irclog2html for #asterisk on 20031121

00:00.08tessierAnd I don't mean tech support for the OS, I mean commercial application support.
00:00.16denonyeah ..
00:00.19denoneverything's redhat
00:00.28denonthey gots the bucks
00:00.35zoathere is probably a bug in ast_sched_add and ast_shed del
00:00.44zoamaybe only on dual processor systems
00:01.13denonseems like I have some weird cvs issues once in a while
00:01.13jsmithzoa: I wouldn't be surprised... Mark thought there was one in there six months ago, only he never had time to track it down
00:01.22ciycould iptables filter on  a zap interfaces?
00:01.32jsmithdenon: I think it's just you, man.  
00:01.34PowerkillI manage to register with a xten client behind a firewall to my asterisk box i see it on sip show peer but i can make calls
00:01.35*** join/#asterisk killerbee (~Killer@ool-44c1013f.dyn.optonline.net)
00:01.45joakois JerJer around?
00:01.45*** join/#asterisk Exomorph (Greg@134-9.bvcompuworks.com)
00:01.49zoajsmith why did he think that ?
00:01.50jsmithPowerkill: Then learn how SIP works with firewalls.
00:01.52Powerkillwhen trying to dial i see this in sip debug SIP/2.0 407 Proxy Authentication Required
00:02.07zoaPowerkill: you are not registered
00:02.14Powerkillyes I am
00:02.21Powerkill<PROTECTED>
00:02.36Powerkill700/700          212.195.110.242 (D)  255.255.255.255  63522    OK (239 ms)
00:02.47PowerkillI can't dial
00:03.26outtolunc239ms ouch
00:04.06Powerkillyes depend sometime it's lower I'm just downloading some stuff
00:04.39Powerkill700/700          212.195.110.242 (D)  255.255.255.255  63522    OK (55 ms)
00:04.46Powerkillso zoa any idea ?
00:04.51*** join/#asterisk stonefly (~trillian@toby.stoneflytech.com)
00:05.02*** join/#asterisk FryGuy (~fryguy@c-24-2-50-122.client.comcast.net)
00:05.28stoneflyHow do I change how many rings it takes for asterisk to answer? right now it takes two...
00:06.22km-it takes two rings to answer?
00:06.24Powerkillyou can't set number of ring you are setting time
00:06.29Powerkillto answer
00:06.36denonWait() before you answer
00:06.52stoneflyWith no Wait() it takes two rings...
00:06.53Powerkillin the dial command you put ,20,r
00:07.12*** join/#asterisk pino (foobar@host115-28.pool21345.interbusiness.it)
00:07.21km-thats weird
00:07.26km-my x100p picks up instantaneously
00:07.36denonPowerkill: he's talking before Answer(), I think
00:07.38jsmithstonefly: Turn off caller-id, and it'll answer on the first
00:07.43jsharpput "usecallerid=no" in zapata.conf
00:08.00jsmithstonefly: Asterisk has to wait for the second ring to get caller-id info
00:08.11stoneflyjsmith, jsharp, ok thank you..
00:08.21Powerkilldenon may be :)
00:08.23km-jsmith: howdy
00:08.44*** join/#asterisk ambassador (~brad@h143.1.39.162.ip.alltel.net)
00:08.46km-jsmith: I think you have to disable chunked encoding in php, not in apache
00:08.59jsmithkm-: Yeah, that's kind of what I remember...
00:10.03enzobye
00:10.06denonsomeoen tell me how to do it once you guys get it ironed out <g>
00:10.07*** part/#asterisk enzo (blueDJ@ALille-208-1-27-146.w81-51.abo.wanadoo.fr)
00:10.21ambassadoranyone using asterisk and some provider as their "only link" to the PSTN?
00:10.25km-denon: oh dont worry, once I get it man...
00:10.26ambassadori.e. no land lines at all?
00:10.31denonhehe
00:10.59jsharpambassador:  I am for my business.
00:11.04stoneflyI finaly figured out how to route calls based on what zapata channel they come in... contexts.. :)
00:11.15ambassadorjsharp: are you satisified with the service and quality?
00:11.25jsharpYes.
00:11.30ambassadorwhat providor?
00:11.33jsharpnufone
00:11.47jsmithstonefly: Yeah... contexts are important :-)
00:11.48*** part/#asterisk Alric (~nbowyer@masq.hyperusa.com)
00:11.50ambassadorI'm seriously considering it for my place
00:12.03ambassadorkeep only a single analog line for fax
00:12.11ambassadorand have all voice go over asterisk
00:12.13espenzNov 21 01:10:25 guestbox kernel: isdn_tty: call from 91625817 -> 33324240 ignored
00:12.18espenzwhy doesnt asterisk answer?
00:12.22stoneflyjsmith, I'm still getting used to asterisk config files...
00:12.39jsmithstonefly: I see...
00:13.12jsharpambassador:  Once the softfax gets more workable, the IAX providers will be able to offer fax services too if they want to.
00:13.35Mikebkw_: sipura is better than grandstream adapter?
00:13.47pinoespenz: might be that you did not tell asterisk to bind to the right MSN
00:14.04denonjsharp: fax worth using yet?
00:14.16ambassadorjsharp: ever have any trouble with people trying to reach you?
00:14.16km-that fax software is weird
00:14.31denonhehe
00:14.32jsharpDunno.  It no workee with my shitty quicknet hardware.
00:14.42jsharpambassador:  Only when my ISDN line falls over.
00:14.45bevinshas anyone got * running on trustix?
00:14.49jsmithjsharp: So sell it on ebay and get real hardware!
00:15.23pinobevins: someone has. don't remember who, but someone has. :)
00:15.51jsharpjsmith:  I'm holding out to afford a T100P...I've got a big beefy channel bank just dying to be used.
00:16.14denongot a few lines that dont hang up on their own very well
00:16.16bevinsseems zaptel.c line 41... has trouble with usr/include with trustix
00:16.34jrollysonhmm.
00:16.42jrollysonjust thought of something.
00:16.46pinoif you give some more details, maybe someone may help you without running trustix...
00:17.02jsmithjsharp: I'm holding out for a bigger box for my TE410P... :-)
00:17.06jrollysonfor hotel and telco installations, operator style barge in.
00:17.29bevinsits kmod.h is where the problem is.
00:17.41jrollysonoperator barges in silently, but hears scrambled audio.
00:17.44denonman, forgot to adjust wavgain before compile
00:18.21bevinsthe compile dies with zaptel.c line 41 which is #ifdef CONFIG_DEVFS_FS
00:19.10pinothe line has nothing wrong in itself or nothing that could trigger an error; probably the compiler is giving you some more hints!
00:19.36*** join/#asterisk nassy (~nassy@24-193-228-121.nyc.rr.com)
00:19.43zoa<espenz> Nov 21 01:10:25 guestbox kernel: isdn_tty: call from 91625817 -> 33324240 ignored --> you don't have an extension for that number in that context
00:20.22bevinspico: /usr/linux/modversions.h:5:1 unterminated #ifndef
00:20.51pinoi suggest that you compile your own kernel and install its headers.
00:22.43bevinsI was just trying trustix... I am looking for a good clean fast dist...any suggestions other than trustix? I will attempt to compile trustix kernel anyway.
00:25.01pinobevins: it's too often a matter of religion rather than technology. :)
00:25.06*** join/#asterisk JerJer[ghost] (~NunYoBizN@pppoe-66-112-51-206-rb2.grp.centurytel.net)
00:26.10zoadebian Doh !
00:27.08ciydebian rocks the house
00:28.56*** join/#asterisk Poincare (~jeff@D577A9AE.kabel.telenet.be)
00:29.15stoneflyI'm using the CLASS-like features from http://lists.digium.com/pipermail/asterisk-dev/2003-July/001070.html, but CALLERIDNUM for a sip client has a "-" and four random characters after it, and it breaks the features for sip clients. Is there anyway to fix that?
00:29.29stoneflyOr is there a better call forwarding example?
00:29.39*** join/#asterisk spr (~spr@cpe-66-191-202-120.spa.sc.charter.com)
00:30.25zoastonefly maybe add a bug report
00:30.57joakostonefly setup something using astdb
00:31.37stoneflyzoa,  I don't think it is a bug, but something I'm doing, anyways the CLASS-like features isn't part of the main package...
00:31.47stoneflyjoako, that's what I was afraid of...
00:31.48jsmithOr just use a regular expression to strip of the -abcd
00:32.09jsmiths/of/off/
00:33.05stoneflyjsmith, I need to learn AGI anyways, so this is a good excuse...
00:33.37jsmithstonefly: Yes, it is...
00:33.41danielqanyone here using ztdummy?
00:33.42stoneflytoo bad you can't do regex in extensions.conf.. :(
00:34.20pinoserver*
00:34.24*** join/#asterisk Exomorph (Greg@134-9.bvcompuworks.com)
00:34.57tholoOoh!  He's alive!
00:35.09pinonow, less than 3MB of RAM for a working idle * ...
00:35.13zoaaha kram !
00:35.14jsmithkram!
00:35.33jsmithkram, kram, he's our man! If he can't do it, uh.. well.. we're screwed!
00:35.40jsharplol
00:35.50zoahehe lol
00:37.14denonaw man, cant you edit the voicemail email subject in th config?
00:37.14denonbummer
00:37.44mrgobyhas anyone used extendb ??
00:38.05stoneflydenon, are you using Voicemail, or Voicemail2?
00:38.10mrgoby(or written it ;-)
00:38.55bevinsdebian.org is down? hehe
00:39.02zoadebian.org is not down
00:39.51stoneflydenon, I believe that with voicemail  you can't edit the subject, but with voicemail2, you can.
00:40.04denonstonefly: there's no such thing as Voicemail1 anymore
00:40.07denonthey're all vmail2
00:40.20denonI see where you can do the body, dont see anything about the subject tho
00:40.20stoneflydenon, oh, I missed that.. :)
00:40.27espenzanyone, with asterisk and ISDN experience?
00:40.36bevinsI'm not able to get to it
00:41.24bevinssprintlink router.....
00:41.42pinobevins, same for me
00:42.00pinobut if you're looking for it in order to download something, you might use one of the foreign mirrors
00:42.10pinoe.g. ftp.uk.debian.org
00:42.19bevinsI wanna read about it forst
00:42.39*** part/#asterisk outtolunc (~me@h-66-166-19-148.SNVACAID.covad.net)
00:42.45bkw_lalalal
00:42.56Mikebkw_: sipura is better than grandstream adapter?
00:43.05danielqbevins: www.uk.debian.org
00:43.05bkw_um yes
00:43.06pinothen http://www.uk.debian.org/ for example..
00:43.11bkw_strings and cans would work better
00:43.20bevinscool thanks
00:43.27pinoespenz: you still stuck with the "ignored" message?
00:43.52Mikebkw_: i asked for 3 adapters from grandstream thou would be enought to put my wireless telefons
00:44.26jetsIs there any way to end a Meetme conference, or kick everyone out...  we have a script using meetme as a "paging system" of sorts...
00:44.41*** join/#asterisk zeta_ (~zeta@207.88.150.254.ptr.us.xo.net)
00:44.48zeta_is there a way to destroy a meetme bridge?
00:44.50espenzpino: yes
00:44.55mrgobywondering if dynextendb has been used by anyone here?
00:44.56espenzand i changed the trunk to:
00:45.01espenzModem/g1
00:45.07espenzand driver: i4l
00:45.41espenzim testet that i works with the ivcall
00:46.44pinoare your incomingmsn's right?
00:46.58bkw_zeta_ what do you mean?
00:47.09*** part/#asterisk ciy (~2mork@node-402405b2.sfo.onnet.us.uu.net)
00:48.04espenzit works
00:48.05espenzyeah!
00:48.16jsmith-awayespenz: I'm glad to hear you got it working!
00:49.05espenzjsharp: :)
00:49.08espenzjsmith-away even
00:49.14espenzthe fail was
00:49.21espenzthat i forgot to take away the
00:49.23espenz;
00:49.27espenzmsn
00:49.27espenz:P
00:50.34espenzjsmith-away: did you call me?
00:50.57zeta_bkw_, I mean, is there a way that I can programatically destroy a meet-me conference room?
00:51.13zeta_bkw_, for example, say 3 people are in a room... can I issue a command that closes that channel?
00:51.32bkw_soft hangup the channels in the meetme
00:52.03zeta_i tried that but it didn't seem to work
00:52.07zeta_all of the other people stayed on the line
00:52.22km-do any of you have problems with dropped calls on x100p?
00:53.19zoanopez
00:53.22Connorzeta, why would you want to do that anyway ??
00:53.45zeta_Connor, trying to implement a paging system
00:54.06km-see my wife says calls always drop for her
00:54.06Connorok, go on.....
00:54.15zeta_i have it all working, except ZapBarge/Monitor will leave the meetme conference immediatly... and softhangup won't kill the users after the "pager" has left
00:54.17km-but nothing weird ever shows up on the asterisk console
00:54.26km-and then I try and sit down and make like ten calls and it doesnt cut out on me
00:54.40km-it really pisses me off
00:54.59Connorkm-, you sure it isn't a ID10T error ?
00:55.15Connoron your wife's part? No offence or anything..
01:00.47*** join/#asterisk nocnoc (~cvenegas@atlas.ifxnw.cl)
01:01.17*** join/#asterisk crontibs (~crontibs@ool-44c02950.dyn.optonline.net)
01:01.20nocnocguys.. anybody know what this might be? NOTICE[57352]: File chan_mgcp.c, Line 1099 (find_subchannel): Gateway '10.0.0.20' (and thus its endpoint 'd001') does not exist
01:09.18*** join/#asterisk tristan2 (~tristan@213.239.44.133)
01:10.30*** join/#asterisk ricky (~ricky@hoochie.digium.com)
01:11.39espenzcould i get asterisk to forward a call remote?
01:11.56*** join/#asterisk ionix- (~ionix@MTL-HSE-ppp202388.qc.sympatico.ca)
01:16.18*** join/#asterisk Muckl (johannes@pD9ED523A.dip.t-dialin.net)
01:17.30Mucklcan someone post me his working [client] section in skinny.conf for a 7940/7960 skinny client?
01:18.46*** join/#asterisk _spr (~spr@cpe-66-191-202-120.spa.sc.charter.com)
01:24.21*** join/#asterisk cfo_ (~cfo@194.19.190.217)
01:25.31*** join/#asterisk MagicMan (~alm971@APointe-a-Pitre-101-1-5-81.w81-53.abo.wanadoo.fr)
01:29.06*** part/#asterisk _spr (~spr@cpe-66-191-202-120.spa.sc.charter.com)
01:30.48*** join/#asterisk _spr (~spr@cpe-66-191-202-120.spa.sc.charter.com)
01:31.01*** part/#asterisk _spr (~spr@cpe-66-191-202-120.spa.sc.charter.com)
01:32.09km-connor: yeah, I double checked it
01:32.23km-connor: her hands are nowhere near the endcall or the receiver hook
01:33.05espenzhm, how do i turn off the default sounds
01:40.55*** join/#asterisk _spr (~spr@cpe-66-191-202-120.spa.sc.charter.com)
01:40.56Mikehow much does digium take to deliver a X100p?
01:41.13km-YES!!!!!!!!!!!!
01:41.13km-DUDES
01:41.23km-denon: WAKE UP I FOUND IT :P
01:42.00Mikekm-: found what
01:42.04Mike?
01:42.11km-mike: the answer to why 7960's stall with web services on apache
01:42.27tholoHuh.  Never stalled for me yet...
01:42.31Mikethats the wireless cisco right?
01:42.38km-tholo: the problem is chunked encoding (like jsmith said)
01:42.40km-adding this:
01:42.44km-BrowserMatch "Allegro-Software-WebClient/3\.10b1" nokeepalive
01:42.44km-downgrade-1.0 force-response-1.0
01:42.51km-to apache.conf has fixed the problem
01:43.04tholoOh.  I guess maybe I haven't sent big enough responses yet...
01:43.18km-its only a problem if you're using php to generate your XML packets
01:43.19Mikekm-: is that the wireless phone from cisco?
01:43.21km-if you're using raw xml it wont happen
01:43.27km-mike: no, it's the 6 line desk phone
01:43.31tholoAh.  I've been using perl so far.
01:43.41Mikeanyone has the cisco wireless?
01:48.07bkw_km- can you email that to me? brian@bkw.org
01:48.16km-I just fired it to the asterisk-dev list
01:48.18km-should be there shortly
01:52.16km-now that I've fixed that I'm going to have to hook up jsmith with some bugfixes for his directory
01:53.32bkw_OMG
01:53.39bkw_ok my blog is copyright Brian K. West
01:54.02km-did someone steal your blog?
01:54.31bkw_some asshole said this about my Fuck Riaa comments:
01:54.32bkw_Comments:
01:54.32bkw_Oh right, and stealing is cool now I guess.
01:54.32bkw_Ever thought that you're ripping off other people?
01:54.32bkw_"Copyright ©2003 Brian K. West " made me laugh - what a fucken hypocrite. 'What you download is copyrighted you moron.
01:55.08bkw_I don't feel thats being hypocrite..
01:55.09bkw_You are free:
01:55.09bkw_to copy, distribute, display, and perform the work
01:55.14bkw_anything on my website
01:55.30Connorwhat website is this bkw?
01:55.42bkw_www.bkw.org
01:56.05Connorno, the site that stole your comment.
01:56.40bkw_nobody did
01:56.52km-bkw: how much space have you exhausted with your ata-186 password?
01:56.53bkw_some ass posted saying I was a hypocrite for having the copyright notice on my site
01:56.58bkw_km- I gave up
01:57.08km-you gave up?
01:57.09bkw_I might hook it back up later this week
01:57.10km-why?
01:57.14bkw_I did about 10% of the space
01:57.19bkw_I lost intrest in it
01:57.23dougheckaOnly during testing did they find that thermonuclear hand grenade's blast radius was further than anyone could throw it.
01:57.29mrgobybkw_  stealing IS cool
01:57.35mrgoby:-d
01:57.44bkw_copyright infringment isn't stealing
01:57.46km-dougchecka: hahahha
01:57.51bkw_the law makes that clear
01:58.04dougheckabkw_: I could let it run on my system
01:58.08dougheckaI have a x100p
01:58.09doughecka:)
01:58.29bkw_na thats ok
01:58.32*** join/#asterisk UnixDawg (~UnixDawg@ny-lasalle6c-203.buf.adelphia.net)
01:58.35bkw_I will hook it back up next week
01:58.41bkw_and let it be for a few weeks
01:58.41dougheckableh :P
01:58.45dougheckaheh
01:58.53dougheckahow do you know if its the correct number?
01:58.59dougheckawont it keep dialing numbers?
01:59.08bkw_yep but it will still reset and stop responding to numbers
01:59.11dougheckaor will it do something that * can recognize?
01:59.13dougheckaah
01:59.34dougheckawhat is that internet phone jack on pulver?
01:59.45bkw_phone patch
01:59.51dougheckahmm
01:59.56bkw_it turns an FXO into an FXS or vice versa
02:00.02dougheckahrm
02:00.08dougheckasounds awfully complex
02:00.25dougheckaso I hook my x100p to it, and I can hook a phone to it?
02:03.07UnixDawganyone know what happen to benjk
02:03.11learathbkw_: how well do they work?
02:03.17UnixDawghe has not responded in 4 days
02:03.43*** join/#asterisk glLoadIdentity (~asdfrt@abn139-91.interaktif.net.tr)
02:04.05bkw_ok the ata is cracking again
02:04.13bkw_started at 200000 and headed up
02:04.38bkw_er 300000
02:04.46bkw_maybe i will get lucky
02:06.46Mikecan i do a menu that says dial 1 to call x dial 2 for the analog phone to ring
02:06.53Mikeis it posible?
02:07.00Mikethat the analog phone rings?
02:07.07Mikeafter the x100p answers?
02:07.54*** join/#asterisk Zebble (~Zebble@Sherbrooke-HSE-ppp3610369.sympatico.ca)
02:09.07tholoAlmost] anything is possible.
02:10.32jrollysonhmm.
02:17.11Muckloh wow, i just tried voicemail feature, its really great!
02:17.47Mucklkm-: do you use your 7960 as SIP or skinny client?
02:22.18km-mucki: SIP
02:23.24*** topic/#asterisk by kram -> From bug 519: "Yes this patch did fix the blocking issue, now it just crashes. Yes this patch did fix the blocking issue, now it just crashes."
02:23.37km-hahahaha
02:23.44*** topic/#asterisk by kram -> From bug 519: "Yes this patch did fix the blocking issue, now it just crashes. Which from my point of view is much better!"
02:24.19km-kram!
02:24.23kramkm-!
02:24.24km-when's the 4 port fxo coming
02:24.24km-:P
02:24.34krami'm about to scream
02:24.37km-"RSN!"
02:24.40krami'm working on it, *really*
02:24.46km-hehehe :P
02:24.47krambut i have this very weird problem
02:24.59km-oh really...
02:25.34km-been there
02:28.11*** join/#asterisk cypromis (~michael@217.11.142.161)
02:28.24km-cypromis: howdy
02:28.38cypromisaloha :)
02:28.57km-cypromis: were you one of the guys who was interested in the answer to my php problem?
02:29.03km-cypromis: I posted to asterisk-dev about it
02:29.07*** join/#asterisk ThunderZ (~lzhang@cm1-off.syd.au.oztell.com)
02:29.24cypromisnope
02:29.30cypromiswe switched from php to java :)
02:29.45km-hehe
02:29.55km-what do you guys do with java and the 7960?
02:30.46UnixDawgok who shot stealth
02:30.49cypromis7960 ?
02:30.54cypromisshot ?
02:31.02cypromisI helped him setup his * connections lol
02:31.06cypromisbut I did not shoot him
02:31.08cypromishehe
02:31.13cypromiskm-: we do telco stuff
02:31.20cypromisno 7960's yet
02:31.22km-hahaha mark's office is messy
02:31.29cypromisalthough we are playing with the tdm40b
02:31.32UnixDawgwell I fixed it and cleaned it up and got his iax working but now he is mia
02:31.40cypromisand are 85% hapy with the results
02:31.46cypromis:))
02:31.51km-kram: dude, get that invoice out of the camera
02:31.52cypromiswhat did you fix ?
02:31.52km-:P
02:31.53cypromis:))
02:32.05cypromisthe extension mess ?
02:32.11UnixDawghis extensions.conf and his sip.conf
02:32.13cypromisok
02:32.16km-cypromis: makes it so php generated xml packets are shown nearly instantaneously
02:32.19UnixDawgthey work now
02:32.20*** join/#asterisk Carp (Carp@ip-204-97-151-185.modem.logical.net)
02:32.24cypromiswe where playing before with g723.1 passthrough
02:32.41cypromisnice
02:32.46CarpWhere is JerJer?
02:32.48CarpAnyone lknow
02:32.51Carpknow*? lol
02:32.58UnixDawgwell I was chatting him 15 min ago and he stopped chatting
02:33.03cypromishmmm
02:33.11cypromisprobably went to get some pizza ?
02:33.27UnixDawgand now when you call him on his ext 7000 it says he is on the phone but he is not
02:33.32UnixDawgahh maybe
02:33.43cypromishehe
02:33.50cypromishe probably played with it again
02:34.02cypromisit was not easy to explain to him that there is no g723.1 yet in asterisk
02:34.03cypromis:))
02:34.05UnixDawgI needed to get back in to the server to see why its saying he is on the phone when he is not
02:34.14UnixDawgI will
02:34.23UnixDawgits not used yet
02:34.36UnixDawgI for get the new ip
02:34.36CarpI ordered an 800 number from JerJer 4 weeks ago and it still doesnt work.
02:35.04km-crap: that sucks.
02:35.04UnixDawgcarp are you sure your authing right
02:35.28CarpUnixDawg: What you mean?
02:35.40CarpI tried to call the 800 number from my phone and ti says its not valid
02:35.43UnixDawgif its threw iax then you need to make sure your getting the connection iax2 show oeers
02:35.51tholoIf you asked for a particular number, NuPhone is probably still waiting for it to be transferred to them.
02:36.06km-yeah
02:36.19km-vanity 800's take a while, as well as transfers
02:36.22CarpUnix: Its a number redirection
02:36.45km-hmm
02:36.50km-so I've got weather on the 7960
02:36.52CarpIf I knew it would take this long I would have had him just turn up a number he already had.
02:36.55km-and now I've got a directory
02:37.12km-what do I implement now for the 7960!
02:37.29tholoVideo!
02:37.38km-hmm video
02:37.42tholo:)
02:37.48km-you can do that on 7960's?
02:38.11km-I thought that was only something that could be done on the skinny ones, not SIP
02:38.22km-i.e., showing a picture of the person who's calling
02:38.40Carphow can i figure out what phone system my school is using? I cant actually look at the system
02:39.08tholoAsk to see the documentation for it? ;-)
02:39.09km-what kinda phones??
02:40.00CarpI dunno lol. They are the yellow analog ones, wall mountable, has just standers buttons and a button with a lightning bolt on it which I believe they oress toi check their mail.
02:40.08Carppress to*
02:40.16km-hmm
02:40.24km-prolly some older comdial system that used analog phones?
02:40.35CarpIts a new system as of last year
02:40.40km-really
02:40.44Carplast year was the first year the rooms got phones in then.
02:40.46Carpthem*
02:40.48km-who knows what it is
02:42.30Carphow can i make my auto-attendant answer right as the call comes in? rather than 2 rings.
02:42.31km-trip over a desk and accidentally pick up the phone while simultaneously dialing random numbers!  Maybe you'll get lucky!
02:44.09CarpOMG! lol
02:44.19CarpI can get into any teachers mailbox at school without a password lol
02:46.05espenzcould anyone send me a "professional" working extensions.conf? ;p
02:46.15learathwatch out they'll throw you in jail for that
02:46.33jrollysonespenz: check the wiki, theres some links
02:48.25cypromisespenz: there is no such thing
02:48.38cypromisit depends what kind of scenario you have
02:48.45cypromisthere is no UNIVERSAL
02:48.48cypromisextensions.conf
02:49.33espenzi know..
02:50.11espenzcypromis: is it possibly to make it do remote calls?
02:50.21CarpEveryones extensions.conf is different, custom to the setup.
02:51.15espenzof course.
02:51.20cypromisespenz: you an do a pletora of things with it
02:51.37CarpDoes anyone know how to setup an intercom using the sound card?
02:51.37km-pretty much if you can think of it
02:51.43carrarplethora?
02:51.44km-and it involves voice transport over phone or internet
02:51.52km-you can do it with asterisk
02:52.17carrarpletora invloves using the tora drivers? :)
02:52.49km-pletora?
02:53.00carrar(scroll back)
02:53.05km-I know, I'm staring at it
02:53.17km-just wondering when the damned crickets will stop!
02:53.17cypromisgreeklish
02:53.19km-:P
02:53.46km-anyone here have the cisco XML SDK
02:53.58km-I'd like to do "Good Things" with it but I cant remember my cisco login
02:54.11*** join/#asterisk cybyc (~cybyc@Ottawa-HSE-ppp258893.sympatico.ca)
03:02.22Mikei can use asterisk as a soft phone?
03:02.41cypromisyes
03:02.57tholoProvided you have a soundcard with OSS drivers.
03:03.20Mikeyes i have a soundcard with oss
03:03.24*** part/#asterisk km- (pgrace@virgil.fierymoon.com)
03:03.29Mikeso what do i need to make it a softphone?
03:03.55*** join/#asterisk Carp (Carp@ip-204-97-151-127.modem.logical.net)
03:04.07CarpI tried to setup an intercom with the soundcard, what does this error mean"
03:04.09Carp:
03:04.10Carp<< Call placed to 'dsp' on console >>
03:04.11Carp<< Auto-answered >>
03:04.11CarpCalled dsp
03:04.11CarpOSS/dsp answered Zap/1-1
03:04.11CarpWARNING[114703]:  File chan_oss.c, Line 402 (soundcard_setinput): Unable to re-open DSP device or resource busy
03:04.13CarpWARNING[114703]: File chan_oss.c, Line 561 (oss_write): Unable to set device to input mode
03:04.15Carp<< Hangup on console >>
03:04.15tholoMake an extension for Console/dsp
03:04.22CarpI did.
03:04.28tholoThen you get commands "dial" and "hangup".
03:04.40Carpexten => 7777,1,Dial,console/dsp
03:04.53Mikeseams to easy:P
03:04.57tholoCarp: Sounds like maybe something else was using the soundcard.
03:05.23Carptholo, maybe Redhat doesnt reconize it?
03:05.39CarpI dont know if my OS thinks its installed, I've never used it with Linux
03:05.41tholoThe actual device opened is /dev/dsp
03:06.01Carphow do I check if my soundcard is setup properly?
03:06.04CarpI dont know alot of Linux
03:06.23tholoI dunno -- I'm not really a Linux person.  I run a couple of Linux systems just for Asterisk, since there is no hardware support on OpenBSD. :)
03:07.22CarpDoes anyone else know?
03:09.12cypromisCarp: get the patch for chan_oss from kapejod at www.junghanns.net
03:09.15cypromisalso
03:09.20cypromischeck if your soundcard is full duplex
03:09.26cypromisand if you have the right drivers loaded
03:09.29cypromisbest use alsa
03:09.31cypromisdrivers
03:11.31Carpcypromis: I dont know much linux, so I dont know how to do any of that
03:12.00*** join/#asterisk _Chotaire (chotaire@irc.chotaire.net)
03:12.43*** part/#asterisk _spr (~spr@cpe-66-191-202-120.spa.sc.charter.com)
03:15.22*** join/#asterisk decode (~decode@ip-wv-68-117-144-207.charterwv.net)
03:17.25Mikesomeone has gotten sipphone.com 1800 toll free numbers to work with asterisk and GS?
03:17.30Mikeor FWD
03:24.33Mikeexten => _1800XXXXXXX,1,SetVar(SIP_CODEC=g711)
03:24.34Mikeexten => _1800XXXXXXX,2,SetCallerID(${FWDUSERID})
03:24.34Mikeexten => _1800XXXXXXX,3,SetCIDName(${FWDUSERID})
03:24.34Mikeexten => _1800XXXXXXX,4,Dial(SIP/*${EXTEN}@fwd.pulver.com)
03:24.37Miketrying that
03:29.29decodei bet my gf loved the away message i have set..
03:29.37decodeerr will love
03:30.18decodeheh, any good web admin shit for *? :)
03:33.10*** join/#asterisk Tekati (~captain@cpe-66-75-211-60.bak.rr.com)
03:33.57*** join/#asterisk JerJer[ghost] (~NunYoBizN@pppoe-66-112-51-206-rb2.grp.centurytel.net)
03:34.15decode"No, no! 'Rare' implies dangerously cooked. When I say rare I mean just let it look at the oven in terror, then bring it out to me."
03:39.30dougheckahahahahahha
03:39.42dougheckawhats that out of?
03:39.44Mikeiaxtel.com still works?
03:40.45decodedoughecka it's from my ex's bitch-box err journal
03:40.59dougheckaah
03:41.16decodei have no idea, but if i ever speak to her again, i'll ask
03:42.04decodewhoa!
03:42.13decodei'm on AIM using 2 clients with the same SN
03:42.15*** join/#asterisk pbxtech (hash@65.204.194.35)
03:42.21*** join/#asterisk Epitaph (~epitaph@pr37.nji.com)
03:42.23dougheckaWOAH!
03:42.27dougheckathats AMAZING!
03:42.33decodeits not supposed to allow that
03:42.33dougheckaI _cant_ believe it!
03:42.40decodebut apparently if gaim is the second client..
03:43.18dougheckaoh boo yea
03:43.20dougheckaI love newegg
03:43.26dougheckaI send a money order...
03:43.37dougheckaand when it gets there, it ships the same bloody day
03:44.48Mikesomeone knows if iaxtel gives servers this days?
03:44.52Mikeor is it dead?
03:45.12dougheckadead
03:45.15dougheckafor me
03:45.35Mikewhat are you using for toll free?
03:45.59dougheckano
03:46.13dougheckaI called manx's phone number
03:48.00denonanyone know if Matt Florell is ever on irc?
03:48.19denonor have any of you played with his gui * call manager yet?
03:48.21dougheckanever heard of him
03:48.41denonhttp://sourceforge.net/projects/astguiclient/
03:48.47denonhe's on the * list
03:48.54denonI cant get the thing to run tho
03:48.58denonon win32, perl ..
03:49.10denonthink Ive got everything it needs .. throwing lotsa errors though
03:49.19dougheckaah
03:49.33denonget it to work?
03:49.40dougheckanever tried
03:49.46denonwell try, man ..
03:49.47denon:)
03:51.47espenzdoes asterisk take mp3?
03:51.53dougheckano
03:51.57dougheckabut it can play mp3s
03:52.00espenzhow?
03:52.00dougheckausing mpg123
03:52.05espenzjust mpg123?
03:52.08espenznot mplayer?
03:52.17dougheckanope
03:52.23dougheckahas to be the REAL mpg123
03:52.44espenzokey
03:52.47espenzhow do i load it?
03:52.50espenzi just want to test
03:52.56espenzso if you got a line for me
03:53.04dougheckaseems you need mysql.pm
03:53.07dougheckanet-mysql
03:53.09dougheckafind it
03:53.09doughecka:P
03:53.13espenzhm
03:53.14espenzmysql?
03:53.14dougheckaunless you have it
03:53.18espenzwhy?
03:53.20dougheckanet-mysql
03:53.21espenzto play mp3, uh?
03:53.22espenz:P
03:53.26dougheckadoh
03:53.28dougheckawrong person
03:53.39espenzhehe
03:53.58dougheckaNet::MySQL
03:54.49Mikebkw_: alive?
03:55.20espenz;exten => 400,1,MP3Player,song8.mp3
03:55.31espenzexten => 400,1,mpg123,song8.mp3
03:55.41espenzwhere must the song be? in wich catalog?
03:56.48dougheckayou can make it like this:
03:57.23dougheckaexten => 2225,2,MP3Player(/root/ThroneEnd.mp3)
03:57.38dougheckaand just tell it where it is
03:57.45dougheckaand make sure you have the REAL mpg123
03:57.55dougheckaand one problem I have is mpg123 crashing *
03:57.58dougheckaYMMV
03:58.17decodenow to go shopping for a new gf, this one has pissed me off.
03:58.29dougheckaLOL
03:58.36denondoughecka: I have net-mysql on it
03:58.41denonI read the readme .. :)
03:58.42dougheckaoh
03:58.48decodei don't tolerate people telling me they'll be at X place at Y time then not being on time..
03:58.50dougheckaand NET::TELNET?
03:58.55denonyes...
03:58.57decodeor even better just never being there
03:59.00dougheckayou have DCC? :)
03:59.11denonthe errors im getting seem to be more along the lines of coding errors
03:59.14denonunless im mistaken
03:59.16dougheckaah
03:59.25dougheckawell I think it was in the alpha stage
03:59.25doughecka:)
03:59.28decodei bill at 60$/hr for wait time.. :)
03:59.30denon"my" variable $dbhA masks earlier declaration in same scope at C:\AstGui\AST_WIN
03:59.30denonphoneAPP_0.7.pl line 2187.
03:59.30denonName "main::user_switching_enabled" used only once: possible typo at C:\AstGui\A
03:59.30denonST_WINphoneAPP_0.7.pl line 141.
03:59.33dougheckahaaa
03:59.34decodechicks just get dumped
03:59.39denonetc etc
03:59.42denonlotsa errors
03:59.52dougheckainteresting
03:59.57decodeif i have to wait 5 minutes for a client, that's an hour of wait time, at 60$/hr
04:00.01dougheckahah
04:00.10espenzWARNING[262161]: File pbx.c, Line 1160 (pbx_extension_helper): No application 'mpg123' for extension (demo, 400, 1)
04:00.10espenz<PROTECTED>
04:00.13espenzwhats wrong?
04:00.15decode1 hour 5 minutes == two hours, etc
04:00.19denondoughecka: got ideas for me?
04:00.21dougheckayou need mpg123
04:00.25espenzi have it?
04:00.33espenzroot@guestbox:~# type mpg123
04:00.33espenzmpg123 is /usr/bin/mpg123
04:00.33dougheckano, did you put my line in there?
04:00.39dougheckaexten => 2225,2,MP3Player(/root/ThroneEnd.mp3)
04:00.46dougheckaits called MP3Player
04:00.49dougheckanot mpg123
04:01.00espenzroot@guestbox:~# type MP3Player
04:01.00espenzsh: type: MP3Player: not found
04:01.01espenz?
04:01.08dougheckain asterisk
04:01.11espenzok
04:01.12dougheckaits called mp3player
04:01.28dougheckathe app that is needed for playing music is called mpg123
04:01.46decodeand you'd want to do type $(which mpg123)
04:01.47decode:)
04:01.55doughecka:P
04:02.05decodeor `` instead of $() but that doesn't nest worth shit
04:02.23dougheckaI WANT MY FILES IN 0.13 SECONDS!
