| 00:02.41 | seanjohn | exten => s,n,GotoIf($[${acode}>=365&{acode}<=384]?blacklisted) what's wrong with that? |
| 00:03.10 | seanjohn | i just tested and it returned 0 with acode 367 |
| 00:05.27 | jaytee | exten => s,n,GotoIf($[${acode}>=365 & <=384]?blacklisted) <- what about trying that? |
| 00:06.11 | seanjohn | reloa |
| 00:08.20 | seanjohn | nevermind, the missing $ |
| 00:08.50 | seanjohn | GotoIf($[${acode}>=365&${acode}<=384]?blacklisted) |
| 00:08.56 | seanjohn | that works |
| 00:09.24 | seanjohn | after i'm done excluding all freakin illegal area codes, would you like a copy lol |
| 00:09.44 | seanjohn | the prefixes are going to be hard |
| 00:10.07 | seanjohn | i'm trying to find a site where it will search through all area codes for an existing prefix |
| 00:10.22 | seanjohn | so i'll know which prefixes to exclude |
| 00:12.33 | seanjohn | 82 checks so far and i haven't reached the 400's of area codes |
| 00:15.00 | jaytee | http://puck.nether.net/npa-nxx/ |
| 00:16.57 | jaytee | actually, this site is better for finding local exchanges in an area code http://www.area-codes.com/area-code/area-code-317.asp |
| 00:17.01 | carrar | seanjohn, those aren't illegal area codes |
| 00:17.43 | seanjohn | i know, i exclude the ones NOT listed |
| 00:17.50 | seanjohn | see my long, tedious job lol |
| 00:17.54 | carrar | It's easier to download the current pnpa db and key off of unassigned NPA's |
| 00:18.13 | carrar | rather then hardcode something into your dialplan |
| 00:18.15 | seanjohn | how would I use it? mysql? |
| 00:18.24 | carrar | I use postgres |
| 00:18.36 | carrar | but you could use mysql if you really wanted too |
| 00:18.52 | seanjohn | i'm still not clear on how to use asterisk with mysql other than CDR( |
| 00:19.27 | carrar | download this, http://www.nanpa.com/npa/AllNPAs.zip |
| 00:19.36 | carrar | import that into your fav db |
| 00:19.54 | carrar | then use that to help control dialing |
| 00:21.24 | seanjohn | thats not text |
| 00:21.27 | carrar | but if you are going to block NPA's you need to keep it up to date |
| 00:21.44 | carrar | doesn't matter if it's text or not |
| 00:21.59 | carrar | it's not difficult to convert from access to postgres |
| 00:22.05 | carrar | or any other db |
| 00:22.57 | carrar | All of 2-3 mins maybe |
| 00:23.02 | seanjohn | http://mdbtools.sourceforge.net/ |
| 00:23.11 | seanjohn | its down |
| 00:23.47 | carrar | Just fireup access with a ODBC connection to whatever db you are gonna use |
| 00:24.19 | carrar | export out each table |
| 00:24.33 | carrar | well the 1 table |
| 00:25.27 | *** join/#asterisk fink (~guest@static-162-84-93-164.fred.east.verizon.net) |
| 00:25.37 | fink | if i want to use speex with asterisk, for example, i need to ensure the voip provider understand speex? |
| 00:25.59 | carrar | unless you want to do transcoding |
| 00:27.02 | carrar | otherwise yes |
| 00:27.55 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 00:28.20 | seanjohn | yes, fink. |
| 00:28.30 | seanjohn | you can use speex with your client phones |
| 00:28.36 | seanjohn | without the provider supporting |
| 00:29.00 | fink | seanjohn: right, but then i will have to transcode at asterisk, and send a different codec to hte provider? |
| 00:30.02 | Naikrovek | yes |
| 00:30.42 | Naikrovek | but it's not like you're transcoding blu-ray or anything. transcoding mono, 16-bit 8khz voice requires very little cpu |
| 00:30.47 | fink | thanks guys |
| 00:30.49 | *** join/#asterisk jsjc (~chatzilla@global269.lnk.telstra.net) |
| 00:30.58 | fink | Naikrovek: cpu isn't my concern as much as latency :| |
| 00:31.08 | Naikrovek | fink: core show translation |
| 00:31.19 | Naikrovek | that'll get you the latency numbers |
| 00:31.22 | Naikrovek | and speex is the worst of them all |
| 00:31.25 | jsjc | I have registered the fax for asterisk 1 free channel so now I was wondering with that module runing in my asterisk do I need spandsp and all that to do faxing? |
| 00:31.28 | fink | Naikrovek: really? oh ok |
| 00:31.43 | jsjc | is there any manual from the roots to understand faxing with asterisk in a simple way |
| 00:32.00 | fink | Naikrovek: do you have a recommendation for the best for a high latency connection? |
| 00:32.01 | Naikrovek | jsjc: the documentation on digium's site should give you what you want, if such documentation exists |
| 00:32.01 | jsjc | becaus everywhere they tell you this can be done this not and this as well.... none of them are too clear (for dumb people like me) |
| 00:32.25 | jsjc | Naikrovek: will read again just in case i missed something |
| 00:32.33 | Naikrovek | fink: transcoding is only going to add maybe a max of 10ms onto whatever latency you already have. |
| 00:32.40 | Naikrovek | fink: so not very much |
| 00:32.51 | fink | Naikrovek: cool, thanks |
| 00:32.58 | Naikrovek | fink: just use a codec supported by the provider (ulaw or alaw, depending on where you are) and you'll be fine |
| 00:33.09 | Naikrovek | if bandwidth is an issue, g729 may warrant some attention |
| 00:33.10 | fink | ok |
| 00:34.15 | Naikrovek | most providers support g729 and most good phones do as well. so, no translatino |
| 00:35.32 | fink | awesome, thanks |
| 00:38.08 | Naikrovek | wants http://cgi.ebay.com/NEW-SEALED-J8700A-HP-ProCurve-5412zl-96G-L3-Switch-/180492950649?cmd=ViewItem&pt=COMP_EN_Hubs&hash=item2a0637dc79 |
| 00:41.54 | TJNII | You should buy it. Only $7500 |
| 00:42.01 | p3nguin | Heck, get a couple. |
| 00:42.02 | Naikrovek | awesome price |
| 00:42.04 | Naikrovek | yeah |
| 00:42.14 | Naikrovek | gotta convince mgmt that they can't live without it first |
| 00:42.24 | Naikrovek | probably will get a couple |
| 00:42.53 | p3nguin | That's the spirit! |
| 00:44.25 | TJNII | We have a couple of those at work, buried behind a sea of cat6 |
| 00:44.27 | jsjc | looks like i do not need spandsp just the new fax for asterisk now i just need to find some sample configs to give me an idea.... anyone know somewhere to get sample configs for fax for sterisk? |
| 00:44.37 | *** join/#asterisk troubled (~troubled@unaffiliated/troubled) |
| 00:44.50 | jsjc | hehe o the next page of the documentation.... |
| 00:44.58 | jsjc | i thought was finsihed sorry for bothering! |
| 00:45.05 | Naikrovek | not a bother at all |
| 00:45.17 | Naikrovek | those switches are awesome. expandable. |
| 00:45.20 | Naikrovek | lifetime warranty |
| 00:45.29 | Naikrovek | they'll probably remain around for 10 years |
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| 00:48.05 | Naikrovek | even list price for those is half what cisco wants for the same thing |
| 00:49.52 | TJNII | We've got a couple ungodly cisco switches floating around. Things are like 20U, I don't even know how many ports |
| 00:51.34 | Naikrovek | 6509s |
| 00:51.42 | Naikrovek | they're oooold |
| 00:51.45 | Naikrovek | if they're the 6509s |
| 00:52.00 | Naikrovek | think they can do 288 10/100 ports |
| 00:52.25 | Naikrovek | what's the name of the tool you use to patch cables into patch panels |
| 00:52.26 | TJNII | They're not in use |
| 00:52.44 | TJNII | One is almost literally "floating around" in a 30U mobile rack |
| 00:52.52 | TJNII | It is buried in a corner of the lab |
| 00:52.59 | Naikrovek | they're worthless these days |
| 00:53.03 | TJNII | Punchdown tool? |
| 00:53.12 | Naikrovek | yeah is it just a 110 punchdown tool |
| 00:53.26 | Naikrovek | or whatever that old phone style punchdown tool was called |
| 00:53.43 | TJNII | I don't think they're 6509s |
| 00:53.53 | TJNII | I'll have to look for a model number Monday |
| 00:54.27 | TJNII | I don't punch cables so I don't know. |
| 00:56.05 | Naikrovek | product page says 110 and krone |
| 00:56.09 | Naikrovek | has his answer |
| 01:03.03 | enyawix | Naikrovek old style punchdown tool was called 66 style. most 110 tools are 66 on the other side |
| 01:03.14 | Naikrovek | ah neat |
| 01:03.16 | Naikrovek | thanks |
| 01:03.42 | enyawix | i would not use a 66 block |
| 01:04.08 | Naikrovek | there is a huge amount of 110 punchdown blocks in the utility room downstairs, probably 1000 lines worth. |
| 01:04.19 | Naikrovek | wonder what this place was before we moved in here... |
| 01:04.33 | enyawix | 110 blocks make better phone connections and they are the same price |
| 01:04.48 | Naikrovek | enyawix: this is for a belkin patch panel, didn't know which tool to use to punch wires in there |
| 01:04.53 | Naikrovek | suspected 110 but wasn't sure |
