irclog2html for #asterisk on 20061115

00:01.37voxtopJT: yeah
00:02.15*** join/#asterisk aao_pwner (n=aao_pwne@c-24-21-91-140.hsd1.wa.comcast.net)
00:02.28voxtopJT: with "opermode=AUSTRALIA" in /etc/modprobe.d/zaptel and with "wctdm" in /etc/modules , wctdm loads fine. the problem is that asterisk fails to start
00:03.31JThave you tried reverting to plain wctdm with no opts?
00:04.16voxtopJT: yeah, and asterisk starts fine then
00:04.57JTcan you pastebin the verbose errors?
00:07.00voxtopJT: i can't reboot the box at this point to generate the errors, but let me see if i can find my paste from last night. sec..
00:07.17JTcan't you just start asterisk
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00:09.25tzafrir_laptopvoxtop, /etc/zaptel/1 generated?
00:09.26voxtopJT: found the errors:   http://rafb.net/paste/results/e4WnIi76.html
00:10.00voxtoptzafrir_laptop: hi again  :)
00:10.16voxtoptzafrir_laptop: /dev/zap/{1-4} exist
00:10.37tzafrir_laptopztcfg is run successfully?
00:10.45HumpBackvoxtop: try to dmesg|less and at tho botton see if there are any errors
00:10.48lucasjbHiyas, I have a user running an IAX soft phone on Windows XP connecting over the Internet (from a different counry) to my Asterisk server. He's complained recently that the audio signal fades in and out constantly with a regular frequency. Can anyone suggest what a likely cause of this might be?
00:12.30voxtoptzafrir_laptop: yes, ztcfg ran successfully
00:13.05voxtopHumpBack: unfortunately i can't at this point, as the * box in question is being used and can't be taken down for several hours
00:13.47Drukendoes anyone get NAME CID on a pri ?? :)
00:15.28HumpBackvoxtop: because i have a similar problem where ztcfg -vv reports all ok but i cannot use the fxo ports
00:15.35tzafrir_laptopvoxtop, in your zaptel.conf you remmed-out channel 1?
00:15.54tzafrir_laptopwhat do you see on /proc/zaptel/1 ?
00:16.02HumpBackvoxtop: s/fxo ports/fxs ports
00:16.09HumpBackvoxtop: and i get errors in dmesg
00:16.49HumpBackvoxtop: http://pastebin.com/824079
00:17.46voxtopHumpBack: i only have fxo ports
00:17.55voxtopHumpBack: never played with fxs ports
00:18.00tzafrir_laptopHumpBack, did you write to asterisk-users ?
00:18.27HumpBacktzafrir_laptop: yes I did
00:18.30HumpBackhttp://lists.digium.com/pipermail/asterisk-users/2006-November/172459.html
00:18.33HumpBackThat's me
00:18.39tzafrir_laptopYou don't seem to have similar problems. You seem to have physical problems with the card, or similar lower-level problems in the system
00:18.40voxtoptzafrir_laptop: /proc/zaptel/1  :  http://rafb.net/paste/results/KGtkFQ81.html
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00:19.18HumpBacktzafrir_laptop: i already tried two cards. seems really strange for both to have the same issue
00:19.21tzafrir_laptopvoxtop, in use? Are you trying to run a second instace of asterisk?
00:19.29voxtoptzafrir_laptop: note that this is with "options wctdm opermode=AUSTRALIA" in /etc/modprobe.d/zaptel and *without* "wctdm" in /etc/modules
00:19.39tzafrir_laptopHumpBack, time to blame the motherboard, I guess
00:20.20tzafrir_laptopHumpBack, seriusly, I don't know those cards well enough to give you a good answer
00:20.49tzafrir_laptopvoxtop, the card is listed as 'in use". Someone else, probably a different astersk, uses it
00:20.58tzafrir_laptopps aux | grep asterisk
00:21.21tzafrir_laptopDo you run xen or a similar virtulized setup?
00:22.01voxtoptzafrir_laptop: i should only have one instance of asterisk running:  http://rafb.net/paste/results/NxH5UA51.html
00:22.20voxtoptzafrir_laptop: no virtualisation; just a standard box
00:22.28HumpBacktzafrir_laptop: hmmm the only other system i have right now has 4 of the 5 pci slots in use. And i dont know if i can use the last due to irq sharing with the vga card
00:23.26tzafrir_laptopvoxtop, just wondering: you have kernel 2.4, right?
00:23.33voxtoptzafrir_laptop: 2.6.16
00:23.49voxtoptzafrir_laptop: on sarge
00:23.55tzafrir_laptopso how come we see so many asterisk processes?
00:24.21tzafrir_laptopThat command shouldn't show separate threads differently, AFAIR
00:25.14ucfMethodnight everyone, seeya tomorrow
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00:27.28voxtoptzafrir_laptop: what does `ps auxf` list on your asterisk boxes?
00:28.15tzafrir_laptopjust a single process
00:28.36voxtoptzafrir_laptop: even with the 'f' argument?
00:28.56tzafrir_laptopCould you run it again to see if the processes remain the same? 'f' shows processes as a tree (somewhat like pstree)
00:31.42voxtoptzafrir_laptop: results from running ps again:  http://rafb.net/paste/results/dC4S6967.html
00:32.53tzafrir_laptopLooks like different threads of the same process (they have exactly the same size)
00:33.03tzafrir_laptopI still wonder why they are all displayed
00:34.28HumpBackvoxtop: stop the asterisk server and see if you still get processes in ps
00:35.21HumpBackvoxtop: forget what i told makes no sence
00:36.16HumpBackvoxtop: run /lib/libc.so.6 and past the result somewhere
00:38.10JTmm not bad
00:38.11JT--- Results after 7252 passes ---
00:38.11JTBest: 100.000000 -- Worst: 99.816895 -- Average: 99.994708
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00:52.21voxtophi guys, sorry i disappeared. the keyboard on my laptop died  :P
00:53.58voxtopHumpBack: you asked me to run /lib/libc6.so.6 . however, that file doesn't exist on my * box
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00:59.10HumpBackvoxtop: you should have some /lib/libc*.so or something
00:59.17HumpBackvoxtop: run it
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01:03.10SavageOne#ext does an unattended transfer, how do i do an attended transfer?
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01:03.57dovid~centosbug
01:03.58jbotfrom memory, centosbug is a problem with the latest Centos kernels (4.2 and 4.3).  To fix it, paste everything inside the quotes into a root shell:  "sed -i s/rw_lock/rwlock/ /usr/src/kernels/*/include/linux/spinlock.h"
01:06.16voxtopHumpBack: what will that do?
01:06.46HumpBackvoxtop: it will "run" the libc that will report some information on the version and extensions
01:06.53SavageOnehey ll
01:06.55SavageOneall
01:07.04SavageOnejust trying to find what I need to dial to do an attended transfer
01:07.09HumpBackvoxtop: it is to see if you are using linuxthreads ou the new NPTL
01:07.14SavageOneor to make it so that # always does attended instead of unattended
01:07.29voxtopHumpBack: http://rafb.net/paste/results/sF6ml424.html
01:08.11HumpBackvoxtop:   linuxthreads-0.10 by Xavier Leroy That is some very VERY old libc
01:08.43HumpBackand no NPTL. That explains why the threads apper in ps aux
01:09.14HumpBackvoxtop: What distribution are you using?
01:09.46voxtopHumpBack: Debian 3.1 (Sarge) with 2.6.16-2-686-smp
01:10.38HumpBackvoxtop: strange. I tought sarge already had nptl. Search in the packages to see if there is a libc that uses nptl
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01:10.59voxtopHumpBack: is it important to use NPTL?
01:11.24HumpBackvoxtop: From a performance perspective the diference in terms of cpu and memory is HUGE
01:11.26voxtopHumpBack: /lib/libc.so.6 links to libc-2.3.6.so
01:11.33voxtopah
01:11.52HumpBackfrom the asterisk side I do not know
01:12.34HumpBackIn theory if one uses thread semantics to code the application there should be no diference between linuxthreads and nptl
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02:03.39Kattyallo.
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02:06.06justinu|laptopmew
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02:09.12Kattyjustinu|laptop: (=
02:13.29fileack it's a mitcheloc
02:13.51fileyay House
02:17.19voxtopdo you set rxgain and txgain for all zap channels, or for each one?
02:19.45aydiosmiomy Zap channels are riding spinnas
02:19.55aydiosmiotwentyfours
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02:33.59voxtopi changed rxgain from 18 to 68, restarted asterisk, and don't hear any change in the volume. why might that be?
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02:39.39sevardSo
02:39.51sevardSince when is libIAX a seperate library?
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02:45.11brookshirei think there is a separate libiax
02:45.17brookshirei didn't realize it was included
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02:59.26JuggielibIAX is for writing iax clients.
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03:09.58droopshey with asterisk 1.2.13, is there a problem with meetme not terminating when no calls are currently going on?
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03:36.50*** join/#asterisk SomethingISODD (n=dan@h109.42.63.69.cable.ottr.cablerocket.net)
03:37.09SomethingISODDhello all question is there a bug in 1.2 that stops people from being about to join a queue?
03:37.27SomethingISODDi keep getting a message when a person calls saying can not join queue
03:39.52jaikeSomethingISODD: check joinempty and maxlen configuration in queue.conf
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03:40.50SomethingISODDjaike i have joinempty commented out and maxlen = 0
03:41.51SomethingISODDjaike do you see any issue?
03:41.56jaikenope
03:42.19SomethingISODDI will paste the queue and extension for that in pastebin, if you could please take a look
03:42.27jaikeSomethingISODD: the member agents cant join the queue? or calls wont go into the queue
03:42.30SomethingISODDi have been trying to figure this out for two days now with no luck
03:42.39fholmesAnyone know of any click to call projects working through asterisk?
03:42.40SomethingISODDcalls can`t join the queue
03:44.39jaikeSomethingISODD: try setting joinempty=yes and leavewhenempty=no
03:45.16SomethingISODDjaike if you have a second i pasted it all
03:45.17SomethingISODDhttp://pastebin.com/824714
03:45.33jaikefholmes: were doing it, php with manager api
03:46.09fholmesreally?  cool.  Can you tell me what your using it for?
03:47.56jaikewebsite, customers enter their tel num and when they click dials their phone and connects to an agent
03:48.08jaikeand vice versa
03:48.45jaikeSomethingISODD: i think you have to explicitly set joinempty and leavewhenempty
03:48.57SomethingISODDoh ok let me try
03:49.01fholmesCool.  I was just thinking of maybe making something for affiliates to use to track phone calls made instead of lead forms filled out.
03:49.20SomethingISODDjaike do i have to set the agent in there as well?
03:49.29SomethingISODDlike i did or can i do the login through the extensions
03:49.59JTreal click to call applications used web soft phones i thought
03:50.07JTentering a phone number is liable to abuse
03:51.52jaikeJT: its on our website but calls coming in is very rare, in case the customer is to lazy to call us up
03:52.28JTheh
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03:52.53JTyeah but it has the potential for misuse, eg. someone putting in the phone number of someone they wish to harras
03:53.13jaikeJT: hmm..good point
03:53.33jaikeSomethingISODD: how do you login your agents? AgentCallbackLogin? AgentLogin
03:53.35fholmesYes, it would have a high potential for misuse.
03:54.21SomethingISODDlet me confirm that second
03:54.24JTyou'd at least need to make it have safeguards in place, like a register of numbers to not call, either manually entered, or automatically enterred if calls to one number are being made at a high rate
03:54.43SomethingISODDAgentCallbackLogin(|${CALLERIDNUM}@tech-queue)
03:55.10fholmesThere would have to be a way to do it somehow with out a softphone on the callers end.
03:56.38fholmesI want to setup a self hosted application that could be used to track calls to a third party.
03:56.46jaikeSomethingISODD: you have a tech-queue context?
03:56.51SomethingISODDyes
03:57.01SomethingISODDthat was the queue i pasted i believe in pastebin
03:57.17JTfholmes: there would be no way that would be foolproof if you let them enter phone numbers to call
03:57.24JTunless it's a restricted website
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03:57.27JTthat's fairly safe
03:58.02jaikeSomethingISODD: i only see a [tech-queue] queuename, how bout tech-queue context in extensions
03:58.13jaikeanyway try the settings first
03:58.47jaikehas anyone done stress testing on 1.4?
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04:04.10SomethingISODDjaike ok its not ringing my phone, so when the caller logs in to the queue it has to be under an extension of [tech-queue] ?>
04:04.21SomethingISODDi mean in the extenstions.conf
04:07.29jaikethe phone extension you are logged in should have an extension in [tech-queue] context, as per your agentcallbacklogin
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04:12.41hd420can I use Astrisk in lieu of my vonage router?
04:12.58hd420yes, I'm a newbie
04:20.17aydiosmiohd420: no, not with a standard residential account
04:20.22aydiosmioyou need to use a spftphone account
04:20.26aydiosmiosoftphone
04:20.40JTdo they restrict it in some way?
04:20.41hd420I have a softphone account
04:20.44hd420for travel
04:20.48aydiosmiothere are several asterisk compatible providers though
04:20.53aydiosmioI'd suggest you switch
04:21.20aydiosmiohd420: then yes, voip-info.org has tutorials
04:21.35aydiosmioyou can get he sip crednetials form the vonage website under your account
04:22.03hd420I'm not sure how everything works
04:23.13aydiosmioyou just set up a trunk with vonage's sip server (sphone.vopr.vonage.net:5061) and your sip credentials (login and password) with a few other specific setigns and you're set
04:23.29aydiosmiosearch voip-info.org for vonage
04:23.39hd420as indicated above, <hd420> yes, I'm a newbie
04:24.21aydiosmiookay
04:24.24hd420what's a "trunk"?
04:24.27aydiosmioso you need me to do the goole search for you then?
04:24.33aydiosmiohttp://www.voip-info.org/wiki/view/Asterisk+and+Vonage
04:26.25JTaydiosmio: i'm curious, how do vonage restrict non softphone accounts from connecting to asterisk?
04:27.10aydiosmiothey do not disclose the SIP credentials for their ATA accounts
04:27.52JTthe sip secret?
04:28.41aydiosmiothat's part of it
04:29.01JTsip username too?
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04:35.42aydiosmiosip username for softphone accounts is the phone number
04:35.46aydiosmionot sure about ATA accounts
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04:37.32hd420I'm missing something
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04:38.11hd420so asterisk is a vonage add-on, in lieu of a vonage replacement?
04:39.05JTaydiosmio: hrm ok, well otherwise what's the other part?
04:39.29JThd420: asterisk is not a vonage add on
04:39.46hd420JT: then?
04:39.59hd420I have Vonage and a softphone here
04:39.59JTasterisk is an open source telephony toolkit
04:40.11JToften used to mak pbxs in software
04:40.15hd420JT: I can read as well as the next guy, mate
04:40.29JTdon't get antsy with me
04:40.30JTyou asked
04:40.32JTi answered
04:40.37JTif you know, why ask?
04:40.52[TK]D-FenderJT : I *like* the sound of my voice personally....
04:41.09JTheh
04:41.15hd420what does "telephony toolkit" mean?
04:41.19[TK]D-Fender</sarcasm>
04:41.53hd420I have Vonage, and a softphone
04:41.56JThd420: it allows you to interface with both voip channels and hardware analogue and digital telephony channels
04:42.08JTso you usually use it in a pbx or ivr capacity
04:42.43[TK]D-Fenderhd420 : Lets sum this up differently : * is a PBX.  You can have it process calls coming in from all sorts of interfaces (analog & digital lines, VoIP accounts, etc), and rprocess those calls any which way yo want including sending them out over any fo said technologies and do any kind of prcessing in between that you can come up with as well.
04:42.56JThd420: what made you look to using asterisk? is there something in particular you were needing done?
04:43.14[TK]D-Fenderhd420 : Just think of it as a supper-massively customizable PBX limited largely by only your imagination.
04:44.14hd420I'd like to be alerted of new voice mails on my mobile phone
04:44.21[TK]D-Fenderhd420 : If you have a soft-phone account with Vonage, it may be possible to use it with * and let * handle voicmail, and offer other routing options to your callers like "conditional fololow-me" and other intelligent stuff
04:44.39JThd420: that's actually not very easy to do i think, unfortunately
04:45.05JTdue to voicemail being handled by your mobile
04:45.08[TK]D-Fenderhd420 :That is quite possible with *.  You can have * take your VMS for you and upon receipt of a new one send you an SMS message or even e-mail it to you.
04:45.18JTahh
04:45.26JTunless i'm looking at this from the other way around
04:45.37[TK]D-FenderJT : you are :)
04:45.47hd420JT: TKD is explaining how to do what I'd like
04:46.03[TK]D-FenderJT : But good of you to note that potential issue before it arouse and caught us by surprise ;)
04:46.19JT[TK]D-Fender: the way he said it could have been taken either way :)
04:46.51[TK]D-FenderJT : Seemed clear to me, but I'm *special* (thats why they send me the little bus)
04:47.38JT[TK]D-Fender: i've heard people in here ask for asterisk solutions to check their mobile phone's voice mail
04:47.41JTnot pretty :P
04:48.32hd420when a new VM comes in, I want * to send me an SMS saying "new voice mail from +############"
04:48.36hd420that's it
04:48.42[TK]D-FenderJT : Yeah, that would be an extremely daunting task who's requirements would be deemed "not remotely reasonable to validate the wasting of time on"
04:48.45hd420is * overkill for this purpose?
04:48.53[TK]D-Fenderhd420 : Pretty easy.
04:49.26[TK]D-Fenderhd420 : not overkill, but try to paint the bigger picture.  How is the call going to land on *?
04:49.39hd420through the softphone number?
04:50.01[TK]D-Fenderhd420 : (as in : from where?  Will * ONLY be taking VM, not performing other duties, etc?)
04:50.19JTvonage
04:50.35hd420TK: it may perform other duties, but at this point, I see no other use for it
04:51.34[TK]D-Fenderhd420 : is your soft-phone account independant of any other accounts you may have?  Are you planning on treateing it solely as a VM box in the end?
04:52.47hd420I use it when/if I'm on wifi at a cafe or something, where there's no wired connection available or it's inconvenient
04:53.04[TK]D-Fenderhd420 : that will pose a porblem then
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04:53.21hd420Fender: it's not billed independantly
04:53.41[TK]D-Fenderhd420 : If * registers to Vonage as your soft-phone, and then you log in somewhere else remotely, it will "steal" the account away from your * box and it won't get calls anymore.
04:54.01JT[TK]D-Fender: so he will have to connect to * when he's out and about
04:54.15[TK]D-FenderJT : That would be the way to do it.
04:54.31hd420[TK]D: that sounds simple enough
04:55.22hd420I could just get rid of the softphone from Vonage and use some other provider's
04:55.24[TK]D-Fenderhd420 : You could as JT suggested connect * to your Voange soft-phone account, and you would then connect a softphone on a laptop,etc to *.  * would then send calls placed through it OUT your Vonage accoun for you.
04:56.00hd420that would work fine, Fender
04:57.10hd420does t-mobile USA have Internet-email to mobile phone?
04:57.27hd420or the other GSM provider?
04:57.33hd420(Singular?)
04:59.14[TK]D-Fenderhd420 : Ok, well now that I've confirmed that * may be up to the task you'll need to go about downloading and learning. it.
04:59.46hd420Fender: yes, I'm beginning to think it may be overkill for the purpose though
05:03.06Qwellhd420: cingular
05:03.23Qwell(that wasn't an answer to your question...just fixing the spelling :)
05:04.55hd420Qwell: it was a question, many thanks
05:05.24*** join/#asterisk aadilismail (n=adilisma@41-247-154-202.wol.net.pk)
05:06.41saamhi
05:08.53riddleboxcan someone help me get the /etc/init.d/asterisk start command to work in ubunt?
05:11.27*** join/#asterisk luke-jr_ (n=luke-jr@user-0c93tj3.cable.mindspring.com)
05:19.51*** join/#asterisk jbot (i=ibot@pdpc/supporter/active/TimRiker/bot/apt)
05:19.51*** topic/#asterisk is Asterisk: The Open Source PBX -=- http://www.asterisk.org -=- Zaptel 1.2.11 released (Nov. 9, 2006) -=- Join #freepbx for freepbx/trixbox support. -=- Join #asterisk-gui to learn about the new Asterisk GUI framework
05:19.53*** join/#asterisk linagee (n=linagee@unaffiliated/linagee)
05:20.11linageehahaha. cool. i am nearing completion to a small project to be able to enter in names to asterisk using DTMF! :o)
05:22.00*** join/#asterisk sahafeez (n=sahafeez@m0b0e36d0.tmodns.net)
05:22.36[TK]D-Fenderlinagee : "show application directory"
05:22.49linagee[TK]D-Fender: huh?
05:23.11[TK]D-Fenderlinagee : Effectively what you described, including reading the name back
05:23.14linagee[TK]D-Fender: names.
05:23.20linagee[TK]D-Fender: people's names. not people in the directory
05:23.27voxtopwhen you do 'sip show peers' and it says the status is "Unmonitored", how can you change the status?
05:23.31linagee[TK]D-Fender: arbitrary names
05:23.32olinuxhe probably just rendered all your hard work and planning useless
05:23.43linageeno
05:23.49[TK]D-Fenderlinagee : If you're referring to doing a "pure" entry tool for unknown names,t hen well yeah, thats kinda neat
05:23.50*** join/#asterisk Faithful (n=Faithful@ns.linuxterminal.com)
05:23.59linagee[TK]D-Fender: yes
05:24.06linagee[TK]D-Fender: i found a big long list of names on the net
05:24.36linagee[TK]D-Fender: so maybe it doesn't have strange aremeic names, but you can always default and have it say, "we don't know wtf you're trying to say, please speak it instead" and just record.
05:24.59linagee[TK]D-Fender: it has every name i have thrown at it from friends. not from other countries though, that can get weird.
05:25.27linagee[TK]D-Fender: similar to how cellphones can guess what words you are trying to say
05:25.43*** part/#asterisk lucasjb (n=lbarbuto@mail.stabat.com)
05:26.31hd420hmm... i wonder if I can script the Vonage site using Perl and then just get that to send out the SMS to my mobile
05:26.57hd420instead of futzing with asterisk
05:27.42linageecool. works with my first name. :)
05:27.44linageei get one match
05:27.51linageewith my last name, it comes up with two matches
05:28.06linageeanyone want to throw a name in numbers at me and see what it comes up with? :)
05:28.08JTvoice recognition, or dtmf entry?
05:28.12linageeJT: dtmf
05:28.26linageeJT: give me some numbers. :)
05:29.04linageea male name, i haven't gotten to female yet. (will take a second or two)
05:30.26hd420Juliet
05:30.44linageehd420: give me the DTMF numbers. that's how it would be when you call in
05:30.48linageehd420: and a male name. :p
05:30.48hd420linagee: 236
05:30.59linageeBEN. :o)
05:31.01hd420"Ben", I hope
05:31.04linageeyup
05:31.07linageeonly one match
05:31.26linageeif it comes up with more than one, i would have to say, push 1 for Ben, push 2 for ___, etc.
05:31.37linageehd420: another! :o)
05:31.53hd42042726
05:32.08linageeunknown....?
05:32.14linageewhat was it?
05:32.20hd420"Hasan"
05:32.23DrkShdwlinagee: 5333
05:32.24techiehaha
05:32.25linageethat's a name?
05:32.34hd420it's my name, linagee
05:32.35linageeDrkShdw: jeff. :)
05:32.57hd42046726
05:33.02linageethis is cool. it's working. :)
05:33.07DrkShdwlinagee: 627846
05:33.14linageehd420: another empty set
05:33.20hd420"Imran"
05:33.28linageeDrkShdw: martin or marvin
05:33.34hd4206999
05:33.35linageeDrkShdw: both are possibilities. :)
05:33.37hd420?
