irclog2html for #asterisk on 20060203

00:00.01Qwell[]I think there is a console version of linphone or something
00:00.15gaupethere is
00:00.22*** join/#asterisk Jun_ (n=wj1918@pool-138-89-62-149.nwrk.east.verizon.net)
00:00.55Flyboy-SR22Does anyone use FreeWorldTel for IAX or SIP termination and if so, how is their service...?
00:01.06Flyboy-SR22http://www.link2voip.com
00:01.46rpmi use them
00:02.02Flyboy-SR22Hey RPM - what do you think of them..?
00:02.17rpmits alright, i don't have anything to complain about from them
00:02.28rpmi get 13ms. lag time to them, best i found so far
00:02.50Flyboy-SR22are you pointing at sw1.link2voip.com..?
00:02.55Flyboy-SR22I get 50ms to them
00:03.20xachen50ms? :O
00:03.21rpmFreeWorldTel     139.142.184.136 (S)  255.255.255.255  4569          OK (16 ms)
00:03.40xachenI get 77ms from Michigan
00:03.41znoGjust wondering if anyone has any suggestions. I have a Sipura ATA and what happens is that call comes in on line 1, person calls line 2 to transfer the call, line 2 doesnt pick up and goes to voicemail, so the person flashes to go back to first person and it actually conferences them as the second call DID pick up (voicemail) and so the conversation gets recorded onto line 2's voicemail. Make sense?
00:03.43xachenand thats bad to Calgary
00:03.46Flyboy-SR22same IP
00:03.50rpmim in calgary
00:03.52trixterI get 1ms to stanaphone
00:03.55Flyboy-SR22an
00:03.55xachennot bad :)
00:04.04trixtermostly becuase they are at NYIIX
00:04.23rpm<PROTECTED>
00:04.24rpmheh
00:04.29Flyboy-SR22what is the ip for stanaphone trixter ..?
00:05.39xachen4.5cents a minute? :O
00:05.42xachenwith Link2voip
00:05.47Flyboy-SR22ouch
00:05.49rpmyeah, its pricey.
00:05.51Flyboy-SR22inbound
00:05.58Flyboy-SR22outbound is like 1.9
00:06.10Flyboy-SR22nufone is 2 cents in and out
00:06.11rpminbound is free if you get the 10$ number
00:06.12xachenI use Asterlink for outbound and my 800 atm
00:06.25xachen4.5cents a minute for 800 I mean
00:06.28_Sam--i have a client on asterlink now
00:06.29xachenwhich is cheap if that includes Canada
00:06.35Flyboy-SR22$10 a month for 2 channels
00:06.40_Sam--asterlink is expensive for canada
00:06.47xachenyes
00:06.47xachen10c
00:06.56rpmnufone is 75ms away for me
00:07.15_Sam--what is the stanaphone ip?
00:07.28xachensip.stanaphone.com
00:07.36Ariel_znoG, wow  I have not seen that problem
00:07.52_Sam--damn, stan is close.
00:07.53Ariel_what does your asterisk say on the cli? do you have in the sip settings canreinvite=no
00:08.16_Sam--trixter:  you can recommend stan?
00:08.21*** join/#asterisk nagl (n=nagl@vie-086-059-104-148.dsl.sil.at)
00:08.26Ariel_I have a number with standaphone but have not used it in a long time.
00:08.37_Sam--Ariel_:  how do you remember the service?
00:09.05Ariel_i got it for a test of a sip provider. It was hard to get working but after it worked it was fine.
00:09.09xachenis link2voip good?
00:09.12_Sam--you CAN use it with asterisk right?
00:09.20Ariel_yes
00:09.31rpmlink2voip works well for me
00:09.54Ariel_I use for my home argh broadvoice due to a customer of mine and also it includes ld to australia.
00:10.33Ariel_I do not recommend BV there sound is not great and drop calls from time to time. But wife so far is happy talking to her sister.
00:11.06Ariel_Flyboy-SR22, do you have an SR22?
00:11.31Flyboy-SR22Ariel_ - I do
00:11.37Ariel_nice
00:11.42Flyboy-SR22Ariel_ - I love it...
00:11.51Flyboy-SR22Ariel_ - I hav owned two of them now
00:11.58Ariel_a group of us was going to get an SR20 but the insurance for it was too high.
00:12.18Flyboy-SR22Yea, that happens...there are 4 of us and we all have over 3000 hours
00:12.37Ariel_ahh we don't have that many
00:12.40Flyboy-SR22I am CFI/CFII Single and Multi and Rotorcraft with almost 200 hours in the SR22
00:12.52Flyboy-SR22been flying sine I was in the marine corps...
00:12.56znoGAriel_: was that to me?
00:13.04Flyboy-SR22but we were hit hard until after we all had 100+ hours in the CIrrus
00:13.20Flyboy-SR22IFR is one of the best tickets
00:13.26Flyboy-SR22also the most fun flying IMHO
00:13.29*** join/#asterisk SoloFlyer (n=soloflye@eth13515.sa.adsl.internode.on.net)
00:13.33Ariel_yes it is
00:13.38znoGAriel_: the main problem is that the Sipura thinks it should create a conference when switching between 2 calls, and I'm not sure how to tell it that it's NOT the case.
00:13.50Ariel_I have a stol airplane... it's a Koliber 150a
00:13.51Flyboy-SR22I live in San Diego, so we don't get much "real" IFR here
00:14.00*** join/#asterisk Wuntherdag (n=alexthew@rrcs-24-227-188-230.sw.biz.rr.com)
00:14.08Flyboy-SR22I started with a 150
00:14.11Flyboy-SR22then a 152
00:14.17Flyboy-SR22then a 172 Hawk XP II
00:14.20Flyboy-SR22then the CIrrus
00:14.25Ariel_Flyboy-SR22, down here we have year round great weather for flying.
00:14.33Flyboy-SR22I do most of my flying for business
00:14.35Flyboy-SR22:-)
00:14.37Flyboy-SR22Where are you at..?
00:14.39Ariel_znoG, but it should say something on the cli
00:14.47Ariel_Homestead,Florida
00:15.00WuntherdagHello all
00:15.08Flyboy-SR22Ariel_, Ah yes...
00:15.22Flyboy-SR22Ariel_, I bet you do !!
00:15.32WuntherdagLooking for help setting up my polycom to auto answer for intercom
00:15.41tainted_wundaboy which model?
00:15.43Ariel_we are selling her.  We are looking at getting an Archer III or Cardinal
00:15.47*** part/#asterisk thosa (n=thosa@p54879ADE.dip0.t-ipconnect.de)
00:15.48tainted_Wuntherdag that is
00:15.58Wuntherdag...
00:16.02rob0<== former enroute ATC
00:16.13Flyboy-SR22Good deal rob0
00:16.20Flyboy-SR22we have great controllers around here
00:16.27Flyboy-SR22fantastic actually
00:16.30Ariel_yes we also have good ones
00:16.48tainted_but we have the best ATCs here
00:16.56tainted_by far
00:16.58Ariel_but we have lots of traffic from the Class B MIA
00:17.05WuntherdagWhat channel is this?
00:17.19tainted_Wuntherdag WHAT MODEL POLYCOM
00:17.22Ariel_Wuntherdag, asterisk
00:17.39*** join/#asterisk fuubu (n=minimark@193.239.129.3)
00:17.42fuubuporno with  Jenifer Lopes  come to http://host4you.land.ru
00:17.44*** part/#asterisk fuubu (n=minimark@193.239.129.3)
00:17.46Wuntherdagoh ok that i clicked the wrong one, the 600 and 501
00:17.49Flyboy-SR22Yea, I have San Diego, Mirimar, CRQ, Camp Pendleton, March, Norton, Ontario, LAX...
00:17.57Flyboy-SR22lots of class B
00:18.13Flyboy-SR22I pretty much file IFR no matter where I go in LA
00:18.16Ariel_MIA is enought for us here
00:18.45Wuntherdaggot the config files set and the ext but its not working
00:19.10Ariel_Wuntherdag, which firmware
00:19.27*** join/#asterisk darkskiez (n=darkskie@bb-87-81-62-203.ukonline.co.uk)
00:19.29Wuntherdag1.4 and 2.6
00:19.41Wuntherdagsip/boot
00:19.47*** join/#asterisk darkskiez (n=darkskie@bb-87-81-62-203.ukonline.co.uk)
00:19.54Ariel_hummm you should get 1.6.2 at least firmware is ok
00:19.57Ariel_boot
00:20.05komnieveHi all - anyone know what the keyword is for tickles? i forgot :-)
00:20.50komnieveso i want to send a message to a desktop ap on a computer - from asterisk -
00:22.54Wuntherdagso is it just a firmware issue, Ariel
00:22.54Ariel_Wuntherdag, maybe but I know I had it working with 1.5 and 1.6.2
00:22.55Ariel_besides transfers work better with 1.5 then 1.4
00:22.58*** join/#asterisk Luhiwu (n=marsosa@200.63.89.240)
00:23.14*** join/#asterisk _flashnet[BNC] (i=flashnet@Darkstar.AceShells.com)
00:23.53Wuntherdaghow did you do your intercom and paging extension set up,
00:24.05Wuntherdagthat is did you use a prefix
00:24.07Luhiwuhi all, i'm using * with chan_h323 and it seems to be transcoding g729-slinear-g729 when it should do a native bridge... is it possible to native bridge rtp between protocols? or it will decode-encode always?
00:24.56*** join/#asterisk crschmidt (n=crschmid@64.92.170.181)
00:24.56Wuntherdaglike * or 4 and then the ext you wish to intercom
00:25.14Ariel_Wuntherdag, intercom is just an extension. and 5*exten is what we use but everyone is different.
00:25.24Ariel_275
00:26.20crschmidthttp://uc.org/read/Zasterisk <- I want to run that. Anyone got any suggestions for what I should do to get 'A working installation of Asterisk', 'A working installation of Festival, configured for use with Asterisk' ?
00:26.41Qwell[]crschmidt: Pay somebody, or read the docs
00:26.42mdavestill trying to find a provider that can provide me a 616 areacode DID and relay it to me over SIP/IAX/etc
00:26.54crschmidtQwell[]: I'm willing to read the docs, but I'm having problems finding 'the docs'
00:26.57mdaveand chage a *flat* monthly fee for it, no 'per minute' crap on an *incoming* call
00:27.16Qwell[]~docs
00:27.21jbot[docs] probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html
00:27.23Ariel_crschmidt, get asterisk@home it predone with all you need
00:27.40mdavecrschmidt, Qwell is a bit of a flamer, you might want to direct your questions elsewhere
00:27.54crschmidtmdave: *shrugs* I asked a shitty question
00:27.59mdaveQwell, playing music nicks are we?
00:28.13mdavemusicAL
00:28.26Qwell[]crschmidt: heh, that guy did a little talk at ETel...funny shit
00:28.28Ariel_flamer???? hummm wow did not know that....
00:28.47crschmidtmdave: I should have tried to set it up first. After looking at a list of 40 different .conf files in /etc/asterisk, I kind of figured i'd look for a 'quick start'
00:28.50*** join/#asterisk Stealthmethod (n=123@adsl-223-239-189.mia.bellsouth.net)
00:28.54Qwell[]I think so anyhow
00:29.04mdavei dont know why he even comes to the channel, since he doesnt seem interested in providing any real help
00:29.23Ariel_damm
00:29.29mdaveperhaps he should start an #asterisk-experts-only-to-boast-about-how-smart-we-are-and-not-help-anyone channel
00:29.46Ariel_crschmidt, if you look at the project for asterisk@home it can get you started complete one iso 20 minutes done
00:29.51rob0/j #asterisk-experts-only-to-boast-about-how-smart-we-are-and-not-help-anyone
00:29.53mdavecrschmidt, youve installed it? done make samples, etc?
00:30.02crschmidtmdave: I just grabbed the debian package
00:30.12Wuntherdagcan i post some lines from my ext.conf to look at Ariel
00:30.25mdaveyou know, im on fedora, and tried the rpm,. didnt get me anywhere.. i removed that, installed from source
00:30.25Luhiwuhi all, i'm using * with chan_h323 and it seems to be transcoding g729-slinear-g729 when it should do a native bridge... is it possible to native bridge rtp when using h323? or it will decode-encode always?
00:30.27crschmidtAriel_: hm... I've already got a system running that I don't really want to kill atm, but I'll consider it, thanks
00:30.31Ariel_Wuntherdag, on pastebin.ca
00:30.34crschmidtmdave: *nod*
00:30.36mdaveand did make install and make samples, and i had a working setup
00:30.38crschmidtI really shouldn't be playing
00:30.44crschmidtI should be working
00:30.51mdavewell if its working then no need to reinstall it
00:30.53crschmidtBut I worked too much today already ;)
00:30.58mdavealrighty
00:31.06crschmidtHow can I test if it's working?
00:31.21mdavewell.. do you have an IP phone, or an ATA, or something?
00:31.28Garak_I'm looking into getting local access to the PSTN, there are no VoIP providers in this area code, so what should I be asking the telco for? I'm looking for something like a T1 PRI I'm assuming
00:31.50mdaverun asterisk -vvvcn, and it should fire up and give you a CLI
00:31.56mdaveer, yeah, a softphone will work too
00:32.03mdaveassuming you have one of those and know how to configure it
00:32.15mdavebasically in sip.conf,
00:32.16Qwell[]crschmidt: You could also use chan_alsa, if you had a soundcard
00:32.28mdaveyou need to add an entry for whatever phone(real or soft) that you want to use
00:32.36Qwell[]Garak_: How many channels do you need?
00:32.45mdavethe part in the [] is the 'username' you tell your phone
00:33.04*** join/#asterisk brian_ (n=bkw_@ppp-70-128-122-10.dsl.tulsok.swbell.net)
00:33.26mdavei can paste you what i have for an SPA-2000 as an example, it might or might not need tweaking to work with your softphone
00:34.17Garak_Qwell[]: not sure yet, we are looking at the feasablity of expanding our webhosting company into VoIP services, so I'm thinking 23 channels should be enough to get us started.
00:34.42*** join/#asterisk Gunnar (n=gunnar@34.80-203-209.nextgentel.com)
00:35.09crschmidtmdave: That should be enough to get me started: just 'sip.conf' gets me closer. asterisk is running now, so I'll play with that and see where I can get, thanks
00:37.53WuntherdagAriel check http://pastebin.ca/39703
00:38.13Qwell[]Garak_: yeah, generally if you call your telco, they'll know whats going on, and refer you to one of their guys
00:39.43*** part/#asterisk Triple1243 (n=Triple12@modemcable171.79-70-69.mc.videotron.ca)
00:42.24Ariel_Wuntherdag, your using amp
00:42.27*** part/#asterisk rnovotny22 (n=rnovonty@198.57.19.112)
00:42.34Ariel_this is going to be a little harder....
00:42.38Wuntherdagyes
00:46.03Qwell[]bkw__
00:48.00bkw__how about a drink?
00:48.11Qwell[]sure, but you'll have to fly here
00:48.16bkw__hehe
00:48.20Qwell[]or walk, or whatever
00:49.05bkw__:P
00:50.37Qwell[]bkw__: You going to VON?
00:51.38*** join/#asterisk zotz (n=zotz@24.231.47.175)
00:52.03*** join/#asterisk spidey (n=spidey@12-216-165-134.client.mchsi.com)
00:53.31*** join/#asterisk elg (n=fugalh@falcon.fugal.net)
00:54.07*** join/#asterisk remmo (n=rem@203.31.238.31)
00:58.02*** join/#asterisk cassio- (i=cassio-@200216087155.user.veloxzone.com.br)
00:58.35WuntherdagLater asteriskes, I have become better at Asterisk (1). Thanks for help Ariel_!
00:58.39Luhiwui'm using * with chan_h323 and it seems to be transcoding g729-slinear-g729 when it should do a native bridge... is it possible to native bridge rtp when using h323? or it will decode-encode always?
00:59.54cassio-does anyone know how can I balance all the lines in asterisk?like I have 2 lines and 3 persons, so I can randomize all the lines they access
01:00.06Qwell[]Luhiwu: What makes you say it's being transcoded?
01:00.16Qwell[]cassio-: use zap groups
01:00.36LuhiwuQwell: this message says something about g729tolin: WARNING[7826]: samples/codec_g729.c:217 g729tolin_framein: Received a G.729 frame that was 2 bytes from RTP
01:00.55*** part/#asterisk crschmidt (n=crschmid@64.92.170.181)
01:00.55*** join/#asterisk RoyK (n=roy@10.80-203-106.nextgentel.com)
01:01.05cassio-Qwell does it use a line each time a person make a call?if he calls on line 1 , next call goes on line 2 and like that?
01:01.17Qwell[]cassio-: yes, it'll pick an available line
01:01.42cassio-Qwell thanks, I will try to find some info on that
01:01.51Qwell[]Luhiwu: it would seem you're right.  Are you sure both sides are g729?
01:01.56*** part/#asterisk Gunnar (n=gunnar@34.80-203-209.nextgentel.com)
01:02.11LuhiwuQwell: yes, h323.conf has disallow=all allow=g729
01:02.16*** join/#asterisk Gunnar (n=gunnar@34.80-203-209.nextgentel.com)
01:02.46Qwell[]Luhiwu: I don't know anything about h323 unfortunately...
01:03.33LuhiwuQwell: do you know who can help me with that besides jerjer?
01:03.52Qwell[]I don't think there are very many people who know h323
01:03.54*** join/#asterisk kyo (n=kio@ool-4577ae5e.dyn.optonline.net)
01:04.16remmoi'm using h323 and g729 and have not seen this problem before
01:04.50remmoQwell: what version of asterisk and h323 are you using?
01:05.05remmoalso do you have g729 codecs on the boxen
01:05.31Qwell[]remmo: I'm not, and I don't
01:05.44ahattarwhy i do not have h323.conf in etc/asterisk?
01:06.28Qwell[]ahattar: It's part of asterisk-addons
01:06.32ahattarok
01:06.45ahattari remember that i install it
01:06.49ahattarlet me check
01:07.01*** join/#asterisk M1 (n=zdeeck@r23s01p06.home.nbox.cz)
01:08.47*** join/#asterisk austinnichols10 (n=austinni@dsl-10-169.cofs.net)
01:08.57ahattarI install asterisk 1.2.4 and addons 1.2.1
01:09.08ahattarstill no h323
01:09.17ahattarh323.conf
01:09.24Luhiwuremmo: i'm the one with the problem with h323
01:13.40remmoluhiwu: whoops. im running an svn trunk and prrior cvs version for over 6 months running fine
01:14.39*** join/#asterisk M1 (n=zdeeck@r23s01p06.home.nbox.cz)
01:16.02*** part/#asterisk Gunnar (n=gunnar@34.80-203-209.nextgentel.com)
01:16.35*** join/#asterisk Gunnar (n=gunnar@34.80-203-209.nextgentel.com)
01:16.42Luhiwuremmo: do you have incoming and outgoing h323? are you using chan_h323 or ooh323?
01:17.32*** join/#asterisk jensd (i=jens@sip-proxy.gratissip.dk)
01:18.47*** join/#asterisk denon (i=denon@sassinak.net)
01:18.47*** mode/#asterisk [+o denon] by ChanServ
01:18.51jensdi'm looking for documentation on the SIP-B extension to SIP - anyone happens to have anything?
01:19.23*** join/#asterisk Gunnar (n=gunnar@34.80-203-209.nextgentel.com)
01:19.54Netgeeksmy poor 7960 is sick
01:20.18Qwell[]Netgeeks: Should have been running sccp...tsk, tsk, tsk
01:21.14NetgeeksI think it's been slumming with my SNOM and caught a STD  (SIP transmitted disease)
01:21.34NetgeeksI think I'll try an *ack* upgrade and see if it fixes things
01:21.48Netgeeksno thanks for sccp
01:22.11*** part/#asterisk jensd (i=jens@sip-proxy.gratissip.dk)
01:25.04*** join/#asterisk Gunnar (n=gunnar@34.80-203-209.nextgentel.com)
01:27.22*** join/#asterisk HamYaI (i=HamYai@125.24.26.220)
01:29.03*** join/#asterisk stealthmethod (n=123@adsl-223-239-189.mia.bellsouth.net)
01:31.10*** join/#asterisk Mark_Halverson (n=mhlvrs@67-139-119-152.dsl1.pco.ca.frontiernet.net)
01:32.37Mark_Halversonanyone try the dtmfmode=auto ??? is it working in 1.2.4
01:33.23SplasPoodztdummy /w my debian sid 2.6.15 is outputting TONS of this into my kernel logs:
01:33.27SplasPoodrtc: lost some interrupts at 1024Hz.
01:33.34SplasPoodanyone have any thoughts?
01:35.40*** join/#asterisk Jun_ (n=wj1918@pool-138-89-62-149.nwrk.east.verizon.net)
01:39.33TokyoJimuDoes asterisk honor the Presentation Restricted bit on an incoming PRI call?  Is this settable.  I see info in the docs about outgoing but not incoming.
01:39.47*** join/#asterisk caitif (n=luis@host15.intranet.digi.com.br)
01:40.00caitifhi
01:40.16caitifanybody use A2Billing???
01:45.22*** join/#asterisk justinu (n=Justin@cpe-72-129-86-208.socal.res.rr.com)
01:47.55tainted_A2CDRDeath
01:49.17*** join/#asterisk mgf (n=mgf@r200-40-218-138-dialup.adsl.anteldata.net.uy)
01:50.50tainted_that's how nugget rolls
01:51.18*** join/#asterisk kn0x (n=atlantic@71.194.235.251)
01:54.53*** join/#asterisk kc5cqm (n=kc5cqm_@cpe-68-206-116-214.stx.res.rr.com)
01:55.35caitifhow i export the tarifs rates from the peoplecall to a2billing??? Anybody know??
01:56.07kc5cqmwhat's the basic filesystem structure required for voicemail?  is it not /var/spool/asterisk/voicemail/default/XXXX (where XXXX = extension or number defined in voicemail.conf).  In particular, do the extenions go inside of "default" or parallel to it?
01:56.34Qwell[]kc5cqm: extensions go inside the context.  The voicemail system will create the structure for you
01:56.50kc5cqmI didn't have theh scripts that set up mailboxes
01:57.00Qwell[]kc5cqm: asterisk does it for you
01:57.06Qwell[]no need to run a script
01:57.09kc5cqmhmmm
01:57.16kc5cqmapp_voicemail.c:2384 leave_voicemail: No entry in voicemail config file for '1000'
01:57.26Qwell[]1.2.1?
01:57.28Qwell[]upgrade
01:57.33kc5cqmyeah yeah...I know
01:57.46Qwell[]no...voicemail is b0rked in 1.2.1.  upgrade
01:57.47kc5cqmwill very soon...just poking around with this thing right now
01:57.49kc5cqmoh
01:57.53kc5cqmthanks
01:58.11*** join/#asterisk [TK]D-Fender (n=joe@66.11.164.239)
01:58.26kc5cqmactualyl I think it's 1.2.3
01:58.30kc5cqmor just 1 ver behind
01:58.34Qwell[]check
01:59.08kc5cqmwhat's the option to display ver?
01:59.11Qwell[]show version
01:59.27*** join/#asterisk frade (n=frade@ip68-104-184-1.ph.ph.cox.net)
01:59.28kc5cqmyeah...its 1,2,1
01:59.29kc5cqmok
01:59.47Qwell[]mmhmm ;]
02:00.24kc5cqmguess it's been a couple versions the last few weeks...moves fast huh?
02:00.28*** join/#asterisk silly (n=silly@cpe-24-174-162-34.satx.res.rr.com)
02:02.36kc5cqmI guess a memory leak is reason enough to upgrade
02:08.26mgfHi! I'm having problems with transfers. In my current dialplan when a call comes in trough a trunk and they dial an extension, that extention does not answer, the call gets sent to a queue.  If a call is transfered to another extension and it is not answered there, it should go to the same queue but instead it get's dropped. I have been looking everywhere but can't find the problem... any clues?
02:09.15*** part/#asterisk Alric (n=nbowyer@masq.hyperusa.com)
02:09.21kc5cqmmy little underpowered Athlon XP2100 is compiling so slow....
02:10.52*** join/#asterisk FastJack_ (i=fastjack@p5091ECA3.dip.t-dialin.net)
02:11.05*** join/#asterisk |db48x| (n=db48x@cpe-66-25-39-5.houston.res.rr.com)
02:14.09*** part/#asterisk |db48x| (n=db48x@cpe-66-25-39-5.houston.res.rr.com)
02:14.52[TK]D-Fenderkc5cqm : You're 100 up on my * box :)
02:18.14*** join/#asterisk ke4qqq (n=chatzill@srv.fgp.com)
02:20.23kc5cqmwhatcha running TK?
02:20.37kc5cqmactually, this is just a sandbox...going to move it all over to a 933 PIII
02:20.54kc5cqmI need to get * off my workstation
02:22.22kc5cqmhey Qwell...same issue (although that may have been another problem).  Does a separate voicemail box have to be created for each incoming context?
02:23.44*** join/#asterisk Lurr (n=pr0ph3t@adsl-11-12-126.mia.bellsouth.net)
02:23.52*** part/#asterisk Lurr (n=pr0ph3t@adsl-11-12-126.mia.bellsouth.net)
02:25.24[TK]D-Fenderkc5cqm : My Server is an XP2000.  Does it all here, gateway, X (on a 52" HDTV), *, the works
02:25.41*** join/#asterisk jbalcomb (n=jbalcomb@gateway.imtco.com)
02:26.19*** join/#asterisk AsteriskNewbie (n=root@63.250.96.18)
02:26.51kc5cqmI like athlon chips
02:26.58kc5cqmseems amd is starting to lead certain markets too
02:27.11AsteriskNewbiehello everyone .. any asterisk guru around? :)
02:27.29kc5cqmwe're all here cutting our teeth AsteriskNewbie
02:27.45kc5cqmbut there are some generous gurus around
02:27.53AsteriskNewbiehehehehe .. I've got an TDM400 card ..
02:28.06AsteriskNewbieFollowed the instructions in the books and elsewhere ...
02:28.34AsteriskNewbiesip/iax work .. not problem .. but I can't get the card to pick my pstn line ..
02:28.35kc5cqmyou're up on me AsteriskNewbie...using a cheapie x101p here
02:28.52AsteriskNewbie:)
02:28.52AsteriskNewbieDoes it work for you?
02:28.53kc5cqmany errors when you load the kernel module AsteriskNewbie ?
02:29.00kc5cqmyeah, I eventually got it working...and echo ch
02:29.06AsteriskNewbieNo errors .. not any that I haven't fixed ..
02:29.06kc5cqmecho cancellation actually works
02:29.14kc5cqmdoes zttool report the channels?
02:29.33AsteriskNewbieYes .. it reports both the fxo and fxs channels no problem ..
02:29.49AsteriskNewbieAnd I have my phone line plugged into the fxo port ...
02:30.09AsteriskNewbieBut when I call the line .. it rings forever and ever, and asterisk doesn't even as much as log anything ...
02:30.47AsteriskNewbieIts almost as if asterisk has no idea someone is calling the line!
02:31.21kc5cqmhmm
02:31.34mgfyour phone line goes in the fxo port an your phone set goes ino the fxs port
02:31.48mgfare you sure you have ther right port?
02:31.50AsteriskNewbieThat's right ...
02:32.13kc5cqmcould it be a context issue for zapata.conf?
02:32.30AsteriskNewbieHmm .. I've been thru the context a million times ..
02:32.51mgffrom asterisk console
02:32.58kc5cqmwell, if ztcfg and zttool shows good, it's probably asterisk config
02:33.00mgfdoes zap show channels
02:33.16mgfshow the right zap channels in the right context?
02:33.48AsteriskNewbieHmm ...
02:33.52AsteriskNewbieThis is what it says:
02:34.02AsteriskNewbie<PROTECTED>
02:34.02AsteriskNewbie<PROTECTED>
02:34.03AsteriskNewbie<PROTECTED>
02:34.33AsteriskNewbieI never saw a working config before ... so am not sure what to expect ???
02:34.46mgfnothing else?
02:34.51AsteriskNewbieNothing else!
02:35.03mgfmmm
02:35.17AsteriskNewbiewhat else should I seeing??
02:35.40AsteriskNewbieztool .. show no alarms/ internally clocked
02:35.53AsteriskNewbieTotal/Conf/At: is 4/2/1
02:35.53mgfsometing is wrog either in your modules, zaptel.conf or zapata.conf
02:35.57mgf<PROTECTED>
02:36.08mgf<PROTECTED>
02:36.17mgf<PROTECTED>
02:36.23kc5cqmreload
02:36.24mgf<PROTECTED>
02:36.28kc5cqmoops
02:36.41mgfetc...
02:36.46AsteriskNewbieHmm ....
02:37.17tronixAsteriskN: look at this: http://pastebin.com/536385
02:37.21tronixthat's what a working zap setup looks like
02:37.30tronixit has my zapata.conf entries and 'zap show channels' output
02:37.44AsteriskNewbieok .. will take a look ...
02:37.52mgfmaybe take a look in /var/log/asterisk/full
02:38.04tronix(it's short)
02:38.21[TK]D-FenderAsteriskNewbie : Do you see WCTDM in "cat /proc/interrupts" ?
02:40.37AsteriskNewbieyes .. it is there ..
02:40.48*** join/#asterisk copantl (n=copantl@205.240.200.77)
02:41.10AsteriskNewbieand by the way .. my zap conf files can be seen at
02:41.13AsteriskNewbiehttp://63.250.96.18/asterisk/
02:42.16AsteriskNewbiemgf .. nothing in the logs ...
02:42.32*** part/#asterisk Igbothom_III (n=HiltonT@office.quarkit.com.au)
02:43.08*** join/#asterisk iq (n=iq@71-214-6-43.omah.qwest.net)
02:45.41AsteriskNewbieso according to my conf .. asterisk should pick up the line when I call .. then echo back to me ..
02:45.57AsteriskNewbiebut it never picks up the line. The logs dont' register anything. The line just keeps ringing forever ..
02:46.00AsteriskNewbie:(
02:47.14*** join/#asterisk angom_w (n=angom@red-corp-200.38.16.10.telnor.net)
02:47.17mgfmm
02:47.52mgfsorry can't see any problems
02:48.04AsteriskNewbie:)
02:48.28AsteriskNewbieI'm wondering if the card is toast ... but as I am a total newbie .. I want to believe the card is fine ..
02:48.45mgfit shouldn't be in extensions.conf
02:49.11AsteriskNewbie??? sorry .. I didn't get that ..
02:49.11mgfyou first need to se the channels with zap show channels
02:49.18tronixwhat's output of 'ztcfg -vvv'?
02:49.50SwKanyone know any good SIP to H323 software
02:49.57AsteriskNewbieoutput of ztcfg is ..
02:49.58QwellSwK: *? :p
02:49.58AsteriskNewbieChannel 01: FXO Kewlstart (Default) (Slaves: 01)
02:49.58AsteriskNewbieChannel 02: FXS Kewlstart (Default) (Slaves: 02)
02:49.59AsteriskNewbie2 channels configured.
02:50.10SwKqwell: * is over kill for what I want it for
02:50.10*** join/#asterisk _DAW (n=bob@adsl-222-26-17.msy.bellsouth.net)
02:50.26tronixAsteriskN: ok, that's good. what about:
02:50.31tronixlsmod|egrep "zaptel|wcfxo"
02:50.40tronix(do they both show up?)
02:50.56*** join/#asterisk YARICK (i=YARICK@usg-206-135-187-99.portnetworks.net)
02:51.11tronixand could you remind me which card this is?
02:51.13tronixTDM400P?
02:51.22AsteriskNewbieyes ..
02:51.25AsteriskNewbieTDM 400P ..
02:51.30tronixahh cool. I have a TDM400P, too
02:51.37AsteriskNewbiewith one fxo module and one fxs module ..
02:51.39tronixfxs module on #1, fxo module on #4
02:51.42AsteriskNewbie[root@localhost ~]# lsmod | egrep zaptel
02:51.42AsteriskNewbiezaptel                206724  11 wcusb,wctdm,wcfxo,wcte11xp,wct1xxp,wct4xxp,tor2crc_ccitt               2113  2 zaptel,hisax
02:51.51AsteriskNewbiethat's the output from lsmod ..
02:51.56tronixhmm.
02:52.07tronixlooks like you have too many unnecessary modules loaded
02:52.11tronixnot sure if that's a problem or not
02:52.14tronixcould do:
02:52.17Qwellcould be
02:52.18tronixmodprobe -r wcusb
02:52.22tronixmodprobe -r wcfxo
02:52.22tronixetc
02:52.32tronixkeep wctdm and zaptel
02:53.34AsteriskNewbieI did ..
02:53.45tronixok. now modprobe -r both wctdm then zaptel
02:53.46tronixthen
02:53.48AsteriskNewbieeverything is silent .. except for zaptel .. which is zaptel is in use ..
02:53.49tronixmodprobe zaptel
02:53.51tronixmodprobe wctdm
02:54.02tronixyou could stop asterisk
02:54.08tronixthen modprobe -r zaptel
02:54.13tronix(-r = unload module)
02:54.25Qwellrmmod is funner
02:55.07*** join/#asterisk mzo (n=moz@ool-435193b3.dyn.optonline.net)
02:55.09AsteriskNewbiesays zaptel is in use .. and asterisk is stoped ..
02:55.30*** join/#asterisk Defraz (n=t0tal@24-119-94-19.cpe.cableone.net)
02:55.34tronixhmm...
02:55.36tronixcould do:
02:55.40tronixlsof | grep zaptel
02:56.06AsteriskNewbiebash      4062   root  cwd       DIR      253,0    12288     328553 /usr/src/zap tel
02:56.12AsteriskNewbieis the result ..
02:56.20tronixhmm. odd. hmm.
02:56.28tronixfastest way is probably to reboot
02:56.29QwellWhat else is uring it?
02:56.32Qwellusing*
02:56.43tronixQwell: that's the problem... asterisk's stopped, lsof not turning up a match
02:56.47AsteriskNewbienothing .. as far as I can see ..
02:56.54AsteriskNewbieby the way .. I have zaptel starting from init ..
02:57.01Qwellrun the modprove grep again
02:57.10AsteriskNewbieok
02:57.21*** join/#asterisk cjk (n=cjk@80.92.64.103)
02:57.35cjkhi, does anyone know a way to do sip registration flood?
02:58.00AsteriskNewbieok .. i did /etc/rc.d/...zaptel stop and that stopped it ..
02:58.05tronixahh ok cool
02:58.12AsteriskNewbiemodprobe  -r zaptel .. nothing ..
02:58.15_Sam--sure...let us help you kill some sip servers
02:58.15AsteriskNewbieso now load zaptel?
02:58.18Qwellno
02:58.23Qwellwhat distro?
02:58.40tronix_Sam--: easier -- just plug in a couple 6-line 7960gs... ;)
02:59.34QwellAsteriskNewbie: What distro?
02:59.45AsteriskNewbieCentOS ..
02:59.47AsteriskNewbiesorry ..
02:59.56Qwelledit /etc/sysconfig/zaptel
03:00.05Qwellremove the uneeded modules
03:01.28Qwellthen start zaptel again
03:01.28AsteriskNewbieOk .. done ..
03:01.44QwellYou should have only wctdm and zaptel in there
03:01.51*** join/#asterisk justinu (n=Justin@cpe-72-129-86-208.socal.res.rr.com)
03:02.58bkw__Qwell, you up from some gang related J-walking?
03:03.08bkw__:P
03:03.15tronixAsteriskN: so you have removed all modules except for wctdm and zaptel from /etc/sysconfig/zaptel
03:03.18[av]banibleh, all the embedded asterisk distros seem to suck :/
03:03.19tronixand started up zaptel, right?
03:03.26AsteriskNewbieyes ..
03:03.35Qwellbkw_: ;]
03:03.40Qwells/_/__/
03:04.03AsteriskNewbieSorry .. brb. Gotta grab something ..
03:04.46tronixI guess you can try starting up asterisk now, when you're back
03:05.18tronixI would probably also check dmesg|tail and do 'cat /proc/interrupts | grep wctdm' first
03:05.29*** join/#asterisk j4m3s_ (n=j4m3s@24.96.145.117)
03:07.35*** join/#asterisk MikeJ[Laptop] (n=vircuser@d14-69-8-30.try.wideopenwest.com)
03:09.13*** join/#asterisk puzzone (n=Timo@e178122020.adsl.alicedsl.de)
03:11.50puzzoneHello *-lovers
03:12.07*** join/#asterisk junbug (n=bouboun@c-69-180-110-32.hsd1.fl.comcast.net)
03:12.15*** join/#asterisk mogorman (i=root@user-24-236-84-48.knology.net)
03:13.39puzzone*cry for help* is anybody familiar with asterisk on OS X? It's the "Bad file descriptor" prob...