| 01:05.10 | enyawix | patch panel is 110 |
| 01:11.34 | *** join/#asterisk Naikrovek (~jjohnson@unaffiliated/naikrovek) |
| 01:12.33 | Naikrovek | wow win7 crashed on me |
| 01:12.39 | Naikrovek | bsod |
| 01:12.49 | Naikrovek | wonder which piece of hardware did that |
| 01:15.23 | enyawix | I wish windows would go to a bsd base |
| 01:16.26 | Naikrovek | far too many things to port to a BSD core to even consider that, i'm guessing |
| 01:16.44 | Naikrovek | they're slowly morphing into their own unix though |
| 01:17.18 | Naikrovek | drop the drive letters and change to a '/' based filesystem and they'll be pretty damn close |
| 01:20.37 | enyawix | i would want a bash clone as well |
| 01:22.14 | enyawix | anyone have 33.6 fax working? how did you go about it? i can not seem to find a foip machine |
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| 01:51.08 | devdvd | can you do ackcall with asterisk 1.6?, and if not, with asterisk 1.4 can i do it with a static agent |
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| 04:01.11 | davidstrauss | I just migrated configuration from 1.4 to 1.6. I've worked out most of the changes on my own, but VoiceMailMain doesn't work. It says first that it's playing vm-login (it doesn't), and then it just says goodbye. |
| 04:02.03 | davidstrauss | I've tested just playing vm-login, and that works. |
| 04:02.19 | davidstrauss | There's no obvious error in a super-verbose shell |
| 04:11.00 | *** join/#asterisk jmcdowell (~airmadnes@173-112-252-75.pools.spcsdns.net) |
| 04:11.03 | jmcdowell | hello all.. |
| 04:11.22 | jmcdowell | anyone with good ptsn termination providers, please msg me |
| 04:11.27 | jmcdowell | I am dropping callcentric. |
| 04:13.23 | davidstrauss | jmcdowell: I like Vitelity |
| 04:13.45 | jmcdowell | I will check them out |
| 04:14.14 | jmcdowell | 1.39 per minute ? |
| 04:15.20 | davidstrauss | jmcdowell: cents, maybe |
| 04:15.45 | davidstrauss | jmcdowell: and only outbound |
| 04:15.51 | jmcdowell | ahhh |
| 04:15.54 | jmcdowell | i get it |
| 04:15.55 | jmcdowell | lol |
| 04:16.21 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 04:19.00 | *** join/#asterisk JAMMAN2110 (~JAMMAN211@unaffiliated/jamman2110) |
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| 04:40.09 | jmcdowell | davidstrauss : Can you give me an example of a bil from these guys for 1 month ? |
| 04:40.12 | jmcdowell | Round numbers ? |
| 04:40.36 | davidstrauss | jmcdowell: We're an 8-person company, and I think we're paying $100-200/mo |
| 04:40.43 | jmcdowell | hmmm |
| 04:40.53 | jmcdowell | I am just a house.. |
| 04:41.07 | jmcdowell | Callcentic went all non-standard.. |
| 04:41.26 | jmcdowell | So I dropped the, starting to look as though we will be without a proiver for while |
| 04:41.31 | davidstrauss | jmcdowell: Vitelity just runs Asterisk |
| 04:42.27 | jmcdowell | CC did all sorts of wierd things that make them a PITA |
| 04:42.40 | jmcdowell | I just need 2 dids with one trunk |
| 04:42.50 | jmcdowell | ppm - |
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| 04:57.31 | TJNII | pokes infobot |
| 04:58.03 | TJNII | is disappointed by the lack of response. |
| 05:02.03 | p3nguin | wonders what's the problem with using Vitelity, VoIP.ms, or Flowroute for services. |
| 05:03.10 | p3nguin | *shrug* I guess I shouldn't care if he's without phone service. |
| 05:07.03 | davidstrauss | Can anyone help me figure out why my VM config for 1.6 just goes right to "Goodbye" on VoiceMailMain? |
| 05:07.33 | *** join/#asterisk pav5088 (~Mark@ppp118-208-88-52.lns20.bne4.internode.on.net) |
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| 05:08.31 | pav5088 | Hi... most Asterisk GUI's seem to be CentOS based... I know there was DeStar years ago, but is there anything packaged for modern Debian based distros? |
| 05:08.59 | *** join/#asterisk Brookss (~SSJGotenk@174.3.119.13) |
| 05:09.34 | *** join/#asterisk cesar_CR (~cesar@201.201.41.242) |
| 05:10.02 | TJNII | Pfft... who needs a GUI. |
| 05:10.15 | TJNII | Just install asterisk and be done with it. |
| 05:11.12 | Brookss | quit tryna sound l33t TJ, we all need FreePBX otherwise how could anyone deal with the madness |
| 05:11.28 | TJNII | Newbie. |
| 05:11.46 | TJNII | I've been running source for 3 years AND THAT"S THE WAY I LIKES IT! |
| 05:12.19 | Brookss | quit tryna sound l33t I said, githead XD |
| 05:12.32 | TJNII | I am ;33t/ |
| 05:12.42 | TJNII | s/;33t/l33t |
| 05:12.52 | TJNII | is also somewhat drung |
| 05:13.20 | TJNII | I earned that title when I found myself explaining the nuances of ARP and DHCP to my coworkers |
| 05:13.22 | JAMMAN2110 | On the internet, no one cares |
| 05:13.34 | TJNII | This amn speeks truth |
| 05:13.58 | TJNII | I could call you a donkey f*cker, and nobody would care but you. |
| 05:14.03 | JAMMAN2110 | Hurrrrrrrr i so l3t3 cuz i haxd da govt ma'n |
| 05:14.09 | TJNII | Indeed! |
| 05:14.14 | JAMMAN2110 | :) |
| 05:14.30 | JAMMAN2110 | But I agree, Asterisk without FreePBX is quite easy |
| 05:14.35 | TJNII | makes a note to get revenge on JAMMAN2110 when sober |
| 05:14.47 | JAMMAN2110 | Why are you seeking revenge? |
| 05:14.54 | TJNII | Oh, you know. |
| 05:15.25 | TJNII | YAY! I HAVE INTERNET FRIENDS! |
| 05:15.56 | JAMMAN2110 | Congratulations |
| 05:16.01 | JAMMAN2110 | I hope none of them play Neopets |
| 05:16.08 | TJNII | wanders off for burritos. |
| 05:17.46 | Brookss | i played yugioh b4... still have my cards XD |
| 05:18.36 | Brookss | before that it was pokemon... |
| 05:18.47 | JAMMAN2110 | I stopped at Pokemon |
| 05:18.58 | JAMMAN2110 | When they added more than the original Pokemon I got pissed off |
| 05:19.14 | Brookss | you mean after the 151, yea |
| 05:19.28 | Brookss | (mewtwo was 1) |
| 05:19.41 | JAMMAN2110 | So realistic "The whole world only has 151 pokemon" |
| 05:19.48 | JAMMAN2110 | Then suddenly "Oh it has 5000 now" |
| 05:20.38 | Brookss | I could sing the song, 'catchem catchem gotta cat..' and then say all the pokemon in song XD ... ya who the hell is going to keep paying attention if they make it that many :o |
| 05:21.43 | Brookss | don't even get me started on digimon |
| 05:24.52 | Brookss | who tells you where to start then learn to use asterisk cli |
| 05:26.56 | pav5088 | I've been writing documentation on how to use GOsa (ie. the software to manage an LDAP based infrastructure)... Noone seems to have heard of it, although apparently Munich uses it for their Linuxification (and other european cities such as Paris and Amsterdam) |
| 05:27.38 | pav5088 | There's an Asterisk module, but it's only for managing handsets, peoples details etc... I don't think it handles dialplans etc... etc... |
| 05:27.48 | p3nguin | pav5088: FreePBX is not "based on" any Linux distribution. You may download and install FreePBX on Debian if that's the sorta thing that makes you feel good. |
| 05:28.36 | *** join/#asterisk sourcode (~code@ppp-61-90-15-25.revip.asianet.co.th) |
| 05:28.56 | dzup | support for * is dropping anytime soon from freepbx ? |
| 05:28.57 | TJNII | p3nguin: Don't encorage him! |
| 05:29.19 | TJNII | My god man, you've been here long enough to know that. |
| 05:29.22 | p3nguin | As far as I can tell, Asterisk is the only thing FreePBX controls. |
| 05:29.30 | pav5088 | Well... users like "easy", so if I'm not going to get the sack when Australia finally gets decent internet because of the cloud offerings from Amazon et al I need to learn how to do "easy". |
| 05:29.38 | p3nguin | But, since I don't use it, I don't actually have any idea. |
| 05:30.12 | pav5088 | Most techies hate Apple too... but the market talks. *shrugs* |
| 05:31.02 | ManxPower | pav5088, we don't care what you use, but FreePBX is off topic on this channel. |
| 05:31.12 | p3nguin | That is true. |
| 05:31.40 | pav5088 | ...and Debian is off topic in the FreePBX channel... *shrugs* things fall between the cracks. |
| 05:31.50 | TJNII | pav5088: Yea, go Macs with 7.2% market share WHOOOO! |
| 05:32.25 | pav5088 | TJNII, it's better than Linux... |
| 05:32.38 | TJNII | Dem's fightin' words, boy.... |
| 05:32.44 | pav5088 | ...though Ubuntu seems to be making a dint by pandering to end users. |
| 05:32.46 | p3nguin | What if my Mac runs Linux? |
| 05:33.32 | TJNII | I thought they all did now, or at least BSD.... |
| 05:33.43 | Brookss | what if your mac is build from parts of freebsd which is close to linux, but you choose to run parallels to do ubuntu over the mac |
