** join/#asterisk intralanman (n=lanman@173-132-223-176.pools.spcsdns.net)
16:49.32*** join/#asterisk rudeboy_xix (n=rudeboy@ded-139-109.eglobalreach.net)
16:51.03*** join/#asterisk simprix (n=simprix@cosmas.supportdept.com)
16:51.03*** join/#asterisk luckyaba (n=Lucky@ip72-194-215-55.sb.sd.cox.net)
16:51.24*** part/#asterisk amazinzay (n=amazinza@67.108.187.186.ptr.us.xo.net)
16:57.18*** join/#asterisk mltlnx (n=mltlnx@asams.mserve.com)
16:57.40phurlwhat is the best open source database for a call center with many users and replicatoin
16:59.25bmoracaassuming all things are equal (service costs, capabilities, etc), which would be the trunk mechanism you would use:  SIP or PRI?  is the hardware timing source that a PRI gives you very important if we're talking about, say, a 50 phone deployment?
16:59.25*** join/#asterisk p4tr0p1 (n=jam@201-34-141-194.fnsce704.e.brasiltelecom.net.br)
16:59.26*** join/#asterisk DLNoah (n=chatzill@72.11.29.130)
17:00.00*** join/#asterisk Connor (n=billy@75.76.32.44)
17:01.58jameswfbmoraca: I say PRI.. sip is okay but alot of break points
17:02.55bmoracawell, technically, the PRI is delivered via SIP...I'm more curious about just how important the external hardware timing source is
17:03.35*** part/#asterisk Connor (n=billy@75.76.32.44)
17:04.42[TK]D-Fenderbmoraca: easy... PRI
17:05.02[TK]D-Fenderand a PRI is not deliveered over SIP
17:05.15bmoracait's an emulated PRI, so, yes, it does get delivered over SIP
17:05.22bmoracalike a media gateway
17:05.28[TK]D-FenderbmAnd what does "emulated PRI" mean?
17:05.29carrarthen it's not a PRI
17:05.30carrarit's SIP
17:05.45[TK]D-Fenderbmoraca: You can't emulate a clock sync.  You can't get telco monitoring like PRI.
17:06.14[TK]D-Fenderbmoraca: And the SIP is never equal.  In the case of ITSP's it goes over the internet where "random shit" happens.  constantly.
17:06.25carrarHows that Q.931 over SIP
17:06.27carrarheh
17:06.45[TK]D-Fendercarrar: Would be more viable to sugget H.323 for that ;)
17:06.48rene-PRI is very stable, actually it remained the ruling party in my country for well over 70 years
17:06.50carrarhaha
17:07.16carrarPRI Party sounds fun
17:07.28carrarno loopbacks required
17:07.33[TK]D-Fenderrene-: Cell party gained some ground I head, but became diconnected from the populace outside of tower range...
17:07.40[TK]D-Fenderheard*
17:08.03bmoraca[TK]D-Fender: the device is a T1 CSU/DSU with a DSX-1 port that emulates PRI functionality.  the device itself trunks out via SIP.  so calls placed through its PRI terminate via SIP to the provider network (whoever that may be).  it uses the clock source of the inbound T1 (or it can generate its own internal clock source) and uses that as the clock for the PRI.
17:08.15bmoracamy question is not so much about this device, as I know it works.
17:08.42carrarYes lets move on to the BEEF of your question
17:08.42bmoracamy question is about asterisk and whether or not its extremely important to have an external timing source, rather than relying on ztdummy or dahdidummy
17:08.51[TK]D-Fenderbmoraca: Well the sync to the local box may be solid, but the guarantees on the packets out of it ... well thats another matter
17:10.39bmoraca[TK]D-Fender: i realize that, but 9 times out of 10, the SIP provider and the T1 provider are the same company.  my concern is for asterisk...as it would be just as easy for me to simply provide a SIP trunk to asterisk from that same provider...but if I'm going to have issues with conference and playback and what not due to not having an external timing source, then I would prefer to go the...
17:10.41bmoraca...route of the PRI
17:11.48[TK]D-Fenderbmoraca: For conferencing and playback you only are recommended to have a timing SOURCE.  That would be the card itself,k not so much being used as the link to a telco
17:12.01ManxPowerWith a PRI if you have problems you can call the telco and say "The PRI is down" and they might fix it.  If you call up your ISP and say my SIP connection is down, the telco will laugh at you.
17:12.10[TK]D-Fenderbmoraca: Card prividers mixer & timing, not the telco
17:12.37*** join/#asterisk davevg-btwtech (n=davevg-b@75.97.64.33.res-cmts.senj.ptd.net)
17:13.04bmoraca[TK]D-Fender: so the card doesn't just take the timing from the line?  interesting.  I thought it required the link to provide accurate timing.
17:13.29[TK]D-Fenderbmoraca: No, which is why many people bought blank TDM cards for this
17:13.58bmoracahmmm...learn something new every day...
17:16.07ManxPowerbmoraca: there are two types of "timing".  There is timing for a T-1 (which I call "sync source") this "sync" comes from the telco usually.  and timing for meetme and IAX2 trunking, that timing comes from a zaptel device,
17:17.01bmoracaManxPower: I realize that, but I always thought that the zaptel device got its timing from its PSTN link (atomic PSTN clock or T1 sync)
17:17.20ManxPowerbmoraca: the telco sync has no time info in it.
17:17.43bmoracano time info, no, but there is definitely clock sync
17:17.56ManxPowerZaptel cards can receive their sync from the T-1(PRI) or it can provide sync to a device connected to the T-1/PRI.
17:18.13ManxPowerbmoraca: I don't usually call things that don't measure time "clocks".  8-)
17:18.41[TK]D-Fenderbmoraca: That sync is only to align communication with the telco.  The alignment has nothing to do with the interval being constant <-
17:18.44ManxPowerBut everyone else seems to.
17:19.52bmoracainteresting
17:20.02*** join/#asterisk wonderworld (n=w@ip-62-143-16-28.unitymediagroup.de)
17:20.12ManxPowerthere is no "atomic clock" involved in t-1 sync, AFIK
17:20.18bmoracai still think I'll push customers toward the PRI...couple thousand extra on the bill and all that
17:20.34ManxPowerbmoraca: it will save them time in the long run.
17:20.36bmoracaManxPower: not T1 sync, no, but on bare copper analog, there is
17:20.45ariel_most PRI's end up being cheaper here in the states...
17:20.56coppiceManxPower then you don't know much about telephony
17:21.17[TK]D-Fenderrut-roh!
17:21.22ManxPowercoppice: I know much less about the internals of the telco.
17:21.35ManxPowercoppice: so correct me.
17:21.55ManxPowerYou *are* the expert  (no sarcasm intended)
17:22.01ariel_sit's back and watches and takes notes
17:22.14*** join/#asterisk n3hxs (n=HAMming@static-151-196-93-200.balt.east.verizon.net)
17:22.14coppicethe T1 clock is atomic sourced. they need to do this to keep the entire net nicely synced without requiring some vulnerable national master clock source
17:22.42coppiceevery public exchange contains a rhubidium clock
17:23.05ManxPowerI sit corrected.  It still doesn't send "time info", just ticks, right?