03:13.57AsteriskNewbieback ..
03:14.10AsteriskNewbiedmesg says:
03:14.12AsteriskNewbieFound a Wildcard TDM: Wildcard TDM400P REV I (2 modules)
03:14.12AsteriskNewbieRegistered tone zone 0 (United States / North America)
03:14.25tronixAsteriskN: okay. that's reasonable.
03:14.32tronix(if you are indeed in the U.S.)
03:14.47tronixztcfg -vvv looks good still, too, right?
03:14.56AsteriskNewbiecat proc/interrupt shows:  11:    5154990          XT-PIC  EMU10K1, uhci_hcd, uhci_hcd, wctdm
03:15.04puzzone1.2.0 to 1.2.4
03:15.11tronixok that looks reasonable (re: /proc/interrupt)
03:15.20AsteriskNewbieYeah ..
03:15.22tronixnice that PCI can do irq sharing.
03:15.33AsteriskNewbieno kidding!
03:15.35tronix:)
03:15.41AsteriskNewbieso I guess I start asterisks and try again?
03:15.45tronixAsteriskN: ok I guess you can try starting up asterisk, yup
03:15.51puzzone:-{
03:16.04tronixpuzzone: I love OS X, but unfortunately, don't have * on that box :(
03:16.31Qwellkram: hey
03:16.32AsteriskNewbieDone ... now I will call my number :p
03:16.43tronixasteriskN: try 'zap show channels' first
03:16.51AsteriskNewbieok
03:17.10AsteriskNewbieNope .. same as before ..
03:17.11AsteriskNewbieChan Extension  Context         Language   MusicOnHold
03:17.12AsteriskNewbie<PROTECTED>
03:17.13AsteriskNewbie:(
03:17.36tronixhmm. how odd. I don't know about centos but it worked first time when I threw in the card , edited files, and started * on Gentoo. sorry. :(
03:17.43tronixI wonder if any error messages?
03:17.46tronixlike in a log file?
03:17.51puzzonethx tronix
03:17.55AsteriskNewbieHmm .. let me look again ..
03:17.57tronixon Gentoo, it logs to /var/log/asterisk/messages
03:17.58Qwellzap show channels won't show channels that aren't in use..
03:18.05copantlany body here use amportal?
03:18.10tronixahh hmm... Qwell: but he has 'em config'd
03:18.12Qwell~amp
03:18.15jbot[amp] NOT supported here! people using it should join #amportal
03:18.18Qwelltronix: So do I
03:18.22tronixhmm.
03:18.24tronixahh.
03:18.25tronixI see.
03:18.26AsteriskNewbieNoe ..
03:18.28AsteriskNewbienone ..
03:18.28tronixokay.
03:18.30AsteriskNewbieother than ..
03:18.32AsteriskNewbiedbus: Can't send to audit system: USER_AVC pid=2978 uid=81 loginuid=-1 message=avc:  denied  { send_msg } for  scontext=root:system_r:unconfined_t tcontext=user_u:system_r:initrc_t tclass=dbus
03:18.32AsteriskNewbieFeb  2 22:16:35 localhost last message repeated 7 times
03:18.32AsteriskNewbieFeb  2 22:17:40 localhost last message repeated 13 times
03:18.42QwellAsteriskNewbie: try calling it, see what happens
03:18.47tronixnot related. yeah agree with Qwell
03:19.33AsteriskNewbieHmm .. I just did ..
03:19.40AsteriskNewbiekeeps ringing and ringing ..
03:20.07tronixokay, guess that sounds like a dial plan issue (extensions.conf)
03:20.12tronixyou can shorten that file for testing
03:20.17QwellWhat's zap show status, say?
03:20.18AsteriskNewbieok ..
03:20.23*** join/#asterisk kimosabe (n=kimosabe@dsl-201-128-128-191.prod-infinitum.com.mx)
03:20.45kimosabeis there a way to configure sipura 2100 as bridge for qos purposes
03:20.53AsteriskNewbieescription                              Alarms     IRQ        bpviol     CRC4
03:20.53AsteriskNewbieWildcard TDM400P REV I Board 1           OK         0          0          0
03:21.09AsteriskNewbiethat's zap status
03:21.13QwellAsteriskNewbie: set verbose 10, you still don't see anything?
03:22.04tronixasteriskN: you are calling your inbound analog number?
03:22.13tronixand it is plugged in which port? #1 or #2?
03:25.29kimosabeis any one using the spa2100
03:25.33AsteriskNewbieyes ..
03:25.44AsteriskNewbieI am calling my .. we.. phone number ..
03:25.51AsteriskNewbieit is plugged into port two ..
03:26.01AsteriskNewbieMy fxo card (red colour) is on Tel 2
03:26.05tronixAsteriskN: okay. where do you want that call to go to?
03:26.23AsteriskNewbieNo Quell .. I don't see anything ..
03:26.33tronixAsteriskN: do you have a voip phone? or an analog phone? you need to tell asterisk which phone to ring
03:26.40AsteriskNewbieWell .. right now am not trying to do anything. I just want asterisk to pick the line when I call it ..
03:27.00AsteriskNewbieI've got s,1,answer() and then s,1,echo();
03:27.26AsteriskNewbieTronix: I have an analog phone .. but at this point am not even asking asterisk to ring anything .
03:27.38AsteriskNewbieI just want it to acknowledge the line ..
03:27.58tronixAsteriskN: Hmm. I guess your dial plan looks reasonable. you could change to exten => s,2,Dial(Zap/1)
03:28.02tronixand plug analog phone in line #1
03:28.07tronix(on the card)
03:28.20tronixthen do in asterisk *CLI: reload
03:28.39AsteriskNewbieok .. will do ...
03:28.47tronixfor [incoming]
03:28.55tronix(exten => s,2,Dial(Zap/1)
03:29.06tronixno ( before exten, I meant.
03:29.51junbugwhats a good billing solution these days
03:30.15AsteriskNewbieK .. done .. still no luck ..
03:30.22AsteriskNewbieringing .. ringing .. ringing ..
03:30.23AsteriskNewbie:(
03:30.49AsteriskNewbieby the way .. I have dsl on the same line ..
03:30.53AsteriskNewbieshouldn't make a diff, should it?
03:30.56QwellI'd recommend calling Digium..
03:31.12Qwellor if you're lucky, mogorman is around
03:31.21AsteriskNewbieQwell ..mogorman?
03:31.23mogormanim here
03:31.24AsteriskNewbiewho is that?
03:31.26Qwellhim
03:31.26mogormanwhats needed
03:31.29AsteriskNewbie:)
03:31.33tronix:)
03:31.46AsteriskNewbiemogorman .. do you work with digium??
03:31.52[TK]D-FenderQwell : My only suspicion is his modules are reversed...
03:31.58mogormanfor a year and half now ^_^
03:32.02AsteriskNewbie:)
03:32.04Qwell[TK]D-Fender: ztcfg shows it fine
03:32.16QwellAsteriskNewbie: What signalling are you using?
03:32.26AsteriskNewbieks ..
03:32.29[TK]D-FenderQwell : Will it show that the physically installed module isn't the right type?
03:32.29Qwellwhat ks?
03:32.37Qwell[TK]D-Fender: thought so
03:32.48mogormanwhats the prob bob
03:32.56mogormanwhats the prob bob
03:32.57QwellAsteriskNewbie: show me the "signalling=" line from the config
03:33.03tronixQwell: he's got a FXS card in #1, FXO in #2, and fxoks=1, fxsks=2
03:33.05AsteriskNewbiefxo_ks for channel1 and fxo_ks for chan 2
03:33.13AsteriskNewbieok ..
03:33.19Qwellfxo_ks for both?
03:33.39tronixhttp://63.250.96.18/asterisk/zaptel.conf
03:33.39AsteriskNewbiefxoks=1
03:33.39AsteriskNewbiefxsks=2
03:33.39AsteriskNewbieloadzone=us
03:33.39AsteriskNewbiedefaultzone=us
03:33.46Qwellk
03:33.56Qwellgreen in 1, red in 2?
03:34.05AsteriskNewbieyes!
03:34.16AsteriskNewbieand incoming telco line plugged into red ..
03:34.19Qwellphone plugged into green, line into red?  k
03:34.28AsteriskNewbieYes!
03:34.42Qwellsomething is fishy
03:34.51tronixdsl freq interference?
03:35.00Qwellno, it'd be the other way around
03:35.15AsteriskNewbieHmm .. I did unplug the dsl last nite to test .. still no luck ..
03:35.55AsteriskNewbiebeats the hell out of me ..
03:36.06QwellYou don't get anything at all on the CLI?
03:36.20AsteriskNewbieNope! Nothing at all!
03:36.27mogormanwhat are you trying to do AsteriskNewbie just get  a tdm card working?
03:36.33Qwellmogorman: yeah
03:36.40AsteriskNewbieyes .. morogman ..
03:36.43mogormanif you give me ssh i can fix it in 5 minutes
03:36.56mogormanand you can watch in screen
03:37.02Qwellmogorman: Miss doing support yet?  heh
03:37.10AsteriskNewbieok mogorman ..
03:37.19mogormanheh i still have some special customers i take care off
03:37.26Qwelloh
03:37.43AsteriskNewbiemogorman ..
03:37.53AsteriskNewbieip is 63.250.96.18 ..
03:37.59mogormanwait wait
03:38.00mogormanwait
03:38.02mogormanmessage me
03:38.08mogormandont put it in open channel....
03:39.59DaminAm I wrong for thinking that this woman is TOTALLY HOT? http://www.eden38dd.com
03:40.40QwellDamin: Needs a NSFW header
03:40.41[TK]D-FenderDamin : No... you're just wrong.
03:41.04mogormandamin, next time nsfw
03:41.23Triffid_HunterDamin: i don't find her attractive in the slightest
03:41.23[TK]D-Fendernsfw?
03:41.28Qwell~nsfw
03:41.30jbotwell, nsfw is not safe for work
03:41.48Qwelljbot: no, nsfw is not safe for work/wife
03:41.49jbotokay, Qwell
03:42.07mogormanlol qwell
03:42.10Qwell:D
03:42.32mogormanheh i was looking at o'rielly's photos of etel
03:42.37QwellI think nsfw should be a tld
03:42.41mogormanand there was you looking like a nubb
03:42.44Qwell:(
03:42.45mogorman^_^
03:42.52mogormanjoking
03:43.02mogorman~tld
03:43.03jboti guess tld is top level domain. .com, .org, .edu, and so on; also includes the country top-level domains .au, .it, .cx and so on, or see http://www.iana.org/cctld/cctld-whois.htm for a complete listing
03:43.07mogormanahh
03:43.15mogormanhey AsteriskNewbie you gonna message me?
03:43.27AsteriskNewbieI did already ..
03:43.31AsteriskNewbiedidn't u get it ..?
03:43.34mogormannever got it
03:43.47mogormanare you authenicated to freenode?
03:43.49QwellAsteriskNewbie: /msg nickserv register
03:43.55mogormanyou have to be authed to send me stuff
03:44.11Qwellor /msg nickserv help register
03:44.36tronixAsteriskNewbie: do what Qwell suggests. it will help you register with freenode... then freenode will let you /msg mogorman
03:44.54AsteriskNewbieregister otutuaka
03:45.02mogormanlol
03:45.07tronixerr I would suggest a different password :-)
03:45.08mogormanalmost
03:45.13AsteriskNewbie:)
03:48.06*** join/#asterisk propagandhi (n=opera@CPE-144-131-132-107.nsw.bigpond.net.au)
03:48.19propagandhihey people
03:48.25tronixtop o' day
03:48.29propagandhican someone give me some advice on a caller ID problem
03:49.06tronixwhat's the problem?
03:49.08tronix(curious)
03:49.35*** join/#asterisk bmg505 (n=leon@dsl-146-50-61.telkomadsl.co.za)
03:49.45propagandhibasically I have two types of phones, I have a sipura SPA-841 and another generic SIP desk phone, bought cheap by my boss
03:49.58propagandhicaller id on queues works on the sipura
03:50.02propagandhibut not on the other
03:50.19tronixhmm, interesting.
03:50.52propagandhihowever, if i use SetCallerID("name_here"${CALLERIDNUM}) - the crappier phones display the word "asterisk"
03:51.21Nuggetthat's asterisk doing that, not the phone.
03:51.34Nuggetif you try to use a malformed callerid string, it'll fall back on "asterisk"
03:52.22mzoyay uptime is a week now
03:52.26propagandhiNugget: yeah, I agree, but if i use SetCIDName("name_here"${CALLERIDNUM}) it shows only the callers ID
03:52.41propagandhiand discards any other text
03:52.50NuggetI lost my 650 day uptime to a failed power supply last week.  Pity, I was hoping I'd break my record.
03:53.09rob0:(
03:53.10mzohaha, =(
03:53.21Nuggetpropagandhi: well you are using setcidname.  maybe you should read more about the callerid functions.
03:53.27mzowell i'm still happy to get over five day uptime, whenever i goes over a week im like 'omg, i didn't break it yet, yay'
03:53.50propagandhiNugget: I have been reading, I assure you, I thought perhaps someone may have had the same issue and overcome it
03:53.56*** join/#asterisk Qwell (n=north@unaffiliated/qwell)
03:54.03Nuggetone way to overcome it is to use the command properly.  ;)
03:54.14Nuggetwhat you're doing is malformed
03:54.22brockj49464What is the quickest way to take a call and send to an ext?
03:54.23propagandhiNugget: I appreciate the sarcasm, but how come it works perfectly on sipura and softphones
03:54.30Nuggetaccident.
03:54.33mzo*cough* feature *cough* :P
03:54.38[TK]D-FenderNugget : Don't worry... you're always a broken record for us ;)
03:54.46Nuggetmysql sucks!  linux is poo!  ;)
03:54.53[TK]D-Fender(j/k)
03:55.00mogormanits fixed
03:55.33Nuggetyou should be using setcallerid() not setcidname() if you really want to push both the name and number in
03:55.49propagandhiNugget: and as I've said, I've used both of those functions
03:55.51Nuggetsetcidname isn't expecting that number, so I'm not surprised you're getting erratic results.
03:56.19propagandhiNugget: and they arent erratic, they're 100% reproducable, it works in every case on the sipuras or on softphones
03:56.38Nuggetyou also lack a space between the name and number, and you're not setting the number in angle brackets as you're supposed to
03:56.54AsteriskNewbieOk .. mogorman fixed it .. :)
03:56.56Nuggetyou're sending malformed text.
03:57.08NuggetI'm not surprised that some phones are not parsing it properly
03:57.10AsteriskNewbiethanks Tronix. And Qwell. And D-Fender
03:57.29tronixAsteriskN: that's great news to hear :) what was the fix?
03:57.55propagandhiNugget: http://www.voip-info.org/tiki-index.php?page=Asterisk%20cmd%20SetCallerID
03:58.04austinnichols10I just found that I can set outbound caller name to anything I want with my telasip account.  Anyone know how they're pulling that off?
03:58.10Nuggetyes.
03:58.11[TK]D-FenderAsteriskNewbie : So what was it?
03:58.19Nuggetyou aren't doing what that page says to do.
03:58.23mogormantypo
03:58.33tronixahhh :)
03:58.39AsteriskNewbiehehehehe ..
03:58.44tronixe-z fixes = good
03:58.49propagandhithe number is to be enclosed in { }
03:58.55AsteriskNewbieturned out I had two [channels] on extension.conf ..
03:58.57AsteriskNewbie:)
03:58.59[TK]D-Fender!
03:59.00tronixahhh! :)
03:59.02NuggetNo.  That's not what it says.
03:59.06propagandhiyes it does
03:59.10propagandhithats the variable
03:59.11NuggetNo.  It really doesn't.
03:59.31NuggetYou should be doing: setcallerid("name here" <${CALLERIDNUM}>)
03:59.43Nuggetthe angle brackets are there on purpose.
04:00.07Nuggetthe curly brackets are just how variables are used anywhere in the dialplan.  That isn't specific to this command
04:00.25propagandhiNugget: I will add those angled brackets now and test
04:00.33Nuggetand put a space in between too.
04:00.56Qwellmeh, Set(CALLERID(num)=5551212) and Set(CALLERID(name)=bob)
04:01.13tronixQwell: do these work prior to v1.2?
04:01.23Qwellno, upgrade
04:01.39propagandhiNugget: I added the space also
04:02.07Qwellmogorman: What is ABE based on, 1.0?
04:02.33propagandhiNugget: going to test
04:03.09*** join/#asterisk _Soul_ (n=Soul@87.196.0.53)
04:04.01*** join/#asterisk X-Rob (n=rob-x@dsl-202-173-151-24.qld.westnet.com.au)
04:04.54tronixg'day mate!
04:07.16mogormanpost 1.0 pre 1.2
04:07.40*** join/#asterisk Skkip (n=Skipper@216.160.91.91)
04:07.54propagandhiNugget: does the name need to also be enclosed in " "
04:08.05Nuggetno, but it's good form to do so.
04:08.19Nuggetthe code can figure it out, even without the quotes, as long as the angle brackets are there
04:08.50propagandhiNugget: exten => 984,2,SetCIDName(984-MacDonald International- <${CALLERIDNAME}>)
04:08.58propagandhidoes that look valid to you
04:09.11Nuggetno, absolutely not.
04:09.29propagandhiNugget: advice?
04:09.34Nuggetwhy are you trying to put CALLERIDNAME where the number belongs?
04:10.10propagandhiNugget: sorry thats meant to be CALLERIDNUM
04:11.24X-RobSetCIDName is deprecated. Use Set(CALLERID(name)=...)
04:11.27austinnichols10Nugget: when you send that on an outbound call, won't the telco override the name based on what's in the number?
04:11.57propagandhiwe have an ISDN link, it sends us the DID and I do inbound routing to translate it to a queue
04:12.03Nuggetif your provider lets you spoof your callerid, you can only do the number portion.
04:12.14propagandhithis is all for inbound
04:12.20Nuggetonce it hits the PSTN, the calleridname will be whatever legitimately corresponds to the number you use
04:12.34austinnichols10nugget: that's what I always thought.
04:12.56propagandhithis isnt for outgoing caller id
04:13.05austinnichols10nugget: why I'm asking is I just got a telasip account and I can spoof NAME but not number.
04:13.21austinnichols10nugget: and I'm scratching my head trying to figure out how they pulled that one off
04:13.56austinnichols10nugget: I just placed a couple of calls under the alias of 'Gene Simmons'
04:14.35mzohaha, that souds fun to spoof calls with ;)
04:14.37austinnichols10they must have some kind of special setup with the telco or be recognized AS the telco
04:16.33[av]banianyone here with a gxp2000?
04:16.39propagandhiNugget,X-Rob: does this look ok
04:16.40propagandhiexten => 999,2,Set(CALLERID(name)=999-StaffDetails)
04:17.00X-Rob'-' can be taken to be a delimiter in some cases.
04:17.07X-Robtry using just a space
04:17.22propagandhiX-Rob: ok, cool
04:17.37propagandhitesting
04:18.32*** join/#asterisk _8ball (n=spunk@64-13-46-242.eug.clearwire-dns.net)
04:23.30propagandhidammit
04:23.31*** join/#asterisk r0d3nt (i=nobody@wsip-24-234-241-145.lv.lv.cox.net)
04:25.54dudesAnyone know what a Dual G4 450Mhz /w 512MB Ram can handle /w *
04:27.29mogormancalls
04:27.35tronixnever tried it but i'd guess 10-15 calls?
04:27.40mogormanit really depends on what you are doing
04:27.43mogormanmuch more than that
04:28.03mogormanlike maybe 15 g729 calls
04:28.08mogormanit really depends on what your doing
04:28.19mogormanlike i run my phone system off a really shitty mips box
04:28.20dudesBasic PBX shit
04:28.25mogormanbut it only does a call or to
04:28.32mogormanwell basic pbx shit ip?
04:28.36mogormantranscoding?
04:28.40mogormanor hardware?
04:28.52dudesg729 to law ... max 9 channels
04:28.53mogormanand is this box only doing asterisk?
04:29.08mogormani think it could handle it
04:29.16dudesOnly *, and I'm only thinking about make a PBX out of it
04:29.19mogormang729 to ulaw is the most intensive thing
04:29.26mogormanyou could test that real eas
04:29.26mogormany
04:29.32mogormanjust buy a g729 lic.
04:29.37mogormanits only 10 bucks for a channel
04:29.42mogormansee the load on a machine with that
04:29.51dudesMy 850Mhz Athlon can handle g729 to ulaw but they sometimes scratch up a bit
04:30.07dudesAnd it's annoying, so thinking maybe the Mac could do better
04:30.12mogormanwell i do g729 on my mips
04:30.15mogormanfor grins
04:30.18mogormanbut only one channel
04:30.34wunderkinmogorman loves to run asterisk on his toaster
04:30.53mogormanyeah
04:30.55mogormani do
04:30.59mogormanas its cheap
04:31.03mogorman35 watt
04:31.06dudesI'd think Dual G4 would should ideally smokie a 850 Athlon
04:31.08filemogorman: you cheap dingo you
04:31.36mogormanheh
04:31.40tronixdudes: probably could, if it used altivec stuff
04:31.46mogormanwell it keeps me from messing my phones
04:31.55mogormanbecause its such a pain
04:32.02mogormanso most of the time my pbx works ^_^
04:32.07[av]baniwhat hardware?
04:32.12[av]banirouterboard?
04:32.22mogorman200mhz cobalt raq 2
04:32.26[av]baniewww
04:32.28[av]baniheheh
04:32.30mogormani run everything off of it
04:32.38mogormanapache, jabber, mail, samba,ssh
04:32.39[av]banirun and 200mhz do not go together
04:32.40mogormanfirewall
04:32.44[av]baniwalk is more like it
04:32.45[av]banior stroll
04:32.49wunderkinthat would be funny to make, an ip enabled toaster
04:32.52mogormanheh load is always at 3
04:32.59mogormanbut it does everything i want it too
04:33.00[av]bani200mhz mips does not 'run'
04:33.05mogormanand only set me back 50 bucks
04:34.05wunderkinthere are ip fridges
04:34.42mogormanheh i saw a toaster running netbsd but it was faster than my box
04:35.57[av]banimogorman: looked at gumstix?
04:36.09mogormanexpensive....
04:36.14[av]bani$160 ?
04:36.30[av]baniyou can run it off AA batteries :D
04:36.50mogormani can run mine off my car battery
04:36.50Triffid_Hunterif i estimate correctly, they're about the size of a cigarette lighter too
04:36.54*** part/#asterisk propagandhi (n=opera@CPE-144-131-132-107.nsw.bigpond.net.au)
04:37.01mogormanthey are a littbigger
04:37.10mogormani saw kristians
04:37.15wunderkini cant even find it on their site :(
04:37.36[av]banigumstix mocks your 35W monster
04:37.46mogormani have been thinking about running my asterisk box off a sharp zaurus or something
04:38.09mogormanso i could have long battery life if power dies
04:38.12[av]baniheh, again, run and zaurus are not two things which go together
04:38.14Triffid_Huntergumstix+audiostix+cfcard = cheap mp3/ogg/whatever player.. am planning on grabbing one or two or three when I can afford em
04:38.18[av]banii had a zaurus. slooooow
04:38.31mogormani had one for a bit
04:38.44[av]baninice hardware, too bad it wasnt faster
04:38.51[av]banii could almost play xmame on it
04:38.59mogorman???
04:39.03mogormani played snes on it
04:42.01austinnichols10good fast cheap - pick any two you want
04:42.17mogormangood and cheap ^_^
04:43.17*** join/#asterisk austinnichols10 (n=austinni@dsl-10-169.cofs.net)
04:44.56*** join/#asterisk jaike (n=a@203.131.137.76)
04:45.32jaikeno 1.2.5? hehe
04:46.00*** join/#asterisk kimosabe (n=kimosabe@dsl-201-128-128-191.prod-infinitum.com.mx)
04:46.27kimosabedoes any one know the best setting in sipura 2100 for Qos
04:46.28*** join/#asterisk Defraz (n=t0tal@24-119-94-19.cpe.cableone.net)
04:48.32[av]banidepends on your individual network
04:48.36[av]baniask your isp
04:49.44kimosabeim pirating my internet from wireless lans
04:49.55kimosabebut i have a clear shot
04:50.00kimosabeiu have 4 conections
04:55.42*** join/#asterisk IOscanner (n=IOscanne@c-24-0-183-125.hsd1.tx.comcast.net)
04:56.30IOscannerhas anyone been able to build zaptel drivers on 2.6.15.2 kernel
04:57.15IOscannerI get an error: too few arguments to function `class_device_create`
04:58.07Triffid_HunterIOscanner: i've built 1.2.1 zaptel drivers successfully, but earlier versions gave me that error
04:58.40IOscannerwill the 1.2.1 drivers work with the older 1.0 asterisk
04:58.51IOscannerI am not ready to upgrade asterisk yet
04:59.11Daminfile: You guys just have a routing anomoly?
04:59.32IOscannereverything works fine and I know I am going to have to change some code to move to 1.2
04:59.47jaikeany commpartners clients here?
05:00.46*** join/#asterisk newsmafia (n=paulr@wsip-68-15-19-142.sd.sd.cox.net)
05:02.17*** join/#asterisk dca[laptop] (n=dca[lapt@c-67-166-23-243.hsd1.co.comcast.net)
05:02.52*** part/#asterisk dca[laptop] (n=dca[lapt@c-67-166-23-243.hsd1.co.comcast.net)
05:05.01IOscannerthat is in v1-2 to download 1.2 correct?
05:08.07blackremedydoes anyone know if it's worth it to upgrade to asterisk@home 2.4 from 2.3?
05:08.26*** join/#asterisk websae (n=icechat5@h69-129-132-18.69-129.unk.tds.net)
05:08.36websaeanyone here familiar with e911 services?
05:09.03*** join/#asterisk Kari1 (n=Karim@243.209-89-66-0.interbaun.com)
05:12.41*** part/#asterisk Kari1 (n=Karim@243.209-89-66-0.interbaun.com)
05:15.26litageis there a particular reason why /var/lib/asterisk is chmod'd 0700?
05:16.34rob0Mine isn't ...
05:18.19konfuzedslePP: eh there
05:18.37*** join/#asterisk kimosabe (n=kimosabe@dsl-201-128-128-191.prod-infinitum.com.mx)
05:18.48kimosabeis any one using sipura 2100 for qos isues
05:23.44IOscanneranyone know if zaptel-1.2 drivers will work with 1.0 asterisk?
05:23.47*** join/#asterisk GeneG (n=GeneG@toronto-HSE-ppp4160385.sympatico.ca)
05:23.53litagerob0: what permissions and ownership do you have on /var/lib/asterisk/ ?
05:24.07rob0755 root:root
05:24.51litagethanks
05:28.45GeneGQuestion about IAX softphone -> Asterisk -> Unlimitel VoIP provider. I'm getting an "unable to transfer" error message when Asterisk attempts to native bridge the softphone and the remote telephone number. Any ideas where I can start looking for the problem?
05:31.44litagedoes ooh323c that comes in asterisk-addons require h323.conf or ooh323.conf ?
05:35.07kc5cqmhowdy.  My callerID coming of my asterisk box isn't setting the timestamp on the callerID phones.
05:35.31kc5cqmany ideas?
05:35.42kc5cqmthe CID info is getting passed correctly
05:35.47kc5cqmjust not the date/time
05:36.09*** join/#asterisk tuxinator_linux (n=Jone_Doe@70-32-106-248.ontrca.adelphia.net)
05:37.15*** join/#asterisk anderiv (n=anderiv@207-67-87-34.gen.twtelecom.net)
05:45.31*** join/#asterisk mattwj2005 (n=Matt@dialup-4.159.59.85.Dial1.Chicago1.Level3.net)
05:46.29mattwj2005*** holds up a sign "Will work for bandwidth"
05:46.50konfuzedmattwj2005: how much bandwidth
05:47.14mattwj2005lol
05:47.20mattwj2005cable modem or dsl
05:47.47mattwj2005dial up and asterisk are not the best of friends
05:48.17mattwj2005but it does work
05:48.22mogormang729
05:49.15konfuzedmattwj2005: i can provide DSL anywhere in North America
05:50.11mattwj2005Ladysmith, WI?
05:50.34konfuzedmattwj2005: but what are you capable of doing? - Most here would likely say that I cant do very much myself ;^)
05:51.47konfuzedif your phone copper runs to a RBOC with ADSL in place then I can have the line activated and service provisioned
05:52.53mattwj2005I have no problem getting DSL or cable modem.....it is just the cost
05:53.14mattwj2005nm me.....I am just missing the high school of the university
05:53.22mattwj2005*high speed
05:53.32*** part/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net)
05:55.04mattwj2005at the university I was able to run ulaw and alaw codecs.....now I am dealing with gsm and ilbc
05:56.39*** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk)
05:56.43konfuzedso what is the cost of standard DSL in LadySmith, is that 3mb down ?
05:57.04mattwj2005about $80 per month
05:57.13mattwj2005USD of course
05:57.40*** join/#asterisk brookshire[home] (n=matt@68.62.235.16)
05:58.28kc5cqmthats pretty high
05:58.37mattwj2005I don't know if that is right
05:58.59mattwj2005you guys are misunderstanding me....I don't live in Ladysmith, WI.....I live in a bigger city
05:59.01Flautosbc charges 14.95
05:59.30mattwj2005what type of contract?
05:59.40Flautodsl
06:00.11*** join/#asterisk tuxinator_linux (n=Jone_Doe@70-32-106-248.ontrca.adelphia.net)
06:00.44mattwj2005my housing in temporary
06:01.02mattwj2005so pretty much dsl and cable are out of the question
06:01.02*** join/#asterisk Assid (n=assid@203.115.64.14)
06:01.29Flautoyou need wireless?
06:01.33Flautohehe
06:02.13mattwj2005I need Ethernet ;)
06:02.15mattwj2005lol
06:02.24Assidi need net
06:02.26*** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net)
06:02.28mattwj2005give me Ethernet or give me death
06:02.29mattwj2005:P
06:03.20kc5cqmI'm spoiled by gbit at work
06:03.39tuxinator_linuxsoo, I have a cisco 7960, do I need 'CON-SNT-PKG1' ($$$) or can I get away with 'CON-SNT-CP7960' ($), for getting the SIP files?
06:03.39mattwj2005you know the prefect stuff.....0 ms jitter......1000 Mbps
06:04.09Qwelltuxinator_linux: get the cheapest smartnet contract you can find
06:04.11kc5cqmford prefect?
06:04.15brookshire[home]that's hot
06:04.25konfuzedmattwj2005: so you're just stuck with on connectivity options then, i understand your lament
06:04.34konfuzeds/on/no/
06:04.38brookshire[home]everyone needs a couple gig drops to their house
06:04.49Qwellbrookshire[home]: What, you don't?
06:04.59konfuzedoh way to gojbot
06:05.00brookshire[home]heh
06:05.18tuxinator_linuxQwell: I know everybody asks that question, I just don't have my logs on this machine.  I found this one, http://www.cdw.com/shop/products/specs.aspx?EDC=672205
06:05.19brookshire[home]i can only afford the syncronous 10mbit wireless ;)
06:05.52mattwj2005I am only stuck at 100 Mbps Ethernet and 54 Mbps Wifi......that doesn't go anywhere :P
06:05.56mattwj2005lol
06:06.50mattwj2005I have a good lan but man I need a wan
06:07.01brookshire[home]doesn't everyone?
06:07.16mattwj2005yeah definitely
06:07.17brookshire[home]in 2 years everyone will have fiber
06:07.19brookshire[home]:D
06:07.21mattwj2005especially in America
06:07.27mattwj2005I hope so
06:07.37brookshire[home]it's already fiber to the pole
06:07.41tuxinator_linuxA bunch of cities in Utah have fiber to the house
06:07.42kc5cqmwhat sip/iax services is everyone using around here?  I've got fwd in/out (iax2), stanaphone in/out (sip), voicepulse in/out (iax2), iconnecthere out (sip),  packet8 in/out (zaptel),  generic sip catchall
06:07.57brookshire[home]why utah
06:08.07wunderkinutah? people live there? wow..
06:08.15tuxinator_linuxI was born there
06:08.21brookshire[home]amazing ;)
06:08.25tuxinator_linuxbut I'm in SoCal now
06:08.25kc5cqmfiber to every block is ideal.  bpl is bad...
06:09.06tuxinator_linux~pbl
06:09.22Qwellkc5cqm: asterlink and nufone
06:09.27kc5cqmhmmm
06:09.36mattwj2005"In a prefect world, every computer would have a connect to every other computer using fiber"
06:09.37kc5cqmI'm looking to get rid of packet8 asap
06:09.47kc5cqmbut would like to port the number
06:09.53tuxinator_linuxI'm not have great service with Broadvice
06:09.56wunderkinbpl is broadband over the power lines
06:10.03brookshire[home]i like voicepulse
06:10.04wunderkinjbot!
06:10.13brookshire[home]but.. they are really the only one with a number in my area
06:10.17kc5cqmdoes newphone still run open callerid?
06:10.32tuxinator_linuxwunderkin, thanks, I remember learning about that a while ago, must reak havok on UPSs
06:12.05mattwj2005brook what do you like about voicepulse?
06:12.21mattwj2005besides the number thing of course
06:12.33brookshire[home]it works!
06:12.38kc5cqmQwell, asterlink's 2c/minute is about what I'm paying with voicepulse...not bad
06:12.53kc5cqmnice that it's iax-based
06:13.19mattwj2005do you get a number and unlimited outgoing?
06:13.23brookshire[home]voicepulse is iax based too
06:13.33brookshire[home]if you get it as voicepulse connect
06:13.35kc5cqmmattwj2005, not on voicepulse... I use packet8 for that.
06:14.15*** join/#asterisk EriSan (n=erisan@81-174-35-6.f5.ngi.it)
06:14.15mattwj2005I actually have been looking at broadvoice
06:14.22mattwj2005what do you guys think of that?
06:14.32kc5cqmthey're SIP only aren't they?
06:15.03mattwj2005yeah it sucks in that regard, but they give you a bunch of free countries
06:15.20kc5cqmvoipbuster has a ton of free outbound
06:15.29kc5cqmI might subscribe
06:15.44tuxinator_linuxmattwj2005: their service has gotten better, I have two accounts with them
06:16.45kc5cqmanyone mess with voipbuster?
06:17.02tuxinator_linuxmattwj2005: but it dropps calls on my wife sometimes (broadvoice)
06:17.03mattwj2005what would be prefect for me is....unlimited inbound and outbound for a low flat rate....and be iax based
06:17.19mattwj2005thanks for the heads up tux
06:17.20Assidwhy not just use sipdiscount
06:17.21tuxinator_linuxwhat's wrong with SIP?
06:17.47tuxinator_linuxI'm all for IAX but SIP works well enough
06:18.03mattwj2005I like the firewall ease
06:18.28tuxinator_linuxI just do port forwarding
06:18.44mattwj2005I prefer to make my own sip channels and have all my inbound/outbound based on iax
06:18.46tuxinator_linuxbut that is a nice feature
06:19.04tuxinator_linuxWife's calling, be back in a bit
06:22.14mattwj2005brook what don't you like about voicepulse?
06:23.24brookshire[home]i dunno.. never had a problem
06:23.38brookshire[home]support is kinda slow
06:23.43brookshire[home]takes about a day or so to get back
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06:25.10mattwj2005sounds godo
06:25.12mattwj2005thanks :)
06:25.15mattwj2005*good
06:25.41junbugwhats a good billing solution these days
06:26.35kc5cqmactually I get fine quality on both iax and sip
06:26.41kc5cqmsip is just more of a pain portwise
06:28.22elgwhy is asterisk asking for proxy auth when I have type=peer only?
06:28.25elgsip.conf
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06:35.08mattwj2005what the heck is a adpcm codec?
06:35.09mattwj2005:P
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06:37.48argos73anyone using a multitech mt5600zdx with caller id?  i'm in hell...
06:38.10mattwj2005thoses argos some cold water
06:38.14mattwj2005*throws
06:38.23argos73ahh - refreshing...:)
06:39.07argos73these things (multiple) simply refuse to report caller id...  they accept the at#cid=1, but no joy...
06:40.32mattwj2005why do you think the modem is causing caller id problems?
06:42.29argos73long and complicated, but I'm trying to get the modem to report CID on incoming calls to route faxes around...  works nicely on a MT5600BL, just not on the ZDX
06:43.08argos73know it's not the phone line or signalling - moved the line to a different modem and it works
06:44.32kc5cqmanyone know about getting asterisk to send a timestamp with the callerid signal?
