| 00:02.13 | X-Rob | RazorJack - for future reference, a list of all the commands that can be used in the dialplan can be viewed by 'show applications' |
| 00:04.26 | drbrown | how would I convert gsm files into wav format? |
| 00:04.31 | X-Rob | sox |
| 00:04.46 | *** join/#asterisk _Simon (n=IRC@i216-58-40-193.cybersurf.com) |
| 00:04.54 | _Simon | good evening everyone. I was trying to search the wiki but couldn't find this info. what do I put in the app and appdata fields if using RealTime in the extentions table if I want an extention to dial a users iax |
| 00:06.00 | tzafrir_laptop | drbrown, you realise, though, that this won't gain you back the quality, as the gsm compression is lossy |
| 00:06.38 | *** join/#asterisk doughecka (n=Miranda@unaffiliated/doughecka) |
| 00:07.07 | syle2 | app=DIAL appdata=IAX2/blah blah blah |
| 00:07.50 | _Simon | syle2: yeah thats what I'm confused what to put after the /, do I put username@hostname? |
| 00:08.04 | X-Rob | IAX2/user:password@host/number |
| 00:08.17 | syle2 | whatever is normally between your () in your dial command |
| 00:09.13 | drbrown | gsm is broke on my t1, and I cant find a plausable solution anywhere |
| 00:09.39 | malverian[work] | Anyone think of a reason why I'd get echo on local numbers (SIP phone -> PRI -> PSTN) but not on cell phones or long distance? |
| 00:10.00 | drbrown | becuase of how echo works |
| 00:10.14 | _Simon | hmm not sure I'm understanding |
| 00:10.21 | drbrown | echo is canceled on long distance calls and cell phones by the carriers |
| 00:10.25 | malverian[work] | Any recommendations for fixing it? |
| 00:10.35 | drbrown | echocancel = 64 |
| 00:10.41 | drbrown | echotraining =800 |
| 00:10.51 | drbrown | and start working up from their |
| 00:10.54 | distortion | drbrown: http://www.asteriskguru.com/audio_conversion.php |
| 00:11.04 | malverian[work] | drbrown, Okay thanks. |
| 00:11.11 | *** join/#asterisk Qwell (n=chatzill@dsl001-136-136.lax1.dsl.speakeasy.net) |
| 00:11.39 | malverian[work] | What's the default echocancel value? |
| 00:12.00 | *** join/#asterisk lilo (i=levin@freenode/staff/pdpc.levin) |
| 00:12.13 | _Simon | asterisk keeps saying "15181@default" does not exist |
| 00:12.25 | drbrown | I can't remember, but valid ones are 8, 16, 32, 64, & 128 |
| 00:12.34 | drbrown | & 256 |
| 00:13.03 | drbrown | I usually use 128 on echo cancel, and 1000 on echotraing, but always start w/ 64/800 |
| 00:14.26 | *** join/#asterisk JamesDotCom (n=james@sweep.bur.st) |
| 00:15.33 | _Simon | X-Rob: so I have Dial in app, and IAX2/username@hostnameofserver.com/extentionofuser ? |
| 00:15.49 | *** join/#asterisk SkramX (n=skramy@vistech.org) |
| 00:18.46 | _Simon | thats in appdata |
| 00:19.10 | malverian[work] | drbrown, Do I turn the values up or down to fix the echo if it still exists? |
| 00:25.21 | *** join/#asterisk r0d3nt (i=nobody@wsip-24-234-241-145.lv.lv.cox.net) |
| 00:26.10 | *** join/#asterisk ronaldl79 (n=chatzill@c-24-8-54-203.hsd1.co.comcast.net) |
| 00:26.34 | ronaldl79 | Okay....are busy tones and 'This number is disconnected' the norm for BroadVoice service? |
| 00:27.11 | denon | more or less |
| 00:27.33 | _Simon | can someone help me with the extentions realtime table please |
| 00:27.42 | _Simon | that would be greatly appreciated :) |
| 00:28.42 | ronaldl79 | Well, that really sucks, denon ... my number is being used by clients, friends and family ... people have been telling me my number is always busy ... this sucks. |
| 00:28.54 | ronaldl79 | Now, I can't even dial out .... |
| 00:29.02 | denon | _Simon: http://www.voip-info.org/wiki/view/Asterisk+RealTime+Extensions |
| 00:29.12 | denon | ronaldl79: www.nufone.net |
| 00:29.44 | ronaldl79 | denon ... I can't afford Nufone, I just left Vonage after two years for Broadvoice ... and I can't continue issuing new phone numbers, let alone spending more activation fees for another provider. |
| 00:29.46 | malverian[work] | Fucking A.. apparently our CO told us the wrong DB value for our CSU. |
| 00:29.56 | denon | activation fees? |
| 00:29.58 | malverian[work] | Apparently we have a -7 not a 0 |
| 00:30.00 | denon | didnt know nufone had them.. |
| 00:30.18 | ronaldl79 | Had what? |
| 00:30.23 | denon | activation fees |
| 00:30.31 | ronaldl79 | I figured they do.... |
| 00:30.34 | denon | nah |
| 00:30.36 | ronaldl79 | Maybe not? |
| 00:30.37 | denon | just per minute |
| 00:30.41 | ronaldl79 | Yeah, that sucks.... |
| 00:30.45 | ronaldl79 | I'm used to unlimited usage..... |
| 00:30.45 | denon | ? |
| 00:30.48 | denon | so get a DID |
| 00:30.50 | ronaldl79 | Nufone would break my bank.... |
| 00:30.52 | denon | like 8 bucks a minute |
| 00:30.54 | denon | er a month |
| 00:30.58 | denon | and 0 per minute for incoming |
| 00:31.09 | ronaldl79 | What's outgoing? |
| 00:31.12 | denon | 2x |
| 00:31.13 | denon | 2c |
| 00:31.15 | ronaldl79 | And why do you recommend Nufone? |
| 00:31.23 | denon | because I like em, and theyre stable for me |
| 00:31.43 | denon | how much did you pay at vonage? |
| 00:31.46 | ronaldl79 | What are you using them for? Biz and personal? |
| 00:31.46 | denon | per month |
| 00:31.49 | ronaldl79 | $24.95 |
| 00:31.52 | drbrown | anyone have any probs w/ T1 cards and distortion under anything gsm including prompts and vmail????? |
| 00:32.02 | denon | so thats 1250 minutes a month |
| 00:32.09 | denon | do you use more than that, outgoing? |
| 00:32.11 | ronaldl79 | I'm paying Broadvoice $19.95 for a BYOD plan .. .can't beat that, but I can't deal with people not being able to reach me either. |
| 00:32.18 | ronaldl79 | Yes, I have, denon. |
| 00:32.24 | *** join/#asterisk SpaceBass (n=SpaceBas@c-24-125-184-203.hsd1.va.comcast.net) |
| 00:32.31 | denon | ronaldl79: outgoing? |
| 00:32.39 | ronaldl79 | Yes |
| 00:32.40 | file | it's not billed per minute though, so you get more then you think |
| 00:32.52 | denon | right .. and 6-second incremented |
| 00:33.01 | ronaldl79 | File, are you using Nufone as well? |
| 00:33.06 | file | no |
| 00:33.11 | ronaldl79 | They don't offer instant provisioning, do they? |
| 00:33.20 | _Simon | denon: I went over this file.. still very confused but I think perhaps I know one thing I'm missing and thats the switch => Realtime/mycontext@realtime_ext statement |
| 00:33.20 | denon | nufone? yeah, I think it goes live within a couple minutes |
| 00:33.24 | ronaldl79 | If I were to jump to another DID provider, I need one tonight.... |
| 00:33.36 | denon | ronaldl79: it happens within a few minutes .. |
| 00:33.37 | SkramX | ronaldl79: what NPA? |
| 00:33.39 | file | NuFone only does toll-free DIDs, and Michigan numbers |
| 00:33.42 | ronaldl79 | However, it would royally suck to tell people yet again...I've got a new number...that blows. |
| 00:33.49 | _Simon | what is this @realtime_ext section? it says its the family name that you enter in extconfig.conf, but not sure what it means by family name, does it mean table name? |
| 00:33.51 | SkramX | denon: you work for nufone? |
| 00:33.52 | denon | ronaldl79: then go gripe to broadvoice |
| 00:33.55 | file | ronaldl79: then complain to broadvoice |
| 00:33.55 | ronaldl79 | Well, that sucks, because I need a DID for Colorado...if I were to leave Broadvoice. |
| 00:33.57 | denon | SkramX: no |
| 00:34.00 | denon | SkramX: I just use them |
| 00:34.00 | SkramX | okay |
| 00:34.02 | ronaldl79 | I am complaining to Broadvoice |
| 00:34.07 | SkramX | ronaldl79: what NPA? |
| 00:34.15 | ronaldl79 | NPA? |
| 00:34.30 | ronaldl79 | That's not registering with me right now.... |
| 00:34.34 | SkramX | Area Code |
| 00:34.37 | ronaldl79 | 303 |
| 00:34.44 | SkramX | Wymoning, right? |
| 00:34.46 | ronaldl79 | of 720 |
| 00:34.48 | ronaldl79 | No, Colorado |
| 00:35.02 | SkramX | Oh Yeah |
| 00:35.12 | SkramX | so you need a 303 DID? |
| 00:35.21 | ronaldl79 | Yeah, or 303 |
| 00:35.24 | ronaldl79 | 720 rather |
| 00:35.26 | ronaldl79 | either or |
| 00:35.36 | SkramX | Hmm |
| 00:35.41 | ronaldl79 | I need a rock solid provider, but hell, I don't want to spend anymore money either.... |
| 00:35.50 | ronaldl79 | BV just needs to fix this shit...... |
| 00:35.56 | _Simon | ok yes it says familyname and table name need to match, so I've done this, still something wrong in my appdata I think, I'm trying to use the default context |
| 00:36.26 | ronaldl79 | And BV is using Global Crossing for termination ... I can't say GC is at fault. |
| 00:37.36 | _Simon | keeps saying "rejected connect attempt from 10.0.0.2, request '13961@default' does not exist" |
| 00:37.50 | file | extension 13961 in context default, does not exist |
| 00:38.19 | _Simon | but its in my extentions table |
| 00:38.20 | _Simon | lol |
| 00:38.33 | file | using a switch? |
| 00:39.02 | _Simon | I have this in extentions.conf: switch => Realtime/default@extentions |
| 00:39.18 | file | you do realize extensions is spelled extensions just a little fyi. |
| 00:39.19 | *** join/#asterisk justinn (n=justinnn@61.95.68.85) |
| 00:39.20 | justinn | hey ppls |
| 00:39.26 | justinn | whos a good international sip provider ?? |
| 00:39.59 | bongfrog | ronald|79 tyied teliax??? they are based in Denver |
| 00:40.06 | *** join/#asterisk elg (n=fugalh@falcon.fugal.net) |
| 00:40.09 | drbrown | file: I figured you used asterlink |
| 00:40.12 | *** join/#asterisk greendisease (n=jack@fedora/greendisease) |
| 00:40.13 | lancey`away | justinn: magrathea works very well for me |
| 00:40.21 | file | drbrown: I do |
| 00:40.26 | justinn | whats there website ? |
| 00:40.34 | lancey | hi all |
| 00:40.51 | lancey | justinn ? magrathea's? |
| 00:40.51 | drbrown | you going to astricon |
| 00:41.03 | file | drbrown: Astricon ends tomorrow, and no - I didn't go - too expensive |
| 00:41.27 | _Simon | file: haha I'm a dumbass, its getting further now, its trying to route the call, thanks so much, still something in my appdata is messed though, getting an unable to create channel IAX2 |
| 00:41.38 | justinn | lancey ya ? |
| 00:42.36 | _Simon | in app I have "Dial" and appdata I have IAX2/test@default |
| 00:42.45 | _Simon | test is the name of the iaxuser in the iaxusers table |
| 00:43.02 | lancey | justinn: www.magrathea.net |
| 00:43.07 | lancey | but you'd better mail them |
| 00:43.15 | lancey | there's almost no info at the site |
| 00:43.27 | file | _Simon: default is the peer entry to use to send the call, and test is the username on the remote server to use... |
| 00:43.30 | justinn | any other ones :) ? |
| 00:43.59 | _Simon | file: peer entry, not sure what you mean by that? the iax username is test |
| 00:44.08 | file | how does it know where to send the call? |
| 00:44.08 | _Simon | I put default for the default context, obviously thats wrong then |
| 00:44.12 | justinn | voipjet seems to be crap |
| 00:44.14 | justinn | it goes up and down all the time |
| 00:44.16 | file | yeah, your usage is wrong |
| 00:44.18 | justinn | we dont ever change our config.. |
| 00:44.28 | _Simon | file: I thought thats what "test" was for, telling it which iax user? |
| 00:44.46 | file | _Simon: it's saying "send a call using IAX2 to the peer named default, and use username test for authentication" |
| 00:45.05 | *** join/#asterisk ManxPower (n=eric@slip-12-65-162-105.mis.prserv.net) |
| 00:45.09 | file | justinn: they're cheap, you can't expect the best |
| 00:45.14 | _Simon | file: so do I need IAX2/test@test ? |
| 00:45.29 | file | _Simon: if you want to send a call to the peer named test, using the username test |
| 00:45.56 | _Simon | file: ok, but what if I want the call to be from the person thats calling? I can't hard code that in the DB lol |
| 00:46.08 | file | that made no sense |
| 00:46.35 | _Simon | ok sorry, I'm really confused, but you said "send a call to the peer named test, using the username test" |
| 00:46.37 | *** join/#asterisk litage (n=nick@203.220.55.70) |
| 00:46.51 | _Simon | I want to send a call to peer named test, using my username which I called from do I not? |
| 00:47.00 | file | you can. |
| 00:47.08 | file | asterisk isn't psychic though, it doesn't know that... |
| 00:47.19 | file | you need to go read up |
| 00:47.26 | _Simon | ok, but I can't hardcode in appdata test@simon |
| 00:47.28 | file | or grab the O'Reilly book, it explains this |
| 00:47.33 | _Simon | or else all calls come from simon? |
| 00:47.45 | lancey | _Simon |
| 00:47.57 | lancey | usernames and passwords are used for authentication/access only |
| 00:48.06 | lancey | not for caller ID as you seem to think |
| 00:48.17 | lancey | or i can't get your idea too ;) |
| 00:48.25 | _Simon | lancey: thats why I'm totally confused why file was saying "using username test" |
| 00:48.34 | _Simon | I understand I want to "call to the peer named test" |
| 00:48.38 | shido6 | ~seen steve |
| 00:48.42 | jbot | steve <n=steve@149.137.209.43> was last seen on IRC in channel #debian, 21h 21m 11s ago, saying: 'exit'. |
| 00:48.42 | _Simon | but the 2nd half I'm lost |
| 00:48.45 | *** part/#asterisk fuzy (n=fuz@oxide.sprintlink.net) |
| 00:48.52 | lancey | _Simon in order for that peer to accept your call |
| 00:48.59 | lancey | you might want to authenticate yourself |
| 00:49.15 | _Simon | iax register you mean? |
| 00:49.16 | lancey | that's what the username is for |
| 00:49.24 | lancey | yup, iax registration |
| 00:49.24 | _Simon | I've registered both peers already |
| 00:49.31 | file | registration only tells the other side, "hey - if you need to call me, you can reach me here" |
| 00:49.34 | lancey | well you might not have to send a username at all |
| 00:49.40 | _Simon | file: yup I understand that part |
| 00:49.44 | ManxPower | _Simon, and "iax2 show peers" or "iax2 show registry" confirms that? |
| 00:49.54 | file | you usually have to authenticate when you're placing a call to the peer as well |
| 00:50.00 | _Simon | ManxPower: umm, lemme check |
| 00:50.03 | lancey | _Simon then you can simply dial IAX2/peername/extension |
| 00:50.28 | ManxPower | iax2 show peers shows what devices are registered TO your asterisk server. iax2 show registry shows what remote services your asterisk has rtegistered with |
| 00:50.53 | lancey | _Simon somewhere was a nice handbook of asterisk |
| 00:50.57 | litage | i've read the wiki () on peoples' experiences in relation to asterisk, hardware, and performance, but i was wondering if there's anyone around right now who's actually put asterisk on a fat box (eg: dual xeon, 2gb ram, ..) and had several hundred concurrent calls going |
| 00:50.57 | lancey | it will really help you |
| 00:50.59 | ManxPower | ~docs |
| 00:51.00 | jbot | docs is probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk |
| 00:51.06 | lancey | yay |
| 00:51.07 | lancey | :) |
| 00:51.16 | wcj | ~seen wcj |
| 00:51.17 | jbot | wcj is currently on #asterisk (55m 34s). Has said a total of 2 messages. Is idling for 1s |
| 00:52.29 | _Simon | hmm weird, maybe my peers aren't registering |
| 00:52.38 | _Simon | when I do iax2 show registry or show peers, they don't show up |
| 00:52.45 | _Simon | but the client apps say "register accepted" |
| 00:52.59 | _Simon | and iax says "restricting registration for peer 'simon' to 60 seconds |
| 00:53.01 | file | if you're using like realtime... they won't show up unless you use caching |
| 00:53.03 | _Simon | does that mean it was accepted? |
| 00:53.07 | lancey | at least with my * box, realtime registrations do not show |
| 00:53.10 | lancey | unless cached |
| 00:53.10 | _Simon | file: ahh ok, yeah its realtime |
| 00:53.18 | lancey | oops |
| 00:53.20 | file | lancey: off my brainwaves |
| 00:53.22 | _Simon | ok that explains that |
| 00:53.23 | lancey | :) |
| 00:53.33 | ManxPower | Why don't people make it work without realtime before trying something as complex as realtime? |
| 00:53.35 | lancey | file yup..... aoeey |
| 00:53.36 | lancey | *sorry |
| 00:53.52 | lancey | ManxPower cause realtime is cool :) |
| 00:54.05 | _Simon | so far I'm loving realtime, I'm quite sure its loading all my tables |
| 00:54.14 | _Simon | I've just got the app/appdata wrong |
| 00:54.21 | malverian[work] | Is there a maximum size for a variable? |
| 00:54.23 | _Simon | and I was missing the switch, which I now have |
| 00:54.32 | lancey | _Simon it would really be better to first experiment with static configs |
| 00:54.36 | lancey | and see how things work |
| 00:54.42 | lancey | you will help yourself a lot |
| 00:55.02 | _Simon | lancey: well I just got it working by doing IAX2/test@test |
| 00:55.20 | _Simon | but I'd like to learn why I need to type test twice, and if its not 100% correct to do so, learn why lol |
| 00:55.46 | ManxPower | _Simon, if you do not have username= in the [test] section, then you would need test@test |
| 00:55.57 | timecop | dongs |
| 00:56.05 | timecop | whats so special about 1.2.0 beta1? |
| 00:56.17 | _Simon | ManxPower: no usernames in the actual files, all loading from realtime |
| 00:56.29 | ManxPower | _Simon, I cannot speak for Realtime |
| 00:56.38 | ManxPower | timecop, nothing, really. |
| 00:56.43 | lancey | _Simon: its IAX2/username@peer |
| 00:56.50 | lancey | so the first test is the username |
| 00:56.54 | lancey | the second one - the peer name |
| 00:57.18 | _Simon | lancey: but both are the call destination? |
| 00:57.34 | lancey | .... |
| 00:57.40 | _Simon | I guess what I don't understand is, when would it be test@somethingelse |
| 00:57.53 | lancey | if the host you are calling is "somethingelse" |
| 00:58.19 | _Simon | host of asterisk or host of test? |
| 00:58.26 | lancey | IAX2/[username[:password]@]peername/extension |
| 00:58.38 | lancey | things in [] being optional |
| 00:58.48 | litage | what is Host Media Processing? |
| 00:58.51 | lancey | _Simon host of what you want to call |
| 00:58.54 | ManxPower | I think the extension is "optoional" too. |
| 00:59.01 | lancey | ManxPower correct, sorry |
| 00:59.03 | ManxPower | as is @context after extension |
| 00:59.27 | lancey | i was just trying to bold up the optional user & password |
| 00:59.34 | _Simon | lancey: so I could of just put IAX2/test and it would of called the person who registeres under the username test? |
| 00:59.41 | lancey | yup |
| 00:59.50 | _Simon | hmm lemme try it hehe |
| 01:00.01 | lancey | it depends, though |
| 01:00.27 | lancey | if it is a user, you should be able to, if he has registered to your * box |
| 01:00.55 | justinn | anyone else recommend a good voip international provider :) ??? |
| 01:01.48 | *** join/#asterisk mcf3782 (n=mcf3782@adsl-065-012-184-148.sip.asm.bellsouth.net) |
| 01:06.22 | _Simon | hmm call keeps getting hung up somehow |
| 01:06.48 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-179-75-202.cust.bezeqint.net) |
| 01:07.17 | _Simon | wonder if its a firewall issue since one of my test "peers" is behind the same firewall the asterisk server is on |
| 01:07.57 | *** join/#asterisk pepepedo (i=ircap@218-161-89-200.fibertel.com.ar) |
| 01:10.11 | _Simon | <PROTECTED> |
| 01:10.14 | _Simon | not sure what that means |
| 01:11.03 | *** join/#asterisk T-Squared (n=ted@hidden.serreyn.com) |
| 01:12.34 | *** part/#asterisk T-Squared (n=ted@hidden.serreyn.com) |
| 01:12.34 | *** join/#asterisk Baph (n=Dave@dirobertson.plus.com) |
| 01:14.53 | ManxPower | _Simon, that means that extenstion 13961 in context exited at priority 1 |
| 01:15.18 | _Simon | ManxPower: so the client hung up? |
| 01:15.47 | ManxPower | _Simon, no way to tell unless you look at IAX2 debug or check the value of Dialstatus |
| 01:17.01 | _Simon | if an iax peer is on the same lan behind a NAT as asterisk, then one peer is on the external internet |
| 01:17.08 | _Simon | would that cause problems? perhaps I just have a bad test case |
| 01:17.24 | *** join/#asterisk Defraz (n=t0tal@72.24.26.215) |
| 01:17.43 | *** join/#asterisk goatmilk (n=goatmilk@130-127-45-11.chouse.resnet.clemson.edu) |
| 01:17.47 | goatmilk | i am here! |
| 01:17.56 | Corydon76-home | Praise be! |
| 01:19.19 | lancey | _Simon it can only cause problems with calls from your * box to the external peer |
| 01:19.23 | lancey | but not to the internal one |
| 01:19.56 | _Simon | lancey: so if my external peer calls the internal peer it should be ok? as long as the internal peer doesn't call the external one? |
| 01:20.04 | *** join/#asterisk alephcom (n=Miranda@207.34.97.130) |
| 01:20.42 | lancey | _Simon can't say that |
| 01:20.55 | lancey | depends on NAT and on * settings |
| 01:21.11 | _Simon | ok, so its not a given * will work in my testing environment then |
| 01:21.20 | _Simon | its weird because it works for 2 seconds then drops |
| 01:21.22 | lancey | you'd better not put asterisk behind nat |
| 01:21.37 | _Simon | lancey: I can't really help that unfortunately |
| 01:21.56 | _Simon | I thought iax was touted as NAT transparent somehow |
| 01:22.09 | goatmilk | drumkilla: lo??? |
| 01:22.30 | drumkilla | yo |
| 01:22.54 | drumkilla | hopefully i'll have wifi up there |
| 01:23.08 | goatmilk | wait |
| 01:23.53 | *** join/#asterisk MrBelvedr (n=tt@ip70-187-237-193.dc.dc.cox.net) |
| 01:24.20 | Starmaker | does anyone know where I can get some free professional sounding on-hold music |
| 01:26.10 | snitt | for free? |
| 01:26.16 | tzafrir_laptop | Starmaker, there is some classical music availble |
| 01:26.19 | mcf3782 | Starmaker: I found some on http://www.incompetech.com/m/c/royalty-free/ that I liked for MOH |
| 01:26.26 | snitt | try freeplaymusic, for low prices |
| 01:27.54 | Starmaker | mcf3782, cool, thanks |
| 01:28.00 | mcf3782 | np |
| 01:29.04 | Starmaker | i need to have something that all the annoying telemarketers i'm gonna put on hold will have to listen to :) |
| 01:29.21 | mcf3782 | umm.. dialtone? |
| 01:29.25 | mcf3782 | ;) |
| 01:29.45 | Starmaker | not as good, I want to make them hope that I'll actually pick up the call :) |
| 01:31.02 | Starmaker | just imagine the suprise of the telemarketer, "Hello, I am calling from company x." "I'll transfer you to our buying department, please hold"' |
| 01:31.04 | Starmaker | :) |
| 01:31.15 | lancey | :)))) |
| 01:31.46 | Starmaker | of course I have to record everything the telemarketer says when he's on hold :) |
| 01:31.46 | lancey | Starmaker |
| 01:31.59 | lancey | i believe there was some url for royalty-free music at asterisk`s site |
| 01:32.30 | tzafrir_laptop | freeplay music, I figure |
| 01:32.30 | lancey | http://www.asterisk.org/ |
| 01:32.34 | lancey | look at the bottom :) |
| 01:32.54 | tzafrir_laptop | sineapps also has a page of classical music mp3s |
| 01:33.49 | Starmaker | cool, lot's to choose from |
| 01:33.51 | goatmilk | i just want everyone to know that you rock |
| 01:34.10 | HiltonT | I'm sure I could make a bunch available too - I'm partners in a local recording studio in Brisbane |
| 01:34.50 | HiltonT | I'll have a chat to my partner and we'll approach the bands that we've recorded for some stuff that will be Asterisk friendly and royalty-free for * users |
| 01:35.19 | HiltonT | mainly acoustic folk music, not really much punk or death metal |
| 01:35.40 | Starmaker | hmm, death metal, that might annoy the hell out of telemarketers (pun intended) |
| 01:35.48 | HiltonT | exactly my thoughts |
| 01:35.49 | mcf3782 | That'd be cool, HiltonT. I'm sure a lot of people would appreciate that. |
| 01:35.59 | HiltonT | np - will speak to Mark sometime soon |
| 01:36.35 | *** join/#asterisk thetalon (n=Administ@pcp05736660pcs.norstn01.pa.comcast.net) |
| 01:38.27 | HiltonT | speaking of MOH, what would be the best format to have this in? |
| 01:38.38 | HiltonT | bitrate, channels, codec, etc |
| 01:38.45 | HiltonT | I'm assuming that mono would be best :) |
| 01:38.59 | Starmaker | what? not 7.2 DTS? |
| 01:39.11 | HiltonT | that was my next chice :) |
| 01:39.14 | h3x0r | great |
| 01:39.16 | Starmaker | :) |
| 01:39.19 | h3x0r | just what we need |
| 01:39.23 | h3x0r | hollywood making money off our phone calls |
| 01:39.30 | Starmaker | :) |
| 01:39.32 | HiltonT | handy for an office with 8 phones :) |
| 01:39.46 | Starmaker | one channel per phone :) |
| 01:40.37 | Starmaker | hmm, what do you think telemarketers would think about polka as hold music? |
| 01:40.50 | HiltonT | Weird Al's polka music! |
| 01:41.00 | Starmaker | no, no, that has royalties |
| 01:41.05 | HiltonT | yeah |
| 01:41.05 | mcf3782 | polka?! |
| 01:41.08 | Starmaker | and you can actually laugh at it |
| 01:41.16 | HiltonT | or something from the Dead Kennedys |
| 01:41.18 | Starmaker | real polka's just sad |
| 01:41.22 | Starmaker | :) |
| 01:41.24 | HiltonT | (if only it were roalty free) |
| 01:42.09 | *** join/#asterisk alephcom (n=Miranda@207.34.97.130) |
| 01:42.16 | HiltonT | someone needs to write a really subtle "telemarketers suck" song and record it :) |
| 01:42.19 | Starmaker | how about death metal/polka fusion (if i can find that anywhere) |
| 01:42.29 | goatmilk | title and registration |
| 01:42.30 | goatmilk | ! |
| 01:42.56 | HiltonT | death metal/polka fusion with a pan pipe solo |
| 01:43.06 | Starmaker | no, no, bag pipe |
| 01:43.10 | HiltonT | rofl |
| 01:43.25 | Starmaker | the bag pipe has to play amazing grace |
| 01:43.26 | mcf3782 | you guys are making my skin crawl. |
| 01:43.30 | HiltonT | I don't really go much for cats, but bagpipes are just CRUEL |
| 01:43.33 | Starmaker | no matter what the other instrumets play |
| 01:44.22 | litage | how is the SER used in conjunction with Asterisk? |
| 01:44.22 | *** join/#asterisk websae (i=websae@207-118-134-96.dyn.centurytel.net) |
| 01:44.36 | websae | If anyone is interested....found some termination for 1.8cents/min |
| 01:46.23 | *** join/#asterisk mm_pt (n=mm@a83-132-248-28.cpe.netcabo.pt) |
| 01:46.55 | mm_pt | hello, some one working with asterisk as an IVR? |
| 01:47.13 | HiltonT | websae; interesting |
| 01:47.39 | mm_pt | i've been doing a case studie but with publicVoiceXML |
| 01:47.45 | mm_pt | some one around that? |
| 01:48.06 | InfraRed | websae: which country |
| 01:48.51 | websae | U.S. |
| 01:49.13 | ronaldl79 | OpenPBX...opinions...comments? |
| 01:49.20 | InfraRed | ok |
| 01:49.28 | InfraRed | i pay £0.007 |
| 01:49.33 | InfraRed | to terminate to the us |
| 01:49.36 | tzafrir_laptop | http://www.goingware.com/tips/legal-downloads.html |
| 01:49.44 | tzafrir_laptop | regarding free music |
| 01:50.00 | *** join/#asterisk desktophero (n=desktoph@ip24-56-30-250.ph.ph.cox.net) |
| 01:50.02 | tzafrir_laptop | ronaldl79, see #openpbx and http://openpbx.org/ |
| 01:50.08 | desktophero | hello all |
| 01:50.29 | InfraRed | which is 1.2c |
| 01:50.34 | ronaldl79 | I've been there, tza....I'm reading about it. |
| 01:51.17 | Starmaker | hmm, just 4 different polka songs |
| 01:51.17 | websae | if anyone is interested be sure to /msg me...1.8cents/min for termination |
| 01:51.25 | Starmaker | that would get annoying really fast |
| 01:55.11 | mm_pt | any one could help me with asterisk? |
| 01:55.30 | alephcom | for use, we are paying $0.045 with $500 deposit. |
| 01:55.36 | alephcom | use s/b US |
| 01:55.41 | *** join/#asterisk Chuji (i=Chuji@pcp09930052pcs.tulipgrove.tn.nash.comcast.net) |
| 01:56.29 | Chuji | Interesting Google hired the lead GAIM developer |
| 01:56.40 | Chuji | Wonder if they pay him well |
| 01:57.01 | Chuji | Hope he thanks Mark for getting that thing started |
| 01:57.08 | *** join/#asterisk jtodd (n=jtodd@dsl001-136-136.lax1.dsl.speakeasy.net) |
| 01:59.14 | *** join/#asterisk MikeJ[Laptop] (n=ircatjer@pcp02696669pcs.roylok01.mi.comcast.net) |
| 01:59.17 | *** join/#asterisk jskcr (n=jskcr@unaffiliated/jskcr) |
| 02:00.07 | jskcr | hy all |
| 02:00.14 | goatmilk | jskcr: what is crackin! |
| 02:00.33 | jskcr | hya goatmilk |
| 02:00.40 | jskcr | just got back from a linux user group |
| 02:00.47 | Nugget | Linux is poo. |
| 02:00.53 | goatmilk | arent they so much excitement |
| 02:00.57 | mm_pt | no one could help me with xlite and asterisk? |
| 02:01.03 | goatmilk | they make a britney concert look like a taping of oprah |
| 02:01.19 | jskcr | this one was, it got moved to a bar |
| 02:01.33 | jskcr | 20 people drinking beer talking about distros lol |
| 02:01.35 | *** join/#asterisk mhnoyes (n=mhnoyes@user-2ivfld4.dialup.mindspring.com) |
| 02:02.18 | goatmilk | entertainment at its finest |
| 02:02.58 | tzafrir_laptop | Is Nugget a bot? |
| 02:03.07 | tzafrir_laptop | Linux |
| 02:04.21 | jskcr | nugget are you a bot |
| 02:04.32 | jskcr | ~seen |
| 02:04.42 | jskcr | noper only jbot is alive |
| 02:04.45 | goatmilk | maybe we can scare it out of its natural habitat |
| 02:04.49 | jskcr | ~karma |
| 02:04.49 | jbot | jskcr has karma of 6 |
| 02:05.07 | goatmilk | the fire truck in its natural habitat is a vicious predator |
| 02:05.34 | jskcr | rot13 jr nyy xabj wfxpe ebpxf |
| 02:05.45 | jskcr | hmm |
| 02:06.56 | drbrown | what could cause gsm to be distorted only on a t1 interface and not on any IP calls????? |
| 02:07.10 | drbrown | all other fuctions on the T1 are very clear |
| 02:08.14 | jskcr | have the t1 tested by the co |
| 02:08.20 | jskcr | then change the smart jack |
| 02:08.25 | jskcr | then change the t1 card |
| 02:08.32 | *** join/#asterisk jeffik (n=Administ@CPE0011505c92d3-CM014350000760.cpe.net.cable.rogers.com) |
| 02:08.53 | jskcr | also it could be caused by a tos issue |
| 02:08.58 | *** join/#asterisk desktophero (n=chatzill@ip24-56-30-250.ph.ph.cox.net) |
| 02:09.15 | drbrown | It's hooked into a Rhino channel bank |
| 02:09.22 | drbrown | what is a "smart jack" |
| 02:09.25 | jskcr | other ip traffic getting equal or higher priority will cause all sorts of strange stuff on ip t1s |
| 02:09.47 | jskcr | follow the cable to where it terminates at the local facility |
| 02:09.54 | jskcr | that is the smart jack |
| 02:09.56 | marc324 | can you have the voice messages stored in db? |
| 02:09.57 | drbrown | k |
| 02:10.21 | jskcr | marc324: odbc strorage is included in asterisk realtime |
| 02:10.46 | marc324 | whats the first step in initializing it? |
| 02:10.53 | websae | anyone here running ASTBILL? |
| 02:11.04 | Starmaker | hmm, another idea is to take telemarketers and just transfer them to some random number using voipbuster |
| 02:14.44 | *** join/#asterisk theblue (n=theblue@pcp04402293pcs.nrockv01.md.comcast.net) |
| 02:15.25 | *** join/#asterisk ManxPower (n=eric@slip-12-65-162-105.mis.prserv.net) |
| 02:16.09 | *** join/#asterisk jdv79 (n=jdv79@u1057064.ul.warwick.net) |
| 02:16.26 | jdv79 | so i take it the berkley DB thing is pretty much depreciated |
| 02:17.08 | jdv79 | where else do people pull stuff for Athenticate from? |
| 02:17.09 | *** join/#asterisk Crontibs (n=frankie@ool-44c02c7f.dyn.optonline.net) |
| 02:17.53 | drbrown | it is not only gsm files it is also hold music that is distorted as well |
| 02:18.32 | marc324 | in th readmes - its written that odbcstorage is not full realtime. what does it mean. |
| 02:18.44 | marc324 | ? |
| 02:21.40 | *** join/#asterisk Snake-Eyes (n=blog@203.220.55.70) |
| 02:24.52 | *** join/#asterisk MikeJ[Laptop] (n=ircatjer@pcp02696669pcs.roylok01.mi.comcast.net) |
| 02:26.45 | *** join/#asterisk MikeJ[Laptop] (n=ircatjer@pcp02696669pcs.roylok01.mi.comcast.net) |
| 02:27.02 | ManxPower | drbrown, Define "distorted" Music is distorted by compressed codecs, that's just the way it is. |
| 02:27.10 | *** join/#asterisk MikeJ[Laptop] (n=ircatjer@d14-69-8-30.try.wideopenwest.com) |
| 02:27.29 | *** join/#asterisk Brijn (n=bas@S0106004063c0fa1f.vn.shawcable.net) |
| 02:27.44 | Brijn | Hello all |
| 02:27.54 | *** join/#asterisk SarahEmm (n=sarahemm@2.35.220-216.q9.net) |
| 02:28.17 | lancey`away | bye all |
| 02:28.25 | SarahEmm | bibi |
| 02:29.08 | X-Rob | Evening SarahEmm |
| 02:29.09 | jskcr | by |
| 02:33.06 | *** join/#asterisk MrBelvedr (n=tt@ip70-187-237-193.dc.dc.cox.net) |
| 02:34.32 | Dr_Ray | I wonder how much these sangoma AA cards are going to be to compete with a channel bank |
| 02:35.01 | SarahEmm | AA? |
| 02:35.19 | Dr_Ray | http://www.telephonyware.com/telephonyware/sangoma_aa.html |
| 02:36.00 | SarahEmm | ahhhh okay |
| 02:36.06 | SarahEmm | sangoma's answer to the TWM-400 then? |
| 02:36.08 | SarahEmm | err |
| 02:36.12 | SarahEmm | TDM-400 |
| 02:37.12 | Dr_Ray | it's supposedly better |
| 02:37.17 | Dr_Ray | I dunno |
| 02:37.46 | jskcr | why get a channel bank when you can get a t1 card |
| 02:37.57 | SarahEmm | err |