| 05:34.11 | pav5088 | TJNII, well, they put a nice front end on it... which is kind of what my first question related to. |
| 05:34.43 | Andrew_M | What if you use your scooter as a bycicle, can you go on the bycycle lane? |
| 05:35.05 | TJNII | pav5088: I'm sorry, what was your first question again? |
| 05:35.11 | p3nguin | If you have a friend with you, you can go in the car pool lane. |
| 05:36.13 | ManxPower | p3nguin, a mac running linux? I think the term is "masochist". |
| 05:36.16 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 05:36.24 | pav5088 | TJNII, how to get myself a deb for an asterisk GUI... asterisk and op-panel seem to be bundled, but I guess I'm looking for the 3rd party repo for FreePBX, Elastix or whatever. |
| 05:36.52 | ManxPower | have fun, don't bother to tell us how it works out. |
| 05:36.53 | p3nguin | Elastix is an entire distro as far as I know. |
| 05:37.12 | TJNII | Oh. right. |
| 05:37.51 | p3nguin | FreePBX is just another piece of software that you install on your existing OS to blow apart Asterisk and make it "easy." *sigh* |
| 05:39.25 | JAMMAN2110 | Ubuntu is growing in market share |
| 05:39.48 | pav5088 | p3nguin, I'm sure the oldschool PBX guys talking same way about VOIP and telephony. |
| 05:41.28 | TJNII | Spin it however you want, man. Grow a pair and configure your PBX without shiny buttons and menus. |
| 05:42.04 | TJNII | 'Cus otherwise in 2 weeks you'll be back here asking why it doesn't do what you want. |
| 05:43.09 | p3nguin | It really doesn't even take testicular fortitude to admin a box without a GUI. It just takes some reading to develop an eventual skill for not needing to point and click to get things done. |
| 05:43.38 | p3nguin | We have a term for people that don't want to read and learn. |
| 05:43.43 | p3nguin | "lazy" |
| 05:44.02 | Naikrovek | not even that. just learn how to read and write the dialplan and contexts. |
| 05:44.10 | pav5088 | TJNII, right after you take up growing your own food, servicing your own car and generating your own electricity. People get simplification wrong lots, but once simplicity evolves enough it wins. Simple, cheap, ubiqitous. |
| 05:44.15 | Naikrovek | but whatever, freepbx fits for some people |
| 05:44.27 | Naikrovek | i use it |
| 05:44.34 | Naikrovek | on another server, i don't |
| 05:44.35 | Brookss | asterisk cant imagine a world without freepbx |
| 05:44.48 | p3nguin | Of course it does. Just because you want to use it does not make you lazy, but using it because you're lazy speaks for itself. |
| 05:44.59 | Naikrovek | asterisk has no imagination, so you're kinda right |
| 05:45.00 | TJNII | pav5088: Eh, 2 outta 3 ain't bad |
| 05:45.07 | TJNII | has a bropwn thumb |
| 05:45.23 | Naikrovek | wtf is a brown thumb |
| 05:45.31 | TJNII | My plants all die. |
| 05:45.34 | Naikrovek | lol |
| 05:45.36 | p3nguin | kills vegetation |
| 05:45.41 | pav5088 | TJNII, well, I've done all of the above... but I prefer not to have to. |
| 05:45.42 | p3nguin | opposite of green thumb |
| 05:45.46 | ChannelZ | Whew. I thought it was a toilet joke. |
| 05:45.49 | p3nguin | lol |
| 05:45.54 | p3nguin | At first I did too! |
| 05:45.58 | TJNII | Yay! ChannelZ is here! |
| 05:46.02 | TJNII | Poop joke time! |
| 05:46.04 | ChannelZ | Toiletpaper Malfunction |
| 05:46.06 | Naikrovek | opposite of green thumb would be red thumb? color wheel |
| 05:46.14 | TJNII | Commie, |
| 05:46.14 | Naikrovek | caca humor is funny |
| 05:46.21 | p3nguin | technicalities, naikrovek |
| 05:46.26 | ChannelZ | Except plants don't turn red when they die |
| 05:46.42 | Naikrovek | your thumb turns red when you jab it with a hand trowel |
| 05:46.50 | Naikrovek | trying to dig up a potato |
| 05:47.00 | pav5088 | TJNII, word of the day - faecolith |
| 05:47.12 | Naikrovek | feh i'm tired, my jokes suck at the moment. at every moment, really |
| 05:47.28 | ChannelZ | Naikrovek: But you're funny LOOKING |
| 05:47.31 | Naikrovek | migrating from a 1tb san to an 8tb san via USB disk |
| 05:47.33 | TJNII | An enema should fix that. |
| 05:47.34 | Naikrovek | ugh |
| 05:47.45 | ChannelZ | USB3 I hope |
| 05:47.58 | TJNII | I don't think that is a USB port. |
| 05:48.01 | Naikrovek | well usb2 but it's going pretty damn slow |
| 05:48.03 | TJNII | Though it may fit..... |
| 05:48.13 | p3nguin | USB enema |
| 05:48.38 | Naikrovek | backed up 1tb san to 250gb USB disks, replaced disks in SAN, all that, now putting data back on |
| 05:48.40 | ChannelZ | that reminds me I was maybe going to upgrade my server to the new Ubuntu tonight |
| 05:48.41 | Naikrovek | what a PITA this is |
| 05:48.55 | ChannelZ | make it x64 now that I have a CPU made in this decade |
| 05:49.10 | ChannelZ | Has anyone installed off a USB stick? |
| 05:49.15 | JAMMAN2110 | Yep |
| 05:49.17 | JAMMAN2110 | USB is easy |
| 05:49.34 | ChannelZ | I think I have a bootable one all setup but the docs are somewhat conflicting and vague.. they say you just copy the .iso file on there too? |
| 05:49.36 | JAMMAN2110 | Considering updating mine |
| 05:49.39 | JAMMAN2110 | But |
| 05:49.43 | JAMMAN2110 | I have Asterisk compiled on it |
| 05:50.23 | ChannelZ | the only part I'm not looking forward to is getting my whole qmail system back up |
| 05:51.14 | TJNII | Qmail is easy once you remember all the stupid shit you had to wade through to get it working in the first place.... |
| 05:51.41 | TJNII | SMTP auth? Who needs that! |
| 05:51.53 | Brookss | t33ch me your l33t ways TJ |
| 05:52.01 | pav5088 | p3nguin, the best nerds are lazy... they spend more effort up front so they don't have to eg. waste brain space on syntax that has nothing to do with the actual problem. |
| 05:52.41 | TJNII | Brookss: Read the RFCs for ARP, IP, UDP, and BOOTP and we'll talk. |
| 05:52.53 | ChannelZ | I have mine pretty well patched up, I just need to make sure I have the right source tree backed up |
| 05:53.13 | Brookss | TJ I can do that EASILY, however I won't understand a bit of it |
| 05:53.20 | TJNII | Touche! |
| 05:53.39 | TJNII | Oh, and ICMP. That is a good one. |
| 05:56.02 | ChannelZ | fo shizzle |
| 06:05.03 | *** join/#asterisk \malex\ (HHNteFDk@unaffiliated/malex/x-000000001) |
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| 06:17.07 | *** part/#asterisk pav5088 (~Mark@ppp118-208-88-52.lns20.bne4.internode.on.net) |
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| 06:46.11 | Naikrovek | i love windows 7 but I hate whoever wrote the "Time remaining" algorithm for file copies. 30 minutes. 12 hours. 2 hours. 24 hours. 15 minutes. 8 hours. |
| 06:48.30 | coppice | Naikrovek: http://xkcd.com/612/ |
| 06:49.08 | Naikrovek | hah |
| 06:49.10 | Naikrovek | exactly |
| 06:51.04 | coppice | Windows 7 is so good its actually not much worse than XP |
| 06:53.55 | Brookss | ahahaha, a genius I once knew... who got fired for other reasons than his knowledge said to count in 15minute increments it makes it easier than saying 'I worked for 24 minutes' instead do '30 minutes' |
| 06:56.30 | Naikrovek | i soooo much prefer win 7 to XP |
| 06:56.41 | Naikrovek | xp came out almost 10 years ago now |
| 06:58.35 | Brookss | remember when everyone preferred win2000 to xp... or 98 to winme? it always ended up being over resource usage |
| 07:01.45 | ManxPower | I prefer to run newer hardware with older OS |
| 07:04.58 | coppice | Win2000 was a terrible resource hog. Machines moved from NT 4,1 to Win2000 were usually more stable, but desperately slow |
| 07:05.52 | coppice | though if you ever used NT 4.1 on an Alpha, you will never have seen it do a BSOD on you |
| 07:06.10 | Brookss | and THATS when they found out 128MB's was NOT going to be enough |
| 07:09.29 | *** join/#asterisk SunnyDP (~scan@bas1-montreal27-1279505166.dsl.bell.ca) |
| 07:09.49 | Brookss | I remember me and my best friend had old computers and he brought his over, I gave him a 32 or a 64mb stick, which was alot but... we were poor so we were also 15 years behind everyone else... ;'-( I miss those days... its nostalgia.. having that 486DX and saying he could now do Win98 instead of 95 even tho XP was out by then |
| 07:12.23 | Brookss | *sigh* ANYWAYS /clear |
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| 07:19.14 | *** join/#asterisk Yudaisrael1984 (~Yuda@77.127.144.138) |
| 07:20.32 | Yudaisrael1984 | can someone here please recomend a good firewall that works well with voip for commercial use? (capabilities needed is nothing more then regular firewall policys with no smart application features or anything like that and that it should be able to work 100% with voip) |