06:44.38kc5cqmmy CID works except for the timestamp
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06:45.58argos73should just use the time from the system clock
06:46.00argos73just tried it
06:46.15kc5cqmall my caller ID's just report 12:00 am for every calltime
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06:46.21argos73DATE = 0203
06:46.21argos73TIME = 0145
06:46.30kc5cqmI'm thinking its a config problem with asterisk
06:46.38kc5cqmbut I've not been able to locate anything on it
06:47.19mattwj2005man oh man
06:47.23mattwj2005time for bed
06:47.27mattwj2005peace everyone
06:47.42tuxinator_linuxnight mattwj2005
06:49.47argos73there's a bunch of knobs in zapata.conf that control it - don't think i had to change anything for US devices, but something could have gotten mangled
06:50.31argos73don't see anything about timestamps, though
06:50.38litageif you direct CDRs to be sent to the Manager interface, how do you grab each CDR as it comes in?
06:51.27argos73you'd need some sort of background job running to grab them
06:52.24kc5cqmcertainly worth a look
06:56.20kc5cqmok, time for sleep...gnite y'all
06:56.23*** part/#asterisk kc5cqm (n=kc5cqm_@cpe-68-206-116-214.stx.res.rr.com)
07:02.21SkramXsupport run # asterisk
07:02.21SkramXsupport run # asterisk -r
07:02.21SkramXUnable to connect to remote asterisk (does /var/run/asterisk.ctl exist?)
07:02.23SkramXeh?
07:02.39SkramXthat doesnt start asterisk anymore? heh.
07:04.55SkramXeh?
07:06.20benjkit never did
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07:09.00SkramXWARNING[22376]: chan_iax2.c:9553 load_module: Unable to open IAX timing interface: No such device or address
07:09.04SkramX:(
07:10.45ThaZZa_WorkSkramX: Are you loading a timing device?
07:10.51SkramXshould be..
07:10.56SkramXhrmm
07:11.14ThaZZa_WorkSkramX: Have you loaded zdummy?
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07:11.49SkramXloaded
07:11.51SkramXnevermind
07:12.39ThaZZa_WorkSkramX: Fix your problem?
07:12.42SkramXYes'Sir
07:14.46sipperanyone know if the requirements for bison have changed for 1.2.4?
07:14.51ThaZZa_WorkSkramX: and ref to asterisk -r that is to connect to an existing console. i.e. asterisk is already running.
07:15.34sipperI'm getting some weird parsing errors Feb  3 00:13:52 WARNING[23665]: ast_expr2.fl:183 ast_yyerror: ast_yyerror(): syntax error: syntax error, unexpected TOK_EQ, expecting TOK_MINUS or TOK_COMPL or
07:15.50SkramXThaZZa_Work: I knpow
07:16.10SkramXhow do i uninstall a res_config_mysql.c? just remove it?
07:16.17SkramX...Feb  3 07:15:35 ERROR[29211]: res_config_mysql.c:615 mysql_reconnect: MySQL RealTime: Failed to connect database server  on . Check debug for more info.
07:16.20SkramXFeb  3 07:15:35 WARNING[29211]: res_config_mysql.c:450 load_module: MySQL RealTime: Couldn't establish connection. Check debug.
07:16.25SkramXI dont want any mysql stuff as of yet..
07:18.15ThaZZa_WorkSkramX: Look at the modules.conf on voip-info.org
07:18.51SkramX<PROTECTED>
07:18.58SkramXis not anywhere in modules.conf
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07:19.46ThaZZa_WorkSkramX: Research mysql cdr and res_config on voip-info.org   The info is there.
07:20.15ThaZZa_WorkSkramX: You are autoloading all modules. so you want to stop * loading a couple of mod's
07:20.21SkramXokay
07:20.22SkramXright on
07:20.46SkramXI havent upgraded in a while (using 1.0.9) so okay
07:23.03ThaZZa_WorkHome Time.. Back soon.
07:23.14SkramXi did noload and i still get errors
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07:24.52redaxgood monring,
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07:25.28websae$8/month UNLIMITED INCOMING with DID
07:25.36moreecehas anyone worked with ASTERISK and routing to different networks, through firewalls and onto different LAN subnet masks?
07:25.42redaxcan I use newlines in SetCIDName ?
07:26.09SkramXredax: nope
07:26.19redaxaha
07:26.23moreeceI assume is relatively simply with firewalls etc, simply using port redirection and assinging the STP and RTP (for sip) port open?
07:26.35redaxand how long should be the CIDName ? (how many chars)
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07:28.43moreecealso within the console is there a simply command that updates asterisk via the internet?
07:29.18Qwellmoreece: svn up, if you're using a svn version
07:29.57moreecesorry Qwell, what is a svn verision?
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07:31.51redaxif I set the caller name with SetCIDName(), and the number with SetCIDNum(),
07:31.55redaxat ringtime the ISDN phone shows the Name
07:31.57redaxbut when you pick up the phone from that, you see the number :)
07:32.01redaxstrange
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07:34.18moreecelol
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07:34.18moreecewoot woot
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07:47.32Ahrimanesanyone using astmanproxy on freebsd?
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08:04.22routerguymornin' folks
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08:09.31thazzaAha.. That was such a wonderful nice flood.
08:11.57routerguyhi all...anybody got a clue why an iaxy works fine in every other respect, but doesn't ring my analog phone?
08:12.01droopsnetsplit?  on freenode?
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08:12.57droopsas in the phone doesnt ring, or it doenst ring or send voice to it
08:13.13duckeoIf a incoming PSTN phone call continues to make my extensions ring, is it likely a Asterisk setting or an ATA setting?
08:13.14routerguyjust no ringer
08:13.27duckeo(Using an ATA for both the PSTN incoming and FXO for the handset)
08:13.33routerguycan answer the call, place calls, etc.
08:13.57droopsand the ringer is turned on at the phone?
08:14.18routerguyyeah...plugged it back in to a pots line to make sure
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08:14.36droopsthen you are past me, i havent set up an iax yet
08:15.20routerguythanks anyway
08:15.59routerguyain't much in the config to tweak, so I'm kinda baffled
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08:43.37fenlandergood morning
08:45.06tuxinator_linuxmorning
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09:03.15movergmorning dudes
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09:13.09robin_szOK  .. anyone used cicso ATA 186 with SIP .. where the fsck do you set the IP of the server huh?
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09:16.43thazzarobin_sz:  I have one of those at home.
09:17.01thazzarobin_sz: Shall have a look for you at the config.
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09:32.07*** join/#asterisk valste10 (n=valste10@adsl-216-133.38-151.net24.it)
09:32.38valste10hi...
09:33.23*** join/#asterisk areski (n=areski@polar.es6.egwn.net)
09:34.00valste10i would try to connect my asterisk with a te110p digium board to a Dialogic d300 board but i have a problem with the protocol...any ideas??? tks, vale
09:34.00Ahrimaneshey areski
09:34.31*** join/#asterisk _Sam-- (n=sam@dca.kneedraggers.com)
09:35.27*** join/#asterisk _di (n=disnider@169-115-207-82.static.ukrtel.net)
09:36.44dudesvalste10 - try restarting the dialogic box ...
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09:38.21valste10i had try...
09:38.30areskiAhrimanes, good morning !
09:38.45valste10the problem that i think is: protocol...cable...
09:39.16Ahrimanesareski: man i was at the office at 6.45 today
09:39.39areskiAhrimanes, what happened u wasnt able to sleep ?
09:40.03Ahrimanesareski: heh girl was sleeping over and she had to get up early so i just followed
09:40.09areskiAhrimanes, I finally sleep 7 hours
09:40.28areskiAhrimanes, hhahaha
09:40.31Ahrimanesareski: nice
09:43.10JohanMy musiconhold doesn't work. Asterisk spawns a mpg123 proces, but I can't hear anything on the phone... Anyone an idea what might cause this?
09:48.53*** join/#asterisk mkl1525 (n=daniel@84.19.203.139)
09:51.23_dimorning guuys
09:51.24_di_di i have just installed a@h 2.5
09:51.24_di_di configured it
09:51.24_di_di and when my soft phone trying to login
09:51.24_di_di i get only timeout
09:51.24_di_di 1 sec
09:51.26_di_di Feb  3 04:36:25 VERBOSE[6464] logger.c:     -- Remote UNIX connection disconnected
09:51.28_di_di Feb  3 04:36:40 DEBUG[3312] chan_sip.c: Auto destroying call 'B20C9FB2BCCA479F8AC0E4448DFF2085@195.5.36.206'
09:51.31_di_di Feb  3 04:36:41 DEBUG[3312] chan_sip.c: Auto destroying call '6954125373034C28AB5FDF6A4685242E@195.5.36.206'
09:51.36_di_di Feb  3 04:38:00 DEBUG[3312] chan_sip.c: Auto destroying call 'B20C9FB2BCCA479F8AC0E4448DFF2085@195.5.36.206'
09:51.39_di_di Feb  3 04:38:01 DEBUG[3312] chan_sip.c: Auto destroying call '6954125373034C28AB5FDF6A4685242E@195.5.36.206'
09:51.42_di_di Feb  3 04:38:24 DEBUG[6631] manager.c: Manager received command 'Command'
09:51.46_di_di Feb  3 04:39:20 DEBUG[3312] chan_sip.c: Auto destroying call 'B20C9FB2BCCA479F8AC0E4448DFF2085@195.5.36.206'
09:51.49_di_di Feb  3 04:39:21 DEBUG[3312] chan_sip.c: Auto destroying call '6954125373034C28AB5FDF6A4685242E@195.5.36.206'
09:51.52_di_di [root@asterisk1 asterisk]#
09:51.54_di_di shit like this
09:51.56_di_di always when it x-lite trying to reg
09:51.58_di_di it rejects
09:52.00_dican anybody help me plzzz
09:52.41_disoftphones are behind nat, and sip.conf users configured with "nat = 1 "
09:53.38*** join/#asterisk Tili (i=Tili@202-133-67-48-dialup.sat.net.pk)
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10:04.07*** join/#asterisk Genman (n=hansenhl@atlrel1.hp.com)
10:04.15GenmanHi Guys
10:04.38davidwhey dudes... question: if I add an address in manager.conf, will it bind to that IP/port, or do I have to do that somewhere else?
10:07.57*** join/#asterisk Bambr (n=Bambr@213-35-241-181-dsl.end.estpak.ee)
10:10.51*** join/#asterisk RoyK (n=roy@80.239.107.70)
10:13.00*** join/#asterisk jan__ (n=jan@ip22.ds1-saen.adsl.cybercity.dk)
10:13.11GenmanHi jan__
10:13.17jan__hi
10:13.48valste10how can i see if the flow arrive to my te110p board? The light is red...can i see a log?
10:14.14*** join/#asterisk RoyK (n=roy@80.239.107.70)
10:16.21bigjb<PROTECTED>
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10:22.14davidwdamn
10:22.16davidwdoesn't look like it
10:22.32RoyK<PROTECTED>
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10:25.57fulgasmorning
10:29.14Genmanmorning
10:29.22valste10morning
10:30.49RoyKmorning
10:30.53*** join/#asterisk Shoragan (n=shoragan@d072.apm.etc.tu-bs.de)
10:38.18fulgasanyone knows why the cdr(answer) and cdr(end) is equal ? even though the cdr have the right duration and billsec ?
10:38.27fulgasusing * 1.2.4
10:39.14fulgashttp://pastebin.ca/39732
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10:55.41darkskiezso, a shit your pants funny thing happened at work today, an order of 500  25cm patch cables has gone a bit wrong: http://www.bryars.me.uk/gallery/Miscellaneous/minipatchlead
10:57.55Genmanlol
10:57.59Genman:-D
10:58.08GenmanWell it is Friday
10:59.58darkskiezoh, im told it was for 2.5m
11:00.10Genmanhehe
11:02.34*** join/#asterisk dsfr (n=dsfr@pdpc/sponsor/digium/dsfr)
11:03.54*** join/#asterisk mogorman (i=root@user-24-236-84-48.knology.net)
11:05.51tronixhahaha ouch!
11:06.02mogorman?
11:06.12tronixdarksiez url
11:06.18davidweheh
11:06.19tronixpriceless photo.
11:06.26oejdarkskiez: That will require a lot of switches
11:06.30oejTo get anywhere
11:07.05darkskiezoej: i'm gutted they are being returned, i want them all
11:07.27oejCute little cablets
11:07.34*** join/#asterisk pa (n=Paolo@unaffiliated/pa)
11:07.37oejThat's evil, returning them
11:07.54tronixkid-sized cablets. :)
11:08.16RoyKcabelette
11:08.21tronixreminds me of the time my college friend brought home a 1U server to install, from work... so his 5 year old daughter
11:08.28tronixupon seeing it, exclaims: "that's a cute computer!"
11:09.24bigjbget a load of couplers and see how far you can get it before the impendance stops it working
11:09.40darkskiezi'm tempted to order some myself for personal use
11:09.47*** join/#asterisk skysport (i=user@212.156.115.17)
11:11.59skysportòåñò
11:12.19*** join/#asterisk dr0ck (n=dr0ck@gateway.digium.com)
11:15.42RoyKæøå
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11:19.26FuriousGeorgestrangest damn thing
11:19.41FuriousGeorgewhen my dynamic ip changes, my did doesnt come back
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11:23.19*** join/#asterisk fourcheeze (n=rich@82.153.215.21)
11:24.08fourcheezeI've got snoms using GSM and sipuras using g726 - My sipuras can call the snoms ok, but when a snom calls a sipura it get 488 - Not Acceptable Here, which I believe means that can't negotiate a codec
11:24.12fourcheezeany ideas
11:24.21fourcheezeI thought that * would reencode
11:24.31RoyKfourcheeze: ethereal trace it :)
11:24.57fourcheezeRoyK: it may come to that
11:25.49FuriousGeorgefourcheeze: you are allowing gz26 in asterisk right?
11:25.54FuriousGeorgeg726
11:27.05skysportDid you check sdp part of INVITE?
11:27.33fourcheezeFuriousGeorge: yes the snoms can dial out with g726
11:27.53fourcheezelisten to MoH, check voicemail, whatever
11:28.01FuriousGeorgedoes this have something to do with passthrough
11:28.07FuriousGeorgeis that a proprietary codec?
11:28.18CaViCcHisorry what is this kernel notice...
11:28.22CaViCcHiPower alarm on module 3, resetting!
11:29.52FuriousGeorge~g726
11:30.35*** join/#asterisk _8ball (n=spunk@64-13-46-242.eug.clearwire-dns.net)
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11:33.19fourcheezeFuriousGeorge: as far as I know it's non-proprietary
11:33.27fourcheezeugh
11:33.31fourcheezesorry s/snoms/sipuras
11:33.35*** part/#asterisk skysport (i=user@212.156.115.17)
11:33.40fourcheezesnoms do gsm sipuras do 726
11:33.48fourcheezecan't get my head around that :-)
11:33.51sleoparduhello : are any docs on how to set a headset for asterisk?
11:34.11RoyKdisallow=all, allow=gsm, allow=g726
11:34.13RoyKthat should do it
11:34.26RoyKsleopardu: directly on the asterisk box?
11:34.28sleoparduhello: to use it to call from the console
11:35.16sleoparduRoyK : yes I need to configure asterisk to call from the console by means of a sound card and a head set, and the problem is that I have two sound cards
11:35.39fourcheezeRoyK: is that in the per-device config or inthe globals?
11:37.11fourcheezethe snoms and sipuras all support G729a - is there a way I can use that when they talk to each other, but fall back to another codec when they dial out or use voicemail?
11:37.39*** join/#asterisk Katty (n=angela@64.82.232.54)
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11:44.23*** join/#asterisk Weezey (n=weezey@206.210.109.233)
11:44.45fourcheezeor can anyone recommend an ATA that supports GSM?
11:45.07WeezeyDTMF over IAX2 doesn't seem to work, any thoughts?
11:45.53WeezeySIP phone -> local * ---IAX2----> remote * VoicemailMain,  only shows first digit pressed in debug.
11:46.22*** join/#asterisk j3g (n=j3g@200.130.8.1)
11:46.46Makenshii have an odd problem with asterisk 1.2 running on fc3 ia64t+smp that when i insmod ztdummy, all calls go silent
11:46.51Makenshiif i remove ztdummy they go back to normal
11:47.04j3gIf I have a PBX system (non-asterisk) with 8 analog ports free and I want to link it to an asterisk box for VOIP calls, I need FXO adapters, right?
11:47.33Makenshij3g, yes, you need FXO
11:47.55*** join/#asterisk dsfr (n=dsfr@pdpc/sponsor/digium/dsfr)
11:47.57j3gWhen would I need FXS adapters?
11:47.57Weezeyj3g: http://www.digium.com/index.php?menu=fxsvfxo
11:48.07j3goh ok
11:48.09Makenshiif you wanted to provide dial tone to analogue phones
11:48.30hypnoxhow can i debug a dialplan? Setting verbosity and global debug to max doesn't show me exactly how a call is progressing through the dialplan, which is what I want
11:50.27j3gMakenshi, but if its my PBX that's going to be talking to the ASTERISK, and the asterisk talking to the VOIP carriers, it would still be FXO as the analog ports on my PBX are FXSs (they will not ring, but will send voltage to analog phones/etc) right?
11:51.23thazzahypnox: Try in the CLI: show dialplan
11:51.26j3gi am ordering the pci adapters so I wanted to be sure i am not getting the wrong ones
11:51.56Weezeyj3g: are you connecting your old PBX to "lines" on the asterisk box?
11:52.14j3gWeezey, yes... We already have a PBX with 200 terminals
11:52.17*** join/#asterisk dr0ck (n=dr0ck@gateway.digium.com)
11:52.27Weezeythen you want FXSs
11:52.35j3gWeezey, but I wanted to provide VOIP
11:52.42Weezeythey act as extensions to your voip system.
11:52.57Weezeyyou want FXOs if you're plugging them into bell lines
11:53.44Weezeyie:  Fax Machine ---FXS--> * ---FXO---> Bell Dialtone
11:53.56j3gthe situation is: I have a siemens pbx with 200 terminals, with a card that provides another 8 analog ports (lines). I am linking the asterisk box to those 8 lines and having the pbx route internatinoal calls to asterisk, which will then send to voip carriers
11:54.38Makenshij3g, we have siemens isdx pbx here, and use FXO on *
11:54.39j3gbut as the 8 ports are regular analog ports (one could plug an old phone in there and get a dialtone from it) wouldn't be FXO?
11:54.54Makenshii wish they wouldnt be so cheap and stump up for an e1
11:55.08j3gyeah but we can't insert an E1
11:55.12*** join/#asterisk Gunnar (n=gunnar@bkkb-gw.bitcon.no)
11:55.14Makenshij3g, the ports on the pbx are FXS, so you need FXO on asterisk
11:55.17j3gI learned that an E1 would provide 30 channels
11:55.32j3gOk that confirms it :) Makenshi thank you
11:55.44*** join/#asterisk HolyGod (i=nobody@got.securebinary.com)
11:56.00valste10i would try to connect my asterisk with a te110p digium board to a Dialogic d300 board but i have a problem with the protocol...any ideas??? tks, vale
11:56.34h3xi wish 5014 would apply to 1.2.4
11:56.34h3xheh
12:00.02CaViCcHiwhat is... Polarity reversed (0 -> -1) ??
12:00.41CaViCcHiany ideas???
12:01.10AndyCapj3g: but how much will the E1 card for the PBX cost you? :-P
12:01.34j3gAndyCap, the problem is that the PBX doesn't have any more free slots, and they ALREADY have the 8 port analog board
12:02.07AndyCapj3g: so it would cost you a new PBX. :>
12:02.35j3gAndyCap,  so doing it using analog would cost nothing (on the pbx side) and doing it on a E1 would have to purchase the board AND the analog board would not be used
12:02.47j3gso we would be wasting the analog board
12:03.00znoGanyone use Sipura or Linksys ATAs in a company?
12:03.16j3gAndyCap, if the usage for voip increases, we will replace the board with an E1
12:03.19fourcheezeznoG: I'm using Sipuras
12:03.24*** join/#asterisk linc (n=lms@cica-adm.fe.up.pt)
12:03.25j3gbut that's something for the future
12:03.58znoGfourcheeze: do you have users complaining that they try to transfer calls to other extensions and if they're not there, their voicemail picks up and when they try to switch back to line 1, they end up conferencing them? do you disable conferencing on your sipuras?
12:04.27linchello :)
12:04.58linccan anybody tell me where i can get a good startup tutorial for asterisk?
12:05.00fourcheezeznoG: Our customer that uses sipuras only transfers to them
12:05.02*** join/#asterisk bigb (n=bigb@static-70-21-248-201.nwrk.east.verizon.net)
12:05.08fourcheezeso I'm not sure if they have the same problem
12:05.27bigbAnyone know anything about the MP108's?
12:05.40AndyCapfourcheeze: the users never transfer calls away? sounds like hell. :-P
12:06.00*** join/#asterisk spunz_ (n=spunz@h081217096096.dyn.cm.kabsi.at)
12:06.20fourcheezeAndyCap: they use their snoms 90% of the time. Users transfer to the sipura if they need to go cordless
12:06.27AndyCapacha
12:06.35AndyCap~thebook
12:06.37jboti guess thebook is Asterisk: The Future of Telephony, released under the Creative Commons license and available at http://www.asteriskdocs.org << Read the book online!
12:06.43AndyCaplinc: ^
12:07.19lincthanks
12:09.42*** join/#asterisk kshumard (n=kshumard@gateway.digium.com)
12:09.43bigbNo one, eh?
12:09.44AndyCapfourcheeze: btw. what did you mean by ata that supports gsm? a sip/rtp - gsm gateway?
12:10.14*** join/#asterisk zotz (n=zotz@24.231.47.175)
12:10.20fourcheezeAndyCap: I mean gsm codec for voice
12:10.27AndyCapah. :> doh
12:11.23fourcheezeAndyCap: if you know of a mobile phone that does sip and gsm I'd also be interested :-)
12:14.33j3gwhat additional options/services does the asterisk bussiness edition provide over the free one?
12:14.49j3gother than support
12:14.54AndyCapj3g: warm fuzzies?
12:15.00mogormanno differnece
12:15.04mogormanerr wait
12:15.06j3ghmmm i already have warm fuzzies
12:15.07mogormannever mind
12:15.09mogormanim tired
12:15.13j3g:)
12:16.45*** join/#asterisk crich1999 (n=crich@p54BF8602.dip0.t-ipconnect.de)
12:16.56AndyCapfourcheeze: nah, I thought you meant something like this: http://www.2n.cz/products/gsm_gateways/voip/voiceblue.html
12:17.13FuriousGeorgeis there any easy way, with a dynamic ips, to get asterisk to act right with 3 other servers?
12:18.05*** join/#asterisk puzzone_ (n=Timo@e178103169.adsl.alicedsl.de)
12:18.21fourcheezeAndyCap: bookmarked that
12:18.36fourcheezeFuriousGeorge: define "act right"
12:18.36*** part/#asterisk davidw (n=davidw@apache/committer/davidw)
12:19.02FuriousGeorgefourcheeze: find eachother again and be able to communicate after ip change
12:19.25FuriousGeorgeif i could get it to uncache the dns address on refresh
12:19.30FuriousGeorgethat would be super
12:19.53FuriousGeorgei.e. do a dns lookup everytime, forgive me if thats not the technical term
12:20.34hypnoxtry using a dns record with a short TTL, like those from dyndns.org
12:21.43FuriousGeorgeis that all it is?  my provider current provider on two of the boxes is dynu.com, but i could use dyndns as it works with my work server
12:21.48FuriousGeorgei mean gateway
12:21.54FuriousGeorgework gw
12:22.03FuriousGeorgeno wait
12:22.33FuriousGeorgeif i dont manually restart asterisk wont it always try to its peers via ip
12:22.41hypnoxit might not work either, as * probably just resolves it once
12:22.48FuriousGeorgeyeah thats what i mean
12:23.14fourcheezeFuriousGeorge: so you've got a moving * ?
12:23.18fourcheezeget a fixed IP
12:23.26FuriousGeorgefloating like a butterfly and stinging like a bee
12:23.42FuriousGeorgefourcheeze: and by that i mean:  yes
12:23.46fourcheezehowever often does it move?
12:23.52hypnoxor script a graceful restart  on every ip change :)
12:23.57*** join/#asterisk iDunno (i=brettp@stef.sommitrealweird.co.uk)
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12:24.12FuriousGeorgea bit too often for my tastes.  maybe 2 or 3 times a week lately
12:24.17t0kehi
12:24.27FuriousGeorgebut there is another service in the area which is almost static
12:24.55*** join/#asterisk sack (n=sack@157.Red-81-33-112.dynamicIP.rima-tde.net)
12:25.34FuriousGeorgemogorman: how hard what it be to get in the code and have it resolve the name every time it refreshes?
12:25.43mogorman?
12:25.45mogormanname?
12:25.48FuriousGeorgedns
12:25.56FuriousGeorgeof peers
12:25.57mogormanthere is stuff for doing dns resolution
12:26.02mogormanover and over again
12:26.09*** join/#asterisk puzzone (n=Timo@e178103169.adsl.alicedsl.de)
12:26.11mogormanfor dealing with dyndns etc
12:26.14mogormanits dnsmanager
12:26.19mogormanand has dnsmanager.conf
12:26.43FuriousGeorgecool
12:26.49FuriousGeorgevery cool
12:27.41*** join/#asterisk secure75 (n=mic@host-82-135-62-14.customer.m-online.net)
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12:28.43fourcheezeIs it possible to display some text on a sip client when it is ringing?
12:28.57dpryocallerid-name? ;P
12:29.19fourcheezeI'd rather leave that
12:29.25fourcheezeI want to display a hunt group name
12:29.56dpryoWhatabout callerid-name "orgingalname TO GROUPNAME"
12:29.56fourcheezeso if the call is for sales it says "sales" etc
12:29.59dpryoor something
12:30.05fourcheezehmm
12:30.11dpryoThat's how avaya does it.
12:30.11fourcheezeok, I'll try that
12:30.13*** part/#asterisk Grubs (n=Miranda@c211-28-119-169.eburwd3.vic.optusnet.com.au)
12:30.14fourcheezeahh right
12:30.19fourcheezegood old avaya
12:31.17fourcheezedpryo: do I need to do that with AGI?
12:31.17FuriousGeorgemogorman: this dnsmanager, is this specifically for * or is it a linux app
12:31.17mogormanits specific to asterisk
12:31.18dpryofourcheeze: No, you could do it in the extensions config
12:33.43blackremedylet me get this right....zaptel is a driver for digium cards?
12:34.50*** join/#asterisk Johan (n=kvirc@194.151.113.2)
12:34.52JohanHi all
12:35.31FuriousGeorgemogorman: u sure its called "dnsmanager"  i did a google search for asterisk & dnsmanager and found 4 unrelated results?
12:35.55mogormanits in asterisk i can look it up but im about to sleep sleep
12:35.59mogormanyes blackremedy
12:36.31JohanMy musiconhold has a extremely poor quality and plays at a high volume. The quality is so bad you can't even recognise the song. I tried quietmp3 and also custom madplay, but the result is the same. are there any options to change the quality?
12:36.49FuriousGeorgefound it
12:42.07*** join/#asterisk linville (n=linville@azure.tuxdriver.com)
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12:47.25fulgason 1.2.4 the duration and billsec doesn't see to work... gettin' from a agi
12:47.29fulgasit was working before
12:47.31znoGgrr the Sipura/Linksys doco is terrible, can't find any documentation for each setting
12:48.15fourcheezedpryo: I'm just playing with this: exten => 601,1,Set(CALLERID(name)=Sales)
12:48.29fourcheezeand I'm getting: 060203-124636 ERROR[21648]: pbx.c:1408 ast_func_write: Function CALLERID not registered
12:48.44fourcheezedo I need to include something first?
12:50.27AndyCapfourcheeze: case sensitive?
12:51.06dpryoNah, that should work ..
12:51.10*** part/#asterisk secure75 (n=mic@host-82-135-62-14.customer.m-online.net)
12:51.20fourcheezehttp://www.voip-info.org/wiki/view/Asterisk+cmd+SetCIDName
12:51.29fourcheezeam I missing a module?
12:52.14fourcheezefunc_callerid.so
12:52.41dpryoAre you running asterisk 1.0.x?
12:52.50fourcheeze1.2
12:52.53dpryook
12:57.43*** join/#asterisk bmg505 (n=leon@dsl-146-25-114.telkomadsl.co.za)
13:01.14*** join/#asterisk NoNeo (n=Artur@193.24.24.10)
13:01.28*** join/#asterisk fugitivo (n=ajf@201.255.179.144)
13:02.25FuriousGeorgewhen mog says the "dnsmanager stuff" is "in asterisk somewhere" does he mean that its in the code but commented out, or something?
13:02.35FuriousGeorgethe code for * itself
13:03.01*** part/#asterisk mogorman (i=root@user-24-236-84-48.knology.net)
13:03.17RoyKdns.c?
13:03.27FuriousGeorgemaybe
13:03.47FuriousGeorgementioned also a dnsmanager.conf
13:09.46muthas anyone setup a polycom 601 via the web interface for it?
13:10.00muti don't want to setup all the stuff to do provisioning for a single customer
13:10.56*** join/#asterisk _Paulo_ (n=paulos@200-168-112-132.dsl.telesp.net.br)
13:11.21fugitivoit shouldn't be hard via web interface
13:12.18mutwell
13:12.26muttheres a sip tab where you can enter servers
13:12.30mutthen theres your lines tab
13:12.36mutwhere you can enter servers and ips
13:12.38muter
13:12.44mutservers and usernames and passwords
13:13.34muti dunno where to fill out what
13:13.48fugitivo?
13:13.51fugitivowhat?
13:15.42*** join/#asterisk tdonahue (n=tdonahue@208.51.101.201)
13:15.44mutdocs don't show anything on the web interface either
13:17.00*** join/#asterisk mkl1525 (n=daniel@84.19.203.158)
13:18.03*** join/#asterisk [TK]D-Fender (n=joe@toronto-HSE-ppp4122655.sympatico.ca)
13:19.06fugitivoyou need to enter you asterisk ip address, username and password there
13:19.46mutwhere tho
13:19.48mutboth places
13:19.50mutor?
13:19.55fugitivounder the sip tab
13:20.05mutthere is no spot for username and password in the sip tab
13:20.12mutonly in the lines tab
13:20.20fugitivothen in the lines tab
13:20.31fugitivojust try until the phone starts working :)
13:20.38mutheh thats what i've been doin :P
13:21.37blackremedyasterisk@home 2.5 released
13:21.50fugitivoduh, who cares
13:22.06*** join/#asterisk X-Rob (n=rob-x@dsl-202-173-151-24.qld.westnet.com.au)
13:23.09*** join/#asterisk Simon-_ (i=fictitio@80.193.211.68)
13:26.12*** join/#asterisk vivekjj (i=1076@203.199.110.93)
13:26.57vivekjjhello everyne
13:27.05*** join/#asterisk Tenkawa (n=Tenkawa@unaffiliated/Tenkawa)
13:27.17jbalcomb[TK]D-Fender ohayo gouzaimasu
13:27.17TenkawaQuestion for you all:
13:27.30vivekjjhow to check return values from agi does anyone know
13:27.38*** join/#asterisk Samoied (n=Samoied@popeye.opens.com.br)
13:27.47TenkawaDoes a 266 mhz geode cpu do you think have enough horsepower to act as a asterisk pbx for voip only?
13:28.10*** join/#asterisk Flyboy-SR22 (i=rsears@gateway.adnc.com)
13:28.46vivekjjslak-:   how to check return values from agi
13:29.00jbalcombTenkawa yes
13:29.17Tenkawanice
13:29.20Tenkawathanks :)
13:29.23*** part/#asterisk Tenkawa (n=Tenkawa@unaffiliated/Tenkawa)
13:29.26vivekjjhow to check return values from agi does anyone know
13:29.52jbalcombvivekjj i think everyone has seen your question.. silence is your answer
13:30.15vivekjjcmon friend, i am stuck up
13:30.58jbalcombif i knew the solution i'd tell you, if you want to me to go figure it out for you its $150.00/hr
13:31.53*** join/#asterisk cassio- (i=cassio-@201008009068.user.veloxzone.com.br)
13:32.02vivekjjgiven
13:32.03muti'll go for $20/hr
13:32.05fugitivommm, i think i should increase my incomings per hour
13:32.20mutcause hell.. i make $9 right now
13:32.21mut:P
13:32.26jbalcombouch
13:32.33fugitivomut: you are a gift
13:32.42mut?
13:32.42*** part/#asterisk FuriousGeorge (n=brian@ool-44c5a9b8.dyn.optonline.net)
13:32.58jbalcombyeah, maybe you can work for me mut. i'll give you $10
13:33.17fugitivofor me too
13:33.19mutdeoends on where i have to move
13:33.32[TK]D-Fenderjbalcomb : Arigato
13:33.42fugitivonot moving, remote work
13:33.48[TK]D-Fenderjbalcomb : Been a while since I brushed up on my Japonese....
13:33.54mutsure
13:33.55cassio-I have a script that does picks up next available line, can someone take a look at it to help me with modules?
13:34.08mutextra income would be great
13:34.26mutgot that phone workin
13:34.59*** join/#asterisk FuriousGeorge (n=brian@ool-44c5a9b8.dyn.optonline.net)
13:35.02mutseems lagged to shit tho, 149ms on wifi, won't even be able to hold a conversation the jitter is so high
13:36.44tzangercool my polycomsarrived today
13:37.07fugitivoi want too
13:37.39*** join/#asterisk Simon-__ (n=byte@ecares35.ulb.ac.be)
13:37.39mutwhats a poly 601 cost?
13:37.40mutany idea?
13:37.40mutw/o me having to go look
13:37.41sivanatzanger: which models?
13:37.42X-Robvoipsupply.com for USD$
13:37.44tzanger501 and 301
13:37.47sivananice
13:37.51tzanger10 501s, 1 301
13:38.00sivanaI have those too.. I like them
13:38.03fugitivoisnt' atacomm cheaper?
13:38.10jbalcomb[TK]D-Fender nihongo o hanashimasuka
13:38.16sivanathe 301 is a bit weak on the lcd display
13:38.42fourcheezeis there a way to persuade a snom 360 to use more than 1 line to show the incoming caller?
13:39.00fourcheezecurrently it tries to fit it all in one line and display an animation below
13:39.25pb__fourcheeze: you can turn off the animation in the web interface, which I think makes it use two lines
13:39.35[TK]D-Fenderjbalcomb : You got me there.... All I got from there was "bookstore" IIRC....
13:39.58[TK]D-Fendermut : $250 @ atacomm, $240 TriTech
13:40.05mutheh
13:40.06mutthats horrid
13:40.16mutthis lady didn't want to buy a spa-2002 ata from us
13:40.18fugitivo[TK]D-Fender: the 601???
13:40.26mutso she went out and bought this thing for her house
13:40.28[TK]D-Fenderfugitivo : yup.
13:40.31mutand is using 1 line
13:40.45fugitivo[TK]D-Fender: that's cheap, i can't buy it here for less than $600 :(
13:40.47[TK]D-FenderRemember as Polycom's least paid pimp I have to know this stuff!
13:40.56[TK]D-Fenderfugitivo : That really blows...
13:41.04[TK]D-Fenderfugitivo : Where are you located?
13:41.16fugitivo[TK]D-Fender: Argentina
13:41.16fourcheezepb_ any idea what the option is called - can't seem to find it anywhere
13:41.29*** join/#asterisk brad_mssw (n=brad@shop.monetra.com)
13:42.13*** join/#asterisk _8ball (n=spunk@64-13-46-242.eug.clearwire-dns.net)
13:43.00[TK]D-Fenderfugitivo : However consider just how cheap housing is in SA.... there is a price to pay for everything. Here were are taxed quite a lot....
13:44.47FuriousGeorgeare snom the only phones with programable LEDs using devstate?
13:45.38*** join/#asterisk CleanerX (n=nix@p54A389E0.dip0.t-ipconnect.de)
13:46.17[TK]D-FenderFuriousGeorge : I believe the devstate stuff just allows hints to be set and should therefor work on any phone with presence support
13:46.23RoyKDo Not Exchange The PID And The Signal Number When Sending SIGHUP To A Process On A Linux Box
13:46.30pb__fourcheeze: under "Preferences", look for "ringer animation"
13:46.37[TK]D-FenderFuriousGeorge : The advantage Snom's have is that they come whit a whole LOT of buttons/lights.