| 02:38.02 | SarahEmm | because you want to connect to analog jskcr? |
| 02:38.03 | X-Rob | jskcr - uh, coz you want to send and receive faxes? |
| 02:38.17 | Dr_Ray | you need a channel bank to hook a t1 to analog lines |
| 02:38.37 | jskcr | oh you want analog |
| 02:38.38 | kb1_kanobe | jskcr: 'cause that many t400p cards generate too much heat? |
| 02:38.41 | X-Rob | no, you need someting with FXS ports. Like a Sipura 20000 |
| 02:38.55 | Dr_Ray | my business you have to use fxs lines |
| 02:40.27 | jskcr | I agree spandsp still has alot of issues |
| 02:40.52 | *** join/#asterisk Soulz--- (n=Soulz---@host-137-132-43-53.imcb.nus.edu.sg) |
| 02:40.56 | Soulz--- | hello all |
| 02:41.00 | *** join/#asterisk lilo (i=levin@freenode/staff/pdpc.levin) |
| 02:41.00 | jskcr | If you dont need a channel bank I would not get one |
| 02:41.03 | Soulz--- | http://pastebin.ca/25452 |
| 02:41.08 | jskcr | but I guess ya need em :) |
| 02:41.17 | tzanger | channel banks are good |
| 02:41.18 | Soulz--- | when i get SIP/2.0 407 Proxy Authentication Required |
| 02:41.26 | tzanger | admittedly if you can just go T1 or PRI it's better (less components) |
| 02:41.27 | jskcr | t38 comming to a asterisk near you soon |
| 02:41.31 | Soulz--- | and it says SIP/2.0 100 Trying |
| 02:41.37 | Soulz--- | SIP/2.0 403 Forbidden |
| 02:42.14 | Soulz--- | how come i can receive calls, but when i make calls i get this on asterisk? |
| 02:42.22 | distortion | someone say t38?? :) |
| 02:42.28 | jskcr | yes t38 |
| 02:42.46 | distortion | "cleanup on isle 3" |
| 02:42.50 | SarahEmm | Soulz---: where is it trying to make the call to? |
| 02:42.57 | distortion | sorry, i just wet myself |
| 02:43.14 | distortion | any details? |
| 02:43.15 | Soulz--- | sarah: to a vsp |
| 02:43.30 | Soulz--- | sarah: but i can get a call from the vsp to my ip phone |
| 02:43.31 | jskcr | t38 good byt spandshit |
| 02:43.44 | Soulz--- | just when i call, i get this http://pastebin.ca/25452 |
| 02:44.14 | distortion | what about sitting in the middle of say 2 cisco's? |
| 02:44.30 | distortion | (more t38 passthrough) |
| 02:45.28 | SarahEmm | Soulz---: vsp? |
| 02:45.30 | marc324 | how do you update *? just install over the previous, or delete previous install first? |
| 02:45.37 | Dr_Ray | make clean |
| 02:45.39 | Dr_Ray | make install |
| 02:45.51 | marc324 | what does make clean do? |
| 02:45.58 | SarahEmm | Soulz---: are you sure your username/password are right? |
| 02:46.01 | Dr_Ray | deletes the old source code |
| 02:46.08 | Dr_Ray | well |
| 02:46.09 | Dr_Ray | wait |
| 02:46.20 | marc324 | no. i mean the old installed asterisk. |
| 02:46.55 | jskcr | distortion: would you like to know how to do t38 |
| 02:47.06 | Soulz--- | sarah: its right |
| 02:47.11 | distortion | YES. |
| 02:47.16 | Soulz--- | sarah: when i do sip show registry it shows its registered |
| 02:47.18 | jskcr | :) |
| 02:47.22 | jskcr | its not that hard |
| 02:47.27 | distortion | jskcr: YES. (incase you missed it) |
| 02:47.33 | jskcr | :) |
| 02:47.35 | SarahEmm | Soulz---: hrm. have you talked to your provider to see what they see? |
| 02:48.19 | jskcr | there ya go |
| 02:48.30 | jskcr | very simple |
| 02:48.33 | Soulz--- | yeah they told me i am disconnecting the call |
| 02:48.40 | Soulz--- | but my sip debug shows |
| 02:48.46 | Soulz--- | proxy authetication problem |
| 02:48.48 | SarahEmm | err... they are returning 403 to you tho |
| 02:48.52 | distortion | haha |
| 02:48.57 | Soulz--- | yeah i know |
| 02:49.18 | *** join/#asterisk Brijn (n=bas@S0106004063c0fa1f.vn.shawcable.net) |
| 02:49.25 | SarahEmm | pastebin your sip.conf and extensions.conf Soulz---? |
| 02:49.47 | Soulz--- | using asterisk@home |
| 02:50.18 | SarahEmm | ugh |
| 02:50.23 | SarahEmm | i have no idea then :P |
| 02:50.46 | HiltonT | is *@H really that fscked aboot with? :( |
| 02:50.55 | marc324 | is odbc storage realtime? |
| 02:51.04 | HiltonT | (I'm running Xorcom Rapid here, and just starting to play with it) |
| 02:51.15 | SarahEmm | HiltonT: i know nothing about *@H so i can't really help with it... |
| 02:51.17 | jskcr | marc324: yes |
| 02:51.20 | HiltonT | aha |
| 02:51.33 | HiltonT | I want to get AMP running on Xorcom Rapid at some point |
| 02:51.43 | marc324 | jskcr -- in the readme-- it says that its not full realtime |
| 02:52.11 | jskcr | marc324: no it basicly transfers the message to the database as a blob |
| 02:52.17 | jskcr | so no and yes |
| 02:52.24 | marc324 | what a blob? |
| 02:52.31 | SarahEmm | Binary Large OBject |
| 02:52.41 | marc324 | does it happen realtime? |
| 02:52.51 | marc324 | what is the delay? |
| 02:52.54 | jskcr | It would make no sense to transfer the voice mail to the data base in realtime |
| 02:53.03 | jskcr | its saved then stored. |
| 02:53.29 | marc324 | the file is saved to disk first? |
| 02:53.34 | jskcr | from the time the voicemail ends till its insertion into the database |
| 02:53.49 | jskcr | stdin > stdout |
| 02:53.54 | Soulz--- | sarah: was on the phone with the vsp |
| 02:54.04 | Soulz--- | sarah: sorry, trying to change some config info |
| 02:54.52 | SarahEmm | okay |
| 02:55.23 | HiltonT | tzafrir; art thee here? |
| 02:55.55 | *** join/#asterisk bjohnson_ (n=bjohnson@i216-58-61-89.cybersurf.com) |
| 02:57.33 | *** join/#asterisk Aquiles (n=Aquiles@63.245.80.243) |
| 02:57.57 | *** join/#asterisk KaBewM (n=kabewm@66-215-7-106.dhcp.psdn.ca.charter.com) |
| 02:57.59 | HiltonT | aahhh, the AMP issues I'm seeing here ALSO occur when trying their demo site!!! |
| 02:58.08 | *** part/#asterisk Aquiles (n=Aquiles@63.245.80.243) |
| 02:59.01 | *** join/#asterisk Igbothom_III (n=HiltonT@static-84.217.240.220.dsl.comindico.com.au) |
| 02:59.15 | websae | anyone here used ASTBILL? |
| 03:03.06 | *** join/#asterisk SwK (i=yfhyrk@12-219-144-126.client.mchsi.com) |
| 03:03.22 | shido6 | whats up, websae? |
| 03:09.50 | Soulz--- | sarah: fixed it thanks |
| 03:09.59 | SarahEmm | ahh |
| 03:10.01 | SarahEmm | waht was it? |
| 03:10.10 | Soulz--- | sarah: was to do with my fromuser= |
| 03:10.12 | *** join/#asterisk irv999 (n=irv999@ool-457249eb.dyn.optonline.net) |
| 03:10.14 | irv999 | lo all |
| 03:10.21 | SarahEmm | ahhh okay |
| 03:10.32 | Soulz--- | :) |
| 03:10.58 | irv999 | Does anyone have a need for used polycom phones? I just took back an installation and don't have a need for them at the moment.. |
| 03:11.25 | irv999 | I figured I would give you guys 1st crack as a thank you for all your help |
| 03:12.39 | *** join/#asterisk brookshire[home] (n=matt@esbrooks3.traveller.com) |
| 03:15.03 | alephcom | irv999 Where are you at? |
| 03:15.07 | BrianR___ | irv999: If they're x01's, I'll take 'em. |
| 03:15.36 | BrianR___ | irv999: how many? |
| 03:16.33 | *** join/#asterisk MikeJ[Laptop] (n=ircatjer@d14-69-8-30.try.wideopenwest.com) |
| 03:21.03 | *** join/#asterisk MikeJ[Laptop] (n=ircatjer@d14-69-8-30.try.wideopenwest.com) |
| 03:24.34 | HiltonT | what's the diff between the x00 and x01 versions, except I know they are updated revisions :) |
| 03:25.39 | SarahEmm | x01 has 4MB of flash |
| 03:25.43 | SarahEmm | x00 has 2MB of flash |
| 03:25.51 | SarahEmm | to use newest bootrom and newst SIP image you need 4MB |
| 03:26.06 | HiltonT | aha, cool, thanks for the info |
| 03:26.25 | SarahEmm | np |
| 03:26.37 | irv999 | ale: new york |
| 03:26.46 | irv999 | brian: no 300 500 and 600.. |
| 03:27.08 | marc324 | The database name (from /etc/asterisk/res_odbc.conf) is in the |
| 03:27.09 | marc324 | "odbcstorage" variable in the general section. |
| 03:27.20 | SarahEmm | how much are you wanting for a 600? :) |
| 03:27.26 | marc324 | I cannot find the "odbcstorage" variable |
| 03:27.29 | irv999 | sarah: $200 + shipping |
| 03:27.39 | SarahEmm | hmmmmmmms |
| 03:27.40 | SarahEmm | tempting :) |
| 03:27.45 | SarahEmm | i shouldn't but... |
| 03:27.46 | irv999 | sarah: I have 4 or 5 in stock.. |
| 03:27.56 | irv999 | sarah: save them from ebay! |
| 03:28.13 | SarahEmm | hehe |
| 03:28.28 | SarahEmm | i just got a 501, it'd be nice to have one on each floor of the house tho... |
| 03:28.30 | *** join/#asterisk nomazda (i=nyyankee@user-0c6tnqf.cable.mindspring.com) |
| 03:28.32 | SarahEmm | and the 600 has a microbrowser in it |
| 03:28.35 | irv999 | sarah: they are expensive phones. still under warranty! |
| 03:28.47 | irv999 | sarah: yep. nice display as well |
| 03:28.52 | SarahEmm | true, nicer than the 501 |
| 03:28.53 | SarahEmm | argh |
| 03:28.55 | SarahEmm | tempting! :P |
| 03:29.43 | *** join/#asterisk giesen (i=giesen@CPE0040f404c559-CM00122546417a.cpe.net.cable.rogers.com) |
| 03:29.48 | SarahEmm | it'd be mainly as another geek toy, i don't even use voice mode phone much :P but.. microbrowser and nice screen and.. hrm. :P |
| 03:30.26 | alephcom | irv999: are you interested in just selling 1 of them? |
| 03:30.35 | giesen | Whenever I try to dial between two SIP extensions, I get Oct 13 23:25:50 WARNING[10603]: chan_sip.c:695 retrans_pkt: Maximum retries exceeded on call 29cd759e6ae19679619d8de05d25ffdf@my.ip.goes.here for seqno 102 (Non-critical Request) |
| 03:30.37 | irv999 | alep: sure.. |
| 03:30.49 | giesen | the extension rings, but it goes straight to voicemail |
| 03:31.02 | giesen | I've set nat=yes in sip.conf for both extensions |
| 03:31.39 | marc324 | in the sql queries-- what is the line KEY ID (ID) ? |
| 03:31.43 | irv999 | if you are interested in the polycoms any quantity e-mail me sethz@mac.com |
| 03:31.48 | marc324 | what does it do? |
| 03:31.52 | alephcom | tks |
| 03:32.17 | giesen | any ideas? |
| 03:32.21 | Dr_Ray | irv999 what model? |
| 03:32.49 | irv999 | dr_: ip300 ip500 ip600.. |
| 03:32.59 | irv999 | dr_: 5 600 |
| 03:33.07 | irv999 | dr_: 5 600's 10 500's 20 300's |
| 03:33.17 | *** part/#asterisk mcf3782 (n=mcf3782@adsl-065-012-184-148.sip.asm.bellsouth.net) |
| 03:34.27 | SwK | ivr999: whats up man? |
| 03:35.00 | irv999 | swk: not much.. just offering these polycom phones to the ppl who helped me.. |
| 03:35.54 | *** join/#asterisk alephcom (n=Miranda@207.34.97.130) |
| 03:35.54 | SwK | kool |
| 03:36.23 | SarahEmm | ooh one thing, what version of the phones are they irv999? SIP, MGCP or H.323? |
| 03:36.36 | SwK | SIP |
| 03:36.36 | irv999 | sarah: SIP.. |
| 03:36.38 | SwK | heh |
| 03:36.43 | SarahEmm | cool |
| 03:36.54 | SarahEmm | hrm. |
| 03:37.29 | SwK | SarahEmm: i helped him with those phones they are all good and have the next to newest firmwares on them |
| 03:37.41 | irv999 | sarah: you know you want one.. its nearly 40% off retail and 25% off wholesale |
| 03:38.50 | SarahEmm | i know i *do* want one :P |
| 03:39.00 | SarahEmm | but i'm not sure if i *should* buy one ;) |
| 03:39.18 | *** join/#asterisk juice (n=juice@mo-67-77-176-229.dyn.sprint-hsd.net) |
| 03:41.18 | Corydon76-home | How much? |
| 03:42.00 | Corydon76-home | SarahEmm: why, because you're a textual person? ;-) |
| 03:42.27 | SarahEmm | Corydon76-home: yeah. i use text telephone 95% of the time and i already have a polycom 501 ;) |
| 03:42.32 | SarahEmm | buying a 600 *might* be overkill ;) |
| 03:42.39 | SarahEmm | but when others are at my house *they* use voice... |
| 03:42.46 | irv999 | cory: depends on quantity.. IP 600's $200 ip500's $180 ip300's $115 |
| 03:42.53 | Corydon76-home | Of course it would be overkill. But that's part of the fun |
| 03:42.56 | SarahEmm | hehe |
| 03:43.01 | SarahEmm | plus i can say the 600s have a microbrowser |
| 03:43.02 | alephcom | :-) |
| 03:43.03 | SarahEmm | which is text |
| 03:43.06 | SarahEmm | and kitriches use text! :) |
| 03:43.24 | irv999 | sarah: the whole asterisk phone system is overkill for a home. we don't do this for adequacy |
| 03:43.38 | SarahEmm | *giggle* |
| 03:43.38 | Corydon76-home | irv999: we'd probably buy the lot |
| 03:43.38 | SarahEmm | true. |
| 03:43.39 | *** join/#asterisk brookshire[home] (n=matt@esbrooks3.traveller.com) |
| 03:43.46 | SarahEmm | Corydon76-home: leave one for kitriches! ;) |
| 03:43.54 | Corydon76-home | irv999: but I'd have to check with my boss |
| 03:44.03 | file[laptop] | KITRICH |
| 03:44.05 | irv999 | cor: np email me when you get an idea.. sethz@mac.cpom |
| 03:44.07 | *** join/#asterisk Piranha- (i=piranha@d198-166-226-58.abhsia.telus.net) |
| 03:44.08 | irv999 | cor: np email me when you get an idea.. sethz@mac.com |
| 03:44.14 | *** part/#asterisk thetalon (n=Administ@pcp05736660pcs.norstn01.pa.comcast.net) |
| 03:45.16 | SarahEmm | file: FILE! |
| 03:45.47 | giesen | anyone have any ideas? |
| 03:45.53 | *** join/#asterisk SplasPood (n=sp@brooklyn.paravolve.net) |
| 03:46.51 | *** join/#asterisk MikeJ[Laptop] (n=ircatjer@d14-69-8-30.try.wideopenwest.com) |
| 03:46.51 | Corydon76-home | irv999: Will do. |
| 03:50.25 | Corydon76-home | MikeJ[Laptop]: did you notice that when you don't mark my new bugs [post-1.2], they get applied? |
| 03:50.49 | wunderkin | magic |
| 03:50.51 | *** join/#asterisk bmg505 (n=leon@rndf-146-4-172.telkomadsl.co.za) |
| 03:51.26 | Corydon76-home | No, I just think it's funny that the lack of that tag gets the patch into CVS, even if it was suggested after the cutoff date |
| 03:51.31 | *** join/#asterisk Pikoro (n=webmaste@db.sunny-net.ne.jp) |
| 03:51.59 | *** join/#asterisk axscode (n=magic@203.213.217.122) |
| 03:52.13 | Pikoro | anyone got a sample config file that will let me use incoming sip? I've about given up |
| 03:52.14 | Pikoro | heh |
| 03:52.19 | Pikoro | i had it working in the past |
| 03:52.27 | giesen | I cant get mine to work either |
| 03:52.31 | giesen | I can seem to dial out |
| 03:52.34 | Pikoro | but as soon as i got outgoing sip calls working, incoming works |
| 03:52.36 | giesen | but it goes straight to voicemail |
| 03:52.39 | Pikoro | err. incoming stopped working |
| 03:52.42 | Pikoro | all i get is a busy signal |
| 03:52.45 | giesen | the receiving phone keeps rining |
| 03:52.50 | giesen | *ringing |
| 03:53.06 | axscode | just want to ask.. what linux kernel needs for asterisk? is 2.4.27-2-386 is fine? |
| 03:53.12 | axscode | or do i have to update it to 2.6? |
| 03:53.34 | Pikoro | 2.4 should be fine |
| 03:53.45 | axscode | ok.. no modules is affected? |
| 03:54.00 | Pikoro | i don't think so |
| 03:54.07 | Pikoro | but i'm far from an asterisk expert |
| 03:54.49 | axscode | hmmm ok.. |
| 03:55.11 | axscode | coz i just remember.. that something might not work on 2.4... |
| 03:55.18 | axscode | just forgot though |
| 03:59.48 | HiltonT | any thoughts on Snom vs Polycom units? |
| 04:01.23 | axscode | is zaptel emulator needs 2.6 kernel. or is it ok to have 2.4 kernel of linux? |
| 04:02.10 | brookshire[home] | it is ok to have 2.4 |
| 04:02.12 | SarahEmm | HiltonT: from everyone i've talked to, polycom is the way to go |
| 04:02.17 | SarahEmm | i'm no expert tho |
| 04:02.23 | brookshire[home] | i think, hehe |
| 04:02.26 | jontow | 2.4 worked great for me, long ago |
| 04:02.27 | brookshire[home] | sarah! |
| 04:02.32 | SarahEmm | brookshire! meow! |
| 04:02.34 | brookshire[home] | jon! |
| 04:02.35 | HiltonT | I *do* like Polycom kit |
| 04:02.36 | jontow | 2.6.9+ works great for me now |
| 04:02.42 | brookshire[home] | heh |
| 04:02.42 | jontow | g'day matt :) |
| 04:02.46 | brookshire[home] | no comment on the 2.6.12 |
| 04:02.48 | *** join/#asterisk ManxPower (n=eric@slip-12-65-114-43.mis.prserv.net) |
| 04:02.50 | jontow | oh |
| 04:02.50 | HiltonT | used some of their high end video conferencing units - the old $25k ones :) |
| 04:02.54 | HiltonT | they were nice :) |
| 04:02.56 | jontow | i've only made it up to 2.6.11 on gentoo :D |
| 04:03.47 | marc324 | where do you set the odbcstorage variable? |
| 04:04.17 | HiltonT | neither are *cheap* units, so if we're going for serious pjhones, we may as well go for the serious, serious ones |
| 04:06.12 | morale | has anyone here got asterisk working with primus or vonage in canada? |
| 04:06.24 | irv999 | night all |
| 04:08.40 | jontow | ethereal is fat |
| 04:11.15 | SarahEmm | HiltonT: if you're going to serious, serious ones, get Cisco ;) |
| 04:12.00 | ManxPower | Polycom is pretty cool |
| 04:14.16 | Brijn | If my external link is working OK tomorrow, i'll probably get the 501.. Use the big polycom units and small poly conference phones. If they are the same quality, i'll be happy |
| 04:14.28 | hardwire | my snoms keep smoking themselves |
| 04:18.38 | *** join/#asterisk littleball (n=littleba@bb220-255-134-65.singnet.com.sg) |
| 04:21.32 | BrianR___ | I think I like the polycom phones slightly better than the cisco ones.. |
| 04:21.42 | BrianR___ | the polycom ones also do 802.1af |
| 04:21.55 | BrianR___ | the cisco ones do proprietary cisco poe only |
| 04:22.17 | BrianR___ | I rolled out about 20 GXP-2000's at my office - nothing but troube |
| 04:22.28 | BrianR___ | buying a whole bunch of ip-301's and ip-501's as replacements. |
| 04:23.12 | BrianR___ | Brijn: the ip501 speakerphone is very nice - fills a small office easily with loud, clear voice. And you don't have to yell to be heard by the other end. |
| 04:23.24 | desktophero | BrianR - what kind of trouble did you have? |
| 04:23.42 | *** join/#asterisk brookshire[home] (n=matt@esbrooks3.traveller.com) |
| 04:24.29 | Brijn | Good! We just immigrated to Canada and will use VoiP to phone with our parents.. It's nice to put it on speaker and just walk around |
| 04:25.34 | *** join/#asterisk jeffgus (n=jeffgus@2002:d856:c704:0:0:0:0:1) |
| 04:26.13 | alephcom | morale: primus is all on mgcp |
| 04:26.29 | *** join/#asterisk Piranha- (i=piranha@d198-166-226-58.abhsia.telus.net) |
| 04:26.31 | Brijn | morale: where in .ca are you? |
| 04:26.38 | alephcom | Alberta |
| 04:26.55 | alephcom | sorry, I answered the wrong line |
| 04:28.08 | Brijn | I'm in Vancouver and hope to get my connection from Dolphintel.ca tomorrow |
| 04:28.56 | *** part/#asterisk Ash (i=aaron@outofband.org) |
| 04:32.30 | BrianR___ | desktophero: phones get very hot in the area behind the lcd |
| 04:32.35 | BrianR___ | sometimes phones lock up |
| 04:32.40 | BrianR___ | dropped calls |
| 04:32.44 | BrianR___ | acoustic echo within the handset |
| 04:32.58 | BrianR___ | rj-h connector on base doesn't hold handset cord tightly, resulting in static |
| 04:33.08 | HiltonT | blutac :) |
| 04:33.24 | BrianR___ | line 1 button gets stuck lit and menu becomes inaccessible |
| 04:33.50 | BrianR___ | audio delay path is quite a bit longer than in cisco and polycom phones |
| 04:34.17 | BrianR___ | also, with 1.0.1.9, there is very bad accoustic echo in speakerphone mode |
| 04:34.28 | desktophero | holy crap! we just deployed about 10 in a remote office |
| 04:34.37 | Brijn | :) |
| 04:34.41 | BrianR___ | with 1.0.1.12, the handset echo problem seems to be a bit more pronounced (maybe they enabled AGC for the handset port?) |
| 04:35.22 | BrianR___ | a few of the sets got worse with time, eventually becoming so unreliable I had to pull them from service. |
| 04:35.59 | ManxPower | desktophero, you didn't test them first??? |
| 04:36.15 | desktophero | ManxPower - I tested them for a week in my office |
| 04:36.15 | BrianR___ | ManxPower: hey.. I did testing too - didn't have many of the problems with my test phone. |
| 04:36.19 | ManxPower | Heck, we ran 3 month tests with several IP phones before deciding on Polycom |
| 04:36.26 | desktophero | I had issues with the speakerphone |
| 04:36.39 | desktophero | contacted Grandstream and they gave me the link to the latest bin for firmware |
| 04:36.43 | BrianR___ | I suspect that a lot of the problems, like the PoE supply getting hot, are manufacturing defects. |
| 04:36.47 | ManxPower | We deployed them to users for 2 months, did a month of testing ourselfes |
| 04:36.51 | desktophero | upgraded, rebooted, and the echo issue went away |
| 04:37.36 | *** join/#asterisk cripito (n=ncripito@c-67-161-130-59.hsd1.co.comcast.net) |
| 04:37.37 | cripito | hi |
| 04:37.40 | desktophero | i have been using the test phone without the same issues as BrianR - i have used it for about 1.5 months |
| 04:37.57 | cripito | <PROTECTED> |
| 04:38.03 | BrianR___ | The polycom ip301/501 are much more mature products, and seem to have a lower defect rate so we're ordering more and phasing out the gxp-2000's. |
| 04:38.19 | desktophero | we may have to go that route |
| 04:38.34 | desktophero | are there any other models - outside of polycom or cisco - that anyone recommends? |
| 04:38.48 | desktophero | or we may just go with a softphone |
| 04:38.51 | BrianR___ | desktophero: the polycom seems to be one of the better models that's not priced crazy insane. |
| 04:38.59 | desktophero | we have had some pretty decent success with SJPhone |
| 04:39.03 | cripito | anyone that can answer a question from h323? |
| 04:39.08 | cripito | ooh323 |
| 04:39.16 | BrianR___ | the cisco phonesa re nice too, but they're too expensive. |
| 04:39.19 | axscode | so all modules works with 2.4 linux kernel? |
| 04:39.25 | BrianR___ | anyway. sleep time. |
| 04:39.47 | axscode | all features works on 2.4linux kernel right? |
| 04:40.00 | desktophero | axscode - i have Asterisk 1.0.9 running on FC1 and FC4 - different servers. |
| 04:40.08 | alephcom | I'm using the 2.4 kernel. I'll grab the exact versoin |
| 04:40.13 | desktophero | are there specific features you are shooting for axscode? |
| 04:40.14 | axscode | Fedora core runs 2.4 kernel? |
| 04:40.20 | desktophero | FC1 |
| 04:40.24 | ManxPower | The problem with Cisco is you have to pay extra for SIP firmware and extra for a power supply. |
| 04:40.30 | morale | has anyone used the voip provider inphonex? |
| 04:40.32 | desktophero | FC3 was the 2.6 switch - I believe |
| 04:40.53 | axscode | desk.. i remember when im installing something.. i dont know whats that.. it requires 2.6 |
| 04:40.54 | morale | inphonex seems to be a smaller little company based in miami, florida.. |
| 04:40.58 | desktophero | Manx - is there a huge downfall with running skinny as opposed to SIP on Cisco? |
| 04:41.00 | axscode | is ztdummy requires 2.6 kernel? |
| 04:41.06 | alephcom | axscode: Linux version 2.4.22-1.2115.nptl |
| 04:41.22 | alephcom | I'm running Fedora core one on that box with absolutely no issues. |
| 04:41.37 | desktophero | ztdummy "CAN" run against one of the USB controllers - I read that somewhere - on a 2.4 kernel |
| 04:41.38 | axscode | ok i have 2.4.27 |
| 04:41.40 | alephcom | no, at least it did not used to |
| 04:41.42 | desktophero | I think...;) |
| 04:41.58 | ManxPower | desktophero, Yes. |
| 04:42.03 | desktophero | I think it is the OPEN USB controller and not the UNIVERSAL - or vice versa |
| 04:42.08 | alephcom | desktophero: I'm running it against a usb controller myself. |
| 04:42.13 | Katty | hmm |
| 04:42.15 | ManxPower | skinny is not well supported by Asterisk. |
| 04:42.26 | desktophero | which usb host controller aleph? |
| 04:42.35 | axscode | coz im just trying to emulate zaptel using ztdummy time.. so im just curious.. so that im not going to install over and over again. |
| 04:42.37 | alephcom | let me look |
| 04:43.03 | desktophero | I was able to get some speed dial buttons working on a 7960, but that was about it - the sound quality was outstanding. tough sell is the ~$260 price tag |
| 04:43.24 | axscode | or its just the voice modem that needs 2.6 kernel? ? |
| 04:43.42 | jdv79 | how does one test a var that for definedness? |
| 04:43.44 | desktophero | what Digium hardware are you running acs? |
| 04:43.59 | alephcom | usb-uhci.c: v1.275:USB Universal Host Controller Interface driver |
| 04:44.02 | jdv79 | how does one test a var for definedness is what i meant to type |
| 04:44.05 | Katty | hmm |
| 04:44.07 | SarahEmm | polycom is said to have great sound quality too desktophero :) |
| 04:44.14 | Pikoro | damnit.. what is the simplest way to get incoming sip calls to route to extension 200? |
| 04:44.22 | ManxPower | jdv79, I don't think you can. You can only test if it's empty of not. |
| 04:44.23 | Pikoro | i have been trying to figure this out for like a week |
| 04:44.24 | Pikoro | heh |
| 04:44.49 | jdv79 | i do that but i think Read() is killing the var on me |
| 04:44.53 | ManxPower | Works on all Asterisk verisons: $[X${VAR} = X] |
| 04:45.02 | jdv79 | i'mm getting warnings about it:( |
| 04:45.11 | ManxPower | that would test to see if ${VAR} was empty or not. |
| 04:45.29 | jdv79 | what are the Xs? i didn't see that |
| 04:45.32 | jdv79 | in the docs |
| 04:45.42 | desktophero | Pikoro - are you routing the call through your contexts correctly? does the call go "anywhere"? |
| 04:45.47 | ManxPower | Pikoro, Where are the incoming SIP calls from? |
| 04:46.24 | ManxPower | exten => 200,1,Dial(whatever). |
| 04:46.31 | jdv79 | ManxPower, that somehow works - thanks |
| 04:46.35 | ManxPower | in the contex the call is coming in. |
| 04:46.38 | jdv79 | is that doc'd anywhere but the code? |
| 04:46.47 | ManxPower | jdv79, it's an old shell scripting trick. |
| 04:47.15 | ManxPower | If ${VAR} is "FNORD" then it would test to see if XFNORD = FNORD |
| 04:47.21 | desktophero | extending Manx's description for the SIP call........Dial(SIP/200)....assuming 200 is registered |
| 04:47.22 | *** join/#asterisk justinu (n=justinu@cpe-72-129-86-208.socal.res.rr.com) |
| 04:47.24 | ManxPower | If the variable is empty than it would test X = X |
| 04:47.37 | jdv79 | oh... |
| 04:47.38 | jdv79 | lame |
| 04:47.48 | ManxPower | desktophero, There is NO reason for extension 200 to be sip user "200" |
| 04:47.54 | desktophero | true |
| 04:47.56 | ManxPower | When you do it that way it limits your thinking. |
| 04:47.59 | jdv79 | ther should be an explicit non-clever/hidden want to do that |
| 04:48.00 | justinu | hey anyone have polycom sip 1.6.2 firmware? |
| 04:48.04 | justinu | it's not here: http://www.freedomphones.net/polycom/files/ |
| 04:48.08 | jdv79 | way |
| 04:48.14 | desktophero | going the simpliest route...that's all |
| 04:48.21 | ManxPower | We actually make our SIP userids be the MAC address of the device with an -a -b -c, etc suffic for multiple lines on a device |
| 04:48.24 | HiltonT | well, we can now resell Polycom phones in .au :) |
| 04:48.40 | ManxPower | justinjas, give it a month or two, it will show up there. |
| 04:48.44 | justinu | well, can you send me a firmware image? :) |
| 04:48.51 | justinu | hmm... a month or two? :( |
| 04:48.57 | wunderkin | ManxPower and you think his way is complicated |
| 04:49.33 | ManxPower | justinjas, why do you need it Right Now? |
| 04:49.46 | justinu | i dunno, i like new stuff |
| 04:50.01 | ManxPower | I hate new stuff. It never works right. |
| 04:50.06 | SarahEmm | justinjas: it's just bugfixes. |
| 04:50.08 | jdv79 | amen |
| 04:50.14 | SarahEmm | justinjas: afaik 1.6.2 does not add any new features. |
| 04:50.16 | wunderkin | justinu: i bet there would be someone there at astricon |
| 04:50.27 | justinu | yeah, i'll ask tomorow |
| 04:50.32 | justinu | were you there? |
| 04:50.39 | wunderkin | not me |
| 04:51.43 | ManxPower | I was pretty annoyed with the last polycom firmware upgrade we did. |
| 04:51.48 | SarahEmm | what bugs have you seen? |
| 04:51.52 | SarahEmm | i'm running 1.5.2 and haven't found any |
| 04:51.59 | justinu | which firmware would you recommend? |
| 04:52.05 | ManxPower | They changed things in weird ways |
| 04:52.22 | SarahEmm | ahh |
| 04:52.23 | ManxPower | But we HAD to upgrade because of the call waiting problem |
| 04:52.27 | SarahEmm | the menus are all different in 1.5, yeah. |
| 04:52.39 | *** join/#asterisk awannabe (n=gti@ip24-251-130-47.ph.ph.cox.net) |
| 04:52.43 | SarahEmm | as this was my first polycom set, and i upgraded to 1.5 as soon as i got it, i was okay with that ;) |
| 04:53.05 | Pikoro | desktophero i believe so |
| 04:54.06 | jdv79 | thanks ManxPower - that was driving me nuts:) |
| 04:54.12 | jdv79 | it still does but less now |
| 04:54.21 | ManxPower | jdv79, it was diriving my USERS nuts. |
| 04:54.21 | Pikoro | ok, so the call hits my asterisk box, then goes to my sip.con for the context right? |
| 04:54.28 | ManxPower | Which in turn drove me nuts |
| 04:54.39 | Pikoro | then the context refers to extensions.conf to figure out where to send it no? |
| 04:54.40 | ManxPower | Pikoro, simple answer: yes. |
| 04:54.52 | Pikoro | ok, that's what i figured |
| 04:54.58 | ManxPower | The call SHOULD have a destination extension associated with it. |
| 04:55.00 | Pikoro | i think the whole dialpan thing is what's kicking my arse |
| 04:55.09 | ManxPower | Pikoro, it does that to all newbies |
| 04:55.14 | Pikoro | heh |
| 04:55.20 | desktophero | Amen to Manx |
| 04:55.33 | Pikoro | i only know about PSPs, this asterisk thing is killing me :D |
| 04:55.35 | Pikoro | i got outbound working |
| 04:55.39 | Pikoro | and used to have inbound working |
| 04:55.40 | justinu | pikoro, you'll get it |
| 04:55.45 | Pikoro | but all i get when i call in is a busy signal |
| 04:55.46 | justinu | pikoro: get the oreilly book |
| 04:55.55 | desktophero | it took me about a week from install to listen to the demo - that included our carrier placing the wrong build on the T1...but anyway |
| 04:55.58 | Pikoro | i know i only need like 1 line to fix this whole thing |
| 04:55.59 | ManxPower | Pikoro, the console is your friend. watch the console. |
| 04:56.07 | Pikoro | i have been |
| 04:56.32 | ManxPower | Pikoro, How is the call coming in? SIP? |
| 04:56.34 | desktophero | Pikoro are you getting "unabel to create channel type of... |
| 04:56.37 | Pikoro | yes, sip |
| 04:56.46 | desktophero | int he cli? |
| 04:56.51 | Pikoro | i can see the incoming call on the console |
| 04:56.57 | Pikoro | but then the call just gets destroyed |
| 04:56.57 | Katty | hmm |
| 04:56.58 | ManxPower | then you should see the call come in on the CLI |
| 04:57.06 | ManxPower | Pikoro, what is the error message on the console? |
| 04:57.11 | *** join/#asterisk escribzz (n=escribzz@wsip-24-249-174-19.ph.ph.cox.net) |
| 04:57.15 | desktophero | there must be something else on the CLI |
| 04:57.16 | Pikoro | there doesn't seem to be one |
| 04:57.23 | Pikoro | this is why i'm getting frustrated |
| 04:57.23 | Pikoro | heh |
| 04:57.39 | morale | has anyone used the provider "inphonex" yet? .. also.. is asterisk ever going to support mgcp? |
| 04:57.43 | ManxPower | Keep in mind that in SIP EVERYTHING is a call. Registrations, pings, notifications, they are all "calls" |
| 04:58.04 | desktophero | when the call comes in, does it immediately go to congestion? |
| 04:58.06 | ManxPower | Pikoro, stop asterisk. Start it as "asterisk -cvvv" (no d's) |
| 04:58.24 | escribzz | Hi has anyone had any luck multi-homing sip services on asterisk? |
| 04:58.25 | awannabe | escribzz: told ya :) |
| 04:58.31 | ManxPower | if you start it with d's you will get all sorts of debug crap that will confuse you. |
| 04:58.33 | escribzz | l0lawannabe |
| 04:58.35 | escribzz | I missed it |
| 04:58.49 | alephcom | morale: probably sometime but.... what are you needing for a .ca provider? |
| 04:59.02 | morale | alephcom, yes.. |
| 04:59.02 | *** join/#asterisk brookshire[home] (n=matt@esbrooks3.traveller.com) |
| 04:59.33 | Pikoro | ok.. got it restarted |
| 04:59.49 | Pikoro | i'm gonna tail /var/log/asterisk/full |
| 04:59.58 | Pikoro | so i can stop the output from scrolling so damn fast ;p |
| 05:00.37 | Pikoro | hmm.. |
| 05:00.49 | justinu | use less |
| 05:00.58 | Brijn | ManxPower: I'm a newbie as well, and started looking at sip.conf after you remark that you should not think in extention terms.. [xxx] header defines the real name.. Later you sue things like DIal(SIP/xxx)? |
| 05:01.16 | Pikoro | ok, now there's practically no output in the logs |
| 05:01.36 | Brijn | But where do you define the actual extention that is being sent to the other phone? |
| 05:01.49 | justinu | extensions.conf |
| 05:02.16 | Pikoro | ok, i get the normal proxy authentication required stuff |
| 05:02.24 | Brijn | If I dial nr 200, it will check the dialplan and find what DIal(xxx) it needs to go out.. That phone receives the call and will show a number.. Where does that num,ber come from |
| 05:02.24 | Pikoro | then the invite |
| 05:02.40 | justinu | or you say: dial(sip/xxx/${EXTEN}) |
| 05:02.41 | Pikoro | then scheduling destruction of the call |
| 05:03.04 | Pikoro | no hangup or errors |
| 05:03.23 | Brijn | justinu: ?? What does that mean? |
| 05:04.30 | Pikoro | any ideas? |
