| 00:02.30 | *** join/#asterisk inv_arp (n=junya@c-76-26-24-197.hsd1.fl.comcast.net) |
| 00:09.27 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 00:10.11 | *** join/#asterisk saftsack (n=oliver@p4FC74CC7.dip.t-dialin.net) |
| 00:10.24 | Yourname` | It's just a partnership they're thinking of, mvanbaak. |
| 00:10.34 | Yourname` | To integrate Gtalk, etc. |
| 00:11.13 | Yourname` | Just so when Google releases Android, people can walk around with integration to Skype and not having to pay a penny to these ridiculous carriers, charging insane amounts of monies for browsing, etc. |
| 00:11.27 | Yourname` | I hope eBay agrees to that, so we can all live a nice and free life. :D |
| 00:12.03 | Strom_M | you know, I don't think my carrier charges insane amounts of money for browsing and talking |
| 00:15.43 | *** join/#asterisk pfn (n=pfnguyen@hanhuy.com) |
| 00:17.33 | *** join/#asterisk ScoobyDooby (n=Me@unaffiliated/andshaggytooby) |
| 00:17.56 | jblack | Yourname`: Open access doesn't imply low rates. |
| 00:24.06 | ScoobyDooby | heys gusy I have a quick question...can I route calls by caller id alpha string as in: exten => Bill Collector,1,Hangup() |
| 00:26.43 | jblack | omg. |
| 00:27.03 | jblack | i can't google. |
| 00:34.41 | *** join/#asterisk gerphimum (n=trekkie@cpe-67-11-228-123.satx.res.rr.com) |
| 00:35.26 | *** join/#asterisk gerphimum (i=Trekkie@cpe-67-11-228-123.satx.res.rr.com) |
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| 00:45.01 | *** join/#asterisk lvl- (n=lvl@145.52.248.101) |
| 00:47.05 | Jumpie | sup |
| 00:47.17 | *** join/#asterisk RoyK (n=roy@ip-108-62-149-91.dialup.ice.no) |
| 01:00.58 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
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| 01:22.26 | BBHoss | What would be do if google simply shut down tomorrow, and never came back again? |
| 01:22.44 | BBHoss | AND wikipedia? |
| 01:25.03 | jblack | BBHoss: I just had something really close to that. |
| 01:25.13 | jblack | I felt like someone had poked my eyes out with sticks |
| 01:25.17 | BBHoss | heh |
| 01:25.46 | BBHoss | i sometimes attempt to use msn live or w/e on clients computers when they dont have FF installed |
| 01:25.50 | BBHoss | FAIL of course |
| 01:26.37 | drmessano | MSN Live doesn't work without FF? |
| 01:28.42 | *** join/#asterisk ZaVoid (n=zavoid@c-67-165-25-195.hsd1.pa.comcast.net) |
| 01:30.01 | jameswf-home | .\/me just watched superhero movie.... Holy crap hat was dumb |
| 01:30.07 | jameswf-home | *that |
| 01:32.35 | BBHoss | no its just the default thing on IE7, and instead of typing google.com, i just accidentally use it thinking it will take me to google |
| 01:32.35 | BBHoss | it blows of course |
| 01:32.38 | BBHoss | especially when loading over RDP, because of all the bullshit ads |
| 01:35.44 | Jumpie | jameswf-home lol i thought it was funny |
| 01:35.53 | Jumpie | supposed to be like that all spoofs are |
| 01:43.15 | Qwell | jameswf-home: holy crap hat? |
| 01:44.28 | Jumpie | <jameswf-home> *that |
| 01:49.17 | *** join/#asterisk bkruse (n=kruz@76.73.154.120) |
| 01:49.17 | *** mode/#asterisk [+o bkruse] by ChanServ |
| 01:50.09 | bkruse | anyone have an asterisk box with manager I can use? |
| 01:50.25 | bkruse | I do not have to have any permissions or anything, just testing (and I am not around any :/ ) |
| 01:51.59 | jameswf-home | bkruse: thats what vmware is for... |
| 01:52.04 | jameswf-home | ~developers |
| 01:52.05 | jbot | developers is probably http://www.youtube.com/watch?v=KMU0tzLwhbE |
| 01:52.16 | bkruse | jameswf-home: I completely realize this |
| 01:52.34 | jameswf-home | :) I like to play captain obvious... |
| 01:52.37 | bkruse | but I am on my friends laptop, no need for that hassel (besides, if I was on my laptop, I would be through the VPN) |
| 01:53.02 | lirakis | does anyone have info on getting grandstream phones to reboot with sip notify ? |
| 01:53.07 | jameswf-home | needs to find dinner.... |
| 01:53.27 | lirakis | .. im finding .. virtually nothing on "the internets" and "the googles" |
| 01:54.49 | *** join/#asterisk ew01f (n=angomg--@201.170.1.55.dsl.dyn.telnor.net) |
| 01:56.38 | TJNII | Is there an easy way to set the moh class for all calls on a specific IAX trunk, or do I need to do it in the dialplan? |
| 02:00.25 | *** join/#asterisk gerphimum (i=Trekkie@cpe-67-11-228-123.satx.res.rr.com) |
| 02:02.08 | TJNII | supposes an exten _X.,1,SetMusicOnHold(class) / exten _X.,2,goto(blah) would work, it just seems.... hackish. |
| 02:03.50 | file | TJNII: that depends, clarify what you mean a bit more |
| 02:04.37 | file | like SIP/potato calls IAX2/myprovider/18005551212, SIP/potato places the call on hold, and you want to make sure IAX2/myprovider/etc always gets the same musicclass? |
| 02:04.39 | Jumpie | TJNII you want a specific music file for all incoming calls on X trunk? |
| 02:05.33 | Jumpie | oh |
| 02:05.45 | TJNII | Specific music class for X trunk |
| 02:06.01 | TJNII | default class for all others |
| 02:06.46 | *** part/#asterisk lirakis (n=lirakis@cpe-68-175-38-65.nyc.res.rr.com) |
| 02:06.46 | TJNII | so if IAX/user1 calls me and I put them on hold, I want them to hear a specific hold class |
| 02:07.09 | TJNII | but if IAX/user2, or anyone else, calls me they get the default |
| 02:10.49 | file | I think... I know a way |
| 02:11.56 | file | setvar=CHANNEL(musicclass)=music class here |
| 02:11.58 | file | in the iax.conf entry |
| 02:12.04 | file | *might* do it |
| 02:13.42 | TJNII | hmmm |
| 02:13.58 | Mavvie | let's see if 1.4.19 will run here without causing too much hassle tomorrow morning :-) |
| 02:14.06 | file | 1.4 and higher... I purged 1.2 knowledge |
| 02:15.55 | *** join/#asterisk outtolunc (n=me@c-67-174-216-60.hsd1.ca.comcast.net) |
| 02:19.32 | jameswf-home | poo |
| 02:19.49 | jameswf-home | file == file 2nd edition |
| 02:20.28 | file | is kept in subversion though, so you could check out a revision that has 1.2 capabilities |
| 02:21.00 | TJNII | file: That seems to work |
| 02:21.20 | jameswf-home | file r218 doesn't like the new API they depricated everything bahhh |
| 02:21.42 | file | jameswf-home: oh snap |
| 02:21.49 | file | TJNII: nifty |
| 02:24.14 | *** part/#asterisk comprookie2000 (n=david@adsl-065-012-210-216.sip.bct.bellsouth.net) |
| 02:24.43 | jameswf-home | I need to get a couple micro controllers so i can get the coffee extenxion to work... maybe rs232.... |
| 02:26.44 | Jumpie | i may try that too file :D |
| 02:27.13 | Jumpie | jameswf-home, lazy bastard |
| 02:28.07 | *** join/#asterisk hohum (n=dcorbe@166.129.124.206) |
| 02:29.44 | jameswf-home | is always thinking of random things to do with asterisk... I was thinking guitar hero... you have to enter in to the keypad what saydigits() says |
| 02:30.43 | jameswf-home | still working out the logistics... |
| 02:31.06 | Jumpie | i dont think you could escape even the minor latency |
| 02:31.44 | jameswf-home | you just need to put an offset 500ms |
| 02:31.56 | Jumpie | oh i was thinking |
| 02:31.59 | Jumpie | integrate with real guitar hero |
| 02:32.00 | Jumpie | somehow... |
| 02:32.02 | Jumpie | lol |
| 02:32.20 | jameswf-home | nah dtmf songs.... |
| 02:32.26 | Jumpie | lol |
| 02:32.32 | Jumpie | thats the nethernerd realm |
| 02:33.01 | jameswf-home | i should bring back wardialing... |
| 02:37.47 | *** join/#asterisk marc7 (n=marc@64.46.14.64.novuscom.net) |
| 02:43.03 | marc7 | wondering if anyone can help me with a quick call forwarding question. It appears that when I place outbound calls on my Cisco 7960, calls leave from the context defined in my specific sip user... but if I've enabled call forwarding on the 7960... calls bounce back and attempt to leave through the context specified from [general] in sip.conf (example: http://pastebin.ca/973421) -- can anybody explain if I can alter this behavior to leave from a spe |
| 02:43.33 | BBHoss | jameswf-home, the coffee thing would definitely be awesome |
| 02:43.38 | BBHoss | it would be easy to do |
| 02:43.41 | BBHoss | just use X10 |
| 02:44.20 | BBHoss | marc7, could you also post your extensions.conf |
| 02:44.38 | *** join/#asterisk gerphimum (n=trekkie@cpe-70-125-151-117.satx.res.rr.com) |
| 02:46.38 | marc7 | BBHoss: it's like 2,000 lines split up across multiple files. |
| 02:46.49 | jameswf-home | heh phones that arent allowed to make long distance calls.... I am going to set the LD Dial pattern to rick roll em :)) |
| 02:46.51 | marc7 | BBHoss: ot |
| 02:46.54 | marc7 | er |
| 02:47.17 | BBHoss | post the context that the phones are supposed to stay in |
| 02:48.24 | jameswf-home | thinks a 2000 line context is a little overkill :) |
| 02:49.57 | BBHoss | i really wish this adhearsion thing worked a bit better, it seems like a really nice concept |
| 02:50.28 | BBHoss | would minimize the lines of code required, and also let you use ActiveRecord (of rails fame) |
| 02:50.53 | BBHoss | and you could even do loops !!!! :) |
| 02:51.01 | bkruse | BBHoss: I know that guy, he is really cool |
| 02:51.03 | Jumpie | jameswf-home wanna know somethin super geeky and a bit wierd |
| 02:51.07 | bkruse | he will fix anything |
| 02:51.12 | bkruse | you using the latest trunk? |
| 02:51.14 | Jumpie | i actually was able to dial a phone using my own voice to create dtmf |
| 02:51.26 | BBHoss | bkruse, well, the thing really lacking is documentation, last time i checked |
| 02:51.29 | marc7 | BBHoss: http://pastebin.ca/973431 again |
| 02:51.38 | bkruse | BBHoss: True, he is a programmer, and he works alone |
| 02:51.47 | Jumpie | i used a tone generator to get the right freqs and hummed it close enough (i wasnt exact, but apparently class5 switches have degree of error tolerance) |
| 02:51.48 | bkruse | Any particular bugs you can post on the bugtracker |
| 02:52.02 | BBHoss | bkruse, i sent him an email about it and he says that he and someone else are writing a book, so i can at least hope for that |
| 02:52.41 | marc7 | BBHoss: again, ten-digit calls that leave directly from the 'marc' peer go out through Internal->Outbound->SIP Peer |
| 02:53.23 | marc7 | but that 302 that my phone replies back... means the call never really "came from"... it's an inbound call being redirected away |
| 02:53.59 | marc7 | and it just happens to finally work its way into whatever context is defined in sip.conf's [general] section |
| 02:54.30 | BBHoss | it looks like you are calling yourself |
| 02:54.43 | BBHoss | thats the only time i've ever seen loop detected |
| 02:54.52 | marc7 | looks like it, caller ID is deceiving |
| 02:55.04 | marc7 | i've found/replaced the numbers to protect the guilty |
| 02:55.36 | marc7 | the callerid field in sip.conf is genuinely different from the number that my phone is redirecting to |
| 02:55.43 | BBHoss | so what are you trying to dial here, it says marc |
| 02:55.47 | marc7 | right |
| 02:56.22 | marc7 | SIP/marc is a friend in sip.conf... the call reaches the phone, the phone says "nope, not answering that." |
| 02:56.47 | BBHoss | temporarily moved is the response, correct? |
| 02:56.50 | marc7 | right |
| 02:56.58 | BBHoss | and forwarding is not turned on? |
| 02:57.06 | marc7 | *is* turned on, that's what generates that response. |
| 02:57.09 | bkruse | BBHoss: That could be me... :] |
| 02:57.12 | marc7 | dialing the phone works just fine |
| 02:57.30 | marc7 | when call forwarding is on... it redirects into some kind of asterisk no-man's-land |
| 02:58.08 | Jumpie | lol |
| 02:58.37 | BBHoss | marc7, what are you redirecting to? |
| 02:59.14 | BBHoss | bkruse, he told me it was Dave Troy |
| 02:59.44 | *** join/#asterisk coppice (n=chatzill@99.166.17.210.dyn.pacific.net.hk) |
| 02:59.55 | marc7 | 2505551234 |
| 03:01.56 | BBHoss | marc7, is 123.45.67.8 your provider's server, or your server running asterisk? |
| 03:02.03 | marc7 | my server running asterisk |
| 03:02.14 | marc7 | BBHoss: here, let me offer this as well |
| 03:02.24 | marc7 | BBHoss: if I call forward it to something that's in the context specified in sip.conf's [general] section, it works!! |
| 03:02.35 | marc7 | sip.conf's [general] section says context=Distribution |
| 03:02.52 | BBHoss | marc7, right, i'm just trying to figure out why its calling itself |
| 03:03.01 | marc7 | [Distribution] is basically where all inbound calls come in from our SIP peers |
| 03:03.34 | marc7 | I'm trying to call forward to a number that isn't in [Distribution]... but that would be reachable from the context in my [marc] user from sip.conf |
| 03:03.53 | BBHoss | the internal context? |
| 03:03.58 | marc7 | that's right. |
| 03:04.10 | marc7 | _NXX... would catch every number |
| 03:04.22 | marc7 | so here's ultimately what it comes down to |
| 03:04.24 | *** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 03:04.24 | BBHoss | ok and is the 250 number on the asterisk server, or like a cellphone or something? |
| 03:04.29 | marc7 | 250 is a cell phone number |
| 03:04.40 | marc7 | so it's not in [Distribution] |
| 03:04.49 | BBHoss | marc7, and you can dial that number fine from the phone? |
| 03:04.54 | marc7 | yes |
| 03:05.09 | marc7 | but the problem is that the call forward doesn't originate from the phone... so it never leaves from the Internal context |
| 03:05.27 | BBHoss | ahh |
| 03:05.46 | marc7 | the call forward is just a "nope, not going to answer you. try this instead." |
| 03:05.47 | marc7 | anyway |
| 03:05.49 | BBHoss | so maybe asterisk is dumping it to the default |
| 03:05.53 | marc7 | exactly! |
| 03:06.05 | marc7 | i could change [general] in SIP.conf to include a line that says context=Outbound |
| 03:06.19 | BBHoss | yeah but thats a security issue |
| 03:06.23 | marc7 | exactly! :) |
| 03:07.45 | marc7 | any idea what options I have in this situation? |
| 03:07.55 | BBHoss | hmm |
| 03:08.00 | BBHoss | thinking... |
| 03:08.14 | marc7 | thanks for your time... I could've done a better job explaining this |
| 03:10.36 | BBHoss | marc7, looks like you're not the only one having this problem |
| 03:10.42 | BBHoss | no solution yet though |
| 03:10.55 | BBHoss | google "CFwdAll 7960 Asterisk" |
| 03:11.41 | marc7 | now I've recently seen someone call forward to 2505551234@10.1.2.3 |
| 03:12.02 | marc7 | is that an option here? is there any way to create a kind of virtual interface for asterisk? |
| 03:12.26 | BBHoss | like what, XML browser or something? |
| 03:13.50 | BBHoss | marc7, if i were you i would just use a feature code to enable forwarding, and disable the CFwdAll button on the phone |
| 03:14.19 | BBHoss | like *72 or something goes to a macro that forwards calls |
| 03:14.29 | BBHoss | or sets forwarding up |
| 03:16.30 | marc7 | yeah.... s'pose that's right. |
| 03:16.48 | unixdog | http://www.kadiak.org/cam3.jpg |
| 03:16.55 | BBHoss | there has to be a way to make it work correctly though |
| 03:18.02 | BBHoss | unixdog, ? |
| 03:18.34 | unixdog | my friends view from his back porch in Kadiak alaska |
| 03:18.39 | unixdog | live |
| 03:19.14 | BBHoss | heh |
| 03:29.23 | bkruse | BBHoss: oh, very nice |
| 03:30.25 | BBHoss | bkruse, i will definitely look into it again |
| 03:31.05 | *** join/#asterisk mitcheloc (n=mitchelo@216.207.245.1) |
| 03:34.06 | bkruse | sup mitcheloc |
| 03:34.52 | BBHoss | all you huntsville guys? |
| 03:35.12 | bkruse | BBHoss: Nah, just, let's see |
| 03:35.27 | bkruse | me, angler, Qwell (and maybe some that are not op'd) |
| 03:36.19 | BBHoss | well mitcheloc is at apid, figured he might be here too |
| 03:36.37 | bkruse | he is in hsv now, but does not live here |
| 03:36.44 | BBHoss | ahh |
| 03:36.50 | bkruse | :] |
| 03:37.16 | BBHoss | its a pretty nice town, i just wish i could get fiber to my house :) |
| 03:37.50 | bkruse | BBHoss: you can, it is just $$$, verizon FIOS is where it is at |
| 03:38.07 | BBHoss | yeah i doubt we'll ever seen verizon around here |
| 03:38.10 | bkruse | BBHoss: It is an alright town. Engineering/Government contracters, not really that fun |
| 03:38.33 | BBHoss | lack of bars and other nightly entertainment |
| 03:39.18 | bkruse | BBHoss: Well, I am 18, but yes. Not a lot of things to do. It is better than others, and in the top50 places to live for young people. |
