| 00:06.37 | *** join/#asterisk snewpy (~markl@203-173-30-79.dyn.iinet.net.au) |
| 00:10.07 | stylee | voip is supposed to be free shitheads |
| 00:10.33 | stylee | fucking commies |
| 00:10.35 | tessier | Since when? |
| 00:10.36 | sxpert | lol |
| 00:10.39 | jcdutton | In the * demo extensions.conf file, it let's one leave a voicemail message. What number can one dial to play back the recorded messages ? |
| 00:12.21 | *** join/#asterisk gafachi (~adam@roc-66-66-46-141.rochester.rr.com) |
| 00:12.26 | *** join/#asterisk angler (~angler@24.214.255.57) |
| 00:13.50 | svanlund | jcdutton: Don't know if any extension is setup in the demo for that, but otherwise you can setup your own and call the VoiceMailMain app. |
| 00:18.49 | *** join/#asterisk neondog420 (~icechat5@dialup-4.235.141.240.Dial1.Orlando1.Level3.net) |
| 00:19.10 | *** join/#asterisk bobman (~bobman@68-169-204-41.agstme.adelphia.net) |
| 00:20.09 | neondog420 | Hi all. I think its ironic looking for asterisk help on irc :) |
| 00:21.11 | svanlund | And the ironic thing is? ;-) |
| 00:21.57 | neondog420 | that we hack internet telephony and should just finish the call :P |
| 00:24.47 | ManxPower | ARG! Anyone here know why my X Server isn't listening on port 6000 but everything works locally? |
| 00:25.05 | ManxPower | xhost + doesn't let run remote X clients |
| 00:25.25 | neondog420 | firewalled most likely. |
| 00:25.29 | matobago | i'm tring to call a sip phone and system tell me "Unable to create channel of type 'SIP'..... |
| 00:25.36 | matobago | i'm tring to call a sip phone and system tell me "Unable to create channel of type 'SIP'..... |
| 00:25.40 | timecop | whats the most usable lunix IAX client |
| 00:25.48 | ManxPower | neondog420: Nope. iptables and ipchains isn't even installed. |
| 00:25.50 | timecop | one that DOES NOT SUCK |
| 00:26.39 | sxpert | timecop: asterisk with a hardware phone ? |
| 00:29.12 | SplasWORK | Request to schedule in the past?!?! |
| 00:29.19 | SplasWORK | Does anyone know what that means and what causes it? |
| 00:29.20 | *** join/#asterisk ManxPower (~eric@dsl-208-164-150-160.datasync.com) |
| 00:30.20 | *** join/#asterisk ManxPower (~eric@dsl-208-164-150-160.datasync.com) |
| 00:30.28 | matobago | i'm tring to call a sip phone and system tell me "Unable to create channel of type 'SIP'..... |
| 00:30.55 | matobago | anyone knos why??? |
| 00:32.12 | Crippled | Sip |
| 00:33.38 | Crippled | do you have SIP or Sip |
| 00:33.53 | *** join/#asterisk chobbs (~chobbs@andor.dreamhost.com) |
| 00:34.03 | matobago | SIP |
| 00:34.37 | chobbs | OK, I've just installed a T400 into my asterisk box, and successfullt loaded modules. I've plugged it into an extension on our existing PBX - what do I need to do to allow SIP phones to make outbound calls through it? |
| 00:34.37 | Crippled | hmm. I woder if it matters. It's listed as Sip, Zap, IAX, IAX2, Phone, |
| 00:35.09 | Crippled | caps usually matters in unix, but I dunno matobago |
| 00:35.27 | matobago | mm ok |
| 00:35.46 | chobbs | I fibbed - it's a TDM10B, not a T400 :/ |
| 00:36.40 | Crippled | have you read the handbook chobbs? |
| 00:36.45 | Crippled | ~handbook |
| 00:36.46 | jbot | i heard handbook is http://www.digium.com/handbook-draft.pdf |
| 00:37.09 | neondog420 | I've got a cisco IP phone I wanna convert to SIP..any suggestions? |
| 00:37.37 | Crippled | neondog: get a sip image and setup a tftp server on your network |
| 00:37.50 | chobbs | Crippled: Yeah, most of it. Is there a straightforward example in there for what I want to do? The key thing is making sure it passes the 9 through with the outbound number. |
| 00:38.15 | neondog420 | got a tftp server..need images |
| 00:38.34 | tessier | neondog420: I'm in the same boat. |
| 00:38.41 | tessier | Actually I need to set up a tftp server also. |
| 00:40.32 | Crippled | tessier: try in.tftp -l -s /path/to/your/files |
| 00:42.33 | denon | so is CVS stable today? |
| 00:42.33 | matobago | someone know what is the configuration file for connect to h.323 server? |
| 00:44.10 | tessier | matobago: No but I have a sneaking suspicion it's in the docs somewhere. |
| 00:44.16 | tessier | Crippled: Thanks, I'll try that |
| 00:46.44 | chobbs | Hmmm - wouldn't my one and only TDM10B port be Zap/1? |
| 00:48.33 | *** join/#asterisk rburkhal (~rburkhall@ip68-229-6-75.lv.lv.cox.net) |
| 00:49.34 | rburkhal | hi all |
| 00:49.51 | rburkhal | anyone using freebsd 5.2.1 with asterisk? |
| 00:50.04 | *** join/#asterisk gbdrbob (~drbob@alltalk.demon.co.uk) |
| 00:50.21 | tessier | rburkhal: Nope. * doesn't work on BSD. |