| 07:24.45 | ChannelZ | those two requests seem to be at odds |
| 07:25.34 | ChannelZ | You need one with 'smart application features' in order to 'work 100% with voip' |
| 07:28.48 | Brookss | Fortigate firewall |
| 07:28.54 | Brookss | if you ever find the cash |
| 07:29.13 | fenrus | stonegate?, cisco asa?, zyxey zywall? |
| 07:29.24 | fenrus | there's probablu hundreds of firewalls that work great |
| 07:33.59 | *** join/#asterisk kartik (~koolkarti@117.199.113.180) |
| 07:53.14 | Naikrovek | cisco asa is nice |
| 07:53.28 | Naikrovek | entry level one is $400 i think |
| 07:53.32 | Naikrovek | probably cheaper solutions about |
| 07:55.40 | *** join/#asterisk lanning (~lanning@173-8-187-197-SFBA.hfc.comcastbusiness.net) |
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| 08:58.08 | Yudaisrael1984 | brookss fortigate firewall DOES NOT WORK WITH VOIP |
| 08:58.24 | Yudaisrael1984 | i have one now it screwed me over big time |
| 08:58.54 | Yudaisrael1984 | anyone know about juniper |
| 09:05.26 | *** join/#asterisk blaines (~blaines@ip68-106-24-21.ph.ph.cox.net) |
| 09:09.57 | Yudaisrael1984 | <Brookss> have u managed to get the fortigate to work with voip?? if so how? |
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| 09:41.21 | *** join/#asterisk JAMMAN2110 (~JAMMAN211@unaffiliated/jamman2110) |
| 09:47.46 | *** join/#asterisk niekie (~niek@CAcert/Assurer/niekie) |
| 09:51.11 | Brookss | test |
| 09:52.13 | JAMMAN2110 | Test failed |
| 09:52.14 | Brookss | back from watching a movie |
| 09:52.16 | JAMMAN2110 | Please try again |
| 09:52.24 | Brookss | Test test... testing 123 |
| 09:52.32 | JAMMAN2110 | had to try that |
| 09:52.57 | Brookss | JAMMAN!!! |
| 09:53.06 | JAMMAN2110 | Hello |
| 09:53.47 | Brookss | what are you doing |
| 09:54.23 | Brookss | is thinking of perusing /. for the latest scoop |
| 10:20.07 | *** join/#asterisk af_ (~getsmart@88-149-240-255.dynamic.ngi.it) |
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| 11:05.30 | *** join/#asterisk Faithful (~Faithful@202.6.145.116) |
| 11:10.57 | *** join/#asterisk ChannelZ (~bobm@burner.com) |
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| 11:50.57 | carrar | Y*A*W*N |
| 11:51.30 | *** join/#asterisk crazybyte (~crzp@unaffiliated/crazypenguin/x-000001) |
| 11:58.39 | ChannelZ | definately |
| 12:00.18 | *** join/#asterisk Yudaisrael1984 (~Yuda@77.127.144.138) |
| 12:01.01 | Yudaisrael1984 | anyone here know fortigate firewalls? or can guide me to the correct place for help with a fortigate firewall |
| 12:02.59 | ChannelZ | Try Google |
| 12:03.19 | Yudaisrael1984 | tried nothing is mentioned there about what i need help with |
| 12:03.45 | Yudaisrael1984 | i need to set up the firewall that the phones behind it will talk to my asterisk server |
| 12:06.57 | carrar | Yudaisrael1984, are they not working? |
| 12:07.09 | Yudaisrael1984 | not with voip |
| 12:07.14 | carrar | Make sure to enable NAT on the phones and Asterisk |
| 12:07.20 | Yudaisrael1984 | i did on both |
| 12:07.26 | carrar | and turn off any ALG or fixup protocols on the firewall |
| 12:07.31 | Yudaisrael1984 | im trying to |
| 12:07.35 | carrar | or any SIP helpers |
| 12:07.38 | Yudaisrael1984 | there are sooooooo many |
| 12:07.49 | carrar | Refer to the instructions for your firewall |
| 12:07.51 | *** join/#asterisk cesar_CR (~cesar@201.201.41.242) |
| 12:07.55 | Yudaisrael1984 | im looking at it |
| 12:08.08 | Yudaisrael1984 | its in front of me this second and i turned most of the things off |
| 12:09.34 | carrar | Yudaisrael1984, your asterisk server is not behind NAT correct? |
| 12:09.39 | Yudaisrael1984 | nope |
| 12:09.50 | Yudaisrael1984 | legal ip not behind any firewall |
| 12:10.14 | carrar | What does tha tmean |
| 12:10.34 | carrar | Are you suggesting the asterisk box is behind a NAT server |
| 12:10.38 | Yudaisrael1984 | that it has a internet ip address (no nat) |
| 12:10.49 | Yudaisrael1984 | no it does not have a nat |
| 12:11.14 | carrar | THen the phones should work fine behind a NAT/fw |
| 12:11.27 | Yudaisrael1984 | and its not thats why im here looking for help |
| 12:11.31 | carrar | assuming they are configured as such |
| 12:11.56 | Yudaisrael1984 | i agree there shouldnt be any problems although there is and thats why i am here |
| 12:12.08 | Yudaisrael1984 | i have been working on it for 5 days now |
| 12:12.13 | carrar | What does your packet dumps say? |
| 12:12.15 | Yudaisrael1984 | went thru every detail |
| 12:12.39 | carrar | In the CLI of the fortigate type the following: |
| 12:12.39 | carrar | config system settings |
| 12:12.39 | carrar | set sip-helper disable |
| 12:12.39 | carrar | set sip-nat-trace disable |
| 12:12.41 | Yudaisrael1984 | i came to the conclusion that the firewall is changing the contact info even though it is set not to |
| 12:12.42 | carrar | then reboot |
| 12:13.08 | Yudaisrael1984 | i have sip status disabled nat-trace disabled |
| 12:13.08 | carrar | Config system session-helper |
| 12:13.09 | carrar | show |
| 12:13.09 | carrar | (now look for SIP, mostly it will be "12") |
| 12:13.09 | carrar | delete 12 |
| 12:14.19 | Yudaisrael1984 | there is no sip |
| 12:14.24 | Yudaisrael1984 | it was deleted already |
| 12:16.44 | Yudaisrael1984 | any other ideas? |
| 12:17.00 | Yudaisrael1984 | i took off the replacing of contact header |
| 12:17.04 | Yudaisrael1984 | still nothing doing |
| 12:17.43 | carrar | Does the phone work if you don't use the firewall |
| 12:17.51 | carrar | verify it |
| 12:18.13 | Yudaisrael1984 | yes |
| 12:18.41 | Yudaisrael1984 | other clients are working (clients that are NOT behind a fortigate firewall) |
| 12:19.12 | carrar | Are they behind some other firewall? |
| 12:19.33 | Yudaisrael1984 | yes |
| 12:19.43 | Yudaisrael1984 | some are behind basic linux firewalls others are not |
| 12:19.44 | carrar | good thing tomorrow is Monday then :) |
| 12:19.52 | Yudaisrael1984 | this is from thursday |
| 12:19.54 | Yudaisrael1984 | my luck |
| 12:20.18 | Yudaisrael1984 | i had my server within the firewall before but then nothing worked |
| 12:20.21 | carrar | Monday being, you can get support from fortigate |
| 12:20.35 | Yudaisrael1984 | do they give support on the phone? |
| 12:20.39 | Yudaisrael1984 | or only with a ticket? |
| 12:20.46 | carrar | I don't use fortigate |
| 12:20.55 | carrar | no idea |
| 12:21.01 | Yudaisrael1984 | oh so then how did u have all that info |
| 12:21.11 | carrar | I was googling for you |
| 12:21.34 | Yudaisrael1984 | hehe i googled couldnt find it (im just not lucky this weekend) |
| 12:21.48 | carrar | "asterisk fortigate" |
| 12:22.06 | Yudaisrael1984 | and my isp experts who claim they know everything about the fortigate say its not possible to do so many sip phones behind a fortigate firewall |
| 12:22.15 | Yudaisrael1984 | and i think that they dont know what they are talking aobut |
| 12:22.19 | carrar | THen it's crap |
| 12:22.23 | carrar | dump that PoS |
| 12:23.05 | carrar | I have DSL Modems that work better then that |
| 12:23.29 | *** join/#asterisk rossand (~aross@CPE000c413a19a3-CM00159a025ad4.cpe.net.cable.rogers.com) |
| 12:23.45 | carrar | but might give fortigate a call just to be sure |
| 12:24.02 | carrar | or email or whatever you have to do |
| 12:33.59 | *** join/#asterisk orioni (~chatzilla@95.107.235.217) |
| 12:34.18 | orioni | hi there , is it possible to use a gsm modem as a gateway on asterisk ? |
| 12:34.42 | orioni | i have 3 for every provider and i want to do a lcr for the outgoing calls |
| 12:36.45 | Yudaisrael1984 | carrar just restarted again this time it worked |
| 12:40.44 | carrar | restarted the fw? |
| 12:40.53 | carrar | you did do that last time correct? |
| 12:41.44 | *** join/#asterisk cesar_CR (~cesar@201.201.41.242) |
| 12:42.39 | carrar | orioni, a cellular GSM modem? |
| 12:42.48 | orioni | yes |
| 12:42.55 | carrar | with a analong output? |
| 12:43.04 | carrar | or sip? |
| 12:43.11 | orioni | analog rj11 |
| 12:43.18 | carrar | All you need is a ATA then |
| 12:43.31 | orioni | which connects to my alcatel pbx |
| 12:43.49 | orioni | so an ATA for every of my gsm |
| 12:43.53 | carrar | yup |
| 12:44.13 | carrar | Audio codes makes nice quality ones that support 4, 8, 16 24 etc.. |
| 12:44.26 | orioni | and how to configure the dialplan to use ata based on the destination |
