IRC log for #asterisk on 20080501

00:04.36*** join/#asterisk mindCrime (n=chatzill@cpe-065-190-188-124.nc.res.rr.com)
00:08.38florzYourname``: what do you mean by "has echo"?
00:09.27Yourname``florz: I mean calls at location A have don't complain of echo.
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00:09.49florzYourname``: how could a call complain?
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00:11.26Yourname``florz: I suggest you open a channel, "#ESL-teacher-for-hire".  If you can't help because someone can't ask a question in accordance with the Book of Webster, then you're clearly not in the right place.
00:12.56florzYourname``: I suggest you rather consider providing as precise information as possible if you are expecting any for-free remote diagnosis that's supposed to help you.
00:13.36Yourname``florz: I've gotten help numerous times by people who don't have a pickle up their ass. So, I can wait.
00:13.57florzYourname``: Sure, have fun ;-)
00:14.04Yourname``Thank you, come again.
00:16.29mwallingladies, please
00:23.56drmessanoloves when people get pissy when begging for free help
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00:24.56Yourname``Wow, amazing day today, huh?
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00:28.56nvrpunkzomgoblinz here, how do I check to see if the peer connection exists between boxA and boxB
00:29.02nvrpunkif it's not a global register
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00:38.35nvrpunkwhat does register_verify: No registration for peer 'teleport'
00:38.36nvrpunkmean
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00:42.28nvrpunkanyone here still?
00:44.18*** join/#asterisk zomgoblinz (n=zomgobli@81.90.21.227)
00:44.28zomgoblinzanyone alive?
00:45.21eXcAliBuReveryone is dead
00:46.33zomgoblinzi keep getting registration rejected
00:46.42zomgoblinzbetween my two asterisk boxes
00:47.38NovceGuruDamn it, i'm getting silly one way audio issues where the reciever of the call can't hear me, and the only thing i've changed is putting my sip clients behind a 1:1 nat </vague_description>
00:48.52drmessano~sipnat
00:48.52jbot[~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions
00:50.06NovceGuru#5 :( The phones till have an internal address
00:50.40drmessanoand?
00:51.28NovceGuruI might not understand how 1:1 nat works, but both the asterisk server is behind nat along with the clients
00:51.46drmessanoWhere are "you" ?
00:52.34NovceGuruThe server is on 192.168.1.x, ipsec vpn connects to a remote network where the sip clients are, on 192.168.2.x
00:53.30NovceGuruit worked, until I did a 1:1 nat with an external ip to the internal ip of the phones (192.168.2.x)
00:54.01NovceGuruI did the 1:1 nat so a bsd based firewall can be used, and I can connect more then one sip client to a wan based sip provider
00:54.29drmessanoIsnt that what asterisk is for?
00:55.30NovceGurulatency issues are making me directly connect to the wan provider, and still using my asterisk box for internal paging on line 2 (cisco 7940s)
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00:56.16NovceGuruI should say latency/multiple POF's
00:57.13drmessanoSounds to me like you got a lot of things screwed up
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00:57.29NovceGuruit's very possible!
00:58.19NovceGuruthe latency issue was that the call was going across the country, then out to the trunk provider, then back through the vpn
00:59.15NovceGuruI'll have to draw up some charts to better explain
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01:02.40edgedoes any body have any experece setting up SIP on a cisco 7961?
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01:06.20zomgoblinzok, http://pastebin.com/m76703313 these are my configs and I can't get box b to register with box a
01:06.27zomgoblinzcan anyone point me in the right direction?
01:11.08edgefor some reason i have the cnf.xml file with the version i want my phone to pickup , and the tftp exchange happens , i just don't know why it wont take the sip image
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01:12.48[hC]anyone here configured an spa-3102?
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01:19.51[hC]anyone with an spa-3102 had a problem where calls coming in from the PSTN line ring the designation extension over and over and over instead of just once?
01:20.12jeffspeff~softphone
01:20.12jbot[~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
01:20.20jeffspeff~ekiga
01:20.20jbot[~ekiga] Ekiga is an OSS SIP soft-phone for Windows, MacOSX, and Linux (Requires GTK+ which is included with the binary releases) that can be found at http://www.ekiga.org
01:24.27*** part/#asterisk chronos_ (n=a23@adsl-99-135-95-53.dsl.emhril.sbcglobal.net)
01:24.58cmantitoso what has to be enabled for automon to work right?
01:26.50eXcAliBuRis a p3-550MHZ with 256MB ram 40GB HD good enough for asterisk ?
01:27.53eXcAliBuRlike as a testing system
01:28.12jblackyeah
01:29.33eXcAliBuR:)
01:29.34eXcAliBuRthanks
01:31.26jeffspeffI tried using a 400mhz processor 20gb hdd and 300 something mb's of ram, and all the audio was extremely choppy
01:31.39jblackIn my experience, ekiga can cause a program with gnet, which in turn leaves ekiga permanantly unstartable.
01:31.46jblacks/program/problem
01:32.11jeffspeffreally?
01:32.26jblacknodes
01:32.30jblacknods, even.
01:32.53eXcAliBuRthanks for the heads up jeff
01:32.59jeffspeffI'm trying to find a softphone similar to xlite, but has transfer features, and things like that.  any suggestions?
01:33.09jeffspeffeXcAliBuR: no problem
01:33.30jeffspeffjblack: any suggestions?
01:33.31jblacktries to remember if twinkle works
01:33.54jeffspeff~twinkle
01:33.54jbot[~twinkle] Twinkle is an OSS SIP phone for KDE/Qt systems and can be downloaded at http://www.twinklephone.com
01:34.15jeffspeffjblack, is twinkle available on windows?
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01:34.43jblackNo idea. I don't care about windows.
01:35.37Yourname``Can I use fxotune, and zapata.conf on a asterisk box that connects to providers via SIP? And has no hardware like TDM, FXS, etc?
01:35.48Yourname``...to try and figure out echo problems?
01:36.27JTno
01:40.37Yourname``Thanks, JT.
01:41.00Yourname``JT: What do you think I could use in such a scenario to be able to diagnose and fix echo?
01:41.09drmessanoHoly crap
01:41.11JTecho from where?
01:41.15drmessanoI just installed apache on Windows
01:41.22*** kick/#asterisk [drmessano!i=north@pdpc/sponsor/digium/Qwell] by Qwell (Qwell)
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01:41.31drmessanoApache >++++ IIS
01:41.47drmessanolol
01:41.59JTactually i think iis performs beter than apache on windows
01:42.01JTlast i heard
01:42.13drmessanoNot in my experience so far
01:42.22Yourname``JT: Well, I use an asterisk box where queues are hosted and agents who rcv calls complain of echo every now and then.
01:42.24drmessanoSeems to be blowing IIS away
01:42.31Yourname``JT: The calls delivered to this box is via sip.
01:42.38JTYourname``: calls are going from what to what?
01:42.39JTok
01:42.43JTsip, from where to where
01:42.44QwellJT: that test was done with Windows on an extremely beefy server..
01:42.49jblacksadly, iis has been catching up
01:42.56Qwelllook it up :)  some IBM mainframe iirc
01:42.57JTQwell: which test?
01:43.03drmessanoIIS still cant handle PHP
01:43.10Qwellperhaps the one you're referring to
01:43.11drmessanoIts PHP support SUXORS
01:43.12Yourname``JT: Provider -> Asterisk -> Queues on another Asterisk
01:43.18JTiis has the advantage of being able to take advantage of poorly documented windows tricks
01:43.33JTYourname``: that can't be the whole scenario
01:43.34drmessanoIIS + PHP = poo
01:43.36JTi see no phones there
01:43.43Qwelllike not having to request priv escalation? ;)
01:43.56drmessanoLOL
01:43.59drmessanoZOMGADMIN
01:44.08Qwell"ohhai, I'm admin, I can haz resources?"
01:44.22drmessanoThis was funny
01:44.25drmessanoI installed apache
01:44.31drmessanoTried to hit a site
01:44.34drmessanodenied
01:44.50Yourname``JT: Alright, call is made from Asterisk 1. Then, when there is a human on the other end, the call is bridged from Asterisk 1 to Asterisk 2 via sip where there is a queue and agents are logged in via eyeBeam. This call that one of the agents answers hears echo in the call.
01:44.51drmessanoAllowed read access for the apache user to that foler
01:44.53drmessanofolder
01:45.01drmessanoIt worked
01:45.06drmessanoZOMGNOTWHOLEFILESYSTEM????
01:45.34eXcAliBuRlol
01:45.40JTYourname``: one end if eyebeam, what is the other end?
01:45.51jblackcan't tell if drmessano is trying to be funny
01:45.53drmessanoIm shocked http://sitename\\\\\\\c:\windows\command\command.com doesnt work anymore
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01:46.00Yourname``JT: A PSTN phone.
01:46.12JTYourname``: pots/pri...?
01:46.25JTYourname``: how is the pstn terminated?
01:46.39Yourname``JT: No idea. Could be anything. If I call your cellphone or your home phone, how could I tell if its PSTN/POTS?
01:46.53drmessanojblack: I give IIS a path.. say c:\windows, and without editing FS permissions, I now have a virtual directory mapped there
01:47.04JTYourname``: do customers hear agents echoing, or just agents hearing some customers echoing?
01:47.19JTYourname``: i can usually tell my the phone number and voice quality
01:47.23Yourname``JT: The latter.
01:47.31JTand i also call known lines to test
01:47.43JTYourname``: ok. how are calls terminated to the PSTN?
01:48.05Yourname``ah, but in our scenario it's a little different because we call current customer phone numbers to cross-promote a feature/product.
01:48.25JTerr i don't see what that has to do with echo to be honest
01:48.26Yourname``JT: VoIP provider.
01:48.32JTYourname``: get a new VoIP provider
01:48.40drmessanoVoIP doesnt echo
01:48.44drmessanoIf it does
01:48.47drmessanoRefer to #1
01:48.49JTyour VoIP provider's equipment is 90% likely to be at fault
01:48.56Yourname``There's 3/4 providers in the mix..
01:48.57edgecan somebody help me fix my Cisco ip phone, it wont take its new firmware
01:49.04Yourname``It's probably not even a VoIP provider.
01:49.06Qwelledge: they do that
01:49.09drmessanoedge: smash it with a hammer
01:49.12JTYourname``: their equipment has insufficient echo cancellation
01:49.19edgedrmessano, helps me 0
01:49.24Yourname``Level3, PacWest and XO along with a couple aggregator providers.
01:49.34JTYourname``: complain to them
01:49.41Qwelledge: there are some decent howtos on the wiki
01:49.43Qwell~wikis
01:49.43jbot[~wikis] VoIP Wiki covering Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, CallWeaver, and YATE.  http://www.voip-info.org (c) Arte Marketing Inc / CommPartners
01:49.44mwallinghmmm... is there a "please wait while your call is being transfered" in the sample sounds?
01:50.12edgeqwell ,i've followed it im kinda at a dead end with it takign the config but not requesting the firmware
01:50.25Qwelledge: yeah...it does that.  what version is it on now?
01:50.45Yourname``JT: That's a good idea, because I wasn't sure if it's asterisk. Because here's another issue. The queues on Asterisk queue are connected to by agents in different locations. One location is in Indonesia, and the agents there do not complain of echo. Then, there is another location in California and those agents complain of echo.
01:50.47edgeQwell, SCCP41.8-2-1es5s
01:50.54drmessano#1 VoIP doesn't echo
01:51.02Qwellswitching to SIP should be trivial
01:51.05edgeQwell, i got that from that status page ,and im sure its not what i want, bc i want sip
01:51.09Qwellare you using the SEP<mac>.cnf.xml?
01:51.21drmessano#2 If VoIP echos, goto #1
01:51.28edgeQwell, yes, it loads it, the TFTP server says it requested it and takes it, just takes no action on it
01:51.46Qwellverbosity up?  no NAK?
01:51.46Yourname``lol drmessano
01:52.48drmessanohttp://i293.photobucket.com/albums/mm45/hotcarltogo/fanboy.jpg
01:53.01drmessanoMAC fanboy
01:53.10edgeQwell, http://rafb.net/p/E1V2P147.html is what im sending it
01:53.13zomgoblinzdrmessano, mind looking at  http://pastebin.com/m76703313   for some reason box b is not registering with box a
01:53.24cmantitoanyone use the automon feature much? I thought I had it setup,  but it tells me the monitor application is disabled when the feature code is pressed.
01:53.49Qwelllooks right
01:54.00JTYourname``: voip doesn't induce echo by itself. echo is cause in analogue sections; eg. POTS lines, telephone handsets.
01:54.14JTYourname``: however, voip can make a small bit of echo sound worse
01:54.17JTdue to latency
01:54.22edgeQwell, and i have the files there, its just the phone doesn't care keeps saying "registering" non stop
01:54.32Qwelledge: welcome to my world...
01:54.44JTYourname``: the only way to properly remove echo is at the analogue to digital conversion stage
01:54.56JTeg. PSTN gateway, or IP Phone
01:55.15edgeand the log says its missing No CTL installed, and File Not Found : CTLFile.tlv
01:56.13Qwelledge: ignore that
01:57.34edgeQwell, so what does a guy do? does it have something to do with my switch? or maybe my phone? does it hate my tftp client?
01:57.40edgedoes it hate me?
01:58.15Yourname``JT: Didn't know the part about voip echo.. thanks..
01:58.45JTYourname``: digital (TDM & VoIP) does not cause echo, only analogue :)
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01:59.31Yourname``JT: Ok, how can I solve this, or attempt to? Location 1, in US complains, and location 2, in Asian never complains, and it's soooo far away from the Asterisk box which is hosted in the US east coast.
02:03.59drmessanoPangea
02:04.03drmessanoThats the only answer
02:11.02JTYourname``: record the numbers that have an issue with bad echo
02:11.10JTcall said numbers using different technology
02:11.20JTsee if echo exists
02:11.59Yourname``JT: Gotcha. What if it doesn't?
02:13.00JTthen your provider is at fault
02:13.36Yourname``JT: Cool. You don't think the Asterisks' need any tweaking?
02:13.59JTif it still exists and you're on a good line, like cell phone or echo cancelled pri, and there is still echo, the echo must be really bad and there's not much anyone can do
02:14.02lanningasterisk does not cause echo
02:14.19JTno, there is no parameters you can tweak in asterisk for echo in voip
02:14.23edgewhats this .TVL file my phone is trying to pull from my TFTP
02:15.53Yourname``Ok..
02:16.08*** join/#asterisk m4sk4r4 (n=m4sk4r4@189.13.71.98)
02:17.23drmessanoAsterisk doesnt prevent echo!
02:17.28drmessanoHappyClownPBX does
02:19.30lanningKrustyBox?
02:19.42drmessano~happyclownpbx
02:19.43jbot[HappyClownPBX] is currently in closed beta, approaching 12GB in size, uses Asterisk for it's core, it pwns, is also now compatible with the Diahatsumashiniriki Keyotason 200LP-A11 SIP phone
02:20.33drmessano~happyclownphone
02:20.33jbot[HappyClownPhone] is a happy softphone using the latest GSM codec, utilizes closed APIs to communicate with HappyClownPBX.  It also pwns, or gay
02:20.50drmessanoHmmm
02:20.58*** join/#asterisk justnuts (n=boonl@202-89-181-213.static.dsl.amnet.net.au)
02:21.04drmessanoI only know one person who adds "gay" to everything
02:21.12drmessanoI guess obnauticus is at it again
02:26.18Strom_Cclearly, he's got penis on the brain
02:26.59drmessanoHe's 17.. That should explain it all
02:31.57plikedge: still around?
02:39.48edgeplik, yes i am
02:39.59edgeplik, still pushing this ip phone around by the cord
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02:41.32plikat some point Cisco chnged the capitilisation requirement for various files to upgrade 79xx phones software, so try duplicating XMLDefault.cnf.xml as xmlDefault.CNF.XML and see if that gets you anywhere
02:42.04Juggiethats fruitless
02:42.05edgeplik, even if it takes it from the ftp?
02:42.06Juggiecheck your tftp log
02:42.11edgeit takes the file
02:42.14Juggiesee what its asking for
02:42.15edgeand never asks for anything ever again
02:42.28edgeit also asks for a TLV file, for the stupid SCCP protocol
02:42.50plikwell, that worked for me,and for NovceGuru a few days ago.... your call
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02:46.00Juggieedge, it in SCCP mode then
02:46.11edgethat is correct
02:46.12Juggie*its
02:46.18edgeand i wish it to be Sip
02:46.28edgehere in lies my frustration
02:46.39Juggiewhat version of firmware is on the phone now
02:46.49edgeit takes its SEP<mac>.cnf.xml file ,but talks not more to the tftp server
02:47.10edgeSCCP43.8-2-1ES5S
02:47.48*** join/#asterisk gerphimum (n=trekkie@cpe-24-167-100-36.satx.res.rr.com)
02:47.52Juggieand what are you trying to upgrade to
02:47.55JuggieSIP version what
02:48.12edgei was told i have to update in a "very" strict path
02:48.18edgei've tried...just about 9 images
02:48.41Juggieyeah there is that
02:49.25Juggietry here, http://www.voip-info.org/wiki/index.php?page=cisco+79xx
02:50.59Juggiei have not done in it years so i would not be the best person for advise
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02:53.39Juggiedo you see it trying to load OS79XX.TXT
02:53.41Juggiefrom tftp
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03:01.02los415hey guys quick question has anyone every tried to connect a stratum 1 bits clock to the recieve pins of a digium t1 card
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03:22.09MrTelephonewhat the heck is going on in here tonight
03:24.52zomgoblinzhow do i fix the error Crypto support not loaded!
03:27.41MrTelephoneasterisk supports crypt now?
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03:30.11*** mode/#asterisk [+o lmadsen] by ChanServ
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03:39.13zomgoblinzrsa
03:39.27zomgoblinzi cant get res_crypto to compile :(
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03:43.24MrTelephoneyou got openssl-dev installed?
03:43.50MrTelephonepaste the line where it says -L and a bunch of libraries afterwards
03:43.56MrTelephoneright before it fails
03:44.34zomgoblinzyeah i got it I think
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03:44.45zomgoblinzlibssl-dev
03:44.49zomgoblinzi think was what i was missing
03:45.10MrTelephonetry that and see what happens
03:47.07zomgoblinzdammit
03:47.14zomgoblinzhow come my iax trunk wont register
03:47.15zomgoblinz:(
03:47.35zomgoblinzhttp://pastebin.com/m76703313
03:47.40zomgoblinzthose are my configs
03:47.49zomgoblinzand no matter what authentication method i use
03:47.51zomgoblinzit rejects
03:48.19golumnhow can I say asterisk to use es sound file not en?
03:50.30MrTelephonei don't use iax
03:50.57MrTelephonelook at your debug more closely if you can
03:51.15zomgoblinzMrTelephone, whats odd is rsa means I cant hear anything
03:51.19zomgoblinzeven once i had it working
03:51.24zomgoblinzbut without i hear just fine
03:51.40MrTelephoneare you encrypting your audio too?