| 03:39.22 | bkruse | I do not think it is though :P |
| 03:39.33 | BBHoss | ahh, so you're not in college yet |
| 03:39.34 | file | Huntsvegas! |
| 03:39.40 | BBHoss | you still at westminster? |
| 03:39.47 | bkruse | file: yes :D |
| 03:40.12 | bkruse | file knows the 256 |
| 03:40.21 | bkruse | BBHoss: Yes, graduating in 7 weeks |
| 03:40.44 | file | adds bkruse on Facebook |
| 03:41.26 | bkruse | signs on just to accept |
| 03:41.47 | bkruse | file: how is your weekend? |
| 03:41.53 | file | great |
| 03:42.00 | Qwell | d'oh |
| 03:42.11 | file | cleaned up my desk a bit, got my XM Radio up and going, all is well |
| 03:42.12 | Qwell | I took my keyboard apart, and don't remember what key was next to print screen |
| 03:42.29 | file | I *should* take the garbage out but Alex and I are seeing who gives in first... |
| 03:42.38 | Qwell | 2 choice - scroll lock (no scroll lock light on this kb) or pause/break |
| 03:42.45 | bkruse | file: xm -=rocks=- |
| 03:43.07 | bkruse | Qwell: How often do you use either is the question! |
| 03:43.23 | Qwell | umm, never |
| 03:43.41 | file | I'm amused how they compress the Canadian news channels and stuff down to virtually nothing |
| 03:44.25 | bkruse | file: Right, XM is never regional, so you get all the awesome american stuff *laughs* |
| 03:44.29 | Qwell | (xev to the rescue) |
| 03:45.02 | bkruse | file: you have been accepted |
| 03:45.24 | file | bkruse: lame |
| 03:46.08 | bkruse | file: I know, sorry :/ Did you know nick h is official married?! |
| 03:46.50 | file | bkruse: I did. |
| 03:47.31 | file | and Russelllllllll |
| 03:51.43 | bkruse | file: Totally. I am happy for em |
| 03:52.07 | file | bkruse: when are you getting married? O.O |
| 03:52.25 | coppice | officially marred? is that different from getting a pirate captain to do it? |
| 03:53.06 | bkruse | coppice: You are officially married, do you recommend it? |
| 03:53.07 | bkruse | file: 3 years. |
| 03:54.06 | coppice | well, it has its ups and downs, but I think I should have joined a monestary |
| 03:56.40 | bkruse | coppice: I guess that is a 75% no, :P |
| 03:56.48 | bkruse | you look happy in your pictures with your wife and kids |
| 03:56.53 | bkruse | takes of stalker hat |
| 03:59.52 | *** join/#asterisk stkn__ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 04:05.15 | Qwell | coppice has pictures? O.o |
| 04:06.10 | Qwell | coppice: wow, you're old :p |
| 04:06.36 | coppice | I'm the little mixed kid. My dad is the old white guy |
| 04:08.25 | Qwell | so, here's a useful little tip. removing keys from a keyboard is very painful |
| 04:09.20 | bkruse | coppice: you are the 5ish year old kid in the picture? |
| 04:09.25 | bkruse | interesting :P |
| 04:10.06 | coppice | I think that picture is maybe 4 yeats old. I'm a bit more mature now |
| 04:10.17 | Qwell | mature, or older? |
| 04:10.48 | bkruse | coppice: http://www.coppice.org/steve_cv.html <--your father? |
| 04:11.01 | coppice | yeah. that's Steve Snr |
| 04:11.13 | bkruse | he has some interesting experience, would you recommend living in hong kong? |
| 04:11.47 | coppice | not if you are an engineer. it used to be a great place for engineers, but now its really bad |
| 04:12.27 | bkruse | hsv is, come move and work for digium now |
| 04:12.37 | bkruse | coppice: how old are you now? |
| 04:19.28 | coppice | 4 years older than in that picture |
| 04:19.56 | coppice | if you are in, say, finance, HK is a great place to live |
| 04:45.19 | *** join/#asterisk dacs (n=haiger@unaffiliated/dacs) |
| 04:47.39 | *** join/#asterisk bakermd (n=none@70.91.165.193) |
| 04:48.35 | bakermd | Hey all - when compiling with ODBC - it's --with-odbc= ?? I forget... is it UNIXODBC that goes there or what? I know the docs call for a path, but I know when I have gotten this working in the past it was not a path it was looking for there |
| 04:59.59 | *** join/#asterisk Toerkeium (i=oo@201.216.206.221) |
| 05:09.03 | carrar | bakermd, ./configure --help | grep -i odbc |
| 05:17.30 | cstar707 | anyone can take a poke at sometimes loosing audio on sip extension over VPN ? |
| 05:18.15 | cstar707 | only one side loosing audio asterisk to sip ata |
| 05:19.51 | carrar | pakcetloss |
| 05:20.01 | cstar707 | no |
| 05:20.18 | cstar707 | sometimes right from the start of call no audio |
| 05:20.39 | cstar707 | seems its getting confused where to send to ? |
| 05:21.06 | cstar707 | vpn is linux box ipsec on each side |
| 05:21.29 | cstar707 | all other traffic is fine |
| 05:22.08 | cstar707 | put ata on the asterisk side of vpn and its fine.. i have 2 locations with ata's and they both doing same thing |
| 05:23.50 | carrar | any nat in the path |
| 05:25.16 | cstar707 | there should not be. ? |
| 05:25.44 | carrar | you tell me |
| 05:26.29 | cstar707 | ok linux box vpn ipsec over internet to linux box ipsec |
| 05:26.46 | carrar | any related logs in your linux box that indicate vpn ip/issues |
| 05:27.09 | cstar707 | ip 10.10.1.0/24 network connected to 10.10.10.0/24 network |
| 05:27.31 | cstar707 | 10.10.10.1 = asterisk 10.10.1.22 = sip ata |
| 05:27.45 | *** join/#asterisk PepOSX (n=angeldav@200.93.28.168) |
| 05:29.43 | cstar707 | i can access web gui in ata and ping it etc.. it registers and sometimes make calls ok but then sometimes there is no audio from asterisk to ata but all other audio is fine |
| 05:30.09 | carrar | might check your tcp flow control channels on your linux boxes |
| 05:30.30 | cstar707 | example please ? |
| 05:30.48 | carrar | talk to you admin who setup your ipsec tunnel |
| 05:31.02 | cstar707 | that would be me .? |
| 05:31.32 | cstar707 | this has been puzzling me for a week |
| 05:31.51 | carrar | well everything works fine with out the vpn right |
| 05:32.07 | cstar707 | yes |
| 05:32.18 | *** join/#asterisk csparks (n=csparks@216-55-149-171.dedicated.abac.net) |
| 05:32.40 | carrar | So you need to re-visit your vpn, route tables, any filtering,e tc... |
| 05:33.41 | cstar707 | there is no filtering its a wide open vpn both directions |
| 05:34.07 | csparks | im needing to write a program for windows, that reads calller id and depending on the callers id opens a webpage, is that easily possible? |
| 05:38.20 | carrar | Try using OpenVPN |
| 05:39.22 | cstar707 | i was thinking same thing.. is it more nat compliant ? |
| 05:39.51 | carrar | You said you are not runing nat |
| 05:41.04 | cstar707 | no i have even told iptables to turn off nat over vpn |
| 05:41.48 | cstar707 | but i have heard that ipsec and nat play funny games and thats why i was thinking that open vpn may solve the issue |
| 05:41.56 | bakermd | I just cannot get this to compile with UnixODBC - I have all the dev libraries installed, and have given it the proper path in --with-odbc but when I get to the menuselect the options are unavailable for odbc - any common mistakes? |
| 05:42.27 | carrar | What dir did you install odbc too when you compiled it? |
| 05:47.34 | bakermd | I used yum |
| 05:47.44 | bakermd | sql.h file is in /usr/include |
| 05:49.48 | bakermd | got it |
| 05:49.53 | bakermd | missing libtool devel |
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| 05:52.42 | *** mode/#asterisk [+o jmhunter] by ChanServ |
| 05:53.54 | jmhunter | buenos noches |
| 05:54.07 | drmessano | It's too late for beany nachos |
| 05:54.14 | jmhunter | hehe |
| 06:07.32 | jameswf-home | ~poop |
| 06:07.33 | jbot | ACTION fertilizes the channel |
| 06:08.36 | *** join/#asterisk drako (n=luisjose@nelug/coreteam/luisjose) |
| 06:11.46 | drmessano | my cat LOVES soy milk |
| 06:11.55 | drmessano | hates regular milk |
| 06:12.01 | drmessano | WEIRD---O |
| 06:16.42 | *** join/#asterisk jklehm (n=john@rrcs-67-53-187-205.west.biz.rr.com) |
| 06:16.59 | jklehm | any feelings on digium x100p cards anyone? |
| 06:17.57 | drmessano | x100p is an x100p |
| 06:18.22 | drmessano | Flaky, flaky, flaky |
| 06:18.52 | cstar707 | yes they are very flakey i agree |
| 06:21.29 | jklehm | no wonder they are so affordable =P |
| 06:21.50 | jklehm | no use if they dont work though |
| 06:23.28 | coppice | their price has nothing to do with how well they work. if you find a properly made one, and use it with OSLEC to clean up the echo, they can work well. this assumes you can find someone making them properly, though |
| 06:24.12 | jklehm | how would i tell? |
| 06:24.35 | coppice | there lies the problem :-) |
| 06:25.29 | jklehm | =P |
| 06:25.30 | drmessano | Who makes a proper one? |
| 06:25.41 | jklehm | I see them on ebay for 30$ =P |
| 06:25.55 | coppice | they should be about $5 or so |
| 06:26.06 | jklehm | i havent seen that |
| 06:26.41 | coppice | why would anyone pay $30? for $40 you can get some complete ATAs |
| 06:26.54 | jklehm | im noob what is ata? |
| 06:27.34 | coppice | complete telephone to SIP adaptors, with an RJ45 on them |
| 06:27.50 | fujin | Linksys |
| 06:27.55 | jklehm | sneaky |
| 06:28.12 | drmessano | You can get a grandstream for $40 |
| 06:28.20 | drmessano | Linksys will run you $70+ for FXO |
| 06:28.42 | drmessano | There's such a gap when it comes to FXO hardware |
| 06:28.45 | drmessano | It's dumb |
| 06:28.48 | coppice | a PAP2 is <$50 here, and all our prices are way over the top |
| 06:28.49 | drmessano | No better word for it |
| 06:28.57 | drmessano | That's a PAP2 |
| 06:29.03 | drmessano | 2 FXS |
| 06:29.47 | coppice | yeah, but it shows the oddity of the pricing. 2 x FXS is more expensive to make than 2 x FXO |
| 06:29.50 | jklehm | let me just put this out there I really new to this and just trying to see how it works, I can buy a DID number from a clec for 10$ /month which gives pstn -> ip:port. What service do I need to connect the other way ip:port -> pstn? |
| 06:29.59 | drmessano | They could probably get by selling X100Ps for $40 with the gap |
| 06:30.03 | ManxPower | The X100P was a fluke - a winmodem so dumb you had to do EVERYTHING in your driver. |
| 06:30.37 | coppice | its interesting that nobody has made any of the other winmodems work with * |
| 06:30.41 | ManxPower | and I'll bet the interface was either documented or so simple mark figured out what he could do with it. |
| 06:31.07 | ManxPower | coppice: the people with the skill have the money to buy quality hardware. The people with out the money don't have the skills |
| 06:31.11 | coppice | these days you can find source code for any of the winmodems |
| 06:31.17 | Qwell | coppice: there was chan_modem, but...meh |
| 06:31.37 | drmessano | It amazes me that no one has come up with a dumber board and everything done in DSP |
| 06:31.43 | coppice | chan_modem was trying to drive a real modem. that's quite different, and very very limited |
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| 06:31.53 | Qwell | ahh, not winmodems? |
| 06:32.06 | Qwell | I never used it, just heard it was pretty terrible (thus why it was removed) |
| 06:32.09 | drmessano | $5 board with a couple RJ-11s and some discreet components, and rest done in software |
| 06:32.40 | adorah | How about Atcom generic zaptel boards? any good? |
| 06:33.03 | coppice | the winmodem code for most modems has a binary DSP chunk, and the source code for the kernel bit that interfaces to the card. i.e. most winmodems now have basic audio driver source code available |
| 06:33.39 | Qwell | I'm surprised nobody has made an AMR fxo card, heh |
| 06:33.40 | tzafrir | some use code from ALSA |
| 06:34.01 | adorah | But basically, the Tiger chipset on which TDM400p is based is originally a winmodem.. |
| 06:34.02 | coppice | who has an AMR slot? |
| 06:34.12 | Qwell | coppice: who has an ISA slot? |
| 06:34.13 | drmessano | LOL |
| 06:34.19 | drmessano | AMR was great.. |
| 06:34.19 | Qwell | people still try to make ISA hardware |
| 06:34.28 | coppice | tzafrir: I think you mean they provide code for ALSA |
| 06:34.31 | drmessano | Lets give modems their own interface just in time for their mass extinction |
| 06:34.37 | drmessano | Way to go, manufacturers |
| 06:34.44 | Qwell | drmessano: heh |
| 06:35.01 | tzafrir | coppice, slmodem can use ALSA for some modems rather than a binary module |
| 06:35.26 | drmessano | I saw a US Robotics modem Advert the other day.. they're now selling them as "backups for your broadband" |
| 06:35.45 | drmessano | Since they're no longer primary use devices |
| 06:35.46 | Qwell | USR still exists? |
| 06:35.53 | drmessano | and soon to be no longer backup devices |
| 06:35.56 | drmessano | Oh yes |
| 06:35.57 | coppice | tzafrir: for what we are talking about here, the code they provide to link their card into the ALSA system is what is relevant |
| 06:36.05 | drmessano | I have no effin idea how |
| 06:36.15 | drmessano | I guess theres enough NetZero users left |
| 06:36.36 | tzafrir | coppice, David Rowe once mentioned an AMR-based design |
| 06:36.56 | tzafrir | a slightly non-standard one, though |
| 06:36.57 | coppice | I haven't seen an AMR slot in 6 or 7 years |
| 06:37.01 | adorah | US Robotics belogns now to 3COM I think.. |
| 06:37.07 | Qwell | adorah: makes sense |
| 06:37.12 | tzafrir | hmm.. I thought you meant the chip |
| 06:37.15 | drmessano | Oh god |
| 06:37.19 | coppice | 3COM sold USR |
| 06:37.34 | drmessano | USR was owned by 3COM years ago |
| 06:37.43 | drmessano | "now" is a bit dated loil |
| 06:37.45 | Qwell | read that as USSR O.o |
| 06:37.46 | drmessano | -i |
| 06:38.03 | Qwell | "3Com defeated USSR?" |
| 06:38.10 | tzafrir | R and A are quite similar in my font |
| 06:38.11 | Qwell | it's clearly late |
| 06:38.22 | adorah | 3COM mostly defeats it self..hehe |
| 06:38.35 | drmessano | US Robotics Modems: "When your broadband fails.. oh wait, you don't have a POTS line anymore.... eh, nevermind" |
| 06:38.40 | coppice | for national security reasons 3COM cannot be bought by HuaWei. It must be alowed to die, instead :-) |
| 06:39.13 | adorah | That was a great deal for Huawei and 3COM.. |
| 06:39.16 | Qwell | coppice: that was silly |
| 06:39.23 | JT | doesn't HuaWei translate to CiscoCopy? |
| 06:39.33 | coppice | er, no |
| 06:39.38 | adorah | coz all that left from 3COM is an Int marketing force.. |
| 06:39.55 | Qwell | "You used to know a guy who once was suspected of committing a crime? Nah, you can't buy that company." |
| 06:40.18 | drmessano | I can't wait for V.94 modems |
| 06:40.33 | drmessano | Gonna blow away V.92 |
| 06:40.40 | coppice | I was amazed anyone bothered cooking up V.92 |
| 06:40.57 | Qwell | nub question - what is v.92 supposed to do? |
| 06:41.00 | drmessano | coppice: They needed to sell ONE more generation of modems |
| 06:41.19 | coppice | V.92 speeds up the back channel a bit |
| 06:41.27 | Qwell | shrugs |
| 06:41.57 | coppice | and has another compression option which works better - V.44, I think |
| 06:41.58 | drmessano | Extended sending from 31kbps to 48kbps |
| 06:42.01 | JT | man |
| 06:42.11 | JT | if the time of dialup was now, i'd be set |
| 06:42.14 | drmessano | v.90 was limited to 31.2 |
| 06:42.23 | JT | i have about 120 ports worth of V.90 gear sitting here |
| 06:42.31 | JT | and 60 ports of 33.6 |
| 06:42.43 | coppice | V.90 should have run at 33.6k, as its back channel was a V.34 one |
| 06:43.03 | cstar707 | can they be used over asterisk ? |
| 06:43.12 | JT | no |
| 06:43.47 | cstar707 | can you do any sort of dialup over sip or iax |
| 06:43.59 | drmessano | I'm not sure why or how, but V.90 ran up to 31,200 up |
| 06:44.04 | drmessano | no |
| 06:44.10 | drmessano | Well, 2400 maybe |
| 06:44.26 | drmessano | I was told someone ran 9600 over asterisk using ATAs |
| 06:44.33 | drmessano | I never got it to work |
| 06:44.40 | cstar707 | so even a virtual pri into a pri USR box ? |
| 06:44.41 | JT | at least 9600 is definitely doable |
| 06:44.54 | JT | i know of sites with ATAs for faxes |
| 06:45.46 | coppice | FAX is one way at a time, and does not use EC. data modem are duplex and rely on EC. Generally, EC won't work over a VoIP path |
| 06:45.54 | adorah | This channel sounds like a telco history channel now... |
| 06:46.31 | cstar707 | dam an internet provider wanted to kill his analog lines and switch to over internet lines guess it can not be done ? |
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| 06:48.02 | drmessano | lol |