| 12:44.27 | carrar | You pick FXO or FXS |
| 12:44.33 | carrar | or mix and match |
| 12:44.51 | orioni | looking for a reference , a general idea |
| 12:44.51 | carrar | depends what ATA but you can tell it what chanel to use |
| 12:45.13 | carrar | google "asterisk ATA config" or analog device |
| 12:45.19 | carrar | lots of examples out there |
| 12:45.22 | orioni | sth like asterisk and analog gsm modems |
| 12:46.15 | carrar | for GSM modems I'd go with a SIP based ATA |
| 12:46.54 | orioni | i have a spa2102 |
| 12:47.00 | orioni | will that work ? |
| 12:47.03 | carrar | Asterisk -> sip -> ATA -> analog line -> gsm modem |
| 12:47.20 | carrar | never used one |
| 12:47.30 | carrar | is it FXO or FXS? |
| 12:47.36 | orioni | fxs |
| 12:47.36 | carrar | and what is the the GSM output? |
| 12:47.58 | orioni | its the ATA not the gsm modem |
| 12:48.03 | orioni | linksys spa2102 |
| 12:48.12 | carrar | but the GSM has a fxs or fxo analog port |
| 12:48.16 | carrar | you sid |
| 12:48.16 | *** join/#asterisk coppice (~chatzilla@191.193.17.210.dyn.pacific.net.hk) |
| 12:48.17 | carrar | said |
| 12:48.28 | orioni | http://www.voip.com/images/spa2102_step1.jpg |
| 12:48.52 | carrar | I suspect the GSM is also FXS |
| 12:49.25 | orioni | you connect a phone to the gsm modems |
| 12:49.36 | carrar | so yea |
| 12:49.38 | carrar | FXS |
| 12:49.40 | orioni | and you can make outgoing calls using the sim card on the gsm modem |
| 12:49.44 | orioni | yep |
| 12:49.49 | carrar | You can;t connect two FXS devices to each other |
| 12:49.59 | orioni | true |
| 12:50.01 | carrar | one needs to be FXO other FXS |
| 12:50.23 | carrar | So you need a FXO ATA |
| 12:50.42 | orioni | any brand / version |
| 12:50.50 | carrar | I'm partial to audio codes |
| 12:50.51 | orioni | low cost :) |
| 12:51.00 | carrar | but they are expensive |
| 12:51.15 | carrar | ot ADIC600 with FXO blades and a T1 card |
| 12:51.17 | orioni | which model of AC |
| 12:51.34 | carrar | how many GSM Modems you gonna connect? |
| 12:52.09 | orioni | 3 |
| 12:52.17 | orioni | one for every provider here |
| 12:52.36 | carrar | They make a 4 port FXO ATA |
| 12:52.47 | carrar | SIP out the otherside |
| 12:53.02 | *** join/#asterisk Bartockbatz (~szilverth@c-24-62-161-95.hsd1.nh.comcast.net) |
| 12:53.17 | orioni | didnt get this |
| 12:53.35 | orioni | so 3 port to the modems and the last to the asterisk ? |
| 12:53.55 | carrar | Audio Codes MP-114 with FXO |
| 12:54.11 | carrar | http://www.audiocodes.com/products/mediapack-1xx# |
| 12:54.29 | carrar | no |
| 12:54.34 | carrar | 3 ports to the GSM Modems |
| 12:54.36 | carrar | 1 spare |
| 12:54.41 | Bartockbatz | Hi - questtion for everyone - Asterisk 1.4 - would like to be able to display incoming caller ID from my SIP trunk. Would like to make sure that I am able to view this in a softphone client such as X-Ten or Ninja Lite. Thanks for anyone and everyone's assistance! |
| 12:54.44 | carrar | 1 Ethernet doing SIP to Asterisk |
| 12:55.32 | carrar | You can register the AC as 1 group with each device as a rollover |
| 12:55.41 | carrar | or 3 seperate ATA's |
| 12:55.59 | carrar | very flexible |
| 12:56.09 | orioni | got it |
| 12:56.13 | orioni | thanx man |
| 12:56.28 | carrar | I am sure there are cheaper ways to do it however |
| 12:56.47 | carrar | but if I were to build it for a client |
| 12:56.51 | carrar | I'd pick that |
| 12:56.57 | carrar | or something compairable |
| 12:58.09 | orioni | i c |
| 12:59.38 | carrar | Bartockbatz, Whats the question? |
| 12:59.43 | coppice | http://www.voip-info.org/wiki/view/VOIP+GSM+Gateways has a number of options for GSM to SIP |
| 12:59.45 | *** join/#asterisk pinoyskull (~pinoyskul@112.198.64.91) |
| 12:59.45 | *** join/#asterisk cesar_CR (~cesar@201.201.41.242) |
| 13:00.16 | carrar | but you already have the GSM modems right? |
| 13:00.57 | carrar | I there was a PCI card with 4 GSM modems in it that can work with asterisk |
| 13:01.02 | carrar | err ^I recall |
| 13:01.28 | carrar | not sure if you can still get that |
| 13:02.36 | carrar | VoiSmart vGSM board in that list |
| 13:04.11 | carrar | later |
| 13:04.19 | coppice | http://www.openvox.cn/products/list.php?catid=62&lang=2 |
| 13:04.55 | Bartockbatz | Hi carrar - I want to be able to see caller ID in my softphone client - I am a little lost |
| 13:05.34 | Bartockbatz | so, incoming calls from the SIP trunk, I would like to see the caller ID info displayed. Am I a giving you enough info |
| 13:05.35 | Bartockbatz | ? |
| 13:20.49 | *** join/#asterisk cesar_CR (~cesar@201.201.41.242) |
| 13:29.57 | Bartockbatz | I guess I should ask what should be in the dialplan for this. |
| 13:34.40 | *** join/#asterisk MiserySoft (~LND@host81-139-167-173.in-addr.btopenworld.com) |
| 13:34.56 | *** join/#asterisk cesar_CR (~cesar@201.201.41.242) |
| 13:35.40 | MiserySoft | Hi, anyone know which version of asterisk is in the current ubuntu server repositories ? |
| 13:36.20 | MiserySoft | Just installing 10.04 server, and hoping my existing asterisk config files will still be OK ? |
| 13:36.27 | fenrus | apt-cache show |
| 13:36.31 | fenrus | <package> |
| 13:36.55 | fenrus | or check packages.ubuntu.com |
| 13:37.17 | MiserySoft | fenrus: Thanks.. |
| 13:39.08 | MiserySoft | looks like 1.6.* my old install was 1.4, are the config files similar syntax ? |
| 13:42.12 | ariel_ | MiserySoft: depends on what you were doing |
| 13:42.31 | ariel_ | 1.6 did change allot, your just going to have to do some reading and testing. |
| 13:43.10 | MiserySoft | ariel_: digium card+ analog adapters... really only extensions.conf I need to keep |
| 13:43.29 | MiserySoft | and sip.conf I guess |
| 13:43.31 | ariel_ | once again depends on what you were doing |
| 13:44.17 | MiserySoft | Nothing fancy, but I guess there's an hours reading.. Thanks. |
| 13:49.42 | ariel_ | just wondering why you upgrade or switched? |
| 13:49.54 | *** join/#asterisk stix (~stix@x1-6-00-1e-2a-87-8d-aa.k1.webspeed.dk) |
| 13:51.15 | MiserySoft | ariel_: Old asterisk server was on debian ... TDM card stopped working after an apt-get upgrade... so did a backup and decided to have a clean start.. |
| 13:51.28 | MiserySoft | only home system, 10 extensions + 2 sip trunks |
| 13:51.33 | fenrus | i'd go back to debian :) |
| 13:52.17 | ariel_ | I would too |
| 13:52.29 | MiserySoft | I prefer debian for stable server tasks like this, but got an ugly message about a kernel flag being deprecated and udev not working properly.. |
| 13:52.35 | MiserySoft | so time for a clean sweep. |
| 13:52.51 | MiserySoft | plus it's sunday afternoon and I'm bored :-) |
| 13:53.02 | ariel_ | I don't (IMO) Ubuntu makes a good server |
| 13:53.05 | fenrus | (: |
| 13:53.29 | ariel_ | I use ubuntu for my desktops |
| 13:53.35 | ariel_ | debian for my servers |
| 13:53.37 | fenrus | i'm running an ubuntu machine as asterisk server somewhere, it's working fine :) |
| 13:53.52 | fenrus | ariel_, sounds like my approach |
| 13:54.05 | MiserySoft | I waited for 10.04 LTS before switching, |
| 13:54.38 | MiserySoft | have a cctv system running dapper drake LTS upstairs |
| 14:11.05 | Kyosh | ip cams? |
| 14:11.25 | MiserySoft | yeah.. ZoneMinder with Axis IP cams |
| 14:11.37 | *** join/#asterisk Benwa (~benwa-ktm@host-212-68-196-120.brutele.be) |
| 14:12.03 | Kyosh | interesting |
| 14:12.25 | Kyosh | i ve been looking for a list of open source NVR's, but never found any |
| 14:12.45 | MiserySoft | ZoneMinder... highly recommended |
| 14:13.06 | Kyosh | have you used any windows nvr's to compare it to? |
| 14:13.38 | MiserySoft | nope. |
| 14:14.01 | Kyosh | because i have cert in alot of commericla stuff, then there's milestone which aint cheap and i got cert in that, but i think my boss can pay me more to do work rather than pay me less and pay vendors more |
| 14:14.37 | *** join/#asterisk riddlebox (~riddlebox@75-132-225-75.dhcp.stls.mo.charter.com) |
| 14:14.49 | Kyosh | hmm |
| 14:15.01 | Kyosh | it has events so i guess it has notifications too |
| 14:15.29 | MiserySoft | yep. |
| 14:15.46 | Kyosh | really nice |
| 14:15.49 | Kyosh | thanks for the heads up |
| 14:15.54 | Kyosh | gonna try it |
| 14:16.11 | MiserySoft | glad I could help.. |
| 14:28.10 | *** join/#asterisk nickaugust (~anonymous@97.102.189.103) |
| 14:29.49 | *** part/#asterisk nickaugust (~anonymous@97.102.189.103) |
| 14:30.17 | Kyosh | misery, does it have a remote-view client to view from other computers? |