03:53.00zomgoblinzi dunno
03:53.01zomgoblinz:p
03:53.04zomgoblinzdont think so
03:53.09zomgoblinzjust using a key
03:54.22MrTelephonemd5 is one way encryption for negotiating passwords
03:54.38MrTelephonersa key will be used to decrypt encrypted audio
03:55.33MrTelephonewhat version of asterisk is that your using?
03:56.42zomgoblinz1.4.16.
03:56.48zomgoblinzand 1.4.16.1
03:57.05zomgoblinzoh
03:57.09zomgoblinzand 1.4.18.1
03:57.10zomgoblinzerr
03:57.12zomgoblinz19.1
03:57.41zomgoblinzshould i use the newest on both?
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04:27.20C4colohello
04:27.59C4colodoes anyone know of any analog cards that allow software programmable FXO/FXS function? i.e. without swapping a module?
04:28.20C4coloI have a customer asking about this, and he seems to think that it is possible/available on the market
04:28.43C4coloI have said I don't know of any, but I wanted to bounce it off you guys before I say for sure they don't exist
04:36.49JTnever heard of them
04:41.26C4colothanks, just want to make sure I'm sane here
04:41.31drmessanoI dont see it being possible
04:41.39drmessanoits two different hardware functions
04:41.50C4colowell, it is possible, if a card had both circuits and then used one or the other for the port
04:42.09C4colobut with most pci solutions the board realestate is far too scare to double the electronics
04:42.18drmessanoOk, tell me how you switch hardware with software
04:42.27C4colopins on a package
04:42.37C4colouse pins 1-4, or 5-8
04:42.51C4colothose pins connect to the specific circuit for fxo or fxs
04:43.37C4coloI know why they don't do it, it is a waste of space, electronics, and complicates the circuit / programming of the device
04:44.00C4coloa module is just that much simpler and straigh-forward, and no waste of board space for a circuit that won't ever be used
04:44.30C4colodrmessano: tell me how onboard sound cards can provide digital audio OR analog audio out the same port?
04:45.09drmessanoMost of the ones I have seen utlilize some form of solid state relays
04:45.18C4colo0.o
04:45.51C4colothen there is your answer
04:46.11C4colouse a "solid state relay" to connect the card to the FXO circuit instead of the FXS
04:46.26C4colobut both circuits would need to be on the card
04:46.33C4colowhich is why I don't think they exist
04:46.59drmessanoNow we're not talking about software differences
04:47.22C4colo<PROTECTED>
04:47.28drmessanoYoure building two modules onto one card and having software switch a relay
04:47.34C4cololike I said, it would have to have both circuits built onto the board
04:47.36drmessanoor set of relays
04:48.36C4colothe modular design is much better in my opinion
04:48.58drmessanoWell, I guess its about the use of terminology
04:49.19C4coloyep, always is =)
04:49.24drmessanoWhen I think of being software programmable, as you originally stated, I think of loading up a DSP chip
04:49.29drmessanoThats "programmable"
04:49.37JTerll
04:49.38drmessanoSwitching two circuits on a card is not
04:49.39JTwell
04:49.46Qwellit really doesn't make a whole lot of sense, to be honest
04:49.49JTyou would not need to double the circuitry
04:49.59QwellI mean, it may be possible, but you would be nearly doubling the cost
04:50.05JTpretty much an FXS port could act as FXO if designed to do so
04:50.10C4coloyea, "Programmable" was the customer's words
04:50.22JTbut it doesn't make sense from a manufacturer pov
04:51.33C4coloyea, maybe in a specific propriatary device with one port that might need to be reprogramed it could make sense, but for a TDM card with multiple ports I can't think of any reason it would be designed that way
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04:57.22JTthe only major electronic difference is an FXS port requires talk battery voltage and ringing voltage
04:57.39JTof course you can always make more money if you sell it as 2 different modules
04:58.42C4cologood point
04:59.10drmessanoHow else are you gonna end up with a box of useless FXS cards?
04:59.40drmessanoI wonder what the ratio of FXS to FXO cards is
04:59.47drmessanoor modules, I should say
04:59.56drmessano10 to 1
05:00.00drmessano20 to 1
05:00.18JTin which direction?
05:00.24drmessanoFXO
05:00.41drmessano20 FXO sold for ever 1 FXS
05:01.06JTmaybe
05:01.21JTneither should be sold, people should just go digital! :P
05:01.28drmessanoIts obviously going to be hugely disproportionate
05:01.38C4colohotels make up for the difference
05:01.39drmessanoBut I am not sure how huge
05:01.45C4colo200 FXS ports with 4 FXOs
05:02.16drmessanoI would think they would buy channel banks and not deal with cards that use daughterboards
05:02.28C4coloyea, probably
05:03.17JT200FXS ports on PCI cards would be rofl
05:03.25JTeither rofl or :~~~(
05:03.52drmessanoTDM20000P sounds like a high power RF tube
05:04.31JTyou'd need 8 * TDM2400P to do 200 ports
05:04.34JTwhich is insanity
05:04.53JTespecially when you consider that digium does not recommend more than 2 PCI cards per system
05:04.58JTso that's 4 servers
05:05.04JTjust to terminate those ports
05:05.05C4coloheh, yea, but
05:05.23C4coloif you consider 99% of those lines will not be in use at any given time
05:05.30drmessanoIf you have a 200 port card, the TDM20000P, and bolted 4 together for 800 ports, would it be the 4TDM20000P?
05:05.35drmessanoNow thats a tube name
05:05.35JTC4colo: how does that help?
05:05.49C4coloPCI bus throughput
05:05.53JTC4colo: no
05:06.02JTthat is not why more than 2 cards are recommended
05:06.13JTthe pci bus can easily handle the throughput
05:06.19JTit's interrupts that are the issue
05:06.21drmessanoThe 2000w power supply you need is
05:06.25drmessanolol
05:06.48C4coloyou are telling me that the 2400s don't synch their clocks?
05:06.50JTand irq sharing too
05:07.04C4coloso you have 5 different 1000hz timers?
05:07.12C4coloall generating interrupts?
05:07.27JTnot sure, but i couldn't recommend putting more than 3 cards in the same pc, and digium has my back on this one :)
05:07.32C4colosounds like an engineering flaw if you ask me
05:08.05JTyeah, but still
05:08.06C4coloif you have that many devices, none of which are generating usage interrupts, then there should be almost no interrupt communication
05:08.17JTget a damn channel bank or sip gateay instead ;)
05:08.19C4coloyou could have 1000 cards, none of which are being used and have no issue
05:08.28C4coloif it were designed properly
05:08.52C4coloif only 2 ports out of 24,000 ports were in use, only two valid interrupt streams would be active
05:09.01drmessanoWell
05:09.03drmessanothe good news is
05:09.17drmessanoBoth of you are having an obsolete convo anyway
05:09.30JTC4colo: it's a stupid way to terminate anyway imho
05:09.39drmessanoBy the time you stuff 3 PCI cards into a box, you've loaded the board up :)
05:10.08drmessanoSo, much like the midgets in a volkswagon dilemna, the problem fixes itself
05:10.18C4coloI agree
05:10.32C4coloI'm just saying that if designed properly there should be no issue
05:11.08drmessanoPeople still use PSTN?
05:11.10drmessanoOh snap
05:11.20JTpstn, of course
05:11.26C4colojust use the first card as the master timing source, there are your 1000 interrupts per second across the bus.  And this would only work in a less than 10% utilization environment
05:11.37drmessanoI dumped all my friends who werent SIP accessible
05:11.42JTit's this global phone network
05:11.43drmessanoWasnt worth the hassle anymore
05:12.02C4colotwo 4-port T1 cards and some channel banks, or a switch and ethernet gateways would work
05:12.20drmessanoTDMoE FTW
05:12.23JTlol
05:12.26C4coloyep
05:12.38JTproprietary protocol that nothing supports
05:12.47JTsorry, asterisk and redfone supports it
05:12.48JT:P
05:12.52drmessanoLinksys LNE100TX V5.1 is the only card you need for TDMoE
05:12.59C4coloanyway, for the record, there is no knowledge of any software switchable FXO/FXS solutions that would interface with a pbx on either a 32-bit PCI bus or ethernet?
05:13.03drmessano$10 on ebay
05:13.14Qwelllne100tx...ugh
05:13.17JTC4colo: not that i know of
05:13.25JTwhat is a lne100tx?
05:13.25Qwellrtl8139
05:13.28JThaha
05:13.29drmessanoHAHHA
05:13.35drmessanoThose are $2.50
05:13.46JTintel NICS > *
05:13.52Qwelljust saved you $7.50 per port
05:14.00C4cololol
05:14.00drmessanoI can get new in box rtl8139 cards for $5
05:14.14drmessanoUsed for 75 cents, I am sure
05:14.15Qwellyou can get a gross of rtl8139 cards for $5
05:14.30JTi've bought used Intel Pro/1000 MT cards for $15
05:14.43QwellSun QFE
05:14.51drmessanoHP 100VG
05:15.04JTvglan is pretty useful in some situations
05:15.05Qwellno, Sun QFE is the best card...like...ever
05:15.13C4coloany reason to choose Digium over Rhino/Sangoma PCI cards?
05:15.18Qwellit's affectionately known as a Happymeal
05:15.28drmessanoC4colo: Qwell gets 5%
05:15.31JTit only does 100Mbit/s
05:15.46QwellJT: It has Fast right in the name.
05:16.05QwellI put one in my SS10...it's so awesome.  Thing can't push 10mbit on a single port.
05:16.22C4coloI mean $650 for the Digium or $400 for the Rhino (and the Rhino can still have 4 more ports added)
05:16.23JTthat's a limitation of the SS10 i suspect
05:16.30drmessano100VG rocks at 10mbit on a standard switch
05:16.31Qwellit is, heh
05:16.48Qwelland to think, the scsi HDs are on the same bus...
05:16.53JTyeah but 100VG can push 100Mbit/s over cat3
05:17.05Qwellso, I've got a 10k RPM drive in there, and it's so wasted
05:17.23drmessano100VG doesn't do so well in a high RF environment lol
05:17.32C4coloit is all that ETHERnet traffic that's getting it wasted
05:17.38drmessanoActually.. it doesnt work at all
05:17.40JTi think thats what fibre is for
05:18.25C4coloI think Ethernet won out over IPX/SPX because it has a cooler name
05:18.39JTC4colo: you're mixing layers
05:18.45JTIPX/SPX runs over ethernet.
05:18.46C4colowell yea
05:18.52drmessanoWe had a 20 runs of Cat5 pulled for rollout years ago, put a VG lan on it, and it sucked out.. 8 years later I am throwing $10 linksys cards on it, and its rockin
05:18.53C4colotcpip
05:18.55drmessanoGo figure
05:18.55C4colonevermind
05:19.07C4coloyou are right, it's 11 pm and I've been up since 6 this morning
05:19.39drmessanoIPX/SPX lost to TCP/IP because IPX/SPX sounded too confused by its own name
05:19.45C4coloI never really memorized the OSI model anyway, just glossed over it
05:19.49drmessanoWhereas, TCP/IP knows the score
05:20.24C4colo"Yea yea, physical layer, transport layer, packets, headers, DUDE IT's 4:20!"
05:20.35drmessanoNetBeui lost out because it sounded like a floatation device or a buttplug
05:20.40C4colohahahaha
05:21.14zomgoblinzquestion, when i add in an iax trunk, it's working but say the phone I am calling has voicemail, its hanging up when the voicemail comes on (fiancee's cell phone voicemail"
05:21.15drmessanoI bet hate crimes were committed over NetBeui
05:21.19JTspeaking of networks
05:21.39JTif anyone has played with open source implementations of MPLS or L2TP, please let me know
05:22.01C4colozomgoblinz: you are calling out to an external voicemail system?
05:22.01drmessanoMy new apache install is blowing the everloving crap out of the IIS I had in place
05:22.16zomgoblinzC4colo, its my fiancees voicemail
05:22.23C4colozomgoblinz: over IAX?  ... and when the voice mail system answers it disconnects?
05:22.29zomgoblinzwithout the trunk it doesnt just hang up
05:22.32zomgoblinzyes
05:22.39zomgoblinzif she picks up herself
05:22.42zomgoblinzit answers
05:22.43zomgoblinzodd?
05:22.45C4coloand by trunk you mean "IAX2 Trunking" mode?
05:22.49zomgoblinzyes
05:22.52C4coloor just a trunk to an ITSP?
05:23.02zomgoblinzC4colo, even without trunk=yes
05:23.04zomgoblinzsame result
05:23.08C4colohmm
05:23.14zomgoblinztrunk between two ast boxes
05:23.17JTa cause code issue perhaps
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05:23.23zomgoblinzthen one hits a ptsn gw
05:23.33C4colodoes this happen on other calls?
05:23.42zomgoblinzno
05:23.43C4colofor example when a person answers the phone?
05:23.45C4colohmm
05:23.52zomgoblinzlike if i call 1800hotsexx it just picks up
05:23.55zomgoblinzif she answers
05:23.58zomgoblinzit picks up
05:24.10zomgoblinzyes, i actually dial that to test
05:24.11C4coloin features.conf do you have ** = disconnect?
05:24.18C4cololol
05:24.23[T]an2I just upgraded my zaptel from 1 4.2 to 1.4.10. without changing the configs that were working when I run ztcfg now, I get the following error:
05:24.23[T]an2ioctl(ZT_LOADZONE) failed: Invalid argument
05:24.23[T]an2Notice: Configuration file is /etc/zaptel.conf
05:24.23[T]an2line 9: Unable to register tone zone 'us'
05:24.42[T]an2I am not finding anything in google to identify what changed between versions.
05:24.49[T]an2can anyone point me in the right direction?
05:25.04zomgoblinzC4colo, no i do not
05:25.19drmessanoUmmm
05:25.35[T]an2here is my zaptel.conf http://pastebin.ca/1003637
05:25.38C4colozomgoblinz: are you using silence supression / CN frames / comfort noise generation?
05:25.42drmessanoYour fiance` answers the phone at 1800hotsexx
05:25.47drmessanoDude, I hate to tell you this
05:25.48C4colohaha
05:25.50drmessanoBut thats a sex line
05:25.53zomgoblinzC4colo, no
05:26.09zomgoblinzC4colo, i actually dont know, base install just an iax trunk
05:26.12zomgoblinznothing special
05:26.27zomgoblinzdrmessano, nah he voicemail on her cell hangs up
05:26.47C4colohave you tried other voice mail?
05:27.08*** join/#asterisk tuxd00d (n=tuxd00d@128.187.132.25)
05:27.13C4colodoes your cell phone voice mail cause it to disconnect?
05:27.26zomgoblinzher verizon cell phone voicemail
05:27.29drmessanoIts a sex line that hangs up on you?  Good god, they finally perfected the simulated marriage line
05:27.31zomgoblinzcauses it to disconnect
05:27.35zomgoblinzover the trunk only
05:27.42zomgoblinzif i just call out another path no trunk
05:27.45zomgoblinzno problem
05:27.59zomgoblinzmiddle ast box is registering with the ptsn gw
05:28.06drmessano"Hi big boy... thank you for calling.... NOT TONIGHT I HATE YOU KTHXBYE"
05:28.14C4coloreinvite=no
05:28.18C4colo?
05:28.25C4colocanreinvite
05:28.28C4colowhatever the option is
05:29.44zomgoblinzah
05:29.46JTanyway
05:29.46zomgoblinzthat fixed it
05:29.47zomgoblinzthank
05:29.51JTit's probably a dialplan issue
05:29.54[T]an2anyone know what I am doing wrong in my config above?
05:29.56zomgoblinzi put that on one box and noth the other
05:29.58JTand how you handle cause codes
05:30.20JTzomgoblinz: do an iax2 debug and record down what cause code you get when it goes to voicemail
05:30.23zomgoblinzJT, it was just canreinvite==no
05:30.28zomgoblinzit works now
05:30.29zomgoblinzjust fine
05:30.31JToh really
05:30.35JTthat is unusual
05:30.47zomgoblinzhmm
05:30.49zomgoblinzwait
05:30.53JTconsidering it otherwise worked
05:30.54zomgoblinzits timing out?
05:31.00JT?
05:31.07zomgoblinzwhen her voicemail picks up on one ring
05:31.09zomgoblinzit works
05:31.13zomgoblinzbut otherwise no
05:31.35zomgoblinzso the call is hanging up after a certain amount of time
05:31.42drmessanoIm confused
05:31.45JTDial timeout
05:31.47zomgoblinzand not waiting for the call to complete
05:31.53zomgoblinzwhere do i fix that?
05:31.54JTso your wife always answers faster than voicemail?
05:31.59zomgoblinzyes
05:32.01JTin your Dial string
05:32.01zomgoblinzalways
05:32.02zomgoblinz:)
05:32.12zomgoblinzto the PTSN?
05:32.18JTright
05:32.22C4coloyea, it woul be in the outbound route
05:32.44[T]an2is loadzone still required in zaptel 1.4.10?
05:32.54[T]an2loadzone=us?
05:33.05zomgoblinzexten => _1NXXNXXXXXX,1,Dial(SIP/${EXTEN}@vitel-outbound,60)
05:33.06zomgoblinz<PROTECTED>
05:33.28C4coloyea
05:33.40C4colowas it 30 before or something?
05:33.44C4colomake it bigger
05:33.48zomgoblinznot set
05:33.49zomgoblinzat all
05:34.00C4colochange the ,60 to ,240
05:34.01C4coloor something
05:34.13C4colooutbound calls should allow much more time for ringing out
05:34.18zomgoblinzit didnt even have 60 before
05:34.19zomgoblinzah
05:34.25JTyeah because we'll definitely wqait 4 minutes for a call to ring :o
05:34.30C4coloI don't know what the default is but it is probably 30
05:34.47C4coloI have sat and waited for 5+ minutes for a line to ring
05:34.49C4colonot often
05:35.01JTyou are crazy
05:35.07C4colobut if someone wants to sit and listen to the ringing sound for 5 minutes hoping someone will "wake up" on the other end
05:35.16JTas it almost always means there's a network failure in my experience
05:35.22C4colowell
05:35.23zomgoblinzthats just stalkerish
05:35.29C4coloalmost every line has voicemail now
05:35.39C4coloso leaving it at a max of 5 minutes doesn't hurt anything
05:35.47zomgoblinzmy parents still have a message machine
05:35.53zomgoblinzwith a tape
05:35.53C4coloand if someone has their phone forwarded, like a hunt group, then it may take 60 seconds
05:35.56JTzomgoblinz: not as stalkerish as calling all of someone's phones simultaneously
05:35.58C4colowell, same thing
05:36.01C4colosomething will answer
05:36.08C4colobefore 5 minutes
05:36.28zomgoblinzJT, I have a number that if you call it rings everyone on our network including the owner of the company in a meetme
05:36.29zomgoblinz:D
05:36.39zomgoblinzhe hates it
05:36.42JTheh
05:36.43zomgoblinzbut its funny
05:36.48C4colobut if something is configured to ring multiple lines, one after the other, without an Answer() happening, and the person finally answers after 61 seconds, your PBX would have cut it off
05:37.12JTi have numbers that when dialling cause all of my 3 mobile phones to ring at the same time
05:37.12C4coloheh, that is funny
05:37.18JTi don't give it out to many people :P
05:37.24zomgoblinzhaha
05:37.26*** part/#asterisk [T]an2 (n=ckwall@c-71-195-194-193.hsd1.ut.comcast.net)
05:37.34C4coloI should create extensions for some of my clients that ring their office/home/cell all at once
05:37.40*** join/#asterisk steliosk (n=Stelios@athedsl-311591.home.otenet.gr)
05:37.41C4colothen I don't have to try one, then the other, then the next
05:38.09C4coloI have three methods of dialing my one cell phone
05:38.16zomgoblinzis there a way to jitterbuffer the transmit portion of the call if you are using two * boxes?