05:33.46linageehd420: another empty set. wtf
05:33.56linageeevery name you have given me is weird. hah
05:34.03hd420"Ozzy"
05:34.08DrkShdwnot wierd,  just not "english"
05:34.10linageehd420: it would have to default and say, "wtf? please speak your name"
05:34.20hd420Ozzy isn't English?
05:34.24linageeno
05:34.36linageehd420: nobody names their kid ozzy
05:34.43linageemaybe as a nickname
05:34.51linageebut not the name on their birth certificate. :p
05:34.54JTlinagee: 22784656639
05:35.09linageeJT: nothing...?
05:35.17JTbartholomew
05:35.18linageewhat was it?
05:35.36linageei only have bart and barton. hrmph
05:35.42JTi purposely used something esoteric :P
05:35.47hd420983883
05:35.52SomethingISODDquestion
05:35.54hd420986663
05:35.57*** join/#asterisk juanjoc (n=juanjoc@248-32-235-201.fibertel.com.ar)
05:36.03SomethingISODDis there wanyway to restrict asterisk to only use one ip
05:36.04linageemy male name database only has 1,500 or so names. like i said, it's suprisingly gotten every friends name i could come up with.
05:36.07SomethingISODDand not take them all over?
05:36.20DrkShdwlinagee: 25273623
05:36.23hd420923
05:36.40linageeDrkShdw: clarence. :)
05:36.54linageeDrkShdw: i could not have done that many numbers by hand that fast. :)
05:37.03JT628426435
05:37.18linageeJT: nathaniel
05:37.19linagee:)
05:37.25JTyeah
05:37.27linageethat's an odd one
05:37.32DrkShdwlinagee: I know,  I remember playing with this in the other channel.  the female DB is even more impressive
05:37.43linageeDrkShdw: i have a female DB too. it's a bit bigger.
05:37.47JTan old long version of nathan
05:37.51linageegive me a few seconds to do the dtmf cache
05:37.53hd420linagee: 983883 (female)
05:37.57linageesec
05:38.04hd420linagee: sure
05:38.15linageecreating cache...
05:38.17linageedone. :)
05:38.20linageeit's that fast. heh
05:38.40linageehd420: wtf?
05:38.48hd420wtf wtf?
05:38.49linageeYUETTE or YVETTE
05:38.52linageewho has those names. hahaha
05:38.59hd420my fiancee is named Yvette
05:39.03linageei would guess it's the second one
05:39.12linageeyuette is just even more weirder. heh
05:39.33linageeso yep. now i have my female db too. :)
05:39.36hd42098663 (female)
05:39.49linageeyvone
05:39.58linageehd420: the female db is a bit larger
05:40.08SomethingISODD??
05:40.09*** join/#asterisk kph100 (n=kph100@206-248-157-22.dsl.teksavvy.com)
05:40.36linageehd420: i plan on running the number through both DBs and saying like, "are you male or female" (for male) and "are you female or male" (for female names)
05:40.37linageeheh
05:40.51kph100anyone can help me setup cdr mysql?
05:40.55linageeor maybe like, "you're male right? 1)yes, 2)no"
05:40.59linageesomething like that
05:42.07linageeit would be "fun" to put unknown callers through this. similar to the freepbx privacy manager.
05:42.20linagee"we don't have a name on file for you..." hah
05:42.42[TK]D-Fenderlinagee : Yes, and I've know sever male's witht he name "Lindsay", etc.
05:42.57*** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk)
05:43.01linagee[TK]D-Fender: i was thinking transexuals, that's why you just can't assume. hrm
05:43.02JTlinagee: yvette is not an uncommon name
05:43.13linageeJT: i think i've heard it before.
05:43.14JTyour idea of common names is pretty narrow
05:43.18linageemy sister had a friend named
05:43.21linageeJT: hehehe
05:43.47JTbe funnier to replace the names dbs with noun dictionary
05:43.49linageeJT: my first name is 5 letters and my last name is 5 letters. :-D
05:43.57JT"did you mean 'toaster"" ARRGH
05:44.01linageeLOL
05:44.01hd420mine is 5 and 5 as well
05:44.02linageetoaster
05:44.22linageeJT: hey, that might be a fun game
05:44.24JTor if it's someone you don't like
05:44.29JTwhen it detects their name
05:44.29*** join/#asterisk kristalino (n=kristali@cl-157.dub-01.ie.sixxs.net)
05:44.30linageeJT: make an adlib for asterisk. :-D
05:44.31hd420242442 (femalr)
05:44.33JT"did you mean fuckwit?"
05:44.37hd420242442 (female)
05:44.45linageehd420: first unknown
05:44.58linageeand second is the same
05:45.14[TK]D-Fenderplenty of Yvette's around here :)
05:45.15hd420that's because there's no difference in the numbers :)
05:45.20JTlinagee: are you going to be releasing the code?
05:45.30hd420but... it's my sister's name "Aicha"
05:45.41linageeJT: i got the DB from US census data. hehehe. :-D
05:45.50linageeJT: it took forever to think of who might have that!
05:45.56[TK]D-Fenderhd420 : Only heard that name once elsewhere.
05:46.11linageeJT: then i put all the names into a DB
05:46.29linageeJT: and i wrote a few lines to turn the names into DTMF. that way i can do the lookup easier
05:46.40JTright
05:46.45linageepretty simple really. :)
05:46.46JTbut what about the dialplan logic?
05:46.55linageeJT: haven't gotten there yet. :)
05:47.04linageeJT: i'm just pushing things at the DB. :)
05:47.16linageeselect * from first_name_male where dtmf="123";
05:47.30linageeer, there is no 1
05:47.32linagee234
05:47.45linageeand it tells me there is no 234 match. :)
05:48.01JTdoes the table have the name in ascii too?
05:48.12linageein ascii?
05:48.18JTtext
05:48.20linageeyep
05:48.23linageethat's how i do the lookup
05:48.24JTnot dtmf digits
05:48.27hd420Fender: where'd you hear the name?
05:48.30linageei have four columns
05:48.35JTi thought you said you converted it to dtmf digits
05:48.37JTok
05:48.39linageeid, name, dtmf, timestamp
05:48.59linageeJT: i precached the dtmf. it makes the lookup instantaneous. :)
05:49.16linageei took the names, turned them into dtmf, stored that in the db. that way, i can query that column. :)
05:49.24JTi reckon the neatest way to do it would be to have some code in a stored procedure in the db to convert tthe name to dtmf
05:49.28JTif you often update the db
05:49.35linageeinstead of having to do some weird logic of creating a bunch of possibilieites
05:49.37JTif the db never changes, it wouldn't help
05:49.52[TK]D-FenderStored procedule on DB add <-
05:49.54linageeJT: it's mysql. no stored procedures. :p
05:49.58[TK]D-Fenderprocedure*
05:50.10JTit's mysql, the db can't do much useful :P
05:50.10linageeer, it's not mysql 5 i should say
05:50.19[TK]D-Fenderlinagee : Time to grow up and use PostgreSQL
05:50.21linageeJT: it does what i need it to do
05:50.33linagee[TK]D-Fender: i do use postgresql when i need something slower
05:50.50linageelike when it's transactional and money is involved or something
05:50.50[TK]D-Fenderlinagee : Then you should get a Cisco router for it too! ;)
05:51.07linagee[TK]D-Fender: same thing. when money is involved. :p
05:51.20JTif speed is the name of the game you can use sqlite or a flat file :P
05:51.25linagee[TK]D-Fender: wtf. why postgresql then?
05:51.34linagee[TK]D-Fender: why not oracle. hahaha. might as well go all the way
05:51.47JToracle costs well, heaps
05:51.48*** join/#asterisk tengulre (n=tengulre@221.11.5.182)
05:52.01linageeJT: mysql is one step above flat files. i don't have to worry about the db logic. :)
05:52.18JTyou don't have to with DBD::CSV
05:52.32linageeJT: and not have indexes? :p
05:52.42JTheh
05:52.58linageeanyhow, it works
05:53.11JTwtf, 5pm
05:53.15JTstupid daylights savings time
05:53.17linagee9:53pm
05:53.34JTlinagee: living in yesterday :P
05:53.58linageecan anyone figure that one out? :)
05:54.36JTnile?
05:54.37bluregardumm, I'm guessing I can ignore the fact that the Dundi draft says it expires April of 05 right?
05:54.40linageeJT: mike
05:55.02JTthe asterisk handbook is still a draft, lol
05:55.34bluregardjust making sure.  I figured since it was right from dundi.com that it was the most current.
05:56.26*** part/#asterisk hd420 (n=hdiwan@c-71-202-20-219.hsd1.ca.comcast.net)
05:57.45linagee1,500 names is not that much. heh
05:57.55linageemaybe i could make a recording for each. :-D
05:58.12JTtext to speech
05:58.25linageeJT: doesn't that sound kind of crappy?
05:58.51linageemaybe i could have a custom pronunciation key for certain names it says weirdly.
05:59.52JToh it might sound crappy heh
06:00.05JTbut how does it currently prompt the user if there are multiple options?
06:00.20linageeJT: like i said, there is no prompting.
06:00.32JTwhat if there are multiple answers
06:00.32linageeJT: right now it's all theoretical beyond having a DB that has the names. :)
06:00.39JToh ok
06:01.00JTrecording 1500 name prompts is not an enviable task, anyway
06:01.06linageei have a good idea of how it should work, just haven't gotten there yet
06:01.19linageewould that take forever? heh
06:01.30JTor lots of $$$
06:01.34linageeJT: huh?
06:01.39linageeJT: why hire someone. :p
06:01.46JTto get them professionally recorded
06:01.51JTwell it's one of the two
06:01.54linageeJT: but why professionally recorded
06:02.08JTand you'd have to fix yours up if you don't want them to sound like crap
06:02.11linageenobody likes my voice? :(
06:02.27linageemy voice is already all over my ivr. hah'
06:02.29JTsamplex beginning and ending at a zero crossing, consistant amounts of silence appended and prepended
06:02.40JTs/samplex/samples/
06:02.53linageejbot: you are annoying when you do that
06:03.02jbotlinagee: I think you lost me on that one
06:03.08JTjbot: lart
06:03.23JTjbot: lart #asterisk jbot
06:05.50linageeLOL
06:06.01linageejbot: lart #asterisk #asterisk
06:06.28linageejbot: lart myself
06:06.58JToh gawd, what have i done? teaching people infobot tricks
06:07.02linageehehehe
06:07.08linageejump jbot, jump
06:07.22linageejbot: botsnack?
06:07.22jbot:), linagee
06:10.25linageehah. the db has "napolean"
06:10.29linageeas if you'd be called that
06:11.22linageeMonty = 66689
06:11.28linagee(Monty burns = evil?)
06:12.01linagee66689 28767
06:12.24JThey napoleon, what did you say you did last summer?
06:12.56linagee2326 7263537
06:12.58linageeanyone? :)
06:13.50*** join/#asterisk hoobastooba (n=ckwall@c-67-169-248-217.hsd1.ut.comcast.net)
06:14.02JTyou sure it was napolean and not napoleon?
06:14.11linageeJT: yes it was napoleon
06:14.15linagee(sp) :)
06:14.28hoobastoobahas anyone else ever experienced something like a memory leak in asterisk? it seems as if the longer asterisk runs, the less memory i have.
06:14.50linagee<PROTECTED>
06:14.56Invertedhoobastooba: the longer you live the less memory YOU have
06:15.19linageeInverted: what if you lived forever?
06:15.42Invertedlinagee: then I'd be one dumb & happy fucker :)
06:15.42linageewould your brain cells keep growing and expanding, or would knowledge just leak out?
06:17.02hoobastoobahelpful... at the end of the day when asterisk is no longer in use, it still has all of my ram consumed and pushed me into swap.
06:17.22Invertedhoobastooba: that is how things are supposed to work with linux
06:17.36hoobastoobaso after three days of asterisk use, my server should crash?
06:17.51Invertedyou said nothing about crashing
06:18.04*** part/#asterisk EmleyMoor (i=ejabberd@hallam.tinsleyviaduct.com)
06:18.39hoobastoobausually with linux, it reallocates the swap back to ram, but asterisk is consuming all of my ram.
06:23.16*** join/#asterisk eltech (i=G00Ds@ool-457c9421.dyn.optonline.net)
06:25.45*** join/#asterisk VibroMax (n=phisto@83.229.70.154)
06:26.37*** join/#asterisk saftsack (n=saftsack@pD9E07F61.dip.t-dialin.net)
06:27.28*** join/#asterisk mosty (n=mostynm@203.143.64.82)
06:33.28*** join/#asterisk Tebi_ (n=rantis@gw.aller.fi)
06:34.58mostyanyone awake? i have a problem with realtime queues, it tries to send calls to lines that are busy, instead of waiting until there is a free agent. is this a bug, or something i have misconfigured?
06:38.33*** join/#asterisk joelsolanki (i=joelsola@202.160.161.94)
06:38.43gmustafahello
06:39.21joelsolankiHello
06:39.56joelsolanki<PROTECTED>
06:39.59joelsolankiis this possible
06:40.12*** join/#asterisk Chris-NB (n=chris@ng1.kurtkrenn.com)
06:40.50linageejoelsolanki: yes
06:41.10linageejoelsolanki: that's child's play with freepbx. with just straight asterisk, you will need voodoo. :)
06:41.21joelsolankihow ?
06:41.31joelsolankii tried but it is not reflecting on my mobile :(
06:42.01linageejoelsolanki: does your ITSP allow you to set your outgoing callerid?
06:42.22joelsolankiYes i can ask them
06:43.05joelsolankithey will allow callerid to pass
06:43.15joelsolankithen where to configure in freepbx?
06:43.23linageeare you using freepbx?
06:43.26joelsolankiyes
06:43.36linageethen you should be asking in #freepbx. :)
06:43.40linageeclick the extension
06:43.42linageeit will be in there
06:43.46joelsolankii asked :)
06:43.49linageeand?
06:44.08joelsolankiYes no reply. they might be busy or sleeping
06:48.24*** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com)
07:01.41*** join/#asterisk kristalino (n=kristali@cl-157.dub-01.ie.sixxs.net)
07:08.52*** join/#asterisk zumbush (n=zumbush@c-2cbce253.038-31-73766c10.cust.bredbandsbolaget.se)
07:11.01zumbushanyone got experiance with Cisco 7941G running with asterisk?
07:12.10Qwellzumbush: should work fine with chan_skinny on 1.4 :D
07:12.28zumbushok..thx.. what if i wanna run SIP?
07:13.10hadsUse chan_sip :)
07:13.21zumbushyea... hehe
07:13.30Qwellsip sucks on cisco...
07:13.44zumbushbut im having trouble figuring out how to get a hold of the SIP firmware
07:14.09VibroMaxanybody using a cisco 7960 SIP phone know if it is possible to have more than 4 lines configured on the phone. There are six physical line buttons on the phone and I have configured  six lines in the phone specific configuration file but only four lines are active. Is the problem my firmware version (P0S302002)  or a limitation of hardware ?
07:15.05zumbushhmm maybee i should try the chan_skinny then
07:15.18*** join/#asterisk [Airwolf] (n=airwolf@attilla.nl)
07:16.27zumbushthats smartnet shit i hate... why should i have to pay more money just to get firmware for a product i already bought..grr
07:16.37zumbush:-P
07:18.56zumbushso if i am to run chan_skinny i need to upgrade to Asterisk 1.4 ?
07:19.13Qwell"need"...yeah, pretty much
07:19.25Qwell1.2 MIGHT work, but I would highly recommend upgrading
07:19.36zumbushk.. is the 1.4 considered stable?
07:19.39Qwellno
07:19.42*** join/#asterisk stephane_ (n=stephane@gw.sortilege.net)
07:19.42zumbushhehe
07:19.42Qwellit's still beta
07:19.54zumbushthen i need to stick with 1.2
07:19.59Qwellbut, I don't disagree.  Ciscos tactics are...meh
07:20.07QwellThey don't "get it" :)
07:20.08zumbushyea
07:20.40*** part/#asterisk stephane_ (n=stephane@gw.sortilege.net)
07:20.58zumbushit will bite them in the ass
07:21.19QwellYou'd be amazed if you saw the emails I've gotten from Cisco guys
07:21.30zumbushyea.. gimme an example
07:21.44zumbushshort verison
07:21.45zumbush:-P
07:21.55Qwell"Only 1% of the phones we sell are not used with Call Manager, so we decided that it isn't worth the effort of making them compatible."
07:21.59Qwell...uhh...BS
07:22.05zumbushgah
07:22.22zumbushstupid
07:22.33zumbushlike saying we dont want to sell more phones
07:23.25zumbushi just have this one phone from Cisco that i bought and now cant use, rest is Polycom that ive gotten to like
07:23.38QwellI asked him bluntly...
07:23.57Qwell"I own a Cisco 7960.  I didn't pay for a license.  Am I violating your license?"
07:24.00Qwell"yes"
07:24.44zumbushbuy this phone for this much money.. but if u dont pay us more u just can use it as a paperweight
07:24.52Qwellwhen I asked what I should do; ie, should I sell it on ebay, he gave me no response
07:25.01zumbushlol
07:25.10Qwellfunny, because he said that if you DID use it as a paperweight, you'd be legal :)
07:25.18zumbushhahaha
07:25.29QwellI can give you a direct quote if you'd like
07:25.36zumbushdo that
07:25.41Qwellone sec
07:26.17QwellQuote 1: "The license is a right to use license, basically the right to operate
07:26.17Qwellthe phone firmware that runs in the phones and register it to a Call
07:26.17QwellAgent."
07:26.44QwellQuote 2: "If you buy the phone as a spare, and use it to prop open the door, or
07:26.44Qwellweight down a pile of paper, that's fine. But if you want to operate
07:26.44Qwellthe software in the phone, have it register and process calls, then
07:26.45Qwellthe right to use license should be purchased. "
07:26.51QwellThose are DIRECT quotes from a Cisco sales rep
07:27.08zumbushROFL
07:28.05QwellYou'll love this one - re; 1%
07:28.15Qwell"As to how many people are using Cisco phones on a non Cisco Call Agent,
07:28.15Qwellit's a much smaller number than many people realize. We have measured this,
07:28.15Qwelland it comes out to less than 1% of our shipping phones. When we consider
07:28.15Qwellthe resources required to maintain compatibility, and to verify interop
07:28.15Qwellwith the different 3rd party Call Agents, it just doesn't make business
07:28.18Qwellsense. Yet. "
07:28.40QwellThe moment I read this, I had completely given up on Cisco
07:29.06zumbushas complying to SIP would be a waste of resources
07:29.11Qwellof course
07:29.31zumbushthey are missing an expanding market
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07:30.03QwellYou're preaching to the choir(sp)...
07:30.11zumbushhehe
07:30.20zumbushjust need to get it out of my chest
07:30.34QwellMy response, which was completely ignored:
07:30.38Qwell"Thanks for the answers - they are quite upsetting (and also quite amusing, to be honest.  Cisco *still* just doesn't get it).  Since Cisco obviously doesn't care about the "1%", I will no longer be recommending Cisco products for use with Asterisk or any other PBX or provider (I'd reconsider that, if Cisco were willing to stop the silly practices).  In addition, should I be selling all of my Cisco hardware on eBay?  It seems that I don't have a legal
07:30.40Qwell<PROTECTED>
07:31.22zumbushgood answer
07:31.52Qwelland since then, I've acquired 3 more Cisco phones, he
07:31.53Qwellh
07:32.09zumbushuh...why?
07:32.23zumbushworks good with chan_Skinny?
07:32.31Qwellbecause regardless of the asshattery, I believe skinny is a good protocol
07:32.36zumbushk
07:32.37Qwell(Which Cisco DID NOT come up with)
07:33.16QwellI actually haven't tested the 2 latest phones..  I need to find a power adapter for them, or get a PoE switch or something
07:33.30zumbushil try the skinny then on my 1.2
07:33.36Qwelldon't bother, honestly
07:33.49Qwellthe changes in chan_skinny between 1.2 and 1.4 were immense
07:34.04zumbushu dont happen to have a cfg example file ?
07:34.13zumbushk
07:34.17QwellI'm biased, because I made about 90% of the changes, but it works a whole hell of a lot better now
07:34.26zumbushhehe
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07:34.47zumbushnice
07:34.58Qwell(which is why I find it extremely funny, but extremely pathetic, of Cisco to pull this BS)
07:35.13zumbushi can imagine
07:35.40JTmake your own PoE injector :)
07:36.28QwellJT: easier said than done...
07:36.41Qwellgoogle 12SP+ and 30VIP :)
07:37.01JTit needs some stupid poe signalling does it? :P
07:37.03Qwellthe 12SP I have is actually not even Cisco branded
07:37.19Qwellthe 30VIP has a sticker OVER the Selsius(sp) branding
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07:37.29Qwellvery, very old
07:41.27Qwellyeah...bed time...
07:41.55zumbushgn.. nice talking to u
07:42.04jaikeanyone done stress testing on 1.4? am hoping it has better scalability than 1.2
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07:42.27Qwelljaike: there is a new "Load Testing Task Force"...
07:42.45Qwellsomething like that anyhow.  Consists of 3-4 people so far
07:43.14saam<PROTECTED>
07:43.19saam?
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07:46.22queloHi to all
07:47.26queloI have a problem is there anyone can help me?
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07:48.11queloHi
07:48.13mazzanetherro
07:48.56mazzanetis ${TIMESTAMP} deprecated in 1.4.0-beta3?
07:49.17Qwellthink so, yes
07:49.53mazzanetthat would explain the lack of a timestamp in my monitor files...
07:49.55queloI have downloaded asterisk-addons by svn in trunk
07:50.02mazzanetwhat's its replacement?
07:50.11quelobut when I compile it...
07:50.18queloit returns this errors...
07:50.22Qwellmazzanet: let's see
07:51.06quelonemo:/usr/src/asterisk-1.2/asterisk-addons# make install
07:51.15Qwellmazzanet: probably something with ${STRPTIME()}
07:51.31Qwellerm, STRFTIME?  something
07:51.32quelohttp://paste.debian.net/16661
07:51.45Qwellmazzanet: core list functions
07:51.49mazzanethave all been deprecated in favor of their related dialplan functions.
07:52.10mazzanetwhat the heck is a dialplan function?
07:52.21Qwellit's like a variable, but with args
07:52.27quelois there anyone can help me?
07:52.37Qwellie; ${CALLERIDNAME} is now ${CALLERID(name)}
07:52.38mazzanetah i see
07:53.42mazzanethttp://www.voip-info.org/wiki/view/Asterisk+Detailed+Variable+List
07:53.44mazzanetah here we go
07:54.02queloI'm using Debian/GNU Linux (etch)
07:54.21quelooh no!!! I'm using Debian/GNU Linux (sarge)
07:54.34mazzanetinterestingly, $CALLERIDNUM still works but $TIMESTAMP doesn't
07:54.52Qwellmazzanet: $CALLERIDNUM was deprecated in 1.2 I believe, but never removed
07:58.34quelocan I solve installing libtool?
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08:01.30Zeeeknyuk, nyuk
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08:04.38queloI have compiling problem for asterisk-addons from trunk on my Debian/GNU Linux (sarge)
08:04.46quelohttp://paste.debian.net/16661
08:04.59queloHow can I solve it?
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08:12.23SoftIcehi, can anyone help me with the snom 190 phone?
08:12.49queloI have compiling problem for asterisk-addons from trunk on my Debian/GNU Linux (sarge)
08:12.52quelohttp://paste.debian.net/16661
08:12.56queloHow can I solve it?
08:13.49Zeeekdo you require h323 ?