17:23.08bmoracaooo...throw some water on that thing and watch sparks fly!
17:23.20bmoracaManxPower: what is time but a series of ticks?
17:23.41coppiceA timing source is also known as a "clock". A "clock"
17:23.42Qwellbmoraca: tocks too
17:23.46*** part/#asterisk awk_r (n=awk@nat/digium/x-ed3c56d7c25d73ba)
17:23.47*** join/#asterisk awk_r (n=awk@nat/digium/x-ed3c56d7c25d73ba)
17:23.59ManxPowerbmoraca: time is a count of ticks.  If you don't count them you don't know the time, do you?
17:24.41ManxPowercoppice: so what should we call zaptel "timing" for things like meetme and iax2 trunking.
17:25.08ManxPowerTo avoid newbie confusion?
17:25.19coppicetime is all relative. times of day are just as relative to midnight as digit clock pulses are relative to each other
17:25.59ManxPowernewbies think they need NTP installed for their T-1 "clock"
17:26.20ManxPowerOK, SOME newbies at least
17:26.36[TK]D-FenderManxPower: Without a timing source it takes newbs 2 hours to watch "60 Minutes" :p
17:26.48jayteeand if you use the better atomic clock in Colorado then your T1 will be accurate to a microsecond :-)
17:27.20coppicemore often people think they need a more accurate clock in absolute terms, rather than one that's just precisely in sync with the far end
17:27.58nkohh[TK]D-Fender!! man! how's it going broheim? i haven't seen you in forever!
17:28.02nkohhwhat are you up to these days?
17:28.30*** join/#asterisk pulpster (n=pulpster@p16.eregie.pub.ro)
17:28.46*** join/#asterisk wonderworld (n=w@ip-62-143-16-28.unitymediagroup.de)
17:30.09[TK]D-Fender~cluebat nkohh
17:30.10infobotACTION pulls out a ClueBat (tm) and thwaps nkohh.
17:31.06nkohhif you keep that up, im going to have to consult with my lawyer, Bob Loblaw
17:31.54nkohhyou may recognize my lawyer from his blog, Bob Loblaw's Law Blog
17:31.57[TK]D-Fendernkohh: I can swing harder and spare you the need for more than a final set of visitors :D
17:32.43*** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir)
17:32.47[TK]D-Fendernkohh: Seriously that random nag script of your is obnoxious, please turn it off in here.
17:33.04nkohhit's not random or nagging
17:33.09nkohhand it most certainly is not a script
17:33.26pulpsterhello
17:33.28[TK]D-Fendernkohh: Fine, "slightly targeted, one shot annoyance the same as the rest" script
17:33.49pulpsteranybody has a configuration of skinny.conf for a 7911G cisco phone or a config that works for that
17:39.35eppigyTRABAJO
17:39.47[TK]D-Fendereppigy: Holiday here :p
17:39.51eppigyYES
17:40.00eppigyI am off thursday
17:40.17*** join/#asterisk javb (n=javb@190.166.136.72)
17:43.17*** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm)
17:43.59*** join/#asterisk admin0 (n=admin0@cm174.delta151.maxonline.com.sg)
17:44.06admin0hi .. is anyone using this ? http://www.voip-info.org/wiki/view/Vovida.org+load+balancer ..
17:47.06*** join/#asterisk Mw3 (i=mw3@mw3.hu)
17:52.00*** join/#asterisk hydra__ (n=hydra@84.91.227.190)
17:52.49BoingoWill having MySQL for CDR on a different computer affect Asterisk to any real degree?
17:53.06[TK]D-FenderBoingo: No.
17:54.14hydra__Hi. I have 2 phones and 2 fixed phone lines. What is the quickest way to create a "load balancer" for those 2 fixed lines?
17:55.01ManxPowerhydra__: call the telco, tell them you want "longest idle hunting" between the two lines.
17:55.06hydra__Example: if someone is using a phone line and I make a simultaneous call, the other line would be used
17:55.18ManxPowerhydra__: um, Asterisk.
17:55.59hydra__ManxPower: I got to have PCI cards for in/out rs232?
17:56.22[TK]D-Fenderhydra__: RS-232?
17:56.24[TK]D-FenderHUH!?
17:56.40bmoracahydra__: staples has 2-line phones for like $99 that don't require any PBX and will allow you to do what you want to do
17:56.43ManxPowerhydra__: Perhaps you should go to digium.com and asterisk.org and learn a little about Asterisk
17:56.59[TK]D-Fenderhydra__: You need a proper FXS/FXO interface to plug all those in.  Ones like the Digium TDM410P, Sangoma A200, etc
17:57.33*** join/#asterisk Victor_Yure (n=victor@unaffiliated/victoryure/x-837844)
17:57.37hydra__bmoraca: how do they work?
17:57.46bmoracayou plug them in...
17:57.58ManxPowerthis is too painful to watch.
17:58.00*** part/#asterisk ManxPower (n=manxpowe@69.73.94.162)
17:58.00hydra__i have to chose which line to use?
17:58.12bmoracaif a line is busy, it shows up as red instead of clear and then you use the other one
17:58.38[TK]D-FenderbmBut taht isn't automatic!  OH NOES!!!!
17:58.44bmoraca~rtfm
17:58.45infobotrtfm is probably Read The F*cking Manual (TM). It is a suggestion to do your homework before posting a question. Sometimes used as RTFM $SPECIFIC_MANUAL to refer to a specific source of information. See also http://en.wikipedia.org/wiki/RTFM
17:59.16bmoracayou know what I always liked?  CyberData VoIP paging speakers and amplifiers has a button on them that's called the "RTFM Button"
17:59.33bmoracathat's litterally how they label it on the device and what they refer to it as in the manual
18:02.18hydra__So, the reasonable way to do this install asterisk on a box with TDM410P?
18:02.27*** join/#asterisk mltlnx (n=mltlnx@mail.rssalesco.com)
18:02.49IBC_jkenneyIf i have a digium PRI card and a Digium FXS card and want to use the FXS card for analog devices can i Push calls thru the PRI interface with the dialplan?
18:03.25IBC_jkenney(I have read i can but the people at digium are making me wonder)
18:04.30IBC_jkenneyI am sure its just a matter of Device A using Context to place call thru DeviceB
18:04.45[TK]D-FenderIBC_jkenney: You can process your calls however you want, its your dialplan
18:05.02hydra__[TK]D-Fender: a box with one TDM410P card + 2 phones with 2 fixed lines is a good combination?
18:05.41IBC_jkenneyI am just checking the digium person i got on the phone stated they didn't know if that would work
18:05.44hydra__(for line balancing)
18:05.45jameswfstill wonders how Digium cards are NOT F.A.C.E. certified....
