irclog2html for #asterisk on 20060411

00:02.41*** join/#asterisk hinckc (n=hinckc@ool-4354c627.dyn.optonline.net)
00:05.41*** join/#asterisk kimosabe (n=kimosabe@dsl-201-133-195-203.prod-infinitum.com.mx)
00:05.48*** join/#asterisk SPoon_TSX (n=klee@h24-83-96-211.sbm.shawcable.net)
00:06.04SPoon_TSXHello there, For all of the experts here. I got a quick question.
00:06.12eric_snight
00:06.16*** join/#asterisk kainam (n=Jake@202.137.160.110)
00:06.42*** join/#asterisk apardo (n=apardo@87.218.45.206)
00:06.43SPoon_TSXI have an asterisk installed and everything works okay except I got some whitenoise at the background when I make a PSTN call. Sip to SIP is prefect.
00:06.57SPoon_TSXI am wondering what could possible causing the problem?
00:07.49*** join/#asterisk hinckc (n=hinckc@ool-4354c627.dyn.optonline.net)
00:07.53SPoon_TSXanyone?
00:08.46kimosabei have a tdm400p card and it gives dial tone and i have a sip account on a remote server any help me with a  config example so that i can dial via tdm400p using sip acount
00:09.29kimosabeis any one here from selectfone ??
00:14.27*** join/#asterisk miztic (n=gerard@rarcoa.com)
00:14.59*** join/#asterisk harryvv (n=none@S010600a0c93f6f7e.vs.shawcable.net)
00:15.14harryvvWhat ntp servers do you all use for your phones?
00:18.48*** join/#asterisk Darwin35 (n=Darwin@c-24-9-75-234.hsd1.co.comcast.net)
00:19.20[av]banimy own
00:19.32[av]banido not abuse happy fun public ntp servers
00:19.49Saturn--I have musiconhold setup
00:19.56Saturn--i can play an mp3 with "mp3player()"
00:20.04Saturn--However when i try and put a call on hold it just says
00:20.09Saturn--started music on hold
00:20.11Saturn--then right away
00:20.14Saturn--stopped music on hold
00:20.16Saturn--no explanation
00:20.43Saturn--lewloal
00:21.52harryvvnaa, just looking for one other then redhat.
00:22.23Saturn--Trying to figure out why it would do this
00:22.49brodiemharryvv, ntpd.
00:23.04*** join/#asterisk miztic (n=gerard@rarcoa.com)
00:25.18generalhananyone in here using multiple digium cards in the same server ? like a TE card and a TDM card ?
00:25.34harryvvbrod I know its technically called ntpd but thats not how is talked about in general conversation or even a google search.
00:26.06*** join/#asterisk znoG (n=gs@109-130-89-200.fibertel.com.ar)
00:31.27*** join/#asterisk miztic (n=gerard@rarcoa.com)
00:33.36harryvvbtw, anyone ever get cwcn running on a ip500?
00:34.15DoktorGregcan anyone post a link to get me started on sorting out music on hold and ivr crackling problems?
00:34.50harryvvsounds like a ground issue. mabey bad mic problem when recording it?
00:35.12DoktorGregdefault sounds, full drop outs...
00:35.23DoktorGregboth soft and hard phones are all working perfectly to each other...
00:35.53DoktorGregonly getting the static when i hit voicemail, ivr and moh
00:36.10DoktorGregthe emailed voice mails seem to sound ok
00:36.35harryvvodd
00:36.52hinckcany relation to server load?  does it happen on when only 1 user is there?
00:37.13DoktorGregyup...  any relation to sound card drivers???
00:37.15Darwin35~amp
00:37.17jbot[amp] "amp is, like, NOT supported here! people using it should join #freepbx (the new name of amp)"
00:37.48k-mananyone know if there is a tapi driver for windows that can direct a call through voip?
00:38.15tzangerman I love when people who don't have clue try to sound technically ept in -users
00:38.33tzangerthat whole te110 interrupt problem thread
00:38.46*** part/#asterisk The_Blob (n=jcragg@24.85.224.102)
00:39.00tzangerthe guy's got a local apic and the cards are on differnet APIC IRQs but the one guy says to turn off APIC to get them on "real" irqs... heh
00:39.29tzangerdude's problem is that he has three digium cards in one box, the interrupt load is pretty high (3000 interrupts/sec not counting anything else)
00:40.21generalhantzanger: i have 2 cards in one system and im having some issues ... is this common ?
00:40.43DoktorGregdigium cards are really really reluctant to work properly and share irq's
00:40.51tzangergeneralhan: high interrupt load isn't cool
00:40.57generalhanwell i checked on that .. they arent charing the IRQs
00:41.01tzangerDoktorGreg: I have next to no issues doing so
00:41.21generalhanwell i really dont have a choice i have a TE210 for the PRI lines ... and i have a TDM for my fax machines
00:41.24tzangeron moderate hardware, yes, of course.  WIth shitty products you're sharing IRQs with, of course...
00:41.36tzangerbut if you have good drivers sharing interrupts ins't too much of an issue with digium cards
00:41.57generalhanwell * refuses to load my TDM card stuff ... it just wont do it
00:41.59DoktorGregwhat mobo do you suggest tzanger?
00:42.23generalhanand im working on a production server so i have to wait until after 8pm MST to shut down the lines to test some things
00:42.40harryvvanyone here on the west coast using a ntpd server thay find reliable and open?
00:42.48tzangergeneralhan: it shoudl load, but work like shit
00:43.22generalhanit wont do it i dont know why ... but im testing some different things out
00:43.30*** part/#asterisk epablo (n=epablo@WLL-24-pppoe205.t-net.net.ve)
00:43.35DoktorGregwhat about digium on sunfire servers?
00:43.48generalhanlike for some reason when i load the wct4xxp it gives me an error about my TDM card ... so im not sure what that is all about
00:44.10generalhani checked zapata.conf and zaptel.conf like 1000 times to see what was happening and it looks good to me
00:44.31harryvvgeneralhan how many phones in your network?
00:44.39Saturn--any suggestions on why the musiconhold would "stop" like immedately after it starts, and never play
00:44.39Saturn--?
00:45.05harryvvSaturn-- do a top and play the vm and see what happens.
00:45.07generalhan30-40 Aastra SIP phones and 15 Cisco 7960s
00:45.15harryvvalso turn on cli in asterisk
00:45.20tzangergeneralhan: ohh
00:45.22generalhanwhy do you ask
00:45.23harryvvthat is, look at cli
00:45.32Saturn--what
00:45.32harryvvgeneralhan just curios
00:45.35tzangerit's complaining because the default config runs ztcfg after the module load
00:45.40Saturn--i am using the cli
00:45.46tzangerand you didn't load BOTH drivers so ztcfg will fail out because some channels are missing
00:45.54Saturn--voicemail isn't related, i donte ven have it on
00:45.58harryvvdoes it return errors in cli when playing vm?
00:46.07harryvvwhats the issue then
00:46.14Saturn--not voicemail, musiconhold
00:46.21generalhanwell i made a script so that i wouldnt get messed up ... it loads zaptel, then wct4xxp then wctdm
00:46.28Saturn--it just says "Started music on hold...."
00:46.35Saturn--then "Stopped music on hold"
00:46.38Saturn--immediately after
00:46.40Saturn--no error, nothing
00:47.05tzangergeneralhan: yes, but if your modules.conf still says "post-install /path/to/ztcfg" it'll error out
00:47.48generalhanhmmm
00:48.52generalhanin modules.conf i just have autoload=yes
00:49.08harryvvSaturn-- first install?
00:49.10generalhanand load=>res_musiconhold.so
00:49.17tzangerno /etc/modules.conf or /etc/modprobe.conf (the kernel module config stuff, not asterisk)
00:49.27generalhanohh
00:49.58generalhaninstall wct4xxp /sbin/modprobe --ignore-install wct4xxp && /sbin/ztcfg
00:50.03generalhanthat stuff ?
00:50.59QwellSaturn--: You need to install mpg123.
00:51.19QwellSaturn--: from the asterisk source dir, type `make mpg123; make install`
00:51.54*** join/#asterisk angom_h (n=angom@red-corp-201.130.165.94.telnor.net)
00:52.15tzangergeneralhan: yeah, or just erase the crap in modules.conf and do it manually :-)
00:52.19tzangerlike a real man :-)
00:52.29generalhanhaha "real man" he says !
00:52.49*** join/#asterisk cced2 (n=dev2003@222.33.36.205)
00:53.45generalhani dont know enough about linux to fill a cup ... you say manual ANYTHING to me and my answer is no ! hahaha !
00:53.45*** join/#asterisk [hC] (n=hardcore@mail.metrobridge.com)
00:53.56tzanger:-)
00:54.19generalhanok well i cant do anything while my reps are still on the phone ... so im going home to wait till after hours to test some stuff
00:54.24generalhanill be back then guys !!
00:54.28generalhanthanks for all the help !
00:55.11Saturn--Qwell
00:55.17Saturn--i just installed mpg123
00:55.22Saturn--the mp3player() function works
00:55.24*** join/#asterisk Jaxxan (n=jaxxan@202.70.125.124)
00:55.29justinuqwell
00:55.31justinufix my shit
00:55.39Qwelljustinu: fix mine
00:55.40generalhanjustinu: HAHAHAHA
00:55.46Saturn--but the musiconhold still does not
00:55.47Qwellor pay me, newb
00:55.48generalhanQwell: HAHAHAHAHAHAHAHAHAA
00:55.48Qwell:P
00:56.08QwellSaturn--: Did you install mpg123 the way I said?
00:56.13Saturn--yes
00:56.19Qwelland you restarted *?
00:56.22Saturn--i probably have the config wrong
00:56.22*** join/#asterisk Eggplant (i=No@dsl-469.cascadeaccess.com)
00:56.22Saturn--yes
00:56.44*** part/#asterisk generalhan (i=general_@ip67-90-64-2.z64-90-67.customer.algx.net)
00:59.03Saturn--But it doesn't indicate any trouble with the config
01:05.51*** join/#asterisk Igbothom_III (n=Hilton@office.quarkit.com.au)
01:06.15Jaxxanyou dont need mpg123 with 1.2.x versions right ?
01:06.23QwellJaxxan: "need", no
01:06.25Qwellbut you need something
01:06.40Jaxxanthought asterisk 1.2.x had it's own player
01:06.49*** join/#asterisk hinckc (n=hinckc@ool-4354c627.dyn.optonline.net)
01:06.54Qwellthere is format_mp3 in asterisk-addons
01:07.00Jaxxandont get me wrong, i think i'm still using mpg123
01:07.39justinugood... i was about to get you wrong
01:07.54harryvvanyone here have a ip500 and know what it takes for a Caller Waiting party to ring the other line ?
01:08.15Jaxxani upgraded from 1.0.9 to 1.2.6 though, that's why i have mpg123 setup, i was just reading something about 1.2.x having it's own player is all. just wondering.
01:08.44*** part/#asterisk alephcom (n=alephcom@host75.net14.mcsnet.ca)
01:09.07Jaxxani have a problem with my queue_log file. it's not recording callerid(name)
01:09.19Saturn--it can play with mpg123 i am assuming, for mp3player function, just onhold it feels it doesn't want to for some reason
01:09.19Saturn--ohwell
01:09.23Jaxxanit only shows callerid(number)
01:09.44Jaxxanand i need it to show the calleridname
01:09.58Jaxxanwell, i need it to print it in the queue_log
01:10.14Jaxxanit printed when i was using 1.0.9, but 1.2.6 doesn't print it.
01:10.44Jaxxanand it sucks cause with the way i do reporting with queuemetrics now
01:11.17Jaxxani label calls based on callerid rather than separate queues and now i can't see what kind of call it was
01:11.43*** join/#asterisk brockj49464 (n=brockj49@63.87.56.236)
01:11.56Jaxxanunless i parse the cdr, but that doesn't do me any good.
01:12.23justinuSaturn--: did you make sure you answered the call?
01:12.34Saturn--yeah
01:12.42Jaxxanhow do i go about getting that changed ?
01:12.44Saturn--if i didn't it would just keep ringing
01:12.54Saturn--even though CLI shows the proc as running
01:12.54*** join/#asterisk tuxd00d (n=tuxinato@netblock-68-183-112-100.dslextreme.com)
01:14.50*** join/#asterisk project_2501 (n=project-@S01060004e2929dc9.br.shawcable.net)
01:20.39justinuSaturn--: got ztdummy loaded?
01:21.00Saturn--isn't working on NetBSD
01:21.11Saturn--I got it to play music, but only if there's like incoming noise
01:21.18Saturn--if I am muted or silent, the music stops
01:21.20Saturn--until i make more noise
01:21.21Saturn--i dont get it
01:21.29QwellSaturn--: turn off VAD/Silence Suppression on your phone
01:21.30*** join/#asterisk MacDome (n=eseidel@c-71-198-177-144.hsd1.ca.comcast.net)
01:22.00Qwell~vad
01:22.01jbot[vad] Voice Activity Detection or Silence Suppression. Asterisk does not currently support this, so please turn it off on the client
01:22.09Saturn--thanks, perfect
01:22.21Saturn--except for being choppy, but what can I expect from california to new york
01:22.24Saturn--thanks
01:22.34QwellYou can expect...perfection
01:23.29Nugget* Void where prohibited by law
01:24.51Saturn--I better read more then
01:24.53Saturn--to tweak this up
01:25.11*** part/#asterisk project_2501 (n=project-@S01060004e2929dc9.br.shawcable.net)
01:25.32Saturn--also the encoding of the mp3's isn't helping
01:27.07Saturn--but that i know how to fix
01:27.50*** join/#asterisk cced (n=dev2003@222.33.36.205)
01:28.08ccedJaxxan :hi
01:28.24Jaxxanyo
01:29.02*** join/#asterisk riddlebox (n=james@24-171-10-102.dhcp.stls.mo.charter.com)
01:29.51Jaxxanhow goes the documentation hunt
01:30.18cceddoing now .Jaxxan.
01:30.29fileNugget: okay that was baddddddddddd
01:30.34Nuggetmoo?
01:30.50fileNugget: mooooooooooo
01:31.00Nuggetapnp 'oow umop apisdn ue w,i
01:31.02*** join/#asterisk gammacoder (n=chatzill@cpe-65-26-178-240.indy.res.rr.com)
01:31.18ccedJaxxan: I want to draw pri state machine figure.
01:31.23QwellNugget: That was painful
01:31.34ccedwhich figure I can use in Q.921 931 spec?
01:32.46Jaxxanno clue
01:32.53Jaxxanthat's to technical for me
01:33.00Jaxxani know know how to get it to work
01:33.05Jaxxanerm, i just know
01:33.50Jaxxanmy boss said, make it work and i did.
01:34.02justinuwhy do you want to draw a pri state machine figure?
01:34.08fileso Qwell
01:34.43ccedfaint. yes .it work well.~~
01:35.27Lino`lol
01:35.35Lino`verizon might want to buy vodafone.airtouch
01:36.10Lino`whole vodafone
01:36.19Qwellso file
01:36.30fileno soup for you
01:36.36*** mode/#asterisk [+o file[laptop]] by file
01:36.43Qwellack!
01:36.56fileI won't hurt you!
01:36.59QwellWould have been FAR better, as a kickban message.
01:37.00Qwellsheesh
01:37.17filebat!
01:37.31fileQwell: translation?
01:37.38fileBuild, Asterisk, Test!
01:38.04Qwellwhy test?
01:38.56filewhy not?!?
01:39.12Qwellbecause it compiles
01:39.34Corydon76-homeIt compiles!  Ship it!
01:39.43Qwellexactly
01:39.50filecrazy crazy people
01:40.10Corydon76-homePot.  Kettle.  Black.
01:40.16QwellI use the following development model:
01:40.30fileoh.
01:40.33Qwellcode, compile, test, code, compile, test, code, compile, release
01:41.12Corydon76-homeQwell version 1.0.0.368b19
01:43.32*** join/#asterisk r0d3nt|m (n=RatMan@foster.stonedcoder.org)
01:43.46Qwellat work, I probably compile every...10 minutes, tops
01:44.27*** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net)
01:46.58*** join/#asterisk Myconid3 (n=myconid@69-164-122-221.sbtnvt.adelphia.net)
01:46.59*** join/#asterisk MacDome (n=eseidel@c-71-198-177-144.hsd1.ca.comcast.net)
01:47.42techman97_andyhey all again...new time of day, new people, hopefully some new ideas.  I'm a 100% SIP environment (SIP xLite for clients, SIP provider for PSTN access), and I'm having an issue where I can dial (or receive) the call, but the RTP traffic won't flow until I place the caller on hold and pull them back off (as quickly as a double click).  I've tested this with the same results from different networks / extensions, and it's alway
01:48.18Myconid3what is the most mature commercial asterisk product
01:48.36tainted-Myconid3 for doing what
01:48.53Qwelltainted-: asterisking
01:48.55Myconid3Running in the enterprise
01:49.00justinutechman97_andy: capture the network traffic and analyze it with ethereal. figure out why rtp doesn't flow by looking at the SIP and SDP messages
01:49.04Myconid36 sites, 170 clients
01:49.18Myconid3"no dropped calls" is concern #1..
01:49.23Myconid3which im not sure asterisk can provide
01:49.25techman97_andyjustinu:  Can you give me some guidance on how to set that up (never used ethereal)
01:49.29tainted-i would have to say Qwell is a pretty mature product of asterisk
01:49.40Qwell~qwell
01:49.44jbotyou are, like, a patented liquid formula that contains three plant-based bio-active agents that work together in a perfectly balanced combination. These agents act synergistically to boost your good cholesterol and slash the bad.
01:49.44tainted-lol
01:50.03Qwellsynergistics, baby
01:50.11tainted-techman97_andy try with other softphones
01:50.14justinutechman97_andy: use "tcpdump -s0 -w filename.cap" to capture the traffic on your * server
01:50.29justinutransfer the cap file to your workstation and load it into ethereal
01:50.39techman97_andywhere can I get ethereal?  is it freeware?
01:50.40tainted-Myconid3 sounds like the project is over your head frankly.. outsource it
01:50.56justinugoogle
01:50.57justinuit's free
01:50.59techman97_andyk
01:51.02techman97_andybrb
01:51.02Myconid3thanks for your expert analysis.
01:51.12QwellMyconid3: Do you have $1 million to spend?
01:51.20*** join/#asterisk cced2 (n=dev2003@222.33.36.205)
01:51.22Myconid3Qwell: if we did, we wouldnt be running Asterisk ;)
01:51.23QwellBecause if not, you're going to have dropped calls, with any solution
01:51.29[hC]Myconid3: good idea, ask for help then be a cock.
01:51.31cced2<cced> Qwell : which part are yo familiar?libpri zaptel asterisk?
01:51.41Myconid3Qwell: We dont have any with our current nortel pbx.
01:51.41tainted-no dropped calls have never been a concern for anyone involved with asterisk or VoIP for that matter
01:51.42*** join/#asterisk MacDome (n=eseidel@c-71-198-177-144.hsd1.ca.comcast.net)
01:51.43Qwellcced: asterisk
01:51.52Qwell[hC]: You!
01:51.58Qwell[hC]: Where are my test results? :P
01:51.59[hC]Qwelzz!@#13123111oneone
01:52.00cced2yes. chan_zap chan_sip channel.c
01:52.09[hC]Sorry, I had a friend over for the weekend
01:52.12[hC]and nerdery ceased progress.
01:52.18Qwellexcuses, excuses
01:52.21[hC]:)
01:52.29[hC]When I go home tonight, I'll load up your channel driver
01:52.35[hC]now that my phone is running sccp
01:52.44Qwellcool
01:52.48Myconid3so asterisk is still in its infancy?
01:52.49cced2Qwell: I want to drwa  SIP state diagrams
01:53.03[hC]I hate the SCCP firmware for 7960 compared to the SIP firmware
01:53.16tainted-Myconid3 not if u know what you're doing
01:53.26Qwell[hC]: because you don't have a good skinny channel driver to use :p
01:53.27[hC]Few reasons: Blind Transfers.  Call Forwarding (maybe just chan_sccp's way)
01:53.30Qwellsccp rocks
01:53.35[hC]well
01:53.37Qwellyes, chan_sccp cfwd is stupid
01:53.43[hC]on chan_sccp, i already want to switch back its that bad
01:53.47Qwellcan only use it offhook...wtf is that?
01:53.48[hC]however on the 7970 its awesome
01:53.51[hC]yeah
01:54.01Qwellmine will let you do it onhook :)
01:54.06[hC]and the blind transfer thing, apparently its a SIP hack to make it work like the way id expect on SIP
01:54.12Qwellhuh?
01:54.17[hC]You cant blind xfer
01:54.24Qwelllame
01:54.27[hC]you either hit transfer the second time REALLY FAST or decide to speak to the person
01:54.30cced2Qwell . how chan_zap read or write data from zaptel?
01:54.39Qwell[hC]: okay, I'll keep that in mind
01:54.55Qwellwill try to think of a solution
01:54.58[hC]I asked sergio and he said that the only reason the sip firmware can do the 100% unattended transfer is because you can use sip headers to do it seamlessly
01:55.08Qwellwell...he's an idiot. :)
01:55.13[hC]Also, Im not sure if this is the phones fault or what
01:55.21[hC]but the 7970 will NOT retain placed/received calls lists
01:55.33QwellThat is a firmware thing, I'm pretty sure
01:55.38[hC]<PROTECTED>
01:55.46Qwellweird
01:55.54[hC]I thought it may have been a locale thing since i get an error updating locale msg
01:56.02[hC]but I dont know how to correct that, Ive searched for the files with no luck
01:56.10QwellHave you tried the 8.0 firmware?
01:56.16[hC]Yah
01:56.22Qwellsame thing?
01:56.27[hC]Yepp
01:56.38Qwellstrange
01:56.43[hC]Im pretty sure it asks for a td-sccp.jar at startup
01:56.44[hC]that I dont have
01:56.47GamercjmHmm can i have asterisk record my voice? like call my DID and have it record a msg for me
01:56.48[hC]and it may be related to that
01:56.55[hC]It seems to come from callmanager
01:56.58[hC]cause it didnt come with the firmware.
01:57.18*** join/#asterisk Tili (i=Tili@61.140.191.181)
01:57.22tainted-i need some asterisk work done. i heard from a friend that it's really cheap and fast and should cost around $50.00 (including hardware).  I need to connect 3 remote offices, ~80 lines and have 100% uptime.
01:57.37Qwelltainted-: heh
01:57.53Ariel_$ 50.00 hummm
01:58.06[hC]tainted-: my friend said the same thing, and said that all you guys will help me do it for free, and you're all SUPER HELPFUL
01:58.06[hC]:)
01:58.09Ariel_wow, phones along cost more then that.
01:58.14Gamercjmill install asterisk for $50 ;)
01:58.16Dream_WEaverheh
01:58.17Ariel_The software is free
01:58.24Gamercjmlol
01:58.29Dream_WEaverInstall, yes.  Configure - forget it :)
01:58.47tainted-well i heard there are like free phones (like skype) so don't try to rip me off
01:58.56DoktorGregIll install a complete asterisk system with 10 phones for 8k if anyone wants it
01:58.56Gamercjmwell softphones
01:58.58Gamercjmare free
01:59.02Ariel_tainted-, there are free softphones yes
01:59.15Ariel_xlite works well
01:59.20Ariel_and yes there free
01:59.22DoktorGregxlite seems to work great
01:59.29Myconid3tainted-: I have 70 7960's if that makes it any easier :P
01:59.29Ariel_but you need to start doing some reading.
01:59.31Ariel_~docs
01:59.33jbotit has been said that docs is probably Documentation can be found at http://www.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 or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html
01:59.33DoktorGregiaxComm works well also, but is fugly
01:59.36Dream_WEavertainted: Clearly you have been mislead if you thought there were some people here that will install your system cheap and effectively :)
01:59.45Dream_WEaverThis is more of a support channel than anything.
02:00.17DoktorGreghow do you deploy software to those 7960's?
02:00.26JaxxanAnyone have any idea how to get the callerID Name to write to the queue_log ?
02:00.28tainted-omg!!11 so ungreateaful 4 work!!!!!!11
02:00.29Ariel_7960 software tftp
02:00.33Myconid3DoktorGreg: cisco call center :~)
02:00.36DoktorGregand if you dont mind me asking...
02:00.43DoktorGregwhat kind of software do you run on em?
02:00.50QwellDoktorGreg: the firmware
02:00.55[hC]cisco's firmware.
02:00.56QwellThat's all you run "on" them
02:01.06Dream_WEaverCisco - the one that doesn't account for DST
02:01.17Dream_WEaverUnless you pay extra for the firmware upgrade.
02:01.21Ariel_polycom, polycom's
02:01.25Dream_WEaver(or it comes with the phone)
02:01.28Dream_WEaverlike polycom
02:01.47DoktorGregso its a phone with a big color screen and ldap support???
02:02.06Myconid3docelm0[QUOTE=BlackMaxima21]my friend has a 2002 se automatic and has this weird shifting problem from 2nd to 3rd.  it doesn't want to shift right away, if you keep your foot on the gas pedal the rpms keep going up but your not accelerating and then they drop down when it shifts into 3rd.  Do you guys have an idea of what this could be??[/QUOTE]
02:02.08Myconid3erm.
02:02.11Myconid3stupid pastey.
02:02.38cced2Hi Ariel_
02:36.26*** join/#asterisk jbot (i=ibot@pdpc/supporter/active/TimRiker/bot/apt)
02:36.26*** topic/#asterisk is Asterisk: The Open Source PBX -=- Asterisk 1.2.6 and Zaptel 1.2.5 Released! (March 27, 2006) -=- http://www.asterisk.org/ -=- AMP/FreePBX users should join #freepbx for support
02:36.35Myconid3omg.. i just downloaded the stupid mac torrent of command and conquer.. grawr.
02:37.26techman97_andyhmmmm
02:37.49DoktorGreghow do i tell what is a good setting for my jitter buffer/.
02:37.52DoktorGreg?
02:38.05techman97_andyI have iptables on the asterisk server, allowing UDP 10000-20000 through.  I see that the RTP packets from my provider are 20574...does that make sense?
02:38.18justinusrc port 20574?
02:38.43techman97_andyGot RTP packet from 64.xx.xx.xx:20574 (type 0, seq 65279, ts 200960, len 160)
02:38.43techman97_andySent RTP packet to 70.xx.xx.xx:10000 (type 3, seq 9500, ts 13280, len 33)
02:38.57techman97_andy70 is me, 64 is my SIP provider
02:39.00justinuyou iptables rule should be allowing packets to dest port 10000-20000
02:39.02techman97_andythat was from the CLI
02:39.11techman97_andyyeah, I'm doing that in iptables
02:39.35justinuthe question is, do you get any RTP from the provider before the phone goes on hold?
02:39.46justinuor from the phone
02:39.48techman97_andynope.  no RTP at all until the hold/offhold thing
02:39.57justinuvverified that w/ ethereal right?
02:40.03techman97_andyas far as I can tell, yes.
02:40.30justinuwell... i dunno... asterisk won't send RTP unless it gets any
02:40.44Jaxxanif i'm using agents.conf to record agent calls as gsm, how can i increase the volume of the file?
02:40.49Jaxxanit's really low
02:40.55Jaxxanor should i be using a different format ?
02:40.57techman97_andyso maybe that 20574 udp port that my provider is sending is being regarded as unsolicited by iptables...
02:40.58techman97_andy?
02:41.06justinuif your behind a nat, possibly
02:41.19techman97_andymy SIP client is behind a NAT, server is just ipchains.
02:41.25techman97_andyand ipchains is on the * server
02:41.36techman97_andymy SIP provider is public
02:41.37justinuserver is public ip?
02:41.44techman97_andyserver is public IP - long story.
02:41.47justinuk
02:42.08justinuwell... you say you're not recieiving any rtp from the provider until the phone goes on/off hold?
02:42.14techman97_andycorrect.
02:43.00justinuwhats weird is the softphone should start sending rtp as soon as it goes off hook
02:43.03*** join/#asterisk cced (n=dev2003@222.33.36.205)
02:43.10justinuthat should be enough to kick off the whole chain
02:43.22cceddo zaptel read or write data from hardware using DMA OR mmap?
02:44.03*** join/#asterisk [TK]D-Fender (n=joe@66.11.164.239)
02:44.18techman97_andyI may have misstated myself - let me back up.  From my understanding of SIP phones, *, and SIP providers, is that I go offhook, dial the number...* places the call on my behalf and acts as an RTP "funnel" between the provider and my softphone, correct?
02:44.55*** join/#asterisk Strom_M (n=strom@gateway.digium.com)
02:44.57techman97_andybut even taking that into account, SIP softphone to SIP softphone within Asterisk still has the same problem....hold/offhold, and we're good.
02:45.24[av]banihttp://www.beckysweb.co.uk/beckysblog/2006/03/conversational-ebonics.asp
02:45.30techman97_andymy environment is the Asterisk server has a public IP, and I have softphones that connect from their homes / private offices....so everyone is flippin' NAT'd.
02:45.48*** join/#asterisk file (n=jcolp@mctnnbsa24w-142167060049.pppoe-dynamic.nb.aliant.net)
02:46.38Hmmhesayswell guys wish me luck
02:46.39Hmmhesaysi'm off
02:46.48techman97_andybye bye
02:47.08justinutechman97_andy: rtp proxy is term
02:47.31*** join/#asterisk Strom_M (n=strom@gateway.digium.com)
02:47.32justinuagain, the problem is with the softphone
02:47.48justinua real phone would start sending rtp right as it goes off hook
02:48.05justinui've used xlite and not had that problem, so it must be a config issue
02:48.14justinubut xlite has a lot of options, so have fun...
02:48.29*** join/#asterisk alephcom (n=alephcom@host75.net14.mcsnet.ca)
02:48.43techman97_andyThis has to be a blocked port thing...I initially set this whole thing up in one location on the same subnet and everything was cool
02:48.58techman97_andyas soon as I seperated and distributed, I've been having problems like this...:S
02:49.17justinuwell, if the phones are on the same lan, doubtful
02:49.35justinuperhaps xlite is using stun and incorrectly determining it's IP
02:49.53*** join/#asterisk Jzalae (n=sk@dsl-66-63-110-48.gwi.net)
02:53.54[TK]D-Fender[av]bani : Welcome to America's linguistic future... not the Portugreek of BladeRunner, but it'll do in a pinch ;)
02:54.38[TK]D-Fendertechman97_andy : Same stupid NAT issues?  Try a 3rd phone, to see if its just one side thats bitchy.
02:54.56techman97_andyI'm back - was reading
02:55.21techman97_andyFender:  I've changed a few things...1 sec
02:55.32ccedwho are familiar with zaptel?
02:56.17[TK]D-Fendercced : just ask the question...
02:59.12tainted-is DUNDi dead?
02:59.23file[laptop]okay, DUNDi is a protocol.
02:59.29cceddo zaptel read or write data from hardware using DMA OR mmap?
02:59.32file[laptop]so it can't exactly be dead
02:59.35justinuhah
02:59.37tainted-yes it can
02:59.39justinuDECnet is dead
02:59.42cced<[TK]D-Fender>  :)
02:59.53justinugopher is dead
02:59.55tainted-smoke signals is a protocol
02:59.58file[laptop]there's a difference, it's no longer used :P
03:00.13tainted-well dead = no longer used in a practical sene
03:00.18tainted-sense
03:00.20*** mode/#asterisk [+o file] by file[laptop]
03:00.20*** join/#asterisk tengulre11 (n=tengulre@221.11.5.180)
03:00.21justinuso is latin not a dead language?
03:00.28file[laptop]but no, some companies are using DUNDi
03:00.30tainted-do poeple use it
03:00.40tainted-no one is in #dundi
03:00.52tainted-is it still actively developed
03:01.03file[laptop]what else is there to develop on it?
03:01.11*** join/#asterisk websae (i=icechat5@CPE-24-167-204-30.wi.res.rr.com)
03:01.30tainted-so development is done, shut down irc channel and all support?
03:01.39file[laptop]did I say that?
03:01.46[TK]D-Fendercced : no clue...
03:01.56justinulol
03:01.58file[laptop]very few changes are done to DUNDi because it's very complete, it's a finished solution
03:02.10file[laptop]and companies do use it
03:02.33file[laptop]and Digium holds no control over who stays in the IRC channel, and who helps with it
03:02.48justinuwell said file
03:03.03cced<[TK]D-Fender>  but I can not find DMA. PCI use mmap
03:03.28file[laptop]justinu: I try
03:04.34tainted-file[laptop] so would DUNDi be a good asterisk load balancing solution?
03:05.07orlockHmm.. i'm trying to configure a ATA
03:05.22orlockand i'm not sure what values i should use for RTP base port, or RFC2833
03:05.22file[laptop]tainted-: that's not what DUNDi is at it's core
03:05.27orlockanybody have any suggestions?
03:05.57mog_workthats why i use it file
03:05.59mog_workthat and failover
03:06.09mog_workdundi + iax2 +regextern = happiness
03:06.17file[laptop]that reminds me
03:06.21Cherebrumorlock: 16384-32767
03:06.28techman97_andyWHOOOO HOOOOO!  It was just a port thing!  I expanded my range and I have full duplex AUDIO!
03:06.42orlockCherebrum: for rtp base port?
03:06.50Cherebrumyes
03:06.58ccedhi :mog_work
03:07.02file[laptop]mog_work: how was the party?
03:07.03techman97_andyI have a fully functioning and configured Asterisk system!  Thank you to ALL of you who helped me out on this since last Thursday!
03:07.09mog_workgrand
03:07.11techman97_andyI'm going to freaking bed.
03:07.12techman97_andynight
03:07.22orlockcool, what about the rfc2833?
03:07.25orlockit wants an integer
03:07.27cceddo zaptel read or write data from hardware using DMA OR mmap ?
03:07.30file[laptop]orlock: 101
03:07.39ccedmog_work.~
03:07.45orlockare you just pulling that out your arse? :)
03:07.59mog_worki dont think we do either cced
03:08.04file[laptop]orlock: no that's the dynamic payload that we and every other sane device likes to use for RFC2833
03:08.04mog_workbut you know where you can find out
03:08.09mog_workzaptel.c
03:08.48*** join/#asterisk Az_au (i=[+MAL4VO@216.127.73.119)
03:08.53orlockfile[laptop]: cool
03:09.23znoGi guess you'd need a nice big network of asterisk servers to make use of DUNDi
03:09.24*** join/#asterisk justinu|laptop (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net)
03:09.48mog_worki do it with just two znoG
03:09.53mog_workbut the more the merrier
03:10.00znoGto achieve what, exactly?
03:10.05mog_workfailover
03:10.08ccedmog_work.yes . zaptel provide interface.
03:10.10znoGi was just reading the DUNDi draft
03:10.10file[laptop]mog_work is just insane
03:10.19mog_workits just as good as an iax2 bridge
03:10.20znoGto understand the protocol a little better
03:10.27mog_workbut it allows for easier growth
03:10.54znoGand it makes sense to use if you have quite a few asterisk servers, and plan on adding more ..
03:11.10ccedmog_work.yes . tor2.c tor->mem32 = ioremap(tor->xilinx32_region, tor->xilinx32_len);
03:11.12mog_workit is more beneficial then
03:11.26mog_workdo not look at to drivers
03:11.45*** join/#asterisk Strom_M (n=strom@gateway.digium.com)
03:11.53mog_worklook at cards still sold today
03:12.13justinu|laptopwhy aren't tormenta cards sold anymore?
03:14.26mog_workwell digium stopped working on them
03:14.44justinu|laptopany idea why? it seemed like a cool architecture
03:14.46mog_workthey arent as good as the newer cards from digium/sangoma etc
03:14.54*** join/#asterisk iq|mobile (n=iq@71-38-73-211.omah.qwest.net)
03:14.58mog_workwell digium's hardware is zaptel design
03:15.00mog_workjust not a tor
03:15.07mog_worktors also where not as scalable
03:15.11ccedyes: most card use PCI slot.yes PCI
03:15.16justinu|laptopyou know any specifics?
03:15.25mog_worki could go into it
03:15.32mog_workbut i only know what was told to me
03:15.34justinu|laptopif you're up to it
03:15.42mog_worki know that driving fulll 4 e1s on a tor2 is very flakey
03:15.54justinu|laptoptoo many interupts/sec or something?
03:16.12ccedyes. vhdl problem.. :)
03:16.32justinu|laptopcced: are you a pot?
03:16.44justinu|laptops/pot/bot/
03:16.50justinu|laptopa bot on pot?
03:18.44*** join/#asterisk jeebusroxors (n=jeebusro@29palms-cuda1-68-170-36-65.losaca.adelphia.net)
03:19.53asterboydid someone say pot?
03:20.00asterboyhigh
03:20.14*** join/#asterisk cced2 (n=dev2003@222.33.36.205)
03:20.25cced2<cced> pot ? what mean?
03:21.27justinu|laptophaha
03:21.50justinu|laptopcced2: what is your native language?
03:22.07cced2Chinese
03:23.43cced2what mean?puzzle
03:25.04*** join/#asterisk Redimido (n=gorv@201.153.212.239)
03:25.11cced2mog_work : in libpri ,where define t203 =10s ?
03:25.47mog_workcced2, i dont know how to explain this politely
03:25.50mog_workim happy to help
03:25.54mog_workbut i am not google
03:25.56mog_workor the source
03:26.04mog_workboth of those you have
03:26.07mog_workuse them
03:26.13mog_workif you have real questions
03:26.15mog_workask me
03:28.19file[laptop]ha
03:28.21harryvvgoogle makes a google box
03:28.53harryvvfile, whats up?
03:29.07harryvvcced cantonese?
03:29.07file[laptop]sitting in bed contemplating sleep
03:29.16orlockman, we are trying to configure a netcomm ata
03:29.19asterboy[park] on the Polycom Phone seems to want to accept some digits and then you press [park] again but it does nothing.   What does work to park a call is to [Transfer] to a parkinglot and then put the call on hold for pickup at another phone.  Anyone know if the Polycom Park is suppoerted and if so what the configuration parameters are?
03:29.19orlockits not fun
03:30.52asterboyAnyone here used the Polycom [Park] feature button?
03:31.14*** join/#asterisk cced2 (n=dev2003@222.33.36.205)
03:31.33cced2mog_work: o thanks . I read codes :)
03:33.18asterboyAnyone use the [Services] button on a Polycom IP600?
03:33.27[TK]D-FenderI do
03:33.30asterboyLooks like a microbrowser
03:33.38[TK]D-Fenderthats exactly what it is.
03:34.33asterboyI thought it was only supported in the IP601 and 501 series.
03:34.43[TK]D-Fendernope, only on 60x
03:36.47asterboyhmmm...time to learn up on what that can do.
03:37.17asterboyhttp://www.voip-info.org/wiki/view/Polycom+Microbrowser
03:37.21[TK]D-Fenderasterboy : not that much.  no control over phone functions yet
03:37.46asterboy"There is very limited information about this feature in the Polycom documentation. "
03:38.21asterboyno doubt, seems to be tantalizing though
03:38.27*** part/#asterisk Redimido (n=gorv@201.153.212.239)
03:38.27[TK]D-FenderYou can use it to display live infor on "idle" and provide basic HTML (VERY) as interactive pages to do something else "useful"
03:38.50[TK]D-Fenderasterboy : the A2H handbook pages ont he WIKI have a lot more detail stangely...
03:38.56[TK]D-FenderA@H
03:39.24asterboyLooks like it would be good for a business card of the business.
03:39.33asterboyTo show address and stuff for temp workers.
03:39.35*** join/#asterisk hinckc (n=hinckc@ool-4354c627.dyn.optonline.net)
03:40.11asterboywow ... even inline images
03:41.19asterboyI might put my business card in there for when the client needs support.
03:42.21[TK]D-FenderI use it for mass-presence checking, company contact list and live queue stats
03:42.37asterboysweet
03:42.44asterboylive queue stat
03:43.10asterboymass-presence...is that like FOP?
03:44.11asterboy"Main Browser Home.
03:44.16asterboyWhere is that set?
03:44.50[TK]D-Fenderasterboy : something like that.  I get "ext  - name" for each phone in use
03:45.15[TK]D-Fenderasterboy typically in your provisioning files.
03:45.20asterboyThat kicks ass.
03:45.38[TK]D-FenderI do it in phonexxx.cfg so as to personalize it per phone
03:45.42asterboyLike <mac>-phone.cfg?
03:45.46asterboyah
03:46.02[TK]D-Fenderno, in the one referenced by <mac>.cfg
03:46.11asterboyok
03:46.37*** join/#asterisk hatamen (n=hatamen@222.183.27.190)
03:47.04asterboyinteresting...nothing in the standard .cfg file.
