irclog2html for #asterisk on 20040115

00:28.51*** join/#asterisk jbot (ibot@c-24-1-99-18.client.comcast.net)
00:28.51*** topic/#asterisk is Asterisk 0.7.1 Released! Please report any bugs at http://bugs.digium.com
00:29.00FuzzyCatjbot you slag!
00:29.03FuzzyCat: are you using Windows?
00:29.03*** join/#asterisk davel (~davel@frobnitz.dsp-services.com)
00:29.24FuzzyCatjbot: shut it slapper
00:29.28FuzzyCat: I don't know, could you explain it?
00:29.33FuzzyCatno
00:33.25mishehuFuzzyCat: whats error 207?
00:33.29mishehuerr issue
00:33.32jimmyzanyone got an idea how i can make all calls conference then dial the number they called?
00:34.34mishehu~theanswer jimmyz
00:34.37jimmyz: 42
00:35.01*** join/#asterisk plc5_250 (~chatzilla@pcp03527486pcs.pntiac01.mi.comcast.net)
00:35.04FuzzyCatmishehu: http://bugs.digium.com/bug_view_page.php?bug_id=0000207
00:35.59UnixDawgdo you have to put anything in modules.con to get it to record to pgsl
00:36.00mishehuFuzzyCat: that does not like to work in links it appears
00:36.00*** join/#asterisk brown (~brown@sp1024.rbs-p01.ewol.com)
00:36.14FuzzyCat?
00:36.16UnixDawgdo you have to load a module
00:37.29jimmyzmishehu: what was that
00:37.34*** part/#asterisk ecd (~dslater@ool-44c17c85.dyn.optonline.net)
00:37.35*** join/#asterisk Zebble (~Zebble@Sherbrooke-HSE-ppp3610678.sympatico.ca)
00:37.50FuzzyCatmishehu: its a SIP issue... if you make a sip call, pull the cable out of your phone (to simulate a client crash) reboot it and put the cable back in  the channel never dies in * unless you restart, or do a soft hangup
00:38.59FuzzyCatthat means if you do billing the billing goes on until you kill it
00:39.23*** part/#asterisk _Yog_ (~magnus@hades.27b-6.de)
00:42.19mishehuFuzzyCat: hmmm...  I did finally get to get the page up...  so SIP doesn't do any checking to see if a connection is still alive or not...  interesting.
00:42.55FuzzyCatyes.. i think it's the same for h323 and mgcp... but I've not tested those...
00:44.16*** join/#asterisk simprix (~simprix@24.247.181.124.gha.mi.chartermi.net)
00:44.27TangentI'm trying to work out where/how I can add in voicemail normalisation to make all the quiet messages louder when they're emailed to me... Would that be app_voicemail.c ?
00:44.31simprixwhen i use asterisk i get this with a sip xten client
00:44.32UnixDawgok anyone out here get *69 working on the pbx
00:44.41simprixJan 14 19:40:44 NOTICE[245776]: chan_sip.c:3635 register_verify: Peer '2000' isn't dynamic
00:44.41simprixJan 14 19:40:44 NOTICE[245776]: chan_sip.c:5405 handle_request: Registration from 'simprix <sip:2000@10.0.0.244>' failed for '10.0.0.242'
00:44.41simprixJan 14 19:40:44 NOTICE[245776]: chan_sip.c:3635 register_verify: Peer '2000' isn't dynamic
00:44.41simprixJan 14 19:40:44 NOTICE[245776]: chan_sip.c:5405 handle_request: Registration from 'simprix <sip:2000@10.0.0.244>' failed for '10.0.0.242'
00:44.42simprixJan 14 19:40:44 NOTICE[245776]: chan_sip.c:3635 register_verify: Peer '2000' isn't dynamic
00:44.44simprixJan 14 19:40:44 NOTICE[245776]: chan_sip.c:5405 handle_request: Registration from 'simprix <sip:2000@10.0.0.244>' failed for '10.0.0.242'
00:44.50LinkIS Build 0.7.1 done?
00:44.51simprixsorry
00:44.53*** join/#asterisk PBXtech (Reggie@65.218.37.248)
00:45.02FuzzyCatLink: topic!
00:45.13km-I wish I remember where I put my tradewars registration code
00:45.20LinkIs this build available?
00:46.39simprixcan someone help me
00:47.01Tangentsimprix: Depends ;)
00:47.21simprixi get this error Jan 14 19:44:00 NOTICE[245776]: chan_sip.c:5405 handle_request: Registration from 'simprix <sip:2000@10.0.0.244>' failed for '10.0.0.242'
00:47.34simprixbut everything looks ok im using the xten client on os x
00:48.30TangentI get those messages for one of my clients too.. but it all seems to work ok
00:48.36TestMasTersimprix,  i`m having the same problems
00:48.44simprixmine won't log in
00:48.51simprixi can't get it working at all
00:48.52FuzzyCatshitty client?
00:49.01simprixxten x-lite
00:49.19simprixi had to do some stupid stuff last time i got it working but i don't remember what
00:49.33TestMasTerFuzzyCat,  agreed
00:49.37TestMasTerbut its needed atm
00:49.59FuzzyCatsjhpone no good TestMasTer?
00:50.06TangentI had to get rid of host=dynamic on my BT101 to make it work
00:50.07FuzzyCatsjphone
00:50.18simprixdid you just take out the line
00:50.31TestMasTerFuzzyCat,  haven`t tryed it i`m using Xlite for other people and it works fine
00:50.37mishehuFuzzyCat: I am gathering that IAX does check to see if a link is still alive then?
00:51.04TangentI made it host=192.168.254.15 instead... but I've no idea if that's valid... but it 'worked for me'(tm)
00:51.19FuzzyCatwell,thats another problem atm... calls don;t appear to disconnect at either end in that situation
00:51.40UnixDawgtest what error
00:51.59UnixDawgI keep getting a 484
00:52.32killerbeeyo yo yo... killerbee is in DA HOUSE
00:52.45*** join/#asterisk Mike1 (~Mike@hoochie.digium.com)
00:53.00Mike1Help!
00:53.02Mike1=)
00:54.23simprixtangent when i do that i get this Jan 14 19:51:11 NOTICE[245776]: chan_sip.c:3635 register_verify: Peer '2000' isn't dynamic
00:54.24simprixJan 14 19:51:11 NOTICE[245776]: chan_sip.c:5405 handle_request: Registration from 'simprix <sip:2000@10.0.0.244>' failed for '10.0.0.242'
00:55.32Tangentsimprix: I just PM'd you the config for my BT101
00:57.12joakoIs it possible to detect a fax over IAX or SIP or only when the call comes from a zaptel device?
00:57.32*** join/#asterisk mvand (~mvand@CPE-24-27-138-147.neb.rr.com)
00:58.15mishehujoako: why would you fax over VoIP instead of scanning in the document and emailing it or something?
00:58.59jimmyzwe voip to a faxserver that e-mails them
00:59.14jimmyzgot to get the call in to a fax server or machine
00:59.48jimmyzif your voip only how do you get faxes then
00:59.53mishehufaxing to a faxserver that then emails them?
01:00.07mishehuI'm still confused...
01:00.13joakonooo, I mean when you have a fax => extension, can asterisk detect that it is a fax call when its going over IAX?
01:00.32jimmyzdon't understand that though....
01:00.42jimmyzwhy not have an extension for faxes only
01:01.21joakodont ask me, ask whoever thought of that feature...
01:02.06*** join/#asterisk dolbe (~fujitsu@pcp04566804pcs.jersyc01.nj.comcast.net)
01:02.39jimmyzmishehu: on my setup some extension has to receive the fax and we are voip only
01:02.52mishehujimmyz: I am probably wrong about this, but I think that's a part of the zaptel and not asterisk itself that does the fax detection
01:03.06jimmyzok could be right
01:03.18joakojimmyz: but does fax detection work over IAX?
01:03.28jimmyzjoako: don't know
01:03.33*** join/#asterisk Cripon (~nothanks@pcp02001453pcs.huntsv01.al.comcast.net)
01:04.44mishehucan't you create virtual zap devices?
01:04.51mishehuif so, you might be able to get that to work.
01:04.53Criponhttp://www.yottadot.org/download.php?op=viewsdownload&sid=10
01:05.08jimmyzjoako: are you trying to share one inbound line for voice and data
01:05.38jimmyzdata=fax
01:13.47dalaberaWould anyone tell me where can I download the cisco 7960 image files to upgrade my phone besides http://www.loligo.com
01:14.47dolbedoes asterisk support those?
01:15.54simprixwhen i dial extension 2999 which is supposed to go straight to voicemail it gives me a 404 error
01:16.12simprixand the console says Jan 14 20:10:51 NOTICE[245776]: pbx.c:1211 pbx_extension_helper: Cannot find extension context 'default'
01:17.55bkw_Corydon76-work that patch works perfect
01:18.03bkw_now we just have to get kram to approve the changes
01:19.52UnixDawgman asterisk is growing so fast
01:20.27jimmyzthink i got my company to switch to it now
01:20.41plc5_250hey folks - I'm having a bit of a problem getting asterisk to work with my (new) TDM-40
01:20.55bkw_ok now lets get sip debug to accept sip debug peername
01:20.55bkw_"sip debug peername"
01:20.58jimmyzwant to guess at the cost to record phone calls on two offices about 150 phones...record only no ccm
01:21.01plc5_250Jan 14 19:32:36 ERROR[16384]: chan_zap.c:5189 mkintf: Unable to open channel 1: No such device
01:21.41jimmyz110k
01:27.08bkw_HAHAHAHA
01:27.10bkw_chan_sip.c
01:27.11bkw_#ifdef THE_SIP_AUTHORS_CAN_SUCK_MY_GONADS
01:28.38jimmyzhow big are gonads
01:29.32rollysonhmmm. is there a way to kill MOH on a snom 200?
01:30.12bkw_GOTTA be a way
01:30.20bkw_sip debug [peername]
01:30.29bkw_that will be handy if you have a loaded asterisk box
01:30.31bkw_DAMN IT
01:30.51*** join/#asterisk VoiceLynx (VoiceLynx@rrcs-central-24-106-64-175.biz.rr.com)
01:31.03rollysonhmm.. everybody leave the conf?
01:31.51joakoyes, everyone's gone
01:31.59joakowhat do you mean by kill MOH on a snom 200?
01:32.29*** join/#asterisk af_ (af@ip381-35-1.adsl.edisontel.com) [NETSPLIT VICTIM]
01:32.29*** join/#asterisk Tangent (authdenied@connerdata-6.dsl.easynet.co.uk)
01:32.29*** mode/#asterisk [+bbb *!*FearOfGOD@*.Level3.net *!*PooPoo@*.ntli.net *!*@*.va.client2.attbi.com] by capek.freenode.net
01:32.29*** mode/#asterisk [+bbb *!*carlossan@*.attbi.com *!*NoHost@*.verizon.net *!*globalthr@*.attbi.com] by capek.freenode.net
01:32.29*** mode/#asterisk [+bbb *!*shane@*.attbi.com *!*tofubar@*.attbi.com *!*@og.latency.net] by capek.freenode.net
01:32.29*** mode/#asterisk [+b *!*dan@194.158.*.*] by capek.freenode.net
01:32.29*** mode/#asterisk [+q sant!*@*] by capek.freenode.net
01:32.41TangentWhee... ride the irc rollacoaster
01:33.12rollysonjoako: I'd like to temporarily disable music on hold ;)
01:33.26rollysonso that when I put someone hold they will get dead air
01:33.33ZX81why
01:33.56rollysonZX81: in case one of the lines on hold is a conf.
01:34.03ZX81aha
01:34.13simprixwhat module's do i have to load for the x100p
01:34.29joakothis is when you press the hold button on the phone?
01:34.37rollysonyeah
01:34.51*** join/#asterisk root (~root@24.214.255.57)
01:35.06rootblah
01:35.36simprixhow do i get the x100p to work
01:35.41*** join/#asterisk angler__ (~angler@24.214.255.57)
01:36.07plc5_250this is weird - working with a tdm40 (4 mod).  I can dial from extension 1 to extension 4 no problem, but dialing from extension 4 to extension 1 gives a busy.
01:36.24joakocreate a new entry in your musiconhold.conf with an empty directory
01:36.34joakoand in your extensions.conf do
01:36.44joakoexten => xxxx,1,SetMusicOnHold(silent)
01:36.50joakoexten => xxxx,2,MeetMe.....
01:38.25*** join/#asterisk juice (~juice@user221.net1199.mo.sprint-hsd.net)
01:38.28simprixi have this in my zapata.conf how do i get this to transfer to voicemail right away and to accept incoming calls ;
01:38.28simprix; Zapata telephony interface sample configuration file
01:38.28simprix;
01:38.29simprix[channels]
01:38.29simprix;
01:38.30simprix; X100P plugged into PSTN
01:38.32simprix;
01:38.36simprixcontext=incoming
01:38.36jimmyzis there anything wrong with the T400P?
01:38.38simprixsignalling=fxs_ks
01:38.40simprixechocancel=yes
01:38.42simprixechocancelwhenbridged=yes
01:38.44simprixrelaxdtmf=yes
01:38.46simprixrxgain=1.5
01:38.51simprixtxgain=1.5
01:38.52simpriximmediate=no
01:38.54simprixbusydetect=no
01:38.56simprixcallprogress=no
01:38.58simprixmusiconhold=default
01:39.00simprixusecallerid=yes
01:39.02simprixcallerid=asreceived
01:39.06simprixchannel => 1
01:39.22Tangentsimprix: put some entries for extension s in your [incoming] section of extensions.conf
01:40.13simprixok i can't get asterisk to open now that i edited zapata.conf
01:40.28simprixdo i need to load the modules first
01:41.26Tangentsimprix: I load the modules from /etc/modules myself.. not sure if asterisk autoloads them
01:41.58simprixwell now that i edited that asterisk won't load
01:42.51plc5_250is IAX1 strictly TCP, or does it use UDP as well?
01:43.37Tangentsimprix: you want wcfxo and zaptel modules loaded
01:43.54simprixJan 14 20:40:41 WARNING[16384]: Unable to specify channel 1: No such device or address
01:43.54simprixJan 14 20:40:41 ERROR[16384]: Unable to open channel 1: No such device or address
01:43.55simprixhere = 0, tmp->channel = 1, channel = 1
01:43.55simprixJan 14 20:40:41 ERROR[16384]: Unable to register channel '1'
01:43.55simprixJan 14 20:40:41 WARNING[16384]: chan_zap.so: load_module failed, returning -1
01:43.55simprixJan 14 20:40:41 WARNING[16384]: Loading module chan_zap.so failed!
01:43.59simprixi have those modules loaded
01:44.55bkw_BUG 850
01:44.57bkw_soemoen care to try that
01:45.51*** join/#asterisk Kumbang (~unknown@167.205.22.54)
01:45.56Tangentsimprix: I can't see anything wrong with the config you pasted
01:47.17plc5_250can someone try to route a call to me - I want to check my firewall forwarding and make sure I have it right.  ip: 68.61.39.105 extension 104
01:49.21simprix<PROTECTED>
01:49.22simprixJan 14 20:46:28 WARNING[16384]: chan_zap.c:659 zt_open: Unable to specify channel 1: No such device or address
01:49.22simprixJan 14 20:46:28 ERROR[16384]: chan_zap.c:5189 mkintf: Unable to open channel 1: No such device or address
01:49.22simprixhere = 0, tmp->channel = 1, channel = 1
01:49.22simprixJan 14 20:46:28 ERROR[16384]: chan_zap.c:7162 setup_zap: Unable to register channel '1'
01:49.23simprix<PROTECTED>
01:49.25simprix<PROTECTED>
01:49.27simprix<PROTECTED>
01:49.29simprix<PROTECTED>
01:49.31simprix<PROTECTED>
01:49.33simprix<PROTECTED>
01:49.37simprix<PROTECTED>
01:49.39simprix<PROTECTED>
01:49.41simprixJan 14 20:46:28 WARNING[16384]: loader.c:312 ast_load_resource: chan_zap.so: load_module failed, returning -1
01:49.41puzzledsimprix: stop flooding the chan
01:49.44simprix<PROTECTED>
01:49.46simprix<PROTECTED>
01:49.50simprixJan 14 20:46:28 WARNING[16384]: loader.c:407 load_modules: Loading module chan_zap.so failed!
01:49.52simprixthats what it says when i try to load it
01:49.52TangentNight all
01:50.01puzzlednite Tange
01:50.04bkw_your config is WRONG
01:50.59bkw_simprix whats up..?
01:51.04simprixbkw_ which config
01:51.11simprixi can't get asterisk to work
01:51.13bkw_zaptel maybe
01:51.21bkw_simprix what are you trying to setup?"
01:51.55simprixa x100p card
01:52.01bkw_let me in the box
01:52.07bkw_give me ssh and root and 2 min
01:52.21bkw_priv msg it to me
01:52.22Exomorph_bkw_: About your reply to bug 571...  That was my plan to add per sip peer debuging.. But wanted to clean up the sip output first.
01:52.44bkw_Exomorph_ ya changing the sipdebug output won't make it into cvs
01:52.56bkw_because the output is the exact format as it is on the wire (or very very close)
01:53.07*** join/#asterisk scat (scat@c-24-126-24-177.we.client2.attbi.com)
01:53.11bkw_atleast not yet
01:53.20Exomorph_bkw_: No its not... and its not at all readable.
01:53.20simprixztcfg says channels not configured
01:53.37scatHey -- I'm new to Asterisk & VoIP -- is there any Newbie FAQs out there?
01:53.42bkw_simprix whats in your zaptel.conf /etc/
01:53.48bkw_Exomorph_ I have no problems reading it
01:53.53Exomorph_bkw_: What I put in that patch was to make it a little more readable.
01:53.59bkw_ah
01:54.09bkw_what I would like to see is > if we are sending a packet
01:54.09Exomorph_bkw_: Only because you have been reading it for how long now?
01:54.18simprixgot it thankis
01:54.19bkw_and < when we receive a packet
01:54.28Exomorph_bkw_: That was in the plans...
01:54.30bkw_Exomorph_ you get that?
01:54.40bkw_I think tha twould make it more clear if its going in and or out
01:54.54*** join/#asterisk kimo_sabe (foobar@ip68-107-131-120.tc.ph.cox.net)
01:54.55bkw_but the peername hack would be SWEET
01:55.01Exomorph_bkw_: It was kinda in that patch... Maybe I'll work on it some more agian.
01:55.14bkw_Exomorph_ you need to be in the next conf call
01:55.28Exomorph_bkw_: When is it?
01:55.33bkw_the conf calls are very constructive
01:55.39bkw_Exomorph_ we have one daily just about these days
01:56.22Exomorph_bkw_: Hmmm Been busy with other stuff lately...  But I'll try for the next one... Just send me a pm when you figure it out.
01:57.03jimmyzbkw: i need your mind fore a sec lol
01:57.37bkw_nope fixing chan_zap tweaks
01:57.44jimmyzok
01:58.00bkw_can you not unregister a manager event?
01:58.12bkw_SURE AS FUCK CAN
01:58.19bkw_chan_zap doesn't even do any of those
01:58.20bkw_man
01:58.31bkw_or unregister most of the cli stuf
01:58.35bkw_NO WONDER it can't be unloaded
01:58.36bkw_hrm
02:03.01jimmyzanyone up for a dialplan ?
02:04.01*** join/#asterisk igor__ (~igor@66.151.220.60)
02:04.09scatCan anyone direct me to some information on using asterisk & a linux pc (ie, plugging a headset into the soundcard) to make voip calls?
02:04.20scatOr anything at all similar to that?
02:04.26bkw_WELL HOLY FUCK
02:04.33bkw_you can unload chan_zap.so now
02:04.34bkw_haha
02:04.35scatI'm sorry if this isn't the right channel for that
02:04.46scatBut if someone could direct me to something else, I'd reall appreciate it
02:04.49scat+y
02:05.17jimmyzscat: you need a softphone to run on the linux pc?
02:05.25bkw_asterisk*CLI> unload chan_zap.so
02:05.25bkw_<PROTECTED>
02:05.25bkw_<PROTECTED>
02:05.25bkw_<PROTECTED>
02:05.25bkw_<PROTECTED>
02:05.26bkw_<PROTECTED>
02:05.28bkw_<PROTECTED>
02:05.34bkw_AND it doesn't crash
02:05.34bkw_haha
02:05.39igor__hi all, is it possible to set up limit of SIP incoming calls for _all_ users? so, in total i'll have no more than 10 SIP calls
02:06.36bkw_yes
02:06.40bkw_incominglimit=
02:07.42bkw_now why can't I reload it
02:08.52TestMasTerbkw_,  wanted to let you know something "Redhat and Fedora" You can reload the config files via asterisk -rx reload restart but you can not do reload restart and have it update well you are us asterisk -rcv
02:11.11dolbeanyone running asterisk on sparc arch?
02:12.00bkw_TestMasTer speak english?
02:12.05bkw_I don't understand that
02:12.05bkw_haha
02:12.22igor__bkw_: incominglimit is per user, I need something for the whole system or per peer basis
02:12.31bkw_per user
02:13.12bkw_ok you can unload chan_zap but you can't load it back now
02:13.38TestMasTerbkw_,  Ok lol let me tell you again  Redhat/Fedora Os is what this is on,   Anyway if you try to reload ( in the * Console )the extensions config insted of Restarting asterisk It doesn`t work    If you do asterisk -rx reload restart it works
02:13.46TestMasTerbkw_,  is that better
02:14.09rollyson"My caps lock key turns itself on every time I try to type"
02:14.12rollysonARGH!
02:15.07bkw_asterisk -r
02:15.08bkw_reload
02:15.15bkw_or asterisk -rx reload
02:15.17bkw_that works
02:15.41TestMasTerrx reload works Fine... From a shell Console but Reload from in the asterisk console Doesn`t
02:15.58bkw_set verbose 10
02:15.59bkw_and type reload
02:16.05bkw_bet verbose is like 0
02:16.45bkw_I bet this is because the fd isn't getting closed
02:17.15TestMasTerbkw_,  Nopper that doesn`t fix  it
02:17.57TestMasTerbkw_,  Ok  it worked that time
02:20.04TestMasTerbkw_,  thanx for the info.... I personal didn`t care much if it worked.. for what i need to do but i just thought it was a bug so i thought i would tell you
02:22.14*** join/#asterisk SladeAKT (~mirc@user-0c8h5qn.cable.mindspring.com)
02:22.25*** join/#asterisk Kumbang (~unknown@167.205.22.54)
02:23.15mishehubkw_: you guys surprised me with actually having a minor version not zero ;-)
02:23.33bkw_haha
02:23.42bkw_i'm trying to smash this bug in chan_zap
02:23.48bkw_I have it where you can unload chan_zap now
02:23.57mishehuwhat is the bug?
02:24.01denonNew! Asterisk 10.0! Even Easier!
02:24.23bkw_oh its not a bug yet
02:24.28bkw_i'm trying to fix it before i open a bug
02:26.44bkw_where is Moc when you need him
02:27.22SladeAKTis it quiet or am i laggeed?
02:29.33kamileonquiet
02:30.10*** join/#asterisk cypromis (~michael@80.51.246.186)
02:30.41kamileonrollyson : try to call my * server
02:30.44rollysonbkw_: where are we at in the next round of bug squashing?
02:30.49kamileoni need to test iax calls
02:30.58kamileonfrom anywhere out there
02:31.06rollysonkamileon: got a 700 number?
02:31.10kamileonno
02:31.14kamileonhow do i get one
02:31.19kamileoni have an ip address
02:31.35rollysonwww.iaxtel.com
02:31.45kamileonone sec
02:32.02bkw_hrm
02:32.09bkw_rollyson let me get JerJer to look at this one
02:32.14bkw_and we can start a conf here in a few
02:32.54SladeAKTis there a test 700 number?
02:34.35bkw_AHHHH HA
02:34.41*** join/#asterisk Lee3 (~khgg@211.24.146.12)
02:35.04jimmyzsladeakt: use an 800 number 8000-555-1212
02:35.43bkw_no you noob
02:35.46bkw_an iaxtel number
02:35.50bkw_:P
02:36.03*** join/#asterisk james1 (~james@hoochie.digium.com)
02:36.12SladeAKTactually i did that before i set up with voicepulse. I'm more interested in a decent fwd number to test to.
02:36.21james1I am ready to install fedora for my asterisk project
02:36.24Lee3hi bkw_
02:36.26jimmyzoh
02:36.31james1is it better to choose a custome pacakge?
02:36.44james1or I shall go for a workstation
02:36.51jimmyzit was a test number to try
02:36.54james1i have a PII 450MHz machine
02:37.05bkw_NOW lets see if this works
02:37.06james1I want to pout the most juice for * rather than something else
02:37.25james1or should I go for custom installation?
02:37.36james1please help.....thanks in advance
02:37.40bkw_HOLY FUCKING SHIT I DID IT
02:38.01kamileoncan i dial like 'username@server'
02:38.06rob--Hi
02:38.48SladeAKTbkw just out of curiosity, what ya workin on?
02:39.00bkw_SladeAKT fixing chan_zap so you can unload it and reload it from the cli
02:39.04bkw_and it works
02:39.04bkw_haha
02:39.27jimmyzwhat was wrong with the 800# to test
02:39.38kamileoncall 6222
02:39.54SladeAKTkamileon on iaxtel or fwd?
02:39.54bkw_http://asterisk.bkw.org/patch2/chan_zap_unload_reload.diff
02:40.07rob--How should I send audio to a channel? Do I have to worry about timing, or do I just send 1 voice frame for every voice frame received?
02:40.18bkw_MUHAHAHAHAHAHHHHAHA
02:40.18kamileonSladeAKT : i really dont know
02:40.33angler__kamileon your 6222 number is only available to people registered with our digium server
02:40.44kamileonahh
02:40.46rob--How do I know how big to make the voice frame (in samples)
02:40.52kamileonokay well how do i talk to other people
02:40.57james1does it matter what type of installation I choose in terms of * performance on 450MHz PII....please let me know...
02:41.03angler__sign up on iaxtel
02:41.06kamileoni talked to someone in germany this morning
02:41.09kamileoni did
02:41.23angler__you can make local calls thru us
02:41.24kamileonbut i have no gnophone, but i have a login/pass
02:41.28angler__but not ld
02:41.38kamileoni want to call people with iax
02:41.39SladeAKTkamileon, what's your 700 number then?
02:41.42kamileonand let ppl call me
02:41.47SladeAKTand do you have it set up in *?
02:41.53angler__sign up for iaxtel
02:41.57kamileoni havent registered my server with iaxtel yet
02:42.00angler__people can call your local number
02:42.06*** join/#asterisk tim27 (tim27@229-29.dr.cgocable.ca)
02:42.13angler__your local number is 428-6222
02:42.16kamileoni dont know how
02:42.17kamileonyeah
02:42.18kamileonok
02:42.40kamileonso how would i call my friend in romainia ?
02:42.50kamileonor recieve my calls from iraq?
02:42.54james1anyone ...please help me choose an installation type with fedora core on my PII 450MHz so as to get the best perf out of *
02:43.05kamileoncvs!
02:43.21SladeAKTkamileon they would have to be on the iaxtel or fwd network themselves
02:43.23angler__kamileon only if you sign up at iaxtel or fwd or something
02:43.36kamileonthats what i mean
02:43.40kamileoni cant figure out how
02:43.43kamileonim slow you know
02:43.47angler__i doubt mark will let you call international off our lines
02:43.54kamileonive been reading for 3 hours!
02:43.59kamileonnoooo
02:44.06kamileoni just want to call them with iaxtel
02:44.08SladeAKTsounds like you're just getting started then ;)
02:44.12kamileonover ip or something
02:44.21angler__damn kdebase-audiolibs
02:44.29SladeAKTthat's what it's set up for...
02:44.29jimmyzkamileon: you need a server there...our use sip computer to computer
02:44.41kamileoni have a server
02:44.54kamileonhere with a live ip
02:44.58james1anyone ...please help me choose an installation type with fedora core on my PII 450MHz so as to get the best perf out of *
02:45.05jimmyzwith asterisk on it
02:45.11kamileonyeah
02:45.15kamileon24.214.198.33
02:45.16tz-afkjames1: are you having performance issues right now?
02:45.20SladeAKTjames1, do you know what differences the choices make?
02:45.35james1my old installation was crap
02:45.40james1i am starting from scratch again
02:45.46SladeAKTwere you running X on the box?
02:45.50james1got a Adtran 750
02:45.59rob--How should I send audio to a channel? Do I have to worry about timing, or do I just send 1 voice frame for every voice frame received?
02:46.10jimmyzwhat about where you are calling
02:46.24rob--from an app
02:46.25tz-afkI am running * on a P233MMX...  one channel granted but no performance issues, just stay away from iLBC :-)
02:46.27james1SladeAKT, should i choose personal desktop, workstation, server, custom
02:46.43kamileoni run on a 200mmx
02:46.50james1if custom, then can someone list me the minimum set of packages.
02:47.02james1kamileon, what installation option did you choose
02:47.05SladeAKTgiven those options i would say workstation or server, more likely server
02:47.19tz-afkkamileon: how many channels?
02:47.26SladeAKTrob, what kind of app are you writing to the port with?
02:47.42joakojames: custom, and just select development and kernel development (if fendora is anything like prior RH versions)
02:47.46mvandkamileon: how many GSM channels can you run on that?
02:47.49tz-afkcrontibs: did you get your voicepulse working?
02:47.53kamileonjames, i just installed debian sarge, installed nessecary stuff to install * and use the box (in console mode only) then i downloaded and installed asterisk from cvs.digium.com
02:48.02kamileonmvand : i dont know
02:48.09tim27my 7960 is working now
02:48.09tim27:)
02:48.20kamileoni have a x100p and a tdm400 with 2 modules
02:48.34james1okay, serverkamileon, what is the necessary stuff?
02:48.37kamileonan OLD tdm
02:48.40james1is there a list or something?
02:48.46kamileonask the digium folk
02:48.57kamileonim hacking my * box together out of crap parts
02:49.05james1hmmm
02:49.11mvandI have * running with an X100P on a P-233 and it works fine with 1 zap and 1 iax channel
02:49.17angler__what kamileon is doing isnt recommended
02:49.17kamileontheres a hardware list on asterisk.org somewhere
02:49.18angler__lol
02:49.23kamileonunder install i think
02:49.39dougheckasize of a box of ceegarrs
02:49.40mvandsame setup didn't work with a p-166
02:49.59jimmyzdoes the T400P work good
02:50.44joakoi think the digium site recommends a 1ghz+ machine for T1/E1 cards
02:50.56james1mvand, what is your system conf. (did you install as server?)
02:51.06angler__800mhz for TE410P
02:51.48*** join/#asterisk james1 (~james@hoochie.digium.com)
02:52.00james1ooops
02:52.02tz-afkdoughecka: wow... a terabyte for under $1500
02:52.03angler__they need to hurry up and fix this bug in kde package
02:52.03james1lost my connection?
02:52.08dougheckatz-afk: :))
02:52.16james1mvand, what was your install option?
02:52.23james1did you install server as well
02:54.06mvandjames1:  That box is running RH9.0 ...
02:54.46SladeAKTimagine if you have to run raid with two those :)
02:54.49james1did you install with server option or with custom packages
02:54.52mvandDid the Custom install with Kernel Development checked, then applied all RH updates
02:55.07jimmyzmvand: http://216.239.37.104/search?q=cache:zKp5qr9jIYAJ:members.lycos.co.uk/wipe_out/asterisk/asterisk_rh9_install-v1.3.pdf+red+hat+9+asterisk&hl=en&ie=UTF-8
02:55.27jimmyzsorry http://members.lycos.co.uk/wipe_out/asterisk/asterisk_rh9_install-v1.3.pdf
02:55.31james1so you have all access such as ssh., telnet etc?
02:56.21bkw_wooohooooo
02:56.27mvandI do have ssh, but telnet is evil
02:56.47james1thanks mvand and jimmyz for the link
02:56.47angler__bkw_ ?
02:56.55james1i will follow the same procedure and see how it goes
02:56.56bkw_angler__ yes
02:56.58mvandI'm pretty sure that ssh is part of the base install.
02:57.03james1i hope it works out as listed there
02:57.06dougheckaEven if a man chops off your hand with a sword, you still have two nice, sharp bones to stick in his eyes.
02:57.26angler__bkw_ was the wooohoo cause you fixed a bug?