18:05.49IBC_jkenneyso I am now just double and tripple checking
18:06.08[TK]D-Fenderhydra__: Sure  Actually, Id get the TDM410P with 2 FXO modules, and get a 2-port ATA for the phones.
18:06.24*** join/#asterisk [intra]lanman (n=lanman@freeswitch/developer/intralanman)
18:06.36coppicejameswf: just another faceless company
18:06.43[TK]D-FenderIBC_jkenney: You dial withever you want in your dialplan <-
18:07.08jameswfoh snap... coppice that should be a tag line
18:07.22hydra__what is  2-port ATA?
18:07.25jameswfwe can all be faceless certified
18:08.07coppicewe can all be certified
18:09.29*** join/#asterisk pepe (n=pepesz@77-120.ipact.nl)
18:09.31pulpsteranyone knows what is the Firmware version identifier for a 7911 to place in skinny.conf  ?
18:10.42*** join/#asterisk arthax0r (i=arthax0r@gateway/shell/blinkenshell.org/x-1bcf6bf4e4736918)
18:14.39*** join/#asterisk Dwayne__ (n=Dwayne@64.42.227.97)
18:14.51ariel_pulpster: you mean the version=P002F202 this should be the one you have on the tftp for upgrading or keeping the devices with same version of software.
18:15.01Dwayne__how can i check if my asterisk box has spandsp installed properly
18:15.15pulpsteryes
18:16.17pulpstershould I leave that one ? is this version the same that should I put in OS79XX.TXT file ?
18:16.57ariel_it's the version you have for the device in your tftp for upgrading
18:17.06ariel_or to maintain a stable build
18:17.16pulpsterI don't have one
18:17.33*** join/#asterisk intralanman (n=lanman@173-133-135-140.pools.spcsdns.net)
18:17.39*** join/#asterisk mltlnx (n=mltlnx@mail.rssalesco.com)
18:17.41*** join/#asterisk ccesario (n=ccesario@200.183.211.66)
18:17.46ariel_then your device will not get upgraded or updated.
18:18.22DeeewayneDwayne__, look for the spandsp libs in your lib dir
18:19.02pepeHi, can someone tell me how to implement "callback on busy" (http://www.voip-info.org/wiki/view/Asterisk+tips+Call+Back) into my system? see http://pastebin.com/d1ef94229. Thanks for help:)
18:19.20pulpsterP003-08-4-00 is the version to load into memory of phone (found it in asterisk book) - is this the same for all cisco phones - in particular for 7911 ?
18:20.01ariel_pulpster: each phone has a diff file, if you have a cal from cisco you can go to there site and get the one you need
18:20.49*** join/#asterisk layne (n=layne@c-68-37-72-150.hsd1.nj.comcast.net)
18:26.41*** join/#asterisk rudeboy_xix (n=rudeboy@ded-139-109.eglobalreach.net)
18:30.54*** join/#asterisk davevg (n=davevg-b@75.97.64.33.res-cmts.senj.ptd.net)
18:32.08[TK]D-Fenderpepe: The sample shows you dialplan that allows you to choose to activate the callback
18:32.19[TK]D-Fenderpepe: And your dialplan looks nothing like it.
18:32.50[TK]D-Fenderpepe: Also, all they do is issue an Originate which calls the target 100 times which, if they have call-waiting-like ability will nag the shit out of them
18:33.11*** join/#asterisk lanning (n=lanning@nat/yahoo/x-d373974d215267e5)
18:33.17IBC_jkenneyis there a digium employee in here?
18:33.18Kattygrooves
18:33.26Kattyi'm never gonna dance again!
18:33.40Katty[TK]D-Fender: waste the chance that i'd been given
18:33.52KattyIBC_jkenney: probably.
18:34.03IBC_jkenneyi'd like to speak to one if possible
18:34.11KattyIBC_jkenney: i would recommend calling them.
18:34.16[TK]D-FenderKatty: I liek the new hard-rock version better....
18:34.16IBC_jkenneyi just did
18:34.24KattyIBC_jkenney: call them back.
18:34.33Katty[TK]D-Fender: oh? linky link?
18:34.34[TK]D-FenderIBC_jkenney: About what?
18:34.45IBC_jkenneyIts a really long story
18:34.54Kattyright. so.
18:34.56IBC_jkenneybut in a nutshell information on a product they carry
18:34.57Kattycall them back.
18:35.07Kattyhit the redial button.
18:35.12eppigyKatty: not the way I danced with you
18:35.27[TK]D-FenderKatty: http://www.youtube.com/watch?v=I7imqO-OBVk
18:35.41[TK]D-FenderIBC_jkenney: Just ask your question already
18:35.41Kattylistens
18:35.51Kattyhmm.
18:36.26Kattythat's like listening to another person sing never gonna give you up
18:36.30Kattyit's just not the same
18:36.37[TK]D-FenderKatty: THANK GOD! :p
18:36.56Kattyi prefer the george michael version.
18:37.15IBC_jkenneyits not important i found my answer.
18:37.22IBC_jkenneyi just expected better people over there
18:37.23IBC_jkenneythats all
18:37.55Kattythat's funny.
18:38.05Kattyeveryone i talk to over there is an absolute joy to work with
18:38.11Kattymaybe it's all about your attitude.
18:38.32Kattyeppigy: ATTITUDE
18:38.36eppigyi have it
18:38.39eppigyI HAVE IT
18:38.42Kattyyou are it.
18:38.47eppigyoh true
18:39.11Kattyalso
18:39.17Kattyin a completely unrelated topic.
18:39.31Kattyi need some sort of 'exchange' and 'sql' aware backup software.
18:39.33Kattythat does not suck.
18:39.45Kattywhich is hard, considering exchange is the suck of all sucky
18:39.47eppigynetbackup
18:39.52[TK]D-FenderIBC_jkenney: You're the one being all mysterious about what you actually want and expecting "any random Digium employee" to have whatever answer it is you're looking for.
18:39.59Kattyeppigy: orly
18:40.12pepeH[TK]D-Fender: I did modified the extentions.conf - changing it to my two digit dialplan. http://pastebin.com/d512b7495 However this doesn't work for me. When line is busy it goes to voicemail directly. If this solution is crapy can you suggest better one
18:40.17eppigyor backup exec
18:40.17Kattyeppigy: is that like symantec backup exec?
18:40.21Kattyohiseethenk
18:40.22eppigyif it is not a huge enviroment
18:40.23IBC_jkenneyi am already on the phone
18:40.29Kattyeppigy: it's like ummm 8 servers?
18:40.32eppigyyeah for like a crap load of jobs
18:40.35eppigyo lol
18:40.38eppigythen backuop exec
18:40.42Kattymmmmk
18:40.42eppigywould eb the way to go
18:40.50Kattydoes it do like...non compression stuff
18:41.00Kattywhere you could actually browse backup directories
18:41.13eppigyWell you have catalogues
18:41.16eppigyyou browse
18:41.22[TK]D-Fenderpepe: And after your 2-digit exten YOU dump them into voicemail and choose not to do anything with them.