| 05:04.49 | desktophero | Pikoro - internally, are you able to dial exten 200? |
| 05:04.52 | Pikoro | yes |
| 05:04.56 | Pikoro | all internal works |
| 05:05.09 | Pikoro | making outbound calls to the pstn or sip numbers work |
| 05:05.13 | Pikoro | just no inbound |
| 05:05.17 | desktophero | sounds like the inbound context is not routing through to the internal sip exten |
| 05:05.39 | Pikoro | yah, but i don't get anything from asterisk like all circuts are busy, or the extension is not available, or whatnot |
| 05:05.43 | Pikoro | just straight to busy signal |
| 05:06.03 | desktophero | are you using an 's' extension for the inbound calls? |
| 05:06.21 | Pikoro | no, just trying to go to an internal extension |
| 05:06.28 | Pikoro | should i route it to start? |
| 05:06.40 | Pikoro | and if so, how could i do that? ;D |
| 05:06.44 | desktophero | how are the calls coming? how are they identified? is this a DID? |
| 05:07.08 | Pikoro | well, the whole DID thing confuses me, but I have no DIDs defined |
| 05:07.11 | desktophero | is this coming in over a Digium card or from another SIP server? |
| 05:07.17 | *** join/#asterisk Blade (i=i@am.blade.nu) |
| 05:07.19 | Pikoro | from another sip server |
| 05:08.10 | desktophero | so you are doing something like Dial(SIP/200@sip_server_ip_address:5060)? from the remote server? |
| 05:08.53 | *** join/#asterisk Igbothom (n=HiltonT@static-84.217.240.220.dsl.comindico.com.au) |
| 05:09.04 | Pikoro | should I be doing that? |
| 05:09.15 | Pikoro | that would be in extensions.conf right? |
| 05:09.19 | Igbothom | well, that was fun! |
| 05:10.15 | *** join/#asterisk outofjungle (n=outofjun@203.101.39.98) |
| 05:10.18 | desktophero | how are you doing it now? |
| 05:11.09 | Brijn | Hmmm, I still don't see where the link is that defines the sending number.. How does the receiving end know the senders numbe |
| 05:11.12 | Brijn | r |
| 05:13.34 | desktophero | Pikoro - have you thought about an IAX2 trunk...;) |
| 05:14.14 | Igbothom | damn xchat is playing silly buggers - bbiab |
| 05:14.30 | *** part/#asterisk Igbothom (n=HiltonT@static-84.217.240.220.dsl.comindico.com.au) |
| 05:14.30 | marc324 | the voicemail storage routine, uses the "%default/" ... gives error. should not be ' ? |
| 05:14.34 | Pikoro | ok, i have exten => s,1,Answer |
| 05:14.39 | *** join/#asterisk brookshire[home] (n=matt@esbrooks3.traveller.com) |
| 05:14.42 | Pikoro | i am in japan.. nobody uses asterisk here |
| 05:15.03 | brookshire[home] | :/ |
| 05:15.04 | Pikoro | after an extensive and exhausting day with google japan, i found nobody who has gotten asterisk to work over here |
| 05:15.13 | justinu | i met some guys at astricon using asterisk in japan |
| 05:15.24 | brookshire[home] | there are a few people using asterisk in japan |
| 05:15.24 | Pikoro | tell them to call me |
| 05:15.25 | Pikoro | haha |
| 05:15.47 | desktophero | Are you able to register to the other server and call the original server? |
| 05:15.49 | brookshire[home] | matt f wrote support for J1 |
| 05:15.50 | brookshire[home] | :) |
| 05:16.05 | Pikoro | yes |
| 05:16.30 | *** join/#asterisk Igbothom (n=HiltonT@static-84.217.240.220.dsl.comindico.com.au) |
| 05:16.41 | desktophero | So for s,2 you could NoOp(${CALLERIDNUM}) to see if your call shows up in the CLI |
| 05:16.42 | Igbothom | and back with a new outlook on life |
| 05:16.45 | Igbothom | well, maybe not |
| 05:16.52 | wunderkin | brookshire: it must be nice working for a relatively small company with a million matts |
| 05:17.01 | justinu | heh\ |
| 05:17.01 | Pikoro | i'll give that a shot |
| 05:17.33 | brookshire[home] | heh |
| 05:17.40 | wunderkin | brookshire: you are the web guy right? do you take suggestions on mantis or who does that go to? |
| 05:17.41 | brookshire[home] | <-- matt b ;) |
| 05:17.56 | brookshire[home] | mantis and I are a touchy subject |
| 05:18.14 | wunderkin | all luvy-dovy? |
| 05:18.35 | brookshire[home] | if i could hold an 9mm to it and pull the trigger, i would ;) |
| 05:18.36 | Pikoro | ok, it doesn't seem to be making it that far |
| 05:18.41 | Pikoro | this is wierd |
| 05:18.42 | wunderkin | heh |
| 05:18.51 | Pikoro | just a busy signal |
| 05:19.01 | brookshire[home] | anyways.. what's wrong with mantis now? |
| 05:19.06 | Pikoro | it's not making it to my context it looks like |
| 05:19.33 | brookshire[home] | pikoro: sip, iax, zap? |
| 05:19.36 | desktophero | without seeing the actual extensions.conf file, this could be a tough one |
| 05:19.48 | wunderkin | brookshire: nothing, just a feature request.. adding a 'not equal to' in the status search would help a lot.. mostly to be able to search for something not closed |
| 05:19.56 | *** join/#asterisk _Thor (i=Christia@dialup-4.237.44.249.Dial1.NewYork1.Level3.net) |
| 05:20.15 | desktophero | Pikoro - where did you add the 's' extension? |
| 05:20.15 | Pikoro | brookshire[home]: sip |
| 05:20.32 | brookshire[home] | ugh.. idon't know sip :( |
| 05:20.41 | brookshire[home] | i really need to learn one day, hehe |
| 05:20.44 | Pikoro | [from-plala] which is included in from-trunk |
| 05:21.15 | desktophero | stupid question - but after making the changes in the extensions.conf, did you reload it in the CLI? |
| 05:21.17 | brookshire[home] | wunderkin: you can search by status |
| 05:21.19 | Pikoro | yes |
| 05:21.20 | Pikoro | heh |
| 05:21.26 | desktophero | had to ask ;) |
| 05:21.30 | Pikoro | i know :D |
| 05:21.39 | Pikoro | i do tech support during the day |
| 05:21.39 | brookshire[home] | oh wait.. that's a different status |
| 05:21.39 | brookshire[home] | wtf |
| 05:21.39 | _Thor | anyone has configured a Sangoma card? |
| 05:21.39 | brookshire[home] | lol |
| 05:22.21 | desktophero | and 200 is defined in the from-plala context? |
| 05:22.36 | Pikoro | no |
| 05:22.43 | brookshire[home] | wunderkin: you can search by resolution, is that what you are talking about? |
| 05:22.48 | Pikoro | i am just trying to get a message from the asterisk box right now |
| 05:22.48 | desktophero | is 200 defined somewhere? |
| 05:22.52 | Pikoro | yes |
| 05:22.57 | Pikoro | in sip.conf |
| 05:23.15 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 05:23.18 | desktophero | but 200 is not a destination in the extensions.conf file? this may be a portion of the culprit.... |
| 05:23.21 | Pikoro | just trying to get a NoOp right now |
| 05:23.28 | _Thor | Has anyone configured a Sangoma card? |
| 05:23.30 | wunderkin | brookshire: well, when you are going to submit a bug report, you have to search to see if it already exists.. so that bug could already be submitted and in a status other than new or whatever |
| 05:23.36 | desktophero | sorry Thor...nope |
| 05:24.03 | desktophero | can you define a 200 in the extensions.conf file? |
| 05:24.11 | desktophero | just a 200,1,Answer |
| 05:24.13 | Pikoro | like, s,2,Dial(SIP/200,25,Ttr) ? |
| 05:24.34 | desktophero | then a 200,2,Dial(SIP/200) |
| 05:24.44 | Pikoro | i had that in there at one point |
| 05:24.50 | brookshire[home] | oh.. i se |
| 05:24.54 | brookshire[home] | i see |
| 05:25.07 | desktophero | what version of Asterisk? |
| 05:25.08 | brookshire[home] | personally.. i think all bugs should be confirmed on the mailing lists, lol |
| 05:25.13 | desktophero | just for my own info ;) |
| 05:25.18 | pauldy | what ports would I need to forward for iax2 if I wanted to run an asterisk server behind a firewall and connect to it remotely |
| 05:25.22 | Pikoro | i think the problem right now is that asterisk is receiving the sip request, but nothing is answering |
| 05:25.23 | wunderkin | brookshire: and if you have been a good boy and already updated to the latest head, then any of the closed bugs wouldn't matter so you are scrolling through all bugs including closed :D |
| 05:25.25 | Pikoro | so i get a busy signal |
| 05:25.34 | desktophero | agreed Pikoro |
| 05:25.50 | Pikoro | it's like my context isn't getting read |
| 05:25.50 | desktophero | you could try a _2XX,1,Answer |
| 05:26.02 | pauldy | just the udp 4569? |
| 05:26.05 | desktophero | this would allow all 2## extensions to be answered |
| 05:26.13 | brookshire[home] | wunder: i understand |
| 05:26.19 | desktophero | then a _2XX,2,Dial(SIP/${EXTEN}) |
| 05:26.22 | brookshire[home] | i might whip something up someday |
| 05:26.29 | wunderkin | cool, just a thought |
| 05:26.32 | brookshire[home] | my priority is to fix thevoice right now though |
| 05:26.36 | brookshire[home] | it's a great idea :) |
| 05:26.38 | wunderkin | thevoice? |
| 05:26.45 | *** part/#asterisk Uberbot (n=Uberbot@69.252.219.76) |
| 05:26.46 | brookshire[home] | thevoice.digium.com |
| 05:26.50 | wunderkin | ah hmm |
| 05:27.23 | desktophero | brookshire - from your conversation - do you work on the digium.com website? |
| 05:27.26 | wunderkin | figured it was for ordering recordings from allison :P |
| 05:27.49 | _Thor | desktophero: Thank you |
| 05:28.27 | desktophero | thor - i have been in contact with them...perhaps one of their DS3 cards will come my way once they are channelized...unless Digium comes out with theirs first ;) |
| 05:28.44 | wunderkin | ahhh haha i also noticed that puregig isn't hosting the site anymore? |
| 05:28.47 | brookshire[home] | yeah... i do |
| 05:28.59 | brookshire[home] | i do mostly internel web stuff though |
| 05:29.02 | brookshire[home] | and the shopping cart |
| 05:29.06 | Pikoro | ok, stupid question, but where does my context line go? for the default |
| 05:29.10 | Pikoro | sip.conf under general right? |
| 05:29.21 | desktophero | brookshire - just wanted to say I like the changes over the last few months..much easier to navigate and such |
| 05:29.39 | brookshire[home] | awesome :) |
| 05:29.43 | brookshire[home] | it's about to get better :) |
| 05:30.05 | desktophero | Pikoro - in sip.conf, my general context is default |
| 05:30.15 | brookshire[home] | yeah.. the digium.com site is now hosted internal :) |
| 05:30.19 | wunderkin | <PROTECTED> |
| 05:30.28 | desktophero | within default in extensions.conf, place the lines I sent earlier in that particular context |
| 05:30.37 | Pikoro | ok |
| 05:30.37 | _Thor | the low profile rumor is that sangomas are better than digiums |
| 05:30.52 | desktophero | i look forward to it brookshire |
| 05:30.53 | marc324 | ne1 tried odbcstorage with postgres? |
| 05:30.54 | jdv79 | how do i know when someone answers the line? |
| 05:30.55 | X-Rob | s'not really 'low profile' |
| 05:31.01 | marc324 | it gives: SQL Execute error! |
| 05:31.03 | jdv79 | from within a context |
| 05:31.08 | X-Rob | the new shark cards are going to significantly kick arse. |
| 05:31.20 | marc324 | i think it uses " instead of ' in the sql |
| 05:31.30 | Pikoro | ok |
| 05:31.31 | desktophero | i think the hardware echo on digium will make some ground on that. from what I think I understand, sangoma is a little better for echo cancellation and cpu load |
| 05:31.47 | Pikoro | grr.. damn this busy signal |
| 05:31.48 | Pikoro | heh |
| 05:31.54 | X-Rob | desktophero - new shark cards have hardware echo on t1/e1/j1 _and_ FXS/FXO |
| 05:32.00 | desktophero | Pikoro - just make sure you reload extensions :0 ha! |
| 05:32.01 | wunderkin | _Thor: the last comparison i saw, each has their advantage in 1 type of connection.. can't remember.. this was on the new model 4 port echo cancel cards.. or was it w/o echo? either way.. |
| 05:32.07 | Pikoro | i did a full reload |
| 05:32.11 | desktophero | X-Rob - from Digium??? |
| 05:32.16 | X-Rob | no, sangoma |
| 05:32.32 | desktophero | but what about the channelized DS3 PCI cards - oh yeah! |
| 05:32.48 | desktophero | sorry for the nose-picking moment there folks. I just can't wait for that |
| 05:32.56 | desktophero | my life will be SOOOO much easier |
| 05:33.22 | brookshire[home] | desktop: ds3 mux :) |
| 05:33.46 | X-Rob | DS3 is 672 channels? *boggle* |
| 05:33.54 | desktophero | i will have so much more rack space and less cable management....I do look forward to that day....ho hum |
| 05:33.55 | X-Rob | if you're running 672 channels into an asterisk box, you're nuts. |
| 05:33.56 | _Thor | yeah, but people who love digium (as I do) feel that purchasing a digium card is like paying back a little for their efforts in giving * for free. ... But sangoma has more experience in manufacturing |
| 05:34.12 | escribzz | thats what I was thinking |
| 05:34.18 | FuriousGeorge | so a tdm fxs officially supports one analog phone, but how many phones worth of voltage does an ATA provide? |
| 05:34.33 | desktophero | i don't need to use all the channels....X-Rob...but it's nice to know it's there |
| 05:34.39 | X-Rob | Most ATA's can supply 3 REN |
| 05:34.50 | wunderkin | _Thor: yeah i think thats how i feel too but they always catch back up |
| 05:35.27 | Pikoro | grr |
| 05:35.29 | FuriousGeorge | X-Rob: how many REN does your average phone require? |
| 05:35.32 | wunderkin | the 2nd gen cards work for me |
| 05:35.48 | ManxPower | And most phones use very little REN |
| 05:36.01 | X-Rob | FuriousGeorge - An old style clapper-bell uses 3 ren. Most new phones use .5. It's often written on the phone somwhere |
| 05:36.02 | desktophero | what's up Pikoro? |
| 05:36.18 | X-Rob | Stands for 'Ringer Equivalence Number' I think |
| 05:36.20 | FuriousGeorge | how many ren in a TDM |
| 05:36.25 | FuriousGeorge | FXS |
| 05:36.25 | Pikoro | ok, under the default context i have this |
| 05:36.36 | X-Rob | I just picked up one phoen and it had 'REN=1.5' on it |
| 05:36.37 | Pikoro | s,1,Playback(vm-goodbye) |
| 05:36.39 | brookshire[home] | it's on asterisk.org/glossary |
| 05:36.40 | brookshire[home] | hehe |
| 05:36.45 | jdv79 | how do i tell the context to wait until the other end picks up? |
| 05:36.47 | Pikoro | s,2,Macto(hangupcall) |
| 05:36.55 | Pikoro | err..Macro |
| 05:37.06 | ManxPower | X-Rob, That MUST be an analog phone with a bell |
| 05:37.07 | Pikoro | then, in sip.conf i have context=default |
| 05:37.26 | FuriousGeorge | my clock phone is one REN |
| 05:37.29 | X-Rob | ManxPower - Nup. 5 year old phone with caller ID and an external plugpack. |
| 05:37.38 | brookshire[home] | The REN of a Digium FXS board is 5 (representing "extension," i.e., parallel-connected telephones). |
| 05:37.46 | kb1_kanobe | REN, Ringer Equivalence Number, is a measure of the load presented to the CO by the device relative to 1 good old fashioned electromechanical ringer. Too many REN and the CO can't drive the line, thus all your mechanical phones would stop ringing. |
| 05:37.58 | FuriousGeorge | X-Rob: got any idea how many ren an FXS module of a TDM supplies |
| 05:38.08 | brookshire[home] | The REN of a Digium FXS board is 5 (representing "extension," i.e., parallel-connected telephones). |
| 05:38.09 | brookshire[home] | :_ |
| 05:38.10 | X-Rob | FuriousGeorge - brookshire just said 5 |
| 05:38.10 | brookshire[home] | :) |
| 05:38.20 | FuriousGeorge | oh |
| 05:38.24 | desktophero | Pikoro - try an answer for the first step |
| 05:38.34 | jdv79 | huh, why would a dialout through a FXS line just "pickup" right away? |
| 05:38.38 | Pikoro | ok |
| 05:38.39 | jdv79 | bad signalling? |
| 05:38.45 | FuriousGeorge | brookshire[home]: thats a lot of REN |
| 05:38.45 | Pikoro | so s,1,Answer |
| 05:38.53 | jdv79 | if i dial a local extention it works fine |
| 05:39.14 | marc324 | is there a bug with odbcstorage and postgresql? the query from asterisk uses double quotes, instead of single quote |
| 05:39.51 | brookshire[home] | so basically... 5 old style phones |
| 05:39.52 | brookshire[home] | :) |
| 05:39.52 | ManxPower | jdv79, You mean dialout thru an FXO. |
| 05:39.54 | ManxPower | ~fxofxs |
| 05:39.57 | jbot | extra, extra, read all about it, fxofxs is An FXO port expects to receive dialtone and receive ring voltage. An FXS port expects to provide dialtone and provide ring voltage. |
| 05:39.58 | jdv79 | i think so |
| 05:40.02 | Pikoro | still just a busy signal |
| 05:40.07 | jdv79 | i have an analong line with a X100P |
| 05:40.10 | ManxPower | Analog FXO ports are considered Answered as soon as they finish dialing. |
| 05:40.15 | jdv79 | i just want to shove calls down it |
| 05:40.27 | brookshire[home] | ~x100p |
| 05:40.28 | jbot | well, x100p is an obsolete card. you don't want to bother trying to make it (or any of the "digium compatible" clones work. Get a TDM01P, you will save your sanity. |
| 05:40.39 | brookshire[home] | :D |
| 05:40.41 | ManxPower | jbot is insane |
| 05:40.44 | X-Rob | ~sangoma |
| 05:40.45 | jbot | i heard sangoma is a company that makes PRI cards the way Digium should have done it in the first place.... |
| 05:40.45 | jdv79 | but i have it |
| 05:40.47 | jdv79 | ;) |
| 05:40.53 | X-Rob | *laugh* |
| 05:41.10 | brookshire[home] | anyways... |
| 05:41.15 | morale | the x100p works fine. |
| 05:41.16 | *** join/#asterisk Corndawg_ (n=notreall@c-66-176-249-51.hsd1.fl.comcast.net) |
| 05:41.19 | ManxPower | LOL! |
| 05:41.22 | ManxPower | That is SO true. |
| 05:41.35 | brookshire[home] | so where is the so called 'shark' |
| 05:41.45 | brookshire[home] | wasn't it suppost to be out 2 months ago? |
| 05:41.50 | jdv79 | how can i get it to consider it answered when the far end picks up? |
| 05:42.10 | *** join/#asterisk Rez (i=lorez@freenode/staff/lorez) |
| 05:42.10 | ManxPower | jdv79, You can't. It's analog. |
| 05:42.21 | X-Rob | jdv79 - how do you plan to figure that out? |
| 05:42.24 | jdv79 | so i can shoot calls out that line at all? |
| 05:42.25 | ManxPower | Analog lines do not provide that information in a machine readable way. |
| 05:42.35 | ManxPower | jdv79, Huh? |
| 05:42.43 | desktophero | signing off...goodnight all |
| 05:42.47 | jdv79 | i want to dial out with that line, that's all |
| 05:43.03 | ManxPower | jdv79, so what's the problem then? |
| 05:43.06 | X-Rob | so you Dial(Zap/2,number) |
| 05:43.21 | ManxPower | Or Dial(Zap/2/5551212) |
| 05:43.22 | jdv79 | but i have a context that plays a file hanging off that |
| 05:43.28 | X-Rob | after it's dialed, the Dial'ed call is connected to the extension that requested it |
| 05:43.39 | jdv79 | it plays the file before i even get the ring at the other end |
| 05:43.44 | X-Rob | So do a wait |
| 05:44.11 | ManxPower | jdv79, So you DON'T just want to "shoot calls out that line". you want to do more stuff. |
| 05:44.11 | jdv79 | how do i know how long to wait? |
| 05:44.28 | jdv79 | i want to "autodial" out the line |
| 05:44.40 | ManxPower | jdv79, You don't If you want to know when the far end answers then you MUST use a PRI. (or a provider that uses PRI, they all should) |
| 05:44.56 | ManxPower | jdv79, no, you want to send calls out the line and play a message when the far end picks up. |
| 05:44.58 | jdv79 | i was hoping there was something i wasn't aware of that would allow it |
| 05:45.14 | jdv79 | its just for testing, i should have my DS3 in a few months:( |
| 05:45.18 | ManxPower | jdv79, there really isn't. |
| 05:45.26 | ManxPower | Analog has significant limitations. |
| 05:45.42 | ManxPower | Too bad Asterisk doesn't support DS3s |
| 05:45.52 | brookshire[home] | well... |
| 05:45.53 | jdv79 | well, via SIP:) |
| 05:45.55 | jdv79 | whatever |
| 05:46.06 | brookshire[home] | ds3 mux -> t1s :) |
| 05:46.09 | jdv79 | when the support it though i'll be all over that |
| 05:46.21 | jdv79 | any word on that card? |
| 05:46.29 | FuriousGeorge | brookshire[home]: i know this is coming in a little late but: or 10 new style phones. i always thought an fxs could dp 3 phones tops |
| 05:46.36 | jdv79 | right now we're using a nextone switch as a GW |
| 05:46.57 | ManxPower | jdv79, I don't believe Digium will ever actually release a DS3 card, but Digium claims they are working on it. |
| 05:47.04 | jdv79 | figures |
| 05:47.15 | jdv79 | how many t1s in a ds3 again? |
| 05:47.34 | FuriousGeorge | !DS3 |
| 05:47.34 | jdv79 | 15? |
| 05:47.37 | X-Rob | 672 channels |
| 05:47.39 | FuriousGeorge | ~ds3 |
| 05:47.41 | jbot | Modem/ISDN sharing tool for linux. URL: http://www.Informatik.Uni-Oldenburg.DE/~prefec2/ |
| 05:48.04 | ManxPower | I told you jbot is insane |
| 05:48.05 | X-Rob | 28 T1s in a DS3 |
| 05:48.10 | X-Rob | ~ds3 |
| 05:48.11 | jbot | Modem/ISDN sharing tool for linux. URL: http://www.Informatik.Uni-Oldenburg.DE/~prefec2/ |
| 05:48.13 | jdv79 | you can get a cisco to do channelized ds3 to voip for ~ 28K |
| 05:48.15 | brookshire[home] | 7 |
| 05:48.24 | jdv79 | so anything more expensive than that is not worth the trouble |
| 05:48.26 | brookshire[home] | wait |
| 05:48.27 | brookshire[home] | hehe |
| 05:48.41 | FuriousGeorge | so what would that look like? a really fat cable from nid to PCI card? |
| 05:48.42 | X-Rob | jbot, no, ds3 is 23 T1 channels, or 672 individual B channels. |
| 05:48.44 | jbot | okay, X-Rob |
| 05:48.46 | brookshire[home] | damn i'm drunk |
| 05:48.48 | X-Rob | ~ds3 |
| 05:48.49 | jbot | well, ds3 is 23 T1 channels, or 672 individual B channels. |
| 05:48.52 | *** join/#asterisk clive- (n=pirch@ndn-165-137-169.telkomadsl.co.za) |
| 05:48.58 | brookshire[home] | 7 quad cards in a ds3 lol |
| 05:49.00 | ManxPower | FuriousGeorge, The only DS3 I've installed came in as two coax cables |
| 05:49.12 | FuriousGeorge | ManxPower: xrazy |
| 05:49.20 | jdv79 | yeah, i've done the coaxs from verizon |
| 05:49.24 | FuriousGeorge | *crazy |
| 05:49.32 | ManxPower | But that was more than 10 years ago. |
| 05:49.32 | wunderkin | brookshire: usually when you are drunk you see double and not 1/4 |
| 05:49.43 | FuriousGeorge | X-Rob: you got some pull with j-bot, huh |
| 05:49.46 | jdv79 | i was doing it 5 years ago |
| 05:50.06 | jdv79 | huh, this sucks |
| 05:50.09 | jdv79 | i need a PRI now |
| 05:50.25 | X-Rob | jbot, no, a ds3 has 28 T1 channels, or 672 individual B channels. |
| 05:50.27 | ManxPower | jdv79, if you just want testing, sign up with Teliax or someone like that. |
| 05:50.55 | FuriousGeorge | ~ds3 |
| 05:50.56 | jbot | ds3 is, like, 23 T1 channels, or 672 individual B channels. |
| 05:51.05 | FuriousGeorge | ~nj |
| 05:51.07 | jbot | i heard nj is home to the Sopranos |
| 05:51.09 | brookshire[home] | jobt: forget ds3 |
| 05:51.15 | brookshire[home] | jbot: forget ds3 |
| 05:51.17 | jdv79 | how many quad cards can you stuff into a box and what kinda box does that require |
| 05:51.21 | jdv79 | just to service as gws |
| 05:51.28 | brookshire[home] | heh |
| 05:51.35 | jdv79 | dual proc boxes? 3 cards each? |
| 05:51.37 | FuriousGeorge | jbot, no nj is home to the sopranos, not for nuthin' |
| 05:51.38 | jbot | FuriousGeorge: okay |
| 05:51.46 | FuriousGeorge | ~nj |
| 05:51.48 | jbot | it has been said that nj is home to the sopranos, not for nuthin' |
| 05:51.54 | jdv79 | nj is where its at |
| 05:51.54 | FuriousGeorge | wow |
| 05:52.03 | FuriousGeorge | jbot, no nj is home to the sopranos |
| 05:52.06 | jbot | okay, FuriousGeorge |
| 05:52.28 | *** join/#asterisk Igbothom_III (n=HiltonT@static-84.217.240.220.dsl.comindico.com.au) |
| 05:52.29 | brookshire[home] | jdv: personally.. i wouldn't shove more than 2.. but i know of people doing 4 or 5 |
| 05:52.33 | X-Rob | jbot nj is also a smelly place |
| 05:52.34 | jbot | X-Rob: okay |
| 05:52.37 | X-Rob | ~nj |
| 05:52.39 | jbot | i guess nj is home to the sopranos, or a smelly place |
| 05:52.55 | *** join/#asterisk djin_ib (n=djin_ib@gridfox.xs4all.nl) |
| 05:53.01 | wunderkin | heh |
| 05:53.05 | FuriousGeorge | jbot: no, X-Rob is a pedarast |
| 05:53.06 | jbot | FuriousGeorge: okay |
| 05:53.06 | jdv79 | jbot, nj is also where jdv79 hails from |
| 05:53.08 | jbot | jdv79: okay |
| 05:53.29 | FuriousGeorge | ~X-Rob |
| 05:53.31 | jbot | methinks x-rob is a pedarast |
| 05:53.31 | Igbothom_III | g'day FuriousGeorge, how's it going? |
| 05:53.56 | FuriousGeorge | Igbothom_III: g'day, mate |
| 05:54.02 | FuriousGeorge | jbot: i agree |
| 05:54.04 | jbot | You agree? |
| 05:54.05 | Igbothom_III | not had time to play more with my * box, but will get onto it tonite/this weekend |
| 05:54.30 | jdv79 | there's 4 quad cards? |
| 05:54.30 | FuriousGeorge | Igbothom_III: i assume ur that Australian bloke i was speaking to yesterday |
| 05:54.34 | FuriousGeorge | Hilton |
| 05:54.39 | Igbothom_III | yup |
| 05:54.50 | Igbothom_III | bloody Aussies |
| 05:55.12 | X-Rob | wot? |
| 05:55.13 | X-Rob | wotwotwot? |
| 05:55.24 | X-Rob | garn giffucked the lot of yaz. |
| 05:55.38 | Igbothom_III | rofl |
| 05:55.38 | ManxPower | ~manxpower |
| 05:55.40 | jbot | extra, extra, read all about it, manxpower is your God. someone you should send lots of money to because he helps so many people. |
| 05:55.42 | Igbothom_III | maaaaate... |
| 05:55.45 | jdv79 | yeah, you can't beat cisco for legacy to voip gwing with digium |
| 05:56.00 | jdv79 | maybe with the phantom ds3 card |
| 05:56.06 | Igbothom_III | I like the Cisco phones, hell nice |
| 05:56.27 | jdv79 | i was talking about the AS5XXX |
| 05:56.29 | jdv79 | things |
| 05:56.35 | ManxPower | ~manxpower |
| 05:56.37 | jbot | you are probably someone you should hire for a job in BelgiumNetherlands, or someone you should send lots of money to because he helps so many people. |
| 05:56.37 | FuriousGeorge | jbot, no fogedaboudit, nj is home to the sopranos |
| 05:56.38 | jbot | FuriousGeorge: I think you lost me on that one |
| 05:56.47 | SarahEmm | ~sarahemm |
| 05:56.48 | jbot | rumour has it, sarahemm is kitrichernesses. |
| 05:57.01 | FuriousGeorge | jbot, no nj is home to the sopranos. Fogedaboudit! |
| 05:57.02 | jbot | okay, FuriousGeorge |
| 05:57.05 | Igbothom_III | I have a mate in .nl |
| 05:57.09 | jdv79 | anyone ever involved with broadvoice? |
| 05:57.10 | FuriousGeorge | ~nj |
| 05:57.12 | jbot | it has been said that nj is home to the sopranos. Fogedaboudit! |
| 05:57.17 | jdv79 | broadsoft or whatever it is |
| 05:57.28 | Igbothom_III | I have a mate in France on Broadvoice |
| 05:57.54 | jdv79 | not broadvoice - that's my old boss |
| 05:57.56 | FuriousGeorge | jbot: no, FuriousGeorge is a knife fighting monkey known to hangout with The Man With the Metal Bat. |
| 05:57.58 | jbot | okay, FuriousGeorge |
| 05:57.58 | jdv79 | :) oops |
| 05:58.03 | jdv79 | i meant broadsoft |
| 05:58.07 | FuriousGeorge | ~FuriousGeorge |
| 05:58.08 | jbot | somebody said furiousgeorge was a knife fighting monkey known to hangout with The Man With the Metal Bat. |
| 05:58.32 | FuriousGeorge | jbot: no, FuriousGeorge is a knife fighting monkey known to hang out with The Man With the Metal Bat. |
| 05:58.33 | jbot | FuriousGeorge: okay |
| 05:58.42 | FuriousGeorge | fun with jbot |
| 05:58.57 | Igbothom_III | lol |
| 05:59.22 | marc324 | is odbc voicemail storage working with postgresql? |
| 06:05.06 | jdv79 | ManxPower, callprogress? |
| 06:08.56 | jdv79 | uh, doesn't really work:) |
| 06:09.00 | jdv79 | at all |
| 06:13.54 | *** join/#asterisk gres (n=serg@62.152.85.99) |
| 06:18.39 | Pikoro | man this is kicking my arse |
| 06:18.42 | *** join/#asterisk KeX (n=chris@83-65-129-46.paris-lodron.xdsl-line.inode.at) |
| 06:19.38 | pauldy | hrm anyone have any good tutorial urls for doing an asterisk to asterisk connection using IAX2 |
| 06:19.44 | *** join/#asterisk JerJer[mobile] (n=jj@68.123.154.34) |
| 06:20.42 | marc324 | can someone help with odbcstorage and postgres? |
| 06:21.14 | JerJer[mobile] | how about asking a specific question about it ? |
| 06:21.26 | wunderkin | brookshire: i was blind sorry, kpfleming point out to me that you can hide status closed in the filter |
| 06:22.09 | brookshire[home] | wunder: but! |
| 06:22.11 | marc324 | is there a problem with odbcstorage and postgresql? the query from asterisk uses double quotes, instead of single quote |
| 06:22.16 | brookshire[home] | you clued me into another demand |
| 06:22.29 | wunderkin | did i? |
| 06:22.34 | brookshire[home] | yeah... hehe |
| 06:22.45 | marc324 | SELECT COUNT(*) FROM voicemessages WHERE dir like "%default/aaa/aaa" |
| 06:22.47 | wunderkin | i just keep making your life more miserable |
| 06:22.53 | brookshire[home] | maybe like.. and easier way to search for patches if you are completely new to bugs |
| 06:22.56 | brookshire[home] | and mantis |
| 06:23.21 | wunderkin | yeah, theres a preset filter for head only open bugs but not open bugs period :P |
| 06:23.27 | marc324 | this gives sql error... needs single quote |
| 06:24.12 | marc324 | jerjer-- do you know whats wrong? |
| 06:24.24 | brookshire[home] | problem exists with postgres :) |
| 06:24.27 | brookshire[home] | MYSQL :) |
| 06:24.51 | brookshire[home] | sorry.. i'm just bitter about postgres |
| 06:24.56 | brookshire[home] | it did me wrong once, lol |
| 06:25.14 | marc324 | mysql uses double quotes, while postgres uses single.... asterisk is send the sql query with double quotes |
| 06:25.53 | wunderkin | brookshire: hows that? |
| 06:26.07 | brookshire[home] | core dumped and deleted my data |
| 06:26.13 | brookshire[home] | well.. corrupted it |
| 06:26.28 | brookshire[home] | this was waaaaaay back in the day though, lol |
| 06:26.32 | wunderkin | ah |
| 06:26.33 | brookshire[home] | it's probably better |
| 06:26.34 | brookshire[home] | now |
| 06:26.44 | wunderkin | hopefully, i use it :P |
| 06:26.56 | brookshire[home] | i learned sql on postgres... hehe |
| 06:27.00 | brookshire[home] | but.. oh well |
| 06:27.20 | wunderkin | yeah me too, i guess thats a good point to still do daily backup even if theres raid |
| 06:27.45 | brookshire[home] | yeah... i do daily dumps on all of our databases, just in case |
| 06:28.09 | marc324 | it cost bw |
| 06:28.14 | brookshire[home] | mysql has never really done me wrong though.. |
| 06:28.19 | brookshire[home] | but 4.1 is crap |
| 06:28.21 | brookshire[home] | :P |
| 06:29.08 | brookshire[home] | too unstable |
| 06:29.11 | brookshire[home] | anyways... ni ni |
| 06:29.18 | wunderkin | cya |
| 06:31.05 | *** join/#asterisk Inv_arp (i=junya@adsl-144-134-75.mia.bellsouth.net) |
| 06:31.23 | jdv79 | if i specify Channel: SIP/1@X.X.X.X where X is another box where should hte call go? |
| 06:31.55 | jdv79 | i'm just confused now |
| 06:33.42 | kaldemar | jdv79: to extension 1 on the box you specify with that ip. |
| 06:34.10 | jdv79 | somehow the sip phone on the originating box is getting rung |
| 06:34.17 | jdv79 | but the call is routing through the other box |
| 06:34.37 | jdv79 | but the other box is just answering and echoing |
| 06:34.43 | jdv79 | that's the entire context on the far end |
| 06:34.59 | *** join/#asterisk Poincare (n=jefffnod@dD577A88C.access.telenet.be) |
| 06:35.33 | jdv79 | my stupidity - sorry |
| 06:37.10 | FuriousGeorge | can someone comment on the relative difficulty of mysql to extensions.conf, if thats possible |
| 06:44.30 | FuriousGeorge | guess not :D |
| 06:51.34 | *** join/#asterisk af_ (n=af@ip-142-250.sn1.eutelia.it) |
| 06:53.37 | *** join/#asterisk jeffik (n=Administ@CPE0020ed8494b8-CM0012c999ca4e.cpe.net.cable.rogers.com) |
| 06:54.53 | jdv79 | Oct 14 02:54:40 WARNING[10694]: channel.c:766 channel_find_locked: Avoided initial deadlock for '0x96f9818', 10 retries! |
| 06:54.55 | jdv79 | hmm? |
| 06:56.52 | *** join/#asterisk mmmToop (n=chatzill@196.31.11.194) |
| 07:04.14 | Rowter | channel.c: Avoided initial deadlock for '0x81f5f30', 10 retries! |
| 07:04.23 | Rowter | whats that? |
| 07:06.18 | tzafrir_laptop | Usually that "deadlock" warning can be ignored, I believe. search for it in google |
| 07:08.02 | Rowter | tzafrir_laptop, looking |
| 07:18.49 | *** join/#asterisk \PsyKo\ (n=xxxxxxxx@golia.caltanet.it) |
| 07:21.08 | *** join/#asterisk syle2 (n=blah@unaffiliated/syle) |
| 07:24.54 | *** join/#asterisk vlrk (n=vlrk@59.93.79.86) |
| 07:25.08 | *** join/#asterisk JunK-Y (n=junky@Toronto-HSE-ppp3781375.sympatico.ca) |
| 07:25.08 | vlrk | does polycom support the MessageWaitingIndicator |
| 07:28.02 | Juggie | yes |
| 07:28.21 | *** join/#asterisk bweschke (n=bweschke@m810f36d0.tmodns.net) |
| 07:33.19 | *** join/#asterisk tessier (n=treed@wsip-68-15-4-13.sd.sd.cox.net) |
| 07:33.51 | *** join/#asterisk orzel (n=orzel@freehackers.org) |
| 07:33.59 | limbique | morning! |
| 07:34.20 | limbique | can anyone help me unlocking cisco 7960? |
| 07:34.28 | limbique | or reset to factory defaults? |
| 07:36.59 | *** join/#asterisk tobiasWolf (n=konversa@195.162.255.10) |
| 07:38.17 | vlrk | Juggie can we configure that option through webpage |
| 07:41.17 | *** join/#asterisk st3v (n=st3v@netblock-66-245-213-120.dslextreme.com) |