| 06:48.08 | drmessano | That's an insane idea |
| 06:48.18 | JT | cstar707: ERR |
| 06:48.29 | JT | cstar707: what Internet providers still use analogue lines? |
| 06:49.22 | drmessano | You can get 256k DSL for $20 a month, with no extra phone line to pay for |
| 06:49.24 | drmessano | Dialup is dead |
| 06:49.44 | adorah | dial-up gives you a dedicated 64kbs upload..excellent for voip connection.. |
| 06:50.28 | JT | adorah: ISDN dialup perhaps |
| 06:50.46 | drmessano | Why the hell would you do that? |
| 06:50.50 | JT | cstar707: ISPs moved to digital PRIs almost a decade ago |
| 06:50.58 | JT | drmessano: available everywhere, almost |
| 06:51.10 | adorah | and you need only 18-32kbs depends on the codec.. |
| 06:51.41 | JT | adorah: if you're talking about voip over dialup, you're forgetting the horrible latency |
| 06:52.30 | drmessano | Most telco's limit their dialup connections to 28.8K anyway |
| 06:52.39 | adorah | A friend on mine worked in sth america and he used 2 softphones with iax/gsm to connect thru my machine at the same time..sound was ok.. |
| 06:52.54 | JT | adorah: okay is about as good as it gets imho |
| 06:53.42 | drmessano | But you had to pay for a line.. and the ISP.. unless the telco was providing the data |
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| 06:55.55 | adorah | Re |
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| 07:06.38 | coppice | JT: if latency is bad enough, its natural to forget things |
| 07:06.54 | JT | heh |
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| 07:56.09 | IguanaNed | anyone help me with PRI info? not technical info just basic pricing |
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| 08:01.12 | cmantito | can sell you a data PRI ;) |
| 08:02.31 | JT | that wouldn't be a pri would it? |
| 08:02.42 | cmantito | sshhh :P |
| 08:02.52 | cmantito | provider's words, not mine ;p |
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| 08:03.28 | cmantito | speakeasy *insists* on calling them data PRIs on their partner docs |
| 08:03.38 | srt | hi, anybody knows if there is an implementation of the asterisk speech api that uses loquenda as engine? |
| 08:17.07 | IguanaNed | can someone hook me up with some info on how PRI pricing works? and what is avtuall the benefit of gettin PRI over say virtual PRI? |
| 08:19.11 | tzafrir | IguanaNed, not me. But it helps to mention where (geographically) |
| 08:19.47 | IguanaNed | ah thnx... Toronto, Ontario |
| 08:20.01 | JT | canada |
| 08:20.04 | IguanaNed | I know there are a lot of options |
| 08:20.41 | IguanaNed | currently I am putting alop of minutes thru an itsp si[ termination |
| 08:21.08 | IguanaNed | trying to figure out if a PRI would be beficial |
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| 08:26.53 | BBHoss | IguanaNed, in the US, they charge you a local loop fee +plus some other fees for unlimited local, then they charge you a low rate for LD |
| 08:39.46 | IguanaNed | BBHoss thnx |
| 08:40.40 | IguanaNed | could you point me to a a paricular one.. perferably one that post rates.. all the ones I see ask you to call for quote |
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| 09:56.37 | sw | hi, which module is responsible for the "register" function.. i'm trying to slim down my asterisk but the ITSP won't register |
| 09:57.00 | sw | so i removed a lot of modules, and I guess i removed the one managing registrations |
| 10:04.10 | xacatecas | sw, afaik sip and iax2 handles their own. i could be wrong. also, you may need the user module (but once again, i'm also pretty new to this so i may be wrong) |
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| 10:24.40 | Hyphenex | Hey, what's the technology called that sends key presses over VoIP? |
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| 10:41.49 | tzafrir | Hyphenex, tthere are several of them |
| 10:42.16 | Hyphenex | tzafrir: oahh, I was thinking of DTMF tones thanks :) |
| 10:42.45 | tzafrir | that's really not VoIP-specific . And rather discourged in voip, actually |
| 10:43.04 | Hyphenex | ahhk |
| 10:43.08 | tzafrir | With the optional compression you might mis-detect digits |
| 10:43.25 | tzafrir | Thus people tend to try to send digits as signalling information |
| 10:45.42 | coppice | there's a strange thing about that. people usually get hopeless results when trying to detect DTMF through a G.729 VoIP path. However, when I test my DTMF decoder through a G.729 encode and decode it won't fully meet all the performance requirements, but it gives only a small number of bad digits. |
| 10:46.28 | tzafrir | not all dtmf ddetectors are the same? |
| 10:46.57 | tzafrir | Anyway, a small number of bad digits is bad enough |
| 10:47.19 | coppice | well, that's true, but people get poor results with the DTMF decoder in *, which is basically mine minus a few performance improvements |
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| 10:48.37 | coppice | I mean really small, as long as the tones are on target. the errors are mostly when the generated signal is pushed to the specified limits |
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| 12:42.16 | stony | hi |
| 12:42.35 | stony | can i activate/deactivate the blindxfer feature from dialplan |
| 12:42.38 | stony | ? |
| 12:44.00 | stony | i'm not talking about the t option of the dial() cmd |
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| 13:09.22 | pa | do you think it's possible to use Asterisk Realtime and keep Sip accounts and Iax accounts in one table? |
| 13:09.29 | pa | after all |
| 13:09.38 | pa | fields with the same name have same purpose |
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| 13:12.40 | mvanbaak | tzafrir: you know if ztdummy can run inside xen ? |
| 13:13.55 | pa | where can i find an explaination for each field i should put in sip or iax buddies entry? |
| 13:14.29 | tzafrir | mvanbaak, not really sure |
| 13:16.44 | mvanbaak | hhmm, I found a closed mantis ticket, going to try that |
| 13:20.35 | mvanbaak | Span 1: ZTDUMMY/1 "ZTDUMMY/1 (source: Linux26) 1" (MASTER) |
| 13:20.36 | mvanbaak | gheh |
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| 13:22.04 | Dovid | anyone here work with PHP + AGI ? |
| 13:43.06 | tzafrir | mvanbaak, yeah, but what does zttest give you? |
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| 13:46.17 | mvanbaak | Best: 99.905 -- Worst: 96.874 -- Average: 99.459537, Difference: 100.540464 |
| 13:53.45 | Dovid | anyone here work with PHP + AGI ? |
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| 14:06.58 | vadim1024 | i'm using regster => in my sip.conf, is there a way to control dialplan context to whih incoming call are routed |
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| 14:26.09 | Dovid | vadim1024: Yes. You can set the context for the peer in sip.conf |
| 14:26.31 | Dovid | or u can use register => some_uid:some_pass@some_IP/some_extension |
| 14:26.53 | Dovid | some_extension in ur default context and route the call apropiately |
| 14:34.55 | *** join/#asterisk dcmwai (n=dcmwai@192.228.184.36) |
| 14:55.52 | *** join/#asterisk UnixDog (n=UnixDog@ppp-69-239-49-62.dsl.irvnca.pacbell.net) |
| 15:04.11 | *** join/#asterisk ZaVoid (n=zavoid@c-67-165-25-195.hsd1.pa.comcast.net) |
| 15:14.11 | *** join/#asterisk CVirus (n=GoD@82.201.178.99) |
| 15:22.23 | *** join/#asterisk dacs (n=haiger@unaffiliated/dacs) |
| 15:23.45 | *** part/#asterisk dacs (n=haiger@unaffiliated/dacs) |
| 15:24.29 | *** join/#asterisk jorgeraidel2000 (n=jorgerai@c-66-176-203-45.hsd1.fl.comcast.net) |
| 15:24.33 | jorgeraidel2000 | hola |
| 15:24.39 | *** join/#asterisk SteveTotaro (n=Administ@pool-151-196-236-69.balt.east.verizon.net) |
| 15:24.40 | jorgeraidel2000 | necesito ayuda caballero |
| 15:24.42 | jorgeraidel2000 | :( |
| 15:25.03 | jorgeraidel2000 | ayuda no , alguien disponible y le apgo pro que me configuren algo |
| 15:25.04 | jorgeraidel2000 | :( |
| 15:25.18 | mvanbaak | English please |
| 15:25.25 | jorgeraidel2000 | my english is not good |
| 15:25.32 | jorgeraidel2000 | jej i will try |
| 15:25.46 | mvanbaak | we wont be able to help you if your questions are not in english |
| 15:26.01 | jorgeraidel2000 | wow |
| 15:26.21 | jorgeraidel2000 | in fact i need someone for paid that set some configuration in my asterisk |
| 15:27.47 | jorgeraidel2000 | hi |
| 15:28.17 | jorgeraidel2000 | nobody? |
| 15:28.18 | jorgeraidel2000 | :( |
| 15:32.02 | *** join/#asterisk tobias (n=tobias@cpe-066-026-085-055.nc.res.rr.com) |
| 15:33.28 | Dovid | ENGLISH !!!!!!!!!!!!! |
| 15:33.37 | mvanbaak | lol |
| 15:36.25 | lesouvage | my features and applicationmap items only work when I use one digit like # or *. If I set #1 or #2 it doesn't work nothing happens. Do I have to set some timeout variable or do something else to have *1 or #2 to work. |
| 15:39.31 | lesouvage | mvanbaak: how are you doing |
| 15:41.51 | *** join/#asterisk axisys (n=axisys@ip70-174-179-120.dc.dc.cox.net) |
| 15:43.08 | *** join/#asterisk nighty^ (n=nighty@p1022-adsau16honb13-acca.tokyo.ocn.ne.jp) |
| 15:43.38 | mvanbaak | lesouvage: not bad, thank you |
| 15:44.39 | *** join/#asterisk ming_zym (n=ming_zym@f0-0-tep-rtr1.corp.cnb.yahoo.com) |
| 15:49.40 | lesouvage | We had dinner in Utrecht a while ago |
| 15:51.04 | mvanbaak | ah :) |
| 15:52.53 | mvanbaak | lesouvage: what version of asterisk are you running ? |
| 15:53.13 | *** join/#asterisk ZaVoid (n=zavoid@c-67-165-25-195.hsd1.pa.comcast.net) |
| 15:53.25 | mvanbaak | the applicationmap stuff should work with multidigit codes |
| 15:54.28 | lesouvage | mvanbaak: 1.4.18.1 |
| 15:55.47 | mvanbaak | are you sure asterisk is still in the mediapath when trying the features ? |
| 15:56.27 | mvanbaak | hhmm, must be because it's working with single character codes |
| 15:56.38 | *** join/#asterisk flush (n=SYN_SENT@ip216-239-68-174.vif.net) |
| 15:56.40 | mvanbaak | let me test |
| 15:57.25 | lesouvage | mvanbaak: Yes, the one digit feautures are working, de multi digits aren't. Packet2Packet bridging SIP/lesouvage-086d5e38 and SIP/312xxxxxxxx-086d47d8 is the message in the cli |
| 15:59.30 | mvanbaak | hhmm, it works fine here |
| 15:59.59 | flush | yo |
| 16:00.03 | flush | i have this |
| 16:00.18 | mvanbaak | lesouvage: but I dont have sip to test it. I'm using iax and skinny |
| 16:00.48 | flush | exten => _9NXXXXXXXXX,1,Dial(${OUTBOUNDTRUNK}/${EXTEN:1},,m) |
| 16:01.09 | flush | so when i call, i first hear my music on hold but it only play for like 2 sec, then after i hear the ringing tone.. |
| 16:01.15 | flush | why doesnt it play music until someone answers ?? |
| 16:02.12 | mvanbaak | flush: maybe your outbound provider only allows 2 seconds of early media |
| 16:03.00 | flush | you mean bell canada is totally full of shit |
| 16:03.11 | flush | and i cant do anything ? |
| 16:03.34 | mvanbaak | no idea |
| 16:04.30 | *** join/#asterisk ctaloi (n=ctaloi@pool-72-90-82-206.syrcny.fios.verizon.net) |
| 16:06.51 | *** join/#asterisk bintut (n=bintut@cm85.omega113.maxonline.com.sg) |
| 16:10.54 | *** join/#asterisk adjohn (n=adjohn@dhcp-172-205-57-69.cf-res.cfu.net) |
| 16:12.52 | Dovid | anyone here work with PHP + AGI ? |
| 16:17.00 | *** join/#asterisk af_ (n=getsmart@88-149-240-26.dynamic.ngi.it) |
| 16:18.27 | JayTee52 | flush, what kind of phones are you using? |
| 16:19.41 | davevg-btwtech | Dovid, I don't use php, but maybe i can help answer your question |
| 16:20.25 | mvanbaak | Dovid: I do php dev |
| 16:22.07 | *** join/#asterisk outtolunc (n=me@c-67-174-216-60.hsd1.ca.comcast.net) |
| 16:23.12 | Jumpie | argh |
| 16:23.41 | Jumpie | no matter what i do, when i set an extension to call an outbound number, it sends the local extension, even though i have set the callerid different |
| 16:23.59 | *** join/#asterisk Rico29 (n=Rico@ARennes-358-1-62-180.w90-49.abo.wanadoo.fr) |
| 16:24.44 | flush | JayTee52 i use the pots |
| 16:24.49 | flush | normal analog phones |
| 16:25.00 | JayTee52 | what ATA's? |
| 16:25.28 | Rico29 | hi |
| 16:25.35 | flush | no ATA |
| 16:25.40 | flush | i have no voip |
| 16:25.54 | flush | a normal analog phone line with bell canada |
| 16:26.02 | JayTee52 | oh, then I can't help you. |
| 16:26.08 | flush | :( |
| 16:26.17 | flush | if i had voip, cause ill get it by july when i move |
| 16:26.20 | JayTee52 | I thought you were running Asterisk |
| 16:26.22 | flush | with voip, it would work good "? |
| 16:26.28 | flush | i am running asterisk... |
| 16:26.35 | flush | only i use normal analog phones |
| 16:26.37 | JayTee52 | now you're confusing me |
| 16:26.43 | flush | i have no voip service |
| 16:26.50 | flush | well goes like this |
| 16:26.55 | JayTee52 | you use pots to connect to Bell Canada. |
| 16:27.12 | flush | i have tdm400p card, bell line is plugged in the fxo module and my other phone are plugged in the fxs module of the card |
| 16:27.12 | riddlebox | flush, all tdm cards? |
| 16:27.14 | JayTee52 | you have an FXS card in your Asterisk box |
| 16:27.17 | Jumpie | its sending my 2000 extension instead of my caller id |
| 16:27.24 | flush | i have tdm400p with 1 fxo and 3 fxs |
| 16:27.38 | JayTee52 | ok |
| 16:28.10 | Jumpie | call me noobish but |
| 16:28.14 | flush | it all works good, music on hold (call parking) and all things i set up in extensions.conf is okay and functionnal.. but when i call i hear the music only like 2 or 3 secondes and then it stops playing, and starts rigning in the phone |
| 16:28.16 | Jumpie | what exactly is the reasoning for fxs? vs fxo |
| 16:28.28 | *** join/#asterisk glaz (i=strke@glaciuz.com) |
| 16:28.42 | JayTee52 | Jumpie, FXS is for phones, FXO is for trunks from your PSTN |
| 16:28.47 | riddlebox | ~thebook |
| 16:28.48 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com |
| 16:29.02 | riddlebox | Jumpie, read that |
| 16:29.21 | Jumpie | riddlebox i have it |
| 16:29.22 | Jumpie | lol |
| 16:29.25 | Jumpie | i just havent gotten it printed |
| 16:29.28 | Jumpie | i can only read so much pdf |
| 16:29.38 | JayTee52 | Foreign Exchange Office and Foreign Exchange Station |
| 16:29.53 | Jumpie | so a fax could go...? |
| 16:30.02 | Jumpie | fxs? |
| 16:30.06 | JayTee52 | i hear ya, I have the PDF on my desktop but I just bought the printed version cause I hate trees |
| 16:30.27 | riddlebox | yes jumpie, a fax can be on fxs, |
| 16:30.34 | JayTee52 | you could send a fax out your fxs port to a fax machine. |
| 16:30.50 | Jumpie | wait..so i think it depends doesnt it? |
| 16:31.01 | Jumpie | a pots line fax would go directly to a fxo? |
| 16:31.07 | Jumpie | but a voip fax would go on fxs? maybe im confused |
| 16:31.13 | pa | excuse me |
| 16:31.19 | riddlebox | Jumpie, fxo=line from phone company |
| 16:31.19 | JayTee52 | but you'd have no way of determining whether your incoming call is a fax or not. How are you routing calls from a single FXO port to your extensions? |
| 16:31.29 | riddlebox | Jumpie, fxs=standard telephone |
| 16:31.38 | pa | anyone uses odbc with postgresql? i dont have the odbc.sql to load into postgres.... |
| 16:31.40 | Jumpie | JayTee52 i thought there was ways to find out if its a fax or not/ |
| 16:31.43 | pa | this file: |
| 16:31.47 | Jumpie | riddlebox...hmm ok |
| 16:31.54 | pa | /usr/share/psqlodbc/odbc.sql |
| 16:31.55 | outtolunc | should build the asterisk gospel channel 24/7/365 tts spewage of ~thebook (or its creators could) |
| 16:32.04 | Jumpie | there are ways to dial a fax like a voip extension |
| 16:32.34 | Jumpie | and then have variable bit rate so you can have multiple pages without issues |
| 16:33.17 | JayTee52 | I have a fax on an Grandstream HandyTone 286 ATA but it's incoming calls have a dedicated FXO port on the Asterisk server. |
| 16:33.39 | JayTee52 | so I'm not sure how you'd do fax detect in Asterisk and then route the call in the Dailplan |
| 16:33.40 | riddlebox | Jumpie, those are not reliable |
| 16:33.50 | Jumpie | riddlebox supposedly you can jury rig it to make it reliable :D |
| 16:34.01 | Jumpie | JayTee52 i dunno ffhand, but a friend of mine that runs an isp seems to have a way |
| 16:34.03 | Jumpie | ill have to ask him |
| 16:34.45 | riddlebox | JayTee52, a voip-ata-fax solution is not going to be reliable, and if you want fax detection I believe there is a package you can install to do it |