18:41.22Kattybut you're like...
18:41.23eppigythat list every file and dir
18:41.28Kattynot able to browse the external hard drive
18:41.33Kattyand copy a file, and paste it somewhere
18:41.44Kattyyou must use some sort of restore wizard thingamaboberoonie
18:41.52eppigywell we use a tape library
18:41.56eppigyif you backup to disk
18:41.57Kattyoh gods.
18:41.59eppigyyou may be able to
18:42.04Kattyi have such horrible horrible luck with tapes.
18:42.09Kattyit's like i'm magnetic or something.
18:42.12eppigylol
18:42.20Kattyi'm serious :<
18:42.25eppigyyeha tapes suck
18:42.31*** join/#asterisk Iamnacho (i=Iamnacho@ip174-70-137-120.ks.ks.cox.net)
18:42.33eppigyI do not touch the backups
18:42.36Kattywe've been using external hard drives and REV disks.
18:42.44Kattyand cobian backup
18:42.48eppigyI replicate my mysql db's and rsync my filez
18:42.50Kattyand batch scripts
18:43.11Kattyi will pastebin my asterisk backup.
18:43.12Kattyit is fun.
18:43.17eppigyfunzors
18:43.39*** join/#asterisk wpbrown (n=wpbrown@wh-gtw-0001.woolfharris.com)
18:43.51pepe[TK]D-Fender: so what should be done - the lines #16 & #19 should be removed? I'm just newbie in asterisk world
18:43.56wpbrownhey guys
18:44.37wpbrownI have Asterisk with a Sangoma t1 card feeding it with a PRI.
18:44.49wpbrownI just got this error. Jul  1 13:41:34 pbx kernel: [652407.149878] dahdi: Disabled echo canceller because of tone (rx) on channel 2
18:44.54wpbrownand it dropped the call.
18:44.57wpbrownany pointers?
18:45.00[TK]D-Fenderpepe: You aren't calling a script
18:45.03[TK]D-FenderKatty: http://www.youtube.com/watch?v=j7r7L4eTXAI&feature=PlayList&p=8C8A7C32138FDDB6&playnext=1&playnext_from=PL&index=22
18:45.22[TK]D-FenderKatty: on the topic of other people's songs ;)
18:45.34Kattyeppigy: http://pastebin.ca/1481061
18:45.55Kattyoh god.
18:45.56Kattyfamily guy
18:46.14Katty:>
18:47.23pepe[TK]D-Fender: from where/how should be the script called
18:47.24pepe?
18:47.50[TK]D-Fenderpepe: Apparently you didn't even read the WIKI page's samples which show you the kind of place they would put it.
18:48.15Kattyeppigy: you like?
18:48.40*** join/#asterisk Whitor (n=Whitor@64.128.237.124)
18:49.40*** join/#asterisk aces1up (n=Now@ip70-173-52-152.lv.lv.cox.net)
18:49.42pepe[TK]D-Fender:  I know I'm missing something byt
18:50.11aces1upwould anyone have the sip firmware files for the cisco 7960 phones?
18:50.21pepeI cant see the example, I'm reading http://www.the-asterisk-book.com/unstable/bk01-toc.html and trying to understand how the scripting works
18:50.43[TK]D-Fenderpepe: They GIVE YOU the damn sample that uses the script
18:50.51[TK]D-Fenderpepe: Wake up and read it
18:51.01Kattyaces1up: i only have polcyom stuffs :< sorry
18:51.30aces1upany idea where i can find it? i really don't want to sign up as  a cisco partner
18:51.43*** join/#asterisk The_Boy_Wonder (n=davidvos@asterisk/batman-developer/dvossel)
18:51.47Qwellaces1up: you don't
18:52.18aces1upqwell i don't have to signup to get the firmware?
18:52.30*** join/#asterisk [intra]lanman (n=lanman@freeswitch/developer/intralanman)
18:52.38Kattyhugs [intra]lanman
18:52.43Qwellaces1up: you don't find the firmware.  your only option is to follow their procedures
18:52.54aces1uphrmm.
18:52.55*** join/#asterisk davevg (n=davevg-b@75.97.64.33.res-cmts.senj.ptd.net)
18:53.04Kattyhi Qwell
18:53.08Qwellwaves
18:53.13Kattyhow're you today dear
18:54.26*** join/#asterisk mythicalbox_desk (n=mythical@rrcs-64-183-110-250.west.biz.rr.com)
18:57.47*** join/#asterisk dfaulk3 (n=dfaulk3@nat/digium/x-8895fc6ea63f4ef3)
19:02.54*** part/#asterisk errr (n=errr@fedora/errr)
19:08.13pepe[TK]D-Fender: I DID letter by letter what was on the wiki and still cant see whats wrong.
19:08.14*** join/#asterisk bbryant (n=brett@74.222.102.238)
19:08.25*** join/#asterisk mltlnx (n=mltlnx@mail.rssalesco.com)
19:09.14[TK]D-Fenderpepe: Really?  Where are you calling the AGI in your code?
19:10.44[TK]D-Fenderpepe: http://pastebin.com/d512b7495 <- I see it in this exten called "callback"  How the hell does any call GET THERE?
19:11.03Jumpiehey guys, in an IVR timeout, does that timer start ticking when the announcement start? or does it wait until the audio announcement file is completeD?
19:11.06*** join/#asterisk mags2 (n=mags2@ampulex.whoi.edu)
19:11.08[TK]D-Fenderpepe: Your 2-digit extension do not do anything with regards to starting a call-back
19:15.32*** join/#asterisk jeffgus (n=jeffgus@2002:ad33:b504:0:0:0:0:1)
19:15.51[TK]D-FenderJumpie: After audio completed
19:18.06jameswftzafrir_laptop: do you have a SRPM for SRPM 1.4.12.9.svn.r4590-Xorcom-trunk-r7049
19:19.33*** join/#asterisk apeiron_ (i=apeiron@isuckatdomains.net)
19:20.29*** join/#asterisk mltlnx (n=mltlnx@mail.rssalesco.com)
19:20.48pepe[TK]D-Fender: ok - after exten => _5[0-46-9],1,Dial(SIP/${EXTEN},30,r) should be exten => _5[0-46-9],2,AGI(callback) ?? is that what I'm missing?
19:21.33Jumpiefender, thanks
19:21.38[TK]D-Fenderpepe: You aparently have no grasp on what their sample is doing.  Go read it again a few hundred more times.
19:23.43*** join/#asterisk xpot-mobile (n=james@static-66-182-88-85.bbsc.net)
19:29.11*** join/#asterisk aiksa[LV] (n=aiksa[LV@mx.fiveplus.lv)
19:29.15aiksa[LV]Hi everyone
19:29.54aiksa[LV]where could I find full documenation of AMI interface changes in 1.6 both - commands and events? Please dont tell me taht the only source is the source code :P
19:33.15*** join/#asterisk blkry (n=blkry@64.147.222.130)
19:43.59Katty[TK]D-Fender: i started doing pushups.