05:38.18C4coloI give them out differently to different people
05:38.38C4colouh, why would you jitterbuffer the tx?
05:38.49zomgoblinzbecause its going over satellite
05:38.52C4colojitter only happens on the far end of a link, i.e. the receive link
05:38.58C4coloreceive end
05:39.06C4coloyou can't have jitter if you are sending the packets
05:39.17C4colothey won't be out of order unless you are sending them out of order, which would be absurd
05:39.21JTC4colo: unless your pc is really screwed
05:39.25C4cololol
05:39.26C4coloyea
05:39.34zomgoblinzbut the receiving end?
05:39.34JTbut yeah, there's only a point having at at rx
05:39.47C4coloyea, you can set a fixed jitter buffer on both ends
05:39.50JTs/at at/it at/
05:39.52C4colowhich will dejitter the rx
05:39.53zomgoblinzok so it transmits out over the satellite
05:40.00zomgoblinzhm
05:40.11JT2 way sat?
05:40.14JTor modem up channel
05:40.24zomgoblinzSCPC/SCPC
05:40.28zomgoblinzsatellite
05:40.28JT?
05:40.32zomgoblinzin iraq
05:40.34zomgoblinzso two way
05:40.35JT2 way satellite?
05:40.35zomgoblinz:P
05:40.37zomgoblinzyes
05:40.37JTok
05:40.45JTuse a codec like ilbc
05:41.03JTilbc is loss tolerant
05:41.07zomgoblinzah
05:41.12C4coloit bloats if needed to add more CRC
05:41.17C4colowell, parity
05:41.20C4colosame-ish thing
05:41.29JTand do NOT use G.726
05:41.38JTG.726 does not tolerate any loss at all
05:41.42JTit's a delta codec
05:41.48C4coloonly helps with loss due to a weak link, not to congestion
05:42.04C4colocongestion + ilbc = bigger issue as the ilbc uses more bandwidth, causeing more congestion
05:42.20zomgoblinzweird
05:42.22C4colos/causeing causing/
05:42.23zomgoblinzit worked once
05:42.27zomgoblinznow the same is happening
05:42.34C4colohmm, jbot doesn't like me or I do'nt know how to format it
05:42.34zomgoblinzit's just not always picking up
05:42.39zomgoblinzquestion
05:42.42JTC4colo: more bandwidth than what?
05:42.53C4colothat the original stream
05:42.58zomgoblinzhow do i setup the pstn gateway to register at the last leg?
05:43.06zomgoblinzand everything to route out the trunk
05:43.14JTC4colo: in what codec?
05:43.20C4coloilbc uses a dynamic amount of bandwidth, if it detects packet loss it adds more redundant data to the packets, making them bigger
05:43.38C4coloif the loss was due to a congested link, the codec uses more bandwidth than before, causing more congestion
05:43.46*** join/#asterisk [T]an2 (n=ckwall@c-71-195-194-193.hsd1.ut.comcast.net)
05:43.51C4coloso, ilbc's tolerance only really works for an uncongested, unreliable link
05:44.16[T]an2my internet dropped. is anyone able to answer my question regarding defaultzone and loadzone = us?
05:44.18JTsame with any codec really
05:44.21[T]an2in zaptel.conf?
05:44.28JTC4colo: you really need to compare it with another codec
05:44.42[T]an2it worked as is in zaptel 1.4.2 and i did an upgrade to 1.4.10 and now it doesnt work
05:44.50[T]an2I see nothing so far on google that helps me
05:45.04C4colowell, yea, just pointing out that if the loss was due to a congested link the loss tolarant aspect of ilbc is useless
05:45.06[T]an2line 9: Unable to register tone zone 'us'
05:45.22C4colowich is the same for about any codec
05:45.30C4coloilbc != magical
05:45.32JTanything real time is useless if the link is congested
05:46.24C4coloexactly
05:46.31[T]an2:-(
05:46.52C4coloI don't know why it would be saying that us is not a valid option [T]an2
05:51.05zomgoblinzhmm
05:51.16zomgoblinzsomehow it just keeps hanging up
05:51.17zomgoblinzgay
05:53.23JTlook at the debug, see if you are doing it, or if the provider is
05:57.48[T]an2i just reinstalled zaptel from scratch and it is still doing it!!!
05:58.21[T]an2am i doing something wrong? /etc/zaptel.conf: http://pastebin.ca/1003647
06:01.56zomgoblinzJT, how do i look at the debug?
06:02.41JTiax2 debug or something?
06:04.19[T]an2i rolled back to an older version of zaptel and it works just fine
06:06.17[T]an2just reinstalled the latest version and it does not work
06:06.36JTthen don't use the latest version?
06:06.59[T]an2anyone else here using it?
06:11.03[T]an2does it matter what verison of zaptel I use with asterisk 1.4.19?
06:11.55*** join/#asterisk gr0mit (n=tim@extrt.txrx.org.uk)
06:16.40*** join/#asterisk MooingLemur (n=troy@unaffiliated/mooinglemur)
06:18.26[T]an2same problem in zaptel-1.4.9 :-(
06:20.39zomgoblinzafter i do iax2 trunk debug where do I find the debug info?
06:25.57Juggie[T]an2, if you upgrade zaptel you should recompile asterisk after
06:27.30[T]an2Juggie: I am simply trying to get through a ztcfg. Not even far enough to install asterisk
06:28.52Juggie[T]an2, when you reinstalled the latest version did you unload the modules and reload them
06:29.02[T]an2yes i did
06:29.12Juggieso you did that and ran ztcfg and what was the output
06:29.20[T]an2deleted everything from /usr/lib/asterisk/modules
06:29.22*** join/#asterisk mbit (n=mark@218-214-57-65.people.net.au)
06:29.30mbithey guys
06:29.44mbiti am having a problem with writing to the cdr mysql table
06:29.45zomgoblinzhow do i fix the warning unable to support trunking on user without zaptel timing?
06:29.57[T]an2output was: http://pastebin.ca/1003666
06:30.12Juggiezomgoblinz, install zaptel and load ztdummy
06:30.19Juggieyou'll need to recompile asterisk too
06:30.22mbitdoes asterisk-addons write the userfield to the mysql database?
06:30.34Juggiembit, yes if you enable it but its enabled in the source
06:30.43Juggiecheck the make menuselect before you compile
06:30.51mbiti use an rpm
06:30.59mbitis there a way to check it after its compiled?
06:31.12Juggieits not compiled by default so i'm guessing it wont be in the rpm
06:31.38mbitwhat fields are compiled in by default that i can write to?
06:31.39Juggieactually, userfield
06:31.44Juggiehmmm
06:31.55Juggiei was thinking of unique id
06:32.00Juggieuserfield i am not sure about
06:32.24zomgoblinzJuggie, i have to install zaptel even if I dont have a zaptel card?
06:32.52zomgoblinznvm
06:32.53[T]an2zomgoblinz: that is correct. You need ztdummy for timing source if you do not have the card.
06:32.53*** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk)
06:32.56zomgoblinzztdummy
06:32.59zomgoblinzah
06:33.01Juggiezomgoblinz, yes.
06:33.14zomgoblinzwhere do i get zaptel?
06:33.19zomgoblinzlooking via google
06:33.21[T]an2asterisk.org
06:33.49Juggie[T]an2, thats an interesting error
06:34.14[T]an2interesting is that I do not get it when using zap 1.4.5
06:34.26*** join/#asterisk ikey (n=ikey@86.99.8.60)
06:34.31[T]an2but i am having other issues and was suggested to use a more current version of zaptel.
06:34.58[T]an2i have done nothing different in the config. but the ztcfg yields this new error I showed you
06:35.03Juggie[T]an2, are you using trixbox?
06:35.07[T]an2no
06:35.12[T]an2asterisk 1.4.19
06:35.15Juggieok.
06:35.22Juggiejust curious the only hits on this problem i found was trixbox
06:35.52*** join/#asterisk stkn (n=stkn@gentoo/developer/pdpc.active.stkn)
06:35.59Juggie[T]an2, it seems like a bug to me, did you try removing the defaultzone=us line?
06:36.03[T]an2yeah
06:36.11Juggiedid it work then?
06:36.20[T]an2it works without defaultzone=us and loadzone=us
06:36.31[T]an2but doesnt that leave me without ringing tones and such?
06:36.43Juggiei am not sure.
06:36.46Juggiebut it seems like a bug
06:36.57Juggiei would report it on the bug tracker if i were you and stick with the working version.
06:37.01Juggiehttp://bugs.digium.com
06:39.22Juggietheres no reason so long as us is in zonedata.c that it should fail
06:41.28Juggiedid you try 1.4.10?
06:41.31Juggie(zaptel)
06:41.46Juggietheres been a couple of zaptel releases since 1.4.9 there was 1.4.9.1 1.4.9.2
06:46.57*** join/#asterisk Mahmoud (n=foo@unaffiliated/mahmoud)
07:03.21los415hey guys is there a way to see which t1 is the active timesource on a 410P card
07:03.37los415i know u set it in the zaptel
07:03.50los415but i'd like to see if it's really getting a clock sync
07:07.56*** join/#asterisk the_5th_wheel (n=edd@webster.cybertek.co.za)
07:09.42the_5th_wheelHi. Can anyone tell me why i get no audio on my sip phone when i phone a local extention, but when i phone an external number, it works fine, and when an external server phones me, it works fine?
07:12.20the_5th_wheelThe SIP phone in question is on a natted internet connection
07:24.50*** join/#asterisk MatBoy (n=MatBoy@wiljewelwetenhe.xs4all.nl)
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07:34.02*** join/#asterisk Mahmoud (n=foo@unaffiliated/mahmoud)
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07:40.54cy3o3yo
07:41.22zomgoblinzthe_5th_wheel, by external you mean something on a pstn gw?
07:41.24zomgoblinzor such
07:42.00the_5th_wheelzomgoblinz: Yes.
07:42.22zomgoblinzthe_5th_wheel, can you pastebin your configs somewhere minus your passwords and usernames
07:42.30zomgoblinzjust the extensions.conf and sip.conf
07:42.40the_5th_wheelI have another Voip server wich handles a calling card system, and it handles the PSTN lines
07:42.44the_5th_wheelok
07:42.49zomgoblinzok
07:43.08zomgoblinzso just internal across the asterisk box isnt working?
07:43.18zomgoblinzie both sips are on registering with it
07:45.07the_5th_wheelOne phone is on the same network as the asterisk server.
07:45.19the_5th_wheelAll phones on the local net can phone each other perfectly
07:45.25zomgoblinzok
07:45.30the_5th_wheelBut when we try and phone a phone outside, no audio
07:45.39zomgoblinzis it via a trunk?
07:45.40zomgoblinzor such
07:45.53zomgoblinzand what do you see with asterisk -r
07:46.00zomgoblinzcore set verbose 9
07:46.04zomgoblinzwhen you make the call
07:46.58*** join/#asterisk Mahmoud (n=foo@unaffiliated/mahmoud)
07:48.53the_5th_wheelok, the PSTN is via an IAX trunk
07:49.18the_5th_wheelmy ext.conf http://pastebin.div0.co.za/results/C4C2B6542.html
07:52.10the_5th_wheelthis is an exerpt of my sip.conf, as it is huuuge. ralfe is an internal extention, edd-home and hein is ext
07:52.49zomgoblinzyeah i just need to see your route out
07:52.50zomgoblinziax
07:52.54zomgoblinznot sip
07:53.00zomgoblinzyou're phones sip
07:53.07zomgoblinzdidnt realize your route out was iax
07:53.18the_5th_wheelbut between phones, its sip
07:53.41zomgoblinzwhat codec
07:53.52zomgoblinzand noreinvite=yes ?
07:54.03the_5th_wheelall the sip phones are on the same sip server
07:54.09zomgoblinzok
07:54.18zomgoblinzand they can dial eachother fine?
07:54.22the_5th_wheeluhm, i have allowed for ulaw,alaw,gsm,ilbc,gsm,g729
07:54.26zomgoblinzie internal to internal
07:54.33*** join/#asterisk jivco (n=jivco@85.187.217.6)
07:54.38*** join/#asterisk [T]an2 (n=ckwall@c-71-195-194-193.hsd1.ut.comcast.net)
07:54.43the_5th_wheelwhen the phone is at the office, yes, they can
07:55.00the_5th_wheelbut, when i take the one phone away, i cant
07:55.03zomgoblinzok, what kind of box is at the other end?
07:55.22zomgoblinzthat the ast * box is connecting to?
07:55.39the_5th_wheelDo you mean the sip clients?
07:55.45the_5th_wheelOr the box doing the PSTN?
07:55.51*** part/#asterisk [T]an2 (n=ckwall@c-71-195-194-193.hsd1.ut.comcast.net)
07:56.00zomgoblinzok, you said the phones can call sip to sip across the ast box fine
07:56.03zomgoblinzwith no issue
07:56.07*** join/#asterisk steveaj (n=steve@82-71-61-44.dsl.in-addr.zen.co.uk)
07:56.14zomgoblinzso ext 1000 can call ext 1001
07:56.16zomgoblinzinternally
07:56.19zomgoblinzno issue
07:56.20zomgoblinzcorrect?
07:56.29the_5th_wheelOnly when they are on the same network....
07:56.44zomgoblinzok... when they are not on the same network
07:56.44the_5th_wheelWhen the phone gets moved off of the network, it doesnt work
07:57.02zomgoblinzok, are they still registering with this box?
07:57.04the_5th_wheelthe phone sitting next to me, is at my home (edd-home)
07:57.07zomgoblinzor a sip server
07:57.08the_5th_wheelThey register fine
07:57.14zomgoblinzok....
07:57.22the_5th_wheelThey can Call PSTN number
07:57.28the_5th_wheelPSTN numbers can call them
07:57.41the_5th_wheelBut when i phone an extention at work, no audio gets exchanged
07:57.55zomgoblinzPSTN is across iax I am guessing
07:58.01zomgoblinzsounds like a port issue
07:58.09zomgoblinzie some ports need to be opened
07:58.17zomgoblinzyou really need to check your logs
07:58.22zomgoblinzto see whats happening
07:58.24zomgoblinzor the console
07:58.32*** join/#asterisk gr0mit (n=tim@dhcp4.zuk40.mot-tools.co.uk)
07:58.50zomgoblinzare you encrypting the audio by chance?
07:59.17the_5th_wheelOk, On the server side, no prts are closed,
07:59.23zomgoblinzok
07:59.29the_5th_wheelAnd where im sitting now, ive disabled the firewall in totality
07:59.38zomgoblinzso let me see if i get the network correctly
07:59.45zomgoblinzyour phone next to you
07:59.49zomgoblinzusing say your ISP
07:59.59zomgoblinzis  registering with the ast box at work
08:00.01zomgoblinzcorrect?
08:00.03the_5th_wheelYes
08:00.11zomgoblinzok, so it registers
08:00.16zomgoblinzit works at work
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08:00.19zomgoblinzbut not home
08:00.21zomgoblinzhmm
08:00.24*** join/#asterisk Mahmoud (n=foo@unaffiliated/mahmoud)
08:00.38the_5th_wheelIf i dial from this phone to a PSTN phone, I get Audio, and it works perfectly
08:00.46zomgoblinzyeah
08:01.07zomgoblinzthat makes almost no sense
08:01.08zomgoblinz:)
08:01.11zomgoblinzseriously
08:01.21zomgoblinzwhat vers ast?
08:01.48zomgoblinzand what happens if you disallow=all
08:01.49the_5th_wheelOk, i just installed a sip client on a PC here, and i  can phone it perfectly when the sip proxy is disabled
08:01.52the_5th_wheel1.2.26-BRIstuffed-0.3.0-PRE-1y-p
08:01.53zomgoblinzand just do 1 codec
08:02.06zomgoblinzah
08:02.35*** join/#asterisk stimpie (n=stimpie@84-104-6-10.cable.quicknet.nl)
08:02.40the_5th_wheelOk, Im doing that now
08:03.10zomgoblinzyour phone may have the same setting btw
08:03.16zomgoblinzwhat kind of IP phone?
08:03.42the_5th_wheelOk, the hardware phone is a grandstream Budget tone, And the Sip client is Linphone
08:03.57zomgoblinzso they can call eachother
08:04.00zomgoblinzat your desk there
08:04.11zomgoblinzwhich is the exact same as calling through a pstn gw
08:04.12zomgoblinz:)
08:04.32the_5th_wheelIf i diable the proxy,they can call each other, yes.
08:04.42the_5th_wheelWhen i enable the sip proxy, i get no audio
08:06.07zomgoblinzso disabling the sip proxy solves the issue
08:06.14zomgoblinzall around?
08:06.34the_5th_wheelBut then i cant phone PSTN....
08:07.07zomgoblinz-.-
08:07.09zomgoblinzwhy do you need a sip proxy to call through a pstn?
08:08.52the_5th_wheelDunno....
08:09.08the_5th_wheelwell i havent got a Public IP here...
08:09.20zomgoblinzso its a NAT at work
08:09.24zomgoblinzset it up for a nat
08:09.37zomgoblinzi have to finish my upgrade
08:09.39zomgoblinzbe back
08:10.56*** join/#asterisk nighty^ (n=nighty@210.188.173.246)
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08:58.50nvrpunkNobody picked up in 30000 ms
08:58.50nvrpunk<PROTECTED>
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10:09.11jblackbut did you reload the extensions.conf after changing it?
10:10.17*** join/#asterisk chrisM8 (n=chatzill@78-105-1-131.zone3.bethere.co.uk)
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10:12.56chrisM8hi, got a problem with Realtime extensions. Looks like "switch" is not accepting variables in context name. My dialplan:
10:12.58chrisM8[ivr]
10:12.59chrisM8exten => s,1,Ringing
10:13.01chrisM8exten => s,n,Set(MENU_ID=29)
10:13.02chrisM8exten => s,n,Wait(2)
10:13.04chrisM8exten => s,n,Answer()
10:13.05chrisM8switch => Realtime/menu-${MENU_ID}@ivr
10:13.07chrisM8and SQL query is:
10:13.08chrisM8MySQL RealTime: Retrieve SQL: SELECT * FROM ivr WHERE exten = 's' AND context = 'menu-' AND priority = '5'
10:13.10chrisM8is it me doing something wrong? or it is not possible to use variables in context names for Realtime switch ?