13:46.46X-Robooh
13:46.49X-RobI love snoms
13:46.50pb__fourcheeze: it's near the bottom of the "general information" block
13:46.52X-RobThey're my bitch
13:46.59*** join/#asterisk Tili (i=Tili@202-133-65-156-dialup.sat.net.pk)
13:47.04X-RobBTW: It's 11:46pm, friday night, and I'm pissed as a parrot.
13:47.28RoyKpoor man
13:47.32*** join/#asterisk Vlasis (n=Vlasis@194.219.121.194)
13:47.37RoyKpissed as a parrot :)
13:48.22fourcheezepb_ I've got "Use Flash Plugin" but that's off
13:48.29Vlasishello all. I have a problem with sending DTMF. I need to send a DTMF tone to the calling party when the call is answered, but sendDTMF is sending the tones to the called party instead! is there a way to solve this?
13:48.56*** join/#asterisk Assid (n=assid@203.115.64.14)
13:49.04Assidheya
13:49.22fourcheezepb_ which firmware version are yours?
13:49.24Assidanyone got the list of the ports that should be open in a firewall for sip + iax
13:49.52pb__fourcheeze: ah, looks like that option is only in 5.x
13:49.57pb__fourcheeze: my 4.x phones don't have it either
13:50.16pb__the one that does is running 5.2
13:50.41*** join/#asterisk zeedo (n=zeedo@reboot-robot.net)
13:53.00*** join/#asterisk mgf (n=mgf@r200-125-29-186-dialup.adsl.anteldata.net.uy)
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13:56.48*** join/#asterisk SibRw0rk (n=DaPhrek@66.234.235.84)
13:57.05muthttp://www.voipsupply.com/product_info.php?products_id=1291
13:57.12mutthats a pretty sweet invention
13:59.32Flyboy-SR22has anybody used the S100-FX v2.0
13:59.34Flyboy-SR22?
14:00.22fourcheezepb_ thanks - I'll upgrade
14:01.02Flyboy-SR22I am looking for a remote box that will allow me to make an IAX connection to it and grab a remote FXO port
14:01.08Flyboy-SR22anyone have suggestions..?
14:02.18fugitivo[TK]D-Fender: that's true, but i don't want housing, i want cheap ip phones, lol
14:02.20*** join/#asterisk svenna_ (n=svenna@p548D0309.dip0.t-ipconnect.de)
14:02.28FuriousGeorge[TK]D-Fender: can you recommend another make/model of phone over the snom 320/360
14:03.02jbalcomb[TK]D-Fender ah, 'hon' is book indeed but using another meaning 'nihongo' is Japanese
14:03.24jbalcomb[TK]D-Fender how long would you need to revamp the configs?
14:03.28[TK]D-FenderFuriousGeorge : All recommendations depend on what you want out of a phone.  I typically suggest Polycom, but like I said "it depends".  Describe their usage....
14:04.12fugitivowhat was the site with cheap prices?
14:04.15fugitivowww.atacomm.com?
14:04.26jbalcombwww.ebay.com
14:04.50[TK]D-Fenderjbalcomb : Several hours clearly.  It'd be done in large stages.  The incoming DID's are a lot to get done in 1 shot, I'd completely rewrite your forward, STDEXTEN and remove the overly generic XXXX method, etc....
14:05.02jbalcombfugitivo the GrandStream GXP-2000s are rather nice phones.. I have about 130 I could sell you.
14:05.10FuriousGeorge[TK]D-Fender: answer, put people on hold, transfer, pickup people off hold
14:05.17fugitivojbalcomb: how much?
14:05.32jbalcombfugitivo $170 ea.
14:05.39[TK]D-Fenderfugitivo : www.atacomm.com and http://www.tritechcoa.com/Brands/9PZ7.html
14:05.42fugitivojbalcomb: if you can pass it to Argentina without custom taxes i'd buy it
14:05.44[TK]D-Fenderjbalcomb : you are evil
14:05.54jbalcombfugitivo i know its kind of high but i need the money to replace them with snoms
14:06.07[TK]D-Fenderfugitivo : Unless your taxes are 100% you'd be losing with that deal :)
14:06.13_Paulo_Hi all...
14:06.22[TK]D-Fenderjbalcomb : So thats the direction you're heading in?
14:06.23fugitivo[TK]D-Fender: well, it's 50% + 10,5%
14:06.23jbalcomb[TK]D-Fender hey, you want this job or not? sssshhhh.
14:06.42*** join/#asterisk glm2k (n=GLM@rrcs-24-199-11-46.west.biz.rr.com)
14:07.05jbalcombfugitivo yeah, i got a guy.. maybe he can smuggle 130 VoIP phones in for ya..
14:07.21fugitivojbalcomb: :)
14:07.42_Paulo_Does somebody know if the settings in the /etc/zaptel.conf may influence timing problems I'm having when sending faxes?
14:08.10fugitivo[TK]D-Fender: what's the difference between the 601 and a cisco 7940g?
14:08.18jbalcomb[TK]D-Fender not sure yet, just passing gas. i'll know more after i get my new server and the cisco phones
14:08.29fugitivo[TK]D-Fender: it's only $10 of difference between them
14:08.42cypromisfugitivo: sound
14:08.55cypromispolycom shoveled all the goodies from their conferencing systems into the phones
14:09.12fugitivopolycom has better sound?
14:09.25jbalcombfugitivo http://bani.anime.net/phonez/
14:09.45[TK]D-Fenderfugitivo : A lot.  Polycom supports standard PoE, doesn't require paid support for firmware, comes with a wall-plug, 6 lines, possibility of adding Attendant modules, etc.  Everything the 7940G DOESN'T.
14:09.56fourcheezepb_ yeah upgrading to 5 does it - pity it takes so long to do each one
14:10.06[TK]D-FenderIP601 pretty much destroys a 7940G
14:10.10*** join/#asterisk mistral (i=mistral@jstevenson.plus.com)
14:10.13fugitivo[TK]D-Fender: what about the lcd and xml support?
14:10.20jbalcombyes/yes
14:10.49*** join/#asterisk cpm (n=Chip@dns1.eruditium.org)
14:10.53[TK]D-Fendercypromis : Cisco is on par with Polycom for voice quality from everyone I've heard from.  It wouldn't be the deciding factor.
14:11.20*** join/#asterisk cyteen (n=soatola@82.153.174.147)
14:11.38[TK]D-Fenderfugitivo : So far no "push" capability for XML, but you have a "services" button with XML, and the idle screen itself can use XML and you can set the refresh rate.
14:11.54cypromis[TK]D-Fender: tries them both next to each other ?
14:12.34[TK]D-Fendercypromis : No, not personally, just talked with people who have both (file for example)
14:12.37*** join/#asterisk elg (n=fugalh@falcon.fugal.net)
14:12.47Luhiwufugitivo: where are you from? 10.5% taxes seems like responsable inscripto, but phones have 21% taxes...
14:12.50cypromisgood way for selling them
14:13.10fugitivoLuhiwu: Argentina :)
14:13.24fugitivoLuhiwu: phones have 21% ? damn
14:13.45[TK]D-FenderFuriousGeorge : What kind of call-volume?  Need speakerphone?
14:13.46Luhiwufugitivo: yep, 21%.
14:14.02Vlasishello all. I have a problem with sending DTMF. I need to send a DTMF tone to the calling party when the call is answered, but sendDTMF is sending the tones to the called party instead! is there a way to solve this?
14:14.06[TK]D-FenderFuriousGeorge : 2nd ethernet port port?
14:15.15_Sam--how would i do something like this on the console...  sip show peers | more
14:15.18*** join/#asterisk donnib (n=aaa@0x555281d0.adsl.cybercity.dk)
14:15.24donnibhi all,
14:16.08fugitivo_Sam--: run asterisk outside the cli
14:16.21fugitivo_Sam--: with the flag for running a command
14:16.25fugitivoi don't remember the flag
14:16.27_Sam--i see...never done that
14:16.35donnibi have a simple question which i can't solve. I am sure it's simple but i can't seem to find an answer. I am running A@Home which works fine but when somebody calls in it takes time before the call is transfered to my (only)extension. What's the cause of that ?
14:16.39_Sam--thanks
14:16.53[TK]D-Fenderdonnib :...
14:16.54[TK]D-Fender~amp
14:16.56jbotextra, extra, read all about it, amp is NOT supported here! people using it should join #amportal
14:16.56*** part/#asterisk redondos (n=redondos@12-207-132-99.client.mchsi.com)
14:17.00_Sam--i think its -x
14:17.29donnib[TK]D-Fender: what do you mean by AMP ? Is it some place i can set that up in AMP ?
14:17.35*** join/#asterisk saw (n=saw@skywalker.patronas.de)
14:17.43donnib[TK]D-Fender: sorry
14:17.47fugitivo_Sam--: yes, asterisk -rx "sip show peers" |more
14:17.52donnib[TK]D-Fender: i will look there then
14:17.54_Sam--ty again
14:18.03[TK]D-Fenderdonnib : A@H uses AMP for its * configuration.  Your problem is AMP, not *.
14:20.17*** join/#asterisk coppice (n=chatzill@238.162.17.210.dyn.pacific.net.hk)
14:20.37X-RobEvening, steve
14:20.39X-RobI'm pissed.
14:21.05fugitivoX-Rob: relax and tell us your problem
14:21.06coppicepissed off or intoxicated?
14:21.17VlasisIs there anyone who knows how to send DTMF to the calling party after the call is answered?
14:21.19X-Robintoxicated
14:21.46*** join/#asterisk FastJack (n=fastjack@reverse-82-141-49-149.dialin.kamp-dsl.de)
14:22.15X-Robwhich is different to pissed _off_,
14:22.19X-Robs/,/./
14:23.50*** join/#asterisk skeffling (n=Andrew_H@andrew.1ec.aaisp.net.uk)
14:24.45*** join/#asterisk sevard (n=kynan@198.174.233.3)
14:24.49coppicewell, getting pissed is the same (americans don't seem to understand that use of the word)
14:25.17*** join/#asterisk dca[laptop] (n=dca[lapt@sta-208-139-193-162.rockynet.com)
14:25.17X-Robyet another vaguery of the english language
14:25.18*** join/#asterisk Utah_Dave (n=boucha@0-2pool130-165.nas28.salt-lake-city1.ut.us.da.qwest.net)
14:25.21X-Robwe shoudl all speak latin
14:25.31sevardlatin is weiiiiiiird
14:25.31fugitivoi speak latino
14:25.33Money5ackhey...
14:25.49Money5ackcan i bind my asterisk to more than one interfaces but not all interfaces ?
14:25.55Makenshio s t mus tis nt... for vit add ed... bat and bant, was'ing and were'ing... that's what i remember from school :)
14:25.58Money5ackfor example comma speretated ?
14:26.20X-RobSadly, my sister did 5 years of latin and can read, write and speak it fluently. She worsk at the oxford university museum, because of this.
14:26.27*** join/#asterisk Mr (n=zdeeck@r23s01p06.home.nbox.cz)
14:26.37*** join/#asterisk Flauto (n=zhao@71.194.194.48)
14:26.57X-RobWhilst I did 3 years of german
14:27.09X-Robof which I rememebr'ich bin, du bist, er/sie/est ich'
14:27.37Money5acker/sie/est ich ? ;)
14:27.53Money5ackyou meas er/sie/es ist
14:27.55Money5ackmean :)
14:28.09X-Robyes
14:28.14X-Robsee!#$
14:28.23cypromisdownundergermanlish
14:28.25cypromisbah
14:28.46*** join/#asterisk spunz_ (n=spunz@h081217096096.dyn.cm.kabsi.at)
14:29.13_Sam--in the usa, there is no tax at all on VOIP calls, right?
14:29.47Money5ackthe only language's i can "speak" is german and english... german is my mother-language and my english is only learnd in school and job...
14:30.29_Sam--my example would be if i used a voip company for termination...a remote gateway...would those calls be taxable in any way
14:30.50Money5ackbut... back to my question: can i bind the * to more than one ip address but not to all...
14:30.53Money5ack?
14:31.06tronixX-Rob: as one of my countrymen has discovered, 'Ich bin ein Berliner' is good for some laughs. :-)
14:32.07tronixMoney5ack: I don't see a startup option for that, alas. Would be useful, indeed.
14:32.16X-Robtronix, I've obviously missed something there
14:32.27X-Robclue me up?
14:32.38zigmanMoney5ack Muttersprache=mother-tounge
14:33.04Money5acktronix: hmm.. okay.. and what about the option "bindaddr" in sip.conf ?
14:33.14zigmanMoney5ack about the ip thing... i think not
14:33.36tronixX-Rob: urban legend (but an amusing one), see: http://en.wikipedia.org/wiki/Ich_bin_ein_Berliner
14:33.41tronix(read the jelly doughnut section)
14:35.09Money5ackwith the ip thing... thats bad...
14:37.08cypromisI'm a pancake ?
14:37.09*** join/#asterisk kpettit (n=keith@69.15.174.114)
14:37.18cypromispeople say a lot of stupid things about themselves
14:37.24tronixhahaha
14:37.37tronixmmm cherry pancakes
14:37.42*** part/#asterisk Mr (n=zdeeck@r23s01p06.home.nbox.cz)
14:37.57jbalcombwoot. my SPA-2002 is here!
14:37.58X-Robtronix, when I was in germany, I never heard them called berliners.
14:38.32cypromisyou didn't ?
14:38.38cypromisyou must have been living in berlin than
14:38.40jbalcombneither did i
14:38.43cypromisin berlin they call them pancakes
14:38.49X-RobPfannkuchen sounds vaguely familiar, but I only had them once, and can only remember referring to them once, but can't remember what I Said
14:38.50cypromiseverywhere else they call them berliner
14:38.58X-Rob(at a different time)
14:39.39X-Roband I said 'zucher bun' whilst pointing at it so they got the idea what I Wanted
14:39.47X-Rob(the first time)
14:39.55*** join/#asterisk santiago (n=santiago@63.245.86.215)
14:40.53*** part/#asterisk santiago (n=santiago@63.245.86.215)
14:42.06*** join/#asterisk Mr (n=zdeeck@r23s01p06.home.nbox.cz)
14:42.19*** part/#asterisk Mr (n=zdeeck@r23s01p06.home.nbox.cz)
14:42.44*** join/#asterisk bweschke (n=bweschke@c-68-36-51-213.hsd1.nj.comcast.net)
14:42.56mkl1525Hi, does anyone know  multiport hfc-s cards?
14:43.26RoyKknow how what?
14:43.30*** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com)
14:43.38coppiceI've been to Hamburg. I've been to Freising. I'm still looking for a German town named after Coke
14:43.38RoyKi've heard of them....
14:44.38*** join/#asterisk telcoDEJ (n=ed@c-68-82-75-221.hsd1.de.comcast.net)
14:45.03Hmmhesays"we've had 20 complaints of hacking coming from ip address x.x.x.x"
14:45.15Hmmhesays"oh yeah? theres no pc on that addy, fark off
14:48.07fourcheezeAnyone know the variable name for the full contact of the user?
14:48.11*** join/#asterisk klictel (n=klictel@207.107.208.137)
14:49.06*** part/#asterisk telcoDEJ (n=ed@c-68-82-75-221.hsd1.de.comcast.net)
14:49.37vivekjjhi friends
14:50.03vivekjjanyone knows how to find the return value of a command like agi or dial , etc.
14:51.34*** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk)
14:52.07sevardI'm looking for a manual for the Cisco 3500 Catalyst XL, does anyone have one on hand?
14:52.38nestarmanual for what?
14:52.44vivekjjwisdom: do u know how to  check the return value of a command like agi, dial, or any other command?
14:52.44nestarit's just IOS?
14:52.52nestaror you need a diagram on how to plug things up
14:52.53nestar:)
14:53.06*** join/#asterisk m_a_g_o (i=maxgluck@200.109.32.127)
14:53.53*** part/#asterisk Vlasis (n=Vlasis@194.219.121.194)
14:54.48*** join/#asterisk Dr-Linux (n=Nothing@202.125.141.9)
14:55.11Dr-Linuxhi all
14:55.17Dr-Linuxhowdy
14:55.25RoyK<PROTECTED>
14:55.29*** part/#asterisk m_a_g_o (i=maxgluck@200.109.32.127)
14:55.52*** join/#asterisk myiagy (n=myiagy@mail.voffice.com.br)
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14:56.33sevardnestar: I need all the docs that I can get on it
14:56.37*** join/#asterisk m_a_g_o (i=maxgluck@200.109.32.127)
14:56.38sevardI'd love a full manual
14:57.04*** join/#asterisk Syrus_ (n=pascal@tahiti.mpl.rullier.net)
15:04.29*** join/#asterisk sigmounte (n=sigmount@LSt-Amand-152-33-38-41.w82-127.abo.wanadoo.fr)
15:06.13Seldon1975when I first came to this channel complaining about crosstalk I was ridiculed... now Digium are sending us new experimental hardware to address the issue of crosstalk
15:06.17*** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk)
15:06.21Seldon1975so I'd just like to say 'In your Face'
15:07.35Seldon1975to the ridiculers
15:09.28*** join/#asterisk juanjoc (n=juanjoc@200.73.189.82)
15:09.32*** join/#asterisk trelane_ (i=trelane@66.93.203.199)
15:09.46FuriousGeorgeecho?
15:09.54FuriousGeorgecrosstalk=echo?
15:10.24m_a_g_ohello, good morning ... i'm having problems with DTMF, precisely receiving duplicte tones and can see that an RTP Duration Event is being received between two RTP End Events, from the 3 regularly received. given all the tests made and their results, i can conclusively say that this is causing the repeated tones. is there a way to fix this? i was thinking of telling dsp.c to drop a packet if its timestamp or sequence number was greater than the last
15:10.38*** join/#asterisk af_ (n=af@ip-142-84.sn1.eutelia.it)
15:10.40[TK]D-FenderSeldon1975 : I'm notsure thats a Digium issue, but I'll check with my customer service dept (the guys who had it happen here) to see if its happened since we replaced our card.
15:11.09m_a_g_oplease let me know your thoughts, thx!
15:11.15*** join/#asterisk Kohat (n=Nothing@202.125.141.9)
15:11.30FuriousGeorge~crosstalk
15:11.42FuriousGeorge~jbot
15:11.43jbotwell, jbot is only marginally useful at best, or a silly little bugger
15:11.56FuriousGeorgethe former
15:12.41FuriousGeorgeis crosstalk echo or that effect you get that you hear someone else's conversation
15:13.10Seldon1975crosstalk: hearing other lines
15:13.18Seldon1975echo: hearing your own line twice
15:13.48FuriousGeorgei know what echo is...
15:13.53Seldon1975D-Fender: Digium are sending us an experimental TDM2400 with hardware crosstalk correction
15:14.14Seldon1975FuriousGeorge: sorry..sorry..sorrry :}
15:14.26Seldon1975(that was 'sorry' being echoed)
15:14.53m_a_g_oja ja ja
15:14.56m_a_g_othat was too
15:15.03Seldon1975heh heh heh
15:15.07m_a_g_owould any one care to suggest please?
15:15.08FuriousGeorgei get it now
15:15.22FuriousGeorgeSORRY SORrry sorry
15:15.55DandanI am looking for someone who has a first hand experience with Voicetronix boards...
15:16.28m_a_g_oduplicte tones and can see that an RTP Duration Event is being received between two RTP End Events, from the 3 regularly received. given all the tests made and their results, i can conclusively say that this is causing the repeated tones.
15:16.57FuriousGeorgem_a_g_o: i got nothing
15:17.16m_a_g_owhat's your anger about?
15:17.21Seldon1975nor me
15:18.07*** join/#asterisk Reko (n=nonsense@cpc4-hem15-0-0-cust755.lutn.cable.ntl.com)
15:18.23*** join/#asterisk Optic (n=dfraser@miso.capybara.org)
15:18.40Optichey, would any of you guys happen to have polycom SIP 1.6.3 handy? :(
15:19.40Kattyhi lads.
15:19.52Optichi
15:19.56RekoHi. I'm experimenting with asterisk and in order to do so I'm running it under a virtual machine. I have ztdummy installed and loaded, but when I connect to 1000 for the demo, the sound is audible but breaks up a bit.
15:20.28Opticasterisk is pretty timing sensitive, i would recommend running it under the reccommended hardware/software configuration
15:20.31Hmmhesaysunder a virtual machine
15:20.39RekoYeah, I figured the VM would probably be the issue.
15:20.40FuriousGeorge~FuriousGeorge
15:20.41jbot[furiousgeorge] a knife-fighting monkey last seen with The Man with the Yellow Bat
15:21.27Seldon1975~Seldon1975
15:21.37Seldon1975darn
15:21.53*** part/#asterisk vivekjj (i=1076@203.199.110.93)
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15:22.17Seldon1975Optic: http://www.freedomphones.net/polycom/files/
15:22.47Seldon1975Optic: oh that only has 1.6.2...
15:23.04*** join/#asterisk rpm (i=russell@S010600111155e117.cg.shawcable.net)
15:23.40rpm[2006-02-02 19:58:29]     -- Executing System("Zap/4-1", "/usr/bin/lame /var/spool/asterisk/monitor/s-02022006-19:57:32.wav /var/spool/asterisk/monitor/s-02022006-19:57:32.mp3") in new stack - can someone let me know why that never executes properly? im calling the System() function properly right?
15:23.49sevardOptic: It sounds like you know what you're talking about.  I'm testing * on a 440mhz box with 256 mb ram, on a 10/100 network I'm experiencing jitters and such.  Shouldn't the processer beable to take _one_ phone call?
15:23.59nestarsevard: yes.
15:24.09nestari run * on a amd 233
15:24.18nestarwith 64 or 128 mb
15:24.26Seldon1975sevard: are you sure it's jitter
15:24.28Opticsevard: yes, it should
15:24.43Opticalthough I don't have any experience running asterisk on slow hardware
15:24.51Seldon1975sevard: jitter can be corrected in the phone devices
15:24.53OpticI've got asterisk on a P4 2.8 with a TE110p card
15:24.58nestarK6-233, 128mb
15:24.59Opticand it's given me no trouble
15:25.11Seldon1975sevard: what phones are you using?
15:25.20*** join/#asterisk SplasPood (i=jwb@206.252.198.100)
15:25.23sevardWhat are some methods one can take to reduce jitters?
15:25.36Seldon1975sevard: depends on what phones are you using?
15:25.36sevardSeldon1975: I'm using a couple X-Lite phones and a hardphone.. one second I'll find out exactly what it is
15:26.04Seldon1975their manuals should tell you how to tweak the setting to elimintate jitter
15:26.09Seldon1975if jitter really is the issue
15:26.18nextimei have an asterisk @ my office on a celeron 233 with 64 MB of ram, it work ok
15:26.39*** join/#asterisk slak- (i=slak@rewted.biz)
15:26.44slak-HELLO!@!#W@#
15:27.10sevardSeldon1975: and my hardphone is a Hirakawa Electronics HOP-1002
15:27.14|vinsik|nextime: how many simultanious calls can it manage?
15:27.19slak-okay i want to use zaptel for local-calls and my sip provider for longdistance
15:27.30slak-can soemone help me reconfigure my all zaptel setup :D
15:27.55Seldon1975sevard: Im not an expert but our Polycom phones have embedded HTTP servers that allow you to tweak jitter buffer settings
15:28.06Seldon1975sevard: your hardware may have a similar thing
15:28.13nextime|vinsik| : 10 concurrently calls with virtually no load ( i have only 10 phones )
15:28.13badboyzanyone ran into the MOH jitter problem when you have more than 1 person holding?
15:28.23slak-embedded HTTPd is AWESOME
15:28.25Seldon1975try doing an http://<ip address of phone>
15:28.27slak-what a great idea
15:28.40slak-did you see that model helicopter with httpd
15:28.42|vinsik|nextime: nice..
15:28.43slak-it really flies
15:28.46slak-and has wifi
15:28.49sevardSeldon1975: This hardphone has a http server that allows me to set SIP settings.  What sort of things should I be looking for to decrease jitter?
15:29.01sevardSeldon1975: when we talk about jitter.. what _exactly_ are we describing.
15:29.03*** join/#asterisk geoffb1974 (n=gbodeman@mail.televerde.com)
15:29.07|vinsik|nextime: i tested linksys openwrt asterisk.. 2-3 calls and sound starts to jitter..
15:29.08*** join/#asterisk tdonahue (n=tdonahue@208.51.101.201)
15:29.09geoffb1974hey all
15:29.13geoffb1974thanks for your help yesterday
15:29.20|vinsik|nextime: so much for cheap and portable solution :D
15:29.26Seldon1975sevard: aha ok I assumed that when you said you were experienceing jitter that you had already identified that
15:29.43Seldon1975sevard: I'm not an expert but I believe it has to do with IP traffic priorities
15:29.47sevardSeldon1975: I noticed different people say jitter for differnt things.
15:30.05Seldon1975sevard: D-Fender can probably help us both with the definition of jitter
15:30.12Seldon1975~jitter
15:30.14jbot[jitter] at http://www.handhelds.org/z/wiki/Kernel%20Documentation
15:30.24Seldon1975try that
15:30.44iCEBrkryo yo
15:30.45Seldon1975no - looks useless
15:31.34Seldon1975sevard: sorry, I'm at the (low) limit of my expertise in terms of jitter
15:31.51slak-i love you
15:31.59Seldon1975thx!
15:32.51slak-exten => 90,1,Dial(Zap/g1/${EXTEN:1})
15:32.51slak-exten => 90,2,Congestion
15:32.53slak-what is that
15:32.56slak-operator support?
15:33.08slak-telco operator
15:33.38*** join/#asterisk Setec_Astronomy (n=iamatall@69-173-5-230.clvdoh.adelphia.net)
15:35.07sevardSeldon1975: do you like the monkey wrench gang?
15:35.20|vinsik|maan did i pick the time to comment "jitter" :D
15:35.37Seldon1975sevard: what is that?
15:35.55Seldon1975sevard: Foo Fighters?
15:36.12sevardSeldon1975: ahh, that makes me sad your nick isn't connected to seldon seen smith.  i wanted to pull out HAYDUKE LIVES and such. :/
15:36.20Seldon1975haha
15:36.32Seldon1975no, Isaac Asimov's Foundation series...
15:36.32*** join/#asterisk svenna_ (n=svenna@p548D0309.dip0.t-ipconnect.de)
15:36.37Seldon1975Hari Seldon
15:36.37sevardheh.
15:36.44sevardgotcha.
15:36.45CaViCcHisorry but...
15:37.33sevardSeldon1975: when I say jitters I mean things like, mid call getting getting cut outs, and it going like "Hi my name is fred and I enn - j j oy - yyy going to the store sometimes"
15:37.39CaViCcHiThere is a category of phones that dont work connected to FXS??????????????
15:37.56CaViCcHimaybe... :P Is There...
15:38.05znoGanyone played much with dialplans in the Sipura/Linksys ATAs?
15:38.15sevardWow, you used 14 too many question marks.
15:38.41sevard14 question marks too many, grr@sentenceformatting
15:38.47CaViCcHi:P
15:38.49Dandanok, i know where my problems with echo come from...
15:39.10CaViCcHiIs there a category of phones that dont work connected to FXS???
15:39.13*** join/#asterisk elephantMan (n=elephant@46.205.103-84.rev.gaoland.net)
15:39.13CaViCcHibetter :P
15:39.15coppicea lot of FXS ports provide a rather low voltage, and many phones don't work well with them
15:39.26CaViCcHioh ok...
15:39.30*** join/#asterisk klasstek (n=nunyobiz@sta-206-168-96-89.rockynet.com)
15:39.30sevardCaViCcHi: Closer, that's two too many.
15:39.42DandanCaViCcHi: yeah I have one of those...
15:39.59*** join/#asterisk tdonahue (n=tdonahue@208.51.101.201)
15:40.10rpmcan someone look at http://pastebin.ca/39760 - i can't figure out why /usr/bin/lame is not running
15:40.15[TK]D-FenderznoG : Whats your question about ATA dial-plans?
15:40.57CaViCcHicoppice: Cause my fxs only works with a cordless... it rings but if you answer nothing more happens...
15:41.30DandanCaViCcHi: o have a ge industrial cordless
15:41.35Dandanand it doesn't work with fxs
15:41.39Dandandoesn't recognize ringtone
15:41.42[TK]D-FenderSeldon1975 : If they blame the card for cross-talk and need a special replacement to handle it then I would seriously reconsider what model I was buying....
15:42.05CaViCcHiIs there a kind of phone that SURELY works?
15:42.29znoG[TK]D-Fender: i have this complicated dial plan entry which I want somebody else to look over and tell me if its wrong, or what
15:42.36coppicequite a few ports use 24V DC, instead of the usual 48 you get with a PSTN line. that should be fine, but a lot of phones don't like it. same thing happens with some traditional PBXes
15:42.39[TK]D-FenderznoG : Ok, paste away
15:43.24Dandanfunny, when I start asterisk, that is what I have in my dmesg:
15:43.25Dandanvpbhp[00]: Host Echo enabled on port 8
15:43.25Dandanvpbhp[00]: Host Echo enabled on port 9
15:43.25Dandanvpbhp[00]: Host Echo enabled on port 10
15:43.25Dandanvpbhp[00]: Host Echo enabled on port 11
15:43.32Dandananyone knows what that might be?
15:43.43Dandani have a terrible echo on the line..
15:43.55Dandanis that a bug? a feature?
15:43.58CaViCcHicoppice. So i have to try an autoalimented phone?
15:44.04znoG[TK]D-Fender: don't think its pastebin-worthy, so I'll paste here :)
15:44.08znoG[TK]D-Fender: ( *xx | #0 | [1-9]xxx | <0,:0> 99# | <0,:0> 08[01]0xxxxxxx | <0,:0> [456]xxxxxxx | <0,:0> 15xxxxxxxx | <0,:0> 1[01][013] | <0,:0> 0[23]xx15xxxxxx | <0,:0> 0[23]xxx15xxxxxx | <0,:0> 0[23]xx1[0-46-9]x. | <0,:0> 0[23]xxx[0-46-9]x. | <0,:0> 0[23]xxx1[0-46-9]x. | <0,:0> 0[23]xxxx[0-46-9]x. )
15:44.44znoG[TK]D-Fender: the problem is dialing a number like 0 002551511111111
15:44.45[TK]D-FenderznoG : OMG
15:45.03rpmthat is such a dirty dialplan
15:45.10znoG[TK]D-Fender: that should be matched by: <0,:0> 0[23]xx15xxxxxx
15:45.21znoGsorry
15:45.24znoGi mean:
15:45.28znoG0 0255151111111
15:45.43[TK]D-FenderznoG : Yeah I saw 1 too many "1"'s there
15:45.44znoGbut, instead, i can keep pressing digits and it never sends the dial
15:45.51*** join/#asterisk Martincit0 (n=martin@litigaractivos1.att.net.co)
15:46.02znoGonce it gets 6 x 1's, it should dial
15:46.10znoGbut instead it waits for the short digit timeout
15:46.24geoffb1974hello all
15:46.31geoffb1974so who is a SIP NAT genius??
15:46.35znoGmeaning, it matched some other entry
15:46.56Martincit0Pls, give me a tip to fix this error:
15:46.56Martincit0Feb  3 10:42:22 WARNING[10888]: chan_zap.c:6468 handle_init_event: Detected alarm on channel 2: Yellow Alarm
15:46.56Martincit0Feb  3 10:42:22 WARNING[10888]: chan_zap.c:1442 zt_disable_ec: Unable to disable echo cancellation on channel 2
15:46.59geoffb1974got my echo test to work yesterday
15:47.00geoffb1974yay
15:47.09sevardAnyone have a good deal of MySQL knowledge?  InnoDB: The error means mysqld does not have the access rights to the directory.   What directory is it complaining about?
15:47.29Martincit0my pri is working fine , but this error hangs all extentions
15:47.33slak-eh who uses relational dabatases these days
15:47.34slak-come on
15:47.34*** join/#asterisk flotox (n=jovan@host211-204.pool8541.interbusiness.it)
15:47.37slak-MODERNIZE
15:47.52rpmsevard: just check the permissions on the directory your db stores its stuff in.. usually /var/lib/mysql or something
15:47.56[TK]D-FenderznoG : Not sure I'm up to speed to help you on this one...
15:48.03*** join/#asterisk mhnoyes (n=mhnoyes@user-2ivfmto.dialup.mindspring.com)
15:48.04flotoxhi
15:48.05slak-!TK
15:48.08slak-you can help me
15:48.12rpmslak-: i prefer 400MB .csv files too
15:48.18[TK]D-Fenderslak- : Depends...
15:48.33slak-well TK
15:48.38slak-let me try and see if what i did worked
15:48.39sevardrpm: I don't know where it's pointing towards to find the databases
15:48.41slak-and then ill bother you sir
15:49.28rpmsevard: check the init scripts for mysqld, check the configuration in my.cnf wherever those files are.
15:50.02*** join/#asterisk bjohnson_ (n=bjohnson@jecinc.tor.istop.com)
15:52.04sevardrpm: and if I don't happen to have a my.cnf ?
15:53.13nestarlol
15:53.17*** part/#asterisk mhnoyes (n=mhnoyes@user-2ivfmto.dialup.mindspring.com)
15:53.35nestarno wonder i don't have any new mail!
15:53.38nestari'm on the wrong box
15:53.40nestar:o
15:54.08sevardnestar: do you have 3500 docs? :)
15:54.38*** join/#asterisk Johan (n=kvirc@194.151.113.2)
15:54.40JohanHi all
15:54.48nestarno, i probably have a stack of 2950 docs, that i've never opened..
15:55.02nestaryou probably need an IOS book, not a manual for the switch
15:55.05Beirdothat's a lot of docs to read
15:55.09mutis there anything ya can do about 'lagged' echo?
15:55.11nestarIOS stuff you can probably just find on the net.
15:55.23mutseems like ~500ms or so the echo comes in
15:55.27sevardnestar: hmmm
15:55.30*** join/#asterisk sulex (n=sulex@pdpc/supporter/active/sulex)
15:55.35mutand i can't get any of the zap echo cancellers to get rid of it
15:55.38sevardnestar: isn't there IOS specific manuals out there?
15:55.41jbalcombthis SPA-2002 has so many more options than the HandyTone-386 its freaking me out
15:55.45mutchanging echotraining taps can't get rid of it
15:55.59JohanI use AGI and agentcallbacklogin(). Now I wonder if it is possible to trigger an AGI script when an agent who's called back answers the line, so the AGI script/code knows who of the agents answerd the line. Is this possible?
15:56.04jbalcombmut what phones?
15:56.05nestarsevard: more than likely
15:56.19nestarthere's a lot of documentation you can read off cisco.com for free
15:56.27nestarif you have a service contract, there's even more.
15:56.46nestara service contract on a 3500 series switch is probably $100 for the year.
15:56.53mutspa-2002's
15:56.57[TK]D-Fendermut : What kind of interface is causing that much echo?
15:57.00sevardI asked a CCNA friend to check out cisco's website for 3500 docs and he said "cisco doesn't support it anymore and the documents aren't online"
15:57.11argentasI'm a bit confused about the +101 priority jump (on busy) following Dial in asterisk 1.0.x - does this only occur if the +101 priority exists or something?
15:57.11[TK]D-Fenderjbalcomb : The Sipura stuff is great....
15:57.28sevardand he had a stupid login for the website
15:57.37nestarthey haven't end of lifed the 3500 series, as far as i know
15:58.05jbalcomb[TK]D-Fender yeah, im digging it. im laughing now though cause i think it has the feature we need to have the fax devices counter our rxgain & txgain adjustments
15:58.15mut[tk]
15:58.56argentasi've got an asterisk box sat between a telco PRI and a separate pbx, just to do voicemail and call recording, and voicemail seems to work fine on busy even if i don't handle the +101 priority (ie just with the next priority after the dial command)
15:59.09muti have a spa-2002 to asterisk to pstn via te405p
15:59.17jbalcombsevard what 3500 model you looking for?
15:59.51nestarsevard: http://www.cisco.com/en/US/products/hw/switches/ps637/tsd_products_support_series_home.html
16:00.02mutnow the echo only comes in when i'm calling off net on verizon landlines
16:00.10mutif i call alltel cellphone there is no echo
16:00.14mutif i call verizon cell there is no echo
16:00.15nestaryour friend is either lazy, lying to you, or can't operate a browser
16:00.22mutif i call inside the local clec ther eis no echo
16:00.32Normanyone here use astlinux? when i create my keydisk on a usb drive, it doesn't seem to save anything
16:00.33mutif i call inside the office thru our adtran/pbx there is no echo
16:00.34nestari'm not logged in, and i can read that.
16:00.47nestarhttp://www.cisco.com/application/pdf/en/us/guest/products/ps637/c2001/ccmigration_09186a00800dc776.pdf
16:00.54*** join/#asterisk salviadud (n=ralfalfa@201.137.161.198)
16:01.12muti'm not really sure what else to try
16:01.19jbalcomb[TK]D-Fender they do not have an option to DTMF Tx via RFC?