19:44.08Katty[TK]D-Fender: i got to 20 before falling on my face.
19:44.15Katty[TK]D-Fender: :<
19:44.33jayteeI like to do at least 10 pushups a year just to maintain my present level of fitness
19:44.43Kattygiggles
19:44.54Kattysadly, i've let myself decline horribly in the last 4 years.
19:46.11Joelbike riding 20 miles every other day is what I do
19:46.12Joelworks wonders
19:46.31Kattysadly, i've no bike.
19:46.37Kattybut i have jogging shoes.
19:46.39Kattyand a yoga mat.
19:46.55jayteeOmmmmmmm
19:46.55Kattymaybe i'll do some sitsup.
19:47.13Kattyjaytee: i don't use it for yoga. just padding on the floor
19:47.15Joelvigerous masturbation was my workout method for a long time, but I apparently eat too much for that to be enough
19:47.30Kattyi hear sex burns a lot of calories.
19:47.36Kattyi've also been lacking in that lately too.
19:47.38Kattysadface
19:48.15jayteeI loved it in Good Will Hunting when Matt Damon lights a cigarette and Robin Williams says, "Those damn things'll kill ya!" and Matt says, "Yeah, they're really getting in the way of my yoga"
19:48.36Kattyhehe
19:49.34*** join/#asterisk defsdoor (n=andy@defsdoor.gotadsl.co.uk)
19:49.43pulpsterI get only dial tone when calling from a cisco to a sip phone using chan_skinny.so - anybody knows what might be wrong - all phones are registering ok
19:50.24pulpsterI dial the number and the tone is still present, during the call (I can transfer no voice payload though)
19:50.39pulpsterany ideas are greatly wellcomed
19:52.11pepe[TK]D-Fender: unless you're willing to help - I'm stuck :(
19:52.42[TK]D-Fenderpepe: If you can't even tell where they are calling it in their sample and which you've got an even bigger problem
19:53.47[TK]D-Fenderpepe: If you can't even tell where they are calling it in their sample and *why* you've got an even bigger problem
19:54.02aiksa[LV]i have found another way to stay fit w/o exercises which bore me
19:54.46aiksa[LV]kitesurfing - a fun way to switch off my brain, and have some really good excersies
19:55.27jayteehmmm, Karl Malden died today @ 97
19:55.50Qwelljaytee: he was almost your age!
19:55.59jayteeLOL
19:56.24Joelkarl who?
19:58.05pepe[TK]D-Fender: PLS, don't judge only because your're experienced...
19:58.24[TK]D-Fenderpepe: Can you read priory steps in an exten?
19:58.39[TK]D-Fenderpepe: Do they call Voicemail if the call doesn't get answered?  No, they do something else.
19:59.04[TK]D-Fenderpepe: This has nothing to dow ith experience.  I READ the code and followed the steps it takes.  You don't seem to notive the steps.
19:59.08[TK]D-Fendernotice*
20:01.00pai have a small problem with asterisk.. I have an HFC ISDN TA, and i configured it in zapata.conf. i then added an extension with my phone number as extension id, and it gets correctly invoked when i receive a call through the ISDN TA. Allright.
20:02.14*** join/#asterisk mltlnx (n=mltlnx@mail.rssalesco.com)
20:02.16panow i tried to add an extension to call regular phones through the ISDN TA . i did something like this (to test): exten => _8XX.,2,Dial(${LINEA}/${MSN1}:${EXTEN:1},60,Tr)
20:02.22pa$LINEA is Zap/g1
20:02.29pa$MSN1 is my phone number
20:03.15pawhen i try to dial 8somenumber via some sip client, i get the phone ringing sound, but in fact it doesnt work
20:03.31pa(im calling some telco number which automatically answer at once)
20:03.41tzafrir_laptopjameswf, http://updates.xorcom.com/astribank/elastix/repo/zaptel-1.4.12.9.svn.4590.xpp.r7049-1.src.rpm
20:03.57paim not sure what im doing wrong
20:05.30[TK]D-Fenderpa : Zap doesn't use MSN's
20:05.54[TK]D-Fenderpa "Za/[channel or group]/numbertocall"
20:06.03QwellZap*
20:06.08[TK]D-Fender^
20:06.38*** part/#asterisk boch (n=fran@200.61.191.9)
20:07.34paah! thanks!
20:08.19pa<PROTECTED>
20:08.59[TK]D-Fenderpa /
20:09.32pfnI forget, cdr's are stored in localtime, right?
20:09.32*** join/#asterisk MindTheGap (n=MindTheG@187.20.141.72)
20:09.33pfnor gmtime?
20:09.38pathanks! i think it's correct : ) now i got answered, but i cant hear anything..
20:09.42pammmh..
20:10.16Kattymmm, walnuts
20:10.19*** join/#asterisk tobias (n=tobias@cpe-071-070-219-040.nc.res.rr.com)
20:10.31pamy MP3Player() extension instead works fine
20:10.34pa(i can hear)
20:11.10paprobably i screwed after my dial command..
20:11.46WindowsUserodd
20:11.58WindowsUsermy wakeup calls aren't generating cdr's
20:12.31WindowsUserpfn: my Master.csv is in gmt
20:12.56pathis is my extension: http://pastebin.com/m5e84e50e
20:13.08pfnWindowsUser, thanks
20:13.32pai think it was working back in the years when i was using some isdn4linux driver
20:13.49pa(instead of zaptel)
20:13.49pfn<PROTECTED>
20:13.49pfn<PROTECTED>
20:13.49pfnhmm, sounds like localtime
20:15.53*** join/#asterisk dwschool (n=Dwayne@64.42.227.97)
20:16.15dwschooltextfmt: No font metric information found for "Courier-Bold". i'm getting this error when trying to send test fax
20:18.05dwschoolwith hylafax or should i ask elsewhere
20:21.12*** join/#asterisk [intra]lanman (n=lanman@freeswitch/developer/intralanman)
20:23.42[TK]D-Fenderdwschool: What is generating that error?
20:23.47aiksa[LV]asterisk manager event *AgentCalled* gets raised when an Agent is called or queue member as well?
20:24.03aiksa[LV]what exactly is a scope of the Agent here?
20:24.55WindowsUserdwschool: is that an error or a warning?
20:25.36aiksa[LV]dwschool: I guess this error comes from a ghostscript
20:25.55aiksa[LV]dwschool how are you trying to send the fax?
20:27.09aiksa[LV]if you submited it to hylafax as a ps file, it shouldnt have those problems
20:27.21*** join/#asterisk andres833 (n=andres83@190.144.75.22)
20:28.17aiksa[LV]however if you are trying to convert it to image from some format which only identifies fonts w/o including them, that would be the prblm.