08:16.33Zeeekpeople of earth attention
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08:16.44Zeeeklook to you skies for a warning
08:17.24queloZeeek how can I solve?
08:17.46Zeeekno idea, I was asking if you need h323
08:17.59Zeeekbecause if not, it may have something to do with the error
08:18.46queloI need some dev libraries ?
08:19.43quelomaybe I need h323
08:20.06Zeeekare you using h323 ?
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08:21.33queloyes
08:24.46linageetoday's random name is randal. :-D
08:25.46Zeeekquelo I can't help, sorry
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08:45.14mazzanethmm
08:45.16mazzanetis the [globals] context deprecated too?
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08:56.12badcfehow do i see what context variables are available in some point of the dialplan.  actually i need the ${VARIABLES} for AccountCode,
08:56.29badcfeDestination and StartTime
08:57.30badcfe?
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08:58.28Zeeekthere is a file called README.variables os something like that
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09:06.34my007msManxPower, are u there
09:09.51Zeeekat this hour? :)
09:11.46my007msso u know him Zeeek
09:11.47my007ms:)
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09:13.05Zeeekya
09:13.21Zeeekbut he has been seen in these parts at weird hours
09:13.23FTexcomsplitttttttttt
09:13.44tzafrirjoin?
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09:14.12ZeeekFrom time to time, I get wrong numbers said to come from something@123.456.123.456
09:14.34ZeeekI think they come directly to a SIP phone
09:14.47Zeeekon a line not registered with asterisk
09:15.19FTexcomZeeek you can disable anonymous sip calls
09:15.23Zeeekin fact, not something, but a US phone number like 17076253454@
09:15.34Zeeekthey come to the phone directly
09:15.42ZeeekI just wondered how and why?
09:16.03ZeeekI believe they are really wrong numbers, not hanckers or spammers
09:16.25FTexcomhave you tried calling them back?¿
09:16.35Zeeekyeah and the last time I did it was like "no I was calling the bank"
09:16.53Zeeekso I'm just curious what scenario would make that happen
09:17.28Zeeekalso got one that said "I was calling another room"
09:17.31Zeeekin fact I think there were two that came from a hotel - it doesn't happen that often
09:17.44FTexcomyou get them everytime?
09:17.57Zeeekget what everytime? The hotel?
09:18.18Zeeekit's often late at night (time diff)
09:18.26Zeeekso I don't usually answer
09:19.03ZeeekI should check the missed calls list and see if the ip is the same, but like I say it doesn't happen often
09:19.26ZeeekI was wondering if it had to do with a voip provider I'm registered to on that line
09:20.05ZeeekThis channel is like confession, I think by describing it, I just figured out what it is :)
09:20.39Zeeekjoin asterisk-psychiatry-couch
09:21.45FTexcomit's a pretty funny thing
09:21.50FTexcomSome years ago
09:22.06FTexcomI had a debt with a ISP
09:22.25FTexcomthey called me around 2 am with an automatic machine saying "you own us...49 euros"
09:22.37Zeeekgreat use for asterisk
09:23.36FTexcommy response was calling they tech support at 2 am saying "you owe two months of a DSL line!"
09:23.49FTexcom*owe me
09:24.28Zeeekbecause they actually had tech support?
09:24.56FTexcom18 year old boys. Working 14 hours for 700 euros...if you want...you can call *THAT* tech support
09:25.43Zeeeksounds like the US Senate pageboys! Did Senators call them with obscene proposals?
09:25.49FTexcomlol
09:26.45FTexcomconservative senators you might add
09:26.55Zeeekglad that elections happened!
09:27.06FTexcomyou're european like me right?
09:27.52Zeeek50-50 USA/French
09:28.03Zeeekexpatriate for 25 years
09:28.14ZeeekI even started voting
09:28.14FTexcomexpatriate from where? USA or france?
09:28.22ZeeekUS Expat in FR
09:28.38Zeeekasked for a got fr citizenship a couple years ago
09:28.50FTexcomyou could always join the foreing legion!
09:28.56Zeeektoo old
09:29.08FTexcomI almost did that...
09:34.16Zeeekwouldn't be cool for me :)
09:35.20queloI have compiling problem for asterisk-addons from trunk on my Debian/GNU Linux (sarge)
09:35.24quelohttp://paste.debian.net/16661
09:35.33queloHow can I solve it?
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09:59.40tzafrirquelo, what version of asterisk do you use?
10:00.02tzafrirYou seem to use add-ons 1.2 and asterisk trunk?
10:00.59tzafrirthat is: the other way around: asterisk 1.2 and addons from trunk
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10:09.58badcfehow do i see what context variables are available in some point of the dialplan.  actually i need the ${VARIABLES} for AccountCode, Destination and StartTime.
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10:24.15skyhawkeri have a wierd problem .. my asterisk stutters a lot .. when i reboot it seems that it works okay for 15 minutes then back to same problem .. it is not hardware related I think
10:24.20skyhawkerany ideas ?
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10:29.05JTskyhawker: what interface?
10:30.13skyhawkerit is on a SIP interface
10:30.16skyhawkerto our sip gateway
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10:35.07backblueany devel here?
10:35.09backbluerussellb: ?
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10:42.33MACscrif i wanted to temporarily disable a trunk in asterisk, as in i just need to stop it from registering, what file do i need to edit?
10:42.34mostyhow can i get the asterisk console to stop showing sip register commands?
10:50.15MACscrmosty, have any idea on my question?
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10:52.48MACscrsweet, i figured it out
10:55.10mostydepends on the type of trunk
10:55.45MACscrwas just a standard sip trunk, i found out which lines i needed to comment out and it worked perfect
10:56.05MACscronly one extension used that line and that extension is not in use, so its no biggie
10:56.37bXidoes one of you have experience running hylafax and iaxmodem on an asterisk?
10:56.37quelotzafir I use asterisk via svf from trunk
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10:58.11mostyhow can i get asterisk to log everything that appears on the console?
10:58.32quelotzafrir I use asterisk via svf from trunk
10:58.50quelotzafrir I use asterisk via svn from trunk
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10:59.07tzafrirtake a look at your build logs. It took headers from an 1.2 tree
11:00.36quelowhere is build log ?
11:02.43quelotzafrir can you tell me how can I solve?
11:02.48Zeeekmosty there is a .conf for that in the asterisk directory
11:03.04Zeeeksomething hard to guess like logging.conf
11:03.10mostyzeeek: i think i have it working now, it's logger.conf - thanks
11:03.19Zeeeknp
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11:16.31rnaapHi2all. Can I get help? =)
11:17.38tzafrirquelo, from what you pasted
11:20.38rnaap... I have a trouble while I try to compile chan_h323 for Asterisk 1.4 beta3 on FreeBSD 6.1... OpenH323 and PWlib versions are equive README. When I try make Asterisk I recieve next messages: [LD] chan_h323.o h323/libchanh323.a -> chan_h323.so   and    /usr/bin/ld: cannot find -lexpat
11:21.19rnaapgmake[1]: *** [chan_h323.so] Error 1
11:21.21rnaapetc....
11:21.34rnaapHelp me, plz...
11:21.39quelotzafrir maybe I solve installing libopenh323-dev ?
11:21.50*** join/#asterisk Tond (n=tond@CPE0014bf30c190-CM0010954a5ffa.cpe.net.cable.rogers.com)
11:21.53tzafrirquelo, no.
11:22.13quelosure?
11:22.31tzafrirquelo, for some reason the asterisk-addons build uses the asters-1.2 version instead of your asterisk trunk version
11:22.53tzafrirPerhaps you need to explicitly point to the asterisk source tree ?
11:23.03Simplixhello, i have no hangup detection with a sip client (nor dtmf detection)
11:23.14TondHi..  I have a small issue i have 2 extentions one _9821.  and another _98. it seems like when i dial 9821xxxx extention _98. gets matched, although _9821. is configured above it.  How can i resolve my issue and get 9821xxxx to match _9821. ?
11:23.24Simplixdtmf is tranmited with sip info
11:24.16quelono I've got asterisk-addons with the command...
11:25.06*** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290)
11:25.07quelosvn checkout http://svn.digium.com/svn/asterisk-addons/trunk/ asterisk-addons
11:26.14rnaapI am put into console pwlibdir=~/pwlib.. export $pwlib... etc..
11:26.28rnaapbut against....
11:27.32tzafrirrnaap, you can pass the path to pwlib to the configure script. If you happen to use Debian, just grab the debs and save yourself the bother
11:27.41quelotzafrir how can I point to asterisk-1.2 to build asterisk-addons?
11:28.01tzafrirquelo, you *should not* point to asterisk 1.2
11:28.14mostySimplix: hangup detection for sip phones should work automatically unless your sip phone is broken
11:28.15tzafrirYou should build addons of trunk with asterisk of trunk
11:28.41tzafrir(1.4 is probably compatible enough by now)
11:28.54tzafrirBTW: at this stage consider using branches/1.4 instead of trunk
11:28.55Simplixmosty: its has worked .....
11:29.06mostytond: you have to include => context1 before context2, the contexts are matched in the order you specify, inside each context there is no guaranteed order of searching
11:29.11quelomy addons IS from trunk
11:29.22tzafrirtrunk of when?
11:29.32quelosvn checkout http://svn.digium.com/svn/asterisk-addons/trunk/ asterisk-addons
11:29.53tzafrirbranches/1.4 is now stabalising and mostly stable. trunk is in the process of morphing (and getting broken, actualy)
11:30.42quelookok I try
11:31.23*** join/#asterisk af_ (n=af@ip-172-242.sn1.eutelia.it)
11:31.31Simplixmosty: it has worked and i have the problem with 2 differents sip phones
11:32.21*** join/#asterisk slvrdrgn (n=Miranda@pd95684d7.dip0.t-ipconnect.de)
11:32.55*** join/#asterisk HarryR (n=Harry@host-83-146-53-46.bulldogdsl.com)
11:33.58*** part/#asterisk Tond (n=tond@CPE0014bf30c190-CM0010954a5ffa.cpe.net.cable.rogers.com)
11:37.43*** join/#asterisk darkskiez (n=mbryars@195-11-205-216.suip.mezzonet.net)
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11:43.27piggieHow does one uninstall Asterisk? make uninstall doesn't work.
11:44.31HarryRrm -rf $prefix/lib/asterisk and the other stuff :)
11:44.32rnaapthnx
11:45.46piggieHarryR: Thanks.
11:47.59quelotzafrir I have installed libopenh323-dev and now the error is...
11:48.02quelohttp://paste.debian.net/16665
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11:49.39rnaaptzafrir... I am use FreeBSD...
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11:50.10*** join/#asterisk coppice (n=chatzill@213.143.17.210.dyn.pacific.net.hk)
11:50.10mostysimplix: what sip phone do you have that cant detect when it's hung up?
11:51.02Simplixmosty: i have 2 sip phone : a Grandstream GXP2000 and a BT100
11:51.47mostySimplix: what does the asterisk console show when you reproduce the problem?
11:52.23Simplix<PROTECTED>
11:52.23Simplix<PROTECTED>
11:53.52Simplixmosty: the phone ring but when i pick up the phone the caller is still ringing
11:54.23Simplixmosty: the caller is iax2
11:54.51Simplixmosty: all sip phone don't work
11:55.05AursSimplix: are you using queue?
11:55.17SimplixAurs: not yet
11:55.40AursSimplix: ok. have had similar problem with sip phones in queue; when 1 picked up, the other phones keeped ringing
11:56.50SimplixAurs: the other problem is that if sip phone call someone (voicemail for example) when i hangup .... the voicemail app doesn't stop
11:59.15rnaapWhich type of SIP phone you`re use?
11:59.33Simplixit's like the phone don't send sip info data or asterisk does't receive them
11:59.45*** part/#asterisk [Airwolf] (n=airwolf@attilla.nl)
11:59.46SimplixGXP2000 and BT100
11:59.53rnaapoh... sorr
12:00.29rnaapBindAddr working finelly?
12:01.04Simplixi can make call with sip phones ....
12:01.15Simplixand called phone ring
12:02.05rnaapCan you put here SIP.conf? ))))))))
12:02.07rnaapsorr ))))
12:02.14mostySimplix: run tethereal and see what packets are flying about
12:02.33Simplixok ... i send sip.conf after that
12:02.52monstedmosty: that's tshark if you're using software that isn't outdated :)
12:04.00mostyi know
12:04.31mostysimplix: what does the called phone hear when you pickup?
12:05.25Simplixmosty: SIP=>IAX i can make a conversation
12:06.29Simplixmosty: but sip hangup is not detected
12:07.33mostySimplix: run tethereal/tshark on the asterisk server, showing packets going to the sip phone's ip. then make a call and see what happens when you hang up
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12:21.24Simplixthe sip phone send a BYE command
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12:21.25*** join/#asterisk mattfletcher (n=matt@heysham-mail.nshc.co.uk)
12:21.25tparcinat38 problems, does anybody have them?
12:21.25Simplixmosty: the sip phone send a BYE command but the server still send data
12:21.25*** join/#asterisk ast_freak (n=jesse@h69-130-171-28.69-130.unk.tds.net)
12:21.25tparcinaI belive that my client has fax that supports t38. and when he tries to send fax to some old fax machine (that doesn't support t38) everything works fine.
12:21.26tparcinabut when he tries to send fax to someone who has fax that also supports t38, then they fax tries to send fax using t38 protocol
12:21.26tparcinathen I have problem... :((
12:21.26mattfletcherHello. http://pastebin.ca/249083 is a copy of the Call Forward code example from http://www.cyber-cottage.co.uk/wiki/index.php/Call_forward (copied to pastebin so I can refer to line numbers) I'm trying to get it to work , and it won't. I spotted that line 23 should refer to ARG2 not ARG1, but it still fails for me. My problem is that different extensions  Iwant to forward to are on different technologies (SIP, Zap and UNISTIM). The dial comm
12:21.26mattfletcherforces me to embed one technology though
12:21.26mattfletcheris there a way round this? The code refers to the technology "Local" but this gives me errors in the console
12:21.26tparcinamy client has panasonic dx600, and they don't know how (or even can it be done) to tourn off t38 support
12:22.40*** join/#asterisk tsurk0 (n=tsurko@80.72.68.86)
12:24.02mattfletcherGot it. It was going to context "default" whereas all my extensions are set in "outbound". Changing it to "exten => s-CFIM,n,Dial(Local/${CFIM}@outbound,30,Ttr)" cracked it
12:28.49backblueanyone use polycoms 601 here?
12:31.14*** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm)
12:34.07quelotzafrir same error for asterisk-addons from branches/1.4
12:35.22quelonemo:/usr/src/asterisk-1.4/asterisk-addons-1.4# make install
12:35.22quelomake[1]: Entering directory `/usr/src/asterisk-1.4/asterisk-addons-1.4'
12:35.23quelomake[2]: Entering directory `/usr/src/asterisk-1.4/asterisk-addons-1.4/asterisk-ooh323c'
12:35.26quelomake  all-am
12:35.28quelomake[3]: Entering directory `/usr/src/asterisk-1.4/asterisk-addons-1.4/asterisk-ooh323c'
12:35.31quelosource='src/chan_h323.c' object='chan_h323.lo' libtool=yes \
12:35.34quelodepfile='.deps/chan_h323.Plo' tmpdepfile='.deps/chan_h323.TPlo' \
12:35.36quelodepmode=gcc3 /bin/sh ./config/depcomp \
12:35.42*** join/#asterisk tsurk0 (n=tsurko@80.72.68.86)
12:35.48quelotzafrir same error for asterisk-addons from branches/1.4
12:36.21quelohttp://paste.debian.net/16672
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12:38.07*** part/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm)
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12:54.59mostyis there a console command that will show me the license id's i have registered?
12:55.05mosty(for g729)
12:55.45HarryRAny help? From a FastAGI (orderlycalls) application, I want to be able to play a sound in the background (IVR menu) then read in DTMF tones... but Background() (and Cepstral()) both block until they've finished playing, or take the first DTMF digit which they shouldn't be
12:55.53rnaapAnybody have chan_h323.so for asterisk 1.4?
13:01.01*** join/#asterisk SwK (n=Silik0nJ@208.44.30.242)
13:02.10fourcheezemosty: show g729
13:03.08mostythat just show's the number available and in use, i want to know the keys
13:03.12*** join/#asterisk apardo (n=apardo@87.217.147.245)
13:05.15*** join/#asterisk Ebola (n=Ebola@host86-136-134-245.range86-136.btcentralplus.com)
13:06.26*** part/#asterisk mattfletcher (n=matt@heysham-mail.nshc.co.uk)
13:11.39fourcheezemosty: not sure if that's done from the console or from another app
13:12.30dlynes_laptopmosty: ls -al /var/lib/asterisk/keys
13:12.38*** join/#asterisk gerphimum (n=trekkie@cpe-68-206-83-62.satx.res.rr.com)
13:13.06dlynes_laptopand on that note...i'm hitting the sack
13:13.11dlynes_laptopg'night peeps
13:13.11mostythanks
13:22.43*** join/#asterisk Sasch (n=Sasch@host102-30-static.107-82-b.business.telecomitalia.it)
13:29.13SaschI have a problem with sip phone ...
13:29.37Saschwhen I recived a call my asterisk direct it into papinicomputer queue
13:30.10Saschand after all telephone ring (my 2 wireless telephone and my 2 grandstream telephone)
13:30.30Saschwhen i call with my wireless telephone a grand stream all work
13:30.45Saschbut when i cal with my grandstream a wireless telephone the Dial() return
13:31.03SaschNov 15 14:25:42 NOTICE[3830]: chan_sip.c:2007 auto_congest: Auto-congesting SIP/10-081a3840
13:31.36*** join/#asterisk jaike (i=jaike@124.106.190.249)
13:31.37Saschcan help me
13:34.55mostyis the phone set to DND?
13:35.02slunkoej if you are about can we discuss the transdirection patch?
13:35.33Zeeektransdirectionality is a personal very private issue
13:37.39oej[training]slunk: I'm here
13:37.55oej[training]slunk: The transdirection patch was merged to svn trunk
13:38.08oej[training]slunk: for the various releases
13:38.14slunkHi  it's stephen dredge. Didn't work for me
13:38.19*** join/#asterisk mosty (i=mostyn@60-241-198-194.static.tpgi.com.au)
13:38.41slunkTags got mixed up when the incoming channel sent a reinvite
13:39.09*** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl)
13:39.36*** join/#asterisk tdonahue-laptop (n=tdonahue@vonmail.vonworldwide.com)
13:39.44oej[training]slunk: With which version?
13:39.58puzzledmorning
13:40.35rnaapGood evening )
13:40.45Zeeekbeau soleil
13:40.48slunkRoot cause is the confused use of the SIP_OUTGOING flag.
13:41.27oej[training]slunk: I know, I know. It's all messed up
13:41.37oej[training]slunk: But which version are you testing with now?
13:42.21slunkI had to insert that change into slightly old svn. I have some mgcp channel changes that i need update to latest to test with same
13:42.55*** join/#asterisk willie (n=willie@itscotland.demon.co.uk)
13:42.56oej[training]slunk: I've changed both 1.2 and 1.4 quite a lot since then. You need to test with new versions, please. Just update chan_sip.c
13:44.18slunklet me just log into work and check version, I have been though the diffs looking for something which should have made a difference
13:45.12mostyi can't get realtime queues to work properly with the rrmemory or leastrecent strategies. sometimes the queue sends multiple calls to one phone (i think it might not be updating the stats until an agent finishes a call, when it probably should also update when an agent accepts a call)
13:45.18*** join/#asterisk mattfletcher (n=matt@heysham-mail.nshc.co.uk)
13:46.10*** join/#asterisk a2lti (n=alti@c.32.169.a475.sta.adsl.cyfra.net)
13:47.48Saschi have two telephone one grand stream and one is samsung wip6000p
13:47.49*** part/#asterisk slvrdrgn (n=Miranda@pd95684d7.dip0.t-ipconnect.de)
13:47.51DrukenHME1.2.12 :)
13:48.13Saschwhen i take a samsung and call grand stream all work
13:48.22ManxPowerDrukenHME: too many reported problems with 1.2.12
13:48.28Saschbut when i call with grand stream my wip 6000p don't work
13:48.30Saschwhy ??
13:48.39ManxPowerSasch: I don't know.
13:48.42jaike1.2.12.1
13:48.48DrukenHMEManxPower: really? i don't have much problem with it at all.....
13:48.57ManxPowerSasch: does "sip show peers" show that the phone is lagged.
13:49.14ManxPowerDrukenHME: try using queues, ChanSpy, or Monitor()
13:49.17mostysasch: set verbose 10 and set debug 10 in the asterisk console, then watch what it says when you try the call
13:49.22DrukenHMEi do use queues....
13:49.36Saschyes 10/10                      192.168.0.6      D   N      5060     OK (100 ms)
13:49.57jaikeManxPower: all those apps work ok on 1.2.12.1
13:50.30Saschhttp://pastebin.ca/249148
13:50.41ManxPowerjaike: I'm still too scared to upgrade.
13:50.47Sasch10 is my wip 6000p and 13 is my grand stream
13:51.01DrukenHME100ms? ouch....
13:51.24jaikeManxPower: know how you feel, been there. wondering if the server will explode on the next upgrade
13:51.37DrukenHMEcourse, my wip300 has a shitty latency as well....
13:51.44jaikebut 1.2.12.1 proved me wrong, am sticking with it til maybe 2.0 comes out
13:51.45DrukenHMEwireless just can't handle it.. hehehe
13:51.46slunkversion 43649 so very old ( has it been that long ? ) I will test with the latest tommorow
13:52.04Saschbut if I call my line the call is redirect to papinicomputer's queue ... and wip6000p ring
13:52.07Saschwhy ???
13:52.23ManxPowerjaike: maybe ?I'll look at 1.2.13 in a few weeks after enough people are running it.
13:52.27Saschwhy when recive a queue ring and when i call with my grand stream don't work
13:53.11Saschis a asterisk bug or is my telephone ??
13:53.12ManxPowerSasch: What version of asterisk?
13:53.25SaschAsterisk SVN-branch-1.2-r46964M built by root @ neo on a i686 running Linux on 2006-11-04 10:38:30 UTC
13:55.07ManxPowertry upgarding to a release
13:55.15*** join/#asterisk zotz (n=zotz@208.196.247.175)
13:55.31slunkthe base request is updated from a received request in 4 places but the outgoing flag is only cleared at two of them
13:55.55Saschok to upgrade i must make zaptel or i can only make new asterisk ??
13:56.21slunkand handle_request i think looks like it expects the flag to be constant for the duration of the call.
13:56.45slunkhas the intention of this flag changed at some point
13:57.42*** join/#asterisk [Airwolf] (n=airwolf@89.205.158.75)
13:57.47ManxPowerSasch: you should always upgrade libpri, zaptel, and asterisk at the same time.
13:57.58Zeeekor not
13:58.20Sasch<ManxPower> ok thanks
13:59.27*** join/#asterisk olivier__ (n=olivier@obs92-4-82-239-116-113.fbx.proxad.net)
13:59.32slunkops meant handle_response
14:00.44*** join/#asterisk devel (n=devel@wiggum.digitalcoven.com)
14:00.59ManxPowerslunk: try #asterisk-dev
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14:04.27*** part/#asterisk Zeeek (n=IceChat7@pdpc/supporter/active/Zeeek)
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14:04.57Chris-NBhi
14:05.29Chris-NBanyone experienced asterisk sending rtp to the private IP of the phone (192.168.2.xxx)
14:06.24*** join/#asterisk danielmendez (n=danielme@201.244.247.15)
14:06.33danielmendezhello there
14:07.10danielmendezi want to  play a DTMF tone when the CALLING party is connected to the CALLED party.