04:02.47decodei just want a cute, geeky chick who understands i'm extremely impatient and will be on time or at least fucking call
04:03.00dougheckahahaha
04:03.01decodeis that as bad as asking for google?:P~
04:03.10decodeI do not wait more than 5 minutes, for anyone
04:03.22decodeunless they wanna pay me to wait
04:03.35denonthink of it this way .. if you had google on your PC .. it'd take so much ram and storage, you couldnt even boot .. :)
04:03.39dougheckabah
04:03.41denonmuch less get 2 second queries :)
04:03.43dougheckathey have google machines now
04:03.50dougheckaa little 1u box
04:03.52denonyeah .. but its not the whole index
04:03.53espenzdoughecka: it works, but it has a crazy sound
04:03.55espenzwhy? :P
04:03.58dougheckathat searches your whole network
04:04.02dougheckahmm
04:04.05denonnetwork yeah
04:04.07denoninternet no
04:04.07dougheckawhat OS?
04:04.11dougheckadenon: yea
04:04.17denonFreeBSD owns you
04:04.22denonand with that, im gonna go watch a West Wing
04:04.24decodehow does it search? i have no ports open on 99% of my boxen
04:04.31denonits runs on your lan
04:04.35doughecka:P
04:04.36dougheckaNFS
04:04.38dougheckawhatever
04:04.53decodewhy not just set my root password to root, and enable remote root logins? :)
04:04.55espenzdoughecka: do i have to choose bitrate?
04:05.02espenzog insert a encoder ?
04:05.07dougheckadecode: haha
04:05.11espenzor
04:05.12dougheckaespenz: no...
04:05.20dougheckawhat os is it? redhat?
04:05.22espenzwhy, is the sound so crazy.. :/
04:05.24espenzslackware
04:05.27dougheckahmm
04:05.28decodethat's my new phrase whenever a client asks for something insecure
04:05.31Corydon76-homedenon:  West Wing?  You're lagged 26 hours, aren't you?
04:05.48decode"Sure.. and while i do that, i'll set root passwd to root and enable remote root logins"
04:06.19dougheckatry getting another copy of mpg123
04:06.24dougheckait might not be the "real" version
04:06.26decodelike a client earlier today insisting we install wu-ftpd
04:06.37dougheckasome distros put mpg321 and just link it
04:06.40decodeeww
04:06.43dougheckaHAHAHAHHAHAHAHHA
04:06.43decodempg321
04:06.50Corydon76-homewuftpd is perfectly safe, as long as it's the current version
04:07.10Corydon76-homeIt's the old versions that hurt.
04:07.12decodempg321: "Now leaves a bitter taste in your mouth and a painful bleeding in your ears!"
04:07.28decodeCorydon76-home *shrugs* based on it's past history, i still don't trust it
04:07.48decodesame with pine and about anything else written by WU
04:08.11decodealmost to the extent i'm literally afraid of sendmail :)
04:09.00Corydon76-homeActually, sendmail has a distinct advantage.
04:09.05bevinsHey guys, I have a bad module on a TDM400. I have no dial tone on it. I change it to another port and the problem moves. My question is how do I order just the port on the digium site?
04:09.21bevinsJust the module.
04:09.27Corydon76-homeIf your machine is ever compromised by a spammer, they'll only be able to send a fraction of the spam of qmail.
04:09.37decodeCorydon76-home i like postfix :)
04:10.05Corydon76-homeBut it's plenty fast for a small company
04:10.13Mikecan i use a quicknet card to plug any telephone i want? and give it an extension?
04:10.52Corydon76-homebevins:  why not place a sales call to Greg or Malcolm and get a swapout for that module?
04:11.44decodebc he accidently connected the 220v instead of 48v to some network gear and fried it resulting in the dead module? :P~
04:12.05Corydon76-homeMike:  I think so, but the TDM400P is ultimately cheaper per telephone.
04:12.12decodei had a moron client do something of that sort then try to say we did a faulty install
04:13.03decodehow can you connect 220 to a PoE switch instead of the 48v bus?
04:13.26bkw_blah
04:13.47jsharpblah indeed.
04:15.11espenzwtf
04:15.13espenzsuddently
04:15.15espenzFrame#  1100 [    0], Time: 0.03 [28.71], [  459314]
04:15.20espenzcame in the cli
04:15.23espenzuuh
04:18.00decodeheh, anyone know where to get some decently cheap serial terms that can do ansi colour? :)
04:18.25decodei just acquired a 16 port cyclades card
04:18.31decodefor 5$
04:20.57daorkheh, cool
04:21.12jsharpGet a bunch of DEC VT520s and you've got yourself a call center.
04:21.18daorknow all you need is some 9k6 modems and you can make an isp!
04:21.20jsharpThey're not ANSI color.
04:21.23jsharpthough.
04:21.36bevinshow do you kill a zap channel to knowck someone off?
04:21.41decodejsharp i'd really like colour
04:23.00decodejsharp and we just use p133's for that
04:23.01decode:)
04:23.11decodewe can get about 150 p133's for 200$
04:23.34decodebut then monitors/etc cost too :)
04:23.41jsharpYeah, but then you gotta add monitors, keyboards, hard drives or network bootable cards...
04:23.52decodejsharp they all have bootable NICs in em
04:24.03decodeyou can get working pulls from .gov installs cheap :)
04:24.11jsharpYeah.
04:24.38jsharpI used to buy them almost by the truckload from a surplus dealer in Baltimore.
04:24.43decodei hate green/amber on black crt's.. and would never subject someone to such pain
04:25.06jsharpThat's why you make sure to get the VT520s with the white phosphor.
04:25.15decodeOOo
04:25.20bevinsIs there someway to knock someone off a Zap channel? destroy?
04:25.22decodethose would be eye friendly :)
04:25.29decodebevins softhangup
04:25.42jsharpYah.  I love mine to death.  I'd never part with it.
04:25.44decodeas for usage, i have no idea
04:25.48decodejsharp heh, est price?
04:26.27*** join/#asterisk _gorman (~lehmann@pD950FE4B.dip.t-dialin.net)
04:27.58decodejsharp i'm pondering getting a 32 port cyclades card, some terminals, some decent ip phones, and setting up asterisk w/ a module to speak to some CRM sw w/ a curses interface heh
04:28.22jsharpI've been working on something similar.
04:28.34jsharpAn answering service package.
04:29.00decodeahh, i'm debating doing outsourced cust. service for local small businesses :)
04:29.06decodeand phone ordering/etc
04:29.22jsharpBut I hadn't thought about doing it with dumb terminals...I was going to wrap the software and softphone all into one package.
04:29.28decodei have about 6 years of experience in marketing :)
04:29.30jsharpSo you just need a PeeCee with a soundcard.
04:29.38decodewell direct marketing
04:29.46decodeheh.. aka telemarketing/fundraising/etcx
04:29.48jsharpOr hell, it doesn't even need to be a PeeCee.  I was doing development on a Sparc 5.
04:30.15atacommwow, AT&T rocks
04:30.27*** join/#asterisk h3x (~hex@user57.net472.lv.sprint-hsd.net)
04:30.28learathBlasphemy!
04:30.56atacomm... lets sue eBay and Paypal because we patented payments over a communications system in 1994, rofl
04:30.59decodejsharp i was thinking about actually having one of our dever's write a decent softphone that will fit into the curses interface for the CRM :)
04:31.46jsharpYup.  That's what I was doing.  That part was easy.
04:31.48decodeheh
04:31.54atacommof course, maybe AT&T will team up with SCO since AT&T used to own it, buy them back, and continue the suing over linux/bsd
04:31.55decodei dunno what approach we'll take yet
04:32.14decodeIt's all going to depend on a few meetings :)
04:37.37Corydon76-homewb, kram
04:37.42kramthanks
04:37.54Corydon76-homeSure has been a busy week on the mailing list
04:38.03Corydon76-homeand contentious
04:39.39bkw_yep
04:41.14*** join/#asterisk mrgoby (~mrgoby@pcp05304587pcs.wanarb01.mi.comcast.net)
04:42.05mrgobyhowdy all !!
04:43.08*** join/#asterisk h3x (~hex@user57.net472.lv.sprint-hsd.net)
04:44.59*** join/#asterisk ant_wood (~ant_wood@hoochie.digium.com)
04:51.42*** join/#asterisk phsdshft (foobar@h00080e299383.ne.client2.attbi.com)
04:51.51phsdshftbkw: you there?
04:54.35phsdshftkram: are you there?
04:54.55krami'm having private chat time with bkw
04:55.43mrgobywhat patch is the topic refering to?
04:55.49mrgobyor rather what is the bug?
04:55.57mrgobywow, i must be lazy
04:56.00mrgobythe number is right there
04:56.05mrgoby:---)
04:56.08bkw_phsdshft yes i'm here...
04:56.09phsdshftkram: oh hehe.. So your probably aware that I revised the original patch to not use strtok
04:56.54phsdshftbkw: I just was monitoring the bug db.. but I take it you are already talking to kram about it
04:57.04bkw_phsdshft yes I informed him
04:57.14bkw_I just msged you what kram recommended
04:57.46phsdshftcool
04:57.47bkw_its more elegant anway.. because we can't assume that strsep will return non-null
04:57.52phsdshftah
04:58.21bkw_also not on my diff we need to move that first if ending } to the bottom of the dring stuff
04:58.34bkw_because we are only going thru that mess if we have them set
04:58.43phsdshftah.. I haven't even looked at your diff yet.. I have had like no time :(
04:58.49bkw_:)
04:58.49*** join/#asterisk Lafinion (~Lafinion@test.incracow.com)
04:59.07bkw_I took the bug tracker today.. turned it upside down.. and beat the hell out of it!
04:59.08bkw_haha
04:59.16phsdshftlol
04:59.55*** join/#asterisk cman (~arun@202.51.76.140)
05:00.01cmanhello
05:00.15cmani cannot dial iaxComm from zap phones
05:00.21cmani can di he reverse
05:03.21*** part/#asterisk decode (~decode@ip-wv-68-117-144-207.charterwv.net)
05:03.30*** join/#asterisk decode (~decode@ip-wv-68-117-144-207.charterwv.net)
05:03.40decodeoh fscking peachy
05:03.53decodewinamp just killed my sound drivers.. god i hate windows...
05:03.58*** join/#asterisk heller (heller@voltar.wacked.org)
05:04.36Adamhow is windows to blame for shitty sound drivers?
05:04.41*** join/#asterisk cocoy (strange123@ipdial-171-178.tri-isys.com)
05:04.50cocoyhi people
05:04.54cocoycan any help me
05:05.03Adamask and you shall see
05:05.10cocoyabout setting SIP user on asterisk
05:05.20*** join/#asterisk Zebble_ (~Zebble@66.207.107.50)
05:05.35Adamask the question
05:05.36cocoyso that my SIP client software can loggin
05:05.51cocoyhow do i create sip user on asterisk?
05:06.34cman[sipuser]
05:06.34Adamlook at examples in /etc/asterisk/sip.conf
05:06.43cmantype=friend
05:06.48cmanusername=sipuser
05:06.55cmansecret=sipuser
05:07.00cmanhost=dynamic
05:07.21cmancontext=sip
05:07.37cmannow in extensions.conf.. u have to diefine sip context
05:07.39cman[sip]
05:07.57cmanexten=>100,1,Dial(SIP/phone1,20,r)
05:08.00cmantr
05:08.05cmanr=tr
05:08.07cmanetc....
05:09.16mrgobywhat are those extra parameters you put in dial cman ??
05:09.41cocoywhat if i have multiple SIP user who want to login to asterisk?
05:09.59mrgobythere is a 'catch all' user conf
05:10.06mrgobyat the top
05:10.11mrgobyof the sample config
05:10.32cmanSIP can be replaced by Zap,IAX etc
05:10.49cmanphone 1 is the username of the phone u dial.. such as sipuser
05:10.56cmanor sipuser2
05:11.10cman20 is the no of sec o dial he called no
05:11.11*** join/#asterisk loko_moko (loko-moko@c-67-165-107-230.client.comcast.net)
05:11.13mrgoby[general]
05:11.13mrgobyi think
05:11.13mrgobyin sip.conf
05:11.18ant_woodAnybody know chan_capi and/or Australian ISDN BRI?
05:11.19cmanand tr helps u to transfer the call
05:12.03mrgobycman wait, i was trying to figure this out before
05:12.26mrgobydo you know how to set up a call extension and have it automatically transfer the second that person picks up?
05:12.35mrgobysomeone else advised i use meetme
05:12.40mrgobyand a .call file
05:13.07mrgobyis there a way i can do it in extensions.conf with a 'transfer' command?
05:14.03mrgobyi don't really want to set up a conference.... more of an automatic 2party bridge
05:15.39bkw_cman username= is pretty much useless since the part in [] is used
05:15.42cmanwhere is the hell password in xlite????????????????
05:15.51bkw_!info xlite
05:15.52AiNFOxlite - a popular SIP softphone available from http://www.xten.com for Windows and OSX. see also the getting started guide at http://www.zebraroaming.com/stuff/X-Lite-Getting-Started.pdf and example configuration for Asterisk at http://www.zebraroaming.com/stuff/X-Lite-and-Asterisk.pdf
05:15.56cmanok
05:16.21jsharpgrrr.  Why do telephony protocols have to be developed by committees.
05:17.16bkw_NYQUIL IS GOOD!
05:17.28mrgobybkw_ is that right???  the username in sip.conf is actually the bracketed name??
05:17.28bkw_jsharp dont think IAX was?
05:17.28cmanmrgroby.. don't know
05:17.36bkw_mrgoby yep
05:17.49mrgobythat is misleading IMHO
05:17.49cmanyes mygroby.. it looks for the name in []
05:18.14cmani had a headache when i named [] and username another...
05:18.21mrgobywell...  it worked by accident in my case then :-)
05:18.41cmani couln't igure out what went wrong
05:19.00ant_wood!info chan_capi
05:19.00AiNFOchan_capi - http://www.junghanns.net/asterisk/
05:19.10ant_wood!info fritz
05:19.13mrgobyyeah.....  that can be troublesome...
05:19.49cmani need info in xlite....
05:19.59cmanhat is the codec o use...
05:20.17cmang711u.. ulaw?
05:20.43bkw_yes
05:20.53cmani see g711u in xlite... if i need to define codec in sip.con i say allow=ulaw?
05:21.53cmani can' find out the password in xlite... damn, where has it gone?
05:21.59cmani want to reset pwd in there
05:22.26cmandoes anyone know it?
05:23.26bkw_!info xlite
05:23.27AiNFOxlite - a popular SIP softphone available from http://www.xten.com for Windows and OSX. see also the getting started guide at http://www.zebraroaming.com/stuff/X-Lite-Getting-Started.pdf and example configuration for Asterisk at http://www.zebraroaming.com/stuff/X-Lite-and-Asterisk.pdf
05:23.28bkw_read those
05:23.35bkw_its pretty much a howto
05:23.53cmanok i found it
05:23.53cmanthx
05:25.32cmanok i got it registered... but thesound quality is worst
05:25.48cmani see errors like.. Unknown RTP codec 72 reeived
05:25.52cmanany idea?
05:26.29*** join/#asterisk cfo (~cfo@194.19.190.217)
05:27.18cocoyi able to to logged my sip user, what extension number can I dial?
05:27.43cmanhow many users are there?
05:29.14*** join/#asterisk decode (~decode@ip-wv-68-117-144-207.charterwv.net)
05:29.15decodeweee
05:29.32decodeblitzrage you alive?
05:29.47cocoyone only..
05:29.54cocoywhen i dial it shows " File chan_sip.c, Line 4489 (handle_request): Failed to authenticate user "
05:30.18cocoysorry guys, am just starting to learn how this asterisk works
05:30.43mrgobyNOTICE[1217662256]: File sched.c, Line 209 (sched_settime): Request to schedule in the past?!?!
05:30.47cocoyi already added my user on "sip.conf" , am i right?
05:30.50mrgobydoes anyone know what this means?
05:30.54mrgobyfrom the current build?
05:30.56decodebrb, one more reboot
05:30.57*** part/#asterisk decode (~decode@ip-wv-68-117-144-207.charterwv.net)
05:31.52bkw_mrgoby it means you need to stop running in debug mode so you wont ask stupid questions
05:32.04bkw_;)
05:32.15mrgobyouch
05:32.26bkw_oh lord poor decode.. thinks he's running windows or something.
05:32.28mrgobybkw_ that really hurts man
05:32.31mrgobyreally
05:32.33mrgoby:-D
05:32.40bkw_mrgoby haha sowwy.. i'm really blunt
05:33.05cmanwhats ur sip user???
05:33.07cmanxlite?
05:33.11bkw_to be honest you shouldn't be in debug mode like that till you understand that most of that stuff is just informational and has really no meaning unless your hunting down a problem.
05:33.20cocoyi use xlite, ren@routpeer.com
05:33.31mrgobyi hear ya....  vvv is force of habit i guess
05:33.38bkw_I use safe_asterisk
05:33.40bkw_then asterisk -r
05:33.46bkw_tail -f /var/log/asterisk/messages if I need debug
05:33.55mrgobygotcha
05:34.08bkw_that way you can exit without killing *
05:34.11mrgobythat's not a bad idea
05:34.16bkw_or having to run it in a screen like some goobs do
05:34.30cocoyi got my user logged in to asterisk, but says can't authenticate user when i dial
05:34.33mrgobyscreen?
05:34.45bkw_mrgoby don't ask
05:34.58cmanbkw
05:35.00mrgobysorry, bad habit that
05:35.15mrgobyi'm an interrogative-type
05:35.19*** join/#asterisk IronHelixz (IronHelix@ool-182c7020.dyn.optonline.net)
05:35.30cmanwhat can be done??? i get unknopwn rtp codec 72 received.. in using xlite and the sound isn't that good
05:35.55cmanif u have any idea?
05:36.27bkw_cman its xlite.
05:37.42cmanany guy from nufone?
05:38.29bkw_just call for the NuFone gods.. they will decend apon you.
05:38.57*** join/#asterisk GotX (~Om3gAnGeL@pcp01540411pcs.huntsv01.al.comcast.net)
05:39.05cocoymy sip user still Fails when dialing someone
05:39.16bkw_cocoy what is the error message?
05:39.44cmandid ur sip regiser ith *?
05:39.57cocoy<PROTECTED>
05:40.03mrgobyboy... sjphone sure is purty
05:40.13cocoyi use xlite
05:40.16cmanthat means u haven't registered
05:40.18bkw_!info xlite
05:40.18AiNFOxlite - a popular SIP softphone available from http://www.xten.com for Windows and OSX. see also the getting started guide at http://www.zebraroaming.com/stuff/X-Lite-Getting-Started.pdf and example configuration for Asterisk at http://www.zebraroaming.com/stuff/X-Lite-and-Asterisk.pdf
05:40.20bkw_did you read that?
05:40.28cman!info xlite
05:40.28AiNFOxlite - a popular SIP softphone available from http://www.xten.com for Windows and OSX. see also the getting started guide at http://www.zebraroaming.com/stuff/X-Lite-Getting-Started.pdf and example configuration for Asterisk at http://www.zebraroaming.com/stuff/X-Lite-and-Asterisk.pdf
05:40.33bkw_cman you don't have to register to make calls.. you just have to have the info right.
05:40.38cmanoh waaa.. ididn't kno tha
05:40.39Mikecan i use a quicknet card to plug any telephone i want? and give it an extension?
05:40.40mrgoby!info sjphone
05:40.42bkw_so when it goes to make the call it can auth with that info
05:40.43mrgobydoh!
05:40.59bkw_cman stick around you will learn grass hopper
05:41.01jsharpMike:  Yes.
05:41.11mrgobybkw_ how you do that?  info?
05:41.20GotX!info
05:41.21cocoymy user already regsitered, why  when i dial any number it says Fail to authenticate user <sipuser>
05:41.24Mikeso i can put my quicknet card and plug the phone and how do i give it an extension
05:41.24Mike?
05:41.25cmani knew that sip don't have to register
05:41.26mrgobyi'm new to irc as well.....  NEEEWWWW-BEEEEE
05:41.33bkw_show us your sip.conf entry
05:41.39cmani didn't know !info could bring that ino
05:42.03GotXwhoa...my screen just turned pink
05:42.06cmanfrom the error it seems that he is trying to register xlite to *
05:42.08GotXthats kinda cool
05:42.11mrgoby!info xlite
05:42.13AiNFOxlite - a popular SIP softphone available from http://www.xten.com for Windows and OSX. see also the getting started guide at http://www.zebraroaming.com/stuff/X-Lite-Getting-Started.pdf and example configuration for Asterisk at http://www.zebraroaming.com/stuff/X-Lite-and-Asterisk.pdf
05:42.18mrgobyyah!
05:42.22mrgobysorry, guys
05:42.23cocoy[general]
05:42.23cocoyport = 5060
05:42.23cocoybindaddr = 203.144.224.167
05:42.29GotXlol @ mrgoby
05:42.33cocoy[ren]
05:42.34cocoytype=friend
05:42.34cocoyhost=dynamic
05:42.34cocoysecret=zxcasd
05:42.34cocoycontext=sip
05:42.34cocoypermit=0.0.0.0/0.0.0.0
05:42.51bkw_oh shit
05:42.53bkw_can I smack you know
05:42.57cmanu don't have to define permi and bind addre....
05:43.14bkw_cman you do if the hostname won't resolve
05:43.21cmanso make 2 or 3 sip users
05:43.22bkw_because chan_sip won't load if the hostname isn't resolvable
05:43.29bkw_that permit part can go
05:43.42cmanok
05:43.56cmanis ur configuration in xlite- ok??
05:44.05cmandoe it say logging in.. and logged in??
05:44.12cmanwhen u start
05:44.16cmanxlite
05:44.41*** join/#asterisk Landrocker (siC591746@203-118-171-237.adsl.ihug.co.nz)
05:45.02cocoyyes says logged in
05:45.16cocoybut whne i dial says fail to authenticate
05:45.19cmanso what did u define in extensions.conf
05:45.36Landrockersetting up x-lite with *?
05:45.39bkw_you don't have to define it to dial
05:45.46cmanap
05:45.48bkw_you just have to have a sane dialplan
05:45.55bkw_or a dialplan at all for that matter
05:46.00bkw_sip show peers
05:46.02bkw_shows it registered?
05:46.16cmanye
05:46.29cocoywhat do i need to put in realm?
05:46.30bkw_show me
05:46.34bkw_no realms
05:46.39bkw_wel
05:46.42bkw_thats the ip of your * server
05:46.45bkw_if I recall
05:46.47cmanin the CLI> type sip show peers
05:46.56cmanu shpould get name/'username hosts etc
05:46.56bkw_I don't use xlite because its crap
05:47.10cmanwhat do u use bk_
05:47.14Landrockerwhat do you use bkw_?
05:47.43cmani treid DIAX.. eco problem.. iaxcomm.. bugs.. no trying lite... sound problems..
05:47.43bkw_hardware phones.. ata-186's and 7960's
05:48.06Adampatience i say, await my soft iax phone
05:48.33cocoyam logged in now, what do i dial?
05:48.43bkw_dial 1000
05:48.45Landrockerbkw_: /me is too poor for voip phones :(
05:48.45bkw_for the demo
05:48.46Adamactually i wanted to ask Mark some questions
05:48.59bkw_Landrocker I am too.. but we all have to sacrifice something
05:49.11cocoy"404 Not found"
05:49.20*** join/#asterisk clive- (~pirch@rndf-ip-nas-1-p160.telkom-ipnet.co.za)
05:49.28bkw_what context you have in your sip.conf peer entry?
05:49.28mrgobyhas anyone had problems with mic capture on linphone with alsa ???
05:49.45*** part/#asterisk GotX (~Om3gAnGeL@pcp01540411pcs.huntsv01.al.comcast.net)
05:50.22AdamMark/Kram: With IAX2, user A isn't NAT'd user B is NAT'd, if * does a supervised transfer, will they realise this (A will receive a packet, B won't) and be able to talk directly?
05:50.26cocoyhow do i call the voicemail? i dialed 1000 , it said "404 Not found"
05:50.50bkw_8500
05:51.01bkw_but WHAT IS YOUR CONTEXT= LINE IN SIP.CONF or do I need to yell?
05:51.03bkw_:P
05:51.12bkw_you have to specify a context for your device
05:51.43cmanwhere do i define incoming callerid so that i can see it?
05:52.21bkw_cman have you even googled yet?
05:52.46mrgobywell, cman , i do not recommend linphone
05:52.54mrgobyor at least not for me
05:52.57bkw_hehe
05:53.01bkw_use hardware phones they ROCK
05:53.14mrgobyi need to make my 'boss' buy one
05:54.17mrgobyi like sjphone so far, although the sip config is limited, or at least non-intuitive
05:54.27mrgobybut
05:54.28cmancan anone tell me how to receive caller id if someone dials from outside
05:54.33mrgobyno leenuchs client
05:54.43mrgoby:-(
05:54.43cmani have usecallerid=yes in zapata
05:54.53cmani don't think that is the one
05:55.08bkw_cman it just happens
05:55.35cmani am not geting caller id
05:55.42bkw_is your telco sending it
05:55.42mrgobyanyone used dynExtenDB ???
05:56.31*** join/#asterisk monsieur (~monsieur@81-86-185-223.dsl.pipex.com)
05:57.03cocoyhow can a sip user make  a call that will pass thru * then to gatekeeper
05:57.12cmanyes.. when i connect directly i gives me caller id.. but if i connect thru * it is not giving
05:57.22monsieurhas someone got a script for running through all the kernel recompile steps in an automated way?
05:57.33mrgobycanreinvite=no forces the audio through * , no?
05:57.42monsieurmrgo: ye
05:58.08mrgobycocoy  i would start by adding canreinvite=no to sip.conf
05:58.28mrgobythen route through gatekeeper
05:58.37cmanbk
05:59.24cocoyanymore things i need to change other than on sip.conf, to do sip->*->gk
05:59.38mrgobydunno cocoy
06:00.01cocoycanreinvite=no, where? on [general] ?
06:00.22*** join/#asterisk Exomorph (Greg@134-9.bvcompuworks.com)
06:00.32*** part/#asterisk Exomorph (Greg@134-9.bvcompuworks.com)
06:01.07mrgobyi don't know [general] confuses me, you set globals there for the whole sip conf .... but which are valid there an which are valid under particular user configs?
06:01.13Mikeanyone can help me with 1800 numbers?
06:01.20Mikewith sipphone or fwd or aixtel
06:01.22Mikeiaxtel
06:01.25Mikeor any?
06:03.34mrgobysssssooooooooooo,  how about that dynamic extensions module, dynExtenDB, boy that's a cool one, eh??  anyone used that?? :-)
06:03.48mrgobyanyone?
06:03.55mrgoby;-)
06:05.09monsieurmrgo: we used it. Apparently it doesn't scale very well but we haven't tried on big installations
06:07.29mrgobythat is what i read from Uriel's post
06:07.33*** join/#asterisk ant_wood (~ant_wood@hoochie.digium.com)
06:07.33mrgobyis that you?
06:07.36mrgoby;-D
06:07.41Mike<PROTECTED>
06:07.41Mike<PROTECTED>
06:07.41Mike<PROTECTED>
06:07.41Mike<PROTECTED>
06:07.41Mike<PROTECTED>
06:07.43Mikeit dies
06:08.22mrgobydid you have to modify the source to get it to work monsieur ?
06:09.01monsieurno
06:09.22mrgobyhmmmm....  very interesting....  
06:10.28cocoycan anyone send me a working sip.conf sample? and extensions.conf too
06:10.29*** join/#asterisk Exomorph_ (Greg@134-9.bvcompuworks.com)
06:10.37*** part/#asterisk Exomorph_ (Greg@134-9.bvcompuworks.com)
06:11.06mrgobymonsieur , by not scaling, do you mean that on a single machine several sip extensions in the DB effects performance significantly??
06:11.57mrgobyunderscore rubbing you the wrong way gorman??
06:12.45*** join/#asterisk tristan2 (~tristan@213.239.44.133)
06:12.52monsieuri'm talking about 250+ extensions. But don't take my word for it.
06:13.32mrgobycocoy:  the sample config files work just fine for me as a basis for building new extensions.....  what are you trying to do ??
06:14.25cmani can't get caller id???????????
06:14.30cmanbut why?
06:15.39mrgobyshoot ...  andreasotto.net is down
06:16.24*** join/#asterisk davidhindmarsh (~dhindmar@CPE-144-137-125-115.nsw.bigpond.net.au)
06:17.21mrgobyMr. Otto .... are you there ??
06:17.59mrgobyandio  where are you??
06:18.38davidhindmarshHi, anybody used the H.323 and Sip to provide a gateway betwwen the two protocols.  SIP > H.323 and H.323 > SIP
06:19.41mrgobydavidc: yes
06:19.52mrgobydavidhindmarsh: yes
06:20.10davidhindmarshI have a request to bring some H.323 users into asterisk,  what components do I need to install and where
06:20.45mrgobyhmmmm ...  honestly, i didn't set it up ...  my partner did ...
06:20.54mrgobyhave you tried it?
06:21.04jsharpread the README in asterisk/channels/h323
06:21.15mrgobyi believe you have to recompile H.323 into your kernel
06:21.19mrgobythat is what we did
06:21.27mrgobyi believe
06:21.49davidhindmarshI have installed the h.323 and that is apparently working, I see it loads and listens, then what?
06:22.02mrgobydid you try calling ?
06:22.33davidhindmarshI could not get netmeeting to register with the asterisk server.
06:23.17mrgobyso, it could be a registration issue then ?
06:23.30monsieurdoes anyone know of a script which automatically recompiles kernel and all steps?
06:23.31mrgobysorry, actually i'm not the guy you wanna talk to ... i'm a sip-er
06:24.15davidhindmarshmost likely, does the endpoint register with asterisk or does there need to be something else involved.
06:24.39jsharpasterisk is not a gatekeeper.  You can'
06:24.50jsharpt register against asterisk with h323.
06:25.13mrgobymonsieur: do you have those dynextendb files handy???  otto's server is offline
06:25.41davidhindmarshIf you can't register how does it work
06:26.20davidhindmarshDo i need to install a gatekeeper
06:26.31mrgobyggk
06:26.41mrgobygnu gate keeper is one i've heard is good
06:26.54jsharpyes.  Gnugk
06:27.34davidhindmarshOk, so endpoint registers with GNUGK, how does it talk to asterisk endpoints who are on SIP
06:27.53jsharpIt can talk directly to SIP clients.
06:27.57jsharpthrough chan_sip.
06:28.33davidhindmarshhow does gnugk now where sip users are.
06:28.42*** join/#asterisk Lafinion (~Lafinion@test.incracow.com)
06:28.58mrgobyset extensions for them in *
06:29.10mrgobyis one way i would imagine
06:29.36davidhindmarshwhat would such a config line look like
06:29.56monsieurmrgob: it was a while ago. I'll ask one of my staff
06:29.58Mikehow long did iaxtel closed?
06:30.12mrgobymonsieur: thnx
06:30.33mrgobyi have it on a server, but it is offline as well !!!
06:31.04danielqmonsieur: if you're using Debian, install the kernel-package deb
06:31.11mrgobydavidhindmarsh:  exten 34 => Dial(SIP/user@host)
06:31.28mrgobysorry taht is the wrong syntax
06:31.30mrgobyjust a sec
06:31.53monsieurdanielq: thx.
06:32.01monsieuron rh8
06:32.03monsieur:-(
06:32.27mrgobyexten => 23,1,Dial(SIP/user@host)
06:32.45mrgobythat would dial directly to the sip user when you call 23@asteriskHost
06:32.49davidhindmarshwhat about the h323 endpoints.
06:33.19mrgobydavidhindmarsh: what do you mean?
06:33.42mrgobyjust call that extension on asterisk with the h323 UA
06:33.57mrgobyi haven't done this, but from my understanding that should work
06:34.07jsharpasterisk itself can register with the gatekeeper, then you have access to all of the aliases that are registered with the GK.
06:34.08mrgobyi'm kinda pulling this outta my ass though
06:34.10mrgoby:-D
06:34.21mrgobythankyou jsharp
06:34.26mrgobyi'll shut up now
06:34.39mrgobyno advice > bad advice
06:34.56mrgobylisten to jsharp davidh
06:35.04jsharpSo if you have a netmeeting client with a username of "crackbaby" registered to your GK and asterisk is registered to the GK as well, then you can set up an extension like "exten => 23,1,Dial(H323/crackbaby)"
06:35.35jsharpand extension 23 will dial the netmeeting client.
06:36.27davidhindmarshHow does the gk know about the asterisk extensions, so h323 can dial the sip ep
06:38.18jsharpyour * server can register multiple aliases with the GK.
06:38.33jsharpYou can register an alias for each extension.
06:39.22davidhindmarshdo you use sip@asterisk type aliases
06:39.30mrgobyjsharp     does the j stand for jonas by any chance ???
06:39.51jsharpnope.  james.
06:40.15mrgoby:-)
06:41.43*** join/#asterisk diana (~diana@home-25022.b.astral.ro)
06:41.45dianaehlo
06:42.09jsharpI think you'd need to register the extension numbers.
06:42.11mrgobyhowdy diana
06:42.29jsharpI've not done a hybrid network before...so I'm just working on what I know.
06:42.43*** join/#asterisk decode (~decode@ip-wv-68-117-144-207.charterwv.net)
06:42.56l-fy_jsharp > what do you mean by hibrid?
06:43.02l-fy_hybrid?
06:43.02decodeanyone happen to know where i can perhaps obtain a database mapping NXX-NPA's to zip codes? :)
06:43.05l-fy_damn english
06:43.33davidhindmarshcan i run the gnugk on the same machine
06:43.50mrgobyhybrid H323  <---> SIP
06:43.53l-fy_davidhindmarsh > yes
06:43.58mrgoby!info hybrid
06:44.02l-fy_mrgoby > use a proxy
06:44.05decodeor hell, a complete list of NXX+NPA's preferably including carrier's for mobile NPA's?
06:44.10decodeand service type designations?
06:44.26decodei just obtained a zip+4 database.. :)
06:44.30mrgoby!info proxy
06:44.35mrgobydoh!
06:45.04mrgobyi do know what that is ... just checking if AiNFO does
06:45.21l-fy_who's AiNFO
06:45.25l-fy_!info hybrid
06:45.30l-fy_k
06:45.42davidhindmarshcan i run the gnugk on the same machine as asterisk
06:45.47l-fy_i hate openh323
06:45.52jsharpTru http://www.nanpa.com/nanp1/AllCodes.zip
06:45.55l-fy_takes way to much to do anything
06:46.21jsharpdavidhindmarsh: No.  Both the h323 channel of * and gnugk try to bind to the same tcp/udp ports.
06:46.33decodehttp://docs.nanpa.com/npa/allnpas.zip that's the NPA's.. hmm
06:46.50decodenow what about exchanges?
06:47.01l-fy_jsharp > you can bind on another port
06:47.16l-fy_anyway gnugk want to bind himself on 1719
06:47.28l-fy_and asterisk on 1720
06:47.50jsharpyes, you can change port bindings...then you have to change ports on everything that needs to talk to the application you changed ports on.
06:48.10l-fy_jsharp > anuway gnugk is on 1719
06:48.14l-fy_les me take a look
06:48.55l-fy_sorry
06:48.57l-fy_on 1721
06:49.01l-fy_i was close
06:49.08l-fy_CallSignalPort=1721
06:50.11cmanwhy ami not getting callerid?
06:50.16cmanthru * box?
06:51.03l-fy_cman > what driver
06:51.03l-fy_?
06:51.05denonso .. someoen tell me .. what's a good "poor man's" packet shaper? open source stuff?
06:52.08phsdshftdenon: um.. your just wanting to do rate-shaping while using linux?
06:52.10l-fy_hi denon
06:52.17jsharpdenon: I'm using OpenBSD's pf.
06:52.38denonhey l-fy
06:52.48cmandriver? meaning?
06:52.55cmanFXO
06:52.57jsharpA combination of priority queing and cbq queing.
06:53.01denonthink I'd stick to FreeBSD, now that it can do true bridging a bit better
06:53.04jsharpI wish I could spell.
06:53.12denonI like OpenBSD for that kinda stuff, but FreeBSD is just so damn smooth to use
06:53.35l-fy_sorry to putting such a stupid question
06:53.42l-fy_but why not linux?
06:53.52l-fy_you have a bandwidth more the 2 Mb?
06:53.52denonI'm not real fond of linux ..
06:54.04l-fy_what's fond?
06:54.13denonwell .. yes, we do .. but I'd not trust our core network to a x86 router
06:54.21denonl-fy: I don't like it.
06:54.24phsdshftbkw: are you there?