02:57.37dougheckano
02:57.37*** join/#asterisk Link (~svoto@h00095b093ebb.ne.client2.attbi.com)
02:57.40dougheckahe _made_ one
02:57.58angler__lol prolly
02:59.25bkw_angler__ yes
02:59.29bkw_I fixed one
02:59.31bkw_bug 851
02:59.33bkw_go read it
02:59.39bkw_and test if you feel like it
02:59.47dougheckaoh, the bug with the bounty..
02:59.47doughecka:P
03:00.38bkw_no twit
03:01.07mvandjames1:  I almost forgot>  You will need to track down the real mpg123
03:01.41mvandSome RH versions install mpg321 and use a symlink.  Other versions don't install it at all.
03:02.01mvandmpg321 plays back too slow
03:02.31bkw_muahahhahah
03:02.41dougheckabwuhahahaha
03:02.56tim27bkw_ can i setup a mailbox and use it on 2 phone ??? i suppose yes ???
03:03.27*** join/#asterisk Kumbang (~unknown@167.205.22.54)
03:03.45dougheckahttp://www.wired.com/news/medtech/0,1286,61889,00.html?tw=wn_story_top5
03:03.53dougheckawho wants to port iax client to it?
03:04.00dougheckaso we can talk by thinking
03:04.01doughecka:P
03:08.09*** join/#asterisk UnixDawg (~UnixDawg@69-160-1-159.bflony.adelphia.net)
03:08.23mvanddoug:  I won't use it if the driver is called "zap"tel
03:08.55*** join/#asterisk A_Guy (~male@211.24.146.12)
03:08.59*** part/#asterisk A_Guy (~male@211.24.146.12)
03:09.09dougheckalol
03:09.29h3xzaptel stands for zapata telephony
03:10.08*** join/#asterisk ursenj_ (~ursenj@ns.ursey.com)
03:10.22ursenj_what is new in .7.1
03:10.35mvandyeah, I know.  It was a joke, son.
03:11.28dougheckahahahahahahhahahahahhhahahaa
03:12.57mvand100 electrodes in my brain, I don't want to hear the word "zap"
03:13.02h3xoh
03:13.44ursenj_is there is anything major in 0.7.1 that would prompt me for any upgrade
03:13.59rollysontimed includes are fixed ;)
03:14.30ursenj_g729 access to vmail and confrenece??
03:16.39*** join/#asterisk mvand (~mvand@CPE-24-27-138-147.neb.rr.com)
03:16.50bkw_jrollyson
03:16.53bkw_try this
03:16.54bkw_http://asterisk.bkw.org/patch2/pri_kill.diff
03:17.12bkw_we aren't kill the PRI D channel monitor thread in my previous patch this should kill it .. "should" being the keyword
03:17.54dougheckais * c or c++?
03:17.57bkw_C
03:18.04dougheckaehm
03:18.04dougheckawhy
03:18.07JerJer[ghost]C
03:18.07bkw_because
03:18.09dougheckaisnt c++ better?
03:18.16bkw_in who's eyes?
03:18.22dougheckaprogrammers
03:18.24bkw_ok lets write it in GWBASIC
03:18.27bkw_NEXT!!
03:18.30JerJer[ghost]if you like to be that far away from the processor
03:18.35dougheckaoh
03:18.41rollysonbkw_: this is in place of the other patch?
03:18.49bkw_http://asterisk.bkw.org/patch2/pri_kill.diff
03:18.54bkw_use that
03:18.55ursenj_will * work with a 3COM VOIP 1102
03:18.56bkw_before I test it
03:19.01bkw_ursenj_ no
03:19.02doughecka:P
03:19.09rollysonbkw_: instead of the other?
03:19.11bkw_yes
03:19.22bkw_the other didn't kill the pri d channel monitor this should
03:19.25bkw_and exit cleanly
03:19.50bkw_then again we may have to change where we kill it
03:20.02*** join/#asterisk afg (joemillion@gafachi.rh.rit.edu)
03:20.03bkw_but we have to kill it
03:20.17bkw_haha
03:20.23bkw_JerJer[ghost] sowwy hit the end button
03:20.23bkw_haha
03:20.33afgHey, I'm trying to get 2 T1 boards to work in a system
03:20.43JerJer[ghost]quick draw magraw
03:20.48bkw_yep
03:21.10bkw_jrollyson I sure hope this fixes it
03:21.26afgis there any reason that I can not put 2 T1 boards in a system?
03:21.36bkw_yes
03:21.39bkw_if you need two T1's
03:21.44bkw_but at that price buy a Quad
03:21.46jimmyzirq's maybe just depends
03:22.02bkw_shit wish I had PRI so I could test this
03:22.16afgeach board gets its own IRQ
03:22.21jimmyzthey are cheap just get one lol
03:23.13ursenj_bkw_, get two T100p back to back
03:23.21bkw_ursenj_ I wish I had another T100P
03:23.23bkw_but I don't
03:23.24afgbkw_: were you talking to me?  if so, do you mean that I can not run 2 PRIs from one system at the same time using 2x t1xxp's?
03:23.40rollysonbkw_: reloading.
03:23.41bkw_afg no
03:23.46rollysonerr, recompiling
03:23.48bkw_rollyson call the conf in a few
03:23.55bkw_msg me when you do
03:24.09rollysonI'm in for a sec
03:24.19ursenj_bkw_, do you know d3i
03:24.39*** part/#asterisk SladeAKT (~mirc@user-0c8h5qn.cable.mindspring.com)
03:25.12*** part/#asterisk Kumbang (~unknown@167.205.22.54)
03:25.16*** join/#asterisk Kumbang (~unknown@167.205.22.54)
03:25.45afgIf I put two t1xxp in one system, one of the cards led will not light up or blink or anything
03:26.08jimmyzeven if you switch slots?
03:26.14angler__afg put it on a different slow
03:26.15angler__slot
03:26.17afgI do not believe it to be the card or the PCI slot, because if I remove the other card, then the one that was not working will work
03:26.36jimmyzmight be bustmastering also
03:26.39*** join/#asterisk brent21 (bdf@paalto-apx-1-144-201.penn.com)
03:26.45afgcat /proc/interrupts shows that they are on different interrupts
03:27.02angler__afg are they taking interrupts?
03:27.06afgand ztcfg shows that the spans are configured, and asterisk loads the zap channels (i'm talking about when both boards are in)
03:27.14afgangler__ yes
03:27.38Mikehows 0.7.1 doing?
03:27.44afgunfortunately the site where I am working on this does not have internet, so I cant try moving the boards around now
03:28.07afgjimmyz: what do I need to do as far as the busmastering?
03:28.21jimmyzit would be in the bios
03:28.29jimmyzon or off
03:28.44jimmyzbut you probably can't get to it if you are not there
03:29.00kamileonangler__
03:29.00afgwell I'll drive back there after I gather some possibilites
03:29.00angler__?
03:29.18kamileoncan you help me with this iaxtel registering thing
03:29.23kamileonand getting iax calls?
03:29.28afgjimmyz: so I want to turn busmastering on?
03:29.28kamileonor can i already?
03:29.42*** join/#asterisk Kumbang (~unknown@167.205.22.54)
03:30.03angler__i think theres an example on iaxtel.com
03:30.13bkw_hrm
03:31.08ursenj_any sucessfuly tied a t100p to a ds1 non-pri card in a Definity
03:31.14ursenj_any-anyone
03:31.46*** join/#asterisk _jackhamr (~jackhamr@64.212.11.53)
03:32.03brent21Are there any good writeups, howto's, etc on QoS configurations geared towards *. E.g. iax2, sip, etc?
03:32.48kamileonangler__ : i see... so what would my 700 number be 700-428-6222 ?
03:33.30_jackhamrhelloooo
03:34.03ursenj_ok simpler questions,.. will a t100p coneccted to a non-pri T1
03:35.18Mikebkw_: should i ask to the ML about my panasonic pbx?
03:35.27bkw_Mike they are going to tell you the same thign I told you
03:35.34bkw_the PBX isn't and can't send the correct singalling
03:35.43Mikebkw_: why?
03:35.56Mikebkw_: what option do i have? if he hangs up the calls when he wants
03:35.57angler__kamileon you have to register a number at iaxtel.com
03:36.18angler__damn i got 3 monitors now working in linux but got 2 desktops
03:36.52Mikebkw_: even if i used sipuras with the panasonic phones the transfer and hold etc wont work
03:37.01Mikebkw_: so i can remove that old pbx
03:37.25Mikeif i could remap the phones keys
03:37.30Mikethat would be great
03:37.34crontibsmike which panasonic pbx do you got
03:37.53Mikecrontibs: its a 308
03:37.59Mikecrontibs: its old and big
03:38.30Mikehttp://www.telephones-online.co.uk/itm00707.htm
03:38.33Mikethat one
03:38.46crontibsi was looking at this other day
03:38.48crontibshttp://www.twacomm.com/Catalog/Model_KX-TA624-5.htm
03:38.49ursenj_don't plug you 12volt AccessPoint powersupply into your BT-101,.. it will smoke
03:39.09afgdoes anyone have 2 T100Ps running in the same system?
03:39.26UnixDawghttp://www.voxilla.com/Article37-nested-order0-threshold0.phtml?POSTNUKESID=6c17b94343252b1114108da28d1ef8db
03:40.37*** join/#asterisk chris007 (~chris007@hoochie.digium.com)
03:40.42chris007sup all
03:40.44_jackhamrhi, i've got a new error on newer cvs  chan_sip.c:3635 register_verify: Peer '1111' isn't dynamic
03:40.44_jackhamr<PROTECTED>
03:41.10chris007i sent a few messages on the mailing list but really didnt get very far...
03:41.54chris007i was wonderign if someone can point me in the right direction on setting up * for only outbound calling with screen pops when the call goes through for my agents
03:42.15chris007with remote agent capibiliaties as well
03:42.38chris007i know that my calling list has to be in the spool
03:43.21chris007but i need to know if anyone has written a agi for importing a .csv file into the sql backend through a web page or something along thoes lines?
03:43.28ursenj_what brands of handsets will work with *
03:43.48*** join/#asterisk PBXtech (Reggie@65.218.37.175)
03:45.06doughecka"CNET News reports that security flaws have been found in products that use VoIP and text messaging, including those from Microsoft and Cisco Systems.
03:45.31_jackhamrThis is because microsoft and cisco suck ass
03:45.37dougheckaamen
03:45.45dougheckabut tis giving voip a bad name
03:45.49afgDo I need to run modprobe twice if I have two T100Ps?
03:46.01angler__no
03:46.37afgangler__ any ideas on how I can track down where the problem is?
03:46.51_jackhamrdough:  Cisco gave voip a bad name when they released call manager.  merrill lynch just dumped CCM and **10,000** 7960 phones because it sucked so bad.
03:47.02dougheckahaahhahah
03:47.08dougheckawow
03:47.23dougheckawhos merrill lynch?
03:47.25kimo_sabe_jackhamr: and I missed that surplus auction? damn :)
03:47.28_jackhamrstock broker
03:47.31_jackhamrinvestments
03:47.37TestMasTerAnyone know can i have a AGI script auto call me at a sertion time?
03:47.45TestMasTerCertain even
03:48.28angler__afg you have another machine to try?
03:48.46crontibsTestMaster you mean like cron job that runs script to send call to asterisk
03:48.59afgangler__ no sorry
03:49.04bkw_angler call me
03:49.10TestMasTercrontibs,  Yes but already got the answer thanx though
03:49.53afgwhen there is only one T100P in it works fine
03:50.03bkw_irq related issue
03:50.13_jackhamrkimo:  I'll hook U up with some cysco phones ;)
03:50.14afgie, card1 is in slot1,  card2 in slot2.     card1 does not work
03:50.22afgif I remove card2,  then card1 works
03:50.35afgbkw_: cat /proc/interrupts shows that each card has its own IRQ
03:50.42brent21doughecka, There are security flaws with analog lines too, e.g. I can hook up a phone to the Demarc on the side of your house.
03:50.50dougheckaindeed
03:50.56dougheckadont care
03:51.35brent21afg you have any extra slots
03:52.31afgbrent21: yes, there is one more slot, I will try that, but I need to drive over to the site...
03:52.41brent21afg, I had a problem with a Promise Array controller, everything was on its own IRQ, but my X100P's would crap out and I couldnt figure out why.  Pulled the Array controller, rebuit the drives, and all works
03:53.08brent21make sure they dont have anything goofy going on with the PCI cards in there, some cards can screew things up even when it looks like its on its own IRQ
03:53.28afgbrent21: thanks, i'll try that, but I think the other slot shares with the ethernet
03:53.45brent21in my situation, everything was on its own IRQ, but you gotta watch out for other devices
03:53.46*** join/#asterisk heison (~heison@CPE000a01d49e6f-CM014300011132.cpe.net.cable.rogers.com)
03:53.51brent21took me 2 weeks to figure that one out ;)
03:54.12afgI only have the 2 x T100Ps in that system... no other cards... of course there is onboard ethernet, vga, 2x serials, keyboard, 2x IDEs....
03:54.46afgthanks, i'll drive over and try it out
03:54.56afgis there anything that else that I could try while i'm there?
03:54.56_jackhamrwhat causes this punk-ass error:   chan_sip.c:3635 register_verify: Peer '1111' isn't dynamic
03:54.56_jackhamr<PROTECTED>
03:55.04brent21if your not using the serials, disable them
03:55.06h3xmake sure you dont put a board in the pci slot next to an agp slot
03:55.13brent21I also disabled USB and firewire support on my server
03:55.21TestMasTerAnyone know where i can get sample.call agi script from?
03:55.24brent21free up as much stuff you are not using, and let the PBX be a PBX
03:55.32rollyson_jackhamr : the host/ip doesn't match and you don't have host=dynamic set.
03:56.03afgI use one of the serials for console, i think I have the other disabled, i'll check to make sure I have everything else turned off, and try that move
03:56.12dougheckaaha!
03:56.15dougheckaserial console?!?
03:56.19afgis there any chance my motherboard doesnt support it?
03:56.21dougheckatis the problem!
03:56.24afg?
03:56.26dougheckaI think
03:56.36dougheckaI read someplace that the serial console may cause problems...
03:56.44dougheckaoh, is this a IRQ problem?
03:56.45dougheckanm
03:56.53_jackhamrrollyson:  thanks :)  I don't have dynamic set, I have specific IP's for each host defined with no host=dynamic in the record.
03:56.54afgwell I'm not sure that it is an IRQ problem
03:56.57dougheckaoh
03:56.57*** join/#asterisk simprix (~simprix@24.247.181.124.gha.mi.chartermi.net)
03:56.57atacommhows 0.7.x? i havent upgraded yet
03:56.59dougheckawhats the problem?
03:57.08dougheckaatacomm: its feeling fine, and sends it greetings
03:57.09_jackhamrata: todays cvs works ;)
03:57.16atacommlol
03:57.22simprixwhat in sip.conf allows a extension to make outbound calls over a x100p
03:57.25afgztcfg -vvvvv works without errors,   asterisk starts and loads up all of the zap channels without errors
03:57.25brent21afg, yeah just try moving stuff around, shouldnt be the motherboard, but the promise arrays shouldnt caused a problem either (and they did)
03:57.40dougheckaso whats the probme?
03:57.46kimo_sabeatacomm: aside from "ast_openstream: File demo-echotest does not exist in any format" it seems to be working for me
03:57.46dougheckaproblem
03:57.56brent21simprix, it basically goes in extensions.conf
03:58.03simprixwhat is it
03:58.17simprixso i don't have to set anything in sip.conf
03:58.21afgwell when there are two cards, the card which isnt working wont light up its LED, and wont run PRI
03:58.38simprixin sip.conf is it context
03:58.48dougheckaah
03:58.49afgif I pull out the second card, then the first card works with no problem (light lights up green and it runs PRI)
03:58.51dougheckaI dont have a pri
03:58.53brent21it could be, it depends on how you have Asterisk setup
03:58.54dougheckaso cant say
03:59.08brent21simprix, check out: http://www.digium.com/handbook-draft.pdf
03:59.34swirlnetsbkw: why would someone not want a unique id on cdrs?
03:59.41afgone more thing to mention: when doing a cat /proc/interrupts, the number for the card which isnt working, in the column CPU is 0
03:59.42brent21the book explains everything really well, sorry to refer you to another place, but you really need to read that in order to understand how * works
04:00.17brent21afg if you can set an IRQ in bios for PCI slots, try that
04:00.25afgsomeone had mentioned bus mastering earlier, does the T100P even utilize bus mastering?
04:00.58afgbrent21: yeah I can do that, it shouldnt matter what IRQ I assign, should it?
04:00.59*** join/#asterisk Zebble_ (~Zebble@Sherbrooke-HSE-ppp3610678.sympatico.ca)
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04:01.36brent21no but if you force it to take an IRQ, and disable auto's on the other, it may help
04:01.42brent21its really an experimenting thing
04:02.26*** join/#asterisk Stuart (~Stuart@65.167.123.238)
04:02.27*** join/#asterisk ursenj_ (~ursenj@ns.ursey.com)
04:02.42afgthanks brent21
04:02.52afghopefully i wont be back later :)!
04:03.45brent21stop back and say hello anyways and let others know what you did to fix it :)
04:03.59brent21that way they dont have the same problem hehe
04:05.01afgwill do, hopefully its not this serial console thing that I just starting reading on
04:06.59*** join/#asterisk blitzrage (~blitzrage@d141-224-202.home.cgocable.net)
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04:11.39Administrator__tim
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04:19.07*** join/#asterisk hmodes (hmodes@66.92.231.2)
04:19.37bkw_hmodes wasabi
04:20.16*** join/#asterisk ^tvt^ (KiepLuHanh@65.218.37.175)
04:20.51hmodeshola bkw
04:21.05hmodeshow is?
04:21.32*** join/#asterisk sobol_ (~druid@80.51.246.186)
04:21.51bkw_good
04:22.31Mikewasim: alive?
04:22.43voip~seen stealth-man
04:22.53voip: i haven't seen 'stealth-man'
04:22.53voip~seen stealth_man
04:22.54stealth_man <Stealth_Ma@80.83.133.249> was last seen on IRC in channel #asterisk, 5d 22h 17m ago, saying: 'how is going ?'.
04:22.54bkw_~seen wasim
04:22.55wasim is currently on #asterisk
04:23.04bkw_wasim wake up boi
04:23.25Mikesomeone doing a farfone now?
04:24.44*** join/#asterisk jsmith (~jsmith@smithfam.dsl.xmission.com)
04:26.08Mikebkw_: you saw the 250$ for bsd drivers?
04:26.14bkw_its 950 now
04:26.19bkw_read all the notes
04:26.21Mikewho raised it?
04:26.26bkw_read the bug notes
04:26.31Mikeokok
04:26.38Mikeare there any developers working on it no?
04:26.41jsmithMike: people like bkw_ did...
04:29.23*** join/#asterisk Koplin (~Koplin@hoochie.digium.com)
04:29.36*** join/#asterisk jorgeraidel (~jorgeraid@ip-69-33-73-90.mia.megapath.net)
04:30.41jorgeraidelwow jajaj and i installed the 0.7.0 one hour ago
04:30.45*** join/#asterisk _gorman (~lehmann@pD9E4ECA0.dip.t-dialin.net)
04:33.27Koplinhow well does asterisk scale?
04:33.31bkw_how far can you kick your server?
04:33.58Koplinvery it intergrated into a football :)
04:34.05hmodeswith or without the wheels on?
04:34.46_jackhamrhow about on mars?
04:34.56_jackhamrdid that damn rover get off of the lander yet?
04:35.20_jackhamris avt=rfc2833 =true?
04:35.57_jackhamrare there any scientologists here? :)
04:37.31*** join/#asterisk PlainWhiteTrash (~matt@user-12hcnqu.cable.mindspring.com)
04:40.33*** join/#asterisk miek (~miek@hoochie.digium.com)
04:41.12miekhello?
04:41.18Koplinhi
04:41.29hmodesmushi mushi!
04:41.34bkw_yo yo yo
04:41.58miekcan someone answer a technical question for me?
04:42.21_jackhamrmiek: shoot
04:42.25bkw_we can try
04:42.39miekI'm getting a "max retries" error on asterisk when using the X-pro soft phone over nat
04:42.43*** join/#asterisk Administrator__ (Administra@61.11.96.40)
04:42.46*** join/#asterisk l-fy (~diana@home-25022.b.astral.ro)
04:42.58_jackhamrsame here...i don't think it causes a performance issue?
04:43.14_jackhamrI think the chan_sip.c warning has something to do with nat traversal
04:43.21miekI can make call but the call is disconnected after a few seconds
04:43.39_jackhamrdo you have the right codecs set up?
04:43.40miekOK. I should just ignore the error then?
04:43.55miekHmmmm. Which codec should I use for the x-pro?
04:43.57_jackhamrI've been ignoring the error but it doesn't make me feel any more comfortable
04:44.08*** join/#asterisk Administrator_ (Administra@61.11.96.40)
04:44.10_jackhamrI wish the error would go away :)
04:44.24Administrator_hi
04:44.49miekWhich codec should I use for the x-pro?
04:45.21_jackhamrwhat are you using now?
04:45.38kimo_sabethis barbietone is kinda cute, and actually seems to be working in my limited test.
04:45.49kimo_sabeAre the Snoms as ugly as the look in the pictures?
04:46.02miekGSM
04:46.15_jackhamrkimo: do you think a black monolithic slab is ugly?
04:46.25kimo_sabe_jackhamr: usually, yes.
04:46.29_jackhamrthat's a shame.
04:46.35_jackhamrheheh
04:46.49kimo_sabe_jackhamr: I'm a bit of a sucker for shiny translucent bits too
04:46.59_jackhamrkimo: like the buttons on the barbietone?
04:47.06kimo_sabe_jackhamr: exactly
04:47.13_jackhamrjellies.
04:47.19kimo_sabethe not-depressing black helps too
04:47.55_jackhamri remember my dress-in-black days ;)
04:54.14hmodespfft
04:54.29kamileonhello
04:55.21_jackhamrhiya.
05:00.01*** part/#asterisk ___log (~stats@asterisk.toad.net)
05:00.02*** join/#asterisk ___log (~stats@asterisk.toad.net)
05:02.00joakobah, IMO the snom's dont look that bad
05:04.03kimo_sabeI need a sugar-mama to buy me a Cisco 7690 and an Snom 105 to play with
05:05.26carrarkimo_sabe, their out there
05:06.23crontibsthose barbietones anygood or should i stick with sipura2000
05:06.28kimo_sabecarrar: I know it. One of my friends is a "Stay at home dad", his wife is a Nurse Practitioner. Another just imported himself a nurse from Kentucky
05:06.57joakostick with the sipura
05:07.11crontibsjoako good idea
05:07.12crontibshe
05:07.13crontibsh
05:08.19joakoalot more features, and per port actually a few bucks less
05:08.24joakoAND THEY ACTUALLY WORK
05:08.35rollysoncrontibs: stay away from barbietones for now, they have a few more bugs to fix.
05:09.31hmodeswhen are they going to fix the ugly as sin bug?
05:10.18bkw_hmodes which one?
05:10.25bkw_yoru momma?
05:10.27bkw_er your
05:10.28bkw_haha
05:10.59hmodespfffft
05:11.25hmodesyou best watch yourself, my mum has quite a temper
05:11.34hmodesshe's also a raving lunatic
05:11.37hmodesnever a good combination
05:11.48loko-mokoAnyone here own any companies big enough to afford donating some $$$ to kids with cancer charity?
05:11.50UnixDawgok what has to be done to get cdr_postgres working
05:12.11UnixDawgor is it active by default
05:12.12timecophaha, they found vulnerabilities in h323
05:12.17timecopdoes that mean they will finally kill that shit off?
05:12.25bkw_timecop NOT
05:12.29timecophehehe
05:12.36bkw_we can only wish
05:12.38timecopbkw_: oh, you like h323?
05:12.42bkw_NOPE
05:12.53*** join/#asterisk roger1 (~roger1@hoochie.digium.com)
05:14.08kimo_sabehmm, Polycom do SIP yet?
05:14.17timecopi wonder how sip session timers are doing
05:14.49roger1I have exten => 2,3,Dial(ZAP/1/${EXTEN},30) but it appears the X100P card never hangs up, if I turn on "callprogress" then the time out works. Any idea why this is happening ?
05:15.01timecop"monitor bug" means I'll get mail about it right?
05:16.44crontibsany advantages of using h323 over sip
05:16.50timecopno
05:16.57crontibsas far as call quality
05:17.12timecopof course not, that has nothing to do with the protocol
05:17.27*** join/#asterisk srinivas (~srinivas@61.11.48.70)
05:17.31timecop8khz a-law is going to sound same over SIP or h323
05:17.35hmodesh323 is teh devil!
05:17.46hmodesit sucks even more then sip, and that takes alot of effort
05:18.00crontibsahh ok
05:18.22crontibsg729 codecs still avilable through Digium
05:18.24timecophaha, thats quotable
05:20.34*** join/#asterisk voip (~voip@c-24-10-202-120.client.comcast.net)
05:21.29hmodesthere needs to be some good motivation for the larger vendors *coughciscocough* to adopt iax2
05:22.09hmodesan iax-speaking 7960 and as5300 would be quite sextastic
05:22.17heisonhow can i do variable substitution in sed?
05:22.32heisonsed -e 's/${cur_ip}/${new_ip}/' /etc/ipnat.conf > /tmp/ipnat.conf
05:22.33heisonthis doesn't seem to work
05:23.02heisoni've tried \${cur_ip} also...
05:23.21hmodesif the var is in a ' it will be taken literally
05:23.28hmodesput it outside
05:25.16crontibsiax2 sipura 3000 would rock
05:25.20crontibsno longer need sip
05:25.24crontibsjust full 100%
05:25.25crontibsiax
05:25.53hmodesfeh
05:26.00heisonhmodes: outside where?
05:26.03hmodesi don't have much use for pots emulation
05:26.16hmodesnothing i'd want to hook to it works quite right anyway
05:26.38hmodesa shame there's no workable provision for data over voip-based pots
05:26.50hmodesnot that i really mind the lack of modems in my life
05:27.44hmodesheison; i think 's/'${cur_ip'/'${new_ip}'/' might be interpreted correctly, don't quote me on that tho
05:27.59hmodesworth a shot at least
05:28.35heisonnot quite... it's okay, i'll look on google
05:28.41UnixDawgok I get a errror wen adding cdr_pgsql.so to the addline in cdr
05:29.00*** part/#asterisk kapejod (~kapejod@pD9E83EEF.dip.t-dialin.net)
05:29.01UnixDawgin the make file
05:29.17UnixDawgwhat has to be done to make pgsql work
05:29.24UnixDawgor is it all active
05:29.49*** join/#asterisk Administrator_ (Administra@61.11.96.40)
05:30.00UnixDawgit give a -lpg -lz error
05:30.01PlainWhiteTrashUnixdawg, have you created a database with a table to store it?
05:30.06jorgeraidelalguien que haya trabajado con el X-lite?
05:30.11UnixDawgI have the db
05:30.17PlainWhiteTrashUnixDawg.. then you probably don't have the developer's libraries installed
05:30.27UnixDawgbut when I try to make the cdr module it errors out
05:30.30PlainWhiteTrashIf it won't compile you don't have the developer's libraries installed or they're in the wrong places.
05:30.50PlainWhiteTrashThat's what I'm saying.  If it will not compile, you do not have the libraries installed is the most likely problem.
05:30.51UnixDawgpostgres developers lb
05:30.59UnixDawgok
05:31.11PlainWhiteTrashyup.  install those and try again
05:31.52jorgeraidelsomebody know works with the X-lite?
05:32.26UnixDawgI dont find them ion the ports
05:32.28UnixDawggrrr
05:32.39PlainWhiteTrashports for what platform?
05:33.12UnixDawgfound them
05:33.14UnixDawgfbsd
05:33.19PlainWhiteTrashugh.  
05:33.22PlainWhiteTrashthat's your problem.
05:33.27PlainWhiteTrashyou didn't say that earlier.
05:33.39UnixDawgexplain
05:33.41PlainWhiteTrashI don't know if they'll be properly found on *bsd.
05:33.42bkw_lkajsdflajsowiejofalksdjfowiejfalksdfj
05:33.48PlainWhiteTrashYou should have said you were on BSD...
05:34.23PlainWhiteTrashI'm not sure what location the libs will get installed to on *BSD, and i'm also not sure that the build system will find them at compile time on BSD (depends on where bsd sends them)
05:34.29hmodesfalk!
05:35.07UnixDawgwell I am installing we will see
05:35.14PlainWhiteTrashk
05:37.11voipHow do I change this to make everything dialed with a 9 go out the zaptel?  exten => _91800NXXXXXX,1,Dial(ZAP/1/${EXTEN:1})  
05:37.34*** join/#asterisk cman (~cman1@202.51.74.250)
05:38.50UnixDawg_9X.
05:39.11voipthe . means ininifty?
05:39.22voipinfinity
05:39.23*** join/#asterisk sLeEpLeSs (~sLeEpLeSs@hoochie.digium.com)
05:40.06cmanhi... i just found out that my gs phone is not registered to my *???
05:40.11cmanhow come?
05:40.57UnixDawgdid you login to the phone and set it up
05:41.11cmanyes i can login from brwoser..
05:41.21cmanbut don't know why its not registered to *?
05:41.28UnixDawgok and you set it all up ip dns server
05:41.32cmanstrange... it was working fine till yesterday evening
05:41.41UnixDawgdo you have a sip exten setup
05:41.51UnixDawgwith the secret= the password
05:42.00cmanyes
05:42.30UnixDawgmake sure the match
05:42.37UnixDawgthen reset the phone
05:42.48UnixDawgand from cli type sip debug
05:42.56UnixDawgand watch the errors
05:46.03angler__grrr....
05:48.17UnixDawgyeah it says it neeeds postgers libs
05:50.34*** join/#asterisk Kumbang (~unknown@167.205.22.54)
05:51.32rollysonbkw_: your box die?
05:54.21bkw_yes
05:54.25bkw_but why are you going in and out of the conf?
05:54.30bkw_you are bouncing all over the place
05:54.38bkw_angler IAX2/guest@asterisk.bkw.org/996
05:54.47rollysontrying to reconnect, I'm not getting audio
05:55.01bkw_rollyson restart * and see if it works
05:55.21rollysonno.
05:55.23rollysonargh
05:55.30bkw_wonder whats up
05:55.35rollysonyou guys still in ?
05:55.45bkw_yes
05:55.54bkw_you keep bouncing in and out
05:55.55rollysonyou hearing me?
05:55.58bkw_nope
05:56.04bkw_who knows it may be my box
05:56.05rollysonbkw_: thought it was still dead
05:56.18*** join/#asterisk cman (~cman1@202.51.74.250)
05:56.27bkw_nope
05:56.38rollysonI'm going to shutdown -r now here.
05:56.45bkw_damn
05:57.01rollysonactually
05:57.06cmanwhy is my GS phone not being registered??
05:57.07rollysonlemme try jeremy's conf first
05:57.12cmanits strange...
05:57.26rollysonmy IAX or my SIP is hosed
05:57.35rollysonor my phone
05:59.25*** join/#asterisk Administrator__ (Administra@61.11.96.40)
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06:00.08*** part/#asterisk PilotPTK-Home (~trillian@pcp02587722pcs.shlb1201.mi.comcast.net)
06:00.17*** join/#asterisk clive- (~pirch@rndf-ip-nas-2-p136.telkom-ipnet.co.za)
06:00.32ReG-Hexercool
06:00.35ReG-Hexerre asterisk
06:01.01*** join/#asterisk Inv_Arp (~junya@adsl-214-36-237.mia.bellsouth.net)
06:03.14h3xyou know, i dont see any DACS documentation in zapata or zaptel config files
06:03.24h3xi suppose i could RTFS but...
06:05.08*** join/#asterisk T` (~total@RAMY8.RES.cmu.edu)
06:06.05ReG-Hexerwhere can i see the changes in 0.7.1 ?
06:06.17ReG-HexerChangeLog?
06:12.06angler__100%[======================================================================>] 37,526,048     2.21M/s
06:12.08angler__how nice
06:12.50kimo_sabeangler__: spiffy
06:13.32voipwow
06:13.46voipwill you be my sugar daddy?
06:13.47Criponangler: where's that from john?
06:14.14hmodeswhy can't i have that at home?
06:14.23angler__i pulled that off a server from my server
06:14.31angler__i wish that was at home
06:14.39hmodeslike seriously, it's been at least 6 years since a significant bump in residential broadband speed :(
06:14.49Criponangler: we
06:14.51angler__Cripon do i know you?
06:14.52hmodeshell comcast is just now getting their caps close to what i had during the early @home days
06:14.53hmodessigh
06:15.09Criponangler: I'd hope so..