14:07.15danielmendezanyone has some hints?
14:08.17BrokenNozethere's an application that does that
14:08.26ManxPowerdanielmendez: you mean like the D() option to Dial?
14:08.32BrokenNozedon't know what its called.
14:08.49tparcinaT.38, anybody?
14:09.01ManxPowertparcina: no thanks.  It doesn't work.
14:09.09BrokenNozeAnyone know how to set the Callerid in an Originate Manager API function
14:09.27tparcinayes, I know that, but my fax machine always tries to send fax using T38
14:09.29*** join/#asterisk spr1te (n=spr1te@213.227.193.75)
14:09.36ManxPowertparcina: tell it not to.
14:09.44BrokenNozeI'm doing Calledid: <My DDI> but it doesn't set the caller id name???
14:09.51tparcinaand it can send fax to some numbers (fax machines) but can't to other
14:09.52ManxPowertparcina: Your fax supports IP?
14:10.04ManxPowerBrokenNoze: the correct format is:  Name <number>
14:10.14ManxPowerOF course, the name will not be accepted by carriers
14:10.26tparcinait seams so - it's panasonic dx600
14:10.26BrokenNozeSo Callerid: MyDDIName 123456?
14:10.30ManxPowerbro no
14:10.41mattfletcherI want to record EVERY outgoing call made. This involves a lot of different extensions (internal, sip, zap channels etc). Is there a way of calling MixMonitor on every call made out of my "outbound" extension?
14:10.43ManxPowerCallerid: MyDDIName <123456>
14:10.53BrokenNozeAh. star cheers guys
14:11.04*** join/#asterisk dasenjo (n=dasenjo@201.228.128.10)
14:11.05ManxPowermattfletcher: yes, run Monitor before the outgoing Dial
14:11.09tparcinaManxPower: how can I tell it not to use t38? I didn't find anything usefull in manual
14:11.44ManxPowertparcina: I don't know.  Is the fax machine connected direct to your Ethernet or connected into an ATA?
14:11.45danielmendezthe problem is that the D() opttion from the Dial application send the DTMF to the calling party when the call has not been answered yet
14:12.21ManxPowerdanielmendez: it will only do that on analog ports since those are always considered answered as soon as Dialing is finished.  And it sends it to the CallED party, not the CallING party
14:12.21tparcinait's panasonic dx600 - handy tone 386 ata - Asterisk - My SIP provider
14:12.37ManxPowertparcina: then the Handytone is trying to do the T.38
14:12.50tparcinaand I can receive fax from everybody (at least it seams like that)
14:12.54danielmendezi need it to start a process reciveing a # DTMF when the call is answered
14:13.10tparcinaand i can send fax to 70% of people, but those 30% allways fail
14:13.43ManxPowerdanielmendez: Which is it?  Send DTMF to the called party?  Send DTMF to the calling party?  Receive DTMF from the called party?  or Receive DTMF from the calling party?
14:14.11ManxPowertparcina: um, FaxOverVoiceOverIP is not reliable.
14:14.12tparcinaManxPower: you don't think that it's FAX problem, but that it's ATA problem?
14:14.19*** join/#asterisk piggie (n=pig@bb219-74-36-37.singnet.com.sg)
14:14.24ManxPowertparcina: no, I KNOW it is an ATA problem.
14:14.32ManxPoweryour fax machine doesn't know anything about T.38
14:15.05tparcinaManxPower: how do you comment that certin numbers allways go thru and some numbers allways fail
14:15.28ManxPowertparcina: I think it's just the fact that faxovervoiceoverip is not reliable
14:15.31*** join/#asterisk iulius (n=iulius@mail1.technologieshq.com)
14:15.57ManxPowertparcina: T.38 is an IP protocol.  Your fax machine isn't even plugged into the network.
14:15.58tparcinaManxPower: ok, thank you
14:16.44danielmendezI need to dial a number, when the call is answered I need to hear a DTMF
14:16.56DrukenHMEi want to see digital voip fax MACHINES!!!
14:17.04piggieHi all, I've managed to successfully install zaptel and I went back to asterisk to do make && make install but chan_zap.so is not being compiled and installed, any ideas? I'm using the latest beta for asterisk and zaptel.
14:17.09ManxPowerdanielmendez: You need to hear DTMF or the Asterisk Dial Plan needs to hear DTMF.
14:17.22danielmendezI need
14:17.40ManxPowerdanielmendez: that should happen by default and I can't think of anything that would prevent that.
14:18.09ManxPowerdanielmendez: So you call me and I answer and then I dial DTMF and you need to hear the DTMF, right?
14:18.24Chris-NBany ideas why a phone sends his private IP in the sip/sdp paket as it's contact information/owner address?
14:18.36ManxPowerChris-NB: because you don't have nat=yes
14:18.44Chris-NBManxPower, I have nat=yes
14:18.46ManxPowerChris-NB: oh, and it will always do that.
14:18.58ManxPowernat=-yes just tells asterik to ignore the ip in the SDP
14:19.09danielmendezthe sYstem must send the DTMF
14:19.19ManxPowerdanielmendez: which system?
14:19.24Chris-NBManxPower, noop. I've two phones behind a router. one is wired, one is wireless
14:19.39Chris-NBthe wireds sends the public ip, the wireless sends the local IP
14:19.42danielmendezan LCD dysplay that counts the time of the call
14:19.54Chris-NBand asterisk sends rtp to the private ip (for the wireless phone)
14:19.56danielmendezwhen it is answered
14:20.05Chris-NBfor the wired phone, everything works fine
14:20.18Chris-NBonly in the reg. contact i can see the private IP
14:20.20ManxPowerdanielmendez: I can't think of an easy way to do that.
14:20.27Chris-NBin sip/sdp there is only the public one
14:20.34DrukenHMEChris-NB: are these both behind the same router?
14:20.40Chris-NBDrukenHME, jep
14:20.40ManxPowerdanielmendez: The dialplan stops until the Dial is finished.
14:21.30Chris-NBDrukenHME, this router is via dsl connected to the interrnet. my asterisk is behind a firewall connected, connected via another dsl to the internet
14:21.37Chris-NBeverything else works fine
14:21.39ManxPowerdanielmendez: Oh!  Yes, they updated the D() to support both called and calling
14:21.51Chris-NBexcept this wireless sip phone
14:22.08ManxPowerdanielmendez: "show application dial" in the Asterisk CLI, pay special attention to the D() option.
14:22.11DrukenHMEChris-NB: i'd be checking the config of the wireless phone then....
14:22.21ManxPowerI think the feature you want is in 1.2.x and later.
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14:22.37Chris-NBDrukenHME, the phone works fine behind a differend router
14:22.50Chris-NBDrukenHME, so I don't think it's the phone config
14:22.54danielmendezi use someting like  D(:#) but the dial application send the DTMF when the call has not been answered
14:23.02DrukenHMEwell, don't think... make sure :)
14:23.20Chris-NBk, the config of the wired and wireless phone is the same!
14:23.21Chris-NB: )
14:23.29ManxPowerdanielmendez: did you miss my statement "Asterisk considers analog ports to be answered as soon as the Dialing is finished"?
14:23.32Chris-NBone works, the other works ... partly : /
14:23.35DrukenHMEdiffrent control ports?
14:23.47Chris-NBDrukenHME, what do you mean?
14:24.03DrukenHMEi mean is the wired and wireless both trying to use 5060 ?
14:24.12Chris-NBjep
14:24.21DrukenHMEchange one of them to 5061
14:24.23Chris-NBhaven't tried simultanously
14:24.25ManxPowerany decent NAT router will remap the source port of 5060 to a different port.
14:24.33danielmendezwell i actually dont know
14:25.09Chris-NBthe curios thing is, registration and call establishment works fine
14:25.27Chris-NBexcept the rtp data is sent to the wrong ip
14:25.37Chris-NBthe private instead of the public one : /
14:25.50ManxPowerIf the public IP is set in the SDP then the phone is set for nat in additon ot Asterisk
14:25.53Chris-NBand i've absolutely no clue why!
14:26.10ManxPowerChris-NB: the phones SHOULD put the private IP into SDP.
14:26.22ManxPowernat=yes tells asterisk to work around that.
14:26.34Chris-NBManxPower, nat=yes is set
14:26.54Chris-NB<PROTECTED>
14:26.56ManxPowerChris-NB: the the incoming call is not matching the sip.conf entry.
14:26.58Chris-NBfrom that peer
14:27.12ManxPowerChris-NB: you don't want both nat=yes AND NAT stuff set on the phone.
14:27.39Chris-NBManxPower, what do you mean by that?
14:27.47ManxPowerChris-NB: put context=INVALID in [general] the put the correct context= line in each of the sip.conf entries.
14:27.48Chris-NBManxPower, what nat stuff?
14:28.12ManxPowerChris-NB: whatever nat stuff the device supports.
14:28.34danielmendezmanxpower:  what would happen with a PRI E1 or PRI T1 ?
14:28.47ManxPowerdanielmendez: it would not happen with any kind of PRI.
14:29.00Chris-NBManxPower, I've nat=yes and context=intern in the [wirelssphone] context
14:29.22ManxPowerChris-NB: and do you have context=INVALID in [general]
14:29.24ManxPowerhave to leave now
14:29.58Chris-NBManxPower, context=default in [general]
14:30.25Chris-NBManxPower, so the phone has to use the own config?!
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14:36.44piggieHi all, I've managed to successfully install zaptel and I went back to asterisk to do make && make install but chan_zap.so is not being compiled and installed, any ideas? I'm using the latest beta for asterisk and zaptel.
14:37.13mattfletcherManxPower: I saw you answer my question about MixMonitor'ing outgoing calls, but my PC then crashed so I don't know if there were more replies. You said to add MixMonitor before the Dial command. That misses my point however. I have lots of extensions which all need recording, is there any way to add a dialplan command for every extension in a context is I suppose what I am asking
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14:43.09mattfletcherAnyone, is there a way to add a dialplan command (MixMonitor) to every extension in a context?
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14:45.23danielmendezi use someting like  D(:#) but the dial application send the DTMF when the call has not been answered
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14:51.40danielmendezMAnxpower:  checking tje dial application, i ve found the L(x:y:z) parameter, where i can define via a VARIABLE LIMIT_CONNECT_FILE the sound file to be played when the call begins.  do you know that option?
14:52.50danielmendezmanxpower: also, do you know what it means "when the call begins"  ?? the start of the dial process or the call connected between the parties
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15:00.15b11dmorning all
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15:12.23dasenjoHi! I have a language=es line at [general] in sip/iax/zapata/voicemail.conf and loadzone=es,defaultzone=es in zaptel.conf. ¿Why am i getting Playing 'digits/5' (language 'en') in CLI?
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15:14.59Dr-Linux|workhi guys
15:16.13Dr-Linux|worki need some help regarding variables
15:16.30Dr-Linux|workmy question is also here >> http://networks.pk/forum/viewtopic.php?p=17#17
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15:17.19Dr-Linux|worki'm appreciate if someone help
15:17.43Dr-Linux|works/i'm/i'll
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15:22.03ManxPowerDr-Linux|work: SetVar(LANG=${SPANISH})
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15:24.21dasenjoI'm using 1.2, ¿should I file a bug?
15:24.25Dr-Linux|workManxPower: thanks but in which portion i should use this variable you mentioned?
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15:25.57dasenjo¿no help? ¿can you guide debugging the problem?
15:28.12db1310installed asterisk on debian sarge via dist package, reading setup from book AsteriskTFOT. Refers to ztdummy, whereis package and do i need it?
15:28.34ManxPowerdb1310: ztdummy *should* be part of Zaptel.
15:28.45ManxPowerWe really can't help much with distro specific things
15:29.48db1310no zaptel card, just trying to setup a simple soft phone for now to experiment. So I do need ztdummy driver
15:30.16dasenjodb1310, apt-get install zaptel-source; m-a a-i zaptel
15:30.21ManxPowerdb1310: I understand.  The zaptel source contains ztdummy
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15:31.23db1310Ok, so i need to get it build it and load the module. Thanks all
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15:44.39Saschto load kernel module is modprobe .. to unload ??
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15:53.29pifiumorning everyone
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15:57.36db1310reading claims on openpbx.org, how do they compare, seriously. pros / cons Asterisk against OpenPbx
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16:01.05CtRiXdb1310, i cannot speack about openpbx here. I have already been banned once for that reason.
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16:05.32dasenjohey, please, I really need help. I think language support in asterisk is almost «poor». Take in account that not all the world speak english
16:10.08CunningPikedasenjo: There are language bundles available for system prompts, and many people have recorded prompts in their own language. Nothing precludes you from doing this in the language of your choice
16:10.43CunningPikedasenjo: Language support in asterisk is as good as the prompts that people record for it
16:12.06dasenjoCunningPike, you did not see my first question
16:12.15dasenjoHi! I have a language=es line at [general] in sip/iax/zapata/voicemail.conf and loadzone=es,defaultzone=es in zaptel.conf. ¿Why am i getting Playing 'digits/5' (language 'en') in CLI?
16:12.56dasenjoCunningPike, I have prompts, I have digits/es es/digits, all the conf ... and asterisk keep playing en messages, ¿why?
16:13.33CunningPikedasenjo: Hmm- not sure - maybe a buglet - have you tried opening a bug on mantis?
16:14.32dasenjouhmmm .. yes .. that was my second ot thrid question .. but karma still make me get confused :P
16:16.18dasenjo:(
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16:28.56Kattyhihi.
16:29.44b11dihih
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16:41.45L|NUXmog : hey
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16:42.43fileKatty!!?!
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16:48.27L|NUXmogorman : hey
16:48.29L|NUXmogorman : you are late ;)
16:48.29L|NUXhehe
16:48.50mogormancable guy screwed me
16:49.11L|NUXoh
16:49.13L|NUXokay :)
16:49.39L|NUXbtw i just come online
16:49.42L|NUXalmost 10 minutes or so
16:49.42L|NUXi have been to a doctor
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16:49.50L|NUXfor my child vaxination
16:50.20L|NUXmog : its good that you are here now :)
16:50.37mogahh okies
16:50.46moglet me get my box set up shouldnt take but 10 minutes
16:51.02L|NUXso will you find out what was issue ?
16:51.04L|NUXok
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16:58.40DerPraktikanhi ! i got a problem by installing Fedora Core 6 Zod on a 64Bit system, can somebody please help me?
16:59.27DerPraktikani boot from the cd , and startet the grafical installation
17:01.04Kattyfile?!?!?!??!!!!
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17:01.41fileKatty: 'chu at work?
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17:01.53Kattyfile: yesh.
17:02.05filethrilling!
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17:02.32Kattymhmm
17:04.08fileeep
17:04.38zx6r...i just want to buy a melon
17:04.42L|NUXmog : back ?
17:04.52mogstill working yes
17:05.09L|NUXok
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17:11.30*** join/#asterisk WGFreewill (n=chatzill@69-170-244-239.atlsfl.adelphia.net)
17:12.36WGFreewillanyone know how to enable at compile time the new jitterbuffer
17:12.41WGFreewillin 1.2
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17:15.17pifwhat labels should I use in GotoIf() when using 'n' priorities ?
17:15.58sangeehow do setup g723 passthrough?
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17:20.03WGFreewillsangee, for g729
17:20.40WGFreewillwe just put g729 in the sip.conf or device entry
17:20.40WGFreewilland set the devices to prefer g729
17:20.40WGFreewilland the calls pass right through no license
17:20.54WGFreewill(I have read that you need to take t and r out of your Dial command)
17:21.08WGFreewillgot g723
17:21.19WGFreewillshould be just allow=g723
17:21.20WGFreewillin the sip.conf
17:21.22WGFreewillif its supported
17:21.34PumasHi, I have a problem with my asterisk: I have configured asterisk with VOIP service, I must record the incoming calls from VOIP, but when the call has 5 minutes, it stops automatically,
17:21.49Pumashelp me please
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17:22.41mogugh L|NUX my jabber server is still down do to cable, let me try connecting elsewhere
17:22.55L|NUXokies
17:22.56L|NUX:)
17:22.57L|NUXbrother
17:23.01L|NUXyou will take your time :0
17:23.24L|NUXmog : just a favor when you get it fixed email me
17:23.25L|NUXSecRECV[<iq to="dost4u@gmail.com/talk0F42BBAE" id="307" type="error" from="CreativeLimit@gmail.com/Talk.v100FF1A9D0D"><session type="terminate" id="3885069672" initiator="creativelimit@gmail.com/Talk.v100FF1A9D0D" xmlns="http://www.google.com/session"/><error type="modify"><sta:bad-request xmlns:sta="urn:ietf:params:xml:ns:xmpp-stanzas"/><sta:text xml:lang="en" xmlns:sta="urn:ietf:params:xml:ns:xmpp-stanzas">unknown session</sta:text></error></iq
17:23.25L|NUXshit
17:23.25L|NUXsorry
17:24.32Pumashelp me please
17:24.35PumasHi, I have a problem with my asterisk: I have configured asterisk with VOIP service, I must record the incoming calls from VOIP, but when the call has 5 minutes, it stops automatically,
17:24.53Pumassomebody knows why?
17:25.00WGFreewillhow are you setup to record
17:25.03WGFreewillmixmonitor
17:25.09sangeethx
17:26.32WGFreewillPumas how is your recording setip
17:26.33PumasI'm using command record
17:26.51fenlanderL|NUX: that looks like a mismatch on the case of CreativeLimit - Google talk can be picky about the difference betwen creativelimit and CreativeLimit in the initiatior string
17:27.03WGFreewillare you setting a maxduration
17:27.08WGFreewillor is the machine out of disk
17:27.13Pumasbut I just have problems when the call comes from VOIP
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17:27.38Pumasby the way I'm totally new in asterisk
17:27.41L|NUXfenlander : okay thanks brother :0
17:27.52mognot that again
17:27.55WGFreewillget some trixbox they use mixmonitor
17:28.01WGFreewilland it works fine usually
17:28.03mogi could have swarn i fixed that a week or so ago
17:28.03L|NUXmog : yeah i am sorry for that brother :(
17:28.07mogor even farther back
17:28.33PumasI have no maxduration in command Record
17:28.37L|NUXmog : are you talking with me ?
17:29.01WGFreewillPumas: you can specify maxduration 0 for no limit
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17:31.23TommyTheKidI have a minor annoyance.. I have a digium te1xxp and te4xxp, both of them continually "reset" (idle?) B channels. I have set the "resetinteval = never" but it just seems to keep on doing it.. is there some other option I am missing?
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17:31.33Pumasyes, I did that, but when the call is in 5 minutes exactly it stops
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17:32.14WGFreewillI see my channels reset
17:32.21WGFreewillon my 405s
17:32.40WGFreewill<PROTECTED>
17:32.45TommyTheKidthere are options in zapata.conf that seem to indicate that you can disable it, but as far as I can tell, they are ignored
17:33.11WGFreewillPumas: is there silence detection?
17:33.13TommyTheKidwith 96-ish channels it gets a bit annoying when they all hit
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17:33.26WGFreewillTommy: I think the network can request the reset as well asn the terminal
17:33.47WGFreewillTommy: I've never had it take down calls
17:33.56TommyTheKidno, I think it only resets idle channels
17:34.16WGFreewillpossibly a q931 debug or something
17:34.16TommyTheKidhence it being a "minor annoyance" rather than a "major issue" :)
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17:34.28WGFreewillto see if there is a message from the net
17:34.29Pumasit is the the command as i use it exten=>  *7,n,record(/usr/asteriskneitek/${folder}/${file}:wav)
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17:35.18WGFreewill*7,n,record(/usr/asteriskneitek/${folder}/${file}:wav|0|0)
17:35.35WGFreewillI have learned to ignore it actually, not that its right
17:36.09PumasI attached a line to asterisk and I called to it and all works fine, BUT when the call comes from VOIP stops at 5 minutes
17:36.32Pumaswhat is the difference between them?
17:36.39WGFreewillthe one I send has
17:36.48WGFreewillsilence 0 and maxduration 0
17:36.51WGFreewillhttp://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Record
17:37.21JuggieWGFreewill, when you call over voip, what device are you using?
17:37.49WGFreewillSJphone, Xten, Polycom Soundpoint, Cisco 79XX, Grandstream
17:37.56Juggieand, are you actually speaking into it, or laying it on the desk.
17:38.20pifwhat code can I use in Hangup() to signal 'normal call clearing'
17:38.43Juggiejust Hangup should do that.
17:39.04WGFreewillJuggie: dont understand, phones go on desks and tables all around me, or the floor or a bucket
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17:39.41Pumasin fact I was specified these 2 parameters, but it didnt work with VOIP,
17:39.42Juggiei was just wondering if it had something to do w/ silence detection.
17:40.09WGFreewillahh the silence was for Pumas, record silence detection
17:40.57WGFreewillare you setting a max threshold on the voip call
17:41.03WGFreewillto cut it after 5 minutes
17:41.09WGFreewillcan you call and echo test for more than 5 minutes
17:41.38Pumaswhere I can configure the threshold?
17:41.55JuggiePumas, my initial reponse would be, how are you testing this,
17:42.04Juggieare you just calling in and putting the phone on the desk.
17:42.07Juggieand waiting 5 minutes?
17:42.14Pumasno
17:42.42WGFreewillPumas: AbsoluteTimeout(seconds)  or setting the timout in the voip Dial command
17:42.43L|NUXmog : i am going see my email and email me
17:42.45Juggieare you sending voice down the line for 5 minutes?
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17:43.56PumasI'm recording the call, I call to VOIP number and my server takes the call, and starts to record the incoming call but when the call has 5 minutes flat it stops the recording
17:44.29Juggieright, but what are you recording
17:44.38Juggieare you talking for 5 minutes, is the phone on the desk, etc.
17:44.50Pumasin this case my voice
17:45.17WGFreewillyou dont daydream and stop talking?
17:45.35JuggiePumas, in your logged.conf enable your full debugging, restart *, make this happen, and then pastebin.ca the 'full' log file.
17:45.41Juggieer, logger.conf
17:45.59WGFreewillshould tell you why the call went down
17:47.01pifwhen getting a BUSY from a zap channel I don't want to relay the busy to the SIP agent, only playtones(busy) and hangup
17:47.22pifbut somehow the SIP client get the dialstatus after my hangup
17:47.22Juggiepif, why?
17:47.24PumasI did another test, I attached a line to Asterisk and call to it, and I'm able to record with no time limit,
17:48.00Pumaswhy just  the voip has this problem?
17:48.15Juggiepumas, follow my instructions and we'll find out.
17:48.23Pumasok, thanks
17:48.25Juggieenable full debug, make it happen, then www.pastebin.ca the output.
17:48.25WGFreewillyes pumas the logs
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17:48.31PumasI''ll do it
17:48.32Pug|WorkSo, about trixbox...
17:48.43WGFreewill#amportal
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17:48.48hmmhesays#freepbx
17:48.49Juggiefor trixbox join #freepbx/#amportal
17:48.53WGFreewillahhh yes
17:48.56WGFreewillnew irc and everything
17:49.13hmmhesaysit must be cold where this chick is performing her stand up
17:49.14WGFreewillhow about that digium GUI
17:49.26WGFreewillslick ajax
17:49.30JuggieWGFreewill, as for the b channels resetting thats nothing to be conerned with.
17:49.39WGFreewillyes I never care
17:49.48WGFreewillbut I see it happen on all ten servers I have
17:49.53Juggieits a tie over from days when asterisk and switches were less friendly to each other, it could probally be removed, but its not a problem.
17:50.07Juggieyeah, asterisk resets all free b channels on the top of every hour.
17:50.27WGFreewillwhat I really need is that new jitterbuffer
17:50.36Juggieit wont affect calls in progress or anything like that.