06:54.33l-fy_denon > ok, i can understand this
06:54.34l-fy_btw
06:54.36denonwe run juniper and packeteer for core applications
06:54.43phsdshftheh packeteer
06:54.44clive-denon I belive lartc.org is good
06:54.48l-fy_i do have my core network on linux
06:55.06denonl-fy_: multiple OC12s and an OC192 sonet ring? :)
06:55.13l-fy_denon > you need much more then i need
06:55.22l-fy_denon > this why i have ask you
06:55.24phsdshftwhy use packeteer
06:55.30phsdshftwhy not use rate-shaping on the juniper?
06:55.38denonwhat I need is a poor man's .. somethin I can toss at remote offices to prioritize voice, etc
06:55.43l-fy_i know the differences between bsd and  linux, and how is junos
06:56.25decodeblah
06:56.35jsharpOpenBSD/pf on a sparc machine.
06:56.41decodebbl, going to try and find more databases >:)
06:57.19phsdshftdenon: Why not use rate-shaping on the juniper?
06:57.27denonphsdshft: these are at remote offices ..
06:57.32denonrunning crappy pipes like cable and dsl
06:57.38denoncheap, dinky little networks
06:57.46denonwhere we cant justify more than a cheapo router and stuff
06:57.50phsdshftYou can specify source/remote ips to rate-shaping
06:57.59phsdshfton your core network (if you so desired)
06:58.02denonnot when I dont control the .. yeah ..
06:58.06phsdshftor your gateway router
06:58.06denonsee .. its not offices on our netowrk
06:58.11phsdshftah well
06:58.28denonthey're crappy places in timbuktu that buy local bw and vpn in
06:58.49phsdshftoh you use packeteer on their sites
06:59.19denonpacketeer on medium sized stuff
06:59.25denonnothing on smaller stuff.. but I need to
06:59.30Mikeanyone has 1800 numbers working with asterisk + GS with FWD sipphone.com or iaxtel??? i just need a little hand
06:59.35denonseeing as though dsl/cable/etc is pretty damn unrepdictable
06:59.36phsdshftlol coo
06:59.39phsdshftanyway
06:59.49denonthis lartc.org  looks kinda cool
07:00.25decodeMike exten => _1800,1,Dial(SIP/${EXTEN}@fwd) or such should work, no?
07:00.37Mikedecode: its hanging up my call
07:00.44Mikedecode: i have about 15 examples
07:01.12decodehmm any way to convert from an access db to a mysql db? :)
07:01.24jsharpexten => _1NXXNXXXXXX,Dial(SIP/${EXTEN}@fwd)
07:01.33decodebesides warez'ing office, installing it, and exporting to csv? :)
07:01.34*** join/#asterisk bwz (~w_w_zhang@66-215-24-240.mpk-eres.charterpipeline.net)
07:01.44jsharpdecode:  Yah, lemme find the software I just had to use.
07:02.04decodecool
07:02.31decodei'm slowly compiling a ton of different db's related to geographic info into one heh
07:02.33jsharpTook me forever to find it originally.
07:02.33bwzanyone knows what this "sip show peers" out mean with (D) in the middle?
07:02.37bwz75561331111/755  210.22.24.65    (D)  255.255.255.255  54328    Unmonitored
07:02.49decodephone exchanges, street level maps, etc
07:02.49decode:)
07:03.30Mikehttp://mike.calle69.net/exten.txt
07:03.33Mikenone of those work
07:03.48jsharpdecode: I think this is the one I used http://www.fonlow.com/zijianhuang/dbconverter/
07:04.21Kumbangguys, how can i make sip client connect to * without password
07:05.05l-fy_Kumbang > you just setup the user without a secret
07:05.14l-fy_ok guys
07:05.30l-fy_is there anyone here who can provide me some DS2155 chips?
07:06.07Kumbanganyone of you guys using pingtel sipphone? kinda hard to connect it
07:06.08jsharpBuilding a T1 card?
07:06.22l-fy_jsharp > not really
07:06.28decodejsharp that appears to a demo that only does 5 records
07:06.36jsharpDammit.
07:06.42decodeheh, don't worry about it
07:06.49decodei'll get my friend to export it to csv tomorrow
07:07.01jrollysonhow hard would that be to build?
07:07.02jsharpMy wife wiped the program I used off her laptop.
07:07.35jsharpShouldn't be that hard to build.  Supporting it on a PCI bus would be tricky.
07:07.51Mike<PROTECTED>
07:07.51Mike<PROTECTED>
07:07.51Mike<PROTECTED>
07:07.51Mike<PROTECTED>
07:07.51Mike<PROTECTED>
07:07.54Mikeit gets hangup
07:09.41decodeextra dry ginger ale + vodka is good :)
07:10.11Mikemaybe i need some speacial code
07:10.16Mikefor 1800 numbers
07:10.17Mike???
07:10.21jsharpMike:  Do you have canreinvite=no in your fwd entry in sip.conf?
07:10.31Mikejsharp: let me see
07:10.47jsharpAnd in the entry for your GS phone?
07:11.02Mikein the entry of my GS i have it on fwd part i dont
07:11.08*** join/#asterisk miller7- (~none@adsl49-static-gw1.access.acn.gr)
07:11.22jsharpAnd where's the extra * in the dialed number coming from?
07:11.27jsharp"SIP/*18006927753@f
07:11.29miller7-g'day people
07:11.31jsharpspecifically.
07:11.41*** join/#asterisk cocoy (strange123@ipdial-171-178.tri-isys.com)
07:11.50Mikethe * is needed by fwd to dial 1800 numbers
07:11.55jsharpOh.
07:11.57decodeit is?
07:12.01Mikeyes it its
07:12.17decodei never used it w/ xlite
07:12.36cocoyneed help on x-lite. "484 Address incomplete"
07:12.47Mikeok i added the canretrive no
07:12.51Mikeand still its getting hangup
07:12.54Mikewhen it answer
07:12.59decodedid you reload?
07:13.04jsharpnot canretrive.  canreinvite.
07:13.06Mikei even stop asterisk
07:13.07Mike:)
07:13.11Mikeand start it back
07:13.13decodecocoy looks like you're trying to call an invalid extension
07:13.20decodei never got x-lite working well w/ fwd tho
07:13.26Mikeusername=77443
07:13.26Mikehost=fwd.pulver.com
07:13.26Mikecanreinvite=no
07:13.26Mikereinvite=no
07:13.34cocoybut it's there on my extensions.conf
07:13.47jsharpHrm.
07:14.26cocoyi noticed i have dynextendb.conf , do i have to change here?
07:14.28Mikejsharp: could it be a codec thing?
07:14.46jsharpdunno.  What codecs are you using?
07:15.19Mikewell
07:15.27Mikedisallow=all
07:15.28Mikeallow=ulaw
07:15.28Mikeallow=alaw
07:15.28Mikeallow=g729
07:15.28Mikeallow=gsm
07:15.28Mikeallow=ilbc
07:15.30Mikeallow=speex
07:15.33Mikeallow=lpc10
07:15.41denonbetter question, what codecs are you not using?
07:15.55Mike:P
07:15.57mrgobycocoy:  do you have the dynextendb package handy ???  otto's site is down
07:16.05mrgobyi was hoping to install it tonight
07:16.35cocoysomeone already installed it on my linux box.. i don't how to install it and where to get it
07:16.57mrgobyk, thnx anyway
07:18.16cocoymrgoby... what can i do to solve my problem?
07:18.24cocoyneed help on x-lite. "484 Address incomplete"
07:18.41cocoycan u give me example extension
07:19.02cmanx-liteecho problem
07:19.53mrgobycocoy what is the problem?
07:20.02cocoyanyone, "484 address incomplete"
07:20.28cocoymrgoby- error in xlite
07:20.29mrgobywhat is generating that error?
07:20.31mrgobyoh
07:20.34decodebbl, going to have to sleep for a bit..
07:20.52mrgobyi don't use xlite but sounds like you have bad syntax on your sip address
07:21.09cocoywhat to u use?
07:21.31mrgobytry variations of sip:user@host , user@host , <sip:user@host>
07:21.46mrgobyalso
07:21.55decodesip:user@domain works well enough
07:22.03mrgobyi noticed that sjphone doesn't like domain names....
07:22.05cocoywhat do i put on domain?
07:22.21mrgobyso....   maybe try to specify the IP number
07:22.28espenz<PROTECTED>
07:22.32espenzhm, its playing
07:22.38espenzbut i get a realy wierd sound
07:22.39espenzwhy?
07:22.39espenz:P
07:23.00cocoymrgoby- i tried also specifying the IP but not happened
07:23.24mrgobysorry cocoy  ...  dunno what to tell you ....  try sjphone
07:23.38mrgobyi've heard lots of people complain about xlite
07:23.38cmando we really need to have register=>line in the onf iles?
07:23.40*** part/#asterisk decode (~decode@ip-wv-68-117-144-207.charterwv.net)
07:23.53mrgobywhat OS cocoy ??  windows?
07:23.57cocoyok, i have my sjphone.. wait
07:24.41cmanwhich is he best sip sotphone?
07:24.57mrgobyi can give you my experience
07:25.06mrgobybut i think all are bad at certain things
07:25.11mrgobythat is
07:25.41mrgobythere are many permutations of interactions that don't work well because of the differences between each implementation
07:26.00mrgobyya know?
07:26.26mrgobyi use linphone or sip-communicator for linux
07:26.32cocoymrgoby- i have my sjphone now, which part of config i need to put my * ip on sjphone?
07:26.55mrgobyand sjphone, sip-communicator, or ubiquity UA for windows
07:27.16mrgobyrephrase that cocoy
07:27.17cmanwhere to download sjphone?
07:27.26mrgobysjlabs.com i think
07:27.34mrgobygoogle sjphone
07:27.37cocoywhat do i need to confgiure on sjphone?
07:27.59mrgobythe configuration of sjphone is non-intuitive for me
07:28.00mrgobybut
07:28.33mrgobyyou need to go into sip tab under preferences or options and set your name, NAT info
07:28.52mrgobythen put the sip address you are trying to dial on the text bar on the UA
07:29.07mrgobybut this is all in the help on the phone :-)
07:32.16*** join/#asterisk renv (strange123@ipdial-171-178.tri-isys.com)
07:32.27*** join/#asterisk Inv_Arp (junya@fiudial2-84.fiu.edu)
07:32.27renvwhat's a SIP URL?
07:32.30*** join/#asterisk FryGuy- (~fryguy@c-24-2-50-122.client.comcast.net)
07:32.38mrgoby!info sip
07:32.39AiNFOsip - Session Initiation Protocol - SIP is the IETF protocol for VOIP. It can also support video and other services (see !info sip-and-nat)
07:32.53cocoywhat's a SIP URL?
07:33.04*** join/#asterisk usam (usam@2001:730:11:29:0:0:0:281)
07:33.08mrgoby!info sip-and-nat
07:33.09AiNFOsip-and-nat - there are various issues with SIP and NAT traversal. The following paper by DeltaThree provides a comprehensive discussion http://www.zebraroaming.com/stuff/SIP-and-NAT-Traversal.pdf
07:34.09mrgobyg'night gentlefolks
07:34.11renv!sjphone
07:34.22renv!info sjphone
07:34.30mrgobyduddn't work... tried it earlier
07:34.33mrgoby:-D
07:34.43renv!info sip.conf
07:35.10renvanyone who knows how to set sphone?
07:35.17renvsjphone
07:36.27renvanyone?
07:36.33renvanyone who knows how to set sjphone?
07:37.36*** join/#asterisk oriontkn (~oriontkn@pool0098.cvx6-bradley.dialup.earthlink.net)
07:37.54oriontknwowzers
07:39.33oriontkndoes anybody know where I can find some updated info on asterisk and dialogic?
07:41.06izooriontkn : mailing list but there isnt much support for it
07:44.48espenzi get a wierd sound when i play mp3s with asterisk, why could that be?
07:44.50*** join/#asterisk edguy3 (~edguy@host-24-225-213-50.patmedia.net)
07:44.51oriontknyeah, that was the impression that I got from looking at the list archives, but some of the posts were older so I thought maybe it had come a ways since then... I have a large dialogic investment
07:45.27cmansip is used to call sip phones only?
07:45.29oriontknalthough the digium t1 cards look kind of attractive
07:45.33cmansjphone i mean
07:45.50*** join/#asterisk Marlow (~marlow@3ffe:200:1:155:0:0:0:2)
07:46.17*** join/#asterisk cocoy (strange123@ipdial-172-68.tri-isys.com)
07:46.47cocoyhelp on configuring sjphone?
07:47.07cmanclick the options tab
07:47.33*** join/#asterisk Marlow (~marlow@ragnarok.marlow.dk)
07:47.36cocoythen
07:48.17cocoycman
07:48.37cmansip
07:48.53cocoythen
07:48.57cmantick use local outbound
07:49.05cocoyok
07:49.23cocoythen
07:49.48cocoyUDP or TCP?
07:49.58cmanudp
07:50.22cocoythen after?
07:50.27cmani don't know but this phone seems to dial ip addresses only
07:50.32cmanclick ok
07:50.57cmanso if u have another sip phone .. dial the ip of that phone.. it should ring
07:51.12cmandid it register with *?
07:51.28cocoyerror says "invalid caller ID"
07:51.42cocoywhat do i need on callerid
07:51.48cmanin caller id pu like tis
07:51.48cmansip:12345@192.168.0.5
07:51.57cmansip:username@ip
07:52.01cocoyok
07:52.04cmansip: user
07:52.38cmanregistered?
07:52.42cocoyyes
07:52.46cocoywhat number can i dial?
07:53.00cmangive ip...
07:53.22cmansuh as 192.168.0.11 and Dial
07:53.23cocoyip of what?
07:53.30cocoyah ok
07:53.32cmansip of another pone
07:54.04cocoyhow abt dialing an extension? like a voice mail
07:54.33cmandon't kno... figure that out.. read the docs.. i tried but ouln't dial other no.. like loal nos..
07:54.36cmanlocalnos.
07:54.39cman66
07:54.39cman77
07:54.44cmanetc
07:55.06cmanit just seems to dial ips
07:56.21cocoyok
07:56.33espenzcould anyone try to call "217.118.59.34" with a voip client?
07:56.47Marlowsip or h.323 ?
07:56.48cmanu can also dial as sip:66@192.168.0.11
07:57.00cmantis is useless
07:57.12espenzMarlow: try both
07:57.13cmanx-lite is better hat sjphone
07:57.13Marlowcman : seems so ..
07:57.25Marlowespenz : eheh .. did you set both up or what ?
07:57.34espenzno, i just dont know what it is
07:57.35espenz:P
07:57.44Marlowespenz : what application ?
07:57.57cmani am not able to get caller id..
07:57.58cocoyok, i have xlite wait
07:58.05cmanwha the heck!
07:58.21espenzMarlow: asterisk
07:58.21Marlowcocoy : RTFM
07:58.42Marlowespenz : and you configured it where ? sip.conf ?
07:59.03cocoywhat do i need to fill up on x-lite?
07:59.10cocoyah ok!
07:59.17cocoyi got u..
07:59.34Marlowespenz : definatly not H.323 ...
07:59.38cmangot who?
07:59.40Marlowespenz : no phone running ..
08:00.18espenzMarlow: try now, sure?
08:00.58espenzdo you have one up, that i can test?
08:03.44discordiamorning gyus
08:04.42Marlowdiscordia : morning ..
08:05.01espenzMarlow?
08:05.11discordiaout of the bed ... in front of coffee ...
08:05.18Marlowespenz : try dial 612@fwd.pulver.com
08:05.26Marlowespenz : that's a test-line ..
08:05.31Marlowespenz : sip, though ..
08:05.43Marlowdiscordia : hopefully real coffee ..
08:06.03Marlowdiscordia : the stuff they serve here can only be used to clean furniture with ..
08:06.18discordia:)
08:06.23discordiano chance
08:06.41discordiaguatemalean it is ... really great!
08:06.50Marlowdiscordia : it's allready that bad, that i've thought of taking my own coffeemachine to work with me ..
08:06.57discordia:)
08:07.09discordiaa man has to do what a man has to do ...
08:07.09Marlowdiscordia : would solve the problem ..
08:07.18Marlowdiscordia : yep ..
08:07.26discordiaand would be cheaper with the time
08:07.46Marlowdiscordia : nah .. coffee is free here ... or .. no .. it's not coffee ..
08:07.54Marlowdiscordia : but what they call coffee ..
08:07.59discordialol
08:08.09*** join/#asterisk JerJer[ghost] (~NunYoBizN@pppoe-66-112-51-206-rb2.grp.centurytel.net)
08:08.18MarlowJerJer[ghost]: morn ..
08:11.36JerJeryep, its certianly morning
08:12.22espenzMarlow: it wont answer
08:12.31cmananyone from xlite?
08:13.47*** join/#asterisk _mwoodj- (~a@user-24-214-189-40.knology.net)
08:16.44voidptrmorning
08:16.53espenzMarlow
08:16.59espenztake phone?
08:16.59miller7-morn voidptr
08:17.01espenz;p
08:17.06espenzlol
08:17.28cmani can't get caller id thru * for incoming calls.. any one>
08:17.37espenzthey have been carried away with monkeys
08:20.21espenzany voip client for windows to recomend?
08:21.29JerJerespenz:  nope
08:21.43JerJercman: you set the caller*id on the source channel or via the exen line
08:22.48discordiaespenz: if you have one please tell me, im searching for my own for good clients ...
08:23.34discordiai think on linux side it'll be * as client, that is possible right? with register in sip.conf im on the way  or not ??
08:30.22blitzrageespenz: x-lite
08:33.10espenzit wont work
08:33.33*** join/#asterisk bwz (~w_w_zhang@66-215-24-240.mpk-eres.charterpipeline.net)
08:34.25bwzanyone know how to config extension.conf to make sip-->zaptel-->outbound to PSTN?
08:35.10*** join/#asterisk Levch (~Levch@217.116.160.6)
08:35.16blitzrageespenz: what do you mean?
08:35.25espenzblitzrage: i wont ring?
08:35.41blitzrageummmm... which way, what are you calling, etc..?
08:36.04espenzim calling the ip 192.246.69.223?
08:36.34LevchHow does IAX2 trunk work?!
08:40.53discordiabkw_: sry, not familiar with zaptel devices ... i can tell you how to get sip-->outbound to PSTN
08:49.22*** join/#asterisk Exomorph_ (Greg@134-9.bvcompuworks.com)
08:54.25blitzrageI'm trying to create a script which will log me into a remote CVS tree.  If I do a cvs login, it asks for a password.  How can I either: pass the password for the prompt, or: pass the password along with the cvs login command?
08:56.12knight-anyone use SmoothWall?
08:57.33discordianope
08:58.17*** join/#asterisk olivier (~olivier@APuteaux-105-2-1-45.w217-128.abo.wanadoo.fr)
09:07.08cman-->
09:16.11discordiaaaarrrrrrgggggl --->
09:16.13discordiaNOTICE[10251]: File app_capiECT.c, Line 125 (capiECT_exec): call was answered
09:16.13discordia<PROTECTED>
09:16.13discordiaSegmentation fault
09:16.58Inv_Arpblitzrage: expect
09:18.02blitzragehuh?
09:19.00Inv_Arpblitzrage: never used it but the "expect" command can be used to pass passowrds etc...
09:19.12blitzrageoh right.. thanks
09:19.14RoyKdiscordia: start with -g and run a backtrace on the core file
09:19.20blitzragehehe.. I wasnt's ure what was going on there for a second :)
09:20.54blitzragehrm... still can't seem to get it
09:22.51*** join/#asterisk Bonbon (~bonbon@62.3.220.66)
09:23.07*** join/#asterisk sobol__ (~sobol@router-1.szczecin.tpnet.pl)
09:23.18discordiaRoyK: i will ...
09:23.21Bonbonsomeone told me about a kernel utility that would compile my kernel and everything required in one step. Does anyone know what I'm talking about?
09:24.35blitzrageBonbon: if you create a bash script, maybe
09:24.53voidptrquick help: modversions.h ... when its missing in usr/include/linux... what is the problem? (missing kernel headers?)
09:25.05Inv_ArpBonbon: its called ik
09:25.16Inv_Arpfreshmeat.net
09:25.33voidptri would guess zaptel needs module headers from the running kernel
09:25.37voidptrand not from libc
09:25.43voidptrbut could be wrong
09:25.52voidptron the other hand, its usespace
09:28.14Inv_Arpvoidptr: what distor?
09:28.19Inv_Arperr distro
09:29.13*** join/#asterisk tristan2 (~tristan@213.239.44.133)
09:29.17voidptrgentoo
09:29.33Bonbonthanks Inv
09:29.41RoyKgrr. security.debian.org is down :(
09:29.43Bonbonwho uses the jitterbuffer in iax.conf?
09:30.34Inv_Arpvoidptr: seems like ya need full kernel src for yer distro
09:30.36*** join/#asterisk fyman (~fyman@CPE-138-130-18-16.nsw.bigpond.net.au)
09:31.02voidptroah, thats so crap
09:31.18af_moin all
09:31.18manyroyk, you should read debian-user.
09:32.22many(and yes, that warning is overdrawn, however it contains true bits)
09:34.18RoyKoh well
09:34.42blitzrage<PROTECTED>
09:34.48*** join/#asterisk Muckl (johannes@pD954CD6F.dip.t-dialin.net)
09:35.23RoyKhttp://www.bangedup.com/Current/MikeysOntherun35.jpg
09:36.39*** join/#asterisk Marlow (~marlow@ragnarok.marlow.dk)
09:37.12Marlowarghh ..
09:38.20*** join/#asterisk mbranca (~matteo@213.140.14.155)
09:38.29mbrancamorning all
09:38.51[Sim]hmm I'm having an odd issue when I try to link an inbound IAX call via IAX to DIAX or iaxComm
09:39.00blitzragembranca: you do any bash scripting?
09:39.00[Sim]seem to be havind trouble agreeing on codecs
09:39.24mbrancambranca, very little. nothing fancy
09:39.34[Sim]anyone else seen this ?
09:39.41mbrancaehm, I meant blitzrage, :)
09:39.43blitzragehehe
09:39.56blitzragedamn... I'm trying to login to CVS with a password from within the script
09:39.57voidptrmorn mbranca, sim
09:40.30mbrancamorning voidptr
09:42.16[Sim]morning :)
09:42.48af_ciao mbranca
09:43.02mbrancaciao af_
09:43.46mbrancablitzrage, automagically?
09:43.52blitzrageyah
09:44.04*** join/#asterisk Unmanaged (~unmanaged@lebanon-24-159-24-23.midtn.chartertn.net)
09:44.31blitzrageif I do a cvs login, then it asks for a password.  I want to pass it automatically from within the script
09:44.45mbrancablitzrage, just use the .cvspass file in the home dir of the user you're running the script as
09:45.01mbranca[root@astro root]# cat .cvspass
09:45.14mbranca<PROTECTED>
09:45.17mbranca:)
09:45.44blitzrageAy=0=h<Z   <<    is this part the password?
09:45.53mbrancais the enc password
09:46.00mbrancafor digium, is anoncvs
09:46.01mbranca:)
09:46.22mbranca~seen wasim
09:46.24wasim is currently on #asterisk.  Has said a total of 92 messages.  Is idling for 17h 26m 58s
09:47.14*** join/#asterisk blitzrage (~blitzrage@dsl-145.sarnia.xcelco.on.ca)
09:48.37mbrancablitzrage, look at that:
09:48.50mbranca[root@astro root]# cat update.sh
09:48.50mbranca#!/bin/sh
09:48.50mbrancaTMPDIR="/tmp"
09:48.50mbrancaCVS="/usr/bin/cvs"
09:48.50mbrancaexport CVS_RSH=ssh
09:48.50mbrancaDATA=`/bin/date`
09:48.54mbrancaDATA2=`/bin/date +"%a_%b_%d_%Y"`
09:48.56mbranca<PROTECTED>
09:48.58mbrancacd $TMPDIR
09:49.00mbrancamkdir cvs-asterisk
09:49.02mbrancacd cvs-asterisk
09:49.04mbranca$CVS -d:pserver:anoncvs:anoncvs@cvs.digium.com:/usr/cvsroot co -P zaptel zapata libpri asterisk gastman asterisk-addons
09:49.07mbranca$CVS -d:pserver:anoncvs:anoncvs@cvs.digium.com:/usr/cvsroot co -P nbs libiax gnophone
09:49.09mbranca$CVS -d:pserver:anoncvs:anoncvs@cvs.digium.com:/usr/cvsroot co -P g723.1 g723.1b$CVS -d:pserver:anoncvs:anoncvs@cvs.digium.com:/usr/cvsroot co -P astconfig libiax2 ztphoned libr2
09:49.15mbrancathis works with the .cvspass file
09:49.19mbrancaall automaagically
09:49.22mbranca-a
09:50.10blitzragewicked!  I will look at that in the morning.
09:50.15blitzragenow I must get some food, and some sleep!
09:50.16blitzragenight all
09:56.09*** join/#asterisk Nix (~Nix@195.174.60.197)
10:06.59*** join/#asterisk Insy (~ask@e146244.upc-e.chello.nl)
10:07.12Insyey guys!
10:08.02Insyquestion: Is it possible to use SIP with a SSH tunnel?
10:08.30l-fy_Insy > why not?
10:08.32cmanwhere can i find the country codes for zaptel... as us, uk,jp, etc
10:08.39voidptrno
10:08.43levonmorning every1
10:08.50mbrancaciao levon
10:08.51Nixgunaydin levon
10:08.53discordiamorning levon
10:09.11Insyl-fy_: can ya give me a hint cause can't seem to get it to work!
10:09.14RoyKmorning
10:09.24levonciao mbranca
10:09.25levonhi discordia
10:09.30levonmerhaba nix, canim, naber....
10:09.35levongod morgen RoyK
10:09.45levonl-fy_, diana, my secret love. good morning to you, too ;)
10:09.54discordiaoh this multiculture
10:09.58discordiai love it ...
10:10.00levontotally
10:10.05Nixiyilik kardes. sen?
10:10.08l-fy_hello levon
10:10.20levoniyilik, canim.
10:10.23l-fy_tunnel everything
10:10.38Nixoptum, tatlim :-)
10:10.39Insyl-fy_: with SSH?????
10:10.39levonNix, sen glLoadEntity`yi taniyormusun?
10:10.53*** join/#asterisk tristan2 (~tristan@213.239.44.133)
10:10.53Insyl-fy_: I'm using putty
10:11.14levonsxpert_work, foood :(
10:11.16Insyl-fy_: can specify the ports to forward.
10:11.21Nixhayir levon. malesef, tanimiyorum :-(
10:11.39RoyKlevon: god formiddag
10:11.47cmananyone from zaptel
10:11.55sxpert_worklevon, I had to get the crappy VCD, because the XSVCD doesn't come...
10:11.55RoyKfrom zaptel???
10:12.05RoyKcman: I'm from oslo... don't know where zaptel is
10:12.07levon"from" zaptel` ;)
10:12.21levonsxpert_work, sh****t.... :(
10:12.22cmanzaptel.. digium...whatever
10:12.33*** join/#asterisk cypromis (~michael@lisa.halo2.pl)
10:12.37cypromisaloha
10:12.37cmani need to find country codes such as us, jp, uk
10:12.43levonNix, glLE ankara`dan geliyor, burada bahsan bisimle konisiyor. tanistiricam sizi ;)
10:12.46cmani am having trouble with caller id
10:12.48cypromiscountry codes ?
10:12.55cmanye
10:12.56levongood morning cypromis
10:12.59cypromiscman: www.numberplan.org
10:13.30RoyKNix: how's istanbul today? quiet again? what did the politicians say?
10:13.47cmani need codes that are used in zaptel.conf files
10:13.49Nixlevon glLE??
10:14.11NixRoyK: well I had trouble getting to my office as I had to divert around the closed road in front of HSBC
10:14.50mbrancaa big bank
10:14.55levonNix: glLE= glLoadEntity (short form)
10:14.56levon;)
10:15.22levonRoyK, the bank that got bombed.
10:16.13NixRoyK: HSBC is the second biggest bank in the world after Citibank
10:16.36Nixthe branch that got bombed is the one that I was about to use 30min later
10:17.22Nixlevon: I am not sure I understood your turkish.. glLE will come from ankara and you will introduce it to me??
10:19.06levonNix: I'm sorry, my turkish is *garbage*. I was trying to say that he is from ankara and is in here frequently too. I'd like to intodruce him to you
10:19.15levonWe can build a turkish-*-force, in here ;)
10:19.39Nixok. sounds good
10:19.42*** join/#asterisk kapejod (~kapejod@pD9E837BF.dip.t-dialin.net)
10:19.49levonmeep meep, mr. kapejod
10:19.50mbrancamorning kapejod
10:19.58discordiamorning kapejod
10:20.03discordiahehe
10:20.04kapejodmorn
10:20.10Nixthat makes me feel beter, I showed your sentence to my wife and she couldn't understand it properly either :-)
10:20.11cypromismoin kape
10:20.14levonkapejod, can you help me installing chan_capi?
10:20.16cypromis13.30 ostbahnhof
10:20.18levonkapejod, can you help me installing chan_capi?
10:20.27mbranca50€
10:20.29levonkapejod, can you help me installing chan_capi?
10:20.40mbranca100€
10:20.43levonrotfl
10:20.46kapejodcypromis: passt gut
10:20.50cypromisok
10:20.55kapejodlevon: what is capi??
10:21.08cypromisich komm entweder aus warszawa oder aus poznan nur mit nem notebook und nem flugticket bewaffnet
10:21.09cypromislol
10:21.09kapejodcypromis: let's met at friedrichstrasse then
10:21.10levon~tell kapejod about capi
10:21.15cypromisok
10:21.18levonpff
10:21.46levonbÀh, jetzt kann ich den nicht mal Àrgern...
10:22.05kapejodcypromis: so gegen 13:45 draussen vor der StäV, das ist auf der nordseite vom wasser
10:22.17cypromisok
10:22.22*** join/#asterisk detten2 (~john@213.219.141.57)
10:22.22cypromisich werds scho finden
10:22.25cypromissonst ruf ich dich an
10:22.26kapejodcypromis: StäV == Ständige Vertretung
10:22.47kapejodk
10:24.26sobol__cypromis: czesc
10:24.33levonczesc sobol__
10:24.45levonkruliczku ;)
10:24.53kapejodw00t w00t , my master server is up again (now with a C3) :)
10:25.02levonkapejod, no, you really did it?
10:25.21cypromishmmm
10:25.27cypromisshould I start in .pl, .ru ?
10:25.40kapejodlevon: yes, that was fun ... because the itx board is so small the pci riser and the case wouldnt really fit
10:25.44MarlowRoyK : de har problemer, hvad ?
10:25.50sxpert_worklevon, ok, it's up ;)
10:26.08RoyKMarlow: tyskere...
10:26.17kapejodlevon: and it's software raid1 with 2.4.22-preempt :)
10:26.27levonkapejod, you're evil ;)
10:26.35MarlowRoyK: ehehe ... og det der ligner ..
10:26.39levonsxpert_work, yeaaah! ;)
10:26.43mbrancaI have a diva server on a mini-itx board.... I had to buy lowprofile ram to make the 4.bri fit in the case
10:27.43kapejodvendor_id       : CentaurHauls
10:27.43kapejodcpu family      : 6
10:27.45kapejodmodel           : 9
10:27.45kapejodmodel name      : VIA Nehemiah
10:27.45kapejodstepping        : 1
10:27.45kapejodcpu MHz         : 999.544
10:27.47kapejod:)
10:27.53levonkapejod, show me da mips
10:28.30kapejodflags           : fpu de tsc msr mtrr pge cmov mmx fxsr sse
10:28.32kapejodbogomips        : 1992.2
10:29.13levonwow
10:29.23voidptrwell
10:29.27voidptrit are bogus mips
10:29.27levonexactly as much mips as the Athlon 1Gig...
10:29.28voidptrso
10:29.28cypromisbogus mips
10:29.28cypromis:)
10:29.29voidptr:)
10:29.30levon;)
10:29.47af_kapejod: which cpu has your mini-itx?
10:30.15voidptrwoah
10:30.19voidptrtotally gey
10:30.26kapejodaf_: C3 1ghz
10:30.40af_C3? what is that?
10:30.49af_it's a VIA stuff?
10:30.57kapejodvia c3
10:31.00af_I see
10:31.12af_I bought a biostar S200, with a celeron
10:31.25RoyKit's a sloow el cheapo low power cpu
10:35.56manyerr
10:36.08manydoes asterisk support regexes like [1-9] in the dialplan?
10:36.14mbrancamany, yep
10:36.23voidptrpom pom
10:36.26*** join/#asterisk Levch (~Levch@217.116.160.6)
10:36.33manymh.
10:36.52af_pum pum
10:36.58mbrancaexten => _[0-2]XXX.1,Dial(blah....)
10:37.00LevchCan I use different codecs to the same server with different extensions?
10:37.22mbrancamatches aany number starting with 0,1,2 , with at least 4 digits
10:37.24manyyea, didnt know.
10:37.30mbrancabut also 5,6,7....
10:38.32*** join/#asterisk nickkknight (~nick@217.206.219.198)
10:39.07nickkknightkapejod hello
10:40.58*** join/#asterisk Tekati (~captain@cpe-66-75-211-60.bak.rr.com)
10:41.00JerJerLevch: use a different user or peer
10:41.08*** join/#asterisk cypromis (~michael@lisa.halo2.pl)
10:43.34kapejodhi nickkknight
10:44.06RoyKkapejod: do you have any nice music for moh?
10:44.18jrollysonJerJer: I can't seem to find pricing on your site.
10:44.37nickkknightkapejod - problems with music on hold - any experience?
10:47.19kapejodnickkknight: you might need zaptel timing for MOH to work correctly
10:47.33*** join/#asterisk Tili (~Tili@202.133.67.121)
10:49.23*** join/#asterisk lele (~fsck@rivendell.windmill.it)
10:49.42levonehila lele
10:49.48mbrancaciao lele
10:50.35mbranca-s
10:50.37nickkknightkapejod how do I find out about zaptel timing?
10:51.02lelemorning levon, mbranca
10:53.59lelehas anybody ever had the need of a queue with priorities?
10:54.07kapejodnickkknight: if you have usb-uhci try ztdummy (in the zaptel dir), else try zaprtc (on my page)
10:57.30JerJerjrollyson; because there is no such information on there
10:58.07levon~tell nickkknight about zaptel-timing
10:58.11levon~tell nickkknight about zaprtc
10:59.00levonupdated ;)
10:59.08voidptrencryption, vpn, blablabla
10:59.17JerJer~tell levon about the mystories of the universe
10:59.24levonlol
10:59.38JerJer01[05:57] <jbot> i dunno what is 'the mystories of the universe'.
11:00.23Areshello all
11:01.01JerJermoo
11:01.04levonmeep meep
11:01.29AresI have problem when I want to access to asterisk remote option /usr/sbin/asterisk -vvvvgr           ERROR[1024]: File asterisk.c, Line 1346 (main): Unable to connect to remote asterisk
11:01.45JerJerthen asterisk is not running
11:01.45l-fy_bwahaha
11:01.53l-fy_sipstack from vovida is a big mess
11:01.56levonlol
11:02.04levonArea: -r is for remote console
11:02.18levonArea: if you haven't a running instance, it's pretty much useless ;)
11:02.19Aresnow if I do /usr/sbin/safe_asterisk
11:02.19Aresbash-2.05a$ Asterisk ended with exit status 1
11:02.19AresAsterisk died with code 1.  Aborting.
11:02.34levonjust start it up, without -r
11:02.38levonlike -vvvvvgc
11:02.39JerJersomething is broke
11:02.42Areslevon: I know
11:02.42JerJer-vvvgc
11:02.43levonyupp
11:03.00levonyou know? then do it ;)
11:03.19nickkknightkapejod that kernel you built me - did you include rtc?
11:04.32knight-why do you need RTC?
11:04.49levon~tell knight- about zaprtc
11:05.00knight-gotcha thanks
11:05.07levon;)
11:05.12knight-:)
11:05.19nickkknightin the readme is says (of zaprtc) it says make sure you don't have rtc compiled in
11:05.32Areslevon : sorry I m bit mess...
11:06.23AresWhen normally I should launch asterisk with safe-asterisk
11:07.19Aresand then when I want access to access to the cli console I have to run /usr/sbin/asterisk -vvvvgr
11:07.25Aresare for remote
11:07.25*** join/#asterisk JerJer[ghost] (~NunYoBizN@pppoe-66-112-51-206-rb2.grp.centurytel.net)
11:07.33Aresare for the remote
11:07.39Aresisn't it ?