06:15.14angler__wish i could get mplayer working with faad right
06:15.23angler__ahh
06:15.26angler__whats up christian
06:15.28kimo_sabehmodes: kazaa not leeching fast enough? There's not much modivation to go any faster
06:15.34Criponangler: that's sad man
06:15.37Cripon:)
06:15.39hmodespffft
06:15.44hmodesscrew kazaa
06:15.45angler__i didnt know your irc name
06:15.57hmodesi want more then two channels on my * box without serious degradation ;p
06:16.20kimo_sabehmodes: there's room to grow on upstreams, but not really down streams
06:16.26Criponangler: did you give mark that kid's resume?
06:16.34hmodeseh, i'm content with 3mbit
06:16.37angler__ummmm
06:16.39hmodesi'd be much happier with 10 tho
06:16.41angler__mail it to me again
06:16.57angler__im not in windows and wont be for awhile
06:17.00hmodesthe upstream really needs to be dealt with tho'
06:17.12*** join/#asterisk Administrator__ (Administra@61.11.96.40)
06:17.25angler__Cripon you get that routing sorted out?
06:17.41hmodesalthough i guess the larger comcast-like megaconglomos are prolly able to get a better price/mbit by gauranteeing async useage
06:17.55Criponangler: for 300 to a pro.. he setup vpn between the boxes.. works like a charm. It's actually faster than it was.
06:18.56CriponI'm just a Bigelow, and everywhere I go people know the part J's playin.
06:19.07angler__haha wtf?
06:19.37CriponI'm just a gigello
06:19.43Cripongigelo
06:19.51Criponi dunno
06:19.57angler__i wish i was then i would be rich
06:19.57hmodesmanwhore!
06:20.33CriponI whore myself out to myself as often as possible.
06:20.51hmodesdo you tip yourself well?
06:21.01angler__lol
06:21.14CriponI'm also a cheap bastartd
06:23.22CriponDid everyone download this and give it a try from their Windows boxes?   http://www.yottadot.org/download.php?op=viewsdownload&sid=10
06:26.33hmodesoh my
06:26.40hmodesthat's cute
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06:29.34joakocripon: what do i seutp on the asterisk side?
06:29.47Criponjust a user in manager.conf
06:30.13*** join/#asterisk Administrator__ (Administra@61.11.96.40)
06:30.15clive-joako what are you tryoing to set up
06:30.33Criponclive: http://www.yottadot.org/download.php?op=viewsdownload&sid=10
06:31.59bkw_wooohooo
06:32.04bkw_bug 851 guys go test it
06:32.05bkw_:)
06:32.43rollysonbkw_: nice work.
06:33.03bkw_thanks for testing the PRI goodie
06:33.04bkw_s
06:33.56clive-cripon whats the URL?
06:35.02rollysonbkw_: wheres the bugs on being unable to unload chan_zap
06:35.03Criponclive: it's an * manager written in C# for windows.
06:35.10rollysonthey need to be resolved ;)
06:36.04clive-cripin sounds interesting, I am a newbie, so I need a windoze and a linux box?
06:36.39Criponclive: yes.  It's coming along, it's not all that robust yet.  But it works and it dials from outlook.
06:38.28clive-outlook is a mail client?....(which I dont use ..pegasus mail rules:)..so how does it work?, and which protocol?
06:38.43rollysonoutlook is a *groupware* client
06:38.46clive-it sounds very interesting:)
06:38.49rollysonnot just a mail client
06:39.02clive-I see
06:39.08Criponclive: install it and play around with it.  it uses the tcp/ip protocal
06:39.11clive-does it support SIP ?
06:39.16joakohow do you dial with outlook??
06:39.26CriponClive: it's not a soft phone.. it's a call manager
06:39.29clive-not outlook express?
06:39.56hmodesuurrrgh
06:40.11Criponjoako:  http://www.yottadot.org/download.php?op=viewsdownload&sid=10  has an outlook dialer built in.
06:40.12hmodesso when does someone donate a 100mbit colo to digium?
06:40.17rollysonCripon: I'll try that tomorrow.
06:40.18clive-ok, so basically a GUI for *
06:40.23Criponhmodes: wouldn't that be nice.
06:40.37hmodesyeeeah..
06:40.44hmodest1s are so 90s ;p
06:41.00*** join/#asterisk Kumbang (~unknown@167.205.22.54)
06:41.02Criponclive: it's an interface.  It's not as "operator" oriented as gastman yet, but that's coming.
06:41.11TestMasTerIs there anyway to give Global Access to asterisk -rx?
06:41.21hmodesthat and i bet it would make their lives less annoying if i wasn't mooching off cvs on the same connection they do their thing
06:41.23joakoyes, i installed it, now what?
06:41.27TestMasTerwhat i mean is so apache has access to beable to run asterisk -rx
06:41.41joakowhat does digium need a 100mbps colo for?
06:41.51kimo_sabeTestMasTer: sudo?
06:41.52bkw_blah timmah
06:41.54Criponhmodes: mark told john and I today that they do 9 million email a day over that pipe, plus cvs.
06:41.56clive-cripon, sounds great....good luck with it...I am not at home where my * box is, but I will keep it in mind....
06:41.59TestMasTerkimo_sabe,  sodo?
06:42.01TestMasTersudo?
06:42.03hmodesyeeeah
06:42.13angler__Cripon lmao
06:42.15hmodesbetween maillists and cvs i bet that poor line is constantly lit
06:42.37angler__mark did say that though  :)
06:42.54joakowhat are they running right now?
06:43.02Criponfractional t1
06:43.22Criponhow many data chans john? 12?
06:43.29rollysonbkw_: so, how long do you think its going to be before we have to do 0.7.2 ;)
06:43.40bkw_rollyson hahah
06:44.00angler__Cripon somethin like that
06:44.12joakohmm i would actually consider giving them a dedicated machine
06:44.30TestMasTerkimo_sabe, What is sudo
06:44.39bkw_super user do
06:44.47Criponon how large a trunk?  then they run the problem of security.
06:44.47joakoanyways, how do i get this thing to dial from outlook??
06:44.55bkw_its a way to give users su access to some stuff
06:45.01Criponjoako. you have to have both items installed
06:45.03kimo_sabeTestMasTer: http://freshmeat.net/projects/sudo/
06:45.06joakoyes
06:45.23*** join/#asterisk ^sly^ (~jelque@adsl-67-66-120-174.dsl.ltrkar.swbell.net)
06:45.34Criponis astring up and running?  do you have a little yellow * in your taskbar?  is it connected to asterisk?
06:45.38angler__Cripon what your wife cook for dinner tonight?
06:45.40joakoyes....
06:45.48TestMasTerkimo_sabe,  thanx
06:46.00hmodessudo is a godsend and teh devil at the same time
06:46.01Criponangler: I cooked chilicheese burritos
06:46.04*** join/#asterisk adkr (~adkr@hoochie.digium.com)
06:46.11joakocripon: was your question directed at me? If I did anything it would be 10mbps and i'd give them 2 or 3 mbps of bandwidth...
06:46.13hmodesspecifically, i really need to ditch %sa    ALL=/bin/su
06:46.22hmodes*sigh*
06:46.29angler__Cripon dang i had some cookies and now im eating cereal
06:46.32rollysonhmm. the only resturant in range is the one across the parking lot
06:46.50Criponangler:  sorry man.. you need to make more money.. come to work with me.
06:46.59angler__Cripon hahaha
06:47.06*** join/#asterisk brc007 (~brc007@ip68-109-131-42.ph.ph.cox.net)
06:47.12angler__Cripon ill work with you on the side
06:47.15joakook i figued it out, you added a button to my outlook toolbar
06:47.16Criponangler: I'm so rich, It's hormel chili everyday.
06:47.23Criponjoako: bingo
06:47.25angler__haah
06:47.35hmodessomeone should hire hmodes!
06:47.40hmodeshe hates his current job
06:47.43angler__lol
06:47.44joakowhen you right click a contact there an option to call contact
06:47.46hmodesor is that the definition of 'job'?
06:47.48joakoyou should use that...
06:47.52angler__does anyone like their job?
06:48.06Criponjoako: you can't mess with the right click menu's in windows.. so it goes to the taskbar
06:48.08angler__JerJer prolly does
06:48.20joakolook @ it
06:48.23Criponmenu's in outlook I mean.. not windows
06:48.23hmodesi would enjoy my job if there weren't so many stupid people negating everything i do
06:48.35kimo_sabeangler__: yes!
06:48.40joakoright click contact -> call
06:48.42CriponI know about the outlook dialer.. doesn't work for this.
06:48.48bkw_angler I love my job
06:48.51joakothen look at dialing options
06:48.55Criponsetup h323 and dial that way then. :)
06:48.58joakoany way to add it there?
06:49.14Criponno.. you can't add right click menu items in outlook
06:49.31angler__i like calling bkw_ and bugging him thru out the day
06:49.32joakono, add it as an option where the modems / h323 are
06:50.04Criponah.. see.. that only works in outlook.. so it's the same boat
06:50.33hmodeswait wait, love and job in same sentence?
06:50.35hmodesthat's possible?
06:50.50bkw_yes
06:50.59hmodesneat
06:50.59bkw_very possible
06:51.25bkw_some days I hate it.. but most days I love it
06:51.28bkw_I work from home...
06:51.39hmodesoooh
06:51.50angler__bkw_ is lucky
06:51.51hmodesthat's gotta be nice
06:51.57pros12bkw: what kind of work do you do besides this?
06:51.59kimo_sabehmodes: what's not to like about working at a liberal arts college that's like 80% female?
06:52.09hmodesi've tried to explain to my work how unnecessary it is that i drive 100mi each way to get there..  they don't seem to get it.
06:52.49bkw_pros12 I am the network bitch for an ISP
06:52.55bkw_I do it all......
06:53.12pros12ahhh.. be my bitch too... lol
06:53.51joakook if i setup this thing propery, should i get something on the console when I dial?
06:54.00bkw_um yes
06:54.05joakoi dont....
06:54.19*** join/#asterisk dias (~dias@hoochie.digium.com)
06:55.46Criponjoako: what do you mean, on the console.. it will just initiate a call to the device you specified for it to use, then call the # you told it to dial
06:56.16pros12is iaxtel down?
06:56.22bkw_down like a french whore
06:56.27pros12:(
06:56.38jrollysonhmm... is mark still around?
06:56.45bkw_nope
06:56.46bkw_why?
06:56.49bkw_got a problem?
06:56.53jrollyson#738
06:57.01pros12i cant make pstn calls now..
06:57.01joakobut on the asterisk console it should say SOMETHING, right?
06:57.17*** part/#asterisk benngard (~mabe@81.26.235.3)
06:57.30Criponjoako: is it doing nothing?
06:58.15hmodeshrmm
06:58.30hmodeseither i'm being retarted or your zap patch b0rk muh shit
07:00.07Criponwhat's everyone's favorite softphone?
07:00.31pros12bkw: can i pm you?
07:00.34jrollysoniaxcomm
07:00.48jrollysonhmodes: lol
07:01.20bkw_pros12 no
07:01.24pros12k
07:01.30bkw_keep it in the channel please
07:01.36*** join/#asterisk scat (scat@c-24-126-24-177.we.client2.attbi.com)
07:01.36bkw_I have too many things going on right now
07:01.37bkw_:P
07:01.47pros12lol np
07:03.25jrollysonhmm.
07:03.33jrollysonI need to get a sound file
07:04.01jrollyson"Due to your continued abusive comments, we are now forced to terminate this call. Thank you for calling, have a nice day."
07:04.30jrollysonand set that up as ext. 666
07:05.02h3xhaahhaha
07:05.56pros12try using this.
07:05.59pros12http://www.rhetorical.com/cgi-bin/demo.cgi
07:07.15jrollysonheh
07:07.54pros12like that..
07:09.23pros12download the wav and use it in *
07:09.39carrarsomewhat illegal
07:09.46pros12????
07:10.13jrollysonthose wavs are only for non-commercial use ;)
07:10.28pros12well use it for home
07:15.31hmodesalrighty bkw, clean checkout, compile, run, stop now, patch, clean, compile, run;
07:15.41hmodesJan 15 02:16:41 ERROR[1024]: chan_zap.c:7513 load_module: zapata.conf has changed since last startup. Will not load chan_zap.so
07:15.52hmodesi haven't touched zapata.conf in months
07:16.19hmodes+ unload chan_zap between run and stop now that is
07:25.02*** join/#asterisk Gazzas (~ghendler@c211-28-134-93.eburwd3.vic.optusnet.com.au)
07:26.39kamileonhello
07:27.32MSpinhmodes: I got the same issue...bkw found the bug and is fixing it tho
07:27.53kamileonbkw_ still having that fax issue?
07:29.41hmodesah, well then i feel doubly useless
07:29.42hmodespfft
07:38.22kamileoncan someone call 7004286222
07:39.50hmodeshrmm, my mwi still doesn't seem to work
07:40.07hmodesam i some kind of freak or something?
07:40.53Criponkamileon: is iaxtel working.. I just go to congestion after a timeout period.
07:41.10*** join/#asterisk deexm (~CraZENiGG@adsl-64-118-253-166.netrox.net)
07:41.59kamileonwait, its fucked
07:42.07kamileonwont boot up atm
07:42.20kamileonsorry
07:42.46kamileoncan i use those voices up there at home since im noncommercial
07:43.05Criponkamileon.. I don't see why not
07:43.13*** join/#asterisk RoyK (~roy@19.80-203-29.nextgentel.com)
07:43.22kamileongood ;)
07:43.38kamileoni want sexy british female
07:43.53Criponkamileon: I'm not the authority, but I say you can use them for whatever you want. Just don't get caught for commercial applications.
07:43.56RoyKGuten Morgen, #asterisk
07:44.04kamileontrue Cripon
07:44.30kamileondoes anyone know if the tdm400 card and x100p will work in a smp box ?
07:44.47Criponyes.. I use them
07:44.57Cripondell poweredge 2500
07:44.57kamileonsmp?
07:45.05kamileoni have a quad ppr0 box
07:45.08Cripondual 933 p3's
07:45.13kamileonand im running them in a p200mmx box right now
07:45.19kamileoni notice no faults yet though
07:45.46kamileoni want to just deck out the intergraph quadppro
07:45.49kamileonjust to do it
07:45.55kamileonwith as much as possible for debian
07:46.04Criponkamileon: you in alabama?
07:46.21kamileonyes
07:46.22kamileonhsv
07:46.24kamileonyou?
07:46.36Criponkamileon: yes
07:46.44kamileonexcellent
07:46.51kamileonwhy up so late?
07:47.33kamileonim kamileon.. dave.. ive posted a few times maybe
08:06.27*** join/#asterisk adam (~arichards@gatekeeper.oremut02.us.wh.verio.net)
08:10.05iewebguy_Hello, Can anyone suggest how to connect my ata 186  it is a former vonage device
08:10.39voidptr_morning
08:17.03clive-iewe, it may be locked in
08:17.26iewebguy_I have it unlocked...
08:18.09iewebguy_It needs sip firmware?  or does it already have it?
08:18.11jrollysonbleh, wrong window
08:18.19clive-cool, then just reconfigure it to point to the SIP server you want
08:18.45iewebguy_ah.... yes there is one big button on the top.  
08:19.15iewebguy_Any reconfig notes available ? URL?
08:19.29*** join/#asterisk vindex (ldm@zenon.apartia.fr)
08:19.41clive-http:// (ip-addyof-ata)/dev
08:20.22iewebguy_k.
08:21.02clive-g'luck
08:21.20clive-bye all
08:26.08*** join/#asterisk dxmdcc (~CraZENiGG@adsl-64-118-253-166.netrox.net)
08:37.49jrollysonanyone in conf?
08:40.07kamileonhttp://www.binarypimpin.com/room/
08:41.11hmodeshah!
08:41.27kamileoni just moved in :(
08:42.16hmodeshttp://matrix.gs/random/desk.jpg
08:42.19hmodestough call ;p
08:43.05*** join/#asterisk oej (~opr@apollo.webway.se)
08:43.46blllI will have to take pictures of my home server room some time
08:44.09hmodesoh, i thought we were comparing desk sizes :)
08:44.15*** join/#asterisk Kumbang (~unknown@167.205.22.54)
08:44.16kamileonno you should see my old one!
08:44.54blllhmodes: what speed processor do you have in that ultra 10?
08:45.01KumbangERROR[8192]: File cdr_addon_mysql.c, Line 298 (my_load_module): Failed to connect to mysql database asteriskcdrdb on localhost.
08:45.04hmodes440
08:45.11hmodesit lacks graphics foo tho :(
08:45.28hmodesthere's 4x400 e450 coming next week :)
08:45.31blllah, I need a 440 for mine
08:45.31hmodesthe u10 is getting sold off
08:45.32Kumbangwhy did it happened, i do exactly what configurtion need for cdr_mysql
08:46.44hmodesnice blurry identification tho
08:46.58blllI have a couple
08:47.16blllused to work for a sun partner that went out of business, got a lot of sun gear at liquidation
08:47.20hmodesthat should be a hot pic :)
08:48.21hmodesa martini and a lamp on a 450 between two couches..  that's what i call a still life.
08:49.27hmodesin the meantime, i shall go play in teh snow
08:50.47*** join/#asterisk dolbe (~dolbe@pcp04566804pcs.jersyc01.nj.comcast.net)
08:50.52bllloh god
08:54.55*** join/#asterisk RoyK (~asdf@213-187-164-3.dd.nextgentel.com)
09:01.34hmodeswow, it's nice n' snowy out
09:01.36hmodesthat's fantastic
09:02.13kamileonwhere?
09:03.35voipDoes anyone know how nuphone charges?
09:03.47kamileonout the ass
09:03.51voipis there a monthly free for a toll-free number plus the 2.9 cents?
09:04.11voipmonthy fee
09:04.55RoyKhm. how many days wer there between 0.7.0 and 0.7.1?
09:05.10kamileon.1 ?
09:06.07RoyKsee topic
09:07.37kamileonwhen is iax gonna be fixed
09:07.39hmodeshrmm
09:07.49hmodesi just spent close to $200 on underarmour
09:07.55hmodesthat seems pretty excessive
09:07.58kamileoncan some tell me how to record my voice greetings and such?
09:08.12kamileonkevlar?
09:08.15RoyKwhere the fsck is the official asterisk download page?
09:08.25kamileonthere isnt one, cvs!
09:08.33hmodesit's less bullet proof then kevlar, but more fulfilling
09:08.36*** join/#asterisk mmco (~irc@pD9E10B6E.dip.t-dialin.net)
09:08.46kamileoni sold my kevlar a month ago
09:08.48RoyKkamileon: then what's the fscking point of releasing versions?
09:08.53kamileoni didnt plan on getting shot
09:09.02kamileoncvs update
09:09.12*** join/#asterisk yaboo (~jsirucka@203-213-113-146-vic.tpgi.com.au)
09:09.32RoyKkamileon: the point of using a release instead of the cvs is that you usually get a more stable version
09:09.41kamileonoh sorry
09:09.45RoyKdunno if that's the case with * but
09:09.45RoyK...
09:10.05RoyKbut then still - why are there releases if it's all a dynamic heap in the cvs?
09:10.21kamileoncompliance?
09:13.33RoyKanyone that knows what's the status of zaptel on 2.6?
09:14.03hmodesi definately noticed special attention getting paid to mantis before .7 was released
09:14.22hmodeshopefully 1.0 will be truely stable.  that would be quitehot
09:14.40hmodestoday's cvs is getting quite close
09:14.56hmodesthe only way i can make it crash is if i do something wouldn't normally
09:15.08hmodesalthough my damn mwi doesn't work ;p
09:17.40RoyKmwi?
09:17.48hmodesmessage waiting indicator
09:18.09hmodeswhen someone leaves me a message in voicemail2 my 7960's 'you've got mail' led doesn't light up for some reason
09:19.11hmodesi have a feeling someone swapped fields around in the sip notifies or something, and the 7960 doesn't turn it on for the new format
09:20.10hmodesi'm too damn lazy to file a formal bug tho, i figure it'll either fix itself or someone else will get irked enough eventually
09:21.14hmodesor i'll just perform shady hacking like i have for my previous problems
09:21.29hmodesi should prolly just make myself a diff against cvs
09:30.33*** join/#asterisk deexm (~CraZENiGG@adsl-64-118-253-166.netrox.net)
09:36.57kamileonsnow where?
09:37.00hmodesas an added bonus the t3 for office use at work is down, and it's certainly not work going in to fight over a couple t1s
09:37.10hmodespennsylvania/newjersey
09:37.43kamileoni have a oc192 in my bedroom!!
09:37.47hmodesonly a couple inches, but around here that's enough to call out of work over
09:37.50kamileonph33r me!!
09:38.23kamileonsomeone suggested i put all my gear in the closet
09:38.42kamileonbut i couldnt get to the back of the cabinet and still close the closet door. :(
09:38.55kamileonits my playroom anyways
09:39.13hmodesmy closet has a closet
09:39.16hmodesit's quite amusing
09:39.23kamileonvery cool.
09:39.44hmodeseh, not so much that as a good conversation peice
09:39.59hmodesapparantley at some point it was a former tenant's baby's room
09:40.03kamileontrue enough
09:40.08hmodesi feel sorry for that kid ;p
09:40.12kamileonanyone have a qlogic scsi controller card?
09:40.16kamileonhuh huh huh
09:40.33kamileonno one ever does
09:41.04hmodesgot plenty in suns, none in leenooks tho
09:41.08kamileonmy alpha wont install the digital unix i bought for it unless it has a 'bootable' scsi card installed so the 'firmware bios' can see the disks on it
09:41.29kamileoni have to boot milo on floppy
09:41.50kamileonthen boot lilo from the ide disk, then mount the scsi disk
09:41.53kamileonwtf!
09:42.09kamileonAND
09:42.15kamileonit doesnt work with my kvm
09:42.27hmodesouch?
09:42.40kamileonyeah, its a good box though
09:42.48kamileonruns redhat 7.1 i think
09:42.52kamileonor 7.0
09:43.29hmodesnot like any 7.x version was any different from any other
09:43.35kamileontrue
09:43.46kamileonim wanting to go boot it now, damn...
09:43.48hmodesspeaking of which, i need to get up on a whitebox clone next week
09:43.56kamileoni dont know why, it serves no purpose really
09:44.16*** join/#asterisk digger_ (~digger@penguin.taide.net)
09:44.20hmodesthe two just don't play well together
09:44.33kamileoni want to put my * hardware in my intergraph 650r
09:44.57kamileonhmodes : yes, and mine is a discontinued, unsupported platform
09:45.04kamileonim lucky to log in to it
09:45.26hmodesyezzz
09:45.29kamileonwhat can i possibly do with 4 processors, any suggestions?
09:48.05hmodeshide one of them and claim smp-ness?
09:48.13kamileonlol
09:48.23hmodesund now it be time for sleep
09:48.25kamileonseriously, what takes advantage of that?
09:48.26hmodesg'nite fockers
09:53.04*** join/#asterisk dnc (~duncan@213.244.224.118)
09:54.06*** join/#asterisk reseaux (~reseaux@host9-132.pool82105.interbusiness.it)
09:54.34reseauxHi To ALL!!!Happy new year!!
09:55.10blllhmodes: I am hoping the same
10:03.42karimddoes anyone know if phonecore is still in development?
10:07.13*** join/#asterisk clive- (~pirch@rndf-ip-nas-2-p189.telkom-ipnet.co.za)
10:14.11*** join/#asterisk geertn (geertn@193.194.136.225)
10:15.31RoyK~seen kape
10:15.36kape <~kapejod@pD9E8297E.dip.t-dialin.net> was last seen on IRC in channel #asterisk, 21h 3m 12s ago, saying: 'ico: doesnt work yet'.
10:15.36RoyK~seen kapejod
10:15.36kapejod <~kapejod@pD9E82915.dip.t-dialin.net> was last seen on IRC in channel #asterisk, 1d 22h 42m 43s ago, saying: 'sniffe: yes'.
10:18.34*** join/#asterisk john (~john@hoochie.digium.com)
10:20.48johnhello everyone...having a question, what do i need to download such that, i could in my asterisk CLI screen the digits i'm sending and receiving?
10:21.45johni downloaded already a fresh version, but it seems doesn't to work..if i dial an extension, my CLI screen won't move..
10:23.42*** join/#asterisk zoa (~john@213.219.141.63)
10:24.01reseauxjohn: have you start * with option -vvvvvvvvvc
10:24.42johnyeah..
10:25.20johnalso, i noticed that my t400p card doesn't want to turn green anymore..
10:25.35johni do already modprobe it
10:27.32reseauxhave you try a local loop on t400p port?
10:28.21johnjust a minute, i'll try to do that..
10:29.57karimdis phonecore still being developed?
10:30.38johnreseaux, still red..
10:41.05voidptr_argh, traditional pabx'es are a biatch!
10:41.36voidptr_i cannot forward an extension without programming it on a port
10:42.04jrollysonvoidptr: thats lame, what system?
10:43.33steeshow can I pass a variable to a context and 'catch' it again in the context ?
10:44.21*** join/#asterisk puzzled (~patrick@puzzled.xs4all.nl)
10:48.16voidptr_jrollyson : philips sopho
10:48.40voidptr_it could be that i am too lame to make it work differently ofcourse :D
10:54.16*** join/#asterisk xs (~CAT@a80-126-102-2.adsl.xs4all.nl)
11:06.08steeshm...
11:06.27steeswhy, oh why cant I just have a voicemail box that doesnt play it's own greeting
11:07.03puzzledstees: you can. don't remember what it was but do show applications and check out the voicemail apps and their settings
11:12.40steespuzzled: cool, got it
11:12.50puzzledenjoy
11:12.51steesVoicemail(s<extension>)
11:12.57steespreceed it with 's' :)
11:25.59reseauxjohn:Sorry...im try to install a server...
11:26.17[Sim]okay this is sick
11:26.18reseauxjohn:so...i still red
11:26.34[Sim]channel variables can get lost if you do a local to iax bridge
11:26.51voidptr_:S
11:26.52[Sim]but not if you do a local to sip bridge
11:27.04voidptr_"can"?
11:27.07voidptr_not always?
11:30.05[Sim]well
11:30.23[Sim]it seems to be related to a very specific combination of things
11:31.06[Sim]scenario: I create a callfile in outgoing which connects channel Local/number to [context],s,1 with a variable set
11:31.34[Sim]if I do this to a local extension that is a SIP phone, everything is fine, and the variable can be used
11:31.54*** join/#asterisk telenieko (~telenieko@80.224.224.228)
11:31.55[Sim]if I do this to a remote extension (that is routed over IAX) the variable is lost
11:32.01teleniekoHi everybody.
11:32.21[Sim]hello
11:32.40[Sim]voidptr; I noticed this before with local to CAPI.
11:32.56reseauxhi!
11:33.16teleniekoProblem there: An H323 Client connects to asterisk and asks to be connected with a 2nd H323 client I use CALL application, but then Asterisk hold both calls it codingg-deconding is very slow. Is tehre abyway to 'transfer' the call so asterisk forgets about it conneting both parties directly??
11:33.19teleniekoThanks ;)
11:34.44*** join/#asterisk khan1 (~khan@203.82.51.38)
11:35.00khan1hello
11:35.19khan1i need some help
11:35.45khan1my phones have registered with the server and yet i am unable to make a call between them
11:36.02puzzledhi [Sim] and voidptr_
11:36.11reseauxkhan1:what kind of problem is?
11:36.13voidptr_hey puzzled
11:36.40khan1well the last time i was here
11:36.46voidptr_[Sim] : hummm ok :(
11:36.49khan1some people told me of this site
11:36.54[Sim]hey puzzled
11:36.55khan1which helped alot
11:37.06khan1but my companys phones don't seem to work with asterisk
11:37.14khan1they keep returning 403 NOT FOUND
11:37.42reseauxkhan1:I think is possible... :-( but you can see some debug trace on *?
11:37.58khan1i have
11:38.19khan1see my server's ip is 192.168.0.23
11:38.23khan1one phone is 12
11:38.29khan1the other is .14
11:38.39khan1when i dial from .12 to .14
11:38.56khan1everything goes fine from .12 to .23 and .23 to .14
11:39.08khan1but .14 returns 403 NOT FOUND
11:39.13reseauxkhan1:from * CLI you can see some information on incoming call?
11:39.39khan1the only thing that seems out of place is that when .23 forwards the call
11:39.48khan1it asks .14 to find .14
11:39.52khan1which ofcourse it can't
11:40.26reseauxkhan1:i cant undestand....
11:40.34khan1which part
11:40.35khan1?
11:41.43khan1how are u supposed to configure phones?
11:41.51khan1you know name their ports?
11:42.05reseauxif you can call from .23 to .14 what you can see from CLI of *
11:42.07khan1they're supposed to have the same names as those in the asterisk server right
11:42.20khan1.23 is the server
11:43.42reseauxyour telephone .14 cant call to any other telephone?
11:44.09*** join/#asterisk Filace (~jon@pluto.geekpeople.net)
11:44.38*** join/#asterisk khan1 (~khan@203.82.51.38)
11:44.51Filaceanyone here used chan_capi with multiple BRI lines?
11:45.59khan1so anyone
11:46.05khan1can you please guide me
11:46.18khan1u know like let me know of any web-site that might help me out
11:46.30Filacewww.voip-info.org
11:46.40Filacehttp://www.automated.it/guidetoasterisk.htm
11:46.53khan1should the sip proxy settings be on on the phone while using asterisk?
11:47.28Filacenot unless you've setup a SIP proxy
11:47.35reseauxkhan1:Check in this site is very usefull...
11:47.36Filaceasterisk doesn't have one built in, afaik
11:47.38khan1well i haven't
11:48.36khan1so what they're supposed to be in the direct mode?
11:49.42khan1yes i've done exactly what it says on these sites
11:49.42khan1i've tried it all
11:49.42khan1the phones register
11:49.57khan1but return 403 NOT FOUND when i dial from one to the other
11:50.08khan1the phones work fine on vocal
11:51.40reseauxkhan1:i cant understand... sorry...
11:52.14reseauxkhan1:you are able to register the .14 but you cant call another phone
11:54.57khan1the phones show when i type sip show peers
11:55.12teleniekoProblem there: An H323 Client connects to asterisk and asks to be connected with a 2nd H323 client I use CALL application, but then Asterisk hold both calls it codingg-deconding is very slow. Is tehre abyway to 'transfer' the call so asterisk forgets about it conneting both parties directly??
11:55.47*** join/#asterisk Aviaa_ (~Aviaa@test.incracow.com)
11:55.48khan1when i dial from one to the other and sip debug is enabled i can see packets going back and forth between both phones but then a busy tone comes
11:55.53khan1and 403 error comes
11:56.20Aviaa_halo if on cisco i have set: dtmf-relay rtp-nte , what dtmfmode i must set into * ?
11:56.48*** join/#asterisk illc0mmm (~illc0mm@142-56.34-65.tampabay.rr.com)
11:57.17reseauxkhan1:have check the conf of sip ?
11:57.24khan1yes
11:57.43khan1i've set up both phones according to the configurations given at both of the above sites
11:57.50*** join/#asterisk ico (~none@pD9531BAA.dip.t-dialin.net)
11:57.54reseauxthe phone is the same?
11:58.01khan1but i haven't setup any sip proxy
12:28.55*** join/#asterisk yoyowon (~yoyowon@hoochie.digium.com)
12:28.59yoyowonhi
12:29.45yoyowonDoes asterisk source include SIP Proxy Server?
12:30.11yoyowonanybody here?
12:30.23yoyowontell me...
12:31.12oejYoyowon: Not a full SIP Proxy server, but a SIP server and SIP user agent
12:33.02johnhi reseaux....sorry, got some problems with our servers here...
12:33.18johnyeah, i tried to loop back, led is still red
12:33.37yoyowonthen...
12:34.39yoyowonAsterisk can connect SIP Proxy Server?
12:34.39yoyowonreal Fully Proxy Server?
12:39.47geertnyoyowon: Please clarify your question
12:42.25YoYo^*BURP*
12:42.41yoyowonDo i wanna know, asterisk can connect to real SIP Proxy Server?
12:43.14jimmyzhow can i make all out bound calls go to a meetme room and then dail the number
12:43.27jimmyzwhere a manager could listen to any call he wanted to live
12:44.00geertnyoyowon: You mean like ser?
12:44.33yoyowonYes..
12:45.46yoyowonSorry.. i can not use Eng.....^^
12:46.42geertnyoyowon: You can register with ser from asterisk... you can also route calls from ser to asterisk.