03:47.40[TK]D-Fenderasterboy : nope, I copied it from sip.cfg.  you can port most of the XML tree fromt he master to overrider whatever you need on a per-phone basis
03:47.54asterboyah
03:49.08asterboyso something like <main mb.main.home="http://domain.com/polycom.html"/>
03:49.17asterboyand then put in supported tags.
03:49.40[TK]D-Fenderyup, just a few basic ones needed.
03:50.30[TK]D-FenderI've done forms on it, minimal font control (avoid where possible), inline images (my queue stats idle page has out company logo in it and info on 2 queues & 2 VM boxes
03:50.47asterboynice
03:51.15asterboyI like the logo idea...can that be put on the front default display with the clock?
03:51.15[TK]D-FenderAnd all ring like Cisco's on "24" :D
03:51.33asterboythat would be creepy.
03:51.46[TK]D-Fenderasterboy : 2 ways on 601.  static image or idle XHTML on time interval.
03:52.22asterboymb.idleDisplay.home="(url)" and mb.idleDisplay.refresh="(seconds").
03:52.29[TK]D-Fenderyup
03:53.06[TK]D-Fendervery usable... thats how I've got "live queue" stats.  every 10 sec I call a PHP'd page that uses AMI to poll queues & VMboxes
03:53.10asterboyonly 4bpp bmp support though
03:53.31[TK]D-Fenderasterboy I believe 8 bit works and is scaled
03:53.39file[laptop]commit commit here, commit commit there
03:53.42asterboythat would be better
03:54.21file[laptop]eep
03:55.02[TK]D-FenderYou know with just a speakerphone, the IP301 would totally rock....
03:55.20[TK]D-FenderAnd my 501 is still on backorder... dammit
03:55.49k-manis there a tapi driver that can communicate with an ip phone
03:55.58asterboyI'm totally happy with my IP600s
03:56.20asterboyEverything works, save [Park] & IM
03:56.20[TK]D-Fenderasterboy : as am I with mine.... I jsut want one of each at home....
03:56.32tainted-anyone work with Vovida's load balancer proxy?
03:56.36asterboylol...ya I have the IP500 and IP300
03:56.49asterboyeBay items
03:57.10asterboyth 01s would be nice for the bigger memory though
03:57.20[TK]D-FenderI boght a 301 & 501 new, billed to my consultancy so hopefully I can write them off as "training tools"
03:57.38asterboytotally write offable.
03:58.16asterboyI want to switch from my computer consultancy to telecom consultancy.
03:58.28asterboyJust enjoy it so much more.
03:58.59asterboyThe telecom company is just starting to take off.
04:00.10[hC]I wanna know what its gonna take to get more than 7 line statuses working on the polycoms
04:00.23asterboyv1.4 when it comes out
04:00.26[hC]Why has nobody implemented the secondary BLF protocol that polycom doesnt have an issue with?
04:00.28asterboyor SIP-B
04:00.40[hC]SIP-B? I wasnt sure what it was called.
04:00.42file[laptop][hC]: implement it then.
04:00.52asterboySIP for Business
04:00.53[hC]Im not a coder or i would have done it already :S
04:00.58[hC]I'm a network engineer/integrator
04:01.13asterboyWho needs it when you can use the microbrowser anyway.
04:01.23xacheni can do perl and php ^_^
04:01.24[hC]My clients receptionists
04:01.26xachenjust not real good C
04:01.53asterboyI can program in anything...when getting paid.
04:02.19[hC]So, any eta on firmware 1.4? or maybe, has anyone attempted to implement SIP-B? I'm more than willing to bounty it, if its worth it
04:02.24asterboyotherwise...I just live at the bash prompt
04:02.26[TK]D-FenderI have XHTML MicroBroser stuff to compesate for "live" presence suppotr"
04:02.41[hC][TK]D-Fender: willing to share it? :)
04:03.05asterboyya, that's what I'm saying [TK]...the microbrowser takes care of that.
04:03.06[TK]D-Fender[hC] : Polycom is supposed to be effectively removing their artificial limi in the next SIP release
04:03.19[TK]D-Fender[hC] : when I'm at work tomorrow, sure.
04:03.32[hC][TK]D-Fender: yeah... any idea when they plan on releasing the new firmware? I mean, have they actually made a statement as to when?
04:03.58[TK]D-Fender[hC] : "When its done" ;p
04:04.53file[laptop]good answer
04:04.56[hC][TK]D-Fender: Figures :)
04:08.37*** join/#asterisk Jaxxan (n=jaxxan@202.70.125.124)
04:12.26*** join/#asterisk tessier_ (n=treed@ppp-71-140-230-121.dsl.sndg02.pacbell.net)
04:14.19[hC]Ive still not touched hte microbrowser at all
04:14.27asterboyJust received my box of Grandstream GXP-2000s
04:14.48tainted-good now box them back up and buy some real phones
04:14.59asterboylol
04:15.01tainted-;)
04:15.19asterboyI tried to sell the Polycoms but the client was cheapo
04:15.42asterboyDidn't want to loose the bid, so I caved in.
04:16.02asterboyNow I get to try out these phones and see if they live up to the garbage name they have.
04:16.16asterboyHopefully they are at least functional.
04:17.13asterboy[hC], I just touched the microbrowser for the first time...and ah...well the page is blank.
04:17.41asterboyback to the playing with the settings...not sure what I did wrong...left only the simple tags.
04:18.31orlockHmm, has anybody here used a netcomm ata?
04:18.48[TK]D-Fenderasterboy : you need to put in the basics, even if empty : HTML, HEAD, TITLE, BOD, and make sure everything you do has a closing tag
04:19.45asterboygot it...tried to get fancy and put a bgcolor field in the body tag...it don't like it.
04:20.32asterboyThanks Polycom you don't have to reboot phone after html changes.
04:20.51bkw_http://www.ngnsky.com/product_info.php?products_id=49
04:21.07justinu|laptopasterboy: you're gonna need a lot of weed if you're trying to work with gxp2000's
04:21.18asterboyLOL
04:21.29asterboyI was just going to twist one before starting.
04:22.01QwellWTF
04:22.05Qwellbkw_: That site is crap
04:22.12justinu|laptopthey're fine for occasional users
04:22.20asterboymicrobrowser does not like <br>
04:22.27justinu|laptopbut for business customers... i dunno
04:22.27asterboywants <br/>
04:22.30Qwell"The version of your browser is not supported.  Please use InternetExplorer version 4 and above!"
04:22.43Qwellsite looks...fine...in mozilla
04:22.58asterboyDillo will break it.
04:23.24Qwellasterboy: <br> isn't valid..
04:23.32Qwell<br/> or <br></br> are
04:23.45asterboyya, it sure lets you know.
04:24.10asterboyI'm thinking CSS is not supported either :P
04:25.35OliverXgood morning asterisk world (;
04:26.07xachenbkw_: thats way cheaper than digium :o
04:26.24[TK]D-Fenderasterboy : Like I said, EVERYTHING has to be closed.. that'd be <br />
04:27.11OliverXis their a interface/api to use mysql with asterisk?
04:27.30*** join/#asterisk drfoomod2 (i=DrFooMod@ool-43501d9f.dyn.optonline.net)
04:27.41drfoomod2has anyone seen a asterisk@home box hang at Checking for new hardware?
04:27.43asterboyyes, it sure complains if you get anyting wrong. Kinda like programming in Pascal
04:28.11drfoomod2this is the first boot after installing 2.7
04:28.25Qwelldrfoomod2: #asteriskathome
04:28.30drfoomod2tx :)
04:28.49justinu|laptopanyone know how this works?
04:28.52justinu|laptophttp://en.wikipedia.org/wiki/Unlicensed_Mobile_Access
04:29.17drfoomod2Qwell: empty chan
04:29.20[hC]f this im gonna go home
04:29.28Qwell[hC]: and test my patch? :D
04:29.33[hC]drfoomod2: they moved, #freeswitch
04:29.39asterboyhome sweet home
04:29.48[hC]Qwell: yep :)
04:29.52asterboy[hC] come on over and twist one with me.
04:29.54Qwellexcellent...
04:30.01QwellJust make sure to test an addon
04:30.09[hC]the 7914 is attached
04:30.09justinu|laptopasterboy: where you live?
04:30.11[hC]where is the patch again?
04:30.15OliverXis their an answer for me?
04:30.16asterboyAlberta bound
04:30.20Qwell6859
04:30.22justinu|laptopah
04:30.22asterboyland locked in Canada eh!
04:30.29[hC]eek that box is running ast 1.2.1
04:30.32[hC]will it compile against that?
04:30.35Qwellno, heh
04:30.37Qwelltrunk only
04:30.37justinu|laptopi live in the wasteland called los angeles
04:30.38asterboyI feast on whale blubber and live in the snow
04:30.42[hC]hmm
04:30.43[hC]okay
04:30.45asterboylost angles
04:30.46[hC]might take some time then
04:30.51[hC]that thing is a production box
04:30.52[hC]:)
04:30.53asterboybeen there at night
04:30.56Qwelleww
04:31.02asterboynot even the cops will help you out.
04:31.02[hC]i know
04:31.03[TK]D-FenderPascal = teh shiznit yo!
04:31.12Qwell[hC]: It'll crash a lot, so...not recommended in production :p
04:31.14xachenasterboy: you from Alberta? :O
04:31.19OliverXhm i must go to work. perhaps i became an answer in this evening(german time +1)
04:31.23[hC]I'll try it on my astlinux box
04:31.30[hC]i'll see if that thing will let me upgrade to trunk easily ;)
04:31.30asterboyyepper...I'm as red necked as an Albertan can get.
04:31.40drfoomod2[hC]: tx
04:31.42justinu|laptopcops are criminals w/ badges
04:31.45[hC]<- vancouver
04:31.58[TK]D-FenderOMG
04:32.05asterboyjustinu, you certainly have that close to the mark...not all, but most.
04:32.09[hC]WTFLOLBBQ?
04:32.17[hC]:)
04:32.21asterboyAlberta is a Police State
04:32.22[hC]sorry bkw
04:32.25bkw_haha
04:32.34[hC]I couldnt RESIST
04:32.44bkw_they moved to #freepbx
04:32.49xachenasterboy: me too :p
04:32.53[hC]freepbx, freeswitch... so close :P
04:33.03justinu|laptopwhy is it a police state?
04:33.10asterboyxachen, Alberta is smoking for business right now.
04:33.20xachenyup
04:33.46alephcomI don't quite see how that makes it a police state but maybe that's cause I'm way out in the country.
04:33.53[hC]Hmm.. I need a LAMP programmer to make a front end for my stuff, anyone know of anyone good in vancouver? or even remote as long as they're really good :)
04:34.13xachenhehe
04:34.14[hC]Since all the asterisk front end config suites are crap, or force you to do things their way
04:34.17xachenaleph is way out in the middle of nowhere
04:34.23asterboyjustinu, because the cops here micro manage everything you do.
04:34.46SplasPood[hC]: heh..
04:35.05alephcomasterboy: What have you done that they watch you? :-)
04:35.16asterboyjustinnu, I've had them come to my door for my dog being out front, drumming at 9pm, shovling snow too close to the sidewalk...etc etc etc.
04:35.37alephcomCalgary?  Edmonton?
04:35.50asterboyalephcom, they watch everyone like a hawk here.
04:36.16asterboyThey even arrested an RCMP officer because for gun possesion.
04:36.17*** join/#asterisk denon (i=denon@synapse.subneural.net)
04:36.17*** mode/#asterisk [+o denon] by ChanServ
04:37.02alephcomxachen is right that I'm in the middle of nowhere.  Let's see.  I saw an RCMP truck on my road 1.5 years ago. lol
04:37.08xachenhehe
04:37.11xachenabout a year here
04:37.22[hC]SplasPood: hey
04:37.30asterboyYa, I need to move out into the country.
04:37.36asterboyThat sounds peacful.
04:37.41[hC]SplasPood: id contract you if you want, but i need something done in like... under 2 months, and you have a full time job already :)
04:37.52asterboyDo you have HighSpeed Internet though?
04:38.11alephcomYeah, of course.
04:38.25asterboycan't be too far from city central
04:38.30alephcomWireless with decent speeds.  It's not a cheap as adsl.
04:38.37xachenim getting wycom :(
04:38.42alephcom:-(
04:38.46xachenshould be in by the end of the week
04:38.47asterboyYa, there is a lot of WISP going on in Alberta now.
04:39.22asterboyAlberta oil sands just got estimated at 1.5 Trillion Dollars.
04:39.44asterboyAlberta is about to become the World's #1 Oil producer.
04:39.50[hC]Im going to send them an email and ask for a pizza, since they can obviously afford it.
04:40.21*** join/#asterisk sjobeck (n=sjobeck@london.sjobeck.com)
04:40.30asterboyI was thinking of going up to Fort McMurray to sell phone systems to the Oil patch.
04:40.56xachenmake them newfie proof first....
04:40.57xachenbig numbers
04:41.01alephcomIt's rather sad that Klein is quitting.  Darren ducks
04:41.03xachenand a big green and red button :p
04:41.11xachenand you suck
04:41.23xachenKlein can go $#$! himself
04:41.47asterboylol
04:42.08drfoomod2can i use a cisco router (such as a 2621xm) with a t1 card to access FXS ports on an adtran channel bank?
04:42.31asterboyI laughed when Klein went down to the homeless shelter, drunk, and threw money at them and told them to back to work.
04:42.43xacheni'm going to sleep
04:43.06asterboyxachen, keep in touch on here.
04:43.45asterboyThe project head for LinuxFromScratch lives in Canmore, Alberta.
04:43.58xachenyeh
04:44.01xachenOpenBSD in Calgary
04:44.07xachenbut I'm out
04:44.09[hC]Did you guys know Erik Reid?
04:44.11asterboynight
04:44.30asterboyErik Reid, no...who is he?
04:44.36[TK]D-Fenderasterboy : yeah, but can his being that far away from "civilization" truely be called "life" ;)
04:44.48[hC]former open/netbsd guy who lives in alberta
04:44.53asterboylol...seriously, Canmore rocks.
04:44.55[hC]he died a couple years ago though
04:45.15asterboywhat was his function in the hive?
04:45.32[TK]D-Fenderok, I'm out... later all
04:45.40asterboynight TK
04:45.53asterboythanks for the info the microbrowser
04:48.21kimosabei have a sip acount i used to use it with a sipura device but now i have a tdm400p with ine fxs on my asterisk box does any one have a config example so that my fxs card will use that sip acount
04:48.26*** join/#asterisk [Mojo] (n=mojito@200-122-80-171.cab.prima.net.ar)
04:51.04asterboykimosabe, its much the same setup save that you can tell your FXS extension to Dial(SIP
04:51.18*** join/#asterisk xbmodder_lappy (i=nobody@unaffiliated/xbmodder)
04:51.23xbmodder_lappyI keep getting this error:
04:51.23xbmodder_lappy== Spawn extension (pushup, 19252097312, 3) exited non-zero on 'SIP/pushup-608e'
04:51.48kimosabeasterboy do you know where i can get an exampl epleazse
04:52.54asterboylooking.
04:53.12kimosabethanks man
04:53.16asterboyI just set this up for a client so I'll see if I can cut and paste.
04:55.21*** join/#asterisk justinu|laptop (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net)
04:55.29*** join/#asterisk florz (n=florz@2001:1a50:503c:0:0:0:0:1)
05:03.39*** join/#asterisk sternn (n=sternn@user-0c938ku.cable.mindspring.com)
05:11.05asterboykimosabe, here is your SIP to FXS example config: http://pastebin.ca/48975
05:11.46asterboyI setup G729, had to buy a license from Digium...so setup your codec as required.
05:14.31*** part/#asterisk FuriousGeorge (n=brian@ool-43536ea8.dyn.optonline.net)
05:18.08*** join/#asterisk kisu (n=daniel@cielkisu.tb.as8758.net)
05:18.14asterboykimosabe, Here is a better version with more contexts: http://pastebin.ca/48976
05:18.22*** part/#asterisk alephcom (n=alephcom@host75.net14.mcsnet.ca)
05:21.50asterboyNice! Check this out: http://www.amooma.de/asterisk/refcard/asterisk-refcard.pdf
05:23.17asterboyjbot, refcard is a nice printout cheet sheet for * found here: http://www.amooma.de/asterisk/refcard/asterisk-refcard.pdf
05:23.18jbot...but refcard is already something else...
05:23.23asterboy~refcard
05:23.25jbotextra, extra, read all about it, refcard is http://people.debian.org/~debacle/refcard/
05:24.05asterboyjbot, refcard is also a nice printout cheet sheet for * found here: http://www.amooma.de/asterisk/refcard/asterisk-refcard.pdf
05:24.07jbotokay, asterboy
05:24.11asterboy~refcard
05:24.13jbot[refcard] http://people.debian.org/~debacle/refcard/, or a nice printout cheet sheet for * found here: http://www.amooma.de/asterisk/refcard/asterisk-refcard.pdf
05:24.27asterboygood boy.
05:25.12Qwelljbot: no, refcard is http://people.debian.org/~debacle/refcard/, or a nice printout cheat sheet for * found here: http://www.amooma.de/asterisk/refcard/asterisk-refcard.pdf
05:25.13jbotQwell: okay
05:25.16Qwell:D
05:25.17*** join/#asterisk FuriousGeorge (n=brian@ool-43536ea8.dyn.optonline.net)
05:25.38asterboy:P
05:25.43FuriousGeorgehey all
05:26.13asterboyFurious, howdy ho.
05:27.33FuriousGeorgehow much does someone who installs a 10-20 extension pbx charge for basic service:  tech support 9-5, next day repair replacement, etc
05:28.18FuriousGeorgewell  assume unlimited phone support 9-5
05:28.40asterboyjbot, refcard is also a vi card here: http://tnerual.eriogerg.free.fr/vimqrc.pdf
05:28.41jbotokay, asterboy
05:28.47asterboy~refcard
05:28.48jboti guess refcard is http://people.debian.org/~debacle/refcard/, or a nice printout cheat sheet for * found here: http://www.amooma.de/asterisk/refcard/asterisk-refcard.pdf, or a vi card here: http://tnerual.eriogerg.free.fr/vimqrc.pdf
05:29.25asterboyvi rulez!
05:29.47FuriousGeorgeasterboy: can you explain something to me
05:30.26*** join/#asterisk ApEtc (i=apetc@ip70-162-216-7.ph.ph.cox.net)
05:30.38Supaplex;)
05:30.41asterboylol
05:31.08FuriousGeorgelol, im not trying to be flip, but from the little ive used vi, even when you got the keystrokes in muscle memory, why would you want to hit all those buttons to edit an asterisk conf?
05:31.16FuriousGeorgekeep in mind i only use nano :)
05:31.39FuriousGeorgenot emacs, i would ask the same to an emacs user
05:31.40Az_auwhen you know all the shortcuts it cuts down on keystrokes ;)
05:31.45asterboyI'm old school...been using Xenix since late 80's
05:31.56asterboyI really don't like GUI
05:32.00FuriousGeorge~xenix
05:32.02jboti heard xenix is NOT that bad.. not that it's that good. or a Unix derivate by M$ aborted to focus on NT.  MS sold it to SCO who made SCO Xenix which became SCO Unix
05:32.27asterboy~MAI
05:32.45Supaplexwell, vi/vim/etc doesn't require most of the keyboard to function :)
05:33.29FuriousGeorgeive been planning to learn some C and PHP so i guess i find out for myself
05:33.53Az_auya.. syntax hilighting is also a bonus
05:34.02asterboyman I must be old, no MAI
05:34.16Az_auand folds
05:34.19FuriousGeorgeasterboy: tell jbot what mai is
05:34.26FuriousGeorgeso that i'll know too :)
05:36.10Supaplexand you won't find emacs on an embeded system :P
05:36.28FuriousGeorgeare learning logical languages like spoken languages in that they suggest you learn one at a time so you dont confuse yourself?
05:36.31QwellSupaplex: But you'll find ed!
05:36.32asterboyjbot, MAI is Management Assistance Inc. which asterboy first cut his programming teeth on. More info here: http://web.archive.org/web/20050305205751/www.science.uva.nl/museum/basicfour_tbl.html
05:36.33jbotasterboy: okay
05:36.35mog_worki thought emacs was an embedded system Supaplex
05:37.54Supaplexreal sysadmins use ed
05:38.04Qwelled is the standard text editor!
05:38.05mog_workbump ed
05:38.12mog_workdump text into device buffer
05:38.17Supaplexcat!
05:38.19Az_auya.. cat
05:38.21mog_workyou should know evreything on box
05:38.26mog_workso no need to read it back out
05:38.30Qwellcat?  pfft, newbs
05:38.31Supaplexor echo :)
05:38.37Qwell< and >
05:38.48mog_workamen brother Qwell
05:38.53Qwellkorn has some nifty stuff for redirection
05:39.04Qwellksh
05:39.39*** join/#asterisk tessier_ (n=treed@ppp-71-140-230-121.dsl.sndg02.pacbell.net)
05:40.23justinuis xenix what made you start smoking pot in the first place?
05:40.29asterboyI really love this site: http://www.old-computers.com/
05:40.44FuriousGeorge~jbot
05:40.45jbotjbot is, like, only marginally useful at best,  He got a C- on his Turing Test
05:40.46asterboylol, no...my wife started me smoking pot.
05:40.53justinulol thats funny
05:40.54*** join/#asterisk _m_ (n=m@fbta199.fbta.uni-karlsruhe.de)
05:40.59justinubut understandasble
05:41.23FuriousGeorgeasterboy: still married to her
05:41.31FuriousGeorge?
05:41.41asterboyoh ya...the only women who could put up with a guy like me.
05:41.49*** join/#asterisk oej (n=oej@apollo.webway.se)
05:41.49FuriousGeorgelol
05:42.11justinui'm getting married next monday
05:42.25asterboyAlthough I'm not gay, I can see why some men go brokeback.
05:42.38asterboyjustinu, seriously?
05:42.40justinuyes
05:43.01*** join/#asterisk jeebusroxors (n=jeebusro@29palms-cuda1-68-170-36-65.losaca.adelphia.net)
05:43.03asterboydon't know if I should congradulate you or feel sorry for you.
05:43.06FuriousGeorgename the artist:  i was in love with a girl on marijuana.  she said "if i'm not stoned, i don't wanna." But she got so paranoid, her place i would avoid. (i was in love with a girl on marijuana
05:43.08justinugoing to NYC tomorow morning for my last trip as a single man
05:43.14asterboynever mind...misery enjoys company.
05:44.10asterboyTommy?
05:44.17FuriousGeorgeyeah
05:44.31FuriousGeorgeTom Petty "Girl on LSD" really funny song
05:44.39FuriousGeorgehaha you knew that.  stoner
05:44.43FuriousGeorge:)
05:44.51justinu~asterboy
05:44.52jbot[asterboy] a weed smoker
05:45.07FuriousGeorgeit could be worse
05:45.11FuriousGeorge~furiousgeorge
05:45.13jbotfrom memory, furiousgeorge is a knife-fighting monkey last seen with The Man with the Yellow Bat
05:45.14justinuyes, far worse
05:45.24justinuwhere is the man with the yellow bat, anways?
05:45.31asterboylol
05:45.38FuriousGeorgehe's about
05:46.00FuriousGeorgehe usually tries to sneak up on me so he changes his nick right before the attack
05:46.13*** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk)
05:46.33justinuwhen i hear man with yellow bat, i think of fooly cooly
05:46.43justinubut that was something like a hockey stick, i think
05:47.05asterboylol, I had one of these: http://www.old-computers.com/museum/computer.asp?st=1&c=1170
05:47.41FuriousGeorgewhy is it that i never get tired of seeing my name in jbot
05:47.54asterboyand one of these: http://www.old-computers.com/museum/computer.asp?c=477&st=1
05:48.16*** join/#asterisk Faithful (n=Faithful@202-6-145-116.ip.adam.com.au)
05:48.27asterboyand this: http://www.old-computers.com/museum/computer.asp?c=102&st=1
05:48.31FuriousGeorgeasterboy: how did you play doom III on that!
05:48.43asterboylol
05:49.18asterboyman I'm not even 40 and I feel old
05:50.13Qwellasterboy: That museum sucks...
05:50.20FuriousGeorgelol
05:50.22QwellDoesn't even have my zenith
05:51.09FuriousGeorgei had an intelivision but i think that one was already a little pop-culture
05:51.34asterboylol...no zenith
05:52.07asterboyI remember beggin for an intelivision
05:52.37FuriousGeorgeso how much is your average basic/silver/whatever service agreement gonna run monthly for 10 -20 extensions?
05:52.41*** join/#asterisk xbit` (n=xbit@frugalware.elte.hu)
05:53.03X-RobOooh, intellivisons were so cool. you could get a keyboard and it was just like a computer!
05:53.20FuriousGeorgeX-Rob: i just remember playing baseball
05:53.31FuriousGeorgei was 4 or so
05:54.28litageare there any devices (other than the TDM400P and similar PCI cards) that have 2 FXS and 2 FXO ports?
05:54.51FuriousGeorgelitage: does the sangoma card do 2X2?
05:54.56FuriousGeorgeor is it 4 per?
05:55.05asterboynow I have 3 netfinities and 3 LH 6000 HP NetServers pilled up doing nothing.
05:55.13*** join/#asterisk glm2k (n=glm@rrcs-24-199-11-46.west.biz.rr.com)
05:55.16asterboyuseless.
05:55.25*** join/#asterisk freat (n=ron@h-72-244-84-43.chcgilgm.covad.net)
05:55.39asterboylitage, not sure if the sangoma has that.
05:56.22asterboysagnoma
05:57.20MikeJ[Laptop]sangoma's analog is based on 2 port mods
05:57.28*** join/#asterisk Telamon (i=telamon@blk-222-22-126.eastlink.ca)
05:57.30MikeJ[Laptop]up to 24 ports
05:57.45litageFuriousGeorge, asterboy: if it does or doesn't, i'm looking for non-PCI card hardware
05:58.06X-Roblitage, usb?
05:58.12FuriousGeorgeT1?
05:58.25FuriousGeorgeyou can get 2 port ATAs from linksys
05:58.25TelamonIs there a variable which gives the login ID (ie, the SIP or IAX2 username)?
05:58.51FuriousGeorgeand some sip -> pots things  i think theyre called iaxy
05:59.12asterboymediatrix and quitum make some nice gates
05:59.27asterboys/qui/quin/
05:59.52SwKscrew mediatrix and quintums
05:59.53litageFuriousGeorge: none of linksys' products support 2 FXOs
05:59.55MikeJ[Laptop]I think for low qantity fxs, sipura is still the best cost/port isnt it?
06:00.00SwKget a Audiocodes
06:00.13MikeJ[Laptop]SwK, what's cost/port?
06:00.18SwKif you need more then 2FXS
06:00.51asterboyFor 2FXS the cheapest is to get an unlocked Linksys PAP2
06:01.03SwKthey arent cheap but they are worth it... 729, 723, T38 etc
06:01.08asterboybut they are a big pain in the ass
06:01.12*** join/#asterisk BugKham (n=lamer@ppp-58.8.4.140.revip2.asianet.co.th)
06:01.16litageasterboy: yes, but that doesn't have any FXO ports
06:01.20asterboytrue
06:01.30SwKlitage: you need mixed FXO/FXS device?
06:01.47litageSwK: yes, i need a device with 2 FXO and 2 FXS ports
06:02.03SwKlitage: only way to get that is TDM400
06:02.05asterboynot cheap: http://cgi.ebay.com/Quintum-Tenor-ASM200-2FXS-2FXO-VoIP-H323-SIP-Gateway_W0QQitemZ5750195904QQcategoryZ61839QQssPageNameZWD1VQQrdZ1QQcmdZViewItem
06:02.18SwKquintums are a POS
06:02.29*** join/#asterisk greendisease (n=jack@fedora/greendisease)
06:02.29*** join/#asterisk marv (n=ilovekim@12-219-145-181.client.mchsi.com)
06:02.38FuriousGeorgelitage: i guess it was just a similar color to a linksys pap2
06:02.47FuriousGeorgeanyway, i know ive seenem
06:02.48asterboyya, I'd much rather go with TDM
06:03.14SwKif you gave me a Quintum, I would ebay it and buy an Audiocodes heh
06:03.52asterboyhttp://cgi.ebay.com/Cisco-VOIP-VG200-Router-NM-2V-with-VIC-2FXO-2FXS_W0QQitemZ9708191990QQcategoryZ51204QQssPageNameZWDVWQQrdZ1QQcmdZViewItem
06:04.38asterboyAnyone use Micronet?
06:04.41asterboyhttp://cgi.ebay.com/2-Micronet-SP5014-VoIP-Routers-2FXO-2FXS-ports_W0QQitemZ9708613398QQcategoryZ61841QQssPageNameZWDVWQQrdZ1QQcmdZViewItem
06:05.01*** join/#asterisk austinnichols101 (n=austinni@dsl-10-169.cofs.net)
06:05.11asterboyThink they are made in New Zealand.
06:05.15asterboyiirc
06:05.35MikeJ[Laptop]what?  newzelanders?
06:06.11MikeJ[Laptop]oh oh SwK!!
06:06.55SwK?
06:07.19MikeJ[Laptop]wassup man
06:07.40SwKnadda
06:07.40MikeJ[Laptop]I want to publically announce btw to never ever use interland
06:07.57SwKhah
06:08.23austinnichols101interland sux0rs
06:08.36MikeJ[Laptop]indeed
06:08.40SwKI would also like to publically say Never use Nocster.com aka burst.net they are the suck
06:08.45MikeJ[Laptop]16hrs to properly escalate a 5 min fix
06:08.58SwKhah
06:09.05MikeJ[Laptop]the whole time fighting with me about it being my prob
06:09.14BugKhamHi there, what's the use of the 'username' parameter in sip.conf? in my client I only need to put the 'section title'
06:09.31MikeJ[Laptop]every time, I carefully explain (clearly over their head) how the problem was their issue in great technical detail
06:09.33X-RobBugKham, I think you're confusing Microsoft Word with a SIP phone.
06:09.41SwKtry 2 days down because they decided my server needed to have the OS reinstalled and then couldnt tell me why no one could connect to the server
06:09.52MikeJ[Laptop]they say they escalated it and it took 8+hrs to get tot hte top of the queue
06:09.53BugKhamfor both username and auth username
06:09.58asterboyThat sounds like an offshore India tech support scenario
06:10.07MikeJ[Laptop]in reality... no one bothered till I was a total jerk
06:10.17X-RobMikeJ[Laptop], and you wonder why I'
06:10.19*** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com)
06:10.22X-RobI've got my own hardware?
06:10.28MikeJ[Laptop]asterboy, there are plenty of offshores that do a good job
06:10.34MikeJ[Laptop]me too...
06:10.41asterboyhave yet to find one
06:10.44X-Robhttp://www.serverpronto.com
06:10.44MikeJ[Laptop]but this was a side deal for my dads office
06:10.46BugKhamX-Rob: huh, my sip client does care about my username parameter
06:10.49MikeJ[Laptop]I just took it over
06:10.57FuriousGeorgeim installing a phone system for a small business (17 users) and i dont wanna offer them a service agreement, anyone in here wanna do it?
06:11.02MikeJ[Laptop]so it basically runs itself till somthing like this blows up
06:11.10BugKhamX-Rob: the section title is all it needs
06:11.15SwKyou dont have it hosted where you work mikej?
06:11.18asterboyFurious, where?
06:11.25litagethe Audiocodes MP-20x datasheet (http://www.audiocodes.com/Objects/LTRM_30008_DS_MP-20x.pdf) says it connects "2 POTS lines or fax machines". how can an RJ-11 port be both an FXO and FXS port?
06:11.26FuriousGeorgecentral nj
06:11.35FuriousGeorgenorthern central kinda east :)
06:11.36SwKFuriousgeorge: remote support ok?
06:11.48FuriousGeorgeSwK sure till the PSU fires
06:11.51asterboyI'm out...I'd have to get a green card.
06:11.51FuriousGeorgefries**
06:11.58FuriousGeorgelol
06:12.10asterboyonce the smell pot on me at the airport...I'm done.
06:12.24FuriousGeorgeasterboy: or is it hash accross the pond?
06:12.32asterboylol
06:12.33austinnichols101asterboy: we'll host for pot
06:12.34QwellSo...
06:12.37X-Robwhy the smeg would you want to live in the US anyway?
06:12.38asterboyLOL
06:12.41asterboyhost for pot
06:13.00Qwellanybody happen to know how to connect a 20 pin 286 harddrive up to a newish computer? :D
06:13.01MikeJ[Laptop]SwK, no, first time I had to touch it was today, when I figured out how badly it's set up..
06:13.06austinnichols101try our weedserver plan
06:13.06SwKhah
06:13.15MikeJ[Laptop]small office with a mail server in house...
06:13.24asterboylike an ESDI drive?
06:13.31Qwellasterboy: no bloody clue
06:13.41brookshireqwell: minix?
06:13.46Qwellbrookshire: ...no clue
06:13.53Qwellprobably some form of DOS
06:13.53MikeJ[Laptop]their mail relays through a spam filter company, to interland, then the clients pop it to their pc's... then it gets saved on the server from the clients
06:14.07Qwell~1982
06:14.09asterboybet that thing makes a good bookend or doorstop
06:14.14MikeJ[Laptop]~1492
06:14.15QwellI can't date it, really
06:14.31Qwelllet's put it this way...
06:14.52X-RobQwell, 20 pin? That's MFM/RLL?
06:14.52Qwellthere are posts on google groups, dating back to 1990, about how people were successful in putting a 20mb hd into the machine
06:14.55asterboyI'd think you could get some sort of ISA Card to hook that up.
06:15.07QwellX-Rob: If I knew, I wouldn't be asking. :)
06:15.08asterboyRun Length Limited
06:15.19X-RobQwell, does it have edge connectors, or pins on the hdd?
06:15.21X-Rob(take a photo of it)
06:15.25Qwelledge connectors?
06:15.30austinnichols101qwell: do you have the old edge connector cable?
06:15.32Qwelllike old floppies has?
06:15.42X-RobQwell, yeah, the connector slides over the board
06:15.44Qwellit has two types...it's using the pines
06:15.45Qwellpins
06:15.46X-Robkinda like a PCI slot
06:15.52brookshireqwell: it would probably be easier to buy an old irix box.. and install it
06:15.52brookshirelol
06:16.00Qwellbrookshire: I want the data off the drive :p
06:16.01brookshireuse it as a fileserver
06:16.14oejbrookshire: You are up early :-)
06:16.18X-RobOoh oej
06:16.23brookshireoej: heh..
06:16.28QwellX-Rob: it does have that type though, yes
06:16.29oejX-Rob: Morning
06:16.52Qwellabout 15 "pins", then a break, and 5 more "pins", on a board extending out
06:16.55*** join/#asterisk [Airwolf] (n=airwolf@83.98.235.222)
06:17.24X-RobQwell, that's MFM or RLL
06:17.32X-Robyou'll need to ebay for that
06:17.38X-Robit'll be an ISA card
06:17.41QwellThey have pci cards for that?
06:17.46X-Robnope
06:17.46austinnichols101qwell: you're going to probably need the original card it was formatted with
06:17.49Qwelllame...I have the isa card in the old box
06:17.58Qwellbut...nowhere to put it
06:17.58asterboyQwell, here ya go: http://cgi.ebay.com/Everex-EV332-16Bit-ISA-Floppy-Hard-MFM-Drive-Controller_W0QQitemZ8792366847QQcategoryZ1247QQssPageNameZWDVWQQrdZ1QQcmdZViewItem
06:18.00Qwellit's HUGE
06:18.16asterboy9.99$
06:18.21asterboyPlace your bid!
06:18.28asterboyAND
06:18.31X-Robooh yeah, the formatting was tied to the controller card
06:18.33asterboyfrom NEW ZEALAND
06:18.34Qwellasterboy: I already have one :D
06:18.38asterboylol
06:18.47QwellThe one it was formatted with, I'd guess
06:18.51austinnichols101qwell: now all you need is an old 286
06:18.57X-RobQwell, so you have the card, and the drive, so what are you missing?
06:18.59Qwellaustinnichols101: I have an old 286 :P
06:19.02X-Roban ISA motherboard?
06:19.08QwellX-Rob: no, it's a full machine
06:19.25X-RobQwell, I'm puzzled what your problem is then
06:19.35asterboyany ISA should do it, no?
06:19.37QwellX-Rob: I don't think it powers on, heh
06:20.03X-RobQwell, ahha. Well, grab an old IBM 300GL on ebay, they've got ISA slots
06:20.09X-Robusually about $10
06:20.10asterboycatch
06:20.32QwellI need to get a NIC for it, or something...
06:20.38Qwellit's got like a 240 baud modem :D
06:20.40austinnichols101and some thin ethernet cable
06:20.56Qwellbigmouth...oh yeah
06:20.57asterboytransfer over 5.25" Disks
06:20.59X-RobQwell, the IBM's are Pii's and come with an onboard NIC.
06:21.12X-Robboot knoppix, it understands MFM and RLL
06:21.16*** join/#asterisk Tili (i=Tili@61.140.191.181)
06:21.20asterboyseriously?
06:21.22QwellX-Rob: My wife would probably kill me. :)
06:21.31X-RobQwell, for buying a $10 computer?
06:21.34Qwellthough, I will check it out
06:21.36QwellX-Rob: yes, heh
06:21.47X-Robput your foot down
06:21.51QwellQuestion is, will the ibm fit an extended length card?
06:21.57X-Robsay 'I want this computer! And I'm not getting any sex until you forgive me for getting it!'
06:22.10asterboyif knoppix can understand MFM & RLL, I'd be so impressed.
06:22.21Qwellasterboy: linux kernel can...why wouldn't knoppix?
06:22.22X-Robasterboy, uh, yes, it does.
06:22.35asterboynever knew that.
06:22.44asterboyshould pay attention to the menuconfig options.
06:22.55QwellSo, knoppix would pick up the isa controller card, and then the drive?
06:22.59Qwells/would/should/
06:23.09X-RobQwell, yup
06:23.12Qwellneat
06:23.15asterboyvery
06:23.18X-Robthe isa card always lives at c800
06:23.42X-Robso it looks there, goes 'ooh, I have a controller card' and hooks into it. It's insanely slow, as it uses like int13 to access it
06:23.51Qwellheh
06:23.52austinnichols101debug: g=c800
06:24.06QwellI knew I asked in the right place...
06:24.07X-Robaustinnichols101, hah. how to format an old hdd 8)
06:24.12Qwellbunch of old telephony guys in here :p
06:24.20austinnichols101don't forget to pick the right interleave
06:24.25*** join/#asterisk apardo (n=apardo@87.218.45.206)
06:24.27X-RobI'm not old, OR telephony
06:24.27X-Robok
06:24.28X-RobI'm old.
06:24.34X-Robbut I'm old and linux-y 8)
06:24.43asterboy*nixy
06:24.47X-Robtrue
06:24.51X-RobSunOS 4.1 ho!
06:25.03asterboyewww
06:25.09FuriousGeorgei used beos once
06:25.13asterboylol
06:25.14austinnichols101out of curiosity, what age is old now?
06:25.26*** join/#asterisk kmilitzer (n=km@office-gw.westend.com)
06:25.27asterboygrey pubic hair
06:25.32MGSsanchoew
06:25.33Qwellaustinnichols101: If you can remember terms such as "MFM and RLL"...
06:25.35FuriousGeorgeyuck
06:25.37austinnichols101yeah
06:25.44kmilitzerMorning everyone ...
06:25.47QwellThere's a good chance you're old. ;)
06:25.53austinnichols10140 next week
06:25.59X-RobI can't remember what MFM stands for, but RLL was 'Run Length Limited'
06:26.07Qwell40's not so old
06:26.15austinnichols101I'm going to have a 'halftime' party
06:26.28freathello
06:26.29Qwell~mfm
06:26.30jbotA graphical frontend for mtools. URL: http://www.core-coutainville.org/mfm/
06:26.41asterboyModified Frequency modulation
06:26.49X-Robthat's it
06:26.51X-Robgo wikipedia
06:26.53SwKST506 y0
06:27.07SwKRun Length Limited ++
06:27.23freatanyone recommend a solution: asterisk server is remote. got Polycom phones on site (obviously) that I would like to be able to connect to an ATA or the like for 911
06:27.41SwKfreat: spa-3000
06:27.59freatSwK: it will handle the phone's SIP connections?
06:28.10X-Robfreat, SPA3000 is your only option, really.