| 16:35.14 | JayTee52 | riddlebox, yeah, Grandstream equipment in general is not very reliable and the HandyTone can be a pain but I have it working at them moment so who cares? |
| 16:35.36 | riddlebox | JayTee52, you can send and receive faxes on it? |
| 16:35.58 | JayTee52 | riddlebox, yes I can |
| 16:36.12 | riddlebox | JayTee52, can you fax multiple pages? |
| 16:36.22 | JayTee52 | yes |
| 16:36.59 | Jumpie | variable bit rate adaptation |
| 16:37.20 | Jumpie | i dont wanna have it to be mission critical riddlebox |
| 16:37.23 | Jumpie | just want it to work :D |
| 16:37.32 | riddlebox | JayTee52, I wish I could do that but I have issues using my charter telephone line with modems and faxes(even though they say its my equipment) |
| 16:37.34 | outtolunc | but he isn't chewing gum also |
| 16:37.55 | JayTee52 | riddlebox, are you using the HandyTone as well? |
| 16:37.58 | riddlebox | Jumpie, the problem is when you need to send an important fax and it doesnt work then what? |
| 16:38.01 | JayTee52 | or trying to? |
| 16:38.29 | riddlebox | JayTee52, I have a handytone 488, for a cordless phone, I just bought a fax switch and put that in front of asterisk |
| 16:38.43 | Jumpie | lol |
| 16:38.44 | Jumpie | go on pots |
| 16:38.48 | JayTee52 | ah, I'm using a HandyTone just for the fax only |
| 16:39.02 | *** join/#asterisk ZaVoid (n=zavoid@c-67-165-25-195.hsd1.pa.comcast.net) |
| 16:39.02 | riddlebox | I have a spare sipura box I could use |
| 16:40.07 | JayTee52 | everyone in here that seems to know Asterisk well all say get fax off your Asterisk box and use pots for it. Not an elegant solution for some but I tend to agree. Faxes should just frakken go away. This is the 'oughts not the '90s. |
| 16:42.17 | Jumpie | JayTee52 i agree in fact |
| 16:42.22 | Jumpie | i'd rather have fax right to desktop |
| 16:42.25 | Jumpie | most are doing that |
| 16:42.32 | Jumpie | extension is mapped to your phone extension or osmethin and you get pdfs |
| 16:43.12 | *** join/#asterisk gego (n=gego@dyndsl-085-016-200-060.ewe-ip-backbone.de) |
| 16:43.59 | JayTee52 | even with a decent telco provider and standard POTS with a good fax machine try sending a 50 page fax and you can still have problems. |
| 16:44.12 | Jumpie | well if you send 50 page fax your an idiot anyway |
| 16:44.14 | Jumpie | lol |
| 16:44.15 | lesouvage | JayTee52: I agree, in tenders there are pages written about fax: fax to fax, fax to e-mail, e-mail to fax and everything elsee. like this are the most exciting features for a PBX. |
| 16:44.21 | JayTee52 | no arguement there :-) |
| 16:45.07 | JayTee52 | I have some fax lines programmed in my Nortel Meridian Option 11c but most fax lines at my work are pots. |
| 16:45.25 | *** join/#asterisk glaz (i=strke@glaciuz.com) |
| 16:46.18 | riddlebox | option 11 ewwww nasty programming LD 21..... |
| 16:46.26 | JayTee52 | lol |
| 16:47.04 | JayTee52 | or LD 10 or LD 11 or LD 23......ewwwww LD 23 "Help me, Obiwan! Your my only hope!" |
| 16:48.09 | JayTee52 | I have 8 analog extensions running into 2 4 port TDM04B cards on one of my Asterisk servers. RPITA. |
| 16:49.25 | JayTee52 | going to try to make Asterisk the "man-in-the-middle" between my telco's PRIs and the Option 11c using 2 of Digium's TE212P cards. |
| 16:51.05 | Jumpie | nerdz |
| 16:51.13 | *** join/#asterisk Skarmeth (n=Skarmeth@iris.aspec.com.br) |
| 16:51.15 | JayTee52 | but currently I'm just using Asterisk as a VOIP bridge for extensions on the Option 11 to SIP phones in a building off our main campus that we couldn't run copper to and my boss is too cheap to buy a fiber channel bank. |
| 16:51.24 | Skarmeth | hi all |
| 16:51.27 | JayTee52 | hi |
| 16:52.47 | Skarmeth | I had installed a new TE220B card on my Asterisk system to replace a TE110P and after configurinf zaptel.conf and zapata.conf it don't go green... |
| 16:53.42 | Skarmeth | My cables are working well (it works with TE110P)... any clues about TE220B install? I am using wct4xxp kernel module and zaptel 1.4.9 |
| 16:59.55 | JayTee52 | are these T1 or E1 circuits? |
| 17:06.15 | Skarmeth | E1 |
| 17:06.34 | Skarmeth | and the two jumpers/straps are connected to PIN 1-2, 3-4 |
| 17:06.50 | Skarmeth | (the manual aren't clear) |
| 17:08.03 | Skarmeth | I just enabled first span (card 0 span 2) as span=2,1,0,ccs,hdb3,crc4 \n bchan= 5-19,21-35 \n dchan=20 |
| 17:08.08 | *** join/#asterisk ManxPower (n=manxpowe@168.sub-70-220-199.myvzw.com) |
| 17:08.55 | Skarmeth | and my zapata.conf was configured to run as PRI CPE with EuroISDN for channels 5-19,21-35 |
| 17:10.43 | JayTee52 | Skarmeth, if you run lspci does it see the card? does lsmod show the wct4xxp module loading? |
| 17:12.04 | lesouvage | I want to add an announcement when # is pressed to start/stop automon (something like "after the beep the recording starts") Is there an easy way (extra parameter or something like that) available to add an announcement. |
| 17:12.07 | Skarmeth | JayTee52, I get /proc/zaptel/{1,2,3} information interfaces loadeds |
| 17:12.29 | Skarmeth | I have the modules (zaptel, wctdm, wct4xxp) up and running |
| 17:13.09 | JayTee52 | what do you get when you run zttool? |
| 17:13.25 | Skarmeth | Status: Provisioned, In Alarm, Down, Active |
| 17:14.03 | lesouvage | with a macro started from applicationmaps I can do the announcement, but now I have to figure out how to actually start en stop the recording from within the macro. |
| 17:14.06 | JayTee52 | did you use the info from the previous zaptel.conf and zapata.conf files? |
| 17:14.25 | Jumpie | you can buy partial pri right? |
| 17:14.58 | JayTee52 | at todays prices why would you want to? |
| 17:15.07 | Skarmeth | for my tdm card ok, for my te220b card on first span yellow/red and for the second unconfigured (ok, I just set up 1 span) |
| 17:15.08 | Jumpie | a full pri is expensive |
| 17:15.30 | Jumpie | im tryin to save client smoney |
| 17:15.30 | Jumpie | hehe |
| 17:15.52 | JayTee52 | 23 channels on T1 is usually less expensive than a dozen pots lines per month unless your telco is a greedy pig |
| 17:16.01 | Skarmeth | JayTee52, I used genzaptelconf |
| 17:16.41 | lesouvage | JayTee52: funny, in Europe we have 30 channels on the same cable. |
| 17:16.55 | JayTee52 | Skarmeth, I've never done E1, just T1 for PRI but I think most circuits use 16 as the D channel. Not sure though. |
| 17:17.34 | Skarmeth | JayTee52, I have it alerdy set for D channel... |
| 17:17.52 | tzafrir | lesouvage, on the same cable you can also pass 100Mb |
| 17:18.09 | tzafrir | not just 2Mb like in a lousy E1 :-) |
| 17:18.17 | Skarmeth | my E1 channel 16 (D-channel) correspond to zaptel channel 20 |
| 17:18.24 | JayTee52 | Skarmeth, you posted earlier that D channel was on channel 20 |
| 17:18.33 | JayTee52 | oh, sorry. |
| 17:18.44 | Skarmeth | I have a 4 ports card... as first one |
| 17:18.53 | JayTee52 | ah, gotcha |
| 17:18.53 | Skarmeth | so, TE220B count start at 5 |
| 17:19.05 | Skarmeth | so 16+4 = 20 |
| 17:19.21 | tzafrir | Skarmeth, with the default /etc/{default,sysconfig}/zaptel the PRI card's module will load before wctdm on boot |
| 17:19.46 | JayTee52 | that was going to be my next suggestion |
| 17:20.16 | Skarmeth | I should be... so, E1 channel count should be 1-31... but it is not.... |
| 17:20.51 | lesouvage | tzafrir: you are totally right. TDM is ald days telephony. A proper sip/iax trunk provider doing the interconection for you is my first choice |
| 17:20.56 | Skarmeth | and I can't remove wct4xxp... I get a log of kernel ops |
| 17:21.00 | JayTee52 | just moving 2 TDM04B cards from one Dell server to another slightly older server flipped my channel assignments because the PCI slot numbers are reveresed between the server models. Ain't this crap fun? |
| 17:21.21 | Skarmeth | heheh I am using a dell server |
| 17:21.30 | Jumpie | JayTee52 |
| 17:21.35 | Jumpie | but do you handle the incoming channels as pri? |
| 17:21.37 | Jumpie | on your cards? |
| 17:21.42 | Skarmeth | my TDM card are on PCI slot and TE on PCIe |
| 17:22.12 | JayTee52 | Jumpie, two different servers. One has the TDM cards and the other has a TE212P card |
| 17:22.39 | *** join/#asterisk bakermd (n=none@70.91.165.193) |
| 17:23.17 | tzafrir | Skarmeth, it generally depends on the order of modules modprobing |
| 17:23.18 | Jumpie | JayTee52 i mean |
| 17:23.22 | Jumpie | if i was to get a chan t1 |
| 17:23.24 | Jumpie | if it was cheaper |
| 17:23.30 | Jumpie | what card would i use? |
| 17:23.39 | Jumpie | i wanna have it dedicatedc channels |
| 17:23.53 | Skarmeth | I am using zaptel default load order |
| 17:24.11 | Skarmeth | it have wct4xxp and wctdm uncommented respectively |
| 17:24.58 | tzafrir | Skarmeth, genzaptelconf / zapconf will generate you a working (though maybe not totally optimal) config |
| 17:26.56 | Skarmeth | it should be... |
| 17:27.37 | JayTee52 | Jumpie, if you only need one T1 circuit then the TE121B for a PCI Express slot or the TE122B for a standard PCI slot. |
| 17:28.30 | lesouvage | mvanbaak: the features are working with codes with more then one digit but I have to press them fast. Is there an option to set the timeout a litlte bit longer? |
| 17:28.51 | mvanbaak | lesouvage: not that I'm aware of |
| 17:29.03 | lesouvage | mvanbaak: at least it's working with an inbound iax connection |
| 17:29.04 | mvanbaak | you can try to set the timeout for digits |
| 17:29.12 | mvanbaak | lesouvage: yeah, that's what I tested |
| 17:29.18 | JayTee52 | Jumpie, I'm not sure what you mean by dedicated channels. On PRI's the telco usually passes your 4 digit extensions over the first available channel and all you need to do is match that in an [Incoming] context to route the call to an extension. |
| 17:29.31 | mvanbaak | I also tested with inbound SIP connection (had to setup XS4ALL voip for that tho) |
| 17:30.06 | lesouvage | mvanbaak: I have to be so fast that it can be only be done by pressing the same digit twice. |
| 17:30.46 | mvanbaak | lesouvage: try: Set(TIMEOUT(digit)=5) |
| 17:30.52 | mvanbaak | dont know if it helps |
| 17:30.58 | mvanbaak | but it's worth a try |
| 17:31.46 | JayTee52 | I have the timeout set to 10 for one phone because the person that uses it is....um...slower than a herd of turtles. |
| 17:32.30 | mvanbaak | lol |
| 17:32.34 | lesouvage | Is it possible to launch a feature by sending the dtmf sequence from within the dialplan, something like exten => s,n,SendDTMF(*1111) ? |
| 17:32.44 | mvanbaak | no |
| 17:32.56 | mvanbaak | the features have no knowledge of the dialplan |
| 17:33.06 | mvanbaak | that is clearly documented in features.conf |
| 17:34.09 | Jumpie | JayTee52 |
| 17:34.10 | Jumpie | right |
| 17:34.13 | Jumpie | but if i get a t1 |
| 17:34.22 | Jumpie | i guess what i dont get is, people say 'oh i use my t1 for voice' yet people also have pri |
| 17:34.28 | Jumpie | in all actuality its the same thing isnt it? its just how its handled? |
| 17:34.33 | Jumpie | its still a copper line that terminate with channels |
| 17:34.41 | pa | anyone uses odbc with postgresql? i dont have the odbc.sql to load into postgres....this: /usr/share/psqlodbc/odbc.sql |
| 17:34.56 | Jumpie | i want the cheapest solution, specifically i have a customer who lny needs 10 concurent calls |
| 17:35.02 | Jumpie | so i could do a 10-12 channel t1 |
| 17:35.07 | Jumpie | basically a 768k |
| 17:35.18 | Jumpie | what interface card do i need? the same card for pri/t1 ? |
| 17:35.21 | JayTee52 | T1 could be all data, all voice or a mix of the two. PRI stands for Primary Rate Interface and is for voice circuits. |
| 17:38.15 | JayTee52 | Jumpie, I'd got with an 8 port FXO and a 4 port FXO card and get 10 or 12 pots lines. |
| 17:39.01 | JayTee52 | cuz the other way is what is refered to as fractional T1 and I don't think most telcos will provision a fractional PRI circuit. I've never used or heard of one. |
| 17:39.31 | Jumpie | JayTee52 i know, but you can run voice over a t1 channel |
| 17:39.36 | Jumpie | relaly? i've heard of frac pri |
| 17:40.13 | Jumpie | the idea is im trying to eliminate all this guy's pots lines |
| 17:40.22 | Jumpie | otherwise he's not saving any money because he's still paying for those PLUS the voip gear |
| 17:41.31 | JayTee52 | I'm not saying fractional PRI doesn't exist, I've just never used it or needed to. All my employers, current and former used full T1 PRIs. |
| 17:41.42 | bakermd | Where is the file that needs to be copied into /etc/init.d to allow for "service asterisk start" ? |
| 17:41.48 | Jumpie | how much would you say is a competative rate for that? |
| 17:47.30 | JayTee52 | it varies so much I can't say. here's a good article to give you a breakdown. Scroll down to item #7 http://idea-pulse.com/blog/2007/09/04/voice-pri-is-it-right-for-my-business.html |
| 17:50.45 | JayTee52 | bakermd, did you install Asterisk by compiling or through a package? |
| 17:51.37 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 17:51.41 | Jumpie | i just want to save these guys money |
| 17:52.36 | JayTee52 | well, then find out how much that many channels are from the telco as fractional PRI, full PRI and POTS. compare the prices and the equipment costs. |
| 17:55.47 | lesouvage | How hard would it be to add a variable and the code to automon to be able to set an announcement in the dialplan played while the key set in features.conf is hit (recording starts/recording stops) |
| 18:01.44 | *** join/#asterisk ToTo (n=ToTo@87.2.134.176) |
| 18:03.31 | *** join/#asterisk tfcarlin (n=tim@c-67-173-252-91.hsd1.co.comcast.net) |
| 18:09.53 | tfcarlin | Is this the correct forum to ask a question about a digium card on asterisk? |
| 18:10.34 | *** join/#asterisk kamanashisroy (n=kamanash@202.56.7.142) |
| 18:11.32 | ectospasm | tfcarlin: it won't hurt |
| 18:12.24 | tfcarlin | I have a digium TDM402B (2fxo) what work fine with one PSTN line (digital phone service from Comcast) but not with my PSTN line from QWEST |
| 18:12.33 | tfcarlin | Any ideas on how to debug? |
| 18:13.07 | ectospasm | try swapping the lines into the card... does the same thing happen (Comcast VoIP works, but PSTN doesn't)? |
| 18:13.27 | tfcarlin | Exactly. |
| 18:13.43 | ectospasm | Have you tried swapping the modules on the card? |
| 18:14.51 | tfcarlin | The Comcast line works find on both channels and Qwest fails on both channels. I'm not sure what you mean by swapping. |
| 18:15.17 | ectospasm | ... to different slots on the card... |
| 18:15.43 | ectospasm | and, an analog handset works on the Qwest line? |
| 18:16.06 | tfcarlin | Yes, the analog handset works fine. In fact, the line works with a SPA-3000. |
| 18:16.23 | ectospasm | that's weird |
| 18:16.36 | Jumpie | anyone got verizon fios pots line to work ok? |
| 18:16.42 | Jumpie | im about to get a 1fxo and hope it works :D |
| 18:16.47 | tfcarlin | Agreed. |
| 18:17.16 | drmessano | 1 FXO? |
| 18:17.17 | drmessano | What sort? |
| 18:17.23 | ectospasm | tfcarlin: have you tried fxotune on the lines? It may be the line characteristics are such that the one fxo won't work with the Qwest line |
| 18:17.44 | Jumpie | its a generi c cloen...for testing purposes, but the chipset shows as a wildcard |
| 18:17.50 | drmessano | X100P? |
| 18:17.53 | drmessano | Oh god |
| 18:17.57 | Jumpie | yup |
| 18:18.04 | Jumpie | scuse me i dont pay 4k for testing crap |
| 18:18.13 | drmessano | Good luck.. X100P is the worst thing you could buy |
| 18:18.15 | drmessano | 4000? |
| 18:18.27 | Jumpie | for the good stuff its pricey |
| 18:18.31 | tfcarlin | I did run fxotune and then fxotune -s. At one point adding "w" to the Trunk worked. |
| 18:18.33 | Jumpie | how its a digium clone? |
| 18:18.43 | Jumpie | why is it the worst thing i could buy? |
| 18:18.55 | drmessano | X100P is a crappy, flaky, shitty card |
| 18:19.05 | JayTee52 | I used some X100P cards when I first started messing with Asterisk. They suck to put it politely. |
| 18:19.16 | *** join/#asterisk djs26 (n=djs@unaffiliated/djs26) |
| 18:19.19 | Jumpie | lol |
| 18:19.22 | Jumpie | define sucks |
| 18:19.23 | drmessano | It's a winmodem that Mark made work with Asterisk |