11:07.43detten2try asterisk -vvvvvvvvvvvvgc
11:07.56detten2and see why safe_asterisk is not working
11:07.58Aresinstead of safe_asterisk
11:08.13Aresno safe asterisk seems to work
11:08.26Aresit's when I want to access to the console
11:08.31zoaasterisk -r
11:08.33cypromis~seen miller7
11:08.34miller7 <~none@adsl49-static-gw1.access.acn.gr> was last seen on IRC in channel #asterisk, 4d 3h 22m 17s ago, saying: 'good morning all'.
11:08.47cypromishmopft
11:09.05voidptrhum
11:09.07Ares<PROTECTED>
11:09.07AresERROR[1024]: File asterisk.c, Line 1346 (main): Unable to connect to remote asterisk
11:09.08voidptrthats borked
11:09.15voidptrmiller was here yesterday
11:09.22cypromisyeah I know
11:11.20nickkknightso in the zaptel directory - I can just run a make install?
11:12.17Bonbonhas anyone got a spanish number that they can sell me?
11:13.12voidptrok i'm going slightly nuts now
11:13.57Aresit seem that the pid files and the cli aren't created inside /var/run/asterisk/
11:15.24Bonbonno-one has spanish number --> sip service?
11:15.29voidptrinstead of module
11:15.35voidptrmodules are just borked by design
11:16.27Pj_yeah!
11:16.31Pj_modules are for sissies
11:16.52voidptraight!
11:16.52voidptr:P
11:16.55Pj_real men patch their kernels in memory
11:17.20voidptr:))
11:18.02AresBonbon: perhaps contact mediafusion if you want to buy
11:18.38l-fy_who have ever install vovida stack?
11:20.03kapejodhey l-fy_ :)
11:21.23*** join/#asterisk kd-93 (~chatzilla@pbd.urtc.ru)
11:21.44kd-93hello people
11:21.51*** join/#asterisk lichen (lichen@58.136.8.67.cfl.rr.com)
11:21.58l-fy_hi kapejod
11:22.35kd-93does * support outgoing h.323 channels, and what is the syntax? "h323/user@peer" or something else?
11:23.41Bonbondoes anyone know how to turn on echo canceller training mode? I think that this was implemented recently.
11:24.20zigmanecho "1" > /dev/hda1
11:24.22zigman:P
11:26.44JerJerBonbon: see zapata.conf.sample
11:26.46zigmanfor wha ?
11:26.48zigman;)
11:26.50Pj_dd if=/dev/echo of=/dev/phone1 count=much
11:27.04zigmanno dev/null
11:27.15zigmanhe doesn't want the echo :P
11:27.23Pj_For training :P
11:27.27zigmanlol
11:27.55many(debian users: http://cert.uni-stuttgart.de/files/fw/debian-security-20031121.txt)
11:28.07mbrancaDial(/dev/null,inf)
11:29.47*** part/#asterisk Nix (~Nix@195.174.60.197)
11:35.26*** join/#asterisk kn0rki (~kn0rki@213.168.83.216)
11:35.30discordiakapejod: wanna have my newest update to app_capiECT? it should be in the chan_capi package cause of bugfixes + features :)
11:36.53BonbonArea: what's the url?
11:38.26kapejoddiscordia: just mail it :)
11:38.34discordiak
11:40.33BonbonAres: what is the url for that?
11:40.49*** join/#asterisk cfo (~cfo@194.19.190.217)
11:41.07kapejodwtf is an ISDM card? ;)
11:42.05mbrancaintegrated services digital moooo
11:42.18levon;)
11:42.30mbrancaa new way to speak to the cows
11:42.31kapejodintegrates services digital mettwurst ;)
11:42.50Pj_a typo.
11:42.58Pj_:D
11:44.11discordiahmmmmmm mettwurst ;)
11:44.53discordialol
11:45.20*** join/#asterisk pino (foobar@host115-28.pool21345.interbusiness.it)
11:45.20levon~time
11:45.23somebody said time was 5:21
11:47.57Bonbonneed a spanish local number to sip.
11:49.56levonm00, ]data[
11:52.46]data[lo
11:54.32JerJerhowdie
11:55.26cypromisunig *!*mspain@hiostu.aim.hsbr
11:55.32RoyKgrr. when using nagios to check for HTTP on grandstream phones, HTTP dies after a couple of days............
11:55.40cypromisbitchx suxx
11:55.52voidptrunig voidptr
11:55.53voidptr:)
11:56.00voidptrtog flood_p
11:56.08cypromistry that on a console
11:56.34RoyKcypromis: use telnet
11:56.43cypromisroyk ????
11:57.21cypromisok found the unignore switch
11:57.22cypromislol
11:57.26RoyKtelnet irc.freenode.net 6667
11:57.53cypromisRoyK: that suxx much more than bitchx
11:57.56RoyK:)
11:58.04RoyKthen stop complaining :)
11:59.18*** join/#asterisk stipe (~stipe@host107-64.pool80180.interbusiness.it)
12:00.07JerJerok the sun is offically rising... us vampires must head to our coffins
12:01.57*** join/#asterisk stipe81 (~stipe@host107-64.pool80180.interbusiness.it)
12:03.12Pj_When I receive a call there's 1-2 seconds with a ring and then it's a different ring and the phones actually starts to ring.. is it normal ? how do I reduce / cancel it ?
12:04.11Pj_(on a X100P)
12:10.46mbrancaI think that's normal, since * must wait for cid on the x100p (the cid is sent between first and 2nd ring)
12:10.54mbrancasame behaviour here
12:11.57mbrancaok. going home. see ya later
12:12.00*** join/#asterisk dnc (~duncan@213.244.224.118)
12:12.34cypromishi duncan :)
12:12.43kd-93does anyone have any experisnce with * and h323 ?
12:12.49cypromisyep
12:13.02cypromisit's somewhere between ok and a major pain in the butt
12:13.05cypromis:)
12:14.15kd-93cypromis: :)
12:15.04Bonboncypromis: do you know where I can a spanish sip number?
12:16.35cypromisBonbon: not yet
12:19.22cmanhey what was the disa format??
12:22.44UnixDawgext,1,DISA,password|where to go
12:23.07UnixDawgcli > show application disa
12:23.38UnixDawguse it only when really nessesary
12:25.24cmanguys help me out with this one..  i spent my whole day figuring tis out
12:25.25cmanhttp://lists.digium.com/pipermail/asterisk-users/2003-November/028027.html
12:26.57rajoany te410p or pri experts around?
12:27.33cypromisrajo: what's your problem with them >?
12:27.45Pj_cman: I guess the fact that you're in nepal is an explanation enough
12:28.10Pj_Asterisk may not be able to handle your country's caller id
12:28.35rajocypromis: te410p, 1 span to the telecom (Deutsche telekom), 1 span to an ascend box. calling out from * to the telecom is working.
12:29.00rajocypromis: calling in from telecom to ascend (using bridging) via analog modem is working, too
12:29.13rajocypromis: calling in via an isdn modem doesn't work (both data calls)
12:29.18rajoare there any issues?
12:30.23cypromisno idea
12:32.52*** join/#asterisk sjoeperd (~sjoerd@213-84-218-42.adsl.xs4all.nl)
12:38.45*** join/#asterisk zw1 (~chris@ewa-denver.com)
12:42.32*** join/#asterisk gadams666 (~wileyuser@63.111.7.137)
12:46.04*** join/#asterisk clive- (~pirch@rndf-ip-nas-2-p53.telkom-ipnet.co.za)
12:49.31rajois there someone (not from germany) who could give me a short call via landline? (just ringing for some seconds)
12:50.23*** join/#asterisk cfu (~kapejod@pD9E8193C.dip.t-dialin.net)
12:53.02maggsure
12:53.10maggrajo : what's your number?
12:53.34rajoMagg +49 681 379680
12:54.06maggcalling now
12:54.12magghidden callerid..
12:54.15maggringing..
12:54.21maggok?
12:54.23rajook
12:54.23rajothx
12:54.28maggno worries
12:54.31rajomagg: no, not hidden
12:54.47maggahh, wrong phone :D
12:55.05magg*grin*
12:55.31cmanis there anyway to change the pattern of * source file so that i can make it compatible with my countrys caller id?????
12:55.36*** part/#asterisk sjoeperd (~sjoerd@213-84-218-42.adsl.xs4all.nl)
12:57.38*** join/#asterisk torment0r|wk| (~someone@207.168.230.31)
12:58.08torment0r|wk|lolol@topic
12:58.08UnixDawgthere is a bug * is not cleaning up mpg123 when you stop now
12:58.20UnixDawgit leaves mpg123 running in the background
12:59.51*** join/#asterisk nitram (nitram@superblob.com)
13:00.04RoyKI've seen that...
13:00.13RoyKshould've reported that...
13:00.15RoyKsori
13:01.23torment0r|wk|i got a cisco 12+ ip phone that uses h 323 for free.  has anyone used one of these with asterisk before
13:01.47rajodoes anyone know how to set the outgoing capability (info transfer capability) of a te410p? want to make data calls, but they're sent out as voice
13:02.30*** join/#asterisk ToyMan (~stuq@smtp.dstoys.com)
13:03.52torment0r|wk|i agree with ya
13:04.04torment0r|wk|it looks like a pain to setup
13:04.22h3xrajo:
13:04.22h3x[Description]:
13:04.23h3x<PROTECTED>
13:04.26h3x<PROTECTED>
13:04.34h3xi think that will do it
13:07.09torment0r|wk|does anyone know of any examples of setting up sphinx with asterisk?
13:07.26torment0r|wk|besides the eagi script that comes with the source
13:07.31rajoh3x: Dial(Zap/g2/$(EXTEN)|160|c)?
13:08.21rajoh3x: this doesn't work :(
13:09.48h3xmaybe its a documented unfeature
13:10.04rajoh3x: :)
13:10.15*** join/#asterisk Prutser (Prutser@bitbucket.capcave.com)
13:10.26h3xor your telco rejects data calls
13:10.51rajoh3x: no, don't think so as this is the main purpose of this line
13:10.54PrutserHi all...
13:11.02PrutserAnyone using * on DM3 hardware?
13:11.18*** join/#asterisk _GiGi_ (gigi@disc.more.pl)
13:14.59cmanis there a way to telnet private ip? like i have pulblic ip in router... and if i want to see my linux box with ip 192.168.0.4
13:15.17dutch_pat
13:17.42*** join/#asterisk Kevorkian (~levo@ool-18bc8bc9.dyn.optonline.net)
13:17.58*** join/#asterisk coppice_ (~Steve@210.17.194.2)
13:18.49*** join/#asterisk lichen_ (~lichen@vanquish.cohpa.ucf.edu)
13:20.45KevorkianIm trying to get my * server to talk to iaxtel .. anyone got a moment to helpo
13:21.46*** join/#asterisk Vco (~Vco@h24-76-11-125.wp.shawcable.net)
13:23.20*** part/#asterisk pino (foobar@host115-28.pool21345.interbusiness.it)
13:38.48*** join/#asterisk Zebble_ (~Zebble@Sherbrooke-HSE-ppp3610369.sympatico.ca)
13:39.47*** join/#asterisk DrJack (~levo@ool-18bc8bc9.dyn.optonline.net)
13:40.41*** join/#asterisk cypromis (~michael@marge.halo2.pl)
13:41.24*** join/#asterisk bjohnson (~bjohnson@jecinc.tor.istop.com)
13:50.01*** join/#asterisk hi (~tauss@210.187.114.238)
13:54.20*** join/#asterisk p0lar (~Miranda@155.101.253.115)
13:57.52dougheckahttp://tuxscreen.net/
13:58.52*** join/#asterisk phiberkut (~phiberkut@tele-free-hotspot.netlinkip.com)
13:59.53*** join/#asterisk EagleRed (~eaglered@a213-22-50-141.netcabo.pt)
14:01.52*** join/#asterisk marrandy (~marrandy@209.216.76.1)
14:02.41marrandyHello.  Trying to get a manual fax working on ch. 4 fxo (tdm400) but seeing problems.  
14:02.59marrandyexten => 22,1,Dial(Zap/4-1,20,d)        ; Ring for 20 seconds, request a low latency call
14:02.59marrandyexten => 22,2,Hangup
14:03.35marrandyignore the -1, that's old and has been changed to Zap/4,
14:03.55marrandylog is as follows
14:05.00marrandyexten => 22,1,Dial(Zap/4,20,d)  ; Ring for 20 seconds, request a low latency call
14:05.00marrandyexten => 22,2,Hangup
14:05.17marrandyanyone see anything wrong with that ?
14:05.46marrandyexten => fax,1,Goto(22,1)
14:06.47marrandyHow many lines before I flood (for the log) ???
14:07.56marrandyanyone here ?  
14:08.44marrandyJerJer ?
14:09.28voidptr<PROTECTED>
14:09.54marrandyhello voidptr:  
14:09.59p0larhello
14:10.22marrandyahh.another one.  Hello p0lar
14:10.27p0larhwy
14:10.29p0larhey
14:10.32marrandyPretty quiet here
14:10.41p0lar<- too early to type or think
14:10.43_GiGi_;-)
14:11.02p0larI think irc is having probs
14:11.09marrandyDoes that fax extions look O.K. ?
14:11.13p0larI only have 20 people in my list
14:11.21marrandyextions = extension
14:11.29p0lar<- has never used faxes
14:11.41p0larbut it looks good.  whats wroing?
14:11.54tzanger~seen citats
14:11.57citats is currently on #asterisk.  Has said a total of 10 messages.  Is idling for 1d 19h 25m 50s
14:11.57marrandyHow many lines before I flood ?
14:12.13p0larflood?
14:12.51marrandyIf you put too many lines of text in, the system can flood and either kick me off or kick everyone off - not good
14:13.05tzangerquestion for you telco folks -- I have a D50 -> BIX cable, and the telco guy who gave it to me says that the buildout is pair 1 and 2, then skip two pairs, then 3 and 4, then skip 2 again, .. and so forth...  why?
14:13.13tzangeras in , why are they punched down in that manner?
14:13.23marrandyLet me break the log into 5 lines to be safe
14:13.58marrandy<PROTECTED>
14:13.58marrandy<PROTECTED>
14:13.59marrandy<PROTECTED>
14:13.59marrandy<PROTECTED>
14:14.00marrandy<PROTECTED>
14:14.00marrandy<PROTECTED>
14:14.21marrandy<PROTECTED>
14:14.22marrandy<PROTECTED>
14:14.22marrandy<PROTECTED>
14:14.22marrandy<PROTECTED>
14:14.37marrandyWARNING[327700]: File chan_zap.c, Line 1364 (zt_call): Unable to ring phone: Device or resource busy
14:14.37marrandy<PROTECTED>
14:14.37marrandy<PROTECTED>
14:14.37marrandy<PROTECTED>
14:14.55p0larahhh ok lemme look
14:15.00marrandyAs I'm a newbie, that is basically  'Duh'  to me
14:15.32p0laris zap 1 your fxs port?
14:15.46p0laris the fax ready? or could it really be busy?
14:16.04DrJackdoes iaxtel still allow 800 service outbound tru them ?
14:16.22marrandyzap 1 = fxo to the landline, zap 2-5 = fxs (tdm400p card)
14:16.24tzangerok marrandy that was a flood
14:16.43tzangerand on this D50->Bix, is pair 25 the first skipped pair?
14:17.15tzangeri..e does the order go  1,2,13,14 2,3,15,16 4,5,17,18 and so on?
14:17.24marrandyp0lar:  You just gave me a thought...let me check
14:18.05p0larok
14:18.13p0larglad someone had a thought :)
14:18.29*** join/#asterisk jp (~jp@hoochie.digium.com)
14:19.32*** join/#asterisk Kevorkian (~levo@ool-18bc8bc9.dyn.optonline.net)
14:22.14*** join/#asterisk jerkface (~me@67.70.231.218)
14:22.36*** join/#asterisk raoul (~raoul@lmepool1.ugr.be)
14:23.49*** join/#asterisk af_ (af@ip314-35-1.adsl.edisontel.com)
14:25.00*** part/#asterisk p0lar (~Miranda@155.101.253.115)
14:25.06*** join/#asterisk p0lar (~Miranda@155.101.253.115)
14:25.07dougheckagee, this sure looks like a nice phone: http://tuxscreen.net/
14:27.56*** join/#asterisk pino (foobar@host115-28.pool21345.interbusiness.it)
14:28.09RoyKgrr
14:28.25RoyKI can park calls, but I can't pick them up
14:28.35pinodoes anyone have a minute? i'd like to test-call and be test-called over iax or anything else to test this phone...
14:29.00raoulhi all, what are the different possibilities in terms of digium hardware if I want to be able to have several conversations at the same time?
14:29.48pinoraoul: several conversations over a single analog line, or something else?
14:30.40raoulpino: well, the only requirement is that we have only one phone number, ie several people can call us at the same time...
14:31.18dougheckadsl
14:31.19doughecka:)
14:31.39Pj_raoul: ask your phone operator I think
14:31.49Pj_you get several lines with the same number
14:32.10Pj_then you connect them to your server either with a few X100P or (far more better), a T1 line
14:32.45RoyKcan someone help me with parking probs?
14:33.10pinoraoul: or, in the middle, a few ISDN lines in a hunt group (if it is called that way... I still don't know too much telco terminology)
14:33.48raoulwell, my first plan was to have a few pstn lines (for example 3), and I asked to my operator to have the same number for all of them
14:33.57raoulthey told me it was impossible
14:34.30raoulnow, if I buy a T1 or E1 line, is it ISDN PRI? or is it something different? I'm lost with all those things ;)
14:34.48h3xraoul: where are you
14:35.07pinowhat they do here (Italy) is this: each single analog line has a different number, but the calls to the main line are "deflected" to the first non-busy of the other lines.
14:35.09raoulh3x: in Belgium
14:35.14h3xoh, heh
14:35.47raoulpino: mmh, so you have for example 3 analog lines with 3 different FXO cards and asterisk will manage that?
14:35.48pinoE1 is ISDN PRI, but it will be extremely expensive... it is usually convenient when you have at least 20 channels ("lines", so to say.)
14:35.48h3xwell anyway as long as you get trunks or PRI then every incoming call in the trunk group (usually the entire span)
14:35.56h3xwill come in with the dialed number and calling number
14:35.59tholoIf you specify you want a 30B+D PRI, that will be delivered as an E1.
14:36.03h3xso your system can route accordingly
14:36.05*** join/#asterisk benjk (~benjamin@f8a01-0359.din.or.jp)
14:36.13pinoraoul: yes, Asterisk will if your telco does that, but I'd tell you right away to use ISDN instead of analog lines.
14:36.29h3xYou don't have to use ISDN
14:36.33h3xto get this functionality
14:36.39h3xits nice but not required
14:36.45h3xif you order E&M Wink start with the right features
14:36.47tholoSome places you can get a PRI with fewer channels actually turned up -- but still delivered on an E1 (in Europe).
14:36.51h3x(ANI/DNIS delivery) you get the same thing
14:37.02pinoh3x: you don't, but i think it saves a lot of trouble...
14:37.19h3xwell in some parts of the world, PRI is too expensive to use
14:37.31pinoand at least here an ISDN 2B+D is cheaper than 2 POTS :)
14:37.36h3xfor the only real features gained is being able to reject a call without answering it, and setting your caller id for outgoing calls
14:37.43raoulmmh and ISDN BRI is 2 voice channels only, to be used with the CAPI channel, right?
14:37.53h3xwhich can be done with MF tones and FGD
14:38.10pinoraoul: that's the setting I'm using.
14:38.16tholoOr with the new, upcoming ZapBRI (should be out any day now, check with kapejod).
14:38.19h3xwell im talking about using robbed bit signalling and tones on a E1 instead of PRI ISDN
14:38.20pinoh3x: fgd?
14:38.24h3xFeature Group D
14:38.34h3xmaybe a north american thing
14:38.45raoulpino: and I guess you have 2 phone numbers too, but you are able to receive 2 different calls on the same number. Right?
14:38.54pinoh3x: maybe I read something about it, but I'm Not An Expert :)
14:38.56h3xi think you guys just use MFC R2 MF
14:39.03pinoraoul: yes, it is very well possible.
14:39.15h3xim sure euroisdn has a better chance of working with asterisk though
14:39.29tholoh3x: Most of Europe uses Q.931, not MFC R2 for PRI signalling...
14:39.35raoulso if I want to have 2 phone calls at the same time, I can use 2 dialup lines with number redirection if busy, or 1 isdn BRI line with chan_capi
14:39.43h3xok well there you have it
14:39.43h3xheh
14:39.52raoulif I want to have more than 2 phone calls at the same time, I have to use ISDN PRI = E1
14:40.01tholoYes.
14:40.09pinoraoul: no, you can just use more than one ISDN BRI ...
14:40.31raoulpino: with the telco redirecting all lines to the same number on a first-available basis?
14:40.33pinoyou will not be able to set the same callerid on all outgoing calls, but you could receive all calls on a single MSN...
14:40.33tholokapejod is about to deliver a product that can do 4 BRIs.
14:40.50raoulI need to check the prices
14:40.50pinoraoul: exactly.
14:41.12raoulok nice :)
14:41.20torment0r|wk|has anyone used a granstream budgetphone? do they work well?
14:41.21raouland what is T1 compared to E1?
14:41.32tholoMight fit nicely in the niche between BRI and PRI.
14:41.33torment0r|wk|or should i just step up
14:41.43pinotholo: I think it could also be done with two "cheap" passive cards, but I suspect ZapBRI will soon be far better in terms of features and support :)
14:41.50tholoT1 is 1.5 Mbit, E1 is 2 Mbit.  T1 is typically used in the US, E1 typically used in Europe.
14:42.02tholopino: Yeah, I agree.
14:42.25raoulis T1 a telephonic norm or is there a relation with the T1/T3 internet lines?
14:42.38tholoThere are also many more channel banks (for use with analog phones / phone lines) available with T1 interface than with E1 interface.
14:43.08raoulok, but I guess E1 is very expensive
14:43.17tholoThere is a relation between T1 and T3, yes...  A T3 has multiple T1's worth of capacity -- the exact number eludes me at the moment.
14:43.31tholoThat varies a lot from place to place.
14:43.36raoulok nice
14:43.50pinoand a T1 for telephone use and a T1 for internet lines are the same thing, but different signalling is used ...
14:43.56tholoYes.
14:44.01torment0r|wk|i think it's 23 t1's worth is a t3
14:44.10raoula last question now (sorry), when they say on digium.com that E1 is 32 channels, how many voice channels (ie how many calls) is it possible to do? And what is the difference with the quad-E1?
14:44.19tholoAnd a T1 (or E1) can even be shared between the two, with some channels for each purpose.
14:44.26*** join/#asterisk Cornfed (~nospam@24.236.221.138.gha.mi.chartermi.net)
14:44.27raoulpino: ok
14:44.38tholoThe Quad E1 is simply a card that can connect 4 E1 lines.
14:44.54raoulah ok
14:45.07tholoAn E1 has 32 64-kbit channels -- when used with a PRI (ISDN), you get 30 voice channels and one signalling channel (30B + D)
14:45.31tholoA T1 has 24 channels, 23 voice and one signalling when used as a PRI.
14:45.54*** join/#asterisk l-fy_ (~diana@home-25022.b.astral.ro)
14:46.23raoultholo: ok, so if I ask for a E1 for phone here, it is an ISDN PRI with 30 voice channels
14:46.50Mikeisnt a t1 a dedicated line for internet?
14:46.50raoulah there is a last thing that worries me, the very last one, I promise :)
14:47.00MikeE1 the same but european standars?
14:47.05Mike2200kbit?
14:47.49kapejod2048 kbit
14:47.50pinoMike: t1 is a dedicated line for whatever you wish; e1 is very similar, but 2048Kbit/second.
14:47.53tholoraoul: Only if you ask for an ISDN PRI.  There are other ways of using an E1.
14:48.00kapejod32 channels x 64 kbit
14:48.16tholoAnd here is the guy with the (upcoming) OctoBRI. ;)
14:48.22Mikeso you talk about t1s as for internet lines?
14:48.27raoulif I'm going for the setup of 3 FXO cards with 3 analog lines, in the asterisk config, I guess I will have to configure it to explicitely try each of the cards to find a non-busy line, but with an E1 or ISDN BRI, it would be automatic, right?
14:48.30kapejodwho?
14:48.36raoultholo: ok
14:48.42tholoT1s are usable both for phones and IP -- not limited to either, Mike.
14:48.47*** join/#asterisk anthm (~anthmct@CPE-24-167-193-161.wi.rr.com)
14:48.54tholoIf you get a PRI in the US, that comes on a T1, for instance.
14:49.04*** join/#asterisk swirlnets (~Mike@dsl093-001-038.det1.dsl.speakeasy.net)
14:49.05jensdE1 32 channels - 1D = 31 but you only gets 30 voice channels - what happens with the last one?
14:49.15pinoraoul: maybe the opposite... that is:
14:49.15Mike23phonelines on a t1?
14:49.18kapejodsome timing syncing stuff
14:49.26tholoMike: Yes.
14:49.37*** join/#asterisk point (1000@195.161.161.248)
14:49.39Miketholo: how much a month for one of those?
14:49.45pinoif you have 3 FXO cards, incoming calls will be directed by the telco; if you have a PRI, you configure * to do what you wish.
14:49.52Kevorkianjensd its 2 chans for the d ..
14:49.59pointhi there all ...
14:50.05RoyKgrr. can someone look through my config? I can't pick up parked calls anymore, after rewriting it...
14:50.12raoulpino: yes but I meant for outgoing calls
14:50.19tholoRewrite it back to what it was? ;-)
14:50.35Miketholo: how much for a t1 of those?
14:50.39RoyKtholo: neppe. it was a complete mess.....
14:51.03tholoMike: The price for a T1 / PRI varies wildly with location and phone company.
14:51.03raoulpino: because in Dial I have to specify the interface to use
14:51.25tholoI thought it was 4pm there?
14:51.30RoyKit is
14:51.33Miketholo: and intermedium price would be?
14:51.58tholoMike: Anywhere from $150 to $1.5k
14:52.05matt-1controlIs anyone else using NuFone right now and having trouble with it?
14:52.10Mike150 is cheap
14:52.15pinoraoul: I think that it is automatic then, but I've never tried to configure a PRI interface on *...
14:52.16tholoYes.
14:52.20pointkapejod has own way ... :)
14:53.01raoulpino: ok thank you :-)
14:53.04KevorkianMike .. 150 would be for a zero mile t .. to your equpment local at the co
14:53.04matt-1controlAlso, how does VoicePulse compare in reliability and customer service to NuFone?
14:53.06tholoAnd you can probably only get one for $150 if you are in a colo at an ILEC.
14:53.21raoulpino, tholo: thanks a lot for your help, everything is more clear for me now!
14:53.39pinoraoul: don't mention it:)
14:53.48raoul:-)
14:53.51tholoI hope I didn't give out too much misinformation. ;-)
14:54.22pointhere is collocation for $25 per unit  + incomming traffic
14:54.23raoulI think I will ask for 3 dialup lines and a redirection of the first of the 3 numbers to the first available one
14:54.24Kevorkiancan anyone give me a good link that explains the systax of extention commands ?
14:55.12pinoKevorkian: the most up-to-date info is got by "show applications" and "show application <name>"
14:56.34pinootherwise you might look at jtodd's sample configs: http://www.loligo.com/asterisk/current/
14:58.06doughecka~seen wasim
14:58.08wasim <~wasim@202.179.137.13> was last seen on IRC in channel #asterisk, 22h 38m 42s ago, saying: 'eww'.
14:59.18*** join/#asterisk bjohnson (~bjohnson@jecinc.tor.istop.com)
15:10.49*** join/#asterisk lucifuge3 (trilluser@pcp05047461pcs.ivylnd01.pa.comcast.net)
15:11.10*** join/#asterisk p0lar69 (~Miranda@155.101.253.117)
15:12.38tzangerwow
15:12.52dougheckawow
15:13.04tzangerI'm happier...  the telco guy was full of shit.  :-)  I was worried I'd blown out an FXS port on this CB (wired it to a PSTN line)
15:13.24tzangeryou'd ring it, it'd buzz like a bee on ludes, then hang up
15:14.06JerJermatt-1control:  everything is good here
15:15.02*** join/#asterisk Kalandor (~storm@inray.spnet.net)
15:15.09JerJertzanger: consider yourself lucky
15:15.32JerJerif u would have done that with a PhoneHack it would have smoked
15:15.50denonman .. another unexplained * crash
15:15.59denonanyone having problems with current?
15:16.01JerJergdb
15:16.13denonim just using safe_asterisk, no special parms
15:16.17*** join/#asterisk jtodd (~jtodd@207.141.153.205)
15:16.27JerJerdenon: run a bt   see /tmp
15:16.29KalandorSorry for disturbing, but  I am a developer just started to examine the asterisk and I have a question. Anyone willing to help me ?
15:16.46JerJerKalandor: it helps if you ask a specific question
15:17.02denonooh .. -rw-------    1 root     root     41967616 Nov 21 09:03 core.9235
15:17.11Kalandorokay I will proceed with it then
15:17.19*** join/#asterisk gage_man (~svoto@host2.216.41.24.conversent.net)
15:17.41KalandorIs there a chance of estabilishing a connection between to IPs in LAN without using the phone line, and how is this done
15:18.23JerJerKalandor: go learn about VoIP
15:18.42KalandorI am doing this now.. this is my first encounter with this subject
15:19.01KalandorI mean using the asterisk for that kind of connection
15:20.11denonper JerJer's privmsg request:
15:20.11denon#0  ast_queue_frame (chan=0x8172840, fin=0x8121a0c, lock=0) at channel.c:368
15:20.11denon#1  0x4069c3d2 in local_write (ast=0x813f328, f=0x8121a0c) at chan_local.c:111
15:20.11denon#2  0x080586f7 in ast_write (chan=0x813f328, fr=0x8121a0c) at channel.c:1392
15:20.11denon#3  0x4042fbe1 in wait_for_answer (in=0x815d8f0, outgoing=0x813f220, to=0xbd5feb2c, allowredir_in=0xbd5feb30,
15:20.12denon<PROTECTED>
15:20.14denon#4  0x40430cbb in dial_exec (chan=0x815d8f0, data=0xbd5ff304) at app_dial.c:635
15:20.16denon#5  0x08060bb0 in pbx_exec (c=0x815d8f0, app=0x8107a48, data=0xbd5ff304, newstack=1) at pbx.c:396
15:20.33lucifuge3Kalandor: Yes, it can be done.  There are many ways and your question is too vague to answer.  Take a look here: http://asterisk.xvoip.com/ and get some basic information on *.  It will probably answer all of your questions.
15:20.36JerJerKalandor: Asterisk is a hybrid TDM / Packet voice (VoIP)  PBX and IVR platform with ACD functionality
15:20.56JerJerlucifuge3: why not refer him to the offical documenation?
15:21.05JerJerhttp://www.digium.com
15:21.16Kalandorok ... thanx... I will take a look and then if I have some more specific questions I'll ask
15:21.17lucifuge3Because that's the firs bookmark in my list.  Why don't YOU refer him to the official docs.
15:21.31JerJeri did
15:21.37loko_mokooh the hostility <G>
15:21.37lucifuge3(Which are pitifully incomplete as of last check)
15:21.50*** join/#asterisk Vco (~Vco@h24-76-11-125.wp.shawcable.net)
15:21.52JerJerlucifuge3: then fucking write better or shutup
15:21.58lucifuge3Yeah...sorry.  Didn't get my entire first cup of coffee down yet. ;)
15:22.01lucifuge3lol...knew that was coming.
15:22.19JerJerMark and Maritn code
15:22.26JerJersomeone else needs to step up and document
15:22.37*** join/#asterisk segal_home (~segal@cuscon4542.tstt.net.tt)
15:22.41]data[didnt you like er just goto bed JerJer ?
15:22.59JerJeryep, i'm sleeping now
15:23.00*** join/#asterisk jules (~jules@63.105.150.196)
15:23.02]data[k
15:23.19tholoThis is JerJer's at-sleep behaviour -- you didn't know?
15:23.23*** part/#asterisk Kalandor (~storm@inray.spnet.net)
15:24.58julesIs there a bot I can bug the crap out of to find out about getting started with IVR? :)
15:25.12Exomorph_Morning people
15:25.24JerJerjules: read the docs (i already posted the url)
15:26.09julesJerJer, I'm currently on Digium's site grepping the FAQ... which is as far as I got. When you say posted, where did you post it?
15:26.51lucifuge3http://asterisk.xvoip.com/
15:26.56JerJeri can tell
15:27.04lucifuge3http://www.voip-info.org/tiki-index.php?page=Asterisk
15:27.16lucifuge3http://www.asterisk.org
15:27.18juleslucifuge3, JerJer, thanks! Very much appreciated.
15:27.44lucifuge3See....I gave him the official one that time too, JerJer.
15:28.34JerJerlucifuge3: there is a Unofficial Links section at digium.com that has all of those url's your spewing
15:28.53JerJerwhy not give out one URL and let them find those links?
15:29.28tholoThat'd be the same page that has the handbook, as well as mailing list information....
15:30.26julesYeah, make the damn newbies work for it lucifuge3.
15:30.43julesSeriously, a few links like that are a great leg up.
15:30.53lucifuge3Sorry.  I'm not pissed off enough this morning.
15:32.34dougheckaat least I am that generous
15:33.07*** join/#asterisk cocoy (strange123@203.76.221.55)
15:33.40cocoycan anyone give me how to create an extension for OH323 ?
15:34.14cocoyhelp
15:35.16*** join/#asterisk cypromis (~michael@marge.halo2.pl)
15:35.34JerJercocoy: can't help you on that one
15:35.58JerJerdamn your old
15:36.16tholoJust kids...
15:40.50cypromis*sigh*
15:40.51cypromiskids
15:41.17*** join/#asterisk mawdawg (dav@205.124.232.173)
15:41.29*** join/#asterisk Tekati (~captain@cpe-66-75-211-60.bak.rr.com)
15:42.35*** part/#asterisk Cornfed (~nospam@24.236.221.138.gha.mi.chartermi.net)
15:46.08JerJerok i think its time to unplug /dev/cerebral/cortex
15:46.56dougheckahahahaha
15:47.20TekatiIs AGI in the cvs repository or do you have to download it somewhere else?
15:49.04*** join/#asterisk raulfragoso (~raulfrago@200-204-142-71.dsl.telesp.net.br)
15:49.21*** join/#asterisk os2doc (~os2doc@hoochie.digium.com)
15:50.58raulfragosoAnyone here using Dialogic cards with Asterisk ?
15:51.22JerJerraulfragoso: no... use zaptel
15:52.00JerJerTekati: read more docs
15:53.00*** join/#asterisk Tddy (bartek@66.178.36.215)
15:53.17JerJerexten => XXX,1,AGI,foo.agi
15:53.28p0lar69anyone seen wasim lately?
15:53.30raulfragosoI'm in trouble to install Dialogic on RedHat 9.
15:53.30TddyHello everone. I need BIG HELP! Just bought Cisco 7960 and I upgrade it to SIP
15:53.45p0lar69ok gor for it
15:53.47p0lar69Tddy
15:53.52TddyBut now I cannot change any configuration because configuration is locked
15:54.04p0lar69in the Cisco phone?
15:54.08TddyYes
15:54.14p0lar69*##*
15:54.20Tddylet me see
15:54.22p0lar69to unlock
15:54.23bkw_no the password is cisco
15:54.30bkw_if its 5.0 and above
15:54.30p0lar69or *##
15:54.36bkw_or
15:54.38p0lar69no dude
15:54.42p0lar69SCCP code
15:54.52p0lar69he does not have SIP on it yet
15:54.54bkw_no *## dont work anymore
15:55.04bkw_he said its upgraded to sip and its now locked
15:55.05p0lar69it does not work on SIP code your right
15:55.09h3xJerJer: what carrier(s) do you use for inbound toll free?
15:55.09Tddyi tryed *## i do not work
15:55.12p0lar69AHHH ^$%$%
15:55.14Tddyalso *##*
15:55.14p0lar69sorry
15:55.18bkw_cisco is the password
15:55.25p0lar69<- still asleep
15:55.27Tddylet me check
15:55.29p0lar69yes
15:55.41p0lar69thx bkw_
15:56.07TddyTHANS
15:56.11TddyIT IS UNLOCKED
15:56.20TddyTHANKS SO MUCH!!!!!!!!!!!!!!!!!!!!!!!!!
15:56.26p0lar69no
15:56.34tholoAw, bwk_, you make it too easy! ;-)
15:56.35p0lar69thx to bkw that he was awake
15:57.11*** join/#asterisk zono (~trillian@inray.spnet.net)
15:57.13TddyThank to all
15:57.19denonbkw_: you know, im surprised ... I reported that crash several minutes ago, and yet I see no patches from you yet .. :)
15:57.25zonohi all
15:57.34bkw_denon I just got up.. and I have a cold
15:57.41denonah
15:57.44zonoI wont to ask about Asterisk
15:57.48denonwell I'll let ya off the hook then ..