12:47.12yoyowoncorrect..
12:47.57yoyowonDo I Wanna konw... how route calls from 'sip proxy' to 'asterisk'..
12:48.49yoyowonasterisk recieved sip Msg... But Forwarding outbound to external line in PRI(E1)
12:49.21vaewyngood 12:49UTC all :}
12:49.28yoyowonNot Forwarding
12:49.41*** join/#asterisk _aggelos_ (~Messenger@cuscon4693.tstt.net.tt)
12:50.42_aggelos_Hi everyone,
12:50.42geertnyoyowon: I'm not sure haven't worked with zap yet only sip. I guess you can setup an extension (with or withouth a prefix) in asterisk which dials out and route the call to the extension.
12:51.16*** join/#asterisk xs (~CAT@a80-126-102-2.adsl.xs4all.nl)
12:51.16_aggelos_I get this strange error from asterisk, it has the same installation as it's sister box:
12:51.30_aggelos_Wait("Zap/1-1", "5") in new stack
12:51.30_aggelos_<PROTECTED>
12:51.30_aggelos_<PROTECTED>
12:51.30_aggelos_<PROTECTED>
12:51.38_aggelos_unable to execture ?
12:51.50bkw_chmod 755 DUH
12:51.55bkw_ok back to bed for me
12:52.01bkw_SOMEONE ELSE TEST BUG 851
12:52.15yoyowonhum....
12:52.19yoyowoni see
12:52.26_aggelos_[daniel@blue sbin]$ ls -l callback.pl
12:52.27_aggelos_-rwxr-xr-x    1 root     root         2754 Jan 12 00:31 callback.pl
12:52.28yoyowonThanks Your comment.
12:52.29vaewyn_aggelos_: make sure it is chmod 755... and also make sure that if it has a "#!/...." line to tell what to run the script under that it points to a real program
12:52.43_aggelos_that script is acctualy running,
12:52.51_aggelos_only that it crashes asterisk every time it does that
12:53.04yoyowoni analyze asterisk....
12:53.37_aggelos_exten => s,1,Wait,5
12:53.37_aggelos_exten => s,2,System(/usr/sbin/callback.pl \"${CALLERID}\")
12:53.41geertnyoyowon: aha:)
12:53.52_aggelos_that's my extensions.conf
12:53.55yoyowongeertn: ^^
12:57.59*** join/#asterisk mitya (~mitya@gw.asylumtel.com)
12:58.02_aggelos_-- Executing System("Zap/1-1", "/usr/sbin/callback.pl 6961737") in new stackJan 15 08:50:45 WARNING[1226062640]: app_system.c:57 system_exec: Unable to execute '/usr/sbin/callback.pl 6961737' ended with -1
12:58.02_aggelos_<PROTECTED>
12:58.16mityaHi everybody
12:58.26_aggelos_hello
12:58.33*** join/#asterisk coppice (~Steve@227.168.17.210.dyn.pacific.net.hk)
12:58.35_aggelos_anyone any ideas ?
12:59.13JerJer[ghost]app_system !?
12:59.15JerJer[ghost]why?
12:59.33YoYo^system(/usr/bin/perl /usr/sbin/callback.pl 5551212)
12:59.52JerJer[ghost]not even close to the right way to dive into a callback system
13:00.11_aggelos_it's not the point,
13:00.24_aggelos_the callerid is sent to another server via a socket,
13:00.42Filacedoes chan_capi support pickupgroups?
13:00.56_aggelos_I pickit up via fxo card in one country, send it to another via internet, and get the callback request from there
13:01.17JerJer[ghost]you don't need app_system for that
13:03.59_aggelos_I did now:    exten => s,2,System(/usr/bin/perl /usr/sbin/callback.pl \"${CALLERID}\")
13:04.00JerJer[ghost]on the 'trigger' machine you run a AGI
13:04.32JerJer[ghost]that sends the call info to the manager port of the other asterisk box
13:04.47_aggelos_and I get Jan 15 09:03:16 WARNING[1217669936]: app_system.c:57 system_exec: Unable to execute '/usr/bin/perl /usr/sbin/callback.pl \"6961737\"' ended with -1
13:05.02JerJer[ghost]chmod u+x callback.pl
13:05.34vaewynJerJer[ghost]: been there... told him that :}
13:05.53_aggelos_strangest thing is when I run asterisk -vvvvgc (console) with no -r (remote), it does not crash the asterisk
13:06.00_aggelos_if I run simple asterisk
13:06.07_aggelos_then asterisk -vvvvr
13:06.19_aggelos_when I get the error it crashes and exits asterisk alltogether.
13:06.35JerJer[ghost]app_system is kludgy
13:06.43_aggelos_[daniel@blue sbin]$ ls -l callback.pl
13:06.44_aggelos_-rwxr-xr-x    1 root     root         2754 Jan 12 00:31 callback.pl
13:06.53_aggelos_it used to work on an old version of asterisk
13:07.04JerJer[ghost]we haven't changed anything in app_system
13:07.13_aggelos_since I updated cvs it gives me this ...
13:07.38_aggelos_\"${CALLERID}\")  <----- is there anything wrong in the notation in your oppinion ?
13:08.13JerJerwhy the slashes ?
13:08.18*** join/#asterisk asterisk-bg (~asterisk-@62.73.103.10)
13:09.33_aggelos_<PROTECTED>
13:09.52JerJerduno
13:10.03JerJeruse AGI
13:11.44_aggelos_thanks
13:12.05*** join/#asterisk sobol_ (~druid@80.51.246.186)
13:16.02*** join/#asterisk Powerkill (~powerkill@host.161.115.68.195.rev.coltfrance.com)
13:16.44*** join/#asterisk f5-Scr^work (~f5-scr@port-212-202-229-162.reverse.qsc.de)
13:17.12f5-Scr^workhi got this failure when i try to use my isdn card WARNING[19473]: channel.c:1644 ast_request: No channel type registered for 'capiCD'
13:17.17f5-Scr^workany idea ?
13:17.33JerJerno such channel type registered for capiCD
13:17.54f5-Scr^workwhen i dial '0' then i want go to my local telefon provider
13:17.58Powerkillhi all
13:18.00f5-Scr^worksorry for my english :)
13:18.10RoyKf5-Scr^work: det er helt greit
13:18.11RoyKlol
13:18.12JerJeri've never heard of caipCD channel type
13:18.23JerJercapiCD
13:18.37[Sim]jerjer: it comes with chan_capi
13:19.28f5-Scr^workcapi_request: didn't find capi device with outgoing msn = 1. you should check your config!
13:19.41f5-Scr^workwhen i try this with normal CAPI channel
13:19.47*** part/#asterisk brown (~brown@sp1024.rbs-p01.ewol.com)
13:19.48f5-Scr^workthis msg comes
13:20.04JerJerdo what the capi god told you to do
13:20.09czmokhi
13:20.11czmokw00p
13:20.15f5-Scr^workin which config ?
13:20.24czmoki am about to have the 7920 working in asterisk :-)
13:20.35czmoki think i know where the error is
13:20.41czmokcurrently i am recoding some stuff :-)
13:22.54mityai would like to compile the asterisk, with openh323-1.12.2 and pwlib-1.5.2
13:22.58mityais it possible???
13:23.03JerJeryes
13:23.06JerJerread the README
13:23.17[Sim]if I want to have a voip-setup and a secretary that can see who is on the phone and who is not
13:23.21[Sim]what would I use ?
13:23.34mityai have already read it
13:23.46mityai says: cd /path/to/openh323
13:23.49mityamake clean opt
13:24.13mityaresult:make: *** No rule to make target `opt'.  Stop.
13:24.47mityawhat is wrong??
13:24.51mityaany idea??
13:27.27geertnI wanna make use of SIP url dialing, is this the right way to go?: exten => _9.,1,Dial(SIP/${EXTEN:1}@${SIPDOMAIN})? (it works but I'm not sure if there is a cleaner and better way to do this.
13:36.51discordiaf5-Scr^work: in capi.conf you must add the outgoing MSNs
13:37.07f5-Scr^workyeah
13:37.16f5-Scr^workjust found my stupid failure
13:37.20f5-Scr^work:)
13:37.30discordiahehe
13:40.26*** join/#asterisk simprix (~simprix@24.247.181.124.gha.mi.chartermi.net)
13:44.49*** join/#asterisk crontibs (Crontibs@ool-44c02950.dyn.optonline.net)
13:45.00[Sim]okay
13:45.14[Sim]has anyone gotten anywhere with sokol's astmgr ?
13:47.22JerJernot here... i even tried it on an XP laptop
13:47.38[Sim]hm
13:47.53[Sim]I got it running here, but I can't get the hang of how to do what
13:48.06*** join/#asterisk Stuart (~Stuart@smtp.dstoys.com)
13:49.07reseauxim back...
13:49.47f5-Scr^workhow i configure asterisk to answer a call from my localnetwork "openphone" ?
13:50.17jimmyzis there something wrong with parking?
13:50.46jimmyzwhen i call sip to another sip and try to park the call it doesn't park...it says there is no one parked on that #
13:50.46JerJerexten => 1234,1,Answer
13:51.24jimmyzit was work two days ago
13:51.30reseauxSomeone Know if is possible to bring a number to call from signal(code preselection function port in DNID the caller in CLI the called nemuber) and made dial before answer a call and if is not busy join the channels?thz
13:55.58*** join/#asterisk sniffe (~sniffe@ti211110a080-3555.bb.online.no)
13:56.03f5-Scr^workhow i configure asterisk to answer a call from my localnetwork for example "openphone" ?
13:57.14JerJerexten => 1234,1,Answer
13:57.19JerJerRTFM
13:57.31f5-Scr^workcool down
13:57.45f5-Scr^worki thought this is a channel to help each other
13:57.46JerJeryou have to help yourself before anyone else can help you
13:58.46f5-Scr^workdont answer my question and ignore me
13:58.50f5-Scr^workdon't flame dude
13:59.05JerJeryou haven't seen a flame yet
13:59.17JerJerread the fuckin docs <---- that's a flame
13:59.41f5-Scr^workmanual :)
13:59.43*** join/#asterisk jtodd (~jtodd@65.199.209.25)
14:00.05f5-Scr^workRTFM <-- not RTMC :P
14:00.10f5-Scr^workehh D
14:00.15f5-Scr^workok im quiete
14:00.28malcolmdg'mornin' #
14:00.41malcolmdhave we ever hit 250 in this chan?
14:00.44JerJerhowdie
14:00.55malcolmddarn thing just keeps on growing
14:03.12*** join/#asterisk erik2 (~eanders@host-127-202-220-24.midco.net)
14:03.23voidptr_omg
14:06.00*** join/#asterisk coppice (~Steve@34.203.17.210.dyn.pacific.net.hk)
14:06.36mityai have already compiled asterisk... OPENH323DIR was wrong... i'm a lamer..
14:07.44_aggelos_JerJer: do you know if anything special needs to be edited in Makefile for asterisk under REDHAT 9 ? tty settings or something like that ?
14:07.57_aggelos_apparently the error I get is related to the terminal.
14:08.10_aggelos_tty related stuff
14:08.25*** join/#asterisk _discordia (disco@calculates.zero-points.for.zetagrid.de)
14:08.42_aggelos_script does not exit with 0 and somehow it bothers asterisk.
14:08.44*** join/#asterisk linuxa (~dave@AMarseille-103-1-3-115.w80-14.abo.wanadoo.fr)
14:09.08*** join/#asterisk martijn (~punisher@213-136-25-66.adsl.bit.nl)
14:09.11martijnhola
14:09.13mityabut, now, when i would like to start it, it dies with the following error: libpq.so.3: cannot open shared object file: No such file or directory
14:09.45martijnis there any thought on when zaptel will be released for linux 2.6 kernels?
14:10.28JerJermitya: plus you gota READ the README in asterisk/channels/h323
14:10.40h3xi <3 Gentoo
14:10.53malcolmdh3x: yeah, gentoo's nice
14:11.09martijnnoone has an idea?
14:11.26vaewynAFKgentoo is nice for single installs... when you gotta admin multiples it sux
14:11.41h3xno way
14:11.52vaewynmartijn: A couple people have alreay hacked it over to work... but no official releases yet
14:12.08martijnah
14:12.15martijnare those patches for download somewhere?
14:12.17h3xa distro shouldnt ever have anything to do with administrating boatloads of machines
14:12.32h3xtheres plenty of applications to aide with that
14:12.47vaewynh3x: yeah... and the best ones arn't in gentoo
14:12.52vaewynyet
14:12.55vaewyn:}
14:12.58h3xsuch as
14:13.24martijnvaewyn: are those patches for download somewhere?
14:13.31heisonis iaxtel down?
14:13.32vaewynsuch as the easiest way to clone production machines   dpkg --get-selections | ssh remotemachine dpkg --set-selections   :}
14:13.50vaewynmartijn: I am not sure myself... I have just heard mention... not sites sorry :{
14:13.54h3xbahg
14:14.08martijnok, then I'll have to do it myself also :)
14:14.47*** join/#asterisk mbranca (~matteo@213.140.14.155)
14:14.53h3xeasiest way to clone a machine is pop the new drive in another as a slave
14:15.00h3xand dump/restore all the shiznit
14:15.24mbrancahi people
14:15.41h3xor
14:15.45h3xuse compactflash cards
14:16.01vaewynmartijn: bkw said to check the README.*26 in the zaptel stuff
14:16.14h3xif you can live with a small installation
14:16.41martijnthanks. :)
14:17.04vaewynh3x: not when you have 200+ "production" machines... that can't go offline long :P
14:17.51vaewynexcept for kernel upgrades these puppies never go down
14:17.52h3xnetboot the mofos
14:17.59*** join/#asterisk Asterisky (~IAX2_Lost@12-220-205-73.client.insightBB.com)
14:18.07*** join/#asterisk ecd (~dslater@ool-44c17c85.dyn.optonline.net)
14:18.13h3xwelcome to the age of gigE
14:18.16vaewynThese have to be full local machines...
14:18.30h3xdepends on your application i guess :)
14:18.42vaewyngigE   hahaha... already saturating 2 channels of those as is :P
14:19.07h3xi was kind of thinking asterisk farms myself but...
14:19.13voidptr_my god
14:19.14vaewynacademia is such a fun place to play in :}
14:19.16voidptr_ripoffs!
14:19.27voidptr_first they deliver the wrong materials
14:19.33voidptr_i call them, so they send extra things
14:19.39voidptr_then they charge me
14:20.04vaewynvoidptr_: easy to deal with... kill them
14:20.05h3xhaha
14:20.15martijnyep
14:20.21martijntake a m4
14:20.34voidptr_i will call them in a bit
14:22.13mbrancahttp://slashdot.org/article.pl?sid=04/01/15/022220&mode=nested&tid=126&tid=172&tid=185
14:22.15vaewynpssh... and hear I thought I would get to see your picture on the news :}
14:22.18mbrancah323 flaws...
14:22.25martijnheh
14:22.41martijnsome 17 year old turkish guy shot his teacher at school here the other day
14:22.45martijnwhole country at hassle
14:23.03martijnbut we all know his face now
14:23.05martijn;)
14:23.26fwd28326anyone heard anymore rumours on the tdm400p FXO modules, has anyone actually seen a prototype, or are they mythical :)
14:24.17h3x*you are getting sleeeeeppy*....  *you know you want a t1....*
14:24.40voidptr_fwd28326 : i'm about to send a mail about that :P
14:25.12fwd28326voidptr_ can we see it before everyone?
14:25.15fwd28326please !
14:25.25fwd28326id love a T1
14:25.32voidptr_no no
14:25.33h3xi have a PRI at home !
14:25.42voidptr_i'm about to send a mail to digium about it
14:25.44fwd28326or even a PRI
14:25.48martijnhehe.. depends on how good the "service" is :P
14:26.05fwd28326but the costs here are off teh scale
14:26.09*** join/#asterisk Cinzas (~cgiirc@p15097463.pureserver.info)
14:26.13h3xwhere are you ?
14:26.17fwd28326ireland
14:26.20mbrancafwd28326, the card exists as a prototype, but not released yet due to fxo module problems
14:26.22h3xhaha so E1
14:26.24Cinzasickserv identify icnopleu
14:26.29fwd28326yes
14:26.38reseauxHi Mbranca ...!!! Im back finaly.... :-)
14:26.49h3xwhats your local carrier out there
14:26.54fwd28326well actually partial PRI
14:27.03fwd28326is all most people ever see
14:27.07fwd28326eircom
14:27.19fwd28326incumbant former stae owned
14:27.26mbrancaciao reseaux, dove cazzo eri finito! :)
14:28.09RoyK"18.2 The Journaling File System (JFS) code contributed to Linux by IBM was almost certainly copied and adapted for Linux from a version of AIX more recent than the one available for comparison"
14:28.11reseauxCiaooo ... non avevo un cazzo e dovevo risolvere un po' di casini!! ex lavoro..
14:28.46mbrancareseaux, :) ok. hope your company started well
14:28.58martijnde mazzel ;)
14:29.11fwd283263200 for an isdn primary rate!!
14:29.18reseauxyeah... in this day i bring my two * asterisk server in CUSS in Milano
14:29.36mbrancaCUSS?
14:30.17reseauxis a MIX where we can join Colt MCI and other telecom company...
14:30.28mbrancauh, ok
14:31.15RoyKfwd28326: 3200 what?
14:31.35reseauxnow i have 2 server with 1 TE410P working on every one and work very well...
14:31.40fwd28326euro
14:31.43fwd28326aplogies
14:31.59fwd28326get so used to (nearly) everyone being euro in europe now
14:32.13fwd28326thats for installation
14:32.38*** join/#asterisk wreckdiver (~goaway@adsl-068-209-107-007.sip.mia.bellsouth.net)
14:33.14fwd28326you get channels in multiples of 16 @ €158 per 16 channels
14:33.36*** join/#asterisk zoa (~john@kn-upc-1.kuleuven.net)
14:33.42zoayellow
14:34.03fwd28326which isnt to bad
14:34.12fwd28326it's the 3200 up front!!
14:34.39*** join/#asterisk dalabera (~Dalabera@206.137.96.110)
14:35.38*** part/#asterisk Cinzas (~cgiirc@p15097463.pureserver.info)
14:36.21heisonhow come the version number never changes as I do cvs updates follow by make or even cvs update -d followed by make?
14:36.24*** join/#asterisk Cinzas (~cgiirc@p15097463.pureserver.info)
14:36.42CinzasHI
14:37.18CinzasI cant make call from H323 to SIP, with asterisk0.7.1
14:37.34Cinzas<PROTECTED>
14:37.51*** join/#asterisk Seba2 (~se@OL236-72.fibertel.com.ar)
14:37.58Seba2hello all
14:38.03CinzasAny idea ?
14:38.10Seba2I nees somebody who can help me
14:38.19mbrancaCinzas, does the user 911 exists in sip.conf AND is authenticated (sip show peers)?
14:38.23Seba2I am having troubles using AGI
14:38.28Seba2with function STREAM FILE
14:38.37CinzasName/username    Host                 Mask             Port     Status    911              (Unspecified)        0.0.0.0          5060     Unmonitored910              (Unspecified)        0.0.0.0          5060     Unmonitored
14:38.44Seba2When I use this function, nothing happens
14:38.45CinzasI can make call from the sip to an h323 endpoint
14:39.09Seba2but if I use playback inside extensions.conf all work OK
14:39.11heisonkram: how come the version number never changes as I do cvs updates follow by make or even cvs update -d followed by make?
14:39.25Seba2any idea?
14:39.43mbrancaCinzas, your sip endpoint must register to * to be able to be called
14:39.58*** part/#asterisk LuchoAR (~ll@200.85.102.10)
14:40.08*** join/#asterisk LuchoAR (~ll@200.85.102.10)
14:40.56*** join/#asterisk rocketman (~rocketman@hoochie.digium.com)
14:40.57AsteriskyI am trying to create an iax user while asterisk is running.  asterisk -rx reload doesn't seem to do it.  Any ideas?
14:41.30rocketmandoes anyone know when some of these drivers will be ported to the 2.6 kernel?
14:41.31Cinzasmbranca: The extensions appear  in "sip show peers"
14:41.41CinzasAren't registered ?
14:41.50kramasterisky: reload should most assuredly do it
14:41.51mbrancaseems not
14:41.51rocketmanor is there a faq I can look at regarding that question?
14:41.54zoaits already ported to 2.6
14:41.56kramwhat are you seeing when you reload?
14:42.06kramthey are in principle ported to 2.6 already
14:42.08krambut untested
14:42.12kramand ztdummy is not yet ported
14:42.22RoyKkram: seems zaptel's broken
14:42.30rocketmanI see ..
14:42.33kramroyk: more details?
14:42.38mbrancaCinzas, see the voice (Unspecified)? there must be an ip addr when a sip endpoint registers with *
14:42.51Asteriskykram, it reloads, but doesn't let the new user registyer.
14:42.55mbrancakram, zaptel un 2.6.1 & wcusb crash my kernel :)
14:42.57Seba2when I use STREAM FILE inside AGI nothing happens, but when I use Playback inside extensions.conf all work OK, any idea?
14:43.01rocketmanthanks guys
14:43.12zoaif i change something in iax.conf i always reload asterisk completely
14:43.21zoaalthough i didnt test it for added users
14:43.31Cinzasmbranca: Ok. I'll see that
14:43.33Nate187can asterisk answer faxes?
14:43.33zoa*restart asterisk completely
14:43.38zoaNate187: yes
14:43.40krammbranca: maybe i can get someone to set it up.  are you running without preemptive kernel?
14:43.46kramand without smp?
14:43.54mbrancakram, yes both
14:44.01kramwith both or without both?
14:44.24mbrancaboth without, sorry. I noticed that preempt caused problems
14:44.26AsteriskyZOP:  you can't restart asterisk completely when other users are on calls.  That's what reload is for
14:44.33mbrancabut still, no luck
14:44.41zoaAsterisky: i know
14:48.10*** part/#asterisk ecd (~dslater@ool-44c17c85.dyn.optonline.net)
14:48.19Seba2when I use STREAM FILE inside AGI nothing happens, but when I use Playback inside extensions.conf all work OK, any idea?
14:48.23*** join/#asterisk wcd (~dslater@ool-44c17c85.dyn.optonline.net)
14:48.29Seba2please somebody can help me?
14:48.36Seba2I don't understand why
14:48.38RoyKkram: seems i was missing some kernel headers
14:49.02kramroyk: i see
14:49.06dalaberaseba what's the command sentence you're using?
14:49.17kramroyk: well, it's fixed now?
14:49.19Seba2I will put the line here
14:49.48Seba2print "ANSWER\n";
14:49.48Cinzasmbranca, i defined the host=ip in sip conf, where ip is the ATA 186 ip !
14:49.54CinzasNow, he creates the channel
14:49.56Seba2print STDERR "6a.  Testing 'record' playback...";
14:49.56Seba2print "STREAM FILE vm-options \"\"\n";
14:50.02Seba2&checkresult($result);
14:50.02Seba2print "HANGUP\n";
14:50.06Seba2that's all
14:50.29Seba2is ok I think
14:50.42Seba2but I can't hear anything
14:51.42rob--kram: can I ask you something about app development?  I'm not sure how to handle the timing when sending voice frames to a channel.
14:51.52Seba2exten => s,6,BackGround(vm-options)
14:51.55Seba2this work ok
14:52.22Seba2any idea=
14:52.23Seba2?
14:52.39kramrob-- what do you mean
14:53.34RoyKkram: need to build 2.4.24 first
14:53.41Seba2dalabera?
14:53.46RoyKkram: btw. is 2.6 support (zaptel) stable with e100p?
14:53.47zoa2.4.24 sux :-p
14:53.52kramroyk: did you read the README.Linux26 already?
14:53.55kramroyk: it's not been tested
14:53.56RoyKyep
14:53.59RoyKok
14:53.59kramroyk: it just compiles
14:54.05zoakram i hope to test it very soon
14:55.16*** join/#asterisk Alric (~nbowyer@masq.hyperusa.com)
14:56.25dalaberazeba your not using the perl Asterisk module. You should look into that. That way it wont that easy...
14:56.55LuchoARzoa why does it sux?
14:57.21Seba2Lucho sos vos?
14:57.27LuchoARsi
14:57.34Seba2oka
14:57.38Seba2dalabera
14:57.44*** join/#asterisk anthm (~anthmct@CPE-24-167-193-161.wi.rr.com)
14:58.03Seba2I test Perl module agi-test but it did not work
14:59.07ScaredyCatSeba2: http://asterisk.gnuinter.net
14:59.07reseauxOPPORTUNITY: I need someone to develop a mysql frontend for * in php? a pay for it... :-) thanks
14:59.35LuchoARreseaux: what do you need it to do?
14:59.50dalaberayou need to intall it in order tu use and it should work
15:00.18Seba2can you tell me how?
15:00.28ScaredyCatfollow the instuciton
15:00.29ScaredyCats
15:00.34rob--kram: I'm trying to write an app for the Cepstral Theta TTS engine.  I need to stream the audio which is generated to the channel, and listen for dtmf. When do I send voice frames, and how many samples should be in each frame? I looked at the code in
15:00.46Seba2what instructions?
15:00.55ScaredyCathttp://asterisk.gnuinter.net
15:00.57ScaredyCathttp://asterisk.gnuinter.net
15:01.06rob--file.c, but I can't get my head arround it. Is there a way of doing it without scheduling a callback?
15:01.14RoyKScaredyCat ScaredyCat ScaredyCat ScaredyCat
15:01.37Seba2I am reading that page
15:01.48ScaredyCatues rob--  do what I go in my app_cepstral .. create the file and stream it back
15:01.57ScaredyCatues? yes
15:02.01Seba2I think if Asterisk comes wiht and agi-test it should work
15:02.20ScaredyCatit does work... we are just suggesting you use the perl module
15:02.36ScaredyCatit will make you life easier
15:02.38ScaredyCatyour
15:02.47Seba2yes
15:02.49Seba2sure
15:03.17jtoddrob--: I look forward to the release of app_cepstral.   :-)  I think that would be pretty cool.
15:03.19Seba2but first I want a simple agi script to work
15:03.24jtoddrob--: Sorry I have no hints for you, though.
15:03.45rob--scaredycat: I could do, but it'd be more efficient to stream from memory. Theta can stream the audio as it is being synthesised.
15:03.46Seba2I am downloading Perl AGI
15:03.52jtoddScaredyCat: do you already have app_cepstral  ?  Is it released?
15:04.15ScaredyCatrob--: this is true.. but i wanted it quick
15:04.16zoagreat, one of my servers is under attack
15:04.29ScaredyCatjtodd: not release but you can have a copy..
15:05.24*** join/#asterisk Shido6_ (~shido@d57-81-103.home.cgocable.net)
15:07.25*** join/#asterisk Takapa (vegard@svanberg.no)
15:07.33rob--It'd be good if the streaming code wasn't file-specific. If I could provide a read function like in file.c and it would just work.
15:08.01rob--I suppose I could use a named pipe and use the file streaming code
15:08.13rajoanyone experience with gnophone and video transmission?
15:08.18rob--But it's a bit of a hack
15:09.15*** join/#asterisk montag (~montag@host166-150.pool80105.interbusiness.it)
15:09.16*** join/#asterisk Tekati (~captain@cpe-66-75-211-60.bak.rr.com)
15:09.18montag<PROTECTED>
15:09.18montagFailed to load driver chan_modem_aopen.so
15:09.22montagwhat can be ???
15:09.22*** join/#asterisk asterisk-bg (~asterisk-@62.73.103.10)
15:09.39*** join/#asterisk Junbug (~junya@adsl-34-194-140.bct.bellsouth.net)
15:09.55montagi use a voice modem connected to serial port
15:09.55LuchoARis anyone developing a prepaid app with the asterisk API ?? we are about to begin a project of it
15:10.31LuchoARand would like to know if anyone is doing something so we can join resources.
15:10.53rob--http://www.voiptalk.org have a pre-pay system using asterisk, but I don't know what it is or if it is released.
15:12.15LuchoARwhere?? it don't see it in the webpage.
15:13.36*** join/#asterisk pattieja (~pattieja@63-252-5-1.ip.mcleodusa.net)
15:14.01rob--they use it for there siptalk and iaxtalk service. I think it's probably a custom app
15:14.23rob--but you could contact them at info@telappliant.com
15:14.29*** join/#asterisk _asr (asr@pimpbox.latency.net)
15:15.14LuchoARaa.. ok..
15:15.16LuchoARtnx!
15:15.42crontibsasr
15:15.44crontibswhats up
15:17.20sobol_any cisco fans here willing to help with 7960
15:18.18*** join/#asterisk junya (~junya@adsl-81-1-250.mia.bellsouth.net)
15:20.31zoarob-- its not a prepay system
15:20.41zoanot yet
15:20.44zoaits a rating engine
15:21.09zoai'm testing it
15:21.44Powerkillzoa it's tholo one ?
15:22.03zoayes
15:22.25zoaoh no
15:22.28zoaits not tholo's
15:22.40zoaits telappliants
15:22.42Powerkillit's better ? cause tholo one have lot of segfault
15:22.44zoai don't think its opernsource
15:22.55Powerkillcan i test it ?
15:23.00zoai don't have it
15:23.05zoadon't think they will release it
15:23.08zoai'll aks
15:25.35Powerkillok thanks
15:25.50zoapowerkill: JerJer has a prepay engine that is supposed to be very good
15:25.53zoaand its for sale
15:25.56Aviaahalo again, anyone know how set into extension.conf if I want make this call: want first dial 123 pause 1sec and then 456 ?
15:27.02ManxPowerDial(Zap/1/123ww456)
15:27.09ManxPowerOnly works on analog ports
15:27.20AviaaManxPower i`ll ...
15:27.34AviaaManxPower let me try
15:27.52ManxPowerw is .5 second pause I think, you might try three w's
15:28.08*** join/#asterisk variable1 (~variable1@hoochie.digium.com)
15:28.16*** join/#asterisk brent21 (bdf@paalto-apx-1-144-142.penn.com)
15:29.58AviaaManxPower not working ...
15:30.06ManxPowerWhat is your dial line?
15:30.08Aviaamaybe "P"
15:30.15ManxPowerNo_Carrier_, "w" is a pause.
15:30.19ManxPowerNo "w" is a pause.
15:30.54[Sim]w00t
15:31.13[Sim]anyone with amsix lines: peer with us :-)
15:31.34Aviaa(SIP/123w456${EXTEN}@some.ip.host|60)
15:31.39ManxPowerAviaa, What Dial line are you using in extensions.conf?  What displays on the Asterisk CLI when the Dial is executed?
15:31.47ManxPowerAviaa, SIP IS NOT ANALOG!
15:32.02Aviaaaj si .... .<ups>
15:32.03ManxPowerYou cannot use "w" with anything except for an analog channel/port.
15:32.48ManxPowerI am not aware of any way to put a pause on a Dial using a digital channel/port
15:32.48AviaaManxPower than we can not set any pause in sip ... am i right
15:33.10ManxPowerAviaa, make it pause on the remote gateway.
15:33.14*** join/#asterisk Stealth_Man (Stealth_Ma@hyp1-19.dialup.online.ge)
15:33.25Aviaaisee
15:33.36bkw_Corydon you around?
15:34.05zoabkw, can i help you ?
15:34.06ManxPowerAviaa, This is a limitation of VoIP protocols, not Asterisk
15:34.16zoaits prolly for some testing right  ? :)
15:35.24bkw_i'm trying to track down the malloc in chan_zap that fails to free when you load it when the signalling is incorrect
15:35.30brent21Should I be worried about seeing these messages occasionally? (ast_readaudio_callback): Failed to write frame?
15:35.36bkw_I think I see it but I don't know why a "return NULL" is there
15:35.38brent21It usually happens when users are in the voicemail system
15:35.38bkw_that doesn't seem right
15:36.32zoaic, i cant help you then i'm affraid :)
15:36.44bkw_line 5192 in mine
15:36.52bkw_<PROTECTED>
15:36.52bkw_<PROTECTED>
15:36.52bkw_<PROTECTED>
15:36.52bkw_<PROTECTED>
15:36.52bkw_<PROTECTED>
15:37.04bkw_i'm going to test something
15:37.14sobol_is it possible to load logo to the Cisco7960 from tftp not ftom logo_url "http://"?
15:37.24bkw_sobol_ yes
15:37.38bkw_it has to be bmp or jpeg(only works in newest firmware)
15:38.32sobol_bkw_: and how it looks on config logo_tftp ?