16:01.20muti can't compile the mark2 echo cancellers because it fails
16:01.26mutfunctions not found or something
16:01.32muttried the kb2 and mg2 tho
16:01.34mutkb1
16:01.57*** part/#asterisk Setec_Astronomy (n=iamatall@69-173-5-230.clvdoh.adelphia.net)
16:01.59sevardjbalcomb: the 3500 itself
16:02.13sevardI just went online to cisco's website and found a BUNCH of docs on the 3500
16:02.17sevardI hate that guy.
16:02.19nestarlol
16:02.37sevardalright, so my next question is that the 3500 does support QoS it just uses an old name for it
16:02.43sevarddo you by chance know what they would call it?
16:02.59nestartry this one.. [11:00] <nestar> http://www.cisco.com/application/pdf/en/us/guest/products/ps637/c2001/ccmigration_09186a00800dc776.pdf
16:03.09sevarddownloading.
16:03.16*** join/#asterisk steve____ (n=steve@store-fw.porchlight.ca)
16:03.22nestarit's the main config guide
16:03.30nestari'm not sure if the 3500 does QoS at all
16:03.58nestarmight need a more expensive cisco for that.
16:04.05[TK]D-Fenderjbalcomb : Not from what I've seen.  use "INFO" for DTMF.
16:04.06nestarbbl, lunch.
16:04.09sevarddamn.
16:04.13sevardthanks for your help nestar
16:04.17nestarno problem
16:04.57nico__a
16:05.05[TK]D-Fendermut : I ditched 2 TE405P's here due to horrific echo problems.
16:05.23mutyea
16:05.54fulgascan't get the CDR(duration) and CDR(billsec) from an PHP AGI script...is there a problem there?
16:06.34Makenshianyone going to ukerna networkshop?
16:07.20brookshiremut: did you call digium support?
16:07.30muti emailed em
16:08.03geoffb1974soory I am aback now
16:08.11geoffb1974so anyone a NAT guru?
16:08.23mutthey just sent me a formmail reply
16:08.27[TK]D-Fendergeoffb1974 : Whats your question?
16:08.43muttelling me to do everything i exploained i had already done in the email
16:08.43geoffb1974getting only one way audio on new test box
16:08.45Makenshispeaking of nat, there is is a sip conntrack module for iptables in the linksys open firmware... noone's ported it yet
16:08.49geoffb1974have registred with FWD via IAX
16:08.51geoffb1974that works
16:08.59geoffb1974local in network sip call work fine
16:09.16geoffb1974but...calls from outside do not work
16:09.22geoffb1974I can hear them, they can't hear me
16:09.37geoffb1974port forwarding to my asterisk box through the router
16:09.38jbalcombsevard http://www.cisco.com/warp/public/473/139.html#q1
16:09.44[TK]D-Fendergeoffb1974 : I trust you added LOCALNET & EXTERNIP to sip.conf, and have forwarded the appropriate ports and disabled reinvites.....
16:09.53jbalcombsevard http://www.google.com/search?hl=en&lr=&q=cisco+3500+QoS
16:10.20*** join/#asterisk zpn (n=xpn@gateway.digium.com)
16:10.32geoffb1974yes indeed
16:10.58*** join/#asterisk MasterYoda (n=mnichols@gateway.digium.com)
16:11.05[TK]D-Fendergeoffb1974 : What ports are you forwarding?
16:11.07sevardjbalcomb: thank you very much.
16:11.30jbalcombsevard: doiitashimashite
16:11.59geoffb1974fender: looking now
16:12.06*** join/#asterisk jimmy_deanPB (n=jhodapp@indianalifesciences.com)
16:12.26jimmy_deanPB[TK]D-Fender, ping
16:13.05geoffb1974fender: I am forwarding 4569 on UDP and TCP for IAX to FWD
16:13.15[TK]D-Fenderjimmy_deanPB, pong
16:13.19filebbl
16:13.21geoffb19745004-5082 on UDP and TCP for SIP
16:13.30[TK]D-Fendergeoffb1974 : Exact ports and protocols please
16:13.32geoffb1974and 10000-20000 on udp and tcp for SIP
16:13.42geoffb1974those are ranges
16:13.54[TK]D-Fendergeoffb1974 : Ok, those look fine.... so you can receive calls?
16:14.03geoffb1974yes
16:14.03jimmy_deanPB[TK]D-Fender, is your asterisk system that you run rock solid or does it usually require hand holding?
16:14.09geoffb1974and I can hear them
16:14.10[TK]D-Fendergeoffb1974 : pastebin your sip.conf
16:14.13fulgas{CDR(billesec}) always get 0
16:14.17jimmy_deanPB[TK]D-Fender, and are you using it for VOIP at all with either SIP and/or IAX?
16:14.22fulgas{CDR(billesec))
16:14.22[TK]D-Fenderjimmy_deanPB : Depends whic, and why :)
16:14.25fulgaserrr
16:14.37fulgasbillsec
16:14.52jimmy_deanPB[TK]D-Fender, just trying to see if it's possible to get a solid Asterisk installation because ours has been having problems almost every day
16:14.55[TK]D-Fenderjimmy_deanPB : My work server is behind NAT and has remote clients (my home & agents for example), and PRI and SIP internal.
16:15.04[TK]D-Fenderjimmy_deanPB : What kind of problems?
16:15.11geoffb1974[general]
16:15.11geoffb1974port=5060
16:15.11geoffb1974bindaddr=0.0.0.0
16:15.11geoffb1974externip=x.x.x.x
16:15.11geoffb1974localnet=192.168.1.0/255.255.255.0
16:15.12geoffb1974srvlookup=yes
16:15.14geoffb1974qualify=2000
16:15.16geoffb1974nat=no
16:15.17[TK]D-Fendergeoffb1974 : PASTEBIN!
16:15.19[TK]D-Fender~pb
16:15.20jbotextra, extra, read all about it, pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca
16:15.27geoffb1974sorry, sorry
16:15.33[TK]D-Fendergeoffb1974 : nat=yes
16:15.55znoGi think i'll just settle for a wildcard dialplan and let asterisk handle the dialplan
16:16.00jimmy_deanPB[TK]D-Fender, sometimes asterisk can't seem to find the remote IAX VOIP provider (can't resolve it in DNS), it has been dropping calls too, and this morning for whatever reason all incoming calls just stopped working until I reloaded the config from the asterisk console
16:16.03znoGthats one thing that sucks about ATAs
16:16.08*** join/#asterisk shmaltz (n=mybox@mail.dmaven.com)
16:16.12*** join/#asterisk [Atlas] (n=whois@216.190.144.90)
16:16.47[TK]D-Fenderjimmy_deanPB : What interface are the dropped calls occurring under?
16:16.53jimmy_deanPB[TK]D-Fender, IAX
16:16.58[TK]D-FenderznoG : Thats for the best....
16:17.12[TK]D-Fenderjimmy_deanPB : I've heard a fair bit about IAX2 dropping calls....
16:17.12jimmy_deanPBthough I can't say that it's not dropping analog Zap channels either but I don't know that for sure
16:17.27[TK]D-Fenderjimmy_deanPB : What kind of card are you using for Zap?
16:17.28jimmy_deanPB[TK]D-Fender, well crap...how is that enterprise ready then?
16:17.39jimmy_deanPB[TK]D-Fender, digium's TDM400P
16:17.50*** join/#asterisk Kohat (n=Nothing@202.125.141.6)
16:17.50[TK]D-Fenderjimmy_deanPB : Who said it was?  And using what tech?  VoIP is not "reliable" in many terms
16:17.52geoffb1974fender: changed nat=yes
16:17.58geoffb1974and reloaded
16:17.59*** join/#asterisk mgf (n=mgf@r200-125-29-186-dialup.adsl.anteldata.net.uy)
16:18.11[TK]D-FenderMy TDM400P dropped calls like "Bambi on Ice"....
16:18.15geoffb1974Fender: do I need to nat = lines for each user
16:18.28[TK]D-Fendergeoffb1974 : Typically not if you have it in General.
16:18.43JunK-Ysome1 has successfully used waitforsilence?
16:19.47*** join/#asterisk skeffling (n=Andrew_H@andrew.1ec.aaisp.net.uk)
16:20.00geoffb1974ok
16:20.23geoffb1974fender : just so you know i am testing from behind a NAT at wotk as well
16:20.29geoffb1974work
16:20.39*** join/#asterisk coppice (n=chatzill@198.168.17.210.dyn.pacific.net.hk)
16:21.28[TK]D-Fendergeoffb1974 : I've done double-NAT'd scenario's here and they work fine...
16:21.35jimmy_deanPB[TK]D-Fender, well would we be better to go all analog lines or digital PRI lines?
16:21.38geoffb1974:fender : ok
16:22.07jimmy_deanPB[TK]D-Fender, surely it's possible to get a reliable VOIP setup
16:23.57jimmy_deanPB[TK]D-Fender, how could any enterprise go with a voip solution if it didn't have a 99.9999% uptime
16:24.18wunderkinjimmy_deanPB, what version of asterisk are you using
16:24.22[TK]D-Fenderjimmy_deanPB : Well digital offers you more possibilities and the tricky points of POTS go away.
16:24.36[TK]D-Fenderjimmy_deanPB : Who offers 5-9's on VoIP?
16:24.37jimmy_deanPBwunderkin, 1.2.4 compiled from source
16:24.38xachenWell you can't get a 99.9999% uptime on * for sure
16:24.56jimmy_deanPBxachen, why do you say that?
16:25.04xachenits still pretty unstable
16:25.30xachenlook at the whole timebomb incident, the fact that the manager crashes by sending too much, etc. It still has a long way to go
16:25.34xachen:)
16:25.56jimmy_deanPBgeez, so how does digium get anyone to install this in an enterprise solution?
16:26.04jimmy_deanPBmy boss is about ready to kill me over asterisk
16:26.24Qwelljimmy_deanPB: people who install in an enterprise generally know what they're doing..
16:26.28xachenyes
16:26.31wunderkini havent had any stability problems, nor manager issues and i use the manager extensively, only a couple manager connections at once but all the calls are generated through it, a lot of manager improvements were made, 1.2 should be stable for the manager
16:26.37_Paulo_My experience is that few services marketed as 5x9 have 99.999% uptime indeed.
16:26.38xachenYou can't just start reading one day and expect to master it the next
16:26.45xachenYou need to know how to work around the flakes
16:26.48MasterYodaxachen: the manager bugs have been largely fixed
16:27.12wunderkinjimmy_deanPB, you need to test before you go into production
16:27.14jimmy_deanPBxachen, yeah, well I've learned a lot by now
16:27.23[TK]D-Fenderjimmy_deanPB : You have to understand that VoIP is about comprimise and flexibility.  Stability is a price you pay.
16:27.27jimmy_deanPBwunderkin, you're telling me :)
16:28.02_Paulo_cheap, easy, stable: pick two.
16:28.12jimmy_deanPB[TK]D-Fender, well then what's a good uptime expectation? and how do I get a setup so that it can notify me when things aren't right instead of customers telling us that our phones aren't working?
16:28.20_Paulo_you cant have all 3.
16:29.49xachenyou could setup nagios with SIP monitoring
16:29.49mutwoah
16:29.49mutdigium hold music is load compared to their voice
16:30.02mutunless that was just a quiet chic
16:30.15jimmy_deanPBxachen, ooh, great idea
16:31.01Seldon1975asterisk, digium, stable-phone-system
16:31.04Seldon1975pick one
16:31.04MikeJ[Laptop]see.. this is all quite funny.. I know many commercial switches who may claim, but really don't have 5-9
16:31.35*** join/#asterisk FastJack (i=fastjack@p5091ECA3.dip.t-dialin.net)
16:31.38_Sam--you think you would notice the last 9 really?
16:31.42_Sam--if you were a customer
16:31.45blintbbt
16:31.48blintbye everyone
16:31.56MikeJ[Laptop]the fact is, your on standard hardware, so you have the uptime of that to start with... if you get a very redundant piece of hardware that is a good start
16:32.12xachenLoad balancing works
16:32.26MikeJ[Laptop]but you are not in a situation like you are on a nortel option 81 where you have full failover from 1 processor card to another..
16:32.32_Sam--my solid state asterisk box has 100% uptime.  but i just built it and started using it 9 days ago :)
16:32.34MikeJ[Laptop]that will require special stuff..
16:32.44MikeJ[Laptop]it is doable... I saw IBM do it with asterisk..
16:32.50xachenindeed
16:32.55MikeJ[Laptop]but no software package alone will do 5-9
16:32.57Seldon1975_Sam--: so you built the version w/o the memory leaks?
16:33.09_Sam--no debian package of 1.2.1.
16:33.22Seldon1975_Sam--: so it's a time bomb
16:33.31_Sam--1.2.1 is fine
16:33.41_Sam--at least ..it has been for 9 days and i have plenty ram free :)
16:33.55Seldon1975read the topic
16:34.04_Sam--yeah that is for 1.2.2 and up
16:34.08Seldon1975aah
16:34.14Seldon1975so the leak was introduced in 1.2.2
16:34.15_Sam--i could be wrong..usually am
16:34.19_Sam--but i think 1.2.1 is fine
16:34.25_Sam--afaik
16:34.35Seldon1975k
16:35.32Seldon1975well, you'll know in a few weeks
16:35.32cpmMan, i so love doing rj21 to 66 blocks
16:35.32Seldon1975:}
16:35.32muthm
16:35.32mutsupport must be busy or something
16:35.32mutnever been on hold for this long
16:35.32*** join/#asterisk coppice (n=chatzill@198.168.17.210.dyn.pacific.net.hk)
16:35.48Seldon1975mut: 15 mins +?
16:35.56*** join/#asterisk Katty (n=angela@64.82.232.54)
16:36.01Seldon1975meow
16:36.08mut~10
16:36.09jbotThere are 10 types of people in this world. Those who understand binary and those who don't.
16:36.18mutheh
16:36.23Seldon1975haha mut
16:36.44muti forgot i was on hold for a while
16:36.57Seldon1975mut: my record is 15
16:36.57mutjust doing other work and listenin to the mudic
16:37.06Qwell15...minutes?
16:37.06mutthen i realized what the music was
16:37.08QwellThat's it?
16:37.09Seldon1975yes
16:37.18mutfor digium or for anything..
16:37.30QwellI've been on hold for > 2 hours, on multiple occasions
16:37.39Seldon1975Qwell: to whom?
16:37.49Seldon1975check this out LOL http://thedailywtf.com/forums/58819/ShowPost.aspx
16:37.54QwellAdelphia, Sprint, UPS
16:38.01Seldon1975Qwell: scheisse
16:38.13muti know way back when i was on hold with packard bell for 3 hrs, then we called again and got put 'higher' in the queue
16:38.19mutand it was another 2 hrs
16:38.23*** join/#asterisk mroth_imm (n=mroth@63.65.26.220)
16:38.23*** join/#asterisk sack (n=sack@179.Red-83-50-147.dynamicIP.rima-tde.net)
16:38.39Seldon1975thats absurd
16:38.49mroth_immanyone have any experience with native MOH + ztdummy timing?
16:39.37_Paulo_I have timing issues with app_txfax... :-)
16:39.46mutwho doesn't
16:39.59JunK-Yi wanna stream a sound-file after 2000ms of blank (not before), how should i do that? waitforsilence(2000) needs the phone to be muted, detect ambient noise as talking, not silence.
16:40.16xachenerm
16:40.21xachenbefore you start the call process?
16:40.30*** join/#asterisk SibRw0rk (n=DaPhrek@66.234.235.84)
16:41.05[Atlas]Seldon1975: Hey bro!
16:41.22Seldon1975[Atlas]: hey
16:41.34Seldon1975[Atlas]: what up
16:42.04[Atlas]not much just fukin with my asterisk install trying to get it to compile with openssl in non default location
16:42.23badboyzanyone experienced the choppy music on hold, when another person is calling in? is this some sata related problem?
16:42.54[Atlas]it worked with a slackware package from slackware.org but on my own it dont like me :)
16:43.02*** part/#asterisk MasterYoda (n=mnichols@pdpc/supporter/sustaining/MasterYoda)
16:43.59mutyay got someone
16:44.36muthe's kinda silent th
16:44.39muto
16:44.39Seldon1975[Atlas]: that sux
16:44.45*** join/#asterisk eKo1 (n=bernd@metrored-gw.tropicohn.com)
16:45.17[Atlas]Seldon1975: eh kinda it keeps me from localizing our software so ..... im content to sit here na mess with it for a few days :D
16:45.20eKo1man, Asterisk 1.2.* is really progressing fast
16:46.01bweschke~seen shmaltz
16:46.06jbotshmaltz <n=mybox@mail.dmaven.com> was last seen on IRC in channel #asterisk, 1d 1h 30m 36s ago, saying: 'anybody here has a teleconferencing server available for the next 6 months for 10 conferences 1.5 hours per week?'.
16:46.06Seldon1975are many ppl using Debian Sarge for Asterisk?
16:46.18Dr-LinuxeKo1: great to know, what else things make it different and good?
16:46.27mutthis guy has no idea what he's doing
16:46.35mutit's like talking to my dialup techsupport
16:46.59tzangerok brainfart time... what's the 't' exension for
16:47.06Qwelltimeout
16:47.10tzangertimeout that was it
16:47.11[Atlas]Seldon1975: had it runnin on sarge but not in production -- trying to get us away from slackware for pkg reasons
16:47.11tzangerthank you
16:48.01Seldon1975[Atlas]:k, im just building a second box for * w/ Deb Sarge cause Digium have some new experimental hardware for us to try out to address crosstalk
16:48.11mroth_immasterisk + debian was a VERY painful experience, imho
16:48.19mutuh
16:48.25Seldon1975mroth_imm: why?
16:48.32mutthats all i run is * & debian
16:48.46mutwasn't painful in any sense of the word
16:48.52mroth_immyeah, it's doable, and i did it, but it's more difficult than other distros...now i use fc
16:49.01mroth_imm(mainly picked because of ABE)
16:49.39mroth_immit was a relatively long time ago, but i just remember really needing to do a lot of digging for packages and such
16:49.48_Sam--Seldon1975:  i have 3 debian sarge asterisk boxes
16:49.50Seldon1975mroth_imm: Fedora Core 4?
16:50.04mroth_immnah, using 3, abe likes fc3 and rhel3 only
16:50.09Seldon1975mroth_imm: someone (D-Fender?) said FC4 sucked
16:50.10Seldon1975ok
16:50.15mutthat just means whatever distro ya have installed more shit by default that ya don't need
16:50.21Seldon1975_Sam--: any setup issues?
16:50.22*** join/#asterisk EriSan (n=erisan@81-174-35-6.f5.ngi.it)
16:50.23*** join/#asterisk justinu (n=Justin@cpe-72-129-86-208.socal.res.rr.com)
16:50.33_Sam--Seldon1975:  none ever for me...but i am a debian zealot.
16:50.36mroth_immbut asterisk was the easy part...amp and xc-ast (queuemetrics) were incredibly painful to get up
16:50.37AndyCapforcing one distros semantics on another is always painful.
16:50.55Seldon1975ok Ive got a really basic config for *
16:51.14*** join/#asterisk nextime (n=nextime@213-140-6-103.ip.fastwebnet.it)
16:51.15Seldon1975only add-on I use is for MP3 MOH
16:51.19[TK]D-FenderSeldon1975 : Its hearsay really... FC4 uses a version of GCC that causes problems and a few other things.  Basically its far more trouble than its worth (so far)
16:51.52Seldon1975cool
16:51.56Seldon1975sounds like I'll be ok
16:52.01mroth_immhttp://www.voip-info.org/wiki/view/Running+Asterisk+on+Debian
16:52.04Seldon1975'course I know where to come for help now..
16:52.12mroth_immthose are my notes from asterisk + deb days
16:52.14mutwow
16:52.17Seldon1975thx
16:52.18_Sam--its the same thing that comes up every day...asterisk works fine on any distribution!
16:52.23mutso hard to get good tech support these days
16:52.24mutit sucks
16:52.29_Sam--if you like debian and know it...then its good for you!
16:52.34mroth_immi'm gonna try to do Asterisk on Knoppix next...figure that'll be about as nasty ;)
16:52.47_Sam--simple
16:52.48Seldon1975mroth_imm: knoppix!? why
16:52.55_Sam--i have asterisk with fluxbox on debian
16:52.59*** join/#asterisk oej (n=oej@apollo.webway.se)
16:53.08mroth_imm_Sam--: you are probably right, it was my first asterisk install, so probably wasn't a good choice to use Deb
16:53.11_Sam--i started with knoppix but its too bloated
16:53.13*** join/#asterisk mogorman (i=root@user-24-236-84-48.knology.net)
16:53.31mroth_immSeldon1975: just for the hell of it : )...don't feel like installing a full distro to work from home
16:53.45Seldon1975k
16:54.02mroth_imm_Sam--: Our FC3 box is very slim...we're shooting for 250 - 500 simultaneous calls with digital recording on it
16:54.20_Sam--that is quite a bit of calls...using what type of channels?
16:54.29Seldon1975maybe you can help me now
16:54.31Seldon1975I did: apt-get install xserver-svga
16:54.38mroth_immall calls sip to sip
16:54.41Seldon1975says no installation candidate
16:55.02mroth_immSeldon1975: it may be outdated...i pretty much abandoned deb when we went to abe
16:55.08_Sam--thats cause xserver-svga inst a valid package
16:55.11mroth_immall calls ulaw codec, pcm recordings
16:55.20mutheh
16:55.21mutno wonder
16:55.26_Sam--at least, its not in any of my list of debian sources.
16:55.31_Sam--and i have a big list of debian apt sources
16:55.41Seldon1975_Sam--, mroth_imm: I got that package name form your doco
16:55.48Seldon1975http://home.comcast.net/~mroth01/phase1.html#conf-desktop
16:56.09_Sam--thats not my doc :)
16:56.11Seldon1975_Sam--: can you tell me how to install xserver
16:56.18Seldon1975_Sam--: sorry that was for MrMagic
16:56.23Seldon1975_Sam--: sorry that was for mroth_imm
16:56.25mroth_immi know, i haven't updated that stuff in ages...sorry if it is no longer accurate
16:56.31Seldon1975ok
16:56.33Seldon1975no worries
16:56.34*** join/#asterisk Tuttle_ (n=Tuttle@kelinat210.keli.cz)
16:56.46Seldon1975but do you know the current way to install xserve
16:56.49mroth_immSeldon1975: if i remember right, you should be able to search using apt to find your stuff
16:56.49*** part/#asterisk zpn (n=xpn@gateway.digium.com)
16:56.57_Sam--apt-get install xfree86-common - X Window System infrastructure transitional package
16:56.57_Sam--xserver-xfree86 - transitional package for moving from xfree86 to X.Org
16:57.00_Sam--er
16:57.05Seldon1975thank you
16:57.12_Sam--that is for the xfree server
16:57.18_Sam--which i use...you can use whatever x server you want
16:57.27mroth_immi should just remove those docs ;(
16:57.59mutwheres digium located?
16:58.03Seldon1975_Sam--: its installed now how do I start x?
16:58.03Qwellmut: AL
16:58.06mroth_imm_Sam--: right now 58 simultaneous SIP channels (around 30 sip to sip bridged calls) 95%idle
16:58.12mutah
16:58.14muttoo far to move
16:58.17_Sam--dude i am not going to be your personal debian guide...i have a real job :)
16:58.27_Sam--but first you will need to configure it :)
16:58.28Seldon1975_Sam--: come on, it will be great fun
16:58.33_Sam--then there is a funky command...called startx
16:58.48Seldon1975_Sam--: I tried that
16:58.50*** join/#asterisk r0d3nt (i=nobody@wsip-24-234-241-145.lv.lv.cox.net)
16:58.53mroth_immSeldon1975: that's why deb is tough if you're new the learning curve is pretty steep
16:58.56_Sam--it doesnt start because you havent configured it.
16:59.02Seldon1975_Sam--: ok
16:59.03Tuttle_when I install Asterisk on some server that is accessible with public IP from everywhere, will it interconnect two sw phones, both behind NAT and firewall restricting most of the output ports (configurable static port group) ?
16:59.10Seldon1975_Sam--: I'll battle on by myself
16:59.16Seldon1975_Sam--: (sniff)
16:59.19_Sam--type xf and hit tab from a bash prompt
16:59.25_Sam--it will will show you somethings you can run
16:59.28_Sam--like
16:59.32_Sam--xf86cfg
16:59.34_Sam--etc
17:00.03Seldon1975thx
17:00.14_Sam--i do have an easier method...i have a debian, xwindows base with asterisk.
17:00.21_Sam--you can download it if you want
17:00.25_Sam--but its meant for a compact flash card
17:00.29jbalcomb[TK]D-Fender iCEBrkr wawoot! got my analog phone working thru asterisk with my new SPA-2002!
17:00.48*** join/#asterisk nico__ (n=nico@34.101.94.80.dslz1.libello.mc)
17:00.58mroth_immyou still stuck with a 2.4 kernel in deb?
17:01.02_Sam--hell no
17:01.04nico__<PROTECTED>
17:01.13Seldon1975_Sam--: yes please
17:01.19Seldon1975_Sam--: can I d/l it
17:01.27skefflingsilly question, is there a way to see which version of zaptel I'm actually running?
17:01.44*** join/#asterisk stevek (n=stevek@slim-eth0.horizonlive.net)
17:01.51Seldon1975mroth_imm: no 2.6\
17:01.55_Sam--Seldon1975:  you are adept enough to untar something to a new partition, and make that partition bootable?
17:02.08_Sam--because what i have is a whole filesystem ready to go
17:02.14_Sam--it just needs to be untarred and booted
17:02.24_Paulo_skeffling, modinfo zaptel
17:02.33Seldon1975_Sam--: no, lol thanks anyway I'll just get * from SVN on this system
17:02.40Seldon1975_Sam--: but thx
17:02.45_Sam--apt-get works fine for debian
17:02.49_Sam--if you have the right apt sources
17:02.53_Sam--i mean for asterisk
17:02.53xachenapt-get moo :)
17:02.56_Sam--apt-get asterisk
17:03.11_Sam--Package: asterisk
17:03.11_Sam--Versions:
17:03.12_Sam--1:1.2.1.dfsg-3(/var/lib/apt/lists/ftp.de.debian.org_pub_debian_dists_testing_main_binary-i386_Packages)
17:03.25Seldon1975_Sam--: nice!
17:03.35skeffling_Paulo_, thanks, I'll try taht
17:03.55Seldon1975_Sam--: is that * version 1.2.1?
17:04.01*** join/#asterisk Jack2B (i=TRAX_@62.162.228.111)
17:04.04_Sam--yep
17:04.10Seldon1975_Sam--: oh
17:04.13Seldon1975_Sam--: i want 1.2.4
17:04.16Jack2BHi to all...
17:04.26_Sam--fair enough
17:04.30_Sam--if you want to be bleeding edge
17:04.32Seldon1975_Sam--: i'll SVN it down
17:04.35_Sam--and worry that your stuff doesnt work tomorrow
17:04.41_Sam--i know my stuff works :)
17:04.42skefflingpauldy, modinfo zpatel shows blank for the version!
17:05.00Jack2BIs it possible to connect quintum on asterisk?
17:05.10*** join/#asterisk websae (n=websae@h69-129-132-18.69-129.unk.tds.net)
17:05.41Tuttle_Q: When I install Asterisk on some server that is accessible with public IP from everywhere, will it interconnect two sw phones, both behind NAT and firewall restricting most of the output ports (configurable static port group) ?
17:06.04[TK]D-FenderTuttle_ : Yes
17:06.17Seldon1975_Sam--: later versions addressed echo problems we were having
17:06.26Seldon1975_Sam--: so I dont have much choice
17:06.45_Sam--i dont know...but i dont think between 1.2.1 and 1.2.4 that echo stuff was introduced
17:06.55_Sam--it was all bug fixing...but i could be wrong...you  have a changelong?
17:07.02Tuttle_[TK]D-Fender:  How can Asterisk make a client C ring, when C is behind NAT ?
17:07.16Seldon1975the Digium guys went in to our box and upgraded us from 1.2.1... echo went away
17:07.23Seldon1975thats all I know
17:07.30fugitivoSeldon1975: from 1.2.1 to what?
17:07.36Seldon19751.2.2
17:07.50Seldon1975and now I need 1.2.4
17:07.50fugitivowhen?
17:07.51_Sam--interesting...i run about 1500 minutes a day through 1.2.1
17:07.53Tuttle_[TK]D-Fender:  Is it true that another Asterisk must be run in LAN where reside all clients that should be able to ring (receive calls)?
17:07.59fugitivoi'm using 1.2.1 too
17:08.01Seldon1975about 3 weeks ago
17:08.02_Paulo_Seldon1975,  bleeding edge can hurt you...
17:08.03Seldon1975or so
17:08.14[TK]D-FenderTuttle_ : You need to tell * that the client is behind NAT, and the client needs to know as well.  Set the nat=yes, Qualify=2000 and that should do it.
17:08.16Seldon1975_Paulo_: it has - many times
17:08.30_Sam--ive had no problems with my 1.2.1's on 3 machines
17:08.31fugitivobut 1.2.2 is known to have problems
17:08.32_Sam--and im not changing
17:08.33[TK]D-FenderTuttle_ : No you DON'T need another * on the receiving side for that.
17:08.34fugitivo1.2.3 too
17:08.46fugitivo1.2.1
17:08.51_Sam--i would like to see a changelog to see if there was any echo changes done
17:08.52Seldon1975yes, I'm hoping 1.2.4 is the last upgrade I'll need in a while
17:08.52Tuttle_[TK]D-Fender:  So the clients maintains a persistent TCP/IP connection to *, waiting for calls?
17:08.53*** join/#asterisk exstatic1 (i=exstatic@redline.mednor.net)
17:08.53fugitivoall versions have problems
17:09.05*** join/#asterisk Wuntherdag (n=alexthew@rrcs-24-227-188-230.sw.biz.rr.com)
17:09.21*** join/#asterisk Samoied (n=Samoied@popeye.opens.com.br)
17:09.29_Paulo_Seldon1975,  I'm having issues with app_txfax: tryed last snapshots, they are segfaulting in my system...
17:10.08sevardIn What Asterisk config file is the path set for the mysql binary?
17:10.17Seldon1975_Paulo_: I consider all of Asterisk 'bleeding edge'
17:10.33_Paulo_Seldon1975, true indeed.
17:10.57_Sam--naw
17:11.46[Atlas]Seldon1975: LOL very true ;p
17:13.23WuntherdagIs a PIII 800 Mhz with 376 M ram ok for 2 TDM400's and 10 sip extensions?
17:14.04*** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-179-75-202.cust.bezeqint.net)
17:14.04[TK]D-Fenderwundaboy : should be fine
17:14.24[TK]D-FenderTuttle_ : No.  SIP is *UDP*.
17:14.24Wuntherdagthanks :)
17:14.45*** join/#asterisk aminorex (n=tony@71-13-40-131.dhcp.dlth.mn.charter.com)
17:14.54[TK]D-FenderWuntherdag <- that was for you.... stupid auto-complete.... I wish they'd have it so it matched based on LAST spoken...
17:15.14Jack2BIs it possible to connect quintum on asterisk, so billing of incoming call is done on asterisk?
17:15.17Wuntherdagnp just thought you were being cute
17:15.45Seldon1975[TK]D-Fender: now that would be a neat trick
17:15.57justinuhey, i run sip over TCP
17:16.00justinujust to be cool :P
17:16.07Seldon1975u da man
17:16.12[TK]D-FenderSeldon1975 : nothing special to do that....
17:16.21[TK]D-Fenderjustinu : FINE be that way!
17:16.28[TK]D-Fender:p
17:16.31justinuSeldon1975: i'm the man who keeps you down
17:16.50Tuttle_[TK]D-Fender:  So I think there must be some keep-alive packets to let the connection open (on the NAT firewall), so the call can be passed behind that firewall to LAN.
17:17.06*** join/#asterisk PupenoL (n=pupeno@200.123.183.89)
17:17.39Seldon1975justinu: no, thats [Atlas]
17:18.21Seldon1975lol
17:19.02[TK]D-FenderTuttle_ : Yes, thats the point of the "nat=yes" and "qualify=2000" I mentioned.  Also letting the device know to reregister and a faster interval.
17:19.05*** join/#asterisk gaspiz (n=gaspiz@86.35.77.154)
17:19.17justinui finally got SER to do clustering right...life is good
17:19.31Tuttle_[TK]D-Fender: I'm glad my deduction were right. Thank you.
17:19.47justinuharry tuttle?
17:19.59justinunot to be confused with buttle...
17:19.59Tuttle_captain one :)
17:20.21Tuttle_everyone thinks i'm harry :-)
17:20.27justinuhe was pretty cool
17:20.55justinuspeaking of that, I figured out the tagline for my business... "Central Services. You do the work, we do the pleasure!"
17:20.56*** join/#asterisk xtr (n=01928375@S0106000c41ed11e1.vf.shawcable.net)
17:20.57[Atlas]hey what  did i miss ,, who am i bringin down ,, wha wha??
17:21.05SkramXshido6: around?
17:21.10gaspizhi, can anyone give me an estimate value of how many paralell calls does asterisk suport with no transcoding or protocol change? the box is a p4 2.4 GHz with 1GB ram
17:21.27[Atlas]im being liquidated??
17:21.30justinuzaptel, or sip?
17:21.34gaspizsip
17:21.44Seldon1975[Atlas]: apparently
17:21.54justinugaspiz: i'd plan on around 120-140
17:21.58[Atlas]ive never been liquidated before
17:22.03[Atlas]i hope its fun
17:22.13justinuit's not... i've been reading this book about stasi
17:22.16Hmmhesaysoff topic, any audiophiles in here right now?
17:22.17[Atlas]im not much for pain -- it hurts me
17:22.24Seldon1975[Atlas]: you're like a turkish rug that used to sell for $80,000
17:22.31gaspizjustinu: Thanks
17:22.31justinui have an adcom GDA600, does that make me an audiophile?
17:22.38Seldon1975[Atlas]: now only $2000
17:23.11[Atlas]all this fun at my expense... wow hoh exciting
17:23.14[Atlas]how*
17:23.48gaspizjustinu: any ideas for SER? i will use it only load balance
17:23.54justinulots of ideas
17:23.58justinuhow much money you got?
17:24.26*** join/#asterisk chrish01 (n=chrish01@wsip-70-183-17-68.oc.oc.cox.net)
17:24.27gaspizjustinu: number of pcalls with ser using the same configuration box
17:24.29justinuif you mean ser capacity for simultaneous calls, fergitabboutit
17:24.32justinua LOT
17:24.34sawhrm. guys, are there any sample files for the digital receptionist out there?
17:24.48justinuser is lightweight, and doesn't have anything to do with the media plane
17:24.51justinuso it should scale well
17:25.07chrish01can anyone give me a pointer on how to setup an extension that dials a different person based on the time of day/week/month?
17:25.18fileI've actually been pondering writing a SER load balancing module... giving it out for free with documentation for Asterisk users for load balancing
17:25.29justinuchris01: gotoiftime
17:25.31filebecause if you didn't record route, only the first INVITE would be seen by SER...
17:25.32gaspizjustinu: so a lot means 1000+?
17:25.37chrish01justinu, thanks
17:25.41justinugaspiz: i would think so
17:26.04justinui'm using SER to load balance right now, but it's going thru the TM module
17:26.07gaspizjustinu: great, I'm on the right track then
17:26.13justinuand I figured out my problem the other night file
17:26.21filejustinu: ooh
17:26.45justinuit turns out SER is pretty cool... but it relies on your DNS to be configured right
17:26.51fileha
17:27.06filejustinu: if you want something else to play with, go download this http://quark.file-radio.com/ser/ser-gemini.tgz and look at the gemini module...
17:27.18justinuso I got it doing a TCP/UDP conversion, and loadbalancing
17:27.33*** join/#asterisk trelane_ (i=trelane@66.93.203.199)
17:27.35justinuthe only issue I have at this point is having phones register with SER instead of ast...
17:27.39justinuNAT issues
17:27.45justinui need to work on nathelper
17:28.00chrish01justinu, any idea if there is a way to use GotoIfTime with an external text file?
17:28.11justinuno idea, sorry
17:28.14chrish01np
17:28.55SkramXwhat does the jitterbuffer do really? should i have =yes or =no?
17:29.32justinujitterbuffer buffers jitter out of incoming RTP packets
17:29.46justinumeaning it corrects the interarrival time of delayed packets before sending them out again
17:30.04justinuwhich of course adds latency into the RTP relay process
17:31.07SkramXjustinu: so its best to have it.. disabled?