10:13.43jblackYes, variables are allowed.
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10:15.58nvrpunkhmm
10:16.06nvrpunki used the script to get ilbc
10:16.11nvrpunkbut it's still not showing
10:16.16jblackat least I think so. I can't see anything wrong with your flood, chrism8.
10:17.09chrisM8sorry for the flood, note "AND context = 'menu-'", should be "AND context = 'menu-29'"
10:17.23jblackyeah. I caught that.
10:17.29jblackI don't see the problem
10:17.36chrisM8but looks like my ${MENU_ID} has not been accepted for some reason
10:18.48jblackI can't find a command named switch.
10:19.05jblacknot in the book, not with show application, not with show function
10:19.11chrisM8I am playing with 1.4.19
10:19.25jblackI am too.
10:19.44jblacksorry. I'm using 1.4.17
10:20.17chrisM8jblack, so you can confirm that "switch => Realtime/${SOME_VAR}@family" should work? (in theory)
10:20.45jblackNo, because I don't have that command here, nor documentation for it
10:21.39jblackwhich is crazy, because I thought it was a valid command
10:22.19chrisM8documentation can be found here: http://svn.digium.com/svn/asterisk/branches/1.4/doc/extconfig.txt and http://www.voip-info.org/wiki/view/Asterisk+RealTime+Extensions
10:23.14chrisM8it is a valid command and it works as expected when I do for example:
10:23.15chrisM8switch => Realtime/menu-29@ivr
10:23.36chrisM8my problem is I would like to use ${MENU_ID} instead of hardcoded "29"
10:24.32chrisM8which seems to be broken... maybe I will fill a bug repor
10:25.52jblackHmm. I've used switch in extensions.conf
10:26.01jblackwith dundi. Since .15 or so.
10:27.02jblackshow switches shows dundi, realtime, iax2 and loopback.
10:28.07jblackyeah. I can't help you.
10:28.30chrisM8ok, thanks anyway
10:29.14chrisM8do you think I should maybe report this as a bug? as far as I know it should be possible to use vars pretty much anywhere in a dialplan
10:29.35jblackit sounds like a bug to me
10:30.02jblackI'd throw in a noop right above it, to make sure it's expanding as intended.
10:30.27chrisM8done it already, NoOp shows var OK
10:30.41jblacksounds like a bug to me
10:35.18Mahmoudupgraded from 1.4.1 to 1.4.19.1 and SIP phones fail to send pressed keys
10:35.21jblackwe should all chip in an get the ubuntu repo a second aol dialup account.
10:35.31Mahmoudtried different dtmf modes, but nothing changed, any idea?
10:35.48nvrpunkhmm, i have installed ilbc with the script in contrib, but upon compiling it's not active
10:35.49nvrpunkat all
10:35.52nvrpunkany ideas?
10:35.58nvrpunkam i missing some packages?
10:36.22jblackMahmoud: Most of the time, when people say they switched things around a lot, they forgot to reload the config to actually check it
10:36.38Mahmoudjblack, reloaded, even restarted
10:36.59jblackok. make sure you set the client to dtmf2833 too
10:37.12Mahmoudjblack, done, using zoiper
10:39.22Mahmoudany major differences between 1.4.1 and 1.4.19.1 that might cause such issues
10:39.46jblack18 bugfix releases? Quite possibly.
10:39.49Mahmoudworking perfect with 1.4.1 same config
10:40.07jblackoh, dtmf should be fine in 1.4.19
10:40.14chrisM8Mahmoud, I can confirm my 1.4.19 is accepting DTMF (keys) from UA (phones/clients) OK
10:40.52Mahmoudhmmm
10:40.53chrisM8Mahmud, when you do:
10:41.39chrisM8asterisk -r
10:41.41chrisM8core set verbose 40
10:41.42chrisM8core set debug 40
10:41.44chrisM8then dial into your asteriks with clien and press some keys can you see anything in the console ?
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10:43.18MahmoudchrisM8, nothing seen
10:47.15Mahmoudboth info and  dtmf2833 aren't working
10:47.19Mahmoudtrying other phones
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10:51.20Mahmoudit works with some phones
10:51.43Mahmoudbut after pressing keys multiple number of times, the voicemail hangs and says nothing
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11:23.35mogiehye people
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11:41.33mogiehey
11:41.38mogiewith one telephone line
11:41.49mogiecan i have multiple people calling in?
11:41.57mogieor i need somethign special
11:42.33JTno
11:42.57mogieso i need something special to do it?
11:43.03JTmore than one line
11:43.19mogieok
11:43.25mogiehow does a call center work?
11:43.34mogiedo they have heaps of line coming in?
11:43.35JTthey have PRIs usually
11:43.39JTwith dozens of lines
11:44.14JTE1 ISDN PRI is available with between 10 and 30 channels per circuit
11:44.21JTand multiple circuits can be used
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11:44.48mogieok
11:44.57mogieis that how googles 411 works
11:45.49phlaxhi - im setting up a new tdm400 board with 2 fxo channels. I have modprobed zaptel/wctdm/wcfxo but get no indication in dmesg that the board has been found. I am starting to think the hardware is faulty. Can anyone suggest something else that i might try?
11:46.05JTmogie: i don't know, what is google's 411?
11:46.22mogieits a service from google
11:46.25mwallingJT: its an IVR that uses google local's search
11:46.51mogiedo they also use PRIs?
11:47.17mwallingmogie: my old rest had 6 PSTN lines. when line 1 was offhook, someoen calling line 1's number would be forwarded to line 2. if line 2 was busy it went to line 3... and so on
11:47.28mwallings/rest/restururant/
11:47.51mwallingwe had that configured through verizon
11:48.05JTmogie: they would use PRIs or wholesale VoIP termination from a major VoIP provider
11:48.41mogieok
11:48.57mogiewhat about out going calls
11:49.10mogieusing voip
11:49.55JTsure, it can be done
11:50.08mogieso the same thing
11:50.15JTless reliable than PRIs unless you have non-Internet data links direct to the VoIP provider
11:50.18mogiejust getting more PRI
11:50.38mogieya i think it would
11:51.11JTyou're looking at doing this in australia yes?
11:51.16mogieyes
11:59.21mogieif all lines are used up on PRI
11:59.32mogiewill the next call ingage?
12:00.40*** join/#asterisk ZaVoid (n=zavoid@66-95-182-90.client.dsl.net)
12:03.05ZaVoidmorning
12:03.29ZaVoidanyone know if the patch that fixed the realtime bug in 1.4.19 is in 1.4.19.1... i couldn't tell from change log
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12:09.33tzafrirZaVoid, svn diff http://svn.digium.com/svn/asterisk/tags/1.4.19 http://svn.digium.com/svn/asterisk/tags/1.4.19.1
12:09.53*** join/#asterisk koukai (n=cel@ABordeaux-256-1-143-178.w90-16.abo.wanadoo.fr)
12:10.59tzafrir'svn log http://svn.digium.com/svn/asterisk/tags/1.4.19.1' might also come in handy
12:12.52JTmogie: the next call will be engaged unless you have configured it at the telco side to redirect elsewhere
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12:27.34DavieyHey, PRI cat5 cable is just a normal cable, not X-over, right?
12:28.11JTthat really depends
12:28.34JTit's either straight through or T1 crossover, does that help? :)
12:28.42Davieyheh
12:28.51DavieyI'll throw a normal patch cable in, and see :)
12:28.54Davieythanks JT
12:29.14JTjust don't use an ethernet crossover cable
12:29.18JTit won't work ever :)
12:29.24Davieydoh
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12:35.18mknerdcan anyone speak for the reliablilty of the aa50 hardware/
12:35.20mknerd?
12:36.47ZaVoidthanks tzafrir
12:37.10DavieyJT: is there are reason an ethernet cable wouldn't work?
12:37.15Davieydifferent pairing or something?
12:37.20JTi said crossover ethernet
12:37.32tzafrirCrossover ethernet is not crossover PRI
12:38.21Davieyahh, a standard ethernet cable might work - if it doesn't require a crossover?
12:38.32florzDaviey: yep
12:38.33JTa T1 crossover, yes
12:38.51DavieyOk, thanks
12:39.42edugas_i'm searching for a way to display the caller id and the number called. anyone has done this?
12:39.57JTedugas_: on?
12:40.56edugas_asterisk, of course
12:41.45JTor
12:41.47JTerr
12:41.52[TK]D-Fenderedugas_: At one point I heard CPID might make it into 1.6.  Go check the featurelist
12:41.54JTyou want what displayed where?
12:41.58ZaVoidahh i see from Qwell This fix is not in 1.4.19.1. You'll need to use svn branch 1.4, or wait for 1.4.20.
12:42.13ZaVoidanyone know how far out 1.4.20 is ?
12:43.28edugas_i want the caller id and what number he called (ie: xxx-xxx-xxxx is 8-5pm support xxx-xxx-yyyy is 24h)
12:43.42edugas_[TK]D-Fender: thanks i'll take a look
12:43.53[TK]D-Fenderedugas_: Then override his CID name with the number he called
12:44.14[TK]D-Fenderedugas_: Sorry I tried to interpret what you were trying to get at and it wasn't CPID
12:44.16*** part/#asterisk mogie (n=opera@ppp121-44-209-93.lns1.hba1.internode.on.net)
12:44.23edugas_oh heh
12:44.24JTedugas_: that's easy
12:44.32JTmodify callerid name or number
12:45.29edugas_so i can't display the callerid name+number+number he called?
12:45.43JTwhy not? you're limited by display space
12:45.49JTjust modify the function
12:46.22edugas_yeah. 3 lines should fit on Cisco 7940's display
12:46.45JTi have 7940s with stuff like SUPPORT <callerid number>
12:48.40[TK]D-Fenderedugas_: Dependsing on the phone you might be able to wrap the text if you can space it out enough.
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12:54.05Davieyhttp://daviey.mooo.com/~dave/pri.jpg <-- does this look like a crossover PRI cable?
12:56.14jasonwootsure
12:56.32Davieyjasonwoot: Thanks!
12:58.28DavieySo i guess the other 7 of you agree :)
12:59.06florzNope, those are obviously just two plugs, no cable whatsoever =:->
12:59.21Davieyheh
13:03.38MrTelephonei agree and i havn't even seen it yet
13:04.00MrTelephonefinally got the visa paid off
13:04.47JTDaviey: are you sure you even need a crossover?
13:05.02MrTelephone15 business days is a long time to wait for a package from united states isn't it?
13:05.25JTit is, that is practically sea freight
13:05.40DavieyJT: no idea :(
13:05.49*** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net)
13:05.57[TK]D-FenderMrTelephone: Depedns where its going, what it is, customs, gravitational constant of the univers, planetary alighnment, and neutron decay...
13:08.09JTDaviey: that picture you have is NOT of a T1 crossover cable
13:08.16JTignore all these people saying it is
13:08.58MrTelephoneits 2 xeon 604 chips from i dunno
13:08.59MrTelephoneone sec
13:09.32[TK]D-FenderDaviey: http://www.google.ca/search?hl=en&q=t1+crossover+cable&btnG=Google+Search&meta=
13:09.32MrTelephoneif i don't get them in im getting ebay to cancel my account
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13:09.33*** mode/#asterisk [+o lmadsen] by ChanServ
13:10.00[TK]D-FenderDaviey: Its not like its a big picture in the FIRST LINK and even viewable IN the search results or anything.....
13:10.18JTDaviey: 1 to 4, 2 to 5
13:10.23JTthat is a T1 crossover cable
13:10.36jasonwootcable's a cable
13:10.55JTpin 1 and 2 must be a pair, pin 4 and 5 must be a pair
13:10.57JTthat is all
13:10.57jasonwoot1 to 4, a to 16, its all good
13:11.40MrTelephonewhen the carrier counter increases is that a problem with the cable to the t1 dsu?
13:16.33*** part/#asterisk MrTelephone (n=test@S010600195b946fa3.ls.shawcable.net)
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13:18.48eric2any reason why the voice quality would be poor sounding as if the person was taking through a microphone from space like the nasa astronauts sound?  Using ulaw...
13:20.15ZaVoidwhat device?
13:21.30eric2linksys 2102's
13:21.46eric2over a cable connection
13:21.51eric25M/800k
13:21.53ZaVoidwhats the far end device/endpoint?
13:22.04eric2what do you mean?
13:22.09ZaVoidwhere you calling to
13:22.13eric2local call
13:22.18JTa linksys 2102 is not a phone
13:22.22JTpots?
13:22.28ZaVoidits an ata that a handset plugs into though
13:22.37JTyes, i realise :)
13:22.57eric2from a pots line to the user... server is 30ms from the end user with the 2102
13:23.13*** join/#asterisk Iamnacho (i=Iamnacho@ip68-103-153-140.ks.ok.cox.net)
13:23.28eric2and I've noticed it myself as I also have a 2102 linksys device
13:23.33eric2ulaw once again
13:23.49ZaVoiddifference with any different codec?
13:24.00JTPOTS <---> Asterisk <---> Internet <---> SPA-2102 ?
13:24.32eric2yes, JT
13:24.44eric2that's the way it's going
13:24.57eric2different codec? I'll try some tests with 729
13:25.10JTdo you experience it with not going from POTS?
13:25.10eric2but it even happens on the outbound calls
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13:25.23eric2when not going through pots, it's fine
13:25.29JTdoes the audio sound bad in both directions?
13:25.30JToh
13:25.35eric2ya
13:25.36JThow are you connecting to pots?
13:25.55eric2I have a 2 way sip channels from a clec
13:26.10JTso you're not connecting to POTS
13:26.13JTPSTN
13:26.28eric2err.. ya.. have a virtual pri
13:26.36eric2so pstn I guess
13:26.36JTthat's not POTS
13:26.46JTbut that definitely sounds like where the issue is
13:26.46eric2sometimes I wish it was  :)
13:26.53eric2so with the pstn provider?
13:26.59JTnah, PRI is way better than POTS :)
13:27.10eric2aka... my virtual pri provider..?
13:27.19JTeither with the provider or the link to the provider
13:27.41JTobviously if the issue doesn't exist when you take the provider out of the equation...
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13:28.17eric2true
13:28.28eric2maybe it's the link...
13:28.44eric2but definitely the problem is tied to the provider at some level
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13:29.43ZaVoidmost likely
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13:40.39jasonwootno freenode PHP channel?
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13:40.57lmadsen##php
13:41.08lmadsen(that ## is not a typo)
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13:45.01adeelhmmm...gotta love upgrading from 1.4.18->1.4.19 causes segfaults
13:45.27lmadsenis this on your test box?
13:45.37jasonwootthat's what I get for doing a slash-list and trying to find it...
13:46.13adeeleric2, what connectivity do you have?
13:46.23lmadsenadeel: since obviously you don't upgrade your production system without testing on a test server, you could get a backtrace and file a bug
13:46.45mwallingjasonwoot: /msg chanserv list > /list (most of the time)
13:46.51eric2adeel  what do you mean connectivity?
13:46.58eric2connectivity from/to what?
13:47.08adeellmadsen, yeah, i will...i need to recompile though as i had set some optimization stuff
13:47.17adeeleric2, nevermind, i scrolled up and saw you were on cable
13:47.41Kobazso, when dialing out a pri, and the other party connects... many times the first 1/2 second of the audio is chopped
13:48.04eric2I've noticed that.. like when the system is bridging the call
13:48.37adeelKobaz, i notice that even on a pure SIP call
13:49.10adeelKobaz, i think it's somehow inherent in asterisk....
13:50.05adeellmadsen, the weird thing, though, was that asterisk would start up fine, and seem to be working normally, but issuing a reload would immediately cause a coredump
13:50.23lmadsenadeel: good news is that you can reproduce easily!
13:50.57adeellmadsen, hehe, yeah...i wonder if the dtmf settings i fiddled with had anything to do with it....i highly doubt it, but still curious
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14:06.32eric2adeel, what version are you running?
14:06.48*** join/#asterisk CunningPike (n=arodgers@204.239.12.183)
14:11.57jackson__Anyone here use an Adit 600 channel bank with Asterisk?
14:12.17[TK]D-Fenderjackson__: Plenty of people.
14:12.49jackson__[TK]D-Fender, I'm looking for some documentation on it, and wondering if those people might have a pdf.
14:14.05*** join/#asterisk grandpapadot (n=anonymou@mail.heavylogic.com)
14:14.29ZaVoidhttp://www.carrieraccess.com/products/index.cfm/fa/adit600.htm that?
14:14.34*** join/#asterisk joshaidan (n=joshaida@S01060090f8009fa6.tb.shawcable.net)
14:15.09[TK]D-Fenderjackson__: http://lists.digium.com/pipermail/asterisk-users/2004-May/040023.html
14:18.49jackson__Oh, looks like CarrierAccess did open up so accounts are now required for access to manuals etc.  Thanks fellas.
14:18.58jackson__er now not required
14:22.08*** join/#asterisk curtn (n=curtis@cl-451.trn-01.it.sixxs.net)
14:22.09grandpapadotGreetings, mighty baud warriors!
14:22.25curtnhi all
14:22.26ZaVoidyerwelcome jackson__ :)
14:22.35ZaVoidwhatcha using it for btw?
14:23.11jackson__ZaVoid, just looking at posibilities for use.  It was used with the outgoing hybrid key system.
14:23.25ZaVoidah
14:23.26jackson__it's populated with 3 8 port fxs cards.
14:23.37ZaVoidi thought i saw ip transcoding on the doc somehwere
14:24.01*** join/#asterisk inv_arp (n=junya@c-76-26-24-197.hsd1.fl.comcast.net)
14:24.06ZaVoidanyone know if the tc400b card does g723r63 as well as r53 yet?
14:24.14Kobazadeel: hmm
14:25.47Kobazadeel: the weird thing, is it seems to depend on the other provider whether the first part of the audio is missed or not
14:26.03*** join/#asterisk isamar (i=1000@voice.maxirede.net)
14:26.07Kobazadeel: if i call a verizon cell phone and it goes to voicemail, it will *always* drop the first 1/2 second of audio
14:26.10isamarhi folks
14:26.44*** join/#asterisk ddunavant (n=David@75.145.240.14)
14:27.22*** part/#asterisk isamar (i=1000@voice.maxirede.net)
14:27.45adeelKobaz, i think it has to do with the way asterisk sets up a call...iirc, asterisk waits to hand off the media stream until it receives some RTP packets
14:27.59Kobazyeah
14:28.01Mahmoudchan_gtalk is intresting
14:29.05adeeli never really got why people get channel banks....considering the cost per port, it's almost the same price as getting a SIP phone
14:29.22Kobazadeel: if i wasn't so busy doing my sysadmin things i would be fixing asterisk bugs
14:29.29adeeloutside of supporting fax
14:29.30Kobazadeel: what about legacy wireing?