17:50.40WGFreewillI implemented a voice over wireless
17:50.47WGFreewilland we are getting killed
17:50.59TommyTheKidok, another minor annoyance... when I dial out to a SIP gateway (cisco) everything is "normal".. however when I dial out using ZAP (pri) it sounds like I am getting a double ring sound... one will start maybe 1/4th of a second before the other then both of them at once then the second one finishes. I have seen many posts about this in mailing lists, but never really seen a resolution. It almost seems like (even without the "r") the Dial comman
17:51.37JuggieTommyTheKid, thats because analog sucks,
17:51.44WGFreewillyour signalling
17:51.48TommyTheKidanalog? who is using analog?
17:52.04WGFreewilltry switching the switch and the ZAP
17:52.16Juggieoh, i didnt read closely hah, that sounded like an analog problem.
17:52.16WGFreewillisdn is supposed to tell you inband or out of band ring
17:52.32Juggieyeah, make sure everything is set to out of band.
17:52.32WGFreewillyou seem to get both
17:52.33TommyTheKidDigium 412p to an Avaya PBX.
17:52.49JuggieTommyTheKid, what switch type?
17:52.53TommyTheKid.. and I am not saying for sure its not all the Avaya
17:53.11WGFreewillI line national, dms-100, 5ess
17:53.17TommyTheKid1,1,0,esf,b8zs,yellow
17:53.21WGFreewillbut I have had to switch both sides
17:53.30TommyTheKidoh, national
17:53.31WGFreewillin some cases to make it work right
17:53.43JuggieTommyTheKid, make sure you are using out of band signalling.
17:53.51WGFreewillDMS-100 doesnt like national
17:53.55WGFreewilleven though both ends support it
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17:54.01WGFreewillyes
17:54.05Juggiei use all dms100
17:54.15WGFreewillI use mostly dms-100
17:54.21WGFreewillbut you know
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17:54.25Juggiecanada is still mostly dms100
17:54.25WGFreewillI have an EICON card
17:54.34WGFreewillwont link to the freaking thing
17:54.37Juggiethough i think we are moving to 5ess
17:54.41JuggieEICOM?
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17:54.58WGFreewillyeah think real expensive PCI T1 card
17:55.07Juggieyah, they bought dialogic
17:55.12WGFreewill(has neato t.30 fax DSPs)
17:55.45ucfMethodis anyone here in the DC area?
17:55.58WGFreewillhow many miles?
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17:56.48WGFreewilljuggie: i have the eicon connected to a sangoma, then sangoma to the nortel works
17:57.54JuggieWGFreewill, you should report that to digium then.
17:58.00Juggieif you cant make them link.
17:58.14TommyTheKidso, I inherited the "zap" portion http://pastebin.ca/249333 of asterisk... I was told this is what works.. back when we had a singe PRI card.. now we have the 412p.. I don't remember the double ring before, but I may not have been paying as much attention before I was the one managing this... does that look setup correctly? (I added the extra "channel" secons and duplicated the "span/bchan/dchan" sections
17:59.48Strom_CTommyTheKid, are you setting up four completely separate PRI spans?
17:59.59TommyTheKidI think so
18:00.22Strom_Cwell, better to find out definitively yes or no before you go any further :)
18:00.23TommyTheKidthey are in one trunk group on the PBX, but as far as I know, they are not sharing D-channels or anything
18:00.49TommyTheKid"the PBX" in this case should be considered as my "telco" :)
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18:01.11Strom_CTommyTheKid, verify how you have things set up on that pbx
18:01.50TommyTheKidwhat do I ask them.. I need specific questions, cause I have to break thru like 2 or 3 tiers of "support" before I get to the people who have a clue :)
18:01.58quidproHmm... anybody using a SPA-3000/3102 here?
18:02.24Strom_CTommyTheKid, "Are these individual PRIs, or do these multiple T1 spans comprise an NFAS group?"
18:02.45Strom_C"NFAS" is pronounced "En-Fass"
18:02.53Strom_Cer, EN-fass
18:03.07arcaninecan i use a generic fax/modem card in replace for the rhino fxo crd
18:03.39JuggieTommyTheKid, take a look at the sample zapata.conf file, and look at some of the options you are missing, like inband/outband settings, make sure you setup out of band etc.
18:07.14TommyTheKidOK, I am sure its not NFAS, the people who set it up are paranoid :) also the PRIs were originally NOT all plugged into the same place. We did change them to all be part of the same "trunk group".. but decided not to setup shared D- channels.
18:07.29Strom_Cwell....define "trunk group"
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18:07.38TommyTheKidI will take a look at the zapata.conf.sample... it seemed to have a lot of "nonsense" it in for analog type stuff
18:07.47Strom_Cbecause "trunk group" to me would seem to suggest NFAS
18:08.32TommyTheKidthats what I thought too, essentially if we have 400 DID's and 4 PRI's one of those DIDs can use all 92-ish lines
18:08.57Strom_CTommyTheKid, find out for certain
18:09.05TommyTheKidwe have only ever had about 60 concurrant calls tho ;)
18:09.11TommyTheKidconcurrent ;)
18:09.18arcaninewhat most suited digium card for inbound toll-free
18:09.24arcaninei hav 4 analog 1-800 lines
18:09.49Strom_Carcanine, for four analog lines, either the tdm400p or the tdm2400p
18:10.21fileStrom_C: where... are you!
18:10.46Strom_Cfile, with Mancini and Jan
18:11.01fileyay Jan
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18:13.47Pumasin logger.conf I have this line: full => notice,warning,error,debug,verbose, but how can I tell asterisk where to write the log output?
18:15.02Pumasare you there WGFreewill?
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18:16.25TommyTheKidi have a ticket in.. meantime I will poke around in zapata.conf.sample looking for inband/outofband stuff .. unfortunately, it seems you can't just "zap reload" :) .. so I can't really test anything till I can do a full restart
18:16.33WGFreewillyeah
18:16.39WGFreewillsorry
18:16.57Kattyumm, umm, file.
18:16.58brif8Hi all,  are 1.4 questions asked here  or on asterisk-gui  I can't assign Service Providers ?
18:16.59Pumasin logger.conf I have this line: full => notice,warning,error,debug,verbose, but how can I tell asterisk where to write the log output?
18:17.23Kattyfile: is there a way to make the cli dump out active extensions?
18:17.36Pumasor in what file is written the output?
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18:18.35WGFreewillPumas: /var/log/asterisk/full
18:18.49TommyTheKid[asterisk@iml-v20z-11 asterisk]$ grep log asterisk.conf
18:18.50TommyTheKidastlogdir => /var/log/asterisk
18:19.06Pumasah ok, thanks
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18:19.36Strom_CI have twelve T1 red alarms all blinking in unison on the back of this box
18:19.39Strom_Cit's like christmas
18:20.01mhz121Anybody know if there will be a performance hit by putting a TDM400 and a TE110P in the same server?  I have a client that uses modems to dial into Air conditioning units and wants to use his new T1/PRI line to dial out with the modems (did any of that make sense?)
18:20.38WGFreewillpumas: you'll probably spot the problem
18:20.50TommyTheKidmodems dial into air conditioning units..
18:20.54KattyStrom_C: yay christmas!
18:21.01Strom_Cmhz121, explain that one more time?
18:21.03mhz121Yep, I was surprised also.
18:21.17Strom_Cwhat do they want to do - connect the modems to the tdm400?
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18:22.24TommyTheKidguessing.. connect the tdm400 to PC's (modems) with some software .. and use the PRI lines to dial out?
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18:29.41TommyTheKidwe had some sort of "virtual modem" software that linked into some expensive fax thing on a NT PC at one company I worked at.. that would be kinda cool
18:29.41mhz121Client wants to purchase a T1 line and send it directly into the * server, I was thinking of using a TE110P to handle that.  However he has analog POTS lines that he would love to ditch and instead use channels on the T1 when needed.  so I thought of also putting in a TDM400 (with FXS modules) so that when the modem dials it creates a channel on the T1.
18:29.41luke-jr_Any way to test if the local telephone company is connected to my apt's wall wiring?
18:29.42luke-jr_short of dialing 911
18:29.42Strom_Cmhz121, ah ok
18:29.42Strom_Cmhz121, if you're doing any kind of data calls at all, you dont want to bridge across the pci bus
18:29.42aydiosmioluke-jr_: listen for a dialtone.
18:29.42Strom_Cmhz121, so i'd suggest a TE2xxP and a channel bank
18:29.42luke-jr_mhz121: I've heard of dynamic phone/data balancing that uses PPP to establish bridged data paths over voice lines
18:29.42luke-jr_aydiosmio: lack of dialtone only means no service, not disconnected wiring
18:29.42Strom_Cluke-jr_, check for talk battery
18:29.42luke-jr_aydiosmio: usually, the wiring would still be connected and dialing 911 would still work
18:29.42luke-jr_talk battery?
18:29.43Strom_Cluke-jr_, yes
18:29.43Strom_Cplug an analog phone in and blow into the transmitter
18:29.43Strom_Cif you get sidetone, you have talk battery
18:29.43luke-jr_hm... an analog phone :)
18:29.43luke-jr_what's sidetone?
18:29.44luke-jr_eg, if I can hear myself?
18:29.44Strom_Cyes
18:29.52fileStrom knows too much... kill him!
18:29.53aydiosmioluke-jr_: usually. but one would assume no dial tone, no dialing
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18:29.56L|NUXbye all
18:30.20luke-jr_Strom_C: if I can't, does that more-or-less guarantee they're unwired?
18:30.21mhz121So, if I understand, better bet is to get a two channel T1 card (TE2xxP) and use one channel for the inbound feed (from service provider) and another channel as a feed to a channel bank?
18:30.21luke-jr_aydiosmio: usually you can dial 911 despite lack of dialtone
18:30.21mhz121Any recommendations on a channel bank?
18:30.21aydiosmioluke-jr_: this telus guide says that if you can hear yourself blowing into the reciever, there is service
18:30.21Strom_Cmhz121, adtran
18:30.21Strom_Cluke-jr_, well, make sure its not a phone that relies on an external power source
18:30.21aydiosmioassumign you're using a line powered phone
18:30.22aydiosmioheh
18:30.22luke-jr_Strom_C: the one I have does, but works w/o external power minimally
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18:30.22TommyTheKidit would be cool if there was some sort of "fake modem driver" that could make a "sip/iax2" connection into asterisk and dial out :)
18:30.22luke-jr_TommyTheKid: there is in development
18:30.22luke-jr_but it doesn't speak SIP/IAX2 natively
18:30.22aydiosmioTo check whether your telephone circuit is busy or has been destroyed, blow into the receiver mouthpiece. If you cannot hear yourself in the earpiece and your telephone is connected to the wall jack, the circuit may be out of service and you will have to try another phone.
18:30.22vader--anyone ever link asterisk with nagios in here?
18:30.22vader--im looking for a way my nagios server can send something to asterisk so asterisk can call me if there is a problem
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18:30.23TommyTheKidsorry.. back to my config (shuts up)
18:32.59luke-jr_yay no AT&T junk
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18:42.33DrAk0SXanyone here using Sugar CRM with Asterisk integration?
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18:49.25ucfMethodanyone know why cell phones wont display CALLERID(name)?
18:49.37TommyTheKidthey don
18:49.54TommyTheKidoh yea, mine does cause it looks them up in the phone book :)
18:49.58Strom_CucfMethod, because mobile phones have never had caller ID name delivery
18:50.09ucfMethodStrom_C: thanks...
18:50.39zx6rucfMethod, depends on your provider.
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18:52.12hoobastoobaon the polycom phones it only shows 9 of the 10 digits of a telephone number, because (I am assuming) the call timer takes up the rest of the char spaces... is there a way to change this? or am i the only person with this issue?
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18:54.56diclophis-workhello all
18:55.06diclophis-worki have a quuestion about timezones
18:55.24aydiosmioum
18:55.38diclophis-workwrong window
18:56.11TommyTheKidhoobastooba: I have a 500sp that displays it fine
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18:58.28mhz121Strom:  Looks like channel banks start at $1000, we're trying to keep the costs down (hence the $350 TDM400 idea).  Do you not think that's reasonable?
18:59.30mhz121With only four modems that are used occasionaly it's really not a justifiable expense
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19:03.09drcodehi all
19:03.21drcodeI want to call to zap card
19:03.36drcodeI need to load somthing in module.conf?
19:04.29mhz121Thanks all!!!
19:04.33mhz121Have  agood day
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19:05.06CunningPikeIs there a way to expand a regexp within Asterisk to dial all extensions that match a pattern?
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19:14.13hmmhesaysdo you capitalize a month in a sentence?
19:16.31CunningPikehmmhesays: ??
19:17.14aydiosmiohmmhesays: yes
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19:18.02CunningPikehmmhesays: Misread your question - yes, you do, as aydiosmio says
19:18.19hmmhesaysit is considered a pronoun right?
19:18.44CunningPikehmmhesays: I think it's a regular noun
19:19.00hmmhesayshmm
19:19.04hmmhesaysbeen too long sing english class
19:19.09CunningPikehmmhesays: iirc, a pronoun is a word like 'it', that refers to something that is itself a noun
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19:20.22hmmhesaysthe wikipedia confirms that
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19:21.58QbYi have mailboxes in a context, is it possible to send a message to every mailbox in that context?  there are about a hundred or so
19:22.00Kattyhmmhesays: mew.
19:22.29hmmhesaysKatty: how are you?
19:22.49Kattyhmmhesays: i'm meh.
19:22.56hmmhesaysyeah about the same here
19:23.14hmmhesaysgot a couple projects that aren't going as well as planned
19:23.44Katty:<
19:24.05hmmhesaysmsn is acting up today too
19:25.19*** join/#asterisk gmfm (n=aaron_pi@rtr.enterprisemtg.net)
19:26.11gmfmNov 15 10:33:48 WARNING[26275]: chan_zap.c:8396 pri_dchannel: Ring requested on channel 0/23 already in use on span 3.  Hanging up owner.
19:26.12gmfmNov 15 10:33:48 WARNING[26274]: chan_zap.c:8396 pri_dchannel: Ring requested on channel 0/2 already in use on span 1.  Hanging up owner.
19:26.29gmfmi'm getting wierd PRI errors and can't make outbound calls reliably today
19:26.41gmfmjust started today on a system that's been working for months
19:26.45*** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com)
19:30.58hmmhesaysi'm not gonna lie, i'll not be a gentlemen
19:31.03hmmhesaysbehind the boathouse
19:31.07hmmhesaysI'll show you my dark secret
19:31.49*** join/#asterisk yogurt2ungue (n=yogurt2u@host236.200-117-208.telecom.net.ar)
19:33.08*** join/#asterisk maverickbna (i=sentinel@wikipedia/Shadowhntr)
19:33.08*** join/#asterisk FastFeet (n=fastfeet@CPE0013109fd25b-CM000f9fa60d7a.cpe.net.cable.rogers.com)
19:33.25hmmhesaysthis ds9 episode where they are flying the defiant manually is funny
19:33.40hmmhesaysthey can warp but they can't get the com system up
19:36.29hmmhesayshey file
19:36.31zx6rwoah.. random possum kingdom :|
19:36.43hmmhesayszx6r: yeah I had to learn it for practice tonight
19:36.43hmmhesayswoo
19:37.08*** join/#asterisk _rnz- (n=jungzzz@66.0.46.210)
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19:40.54*** join/#asterisk drcode (n=chatzill@87.69.59.186.cable.012.net.il)
19:40.59drcodehi all
19:41.08drcodeI got unable to connect ZAP device
19:41.13drcodeany idea?
19:41.19drcodeI compile ZAP module
19:41.22drcode1.2
19:41.41drcodeand I am using Astrisks from debian on ubuntu
19:44.03*** join/#asterisk etaoins (n=ryan@209.139.250.204)
19:44.34etaoinsAnyone have problems with one way audio on the Polycom SoundStation 4000?
19:45.11etaoinsIf I call from a Grandstream to the 4000, audio works both ways
19:45.33etaoinsIf I call the the 4000 to the Grandstream, audio only goes from the 4000 to the Grandstream
19:45.59etaoinsIf I dial out from the 4000 through zap, audio only goes from Zap to the 4000
19:46.08_rnz-etaoins
19:46.14_rnz-why use asterisk, when you can use Call Manager?
19:46.29_rnz-._.
19:46.44_rnz-is the budget a little low? :)
19:46.52etaoinsAre you trolling?
19:47.15_rnz-No Ryan, but you may start cumming
19:48.58etaoinsWay to use NickServ
19:49.07etaoinsAnyone else have any ideas?
19:49.46*** join/#asterisk osiris (n=osiris@c-71-205-27-131.hsd1.mi.comcast.net)
19:50.20CunningPikeetaoins: Could it be a codec issue?
19:51.06ucfMethodanyone know if there is a website that has all the standard asterisk sounds so I dont have to constantly create a test extension and use Playback to hear the sounds?
19:51.25etaoinsCunningPike: Both ends are set up for ulaw
19:51.37etaoinsCunningPike: I can send you sip debug if you'd like
19:51.49CunningPikeetaoins: Pastebin it so others can see also
19:52.39_rnz-pastebin.com is mad slow
19:52.41_rnz-the past few months
19:52.46_rnz-i use pastebin.co.uk now
19:52.54drcodewhy I can't run zap show channel?
19:52.56drcodeany idea
19:53.01drcode"zap show channels"
19:53.07drcodeno such command
19:53.24etaoins<PROTECTED>
19:53.42etaoinsThat's a call from the 4000 to a Grandstream
19:54.00etaoinsWhere audio only passes from the 4000 to the Grandstrean
19:54.06*** join/#asterisk Luke-Jr (n=luke-jr@user-0c93tj3.cable.mindspring.com)
19:55.54etaoins192.168.97.231 is the Polycom
19:56.00etaoins192.168.98.8 is the Asterisk server
19:57.49drcodeany idea why "zap show" dosnt work??
19:57.53*** join/#asterisk luke-jr_ (n=luke-jr@user-0c93tj3.cable.mindspring.com)
19:58.01*** join/#asterisk eivindtr (n=eivindtr@062016176152.customer.alfanett.no)
19:58.11dlynes_laptopdrcode: load chan_zap.so
19:59.01*** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com)
20:00.10drcodeno such file
20:00.12drcodedlynes_laptop:
20:00.38dlynes_laptopdrcode: that's why you don't have zap show channels then
20:00.47dlynes_laptopdrcode: you didn't build the module
20:00.50drcodeI use astrisks from debian
20:00.57FastFeetI need a recommendation for some hardware. I need a device which will allow me to use my existing office telephones (3 users), and allow output to a PSTN line.
20:00.57drcodeI need to recompile it or somthing ?
20:01.01dlynes_laptopdrcode: Did you install zaptel and libpri?
20:01.22drcodezttools give me ok
20:01.26drcodelibpri?
20:01.29*** join/#asterisk alamantia (i=alamanti@nat/digium/x-49cc8f76fec0dd96)
20:01.29drcodeits module?
20:01.34dlynes_laptopdrcode: I didn't say zttools
20:01.38dlynes_laptopdrcode: i said zaptel
20:01.48dlynes_laptopdrcode: i have no idea what zttools is
20:01.50drcodeyes
20:01.56drcodewhat is libpri?
20:02.10dlynes_laptopdrcode: it's a library for T1 lines
20:02.24drcodeI use x100p
20:02.27dlynes_laptopdrcode: but if I remember correctly, chan_zap.so doesn't get compiled if it's not there
20:02.47drcodeits in zaptel module?
20:02.53dlynes_laptopdrcode: no...it's separate
20:03.09drcodehow I can download it?
20:03.19dlynes_laptopdrcode: install zaptel, then libpri, then asterisk, then asterisk-addons
20:03.32dlynes_laptopdrcode: you can install asterisk-sounds at any part of that cycle
20:03.49dlynes_laptopdrcode: www.asterisk.org has all the code available for download
20:04.08dlynes_laptopdrcode: you can also try asking on #debian to find out what the package name is you have to download for it, if you want to use apt-get
20:04.14*** join/#asterisk martin5519 (n=marty@host-209-50-87-3.dyn.295.ca)
20:04.43FastFeetI need a device recommendation that will allow me to use my internal phones (3 users), PSTN connect, and Ethernet. I was looking at the SIPura SPA3000, but I am unable to locate them, as they have been replaced by Linksys models. Will they work?
20:04.54martin5519Hello... can anyone tell me if they have ever used Asterisk to deliver VOIP to an appartment building?
20:05.26dlynes_laptopmartin5519: what's the difference between an apartment building and any other building?
20:05.31monstedmartin5519: someone has - anything special you want to do?
20:05.32FastFeetlol
20:06.02dlynes_laptopFastFeet: you're wanting to use how many analog phone lines and how many analog phones?
20:06.14*** join/#asterisk alerios (n=alerios@190.24.100.110)
20:06.24FastFeetI I have 3 analog phones.
20:06.31FastFeetI telco line.
20:06.34FastFeet1
20:06.45martin5519monsted... nothing fancy..
20:06.53dlynes_laptopFastFeet:  I would get a tdm400p with 3 fxs ports and 1 fxo port, myself
20:07.00monstedwill the phones be sharing a single FXS port or three seperate?
20:07.05dlynes_laptopFastFeet: throw that into a pc
20:07.24dlynes_laptopFastFeet: or if what monsted says, then one fxs port and one fxo port
20:07.33FastFeetSupport PSTN?
20:07.34martin5519monsted.. essecntially, just deliver on a Amphonol interface and make sure the Asterisk box supports all the features and that users can active those features from their phone and that Asterisk can pass all calls to a VOIP gateway at anothr elocation
20:08.40martin5519We had problems with Quintum... Mitel and Cisco are outrageously expensive
20:08.43dlynes_laptopmartin5519: so you're wanting to use the existing cabling infrastructure, and hook up to the 24 pair demarc via amphenol tail?
20:08.48FastFeetBTW, I have a low budget...
20:08.50martin5519exactly!
20:09.03dlynes_laptopmartin5519: or 50 pair demarc, depending on the size of the building
20:09.16FastFeetMy intenstions for this Asterisk box, is for a College Project.
20:09.18dlynes_laptopmartin5519: yeah...no difference between an apartment building and any other building for that
20:09.25dlynes_laptopmartin5519: how old is the building?
20:09.59monstedmartin5519: cisco may be more expensive, but you usually get what you pay for
20:10.00martin5519Well, the boards I saw on Digium that support the Amphenol are 24 ports - that's cool anyways... we can add later..  oh, these buildings have CAT3 and are probably about oh.. I do't know 30 to 50 years old maybe
20:10.16*** join/#asterisk oej_ (n=oej@apollo.webway.se)
20:10.29dlynes_laptopmartin5519: Yeah...you'll want to get the tdm2400p's with hardware echo cancellers then
20:10.41martin5519monsted:  Big difference though between $30,000 solution and $3,000 box that I can even customize in Perl
20:10.52dlynes_laptopmartin5519: You'll probably also want to get the single port or dual port pri cards with hardware echo canceller, too
20:10.54martin5519dlynes: And echo cancellers?
20:11.20martin5519Well, we don't need PRI cards though as we're passing off the calls via Ethernet to a VOIP Peer
20:11.23dlynes_laptopmartin5519: then you can have a pri/t1 coming in, instead of a 25 pair amphenol
20:11.40dlynes_laptopmartin5519: ah...ok...nvm then
20:11.43monstedmartin5519: oh, i didn't mean call manager, but their gear in general
20:12.12martin5519monsted... show be a router that can do all that for less than 5 gran.. and I'll be interested - my expertise is actually Cisco
20:12.30dlynes_laptopmartin5519: there are other solutions from sangoma that use regular rj9 jacks instead of amphenol tails, as well
20:12.55SheriF_SpacEdose asterisk 1.4 supporsts video confrancing ?