17:31.18justinudepends on what you want to do
17:31.22SkramX...
17:31.26justinui think that jitterbuffer you're talking about is IAX only
17:31.31SkramXRight
17:31.42justinuare you taking IAX calls and gatewaying them to PSTN?
17:31.45justinuwith a zaptel card, or something?
17:31.48SkramXnop
17:31.49SkramXe
17:31.58SkramXtaking incoming, and possibly doing outgoing, but all virtual.
17:32.02SkramXno hardware
17:32.24justinuk... if you have clients connecting via Internet, you may consider leaving it on
17:32.38*** part/#asterisk RoyK (n=roy@80.239.107.70)
17:32.41justinubut if your link to your PSTN gateway is big I Internet also, it might be pointless
17:32.50justinudepends on link quality, etc.
17:33.06justinuit's voip, there's never a simple answer
17:34.25*** join/#asterisk roulduke_ (i=m4vc2wzm@p508D0BED.dip0.t-ipconnect.de)
17:39.42*** join/#asterisk s34n (n=s34n@ip-206-159-190-125.mvdsl.com)
17:41.46*** join/#asterisk Reko_ (n=nonsense@cpc4-hem15-0-0-cust755.lutn.cable.ntl.com)
17:41.48Wuntherdagnoob? Is it possible to direct calls from the PSTN to certain extensions based on CID?
17:42.15darkskiezyes
17:42.16mdaveexten => s/calleridinfo,1,Dial(etc)
17:42.30mdavesubstitute appropriate values for s, 1, and etc
17:42.34[TK]D-FenderWuntherdag : plenty of ways.  that method is one.
17:42.46*** join/#asterisk pengyong (n=lala@218.108.12.98)
17:42.49mdaveand calleridinfo, for that matter
17:43.41sawany MeetMe users alive?
17:44.30Wuntherdagok thanks, we have no DID, so that means that the girl in accounting that gets 40 calls a day from her BF could get them routed to her ext
17:45.02zoayeah, send the boyfriend calls to her ex
17:45.15Wuntherdagrawr
17:45.21zoathat will teach em!
17:45.31mutor tell her to stop wasting time at work
17:45.44muti'de be firing that woman
17:46.23Wuntherdagno accounting girl = no paycheck for me :(
17:46.33*** join/#asterisk gaspiz (n=gaspiz@86.35.77.154)
17:46.35mutget a new one
17:46.41mutmaybe you'll start getting bigger paychecks
17:46.59Wuntherdaghehe, not my call but you are prb right
17:47.44fugitivoi'm using monitor-join=yes for my queues but it isn't joining the files, any idea?
17:48.13gaspizfor more paralell calls what's better: better CPU or more RAM?
17:50.06Qwellgaspiz: which your machine is currently lacking in.  It takes research
17:50.38Qwellwhichever*
17:50.48justinuasterisk runs into a wall at around 15mbit/sec of RTP thruput
17:50.53*** join/#asterisk pengyong (n=lala@218.108.12.98)
17:51.06justinuadditional CPU/RAM seem to help little
17:51.14*** join/#asterisk jonathh (n=asd@host86-142-221-157.range86-142.btcentralplus.com)
17:51.45gaspizfor g711 it's 64kbit/sec so I can calculate it from there ?
17:52.03Dr-LinuxWuntherdag: that girl has 40 boyfriends? :O
17:52.48Dr-LinuxWuntherdag: what kind of office you guys have? :S
17:52.48sevardERROR 1045 (28000): Access denied for user 'root'@'localhost' (using password: NO)
17:52.51gaspizjustinu: if yes it's 234 calls
17:52.58sevardhow do I reset ALLLLLL of mysql's passwords
17:53.20*** part/#asterisk jonathh (n=asd@host86-142-221-157.range86-142.btcentralplus.com)
17:53.29justinui never actually got 234 calls out of a more powerful system than you have
17:53.35justinuit hit a wall around 180
17:54.00Qwelljustinu: asterisk hits a wall, or your NIC/switch does?
17:54.05justinuasterisk
17:54.12justinuwell, something on the host
17:54.19justinuit's not the NIC, because I can transfer files much faster
17:54.41justinui think it's because the RTP proxy is in userspace
17:54.45justinutoo much context switching
17:54.45tzafrir_laptopsevard, mysql?
17:54.52Qwelljustinu: perhaps
17:54.55gaspizjustinu: so for safe running if I plan 100 calls it should resist to some load spikes too?
17:54.59tzafrir_laptopsevard, is that a Debian system, by any chance?
17:54.59WuntherdagDr-Linux : she has more than one, but the one that calls, calls a lot, must be gudstuff
17:55.12justinugaspiz: yeah, it should be happy as a clam at 100 g711 calls
17:55.55Dr-LinuxWuntherdag: then the girl is not good, change her with new one
17:55.56tzafrir_laptopsevard, you can start mysql with the authentication part disabled, so it will let you login as whoever you'd like
17:56.21sevardtzafrir_laptop: yes, I do mysql --password, it asks for a password, i put in 'hello' and it  gives ERROR 1045 (28000): Access denied for user 'root'@'localhost' (using password: YES)
17:56.32tzafrir_laptopsevard, also, On Debian there's a sepacial priviliged mysql user for system cron jobs and alike
17:56.38sevardhow would I start it with authentication disabled?
17:56.39*** join/#asterisk websae (n=websae@h69-129-132-18.69-129.unk.tds.net)
17:56.48*** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com)
17:56.53tzafrir_laptopsevard, that field in the database holds the md5-hash of the password
17:57.09fugitivodid anyone realize that newer asterisk releases (1.2.x) and maybe older have an incomplete set of audios?
17:57.19Wuntherdagnot referencing her biz skills, i am talking about her personal skills, hehe
17:57.21fugitivofor example, vm-youhaveno is missing
17:57.24sevardtzafrir_laptop: in ibdata ?
17:57.42fugitivoso you call voicemailmain and if you don't have any voicemail it'll hangup without any message
17:58.32Dr-LinuxWuntherdag: is she not a whore ? :O
17:58.59Dr-LinuxWuntherdag: and i think you LIKE her .. :S
17:59.15*** join/#asterisk Defraz_ (n=t0tal@tim.mychoice.cc)
17:59.22tzafrir_laptopsevard, what is bdata? Looking for that switch name...
17:59.25*** join/#asterisk pa (n=Paolo@unaffiliated/pa)
17:59.40*** part/#asterisk Utah_Dave (n=boucha@0-2pool130-165.nas28.salt-lake-city1.ut.us.da.qwest.net)
18:00.38_Paulo_Do I have to enable something int the linux 2.6 kernel to use asterisk "-p"???
18:00.47_Paulo_s/int/in/
18:00.59sevardtzafrir_laptop: I don't know.  It's in the database dir ibdata1 and iblogfile1, i thought it had to do with configurations and have the password in there, but they hold binary data.
18:01.37sevardWow, jbot does sed edits? omg that's AWESOME
18:02.33tzafrir_laptopsevard, the switch is '--skip-grant-table' . Naturally it is dangerous!
18:02.49sevardwith that switch can I throw a _new_ password at it?
18:03.20_Paulo_sevard, yes.
18:03.36tzafrir_laptopsevard, google produced the following, which seems correct: http://www.spikesource.com/docs/cs_1.6.2-linux/doc/mysql41/mysql41_troubleshoot.html
18:03.53tzafrir_laptopis jbot sed-compliant?
18:03.55_Paulo_sevard, I advice to turn tcp off in my.cns
18:04.25_Paulo_sevard, before using --skip-grant-table
18:04.28tzafrir_laptops/is [jk]bot /am I/
18:04.53sevard_Paulo_: ?>
18:05.10fugitivotzafrir_laptop: it doesn't work, i tried the other day
18:05.11*** join/#asterisk razu_ (n=razu@217-159-187-162-dsl.prn.estpak.ee)
18:05.50tzafrir_laptopthe following might have been useful:
18:06.15tzafrir_laptopso who's that jbot anyway?
18:06.22tzafrir_laptop$d
18:06.26Dr-Linuxbot
18:06.36_Paulo_sevard, with --skip-grant-table you will connect without giving any password.
18:07.00sevardbut you wanted to turn off something in cns
18:07.03sevardoh, outgoing connections
18:07.03tzafrir_laptopDr-Linux, tht line was the target for $d...
18:07.13sevardno worries, it's NATd twice :/
18:07.20tzafrir_laptops/my.cns/my.cnf/ ?
18:07.21sevardfirewalled 3x nothing in or out yet, heh
18:07.31_Paulo_sevard, s/my.cns/my.conf/
18:07.31sevardthis is a line here and I like soda
18:07.39sevards/soda/apples
18:07.43sevards/soda/apples/
18:07.46Dr-Linuxi don't understand
18:07.47Qwellheh
18:07.47sevardNEAT
18:07.51sevardthis is a line here and I like soda
18:07.52Qwellstop comparing apples to apples
18:07.53sevards/soda/apples/
18:07.59sevardthat is so cool.
18:08.11Qwelltest
18:08.17Qwells/e/o/
18:08.20Qwells/s/b/
18:08.24Qwell:D
18:08.24sevardhahaha
18:08.26Dr-Linuxjbot: who is your father? qwell? twisted[asteria]?
18:08.29jbotDr-Linux: what are you talking about?
18:08.29Qwellhe's good
18:08.36*** part/#asterisk mogorman (i=root@user-24-236-84-48.knology.net)
18:08.44sevardI really like that function in that bot.
18:08.51*** join/#asterisk Tili (i=Tili@202-133-67-25-dialup.sat.net.pk)
18:08.55Dr-Linuxjbot is a nice eggdrop :P
18:09.04QwellI don't think he's an egg
18:09.18*** join/#asterisk klictel (n=klictel@207.107.208.137)
18:09.52sevard_Paulo_: after I start mysqld_safe with --skip-grant-table how do I reset the password?  with mysqld --password ?
18:10.01*** part/#asterisk Skkip (n=Skipper@216.160.91.91)
18:10.03*** join/#asterisk Cisco_VPN_God (n=cisco_ki@adsl-69-225-45-137.dsl.skt2ca.pacbell.net)
18:10.08Cisco_VPN_GodHello
18:10.43tzangerheh
18:10.46tzangerSome people call them "cars" or "trucks"; I call them "dimensional transmogrifiers" because they change three-dimensional cats into two-dimensional ones.
18:10.54Nuggetto some people all bots are eggrdops, all facial tissues are kleenex, all photocopiers are xeroxes, all unixes are "linux", and all databases are mysql.
18:11.03*** join/#asterisk darein (n=kir@62.64.107.70)
18:11.27xachentzanger: haha
18:11.37sevardtzanger: Nice.
18:11.39Cisco_VPN_GodAny one have a solution for the following problem : SIP Client ---> NAT --> Internet <-- NAT w/ port Forwarding <----> Asterisk?
18:11.54tzangerCisco_VPN_God: yeah: IAX2.
18:11.56_Paulo_sevard, grant all on mysql to root@localhost
18:11.56NuggetSurely a VPN God such as yourself can solve that one.  :)
18:12.20Cisco_VPN_Godtzanger, I know, but Cisco 7960 does not support it
18:12.23sevard_Paulo_: I'm not sure what you're saying.  Is that the command I throw at mysql> prompt?
18:12.30tzangerCisco_VPN_God: set up an IP tunnel then
18:12.42Qwell7960 supports NAT..
18:12.43Cisco_VPN_GodWithout IPSEC
18:12.54tzangerCisco_VPN_God: if you're going to keep adding criteria nobody will help you
18:12.54_Paulo_sevard, grant all on mysql.* to root@localhost identified by 'newpassword'
18:13.05_Paulo_sevard, but go to #mysql !
18:13.11_Paulo_:-)
18:13.17sevard_Paulo_: <3
18:13.25Cisco_VPN_GodOk let me rephrase
18:13.27chrish01Cisco_VPN_God, ipsec is fine as long as you keep your latency under about 70ms
18:13.34chrish01Cisco_VPN_God, ive done it that way for a few customers
18:13.58chrish01thats 70ms one way btw
18:14.32Dr-LinuxCisco_VPN_God: yes
18:14.42rob0OpenVPN is the way to go with IP tunnels. SSL encryption, wide OS support.
18:14.53rob0nat friendly too
18:14.54Cisco_VPN_GodI want to take my phone to any customer's site and make and recieve calls without the need to establish a VPN tunnel.  Currently I can call out with no problems, but asterisk will not initiate a new call to my SIP phone
18:14.54sevard_Paulo_: :( ERROR 1290 (HY000): The MySQL server is running with the --skip-grant-tables option so it cannot execute this statement
18:15.22NuggetCisco_VPN_God: I just run asterisk on my laptop and connect my sip phone to the local asterisk then have it talk iax to my "real" asterisk server.
18:15.44_Paulo_sevard, "use mysql"
18:15.57_Paulo_sevard, "select * from user;"
18:16.02Cisco_VPN_GodNugget - I can get SIP soft phone to work with no problems with Cisco IP VPN.
18:16.13*** part/#asterisk Optic (n=dfraser@miso.capybara.org)
18:16.24NuggetI was just offering a possible solution, not inviting you to further complicate the scenario.
18:16.26_Paulo_sevard, "show tables from mysql;"
18:16.57Cisco_VPN_GodI just would like to to use my Cisc IP Phone to receive calls.  Any takers?  I am willing to pay for a solution, if one exists, I was just asking.
18:17.15NuggetI offered one such solution.
18:17.19Nuggetwhere's my fucking money?
18:17.20sevard_Paulo_: then the password command?
18:17.26tzangerCisco_VPN_God: there are numerous solutions but double-nat is always a problem
18:17.27_Paulo_sevard, these tables keep authorization stuff.
18:17.37*** join/#asterisk MrBrown (n=owned@68-169-204-147.agstme.adelphia.net)
18:17.43tzangeryou can register to a third (public-ip) SER server and eliminate most of this though
18:18.01MrBrownis this a good deal? http://cgi.ebay.com/Hosted-Asterisk-PBX-Polycom-IP-501-601-with-Dial-Tone_W0QQitemZ5862473555QQcategoryZ11908QQrdZ1QQcmdZViewItem
18:18.03_Paulo_sevard, you will have to insert or update lines at these tables.
18:18.32Cisco_VPN_GodTzanger -- tell me about it.  Thats why I decided to come here.  To see if there was a viable solution just using asterisk.
18:18.38*** join/#asterisk mogorman (i=root@user-24-236-84-48.knology.net)
18:18.46tzangerCisco_VPN_God: yes there is, put an asterisk box on a public IP
18:18.50QwellMrBrown: pbx hosting?
18:18.52_Paulo_sevard, like "update user set password=password('mynewpassword') where user='root'"
18:19.19Cisco_VPN_GodIt looks more likely that SER or what you just mentioned is the only solution.  But I just don't like hanging stuff out there like that.
18:19.21sevard_Paulo_: i'm in the dark man.  I wanted SQL working so i could play  with it.
18:19.30_Paulo_sevard, I cant recall the right field names.
18:19.48*** join/#asterisk Umaro (n=umaro@68.142.142.105)
18:19.52_Paulo_sevard, you got the mysql prompt?
18:20.01sevardYes
18:20.06UmaroHey guys.. does anyone know of a digium supplier (or anyone who can sell me a digium card) in india?
18:20.12sevardand I've typed in everything you asked for except the line about passwords
18:20.43Dr-LinuxQwell: i wanna ask you something ..
18:20.54Qwell?
18:20.54Dr-Linuxhelp*
18:20.59_Paulo_sevard, well...
18:21.16Dr-Linuxi have 2 servers in different location, on is in US another one is in Pakistan.
18:21.37MrBrownQwell, it seems like a good deal, i mean a dedicated highspeed machine, with the dial tone already connected and paid for an entire year, phones, maintenance, dialtone, LD, a toll-free number, and all?
18:21.41Dr-LinuxUS server's SIP users talk to each other, make inbound/outbound calls good
18:21.55Dr-Linuxsame Pakistan users with pakistan asterisk server
18:22.03_Paulo_sevard, not to be rude, but: http://www.google.com.br/search?q=mysql+password+recover&sourceid=mozilla-search&start=0&start=0
18:22.24sevard_Paulo_: It's not rude to tell somebody to RTFM
18:22.36*** join/#asterisk chuy (n=Miranda@69.172.199.82)
18:22.45Dr-Linuxbut if US users want to make a call to pakistan, then there is very low voice from pakistan regular phone ..
18:22.56De_Mon_Paulo_ or to show hem how to google correctly
18:23.04chuylo.. trying to find a piece of software i sumbled upon a few weeks ago and neglected to bookmark
18:23.05De_Mon*them
18:23.07Dr-LinuxQwell: both servers are linked with each other via SIP
18:23.07Hmmhesaysat it again huh Dr-Linux?
18:23.16chuywin32 app that showed status of various phones in the office
18:23.24_Paulo_sevard, you are almos there...
18:23.29Nuggethttp://justfuckinggoogleit.com/?q=mysql+password+recovery is a better url to use.  :)
18:23.36chuyalmost like an IM client.. tiny
18:23.37_Paulo_s/almos/almost/
18:23.40mutomg this guy borrowed my analog phone to test some stuff and he's hacking and coughing all over it
18:23.47Qwellchuy: something from asteriskguru perhaps?
18:23.51De_Monlastlog app_queue
18:23.55*** join/#asterisk beefsalad (n=crash3m@unaffiliated/crash3m)
18:24.00chuyi don't know if it's asterisk specific
18:24.09_Paulo_sevard, just issue the "update user set password = password('yournewpassword') where user = 'root' and host='localhost'"
18:24.13chuylooked handy.. IIRC, it would act as dialer too
18:24.44Qwellchuy: http://www.asteriskguru.com/tools/switchboard.php ?
18:24.49Dr-LinuxQwell: did i ask wrong question? :S
18:24.55*** join/#asterisk spatulamaan (n=ggilmore@ip66-107-33-196.z33-107-66.customer.algx.net)
18:25.13chuyQwell: no.. it was much smaller app
18:25.25sevard_Paulo_: you want me to issue that or read these documents and format your command to my database
18:25.43*** join/#asterisk puzzled (n=yeahrigh@53533C13.cable.casema.nl)
18:26.07gaspizjustinu: from your experience in peak hours how mutch does the traffic increase? (traffic=simultanious calls) 120%, 150%, 200%, 250%?
18:26.17_Paulo_sevard, just issue that, if its a standard install.
18:26.23sevard_Paulo_: 10-4
18:26.31*** join/#asterisk mogorman (i=root@user-24-236-84-48.knology.net)
18:26.36_Paulo_sevard, then, type "flush privileges; "
18:26.44_Paulo_sevard, then "quit"
18:26.46chuyit was little desktop app so users could determine if coworkers are on phone, away, etc.
18:26.53*** join/#asterisk angom_w (n=angom@red-corp-200.38.16.10.telnor.net)
18:27.03*** join/#asterisk pa (n=Paolo@unaffiliated/pa)
18:27.27_Paulo_sevard, then restart mysqldaemon without --skip-bla-bla
18:27.43*** part/#asterisk beefsalad (n=crash3m@unaffiliated/crash3m)
18:27.48chuyaha.. jivesoftware.org
18:27.57mutargh
18:28.01mutslacking off at work is hard to do
18:28.03muti really don't see how people and sit and do nothing all day
18:28.08mogorman?
18:28.24[TK]D-Fenderchuy : Easy enough to do with just a script on Apache......
18:28.35sevard_Paulo_: I love you man.
18:28.39sevard_Paulo_: I frickin love you.
18:28.49Nuggetmut: irc helps
18:28.51chuythere's asterisk plugin for spark
18:28.56mutsomewhat
18:28.59mutbut still
18:29.05chuylooks pretty slick to me :)
18:29.05mutit just feels dirty and wrong
18:29.41[TK]D-Fendermut : I am *no* dirty! ...... well maybe :)
18:29.52mutheh
18:29.53_Paulo_sevard, It worked, I presume...
18:30.08mutit's just more difficult than i'de have figured
18:30.10sevard_Paulo_: when I type mysql --password and give it the new password it works.
18:30.10*** join/#asterisk sigmounte (n=sigmount@LSt-Amand-152-33-38-41.w82-127.abo.wanadoo.fr)
18:30.19sevardThus, I love you and will have your children
18:30.27*** join/#asterisk KalD (n=Kal@c-24-16-117-210.hsd1.wa.comcast.net)
18:30.29chuyare there any IP phones to _avoid_ ?
18:30.29sevardI will have to undergo many surgical processes first.
18:31.05chuyI've been looking at Grandstream and Astra
18:31.07tzafrir_laptopchuy, some say that the BT10x stand for Bug-Tone
18:31.09*** join/#asterisk Reko (n=nonsense@cpc4-hem15-0-0-cust755.lutn.cable.ntl.com)
18:31.16tzafrir_laptopOthers like them
18:31.41chuyI could just go with cheap Sipura over BudgetTone
18:31.45hypnoxChuy grandstream works in the most cheap shoddy sense
18:31.45tzafrir_laptopchuy, is it for you or for a client?
18:31.55chuyer.. I mean LInksys :)
18:32.08websaewhen I call people, people say it sounds like I quite far away, anyone know why that would be?
18:32.12rob0I've seen *many* here say to avoid Grandstream, no experience tho.
18:32.13SibRw0rkin sip.conf what's the difference between peer and friend?
18:32.15tzafrir_laptopchuy, do test them. Try working with them extensively. Many IP phones are simply badly-designed
18:32.17*** join/#asterisk iCEBrkr (n=icebrkr@6532244hfc169.tampabay.res.rr.com)
18:32.18jaigerChuy, depends on your needs but I don't consider a grandstream a phone for anything other than home/play use
18:32.21chuytzafrir: both actually
18:32.43cpmSheesh, anyone here clueful on a CAC Bank Bank 1 and a wtc11xp ?
18:32.48chuydo any brands have a consistantly good track record?
18:32.58cpmI've put days into this, can't get the alarms to clear
18:33.03tzafrir_laptopchuy, also consider the option to use an ATA and existing phones. Expecially for wireless phones
18:33.20chuytzafrir: yes.. i've considered ATA for wireless
18:33.26*** join/#asterisk justinu (n=justin@72.18.13.34)
18:33.33chuyAstra's wireless offering is a bit on the pricey side
18:33.51sevard_Paulo_: want an asterisk + sql based question?
18:33.59jaigerI thought the aastra wireless is a decent price
18:34.18chuywell..  i was looking at the 480i
18:34.20jaigerChuy, we like our polycom phones
18:34.21tzafrir_laptopsevard, who cares if he wants it. Ask! ;-)
18:34.24chuyit's like $400
18:34.35_Paulo_sevard, Its not my game, but go on.
18:34.42_Paulo_sevard, I will try my best.
18:35.06chuyjaiger: 301's?
18:35.11sevardActually, give me 5 minutes, it's a library problem, i'm going to hack it out for a second.
18:35.40jaigerchuy, no.  we use 500 and 600 phones.  we sold 301s to a customer and they didn't like the limited features.  they really wanted 500s
18:36.49*** part/#asterisk chrish01 (n=chrish01@wsip-70-183-17-68.oc.oc.cox.net)
18:36.51chuyjaiger: use PoE in any deployments?
18:37.09chuyjaiger: and if so, what switch was chosen?
18:37.09jaigerchuy, not yet but I have cabling and a switch here to test out
18:37.18chuycool
18:37.23chuywhich switch are you testing with?
18:37.34jaigeran 8 port netgear
18:37.43chuyRed Hawk's setup seems decent
18:37.46chuyi like the modularity
18:38.52chuytho they call it a hub.. if it is indeed a dumb hub, not worth the price
18:39.47MrBrownSo, Qwell, what did you think about that? Anyone elses opinion seriously: http://cgi.ebay.com/Hosted-Asterisk-PBX-Polycom-IP-501-601-with-Dial-Tone_W0QQitemZ5862473555QQcategoryZ11908QQrdZ1QQcmdZViewItem
18:40.12sevard_Paulo_: got it
18:40.15sevardlunch time!
18:40.18QwellMrBrown: $5500 for a hosted server is far too much...  I bet they charge like 5c/min above the 14k
18:40.25sevard_Paulo_: love you snookums thank you for your help
18:40.56_Paulo_sevard, you are welcome, good luck.
18:41.18[TK]D-FenderMrBrown : Not worth it.
18:41.44MrBrownwell it's not $5500 for just the server... the server is only $1499
18:41.45*** join/#asterisk Dr-Linux (n=Nothing@202.125.141.6)
18:41.46Qwellplus it's like $3k/year
18:43.04[av]bani[TK]D-Fender: i have a new linksys phone for you
18:43.15[TK]D-Fender[av]bani : No thanks... you can have mine :)
18:43.29mistralmrbrown: is that a hosted server ?
18:43.34mistralas in in a datacenter ?
18:43.39MrBrownbut I already pay $2800 / year for my 5 co lines, and I pay $1500 / year for about 30,000 minutes of LD... so that's $4300 per year i'm paying already...
18:43.39[av]bani[TK]D-Fender: http://bani.anime.net/phonez/   PHM1200
18:43.50*** join/#asterisk saftsack_ (n=saftsack@p54A7FFDC.dip.t-dialin.net)
18:43.57saftsack_many, hi
18:44.15MrBrownmistral yes!, not a vmware machine either...
18:44.15QwellMrBrown: 14000 minutes with any decent provider is...$280
18:44.30[av]bani[TK]D-Fender: color backlit lcd, 2 10/100, fd spkphone, poe, 24 blf, xml... $270
18:44.32Qwelloh, 24000...whatever
18:44.40MrBrowni use 30,000
18:44.42Qwell480 then
18:44.43mistralmrbrown: what are co lines ? and why are you paying minutes ?
18:44.45MrBrownper year
18:44.48MrBrownabout
18:44.50QwellMrBrown: okay, $600
18:44.59[TK]D-Fender[av]bani : Only supposed to work with LinksysOne.  Wouldn't touch it if thats the case
18:45.03Qwell$600 * 5 = $3000
18:45.04[av]bani[TK]D-Fender: it says 'sip'
18:45.06MrBrowni pay 1500, it's 5 cents per minute for me
18:45.25MrBrownmistral, a CO line is a phone line from the phone company
18:45.25[av]bani[TK]D-Fender: and other press releases dont mention linksysone requirement at all, but do say 'sip phone'
18:45.33MrBrowni pay $48 per month per line
18:45.40[TK]D-Fender.05 / min?  RIDICULOUS.  I pay .03 CDN for LD here and I COULD get less....
18:45.40chuythat chick's teeth need photoshop :)
18:45.40MrBrownplus 5 cents / min for long distance
18:45.43QwellMrBrown: how many simultaneous calls?
18:45.45mistralmrbrown: just called a phone line here :D. How big is it ?
18:45.48Qwelllike 5-10?
18:46.08MrBrown5-10 intercom calls probably... no more than 5 pstn calls
18:46.13MrBrownonly got 8 extensions
18:46.19mistralouch
18:46.33MrBrownhow big is it?
18:46.37MrBrownthe phone line mistral?
18:46.39*** join/#asterisk sigmounte (n=sigmount@LSt-Amand-152-33-38-41.w82-127.abo.wanadoo.fr)
18:46.44mistralyup
18:46.50[TK]D-Fender[av]bani : What do you think that SRV9000 requirement is?
18:47.10MrBrownit's 1 pair of copper wires delivering 1 analog phone line
18:47.42chuywhy is guy who's selling AA aimer bots on eBay moving to selling VoIP service? :P
18:47.55MrBrown?
18:47.58mistralmrbrown: ahh when you get bigger than 4/5 lines here you put in an isdn 30 or so. For what your paying isdn 30 with 10 live lines would be way cheaper
18:48.03chuyAmerica's Army SMiRLbot Multi Hack Aimbot for AA 2.5
18:48.16chuythe guy running that VoIP action :P
18:48.31chuykind of hard to take seriously
18:48.31MrBrownmaybe he owns 2 businesses?
18:48.36mistral5 pstn lines in the uk would cost oyu £900 and the server hosting would be £30/month with 100mbit net connection with a quota of 1TB / data a month
18:48.55QwellMrBrown: You could get a dedicated server (with 30000 minutes) for $2400/year, and all you have to pay for is the phones
18:49.08[TK]D-Fender[av]bani : "The PHM1200 must be connected to a Linksys One-enabled switch port."
18:49.28Qwellyou could possibly get far cheaper if you were to buy the server and have it racked somewhere
18:49.43mistralqwell thats what i thought he was doing
18:49.48[TK]D-FenderMrBrown : Yup... "roll your own"....
18:49.48hypnoxmistral who co-los a server in the UK for £30/mo?
18:50.00mistralhypnox: most of them now if you look around
18:50.08Qwellmistral: he's paying somebody else to do it though...at a far higher cost
18:50.09gaspizfrom your experience in peak hours how mutch does the traffic increase? (traffic=simultanious calls) 120%, 150%, 200%, 250%?
18:50.12tdonahuemistral: you are only paying ~$52 per month for 1 TB?
18:50.41MrBrownQwell, does that include configuration and maintenance?
18:50.44mistralyeah bandwidth prive in the uk has dropepd alot
18:50.58mistral5 years ago it used to be 1p/mb
18:51.07MrBrownWhat about the phones, how much should 8 polycom ip 501/601's cost?
18:51.15QwellMrBrown: Theirs only gives 1 year
18:51.21Qwellabout $200 each?
18:51.35MrBrownyea, it says $212 on that auction, and that includes shipping...
18:51.41tdonahueknowing what a 100 Mb link goes for in the US that is unbelievable
18:51.49hypnoxcheapest i've seen is around the £50 mark, and i've looked plenty
18:51.53MrBrownshipping and configuration*
18:51.56Qwelland I would be willing to bet that they charge an arm and a leg for each minute above the 24000
18:52.04MrBrown4.4cents
18:52.04justinuanyone know how to calibration a fluke i410 AC/DC current clamp?
18:52.07MrBrownno wait
18:52.09MrBrown3.8 cents
18:52.11Qwellyeah, that's expensive as hell
18:52.12justinus/calibration/calibrate
18:52.17[TK]D-FenderMrBrown : IP 501 = $170, IP 601 = $250
18:52.21*** join/#asterisk zotz (n=zotz@24.231.47.175)
18:52.23MrBrown4.4 cents on the toll-free number
18:52.25mistralhypnox: depends you need ot dump certint hings to get the price down more. eg restricted access to server etc.. etc.. if you require staff todo things to it its a rip off
18:52.27Qwelljesus
18:52.41Dr-Linuxwhat will be the differnece if i change rxgain/txgain values from 1.0 to 3.0 ?
18:54.04MrBrownhere's the thing about the service... the dialtone provider: http://www.voipnettechnologies.com provides the trunk replacement service. their entire network is on Level3 and the entire network is QoS controlled. There are no audio problems ever
18:54.14*** join/#asterisk angom_w (n=angom@red-corp-200.38.16.10.telnor.net)
18:54.26MrBrownthe server in the datacenter has unmetered bandwidth
18:54.51QwellMrBrown: They can't guarantee that
18:54.55*** join/#asterisk elg (n=fugalh@falcon.fugal.net)
18:54.56Qwellit goes over the public internet
18:55.02MrBrownno it doesn't actually
18:55.13QwellSo, you have to move your office to their datacenter?
18:55.38mistralmrbrown: the address of that site sucks for uk routing :D
18:55.49MrBrownoh no... that part still goes over the public internet
18:56.16*** join/#asterisk crich1999 (n=crich@port-212-202-201-26.dynamic.qsc.de)
18:56.19Umarohi guys.. anyone have any tips about echo that seems to be caused by meetme?
18:56.20QwellMrBrown: and their L3 traffic likely does too
18:56.23MrBrownbut the dialtone connections between all of the CO's are all QoS on level 3
18:56.58MrBrownfrom the PSTN to the server it's all super highspeed very low latency QoS prioritized voice
18:57.03mistraltheres me thinking level 3 was part of the public internet
18:57.32*** join/#asterisk Reko_ (n=nonsense@cpc4-hem15-0-0-cust755.lutn.cable.ntl.com)
18:58.21MrBrownlevel3 provides the top level internet connectivity to a very large part of u.s. internet
18:58.35mistraland most other countries
18:58.57muti bet they pay pretty good
18:59.03mistralLevel 3 tends to interconnect all the major isp in the uk us fr gr and various other countries
18:59.16*** join/#asterisk ipso (n=ipso@d207-81-249-35.bchsia.telus.net)
18:59.32mistralalso the l3 providers also have datacenters for servers
19:00.01MrBrownYes, this server is hosted in a cage in a L3 datacenter
19:00.22mistralalso level 3 is having polictal arguments latly and alot of companies were goign to leave it
19:00.45MrBrownsurvival of the fittest :-)
19:00.51[av]bani[TK]D-Fender: wnder what that means, because http://biz.yahoo.com/prnews/051114/lam072.html?.v=31 mentions the requirement naught
19:00.58[av]bani[TK]D-Fender: it does say 'sip based' though
19:01.46mistralmrbrown: yeah the fittest were going to leave because they could afford it
19:01.47mistral:D
19:01.55mistralit would have played a massive problem in l3
19:01.57*** join/#asterisk Alric (n=nbowyer@masq.hyperusa.com)
19:02.01*** join/#asterisk SpaceBass (n=sp@static-71-251-230-2.rcmdva.fios.verizon.net)
19:02.05mistralas in break it up
19:02.10[TK]D-Fender[av]bani : think of the movie terms "based on actual events".  Could imply its a story about a person on EARTH, and thats about it....
19:02.39hypnoxgblx seems like the best international carrier
19:02.49[TK]D-FenderAnd then the "writing between the lines" begins....
19:02.50mutanyone know any adsl modems with built in sip/fxs ports?
19:03.23muti think there is a bug in the firmware of these zoom modems
19:03.35mutcause they freeze when using nat & sip through them
19:03.49muthad to turn off nat and route ips to the internal network
19:03.53mutand it stopped the problem
19:04.04[TK]D-Fender[av]bani : So until someone does some "hard" testing of it I'd avoid it like the plague.  And even then, it costs more than an IP 601.  If they sheap out guys like us on it, forget it.
19:04.16[TK]D-Fender[av]bani : It'd be just another stupid "key" system
19:04.23hypnoxmut SpeedTouch 716 WL
19:04.27Normi'm just now playing with asterisk... i've got a soft phone hooked up to the test extension (200). i've just configured a cisco hardphone to use a new extension (120), but it can't seem to call the echo-test application
19:04.43Normi get a dial-tone though... any ideas?
19:05.07Flautohey guys
19:05.15Flautoi just upgraded to 1.2.4
19:05.19MrBrownNorm, the dialtone you hear is coming from the phone
19:05.19Flautowhat is new
19:05.33*** part/#asterisk MrBrown (n=owned@68-169-204-147.agstme.adelphia.net)
19:05.34mutare they worthwhile dsl modems?
19:05.38muti froogle SpeedTouch 716
19:05.40mutand nothing comes up
19:05.42muto_O
19:05.52hypnoxmut i saw it here http://www.plus.net/products/hardware/voice/voice_router_kit.shtml
19:05.57[TK]D-Fendermut : S518 :)
19:06.04NormMrBrown: when i configured the phone's extension via AMP, it gave me a dialtone and the date/time is displayed on the LCD... before, it just gave me errors... so i suspect i've made some progress, somehow :)
19:06.47[TK]D-Fender~amp
19:06.48jbotit has been said that amp is NOT supported here! people using it should join #amportal
19:07.13mutumm
19:07.15muttk?
19:07.32*** join/#asterisk manolo (n=manolo@200.124.172.72)
19:07.37muti'm looking for external stuff
19:07.42[TK]D-Fendermut : Sangoma S518 ADSL PCI card :)  Its what I run in my * server :)
19:07.48mutfor cpe end
19:07.49[TK]D-FenderExternal?  BAH!
19:08.08[TK]D-FenderI have a GNET BB00060 for sale if you like :)
19:08.09mutcause zoom modem + spa-2002
19:08.12mutdoesn't seem to do it
19:08.25hypnoxnot from behind nat no
19:08.32[TK]D-Fendermut : Sure you did allthe NAT settings right for it?
19:08.33manoloHey i wannna know if it's possible to assign a different context to incoming calls depending on which channel is it coming from
19:08.38mutyea
19:08.41[TK]D-Fenderyup
19:08.44mutit's dmz'd
19:08.48mutit locks up the adsl modem tho
19:08.53[TK]D-Fendermut : Shouldn't need to DMZ it!