| 07:41.34 | st3v | do the Digium cards come with a T1 cable? I ordered a digium 2 port card and a rhino channel bank. I am not sure if they come with a cable to connect the two |
| 07:41.48 | h3x0r | maybe in the combo |
| 07:41.52 | h3x0r | but i doubt it |
| 07:42.27 | st3v | does it just use a crossover cat5e cable? it's rj45 isn't it |
| 07:43.26 | X-Rob | no |
| 07:43.36 | X-Rob | PRI uses pairs 1 and 2 |
| 07:43.40 | X-Rob | uh |
| 07:43.45 | X-Rob | pins 1,2,3,4 |
| 07:43.59 | X-Rob | where pair 1 and 2 on an ethernet crossover is 1,2,3,6 |
| 07:44.47 | st3v | can I can crimp my own cable with that pinout, or do I have to get a special one |
| 07:44.54 | X-Rob | you sure can |
| 07:44.59 | X-Rob | 12/34 crossover |
| 07:45.06 | h3x0r | what the hell |
| 07:45.07 | X-Rob | 1->3 2->4 |
| 07:45.19 | h3x0r | you dumb asses |
| 07:45.22 | h3x0r | its 1,2 4,5 |
| 07:45.30 | h3x0r | left two and middle two |
| 07:45.36 | jdv79 | is there a way to get the active channel count without the whole listing? |
| 07:46.25 | X-Rob | fucker. so it is |
| 07:46.28 | X-Rob | thanks h3x0r |
| 07:46.31 | h3x0r | heh |
| 07:46.34 | h3x0r | brain farts |
| 07:46.36 | h3x0r | everywhere |
| 07:46.46 | h3x0r | fuck T1s |
| 07:46.47 | h3x0r | use DS3s |
| 07:46.50 | h3x0r | two coax cables |
| 07:46.53 | h3x0r | you can't fuck that up |
| 07:46.55 | snitt | 1,2 => 3,6 |
| 07:46.57 | snitt | not 3,4 |
| 07:46.59 | h3x0r | at least i hope not |
| 07:47.17 | X-Rob | snitt - no. That's ethernet |
| 07:47.25 | snitt | i'm talking about ethernet :) |
| 07:47.41 | X-Rob | ok |
| 07:48.16 | h3x0r | actually |
| 07:48.25 | X-Rob | ble D-4: Standard Crossover PRI Patch Cable |
| 07:48.25 | X-Rob | Standard Interface Crossover Interface |
| 07:48.25 | X-Rob | Pin 1---Receive (+) Ring 2 |
| 07:48.25 | X-Rob | |
| 07:48.25 | X-Rob | Pin 4---Transmit (-) Tip1 |
| 07:48.25 | X-Rob | Pin 2---Receive (+) Tip2 |
| 07:48.26 | h3x0r | its 1-8 => 1-8 coz most ethernet switches have auto crossover :PPPPPPPPPP |
| 07:48.27 | X-Rob | |
| 07:48.29 | X-Rob | Pin 1---Receive (+) Ring 2 |
| 07:48.31 | X-Rob | Pin 4---Transmit (-) Tip1 |
| 07:48.33 | X-Rob | |
| 07:48.35 | X-Rob | Pin 5---Transmit (-) Ring1 |
| 07:48.39 | X-Rob | |
| 07:48.41 | X-Rob | Pin 2---Receive (+) Tip2 |
| 07:48.43 | X-Rob | that's what cisco thinks a PRI crossover cable is |
| 07:48.52 | *** join/#asterisk stdio (n=stdio@pcp01473275pcs.lncstr01.pa.comcast.net) |
| 07:49.05 | snitt | :D |
| 07:49.08 | h3x0r | is that a db-15 t1? |
| 07:49.14 | *** join/#asterisk srsergio (n=asterisk@194.red-217-216-155.user.auna.net) |
| 07:49.25 | X-Rob | http://www.cisco.com/en/US/products/hw/video/ps1870/products_user_guide_chapter09186a008007edf9.html |
| 07:49.57 | h3x0r | The IP/VC 3525 Gateway uses a proprietary PRI interface. An adapter cable that converts the proprietary PRI configuration to the standard PRI configuration is shipped with the unit |
| 07:50.00 | h3x0r | proprietary |
| 07:50.04 | h3x0r | proprietaryt |
| 07:50.05 | h3x0r | heh |
| 07:50.05 | X-Rob | scroll down |
| 07:50.20 | X-Rob | Table D-4 |
| 07:50.22 | h3x0r | ok |
| 07:50.40 | h3x0r | yeah that one makes sense |
| 07:50.43 | h3x0r | but not when you paste it |
| 07:50.43 | h3x0r | heh |
| 07:51.19 | h3x0r | this sucks, ive been pulling electrical wiring all night in my data center |
| 07:51.38 | X-Rob | sucks to be you |
| 07:51.45 | X-Rob | I pay people to do that |
| 07:51.46 | X-Rob | 8) |
| 07:51.50 | h3x0r | i decided after working in the hot attic to pull off the cap on the a/c stub up there, close the cieling tile and turn it on full blast to cool that shit down |
| 07:51.50 | h3x0r | heh |
| 07:51.56 | X-Rob | fuck crawling around dragging cables under floors. |
| 07:52.14 | h3x0r | fuck raised floors |
| 07:52.17 | X-Rob | even worse, in celings |
| 07:52.20 | h3x0r | that is so 80's |
| 07:52.29 | h3x0r | we don't have forced air cooling mainframes anymore |
| 07:52.39 | h3x0r | why do people still buy those damn raised floors |
| 07:53.05 | X-Rob | because they're a smegload easier to cable than anything else |
| 07:53.17 | h3x0r | cable trays are easy |
| 07:53.35 | *** join/#asterisk alphaque (n=alphaque@218.111.204.40) |
| 07:54.13 | h3x0r | it takes longer to pull up tiles and run shit than it does to use a step ladder and throw it up on a cable tray |
| 07:54.30 | h3x0r | when you consider all the maintenance you have to do to a raised floor |
| 07:54.50 | *** join/#asterisk fenlander (n=neils@82.152.81.57) |
| 07:54.52 | h3x0r | you usually can't pull up the tiles near the cabinets because the cabinets are sitting on some of them |
| 07:55.06 | h3x0r | you have to get 9' racks and screw them into the subfloor which is a lot of extra work |
| 07:55.26 | h3x0r | and then cut holes in the raised floor to fit it all. its just a pain in the ass |
| 07:55.39 | h3x0r | there is a neat product i found called netfloor though, its more like lego bricks |
| 07:55.42 | h3x0r | and its only 6" thick |
| 07:55.55 | h3x0r | you just clip them all together and its free standing |
| 07:55.56 | st3v | so a PRI pinout is the same as a T1 pinout? |
| 07:56.05 | wunderkin | yes st3v |
| 07:56.08 | h3x0r | PRI is a protocol that runs over T1 |
| 07:56.20 | h3x0r | it just got lost in marketing jumbo |
| 07:56.43 | X-Rob | h3x0r - wrong way around. |
| 07:56.51 | X-Rob | A PRI == Primary Rate Interface |
| 07:57.03 | X-Rob | T1 or E1 is the protocol used on the PRI wire |
| 07:57.11 | st3v | I see |
| 08:00.35 | tzanger | T1 defined a logical and electrical organization |
| 08:00.48 | tzanger | PRI defines a layer 'higher' than that |
| 08:00.55 | tzanger | think of the OSI network model |
| 08:01.15 | st3v | regarding that Cisco table D-4 on PRI Crossover Interface, how do I know what colors go on each end? I can't find anything on that |
| 08:01.16 | limbique | can anyone help me unlocking cisco 7960? |
| 08:01.18 | limbique | or reset to factory defaults? |
| 08:01.28 | tzanger | T1 is the first two layers (physical and data link |
| 08:01.30 | tzanger | PRI is a protocol |
| 08:01.47 | X-Rob | well that's two wrongs. I'll STFU I think. |
| 08:02.00 | tzanger | it's actually pretty rare to find a "real" T1 anymore |
| 08:02.07 | tzanger | they're all DS1 over HDSL or more commonly now HDSL2 |
| 08:02.26 | tzanger | DSX-1 is common |
| 08:03.02 | tzanger | st3v: with the RJ45 connector pointing "up" and iwth the contacts facing you |
| 08:03.20 | tzanger | pin 1 -> 4 |
| 08:03.23 | tzanger | pin 2 -> 5 |
| 08:03.41 | tzanger | that's it |
| 08:03.56 | tzanger | 12x45xxx -> 45x12xxx |
| 08:04.27 | tzanger | I always forget the wire colour codes (the resistor colour code I've got burned into my head from years and years of working in it) |
| 08:04.48 | X-Rob | ORANGE IS 5!!!! |
| 08:04.48 | tzanger | IIRC pair 1 is blue, pair 2 is orange, pair 3 is green, pair 4 is violet |
| 08:04.50 | tzanger | I *think* |
| 08:05.19 | X-Rob | Blogb is the way I remember it, which I believe is wrong, but as long as you're consistant it doesn't really matter |
| 08:05.26 | X-Rob | BLue Orange Green BRown |
| 08:05.27 | tzanger | very true |
| 08:05.35 | tzanger | brown that is what it was |
| 08:05.39 | tzanger | brown and then pair 5 is violet |
| 08:05.49 | tzanger | and then you start getting into stripe groups |
| 08:06.05 | tzanger | white, yellow red black slate (I think) |
| 08:06.54 | tzanger | http://www.shout.net/~wildixon/telecom/cable/color.html |
| 08:06.56 | tzanger | that is it |
| 08:07.12 | tzanger | blue/orange/green/brown |
| 08:07.25 | tzanger | er blue/orange/green/brown/slate |
| 08:08.09 | tzanger | then binders are white/red/black/yellow/violet |
| 08:08.15 | tzanger | there's a 25-pair cable |
| 08:08.46 | tzanger | then you start with binders which group up 25 pairs |
| 08:09.30 | tzanger | I always forget what tip is... it's red in 4-pair cable, heh |
| 08:09.42 | tzanger | man this brings back memories of pissing around in telecom pedestals |
| 08:10.02 | tzanger | identifying pairs by the binder/wire colour/stripe |
| 08:11.05 | jdv79 | how do i know what codec a call is using? |
| 08:11.11 | jdv79 | do i have to run debug? |
| 08:11.17 | tzanger | jdv79: show channel [callid] shows it |
| 08:11.57 | limbique | can anyone help me unlocking cisco 7960? |
| 08:11.58 | limbique | or reset to factory defaults? |
| 08:12.21 | jdv79 | thanks |
| 08:16.10 | *** join/#asterisk lilo (i=levin@freenode/staff/pdpc.levin) |
| 08:19.55 | jdv79 | oh wow |
| 08:20.04 | limbique | can anyone help me unlocking cisco 7960? |
| 08:20.05 | limbique | or reset to factory defaults? |
| 08:20.18 | jdv79 | i was wondering why there was smoke coming out of one of my boxes...transcoding - duh... |
| 08:21.23 | *** join/#asterisk lilo (i=levin@freenode/staff/pdpc.levin) |
| 08:26.07 | *** join/#asterisk nextime (n=nextime@213-140-6-96.ip.fastwebnet.it) |
| 08:27.17 | *** join/#asterisk Champi (i=Champi@damn.e-leet.be) |
| 08:27.44 | jdv79 | does asterisk fork itself at all? |
| 08:28.15 | jdv79 | as in would a quad box even be useful? |
| 08:30.18 | Igbothom_III | that's more related to multithreading than just forking |
| 08:30.21 | Igbothom_III | but, still... |
| 08:30.22 | Igbothom_III | :) |
| 08:30.45 | jdv79 | threading, forking, whichever asterisk does it |
| 08:30.49 | Igbothom_III | :) |
| 08:31.18 | Igbothom_III | hhmmm, about to start looking at getting my Wildcard 100 clone card working in Xorcom Rapid |
| 08:31.26 | Igbothom_III | so I can accept and place calls on my landline |
| 08:31.36 | Igbothom_III | then to play SIP to an external SIP personage |
| 08:33.57 | tzafrir_laptop | jdv79, every channel is handled by a separate thread |
| 08:34.18 | Igbothom_III | tzafrir; cool, that's nice to know |
| 08:34.26 | jdv79 | thanks |
| 08:34.35 | Igbothom_III | now, if only there were some halfway decent docs for Xorcom Rapid |
| 08:34.52 | jdv79 | i can only get 100 concurrent calls on my celeron 2.4 box for some reason |
| 08:34.58 | jdv79 | i know i've done more:) |
| 08:35.02 | Igbothom_III | (or Asterisk on configuring zapata.conf...) |
| 08:35.17 | tzafrir_laptop | why do you need docs if it's working? or to rephrase it: what's not working/clear? |
| 08:35.39 | Igbothom_III | how do I configure the POTS card (Wildcard 100 clone) |
| 08:35.47 | Igbothom_III | it isn't working, afaict |
| 08:35.58 | tzafrir_laptop | was it on the system when you installed it? |
| 08:36.11 | Igbothom_III | and then I need to sort a SIP provider out (prolly my SIPPhone account) to place external SIP calls, then it will ne sweet |
| 08:36.14 | Igbothom_III | yup |
| 08:36.56 | tzafrir_laptop | so it should have been identified... In the asterisk => asterisk info menu, zaptel channel, is there channel "1"? |
| 08:37.50 | tzafrir_laptop | As for the sip provider: one starting point is: 'man ast-cmd' . Note the section about add-trunk |
| 08:38.07 | Igbothom_III | cool, will read |
| 08:38.12 | Igbothom_III | and only "pseudo" |
| 08:38.47 | tzafrir_laptop | so the card has not been identified. Do you see a tigetjet-something card in the output of lspci? |
| 08:39.04 | Igbothom_III | I do like the sip-phones.d/ way of having conf files WAY better than all in a single file |
| 08:39.41 | Igbothom_III | <PROTECTED> |
| 08:39.57 | tzafrir_laptop | next: lspci | grep zaptel |
| 08:40.04 | tzafrir_laptop | does it report anything? |
| 08:40.10 | Igbothom_III | nothing |
| 08:40.26 | Igbothom_III | it detects it during boot, so sayeth dmesg |
| 08:40.41 | morale | is your pci.ids up to date? |
| 08:40.41 | Igbothom_III | it is a WC 100 clone card |
| 08:40.50 | Igbothom_III | (to play with until I buy a real Digium card) |
| 08:40.56 | Igbothom_III | wassat? |
| 08:41.17 | tzafrir_laptop | what does it report? |
| 08:41.27 | Igbothom_III | fresh Rapid 1.1 install as of yesterday |
| 08:41.29 | tzafrir_laptop | try: /etc/init.d/asterisk stop |
| 08:41.42 | tzafrir_laptop | (better be safe when rmmod-ing zaptel modules) |
| 08:41.54 | tzafrir_laptop | then: rmmod wcfxo |
| 08:42.02 | tzafrir_laptop | insmod wcfxo |
| 08:42.33 | limbique | can anyone help me unlocking cisco 7960? |
| 08:42.35 | limbique | or reset to factory defaults? |
| 08:42.40 | tzafrir_laptop | what does it report in dmesg? (dmesg| tail, or tail /var/log/kern.log ) |
| 08:42.45 | Igbothom_III | can't read wcfxo: no such file/dir |
| 08:43.02 | tzafrir_laptop | limbique, the default password is cisco, fwiw |
| 08:43.31 | tzafrir_laptop | modprobe wcfxo |
| 08:43.32 | tzafrir_laptop | ? |
| 08:43.40 | *** join/#asterisk clive- (n=pirch@ndn-165-137-169.telkomadsl.co.za) |
| 08:43.41 | limbique | tzafrir !! YOU ROCK!!!! |
| 08:43.45 | limbique | tnx!!! works! |
| 08:43.54 | Igbothom_III | modprobe was fine |
| 08:44.07 | Igbothom_III | well, no error |
| 08:44.18 | tzafrir_laptop | anything in /proc/zaptel ? |
| 08:44.36 | Igbothom_III | "1" |
| 08:44.48 | tzafrir_laptop | is it something for 'ztdummy'? |
| 08:44.55 | tzafrir_laptop | cat /proc/zaptel/1 |
| 08:45.16 | tzafrir_laptop | hopefully it should be for the FXO card |
| 08:45.20 | Igbothom_III | Span 1: WCFXO/0 "Generic Clone Board 1" RED |
| 08:45.20 | Igbothom_III | <PROTECTED> |
| 08:45.45 | tzafrir_laptop | good. So why was the module not loaded at boot time/install time? |
| 08:45.50 | Igbothom_III | nfi |
| 08:46.28 | Igbothom_III | want me to reboot to check it didn't selfdestruct sometime today? |
| 08:46.42 | tzafrir_laptop | generally running 'genzaptelconf -s' at this point should 'fix' your configuration and restart asterisk. HOwever, I wonder what happens on next boot |
| 08:47.13 | Igbothom_III | about to find out :) |
| 08:48.22 | *** join/#asterisk Gunnar (n=gunnar@34.80-203-209.nextgentel.com) |
| 08:49.50 | Igbothom_III | it appears to be up, rebooted, with "1" there now |
| 08:49.58 | *** join/#asterisk kippi (n=chrisfro@untrust-gct.equinoxit.net) |
| 08:50.00 | *** join/#asterisk Speeder (n=psilva@80.172.156.1) |
| 08:50.10 | kippi | hey |
| 08:50.22 | Igbothom_III | damn, phone cord not long enough now I moved it :( |
| 08:50.26 | Speeder | hey |
| 08:50.29 | Igbothom_III | will have to relocate the box, back in a few |
| 08:52.00 | kippi | Is there away to have a message where you can type in what you want it to say, so you don't have to record your message |
| 08:52.02 | snitt | god bless the patchpanels |
| 08:52.12 | Speeder | i'm having a problem with asterisk & zaphfc, i can't create zap channels... |
| 08:52.31 | Speeder | could u help me? |
| 08:53.30 | tzafrir_laptop | http://tzafrir.org.il/genzaptelconf may be able to generate config files... |
| 08:53.34 | snitt | what is hfc? |
| 08:54.03 | tzafrir_laptop | do you see the channels in /proc/zaptel/* ? |
| 08:54.44 | tzafrir_laptop | snitt, it uses the zaptel interface to support some simple ISDN cards (with the HFC chipset?) |
| 08:55.54 | Speeder | ls /proc/zaptel/ says 1 |
| 08:56.14 | tzafrir_laptop | Speeder, cat that file |
| 08:56.37 | Speeder | Span 1: ZTHFC1 "HFC-S PCI A ISDN card 0 [TE] layer 1 DEACTIVATED (F4)" AMI/CCS |
| 08:56.37 | Speeder | <PROTECTED> |
| 08:56.38 | Speeder | <PROTECTED> |
| 08:56.38 | Speeder | <PROTECTED> |
| 08:56.45 | Speeder | 3 channels |
| 08:57.33 | tzafrir_laptop | module loaded, card identified. Now you need to generate config files. The bristuff distro has sample zaptel and zapata confs in the zaphfc directory |
| 08:58.44 | Speeder | my zaptel.conf is simple it only defines the 2 voice channels and one data channel |
| 09:00.29 | Igbothom_III | ok, tzafrir, box is back on, wcfxo seems to be found... now I need to know what to do in Rapid to use it :) |
| 09:02.37 | jdv79 | looks like its just totally CPU starved:( |
| 09:02.41 | Igbothom_III | I call that line, Rapid answers and gives me deathly silence |
| 09:02.52 | Igbothom_III | jdv79; damn |
| 09:03.12 | jdv79 | SIP using ulaw playing recordings in ulaw should be decently light on the load, right? |
| 09:03.36 | Igbothom_III | well, it still has to transcode |
| 09:03.46 | jdv79 | wy's that? |
| 09:03.55 | Igbothom_III | mu-law to a-law |
| 09:04.22 | jdv79 | huh? the file i'm playing is ulaw and the connection is ulaw |
| 09:04.38 | Igbothom_III | sorry, misread :) |
| 09:04.52 | Igbothom_III | in that case, yes, it should be light on the CPU |
| 09:05.03 | Igbothom_III | oh, asterisk reported "Oct 14 19:01:59 NOTICE[2085]: app_dial.c:777 dial_exec: Unable to create channel of type 'Zap'" and that everyone was busy |
| 09:05.11 | Igbothom_III | when the POTS was answered |
| 09:05.48 | jdv79 | is a 2.2 celeron slow these days? |
| 09:06.10 | Igbothom_III | only just |
| 09:06.22 | jdv79 | i think my almost new laptop is only 1.6 |
| 09:06.23 | Igbothom_III | but remember, a Celeron is slower than a P4 or Athlon64 |
| 09:06.29 | jdv79 | 1.6 M or whatever is it |
| 09:06.31 | jdv79 | it is |
| 09:06.35 | Igbothom_III | that's a Celeron M - a TOTALLY different beast |
| 09:06.56 | alphaque | Igbothom_III: how different ? |
| 09:07.17 | jdv79 | it doesn't say its a celeron |
| 09:07.32 | jdv79 | it says "Pentium M" |
| 09:07.39 | Igbothom_III | with the "M" series, Intel went back to the Pentium-III core and rebuilt with what they have learned from the P4, and also with power friendliness in mind |
| 09:08.11 | jdv79 | Xeon is fastest tey have these days, right? |
| 09:08.14 | Igbothom_III | Celeron M or Pentium M will run at a much slower clockspeed than an equivalent Celeron or Pentium 4 |
| 09:08.22 | Igbothom_III | actually, Opteron is |
| 09:08.25 | h3x0r | P4 Xeon |
| 09:08.36 | Igbothom_III | but yeah, in the Intel world, Xeons are due to their larger cache |
| 09:08.37 | jdv79 | eh, we've got dell servers already at the office |
| 09:08.52 | jdv79 | dual an quad xeons |
| 09:08.55 | h3x0r | the lga775 (socket T) P4 3.0+ are mostly 2MB cache |
| 09:08.57 | h3x0r | fuck xeon |
| 09:09.14 | Speeder | [channels] |
| 09:09.18 | Speeder | language = pt |
| 09:09.19 | jdv79 | ok |
| 09:09.22 | Speeder | switchtype = euroisdn |
| 09:09.26 | Speeder | signalling = bri_cpe_ptmp |
| 09:09.30 | Speeder | pridialplan = local |
| 09:09.34 | Speeder | ;priindication=outofband |
| 09:09.38 | Speeder | pritrustusercid = yes |
| 09:09.39 | jdv79 | i wasn't involved in the purchasing at the time |
| 09:09.40 | Igbothom_III | tzafrir; are you still here? |
| 09:09.42 | Speeder | echocancel = yes |
| 09:09.46 | Speeder | overlapdial = yes |
| 09:09.49 | jdv79 | had i been i would have tried for AMD |
| 09:09.50 | Speeder | echocancelwhenbridged=no |
| 09:09.55 | Speeder | echotraining=no |
| 09:09.57 | Igbothom_III | either is quite decent :) |
| 09:09.58 | Speeder | immediate = no |
| 09:10.02 | Speeder | usecallerid = yes |
| 09:10.06 | Speeder | group = 1 |
| 09:10.10 | Speeder | context = default |
| 09:10.14 | Speeder | channel => 1-2 |
| 09:10.18 | Speeder | usecallingpres=yes |
| 09:10.19 | Speeder | nationalprefix = 0 |
| 09:10.19 | Speeder | internationalprefix = 00 |
| 09:10.20 | Speeder | my zapata.conf |
| 09:10.20 | Speeder | it should work... |
| 09:10.30 | jdv79 | who makes production AMD boxes? |
| 09:10.36 | Igbothom_III | HP |
| 09:10.46 | Igbothom_III | Dell, probably |
| 09:11.05 | Igbothom_III | IBM may, but I don't "do" IBM nor Dell |
| 09:11.17 | jdv79 | dell dropped the line i think |
| 09:11.21 | jdv79 | after trialing it last year |
| 09:12.01 | tzafrir_laptop | Igbothom_III, need to go for a while... |
| 09:12.12 | Igbothom_III | ok, roughly how long? |
| 09:12.16 | Igbothom_III | (I can get dinner) |
| 09:12.16 | tzafrir_laptop | Speeder: |
| 09:12.18 | tzafrir_laptop | ~pb |
| 09:12.20 | jbot | hmm... pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca/ |
| 09:12.42 | Speeder | ? |
| 09:12.53 | Igbothom_III | saves flooding the channel |
| 09:13.10 | tzafrir_laptop | Speeder, does asterisk successfully start? it is known to fail if there are incorrectly-configured channels in zapata.conf |
| 09:13.17 | Speeder | yes |
| 09:14.09 | jdv79 | this sucks, i have no fast boxes here |
| 09:14.18 | jdv79 | oh well, time to sleep i guess |
| 09:14.21 | Igbothom_III | I'm using a Dual Cel-525 here :) |
| 09:14.26 | Igbothom_III | with 512 MB |
| 09:14.53 | jdv79 | all my boxes have 512 or 1G RAM but it seems that i've skimped on the CPU:) |
| 09:15.09 | Igbothom_III | and I mean Celeron-III :) |
| 09:15.15 | *** join/#asterisk ful|work (n=fulgas@213.58.130.46) |
| 09:15.52 | Igbothom_III | but this is only for me to learn * (and Xorcom Rapid) on and for use here at the office if it can handle it |
| 09:16.12 | Speeder | when I try make a call I receive this mesage zaphfc: empty HDLC frame or bad CRC received on dmesg |
| 09:17.00 | h3x0r | you skimped on the ram too |
| 09:17.04 | h3x0r | my boxes have 2G ram |
| 09:17.37 | vp7 | Can anyone help me with asterisk (cvs) + asterisk-oh323-0.7.3 (+cvs pwlib&openh323)? Without chan_oh323.so asterisk works fine, but after adding it - it says "Segmentation fault" and didn't start at all :( |
| 09:17.50 | jdv79 | dual core, 64bit - what is al this about? |
| 09:18.06 | Igbothom_III | speed, jdv79, speed! |
| 09:18.16 | jdv79 | what should i shoot for? |
| 09:18.19 | jdv79 | both? |
| 09:18.24 | Igbothom_III | fast, jdv79, fast! |
| 09:18.31 | jdv79 | yeah... |
| 09:18.34 | Igbothom_III | all Dual Core CPUs are 64-bit |
| 09:18.47 | jdv79 | is it just 2 32 bit cores welded together? |
| 09:18.48 | h3x0r | i wouldnt run asterisk on a x64 os |
| 09:18.57 | h3x0r | no |
| 09:19.01 | Igbothom_III | dual core is *basically* 2 CPUs on the one package, so about 75 - 80% faster than a single CPU |
| 09:19.02 | h3x0r | this isnt Neo-Geo |
| 09:19.28 | jdv79 | you can run in 32bit mode though, right? |
| 09:19.30 | Igbothom_III | and as for 32/64, it is a LOT more than 2 ALUs banged together :) |
| 09:19.36 | Igbothom_III | yeah, easily |
| 09:19.56 | h3x0r | man i tried a bunch of different x64 os's and they all ran so damn slow |
| 09:19.56 | Igbothom_III | I'd like to see * running on a Dual Core x64 CPU in 64-bit - it would hammer |
| 09:20.03 | h3x0r | no it woudlnt |
| 09:20.06 | Igbothom_III | yeah, right now they do |
| 09:20.08 | h3x0r | asterisk has no use for 64 bit registers |
| 09:20.11 | Igbothom_III | as does Win XP x64 |
| 09:20.21 | h3x0r | you believe in video game marketing style |
| 09:20.31 | h3x0r | 64 bit does nothing at all for most applications |
| 09:20.32 | Igbothom_III | I don't play video games |
| 09:20.43 | Igbothom_III | right now it doesn't, agreed |
| 09:20.53 | h3x0r | its like a human being able to count to 10 trillian instead of 10 billion |
| 09:20.59 | h3x0r | Trillion i mean |
| 09:21.02 | h3x0r | fucking trilian |
| 09:21.18 | h3x0r | and it never will |
| 09:21.23 | Igbothom_III | running Windows Server 2003 on 64-bit will allow about 50% more Terminal Server performance |
| 09:21.30 | h3x0r | the only thing that might be remotely useful is re-writing codecs for 64 bit mode |
| 09:21.33 | Igbothom_III | (if the damned apps run in 64 bit Windows, that is :) |
| 09:21.40 | h3x0r | That is some bullshit. |
| 09:21.46 | Igbothom_III | nope, observed |
| 09:21.51 | Igbothom_III | we trialled it here |
| 09:21.59 | Igbothom_III | before we deployed it in 32-bit mode |
| 09:22.01 | Igbothom_III | due to some apps |
| 09:22.16 | h3x0r | the real reason is most likely because they deleted compatibility code |
| 09:22.20 | h3x0r | with older OS revisions |
| 09:22.25 | Igbothom_III | the thin clients behaved a LOT snappier with x64 Win Server |
| 09:22.30 | h3x0r | and turned on some compiler optimizations for once |
| 09:22.35 | Igbothom_III | yeah, no probs with that |
| 09:22.47 | Igbothom_III | 16-bit should have died when XP came on the scene |
| 09:22.47 | h3x0r | because anything you compile 32 bit you need to use a 486 instruciton set so it will run on everything |
| 09:22.59 | *** join/#asterisk folsson (n=filip@h147n1fls32o985.telia.com) |
| 09:23.02 | h3x0r | it dosent have anything to do with 16 bit api's |
| 09:23.11 | h3x0r | win32 |
| 09:23.22 | Igbothom_III | Windows XP/2003 won't run on a 486, so they could EASILY optim for a P5 or above |
| 09:23.24 | h3x0r | anyway |
| 09:23.29 | h3x0r | yeah well |
| 09:23.38 | h3x0r | gcc optimizes for p4, p4m, prescott, nocona, etc. |
| 09:23.40 | h3x0r | if you tell it to |
| 09:23.46 | Igbothom_III | uyup, I know :) |
| 09:23.53 | Igbothom_III | played with Gentoo for some time |
| 09:24.07 | h3x0r | if you compile something x64 theres only two types of CPUs it will run on |
| 09:24.21 | h3x0r | anyway, 64 bit if anything slows shit down |
| 09:24.25 | Igbothom_III | P4 and Ath64 and Xeon and Opteron - I can count 4 there |
| 09:24.27 | h3x0r | and it makes everything take twice as much ram |
| 09:24.29 | h3x0r | even if it dosent need it |
| 09:24.29 | jdv79 | you think upping the -O on a asterisk build would be noticable? |
| 09:24.48 | h3x0r | it just means less operations if you need to calculate something that won't fit in a 32 bit register |
| 09:25.02 | Igbothom_III | I'm getting food, back and chatting to tzafrir_laptop later to get my Xorcom happening with FXO card |
| 09:25.11 | h3x0r | asterisk already ups the -O for certain pieces of it |
| 09:25.44 | h3x0r | i tried both linux and freebsd on amd64 and intel x64 |
| 09:25.50 | h3x0r | and it ran like a 486 |
| 09:25.59 | jdv79 | alright |
| 09:26.00 | h3x0r | it was terrible |
| 09:26.08 | jdv79 | too bad nobody sells 32 bit anymore |
| 09:26.25 | h3x0r | yeah well too bad everybody uses a 32 bit os on a 64 bit cpu |
| 09:26.46 | jdv79 | not everything with the world is in sync dude |
| 09:27.16 | jdv79 | there is always friction on timelines and huge movements and stuff |
| 09:27.31 | h3x0r | but its something the average consumer dosent even need |
| 09:27.45 | jdv79 | you can't well have the OS if you don't have the CPU for it |
| 09:27.47 | h3x0r | why dosent microsoft do something useful like remove the 65536 row limitation on excel spreadsheets |
| 09:27.51 | jdv79 | one step at a time |
| 09:28.10 | vp7 | Can anyone help me with asterisk (cvs) + asterisk-oh323-0.7.3 (+cvs pwlib&openh323)? Without chan_oh323.so asterisk works fine, but after adding it - it says "Segmentation fault" and didn't start at all :( |
| 09:28.12 | h3x0r | there have been 64 bit cpus for 15 years now |
| 09:28.22 | jdv79 | because that's less profittable than ushering in the 64bit shit |
| 09:28.52 | jdv79 | yeah but not mainstream |
| 09:29.03 | jdv79 | vp7, i got something similar too |
| 09:29.06 | jdv79 | no idea |
| 09:29.15 | Speeder | can anybody help me with the zaphfc problem?... |
| 09:29.36 | tzafrir_laptop | Igbothom_III? |
| 09:29.55 | tzafrir_laptop | Speeder, are you sure you use the right ISDN signalling? |
| 09:30.18 | vp7 | jdv79: No idea at all? Bad, too bad... i tried to run asterisk under strace and the last message was "socket(PF_INET6, SOCK_DGRAM, IPPROTO_IP) = -1 EAFNOSUPPORT (Address family not supported by protocol)".. maybe i'll try to find place in code where error occures... |
| 09:30.33 | *** join/#asterisk Speeder (n=psilva@80.172.156.1) |
| 09:30.35 | jdv79 | ouch |
| 09:30.41 | jdv79 | why not use stable? |
| 09:30.45 | tzafrir_laptop | vp7, what distro? what versions of pwlib and oh323? |
| 09:30.56 | Speeder | I think so, is there any way to see what signalling is used? |
| 09:31.30 | vp7 | jdv79: I tried. but didn't find correct versions of asterisk & oh323 & pwlib & openh323 that will work correct togather. if you have it - please, give me URLS |
| 09:31.36 | snitt | someone was hilighting me |
| 09:31.38 | tzafrir_laptop | both in the span definition in zaptel.conf and in the one in zapata.conf |
| 09:31.48 | snitt | ah, it was tzafrir_laptop |
| 09:31.49 | tzafrir_laptop | not sure exactly |
| 09:31.52 | snitt | tnx for info |
| 09:32.18 | h3x0r | the shit just aint gonna fly |
| 09:32.19 | *** join/#asterisk Mother__ (n=Mother@53.Red-217-126-93.staticIP.rima-tde.net) |
| 09:32.21 | h3x0r | just like IPv6 |
| 09:32.24 | Mother__ | greetings |
| 09:32.40 | h3x0r | it addresses a non-problem even, on top of that |
| 09:32.42 | vp7 | tzafrir_laptop: last cvs versions of pwlib&openh323 (downloaded 2-3 hours ago), asterisk is also last cvs + asterisk-oh323-0.7.3 |
| 09:32.50 | Mother__ | anyone have had checksum error problems in IAX2 channels? |
| 09:33.03 | Mother__ | they don't seem to cure restarting asterisk either, only a reboot will do |
| 09:33.07 | Speeder | is a normal isdn line with 2 voice channels e one date channel, this span definition shoul be right, span=1,1,3,ccs,ami |
| 09:33.22 | h3x0r | 64 bit addresses a problem that maybe .05% of computer applications really need |
| 09:33.31 | Mother__ | and yes, I have tried all the ignore checksum settings etc. from what it seems it's the OS that's rejecting the packets |
| 09:33.33 | tzafrir_laptop | Speeder, not sure. I believe it is a bit coutry-dependent |
| 09:33.42 | h3x0r | except in 2038 |
| 09:33.44 | tzafrir_laptop | But don't take my word for it |
| 09:33.52 | h3x0r | when we run out of unix epoch time |
| 09:33.54 | h3x0r | and its 1969 again |
| 09:34.05 | h3x0r | id rather it be 1969 anyway |
| 09:34.20 | tzafrir_laptop | h3x0r, assuming we're not 64bit (or worse: 128bits) by then |
| 09:34.45 | h3x0r | if we're using CPUs still |
| 09:35.13 | *** join/#asterisk szer (n=szer@217.116.36.22) |
| 09:35.26 | szer | hi all |
| 09:35.39 | tzafrir_laptop | heck, we're still using "teletypes", don't we? and on mainframes they still use "virtual punch card readers |
| 09:35.46 | tzafrir_laptop | " |
| 09:36.47 | Mother__ | nobody? |
| 09:36.52 | Mother__ | this sucks |
| 09:36.59 | Mother__ | i.e. the problem sucks |
| 09:37.31 | taec_ | That might suggest it's a hardware problem |
| 09:37.33 | Speeder | what options do i have to change the signalling for a simple isdn line |
| 09:37.33 | taec_ | but I'm guessing |
| 09:37.39 | Mother__ | it's happening on two boxes |
| 09:38.07 | Mother__ | they are linked to each other over IAX2, and when this problem comes up, one shows the other as unavailable |
| 09:40.35 | vp7 | btw, did anyone worked with ooh323c? i've successfully installed it,but it strangely cant' establish connection to H323 gateway - seems codec problem or smth like that. |
| 09:42.53 | Mother__ | maybe running an agi before each call to check iax status, and rebooting the box if unavailable |
| 09:43.11 | Mother__ | sigh |
| 09:43.58 | tzafrir_laptop | Mother__, can the two boxes use other forms of IP communication without checksum errors? |
| 09:44.13 | tzafrir_laptop | Mother__, anything in the kernel logs? |
| 09:48.04 | *** join/#asterisk RoyK (n=roy@80.239.107.70) |
| 09:52.35 | *** part/#asterisk Speeder (n=psilva@80.172.156.1) |
| 09:53.14 | vp7 | btw, did anyone worked with ooh323c? i've successfully installed it,but it strangely cant' establish connection to H323 gateway - seems codec problem or smth like that. |
| 09:54.07 | Mother__ | tzafrir_laptop: yes, I can for example SSH from one to the other no problem |
| 09:54.14 | *** join/#asterisk jaxkz (n=cyrieldo@tbnb-165-194-88.telkomadsl.co.za) |
| 09:54.41 | Mother__ | and tcpdump shows the iax2 packets going out with a bad checksum |
| 09:55.05 | jaxkz | Are the company's available wich provide DID trough sip? |
| 09:55.18 | jaxkz | without paying for every connection? |
| 09:57.25 | L|NUX | jaxkz : www.didx.org |
| 10:00.59 | Mother__ | tzafrir_laptop: nothing in kernel logs either |
| 10:03.34 | Mother__ | bbiab |
| 10:19.40 | *** join/#asterisk r0d3nt (i=r0d3nt@tinfoilhat.net) |
| 10:23.12 | *** join/#asterisk twisted|astricon (n=twisted@asterisk/friend-and-developer/pdpc.professional.twisted) |
| 10:23.12 | *** mode/#asterisk [+o twisted|astricon] by ChanServ |
| 10:23.34 | *** join/#asterisk lilo (i=levin@freenode/staff/pdpc.levin) |
| 10:23.41 | twisted|astricon | hi lilo |
| 10:24.25 | bon | lilo grub :) |
| 10:31.49 | *** join/#asterisk queuetue (n=queuetue@toronto-HSE-ppp4122670.sympatico.ca) |
| 10:39.40 | *** join/#asterisk |cleric| (n=dacleric@p5482C0DB.dip0.t-ipconnect.de) |
| 10:51.09 | *** join/#asterisk danalien (n=danalien@unaffiliated/danalien) |
| 10:55.27 | *** join/#asterisk tatuman (n=Miranda@joltid-gw.joltid.org) |
| 10:55.38 | tatuman | hi |
| 10:55.39 | tatuman | <PROTECTED> |
| 10:56.07 | *** join/#asterisk gvag11 (n=g@ipa243.5.tellas.gr) |
| 10:56.56 | tatuman | can someone help me out |
| 11:00.05 | tatuman | anyone here understands about meetme? |