| 18:19.26 | Jumpie | doesnt work, crapy quality, etc |
| 18:19.39 | drmessano | Crap quality, LOTS of echo issues, etc |
| 18:20.03 | JayTee52 | sucks=locks up line, sucks=having to reboot server to reset the card, sucks=noise, echo, lousy voice quality. |
| 18:20.05 | Jumpie | drmessano i need it for testing, plans, etc |
| 18:20.08 | Jumpie | not for actualy product, ever |
| 18:20.20 | drmessano | Jumpie: You're gonna have more problems than not |
| 18:20.29 | drmessano | It's not worth even testing with |
| 18:20.34 | JayTee52 | Jumpie, how much you payin? |
| 18:20.36 | Jumpie | well it was $30 so ill take my chances |
| 18:20.47 | Jumpie | i'd never use it for clients |
| 18:20.48 | Jumpie | i know that :D |
| 18:20.55 | JayTee52 | hell, I'd have sold you two of them for that price, plus shipping |
| 18:21.00 | Jumpie | no kidding |
| 18:21.03 | Jumpie | lol |
| 18:21.18 | drmessano | Jumpie: It sounds to me like you're doing everything possible to learn Asterisk and it's associated hardware the worst way possible |
| 18:21.24 | *** join/#asterisk lirakis (n=lirakis@cpe-68-175-38-65.nyc.res.rr.com) |
| 18:21.44 | drmessano | ~x100p |
| 18:21.45 | jbot | extra, extra, read all about it, 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 TDM01B, and you will save your sanity, your hair, and countless other things. |
| 18:21.45 | JayTee52 | drmessano, not Asterisk, Jumpie is a trixbox boy |
| 18:22.28 | *** join/#asterisk LordScinawa (n=adsf@host31-235-dynamic.5-87-r.retail.telecomitalia.it) |
| 18:22.30 | tfcarlin | Any ideas on how to debug a TDM not recognizing a PSTN line? |
| 18:22.58 | drmessano | When people tell you "This sucks", we're not getting kickbacks for you buying a card.. it really does suck. If you told me you bought an eMachine for gaming, and I called you an effin moron, would you dismiss it? |
| 18:23.07 | Jumpie | lol it depends on the game |
| 18:23.14 | drmessano | Wow |
| 18:23.21 | JayTee52 | Pong works well on the eMachines |
| 18:23.47 | drmessano | I'm convinced now that you're really just trying to cheap ass your way into Asterisk with little effort and investment.. |
| 18:23.52 | Jumpie | yes |
| 18:23.53 | Jumpie | your 100% right |
| 18:24.00 | Jumpie | i dont have the budget to tinker with it on costly gear |
| 18:24.03 | Jumpie | for what i need to 'learn' itll suffice |
| 18:24.07 | drmessano | So who would ever buy a system from you.. ever? |
| 18:24.11 | Jumpie | many people |
| 18:24.13 | JayTee52 | Jumpie, aren't you running trixbox? |
| 18:24.14 | Jumpie | because THEY give the money |
| 18:24.23 | Jumpie | drmessano throw me $10k for testing, and then you can talk |
| 18:24.27 | drmessano | But you havent learned shit by using Trixbox and an X100P card... |
| 18:24.30 | drmessano | Thats laughable |
| 18:24.35 | Jumpie | i've actually learned quite a bit |
| 18:24.56 | Jumpie | unfortunately there is a 'higher than thou' complex in this room |
| 18:25.10 | Jumpie | because we dont have the biggest and baddest of everything, im geting there |
| 18:25.20 | JayTee52 | only for those who have failed to "do their homework" |
| 18:25.27 | Jumpie | no i didnt fial to do shit |
| 18:25.33 | Jumpie | i just am not paying for costly gear to learn basic things |
| 18:25.42 | Jumpie | for incoming plans or testing |
| 18:25.46 | Jumpie | having a 100000$ card or a $30 card |
| 18:25.49 | Jumpie | will have no difference |
| 18:25.53 | drmessano | Jumpie: You're an idiot.. You keep throwing huge dollar figures out there.. 4k, now 10k.. if you got a cheap system (that you may have had lying around) with a single FXO digium, rhino, or sangoma card for under $150, you would be set |
| 18:26.03 | drmessano | You obviously know nothing about card prices |
| 18:26.07 | ectospasm | costly? You think a current analog card is costly? It'd be more costly to use something cheap and broken... |
| 18:26.08 | Jumpie | all the sangoma crap i saw was over $500 |
| 18:26.09 | drmessano | $100000 card now? |
| 18:26.17 | drmessano | When is the Million dollar card coming out, I want one |
| 18:26.27 | Jumpie | probably at the school where you learned manners |
| 18:26.47 | Jumpie | of course i want more reliab le gear, it'll come |
| 18:26.50 | Jumpie | this is all out of my OWN pocket |
| 18:26.51 | drmessano | Have you even REALLY priced hardware, or did you just jump at an X100P because it was cheap |
| 18:26.59 | Jumpie | yes i ahve |
| 18:27.01 | Jumpie | and i know what i wanna get |
| 18:27.05 | Jumpie | of course i was being slightly sarcastic |
| 18:27.10 | Jumpie | this was a recommendation from a friend to get me started |
| 18:27.16 | drmessano | $150 for a card is not a huge investment if you expect people to pay you for whole systems |
| 18:27.20 | Jumpie | well i never saw a 150 card |
| 18:27.25 | Jumpie | so im sorry fo rmy ignorance |
| 18:27.26 | drmessano | Not "I learned on an X100P, I R N XPART" |
| 18:27.32 | Jumpie | i was lookin on sangoma and digium and they were costly |
| 18:27.46 | ectospasm | going to resellers is usually cheaper |
| 18:28.17 | JayTee52 | a TDM01B (TDM400 card with one FXO module) costs around $119.00 retail and lets you expand to 4 FXO ports by adding modules. |
| 18:28.39 | Jumpie | well i guess i just wanted to start asap with somethin basic |
| 18:28.43 | Jumpie | you have to realize |
| 18:28.48 | Jumpie | when i first got it, i knew less than i know now |
| 18:28.51 | Jumpie | which still isnt much |
| 18:28.54 | Jumpie | its already bought |
| 18:29.04 | Jumpie | ectospasm whats a good reseller? |
| 18:29.09 | Jumpie | telephonydepot? |
| 18:29.16 | Jumpie | i heard not so good things about voipsupply |
| 18:29.19 | ectospasm | I dunno, I haven't looked in a while |
| 18:29.20 | bitzero | would contend that you can learn a lot with an x100p as your first Zap channel device... You've got to start somewhere afterall. |
| 18:29.27 | Jumpie | yea |
| 18:29.30 | bitzero | but trixbox .... |
| 18:29.31 | Jumpie | i mean for my incoming routes |
| 18:29.35 | Jumpie | plans whatever |
| 18:29.48 | Jumpie | honestly bitzero i've gotten it to do exactly what i've wanted to for the time being |
| 18:29.59 | Jumpie | and im learning how the files interact |
| 18:30.06 | Jumpie | so ill end up dumping it shortly |
| 18:30.07 | bitzero | Thats great, good for you - I'm not knocking it if you're just building something for your own use. |
| 18:30.13 | Jumpie | right |
| 18:30.20 | Jumpie | i wanna learn i guess on a less steep learning curve |
| 18:30.27 | Jumpie | to ulltimately provide somethin better later for clients, |
| 18:30.28 | bitzero | But I am saying there are a lot of steps between trixbox and qualified VAR. |
| 18:30.29 | Jumpie | not like its omorrow |
| 18:30.31 | drmessano | trixbox makes it harder to learn |
| 18:30.33 | ectospasm | You'd be better off getting the book, and learning that way |
| 18:30.38 | bitzero | would agree with drmessano |
| 18:30.41 | ectospasm | ~thebook |
| 18:30.42 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com |
| 18:30.42 | drmessano | you have to unlearn thinhs with trixbox |
| 18:30.42 | bitzero | you're learning it "wrong" first. |
| 18:30.49 | bitzero | so you'll have to unlearn a lot when you transition. |
| 18:30.49 | Jumpie | well when i first tried i was doing ubuntu and alot of conflicts with their guides so i was very confused |
| 18:30.59 | bitzero | its your time, so whatever floats your boat - but you will be handicapped. |
| 18:30.59 | Jumpie | ill load fresh centos |
| 18:31.21 | Jumpie | i guess i was worried i'd never know if everything is 'up and running' cause i dont have a package gui |
| 18:31.23 | Jumpie | im very graphics based |
| 18:31.32 | Jumpie | once i learn somethin, then i like to go cli |
| 18:31.37 | bitzero | well telephony isn't |
| 18:31.38 | Jumpie | if its my way of learning , and the end justifies the means so be it |
| 18:31.45 | drmessano | You're doing it the wrong way.. and lots of folks have told you that |
| 18:31.59 | bitzero | It's not like it's any skin off your back, drmessano. |
| 18:32.05 | Jumpie | what exactly do i need to unlearn? |
| 18:32.18 | Jumpie | the only difference is you cant directly edit a file, you have to edit the includes |
| 18:32.19 | bitzero | you've let him know your opinion - made sure he understands that it's an exceptionally common opinion... |
| 18:32.31 | Jumpie | im not saying im gonna go production with trix |
| 18:32.38 | Jumpie | but i honestly think im beter off now, and ready to go vanilla |
| 18:32.47 | JayTee52 | well, good luck with that |
| 18:32.53 | bitzero | we're saying you'll get some ideas of "how things are" from what you're playing with, and you'll have to overcome those later. |
| 18:32.54 | Jumpie | of COURSE i want to be as high speed as you gusy |
| 18:33.16 | bitzero | thats all - so while drmessano seems to want to save you from yourself with tough love right now... |
| 18:33.19 | Jumpie | ok well if i was to do it 'the normal way' |
| 18:33.25 | Jumpie | is there a particular platform that is good? |
| 18:33.27 | drmessano | True.. Except when the person likes to come in and argue for help, and insists on using something sub standard, then argues over the lack of help or being given an answer that isn't within the context of what they're doing. |
| 18:33.28 | Jumpie | dell power edge, proliants? |
| 18:33.30 | bitzero | I'd just caution you "You're making it harder for yourself in the long run." |
| 18:33.37 | Jumpie | drmessano but som eof the questions i ask are not trixbox related |
| 18:33.46 | Jumpie | they are things that are generic and you immediatley pound on me about trix |
| 18:33.50 | Jumpie | im not sayin 'hey this gui button does that' |
| 18:33.53 | drmessano | Jumpie: If you're on a trixbox system, they're ALL trixbox related |
| 18:33.54 | Jumpie | i know your point |
| 18:33.58 | bitzero | I wasn't around for the start of the conversation. |
| 18:34.08 | Jumpie | a syntax for X is the same |
| 18:34.11 | Jumpie | a match for y is the same |
| 18:34.16 | bitzero | if you're in here asking for help with Trixbox - don't think you're owed an answer. |
| 18:34.16 | Jumpie | its how i get to it thats diff |
| 18:34.25 | bitzero | there are trixbox specific forums for you. |
| 18:34.32 | JayTee52 | Dell servers work ok for me. If I was going to do it on the cheap I'd buy a decent low cost Intel motherboard, mid range Core2Duo and run CentOS 5 on it. |
| 18:34.33 | drmessano | If you open a FreePBX written config file in nano and ask for help, you're not "just editing an asterisk config file, so it's the same" |
| 18:34.39 | Jumpie | ok well, i guess im officialy tired of the trix flaming so i ghuess ill dump it |
| 18:34.49 | JayTee52 | but lots of people run * on Debian or Fedora too |
| 18:35.01 | Jumpie | JayTee52 the book is centos based |
| 18:35.04 | Jumpie | i want to get as close as i can |
| 18:35.07 | bitzero | Jumpie: we could give you answers to specific questions that may well break other parts of trixbox. |
| 18:35.08 | Jumpie | to me the book is a bit lacking though |
| 18:35.20 | bitzero | because trixbox and freePBX and all the others make a lot of assumptions about how things are configured. |
| 18:35.27 | bitzero | if you go outside of that, you loose. |
| 18:35.34 | JayTee52 | by the book, you mean Asterisk TFOT 2nd ED? |
| 18:35.38 | Jumpie | yes |
| 18:35.47 | Jumpie | it seems like it gives you alot of resources |
| 18:35.50 | Jumpie | but not really throws it all together |
| 18:35.55 | lirakis | Jumpie: its a tutorial .. its not all inclusive |
| 18:35.57 | Jumpie | k |
| 18:36.07 | lirakis | Jumpie: ive not found any "source" for all things asterisk |
| 18:36.12 | Jumpie | so let me get this straight, you'd rather have me asking more simple questions, on a normal isntall than trix :D |
| 18:36.13 | JayTee52 | CentOS is just Red Hat Enterprise Linux without the branding. |
| 18:36.20 | lirakis | except [TK]-Dfender |
| 18:36.21 | lirakis | he he |
| 18:36.22 | drmessano | I use FreePBX, and I know a lot about what is under the hood... at least, how things interact and how I can work around them.. and have even added lots of packages to a box with FreePBX written config files, and worked around the contexts and whatnot.. But there's a method to that, and an acceptance that some of what you do may not work out for reasons you cannot change. |
| 18:36.40 | Jumpie | drmessano well i definately want flexibility |
| 18:36.42 | Jumpie | and customization |
| 18:36.44 | JayTee52 | I've run * on Ubuntu 6.06 server, CentOS 5 and RHEL 5 64 bit. |
| 18:36.45 | drmessano | That's NOT.. NOT the same as learning asterisk from the ground up |
| 18:36.50 | drmessano | and you're NOT doing it the right way |
| 18:36.55 | bitzero | runs * on Slackware |
| 18:37.02 | Jumpie | well i guess its that i hear peole who are BOTH reputable and seem to have knowledge |
| 18:37.07 | Jumpie | ahlf say 'get throw in the deep end first' |
| 18:37.11 | Jumpie | and others say 'trix is a good start' |
| 18:37.14 | Jumpie | so who is wrong? who is right? |
| 18:37.25 | lirakis | Jumpie: jump in the deep end |
| 18:37.27 | drmessano | the people that have used both are correct |
| 18:37.36 | Jumpie | ok well |
| 18:37.40 | Jumpie | im gonna jump in tonight |
| 18:37.40 | bitzero | The trix people are wrong. |
| 18:37.48 | Jumpie | im gonna blow it away, fresh centos install |
| 18:37.52 | Jumpie | i wanna be sure i have all the packages thought |
| 18:38.02 | lirakis | Jumpie: i started out using trixbox for like.. 3 weeks .. then i wiped it out .. and did a centos install with asterisk from source. |
| 18:38.03 | Jumpie | like i said, ubuntu compilation and packages was way different and the book wasnt much help on that aspect |
| 18:38.10 | Jumpie | lirakis ok well thats wha ti'll do |
| 18:38.14 | Jumpie | :) |
| 18:38.18 | lirakis | Jumpie: .. it wont work right away.. but that will just show you .. how much you had no clue about |
| 18:38.18 | drmessano | The people that are die hard asterisk users with no GUI use can't speak for both sides.. they just know what the GUIs can do.. the ones that have used both can tell you with certainty, and I can say that a lot of the experts in here have used both |
| 18:38.32 | JayTee52 | Jumpie, this will serve you well for a start: http://www.voip-info.org/wiki/view/CentOS+5+and+Asterisk+1.4.x+installation |
| 18:38.35 | lirakis | Jumpie: but you will learn a lot.. and have an infinitely more flexible, and stable system |
| 18:38.48 | Jumpie | lirakis its just frustrating because im a very technical person, and a good troubleshooter, im more about efficiencyu and i thought the gui was more efficient |
| 18:38.54 | Jumpie | lirakis well of course..thast what i want |
| 18:38.56 | Jumpie | thanks JayTee52 |
| 18:39.03 | lirakis | Jumpie: the gui is a severe limitation.. really |
| 18:39.08 | lirakis | Jumpie: thats about it |
| 18:39.09 | Jumpie | ok |
| 18:39.30 | Jumpie | well i dont not beleive you, i just feel like you burnt out my torche |
| 18:39.31 | JayTee52 | GUI's just slow the system down running native and web interfaces usually just hose things half the time. |
| 18:39.34 | Jumpie | and im going into a cave in darkness :D |
| 18:39.34 | bitzero | Jumpie: if you thought a GUI (ANY GUI) was more efficient ... I question calling yourself "a very technical person" |
| 18:39.43 | lirakis | Jumpie:.. and when you get into managing 300 users on a pbx.. a gui is not going to make it any more efficient.. probably a lot less |
| 18:39.46 | Jumpie | more efficient as in package conslidation |
| 18:39.52 | Jumpie | ok ok...granted |
| 18:39.55 | lirakis | Jumpie: sorry it sounds like im trying to convince you.. im not.. |
| 18:39.56 | Jumpie | ill tackle this tonight :) |
| 18:40.03 | lirakis | Jumpie: im just telling you my exp. |
| 18:40.06 | Jumpie | lirakis i know you guys are right |
| 18:40.12 | Jumpie | i also know and hav eseen the support is much better on vanilla |
| 18:40.19 | Jumpie | standard, whatever u call it |
| 18:40.22 | lirakis | Jumpie: b/c i have used trix, freepbx ontop of asterisk, and just plain asterisk |
| 18:40.27 | bitzero | What we're all getting at, Jumpie, is: Do what we say or we wont be your friend. |