15:57.53p0lar69ok zono
15:58.36doughecka~eeks
15:58.37somebody said eeks was http://eeks.convergence.com.pk
15:59.04Exomorph_wb jsmith :)
15:59.23*** join/#asterisk xupinet (~xupinet@150.162.248.8)
15:59.26dougheckahey
15:59.32dougheckaeeks.confergance.com is down
15:59.35xupinethi all! one easy question
15:59.42denon+.pk
15:59.43jsmithxupinet: What's the question?
15:59.44xupinetin extensions.conf file
15:59.55xupinetwhat does include mean?
16:00.01xupinetwhat does it do?
16:00.05dougheckayea
16:00.07xupinetthat's the question
16:00.30xupinetjsmith: can you answer me?
16:00.31jsmithxupinet: It includes the extensions from another context in the current context...
16:00.32p0lar69it allows you to "include" other files with * configugratuon in them
16:00.51*** join/#asterisk loko-moko (loko-moko@c-67-165-107-230.client.comcast.net)
16:00.56Exomorph_So I'm gonna mod the code to display only sip messages from xxx ip.
16:01.19p0lar69so you can keep all extensions from each context in seperate files
16:01.22jsmithExomorph_: Just look at it with a packet sniffer, and use that to filter...
16:01.28xupinetbut aren't they all include? why you must include them in context what just two lines above you have declared them?
16:01.38jsharp"include" in extensions.conf allows you to include contexts into other contexts...not include files.
16:01.54jsmithxupinet: Let me give you an example...
16:02.06Exomorph_jsmith: That doesn't show all the logic/etc going on on the asterisk side tho.  It only show's it they are talking to each other.
16:02.16p0lar69#include is what I was talking about......
16:02.24p0lar69<- is going back to sleep
16:02.27Exomorph_jsmith: and tcpdump doesn't show whats inside the packets themselves.
16:02.39xupinetjsmith: i am listening
16:02.42jsmithExomorph_: Use ethereal....
16:02.50JerJerExomorph_: wrong
16:02.53jsmithxupinet: Give me just a second... I'm trying to build a good example...
16:02.58*** join/#asterisk detten2 (~john@213.219.141.57)
16:03.05xupinetjsmith: yes of course!
16:03.06Exomorph_JerJer: Explain? :)
16:03.19JerJerman tcpdump
16:03.29Connor-tcpdump can.. -s0
16:03.31zoa<-- looking for a mirror with the debian 3.0r2
16:03.37Pj_Use ngrep
16:03.48Exomorph_Ya, but it still doesn't show the nice logic that asterisk can. :)
16:04.18JerJerConnor: incomplete answer
16:04.18*** join/#asterisk Shido6 (~shido@d57-81-103.home.cgocable.net)
16:04.25JerJertcpdump -s0 -x
16:04.32h3xJerJer: what carrier(s) do you use for inbound toll free?
16:04.34*** join/#asterisk anthm (~anthmct@CPE-24-167-193-161.wi.rr.com)
16:04.34*** join/#asterisk ]data[ (~data@193.138.95.4)
16:04.42h3xim trying to figure out something for rozo
16:04.42Connor-tcpdump -s0 -xxxv -ieth#
16:04.51rozohi
16:05.00*** join/#asterisk llboy (~llboy@hoochie.digium.com)
16:05.01JerJerh3x:  Global Crossings, Qwest and ATT
16:05.10h3xare you LCRing it ?
16:05.10rozogx here as well
16:05.15JerJerno
16:05.16Connor-oh and -X also
16:05.23h3xheh
16:05.24JerJer-x
16:05.25h3xok
16:05.40Connor-no -X prints ascii
16:05.54JerJerh3x: we have batches of numbers from each
16:06.00Exomorph_Connor: That gives some nice hex output...  but no text.
16:06.08Exomorph_Connor: LOL  read my mind.
16:06.25Connor-tcpdump -s0 -xxxvvvX -ieth#
16:06.33p0lar69use ethereal
16:06.34p0lar69:)
16:06.35h3xwell i was thinking he could send his WA calls to you with sms/800 or toll free enhanced routing
16:06.35Connor-the capital X gives you ascii
16:06.55h3xbut that might tilt the non-rboc in the wrong direction....
16:07.14*** part/#asterisk zono (~trillian@inray.spnet.net)
16:07.15JerJer?
16:07.21h3xintrastate calls
16:07.28JerJerhe can
16:07.39JerJerwe have CLECs sending us traffic
16:07.49*** join/#asterisk klicTel (~Claude@modemcable115.119-131-66.mc.videotron.ca)
16:07.53h3xthats cool
16:08.28h3xi think enhanced routing wont work because it would span two customer accounts
16:08.34zigmandoes anyone know a
16:08.35Exomorph_Connor: Ok, that might give me text... But its not formated anywhere near what asterisk does...  Nor does it show the logic behind what the Asterisk is doing...
16:08.41zigmanNIC with ROL  ?
16:08.43JerJer?
16:08.49h3xhes got gx
16:09.01h3xit would prob have to be done in sms/800
16:09.27Connor-tcpdump just gives you raw data..  you can tee it to a file then debug after the session
16:09.32p0lar69Exomorph can you use ethereal?
16:09.48JerJerh3x: i guess i'm not following you
16:09.59JerJerprolly cuz i haven't slept in like 40 hours
16:10.08h3xme either :(
16:10.18p0lar69Exomorph?
16:10.25Exomorph_01p0lar69: Could....   Have to install it first.
16:10.35jsharpJerJer: sleep gud.
16:10.37p0lar69it decodes SIP nucly
16:10.41p0lar69nicly
16:10.55Exomorph_Hmmmm
16:11.00JerJerjsharp: wish i could... i keep hearing ringing in my sleep
16:11.07JerJerthe fone that is
16:11.10p0lar69i also found a HTML prog that took a dump file and made a web page that traced the SIP messages out
16:11.24p0lar69you like something like that?
16:11.30jsharpI know the feeling.
16:11.43jtoddp0lar69: Yes, that would be pretty handy.  What's the name of the parser?
16:11.51p0lar69lemme look
16:12.05JerJerfun fun
16:12.13jsharpfrom Texas.
16:12.15jtoddjsharp: thxgiving holiday festivities starting early?
16:12.19Exomorph_Ya... send me the url...   If nothing else I'll include it in the asterisk doc some of us are making.
16:12.19JerJeroh shitty
16:12.37p0lar69ok
16:12.40JerJeri thought driving to chi from michigan sucked
16:12.42p0lar69its really cool
16:12.49jsharpyah.  We're headed up to visit family.
16:12.51p0lar69shows the whole call flow and SIP setup etc
16:13.06jsmithp0lar69: So give us the URL already :-)
16:13.17p0lar69looking.........
16:13.28p0lar69dude you do need some food
16:13.30p0lar69:_)
16:13.48p0lar69found it
16:14.06jsharpAnd hunt around chiland for employment as well.
16:14.25jsmithp0lar69: Sorry... low blood sugar this morning...
16:14.59p0lar69http://www1.cs.columbia.edu/sip/download/sip_scenario/
16:15.11p0lar69use this
16:15.11p0lar69tcpdump -s 0 -i eth0 'port 5060' -w /var/log/sip1.dump
16:15.14jtoddAnother one of Henning's tools, no doubt.
16:15.17p0lar69to make the dump file
16:15.24p0lar69heh
16:15.31p0lar69I smell EDU all over it
16:15.43p0lar69http://www1.cs.columbia.edu/sip/download/sip_scenario/
16:15.46p0lar69base url
16:15.51*** join/#asterisk ]data[ (~data@193.138.95.4)
16:16.16p0lar69got it Exomorph?
16:16.45Exomorph_Yup.  Thanks.
16:16.56Exomorph_Just a dir with files in it... :)
16:17.08p0lar69yeah
16:17.21p0lar69those are the tar files
16:17.30jtoddp0lar69: What I'd really love is a color-coded version that runs via an ANSI terminal session that hands out the same kind of data, instead of post-processing...
16:17.34p0lar69use the tcpdump command and create a file
16:17.34*** part/#asterisk xupinet (~xupinet@150.162.248.8)
16:18.08*** join/#asterisk p0lar (~Miranda@155.101.253.117)
16:18.12p0larok yes
16:18.15Exomorph_jtodd: So you'd like the sip debug fixed up to output a cleaner message format?
16:18.17p0larthat would be nice
16:18.41p0larcheck this out  it may help yer prob
16:19.51*** join/#asterisk cocoy (strange123@ipdial-172-28.tri-isys.com)
16:20.21cocoyhow do i make an extension for OH323 ?
16:20.33JerJercocoy: can't help you on that one
16:20.42cocoytnx jerjer
16:20.51bkw_fuck
16:20.54bkw_someone shoot me
16:20.57jetshey whats up p0lar
16:21.05p0larhey
16:21.05*** join/#asterisk ram (~ram@hoochie.digium.com)
16:21.09Exomorph_:)
16:21.09p0lartrying to wake up
16:21.21cocoyanyone pls help me
16:21.23*** join/#asterisk ManxPower (~eric@ip68-109-105-237.pn.at.cox.net)
16:21.37p0larwhats up bkw_?
16:21.56p0laryou cant be doing as bad as I have already today
16:22.19bkw_good
16:22.24bkw_denon what are you doing to crash *
16:22.28JerJercocoy: i wrote the H.323 channel driver that's distrubuted with Asterisk
16:22.30bkw_and are you running latest cvs?
16:22.42Shido6hey
16:23.08p0larHey jets
16:23.37p0larI talked to our Man over at Mountain States.... he said he just talked to you guys yeaterday too....
16:23.47cocoyi think so.. my friend installed it, am just configuring * so it can connect to GK then make call
16:23.56p0larhe told me you names but all I know is your aliases
16:24.11p0larRyan Nelson
16:25.27JerJercocoy: then he installed a useless driver, imho
16:25.33*** join/#asterisk i8kl (~lar@indo1.indosoft.unb.ca)
16:28.17denonbkw_: see that bug .. cvs as of yesterday, didnt do anything, it just happened this morning
16:28.33bkw_updated zaptel lately?
16:28.39denonupdated everything yesterday
16:28.41denonI do all at once
16:29.05jsmithp0lar: Yeah, I pointed Jets over to Ryan at Mt. States... do I get a commission?
16:29.11bkw_"bt full" gives more goodies when you do another backtrace
16:31.01torment0r|wk|JerJer: do you think it's possible to get a cisco +12 ip phone working with asterisk?
16:32.41*** join/#asterisk skeeziks (tim@66-23-208-2.clients.speedfactory.net)
16:33.43skeeziksIs it possible to set the bitrate on audio recordings coming out of asterisk?  On .wav files for example?
16:36.49ManxPowerskeeziks, no,  It's always 8000 since that's what telephony audio is.
16:36.59skeeziks16-bit?
16:37.06ManxPowerI don't recall.
16:37.13ManxPowerprolly 16 bit
16:38.21Corydon76-workYes, 16 bit
16:38.37skeeziksCool, thanks guys
16:38.45Corydon76-workAlthough it's the frequency that matters, not the number of bits
16:39.01ManxPoweryou can always resample it to a higher bitrate if you app requires it.
16:39.44i8klsend text agi, what channels support it
16:40.00Corydon76-workYes, but upsampling will give you worse results than downsampling
16:40.06i8klI'm having a hard time finding info on it
16:41.03skeeziksSo it's effectively a 128kb/sec stream...
16:41.44skeeziksThanks guys
16:42.53znoGkapejod: around?
16:43.07*** join/#asterisk mrgoby (~mrgoby@pcp05304587pcs.wanarb01.mi.comcast.net)
16:43.29mrgobymorning all
16:44.06*** join/#asterisk l-fy_ (~diana@home-25022.b.astral.ro)
16:45.49klicTelcan anyone give me hand with extensions.conf? My problem is that when an extension is on the phone, and another call comes in for that extension, it just keeps on ringing and ringing in the party's ear... is there a way to disable the second line? or to force a transfer to voicemail directly?
16:46.07bkw_klicTel grandstream?
16:48.07klicTeli have a channel bank with analog phones
16:48.33ManxPowerklicTel, uh, callwaiting=no and threewaycalling=no in zapata.conf
16:48.34*** join/#asterisk stan (~stan@213.78.66.69)
16:48.57ManxPowerGo to http://www.digium.com/index.php?menu=documentation and look at the "Unofficial Links" section.  This section has links to a wide variety of 3rd party Asterisk related pages.  My page is the "Asterisk Resource Pages".
16:49.09klicTelManxPower: that completly deactivate the second line?
16:49.23ManxPowerklicTel, It's not a second line, it's call waiting
16:49.42klicTelyes call waiting... sorry
16:50.11*** join/#asterisk outtolunc (~me@h-66-166-19-148.SNVACAID.covad.net)
16:50.14ManxPowerklicTel, I think my sample zapata.conf file on my asterisk page has callwaiting disabled, but you'd have to look to be sure.
16:50.37klicTelManxPower: I'm gonna try it... thanks
16:50.39h3xthree way calling is a good thing to have
16:50.49h3xcall waiting is annoying as hell
16:51.21h3xi still wish there was a way to set up slave call appearances of zaptel devices
16:51.24bkw_http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&item=3059326125&category=51256
16:51.35h3xso you could have it roll over to line 2 key for instance on a phone
16:51.50h3xman this fuckin rules
16:52.17gadams666hell, I'm just happy there is finally multiple presentations for a single extension on the cisco 6.0 SIP cdoe
16:52.30h3xi found a office space in the same building as williams, xo, xspedius, broadwing, and qwest for $0.63/ft^2 per month here in vegas
16:52.43h3xer .64
16:53.32znoGisnt there some way to set the caller on hold for X secs/mins until the extension becomes free, otherwise voicemail?
16:53.45h3xznoG: call queues.
16:53.48outtolunchttp://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&item=3059325876
16:54.08gadams666h3x: where's that at (vegas native here)
16:54.23znoGh3x: isn't that a good solution to what klicTel wants?
16:54.29h3xsahara & lamb is where all the telcos are
16:54.33h3xi was gonna get colo space
16:54.47h3xbut i can get like 2500 ft^2 for less than the price of a 8x8 cage
16:54.47h3xheh
16:55.10gadams666environmental controls too?
16:55.18h3xand the money saved in like 3-4 months i can just buy a damned dc power plant, batteries, rectifiers and shit off ebay
16:55.28gadams666only good thing I like about my co-lo space, A/C, power, and security all taken care of
16:55.41gadams666h3x: ;) there ya go
16:55.57h3xhahahahaha.   Rack Mount...  ROTFL
16:56.15h3xI could just steal power from the neighbors :P
16:56.30i8klis it possible to send custom text to display on snom200 sip phones
16:56.35h3xmost of the spaces in this building have iron bars over all the windows and shit
16:57.31gadams666well, considering the area, don't blame them. hehe. Sam's Town brings in a rough crowd.
16:58.15h3xyeah it sucks
16:58.28h3xbut what the hell
16:58.30znoGwhich area?
16:58.37h3xeven though i live in the west side, DI is a quick way to get there
16:58.54h3xsince it turns into lamb anyway
16:58.56gadams666yup - bars are good protection. get space to park your car inside too!
16:59.02h3xHAHAHAH
16:59.09tclarkznoG:http://bugs.digium.com/bug_view_page.php?bug_id=0000509
16:59.26gadams666we used to do that up near Valley View and Industrial
16:59.28*** join/#asterisk erik (~eanders@host-127-202-220-24.midco.net)
16:59.44h3xtommy rockers or something?
17:00.02gadams666razor wire fencing didn't keep the punks from throwing rocks at his car. so he started parking in the bulidng
17:00.14h3xshit i live at flamingo & decatur and have a call center across the street from me
17:00.18znoGtclark: haha, thanks. so many cool things you can do with *!
17:00.25gadams666Tommy's is up at Decatur, is it not? been a few years since I've been in town
17:00.26h3xand they are paying like $2.10 or some crap for their least
17:00.29h3xlease
17:00.30h3xno
17:00.42h3xactually its like flamingo & industrial
17:01.40gadams666ah, prolly moved since he orniginally opened it. Tommy's ok, knew him back in UNLV daze
17:01.40h3xthe reason all that shit is over there is Excel's fault
17:01.52h3xthey put a big phone switch next door so all the telcos built up around there
17:01.55h3xtheres 24 of them on that block
17:02.27h3xplus its a easy exit towards the phoenix/california/salt lake corner of the state where they have their fiber run
17:02.28gadams666lol. better be good service to Green Valley and Henderson then.
17:02.57h3xshit...
17:03.11h3xi got some guys that are moving their call center onto greenspun property
17:03.12gadams666yah. was never in the telecoms b4 moving out of Vegas
17:03.17h3xand you can't get anything but XO on that block
17:03.41h3xyou cant even order a sprint POTS line anywhere near 215 and green valley pkwy
17:04.01gadams666damn, that just aint right.
17:04.07h3xi wish we had global crossing here
17:04.20h3xwhen enron went BK, GX got wiped out too
17:04.24gadams666GBLX - now that was an intersting company
17:04.24h3xthey were coloed with em
17:04.41h3xso now we have these 230+ mile loops to LA
17:04.56oriontknouch
17:05.57*** join/#asterisk GhostNr1 (~Ashmed@193.10.185.3)
17:06.02gadams666STM-1 or above?
17:06.06*** join/#asterisk marrandy (~marrandy@209.216.76.1)
17:06.23h3xis what stm-1
17:06.27marrandyHello. Fixed my earlier fax machine issue
17:06.45marrandyOn to the next issue.
17:07.06h3xer well lets just say a t1 local loop is $750
17:07.08bkw_if I had 10 bucks for every time I wanted to shoot the telco.. I would be one rich mofo
17:07.14gadams666stm-1 == oc-3 I think. then submarine fiber guys seem to talk STM only. sigh
17:07.15h3xactually now they sell em for 200 but
17:07.27h3xi know, but i didnt know what you were referring to
17:07.36gadams666bkw_: $0.05 and you'd still be richer than Gates
17:07.36marrandyOn a conventional fax/ans phone, you can monitor as people leave a message and pickup.
17:07.38mrgobyshoot, bkw_  if i had $0.01
17:07.43h3xthe problem is theres no way for me to get a 0 mile loop to GX
17:08.16oriontknor if your cliped onto the vm ports :)
17:08.17marrandyHow do you monitor/put through the pc speaker in * ?
17:08.32mrgobykram:  I suggest a new topic "Ma Bell beats her kids"
17:08.34ManxPowermarrandy, Actually that's only the case with an answering machine.  Not for PBX or Key Systems
17:08.48ManxPowerOr voicemail systems.
17:08.59oriontknpanasonic, iwatsu and a few others have the ability to listen to messages as they are left
17:09.08ManxPoweroriontkn, celever.
17:09.10raoulcu all :)
17:09.11oriontknthe kx-TD series in perticula
17:09.12oriontknr
17:09.12ManxPowerclever, even
17:09.15marrandyso you can't monitor as a message is being left in VM ?
17:09.18gadams666GBLX still got US presence on their Frontier network?
17:09.20bkw_People names Johnny Depp sexiest man
17:09.25bkw_RIIIGHT
17:09.32ManxPowermarrandy, Not without doing dialplan hacks
17:09.33bkw_he's not sexy at all.  eww eww eww
17:09.51ManxPowerbkw_, I think he's kinda cute
17:10.09bkw_ewww
17:10.16gadams666god actor. esp in Tim Burton films. IMO
17:10.19bkw_gag me with E.T.'s finger.
17:10.25mrgobyheheh
17:10.45marrandyhmm...that could be a problem.  I am testing this at home
17:11.07ManxPowermarrandy, Then buy an answering machine and don't use the voicemail system
17:11.37oriontkntheres no way to intercept the stream and copy it to another timeslot on another port?
17:11.41marrandyI have one.  But in the testing phase, I want to replace it as I laern the system
17:11.45h3xYeah, well they sell a local enhanced service product on the class 5 switches.
17:11.45ManxPowerThat feature can't be all THAT popular since almost everyone I know uses either the voicemail on their cellphone or the voicemail service from their LEC and neither of those support that feature.
17:12.04ManxPowermarrandy, Yes.  Be prepared to do some coding in C
17:12.06oriontknor CLAES
17:12.09marrandylaern = learn
17:12.34ManxPowermarrandy, You could prolly hack app_monitor to spit the audio at /dev/audio
17:12.34mrgobymarrandy   you could write an AGI that plays your voicemail for you after they leave it ...  although that might not be the desired effect
17:13.03ManxPowerHeck you might even be able to give app_monitor /dev/audio as an output filename.
17:13.05*** join/#asterisk HeeD (~db@s216-232-111-32.bc.hsia.telus.net)
17:13.06ManxPowerI don't know.
17:13.30marrandyWell, you know how it is, you call screen and monitor the line and then decide whether to pickup.  I was hoping to suplicate this or the misses is going to be unhappy
17:13.31gadams666gotcha. GBLX, Southern Crossing, and the rest of the sub companies were HQ'ed down the street from us in Bermuda
17:13.40marrandysuplicate = duplicate
17:13.42ManxPowermarrandy, You don't have callerid?
17:13.42*** join/#asterisk ]data[_ (~data@193.138.95.4)
17:14.13marrandyIt costs $6.95 plus taxes per month
17:14.15gadams666ironically they were denied the abiltiy to land a fiber there. cust care and all data servers provided by clueless and worthless at the time.
17:14.25ManxPowermarrandy, put an answering machine on the misses FXS port.
17:14.47mrgobyDoes anyone know what is going on with DynExtenDB ??  The site is down, or at least not accepting my pings/http requests
17:15.08marrandyI want to play with it here, but obviously, have to keep the boss happy  ;-)
17:15.12*** join/#asterisk cocoy (strange123@203.76.221.68)
17:15.28ManxPowerYet another reason not to get married, but I digress.
17:16.00marrandyManPower: you a * developer ?
17:16.08cocoywhat do i need to put on extensions.conf so i can call thru a GK?
17:16.32marrandyManxPower: you a * developer ?  Damn...my typing is going to pot
17:16.32tclarkmarrandy ManxPower: invoke zapbarge on the fxo channel to listen to vms ..
17:17.20marrandyoohh...ahhh...can you also pickup ?
17:17.23ManxPowermarrandy, No, but I play one on IRC.
17:17.36mrgobycocoy    i think last night someone said that all you have to do is register asterisk with the GK and then you can call through with Dial(h323/blah@blah)
17:17.43ManxPowermarrandy, sapbarge is designed for bosses to evesdrop on their employee's phone calls.
17:17.57ManxPowerzapbarge, even
17:18.01h3xsapbarge? is that a tree version?
17:18.13ManxPowertclark, how would you send the zapbarge audio out the sound card?
17:18.19h3xyou hear the next 2 people under you at the same time? :)
17:18.49tclarkManxPower: missed that part, yea that would be a mod to zapbarge
17:18.56marrandymarrandy returns to learning mode as the discussion ensues
17:18.57h3xI have an idea, How about you send voicemail calls to a meetme
17:19.10ManxPowertclark, why not using app_monitor and give it a filename of /dev/audio?
17:19.15h3xand somehow set up that meetme to invite the voicemail and monitoring party.   naaah  nevermind
17:19.23mrgobyh3x   that might work ...  and make your console a silent listener
17:19.41h3xzapbarge would be better
17:19.46HeeDI got my g.729 license just now.... setup on * and cant wait to go home to try on ata-186 :)
17:19.47mrgobymaybe???  hehehe  we are searching here :-)
17:20.07tclarkManxPower: wee with a small mod i have to invoke zapbarge I can talk as well as listen not sure about app_monitor ..
17:20.35cocoymrgoby- can u give me an idea how can i make call from SIP to GK?
17:20.42h3xI don't get why monitor and barge are named what they are
17:20.43h3xmonitor should be log or record
17:20.44mrgobymaybe
17:20.48ManxPowertclark, Can you also play a sound file?
17:20.48h3xand barge should be monitor
17:21.11ManxPower"Please disconnect now.  The line is required for a 911 emergenct call" sort of thing.
17:21.16mrgobycocoy   this is just my understanding, which might be limited so don't take this to be from an expert, but
17:21.27tclarkManxPower: not right now but I guess you could make that an option & do ast_streamfile ..
17:21.33h3xwho the hell would care for a 911 call.  Just Hangup() the damned line
17:21.35oriontknit needs to support autovon precedence
17:22.04*** join/#asterisk Vco (~Vco@h24-76-11-125.wp.shawcable.net)
17:22.05*** join/#asterisk Damin ([rtFCpCntc@nucleus.nacs.net)
17:22.07cocoymrgoby - i'll gladly accept what u can give me
17:22.16oriontkn<beep> <beep> and dumps the caller
17:22.23DaminAlright.. So I got my first live demo of asterisk last night..
17:22.42mrgobyyou can register your * machine with the GK ...  this allows * to call h323 clients...  then, just make an extension that calls an h323 client who is also registered with the GK
17:22.43tclarkManxPower: if you look at zapbarge it is realy bridgeing the channels to a meetme conf, its the same code as meetme
17:22.50mrgobythen just call that extension on a SIP client
17:23.01zoaanyone with a sipbarge ?
17:23.10DaminI've known about it for a while, but I was just clued into how totally kick ass it is!
17:23.10mrgobycocoy does that help?
17:23.13ManxPowerh3x, I'm concerned about someone trying to dial right after they are disconnected
17:23.20DaminAnyone from Digium online?
17:23.55mrgobyis there a doctor in the house?
17:24.02cocoymrgoby - that's what i need an extension syntax i used exten => _X.,1,Dial(OH323/h323:${EXTEN}@111.111.111.1,60,r)
17:25.06ManxPowerFor some reason I thought it's supposed to be OH323/ip:${EXTEN}@111.111.111.1,60,r
17:25.13t4khow do you increse taps for echocancelation?
17:25.17ManxPowerBut I don't run H323 so I don't know for sure.
17:25.18mrgobycocoy , unfortunately my associates took the box that has our h323 extensions offline, and i boked too much smot in college to remember the syntax exactly for h323
17:25.29cocoymrgoby- i will try
17:25.40*** join/#asterisk maharzan (~chandra@202.51.76.74)
17:25.44mrgobysorry
17:25.49mrgobythat was poked too much smot
17:25.53mrgoby:-D
17:26.04mrgobyindeed
17:26.19maharzanwasim akram????
17:26.25maharzanwhere r u?
17:26.33mrgobykram
17:26.37mrgobythere he is
17:26.47maharzanhey any body outside US successful in setting up caller ID???
17:27.24HeeDI am using voicemail... when dialing in on FXO card and as soon as woman's voice answers prompting to leave a msg after beep... if person calling hangs up it takes a few secs to recognize hangup and records a second or 2 of silence. Any way to add more time after the beep prompt?
17:27.59ManxPowerHeeD, Why do you want more time after the beep prompt?
17:28.17maharzanbeep, more seconds
17:28.27maharzanLOL
17:28.28ManxPowerHeeD, Just fiddle with the silence stuff in voicemail.conf
17:28.28HeeDmanx because if someone hears the voicemail prompt and decides to just hangup and not leave a msg, it still records a couple secs of silence
17:28.37maharzan8-)
17:28.49ManxPowerHeeD, You would rather the system not record the first few seconds of their message?
17:29.04cocoymrgoby - do i need to change something else on extensions.conf ? any more ? how abt "default"
17:29.15*** join/#asterisk rusty_ (~rusty@65-101-255-24.dnvr.qwest.net)
17:29.26HeeDno, I want the voicemail to wait a few seconds after the prompt to leave after beep before it actually starts recording
17:29.40denonHeeD: make a beep.gsm with a delay
17:29.44denonsome silence
17:29.57ManxPower*beep* *wait* [missed the name and phone number] "so if you want to get togather tonight give me a call"
17:29.59HeeDdenon that sounds like it would work
17:30.19denonHeeD: it'll work fine, but make sure you have a script copy your beep back after every CVS, cause it'll get overwritten
17:30.20mrgobycocoy    i'm not sure...    i would put it in your default context if you don't understand how contexts work
17:30.40ManxPowerHeeD, Do you know how every other user of Asterisk solves that problem?
17:30.56HeeDManx no
17:31.02*** join/#asterisk Mike (~mike@dsl-200-67-4-11.prodigy.net.mx)
17:31.14cocoymrgobt - error "Channel 'H323:1820' sent into invalid extension 's' in context 'default', but no invalid handler"
17:31.19ManxPower; How many seconds of silence before we end the recording
17:31.19ManxPowermaxsilence=10
17:31.19ManxPower; Silence threshold (what we consider silence, the lower, the more sensitive)
17:31.19ManxPowersilencethreshold=128
17:31.33Connor-So, can anyone tell me why when the IVR answers, that 80% of the time, it custs of the first few seconds of speach ?
17:31.43mrgobyhmmmmmmmmmmmm    what does your exention look like cocoy ??
17:31.52ManxPowercocoy, Looks like the call is being sent to extension "s" in the [default] context.
17:31.52mrgobygive me the line in extensions.conf
17:31.53Connor-Mostly happens with SIP, but, sometimes with PSTN too
17:31.56mrgobyyes
17:32.38maharzananybody not from US
17:32.39cocoymrgoby - [sip]
17:32.40cocoyexten => _X.,1,Dial(OH323/ip:${EXTEN}@203.144.224.167,60,r)
17:32.44maharzanye yeah
17:32.49maharzansay yeah
17:33.29cocoymrgoby i have no [default] context under extensions.conf
17:33.30Mikewhere do i configure h323 in asterisk?
17:33.32ManxPowermaharzan, CallerID works in a lot of countries.  Not BT UK lines, however.
17:33.47maharzanindia?
17:33.54ManxPowerMike, That question is a lot like "How do I drive a car."  Read the README file.
17:34.01ManxPowermaharzan, prolly not. 8-)
17:34.03maharzani am in nepal.. they should be similar to india pk..
17:34.17bkw_ATTENTION H.323 SUCKS.. DONT USE IT IF YOU DONT HAVE TOO!
17:34.20ManxPowermaharzan, How is CallerID delivered to you?
17:34.21bkw_ok that said...
17:34.26rozomaharzan: we weren't able to get callerid to work in new delhi, india
17:34.56ManxPowermaharzan, Is it FSK or DTMF.  When is it delivered?  Before the first ring or between the first and second ring?  Does it support name as well as number
17:35.25mrgobybkw_   didn't you field this last night ??
17:35.30maharzani 'll have to figure that out.... but i seee that when i connect directly to pstn.. the callerid appears after one ring.. as that os US.. don't know what they use.. FXS or other..
17:35.41i8klis it possible to send custom text to display on snom200 sip phones
17:35.47mrgobyan h323 question very similar
17:35.56mrgobyi can't remember if that was you
17:35.56*** join/#asterisk anachron (~fradooj@adsl-64-169-89-27.dsl.snfc21.pacbell.net)
17:36.00bkw_i8kl you have asked that like twice
17:36.03bkw_post to the mailing list
17:36.06maharzanit gives me number only.. like 012222555 for local no
17:36.21cocoymrgoby - that's not me, am just learning * now
17:36.33ManxPowermaharzan, If it's sent as FSK after the first ring then either the X100P supports that callerid or it can easily be patched to do so.
17:36.39cocoybut am vey familiar with opengk
17:37.10maharzanwhats DTMF?bbefore first ring? as in UK?
17:37.12ManxPowerActually if it's sent as FSK or DTMF after the first ring chances are that it can be made to work eventually
17:37.13MikeManxPower: im trying to figure out if the quicknet card can do sip would make it easier
17:37.19ManxPowermaharzan, Yes.
17:37.21MikeManxPower: do you know any software?
17:37.54Daminq
17:37.54ManxPowerMike, No, the Quicknet card does not do SIP.  Just like the ZapTel cards don't do SIP and the Dialogic cards don't do SIP.
17:37.54ManxPowerMike, I know lots of software.
17:37.56cocoymrbogy - do i need only one line for OH323 to work?
17:38.10MikeManxPower: so i need to get it working under h323?
17:38.14*** join/#asterisk FlowerPower (~michal@otho.nask.waw.pl)
17:38.30ManxPowerMike, I have no idea.  I know Netscape, MS Ofice, Linux, GNOME, but I don't know H323
17:38.37FlowerPowerhi
17:38.44ManxPowerMike, NO!  These cards are PSTN cards not SIP cards.
17:39.01mrgobywell....   i don't know really    ....  exten => 420,1,Dial(SIP/600@111.111.111.111)
17:39.09mrgobythis is one that works for me with sip
17:39.15*** join/#asterisk segal4 (~segal@cuscon4728.tstt.net.tt)
17:39.21ManxPowerIf you want to interface with SIP do it just like you would with a ZapTel card.  Load the kernel drivers and set up your extensions.conf and the rest of the .conf files.
17:39.23mrgobyi would imagine changing the SIP to H323 would work
17:39.26mrgobybut i don't know
17:39.35maharzancallerid comes after the first ring.. b4 second... what can be done to fix it?
17:39.41MikeManxPower: i have it on a windows computer
17:39.45ManxPowermaharzan, Post a bug report and pray
17:39.46FlowerPoweris here any1 who has succsessfully configured isdn with *?
17:39.51ManxPowerMike, I can't help you.
17:39.59maharzan:D
17:40.13maharzansome one told me that i can change callerid.c file
17:40.15ManxPowerMike, But I guess you would have to load SIP or H323 software on the Windows computer.
17:40.22maharzani'll try to have a look around
17:40.48FlowerPowermaybe you know some sites where the problem is described
17:40.50FlowerPower>
17:40.51FlowerPower?
17:41.03ManxPowerGo to http://www.digium.com/index.php?menu=documentation and look at the "Unofficial Links" section.  This section has links to a wide variety of 3rd party Asterisk related pages.  My page is the "Asterisk Resource Pages".
17:41.11ManxPowerLook at the CAPI or kapejod pages
17:41.14MikeManxPower: thats whay im looking sip software compatible with quicknet card
17:41.32ManxPowerMike, I doubt you are going to find anyone here that runs Quicknet on Windows.
17:41.47ManxPowerMike, Call up the makers of the card.
17:42.10MikeManxPower: i will look for some box running linux with isa
17:42.14bkw_kram kram kram
17:42.16bkw_:)
17:42.17Mikei just found this windows box has isa
17:42.21kramlol
17:42.26*** topic/#asterisk by kram -> "Ma Bell beats her kids"
17:42.27Mikethats my problems ISA is hard to find this days
17:42.36ManxPowerMike, Would it not just be cheaper to but an X100P?
17:42.38Mikeand the only computer with an isa slot is a winXP computer
17:42.47marrandyConnor:  Do you mean the first part of the announcemnet
17:43.07MikeManxPower: a friend just told me to use it
17:43.23*** join/#asterisk os2doc (~michael@65.115.136.98)
17:43.26MikeManxPower: i dont want to use it as a zaptel card
17:43.28*** join/#asterisk jdg (~chatzilla@202.3.246.151)
17:43.34mrgobykram   YOU RULE
17:43.39MikeManxPower: i just want to plug a telefone to put an extension
17:43.50ManxPowerYou mean like a TDM10B?
17:44.07Mikesorry i dont know alot of the hardware yet
17:44.09marrandyTDM400
17:44.17ManxPowerMike, I seriously doubt you will find any Asterisk user that has the configuration you are trying to do
17:44.25marrandyhas expansion to 4 extensions
17:44.30ManxPowermarrandy, No, that's a 4 ports card, not the 1 port card
17:44.48*** join/#asterisk denon (denon@synapse.subneural.net)
17:44.51ManxPowerWell technically they are both the same card.
17:44.58MikeManxPower: this card can also be useful as a x100p?
17:45.05ManxPowerMike, No.
17:45.13Mikeok
17:45.17Mikethen ill just return it
17:45.18denonhaha .. that topic is great
17:45.24ManxPowerThe X100P is for using with a telephone line.  The TDMx0B is for using with a phone
17:45.31marrandyNot at the moment (TDM400) they are working on FXO modules
17:46.11marrandyI bought the TDM400 with 1 module, then bought the extra 3 fxs modules
17:46.35marrandyI also have the X100P as I couldn't wait for the FXO modules to appear
17:46.43*** join/#asterisk os2doc (~michael@65.115.136.98)
17:46.44ManxPowermarrandy, The TDM400P is the card with no modules, the TDM10B is the card with 1 mondule.
17:47.04*** join/#asterisk lucifuge3 (trilluser@pcp05047461pcs.ivylnd01.pa.comcast.net)
17:47.22UnixDawgdoes anyone know what stealths name is
17:47.24ManxPowermarrandy, if you don't put P or B at the end of the name when you refer to the card people can get confuzzled.
17:47.48silugso is digium replacing the original tdm400p cards?
17:47.53marrandyhttp://www.digium.com/index.php?menu=wildcard_tdm400p
17:48.06marrandyOh, I see.  They have changed the terminology
17:48.39bkw_what changed?