14:29.59adeelKobaz, you can run ethernet over it...not the fastest speeds, but it's VoIP, you don't need more than a 10mbit ethernet link
14:30.37Kobazadeel: what about those 3000 foot runs of cat3
14:30.58adeelas i said, you can run ethernet over it
14:31.16Kobazmax ethernet distance is 100 meters
14:31.44[TK]D-FenderKobaz: repeaters/switches
14:31.58[TK]D-FenderKobaz: And you can get amplifiers as well that push that.
14:31.58adeelKobaz, i think you'd be hard pressed to find a building where its not wired for Cat5 as well as Cat3
14:32.12adeel3000 foot runs is more than half a mile by the way
14:32.14[TK]D-Fendercat3 = total waste
14:32.32Mahmoudno one uses amplifiers these days, except old dusty networks in afghanistan caves
14:32.34*** join/#asterisk ajohnson (n=ajohnson@63.147.46.186)
14:32.47ZaVoidafghan caves have goat's with sat dishes on them now :)
14:32.57Kobazadeel: i've seen some hack jobs where you do get those sorts of distances
14:33.00adeeli can't think of a single install that would have 3000 foot Cat3 running except for maybe old telco wires
14:34.09adeelKobaz, i have a client in san fran who has 1500 feet of cat3 running a total distance of 300 feet, but that's just because he bought a ton of cat3 cable and never bothered to cut it...
14:34.47Kobazhaha
14:35.32adeelhe wanted to get 20 fxs ports for his old telephones, and i told him it's a waste....for an extra 10 bucks a phone i could get him the new Polycom phones
14:38.40SwKincase anyone is interested 2600+ Ratecenters, $2 FlatRate incoming DIDs, Minimum 50 DIDs... email me krice@rmktek.com or msg me for details </spam>
14:39.53MahmoudKobaz, they are connected to YouTube to distribute their media :-P
14:41.09adeelwhere was the beginning of that <spam> tag?
14:41.29SwKooops I forgot it
14:41.37adeelhahah
14:42.18adeelSwK, that's actually pretty pricey for DID's...i'm getting them at around $0.25 for most LATA's
14:42.47SwKadeel, you arent getting them at that rate with no incoming usage charges :P
14:43.01adeelSwK, well it's not 2600 rate centers, that's the only catch
14:43.12*** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net)
14:43.38Mahmoudany one here uses chan_gtalk? How can I allow gtalk callers to access voice menu for example?
14:44.33SwKadeel, the only way you are getting them flat rate and that cheap is if you are getitng them rlecs and they are making any money off of cabs... otherwise lemme know where you are getting them from canse i could use a stack of them unless there are other charges like trunking charges to go along with it
14:45.38Kobazswk: i have some pris from one communications, i'm paying 50 cents a did with no incomming charges
14:45.56SwKKobaz, you are paying for PRIs tho
14:46.18Kobazyeah
14:46.29Kobazwhat are you using for transport
14:46.31SwKsee there is no PRI charges or anything like that with these DIDs
14:47.35SwKthese are DIDs from a Tier1 RBOC with a national presence... and my data center is also a teir1 facility
14:49.10Kobazsounds good, but how are we going to get voice traffic from your data center to ours
14:49.18SwKSIP
14:49.48intralanmanSwK: can you do tls/srtp?
14:49.58SwKintralanman, yes I can :P
14:50.02intralanman:-D
14:51.57adeelwhat's it take to setup tls/srtp?
14:52.00SwKKobaz, and have interoped with a stack of SIP based PBXs and SoftSwitchs from OpenSource to commercial... from Asterisk, OpenSER, FreeSwitch to Sonus, VEraz, Metaswitch, CopperCom, AcmePacket etc etc etc
14:52.55SwKadeel, do you have a softswitch/PBX that supprts it?
14:53.17adeeli was referring to setting it up on asterisk
14:53.52SwKI havent tried... our softswitch supports it and we've used it with polycoms, snoms, and a few other UACs
14:53.58adeelnot sure if the rlec supports it or not though
14:57.04SwKan I think at this time TLS is in the works for asteirsk w/ 1.6.. i havent had a chance to test it... looks like SRTP is still on the plate for later
14:57.21adeeleh, simplest way is just to run it through openvpn
14:57.33Kobazopenvpn > *
14:57.52Kobazadeel: but disable compression
14:58.14*** join/#asterisk Idle (n=brian@S010600a024969312.ed.shawcable.net)
14:58.34intralanmanoh, do a lot of providers offer openvpn support?
14:58.39Kobazno
14:59.09adeelintralanman, mine does
14:59.16adeelat least they're rolling it out soon
14:59.41IdleI have a quandry about planning a setup. We have a client with a nortel ICS, seems to work fine but they want some features that asterisk can handle easily, such as unified messaging/voicemail to email, and remote phone calls via voip, etc
14:59.54*** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla)
14:59.54*** mode/#asterisk [+o russellb] by ChanServ
15:00.18Idlewould the best way to set it up be to put in a dual T1 card in the asterisk box, then put asterisk inline? and if so, how is transfers to alternate extensions handled?
15:02.44*** join/#asterisk a-s (n=user@89.38.174.194)
15:02.53a-shello
15:03.16a-scan somebody help me with some suggestions in order to make speex work in asterisk, please?
15:03.49a-sI have installed speex, `show translations` show it present, however it does not work
15:04.10adeelIdle, i'm not sure on your exact setup, but you wouldn't necessarily need to take the T1 into asterisk if it's being terminated by something else...and as far as setting up asterisk, you could do the nortel as a trunk and have dialing rules to match
15:05.21a-s" chan_sip.c:5256 process_sdp: No compatible codecs, not accepting this offer!"
15:05.22MahmoudIs chan_gtalk able to answer gtalk calls? or is it just initiating calls only?
15:05.36adeela-s, if i may ask, why use speex? i've found speex to be pretty resource intensive compared to others that give comparable compression
15:06.20seanbright'wrap up time' is only respected across queues when the agent channel is used, correct?
15:06.24a-sadeel: I have a phone that uses speex, and I want to make it work
15:06.35Idleadeel: so, add another t1 card to the nortel, then add a single t1 to the asterisk box?
15:06.51*** part/#asterisk intralanman (n=lanman@209.85.58.2)
15:06.56a-sbut when I call from one ohone to the other, asterisk do not translate.
15:07.02a-seven the module is loaded
15:08.18adeelIdle, ohh, i understand your setup now....hmmm...what are the capabilities of the nortel? can it terminate to different types of phones, or just digital?
15:08.36Idlejust digital, as it is now
15:08.42a-sit workes, finally :)
15:08.48kingsobI get this error when i log into the gui, "GUI does not have write access!! Please check manager.conf"  but i have this line in my manager.conf file "write = system,call,log,verbose,command,agent,config"
15:08.56Idlethey mainly need more voicemail inboxes, and getting them via email would be perfect
15:09.06Idlebut, they have been quoted at 15k for 40 voicemail boxes
15:09.10Idlewhich, is disgusting
15:09.15kingsobdo i need to set the permissions on these files to something other than root/root?
15:09.30a-sI hear nothing
15:09.55adeelkingsob, you need to have asterisk config files owned by the user asterisk is running as
15:10.41adeelIdle, hmmm, so you'll need a t1 card in asterisk, and just setup asterisk as an extension of nortel, and just have the dial plan of asterisk just be straight to voicemail
15:10.56kingsobshould work then, im running asterisk as root for testing
15:11.23kingsobill just chmod 777 them and see if that solves it
15:12.11adeelkingsob, 777 is typically bad....644 is typically best
15:13.21kingsobsame error.. in the asterisk console i see   Unable to write /etc/asterisk/http.conf (Permission denied)  == Parsing '/etc/asterisk/http.conf': Found
15:13.36Idleadeel: but making outbound calls
15:13.43Idleand arn't nortel T1 cards retarded expensive?
15:13.51Idlefor a ICS that is
15:14.27adeelIdle, well it all depends on what you're trying to do with *...if you just want to use it as a voicemail server, then you don't need to do much
15:14.53adeelIdle, but if you're planning on having users register with * and act as a normal pbx, then you need to setup the nortel as a trunk
15:15.05adeelas far as asterisk cares, the nortel is just a carrier
15:15.27Idlethey want to basicly just use the phones on the nortel merely as phones, with the ability to transfer to extensions, then have virtual extensions, as well as have asterisk do a frontend 'press 1 for blah, press 2 for the company directory'
15:15.48*** join/#asterisk NDT (n=xwebs@cpe-74-70-217-136.nycap.res.rr.com)
15:17.37adeelIdle, well i'm not well versed on nortel, but all you need to do is connect the nortel to asterisk...and then jsut do your logic
15:17.42mikkelIs anyone using mISDN at all, it crashes my server (CentOS 5 and mISDN 1.1.7.2) ?
15:17.46adeelor you could put asterisk in front of nortel, but then it gets a little messier
15:18.23Idleadeel: well
15:18.36Idlethe nortel cannot do any of the things they want, aside from making phone calls
15:18.51Idlethe idea was to put asterisk in front, and have the nortel only handle their existing phones
15:19.05Idlethen asterisk could handle all of the logic, voicemail, etc
15:20.19*** join/#asterisk nny_1 (n=Scott@64.20.159.192.dyn-e-pool32.pool.hargray.net)
15:23.31*** join/#asterisk l2cache (n=chatzill@97.101.178.117)
15:23.46l2cacheWhat is the best open source dialer software for asterisk?
15:27.46*** join/#asterisk sun_moon (n=RaviRaja@ip72-206-113-190.om.om.cox.net)
15:27.58*** part/#asterisk sun_moon (n=RaviRaja@ip72-206-113-190.om.om.cox.net)
15:28.42*** part/#asterisk jivco (n=jivco@85.187.217.6)
15:29.12*** join/#asterisk nighty^ (n=nighty@p4069-adsau17honb13-acca.tokyo.ocn.ne.jp)
15:30.49nny_1What's the best way to have multiple separate extensions.conf on one running version of asterisk? I am going to be setting up some small test companies on one box, and have the incoming context for each be separate. I am wondering if it is possible to have 2 dialplans running in parallel on a box.
15:31.09nny_1maybe each company uses a set of unique contexts?
15:31.37adeelnny_1, yes, you can do that
15:33.01*** join/#asterisk xenonex (n=xenonex@92.47.0.186)
15:33.06nny_1adeel: should i just use a unique naming scheme for each companies series of contexts? (i.e. [local-companya] [local-companyb] etc. I am going for the maximum degree of separation
15:33.07*** join/#asterisk wordzilla (n=me@d122-109-83-20.sbr12.nsw.optusnet.com.au)
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15:37.30l2cacheWhat is the best open source dialer software for asterisk?
15:37.45adeelnny_1, yeah, more or less...
15:38.37nny_1adeel: ok, if there is a more efficient or ideal way, feel free to let me know.. this is more or less me feeling my way around
15:38.48nny_1can I use a global variable in a context name?
15:38.54Qwellno
15:39.15nny_1heh understandable
15:39.44nny_1hmm
15:40.53nny_1I am slicing up a pbx for a company with many small companies and looking to make the process a little easier.. any way duplicate the contexts, but make each unique? Guess the only way is to manually name each context A B etc
15:42.30l2cacheI would have extensions.conf just contain the company information, then use and include to keep each separate company's dialplan in their own file.
15:42.51l2cacheuse includes to a file for each company's dialplan
15:43.28nny_1l2cache: yeah i am going to have my incoming going to each [incoming-A] [incoming-B] etc, and then have an #includes ./companyA/extensions.conf etc
15:43.34nny_1same for sip.conf and voicemail.conf
15:44.22nny_1hmm I am just testing this out now, gonna barrel forward and see where the issues come about
15:48.59l2cacheI dont see any issues.  It is the equivalent of having all of the info in one file.  Just segmenting the same information into multiple files with includes
15:49.28l2cacheDoes anyone have experience with asterisk-based dialers?
15:49.51nny_1yeah i haven't thought about how to separate sip.conf (vm is easy) to allow multiple extension 100 etc
15:50.10l2cacheYou wont be able to have the same extensions registered.
15:50.23l2cacheJust do 100 - 120 on one box.  And 200 - 220 on the other
15:50.37l2cachebox = sip.conf include
15:50.53nny_1hmmm that's what i thought
15:51.11nny_1wash oping though but yeah SIP/100 = SIP/100 no way around that
15:52.15*** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com)
15:54.05l2cacheDoes anyone know of a dialer solution where agents can use the phone only (no PC required)
15:54.17*** join/#asterisk hfb (n=hfb@pool-71-118-254-245.lsanca.dsl-w.verizon.net)
15:54.45Qwellerr, what?
15:55.15l2cacheA asterisk-based predictive dialer that allows the agent to log in, and work outbound leads, without logging on through a PC.
15:55.28l2cacheAll phone-based logins
15:56.48IdleQwell smells
15:57.06*** join/#asterisk RoyK (n=roy@c51008389.inet.catch.no)
15:57.12x86http://news.bbc.co.uk/2/hi/europe/7376919.stm
15:57.15x86shakes head
15:57.18*** join/#asterisk draygon (n=draygon@208.76.99.5)
15:58.11QwellIdle quits IRC.
15:58.23Idleyea, cept when I need asterisk help :P
15:58.37Idleso, asterisk in front of a nortel ICS, good idea?
16:01.05*** join/#asterisk cuco (n=cuco@bzq-79-183-133-62.red.bezeqint.net)
16:01.19cucotzafrir: ping
16:01.52nny_1so say you wanted to slice up asterisk 60 ways or more, i take it VM is the method at that point, for all kinds of reasons (restarting 1 system vs. 60 for example)...
16:02.11*** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca)
16:03.11l2cacheI have found that asterisk runs great in a VM
16:03.12tzafrircuco, hi!
16:03.22l2cacheI am running 2 on my MAC mini right now.. for business
16:03.30cucotzafrir: reply to icq
16:03.39cucomove to the left by... 90 cm?
16:10.14*** join/#asterisk stimpie (n=stimpie@84-104-6-10.cable.quicknet.nl)
16:12.24*** join/#asterisk jong2 (n=chatzill@65.100.10.89)
16:13.22nvrpunkquestion, are there any dependencies for ilbc to compile into asterisk?
16:13.32*** join/#asterisk deneenm (n=fullstop@neptune.saucontech.com)
16:13.37nvrpunkI have two boxes one has the codec no problem
16:13.45nvrpunkthe other won't seem to install the codec
16:14.02*** join/#asterisk klimonso (n=eddy@dxb-b13491.alshamil.net.ae)
16:14.10*** join/#asterisk fluff (n=dune@snowflake.fluffigt.net)
16:14.11deneenmI'm having a problem with AGI... are there any known problems with STREAM FILE ?
16:15.03klimonsolookin for help linking 4 pbx together, paid service is fine with me, please contact me
16:15.13[TK]D-Fenderdeneenm: Check the bug tracker.  Aside from that, don't go fishing. Pastebin your problem.
16:15.40deneenm[TK]D-Fender: sorry.  I'll gather more info and pastie it
16:16.00[TK]D-Fendernvrpunk: * no longer contains ILBC in its distribution.  there are new dependencies to link in the libraries to gain support.
16:16.09*** join/#asterisk mknerd (i=3f951603@gateway/web/ajax/mibbit.com/x-d5e344bd9732d706)
16:16.13[TK]D-Fender~pb
16:16.13jbot[~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
16:16.16[TK]D-Fender^^^^^^
16:16.33mknerdanyone here use spinx2 on a asterisk appliance?
16:16.36nvrpunk[TK]D-Fender, I used the script in asterisk/contrib/scripts  shouldn't that be enough?
16:16.37[TK]D-FenderIdle: No, burning your ICS is a good idea.
16:16.39nny_1l2cache: thanks for the input, off to tweak
16:16.43*** part/#asterisk nny_1 (n=Scott@64.20.159.192.dyn-e-pool32.pool.hargray.net)
16:16.46[TK]D-Fendernvrpunk: Show us the problem.
16:17.04l2cachenp
16:17.28nvrpunkshow core translation
16:17.30nvrpunkilbc    -
16:17.35nvrpunkall are -'s
16:17.40nvrpunkfor ilbc
16:17.53nvrpunkafter running the get ilbc script and recompiling
16:18.29Idle[TK]D-Fender: heh
16:18.33nvrpunk[TK]D-Fender, is there a list of dependencies anywhere for ilbc?
16:18.35Idlewell, they have it already
16:18.39Qwellnvrpunk: did you enable it in menuselect?
16:18.41Idletheyre not gonna replace all of their phones :P
16:19.07nvrpunkQwell, menuselect?  I am guessing not since i have no clue what that is
16:19.47*** join/#asterisk fakhir (i=a7ce8021@gateway/web/ajax/mibbit.com/x-0dd06e87ee0a060e)
16:20.05Idle[TK]D-Fender: it can be done tho...
16:20.28nvrpunkQwell, inside the asterisk/menuselect folder?
16:20.52[TK]D-FenderIdle: You can put * beside it, in front of it, on top of it, sprinkled over with cherries & whipped cream. Whatever.
16:20.55Idlethe real question tho, is when they transfer from a nortel phone to another nortel extension, does it actually have 2 channels on the trunk open?
16:22.29*** join/#asterisk nvrpunk_ (n=zomgobli@81.90.21.227)
16:22.40nvrpunk_how do I enable it in menuselect?
16:24.27lmadsenilbc code is now in asterisk-addons
16:24.27deneenmok, I'm back.  My problem is that I can not get any audio streamed using STREAM FILE.  Here's a log: http://www.pastie.org/189943
16:24.51lmadsendidn't read far enough back to give anything useful though
16:25.00nvrpunk_Qwell, thanks i got it
16:25.18lmadsenQwell: I like your hustle
16:25.24lmadsenQwell: that's why it was so hard to cut you
16:25.33Qwellyou want to cut me?
16:26.10l2cacheA asterisk-based predictive dialer that allows the agent to log in, and work outbound leads, without logging on through a PC.
16:26.19l2cacheAnybody know of one?
16:26.23nvrpunk_Qwell, now i get, codec_ilbc.c:50:30: error: ilbc/iLBC_encode.h: No such file or directory
16:26.23Qwelllmadsen: no it's not
16:26.39Qwellnvrpunk: then you did something wrong
16:26.40lmadsenQwell: seriously? I thought it got moved there after the license stuff?
16:26.47Qwellit was removed
16:26.52nvrpunk_Qwell, selected the codec in the translator
16:26.59lmadsenQwell: just removed altogether?
16:27.00nvrpunk_and ran ./configure
16:27.02Qwellyep
16:27.03nvrpunk_then make
16:27.09lmadsenQwell: so how do you get ilbc now then?
16:27.09Qwellnvrpunk: no script?
16:27.15Qwelllmadsen: UPGRADE.txt :D
16:27.27lmadsenQwell: pfffft... too logical
16:27.33nvrpunk_Qwell, script?  to make it?