| 11:02.26 | tatuman | :( |
| 11:05.37 | *** join/#asterisk gordonjcp (n=gordonjc@cpc3-broo2-5-0-cust232.renf.cable.ntl.com) |
| 11:07.35 | *** join/#asterisk Tjief (n=Tjief@r0x0r.dk) |
| 11:07.38 | kshumard_home | tatuman, I think the outgoing stream for a muted user is stopped so is there no transcoding of their audio |
| 11:08.04 | kshumard_home | tatuman, but if they're still listening to the audio from others in the conference that stream will still be transcoded |
| 11:08.19 | kshumard_home | if I'm awake enough to think clearly, that is. : ) |
| 11:09.18 | tatuman | kshumard_home, but if they are listening only |
| 11:09.35 | tatuman | kshumard_home, why should their input stream be transcoded |
| 11:09.50 | tatuman | kshumard_home, unnecessary cpu |
| 11:10.08 | tatuman | kshumard_home, where can i chang this? |
| 11:17.15 | *** join/#asterisk Gunnar (n=gunnar@bkkb-gw.bitcon.no) |
| 11:17.37 | skrusty | does anyone here know anything about the new domain code in SIP? as it's causing me some right problems |
| 11:26.18 | Dr_Ray | http://cgi.ebay.com/TE410P-Quad-T1-PCI-Digium-VOIP-Asterisk-FREE-SHIPPING_W0QQitemZ5818866537QQcategoryZ99269QQssPageNameZWDVWQQrdZ1QQcmdZViewItem |
| 11:26.22 | Dr_Ray | that price has to be wrong |
| 11:26.31 | Mother__ | back |
| 11:27.01 | Mother__ | tzafrir_laptop, you still around? |
| 11:29.07 | kippi | hey |
| 11:29.19 | kippi | i am trying to download asterisk using cvs |
| 11:29.28 | kippi | it gets to cvs server: Updating asterisk/sounds |
| 11:29.32 | kippi | and seems to stop |
| 11:29.42 | kippi | has anyone got any ideas? |
| 11:32.33 | nfi|ermes | anyone can suggest me a good free implementation of G.729 ? |
| 11:34.04 | morale | is g.729 not a free protocol to use? |
| 11:35.18 | nfi|ermes | dont you need a license to use it ? |
| 11:35.45 | morale | not that i know of. |
| 11:36.20 | nfi|ermes | http://www.voip-info.org/wiki/view/Asterisk+codecs |
| 11:36.51 | nfi|ermes | instead, this should be a free implementation: http://www.readytechnology.co.uk/open/ipp-codecs-g729-g723.1/ |
| 11:38.28 | tzafrir_laptop | kippi, there are some large files in that directory |
| 11:38.40 | kippi | so just give it time? |
| 11:38.53 | limbique | hi, question, what linux dest. you use for a latests version of asterisk? |
| 11:39.02 | tzafrir_laptop | 3 large mp3 files, actually. Try to see if cvs has some verbosity flags |
| 11:39.54 | tzafrir_laptop | Mother__, I don't have any new ideas about your problems |
| 11:40.47 | tzafrir_laptop | limbique, distro? debian , of course: deb http://rapid.dotsrc.org/ experimental/ |
| 11:41.34 | kippi | is there no way to get a progress bar to see how much has been downloaded? |
| 11:41.52 | tzafrir_laptop | nfi|ermes, there is an independent implementation of g729 and g723.1, however it is probably illegal to use it in just about anywhere in the world |
| 11:42.24 | nfi|ermes | and what about http://www.readytechnology.co.uk/open/ipp-codecs-g729-g723.1/ ? |
| 11:42.29 | nfi|ermes | is this illegal too ? |
| 11:42.55 | tzafrir_laptop | nfi|ermes, this is what I meant. Read that page. |
| 11:43.25 | nfi|ermes | i've read the instructions |
| 11:43.33 | tzafrir_laptop | near the bottom |
| 11:43.52 | kippi | hmm, i am pritty sure that cvs is stoping on asterisk/sounds |
| 11:44.37 | nfi|ermes | what a damn |
| 11:45.36 | nfi|ermes | which codec should i use for pstn ? |
| 11:46.03 | nfi|ermes | g.711 ha too high bandwith reservation |
| 11:47.57 | nfi|ermes | i'll try G.726 |
| 12:01.18 | limbique | tzafrir: i'm a noob with linux.. |
| 12:01.43 | limbique | is that a release with linux and asterisk? |
| 12:02.47 | case_ | limbique: there is not a single Linux, there are linux distributions (red-hat, debian, suse, etc...) |
| 12:03.05 | case_ | limbique: each of them may have a package to install astrisk smoothly |
| 12:03.35 | case_ | but of course, Debian beats them all ;p </troll> |
| 12:03.37 | limbique | ok! cool |
| 12:03.44 | limbique | why debian beats them? |
| 12:03.58 | limbique | it's for a amdk6-2 400mhz |
| 12:04.15 | case_ | in my opinion, debian has the best package system out there |
| 12:04.24 | *** join/#asterisk Blazint (n=blazin@cm225.epsilon203.maxonline.com.sg) |
| 12:04.42 | limbique | heavy distro? |
| 12:05.00 | case_ | limbique: in my case, i had just to type "apt-get install asterisk", et voila. |
| 12:05.12 | case_ | debian is heavyly scalable |
| 12:06.21 | limbique | ok, but also on my amdk6-2 400mhz? |
| 12:06.28 | limbique | only for 1 or 2 sip clients |
| 12:06.38 | limbique | and a soft-modem |
| 12:06.57 | *** join/#asterisk razu_ (n=razu@217-159-178-92-dsl.prn.estpak.ee) |
| 12:07.31 | *** join/#asterisk jimandreu (n=jim@user-0c99nq6.cable.mindspring.com) |
| 12:08.15 | case_ | limbique: i have a pii-400 runing debian as a server |
| 12:08.22 | limbique | cool |
| 12:08.29 | case_ | limbique: with ssh, smtp, pop, www, dns |
| 12:08.34 | limbique | i don't need any graphical interface |
| 12:08.49 | limbique | graphical interface and server sux :) |
| 12:09.05 | limbique | like with putty |
| 12:09.13 | limbique | so u say, use debian? |
| 12:09.15 | case_ | the debian installer only set up the minimum, then you have to add everything you want |
| 12:09.43 | case_ | it's a little tough for beginer, but it's a realy nice system when you know how to use it |
| 12:09.59 | limbique | ok, |
| 12:10.22 | limbique | but how do i know what to add, if i don't know what i mis? |
| 12:10.45 | case_ | the most important is tu learn how to use the package system (apt) |
| 12:11.03 | case_ | when you ask apt to install something, it will install any depences too |
| 12:11.05 | limbique | ok |
| 12:11.09 | case_ | so you don't have to bother |
| 12:11.13 | limbique | ok, got it |
| 12:11.23 | case_ | (in fact, it will download, and then install) |
| 12:11.28 | *** join/#asterisk shanky (n=shanky@238.Red-80-33-29.staticIP.rima-tde.net) |
| 12:11.34 | shanky | hi, good afternoon |
| 12:11.39 | limbique | so you advice me first to play with debian before i start with asterisk on that computer |
| 12:12.01 | shanky | anyone knows is there any way to get the dcap in europe? |
| 12:12.16 | limbique | at the moment we use fedora here (some other company installed that) |
| 12:12.51 | case_ | limbique: do you have unix skills? |
| 12:13.28 | *** join/#asterisk MozartsGhost (n=ice@dyn.isogo.co.za) |
| 12:13.39 | limbique | nah :( |
| 12:14.01 | *** join/#asterisk pa (n=Paolo@unaffiliated/pa) |
| 12:14.34 | MozartsGhost | Oct 14 14:10:36 WARNING[2378]: chan_capi.c:3125 capi_handle_connect_indication: did not find device for msn = 5162 |
| 12:14.34 | case_ | limbique: you have to know how to use the shell and everything before trying to install a deiban, i think |
| 12:14.39 | MozartsGhost | anyone get this message before ? |
| 12:15.08 | MozartsGhost | can't find a dam thing about it in posts. |
| 12:15.15 | limbique | ok, |
| 12:15.22 | limbique | i can login a system |
| 12:15.31 | limbique | do some file manipulations |
| 12:15.37 | limbique | edit conf file etc. |
| 12:15.43 | limbique | find locate etc. |
| 12:15.51 | limbique | but after that it stops |
| 12:15.52 | case_ | limbique: what editor do you use ? |
| 12:15.55 | limbique | vi |
| 12:15.58 | limbique | and mc |
| 12:16.07 | limbique | there is a pico |
| 12:16.11 | jaxkz | When i only have output termination, how can i supply my customers with DID? |
| 12:16.12 | limbique | but never played with it |
| 12:16.53 | case_ | limbique: you won't have mc at the begining, but vi is here |
| 12:17.01 | limbique | ok |
| 12:17.08 | limbique | i like mc :) |
| 12:17.16 | case_ | i don't :) |
| 12:17.17 | limbique | like my old norton commander |
| 12:17.33 | MozartsGhost | You kinda get the perception this will be like the central Oracle of Asterisk since it is the official channel. but are questions allowed here ? |
| 12:18.09 | case_ | limbique: anyway, if your network is ok and the instllation well done, you'll only have to type "apt-get install mc" to get it installed |
| 12:18.33 | limbique | the biggest probs i got, is 'i don't know how the os works like with drivers etc' |
| 12:18.43 | limbique | cool!!! |
| 12:18.49 | kippi | help! trying to install asterisk and I am geting this error configure: error: installation or configuration problem: C compiler cannot create executables. |
| 12:18.57 | limbique | apt-get is that standard on debian? |
| 12:19.06 | MozartsGhost | yes |
| 12:19.10 | *** join/#asterisk zotz (n=zotz@24.231.36.100) |
| 12:19.23 | case_ | most of the time the debian auto-detect everything, at least for the old hardware like yours |
| 12:19.28 | MozartsGhost | just make shure yorur sources are up to date aswell. |
| 12:19.30 | *** join/#asterisk nfi|ermes (n=ermsewrk@217.220.121.62) |
| 12:19.37 | MozartsGhost | s/yorur/your |
| 12:19.46 | *** join/#asterisk littleball (n=littleba@cm52.epsilon173.maxonline.com.sg) |
| 12:19.47 | case_ | well, everything else ISA cards :) |
| 12:19.48 | RoyK | yououyur |
| 12:20.27 | littleball | hello, is it possile to record the call when i am make a phone call? |
| 12:20.27 | MozartsGhost | lol |
| 12:20.38 | case_ | O_o |
| 12:20.40 | RoyK | littleball: show application monitor |
| 12:20.41 | shanky | anyone here has the Digium Certified Asterisk Proffesional? |
| 12:20.56 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
| 12:21.06 | littleball | thanks. RoyK |
| 12:21.30 | RoyK | littleball: also see the m and M flags on appliocation Dial in cvs head |
| 12:22.40 | MozartsGhost | guys, I got an HFC isdn card running right. got my extension setup for capi-in. exten => 5162,1,Dial(SIP/rasmus). but when I try dial my line. I get capi_handle_connect_indication: did not find device for msn = 5162. i;m pretty shure my capi.conf is correct. |
| 12:22.44 | MozartsGhost | anybody have advice ? |
| 12:23.34 | MozartsGhost | also, from the phone I call from, I get "error in connection". or No Answer. |
| 12:24.34 | nfi|ermes | i have sip channels for internal hosts, and zaptel channels for pstn. How can i use different codecs ? Should i use: allow=codec in zapata.conf and in zip.conf ? |
| 12:25.57 | kippi | anyone got any ideas why i could compile it on redhat WS and not on Feroda 3? |
| 12:26.16 | *** join/#asterisk Samoied (n=Samoied@popeye.opens.com.br) |
| 12:27.31 | MozartsGhost | kippi, install debian :) |
| 12:27.40 | MozartsGhost | nah, kidding. make shure u got all your "dev" stuff in. |
| 12:27.46 | kippi | nope, got to use redhat ::( |
| 12:27.47 | MozartsGhost | libraries etc.. |
| 12:27.55 | RoyK | MozartsGhost: you can't use chan_capi with hfc pci cards |
| 12:28.02 | RoyK | MozartsGhost: use visdn or bristuff |
| 12:28.03 | MozartsGhost | :( |
| 12:28.18 | RoyK | or isdn4linux, although that sucks |
| 12:28.18 | MozartsGhost | lol,ok. ta. I feel like a real tart now. |
| 12:28.24 | MozartsGhost | ye |
| 12:28.24 | kippi | MozartsGhost: how can i check that? |
| 12:28.27 | MozartsGhost | ok, will go for bristuff |
| 12:28.40 | RoyK | ~bristuff |
| 12:29.14 | *** join/#asterisk lehel (n=asd@82.79.20.17) |
| 12:29.27 | RoyK | jbot: bristuff is a patch collection with a hfc-pci driver supporting NT and TE mode ISDN. See http://junghanns.net/ for more info |
| 12:29.28 | jbot | okay, RoyK |
| 12:29.30 | RoyK | ~bristuff |
| 12:29.31 | jbot | hmm... bristuff is a patch collection with a hfc-pci driver supporting NT and TE mode ISDN. See http://junghanns.net/ for more info |
| 12:30.20 | nfi|ermes | i use bristuff |
| 12:30.29 | nfi|ermes | it works well |
| 12:30.59 | RoyK | yes |
| 12:31.17 | RoyK | although I had problem using asterisk stable with it last i tried |
| 12:31.17 | nfi|ermes | RoyK, can u help me with mmy question |
| 12:31.24 | RoyK | perhaps |
| 12:31.26 | RoyK | maybe |
| 12:31.27 | RoyK | possibly |
| 12:31.29 | nfi|ermes | :D |
| 12:31.38 | MozartsGhost | kippi, not 100% shure how fedora's packages work. don't use it at all. |
| 12:31.45 | MozartsGhost | should be able to google it easily tho |
| 12:31.53 | MozartsGhost | ta RoyK |
| 12:33.21 | nfi|ermes | [14:24] <nfi|ermes> i have sip channels for internal hosts, and zaptel channels for pstn. How can i use different codecs ? Should i use: allow=codec in zapata.conf and in zip.conf ? |
| 12:33.45 | nfi|ermes | i d like to use g.711 in my lan and g726 for pstn |
| 12:34.11 | nfi|ermes | now i m using g711 everywhere |
| 12:34.50 | RoyK | er |
| 12:34.56 | RoyK | pstn is g.711a only |
| 12:35.03 | *** join/#asterisk ful|work (n=fulgas@213.58.130.46) |
| 12:35.05 | RoyK | unless you're in northern america |
| 12:35.09 | RoyK | there it's g.711u |
| 12:35.28 | RoyK | nfi|ermes: why would you want to use anything else for pstn?????? |
| 12:35.38 | nfi|ermes | to take less bandwith |
| 12:35.41 | RoyK | you won't get more bchans on the PRI if that's what you think....... |
| 12:36.00 | RoyK | pstn is channel switched/multiplexed |
| 12:36.10 | RoyK | you don't talk about bandwidth in there |
| 12:36.31 | nfi|ermes | codec implementation speak about bandwith |
| 12:39.41 | RoyK | yes |
| 12:39.43 | RoyK | on IP |
| 12:39.52 | RoyK | ISDN uses 64kbps per channel |
| 12:40.01 | RoyK | multiplexed timeslots |
| 12:40.22 | nfi|ermes | ok |
| 12:40.25 | RoyK | analog phones use audio over a pair of wires |
| 12:40.27 | nfi|ermes | thx so much |
| 12:40.48 | nfi|ermes | i misss some basic knowledge on telephony system |
| 12:41.34 | *** join/#asterisk toddf (n=toddf@ns0.fries.net) |
| 12:42.55 | *** part/#asterisk Mike9 (n=sturdee@ireland.pathwaynet.com) |
| 12:46.33 | *** join/#asterisk damned (n=Victor@prior.lanck.net) |
| 12:46.45 | damned | hi all |
| 12:47.03 | damned | is there any doc how to start with asterisk ? |
| 12:47.55 | shanky | damned: sure |
| 12:48.15 | shanky | wait a minute |
| 12:48.43 | shanky | http://www.voip-info.org/wiki-Asterisk |
| 12:48.57 | lehel | ;) |
| 12:49.10 | RoyK | Damin: |
| 12:49.14 | RoyK | damned: see |
| 12:49.16 | RoyK | ~docs |
| 12:49.19 | jbot | docs is probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk |
| 12:49.38 | RoyK | damned: if you can afford it, try the book from oreilly |
| 12:50.58 | kippi | anyone used Text-to-speech? |
| 12:51.08 | damned | yup.. thanx |
| 12:54.26 | *** join/#asterisk corne (n=corne@ndn-165-141-208.telkomadsl.co.za) |
| 12:54.59 | *** join/#asterisk iCEBrkr (i=icebrkr@rrcs-24-129-130-158.se.biz.rr.com) |
| 12:55.06 | corne | lo |
| 12:56.04 | Katty | mew. |
| 12:56.43 | skrusty | anyone know what `Error in PREPARE -1` means from cdr_odbc ? |
| 12:56.49 | *** join/#asterisk coppice (n=chatzill@190.196.17.210.dyn.pacific.net.hk) |
| 12:56.59 | corne | hi all. i am a newbee both in asterisk. i would like to know if anybody here would be able to help me setup speed dial on asterisk |
| 12:58.41 | iCEBrkr | Wouldn't a speeddial just be a *123 type extension that just does a Dial()? |
| 12:59.51 | iCEBrkr | exten => *1,1,Macro(setCID) |
| 12:59.51 | iCEBrkr | exten => *1,2,Dial(IAX2/${VPID}@vpconnect-t01/12165551212) |
| 12:59.54 | iCEBrkr | Something like that? |
| 13:01.09 | corne | iCEBrkr, thanx |
| 13:01.37 | iCEBrkr | corne: Obviously that's going to need some tweaking for your configuration |
| 13:01.53 | iCEBrkr | corne: But if you look up how to use the Dial() app, you should figure it out |
| 13:02.16 | *** join/#asterisk prh (n=paul@wacka.mjr.org) |
| 13:02.28 | *** join/#asterisk viLeR (i=1000@66.128.47.232) |
| 13:02.36 | corne | iCEBrkr i figured. you gave me direction, much apreciated |
| 13:02.47 | iCEBrkr | np |
| 13:03.20 | riksta | anyone here using Sangoma A101 cards? |
| 13:12.31 | RoyK | only A102 anad A104 |
| 13:12.33 | RoyK | er |
| 13:12.37 | RoyK | no I have one A101 as well |
| 13:12.47 | morale | great.. i signed up with inphonex.com last night asked for a DID in area code 403 and they gave me one in 786 which is florida.. 3000 miles away. |
| 13:13.18 | *** join/#asterisk MikeJ[Laptop] (n=ircatjer@mi.origenfinancial.com) |
| 13:14.56 | iCEBrkr | Doh |
| 13:15.01 | iCEBrkr | morale: I'll take it :P |
| 13:15.33 | iCEBrkr | Miami is only 4hrs from me. |
| 13:18.25 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 13:23.40 | RoyK | calculated in timezones |
| 13:26.47 | *** join/#asterisk e3g (i=ee@u15157627.onlinehome-server.com) |
| 13:26.52 | e3g | hi |
| 13:27.56 | e3g | I am getting "choppy sound" in calls recording ...... but no problem in conversation ... |
| 13:28.03 | e3g | any help ? |
| 13:31.13 | *** join/#asterisk hotgrits (n=hotgrits@192.160.238.156) |
| 13:34.30 | *** join/#asterisk usm (n=usm@60.49.71.133) |
| 13:35.00 | riksta | RoyK: have you had any trouble with the cards not detecting hangups correctly? |
| 13:35.20 | *** join/#asterisk gambolputty (n=gambolpu@72.240.241.108) |
| 13:36.55 | e3g | how do I uninstall SOX ? |
| 13:37.24 | *** join/#asterisk Ariel_ (n=Ariel@adsl-068-157-125-248.sip.mia.bellsouth.net) |
| 13:38.16 | Ariel_ | Hello, hello, Hello. Hope everyone is doing great. |
| 13:38.34 | e3g | hi Ariel_ |
| 13:38.57 | e3g | u gonna be helpful today :) |
| 13:38.59 | e3g | hi Ariel_ |
| 13:39.06 | e3g | how do I uninstall SOX ? |
| 13:39.35 | coppice | just pull them off, and throw them in the wash |
| 13:39.44 | Ariel_ | e3g, today only.... humm I feel I am helpful all the time. |
| 13:39.55 | Ariel_ | ROFL |
| 13:40.14 | Ariel_ | e3g, that depends on which distro you have. |
| 13:40.19 | e3g | :D |
| 13:40.22 | e3g | REH HAT 9 |
| 13:40.26 | e3g | actually ...the problem is |
| 13:40.28 | snitt | LOL |
| 13:40.35 | e3g | I am getting "choppy sound" in calls recording ...... but no problem in conversation ... |
| 13:40.50 | Ariel_ | what does sox have to do with that? |
| 13:41.09 | e3g | everything was fine till yesterday :S |
| 13:41.13 | e3g | then ??? |
| 13:41.32 | Ariel_ | well what did you do? |
| 13:41.40 | e3g | nothing..... |
| 13:42.02 | e3g | believe me .. I didnt do anything... |
| 13:42.26 | e3g | conversations are fine.....but getting so much distortion in the recording.. |
| 13:42.34 | *** join/#asterisk jcollins (n=chatzill@ptech7-127.acdmis.com) |
| 13:43.29 | Ariel_ | nothing never happens to it's self. |
| 13:43.43 | jcollins | is there any way via the Asterisk Manager API to place an outbound call (got this part done) and play DTMF tones to the external side? |
| 13:44.03 | Ariel_ | You did not upgrade? Your network did not change? you have not run out of disk space? you did not reboot? |
| 13:44.05 | e3g | Ariel_ : should I check it by changing codec? |
| 13:44.24 | e3g | I did ..just reboot .. |
| 13:44.34 | e3g | Asterisk CVS HEAD built by root@localhost.localdomain on a i686 running Linux on 2005-10-11 07:47:38 UTC |
| 13:44.43 | e3g | see! ... it's a new box |
| 13:45.46 | MozartsGhost | ~Schweet ! |
| 13:45.48 | MozartsGhost | mweheahea |
| 13:45.51 | MozartsGhost | got it working :) |
| 13:46.14 | MozartsGhost | only thing is. like. sounds like someone is tapping their finger on the mic. the whole time. |
| 13:46.17 | MozartsGhost | what would that be ? |
| 13:46.37 | coppice | the rhythm section? :-) |
| 13:47.09 | MozartsGhost | lol |
| 13:48.59 | e3g | Ariel_ : I got it |
| 13:49.01 | e3g | I got it |
| 13:49.11 | e3g | ohhh man I cant believe it !!! |
| 13:49.20 | *** join/#asterisk nomazda (i=nyyankee@user-0c6tnqf.cable.mindspring.com) |
| 13:49.25 | *** join/#asterisk mcf3782 (n=mcf3782@adsl-065-012-184-148.sip.asm.bellsouth.net) |
| 13:49.28 | Ariel_ | e3g, well? |
| 13:49.55 | e3g | Ariel_: I did ftp from Linux to my machine and uploaded those files ....and got choppy sound |
| 13:50.14 | e3g | but when i do Background() with the same file........ |
| 13:50.20 | e3g | no problem at all :S |
| 13:51.30 | e3g | what is this shit |
| 13:51.38 | e3g | eeekhh |
| 13:51.39 | e3g | thanks |
| 13:51.40 | e3g | bye |
| 13:51.43 | *** part/#asterisk e3g (i=ee@u15157627.onlinehome-server.com) |
| 13:51.45 | MozartsGhost | o_O |
| 13:52.16 | MozartsGhost | ~codec |
| 13:52.18 | jbot | methinks codecs is http://snipurl.com/wiki_codecs |
| 13:52.18 | MozartsGhost | ~codecs |
| 13:52.19 | jbot | codecs is probably http://snipurl.com/wiki_codecs |
| 13:52.38 | MozartsGhost | jbot, you rock. |
| 13:52.38 | jbot | MozartsGhost: thanks |
| 13:52.53 | MozartsGhost | np. |
| 13:53.34 | mcf3782 | I would like to have a VoIP provider who can provide me with Atlanta, Georgia inbound and outbound service. Anyone in here have any providers I should definitely steer clear of as I start my search? |
| 13:53.39 | *** join/#asterisk wmandra (n=me@pcp04943183pcs.verona01.nj.comcast.net) |
| 13:53.58 | wmandra | morning all.... |
| 13:55.48 | Ariel_ | mcf3782, I would say the good ones that I use are asterlink, VoicePulse, I have also used Teliax they have all been good. The one that I have had mixed feeling about has been broadvoice. |
| 13:56.51 | mcf3782 | Ariel_: Thanks for the info. How's the baby girl? :) |
| 13:57.23 | riksta | RoyK: ping |
| 13:58.54 | Ariel_ | mcf3782, great thanks |
| 13:59.11 | rayvd | What is the proper way to send ANI over SIP? From: field? or Calling-Party-ID field? |
| 14:01.28 | bon | hm |
| 14:01.34 | bon | if i have * with sip phones |
| 14:01.39 | bon | and i have some old h323 phones |
| 14:01.42 | bon | can i make a gw or something? |
| 14:01.44 | wmandra | hey all, maybe someone here can give me some advice... I've been using * for almost 2 years, originally I set it up with a Broadvoice account, but I think it's time to switch, I just can't justify paying fees for a service when I don't use 90% of the features. The only thing I need is outbound/inbound DID termination through either SIP/IAX. anyone have any suggestions? |
| 14:01.53 | Ariel_ | I was under the impression ani was something you send via a PRI or E1 |
| 14:03.10 | jcollins | is there any way to automate the playing of dtmf tones to the external side of a call using asterisk? |
| 14:05.09 | *** join/#asterisk dalfry (n=dalfry@gateway.ishisystems.com) |
| 14:05.58 | kippi | has anyone linked a avaya system or a cisco phone system into asterisk? |
| 14:06.43 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 14:07.09 | puzzled | afternoon all |
| 14:07.29 | BrianR___ | jcollins: senddtmf |
| 14:07.56 | BrianR___ | jcollins: or the D() option to dial if you just want to send digits right after dialing. |
| 14:08.10 | jcollins | BrianR___: I don't see that in the Asterisk Manager API |
| 14:08.16 | tzafrir_laptop | e3g: rpm -U sox ? |
| 14:08.35 | *** join/#asterisk Assid (n=assid@203.115.64.61) |
| 14:08.37 | Assid | heya |
| 14:08.47 | Assid | umm.. anyone tried the sipura ata 2000 |
| 14:08.49 | jcollins | or is that something I would put into the extension that Originate sends the call to? |
| 14:09.35 | tzafrir_laptop | Assid, yes, we have two Sipura 2000 boxes in the office. nice. |
| 14:10.49 | BrianR___ | jcollins: You're trying to do it from asterisk manager? |
| 14:10.49 | Assid | any issues with configuring or anything? |
| 14:10.51 | wmandra | who is everyone here using as their voip provider?? |
| 14:11.01 | tzafrir_laptop | Assid, basically it gives you two independent "sip phones" to configure. |
| 14:11.10 | jcollins | BrianR___: that's how I'm initiating the external call, yes |
| 14:11.33 | tzafrir_laptop | Simply tell them that Asterisk is their SIP proxy. |
| 14:12.00 | jcollins | and I need the DTMF tones sent to the external party, not the the asterisk side... |
| 14:12.19 | BrianR___ | jcollins: So you're using Dial as the application in originate? |
| 14:12.28 | Assid | tzafrir: so i can connect my normal wireless phone ? |
| 14:12.36 | tzafrir_laptop | Assid, yes |
| 14:13.03 | jcollins | BrianR___: no, I'm using the Originate "action" in the Manager API |
| 14:13.11 | *** join/#asterisk \Grooby\ (n=Grooby@ip24-250-126-171.dc.dc.cox.net) |
| 14:13.18 | *** join/#asterisk Pr0ph37 (i=user@adsl-067-034-122-207.sip.mia.bellsouth.net) |
| 14:13.20 | BrianR___ | jcollins: Yes. What arguments to originate are you using? |
| 14:13.23 | mcf3782 | jcollins:Have you looked at "show application SendDTMF" from the Asterick console CLI to see if that will do what you need? |
| 14:13.24 | bjohnson | bon: yes |
| 14:13.31 | kippi | has anyone got a demo voice of what festival sounds like?? |
| 14:13.49 | jcollins | ACTION: Originate |
| 14:13.49 | tzafrir_laptop | kippi, not grand |
| 14:13.57 | jcollins | Channel: ZAP/1/xxxxxx |
| 14:14.09 | jcollins | Exten: s |
| 14:14.12 | Assid | tzafrir_laptop: anything for gsm? |
| 14:14.14 | jcollins | Priority: 1 |
| 14:14.18 | jcollins | Context: default |
| 14:14.23 | kippi | if I was setting up a AA for a cinema saying what is on, what it do the trick? |
| 14:14.30 | *** join/#asterisk olivier_ (n=olivier_@obs92-4-82-239-116-113.fbx.proxad.net) |
| 14:14.43 | tzafrir_laptop | Assid, if you connect it in a LAN, you'll use ulaw anyway |
| 14:14.47 | mcf3782 | kippi: I think there is some demo speech on Festival's web site. |
| 14:14.48 | jcollins | it's just been for testing to get the Asterisk to actually place the external call, which it is now doing... |
| 14:15.01 | BrianR___ | jcollins: Ok. You do relize you can specify Application: Dial instead of a channel, right? |
| 14:15.08 | Assid | tzafrir_laptop: nah.. im thinking of making it my intl. line.. when the computer isnt on |
| 14:15.19 | BrianR___ | jcollins: Also, if default,s,1 does a SendDTMF, it will be sent to Zap/1 in your example. |
| 14:15.40 | jcollins | BrianR___: no I wasn't aware of that |
| 14:15.55 | BrianR___ | Hmm... No.. I take that back about using Application: dial - that won't do what you want. |
| 14:15.58 | tzafrir_laptop | Assid, maybe it supports g726. I'm not sure |
| 14:16.13 | jcollins | the wiki made no meantion of Application: dial for Originate |
| 14:16.17 | BrianR___ | But SendDTMF run from the target context should do what you want. |
| 14:16.33 | BrianR___ | jcollins: You can use APplication: dial for originate in order to implement things like click to dial |
| 14:16.50 | jcollins | cool, the console information from "show application senddtmf" isn't clear on which side is getting the DTMF |
| 14:17.05 | BrianR___ | jcollins: Like Channel: SIP/3663\nApplication: Dial\nData: Zap/411 would cause my phone to ring and 411 to be connected when I pick it up. |
| 14:17.36 | jcollins | very interesting... |
| 14:17.58 | BrianR___ | jcollins: The connected party gets the DTMF, so it's the caller (or callee if you're using a outgoing spool file or manager to start the call) |
| 14:18.19 | *** part/#asterisk Pr0ph37 (i=user@adsl-067-034-122-207.sip.mia.bellsouth.net) |
| 14:18.29 | Assid | g726 ? |
| 14:18.34 | Assid | yes it does |
| 14:18.36 | Assid | but what is it? |
| 14:18.42 | jcollins | sounds like what I need then, I appreciate your help... |
| 14:18.45 | Assid | isnt g726 commercial? |
| 14:18.56 | coppice | no |
| 14:19.10 | Assid | how does g726 differ from gsm? |
| 14:19.10 | olivier_ | u just have to install it |
| 14:19.11 | tzafrir_laptop | g711 isn't patented either |
| 14:19.18 | wmandra | g729 requires a licence |
| 14:19.52 | tzafrir_laptop | Assid, not sure exactly |
| 14:20.14 | tzanger | Assid: how does a CD differ from a DVD? |
| 14:20.27 | Assid | hrmm |
| 14:21.00 | Assid | G.711 a-law, G.711µ-law, G.726, G.729 A, G.723.1 |
| 14:21.06 | Assid | thats all this device supports |
| 14:21.24 | Assid | linksys pap2 |
| 14:21.57 | *** join/#asterisk pa (n=Paolo@unaffiliated/pa) |
| 14:22.11 | tzafrir_laptop | Assid, 'show translations' shows that g726 is a bit more cpu-intensive than gsm, but not much |
| 14:23.59 | Assid | cant see anything comparing ilbc to 726 |
| 14:25.11 | *** join/#asterisk nexis (n=nexis@12-219-60-252.client.mchsi.com) |
| 14:25.13 | coppice | Assid: G.726 is much higher bit rate than iLBC |
| 14:25.28 | Assid | how much bandwith does it require? |
| 14:26.13 | coppice | 32Kbps for the codec, plus the usual massive RTP overhead. |
| 14:26.33 | Assid | hrmm |
| 14:26.57 | Assid | so around 55kbps line |
| 14:27.45 | Assid | say around 7K/sec |
| 14:27.48 | *** join/#asterisk Xen^ (i=linux@202.5.131.69) |
| 14:29.17 | *** join/#asterisk mithro (n=tim@tagung-233-168.tagung.uni-hamburg.de) |
| 14:29.28 | *** join/#asterisk wrmem (n=monnin@monnin-win.cso.uiuc.edu) |
| 14:31.05 | limbique | apt-get on fedora? |
| 14:31.16 | ian_k | yum |
| 14:31.21 | ian_k | not apt-get |
| 14:31.55 | tzafrir_laptop | apt-get can be installed on Fedora, if you really insist |
| 14:32.12 | limbique | how? |
| 14:32.23 | tzafrir_laptop | I recall yum used to be really slow. But maybe it was fixed in later versions |
| 14:32.36 | \Grooby\ | FC3 seems to be faster than FC2 |
| 14:32.42 | tzafrir_laptop | limbique, rpm -Uv apt-whatever.rpm |
| 14:32.44 | \Grooby\ | and don't really have much prob with centos 4 |
| 14:32.51 | \Grooby\ | so they *should* be fixed |
| 14:33.32 | tzafrir_laptop | This is from two or three years ago when I had RH at work and Debian at home |
| 14:34.58 | limbique | ok, |
| 14:35.28 | limbique | wow, many options on rpm |
| 14:35.39 | *** join/#asterisk mhnoyes (n=mhnoyes@user-2ivfk05.dialup.mindspring.com) |
| 14:36.29 | wmandra | who is everyone here using as their voip providers?? |
| 14:36.56 | limbique | sipo.voipbuster.com |
| 14:37.02 | limbique | -o |
| 14:37.08 | limbique | getting coffee |
| 14:38.58 | BrianR___ | crap. The changelog for 1.0.x zaptel is not very useful :( |
| 14:39.11 | ian_k | do cvs diff's |
| 14:39.51 | *** join/#asterisk nshm (n=shmyrev@t25.niisi.ras.ru) |
| 14:40.21 | BrianR___ | ian_k: good idea. Is there a cvsweb somewhere to make this easy? |
| 14:40.36 | *** join/#asterisk iCEBrkr (i=icebrkr@rrcs-24-129-130-158.se.biz.rr.com) |
| 14:41.32 | BrianR___ | found it http://cvsweb.digium.com/index.cgi/ |
| 14:42.54 | bjohnson | for red hat, apt is much faster than yum |
| 14:43.12 | bjohnson | but for whatever reasons, many of the repos have dropped apt support and gone with yum |
| 14:43.14 | limbique | a, yum is a application, not a sound :) |
| 14:43.24 | limbique | stupid names in linux world |
| 14:43.37 | bjohnson | yeah, not cool like longhorn |
| 14:43.38 | tzafrir_laptop | "rpm at idle" |
| 14:43.44 | ian_k | BrianR___ : Not that I am aware of. You can certainly use regular CVS. Let me know if you need CVSROOT or anything |
| 14:43.48 | limbique | hehe |
| 14:43.58 | bjohnson | who's getting the horns on that deal? |
| 14:44.03 | tzafrir_laptop | (this is the subtitle of the rpm book) |
| 14:44.25 | ian_k | BrianR___ : Oh yeah.. forgot about that one. :) |
| 14:44.38 | limbique | windows is easy |
| 14:44.46 | limbique | linux i'm a noob with that |
| 14:45.39 | BrianR___ | just trying to see what's changed in the echo canceller stuff to see if upgrading from 1.0.6 to 1.0.9 will fix my problem. |
| 14:46.17 | tzafrir_laptop | BrianR___, there is zaptel 1.0.9.2 . Echo canceller is in zaptel |
| 14:46.37 | BrianR___ | tzafrir_laptop: Yeah.. I'm diffing the zaptel package. |
| 14:46.53 | BrianR___ | Is the EC in the kernel or userspace part of zaptel? |
| 14:47.09 | ian_k | zaptel = modules = kernel |
| 14:47.17 | tzafrir_laptop | BrianR___, what's wrong with cvs diff -rv1-0-6 -r v1-0 ? |
| 14:47.35 | tzafrir_laptop | the EC is in the module zaptel, IIRC |
| 14:47.49 | BrianR___ | Yeah.. Looking at it now, the only userspace stuff is a handful of config utils. |
| 14:48.35 | tzafrir_laptop | There are a bunch of ECs, implemented in a separate .h file (zconfig.h ?) |
| 14:48.38 | MozartsGhost | aah, for all those using HFC cards with asterisk, I just found that if u remove the modes=1 from loading the modules zaphfc. then it stops the choppy sound. and improves the quality |
| 14:48.42 | MozartsGhost | hope that helps someone ! |
| 14:48.46 | tzafrir_laptop | grep for _ec or something similar |
| 14:49.03 | BrianR___ | tzafrir_laptop: yeah.. .looking at it now. |
| 14:52.15 | BrianR___ | tzafrir_laptop: nothing - just have to pull the trees down off cvs in order to do that |
| 14:52.34 | *** join/#asterisk wolfson (n=hehe@usr-kdh-208-6-58-26.beachlink.com) |
| 14:56.13 | *** join/#asterisk danalien (n=danalien@unaffiliated/danalien) |
| 14:56.31 | *** join/#asterisk brimstone (n=brimston@pdpc/sponsor/digium/brimstone) |
| 14:57.27 | nshm | Hello all |
| 14:57.51 | nshm | Does anyone knows any free analog vocoder? |
| 14:58.04 | *** join/#asterisk Craziman2 (n=Craziman@boromir.apid.com) |