06:28.23freatX-Rob: thanks I will read up on it
06:28.24X-Robif sipura didn't have one, someone else would have made one by now
06:28.25*** join/#asterisk BenderNZ (i=bender@nz1.recoil.net.nz)
06:28.42SwKit wont play sip proxy but if it cant contact the sip proxy (asterisk) it connects the FXS port on it to the FXO port for a life line
06:28.54*** join/#asterisk tessier_ (n=treed@ppp-71-140-230-121.dsl.sndg02.pacbell.net)
06:29.05asterboyst506...I remember those...great drive
06:29.24freatSwK: hmm... so they would have to pick up the phone that's connected to it? Sure would be nice to config the polycoms to hit it somehow
06:29.46BenderNZhi - I've got asterisk setup and incoming calls are coming in via SIP fine and my phones are ringing, however outgoing calls ring once then in the asterisk log I see Failed to authenticate on INVITE
06:30.02freatPolycoms allow for a separate server config for emergency dialing
06:30.10BenderNZI can't see why they aren't authenticated though because they're registering and they ring when I ring them
06:30.18QwellSo, does 1982 sound about right for that hardware?
06:30.53wasimi got a vic-20 in 1982
06:31.08wasimmy first real honest to goodness computer
06:31.54asterboyvic was good, but c64 was a god send
06:32.02X-RobHmmmm. 1982 sounds early, but possible
06:32.04*** join/#asterisk yxa (n=diablo@58.185.90.101)
06:32.13X-RobI'd think more like 83-84
06:32.21X-Robwhat's the model number of the HDD?
06:32.33QwellX-Rob: no clue...haven't bothered taking it out
06:32.46Qwellbut it's from a Zenith z-152
06:33.19X-Roboooh
06:33.23X-RobI used to sell them!
06:33.26Qwellhah
06:33.38X-RobI got better!
06:33.43FuriousGeorgeanyone wanna guesstimate *'s market share?
06:33.53X-RobFuriousGeorge, - 100%
06:33.55QwellSo, yeah...a 20mb was a major upgrade, as was 640k ram
06:34.00X-Robno, um 672%
06:34.02austinnichols10182 = vic 20
06:34.10FuriousGeorgeX-Rob: stop liein'
06:34.24FuriousGeorgedo you think it reaches 1%
06:34.25FuriousGeorge?
06:34.28austinnichols10184 = mac 128
06:34.31X-RobFuriousGeorge, of what?
06:34.50FuriousGeorgeof total pbx installed in the universe
06:34.51asterboywhen was the apple iie?
06:34.56X-RobFuriousGeorge, shit no.
06:35.23X-Robthat's bazillions of 20 year old pabx's doing their stuff
06:35.25FuriousGeorgetrue
06:35.32austinnichols101we made a shitload back then selling ast research six-pack plus cards
06:35.38asterboy84 =apple iie
06:35.41FuriousGeorgei got one sitting on top of an * server :)
06:35.47austinnichols101and tallgrass technologies disk/tape
06:35.56X-RobI'd say it would be 5-10% of SIP PABX's though.
06:36.16FuriousGeorgewhats the A for
06:36.20FuriousGeorge~pabx
06:36.21jboti guess pabx is Private Automatic Branch eXchange
06:36.21asterboyI just upgraded a Nortel Vantage 12
06:36.26X-RobAutomatic
06:36.36asterboyAnyone working on Vantage?
06:37.19*** join/#asterisk r0d3nt (n=RatMan@foster.stonedcoder.org)
06:37.26X-Genho hum
06:38.07asterboyRemember the Trash 80s?
06:38.14SuperLagVic20++
06:38.22asterboyhttp://www.old-computers.com/museum/computer.asp?c=409&st=1
06:38.25FuriousGeorgeme had an 8088 tandy
06:38.34asterboytandies were dandy
06:38.41FuriousGeorgei think it was the tandy 1000 ir sinmething
06:38.55FuriousGeorgehell yeah, it had a 3.5" floppy
06:39.23asterboyI had an Adam, but took it back along with my garbage Commodore 128
06:39.57FuriousGeorgei remember looking at that floppy in 1986 and thinking "i wonder what we'll be using in 20 years.   then 20 years later i bought a mb and the sw raid driver for winxp is on a floppy
06:40.23asterboythat's funny
06:40.44austinnichols101get your hayes 1200 longcard and fire up your bbs: http://software.bbsdocumentary.com/
06:40.53Qwell1200, pfft
06:41.20SupaplexI still have 4 8port 9600 bps multiport serial cards. =-)
06:41.25asterboyThere must be some die hard BBSs running still.
06:41.27Supaplexisa even
06:41.34X-RobFuriousGeorge, pfft. The _REAL_ TRS80's were the old Z80's. Astoundingly good machines.
06:41.48FuriousGeorgecome to think of it, if windows xp was able to pull a driver off a pendrive for install, there would be absolutely no reason to use'em anymore (finally)
06:41.50*** join/#asterisk mogorman (n=mogorman@68.62.237.103)
06:41.52asterboycoleco!
06:41.56FuriousGeorgeX-Rob: i was too young for those
06:42.13Supaplexgameboy uses z80
06:42.46FuriousGeorgeat the time the only thing i knew about BASIC was that it was that annoying screen that came up when the game cartridge wasnt loaded right in my commodore 64
06:43.04Supaplex10 goto 10
06:43.06Supaplexrun
06:43.13FuriousGeorgeand i couldnt play jungle hunt
06:43.14asterboyI sure got to know basic in a hurry in order to hack games.
06:43.43asterboyhex editor for the 1541
06:43.43FuriousGeorgeon what the old commodore?
06:44.02asterboyyep
06:44.29FuriousGeorgei knew some kid in spain, lived in the middle of nowhere, had some old comuter that ran on data cassettes.  as in audio cassettes with data on them
06:44.42asterboyso that was say 20 year ago...can you imagine in the next 20?
06:45.00austinnichols101I had the cassette drive for my vic 20
06:45.01FuriousGeorgethe floppy's should hold 1.44 megs by then
06:45.29asterboyThe Timex Sinclair (Z81) had a cassette
06:45.32wasima nice white cassette tape recorder
06:45.40asterboylol, nice white
06:45.46*** join/#asterisk tessier_ (n=treed@ppp-71-140-230-121.dsl.sndg02.pacbell.net)
06:45.48wasimnot the silly beige of the c64
06:45.59asterboythat would turn orange if left in sunlight
06:46.14asterboydig dug
06:46.22asterboyM.U.L.E.
06:46.25FuriousGeorgedig-dug was great
06:46.38asterboyIndiana Jones
06:46.40QwellX-Rob: If you can remember the z152...how about the "centaur II"?
06:46.42asterboyBruce Lee
06:46.56X-RobQwell, nope, that doesn't ring a bell.
06:46.57FuriousGeorgei had indiana jones too but i dont remember playing it
06:47.06X-Robisn't centaur a VIA cpu?
06:47.07asterboyit had a snake pit
06:47.16austinnichols101isn't that the one where you would swing with the whip?
06:47.19Qwelldunno
06:47.21asterboyMoon Lander
06:47.28asterboyerr...Moon Rover
06:47.58asterboyQbert
06:49.34Supaplexok, the $brain is drained. time for bed(); nite.
06:49.42asterboyhear ya there.
06:50.01FuriousGeorgei used to hate when jumping on a block 1 extra time turned it the wrong color
06:50.09asterboygreat to reminice.
06:50.35FuriousGeorgealthough i do still play cupert on the nintendo emulator
06:50.38*** join/#asterisk r0d3nt|m (n=RatMan@foster.stonedcoder.org)
06:50.55FuriousGeorgeits legal cuz i had the game and the system, OK
06:51.01asterboyseriously, after remembering all that, I can't believe where I am in computers and technology today.
06:51.06FuriousGeorgeas for the other 700 games that came in the 10 meg file
06:51.14FuriousGeorgei deleted those :)
06:51.16asterboyit really does seem like a dream
06:51.28asterboyya I had a ton of games.
06:51.37asterboydisk upon disk
06:51.52asterboysome good porn ones too
06:51.58FuriousGeorgeasterboy: you can get them all again. they're in a torrent somewhere
06:52.03FuriousGeorgei bet Qwell is hosting it
06:52.23asterboyI know, there are some amazing sites doing java emulation of the C64 chip
06:52.36Qwell10mb?  please, my zenith can't even hold that much :p
06:52.37FuriousGeorgedont get me started please
06:52.52FuriousGeorgelol
06:53.05asterboyNow look at battlefield 2 and the next gen of games.
06:53.18asterboyonline gaming is a culture all on its own.
06:53.30tainted-that's deep
06:53.53asterboythat it is.
06:54.07FuriousGeorgebrb
06:54.11asterboyI'll fade to black on that note.
06:54.32asterboyBuilding an * box in the morrow
06:54.58tainted-men amongst men
06:54.58asterboythanks for the step back into computer history.
06:55.04asterboyyes
06:55.08tainted-:D
06:55.25asterboynext ime lets go back to Charles Babbage...the supposed father of computers.
06:56.20asterboywell..you could say the abbacus started the road.
06:56.43asterboynight guys.
06:56.47tainted-night
07:03.27*** join/#asterisk Creperum (n=ilya@mail.tex.kiev.ua)
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07:20.46*** join/#asterisk tetsuzan (n=raizen@200.180.124.12)
07:22.27tetsuzananyone knows if zaptel already works with freebsd 6?
07:22.42zoadont think so
07:22.53tetsuzantime ago, i had test
07:23.03tetsuzanand, it fails
07:24.41*** join/#asterisk subdolus (n=subby@subby.afraid.org)
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07:28.16*** part/#asterisk mogorman (n=mogorman@68.62.237.103)
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07:35.03*** join/#asterisk SHad|Work (n=kvirc@popust.net)
07:35.14SHad|Workhello
07:35.40SHad|WorkI'm a bit puzzled with how asterisk and sip phones use the codec settings
07:36.14*** join/#asterisk ReD-MaN (i=redman@dhcp-0-2-b3-9a-4a-5b.cpe.quickclic.net)
07:36.55SHad|Workis it normal for the phones to use just the first codec from the allow list? asterisk then always tries to transcode the calls instead of requesting the same codec on both phones
07:41.19*** join/#asterisk plasko (n=plasko@triana.kmpanilla.com)
07:41.25tetsuzanyou have to add the codecs as a priority list
07:41.33plaskowow its busier in here than I expected.
07:41.42tetsuzanif your phone supports the first codec, the codec will be used.
07:42.01kaldemarSHad|Work: the allow's are in order of preference.
07:42.33kaldemarSHad|Work: e.g. if your phone supports the first codec, it will use it.
07:43.38plaskohas anyone used RAGI?
07:44.34kaldemarSHad|Work: do you have canreinvite=no in your sip.conf?
07:45.59*** join/#asterisk [hC] (i=turnerd@donkey.voxter.ca)
07:46.53*** join/#asterisk Shaun2222 (n=ndci@ip68-5-63-223.oc.oc.cox.net)
07:47.05Shaun2222are their any advantages to using sccp?
07:47.12*** join/#asterisk psk (n=psk@golia.caltanet.it)
07:52.03Shaun2222ok..
07:52.47Shaun2222anyway to make the phone show parked calls, i've used a few pbx's where it shows calls parked on the phone, so far the asterisk config just tells you where it was parked and then you have to remember it i guess...
07:57.10SHad|Workkaldemar: no, I have canreinvite=yes
07:57.42*** join/#asterisk andrebarbosa (n=andrebar@62.48.215.190)
07:57.46SHad|WorkI thought asterisk tried to negotiate a native call between the phones
07:57.56Zhadnostit depends on the config.
07:58.07Zhadnostdepends on the nat= and the canreinvite=
07:58.09SHad|Workwhat part of the config?
07:58.19SHad|Worknat=yes, canreinvite=yes
07:58.31SHad|Workthat's what I've got
07:58.31kaldemari think it's supposed to if you have canreinvite=yes and the phones are on the same lan.
07:58.47Zhadnostthen (from vague memory) asterisk assumed the phones are behinfd a NAT and can't directly talk to each other.
07:59.07Zhadnost(I'm not sure, but it's all on voip-info).
07:59.11SHad|Workbut g729 works in passhtrough
07:59.17SHad|WorkI don't have the g729 licence
07:59.29Zhadnostyou don't need to if it's on passthough
07:59.35Zhadnosteven if the call is being routed through asterisk
07:59.50x86morning :)
08:00.01Zhadnostthe SVN branch supports T.38 in passthrough but that doesn't mean asterisk has any idea aregarding the traffic.
08:00.10SHad|Workso is there any possibility of knowing if the calls are routed through asterisk?
08:00.45ZhadnostI guess it'd appear in sip show channels
08:01.03SHad|Workso the nat setting might be the culprit
08:01.07Zhadnostalso (from memory) asterisk wont pass over the call if you have other items configured, like interactive calling or recording).
08:01.45*** join/#asterisk Telamon (i=telamon@blk-222-22-126.eastlink.ca)
08:01.50Shaun2222is ztdummy needed with 2.6?
08:02.15Telamonshaun222: If you don't have a TDM or TE card, then yes, I believe so.
08:02.34Zhadnostif you want to use Meetme rooms, MoH etc.
08:02.44Shaun2222k
08:02.45*** join/#asterisk sulex (n=sulex@pdpc/supporter/active/sulex)
08:03.00Telamonztdummy is used to provide a timing source.  If you have a real Digium PRI card, it uses the timing from that instead.
08:03.20SHad|Workso in regard to scallability if I use MoH the server network infrastructure should be able to handle all the calls bandwidth
08:03.25SHad|Worknever thought about that
08:03.27SHad|Workthank you
08:03.45SHad|WorkI'll try the nat setting as soon as get the server up again
08:04.12TelamonAnyone know why my Asterisk would be crashing when running certain macros?  Is there a limit to the number of macros you can call?  IE, macro a calls macro b two or three times.
08:05.50*** join/#asterisk cced (n=dev2003@222.33.36.205)
08:06.09Zhadnostweird thing happened on Sunday, I put a TDM400 card in a machine, configured it and ran ztcfg, ran asterisk and everything was fine. A couple of hours later asterisk had stopped running and wouldn't start (card hadn't been configured) zttool confirmed that the card hadn't been configured and then after runnign ztcfg again everything was fine.
08:06.16ZhadnostDid I miss a step in the setup?
08:07.32*** join/#asterisk duckz (n=duckz@193.192.47.26)
08:09.20TelamonZhadnost: Did you reboot or reload any modules in between the first ztcfg run and the crash?
08:09.48Telamonreload = remove or add, not necessarily just remove and add back in
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08:10.10ZhadnostI don't remember doing it, I may have done an asterisk reload or similar.
08:10.40ccedAriel_ :hi
08:10.52ZhadnostIt hasn't happened since
08:11.24TelamonHmm, if you unload the module (which I don't think you can do with Asterisk running, as the device would be in use) or possibly if you load another ZT module it might cause problems.  Or it might have just been gremlins. :)
08:11.46*** join/#asterisk sercz (n=serz@i3ED6F0DD.versanet.de)
08:16.47DoktorGregI remember MFM and RLL though i dont remember what they mean
08:17.27ZhadnostI'm betting on cosmic rays
08:18.26DoktorGreggremlins are real!
08:18.52DoktorGregyou can have zero stress on a cable
08:19.03DoktorGregbut if you dont have it battened down, it will get pulled out
08:19.13DoktorGregwhen there is no possible way for it to get pulled out
08:19.29DoktorGregI have experienced this effect a lot
08:19.41Zhadnostthat sounds like an office to me
08:19.49DoktorGregnow everything gets a zip tie
08:20.09DoktorGregzip tie's are gremlin bane
08:20.56DoktorGregholy shite, boot camp has made me insane
08:21.11DoktorGregi just read mac vs pc blogs for 4 hours
08:21.17*** join/#asterisk sulex (n=sulex@pdpc/supporter/active/sulex)
08:21.41Zhadnosteww
08:22.05DoktorGregmy first computer after my atary 800 was a mac 512
08:22.34DoktorGregive been bent ever since then
08:22.51TelamonMFM and RLL were old disk drive standards, before ATA and SCSI came around, if memory serves me correctly.
08:23.06*** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25)
08:23.15Zhadnostthe eww wasn't in reference to macs, more the reading articles comparing them for 4 hours.
08:23.37DoktorGreglol, like i said Boot camp made me insane
08:23.42Zhadnostdid it go along the lines of, my dicks bigger than yours, yeah but mine has a go faster stripe on it? (etc. etc.).
08:24.03wasim11:26 < asterboy> Modified Frequency modulation
08:24.08wasimand Run Length Limited
08:24.09DoktorGregum, again, I want a mac
08:24.16DoktorGregi really want one
08:24.21Zhadnosta mini?
08:24.21DoktorGregbut logically i cant justify it
08:24.32DoktorGregno i want a core duo
08:24.35DoktorGregimac
08:24.38DoktorGreg2 ghz
08:25.01DoktorGregbut i keep specing computers twice as fast for same mony
08:25.14DoktorGregor...
08:25.36DoktorGrega couple of computers, same as fast...
08:26.09DoktorGregand i cant do it for osx
08:26.26DoktorGregbecause with... 15 years of legacy this direction....
08:26.54DoktorGreg...
08:27.45DoktorGregmostly my observation is that, like me
08:27.54DoktorGregmac users are generally insane
08:28.04DoktorGregwell the ones posting to blogs anyhow
08:28.19*** part/#asterisk SyrusMPL (n=pascal@tahiti.mpl.rullier.net)
08:28.30DoktorGregThey make these wild eyed assertions like...
08:28.34DoktorGregwindows in unstable
08:28.39DoktorGreg???
08:28.51DoktorGregwindows hasnt been unstable for like a decade....
08:29.05DoktorGregwell whenever it was that i started using nt...
08:29.41DoktorGregbut even 98 wasnt as bad as they make it out to be
08:29.46*** join/#asterisk Syrus_ (n=pascal@tahiti.mpl.rullier.net)
08:29.49Zhadnostit could be
08:30.18TelamonMac users have "drunk the cool-aid", so to speak.  They don't just think their Macs are better, they *BELIEVE* their Macs are better.  Personally, I use Linux and it works fine for me.
08:30.21ZhadnostDid anyone else notice that there was no fax server in 98 (and there was in 95).
08:30.55ZhadnostOh, and it alxso isn't in any subsequent version.
08:30.57DoktorGregI was using slackware for server things when 95 came out
08:30.57*** join/#asterisk Syrus_ (n=pascal@tahiti.mpl.rullier.net)
08:31.21Zhadnostnow I mostly use debian.
08:31.24*** join/#asterisk Money5ack (i=moneysac@wer.will.spontanficken.de)
08:31.39DoktorGregIIRC windows 95 had no TCP/IP stack installed by defaut
08:31.41TelamonSlackware.  Now there was a real man's OS.  Here are some tar files, unpack them and have fun... :)
08:31.58DoktorGregslack is still one of the largest distros
08:31.59ZhadnostThere was nothing Unixy at the time that fitted into the windows environment as well as the w95 Fax server though.
08:32.31ZhadnostHad IP/X by default, but that was usually changed during setup
08:32.35*** join/#asterisk cced (n=dev2003@222.33.36.205)
08:32.43cced<PROTECTED>
08:32.59TelamonDoktorGreg: Yep, it's BSD with drivers.  We used to be 100% Slackware servers at the office, but the upgrades were killing me, so I'm migrating to Gentoo.  Still a lot of old Slack boxes kicking around our server room though.
08:34.36DoktorGregI pretty much cut my teeth on slack, so now when i use just about anything else
08:35.07DoktorGregi head straight for the cli, /etc and vi
08:35.23DoktorGregIve tried vi on windows
08:35.36DoktorGregit just doesnt work the same anyhow
08:35.45DoktorGregi keep meaning to work more with vim
08:36.43DoktorGregagain, boot camp has driven me over edge
08:36.50DoktorGreglol
08:36.56DoktorGregbut dang it
08:37.03DoktorGregthey cost too darn much
08:37.09Shaun2222where does the DIALTEMPLATE go, does it just go in the SIPDefault or SIP<MAC>.xml?
08:37.48*** join/#asterisk dlynes (n=dlynes@S010600c09f9a0fc4.vc.shawcable.net)
08:38.43DoktorGregI cant figure it out
08:38.59DoktorGregis it having that mac 512 20 years ago???
08:39.15DoktorGregthat machine that barely worked???
08:39.26DoktorGregwith 400kb floppy disks?
08:39.44DoktorGregthat i had to swap to get the os to load?
08:40.29DoktorGregand i had to copy them every couple of weeks because the floppies that the os was on would wear out???
08:40.55*** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com)
08:41.31DoktorGregwhy on earth do i have happy memories about that???
08:41.55DoktorGregand that little 9" screen
08:42.19DoktorGregthen one day
08:42.19Zhadnostdoes insanity run in the family?
08:42.29DoktorGreghmmmm
08:42.37DoktorGregfamily is all small biz people
08:42.42DoktorGregdoes that count?
08:42.57Zhadnostpretty much
08:43.16DoktorGregwell we would actually be considered mid sized now
08:43.27Zhadnostcool
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08:44.20DoktorGregasterisk has been a lot of fun
08:44.26DoktorGreghahaha check out this email i got today
08:44.38DoktorGregoh what that copy paste server?
08:45.11Telamonpastebin.ca
08:45.23*** join/#asterisk RoyK (n=roy@80.239.107.70)
08:45.37*** join/#asterisk RoyK (n=roy@80.239.107.70)
08:46.02DoktorGreghttp://pastebin.ca/48992
08:46.11DoktorGreganyhow i was trying to order a 1000 dollar thing
08:46.31DoktorGregand i couldnt get the order to go through their insane security checking thing
08:46.45DoktorGregbecause it is a business location
08:46.53DoktorGregand we do all mail to a po box
08:47.00DoktorGregand everyhing is shipped to a building
08:47.16ZhadnostOn the third day of speaking to sales reps, one finally agreed to fax a quote/proforma.
08:47.47ZhadnostWhen I called him back 5 hours later, he said. 'What? You didn't really expect to get the Fax today did you?'
08:48.00DoktorGregfrom voip supply?
08:48.23ZhadnostIn the end we couldn't buy the kit because the fundholding manager was going to Canada the next day and by the time  he'd come back it was a new financial year. (Which I had explained to the salesman, but did no good).
08:48.55Zhadnostno-one seems to like you spending lots of money with them nowadays, it's weird.
08:49.06DoktorGregwell anyhow
08:49.21DoktorGregi didnt pay the ship same day premium from the digium site
08:49.24ZhadnostWhat was the security problem?
08:49.43DoktorGregthey wanted me to type in po box 239 for both shipping and billing
08:50.02Zhadnostweird
08:50.09Zhadnostthe card is registered to a PO BOX?
08:50.20DoktorGregyah, company po box
08:50.33DoktorGregindustrial district no mail address...
08:51.42Zhadnostweird
08:52.18DoktorGreganyhow
08:52.37DoktorGregtheir site said i had to call and register extra addresses to the credit card
08:52.45DoktorGregso i called controller
08:52.48Zhadnostunderstandable
08:53.04DoktorGreghe was like, huh?
08:53.41DoktorGregwe have shipping address
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08:53.58DoktorGregusps doesnt deliver mail there
08:54.15DoktorGregi dont know particulars
08:54.26DoktorGregits never been a problem before
08:54.28Zhadnoststrange
08:54.50Zhadnostmind you, the postal service here will moreorless deliver to anywhere
08:55.03ZhadnostIf you give a drain as a postal address they'd deliver to it.
08:55.23Ahrimaneshm here they hardly deliver to actual adresses
08:55.45ZhadnostIt will be like that here in a few years when the postal service is finally privatised :-(
08:56.02DoktorGregwell the best deal in town for business is fedex
08:56.16DoktorGregfedex lets us buy shipping at wholesale rates
08:56.31DoktorGregwe then sell shipping to our customers at the standard fedex retail rates
08:56.54mtryfossasterisk seems to dealock when originating a call into a queue (auto callback). is this a known issue ?
08:57.02*** join/#asterisk slav_jb (n=k@pirus.securax.be)
08:57.15DoktorGregWe make like a 100k a year on that plan
08:57.26ZhadnostOver here rates seem to depend on the exact size and weight of a package as to the cheapest shipping, it's almost a science to work out.
08:57.32Zhadnostcool.
08:57.34ccedwho is in CHINA?
08:58.24DoktorGregoh someone in china
08:58.43*** join/#asterisk psk (n=psk@golia.caltanet.it)
08:58.48DoktorGregcan you do, or know someone who can, do home decor manufacturing?
08:59.16Shaun2222anybody know what the name of a normal dialtone is with these bellcore-* tones?
08:59.40DoktorGregwhoops, got caught selling out my fellow americans again:P
08:59.57FrogzooI'm wondering if China allows IRC?
09:00.29DoktorGregthey get Google lite in china
09:00.36ZhadnostI must say, I don't think I''ve ever spoken to someone from China on IRC.
09:01.04DoktorGregIve spoken to people who claim to be from china in Second life
09:01.09*** join/#asterisk kisu (n=daniel@2001:618:400:0:0:0:da26:a0d2)
09:01.25DoktorGregbut its the internet so.....
09:01.50DoktorGregand its second life
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09:02.07cced<PROTECTED>
09:02.08Zhadnostwith all the free proxies out there, it's hard to work out how they can block traffic.
09:02.13DoktorGregwhich seems fundamentally based in the idea of being more interesting than you actually are
09:02.17DoktorGregthe game second life
09:02.40Zhadnostcced> you appear to be
09:03.03ccedi want to local language help. yes. who else? I want code dev help
09:04.39ccedZhannost: where a u?
09:04.52ZhadnostUK
09:05.02*** join/#asterisk vlrk (n=vlrk@59.93.77.120)
09:05.06tzafrirhe can tell you, but he'll have to kill you
09:05.22vlrkdoes any body have idea on sipura auto provisioning
09:05.31ccedZhannost: are you familiar with libpri?
09:05.52Zhadnosttzafrir> They gigve you software for it if you buy >200 units.
09:06.00Zhadnostcced> Fraid not.
09:06.17tzafrircced, why do you ask?
09:06.50ccedtzafrir: I want to write some document about  libpri
09:07.20ccedabout SIP signalling <-> isdn pri signalling exchange .
09:07.22tzafrircced, first-off, try #asterisk-dev
09:07.36ccedZhannost: thanks ~
09:07.48ccedirc channel?
09:07.56tzafrircced, also, it is generally more useful to ask a specific question
09:08.08wasimcced: http://www.packetizer.com/rfc/rfc.cgi?num=3398
09:08.51ccedthanks
09:12.13*** join/#asterisk MacDome (n=eseidel@c-71-198-177-144.hsd1.ca.comcast.net)
09:15.00dlynesanyone know what the diff is between a digium x100p and a digium x101p?
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09:15.42ZhadnostFrom memory, one's a motorola modem and the other one's an Intel.
09:15.56ZhadnostThe x100p was suppossed to be a superior product.
09:16.02dlynesah
09:16.09dlyneswhy is the 101p a higher version number then?
09:16.19Zhadnostbecause it was released later.
09:16.25Telamondlynes: It was made later, with cheaper parts.
09:16.27Zhadnostand as a replacement.
09:16.35dlynesthat's lame...bring out a cheaper product later?
09:16.42Zhadnostall companies do it.
09:16.44dlynesyou'd think they'd improve it
09:16.49dlynesnot cheapen it
09:17.06Zhadnostan SPA2000 is a cost reduced cisco 186, and a PAP2 is a cost reduced SPA2000
09:17.27dlynesbut the spa2000 is a better product than the cisco 186, isn't it?
09:17.36dlynesand the pap2 is a better spa2000 too, right?
09:17.44Zhadnostthe software is, but it is still a cheaper build.
09:17.53dlynesah
09:18.11ZhadnostI don't think the PAP2 was any better than the 2000.
09:18.27dlynesI heard the pap2 supported faxing slightly better than the 2000
09:18.39Zhadnostnto afaik
09:18.44dlynesah
09:18.52dlyneswhat about the 2100 then?
09:18.57Zhadnostthe hardware is identical. (infact later SPA 2000's had the same board).
09:19.17Zhadnostfunnily enough the ATA 186 could recognise fax traffic.
09:19.36dlynesdunno...my stupid sipura 2000 can't upgrade higher than 2.0.13g
09:19.39Zhadnostdoesn't
09:20.01dlynesI heard the ata186 used more bandwidth than the spa2000, though
09:20.15dlynesi don't know how it's possible though, if it's using hte same codecs
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09:20.27ZhadnostI don't remember seing 723/729 on it.
09:20.40Ahrimanesif only grandstream ata's would have working callerid on B&O telephones..
09:20.58Zhadnosteither way, the PAP 2, doesn't handle compressed codecs in more than one channel at a time.
09:21.15dlynesneither does the spa2000
09:21.30dlynesatcom can handle two g729 streams?
09:21.37Zhadnost4 channels, all of which can simulateously use G.729
09:21.42dlynesah
09:21.43dlynescool
09:21.49dlyneshow much are they?
09:22.08ZhadnostOnly problem is, the firmware is buggy (if it gets a NOTIFYU command from asterisk, it falls over).
09:22.13dlynesoh
09:22.20Zhadnostquite expensive
09:22.36Zhadnost(basically means you don't configure a mailbox in sip.conf).
09:22.39dlynesso, a digium tdm400 cheaper?
09:22.53dlyness/cheaper/is cheaper
09:23.46Zhadnost$119.99
09:23.55Zhadnostproba bly down to $100 if bought from the factory
09:24.05dlynes' $100/port?
09:24.11dlynes' or $100 for all four?
09:24.14Zhadnostno, it's an ATA, that's all 4 ports.
09:24.21Zhadnost4 X FXS ports
09:24.24dlynesI thought you said it was expensive?
09:24.28dlynesThat's dirt cheap
09:24.36Zhadnostthat is when you're talking about grandstreams
09:25.01dlynesIt's about $100 for a sipura unit that only has two ports
09:25.11dlynesWell...$110
09:25.16dlynes$CDN
09:25.21Zhadnostif you buy from the factory you will probably need to order 20 at a time, and the factory is in Shenzhen
09:25.37dlynesah...that's where I've seen the name before
09:25.44AhrimanesZhadnost: do atcom have an english product page for that ata?
09:26.00dlynesThey're always sending me catalogues from China and Taiwan
09:26.24ZhadnostAhrimanes> http://www.atcom.cn/En_products_AG468.html
09:26.38AhrimanesZhadnost: thanks
09:26.58ZhadnostIt's not based on the PA1688 (like the rest of their products) It's a Myson Century CS3220
09:27.12*** join/#asterisk sercz (n=serz@i3ED6F067.versanet.de)
09:28.01vlrkAny idea on "sipura spa 841 auto provisioning" ?
09:28.03dlyneshttp://www.atcom.com.cn/ is the Chinese site; http://www.atcom.cn/ is the English site
09:28.18vopiahhh I have  thats model
09:28.23vopiAG468
09:28.35vopibut it is sleeping
09:28.41dlynesPA1688...isn't that based on something from Advantage Century Telecom?
09:29.01ZhadnostAll I can tell you regarding the T.38 support is that if you plug a fax machine into it, it will detect the fax signal and request T.38. (Can't really test any further than that).
09:29.05AhrimanesZhadnost: hm.. the 4 ports.. can they be configured for 1 number or only as seperate numbers?
09:29.30dlynesAhrimanes: That would all depend on your asterisk configuration
09:29.44ZhadnostAhrimanes> you get to comfigure 2 servers, and set usernames and passwords for each port (second server as a backup server).
09:30.41ZhadnostSo you can do it by configuring asterisk to let all 4 ports use the same account or set up accounts that are equivalent in asterisk, there is nothing inherent in the ATA.
09:31.08Zhadnostafaik you can't route between the ports with a dialplan like you can with sipura stuff
09:31.11*** join/#asterisk julesvd (n=julesvd@pdpc/supporter/student/julesvd)
09:31.11AhrimanesZhadnost: ok thanks.. hm i need a 3+ ata that just has the same account for all ports.. damnit
09:31.27julesvdhi all !
09:31.45Ahrimanesoh well, i should get to the office.. ttyl
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09:35.27backbluehi, anyone with x100p in europe, working with CLIP ( callerid ) ?
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09:45.27zoai never got the callerid to work on those crappy x100ps
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09:49.48backblueexacly my problem too.
09:54.14kaldemarwould anyone know where the "starting context" of a call is defined, and whether it can be changed from the dialplan?
09:54.31kaldemarby starting context i mean the context where for example the hangup extension is searched in when either the caller or the callee hangs up.
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10:05.00Mystiqkaldemar: for an incoming call ? check zapata.conf, iax.conf or sip.conf
10:06.26kaldemarMystiq: no no, they're all incoming in asterisk's sense.
10:07.23kaldemari know all the context=blablah rows in conf files, but when you change the current context in the dialplan with app Goto, it changes the context that is seen as the start context for the call.
10:07.44kaldemarand it doesn't find the hangup context.
10:08.52kaldemari've found an ugly solution for that already, but... it's ugly.
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10:13.08Mystiqyou need to have a hangup extension in every context ofcourse
10:14.10RoyKMystiq: i beleive he means if starting off in [a], goto(b,s,1) then [a]'s hangup extension is run when the call is hung up
10:15.11backbluezoa: which country are you in?
10:15.28dlynesRoyK: but it's currently looking for a hangup extension in [b], not [a]
10:15.41dlynesRoyK: because he entered the [b] context which is included in the [a] context
10:15.57dlynesRoyK: and then issued a dial command from the [b] context
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10:16.22dlynesright, kaldemar?
10:16.51kaldemarnot exactly, otherwise, but b isn't included in a.
10:16.58kaldemari'm using a goto to b from a.
10:17.10dlynesYeah, so it switches contexts
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10:17.23UnderMinelo
10:17.23kaldemari'm trying to avoid putting the hangup context in every context.
10:17.25dlynesit's included in a, just not using the include statement
10:17.36kaldemardlynes: yes.
10:17.38MystiqRoyK: ehm, if you goto to b, then the hangup extension in b should be run
10:17.40dlynesso it's not actually part of hte a context, but you get my point
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10:18.20dlyneskaldemar: this is where objects would come in handy
10:18.34dlynesmake every context inherit the hangupable context :)
10:18.40kaldemarMystiq: it is, but as i said i don't want to have hangup extension in every context because i have a hierarchical structure and it would be so pretty if i only could have it in one context. :)
10:18.45RoyKkaldemar: and then which hangup extension is run?
10:19.22Mystiqkaldemar: no other way.. you would need a hangup extension in every context, could be a Macro, but you need a hangup extension i'm afraid
10:19.22kaldemarRoyK: exactly. at the moment one hangup extension is enough for every call class, but in the future...
10:19.49kaldemarthat's why i'd like to see and option for that in the dial app.
10:20.12dlyneskaldemar: You could always write it, and then submit it to digium for inclusion in the next version of asterisk
10:20.17kaldemarthat you could trigger a macro when the call gets hung up, no matter if it's the caller or the callee that hangs up.
10:20.21Mystiqnot related to dial i think, would be pbx.c
10:20.29syleanyone have the patch for ibm's pthread error that happens with asterisk 1.2.x for opensouce g729?
10:20.41Vahramsyle, yep
10:20.51sylecan you please send to me
10:20.58Vahramshure
10:21.01sylety
10:21.03kaldemarMystiq: it is in pbx.c, but it would be a nice feature in app_dial
10:21.19kaldemardlynes: i could write it if i could write c.
10:21.47kaldemarmaybe i'll just have to quit whining and learn me some. ;)
10:29.21grem_linHi there, could someone possibly give me some help/advice on my dialplan http://pastebin.ca/49001 . My problems are that when you enter an invalid extension, it seems to be going to _X. again - I would use 's' however when I dial the inbound number I get no reply and it simply "times out". Also, is there a way to differenciate from no reply and an invalid 2xx extension, which are defined in my [Outgoing] context?
10:32.42backbluecallerid with x100p? anyone?
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10:35.27wmandramorning all
10:35.57*** join/#asterisk cced (n=dev2003@222.33.36.205)
10:37.44*** join/#asterisk kmilitzer (n=km@office-gw.westend.com)
10:39.16grem_linDoes anyone have any idea why the 's' extension might not be working as one would expect it to? Any help would be greatly appreciated...
10:41.39sternngrem_lin: I'm not positive, but you might check your capitalization for the "i" section Playback vs. playback, etc.
10:43.13grem_linSurely I'd see that on the console though? I beleive that it's not actually reaching the "i" section at all, and the _X. is picking up on anything at all, which is why I'm thinking I need to try and get the "s" extension to work to take the initial call
10:46.10*** join/#asterisk cj-rm (n=cjrm@81-86-30-78.dsl.pipex.com)
10:46.17cj-rmHey people...
10:46.57cj-rmFor some reason I'm getting problems with Asterisk detecting answered calls when making an outgoing call on an FXO line via a TDM400.
10:46.59backblueUnderMine: easy cake. :P
10:47.02cj-rmAnyone any ideas???
10:49.04*** join/#asterisk ManxPower (n=ewieling@stirprop-s4-0-0-21.ndcr2.datasync.net)
10:50.40andrebarbosaanyone can help me with Huawei softswitch integration? (SIP)
10:50.47cj-rmhmmm.... Ok, I'm in the UK (on BT) and I've just set callprogress=no and it works, but asterisk doesn't wait for the call to be answered... Any ideas on how to fix this, callprogress=yes halts execution of the dial plan after the Dial() but doesn't detect the call being answered...
10:50.52cj-rmis there a UK setting for this??
10:51.11andrebarbosahttp://forums.digium.com/viewtopic.php?t=5751&highlight=huawei
10:51.32andrebarbosaif anyone have similar problems let me now please :)
10:51.39andrebarbosahad*
10:54.26cj-rmDoes anyone know if there are any unit testing frameworks for asterisk??
10:57.40*** join/#asterisk Gunnar (n=gunnar@101.Red-80-24-171.staticIP.rima-tde.net)
11:01.45ManxPowercj-rm, define unit testing
11:02.10*** join/#asterisk syle2 (n=blah@unaffiliated/syle)
11:02.32UnderMinebackblue: not when the C2 card didn;'t want to talk
11:02.46ManxPowerThe docs for WaitExten suck
11:04.07*** join/#asterisk shiznatix (n=shiznati@213-35-236-128-dsl.end.estpak.ee)
11:04.36shiznatixcan anyone give me a list of all the characters that can be used in the dialplan that asterisk will recognize?
11:05.01*** join/#asterisk rkr245 (n=ravi@office.callsat-telecom.com)
11:05.22rkr245hello
11:05.25rkr245and hi
11:07.48backblueshiznatix: read the rfc, it depends on the protocol
11:09.32shiznatixbackblue, what is the rfc
11:11.18backblueshiznatix: do you work in this area, or just for fun?
11:13.08Mystiqanyone installed an asterisk (using zap) in ukraine ? apparently you need some kind of certification for that ?
11:13.25shiznatixbackblue, just for fun really
11:15.08ManxPowershaun222, All printable ASCII characters
11:15.16ManxPower..er..
11:15.23ManxPowershiznatix,  All printable ASCII characters
11:15.42shiznatixManxPower, thanks
11:15.57ManxPowerIf you give more details about specifically WHAT you are asking about.  are you asking about extensions, pattern matches, Dial options?
11:18.02rkr245hi can any body tell the process of adding friends in sip.conf and extensions .conf with ip address
11:19.09shiznatixManxPower, patterns for the number that was dialed like: _95XXXXXXX. then I also need for the extension do dial like: {$EXTEN:1}.
11:19.37shiznatixManxPower, basically what chars will be used by asterisk in those 2 situations so I can make sure nothing else is put in
11:19.40ManxPowershiznatix, You mean dial=> ${EXTEN:},1,Dial(....
11:19.48ManxPower..er
11:19.55ManxPowershiznatix, You mean exten => ${EXTEN:},1,Dial(....
11:20.10shiznatixManxPower, yes
11:20.25ManxPowershiznatix, I doubt you can do that.
11:21.18ManxPowerrkr245, not any different from adding them without an IP address, except you use host=ip.ad.dr.ess instead of host=dynamic
11:21.24ManxPower~thebook
11:21.26jbotmethinks thebook is Asterisk: The Future of Telephony, released under a Creative Commons license and available at http://www.asteriskdocs.org << Read the book online!
11:21.29ManxPoweralso see the Asterisk book
11:22.37shiznatixManxPower, no my dialplan is like this: exten => _95XXXXXXX,1,Dial(Zap/5/{$EXTEN:1})
11:22.51ManxPowershiznatix, that is fine.
11:22.56ManxPowerthere's NOTHING special about htat
11:23.00shiznatixManxPower, but I am using a php script to be able to change the _95XXXXXX and the {$EXTEN:1}
11:23.02rkr245o.k
11:23.07rkr245thanx
11:23.19*** join/#asterisk Shoragan (n=shoragan@d072.apm.etc.tu-bs.de)
11:23.20shiznatixManxPower, I want to make sure that the person does not input any characters that asterisk won't use
11:23.31ManxPowershiznatix, Any time you change extensions.conf you have to issue a reload
11:23.38shiznatixManxPower, I know this
11:24.03cj-rmManxPower: Being able to simulate certain interactions with asterisk on particular channels and check whether the dialplan/configs behave as expected.  If they don't then I'd like to know that the test failed.