16:28.21lmadsenQwell: I'll stick it on the list of docs to be written...
16:28.32Qwelllmadsen: hold that thought
16:28.40lmadsenholds
16:28.54Qwellthe world may change soon
16:30.41Idlehow are transfered calls handled over a t1? are 2 channels left open for the call? is it bounced back?
16:30.48nvrpunk_so why was ilbc taken out?
16:30.54*** join/#asterisk atis_home (n=chatzill@193.238.213.215)
16:30.54nvrpunk_fails to see the logic
16:31.34l2cacheDoes anyone know of a asterisk-dialer that will enable agents to log in via phone (no pc at all) and work outbound leads?  Vici Dial? GNU Dialer? etc.
16:31.56[TK]D-FenderIdle: unless your PBX can trigger a 2BCT the it always sits in between.
16:32.21Qwellnvrpunk: UPGRADE.txt
16:32.53Idle[TK]D-Fender: so there will be 2, ok :P
16:33.07nvrpunk_Qwell, will that answer why i have that error?
16:36.48deneenmso everyone else is happily streaming files through AGI?  :-)
16:39.34nvrpunk_ok now i get this error, anaFilter.c:98: error: redefinition of âanaFilterâ
16:39.37nvrpunk_:(
16:45.04adeeli swear i hate dtmf
16:45.22BCS-SatoriOther then AGI or SQL, is there a way to route calls based on CID, I'm talking about 100 numbers in a list.
16:45.37deneenmadeel: mine wouldn't work until I ran fxotune
16:45.45l2cacheDoes anyone know of a asterisk-dialer that will enable agents to log in via phone (no pc at all) and work outbound leads?  Vici Dial? GNU Dialer? etc.
16:45.52adeeldeneenm, well this is a pure sip connection, so need for fxotune
16:46.01*** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose)
16:54.45*** join/#asterisk CrashSys (n=kumba@azrael.crashsys.com)
16:55.06*** join/#asterisk atis_work (n=atis_wor@193.238.212.171)
17:01.03deneenmok, so I've managed to get even more debug information and I am truly stumped.  As far as I can tell, STREAM FILE just does not work.
17:01.05deneenmhttp://www.pastie.org/189971
17:01.33deneenmit thinks it plays it... but it never does!
17:01.53*** join/#asterisk harryv (n=harry@67-207-147-205.slicehost.net)
17:02.50harryvhi, i'm having troubles setting up a sip client: http://pastie.org/189974
17:03.00*** join/#asterisk cubber (n=ronny@static-74-41-185-190.br1.glv.ny.frontiernet.net)
17:03.28jjshoeis there a way to make polycoms log everytime someone tells them to reboot?
17:03.30deneenmharryv: what is with the extension name?
17:03.38cubberI have an asterisk 1.4 server set up on gentoo and I am trying to get a conference room set up with Meetme, do I need to add the zaptel use flag to get the ztdummy module working?
17:04.22harryvdeneenm: ?
17:04.40deneenmsorry the _99400701
17:04.42Qwellcubber: don't use portage
17:04.47Qwellnot for zaptel/asterisk, IMO
17:04.52adeelcubber, yes
17:05.03deneenmharryv: that is not your problem though
17:05.13adeelcubber, if you install the portage overlay, you can install asterisk/zaptel through portage
17:05.20adeelcubber, errr. voip overlay
17:05.39kingsobany ideas why im getting permission denied when i get a voicemail?
17:05.41kingsobhttp://rafb.net/p/HOyfdO23.html
17:06.01Qwellbecause you don't have permission to that path
17:07.24cubberI am using the voip overlay in portage
17:07.42cubberthe server runs perfectly I just want a conference room and never installed zaptel
17:07.48kingsobbut asterisk is running as root
17:08.09adeelcubber, you could use the app_conference rather than meetme, but if you want meetme, then you NEED zaptel
17:08.12harryvroot might not have write-permission.
17:08.21kingsobo
17:08.23cubberooo app_conference does not need zaptel?
17:08.25kingsobthats true, ill take a look
17:08.39cubberI would rather not install it if I don't need it.  Would rather keep the install as simple as possible
17:08.46deneenmi figured out my agi problem.  I wasn't waiting for the sound to finish streaming before sending the next command.
17:08.59adeelcubber, adding zaptel doesn't really add that much complexity...although it adds a few more packages though
17:09.15cubberhow do I configure app_conference?
17:11.07kingsobi manually created the folder it wanted
17:11.12kingsoband now i get
17:11.14kingsob<PROTECTED>
17:11.25Qwellagain
17:11.27kingsoband root has permissions
17:11.30Qwellyou don't have permission to that path
17:11.31jjshoekingsob what's hard to understand?
17:11.49gitguyit is ok to charge for trial job periods?
17:11.50jjshoekingsob what user does asterisk run as (waiting for your lightbulb)...
17:11.56gitguyor thats anti-ethic?
17:11.58kingsobroot
17:12.10kingsobwaiting for u to stop thinking im a total idiot
17:12.21cubberis app_conference a separate package I need to install from the voip overlay? Or do I just edit a config file?
17:12.22jjshoekingsob put the output of ps aux on a pastebin and link us.
17:12.39harryvps aux |grep asterisk && chmod 755 -R /var/spool/asterisk
17:12.47harryvor, skip that last part.
17:12.49jjshoekingsob then do an la -al on every dir leading up to that dir on pastebin.
17:13.14*** join/#asterisk JayTee52 (n=jforde@207-67-84-181.static.twtelecom.net)
17:13.30*** join/#asterisk gr0mit (n=tim@extrt.txrx.org.uk)
17:13.34jjshoeI'll quit my job and be Qwell's lawn boy if there's a bug regarding file perms in app_voicemail.
17:13.58Qwellmakes a small change to app_voicemail.c
17:14.32cubberlooks like its in the overlay, gonna emerge it
17:14.49jjshoe:D
17:14.57kingsobhttp://rafb.net/p/28nuPR63.html
17:15.06jjshoeQwell is it ok if I wear a bikini while I wash yo ride?
17:15.18Qwellkingsob: SO, asterisk isn't running as root.
17:15.39kingsoblol
17:15.41*** part/#asterisk MonkeyHugs (n=asdf@206.71.78.158)
17:15.43kingsobok so i am an idiot
17:15.48Qwellpretty much, yeah ;/
17:15.52jjshoekingsob isn't that what we were telling you?
17:16.02cubberdoes app_conference work with SIP or is it just IAX?  I tried to pull it down and got an error saying it can't fetch it from repository: https://svn.sourceforge.net/svnroot/iaxclient/trunk/app_conference
17:16.06Qwellkingsob: need to give the asterisk user perms to those dirs.
17:17.03QwellI think the book goes into detail about what needs to be done to run asterisk non-root
17:17.04Qwell~book
17:17.05jbotAsterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook
17:17.12l2cacheDoes anyone know of a asterisk-dialer that will enable agents to log in via phone (no pc at all) and work outbound leads?  Vici Dial? GNU Dialer? etc.
17:21.57kingsobi started it as root, i dunno why its running from a different user
17:22.18kingsobo nm, itsrighht in my command
17:22.19kingsoblol
17:22.19jjshoekingsob you shouldn't run it as root.
17:22.37adeelcubber, it should work with either sip/iax/zap/etc
17:23.16cubberadeel: ok it won't pull down the package from the voip overlay, looks like the link its trying to grab it from is either down or non-existant.  I will wait for an update I guess
17:23.34Qwellcubber: or, just don't install from portage
17:23.48*** join/#asterisk [hC] (n=hardcore@mail.metrobridge.com)
17:25.05*** join/#asterisk justdave (n=dave@unaffiliated/justdave)
17:26.13adeelcubber, app_conference should be installed
17:26.20jjshoeQwell wouldn't it be easier to have someone at digium make packages for all the common distros instead of yelling at newbs all day long?
17:26.29cubberadeel: installed by default in asterisk?
17:26.37Qwelljjshoe: nobody would use them.
17:26.43Qwellthe distros, that is
17:27.02*** join/#asterisk eXcAliBuR (n=dont@207.134.8.33)
17:27.02adeelcubber, nevermind, i thought it was but i don't see it...simplest thing is just to set the zaptel use flag
17:27.11*** join/#asterisk oej (n=olle@65.42.208.134)
17:27.14jjshoeQwell ? I'm saying make packages of zaptel + asterisk for redhat, centos, etc.
17:27.35cubberadeel: then it will use ztdummy for meetme?  I don't have any zaptel devices installed
17:27.45*** join/#asterisk thelodger (n=svf@67.106.246.227.ptr.us.xo.net)
17:27.45kingsoball rite, all is working now, thanks
17:28.02adeelcubber, yes
17:28.24adeelcubber, but you need to set the ztdummy useflag for the zaptel package
17:28.39adeeland honestly, i don't see the app_conference app anywhere in 1.4
17:29.51cubberadeel: i will give it a shot, looks like the meetme config is much simpler than the app_conference config anyway
17:31.35cubberadeel: bah the package net-libs/libpri is masked by missing keyword.  So I can't add zaptel
17:32.00fluffcubber: unmask it then
17:32.02Qwellyou don't need libpri for zaptel
17:32.13Qwelland seriously...for like the 4th time.  don't install from portage ;/
17:32.20cubberfluff: i did try to unmask it in package.unmask as well as gave it the ~x86 flag
17:33.08cubberqwell: I hear you but its already set up and running, and I don't feel like rebuilding it now.  Guess I will just hold out, don't really need the function right now anyway.
17:33.15fluffcubber: package.keywords are you looking for if you've got a reasonable up-to-date portage...
17:33.41fluffcubber: package.unmask isn't supposed to be used anymore i think.
17:33.49cubberfluff: the system is a fresh install from minimal as of last week using the 2008.0 beta cd
17:34.14cubberfluff: I had to use package.unmask in order to pull down asterisk 1.4 from voip overlay, otherwise I was stuck with 1.2 no matter what I tried
17:34.31Qwellwhat version of 1.4 is it stuck on?
17:35.17cubberI am using 1.4.19.1 from voip overlay that is what gets pulled in when I put net-misc/asterisk ~x86 in package.keywords and net-misc/asterisk in package.unmask
17:35.23[TK]D-Fendercubber: Stop inisting on using packaged * and then whining that you don't have control.  Thats exactly what you sold out on.
17:36.59gitguycan i have 90000-99999 conferences with MeetMe() by saying _9XXXX in meetme.conf and extensions.conf ?
17:37.00cubberD-Fender: I am not whining, I am just trying to figure this out.  I did not realize that I should not have used portage to install Asterisk 1.4.  I realize the limitations of using portage for this kind of thing, and also the advantages
17:37.21gitguyor create dynamic conferences
17:37.24gitguy...
17:37.40[TK]D-Fendercubber: So follow Qwell's advice and compile from source.
17:37.54cubberI will try that when I have more time, thanks for the info guys
17:38.01[TK]D-Fendergitguy: Yes, go read the book, and the applications instructions.
17:38.01*** part/#asterisk harryv (n=harry@67-207-147-205.slicehost.net)
17:38.12gitguy[TK]D-Fender: ok
17:38.34jasonwootyou guys should program the bot to say that
17:38.54[TK]D-Fender~rtfm
17:38.55jbotwell, rtfm is 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
17:38.58gr0mithehe
17:39.06jasonwootLoLoLoL
17:39.08[TK]D-Fenderjasonwoot: I do believe it does... I was just being kinder :)
17:39.17gr0mitstarted with packaged asterisk
17:39.24jong2auch
17:39.25gr0mitand then saw the light!
17:39.41jasonwootin the military, we said RTFM squared, as in "read the F'ing manual, twice"
17:39.46gr0mitcompiling from source is ALWAYS worth doing
17:40.06jong2read the friendly manual whatever you suit.
17:40.56jjshoeQwell what you said finally clicked :P the distro's wouldn't use the packages you made. hehe.
17:41.02cubberOk one last thing, portage does compile asterisk from source, it is just having an issue with one of the dependencies
17:41.14Qwelljjshoe: right - thus nobody else would
17:41.21Qwellbecause they would install from the distro still
17:41.25[TK]D-Fendergr0mit: If you're lucky, "seeing the light" doesn't come at the end of a 5 tera-watt laser :)
17:43.25gr0mitonly if people don;t use the source!
17:48.10*** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com)
17:52.03jeffspeff2I have my timebased rule defined in extensions.conf, what is the syntax for the voice menu to reference the time rule?
17:54.57*** join/#asterisk RoyK (n=roy@253.80-203-34.nextgentel.com)
17:57.34*** join/#asterisk CVirus (n=GoD@82.201.222.72)
18:00.39[TK]D-Fenderjeffits already in your dialplan, jsut call it where you need it.
18:07.55*** join/#asterisk schreckdebase (n=tpdc@koe67-2-82-238-198-119.fbx.proxad.net)
18:09.21*** join/#asterisk justdave (n=dave@unaffiliated/justdave)
18:16.30l2cacheDoes anyone know of a asterisk-dialer that will enable agents to log in via phone (no pc at all) and work outbound leads?  Vici Dial? GNU Dialer? etc.
18:17.05[TK]D-Fenderl2cache: go look on the WIKI listing.
18:17.14l2cachewhich wiki?
18:18.05l2cacheI need to know if anyone has experience with a similar dialer for asterisk.
18:18.46jeffspeff2[TK]D-Fender: but how do i call it?  exten=1,n,Goto(timebasedrules)   returns the following errors...   [May  1 13:16:09] NOTICE[10722]: pbx.c:1881 pbx_extension_helper: No such label 'timebasedrules' in extension 's' in context 'voicemenu-custom-5'
18:18.46jeffspeff2[May  1 13:16:09] WARNING[10722]: pbx.c:6381 ast_parseable_goto: Priority 'timebasedrules' must be a number > 0, or valid label
18:20.31[TK]D-Fenderjeffspeff2: what is "timebasedrules"?  Read the instructions for "goto".  You need to specify a priority, and possible an extension, and then context.
18:21.16[TK]D-Fenderjeffspeff2: PASTEBIN is your friend.
18:21.18[TK]D-Fender~pb
18:21.18jbot[~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste
18:25.19jeffspeff2timebasedrules is a context within my extensions.conf. It defines the time rules for inbound call routing.
18:25.31jeffspeff2http://pastebin.ca/1004155  look down at the bottom
18:26.19*** join/#asterisk SirThomas (n=tomc@mail.kendeco.com)
18:27.27*** join/#asterisk oej (n=olle@2.234.241.83.in-addr.dgcsystems.net)
18:28.16gitguyloves the asterisk music on hold
18:28.34unpaidbilli have the 2live crew for my moh
18:28.40unpaidbillthey love it at the monestary
18:29.19gitguy-- Started music on hold, class 'default', on SIP/diego-17f67490
18:29.33*** join/#asterisk sysreq` (n=sysreq@unaffiliated/sysreq)
18:29.56[TK]D-Fenderjeffspeff2: Looks like you want to INCLUDE the extens in that context into your other IVR.  You would then add "include => timebasedrules" into your IVR context and it will gain access to them.
18:30.35jeffspeff2ahh... thanks. that makes sense. lol
18:30.43gitguy[TK]D-Fender: yesterday i tried to "include => internal" but when i tried to dial a extension from internal into default it didn't work...
18:31.08[TK]D-Fendergitguy: And you're not showing me much and I don't trust much of what I don't see...
18:31.19gitguy[TK]D-Fender: 1min
18:31.28tzafrirputs gitguy on hold
18:31.32gitguy:S
18:31.49gitguymy exgf used to put me on hold :(
18:32.41x86<PROTECTED>
18:32.42x86shakes his head
18:33.02nvrpunk_I have a questions regarding connecting over satellite
18:33.19nvrpunk_our network is wireless, then shoots over a satellite to a teleport
18:33.29nvrpunk_I was thinking of trying ilbc
18:33.45nvrpunk_im looking for slightly better voice quality than g729
18:33.59nvrpunk_but ilbc transcoding is an addition 18ms
18:34.06nvrpunk_from ulaw internally
18:34.20nvrpunk_and ulaw is high overhead even for the wireless part
18:34.38nvrpunk_but i dont want to transcode between 729 and ilbc, and our IP phones dont support ilbc
18:34.46nvrpunk_neither does our pstn GW
18:34.53nvrpunk_so ill be transcoding two time
18:34.54nvrpunk_s
18:35.03tldnvrpunk_: Why do you want to try something else?
18:35.08tldnvrpunk_: What's your problem?
18:35.11coppiceilbc doesn't sound any better than G.729, unless the packet loss is high
18:35.11tldPacketloss?
18:35.12nvrpunk_than g729?
18:35.23nvrpunk_yeah
18:35.30nvrpunk_we are blasting over satellite
18:35.34nvrpunk_there will be packet loss
18:35.40nvrpunk_and latency
18:35.47tldsatellite gives latency, but doesn't have to give packet loss.
18:35.55tldI'd look into tuning the jitterbuffer first.
18:36.14tld18ms*2 might not be such a bad extra latency compared to the sat link itself though
18:36.37gitguy[TK]D-Fender: it works now, looks like i made something wrong yesterday...
18:36.53Qwellnvrpunk_: 18ms for 1 second of audio
18:37.01nvrpunk_tld, any good links for jitterbuffer tuning?
18:37.14nvrpunk_Qwell, yeah not a very high ratio
18:37.16eXcAliBuRdoes the asterisk gui work on 1.6 ?
18:37.51tldnvrpunk_: Which kinds of phones to you have?  Good place to start might be to just use phones that'll actually tell you what jitter they're seeing.
18:38.00tldnvrpunk_: Or phones with good auto-sizing jitterbuffers.
18:38.21tldnvrpunk_: Keep in mind that jitterbuffers basically replaces jitter with latency, so you might not want to be too aggressive either.
18:38.22nvrpunk_engenious, linksys, grandstreams, dlinks....
18:38.23nvrpunk_a lot
18:38.38tldnvrpunk_: Put some 7960s in?
18:38.52tldnvrpunk_: They'll teach you more about your network conditions, and are great phones by themselves as well
18:39.23nvrpunk_ah not a bad idea
18:39.33nvrpunk_yeah i have access to some of those actually :D
18:39.45tldneat
18:39.47tldgood to go then. :)
18:39.53nvrpunk_granted those are expensive phones
18:39.58nvrpunk_but the military buys them :)
18:40.05tldThey're way cheaper now than they used to be.
18:40.21tldI think the most important thing is figure out what your problem actually is, and where packets are lost.
18:40.27tldIf it's over the sat link, iLBC might help.
18:40.43tldIf it's because they arrive too late for the jitterbuffer, tweaking it might be a better choice.