19:08.53[av]bani[TK]D-Fender: of course, could be something like sccp... just waiting to be reverse engineered for asterisk
19:08.55mutand you have to hard reset it
19:09.11mut[TK]D-Fender: i know but that what the exhaustion of all options
19:09.17[TK]D-Fender[av]bani : I'm just waiting for one to be FORWARD engineered properly :D
19:09.21[av]bani[TK]D-Fender: probably just means it autoprovisions via the linksysone switch
19:09.25mutwe turned nat off completely after that and routed ips to it and it works fine
19:09.35[TK]D-Fendermut : So you told * all about the NAT, AND the SPA as well?
19:09.38mutdid it with msn voice chat too
19:09.49mutno ata involved
19:10.08[TK]D-Fendermut you should not have to forward anything for it to work....
19:10.20mutthe modem just has problems
19:10.23manoloFor instance i have 3 FXO ports and each one of them has a ZAp exten assigned to it, so i want the Zap 1 to go to Digital Recepcionist, in context 1, Zap 2 to go to Digital Recepcionist in context 2 and so on... How to do that?
19:10.28[TK]D-Fenderand the modem should have no impact over routing unless its an "all-in-one"
19:10.33mutmaybe it's overloaddd with the huge udp rtp stream i dunno
19:10.37[TK]D-Fendermanolo : Yes, all very easy....
19:10.42mutum
19:10.43mutwell
19:10.51mutmsn voice chat (sip) locks up the modem
19:11.01mutand spa-2002's (sip) lock up the modem
19:11.04[TK]D-Fendermut : you told the ATA that its behind nat, as well as * and set the keep-alive stuff and everything?
19:11.11mutyes
19:11.21[TK]D-Fendermut : UGH.  NEver heard of this before...
19:11.22muti have other customers behind linksys routers and they work fine
19:11.36manolo[TK]D-Fender: COuld you give me some hints.?
19:12.06Hmmhesays[TK]D-Fender how goes it today?
19:13.11[TK]D-Fendermanolo : Its all in the WIKI..... just read up on zapata.conf.  thats where you define the contexts used by your zap channels.
19:13.20SibRw0rkanyone get asterisk servers to talk to one another
19:13.24SibRw0rkto pass numbers thru?
19:13.32justinunever
19:13.52[TK]D-FenderHmmhesays : *BLEH*  I'm &^%@ing drained and there's an emotional mess waiting for me on Sunday.  I haven't worked out in months and feel like shit on a stck :/
19:14.01justinuemotional mess?
19:14.12Hmmhesays[TK]D-Fender grand
19:14.15Hmmhesaysemotional mess?
19:14.29HmmhesaysYou want to talk emotional mess, I'm dating an engaged girl
19:14.35*** join/#asterisk z-killz (i=koma@c211008.adsl.hansenet.de)
19:14.36z-killzmoin
19:14.50[TK]D-Fenderjustinu : GF is coming back from a vacation after we fought 2 days prior (after which I'd been avoiding contact even though we live together), and on her return she'll be finding that we're "over".
19:15.11rob0uh oh
19:15.18justinulol
19:15.26justinudude, that sucks
19:15.28Hmmhesays[TK]D-Fender my girlfriends fiance is out of town for week, and it a week my employer decided i'm flying down to north carolina
19:15.34[TK]D-FenderHmmhesays : You are choosing to continue something that seems easy to stop.  not so with mine as I'm leaving her with her place which she will go in debt for.
19:15.46rob0[TK]D-Fender: I know you don't drink (much) but perhaps it's time to start :)
19:15.50Hmmhesaystomato tomAto
19:15.57justinui had a live in girlfriend, i had to throw her out of my house
19:16.01justinuthat was unpleasant
19:16.05[TK]D-Fenderrob0 : Sorry... NO.
19:16.13HmmhesaysI drink a fair amount
19:16.17Hmmhesaysmakes life tolerable
19:16.17rob0(it was a joke)
19:16.34*** join/#asterisk ast_freak (n=ast_frea@68-112-130-237.dhcp.stcl.mn.charter.com)
19:17.06*** join/#asterisk OnwealdAndy (n=arodes@MT1-28.aircanopy.net)
19:17.34justinuto alcohol! the cause of, and solution to all of life's problems!
19:17.45iDunnoit's not the cause.
19:19.39rob0[TK]D-Fender: how long were you together?
19:19.55Hmmhesaysi find being in a band helps you break up with females
19:20.03Hmmhesayswhy, because when you play guitar all the chicks want you
19:20.52*** part/#asterisk OnwealdAndy (n=arodes@MT1-28.aircanopy.net)
19:21.01[TK]D-Fenderrob0 : days under 2 years
19:21.28_Paulo_Alcohol ! Helping ugly people to get laid since mankind roots.
19:21.35[TK]D-FenderHmmhesays : Yeah... I gotta get back into stage work....
19:21.39rob0<== had a divorce after just shy of 10 years
19:21.44justinuouch
19:21.56rob0It was a good thing.
19:22.06justinui'm marrying my fiance on our 6th anniversary (of meeting)
19:22.17rob0Long engagements are good too.
19:22.32justinuyeah, i hope so
19:23.09rob0I'm remarried, did much better the 2nd time. :) 11 years now.
19:23.44[TK]D-FenderBon Jovi said it best "We've been living together 3 years, 2 hours, 20 minutes maybe four houndred beers"
19:23.53rob0Great sex has always helped us work out our problems :)
19:23.54*** join/#asterisk Maxxed- (n=whyman@66.195.105.87)
19:24.02Maxxed-man this chan is big now!
19:24.10Maxxed-go * !
19:24.11Maxxed-heh
19:24.14DaminAnd, (I think this is possible) I would like to use cdr_odbc to log to multiple back-end datasources.. one on MySQL and the other on MS-SQL.. is that doable?
19:24.33rob0ugh, ms-sql :(\
19:24.34DaminI.E. can I get cdr_odbc to log the record to both datasources at once?
19:24.58Maxxed-i have an idea.. not sure how to implement it..
19:25.00*** join/#asterisk sulex (n=sulex@pdpc/supporter/active/sulex)
19:25.14Hmmhesays[TK]D-Fender yeah man it everything just seems OK when you are up there
19:25.17Maxxed-basicly, when i have to call sombody, i leave a vmail for them
19:25.39Maxxed-well i end up calling there office, cell, home, and have to repeat my message
19:25.50Maxxed-can anybody think of a clever way to record the message once
19:25.58Maxxed-then dial the next number and have it play back
19:26.02Maxxed-as im on the line
19:26.08Maxxed-kind of like a confrence call
19:26.27Hmmhesayscascading dial that ultimately ends up as an asterisk voicemail?
19:27.10nroejhi all
19:27.38Hmmhesaystiming would be hard on the Maxxed-
19:27.44Hmmhesaysbeing each vmail system is diff
19:32.36Dr-LinuxHmmhesays: WHAT was the song ? that you were singing yesterday?
19:33.18Maxxed-Hmmhesays: yeah
19:33.30Maxxed-Hmmhesays: well, the idea is-its not automatic
19:33.38Maxxed-i still dial the number as i normaly would
19:33.47HmmhesaysDr-Linux: sugar by fallout boy
19:33.51Maxxed-then bash an extention
19:33.57Maxxed-to get the temp wav to playback
19:34.02Maxxed-while im on the line
19:34.41Hmmhesaysthe problem is getting the message to play when you want it to
19:34.53Normokay so i have my cisco hardphone (SP12) configured via skinny, and i see it's output in my asterisk console.... i dial a number and the console registers "collected digit...", but it never connects to the echo-test application. any ideas?
19:35.10Maxxed-leave a message at the tone {i transfer the call to a confrence ext where the wav plays}
19:35.11HmmhesaysIE: after the beep
19:35.31Maxxed-i dont mind a sec or two delay, thats not biggie
19:35.35HmmhesaysMaxxed- yes you could do that
19:35.54Maxxed-its how to trigger the wav to play..i cant think of to many ways to do it
19:36.08Maxxed-yeah, best idea i have so far
19:36.09Hmmhesaysum, playback
19:37.01Hmmhesaysexten => 111,n,Playback(boobies)
19:37.48Dr-LinuxHmmhesays: why you always use bad words?
19:38.02Hmmhesayslets take a vote, did I use bad words today?
19:38.09Qwellonly good words
19:38.20Dr-LinuxHmmhesays: yes you did
19:38.28Maxxed-heh
19:38.30AlricOnly in America, I suppose.
19:38.31Maxxed-boobies isnt bad
19:38.34Dr-Linux<Hmmhesays> exten => 111,n,Playback(boobies) <<<
19:38.36shido6there is no existence of 'boobies' context
19:38.38Hmmhesaysin fact boobies are good
19:38.44Maxxed-yes sir
19:38.47Dr-Linuxboob >> boobs >> boobies
19:38.53Hmmhesayspreferably when they come in a set
19:39.03Hmmhesaysand aren't on a guy
19:39.24shido6Your application(s) bobbies is (are) not registered
19:39.26sawhrm. can one configure different languages depending on where the calling number comes from?
19:39.52Dr-Linux:S ...
19:39.56Martincit0pls, give me tips to solve this warning:
19:39.57Martincit0WARNING[10888] chan_zap.c: Unable to disable echo cancellation on channel
19:40.05AlricIsn't there like a set(language()=) thing?
19:40.07Hmmhesaysdon't call out chan_zap?
19:40.55Martincit0pri is fine in and out (until that message)
19:41.34Hmmhesayssetting up cascading dialing sucks
19:42.09Hmmhesayssetting up voip in general to integrate with an existing pbx and be totally transparent to the end user (who usually has an IQ of aobut 85) is rough
19:42.41Hmmhesaysso true
19:42.57Martincit0:(
19:43.06AlricLike this one lady, who couldn't understand why she couldn't transfer a call... when she had no active call.
19:43.10Maxxed-iv not done anything with the confrence stuff, so il have to hit the books some
19:43.13*** part/#asterisk austinnichols101 (n=austinni@70.46.69.130)
19:43.20HmmhesaysMaxxed- why o you need to conference?
19:43.40*** join/#asterisk dioz (n=d@tor/session/x-64f0044e1dafcd61)
19:43.43Maxxed-well, wouldnt i inorder to get the playback to work
19:43.58Maxxed-i couldnt just dial 22 and have it play while in a call could i?
19:44.07Hmmhesaysyou mean you want to transfer the call  to the playback extension just once? for multiple voicemail systems?
19:44.16Maxxed-just once
19:44.22Maxxed-but i want to bable to hear it
19:44.26Maxxed-well.. i would like to beable to hear it
19:44.30Hmmhesaysgood luck with that
19:44.32Maxxed-to make sure its the tright shit
19:44.32Maxxed-yeah
19:44.45Hmmhesayswhat happens when you hit a cell phone that vm answers right away
19:44.49Hmmhesayswhile the others are ringing?
19:45.00*** part/#asterisk dioz (n=d@tor/session/x-64f0044e1dafcd61)
19:45.00Maxxed-im doing one call at a time
19:45.09_Paulo_Maxxed-, why dont you just record a message and then place callfiles with a macro?
19:45.11Hmmhesaysahh gotach
19:45.16Maxxed-i call home #, no after, leave a vmail and record it
19:45.27HmmhesaysMaxxed- that is pretty simple
19:45.30*** join/#asterisk Garak_ (i=1000@209.5.171.170)
19:45.37Maxxed-i call cell #, no, play back the recorded home vmail
19:45.51Maxxed-yeah, but the ability to listen to it at the same time, not so sure
19:45.52Maxxed-heh
19:45.55[TK]D-FenderHmmhesays : Doesn't matter.  all PSTN channels are considered "answered" as soon as the link accepts digits IIRC....
19:45.56Hmmhesayschanspy
19:46.02Maxxed-i guess il have to script somthing up
19:46.20Hmmhesays[TK]D-Fender if thats the case then I will never use a digium card
19:46.38Hmmhesaysfas no matter what? I doubt that
19:47.07Maxxed-im just trying to get an idea of what way to roll with this
19:47.14Hmmhesayscheck out chanspy
19:47.25Maxxed-as far as the syntax and such.. il figure out thru the docs
19:47.27Maxxed-chanspy?
19:47.36Hmmhesaysallows you to listen to a channel
19:47.42Maxxed-ah!
19:47.53Maxxed-sounds like that could be used here
19:47.55HmmhesaysIE: the channel between your vmail and the playback
19:48.08Hmmhesaysthat'll be 20 bucks or a case of budweiser please
19:48.17Maxxed-hah ;)
19:48.28Maxxed-i'll pay pal a case of bud over if you just to it all for me ;)
19:48.50HmmhesaysOh I could set it up for you, but it'd cost you more than a 20
19:49.05Hmmhesayswhen you want it done by?
19:49.14Maxxed-pssh, il do it my self, learn somthing, then get a case and forget what i learned
19:49.17Maxxed-heh
19:49.26Maxxed-no rush, im just fugin around with the idea right now
19:49.39Maxxed-seeing if i can do it (and it looks like i can) and how to do it
19:50.08*** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn)
19:50.54Maxxed-thx for the pointers so far Hmmhesays :)
19:51.00Hmmhesaysbasically, 1. record the first vmail, 2. second call, transfer to playback after beep and chanspy in
19:51.02Hmmhesaysor....
19:51.09Hmmhesaysyou can do it with a manager redirect
19:51.13Hmmhesayswhich is also pretty simple
19:51.24Maxxed-manager redirect?
19:51.50Hmmhesaysthat would require you do some scripting though
19:52.01Maxxed-im not afraid of that any
19:52.07Maxxed-im a scriptin' mofo
19:53.10Hmmhesayshttp://www.voip-info.org/wiki/index.php?page=Asterisk+Manager+API+Action+Redirect
19:53.24Hmmhesaysnot sure how that would work redirecting to a playback extension though
19:53.31Hmmhesaysit might freak, but you can try it
19:53.46nroejhmm i is it possible to start vocimail recording while the announcement is running?
19:53.50*** part/#asterisk Martincit0 (n=martin@litigaractivos1.att.net.co)
19:53.59Maxxed-im toy around with it
19:54.05Maxxed-s/im/il
19:54.16*** join/#asterisk tainted- (n=somewher@mail.k2usa.com)
19:56.03Hmmhesaysand just for Dr-Linux .... boobies
19:56.19Maxxed-( . ) ( . )
19:56.24Maxxed-wooo yeah
19:56.28Maxxed-ascii art gone wild
19:56.38tainted-hey u guys are pretty funny
19:56.42Hmmhesayslol
19:56.44Flautois there a way to set caller id if the service provider does not support it
19:56.45Maxxed-show us your pixles!!
19:56.46Maxxed-heh
19:56.46Hmmhesaysits friday dude
19:56.51QwellFlauto: no
19:56.57HmmhesaysFlauto: maybe
19:57.23tainted-Flauto yes, as long as you don't call outside through your service provider
19:57.41Flautotainted, i know that
19:57.51justinuis flauto anything like a Flouta?
19:57.59HmmhesaysFlauto: you can set it statically in your dialplan
19:58.15tainted-Flauto u can also set the caller id through that provider by using another provider
19:58.26Hmmhesaysassuming you are talking about a call coming in from the tdm world and going out IP
19:58.47*** join/#asterisk stkn__ (n=stkn@gentoo/developer/pdpc.active.stkn)
19:59.06filehot'n'sexy Hmmhesays!
19:59.09Hmmhesaysfile
19:59.17Hmmhesayswhat up brotha
19:59.23Flautohmmhesays, i did set callerin(num)=xxxxxxxxxx
19:59.33Hmmhesaysyeah what sure wouldn't work
20:00.00fileHmmhesays: porting drama...
20:00.17_Paulo_bye everybody. In Brazil it's time to grab a beer.
20:00.22HmmhesaysSet(CALLERID(num)=XWhatevaX) would
20:00.26Hmmhesaysfile fun fun
20:00.32*** part/#asterisk _Paulo_ (n=paulos@200-168-112-132.dsl.telesp.net.br)
20:00.36Flautoi did that
20:00.44Hmmhesayswithout typo's?
20:00.53Flautoyes, sir
20:01.06Hmmhesayspastebin.ca your dialplan
20:01.16*** join/#asterisk lithi (n=interp3@Toronto-HSE-ppp3858329.sympatico.ca)
20:01.32Hmmhesaysfile: I'm going to drink heavily and dance with hotties tonight, its going to be grand
20:01.40fileHmmhesays: excellent
20:01.50Hmmhesaysyes indeedy
20:01.51mzowill your asterisk server stay up while you go dance with hotties?
20:01.58Hmmhesayslikely
20:02.11Hmmhesayswell most of them should
20:02.51Hmmhesaysthen tomorrow I shall purchase aerosmith tickets
20:03.13mzoconfess, you're just going because you have a crush on Liv Tyler :P
20:03.28Hmmhesayswell I wouldn't kick her out of my bed
20:03.36Flautohttp://pastebin.com/537447
20:03.40Maxxed-heh
20:03.45mzoshe has to have her elf ears on, otherwise she's getting booted :P
20:03.55Hmmhesaysgood point mzo
20:04.21*** join/#asterisk harryvv (n=none@S010600a0c93f6f7e.vs.shawcable.net)
20:04.23Hmmhesaysum Flauto, you ain't going to get a call going anywhere with that
20:04.42Flautothat is how i set the callerid
20:04.46harryvvis the memory leak the result of asterisk not shutting down when it should?
20:04.59Hmmhesaysshow the rest of your dialplan you're using to call out
20:05.10*** join/#asterisk Martincit0 (n=martin@litigaractivos1.att.net.co)
20:05.12Hmmhesaysand preferable some cli output with verbose at least 3
20:05.33lithiAny ideas why when I mute my microphone via the recording control pannel in windows idefisk mutes both the mic and sound from the call? This only happens with this softphone.
20:06.44HmmhesaysI'm curious if I should go with a digial eq or an analog one
20:07.14*** join/#asterisk elg (n=fugalh@hfugal.NMSU.Edu)
20:07.25Flautohttp://pastebin.com/537454
20:07.54Hmmhesaysdoes voipstunt support setting your CID?
20:08.06Flautoi don't know
20:08.08Flautoi tried
20:08.08*** join/#asterisk Dr-Linux (n=Nothing@202.125.141.6)
20:08.12Flautobut not working
20:08.13Hmmhesaysobviously it doesn't
20:08.13donnibi have been told that this forum is not for AMP but i think i have a problem wich is related to Asterisk. When i call into my PSTN number(SIP) it takes 11 seconds before the call is transfered to my extension. here is what the CLI tels me. http://pastebin.com/pastebin.php?dl=537450 Can somebody take a look please.
20:08.15Flautoso it might not
20:08.17*** part/#asterisk Tuttle_ (n=Tuttle@kelinat210.keli.cz)
20:08.41Hmmhesaysno it sure doesn't look like it
20:08.43Flautothere is notthing i can do then
20:08.45HmmhesaysDr-Linux
20:08.46De_Monlithi assterix gets unhappy when the phone doesn't transmit silence, fix the softphone so it transmits silence
20:08.48*** join/#asterisk tdonahue (n=tdonahue@208.51.101.201)
20:08.53De_Monasterisk
20:09.06donnibit get's pretty fast to -- Launched AGI Script /var/lib/asterisk/agi-bin/dialparties.agi
20:09.25donnibbut then it takes 11 sec to get further.
20:09.25Dr-LinuxHmmhesays: yes?
20:09.25Hmmhesaysboobies
20:09.30Hmmhesaysdonnib: using aah?
20:09.47mzoI thought that was normal?
20:09.52donnibHmmhesays: yes i do but i am not sure it's related to that or ???
20:10.08Dr-LinuxHmmhesays: you like boobies?
20:10.15Hmmhesaysyou mean it just sits at dialparties.agi? or does it continue
20:10.43donnibit continues after 11 sec. it gets there pretty fast when i call but then it hangs a wile and continues
20:10.45Hmmhesaysbecause there is a bunch of waits in aah dialplan that you can remove if you arne't faxing
20:10.54mzois that what that is?
20:11.01mzoargh *murder* :P
20:11.21Flautoit is not good not being able to set callerid
20:11.26Hmmhesaystheres at least 6 seconds, that I can recall
20:11.27donnibohhh i get it so i need to get rid of those ? any idea were they are i mean in which file
20:11.31HmmhesaysFlauto, get a new provider
20:11.37Hmmhesaysdonnib, look at your dial macro's
20:11.40mzoyou need them to receive faxes, or send?
20:11.52donnibno i don't
20:11.52Hmmhesayssome of the waits are for fax dectection
20:12.00lithiDe_Mon: Send CN frames when slience detected?
20:12.09*** join/#asterisk Mw3 (n=mw3@national.t-error.hu)
20:12.14donnibi don't need fax at all
20:12.19Hmmhesayssome are there for others reasons which I don't really care
20:12.26lithiDe_Mon: Thats one of the options, but still is muting the call sound when I mute my mic.
20:12.30De_Monlithi no idea my phone says 'transmit silence'
20:12.48donnibso any idea where i should start ?
20:12.52Hmmhesaysmzo, look at the cli while its dialing, there are a lot of 2 second waits
20:13.10De_Monlithi voip-info.org will probably have the specifics for your phone
20:13.31Flautohmmhesays, this one is free
20:13.43HmmhesaysFlauto, then don't complain
20:13.43Flautoi guess thati have to find a back up
20:13.46mzoi'll have to check, i never noticed that before
20:13.49Flautohaha
20:13.52Flautothanks man
20:13.53mzoi always thought the pauses were normal.
20:13.59Hmmhesayshaha
20:14.00Hmmhesaysyeahright
20:14.04mzoreally!
20:14.08HmmhesaysI got pissed off at mine and hacked it up
20:14.08mzoi thought the server was too slow
20:14.12Hmmhesaysnow it dials right
20:14.15lithiDe_Mon: Yea I think its the softphone cause it says "Send CN frames when silence is detected - if silence is detected, it sends empty frames"
20:14.19Hmmhesayswell the way I want it to anyway
20:14.25lithiDe_Mon: Ill try on the idefisk fourms
20:14.30mzoi messed up my aah install by upgrading to 1.2.3
20:14.37mzoit runs fine but all the web stuff is busted, no big deal though
20:15.17Flautohmmhesays, what is the new command to set caller id name in 1.2?
20:15.34AlricSet(CALLERID(name)=boobies) ?
20:15.34HmmhesaysFlauto: guess
20:15.38mzohaha, i was hitting message randomly on my phone and it was actually making calls when i did that, oops :P
20:15.40Flautookay
20:15.43Flautohaha
20:16.08Hmmhesaysfridays were made to be fun
20:16.17Flautoalric, i will do that
20:16.20*** join/#asterisk areski (n=areski@88.Red-88-5-209.staticIP.rima-tde.net)
20:17.50*** join/#asterisk santiago (n=santiago@63.245.86.215)
20:19.15justinu~seen [av]bani
20:19.19jbot[av]bani is currently on #asterisk (2d 14h 26m 49s). Has said a total of 316 messages. Is idling for 1h 9m 58s, last said: '[TK]D-Fender: probably just means it autoprovisions via the linksysone switch'.
20:19.20*** join/#asterisk L|NUX (i=linux@203.101.168.138)
20:19.29justinubani, you around?
20:20.05[av]banino, im asquare
20:20.21justinuyou know how to turn on STUN on the snom360?
20:20.28[av]baniyes
20:20.46justinui filled in the stun server fields, but it still sends SIP messages with it's internal IP
20:20.54Hmmhesaysugh stun
20:20.55[av]baniyou need to set per line
20:20.58*** part/#asterisk santiago (n=santiago@63.245.86.215)
20:21.18justinudid that
20:21.21[av]banitheres a radio button you select under nat settings to enable sending something, i forget what the name is
20:21.26justinuoh, maybe not
20:21.29[av]banihaha
20:21.31[av]bani<3
20:21.44[av]banilet me check
20:22.38Hmmhesays[TK]D-Fender you still around?
20:23.02[av]banijustinu: you just got a snom 360?
20:23.11justinuneat, it works
20:23.14justinui've had it for a while
20:23.18mzosnomstunsnomsipsipsip!
20:23.34[av]banijustinu: just one?
20:23.44[TK]D-Fenderhere
20:23.46justinuyep
20:23.46justinuit's for my lab
20:24.12[av]bania couple things. you hate the icons right?
20:24.24[TK]D-FenderI'm considering getting a Snom 360 to replace my 601 here for my receptionist....
20:24.32[av]bani[TK]D-Fender: why?
20:24.41[av]banineed the 12 blf's?
20:24.47justinuthe icons? i dunno...
20:24.53[av]banijustinu: you can make them text.
20:24.53justinuthe phone interface itself is a little funky
20:24.59[av]baniyeah, the interface is pretty bad :)
20:25.20justinui like the record button tho
20:25.26justinuvery STASI of them
20:25.26[av]baniyeah, that is very cool
20:25.39[av]banione touch record
20:25.40[TK]D-Fender[av]bani : All about Presence.  Thats the point of my receptionist phone.
20:25.53[av]bani[TK]D-Fender: what, no 601+sidecar?
20:26.02De_Monasterisk is quitting, but it's not saying segmentation fault or anything, the processes just die
20:26.14[TK]D-Fender[av]bani : Actually 12 is not enough and when I factor in the sidecar on the 360 it'll cost a fair bit less than the Polycom.  AND it'll work...
20:26.23justinuinteresting... i've got the snom360 talking to SER via TCP
20:26.23[TK]D-Fender[av]bani : I have a 601 + 2 Modules.
20:26.27[av]bani[TK]D-Fender: LOL
20:26.29justinuand SER is talking to Asterisk via UDP
20:26.31justinuneat-o
20:26.36[av]bani[TK]D-Fender: what about xml though? surely thats enough
20:26.51[av]banijustinu: now try srtp :)
20:27.02*** join/#asterisk AlexCTI (n=alex@pembrkfl-bellsouth-24-53-200-134.miamfl.adelphia.net)
20:27.07[TK]D-Fender[av]bani : I mean I wouldn't change ANY other phone here (except maybe to downgrade my CSR's), but the 360 makes sense for my receptionist NOW.  And it DOES cost less.
20:27.15justinuhmm, i dunno if SER is gonna convert that to plain RTP
20:27.19De_MonI'm on 1.2.1, thought it was related to queues but it's still going down with queues off... Looks like I'll be compiling a copy for useful cores now
20:27.20Hmmhesaysyou ever have any experience with behringer DEQ's [TK]D-Fender?
20:27.26[av]banijustinu: there are srtp patches for *
20:27.26justinuSER doesn't have anything to do with the media plane
20:27.39[av]banijustinu: and apparently, snom tested them
20:27.41justinupatches, perpetual patches
20:27.45[TK]D-FenderHmmhesays : NOpe.. I've used a VAMP before from them, but thats it... they are like Line6
20:27.49*** join/#asterisk elg (n=fugalh@hfugal.NMSU.Edu)
20:27.55Hmmhesayshmm ok
20:28.01[TK]D-FenderHmmhesays : I just run a Boss GT-8 now.
20:28.07Hmmhesaysah i was talking about for an EQ
20:28.13[TK]D-FenderI used to be on the Boss VF-1 half-rack.
20:28.23Hmmhesaysbehringer makes a pretty kickass looking digital EQ for 300 bucks
20:28.35[TK]D-FenderHmmhesays : EQ's been on-board every effects processor I've ever had.  Never a seperate unit
20:28.44Hmmhesaysthis is for a PA
20:28.49[TK]D-FenderHmmhesays : Yup.... Behringer is a great value....
20:28.55Hmmhesaysand to quelch feedback
20:29.30[TK]D-FenderPA.. ok, that'd be a thing to do it for.  Of course you could go all psycho and get the "Ashlee Simpson" processor... it even makes it sound like you can sing!
20:29.39Hmmhesayslol REVERB
20:30.06[TK]D-FenderI use Chorus/Reverb/Delay on all of my settings.  Sometimes swap for Flanger...
20:30.11justinulol, now my snom won't reboot
20:30.14*** join/#asterisk Kernel-Kris (n=kkirklan@lfkn-fw.angelinacounty.net)
20:30.36justinustrange phone
20:30.39[TK]D-FenderYesterday learned how to play "DOA" from Foo Fighters....
20:30.40Kernel-Kriscan someone assist me with a vonage softphone and asterisk issue
20:31.10Kernel-Kris:P willing to pay
20:31.30Hmmhesayshttp://www.musiciansfriend.com/product/Live/Sound/Signal/Processors?sku=182484
20:31.33mzoin sex? :P
20:31.33Hmmhesayscheck that baby out
20:31.42HmmhesaysKernel-Kris what issue
20:31.47Netgeeksgreat...  cnn.com leads with two hi profile stories... 6 churches burned in alabama in three hours, and Paris Hilton wants her diaries back.....
20:31.55Netgeekshigh profile
20:32.01*** join/#asterisk dasuberdavid (n=dasuberd@gateway.digium.com)
20:32.04Beirdothey musta been in one of the churches
20:32.08Kernel-Krisincoming calls work fine but outgoing calls say that all lines are busy
20:32.14Hmmhesayswhat did you do?
20:32.20Beirdohow is Paris Hilton STILL news?
20:32.29HmmhesaysI can probably help you, you can pm if you think you wanna pay for it
20:32.29mzoher last name.
20:32.31Beirdostupid rich people :)
20:33.08Hmmhesaysi'd park my car in the paris hilton
20:33.10Kernel-KrisHmmhesays: depens on how much and yes i will im you
20:33.14Hmmhesayspm me
20:33.25[TK]D-FenderHmmhesays : Digital EQ? hmmmm....
20:33.38Hmmhesays[TK]D-Fender yeah, good reviews, lots of features
20:33.42*** part/#asterisk Samoied (n=Samoied@popeye.opens.com.br)
20:33.46*** join/#asterisk dasuberdavid (n=dasuberd@gateway.digium.com)
20:33.47Hmmhesays24bit dac
20:36.08[TK]D-FenderHmmhesays : You know that Eddie Van Halen though the cheap-ass audio demo tape he did for VH3 came out better sounding than the CD due to the difference in dynamic range.  Thats the problem with "digital".  People think "its all that", but you lose something to it every time....
20:36.18*** join/#asterisk j4m3s (n=debbie@gateway.digium.com)
20:37.26justinuit's not just digital... it's the recording engineers mix downs
20:37.26justinusome asshole engineers compress everything into oblivion
20:37.26j4m3scan someone tell me what compiling with "profiling info" (-pg) means?
20:39.07Qwellj4m3s: in what context?
20:39.25j4m3sbuilding asterisk
20:40.06Hmmhesays[TK]D-Fender this is true
20:40.07j4m3s#Include debug symbols in the executables (-g) and profiling info (-pg)
20:40.07j4m3sDEBUG=-g #-pg
20:41.07Hmmhesaysbut digital now is a lot different than digital even 6 years ago
20:41.23Hmmhesaysmy solid state head sounds pretty damn good
20:41.27mzohaha, my audio engineer friend was like 'he's insulting my work, what's his screen name'
20:42.16Hmmhesayswtf
20:42.18Beirdowonder if I have VH3 ripped on my harddrive
20:42.33Beirdodamn, no
20:42.55*** join/#asterisk mogorman (i=root@user-24-236-84-48.knology.net)
20:43.18EriSananyone also has problems using voipbuster, sipdiscount and voipstunt as sip trunk ?
20:43.33dpryoWho doesn't have problems with siptrunks? :)
20:43.36NetgeeksI don't know about now, but back when I worked at a studio, not one person beleived that analog wasn't better than digital for the best master.  However, for home use (where the majority of the arguments occur), it's alot cheaper to get a good sounding digital setup than a good sounding analog setup.  Digital is just cheap
20:43.45Qwelldpryo: people who know what they're doing? :)
20:43.49tainted-did u try voipwanker, rocketvoip, or voip-r-us
20:43.50*** join/#asterisk _cleric_ (n=dacleric@p5482B16F.dip0.t-ipconnect.de)
20:43.57Beirdohahahaha
20:44.24justinuvoip-on-a-stick
20:44.25tainted-voipmart is a good sip trunk for reliability
20:44.28Beirdooh wow
20:44.34Beirdovoipwanker.com is available
20:44.39EriSanif i do a traceroute to them the route seems broken
20:44.41QwellBeirdo: not for long
20:44.46*** join/#asterisk brockj49464 (n=brockj49@233.101.dhcp.hope.edu)
20:44.52BeirdoI'm half tempted to register it and redirect to vonage.com
20:44.53tainted-EriSan well for their dsl connection to cool off
20:45.05tainted-s/well/wait
20:46.02tainted-personally i like sipnsuck, voiplingus and isthatvoiporyourballzonmyforehead
20:46.06*** part/#asterisk UberDuper (n=jsatter@elrond.uberduper.com)
20:46.20*** join/#asterisk mogorman (i=root@user-24-236-84-48.knology.net)
20:46.43*** join/#asterisk nagl (n=nagl@vie-086-059-104-148.dsl.sil.at)
20:46.58[TK]D-Fendertainted- : How can you tell with the dong just hanging in front blocking your view? :D
20:47.21tainted-i use STUN
20:47.24tainted-gets around it
20:47.34mzothat's the phone
20:47.47mzoi want to hook up my shoe to asterisk so i can talk to people over voip maxwell smeart style
20:47.57*** join/#asterisk A-Tuin (n=a-tuin@steves.ip.v4.me.uk)
20:47.59*** join/#asterisk h4mm3r` (n=h4mm3r@85-18-14-10.fastres.net)
20:48.11dpryoWhat should I tell my boss when he sees I sendt a registration for voipwanker.com?
20:48.21Beirdobetter put a 802.11g adapter in the shoe, mzo :)
20:48.23HmmhesaysLOL
20:48.24tainted-lol
20:48.36Beirdo"domain squatting"
20:48.38mzovoipwanker.com is a real domain?
20:48.44Beirdoit could be
20:48.46tainted-looks like it will be in a minute
20:48.48mzohaha, comedy
20:48.50dpryomzo: Soon to be ;P
20:48.57mzocomedy
20:49.01NetgeeksSo I've got a client who has 24 GXP-2000, should I tell him to trash them, or is it even worth trying to make the work?
20:49.03mzopoovoip.com
20:49.08*** join/#asterisk chaos1 (i=anonymou@klover226.bitnet.nu)
20:49.19chaos1hej
20:49.21[TK]D-FenderNetgeeks : Trash 'em and join your fellow legions!
20:49.24_Sam--i'll buy them all
20:49.25*** part/#asterisk mogorman (i=root@user-24-236-84-48.knology.net)
20:49.32NetgeeksYou will Sam?
20:49.35_Sam--yep
20:49.44tainted-voipgoldenshower.com still available
20:49.51Beirdoewww
20:49.51tainted-just letting u all know
20:49.53mzoi'd take one, but if someone buys the set :P
20:49.55Netgeekssend contact info to chris@netgeeks.net  I'll pass it on to my client
20:50.09Hmmhesaysi kind of like voipfister.com
20:50.14*** join/#asterisk miguel3239_ (n=chatzill@h-68-167-124-170.cmbrmaor.covad.net)
20:50.17tainted-voip-n-linens.com still available
20:50.34Hmmhesaysvoipfister.com listen to girls get um...
20:51.03*** join/#asterisk mogorman (i=root@user-24-236-84-48.knology.net)
20:51.06Qwellmeh, chan_bluetooth
20:51.12manygetting fi...?
20:51.20manyqwell: sup with bluetooth?
20:51.42[TK]D-FenderHmmhesays : Plug & Play!
20:51.48chaos1Qwell:is chan_blue stable
20:51.51Qwellgot me
20:51.57Hmmhesaysthe key is not making a fist till your hand is up um... there
20:52.01chaos1Qwell:it used to be an alpha when i last checked
20:52.04HmmhesaysLOL
20:52.08manyqwell: its not.
20:52.11manyerr.
20:52.14manychaos: its not.
20:52.23manyand it is very picky on headsets atleast.
20:52.32Hmmhesayswatch this will work and this dude will disappear
20:52.42*** join/#asterisk Reko_ (n=nonsense@cpc4-hem15-0-0-cust755.lutn.cable.ntl.com)
20:52.46mzovoipfister.com is good
20:52.49Hmmhesaysbetter get his whois info
20:52.57tainted-voipfister lol
20:52.59manys/die/he/
20:53.08justinuHmmhesays: thanks for the tips
20:53.15EriSanhow should i configure asterisk to use a stun server ?
20:53.17tainted-menageavoip
20:53.23Hmmhesaysnp justinu
20:53.32Hmmhesaysis voipfister.com really available?
20:53.46tainted-EriSan http://www.voipfister.com/asterisk-stun.htm
20:54.10BeirdoHmmhesays: yes it is
20:54.12Hmmhesaysbwhahaa
20:54.12chaos1I am trying to forward calls received on asterisk to agents registered on a sip proxy(ser). How would asterisk find whether agents are busy or not as they are not registered on asterisk?