| 18:40.31 | Jumpie | lol |
| 18:40.37 | Jumpie | bitzero thast what i think i was beginning to understand |
| 18:40.38 | Jumpie | :D |
| 18:40.48 | Jumpie | so do you not even recommend getting freepbx ove rasterisk? |
| 18:40.58 | lirakis | Jumpie: no |
| 18:41.04 | Jumpie | ok |
| 18:41.06 | Jumpie | so just filez |
| 18:41.11 | lirakis | Jumpie: yes |
| 18:41.20 | tzafrir | Jumpie, FreePBX is a complete PBX built on top of Asterisk |
| 18:41.41 | Jumpie | so again, more constraints then |
| 18:41.42 | tzafrir | Asterisk is basically a PBX toolkit |
| 18:41.43 | lirakis | Jumpie: and download or buy the book... its a great way to get started.. and can answer A LOT of questions that beginners have |
| 18:41.45 | JayTee52 | Jumpie, nano and vi are your friends, gedit hates you! :-) |
| 18:41.45 | lirakis | ~the book |
| 18:41.46 | mvanbaak | FreePBX is a virus |
| 18:41.50 | tzafrir | You can build your own |
| 18:41.57 | Jumpie | what i dont get is...if all this other stuff is SO aweful, why do people even bother making them, selling books on them, etc? |
| 18:41.59 | lirakis | err.. |
| 18:42.02 | lirakis | ~thebook |
| 18:42.03 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com |
| 18:42.05 | JayTee52 | money! |
| 18:42.09 | Jumpie | JayTee52 i can use vi |
| 18:42.13 | tzafrir | If FreePBX suits you - fine. The problem is: what if it doesn't? |
| 18:42.13 | Jumpie | im not anti unix at all |
| 18:42.25 | Jumpie | and i worked with solaris for 6 years..i know thats a bit diff but |
| 18:42.26 | lirakis | Jumpie: how about anti Linux ?? |
| 18:42.26 | tzafrir | That's the tradeoff |
| 18:42.27 | lirakis | he heh |
| 18:42.31 | *** join/#asterisk qdk (n=qdk@195.242.194.42) |
| 18:42.32 | Jumpie | *nix |
| 18:43.01 | drmessano | Trixbox is great if you're wanting to throw a system in place, put 10 to 50 users on it, and be done.. |
| 18:43.14 | Jumpie | drmessano true, but i'd rather know one way and stick with it |
| 18:43.18 | drmessano | It makes it fast and easy |
| 18:43.19 | mvanbaak | and not having to troubleshoot anything |
| 18:43.20 | Jumpie | some of my clients potentially ARE 10-20 users |
| 18:43.25 | lirakis | Jumpie: trixbox etc. is made for .. hmmm.. not so smart people .. or lazy people .. who want a boxed solution |
| 18:43.29 | Jumpie | but i dont wanna like have to do some clients trix and some clients non |
| 18:43.33 | Jumpie | i'd rather go big or go home |
| 18:43.45 | Jumpie | and with telco, there is no such thing as 'never having to troubleshoot' |
| 18:44.00 | drmessano | If you are going to use Trixbox, expect to make backups, and expect to blow the box up and restore from a FreePBX backup if the Fonality folks push out a shitty RPM next week and you decide to update |
| 18:44.09 | outtolunc | the issue is when you or the client wants to do something 'non-standard' to the existing trixbox setup |
| 18:44.24 | outtolunc | and noone!!! wants to help you <G> |
| 18:44.25 | lirakis | Jumpie: yeah.. there is no stability or upgrade path with Trixbox or Freepbx |
| 18:44.28 | Jumpie | one client, has 22 plain pots lines coming in |
| 18:44.32 | drmessano | Also plan to have that box locked into one minor version, because minor updates have never been upgradable with TB |
| 18:44.33 | Jumpie | only needs to make 10 concurrent |
| 18:44.40 | Jumpie | sigh |
| 18:44.42 | drmessano | lirakis: yep, exactly |
| 18:44.49 | Jumpie | ok |
| 18:45.15 | drmessano | The biggest problem with trixbox.. |
| 18:45.17 | drmessano | IMHO |
| 18:45.28 | outtolunc | the dialplan <G> |
| 18:45.33 | drmessano | Without being my usual drmessano grouchy self |
| 18:45.36 | drmessano | Naah |
| 18:45.37 | mvanbaak | lol outtolunc |
| 18:45.40 | lirakis | .. different subject here... |
| 18:45.43 | drmessano | The RPMs |
| 18:45.52 | ManxPower | The biggest problem with Trixbox is that it's users seem to have some odd mental defect where they come here for support, instead of the correct support channels. |
| 18:45.52 | lirakis | ive got a new 1.4 install on a Xen domU |
| 18:46.06 | drmessano | Relying on Fonality RPMs for Asterisk, Zaptel, etc |
| 18:46.09 | drmessano | They ALWAYS break them |
| 18:46.10 | tzafrir | ManxPower, that's not a mental defect |
| 18:46.20 | *** join/#asterisk bminish (n=bminish@78.152.241.185) |
| 18:46.23 | lirakis | and i get NO audio when I call from phones in DIFFERENT LAN's .. but if i call from phones in the same lan it works fine |
| 18:46.25 | lirakis | .. but |
| 18:46.25 | tzafrir | It's because #trixbox is quite abandoned |
| 18:46.25 | ManxPower | tzafrir: I can't think of any other reason. |
| 18:46.32 | drmessano | They ALWAYS create a bunch of bricked boxes every few months with untested RPMs |
| 18:46.39 | lirakis | .. the Xen and DomU are on public IP |
| 18:46.40 | ManxPower | then perhaps people should not be using a product that they can't get support for. |
| 18:46.45 | *** join/#asterisk ZaVoid (n=zavoid@c-67-165-25-195.hsd1.pa.comcast.net) |
| 18:46.50 | drmessano | They can, in the forums |
| 18:46.53 | lirakis | so basically |
| 18:46.56 | drmessano | Which is where I direct them |
| 18:47.12 | lirakis | lan1 -> internet -> PBX -> internet lan1 works fine |
| 18:47.15 | lirakis | but |
| 18:47.20 | ManxPower | drmessano: apparently not, they still show up here -- like rats carrying the plague. |
| 18:47.29 | lirakis | lan1 -> internet -> PBX -> internet -> lan2 |
| 18:47.31 | lirakis | i get no audio |
| 18:47.41 | outtolunc | glares at jumpie <G> |
| 18:47.41 | lirakis | asterisk stays in media all the time |
| 18:47.43 | ManxPower | lirakis: I think you need NAT to get from lan1 to the internet. |
| 18:47.46 | tzafrir | This channel has no GUI users for a very self-selecting reasons |
| 18:47.47 | Jumpie | lol |
| 18:47.57 | lirakis | .. ive got NAT setup on ALL sip peers |
| 18:47.59 | Jumpie | i was gonna get the .pdf printed out at kinkos |
| 18:48.03 | drmessano | FreePBX has it's place.. but not if you're going to edit your dialplan manually.. you can't have both.. FreePBX is great for what it does.. but it's a different set of rules. |
| 18:48.07 | Jumpie | double sided, 2 pages per page |
| 18:48.25 | ManxPower | drmessano: Of course Asterisk GUIs have their place -- its just not THIS place. |
| 18:48.33 | drmessano | ManxPower: Yep |
| 18:48.33 | Jumpie | drmessano would freepbx be ok if your aim is 50 or less employees with basic needs? |
| 18:48.38 | Jumpie | or should i just say screw it |
| 18:48.47 | lirakis | ManxPower: yeah .. like i said.. ive got NAT on all peers... and on the phones themselves |
| 18:48.49 | outtolunc | if you do that and flip the pages fast enough it says 'use the force luke' <G> |
| 18:48.57 | ManxPower | and yet you (the royal you, of course) still seem to be encouraging them to be here. |
| 18:48.58 | Jumpie | ya tha thappened once.. |
| 18:49.01 | lirakis | ManxPower: so im not sure if i am missing it some where |
| 18:49.03 | drmessano | Jumpie: Do you expect to tweak? |
| 18:49.04 | lirakis | else |
| 18:49.14 | ManxPower | lirakis: you were missing it in your diagram |
| 18:49.21 | Jumpie | drmessano i dont know what to expect, but i'd rather know HOW to, and do it sometimes, than assume not to, and need it and im up shit creek |
| 18:49.23 | tzafrir | ManxPower, well, that depends. For me: as long as they can intelgently talk Asterisk speak, and not FreePBX speak, they're welcome here |
| 18:49.32 | drmessano | I am encouraging them to be here? |
| 18:49.36 | Jumpie | never assume what the client will or wont need |
| 18:49.54 | Jumpie | i dont anticipate needing some bizzzare tweak for these guys |
| 18:49.59 | ManxPower | tzafrir: I've just not seen many users manage to figure out how to even troubleshoot Asterisk in a GUI enviroment. They just don't know enough |
| 18:50.02 | Jumpie | but i'd still rather learn what i need to learn |
| 18:50.06 | drmessano | Then that's something you need to figure out |
| 18:50.22 | lirakis | ManxPower: .. lan1 -> NAT -> internet ??? ha ha |
| 18:50.28 | drmessano | Whether or not you need the ability to greatly customize |
| 18:50.33 | tzafrir | ManxPower, no wonder. They were never given a chance |
| 18:50.35 | Jumpie | however, i did find a trixbox bug and i bugreported it :D |
| 18:50.47 | ManxPower | tzafrir: Me: Show us the CLI output of a failed call. User: pastebins 80 lines of dialplan crap created by the GUI. |
| 18:50.48 | outtolunc | feels like were are in time bubble as the same people were having the same convo not but 2 days ago <G> |
| 18:50.56 | Jumpie | outtolunc its the matrix |
| 18:51.16 | ManxPower | Me: Show us the contents of sip.conf User: pastes sip.conf which contains one line #include /etc/asterisk/sip.conf.screwedup.conf |
| 18:51.26 | Jumpie | lol |
| 18:51.29 | outtolunc | jumpers is a movie.. you should have said you were a char from it |
| 18:51.30 | Jumpie | ok ok... |
| 18:51.38 | Jumpie | saw it...decent flik |
| 18:51.39 | bitzero | There is the theory of the mobius... |
| 18:51.42 | bitzero | where time becomes a loop. |
| 18:51.55 | Jumpie | i read some books bout this guy that jumps in mobius contiuum |
| 18:52.05 | ManxPower | outtolunc: and yet it looks like the GUI users didn't get the hint. |
| 18:52.14 | Jumpie | ManxPower no.....i got the hint |
| 18:52.18 | tzafrir | ManxPower, so? I just skip most of it and look for the Dial line |
| 18:52.19 | Jumpie | i saw the target |
| 18:52.26 | ManxPower | Jumpie: you are still here so apparently not. |
| 18:52.33 | Jumpie | not sure if you saw my earlier where i said im scrapping it |
| 18:52.49 | outtolunc | sorry, but i'm still laughin over here.. sorry manx |
| 18:53.02 | Jumpie | im tired of being treated like i have bubonic plague |
| 18:53.05 | Jumpie | so yea im gonna conform |
| 18:53.13 | drmessano | lol |
| 18:53.26 | ManxPower | Jumpie: sort of like if you went to a Debian channel and asked about CentOS. |
| 18:53.33 | Jumpie | ManxPower i was NEVER intending to stick with trix |
| 18:53.35 | drmessano | It's not about conforming, it's about being in the right place.. #asterisk is not the place to get help on freepbx or trixbox |
| 18:53.39 | Jumpie | i was just hoping it'd be easier to learn initially |
| 18:53.45 | Jumpie | i didnt think it'd be all that hard to unlearn |
| 18:53.48 | Jumpie | but i was wrong |
| 18:53.52 | Jumpie | i am man enough to say that |
| 18:54.04 | ManxPower | We don't CARE if you use a GUI or non-GUI, we just care that you don't use #asterisk as #trixbox |
| 18:54.07 | lirakis | ManxPower: ive got a full paste of a SIP debug of the call between peers |
| 18:54.16 | lirakis | ManxPower: http://rafb.net/p/5Lzxs371.html |
| 18:54.31 | lirakis | ManxPower: im going to scour it.. but let me know if you spot anything glaring |
| 18:54.36 | ManxPower | lirakis: today is my complaining day, not my support day. |
| 18:54.49 | tzafrir | ManxPower, but this is just what I said. If one can use Asterisk terminilogy, he's welcomed here |
| 18:54.57 | lirakis | ManxPower: fair |
| 18:55.04 | outtolunc | on the 7th day reverted to a GUI user <G> |
| 18:55.06 | tzafrir | (or she. Or it. jbot's also welcomed) |
| 18:55.20 | outtolunc | runs screaming |
| 18:56.05 | ManxPower | tzafrir: I agree with you, except I think they should also not be using any #includes in their config files. |
| 18:56.31 | Jumpie | im curoius how this book is allowed to be free download |
| 18:56.58 | drmessano | Personally, I use both.. and if I can make something work on a FreePBX install that I made work on a vanilla asterisk box, then cool.. but I make damn sure when something doesn't work, and I ask for help for it, I am testing in a sanitary environment, or I go to the proper PLACE to confer with other users.. it's about PLACE. |
| 18:56.58 | lirakis | ManxPower: some thing is def. jacked up with NAT though.. i only see the internal IP of my local phone here in the whole sip debug |
| 18:57.18 | tzafrir | ManxPower, using #include-s is very logical - it seperates default ("sample"), parts generated from the database, and custom parts |
| 18:57.31 | *** join/#asterisk Great_Randew (n=Andrew@stjhnbsu84w-156034184245.nb.aliant.net) |
| 18:57.34 | ManxPower | tzafrir: And it has no place in troubleshooting. |
| 18:58.21 | ManxPower | #include is like turning on call forwarding on your phone and then expecting the phone company to he able to figure out why you can't receive calls |
| 18:58.21 | tzafrir | ManxPower, now, you ask someone to show you the dialplan , and get a bunch of includes . This is practically the same |
| 18:58.23 | Jumpie | hey JayTee52 |
| 18:58.39 | tzafrir | Many (Asterisk) beginners will give you the same answer |
| 18:59.06 | *** join/#asterisk fakhir_ (n=fakhir@unaffiliated/fakhir) |
| 19:01.10 | drmessano | That's also why I push people to learning vanilla asterisk if they want to tweak FreePBX.. "Hey I saw all these comments in the config files, so I cleaned them up" |
| 19:01.11 | outtolunc | lirakis: i do not have time either but it looks like the 6400 ms invite timer was hit so i sent the BYE |
| 19:01.23 | drmessano | Then, you |
| 19:01.27 | mocker | Guh, why won't this iax server register? |
| 19:01.33 | drmessano | Then, you're screaming for your weekend back |
| 19:01.36 | mocker | It just stopped working for no reason. |
| 19:02.19 | Jumpie | drmessano |
| 19:02.29 | Jumpie | ill be askin alotta questions later most surely |
| 19:02.31 | Jumpie | but at least ill be 'on track |
| 19:02.32 | Jumpie | :D |
| 19:02.53 | drmessano | Get the book, learn asterisk.. then if decide to build boxes, or even quickie deploy ISO based systems, you can actually REALLY troubleshoot the box. |
| 19:03.14 | Jumpie | yes thats what i wanna do |
| 19:03.18 | Jumpie | i wanna quick deploy solution |
| 19:03.19 | drmessano | You know enough of the engine to be able to see something screwed up on top of it |
| 19:03.35 | Jumpie | however, i want to keep the same hardware platform if possible |
| 19:03.52 | Jumpie | when you consider an appliance by trix or digium is like 2000-3000 |
| 19:03.52 | drmessano | Dude, why would the hardware change? |
| 19:04.01 | Jumpie | no im saying |
| 19:04.05 | Jumpie | i dont wanna go client a, this hardware solution |
| 19:04.07 | Jumpie | client b, that |
| 19:04.12 | Jumpie | i wanna find a good one, good price, and stick with it |
| 19:04.49 | drmessano | If you can load an ISO based PBX-on-a-disk on it, certainly you can roll a vanilla asterisk box on it |
| 19:04.55 | lirakis | outtolunc: .. well i have the far end set to autoanswer... so i dont know what it does once i hangup |
| 19:05.02 | lirakis | outtolunc: the messaging works fine though |
| 19:05.12 | lirakis | outtolunc: its just rtp that is not getting through |
| 19:05.22 | lirakis | *signalling |
| 19:05.45 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
| 19:06.10 | mocker | I just keep getting LAGRQ on my IAX2 registration to my second server. |
| 19:07.20 | Jumpie | drmessano im not worried about deployment of it |
| 19:07.48 | drmessano | Why not? |
| 19:07.49 | Jumpie | when you say iso based, you mean all the files on the disk as well as the os? |
| 19:07.59 | Jumpie | drmessano i can install the operating system |
| 19:08.04 | drmessano | I mean Trixbox, PIAF, etc |
| 19:08.04 | Jumpie | i just wanna pick a favorite platform |
| 19:08.06 | Jumpie | oh |
| 19:08.08 | Jumpie | haha no |
| 19:08.10 | drmessano | Thats an ISO based PBX |
| 19:08.12 | Jumpie | im tryin to steer from that |
| 19:08.18 | Jumpie | i just would like a one sotp shop for all my files |
| 19:08.35 | Jumpie | lets say i have no internet connection |
| 19:08.37 | Jumpie | theoretically |
| 19:09.02 | Jumpie | il get what i need, |
| 19:09.06 | Jumpie | im just sounding silly |
| 19:09.08 | Jumpie | thx |
| 19:10.01 | JayTee52 | I'd avoid selling to customers that have no theoretical internet connection |
| 19:10.16 | ectospasm | heh, try someone at a military installation... |
| 19:10.38 | JayTee52 | "Gee, I need to download a firmware update...I'll be back in an hour." |
| 19:10.54 | Jumpie | JayTee52 no i meant, when i have to initially deploy |
| 19:11.00 | Jumpie | er..i'd have internet there..so nm |