| 14:58.19 | nshm | For example one that will change speech freqency range to some predefined range? |
| 14:59.46 | Craziman2 | Is there is a way to show Alarms from the asterisk console on the 4-port T1 cards? |
| 15:01.14 | coppice | nshm: there is software to change the speed without changing the pitch. that can easily eb reworked to do the opposite |
| 15:01.54 | *** join/#asterisk desktophero (n=chatzill@ip24-56-30-250.ph.ph.cox.net) |
| 15:02.17 | kippi | what is Asterisk Head |
| 15:02.18 | nshm | actually I'd like to do some compression of bandwidth also, like using freqency +- 500 Hz |
| 15:02.38 | nshm | I wonder why there is such free linux coder :( |
| 15:03.26 | *** join/#asterisk Xen^ (i=linux@202.5.132.26) |
| 15:03.28 | nshm | Coder with formant detection would be optimal |
| 15:05.19 | coppice | nshm: what are you actually trying to achieve? |
| 15:06.05 | nshm | encode speech to pass through analog device that has small bandwidth |
| 15:06.16 | nshm | and then decode it |
| 15:06.45 | *** join/#asterisk ful|work (n=fulgas@213.58.130.46) |
| 15:07.02 | nshm | coppice: thanks for suggestion anyhow |
| 15:07.25 | h3x0r | uh |
| 15:07.29 | tzafrir_laptop | a review of free codecs: http://software.newsforge.com/article.pl?sid=05/09/28/1646243&from=rss |
| 15:07.36 | h3x0r | its called speex and ilbc |
| 15:07.47 | h3x0r | you are talking about a codec |
| 15:07.53 | h3x0r | asterisk has tons of those |
| 15:08.25 | *** join/#asterisk supaigtr (n=yurplsl@152.53.16.10) |
| 15:08.26 | tzafrir_laptop | h3x0r, that article is an interesting reading. Well, sort of. |
| 15:08.38 | *** join/#asterisk |cleric| (n=dacleric@p54829745.dip0.t-ipconnect.de) |
| 15:08.39 | coppice | its full of errors |
| 15:08.58 | tzafrir_laptop | could you point some? |
| 15:09.37 | coppice | the descriptions of what the codecs are is mostly wrong |
| 15:10.43 | skrusty | is there still the issue of datetime with mssql when using cdr_odbc? |
| 15:11.30 | nshm | they all are digital :( |
| 15:11.34 | h3x0r | and the guy that did the first reply to the article is a retard |
| 15:11.44 | h3x0r | Even Digium, the Atserisk developers have chosen to charge for their G.729 codec because it took a lot of development effort and is the best codec. Until someone develops the best codec and chooses to make it free, the better systems will use proprietary codecs. |
| 15:11.52 | h3x0r | digium didnt "choose" to charge for it |
| 15:11.55 | h3x0r | and they didnt even write it |
| 15:12.04 | lehel | where is the asterisk 1.0.9 stable debian package? [need to update sources.list] |
| 15:12.12 | Craziman2 | Can some one point me to a doc that explains Kewlstart? |
| 15:13.04 | coppice | h3x0r: rubbish. G.729 just got there first and grabbed the high ground. now you need to use it for compatibility. |
| 15:15.16 | h3x0r | no man i mean |
| 15:15.25 | h3x0r | hes saying that digium wrote that code and "decided" to charge for it |
| 15:15.48 | coppice | there are a number of better codecs than G.729 these days. I bet the damn thing has enough traction to stick around, though |
| 15:16.04 | brookshire | a whole bunch of people wrote the codec... |
| 15:16.11 | coppice | h3x0r: oh, he's the usual clueless idiot |
| 15:16.22 | brookshire | but digium did not write it |
| 15:16.33 | coppice | brookshire: do you mean wrote, or developed? |
| 15:16.49 | brookshire | well.. |
| 15:17.08 | *** join/#asterisk mkrufky (n=mk@68.160.103.77) |
| 15:17.11 | coppice | even the reference code was not written by the main developers |
| 15:17.20 | brookshire | there are serveral patents wrapped up into g729 |
| 15:17.28 | brookshire | i think nokia has a few, lucent, etc |
| 15:18.00 | brookshire | and they want money for each channel they give you permission to use |
| 15:18.02 | coppice | not nokia. lucent I think. the main ones are Sherbrook U and France Telecom |
| 15:18.08 | file | brookshire: Matttt |
| 15:18.36 | brookshire | hey file :) |
| 15:18.55 | brookshire | about the only way g729 can be given away in gpl is for testing purposes.. |
| 15:19.12 | brookshire | any kind of commerical use must be charged... |
| 15:19.38 | MozartsGhost | ~hfc |
| 15:19.45 | h3x0r | s/testing/educational/ |
| 15:19.46 | MozartsGhost | ~hfcpci |
| 15:19.53 | coppice | the stuff in the GPL about patents seems to stop you even calling an educational implementation GPL |
| 15:20.11 | olivier_ | ~google hfc |
| 15:20.15 | olivier_ | arf :) |
| 15:20.20 | MozartsGhost | lol |
| 15:20.22 | nshm | Btw, I did work on making g729 thread safe |
| 15:20.22 | nshm | http://nshmyrev.narod.ru/g729/openh323-g729-2.tar.gz |
| 15:20.46 | nshm | It's modified ITU reference code |
| 15:21.02 | h3x0r | what does g.729 have to do with h323 |
| 15:21.21 | brookshire | heh.. |
| 15:21.28 | brookshire | well h323 is a voip protocol |
| 15:21.30 | nshm | It's codec for opeh323 |
| 15:21.44 | *** join/#asterisk bongfrog (n=winston@206.165.75.198) |
| 15:22.13 | bongfrog | whooo hoooo thanks to DIGIUM for the Hang Over ;) |
| 15:22.26 | nshm | Interesting to know, do I understand correctly that it will expire soon? |
| 15:22.35 | nshm | I mean patent on G.729 |
| 15:22.42 | awannabe | bongfrog: astricon 2005? |
| 15:22.53 | h3x0r | by the time that patent expires |
| 15:22.58 | h3x0r | we won't use voip or phones anymore |
| 15:22.58 | h3x0r | haha |
| 15:23.01 | brookshire | it probably won't expire in the usefully lifetime |
| 15:23.01 | *** join/#asterisk Lee__ (n=Lee__@dsl254-122-010.nyc1.dsl.speakeasy.net) |
| 15:23.09 | brookshire | useful also, lol |
| 15:23.46 | coppice | there is no "patent" of G.729". There are a number of patents on things which G.729 needs. Some have about 10 years to run outside the US, and 15 inside |
| 15:24.08 | nshm | 10 years from 1995 = 2005? |
| 15:24.25 | h3x0r | 2015 |
| 15:24.42 | h3x0r | 10 MORE years |
| 15:24.54 | h3x0r | that type of patent is 18 years here i thinkj |
| 15:25.22 | brookshire | asterisk will support a high quality codec by then, definately |
| 15:25.30 | brookshire | one with mp3 quality :) |
| 15:25.32 | tzafrir_laptop | speex |
| 15:25.38 | coppice | 10 years from now. patents last 15 years in most countries, and some were only issued in 1999 (OK, that's 9 years) |
| 15:26.32 | tzafrir_laptop | And in 10 years we won't really care about the extra CPU speex takes |
| 15:26.45 | nshm | too long :( |
| 15:27.00 | h3x0r | speex sounds like crap |
| 15:27.05 | coppice | speex no longer takes massively more CPU |
| 15:27.25 | tzafrir_laptop | no longer: which version? |
| 15:27.26 | h3x0r | somebody rewrote it? |
| 15:27.27 | coppice | h3x0r: you did something wrong. speex is very comparable with G.729 |
| 15:27.41 | h3x0r | it is? |
| 15:27.50 | h3x0r | that code must have really sucked then |
| 15:28.18 | *** join/#asterisk Seyr (n=Seyr@rrcs-24-227-133-226.sw.biz.rr.com) |
| 15:28.29 | tzafrir_laptop | 'show translations' shows speex as no. 1. pitty the larger the number the worse |
| 15:28.35 | coppice | there are implementations for some DSP chips now, as well |
| 15:28.52 | coppice | duh! its *. it never keeps up with the times |
| 15:28.54 | Seyr | im trying to get ztdummy working on 2.6 kernel and when I do a modprobe, I get this error: FATAL: Error inserting zaptel (/lib/modules/2.6.9-11.ELsmp/kernel/extra/zaptel.ko): Invalid module format |
| 15:29.43 | Seyr | anyone know what I might have done wrong/not done? |
| 15:30.27 | nshm | wrong compiler? |
| 15:30.43 | mutilator | anyone have a copy of the EMI output specification |
| 15:30.45 | mutilator | ? |
| 15:31.02 | supaigtr | Anyone have any exp with SER, * VM, and Maxtnt? |
| 15:31.09 | mutilator | noti |
| 15:31.09 | supaigtr | mutilator: CABS? |
| 15:31.40 | mutilator | emi.. |
| 15:31.44 | *** join/#asterisk marc324 (n=marc3234@206-248-159-56.dsl.teksavvy.com) |
| 15:31.50 | supaigtr | Yea EMI used with cabs? |
| 15:32.21 | mutilator | for processing emi records myself |
| 15:33.24 | mutilator | cabs stand for? |
| 15:35.11 | szer | bye |
| 15:37.31 | *** join/#asterisk hellagony (n=egutierr@host196-60.pool82107.interbusiness.it) |
| 15:38.29 | *** join/#asterisk Guggemand (i=irc@tester2.har-tabt.dk) |
| 15:38.29 | *** join/#asterisk Defraz (n=t0tal@72.24.26.215) |
| 15:38.29 | supaigtr | Carrier access billing. |
| 15:39.41 | *** join/#asterisk Tili (n=Tili@202-133-67-235-dialup.sat.net.pk) |
| 15:40.13 | morale | i can't get asterisk to register my account.. anyone use inphonex? |
| 15:40.34 | morale | Oct 14 09:36:30 NOTICE[18687]: chan_sip.c:6882 handle_response: Failed to authenticate on REGISTER to '<sip:7701234@sip.inphonex.com>;tag=as531019ca' |
| 15:40.49 | InfraRed | check your password |
| 15:41.19 | morale | the password is correct. |
| 15:41.32 | InfraRed | run sip debug and log the output |
| 15:41.35 | InfraRed | then read it |
| 15:41.48 | InfraRed | HERE LIES CLUES |
| 15:42.08 | *** join/#asterisk n3u7 (n=neutrin0@CPE000d8802a707-CM0011e6c7edb1.cpe.net.cable.rogers.com) |
| 15:42.15 | *** join/#asterisk RoyK (n=roy@host-81-191-147-248.bluecom.no) |
| 15:42.41 | n3u7 | kernel preemption error is riuining my install |
| 15:42.42 | mutilator | ya supaigtr |
| 15:42.49 | n3u7 | this sucks |
| 15:42.50 | Tili | morale: check auth type in your sip.conf |
| 15:42.53 | mutilator | i've got to import emi records to a db |
| 15:42.53 | mutilator | then bill it |
| 15:42.53 | n3u7 | not documented |
| 15:43.19 | n3u7 | http://www.asteriskguru.com/board/viewtopic.php?p=688#688 |
| 15:43.25 | *** part/#asterisk Seyr (n=Seyr@rrcs-24-227-133-226.sw.biz.rr.com) |
| 15:43.25 | n3u7 | nyone help? |
| 15:43.46 | *** join/#asterisk tholo (n=tholo@dsl001-136-136.lax1.dsl.speakeasy.net) |
| 15:44.31 | jontow | curiousity.. say i have #include'd another file with contexts/extensions in it for the dialplan, can i have a [globals] 'context' in that file as well, to set variables in? |
| 15:44.44 | mutilator | the heck is preemting? |
| 15:45.05 | nexis | when a call comes in, the caller id number gets put into ${CALLERIDNUMBER} correct? |
| 15:45.23 | *** join/#asterisk ohad (n=ohad@19-231-13-72.cosmoweb.net) |
| 15:45.31 | n3u7 | mutilator:it has to do with kernel scheduling |
| 15:45.51 | bon | bjohnson: and so? ;) |
| 15:45.52 | *** join/#asterisk JerJer[mobile] (n=jj@dsl001-136-136.lax1.dsl.speakeasy.net) |
| 15:45.53 | n3u7 | mutilator:fact is i would rather be writing agi |
| 15:46.01 | ohad | hi, i have just updated my snom 360. i get an .htaccess user and pass prompt when i try to access the web interface. does anyone know the default user and pass for the snom 360 |
| 15:46.01 | ohad | ? |
| 15:46.20 | morale | tili, the auth type.. is "type = peer" thats right? |
| 15:46.28 | h3x0r | there is no default |
| 15:46.55 | h3x0r | but you can set it up in the phone's menu keypad |
| 15:47.10 | *** join/#asterisk FaithX (n=FaithX@202-6-145-116.ip.adam.com.au) |
| 15:47.14 | ohad | h3x0r, how? |
| 15:47.37 | ohad | got it |
| 15:47.51 | n3u7 | http://www.linuxquestions.org/questions/showthread.php?s=&threadid=373030 |
| 15:47.57 | n3u7 | oops |
| 15:48.05 | n3u7 | sorry wrong window |
| 15:48.12 | h3x0r | yeah its in Network |
| 15:48.13 | Tili | morale: that is fine. also check if you have host=dynamic or set to ip/netmask of your iphone |
| 15:48.24 | n3u7 | I'm trying to get around this preemption thing |
| 15:49.08 | morale | tili, http://voxilla.com/forum-viewtopic-t-4495-view-next.html - i followed that.. thats the extent of my asterisk knowledge. |
| 15:50.31 | ohad | h3x0r, what is the username? i tried 'admin' and 'administrator' with the new passwd i've setup |
| 15:50.38 | ohad | h3x0r, doesn't work |
| 15:53.42 | *** join/#asterisk terrapen (n=cjs@fw-01.satx.bikeworld.net) |
| 15:53.53 | *** join/#asterisk Cresl1n (n=matt@dsl001-136-136.lax1.dsl.speakeasy.net) |
| 15:55.42 | *** join/#asterisk tholo (n=tholo@dsl001-136-136.lax1.dsl.speakeasy.net) |
| 15:56.15 | ohad | h3x0r, ideas? default user and pass? |
| 15:56.16 | h3x0r | you set that in there too |
| 15:56.24 | ohad | h3x0r, ?? |
| 15:56.41 | h3x0r | Menu -> Network -> HTTP Admin and HTTP Password |
| 15:56.45 | *** part/#asterisk terrapen (n=cjs@fw-01.satx.bikeworld.net) |
| 15:57.02 | morale | how do i verify i am connecteD? |
| 15:57.10 | morale | connected, to my sip provider |
| 15:57.20 | h3x0r | by the way if you didnt do so already, you need to hit the settings button and switch to "Administrator Mode" |
| 15:57.48 | Tili | morale: sip show registry |
| 15:58.21 | *** join/#asterisk Cherebrum (i=jgarland@72.36.136.226) |
| 15:58.56 | *** join/#asterisk AsteriskNoob (i=BoredBoz@207-114-232-10.gen.twtelecom.net) |
| 15:59.04 | AsteriskNoob | wham! |
| 15:59.13 | *** join/#asterisk Guggemand (i=irc@tester2.har-tabt.dk) |
| 15:59.42 | Cherebrum | I have a customer that is bitching about call setup time on their Cisco 7960 sets. Anyone know how to make them shut up? |
| 15:59.54 | iDunno | call setup time? |
| 15:59.55 | AsteriskNoob | what firmware? |
| 15:59.57 | ohad | h3x0r, hitting menu - i don't have network |
| 15:59.59 | ohad | option |
| 16:00.38 | AsteriskNoob | i find that the biggest delay in call setup comes from my PRI... but then again there is a big problem with latency in the 7.5 firmware that I am currently working on with cisco |
| 16:00.42 | ohad | h3x0r, these phones suck |
| 16:00.42 | AsteriskNoob | it didnt exist in 7.5 |
| 16:00.44 | AsteriskNoob | it didnt exist in 7.4 |
| 16:01.42 | morale | hmm..now it is not even trying to login.. |
| 16:01.42 | h3x0r | no they dont suck |
| 16:01.44 | Cherebrum | AsteriskNoob: 7.4 |
| 16:01.48 | AsteriskNoob | hmm |
| 16:01.48 | h3x0r | you just didnt become the administraotr in the settings button |
| 16:02.02 | h3x0r | by the way another problem you may run into is if you just updated to 4.x sip firmware |
| 16:02.07 | h3x0r | you need to load a new license key or sip wont work anymore |
| 16:02.14 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-170.dslextreme.com) |
| 16:02.22 | AsteriskNoob | well cherebrum, there were lots of fixes and improvements in 7.5, however.... it seems to have 100ms more latency than 7.4 |
| 16:02.23 | h3x0r | you will get a big fat X in a box next to your line appearence on the lcd screen |
| 16:02.24 | *** join/#asterisk wunderkin (i=kev@12-219-162-233.client.mchsi.com) |
| 16:02.27 | AsteriskNoob | just so ya know |
| 16:02.49 | ohad | h3x0r, they echo and i'm using PRI. they do not work well with headsets.. the volume is all screwed up. i am trying to set these up at a call center and they are sucky. they get stuck as well. i just upgraded the fireware and they still suck |
| 16:03.05 | h3x0r | Uh |
| 16:03.09 | h3x0r | that echo problem isnt your phone |
| 16:03.15 | AsteriskNoob | ohad and h3x0r what phones are you talking about? |
| 16:03.19 | h3x0r | its the gain setting in zaptel on your asterisk box probably |
| 16:03.23 | h3x0r | Snom 360 |
| 16:03.26 | AsteriskNoob | ah |
| 16:03.37 | h3x0r | they are the best voip phones ive used and i have a ton of them |
| 16:03.48 | h3x0r | some setup things are not intuitive |
| 16:03.49 | AsteriskNoob | lol, i had one of those, i always go back to my cisco phones |
| 16:03.52 | h3x0r | but you only have to do that shit once, usually |
| 16:04.01 | ohad | AsteriskNoob, snom 360. i have just updated the fireware and i can't login. i get an .htaccess prompt |
| 16:04.02 | h3x0r | dude, cisco phones dont do shit with sip firmware |
| 16:04.12 | Cherebrum | h3x0r: Have you used the Polycom phones? |
| 16:04.15 | *** join/#asterisk dalabera (n=Dalabera@pmr.pmrtechnologies.com) |
| 16:04.17 | jake1932 | tey make and receive calls |
| 16:04.23 | ohad | AsteriskNoob, i can't set the user and the pass on the phone since i can't get to the network pref. |
| 16:04.24 | h3x0r | You can't legally buy a polycom without owning polycom software |
| 16:04.38 | h3x0r | like, to the point where most resellers wont give you one |
| 16:04.45 | AsteriskNoob | ohad, i havent done anything with that phone for a looong time |
| 16:04.46 | jake1932 | making and receiving calls - isn't that what a phone is for? |
| 16:04.57 | Dr_Ray | legally? |
| 16:05.01 | AsteriskNoob | my cisco phone does a lot h3x0r, i just had to do alot of custom work on them |
| 16:05.05 | ohad | h3x0r, how do i reset the phone without the web interface? |
| 16:05.16 | ohad | h3x0r, i want to be able to access the web interface again |
| 16:05.17 | h3x0r | by the way the next snom firmware will support XML browsing like the cisco |
| 16:05.34 | h3x0r | settings -> reset values |
| 16:05.54 | AsteriskNoob | i will admit, i like the cisco's better when they run sccp, and i actually do run sccp on the one in my office :) (from asterisk) |
| 16:06.08 | h3x0r | you cant drop a conference party on cisco sip stack |
| 16:06.20 | lehel | anyone using mysql 5? |
| 16:06.22 | h3x0r | it dosent haev web configuration at all |
| 16:06.27 | AsteriskNoob | sure you can h3x0r, as long as it's you HAHAHA |
| 16:06.34 | h3x0r | ha |
| 16:06.38 | h3x0r | its a piece of garbage dude |
| 16:06.39 | *** join/#asterisk gclark (n=gclark@cpe-069-132-101-069.carolina.res.rr.com) |
| 16:06.41 | AsteriskNoob | you can hang up and the remaining stay |
| 16:06.48 | h3x0r | even a sipura spa-841 is better |
| 16:06.54 | gclark | hello _ i have a question? |
| 16:06.57 | jake1932 | not my 841 |
| 16:07.00 | h3x0r | except for the small screen |
| 16:07.09 | jake1932 | and crappy buttons |
| 16:07.17 | morale | why is asterisk skipping my sip.conf file now and not loading it on startup? i turned debugging up to see if it may be a syntax error in the file but it is not reporting anything |
| 16:07.23 | AsteriskNoob | like i said i run sccp |
| 16:07.31 | gclark | Is there any sip phones or modules out there that will allow for user a to put someone on hold and user b pick up that call by hitting a button? |
| 16:07.41 | Lee__ | morale: are you /sure/ it's skipping it? |
| 16:07.45 | h3x0r | yeah its called a shared line appearance key or a park orbit |
| 16:07.53 | h3x0r | too bad asterisk dosent support it for shit |
| 16:08.05 | jake1932 | gclark: use asterisk parking |
| 16:08.09 | Lee__ | gclark: that's called call parking |
| 16:08.20 | morale | lee__ i think so, its not trying to connect to my sip provider anymore |
| 16:08.42 | gclark | I have offered that to our customer but they the old key system style where they hit a button then go to another phone and press that line key and have the call |
| 16:08.54 | AsteriskNoob | sip doesnt do shared line appearances without jumping through hoops of fire |
| 16:08.58 | Lee__ | morale: how do you know? and why "anymore"? what changed between then and now? |
| 16:09.04 | jake1932 | gclark: suppose multiple people park calls |
| 16:09.05 | jake1932 | ? |
| 16:09.08 | AsteriskNoob | its not asterisk, its sip |
| 16:09.18 | jake1932 | gclark: you have to know which one to pick up |
| 16:09.43 | Lee__ | gclark: you could define each button to a call parking extension and monitor it's state. the snom 360's have LEDs that can do that. |
| 16:09.55 | morale | lee__ i really dont know what changed, all i have been modifying is the sip.conf in /etc/asterisk |
| 16:10.54 | gclark | Lee__: thats a good idea. they currently use polycom phones and I see they have the new expansion module - has anyone used this yet or is this product to new? |
| 16:11.56 | *** join/#asterisk Seyr (n=Seyr@rrcs-24-227-133-226.sw.biz.rr.com) |
| 16:12.00 | Lee__ | morale: you just said you changed sip.conf. revert to the version that worked and see what the differences are. |
| 16:12.00 | *** part/#asterisk Seyr (n=Seyr@rrcs-24-227-133-226.sw.biz.rr.com) |
| 16:12.14 | *** join/#asterisk Seyr (n=Seyr@rrcs-24-227-133-226.sw.biz.rr.com) |
| 16:12.30 | marc324 | does odbcstorage work with postgresql? |
| 16:12.31 | Lee__ | that's what version control is for. rcs for starters. cvs for advanced. |
| 16:12.41 | h3x0r | ohad: did you do the administrator thing in the "Settings" button? You won't see those HTTP options in Menu without doing so |
| 16:12.47 | Seyr | Im trying to get ztdummy to work on kernel 2.6 and I am getting the following error: FATAL: Error inserting ztdummy (/lib/modules/2.6.9-11.ELsmp/kernel/extra/ztdummy.ko): Invalid module format |
| 16:12.49 | Seyr | anyone know? |
| 16:13.24 | h3x0r | Seyr: I would guess you compiled zaptel against the wrong kernel source tree |
| 16:13.26 | Lee__ | damn RHEL. ztdummy works fine here on Debian. sorry. |
| 16:13.39 | Lee__ | Seyr: what's uname -a output? |
| 16:13.39 | h3x0r | RHEL sucks, too |
| 16:13.40 | marc324 | the sql sent to postgres uses double quotes. instead of single. |
| 16:13.46 | iDunno | zaphfc works fine here on debian sarge :) |
| 16:14.00 | Seyr | uname shows "2.6.9-11.ELsmp" |
| 16:14.23 | iDunno | only took several hours, a lot of swearing and crying, and then a single moment of realisation and 2 changes to 2 lines for it to work ;) |
| 16:14.31 | Lee__ | that's weird. must be a kernel source thing. does /usr/src/linux exist? |
| 16:14.32 | n3u7 | http://www.asteriskguru.com/board/viewtopic.php?p=688#688 |
| 16:14.37 | h3x0r | \\ |
| 16:14.38 | n3u7 | :( |
| 16:14.40 | h3x0r | crap |
| 16:14.46 | morale | if i load chan_sip.so by hand it attempts to connect again, what file contains a list of the modules loaded on startup? |
| 16:15.37 | Lee__ | <PROTECTED> |
| 16:15.55 | Seyr | Lee: yeh, /usr/src/linux-2.6 is a symlink to kernel source |
| 16:16.18 | morale | ahh.. that is why its not loading it :) i must have typed something in the cli interface to update the configs |
| 16:16.32 | Lee__ | Seyr: rename that file to /usr/src/linux |
| 16:16.46 | Lee__ | you probably compiled it against 2.4 sources |
| 16:17.12 | *** join/#asterisk damned (n=Victor@damned.vpol.org.ru) |
| 16:17.21 | morale | there now it is failing to login again, its better than nothing though |
| 16:17.40 | ohad | h3x0r, how do i upload a custom melody? |
| 16:17.45 | Seyr | Lee: no, only have ever had 2.6 on here |
| 16:18.12 | *** join/#asterisk fugitivo (n=ajf@209.13.243.83) |
| 16:18.13 | fugitivo | hello |
| 16:18.22 | Lee__ | rename file, try again. other than that I have no idea. what's 'modinfo zaptel' say? |
| 16:20.11 | *** join/#asterisk wolfson (n=hehe@65.174.122.198) |
| 16:20.21 | h3x0r | i havent done that but im sure its well documented |
| 16:20.36 | Seyr | lee: sent you a PM with the info |
| 16:21.47 | n3u7 | my modinfo zaptel reports vermagic 2.6.11.10 preempt 386 gcc-3.3 |
| 16:22.43 | n3u7 | but modprobe zaptel reports unknown symbol which dmesg informs is sub an add _prempt_count |
| 16:22.44 | *** join/#asterisk SpaceBass (n=sp@c-24-125-184-203.hsd1.va.comcast.net) |
| 16:22.46 | SpaceBass | hey folks |
| 16:22.53 | SpaceBass | is there an easy way to search for DIDs? |
| 16:22.54 | n3u7 | ? |
| 16:23.19 | morale | http://deadbolt.ca/~russell/screen.txt - that is the log file i got from asterisk, it just keeps saying "Failed to authenticate on REGISTER" |
| 16:23.22 | Lee__ | Seyr: weird. sounds like your kernel is fuxored. I haven't worked with anything other then the kernel in Debian stable so I can't be of much more help. |
| 16:23.35 | SpaceBass | also, I have a probem with zap channels that appear to jump context |
| 16:23.47 | Lee__ | morale: sip debug peer <trunkname> |
| 16:24.24 | kippi | can asterisk cope wioth pushing e-mails? |
| 16:24.39 | n3u7 | i can't believe *I am having a problem that is undocumented |
| 16:25.01 | n3u7 | and apparently unsolvable |
| 16:25.11 | n3u7 | LIEF!!!!!!!!!!!! |
| 16:25.23 | n3u7 | :( |
| 16:25.50 | morale | lee__ same thing.. |
| 16:26.15 | Lee__ | that didn't give you any more interesting information? |
| 16:26.20 | Seyr | Anyone know why my MWI would not work on my 7960 with Realtime even though rtcachefriends=yes? |
| 16:26.34 | SpaceBass | i have zap channels that appear to jump contexts... anyone ever seen that? |
| 16:26.42 | morale | nope.. i typed "sip debug peer inphonex" and it said SIP Debugging Enabled for IP: 208.239.76.157:5060 |
| 16:28.29 | morale | there that fixed it |
| 16:29.10 | SpaceBass | this is my zapata-auto.conf |
| 16:29.12 | SpaceBass | http://pastebin.ca/25489 |
| 16:29.39 | SpaceBass | and zapata.conf |
| 16:29.40 | SpaceBass | http://pastebin.ca/254903 |
| 16:32.36 | *** join/#asterisk hellagony (n=egutierr@irc.americatelnet.com.pe) |
| 16:32.48 | InfraRed | what's the linux of choice these days for asterisk |
| 16:33.27 | denon | InfraRed: that's a pretty loaded question.. |
| 16:33.37 | denon | but I use it and like it on debian |
| 16:34.15 | n3u7 | InfraRed:good question :I'mgetting nowhere with ubutu |
| 16:34.21 | n3u7 | *ubuntu |
| 16:34.29 | InfraRed | i am pondering centos |
| 16:34.32 | InfraRed | or debian |
| 16:34.36 | n3u7 | http://www.asteriskguru.com/board/viewtopic.php?p=688#688 |
| 16:34.49 | *** join/#asterisk myiagy (n=myiagy@200.138.215.78) |
| 16:35.04 | Lee__ | I hear OS X uses Linux </sarcasm> |
| 16:35.05 | tzafrir_laptop | n3u7, which version? The ubuntu Asterisk packages are generally repackaged Debian packages. Sometimes they're a bit out of date |
| 16:35.11 | *** part/#asterisk Seyr (n=Seyr@rrcs-24-227-133-226.sw.biz.rr.com) |
| 16:35.13 | *** join/#asterisk Cresl1n (n=matt@dsl001-136-136.lax1.dsl.speakeasy.net) |
| 16:35.43 | Lee__ | InfraRed: "these days" you should use Debian. |
| 16:35.58 | InfraRed | "these days" ? |
| 16:36.02 | SpaceBass | i was playing with the latest ubuntu the other day... it was ok |
| 16:36.05 | *** join/#asterisk Weezey (n=chatzill@lo31.loit.ca) |
| 16:36.36 | syzygyBSD | has anyone had a problem with the command "rmmod wcte11xp" crashing their box? |
| 16:36.40 | SpaceBass | when using DISA is there a way to get back to dialtone when you are done wiht a call? |
| 16:36.42 | syzygyBSD | all the other modules unload fine |
| 16:36.43 | Weezey | i'm having a wierd issue where my box isn't answering when the call is going to voicemail, the line just rings and then hangs up. |
| 16:36.45 | Lee__ | speaking of, I remember reading about a preconfigured debian/ directory for building a .deb out of Asterisk CVS. Has someone published this? |
| 16:37.11 | tzafrir_laptop | syzygyBSD, I saw this mentioned on asterisk-users by two users. Are you one of them? |
| 16:37.26 | syzygyBSD | No, I will take a look, thanks |
| 16:37.31 | tzafrir_laptop | syzygyBSD, is it a bristuffed version? which version? |
| 16:37.44 | tzafrir_laptop | syzygyBSD, I didn't see any solution posted |
| 16:38.33 | n3u7 | tzafrir_laptop:i have installed from the cvs |
| 16:39.08 | *** join/#asterisk greekman (n=alex@host254.209.113.199.conversent.net) |
| 16:39.09 | tzafrir_laptop | Lee__, if you want debs for 1.2beta1: deb http://rapid.dotsrc.org/ experimental/ |
| 16:39.14 | syzygyBSD | I dont' know which version let me check |
| 16:39.48 | tzafrir_laptop | Lee__, I figure you should be able to use those debs (the sources are included there) to build for HEAD. |
| 16:39.59 | Lee__ | thanks tzafrir. I'm scared of repositories labled "experimental" :) |
| 16:40.02 | tzafrir_laptop | Though I know at least one of the patches there will not apply |
| 16:40.19 | Lee__ | but you post a lot to the voip-pkg list so I'll trust you :) |
| 16:40.30 | taec_ | Hi, 'zap show channels' so no channels, even though 'zap show status' seems to show one active line. Where should I start? |
| 16:41.08 | tzafrir_laptop | Lee__, well, it is. But that's also because those are not ready yet to replace the packages in debian Unstable |
| 16:41.49 | tzafrir_laptop | Lee__, those packages are basically what you'll get with a svn-buildpackage from the pkg-voip svn |
| 16:41.52 | *** join/#asterisk g__ (n=g@itd01fw-fibre.itdepartment.com) |
| 16:42.14 | n3u7 | tzafrir_laptop:http://www.asteriskguru.com/board/viewtopic.php?p=688#688 |
| 16:42.22 | Lee__ | excellent. I didn't you can do that with SVN. |
| 16:42.41 | *** join/#asterisk steve___ (n=steve@store-fw.porchlight.ca) |
| 16:42.54 | *** join/#asterisk shido6 (n=curtis@d221-68-210.commercial.cgocable.net) |
| 16:42.58 | Lee__ | I'm just compiling Asterisk from source each time against Sarge. |
| 16:44.12 | tzafrir_laptop | n3u7, are you sure you built the package vs. the right kernel headers? |
| 16:44.14 | g__ | My users are reporting their phone calls are getting dropped after 45 seconds. The console is displaying "Nobody picked up in 45000 ms" (even though the third party answered).. any idea why this might be happening? (Is it the zap module that decides when a phone call is "answered"?) |
| 16:44.27 | *** join/#asterisk garyr (n=garyr@cpc1-rdng4-3-0-cust102.winn.cable.ntl.com) |
| 16:44.29 | tzafrir_laptop | Did you use m-a ? |
| 16:44.32 | *** join/#asterisk corne (n=corne@ndn-165-141-208.telkomadsl.co.za) |
| 16:44.49 | syzygyBSD | tzafrir_laptop: I am using the version compiled from the zaptel source v 1.0.4 |
| 16:45.12 | g__ | tzafrir: is that to me? |
| 16:45.31 | n3u7 | tzafrir_laptop:no I'm not sure, how to I specify? |
| 16:46.52 | Lee__ | g__: NAT problems |
| 16:46.58 | Lee__ | what do your SIP headers say? |
| 16:47.31 | garyr | Anyone have time to help me on a NO Audio issue with Asterisk under FC4? Tried the FAQ's and docs and still lost. |
| 16:47.44 | jcollins | BrianR___: that senddtmf worked perfectly, thanks again. |
| 16:47.44 | Lee__ | garyr: NAT problems. read the wiki. |
| 16:47.51 | g__ | NAT, like network address translation? This is on a zap line.. but hang on, I'll check for sip stuff. |
| 16:48.22 | Lee__ | are your phones and Asterisk on the same subnet? |
| 16:48.35 | garyr | Lee_: tried that no nat, I can call each phone answer but no sound yep same subnet |
| 16:48.49 | g__ | No, but they are directly routable in both directions. |
| 16:50.36 | tzafrir_laptop | syzygyBSD, hmmm... Debian did not have bristuff in at that version, I believe. anyway, isn't 1.0.4 buggy (and was fixed in 1.0.5?) |
| 16:50.50 | tzafrir_laptop | sorry: it was with zaptel |
| 16:50.51 | syzygyBSD | I just rebuilt the drivers and get the warning "dereferencing type-punned pointer will break strict-aliasing rules" several times, is that a problem? |
| 16:50.54 | garyr | Lee__: I even have it so I have it connected to Nikotel and recieve calls ( but again no sound ) no errors full debug mode show s no errors |
| 16:50.54 | *** join/#asterisk tholo (n=tholo@dsl001-136-136.lax1.dsl.speakeasy.net) |
| 16:51.08 | Lee__ | Sorry, I can't help you anymore g__ and garyr. not sure what's happening. You'll have to monitor the SIP headers of your calls. |
| 16:51.28 | tzafrir_laptop | syzygyBSD, you can ignore those, AFAICT |
| 16:51.45 | garyr | Lee_: thanks for your time anyway |
| 16:51.52 | syzygyBSD | ya, it worked fine for the last 10 months, didn't think those would be a problem |
| 16:52.57 | tzafrir_laptop | n3u7, how did you build the modules? vs. what kernel headers? module-ass-stant (m-a) is a debian tool for building kernel module packages |
| 16:57.59 | n3u7 | tzafrir_laptop:this sucks,I finally get help and I haveto go to work |
| 16:58.14 | g__ | Thanks Lee__.. I'm still looking into your theory, because I now see some dropped packets on the firewall (one at the begining of outgoing phone calls). |
| 16:58.25 | n3u7 | tzafrir_laptop:I will reinstall from HEAD an update the kernel |
| 16:58.41 | n3u7 | I will have asterik! |
| 16:58.45 | n3u7 | thanks all |
| 16:58.47 | tzafrir_laptop | FWIW, I gtg too |
| 16:58.48 | n3u7 | lates |
| 16:59.01 | g__ | I'll dig around on the 'net as well.. perhaps it has something to do with the recent upgrade to 1.0.9 |
| 16:59.24 | Lee__ | g__: you need two way communication on ports 5060 (SIP) and 10000-20000 (RTP) |
| 17:01.28 | g__ | Lee__: for some reason it hasn't been an issue before. Also, it seems to only affect outgoing calls. |
| 17:02.05 | lehel | sorry anyone google earth?;) |
| 17:02.56 | *** join/#asterisk Trionnis (i=fsn@12-215-249-177.client.mchsi.com) |
| 17:04.01 | *** join/#asterisk kimosabe (n=kimo@dsl-200-67-12-220.prod-empresarial.com.mx) |
| 17:05.09 | kimosabe | is there agood how to on how to configure a a conection from my server to another server and for me to be able to take the line via one of my sipuras with dial 9 |
| 17:07.08 | malverian[work] | Can anyone think of why ChanSpy would sound really bad? Sounds like packets are being dropped like mad or something. |