| 14:33.52 | riddlebox | hey guys my salesman and I demo'd a system the other day and the customer asked, if there was anything where he could go to a website on his network and change his forwarding options? |
| 14:34.10 | riddlebox | does anyone know of anything to do that? |
| 14:42.24 | Kyosh | for an individual extension? |
| 14:43.22 | Kyosh | sure |
| 14:43.26 | Kyosh | have the user log in |
| 14:43.32 | *** join/#asterisk MhaddogM1 (~MhaddogM1@adsl-64-223-140.mia.bellsouth.net) |
| 14:43.46 | Kyosh | its under vmx locator or follow me or even phone features |
| 14:43.51 | Kyosh | its all right there |
| 14:49.25 | *** join/#asterisk Bartockbatz (~chatzilla@c-24-62-161-95.hsd1.nh.comcast.net) |
| 14:50.00 | Bartockbatz | Hey - question about caller ID with asterisk and a SIP trunk - anyone got a minute?? |
| 14:50.24 | florz | Bartockbatz: no, that's off topic in here |
| 14:50.46 | Bartockbatz | okay - can you recommend a channel? |
| 14:51.45 | Bartockbatz | florz can you recommend a channel? |
| 14:52.06 | *** join/#asterisk Brookss (~Brookss@174.3.119.13) |
| 14:52.09 | TJ^ | Bartockbatz i think hes pulling yr leg man |
| 14:52.26 | TJ^ | Bartockbatz just ask the question :) |
| 14:52.32 | Bartockbatz | hard to tell - I guess I have been 'owned' |
| 14:52.37 | florz | I'd recommend you rather ask the question the answer to which you are actually interested in, instead of waiting for 200 people to say "yes, I am here, but I have no clue whether I can help you" |
| 14:53.15 | Bartockbatz | Okay - it has been years since I used IRC - sorry - I forgot the 'nettequite' |
| 14:53.34 | *** join/#asterisk DennisG (DennisG@2002:541e:88d0:0:151f:c3f8:13cd:37eb) |
| 14:54.26 | Bartockbatz | okay - SIP trunk - incoming calls do not show the caller ID info in the CLI (with set sip debug) - shows 'anonymous' |
| 14:55.30 | Bartockbatz | so, I am to assume that there is a change/modification I need to make to the dialplan to show the caller ID info - follow? If I seem like I don't have a clue, you are right - not an asterisk guru |
| 14:56.13 | florz | is the caller ID included in the SIP messages? |
| 14:56.25 | florz | (and if so, where?) |
| 14:56.55 | Bartockbatz | the provider claims the include it |
| 14:57.11 | Bartockbatz | they include it |
| 14:57.16 | florz | well, have you verified that? |
| 14:57.48 | Bartockbatz | yes - their tech folks have told me that - ( after asking 3 times, and getting the same answer) |
| 14:59.06 | florz | well, have _you_ verified that? |
| 14:59.11 | florz | like, using a packet sniffer |
| 14:59.50 | Bartockbatz | I don't remember - however, I shall look at get back to you folks - good place to start |
| 15:03.10 | *** part/#asterisk Brookss (~Brookss@174.3.119.13) |
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| 15:24.24 | *** join/#asterisk smooth_penguin (~smoove@59.95.28.15) |
| 15:25.24 | smooth_penguin | HEY |
| 15:29.07 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 15:38.44 | Kyosh | miserysoft, got a sec? |
| 15:43.18 | *** join/#asterisk cosmicwombat (~cosmicwom@69.7.44.68) |
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| 16:18.02 | MiserySoft | Kyosh: I'm here now |
| 16:21.19 | *** join/#asterisk MhaddogM1 (~MhaddogM1@adsl-64-223-140.mia.bellsouth.net) |
| 16:21.53 | *** join/#asterisk SaiSoma (~SaiSoma@adsl-074-167-136-030.sip.mob.bellsouth.net) |
| 16:25.11 | ManxPower | ~mailinglistr |
| 16:25.14 | ManxPower | ~mailinglist |
| 16:25.15 | infobot | [~mailinglist] The Asterisk mailing lists can be found at http://lists.digium.com , http://www.asteriskguru.com/archives Search the archives by adding "site:lists.digium.com" to your Google search. |
| 16:25.19 | ManxPower | ~answers |
| 16:25.20 | infobot | [~answers] Asterisk Book: http://astbook.asteriskdocs.org & SIP w/ NAT: http://www.aocomputing.net/?p=3 & Wiki: http://www.voip-info.org & Docs: /path/to/src/asterisk/doc & 1.4 .vs. 1.6.0 .vs. 1.6.1: /path/to/src/asterisk/UPGRADE*.txt |
| 16:31.45 | *** join/#asterisk Andrew_M (~485018b7@gateway/web/freenode/x-qkjrkydujjzlxerl) |
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| 17:22.40 | flip__ | i need to configure a group of gsm phone numbers in my asterisk to forward a call from numer 23 to 5 mobile phones... how can i configure it to prevent that a mailbox of a mobilephone s answering the call? |
| 17:23.10 | flip__ | something like press # to get the call at the beginning would be ok |
| 17:23.29 | flip__ | any hints how i can do that or any manualpages? i didn't find anything that fits |
| 17:26.31 | TJNII | Answering machine detection, find me / follow me does that IIRC / make the dial drop the callee into a context with a menu... |
| 17:27.24 | ManxPower | flip__, you basically can't. Answering machine detection might help, and timeouts might help, but the only way for it to work 100% of the time is to disable voicemail on the cell/mobile |
| 17:28.05 | ManxPower | flip__, if you are willing to do answer confirmation then all things are possible. See "core show application dial" |
| 17:38.13 | [TK]D-Fender | flip__: "core show application dial" <- M() |
| 17:39.42 | *** join/#asterisk Heretic (~Fallen@dsl-246-75-190.telkomadsl.co.za) |
| 17:40.07 | Heretic | hi al |
| 17:40.52 | flip__ | ManxPower: [TK]D-Fender: thx |
| 17:40.59 | flip__ | will have a look |
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| 17:51.19 | *** join/#asterisk torrio (~f3k@p54A2C7A4.dip.t-dialin.net) |
| 17:55.07 | *** join/#asterisk RypPn (~TuMbL@rosscom.co.uk) |
| 17:56.40 | torrio | hi guys, i set asterisk up these days and was playing with the dailplan. Everything worked fine. When i powered my asterisk host on today ... i still get a timeout, now. My softphone is registered in "sip show peers", when calling i recieve a 408 ! When doing a "dailplan reload" no errors are shown ... someone can help me out of this ? |
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| 18:01.08 | *** join/#asterisk Tim_Toady (~moi@77.49.61.52.dsl.dyn.forthnet.gr) |
| 18:03.05 | torrio | ... when calling there is nothing in the CLI |
| 18:06.00 | p3nguin | increase verbose level and reload the dialplan again. |
| 18:06.11 | p3nguin | Turn on sip debug and try to make a call, too. |
| 18:12.32 | torrio | verbose is 20, sip set debug -> SIP Debugging re-enabled, is the sip debug, was "Ignoring this INVITE request" ! |
| 18:13.43 | *** join/#asterisk Mhaddog (~Mhaddog@adsl-074-186-011-199.sip.mia.bellsouth.net) |
| 18:15.54 | *** join/#asterisk aidinb (~Aidin@71-94-148-218.static.mtpk.ca.charter.com) |
| 18:17.00 | *** join/#asterisk atis_work (~atis_work@193.238.212.171) |
| 18:20.52 | p3nguin | ~pb |
| 18:20.52 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood 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://bin.cakephp.org/ , http://asterisk.pastey.net/ , or install pastebinit with yum or aptitude. |
| 18:20.56 | p3nguin | torrio: ^^ |
| 18:22.41 | SaiSoma | question regarding queues: I cannot get periodic announcements to function properly. they just never play. any ideas? here is the config: http://pastebin.com/3Uv8D19B |
| 18:24.40 | p3nguin | saisoma: periodic-announce is supposed to be a file name. Is that value that you have set a file name? |
| 18:25.01 | SaiSoma | yes. verified functional by using dialplan to play it prior to entering the queue |
| 18:25.09 | SaiSoma | it is a wav file |
| 18:25.19 | SaiSoma | does it need the extension? (unlike in the dialplan?) |
| 18:25.39 | [TK]D-Fender | SaiSoma: Where is the CLI output of a failed attempt? What version are you running? |
| 18:26.07 | SaiSoma | ver 1.6.2.6 is the ver, getting cli, one sec |
| 18:26.12 | p3nguin | I don't even know what "does it need the extension?" means. |
| 18:26.34 | [TK]D-Fender | SaiSoma: periodic-account-frequency = 15 M---- wuold help if this said ANNOUNCE, not ACCOUNT |
| 18:26.57 | SaiSoma | oh holy . . .ok. thanks. stupid typos. i've been looking at it for 30 mins. |
| 18:27.07 | SaiSoma | [TK]D-Fender: thanks . a LOT |
| 18:27.09 | SaiSoma | :) |
| 18:27.26 | p3nguin | Oh, "periodic-accounce" |
| 18:27.40 | SaiSoma | yea, caught that one too |
| 18:27.49 | SaiSoma | knocks self on head. |
| 18:28.31 | p3nguin | I would have expected the core to complain about those at some point. |
| 18:29.06 | ManxPower | SaiSoma, take a break. you are suffering from fatigue. |
| 18:29.12 | SaiSoma | never saw anything in the cli output. verbosity is at 101. but hey, i'm supposed to be smarter than the PC eh? |