| 19:11.23 | JayTee52 | I'll be back later |
| 19:11.32 | ectospasm | No Internet means no way to remotely admin it |
| 19:14.23 | *** join/#asterisk Navion (n=billp@208-106-107-201.dsl-s.california.com) |
| 19:14.27 | mocker | Guhh, and I verified w/ nmap that the iax2 port is open. |
| 19:14.37 | drmessano | mocker, who is the ITSP? |
| 19:14.55 | Navion | Anyone have thoughts about inbound routing that directs all calls from a specific trunk to a destination extension? |
| 19:15.09 | mocker | drmessano: Just connecting two boxes up. |
| 19:15.23 | mocker | And it *was* working |
| 19:15.30 | mocker | But decided to quit on me. :( |
| 19:15.37 | tzafrir | BTW: any comments on http://www.voip-info.org/wiki/view/Asterisk+Data+Configuration ? |
| 19:15.43 | tzafrir | Feel free to remove old junk |
| 19:15.54 | tzafrir | There's plenty of it there, anyway |
| 19:20.25 | SteveTotaro | go to 1.2.x and it will work |
| 19:20.30 | styelz | can i count the number of rings in a dial plan for incomming call some how ? |
| 19:20.56 | SteveTotaro | you can cound time |
| 19:21.14 | SteveTotaro | wai(5) |
| 19:21.29 | SteveTotaro | wait(5) |
| 19:21.38 | styelz | i want to count rings... like a modem does |
| 19:21.40 | SteveTotaro | man, i have too many crumbs in my KB |
| 19:21.55 | SteveTotaro | modem probably fakes is |
| 19:21.58 | SteveTotaro | fakes it |
| 19:22.32 | SteveTotaro | call a number and time how long it takes to reach that number of rings |
| 19:22.56 | SteveTotaro | call a POTS line to be more accurate, POTS to POTS |
| 19:23.05 | SteveTotaro | that's the usual fax method |
| 19:23.21 | styelz | just wondering |
| 19:24.34 | SteveTotaro | there may be, but why? |
| 19:24.34 | mocker | weird, I have a iax2 host I'm registered too, but I can't find the line that's registering me to that host. |
| 19:24.34 | SteveTotaro | never understood why people take the hard way when something apparently easy and simple is right in front of them |
| 19:24.57 | SteveTotaro | mocker: are you using trixbox/freepbx |
| 19:25.04 | styelz | got a pstn line comming in which has normal phones on it, and one goes to an fxo. just wanted to get around asterisk picking the call up for voicemail if the pstn was answered by a normal phone on the line. |
| 19:25.37 | styelz | waits for vomit sounds |
| 19:25.49 | SteveTotaro | i did that setup and had all kinds of troubles |
| 19:26.15 | SteveTotaro | zaptel with "start simple switch" when it detects ring |
| 19:26.29 | *** join/#asterisk lvl- (n=lvl@101.248.dhcp.lht.hhs.nl) |
| 19:26.44 | SteveTotaro | and when you pick up the regular POTS line i think that also messes with zaptel |
| 19:26.56 | mocker | SteveTotaro: No.. |
| 19:27.30 | SteveTotaro | then vi /etc/asterisk/iax.conf |
| 19:27.39 | SteveTotaro | <PROTECTED> |
| 19:27.46 | SteveTotaro | <PROTECTED> |
| 19:27.49 | styelz | i think asterisk still thinks its calling .. maybe the voltage on the line or something |
| 19:27.57 | styelz | when offhook |
| 19:28.18 | SteveTotaro | styelz, it will not work work, period |
| 19:28.24 | styelz | yea |
| 19:28.29 | styelz | thought so |
| 19:28.51 | mocker | SteveTotaro: Heh, been there. |
| 19:29.03 | mocker | I'm in the users.conf now.. |
| 19:29.07 | SteveTotaro | you could get a second number with distinctive ringing pointed at your pots line |
| 19:29.31 | SteveTotaro | and there is a device that will split calls based on ring type |
| 19:29.47 | SteveTotaro | so one type of ring will go to pots, the other to asterisk |
| 19:30.09 | SteveTotaro | mocker: use grep |
| 19:32.03 | mocker | SteveTotaro: Yup, grep -ril is what led me to the users.conf |
| 19:32.15 | drmessano | SteveTotaro: Then you get to deal with weird voltages on the ring decipher boxes |
| 19:32.27 | mocker | I found it, just was weird that I didn't have a register line for it in iax.conf |
| 19:32.36 | *** join/#asterisk Daviey (n=dave@ubuntu/member/daviey) |
| 19:33.35 | SteveTotaro | stylez: you could also pickup a a cheapo two port card from www.openvox.com, one FXO one FXS |
| 19:34.48 | SteveTotaro | http://www.openvox.com.cn/ sorry for the bad link |
| 19:35.44 | SteveTotaro | they used the same reference design that digium did and the build quality of the cards are just as good as Digium, don't know about warranty |
| 19:37.17 | styelz | yea i have a tdm800 |
| 19:38.33 | SteveTotaro | http://www.openvox.com.cn/products_detail.php?genre_id=9&id=36 |
| 19:38.39 | styelz | 6 fxo 2 fxs |
| 19:38.52 | SteveTotaro | so you just don't have enough pots lines, is that the problem? |
| 19:39.22 | styelz | no im trying to work with an existing nec pabx |
| 19:40.04 | styelz | and am being restricted |
| 19:40.53 | SteveTotaro | so you only have one FXS port on your NEC? |
| 19:41.46 | *** join/#asterisk brut- (n=brut@h66-173-4-254.mntimn.dedicated.static.tds.net) |
| 19:41.57 | SteveTotaro | just get another analog FXS card for the NEC |
| 19:42.15 | styelz | i want to, but they wont let me |
| 19:42.16 | SteveTotaro | or get T1 cards for both systems ;) |
| 19:43.09 | SteveTotaro | what does the asterisk box do? IVR? VM? |
| 19:43.23 | styelz | just voicemail |
| 19:43.37 | styelz | but between 5pm and 9am mon - fri |
| 19:43.37 | SteveTotaro | one port VM, lol |
| 19:44.32 | SteveTotaro | so what is the problem, let asterisk take the call and then ring the analog phone |
| 19:44.52 | SteveTotaro | then go to voicemail or dial your on-call person |
| 19:45.11 | styelz | i did have it go direct to VM .. no rings.. that was fine |
| 19:45.13 | SteveTotaro | over voip since you are using your only trunk |
| 19:45.29 | SteveTotaro | so what is the problem? |
| 19:45.31 | styelz | but then .."boss".. compain they want to receive calls after 5pm (sometimes) |
| 19:45.44 | styelz | so i thought i could delay the VM by 20 seconds. |
| 19:45.50 | SteveTotaro | so my solution is easy and makes the boss happy |
| 19:45.51 | styelz | but not so simple |
| 19:46.30 | SteveTotaro | no, just go into asterisk, dial the zap pots phone for 20 seconds, then go to VM |
| 19:46.32 | styelz | its just temporary until the nec lease ends |
| 19:46.47 | SteveTotaro | you could do it in half an hour |
| 19:47.02 | styelz | yea i do |
| 19:47.29 | styelz | but if someone in the office picks a nec extension up. after say 5 seconds. |
| 19:47.43 | styelz | 15 seconds into the conversation, the VM answers |
| 19:47.44 | SteveTotaro | then the call is bridged |
| 19:47.47 | *** join/#asterisk micander (n=Michael_@ip70-181-134-119.sd.sd.cox.net) |
| 19:47.53 | styelz | nec is not voip |
| 19:48.05 | styelz | and it only has 6 pots |
| 19:48.07 | micander | How do you stop a running valgrind instance and not have it lose everything it's gathered? |
| 19:48.11 | SteveTotaro | no, you let asterisk answer the call over the FXS to FXO port on asterisk |
| 19:48.19 | styelz | ive just split them into the fxo ports in asterisk |
| 19:48.38 | SteveTotaro | you are making things way too complex |
| 19:48.58 | styelz | ok i see |
| 19:49.02 | styelz | so no splice |
| 19:49.14 | SteveTotaro | you have six fx0 and two fxo right? |
| 19:49.17 | SteveTotaro | no splice |
| 19:49.18 | styelz | i should be using the fxs ports for the nec ? |
| 19:49.22 | SteveTotaro | throw that wire away |
| 19:49.36 | styelz | ok |
| 19:49.41 | styelz | that would be nicer |
| 19:49.54 | SteveTotaro | are they analog FXS ports on the NEC or digital |
| 19:49.55 | styelz | 2 lines is enough for VM |
| 19:50.01 | styelz | analog |
| 19:50.07 | SteveTotaro | perfect |
| 19:50.22 | SteveTotaro | ok so you have two spare FXS on your NEC? |
| 19:50.33 | styelz | no |
| 19:50.38 | SteveTotaro | only one? |
| 19:50.49 | styelz | on * i have 6 fxo and 2 fxs. |
| 19:51.01 | SteveTotaro | i am talking about your NEC |
| 19:51.18 | SteveTotaro | (3:50:20 PM) irc: ok so you have two spare FXS on your NEC? |
| 19:51.25 | styelz | no |
| 19:51.42 | SteveTotaro | can you please provide more info then yes and no |
| 19:51.49 | styelz | sorry |
| 19:52.09 | styelz | no i dont have two spare fxs ports on the nec |
| 19:52.13 | SteveTotaro | you have one spare fxs on the NEC? |
| 19:52.19 | styelz | none |
| 19:52.37 | SteveTotaro | then what are you spicing? |
| 19:52.41 | SteveTotaro | i dont get it |
| 19:53.07 | SteveTotaro | you have a phone or two for support? |
| 19:53.20 | SteveTotaro | connected to the NEC FXS ports |
| 19:53.30 | styelz | the office is nec pabx |
| 19:53.49 | SteveTotaro | yes, i understand that, I am NEC certified |
| 19:53.59 | SteveTotaro | and you said it at leat ten times noow |
| 19:54.15 | styelz | its just got 6 lines into it |
| 19:54.19 | styelz | PSTN lines |
| 19:54.24 | SteveTotaro | ok |
| 19:54.27 | SteveTotaro | then what |
| 19:54.52 | outtolunc | one it goes into the magic box it is anyones guess <G> |
| 19:55.03 | outtolunc | giggles |
| 19:55.07 | styelz | heh |
| 19:55.14 | styelz | i got the fairies to do it |
| 19:55.25 | SteveTotaro | seriously do you want help or not |
| 19:55.26 | styelz | .. |
| 19:55.34 | drmessano | app_pixiedust.so |
| 19:55.35 | styelz | you've helped me already |
| 19:56.13 | SteveTotaro | just take the NEC FXS lines that go into the phones and plug them into asterisk |
| 19:56.22 | styelz | yea |
| 19:56.34 | styelz | thats a nicer way... |
| 19:56.37 | styelz | will do that |
| 19:56.39 | *** join/#asterisk ZaVoid (n=zavoid@c-67-165-25-195.hsd1.pa.comcast.net) |
| 19:56.57 | SteveTotaro | plug the unplugged phones into the asterisk FXS ports |
| 19:57.12 | styelz | thanks SteveTotaro.. i just cant seem to explain it too well |
| 19:57.57 | SteveTotaro | then designe your dialplan like any other voicemail backed extension |
| 19:58.25 | styelz | yep |
| 19:58.36 | SteveTotaro | i don't think the explaining is the problem, i think it is the comprehension |
| 19:58.54 | styelz | its messy |
| 19:59.02 | hwt | is realtime extensions broken in 1.6? |
| 19:59.38 | SteveTotaro | everything is broken in 1.6 |
| 19:59.42 | hwt | :) |
| 19:59.44 | hwt | really? |
| 19:59.46 | SteveTotaro | realtime is broken period |
| 19:59.54 | SteveTotaro | in 1.4 and 1.6 |
| 19:59.56 | hwt | well, yeah. |
| 20:00.01 | hwt | but are there any other options? |
| 20:00.14 | hwt | i need db backend |
| 20:00.30 | SteveTotaro | realtime? |
| 20:00.41 | SteveTotaro | or lookups |
| 20:00.48 | hwt | i need to keep users and extensions in a db |
| 20:01.02 | SteveTotaro | mysql is nice |
| 20:01.13 | JayTee52 | i need to keep users locked in a cage in the basement |
| 20:01.35 | hwt | SteveTotaro: oh, so there is another way of keeping that in mysql in 1.6? |
| 20:01.44 | SteveTotaro | you need to wait for a new realtime |
| 20:01.54 | SteveTotaro | it is being re-written properly |
| 20:02.14 | hwt | is there no way to make it work? |
| 20:02.24 | SteveTotaro | you can make it work, sure |
| 20:02.31 | SteveTotaro | reliably, no |
| 20:02.49 | SteveTotaro | depends on your definition of reliability though |
| 20:02.57 | hwt | yeah, i don't need 100%. |
| 20:03.08 | hwt | it's just that it broke on my test box when i upgraded. |
| 20:03.17 | SteveTotaro | why 1.6? |
| 20:03.27 | hwt | might be change of separators or something. |
| 20:03.32 | SteveTotaro | is there something there you need that is not in 1.4? |
| 20:03.35 | *** join/#asterisk maagic (i=maagic@fsck.fi) |
| 20:03.35 | hwt | SteveTotaro: just experimentingg right now. |
| 20:03.43 | SteveTotaro | or 1.2 has realtime also |
| 20:03.44 | hwt | SteveTotaro: i heard video works better in 1.6 |
| 20:04.15 | SteveTotaro | not a video fan |
| 20:04.30 | SteveTotaro | only good for porn or important meetings |
| 20:04.49 | SteveTotaro | in which case, there are off the shelf cheap solutions |
| 20:05.05 | *** join/#asterisk Greek-Boy (n=email@41.221.58.13) |
| 20:05.14 | outtolunc | aww comeon, those nosecone camera videos are pretty slick <G> |
| 20:05.26 | drmessano | Video works fine in 1.2 and 1.4 |
| 20:05.33 | drmessano | 1.6 has high def video |
| 20:05.40 | drmessano | or high quality |
| 20:06.15 | SteveTotaro | yeah, i really want to clearly see the hair sticking out of someone's mole |
| 20:06.17 | drmessano | I found the video modes in 1.2 and 1.4 to be as good or better than anything exisiting in your standard IM clients |
| 20:06.47 | drmessano | So for Vid Conf with someone using X-Lite or whatever, it worked great |
| 20:07.04 | SteveTotaro | whiteboard rulez |
| 20:07.30 | styelz | SteveTotaro: this is exactly what i WAS trying to do. http://archives.free.net.ph/message/20060718.154149.3819dd9b.en.html |
| 20:07.34 | drmessano | I have whiteboard modules in my Spark IM clients.. couldn't live without them |
| 20:07.35 | SteveTotaro | freemind with video and voice would rule for brainstorming |
| 20:07.56 | outtolunc | starts printing bumper stickers ..' you are too close to my * ' |
| 20:08.10 | styelz | order 10 |
| 20:09.44 | SteveTotaro | styelz, it sounds like you are not giving up on your splicing the analog line idea |
| 20:10.04 | styelz | yea i am.. i even premted with a caps "was" |
| 20:10.10 | outtolunc | laterz |
| 20:10.14 | SteveTotaro | bye |
| 20:10.24 | *** join/#asterisk Greek-Boy (n=email@41.221.58.13) |
| 20:10.57 | SteveTotaro | how's the new gig DRM? did you replace their pbx yet? |
| 20:11.26 | *** join/#asterisk cyberjackal (n=jackal@pool-70-109-59-80.clppva.fios.verizon.net) |
| 20:11.26 | drmessano | lol |
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| 20:12.02 | drmessano | It's good.. Been busy as hell.. Was thinking about talking to the boss this week about looking at Asterisk |
| 20:13.32 | hwt | drmessano: no, not bidirectional with Eyebeam and canreinvie=no |
| 20:13.53 | hwt | canreinvite=no* |
| 20:14.22 | drmessano | So are you telling me I didn't actually use it? |
| 20:15.00 | drmessano | Becasue I had bi-directional videp |
| 20:15.01 | drmessano | Becasue I had bi-directional video |
| 20:15.06 | drmessano | Argh, whatever |
| 20:25.24 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
| 20:25.54 | *** join/#asterisk adjohn (n=adjohn@dhcp-172-205-57-69.cf-res.cfu.net) |
| 20:28.09 | BBHoss | does asterisk 1.4 certifiably support h.264 now? |
| 20:28.29 | Corydon76-dig | It always has |
| 20:28.46 | BBHoss | thats what i thought |
| 20:28.48 | Corydon76-dig | What it won't do is transcode video |
| 20:28.57 | drmessano | h.263p and h.264 were added from 1.2 > 1.4, correct? |
| 20:28.59 | BBHoss | another thing is, why can't we transcode g722? |
| 20:29.13 | Corydon76-dig | You can |
| 20:29.30 | BBHoss | hmm, i guess i'm behind the times then |
| 20:29.43 | Corydon76-dig | g722 support was only added in 1.6, though |
| 20:30.39 | BBHoss | ahh, ok |
| 20:30.44 | BBHoss | now i remember that being in there |
| 20:30.53 | BBHoss | passthrough is in 1.4 though, correct? |
| 20:31.25 | Corydon76-dig | No idea |
| 20:31.39 | Corydon76-dig | Doubtful |
| 20:31.42 | drmessano | I _think_ it is |
| 20:32.07 | Corydon76-dig | If the bit isn't defined, then it's unsupported |
| 20:32.12 | drmessano | I'll check.. I was testing G.722 passthru but I forget if it was after I got on 1.4 or during my brief time with 1.6 |
| 20:32.45 | drmessano | Does 1.6 only support g722 passthru? |
| 20:33.11 | Corydon76-dig | No, 1.6 supports transcoding |
| 20:33.31 | drmessano | Then it had to be 1.4 then.. because I remember only being able to do passthru |
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| 20:44.43 | *** join/#asterisk mh166 (n=mh166@p5B0E5F15.dip.t-dialin.net) |
| 20:44.53 | mh166 | hi there |
| 20:45.35 | mh166 | i read on the mISDN-page that i may get support for it here |
| 20:45.46 | mh166 | so i'm going to try my luck ;) |
| 20:46.24 | mh166 | i just tried to compile my kernel (2.6.24.4) with mISDN (v.1.1.7.2) |
| 20:47.02 | mh166 | but the compiling stopped with the message: |
| 20:47.19 | mh166 | scripts/Makefile.build:46: *** CFLAGS was changed in "/mnt/hda/linux/kernel/zion/linux-2.6.24.4/drivers/isdn/hardware/mISDN/Makefile". Fix it to use EXTRA_CFLAGS. Schluss. |
| 20:47.20 | mh166 | make[3]: *** [drivers/isdn/hardware/mISDN] Fehler 2 |
| 20:47.20 | mh166 | make[2]: *** [drivers/isdn/hardware] Fehler 2 |
| 20:47.20 | mh166 | make[1]: *** [drivers/isdn] Fehler 2 |