11:24.03key2!seen kram
11:24.05ManxPowershiznatix, to be safe only allow a-z, A-Z
11:24.29ManxPowercj-rm, I know of no such regression testing at this time, but you should ask on #asterisk-dev
11:24.49*** join/#asterisk brif8 (n=Techno@lazyjtrainingcenter.com)
11:24.52shiznatixManxPower, but what about the { and $ and stuff that are used
11:25.10shiznatixManxPower, and all the numbers like 9 and 5
11:25.10ManxPowerso allow those too.
11:25.39shiznatixManxPower, but what else is there that can be used is what i am looking to get. what is the full list of stuff that can be used
11:25.50ManxPower, is a valid character in extensions.conf but if you use it in the wrong place bad things happen
11:26.27brif8What causes  "ss_thread: CallerID feed failed: Success"    and     "ss_thread: CallerID returned with error on channel 'Zap/4-1'"  ?  When I call in on a PSTN line I get this and callerid is blank. using a std phone I see the caller ID information ?
11:26.33ManxPowerSo is @ but you don't want to allow people to enter exten => _123@bob.com,1,Dial(Zap/5/whitehouse)
11:27.10ManxPowerbrif8, the rxgain is too high or too low
11:27.29brif8ManxPower: rxgain = 0.0   txgain = 0.0
11:27.42brif8what would you suggest ?
11:27.54ManxPowerbrif8, I suggest you try different values for rxgain
11:28.08ManxPowerincrease by 2 at a time or decrease by 2 at a time
11:28.23brif8ok I'll first take it up and then try down. thanks
11:28.32cj-rmDoes anyone know how to terminate an internal Local/extension which is running?  soft hangup Local doesn't seem to work
11:29.18ManxPowershiznatix, a better way is to not allow them to enter ${EXTEN:1}, but ask them how many digits to strip and build the ${EXTEN:1} for them
11:29.52ManxPowercj-rm, it will terminate when it falls off the end of the dialplan or when the physical channel is hungup
11:30.25grem_linTo be able to use the System() application is there a module that I need to load?
11:30.31ManxPowergreendisease, no
11:30.42ManxPowergrem_lin, not that I know of.  Why?
11:30.43shiznatixManxPower, I thought about that but what if they want all calls that meet a certain criteria to goto one number
11:31.05grem_linI'm getting the message  No application 'System' for extension in the console
11:32.10ManxPower[root@pbx-1 bin]# asterisk -rx "show modules" | grep -a system
11:32.10ManxPowerapp_system.so                  Generic System() application             0
11:32.40ManxPowerunless you are doing something really stupid with /etc/asterisk/modules.conf app_system.so will autoloafd
11:33.33grem_linI'm probably doing something really stupid, I'm specifying the modules I want to load rather than using autoload
11:33.44ManxPowerthat's stupid.
11:33.55ManxPowerspecify the modules to NOT load.
11:34.04*** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl)
11:34.25puzzledmorning all
11:34.38ManxPowergrem_lin, http://pastebin.ca/index.php
11:34.41ManxPowerhello puzzled
11:35.08*** join/#asterisk UlbabraB (n=caplaz@host241-43.pool8172.interbusiness.it)
11:35.25cj-rmManxPower: It's in an infinite loop, and it's a Local extension so there is no physical channel to hangup
11:35.31grem_linThanks for all of your help, ManxPower - I'll do that :)
11:35.41ManxPowercj-rm, no way other than to stop the PBX that I know of
11:36.02ManxPoweryou have to be VERY careful about loops
11:36.14puzzledhey ManxPower. how's life?
11:36.14ManxPowerthat's why I normally put in a loop counter.
11:36.24ManxPowerpuzzled, Customers are morons
11:36.29vgsterdoes i option not work?  i am including 2 contexts which have an i option in but when i use the call context i am using it is using gthe i options from the first context it comes across with an i option
11:36.44*** join/#asterisk sysdebug (n=sysdebug@200.250.222.8)
11:36.54ManxPowervgster, which of the 5 or 6 i options are you referring to?
11:37.02vgsterinvalid
11:37.13shiznatixCan anyone help me with my zapata.conf configuration. Im having some strange problems with it
11:37.16ManxPoweryou mean the "i" EXTENSION
11:37.20vgsteryes
11:38.09ManxPowervgster, the special extensions like i always have a higher precidence in the local context than in include => contexts.
11:38.22ManxPowerIn fact, I didn't even think they worked in an included context
11:38.25*** join/#asterisk zotz (n=zotz@24.231.32.85)
11:38.46vgsterhmmm
11:39.47vgsteri was hoping that wasnt the case and I could special a invalid option handler per context
11:39.54*** join/#asterisk fulgas (n=fulgas@207.226.175.10)
11:40.23vgsterany alternatives?
11:40.59*** join/#asterisk cced3 (n=dev2003@222.33.36.205)
11:41.38backbluedamm x100p does not sends any callerid.
11:42.25ManxPowerbackblue, SEND?
11:42.41ManxPowerNo analog FXO card will send Caller*ID
11:42.53backbluereceive
11:42.54backbluesorry
11:43.00backblue:x
11:43.04ManxPowerWhat country?
11:43.11backblueportugal
11:43.31ManxPowerThe X100P only supports USA FSK Caller*ID
11:43.54ManxPowerone of the many reasons Digium has not sold that card in several years
11:44.34backbluebut there are patchs for germany and uk
11:44.44backblueor options for zapata.conf
11:45.31ManxPowerbackblue, Are you sure those are for the X100P?
11:45.49backblueManxPower: yes.
11:45.53ManxPowerand does your country use the same Caller*ID method at DE and UK?
11:46.37backbluei dont know, but that i think depends on the providers equipment, and not the country! or i'm wrong?
11:49.19*** join/#asterisk Seyr (n=Seyr_@cpe-67-10-139-141.houston.res.rr.com)
11:49.43ManxPowerbackblue, most providers in a country use the same protocols
11:50.07SeyrWhen your on a call and press "#1" to do a transfer (from features.conf), where would the digit timeout be set? for the time between # and 1?
11:51.02backblueManxPower: i hope so! :D
11:51.25ManxPowerSeyr, in features.conf
11:51.52backbluemaybe this clone card, does not suport callerid.
11:51.55Seyrdoh!
11:51.58Seyrthanks ManxPower
11:53.23sternnexit
11:53.32*** join/#asterisk apardo (n=apardo@62-15-114-125.inversas.jazztel.es)
11:54.41cced3who is in CHINA?
11:58.15Hmmhesaystrick question?
11:58.19Hmmhesayschinese people, duh
11:58.49kaldemarwho _are_ would be better.
11:58.54Zhadnostcced3> Did you look on #asterisk-dev ?
11:59.47*** join/#asterisk Grizzy (i=Generic@ppp-71-133-231-94.dsl.pltn13.pacbell.net)
12:01.26Hmmhesaystoday is going to suck
12:02.36Zhadnostit already does
12:03.02Hmmhesaysi have a chick upstairs I have to take through 50F weather on a motorcycle
12:03.06Hmmhesaysshe will not be impressed by that
12:03.38X-RobDepends what sort of bike it is
12:03.49Hmmhesaysa bicycle
12:03.50X-Robif it's a harley, she's not gunna be impressed anyway.
12:03.58X-Robpushbikes are cool
12:04.03Hmmhesaysi joke
12:04.12Hmmhesaysit's an m50, we went out riding last night and ended back here
12:04.29Hmmhesaysshe might be under the impression that I actually have a car right now
12:05.32Ahrimaneshow did she get that impression?
12:05.39Hmmhesayslogical conclusion
12:05.43Ahrimanesah
12:05.48Hmmhesaysnice bike, decent home
12:05.55Ahrimanesshe's a girl tho.. so.. logical?!
12:05.59Hmmhesayslol true
12:06.19X-RobHmmhesays, m50? A scooter?
12:06.26Hmmhesayssuzuki
12:06.34X-RobHeh
12:07.05Hmmhesayshttp://motorcyclecruiser.com/newsandupdates/blvd-m50-oar-xl.jpg
12:07.38X-Robhttp://www.montesa.hpg.ig.com.br/ima/t315R2001.jpg
12:07.47X-RobOooh
12:07.50X-Robthat's so not a scooter.
12:07.59X-Robit's a harley wannabe
12:08.10Hmmhesaysthat bike don't wanna be a harley
12:08.20Hmmhesaysfuel injection, computer controlled timing, shaft drive
12:08.41*** join/#asterisk [TK]D-Fender (n=joe@toronto-HSE-ppp4122655.sympatico.ca)
12:08.52X-Robhttp://www.xaraxone.com/FeaturedArt/ps/assets/images/Honda_Firestorm.jpg
12:08.53Hmmhesaysthat bike surpasses harley in every way except namesake
12:09.01X-Rob200kph in 3rd gear.
12:09.03X-RobI win
12:09.03X-Rob8)
12:09.16Hmmhesaysa 1 cylinder bike ?
12:09.18Hmmhesaysyeahright
12:09.18[TK]D-FenderNice donor-cycle.....
12:09.25X-RobV-Twin, 1000cc
12:09.46X-Rob[TK]D-Fender, oddly enough, it's the cruisers that have FAR more accidents than the sportsbikes
12:09.58Hmmhesaysnot around here
12:09.59X-Robpossibly because the oldies tend to own the cruisers in .au
12:11.00X-Robours is slightly more custom than that
12:11.14X-Robhttp://www.gladstonewireless.net/users/taz/mybike.html
12:17.00shiznatixhey can anyone help me with my zapata.conf file? I am having some troubles getting Zap to work properly
12:17.16Zhadnostit's vaguely possible, go ahead
12:17.27Zhadnosthave you read the voip-info zapata.conf page?
12:17.47shiznatixZhadnost, yes I read it several times
12:18.15Zhadnostdid you configure zaptel.conf first and run ztcfg ?
12:18.59shiznatixZhadnost, here is my problem. When I try to dial out from my SIP phone through Zap it does not ring the destination phone but instead just says that the call was answered and it gives me this weird high pitched beeping
12:19.07*** join/#asterisk NirS (n=NirS@62.90.49.98)
12:19.11NirShello all
12:19.25Zhadnostweird, anything thin the log files?
12:20.55shiznatixZhadnost, not really. here is my stuff: http://pastebin.com/653321
12:22.11NirSanyone has an idea why L flag and S flag using dial application won't work ?
12:22.54NirSanybody home ?
12:24.09ManxPower1200 ft of 24 strand multimode fiber.  must.  resist.  bidding.
12:25.05Zhadnostsounds like mega-money
12:26.01*** join/#asterisk fuzzbawl (i=fuzzbawl@69.44.167.80)
12:26.03shiznatixZhadnost, any ideas on my Zap situation?
12:26.36Zhadnostbut then I'm not that experienced with Zap
12:27.32iCEBrkr*** glibc detected *** double free or corruption (fasttop): 0xf6411f58 ***
12:27.37iCEBrkrGrrrr
12:28.08ManxPowershiznatix, analog ports are considered answered as soon as dialing is finished.
12:28.45shiznatixManxPower, how do I stop that? how do I make it wait so it will actually ring my the outside line and wait for a answer?
12:29.10ManxPowershiznatix, you cannot have it wait for an answer on analog ports.  I don't know why your calls are failing.
12:29.22ManxPoweryou should hear ringing from the telco
12:30.25shiznatixManxPower, there is no ringing anywhere
12:30.49*** join/#asterisk skyhawker (n=skyhawke@a62-216-22-13.adsl.cistron.nl)
12:30.49ManxPowershiznatix, THAT is the problem.
12:31.00ManxPowerYou did not find anything helpful when you searched the mailinglist archives?
12:31.22skyhawkeri have a mitel telelphone and was wondering if i could see which external number somebody calls on the phone .. we have two companies here and need to answer accordingly
12:31.24shiznatixManxPower, I don't even know where to begin searching. I don't know what I am looking for
12:31.52Zhadnostsilly question, but is the correct zone loaded in zaptel.conf
12:31.53skyhawkershiznatix: i have that problem toio
12:32.11ManxPowertry X100P and your country name with site:lists.digium.com at google
12:32.40Ahrimanesskyhawker: many possible ways.. one would be to set callerid to the external number that was called..
12:32.59ManxPowerskyhawker, callerid=Company A or callerid=Company B  BEFORE each channel=> line in zapata,conf
12:33.30skyhawkerManxPower : thanks .. i am using a external SIP gateway though
12:33.35ZhadnostManx> Is setting callerid mandatory in zapata.conf ?
12:33.56ManxPowerZhadnost, no
12:34.00NirSanyone has an idea why the L parameter on Dial and S parameters will not work ?
12:34.27ManxPowerskyhawker, then the calls will come into Asterisk and match and exten => line for that dialed number and you can SetCIDName there.
12:34.40NirSI'm using 1.2.6
12:34.45ManxPowerNirS, paste a non-working Dial line that uses those options.
12:34.50NirShold on
12:35.07ManxPowerZhadnost, "show application dial"
12:35.13skyhawkerManxPower : thanks
12:35.16key2how can I call a number and dial DTMF keys for calling a number and dialing an extention ?
12:35.37ManxPowerkey2, see the D option to Dial
12:35.44ZhadnostD(<digits).
12:35.47Zhadnostin the dialstring
12:35.54ManxPowerZhadnost, make them look it up
12:36.00Zhadnosts/<digits/<digits>/;
12:36.05key2ManxPower: thx
12:36.13ManxPowerand it's in the OPTOINS string, not the dial string
12:36.26NirSSIP/972544482826@62.90.49.50|120|S(60)M(ngx1_originator_connect^227^d41d8cd98f00b204e9800998ecf8427e443bb091511f9^972544482826^1^moh_promo1^60)L(60000:30000)
12:36.48Zhadnostwow, that hurts to read.
12:36.57NirSsorry
12:36.59NirSthat is the dial string
12:37.02NirSany idea Manx ?
12:37.04Ahrimanesalmost like reading regex's
12:37.34*** join/#asterisk io_error (n=error@87.236.196.130)
12:38.09ManxPowerNirS, are you setting the variables to specify the file to be played.
12:38.24NirSno, I'm using the defaults
12:38.31ManxPowertry setting them.
12:38.35NirSok, will try that
12:38.42Zhadnostas a curiousity, can you get spandsp to turn a fax into an email?
12:39.05Zhadnostor even into a tif file. (after that it'd be simple).
12:39.08tzangerZhadnost: yes you can
12:39.44tzangertiff2ps and ps2pdf
12:39.44tzangerthen email the pdf
12:39.44tzangerthis is well documented on the wiki and google
12:39.46Zhadnostno doubt,, I was just curious.
12:40.09tzangerbut good lord is spandsp and libtiff picky... it's not exactly spandsp's fault... libtiff has some *fucked up* releases for faxing
12:40.19shiznatixManxPower, There was nothing helpful there. I found one guy who has the same problem but it was not answered :(
12:40.36znoGZhadnost: you should look into iaxmodem. Great solution and you can use HylaFAX which does just that (converts to PDF and emails)
12:40.40io_errorblah...I can't call in or out on FWD... but I seem to be registered OK. Where do I start on this?
12:40.45tzangeryes I have to play with iaxmodem again
12:40.54tzangerI hear redder86 and coppice got 14k4 working
12:41.22ZhadnostznoG> I already use them together, seems pretty good, never configured asterisk to send emails instead of faxes before.
12:41.35ZhadnostznoG> been using hylafax for years.
12:41.48tzangerI want to get my old Ascend Max plugged into hylafax
12:41.57tzangertelnet to port 5000 or 9000 and you hit a modem
12:42.05tzangerI just need to write a simple telnet client for hylafax
12:42.14tzangerthere are dozens of telnet SERVERS to serial, but nothing reverse
12:42.15austinnichols101~SER
12:42.25jboti guess ser is Sip Express Router - see http://www.iptel.org/ser/
12:42.25znoGZhadnost: oh, you wanted to know if spandsp can do it.. well spandsp receives into tif, so then you run a script on the "h" extension to turn it into pdf and email. It's pretty simple.
12:42.29ManxPowerYay!  I finished adapting my voicemail outcall stuff to be 1.2 specific  (more or less)
12:42.31NirSManx, that didn't help
12:42.39tzangerManxPower: what you say?
12:43.02tzangerSELECT a.red_mbox as "Property", to_char(sum(b.red_ctime - c.red_ctime),'MI:SS') as "Time Spent", cdr.clid as "Caller*ID"
12:43.05tzangerFROM realestate_detail a, (select red_uid, red_mbox, red_ctime from realestate_detail WHERE
12:43.08tzanger<PROTECTED>
12:43.11tzanger<PROTECTED>
12:43.14tzanger<PROTECTED>
12:43.17tzangerWHERE   a.red_uid = b.red_uid
12:43.19ManxPowertzanger, my users are too stupid to have the system send notification of new voicemails via SMS to their cell phone, so the system has to call them to tell them they have new voicemail
12:43.20tzangerAND     a.red_uid = c.red_uid
12:43.22tzangerAND     b.red_uid = c.red_uid
12:43.24tzangerAND     b.red_mbox = c.red_mbox
12:43.27tzangerAND     a.red_mbox <> '?'
12:43.29tzangerAND     cdr.uniqueid = a.red_uid
12:43.32tzangerGROUP BY a.red_uid, a.red_mbox, cdr.clid
12:43.35tzangeroh shit
12:43.37tzangerwrong window guys, sorry
12:43.40tzangerManxPower: :-)
12:43.42[TK]D-Fendertzanger : Sure you're going slow, but its obnoxious!  Pastebin!
12:43.48iCEBrkrtzanger: Hey! thanks man!
12:43.52tzanger[TK]D-Fender: that was a mistake, sorry :-)
12:43.57tzangerdoing some real estate crpa
12:43.59tzangerer crap
12:44.15*** join/#asterisk brockj49464 (n=brockj49@41.105.dhcp.hope.edu)
12:44.17austinnichols101~siproxd
12:44.30tzangerthat was supposed to be basted into my pgsql window, not this one
12:45.29key2Zhadnost: once the answer machine I call has answered, it does,'t Dial the extension, do you have more info on what I should look for ?
12:45.50NirSManx, setting the variable didn't do any help
12:46.19NirSis it possible that if the dialing activated a Macro to pass the called user into a meetme room, then the L and S parameters can't be used ?
12:46.45shiznatixIs there anyone that can help me with my Zap problem?
12:47.05NirSshiz, maybe I can
12:47.06NirSshot
12:47.57shiznatixNirS, alright well I am trying to use Zap to call outside lines. I have everything setup but when I call from the SIP phone it just automatically answers and gives me these high pitched beeps
12:48.06shiznatixNirS, this is the output and my setup: http://pastebin.com/653321
12:48.47ManxPowershiznatix, ignore the answer
12:49.31Zhadnostkey2> you may need to write a macro to do a pause, then dial the extra extension.
12:49.48Zhadnostkey2> I've seen stuff liekt his on the mailing list but I'm damned if I can remember how it was done.
12:49.55key2Zhadnost: I do a wait
12:50.07shiznatixManxPower, ok ill ignore the answer but whats up with this horrible beeping and the failure of the dialed phone to ring
12:50.10*** join/#asterisk Hali_303 (n=surfk@dsl51B6E6EB.pool.t-online.hu)
12:50.14Hali_303hi
12:51.11ManxPowershiznatix, I CANNOT help you with BRI ISDN issues
12:51.15*** join/#asterisk Cheetah (n=Snak@62.217.48.111)
12:51.21*** join/#asterisk linville (n=linville@nat-pool-rdu.redhat.com)
12:51.22ManxPowerI thought you were running an X100P
12:51.29NirSdo you also have an E1/T1 card in your box ? or just a TDM 400 ?
12:51.48Cheetahheya
12:51.59key2Zhadnost: that's the D<> that * doesn't like, what docs should I look for getting the syntax on Dialing DTMF during the call?
12:52.10*** join/#asterisk brif8 (n=chatzill@rrcs-71-41-50-162.se.biz.rr.com)
12:52.10shiznatixManxPower, ok no problem because this card is not a BRI ISDN card. its a zapata card
12:52.19brif8I'm getting can't locate Asterisk/AGI.pm  how do I install this using gentoo's emerge ?
12:52.44ManxPowershiznatix, then why do you have signalling=euroisdn?
12:52.44*** join/#asterisk cced (n=dev2003@222.33.36.205)
12:52.50ManxPowerbrif8, I don't know if you can.
12:52.56shiznatixManxPower, the BRI ISDN card is another card I have installed on the asterisk box
12:52.57CheetahWhat is the better decision for a 30-user phone upgrade?  Linksys SPA942  or  SNOM 360?
12:52.57ManxPowerjust download it and install it
12:53.04Cheetahanyone has some experiences with those brands?
12:53.08*** part/#asterisk io_error (n=error@87.236.196.130)
12:53.16brif8ManxPower: from where ?
12:53.34ManxPowerbrif8, search for "asterisk-perl" in Google
12:53.56ManxPowerkey2, it's D() not D<>
12:55.02key2ManxPower: like Dial(blah,D<2>) ?
12:55.07ManxPowerNirS, It would not suprize me because then you do't have a DIAL
12:55.15key2i mean
12:55.15ManxPowerkey2, read the damm docs for dial
12:55.21key2ManxPower: like Dial(blah,D(2)) ?
12:55.29NirSManx, I pasted the options
12:55.34NirSnot the actual dial command
12:55.36*** join/#asterisk fuzzbawl (i=fuzzbawl@69.44.167.80)
12:55.37Cheetah:'(
12:56.16ManxPowerNirS, is your Dial command secret or something?
12:56.19*** join/#asterisk apardo (n=apardo@62-15-114-125.inversas.jazztel.es)
12:56.25NirSno
12:56.27NirShold on
12:56.38NirSDial(SIP/${originator}@62.90.49.50,120,S(${call_limit})M(ngx1_originator_connect^${meetmeroom}^${ivr_session_id}^${originator}^${recording}^${moh_class}^${call_limit})L(${call_limit}000:30000))
12:57.13Hali_303is there a hacker's guide for asterisk? what I mean is something like an introduction of the source code, internal mechanisms, etc.
12:57.26ManxPowerNirS, When you look at the console you see the VALUES of the variables, not the actual ${meetmeroom} string, right?
12:57.44NirSI see the actual values
12:57.50ManxPowerNirS, I think macros are run before Dial actually happens so I doubt that the call limits apply to the macro.
12:58.24ManxPowerNirS, does everything work EXCEPT the call time limits?
12:58.28NirSThe macro actually is run after the call is answered, and that really happens
13:00.08*** part/#asterisk sercz (n=serz@i3ED6F067.versanet.de)
13:00.29*** join/#asterisk sercz (n=serz@i3ED6F067.versanet.de)
13:01.20ManxPowerNirS, Perhaps it's a string length limitation.  Try putting the limit options before the M() option
13:02.29NirSHmmm... interesting
13:02.30NirSwill try
13:03.14*** join/#asterisk PakiPenguin (n=uppal@linuxpakistan/admin/pakipenguin)
13:03.23PakiPenguinhi there , how are you guys today?
13:03.26Cheetahfolks, where (except google) can I ask/find some users who have experience with snom 360 and/or linksys products?
13:03.44Cheetahwe are switching from our old ISDN phone box to VOIP
13:03.44ManxPowerNirS, I encountered a string length problem when doing group voicemail
13:03.59Kattymorning.
13:04.31*** join/#asterisk Ariel_ (n=Ariel@adsl-068-157-125-248.sip.mia.bellsouth.net)
13:04.48Ariel_Morning everyone
13:07.47mitchelocgood morning
13:09.49Ariel_morning hope your day is start great
13:09.54GrizzyGrizmorning.
13:10.17*** join/#asterisk Gunnar (n=gunnar@101.Red-80-24-171.staticIP.rima-tde.net)
13:10.28ManxPowerWhere are dialplan labels documented?
13:11.12KattyAriel_: i'm caffeinating....that's about as far as i've gotten thus far ;)
13:12.20KattyGrizzy: my legs are killing me :<
13:12.28Hmmhesayswell that went suprisingly well
13:12.37KattyHmmhesays: make my legs stop hurting.
13:12.59brif8I'm using teliax as my outgoing LD provider (with SIP connection).  (1) if I SetCallerID() then I get a busy signal. (2) if I don't then it reports as Denver convention center, which I'm not.  Any ideas why ?
13:13.12KattyHmmhesays: thanks.
13:13.31Kattygotta stop doing so many lunges.
13:13.54*** join/#asterisk brad_mssw (n=brad@shop.monetra.com)
13:14.01Ariel_brif8, sounds like they don't allow you to change the callerID
13:14.05*** join/#asterisk FlyboySR22 (n=rsears@sdtc.ar01.f2-40.host2.1.americanis.net)
13:14.50PakiPenguinhi , i have a t1 ( sangoma ) and an astribank connected to a server  , when i try ztcfg -vvvvv , it gives me this error , can someone tell me what can be wrong please ? http://pastebin.com/653419
13:15.20x86hi PakiPenguin :)
13:15.34brif8Ariel_: why would it show incorrectly have they missed something or have I
13:15.48grem_linHey, could anyone give me any clues as to why this dial plan doesn't work as intended http://pastebin.ca/49019 , when I dial a number after authenticating I'm simply getting a fast-busy tone. Any help would be greatly appreciated.
13:16.17Ariel_brif8, post how you set your callerID on pastebin so we can see it.
13:16.44*** join/#asterisk DiggerDan (n=chatzill@adsl-69-107-142-7.dsl.pltn13.pacbell.net)
13:17.43brif8http://pastebin.com/653426
13:17.57key2how do I set the time lenght for the DTMF i dial ?
13:18.19key2if i want each dtmf I dial to last 1s for example
13:18.22*** join/#asterisk op3r (n=op3r@202.71.189.66)
13:18.39op3ranybody knows how to improve the quality of the calls on asterisk?
13:18.51ManxPowerop3r, that would depend on what is causing the problem
13:18.57taecHave you set Quality(high) ?
13:19.05op3rtaec: nope
13:19.07*** join/#asterisk Router19 (i=SMOKEY@modemcable075.195-131-66.mc.videotron.ca)
13:19.17taecop3r, oh, well it defaults to Quality(very-very-very-poor)
13:19.20ManxPowerkey2, I don't think you can for SIP
13:19.30op3rbecause we had a problem dialling new south wales yesterday
13:19.36op3rand the line quality sucks bad
13:19.37op3r:(
13:19.45ManxPowerthe DTMF length is determind by the device that interfaces with the telco
13:20.01Ariel_PakiPenguin, those settings don't look right. since it's telling you that your using the wrong signal
13:20.15key2ManxPower: when I do a Dial(tech/chan,,D(123456789)) I only hear one beep, is it normal ?
13:20.19ManxPowerop3r, My car is having problems.  How do I fix it?"
13:20.27Kattywith duct tape!
13:20.40ManxPowerkey2, no you should hear DTMF for 12345678 and 9
13:20.58brif8Ariel_: do you see any problem with it ?
13:21.00nokyi'm trying to install zaptel in a fc3 follow the wiki...(i have a 2.6 kernel)... but it create a /lib/modules/2.6.9-1.667smp/misc/ztdummy.o
13:21.13sylelets troubleshoot ManxPower, did you remember to put gas in your car? :)
13:21.19nokyshould is not ztdummy.ko ?
13:21.19Ariel_just started to read it give me a minute
13:21.51ManxPowerkey2, who is your provider?
13:22.06key2ManxPower: I do it in internal, just for test purpose
13:22.28key2ManxPower: all I want is when I call an extension, it answers and play the dtmf 123456789
13:22.33ManxPowerkey2, Just paste the damn Dial line so I don't have to waste my time asking you for more information
13:23.27key2exten => 5665,4,Dial(iax2/key2,,D(123456789))
13:23.47ManxPowernow what device is "key2"?
13:23.59key2my idefisk phone
13:24.16ManxPowerkey2, then you need to talk to the idefsk people.
13:24.38key2it's not asterisk that generate the dtmf sound ?
13:24.45ManxPowerasterisk sends the messages DTMF 1  DTMF 2  DTMF 3, etc.  The idefsk then actually generates the tones and length of tones.
13:24.59ManxPowernoky, asterisk does NOT do inband DTMF with IAX2 EVER
13:25.11key2ohh ok
13:25.37Hmmhesaysi see there is some skype to sip software now
13:25.53key2but basically, if I want to call an exten, and it answers and play dtmf ? like i did ?
13:25.54noky?¿
13:25.57nokywhat?
13:26.03ManxPowerkey2, correct.
13:26.05key2Hmmhesays ?? where ?
13:26.23Hmmhesayskey2 are you noky
13:26.24ManxPowernoky, you got caught in my "no, asterisk does NOT do inband DTMF with IAX2 EVER" auto complete
13:26.41nokyok
13:26.52nokyztdummy.ko or ztdummy.o ?
13:27.00nokysomething is wrong
13:27.13key2ManxPower: but with this solution, it calls back the phone, it doesnt just answer it and play the dtmf
13:27.22*** join/#asterisk umay (n=chris@70-101-61-50.dsl2-plymouth.roc.ny.frontiernet.net)
13:27.28op3rexten => i,1,PlayBack(bad) <---------------- is this the reason why chanspy suck bad?
13:28.30*** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com)
13:28.30*** mode/#asterisk [+o anthm] by ChanServ
13:28.46*** join/#asterisk pdunkel (n=pdunkel@213.235.231.189)
13:29.32ManxPowerI guess I should head out to the other office.
13:30.08*** join/#asterisk brif8 (n=chatzill@rrcs-71-41-50-162.se.biz.rr.com)
13:30.17brif8Areil_: sorry short power failure
13:30.18NirSManx, I changed the order, it didn't do much help
13:30.22NirSstill S and L don't work
13:31.06*** join/#asterisk sorryIdontknow (n=gabriela@200.122.94.137)
13:31.16iCEBrkrGrrr
13:31.17Hmmhesaystobad this bullshit runs on windows
13:31.25iCEBrkrDamnit.. Asterisk keeps dying
13:31.52sorryIdontknowHello, I have a question. Can asterisk record/monitor calls by itself, or does in need another program to do that (monitor/mixmonitor)?
13:32.18*** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl)
13:32.40*** join/#asterisk dsfr (n=dsfr@pdpc/sponsor/digium/dsfr)
13:32.56NirShere's something weird
13:33.06NirSI've brought up the debug level to show debug messages
13:33.23NirSasterisk doesn't even show anything happen where the time of alert is supposed to happen
13:33.26NirSwhich is really weird
13:33.35pdunkelsorryidontlnow: see res_features it can do it on its own. You usually activate ist with *1 (see features.conf)
13:33.39Ariel_brif8, try this: http://pastebin.com/653451
13:33.40docelm0MEW!
13:33.47docelm0iCEBrkr, NEWB!
13:33.50sorryIdontknowAh, thank you.
13:34.23iCEBrkrdocelm0: U MAK BABBEE JEBUS KRY
13:34.31docelm0DAMN RIGHT!
13:34.33iCEBrkrhaha
13:34.44Kattyhey docelm0
13:34.54docelm0whadup?   I see you sitting on myspace..
13:35.00iCEBrkrI think I may have to upgrade my Asterisk install :(
13:35.00pdunkelsorryidontknow: Also look at http://www.voip-info.org/wiki-Asterisk+cmd+monitor
13:35.08Kattyi'm trying to wake up
13:35.09*** join/#asterisk Vagabond (n=Vagabond@pdpc/supporter/active/Vagabond)
13:35.14sorryIdontknowThanks, pdunkel. :)
13:36.08Vagabondhey, I'm playing with realtime support, I've got it working for sip peers, but now I'd like to store agents in the database too. I can't find any documentation or discussion about this, is it possible?
13:36.35KattyVagabond: hi.
13:36.40*** part/#asterisk sorryIdontknow (n=gabriela@200.122.94.137)
13:36.54KattyVagabond: so nice of you to be /nice/ instead of just waltzing in and asking
13:38.01iCEBrkrDamnit, I need another 727 DID
13:38.02iCEBrkrc/lear
13:38.15*** join/#asterisk Samoied (n=Samoied@popeye.opens.com.br)
13:38.22[TK]D-FenderKatty : Potentially preferable to "Why doesn't my A@H work?  I put the CD in and EVERYTHING!" ;)
13:38.28SamoiedHello all!
13:38.30tzanger[TK]D-Fender: haha
13:38.51Vagabondso I'm expected to socialize before I get to ask a question? ;)
13:38.58tzanger[TK]D-Fender: I managed to get my ip501 rollout (12 phones) fully ftp-provisioned this weekend
13:38.59SamoiedAnyone knoe the possible values to _ALERT_INFO for grandstream Handytone?
13:39.04znoGwell, considering the people in here are humans, it's not a bad idea Vagabond.
13:39.08nokyhow can i know if ztdummy is OK? i'm trying to configurate the conference (meetme) and i can't... i'm following the wiki for the installation of zaptel...
13:39.15SamoiedI have tried Bellcore-r1,r2, without success
13:39.22[TK]D-Fendertzanger : Whats scary is just how many of those we get here... Should ask Russell to "enhance" the topic list of "outlawed" junk in here )
13:39.25nokywhat could be wrong?????
13:39.29Vagabondyeah, I'm just trying to make some progress with this today, not really idle on company time ;)
13:39.35noky[TK]D-Fender: help please
13:39.35Ariel_Vagabond, it would be nice. But no you don't have to. It's just that people tend to ignor
13:39.40tzangernot a single thing is configured on the phone.  hell even the dhcp server gives a separate ip range and tftp option to the polycom MACs :-)
13:40.09[TK]D-Fendertzanger : Yeah, they go so fast when you have limitied profile types.... I did my 301 from complete scratch within 15 minutes including SIP upgrades.
13:40.21VagabondAriel_: yeah, I guess I'll hang around in here then, it looks like I'll be doing asterisk work for a while
13:40.42noky:D
13:40.43[TK]D-Fendernoky : Do you see ztdummy loaded?
13:40.45Ariel_Vagabond, ok. I wish I could help you. But I do not use realtime and don't belive in it either.
13:40.49nokyyes.. is it...
13:40.52KattyVagabond: well i'm social :P
13:40.54nokyappears in 'lsmod'
13:41.09tzanger[TK]D-Fender: yes NOW that I know what I'm doing it's easy
13:41.09[TK]D-Fendernoky : and cat /proc/interrupts?
13:41.20tzangerit took me 2.5 hours to get the fucking DHCP server to match a partial MAC though
13:41.27RoyK[TK]D-Fender: I heard a good that goes back few years. There was a Word Perfect course, starting with installing WP5.1 from 5 1/4" floppies, so teacher says "Insert disk one and press enter"... done with that he goes "Now insert disk two..", and "Now insert the final floppy.." and a lady in the room raised her hand and commented that there weren't room for more...
13:41.40*** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com)
13:41.40*** mode/#asterisk [+o anthm] by ChanServ
13:41.41[TK]D-Fendertzanger : Why bother matching a partial MAC?
13:41.42Ariel_tzanger, well at least  you got it working.
13:42.00tzanger[TK]D-Fender: because I wanted a separate pool/dhcp options for them and I did NOT want to match individual MACs
13:42.01[TK]D-FenderRoyK : Golden Oldie....
13:42.03tzangeroffhand, you do this
13:42.10tzanger# matches any polycom hardware (vendor MAC 00:04:f2):
13:42.10tzanger<PROTECTED>
13:42.10tzanger<PROTECTED>
13:42.13tzanger<PROTECTED>
13:42.14tzangernow how fucking ugly is that
13:42.15russellb[TK]D-Fender: I heard my name!!!
13:42.18tzangerAriel_: indeed
13:42.23nokymmm
13:42.38[TK]D-Fendertzanger : I just give all options to all devices on my lans... basically Windows PC's ignore the rest so what the hell...
13:42.45tzangeractually Ariel_ and [TK]D-Fender do you know offhand what the DHCP options are to tell the polycom phones what the FTP username/pass is?  I cannot find that info and I'd like to change the user/pass to tighten up security a little
13:42.52tzanger[TK]D-Fender: yeah, I didn't want to do that :-)
13:42.56[TK]D-Fenderrussellb : !!!
13:43.16Ariel_tzanger, I have not done that yet.
13:43.31Ariel_but it's something I will oneday get to as I would like to change that my self.
13:43.58[TK]D-Fendertzanger : sorry, gotta do it at the phone level..
13:43.59noky[TK]D-Fender: http://pastebin.com/653473 look this
13:44.14tzangersimiarly, I cannot figure out how to specify http:// instead of ftp stuff but it doesn't look possible either
13:44.31tzangerseems like a rather egregious omission on polycom's part... they made it SO nice to provision but tripped right at the finish line
13:44.31nokyplease
13:45.10pdunkelHi, does anyone know which Event headers for "sip notify" are supported by the Snon190/ElmegIP290 phones ?
13:45.25[TK]D-Fendernoky : I don't see ZTDUMMY in your interrupts list..... modprobe it...
13:46.26*** join/#asterisk Darwin35 (n=Darwin@c-24-9-75-234.hsd1.co.comcast.net)
13:46.48nokymodprobe ztdummy... and the interrupts is the same..
13:46.50*** join/#asterisk bkw_ (n=brian@c-68-32-112-142.hsd1.md.comcast.net)
13:46.50noky:(
13:46.55[TK]D-Fendertzanger : http you need to do in the provisioning files the first time and doesn't get passed except in a retreived file later
13:47.08tzanger[TK]D-Fender: eh?
13:47.56[TK]D-Fenderbasically you need to feed a config to it that contains the HTTP provisioning server info so that the NEXT time it boots it goes there instead I believe
13:48.14[TK]D-Fendertzanger : either way though... I like FTP just fine, thanks :)
13:49.14*** part/#asterisk mitcheloc (n=mitchelo@70-32-188-167.lmdaca.adelphia.net)
13:49.20[TK]D-Fenderrussellb : Hey, care to add "elaborate the "unspeakables" list a bit in the topic? "AMP/FreePBX/A@H" and any others I've missed....
13:50.09backbluerussellb: do you know if x100p (md3200) clones, do suport callerid?
13:50.09tzanger[TK]D-Fender: true enough
13:50.58RoyKthe x100p is stupid and doesn't support anything, but zaptel/libpri supports callerid
13:51.07[TK]D-Fenderrussellb : Yeah, add X100P's in there too! ;)
13:52.41tzangerx100p can so do callerid
13:52.50tzangerit's not done on the card, but neither is any other digium card's callerid :-)
13:53.05nokycrws-----T  1 root root 196,   0 Apr 11 10:13 ctl
13:53.10noky/dev/zap/ctl
13:53.15nokyis it ok ?
13:53.22noky0 byte? o_O
13:53.48[TK]D-Fendernoky : no clue... I've never successfully installed ZTDUMMY before and really need to....
13:53.50russellbis there a #aah?
13:54.04brif8Ariel_: thanks it works
13:54.30Ariel_russellb, no it's support at freepbx
13:54.42sylenoky do you have ztdummy?
13:54.42nokymm?
13:54.46nokyyes
13:54.48russellbso confusing :)
13:54.51nokyi compile zaptel for this
13:54.59*** topic/#asterisk by russellb -> Asterisk: The Open Source PBX -=- Asterisk 1.2.6 and Zaptel 1.2.5 Released! (March 27, 2006) -=- http://www.asterisk.org/ -=- AMP/FreePBX users should join #freepbx for support
13:55.04Ariel_russellb, aah is just asterisk plus freepbx as an iso
13:55.20syleso what happens on modprobe ztdummy?
13:55.28*** topic/#asterisk by russellb -> Asterisk: The Open Source PBX -=- Asterisk 1.2.6 and Zaptel 1.2.5 Released! (March 27, 2006) -=- http://www.asterisk.org/ -=- AMP/FreePBX/Asterisk@Home users should join #freepbx for support
13:55.32russellbAriel_: yeah, i know
13:55.45tzangerljam: my gf likes to play tennis... we should get together so you can show me up :-)
13:55.56pdunkelrussellb :  :)
13:56.12ljamtzanger: like that Seinfeld episode? :)
13:56.22tzangerno idea
13:56.26nokysyle: i use /etc/init.d/zaptel start
13:56.52nokyand i don't log an error...
13:57.04syleyeah but what happens when you run modprobe ztdummy?
13:57.15*** join/#asterisk [ProB]CrazyMan (n=crazyman@p549F3E1C.dip0.t-ipconnect.de)
13:57.33nokywait...