18:40.54*** join/#asterisk CanWood (n=chatzill@24.108.64.80)
18:41.16nvrpunk_we have more jitter than packet loss
18:41.21tldAnyone can make a SIP phone, but the quality of the jitterbuffer is one of the things that separates the boys from the men. ;)
18:41.25nvrpunk_i monitor the network constantly
18:41.27tldThen better equipment
18:41.30tldnice
18:41.38tldI'd start with trying the 7960 thing
18:42.58CanWoodhey folks.  MixMonitor's v and V options, heard and spoken, respectively: which is which for external calls.  is heard always the extenal and spoken the asterisk side?
18:43.00tldIf you want to really get a gold star, you could get some AMD-WB (G722.2) phones, and try those on the thing.
18:43.08tldWideband is *very* nice, and it's fairly low bitrate.
18:43.12tldhttp://en.wikipedia.org/wiki/AMR-WB
18:43.20tldShould survive latency if the jitterbuffer is decent
18:44.17tldnvrpunk_: In theory, AMR-WB at 12.65 kbit/s gives you better quality than G722 at 56kbit/sec.
18:44.20tldCan't beat that? ;)
18:48.04nvrpunk_tld, what phones support that?
18:48.27tldnvrpunk_: Not sure. :/
18:48.44nvrpunk_i see mobiles
18:48.48nvrpunk_but no just voip
18:48.51nvrpunk_:/
18:49.10*** join/#asterisk mandd (n=dache@dsl-134-153.aei.ca)
18:49.28tldnvrpunk_: I know this does: http://www.tandberg.com/products/video_systems/tandberg_1700_mxp.jsp
18:49.42manddcan I force asterik to send Request REGISTER packets much more ofter?
18:50.02[TK]D-Fendermandd: Why would you want to do that?
18:50.03nvrpunk_often?
18:50.07nvrpunk_yeah
18:50.10nvrpunk_you could
18:50.12manddhello [TK]D-Fender
18:51.14manddbecause for some reason, after a while, my voip provider sends  Rqeust: CANCEL  when someone calls my number
18:51.28manddand asterisk isnt receiving that incmoing call
18:51.44manddhowever after  Request Register it works fine
18:51.50manddfor like 10-15 minutes
18:51.53manddrandom really
18:52.12[TK]D-Fendermandd: Sounds like something else...
18:52.28[TK]D-Fendermandd: * sends register attempts at the speed your provider requests
18:52.29manddwell, it started as a NAT issue
18:52.47[TK]D-Fendermandd: If you're running * behind NAT and didn't configure things right, then that would be a reason.
18:52.58[TK]D-Fendermandd: Sounds like one you didn't fix properly then.
18:53.00[TK]D-Fendermandd: ....
18:53.02[TK]D-Fender~sipnat
18:53.03jbot[~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions
18:53.04[TK]D-Fender^^^^
18:53.21mandd[TK]D-Fender i've been at it for a long while now
18:53.30[TK]D-Fendermandd: the pastebin it all up.
18:53.35manddliterally placing packets (using PF) into asterisk's hands
18:53.41[TK]D-Fendermandd: masking only passwords
18:53.42manddok, thanks, one minute
18:55.36*** join/#asterisk atis_work (n=atis_wor@193.238.212.171)
18:56.06[hC]anyone here use Netgear FS752TP or FS726TP switches with VLANs?
18:57.09CanWoodFMS726S here
18:59.01[hC]May or not still apply. You using 802.1q VLAN tagging?
18:59.32CanWoodrightnow it's on, but we just removed the Mitel phone system that was using it yesterday.  Our current phones are not tagging traffic
19:00.22[hC]In the web interface, do you set it up by clicking on all the port icons and setting some to "U" and some to "T"?
19:00.43[hC]Im asking because ive set up some switches that do it before, and some that still do, and upon setting up some new switches again last night, its not working
19:00.47[hC]yet i cant see any reason why not
19:00.56[hC]Thinking i might have missed something, im not sure.
19:02.52CanWoodI had all of the ports set to untagged for the data traffic and all set to tagged for the phone traffic, so it would honour the priority in the VLAN tag for phone packets
19:04.09[hC]yeah thats what im trying to do.. untagged vlan1, tagged vlan 20.. im gonna see if I can remote into another switch to compare configs
19:04.14[hC]makes no sense why its not working..
19:11.59*** join/#asterisk lanning (n=lanning@ip67-152-85-190.z85-152-67.customer.algx.net)
19:14.24gitguycvs/svn sucks
19:14.25gitguygit ftw
19:14.32tldhg ftw
19:14.42l2cachewhats better gnudialer or vicidial?
19:14.43manddhttp://pastebin.com/m327b1c02
19:14.45gitguygit ftw
19:14.53tldmercurial ftw
19:14.59gitguygit teh faster
19:15.05mandd[TK]D-Fender I put two wireshark versions, one where it works, and one where it doesnt
19:15.07tldhg teh beter
19:15.11manddie.  Before reload, after reload
19:15.21gitguytld: git is better :)
19:15.30Qwellgitguy: are you here for any reason besides git trolling?
19:15.38gitguyQwell: i'm not trolling
19:15.47Qwellyes or no?
19:15.55gitguyi'm just expressing my love for the best scm
19:16.58gitguywell, no... i'm an asterisk guy too
19:17.47unpaidbillyou just like git because linus does
19:17.51gitguyi made it work behind a nat, that makes me an asterisk guy right? :)
19:17.58gitguyunpaidbill: linus is cool too
19:18.47jeffspeff2[TK]D-Fender: hey, thanks for your help. i got the time macro working now. :)
19:18.58*** join/#asterisk P424D0X (n=DL7RAY@p57BCF923.dip.t-dialin.net)
19:19.14unpaidbillasterisknow is pissing me off so badly
19:19.34unpaidbilli highly recommend staying far, far away from it :(
19:19.34P424D0Xhow connect the cisco7940 to asterisk?!
19:19.42gitguyunpaidbill: use asterisk alone
19:19.53l2cacheget the SIP firmware for the phone. and setup a tftp server for provisioning
19:20.35jeffspeff2unpaidbill: it's not all that bad. well at least for noobs like myself.
19:20.58unpaidbillyeah well i normally do, i tried it out for an install a week ago and quickly ran into a billion bugs
19:20.58P424D0Xl2cache: i have, but the phone shows "invalid application protocol"
19:21.17*** join/#asterisk skyphyr (n=quassel@92-234-8-140.cable.ubr07.hari.blueyonder.co.uk)
19:21.24unpaidbillthe most frustrating is how the hell it dials users extensions... it's like some kind of magic trick that i cant figure out
19:22.28russellbthe GUI mainly manages users using users.conf
19:22.31skyphyrhi all - I've got a strange issue that google isn't helping me with much. I've just switched distro (dualboot to start with) and using the same config files on each, but the new one callers can't hear me
19:22.42russellbwhen you put a user in there, it automatically creates an extension for them
19:22.45skyphyrusing a wildcat fxo card
19:22.47russellbwhich calls the stdexten macro for their device
19:22.53skyphyralso when you dial in from another phone
19:22.55russellbthe stdexten macro can be seen in extensions.conf
19:23.01skyphyryou don't hear the ringing on the dialers end
19:23.26skyphyrwas running 1.2.21 new one is 1.2.27
19:23.35l2cachep424d0x: do you have the Cisco SIP firmware installed?
19:23.47skyphyrsorry was 1.2.21.1
19:24.46*** join/#asterisk zoid99 (n=chris@router.asteriasgi.com)
19:24.59P424D0Xl2cache: P003-08-8-00 - FW-Package is on the tftp-server..
19:26.48P424D0Xl2cache: And i've installed the SIPDefault.cnf, SEPxxxxx.cnf.xml and xmlDefault.CNF.XML
19:27.58l2cacheWhat are all of the errors on boot?  Is the phone hitting the tftp server?
19:28.52*** join/#asterisk klictel (n=klictel@atelka.info)
19:29.50gitguyunpaidbill: no, seriously... i been using svn for like, 6 months and the fact that i can do branching, merging, commiting, and everything else in my local computer, is great... and then push/pull/clone from anywhere, part of why i like git are those reasons
19:30.46P424D0Xl2cache: i can read in the logfile, that dhcpd is given a ip-number to the phone.. but nothing to read about tftpd.. :-/
19:31.09nvrpunk_question, if I have a user and a peer, and I set a jitterbuffer is it the user who gets the buffer always?
19:31.52*** join/#asterisk xenonex (n=xenonex@89.218.236.42)
19:32.35skyphyrok downgrading to 1.2.21.1 made no difference
19:33.52l2cachep424d0x: What errors do you get on the phone when it boots?
19:34.03gitguyunpaidbill: linus is just another reason why i like git too ;)
19:35.51skyphyrsorry it's an fxs not an fxo
19:35.58skyphyruses fxo signaling for fxs :)
19:37.07*** join/#asterisk RoyK (n=roy@253.80-203-34.nextgentel.com)
19:38.29*** join/#asterisk putnopvut (n=putnopvu@216.207.245.1)
19:38.29*** mode/#asterisk [+o putnopvut] by ChanServ
19:42.01C4colodo the zap drivers detect hand-crank ring signal?
19:42.39C4colowould an old hand-crank phone need to be connected to an FXO instead of an FXS?
19:43.01[TK]D-FenderC4colo: if hand crack is a stable-timed AC induction of the appropriate range I don't see why not.
19:43.08[TK]D-Fendercrank*
19:43.40C4coloand that port pointed to a context that connects since there would be no method for dialing, well maybe pulse by clicking the switch hook
19:43.58C4coloor voice recognition!
19:43.59C4colohaha
19:50.00*** join/#asterisk hi365 (n=hi365@bzq-219-141-66.static.bezeqint.net)
19:50.20JayTee52is there a way to add the time to the console output for each of the calls being placed? is this a higher level of verbosity?
19:50.37*** join/#asterisk talntid (n=t@66.208.251.170)
19:53.02*** join/#asterisk critch (n=critch@c-71-228-211-57.hsd1.tn.comcast.net)
19:54.01*** join/#asterisk CrashHD (n=CrashHD@65.74.161.225)
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20:00.43jeffspeff2what are the going rates for an asterisk server setup with voip, ip phones, voicemail, ivr, a few conference rooms, and around 10 or so users?
20:01.55JayTee52lebenty-leven buck o' five
20:02.45Qwell$X + $Y + ($Z * 10)
20:03.02Qwellwhere X is the price of the server, Y is the cost of the consulting time (if any), and Z is the cost of the phones
20:04.14M1s3ryjeffspeff2, they really aren't kidding :/
20:05.07critchnot directly an asterisk question, but any suggestions for a low cost toll free provider for 200k minutes a month or more.
20:05.15draygonanyone here use route5060.com?
20:05.23adeelcritch, define low cost
20:05.32critchlower the better
20:06.12jeffspeff2i'm new to all of this, have been playing with asterisknow, i'm sure there are several small business in my area that would love to have a more features with their phone system. i'm not sure how to price the server and setup.
20:06.22*** join/#asterisk jyap (n=jyap@66.135.230.101)
20:07.00adeeljeffspeff2, you'll probably be spending a decent amount of time with support, as setup and install isn't too long
20:07.06adeelonce you get the hang of it
20:08.40jyapis it normal for the command 'sip show channels' to show a value 25 like but 'module show like chan_sip' to show something really high like 441 as the use count?
20:10.16critchjyap: how many registered users?
20:10.39jyapmany..  but through openser
20:10.59adeeljyap, sip show channels will tell you how many active sip conversations is going on
20:11.08adeeljyap, that takes into accounts registrations, options, etc
20:11.54adeeljyap, on a box that has only 10 extensions, i have a USE count of 222
20:12.25*** join/#asterisk IPGHOST (n=imrouter@116.58.16.175)
20:12.31jyapi see.  do you know what the use count refers to?
20:12.56adeelno idea....maybe how many times the channel was invoked?
20:13.10adeelsince last reload/reset ?
20:13.37jyapit seems to slowly climb up.  was wondering if it was some leak.
20:15.35adeelinteresting, on one box the counts at 222, the other it's at 0
20:15.39unpaidbillanyone here tried out the snom m3
20:16.11jyapunpaidbill: yes
20:16.21adeeljyap, it probably has something to do with the number of sip messages in a session
20:16.22unpaidbillhow is the range/reception on that thing?
20:17.15jyapunpaidbill, audio quality is good.  range can differ.  kinda flaky.
20:18.21unpaidbillyeah range is always a pain, says 50-100 meters indoor/outdoor on the site, sounds pretty decent.
20:18.35jyapso you've tried it?
20:18.42unpaidbillno, im readign about it right now
20:18.58*** part/#asterisk critch (n=critch@c-71-228-211-57.hsd1.tn.comcast.net)
20:19.29jyapthat range is under ideal conditions.  it varies.  i think it's very prone to interference
20:19.47*** join/#asterisk hunmonk (n=hunmonk@drupal.org/user/22079/view)
20:19.48unpaidbillyeah, and dect was supposed to alleviate interference i thought :(
20:20.26adeeli keep seeing __ast_smoother_feed: Dropping extra frame of G.729 since we already have a VAD frame at the end......in my logs...are there options i can pass to supress the VAD?
20:24.17DarKnesS_WolFanyone running asterisk + zaptel with openBSD?
20:24.27*** join/#asterisk Maveric (n=maveric@mavbox.gotskill.net)
20:24.43*** join/#asterisk kyron (n=kyron@modemcable086.140-70-69.static.videotron.ca)
20:24.50DarKnesS_WolFadeel: ah i recall an option but really not sure that was long time ago
20:24.54*** join/#asterisk theHub (n=theHub@69.177.93.21)
20:25.05DarKnesS_WolFunpaidbill: yes i have M3 they are cool !
20:25.13kyronAhoy, anyone here has had experience with "[HBB1] Broadband Booster" (traffic shaping) ?
20:28.37Maverici need to bridge meetme confernces across asterisk servers i know this can be done with call files but the issue i see is once they are bridged they will continue to be bridged until manual intervention  does anyone know of a cleaner way of doing this?
20:29.13*** join/#asterisk funxion (n=x@63.214.236.169)
20:30.15funxionI'm having prroblems with the codec_g729.so file that I used from digium ftp after adding licenses
20:30.16*** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq)
20:30.22*** join/#asterisk harryv (n=harry@67-207-147-205.slicehost.net)
20:30.23harryv[May  1 22:28:59] WARNING[7892]: app_dial.c:743 wait_for_answer: Unable to forward voice or dtmf
20:30.28harryvwhat would this mean?
20:31.12funxionharryv what version are you using
20:31.40funxionand when are you getting it
20:32.14harryv1.4.18 and i'm calling some AGI-script from a local sip phone. must have changed something, was working just before
20:32.44funxionwhat did you change
20:33.56hunmonkis there any way to disable the mailbox options for voicemail?  as in, the user is given no options to do greetings, change their password, etc
20:35.23funxionare you using realtime?
20:35.29hunmonkfunxion: no
20:37.31funxionI beleive you can turn the options off in voicemail.conf
20:38.31adeelis it possible to redirect the rtcp output to its own log file?
20:38.38adeelwithout hacking the source
20:40.37funxionhas anyone in here adding g729 licenses on a box with a xeon processor
20:40.38funxion?
20:41.13DarKnesS_WolFhunmonk: i make it like exten => 555,1,VoiceMailMain(${EXTEN}@${CONTEXT})
20:41.38DarKnesS_WolFhunmonk: or it was ${ARG1}don't recall :-s
20:41.56hunmonkDarKnesS_WolF: i mean more when they are in the admin section of their voicemail
20:42.06funxionhttp://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+VoiceMailMain
20:42.06hunmonkDarKnesS_WolF: ie, where they check messages
20:42.23funxionhummonk read that
20:42.55DarKnesS_WolFhunmonk: i know u wanna user dial a number and only enter the password not the mailbox number
20:43.04funxionread the part at the bottom about oicemail.conf
20:43.44DarKnesS_WolF<PROTECTED>
20:45.41hunmonkfunxion: mm, i've been to that page.  perhaps i'm not making myself clear.  what i want is: user accesses their voicemail box to listen to messages -- there's also an option in there for them to do stuff like change their main greeting, set a new passcode, etc -- and _that_ is what i want to disable
20:46.08nvrpunk_how do I fix  DEBUG[17556]: jitterbuf.c:535 jb_put: Attempting to exceed Jitterbuf max 600 timeslots
20:46.19nvrpunk_as soon as this hit the console audio dropped
20:46.24nvrpunk_during the call
20:46.25hunmonkfunxion: on the voicemailmain page you pointed me to, that would be disabling the entire '0 Mailbox options' section
20:47.23*** join/#asterisk xacatecas (n=jkroon@196.46.172.142)
20:47.55*** part/#asterisk harryv (n=harry@67-207-147-205.slicehost.net)
20:48.11xacatecashi guys, random question, if i connect to an upstream sip connection (which happens to be asterisk), is it possible to obtain the remote uniqueid in some way?
20:48.11C4coloRandom answer[148]: RTFM! (from DJpyro)
20:48.45xacatecasspecifically I want to set my "userfield" to the remote "uniqueid".
20:51.32funxionhunmonk what I wanted you to read was "(1) Prompts for these are only played if these options are enabled in voicemail.conf"
20:52.05*** join/#asterisk Rico29 (n=Rico@ARennes-256-1-155-252.w86-195.abo.wanadoo.fr)
20:52.32hunmonkfunxion: yeah, i saw that part.  i thought that only applied to the stuff w/ (1) by it in the menu tree on that page
20:53.25plikhunmonk: yeah, that's how I understood it too
20:56.00*** join/#asterisk tuxd00d (n=tuxd00d@128.187.132.25)
20:56.46*** join/#asterisk atis_work (n=atis_wor@193.238.212.171)
20:58.09[hC]anyone loaded the newer xml-config-based cisco 79[46]1 firmware onto an older cisco 79[46]0 ?
20:58.14*** join/#asterisk rpyne (n=richard@69.77.169.14)
20:59.16rpyneDoes anyone know how to allow asterisk to return a device ID (not the callerID)
21:01.40rpyneAnyone has the problem where the called phone keep ringing after the caller phone hangs up if the call is not answered before the timeout period?
21:02.21*** join/#asterisk seaq (n=seaq@190.25.104.12)
21:02.27seaqhello
21:03.09seaqbesides NAT any other reason to get one way audio at INCOMING calls from a SIP trunk_  outgoing works fine both ways...
21:03.28lmadsenfirewall?
21:03.37lmadsenblocking the RTP ports
21:04.42*** join/#asterisk jong2 (n=chatzill@65.100.10.89)
21:05.45*** part/#asterisk lirakis_work (n=lirakis@65.200.191.241)
21:06.15jong2have anyone tried voicexml thing?
21:06.49*** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir)
21:07.37*** join/#asterisk [TK]D-Fender (i=Joe@64.235.218.194)
21:10.51lmadsenjbot: I love you
21:10.52jbotYou love you?
21:11.08lmadsenjbot: yes I do
21:11.08jbotYou do?
21:11.15lmadsenjbot: who do you do?
21:11.25lmadsenstumped!
21:11.45talntidjbot, don't let lmadsen bug you
21:12.03talntidrumor on the streets has it.. that he won't be around much longer...