| 20:47.20 | mh166 | make: *** [drivers] Fehler 2 |
| 20:47.21 | mh166 | root@zion:/mnt/hda/linux/kernel/zion/linux-2.6.24.4# |
| 20:47.24 | drmessano | EEK FLOOD |
| 20:47.30 | drmessano | ~pastebin |
| 20:47.31 | jbot | [~pastebin] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
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| 20:47.52 | *** mode/#asterisk [+o anthm] by ChanServ |
| 20:48.03 | mh166 | so i changed every "CFLAGS"-line in the makefile to "EXTRA_CFLAGS" |
| 20:48.23 | mh166 | but this didn't work either |
| 20:48.56 | mh166 | now he complained about something other: |
| 20:49.14 | lotho | mh166: use misdn-git repository |
| 20:49.37 | mh166 | http://pastebin.com/m1061c5e9 |
| 20:49.46 | lotho | it is fixed there |
| 20:49.51 | mh166 | uuh |
| 20:49.52 | mh166 | nice |
| 20:49.57 | mh166 | so i'll look for it |
| 20:51.04 | mh166 | is it this what you mean: http://misdn.org/index.php/GIT ? |
| 20:51.41 | lotho | http://misdn.org/index.php/Installing_mISDN |
| 20:52.19 | mh166 | okay thx, so i got the right link |
| 20:52.23 | mh166 | thank you so far :) |
| 20:52.30 | lotho | np |
| 20:52.41 | mh166 | if i should get in trouble again i'll be back ;-) |
| 20:53.12 | mh166 | cu |
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| 21:36.16 | JayTee52 | Asterisk makes it into Forbes magazine and the top of Digg's page 1!!!!! |
| 21:36.18 | JayTee52 | http://www.forbes.com/forbes/2006/0410/063.html |
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| 21:38.26 | *** part/#asterisk RoyK (n=roy@ip-108-62-149-91.dialup.ice.no) |
| 21:43.18 | drmessano | Wow |
| 21:43.24 | Qwell | 2006 O.o |
| 21:43.34 | drmessano | I just got a Twitter flood of Digg posts for Asterisk |
| 21:43.52 | JayTee52 | yeah, I posted the link above. it's at the top of the page |
| 21:44.10 | JayTee52 | good article |
| 21:44.51 | JayTee52 | but 2 years old |
| 21:45.38 | drmessano | I can't wait for FF3 |
| 21:45.46 | drmessano | Trying to view this DIgg article |
| 21:45.51 | JayTee52 | Digg has really gone downhill over the last year or so. It used to have mostly current stuff, not hyped retread crap. |
| 21:45.54 | drmessano | FF froze.. 545MB Ram used |
| 21:46.03 | JayTee52 | on which OS? |
| 21:46.09 | drmessano | Windows, of course |
| 21:46.17 | drmessano | It works elsewhere :) |
| 21:46.20 | JayTee52 | oh, cuz it's a pig on Ubuntu as well |
| 21:46.38 | drmessano | The memory management changes in FF3 are supposed to be substantial |
| 21:46.42 | drmessano | They've said that before |
| 21:46.45 | drmessano | :/ |
| 21:46.53 | JayTee52 | mine always locks up when trying to leave YouTube for another page after watching more than one video. |
| 21:47.52 | *** join/#asterisk jameswf-home (n=james@ip72-204-221-181.ph.ph.cox.net) |
| 21:48.36 | JayTee52 | and it bleeds memory like a stuck pig. If I reboot and come up clean with Compiz Fusion and AWN running only I'm using like 240MB of RAM. I run FF2 and it bumps it to over 350MB and continues climbing the longer it's open till it plateaus with 450+ MB of RAM in use. Then if I close FF I don't get the memory back. |
| 21:49.17 | drmessano | heh |
| 21:49.26 | jameswf-home | JayTee52: yor running asterisk on a box with X and all that eye candy |
| 21:49.29 | drmessano | on my machine, Firefox ITSELF uses 600MB Ram at times |
| 21:49.45 | JayTee52 | jameswf-home, umm, no. |
| 21:50.00 | drmessano | I've had it burn up to a GB of RAM on one horrid session.. was a Flash memory leak |
| 21:50.13 | JayTee52 | why would I run Compiz on an Asterisk server. I'm talking about one of my home computers. |
| 21:50.24 | jameswf-home | I have FF with like 30 tabs and thunderbird it crashes maybe 1 time a day but isnt too bad |
| 21:50.41 | jameswf-home | JayTee52: I wouldnt put it past people |
| 21:51.02 | JayTee52 | jameswf-home, yep neither would I. Jumpie maybe? |
| 21:51.20 | jameswf-home | jumoie says he runs Unix :) |
| 21:51.38 | drmessano | lol |
| 21:52.17 | jameswf-home | he wouldnt tell me what unit |
| 21:52.21 | jameswf-home | *unix |
| 21:52.50 | JayTee52 | I've got * 1.4 running on RHEL 5 64bit at work, it has 4GB of RAM and a Quad Core XEON and came with RHEL prestalled so I have Gnome loading at boot. It doesn't seem to faze Asterisk at all. |
| 21:52.52 | jameswf-home | I have a Solaris DVD havent had the time to actualy use it.... |
| 21:54.50 | JayTee52 | I had an earlier build of Nevada but it kept borking during the install on a couple machines. too busy to mess with it nowadays. |
| 21:58.01 | JayTee52 | I just had a vision of someone running Trixbox with Compiz Fusion and doing a demo for a customer. "And here's the management interface and you can put it on this desktop then you can use Ctrl-Alt-left or right arrow to switch to another desktop and run Evolution for email like this.........damn, it froze again." |
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| 22:28.32 | _ShrikE | errr. this is frustrating |
| 22:28.58 | _ShrikE | what is the command to concatenate two gsm files in sox? |
| 22:30.32 | *** join/#asterisk CCFL_Man2 (i=be2744df@pool-71-241-117-125.scr.east.verizon.net) |
| 22:31.12 | CCFL_Man2 | i was just reading an awesome article on chosing voip for a business |
| 22:32.57 | CCFL_Man2 | it says how people expect voip to work the way the pstn line does |
| 22:33.35 | CCFL_Man2 | how they get the shitty, made in china atas and expect them to work with their current phone system |
| 22:34.34 | CCFL_Man2 | most of the time they get those adapters rather than a sip account or iax trunk or whatever |
| 22:35.33 | tzafrir_home | _ShrikE, cat file1.gsm file2.gsm >concat.gsm |
| 22:35.42 | tzafrir_home | No need for sox |
| 22:37.01 | _ShrikE | tzafrir_home: thanks for the tip |
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| 22:39.44 | *** join/#asterisk R-MAN (n=no@i-83-67-127-138.freedom2surf.net) |
| 22:39.48 | R-MAN | hey guys |
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| 22:51.52 | CCFL_Man2 | some channel banks are not perfect either |
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| 22:56.32 | CCFL_Man2 | what makes the tones that you hear from a channel bank? |
| 22:56.38 | CCFL_Man2 | like dialtone, etc |
| 22:57.02 | CCFL_Man2 | does the channel bank itself make it or do they come from the T1? |
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| 23:06.40 | drmessano | CCFL_Man2: A lot of people make cheap mistakes with VoIP |
| 23:07.14 | CCFL_Man2 | yeah, they do |
| 23:08.29 | drmessano | The PBX, in a lot of businesses, is their most important piece of equipment.. so how do they respect that? They cut corners. |
| 23:08.51 | _ShrikE | CCFL_Man2: On most channel banks, the dialtone and ringing is generated on the channel bank. |
| 23:09.42 | drmessano | Imagine having a location with a PBX and being too cheap to buy a voicemail card.. so you get a dozen users Radio Shack answering machines |
| 23:10.45 | CCFL_Man2 | _ShrikE: ahh, thanks |
| 23:10.49 | JT | errr |
| 23:10.52 | JT | that is not correct |
| 23:11.00 | JT | the dialtone and ringing comes from the T1 |
| 23:11.21 | JT | obviously ringer bell voltage comes from the channel bank |
| 23:11.28 | JT | after being signalled to by the T1 |
| 23:11.44 | CCFL_Man2 | ringback and dial tone come from the T1? |
| 23:11.59 | JT | correct |
| 23:12.58 | CCFL_Man2 | when i had that cisco voice gateway originating a T1 for my channel bank, the dialtone was comming from it? |
| 23:13.15 | _ShrikE | Sorry about the confusions. I am referring to the actual ring and dt voltages. Yes, they are signaled by the T1. |
| 23:13.41 | drmessano | Signalled.. but the dialtone is not passed through as analog audio |
| 23:13.56 | CCFL_Man2 | but the audio signals are digitised on the T1? |
| 23:13.57 | *** join/#asterisk guille1983 (n=guille19@190.73.188.118) |
| 23:14.33 | JT | drmessano: you mean bell or the tone you hear when you pickup a handset |
| 23:14.46 | JT | because i refer to dialtone as the tone i hear when i pick a handset up[ |
| 23:14.51 | drmessano | Yes |
| 23:14.53 | drmessano | The dialtone |
| 23:15.13 | JT | that comes from the T1 |
| 23:15.15 | CCFL_Man2 | drmessano: so many people believe that the cheap chinese-made ATAs are just plug and play |
| 23:15.16 | JT | sampled audio |
| 23:15.20 | lesouvage | does anybody know where I can find the sourcecode used by automon. I just downloaded the sourcecode but I can't find it. |
| 23:15.31 | lesouvage | of asterisk |
| 23:16.20 | CCFL_Man2 | JT: so asterisk or whatever generate the digitally sampled dialtone? |
| 23:16.27 | JT | yes |
| 23:16.37 | JT | you can change it to any tone you want |
| 23:16.43 | JT | that is in the zaptel source |
| 23:16.46 | *** join/#asterisk baalim (n=baalim@190.199.241.35) |
| 23:16.51 | Strom_M | it's not sampled |
| 23:16.53 | Strom_M | it's generated |
| 23:17.01 | drmessano | If you can change the tone, it's not sampled |
| 23:17.07 | drmessano | It does't pitch shift |
| 23:17.09 | baalim | hello, i have a question about the asterisk appliance, is this the right place? |
| 23:17.18 | JT | it's generated, and samples are sent across the T1 |
| 23:17.19 | drmessano | It may detect DT and then generate it |
| 23:17.40 | CCFL_Man2 | yeah |
| 23:17.49 | CCFL_Man2 | samples are sent alone the T1 |
| 23:17.50 | drmessano | So how can I change the tone then, if it's sampled? |
| 23:18.03 | JT | fine, it's generated in asterisk |
| 23:18.46 | CCFL_Man2 | neat, i never knew that |
| 23:19.20 | CCFL_Man2 | so my cisco mc3810made the dialtone for the channelbank |
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| 23:20.55 | JT | CCFL_Man2: it's in zonedata.c in the zaptel source |
| 23:21.16 | JT | *if* asterisk was more flexible, you could in theory use sampled audio like a .wav for a dialtone |
| 23:21.24 | CCFL_Man2 | JT: even for the T1 cards? |
| 23:21.34 | JT | sure, why not? |
| 23:22.10 | CCFL_Man2 | oh, it's a function of zaptel itself, not the hardware drivers |
| 23:22.58 | drmessano | Dialtone, in theory, could be much of anything.. unless you have connected equipment expecting something else |
| 23:23.25 | CCFL_Man2 | could be a farting sound |
| 23:24.19 | drmessano | DT really doesn't *DO* anything |
| 23:24.19 | jameswf-home | lol I shal replace my dial tone with farts |
| 23:24.19 | CCFL_Man2 | heh |
| 23:24.19 | JayTee52 | Oompah-Loompahs singing |
| 23:24.26 | jameswf-home | Asterisk provides DT not zaptel |
| 23:24.48 | drmessano | A .wav file from your boss "This better be local!" |
| 23:25.00 | CCFL_Man2 | lol |
| 23:25.20 | jameswf-home | I am going to set al LD dialpaterns to Playback(rickroll) |
| 23:25.52 | lesouvage | drmessano: you can do that using features |
| 23:26.21 | lesouvage | drmessano: you can do that using features |
| 23:26.50 | lesouvage | drmessano: or with some other Asterisk trix |
| 23:26.53 | jameswf-home | is there an echo in here |
| 23:27.11 | jameswf-home | lesouvage: just said use trixbox |
| 23:27.12 | CCFL_Man2 | there is an echo on teh line |
| 23:27.40 | lesouvage | jameswf-home: no my mistake sorry ;-( |
| 23:27.59 | lesouvage | and I think I ment tricks |
| 23:28.24 | drmessano | lol |
| 23:28.28 | drmessano | Dont say the t word |
| 23:28.44 | jameswf-home | t = Tandy computer corp ? |
| 23:28.48 | drmessano | I don't want to be accused of starting a GUI uprising again |
| 23:29.10 | JayTee52 | hehehe |
| 23:29.25 | drmessano | ~drmessano |
| 23:29.25 | jbot | [drmessano] the leading cause of censorship in #asterisk, maybe a Doctor, not really a mess um a no..... The Doctor is accepting new patients, <drmessano> I am not a OB/GYN but I'll have a look anyway |
| 23:30.08 | lesouvage | ~lesouvage |
| 23:30.33 | lesouvage | Just curious |
| 23:30.44 | lesouvage | But this seems ok |
| 23:30.55 | CCFL_Man2 | drmessano: even for ugly womenz? |
| 23:31.14 | jameswf-home | ~me |
| 23:31.15 | jbot | when -home is added it means he is on his own time dont call his boss, or http://jameswf.info |
| 23:31.49 | drmessano | ~drmessano |
| 23:31.50 | jbot | [drmessano] the leading cause of censorship in #asterisk, maybe a Doctor, not really a mess um a no..... The Doctor is accepting new patients, <drmessano> I am not a OB/GYN but I'll have a look anyway, and is earning his reputation daily |
| 23:31.58 | drmessano | Damn skinny |
| 23:32.04 | jameswf-home | ~jameswf |
| 23:32.04 | jbot | jameswf loves unsolicited technical support, or http://jameswf.info |
| 23:32.33 | CCFL_Man2 | ~cisco |
| 23:32.34 | jbot | cisco makes the routers that move the pr0n across the internet at a very high rate of exchange. a company full of patent protected punks!, or <reply>Cisco phones are expensive crap which should be avoided with extreme prejudice |
| 23:32.44 | drmessano | hahah |
| 23:33.36 | drmessano | ~failburger |
| 23:33.37 | jbot | You fail at life. Have a failburger with fail fries and a large diet fail. |
| 23:33.43 | drmessano | Yay |
| 23:33.46 | CCFL_Man2 | lol |
| 23:33.48 | drmessano | feel free to use that |
| 23:34.17 | lesouvage | This week I read "Failure is not an option, it comes as part of the window package" I thought it was funny. |
| 23:34.38 | drmessano | niiice |
| 23:35.17 | JT | jameswf-home: if asterisk provides dialtone, why are the tones set in the zaptel source? |
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| 23:44.00 | *** part/#asterisk time2code (n=dingo@c-67-190-92-8.hsd1.co.comcast.net) |
| 23:44.59 | *** join/#asterisk cyberjackal (n=jackal@pool-70-109-59-80.clppva.fios.verizon.net) |
| 23:46.08 | plik | I think asterisk only genereates a dialtone for a DISA call - otherwise its the FXO signalling, so xaptel, or faked at the local station for most SIP/IAX hardware |
| 23:46.35 | plik | s/xaptel/zaptel/ (obvoiusly) |
| 23:47.05 | JT | correct |
| 23:47.19 | *** join/#asterisk Mavvie (n=edwin@ppp121-44-91-162.lns10.syd6.internode.on.net) |
| 23:47.20 | JT | people forget that there are two tone generators |
| 23:47.23 | JT | one in asterisk |
| 23:47.26 | JT | one in zaptel |
| 23:47.27 | plik | gone are the days when it actually signified completion of a circuit |
| 23:47.56 | plik | (well, as far as we're concerned) |
| 23:48.24 | drmessano | That's a lot different than it being generated on the T1 though |
| 23:49.12 | JT | robbed bit signalling tones are generated from zaptel |
| 23:49.25 | plik | I once read a really nice description of what exactly happend traditionally in some Mitel docs - it was about 2 pages long describing everything that went on in the fraction of a second between lifting the handset & getting dialtone |
| 23:49.53 | fujin | garrrh! mitel |
| 23:49.55 | fujin | blows up mitel |
| 23:50.20 | drmessano | I still remember the first time I picked up an ATA and realized it generated the DT and the ringing.. all in this tiny little box |
| 23:50.25 | drmessano | Just blew my mind |
| 23:50.58 | drmessano | That nothing that went on inside that box had a damn thing to do with what was occuring between the ATA and asterisk |
| 23:51.04 | JT | i don't think most ATAs generate full DC battery voltage though |
| 23:51.20 | drmessano | 48V on the PAP2s |
| 23:51.32 | JT | nice |
| 23:52.06 | drmessano | I've seen ring deciphers that didn't lol |
| 23:52.09 | drmessano | 24V.. 20V |
| 23:52.13 | ManxPower | But what is the RING voltage of them? |
| 23:52.26 | drmessano | 90V.. although I set mine at 70V |
| 23:52.27 | jameswf-home | Its funny I spent 4 hours trying to get the telco to get rid of my dialtone or stop charging me a dialtone fee.... |
| 23:52.41 | drmessano | They may go higher.. not sure |
| 23:53.41 | jameswf-home | I said if I have to have it dont charge if its optional I dont want it |
| 23:54.26 | drmessano | The PAP2s don't quite have the current your POTS lines do.. but it's more than most people seem to claim, in my testing |
| 23:56.59 | *** join/#asterisk ZaVoid (n=zavoid@c-67-165-25-195.hsd1.pa.comcast.net) |
| 23:57.07 | jameswf-home | my pap 2 powered 3 phones without flinching... |
| 23:57.45 | drmessano | I got 6 of them with electronic ringers to work just fine.. It will handle a full Western Electric bell phone too |
| 23:58.02 | drmessano | More than one phone with a bell, dunno |