13:57.40syledon;t use scripts if you have a problem, use them after everything works ok
13:57.44ljamtzanger: doh -- it's a funny episode -- but yah -- lets play some tennis! it's fun -- only started last summer, so I'm not very good, but I can rally alright. I've had a bunch of people ask me to play tennis this week already
13:57.45nokyi will try again, because the zaptel load automatic
13:57.49nokyok
13:57.58tzangercool.  I suck mightily at it
13:58.05ljamlol
13:58.20ljamI suck at overhand serves.... finally getting to the point that I can get some in now
13:58.34pdunkellet's open a new room: #tennis  :)  Anyone up for it?
13:58.50russellbooh, i'll play someone :)
13:59.20ljamI prefer real-life tennis, not virtual :)
13:59.25russellbyes, me too
13:59.30russellbljam: next Astricon, it's on
13:59.39syleno offence but what does tennis have to do with asterisk?
13:59.41ljamrussellb: done and done!
13:59.49ljamwhat does asterisk have to do with asterisk? :)
13:59.58pdunkelljam: Now that was an ace if I ever saw one! :)
14:00.05russellblol
14:00.05ljamlol
14:00.10russellbwe got burned.  :(
14:00.13ljamlol
14:00.18Vagabondhmm, I've been browing the source of chan_agent.c and I don't see anything about realtime storage, am I to assume that agents cannot be stored in the realtime DB right now?
14:00.23ljamwe?  I think he was talking to you
14:00.30[TK]D-Fenderljam : Tennis is cool because of the number of free municipal courts.  A good change of pace and gets your outside.  I've taken up vollyball and Tenshin Shoden Katori Ryu (one of the oldest schools of Japanese swordsmanship) lately... I may add squash to that list shortly...
14:00.41russellbVagabond: i don't think they can, no
14:00.45*** join/#asterisk bweschke (n=bweschke@66.152.225.74)
14:00.50[TK]D-Fenderljam : and in VB I still haven't recovered my overhead serve :)
14:00.59russellbVagabond: if you search for the word "realtime" and it's not there, then yeah, it's not supported :)
14:01.03Vagabondrussellb: do you know if that's planned at all? Or will I have to hack it in?
14:01.21backbluerussellb: do you know if x100p (md3200) clones, do suport callerid?
14:01.22[TK]D-FenderEVERYTHING!
14:01.23russellbsure, it's probably on the list of the 5 billion other things we want to do
14:01.30russellbbackblue: they should
14:01.38pdunkelsyle: see chan_tennis.c (will be checked into source as soon as I have approval) Concept: Transport of voice data via tennis balls!
14:01.44syleplay a real sport like hockey :)
14:01.45Vagabondrussellb: anywhere I could look to see what is planned? would it be on the bugtracker?
14:01.50sylegirls play tennis
14:01.54backbluedoesn't work with me! :(
14:01.58[TK]D-Fenderpdunkel : Packet loss must get expensive ;)
14:02.02ljamsyle: EXACTLY -- girls play tennis
14:02.12nokysyle: i have an error
14:02.16[TK]D-Fendersyle : Sure, you play hockey, we'll take the girls ;)
14:02.19russellbI don't think there is an existing patch on there, but you could look around.  Threre are about 300 open issues so it's hard to keep upp
14:02.20[TK]D-Fender:D
14:02.23cj-rmhmm... My Zap groups don't seem to be working properly for outgoing calls.... Dial() appears to dial twice!  Any ideas?
14:02.24pdunkel[TK]D-Fender : Yes but is sponsored by ATP!
14:02.30*** join/#asterisk mhnoyes (n=mhnoyes@dialup-4.246.236.47.Dial1.SanJose1.Level3.net)
14:02.32Vagabondrussellb: thanks
14:02.34ljam[TK]D-Fender: I play beach volleyball in the summer ---mmmmm... summer girls :)
14:02.41nokysyle: http://pastebin.com/653503
14:02.48russellbljam: on your fake beach!
14:02.49nokyi don't know how fix
14:02.57syleis asterisk running as root?
14:02.59nokyi follow the wiki... i don't understand
14:03.00ljamrussellb: with the fake... you know :)
14:03.00pdunkelAny takers on my sip notify with snom question?
14:03.10nokyasterisk is doesn't running at this moment
14:03.14nokyshould be run?
14:03.20sylewhen you start it, as what user?
14:03.27*** join/#asterisk A-Tuin|work (n=A-Tuin@212.41.185.81)
14:03.38nokyroot
14:03.43nokysorry, is running now...
14:03.45nokyroot      2090  0.0  0.7 19220 7348 ?        Sl   10:24   0:00 /usr/sbin/asterisk -vvvg -c
14:03.45bweschkeVagabond: what are you looking to do?
14:03.47*** part/#asterisk mhnoyes (n=mhnoyes@dialup-4.246.236.47.Dial1.SanJose1.Level3.net)
14:03.49[TK]D-Fendernoky : If you don't see ZTDUMMY loaded, forget it.  You sure you uncommented it in the source file and recompiled and everything?
14:03.57nokyyes..
14:03.58cj-rmI'm not sure if it detects that my channel is busy
14:04.08pdunkelsyle: While we are at * and users. What are the pros and cons of running * as root?
14:04.11nokyi'm sure
14:04.49nokyin Makefile i uncomment the ztdummy... make clean && make linux26 && make install ... and then (i have use udev) configurate the udev follow README.udev
14:04.50[TK]D-Fenderpdunkel : If there's something exploitable you could comprimise your entire server.....
14:05.05Vagabondbweschke: I'd like to be able to have agent numbers/passwords in the realtime db so they're easier to add
14:05.11Vagabondand generally manage
14:05.20syleso you have your udev entries in /etc/udev/rules.d/50-udev.rules ?
14:05.29ljamrealtime? ugh -- just use a DB :)
14:05.35nokyi test with 'make' instead of 'make linux26'...
14:05.40nokyyes...
14:05.48syledid you reboot after adding them?
14:06.00nokyyes.. i reboot
14:06.08nokybut /dev/zap/ctl appears with 0 bytes
14:06.09cj-rmWhy is it when I try and make an outbound dial with Zap groups that my extension runs twice??????
14:06.16nokyi don't think that is ok..
14:06.30ljamok -- off to get some breakfast, then off to implement slony-I :)
14:06.45cj-rmYet when I specify the specific channel to call on e.g. Zap/3 then it doesn't.
14:06.52pdunkel[TK]D-Fender: Yeah, the question is more like what issues are to watch out for if I decide not to run as root as well as likelyhood of exploits. Of course these can only be experience reports and I do know the general issues well. I was hoping for something more along the line of experience reports.
14:06.59sylemake linux26 was suppose to be depreciated anyways in recent code bases , shouldn;t make a difference
14:07.14[TK]D-Fendercj-rm : pastebin your entire extensions.conf and zapata.conf.
14:07.39syleits suppose to be 0 bytes
14:07.44nokyok..
14:07.50pdunkelljam: just modprobe coffe && modprobe cigarette (That does it for me!)
14:08.10[TK]D-Fenderpdunkel : To tell you the trusth I'm not aware of more that 1 occurance where anything serious could happen, its just a "purist approach" to not run daemons as root.
14:08.12*** join/#asterisk blaylock (n=seth@snap.helixsystems.com)
14:08.38sylepastebin 3 things will you, modprobe ztdummy, ls -al /dev/zap , and cat /etc/udev/rules.d/50-udev.rules ok
14:08.44cj-rm[TK]D-Fender: Can Dial() on a group match a Local channel?
14:08.57pdunkel[TK]D-Fender: Well I guess I'm about to find out how much of a "purist" I really am. Thanks
14:09.18sylelsmod as well
14:09.18nokyoks
14:09.28pdunkelAnyone here has any SIP knowledge (relating to SIP NOTIFY) ?
14:09.40pdunkelOr are you all ZAP guys ?
14:10.13sylelol
14:10.31[TK]D-Fenderpdunkel : I wouldn't and don't personally bother...
14:10.36*** join/#asterisk pif (n=ldm@zenon.apartia.fr)
14:11.04nokysyle: http://pastebin.com/653518
14:11.06pdunkel*pdunkel sizzles
14:11.15*** join/#asterisk cj-rm (n=cjrm@81-86-30-78.dsl.pipex.com)
14:11.15iCEBrkrGrrr
14:11.23iCEBrkrEveryone elses system runs just fine *Grumps*
14:11.30pdunkelsomeone give me a raygun so i can zap some as well! :)
14:12.40[TK]D-FenderiCEBrkr : * is perfectly stable!  When it crashes it doesn't move at all!
14:12.45iCEBrkrhaha
14:12.55iCEBrkrWell, I *Am* dropping 35 call files at a time.
14:13.07pdunkeliCEBrkr: No mine crashes about 2 a day, so I have implemented a cron check script until I have found out why the crash happens. *pdunkel sends sympathy
14:13.15iCEBrkrOdd thing is.  I just hit my up-arrow once to relaunch asterisk -cvvvvv and it worked.
14:13.37*** join/#asterisk mut (n=animenod@65.111.222.120)
14:15.31nokysyle: something wrong ?
14:15.35*** join/#asterisk opc0de (i=adam@CPE006008148866-CM000f9fa8c50a.cpe.net.cable.rogers.com)
14:15.43opc0dehey anyone here use chan_btp ?
14:15.46*** join/#asterisk pjz (n=pj@66.219.59.183)
14:15.59pjzhow is the soundpoint 501 different from the soundpoint?
14:16.17pjzmy autoboot-and-upgrade-firmware stuff doesn't seem to work on the 501 but did on the 500
14:16.38pjzthe 501 is saying 'Error updating BootROM'
14:16.43opc0deI'm trying to figure out why "client => user,00:11:22:33:44:55,Zap/4/1234567891" won't dial 1234567891.. it simply picks up Zap/4 and connects me, without dialing then umber
14:16.49pjzI've got bootrom v2.6.1
14:17.07tzangerpjz: hmm
14:17.14tzangerI didn't try to upgrade my bootrom, I'm using 2.6.1
14:17.52opc0depjz:
14:17.56[TK]D-FenderLeave your BR alone if at all possible....
14:17.57opc0deI've got bootrom 3.1.3 if you want
14:17.59syleit looks good noky
14:18.09syledid you make install latest libpri first
14:18.11pjzI don't really want to go to 3.1.3 since that's all https and stuf
14:18.17opc0depjz: ar eyou using ftp or tfp?
14:18.22cj-rm[TK]D-Fender: http://pastebin.com/653535
14:18.22pjzopc0de: ftp
14:18.30cj-rm[TK]D-Fender: The relevant bits are there.
14:18.36opc0depjz: does your ftp log show the phones downloading the software?
14:18.47tzanger2.6.1 and sip version 1.6.5.0043
14:18.50opc0deno one here uses bluetooth?
14:18.52OliverXwich ports must i forward in my nat to register asterisk to the sipgate account?
14:18.58nokylibpri ?
14:19.35stoffellhm, why does the MACaddr-app.log of polycom phone keeps getting written in ftp homedir? (i specified logs/ dir in xxx.cfg)
14:19.39bkw_OliverX, if asterisk had a STUN client it wouldn't matter :P
14:19.49*** join/#asterisk mitcheloc (n=mitchelo@70-32-188-167.lmdaca.adelphia.net)
14:19.58syletry modprobe zaptel first
14:20.01sylesame error?
14:20.37nokymm wait me please
14:21.10opc0deis there any way to define a channel which by default dials a certain fphone number?
14:21.17nokymodprobe zaptel is ok
14:21.26nokyi don't see some error.
14:21.27opc0deby using Zap/4/1234567891, it never dials the phone number, it only picks up the channel
14:21.29[TK]D-Fendercj-rm : I don't see where those variables you use are being set.. include MORE....
14:21.37pjzopc0de: hrm, no they're not logging in
14:21.43[TK]D-Fendercj-rm : Show me everything that toushes it...
14:21.51pjzopc0de: do 501s have different usernames?
14:22.14cj-rm[TK]D-Fender: They're being set in the call file which is copied into /var/spool/asterisk/outgoing
14:22.15opc0depjz: you've gotta set the username/password in t he admin config screen on the phone
14:22.43[TK]D-FenderOliverX : typically 5060, 10000-20000, all UDP
14:22.54[TK]D-Fendercj-rm : Ok, I don't know call-files really...
14:22.58pjzopc0de: oh, nm, it's logging in correclty
14:23.20pjzopc0de: oh wow
14:23.23nokymust i wait seconds to modprobe ztdummy?
14:23.28opc0depjz: what uid/password are you using right now?
14:23.32pjz| ProFTPD terminating (signal 11)
14:23.34opc0de456/Polycom ?
14:23.36opc0dethat's not good
14:23.40nokybecause if i wait some seconds ... the modprobe ztdummy works ok
14:23.45pjzyeah, my guess is that that's the problem :)
14:23.46nokyafter the modprobe zaptel
14:24.02pjzSig 11 is Segfault
14:24.03SamoiedAnyone know the possible values of _ALERT_INFO for Grandstream Handytone?
14:24.08syleis it in lsmod now?
14:24.13syleztdummy
14:24.55nokywhen i do modprobe zaptel doesn't appear ztdummy
14:25.08syleis it in lsmod now?
14:25.10nokybut i do modprobe ztdummy... and i don't see error... and now is ok
14:25.12nokyyes
14:25.25greendiseasecan someone lastlog me please?
14:25.37*** join/#asterisk nitram (i=foo@superblob.com) [NETSPLIT VICTIM]
14:25.41sylegreat well script it to start it like that then, have a good day
14:25.59nokyhaha
14:26.09nokysyle: i will try if works
14:26.19nokythanks for your time buddy
14:26.50sylei don;t know why you have to load zaptel first but whatever , just do modprobe zaptel;sleep 5; modprobe ztdummy then in some startup script
14:26.50cj-rm[TK]D-Fender: The variables are set correctly, it works perfectly when the values for ${OUTBOUNDTRUNK} are set to Zap/3 for the 1st Dial() and Zap/4 (for the second).  But not when I use the channel group.
14:27.25cj-rm[TK]D-Fender: By that I mean when ${OUTBOUNDTRUNK} is replaced by Zap/3 and Zap/4 btw...
14:27.35sylei don;t know what release etc your running, nor do i care but at least its working, glad to help
14:28.45*** join/#asterisk vader-- (n=johndoe@204.183.88.101)
14:28.47vader--hello
14:29.51nokythanks
14:33.48nokysyle: i follow heard in asterisk the operator saying "..please try again..", my extension look good.. and in /proc/interrupts doesn't appear the ztdummy... like say [TK]D-Fender must to appear, not? ....
14:34.57nokymy extension of Meetme
14:35.33*** join/#asterisk nitram (i=foo@superblob.com) [NETSPLIT VICTIM]
14:35.40syleit will appear on kernel version 2.4 not in 2.6
14:36.26syleyour concern is it loads and you can see it in lsmod
14:36.33[TK]D-Fenderop3r : which user/pass are you trying to figure out?  there is another mixed case one "plcmspip" for FTP which is "default"
14:36.47*** part/#asterisk kmilitzer (n=km@office-gw.westend.com)
14:36.52nokyok
14:36.55*** join/#asterisk nextime (n=nextime@213-140-6-103.ip.fastwebnet.it)
14:37.31nokydo u use realtime extensions syle ?
14:37.31syleyes
14:37.42noky|  15 | default | 999           |        4 | MeetMe             | 1234|M                               |
14:37.44sylebe a pain in the ass to interact with asterisk froma  webpage otherwise
14:37.50nokythis extension is ok, not ?
14:37.58nokyhehe
14:38.34nokyin meetme.conf only have
14:38.35noky[default]
14:38.37nokyconf => 1234
14:38.52brif8anyone using a Cisco 7920 (wireless) I keep getting  No AP found ??
14:38.54nokywhen i dial 9999 i listen "please try again.." :(
14:39.03noky999*
14:39.45nokythe priority 1 is Answer, the priority 2 is a wait 1 second and the priority 3 is a wait 2 seconds...
14:40.02nokythen the priority 4 is Meetme .. and the priority 5 is Hang up
14:40.08nokymust be ok...
14:40.55sylelooks good to me, haven;t played with meetme , so can;t help you there, realtime looks good, check /var/log/asterisk/debug as well for info
14:41.02nokyApr 11 11:04:40 DEBUG[3949] app_meetme.c: 1234 isn't a valid conference
14:41.04noky:o
14:41.08nokyappears in log...
14:41.16nokyoks.
14:43.44*** join/#asterisk gandhijee (n=loser@host-66-202-34-162.spr.choiceone.net)
14:43.56gandhijeeanyone know of a softphone that can do IP dialing?
14:43.57*** join/#asterisk jsharp (n=jsharp@65.88.255.245)
14:44.37gandhijee??
14:45.06*** join/#asterisk freat (n=ron@h-72-244-84-43.chcgilgm.covad.net)
14:45.08sylewhy not just use your cell phone, switch out your sim card, put in a gsm gateway
14:45.21sylehate a phone that is not wireless hehe
14:46.16gandhijeei c.
14:46.33gandhijeei have wifi phones
14:46.48sylekewl, what kind?
14:46.48Lino`brif8
14:46.54gandhijeei am just lookin for a softphone that can do IP to IP calling
14:46.55Lino`which ap do you have?
14:47.06gandhijeeZyXEL
14:47.10gandhijeeand i just got 2 linksys WIP300's
14:47.21gandhijeehaven't got to test those yet, they are in VA
14:47.36brif8Lino` D-Link 3200AP
14:47.38*** join/#asterisk amdtech (n=amd011@ab1-1-246.shsu.edu)
14:47.39Lino`hmmm
14:47.44Lino`you need to broadcast the SSID
14:47.49brif8I got it had to set SSID
14:47.54Lino`you need to set the encryption parameters for the 7920
14:48.00*** part/#asterisk amdtech (n=amd011@ab1-1-246.shsu.edu)
14:48.00*** join/#asterisk Strom_M (n=strom@gateway.digium.com)
14:48.09Lino`and you need to allow the phones mac address
14:48.10*** join/#asterisk argos73 (n=mike@w010.z208036240.chi-il.dsl.cnc.net)
14:48.12brif8now getting Connecting to CallManager 0  (it cycles 0 to 5 and back again)
14:48.17sylei find desktop phones are only good for speaker phone personally, polycoms are best for that, otherwise use your cell phone or wifi phones, at least you can still take a piss and talk on the phone
14:48.17Lino`ok
14:48.20Lino`thats fine
14:48.26Lino`you have a call manager or asterisk?
14:48.30brif8*
14:48.35Lino`most probably asterisk in this channel *stupid me*
14:48.41jsharpI hate people who take a piss while I'm on the phone with them.
14:48.47Lino`now you need sccp
14:48.50Strom_Msyle, sure, if you don't care about the sound of GSM compression
14:48.54Lino`either chan_sccp2 or chan_skinny
14:48.56sylecall me then, i need to take one :)
14:49.04Strom_Mor if you never have to worry about battery life
14:49.22Lino`:D
14:49.32file[laptop]meep meep
14:49.34brif8I followed the HOW to SCCP  but so far no joy
14:49.37sylecell phones are battery life, i think they are most popular :)
14:49.48[TK]D-Fenderor range, or risk of being dropped....
14:50.00sylethen install access points
14:50.00brif8Lino`: do you use one and do you have it working ?
14:50.06sylesolution for everything
14:50.18gandhijeemy wifi phones run PCMU
14:50.32Strom_Mbasically, mobile and cordless phones are only good for situations where you need to be in many different locations all the time.  Otherwise, corded phones are the way to go
14:50.49gandhijeesyle: installin AP's don't get rid of the handoff problem
14:50.52syleyou have no life if thats what you think , no offence :)
14:51.12gandhijeeany of you guys check out RoamAD's software?
14:51.25gandhijeeits pretty cool stuff
14:51.25file[laptop]Strom_M: are you involved in the... meeting of doom today?
14:51.32brif8do I need to get the sccp2  or is sccp fine , how do I know I have the right one ?
14:51.33Strom_Mmeeting of doom?
14:52.07Hmmhesaysthank you federal guberment
14:52.08Kattyhey Hmmhesays, i found a pretty girl for ya!
14:52.12Hmmhesaysoh yeah?
14:52.36iDunnocan you find me one too?
14:52.40*** join/#asterisk syle (n=blah@unaffiliated/syle)
14:52.48brif8Lino`:   I have /etc/asterisk/sccp.conf  what else is needed ?
14:52.57FlyboySR22Has anyone run into any issues running * on a x64 platform...? We are looking at Suns new Sunfier X2100 servers to deploy * on....
14:53.15sylewaste of money
14:53.28KattyiDunno: i don't share my girls with strangers :P
14:53.28tzafrirFLeiXiuS, I know Debian has occasionally some patches...
14:53.37FlyboySR22syle, WHy..they are cheap and fast
14:53.43FlyboySR22??
14:53.46syledefine cheap
14:53.53FlyboySR22sub $1K
14:54.03FlyboySR22and they fit in with the rest of my sun gear :-)
14:54.27sylethat might be kewl, i;ve always found i could buy 3 intel servers for the price of 1 sun machine
14:54.43FlyboySR22white-box server...?
14:54.57FlyboySR22I have yet to find a good $300 server
14:54.59syle1k is good though if they are that cheap
14:55.06*** join/#asterisk bkw_ (n=brian@c-68-32-112-142.hsd1.md.comcast.net)
14:55.12gandhijeeFlyboy: then just get a linux and run them in 32-bit mode....
14:55.25*** join/#asterisk nitram (i=foo@superblob.com) [NETSPLIT VICTIM]
14:55.25iDunnoKatty: awww :(
14:55.27syleinterested to hearing about asterisk in 64 bit code
14:55.29gandhijeei got my dual core dell for like 900 w/ 2 gigs for ram
14:55.42FlyboySR22gandhijee, kind of defeats the purpose of a 64 bit system dosn't it..?
14:55.43gandhijee*of
14:55.45syleyou have the 64 bit optimizations ready?
14:55.46FlyboySR22gandhijee, New..?
14:56.12gandhijeeFlyboy: no, but IRC the zaptel drivers are not ready for 64 bit yet.
14:56.25FlyboySR22ah
14:56.28gandhijeethat is if u plan on running any PSTN lines
14:56.41Seyrsyle: the $900 Sun server is running AMD, thats why its so cheap
14:56.44gandhijeei just dont come by often
14:56.46FlyboySR22how about the libpri stuff..? I am only going to run PRI cards...
14:56.49syleyeah, he;ll be running in 32 bit mode anyways
14:56.58syleso won;t matter much hehe
14:57.06FlyboySR22Opteron is 64 bit I thought..?
14:57.08*** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn)
14:57.16FlyboySR22Sun X2100 = Opteron
14:57.17gandhijeeFlyboy: you still need zaptel for the PRI stuff
14:57.24FlyboySR22for timing I assume
14:57.27Vagabondhmm, when you use AgentCallbackLlogin, and it asks for an extension, how do I define one it won't say is invalid?
14:57.44gandhijeeFlyboy: no to make the zaptel channels, i assume you are getting a T1 or E1?
14:57.52gandhijeeas i dunno where u are
14:58.04FlyboySR22Yes - sorry. PRI
14:58.07FlyboySR22I am in the US
14:58.11FlyboySR22San Diego
14:58.12jsharpYou'll need zaptel and libpri.
14:58.17*** join/#asterisk plasko (n=plasko@triana.kmpanilla.com)
14:58.30FlyboySR22OK - so stay away from 64 bit for now...
14:58.41gandhijeeok
14:58.42gandhijeeya, you still need the Zaptel
14:58.42gandhijeethats what makes those nice little ZAP/1 channels for you so you can get an incomming call.
14:58.54sylehell no
14:59.00gandhijeeunless you want to have fun =)
14:59.03sylei want to hear results :)
14:59.03jsharpzaptel drives the card.  libpri handles layer 3 ISDN signalling.
14:59.18*** part/#asterisk Seyr (n=Seyr_@cpe-67-10-139-141.houston.res.rr.com)
14:59.27FlyboySR22but zaptel does not work well on 64 bit..? These would be production boxes
14:59.33*** join/#asterisk inv_arp[work] (i=junya@adsl-11-225-195.mia.bellsouth.net)
14:59.44sylebecause a box is 64 bit doesn;t mean it can;t run 32 bit code
14:59.51syleyou just adjust your compiler
15:00.27FlyboySR22syle, got it, but again, then why get a 64 bit processor if I have to run in 32 bit mode..?
15:00.32brif8Lino`:  are you using the Cisco 7920  or anyone else ?
15:00.38FlyboySR22are there other benefits ..?
15:00.41jsharpYour kernel drivers have to be 64-bit clean otherwise bad stuff happens.
15:00.41sylewell that was my point, not really needed
15:00.54sylei was just interested to hear how it works for you
15:00.56FlyboySR22syle, :-) Got it
15:01.37*** join/#asterisk oej (n=oej@apollo.webway.se)
15:01.40syleif your running production yeah i wouldn;t chance it personally
15:01.41brif8I have sccp loaded but it still seems that the 7920 won't connect to * ?
15:02.13*** join/#asterisk gandhijee (n=loser@host-66-202-34-162.spr.choiceone.net)
15:02.28FlyboySR22syle, Thanks for the informaiton, I appreciate it. I have not done anything with a 64 bit processor yet so was looking at trying it out
15:02.36Hmmhesaysso who do ya'll register your domain names through
15:02.53sylei thought you had lots of SUN equipment, they are usually all 64 bit these days hehe
15:03.04Kattywe use register.com
15:03.06Kattybut don't use it
15:03.06FlyboySR22yes, but they are running Solaris
15:03.07gandhijeeThe Sparc boxes are
15:03.11Kattythey don't support reverse dns lookup
15:03.11*** join/#asterisk [ProB]CrazyMan (n=crazyman@p549F3E1C.dip0.t-ipconnect.de)
15:03.12Hmmhesayswhat do you run on them syle, I have a sparc 3000 doing nothing
15:03.20FlyboySR22we have the E450s and some of the Netras
15:03.22jsharpsparc 3000?
15:03.28sylei;ve only run solaris on sparcs
15:03.30*** join/#asterisk wunderkin (i=kev@ip68-226-113-228.ph.ph.cox.net)
15:03.33FlyboySR22we got rid of the E250s
15:03.33gandhijeex86 netras?
15:03.46syleyour big ass fiber array channels and veritas
15:03.49FlyboySR22not sure, I would have to ask one of my Sun guys
15:05.05syleSUN is stable, never had a problem with it ever
15:05.21sylei just couldn;t justify the cost is all
15:05.31jsharpYah, you pay for the stability.
15:05.53FlyboySR22Netra T1 = Sparc 64 bit according to my Sun guy
15:05.57syleyeah well you can argue you pay your sys admin for stability
15:06.03syleso whatever works :)
15:06.16FlyboySR22:-)
15:06.20FlyboySR22how true is thata
15:07.59brif8anyone using the 7920 where did I goof. I keep getting (Connecting to CallManager 0)  it cycles 0 -> 5
15:08.28brif8I have sccp loaded and sccp show devices and channels are emty  version is 20060408
15:08.52*** join/#asterisk nitram (i=foo@superblob.com) [NETSPLIT VICTIM]
15:09.10jsharpSounds like your sccp.conf isn't right.  If show devices is empty, there are no devices properly configured.
15:09.14X-RobGrrrr.
15:09.30X-RobAny way in the dialplan, that anyone knows of, to check if an exten => 1234 exists?
15:10.17tzangerX-Rob: that's a damn good question.  Short of a System(grep...) I odn't think there is a way
15:10.25pdunkelX-Rob: Do you mean you want to check with a dialplan app whether an ext exists?
15:10.30*** join/#asterisk SplasPood (n=jwb@206.252.198.100)
15:10.58X-Robpdunkel, I want to GotoIf (${exten_exists(1234)}?1234:fail)
15:11.11Hmmhesaysdamnit
15:11.20pdunkelIf that's what you mean, you coud do a TryExec(Goto(<context>,<exten>,<priority>) as of the current svn trunk
15:11.44X-Robpdunkel, bugger. Can't use trunk yet, as I want this to work in 1.2
15:11.47pdunkelAfter that you can check ${TRYSTATUS} for FAILED or SUCCESS
15:12.23tzangerX-Rob: offhand, why do you need this?
15:12.28pdunkel(This is something I submitted) I can giv you the patch. I have it working against 1.2.4 and 1.2.6 without a problem!
15:12.39X-Robtzanger, routing based on Zap channel names.
15:12.51X-RobI've got it _almost_ working.
15:13.04Hmmhesayshow could someone waste a good domain name like lostpacket.org
15:14.24X-RobThe issue is, I have to do it in a macro.........ooh, you know, I might NOT have to
15:14.35X-Robno
15:14.36X-Robbugger
15:14.37X-RobI do.
15:14.58X-RobI don't want to have to set all the zap channels to a different context
15:15.03pdunkelX-Rob: You want the replacement for app_exec.c?
15:15.20X-Robpdunkel, I can't change asterisk - we've got enough dependancies as it is 8)
15:15.35X-Rob(this is for freePBX 2.1)
15:15.57pdunkelX-Rob: Ahh. Well suit yourself :)
15:16.16RoyK~disclaimer?
15:16.17jbotI disclaim all of you!, or "fortune -m 'Void where'"
15:16.47RoyK~jbot?
15:16.48jbotsomebody said jbot was only marginally useful at best,  He got a C- on his Turing Test
15:17.07*** part/#asterisk sercz (n=serz@i3ED6F067.versanet.de)
15:17.22X-Robhow the smeg am I going to do this
15:17.54nokysyle, [TK]D-Fender: it's work!
15:18.00iDunnomagic.
15:18.01X-RobI think I might just go 'if you're routing via channel, then you can't route via Caller ID. Accpet this and move along'
15:18.28Hmmhesayswow godaddy is cheaper than registerfly
15:18.38[TK]D-Fendernoky : congratulations
15:19.02pdunkelX-Rob: The problem is that Goto returns -1 if it can't find the exten. This tells * to stop dialplan execution. There really is no way aound this. I was laboring over the same problem. That's why I came up with this new app.
15:19.11sylegoddaddy is run in USA though so they shut down domains as they see fit
15:19.27syleoffshore is abit more secure
15:19.32X-RobHmmhesays, you're not _serously_ considering using godaddy after what they did to me, are you?
15:20.20syledid they make you their bitch?
15:20.27X-Robthey fucked my arse well.
15:20.41sylelubed or 12 inches dry?
15:20.48X-Rob'We've got all your data, and all your backups. Fuck off. Any questions? email legal@godaddy.com'
15:21.15syleoww you hosted with them
15:21.18X-RobYup.
15:21.33sylei don;t think i;d ever host without colocated box
15:21.37pdunkelThere probably are ladied present. goto #inappropriate or #fun for some #fucking :)
15:21.49nokythanks [TK]D-Fender & syle !!!
15:21.52noky:D
15:22.38*** join/#asterisk salviadud (n=ralfalfa@201.135.13.124)
15:22.51pdunkel*pdunkel just slipped on it :)
15:23.01*** join/#asterisk timscott (n=a@d198-53-19-216.abhsia.telus.net)
15:23.06key2what is the name of the zaptel monitoring tool ?
15:23.40sylecd /usr/src/zaptel
15:23.48syleall your binaries are there
15:23.56a1faall your binz belong to us
15:24.04mitcheloc* are belong to us
15:24.28Kattywhat a bunch of oddballs.
15:24.30mitcheloca1fa: personally, i think the channel logo topic should be "all your pbx are belong to us"
15:24.41a1faKatty, hi sweetie
15:24.52a1fahehe
15:24.54a1fathen change it
15:25.01syleKattie likes being center of attention so tell her how nice her tits look today :)
15:25.04a1faanybody experiencing odd issues with 1.2.6 and sip?
15:25.13a1famy pbx picks up calls on random
15:25.13Kattywow, syle really doesn't know me
15:25.20Kattythat's kinda sad.
15:25.29salviaduddon't be dissin' on Kat
15:25.36a1fasalviadud : i love me some katty
15:25.49pdunkela1fa: Yes, I keep having open channels that never go away.
15:25.49salviadudshe's funny
15:26.02sylesalviadud how much do you like that cock sucking you do :)
15:26.03Kattyi just randomly drop by and say something and disappear again
15:26.03salviadudshe's got some funny pics in her site
15:26.04a1fashe is hot
15:26.09a1fakatty is hot
15:26.11a1fawoom
15:26.15*** join/#asterisk klasstek (n=nunyobiz@sta-206-168-96-67.rockynet.com)
15:26.16X-RobSigh.
15:26.17Kattyoh for goodness sake
15:26.19Kattygrow up
15:26.22a1fahaha
15:26.27a1faeverbody msg katty -> asl
15:26.30Kattyi am not a slab of meat :P
15:26.38salviadudsyle, i'm not gay...
15:26.38syleof course you are, get use to it
15:26.41a1fai believe katty has enum too :) i am sure we can peer to her :P
15:28.03sylesalviadud thats to bad, i know a guy that was asking about you
15:28.54salviadudriiiiight
15:29.31*** join/#asterisk nitram (i=foo@superblob.com) [NETSPLIT VICTIM]
15:32.33Lino`oh
15:32.44DoktorGreg<PROTECTED>
15:32.52DoktorGregmy pri card gets here tomorrow
15:33.22a1fagood for you gay
15:33.25a1fa:)
15:33.27a1faehhe
15:34.22a1faanybody using 1.2.6 and broadvoice?
15:34.47[TK]D-Fendera1fa : A company I consult at.
15:34.52*** join/#asterisk Kokey (n=jramirez@dsl-200-78-65-27.prod-infinitum.com.mx)
15:35.03sylei;m surprised their still around
15:35.10a1fa[TK]D-Fender : hehe.. asterisk wont pickup inbound calls
15:35.15*** join/#asterisk tdonahue-laptop (n=tdonahue@208.51.101.201)
15:35.16a1fait worked fine in 1.2.5
15:35.25a1fai see a call come in, it rings, and it rings out
15:35.29a1faSIP Debugging Enabled for IP: 147.135.0.128
15:35.33a1fanothing in the log file
15:35.44a1faor debug
15:36.05a1faExpiry for sip.broadvoice.com is 3599 sec (Scheduling reregistra                        tion in 3584 s)
15:36.12a1fathat is on sip reload
15:36.23[TK]D-Fendera1fa : news to me...
15:36.26a1faok
15:36.29a1fait picked up this time
15:36.33a1faafter i did sip reload
15:37.22a1facanreinvite=no
15:37.28a1faright?
15:37.42salviadudyou guys think linking vonage to asterisk is a pain, or a breeze?
15:38.22[TK]D-Fendera1fa : re-invites = evil
15:38.31sylei know vonage is scared of AT&T
15:38.42sylethey dropped their rates 5 bucks after they did heh
15:38.55cj-rmHow many people can asterisk realistically support in a single conference call?
15:39.01[TK]D-Fendersalviadud :well you can't do it direct, you'd need an FXO channel for it... so "pain" it is
15:39.12shiznatixis there any reason why when i make a call through a zapata device to a regular phone line that my calls would go to a random phone number about 60% of the time?
15:39.29key2how comes my x100p can Dial but can't detect when the line is ringing, any idea ?
15:39.38Kattyit's deaf.
15:39.39*** join/#asterisk Letron (n=asd@216.94.46.194)
15:39.47[TK]D-FenderKatty : lol
15:40.07pdunkelshiznatix: That shoul make you popular in you neighborhood. I hope you don't develop during night-time!
15:40.40Katty[TK]D-Fender: lolerskater.
15:40.43*** join/#asterisk Ansonmus (n=ahaeser@dsl97-13-100.fastxdsl.nl)
15:40.56cj-rm[TK]D-Fender: Did you find out what was up with my dialplan for channel groups?  I'm sorry but I got dragged away by some other work...
15:40.58Ansonmushello, do anyone have experioence with Express Talk?
15:41.01shiznatixpdunkel, hahahahaha. so far i have gotten 3 different people and im like uhhhh sorry my PBX is sending out random singlas....
15:41.23*** join/#asterisk nitram (i=foo@superblob.com) [NETSPLIT VICTIM]
15:41.23[TK]D-Fenderkey2 : Do you have a context defined in zapata.conf and properly matching in extensions.conf for that channel?
15:41.56pdunkelshiznatix: What channels are you using?
15:42.21*** join/#asterisk justinu|laptop (n=Justin@wirelessdata-031-029.mycingular.net)
15:42.38stoffelloh boy, Ukraine does not "certify" asterisk/digium equipment ??!
15:42.42*** part/#asterisk justinu|laptop (n=Justin@wirelessdata-031-029.mycingular.net)
15:42.55a1fa* > Ukraine
15:42.57a1falol
15:43.12pdunkelstoffel: ? Where's that from? Ukraine? So?
15:43.24salviadudare there sexy girls in ukraine?
15:43.35*** join/#asterisk AlexCTI (n=alex@68-66-149-78.miamfl.adelphia.net)
15:43.54stoffellpdunkel, a company doesn't want to use asterisk because they don't have a certificate for digium equipment on the telco's in ukraine
15:43.58pdunkelsalvidud: Yes there are. I guess I'll have to video call them via an * in Europe then :)
15:44.41Kattythere are pretty girls everywhere.
15:44.44[TK]D-Fendercj-rm : Really not sure...
15:45.28pdunkelstoffel: Well, I guess that makes that company not very smart. I also guess the don't want to use */digium for other reasons and use Ukraine as an excuse. Sort of like me saying I'm not using asterisk because it's not ceritfied by (me).
15:46.22cj-rm[TK]D-Fender: It's really strange, it looks like when I Dial(Zap/g1) that it somehow triggers the same internal extension that is running.
15:46.28pdunkelshiznatix: Still want me to try and help with your dialout problem?
15:46.34zoalike anyone in the ukraine cares about certificates
15:46.44pdunkelshiznatix: What channels?
15:46.58key2pdunkel: well even if I didn't have, I would see something in the asterisk console ?
15:47.10stoffellzoa, that customer does :(
15:47.11salviadudwell, there are pretty girls here, yet, i don't like their style
15:47.21salviadudi want me a black gurl!
15:47.25pdunkelkey2?
15:47.30salviadudor something exotic, from the amazon
15:47.40salviaduda girl that likes to talk on the phone
15:47.50DoktorGregWhat does the sendText command do?
15:47.51salviadudand have phone with my wacky ivr systems
15:47.57*** join/#asterisk Darwin35 (n=Darwin@c-24-9-75-234.hsd1.co.comcast.net)
15:47.59*** join/#asterisk amonroy (n=chatzill@wireless-121.media.mit.edu)
15:48.04salviadudi meant, have fun
15:48.10salviadud... damn, i'm sleepy
15:48.12key2pdunkel: well if I ring the line, I would see something in the asterisk console no ?
15:48.27pdunkelsaliva dud: mabe you should try the real thing some time. It is much more fun than on the phone!
15:48.44DoktorGregwill sendtext work with ISDN phones?
15:48.46pdunkelkey2: That depends on verbosity.
15:49.14*** join/#asterisk rollot (n=rollotom@c-68-32-33-163.hsd1.pa.comcast.net)
15:49.32[TK]D-Fenderkey2 : pastebin your zapata.conf and extensions.con
15:49.42pdunkelkey2 == shiznatix ?
15:50.06salviadudpdunkel, of course i've tried the real thing
15:50.13salviadudlast night, i was doctor gonzo
15:50.13*** join/#asterisk lecter___ (n=lecter__@200.218.192.10)
15:50.16*** join/#asterisk leonk24 (n=adf@200.62.141.190)
15:50.38pdunkelsaliva dud: You sure? And you still want phones**?
15:50.51shiznatixpdunkel, sorry, hold on let me try something real quick
15:50.59leonk24Hi... somebody from Peru? I have a problem with circuit Telmex and TE110P
15:51.00salviadudhey, i like to talk on the phone too
15:51.13lecter___Hi fellows. I'd like to know if ASTERISK can listen SIP messages at 5060 TCP. Is it possible? Or just UDP?
15:51.24zoaonly udp
15:51.27zoaNEXT!
15:52.31*** join/#asterisk nitram (i=foo@superblob.com) [NETSPLIT VICTIM]
15:52.54rollot<PROTECTED>
15:53.15*** join/#asterisk ctooley (n=ctooley@rrcs-24-227-212-162.sw.biz.rr.com)
15:53.33lecter___second question: When I select TCP transport at a SIP device (I tried with Eyebeam 1.1), it just insert the field 'transport=TCP' in the headers but it sends using UDP. Is it right?
15:54.05rollot<PROTECTED>
15:55.10nokycan i avoid to proxy rtp of asterisk ? ... i like that rtp will be between sipA & sipB directly... could be or asterisk have a problem with this or isn't implement of this form ?
15:55.39ctooleyexten => _NXXNXXNXXX,1,Set,CDR(accountcode="14 - 2352").  I thought that was right, but it's not... what am I doing wrong?