20:13.06martin5519dlynes.. yes but the problem is the lines come down from teh appartments onto BIX blocks so then we just punch down an amphonel cable on the bix
20:13.23monstedmartin5519: 24 port FXO thingies? no idea, i only work on the server end of things :)
20:13.28monstederr, FXP
20:13.37dlynes_laptopmartin5519: yeah..I would just use cross connect wire to hook up the rj9's
20:13.47dlynes_laptopmartin5519: you must be in Canada :)
20:13.54martin5519monsted - not sure what you call that interface - all I know it's a 25-pair Amphenol...
20:14.07dlynes_laptopmartin5519: And it must be all new wiring
20:14.08martin5519LOL! I am actually! how did you know?
20:14.17dlynes_laptopmartin5519: because americans don't use bix blocks :)
20:14.28martin5519oh.. how does it get wired?
20:14.46martin5519Come to think of it, I've never seen a Telecom Closet down south
20:14.46dlynes_laptopmartin5519: all the old crap in Canada is all 110 block or 66 block
20:14.57monstedmartin5519: i'm pretty sure they have a media gateway that does SIP/MGCP/Skinny/H323 with just about any kind of interface you need
20:15.00martin5519We;re behind..
20:15.04dlynes_laptopmartin5519: all the newer stuff is bix, and all the really new stuff is a newer standard
20:15.27dlynes_laptopmartin5519: Telus just started experimenting with a new standard that has gel in the slots
20:15.32dlynes_laptopmartin5519: and it's toolless
20:15.44dlynes_laptopmartin5519: so even alarm guys can figure out how to use it
20:15.49martin5519ok.... well, what do you suggest?
20:16.10martin5519I'm no cabling expert by any means... if there's a better way, i'm all ears
20:16.22dlynes_laptopmartin5519: well, it's a new enough job...check to see if the cabling's new enough to be cat5
20:16.25fileonly ears? no body?
20:16.42martin5519Right now, in the buildings, we've been redoing the cross-connects so that it connect to our amphenol - that's all I  know.
20:16.51dlynes_laptopmartin5519: most bix strips are cat5e compliant
20:17.13martin5519We can't start running cat 5 to every appartment - too expensive
20:17.34dlynes_laptopmartin5519: ok, so it's all cat3 for sure?
20:17.51martin5519I'd have to double check but probably yes
20:18.10dlynes_laptopmartin5519: if so, you're going to need echo can for sure...if it's cat5e you might be able to scrape buy with just a software echo can
20:18.12martin5519It just comes down and currently punched down on the Bell BIX... I have pics too... on my laptop though
20:19.17dlynes_laptopmartin5519: the cables are generally labelled
20:19.28martin5519ok... well not CAT5E or CAT5 - so I'll get echo cancellation ... but do you thin kI can do all this with an Asterisk box?  I mean... I need the low tech guy in teh office to go to a web browsser or some type of managmenet interface and be able to enable/disable ports and manage features for users.
20:19.40dlynes_laptopmartin5519: whether the cables are blue or grey or white doesn't mean anything either...you can get cat3/cat5/cat5e in all colors
20:19.52martin5519I'm in Ottawa - buildings are in Toronto though.. I'll check it out next tiem I'm there
20:20.00monstedmartin5519: can't be worse than most PBX'es :)
20:20.12dlynes_laptopmartin5519: easily
20:20.24dlynes_laptopmartin5519: the only question is how many problems you'll run into with call quality
20:20.28martin5519Really eh! that's awesome - are you an expert on ASTERISK?
20:20.48dlynes_laptopmartin5519: I wouldn't say I'm an expert, but I have been using it for a couple of years now
20:21.03martin5519Ah... call quality - that's were I come in - I am a networking expert ;) -... serously though... so no prob there... I've done VOIP before withouth issue
20:21.27dlynes_laptopmartin5519: I'm talking about between the analog phones and the asterisk box
20:21.33dlynes_laptopmartin5519: not on the voip end
20:21.36martin5519oh....
20:21.58martin5519Well... it's just analog though at that point...
20:22.02dlynes_laptopmartin5519: 99% of your call quality issues when dealing with analog devices is usually on the analog end
20:22.12martin5519yeesh...
20:22.17martin5519Good to know actually
20:22.36dlynes_laptopmartin5519: that's why hardware echo can is a must if you want to be guaranteed of less problems right from the get go
20:22.42martin5519Well, we have on ebuilding right now operating off a Quintum with 12 users .... had only 1 issue but I see what you're saying
20:22.56martin5519So, what are these hardware devices called?
20:23.03martin5519how do they fit on the line?
20:23.07dlynes_laptoptdm2400p/tdm400p/a200d
20:23.10martin5519Is it a board I put inside the server?
20:23.18dlynes_laptoperm tdm400p doesn't have hardware echo can
20:23.21martin5519.... Oh well yea... the tdm2400p is actually what I'm looking to get
20:23.33dlynes_laptopThe 2400p has a hardware echo can option
20:23.41martin5519Because that tdm2400p is the proper physical interface too... we have a winner!
20:23.45dlynes_laptopYou can buy it with or without hardware echo can
20:23.50martin5519I'
20:23.54martin5519d buy with obvously
20:23.58martin5519We want a rock solid service
20:24.25*** join/#asterisk xnon (i=xnon@200.8.30.3)
20:24.49martin5519See th eproblem we ran into with Quintum is it deosn't have a good interface to enable/disable ports by non-tech admin. and it deosn't do call forward and something else... and the users are not able to program features via their phones
20:24.54Corydon-wIf you wanted rock solid, you wouldn't be using analog
20:25.01*** join/#asterisk zotz (n=zotz@208.196.247.175)
20:25.01Corydon-wYou'd be using PRI
20:25.26martin5519Corydon - Yeah I know... but that' $500/month
20:25.41dlynes_laptopCorydon-w: if he wanted rock solid, he wouldn't be using a voip line, either
20:25.57martin5519From an engineering standpoint, you can pull all the best stuff out to be stable... but there's a cost issue there..
20:26.00Corydon-wmartin5519: so is 12 lines of analog
20:26.06martin5519yes
20:26.41Corydon-wYet a PRI gives you nearly double the channels
20:26.45martin5519yep - our current building is 12 analog lines being delivered on an amphonol interface into the Quintum which passes the calls over a DSL line (no PPP header and no data) to a VOIP Peer
20:27.35Corydon-wWould you like it good or cheap?  ;-)
20:27.39martin5519Corydon - w:  Sorry, the lines come in as wire pairs form the appartments.... are you suggesting we put those wire pairs on a digital interface?
20:28.00martin5519Then, we'd have to get Digital phones for all the units???
20:28.07FastFeetAny alternatives to the TDM400P? Honestly, it is a little over my budget. looking to spend no more than $200 if I can help it.
20:28.08aydiosmiogood, cheap, fast / good, cheap, legal
20:28.34dlynes_laptopFastFeet: are all analog phones going to be wired in parallel?  Or are they going to be three separate stations?
20:28.34Corydon-wFastFeet: nope, not really
20:28.47FastFeetparallel
20:29.00dlynes_laptopFastFeet: go with the sipura 3000 or the grandstream ata-186 then
20:29.20FastFeetI was looking at the sipura 3000, but I see Linksys bought them
20:29.21dlynes_laptopFastFeet: erm...forget the grandsteam ata-186...it's not a gateway
20:29.26FastFeetare their devices the same?
20:29.28dlynes_laptopFastFeet: Sipura still exists
20:29.35dlynes_laptopFastFeet: it's just a division of Linksys now
20:30.10dlynes_laptopFastFeet: They're gradually rebranding all sipura devices as Linksys, but not all of them have been rebranded yet
20:30.13martin5519Corydon... if we delivered the appartements on a PRI interface, wouldn't each unit need a digital phone?
20:30.21dlynes_laptopFastFeet: I think sipura 3000 is one of the ones that hasn't been rebranded yet
20:30.26dlynes_laptopmartin5519: no
20:30.34dlynes_laptopmartin5519: you can give them all analog phones
20:30.43FastFeetOK, any recommendations one where to purchase one?
20:30.54dlynes_laptopFastFeet: What country are you in?
20:31.15FastFeetCanada, but would be willing to look within the US as well.
20:31.23dlynes_laptopFastFeet: Are you in the east, or the west?
20:31.37FastFeetEast
20:31.39Corydon-wmartin5519: generally for that kind of setup, I'd do a PRI to the apartments and a channel bank out to each individual
20:31.45dlynes_laptopFastFeet: Try voxilla.ca
20:31.53dlynes_laptopFastFeet: or voipdepot.ca
20:32.17martin5519Corydon-w:  A channel bank - what's that?
20:32.31FastFeetThanks,  a big help!
20:32.34Corydon-wmartin5519: splits T1 CAS into individual analog lines
20:32.35dlynes_laptopFastFeet: Or, we might even have a used one kicking around...I'd have to see how much the boss would let it go for, though
20:32.40dlynes_laptopFastFeet: We're in Vancouver
20:33.08FastFeetcool....
20:33.18FastFeetJust looking at the linksys once again....
20:33.18martin5519Corydon-w:  You got my curiosity... but if you break that into anaolog - aren't you back to square 1?
20:33.22FastFeethttp://www.linksys.com/servlet/Satellite?c=L_Product_C2&childpagename=US%2FLayout&cid=1149639756784&pagename=Linksys%2FCommon%2FVisitorWrapper
20:33.32FastFeetThis is the same thing as the Sipura 3000?
20:33.41martin5519By the way, what deos CAS stand for?
20:33.42Corydon-wmartin5519: nope, you have a superior interface to the phone company
20:33.52Corydon-wChannel Associated Signalling
20:34.13FastFeetjust renamed??
20:34.13dlynes_laptopFastFeet: nope...that one's a sipura 3000 built into a router
20:34.22dlynes_laptopFastFeet: the sipura 3000 doesn't have a router functionality
20:34.23FastFeetConfussed...
20:34.28FastFeetOK
20:34.32Corydon-wmartin5519: analog signalling to the phone company is bad when you figure that they charge if you don't hangup properly
20:34.34dlynes_laptopFastFeet: that one's probably more expensive
20:34.45martin5519Corydon-w: Ok... I think you may have missed something... we are not sending the calls to a phone company, we are sending the calls via IP to a VOIP Peers over Ethernet.  We don't deliver the calls to any LE or CO
20:34.50FastFeetI C..  Your probley right....
20:35.03martin5519Here's a sorta map
20:35.08FastFeetI will check those links out.....
20:35.37_rnz-heh
20:35.40_rnz-i just socially engineered
20:35.41martin5519appartements -------- cat3 ------- amphenol (VOIP BOX )Ethernet ------------- ATM CLOUD ------------ VOIP Peer
20:35.42_rnz-#freebsd
20:35.47_rnz-into a religous argument
20:36.22dlynes_laptopFastFeet: http://www.linksys.com/servlet/Satellite?c=L_Product_C2&childpagename=US%2FLayout&cid=1125638798035&pagename=Linksys%2FCommon%2FVisitorWrapper
20:36.50FastFeetThat would be really sweet
20:37.10FastFeetif you had one kicking around.
20:37.11dlynes_laptopFastFeet: that's the rebranded sipura 3000; the model number is still spa 3000
20:37.20*** join/#asterisk CunningPike_ (n=CunningP@204.239.12.189)
20:37.22FastFeetok....
20:37.30dlynes_laptopFastFeet: we've got the old sipura branded device
20:37.59dlynes_laptopFastFeet: basically the only difference is the linksys looks nicer, and it's in a cheaper feeling enclosure
20:38.13FastFeetok, I wasn't sure...
20:41.01FastFeet<---| Newbie...
20:41.01_rnz-dlynes
20:41.01_rnz-that reminds me
20:41.01FastFeetI am looking to set up and show off Asterisk for a college senior project.
20:41.01_rnz-i have a wrt54g at home
20:41.02_rnz-i saw linksys
20:41.02FastFeetthanks for your help.
20:41.02_rnz-arent wrt54g's intensely configurable
20:41.02_rnz-, ie: to hack around
20:41.02martin5519Corydon?
20:41.02FastFeetyup, I have linux running on mine.
20:41.02Corydon-wmartin5519: what?
20:41.03martin5519Did you see what I meant?
20:41.03Corydon-wYes
20:41.03martin5519so - what do you think?
20:41.03martin5519We have this working fine right now - jsut poor features on VOIP box we're using
20:41.03*** join/#asterisk moon06 (n=michael@cim06-1-82-228-240-97.fbx.proxad.net)
20:41.03Corydon-wI'd probably still go with a channel bank
20:41.03martin5519which side the the VOIP box in the diagram - you mean facing the appartments?
20:41.03Corydon-wYep
20:41.14Corydon-wchannel bank + t1 interface
20:41.32martin5519But does that mean each appartment gets digital and that we need a digital phone in each unit?
20:41.36martin5519or do you mean something like:
20:41.39Corydon-wwhat features are you looking for?
20:41.47Corydon-wNo, each apt is still analog
20:41.52dlynes_laptop_rnz-: only certain hardware versions
20:42.02martin5519appartments----------- cat3-------- channelbank-----T1(VOIPBOX)..........
20:42.12Corydon-wYep
20:42.13dlynes_laptop_rnz-: certain hardware versions allow you to replace the OS on them
20:42.35martin5519All features typical to voice ... call forward... call waiting... users who can activate/deactivate features from their phone etc...
20:42.41martin5519give me 2 seconds... client in here
20:42.43FastFeetThat Grandstream ATA-186 device.... have they been bought out by Cisco as well?
20:43.02Corydon-wGenerally T1 interface + channel bank is cheaper than a TDM2430
20:43.08dlynes_laptop_rnz-: there's a mailing list dedicated to throwing asterisk onto those wrt54g's and the Linksys network storage devices
20:43.43dlynes_laptopFastFeet: If you'll scroll up, you'll see where I said forget about the ata-186, because it's just an ata, not a gateway...and afaik, it hasn't been bought by cisco...but I could be wrong
20:44.12FastFeetYour right, I missed that comment.
20:44.15FastFeetSorry.
20:44.28FastFeetSipura it is...
20:45.02*** join/#asterisk bungalow (n=tasat-@c-24-7-62-81.hsd1.ca.comcast.net)
20:45.11martin5519back...
20:45.48martin5519So Corydon... oh right.. I see... makes sense.. so what type of channel bank could I buy?
20:46.08bungalowhi -- I'm getting chan_sip warning: Maximum retries exceeded on transmision ... for seqno ### (Critical Response) -- can someone explain what's happening and suggest a way to fix the problem? Is this me, my sip provider, or both?
20:46.28martin5519And the Channel bank in that case would have the AMphenol interface...
20:46.35Corydon-wmartin5519: correct
20:46.52*** join/#asterisk Skarmeth (n=Skarmeth@201009095121.user.veloxzone.com.br)
20:47.14martin5519Would it then just be easier for me to buy a Cisco Router with a PRI interface that can be configured for Voice?
20:47.26martin5519... considering I have a lot of experience with Cisco products
20:48.39Corydon-wI have no idea what Cisco would do this
20:48.56martin5519Right... ok..
20:49.17martin5519So Corydon-w: have you done this with a Channelbank and PRI on a box?
20:49.20Corydon-wCisco is generally far more expensive than anything else
20:49.27Corydon-wForget PRI
20:49.28martin5519Agreed...
20:49.34Corydon-wPRI is for the connection to the telco
20:49.50martin5519Well, I could have done it facing the appartments as well
20:50.29*** join/#asterisk letoto (n=paul@tla.xelerance.com)
20:50.35martin5519how much deos a channel bank cost and how much does a PRI card from Digium cost?
20:50.47Corydon-wYou can usually get the CAC1 channel bank for just under $400
20:50.57letotoeasy question. Does Asterisk work inside a Xen virtual machines, including timings for conference calls?
20:51.19Corydon-wletoto: probably not
20:51.42letotoso i need to give the xenU the USB bus. That's possible
20:51.51Corydon-wletoto: I wouldn't even trust it within VMware
20:52.01letotoI trust xen more then vmware
20:52.19*** join/#asterisk dahunter3 (n=dahunter@168.sub-75-214-69.myvzw.com)
20:52.19martin5519AND PRI for about the same price I imagine yet the card I need is $900 - kind of works out to the same
20:52.19letotobut what are timers used for conference? Is it still usb timings? or did it use other kernel internals?
20:52.54Corydon-wletoto: no, it uses RTC
20:53.14Corydon-wIt used USB only under 2.4
20:53.19*** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com)
20:53.32letotoah. so there is no point gating the usb bus to the xenu
20:53.44Corydon-wletoto: not unless you're running 2.4
20:53.56CunningPike_bungalow: Some critical call signaling packets are going missing - likely due to latency
20:54.12*** part/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com)
20:54.20martin5519Anyone on the board know who's the biggest ASTERISK expert?
20:54.36Corydon-wThis isn't a board; it's a channel
20:54.42martin5519channel..
20:54.54monstedhmm, these channel banks would take a PRI "FXS" port and turn it into 24/30 POTS ports?
20:55.03monstednot an item i've come in contact with before :)
20:55.29Corydon-wmonsted: You're combining different technologies
20:55.45martin5519I need an Asterisk expert
20:55.53Corydon-wmartin5519: see voip-info.org and find a local consultant
20:56.54bungalowhi -- I'm getting chan_sip warning: Maximum retries exceeded on transmision ... for seqno ### (Critical Response) -- can anyone suggest an approach to debug this?
20:56.57monstedCorydon-w: ok, i must admit that ISDN in general confuses me :)
20:57.14martin5519oh no... I don't want to go through a company... I'd rather find someone local I can pay via PAYPAL once I know this is all doable with Asterisk
20:57.39Corydon-wmartin5519: it's all doable with Asterisk
20:57.39monstedi'd rather pay a company that could actually service the crap when it breaks ;)
20:57.42letotomartin5519: companies use paypal too :P We accept paypal for our Openswan contracts
20:58.16martin5519letoto:  Right, but I just want an individual Asterisk expert I can work with
20:58.43*** join/#asterisk dasenjo (n=dasenjo@201.228.128.10)
20:58.44monstedmartin5519: why not someone who can cover you even if one person is sick or on vacation?
20:59.03martin5519ok
20:59.15martin5519maybe a company would be a good thing
21:00.34*** join/#asterisk jjasper (n=jjasper@h-66-112-162-129.connactivity.com)
21:01.17CunningPike_martin5519: Do you have a specific problem, or are you just looking for support in general?
21:01.40*** join/#asterisk [Airwolf] (n=airwolf@attilla.nl)
21:02.13brif8Hi All,  Do we ask here about 1.4 GUI or  asterisk-gui  I have installed 1.4 to test and can't configure Service Providers  or Calling Rules.  I have got Users configured but that's about it ??
21:02.15martin5519CunningPike:  I have a requirement that I know I'm going to need help with getting rolling.   I
21:02.26aydiosmiocraigslist!
21:03.11CunningPike_brif8: Try #asterisk-gui
21:03.14martin5519CunningPike:  Essntially, I need someone who knows Asterisk so well that they can assurem it can deliver what we need.  Then, we order th part, bring up the server, give the consultant a an account via SSH and log in and help us out.  I'm very techincal but it will take me 10 times the amoutn of time figuring things out
21:03.49*** join/#asterisk hoobastooba (n=ckwall@63.149.122.93)
21:04.04Strom_Cmartin5519, I do consulting, and let me tell you - having someone on-site is EXTREMELY useful
21:05.05martin5519Strom_C:   For us, we have our setup as if we were on site:  Remote Reboot box (APS Masterswitch) and Console Access via backup dial-up
21:05.19Strom_Cand what about the phones?
21:05.20martin5519We can't be everywhere... :)... our buildings are in Toronto - w'ere in Ottawa
21:05.28martin5519The phones?
21:06.14CunningPike_martin5519: I'm in BC - drop me an email and I'll see if I can help you
21:06.17hoobastoobai see that in "/var/spool/asterisk/voicemail/context/extension" there are four versions of the recording... gsm,wav,WAV and text. I have set in the [general] context format=wav and in the context i am using for those extensions format=wav. is that all i have to change? or is there something else... I only want the .wav files in voicemail
21:08.30CunningPike_hoobastooba: That should be it - if there are multiple versions of a file present, I think asterisk will pick the 'least cost' one - the .conf settings only affect the recording of new files. You will need to delete the others if you want the wav ones to play
21:09.21hoobastoobaand once the recordings are listened to are they supposed to be moved from the INBOX to a new location?
21:10.59CunningPike_hoobastooba: Yes - they should end up in the 'Old' folder
21:11.24hoobastoobaok, thanks
21:11.36*** join/#asterisk adorah (n=admin@87.68.144.118.cable.012.net.il)
21:11.37*** join/#asterisk dahunter3 (n=dahunter@168.sub-75-214-69.myvzw.com)
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21:14.44*** join/#asterisk CunningPike_ (n=CunningP@dhcp-10-153.district.north-van.bc.ca)
21:15.45letotomartin5519: Xelerance has a presence in Ottawa and Toronto. www.xelerance.com
21:20.44*** join/#asterisk Jamez^7 (n=martini@modemcable131.214-131-66.mc.videotron.ca)
21:26.53*** join/#asterisk jm|work (n=jamiem@zen.jamiem.com)
21:27.32martin5519CunningPike - did you get my e-mail?
21:27.47CunningPikeYes - I did, thanks
21:29.19*** join/#asterisk IOscanner (n=IOscanne@cpe-76-187-205-211.tx.res.rr.com)
21:33.35CunningPikemartin5519: Check your pm
21:37.46arcaninewhat is digium's equivalent for rhino r4fxo card
21:38.04*** join/#asterisk icel (n=dan@63.78.162.77)
21:38.29*** join/#asterisk DaveCanoe (n=Dave@H6.C30.B96.tor.eicat.ca)
21:38.35Strom_Carcanine, four FXO ports?
21:38.56arcaninetdm04b
21:38.58*** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-88-153-235-141.red.bezeqint.net)
21:39.06Strom_Cyes
21:39.12arcaninek thanks
21:39.48*** join/#asterisk icel (n=icel@63.78.162.99)
21:41.13*** join/#asterisk qdk (n=qdk@0x535eae17.boanxx9.adsl-dhcp.tele.dk)
21:41.36hoobastoobais it possible to make it so that when voicemail creates a directory for an extension, it creates it with different read write permissions that what is currently done by default? I need each directory to be 744 and i dont want to have to go back and modify it each time one is created.
21:41.39hoobastoobapossible?
21:42.28CunningPikehoobastooba: umask?
21:42.36*** join/#asterisk Waverly360 (n=mirc@209.12.249.243)
21:42.58*** join/#asterisk db1310 (n=danny@216.77.58.40)
21:43.09hoobastoobaCunningPike: what is umask?
21:43.19CunningPikehoobastooba: Scratch that - won't work anyway :)
21:43.30CunningPikehoobastooba: What are you trying to accomplish?
21:44.03CunningPikemartin5519: Did you get my pm?
21:44.05db1310can someone please tell me where the xilink files for the zapata open Tormenta 2 card is?
21:44.17hoobastoobai need to be able to send files from another server/user to the "/var/spool/asterisk/<context>/<extension>" directory for an application I am writing.
21:44.41*** join/#asterisk FuriousGeorge (n=brian@ool-43536ea8.dyn.optonline.net)
21:50.05*** join/#asterisk bblack (n=bblack@129.174.111.120)
21:50.36db1310anyone know where i might get support for the zapata hardware project?
21:51.11CunningPikehoobastooba: Hmm - well, we actually store our voicemail on an NFS mount, so we use the anonuid argument in /etc/exports to make sure that the remote server uses a particular uid to create the files
21:51.34justinu|laptopdb1310: good luck
21:51.45Waverly360I need NTP help.  My NTP..it no worky..and I'd like to know why.