15:38.44ScaredyCathas anyone got the doxygen docs for * online - wassim gave me a link but i lost it
15:40.22bkw_logo_url: http://alsdkfjlasdf.com/blah.bmp
15:40.47dougheckaCOOL WEBSITE!
15:40.51doughecka:P
15:42.42ScaredyCat/s/in//
15:42.54sobol_bkw_: but how t dowload logo from tftp not http
15:44.10bkw_you can't
15:44.15scrI came in this morning and asterisk had segfaulted on me.  I'm looking at the backtrace, but I don't really know what I'm looking for
15:44.21*** join/#asterisk huats (~chris@AToulouse-104-2-1-24.w217-128.abo.wanadoo.fr)
15:44.22bkw_i'm not aware.. sobol_ why not read cisco's site on this?
15:44.37bkw_scr what does the first three lines of the bt say?
15:44.54scr#0  0x40239f2c in mysql_fetch_row () from /usr/lib/libmysqlclient.so.12
15:44.54scr#1  0x4022697e in do_directory (chan=0x82f5170, cfg=0x82eca78, context=0x0, digit=55 '7') at app_directory2.c:227
15:44.54scr#2  0x4022669b in directory_exec (chan=0x402275f6, data=0xbd5ff274) at app_directory2.c:300
15:45.10bkw_1 you are using app_directory
15:45.12bkw_with mysql
15:45.14bkw_NEXT!!!
15:45.15martijnZT_SPANCONFIG failed on span 1: No such device or address (6)
15:45.21martijndoes anyone have a clue?
15:45.26martijnztcfg output
15:45.29bkw_no such device or address
15:45.31bkw_your configs are wrong
15:45.35bkw_double check your configs
15:45.37*** join/#asterisk simprix (~simprix@ip-64-32-242-201.dsl.iad.megapath.net)
15:45.41Seba2hello
15:45.42bkw_and make sure you kernel modlues are in
15:45.46martijnthey are
15:45.59scrwhat's wrong with using directory and a db?
15:45.59Seba2somebody knows how to use musiconhold inside AGI?
15:46.01martijnand i copies the configs from a running box
15:46.13*** join/#asterisk nighty (~nighty@hoochie.digium.com)
15:46.26martijns/copies/copied/
15:46.34bkw_Seba2 print "SET MUSIC ON default\n";
15:46.40Seba2tx
15:46.43huatsdoes anybody know a way to connect to a call and just speak to one of the people who are in this call ?
15:46.58MSpinmorning bkw
15:47.17martijnbkw_- the configs are from a stable and running box
15:47.24martijnand the mods are loaded
15:49.03bkw_lsmod
15:49.06bkw_ztcfg -vvvv
15:49.12martijndid that
15:49.14bkw_rmmod's
15:49.18bkw_reinstall the mods
15:49.22martijndid that
15:49.22bkw_ztcfg -vvvvv again
15:49.26martijndid that
15:49.31dougheckashutdown -r now
15:49.32bkw_cat /proc/interrutps
15:49.39dougheckainit 0
15:49.42bkw_doughecka SHUT UP this isn't windows
15:49.51bkw_martijn but have you rebooted the box
15:49.52martijnit says that all channels are configured
15:49.58dougheckashutdown -r now is the command for redhat
15:50.02doughecka:P
15:50.13martijndoughecka- that's not what bkw_ meant
15:50.13bkw_doughecka grow up and be helpful please :P
15:50.17martijn;)
15:50.31doughecka:P
15:50.36bkw_martijn just for the sake or argument reboot the box
15:50.41martijnbah
15:50.44dougheckahumbug
15:51.00martijnbox has been rebooted once since install
15:51.07martijnso that should not be an issue either
15:51.11martijnmods are in
15:51.17bkw_well I hear it has in the past been an issue
15:51.21bkw_are you using latest cvs on zaptel?
15:51.26martijnztcfg tells me all channels ar configuyred
15:51.31martijnyea
15:51.32bkw_asterisk -vvvvgc
15:51.34bkw_what does it say?
15:51.36martijnjust got it out
15:51.42martijnChannel 123: Individual Clear channel (A-law) (Slaves: 123)
15:51.42martijnChannel 124: Individual Clear channel (A-law) (Slaves: 124)
15:51.42martijn124 channels configured.
15:51.43martijnZT_SPANCONFIG failed on span 1: No such device or address (6)
15:51.57bkw_you sure your have it rigiht?
15:51.58martijnzaptel and tor2 are loaded
15:52.06bkw_show me your spans in zaptel.conf
15:52.06martijni copied the configs
15:52.18martijnspan=1,0,0,ccs,hdb3,crc4
15:52.18martijnspan=2,0,0,ccs,hdb3,crc4
15:52.18martijnspan=3,0,0,ccs,hdb3,crc4
15:52.18martijnspan=4,0,0,ccs,hdb3,crc4
15:52.27bkw_you sure you need that crc4?
15:52.29*** join/#asterisk cypromis (~michael@80.51.246.186)
15:52.39cypromishmmm
15:52.47[Sim]hmm?
15:52.50cypromisanybody has callerid display with the latst firmware of the barbietones ?
15:52.51bkw_hoo?
15:52.56*** join/#asterisk TeleRidd (~TeleRiddl@hoochie.digium.com)
15:52.56martijnthat's what my other config runs
15:53.04bkw_martijn very strange
15:53.08martijnyea
15:53.13bkw_reboot the box
15:53.14martijni run linux 2.6 though
15:53.20bkw_oh and the truth be told
15:53.21martijn2.6.1
15:53.22*** join/#asterisk montag (~montag@host166-150.pool80105.interbusiness.it)
15:53.22bkw_NEXT!!
15:53.28martijn;)
15:53.30TeleRiddIs this support channel an "official" digium support forum or is it just users?
15:53.31bkw_2.6 isn't totally supported yet
15:53.32martijnshould not be a problem
15:53.34montagany tips for a voice modem with asterisk ???
15:53.41bkw_TeleRidd its for the users
15:53.56TeleRiddOkay, thank you
15:54.09TeleRiddHere is my problem, as many might have heard from yesterday
15:54.40TeleRiddOur system was working fine with all configs until we had some power spikes as indicated by our Backup Power Supplies
15:55.05bkw_have you treated yoru boxes like windows?
15:55.07TeleRiddWe can receive incoming calls just fine but outgoing calls yield no audio
15:55.17bkw_is this a T410P?
15:55.17TeleRiddYes, rebooted
15:55.44bkw_cards?
15:55.50TeleRiddWhen we listen in on the X100P card dialing out we can hear numbers being dialed
15:56.05*** join/#asterisk Jon_UK (Jon_UK@host217-35-96-128.in-addr.btopenworld.com)
15:56.16Jon_UKHi everyone!
15:56.19bkw_wonder if it was fried
15:56.22TeleRiddthen we receive a message from the phone company that the number could not be completed as dialed
15:56.29Jon_UKIs anyone here using a SNOM 200 phone with *
15:56.48TeleRiddBut if it still receives incoming calls, how could it not dial out?
15:56.49bkw_snom are fugly phones.. CISCO OR DEATH
15:57.02bkw_TeleRidd have you modified digits.h in zaptel any?
15:57.07zoasnoms are not too ugly
15:57.08TeleRiddusing Snom200 phones here.  Mess to setup but getting better
15:57.16*** join/#asterisk tim27 (tim27@229-29.dr.cgocable.ca)
15:57.16Jon_UKbkw_ easy to say if you've got the money for a cisco phone!
15:57.31TeleRiddNo, phone.conf was missing when we rebooted but that is it
15:57.32tim27Jon_UK i got the money for 2 7960
15:57.35tim27and 2 7905
15:57.38bkw_Jon_UK I don't have the money.. I just want reliable phones
15:57.40TeleRiddjust regenerated from "make samples"
15:57.57Mikeis queue a good idea insted of call wait?
15:58.00bkw_TeleRidd make samples will remove all your stuff
15:58.06bkw_and phone.conf is for a quicknet card
15:58.23Jon_UKI am trying to get my SNOM 200 to light up a different line button depending on which extension is receiving the call, without success! has anyone managed this?
15:58.26tim27bkw_ : my 7960, are working now
15:58.28tim27:))))
15:58.35TeleRiddI know this, I backed up all my conf files before I did this and then moved them back into the directory with the new phone.conf
15:58.41tim27register to proxy was set to off
15:58.47RoyKHow many calls would you think I should be able to handle with a Duron 1400?
15:58.54zoadepends royk
15:58.58zoawhat do you want to do ?
15:59.01zoatranscoding ?
15:59.01bkw_phone.conf don't even need to be there
15:59.04zoatermination ?
15:59.07zoawhat codecs ?
15:59.14TeleRiddokay, that is what I thought.
16:00.13Jon_UKTeleRidd, have you got the function buttons programmed on your SNOM200 ?
16:00.14RoyKI just setup a ultra-lowcost system with an MSI mb, a duron 1400, 256MB RAM and two ISDN cards for < ¤300
16:00.23tim27bkw_: cisco are reliable, but the LCD screen on the 7960 look cheap as the grandstream phone
16:00.26TeleRiddyeah I do
16:00.34bkw_tim27 UM now
16:00.36bkw_er no
16:00.44zoaroyk: what isdn card ?
16:00.48RoyKso I want to produce an ugly-looking ultra-low-cost pbx
16:00.50RoyKhfc-pci
16:00.51bkw_it may not be highres but damn it its nice
16:00.53TeleRiddI have just renamed phone.conf -> phone.conf.bak and then restarted Asterisk
16:00.53zoathats like the price of 1 isdn card !!!
16:01.01TeleRiddit says it is unable to load now
16:01.10RoyKkapijod will come up with zaptel drivers for them soon
16:01.13simprixhow much do channel banks run these day s
16:01.16bkw_TeleRidd do you use chan_phone?
16:01.17TeleRiddso I must need phone.conf then?
16:01.27bkw_you must not have an X100p then
16:01.28RoyKzoa: not el cheapo hfc pci
16:01.40TeleRiddI have 3 of them in the slots
16:01.45Mikeanyone of you guys know of any cellphone that has wifi in it and that has a sip client that works with asteirsk?
16:01.47TeleRiddwhat do I use chan_phone for?
16:03.25TeleRiddWhat is phone.conf used for?  I know it says linux telephony driver.  I must need it if * won't start
16:04.05bkw_no you don't need it unless you have quicknet phone cards
16:04.11UnixDawgMIKE yes the palm phone
16:04.24Jon_UKMike X-Lite works well from a pocket PC
16:04.24bkw_if you have quicknetphone cards you might as well jump off a off a bridge now
16:04.26pros12mike: what phone is this?
16:05.36MSpinmorning kram...
16:06.19TeleRiddOkay well here is what asterisk is telling me when I start.
16:06.26RoyK~x100p
16:06.35zoakram: are the E400p's still for sale ?
16:06.50TeleRidd..Jan 15 09:59:43 ERROR[1074432736]: chan_iax.c:4826 set_config: Unable to load config iax1.conf
16:07.01TeleRidd.....Jan 15 09:59:43 WARNING[1074432736]: chan_iax2.c:5510 set_config: Ignoring port for now
16:07.04zoathats normal
16:07.07zoateleriid
16:07.09bkw_TeleRidd that has nothing to do with it
16:07.10TeleRidd...Jan 15 09:59:43 ERROR[1074432736]: chan_phone.c:1123 load_module: Unable to load config phone.conf
16:07.18TeleRiddJan 15 09:59:43 WARNING[1074432736]: loader.c:312 ast_load_resource: chan_phone.so: load_module failed, returning -1
16:07.25TeleRiddJan 15 09:59:43 WARNING[1074432736]: loader.c:312 ast_load_resource: chan_phone.so: load_module failed, returning -1
16:07.26bkw_then noload => chan_phone.so in modules.conf if you don't need it
16:07.27zoateleridd: unload chan_phone,
16:07.32zoaunload iax1.so
16:07.41zoabkw, or is that still iax.so ?
16:07.46bkw_noload => chan_iax.so
16:07.52bkw_yes its still chan_iax.so
16:07.53zoaah k
16:07.56RoyKWhen should one use IAX1 and when should I use IAX2?
16:08.04simprixwith a sip client do i have to dial 9 to get a outside line for a pstn call
16:08.07bkw_use IAX2 at all times
16:08.24TeleRiddHow do I unload the .so modules
16:08.32bkw_modules.conf
16:08.38TeleRiddsorry for the newbie question, never had to do that before
16:08.39bkw_put noload => chan_phone.so
16:08.49TeleRiddin modules.conf right?
16:08.54bkw_yes
16:09.52Powerkillwhat the cvs date to upgrade to if i want the 0.7.1 ?
16:10.27bkw_you download the tarball from ftp.digium.com
16:10.34tim27bkw_ : the half tones on the lcd look crapy... ( i mean the grey tab behin the text of the soft key) ... same on the 2 phone
16:10.46TeleRiddalright, Asterisk started
16:11.08*** join/#asterisk extremis (extremis@121-17.waldenweb.com)
16:11.10bkw_tim27 don't halftone anything then
16:11.24tim27what you mean
16:11.35tim27i cant no halftone them
16:11.47extremisso... to change the musiconhold context per extension that gets dialed, do you just add a priority to that extension before it gets dialed for MusicOnHold?
16:11.59tim27when you press the softkey HOld, there is a grey tab behin the hold text
16:12.14tim27this look very crapy heaven if i ajust contrast
16:12.24bkw_oh thats not halftone.. adjust the contrast
16:12.29bkw_mine looks perfect
16:12.34bkw_maybe your being too picky
16:12.40extremisbkw_: I've seen it vary between 7960's
16:12.46extremisthe lcd's on the 7960's suck
16:12.49bkw_extremis really?
16:12.52bkw_mine both look perfect
16:12.53extremisall mine have different splotches
16:12.57extremisI have 3
16:12.59extremisthey all vary
16:13.00tim27mines 2
16:13.10MSpinthe ones we had looked great
16:13.12simprixwith a sip client do i have to dial 9 to get a outside line for a pstn call
16:13.12tim27the light grey synch
16:13.16MSpin(my old company)
16:13.28extremistim27: just bitch to cisco and have em send you another one :)
16:13.35extremisso... about my musiconhold question?
16:13.35*** join/#asterisk Inv_Arp (~junya@68.223.139.104)
16:13.44tim27extremis: I didnt buy them new
16:13.50zoaah voila
16:13.56zoayou bought them at ebay
16:13.57zoafor 100$
16:14.02zoaand they ripped you off
16:14.03tim27no for 350 $ CAN
16:14.20tim27maybe thoses guy selling on ebay sell 2nd quality phone
16:14.23tim27i suspect that
16:14.23extremistim27: if you buy from ebay you should know your repair options
16:15.05extremistim27: honestly I wouldn't worry about it... I have splotchy backgrounds on my 7960 too
16:15.07tim27bkw_: buy some phone on ebay too... he like to bid there
16:15.08extremisthe pure black is fine
16:15.16extremisits only some of the greys that look like ass
16:15.17tim27pure black is fine
16:15.24tim27greys look ugly
16:15.27tim27that same here
16:15.28tim27:)
16:15.40tim27but phone speak well
16:15.41tim27:)
16:16.16extremiscan you add a mmusiconhold context to sip.conf?
16:16.33alex-homeWhy would you want to?
16:16.38extremisexten => 2600,1,MusicOnHold(extremis)
16:16.38extremisexten => 2600,2,Dial,SIP/2600|20
16:16.43extremisthat doesn't change the context
16:16.50extremisit still uses default (defined in zaptel.conf)
16:17.28extremisand yes, mpg123 is running for that context
16:17.52Filacecould anyone here help me with configuring 2x BRI lines with asterisk.. i've got two of the lines working (except for the fact that BT have broken one of them)
16:18.33*** join/#asterisk s3gal (~leon@cuscon4743.tstt.net.tt)
16:19.35extremishrm, maybe I need SetMusicOnHold
16:20.00*** join/#asterisk adke (~adke@hoochie.digium.com)
16:21.02*** join/#asterisk jsmith (~jsmith@209.180.83.10)
16:21.12zoawtf:
16:21.15zoaBANNED FILENAME ALERT
16:21.15zoaOur content checker found
16:21.15zoa<PROTECTED>
16:21.15zoabanned filename in an email to you from:
16:21.15zoa<PROTECTED>
16:21.16zoaNot quarantined.
16:21.24zoathats not my mailserver doing that
16:21.30zoaso it must be theirs
16:21.32alex-homeThat could be your ISP?
16:21.35zoanopez
16:21.42zoait the mailserver of that other dude
16:21.44alex-homehmm, probably be theirs then yea.
16:21.49alex-homeGet him to zip the file and send ;)
16:22.10zoai don't even know that guy
16:22.10*** join/#asterisk Sobek (~btatton@209.180.83.11)
16:22.10extremisnope, its not SetMusicOnHold
16:22.10jsmith!lart Sobek
16:22.12jsmith~lart Sobek
16:22.19ManxPowerThat is the usual recommendation for sending EXE files via virus scanning mail servers
16:22.20Sobek~whaleslap jsmith
16:22.24ACTION slaps jsmith upside and over the head with one freakishly huge killer whale named hugh
16:22.25jsmith~meepgun Sobek
16:22.29ACTION shoots Sobek with a hyper-charged flux neutron gun
16:22.29ManxPowerOr just rename the damn file.
16:22.38Sobek~lart jsmith
16:22.40zoaManxPower it should send the virus message to the sender
16:22.47zoanot to the recipient of the other server
16:23.02zoai might as well get 1000 messages with virus warnings now :(
16:23.05ManxPowerzoa, Our mail server used to do that.  Since most virii these days fakes the sender address it's pretty pointless.
16:23.15zoatrue
16:23.17*** join/#asterisk PBXtech (~PBXtechco@67.107.241.3.ptr.us.xo.net)
16:23.48zoaooowz
16:24.39ScaredyCatanyone know what pbx_substitue_variables_temp is?
16:25.24*** join/#asterisk Aharonov (~mcr@desk.marajade.sandelman.ca)
16:25.29extremisok, so I want the music on hold context to be changed for all outgoing calls originating from a specific SIP user... is this possible?
16:25.36bkw_ScaredyCat oh their is a workaround for your UNIQUEID stuff you wanted
16:25.50bkw_it was posted on the bug note before I closed it.. did you get that info?
16:25.51ScaredyCatoh kewl where?
16:26.00ScaredyCatno
16:26.06ScaredyCatagain
16:26.10bkw_citats I think came up with it
16:26.21bkw_its a diffrent API callyou need to use for those
16:26.35extremisI know how to do it with overloading variables... but I want to do it in sip.conf
16:26.37bkw_because they aren't what you would call "channel" variables
16:26.45UnixDawgcannot find -lpg
16:26.52bkw_install pgsql
16:26.54bkw_NEXT!!
16:27.05UnixDawgI have it installed
16:27.28UnixDawgand pgsql-devel
16:27.35bkw_the linker can't find it apparently
16:27.37ScaredyCatbkw_ is you mean use pbx_substitue_variables* thenit's wrong... or at least the example code is very wrong
16:27.53*** part/#asterisk Aharonov (~mcr@desk.marajade.sandelman.ca)
16:28.01bkw_was it james that came up with that?
16:28.03UnixDawggrr this means we have to fix the paths
16:28.05UnixDawggrrr
16:28.12UnixDawgti fbsd
16:28.27ScaredyCatbkw918 01-11-04 02:56  use pbx_substitue_variables_*  
16:28.45bkw_yes I was told that would work
16:28.53bkw_check the api specs
16:28.53ScaredyCatno...
16:28.56bkw_hrm
16:29.02bkw_then we need to expose those vars also
16:29.10bkw_what was that bug number?
16:29.13UnixDawgfrik
16:29.15ScaredyCatthe example uses  pbx_substitue_variables_temp - which doesn't exist
16:29.23ScaredyCathttp://bugs.digium.com/bug_view_page.php?bug_id=0000741
16:29.49drgalaxymy skinny phone works when I register it with our main * box, but when I connect it to my * box and switch its context to the remote one, the sound from me talking into my phone is not transmitted.  any ideas?
16:30.06bkw_ScaredyCat ok its reopened
16:30.12*** join/#asterisk VoiceLynx (VoiceLynx@rrcs-central-24-106-64-175.biz.rr.com)
16:30.15ScaredyCatsubstitute
16:30.43citatsif you look at app_cut you can get an idea of how to get at those special variables.... ideally pbx_builtin_getvar_helper would call pbx_substitute_variables_temp within it to subst those vars
16:31.27jimmyzwhen you do a sip show channels does it tell you what zap channel the phone is using
16:31.54jimmyzi don't have a zap device yet to check this
16:32.08bkw_I need someone with a bit more C clue than myself
16:32.24citatsapp_cut doesn't use variables_temp iirc, it uses variables_helper but it addes ${ } around the var so it doesn't need to do some trickery
16:32.44citatswhen i get a chance i'll make it work right, but i dont know when I'll have a chance
16:33.21f5-Scr^workwhen you dial an empty number, and have early B3 enabled, with:
16:33.21f5-Scr^workDial,CAPI/12345678:b
16:33.21f5-Scr^workthe channel will come up at once and give you the dialtone it gets from the local exchange.
16:33.21f5-Scr^workat this point the channel is like a legacy phone, now you can send dtmf digits to dial.
16:33.26f5-Scr^worki dont understand this
16:33.41f5-Scr^worki want to get a dialtone when i dial the number 0
16:34.08bkw_does it?
16:34.10f5-Scr^worki mean the dialtone of my local provider
16:34.15bkw_your TA should do that
16:34.25bkw_in this case Asterisk right?
16:34.41*** join/#asterisk coolhp (~Dude@unknown-141-roc.globalcrossing.com)
16:34.42ScaredyCatcitats: thanks for the pointer, I'll take a look
16:34.46f5-Scr^workhmm dont know
16:34.55f5-Scr^workTA ?
16:35.02f5-Scr^worksorry im a german
16:35.19citatsScaredyCat: hopefully i'll have a chance in the next few days to come up with a patch for that
16:35.29*** join/#asterisk Shido6_ (~shido@d57-81-103.home.cgocable.net)
16:35.47f5-Scr^workneed a dial rule where i only get to my provider and kann normal dial ...
16:35.54f5-Scr^workhas any one a idea ?
16:36.18Filaceif you dial the B channel on a BT provided ISDN2e circuit, you get a regular dialtone
16:36.33bkw_HOW?
16:36.45bkw_its digital your terminal adapter should provide dialtone
16:36.52Filaceok, it could be the TA
16:36.56f5-Scr^work;exten => _0.,1,StripMSD,1
16:36.56f5-Scr^work;exten => _XX.,2,Dial,capi/${CALLERIDNUM}:bBYEXTENSION
16:36.56bkw_or thats how it is in the US
16:36.58FilaceI said you get a dialtone, not where it came from :)
16:37.00f5-Scr^workwhen i do this
16:37.04f5-Scr^worki can phone out
16:37.08bkw_dont' use BYEXTENSION
16:37.15f5-Scr^workbut i want to dial 0 and get the dialtone
16:37.19bkw_its gonig BYE BYE
16:37.38Filacebkw_: you know anything about connecting more than one BRI line to a * box?
16:37.48f5-Scr^workhmm
16:37.54f5-Scr^workso no one have a idea ?
16:38.05f5-Scr^workor can help me ?
16:38.06f5-Scr^work:)
16:38.15puzzledbkw: chan_capi sports "real" dialtone with e.g. Dial,CAPI/12345678:b
16:38.29Filacef5-Scr^work: why not just setup sommat like exten => _9.,1,Dial,CAPI/MSN:${EXTEN:1}
16:38.37bug847bkw's a nub... don't listen to him
16:38.51Filaceso you can dial 9{number} to call anything external
16:39.02*** join/#asterisk heison (~heison@dyn-65-97.ham.dial.tht.net)
16:39.25f5-Scr^workhmm dont works
16:39.30*** join/#asterisk Jacky (~Jacky@hoochie.digium.com)
16:39.39puzzledf5-Scr^work: look in the chan_capi README under "Overlap sending"
16:39.59f5-Scr^workOverlap sending (a.k.a. real dialtone)
16:40.00f5-Scr^workwhen you dial an empty number, and have early B3 enabled, with:
16:40.01f5-Scr^workDial,CAPI/12345678:b
16:40.03f5-Scr^workthe channel will come up at once and give you the dialtone it gets from the local exchange.
16:40.04f5-Scr^workat this point the channel is like a legacy phone, now you can send dtmf digits to dial.
16:40.14f5-Scr^workbut i dont understand this
16:40.29ScaredyCatbkw_: you can close it again
16:41.00bkw_ScaredyCat ok
16:41.18scrbkw: you objected earlier to my using mysql and directory...  was that opposition to mysql (over odbc) or should I not be using the directory w/ a db at all?
16:41.18UnixDawgfrick
16:41.32bkw_scr no
16:41.40bkw_but you are using something thats not in CVS
16:41.45bkw_or is mysql directory in CVS now?
16:41.58bkw_nope doesn't appear to be
16:42.01scrnope, it's my code, I know it's my bug...
16:42.18bkw_scr I have an app_directory thats odbc aware but its not pretty
16:42.32bkw_I also have voicemail routines that are odbc aware
16:42.43bkw_just havne't had time to tweak with it
16:43.12bkw_wooo deadlock
16:44.13jsmithbkw_: Wow... more dreadlocks?  I always wanted dreads...
16:44.19Filaceheh
16:44.23UnixDawgshould pq not be pg
16:44.51bkw_jsmith No I created this on purpose and it worked
16:44.51bkw_haha
16:44.55bkw_i'm trying to trap a bug
16:45.23jsmithbkw_: Oh...
16:46.01JackyDoes any one use VoiceTronix Openswitch 12 for FXO/FXS card?
16:46.02bkw_this is a very strange one
16:46.58vaewynHmm... ok.. so now I need to figure out why my Zap FXS card spent 8 hours yesterday ringing the line after I had hung up
16:47.23vaewynwondered why the cordlesses batteries were dead when I had just charged it :P
16:48.51zoabkw, how can i retrieve the accountcode from within an AGI ?
16:49.14zoaor from in  a dialplan ?
16:50.30UnixDawgwich lib pg++ or XX
16:50.30UnixDawgis needed
16:50.30UnixDawgi need pgsql suport
16:50.31citatszoa: accountcode is passed to AGIs as the * vars when the AGI starts
16:50.31zoaand how can i retrieve it ?
16:50.31zoain what variable can i find it ?
16:50.31citatsand from the dialplan as ${ACCOUNTCODE}
16:50.31citatszoa: agi_accountcode
16:50.31zoathnx a lot
16:50.35*** join/#asterisk Havik (~Havik@hoochie.digium.com)
16:50.35HavikHey guys, just getting ready to start on my first asterisk system.... was wondering if anyone had any recommendations for phones?
16:50.49crontibs7960's
16:50.54crontibshavik
16:51.00zoa7960's or snoms
16:51.05bug847Panasonic 1 line corded..  $19.95 @ Office Max
16:51.11bug847works wonderful
16:51.22bug847we also use a mix of Uniden and Panasonic cordless phones
16:51.39Haviksorry, good VoIP phones is what i meant   :-)
16:51.42UnixDawgbkw
16:51.49bug847crisco or snom
16:51.50Havikfor remote extensions
16:51.51Alric7960!
16:52.04bug847if you're experimenting, just get a couple grandstreams
16:52.11drgalaxyboo snom.com resized my browswer window
16:52.14drgalaxyerr snom.de
16:52.15bug847but most people will tell you that GS phones suck for production use
16:52.36bug8477940 has 2 line appearance?
16:52.42drgalaxyare the grandstream ATAs worth anything?
16:52.53bug847drgalaxy: dunno... wait for the Eaxy though
16:53.02bug847(if you trust Digium hardware that is)
16:53.03drgalaxywhat is Eaxy?
16:53.03bug847=D
16:53.09bug847iaxy... sorry
16:53.09zoaIAXy
16:53.14rob--drgalaxy I'm using one and it works fine.
16:53.15zoadigium's ata
16:53.21drgalaxyahh excellent
16:53.46drgalaxyhow long until we can get them?
16:53.48rob--drgalaxy: If you want to phone me to here the quality my fwd is 212811 and my iaxtel is 17006602034
16:54.04UnixDawggrr this pisses me off
16:54.21drgalaxyrob--: thanks, I trust it will be a good unit.  (also, don't have iaxtel setup)
16:54.34*** join/#asterisk jorgeraidel (~jorgeraid@ip-69-33-73-90.mia.megapath.net)
16:54.37drgalaxyfor some reason my skinny phone doesn't transmit the audio I say into it to my office's * box
16:54.39jorgeraidelhello
16:55.02drgalaxyHavik: a good place to look for info about phones and such is www.voip-info.org
16:55.05jorgeraidelhey my asterisk it have one error when I make ectensions reload
16:55.16jorgeraidelJan 16 00:54:22 WARNING[1226400960]: pbx.c:4734 ast_context_verify_includes: Context 'asterisk' tries includes non-existant context 'iaxtel'
16:55.16jorgeraidelvoiceip*CLI>
16:55.39zoa[iaxtel] doesnt exist in extensions.conf
16:56.36*** join/#asterisk Lemmiwinks (~sean@adsl-065-083-169-002.sip.mco.bellsouth.net)
16:56.38rob--drgalaxy I was talking about the gs ata, not the iaxy
16:57.03drgalaxyrob--: ooooh
16:57.20crontibsyoyo
16:57.23drgalaxyrob--: either way, I can't speak and be heard currently.. going to work on it later
16:57.24crontibswhats going with freebsd
16:57.27crontibs847 thing
16:57.57jorgeraidelsomebody can helpme
16:58.09*** join/#asterisk seemore (~craig@d205-206-9-104.abhsia.telus.net)
16:58.20*** join/#asterisk Starbuck (~Starbuck@hoochie.digium.com)
16:58.33rob--drgalaxy: It's not too bad, although it hasn't got many features.  
16:58.55StarbuckDoes anyone know an easy way to make GSM audio files that I can use for the phone menu system?
16:59.05UnixDawgman this is a pain
16:59.24rob--drgalaxy: sometimes when I answer a call I get a loud beep, and have to hangup the phone and pick it up again to answer the call.
16:59.34*** join/#asterisk Stuart (~Stuart@smtp.dstoys.com)
16:59.59swirlnetsdoes anyone know of a cheap headset for a cisco 7960?
17:00.13jimmyzwe get ours for $50
17:00.19StarbuckMy company is selling headsets for the 7960.
17:00.26swirlnetsstarbuck: URL?
17:00.32StarbuckAbout 30 used.
17:00.42swirlnetsstarbuck: I don't want used
17:00.52swirlnetsi can get used for 20 bucks
17:01.08rob--Kram: what is the best way to stream audio from memory in an app?  
17:01.24jimmyzswirlnets: http://www.acousticalinnovations.com $50 new
17:01.37jimmyzpart number AT-904-6400
17:02.20swirlnetsjimmyz: do you have a direct link?
17:02.41jimmyzno i e-mail our order in
17:03.17jimmyzlooks just like this one http://www.acousticalinnovations.com/LX-900.htm
17:03.33*** join/#asterisk Sobek (~btatton@209.180.83.6)
17:03.36swirlnetsjimmyz: thanks
17:03.39jimmyznp
17:04.01jorgeraidelalguin me puede ayudar?
17:04.43zoawhat is the price of an LX-900 ?
17:04.52zoaand how easily can you break them ?
17:05.01zoaare they the same quality as the GN-Netcoms ?
17:05.19jorgeraidelbut this lx-900 with what kind of voice works?
17:05.31*** join/#asterisk jsmith (~jsmith@209.180.83.18)
17:05.33jimmyzvoice works? it's analog
17:05.46*** join/#asterisk rumba (rumba@cpe-24-160-4-31.sw.rr.com)
17:05.51jorgeraidelbut how I conected?
17:06.02jimmyzthe headset jack on the 7960
17:06.10jorgeraideloh ok
17:06.16*** join/#asterisk benngard (~mabe@81.26.235.3)
17:06.19jorgeraidelI don't have the 7960
17:06.30jorgeraidelonly works in this voip?
17:06.44PowerkillI receice this NOTICE in asterisk
17:06.44PowerkillJan 15 18:05:06 NOTICE[1217755952]: rtp.c:418 ast_rtp_read: Unknown RTP codec 19 received
17:06.58Powerkillkram can you add it i think it's g726
17:07.15Powerkillit's from grandstream phone that have it :)
17:08.05jimmyzJorgeraidel: don't know...it's what i use on our 7940 and 7960
17:12.02LemmiwinksHey guys, anyone have any good recommendations for VoIP phones (remote extensions) for the asterisk system?