15:56.54*** join/#asterisk skkip (n=Skipper@216.160.91.91)
15:57.14*** part/#asterisk pdunkel (n=pdunkel@213.235.231.189)
15:58.09lecter___there are a ',' after set
15:58.17[TK]D-Fenderctooley : exten => _NXXNXXNXXX,1,Set(CDR(accountcode)=14 - 2352)
15:58.31ctooley[TK]D-Fender, finally found the wiki page that says that.
15:58.33ctooleythanks
15:59.33leonk24Hi ... help me please with configuration for E1 Peru with Telmex.
16:00.18leonk24I configured span with 1,1,0,ccs,hdb3,crc4 and load modules in the kernel but the led is red.
16:00.55nokyis not possible that ast* disable the forward rtp between SIP_A & SIP_B ?... else ast* must control the signalling between SIP_A & SIP_B ?
16:02.42shiznatixOK! I have finally started getting much closer to getting faxes to work but with spandsp it is not really working
16:03.10shiznatixAsterisk understands its a fax, starts saving the fax, but then says that it was complete but there is no file saved anywhere
16:03.40X-Robpdunkel, end result is, I left the code in there, commented out, and went 'This will be fixed when asterisk 1.4 is released' as a comment. Also hinted that TryExec will be the trick. Thanks 8)
16:04.00*** join/#asterisk spunz_ (n=spunz@h081217096096.dyn.cm.kabsi.at)
16:04.13noky"The audio channels (RTP) may go directly from phone to phone or may go through Asterisk's media bridge. "
16:04.14cybergypsyi have an X100 card working for incoming calls , how do i dial( using it ?
16:04.29nokyoh nice... where can i configurate that rtp may go between phones ?
16:04.32backbluecybergypsy: dial (Zap/1) ?
16:04.33salviadudasterisk 1.4?
16:05.07salviadudX-Rob, you workin' pretty close with the main Devs on the project?
16:06.11nokywhat should I do to make audio channels to go directly from phone to phone
16:06.21cybergypsythanks backble - i tried that and it gave me an error , it seemed to obvious
16:06.54cj-rmWhat does /n do to the Local channel???
16:06.58*** join/#asterisk X-Rob (n=rob-x@dsl-202-173-151-24.qld.westnet.com.au)
16:07.59*** part/#asterisk hensema (n=erik@scrat.hensema.net)
16:08.03rollot<PROTECTED>
16:11.57cj-rmDoes the local channel work properly with Zap channel groups?
16:12.01*** join/#asterisk X-Rob (n=rob-x@dsl-202-173-151-24.qld.westnet.com.au)
16:15.17*** join/#asterisk Gamercjm (n=chris@pool-71-254-175-120.lsanca.fios.verizon.net)
16:15.30*** join/#asterisk greendisease (n=jack@fedora/greendisease)
16:15.45*** join/#asterisk mjackson (n=mjackson@69.85.202.2)
16:15.54mjacksonhowdee howdee
16:16.57salviadudhowdee partner
16:16.59mjacksonThank you for calling #asterisk.  Your call is very important to us.  Please stay on the line.  Your call may be recorded for quality control purposes.
16:17.25[TK]D-Fendermjackson : No, probably more for future public ridicule ;)
16:17.29salviaduddamn, i a called a friend at work
16:17.44salviadudi was all uppity, and the guy was like "dude.... i'm at work"
16:18.03salviadudman, i felt the coldness
16:18.38salviadudwhere i work, it's madness, so i can fun at work... i've never really had a serious job in my life
16:18.56salviadudi mean, have fun
16:19.07mjacksonYou at work now? :-)
16:19.12salviadudyeah :)
16:19.30mjackson^^
16:19.30*** join/#asterisk nitram (i=foo@superblob.com) [NETSPLIT VICTIM]
16:19.36*** join/#asterisk austinnichols102 (n=austinni@70.46.69.131)
16:19.41mjacksonAnybody got a preference between gnudialer and vididial?
16:20.11mitchelocmjackson: can you link me to vididial?
16:20.31mjacksonI just got done using festival to set up a new submenu in the menu system... it's I wonder if anybody will notice the 'accidental' studdering the robot voice does
16:20.43salviadudi'm like "hey mr. boss man, i'm gonna get us a toll free number service so we can call those hotlines" he'll go something like this "COUGH!!!@@1!umm... yeah, whatever, you do what you gotta do"
16:20.46*** part/#asterisk leonk24 (n=adf@200.62.141.190)
16:20.54*** join/#asterisk leonk24 (n=adf@200.62.141.190)
16:21.12shiznatixAlright im back! My Zap device is dialling a wrong number about 40% of the time can anyone help me out?
16:21.34salviadudsome dude was playing with festival the other day. he made it go "domo origato, mr. roboto"
16:21.50salviadudit sounds kinda funky
16:21.55mjacksonLoL
16:23.26mjacksonack... he says it all wrong... would take some tweaking
16:23.53salviadudyeah, nothing like fine tuning the instruments we use
16:23.54austinnichols102festival is excellent for the stephen hawking voice
16:23.58austinnichols102"Einstein was very unhappy about this apparent randomness in nature. His views were summed up in his famous phrase, 'God does not play dice'. "
16:24.00*** join/#asterisk Cybertoy (n=maxim@dsl254-123-241.nyc1.dsl.speakeasy.net)
16:24.30cj-rmHas anyone here looked at getting Nuances ( http://www.nuance.com/ ) excellent tts voices working with asterisk?
16:24.32mjacksonmchawking.com
16:24.39mjacksonhawking's voice makes great rap music
16:24.43austinnichols102cepstral diane is excellent
16:24.44cj-rmThey've got a good interactive demos on their site.
16:25.23cj-rmThe British Male voice is particularly good... very much the BBC newscaster.
16:26.24*** join/#asterisk Qwell[] (i=north@unaffiliated/qwell)
16:26.29*** join/#asterisk MooingLemur (n=troy@shells200.pinchaser.com)
16:27.12*** join/#asterisk pdunkel (n=pdunkel@213.235.192.27)
16:27.34austinnichols102mjackson: nice link for mchawking.com
16:27.59[TK]D-FenderCepstral seems an affordable TTS solution...
16:28.02salviadudi'm checking the faq out
16:28.07salviadudvery funny stuff
16:28.09[TK]D-FenderIn as much as I'd like "free"
16:29.07*** part/#asterisk rollot (n=rollotom@c-68-32-33-163.hsd1.pa.comcast.net)
16:32.08mjacksonAnybody got some advice for somebody about to set up their first predictive dialer center?  40 people to start, growing to 200?
16:35.21xachenwith Asterisk?
16:35.29mjacksonyah
16:35.33mjacksoni think we're going to use gnudialer
16:35.37*** join/#asterisk sevard (i=sev@24-179-181-160.dhcp.dlth.mn.charter.com)
16:38.54mjacksongnudialer is an asterisk addon :-P
16:40.56docelm0mjackson, check out vicidial..
16:41.02docelm0I know the author..  Good guy
16:41.22op3rmjackson: vicidial is cool
16:41.51docelm0He lives about 20 miles from me
16:42.32cj-rmmjackson: My advice is dont... The world could use fewer cold calls.
16:43.10docelm0cj-rm, true..  but gotta make money somehow
16:43.25op3rhahahahaha
16:44.10cj-rmdocelm0: So do drug dealers, pimps and the mafia...
16:44.32*** join/#asterisk spunz_ (n=spunz@h081217096096.dyn.cm.kabsi.at) [NETSPLIT VICTIM]
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16:44.32*** join/#asterisk [av]bani (n=[av]bani@washuu.anime.net) [NETSPLIT VICTIM]
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16:45.37Hmmhesaysanyone familiar with MS's dns server?
16:46.05Hmmhesaysyeah, i have no choice in this matter
16:46.14Hmmhesaysso save your sarcasm
16:46.32mjacksonlol
16:46.34salviadudlol
16:46.55Hmmhesaysyes we're all comics
16:46.59xachencall centers... peh :(
16:47.24xachenI don't care if you want to sell me a $2000 Mr. Cosmo vacuum when I can get a just as good one from Sears for $200 :P
16:47.54sevardxachen: telemarket trap script
16:47.55mitchelocHmmhesays: use dnsmadeeasy ;)
16:48.10xachenheh
16:48.27xachenyeah when I get my new system up it'll be knocking off telemarketers
16:48.51sevard<PROTECTED>
16:49.12Hmmhesaysi just have a simple question on how ms sets up their A records, because I can ping this www.foo.bar  but not foo.bar
16:49.49Hmmhesaysin bind this is simple, in ms... not so much
16:49.52docelm0Hmmhesays, ya
16:49.57docelm0What do you wanna know?
16:50.04docelm0I use it for my primary DNS
16:50.14Hmmhesaysms?
16:50.18docelm0yep
16:50.20docelm0I use active directory
16:50.28docelm0No choice..  Was in place when I got here
16:50.29Hmmhesayshow do I make foo.bar return the ip
16:50.34docelm0@
16:50.58docelm0or setup an A record with foo.bar.  point to the IP
16:51.08NirSany digium wizards around here ?
16:51.08docelm0make sure you use the . at the end of the same
16:51.08Hmmhesayswon't let me add it in the mmc
16:51.50docelm0Your wanting to resolve foo.bar to an ip..    Under Zone and then the name tell it you want to add a new A record with foo.bar.  as the name and your IP
16:52.01Hmmhesaysoohhh gotcha
16:52.03iCEBrkr*** glibc detected *** corrupted double-linked list: 0x08259650 ***
16:52.03iCEBrkrAborted
16:52.14docelm0haha
16:52.37iCEBrkrFind.. I'll build 1.2.6
16:52.38iCEBrkrgeesh
16:52.39xachenhaha
16:52.48xachenasterisk is evil to compile
16:52.53docelm0#0  0x008b5911 in ____strtod_l_internal () from /lib/tls/libc.so.6
16:52.53docelm0#1  0x008b2f68 in __strtod_internal () from /lib/tls/libc.so.6
16:52.53docelm0#2  0x008aefcd in atof () from /lib/tls/libc.so.6
16:52.53docelm0#3  0x006f1f43 in ?? () from /usr/lib/asterisk/modules/cdr_addon_mysql.so
16:52.57docelm0explain that one
16:52.58docelm0:P
16:53.23Hmmhesayswtf, the new host section of the mmc won't let me enter a  .
16:53.38docelm0Hmmhesays, Im a 2000 and 2003 MCSE
16:53.54docelm0You cant enter a .   you enter:   foo.bar.
16:53.59*** join/#asterisk timscott (n=a@d198-53-19-216.abhsia.telus.net)
16:54.32Hmmhesaysfoo.bar. yes  it comes out  foobar
16:54.33docelm0atof..  shit.. I know where the problem is..  DAMNIT!
16:54.51docelm0um, hmmm
16:55.03docelm0hold on lemme get into my DNS
16:56.05docelm0ok load up the DNS manager..  Then click on Forward Lookup Zones then the domain name.   Once you click on the name
16:56.42Hmmhesaysis this something different than what i'm using? i just got the dns mmc up
16:57.24docelm0no..  DNS MMC
16:57.31Hmmhesaysi got a window with the name/mail/sub domains..
16:57.33docelm0Tell it to add a new host..   A record
16:57.44docelm0What version windows are you using?
16:57.46Hmmhesaysright click, new host
16:58.03Hmmhesays2000 server sp4
16:58.22docelm0When you add the new host leave "NAME" bank the first field..  2nd will be greyed out and enter your IP in the 3rd..
16:58.34docelm0ya..  same MMC plugin then
16:59.01mjacksonyou don't have to use the mmc
16:59.09mjacksoni think you can directly edit your zone files on an ms dns server
16:59.14mjacksonand they're standard format zone files
16:59.19docelm0NOPE!
16:59.25docelm0You have to use the GUI
16:59.32mjacksonLoL
16:59.48docelm0And they are not standard format..  M$HIT once again has their own way to do it.
17:00.05Hmmhesaysyeah that worked docelm0, thanks a bunch
17:00.15docelm0No prob dude..
17:00.42docelm0What can I say Im a well rounded engineer..   Sun, Linux, WinDOZ, networking, cisco, asterisk, etc..   I could go on FOREVER
17:00.48mjacksonhttp://en.wikipedia.org/wiki/Microsoft_DNS  DNS Data can be stored in zone files, or in Active Directory.  MS DNS can be administrated via a GUI, the MMC, or a command line interface dnscmd
17:01.20docelm0mjackson, read the last thing is says..  dnscmd from the CLI
17:01.27docelm0you cant not directly edit the files
17:01.30mjackson*nod*
17:01.31docelm0its a BAD ideas
17:01.32sevarddocelm0: I want to hear you go on forever.
17:01.33docelm0err idea
17:01.36mjacksonright... if it's stored in active directory
17:01.57docelm0I never store my in AD..  too much headache
17:02.00mjacksoni bet each method handles editing the information differently... and each has it's own quirks.  Yuck.
17:02.02docelm0anywho..  going to lunch.
17:02.23mjacksonyay lunchtime!
17:03.08*** join/#asterisk FlyboySR22 (n=rsears@sdtc.ar01.f2-40.host2.1.americanis.net)
17:04.39iCEBrkrdocelm0: you're well 'rounded' alright.
17:04.43sevardi ate my lunch at 11 :/
17:05.42timscottit's 11 right now. :/
17:05.45timscottMST
17:05.46timscott:/
17:06.39*** join/#asterisk spatulamaan (n=ggilmore@ip66-107-33-196.z33-107-66.customer.algx.net)
17:09.08Hmmhesaysi need a cigarette
17:09.23*** join/#asterisk Nix (n=Nix@81.214.255.57)
17:18.18zafAD-integrated DNS doesn't work too bad
17:23.21*** join/#asterisk [hC] (n=hardcore@mail.metrobridge.com)
17:26.37Hmmhesayswow trying to troubleshoot ser with techs that don't know sh1@t it rough
17:31.23*** join/#asterisk nite (n=nite@gateway.digium.com)
17:31.39timscottser?
17:32.03timscottlooks like the techs aren't the only ones who don't know anything about it...
17:32.04timscott:S
17:32.43*** join/#asterisk gmonxx (n=gg@65.172.4.34)
17:33.24Qwell[]timscott: What's wrong with SER?
17:34.13gmonxxanyone know why my voice volume is really low when making an outside call?
17:35.19timscottQwell: I don't know. I don't know _what_it_is_.
17:35.21timscott:/
17:37.02*** join/#asterisk nettie (i=esivieri@85-18-54-38.ip.fastwebnet.it)
17:37.07[av]bani~ser
17:37.13jbotmethinks ser is Sip Express Router - see http://www.iptel.org/ser/
17:38.39*** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com)
17:40.50*** join/#asterisk brif8 (n=chatzill@rrcs-71-41-50-162.se.biz.rr.com)
17:41.58*** join/#asterisk klasstek (n=nunyobiz@sta-206-168-96-67.rockynet.com)
17:44.35a1fai need to pee
17:44.46Qwell[]good to know
17:44.52*** join/#asterisk Dandan (i=dandan@jestem.lama.ale.mam.super.konto.na.pacanka.com)
17:44.54Dandanhey all
17:45.50a1fathankz qwell
17:45.53docelm0Hay TIM uhh screw you there bub
17:46.06Dandani have a question: I will be getting a PRI, what card would you recommend?
17:46.16Dandan(I decided to drop 15 copper lines...)
17:46.30Qwell[]Dandan: a single PRI?  Digium TE110P or TE105P
17:46.50mog_workwe have no 105...........
17:46.51DandanQwell: cool, no problems whatsoever?
17:46.56Qwell[]oh
17:47.00Dandani decided to drop voicetronix
17:47.09Qwell[]well, I just made that one up then :p
17:47.10Dandancouldn't get it to work one way or another...
17:47.11[av]baniyay software EC
17:47.25Qwell[]mog_work: Do you only have one voltage for the 1 port?
17:47.38mog_workrigth it does 3.3 or 5 for that board
17:47.47Qwell[]ahh, okay
17:47.48Dandananyone heard anything good/bad about CTC Comm as providers?
17:48.04Qwell[]So it's the TE110P
17:48.15Qwell[]mog_work: Why don't the other boards do both?
17:48.46*** join/#asterisk jeffik (n=Jeff@208-41-192-106.client.dsl.net)
17:48.57mog_workxylinx thing
17:49.01Qwell[]oh
17:49.07Qwell[]silly xylinx :D
17:49.10wunderkin:(
17:49.12mog_workman i just butchered that word
17:49.21mog_worki think its xilinx but i have no clue
17:49.41Qwell[]yeah, it is
17:49.51Qwell[]per google
17:50.12a1fabrb
17:50.14*** part/#asterisk a1fa (n=a1fa@207.210.210.202)
17:51.58jeffikAnybody know how to access sipura 1001 remotely?
17:52.04Dandananyone has any experience with sangoma a101
17:52.06Dandan?
17:52.26brif8I have bought a cisco 7920 but they say without the CM license where can I just buy the CM license so I can get the cmterm file anyone >?
17:52.51*** join/#asterisk maffro (n=furby@n156.dkm.cz)
17:52.59maffroHi all
17:53.18*** join/#asterisk a1fa (n=a1fa@207.210.210.202)
17:53.24a1fals
17:53.24*** join/#asterisk oconnect (i=lukash@sip.pekelnik.net)
17:53.26a1fasudo su-
17:53.30a1far0#*@($#*$#R#RJ
17:53.36a1faomfg ;)
17:53.36*** join/#asterisk techman97_andy (n=me@70-98-31-249.dsl1.rsm.mn.frontiernet.net)
17:53.41a1fayou have my rewt password
17:54.03[TK]D-FenderDandan : All the Sangoma cards work the same.  I have an A104d
17:54.20a1fa[TK]D-Fender : can you make a decent living doing voip consulting?
17:54.24Qwell[]brif8: Any cisco reseller should be able to sell you a CM license...not that you need it
17:54.24Qwell[]You do need a smartnet contract though
17:54.30[TK]D-Fenderjeffik : Tried using a web-browser?
17:54.36Qwell[]a1fa: You can, sure
17:54.46a1faQwell[] : what planet?
17:54.47Dandan[TK]D-Fender: hey :)
17:54.50a1faor city?
17:54.55Dandani decided to return the voicetronix
17:55.08techman97_andyhello all again - my system accepts calls from a SIP provider, and I'd like there to be 1-2 rings before Asterisk picks up and dumps into my IVR.  I currently have a "wait(2)" in my extensions.conf file, but that's just dead noise.  Any idea on what commands I can use to provide ringtone to the caller?
17:55.10Dandani had SO many issues that it was rediculous
17:55.14Qwell[]a1fa: There are quite a few people who do it
17:55.16Dandanwent with CTC Comm and PRI
17:55.22Dandanso now I need a new card
17:55.28a1fatechman97_andy : yes
17:55.31[TK]D-FenderDandan :REALLY?! ;)
17:55.38Dandanand I came here for an advice
17:55.44a1faexten => s,2,Playtones(ring)
17:55.47[TK]D-FenderLooking for PRI this time are we?
17:55.52techman97_andya1fa:  cool - thanks!
17:55.52brif8Qwell[]:  How and where I've tried calls to cisco atacomm ingram micro I just want the cmt bin files
17:55.55a1faok
17:56.02a1fastupid * is not picking up a call again
17:56.05a1fawtf
17:56.21ruzalukash: ':)
17:56.26lukashruza: :)
17:56.27a1faok
17:56.29a1fawtffff
17:56.30a1fai am pissed off
17:56.34*** join/#asterisk noky (n=Noky@200.69.211.18)
17:56.35nokyhi
17:57.17brif8Qwell[]: can you recommend a reseller who can sell it to me now on line ?
17:57.26Dandanfenlander: oh, shshsh...
17:57.26Dandan[TK]D-Fender: yes :/
17:57.26Dandando not say "didn't I say so..."
17:57.26Dandan15 lines are a bit too many to interface with voicetronix...
17:57.30Dandan[TK]D-Fender: so there are no issues with IRQs/drivers under linux 2.6?
17:57.48docelm0MEW MEW MEW MEW MEW MEW MEW MEW MEW
17:57.54Qwell[]techman97_andy: Wait() should work...just don't Answer() beforehand
17:57.55a1fabroadvoice's sip proxies are as reliable as your friendly neighborhood drug dealer
17:57.56[TK]D-FenderDandan : With what?
17:58.02Dandansangoma
17:58.14[TK]D-FenderDandan : Not one case I can recount.
17:58.16sylenot reliable then hehe
17:58.22Dandana1fa: I had no problems with BV today...
17:58.31Dandanit was really bad friday but not today
17:58.33a1faDandan : what proxy?
17:58.35Qwell[]brif8: look on the wiki, for cisco smartnet
17:58.43noky"The audio channels (RTP) may go directly from phone to phone or may go through Asterisk's media bridge. "
17:58.45Dandanhm, do not recall, hold on :D
17:58.46Qwell[]it'll list a reseller or two
17:58.49jeffikD-Fender: yes, using extenal ip and set the port to 8022
17:58.54nokywhat do u trhink about this [TK]D-Fender ?
17:58.58nokythink*
17:59.00[TK]D-FenderDandan : Let me just say that unless your company is UBER-cheap (aka STUPID), get the A104d.. the hardware EC is entirely worth it.
17:59.02Dandan[TK]D-Fender: ty going to tell my purchasing dept to buy it for me
17:59.22a1faApr 11 17:59:10 NOTICE[11439]: chan_sip.c:9686 handle_response_register: Outbound Registration: Expiry for sip.broadvoice.com is 30 sec (Scheduling reregistration in 23 s)
17:59.25a1fai dont get it
17:59.26[TK]D-Fendernoky : Re-invites are EVIL.  <-
17:59.37a1fawhy is it re-registering every 30s
17:59.42a1fai told it to re-register every hour
17:59.44Dandan[TK]D-Fender: quad pris?
17:59.44a1fagod damn it
17:59.46techman97_andya1fa:  that did it - thanks again!
17:59.57a1fayeah, you are welcome
18:00.00HmmhesaysIIS is going to make me rip my eyeballs out
18:00.08[TK]D-FenderDandan : Yes, I know you don't need 4 ports, but thats the only size out now with the HWEC on it.
18:00.08a1faguys
18:00.13a1faheeeeelp
18:00.20[av]baniwait 12mo for sangoma to make a 2port ver
18:00.22Dandanhm
18:00.23a1faOutbound Registration: Expiry for sip.broadvoice.com is 30 sec
18:00.24[av]banisimple
18:00.26a1faits not
18:00.31Dandanwell, voicetronix was supposed to have EC...
18:00.37Dandanand all I was getting was ECHO :D
18:00.41[TK]D-Fender[av]bani : Should be sooner than that....
18:00.45nokymmm... i want to know if is posibble that Asterisk could not forward rtp...
18:00.51a1famaxexpirey=180
18:00.51a1fadefaultexpirey=160
18:00.52[av]banior buy an external sip gateway
18:01.02Dandana1fa: 147.135.20.128          sip.broadvoice.com
18:01.07a1fanyc
18:01.09a1fai got the same one
18:01.16nokyi want to my rtp goes between Phone A & Phone B... not Phone A => Asterisk => Phone B
18:01.18Dandanon choiceone as a provider...
18:01.32nokypossible**
18:01.36a1faok
18:01.37a1fawtf
18:01.42a1fawhy is it re-registering every 30s
18:01.48a1fawhen it is told to re-register every fucking hour
18:01.53[TK]D-Fendernoky : If you use "canreinvite=yes" on both sides of the call then they will reconnect directly to each other (usually a BAD idea).  So I suggest "canreinvite=no" as a global setting
18:02.00DoktorGregoh man
18:02.11DoktorGregsorry ive been deeply in identity crisis
18:02.24nokyok thanks [TK]D-Fender !
18:02.41DoktorGregBut apple has release a rdc that lets you drag and drop accross computers...
18:02.46[av]baniwhy is reinvite bad, unless you're in nat?
18:02.57a1fadoes anybody know why am I getting this stupid shit
18:02.59[av]banireinvite is the only way to scale to 1000's of extensions on a single box
18:03.04a1faOutbound Registration: Expiry for sip.broadvoice.com is 30
18:03.20Hmmhesaysturn off your monitor
18:03.31a1fa?
18:03.34Dandan[TK]D-Fender: thx, i will look into getting that 104d
18:03.34a1faHmmhesays : lol
18:03.45sevardwhere are you tftpd
18:04.11[av]baniDandan: the te1** cards are also very finicky about the motherboards they will work in, the 104d is guaranteed to work in anything
18:04.22[TK]D-Fender[av]bani : because typically phones don't end up getting public IP's.  I guess you could isolate your outside peers, but better to just let it flow through unless you're huge
18:04.29a1fa[TK]D-Fender : yo yo yo
18:04.35Dandan[av]bani: i have a dell sc430
18:04.37[av]bani[TK]D-Fender: if your extensions arent talking to the outside world...
18:04.39Dandanas my test platform
18:04.58[av]bani[TK]D-Fender: which is the typical case... your extensions always talk through the pbx, not directly to the intarweb
18:04.59[TK]D-Fender[av]bani : they do sometimes.
18:05.20[av]bani[TK]D-Fender: i would say 99% of the time that is not hte case... 99% of the time people are using * for internal pbx, not public intarweb pbx
18:05.21Hmmhesaysif you don't have a massive number of calls, who cares
18:05.35grem_linHey, could anyone comment on the suitability of the Zyxel Prestige 2002 for use in conjunction (or, indeed general use) with asterisk before I go and buy one ? :)
18:05.36[TK]D-FenderHmmhesays : My thoughts exactly...
18:05.48[TK]D-Fendergrem_lin : Bleh <-
18:05.56[TK]D-FenderWifi + SIP = shit today.
18:05.59noky[TK]D-Fender: thanks.
18:06.00[av]baniHmmhesays: if you're running on low end hardware (wrt54gs, gumstix), then you want reinvite
18:06.08a1fawifi + iax = future
18:06.13a1fa[TK]D-Fender : hey fender
18:06.14[av]baniHmmhesays: of course, you could wimp out and say "dont run asterisk on low end hardware", which is lame
18:06.32[TK]D-FenderDandan : the SC430 is on Digium's "shit list" IIRC with regards to several of their cards.
18:06.33[av]baniHmmhesays: think outside the box. it must get cramped in there.
18:06.42a1fagumstix * > any
18:06.54Dandan[TK]D-Fender: sh*t, you think I should get a sangoma then?
18:06.54[TK]D-Fenderiax + anything other than other IXA server = toy
18:07.03a1fahehe
18:07.03Dandan(I have to present a working asterisk server by the end of the month)
18:07.11grem_linOh don't say that :( I've spent ages trying to find the right piece of kit
18:07.16Dandanotherwise avaya here i come...
18:07.18a1fa[TK]D-Fender : hey man.. i am having an issue with re-registration.. it is expriging every 30s
18:07.30a1fa[TK]D-Fender : expirying
18:07.39a1faor howerver you spell expire + ing
18:07.42[TK]D-FenderCould be the OTHER side.... not you at all.. maybe they're assy today
18:07.57a1fa[TK]D-Fender : its been doing this since i updated to 1.2.6
18:08.12a1fa[TK]D-Fender : and they keep telling me they dont have that setting set to 30s
18:08.38a1fai know they do
18:08.45Dandanwell, recently asterisk's pace of development outpaces kernel's. Amazing. TVG train should dash for cover...
18:09.00a1fahandle_response_register: Outbound Registration: Expiry for sip.broadvoice.com is 30 sec
18:09.31[TK]D-FenderDandan : If the telco isn't stupid you can set a small company up in a day from scratch.
18:09.47[TK]D-FenderDandan : Pick good phones....
18:10.26a1faomfg
18:10.31a1fa[TK]D-Fender : you are right
18:10.36a1fait is coming from broadvoice
18:10.59Dandan[TK]D-Fender: i just need a foolproof setup, i grandfathered the sc430, now I need a pri card for it...
18:11.17[TK]D-Fenderrule of thumb : ITSP's suck.  ALL of them.  Just some more than others at varying intervals...
18:11.50[TK]D-FenderDandan : the SC430 is your first mistake.  PM your expected setup / needs / expectations.
18:13.02a1fahttp://pastebin.ca/49047
18:13.04a1fa[TK]D-Fender http://pastebin.ca/49047
18:13.07a1faSIP READ
18:13.16a1fais config coming down from the server.. rgith?
18:13.39a1faExpires: 30
18:13.44a1fagod damn fucking bastards
18:13.51a1fai fucking hate these motherfuckers
18:14.03a1fai am not able to recieve a call forever
18:14.07[TK]D-Fendera1fa : since you changed the names I presume the "from" was THEM, and the ""to" was YOU?
18:14.32a1fayes
18:14.36maffro000000000000.
18:14.40a1faFrom: was me
18:14.42a1fabrimstone: was me
18:14.48a1fa"To:" was me
18:15.23[TK]D-Fender:/
18:15.38[TK]D-Fenderwierd... not sure how to interpret that...
18:15.51a1fa:)
18:16.06a1faits crazy man
18:16.15a1faSIP READ FROM  147.135.20.128:5060:
18:16.28a1fathis means that i am getting data from them, right?
18:16.51a1fa"From: <sip:me@sip.broadvoice.com>"
18:16.55a1fa"To: <sip:me@sip.broadvoice.com>"
18:17.04a1fa"Expire: 30"
18:17.08a1fagod damn i hate these people
18:17.11[TK]D-Fendera1fa : I am thinking that'd be the case
18:17.32[TK]D-Fenderyou set "defaultexpiry" and all that for the peer?
18:17.34a1fai am on hold
18:17.46a1fadefaultexpiry=3600
18:18.16[TK]D-Fendera1fa : Global & peer?
18:18.27a1fapeer only
18:18.35*** part/#asterisk Hali_303 (n=surfk@dsl51B6E6EB.pool.t-online.hu)
18:18.58a1faglobal & peer now
18:19.02a1fadefaultexpiry=3600
18:19.10a1fai did sip reload
18:19.11a1fafirs message that came upo
18:19.21a1fa<PROTECTED>
18:19.52GamercjmDoes sox hangle ulaw?
18:19.59Gamercjmhandle*
18:20.14[TK]D-Fendera1fa : Fuck the fucking fuckers ;)
18:20.15Qwell[]Gamercjm: yes
18:20.17*** join/#asterisk gammacoder (n=chatzill@64-132-192-33.gen.twtelecom.net)
18:20.37a1fa[TK]D-Fender : those mother fuckers, i've been on hold for 30 minutes now
18:20.46a1fagod damn that fucking bastard and his call center
18:20.47wunderkindont bother
18:20.56pdunkelGamercjm: You might just as well type the words "sox" and "ulaw" into goolgle. Al the top posts tell you the answer.
18:21.40maffrohey folks, any of you enjoyed call recordings out of sync when using monitor to wav49 ?
18:23.09Dandana1fa: nothing new
18:23.14*** part/#asterisk Nix (n=Nix@81.214.255.57)
18:23.26*** join/#asterisk Lino` (n=Lino@i577BCF08.versanet.de)
18:23.28GamercjmWell i tried looking for it, but couldnt find the way its put into syntax "sox test.wav -r 8000 -c 1 -s -w test.ulaw resample -ql"
18:23.32skyboyhello has anyone used SER to load balance asterisk here? specifically what is the setup required to get it to go with srv records etc?
18:23.34Gamercjmdoes that look correct?
18:23.47a1faok
18:23.50a1fai got those fuckers on the line
18:24.24Beirdo976 number?
18:26.27[hC][TK]D-Fender: any chance i could get that minibrowser stuff from you today?
18:26.59a1fa[TK]D-Fender : he sais he doesnt have that setting
18:27.04a1faliar fucking bitch
18:27.10a1fai am going to do a tcpdump
18:27.16a1faand send him the binary log
18:27.19a1faso he can use ettercap
18:27.22*** join/#asterisk oej (n=oej@apollo.webway.se)
18:27.23a1faand jerk himself to death
18:28.04pdunkelGamercjm: Almost there is no such extension .ulaw. the one used by sox to tell it is the pseudo extension .lu
18:28.19*** join/#asterisk jpablo (n=jpablo@200.94.130.194)
18:28.39pdunkelGamercjm: This information was just obtained for you by the command "man sox" any more quesstions and the next response is RTFM!!!
18:29.00*** join/#asterisk kratzers (n=kratzers@martha.pa.net)
18:29.04jpablopeople I have a big problem. I have a PRI when i dial a number that does not exists it gives a normal busy tone instead of playing the telco's "the number you are trying o dial does not exists, etc", how can i fix that ?
18:29.43[TK]D-Fender[hC] : sure.
18:30.11[hC][TK]D-Fender: email work the best, or?
18:30.14nettiehey guys, I'm looking for a way to assign an extension number as CID for internal calls. I would like to have the extension displayed instead of the actual sip phone username. Anyone can suggestme something? I suppose I should use SetCallerID..
18:30.20jpabloany idea how can i fix that ?
18:30.40pdunkelnettie: That might work :)
18:30.49jpablothe same pri works fine with a panasonic pbx
18:31.11nettiepdunkel yeah the problem is that if I user SetCallerID({$MACRO_EXTEN})
18:31.17netties/user/use
18:31.31nettieit doesnt work
18:31.43nettieI get {$MACRO_EXTEN} displayed
18:31.51*** join/#asterisk x86 (n=x86@p3m/member/x86)
18:32.16pdunkelnettie: Aren't there new CallerID functions? Hold I'll check.
18:32.22kratzerstry ${MACRO_EXTEN}
18:32.53kratzersdollar sign then freedom brace
18:33.05pdunkeloej: Do you happen to know of hand? (If we have the honor of a guru present...)
18:33.12Katty[TK]D-Fender: have you seen clockwork orange?
18:33.18nettieand anyway I doubt it will help.. considering I would like to display the caller extension and not the called one
18:33.20[TK]D-FenderKatty : nope...
18:33.25Kattykk
18:33.32nettiekratzers I'm actually using curly
18:33.33[TK]D-Fender[hC] : PM me your e-mail addy
18:33.37nettieas you suggested
18:33.39nettiehey TK
18:34.19pdunkelnettie: http://www.voip-info.org/wiki/view/Asterisk+cmd+SetCIDNum details the new functions.
18:34.56nettiethanx a lot pdunkel checking...
18:34.58Dandanwhat is asterisk-addons-1.2.2-patch.gz ?
18:35.05kratzerssay I have two agents logged in, and both go on do not disturb... is there a way for calls to be queued instead of goign to voicemail boxes?
18:35.17pdunkelnettie: That should evaluate the ${MACRO_EXTEN} expression since Set usually does that.
18:35.33*** join/#asterisk Ariel_ (n=Ariel@adsl-068-157-125-248.sip.mia.bellsouth.net)
18:35.38*** join/#asterisk Luda1 (n=nechci@rb1j38.chello.upc.cz)
18:36.05*** join/#asterisk jimmy_deanPB (n=jhodapp@indianalifesciences.com)
18:36.38Luda1hi, pls i need register my asterisk to voipcheap, can anybody help me ?
18:37.01kratzersit seems that agents who are on the phone or are on DND should not be considered eligible to take calls
18:37.41malverianI have a question about CDR...
18:38.09*** join/#asterisk Dabian (n=M0RTEN@fsf/member/dabian)
18:38.27oejpdunkel: Know what?
18:38.33nettiepdunkel is does it.. the problem is that it sets the wrng parameters.. :( Executing Set("SIP/poly2-7260", "CALLERID(number)={$MACRO_EXTEN}") in new stack
18:39.33Dandan<PROTECTED>
18:39.34kratzerswhy is the curley brace before the dollar sign?
18:41.34[TK]D-Fenderkratzers : to prevent it from working of course!
18:41.44a1fa[TK]D-Fender : lol.. they opened a ticket with broadvoice
18:41.46a1faengineers
18:41.47kratzersah
18:41.51a1fastupid dump dumb fucks
18:41.59a1fasome fag changed expiry to 30
18:42.06a1fanow they cant change it back to 3600
18:42.26kratzersand by freedom brace in my previous answer, I meant the politically correct American version of french brace
18:43.22mitchelocis anyone subscribed to the asterisk mailing list? i'm wondering if my message didn't go through
18:43.37kratzersI am
18:43.38maffrokratzers: heh, I for one was wondering
18:44.07kratzersmaffro, it's a term my one Unix teacher used a few years back
18:44.14mitchelockratzers: could you check if a message from me went through (mitcheloc@gmail.com)
18:44.31*** join/#asterisk riddlebox (n=blah@24-171-40-167.dhcp.stls.mo.charter.com)
18:44.40[TK]D-Fender[14:42] <kratzers> and by freedom brace in my previous answer, I meant the politically correct American version of french brace <- since when is America "correct"? ;)
18:45.08kratzersmitcheloc: I don't see one
18:45.09maffro[TK]D-Fender: that comes after it is free
18:46.08kratzersso, why if there are two agents who are on calls and a new call comes in, is the new call sent to voicemail rather than queued?
18:46.14mitchelockratzers: hmm, is the list moderated? maybe they marked it as spam =/
18:46.16*** join/#asterisk Leob (n=chatzill@w2kvpn-22.media.mit.edu)
18:46.22kratzersdoesn't that defeat the purpose of queueing?
18:47.00kratzersmitcheloc: I'm not sure, when did you send it? Maybe delivery is delayed?
18:47.09mitchelocabout an hour and a half ago
18:47.31kratzershmm, try again maybe
18:47.43[TK]D-Fenderkratzers :  Using AgentCallbackLogin?
18:47.57kratzers[TX]D-Fender: yes
18:48.04kratzersbah
18:48.15kratzersstupid highlighting in irssi
18:48.46kratzerscan't read your name because it is in yellow agains white
18:48.52[TK]D-Fenderkratzers : then stop using a macro that answers the phone and dumps them to voicemail!~
18:49.02kratzershmm
18:49.05*** join/#asterisk oej (n=oej@apollo.webway.se)
18:50.10*** join/#asterisk oej (n=oej@apollo.webway.se)
18:50.40Luda1hi, pls i need register my asterisk to voipcheap, can anybody help me
18:50.45kratzersI'm not sure that I am
18:50.47[TK]D-FenderWhat are you talking about?
18:50.47kratzershttp://pastebin.com/654036
18:52.09[TK]D-Fenderkratzers : show us where agents loging and the context it dials into
18:52.13PakiPenguin[TK]D-Fender  go  the card link up atleast , but getting zaptel errors now
18:52.38PakiPenguinhttp://pastebin.com/653419
18:53.27*** part/#asterisk mitcheloc (n=mitchelo@70-32-188-167.lmdaca.adelphia.net)
18:53.30[TK]D-FenderPakiPenguin : the FXO stuff looks bad...
18:53.59PakiPenguinthe fxo is the astribank i have
18:54.00*** join/#asterisk nite (n=nite@gateway.digium.com)
18:54.16PakiPenguini mean fxo signaling
18:54.19kratzersrelevant stuff -> http://pastebin.com/654046
18:54.22PakiPenguinits fxs actuallly
18:54.24*** join/#asterisk MacDome (n=eseidel@c-71-198-177-144.hsd1.ca.comcast.net)
18:55.59*** join/#asterisk MacDome (n=eseidel@c-71-198-177-144.hsd1.ca.comcast.net)
18:58.19PakiPenguin[TK]D-Fender can you solve it ,o r point me where to solve it please
18:59.25timscottERGGG
18:59.26timscottNAT
18:59.30timscottI hate NAT.
18:59.33timscottSeriously.
19:01.51*** join/#asterisk terrapen (n=cjs@166.70.183.108)
19:01.57terrapenman, the SPA942 ROCKZZZ
19:02.52[hC]I have a question actually, for anyone with the spa941/942... do you hear double-ringing with it? I dont hear double ring with my ciscos but i do with these phones. (and no im not specifying ,r to Dial) - also some people claim that these phones are a bit echoey
19:03.04terrapendouble ringing?
19:03.23[hC]yeah, like you dial a number and rather than hearing one ring 'tone'
19:03.26[hC]you hear two, over lapped
19:03.28terrapennope
19:03.33[hC]Hmm.
19:03.41[hC]I have it happen on two completely different systems.
19:04.19terrapeni'm doingprogressinband=no
19:04.21terrapenerr
19:04.21terrapenprogressinband=no
19:04.28terrapentry that
19:04.41[hC]i just read that in mantis
19:04.42[hC]i'll try it.
19:04.51[hC]im surprised it would be set to yes as default..
19:05.30[TK]D-FenderPakiPenguin : Not sure about what to do with that...
19:05.55[TK]D-Fender[hC] : SPA-x4x = waste.
19:06.03[hC]hmm. that seems to have helped it.
19:06.25Leobhello there, can anyone help me with ODBC? Asterisk crashes all the time during VoiceMailMain...
19:06.52[TK]D-Fenderkratzers : Well you are using a macro with an answer in there.. that screws your queues... you need to make another means of dialing them so they don't answer...