| 18:29.25 | SaiSoma | ManxPower: you are more than likely correct sir! lunch time anyway |
| 18:32.56 | p3nguin | Is Dial()'s M() option supposed to return after the macro finishes if there was no Hangup() in the macro? |
| 18:38.04 | *** join/#asterisk orioni (~chatzilla@95.107.235.217) |
| 18:38.12 | orioni | anyone used sth like this http://www.suncomm.com.tw/Product_detail.asp?P_ID=6008485&F_ID=1000344 |
| 18:38.24 | orioni | cant find a way how to configure the asterisk to use that as a gateway |
| 18:43.53 | [TK]D-Fender | p3nguin: there is a variable associated with that... read up |
| 18:44.29 | [TK]D-Fender | orioni: Where do we see you try? |
| 18:45.42 | orioni | [TK]D-Fender : sorry |
| 18:45.50 | orioni | didtn got that |
| 18:52.40 | *** join/#asterisk crazybyte (~crzp@unaffiliated/crazypenguin/x-000001) |
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| 18:56.02 | *** join/#asterisk LemensTS__ (~LemensTS@166.137.12.42) |
| 18:56.25 | LemensTS__ | Hello |
| 18:58.31 | LemensTS__ | If i do g729 it still has to transcode moh, voicemail, anything else? |
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| 19:03.24 | *** join/#asterisk TehRabbitt (~rabbott@c-71-59-82-2.hsd1.pa.comcast.net) |
| 19:03.55 | *** join/#asterisk LND (~LND@dyn-62-56-86-50.dslaccess.co.uk) |
| 19:04.52 | TehRabbitt | Quick question, does anyone have experiance setting up SCCP phones? |
| 19:04.58 | TehRabbitt | Cisco 7921G |
| 19:05.22 | TehRabbitt | using a LaFonera as an AP connected to my linux box directly |
| 19:06.37 | p3nguin | I've only used chan_skinny, and it's not that wonderful. |
| 19:06.52 | TehRabbitt | p3nguin: what do you mean "not that wonderful" :-\\ |
| 19:07.57 | p3nguin | Using the skinny channel driver that comes with Asterisk, there's a lot of SCCP functionality that isn't present or doesn't work. There are 3rd-party SCCP channel drivers that address these problems, though. |
| 19:08.07 | TehRabbitt | i just picked up this phone thinking I could use asterisk as an intermediary between 3 different SIP lines I have through Viatalk.com but I cant figure out how to get the phone to connect |
| 19:09.00 | p3nguin | Configure skinny.conf. Load or reload chan_skinny. Use the phone. |
| 19:10.30 | TehRabbitt | hm... well basically this was my thinking originally since I know almost *nothing* about asterisk right now and have just started dabbling in it the other day... I want to set up one of the 3 lines as an outgoing trunk, 2 as incoming, and also allow my friend from his house to use the asterisk server (if possible... i'm guessing it's just a matter of port fowarding) to connect and place calls using a SIP Linksys PAP2 from his house |
| 19:11.44 | p3nguin | If your Asterisk computer is behind a NAT router, then you'll need port forwarding and also need to correctly configure Asterisk for NAT. |
| 19:11.48 | p3nguin | ~sipnat |
| 19:11.49 | infobot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 19:12.08 | TehRabbitt | Ah thanks |
| 19:12.46 | TehRabbitt | suprisingly it seems rather easy to set up, it's just a matter of planning everything out first i suppose... In theory, once it's set up, I would be able to use my IP phone here to call my friend's IP phone correct? |
| 19:12.55 | TehRabbitt | just by dialing extention? |
| 19:13.01 | p3nguin | Yes. |
| 19:13.18 | p3nguin | Each extension you create in extensions.conf can be configured to do just about anything you want. |
| 19:13.43 | p3nguin | This includes dialing SIP devices by SIP URI, even. |
| 19:13.58 | TehRabbitt | nice... and as long as I use regular SIP phones etc I don't need any special hardware correct? |
| 19:14.21 | p3nguin | You can use SIP phones or even an ATA and a regular phone. |
| 19:14.45 | p3nguin | As long as your device(s) speak a language that Asterisk has a channel driver for, you can communicate with it. |
| 19:15.00 | p3nguin | I use both SIP and IAX2 for phones right now, and I have used SCCP before. |
| 19:15.20 | TehRabbitt | Hm so SCCP is not too hard to get working? |
| 19:15.33 | TehRabbitt | because I can't seem to get this phone to "see" the server :-( |
| 19:15.48 | p3nguin | No, but the chan_skinny driver that comes with asterisk isn't great. |
| 19:16.29 | TehRabbitt | oh... :-\ so what would you reccomend I do heh basically I just want this phone to be an extension I can take outside / into another room so i'm not tied down to my desk |
| 19:16.41 | p3nguin | phones are not extensions |
| 19:16.44 | p3nguin | phones are phones |
| 19:17.01 | p3nguin | Extensions are the rules in extensions.conf that tell asterisk how to process calls. |
| 19:17.12 | TehRabbitt | ah. aka which phone to send it to? |
| 19:17.24 | p3nguin | Extensions don't even have to send calls to phones. |
| 19:17.39 | TehRabbitt | i'm guessing they can also be sent directly to voicemail or such? |
| 19:17.49 | p3nguin | exten => 100,1,Playback(tt-weasels) |
| 19:17.53 | *** join/#asterisk eliel (~eliels@186.18.108.106) |
| 19:17.58 | p3nguin | extension 100 plays that sound file and hangs up. |
| 19:18.02 | p3nguin | no phone involved. |
| 19:18.12 | p3nguin | Sure, to voicemail or anything. |
| 19:18.14 | [TK]D-Fender | TehRabbitt: they do whatever you tell them to |
| 19:18.16 | TehRabbitt | So in other words one of those "our normal business hours are XYZ thank you ... click" |
| 19:18.52 | p3nguin | exten => 9934567,1,Goto(hours,s,1) |
| 19:19.11 | [TK]D-Fender | TehRabbitt: it can make you coffee. It can change tracks on a jukebox playback, it can send an e-mail toa firend for you. If can report if a server is responding to pings, or.... whatever |
| 19:19.22 | p3nguin | Then in the hours context, exten => s,1,Playback(our-hours-are) |
| 19:19.27 | TehRabbitt | I guess basically what i'm asking... this SCCP phone I have, is there a basic way I can get it so I can just make and receive calls on it just basic functionality? |
| 19:19.40 | p3nguin | (1408.59) <p3nguin> Configure skinny.conf. Load or reload chan_skinny. Use the phone. |
| 19:19.48 | TehRabbitt | [TK]D-Fender: hm so basically extensions acn do almost anything? |
| 19:19.58 | [TK]D-Fender | TehRabbitt: yes |
| 19:20.31 | TehRabbitt | p3nguin: alright, i'm guessing once I configure skinny.conf and reload chan_skinny, It should basically be the equivilant to a regular phone running off an ATA? (you dont get the full package of SCCP features) |
| 19:20.51 | p3nguin | similar |
| 19:21.10 | p3nguin | chan_skinny does allow making and receiving calls. I do know that much. |
| 19:21.25 | p3nguin | It didn't allow me to make transfers from the phone, though. |
| 19:21.40 | TehRabbitt | i'm guessing the "roaming phonebook" and "push to talk" that the phone's firmware supports wouldn't work though correct? |
| 19:22.00 | p3nguin | That depends on how those things are incorporated. |
| 19:22.04 | *** join/#asterisk iscario (~div@laureades.davout.pck.nerim.net) |
| 19:22.15 | TehRabbitt | I think it's all "Cisco Call Manager" that normally handles it :-\ |
| 19:22.34 | fenrus | the CME |
| 19:22.37 | p3nguin | If the phone book uses http and xml, you can set up those things. |
| 19:22.55 | fenrus | isnt there a SIP software available for that phone ? |
| 19:22.57 | p3nguin | I wouldn't expect PTT to work, though. |
| 19:23.00 | [TK]D-Fender | TehRabbitt: forget PTT. Phonebooks may not directly involve SCCP |
| 19:23.02 | p3nguin | no, there isn't. |
| 19:23.11 | fenrus | that's a shame |
| 19:23.13 | TehRabbitt | fenrus: apparently SIP was never made for this phone |
| 19:23.20 | p3nguin | The 7921 wireless phone currently only has an SCCP image. |
| 19:23.27 | TehRabbitt | yea :-\ |
| 19:23.46 | fenrus | hm, i read 7912G somewhere |
| 19:23.57 | p3nguin | It's too bad, because that's a desired phone. |
| 19:24.20 | TehRabbitt | p3nguin: I know, i mean other than the lack of SIP it's a great little phone... I just wish it had built in SIP because it'd make things 10x easier |
| 19:24.47 | p3nguin | Use chan_skinny for now just to get it "online" with Asterisk. |
| 19:25.04 | TehRabbitt | p3nguin: have you ever used the 7921? |
| 19:25.11 | fenrus | hm, we had problems with that phone roaming between different access-points |
| 19:25.22 | p3nguin | I'm familiar with it a little, but I've never owned my own. |
| 19:25.33 | fenrus | we had some older phone version |
| 19:25.58 | TehRabbitt | Ah... yea apparently for it to even "find" the asterisk server, I need to specify the location within an XML file hosted by TFTP |