20:30.20*** join/#asterisk illizit (n=cengroba@c-76-109-84-129.hsd1.fl.comcast.net)
20:30.59*** join/#asterisk andres833 (n=andres83@190.144.75.22)
20:32.28*** join/#asterisk grEvenX (n=even@cB78D5AC1.dhcp.bluecom.no)
20:35.07*** join/#asterisk pcmedic (n=pcmedic@213.63.137.210)
20:35.22*** join/#asterisk callenb (n=callenb@host86-168-62-181.range86-168.btcentralplus.com)
20:35.28*** join/#asterisk aces1up (n=hey@76.3.153.228)
20:37.10aces1upi have a cisco 7960 connected to my asterisk box, and it says proxy unavailable but i have the right ip in the .cnf files that the cisco phone loads..
20:37.29aces1upis there a way i can check to make sure asterisk is running correctly as in receiving SIP requests?
20:38.12pcmedicls
20:39.02*** join/#asterisk awk_r (n=awk@nat/digium/x-092d80a6f9163896)
20:41.26bmoracaaces1up: enable sip debugging
20:44.05dwschoolaiksa yes i am trying to send a fax
20:44.36dwschool[TK]D-Fender: I am trying to send a fax
20:44.47dwschoolWindowsUser: its an error
20:45.54dwschoolaiksa[LV]: sendfax -n -d <faxnumber>  <file.txt>
20:50.44*** join/#asterisk aces1up (n=hey@76.3.153.228)
20:50.59aces1updumb question, but where are the asterisk config files typically located?
20:51.37dwschoolwhat are u running?
20:52.55dwschool/etc/asterisk
20:53.36[TK]D-Fenderdwschool: Yeah, that part was very clear.  And I asked you WHAT piece of software spat out that message
20:54.43aces1upis there a way to see what extensions are currently connected to asterisk?
20:54.53dwschoolhylafax
20:55.06dwschoolaceslup: you mean sip?
20:56.00aces1upi guess yeh, so if i have an extension configured as 100 how can i tell if the cisco phone is connecting to my box on that extension.
20:56.27aiksa[LV]dwschool: ok, got it
20:56.34awk_races1up: what technology are you using to connect the phone to Asterisk? if its sip then do "sip show peers"
20:56.43aces1upsip sorry
20:56.43aiksa[LV]dwschool: install ghostscript font metrics package
20:56.46aces1upok will do.
20:57.27aces1upok so i got this for my extension
20:57.29aces1up100                        (Unspecified)    D   N      0        UNKNOWN
20:57.29aces1up1 sip peers [Monitored: 0 online, 1 offline Unmonitored: 0 online, 0 offline]
20:57.36awk_races1up: also, to help "extensions" is usually a reserved word meaning a dialplan extension, a "peer" is what you were trying to refer to :-)
20:57.37aces1upso 0 are online.
20:57.52aces1upwhere do i start troubleshooting to get it to go online?
20:57.57awk_races1up: also, don't mass paste, use something like pastebin
20:57.58awk_r~pastebin
20:57.59infobot[~pastebin] 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
20:58.02aiksa[LV]dwschool: the point is hylafax tries to convert txt document into something sendable. for that it needs fonts to represent chars in your text file
20:58.04aces1upawk ok sorry.
20:58.11awk_races1up: np :-)
20:58.15aces1upit was only 2 lines so didn't think it was a prob.
20:58.20aiksa[LV]dwschool: what linux distro are you using?
20:58.32dwschoolaiksa[LV]: i know, what package would it be for debian?
20:58.50aces1upawk where can i start troubleshooting as to why my phone isn't online?  i am also using freepbx BTW.
20:59.31aiksa[LV]lemme consult google
20:59.31dwschoolaiksa[LV]: debian
20:59.44dwschoolaces1up: what phone make?
20:59.46awk_races1up: turn on sip debug ("sip set debug on") and see if your phone's registration is actually making it to Asterisk. If so, sift through and see if it is getting rejected
20:59.49aces1upcisco 7960
20:59.52aiksa[LV]ghostscript-fonts perhaps
21:00.14aiksa[LV]i dont have any debian family linuxes at my disposal at the moment so cant check this
21:00.28awk_races1up: change that...first uninstall FreePBX and install Asterisk-GUI (lol kidding...<for those that know me>)
21:01.50aiksa[LV]dwschool: where you able to locate it?
21:02.06*** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net)
21:02.12dwschoolno - i know ghostscipt is installed
21:02.20dwschoolpackage is gs
21:02.36aces1uphrmm ok so im getting a buncha sip messages most of em say sip/2.0 401 unauthorized
21:02.41aces1upas the header
21:02.49aces1upthat can't be good i suppose.
21:03.07pepe[TK]D-Fender: exten => _5[0-46-9],1,Dial(SIP/${EXTEN},30,r) and exten => _5[0-46-9],2,Goto(s-${DIALSTATUS},1)   ??  where DIALSTATUS will be BUSY when callee is busy
21:03.52aiksa[LV]hmm, the just follow this guide
21:04.00aces1upwhen i register a 7960 and editing the sip<mac>.cnf file is ther a authorization line i need to put in there?  is this the correct place to be putting it?
21:04.02aiksa[LV]http://linux.about.com/library/howto/font/blfont5.htm
21:04.18*** join/#asterisk mltlnx (n=mltlnx@mail.rssalesco.com)
21:04.49aiksa[LV]perhaps you will need to move/copy some of the existing font files and convert them or just change a path to exact location
21:06.17aiksa[LV]dwschool: try this one: gsfonts
21:06.45dwschoolalready installed
21:07.31aiksa[LV]strange. never the less the error states that gs is unable to locate these files. could be path issues
21:07.35awk_races1up: usually Asterisk sends 1 unauthorized...if it sends it twice in a conversation its bad yes. I'm not familiar with Cisco phones, so someone else will have to chime in
21:08.43*** join/#asterisk xpot-mobile (n=james@static-66-182-88-85.bbsc.net)
21:09.06aiksa[LV]dwschool: enter #gs --help and pastebin the output
21:10.29aiksa[LV]in that output you should see a folder in which gs is looking for fonts
21:10.33dwschoolhttp://pastebin.com/m4d164aee
21:10.58dwschooli see it and its there when i browse for it
21:12.04dwschoolaces1up: is this the first cisco phone u are setting up?
21:12.09aiksa[LV]you have a number of pfb and afm files there?
21:12.17aiksa[LV]in one of those folders i mean?
21:12.42aces1updwschool yes
21:13.11dwschoolis the [device] and username in sip.conf the same?
21:13.52dwschoolaiksa[LV]: yes tons
21:14.21rudeboy_xixwhen using the AMI of asterisk, and then i execute a command like 'Originate' then the other party answered, can i use extenspy on it? is it possible?
21:14.22aces1updwschoool, well i am using freepbx to configure the extensions.
21:14.31aces1upand i don't see in what sip.conf file it is defined.,
21:15.21*** join/#asterisk gr0mit (n=tim@2001:8b0:3d9:0:b01e:f686:4124:ac1d)
21:15.26dwschoolaces1up: there are three file containing sip in /etc/asterisk open them and check. I always have to go back and redit the sip.conf
21:15.28aces1upis there a way in asterisk cli to see what sip extensions are registered but not connected?