19:07.29[TK]D-Fenderterrapen : Polycom IP501 can be had for less than the SPA-942 and is of superior usability & quality
19:07.43[hC]Yeah.
19:07.49[hC]I carry the 941/2 but i prefer the polycom
19:08.05[hC]if only i could get boot up time shortened, i'd be happy :)
19:08.17[hC]and this 7 line BLF resolved, we'd have a solid winner
19:08.17kratzershmm, I suppose I need to look at a complete set of sample configs using agents to see how it should be done
19:08.44*** join/#asterisk mrtwister|mobile (n=andrius@cable-10-68.cgates.lt)
19:08.53[TK]D-Fenderkratzers : you did it in a "right way", just one with consequences... you need to make a context for your agents that only Dial's them, and nothing more...
19:09.57[TK]D-Fender[hC] : then again, how often do you need to reboot them?  I did mine twice.  Once to pick-up the config & firmware coming fresh out of the box, and the other to go live :)
19:10.08kratzersand that's the one they login to?
19:10.17[hC][TK]D-Fender: power outages, etc..
19:10.20[hC]for some clients
19:10.30[TK]D-Fender[hC] : UPS & PoE :)
19:10.48terrapenTDK, I'm not seeing how the polycom is more usable
19:10.56terrapenthis has a backlit, very easy to read screen
19:10.57[hC][TK]D-Fender: like i said.... 'some' customers. :) the cheap ones who dont vote for ups+poe
19:11.01terrapensound quality is fine
19:11.09terrapenweb interface is incredible, if you need it
19:11.16*** join/#asterisk Nix (n=Nix@81.213.125.220)
19:11.17[hC]the 942 is backlit?
19:11.22terrapenyes
19:11.24terrapenwhite-lit
19:11.28[hC]i only have 941's
19:11.50terrapenoh, and the Linksys boots in less than 7 seconds
19:11.56terrapencompare to the Polycom's 2 minutes+
19:12.10brad_msswhuh, the 942 is backlit ?? really?
19:12.11terrapenI've loved polycom but really, I think Linksys has them beat on this one
19:12.14brad_msswcrap, we just bought 941s
19:12.16terrapenbrad, heck yeah, it's sweet
19:12.23terrapenits a white-lit LCD
19:12.27Luda1hi, pls i need register my asterisk to voipcheap, can anybody help me  ???
19:12.34terrapenbrad, but it's not as cool as my new mountain bike ;-)
19:12.37[hC]the 941 has a half-ass speakerphone too.
19:12.45[hC]the polycom beats it on speakerphone hands down
19:12.49brad_msswterrapen: how is it that I didn't see that on the 942 spec sheet
19:12.52h3x0rthe 941 is god awful
19:13.03h3x0rthe sound quality is like a chinese phone
19:13.04*** join/#asterisk PakiPenguin (n=uppal@linuxpakistan/admin/pakipenguin)
19:13.05terrapenhc, the speakerphone on the 942 is not the best evar but it's fine for desk use
19:13.13terrapenI may buy some 501's for the conference rooms
19:13.16h3x0ron the handset im talking about even
19:13.25[TK]D-Fenderterrapen : Polycom uses the LCD more effectively, is larger and more readable under proper lighting.
19:13.31[hC]the sound quality could be better, for sure.
19:13.33terrapenthe sound quality on the handset for the 942 is great
19:13.43Shaun2222where can i find a list of valid tone's for a dialplan on a 7960 cisco phone?
19:13.45[hC]it sounds like they may have improved the phone alot w/ the 942
19:14.04[TK]D-Fenderh3x0r  I wouldn't say the 941 is "bad", just not comparable to something similarly priced.
19:14.06terrapenTK, dunno, I have them side-by-side right in front of me.  This is the 942 I'm talking about
19:14.17terrapenBesides, the polycom's minibrowser sucks
19:14.20kratzersthink it's working now, but I'll have to play with it more tomorrow
19:14.22kratzersthanks
19:14.29terrapenTK, I'm starting to think that the 942 is much-improved over the 941
19:14.33[hC]terrapen: the linksys doesnt even have a minibrowser. does it?
19:14.40terrapenyeah, I think it does.
19:14.43terrapenthis one does.
19:14.46[hC]hm.
19:15.03[TK]D-Fenderterrapen : But costing more than the 501 really kills it... and I doubt the display is any better utilized.  backlight IS a nice bonus though....
19:15.04[hC]the 941 has 'directory'
19:15.04[hC]but thats it
19:15.08terrapenwell, maybe not.  i'm not sure
19:15.12[hC]haha
19:15.17[hC]the 941 also has an option in the menu
19:15.20[hC]'Call Foward'
19:15.21[TK]D-FenderNo, Linksys does NOT have a microbrowser
19:15.24terrapenhow much is the 501, tk
19:15.49[TK]D-Fender[hC] : Polys have forwarding as well, and can do it on a per-contact basis too
19:15.59[TK]D-Fenderterrapen : Seen for jsut under $170USD
19:16.10[hC]I was just making fun of how they missed the r in forward.
19:16.12[hC]'Foward'
19:16.16brad_msswterrapen: dunno, dude, can't find anything about backlights on the 942
19:16.23terrapenok, no minibrowwser
19:16.38terrapenbrad, i swear to you, it is.  :) I will take a picture tonight
19:16.48terrapenit also helps greatly in bright light situations
19:17.05brad_msswterrapen: yeah, i'd like to see it, these damn 941s can be hard to read
19:17.47*** join/#asterisk justinu|laptop (n=Justin@66.209.15.236)
19:17.59terrapeni used this phone last night in the dark office and it was great
19:18.17brad_msswthey come with the 2-port built-in switch as well, right ?
19:18.23terrapenyup, and PoE
19:18.32brad_mssweh, don't care about PoE
19:18.41terrapenwe just ordered a 48-port Foundry 10/100/1000 PoE switch today
19:18.46*** join/#asterisk brockj49464_ (n=brockj49@41.105.dhcp.hope.edu)
19:19.01[TK]D-FenderWell I hate the fact that linksys.com hides the entire line from public view... says something about them...
19:19.08[TK]D-Fenderterrapen : $?
19:19.11brad_msswterrapen: we use all HP Procurve switches here
19:19.18malverianI forgot to ask my question ;)
19:19.21brad_msswterrapen: btw, where did you buy your 942s ?
19:19.22[TK]D-Fenderterrapen : must be NASTY to have 1000 as well
19:19.27terrapenvoipsupply
19:19.38timscottDo any of you have suggestions regarding SIP and NAT? I'm trying to associate a phone inside of a NAT with a server inside of a different NAT.
19:20.10[TK]D-Fendertimscott : WIKI tells you what to do.
19:20.19timscottvoip-info.org, you mean?
19:20.26malverianIf I'm using call parking, what's the best way to make sure CDR is handled sanely? I have an extension (say 600) that is a call parking extension.. when a representative transfers a customer there, and then a different representative picks up the person from that parking spot, I want to see two separate CDR entries.
19:20.32malverianAny recommendations on how to accomplish this?
19:20.39terrapenI'm geting 10/100/1000 because I'm going to re-purpose this switch when our big Foundry chassises arrive this summer
19:21.04*** join/#asterisk oej (n=oej@apollo.webway.se)
19:21.12Leobcan anyone help me with ODBC?
19:21.22malverianCurrently, when I do this, all of the CDR variables are lost when the customer is transfered, so I lose the "dst", "src" etc variables.
19:21.24terrapentk, the 942's list at $199 but are cheaper in bulk
19:22.33*** join/#asterisk x86 (n=x86@p3m/member/x86)
19:23.05*** join/#asterisk timscott (n=a@d198-53-19-216.abhsia.telus.net)
19:23.24malverianSo I guess my question is.. how can I make CDR variables carry over across blind transfers?
19:23.54[TK]D-Fenderterrapen : Better off getting sperate switches and picking your jacks... because anywhere you will use PoE can't support GBIT at the same time... better off cross-conencting.  Keeps your cost MCUH lower
19:24.35*** join/#asterisk funxion (n=nunya@63.214.236.169)
19:27.35[TK]D-Fenderterrapen : lets say 25 phones.. how low could you go (in retail, no "extra" special pricing)
19:28.25[av]bani[TK]D-Fender: gbit works with poe
19:29.34[TK]D-Fender[av]bani : Never heard of it at the same time... PoE uses 2 pairs w/o data
19:30.01[TK]D-Fender[av]bani : and every device quoting it seems to say that GBIT only applies if its wall-warted
19:30.03[av]bani[TK]D-Fender: there's literally billions of gbe-poe products...
19:30.22[TK]D-Fender[av]bani : Got a decent link to one that well douments this ability?
19:30.26[av]banihttp://www.google.com/search?q=%2B%22gigabit%22+%2B%22poe%22&hl=en&lr=&start=10&sa=N
19:31.00Shaun2222what does this do? ignorepat => 9
19:31.11timscottyou press 9, it doesn't break the dialtone.
19:31.17Shaun2222ok
19:31.35*** join/#asterisk shido6 (n=shido6@d38-45-81.commercial1.cgocable.net)
19:31.36malverianI wish there was a better explanation of how CDR works..
19:32.00Hmmhesaysfreaking IIS
19:32.21[TK]D-Fender[av]bani : Can you link me a specific product... I hate the hunting...
19:32.26Shaun2222timscott: doesnt the phone send the dialtone? mine does i think, i dont see a connection to asterisk until it times out and attempts to dial or i press dial
19:32.42timscottit depends how you're using it, yeah.
19:32.53Shaun2222can that be changed?
19:32.57timscottif your phone is generating the dialtone, then it doesn't really matter if you hit "9" or not
19:33.08timscottbecause your phone is waiting for the whole string to be completed before sending the string
19:33.22timscottyeah, I guess, depending on your phone, there might be a setting for early dialing
19:33.28timscottor something similar
19:33.39Shaun2222i'm using the cisco 7960 phone, is their a better phone to use with asterisk... this phone seams kind of lame and it seams almost like asterisk and this phone lack the features of a real pbx.
19:33.46[av]bani[TK]D-Fender: http://www.eeproductcenter.com/powersources/brief/showArticle.jhtml?articleID=168602163
19:33.54timscottI'm not sure, you'd have to ask someone else about that phone
19:34.01timscottI'm not familier with cisco products, really.
19:34.03timscottsorry mate.
19:34.04[av]bani[TK]D-Fender: http://cs.pennnet.com/Articles/Article_Display.cfm?Section=ONART&cat=INDUS&p=42&ARTICLE_ID=225388&VERSION_NUM=1
19:34.17*** join/#asterisk greendisease (n=jack@fedora/greendisease)
19:34.19Shaun2222timscott: what type of phones do you use?
19:34.31h3x0rsha
19:34.38h3x0rsha: the 79xx series was designed to use with ccm
19:34.45*** part/#asterisk Grizzy (i=Generic@ppp-71-133-231-94.dsl.pltn13.pacbell.net)
19:35.03Shaun2222h3x0r: yes, is that the reason, is their a better phone?
19:35.05[av]bani[TK]D-Fender: http://www.powerdsine.com/news/prs/pr_030505_Gigabit_Midspan.html
19:35.14jsharpIf you're waiting for timeout to dial on your 7960s, then you don't have a proper dialplan set for it.
19:35.17timscottShaun: Grandstream GXP2000, and Snom190, and an SPA3000
19:35.36timscottjsharp: I think it was just an example.
19:35.40h3x0rsnom 320 and 360 is the way to go
19:35.49h3x0r190 is a elmeg phone
19:35.50Shaun2222jsharp: i could change it in the dialplan sure, but so far i dont see the point in even having a phone with a tone of buttons...
19:35.52timscottIf you can afford it, h3x0r.
19:35.54[av]banisnom 360 .. lol
19:35.58timscottWe aren't all made of money, like you.
19:36.00h3x0rthey dont cost much more than the other phones
19:36.02h3x0rthe 320 is nice
19:36.07timscottMmhmm, only like 100 bucks more.
19:36.18timscottwhatever
19:36.18[av]banih3x0r: the snom hardware is ok, the firmware is *horrible*
19:36.20h3x0rno like 40 bucks more
19:36.26timscottyes, right, you win.
19:36.27[av]banii would not recommend snom 3xx toanyone
19:36.28h3x0rthe new firmware is better
19:36.32[av]banino, it isn't.
19:36.35h3x0rthey fixed most of the bugs
19:36.37[av]baninope.
19:36.43h3x0rwhens the last time you upgraded
19:36.43timscott190's firmware seems alright. haven't had any real big problems with it
19:36.44*** join/#asterisk Grizzy (i=Generic@ppp-71-133-231-94.dsl.pltn13.pacbell.net)
19:36.45[av]baniit still locks up. still has horrile UI
19:36.51[av]banihmm... last week?
19:37.05[av]banirunning 5.5.1b
19:37.10*** join/#asterisk boch (n=fran@unirc.com.ar)
19:37.16bochhi guys, need your help to set the pppoe password via the ivr menu of a sipura spa-2100
19:37.27[av]baniit still locks up, still has a HORRIBLE ui, still has incorrect US indications
19:37.33jsharpI like the 7940s we have.  Once we got over the heartburn of reflashing them to SIP, that is.
19:37.33[av]banistill has 90% of the bugs I reported to snom MONTHS ago
19:38.25h3x0rlike what
19:38.37[av]baniwhy? can you fix them?
19:38.41*** part/#asterisk justinu|laptop (n=Justin@66.209.15.236)
19:39.03timscottbani: what's wrong with the 3xxx's that isn't wrong with the 190's?
19:39.06timscotterr
19:39.09timscott*3xx's
19:39.12[av]banitimscott: dunno, i dont have a 190
19:39.25timscott:/
19:39.25[av]banibut the 3xx is poo
19:39.28h3x0rthey dont sell 190s anymore
19:39.40timscottmy only real bug with the 190 is that it's kind of light...it slides around on my desk. :)
19:39.44timscottOther than that, they're fine
19:39.56Shaun2222what does include => default load?
19:40.08timscottthe [default] context, if there is one
19:40.15h3x0rsnom fixed the bugs i bitched about fairly promptly
19:40.26[av]banih3x0r: snom just blew off my last bugreport, saying it's a deliberate feature and won't ever be fixed.
19:40.46Shaun2222i see
19:40.46h3x0rand whats that
19:40.52timscottbani: what's that?
19:41.11[av]banion the idle screen, if you adjust ringer volume, it plays ringer 4 always. regardless of the actual ringer you have configured (eg custom ringtone, with completely different volume than the phone's internal ringer 4)
19:41.12timscottShaun: By the way, don't take _anything_ that I say as 100% truth and fact...
19:41.22timscottHonestly, I don't know half the time, it's just educated guessing.
19:41.26[av]baniso the ringtone voluem on the idle screen will likely not be ANYWHERE close to the actual ringer.
19:41.37h3x0rand this is the latest 5.x firmware?
19:41.44[av]baniyes. and snom says they won't fix it.
19:41.48timscottOh snap!
19:41.50[av]banithey say it's a deliberate decision.
19:42.07h3x0rhow does that interfere with you using your phone
19:42.11timscottbani, you're obviously a purist. ;)
19:42.29[av]banih3x0r: because the ring voluem adjustment on the idle screen has NOTHING to do with the actual ring volume.
19:42.38[av]baniit is INCORRECT, not only playing the wrong ringtone, but the wrong volume
19:42.50[av]baninow if you want things that interfere with me using my phone...
19:42.53h3x0rthe only thing that pisses me off on the 360 is the font size
19:42.55[TK]D-Fendertimscott : No, he just has the worlds largest raging bile ducts EVER.
19:42.56[av]baniit LOCKS UP all the time
19:43.03h3x0rand they told me its that way to display unicode characters
19:43.06[av]baniyeah, the font size is annoying too.
19:43.14h3x0rso thats why i use 320s :P
19:43.20[av]bani90% of my caller IDs are cutoff
19:43.21timscottD-Fender: :D
19:43.23h3x0rthe larger screen on the 360 is useless
19:43.27[av]banihell, snom's OWN MENUS get cut off...
19:43.41timscottWow, I'm so glad I didn't buy a 3xx.
19:43.43timscott190 <3
19:43.54[av]baniyou can't force the backlight on either.
19:43.55h3x0rthe 320 has 14 line keys
19:43.58h3x0renough said
19:43.58timscottNext time I get the chance, I'll try one out and see for myself.
19:44.01[av]banisnom also said they won't fix that.
19:44.15h3x0ri wouldnt leave a electronumicandescent backlight on all the time either
19:44.21[av]banithe US indications are wrong.
19:44.21h3x0rit would burn out in a couple years
19:44.24austinnichols102snom is a stupid name
19:44.29timscotthaha zing!
19:44.43h3x0rsnom 3xx sucks but everything else sucks more
19:44.45[av]banibut the biggest annoyance is the phone locks up all the time...
19:44.53[av]banino, i have a cisco and it is very nice.
19:45.02timscottI've never had a problem with locking up on any of my phones...
19:45.03[av]banihell, even the grandstreams don't f*cking lock up
19:45.08[av]banionly snom does
19:45.09h3x0rmine has never locked iup
19:45.10h3x0rup
19:45.13h3x0rmaybe yours is defective
19:45.17[av]baninope
19:45.23[av]baniand... lots of other people have it lock up too
19:45.29[av]baniit locks up mostly on transfers or conf calls
19:45.30timscottFlame war!
19:45.37timscottSnom sucks!
19:45.39timscottNo it doesnt!
19:45.41timscottYes it does!
19:45.43timscottNo it doesnt!
19:45.45timscottwhatever guys,.
19:45.47[av]baniit's not an isolated problem with defective hardware. snom knows about it, they've acknowledged the bug
19:45.52[av]banithey keep saying it's fixed, but it's not
19:45.54h3x0rat least it does conference calls right
19:46.17[av]banior rather, they keep promising it's fixed in the latest firmware. but we all keep having lockups anyway
19:46.33h3x0ryou are using 5.3.6?
19:46.59[av]banii've tried everything from 5.0 to 5.5.1b
19:47.20h3x0r5.5 ?!
19:47.27[av]baniyes
19:47.32mswanyone know what might give me "Cause: Invalid number format (28)" on my PRI?
19:47.37msw(this is my initial bringup)
19:47.44h3x0rwhere do you get that
19:47.51mswI've tried national and unknown pridialplan
19:47.59[av]banisnom really broke 5.4 also... totally broke autoprovisioning
19:48.02mswh3x0r: pri debug span N
19:48.09[av]banithen blamed me for having "incorrect programming"
19:48.11brif8Is there a way to route calls from one * box to another via the Internet where both * boxes are behind NAT ?
19:48.13h3x0rno i mean snom firmware 5.5
19:48.16timscottbrif: yes
19:48.24[av]banii sent them screenshots which proved it was their fau
19:48.26timscottThat's the problem I am actually working on at this very moment... >_<
19:48.26[av]banifault
19:48.32mswh3x0r: sorry, that wasn't for me, was it?
19:48.36timscottbrif: http://www.voip-info.org/wiki-Asterisk+SIP+NAT+solutions
19:48.37[av]banigetting them to acknowledge the bug and fix it was a major PITA
19:48.39h3x0rnope msw
19:48.49timscottbrif: http://www.voip-info.org/wiki/view/NAT+and+VOIP
19:49.09sevardSo I have a question about register times
19:49.23Kattythe answer is 42
19:49.26h3x0rare you using taht phone on the public internet by chance
19:49.33sevardIf * gets rebooted (and it does need to rebooted often) the phone stops ringing although * calls it
19:49.39sevardKatty: six by nine?
19:49.39timscottbrif: hope that helps, it's where I'm at right now... :/
19:49.53Kattysevard: this is not algebra.
19:49.53timscottNo, 6 by 7.
19:50.03Kattysevard: besides, that's 54
19:50.07[av]banih3x0r: it's obvious that snom has never tested the 360 outside the confines of their german offices.
19:50.20sevardKatty: I was just at a trade show demoing voip, i named one trillian, one arthur, one marvin, and one peter jones, phones.  and nody got my theme.
19:50.28sevardobviously you guys don't get the six by nine either :(*
19:50.28timscotthahaha
19:50.34timscottNo, sorry.
19:50.39timscottI only got the funny joke.
19:50.40timscott:S
19:50.41sevardoh, another phone was Ford
19:50.45timscottWhat's the six by nine all about?
19:50.58timscottIf you don't tell me, I'm going to have to waste my time googling. :S
19:50.58sevardthat's the ultimate question
19:51.03sevardwhat is six by nine
19:51.14sevardthe ultimate answer is 42
19:51.14timscottoh right
19:51.16timscottdurr
19:51.17timscottsorry
19:51.22[av]banih3x0r: http://www.snom.com/wiki/index.php/Beta_Firmware
19:51.27sevardbbiab have to move my car so the parking bitch doesn't slaughter me
19:51.31h3x0ryea i just found it on the wiki
19:51.31mswnevermind - bad Dial() cmd
19:51.37timscottforgot, base 13
19:51.57jsharpmsw:  Sending alpha characters onto the PRI?
19:52.30h3x0rhttp://www.voip-info.org/wiki/view/snom+360
19:52.32h3x0rhaha
19:52.34*** join/#asterisk justinu|laptop (n=Justin@wirelessdata-031-031.mycingular.net)
19:52.34h3x0rits the bug list
19:52.39timscottGuys. Who cares.
19:52.51Kattyi'm feelin pretty lost right now
19:52.56Kattybut that's ok!
19:52.59Kattyi'm used to it.
19:53.00timscottyoure in #asterisk on freenode.ent
19:53.02timscott*net
19:53.03Hmmhesaysi'm feeling like i want to knock some head
19:53.06*** join/#asterisk techman97_andy (n=me@70-98-31-249.dsl1.rsm.mn.frontiernet.net)
19:53.20Kattyoh thank goodness.
19:53.22[TK]D-FenderKatty : There, you're found!
19:53.22Kattyi'm no longer lost.
19:53.24Kattyk, all better.
19:53.27timscott*follows
19:53.30timscott:(
19:53.36KattyHmmhesays: i'm up for a little sparing
19:53.53[av]banih3x0r: that's not all the bugs, just the ones i was annoyed enough to post publically because snom ignored me
19:54.00mswjsharp: it wasn't stripping off the prefix properly
19:54.10techman97_andyhey all - question.  My SIP provider allows me to change the codec from uLAW to G729A.  I did that, but all of my calls were inaudible...scratching, popping, and the CLI was going nuts with errors.  I thought I saw G729A in the list of supported codecs that came with Asterisk - any ideas?
19:54.45[av]banih3x0r: mike240se is one of the other people having lockups with his snom phones... if it is "defective hardware" then snom really has severe QC issues with their manufacturing.
19:54.59[TK]D-FenderHmmhesays : Yeah... spare her ;)
19:55.09h3x0rits like a german car huh :)
19:55.16RaYmAn-Bxtechman97_andy: it's only supported in pass-through mode unless you buy a G729 license
19:55.32*** join/#asterisk badboyz (n=will@216.87.37.130.primary.net)
19:55.46[av]banih3x0r: also.. the snom 360 runs REALLY warm... all other phones I have don't get anywhere near that hot
19:55.53[av]banieven my cisco 7970
19:55.58techman97_andyso if I had SIP phones hanging off of Asterisk going to this SIP provider in G729A, I would need a license for every channel?
19:56.04badboyzso if i have a t1, and i want to connect it to an asterisk box, to use as a backup for outgoing phone calls... which interface card do i need?
19:56.21sevardAlright, so
19:56.33[TK]D-Fendertechman97_andy : Every channel in use at a given time.  But thats only in places where * neds to inject audio (like in the voicemail system, or IVR's, etc)
19:56.33sevardIf * gets rebooted (and it does need to rebooted often) the phone stops ringing although * calls it
19:56.55sevardIt sounds like a dumb idea for the phjone to re register to the server every 60 seconds to fix this problem
19:57.00sevardphone*
19:57.10sevardbut.. that's the only answer I can come up with.
19:57.33techman97_andyok - so my SIP provider only has one channel that I go out...although it has multiple pipes that are accessible...9 SIP phones, and 1 IVR...would that by two G729A licenses?
19:57.43h3x0ranybody use that linksys cordless voip phone yet
19:57.57[av]banih3x0r: the snom 360 is a real letdown for such an expensive phone. other phones in the same price bracket are 100x better.
19:58.11[av]baniand most of the issue is with snom's shitty firmware.
19:58.12h3x0rlike what
19:58.14h3x0rciscos phones suck
19:58.21timscottuhh
19:58.23timscottheathen?
19:58.24h3x0rpolycoms you cant legitamtely use with asterisk
19:58.27[av]banii have cisco, grandstream, polycom, and snom
19:58.34[av]banium what?
19:58.38mog_workwhy cant you use polycoms?
19:58.38h3x0rgrandstream is awful
19:58.40*** join/#asterisk exonic (n=exonic@sig.triton.net)
19:58.44sevardh3x0r: I just got an Aastra 408i CT and it's everything I'ev ever wanted in any phgone ever.
19:58.46[av]banipolycom has no problems using with asterisk.
19:58.47sevardphone*
19:58.51timscottAastra = awesome.
19:58.53h3x0rits not "legal"
19:58.54[av]baniin fact asterisk is an _officially supported platform_ by polycom
19:58.55timscottAastra for the win.
19:59.04tainted-why can't u use polycom
19:59.04[av]banih3x0r: um. sorry, but... BS
19:59.04mog_workwhat do you mean its not legal?
19:59.06mog_workpolycom supports astersik
19:59.08mog_workerr asterisk
19:59.09exonicHey folks, What' s agood solution to peer asterisk with 4+ lines? I would like to integrate into FXO ports that are already in existance.
19:59.09[av]baniit's perfectly legal, and even supported.
19:59.10mog_workas guess what
19:59.14sevardtimscott: This phone is _so god damn awesome_
19:59.18mog_workpeople want to buy phones to use with asterisk.
19:59.20timscottsevard: that's what I've heard.
19:59.22timscottI want to get one. :D
19:59.25mog_workmore than they want a polycom pbx and phones
19:59.30h3x0r<PROTECTED>
19:59.33h3x0rvoipsupply
19:59.34h3x0ratacomm
19:59.35h3x0retc.
19:59.36[av]banih3x0r: i have no idea where you heard that, but it's bullshit
19:59.37NetgeeksI like my cisco phone, it has worked perfectly for me for years and I spend many hours a day on the phone...
19:59.37tainted-sevard what phone
19:59.37mog_workyou are wrong
19:59.42mog_workits total bullshit
19:59.44sevardtimscott: more expensive than most but BY FAR _the best_ phone I've ever used ever.
19:59.52sevardtainted-: Aastra 480i CT
19:59.54mog_workpolycom sent several phones to digium
19:59.57mog_workto be sure they all worked
20:00.00tainted-sevard i heard it's shit
20:00.01mog_worki have one on my desk
20:00.04justinu|laptopi'v never had that problem with anyone
20:00.04[av]banih3x0r: um. voipsupply and atacomm know perfectly well polycom is legal to use with asterisk.
20:00.07mog_workkevin is talking on one right now
20:00.07h3x0rok
20:00.08timscottAastra built stuff for TDM pbxs before they built IP, mmhmm.
20:00.12timscottunless I am mistaken
20:00.13[av]banih3x0r: voipsupply even advertises polycoms for that purpose.
20:00.13sevardtainted-: My review is it's the absolute best.
20:00.15h3x0rmaybe they just dont make money on them
20:00.22mog_workheh
20:00.23mog_worksure..........
20:00.24sevardtimscott: you're not mistaken
20:00.31timscottthey've had a while in the industry to learn how things should work
20:00.35tainted-sevard have u used snoms, polycoms, or ciscos? maybe u have nothing to reference to
20:00.37sevardIt doesn't feel cheap, it feels like a real phone, it has NO draw backs o that I can tell
20:00.46timscott'cept $$$++
20:00.48timscott:D
20:00.50h3x0rthe cisco 94x phones are awful
20:00.51timscottBut it's worth it
20:00.55tainted-sevard that's the one with the xmlbrowser right?
20:00.56h3x0rand the legal issue with 79xx applies
20:01.01sevardI've used snom and polycoms and used a HOP1--2 for a long time
20:01.03sevardHOP 1002
20:01.09austinnichols102what's the 79xx legal issue?
20:01.10h3x0ror i should say, the fact that they suck with sip
20:01.17Netgeekslegal issues with the 79xx?
20:01.25sevardtainted-: I haven't dug into xml on this phone yet
20:01.25h3x0ryou have to pay license fees to use the phone
20:01.25jsharpSIP licensing and all that.
20:01.38[av]banithere's no licensing issues. i got that direct from cisco.
20:01.51austinnichols102no problem with sip licensing.
20:01.52[av]banithe only license is for ccm
20:01.52h3x0rregardless the sip firmware for 79xx is a joke
20:01.57Netgeeksyou buy the phone for a authorized cisco reseller, and you are fine
20:02.08*** join/#asterisk brodiem (i=1000@cpe-66-69-222-36.austin.res.rr.com)
20:02.08austinnichols102I'm not laughing
20:02.18Netgeeksum okay h3x0r
20:02.29h3x0rpots phones on an ata do way more than a 7960
20:02.43brodiemanyone have experience with using fax w/ ATA (specifically SPA1001)?
20:02.46austinnichols102example?
20:02.54malverianIf I do a "NoCDR" sometime earlier on in the dialplan.. will ResetCDR undo that?
20:02.57[av]banih3x0r: no, you don't.
20:02.59h3x0rnot being able to drop a conference party
20:03.11[av]banithe only license you need is for CCM. please stop.
20:03.35h3x0ryou need cco to download the sip firmware
20:03.48h3x0rto use cco means you are supposed to buy the support contract on your phone
20:04.06austinnichols102they charge for fw updates, so what...
20:04.44h3x0rupdates?  they ship with SCCP firmware
20:05.39[av]banih3x0r: there are various support contracts, the cheapest one is $8
20:05.52Netgeeksso is your dislike for the 79xx phones based upon cisco business practices, or do you actually not like the way the phone functions, or both?
20:06.30timscottIs anyone here at all familier with SIP and NAT workarounds for Snom190's?
20:06.32austinnichols102I would like to have keepalive and a couple of other features in the phone, but I'm far from not liking the phone
20:06.47timscottIs anyone here at all familier with SIP and NAT workarounds for Snom190's?
20:06.48timscott>_<
20:07.28[av]banicisco is pretty lax with firmware policies though. you can usually call them up without any support contract and ask for the latest sccp and they'll send it along.
20:07.35brif8can I still join two * boxes with SIP if the extensions are the same on both?  I basically want to route all incoming calls on Box 1  to [incoming] on box 2
20:07.40[av]bani(they will also do this with ios, when there are security alerts)
20:08.27[av]baniso um h3x0r, what phones do you actually have besides a snom 3xx?
20:09.12timscottsooo...I'm guessing no one, then. :'(
20:09.16[TK]D-Fenderbrif8 : You can do most things you can imagine with *
20:09.17sevardI strongly suggest getting an aastra 480i ct from a retailer that you can return it to if you don't like it
20:09.21brif8I'm reading WIKI but with the day I've had it isn't making much sense  http://www.voip-info.org/wiki-Asterisk+-+dual+servers
20:09.42[TK]D-Fenderbrif8 : you can send calls from anywhere to anywhere.. its up to you.
20:10.23brif8[TK]D-Fender: care to please just clarify what I need to do.  I tried simply adding a extension on box 1 to DIAL(SIP/nex@serverBIP,80,Ttr)  but it rings once and fails
20:10.32malverianI need some help with a CDR problem I'm running into..
20:10.39brif8I probably need to have a register => but I'm not sure
20:10.45malverianIf anyone has a moment to give me some pointers, I'd greatly appreciate it.
20:10.54*** join/#asterisk ToTo (n=ToTo@host55-145.pool870.interbusiness.it)
20:11.50DandanIAX2 to SIP/broadvoice translation doesn't work...
20:11.50Dandanno errors reported
20:11.52*** join/#asterisk Jaxxan (n=jaxxan@202.70.125.124)
20:11.53timscottI would also greatly appreciate it if someone would take a moment to give me pointers :S
20:11.57malverianThe scenario is like so: 1) Customer calls in to main line and a sales rep answers the phone, 2) Sales rep puts the customer in "parking lot", 3) Another sales rep picks up the customer from parking lot
20:13.14malverianThe following call detail records get logged: 1) src = customer, dst = rep, 2) src = rep, dst = parking lot
20:13.53*** join/#asterisk oej (n=oej@apollo.webway.se)
20:14.06malverianI need #2 to instead show src = customer, dst = rep. The Set(CDR(src)=....) functions say that the CDR variable is read only.. do I have any chance here?
20:14.18Kattywhat's better, the ip500 or the 501?
20:14.30Kattyare there any major differences?
20:14.35Kattydid the 501 sprout wings?
20:14.38Kattydoes it do a little dance?
20:14.47timscotthmm
20:14.57timscottsometimes the "X01"s have a second port
20:14.59timscotta mini-hub
20:15.02malveriantimscott, I have no idea.. there is a wiki on the SNOM site, have you checked tha tout?
20:15.15timscottmalverian: reading it at the moment :S
20:15.18brodiemwhat is the most reliable way of using a physical fax machine? Getting about 90% fail rate using SPA1001..
20:15.25timscotthaha, use POTS
20:15.33timscottbrodiem :S
20:15.39malverianbrodiem, Install FXS card, run analog cable :-P
20:16.04[TK]D-Fenderbrif8 : That Dial sample sends an UNAUTHENTICATED call to the other server, which would have to be set up to accept them.  usually an unsafe thing when "forwarding" calls...
20:16.13brodiemmalverian, my only worry is that Digium doesn't recommend their FXS for faxing because of the timing device
20:16.31malverianbrodiem, Does your trunk have a timing device?
20:16.39brif8[TK]D-Fender: I accept that but this is just a temporary test
20:16.46[TK]D-FenderKatty, timscott : Incorrect.  X01's have more ram to support larger SIP & BR images
20:16.52brif8how would I authenticate two * boxes
20:16.54malverianbrodiem, We have a PRI here, and our fax machines work almost 100% successful with wildcards.
20:16.59Katty[TK]D-Fender: thanky.
20:17.04timscottFender ok
20:17.09[TK]D-Fenderbrif8 : then you need to have a context set in [general] and allowguest=yes in sip.conf
20:17.16gandhijeeDandan: what do you mean it doesn't work?
20:17.25[TK]D-FenderKatty : and in the case of the 601, the ability to add on the attendant modules.
20:17.27*** join/#asterisk Cybertoy (n=maxim@dsl254-123-241.nyc1.dsl.speakeasy.net)
20:17.31timscottI just ate like a whole salmon
20:17.39brodiemmalverian, yes I believe so (at least I have the telco setup as the master timing source in zaptel.conf)
20:17.44*** join/#asterisk mitcheloc (n=mitchelo@70-32-188-167.lmdaca.adelphia.net)
20:17.46gandhijeeDandan: are you sure you are registered w/ broadvoice?
20:17.47NetgeeksTeh IP501 sprouts wings?
20:17.51NetgeeksThe, even?
20:17.53malverianbrodiem, Then you should be fine.
20:18.01[TK]D-Fendertimscott : I doubt that unless you got a tiny baby one... which typically would get thrown back...
20:18.02malverianbrodiem, We had someone come test the line and we had no timeslips.
20:18.10brodiemmalverian, cool
20:18.11timscottdude it was like three lbs of salmon
20:18.15timscotti'm stuffed
20:18.19timscottwith salmon.
20:18.21brodiemmalverian, is there a single FXS card or just the 400p w/ module available?
20:18.23Dandangandhijee: yeah
20:18.23Dandani can do sip-sip through bv
20:18.24Dandanno probs
20:18.25malverianbrodiem, Er.. frame slips.
20:18.34[TK]D-Fendertimscott : Do you have any real idea just how big a mature salmon gets?
20:18.38mog_workmalverian, any chance you might finish you sphynx pre 1.4?
20:18.41malverianbrodiem, I'm not certain.. I _think_ there are dual span.
20:18.41gandhijeeDandan: what version you running?
20:18.53malverianmog_work, Yes, as soon as I figure out how to fix this damn CDR issue :-P
20:18.55timscottFender: Relax, man...it's just talking.
20:19.03gandhijeeDandan: i was using my asterisk box w/ broadvoice till last week, has IAX, MGCP and SIPs phones
20:19.06gandhijeeno problems
20:19.10[TK]D-Fenderbrodiem : TDM for FXS is pretty wasteful $ wise, and less functional / portable
20:19.27malverian[TK]D-Fender, But more reliable for faxing in my experience here.
20:19.31LeobNOVICE QUESTION: can anyone help me configure ODBC?  I've been trying for about 8 hours and I have no idea of how to check if I'm doing something wrong... Any suggestion would be more than welcome!!
20:19.34mog_workwhats wrong with cdr
20:19.42[TK]D-Fendermalverian : than a decent ATA?
20:19.43brodiem[TK]D-Fender, what is the alternative for fax? besides a completely seperate pots
20:19.46malverianmog_work, Just a problem with a specific thing I am trying to do with it.
20:19.50Dandan1.2.6
20:19.50Dandani use cubix
20:19.50Dandanas iax
20:19.51Dandanto call sip via bv
20:19.51Dandanif i do iax - sip internally all is good
20:19.52malverian[TK]D-Fender, Ah.. no.
20:19.55malverian[TK]D-Fender, But that's usually overkill.
20:19.57[TK]D-Fendermalverian : OH, faxing... nevermind ;)
20:20.02brodiemlol
20:20.03[av]banihmm.. gxp2000 does rfc2833 fine, polycom 601 doesn't...
20:20.04[av]baniweird
20:20.08austinnichols102anyone familiar with US LEC as a carrier?  I was shocked to find out that they support asterisk.
20:20.13*** join/#asterisk MacDome (n=eseidel@A17-255-104-111.apple.com)
20:20.17malverianaustinnichols102, Yeah, we use them in Gainesville.
20:20.18[TK]D-Fender[av]bani : Mine seems to be fine here with RFC...
20:20.18Dandanaustinnichols102: I am
20:20.27Dandanactually my current provider is taking them over
20:20.30malverianaustinnichols102, They are the worst customer service I've ever epxerienced :-P
20:20.34malverian*experienced
20:20.37gandhijeei used idefisk as my IAX client, a Yxtn phone w/ MGCP and a snom 360 w/ sip, no problems
20:20.42[av]bani[TK]D-Fender: i can't get our 601's to do dtmf properly. gxp2k's work fine. so it's not an asterisk config problem.
20:20.44malverianaustinnichols102, Their "professionals" are complete noobs.
20:20.48austinnichols102great
20:20.49timscotthappy noob
20:20.50[av]bani[TK]D-Fender: and i'm using the stock sip.cfg from polycom...
20:21.05brodiemmalverian so a $140 tdm10b is probably the cheapest way to get analog fax?
20:21.15gandhijeeDandan: taking who over?
20:21.15h3x0rlets see i have cisco 7960, grandstream gxp2000, snom 320 and 360, linksys 941, sipura 841,
20:21.23Dandanus lec
20:21.32gandhijeeis taking over broadvoice?
20:21.34Dandanctc is taking us lec over
20:21.35Dandannoooo
20:21.41[TK]D-Fender[av]bani : I use basic stock setups too.. never a problem, and I use them at home a lot now too.
20:21.58[av]bani[TK]D-Fender: well i'm stuck, i can't figure out why the 601 isn't sending dtmf. sucks :(
20:22.07[TK]D-Fender[av]bani : I hear ya...
20:22.13[av]bani[TK]D-Fender: i keep bashing my head against stupid polycom documentation
20:22.20gandhijeespeakin of the 601's is there an XML editor for those bitches?
20:22.25austinnichols102who's CTC?
20:22.27[av]banino
20:22.28gandhijeeor an easier way to edit that file
20:22.34[av]baninope
20:22.38gandhijee=(
20:22.38h3x0ri sold the other stuff i had
20:22.40[TK]D-Fendergandhijee : there are a number of XML editors out there....
20:22.43h3x0ri should try polycom
20:22.59[TK]D-Fenderh3x0r : I did the same... to BUY my Polycom's :)
20:23.01[av]banipolycom are harder to provision than even cisco.
20:23.13malverianmog_work, I queried you to avoid sending repeat information to the channel.
20:23.13gandhijeeyeah they are
20:23.20[av]baniyou keep finding these stupid little hidden undocumented switches in the configs
20:23.21timscottI thought ciscos were nice to provision
20:23.32timscottCan't you import a CSV file to configure them?
20:23.34[av]banitimscott: not with asterisk
20:23.35gandhijeebut they sound great
20:23.38[TK]D-Fender[av]bani : Maybe a little slower on the start, but once you know, I doub't cisco is really much better...