21:52.02CunningPikehoobastooba: So, your other server could NFS to your asterisk server using the same uid that owns the existing files
21:52.04Waverly360What's wrong you ask?
21:52.07*** join/#asterisk emp (n=emp@host-69-144-157-39.bln-mt.client.bresnan.net)
21:52.18CunningPikeWaverly360: What's wrong?
21:52.20CunningPike:)
21:52.22Waverly360:)
21:52.25aydiosmioNo, I ask "Why are you askign here?"
21:52.26pigpenCan anyone tell me if Asterisk can kill itself, then relaunch on it's own?
21:52.40db1310justinu|laptop: so, former developers to busy making money on the project?
21:52.53Waverly360Phones aren't getting time...they're just flashing.  They're polycoms...I've had port 123 opened from the asterisk box to the outside world for udp
21:52.56ucfMethodAnyone know what format wav file, if any the blackberry's support? The emails w/ wav attachments that are being sent from Asterisk are not playable by the phones.
21:52.57CunningPikepigpen: No - it can relaunch itself if something else kills it, but it doesn't kill itself ;)
21:53.16aydiosmioyou can send signal HUP
21:53.19empwhat's a good distro to run asterisk? for the lazy :)  trixbox? something else?
21:53.21justinu|laptopdb1310: that's a good question... i haven't heard about tormenta cards in a while
21:53.34CunningPikeemp: The best distro is the one you know the best
21:53.35Waverly360Unfortunately for me..I'm not really sure how the ntp stuff works.  Does asterisk talk to the ntp server, and then serve time to the phones? or do the phones talk to the ntp server themselves?
21:53.36aydiosmiothat usually will do a clean reload on a process
21:53.51CunningPikeWaverly360: The phones speak directly to an NTP server
21:53.55aydiosmioNTP is sepearate
21:54.01pigpenhmm...I had a box kill itself off, then bring it back up.  Originally, it was running as a process "/usr/sbin/asterisk -U asterisk"  now it is "asterisk" and my new MOH music is playing....
21:54.10Waverly360Ok..how do I tell the phones to talk to the ntp server I have running on my pbx?
21:54.27pigpenie: I added several new music files, but I had not reloaded yet....old music was left intact.
21:54.46CunningPikeWaverly360: In your sip.cfg, you need to set the IP address of the NTP server - hang on a sec.....
21:54.52hoobastooba<PROTECTED>
21:54.55Waverly360CunningPike: kk
21:55.21pigpenSorry to sound like a dumbass...but ....  "safe_asterisk" ?
21:55.40Waverly360pigpen: it's a process that watches asterisk.  If asterisk dies..it restarts it.
21:55.46pigpenhmm....
21:55.53CunningPikeWaverly360: <SNTP tcpIpApp.sntp.resyncPeriod="86400" tcpIpApp.sntp.address="132.246.168.164" tcpIpApp.sntp.gmtOffset="-28800" etc etc etc
21:55.57pigpenworks for me...that would at least explain it.
21:56.09*** join/#asterisk tsurk0 (n=tsurko@vpn-pppoe-213-240-242-237.megalan.bg)
21:56.27pigpenIt was down for about 5 min...but of course 250 phones will take a few min to re-register...
21:56.27Waverly360CunningPike: Ok...so even better question...if tcpIpApp.sntp.address="" then where would my phones get their time from?
21:56.41CunningPikeWaverly360: Nowhere :D
21:56.49Waverly360then how...do my phones have time?
21:57.10CunningPikeWaverly360: Does your DHCP server serve an NTP option?
21:57.25Waverly360CunningPike: *gasp*  I think so....forgot about that.
21:57.37Waverly360CunningPike: hold please :)
21:57.40CunningPikeWaverly360: ;)
21:57.41hoobastoobapigpen: check the wiki on safe_asterisk
21:57.50pigpenk
21:57.50hoobastoobaeasier than explaining.
21:58.04Waverly360CunningPike: It does!  oooh...ok..it's all coming together..either that or I'm blacking out again..we'll see
21:58.11CunningPikeWaverly360: heh heh
21:59.17*** join/#asterisk CharlesR (n=charlesr@cpe-76-188-71-88.neo.res.rr.com)
21:59.59db1310anyone know who to ask to get the full set of design files for the pci zapata card, its open license but all of the files are not available. Doesn't that violate the license?
22:00.52_rnz-(BREAKING NEWS)(AP/REUTERS) - Iran has launched 4 nuclear missiles into Israel causing unprecedented, and widescale devastation in Tel Aviv.  Details Soon.
22:01.52*** join/#asterisk CunningPike_ (n=CunningP@204.239.12.189)
22:02.46erubrightexit
22:04.59*** join/#asterisk linlin (i=will@c-71-194-70-13.hsd1.il.comcast.net)
22:07.09*** join/#asterisk ToyMan (n=stuq@user-12lcqia.cable.mindspring.com)
22:11.22Dovid_rnz-: where did u see that report ?
22:11.36pigpenyeah...I can't find anything.
22:11.59Dovidsome one must be bored
22:12.20*** join/#asterisk Tond (n=tond@CPE0014bf30c190-CM0010954a5ffa.cpe.net.cable.rogers.com)
22:12.28Dovidi am in israel now - if it did happen i would know about it
22:13.11pigpenlets hope it didn't happen...
22:13.16justinu|laptopthat's a joke
22:13.20justinu|laptopa poor idea of a joke, at least
22:13.46Tondhi in my extentions i have a _98. extention as well as _9821. extentions pattern, but when i dial 982144 it matches _98. pattern, whyc does it do that?  Besides _9821. pattern is created before _98. !
22:14.16Dovidwhere is a mod when u need some one to bounce an ass ? i freaked out for a sec. but on the other hand i am 30 min from TLV. had it happend i would of heard it
22:14.41bblackTond: http://www.voip-info.org/wiki/index.php?page=Asterisk+config+extensions.conf+sorting
22:14.46puzzledTond: if you do "show dialplan" from the cli then you can see how asterisk interprets the order
22:15.06TondThanks guys...
22:15.07Tond:)
22:16.09Dovidtest
22:16.19*** join/#asterisk jm|work (n=jamiem@zen.jamiem.com)
22:17.33_rnz-Dovid
22:17.36_rnz-it was a fake headline
22:17.40_rnz-I take it by your name: Dovid
22:17.42_rnz-you are israeli :)
22:17.45DovidNOPE
22:17.47Dovidamerican
22:17.58_rnz-why o, and not a?
22:18.00Dovidbut that was real sweet for scrain the shit outa sme one
22:18.02_rnz-Dovid sounds cripled, heh
22:18.04_rnz-lol
22:18.08_rnz-you get scared? :
22:18.08Dovidits my legal name
22:18.19_rnz-Dahvid = Dovid
22:18.23_rnz-David = Day-vid
22:18.28Dovidfor a moment then realized had there been one i woulda hear it from here
22:18.34DovidDovid = Duh-Vid
22:19.40icelwhat is the best way to hook up multiple (>100) digital phones?  Surely you don't have to have a fxs module for each one?
22:20.39monstedicel: IP phones? a switch port :)
22:20.42CunningPike_icel: What do you mean by 'digital' phones?
22:20.49icelnot a soft phone
22:21.04monstedthere are physical IP phones too
22:21.24icelthat might be a good option then
22:21.42*** part/#asterisk hoobastooba (n=ckwall@63.149.122.93)
22:21.49monstedlooks like a regular phone but has a standard ethernet port (or two) in the back
22:21.56Dovidthen u use a switch
22:22.11Dovidthats when asterisk gets cheap
22:22.16monstedwith some of them, you can even use the switch port peoples PC is in and plug the PC into the phone
22:22.17Dovidu dont have to pay for ports
22:22.18icelCunningPike: digital as in receives power from phone switch
22:22.35JThuh
22:22.42monstedicel: PoE switch ports and a compatible phone
22:22.44JTthat's got nothing to do with the definition of digital
22:22.57CunningPike_icel: So, IP phones?
22:23.02icelSo if I have a voice T1, a digium card to use it, then my cheapest option is to use IP phones and throw them onto a switch
22:23.11JTboth analogue phones, digital key system phones, and ip phones receive power from the switch
22:23.12monstedicel: correct
22:23.24JTa PoE capable switch
22:23.26icelthis is like school, it is good for me
22:23.26Dovidicel: be carefull with what phones u get. in general cheap phones suck (IMHO)
22:23.42_rnz-NEW YORK - In an account his publisher considers a confession and some media executives call revolting, O.J. Simpson plans a book and TV interview to discuss how, hypothetically, he could have killed his ex-wife and her friend.
22:23.44JTit's probably not the *cheapest* solution, but one of the best solutions
22:23.46_rnz-http://www.msnbc.msn.com/id/15723351/
22:24.05icelEven with good phones it should be cheaper than having to buy billions of modules and hardware though I would think
22:24.12Dovidyup
22:24.17Dovidthats y we luv asteris
22:24.19Dovidasterisk*
22:24.19_rnz-One expert noted that the justice system’s protection against double jeopardy means Simpson’s book, explosive as it may be, should not expose him to any new legal danger.
22:24.40monstedicel: it's certainly a lot easier
22:24.41JTicel: it wouldn't be cheaper than getting cheap analogue phones, T1/E1 card and channel bank
22:24.58icelJT: ok
22:25.04JT_rnz-: do you really need to paste all that?
22:25.17icelis he still talking?  I did an ignore a while ago
22:25.22JT_rnz-: it's offtopic, and that last paragraph is obvious to most people who know anything about the law
22:25.49icelIs there any good documentation about ways to design the whole bloody phone system?
22:26.01JT~thebook
22:26.02jbotfrom memory, thebook is a book called  Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11
22:26.07icelI find lots of good documentation, but it seems to lack in architecture
22:26.08icelk
22:26.14JTit won't tell you how to design it really
22:26.19JTthat's still up to you
22:26.24JTwhat sort of architecture?
22:26.45icelI just want to know good and feasible ways that I can connect hardware to software in a 100+ phone environment
22:27.02justinu|laptopicel that comes from experience, and i have yet to see a single good telephony architecture handbook
22:27.05Dovidicel: read the book
22:27.08*** part/#asterisk letoto (n=paul@tla.xelerance.com)
22:27.14justinu|laptoptry newtons telecom dictionary for a bit of help maybe
22:27.15JTyou'd really want to go VoIP phones unless there was an overriding reason not to
22:27.39bblackicel: a good read: http://www.amazon.com/Carrier-Grade-Voice-Over-IP/dp/0071406344/sr=8-10/qid=1163629613/ref=pd_bbs_sr_10/104-6750231-8639109?ie=UTF8&s=books
22:27.43_rnz-icel, a good solution to that is
22:27.47_rnz-Call Manager Express
22:27.52icelI figured that would be good so I messed around with SIP a lot and got softphones working how I want, but now its time to try hardware
22:28.11monstedwith 100 phones, i'd go for a full Call Manager (or asterisk)
22:28.18icelthanks so far on the uRLS and ideas, keep em comin!
22:28.34_rnz-monsted, im using the ccvp call manager express simulator
22:28.39_rnz-and it has the scenarios which he mentions
22:28.44Waverly360CunningPike: Oh..by the way..I believe you solved my problems.  Mucho appreciation.
22:29.11icelwhaddya mean a full call manager or asterisk
22:29.13CunningPike_Waverly360: Great!
22:29.20Waverly360CunningPike: Oh..and btw...remember WAAAAY back when I was asking all kinds of questions about presence with the Polycoms in asterisk?
22:29.21JTcall manager would take the problems out of your hands
22:29.23JTbut cost++
22:29.33JTicel: they're spruiking cisco call manager
22:29.34iceldo you mean something different from the asterisk software pbx i am using?
22:29.53justinu|laptopyeah, they're talking about cisco's integreated voip pbx
22:29.55Waverly360CunningPike: Well..turns out the Polycoms will send out the information properly...but only after some weird freaking subscribe is sent to it
22:30.08monstedicel: Cisco Call Manager is a similar bit of software, but a bit more "slick", CCM Express is a smaller version on Cisco routers
22:30.13*** join/#asterisk AsteriskMonkey (n=admin@wireless-net143-ip170-toronto.ica.net)
22:30.24Waverly360CunningPike: Live Communication Server is currently the only thing that supports it, and the way polycom is doing presence is different than any of the other phones out there.
22:30.24_rnz-monsted, express is for a small business primarily
22:30.27_rnz-that has a lower budget
22:30.33monsted_rnz-: yep
22:30.52AsteriskMonkeyhas any one experinece a problem with voicemail where is continually fills up with 5 second blank voicemails from the same caller id?
22:30.54CunningPike_Waverly360: Are we still talking about NTP?
22:31.02_rnz-what is digiums relevance to asterisk again?
22:31.05_rnz-their HQ is in town here
22:31.06Waverly360CunningPike: no...presence from a long time ago
22:31.08_rnz-in huntsville, AL
22:31.08monsted_rnz-: they made it :)
22:31.18Waverly360CunningPike: in reference to the MyStat softkey on the phones
22:31.19CunningPike_Waverly360: Oh - sorry :)
22:31.29_rnz-when you say they? the one guy, or a team of engineers?
22:31.34_rnz-if its open source now, what does the company do?
22:31.40JTmakes hardware
22:31.41JTseriously
22:31.43_rnz-fabricate hardware asterisk integration solutions?
22:31.44JTwww.digium.com
22:31.45justinu|laptopthey made parts of it, other parts were contributed by ppl not connected with digium at all
22:31.45CunningPike_Waverly360: Did you find out what the 'weird freaking subscribe' was?
22:32.27_rnz-where can I download it? id be curious to lok at it at home
22:32.34_rnz-and compare it to ccm
22:32.36Waverly360CunningPike: My friend has a copy of it.  He's currently working to implement it in Freeswitch..but he's having a helluva time.
22:32.37JTwww.asterisk.org
22:32.48JTnext url you need is www.google.com
22:32.51justinu|laptopWaverly360: who's your friend?
22:32.52justinu|laptoppdt?
22:32.59_rnz-in the 3 of 5 tests ive passed in the CCVP test(s) so far
22:33.03_rnz-i havent heard asterisk mentioned once
22:33.09Waverly360justinu|laptop: yep
22:33.11justinu|laptopWaverly360: great guy
22:33.29Waverly360justinu|laptop: Yep...killer smart too
22:33.29monsted_rnz-: well, are you really surprised that Cisco doesn't talk about Asterisk in their tests?
22:33.31*** join/#asterisk h3x0r4t0r (n=hex@ip68-224-236-92.lv.lv.cox.net) [NETSPLIT VICTIM]
22:33.50JTcisco only supports cisco for their cisco phones
22:34.05bblackWaverly360: did you also play with agent login on Polycoms?
22:34.05monstedCisco only supports Cisco for anything :)
22:34.06JTif you only listen to stuff cisco says, you'll get a very narrow world view :]
22:34.14justinu|laptopheh
22:34.16Kattywhich conf file is responsible for the master.csv?
22:34.27JTmonsted: oh, they're at least forced to make their gear talk tcp/ip properly for other stuff
22:34.29justinu|laptopcdr-csv.conf?
22:34.31justinu|laptopsomething like that
22:34.34_rnz-cisco supports only cisco for the cisco file, because cisco ciscos the cisco chip on the cisco phone, that only works on the cisco software?
22:34.36Waverly360bblack: I used to, until we realized it was locking up our PRI
22:35.08JTtheir sip phones don't talk the sip standard very well though
22:35.09monstedthere are three realities in the cisco tests - there's the way it's supposed to work, there's the way cisco works and there's the way the real world works ;)
22:35.21_rnz-cisco = $$$
22:35.24_rnz-open source = potential $$$
22:35.31_rnz-salary talk
22:35.38Kattyjustinu|laptop: do you think that the postgres conf file will play nicely with mysql?
22:35.43_rnz-havent heard one major defense contractor in town
22:35.46_rnz-that uses asterisk here
22:35.46*** join/#asterisk tsurk0 (n=tsurko@vpn-pppoe-213-240-242-237.megalan.bg)
22:35.46bblackWaverly360: I saw the work done by bweschke in 6119 but it's out of date for 1.4 and didn't really have time to see what's going on and update it
22:36.06monstedNobody ever got fired for buy IBM^H^H^HCisco :)
22:36.08justinu|laptopKatty: play nice how? you want to log CDR to a mysql database, and a postgres database?
22:36.09_rnz-Im thinking asterisk is used by individuals or companies
22:36.14_rnz-who want to be "bleeding edge"
22:36.14monstedbuying, even
22:36.15_rnz-or "alternative"
22:36.22CunningPikeWaverly360: Does your friend happen to have the signaling that Polycoms need to display the ID of the called party?
22:36.24Kattyjustinu|laptop: to a mysql database....we don't have postgres installed on any machines.
22:36.35JT_rnz-: i'm thinking you like to talk a lot
22:36.36monsted_rnz-: it's used by thousands of companies who want a good, cheap soft-pbx
22:36.37Kattyjustinu|laptop: and another guy's going to write a lil website app to query it, etc
22:36.41JTand have little knowledge
22:36.47justinu|laptopKatty: for mysql, i think you need a cdr logging module out of asterisk-addons??
22:37.04Waverly360CunningPike: It's possible.
22:37.05_rnz-monsted, the 8 companies ive setup the ccm here, none of em even think twice
22:37.09_rnz-about having a free solution
22:37.15_rnz-they want a product that has a good foundation/history :)
22:37.29Waverly360CunningPike: Sadly I don't know near as much about it as he does.
22:37.35monsted_rnz-: i know - i make CCM solutions for a living too
22:37.37JT_rnz-: get the pole out of your arse
22:37.41CunningPikeWaverly360: Does he some here?
22:37.42_rnz-It's like telling a windows user: Use linux, because its free
22:37.43JT_rnz-: seriously
22:37.49JT_rnz-: did you come here just to troll?
22:37.51monsted(well, CCM and Nortel CS1000 stuff)
22:38.04Waverly360He used to, but he sticks to the freeswitch channels mostly these days.
22:38.05JTerr there are other reasons to use asterisk apart from price
22:38.10JTand you can pay for it if you want to
22:38.24CunningPikeIt's funny when you have /ignore set -you can see all the replies but none of the trolling :)
22:38.27Waverly360CunningPike: I can maybe convince him to login if you want.
22:38.28Kattyjustinu|laptop: umm umm, i found it.
22:38.36Kattyjustinu|laptop: any documentation on how to actually use it? (=
22:38.37CunningPikeWaverly360: That would be great........
22:39.03justinu|laptopKatty: i can only think to check the README files, and the sample config files
22:39.09CunningPikeWaverly360: What we are interested in is sending whatever message to the Polycom that it needs to display the CID of the called party
22:39.43justinu|laptopKatty: they might also tell you to set it up using ODBC, but that's more complicated, and if you ask me ODBC is evil
22:41.03Waverly360CunningPike: Hmm..y'know I think he wrestled with something similar awhile back...not sure he found an answer though :(
22:41.21Kattyjustinu|laptop: i found a lil doc on it, dankou!
22:41.23Waverly360CunningPike: Lemme forward that stuff onto him, and see if he won't log in and chat with you.
22:41.26CunningPikeWaverly360: OK - we're bugging our tech rep about it
22:41.30justinu|laptopno prob ;)
22:41.30CunningPikeWaverly360: Cool
22:41.57justinu|laptopCunningPike: i know how to do that.
22:42.08CunningPikejustinu|laptop: You do? Do tell........
22:42.14JT_rnz is afk - tossing
22:42.23justinu|laptopCunningPike: who's "we"? ;)
22:42.32JT_rnz-: it's "asterisk" btw, no s on the end
22:42.39monstedI would like to point out that not all cisco geeks are such insufferable twats ;)
22:42.49JTyeah
22:42.50CunningPikejustinu|laptop: We are the District of North Vancouver in BC, Canada? Why
22:43.01justinu|laptopjust curious...
22:43.07justinu|laptopi'll dig up the specs
22:43.08JTi've done the ccna
22:43.17JTnot an excuse not to use my own brain though :)
22:43.18*** join/#asterisk rajiv|work (n=rajiv@gentoo/developer/rajiv)
22:43.21CunningPikejustinu|laptop: Cool
22:43.32CunningPikejustinu|laptop: That would be awesome
22:43.37justinu|laptopbasically, it's done by sending the phone a particular Remote-Party-ID tag in the 180/183/200 OK message
22:43.44justinu|laptopof the party you connected to
22:43.47justinu|laptopor are ringing to
22:45.12bblackCunningPike: maybe you want to check this: http://bugs.digium.com/bug_view_page.php?bug_id=6643
22:45.50*** join/#asterisk jtf0518 (n=jaytee@c-69-137-243-25.hsd1.in.comcast.net)
22:45.51justinu|laptopthat sounds like the right patch, yeah
22:46.07_rnz-if everyone in here actually has any legitimate voip desires/interests
22:46.12_rnz-ccvp would be beneficial to anyone
22:46.19justinu|laptopi even got credit for help in that patch ;)
22:46.25_rnz-just because theres an anti cisco sentiment in here
22:46.27_rnz-is no reason to take it
22:46.30puzzledjustinu|laptop: does #6643 work on 1.2 also?
22:46.32_rnz-to not take it rather
22:46.40CunningPikebblack, justinu|laptop: Great - thanks!
22:46.53*** join/#asterisk tsurk0 (n=tsurko@vpn-pppoe-213-240-242-237.megalan.bg)
22:46.53justinu|laptoppuzzled: yeah, it was made for 1.2
22:47.15*** join/#asterisk shido6 (i=shido6@d221-68-200.commercial.cgocable.net)
22:47.18bblackpuzzled: patch made for 1.2.4
22:47.20jtf0518I have a couple questions about Asterisk. First one is how well does it scale?
22:47.31justinu|laptoplol, talk about a loaded question!
22:47.36_rnz-jtf, when compared to call manager,
22:47.36shido6to the sky
22:47.38JT_rnz-: why would you convince anyone to take the ccvp after your prior behaviour in here?
22:47.40_rnz-its not looking to good :)
22:47.53CunningPikejtf0518: Search the wiki for Asterisk-at-large
22:47.58bblackjtf0518: no offence, but you can write a book about it ;)
22:48.00JTjtf0518: ignore _rnz-, he's just a cisco troll
22:48.02jtf0518Thanks CunningPike
22:48.05puzzledjustinu|laptop: cool. which one is for 1.2? gork-calledparty.diff or gork-calledrpid-trunk.diff?
22:48.17jtf0518JT, I kinda figured that already
22:48.19_rnz-jtf, ignore JT, hes just bitter
22:48.23_rnz-:)
22:48.31JTwho won't even enter into any discussion with people
22:48.32_rnz-hes to lazy to go for a ccvp, or ccie in voice
22:48.32bblackpuzzled: first
22:48.35JThe just keeps trolling
22:48.41JT"too"
22:48.41puzzledbblack: thanks
22:48.42_rnz-he sticks to generic networking knowledge
22:48.49justinu|laptopman, gork didn't even get any karma points for that patch
22:49.00_rnz-if JT asked someone how RIP or ospf worked, he would say: IP routing works by routing a packet of kluged packets through a maze of mess
22:49.03_rnz-without even knowing theory
22:49.04_rnz-:)
22:49.04AsteriskMonkeyno zconfig.h anymore?