| 17:08.57 | taec_ | Extension '3600' in context 'default' from '<blah>' does not exist ... it seems my extensions are coming up as 4 digit extensions instead of 3 digit ones, is there a way to correct that? |
| 17:11.25 | kimosabe | i have 10 sipuras configured on my asterisk box using them for voice overip between offices |
| 17:12.06 | Trionnis | kimosabe: what you're looking for is called "trunking" |
| 17:12.15 | Trionnis | use IAX |
| 17:12.27 | Trionnis | ~trunking |
| 17:12.45 | Trionnis | hmm, maybe I forgot the command |
| 17:12.51 | Trionnis | ~trionnis |
| 17:12.52 | jbot | somebody said trionnis was a moron who should read the *whole* config file to fix his problems |
| 17:12.55 | Trionnis | haha |
| 17:12.56 | Trionnis | nope |
| 17:14.05 | Trionnis | jbot, trunking is http://www.voip-info.org/wiki/index.php?page=Asterisk+-+dual+servers |
| 17:14.06 | jbot | Trionnis: okay |
| 17:15.01 | Trionnis | jbot, trunking is connecting 2 asterisk servers together with a shared dialplan. See: http://www.voip-info.org/wiki/index.php?page=Asterisk+-+dual+servers |
| 17:15.02 | jbot | ...but trunking is already something else... |
| 17:15.06 | Trionnis | gr |
| 17:15.15 | Trionnis | jbot, forget trunking |
| 17:15.15 | jbot | Trionnis: i forgot trunking |
| 17:15.22 | Trionnis | jbot, trunking is connecting 2 asterisk servers together with a shared dialplan. See: http://www.voip-info.org/wiki/index.php?page=Asterisk+-+dual+servers |
| 17:15.24 | jbot | Trionnis: okay |
| 17:15.29 | Trionnis | ~trunking |
| 17:15.30 | jbot | well, trunking is connecting 2 asterisk servers together with a shared dialplan. See: http://www.voip-info.org/wiki/index.php?page=Asterisk+-+dual+servers |
| 17:15.35 | pauldy | is there not a howto somewhere that describes how to create a link between two asterisk pbxs using iax2 |
| 17:15.41 | file[laptop] | hrm? that's not exactly trunking |
| 17:15.47 | Trionnis | it's close enough |
| 17:15.49 | Trionnis | ;) |
| 17:16.08 | file[laptop] | that's dialplan sharing :P |
| 17:16.30 | Trionnis | ok |
| 17:16.32 | Trionnis | :P |
| 17:16.38 | Trionnis | jbot, forget trunking |
| 17:16.38 | jbot | i forgot trunking, Trionnis |
| 17:16.48 | kimosabe | i purchased a voice over ip account or a pstn acount from a provider and i have an asterisk box but i want my asterisk users to be able to take that provider line i just purchased how do i do it ? |
| 17:17.13 | kimosabe | i alreday have it registered via my asterisk box i do sip show registry and it shows registered |
| 17:17.25 | kimosabe | but i want to take the line by dialing 9 |
| 17:17.27 | file[laptop] | registration is not for outbound calls |
| 17:17.42 | file[laptop] | it's a method of saying to the other side, "hey - if you need to send me an inbound call, I can be reached here!!!" |
| 17:18.30 | kimosabe | oki |
| 17:18.41 | kimosabe | filelaptop so what do i need 2 do ? |
| 17:18.57 | file[laptop] | set up a peer entry to the other side with secret, host, fromuser, and username |
| 17:19.09 | file[laptop] | then send calls to it. |
| 17:19.12 | *** part/#asterisk garyr (n=garyr@cpc1-rdng4-3-0-cust102.winn.cable.ntl.com) |
| 17:19.48 | Trionnis | so does anyone at nufone.net ever answer the sales email address? |
| 17:19.50 | kimosabe | and that goes in sip.con or extensions.conf |
| 17:19.58 | file[laptop] | sip.conf |
| 17:20.10 | Trionnis | I've been trying to find out about porting a tollfree did for a month now, and can't get a reply |
| 17:20.43 | kimosabe | file lap top then i also need an entry in extensions.conf ? |
| 17:20.50 | file[laptop] | yes |
| 17:20.59 | outtolunc | yet you wait till astricon is happening to come in here and ask about it <G> |
| 17:21.03 | kimosabe | filelaptop do you have a how to on this ? |
| 17:21.09 | file[laptop] | kimosabe: no, Google. |
| 17:21.18 | *** join/#asterisk sahafeez (n=sahafeez@67.109.14.227.ptr.us.xo.net) |
| 17:21.31 | kimosabe | cool what is this called so i can gogle it ? |
| 17:21.35 | Trionnis | outtolunc: you're talking to me? |
| 17:21.43 | outtolunc | Trionnis: yes i was |
| 17:21.45 | Trionnis | ah |
| 17:21.53 | Trionnis | I've actually asked about it in here quite a few times |
| 17:21.54 | Trionnis | :) |
| 17:22.01 | Trionnis | just never have gotten an answer |
| 17:24.07 | syzygyBSD | what is the variable for the incomming number? |
| 17:25.53 | *** join/#asterisk tholo (n=tholo@dsl001-136-136.lax1.dsl.speakeasy.net) |
| 17:26.38 | ohad | hi, i am using * with T1 pri lines. for some reason every once in a while i hear myself echo on my phone and the other person echo. ideas why and how to resolve this problem? |
| 17:27.32 | *** join/#asterisk svenna (n=svenna@p548D2019.dip0.t-ipconnect.de) |
| 17:27.48 | Lee__ | ohad: buy an echo canceler or mess with the zaptel driver's software one. |
| 17:28.06 | Lee__ | the PSTN echos, you can't get around that. |
| 17:28.10 | kimosabe | filelaptop i have an entry in sip.conf it says register=> and m info |
| 17:28.47 | file[laptop] | I already said |
| 17:28.54 | file[laptop] | registration has nothing to do with outbound calls |
| 17:29.02 | ohad | Lee__, what is an echo canceler? |
| 17:29.10 | ohad | Lee__, where do i add it? |
| 17:29.12 | Lee__ | I don't have any registrations and I make outbound calls all the time :) |
| 17:29.20 | file[laptop] | you need to setup a peer entry in sip.conf, with your host to send calls to plus your authentication information, and then dialplan logic to use that peer entry |
| 17:29.40 | Lee__ | ohad: please RTFM. you should figure out what an echo canceler is before you bought a PRI. |
| 17:29.58 | Lee__ | there's copious documentation on the wiki about it. |
| 17:30.10 | Lee__ | and in books on VoIP. |
| 17:31.16 | Lee__ | file[laptop]: or just use IP filtering on both ends to allow what IPs are allowed to send SIP traffic in and out of your network. |
| 17:31.40 | file[laptop] | Lee__: he's using some provider |
| 17:31.50 | *** join/#asterisk wh|tey (n=wh|tey@c-24-8-158-104.hsd1.co.comcast.net) |
| 17:32.05 | *** part/#asterisk Trionnis (i=fsn@12-215-249-177.client.mchsi.com) |
| 17:32.24 | wh|tey | HI all, does anyone here happen to know how to swap the codec that is used w/in AMP on asterisk at home |
| 17:32.52 | Lee__ | wh|tey: disallow=all, allow=<codec> |
| 17:33.01 | wh|tey | right now I am using G711 ulaw codec but want to use g726 codec to save about 1/2 the bandwidt adn to be abkle to make more simultaneous calls |
| 17:33.13 | Lee__ | but you should go to #amportal for that question |
| 17:33.26 | devel | afternoon, all. i'm trying to set the caller id on a call to the extension of the person transferring, can that be done? |
| 17:33.26 | wh|tey | ohh sorry |
| 17:33.42 | Lee__ | devel: yes |
| 17:33.54 | devel | excellent. |
| 17:35.26 | *** join/#asterisk eKo1 (n=bernd@metrored-gw.tropicohn.com) |
| 17:36.02 | devel | ah! can i just "set callerid ${EXTEN}" when i answer the call before i transfer it? |
| 17:36.44 | *** part/#asterisk nexis (n=nexis@12-219-60-252.client.mchsi.com) |
| 17:38.14 | wh|tey | Lee__ , sorry to bug ya but is that statment on one of the config files w/in AMP, like per say under maintanence-> config edit -> then one of the files in there? |
| 17:38.52 | Lee__ | wh|tey: /join #amportal |
| 17:39.02 | wh|tey | k |
| 17:39.44 | *** join/#asterisk JerJer[mobile] (n=jj@dsl001-136-136.lax1.dsl.speakeasy.net) |
| 17:39.44 | desktophero | has anyone purchased and used the G729 codec from digium? |
| 17:39.53 | *** join/#asterisk bkw_ (n=bkw_@adsl-69-154-1-104.dsl.tulsok.swbell.net) |
| 17:40.03 | *** part/#asterisk bkw_ (n=bkw_@adsl-69-154-1-104.dsl.tulsok.swbell.net) |
| 17:40.04 | devel | yes, desktophero |
| 17:40.17 | Lee__ | desktophero: I would hope so, or they'd stop selling it! |
| 17:40.57 | desktophero | sorry for the phrasing of the question, Lee |
| 17:41.14 | *** join/#asterisk Cresl1n (n=matt@dsl001-136-136.lax1.dsl.speakeasy.net) |
| 17:41.20 | desktophero | i mean is there anyone on this list that has purchased and implemented the codec ;) |
| 17:41.37 | desktophero | that could answer a couple of questions for me? |
| 17:41.42 | devel | yes, desktophero |
| 17:42.10 | desktophero | devel - i currently have 2 * servers communicating through an IAX2 trunk |
| 17:42.26 | desktophero | devel - on the far end, I have hard phones that will support the g729 codec |
| 17:42.31 | desktophero | the trunk is setup for gsm now |
| 17:42.42 | desktophero | but I am wondering if i use g729 through the trunk |
| 17:42.45 | desktophero | will that work? |
| 17:43.04 | eKo1 | try it and see |
| 17:43.07 | desktophero | and if so, do i just need to buy enough licensing for the highest number of possible calls over the trunk? |
| 17:43.21 | desktophero | eko1 - my question is do I need 2x licenses per call |
| 17:43.34 | desktophero | one for each server since the channel exists within that context? |
| 17:43.35 | kimosabe | filelaptop u there |
| 17:43.43 | desktophero | or 4 for each leg of the call? |
| 17:43.51 | eKo1 | i think the floks at digium will answer your question desktophero |
| 17:44.00 | desktophero | Phone1 ->> Ast_1 <<->> Ast_2 <<->> Phone2 |
| 17:44.01 | bjohnson | and I think the answer is 4 |
| 17:44.06 | devel | desktophero, it should work fine. if the server stays in the call (i.e. transcoding), it would need two licenses on each box for each call. |
| 17:44.23 | devel | so, what bjohnson said |
| 17:44.25 | ohad | how to i cancel each on zaptel? lots of my users are experincing echo esp on the snom 360 phones. |
| 17:44.34 | ohad | i have read - http://www.voip-info.org/wiki/view/Causes+of+Echo |
| 17:45.08 | bjohnson | although you might be able to do it with 0 if you let the phones reconnect between themselves and NAT doesn't cause problems |
| 17:45.23 | desktophero | my guess is that I could potentially go from G729 across the trunk to GSM on each far end (at the phone), but then I lose some of the intent of going with g729 in the first place |
| 17:45.58 | desktophero | bjohnson - 0 licensing across the trunk? |
| 17:45.59 | *** join/#asterisk mutilator (n=animenod@65.111.201.79) |
| 17:46.15 | bjohnson | generally, on a LAN just use ulaw unless you have a big reason not to |
| 17:46.31 | malverian[work] | Our current phone system has a "Voicemail Transfer" function. It works identically to normal transfer except that it goes directly to the voicemail of the person you transfer to. Can anyone think of a way to do this with Asterisk? |
| 17:46.39 | bjohnson | yes |
| 17:46.51 | bjohnson | make extens for it |
| 17:47.26 | desktophero | bjohnson - bandwidth is the main concern. we have a min of 764 K with a max of 1.5 M. There are only about 15 people in the office so we are talking ~5 concurrent calls |
| 17:47.34 | bjohnson | eg 2001 rings and 3001 goes stright to vm for 2001 |
| 17:47.49 | malverian[work] | bjohnson, That's too complicated for our users :-P |
| 17:47.52 | gordonjcp | stupid question here |
| 17:47.54 | desktophero | we just end up with a choppy sound - and unfortunately jitterbuffer in * 1.0.9 does not work too well |
| 17:48.00 | Xen^ | hello |
| 17:48.10 | Xen^ | can some one tell me how can i do live streaming with * |
| 17:48.15 | malverian[work] | I need to have them press a button and it say "Type the extension of the voicemail to transfer to" and then transfer the person they were on the phone with to that voicemail. |
| 17:48.17 | *** part/#asterisk mcf3782 (n=mcf3782@adsl-065-012-184-148.sip.asm.bellsouth.net) |
| 17:48.19 | gordonjcp | if I'm using GSM, why does my bandwidth graph go up to 86kbps? |
| 17:48.25 | bjohnson | desktophero: you have * at each end right? so you have more bw between the phones and * |
| 17:49.00 | bjohnson | Xen^: mplayer and slimserver |
| 17:49.07 | eKo1 | these old iaxys are a piece of shit, ugh |
| 17:49.25 | desktophero | bjohnson - correct, the bandwidth is between the two * servers |
| 17:49.26 | Xen^ | i did installed slimserver |
| 17:49.34 | Xen^ | but what to do in musiconhold.conf |
| 17:49.35 | desktophero | we have a 100/full at each LAN |
| 17:49.48 | bjohnson | desktophero: so you can use ulaw between the phones and * and gsm between the 8 servers |
| 17:49.54 | bjohnson | * servers |
| 17:50.03 | bjohnson | Xen^: read the wiki |
| 17:50.16 | malverian[work] | Xen^, http://www.voip-info.org/wiki-Using+Slimserver+for+Music+on+Hold |
| 17:50.21 | Xen^ | bjohnson : i did :0 |
| 17:50.40 | Xen^ | i am but it said add a line slim => in musiconhold.conf |
| 17:50.44 | Xen^ | but where i did not :) |
| 17:51.17 | desktophero | that is what I am going now...but the sound is sometimes worse than a cell call with the choppiness and "jitter". G.729 seems like the next best thing. I don't think I could handle ilbc transcoding (eh, Mr. Roboto?) |
| 17:53.34 | *** join/#asterisk ACiDV (n=acidvici@26-121.dr.cgocable.ca) |
| 17:55.14 | *** join/#asterisk zotz (n=zotz@24.231.36.100) |
| 17:58.52 | kimosabe | when i do the server defenition for out going calls where doies this go in ip .conf ? |
| 17:59.02 | kimosabe | sip.conf or extensions.conf ? |
| 18:02.45 | *** join/#asterisk Cresl1n (n=matt@dsl001-136-136.lax1.dsl.speakeasy.net) |
| 18:03.08 | kimosabe | look out room im using this how to http://www.voip-info.org/wiki-Asterisk+config+sip.conf |
| 18:03.35 | kimosabe | so that via my sipuras we can use a ip providers pstn but it doent work can i get a hand ? |
| 18:06.29 | *** join/#asterisk CrazyYoss (n=nobody@adsl-69-236-44-222.dsl.pltn13.pacbell.net) |
| 18:06.36 | *** join/#asterisk jeffgus (n=jeffgus@2002:d856:c704:0:0:0:0:1) |
| 18:08.21 | CrazyYoss | <-- Noob, When you sign up for a VOIP carrier with a DID (I think thats right) does the carrier take the signal from the caller compress it then send it to the pbx box? Or is it uncompressed? |
| 18:08.38 | jdv79 | can asterisk do LCR via pg? |
| 18:08.54 | Ahrimanes | jdv79: postgres? |
| 18:08.59 | jdv79 | yes, sorry |
| 18:09.04 | h3x0r | international ? |
| 18:09.09 | Ahrimanes | jdv79: i'm doing that with astcc |
| 18:09.19 | jdv79 | i'm not familiar |
| 18:09.21 | hypa7ia | CrazyYoss: you can often tell the carrier what codec to send... that's where compression is determined |
| 18:09.22 | jdv79 | i'll look it up |
| 18:09.33 | h3x0r | jdv79: what type of calls are you LCRing |
| 18:09.50 | jdv79 | international and local |
| 18:09.57 | jdv79 | could be any type i guess |
| 18:09.58 | h3x0r | oh ok |
| 18:10.05 | h3x0r | well us domestic is tricky |
| 18:10.11 | jdv79 | it is? |
| 18:10.15 | jdv79 | its just by lata isn't it? |
| 18:10.16 | h3x0r | yeah if you are doing wholesale |
| 18:10.22 | jdv79 | uh huh |
| 18:10.27 | h3x0r | LATA by Class, and Class includes an OCN list |
| 18:10.30 | CrazyYoss | hypa7ia: thank you |
| 18:10.31 | h3x0r | and numbers can be ported |
| 18:10.36 | jdv79 | hmm |
| 18:10.42 | h3x0r | you cant simply LCR by NPA/NXX |
| 18:10.48 | h3x0r | 35% of numbers in the US are ported to another carrier |
| 18:10.51 | jdv79 | how do you do it? |
| 18:11.00 | h3x0r | i do ss#7 lidb queries |
| 18:11.13 | h3x0r | i am working on a way to do it from asterisk and ser more directly |
| 18:11.31 | jdv79 | i don't really know what i'm talking about but the guy i'm working with is about to buy a mera box or something to do it |
| 18:11.42 | jdv79 | and i told him i think we can handle that with the * boxes |
| 18:12.01 | jdv79 | 60K for a box that does nothing but routing |
| 18:12.05 | jdv79 | BS |
| 18:12.15 | h3x0r | well |
| 18:12.21 | h3x0r | you cant do it in asterisk, yet |
| 18:13.02 | h3x0r | take this scenario in mind |
| 18:13.13 | h3x0r | lets say you call a US West (Qwest) number, and somebody ported it to XO |
| 18:13.16 | *** join/#asterisk Gunnar (n=gunnar@34.80-203-209.nextgentel.com) |
| 18:13.27 | h3x0r | you might send it to the wrong carrier and pay .03/Min when you think its .009 a minute |
| 18:13.47 | h3x0r | whereas if you knew the real LEC on the other end you could have sent it somewhere else for like .012/Minute |
| 18:14.16 | jdv79 | where's the DB for that? |
| 18:14.24 | h3x0r | furthermore, most of the SS#7 database providers (tehre are only 14 of them in the US) |
| 18:14.24 | jdv79 | its not the LERG is it? |
| 18:14.28 | h3x0r | do not have a way to access it over IP |
| 18:14.28 | h3x0r | no |
| 18:14.32 | jdv79 | i forget all this crap |
| 18:14.33 | ohad | how to i cancel each on zaptel? lots of my users are experincing echo esp on the snom 360 phones. |
| 18:14.37 | h3x0r | the biggest ones are Verisign and um |
| 18:14.58 | jdv79 | so its ss7 then? |
| 18:15.10 | h3x0r | well, its a ss7 TCAP database |
| 18:15.16 | h3x0r | a box to do that for 60 grand is cheap |
| 18:15.28 | h3x0r | im using a provider that lets me run it through their gateway over SSL |
| 18:15.28 | jdv79 | we're demoing it |
| 18:15.48 | *** join/#asterisk Assid (n=assid@203.115.64.62) |
| 18:16.16 | Assid | umm.. whats the app to 'show' to see jitter etc.. for packet loss |
| 18:17.18 | h3x0r | most of them dont wanna talk to you unless you are doing millions of queries a month |
| 18:17.29 | h3x0r | we did a deal with TNS for 5M transactions a month minimum |
| 18:17.29 | ohad | h3x0r, hi, do you know how to cancel the echo on zaptel/asterisk? |
| 18:17.33 | jdv79 | that'll be me |
| 18:18.04 | h3x0r | but you have to decide if you want to pay the .0075 a query |
| 18:18.08 | h3x0r | or just pay for misrouted calls |
| 18:18.12 | jdv79 | wow |
| 18:18.18 | h3x0r | some lecs you have to pay more |
| 18:18.20 | *** join/#asterisk meriad (i=mreith@snoid.gfed.net) |
| 18:18.23 | h3x0r | Qwest is .018 |
| 18:18.37 | jdv79 | so asterisk doesn't support ss7 directly yet? |
| 18:18.38 | h3x0r | sprint, snet, united, some otehrs are as high as .07 per query |
| 18:18.53 | h3x0r | No, and it probably never will support TCAP internally, it dosent make any sense |
| 18:18.54 | ACiDV | Someone have working example of "limit concurrent call (in/out)" with GROUP() and GROUP_COUNT() ? |
| 18:18.56 | h3x0r | thats a database server activity |
| 18:19.05 | h3x0r | what we are doing is actually implementing a postgres trigger |
| 18:19.09 | h3x0r | so that when you do a SELECT statement |
| 18:19.17 | h3x0r | it queries and caches the result for so many days |
| 18:19.35 | h3x0r | this way you can piggyback a dns server on it that supports pgsql, and do enum through it |
| 18:19.41 | h3x0r | or you can do web queries through it or whatever |
| 18:19.59 | jdv79 | what's wrong with doing it direct from ss7 again? |
| 18:20.00 | h3x0r | we're going to give away our transactions at cost to help build up the cached data and give ourselves a bigger volume discount |
| 18:20.07 | *** part/#asterisk meriad (i=mreith@snoid.gfed.net) |
| 18:20.10 | h3x0r | expensive shit to make it work |
| 18:20.10 | jdv79 | i'm trying to get my head around all this legacy shit |
| 18:20.14 | h3x0r | you might be able to use ss7box to do it |
| 18:20.20 | h3x0r | which uses a sangoma card and ss7 stack |
| 18:20.31 | h3x0r | it isnt legacy dude |
| 18:20.33 | ACiDV | Hmmm I know how to set Set(GROUP()=${EXTEN}) but GROUP_COUNT is always = 0 and so the GotoIf GROUP_COUNT() > 1 doesnt work |
| 18:20.37 | h3x0r | the stuff came out in like 1992 or something |
| 18:20.37 | h3x0r | heh |
| 18:20.47 | kimosabe | if im trying to add a sip on my box as a client and all my sipuras are in context home how can i handle the call |
| 18:20.49 | jdv79 | so your setup doesn't work yet? |
| 18:20.55 | jdv79 | :) |
| 18:21.11 | h3x0r | it works, we just are doing a lot of testing before production |
| 18:21.18 | h3x0r | and we need to add some features before reselling it |
| 18:21.22 | h3x0r | like knowing how much each query costs |
| 18:21.25 | h3x0r | they are working on that on their end |
| 18:21.35 | jdv79 | we might be interested in such a service |
| 18:21.45 | h3x0r | their system is super retarded |
| 18:21.47 | h3x0r | its all java crap |
| 18:22.06 | jdv79 | what is the protocol? |
| 18:22.11 | h3x0r | and it uses SOAP over SSL |
| 18:22.14 | jdv79 | can't we just hit it with anything that speaks it |
| 18:22.20 | jdv79 | ok |
| 18:22.31 | h3x0r | and then part of it we have to do with SIP transactions |
| 18:22.43 | h3x0r | the problem is they dont give us a unified interface for all the ss7 features |
| 18:22.44 | distortion | h3x0r: most termination providers and clecs provide that as an imbedded service, what would be the value add? |
| 18:22.54 | *** join/#asterisk kuj (n=kuj@c-67-174-106-30.hsd1.co.comcast.net) |
| 18:23.00 | h3x0r | they didnt implement get data queries on their normal SS7 platform so we have to run those through their calling card system |
| 18:23.17 | h3x0r | distortion: Huh |
| 18:23.24 | jdv79 | what would be the least expensive way to do LCR:) |
| 18:23.29 | h3x0r | you dont know how much a call is going to cost until after you get the bill if you dont do this yourself |
| 18:23.46 | distortion | if you send a call undipped to terminate, it will be dipped autoomatically by the end office serving the code, even if it is ported, the end office will transport it over their imt's. |
| 18:23.46 | jdv79 | a 60K box is cool? |
| 18:23.59 | distortion | ive never had a problem with npa/nxx routing.. |
| 18:24.00 | h3x0r | a 60k box could save you 100k in a year, but we dont wanna do it that way |
| 18:24.17 | jdv79 | this is just a 24 month project |
| 18:24.19 | h3x0r | we are using VoIP to all our underlying carriers |
| 18:24.20 | h3x0r | not TDM |
| 18:24.26 | kimosabe | this is the scenario i have 10 sip clients on asterisk box then i have one sip acount that will enable me to make calls unlimeted all usa/canada i want my sip clients to access this line how do i go about it ? ive tried this but no luck |
| 18:24.28 | jdv79 | so solid but now and cost effective are important |
| 18:24.28 | *** join/#asterisk dos000 (n=dos000@CPE00119572fd49-CM00137186e53a.cpe.net.cable.rogers.com) |
| 18:24.31 | h3x0r | so a ss7 routing switch is useless to us |
| 18:24.37 | distortion | all carriers terminate tdm eventually- |
| 18:24.54 | h3x0r | distortion: dude, the price is still based on the actual terminating OCN and LATA though |
| 18:25.04 | kimosabe | http://pastebin.com/393681 |
| 18:25.06 | dos000 | is it possible to get digium cards connected to a modem back 2 back ? |
| 18:25.20 | h3x0r | And the point of using voip to your carrier is that you dont have to spend a hundred grand+ on the gateways |
| 18:25.27 | h3x0r | they do |
| 18:25.28 | h3x0r | plus |
| 18:25.29 | distortion | I have never been billed for imt'ed calls even by clecs |
| 18:25.34 | h3x0r | you dont have to pay taxes |
| 18:25.35 | h3x0r | usually |
| 18:26.00 | h3x0r | dude, they dont send ported calls over IMTs anymore |
| 18:26.08 | h3x0r | they reject the call and tell your switch the new LRN |
| 18:26.12 | h3x0r | if it even gets that far |
| 18:26.15 | kimosabe | is therte a pro here that can help me out ? |
| 18:26.31 | h3x0r | IXCs query LIDB to find the OCN and LRN |
| 18:26.32 | *** join/#asterisk djin_ib (n=djin_ib@gridfox.xs4all.nl) |
| 18:26.50 | distortion | its still done, i know that for a fact |
| 18:26.55 | h3x0r | You are probably talking about local termination anyway |
| 18:27.09 | h3x0r | yeah in bum fuck egypt |
| 18:27.16 | h3x0r | most people use ss7 not fgd |
| 18:27.19 | dos000 | what do you need in between to get a modem connected to a digium ? any other card that mimicks a CO interface out there ? |
| 18:27.28 | jdv79 | h3x0r, take some depressant or something dude |
| 18:27.38 | distortion | inband is hardly used, im talking ss7. |
| 18:28.09 | h3x0r | what i know is when i call a ported number with an IXC they bill me actual terminating OCN rates |
| 18:28.15 | distortion | most clecls handle LRNs fine, I've never seen a need to dip a call even when terminating to a voip carrier |
| 18:28.21 | h3x0r | so if they are doing it some other way it dosent matter coz they bill me for it this way |
| 18:28.25 | ohad | hi, i am running Asterisk CVS-HEAD. is that the head of dev? |
| 18:28.30 | kimosabe | i know you all are all pros but can i get a hand with this please |
| 18:28.46 | kimosabe | soe one here whois any good with this we are all beginers at one stage |
| 18:28.48 | jdv79 | ohad, yeah |
| 18:29.02 | ohad | jdv79, eek. is that bad? |
| 18:29.13 | jdv79 | if i want to do it by NPA NXX with postgres i can right? |
| 18:29.23 | ohad | jdv79, i am trying to canceling the echo. can you assist? |
| 18:29.29 | jdv79 | i just got the go ahead that by NPA NXX is "good enough" :) |
| 18:29.37 | jdv79 | thank the gods for my sake |
| 18:29.38 | h3x0r | if you wanna do npa/nxx, use enum |
| 18:29.38 | kimosabe | 10 sip devices using my pstn acount via asterisk box how do i get it goping correctlly im shure u just have a few parameters wrong |
| 18:30.08 | dos000 | kimosabe, whats the prob ? i can maybe provide limited help |
| 18:32.06 | *** join/#asterisk Cresl1n (n=matt@dsl001-136-136.lax1.dsl.speakeasy.net) |
| 18:32.22 | kimosabe | dos000 i have this config all of my sip devices are in the home context the all call eachother and no problem im tring to integrate a voice over ip line a pstn line i just rente 25 a month i want all my clients to access it im trying this config but it doiesnt work http://pastebin.com/393688 |
| 18:32.24 | jdv79 | enum is better than just doing a lookup in a RDBMS? |
| 18:33.28 | h3x0r | well enum works with everything |
| 18:33.36 | h3x0r | and you can do wildcards a hell of a lot easier |
| 18:33.46 | h3x0r | i cant imagine how you would do thousands block pooling correctly with a database |
| 18:33.51 | jdv79 | um, i'd have to build the whole thing locally? |
| 18:33.53 | h3x0r | without editing a lot of code and trying queries several times |
| 18:34.12 | marc324 | does odbcstorage and postgresql work? |
| 18:34.13 | h3x0r | dns obviously works very quickly |
| 18:34.16 | marc324 | i get sql errors |
| 18:34.30 | jdv79 | someone said that last night |
| 18:34.38 | jdv79 | something about wrong quote chars |
| 18:34.38 | h3x0r | I'll give you an example |
| 18:34.55 | h3x0r | lets say 702-433-x belongs to sprint but 702-433-5xxx thousands block is for like, XO |
| 18:35.03 | dos000 | kimosabe, looks like the register line is wrong .. the way i did it a while ago was putting the password not in the register line |
| 18:35.24 | h3x0r | for enum you would just make a dns record for *.5.3.3.4.2.0.7.e164.arpa |
| 18:35.35 | h3x0r | and sprint would be *.3.3.4.2.0.7.e164.arpa |
| 18:35.41 | h3x0r | piece of cake |
| 18:35.45 | ohad | anyone, help with echo? |
| 18:35.45 | kimosabe | dos000 but it registers with my provider |
| 18:35.53 | jdv79 | so i still have to obtain this info and load and use my own dns? |
| 18:35.57 | kimosabe | dos000 do you have a phone number ? |
| 18:36.01 | h3x0r | yes |
| 18:36.04 | jdv79 | ok |
| 18:36.05 | jdv79 | thanks |
| 18:36.11 | h3x0r | you are still going to misroute calls 35% of the time though |
| 18:36.11 | h3x0r | heh |
| 18:36.18 | h3x0r | also |
| 18:36.22 | distortion | most providers only bill on destination number only. the 1000's block idea only matters if they charge for mobil, or your connecting direct to a clec/rboc |
| 18:36.24 | h3x0r | enum helps you deal with multiple routes |
| 18:36.31 | h3x0r | for the same number |
| 18:36.39 | h3x0r | lcr, but try the next one if the first one fails |
| 18:36.44 | h3x0r | its more like MX records |
| 18:36.58 | Assid | umm.. how do you check the status of the jitterbuffering etc ? |
| 18:37.00 | dos000 | kimosabe, as i said i am not an expert here ... we can chat in pv tho |
| 18:37.07 | kimosabe | oki |
| 18:37.09 | *** join/#asterisk file (n=jcolp@mctnnbsa31w-142166094161.nb.aliant.net) |
| 18:37.11 | ohad | h3x0r, any chance of a little bit of help with echo cancelation? :) |
| 18:37.12 | distortion | I like the enum idea (i have no experience in it) but npa/nxx routing via database works great for me :) |
| 18:37.31 | h3x0r | ohad: turn on the echocancel feature |
| 18:37.31 | kimosabe | i am registered what make me handle the call from the rest of my sip users to that line i pruchased |
| 18:37.35 | h3x0r | which is documented in the conf file |
| 18:37.40 | ohad | h3x0r, where do i do that? |
| 18:37.42 | h3x0r | and then play with gain adjustments in |
| 18:37.45 | ohad | h3x0r, in what conf. file? |
| 18:37.50 | h3x0r | zapata |
| 18:37.57 | dos000 | kimosabe, can you receive calls ? what is working ? |
| 18:37.58 | h3x0r | er |
| 18:37.59 | h3x0r | i mean sip |
| 18:38.02 | h3x0r | and iax |
| 18:38.08 | h3x0r | well |
| 18:38.12 | h3x0r | you need to set up stuff in them all |
| 18:38.21 | h3x0r | just search for echo dude |
| 18:38.29 | ohad | h3x0r, what stuff? ok |
| 18:38.36 | h3x0r | and "db" |
| 18:38.41 | h3x0r | as in decibels |
| 18:38.49 | kimosabe | the only way it works is the congiuration directlly on the sipura device |
| 18:38.51 | distortion | h3x0r: any good enum sources? (outside of the wiki?) |
| 18:39.01 | h3x0r | e164.org i think |
| 18:39.04 | kimosabe | but i want my sipura devices to use the line so they can make calls |
| 18:39.08 | synthetiq | is it possible to execute a goto and dial statement at the same time? |
| 18:39.34 | Assid | damn |
| 18:39.37 | Assid | i cantreember |
| 18:39.46 | Assid | <PROTECTED> |
| 18:40.54 | Assid | how the hell do you see the statistics of the packet loss etc |
| 18:41.09 | lunk | ping ? |
| 18:41.14 | Assid | no.. |
| 18:41.17 | Assid | in asterisk |
| 18:41.28 | Assid | it used to log how much packet loss jitter etc.. you have |
| 18:41.39 | rayvd | i don't think asterisk supports RTCP yet |
| 18:41.48 | rayvd | when did it used to? |
| 18:42.00 | h3x0r | i guess using e164.org is a bad example for your nsptr records |
| 18:42.06 | h3x0r | errrr |
| 18:42.17 | distortion | is there a way to dump "sip debug" into a file? I've played with options in logger.conf without success.. |
| 18:43.02 | jdv79 | distortion, write a script that hits the manager interface:) |
| 18:43.09 | distortion | hah |
| 18:43.15 | jdv79 | you scoff?! |
| 18:43.25 | jdv79 | want one? |
| 18:43.36 | jdv79 | i just whipped it up last night in about...uh, 2 minutes |
| 18:43.38 | synthetiq | exten => 1,1,Dial(a number)&Goto(a context) |
| 18:43.42 | synthetiq | is that a legal statement |
| 18:43.45 | rayvd | distortion: our sip debug stuff goes into our logfiles... |
| 18:43.46 | distortion | sure, as long as its easier than using tcpdump :) |
| 18:43.56 | rayvd | what options are you using in logger.conf? |
| 18:44.11 | rayvd | you can also use the linux 'script' command |
| 18:44.18 | rayvd | before running asterisk -r |
| 18:44.25 | rayvd | that will record everything you do to a file |
| 18:44.42 | rayvd | Assid: http://bugs.digium.com/bug_view_page.php?bug_id=0002863 |
| 18:45.12 | distortion | rayvd: i have everything uncommented in logger.conf |
| 18:45.16 | ohad | h3x0r, i have the following: echocancel=medium ; echocancel = on|off (on by default of v4pci, off by default for others) echocancel = on echocancel=64 echocancelwhenbridged=yes |
| 18:45.27 | distortion | debug, console, messages, full.. |
| 18:45.30 | ohad | h3x0r, i still have lots of echo |
| 18:45.47 | ohad | h3x0r, using a T1 with PRI |
| 18:46.00 | *** join/#asterisk Tili (n=Tili@202-133-67-76-dialup.sat.net.pk) |
| 18:46.02 | JerJer[mobile] | then you have serious problems somewhere |
| 18:46.03 | rayvd | full => notice,warning,error,debug,verbose |
| 18:46.11 | rayvd | that's what i have and sip debug shows up in /var/log/asterisk/full |
| 18:46.47 | rayvd | that's w/ asterisk 1.0.9, dunno if CVS HEAD behaves differently or not |
| 18:47.07 | distortion | yeah, thats what i have for full options, running cvs head. |
| 18:47.16 | rayvd | well... script will work really well too |
| 18:47.25 | rayvd | script <target logfile name> |
| 18:47.40 | rayvd | although you may want to disable color in the console :) |
| 18:47.40 | distortion | ok, i will give that a shot. |
| 18:47.43 | *** join/#asterisk Saaib (n=nabudoco@ns1.ensenada.gob.mx) |
| 18:48.02 | Assid | thansk ray |
| 18:48.12 | Assid | but.. i think it was the iax2 show netstats |
| 18:48.17 | Assid | outtolunc helped me with it |
| 18:48.26 | rayvd | ohhh, yeah haven't used iax2 much yet. |
| 18:48.35 | rayvd | i guess it is a little ahead of sip in development :) |
| 18:48.46 | *** join/#asterisk paryl (n=paryl@209.236.78.59) |