20:23.48[av]bani[TK]D-Fender: cisco is a lot simpler
20:23.56[av]banitheres no 5 billion config entries
20:24.01timscottbani: Can't you import a CSV file into most cisco phones for provisioning?
20:24.05[av]banipolycom is a real mess
20:24.06timscottMakes for easier automation
20:24.12[TK]D-Fender[av]bani : I dunno... I got my news ones done from scratch in no time flat...
20:24.14gandhijeeyou'd think polycom would put out a file to help you config them easier, specailly since they "Now Support Asterisk"
20:24.17DandanCtc communications
20:24.23Dandanwww.ctcnet.com
20:24.26[av]bani[TK]D-Fender: because your config is simple and you had a ready made template.
20:24.27austinnichols102tks
20:24.39[av]bani[TK]D-Fender: someone coming to polycom from scratch isn't nice
20:24.40Dandanthey are getting bigger by the month...
20:24.53[av]baniout of the box they are a major pita
20:24.59austinnichols102dandan: considering us lec, but don't have time to really deal with more carrier-level stupidity
20:25.33gandhijeei still don't understand everything bout those phones
20:25.36h3x0ryou know whats crazy
20:25.37austinnichols102had enough of that with FDN this year
20:25.37timscottbani: you gotta stop talking about pitas like they are a bad thing
20:25.46h3x0rit would be cheaper to deploy computers and headsets than most voip hard phones
20:25.52*** join/#asterisk websae (i=icechat5@CPE-24-167-204-30.wi.res.rr.com)
20:25.57h3x0r*cheap* computers anyway
20:26.00Netgeeksaustinnichols, are you looking for termination and origination?
20:26.05austinnichols102yup
20:26.06*** join/#asterisk goodjoke (n=Goodjoke@rrcs-24-97-65-74.nys.biz.rr.com)
20:26.15timscottHey look, it's a good joke.
20:26.21Netgeeksdo you need any specific DIDs (location wise?)
20:26.26goodjokeis this room for a@h as well as *?
20:26.36h3x0rgood: read the /topic
20:26.37[TK]D-Fender[av]bani : No I didn't have atemplate for the one I just set up.  like I said, from SCRATCH.
20:26.43goodjokethanks..just saw it
20:26.46timscottoh darn.
20:26.54timscottDon't leave, it's been a while since I've heard a good joke.
20:26.56timscott:(
20:26.59Netgeeksaustinnichols102, do you need any specific DIDs (location wise?)
20:26.59timscottha ha. :S
20:26.59[TK]D-Fender[av]bani : I just knew what to look for.
20:27.21goodjokei used to have a website called goodjoke.com... then i got married
20:27.33[av]bani<DTMF tone.dtmf.level="-15" tone.dtmf.onTime="50" tone.dtmf.offTime="50" tone.dtmf.chassis.masking="0" tone.dtmf.stim.pac.offHookOnly="0" tone.dtmf.viaRtp="1" tone.dtmf.rfc2833Control="1" tone.dtmf.rfc2833Payload="101"/>
20:27.34[TK]D-Fender[16:26] <goodjoke> is this room for a@h as well as *? <- timscott, thats good enough for me :)
20:27.35timscottit's nice to hear a good joke.
20:27.38[av]banihm. that should work, but it doesn't.
20:27.47[av]baniyou punch dtmf and the remote hears nothing.
20:27.51[av]banigxp2k's work perfectly.
20:28.03timscottbani, 'cept for mad lack of echo cancellation
20:28.11[av]banitimscott: what?
20:28.15timscottyou read me.
20:28.25[av]banibe more specific?
20:28.26timscottmad lack of speakerphone echo cancellation
20:28.32[av]banier
20:28.40[av]banithat was fixed in 1.0.2.8 or so
20:28.43[av]banilast year
20:28.45timscottOh, was it?
20:28.52timscottI updated like four months ago
20:28.53[av]baniabout 8 months ago?
20:28.55h3x0rhttp://www.polycom.com/techpartners.htm
20:28.56timscotti'll update again, I guess.
20:29.01h3x0rheh they list asterisk business edition
20:29.04[hC]er? rfc2833 would not let the remote end hear dtmf audibly.. you'd have to use inband for that
20:29.22[av]bani[hC]: what?
20:29.31[av]bani[hC]: the gateway converts rfc2833 to audio...
20:29.36h3x0rand the 600 isnt listed
20:29.49[av]bani[hC]: if the remote end couldnt hear dtmf audibly, then the gxp2k's wouldnt be working either. but they work.
20:29.56[hC][av]bani:: okay, most phones that i use, when using rfc2833, if i press dtmf, talking to someone, they cant hear it
20:30.02[hC]yet my dtmf works on IVR's and such
20:30.15*** join/#asterisk Jaxxan (n=jaxxan@202.70.125.124)
20:30.38[av]bani[hC]: um, that's not possible, unless you're talking 100% IP to the ivr
20:30.47[av]baniif you're going out on pstn, it HAS to be converted to audio.
20:31.05[av]banithink about it for a minute and i'm sure you'll figure it out :)
20:31.43timscottyay reboot.
20:31.50timscottnick war
20:31.55timscotterr
20:31.59timscottthat was the wrong server.
20:32.38[hC][av]bani: yes, i agree, it makes sense. Im not sure, i havent really tested  what happens extensively. I just know its not an issue :)
20:33.32timscotthot!
20:33.36timscottI got my phone to register.
20:33.37timscottNice.
20:33.41timscottNow let us try calling.
20:34.03*** join/#asterisk zotz (n=zotz@24.231.32.85)
20:34.33[TK]D-Fenderok, I'm out, later all
20:35.46*** join/#asterisk CrummyGummy (n=wayne@dsl-145-122-123.telkomadsl.co.za)
20:36.10*** join/#asterisk tuxd00d (n=tuxinato@netblock-68-183-112-100.dslextreme.com)
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20:37.34hackeronHey, for some reason I hear the occational bleep when going out through ZAP channels (analog lines) and Verizon swear its not them, any ideas what it could be?
20:39.10jsharpcall waiting?
20:40.23*** join/#asterisk lokkju (n=lokkju@unaffiliated/lokkju)
20:40.54hackeronjsharp: hmm, not a feature on the line and isnt it normally a beep? -- I hear like a test blip, its very short and not always clear, just like a blupi type sound, lol
20:41.16docelm0Say who in here was building a call center?
20:41.31brif8http://pastebin.com/654287  what am I missing ??
20:41.42jsharpDunno, then.
20:41.54lokkjuwondering if someone could give me some suggestions on where to start debugging here.  I recently installed * from the debain packages, and am now having trouble with sound - no matter what, I can not seem to get any sound to work.  There is no firewall between my client and my server, and there are no error messages in my full log
20:43.21brif8getting multiple Registration errors  on both servers
20:45.51*** join/#asterisk Gamercjm (n=chris@pool-71-254-175-120.lsanca.fios.verizon.net)
20:46.38GamercjmHas anyone used voipuser.org and set up the DID using SIP?
20:46.39freatlokkju: you got iptables running on the asterisk server?
20:46.40hackeronjsharp: I mean there cant really be call waiting, its a hunt group, but maybe it blips when failing over to the next line? - even so, the dialing out lines and incoming lines are at opposite ends
20:47.00GamercjmIm having trouble with the SIP URI
20:47.16lokkjufreat, yes, but totally open on the internal interface, which is what I am connecting to
20:47.37freatlokkju: for grins I would try turning off iptables just to see...
20:47.53lokkjufreat tried it already
20:47.56freatlokkju: oh ok
20:48.06brif8[TK]D-Fender:  http://pastebin.com/654287  what am I missing ?? getting multiple registration failure  on both servers
20:48.19freatlokkju: no audio in either direction? if you call a conference room or something with MOH do you hear the music?
20:49.08freatlokkju: I take it that the phone is registered w/ Asterisk OK? sip show peers
20:49.46freatalso try sip debug and place a call, then sip no debug and scroll up
20:49.49lokkjuyes, and in the debug I can see the CLI saying it is playing sound
20:50.10lokkjuhmm
20:50.20freathandset plugged in all the way? ;)
20:50.53lokkjuinteresting - I put a festival line and a playback line one right after the other, and the festival line I can hear, but not the playback
20:51.03freatahh
20:51.27freatsounds like no timer source
20:51.46freatcheck your loaded modules... look for ztdummy
20:51.51*** part/#asterisk goodjoke (n=Goodjoke@rrcs-24-97-65-74.nys.biz.rr.com)
20:51.53freator do you have a zaptel card?
20:51.57DabianI wonder if you guys could give me some tips on how to set up a PSTN gateway.  Both hardware, business model, etc.
20:52.17lokkjuztdummy *IS* loaded
20:52.39freatlokkju: hmm.. try re-running ztconfig or whatever the debian equivalent service
20:52.40Dabian(PSTN/ISDN)
20:53.07*** join/#asterisk MacDome (n=eseidel@A17-255-104-111.apple.com)
20:53.15freatDabian: I think people need more specifics. I mean, it could be very simple or very complex depending.
20:53.15DabianI assume I will need FXO hardware.. but I also wonder how to scale .. eg. do I need 9 lines for 10 users, or only 5?
20:53.18lokkjufreat, I don't know of any such thing...
20:53.27*** part/#asterisk austinnichols102 (n=austinni@70.46.69.131)
20:53.27jsharpDabian:  PRI + Asterisk, buy cheap, sell high.
20:53.45Dabianfreat : Scaleable .. I plan on doing this in my free time.
20:53.51freatlokkju: there should be a service that loads the ztdummy module
20:53.59Dabianjsharp : PRI?
20:54.09lokkjufreat, um, loads it into the kernel?
20:54.55freatwell, I dunno about debian. But on linux there's a service that looks for zaptel devices, then if it doesn't find them, it loads up the ztdummy module.
20:55.15jsharpPRI = Digital phone lines
20:55.17freattry typing ztcfg at the command line
20:55.24freatPrimary Rate Interface
20:55.36jsharp23 lines, brough in on a single circuit.
20:56.00freat23 B + 1 D
20:56.01lokkjuno, I just directly load through /etc/modules - and if debian is not linux, then what do you think it is?
20:56.13freatlokkju: heh yeah I just meant distro
20:56.15Dabian23 B?
20:56.16Dabianjsharp : Sounds nice.  How much are they?
20:56.23Dabian(+/-)
20:56.38freat$400 month
20:56.41freator more
20:56.47freatdepending on your package
20:56.51freatcould be thousands
20:57.01DabianOh .. thats including stuff
20:57.22jsharpYah.  Price depends on a lot of things.
20:57.29lokkjufreat, got it, all my zt* bins are in my build directory, since I did not want to do a full install - no plans on installing a zaptel card in a 1U in a colo :)
20:57.44jsharpThen you also have to pay your LD charges on top of the PRI loop charges.
20:57.58DabianI would like to route danish PSTN numbers over IP.
20:58.13DabianLD?
20:58.18jsharplong distance
20:58.23b4kaanyone knows if the sangoma cards work no openbsd with asterisk?
20:58.32freatDabian:  you should probably read the forums and such
20:58.36Dabianahh .. I plan on one way .. PSTN -> IP
20:59.03*** join/#asterisk austinnichols102 (n=austinni@70.46.69.131)
20:59.10DabianAt least for starters.  Maybe 1 or 2 lines IP->PSTN.
20:59.29Dabian23 lines sounds like a nice starting package.
20:59.36jsharpOh.  Inbound from the PSTN out to IP.
20:59.36Dabianfor PSTN->IP
20:59.40lokkjufreat, ok, so I found the tools - what did you want me to check
20:59.44freatthen just get some regular lines, a digium card w/ 4 ports and go to town
20:59.52freatlokkju: try running ztcfg
20:59.57jsharpyeah, what freat said.
21:00.07freatjsharp: heh
21:00.09lokkjufreat, ok, runs no problems
21:00.32Dabianjsharp : Yeah .. like someone on PSTN dials "12345678", I pick up the number, look it up in a db or asterisk handles it, and routers it to the SIP client or voicemailbox.
21:01.45jsharpThat's easy, then.
21:02.07DabianHeh .. I didn't dare hoping you would say that. :-)
21:02.28jsharpIts just all a matter of how much you want to spend on hardware and phone line charges.
21:02.39lokkjuwait a sec - this thing about needing usb-uhci still true?
21:02.45Dabianjsharp : I guess its because I forgot to say I want number-series and numberporting.
21:02.54*** join/#asterisk lyroy (n=toor@modemcable146.87-83-70.mc.videotron.ca)
21:03.30jsharpThen you'll need a PRI line of some sort, then.
21:03.41lyroyDoes someone here ever experience problems when compiling asterisk-addons on a VIA processor...?
21:03.41websaehas anyone used VoIP-Discount?
21:04.19*** part/#asterisk maffro (n=furby@n156.dkm.cz)
21:05.35lokkjuseems that ztdummy is supposed to rely on usb-uhci, but my system uses uhci_hcd
21:05.39Dabianjsharp : Eg. lets say Smith has the PSTN number X, and he uses VoIP_Y for IP.  Now he wants to go all IP.  I tell him, "Pay me some $$, and I'll forward your number."
21:05.42lokkjucould this be causing the issues?
21:05.47*** join/#asterisk marv (n=ilovekim@12-219-145-181.client.mchsi.com)
21:06.11*** join/#asterisk nitram (i=foo@superblob.com) [NETSPLIT VICTIM]
21:06.23Dabianjsharp : When someone on PSTN dials X, the number gets on my PRI? and I router it to VoIP_Y.
21:06.35lokkju(if it makes a diff, I am on the 2.6 kernel)
21:06.36jsharpRight.  Sounds good so far.
21:06.37*** join/#asterisk Aurs (n=Aurs@host-81-191-123-189.bluecom.no)
21:06.51Dabianjsharp : The PRI, they're not COPPER lines?
21:06.57Dabianjsharp : They're like VPN?
21:07.08jsharpThey are copper lines, yes.
21:07.11b4kaanyone knows if the sangoma cards work on openbsd with asterisk?
21:07.20Dabianjsharp : So I need a bunch of copper?
21:07.36jsharpInstead of having 24 separate 2-wire phone lines come in, they come in as 24 channels multiplexed together on a T1.
21:07.47jsharp4 wires versus 48 wires.
21:07.53jsharpOne port on a switch versus 24 ports.
21:07.54Dabianok
21:08.31MikeJ[Laptop]b4ka, sangoma cards still require zaptel on asterisk
21:08.42MikeJ[Laptop]so anything that has base zaptel support
21:09.12lyroyDoes someone here ever experience problems when compiling asterisk-addons on a VIA EPIA (mini-itx board with CPU on board)...?
21:09.14Dabianjsharp: I guess ideally I would prefer that to be handled at the ISP, so I get all the traffic over IP..
21:09.34jsharpWhere you put the equipment is up to you.
21:09.47Dabianjsharp : Nice
21:09.48jsharpBut if you put it in a colocation facility, you often get a much better price on the PRI.
21:10.04Dabianjsharp : Ahh of course.. Colocation!
21:10.41Dabianjsharp : That way I might not have to pay for digging down copper (which would probably make it virtually impossible for me)
21:11.07*** join/#asterisk fr3aky (n=fr3aky@nar.macol.ru)
21:12.13*** join/#asterisk dwmw2 (n=dwmw2@baythorne.infradead.org)
21:15.03Dabianjsharp : I don't really want to offer outgoing PSTN - at least not for starters, since I assume thats where you put the heavy money down.
21:15.47shido6anyone in India?
21:15.52freatlokkju: sorry got a phone call
21:16.12freatlokkju: can you place outbound or inbound calls?
21:16.34Dabianjsharp : The idea is to make it easy for people to switch between different VoIP companies offering peering, pertaining their old PSTN number for a small fee they pay me.
21:17.23Dabianjsharp : They wont get CID on the number though.
21:18.16*** join/#asterisk key2 (n=key2@gob75-2-81-56-64-17.fbx.proxad.net)
21:18.36Dabianjsharp : I am trying to figure out if its possible to do within a buget.  I don't want to rip off people, on the other hand, I don't want to work for free.
21:19.14Dabian!ping
21:19.18lyroyDoes someone here ever experience problems when compiling asterisk-addons on a VIA EPIA (mini-itx board with CPU on board)...I have some message like that ( mysql.h No such file...) ?
21:19.48Dabianjsharp : Any idea about porting numbers, and number series?
21:20.42Dabianjsharp : I mean, every time I get a number ported, I guess I have to tell the PRI provider, "Hey, would also route the number XXXXXXX for me?"
21:21.20lokkjufreat, inbound, no problem
21:21.41lokkjufreat, call <anything>@ns2.ifpdx.com
21:22.09lokkjuyou are supposed to hear a festival voice, then three beeps
21:22.14Dabianfreat: How much is a digium 4 ports card, btw?  (Not exact price, just the ball park)
21:24.11Nuggethttp://store.digium.com/
21:24.18lokkjuDabian, http://www.gmprice.com/index.php?qstring=digium+4&cat=61838
21:24.44lokkjuaverage ebay price is $342, as you can see on the site I just pasted
21:24.59*** join/#asterisk r0d3nt (n=RatMan@foster.stonedcoder.org)
21:25.00lokkju(my own little tool, and yes, I'll plug it when I can)
21:25.13websaeanyone here manage or own a call center, where you use only SIP or IAX trunks for termination? How is your call quality and uptime?
21:26.34websaei am curious how many businesses are going to VoIP trunks either via SIP or IAX
21:27.26lokkjufreat, please, some more ideas on debugging this
21:29.57brodiemIs it possible to dial out in a Meetme room?
21:30.13badboyzso if i have a t1, and i want to connect it to an asterisk box, to use as a backup for outgoing phone calls... which interface card do i need?
21:32.03*** join/#asterisk glm2k (n=glm@rrcs-24-199-11-46.west.biz.rr.com)
21:32.22PakiPenguinis there a difference between an integrated t1 and mixed mode t1
21:32.35Dabianwebsae : Maybe that would be interesting for me..
21:32.45Dabianlokkju : Sounds kinda expensive.
21:32.53Dabianlokkju : I assume its high quality?
21:34.40*** join/#asterisk stoffell (n=stoffell@pot.catsanddogs.com)
21:35.01lokkjuDabian, they are digium *shrug* - and that actually pretty damn cheap compared to normal phone hardware
21:35.50*** join/#asterisk jtodd (n=jtodd@dsl027-191-178.sfo1.dsl.speakeasy.net)
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21:37.44DabianOh .. I don't mean expensive .. even though thats what I wrote .. just more pricey than I expected.
21:38.31Nivexbrodiem: not directly.  You could use the manager interface or a call file to bring a party in though
21:39.18skyboyhello, can anyone help out with a config regarding ser to asterisk connectivity?
21:40.10*** join/#asterisk RoyK (n=roy@28.80-203-106.nextgentel.com)
21:40.45*** join/#asterisk Renacor (n=kvirc@ip21.farheap.net)
21:41.02RenacorIs there an app that can take the asterisk cdr csv and parse it for statistics?
21:41.38AursRenacor: have you considered cdr_odbc or cdr_mysql?
21:43.24Renacoris there a howto on it?
21:43.33RenacorIm guessing you can make cdr_odbc work with postgres?
21:44.25*** join/#asterisk MacDome (n=eseidel@A17-255-104-111.apple.com)
21:45.01PakiPenguinapple.com :)
21:45.04PakiPenguinhehe
21:45.07*** join/#asterisk eric_hill (i=EricHill@204.94.175.11)
21:45.10PakiPenguinsend me a free imac
21:45.12PakiPenguinplease
21:45.21tainted-skyboy what are u trying to do?
21:46.56*** join/#asterisk dsfr (n=dsfr@pdpc/sponsor/digium/dsfr)
21:46.59skyboytainted: I am actually trying to track down a weird situation in SER config to proxy to asterisk. Ive come to understand the "vernacular" used in ser but doesnt understand the resolution bits.
21:47.05AursRenacor: don't know if there is a howto... but if there is, google will know ;)
21:47.10skyboyhere is an example -
21:47.27DabianWhat makes digum cards pricey?  What can they do, that your old soundcard with FXO wont do?
21:47.46*** join/#asterisk L|NUX (n=linux@202.5.145.58)
21:48.15Nuggetwork without you having to write your own drivers.
21:48.21eric_hillCan anyone tell me how to automatically log off a dynamic ACD agent if they don't answer?  The automatic logoff function only seems to work for agents defined in agents.conf...
21:48.23DabianDo they encode the voice in the right codec?
21:48.55skyboyif (uri=~"^sip:\+[0-9]+@192.168.0.1") {
21:48.55skyboy<PROTECTED>
21:48.55skyboyrewriteport("");
21:48.55skyboy<PROTECTED>
21:48.55skyboy<PROTECTED>
21:49.01skyboysorry about that...
21:49.07DabianNugget : Ahh ok.  Thats of course an interesting point.  Writing drivers can be quite timeconsuming. :-)
21:49.11skyboybut thre it is...
21:50.00skyboytainted: basically the there is a forward im guessing to a machine known as pbx.foo.org but when I do a dns lookup there is no such machine..weird.
21:50.10skyboyam I reading the config file correctly?
21:51.37Renacorhmm i got the cdr in a postgres database, however I don't think there is any way to get call statistics for agents answering a queue from the data in the cdr is there?
21:52.48skyboyany ideas
21:54.43*** join/#asterisk DeeJay[2] (n=bleh@office.abi.ca)
21:54.45DeeJay[2]Hi!
21:54.55DeeJay[2]Has anybody been able to use BLF with polycom phones?
21:55.40Kattyhihi.
21:55.42triple-ewhats BLF ?
21:55.53Qwell[]~blf
21:55.55jbotextra, extra, read all about it, blf is Busy Lamp Field, aka little lights next to speed dials that light up when the person is on the phone and blink when that line is ringing.  hint extensions are static mapped to SIP or other channels.
21:56.34Dabianjsharp : Thanks for the advise .. I will look into the PRIs.
21:59.02websaedoes anyone here have experience with ASTBILL?
22:00.01lokkjuwtf is up with this - no sound from Playback, using ztdummy - everything looks ok, but it as if it has no timer source - any way to actually test that ztdummy is fully working?
22:11.15*** join/#asterisk zaf (n=zaf@wsip-68-228-9-79.br.br.cox.net)
22:11.51*** join/#asterisk Saturn-- (i=Saturn@24.50.85.195)
22:13.51DabianIs it true that asterisk doesn't support crypto-sip?
22:14.09*** join/#asterisk MattB2 (n=MattB2@mail.tricycleinc.com)
22:14.15MattB2hi all
22:14.17MattB2got a qq
22:14.41MattB2if someone calls my asterisk and in the dialplan i set a variable, will that variable already be set when another user calls in?
22:14.47MattB2or are they per-call
22:18.28*** join/#asterisk Strom_M (n=strom@gateway.digium.com)
22:19.38*** join/#asterisk alephcom (n=alephcom@host75.net14.mcsnet.ca)
22:21.45PakiPenguintzafrir, around?
22:22.05Renacoranybody know an app that can work with a postgres cdr and create statistics for answered calls by agents?
22:22.20*** part/#asterisk MattB2 (n=MattB2@mail.tricycleinc.com)
22:22.21GamercjmIm having problem with SIP to SIP, like the users cannot hear each other
22:22.44websaeSIP to SIP is terrible
22:22.47websaebecause of RTP
22:22.50websaeneed to use IAX :)
22:23.02Gamercjmyah but i need to use sip
22:23.10Gamercjmi have the RTP ports open on the server
22:24.26*** join/#asterisk MacDome (n=eseidel@A17-255-104-111.apple.com)
22:25.10*** part/#asterisk austinnichols102 (n=austinni@70.46.69.131)
22:26.20mitchelocRenacor: snap snap snap ;) (well not yet, but it will)
22:27.06lokkjuRenacor, would be easy to do - just write a couple sql queries, and a php page to display em
22:28.57*** join/#asterisk Gunnar (n=gunnar@101.Red-80-24-171.staticIP.rima-tde.net)
22:30.40*** join/#asterisk Maxxed (n=user@cpe-72-177-150-20.houston.res.rr.com)
22:30.45Maxxedyo fellas :)
22:31.42Maxxedgota odd one
22:31.42Maxxed> reload chan_zap.so
22:31.42Maxxed-- Reloading module 'chan_zap.so' (Zapata Telephony)
22:31.42Maxxed== Parsing '/etc/asterisk/zapata.conf': Found
22:31.42MaxxedApr 11 16:31:14 ERROR[4284]: chan_zap.c:10309 setup_zap: Unable to reconfigure channel '1'
22:31.42MaxxedApr 11 16:31:14 WARNING[4284]: chan_zap.c:11069 reload: Reload of chan_zap.so is unsuccessful!
22:31.54Maxxedand idea?
22:31.58*** join/#asterisk Gamercjm (n=chris@pool-71-254-174-212.lsanca.fios.verizon.net)
22:31.59Maxxedmy cfgs look stright
22:32.03Maxxedmods look loaded
22:32.10Maxxedztcfg is showing sweet
22:32.38Maxxedtdp400p
22:32.44Maxxedtwo fxo ifaces
22:32.50Maxxedusing fxsks signaling
22:33.19Maxxedim about to recompile
22:35.31key2someone has an idea of why caller ID on X100P doesn't work ?
22:35.49Maxxedim not sure, iv never used the x100p's
22:35.54Maxxedit should suport it if im not mistaken
22:36.23Maxxedcheck ye ol zapta.conf
22:36.33Maxxedusecallerid=yes
22:36.47key2yeah
22:37.02Maxxeddo you have caller id service?
22:37.14Maxxedi have to pay a few bux extra a month for it on my analog crap
22:37.25key2yeah
22:37.32key2since with a phone I see the number
22:37.47key2so I do have
22:37.53Maxxedare you trying to set the outgoing caller id ?
22:37.58Maxxedor just view the incoming
22:37.58key2no
22:38.05key2view incomming
22:38.07Maxxedum
22:38.15key2it's not possible on analog line to set outgoing callerid
22:38.24Maxxedok, so you know that much ;)
22:38.35Maxxedive seen alota folks come thru that didnt know that
22:38.36Maxxedheh
22:38.36key2lol
22:38.42Maxxedlol n'deed
22:39.05key2is it possible to be a Line/Phone issue ?
22:39.55key2I tryed with both actually and it's the same :)
22:39.56Maxxedit may be.. do you have a caller id u can plug in
22:39.56Maxxedto test if your id service is working
22:39.56skyboyasked on the ser forum..but maybe someone can help me here ---how do you query a dns server for srv records?
22:39.56key2Maxxed: yeah, I tryed with an analog phone
22:39.56Maxxedor a plane ol phone that supports callerid
22:40.03*** join/#asterisk CpuID2 (n=none@gentoo/contributor/cpuid)
22:40.08key2and it gives me the id
22:40.08Maxxedskyboy: are you familair with dig?
22:40.24Maxxedkey2: um, sounds like a config some where
22:40.40Maxxedkey2: cuz im pretty sure the xp100p's have caller id support
22:40.42skyboyMaxxed: familiar in that I have used to do a dig against a dns server..
22:41.38skyboyMaxxed: but im perplexed because I have record that forwards in SER to pbx.foo.org but doing an nslookup against that resolves to nothing...so im thing where is that machine?
22:42.05Maxxedkey2: dig @dns1.whateversrever.com domain.net any
22:42.08skyboyMaxxed: Am I thinking of it wrong?
22:42.09Maxxedkey2: i think.. ;p
22:42.37Maxxedwops
22:42.38Maxxedwrong nicks ;p
22:42.38skyboyn/p
22:42.38skyboyI got it ;)
22:42.39Maxxedrockin :)
22:42.46*** join/#asterisk Gamercjm (n=chris@pool-71-254-174-51.lsanca.fios.verizon.net)
22:43.00Gamercjmok so im still having a sip to sip audio problem
22:43.02skyboybut ill have to check if it resolves back :)
22:43.07Gamercjmanyone know how to get that working
22:43.39Maxxedkey2: go over the configs, check logs, what kinda phones are you using?
22:44.05Maxxedkey2: might have some wacky extentions.conf messin it up
22:44.13*** join/#asterisk doughecka (n=Miranda@unaffiliated/doughecka)
22:44.56Maxxedanyone have any idea why im geting this " setup_zap: Unable to reconfigure channel '1' " crap ?
22:45.12Maxxedim about to recompile, maybe somthing gone south the frist go around
22:45.36Maxxedasterisk seems to be working well, minus the tdm400p
22:45.49Az_audid the kernel modules cause any errrors?
22:45.54Maxxednope
22:45.57Maxxedthey loaded fine
22:46.02Az_autry rmmod and modprobe again?
22:46.21Az_auif not i'd recompile both zaptel and asterisk
22:46.31Maxxedyeah..
22:46.34MaxxedApr 11 15:29:55 host kernel: Zapata Telephony Interface Registered on major 196
22:46.34MaxxedApr 11 15:29:55 host kernel: Zaptel Version: 1.2.4 Echo Canceller: KB1
22:46.34MaxxedApr 11 15:29:55 host kernel: ACPI: PCI interrupt 0000:01:01.0[A] -> GSI 16 (level, low) -> IRQ 193
22:46.34MaxxedApr 11 15:29:55 host kernel: Freshmaker version: 71
22:46.34MaxxedApr 11 15:29:55 host kernel: Freshmaker passed register test
22:46.35MaxxedApr 11 15:29:55 host kernel: Module 0: Installed -- AUTO FXO (FCC mode)
22:46.37MaxxedApr 11 15:29:55 host kernel: Module 1: Installed -- AUTO FXO (FCC mode)
22:46.39MaxxedApr 11 15:29:55 host kernel: Module 2: Not installed
22:46.41MaxxedApr 11 15:29:55 host kernel: Module 3: Not installed
22:46.43MaxxedApr 11 15:29:55 host kernel: Found a Wildcard TDM: Wildcard TDM400P REV E/F (2 modules)
22:46.45MaxxedApr 11 15:29:55 host kernel: Registered tone zone 0 (United States / North America)
22:46.49Maxxedthx for the pointers Az_au
22:46.50eric_hillCan anyone tell me how to automatically log off a dynamic ACD agent if they don't answer?  The automatic logoff function only seems to work for agents defined in agents.conf...
22:47.04skyboyMaxxed: it comes back with the authoritative dns server but no record per say
22:47.09Maxxed# lsmod
22:47.09Maxxedwctdm 40192 0
22:47.10Maxxedzaptel 228772 5 wctdm
22:47.26Maxxedskyboy: try dig @dns1.whateversrever.com domain.net ser
22:47.34skyboyokay
22:47.50Maxxeddnstools.com might have an easy point click way to find out
22:48.02Maxxedwe i mean dnsstuff
22:48.02Maxxedhttp://www.dnsstuff.com/
22:48.06Az_auMaxxed: i've got 3 modules loaded (i'm using fxs and fxo)
22:48.07Az_auwcfxo                  17312  0
22:48.07Az_auwctdm                  41920  2
22:48.07Az_auzaptel                192516  10 wcfxo,wctdm
22:48.07Maxxeds/we/er
22:48.08Maxxedheh
22:48.24Maxxedum, i dont have the 3rd
22:48.27Maxxedwcfxo ?
22:48.34Az_aui think it's an alias
22:48.35Maxxedyou have a tdm400p ?
22:48.36Az_aulemme check
22:48.37Az_auya
22:49.02Maxxedthe readme states, wctdm or wcfxs
22:49.05Maxxedi think..
22:49.15*** join/#asterisk GTX (n=charlie@pdpc/supporter/monthlybronze/GTX)
22:49.17Az_aualias wcfxs wctdm
22:49.22GTXGuys, I have an snom 360 and in the SIP settings it says "Music On Hold Server" does anyone know one of these?
22:49.23Maxxedwctdm or TDM400P - Modular FXS/FXO interface (1-4 ports)
22:49.23Maxxedwcfxs
22:49.29Az_auso my fxs is covered by wctdm but fxo seems to be it's own
22:49.45Az_aumy modprobe.conf lists as follows for fxo:
22:49.46Az_auinstall wcfxo /sbin/modprobe --ignore-install wcfxo && /sbin/ztcfg
22:49.47Maxxedi have two fxo modules on the tdm400p
22:49.51skyboyMaxxed: okay...the dig im doing is supposed to be asterisk box so does the prvious dig command still hold?
22:50.02skyboyHere is the config -
22:50.44skyboyif (uri=~"^sip:\+[0-9]+@someip") {
22:50.44skyboy<PROTECTED>
22:50.44skyboyrewriteport("");
22:50.44skyboy<PROTECTED>
22:50.44skyboy<PROTECTED>
22:50.56Maxxedinstall wcfxo /sbin/modprobe --ignore-install wcfxo && /sbin/ztcfg
22:51.19skyboyMaxxed: the pbx.foo.org appears to me where SER is forwarding correct?
22:51.22Maxxedalias wcfxs wctdm
22:51.44*** join/#asterisk Op3r (n=op3r@203.82.42.10)
22:51.46Maxxedskyboy: um.. not sure, i havent toyed with any dns stuff in a long while ;\
22:51.51*** join/#asterisk melange8272 (n=melange8@ool-4576ab1f.dyn.optonline.net)
22:53.02Az_aumaxed: my startup modprobe goes like this:
22:53.05Az_aumodprobe zaptel
22:53.05Az_aumodprobe wcfxs
22:53.05Az_aumodprobe wcfxo
22:53.29Maxxedi dont have any fxs modules, so i wouldnt think i would need it
22:53.49Az_aunah proll ynot
22:54.01Maxxedim gona try and recompilke
22:54.05Maxxedblah
22:54.07Maxxedrecompile
22:54.46Maxxedi do remember having a few small issues with the zaptel compile. dependcy issues i think
22:54.59Maxxedil swing back by and harass you guys if i need anything
22:55.03Maxxedthanks again Az_au
22:55.06Az_aunp
22:55.25*** join/#asterisk Kokey (n=jramirez@dsl-200-78-65-27.prod-infinitum.com.mx)
22:55.52*** part/#asterisk GTX (n=charlie@pdpc/supporter/monthlybronze/GTX)
22:57.13GamercjmIs there anyway to get the rtp thing in SIP to not have to open rtp ports on the client router?
23:00.00*** join/#asterisk MacDome (n=eseidel@A17-255-104-111.apple.com)
23:00.22Sedoroxexit
23:00.22Sedoroxexit
23:00.24Sedoroxerrrrr
23:04.16GamercjmHow do i add a SDP thing
23:08.29melange8272anyone know if sip digits are spit out over the wire in call (sniffable?) I know the invite is easy to snag, but not sure if digits are sent rtp during the call
23:10.51Maxxedwell hell ;p
23:11.03MaxxedAz_au: i recompiled the zzaptel drivers, they work like a champ now
23:11.04Maxxedheh
23:11.07Maxxedhow bout that ;)
23:11.18Qwell[]melange8272: yes, dtmf is in the rtp
23:12.03melange8272sniffable in ethereal.. Guess I won't check my bank balance from my sip phone ;)
23:15.20*** join/#asterisk timscott (n=a@d198-166-221-177.abhsia.telus.net)
23:15.23*** join/#asterisk gbodemantv (n=gbodeman@216.142.38.154)
23:15.25gbodemantvhi all
23:15.31gbodemantvso who is using xlite?
23:17.11*** join/#asterisk MRH2 (n=Mr_happy@fcirc-adsl.demon.co.uk)
23:17.37tainted-they're all trying to figure out what's wrong with it right now
23:17.49MRH2hi can anyone tell what : #define MONITOR_CONSTANT_DELAY does in channel.c
23:18.10*** join/#asterisk nite (n=nite@gateway.digium.com)
23:19.30*** join/#asterisk Jaxxan (n=jaxxan@202.70.125.124)
23:19.58*** join/#asterisk MoutaPT (n=MoutaPT@85.139.183.36)
23:20.47MoutaPTHi, does any one knows or could guess why Xlite is taking like 5 or 6 seconds to Hung up a call, ASterisk is 1.2.5?
23:24.00FLeiXiuSlol to hung it up?
23:24.05FLeiXiuSHang up a call maybe?  ;_)
23:25.37MoutaPTyou r right... thks
23:25.46MoutaPT:) my english isn't so good
23:26.05MoutaPTanyways you got the question?:)
23:26.15MRH2is there some emergency going on?
23:26.59MRH2this channel seems a bit lite at the moment
23:26.59MoutaPTMRH2 what kind of emergency? not just a normal call...
23:27.13MoutaPTyeah u r right MRH2
23:28.48gbodemantvtrying to turn off call waiting sound in xlite
23:28.53gbodemantvanybody have any idea how
23:28.54gbodemantv??
23:31.25*** join/#asterisk Pazzo (n=thomas@host130-250.pool8172.interbusiness.it)
23:32.40*** join/#asterisk mrbnet (n=sureal@cust-static194-37.BHI.COM)
23:33.46mrbnethas anyone here setup a DID using virtualphoneline.com that could answer a couple q's for me?
23:38.06ljamhow might one go about logging an agent out of a queue?
23:43.16[av]banihttp://siln.livejournal.com/235478.html
23:44.41*** join/#asterisk brif8 (n=Techno@lazyjtrainingcenter.com)
23:45.53brif8I have  IP Phone <----> * Box 1  <--->  Gateway (NAT) <---> Internet  <---> Gateway Nat  <---> * Box 2  <---> T1 Card.  How can I route calls from the T1 Card on Box 2 to IP Phones on Box 1  ?
23:47.52tzangerwerd to the goat herd
23:48.06timscottbrif: over the internet
23:48.07timscott:p
23:48.28timscottasterisk, man. install and set it up.
23:50.04skyboyhello..im incurring some errors when bulding zaptel-1.2.5 on centos4 --> errer:syntax error before "zone_lock" can someone help out?
23:50.14skyboythen followed by a few more ;)
23:51.09skyboyI set up the simlinks before the make properly so Im perplexed as to what package Im missing..or??
23:51.45X-Robskyboy,
23:51.50X-Rob~centosbug
23:51.51jbotfrom memory, centosbug is a problem with the latest Centos kernel (4.2 and 4.3).  To fix it, edit the file /usr/src/kernels/2.6.9-34.EL-i686/include/linux/spinlock.h and change 'rw_lock_t' on line 407 to 'rwlock_t'. This is part of the 'kernel-devel' package.
23:52.23skyboyjbot: thanks...
23:52.23jbotskyboy: sure thing
23:52.48brif8timscott:  I'm trying to connect via IAX2 but and I read the wiki on dual servers but for some reason when I dial the exten on Box 2 to ring a phone on Box 1 it rings once and then disconnects ??
23:52.52*** join/#asterisk wgroh (n=chatzill@69-163-232-176.atlsfl.adelphia.net)
23:53.20*** join/#asterisk nite (n=nite@gateway.digium.com)
23:53.37skyboyjnot: one more thing am I playing with "fire" with gas on my hands ;) by using 4.3 or shoud the rest of the install go smoothly. In particular what is the recommended version to go with for stability..
23:53.48X-Robskyboy, jbot is a bot.
23:53.53brif8I was hoping someone might be able to assist what dumb mistake I am making in the config or something.
23:54.16*** join/#asterisk rva (n=rafa@200.210.51.130)
23:54.17X-Robbrif8, usually codec incompatibilites. look in /var/log/asterisk/full
23:54.23skyboyahh...learning something new everyday ;)
23:54.38rvahi guys....could someone give me a little help with realtime?
23:54.59rvaasterisk seems not to be sending the user authentication to the realtime....mysql
23:55.20skyboyhmm...is 4.3 version of centos stable for * or should I be using a previous version for stability?
23:55.22brif8X-Rob: there is no /var/log/asterisk/full
23:55.32brif8events and messages
23:55.37X-Robbrif8, edit /etc/asterisk/logger.conf and turn it ou
23:55.50wgrohanyone having DTMF problems with the TE406
23:55.51X-Robskyboy, centos4.3 is heavily used
23:55.54wgrohthe VPM
23:56.18skyboysweet
23:56.28skyboyX-Rob: thanks
23:56.40brif8messages => error,warning   and console => error,notice,warning;,info
23:58.41*** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net)
23:58.55brif8X-Rob: I see   Unable to create translator path for unknown to slin on IAX2/lecanto-16384  in messages
23:59.04wgrohlooks like the TE420 is the same as TE406 but with hardware DTMF
23:59.14Qwell[]hardware dtmf?
23:59.23Qwell[]I think you mean echo cancel...
23:59.27tzangerno
23:59.28wgrohnegative
23:59.37tzangerhardware DTMF detection and other little goodies
23:59.38Qwell[]does the TE420 even exist?
23:59.40brif8also  chan_iax2.c: Max retries exceeded to host 71.41.50.162 on IAX2/lecanto-16384 (type = 6,
23:59.42wgrohchan_zap.c: Detected digit 'A'
23:59.45wgrohchan_zap.c: Detected digit 'D'
23:59.57wgrohTE406 is killing my team

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