| 00:52.26 | rburkhal | are u sure?, bugtrack on digium claims you can |
| 00:52.45 | ManxPower | Actually ASTERISK works on *BSD, but none of the HARDWARE works on *BSD |
| 00:52.58 | tessier | oh, ok |
| 00:53.00 | ManxPower | i.e. no drivers for Digium cards work on *BSD |
| 00:53.14 | rburkhal | i dont need the hardware to work |
| 00:53.25 | rburkhal | can you help me get it running |
| 00:53.55 | rburkhal | the port is version 0.5.0 |
| 00:55.49 | *** join/#asterisk Pkunk (~imranyo@61.11.73.47) |
| 00:56.22 | ManxPower | rburkhal: 0.5.0 is generally considered "so old you would have to be a lunatic to use it" |
| 00:56.53 | Pkunk | is it possible to setup asterisk to make only one outgoing call per SIP peer ? i want to setup a round robin of 3 SIP peers for outgoing calls |
| 00:58.38 | Pkunk | like is there something i can set in sip.conf to do that ? |
| 00:59.04 | *** join/#asterisk Salvador_VE (Salvador@200.109.13.97) |
| 00:59.13 | *** part/#asterisk Salvador_VE (Salvador@200.109.13.97) |
| 01:00.22 | rburkhal | that's the only version that's ported on BSD, should I use a different OS, I really like FreeBSD\ |
| 01:01.43 | brc007 | sup h3x |
| 01:01.54 | brc007 | jbot seen h3x lately? |
| 01:01.55 | jbot | i haven't seen 'h3x lately', brc007 |
| 01:02.05 | brc007 | jbot seen h3x? idiot |
| 01:02.05 | jbot | i haven't seen 'h3x? idiot', brc007 |
| 01:02.18 | fskfsk | Is iLBC less computation intensive than G729 ? |
| 01:02.44 | rburkhal | any responses to my OS question? |
| 01:03.07 | Pkunk | http://www.mail-archive.com/asterisk-dev@lists.digium.com/msg00700.html <--- is this code already in asterisk ? |
| 01:03.35 | Crippled | where can I find where nufone can terminate and originate to on their site? Anyone wanna help me out? |
| 01:03.50 | Crippled | meaning.. what cities and exchanges |
| 01:04.25 | brc007 | huh? |
| 01:04.34 | brc007 | don't understand exactly |
| 01:04.50 | brc007 | they don't offer many local dids last time I checked |
| 01:04.51 | Crippled | I want to know if I can get a local number |
| 01:04.55 | brc007 | I doubt it |
| 01:05.07 | brc007 | you can get an 800 number for 2.9c/min though |
| 01:05.13 | brc007 | no monthly fee |
| 01:05.22 | rburkhal | what os are u guys using for asterisk? |
| 01:05.43 | Crippled | that sounds good.. I'm looking for a provider in Irvine California |
| 01:06.23 | brc007 | rburkhal: almost everyone uses <insert favorite flavor> linux |
| 01:06.29 | Pkunk | how many SIP uLaw/aLaw calls can a celeron-466 handle ? or should i use reinvite ? |
| 01:06.49 | Crippled | wow.. celeron 466 |
| 01:07.07 | Crippled | THat's like a p1 |
| 01:07.09 | Crippled | :( |
| 01:07.21 | brc007 | I'm either going to use debian or gentoo. |
| 01:07.22 | MustDie | Pkunk: stop checking garbage cans. |
| 01:07.23 | FryGuy | grr |
| 01:07.27 | brc007 | or fedora |
| 01:07.41 | FryGuy | i just typed out a long howto for Adtran TSU + T100P and voip-info.org borked |
| 01:08.00 | rburkhal | is geetoo good or debian better |
| 01:08.00 | FryGuy | luckily i use mozilla firefox so I just got what I typed out from the page info |
| 01:08.02 | Pkunk | my main linux server is a cel-400 , Crippled . and it does RAID , dns , NAT , samba ,squid and the CPU is usually abt 3-4% |
| 01:08.05 | Crippled | FryGuy: That stinks.. I hope you're inclined to do it again. |
| 01:08.08 | FryGuy | gentoo > debian |
| 01:08.33 | Pkunk | yep gentoo is nice . esp.if you optimize during nstall |
| 01:08.43 | rburkhal | ok I reformat the laptop and put debian on it |
| 01:08.53 | Crippled | pkunk: my main gateware does all those minimal services you say as well and it's a k6/2 450 |
| 01:09.28 | Pkunk | Crippled : k then does asterisk need a P4 ? |
| 01:10.33 | fOSSiL | Pkunk, umm, Star Control? you should hang out in #sc2 :) |
| 01:10.34 | FryGuy | http://www.voip-info.org/tiki-index.php?page=adtran |
| 01:10.36 | Pkunk | MustDie : i never check . i just want to avoid throwing all that old hardware i bought for lotsa moolah into one |
| 01:10.40 | FryGuy | does that look good to you guys? |
| 01:11.15 | *** join/#asterisk ChulJin (~chuljin@adsl-68-121-94-237.dsl.irvnca.pacbell.net) |
| 01:13.13 | timecop | uh what? |
| 01:13.18 | timecop | asterisk does NOT need a p4 |
| 01:13.25 | timecop | its running just fine off a p2 350 at my house |
| 01:13.37 | timecop | routing like 5 voip calls at a time and running one zap channel |
| 01:13.42 | ManxPower | That would be a cool name for a Linux Firewall Distro: Gateware! |
| 01:13.44 | ChulJin | p3 500 here, $60 on ebay, runs fine |
| 01:14.14 | ChulJin | (not that I saw what/whom timecop was replying to :P) |