21:15.40dwschoolsip show peers
21:16.04dwschoolif address is unspecified, its not registered
21:16.11aces1upso when it show extension 100 in that list...  that means it is configured somewhere is some .conf file/
21:16.19dwschoolyes
21:16.33dwschooldoes address say unspecified?
21:16.43aces1upyes
21:16.46rudeboy_xixcan anyone answer me?
21:17.05aces1upi dunno i have the secret and everything matching in the sip<mac>.cnf file as well.
21:17.30dwschoolok, go to sli - set verbose to 10 and see if your phone is trying to auth
21:17.48dwschoolcli i mean
21:18.05Katty[TK]D-Fender: :<
21:18.12Katty[TK]D-Fender: my sister was considering taking lipozene.
21:18.22Katty[TK]D-Fender: my family is a bunch of dingbats.
21:18.45Katty[TK]D-Fender: i told her to take the money she was going to spend on lipozene, and go see a doctor.
21:18.58aces1updwschool it is.
21:19.15dwschoolfind it in sip.conf
21:19.17aiksa[LV]dwschool: - then sorry
21:19.41aiksa[LV]I dont have any idea why it would not find font metrics file
21:20.22*** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net)
21:20.27aiksa[LV]thats it - I am leaving now. it is over midnight here already
21:20.36aiksa[LV]dwschool: good luck!
21:20.44dwschoolthanks
21:21.44aiksa[LV]dwschool: just last thing
21:21.50aiksa[LV]in those dirs
21:21.56aiksa[LV]there should be two additional files
21:21.58Kattyfile: OH EM GEE
21:22.12Kattyfile: so pretty :>
21:22.25aiksa[LV]fonts.dir && fonts.scale
21:22.28Kattyfile: will there be cats, or dogs?
21:22.41aiksa[LV]if i am not mistaken fonts.dir acts as a roadmap
21:22.55aiksa[LV]links specific font name to a pfb file
21:23.02aiksa[LV]check if you have that file
21:23.26aiksa[LV]and if it lists the Courier-Bol in there
21:23.44[TK]D-FenderKatty: Agree... Wingbats......
21:23.50dwschooldont have both
21:24.04dwschoolaiksa[LV]: don't have both
21:24.49Katty[TK]D-Fender: you see file's new house?
21:24.59[TK]D-FenderKatty: Nope....
21:25.06Katty[TK]D-Fender: you still on facebook?
21:25.40[TK]D-FenderKatty: Extrememly rarely, and I remove almost all material about myself.... Facebook = EVIL
21:25.50*** join/#asterisk pha3drs (i=nobody@static-96-254-70-2.tampfl.fios.verizon.net)
21:26.01[TK]D-Fenderkicks Novation's MIDI implementation in the ndas
21:26.06[TK]D-Fendernads*
21:26.16aiksa[LV]dwschool: cd into that directory
21:26.42[TK]D-Fenderkicks Roland's MIDI implementation in the nads for not letting him cheat with SysEx either
21:26.46pulpsterhello
21:26.49aiksa[LV]and perform #mkfontdir
21:26.50pulpsterplease help
21:26.55pulpstermy problem is here:  http://pastebin.com/m268ac740
21:27.09Katty[TK]D-Fender: well you should go have a look anyway.
21:27.14pulpsterI have tried anything can't seem to figure out what I do wrong
21:27.17aiksa[LV]dwschool: documentation says it should rebuild the fonts.dir and fonts.scale
21:27.49dwschoolok,ok
21:28.14aiksa[LV]I`ll hang in to see if you suceed
21:28.15pha3drsHello...  I am using Trixbox v2.6.1.10 - I was wondering what controls my ability to call Canada, as everytime I try to call any number in CA I get a busy....
21:28.15aiksa[LV]:)
21:28.45aiksa[LV]pha3drs: your telco controlls your ability to call Canada
21:29.09*** join/#asterisk intralanman (n=lanman@70-10-178-143.pools.spcsdns.net)
21:29.19Kattyhugs intralanman
21:29.24dwschoolok, i fixed the path in hyla.conf was incorrect
21:29.24aiksa[LV]dwschool: did it work?
21:29.29pha3drsright, I called Voicepulse and they didn't see anything in the logs when I dialed
21:29.41dwschoolbut u pointed me in the right direction - thanks
21:29.55aiksa[LV]dwschool: take care
21:30.01dwschool:D
21:30.09aiksa[LV]missing font packs is a traditional issue with slackware
21:30.31aiksa[LV]they doesnt have this as dependancy for GS, so everytime I setup another hylafax - oh the fun ...
21:30.41pha3drsCalling any number in the US works, but not CA and voicepulse said that it was on my end, is that not the case?
21:30.53aiksa[LV]pha3drs: check your logs
21:30.59pha3drsok
21:31.03aiksa[LV]see how call gets procesed
21:31.09aiksa[LV]and if it even leaves your server
21:31.15aiksa[LV]thats it for now
21:31.18aiksa[LV]end of show
21:32.00*** join/#asterisk [acer]lanman (n=lanman@70-10-178-143.pools.spcsdns.net)
21:32.42pulpsteranybody configured cisco phones + asterisk - need someone with experience to ask her/him some questions
21:32.45pulpsterplease help
21:33.05pulpsterI will soon have no hair left on my head
21:33.20pha3drshere is the log: http://pastesite.com/8449
21:33.21dwschoolpulpster - whats the problem?
21:33.26pulpsterhttp://pastebin.com/m268ac740
21:33.32dwschooli have quite a few - what model?
21:33.36pulpster7911
21:33.39pulpster7911G
21:33.59pulpsterI have tryed both chan_sccp.so and chan_skinny.so
21:34.07pulpsterI cannot send voice payload
21:34.13pulpsterthis is wher I am stuck
21:34.15dwschoolwell i iuse sip
21:34.32pulpsteryou've changed the firmware of your phone to sip ?
21:34.56pulpsterI cannot change anything in my phone..
21:35.05*** join/#asterisk errr (n=errr@fedora/errr)
21:35.08pulpsterI'll just have to use it as it is I guess
21:35.29pulpster...or tear it appart..eventually
21:36.05*** join/#asterisk mltlnx (n=mltlnx@mail.rssalesco.com)
21:36.40dwschooli've decided to switch to mitel phones - i like them way better
21:37.08pulpsterI will switch to any other vendor next after I finish this job configuring the cisco
21:38.20pulpsterdo you guys have any ideas, what might cause this behavoiur ?
21:39.24pulpsteram I stupid and do smtg obviously wrong ? or is it because skinny.so / sccp.so are not well suited to my 7911 phone ?
21:39.40*** join/#asterisk [intra]lanman (n=lanman@freeswitch/developer/intralanman)
21:41.46*** join/#asterisk ttl- (n=patrick@d5153AE82.access.telenet.be)
21:42.12Kattyeppigy: hungry :<
21:44.00pulpsterdo you know anyone who might solve my problem ?