21:13.09seanbrightwhere i'm from dems fightin words
21:13.40adeeli've set my DTMF mode to auto, but it seems that * forces 2833 even though the carrier sends the 200 saying inband, is there any insight anyone can offer?
21:14.26talntiduse a bigger hammer? :)
21:14.30*** join/#asterisk deeperror (n=deeperro@69-215-202-195.ded.ameritech.net)
21:14.39talntidwish i knew
21:15.18deeperrorwhen asterisk detects a hook-flash is the stutter tone standard or is there some way to disable this behavior?   There is no stutter when the phone is picked up just when pressing flash...
21:15.47adeeldeeperror, typically, the stutter tone indicates voicemail waiting
21:15.56*** join/#asterisk gitguy (n=diego@adsl-132-73.click.com.py)
21:16.05*** part/#asterisk IPGHOST (n=imrouter@116.58.16.175)
21:16.09deeperrori do have voicemail setup but nothing configured on the channels (zapata.conf)
21:17.45adeeldeeperror, you should be able to change that behavior....can't recall where though, maybe in zapata.conf
21:18.22deeperrorwell i know there is a mailbox setting in zapata but that i think is for the initial pickup of the line not the flast
21:18.28deeperrors/flast/flash
21:23.52*** part/#asterisk deeperror (n=deeperro@69-215-202-195.ded.ameritech.net)
21:24.41*** part/#asterisk xai (n=pasta@about/networking/0.0.0.0/xai)
21:26.25*** join/#asterisk harryv (n=harry@67-207-147-205.slicehost.net)
21:26.36harryvwhen using Record() the beep signaling the start of the recording is very very low
21:29.42[TK]D-Fenderharryv, its the endpoint listening to it that needs adjusting
21:29.48*** join/#asterisk UnixDog (n=UnixDog@ppp-69-238-167-52.dsl.irvnca.pacbell.net)
21:29.54UnixDogwhat the hell
21:30.05UnixDogasterisk.com went flash
21:30.13UnixDogthats fricking stupid
21:30.20UnixDogflash is crap
21:30.37QwellUnixDog: so complain to the owners of asterisk.com
21:30.46UnixDogdigium
21:30.49Rico29UnixDog>  i agree
21:30.53jong2trash
21:30.53Qwelltry again
21:31.25harryvand the download links for the source is really incompetent and not all too compatible with wget
21:33.08UnixDogi dont use flash  not I cant get to the asterik site.
21:33.15UnixDogasterisk even
21:33.18QwellUnixDog: again - complain to the owners of asterisk.com
21:34.32UnixDogwhy does digium not keep the sourcecode on thier own server
21:34.33unpaidbillwhat software do you guys use to load test asterisk?
21:34.42QwellUnixDog: who says we don't?
21:34.56UnixDogI dont find a link on the digium site for svn
21:35.24jjshoeUnixDog look. harder.
21:35.43jjshoegoogle.com/?q=ssite: digium.com svn
21:36.17UnixDogasterisk.org
21:36.19UnixDoglol
21:36.27UnixDogbrain fart day
21:37.30*** join/#asterisk ZaVoid (n=zavoid@c-67-165-25-195.hsd1.pa.comcast.net)
21:37.36Maverichey Qwell how are you doing?
21:37.40UnixDogok bbl
21:37.43*** part/#asterisk UnixDog (n=UnixDog@ppp-69-238-167-52.dsl.irvnca.pacbell.net)
21:37.53QwellMaveric: not bad
21:38.04Mavericis malcolmd not working for digium any more?
21:38.10*** join/#asterisk cabbiepete (n=cabbiepe@host86-144-249-12.range86-144.btcentralplus.com)
21:38.11Qwellsure he is
21:38.19Mavericah ok i wasn't sure
21:38.29Maveric<----- the Easynews guy he used to talk to
21:38.32Qwellwe probably wouldn't let Malcolm leave
21:38.35Mavericif you remember
21:38.43Qwellmight have to threaten him if he ever tried. ;)
21:38.48Mavericlol
21:38.49Mavericyeah
21:38.54Mavericmalcolmd is a good guy
21:38.59*** join/#asterisk notbright (n=chatzill@unaffiliated/yourname/x-837320)
21:39.10Maverici'm back taking care of asterisk office again
21:39.16Mavericso here i sit
21:40.24*** join/#asterisk jsmith (n=jsmith@72.21.36.138)
21:40.24*** mode/#asterisk [+o jsmith] by ChanServ
21:40.36jsmithjumps back in #asterisk for a few minutes
21:41.16Kattycries.
21:42.03jsmithKatty: You don't have to cry just because I'm here!
21:42.17Kattyjsmith: has nothing to do with you
21:42.23Kattyjsmith: been a bad day :<
21:42.34jsmith:-(
21:44.23harryvwhen i Record() the recordings are very obfuscated and bad quality. I have tried different phones.
21:45.31jsmithharryv: Sounds like a network or codec or CPU overload problem... can you be more specific?
21:46.40Kattyjbot: obfuscated
21:46.41jbotfrom memory, obfuscated is something is intentionally made confusing and hard to understand.
21:46.58Kattyjbot: can you pronounce that for me
21:47.14Kattyahhh b few skated?
21:47.24*** join/#asterisk justdave_ (n=dave@unaffiliated/justdave)
21:47.31outtoluncall-buggered-up
21:47.41Kattyouttolunc: i like that one better.
21:48.50*** join/#asterisk ZaVoid (n=zavoid@c-67-165-25-195.hsd1.pa.comcast.net)
21:49.21harryvjsmith: load is 0.1, so, that's not an issue, and i've tried callnig from a sip phone at the local net, and also accessing it from a normal pstn phone -- it's isdn
21:49.45harryvit's difficult to hear what's actually said on the recording, it's fuzzy.
21:50.05jsmithHmmmn... most likely a codec issue of some kind then...
21:50.09jsmithWhat version of Asterisk?
21:50.17jsmithWhich codec (on the SIP call)?
21:50.31jsmithWhat format is Record() using to save the file?
21:52.37harryv1.4.18, alaw codec for the sip, but the pstn was just as bad, and it's saved as gsm
21:56.29jong2is echo canceller card worthwhile investment?
21:58.33jsmithjong2: If you have echo, yes!
21:58.50talntid;)
21:58.56[hC]Anyone using SIP firmware 8.9 or near there with a cisco 7960?
21:58.57jsmithjong2: (I usually suggest buying it just in case -- audio quality should be one of your primary goals)
21:59.05jong2i thought asterisk comes with echo.
21:59.17jsmith[hC]: Nope... didn't know 8.9 was out
21:59.18talntidcups with a connecting string doesn't cause echo.
21:59.40jsmithtalntid: My tin can has a bunch of echo!
21:59.41[hC]jsmith: yep, i just pulled it off CCO earlier. I went from 7.x to it, and its rebooting itself over and over again. not sure why.
21:59.59jsmith[hC]: Usually a bad config file
22:00.29[hC]jsmith: yeah, could be. It says copying to bank 1 after the upgrade, then reboots and starts over.. I'll check the config file.
22:03.36[hC]jsmith: lovely. the way cisco shipped their OS79XX.cfg, it was pointing to the P00 file, I had to change it to P0S
22:03.37[hC]:|
22:03.42[hC]now its happy..
22:04.04*** join/#asterisk anthm (n=anthm@70-9-119-28.area4.spcsdns.net)
22:04.15talntidjsmith, to keep your tin can from echoing, simply change the string to a wire. no more echo.
22:05.40jsmithtalntid: Maybe I need a smaller can?  The echo comes from the can, even when I disconnect the string (or the wire)
22:05.56talntidhave you checked to ensure it's not a corn can, but rather a bean or soup can?
22:06.14talntidomg. please don't tell me it's a coffee can.
22:06.27talntidthose are sooo old.
22:06.31Qwellplastic coffee cans are acceptable, and do not produce echo
22:06.39talntidtrue, but not the tin ones
22:06.48talntidplastics make it possible :)
22:06.50CanWoodit's wireless...  so obviously a pringles can
22:07.12talntidCanWood, problem is, when I use pringles cans... i hear a lot of popping on the line... :\
22:07.15talntidso i stopped using them
22:07.20CanWoodwell played
22:07.21Qwellyou can't stop
22:07.31Qwellonce you hear the popping, that is
22:07.36talntidright.
22:08.15talntidpersonally, i have found great success with the low-profile tuna can...
22:08.35talntiddownside, my cat tends to attack it.
22:08.40Qwellthe latency with those is lower.  there is still echo, but it's less noticeable.
22:09.30talntidi added a foam pad to the back where the string connects. seems to drop the echo out
22:09.58talntidand some LED lights.
22:10.59*** part/#asterisk harryv (n=harry@67-207-147-205.slicehost.net)
22:12.10manddis there s reson why my VOIP provider server changes destenation ports in SIP packets
22:12.12manddafter a while
22:12.16manddhttp://pastebin.com/m327b1c02
22:12.32manddand until I reload, i cant receive incomign calls
22:12.51manddafte reload, incoming calls work (invite packets are addressed to 5060)
22:13.02manddand then it starts sending themn to another port again
22:13.04talntidwireshark owns :)
22:13.12manddwireshark ++
22:13.54nvrpunk_question, I have a PTSN gw on a SIP connection to  BoxA, and a trunk from A to B with A set as the user and B set as the peer,   before A was peer and B was user and it worked now it does not
22:13.58manddan ideas?
22:14.00manddany
22:14.06nvrpunk_can the initiating called only be set to user?
22:14.10nvrpunk_caller*
22:15.53*** join/#asterisk jm|laptop (n=jm|home@zen.jamiem.com)
22:21.11rje`cfanyone know if you can make an allworx 9112 phone work with asterisk?
22:22.05*** join/#asterisk ManxPower (n=manxpowe@adsl-35-244-48.msy.bellsouth.net)
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22:38.35harryvbloody, being thrown off all the time. but: can't i replade the beep -sound Record() uses?
22:39.54*** join/#asterisk ChkDigit (n=mrw@static24-89-65-166.regina.accesscomm.ca)
22:40.15*** part/#asterisk harryv (n=harry@67-207-147-205.slicehost.net)
22:47.04*** join/#asterisk edge (n=edge@h75-100-218-214.lncswi.dsl.dynamic.tds.net)
22:51.48*** join/#asterisk fnordus (n=dnall@24.84.160.227)
22:56.54[hC]jsmith: seen a 7941/7961 get stuck in a dhcp request loop trying to upgrade itself?
22:57.18Qwelllooks over at his 7970
22:57.24Qwell"Updating CTL"
22:57.29Qwellbeen doing that for about 2 months now
22:59.33*** join/#asterisk Keltus (i=Keltus@about/cooking/nakedchef/beefstew/Keltus)
23:00.06[hC]Hah.
23:00.09[hC]Ive had that before too
23:00.15[hC]I frigging hate cisco + firmware upgrades.
23:00.33jjshoecisco == paperweight
23:01.12[hC]Qwell: why oh why is there no scp on the aa50? or tcpdump? or why is it that when i start an ftp download, the box immediately reboots?  grumble grumble.
23:01.24[hC]trying to debug / download / upload anything on this thing is a pain
23:01.39[hC]ive been sending files with netcat!
23:04.14*** join/#asterisk fnordus (n=dnall@24.84.160.227)
23:08.23*** join/#asterisk RoyK (n=roy@ip-115-24-149-91.dialup.ice.no)
23:09.01*** part/#asterisk RoyK (n=roy@ip-115-24-149-91.dialup.ice.no)
23:18.02*** join/#asterisk _polto_ (n=polto@elphelut.fttp.xmission.com)
23:18.10_polto_hello all
23:19.23*** join/#asterisk ManxPower (n=manxpowe@adsl-35-244-183.msy.bellsouth.net)
23:21.44nvrpunk_anyone mind looking at the following two call console logs?
23:21.46nvrpunk_http://pastebin.com/m1bec92bb
23:22.04nvrpunk_whenever I force a jitter on the middle end of the trunk all hell breaks loose
23:22.20_polto_can somebody suggest me a good ISP for DID in UDA ? (I tried http://www.ipcomms.net/ for a month, and they was not able to make outbound calls working, and it seems that they have only 1 person-support)
23:23.00nvrpunk_you mean a PSTN Gateway provider?
23:23.24nvrpunk_vitelity, teliax, junctionnetworks
23:23.33nvrpunk_voice pulse
23:23.38nvrpunk_there s a ton
23:23.45nvrpunk_voip-info.org has a good list
23:24.35jblackpolto: UDA = USA?
23:24.47_polto_oups, yeh - USA
23:25.15unpaidbilli use callcentric
23:25.20_polto_I am searching for a reliable provider. ipcomms are just .....
23:25.27nvrpunk_vitelity is like 1dollar a month for one chanel unlimited in
23:25.29unpaidbill2.95 dirt cheap DID
23:25.31ManxPowernvrpunk_: Maybe the IAX2 trunking has problems with the jitterbuffer support in Asterisk?
23:25.32unpaidbillper month
23:25.53ManxPowerI suggest vitelity.net or teliax.com
23:26.21_polto_unpaidbill, I am not looking for the cheapest. I look for a reliable IAX2 provider for DID in Utah, USA
23:26.50unpaidbilloh iax, i dont think callcentric supports that
23:27.20ManxPowernvrpunk_: have you tried turning off trunking?
23:27.28unpaidbillbut - callcentric has worked great for me via sip!
23:27.29nvrpunk_ManxPower, well here is the concept, we have a satellite link, and a trunk that goes across is.  The PSTN is with the teleport which is the in the middle of us and the PSTN GW.  The buffer is on the Teleport side so when it receives our outbound traffic it's supposed to buffer it so our outbound sounds better
23:27.44_polto_ok, i'll read about vitelity.net and teliax.com
23:27.46nvrpunk_ManxPower, not yet, been messing with it for a day doing various tests :)
23:27.48_polto_thanks
23:27.56ManxPowernvrpunk_: you realize that an IAX2 trunk is different from an IAX2 connection, right?
23:28.10nvrpunk_ManxPower, yes one trunk=yes
23:28.13nvrpunk_other doesnt have that
23:28.15nvrpunk_:)
23:28.17nvrpunk_well aware
23:28.20nvrpunk_and what it does
23:28.28ManxPowernvrpunk_: you need trunk=yes on BOTH ends
23:28.30nvrpunk_and the trunks and jitterbuffers mess up
23:28.32nvrpunk_yes
23:28.34nvrpunk_got that
23:28.36nvrpunk_and timestamps
23:28.39nvrpunk_and ztdummy
23:28.52ManxPowernvrpunk_: if turning off trunking fixes it, then report it on bugs.digium.com
23:29.52nvrpunk_ManxPower, you got what I meant about the middle point receiving our outgoing traffic and buffering it though right?
23:30.14nvrpunk_i was just making sure there was nothing wrong with doing that
23:30.31ManxPowernvrpunk_: is the 'middle point" IAX2?
23:30.46nvrpunk_its IAX2 on the trunk
23:30.52nvrpunk_and SIP on the PSTN GW
23:31.07ManxPowerWhat protocol does the "middle point" run?
23:31.25nvrpunk_codec?
23:31.27nvrpunk_o0
23:31.41ManxPowerGenerally you only want to dejitter at the END POINTS of the call.
23:31.42nvrpunk_confused by what you mean and tcpip pops in my head
23:32.01ManxPowernvrpunk_: no, is your "middle point" a router, an asterisk, some super satellite buffering device?
23:32.11nvrpunk_asterisk also
23:32.18nvrpunk_iax.conf to iax.conf
23:32.32nvrpunk_the USA has the PSTN GW
23:32.40nvrpunk_the middle is sweden with an ast box
23:32.46nvrpunk_and our end is iraq with ast box
23:33.19ManxPowerAs I said, you only want to dejitter at the end phones if possible, but since IAX2 does not use RTP, I have no idea how that would impact it.
23:33.34ManxPowerBut as I said, if turning off trunking makes the problem go away, then report it as a bug.
23:33.44nvrpunk_ManxPower, did you see my outputs
23:33.46nvrpunk_form my console
23:33.52nvrpunk_http://pastebin.com/m1bec92bb
23:33.56nvrpunk_from*
23:34.11nvrpunk_RTT 1000 a lot
23:34.12ManxPowerI just finished a 12 hour say dealing with whiny real estate agents that think "the internet is down"
23:34.28ManxPowerI'm not lookking at ANYTHING to close.
23:34.52nvrpunk_I had my fiancee's sister who troubleshot for 2 hours with bell south on her DSL and I spent 5 minutes to figure out it was her Cat5e :P
23:35.06nvrpunk_she didnt really believe it was that simple
23:35.16nvrpunk_since the techs spent hours troublshooting
23:35.42ManxPowerWe were renumbering our network at that office.  Of course there are going to be problems, that's why we warned them about the upgrade on monday
23:37.59timothy_jonesguys, is possible to find out what protocols sip phone (hard) support?
23:38.08nvrpunk_no
23:38.16nvrpunk_timothy_jones, manuals for the phones
23:38.20nvrpunk_settings on the phones
23:38.26nvrpunk_asteriskguru.com
23:38.33nvrpunk_there are many places to find the info
23:38.47timothy_jonesno possible from asterisk cli?
23:39.06nvrpunk_sure, see if it make a call with a given protocol
23:39.07nvrpunk_:P
23:39.13nvrpunk_total waste of time
23:39.19nvrpunk_given you could read the manual
23:39.40nvrpunk_and no not possible from the CLI
23:40.01timothy_jonesnvrpunk_: thank you my friend
23:40.01nvrpunk_not without placing a call at some point
23:40.36nvrpunk_although that would be kinda nice to have asterisk poll the device
23:40.54nvrpunk_bit of overhead though that's not needed
23:41.03timothy_jonesthese are cisco 7912. but I know they were flashed to support sip. maybe need to ask cisco
23:42.38timothy_jonesprobably possible diasllow=all, allow=g729 (etc), but yes, waste of time
23:45.31nvrpunk_I am quite certain you can find out online
23:45.40nvrpunk_especially being that they are cisco devices
23:45.48nvrpunk_plus, they probably have all the info in the phone
23:45.58nvrpunk_infact, i know they do
23:47.06timothy_jonesI checked the settings and not, maybe I need to check web page
23:54.32*** join/#asterisk ZaVoid (n=zavoid@c-67-165-25-195.hsd1.pa.comcast.net)
23:55.02*** join/#asterisk budol (i=budol@202.124.138.70)
23:56.58*** join/#asterisk SublimE-wk (n=noneya@75.160.195.50)
23:57.21SublimE-wkI was hoping to find some help... we are in contact with Qwest because starting about two weeks ago we started getting intermitant call issues, when we call the main line, it just rings and rings but the phone system isnt picking up and ringing all the way through, then sometime it goes right through.
23:57.29SublimE-wkI dont know if it's stopping at the box or not. I thought maybe this log file contained info about the problem... --> http://freebsdonirc.pastebin.ca/1004219

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