17:13.24RoyKLemmiwinks: snom is nice, but quite expensive
17:17.58swirlnetslemmiwinks: the best options are Cisco 7960
17:18.15*** join/#asterisk cypromis (~michael@80.51.246.186)
17:22.58ManxPowerPowerkill, check the archives for the reason and the fix
17:24.10*** join/#asterisk rajo_home (~rainer@p508AE16C.dip.t-dialin.net)
17:24.13*** join/#asterisk clh (~me@216.253.86.210)
17:24.40clhis the Cisco ATA configurable via browser?
17:24.45clhanyone..
17:24.55ManxPowerclh, Yes, see my sample config files.
17:25.17clhthey reachable through links page on asterisk.org?
17:25.24ManxPowerGo to http://www.digium.com/index.php?menu=documentation and look at the "Unofficial Links" section.  This section has links to a wide variety of 3rd party Asterisk related pages.  My page is the "Asterisk Resource Pages".
17:25.42*** join/#asterisk dnc (~duncan@213.244.224.118)
17:26.11clhk, thanks!  I assume you can just plug an el cheapo 900 MHz phone to it and have a "wireless VoIP phone" within the range of the phone base station....is this true?
17:26.46ScaredyCatbkw_:
17:27.31jorgeraideltext 'asterisk' tries includes non-existant context 'shortcut'
17:27.32jorgeraidelJan 16 01:26:34 WARNING[1226400960]: pbx.c:4734 ast_context_verify_includes: Context 'asterisk' tries includes non-existant context 'iaxtel'
17:27.45jorgeraidelI have this eror when I make extensions reload
17:28.06*** join/#asterisk outtolunc (~me@h-66-166-19-148.SNVACAID.covad.net)
17:28.51ScaredyCatnm
17:29.06clhanyone used cordless phone with Cisco ATA?
17:29.06bkw_zoa told you once
17:29.10outtoluncmorn'n
17:29.13bkw_iaxtel doesn't exist
17:29.16bkw_the [iaxtel] context that is
17:29.54zoabkw, he's not listening i think :)
17:29.58jorgeraidelbut is my asterisk?
17:30.01zoayes
17:30.10jorgeraidelI installed the 0.7.0 yesterday
17:30.13zoacheck extensions.conf
17:30.15jorgeraideloh
17:30.17zoalook for include
17:30.18jorgeraidelahd them
17:30.26zoacheck if there is an include => iaxtel
17:30.29zoaif there is,
17:30.33zoacheck if you also have an
17:30.36zoa[iaxtel]
17:30.41zoaif you don't have an [iaxtel]
17:30.41jorgeraidelok
17:30.46zoacomment the include line
17:30.51jorgeraidelok
17:30.58tclarkclh: sure no problems
17:31.01zoaNEXT!
17:31.38zoakram: that TE410p did work after all
17:31.47UnixDawggrrr
17:31.48zoadunno why it didnt work at first
17:31.49clhtclark: tx
17:31.55UnixDawgthis pisses me off
17:32.05zoakram: i noticed in the changelog that there are now releasecodes for the pri's
17:32.12zoawhere are those being logged ?
17:32.17zoaCDR ?
17:32.34krami don't rembmer
17:32.45zoa:)
17:33.21kramhey guess what!
17:33.59zoaMOOSE PENIS ?
17:33.59kramiconnecthere wrote me
17:34.10kramthey want someone to help come up with a guide for them for how people can connect with asterisk
17:34.10malcolmdkram: and?
17:34.13JerJerahh sometimsconnecthere
17:34.13bkw_kram what did icantconnecthear say?
17:34.20malcolmdheh
17:34.24zoalol
17:34.40kramalso i'm looking for the origination of the term "barbietone"
17:34.51zoathats jerjer or bkw
17:34.51zoai'm sure
17:34.56zoai think i have it logged
17:34.57zoa:)
17:36.11*** join/#asterisk heison (~heison@dyn-65-97.ham.dial.tht.net)
17:37.31zoa<Blaisen1> but when i call extension to extension on my grandstream barbietones i get no compatible cdoecs
17:37.31zoa<Blaisen1> i don't have anything disallowed in sip.conf so i don't understand
17:37.31zoa<atacomm> HoopyCat: not at all huh? driver issues getting to the Zaptel cards in a virtual machine?
17:37.31zoa<benjk> Blaisen: make sure that you specify a common codec in your sip.conf entries for those phones and make sure they are codecs that the GS supports
17:37.31zoa<benjk> for example:
17:37.32zoa<blitzrage> hmmm.. for some reason I can't seem to get my tdm400p to work in this Celeron 633.  I know i
17:37.42zoais the first thing i can find for barbietone in my logs
17:37.51JerJerwhat's the date of that ?
17:38.02zoadidnt have timestamps :(
17:38.10zoabut someone else that is logging might have it
17:38.13*** join/#asterisk heison (~heison@dyn-65-97.ham.dial.tht.net)
17:38.25JerJeri remember going to shido's place and seeing one for the first time and sayng, "that looks like a toy, man"
17:38.33zoa:)
17:38.57zoai have at least one month worth of logs before that statement
17:38.57UnixDawgok this is a pain I need more fbsd people to work on this
17:39.06zoabut i'm not in the chan 24/24 7/7
17:39.10UnixDawgto get postgres usable
17:39.14zoabut i think that was the first one
17:39.30*** join/#asterisk heison (~heison@dyn-65-97.ham.dial.tht.net)
17:40.01Shido6_i think jj started the barbietone name
17:40.11*** join/#asterisk russT (~rusty@65-101-255-24.dnvr.qwest.net)
17:41.05ManxPowerBlaisen's problem is that he does not do a disallow=all and an allow=compatablecodec in his sip.conf
17:41.23ManxPowerJerJer, The skirt does not look good on you.
17:41.37JerJerdamnit that's a kilt
17:41.42JerJerheheh
17:41.44citatszoa: i have the same logs in mine and it looks like that Blaisen1 message was about oct 11th or so
17:42.19ManxPowerJerJer, not if you are giggleing like a school girl, it isn't. 8-)
17:42.30JerJerlol
17:43.21bkw_damn it to hell chan_zap will be my bitch today
17:44.08jetshaha
17:45.58LuchoARI have installed an E100P (Euro ISDN PRI), which modules should I load before staring asterisk
17:47.38citatsLuchoAR: wct1xxp
17:48.18reseauxhi luchoAR!!
17:48.20LuchoARhi!
17:48.26LuchoARJan 15 14:41:48 WARNING[1074404064]: chan_zap.c:659 zt_open: Unable to specify channel 1: No such device or address
17:48.34LuchoARJan 15 14:41:48 ERROR[1074404064]: chan_zap.c:5189 mkintf: Unable to open channel 1: No such device or address
17:48.39*** join/#asterisk hclai (~hclai@hoochie.digium.com)
17:48.39LuchoARhere = 0, tmp->channel = 1, channel = 1
17:48.44LuchoARJan 15 14:41:48 ERROR[1074404064]: chan_zap.c:7162 setup_zap: Unable to register channel '1-3'
17:48.51LuchoARthat's what I am getting
17:49.15*** join/#asterisk camoa (~camoa@200.71.58.37)
17:49.16reseauxLuchoAR: have you configure the /etc/zaptel.cfg
17:49.28LuchoARchannels]
17:49.28outtolunc.conf
17:49.28LuchoARcontext=default
17:49.28LuchoARswitchtype=euroisdn
17:49.28LuchoARpridialplan=national
17:49.28LuchoARsignalling=pri_cpe
17:49.29LuchoARusecallerid=yes
17:49.31LuchoARhidecallerid=no
17:49.33LuchoARcallwaiting=yes
17:49.35LuchoARusecallingpres=yes
17:49.35camoahello
17:49.37LuchoARcallwaitingcallerid=yes
17:49.39LuchoARthreewaycalling=yes
17:49.41LuchoARtransfer=yes
17:49.43LuchoARcancallforward=yes
17:49.43camoaone question is the last CVS stable?
17:49.45LuchoARcallreturn=yes
17:49.47LuchoARechocancel=yes
17:49.49LuchoARechocancelwhenbridged=yes
17:49.51LuchoARrxgain=0.0
17:49.53LuchoARtxgain=0.0
17:49.55LuchoARcallgroup=1
17:49.57LuchoARpickupgroup=1
17:49.59LuchoARimmediate=no
17:50.01LuchoARcontext = default
17:50.03LuchoARgroup = 1
17:50.05LuchoARchannel => 1-3
17:50.07LuchoARthat's my zapata.conf
17:50.12camoakram
17:50.22hclaihi, LuchoAR
17:50.26UnixDawgcvs is never stable
17:50.35LuchoARhi!
17:50.38hclaido u have a asterisk with u now?
17:50.40camoaso, where can i get the last stable version?
17:50.44LuchoARyes
17:50.46hclaican u try to call my IAX?
17:50.52camoaso, where can i get the last stable version? please
17:50.54UnixDawgget 7.1 tarball
17:50.56LuchoARbut I don't have it up
17:50.59hclaimy IP =220.255.46.213
17:51.04reseauxhave you define D channel
17:51.05camoaunixdawg, where?
17:51.12UnixDawgfrom the website
17:51.20LuchoARmmmm.. nope..
17:51.20reseauxLuchoAR: span=1,1,0,ccs,hdb3,crc4,yellow
17:51.30LuchoARthat should go in zapata.conf ?
17:51.31jetshclai: guest ?
17:51.32reseauxLuchoAR: bchan=1-15,17-31
17:51.32reseauxdchan=16
17:51.54hclaiwho has a asterisk up now???? can u  guys try to call my asterisk using IAX?
17:51.59reseauxLuchoAR: no zaptel.cfg
17:52.02jetshclai: i do, and what user, guest?
17:52.17hclaileaptron
17:52.35reseauxhclai: if i want i try!
17:52.38hclaileaptron@220.255.46.213
17:53.00hclaijust want to make sure it works
17:53.12hclaianyone can let me try theirs?
17:53.16jets<PROTECTED>
17:53.16jets<PROTECTED>
17:53.16jets<PROTECTED>
17:53.37jetsnothing.
17:53.46jetstry the guest@misery.digium.com or whatever
17:54.04hclaican u try to ping me using that address?
17:54.51camoasorry for asking, and the most stable zaptel would be???
17:55.02*** join/#asterisk AtoB (~shahinkha@62.3.220.66)
17:55.07*** join/#asterisk _jackhamr (~jackhamr@64.212.11.53)
17:55.08camoathe one from the FTP or the one from CVS
17:55.08jets64 bytes from 220.255.46.213: icmp_seq=1 ttl=49 time=536.7 ms
17:55.17hclaiok
17:55.18jetsdid you have any logs from me trying to use your IAX?
17:55.30hclainoo
17:55.37hclaino response in the log
17:55.46jetsnothing is in the logs?
17:55.53jetsis your machine behind nat?
17:56.00reseauxTO ALL: Someone have experience in free PRI messages "The caller is not rechable"?thz
17:56.01hclaiwhat should i config if i want to call the other IAX?
17:56.04hclaiyes
17:56.13jetsjust dial 500 on a phone or soft phone
17:56.18jetsif you still have the demo context
17:56.22hclaii had forward the port to this pc
17:56.30hclaiyes
17:56.42hclaiit did dial to the digium IAX
17:58.50jetshclai: i bet you need to forward in the IAX ports
17:58.58jetsotherwise other people cannot use your iax
17:59.01hclaican u try guest@220.255.46.213?
17:59.20hclaii already forward the port.but let me confirm again
18:00.00jetsya it's not getthing through to you
18:00.14hclaithe IAX using port 5036 ya?
18:00.24FuzzyCatby default
18:00.27reseauxTO ALL: Someone have experience in free PRI messages "The caller is not rechable"?thz
18:00.32hclaiis it TCP or UDP?
18:00.53Bonbonanyone here from Ireland (UK)?
18:01.09FuzzyCatooo... dangerous words Bonbon
18:01.22BonbonFuzzy?
18:01.25rob--hclai: udp
18:01.27jetshttp://www.voip-info.org/wiki-Asterisk+firewall+rules
18:01.36jetsthe above is a great url for letting in iax, sip, etc
18:01.40hclaijets, can i call ur IAX?
18:01.45FuzzyCatIreland is not part of the UK ... they would prolly tell you in less words...
18:01.59BonbonFuzzy: ok, take your point.
18:02.01Bonbonha ha
18:02.02jetshclai: sure, i'm not sure if i have a guest account
18:02.06jetstry guest@12.174.38.5
18:02.09FuzzyCat:)
18:02.33*** join/#asterisk alex4152 (~alex4152@hoochie.digium.com)
18:03.28jetshclai: if you call it, dial extension 6005
18:04.16h3xoops
18:04.32h3xi was using quickbooks, accidently typed my account number in the $ amount and almost sent nevada power 1.6 million dollars.
18:04.45h3xthat would have sucked.
18:04.51ManxPowerh3x, you have  1.6 million dollars in your account
18:04.58ManxPower?
18:04.59citatsheh so now we know the first 2 numbers in your account number are 16
18:05.01h3xno
18:05.02citats:)
18:05.06h3xi guess then it wouldnt have worked huh
18:05.08hclaijets, i can get through
18:05.12h3xit would be cool to bounce a check for a million though
18:05.14hclaican i speak to u?
18:05.14jetsokay dial extension 6005
18:05.45rob--hclai: you can also try guest@haylott.plus.com if you like.
18:06.02jetshclai: call my iax and dial 6005
18:07.00ManxPoweryou can also try Dial(IAX2/guest@ext-1.fnords.org/2101) if you want to test IAX2
18:07.42*** join/#asterisk hclai (~hclai@hoochie.digium.com)
18:07.48hclaijets
18:08.09hclaiwhats the extension i should dial?
18:08.31hclaii would like to try and listen to the quality of the voice
18:08.54jorgeraidelzoa?
18:08.58hclaii already did the port forwarding on the router side
18:09.02rob--hclai: he said 6005.
18:09.07hclaiwhy u still cannot call in?
18:09.09jorgeraidelI have the same problem and i have include iaxtel
18:09.12reseauxTO ALL: Someone have experience in free PRI messages signalling like this "The caller is not rechable"?thz
18:09.16hclaiok
18:09.17rob--hclai: you can also try guest@haylott.plus.com if you like.
18:09.19hclaithanks
18:09.24jorgeraideltext 'asterisk' tries includes non-existant context 'asterisk-usa'
18:09.24jorgeraidelJan 16 02:10:11 WARNING[1226400960]: pbx.c:4734 ast_context_verify_includes: Context 'asterisk' tries includes non-existant context 'sipphone'
18:09.24jorgeraidelJan 16 02:10:11 WARNING[1226400960]: pbx.c:4734 ast_context_verify_includes: Context 'asterisk' tries includes non-existant context 'shortcut'
18:10.12JerJerisn't it obvious?
18:10.28extremisheh, any digium sales monkeys around?
18:10.40ManxPowerjorgeraidel, ANY time you see that message that means there's an include => thatcontext with not [thatcontext] line in extensions.conf.  If you get this with the default extensions.conf then report it as a bug to bugs.digium.com
18:11.13jetshclai: that is jesse you are talking to
18:11.16jetsmy coworker
18:12.31extremisI love getting voicemail when calling digium :)
18:12.45hclaii see
18:12.56jetsthank you hclai
18:13.02jorgeraidelok
18:13.12hclaibut i still cant let u to call in
18:13.44hclaii got to make this thing work. i tried that day and it waas working
18:13.54hclaii am not sure what is the problem now
18:14.36jetshclai: did you look at the URL i gave you?
18:14.42hclaiyes
18:14.59rob--hclai: what is your ip address?
18:16.02*** join/#asterisk erubright (~erubright@216.229.189.150)
18:16.09hclaiguest@220.255.46.213
18:16.15hclaican u try to call in
18:16.37hclaii saw my log file and there is someone call to my asterisk
18:16.40hclaidial 104
18:16.44rob--I can connect with iax1, but not iax2
18:16.45hclaiextension
18:16.52hclaiya...only iax
18:16.56jetsoh sorry
18:16.58jetsi was trying iax2
18:17.00hclaican u dial wxtension 104?
18:17.12hclaii would like to talk to u
18:17.18*** part/#asterisk _jackhamr (~jackhamr@64.212.11.53)
18:17.19hclaithanks for ur help
18:18.04*** join/#asterisk token (~saa@host2.216.41.24.conversent.net)
18:18.15tokenbkw you there bud
18:18.35hclaithanks guys
18:18.51hclaiwhats the differnce of IAX and IAX2?
18:19.36rob--no prob.
18:19.36jetshclai what is your extension?
18:19.45jjanzeri was pretty impressed with the quality of that call hclai, sounded really clear
18:20.03rob--I think iax2 uses udp and is better.
18:20.08hclai104
18:20.16bkw_most voip uses udp unless its CRACK
18:20.23bkw_well let me say all voip uses udp
18:20.28bkw_NEXT!!!
18:20.38jetshclai: now get an iaxtel.com account, my iaxtel number 1 700 226 4222
18:20.46ManxPowerIAX and IAX2 both use UDP
18:20.51hclaii see
18:20.56hclaicool
18:20.57*** part/#asterisk glyph (glyph@h00095b4e65ab.ne.client2.attbi.com)
18:21.01rob--Ok, so what's better about iax2?
18:21.09ManxPowerrob--, trunking
18:21.20hclailet me check my iaxtel account
18:22.39jorgeraidelwhat is asp % and udp ?
18:22.44jsmithrob--: And, IAX1 is going away :-)
18:23.06jsmithrob--: Mark wrote IAX2 because he didn't like certain things about IAX1...
18:23.29*** join/#asterisk Junbug (~junya@adsl-158-118-24.mia.bellsouth.net)
18:23.38ManxPowerjorgeraidel, I don't know what "asp" is, but % is Percent and UDP is Unicast Datagram Protocol
18:23.57*** join/#asterisk kapejod (~kapejod@p509241B8.dip0.t-ipconnect.de)
18:24.03kapejodhi there
18:24.17txhi kapejod.
18:25.06hclaijets, i think i will sign up the account later
18:25.14hclaii face another problem here
18:25.22hclaii cant have music on hold
18:25.35jetshclai: do you have mpg123 installed?
18:25.48hclaithe asterisk is playing the musiconhold"default"
18:25.56camoahello, i've just compiled zapter 0.8.0 but now i get tons of unresolved symbols with depmod, i got a pretty short kernel an modules, so i don't know what i'm missing, anyone may help me? please
18:25.59jorgeraidelI can't enter in downlaod asterisk for 0.7.1
18:26.02hclaibut i cant hear anything from the phone
18:26.23jetshclai: make sure you have the program mpg123 installed
18:26.23hclaihow to check that?
18:26.29jetsat a shell
18:26.31jetstype mpg123
18:26.33jetsor locate mpg123
18:27.04*** join/#asterisk simprix (~simprix@ip-64-32-242-201.dsl.iad.megapath.net)
18:27.26hclaiafter that?
18:27.35jetsdoes it run
18:27.42hclaii typed "locate mpg123"
18:27.47hclainothing happen
18:28.26*** join/#asterisk DL4GRC (~DL4GRC@pD9ECD87A.dip.t-dialin.net)
18:28.28jetstype just mpg123
18:28.33hclaii just downloaded the mpg123-0.59q.tar.gz
18:28.40camoaand the symbols are those like vprintf and stuff
18:28.49hclaicommand not found
18:30.23camoathe mpg123 executable should be located at /usr/bin or /usr/local/bin not remember exactly, but that makes * fail on MOH
18:30.31hclaithe binary program of mpg123 is in the directory but it said command not found
18:32.46*** join/#asterisk lorena26 (~lorena@78.Red-80-37-200.pooles.rima-tde.net)
18:32.50lorena26hi
18:33.31reseauxhi
18:34.47FuzzyCathclai: you need to do an 'updatedb' if you just installed it
18:35.06FuzzyCatthen u can do a locate
18:35.39hclaihow to locate?
18:35.54citatsif its in your path type: which mpg123
18:37.09hclaithe directory i install the mpg123?
18:37.26FuzzyCat<FuzzyCat> hclai: you need to do an 'updatedb' if you just installed it
18:37.26FuzzyCat<FuzzyCat> then u can do a locate
18:38.38hclaiyes
18:38.41lorena26work now the Cvs ?
18:38.41*** join/#asterisk slacker (~dhollis@washdc3-ar10-4-63-121-042.washdc3.dsl-verizon.net)
18:38.41hclaii done it
18:38.44hclaithank u
18:38.47FuzzyCatnp
18:39.35reseauxlorena:i dont know
18:40.04reseauxlorena:download the 0.7.1
18:42.24lorena26yes.. thats i Have
18:42.25*** join/#asterisk dannie (~dannie@s1-09.colo.iglou.com)
18:42.41lorena26but .. I have a E100P 1 E1 and take audio in Only One way..
18:43.03vaewynanything bigtime wrong with current cvs? I want to check if it will fix my hanging zap channel
18:44.25*** join/#asterisk mbranca (~matteo@ppp-217-133-173-173.cust-adsl.tiscali.it)
18:45.46mbrancahi all
18:48.37*** join/#asterisk simprix (~simprix@ip-64-32-242-201.dsl.iad.megapath.net)
18:48.43*** join/#asterisk ok_ (~ok@DSL01.212.114.231.34.NEFkom.net)
18:49.20jtoddI'm behind the curve here - is there a g.726 codec available for integration into * now?  I see it in the translation list, but nothing in the cvs trees that seems to reference it...
18:50.32vaewynjtodd: I beleive that is the one you have to purchase from digium for licensing reasons
18:50.45bkw_its coming
18:50.47vaewynnot sure though... that might be 729
18:50.55bkw_its not even written yet I don't think
18:50.55jtoddI was under the impression that the g.726 stuff was "open source" - yeah, what bkw_ said.
18:51.03jtoddOK.
18:51.10bkw_its coming soon from what I hear
18:51.16jtoddI'll wait patiently.  I don't have any g.726 devices yet anyway, I think.
18:51.27PBXtech726 like 729?
18:51.52mbrancanope. you can buy only g.729
18:52.06mbrancag.726 isn't anywhere till now...
18:52.22citatsisnt 726 ADPCM?
18:52.47reseauxlorena:is not a problem of asterisk i think you use a sip phone?
18:52.50bkw_nope
18:52.52reseauxhi mbranca!!!
18:52.53bkw_its like it
18:52.55bkw_but diffrent
18:53.04*** join/#asterisk Tili (~Tili@202.133.65.187)
18:53.25puzzledwhat's with normal calls getting detected as fax calls in today's cvs?
18:53.35citatsbkw_: actually it is ADPCM, i just looked it up... capable of 40, 32, 24, and 16 Kbps
18:53.47FuzzyCatUSE_OLD_CVS_VERSION :P
18:53.55vaewynpuzzled: is broken DSP... if you uncomment the "OLD_DSP" ish line it fixes it
18:54.00bkw_citats no the header stuff is diffrent from what I was told
18:54.14citatsbkw_: different than what?
18:54.16*** join/#asterisk Om3gAnGeL (~Om3gAnGeL@vwmcches.uncg.edu)
18:54.25vaewynpuzzled: OLD_DSP_ROUTINES in dsp.c
18:54.30puzzledvaewyn: thanks but I don't understand why things get broken after a stable brnach release. It used to work fine afaik
18:54.45bkw_you can't use ADPCM as g726
18:54.47bkw_it just wont work
18:54.53vaewynpuzzled: they fixed another bug and created this one
18:54.59bkw_we have tried :P
18:55.04*** part/#asterisk Om3gAnGeL (~Om3gAnGeL@vwmcches.uncg.edu)
18:55.20citatsbkw_: thats not what i'm saying... i'm saying that g726 is ADPCM transcoding technique
18:55.59bkw_ah ok
18:56.01bkw_yes that is true
18:56.07bkw_but we can only use g726-32
18:56.15bkw_the rest are still under patent
18:56.18vaewynpuzzled: see  http://bugs.digium.com/bug_view_page.php?bug_id=0000696   and http://bugs.digium.com/bug_view_page.php?bug_id=0000649 for more information
18:56.20bkw_g726-32 is accually g721
18:56.27puzzledvaewyn: thanks
18:56.32vaewynnp :}
18:56.37bkw_which is really old but compatible with g726-32 (Because they are the same damn thing)
18:56.46Seba2hello
18:57.10PBXtechis there an autoconnect function? so if a handset is picked up it automatically calls security?
18:57.36citatsPBXtech: immediate = yes
18:57.40Seba2somebody knows wich variable can tell me information about calltime and duration of a call in AGI?
18:57.57vaewynPBXtech... easy... just make your s extension dial them :}  drop them into that context and voila! :}
18:58.35vaewynOk... who wants to donate me a quad span T1 card?  ;P
18:58.47PBXtechso it wont want for any digits to be input? pickup and it rings.. right?
18:59.03vaewynPBXtech: pretty much :P
18:59.30PBXtechso how would the dial string look?
19:00.04vaewynPBXtech: like   exten => s,1,Dial,Zap/1      or such...
19:00.20vaewynmake the default context for the line include that exten and you are set
19:00.24PBXtechoh the s thinger.. gotcha thanks
19:00.26puzzledvaewyn: will I get the OLD_DSP_ROUTINES back also when I enable it in the Makefile or do I need to do it in dsp.c?
19:00.59vaewynpuzzled: Up to you.. if you define that in the makefile it will do it... if you uncomment the define in dsp.c it works also... up to you :P
19:01.25puzzledvaewyn: ok, thanks
19:01.29vaewynnp
19:03.48tclarkvaewyn: current code does not have flag in dsp.c any longer:)
19:04.00vaewyntclark: ahh :P
19:04.20vaewynam just updating now... stupid fxs card keeps hanging in ring state sol..
19:07.05*** join/#asterisk zoa (~john@kn-upc-1.kuleuven.net)
19:07.42*** join/#asterisk Moc (~Moc@modemcable213.103-70-69.mc.videotron.ca)
19:09.05bkw_Moc
19:09.09bkw_we have a patch for you to test
19:09.33bkw_who here things asterisk should continue on if it can't load chan_zap or other modules?
19:09.46zoadepends bkw
19:09.50zoaon a start ?
19:09.52bkw_(right now it segfaults on misconfigured stuff)
19:09.54zoaon a startup ?
19:09.55vaewynshould yell no matter what
19:10.04vaewynyell loudly
19:10.07zoabkw, i think it should work if G729 license starts fucking up
19:10.13bkw_haha
19:10.14zoaas happened on my server this week
19:10.26zoasuddenly safe_asterisk coredumps all the time
19:10.34zoaand restart
19:10.34FuzzyCatouch!
19:10.34zoaand restart
19:10.34FuzzyCatJan 15 20:09:06 WARNING[180245]: chan_zap.c:1423 zt_call: Unable to ring phone:
19:10.34FuzzyCatDevice or resource busy
19:10.34zoaand restart
19:10.40bkw_ok right now if you have a mismatch in your signalling it will segfault
19:10.41zoaturns out my g729 license is gone again
19:10.54zoabkw, that should not happen !
19:10.55bkw_ok anthm
19:10.58bkw_lets do this
19:10.58vaewynwb kram
19:11.03kramthanks but it's only for a sec
19:11.06krambut i'll be back sooon
19:11.07vaewyn:P
19:11.19zoabkw, a question i have had several times since 0.7.0
19:11.34zoacould people use asterisk without compiling zaptel before 0.7.0 ?
19:11.39bkw_yes
19:11.47zoabecause now it doesnt work
19:11.57denonzoa: thats a feature :)
19:12.07bkw_here in a few chan_zap will no longer segfault
19:12.14bkw_when its totally screwed
19:12.18bkw_it will just stop and say BLAH
19:12.21bkw_like it should
19:12.44denonZaptel: URGENT: I have wet my pants. Please reboot.
19:13.42vaewynhehehe I like that
19:15.22bkw_that seems to have fixed it
19:16.16vaewyneven cooler would be config definable actions when things fail... then could have it reboot itself... or page you when things go awry :P
19:16.23vaewynbut that is a daydream :P
19:16.43vaewynliving in the past I tell you :P
19:16.58FuzzyCatwell the present isn;t working
19:17.42vaewynHeh... FuzzyCat is playing "butterfly effect" with his * install
19:18.03vaewynonly more serious... and without the moronic banter
19:18.08vaewynI hope at lest
19:18.12vaewynleast even
19:19.40lorena26zoa have u problem with zaptel in the last 0.7.1?
19:19.56zoanopez i was wrong
19:20.00zoai got a flood of messages
19:20.05zoabut when i retried all went fine
19:20.19lorena26u have card for E1 or T1?
19:20.21zoaprobably a little synching problem when i unloaded en reloaded the modules that fast
19:20.23zoaE1
19:20.30lorena26ok, I the same
19:20.37lorena26take u only way AUDIO ?
19:20.45zoanopez, i prefer 2 way
19:20.46zoa:-p
19:20.50lorena26jajajaja
19:20.57zoahey, where are you from ?
19:21.05zoaare you dutch ?
19:21.05lorena26sorry.. U undertand my question ?
19:21.05zoagerman?
19:21.07lorena26Spain
19:21.14zoayes,
19:21.17zoai don't have that problem
19:21.25vaewynOk... if I am ringing a remote context on another server do I need anything more than:
19:21.25vaewyn[bell]
19:21.26vaewynexten => s,1,Dial,Zap/1  
19:21.26vaewyn?
19:21.37vaewyn(That is the context I am calling
19:21.39vaewyn)
19:21.41lorena26can u give me a test config ?
19:21.45bkw_Moc go test bug 851 now
19:21.46lorena26may be is my setup problem
19:21.48bkw_tell me if you like that
19:21.59zoa851 ? we are already at 851 ?
19:22.06bkw_yes
19:22.06zoawhat happened while i was eating !
19:22.13vaewynzoa++   :}
19:22.29zoabkw, how could i retrieve the caller id from an asterisk c app ?
19:22.31zoanot agi ?
19:22.37zoapbx_helper blabla ?
19:23.30FuzzyCathang zoa
19:23.43vaewynthat's kindof mean Fuzzy
19:23.45vaewyn:}
19:24.01anglermy coworker has a turbo in his prelude
19:24.09zoabut i already know my last wishes !!
19:24.13vaewynhahah nice angler
19:24.16puzzledhehe
19:24.18FuzzyCat<PROTECTED>
19:24.22anglerhe is lying
19:24.23anglerhaha
19:24.31zoaoh shit
19:24.33zoai asked caller id
19:24.36zoai want accountcode :)
19:24.37anglernother guy at my pc
19:24.38zoasorry dude
19:24.39zoa:)
19:24.42FuzzyCatast_callerid_parse (old_cid, &name, &num);      /* this destroys the original string */
19:26.24zoabkw, i'll test that puppy
19:27.44*** join/#asterisk juice (~juice@user221.net1199.mo.sprint-hsd.net)
19:28.21*** join/#asterisk Mike (~mike@dsl-200-67-4-96.prod-infinitum.com.mx)
19:28.25zoabkw, look at my remark for chan_zap 851
19:29.03bkw_zoa it doesn't
19:29.11zoayou sure ?
19:29.11bkw_you can accually unload chan_zap during a meetme
19:29.16bkw_done it many times lastnight
19:29.23bkw_all that stuff doesn't tie to the channel usage
19:29.27zoahow about during trunking ?
19:29.34bkw_same thing
19:29.48zoak, so its the kernel modules that do timing ?
19:30.00bkw_yes sir
19:30.05zoagroovy
19:30.13bkw_so chan_zap will no longer seg either on load
19:30.22bkw_grab the latest patch
19:30.29bkw_I changed some of the verbose stuff we added for testing
19:30.33bkw_I forgot to remove one line
19:30.38zoaSYSTEM WARNING: extract() expects first argument to be an array
19:30.38zoa(file_download.php: line 50)
19:30.43zoamantis is fucked
19:30.48bkw_also anthm did most of the work tracking down that problem
19:30.51bkw_reload the page
19:31.41Seba2hello
19:31.52bkw_now if chan_mgcp and chan_skinny would just unload
19:32.27zoaFuzzyCat= do you also know how to grab the accountcode ?
19:32.32zoaas i made a typo ?
19:32.36zoai didnt want the caller id
19:32.38zoabut the accountcode
19:32.57FuzzyCattry chan->accountcode :)
19:32.58bkw_muhahahahahahahhaha
19:33.20bkw_if you have the cdr struct its cdr->accountcode
19:34.03bkw_ok who wants to join the conf
19:34.11bkw_JerJer you alive?