21:44.02*** join/#asterisk watchy (n=dolphinr@76.196.98.139)
21:44.07watchyhi
21:44.12*** join/#asterisk ElSonico (n=tav@dondo.ampiainen.net)
21:44.27pulpsteran email / site or a developer from sccp who might know these stuffs ?
21:44.57watchywhats a easy way to send a cell SMS messages from asterisk?
21:45.42WindowsUserwatchy: via a phone over bluetooth or via an email gateway?
21:48.31watchywell a customer wants a text sent to his cell everytime he gets a new voicemail on his landline
21:49.01dwschooli am getting hylafax error no local dialtone
21:49.31IBC_jkenneywatcy thats easy
21:49.36IBC_jkenneywatchy thats easy
21:49.48IBC_jkenneywhy send sms get his cell phone e-mail address
21:50.03IBC_jkenneymost cell providers let you send number@carrier. and it goes right to the phone as sms
21:50.06IBC_jkenneyi use it all the time
21:50.49IBC_jkenneyyou can do it in voicemail.conf add pager=emailaddress
21:50.54IBC_jkenneyto the line and you should be all set
21:51.23watchyoh
21:51.25watchygood call
21:52.04IBC_jkenneywant an example?
21:52.21watchynah i just never thought about that now that you brought it up
21:52.31watchythats a very easy way without actually sending an SMS
21:52.54IBC_jkenneyYeah i fiddled with it for myself
21:53.01watchydid it work well?
21:53.38IBC_jkenneySure did if you have a smart phone with windows on it you can also attach the VM as a wave and send it to the phone
21:54.50watchywow
21:55.48*** join/#asterisk Dr1 (n=AXON@pool-71-105-96-98.lsanca.dsl-w.verizon.net)
21:56.07Dr1are there any free ITSPs?
21:56.30watchyi guess it wont work with a iphone?
21:56.43IBC_jkenneyyeah it will work with a iphone
21:56.50IBC_jkenneyas long as you can open a wav file
21:57.02IBC_jkenneymake sure to specify in voicemail.conf wav not wav49
21:57.11*** join/#asterisk DrAk0 (n=luisjose@nelug/coreteam/luisjose)
21:57.13watchywell you can't send MMS to a iphone
21:57.34IBC_jkenneybut you can send e-mails to the I phone
21:58.39watchythey arent allowed to use email
21:58.47watchythats why they need it to be an SMS
22:01.31WindowsUserwatchy: cellphone companies usually have a email to txt gateway
22:01.44IBC_jkenneythat was what i said
22:01.46watchyyea, thats what i m gonna have to use
22:01.54watchybut i cant attach a wav to that
22:02.09IBC_jkenneythe pager= will only send the text of the email
22:03.32*** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage)
22:03.32*** mode/#asterisk [+o lmadsen] by ChanServ
22:03.49watchyyea thats what i need
22:04.34*** join/#asterisk qdk (n=qdk@0x573d8dd1.bynqu1.dynamic.dsl.tele.dk)
22:12.11*** join/#asterisk mltlnx (n=mltlnx@mail.rssalesco.com)
22:19.40*** part/#asterisk callenb (n=callenb@host86-168-62-181.range86-168.btcentralplus.com)
22:22.48*** join/#asterisk Shazaum (n=oidjiowe@unaffiliated/shazaum)
22:23.02Shazaumany doc srtp in asterisk 1.4
22:23.02Shazaum?
22:24.17leifmadsenShazaum: just the issue open in the issues tracker
22:25.44*** part/#asterisk squish102 (n=squish10@cpe-075-181-098-059.carolina.res.rr.com)
22:25.54Shazaumleifmadsen, so
22:26.06Shazaumleifmadsen, srtp only in asterisk 1.6? right?
22:26.20leifmadsenShazaum: yes, 1.6.3, or whenever it gets merged
22:26.27leifmadsenit has not been merged -- see the issue
22:28.26Shazaumleifmadsen, ok, tks 4ur attention
22:29.04*** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee)
22:33.30*** join/#asterisk Toerkeium (i=Toerkeiu@201.216.206.221)
22:38.09*** join/#asterisk mltlnx (n=mltlnx@mail.rssalesco.com)
22:46.57*** join/#asterisk jedaustin (n=chatzill@austin-j.its.dist.maricopa.edu)
22:57.44*** join/#asterisk fakhir_ (n=fakhir@unaffiliated/fakhir)
23:01.09*** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm)
23:01.35*** join/#asterisk galeras (n=galeras@186.80.186.118)
23:03.14galeraslittle help: why call is not starting after mv sendfax.call /var/spool/asterisk/outgoing/
23:05.09drmessanogaleras: I just looked in /var/spool/asterisk/outgoing/ and it's empty
23:06.09*** join/#asterisk securevoip (n=securevo@c-76-123-20-170.hsd1.va.comcast.net)
23:06.28galerasdrmessano: mine is not empty
23:07.36drmessanogaleras: I wouldn't know.. you've shown us NOTHING yet
23:09.38galerasnever mind, i missed autoload=yes in modules.conf
23:13.39*** join/#asterisk lasko (n=verilan@66.178.162.34)
23:14.09laskoCould any tell me why I would be getting "SIP response 488 - Not Acceptable Here" when I'm making a call?
23:16.31*** join/#asterisk propellerhead (n=yogurt2u@host176.190-136-58.telecom.net.ar)
23:17.30*** join/#asterisk seanbright (n=sean@asterisk/contributor-and-bug-marshal/seanbright)
23:17.39jayteelasko, is it all calls or just some?
23:20.11[TK]D-Fenderlask488 = codec mismatch
23:20.17*** join/#asterisk Octothorpe (n=Octothor@pdpc/supporter/professional/octothorpe)
23:20.28[TK]D-FenderDarn, missed the exit
23:22.26jayteesame here
23:40.59Dr1Is there a free ITSP that you can use to test your asterisk VoIP connection?
23:43.18*** join/#asterisk mltlnx (n=mltlnx@asams.mserve.com)
23:44.27securevoiphttp://www.sipgate.com/
23:44.58securevoipOR http://www.ipcomms.net/html/freedid.html
23:45.14*** join/#asterisk SaiSoma|AFK (n=SaiSoma@74.167.136.30)
23:45.33*** join/#asterisk mltlnx (n=mltlnx@asams.mserve.com)
23:48.30*** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm)
23:51.04[TK]D-FenderDr1: www.ekiga.net
23:53.24*** join/#asterisk mltlnx (n=mltlnx@asams.mserve.com)
23:57.21Dr1If I'm reading this right, it looks like ipcomms is free dial-in only.  Do SIPGate and ekiga both do free inbound and outbound?  What are the limitations (cap on minutes per day, etc)?  The Quality?

Generated by irclog2html.pl Modified by Tim Riker to work with infobot.