17:48.58*** join/#asterisk rainer_home (~rainer@p508AF127.dip.t-dialin.net)
17:49.40ManxPowerTDM400P is the card with no modules installed.
17:50.59marrandyProbably...I don't remember.  I ordered mine in Mid August and it was delayed till September until the new TDM400P's came in
17:51.00cocoymrgoby - can u give urls where can i see setting-up OH323
17:51.18bkw_cocoy don't do it if you don't need it
17:51.24bkw_h323 is pure and evil..
17:51.29bkw_at the same time.
17:51.35ManxPowercocoy, There are no docs with the tarball you downloaded?
17:52.00cocoybkw - i need to call from SIP to GK
17:52.30*** join/#asterisk os2doc (~michael@65.115.136.98)
17:52.47cocoymanx - few docs on it
17:53.00ManxPowercocoy, Have you tried the H323 driver that comes with Asterisk
17:53.40os2docAnyone want to answer a newbie question?  Setting up a phone system
17:54.00cocoymanx - yes that's what am using
17:54.02marrandyIf you want a connection to the CO and have extensions, then get  http://www.digium.com/index.php?menu=developerskit_tdm
17:54.17JerJerTHAT IS NOT OH323 THEN!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!
17:54.21ManxPowercocoy, Uh, the chan_oh323 is not included with Asterisk.  chan_h323 is included with Asterisk
17:54.57ManxPowercocoy, They are TOTALLY different software.
17:55.27marrandyI would get Two fxs modules so you can test between Two regular phones
17:55.45cocoymanx - how will i check if i have chan_oh323 , my friend said he already installed it
17:55.58Corydon76-workWell, they aren't totally different.  But they are different enough that they aren't the same.
17:56.20ManxPowerCorydon76-home: The codebase is totally different.  The design is totally different.
17:56.21JerJerCorydon76-work: oh but they are tottally different
17:56.35JerJerthe only simularity is they use the same H.323 stack
17:56.36Corydon76-workThey implement the same protocol.
17:56.59JerJerCorydon76-home:  LInux and windows use X86 processors also
17:57.06ManxPowerCorydon76-home: Grandsream and Cisco 79xx both implement the same protocol too 8-)
17:57.59Corydon76-workJerJer:  they aren't totally different, either.  They're very different, but totally is not an appropriate qualifier
17:58.21Exomorph_JerJer: you have to admit that having them both named so simular, can confuse alot of people.   When I first looked at h323 channels for asterisk, I initially thought they were both the same project, just at some point they changed the name.  (of course I now know different)
17:58.33ManxPowerCorydon76-home: Think "totally different" is the correct term when talking to someone that's confusing the two. 8-)
17:58.47bkw_ManxPower I agree
17:58.56Corydon76-workhyperbole
17:58.57ManxPowerThey should just call them chan_jerjer and chan_mbraca
17:59.12Corydon76-workmbranca didn't write chan_oh323
17:59.26ManxPowerCorydon76-home: or chan_whoeverwroteoh323
17:59.32*** join/#asterisk feanor (~feanor@63.245.86.109)
17:59.52Corydon76-workManxPower:  that's even more confusing
18:00.11JerJertechnically  its chan_h323 and asterisk-oh323
18:00.13ManxPowerCorydon76-home: Maybe so, but people would stop confusing the two.
18:00.43Corydon76-workManxPower:  considering what they do, not a chance
18:01.39Exomorph_What needs to be done is a good Asterisk Doc made up (we're in the process, right jsmith?) that clearly states they are diffrent, and all the details with them.
18:03.43Corydon76-workhow about chan_oh323_dynamic and chan_h323_static ?
18:04.01JerJerhow about chan_broken_h323 and chan_h323 ?
18:04.29*** join/#asterisk elbandy (~elbandy@pD9E3E5BB.dip.t-dialin.net)
18:04.35bkw_FIGHT.. FIGHT.. FIGHT.. FIGHT..
18:04.38Corydon76-workUnfortunately, oh323 is broken, so both channels are broken
18:04.51*** part/#asterisk elbandy (~elbandy@pD9E3E5BB.dip.t-dialin.net)
18:04.52os2docWould anyone be able to answer a few basic questions about setting up a basic phone system.  I am having difficlty getting specific advice and information...
18:04.52cypromischan no323
18:04.53jsmithsuggestion, not implementation...
18:04.59JerJer:)
18:05.02jsmithos2doc: Just ask...
18:05.08bkw_chan_ho323
18:05.17JerJerlol
18:05.26Corydon76-workchan_oink323 and chan_moo323
18:05.49bkw_its chan_orange and chan_apple
18:06.01jsmithchan_die_H323
18:06.14Exomorph_LOL
18:06.14cypromisyou have 80 fowl oranges left .... beep ... select your destination party ...
18:06.23mrgobychan_can_string.so
18:06.40JerJerwe need an independant third party to setup and deploy each one, then whichever one he picks the other one goes totally away, forever
18:06.44os2docCool.  ;)  I am setting up a phone system for my practice, and want to use as much gnu software as possible.  It is a medical practice.  Anyway, 5 B1 lines coming in, and want to use asterisk for the phone system.  The problem is trying to figure out what phones I can use with the system.  Is it safe to assume that any PBX phone will work with Asterisk?  Like the Nortel Meridian 2317 for example?
18:07.05Corydon76-workJerJer:  could you really live with yours going away, though?
18:07.17jsmithos2doc: No... only analog phones, or VoIP phones
18:07.28JerJerCorydon76-work: sure...i've purged H.323 totally from my operation and will never go back to it
18:07.29jetsos2doc why use old phones, when you can get nice voip phones
18:07.31marrandyI don't think any proprietry phone will work with asterisk
18:07.44jsmithos2doc: PBX phones are usually a proprietary digital format, which Asterisk has no way of talking to
18:07.52cypromis:)
18:08.05cypromisI am trying to rid my operations of h323 as well
18:08.07marrandyUnless you link asterisk to your PBX which sort of defeats the purpose
18:08.09JerJerPlus, its my code, if for some dumb reason there I find a need for it, i'll pull it out of my own cvs
18:08.23JerJer:P
18:08.44Corydon76-workThat's not exactly your code going away, forever... :-P
18:09.04JerJerit wouldn't be in asterisk and i wouldn't support it
18:09.09marrandyJerJer:  do you know if you can monitor and pickup someone leaving a Voicemail ?
18:09.16JerJerif it happened to loose the test
18:09.16*** join/#asterisk anthm (~anthmct@CPE-24-167-193-161.wi.rr.com)
18:09.20os2docjsmith: ok, so I use standard phones or voip phones.  The problem is trying to figure out which phones are VOIP.  I have been looking at phones on ebay to save money, but am getting really confused...  I had assumed that the PBX phones would be voip.  Bad assumption?
18:09.44Corydon76-workbad assumption.
18:09.45JerJerbut anyone that has acutally used that other one knows the limitations
18:09.56Corydon76-workMost PBX phones are not VoIP phones
18:10.06ManxPoweros2doc, PBX phones are NOT "standard phones"
18:10.41JerJerhence why i wrote mine (after the author wouldnt listen to me and told me i couldn't do any better)
18:10.48ManxPowerPBX phones are "vendor phones made incompatable with any other system so the vendor can lock you into their phones and charge you incredibly high prices"
18:11.02os2docThat answers some questions. Now where to find phones.
18:11.09Corydon76-workFace it, you can either go for many different proprietary phones, that all implement a protocol in various different ways, or you can use expensive channel banks and cheap phones from Walmart...
18:11.52ManxPoweros2doc, You'll find a lot of different suggestions. *I* think VoIP phones at a decent pricepoint at NOT mature and should NOT be used.  I feel that people should use analog phones on a Zap port.
18:11.56marrandyHow many incoming lines ?  How many extenions ?
18:11.59Corydon76-workPick your poison, as it were.
18:12.03marrandyT-1 ?
18:12.04ManxPowerThere are other people that think VoIP phones are just fine.
18:12.07marrandypri ?
18:12.10UnixDawgwhere in the hell is stealth
18:12.19UnixDawgI am going to beat him with asterisk
18:12.28os2docmarrandy: 5 incoming lines to start, provably 15 extensions at max
18:12.37Corydon76-workUntil VoIP phones are all $20 apiece, it's a still a problem of pick-your-poison
18:13.12ManxPowerCorydon76-home: I'm waiting for phones similar to Cisco 79xx phones to be $100/each
18:13.14h3xwhich will never happen unless they make them with no dsps and no itu algos
18:13.43JerJerh3x: which would be fine
18:13.50marrandysounds like a T-1 and a channel bank.  Adtran 650 ?  (newbie alert) that's what i've heard
18:13.55h3xyeah for asterisk users
18:13.56JerJerfuck the bastards running anything that's not asterisk
18:14.01os2docMarrandy: no, pots analog b1 lines coming in.  I know I willn eed a channel bank to multiplex to a data intefacce into Asterisk box
18:14.25JerJerthey should be punshed for blantely wasted all that money
18:14.26ManxPowerI have no problem leaving out phone systems that do not use codecs like Speex, iLBC, GSM, etc
18:14.42JerJerwaisting
18:15.13JerJerI have faith in wasim and his crew
18:15.43ManxPowerJerJer, Unfortunatly, I don't.
18:15.48os2docI would like to be able to run just one set of cables for phones and network, wich I thought I could do with IP phones.  But with analog phones, I get to run a second wire.  Not a big deal, but cooler iwth no wires.
18:16.07JerJerManxPower: i've already talked to him on his IAX hardfone
18:16.11HeeDwhat would I use to connect 6 analog overlines to asterisk? WOudl I have to get provisioned differently from telco...ie. T1 instead of 6 lines?
18:16.34marrandyprice it out
18:17.03JerJerHeeD: you can start out with a T100P and a adtran or cac channel bank
18:17.09ManxPowerJerJer, Oh, I'm sure he can build the EEKS phone, I just don't know if 1) he can get mfg costs down and 2) release it soon enough for me to care.
18:17.32JerJerheed: then when u get to about 11 lines upgrade to a PRI and flebay the channel bank
18:17.37ManxPowerJerJer, wasim is a smart guy.
18:17.49HeeDjerjer ok thanks
18:17.58JerJerManxPower:  they are going after the barbietone market
18:18.02marrandyListen to JerJer:  he's a * guru
18:18.09h3xyou can get fractional pri's
18:18.16h3xsome clecs have minimum 4 lines
18:18.28JerJerh3x: yeah, sometimes they don't make much sense financially
18:18.30os2docI priced a T1 with the local  telco, 5 pots with hunt, 1 dedicated, 1 fax/DSL line, 280$, T1 890$ starting.  No contest.  the extra 600$ gets me a channel bank on ebay, and the asterisk server.
18:18.35ManxPowerh3x, We are having a line installed today that's PRI with 6 channels along with 384K internet
18:18.36MikeManxPower: in linux i just load the quicknet module and can i use it to conect a phone and give it an extension or just PSTN?
18:18.37torment0r|wk|i wanna use my bonephone.. lol.. the name is silly
18:18.39torment0r|wk|http://iptel.org/products/bonephone/
18:18.43h3xJerJer: well xspedius here in vegas for 4 lines + d channel is like $200/mo
18:18.47ManxPowerMike, I don't know.
18:18.48h3x$100 if its robbed bit
18:19.01h3xmany: thats an even better idea
18:19.09h3xer.  manx
18:19.12ManxPowerI think dog was smileing on me when UPS lost my quicknet order and the vendor no longer stocked them.
18:19.15mrgobytorment0r|wk|:   did you get bonephone to work ???
18:19.23ManxPowerSo I got an X100P and a TDM10B instead
18:19.24JerJerMike:  your first mistake was acutally spending money on quicknet hardware
18:19.26mrgobyi tried but never got it to run
18:19.32torment0r|wk|no.. i just seen it and started to laugh
18:19.46mrgobyit is awful ...  i would stay away
18:20.07ManxPowerMike, Quicknet is NOT well supported with Asterisk.
18:20.29JerJermore correctly:   01Mike, Quicknet is NOT well supported
18:20.36ManxPowerI don't know if that's the fault of the card/drivers (as most people seem to think) or the fault of Asterisk (which other people thing), but regardless it's not well supported.
18:21.20JerJerManxPower: their LTAPI is a joke
18:21.42JerJerManxPower: run valgrind and load chan_phone.so and u'll see for yourself
18:21.52*** join/#asterisk l-fy_ (~diana@home-25022.b.astral.ro)
18:22.19JerJeror hell run any of the ixj sdk sample apps in valgrind
18:24.10torment0r|wk|i have 3 quicknet cards looking at me.. eak!
18:24.19JerJerflebay them, quickly
18:24.54JerJeri've got like 4-5 phonehacks and a linehack or three collecting dust around here somewhere myself
18:24.58]data[_valgrind, uck
18:25.11torment0r|wk|i've been searching today to find an cheap sip phone.. without and luck
18:25.19JerJer7960
18:25.25torment0r|wk|there like $300
18:25.29JerJeryep
18:25.31*** join/#asterisk stan_ (~stan@213.78.71.85)
18:25.38JerJerworth every penny
18:25.52torment0r|wk|i'd believe it.. but the issue is i'm going to college and i'm poor
18:26.11torment0r|wk|it's bad enough i live off tuna
18:26.12JerJerbarbietone
18:26.18]data[_JerJer: innit :}
18:26.18HeeDmy * box is takin a couple seconds to realize a call is hung-up on FXO card. Is this normal or is there some setting? Does have to do with speed of server (433Mhz Celeron)
18:26.23JerJerahh that's the smell
18:26.30torment0r|wk|yep
18:26.44jtoddAnyone remember the three-finger salute to get a SNOM 200 to reboot without plug-yanking?
18:26.48h3xthats because the LEC is slow
18:26.50]data[_torment0r|wk|: if you want something CHEAP and CHEERFUL, get a grandstream
18:27.04JerJerbarbietone
18:27.25ManxPowerHeeD, That's normal on analog lines.
18:27.25h3xthey have black phones now :PPPPP
18:27.45torment0r|wk|has anyone used one before though
18:27.54JerJerso Jamican barbietones
18:28.02miller7-HeeD: you're in luck cause my analog card does not understand hang up at all :P
18:28.07h3xhaha
18:28.16HeeDmanx ok, still tryin to fix my voicemail issue... I think I may attempt to make an IVR that answers and prompts user to press a key to leave voicemail
18:28.17torment0r|wk|i wanted a pink one
18:28.25HeeDmiller haha
18:28.40*** join/#asterisk glLoadIdentity (~asdfrt@abn139-91.interaktif.net.tr)
18:28.44JerJermiller7-: you need disconnect supervsion on ur line
18:28.49HeeDwhat is voicemail2, maybe that would help me ?
18:28.53miller7-JerJer: what do you mean?
18:29.20torment0r|wk|HeeD, you need to update if you still have voicemail2
18:29.23torment0r|wk|that's so last week
18:29.25JerJermiller7-: find a old school fone...like one that lights up or something powered only by the PSTN voltage
18:29.43JerJerthen make a call and let the far end hang up first, see if the lights go out for a fraction of a second
18:29.45HeeDI am using voicemail, not voicemail2
18:30.16HeeDI d/l latest cvs last week, how do I check quickly if there is newer cvs ?
18:30.21torment0r|wk|you shouldn't have voicemail2 is what i'm getting at.. they replaced voicemail with voicemail2
18:30.35JerJerHeeD: cvs status
18:30.39JerJerin the asterisk/
18:30.54miller7-JerJer: well, the thing is that no matter how much I leave the card does not hang up. This happens in one line, others work fine (using busydetect=yes)
18:31.01marrandy[data[-:  whats your config for the grandstream ?
18:31.23JerJermiller7-: what wakky country u in?
18:31.53miller7-JerJer: the specific telco has problems in the area I am in...
18:31.56HeeDand to update I type cvs update ?
18:32.06JerJerbitch, loudly
18:32.15JerJerHeeD: yeppers
18:32.21miller7-JerJer: mixed hardware from 2 vendors, old hardware etc... very old pop
18:32.23]data[_marrandy: asterisk config?
18:32.32HeeDJerJer, and then just make install again ?
18:32.47JerJermiller7-: show them asterisk
18:32.53bkw_blah
18:32.56JerJertell em u'll deploy it for a free PRI
18:32.58JerJer:)
18:33.15torment0r|wk|HeeD, do a make update
18:33.16]data[_mmm pri
18:33.21torment0r|wk|then make install
18:33.21HeeDok
18:33.30miller7-JerJer: hell, yeah... they are slower than replay... would take them 50 years to browse through first screenshot
18:33.33JerJertorment0r|wk|: make update just updates the cvs date
18:33.43JerJerheed: make install will work
18:34.01HeeDI gonna backup my .conf files just in case :)
18:34.05marrandyanything special on the grandstream setup.  I initially had it going to FWD but have registration errors.  Anything special about settings on the grandstream itself ?
18:34.06JerJermiller7-: run your own copper then
18:34.18miller7-JerJer: :P
18:34.24*** join/#asterisk Santo (~santosh@209.78.110.175)
18:34.53marrandynow I'm going to the asterisk box instead.
18:35.00]data[_marrandy: nope, its a doddle
18:35.00torment0r|wk|does anyone know why i can't get any woman?
18:35.24jtoddtorment0r|wk|: Because you spend too much time on IRC.
18:35.29]data[_torment0r|wk|: tried 'cvs co woman' ?
18:35.33marrandydo you use inband signalling ?
18:35.36torment0r|wk|lolol
18:36.00miller7-JerJer: anyway, I won't use analog, this card was just for meetme tests. I am waiting for kapejod's ISDN card for small deployment before going to PRI
18:37.11marrandyjtodd:  thought you were busy  ;-)
18:37.35torment0r|wk|d00d.. people become unbusy when it's comes to chicks
18:37.40jtoddI am never to busy to slap someone around.
18:37.43jtodd:-)
18:38.20tzangerhahaha
18:38.43marrandyWell if you want to check my configs for security issues, or any other problems, the offer still stands
18:38.43tzanger~seen citats
18:38.45citats is currently on #asterisk.  Has said a total of 10 messages.  Is idling for 1d 23h 52m 38s
18:38.53tzangerwow two days idle
18:38.54]data[_marrandy: dtmf is 'in-audio'
18:39.21*** join/#asterisk sjoep125 (~sjoerd@213-84-218-42.adsl.xs4all.nl)
18:39.46jtoddHas anyone seen their SNOMs (2.02t) recently go into "endless loops" during authentication on any SIP messages?  
18:40.18Connorchange dtmf to info or rfc2833
18:40.33SantoI am having some telnet issue
18:40.38SantoRedhat 7.2 --> telnet localhost 3128  ..... after few seconds I get connection reset by foreign host  ????
18:41.02*** part/#asterisk raulfragoso (~raulfrago@200-204-142-71.dsl.telesp.net.br)
18:41.11Santoany telnet Gurus
18:41.23Santobkw_ you there
18:42.01SantoI want my Voicegenie box to talk to *
18:42.18ManxPowerSanto, Try #linux
18:42.19Santoanyone with VXML and *
18:42.40ManxPowerIsn't 3128 the default Squid proxy port?
18:42.45Santoyes
18:43.02ManxPowerSanto, What does this have to do with Asterisk?
18:43.51Santono Once I set my VoiceGenie box I could use H.323 Gateway -->*-->Voicegenie box
18:44.58SantoVoiceGenie (VXML server) is SIP only Gateway is H.323 Only
18:45.15*** join/#asterisk Keso (~kddand@63.80.216.3)
18:45.27Santoso * has to talk to VXML server
18:46.08SantoManxPower any idea about telnet
18:46.25ManxPowerSanto, No idea.
18:46.40*** join/#asterisk qiu (~qiu@194.102.203.13)
18:47.53Santohmm bkw_ you there
18:48.02HeeDhmm I updated cvs and now sip phone cant login
18:48.11HeeDI notice there is no sip show peers command in new console
18:49.38HeeDdo I have to update libpri and zaptel also maybe ?
18:49.41torment0r|wk|no
18:49.51torment0r|wk|it should just work.. have you stoped and reloaded asterisk?
18:49.54HeeDyes
18:50.00torment0r|wk|bkw_!@#!
18:50.03torment0r|wk|<3
18:50.15UnixDawgBKW is always here
18:50.24UnixDawgthis is nothing new
18:50.58HeeDit is like sip is missing
18:51.27*** join/#asterisk denon (denon@synapse.subneural.net)
18:51.30torment0r|wk|that's odd
18:51.30HeeDfrom /usr/src/asterisk I did.... make update and then make install
18:51.35marrandySanto:  telnet, what's up ?
18:51.37ManxPowerHeeD, try sip show users
18:51.39denonheh .. guess someone likes dell.
18:51.53HeeDmanx, it says no command
18:51.54*** part/#asterisk marrandy (~marrandy@209.216.76.1)
18:52.01HeeDif I type help, there is no sip stuff at all
18:52.04*** join/#asterisk marrandy (~marrandy@209.216.76.1)
18:52.15HeeDmaybe I should try to re-download the full cvs ?
18:52.29UnixDawgsip show peers
18:52.37UnixDawgand sip show regisrty
18:52.46bkw_http://bugs.digium.com/bug_view_page.php?bug_id=0000562
18:52.49bkw_JerJer you see that?
18:52.49UnixDawgsip show clevage
18:53.10HeeDno sip commands are working
18:53.17torment0r|wk|mmm.. clevage
18:53.29ManxPowerHeeD, Looks like chan_sip didn't load then
18:53.49bkw_Clearing connection ip$localhost/14668 reason=EndedByTransportFail
18:53.52bkw_that dont look right
18:54.16HeeDhmm maybe cvs didnt update something... I noticed I was getting verify erros and said it was gonna retry
18:54.46Santobkw_ howdy
18:56.12HeeDit doesnt even recognize command... stop now
18:56.13HeeDhmm
18:56.20ManxPowertry show modules
18:56.24ManxPowerlook for chan_sip
18:56.44HeeDModule               Description                              Use Count
18:56.45HeeDchan_sip.so          Session Initiation Protocol (SIP)        0
18:56.45HeeDchan_iax.so          Inter Asterisk eXchange                  0
18:56.45HeeDres_monitor.so       Call Monitoring Resource                 1
18:56.45HeeDres_indications.so   Indications Configuration                0
18:56.45HeeDres_crypto.so        Cryptographic Digital Signatures         1
18:56.47HeeDres_parking.so       Call Parking Resource                    1
18:56.49HeeDres_adsi.so          ADSI Resource                            1
18:56.51HeeDres_musiconhold.so   Music On Hold Resource                   1
18:56.53HeeDchan_modem_aopen.so  A/Open (Rockwell Chipset) ITU-2 VoiceMod 0
18:56.55HeeDchan_modem.so        Generic Voice Modem Driver               0
18:57.18manyblerg.
18:58.07ManxPowerare you sure you are not typing show sip users and not sip show users
18:58.17HeeDyup
18:59.06HeeDif I type help, it does not list any commands for sip
19:00.51*** join/#asterisk sp3cialk (~alex@d141-82-32.home.cgocable.net)
19:01.55HeeDmaybe I have to restart the server
19:03.35HeeDIm rebootin my linux box now
19:05.06bkw_HeeD its not windows
19:05.39HeeDyes I know, but wasnot workin so I thought I would try that
19:07.02HeeDnow when I type asterisk -r it just hangs after displayin a couple lines grrr
19:07.17ManxPowerHeeD, type asterisk -c
19:09.05HeeDmanx it registers some countries.... then says SIP seeding and name of a sipphone at 192.168.0.41:5060 for 3500
19:09.18HeeDand is stopped there
19:09.24*** join/#asterisk sobol_ (~sobol@router-1.szczecin.tpnet.pl)
19:10.20*** part/#asterisk lucifuge3 (trilluser@pcp05047461pcs.ivylnd01.pa.comcast.net)
19:11.07ManxPowerlooks like something is blocking
19:11.51HeeDyes strange
19:12.55HeeDeven an iax connection wont connect to it
19:12.59denonany of you guys deal at all with those dinky little Dell PE 400SCs?
19:13.02HeeDie. friend's * box
19:13.42Bonbondoes anyone have a spanish number which they can sell / give me? Need it for sip termination.
19:13.43ManxPowerIf SIP is blocking then nothing will work
19:13.53HeeDmaybe I should try to re-download entire cvs and make again
19:14.01HeeDrather than update
19:14.17*** join/#asterisk adam_gafachi (~diddy@69-55-69-130.da.netsville.net)
19:14.25bkw_HeeD that makes no diff
19:14.26ManxPowerIf your original Asterisk is more than 2 weeks old you need to do that as was announced on the mailing list.
19:14.39HeeDahh
19:14.55HeeDmaybe I should join the mailing list? :)
19:14.59ManxPoweryou have to delete the asterisk source code an and other of the modules.
19:15.00bkw_um yes
19:15.09bkw_ManxPower Accually you don't
19:15.13ManxPowerHeeD, I don't help people that I know are not on the mailing list.
19:15.16bkw_it you make update twice it will be fine
19:15.20bkw_thats all I did
19:15.25HeeDmanx, ok Ill join the list now
19:15.33*** join/#asterisk phiberkut (~phiberkut@tele-free-hotspot.netlinkip.com)
19:16.18*** part/#asterisk sjoep125 (~sjoerd@213-84-218-42.adsl.xs4all.nl)
19:16.31t4kis there any known issues w/ echo cancel over tdmoe?
19:16.39t4kfor example... it dosen't work?
19:17.21t4kif I'm doing tdmoe from hostA -> hostB, what host should have echocancel turned on? both?
19:17.50t4kwill echocancel work if it's just turned on, on hostB?
19:18.18*** join/#asterisk defian (ircuser@client80-83-46-147.abo.net2000.ch)
19:18.44defianhi
19:20.46Exomorph_Is there any way to run an agi script when the asterisk starts up and/or stops?
19:20.48bkw_<PROTECTED>
19:20.48bkw_<PROTECTED>
19:20.48bkw_<PROTECTED>
19:20.48bkw_<PROTECTED>
19:20.48bkw_<PROTECTED>
19:20.50bkw_muhahahahah
19:20.54bkw_I love privacyManager
19:21.07bkw_my phone dont ring much at all
19:21.55Corydon76-workIt don't?
19:22.03bkw_nope
19:22.16*** join/#asterisk Vco (~Vco@h24-76-11-125.wp.shawcable.net)
19:22.20Corydon76-work--> doesn't
19:22.30HeeDok, I subscribed to both lists... waitin for confirmation emails
19:22.36bkw_ok english guru.. i'm in oklahoma.. we say don't
19:22.43bkw_:P
19:23.04jsmithbkw_: You can add me... as long as you don't tell anyone :-)
19:23.14bkw_haha online friends list is HUGE
19:23.15HeeDbkw, I did a make clean, make update and make install.... noticed make update didnt do anythin 2nd time I ran
19:23.26bkw_HeeD it usually wont
19:23.26HeeDwaitin for it to compile... fingers crossed
19:23.32jsmithHeeD: It's not supposed to...
19:23.47espenzwhat is the best method to use asterisk and hardware as a telephone server?
19:23.53bkw_but I have seen CVS be crack headed once or twice and it has checked stuff out the second time around
19:23.56espenzi mean, which hardware should I have?
19:24.04bkw_espenz depends on your goal
19:24.19espenzmy goal is to have, lik this:
19:24.22espenzpress one for billing
19:24.27espenzpress two for sales
19:24.46bkw_My last set of friends all got mad at me.. they showed up every single day... ate all my food.. drank all my soda/pop/coke(what ever you call it in your neck of the woods) and I finally got sick of that and told em off
19:24.46espenzand own people answering the telephone
19:25.07bkw_espenz you apparently didn't understand my statement
19:25.17defianespenz: do you live in the US or Europe?
19:25.25espenzEurope
19:25.34bkw_how many lines?
19:25.35defianespenz: in Europe the definitce choice is an ISDN PCI card
19:25.35espenzNorway that is
19:25.47espenzwhy?
19:25.49Corydon76-workya damn oklahomo's...  :-P
19:25.57defianespenz: for example with the AVM c4 you can have upto 4 ISDN lines, meaning 8 simultaneous connections
19:25.57espenzhow could I then transfer calls?
19:26.01bkw_Corydon76-work you got it!
19:26.05*** join/#asterisk denon (denon@synapse.subneural.net)
19:26.19defianespenz: you then either transfer call using a P2P (point-to-point) ISDN bus on the c4, or via SIP to a cheap Grandstream phone
19:26.32defianespenz: I am doing just that on my server (AMD + c4 + SIP phones)
19:26.38espenzhm
19:26.45espenzP2P isdn bus?
19:26.49espenzmean, ordinary isdn card?
19:26.51bkw_I wish ISDN was cheaper here
19:27.02bkw_189/mth for 2B+D
19:27.04espenzisdn is realy cheap in norway
19:27.05espenz:)
19:27.05defianyes, with the c4 you can in theory (never did that) connect a phone and the c4 does the NT
19:27.15defianhere I don't do that I do SIP phones
19:27.20Corydon76-workMove to Tennessee...
19:27.23defianeasier to install, cheaper, works quite well
19:27.48defianthe c4 however is not cheap: if you only have one ISDN line (two simultaneous connectins) maybe a cheaper ISDN card might be better
19:27.50manyhuh? 189USD for dual channel isdn? per month?
19:27.51espenza subscription costs about, 14 dollars pr month
19:28.09espenzand a isdn card, about 28 dollar
19:28.10bkw_many yes from SBC (link extension charge is most of that)
19:28.14HeeDok, that didnt work, so I should just del evertying in my /usr/src/asterisk and re-download the cvs? Should I delete the modules too?
19:28.19Corydon76-work$50/month for ISDN... same cost as two analog lines...
19:28.20manyphew.
19:28.23espenzomg?
19:28.24espenz:P
19:28.32defianmany: anyway in US ISDN is not really ISDN, it depends on the provider
19:28.42defianmany: (compatibility is not assured)
19:28.44defianmany: will all equipment
19:28.45manyyes. 56k isdn sucks. :-P
19:28.45espenzis SBC a provider?
19:28.55Corydon76-workHellSouth here...
19:28.56bkw_DOVBS
19:29.06defianmany: in the US it's not that important since POTS lines have caller-ID
19:29.10defianmany: and are quite cheap
19:29.20defianmany: you don't really need ISDN
19:29.28Corydon76-workbut the TN Public Utility Commission forced BS to give ISDN *anywhere* in TN for cheap
19:29.38manyi like isdn alone for the fact that it transfer the type of the call.
19:29.51manycallerids are on analog lines here, too.
19:29.55manyif you want that.
19:30.54espenzdefian: anyways, i want a simple solution, number to choose type of customerhelp
19:31.14espenzhow can I do that? i have a 2 channels ISDN card, that works fine in asterisk using the i4l-driver
19:31.16defianespenz: so even with an analog voice modem it could (?) work.
19:31.26defianespenz: plus either softphones or SIP phones
19:31.39espenzokey, but where do i connect the SIP ?
19:31.52defianto Ethernet
19:32.26espenzhm
19:32.32espenzcool
19:32.38espenzhow much does a SIP phone cost?
19:33.16ManxPowerespenz, $60 - $1,000
19:33.33high-rezDo you guys usually enable MMX on zaptel interfaces?
19:33.34espenzcheap, where could I order then?
19:33.52ManxPowergrandstream.com I think.
19:34.12*** join/#asterisk raulfragoso (~raulfrago@200-204-142-71.dsl.telesp.net.br)
19:34.13espenzUSA is early with technology
19:34.20*** part/#asterisk raulfragoso (~raulfrago@200-204-142-71.dsl.telesp.net.br)
19:34.24espenzand, sorry my bad english ;-P
19:34.28ManxPowerOr www.sipphone.com or any of a number of sites that carry what people here call the BarbieTone (I don't recalls what the actual name is since I'd NEVER buy one of those gorrible phones)
19:35.13defianBudgeTone
19:35.25defianManxPower: I have one on my desk and it does what it should, especially for the price.
19:35.50ManxPowerdefian, You don't have problems with stuck keys or the phone slideing across the desk?
19:35.56bkw_'SeaQuest DSV' actor Brandis dead at 27
19:36.09*** join/#asterisk ricky1 (~ricky@hoochie.digium.com)
19:36.18defianManxPower: for the sliding I put some specially crafted anti slipping glue, and for the keys, not yet.
19:36.28ricky1hello...
19:36.47HeeDmanx, which mailing list wast that mentioned on, I am checking november archives and cant find... but there are 3 or 4 lists
19:37.20blitzrageoff to work, have a good day all!
19:37.27Corydon76-workManxPower:  those phones do have a bracket on the bottom, so you can bolt them down, if that's a problem
19:37.31blitzrageI love the topic
19:37.43espenzdefian, hm, i dit not find, orderstuff on the grandstream
19:37.43espenzdid
19:37.46blitzrageif someone knows the key mappings of comedian mail, please email me at leif@hacklocalhost.com
19:37.59defianespenz: a customer of me bought it
19:38.07defianespenz: AFAIK it's referred from sipphone.com
19:38.19*** join/#asterisk Beave (~beave@bundy.vistech.net)
19:38.19bkw_hahahahaha
19:38.22Corydon76-workManxPower:  and I haven't noticed the keys sticking on the GS phones... (yes, though, on the SNOM 100)
19:38.22bkw_love the topic
19:38.24Beavehey all.
19:38.53Corydon76-workHey, Beave, where's Wally?
19:39.22BeaveI have a TDM400P,  with only one port enabled.  It appears to be upgradable to 4 ports.  I looked for pricing the upgrades but dont see it on there webpage.  any ideas how much it cost for a upgrade module for the TDM400P?
19:39.36BeaveCorydon76: yeah.. Thats new..  :)
19:39.46Exomorph_I'm making an outgoing call via sip to another provider, and for some reason the asterisk sip stack freezes... :(
19:39.47Corydon76-workCall Digium and hit the button for sales
19:39.54ManxPowerBeave, $99, I think, you have to call Digium direct to order those modules
19:40.04Exomorph_I've just upgraded to the latest cvs and thats when I started getting this problem.
19:40.15Beaveok..  thanks manxpower.
19:40.23ricky1anyone knows where can i get to read on iaxcomm?
19:40.55ManxPowerExomorph_, Did you read the announcement about the CVS changes?
19:40.57*** join/#asterisk G0L3M (~golem@CPE0040f42c4568-CM013529902742.cpe.net.cable.rogers.com)
19:41.15Exomorph_Nope.  What changes?
19:41.28ManxPowerExomorph_, Are you on the mailing list?
19:41.42Bonbonwhat does this mean: -- SIP Seeding '3991' at 192.168.254.189:5060 for 3600
19:41.59Exomorph_ManxPower: Just the Users list.
19:42.02Bonbonasterisk just hangs on it
19:42.05bkw_it seeds the sip registrations from astdb
19:42.09bkw_at the cli
19:42.14bkw_do a database show
19:42.18bkw_you will see them
19:42.30bkw_that way if * crashes you have your registrations still in the db
19:42.40Bonbonbut I don't get "Asterisk Ready"
19:42.50ManxPowerMessage-Id:  <200311131556.hADFuhmT005139@mail.sigmasoft.com>
19:43.14ManxPowerThat's the message that talks about the CVS changes
19:43.23defiancould anyone just check that my IAX connection works?
19:43.38marrandydefian:  On the Grandstream, have you managed to get any of the 'feature' buttons to work ?
19:43.39defian(by dialing my IAXtel/gnophone number 1-700-895-5211)
19:43.56defianmarranty: TRANSFER worls; CALLED works; CALLER works
19:44.00ManxPowerdefian, You can test that by calling your own IAXTel number
19:44.09defianmarrandy: I didn't try the rest
19:44.24bkw_ACK
19:44.29JerJerREJ
19:44.31marrandydefian> really, none of mine work, not even the called and callers
19:44.32bkw_new zaptel changes cause kernel panic
19:44.39marrandyor the redial
19:44.40defianmarrandy: with Asterisk?
19:44.46marrandyor flash
19:44.48defianmarrandy: SEND also works for redialing
19:44.49Bonbonyeah, and asterisk doesn't seem to start with latest cvs
19:44.54defianmarrandy: strange.
19:45.17ManxPowerBonbon, as I understand it that only happens if you didn't read the message about the changes to the CVS server.
19:45.37BonbonManx: oh, obviously not. Which changes/
19:45.38Bonbon?
19:45.50ManxPowerBonbon, Read the message ID...
19:45.52ManxPowerMessage-Id:  <200311131556.hADFuhmT005139@mail.sigmasoft.com>
19:45.58marrandyYou would think that it would save the last number it had dialed, but no
19:46.06ManxPowerBonbon, it was posted to bother the users and developer lists.
19:46.13bkw_GREAT
19:46.29BonbonManx: what was it then?