19:34.13zoabkw, i have no headset here :)
19:34.24zoado you have speech2text ? :)
19:34.25sxpert~seen levon
19:34.35levon <~levon@mail.feature-it.de> was last seen on IRC in channel #asterisk, 16d 10h 15m 4s ago, saying: ';)'.
19:34.35bkw_Moc you testing bug 851 since you were complaining about it so much the other day?
19:34.35FuzzyCatlook at the doxygen docs for the struct ast_chan ZOP
19:34.35FuzzyCaterrm zoa
19:35.40Seba2guys
19:35.52Seba2how can I back from an AGI file?
19:36.19Seba2after processing AGI * continues processiong next in extensions.conf?
19:37.42bkw_next on my list
19:37.42*** join/#asterisk ^sly^ (~jelque@65.70.27.77)
19:37.50bkw_res_musiconhold should be able to unload now
19:37.52bkw_muhahahahhahahahaha
19:38.12Mikehows 0.7.1 doing?
19:38.26bkw_if we can unload res_musiconhold you can in theory unload the kernel modules
19:38.42*** join/#asterisk cypromis (~michael@217.11.142.161)
19:39.12bkw_then load them back
19:39.21bkw_and bring the modules back in without restarting *
19:39.31bkw_har har har
19:39.54FuzzyCatand I would want to do that ... why?
19:40.24vaewynlets you keep SIP or H.323 stuff running even though your zap is messed and such
19:40.47Seba2NOTICE[1296014640]: rtp.c:418 ast_rtp_read: Unknown RTP codec 19 received
19:40.52FuzzyCatis my zap aint working I don;t want sip working
19:40.52Seba2What's that?
19:40.53bkw_turn off VAD
19:40.57puzzledanyone seen "stop when convenient" drop a bunch of IAX channels?
19:40.58*** join/#asterisk Inv_Arp (~junya@65.2.9.250)
19:41.10cypromismaybe they where inconvenient ?
19:41.11Seba2ok
19:41.13bkw_FuzzyCat if you setup correctly you can direct calls out another provider if zap is down
19:41.16zoabkw, we really need a FAQ :(
19:41.36zoaand people get kicked for asking a question in the FAQ
19:42.08FuzzyCatbkw_: if my zap if fucked i don;t want to redirect... I want to fix zap
19:42.46zoacleopatra*CLI> unload chan_zap
19:42.46zoaUnable to unload resource chan_zap
19:42.46zoa<PROTECTED>
19:42.46zoaJan 15 20:42:30 NOTICE[18451]: chan_zap.c:3588 zt_read: Fax detected, but no fax extension
19:42.46zoacleopatra*CLI> unload chan_zap.so
19:42.47zoaUnable to unload resource chan_zap.so
19:42.49zoaJan 15 20:42:35 WARNING[17426]: loader.c:119 ast_unload_resource: Soft unload failed, 'chan_zap.so' has use count 1
19:43.03zoacool
19:43.07jetsFuzzyCat: But in in the intereium of 45 minutes i don't want my CEO mad he can't make a call
19:43.08zoayou cant unload if someone is calling :)
19:43.28bkw_nope
19:43.31bkw_it won't let you
19:43.34bkw_usecount is > 0
19:44.00bkw_zoa if you have E1 it will take like 3 seconds for it to respond to the unload
19:44.11bkw_because it has to kill the monitor threads for the D channels
19:44.16FuzzyCatjets: if you zap is fucked then you should hide under your desk to get away from the CEO
19:44.22bkw_and if that fails you have larger problems
19:44.34zoayeah
19:44.36zoait works great
19:44.43zoatried to break it
19:44.45zoadidnt work :)
19:44.54zoagot an S-frame while it was reloading
19:44.57vaewynFuzzyCat: for some installations yes... for some if the zap is down ohh well.. you have lost 10% of your call volume... woopty do :}
19:45.06zoabtw bkw
19:45.20zoait would be great if soft hangup ZAP/1-1 would work
19:45.22zoaeuh
19:45.27bkw_it does
19:45.28zoasoft hangup ZAP/1
19:45.32bkw_press tab
19:45.33zoanot /1-1
19:46.17bkw_zoa did you restart before you did the unload with the new chan_zap in place the preivous one wouldn't unload
19:46.32zoano
19:46.35FuzzyCatvaewyn: yes, but my point was more to do with the massive effort to allow the unloading and loading of modules when there are still bugs to fix
19:46.38bkw_thats why it didn't unload
19:46.44zoano no it unloaded fine
19:46.46zoait worked great
19:46.47bkw_ah ok
19:46.49bkw_thank you
19:46.50bkw_haha
19:46.56zoai did restart asterisk before testing yes
19:46.58bkw_was about to say we tested that part ALOT lastnight
19:47.04zoai did unload
19:47.05zoaand reload
19:47.08zoaand it came back up
19:47.11bkw_:)
19:47.13vaewynFuzzyCat: Ahh yes.. but for some of these bugs this lets them restart faster to fix it :P
19:47.14bkw_handy dandy eh?
19:47.24zoahad to hangup some conversations though :)
19:47.25zoassst
19:47.39bkw_haha
19:48.09bkw_now chan_mgcp and chan_skinny need to be fixed so they can unload
19:48.18zoathat would also be a great app: something to no longer accept new calls
19:48.25bkw_correct
19:48.33vaewynamen... like an /etc/nologin for * :P
19:48.36bkw_yep
19:48.46zoawithout having to reload and also kill the other channel types
19:49.08bkw_well you can unload and change signalling also and load it back
19:49.12bkw_just amke sure you run ztcfg first
19:49.13bkw_haha
19:49.24bkw_if you don't it will bitch at you and refuse to load it
19:50.09zoabkw, i think a patch to stop chan_zap from flooding when something is wrong with the signalling would be great too
19:50.13zoaif something goes fubar
19:50.22zoai sometimes start flooding your logfiles
19:50.40*** join/#asterisk Bubbag1 (~Bubbag1@12.10.10.226)
19:50.42zoaand believe me, a 16krpm disk fills very fast :(
19:50.52*** join/#asterisk Ramereth (~lance@ramereth.registered.freenode)
19:51.04jjanzer16k, that's an odd number.... usually you see around 15k
19:51.19bkw_like a no signalling errors
19:51.22bkw_and a signalling errors
19:51.37extremisso if the telco hands me a t1 with the first 6 channels allocated for voice is there a D channel (he said it wasn't a pri) for signaling?
19:51.53bkw_its a CT1 I bet
19:52.05zoa15 it is :)
19:52.09extremisyyeah, I heard him say that
19:52.15extremiswhats a ct1?
19:52.21bkw_Channelized T1
19:52.36extremisanything special in my zaptel.conf I need?
19:52.46bkw_the signalling type is em_w or em
19:53.05bkw_chan_zap is my bitch now
19:53.09bkw_MUHAHAHA
19:53.22outtoluncut oh
19:53.24extremisbkw_: so is there any drawbacks to a CT1?
19:53.29extremisare there any drawbacks rather
19:53.32bug847WOW!  $1050 bounty on the zap drivers now
19:53.45Seba2SOMEBODY CAN GIVE ME A URL ABOUT AGI WITH EXAMPLES AND GOOD EXPLANATION?
19:53.48bkw_extremis you can't sound outbound callerids
19:53.55extremissound?
19:53.56bkw_Seba2 SEARCH THE MAILING LIST AND STOP YELLING
19:54.02bkw_s/sound/set/
19:54.05bkw_fuck I can't think
19:54.12bkw_Seba2 www.voip-info.org
19:54.25Seba2tnx
19:54.43extremisahh... bkw: they are delivering me a T1 with voice on some channels and data on the other, and it hits an adtran and it breaks it out to ethernet and a T1... should I tell them to configure it a specific way?
19:54.53extremisbecause the guy I'm talking to can do it how I want it
19:54.59*** join/#asterisk zoa (~john@kn-upc-1.kuleuven.net)
19:54.59bkw_extremis asterisk can do what you want so you don't have to break it out
19:54.59extremisI just don't know how I want it :)
19:55.00zoaMantis - All Projects
19:55.00zoaAPPLICATION ERROR #200: ERROR: A required parameter to this page was not found.
19:55.02zoa(gpc_api.php: line 37)
19:55.06zoamatis is not doing so good today
19:55.14bkw_mantis is crap
19:55.17extremisbkw_: whatcha mean?
19:55.32extremisI don't want my internet traffic terminating on my pbx :)
19:55.36bkw_it can
19:55.45bkw_otherwise I have no clue how to tell youto set it up
19:55.51bkw_:(
19:56.10zoawhat do you mean extremis :
19:56.15zoai missed part of your question
19:56.21bkw_<extremis> ahh... bkw: they are delivering me a T1 with voice on some channels and data on the other, and it hits an adtran
19:56.21bkw_<PROTECTED>
19:56.36extremisI like having it broken out via ethernet
19:56.53extremisthat way my PBX isn't exposed to the internet
19:57.04bkw_why not no voip?
19:57.07extremisbut...  I do want to send caller id info on outbound calls
19:57.13bkw_you can't with CT1
19:57.42extremisso when it breaks out the data dn voice can it be built diff on the voice side?
19:57.59zoashould chan->account also contain the accountcode bkw ?
19:58.07bkw_extremis prob not
19:58.13zoai'm telling my programmer what to do
19:58.16*** join/#asterisk oej2 (~opr@apollo.webway.se)
19:58.19zoabut he doesnt know shit about asterisk
19:58.24zoaand he has no axx to the server :)
19:58.32extremisbkw_: so the best option is to terminate teh full t1 at teh T100P card and then route the data to a seperate NIC
19:58.36bkw_i'm learning this stuff so FAST
19:58.53bkw_extremis still without PRI you can't set the stuff
19:59.02extremisoh, so I need a PRI
19:59.05bkw_yes
20:00.04jetsya sending digits down a normal t1 with dtmf is kind of a pain
20:00.14bkw_if they let you do that
20:00.21bkw_MF is what you would have to send
20:00.25*** join/#asterisk brent21 (~bdf@dpvc-207-68-114-121.alt.east.verizon.net)
20:00.26jetsi suffered getting this done correctly with a cisco router
20:00.28jetsya sorry, mf
20:00.30*** join/#asterisk clh (~me@216.253.86.210)
20:01.02extremisbkw_: how does the zaptel drivers break out the data? does it create a network interface like the NICs do?
20:01.14kramgreets brent21
20:01.48*** join/#asterisk glLoadIdentity (~ghjkl@abn143-215.interaktif.net.tr)
20:02.18brent21hello, hows it going today?
20:03.16brent21Just got my t100P card from digium, great work!  reading all about my new Adit 600 right now :)
20:03.39*** join/#asterisk ecs (~ecs@hoochie.digium.com)
20:03.45Criponanyone use diax in here?
20:03.55CriponI can place calls, but I've only been able to ring it once
20:04.08Criponnow it says, "Event for non-existant even. dropping."
20:04.36Criponsubstitute session for even there
20:05.12kramcool :)
20:05.14bkw_extremis depends on how they send you the data
20:06.26vaewynmmmm channel bank... can't wait for mine to come in :P
20:07.03UnixDawgok grr this bites
20:07.05tclarkCripon: it might be that bug (out of sequence inbound iax2 event) in the iaxlcient ml that is having a hard time getting resolved in IAX2 that the iaxclient.sf.net apps use
20:07.31Cripontclark: that sucks. :(
20:08.15kramtclark: what's your issue with iax2?
20:08.27*** join/#asterisk PilotPTK-Home (~trillian@pcp02587722pcs.shlb1201.mi.comcast.net)
20:09.05brent21for the Adit 600, do I leave it on the CB to Loop Start and just use Kewlstart in * config files, or enable Loop Start with Calling Party Disconnect and still use Kewlstart?
20:09.09Criponkram: hey mark.. it's christian.. I'm having problems ringing a diax client.
20:09.10UnixDawganyone get *69 to work on asterisk
20:09.14tclarkmark steven sokol has track down an issue on the iax2 on win32 where iax2 event come in out of sequence
20:09.47*** join/#asterisk russT (~rusty@65-101-255-24.dnvr.qwest.net)
20:09.56Cripontclark: bingo.. switched to iax signalling and it works like a charm
20:10.04reseauxhi to all!!
20:10.21Mocsalut
20:10.22reseauxI have a problem with Signaling of libpri...
20:10.23Criponreseaux: hello
20:10.26PBXtechis there a design flow for *?
20:10.27tclarkyea mark even you could work with steve to solve that we be doing a major ahhpy dane
20:10.31reseauxhi Cripon
20:10.39tclarkerr HAPPY dance
20:11.13reseauxfocus in  "establishing before answer"
20:11.30reseauxsome help?
20:11.31reseaux:-)
20:11.54reseauxthis type of function is called "early B3" in CAPI
20:12.46ecsI am having issues with astman, I get an error, broken pipe
20:12.53ecsafter i log in
20:13.14hclaijets, are u there?
20:13.37reseauxsome help please :-)
20:13.53hclaianyone try to use GotoIfTime?
20:14.50hclaiis this syntax correct? 200,1,GotoIfTime(4:00-5:00,mon-fri,*,*,sleep,s,1)  ?
20:14.58hclaianyone can help?
20:15.05*** join/#asterisk ssokol (~ssokol@64-151-42-28-dhcp-kc.everestkc.net)
20:15.14jetshclai: yes
20:15.35hclaibut it didnt jumo to that context
20:15.46hclaiit just play the next priority
20:15.57jetsis it from 4am to 5am mon-fri?
20:16.02hmodesgrrrr
20:16.18hmodesit's difficult to justify telecommuting when * keeps thinking conference calls == busy signals
20:16.20PBXtechbkw, is there a design flow doc?
20:16.24hclaiyes
20:16.40*** join/#asterisk Junbug (~junya@adsl-80-16-148.mia.bellsouth.net)
20:16.56hclaiany idea why it doesn't works?
20:17.16jetsnope
20:17.18extremis<PROTECTED>
20:17.25extremisyou don't mean 5pm?
20:17.29jetsdid you check voip-info.org
20:17.32hclaidid u  use this before?
20:17.35hclaiya
20:17.37hclaii check
20:17.45jetswhat time is it
20:17.49jetsin 24 hour format
20:17.57jetsin singapore i mean
20:18.02vaewynon that machine :}
20:18.09hclaiyes
20:18.16extremisyes
20:18.35hclaiu meant the machine got to be 24 hours format?
20:18.44*** join/#asterisk mike1 (~mike@hoochie.digium.com)
20:18.45jetson your server type
20:18.46hclaibut here is morning
20:18.46jetsdate
20:18.48jetsand paste it here
20:19.26extremisso does CT1 support e&mwinkstart ?
20:19.42hclaiThu Jan 15 04:19:09 SGT 2004
20:19.57CriponJan 15 14:18:04 NOTICE[114696]: chan_iax2.c:4718 socket_read: Registration of 'kraven' rejected: Registration Refused
20:19.57Cripontoonville*CLI> /usr/sbin/safe_asterisk: line 77: 12198 Killed                  asterisk ${ASTARGS} 1>&/dev/${TTY} </dev/${TTY}
20:19.57CriponAsterisk ended with exit status 137
20:19.57CriponAsterisk exited on signal 9.
20:19.58CriponAutomatically restarting Asterisk.
20:20.00CriponDisconnected from Asterisk server
20:20.02Criponanyone?
20:20.17*** join/#asterisk lkc (~lkc@hoochie.digium.com)
20:20.25Seba2Sorry again
20:20.27mike1Can someone tell me why this does not work ==> exten => 0,1,Dial,ZAP/21
20:20.33Seba2What about EAGI?
20:20.44Seba2I don't know nothing about that
20:20.47jetsmike1: zap/2-1 ?
20:20.48Seba2What is?
20:21.21mike1zap/21 - I am assuming that the 21 is the channel. Is that correct
20:21.24hclaijets, did u see my date just now?
20:21.30zoamike: correct
20:21.34hclaiwhat should i do?
20:21.39jetsmike1: well you have spans, and channels
20:21.48zoathat should work
20:21.51mike1Asterisk complains that it can find "ZAP"
20:21.52zoajets no
20:21.57zoaits is correct like that
20:22.06*** join/#asterisk PilotPTK-Home (~trillian@pcp02587722pcs.shlb1201.mi.comcast.net)
20:22.09zoayou don't need to use the span when you dial
20:22.18zoayou prolly dont have zaptel loaded
20:22.19mike1unable to create channel type Zap
20:22.39zoado a cat /proc/zaptel/*
20:22.45zoatell us what it gives you
20:23.26mike1shows all 24 channels as FXOKS
20:23.45mike1do I need to specify ZAP somewhere?
20:23.49*** join/#asterisk c4uldr0n (~korri@ns.elitecomp.net)
20:23.51mike1is that a context
20:24.16jetsno zap is a device much like sip/
20:24.28jetsyou need a zap interface in zapata.conf
20:24.29jetsetc
20:24.43mike1A zap interface?
20:25.19jetsLike a T1 span, or an fxs or fxo card
20:25.21jetsetc
20:25.46c4uldr0nI'm very new to the Asterisk program, I'm having a problem with the zaptel installation... it fails when I do a "make config"..... error: make: *** [config] Error 127
20:26.07c4uldr0ncan somebody give me some insight?
20:26.31hmodesoh, i had tone and silence turned off
20:26.35c4uldr0nthis is with the CVS install
20:26.36hmodessilly clean tree
20:28.28lorena26Jan 15 21:28:12 NOTICE[1258761520]: rtp.c:418 ast_rtp_read: Unknown RTP codec 19 received
20:28.28lorena26how I can fix it ?
20:28.34lorena26I call with G729
20:28.35vaewynOk... doing a Dial(IAX/bridge:testing@143.207.x.y/s@bell,15)  (bell being context with a single extension to Dial the Zap interface) should it be ringing by itself... or does the dial have to have the ,r for the user to hear the ring?
20:29.31vaewyn(end phone is rining fine... just caller doesn't seem to be getting ring indication)
20:29.59*** join/#asterisk point (~litw@195.161.106.222)
20:31.38vaewynand another question... do I have to call a context? or can I call Zap/1 directly on the remote machine?
20:33.09vaewynHmm.. no.. it seems to want a context/extension  :{
20:35.42*** join/#asterisk krishna (~krishna@hoochie.digium.com)
20:36.34*** join/#asterisk Inv_Arp (~junya@adsl-80-16-148.mia.bellsouth.net)
20:38.37data[out^londondoh? 0.7.1 :)
20:39.08mike1How do I correct this ==> No channel type registered for 'ZAP'
20:39.19*** join/#asterisk groz (whocares@h24-87-70-66.vc.shawcable.net)
20:39.41c4uldr0nI had to manually install my zaptel init scripts because "make config" gives me an error... that sucks huh
20:40.05UnixDawganyone have calltrace working
20:40.22UnixDawgaka last nmbr callback
20:40.35grozgood afternoon folks, is it acceptable to ask a couple of probably dumb newbie q's here ?
20:40.51Cripongroz: sure
20:41.06Cripongroz: they may not get answered, but it doesn't hurt to ask
20:41.08grozok, the thing i have not been able to quite fathom from rtfm is this
20:41.18vaewyn42
20:41.19vaewyn:P
20:42.02grozif i set up asterisk at 2 locations, can i interconnect them in such a way that they have the appearance of being 'one big pbx'  with my link between them being all ip on private network ?
20:42.44vaewyngroz: yes... VERY easily... :}
20:42.50grozeffectively what i'm looking at doing, is trying to get  seamless link between 2 locations, one of them is very remote, but it has data via satellite link
20:43.04grozwe have 128K dedicated link over a satellite connection
20:43.06vaewyneww... sattelite... your gonna have latency issues
20:43.15grozahh, latency we already deal with
20:43.16vaewynbut yes... it would work
20:43.24grozusing globalstar phones at $1+ per minute
20:43.31vaewynouch hhehe
20:43.31grozso, latency is the least of the issues, never get rid of it
20:44.16reseauxI NEED HELP With "early B3 connect" on libPRI Thz
20:44.24grozok, that's basically the answer i was looking for, think i'm gonna start setting up a test scenario here, and ordering some hardware to try it all out
20:44.40vaewynbasic answer is... the remote * install can have a switch statement in the config pointing to your main * box... the dialplan on the main is available to both machines... and it figures everything out for you :P
20:44.44grozsuggestions on the 'right' stuff to buy for setting up a test bench to demo it all ?
20:45.03vaewyndigium hardware... whatever is appropriate for your needs
20:45.07grozi was thinking the digium stuff just cuz they seem to be behind it all
20:45.12grozbut never hurts to check
20:45.16vaewyn*nods*
20:45.25grozok, answers that one too, this was 'to easy' today
20:45.37vaewyn:}
20:45.53Cripongroz: you want to stick with digium hardware.  What kind of test bed are you looking at?
20:46.06vaewynIf you want H.323 gear or such you would go cisco but.. other than that digium rocks da house :P
20:46.24grozi'm looking at  a couple of pc's with just enuf hardware to support one connection at a time
20:46.32reseauxI NEED HELP With "early B3 connect" on libPRI Thz
20:46.35grozbasically what i need to do for a true test is this
20:46.44grozplug an analog phone in on one  end, another on the other end
20:46.44*** join/#asterisk Buana (~thomasn@Gcb0a.g.pppool.de)
20:46.50grozhave the ability to dial between them
20:47.02Cripongroz: why not test with soft phones?
20:47.05grozthen i'll put em on opposite ends of the satellite link, and we'll see how it works
20:47.14Cripongroz: have you used linux before?
20:47.16grozoh crip, i have to demonstrate 'real phones' in this case
20:47.23grozyes, do embedded stuff for a living
20:47.27grozso not scared of linux
20:47.35groz<---- rolls own lfs for every project
20:47.40vaewyn:}
20:47.57Cripongroz: ok.. for testing you could use s100u's but they're flakey.. You'd need tdm400p's to do a box to box call with no external lines
20:48.01Buanacool, just found mail-archive from may-2016 on asterisk-side:)
20:48.03grozfirst thing i'll be looking at, is load this all up on a dedicated box, busybox boot and 'just run' from flash
20:48.09*** join/#asterisk PlainWhiteTrash (~PlainWhit@ps-ast-router.papersoft.com)
20:48.20Cripongroz: well good luck
20:48.23grozok crip, thanks
20:48.36grozwhat i was looking at was the digium devloper kit that comes with one of each
20:48.40grozjust order 2 of those
20:48.52Cripondo you need the x100p?
20:48.52grozlets me play with any scenario
20:49.00vaewynthat would work... would let you take in external calls also to play with
20:49.03grozit leaves me options to connect it to the existing systems
20:49.17Criponyep.. get the tdm dev kit.. don't get light
20:49.20Criponit sucks
20:49.24vaewyn:}
20:49.30grozyah, already saw that, saw 'usb' and said 'nope'
20:49.50Exomorphbkw_: You around?
20:50.36grozmy plan is basically a dedicated box running from a flash disk, appropriate hardware plugged in, no monitors or keyboards, just treat it like an appliance, has an off and on switch, that's it
20:53.27fonzaihi guys! I try to run asterisk on other machine than it was compiled on. But now I get "...Illegal instrucion" when trying to start it. that happens after successfully loading some of the modules. The computers have exactly the same kernel, but different CPUs. any ideas? TIA
20:54.24reseauxfonzai: different library... dirrent type distribution... i think binary is not good...
20:54.51fonzaireseaux: the libraries and the distribution are the same, as well
20:55.09reseauxthe kernel is the same?
20:55.23extremisshould I get dtmf on my T1 or DT or MF?
20:55.25fonzaireseaux: yes
20:55.44reseauxsame hardware?
20:56.07extremisI need someone with a clue to answer, I have the telco on the phone
20:56.08fonzaireseaux: not exactly, different cpu etc.
20:56.11extremisand they are building the circuit
20:56.23extremisplease
20:56.45reseauxextremis: I cant understand!please explain more.thz
20:57.14fonzaireseaux: I looked at Makefile, but it seems that it doesn't make any optimizations based on CPU type automaticly. or such..
20:57.49extremiswhats glare action?
20:58.55grozanother dumb question guys, how much cpu horsepower is needed for a system to handle say 4 lines, and have no problems keeping up with everything, including compression etc ?
20:59.12jetswe need an on,ine guestimator "
20:59.32grozi looked, but i did ask politely if dumb qs were ok :)
20:59.41grozcouldn't find this stuff online
20:59.45jets"X t1's, X pri's, X SIP phones" "You will need about a .... xx processor, xx memory"
21:00.01PBXtechgreat idea jets
21:00.14grozyah, i'm just wondering if a epia board with the 533mhz fanless processor will be able to keep up
21:00.38grozi see a huge value to having a solid state fanless system
21:00.56ExomorphUgs.... I'm having lots of problems with Asterisk hanging...
21:00.57extremisyou won't be able to get 4 x100p's in there
21:01.04extremisirq sharing is a big problem
21:01.25PBXtechdamn the stable release!
21:01.46grozeven on the 4 port board ?
21:02.04extremisthere isn't a 4port x100p
21:02.13grozno, i meant 4 phones connected
21:02.16PBXtech12[4jets12]1: if you have the info I could put up a page to do that
21:02.16grozno external lines
21:02.22grozthat goes out over data link
21:02.26extremisoh, helll I have 10 with a 450mhz
21:02.39grozok cool
21:03.27grozcuz of my remote location, solid state is a HUGE benefit, it's in the wheelhouse of a vessel at sea
21:04.02extremismmm saltwater
21:04.11grozyah i know, already got a dozen systems on the boat
21:04.21c4uldr0nI just started making my Asterisk box... I have my 2 x X100P installed (I have two incoming phone lines)... I'm waiting for my 3 SIP phones to come in a couple days but I want to have my * box ready when they arrive... Can somebody help with a X100P/SIP config
21:04.28grozsaltwater is nothing compared to the problems of the antenna
21:05.06c4uldr0nif somebody can help, plz private msg me ;)
21:06.48mishehuhmm...   Asterisk 10! hasn't yet been released...  I'm disappointed
21:06.49mishehuheh
21:07.10*** join/#asterisk j35 (~j23@runningawayfrom.avatara.org)
21:08.55reseauxI NEED HELP With "early B3 connect" on libPRI Thz
21:09.27jetsPBXtech: I don't have that info ;)
21:09.31jetsi have no idea
21:10.36*** join/#asterisk jsharp (~jsharp@www.thegeekworks.com)
21:10.49PBXtechbut it is a great idea :)
21:11.00jsharpWill setting nat=yes on a SIP peer do bad things if the phone really isn't behind nat?
21:11.48mishehuit will make your hair turn green
21:11.50jetsjsharp: i don't think so
21:12.14jsharpThat sure beats the grey it is right now.
21:12.35outtolunche meant florenscent green
21:12.51outtoluncer -n
21:13.09oejjsharp: It means that Asterisk will check for the IP address of the SENDER instead of reading the SIP message.
21:13.15j35so i'm trying to install zaptel, and i complie it and then install it and modprobe zaptel and get a BUNCH of unresovled symbol's.. any idea?
21:13.24*** join/#asterisk lkc1 (~lkc@hoochie.digium.com)
21:13.28oejjsharp: So in most cases, it should not mean anything to turn NAT=on
21:13.35jsharpAs long as its not phosphorescent green & doesn't glow in the dark.
21:13.51mishehuj35: you run depmod -a recently?
21:13.58mishehusometimes taht fixes it.
21:14.11jsharpoej:  Excellent.  That's what I thought.  I'm building sip.conf out of a database & don't wanna deal with users knowing if they're behind nat or not.
21:14.12mishehujsharp: if your uptime is greater than 3 hours, it will
21:14.15j35mishehu: it still gives the unresolved symbols in........ i'm using a stock debian kernel
21:14.18j35should i build my own?
21:14.26jsharpI'm safe then, I just rebooted the asterisk box.
21:14.33mishehuj35: *shrug*, i use slack, and always build my own.
21:14.35*** join/#asterisk w0ss (~w0ss@h00e01455ec48.ne.client2.attbi.com)
21:14.55j35lol can't hurt to try
21:15.02PBXtech4y5u6p
21:15.04PBXtecherr
21:15.12PBXtechwrong winder
21:15.18mishehuwrong colors too
21:15.24PBXtech1H2e3H4e5H6e
21:15.24PBXtechoops
21:15.25PBXtechhah
21:15.27jetsWhats the best way to do contexts... say all of my phones are in the [corporate] context.
21:15.43jetsand i want then to use long distance dialing say a context [longdistance]
21:15.56jetsin the [longdistance] context do i want to include the corporate context
21:15.57jetsor vice versa
21:16.21mishehuvice versa i think.
21:16.37PBXtechwhy
21:16.39jetsbut that means someone at the default s, menu can use the longdistance...
21:16.48jetsbecause [longdistance is included]
21:17.08PBXtechwhat if you had a lobby phone
21:17.13mishehujets: but the other way around, you might be putting [longdistance] into other contexts, and you don't want to include corporate in all other contexts
21:17.40mishehujets: what is the "default s, menu" ?
21:19.02*** part/#asterisk Buana (~thomasn@Gcb0a.g.pppool.de)
21:19.09PBXtechi would have the include in the corporate context
21:19.20PBXtechcause if you had a lobby phone you would just give it a local context
21:19.29mishehuor a [lobby] context
21:19.31PBXtechwhat about international calls? that open for everyone
21:19.39mishehuor a [whateverthefuckyouwant] context
21:19.48PBXtechgoes against regular PBX methods thou
21:20.12PBXtecheach phone has a restiction a restriction
21:20.20*** join/#asterisk drgalaxy (~bduhan@adsl-66-140-100-19.dsl.lbcktx.swbell.net)
21:20.51mishehuand what are 'regular pbx methods' then?
21:21.25jetshrm the doc/SECURITY in the source does it a weird way
21:21.25reseauxI NEED HELP With "early B3 connect" on libPRI Thz
21:21.33PBXtechyou assign a phone a restiction
21:22.13mishehuPBXtech: well, that's not exactly the same as going against an rfc.  ;-)
21:22.20jtoddCan someone give me an example of a toll-free prefix in the UK?  Here in the US, an example would be 1800xxxxxxx.
21:23.26jtoddNever mind - found a list.
21:23.30PBXtechguess what your used to
21:23.35mishehuhope it's an accurate list ;-)
21:24.22jtoddprobably not.  http://mirror.lcs.mit.edu/telecom-archives/archives/country.codes/toll.free.prefixes
21:24.30jtodd0345 still valid?
21:25.07zoa=============
21:25.08zoaB e l g i u m
21:25.08zoa=============
21:25.08zoa<PROTECTED>
21:25.08zoa11 xxxx
21:25.08zoa<PROTECTED>
21:25.10zoathat is so wrong
21:25.15zoadon't believe that list !
21:25.28jtoddzoa: so give me an example of a toll free prefix in Belgium.  I just need an example for a demo script.
21:25.41mishehubelgium is the most horrendous swear word in the galaxy
21:25.45zoa08
21:25.52zoamishehu: why ?
21:25.55jtoddOK, that's fine.  What's belgium's country code?
21:25.58zoa32
21:26.01jtoddthanks.
21:26.07zoabut that won't work internationally
21:26.07[Sim]*yawn*
21:26.46*** join/#asterisk heison (~heison@ns1.somanetworks.com)
21:26.49reseauxThere is someone that offer a big Voip Termination for 30000 Euro/month of call
21:26.55ExomorphAnyone having problems with Asterisk always sending calls to the fax extention?
21:27.09zoareseaux what do you mean ?
21:27.11*** part/#asterisk VoiceLynx (VoiceLynx@rrcs-central-24-106-64-175.biz.rr.com)
21:27.16mishehuzoa: you never read The Guide?
21:27.24zoawhat guide ?
21:27.32mishehu~theanswer zoa
21:27.36zoa: 42
21:27.39jtoddzoa: what won't work internationally?  the country code or dialing toll-free stuff from outside of Belgium?
21:27.40reseauxi have a company in italy specialized in Etnic traffic...
21:27.43jetsokay
21:27.45jetsi have a problem
21:27.48jjanzerheh
21:27.54zoafrom outside of belgium
21:27.58mishehuzoa: The Hitchhiker's Guide to the Galaxy
21:28.01zoaaaah no
21:28.04zoadont know that
21:28.11jtoddzoa: Maybe not for much longer.  :-)  You'll see in my note shortly to the list.  
21:28.15daorkdoes jbot know the question?