20:54.20Hmmhesaysthat might just be worth registering for 10 bucks
20:54.26Beirdoaccording to whois :)
20:54.44Hmmhesaysit could be like the rejection hotline
20:54.53lithiAre you guys trying to come up with asterisk & porn hosting domain? Asterisk & porn your one stop shop for everything useful.
20:55.06mzoyou need hot girls modelling voip handsets then
20:55.09justinunot just porn
20:55.10mzoin lasvicious positions
20:55.10HmmhesaysHmm lithi I think you're onto something
20:55.12justinufetish porn
20:55.13EriSantainted-, LMAO
20:55.15AndyCap"modeling"?
20:55.18Hmmhesaysthat can arranged
20:55.19dpryovoipfetish.com
20:55.26lithilol voip fetish
20:55.31chaos1lithi::D
20:55.40manyunivoip, metrovoip
20:55.40dpryoHow'bout a .xxx domain? ;P
20:55.41tainted-EriSan i think u've started something really beautiful
20:55.42justinuthose RTP packets you're sending sure get me hot
20:55.45Hmmhesaysyou can make a crack addicted do just about anything for blow, just gotta find one that hasn't wrecked herself yet
20:55.50[TK]D-Fendermzo : And henceforth "handsets" shall now be referred to as either "handjobs" or "fistsets" :O
20:56.04mzoYES
20:56.09Beirdooh jeez
20:56.09mzomy handjobs don't work nomore
20:56.15tainted-voipwhores.com
20:56.15[TK]D-Fender:D
20:56.17Hmmhesaysfistsets
20:56.21mzoI'm a fetish photographer by trade.  Let me know if you're interested. :P
20:56.30*** join/#asterisk ToTo (n=ToTo@host10-211.pool872.interbusiness.it)
20:56.47*** join/#asterisk VJ (n=vijay@203.123.32.80)
20:57.01VJHello,
20:57.18VJi need to do the external conferences using asterisk
20:57.24EriSanAnd this is supposed to be the support channel for Asterisk, pfff
20:57.44lithiEriSan: Didint you know all open source project IRC chatrooms come down to porn?
20:57.48Hmmhesaysits friday people
20:57.49mzoof cpirse
20:57.49dpryoVJ: You want to lookup meetme
20:57.53mzohttp://www.openporn.com
20:58.05mzoopen-source porn, free holes for one and all!
20:58.17Qwelllithi: s/open source project IRC chatrooms/computers/
20:58.17mzogives new meaning to the term 'run time compiler'
20:58.22dpryoVJ: http://www.voip-info.org/wiki-Asterisk+cmd+MeetMe
20:58.35Hmmhesaysi prefer the left nostril
20:59.54EriSanYou guys should open your eyes instead on walking behind your dick
21:00.08EriSans/on/of
21:00.08justinulol
21:00.11dpryohehe
21:00.20dpryovoipdicks.org
21:00.51VJfrom meetme i got an idea how to go the conferencing
21:00.55lithivoipdicks.org - Finishes in 10 seconds or less (tm)
21:00.57justinuby the power of greyskull! I HAVE THE POWER!
21:01.05VJanyone calls up that internal number gets into the conference
21:01.14VJnow lets say i am speakign with a customer
21:01.30VJand i need another customer to be in conference
21:01.40*** join/#asterisk beefsalad (n=crash3m@unaffiliated/crash3m)
21:01.44dpryoAh, and you want to put him in "realtime"?
21:02.01VJand i need to call both the customer
21:02.02dpryoMany phones support that, and also implements it via SIP
21:02.11dpryoYou could alw
21:02.12dpryoerr
21:02.20dpryoYou could also originate all calls to the meetme()
21:02.28*** part/#asterisk beefsalad (n=crash3m@unaffiliated/crash3m)
21:02.32VJhow?
21:02.53*** join/#asterisk [dc] (n=dc@24-205-223-175.dhcp.slto.ca.charter.com)
21:02.59dpryoWould need a little coding, against the asterisk manager
21:03.10[TK]D-Fenderjustinu : That was just scary...
21:03.21[TK]D-Fenderjustinu : Kinda like a friend of mine DLing all of them....
21:03.25dpryoVJ: Does your phone support conference?
21:03.34justinufender: ?
21:03.44[TK]D-Fender[16:00] <justinu> by the power of greyskull! I HAVE THE POWER!
21:03.54justinuoh, heman
21:04.07justinuit just popped into my head
21:04.22_Sam--lol
21:04.37[dc]anybody know where to configure the URL that is sent to a user when they get voicemail notification? (using aah)
21:04.55[dc]presently my users get this url which is useless: > Dial *98 to access your voicemail by phone.
21:04.56[dc]> Visit http:///cgi-bin/vmail.cgi?action=login&mailbox=203 to check your voicemail with a web browser.
21:05.14VJmy phones are only the analog phones
21:05.34VJit might not support conference by itself
21:05.57dpryoVJ: Okay, the best solution might be to get your customers call your conference number.
21:06.02*** join/#asterisk slak- (i=slak@rewted.biz)
21:06.06slak-well hello hello
21:06.13dpryoVJ: Or try to get origniating calls working.
21:06.18slak-i need to add support for international number formet.. 0110xxxxx'
21:06.24slak-to this sipira dialplan stringf
21:06.25slak-(*xx|[469]11|9x.T|0|[1-8]xx)
21:06.26dpryoVJ: So that you can call them, and automaticly connect them to the meetme()
21:06.27slak-HELP!
21:06.36[TK]D-Fender[dc] : ......
21:06.37[TK]D-Fender~amp
21:06.39jbotit has been said that amp is NOT supported here! people using it should join #amportal
21:06.44slak-OH ! TK MY FRIEND
21:06.53slak-help me modify (*xx|[469]11|9x.T|0|[1-8]xx) to support international
21:06.59[TK]D-Fenderslak- : (*xx|[469]11|9x.T|0|[1-8]xx|011.T)
21:07.01[TK]D-Fenderdone
21:07.06VJlets say i have a entry in meet me as 6080
21:07.14VJi got into the conference
21:07.18[TK]D-Fenderactually.. you want to dial 9 in front still for that don't you?
21:07.22VJby dialing 6080
21:07.23slak-TK you're the best *kiss*
21:07.29slak-ya
21:07.35slak-i want to dial 9 for all outgoing
21:07.46slak-(*xx|[469]11|9x.T|0|[1-8]xx|9011.T)
21:07.47slak-?
21:08.09[TK]D-Fenderslak- : ACTAULLY I guess you could say that technically you can use the 9x.T thats already in there as-is.  its a question of what in *'s dialplan to do the routing.
21:08.23slak-well its not working
21:08.28slak-and its in the dialplan
21:08.30slak-in asterisk
21:08.33slak-but not in sipura
21:08.39slak-i get fast busy and nothing in my asterisk console
21:08.47slak-so it crapps out at the ata
21:09.07[TK]D-Fender9 +011 + whatever should work fine....
21:09.07VJthen how to originate the calls from here
21:09.15dpryoVJ: Yeah, you'd need an external number to let the customers call in. Or configure originate
21:09.29slak-Tk you're saying the way it wasbefore should work?
21:09.31slak-well it doesnt!
21:09.32VJcustomers cannot call me
21:09.48VJso, how to configure this originate
21:09.53dpryoVJ: Then figure out originate: http://www.voip-info.org/wiki-Asterisk+Manager+API+Action+Originate
21:09.56dpryo:)
21:10.06pifiuwhy doesnt * support SLA? (shared line appearance)?
21:10.17[TK]D-Fenderslak- : pastebin your extensions.conf and tell me what that phones target context is....
21:10.23NuggetYou haven't written support for SLA yet.  :)
21:10.27[TK]D-Fenderpifiu : Because we're all out to get you!!!!!!
21:10.34pifiulol
21:10.35pifiuseriously
21:10.37pifiuit bothers me
21:10.46pifiui never know if the call is direct to my line or to the general sales line
21:10.47pifiu=(
21:10.49pifiulol
21:10.56[TK]D-Fenderpifiu : like they said... no one has written it.  Its not the most necessary of options...
21:11.05VJok, moreover one major problem i am facing is with DTMF signalling
21:11.06*** join/#asterisk bkw__ (n=bkw_@ppp-70-128-122-10.dsl.tulsok.swbell.net)
21:11.18VJi have an IVR where i need to take customer into conference
21:11.20[TK]D-Fenderpifiu : Knowing who's calling you doesn't require a shared appearance.
21:11.32slak-wait tk
21:11.36slak-i think i see the problem
21:11.38pifiuwell it does when my phone rings both the main line and my direct line
21:11.46VJbut to make the IVR work, the inputs i give to the IVR are not accepted by IVR
21:11.52slak-TK: exten => _011X.,1,Dial(SIP/${EXTEN}@asterlink)
21:11.57slak-that assumes im NOT dialing 9?
21:11.58slak-right?
21:12.04pifiufender, it is in my opinion an important feature, plenty of people are using IP phones with SLA support
21:12.17chaos1will someone care for my question
21:12.25[TK]D-Fenderslak- : You don't have a 9 in front!!!
21:12.33[TK]D-Fendercorrect
21:12.34slak-ya
21:12.36slak-ok
21:12.38slak-solved
21:12.39[TK]D-Fenderyou should add the 9
21:12.44slak-woooooooooooooooooohoooooooooooooooooooooooooooooooooooooo
21:12.46chaos1how to forward calls from an asterisk queue to remote agents registered on SER proxy (non-asterisk)
21:12.49slak-shit stuill busy
21:12.56[TK]D-FenderJust for the halibut!!!!! Diet Coke!
21:12.59chaos1how would asterisk find out whether the other agent is busy or not
21:13.05sevardhalibut, yum.
21:13.13slak-<PROTECTED>
21:13.16slak-wtf
21:13.32[TK]D-Fenderslak- : paste the new dialplan and dial-line
21:14.08slak-what?
21:14.12pifiufender you got PM
21:14.13slak-the new dialplan?
21:14.15slak-nothing got changed
21:14.19slak-i just appended a 9
21:14.25slak-i didnt mod my sipura dialplan
21:14.25*** join/#asterisk n4y (n=tmalkut@fw.orasoft.net.pl)
21:14.34slak-i think its a config issue with my sip provider
21:15.12[TK]D-Fenderslak- : JUST DO IT.
21:15.15slak-ok
21:15.19slak-the 9 messed u
21:15.20slak-p
21:15.22*** join/#asterisk darby_t (i=darby_t@dkp220.neoplus.adsl.tpnet.pl)
21:15.24slak-the 9 gets passed to the SIP
21:15.25slak-heh
21:15.28slak-whihc is ERRRRR
21:15.30[TK]D-FenderEXTEN:1 <-
21:15.37slak-what??!!
21:15.41VJlets say IVR says please enter the customer phone number and i press the phone number, still it says invalid entry
21:15.46[TK]D-FenderI knew it... I just wanted to see if you'd paste it :)
21:15.50VJand it does not accept my input properly
21:16.07VJseems like DTMF singnalling is not happening properly
21:18.06*** join/#asterisk Drew (n=foo@zux221-065-169.adsl.green.ch)
21:18.26[TK]D-Fenderslak- : So.. paste your current dial line....
21:18.55slak-thanks
21:18.58slak-it all ok
21:18.59slak-thanks
21:19.42[TK]D-Fender:D
21:20.02slak-awesome
21:20.03slak-TK
21:20.10slak-i just switched the 20 users we have
21:20.15slak-to use sip for long distance
21:20.23slak-the only issue is that we only have a t1
21:20.27[TK]D-Fendercool..... Whats your termination rate?
21:20.29slak-i'd like to find the optimal codec
21:20.32slak-like 2 cents
21:20.36chaos1will someone care for my question
21:20.38[TK]D-FenderYou can do better....
21:20.39chaos1how would asterisk find out whether the other agent is busy or not
21:20.43slak-im using asterlink..its a great service
21:20.50chaos1I am trying to forward calls received on asterisk to agents registered on a sip proxy(ser). How would asterisk find whether agents are busy or not as they are not registered on asterisk?
21:20.53slak-and i get really amazing latency to them
21:20.56slak-but TK
21:20.59[TK]D-Fendert1?  Surprised you don't ditch the T1 data for DSL.  Big savings.
21:21.06slak-id like to find the best codec
21:21.09slak-well we had dsl
21:21.15slak-but they could only provide so much upstream
21:21.18[TK]D-Fenderslak- : "best" depends on many factors
21:21.23slak-well
21:21.29slak-good quality, and moderate bandwidth
21:21.30[TK]D-Fenderslak- : how many concurrent calls?
21:21.33slak-cos right now i use ulaw
21:21.44slak-like probobly up to 6
21:22.06[TK]D-FenderFor 6 * 10kbps easily fits into your T1....
21:22.17slak-well ulaw isnt 10kbpos
21:22.20slak-its 50k/sec
21:22.20[TK]D-Fenderthats max SIP usage in poor scenarios.
21:22.20*** part/#asterisk marv[work] (n=timr@64.89.118.139)
21:22.21_Sam--plus dsl has no SLA if your circuit goes down
21:22.21*** join/#asterisk DrJones1 (i=mrjones1@ip68-105-251-187.lu.dl.cox.net)
21:22.27*** join/#asterisk marv[work] (n=timr@64.89.118.139)
21:22.30DrJones1anyone know what the telnet command to reboot a 7960 is?
21:22.44pifiufender did you check your PM?
21:22.44mzoboot + foot  = out the window
21:23.01tdonahueis it possible to debug a specific IAX2 peer (or ip address) instead of dumping all the IAX traffic from all our peers?
21:23.05slak-who the hell uses pm's on freenode
21:23.06slak-damnit
21:23.07[TK]D-FenderStill it fits... I'd suggest G726 then.  No license problems, half the bandwidth and better than "psycho" compressed
21:23.19Flautois anyone using stanaphone here? i have been timed out for a while out
21:23.23slak-ok g726
21:23.24slak-thanks
21:23.28slak-and thats all set in sip.conf?
21:23.45slak-will asterisk convert the codec coming from my ATA
21:23.52_Sam--Flauto:  how has your stanaphone service been?
21:23.53slak-that gets passed to my SIP provider
21:23.54Kernel-Krisany suggestions on a sip trunk provider that plays well with asterisk
21:24.06slak-i like asterlink
21:24.07slak-try them
21:24.22_Sam--they dont do local DID
21:24.25_Sam--asterlink that is
21:24.54slak-i like it mainly because of:
21:24.55slak-rtt min/avg/max/mdev = 22.661/24.881/26.998/1.325 ms
21:25.02Flautohi sam
21:25.08slak-and thats unusually high for them
21:25.18[TK]D-Fenderslak- : YUP
21:25.29[TK]D-Fendereek
21:25.39slak-allow=ulaw,g726,gsm            ; Allow codecs in order of preference
21:25.43Flautoit has been okay sam
21:25.43file[TK]D-Fender has met me in real life, he can vouch that I'm a great person and that I know my way around Asterisk and that Asterlink rocks... yes, yes he can
21:25.44slak-i have that in [general[
21:25.52file[TK]D-Fender: yes. you. can.
21:25.53Flautobut now, it is timed out
21:26.16[TK]D-FenderI *could* .......
21:26.23Garak_I'm working on a proposal for our telecom lab at school, I'm looking for some feedback on my idea http://insec.ca/telecom.jpg I'm trying to demo the posible VoIP setups using the mixed bag of equipment on site...
21:26.26EriSanso noone knows how to configure * to use a stun server ?
21:26.28_Sam--file:  all asterlink originations are trunked?  in that i mean if 10 people call my toll free number, your server will keep sending all 10 calls to me
21:26.36file_Sam--: yes
21:26.43_Sam--thanks
21:27.17fileEriSan: Asterisk doesn't have a STUN client
21:27.27slak-i urge everyone to get an asterlink account and give them your credit cards
21:27.31slak-cos its like awesome
21:27.35EriSan:/
21:27.46fileit's awesome when you send the right dial string to us :)
21:27.53sawguys, any of you experienced with DevState()?
21:27.58*** join/#asterisk AlexxGreenberg (n=alex@64.201.13.170)
21:28.13AlexxGreenberg#asterisk
21:28.37slak-file
21:28.45slak-asterlink does support g726?
21:28.49fileyes
21:28.50Hmmhesaysi think file should give me some free asterlink minutes
21:28.59fileHmmhesays: I'll give you a hug!
21:29.26Hmmhesayswoo
21:30.13Hmmhesaysi need some beer and pizza like now
21:30.27fileyes you do
21:30.48_Sam--that is what the hug made you need?
21:30.51Hmmhesaysno
21:30.52_Sam--beer and pizza from file's hug
21:31.00Hmmhesaysi'm going to be away all next week
21:31.02tzangerok this polycom phone's kind of cool
21:31.06filetrelane_: eep
21:31.07tzangerbut ALL the provided rings suck ass
21:31.08_Sam--he made you feel homophobic...so you needed a beer and pizza to prove your manlineess!
21:31.17HmmhesaysI never feel homophobic
21:31.54Hmmhesaysi'm secure in my sexuality thank you
21:32.09_Sam--fair enough
21:32.45chaos1Garak_:that jpg sucks
21:32.45fileHmmhesays: speaking of which - was it lastnight or tonight you were going out with the ladies?
21:32.47*** join/#asterisk elg (n=fugalh@dhcp25.cs.nmsu.edu)
21:33.19fileI'm an #asterisk senior, respect my authority!
21:33.32_Sam--file:  you guys use SER at asterlink?
21:33.46file_Sam--: we use it internally for routing between things
21:33.53Drewi have the right of free speach - respect my oppinion!
21:33.57filenothing visible to the end user
21:34.09_Sam--you couldnt improve the 7 proxy registration setup somehow by using ser?
21:34.26filealready on my list
21:34.32filebut major projects are at the top of my list
21:34.34jbalcomb[TK]D-Fender Turns out we were using rx_fax<SP> to handle our incoming faxing. We took that out and replaced the HT-386 with the SPA-2002. They got thier first fax in a WEEK!
21:34.38Hmmhesaysfile: tonight, last night I was at the warehouse playing guitar
21:34.47fileswitch.asterlink.com will give you round robin though :)
21:35.34fileI just threatened the Florida call center people... that was great
21:35.40file"next person who talks to me about that gets their pay reduced"
21:35.55[TK]D-FenderOK, i'M HEADING HOME.....
21:36.03filettyl
21:36.04*** join/#asterisk Mother_zzZ (n=mother@93.Red-80-32-127.staticIP.rima-tde.net)
21:36.20*** join/#asterisk fulgas (n=fulgas@a81-84-117-84.cpe.netcabo.pt)
21:37.23Hmmhesaysnice
21:39.27trelane_file: poke, check /msg
21:39.58*** join/#asterisk jeffik (n=jeffik@CPE0050babf4cd5-CM014350000760.cpe.net.cable.rogers.com)
21:40.03filemessage not found
21:40.05fileare you registered?
21:40.09Hmmhesaysthere's a few of us that have been around awhile
21:40.10fileand identified?
21:40.13fileif not, no private messages for you
21:40.22Hmmhesaysyou can't get in here unidentified
21:40.28_Sam--hey file, there is tax on VOIP calls through you?
21:40.31fileyes... you can
21:40.45_Sam--i dont know the current state of VOIP taxation
21:40.46Hmmhesayslast time it booted me to asterisk-unregistered
21:40.57file_Sam--: it's the... blast what's that called... universal recovery thing? it's some US thing, I'm Canadian so I'm not the best to answer... but I can override it
21:41.05AlricUSF
21:41.23AlricHmmm.  That sounds wrong.
21:41.33_Sam--Alric:  calls terminated to remote VOIP gateways, are subject to taxation?
21:42.22fileall that stuff was done before I was hired sadly enough
21:42.22Hmmhesaysyou can't tax love
21:42.31fileHmmhesays: yes I can... I can tax it heavily!
21:42.36Hmmhesaysoh that's right
21:48.27VJcan anyone guide me on DTMF signalling
21:48.37Hmmhesaysparanoid people are funneh
21:48.47VJi am connecting to an IVR from asterisk server
21:49.32VJbut server is not accepting the input i am providing
21:49.49VJi mean IVR number is not accepting the input i am providing
21:49.55VJany idea what can be the issue
21:50.01_Sam--dtmfmode =
21:50.26VJits set to rfc2833 in sip.conf
21:50.50_Sam--what is the device
21:50.59VJbut i am dialing through iax
21:51.16VJmy voip device is audiocode
21:51.22Hmmhesaysouch
21:51.24_Sam--you could try dtmfmode = inband and see if it works
21:51.33Hmmhesaysaudiocodes settings suck
21:51.55Hmmhesaysfxs or fxo device?
21:51.59VJfxs
21:52.11VJbut this setting would be in sip.conf
21:52.25Hmmhesaysit has to match what audiocodes has
21:52.31Hmmhesaysand by default inband signaling is used
21:52.42VJaudiocode is set to rfc2833
21:52.45Hmmhesaysi'm growing to hate audiocodes and their fscking voicexml
21:55.10*** join/#asterisk mutilator (i=WebChat@i.think.napoleon.dynamiteblows.com)
21:56.25*** join/#asterisk scon (n=scon@dslb-084-057-000-058.pools.arcor-ip.net)
22:01.55*** join/#asterisk L|NUX (i=linux@203.101.168.138)
22:02.59*** join/#asterisk VJ (n=vijay@203.123.32.80)
22:03.16*** join/#asterisk diegonix (n=diego@200.162.8.226)
22:04.37diegonixtest
22:06.01*** join/#asterisk Reko_ (n=nonsense@cpc4-hem15-0-0-cust755.lutn.cable.ntl.com)
22:06.36*** join/#asterisk P4C0 (n=ash@200.124.22.34)
22:06.42P4C0hello guys
22:06.49*** join/#asterisk Snuffkin (n=nonsense@cpc4-hem15-0-0-cust755.lutn.cable.ntl.com)
22:07.05P4C0how knows a good sip soft phone that works well with asterisk and support artsd?
22:08.36brookshireneat
22:08.40brookshirefor you debian people
22:08.41brookshirehttp://www.backports.org/debian/pool/main/a/asterisk/
22:09.21dpryobrookshire: cool
22:10.07manyill tell nobse thankyou on monday.
22:11.52*** part/#asterisk jimmy_deanPB (n=jhodapp@indianalifesciences.com)
22:12.51X-RobYay. Got paging to work on the snom 220
22:14.55*** join/#asterisk [TK]D-Fender (n=joe@66.11.164.239)
22:16.52Kattyanyone using ingdirect.com?
22:18.21[TK]D-Fender.. And save YOUR money...
22:18.31[TK]D-FenderKatty: Mew.
22:18.43_Sam--brookshire:   Package: asterisk
22:18.43_Sam--Versions:
22:18.44_Sam--1:1.2.1.dfsg-3(/var/lib/apt/lists/ftp.de.debian.org_pub_debian_dists_testing_main_binary-i386_Packages)
22:19.40*** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl)
22:19.55Katty[TK]D-Fender: mew.
22:19.58Katty[TK]D-Fender: was that a yes?
22:20.18[TK]D-FenderKatty: No, I'm just the product of so much otherwise wasted advertising :)
22:20.45[TK]D-FenderTechnically my insurer is owned by them so in a way yes, but not as a "bank"
22:21.01Kattyhmmmmmmmmmmm. k
22:21.16[TK]D-FenderKatty : They are reputable if thats what you're looking for.
22:23.11*** join/#asterisk bigjb (n=nnbigjb@195.60.10.114)
22:24.28Katty[TK]D-Fender: kk (=
22:24.50sevardX-Rob: Are you around?
22:24.55X-RobI am
22:25.21lesouvageWhat is the best place to put a modprobe zaphfc in the start procedure on a debian system (maybe ot but I try it anyway).
22:25.24sevardAwesome, can you help me with an AMP issue?  I got it all narrowed down, I fixed ALLLLLLL of the sql bugs omg i hate sql, now I have a php error that I can't seem to track down.
22:25.43X-Robuhoh. I am the suxx0rz at php, but I'll help 8)
22:25.54X-Robs/help/try to help/
22:25.57Hmmhesaysphp is supa easy
22:26.06sevardX-Rob: When I add a new extension I  get a warning before I click the "red button":
22:26.22sevard<PROTECTED>
22:26.24sevard<PROTECTED>
22:26.41X-Robooh. I think that's a php-version-issue.
22:26.48X-RobI remember seeing that on the mailing list.
22:27.00sevardI didn't get that error before I fixed the SQL issues
22:27.23sevardIt's putting the extensions in the database but I don't think it's putting the extension in the asterisk config files, meaning the phones won't log in
22:27.25dpryoPerhaps it's because the tables are empty? :)
22:27.32sevardThe phones _did_ log in before.
22:27.43sevarddpryo: Which table would be empty?
22:28.03dpryosevard: whatever table that fills the array you're foreach()'ing
22:29.44diLLecare there any recommendations in kernel config for Asterisk ?
22:29.49sevard<PROTECTED>
22:30.02sevardI'm kind of sick of sql and php. my god.
22:30.05Dr-LinuxHmmhesays: please suggest me any php damn good tutorial ?
22:30.09sevardlike physically sick
22:30.21X-Robthat's not a foreach()
22:30.41Dr-Linuxphp is just like Mula omer to me, i never seen it
22:31.10sevardAlright, I don't know _which_ foreach it's freaking out on
22:31.15cpmOkay, question by the think headed. My channel bank channels 1-12 are FXO loop-start ports, channels 13-24 are FXS ports. How to I set zapata.conf and zaptel.conf ? I can't get this right.
22:31.30dpryosevard: That one at line 2597 and 2460 in /opt/lampp/htdocs/admin/functions.php
22:31.30*** join/#asterisk Kohat (n=Nothing@202.125.141.6)
22:31.35dpryoObviously.
22:32.39Hmmhesaysfooooood
22:32.39*** join/#asterisk groogs (n=greg@d221-73-237.commercial.cgocable.net)
22:32.41Hmmhesaysbeeeeeeeer
22:32.45Hmmhesayswhere art though!
22:33.27*** part/#asterisk Norm (n=Norm@normmac.net.wm.edu)
22:33.33many_Sam--: you missed the point entirely.
22:33.54manyand otoh, someone running debian testing.. ohwell.
22:34.16*** join/#asterisk Skkip (n=Skipper@216.160.91.91)
22:34.32*** join/#asterisk Reko_ (n=nonsense@cpc4-hem15-0-0-cust755.lutn.cable.ntl.com)
22:35.15Seldon1975cpm: you using a TDM2400
22:35.39P4C0how knows a good sip soft phone that works well with asterisk and support artsd?
22:36.09*** join/#asterisk elg (n=fugalh@dhcp25.cs.nmsu.edu)
22:36.33dpryoP4C0: artsd?
22:36.44sevarddpryo: http://pastebin.com/537674
22:36.51dpryoP4C0: Like kde arts?
22:37.23[TK]D-FenderHmmhesays : Just got a call from the GF in tears.... lovely
22:37.44dpryosevard: Okay, so $devices never gets defined, nor does $extens whenever that code is ran
22:37.46Seldon1975cpm: zaptel.conf should have    fxoks=1-12   fxsks=13-24
22:37.55manyonly when your gf is pregnant :)
22:37.56*** join/#asterisk Libila (n=vye@ip68-8-174-154.sd.sd.cox.net)
22:37.57Qwell[TK]D-Fender: Gonna have to delay the inevitable now? :p
22:38.03*** join/#asterisk jr_ewing (n=jeanmaro@d80-170-33-44.cust.tele2.fr)
22:38.14[TK]D-FenderQwell : More like NO avoiding it....
22:38.20Qwellheh
22:38.25P4C0dpryo, yes kde arts
22:38.27[TK]D-Fendermany : Not possible....
22:38.39manycant be lovely then =)
22:38.51dpryoP4C0: I think kphone should support it.. though kphone sucks
22:38.54[TK]D-Fendermany : Thats called sarcasm
22:39.01P4C0dpryo, I'm using kphone but the problem is that it takes my dev, so i can't heard music and wait for calls :(
22:39.09manypoor sarc.
22:39.27sevarddpryo: so um :(
22:39.42elgP4C0: have you tried dmix?
22:39.53dpryosevard: I bet there's some configuration to do somewhere
22:39.54elgor arts
22:39.59P4C0elg, no, what's that?
22:40.13elgalsa magic to allow more than one audio channel
22:40.16dpryoP4C0: Checked out x-lite? Dunno what audiodevices it supports... maybe only raw /dev/dsp ...
22:40.28elgactually there by default in newer versions of alsa
22:40.34P4C0dpryo, ok I will
22:40.57cpmSeldon1975, naw, a te110p to a cac channel bank 1 with 1 fxo and 1 fxs card
22:41.15elghowever if it is using oss, e.g. /dev/dsp, then dmix won't help, you'd need aoss
22:41.17cpmso, both should be cool start then?
22:41.24cpmkewl start rather?
22:41.44sevarddpryo: Dude, I'm so in the dark you have no idea :'(
22:42.28cpmSeldon1975: so, for a channel bank, they should be set for what they are, and not the opposite (like I keep reading) ? Yes?
22:44.30dpryosevard: Well, seems like the variables the php-script wants never contains anything useful, they might be defined from a sql-table, and if the table is empty, there won't be an array.
22:45.06Dr-LinuxHmmhesays: please suggest me any php damn good tutorial ?
22:45.08Dr-Linuxphp is just like Mula omer to me, i never seen it
22:45.09dpryosevard: So the problem probably is your database, and not the code it self (though it could be somewhat more spesific on its problems ;P)
22:45.13sevardYou're very right.  I'm unsure how see if the SQL tables are filling in with the information.
22:45.39dpryoselect * from your_table
22:45.40dpryo:)
22:46.26sevardI see my extension in there.
22:48.31*** join/#asterisk Pegger (n=peg@pool-68-163-180-64.bos.east.verizon.net)
22:49.17sevardi can feel the frustration inside of me.
22:52.30sevarddpryo: I go to the portal, add an extension, it shows up in the extensions table under db asterisk
22:52.51brookshire_sam--: what was that up on the top?
22:53.18brookshiretop.. i mean link ;)
22:53.41sevarddpryo: if not there where should I start?
22:56.44*** join/#asterisk bjohnson (n=bjohnson@i216-58-62-9.cybersurf.com)
22:56.56*** join/#asterisk scon (n=scon@dslb-084-057-000-058.pools.arcor-ip.net)
22:57.32*** join/#asterisk darwin_35 (n=darwin35@sta-208-139-193-162.rockynet.com)
22:58.56darwin35ok
22:59.20darwin35any one here know where I can get sipware for some oplder 7905 cisco sip phones
23:01.07Peggersipware?
23:01.22darwin35sip firm ware
23:01.26*** part/#asterisk puzzone (n=Timo@e178103169.adsl.alicedsl.de)
23:01.33*** join/#asterisk mjmtaiwan (n=michael@ip68-100-65-52.dc.dc.cox.net)
23:01.44Peggerhumm cisco locks that stuff down pretty tight
23:01.56*** join/#asterisk elg (n=fugalh@128.123.232.3)
23:03.20darwin35they seem not to want to support these older phone now days but have a client with5 of them they want to get working
23:03.21*** join/#asterisk mcn (n=mcn@ext-gw.newtoncomputing.co.uk)
23:03.50darwin35so thoight maybe someone here might have a idea
23:04.13[av]baninope, no ideas here
23:05.08darwin35yeah its friday I am outa here
23:05.27darwin35have fun weekend time to go do the bar clothes and go out and get drunk
23:06.08*** part/#asterisk darwin35 (n=darwin35@sta-208-139-193-162.rockynet.com)
23:11.32sawdoes one of you guys use a phonebook reverse lookup script?
23:11.48*** join/#asterisk Katty (n=angela@64.82.232.54)
23:12.03*** join/#asterisk scon (n=scon@dslb-084-057-003-105.pools.arcor-ip.net)
23:12.39sevardI'm having problems.  Each phone can dial asterisk (like *98 for mailboxes) but no phones can dial eachother (140 can't dial 777)
23:14.24*** join/#asterisk ard (n=ard@2001:7b8:32d:0:20c:6eff:fe18:d11f)
23:17.30*** join/#asterisk brockj49464 (n=brockj49@63.87.56.252)
23:23.47Zodiacaldoes anyone use asterisk with pstn lines with diguim hardware.  can you make outgoing calls immediatly? mine has a anoying 5 second delay before phones ring on both sides.  im just wondering if its fast for anyone, or if this is a common problem...
23:26.19sevardanyone here? :/
23:26.39*** join/#asterisk tainted_ (n=identd@adsl-71-129-32-116.dsl.irvnca.pacbell.net)
23:31.26*** join/#asterisk tainted- (n=identd@adsl-71-129-32-116.dsl.irvnca.pacbell.net)
23:33.05*** join/#asterisk MarkO (n=info@c-68-39-136-214.hsd1.nj.comcast.net)
23:34.05MarkOlooking for help with outbound chanels
23:36.00MarkOonly uses the last channel
23:36.14MarkOwill not to to next channel for next call
23:36.26*** part/#asterisk SwK (n=SwK@12-219-147-107.client.mchsi.com)
23:37.11*** part/#asterisk MarkO (n=info@c-68-39-136-214.hsd1.nj.comcast.net)
23:40.15*** join/#asterisk MarkO (n=info@c-68-39-136-214.hsd1.nj.comcast.net)
23:40.27MarkOneed help
23:40.53MarkOoutgoing calls only going out on Zap/4-1
23:41.21MarkOincoming calls are hunting 1-4 just fine
23:41.41MarkObut outgoing will not pick up 3,2,1 for outbound calls
23:44.39MarkOname
23:46.14*** join/#asterisk elg (n=fugalh@falcon.fugal.net)
23:46.40MarkOis anyone even here?
23:47.16niZonboo?
23:51.35*** join/#asterisk mut (n=animenod@65.111.201.79)
23:51.52*** join/#asterisk chaos1 (i=anonymou@klover226.bitnet.nu)
23:51.55chaos1hello folks
23:52.01chaos1i am on old redhat  user
23:52.11chaos1i just started using linux again
23:52.23rob0Hello old Red Hat user
23:52.24chaos1i have found that redhat has become rude
23:52.33chaos1so which distro i should use for running asterisk only
23:52.49*** join/#asterisk Brixius (n=Brixius@206.196.34.25)
23:52.52chaos1all my old asterisk 0.4 cvs configs were done on rh7.3
23:52.52Nuggetit doesn't matter in the slightest.
23:52.57chaos1recommend one
23:52.59rob0Use what you like best, nothing is "better"
23:53.02chaos1i like knoppix
23:53.03NuggetMac OS X.
23:53.13chaos1but i want a knoppix that i could install on my laptop
23:53.19rob0Knoppix is fine. I like Slackware.
23:53.34chaos1will asterisk 1.2 work with redhat 7.3 (would there be a glibc issue)
23:53.40chaos1version mis-match
23:53.56chaos1rob0:yes,slack is nice but i am more used to of redhat
23:54.01rob0oh ouch. I wouldn't try something that old.
23:54.05Nuggetprobably not if you try to use a binarty package, but there can't be a glibc mismatch on something you compile yourself.
23:54.10chaos1hey
23:54.17chaos1have you ever thought about trying engarde
23:54.23chaos1engarde linux
23:54.33chaos1i heard its very secure
23:54.38NuggetI dont like linux.  I don't spend any time thinking about trying new flavors of it.
23:54.47rob0No, I don't go distro-hopping, I am settled where I am.
23:54.49NuggetReally, asterisk doesn't care at all.  Use whatever makes you happy.
23:55.22sevardCan anyone give me a hand?  SIP phones are able to call asterisk services such as *98 and such, but SIP phones aren't able to dial SIP phones.
23:55.37chaos1GLIBC version
23:55.45chaos1please tell me which glibc version does 1.2 use
23:55.55Nuggetwhatever version is on your machine when you compile it.
23:55.57*** part/#asterisk brockj49464 (n=brockj49@63.87.56.252)
23:55.59iDunnochaos1: if you want to go redhat, why not one of the newer redhat based distros? CentOS or Fedora spring to mind...
23:56.13iDunno(though, I'm a debianite, so feh)
23:56.18chaos1iDunno:thanks
23:57.54*** join/#asterisk RoyK (n=roy@10.80-203-106.nextgentel.com)
23:58.55*** join/#asterisk austinnichols101 (n=austinni@dsl-10-169.cofs.net)
23:59.56mjmtaiwanHey guys.
23:59.57MarkOneed help outgoing call only going out on Zap/4-1

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