19:46.41ManxPowerBonbon, read the archives. I'm not your personal assistant
19:46.43defianmarrandy: what firmware version do you have?
19:46.53Bonbonok, ok. But would that cause * not to start?
19:47.17ManxPowerBonbon, I don't know.  I did what the message told me so I don't have those problems
19:47.51espenzdefian: now I have bought a SIP phone ;-)
19:48.00julesmarrandy, is that you martin?
19:48.00defianespenz: can you buy just one?
19:48.06espenzUPS Worldwide Express
19:48.08espenzdefian: yes?
19:48.11marrandyyes.  Hi Jules
19:48.12espenz:)
19:48.12*** join/#asterisk Marlow (~marlow@3ffe:200:1:155:0:0:0:2)
19:48.19espenzSub-Total:    $79.99  
19:48.19espenzShipping & Handling:    $85.46  
19:48.19espenzTax:    $0.00  
19:48.19espenz--------------------------------------------------------------------------------
19:48.19espenzTotal: $165.45  
19:48.29denonman, im gettin damn sick of waiting for the Pioneer A07 drive to come out
19:48.29defianexpensive
19:48.31defianshipping
19:48.33julesmarrandy, I didn't know you were a IVR type ;)
19:48.35denonmaybe I'll just buy an A06 for the house
19:48.48espenzdefian: its norway
19:48.50defianespenz: my customer had to buy two, but it was about the same price (?)
19:48.59defianespenz: an Expensive Country?  but how nice!
19:49.00espenzhuh
19:49.02defianespenz: :)
19:49.05espenz:P
19:49.17espenzwierd shopping, it was on lindows
19:49.17julesmarrandy, drop me an email. I might have some work for you.
19:49.25marrandydefian>:   SEND, there is no SEND button on my phone ???
19:49.42marrandyOohhh...work...yummy
19:49.44defianmarrandy: you probably don't have a BugdeTone-100 then
19:49.56marrandyI have the 102 with the Two ports
19:50.13marrandyGoes and checks
19:50.14defianaha
19:50.15marrandy....
19:50.20defiantwo ports but no send key
19:50.24defiancan't have everything :->
19:50.26Marlowespenz : what was that you ordered ?
19:51.18espenzhttp://www.sipphone.com/images/grandstream.gif
19:51.38Marlowespenz : ah .. are you sure, that there is no dealer in europe for that ?
19:51.50Marlowespenz : i wouldn't order it in the US .
19:52.01espenzhm,
19:52.04t4kdoes echo cancel work over tdmoe?
19:52.05espenz:/
19:52.13JerJerkram: this time it wasn't me who broke CVS  :P
19:52.30ManxPowert4k, Why are you using TDMoE and not IAX2?
19:52.32kramwhat's broken in cvs?
19:52.34*** join/#asterisk Mike (~mike@dsl-200-67-4-11.prodigy.net.mx)
19:52.41high-rezYeah he didn't break cvs, he just inserted broken code.
19:52.41Marlowespenz : jeg vil se, om jeg kan få fingrene i en Cisco i stedet ..
19:52.43high-rez;)
19:52.47t4kManxPower -> uh.
19:53.00t4kdoes echo cancel work over tdmoe?
19:53.04espenzMarlow: Hva mener du?
19:53.09JerJerkram: kernel panic on new zaptel changes (see bkw)
19:53.19ManxPowerkram, People are claiming chan_sip is blocking on startup.  I don't know if it's a real problem or the fact they are updating from a pre cvs layout changes to an after cvs layout changes
19:53.24Marlowespenz : den er noget dyrere .... men også en bedre telefon ..
19:53.28kramjerjer: no, he just didn't unload it right
19:53.34JerJerahh ok
19:53.41espenzMarlow: Sikkert, men jeg skal bare teste det først.
19:53.41marrandydefian>:  it's a 102.  Do you really have a 100 ?  or is it a 101
19:53.47ManxPowert4k, I am not aware of anyone using TDMoE so I don't know the answer to that.
19:53.47HeeDI just downloaded an entire new cvs and it doesnt work... same sip problem
19:53.48Marlowespenz : klart ..
19:53.54espenznår ble du norsk?
19:53.56espenz:p
19:54.00ManxPowerHeeD, you deleted your old source right?
19:54.02Marlowespenz : nej .. dansk :)
19:54.13HeeDyes manx, I moved /usr/src/asterisk to /usr/src/asteriskbak
19:54.15Marlowespenz : men ... i skift er der jo ikke den store forskel ..
19:54.19HeeDand cvs made entire new dir
19:54.26Marlowespenz : faktisk dansker i Sverige :o)
19:54.28espenzMarlow: vet du om en forhandler for SIP i Norge eller europa generelt?
19:54.32JerJerno comprenda ur jive, dig
19:54.42Marlowespenz : jeg søger ....
19:54.51ManxPowerHeeD, Are you using zaptel or pri as well?
19:54.52Marlowespenz : skal nok finde nogen på et tidspunkt  ..
19:54.57high-rezJerJer: Does you use MMX optimizations in zaptel? :)
19:54.58espenzokey, gi meg output hvis du finner noe da ;-)
19:55.00HeeDmanx yes zaptel
19:55.09ManxPowerHeeD, I assume you did the same for zaptel?
19:55.10Marlowespenz : selvfølgeligt ..
19:55.22HeeDno, someone here told me I didnt need to
19:55.28JerJerd'oh
19:55.29HeeDIll try that
19:55.35bkw_man ok... i'm happy now
19:55.36ManxPowerHeeD, I don't know if you need to or not, but try it.
19:55.38defianje ne comprends pas non plus :)
19:55.44ManxPowerIf that doesn't fix it, file a bug report
19:56.04JerJeri'll be brave
19:56.18Marlowdefian / JerJer : :o)
19:57.08marrandydefian>:  it's a 102.  Do you really have a 100 ?  or is it a 101
19:57.18bkw_haha
19:57.32espenzMarlow: men det er uansett litt for sent nå, kontoen har blitt belastet
19:57.41defianMarlow: was that norvegian?
19:57.42Marlowespenz : jup ..
19:58.03Marlowdefian : for espenz norwegian, for me danish .. but they are similar, especially in writing ..
19:58.28defianMarlow: ok
19:58.40defianmarrandy: it's written BudgeTon-100
19:58.46defianmarrandy: it's written BudgeTone-100 (sorry)
19:58.49bkw_we know
19:58.51JerJerBarbieTone
19:58.53bkw_or BarbieTone
19:58.57bkw_or CrapStream
19:59.00JerJerGMTA
19:59.09bkw_JerJer yeppers
20:00.14bkw_asterisk*CLI> stop now
20:00.15bkw_No such command 'stop' (type 'help' for help)
20:00.16bkw_haha
20:00.32Marlowhmmm ...
20:00.34Marlowbkw_ : panic mode ?
20:00.37HeeDI tried updating zaptel and reloaded modules, asterisk still has problem with sip
20:01.00bkw_ACK
20:01.06JerJerREJ
20:01.06JerJerHeeD: you have any host names in sip.conf that aren't resolving
20:01.08bkw_broken cvs
20:01.24HeeDbks I get same error.... that stop now wont work either
20:01.53HeeDjerjer not that I know of
20:02.06JerJermy slow ass home box is chewing on the compile now
20:02.27Exomorph_For those that need the URL for the cvs compile problem(s)...  Go to:
20:02.27Exomorph_http://lists.digium.com/pipermail/asterisk-users/2003-November/027050.html
20:02.40bkw_HeeD ack.. evil thing
20:02.47defianbkw_: works very well. (GS)
20:03.12JerJeroddness...mine hangs on chan_iax.so (like it does when gethostbyname() is blocking)
20:03.19HeeDhmm and I didn't backup my old cvs... have to get from friend later
20:03.31JerJerHeeD: cvs update -D '1 day ago'
20:03.36HeeDoh
20:04.33marrandyGot to go. Catch you all later.  Thanks for the info.  Regards and best wishes !!!
20:05.09HeeDjerjer ok I typed that, make install again... see what happens
20:05.46JerJeryeah thre is defainatly a problem
20:06.14Exomorph_JerJer: With the current changes?
20:06.21Exomorph_err cvs I mean.
20:06.41JerJeryes, cvs head
20:07.12bkw_I compiled lastnight and it was fine
20:07.23bkw_after we did all the bug fixes.. just to double check it
20:07.36mrgobyis dynExtenDB on the asterisk FTP site ????
20:08.11Marlowmrgoby : it's here: http://andreasotto.net/asterisk/
20:08.18JerJermrgoby: don't bother...its not the right way to deal with dynamic extensions
20:08.22Exomorph_JerJer  I'm compiling everything from scratch right now and installing to see if that fixes the problem.
20:08.36mrgobyJerJer:  what is ??
20:08.49mrgobyMarlow:   the site is down
20:08.54JerJernot an asterisk application
20:09.01jsharpAny recommendations on a SIP client for MacOS X?
20:09.09Marlowmrgoby : not here ..
20:09.22mrgobyMarlow:  really....hhhhmmmmmmmmmmmm
20:09.31mishehuoh where oh where has my cisco 7960 gone...  oh where oh where can it be...
20:09.33MarlowJerJer : better suggestion to solve that then ?
20:09.43mrgobyJerJer: yes please elaborate
20:09.52mrgobyi find this very intereseting
20:10.04JerJerMarlow:  see retrieve_extensions_from_mysql.pl
20:10.23MarlowJerJer : i'll have a look at that ..
20:10.40JerJerthat's a decent way.. we do something sorta simular
20:11.06*** join/#asterisk os2doc (~michael@65.115.136.98)
20:11.11jsmithmishehu: I didn't steal it!
20:11.48bkw_ok cvs is fixed
20:12.05mishehujsmith: well, smoebody must have, becuase it's not arrived from shido yet...
20:12.09Exomorph_bkw: What did you do to get it working?
20:12.23jsmithmishehu: I've only got 40 or so of them right now :-)
20:12.26bkw_Exomorph_ kram just fixed it. cvs update again
20:12.38Exomorph_Ok
20:12.38HeeDjerjer that 1 day ago worked :)
20:12.47bkw_HeeD todays cvs will work now too!
20:12.53bkw_asterisk*CLI> show version
20:12.54bkw_Asterisk CVS-11/21/03-14:10:15 built by root@asterisk on a i686 running Linux
20:12.57Exomorph_HeeD: You can use the latest cvs now... its fixed.
20:13.01HeeDoh haa
20:13.07*** part/#asterisk stan (~stan@213.78.71.85)
20:13.08bkw_Heed things move fast around here
20:13.13HeeD:)
20:13.14mrgobyJerJer:  where can i find the ....mysql.pl
20:13.18mrgoby?
20:13.20os2docIs there any way to attach one of the big fancy multiline phones to the asterisk system, like one of the receptionist phones.  It seems that all I can find are propriatary.  
20:13.23mishehujsmith: care to donate one?  ;-)
20:14.02jsmithmishehu: Uh, no.  They're all in production, and you'd have to pry them from my cold dead fingers... Everyone here *loves* them.
20:14.12mishehuos2doc: a voip phone doesn't need multiple physical lines
20:14.20mrgobyi just googled it and only found a message from Uriel asking you ( it said Jeremy) the same thing
20:14.32mrgoby:-D
20:14.39MarlowJerJer : but as far as i see, you write the config from the table to a static file here ..
20:14.48MarlowJerJer : and reload asterisk ..
20:14.54JerJerand the problem is?
20:15.12Mucklis the name in the [brackets] important for the registration of a SIP client or just for internal use in the extensions.conf?
20:15.21MarlowJerJer : tjaeh .. not a big one .. but it ain't nice ..
20:15.37JerJer?! we send multi-megabit text config files at our class 4/5 switches
20:15.44JerJerand issue a reload command
20:15.50os2docmishehu: yes, that seems the only option.  It seems that my phone system if I use asterisk is going to be analog phones with my employees having to punch in the codes, with the receptionist with the large phone, or maybe a computer terminal phone.  Thats a though.
20:15.51MarlowJerJer : what do you have against using the sql table directly ?
20:16.01*** join/#asterisk pawpa (~bob@216.253.86.210)
20:16.04bkw_Marlow stability
20:16.08pawpahi
20:16.13JerJerwhat happens WHEN your db goes down?
20:16.14bkw_why put more chances of instability into *
20:16.20JerJernot if
20:16.23JerJerWHEN
20:16.33mrgobyJerJer:   nevermind  i think i found it
20:17.19Marlowi know, that it can be an issue ..
20:17.33Marlowso the question is creating the possibilty of an fallback ...
20:17.52Marlows/an/a/
20:18.11os2docmishehu: an IP phone doesn't need multiple lines running to it, but what about access to multiple lines, like all 6 incoming pots lines, intercom, etc, without having to type in codes.
20:18.24JerJerMarlow: it is all in proper design of the system
20:18.28mrgobyJerJer:   is that the only reason for doing it the other way??  just because MySQL is not dependable to be running always?
20:18.42JerJerNuFone runs on a totally databased method of config and we have zero trouble
20:18.51espenzMarlow: hvordan fungerer det når jeg får telefonen? Setter den opp med IP, så den treffer burken min.. Men skal jeg bruke asterisk som SIP-gw på en måte?
20:18.55mrgobyi understand that point, don't get me wrong
20:18.57mishehuos2doc: not sure about a good phone yet...  I'm waiting for a cisco 7960 to arrive
20:18.58JerJermrgoby:  01[15:14] <bkw_> why put more chances of instability into *
20:19.06espenzSlik at andre SIP telefoner kan tryke f.eks "2" for å komme til en telefon
20:19.13Marlowespenz : jep .. det er helt rigtigt ..
20:19.14torment0r|wk|me no hablo espano
20:19.17pawpaJerjer: i thought you said not to use a DB because you are at the mercy of it's availability
20:19.18mrgobyi see
20:19.18espenz(må seff) sette opp det også
20:19.26Marlowespenz : fungerer på samme måde, som med en softphone ..
20:19.28espenzMarlow: men er det instillinger for SIP gw?
20:19.28JerJerpawpa: that's not what i said
20:19.31espenzpå telefonen
20:19.36kapejodoh, the vikings again ;)
20:19.40espenzburde vel være det
20:19.41espenz;P
20:19.41torment0r|wk|hola?
20:20.09*** join/#asterisk okrumm (~okrumm@dsl-200-95-104-110.prodigy.net.mx)
20:20.15pawpajerjer: i thought you were saying that it isn't a good idea to use a DB (mysql) because you have to maintain an extra server
20:20.28JerJerum no...where'd u read that?
20:20.32espenzMarlow: har du satt opp noe opplegg hjemme som funker?
20:20.48pawpahemm let's see
20:21.00Marlowespenz : jeg har ikke en hardphone, men jeg har * med softphone kørende ..
20:21.01okrummAnyone has a Windows SIP phone working?
20:21.07JerJerwana bring a Cisco Call Manager to its knees:  stop the MSSQL service
20:21.14espenzMarlow: kan jeg få teste?
20:21.15Marlowkapejod : wir können auch anders ..
20:21.17jsmithJerJer: Yup... :-)
20:21.20JerJeror simply DoS it
20:21.21Marlowespenz : klart ..
20:21.24espenztenkte på innringer nr med routing til softphone?
20:21.27mishehuokrumm: a number of ppl have gotten xlite to work.
20:21.35jsmithJerJer: Lemme guess... it install MSSQL with a blank "sa" password
20:21.41torment0r|wk|is there anything that the cisco call manager does that asterisk can't do?
20:21.46pawpajerjer: i have heard that  cdr_mysql does not build correct CDR's...true?
20:21.51kapejodMarlow: lol
20:21.58jsmithtorment0r|wk|: Cost tons of money?  Crash often?  Run on windows?
20:22.05torment0r|wk|well i know that
20:22.10jsmith:-)
20:22.10JerJerpawpa: absolutely...incomplete and implemented poorly
20:22.11torment0r|wk|but.. i've never seen one in action
20:22.27JerJertorment0r|wk|: you don't want to
20:22.45torment0r|wk|is it like compairing IIS to Apache kinda thing
20:22.50pawpajerjer: are there modules available that generate correct CDRs or is the some thing i need to write?
20:22.56bkw_I love this
20:22.56bkw_asterisk*CLI> show uptime
20:22.57bkw_System uptime: 3 weeks, 22 hours, 35 minutes, 3 seconds
20:22.57bkw_Last reload: 2 weeks, 1 hour, 19 minutes, 29 seconds
20:23.03bkw_not bad eh?
20:23.09bkw_thats the production one
20:23.27JerJerpawpa: i have written my own entire real-time rating engine, not just a cdr inserter
20:23.31mishehubkw_: uhm...  aren't those supposed to match?  ;-)
20:23.39bkw_mishehu nope
20:23.41JerJermishehu: nope
20:24.01pawpajerjer: reasonable undertaking or daunting?
20:24.02Marlowdu mener sip ?
20:24.29mishehu*CLI> show uptime
20:24.30mishehuSystem uptime: 2 weeks, 3 days, 11 hours, 40 minutes, 57 seconds
20:24.54HeeDyup works now.... CVS-11/21/03-12:08:31
20:24.54mishehunot bad for mine, considering the fact that I keep changing shit in it.  ;-)
20:24.57bkw_mishehu do a reload and it will show las realod
20:25.19JerJergood lordy... two small charectors in sched.c blew up everythign :)
20:25.43JerJer-     ast_mutex_lock(&con->lock);
20:25.43JerJer+     ast_mutex_unlock(&con->lock);
20:25.47*** join/#asterisk TK (~sudsboy@53.int43.dsl.garlic.net)
20:26.50pawpajerjer: never heard of a AAA oriented object called a rating engine...where does the definition come from?
20:26.51bkw_JerJer thats usually the case
20:26.54mrgobyJerJer:   so do these implementations of the dynamic configurations and extensions that you referenced, do they put the configuration into a flat conf file and then just reload?
20:27.01mishehugarlic.net....  the dsl provider that smells bad
20:27.18bkw_RADIUS... let the fight begine!
20:27.20bkw_er begin!
20:27.29bkw_Radius Kombat!
20:27.39bkw_oh nice bug in radius today.. did you guys see it?
20:27.51Marlowbkw_ : radius aint bad ..
20:27.54Exomorph_JerJer: THATS MY BUG!  THATS MY BUG!   Sweet... Now I won't have to reboot the linux box agian! :)
20:27.55mrgobyi find adding dynamicism to * very intriguing
20:28.30JerJerpawpa: don't sware at me
20:28.46pawpahuh?
20:28.57JerJerRADIUS will never be implemented in ANY VoIP operaiton I have control over
20:29.43Connor-I still don't see what you all have against radius
20:29.44Marlowno .. but not everybody shares that point ..
20:30.07pawpajerjer: i can see plainly how authentication and authorization would be simple, but how do you send the time elapsed to the script after the hangup
20:30.14MarlowJerJer : nobody forces you ..
20:30.33JerJerwhy add another layer of potential failure?
20:30.50JerJerthere are many different methods to talk to a databas without involving the complexitities of RADIUS
20:31.14MarlowJerJer: because you can add scalability ... and radius is implemented in many places allready ..
20:31.21MarlowJerJer: use of existing ressources ..
20:31.45Connor-That's all well and good.. Radius makes cross platform easier too. You ever try to talk to a MSSQL from linux?
20:31.50JerJerand as you scale your operation not only do you have to worry abuot keeping your database online (and available) you have to also worry about your RADIUS server(s).
20:32.15MarlowJerJer: who says, that i talk to a database server behind radius ?
20:32.21HeeDwhat I done movin my /usr/src/asterisk dir to a backup, will that affect my g.729 licenses in any way... asterisk says I still licensed
20:32.29JerJerMarlow: by default... because some lazy Cisco developer implemented it into their boxes
20:32.29sizzzungConnor-: have you ever tried to talk to mssql from linux?
20:32.33sizzzungConnor-: it seems not
20:32.39JerJersizzzung: simple
20:32.40MarlowJerJer: most radius work from plain textfiles, but you would have a central repository for multiple boxes ..
20:32.41sizzzungConnor-: because you can do it. quite easily.
20:32.44sizzzungConnor-: FreeTDS
20:32.54MarlowJerJer: cisco is not the default ..
20:32.54pawpafreetds?
20:32.56HeeDie. does the Registration do somethin ... cuz they say it has to be run from /usr/src/asterisk initially
20:32.56JerJerMarlow: like that will scale
20:32.59Connor-Last time I tried it was a pain in the arse
20:33.17sizzzungi was doing it 3 years ago easily.
20:33.31MarlowJerJer: nope ... it will not .. but you are not required to talk to a database .. there are other possibilities ..
20:33.52sizzzungThis point release fixes some memory leaks.  (3 August 2003)
20:33.57kapejodflat files!!! ;)
20:33.58sizzzungi guess they got it to the point where it's perfect!
20:34.04pawpaif not radius, how can you pass accounting infor (ie time elased) back your accounting script for the sql insert
20:34.15JerJerpawpa: there is no script
20:34.29JerJerAsterisk has built in facilities for dealing with accounting
20:34.30Marlowwhy shouldn't anybody have the possibility to choose for themselves ? ..
20:34.34sizzzunghrmm
20:34.36sizzzungdamn it
20:34.38pawpathe acct flag?
20:34.42sizzzungwhen will my T1 be in?
20:34.57JerJerMarlow:  by adding all the complexities of RADIUS?  i think not
20:35.02JerJercdr_csv.so
20:35.05JerJerthere is your flat file
20:35.12pawpaok
20:35.14kapejodyeeehaaa ;)
20:35.16JerJereven seperated out by account code
20:35.17Connor-JerJer too bad those facilites doen't talk to my accounting system, which DOES talk to radius.
20:35.28MarlowJerJer: nobody says, that you will have to use radius .. others allready base their business on it ..
20:35.31pawpaok...politically correct CDRs?
20:35.44*** join/#asterisk point (~litw@195.161.106.222)
20:35.56JerJerMarlow: time will tell how smart of a business decsion that will be
20:36.12sizzzungheh
20:36.22MarlowJerJer: might .. might not ..
20:36.24JerJeri've been there and seen how RADIUS fails with VoIP
20:36.30MarlowJerJer: it's still the freedom of choice ..
20:36.33sizzzunghow many real telecom companies uses RADIUS for accounting?
20:36.39sizzzungMarlow: it's a freedom to make dumb decisions.
20:36.46JerJersizzzung: right on
20:36.56Connor-How many ISP's use it for Dialup, DSL and other things?
20:36.57mrgobyOUCH
20:36.58sizzzungJerJer: high five, one owner to another.
20:37.01Marlowwhy should it be a dumb decision, when it has been used for account for years ..
20:37.10sizzzungMarlow: obviously you're not pushing enough traffic.
20:37.16JerJerConnor-: which is EXACTLY what radius was designed for
20:37.20sizzzungwait till you're pushing millions of CDRs a month.
20:37.34sizzzungyou're going to use RADIUS for that?
20:37.36JerJersizzzung: or even a few hundered a second
20:37.39Marlowfor modem-dialup
20:37.39Marlowwhere is the difference ?
20:37.45Connor-Let see, I track dialup time via radius, hmm.. What's the damn difference?
20:37.50JerJera whole hell of a lot
20:37.50bkw_we do not use radius for dialu
20:37.50Marlowit simply depends on what setup you want to archieve ..
20:37.50sizzzungMarlow: there are a lot more records
20:37.50bkw_p
20:37.54sizzzungyou guys have no fucking experience
20:37.56bkw_er DSL
20:37.58sizzzungjesus fucking christ
20:38.01sizzzungi swear to god
20:38.01bkw_we use radius for dialup.. but not DSL
20:38.09pawpawhoa
20:38.16pawpai believe you
20:38.16pawpa:)
20:38.18sizzzungmost of you are fucking nerds sitting in your mom's basement trying to start a carrier with a linux box and PRI
20:38.22JerJersizzzung:  i feel your pain
20:38.24pawpahahaha
20:38.26pawpahah
20:38.26pawpaha
20:38.32Marlowsizzzung : no . we have other demands ..
20:38.34sizzzungnote, i said most, and not all.
20:38.38pawpahah
20:38.38pawpahya
20:38.42pawparotfl
20:38.43sizzzungMarlow: demands like pushing CDR's to a RADIUS box?
20:39.08sizzzungso much lack of clue.
20:39.13kapejodsizzzung: i dont even have a pri :(
20:39.16Connor-I'm not really wanting it for CDR's as much as replacing sip.conf using radius attributes
20:39.18Marlowsizzzung : that would be for integration in the current billing system, instead of developing a new one ..
20:39.23JerJerRADIUS is for sheep
20:39.33sizzzungMarlow: then your current billing system sucks. rewrite it, or buy mine.
20:39.46JerJerConnor-: you most certianly do not need radius to do that
20:39.47pawpasizzung: i will HAPPILY peruse any documentation on an alternate setup for voip AAA...can you direct me?
20:40.01pawpasizzzung: how much for your solution?
20:40.07JerJerpawpa: it doesn't exist in Asterisk
20:40.08sizzzunghaven't decided on a price.
20:40.15sizzzungi'm thinking $125-$200k
20:40.26Connor-No, But sinze my freaking accounting package can handle that, it makes things damn easy.
20:40.26sizzzungmattering how much more mature it gets in the next two months
20:40.35Marlowi'm not saying radius is the solution .. but it is one way to go .. especially because many base their business on it allready ..
20:40.42pawpajerjer: so i must roll the solution in house, eh?
20:40.42sizzzungokay. do this.
20:40.51JerJerpawpa: not necessarily
20:40.56sizzzungcreate a perl pre-parser that'll parse the csv cdrs
20:40.58Connor-granted, I probably could talk directly to the SQL server, reading those same damn attrbitues
20:41.02sizzzungor runs through the sql server
20:41.02ManxPowerDoes anyone know what message waiting indication Zaptel analog FXS cards use?  FXS, 90v, 48v?
20:41.07sizzzunggenerate the summaries
20:41.11sizzzungand drop the summaries into RADIUS
20:41.15sizzzungthere are plenty of radius perl modules.
20:41.16pointJerJer, I have fixed my main problem with chan_h323 ... only after installation of new ethereal :) - the option noh245tunneling does not work for outgoing calls at last without gk ..
20:41.20Marlowsizzzung : a billing system is not easy replaced ... if it's integrated with other stuff ..
20:41.52JerJerMarlow: who is saying replace anything?  he just gave you the way to hook asterisk into your existing crap
20:41.58Connor-btw, I WANT TO KNOW. WTF do you think RADIUS FAILS with VoIP ??
20:42.06sizzzungConnor-: TOO MUCH TRAFFIC!
20:42.16JerJerVendor Specific Attributes
20:42.19MarlowJerJer : i had a lag here .. got the last 15-20 messages in a bunch
20:42.24JerJerimproperly impleemnted records
20:42.32MarlowJerJer: after i wrote ..
20:42.36sizzzungi'm just blaming traffic.
20:42.45Connor-Bullshit!
20:43.06Connor-If your damn radius can't handle the traffic, then it's not scalled correctly or done correctly.
20:43.19kapejodor spelled correctly
20:43.35JerJerConnor-: so your saying buy two high end servers instead of one?
20:43.49JerJerfor each time you need to scale up
20:43.59JerJerwhere is the financial sense there?
20:44.02Connor-JerJer, You better have 2 if you want to have failover/reduntant server.
20:44.12JerJerlol no..doesn't work like that
20:44.17ManxPowerI figure that is RADIUS support was so important someone would have written a RADIUS logger module for Asterisk already.
20:44.17dantbut if 1 can't handle the load, then you need 3
20:44.19kapejodbut you dont want to have 4
20:44.31pawpasizzzung:  using radius on the back end for it's accounting abilities?
20:44.40pawpasizzung: jerjer: > what about prepaid?
20:44.51JerJerpawpa: we do pre-paid all day long
20:44.54pawpado you use the cdr_csv the same way?>
20:44.58JerJerno
20:45.01]data[_cdr_flow 0wnz me
20:45.02]data[_:]
20:45.34Connor-look, the main reason, like I said, that I want radius, is for provisioning new VoIP customers.
20:45.50JerJerConnor-: and i'm asking why?
20:45.59JerJerits a blantent waste of resources
20:46.02]data[_the whole configfile structure of asterisk is a mess imo :-)
20:46.02ManxPowerDoes *does* RADIUS send a "disconnect" message when a dialup user has exceeded their monthly allowance?
20:46.12Connor-Because, My freaking billing/accounting solution makes it easy to do.
20:46.21ManxPowerSeems to me that's close to pre-paid voiuce.
20:46.23torment0r|wk|there's really no point in using radius for VoIP..
20:46.41]data[_there are entire radius billing platforms out there you can buy off the shelf
20:46.41ManxPowertorment0r|wk|, There are lots of reasons to use RADIUS for VoIP billing. 8-)
20:46.43]data[_thats the point
20:46.48JerJerConnor:  so your solution can set specific codecs on a per user/peer basis?
20:46.48*** join/#asterisk tim27 (doug25@229-29.dr.cgocable.ca)
20:46.56tim27hello everyone :)
20:46.58torment0r|wk|ManxPower, it's not accurate
20:46.59pawpajerjer: sizzung's recommendations are very straightforward, but you mentioned that I wouldn't have to roll the solution myself...were you meaning that i wouldn't have to write it completely because there are some helper modules available?
20:47.10rusty_jerjer - what are the readily available alternatives to radius, then?
20:47.17Marlowtorment0r|wk| : wrong .. Connor- gave you one ..
20:47.26torment0r|wk|i work for an isp.. trust me.. my time and the time of Qwest never matches
20:47.29ManxPowertorment0r|wk|, *shrug*  That's not my problem, that the problem of someone that wants to use RADIUS.
20:47.48torment0r|wk|and it always leads to problems
20:47.49JerJertorment0r|wk|:  sync to a stratum 2 time source
20:47.51Connor-JerJer: I can set any damn attribute I want via radius Attributes as long as I have them defined for Vender specific.
20:48.02Exomorph_wb karm
20:48.05JerJerConnor: i think not
20:48.21Connor-Oh?
20:48.23mrgobypawpaw:  that is a good question   Jer???
20:48.24ManxPowerkram: What message waiting indication does Zaptel analog FXS cards use?  FXS, 90v, 48v?
20:48.45rusty_JerJer: what are the readily available alternatives to RADIUS?
20:48.51JerJerrusty_: i've let bkw_ play with a first public version of my nufone_rating.so
20:48.55tim27i read on the web that many people get echo problem when mixing FXO phone line and ip phone, any know solution to this prob... (some phone better than other ??? ) and there is a way to get a isdn for SOHO (BRI) ...
20:49.11*** join/#asterisk Marlow_ (marlow@as2-6-3.tbg.s.bonet.se)
20:49.16pawpajerjer: how much for your rating engine?
20:49.17rusty_JerJer: last I heard that was being kept a secret
20:49.38rusty_JerJer: and it sounds like it's going to be a commercial release (as opposed to OS)?
20:50.05JerJerrusty_: if your rating calls that means you are going to be billing customers
20:50.11JerJerand if your billing customers you are making money
20:50.22rusty_JerJer: and?
20:50.27pawpajerjer: i am interested in your solution, but see no info on nufone
20:50.35bkw_nufone_rating.so rocks... very nice
20:50.43Connor-JerJer: Why do you say I can't added custom Radius attributes for codec's?
20:50.48*** join/#asterisk ToyMan (~stuq@user-0cevdks.cable.mindspring.com)
20:51.08JerJerI have worked out a non-gpl license with Digum, which will help with the continued the development of Asterisk
20:51.18mrgobyyes...  i'd be interested too jer
20:51.22pawpaso licensing is through digium
20:51.23pawpa?
20:51.29rusty_JerJer: so it's going to be a commercial license?
20:51.48JerJerConnor-:  there is only so much u can do with a VSA
20:52.20JerJerwhy you think those billing software packages out there cost so fsckin much?
20:52.20defiancu )
20:52.26pbxtechits everywhere you want to be
20:52.32tim27any got a clue :)
20:52.34rusty_JerJer: because they can
20:52.40rusty_JerJer: it's a niche market
20:52.48pbxtechthere are open source billing software
20:52.52JerJerlolololol
20:52.57JerJerpbxtech:  ok
20:53.00pawpawhat about freeside?
20:53.09*** join/#asterisk stonefly (~trillian@toby.stoneflytech.com)
20:53.10JerJeror tarbas
20:53.26pawpadoesn't seem like trabas even has any SW there
20:53.56pawpaif you are dropping cdrs into radius via perl, couldn't you use freeside to maintain your accounts?
20:53.59rusty_JerJer: so I'm still not clear on what kind of licensing you intend to have for your rating engine
20:54.11pointwho use trabas ?
20:54.24torment0r|wk|what exactly does your rating engine do.. just keep track of billing?
20:54.38]data[_it rates the calls :)
20:54.39bkw_torment0r|wk| you know that bill you get from the phone comany?
20:54.40JerJerrusty_:  non-gpl...you will recieve the shared object an example config and some support
20:54.49tim27anyknow about echo ... ???
20:54.50bkw_with all the calls list and the prices.. you need a rating engine to do that
20:54.51rusty_JerJer: cost?
20:54.59Connor-JerJer: I see nothing in sip.conf that couldn't be done with a VSA...
20:55.03*** join/#asterisk FryGuy (~fryguy@c-24-2-50-122.client.comcast.net)
20:55.07JerJerConnor-: then do it
20:55.13torment0r|wk|bkw_, someone couldn't do that with the cdr_myql db
20:55.18]data[_i'm writing my own one for fun too btw :)
20:55.19mrgobywhat about echo tim?
20:55.44bkw_torment0r|wk| if you think you can rely on that.. go for it
20:55.48torment0r|wk|i'd be interested in it though
20:55.49JerJerConnor: and when you hit that brick wall you'll wish you would have listened to someone that has been there already
20:55.55rusty_JerJer: doesn't nuphone use asterisk?
20:55.57tim27i want to purchase equipment, is that true that i will get echo on SIP phone trought FXO phone line ???
20:55.58bkw_torment0r|wk| but you can't do prepaid.. and such with it.
20:56.05JerJernufone is 100% asterisk
20:56.06dantJerJer, can I have the rating thing for free so I can bill the gf?? ;)
20:56.18rusty_JerJer: gpl software
20:56.22pawpasizzzung: are you using off the shelf solution for billing once the cdrs are in radius?
20:56.26tim27on the webforum, many people complain about echo
20:56.31JerJercodec_g729 is not gpl
20:56.39rusty_JerJer: right...so?
20:57.04JerJerso what?
20:57.20*** join/#asterisk Padre (Padre@67.60.50.146)
20:57.33mrgobyyou are saying that your system just works with open source, and does not contain open source, right Jer?
20:57.55mrgobyor GPL rather
20:57.58bkw_You can do prepaid. postpaid.. any paid.. in any currency with nufone_rating.so
20:58.14pawpahow much is it?
20:58.16JerJermrgoby: nufone and digium have worked out a non-gpl license...
20:58.23pawpais digium licensing it at themoment?
20:58.25rusty_mrgoby: yet his entire business is run on OS software (except for the codecs)
20:58.44rusty_not just OS, GPL
20:58.46pawpasounds like an efficient model to me
20:59.02mrgobysure
20:59.06pawpayou can't feel bad about making money with OS
20:59.08*** join/#asterisk sleepy_ (~sleepy@80.96.155.10)
20:59.12mrgobyno
20:59.12pawpathat's what itis there for
20:59.23pawpai'm sure jerjer contributes in many ways
20:59.40pawpaanswering 5 * 10^1200 questions a day is one way
20:59.43bkw_um
20:59.49mrgobyPirates, all a ya !!!  pinko commies !!!!
20:59.49bkw_chan_sip keeps losing my sip registrations
20:59.51bkw_strange
20:59.54JerJerrusty_: show me the paragraph that says I can't make money using GPL
21:00.10sleepy_hello all
21:00.17Connor-bkw_ you sure it isn't GS firmware? Had that problem already
21:00.19mrgobyhowdy sleepster
21:00.22dantJerJer, I'm sure Microsoft have some papers on it ;)
21:00.25rusty_JerJer: I realize that you can...show the paragraph that says I can't critize your method of making money
21:00.27pawpajerjer: i'll probably be licensing your engine until i can get my own solution ready
21:00.30tim27mrgoby, they said :Some changes were implemented around Oct 27th that improved the echo
21:00.30tim27problem. To use that, put "echotraining=yes" in zapata.conf for each
21:00.30tim27x100p card installed. As I understand this, it changed the way the
21:00.30tim27echo canceller samples the analog line. Using the parameter is far
21:00.30tim27better then previous, but there can still be some echo involved in
21:00.31tim27some cases.
21:00.46bkw_Connor- i'm talkin about my Cisco
21:00.53kapejodactually JerJer could have done it also without paying for a non-gpl license from digium
21:01.15mrgobyanyone wanna help tim27   ???  not I said the fly
21:01.16bkw_kapejod how?