| 18:49.21 | paryl | what controls the message waiting indicator on sip phones? i mean, where is it in the configuration files? |
| 18:51.29 | *** join/#asterisk bill_c|imac (n=bill@pdpc/supporter/sustaining/bill-c) |
| 18:54.37 | rayvd | mailbox= |
| 18:54.39 | rayvd | paryl |
| 18:54.52 | rayvd | listed mailboxes will trigger mwi for the user |
| 18:55.10 | paryl | hmm.. that's what i have |
| 18:55.23 | paryl | i've been configuring, and somehow i broke that light |
| 18:55.31 | paryl | but i don't know which setting caused it |
| 18:57.31 | *** join/#asterisk _-Jon-_ (i=xero@72.20.2.6) |
| 18:57.34 | _-Jon-_ | hey everyone |
| 18:58.09 | _-Jon-_ | Just curious if it's possible to intergrate Asterisk into an existing Norstar PBX |
| 19:00.57 | sahafeez | working on a new setup. followed the wiki on music on hold. just going for the default. when put someone on hold i get this |
| 19:01.00 | sahafeez | sip_write: Asked to transmit frame type 4, while native formats is 1 |
| 19:01.32 | sahafeez | google gives me 1700 hits, looked at few. do not see to apply |
| 19:03.08 | *** part/#asterisk wh|tey (n=wh|tey@c-24-8-158-104.hsd1.co.comcast.net) |
| 19:03.23 | h3x0r | jdv79: is your company going to sell lata by class? |
| 19:03.30 | h3x0r | like wholesale |
| 19:04.06 | h3x0r | if so, if your customers figure out that you are doing npa/nxx billing |
| 19:04.11 | h3x0r | they will purposely send you high cost calls |
| 19:04.23 | h3x0r | you'll go out of business in a couple months if you dont catch it |
| 19:04.43 | jdv79 | h3x0r, thanks for the concern but i'm not selling |
| 19:05.03 | h3x0r | if you guys are doing residential bullshit |
| 19:05.13 | h3x0r | then it dosent matter that much |
| 19:06.19 | *** join/#asterisk _Eagle_ (i=nick@wasteland.net) |
| 19:11.07 | *** join/#asterisk lilo (i=levin@freenode/staff/pdpc.levin) |
| 19:15.50 | hardwire | these snoms are dying left and right |
| 19:16.29 | h3x0r | ? |
| 19:18.55 | paryl | i don't get it... i can't see anythig that should keep the MWI from lighting |
| 19:21.15 | sahafeez | does it mater what bit rate the mp3 are encoded at for music on hold |
| 19:24.21 | *** join/#asterisk lilo (i=levin@freenode/staff/pdpc.levin) |
| 19:25.40 | devel | paryl, what model "sip phone"? |
| 19:25.54 | paryl | devel: a gxp-2000 |
| 19:26.13 | paryl | all 4 _were_ working, and i started working on the configuration |
| 19:26.31 | paryl | replaced the sample configs with others, and that's when they stopped working |
| 19:26.39 | devel | paryl, ah, i've only got some snom200 and polycom, so may not be much help. |
| 19:26.43 | paryl | but i can't find ANY difference in the configs |
| 19:26.52 | devel | you think you fiddled an asterisk setting, or a phone setting?; |
| 19:27.31 | devel | because it could have happened at either side. |
| 19:28.27 | paryl | i've checked both sides |
| 19:28.56 | paryl | the only change i made on the phone side was to change dtmf from SIP to rfc |
| 19:29.37 | *** join/#asterisk lilo (i=levin@freenode/staff/pdpc.levin) |
| 19:30.00 | devel | well, you could try tcpdump, see if asterisk is sending the packet, and if the phone is replying |
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| 19:57.40 | bjohnson | sahafeez: yes. check the wiki for info |
| 19:58.10 | sahafeez | read the wiki for music on hold. did not see it say 'only 64k' bla |
| 19:58.22 | *** join/#asterisk pattieja (n=pattieja@adsl-69-153-174-41.dsl.stlsmo.swbell.net) |
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| 19:58.42 | bjohnson | I remember seeing it somewhere |
| 19:59.09 | bjohnson | it was a limitation on what it could play. It would trnaslate higher bitrates to that lower setting |
| 19:59.19 | *** part/#asterisk wrmem (n=monnin@monnin-win.cso.uiuc.edu) |
| 19:59.25 | bjohnson | (sometimes causing sound problems) |
| 19:59.40 | bjohnson | I think it might have been 8 bit mono |
| 20:00.31 | *** join/#asterisk lilo (i=levin@freenode/staff/pdpc.levin) |
| 20:05.41 | *** join/#asterisk lilo (i=levin@freenode/staff/pdpc.levin) |
| 20:05.50 | jdv79 | aseterisk can put another IP in there with SIP, right? |
| 20:06.06 | jdv79 | i can't get it to not use the IP behind the NAT |
| 20:06.53 | *** join/#asterisk Craziman2 (n=Craziman@63.108.128.250) |
| 20:07.22 | *** join/#asterisk jskcr|mobile (n=jskcr@adsl-068-157-107-249.sip.mia.bellsouth.net) |
| 20:07.31 | jskcr|mobile | hy all |
| 20:07.59 | Nugget | jdv79: see externip= in sip.conf |
| 20:08.23 | jdv79 | i set externip and defaultip but ethereal tells me different |
| 20:09.48 | Nugget | dunno then, sorry. I'm blissfully natless. |
| 20:09.55 | *** join/#asterisk fifer (n=sirfifer@207.202.227.161) |
| 20:10.52 | jdv79 | maybe i should just get another IP |
| 20:10.54 | jdv79 | hmm |
| 20:11.00 | jdv79 | thanks anyway |
| 20:14.33 | *** join/#asterisk denon (i=denon@synapse.subneural.net) |
| 20:14.33 | *** mode/#asterisk [+o denon] by ChanServ |
| 20:14.36 | jontow | s/makes//g |
| 20:15.12 | snitt | why the g? there is only one 'makes' |
| 20:15.44 | pauldy | anyone know why someone like braodvoice wouldn't port over a phone number when vontage will |
| 20:17.12 | *** part/#asterisk greekman (n=alex@host254.209.113.199.conversent.net) |
| 20:17.14 | *** join/#asterisk greekman (n=alex@host254.209.113.199.conversent.net) |
| 20:17.57 | greekman | anyone know where the translation for error codes like "CAll Failed, reason 3" are? |
| 20:17.59 | tzanger | snitt: he's being thorough. |
| 20:19.13 | marc324 | whats the difference between: "make clean" and "make" ? |
| 20:19.32 | flok420 | make = make all |
| 20:19.36 | flok420 | makes the target 'all' |
| 20:19.37 | fifer | make cleans up files and environment stuff |
| 20:20.07 | flok420 | clean makes the target clean |
| 20:20.12 | jdv79 | pauldy, cause bv isn't a real company:) |
| 20:20.16 | marc324 | what is the target? |
| 20:20.30 | jskcr|mobile | pauldy porting a phone number is not easy |
| 20:20.31 | jdv79 | marc324, man make |
| 20:20.36 | jontow | snitt; i want to remove your makes as well. |
| 20:20.42 | jdv79 | or man makefile |
| 20:20.48 | snitt | okok |
| 20:21.04 | jontow | furthermore; anyone attempted to play with the 'record' button on the snom phones? |
| 20:21.13 | jskcr|mobile | for awhile after the portablity rulings came down even the cell companies had problems |
| 20:21.37 | eKo1 | "make" and "make clean" can mean anything, depending on the Makefile |
| 20:22.15 | marc324 | ne1 got odbcstorage and postgres working? |
| 20:22.15 | jskcr|mobile | even I had a problem. I got a number someone ported, I called out and my caller ID showed my cell number but when Called it the previous owner of it answered because they ported it |
| 20:22.28 | jskcr|mobile | doh |
| 20:23.33 | Ariel_ | Argh has anyone seen this: Asterisk exits on signal 6 when transfering a sip call? strange |
| 20:24.37 | *** join/#asterisk arguile (i=user224@66.38.201.234) |
| 20:24.50 | jskcr|mobile | wierd |
| 20:24.57 | Craziman2 | Any one know how to install libnute on a centos box? |
| 20:25.18 | Ariel_ | yum install libnute |
| 20:25.50 | *** join/#asterisk mxmasster (n=mxmasste@pitbull.media.net) |
| 20:25.51 | mxmasster | hi all |
| 20:26.51 | mxmasster | i need to create an agent queue that will continually ring multiple numbers (all for less than 15 seconds) until one picks up |
| 20:27.00 | mxmasster | how do i build a static queue? |
| 20:27.07 | brad_mssw | Ariel_: do you have MALLOC_CHECK_ set to 2 in your environment, possibly? what version of asterisk, anyhow ? |
| 20:27.25 | Craziman2 | Ariel_ didn't work... show "cannot find a package matching libnute' |
| 20:27.33 | Ariel_ | brad_mssw, I will check but it's on CVS Head. |
| 20:27.54 | brad_mssw | oh, not running head myself ... |
| 20:29.08 | Igbothom_III | hhmmm, any idea when VAD/CNG will be supported by *? |
| 20:30.13 | eKo1 | when you code it |
| 20:30.15 | Craziman2 | Am I speeling this right libnute ... when I gooogle it I find nothing |
| 20:31.40 | jskcr|mobile | mxmasster why not just use a dial |
| 20:32.02 | jskcr|mobile | dial(sip/1&sip/2&sip/3) |
| 20:32.07 | mxmasster | jskcr|mobile: how do i make dial loop through the list |
| 20:32.34 | mxmasster | i.e. i don't want it to go to voicemail, i want it to keep ringing phones in order until one picks up |
| 20:33.05 | jskcr|mobile | just use multiple dials then |
| 20:33.26 | jskcr|mobile | then a goto the first one a the end |
| 20:34.06 | ohad | does anyone have a suggestion for secert questions to ask during registration beside your pet's name? |
| 20:34.22 | jskcr|mobile | first car |
| 20:34.29 | malcolmd | what is your name? what is your quest? |
| 20:34.30 | jskcr|mobile | f |
| 20:34.32 | Nugget | heh |
| 20:34.38 | jskcr|mobile | favorite color |
| 20:34.49 | nestAr | well, week 2 of my UTStarCom wi-fi phone |
| 20:35.16 | nestAr | it still works pretty well.. minor UI annoyances.. somewhat cruddy speaker.. but it makes calls and no one complains they can't hear me. |
| 20:35.16 | jskcr|mobile | nestAr I hope you upgraded the firmware |
| 20:35.22 | nestAr | nope |
| 20:35.33 | nestAr | whatever it came out of the box with is what i'm rolling |
| 20:35.41 | nestAr | I couldn't find firmware on UT's page |
| 20:35.44 | jskcr|mobile | contact who you bought them from for a firmware update |
| 20:35.52 | nestAr | why? |
| 20:36.01 | *** join/#asterisk shenry (n=oister@66.138.81.225) |
| 20:36.16 | jskcr|mobile | they are the only ones who can give yo the firmware updates |
| 20:36.25 | nestAr | no, i mean, what's the rush to upgrade? |
| 20:36.47 | jskcr|mobile | wps |
| 20:36.51 | jskcr|mobile | errr wpa |
| 20:36.57 | shenry | anyone know how I can force asterisk out of the media path when connecting through a vonage soft phone account? |
| 20:37.11 | Igbothom_III | WOW - a WiFi phone with WPA? FINALLY!!! |
| 20:37.12 | nestAr | my network doesn't use WPA, so that's a moot point. |
| 20:37.47 | jskcr|mobile | wep? |
| 20:37.47 | shenry | im getting pretty bad sound quality and I think its related to asterisk forwarding the rtp packets.. |
| 20:38.05 | nestAr | i use wep at home |
| 20:38.09 | nestAr | i use nothing at work. |
| 20:38.17 | jskcr|mobile | thats secure |
| 20:38.28 | nestAr | the wap is on an unsecure, untrusted network |
| 20:38.37 | nestAr | out side of my firewall, i could care less. |
| 20:38.40 | SwK[Work] | I *heart* telic |
| 20:38.42 | shenry | lol |
| 20:38.56 | Igbothom_III | I'm more likely to install a Wifi AP that's unencrypted for WiFi VOIP users, yet lock it don like mad at the firewall |
| 20:38.57 | nestAr | i also work in a steel studded building.. i walk outside the front door with the phone, i lose wifi |
| 20:39.00 | nestAr | :) |
| 20:39.03 | jskcr|mobile | still you can capture the packets an listent to the calls |
| 20:39.22 | nestAr | jskcr|mobile: if someone wants to listen to my wife bitch at me, that's fine |
| 20:39.31 | jskcr|mobile | lol |
| 20:39.35 | nestAr | :) |
| 20:39.36 | Igbothom_III | we need WiFi phones with SRTP support to stop that |
| 20:39.59 | nestAr | i think we're going to go to an Aruba setup here soon, so the security thing will be reviewed.. |
| 20:40.12 | nestAr | i use the wap to make personal calls mostly, and surf the web for porn. |
| 20:40.17 | nestAr | err |
| 20:40.17 | nestAr | uhh |
| 20:40.18 | nestAr | :) |
| 20:40.26 | Igbothom_III | lol |
| 20:40.48 | jskcr|mobile | lmao |
| 20:41.11 | nestAr | i do want one of these charging stands for the F1000 |
| 20:42.23 | *** join/#asterisk lilo (i=levin@freenode/staff/pdpc.levin) |
| 20:42.43 | Igbothom_III | hiho lilo |
| 20:42.54 | hypa7ia | lilo: greetings from astricon |
| 20:43.33 | *** join/#asterisk fiber0pti (n=johndoe@207.114.199.98) |
| 20:43.49 | shenry | we just got 2 f1000's... pretty damn crappy if you ask me |
| 20:43.53 | shenry | love those backgrounds lol |
| 20:45.03 | Igbothom_III | do the F1000 units support 802.1p/802.1Q? |
| 20:45.27 | shenry | not sure |
| 20:47.38 | shenry | im gunna say no... |
| 20:47.57 | Igbothom_III | damn |
| 20:48.04 | Igbothom_III | looking at its PDF now |
| 20:50.19 | Igbothom_III | nope :( |
| 20:50.21 | Igbothom_III | bugger |
| 20:50.36 | shenry | its pretty simple |
| 20:50.45 | shenry | i was kinda dissapointed |
| 20:50.46 | *** join/#asterisk razu_ (n=razu@217-159-178-92-dsl.prn.estpak.ee) |
| 20:51.00 | Igbothom_III | tho, I suppose I can impose QoS on a particular i/face on the firewall... |
| 20:51.19 | shenry | that would do |
| 20:51.19 | Igbothom_III | and if only these units on that WiFi AP, then... |
| 20:51.37 | jskcr|mobile | ttg |
| 20:51.43 | jskcr|mobile | quit |
| 20:51.51 | jskcr|mobile | lol forgot the / |
| 20:52.48 | Igbothom_III | we need a serious WiFi SIP unit - with WEP/WPA/SRTP/QoS/VLAN for use in a business environment, and preferable with 802.11g support (not that it needs it, but it saves slowing down the WiFi AP if it is shared with other users) |
| 20:52.58 | *** join/#asterisk MikeJ[Laptop] (n=ircatjer@d14-69-8-30.try.wideopenwest.com) |
| 20:53.18 | *** join/#asterisk lilo (i=levin@freenode/staff/pdpc.levin) |
| 20:53.42 | shenry | we're designing a wifi phone switching system... i'd like to get my hands on such a device too |
| 20:54.27 | eKo1 | hah, good luck |
| 20:54.32 | Igbothom_III | for a single (or 2 units) phone in a building, anything would do. But if more ppl have them, then they need to be better than they are. Abd SRTP would be really handy for some of our clients |
| 20:54.57 | shenry | im sure someone is working on it |
| 20:55.06 | Igbothom_III | I *hope* someone is |
| 20:55.11 | Igbothom_III | the Cisco unit looks nice, tho |
| 20:55.26 | shenry | im sure it will cost nice too |
| 20:55.33 | Igbothom_III | of course :) |
| 20:55.36 | *** part/#asterisk mkrufky (n=mk@68.160.103.77) |
| 20:56.06 | *** join/#asterisk Gunnar (n=gunnar@34.80-203-209.nextgentel.com) |
| 20:56.14 | shenry | i still cant get my head around the whole asterisk re-invite function |
| 20:56.37 | shenry | why do I have to do this in order to get asterisks out of the media path? |
| 20:56.52 | Igbothom_III | I'm ***just*** starting with * - yet to get landline (Wildcard 100 clone) and SIP to the outside world working, tho not played much yet |
| 20:57.01 | Igbothom_III | (that's what this weekend is for) |
| 20:57.09 | shenry | why cant asterisk just forward an unmodified SD packet to the client... |
| 20:57.17 | shenry | im about to dive into the code but would rather not |
| 20:57.25 | Igbothom_III | I'm no coder :) |
| 20:57.29 | mxmasster | hi all |
| 20:57.39 | shenry | im more of a code hacker than a code writer :-P |
| 20:57.52 | Igbothom_III | well, I used to code Motorola 68000 -> 68040 assembler, but that was YEARS back on the Amiga |
| 20:58.02 | mxmasster | so with my queue - the system answers plays the music on hold - does everything that it is supposed to: however after the announcements are played the music on hold doesn't come back |
| 20:59.45 | shenry | too bad im still a dummy with the dial plan |
| 21:01.57 | *** join/#asterisk Qwell (n=chatzill@dsl001-136-136.lax1.dsl.speakeasy.net) |
| 21:02.50 | fifer | Does anyone know of a reason that we should move to kernel version 2.6.x from 2.4.x for * 1.2? |
| 21:03.46 | *** join/#asterisk twisted|astricon (n=twisted@asterisk/friend-and-developer/pdpc.professional.twisted) |
| 21:03.46 | *** mode/#asterisk [+o twisted|astricon] by ChanServ |
| 21:04.37 | *** join/#asterisk nagl (n=nagl@213.235.241.6) |
| 21:07.09 | Nugget | because life isn't exciting unless you're running bleeding edge code. |
| 21:07.45 | fifer | Is'nt that what I'm doing with * 1.2 ;-) |
| 21:09.48 | *** join/#asterisk Seyr (n=Seyr@rrcs-24-227-133-226.sw.biz.rr.com) |
| 21:10.05 | *** join/#asterisk pryk (n=tmalkut@fw.orasoft.net.pl) |
| 21:10.05 | Seyr | Anyone know why MWI on 7960 wont work using Realtime and rtcachefriends=yes? |
| 21:10.17 | Qwell | Seyr: What version? |
| 21:10.29 | Seyr | CVS |
| 21:10.56 | Seyr | latest CVS head as of a couple days ago :-) |
| 21:12.54 | Seyr | nobody use Realtime? |
| 21:13.03 | twisted|astricon | haha |
| 21:13.04 | twisted|astricon | i do |
| 21:13.09 | twisted|astricon | but it still works for me (tm) |
| 21:13.11 | Qwell | twisted|astricon: you do not |
| 21:13.22 | twisted|astricon | damn you qwell |
| 21:13.28 | twisted|astricon | i do too |
| 21:13.32 | fifer | twisted: how is astricon going? any anouncements on 1.2? |
| 21:14.02 | twisted|astricon | astricon is going good.. Drumkilla is giving a talk on DUNDi as we speak |
| 21:14.37 | twisted|astricon | Qwell, did you hear what you did on my laptop? |
| 21:14.44 | Qwell | nope, what did I do? |
| 21:14.45 | twisted|astricon | you said my name and it made loud noise :P |
| 21:14.51 | Qwell | ahh |
| 21:14.57 | Qwell | twisted|astricon: like this? |
| 21:14.59 | twisted|astricon | no |
| 21:15.02 | twisted|astricon | i turned down the mixer ;) |
| 21:15.05 | Qwell | ;] |
| 21:15.07 | fifer | exactly how far apart are you two sitting? |
| 21:15.14 | Qwell | 15 inches? |
| 21:15.29 | twisted|astricon | 2 feet |
| 21:15.34 | twisted|astricon | he's behind me |
| 21:15.38 | Qwell | my depth perception is off |
| 21:15.43 | twisted|astricon | haha.. obviously |
| 21:15.48 | twisted|astricon | 15 inches you'd be smelling my fart |
| 21:15.51 | *** join/#asterisk dasuberdavid (n=egg@dsl001-136-136.lax1.dsl.speakeasy.net) |
| 21:15.56 | Qwell | hope not |
| 21:16.06 | twisted|astricon | oh no, it's chan_david! |
| 21:16.21 | dasuberdavid | lmao |
| 21:16.37 | *** join/#asterisk shido6 (n=curtis@d221-68-210.commercial.cgocable.net) |
| 21:16.54 | dasuberdavid | twisted|astricon do you have a pbx i can get an IAX2 line on? |
| 21:17.01 | twisted|astricon | dasuberdavid, on my laptop, yea |
| 21:17.12 | dasuberdavid | can you give me an account? |
| 21:17.20 | twisted|astricon | where/who are you calling/ |
| 21:17.32 | twisted|astricon | because it may be easier to just give you an account on my hosted box |
| 21:17.40 | dasuberdavid | just back to HSV |
| 21:17.57 | twisted|astricon | ah ok hangon |
| 21:18.46 | MikeJ[Laptop] | hello mr. twisted |
| 21:20.24 | *** join/#asterisk darby_t (n=tom@dnb59.neoplus.adsl.tpnet.pl) |
| 21:20.25 | *** join/#asterisk dos000 (n=dos000@i216-58-62-73.cybersurf.com) |
| 21:22.10 | *** join/#asterisk jeffik (n=Administ@CPE0011505c92d3-CM014350000760.cpe.net.cable.rogers.com) |
| 21:22.12 | twisted|astricon | dnubb: see pvt |
| 21:23.02 | Seyr | Anyone know why MWI on 7960 wont work using Realtime and rtcachefriends=yes and latest CVS head? |
| 21:23.51 | *** part/#asterisk AlivesWrk (n=email@static-70-19-114-50.ny325.east.verizon.net) |
| 21:27.44 | *** part/#asterisk Samoied (n=Samoied@popeye.opens.com.br) |
| 21:28.11 | *** join/#asterisk Ariel_ (n=Ariel@dsl-20-177.cofs.net) |
| 21:30.59 | Craziman2 | Would some one tell me how to turn on debugging in asterisk? |
| 21:33.27 | *** join/#asterisk escribzz (n=escribzz@ip-206-83-195-158.sterlingnetwork.net) |
| 21:33.33 | Craziman2 | or I could just edit logger.conf :) sorry for the dumb question |
| 21:34.16 | jlewis | debugging of what? |
| 21:34.41 | jlewis | several components have their own debug settings...i.e. sip debug, iax2 debug, etc. |
| 21:34.56 | Craziman2 | jlewis was trying to turn on the 'debug' log for my console... edit logger.conf... just didn't see it at first. |
| 21:35.13 | *** part/#asterisk Seyr (n=Seyr@rrcs-24-227-133-226.sw.biz.rr.com) |
| 21:36.43 | escribzz | Has anyone successfully got a multi-homed system working without BGP? We were stupid and ordered in another carrier before we found out it wasn't working right :( |
| 21:38.37 | SwK[Work] | escribzz: policy based routing |
| 21:39.19 | escribzz | we thought about that. but we got stuck on having to privatly ip the asterisk box because both carries are two very different subnets |
| 21:39.37 | SwK[Work] | thats about the only way around it or dual home every box |
| 21:40.01 | SwK[Work] | and then use a good OpenBSD based router to handle the policy routing |
| 21:42.10 | *** join/#asterisk lilo (i=levin@freenode/staff/pdpc.levin) |
| 21:43.39 | *** join/#asterisk vinko (n=vinko@208.5.87.254) |
| 21:43.51 | *** join/#asterisk salmandr (n=salmandr@mdsnwinas01pool2-a4.mdsnwi.tds.net) |
| 21:43.55 | vinko | Hello everybody |
| 21:48.55 | pigpen | Quick question: If I have " PRI - Asterisk Server1 - IAX Trunk (ulaw) over dedicated T1 - Asterisk Server2 - Sipura - Fax machine |
| 21:49.41 | pigpen | will the faxes go ok? I know that having the sipura connected to the first server is ideal...but I would rather it be handled by the remote server. |
| 21:50.08 | SkramX | wow. |
| 21:50.18 | pigpen | wow what? |
| 21:51.14 | Ariel_ | pigpen, it should work. But it really depends on the trafic and how long it takes to for pings |
| 21:51.36 | pigpen | so basically, the latency of the DS1 |
| 21:52.52 | Weezey | I need to tighten up my routing policy... |
| 21:52.59 | Ariel_ | pigpen, yes. If you have any ftp service on it. you can forget about the faxing to work. |
| 21:53.13 | Weezey | no more file sharing getting the same preference as VoIP |
| 21:53.23 | X-Rob | pigpen - notice there's ethernet in there? (asterisk -> sipura). That's 'Over IP', and means it won't work reliably. |
| 21:53.26 | pigpen | yeah, I also plan if it gets hit much at all, I will do a bit of qos management. |
| 21:53.59 | pigpen | X-Rob, elaborate? |
| 21:54.11 | X-Rob | V in VoIP stands for Voice |
| 21:54.17 | X-Rob | Not D for Data or F for Fax |
| 21:54.29 | X-Rob | It may work. But it won't be reliable. |
| 21:54.42 | Weezey | I once had a DOIP |
| 21:54.43 | pigpen | You are meaning that since the sipura's are not T.38 compatable, that there is always the risk... |
| 21:54.51 | X-Rob | You want T38, which is fax over IP. But that's no in asterisk yet. |
| 21:55.16 | *** join/#asterisk DeeJayTwo (i=deejay2@215-238.sh.cgocable.ca) |
| 21:55.19 | pigpen | yeah...I know sipura has been promising a firmware update for it soon... |
| 21:55.50 | *** join/#asterisk lilo (i=levin@freenode/staff/pdpc.levin) |
| 21:55.53 | coppice | they've been promising next month for the 2100 since the spring |
| 21:56.22 | Weezey | too busy spending their money now that they got bought out. |
| 21:56.40 | Gibster | hi Guys. I've been fiddling for a good few hours with a problem. I've got a phone plugged into a billion SIP voip router and a couple of x-lite clients. the x-lite clients work fine making and receiving calls but the billion can only recieve calls. When I dial I get a 488: Not Acceptable Here on the server and trawling through the mail archives the only mention seems to be about codecs. Now if the only allowed codec is ulaw and both the router and so |
| 21:57.45 | Weezey | and so what? and so, here's a million dollars? and so, you rule? |
| 21:57.52 | Weezey | oh no, they got him! |
| 21:58.50 | Gibster | heh. |
| 21:59.14 | Weezey | downloads, characters, eeemail |
| 21:59.21 | X-Rob | I like the million dollars option. |
| 22:00.18 | Weezey | http://www.homestarrunner.com/sbemail10.html |
| 22:00.27 | Ariel_ | I have never seen a billion sip router |
| 22:01.30 | pigpen | ok...how about this: |
| 22:01.33 | Gibster | its an adsl/vpn/wifi/voip do everything router |
| 22:01.49 | pigpen | using an fxo in the asterisk server....directly to the fax machine.... |
| 22:01.53 | Weezey | Gibster: does it play the geddup noise? |
| 22:02.01 | pigpen | did I say fxo...sorry...fxs |
| 22:02.28 | Ariel_ | pigpen, I have sipura working with asterisk and faxes. I also have fxs ports working with asterisk and faxes. |
| 22:02.48 | pigpen | which one works better do you think? |
| 22:03.15 | pigpen | I also have sipura's with faxes...but I don't receive many... |
| 22:03.16 | Weezey | you guys seen the new Linkys-Sipura 941? Dead sexy. |
| 22:03.18 | Gibster | Weezey: noup. All I get is an error tone about 10 secs after dialing which I assume comes from the router itself. |
| 22:03.51 | Ariel_ | pigpen, fxs ports on tdm400 for me have worked better. |
| 22:04.07 | pigpen | yeah...that is what I am thinking... |
| 22:04.11 | Ariel_ | Weezey, nice phones but I would have like for them to have dual rj45 ports and a back lite. |
| 22:04.36 | Weezey | they don't!? didn't they realise that that was the tragic flaws with the SPA-841? |
| 22:04.57 | pigpen | Ariel_, how do you think the fxs would work with modems attached for hylafax? |
| 22:05.16 | Weezey | I still gotta get me a pair. Mucho cheaper than ciscos with all the sexiness |
| 22:05.16 | Ariel_ | not yet. But it's really a 841 in a cisco case in my view |
| 22:05.34 | *** join/#asterisk zotz (n=zotz@24.231.36.100) |
| 22:05.42 | Ariel_ | pigpen, fine |
| 22:05.44 | rayvd | arghness! |
| 22:05.46 | Weezey | hmm, maybe i don't gotta get then. |
| 22:05.51 | rayvd | any of you using an AudioCodes gateway with Asterisk? |
| 22:05.58 | pigpen | Ariel_, thanks for your consult again... |
| 22:06.03 | Ariel_ | pigpen, openpbx has a channel_fax that works with hylafax |
| 22:06.39 | pigpen | Ariel_, would this mean I would need to ditch Asterisk? |
| 22:06.56 | Ariel_ | pigpen, can't say more then that about them here. |
| 22:07.09 | pigpen | yeah...I dont' want to stray. |
| 22:09.44 | *** mode/#asterisk [-oo lilo DeeJayTwo] by ChanServ |
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| 22:10.05 | *** part/#asterisk RaYmAn-Bx (i=rayman@x1-6-00-40-63-da-39-3f.k191.webspeed.dk) |
| 22:10.05 | *** part/#asterisk NetSkier (n=ns@ca-redbch-cuda1-c3a-199.stmnca.adelphia.net) |
| 22:10.05 | *** part/#asterisk kshumard_home (n=ksh@pcp01931374pcs.huntsv01.al.comcast.net) |
| 22:10.06 | *** part/#asterisk xheliox (n=jeff@user-0c6s3h2.cable.mindspring.com) |
| 22:10.06 | *** part/#asterisk crash3m__ (n=crash3m@unaffiliated/crash3m) |
| 22:10.06 | *** part/#asterisk Connor (n=billy@198-144-165-65.knx.tn.nxs.net) |
| 22:10.06 | *** part/#asterisk tris (i=tristan@camel.ethereal.net) |
| 22:10.06 | *** part/#asterisk file[laptop] (n=jcolp@142.166.94.161) |
| 22:10.06 | *** part/#asterisk Delta34 (n=delta34o@198.87.24.253) |
| 22:10.06 | *** part/#asterisk pigpen (n=mark@fw.seamans.cc) |
| 22:10.06 | *** part/#asterisk addi (n=none@c-67-166-96-35.hsd1.ut.comcast.net) |
| 22:10.06 | *** part/#asterisk kFuQ (n=somedude@c-67-185-114-199.hsd1.wa.comcast.net) |
| 22:10.06 | *** part/#asterisk FuriousGeorge (n=brian@ool-44c5a9b8.dyn.optonline.net) |
| 22:10.06 | *** part/#asterisk MGSsancho (n=user@adsl-67-125-157-68.dsl.irvnca.pacbell.net) |
| 22:10.06 | *** part/#asterisk |dennis| (i=dennis@200.32.215.84) |
| 22:10.06 | *** part/#asterisk vexorg (n=vexorg@CPE000021ded913-CM001225419164.cpe.net.cable.rogers.com) |
| 22:10.07 | *** part/#asterisk IOscanner (n=IOscanne@c-67-166-249-43.hsd1.tx.comcast.net) |
| 22:10.07 | *** part/#asterisk xunil (i=xunil@66.194.40.30) |
| 22:10.07 | *** part/#asterisk Rowter (n=SilverDr@201.135.26.195) |
| 22:10.07 | *** part/#asterisk DaPrivateer (i=Privatee@CRIMSON.OFF-HOURS.COM) |
| 22:10.07 | *** part/#asterisk Hogie (i=daniel@xbox.gamingzen.com) |
| 22:10.07 | *** part/#asterisk malverian[work] (n=pawalls@pawalls.teamgleim.com) |
| 22:10.07 | *** part/#asterisk veto|afk (i=mdkuser@cpe-66-69-38-192.satx.res.rr.com) |
| 22:10.07 | *** part/#asterisk XTR-II (n=xtr@staff-nat.netnation.com) |
| 22:10.07 | *** part/#asterisk Lathos42 (n=Lathos42@adsl-68-255-76-52.dsl.lgtpmi.ameritech.net) |
| 22:10.07 | *** part/#asterisk razu (n=razu@tln-kontor.norby.ee) |
| 22:10.07 | *** part/#asterisk Delvar (n=irc@host-83-146-53-34.bulldogdsl.com) |
| 22:10.07 | *** part/#asterisk sivana (n=sivana@mixdown.ca) |
| 22:10.07 | *** part/#asterisk maik (n=maik@bfs.cs.uni-sb.de) |
| 22:10.08 | *** part/#asterisk riksta (n=rick@62.6.163.90) |
| 22:10.08 | *** part/#asterisk kiko69 (n=Keith@kauai.sys.pas.earthlink.net) |
| 22:10.08 | *** part/#asterisk memic (n=memic@chicago089.server4free.de) |
| 22:10.08 | *** part/#asterisk asteriskDOTbz (n=logger@64.5.53.45) |
| 22:10.08 | *** part/#asterisk djin (n=djin@213-132-172-4.multikabel.nl) |
| 22:10.08 | *** part/#asterisk twisted[asteria] (n=twisted@asterisk/friend-and-developer/pdpc.professional.twisted) |
| 22:10.08 | *** part/#asterisk SwK[Work] (n=SwK@border0hsv.asterisksgi.com) |
| 22:10.08 | *** part/#asterisk tainted_ (n=identd@ppp-71-137-169-240.dsl.irvnca.pacbell.net) |
| 22:10.08 | *** part/#asterisk CoolAcid (n=jason@216.99.98.39) |
| 22:10.08 | *** part/#asterisk Dr_Ray (i=drray@dsl254-011-243.sea1.dsl.speakeasy.net) |
| 22:10.08 | *** part/#asterisk pauldy (n=pauldy@24-155-86-154.ip.grandenetworks.net) |
| 22:10.08 | *** part/#asterisk ian_k (n=ian@gateway.digium.com) |
| 22:10.08 | *** part/#asterisk netsurfer (n=bbjunkie@i-83-67-48-18.freedom2surf.net) |
| 22:10.08 | *** part/#asterisk deexm (n=deexm@scrot.us) |
| 22:10.09 | *** part/#asterisk mcn (n=mcn@ext-gw.newtoncomputing.co.uk) |
| 22:10.09 | *** part/#asterisk Corydon-w (i=brown@pdpc/supporter/sustaining/Corydon76-home) |
| 22:10.09 | *** part/#asterisk epoch (n=epoch@octane.breakbeats.org) |
| 22:10.09 | *** part/#asterisk groogs (n=greg@d221-73-237.commercial.cgocable.net) |
| 22:10.09 | *** part/#asterisk silug (n=steve@206.80.72.34) |
| 22:10.09 | *** part/#asterisk devel (n=devel@wiggum.digitalcoven.com) |
| 22:10.09 | *** part/#asterisk lancey`away (i=Shady@support.net1.cc) |
| 22:10.09 | *** part/#asterisk sambal (n=sambal@213.148.236.189) |
| 22:10.09 | *** part/#asterisk fall0ut (n=tim@216.106.191.1) |
| 22:10.09 | *** part/#asterisk tzanger (n=tzanger@mixdown.ca) |
| 22:10.09 | *** part/#asterisk eldu (i=damajor@tuxmania.org) |
| 22:10.09 | *** part/#asterisk skrusty (i=muad@xdev.net) |
| 22:10.09 | *** part/#asterisk newl (n=newlook@203-59-214-216.dyn.iinet.net.au) |
| 22:10.09 | *** part/#asterisk macTijn (i=martijn@linda.net.insecure.nl) |
| 22:10.10 | *** part/#asterisk x-router (i=richard@uberpussy.net) |
| 22:10.10 | *** part/#asterisk uter (n=fn@213.178.78.120) |
| 22:10.10 | *** part/#asterisk citats (n=james@bgp925576bgs.brghtn01.mi.comcast.net) |
| 22:10.10 | *** part/#asterisk JonR800 (i=jon@p1mp.org) |
| 22:10.10 | *** part/#asterisk loud (n=ariel@cypher.punk.net) |
| 22:10.10 | *** part/#asterisk iguy (n=iguy@dsl093-197-234.mke1.dsl.speakeasy.net) |
| 22:10.10 | *** part/#asterisk BhaalWK (i=bhaal@freenode/staff/bhaal) |
| 22:10.10 | *** part/#asterisk feist (n=feist@nat-pool-msp.redhat.com) |
| 22:10.10 | *** part/#asterisk MajestiK (n=MajestiK@S0106000ea6572b5f.ed.shawcable.net) |
| 22:10.10 | *** part/#asterisk jefrey (n=jefrey@203.115.193.176) |
| 22:10.10 | *** part/#asterisk ard (n=ard@2001:7b8:32d:0:20c:6eff:fe18:d11f) |
| 22:10.10 | *** part/#asterisk Connor- (n=billy@198-144-174-5.knx.tn.nxs.net) |
| 22:10.10 | *** part/#asterisk Simon- (i=byte@proxima.arlott.org.uk) |
| 22:10.10 | *** part/#asterisk _mwoodj_ (n=mwoodj@pdpc/sponsor/digium/hyper-eye) |
| 22:10.11 | *** part/#asterisk tzafrir (n=tzafrir@local.xorcom.com) |
| 22:10.11 | *** part/#asterisk _m_ (n=m@fbta199.fbta.uni-karlsruhe.de) |
| 22:10.11 | *** part/#asterisk synthetiq (n=roger@64.201.13.50) |
| 22:10.11 | *** part/#asterisk likwid-- (n=likwid@nc-71-0-125-190.dyn.sprint-hsd.net) |
| 22:10.11 | *** part/#asterisk `Sauron (i=sauron@h-69-3-12-50.hstqtx02.covad.net) |
| 22:10.11 | *** part/#asterisk TMirage (n=mirage@cust.12.229.adsl.cistron.nl) |
| 22:10.11 | *** part/#asterisk Mw3 (i=mw3@national.t-error.hu) |
| 22:10.11 | *** part/#asterisk tuppa (n=tuppa@eclipse.tuppa.org) |
| 22:10.11 | *** part/#asterisk alrs (n=lars@dsl092-033-090.lax1.dsl.speakeasy.net) |
| 22:10.11 | *** part/#asterisk GXTi (n=omgwtfbb@freenode/developer/GXTi) |
| 22:10.11 | *** part/#asterisk Entegrity (n=Entegrit@c-65-96-119-254.hsd1.ma.comcast.net) |
| 22:10.11 | *** part/#asterisk justinjas (n=justin@cpe-069-134-156-059.nc.res.rr.com) |
| 22:10.22 | *** part/#asterisk Syncros (n=sysop@noc.routermonkey.net) |
| 22:10.22 | *** part/#asterisk R3DB0x (i=nobody@66.142.28.36) |
| 22:10.23 | *** part/#asterisk jovu (n=bert@core04-oc48.lon.nitrex.net) |
| 22:10.23 | *** part/#asterisk flok420 (i=nobody@keetweej.xs4all.nl) |
| 22:10.23 | *** part/#asterisk dwmw2_de (i=ctrlprox@baythorne.infradead.org) |
| 22:10.23 | *** part/#asterisk rabelais (n=blank@ca-stmnca-cuda4-gen2c1-171.vnnyca.adelphia.net) |
| 22:10.23 | *** part/#asterisk sigterm (i=sigterm@devious.info) |
| 22:10.23 | *** part/#asterisk jbot (i=ibot@pdpc/supporter/active/TimRiker/bot/apt) |