21:28.20vaewynAFKExomorph: yes... see http://bugs.digium.com/bug_view_page.php?bug_id=0000696 and http://bugs.digium.com/bug_view_page.php?bug_id=0000649
21:28.28tclarkExomorph: beaten to death on the ml, new dsp is fsckd, im makefile set it back to odd_dsp_routines
21:28.29zoajtodd: i can dial to it from out of europe
21:28.36zoabut i still have to pay to it though :(
21:28.55Exomorphtclark: AHHHHHH!   Thank-you!
21:28.59jtoddzoa: OK, so from Germany, as an example, you can dial 3208xxxxxx but you get charged for an international call, right?
21:29.08zoano no no
21:29.11zoafrom germany
21:29.18zoayou need to call 00328 prolly
21:29.21Exomorphvaewyn: And thanks as well. :)
21:29.25zoabut you can't call that number
21:29.31zoa0800 numbers are for belgium only
21:29.35zoathey are blocked by the isp's
21:29.37vaewynExomorph: no prob :}
21:30.21jtoddzoa:  So, what would you dial if you were in Belgium, and were dialing a hypothetical toll-free number?
21:30.51zoa0800
21:31.15jtoddzoa: Let's see a full number, if you have a "real" example.
21:31.21zoa080060800
21:31.27jtoddok.
21:31.46jtoddso is 08 the prefix for toll-free or is 0800 the prefix for toll-free?
21:31.51zoa0800
21:31.58jtoddOK.
21:32.07zoayou could have 087 as an area code i think
21:32.13zoai could look it up if you want
21:32.25jtoddzoa: not that important;I just wanted one valid example.
21:32.44zoawhat are you up to ?
21:32.55jtoddWriting up ENUM examples for freenum.org
21:33.06zoacool
21:33.52jetsyou shut up jjanzer
21:34.02jetswhen you do a [longdistance] -- and then include default
21:34.03jjanzereh?
21:34.12jetsoh nevermind
21:34.15jetsnobody will be of any help
21:34.29bkw_IDSKLFJLSDFIJSLEIF
21:34.36bkw_I ready to shoot res_parking and res_musiconhold
21:34.51bkw_why the fuck are they modules?  if you can't unload them they should be static
21:35.12mishehujets: why would you include [default] in [longdistance] ?
21:39.57ManxPowerbkw_, because you could load a different one that exported the same symbols?
21:40.03tokenANyone know is * can Fork SIP Calls
21:40.12ManxPowertoken, Define "fork"
21:40.13denonfork?
21:40.20denonyou can conference em
21:40.24ManxPowerUsually it's the SIP client that forks up the call, not Asterisk.
21:40.33tokenor Multiple users are allowed to register with the Same SIP Reg/Auth different phones
21:40.48ManxPowertoken, that's not supported.
21:41.04denonnot entirely wise either
21:41.06ManxPowerSince registration is for Asterisk routing a call to the SIP endpoint it doesn't make a lot of sense to support it.
21:41.46tokenWell what if you had A hpone in your office and you had a SIP usb phone in China
21:42.03token*hpone phone
21:43.05ManxPowerThen deal with it with extensions.conf and Dial(SIP/office&SIP/china)
21:43.37jorgeraidelque manera de darme problemas el 0.7.0 jajaja
21:43.38jorgeraidelwow
21:45.31*** join/#asterisk point (~litw@195.161.106.222)
21:46.25[Sim]~seen kapejod
21:46.28kapejod <~kapejod@p509241B8.dip0.t-ipconnect.de> was last seen on IRC in channel #asterisk, 3h 22m 25s ago, saying: 'hi there'.
21:49.12point~seen JerJer
21:49.13jerjer is currently on #asterisk.  Has said a total of 25 messages.  Is idling for 3h 39m 1s
21:49.17point:)
21:49.46point~help
21:53.43voidptr_you see
21:53.44PBXtechwhats a cheep price for a 7960
21:53.49voidptr_eating and drinking is better
21:53.54Corydon76-home~seen all-the-new-asterisk-sounds
21:53.55Corydon76-home: i haven't seen 'all-the-new-asterisk-sounds'
21:53.58voidptr_i'm noyt completely drunk now
21:54.11voidptr_just a biot
21:54.27*** join/#asterisk clh (~me@216.253.86.210)
21:56.00jtoddyeah, yeah... .I have a real job that takes up 14 hours a day... the sounds are open on my desktop; alphabet and like 20 phrases done... more work this PM.
21:56.28Corydon76-homeNo pressure.  ;-)
21:56.33PBXtechhate those kind of jobs
21:56.36jtoddnot saying that othes don't have real jobs, but this one is a real doozy.
21:56.58PBXtechtelecom jobs are boring
21:56.59jtoddand Ijust got handed two more full time jobs that I get to do, on top of the two I already had.
21:57.05jrollysonhello
21:57.34PBXtechhi
22:00.04bkw_anthm I blew it up
22:00.05bkw_haha
22:00.06SobekAre there no power supplies shipped with an IAXy?
22:00.19zoayou have an iaxy ?
22:01.31JerJeriaxy iaxy iaxy!
22:01.41jsharptelecom jobs sure beat the hell out of "internet" jobs.
22:01.52jtoddjsharp: there is a difference?
22:02.01PBXtechpackets suck
22:02.03PBXtechheh
22:02.19grozjtodd, yes there is, folks in telecom are still working
22:02.36jtoddgroz: I suppose you consider VoIP telecom, eh?
22:02.39*** join/#asterisk Tjardick (Tjardick@15-216.240.81.adsl.skynet.be)
22:02.39PBXtechand underpaid
22:02.56grozit's hybrid depending on installation i guess
22:02.57PBXtechexcept me
22:03.08*** join/#asterisk omer (~omer@hoochie.digium.com)
22:03.10tholoOn the other hand, some of us that were working in "internet" jobs just plain retired...
22:03.29jsharpjtodd:  Yeah, at least in the telecom industry you don't have a bunch of people hiding in their mommy & daddy's basement claiming to be a "web engineer" cause they know how to use frontpage.
22:05.36jtoddjsharp: Yeah, you get a bunch of people hiding in their mommy * daddy's basement claiming to be a "telco" because they know how to use asterisk.  LOL.
22:05.54jsharpHeh.  True.
22:06.46*** join/#asterisk digger_ (~digger@almestien.com)
22:08.33tclarkjealous
22:08.35bkw_har har har
22:11.07JerJerlocal config isssue,  NEXT
22:12.06bkw_tholo
22:12.17bkw_this codec problem is a combo of fwd and the 7960
22:12.32bkw_the 7960 offers it so FWD sees that and forces it
22:14.43bkw_tholo have you set the perfered code on the 7960 to ulaw or alaw?
22:14.45bkw_does it happen?
22:14.51*** join/#asterisk rocketman (~eedgar@c-66-41-182-23.mn.client2.attbi.com)
22:16.41c4uldr0nif I have 2 X100P's and my phones are all SIP, can I disable chan_modem.so  in modules.conf?
22:19.20*** join/#asterisk calvis (~calvis@hoochie.digium.com)
22:19.42Tjardickc4uldr0n, sure, you only need to modules for things you use ;)
22:20.31Tjardickbut don't forget to disable the other modem modules two
22:20.32Tjardicktoo
22:20.37Tjardickas they depend on the main one
22:22.06ManxPowerc4uldr0n, YES!
22:22.14Corydon76-home~seen coil
22:22.24coil <0@user-69-1-15-56.knology.net> was last seen on IRC in channel #asterisk, 40d 16h 48m 21s ago, saying: 'no'.
22:22.34Corydon76-home~seen mr
22:22.35mr <~fusi@pD953224A.dip.t-dialin.net> was last seen on IRC in channel #debian, 7h 9m 32s ago, saying: 'thanks i try it'.
22:23.15FuzzyCatfekcing reload fucked again
22:23.36*** join/#asterisk easydone (~easydone@eksel.demon.nl)
22:24.06SobekLOL I just read the iax song  
22:24.50voidptr_oh man
22:24.54Connorokay.. I really need a SIP--H323 converter or something.. I MUST get Asterisk talking to CCM
22:25.03dnchey FuzzyCat, whats up with nl.nixhelp.org
22:25.20c4uldr0nI just ran "make samples" so I'm getting rid of the unneeded items right now
22:25.24voidptr_hey dnc
22:25.25jetswhat is the function of ignorepat
22:25.32jetsi don't really understand it i suppose
22:25.35mbrancaConnor use * with chan_h323
22:25.45jetsoh nevermind
22:25.45voidptr_jets: when you press that digit it keeps dialtone
22:25.46jetsyes i do
22:25.54voidptr_iirc
22:26.00Connormbranca: chan_h323 won't work with CCM
22:26.01JerJerdon't run make samples then
22:26.13easydonewasim: http://farfon.convergence.com.pk/ is not working.....
22:26.24c4uldr0nheh, doesn't that just move the conf files into the /etc/asterisk folder?
22:26.24voidptr_: idle     : 178 hours 46 mins 32 secs (signon: Tue Jan  6 07:25:48 2004)
22:26.33voidptr_i dont think wasim will respond easydone ;)
22:26.42easydonewhy not?
22:26.57voidptr_he hasn't said anything for 178 hours
22:27.23lorena26jerjer, for what when i call to cisco in g729, give me error of codec 19 ?
22:27.30voidptr_he's quickly approaching the top idlers
22:27.30easydonevoidptr: meditating?
22:27.34lorena26Jan 15 21:30:01 NOTICE[1267154224]: rtp.c:418 ast_rtp_read: Unknown RTP codec 19 received
22:27.40voidptr_i guess ;)
22:27.45JerJerlorena26: no cng
22:28.01ManxPowerlorena26, search the mailing list archives
22:28.38lorena26I used g729r8 in cisco with No VAD
22:28.51mbrancaConnor, sorry, I'm not a ccm expert... what protos ccm speak, so?
22:29.31Connormbranca: h323, sccp and mgcp I think... but sccp and mgcp are for phones only.. not trunking.. which is what I'm needing
22:30.42bkw_CUSOTMERS SUCKS
22:30.50PBXtechyes they do
22:30.52dougheckalol
22:31.29PBXtechif they are cute ill let them really suck
22:31.32voidptr_i'm not complaining if they pay good
22:31.33voidptr_but they dont
22:31.34voidptr_:P
22:31.39PBXtecherrr did I say that out loud?
22:31.59mbrancaConnor so why chan_h323 won't work with ccm?
22:32.43PBXtechif they are complaining they get a higher percentage
22:32.44Connormbranca: Ask JerJer. :)  It won't hand off the RTP session to the phone.
22:32.56tokenBKW is * 0.7.0 stable now
22:33.01tokenany notable problems
22:33.12PBXtech0.7.1.4.7 is good
22:33.30tokenwill I get that from default CVS
22:33.34dnchey voidptr, hows life?  
22:33.35token:)
22:33.38PBXtechend of the day silly
22:33.49dnci played with a motorola canopy 5ghz antenna today
22:33.55dncit was lovely
22:34.01PBXtechlovely
22:34.07tokenSeriously is 0.7.0 Stable!!
22:34.20PBXtechheard lots of problems dog
22:34.23JerJernobody said 0.7.0 was stable
22:34.31rocketmancan anyone tell me a good linux softphone?
22:34.33pros12is anyone running asterisk with pstn in montreal?
22:34.43FuzzyCatdnc
22:34.47tokenok!  that is why the question came up I guess
22:34.47FuzzyCatnothing...
22:35.00dantCVS still closed?
22:35.12voidptr_dnc: ok... it was free beer again this evening
22:35.45ConnorOKay.. someone explain the gk funtion of h323 in chan_h323
22:36.30mbrancause it for register aliases to a gk
22:36.40jjanzerrocketman, kphone is decent
22:37.07rocketmanok I will look into that
22:37.12Connorso, is ccm considered a gk ?
22:37.19mbrancaso you can tell to a gk to handle (for example) all number beginning with '2' by passing them to *, in the context specifed in h323.conf
22:37.45*** join/#asterisk Cinzas (~AsHeS@gil.di.uminho.pt)
22:40.26mbrancanite all
22:40.54jrollyson"If SCO thought threat-born licensing fees would provide a quick boost to the bottom line, it looks to have miscalculated. As fellow Fool Tom Taulli noted last month, SCO has tried to cast its lawsuits in apocalyptic terms. But with the entire computing world putting its money behind Linux, it appears that, for SCO, the apocalypse is now."
22:41.51c4uldr0nis there a good tutorial for configuring Asterisk? I'd rather read than bug the hell out of everybody here ;)
22:42.43Cinzasthe best tutorial that i found ..... google
22:42.48Cinzas:(
22:43.56c4uldr0nduh ;)
22:44.04Cinzasreally
22:44.59dncc4uldr0n: http://www.automated.it/guidetoasterisk.htm
22:45.32c4uldr0nthanks
22:46.24grozthanks dnc
22:46.29grozthat's a nice starting point
22:46.39groz<--- just starting to do first configure after building
22:47.11c4uldr0nI don't suppose the X100P can dial-out as well as receive phone calls ;)
22:47.16*** join/#asterisk draconius (~j00momma@12-218-60-36.client.mchsi.com)
22:47.21c4uldr0nsorry for the lame ?
22:47.27PBXtechit can
22:47.31c4uldr0nk, whew
22:47.35PBXtechheh
22:48.14c4uldr0nI was hoping to get this up and running in one day, it has so many features (Asterisk) that it's hard for me to learn it all in one day
22:48.33PBXtechtry 3 months
22:49.13c4uldr0nI only have 2 phone lines... I just want to receive calls and have extentions, transfer phone calls between 3 SIP phones in the office
22:49.22c4uldr0nyou would think, 1 day project ;)
22:49.26c4uldr0nbut I'm thinking more
22:49.34*** join/#asterisk FryGuy (~fryguy@c-24-2-50-122.client.comcast.net)
22:50.57Cinzas:)
22:51.40CinzasTOday i've tested * with gatekeeper (h323), ata (h323) anda ata(sip), anda i4l (isdn)
22:51.53CinzasWorking fine ! Rox
22:52.06Cinzasanda = and !
22:53.06PBXtechi cringe when people say h323
22:53.11denonh.323
22:53.13bkw_me too
22:53.13denonmwahahaha
22:53.24PBXtech1L2a3u4g5h6i7n8g 9O11u11t 12L13o14u19d
22:53.31PBXtechwoooo pretty
22:53.46Cinzassame thing
22:53.47denonI live if only to annoy bkw
22:54.08denonso PBXtech, you gonna kick in some $$ on my bounty?
22:54.32PBXtechI cringe when people say BSD
22:54.46denonwhat's wrong with FBSD and OBSD?
22:54.54PBXtechwhats wrong with linux
22:55.01denondont get me started
22:55.06Cinzaswhats wrong with u ?
22:55.21PBXtechumm dont get me started
22:55.47tokenanyone know how to get dial by name directory to work?
22:55.48denondon't worry, I'm not even gonna LET ya get started..
22:55.49denon:)
22:55.53tokenIs it at all possible?
22:56.17crontibsdenon the bounty starting to look good
22:56.37tokenAnyone Know?
22:56.47*** join/#asterisk haighis (~chatzilla@216.58.40.2)
22:56.53tokenAnyone?  Bueller?  Bueller?
22:57.22*** join/#asterisk zwi (~zwi@216.88.131.43)
22:57.23jrollysonkram: good timing
22:57.38jrollysonkram: can you look at bug #853
22:57.45denoncrontibs: yeah? I havnet looked today
22:57.46PBXtechand #823
22:58.05denoncrontibs: oh, nothin new ..
22:58.09haighisbkw_, regarding callerid/canada/stentor switch i tried the same x100p at a different location..in Burlington...still getting now callerid coming through.. i wanted to try on a different switch to see if i get the same errors..
22:58.13PBXtechwhats the bounty up to now anyway?
22:58.22jrollysonPBXtech: 823 is already fixed ;)
22:58.23PBXtechI cringe when people say canada
22:58.28denonPBXtech: $350
22:58.33PBXtechjust wanted him to look at it
22:58.36PBXtech:)
22:58.42PBXtechthough it was like 800 ish
22:58.45haighisPBXtech,  you love canada
22:59.03PBXtechnuke em
22:59.07c4uldr0nanybody use MSN to test sip?
22:59.13haighiswe bring lots of good things...like a the Toronto Seal hunt..
22:59.28PBXtechcant buy drugs there anymore so nuke me
22:59.30PBXtechem
22:59.33PBXtechheh
22:59.49tokenAnyone Know if directory sservice dial by name is available on asterisk?
22:59.54haighisdid you hear about the big bust in Barrie..in the molson brewery?
23:00.13haighisgood stuff.....
23:00.17PBXtechwe went to that big old waterfall a year ago
23:00.20tokenAnyone?
23:00.22voidptr_haggis?
23:00.23voidptr_:)
23:00.23haighisleave some skin for the kids..
23:00.43tokenHe puts the lotion on the skin???
23:00.53haighisyeah...haighis...
23:00.58haighisdo you know a haighis?
23:01.13PBXtech12[4token12]1: I thought you could. never tried it myself
23:01.22zwican anybody tell me the exten pattern I need if I want to just direcly dial (10 digits) to an outside line?
23:01.40PBXtechnxxnxxxxxx
23:01.57tokenn= Digits 2-9 x= digits 0-9
23:02.06*** join/#asterisk johnny_ (~johnny@adsl-63-202-210-220.dsl.snfc21.pacbell.net)
23:02.16zwiOh...and here I thought it was much harder than that thanks!
23:02.17haighishas problem with all calls coming in as fax been fixed or acknowledged as a problem?
23:02.19johnny_hi all, i have some dumb questions if ya all wouldnt mind
23:02.21tokenPBXTech you know where I can get a working config for Dial directory
23:02.28token<PROTECTED>
23:02.36tokenheading out to sea
23:03.01haighistoken: xten => 411,1,Directory,default
23:03.12haighissorry that exten => 411,1,Directory,default
23:03.19PBXtechhttp://www.bkw.org/~brian/asterisk-conf/
23:03.23PBXtechhe has a config
23:03.44haighisso no thoughts on the callerid?
23:03.49tokencan I make # Dial by directory
23:03.55haighis??
23:03.56PBXtechoh dial by name config.. umm
23:04.07*** join/#asterisk pchitescu (~r00t@home-25022.b.astral.ro)
23:04.22haighistoken, use 411 or some other number
23:04.38c4uldr0ndo I have to do anything special to Asterisk to allow MSN login via SIP for testing purposes?
23:04.47tokenI don't mean directory service though
23:04.58haighiswhat do you mean then?
23:05.02PBXtechut oh a root user  hack HIM!
23:05.15tokenmy 411 no calls out to directory assistance
23:05.29haighisokay..then make it 511 or *411
23:05.31johnny_I need to have a VoIP solution between remote sites and am I correct in saying that if we get a t1 pri, we would need one Wildcard TE410P w/ asterix?  do people ever use asterix w/ cisco routers(sorry for the dumb question)
23:05.56johnny_and we just configure the amount of B-channels in the conf?
23:06.05jetsjohnny_: correct
23:06.07voidptr_t100p should be ok too
23:06.16jetsand yes asterisk can use mgcp
23:06.34voidptr_te410p also allows use of channel banks
23:06.51johnny_ok, cool..im just really wondering if it will magically work with the PSTN, or how it determines data from voice...?  can someone shoot me a relavent FAQ link? im going through as many pages as possible
23:06.56PBXtech12[4token12]1: try this exten => *,1,Directory,default
23:06.59ManxPowerDoes anyone have any ideas of how to do this: Transfer a call to the parking extension, the parking application reads back the extension the call is parked on, then plays a beep, then connects you to another extension (overhead paging).  I can do all of that except automatically connect to the overhead paging.  Currently users have to hang up and then dial the paging extension
23:07.26johnny_im somewhat confused still about channel banks, all i know is that Direct-ID and caller ID are critical
23:08.01tokenPBXtech let me try thanks
23:08.16PBXtechthats what i had in my config.
23:09.02tokenwhere should I put it under greeting message you think?
23:09.24PBXtechthat was under my autoattendand  press * to hear company directory
23:09.55tokenPBXtech let me try thanks
23:10.02*** part/#asterisk haighis (~chatzilla@216.58.40.2)
23:11.11JerJerOpenh323 is not vulnerable. This is the old SNMP asn1 stack vulnerability.
23:11.11JerJerSome vendors might have used this in implementing h323 stacks, but
23:11.11JerJeropenh323 created its only asn1 code, which to my knowledge, does not have
23:11.11JerJerany such vulnerability.  h323 software that is based on openh323 should
23:11.11JerJernot have any problems related to this CERT.
23:11.42JerJers unfortunate that the one has to dig so deep to find out that it was
23:11.42JerJera particular ASN1 stack that has the problem, and not h323 in general.
23:11.45denonJerJer: they claim SIP/etc has the same issues
23:12.01JerJerSomeone should probably contact CERT to explain the difference between
23:12.01JerJerASN1 implmenetations and those things that are implemented with ASN1.
23:12.01JerJerThis CERT advisory is like finding a flaw in Linux rpc.statd UDP, and then
23:12.01JerJerreporting that anyone using IP is vulnerable.
23:12.19voidptr_:))
23:12.21JerJer-- Dean Anderson, a major Open H.323 contributor
23:14.06MikeJerJer: alive?
23:16.05JerJernope
23:16.15MikeJerJer: i just want your price list
23:16.16*** join/#asterisk wreckdiver2 (~goaway@216.82.109.5)
23:18.02denonMike: give him $5k and he'll do anything
23:18.09denoncouldnt get any simpler than that
23:18.22JerJeryep
23:18.29Mikebah
23:18.42JerJersend an email to sales@nufone.net requesting our rate table, if that's what ur lookin for
23:18.46JerJerdinner time
23:20.14*** join/#asterisk jsharp (~jsharp@www.thegeekworks.com)
23:20.27jsharpBORK BORK BORK.
23:20.38ConnorHow good is a 7905 phone?
23:21.44*** mode/#asterisk [-o denon] by ChanServ
23:21.53*** join/#asterisk zoa (~john@kn-upc-1.kuleuven.net)
23:22.24jsharpHas Digium/VoiceAge worked out the stupidness of the G729 codec registry on SCSI devices?
23:22.26rozoConner: it works well. Only thing I don't like is the speaker phone (or lack of the speaker phone).
23:22.34zoajsharp: it works on scsi !!!
23:22.38jrollysonkram: bug 853
23:22.39Connorwhats the diff with it and the 7912?
23:22.39rozoit does?
23:22.43zoai use it on scsi already for a very long time
23:22.57jsharpI've heard mixed reviews.  I couldn't get it to run on SCSI myself.
23:23.02rozowhen we run the register program on a machine with SCSI, it locks up the machine and requires a hard reboot.
23:23.09zoafunky :)
23:23.10jsharpThat's what I ran across.
23:23.12kramjsharp: i think there may be a workaround
23:23.19kramyou'll have to talk to martin
23:23.22JerJernews at 11
23:23.27krambut the short answer is that voiceage refuses to fix their bug
23:23.30zoawhat will martin say ?
23:23.35JerJerhehe
23:23.35jsharpI'm shocked and surprised.
23:23.44zoamartin = digium martin ?
23:23.48kramyah
23:24.04jsharpOkee.  martin it is.
23:24.53dext0rwhere is struct ast_channel defined?
23:24.55johnny_one more dumb question..if i have 1 T100P and a full t1, I can use that card for both data for our lan and voip to the pstn?
23:24.57rozomartin was the one that got working for us but he has to register the codec himself.
23:25.06Connorcan someone tell me what the diff is between a cisco 7905g and 6912g ?
23:25.09zoahow do you mean ?
23:25.10Connorthey look the same...
23:25.23zoahow did he do that ?
23:28.23*** join/#asterisk s3gal (~leon@cuscon4743.tstt.net.tt)
23:28.48*** join/#asterisk jsharp (~jsharp@www.thegeekworks.com)
23:29.00*** join/#asterisk haighis (~chatzilla@216.58.40.2)
23:29.26dalaberaconnor > http://cisco.com/en/US/products/hw/phones/ps379/products_qanda_item09186a00801739d1.shtml
23:29.42*** join/#asterisk guibert (~guibert@hoochie.digium.com)
23:31.02*** part/#asterisk mmco (~irc@pD9E10B6E.dip.t-dialin.net)
23:31.37guiberthi, I  have a TDM400P with 4 FXS ports. On the other side 4 FXS ports given by my provider. Is there a way I can connect one port from the board to the ports of my provider to place calls through there?
23:31.42johnny_asterix does not support LDAP, but CCM does, right?
23:31.46*** join/#asterisk scud (~scud@12-219-155-152.client.mchsi.com)
23:32.06Connordalabera: Ethernet switch is the diff.. okay..
23:32.19Tangentguibert: Isn't that whay FXO cards are for?
23:33.18c4uldr0nwhen are they going to make a 4 port FXO?
23:35.23Tangentc4uldr0n: There's not much point making one when there's BRI and PRI cards already available
23:36.11c4uldr0nheh, sorry, but what are BRI and PRI?
23:36.19*** join/#asterisk blitzrate_sk00l (blitzrage@CPE0080c6f83fe1-CM.cpe.net.cable.rogers.com)
23:36.28jsharpExcept there's no cheap BRI cards that play well with US National ISDN U interfaces and PRI is damn expensive unless you've got more than 6 or 8 lines.
23:36.29TangentBasic Rate ISDN (2 channels) and Primary Rate ISDN (30 channels)
23:37.04*** join/#asterisk simprix (~simprix@24.247.181.124.gha.mi.chartermi.net)
23:37.04c4uldr0nk
23:37.04TangentHi Simon_ca
23:37.06TangentHi simprix
23:37.12TangentSimon_ca: Not you.. sorry
23:37.20simprixIm kinda new to asterisk I can't call between extensions i get a 404 error but i can make outbound calls over my pstn line
23:38.14c4uldr0nI just got asterisk working with x-lite, hehe, this is a huge advance for me ;)
23:38.43c4uldr0nnow I guess I have to configure the rest of asterisk so I can hook it up to my phone lines
23:39.00c4uldr0nw00t
23:39.06guibertyes, I should have FXO cards but i thought perhaps I can trick this someway. Perhaps using a small pbx as a bridge. Any ideas?
23:39.14Tangentc4uldr0n: You've leapt the first hurdle.. it gets easier from now
23:39.25c4uldr0nhehe, cool
23:39.31c4uldr0nI'm just pissed my phones aren't here yet
23:39.40c4uldr0nI ordered them Sunday, they shipped them yesterday
23:39.57Tangentc4uldr0n: That's not overnight delivery
23:40.13c4uldr0ngood thing I found X-lite so I can test and configure until they get here ;)
23:40.27Tangentc4uldr0n: I installed xlite yesterday :)
23:40.29*** join/#asterisk guibert1 (~guibert@hoochie.digium.com)
23:40.32TangentHadn't heard of it before then
23:40.41c4uldr0nit's pretty neat
23:40.50TangentYep.. nicer than SJPhone
23:41.00*** join/#asterisk pointer (pointer@aj.catt.com)
23:41.02c4uldr0nhaven't tried anything else, so i wouldn't know
23:41.10c4uldr0nall I know is that MSN didn't connect properly with SIP
23:41.14c4uldr0nso I looked further
23:41.18c4uldr0nand found X-lite
23:41.22derrickxlite works under wine too fyi
23:41.32Tangenteven better :)
23:41.34*** join/#asterisk {^DaNi^} (~dani@hoochie.digium.com)
23:41.40c4uldr0nwine rules ;)
23:41.44zoaderrick: is it good ?
23:41.51zoanot too much overhead / delay ?
23:41.51c4uldr0nI should see if Quickbooks works under wine, then I could get rid of Windows ;)
23:41.53derrickzoa, i didn't use it extensively, just for some testing
23:42.03derrickzoa, nope, not that i noticed but i keep beefy workstations
23:42.05zoai am thinking of using it bugtime
23:43.01*** join/#asterisk {^DaNi^} (~dani@atm0-1.r1-dcp.d1n.net)
23:45.07zoaeverybody go look at the latest bug report
23:45.13zoaand go add what you'd like to see documented
23:45.23zoalets give the documentation project some work !
23:45.26crontibswouldn't this be good error sound for *
23:45.26crontibsOperator: The fingers you have used to dial are too fat.  To obtain a
23:45.27crontibs<PROTECTED>
23:45.27crontibs<PROTECTED>
23:45.37crontibssimpsons ref
23:45.37zoayes i'm sure
23:46.30*** join/#asterisk fyman3 (~fyman@CPE-138-130-19-193.nsw.bigpond.net.au)
23:47.14*** join/#asterisk yaboo (~jsirucka@203-213-113-146-vic.tpgi.com.au)
23:47.46heisonanyone see this NOTICE?
23:47.51dantanyone fancy helping me test how many people I can fit in a meetme?
23:47.52heisonJan 15 18:44:04 NOTICE[1225991360]: chan_zap.c:3587 zt_read: Fax detected, but no fax extension
23:48.21Tangentdant: What do I have to do?
23:48.41zoaheison: a lot of people
23:48.47zoaits a little issue with new DSP
23:48.48dantDial(IAX2/guest@195.102.146.5/7777)
23:48.55dantshould do it
23:49.18heisonis this supposed to fix the WARNING: http://lists.digium.com/pipermail/asterisk-users/2003-December/030592.html
23:49.28zoarevert to old dsp in the makefile
23:49.30Tangentdant: In that case.. no.. I haven't worked out IAX yet or allowed it through my firewall
23:49.31zoais all you need to do
23:49.43heisondant: #define CONF_SIZE 160
23:49.51ConnorAre snome 200's fully functional in asterisk? I.E. Multiline support, attended and blind transfer, PARKING, and speed dial etc?
23:50.17dantheison, it's more the size of the box and the quality of the line I want to test :)
23:50.46ConnorI'm putting together a 10-12 phone system and need some cost effective phones..
23:51.27heisondant: depends how you gonna test this... 160 ppl via IAX will fill up a T1 pretty quickly.
23:52.03c4uldr0ncontext=sip                     ; Default for incoming calls
23:52.03c4uldr0nallow=ulaw
23:52.03c4uldr0nallow=alaw
23:52.03c4uldr0nallow=gsm
23:52.03c4uldr0nallow=ilbc
23:52.10c4uldr0ncrap, wrong button
23:52.12c4uldr0nfux0r
23:52.18c4uldr0nsorry guys
23:53.08dantwell, anyone else want to try and kill this p3 733?
23:53.29{^DaNi^}Hi people, I want deploy * as pbx in one of our offices. 150 cisco voip phone. 2 * box.
23:53.49{^DaNi^}And iax trunk with 2 * box located in our main office with PRI
23:54.07jetsdant: how many are in this meetme
23:54.10{^DaNi^}any help about the hardware for this purpouse
23:54.33heisonwhere can i find documentation for meetme? I wanna use the PIN feature, but I can't seem to find up to date docs
23:54.36dantjet's 3 so far :)
23:54.45{^DaNi^}in this moment i am thinking in use digium hardware for cards and supermicro as servers
23:54.51{^DaNi^}with debian as distro
23:55.32rob--<PROTECTED>
23:56.09zoadani: how much simultaneous calls + what codecs
23:56.12zoais you only concern
23:56.33zoagive us an exact layout
23:56.37zoaand we will try to help you
23:56.41{^DaNi^}zoa 30 calls as peak with g729
23:56.49dantnice
23:56.53jetsrob--:i'm in the meet me
23:56.54zoak, and those 150 cisco's
23:56.55jetswtf is that
23:57.00zoawhy 2 asterisk servers ?
23:57.10zoawhy two times two ?
23:57.13zoaredundancy ?
23:57.15{^DaNi^}zoa i need redundancy in all of our proyects
23:57.37{^DaNi^}i work for a bank if they dont see two box
23:57.40zoak, so i suppose you also have 2 pri's ?
23:57.59{^DaNi^}in our main office we will have more pri
23:58.05zoak
23:58.09{^DaNi^}this first office is only a test
23:58.19{^DaNi^}if work we will deploy asterisk in more offices
23:58.19zoathought about how handling the redundancy of the incoming pri lines btw ?
23:58.29zoai think you need a single xeon
23:58.33zoafor 30 times g729
23:59.04zoaalthough i have little information on the cpu overhead for trunking
23:59.07zoaas there seems to be some
23:59.10_aggelos_Dani: do you want to run the ip phones instead of phone cables ? evertying via ip ?
23:59.32zoai only have 10 shitty g729 licenses
23:59.40{^DaNi^}_aggelos_ yes, all ip  
23:59.55*** join/#asterisk dant (~dan@81-86-69-213.dsl.pipex.com)

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