21:01.20bkw_kapejod do tell!
21:01.26Connor-bkw_ not good
21:01.26kapejodbkw_: read the gpl
21:01.32Exomorph_JerJer: Ok... New problem with the lastest cvs...  When it does a restart on the B-Channel's, (pri's) asterisk hangs.
21:01.38rusty_I'm sure that jerjer's engine will be very useful for some people
21:01.44torment0r|wk|JerJer, happen to have any documentation on exactly what it does.. i'd be interested to get something like that
21:01.51sleepy_i tried to install asterisk on my computer with my gvc external modem and it can't be recognized
21:01.54stoneflytim27, does echotrainning=yes apply to T100p cards hooked up to a channel bank too?
21:02.02Marlowthe GPL is not about free beer, it is about the freedom of Source ..
21:02.02sleepy_can someone help me?
21:02.12JerJertorment0r|wk|: no, i'm still working on the code still... as i get feedback from the few ppl that are currently using it
21:02.16PadreFree BEER? HUH HUH!
21:02.23PadreWho's buying?
21:02.28manybeer
21:02.29ManxPowersleepy_, That is correct.  You will have hack the asterisk source code to make it work with a voicemodem.  Even then it won't work very well
21:02.30manygood one.
21:02.38manyits 10pm, good time for a beer.
21:02.46pawpa2 pm, here
21:02.47Marlowi've got mine allready ..
21:02.51Marlowbefore i came home ..
21:02.52pawpastill sounds like a good idea, though
21:02.54rusty_But there is a constant schism between Qt vs GTK camps for the same reason.
21:03.02rusty_GPL vs non-GPL
21:03.06Padrehey I am in the west coast only 1pm here time for a beer lunch brake
21:03.09Marlowrusty_ : nah .. that's another issue ..
21:03.12Connor-SO... If someone DID want to write a radius module for *, where would be good place to start?
21:03.13torment0r|wk|bbiab
21:03.21Marlowrusty_ : and has nothing to do with making money of source ..
21:03.27rusty_Marlow - no, it's not. One's proprietary, and one isnt.
21:03.30pawpasomeone is GPLing one at the end of the year
21:03.31JerJerConnor:  good luck
21:03.38Marlowrusty_ : yes .. but it's not about money .
21:03.38ManxPowerConnor: cdr_mysql or cdr_csv
21:03.49Connor-not talking for cdr's
21:03.57pawpawell, cdr_mysql is messed up
21:03.59Connor-for configing sip.conf
21:04.02rusty_Marlow: I was never debating money - jerjer brought that up
21:04.16ManxPowerConnor: oh.  put it in ast_load then any of the config files could use it.
21:04.33pawpaast_load?
21:04.39rusty_Marlow: I was debating the principle.
21:04.47ManxPowerast_load is the function that loads and parses the asterisk config files.
21:04.54Marlowrusty_ : of course .. but that is his freedom to choose ..
21:05.00ManxPowerat least I *think* it's called ast_load
21:05.02pawpaso put sip.conf in ast_load...don't follow
21:05.04Marlowrusty_ : as is yours to do it different ..
21:05.12rusty_Marlow: yes, of course
21:05.31*** join/#asterisk ssokol (~ssokol@64-151-38-185-dhcp-kc.everestkc.net)
21:05.32Marlowrusty_ : he has released the h323 chan driver, if i understood that right .. ... so he contributes ..
21:05.36ManxPowerpawpa, Connor was asking where the best place to put RADIUS support was, I suggested he patch the ast_load function in the Asterisk source code.
21:05.50pawpamanx: sorry
21:05.53Connor-as there any sort of dynamic config file stuff * out yet?
21:05.53Marlowrusty_ : but nobody forces him to release everything, just because he uses OS
21:06.02JerJercan't forget about chan_skinny
21:06.26ManxPowerConnor: No, but if you put RADIUS support in the asterisk config file parser....
21:06.54Connor-Manx: That's the plan.
21:07.01bkw_anyone else notice chan_sip just stops responding?
21:07.13ManxPowerOf course then you would need to modify Asterisk to reload it's configs when some event happens
21:07.17Exomorph_bkw: Me... I'm looking into it now.
21:07.45pawpasizzung: you aroung?
21:07.55MarlowManxPower : does dynextendb do that ?
21:08.10Connor-Manx: or setup a event schedule... is there a way to force a reload on individuale conf files?
21:08.10MarlowManxPower : reload on any event ?
21:08.10pawpai heard extendb is not the best way for dynamic extensions
21:08.38ManxPowerMarlow, I have no idea.  I am not aware of anyone using that external software
21:08.53*** join/#asterisk jgaviria (~jgaviria@63.245.86.109)
21:09.17pawpajerjer said that extendb was not the best way, but didn't mention a better way
21:09.28JerJerpawpa i certainaly did
21:09.33pawpasorry
21:09.37pawpawhat was that?
21:09.41pawpamissed that
21:09.53Marlowpawpa : yes .. but he didn't say way ... beyond that he doesn't think radius is the way either ..
21:10.02ManxPowerI always thought "JerJer's way" was "modify the config files and issue reloads automatically"
21:10.09JerJerretrieve_extensions_from_mysql.pl
21:10.17pawpatx!!
21:10.19pawpayou da man
21:10.40pawpaeverybody: here this............retrieve_extensions_from_mysql.p is the way
21:10.44pawpa:^))
21:10.50pawpapawpa happy
21:10.51rusty_I think radius will be fine for some installations - not everything has to scale to huge proportions.
21:10.58JerJerManxPower:  nufone runs on a database
21:11.02JerJerconfig
21:11.25MarlowJerJer: so when you add something .. when is it active ?
21:11.34pawpaimmediately i'd guess
21:11.42pawpait is dynamic
21:11.47pawpa?
21:11.57JerJerdynamic enough
21:12.06pawpaheh
21:12.07mrgobyJerJer: what does that mean?
21:12.12Marlowdoes still not answer my question ..
21:12.18ManxPowerJerJer, Yes, but the database writes the config files and then issues a reload doesn't it?
21:12.27Connor-So where IS retrieve_extensions_from_mysql.pl ?
21:12.30mrgobyis that how it works?
21:12.37JerJerManxPower: not directly, no
21:12.44pawpamrgoby: i believe itmeans you can modify and maintain your extensions via php or something and have dynamic control of them vs. a static file
21:12.49mrgobyyeah....asterisk.gnuinter.net/files/digium/asterisk-ng/retrieve_extensions_from_mysql.pl
21:12.55dantConnor, /usr/src/asterisk/retrieve_extensions_from_mysql.pl
21:13.03ManxPower./asterisk/messages-expire.pl
21:13.03JerJermrgoby: how about /path/to/asterisk/
21:13.03ManxPower./asterisk/retrieve_extensions_from_mysql.pl
21:13.03ManxPower./asterisk/retrieve_sip_conf_from_mysql.pl
21:13.03mrgobyheheh
21:13.27mrgobyJerJer:   sorry, google was closer
21:13.29mrgoby:-D
21:13.35MarlowJerJer : how often do you write the configfiles and reload ?
21:13.48JerJerMarlow: doesn't work like that
21:13.57*** join/#asterisk bobman (~bobman@mube.psouth.net)
21:13.59MarlowJerJer: but ?
21:14.10pawpajerjer: clarification needed....does retrieve_extensions load it once or lookit up like an AGI script...like extendb
21:14.26JerJerpawpa: its not a AGI script, no
21:14.28ManxPowerpawpa, Try reading the file.
21:14.36pawpaok
21:14.37pawpagood idea
21:14.40pawpano, great idea
21:14.42JerJeryeah, do i smell a RTFM bomb again?
21:14.49pawpadamnit, guys
21:14.51pawpadon't do that
21:14.52mrgobyhehehe
21:14.55pawpajeez
21:14.58JerJerlast time i released one i brought down freenode
21:14.59pawpai'm sorry
21:15.06pawpai'm soorrryyyyyyyyyyy
21:15.15mrgobygrovel pawpaw
21:15.23mrgobybow down like the newbie you are
21:15.26pawpai just want to make money on GPL's software....i will RTFM again and again
21:15.33*** join/#asterisk MamboKing (~mambo@66.207.107.50)
21:15.45MamboKingwhat's up with ma bell?
21:15.56pawpahalitosis
21:15.58pawpasp?
21:16.45MamboKinganyways, anyone developed pageing via ADSI yet?
21:16.52mrgobyheheh   kram changed the topic to that at my suggestion :-)   i think it is funny
21:17.05Connor-What do those perl scripts do, make a new conf file everytime something is changed and issue a reload?
21:17.07bkw_access-list 101 deny   ip any 64.254.234.0 0.0.0.255
21:17.08bkw_access-list 101 deny   ip any 65.49.50.0 0.0.0.255
21:17.10bkw_does tha tlook right?
21:17.15pawpais there a repository of contrib files for asterisk
21:17.27manybkw: depends on whaddya want :-P
21:17.30Connor-bkw_ don't for get the permit ip any any at the end
21:17.39bkw_Connor- um no
21:17.44bkw_you never permit ip any any
21:17.49bkw_you boob
21:17.52pawpaheh
21:17.56bkw_you only permit things that are destin for your network
21:18.15Connor-bkw_ umm you do. Maybe not for what your doing.. But you do.
21:18.27pawpau oh
21:18.33bkw_no you don't you only allow your network in.. unless you like having traffic on your network that shouldn't be there.
21:18.35dantConnor, when doing logging? :)
21:18.38pawpawell what is it?...you do or you don't
21:18.43bkw_anyway
21:18.44bkw_next
21:18.51pawpaso you do?
21:18.51bkw_I wanna block those IPs from my network
21:18.57jensdhow would you reload asterisk as best way - through the manager interface or through shell? (in a perl script)
21:19.08bkw_asterisk -r
21:19.10bkw_type relaod at the cli
21:19.16bkw_asterisk -rx reload
21:19.16MamboKingthrough the cli is better
21:19.26Connor-then you need to have access-list 101 permit ip any any at the end of it as well.
21:19.40Connor-the deny takes precedence, then the permit allows everything else.
21:19.50jensdwhy not through manager interface? (when there is this nice perl manager interface :-P)
21:19.55Connor-once it's denied, it breaks out of the access list.
21:19.56bkw_Connor- yes I know this...
21:20.16MamboKingbkw_: do you know if kram has had a chance to look at that bug yet? (487)
21:20.40bkw_not yet
21:20.52MamboKingawesome
21:21.00MamboKingkram: how's it going?
21:21.25Connor-Do I need to send in a disclaimer for that little patch I did?
21:21.58kramsurviging at the moment
21:22.15MamboKinggood to read
21:22.24*** join/#asterisk Vco (~Vco@h24-76-11-125.wp.shawcable.net)
21:22.25izoanyone here has ETSI ISDN specification ?
21:22.30rozohey kram, what kind of cam is that in your office? The one that's taking 180 degree pictures
21:23.48MamboKingkram: a colegue of mine was talking to you about enableing pageing from phone to phone via ADSI some time ago.. you had mentioned that it was possible but it would require development
21:24.07MamboKingwhat kind of cost do you think would be involved in such an undertaking?
21:24.26jetswhats up gents
21:25.22*** join/#asterisk voipguy (~voipguy@hoochie.digium.com)
21:27.45JerJerrozo: i think its homemade (if memory services me :)
21:28.11kapejodizo: www.etsi.org
21:28.18MamboKinganyone here done any poking around with ADSI?
21:29.51izokapejod : i cant find it there :-( maybe you have downloaded ?
21:30.20izokapejod: someone told me today that SUSPEND ISDN message is not available in PRI ETSI specification
21:30.29kapejodizo: click on "get a standard", sign up and download them for free
21:30.47Marlowkapejod : what is the hfcdummy for ?
21:31.03rusty_JerJer: how would you quantify the difference between the way that RADIUS scales (or not), and the way that your rating engine would scale?
21:31.09kapejodizo: yes that makes sense
21:31.15kapejodMarlow: it generates zaptel timing out of a hfc-s pci a isdn card
21:31.29JerJerrusty_: my solution is intergrated into asterisk
21:31.35izokapejod: well I've seen it working :
21:31.56Marlowkapejod : from the isdn network ?
21:31.57rusty_JerJer: that doesn't tell me much.
21:32.08JerJerrusty_: and doesn't had a whole sepearate protocol worth of overhead just to rate calls and instert them into a DB
21:32.08kapejodizo: it depends on the switch i'd say
21:32.25atacommhey jets, how goes it
21:32.28pawpathere is also a company that was listed on the voip-info list ( Buena Vista Communications ) and they have a nice cheap AAA system for asterisk, although they have told me that this release is not scalable...meant more for call center applications and whatnot...soho/smo
21:32.29Marlowkapejod : or does  the chipset provide that ?
21:32.32rusty_JerJer: how does that translate into actual numbers?
21:32.36izokapejod: u r probably right thats why I want to check ETSI standards
21:32.40kapejodMarlow: no network needed, just the card
21:33.05kapejodizo: suspend is not a mandatory message
21:33.27rusty_JerJer: I'm a PHB...help me understand why I'd choose your engine over a free RADIUS solution.
21:34.03JerJerrusty_: the radius solution isn't free
21:34.03JerJerthre is no radius hooks into asterisk
21:34.12izoJerJer there is
21:34.15izobut not published
21:34.21izothere was posts on mailingilst
21:34.25pawpayes i saw that on google
21:34.27JerJerthat was a troll
21:34.30izonaah
21:34.33pawpatroll?
21:34.35JerJervaporware
21:34.36izojust contact those people
21:34.41rusty_hell just grep the source
21:34.41izoand pay him a little
21:34.43JerJerno need
21:35.01izoJerJer did you try to contact them ?
21:35.10rusty_JerJer: so you're saying that there's no way to use radius with asterisk?
21:35.11JerJeri have a little bird
21:35.16JerJerthat told me
21:35.18pawpabtw, that same company with the AAA system has a radius solution rolling out soon
21:35.22tzangerso in an answering dialplan (s,1,Answer) what is the timeout it waits for a response...  is that the DigitTimeout?  
21:35.23pawpabut it isn't open
21:35.45JerJerlooks like the AstGod has bestowed another fix into the cvs
21:35.47Exomorph_bkw_: cvs update to fix your sip problem.
21:35.57pawpabut it is cheap
21:36.11bkw_Exomorph_ just checking
21:36.15bkw_Stealth_Man wake up
21:36.30pawpathey are the only company besides nufone that i've seen offereing these products
21:37.00JerJerum our solution isn't AAA garbage
21:37.08mrgobypawpaw: who?
21:37.19pawpabuena vista communications
21:37.26Marlowwhat happened to the daily debian builds of the cvs ?
21:37.27pawpawell i've tried it out and it works well
21:37.34pawpafor the application that it targets
21:37.40pawpaVERY intuitive
21:38.02pawpawhich is really nice for me, because the operators of our call centers don't have to know diddly
21:38.17Connor-what ths addi pawpa
21:38.40pawpajerjer:  like i mentioned earlier, though, they told me that it was not meant to scale...only for soho/smo call center type app...does prepaid, too
21:38.48bevinscan I use irq 3 or 4 for pci slots with x110p and tdm400? I know these are for serial ports usually but....cat /proc/interrupts doesn't show these, so I guess they are not used.
21:39.08bevinsasterisk -vvvvr
21:39.16pawpaand they offer hosted billing, too...which can have its advantages
21:39.23JerJerpawpa: why buy something that won't grow with you?
21:39.49pawpajerjer: cost/performance (right tool for the job) i don't need scalability for my applications
21:39.56JerJerlol ok
21:39.57rusty_JerJer: not everyone needs to grow in a way that requires the degree of scaling that you're talking about.
21:40.04pawpajerjer:  why do you say?
21:40.13JerJerfamous last words
21:40.14pawpajerjer: i'm only needing 24 phones at a time
21:40.20JerJerpawpa: today
21:40.20bkw_Exomorph_ seems to fix it
21:40.38pawpai'm not a carrier, jerjer, i'm a callcenter guy...that's what i do
21:40.46JerJerdoesn't matter
21:40.52bevinscan I use irq 3 or 4 for pci slots with x110p and tdm400? I know these are for serial ports usually but....cat /proc/interrupts doesn't show these, so I guess they are not used.
21:40.58JerJerwhat if u land a hell of a deal for 500 phones?
21:41.01MarlowJerJer : not everybody wants to grow endless ..
21:41.04ManxPowerbevins, Try it and see.
21:41.24ManxPowerbevins, Your motherboard will assign the IRQs to the cards.
21:41.24MarlowJerJer : if it doesn't benefit in the long run ..
21:41.39ManxPowerbevins, IF your motherboard does not generate interrupts for devices that are disabled then in theory it should work
21:41.50pawpajerjer:  your point is very well taken, it is just that currently i don't have the capital to get into a more scalable solution and also, the BVC solution is working perfectly fo rme
21:41.50sizzzungyo jerjer.
21:41.59MarlowJerJer : callcenter business is hard .. and keeping people with work isn't easy ..
21:42.05ManxPowerJerJer, Then he will pay for his shortsightedness 8-)
21:42.07sizzzungcellcenter business is a bitch.
21:42.08bevinsManxPower: thats what I am trying not to do because it assigns wcfxo and eth0 the same irq....irq sharing and it has locked up the channel a few times.
21:42.17pawpajerjer: we are very happy with it...i don't even have to train anyone to use it becuase it is so intuitive
21:42.18bkw_Specifically, the law contains: opt-out, authority for the FTC to set up a "Do-Not-SPAM" registry, criminal charges for fraudulent spam, including five years in prison, statutory damages of $2 million for violations, tripled to $6 million for intentional violations, unlimited damages for fraud and abuse.
21:42.19sizzzungyou have to show up at their office with a gun to get them to pay their bill.
21:42.20bkw_woot
21:42.20JerJerManxPower: yep
21:42.42ManxPowerbevins, Yup.  Change the slot the card is in or figure out a way to make the motherboard assign a different IRQ for that slot.
21:42.44pawpasuzzzung: we've never had any problems with call center apps
21:42.49JerJerso much for sleeping today
21:43.01sizzzungpawpa: what kind of call center?
21:43.07ManxPowerbkw_, It'll never pass.
21:43.10pawpapre and post paid  and internet cafe
21:43.28sizzzungahh
21:43.36pawpawe were using quintum
21:43.45bevinsManxPower: I have another box that put wxfxs,wxfxo,eth0 on the same irq....That doesn't work too well
21:44.06pawpabut have switched to asterisk becuase of the cheaper entry costs and the BVC AAA system fits nicely with our needs
21:44.12ManxPowerbevins, IRQ sharing is not supported with Digium cards
21:44.19Marlowi've been working for callcenters starting from 10 agents to over 300 now .
21:44.37bevinsManxPower: I know....thats what I have to tell my motherboard...:-)
21:44.51pawpamarlow: wow
21:45.11ManxPowerbevins, on the motherboards I've used you tell the system to not auto assign IRQs and then it will let me spcify per slot IRQ assignments
21:45.32pawpaanother reason i like the BVC is because they will do hosted billing and handle customer accounts, too
21:45.33Marlowand it's allways a bitch
21:45.51bevinsManxPower: thats why I wanted to know about irq 3 and 4..I will just try it.
21:45.56pawpawhich is great for ease of setting up a new call-center
21:46.10pawpathey will ship a unit preconfigged
21:46.16pawpaand we just plug it in and go
21:46.18ManxPowerbevins, I don't think 3 and 4 are "officially supported" but I suspect it will work.
21:46.21*** join/#asterisk zwi (~chris@216.88.131.43)
21:46.57Marlownot only is it a bitchy job .. you also allways run into a M1
21:47.06bevinsas long as I don't run setserial on those tty's, even better disable onboard serial ports...
21:47.13JerJerpawpa: hosted billing is nothing special... hell we host asterisk based billing for a few of our  highly motivated customers
21:47.33*** join/#asterisk glLoadIdentity (~asdfrt@abn139-91.interaktif.net.tr)
21:47.54ManxPowerI have a question!  /etc/zaptel.conf what signaling should I use for a PRI?
21:48.13pawpajerjer: i know it is nothing special i was just saying that I like BVC and that it is a service they offer
21:48.16ManxPowerpri_cpe is only listed in /etc/asterisk/zapata.conf not in /etc/zaptel.conf
21:48.25pawpajerjer:  it is nice for my situation
21:48.34pawpajerjer: and they are very reasonable
21:48.35tzangerI have 12 lines in a trunk group...  how do I dial that group?
21:48.44tzanger(i.e. so it picks up any available in the group)
21:49.34JerJerpawpa: then run with it
21:49.39bkw_lalllalalal
21:50.38*** join/#asterisk anachron (~fradooj@adsl-64-169-89-27.dsl.snfc21.pacbell.net)
21:52.23Marlowbkw_ : what have you eaten ?
21:52.39jets801 805 6034
21:52.45bkw_Marlow just being me
21:53.28jsharpAnyone available to help me with some PRI problems?
21:53.30zeta_jets, eh?
21:53.31Marlowjets : and ? what happens if i dial that no. ? :o)
21:53.54jetsyou get tt-monkeys
21:53.56jetsits awesome
21:54.18tzangerwhat is the difference between 'group' and 'callgroup' in zapata.conf?
21:54.20zeta_heh
21:54.46zeta_Does anyone know if it is possible to destroy a meetme bridge, or to kick users out of the meetme conference room?
21:55.04Marlowehe ... i've got tt-allbusy, before you get to my VM
21:55.16Marlow<-- jets
21:56.32jetshuh?
21:57.17JerJerzeta_: sorta... i've been playing with some manager hooks for app_meetme
21:57.24Marlowjets : i just found it too amusing, when i heard that the first time on 612@fwd
21:58.04jsharpExtension '8005095639' in context 'incomingpri' from '008307519917' does not exist.  Rejecting call ecting call on channel 1, span 4
21:58.16jsharpExcept the extension does exist in the context.
22:00.35bkw_does it now?
22:01.02tzangerbah
22:01.04tzangeryou guys are right
22:01.11tzangerno far end disconnect on the channel bank I
22:01.49*** join/#asterisk af_ (af@ip314-35-1.adsl.edisontel.com)
22:01.52jsmithtzanger: Yup... use an Adit 600 for FXO ports...
22:02.08tzangerjsmith: yup
22:02.12tzangerhow do I kill the channels?
22:02.18tzangercan I tell * for forcibly hang them up?
22:02.22Marlowanyone an idea, what codec the fwdnet 0800 gateway requires now ?
22:02.23tzangershow applications
22:02.23jsmithsoft hangup Zap/1
22:02.26tzangerhaha wrong one
22:02.28tzangerer window
22:03.44tzangerjsmith: thanks
22:03.58tzangerdamn liars...  Carrier Access says CB1 does Far-end discon
22:04.00tzangeroh well
22:04.13]data[_ooh er
22:04.21]data[_* Run-Time Check Failure #0 - The value of ESP was not properly saved across a function call.  This is usually a result of calling a function declared with one calling convention with a function pointer declared with a different calling convention.
22:04.47tzangerlater all tiem to pick up the kids
22:06.51*** join/#asterisk woodsy (~woodsy@64.6.35.237)
22:08.58zeta_JerJer, have you had any success at it?
22:10.20JerJerwho what when where why?
22:10.37zeta_JerJer, with kicking users out of a conference/channel?
22:11.10outtoluncseems to me, last time i viewed app_meetme.c the 'admin menu' was a bit lacking
22:11.16zeta_heh yeah
22:11.18*** part/#asterisk mrgoby (~mrgoby@pcp05304587pcs.wanarb01.mi.comcast.net)
22:11.29JerJerouttolunc: yeah thats a stubb for someone else to implement
22:11.32JerJerbut its not needed
22:11.37outtoluncah
22:11.41zeta_I'm trying to find out if I have to code in support into the meetme source to actually accomplish this task
22:11.45JerJer_EAGLE_'s agi hook is ABSOTIVELY KILLER!
22:12.06pawpawhat agi hook?
22:12.16JerJershow application meetme
22:12.16outtoluncyeah cough it up <G>
22:13.07*** join/#asterisk espenz (espen@a217-118-51-166.bluecom.no)
22:13.13espenzyeh, yo
22:13.16outtoluncah
22:14.04pbxtechwhere is his agi app at?
22:14.25*** join/#asterisk FredtheBadger (~fred@128.113.36.123)
22:14.38FredtheBadgerhi, I have a question about asterisk and DynExtenDB
22:15.21Marlowhmm .. skal lige se, om jensd også kender danske el. svenske .. han er jo dansk ..
22:15.28JerJerhaven't we beat that dead horse enough ?
22:15.34FredtheBadgerI need database support for extensions (unless there's an easy way to deal with a list of extensions that changes rediculously often in large quantities), but I've heard a lot of bad things after searching
22:15.51FredtheBadgerI figured it was a dead horse, but I still need info on it :)
22:16.05JerJerits a hack
22:16.23pbxtechJerJer, is eagles agi hook public?
22:16.30FredtheBadgeris there a better way to do it?
22:16.36FredtheBadgerI'm assuming the answer's yes
22:17.52JerJeryes,  vi extensions.conf  
22:18.07FredtheBadger?
22:18.39JerJerpbxtech: show application meetme
22:18.54JerJerfor the lazy
22:18.55JerJer<PROTECTED>
22:18.55JerJer<PROTECTED>
22:19.37bkw_Windows the Internet Whore.. catches more viruses and worms than a french whore in pheonix
22:19.55JerJerthe zaptel only problem can be fixed, with proper motivation
22:20.17bkw_well fix it
22:20.18bkw_:)
22:20.24bkw_hehe
22:20.24JerJerhence proper motivation
22:20.46JerJerit works absolutely flawlessly for me
22:20.50JerJercurrently
22:21.20mishehuare there many french whores in phoenix?
22:21.37sizzzungjerjer.
22:21.47sizzzungJerJer: when are we going to party?
22:21.53sizzzungwhen are you coming down to south florida?
22:22.31JerJerwhen a ticket shows up here :)
22:22.47sizzzungplz
22:24.12JerJernow if u could provide some clean, geek friendly female units to party with, that might encurage me to jump in the C172 and head down
22:24.54pbxtechjerjer, did shido9 ever talk to you about being able to record a conference call?
22:25.06bkw_its possible
22:25.18bkw_and easy if you stop and think about it
22:25.23JerJerthe recording it not hard at all
22:25.38bkw_nope
22:25.42bkw_I record all calls
22:25.47pbxtechjust recording then mixing right
22:25.53JerJerits how you plan to manage those recordings after the fact is what gets tricky
22:25.57JerJeror can get
22:26.12pbxtechhe never got back to me on a price to get something built
22:26.36pbxtechcant you have some monitor extension dial into it an record off 1 line?
22:26.39pbxtechor something
22:27.23bkw_pbxtech you can.. or setup the conf admin as the monitor
22:27.34bkw_once they enter.. record them
22:27.50*** join/#asterisk indiam (~sipjic@81-86-244-189.dsl.pipex.com)
22:28.01pbxtechi just need something build and I was willing to pay to get it done
22:28.36*** join/#asterisk Spaceboy (furgaw@205.124.232.209)
22:28.39indiami have a specific question to use asterisk only as a User Agent to Register to SIP
22:28.48indiamcan anybody help me ?
22:29.55*** join/#asterisk RichA (~Rich@vegas.routers.com)
22:30.22indiamis nobody in this channel ??
22:30.39bkw_indiam no you just don't barge in and start asking questions.. thats rude
22:30.43bkw_say hi
22:30.44bkw_or something
22:30.47bkw_jesus
22:30.48indiamhi
22:30.50indiamyes
22:30.54indiamsorry about that
22:30.59bkw_you want to change the user agent on outbound sip registrations right?
22:31.06bkw_if so then you need to edit the src code
22:31.14*** join/#asterisk bevins (~bob@modemcable197.44-202-24.mc.videotron.ca)
22:32.07indiamI want to just use it as a SIP UA
22:32.36drgalaxyI keep getting "ast_load_resource): /usr/lib/asterisk/modules/chan_skinny.so: undefined symbol: ast_pickup_call" when I try to load module chan_skinny.so
22:32.54drgalaxyanyone have any ideas?
22:34.17Corydon76-workYou forgot to load res_parking.so first
22:34.23SpaceboyI need some help with installing the h323 channel driver is there any one here that has done it before?
22:34.55JerJerSpaceboy: depends on which one your talking about
22:35.03Corydon76-workSpaceboy:  follow the instructions in the README *to* *the* *letter*
22:35.34JerJeryeah what Corydon76-work said
22:38.07JerJerdata[Zzz]:  whimp
22:38.33JerJerive been up almost two days straight now
22:38.59mishehuhmm...
22:39.12mishehuI wonder if The Precursors would sound good as music on hold
22:39.28JerJeri like russian-monkeys myself
22:40.27bkw_JerJer
22:41.40bkw_show me on the doll where the bad man thouched you.
22:42.28mishehuwas the bad man who thouched you named michael jackson?
22:42.37bkw_haha
22:42.42bkw_mishehu I was thinking along those lines
22:43.59mishehubkw_: criminal minds think alike
22:44.15bkw_i'm not criminal
22:44.20bkw_doesn't matter what my record says
22:44.27bkw_I'M NOT NOT NOT NOT NOT
22:46.15gage_manHey anyone Know max CALL-waiting callers
22:46.19gage_manI only have 2
22:47.18JerJer1-517-679-7772  <--- russian monkeys
22:47.54sizzzungi stink
22:47.55mishehumy record says nothing...  I stopped buy lp's when I was...  hm...  probably 10...
22:48.07*** join/#asterisk Apophis (~btatton@209.180.83.2)
22:48.15sizzzungJerJer: gvive me your phone number
22:48.46JerJer1-517-679-7772
22:48.49JerJerlol
22:48.55sizzzungdamn it
22:49.09sizzzungseriously.
22:49.11jetsawesome asterisk is awesome
22:49.13sizzzunglet's talk.
22:49.16sizzzungi have questions.
22:49.38tim27any know here if i will get echo using a SIP phone to make or receive call from PTSN
22:49.47sizzzungtim27: probably not.
22:49.50sizzzungbut anything is possible.
22:50.28mishehuJerJer: I hope the russian monkeys weren't the products of soviet nuclear research
22:50.29mishehuheh
22:50.51JerJerlol
22:51.04stoneflyPOT lines hooked up to a channel bank are very frustrating!!! One call it is too quiet, another call it's too echo'y another call there is feedback... I wish frac T1's were cheaper here!
22:51.17tim27sizzung : i read on the web many post complaining about echo ... on sip phone... why ???
22:51.36sizzzungtim27: because people suck.
22:51.37jsharpbleh.  I can't get this PRI problem straightened out.
22:51.44tim27lol
22:52.08tim27i have only 3 phone line... so i can only affort ... analog...
22:52.33stoneflytim27, same here...
22:52.58tim27stonefly: you use fxo with sip phone ... or with analog phone...
22:56.42*** join/#asterisk daork (~daork@202.89.35.252)
23:01.48*** part/#asterisk Exomorph_ (Greg@134-9.bvcompuworks.com)
23:02.22*** join/#asterisk cfo (~cfo@194.19.190.217)
23:15.01*** join/#asterisk xeno42 (~xeno42@wtf.cx)
23:16.15*** part/#asterisk stonefly (~trillian@toby.stoneflytech.com)
23:16.24*** join/#asterisk stonefly (~trillian@toby.stoneflytech.com)
23:17.22*** join/#asterisk Exomorph_ (Greg@134-9.bvcompuworks.com)
23:19.23sizzzunghaha jerjer.
23:19.27sizzzunggizzzone.
23:21.02*** join/#asterisk juhas (juhas@hot.juhas.net)
23:22.42juhasI'm having a problem with sync sources on a TE410P, could anybody help? Mainly, zttool tells that sync source is "internal" even when it should be the line itself.
23:23.01JerJeryeah gdb tends to cause a lil blocking
23:23.31*** join/#asterisk okrumm (~okrumm@dsl-200-95-104-110.prodigy.net.mx)
23:24.25jsmithJerJer: Do you happen to have a CVS server we could use for this documentation stuff?
23:25.04dougheckahttp://www.st.cs.uni-sb.de/askigor/
23:25.06dougheckainteresting
23:25.40izojuhas when you change /etc/zaptel.conf do you do ztcfg -vvv
23:25.49ManxPowerHmm.  Took me 5 mins to make my Asterisk server work after the PRI was installed.
23:29.21jsharpCome fix my PRI, then.
23:31.24*** join/#asterisk espenz (espen@a217-118-51-166.bluecom.no)
23:31.40*** join/#asterisk fyman (~fyman@CPE-138-130-18-16.nsw.bigpond.net.au)
23:33.08*** join/#asterisk cfo (~cfo@194.19.190.217)
23:33.17*** join/#asterisk pbx_noob (~pbx_noobi@hoochie.digium.com)
23:33.29*** join/#asterisk anthm (~anthmct@CPE-24-167-193-161.wi.rr.com)
23:35.12ManxPowerjsharp, They way you fix your PRI is to go call up a good CLEC 8-)
23:35.20indiamjsharp he said it is only for his asterisk server, not for yours dear
23:35.23ManxPowerjsharp, what's the span= line in your zaptel.conf?
23:37.43jtoddCan accountcode be set from within the dialplan?  I see that it's frustratingly only set in the channel configs...
23:38.55ManxPowerjtodd, yes
23:39.11jtoddSo, I can SetVar(ACCOUNTCODE)=blah   ?
23:39.22ManxPowerexten => 91411,1,SetAccount(${CALLERIDNUM})
23:39.38jtoddCool cool.  I'll try...
23:40.51jetsany way to import the "show channels" in to an agi script?
23:41.40*** join/#asterisk Exomorph_ (Greg@134-9.bvcompuworks.com)
23:42.00ManxPowerjsharp, what's the span= line in your zaptel.conf?
23:44.39zoaany asterisk internals for rent ?
23:44.46pbx_noobdoes anyone have a link to good doc for CLI?
23:44.51jsmithzoa: Rent?
23:45.03jsmithpbx_noob: Hit tab in the CLI...
23:45.20pbx_noobthanks
23:45.25jsmithpbx_noob: For example, type "iax [TAB]"
23:46.13zoayes i need someone to take it up the ...
23:47.24*** part/#asterisk tim27 (doug25@229-29.dr.cgocable.ca)
23:47.54ManxPowercontext = incoming
23:47.54ManxPowersignalling = pri_cpe
23:47.54ManxPowergroup = 1-4
23:47.54ManxPowerchannel => 1-4
23:47.59ManxPowerthe group = is correct?
23:49.19jtoddManxPower: Nope, no luck.  ACCOUNTCODE is not settable.
23:49.40ManxPowerjtodd, Maybe not on your asterisk, but it is on mine, using the line I pasted.
23:50.23jtoddAh... sorry, I mis-read your line.  I'll try again.  (getting sloppy - I want to get the hell out of the office but I keep getting new jobs piled on me as I reach for my bags...)
23:50.41jsmithjtodd: I know that feeling!
23:52.15*** join/#asterisk fmany (~fyman@CPE-138-130-18-16.nsw.bigpond.net.au)
23:52.18*** join/#asterisk Conro (~rolko@pcp01843985pcs.lncstr01.pa.comcast.net)
23:52.28ConroHey, I'm in need of some help :)
23:52.40ConroIf anyone is available to help an oldbie turned newbie, PM me plz =)
23:52.56Conro*cries*
23:52.59jsmithConro: Just ask your question...
23:53.03ConroAlright, thanks :)
23:53.09ConroI need to know how to register myself a channel on this server
23:53.17ConroI'm going to be using to to host a meeting for a website I work for :)
23:53.20ConroI forget how =p lol
23:53.36Conro#nexusatlas
23:53.51jsmithjust type "/join nexusatlas"
23:53.59jsmithAnd you'll be in that channel
23:54.09ConroDoesn't exist, I'll need to create :)
23:54.12ConroHow do I do that? lol =p
23:54.14*** join/#asterisk bkw__ (~brian@hoochie.digium.com)
23:54.21jsmithjust type "/join nexusatlas" and it'll create it
23:54.26*** join/#asterisk dasenjo (~dasenjo@200.21.83.174)
23:54.41Conro-
23:54.41Conronexusatlas That channel doesn't exist
23:54.59bkw_what evah
23:55.03Conroerr, forgot to put the # in front of it :)
23:55.07ConroIt's created now :)
23:57.40bkw_blah blahb blah
23:57.52*** join/#asterisk bjohnson (~bjohnson@ip114-165.tor.istop.com)
23:58.08*** join/#asterisk forkenb (~forkenb@dailup-219-90-63-226.appscorp.net)
23:59.16sizzzungyou ain't noting but a hoochie momma

Generated by irclog2html.pl by Jeff Waugh - find it at freshmeat.net! Modified by Tim Riker to work with blootbot logs, split per channel, etc.