22:49.06justinu|laptopand he's still active on the issue, it seems
22:49.12_rnz-or if someone asked JT that rather
22:49.12JTi have more networking and telecommunications knowledge than your small little head could fathom, _rnz-
22:49.33_rnz-if someone asked JT how rip or ospf worked, he would say: IP routing works by routing a packet of kluged packets through a maze of mess
22:49.34*** join/#asterisk stkn (n=stkn@gentoo/developer/pdpc.active.stkn)
22:49.39_rnz-he wouldnt know BGP if it smacked him in the face
22:49.44JTthat makes no sense
22:49.49jtf0518I have a Nortel Meridian Option 11C and I'd like to connect an Asterisk server to it so I could route internal calls in our organization both ways with 4 digit dialing. I don't want to go with the Nortel solution and I've been looking at alternatives.
22:49.51AsteriskMonkeynor an AS number :P
22:49.52aydiosmioso
22:49.53JTworst pay out i've ever seen
22:49.57aydiosmioare you two gonna shut up?
22:50.00_rnz-aydiosmio
22:50.02_rnz-ask him
22:50.03_rnz-dont know :)
22:50.12monsted_rnz-: just because someone doesn't use Cisco and has no Cisco certs doesn't mean they know nothing about networking
22:50.17JTAsteriskMonkey: who was that in reference to?
22:50.17shido6thats on the wiki jtf0518
22:50.37jtf0518thanks shido, that would be www.asterisk.com?
22:50.42shido6no
22:50.46shido6www.voip-info.org
22:50.50PumasI have a problem when I try to record an Voip incoming call it stops at 5 minutes, I just can record 5 minutes of conversation
22:50.55jtf0518thank you, shido6
22:51.00Pumashelp me please
22:51.00CunningPikejtf0518: We did exactly that :)
22:51.15monstedi find the Nortel (CS1000-friendly) phones quite horrible
22:51.17jtf0518CunningPike, can I ask you a specific question?
22:51.24CunningPikejtf0518: Of course
22:51.44*** join/#asterisk hoobastooba (n=ckwall@63.149.122.93)
22:51.58hoobastoobawhat is the dial string for dialing more than one extension at the same time?
22:52.02CunningPikejustinu|laptop: Is SIPCalledRPID (http://bugs.digium.com/bug_view_page.php?bug_id=6643) in 1.4?
22:52.03shido6whoops, http://www.pham.org/asterisk/asterisk-meridian-a1.pdf is missing.
22:52.04jtf0518I've looked at the Digium products for Asterisk. How did you interface your Option 11C with Asterisk?
22:52.18CunningPikehoobastooba: Dial(SIP/ff&SIP/bar)
22:52.49shido6what kind of interfaces do u have jtf0518  ?
22:52.53justinu|laptopCunningPike: apparently not
22:52.59CunningPikejtf0518: We have a TE411P and run 2 PRIs to the Nortel - we needed 2 to get the CID to pass in both directions
22:53.03justinu|laptopone of those things that wasn't very important, I guess
22:53.18justinu|laptopit required a revised ast_indicate API to work "right"
22:53.23justinu|laptopbut that is in 1.4, I believe
22:53.44jtf0518which Nortel PRI cards are you using?
22:54.00CunningPikejtf0518: Now you're asking tough questions lol
22:54.01hoobastoobadid not understand that... if I want to dial sip/7000 and sip/7001 what would that look like? I thought it was Dial(SIP/7000),(SIP/7001) but its not working... looks like by your example i am way wrong
22:54.13CunningPikejtf0518: Hang on while I ask
22:54.21PumasI have a problem when I try to record an Voip incoming call it stops at 5 minutes, I just can record 5 minutes of conversation
22:54.23CunningPikehoobastooba: Dial(SIP/7000&SIP/7001)
22:54.30hoobastoobaah, ok,thanks
22:55.26CunningPikejustinu|laptop: I wonder is there a SIP-Header that would do the same thing.....
22:56.32justinu|laptopthere's another way to do it, with P-Asserted-Identity
22:56.36justinu|laptopi think that's the RFC compliant way
22:56.45justinu|laptopi know for a fact, polycom supports the RPID method.
22:57.00justinu|laptopbut it might also support the P-Asserted-identity method
22:57.16JTAsteriskMonkey: haven't had that problem myself, but who was that comment regarding ASNs aimed at, before?
22:58.04AsteriskMonkeydont know just saw someone making fun of someone who knew nothing about bgp so i threw in the asn comment :)
22:58.05AsteriskMonkeylol
22:58.35*** join/#asterisk aao_pwner (n=aao_pwne@c-24-21-91-140.hsd1.wa.comcast.net)
22:58.48JTAsteriskMonkey: you aided a troll who knew nothing about the knowledge of the person they were bagging out, actually :P
22:59.36JTno matter :)
23:00.07*** join/#asterisk r0d3nt|m (n=RatMan@foster.stonedcoder.org)
23:00.26monstedhe was just pissed when he found out that having cisco certs didn't mean you could actually do anything useful
23:00.34justinu|laptopCunningPike: i consider it a pleasure to help out the District of North Vancouver, BC, Canada ;)
23:00.44JToh i think he was here for no good since the beginning
23:01.06aydiosmioamg r0d3nt|m
23:01.09monstedJT: possibly just attempting a dicksize competition
23:01.15JTpossibly
23:01.22monstedJT: "Yeah, that's fine, but i'm better than you because i know Cisco!"
23:01.31CunningPikejustinu|laptop: ;) - bblack just sent me this: http://www.voip-info.org/wiki/view/P-Asserted-Identity+and+Remote-Party-ID+header
23:02.07JTmonsted: grasping at straws really "* _rnz- chuckles at asterisks fetid vermin ROI"
23:02.13r0d3nt|mamg aydiosmio
23:02.23AsteriskMonkeylol last time i used a cisco cert was when i ran out of rolling paper
23:02.24justinu|laptopCunningPike: ok, but my feeling is that will make aserisk send RPID/Asserted Identity on INVITEs when making the outbound call
23:02.24AsteriskMonkey:P
23:02.33monstedtoo bad he didn't try for real... i've got the login for enough stuff to make most trolls fold ;)
23:02.40justinu|laptopCunningPike: you need asterisk to send your phones RPID on 183/180/200 OK responses
23:02.44*** join/#asterisk yassine (n=yassine@xdsl-87-78-21-126.netcologne.de)
23:02.48yassinehi everyone
23:02.48CunningPikejustinu|laptop: OK
23:02.58yassineanyone here running asterisk on debian ?
23:03.05monstedyassine: many
23:03.16CunningPikejustinu|laptop: I'll try it and see what happens
23:03.49yassinemonsted, good news are there any prepackages packages or debian is only being used as a platform and everything is done manually ?
23:04.08*** join/#asterisk mosty (i=mostyn@60-241-198-194.static.tpgi.com.au)
23:04.23jtf0518If we try out this Asterisk solution we're thinking of either Suse Linux or Ubuntu server. I'd like to hear anyone's opinion on those platforms as a choice.
23:04.32justinu|laptopgentoo
23:04.38jtf0518now way!
23:04.42justinu|laptopubuntu then
23:04.50JTi always say go for the platform you are most comfortable with administering, jtf0518
23:04.52mostyi have just registered some more g729 licenses, can i get asterisk to use them without restarting?
23:04.56AsteriskMonkeylol apparently someone is trying to war dial my asterisk box lol... 4 channels vs my 8 pris...
23:05.00jtf0518I have very little hair on top of my head justinu|laptop :-)
23:05.12justinu|laptopgentoo is really slick one you get the hang of it
23:05.17NuggetLinux is poo.
23:05.42justinu|laptopjtf0518: if you're in a hurry and don't want to compile, go with ubuntu
23:05.42jtf0518I've heard but it's a nightmare to setup for most unless you're 133t
23:05.50KattyNugget: you!
23:06.00aydiosmioI've used CentOS and Debian for Asterisk
23:06.04justinu|laptopjtf0518: you're end up learning a lot of stuff you really should have known in the first place!
23:06.09aydiosmioI prefer Debian, as it's the platform Digium develops on
23:06.28Kattyall my stuff runs on debian.
23:06.33Kattyi'm an apt-get addict.
23:06.34justinu|laptops/you're/you'll/
23:06.35dlynes_laptopKatty!!!
23:06.45Kattydlynes_laptop: allo (=
23:06.51puzzledaydiosmio: iirc mark developed on fedora core for at least a while
23:06.57jtf0518I believe you justinu and I've heard good things about Gentoo (in fact I work at a zoo and we have real Gentoo penguins here) but time is of the essence.
23:07.08JTmy choice is debian
23:07.09Kattyjtf0518: penguins :>
23:07.10JTbut meh
23:07.21justinu|laptopjtf0518: understandable... got any pics of your Gentoo's btw??
23:07.23jtf0518We also have Rockhoppers too but no one's made a distro by that name yet.
23:07.44jtf0518justinu, not handy with me but I could get some and post them some other time.
23:07.50justinu|laptopthat would be cool
23:07.58jtf0518I'll make a point to do that.
23:08.11KattyNugget: i've got new projects going :>
23:08.11Waverly360Hmm..for you linux gurus out there..I know this really isn't the forum for it...but anyone have a clue what would keep services from starting up in the order they are specified in /etc/rc3.d on a fedora core box?
23:08.16KattyNugget: including bananapos :>
23:08.24jtf0518I can also get stuffed penguins at our gift shop at a discount too if someone wants a linux mascot toy.
23:08.25Nuggetyay
23:08.33*** join/#asterisk suma (n=suma@cm53.omega182.maxonline.com.sg)
23:08.34KattyNugget: and xen :>
23:08.43KattyNugget: xen is hotnessss
23:08.53sumahi how can i make asterisk to access extensions directly from mysql database ?
23:09.03dlynes_laptopsuma: real time extensions
23:09.12Nuggetsuma: you want the (catastrophically mis-named) asterisk "realtime" stuff.
23:09.13Corydon-wjtf0518: any 36 inch plush penguins?
23:09.26puzzledWaverly360: maybe you need to do a reset or resetpriorities on the funky service(s). check man chkconfig
23:09.26dlynes_laptopNugget: no doubt
23:09.41jtf0518Corydon, the ones I've seen are about 12 - 18 " but I can check for  you.
23:09.46sumadlynes_laptop: all the configurations taken from mysql ?
23:09.56Nuggetall the ones that matter.
23:09.57dlynes_laptopsuma: if that's what you want to use, yes
23:10.19sumadlynes_laptop: any doc how to configure the same? please
23:10.19dlynes_laptopsuma: basically any database supported by unixODBC
23:10.25dlynes_laptop~doc
23:10.29jbotDocumentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk
23:10.29dlynes_laptop~docs
23:10.30jbot[docs] Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html
23:10.31dlynes_laptop~wiki
23:10.32jtf0518Today we played around and got Asterisk working in a VM running Suse 10.1 with softphones.
23:10.36Corydon-wjtf0518: I haven't been able to find 36" penguins since linuxmall.com went out of business
23:10.36*** part/#asterisk hoobastooba (n=ckwall@63.149.122.93)
23:10.53Corydon-wjtf0518: and mine is beginning to show age
23:10.55justinu|laptopKatty: all well in mysql land?
23:10.55dlynes_laptopthere ya go, suma
23:11.06Kattyjustinu|laptop: i'm aptgetting teh package of sqlly goodness.
23:11.07aydiosmioI had Debian running in VMWare with * for a long time
23:11.10aydiosmioworked really well
23:11.50Kattyjustinu|laptop: 2 minutes to go (=
23:11.51Waverly360puzzled: doesn't the linkname of the startup script in that directory specify which order it's started though?  Or am I mistaken in believing that it's based on directory listing order?
23:12.03justinu|laptopKatty: what are you on, dialup??
23:12.10Kattyt1
23:13.08puzzledWaverly360: I think it's the link name but the priority is also noted in the service file name in /etc/rc.d/init.d
23:14.25jtf0518darn, that pdf file link in the wiki for Legacy Integration for an Option 11C is broken!
23:14.28monstedWaverly360: it's in numerical order (that is S01blah starts before S02blah)
23:15.26jtf0518Is that O'Reilly book on Asterisk any good?
23:15.51*** join/#asterisk slayer192 (n=slayer19@66.138.39.225)
23:16.04Nuggetimho asterisk development is too chaotic for dead tree books to be useful.
23:16.09Nuggetmaybe in another year
23:16.23Waverly360monsted, puzzled: Well here's the problem.  I have 'S19sangoma_config -> /etc/init.d/sangoma_config', then 'S20wanrouter -> /usr/sbin/wanrouter', then 'S40asterisk -> ../init.d/asterisk'
23:16.29*** join/#asterisk gerphimum (n=trekkie@cpe-70-114-42-210.satx.res.rr.com)
23:16.40Waverly360That should run the sangoma_config script, then wanrouter, then asterisk
23:16.48monstedWaverly360: correct
23:16.58Waverly360but in the var/log/messages file, it shows asterisk trying to startup before wanrouter
23:17.16Waverly360which would cause asterisk to bomb out, because there are no channels for it to find
23:17.21Waverly360er..zaptel devices
23:17.25Waverly360whatever you wanna call it
23:17.42*** part/#asterisk dasenjo (n=dasenjo@201.228.128.10)
23:17.53Waverly360what's killing me
23:17.57Waverly360is that I have two identical servers
23:18.03Waverly360both running the same stuff..setup the same way
23:18.07Waverly360one works right..the other doesn't
23:18.21Waverly360my problem is..asterisk isn't starting when the box has a power outage, or when it's rebooted
23:18.36Waverly360I can't recreate the problem on my test box...ONLY on the live one..of course
23:19.10monstedare the links in the right rcX.d directory?
23:19.11justinu|laptopsame filesystem?
23:19.22Waverly360yep
23:19.25Waverly360identical
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23:19.33justinu|laptopodd
23:19.39Waverly360tell me about it *headdesk*
23:20.35Waverly360I'm no linux noob..but this is killin me
23:20.36monstednote that they probably all start just about simultaneously - you could try putting in a "sleep 10" in the wanrouter script to give it a chance to find all the hardware before the script runs off to start asterisk :)
23:20.36Nivexwacky idea: Can I use an USB ACM capable cellphone as a channel for Asterisk?
23:20.39Waverly360well..and I can do that..but that still doesn't explain why it works on one box..and not the other...unless..
23:20.54Waverly360unless maybe the lab test box doesn't have the same amount of pri/analog cards in it.
23:21.07CunningPikejustinu|laptop: Ya - I tried the RPID and P-Asserted-ID and it sends them to the dialed phone
23:21.09Waverly360I guess wanrouter would take longer to startup if there are more devices for it to configure?
23:21.22monstedWaverly360: i'd think so
23:21.55justinu|laptopCunningPike: yeah, so you need gork's patch
23:22.39Waverly360and here's another little bit of weirdness that I don't understand.  In my var/log/messages file on the broken box..I have dates and times of Nov 15 13:46...but at one point..the time changes to 07:45...then back to 13:45
23:23.07Waverly360is /var/log/messages not guaranteed to be in order?
23:23.32justinu|laptoponly reason I can thing for that is that syslog might have restarted, and dumped the klog messages twice??
23:23.44justinu|laptopbut that doesn't sound right
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23:24.30Waverly360the time changes on this command rc.sysinit: -e
23:26.05CunningPikeWaverly360: I'm going to bet you're in CST
23:26.08*** join/#asterisk sjobeck (n=sjobeck@66-182-49-26.atgi.net)
23:26.16Waverly360Yes I am
23:26.25*** join/#asterisk Choubaka (n=odiq@c66.110.141-241.clta.globetrotter.net)
23:26.27CunningPikeWaverly360: Your timezone is changing
23:26.43Waverly360....
23:26.44CunningPikeWaverly360: From local to GMT and back
23:26.45Waverly360hah
23:26.48Choubakaanyone can help me with zaptel module?
23:26.49justinu|laptophow does that happen?
23:27.21CunningPikejustinu|laptop: Possibly something borked in the setup
23:27.25Waverly360hmm...I suppose that would explain it.
23:27.45CunningPikeWaverly360: When you installed linux, did you opt to use UTC for the system clock?
23:28.10Waverly360CunningPike: to be honest, I don't know.  It was an image that's auto installed on the box.
23:28.29CunningPikeWaverly360: That would be my suspicion anyway.......
23:28.31Waverly360hmm
23:28.47Waverly360How would I figure that out now?
23:28.55Waverly360Any thoughts?
23:29.35Waverly360here's the thing..this has been happening on several boxes
23:29.41Waverly360but not my test box..
23:29.47Waverly360I need to figure out what's different
23:30.05Waverly360Do you think my problems with services starting at the wrong time could be time related?
23:30.17NuggetI doubt it.
23:30.58Nuggetif any of the running processes has a TZ environment variable set to a different timezone than the system time (as defined by /etc/localtime) this will result in conflicting times reported by syslog logs.
23:31.09Nuggetbut there's no actual problem, just awkward logs.
23:32.49Waverly360well, then there's not much point in chasing that problem down
23:34.17Nivexhmm, looks like I'd have better luck with chan_bluetooth
23:36.19ChoubakaI can't place outgoing call, I've got this: fixlocalprefix: Could not open /etc/asterisk/localprefixes.conf
23:36.22Choubakaanyone?
23:36.38Nugget"can't" is a little vague.
23:36.58Nuggetdoes that file exist?
23:37.05Choubakanop
23:37.18Nuggetwhat's supposed to be in it?
23:37.23Choubakadunno :P
23:37.40yassineim trying to  set up asterisk on debian etch and while trying to run : m-a a-i zaptel i get this error : http://rafb.net/paste/results/WnPWpw63.html any help is apreciated
23:38.32Nuggetgoogle indicates that localprefixes.conf is an AMP thing.
23:38.57Nuggetor perhaps asterisk@home?
23:39.02ChoubakaNugget, yep I wee it.. but I think is not very important
23:39.09JTyassine: that's a really easy error
23:39.17ChoubakaI'm using trixbox v2 beta
23:39.18JTyassine: exactly like it says, you need to install gcc
23:39.29Nuggetyou won't get much help for that in here.
23:39.39yassineJT i also think so but i have gcc installed a newer version :s
23:39.55JThmm
23:40.02JTit's looking for that specific one
23:40.31yassineJT gcc version 4.1.2 20061028 (prerelease) (Debian 4.1.1-19)
23:41.58JTthere might be a package that allows you to install gcc-3.3 concurrently
23:42.43ChoubakaI got this message now.. localprefixes is now ok.. All circuit is busy now, please try again later..
23:43.15*** join/#asterisk e-horn (n=bp@cr2.pfn.citynetwireless.net)
23:43.20yassineJT im wondering how i can set the RELAX_CC_CHECK VAR
23:43.28*** join/#asterisk darkskiez (n=mbryars@bb-87-81-62-203.ukonline.co.uk)
23:44.01JTdebian often has packages for different versions of gcc
23:44.05JTi suggest you check into it
23:44.19NuggetChoubaka: I suggest you ask on the trixbox forums.
23:44.23JTmuch more likely to work, in case it doesn't like gcc4 for whatever reason
23:44.27NuggetI don't think anyone here will be able to help
23:44.29Choubakak, cool
23:44.38JTor #freepbx
23:45.28e-hornI'm having a problem where I'm trying to use * (a) to drop people into their voicemail menu when the greeting starts playing (so when traveling people can check their voicemail)... it's working fine when you call from a voip phone, but on a outside line (landline/cell/etc), the DTMF tones are ignored... however I know it's not a problem with dtmfmode or anything like that because calling into my IVR and choosing options works, any ideas?
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23:46.01*** join/#asterisk bulatitoy (n=rmn@adsl-70-231-130-250.dsl.snfc21.sbcglobal.net)
23:46.34bulatitoyi need help again
23:46.45shido6how are the calls coming in, e-horn?
23:47.24bulatitoycan i use a single phone line, X100P card then have 2-3 extensions?
23:48.07JTyes
23:48.55bulatitoywhat kind of card do i use to connect the phones?
23:49.05bulatitoysorry im really new to this
23:49.16JTif you need physical phones
23:49.32justinu|laptopethernet card
23:49.49JTthe main options are analogue phones + TDM400P with FXS modules, T1 + channel bank, or SIP VoIP phones via ethernet
23:50.04*** join/#asterisk rollergrrl (n=0x3e44d@71-213-6-123.slkc.qwest.net)
23:50.20bulatitoyso i will install the X100P on the linux machine running asterisk
23:50.29rollergrrlHas anyone heard of a new federal DID tax?
23:50.32JTSIP VoIP phones usually offer the best flexibility
23:50.49JTbulatitoy: do the extensions need to be different, or is shared ok?
23:51.05bulatitoyJT: shared is fine
23:51.28bulatitoyi just want to know how everything works and I will do more experiments when i get this working
23:51.46JTright so there bare minimum for that if you went with an analogue setup is a TDM400P with 1 FXS module
23:51.53JTand if you did that
23:51.59JTyou may as well get an FXO module too
23:52.03JTbecause X100Ps are crappy
23:52.07bulatitoyi see
23:52.09JTand no longer produced by digium
23:52.22JTyou might get one that works, you might not
23:52.26bulatitoyso the setup would be like this
23:53.00bulatitoyPSTN---X100P or TDM400P on Linux/Asteris machine --- phones
23:53.05bulatitoyis that correct?
23:53.09JTof course you can always us PC based softphones for testing
23:53.14JTyeah pretty much
23:53.36JTphones would connect via the TDM400P too if they were analogue
23:53.41JTor you could go voip
23:53.43bulatitoyif i use softphones, is it safe to say, i was like testing phone hardware?
23:53.57e-hornshido6: from a SIP provider
23:53.58JTbulatitoy: what do you mean?
23:54.15shido6do u see "*" come in when u debug?
23:54.34e-horndebug will show that?
23:54.34bulatitoyare they the same? softphone and the actual phone device/unit?
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23:54.59JTbulatitoy: they aren't exactly the same, but they can still access the phone lines and what not
23:55.09JTthe differences will be in the features, and how you use them
23:55.32e-hornlike I said it works fine when other voip phones call the number and hit *... so you would think if it does nothing when pressing * on outside line that it's just not coming in, but if maybe it's ignoring it for some reason.... I don't know
23:55.35JTanalogue phones don't have that many features, but are easy to use to an extent
23:55.53e-hornand whether it's a voip phone or outside call it all drops into the same context/extension/etc
23:55.59bulatitoyi would like to try first analog, just to see how they are interconnected
23:56.04JTyou'd need to use feature codes to do anything much pabx like on analogue phones
23:56.04e-hornso there's nothing different
23:56.20JTok, that's fine as long as you're prepared to pay for the TDM400P :)
23:56.27JTsoftphones are cheap (free mostly)
23:56.42bulatitoyi am having a hard time looking for docs that shows how to connect the necessary hardware
23:56.58JThave you found voip-info.org?
23:57.17bulatitoyill check on that
23:57.19JTand the book
23:57.21JT~thebook
23:57.30jbotfrom memory, thebook is a book called  Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11
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23:57.46bulatitoythanks for the link!
23:57.52JTnp
23:57.59JTdigium also put out a handbook too
23:58.04JTand there's other random stuff online
23:58.18bulatitoyso i can go TDM400P and a single phone line (my home phone) + softphones?
23:58.23JTvoip-info is a real resource provided you can find what you're looking for
23:58.29JTbulatitoy: yes
23:58.39JTthe TDM400P is a board with a space for 4 modules
23:59.01JTthere are 2 types of modules, FXO and FXS, for exchange connection, and handsets, respectively
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23:59.20e-hornshido6: I just tried turning on sip debug for a peer, and it didn't print anything when I pressed * calling in from outside
23:59.40*** part/#asterisk kschava (n=kschava8@blk-222-18-178.eastlink.ca)
23:59.54e-hornshido6: but it doesn't show anything even if I do the debug on a working phone, so...
23:59.57bulatitoyi can see that some TDM400P has 4 FXO

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