| 19:26.15 | p3nguin | That's easy enough. |
| 19:26.30 | fenrus | some dhcp-options and an tftpd and youre set |
| 19:26.31 | iscario | please : "IAX includes the ability to authenticate in three ways: plain text, MD5 hashing, and |
| 19:26.31 | iscario | RSA key exchange. This, of course, does nothing to encrypt the media path or headers |
| 19:26.31 | iscario | between endpoints." ----> does it means that a man in the midle won't be able theorically to uncipher the voice data, but will be able to know from and where the packets are going ? |
| 19:26.49 | *** join/#asterisk crazybyte (~crzp@unaffiliated/crazypenguin/x-000001) |
| 19:27.11 | TehRabbitt | only other thing I didn't like about this phone is it's lack of WPA/WPA2 |
| 19:27.28 | TehRabbitt | it "supports" it but only if it also uses a username/password or something 0_o |
| 19:27.46 | p3nguin | Install the hpa tftpd on the same computer and put your files there. Rather than make your dhcp server tell the phone where to look for the tftpd, just set it manually in your phone. |
| 19:27.52 | TehRabbitt | hence the use of a seperate AP directly attached to the server |
| 19:28.06 | TehRabbitt | hpa? |
| 19:28.53 | p3nguin | http://freshmeat.net/projects/tftp-hpa/ |
| 19:29.06 | p3nguin | tftp-hpa... should be available in your repos. |
| 19:29.12 | TehRabbitt | aight |
| 19:29.31 | TehRabbitt | it's better than regular tftpd? |
| 19:29.45 | p3nguin | What do you mean by "regular"? |
| 19:29.50 | p3nguin | extra/tftp-hpa 5.0-3 Official tftp server |
| 19:30.11 | TehRabbitt | debian has 2 packages available "tftpd" and then "tftpd-hpa" |
| 19:30.19 | p3nguin | Oh. |
| 19:30.27 | p3nguin | *shrug* choose one. |
| 19:30.43 | TehRabbitt | well I chose the HPA one and it's installed now i'm just trying to find it's .conf file lol |
| 19:30.55 | p3nguin | I don't use Debian, so I don't know what the other one is. |
| 19:31.39 | p3nguin | There probably isn't a conf for it. |
| 19:32.03 | TehRabbitt | oh... hm how do I specify the directory the files will be in then? |
| 19:32.18 | ChannelZ | you run it that way |
| 19:32.24 | TehRabbitt | Ah |
| 19:32.46 | p3nguin | The init script might have something for tuning, or it could just be hard-coded in the script. |
| 19:32.57 | TehRabbitt | http://www.davidsudjiman.info/2006/03/27/installing-and-setting-tftpd-in-ubuntu/ good tutorial I just found... though it's for Ubuntu, debian uses the same exact commands since ubuntu is built upon debian |
| 19:33.13 | ChannelZ | the one I'm using you can run from inetd.conf for instance |
| 19:33.19 | TehRabbitt | hm |
| 19:33.23 | p3nguin | I actually have /etc/conf.d/tfptd which sets the TFTPD_ARGS value that my /etc/rc.d/tftpd script uses. |
| 19:33.38 | TehRabbitt | i'm gonna try to get this working and see if the phone can register to the asterisk server heh |
| 19:33.40 | p3nguin | /etc/conf.d/tftpd, that is |
| 19:34.50 | TehRabbitt | are you running gentoo? |
| 19:35.00 | p3nguin | Me? No, ArchLinux. |
| 19:35.10 | TehRabbitt | ah |
| 19:35.39 | p3nguin | If you've already got xinetd going, it's certainly a reasonable idea to pop your tftpd stuff in there, since it only spawns the processes as needed. |
| 19:36.08 | fenrus | *shrug* |
| 19:36.09 | TehRabbitt | true |
| 19:36.13 | *** join/#asterisk atis_work (~atis_work@193.238.212.171) |
| 19:36.28 | ChannelZ | Do you need tftp all the time? |
| 19:36.48 | TehRabbitt | the phone polls the TFTP server every few minutes to make sure it's still connected 0_o |
| 19:36.49 | p3nguin | I wouldn't want to have to turn it on and off when needed. |
| 19:37.05 | TehRabbitt | that's how the phone can tell if it's "in range" or not |
| 19:37.23 | ChannelZ | That's... a choice.. |
| 19:37.53 | p3nguin | Either way (tftp as a daemon or tftp from xinetd), it isn't going to create a lot of load on the system. |
| 19:37.54 | TehRabbitt | ChannelZ: howso? the phone is hardcoded to auto-check for the XML file every erm 2 min or so |
| 19:38.19 | ChannelZ | I just mean that's a strange and dumb design |
| 19:38.20 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 19:38.36 | TehRabbitt | ChannelZ: Cisco Designed it, I just bought it heh |
| 19:39.07 | TehRabbitt | they want you to use the phone on THEIR AP's with built in TFTP servers, along with THEIR server software (call Manager) |
| 19:40.24 | TehRabbitt | All the XML file contains is what "extensions" the phone will be able to dial out from and also the IP address of the asterisk server |
| 19:41.08 | TehRabbitt | if it can't find the XML file it just goes into a constant reboot until it does... I did get it working using TFTP on my windows machine but it just says "invalid SCCP server" |
| 19:42.34 | *** join/#asterisk Akiraaa (~Akiraaaa@79.112.44.66) |
| 19:43.14 | p3nguin | Until you configure chan_skinny or another channel driver, you do have an invalid SCCP server. :/ |
| 19:43.25 | TehRabbitt | exactly 0_o lol |
| 19:44.03 | TehRabbitt | so i figure since I got it working using the XML file on the windows machine, once i figure out TFTP on this machine, I can actually have the phone connect to the same machine for both TFTP as well as SCCP |
| 19:44.37 | p3nguin | That's how I do it. No reason to NOT have the tftpd on the same box as Asterisk. |
| 19:45.01 | TehRabbitt | exactly but I can't figure out how to get TFTP to run now haha |
| 19:45.02 | p3nguin | Now when you start doing hundreds of phones checking the tftpd all the time, there could be a reason. |
| 19:45.09 | TehRabbitt | lol true |
| 19:46.08 | p3nguin | Install it. See if there is a config file under /etc/sysconfig (or where ever debian puts them), then adjust it if needed. Then start the server with /etc/init.d/tftpd start. |
| 19:46.40 | TehRabbitt | kk *crosses fingers* lol |
| 19:47.00 | p3nguin | The tftp root is probably something like /var/tftp or /var/tftpboot. |
| 19:47.46 | TehRabbitt | yea, i figured that out it's in inetd.conf |
| 19:47.49 | TehRabbitt | but now i get this: |
| 19:47.49 | TehRabbitt | thoth:~# /etc/init.d/tftpd-hpa restart |
| 19:47.49 | TehRabbitt | tftpd-hpa disabled in /etc/default/tftpd-hpa |
| 19:48.10 | p3nguin | sigh |
| 19:48.15 | p3nguin | adds this to another reason to never use Debian. |
| 19:48.33 | TehRabbitt | fixed it heh "run as dameon" was set to "no" |
| 19:48.34 | TehRabbitt | lol |
| 19:48.52 | TehRabbitt | Annddddd its alive! |
| 19:49.26 | p3nguin | Do you have the files (or at least a template) that the phone wants from tftp? |
| 19:50.19 | TehRabbitt | Yes there's actually a tutorial on how to set up those files but they expect you to already have experiance with tftp / asterisk |
| 19:50.19 | TehRabbitt | they give you a nice XML template though |
| 19:51.06 | TehRabbitt | http://www.voip-info.org/wiki/view/SCCP-HOWTO2 |
| 19:53.03 | *** join/#asterisk Raden (~Raden@71.89.121.119) |
| 19:56.03 | iscario | hi, i am looking forward a good IAX softphone (client) with alphanumerics touch , and if possible open-source (at least free). Does anyone know one ? thx |
| 19:57.18 | p3nguin | iscario: zoiper |
| 19:58.29 | iscario | i know this one, what bother me was that i could only use ABCD on the visual pad.... thx anw p3nguin |
| 19:59.18 | Baylink-afk | Yeah; zoiper's pretty nice. |
| 20:00.25 | Baylink-afk | It's free predecessor had a few more features, but also possible a nasty crash-your-server IAX bug. |
| 20:00.56 | [TK]D-Fender | iscario: what is the point of having more than ABCD on the dialpad? |
| 20:01.17 | p3nguin | The keyboard can enter all the other keys. |
| 20:01.53 | Baylink-afk | ABCD are on there for full 16-key TouchTone<tm> support, not for alpha entry. JFYI. |
| 20:02.20 | [TK]D-Fender | precisely |
| 20:02.23 | TehRabbitt | hm what does this mean from a tutorial i was following: |
| 20:02.24 | TehRabbitt | you will also need chan-sccp-b svn (March 2008) for the device to show correctly in asterisk (im using 1.4.13 with svn pulled -06/03/08 dod) |
| 20:02.35 | TehRabbitt | do I need to worry about that? |
| 20:02.39 | [TK]D-Fender | TehRabbitt: that version is ancient |
| 20:02.42 | TehRabbitt | k |
| 20:02.52 | iscario | i thought that i could ask the user to enter a name instead of nums, so that an alphanumeric keyboard would have been helpful [TK]D-Fender |
| 20:03.11 | [TK]D-Fender | iscario: You can, but there is no point of being on the pad since you can't use it once the call is placed |
| 20:03.51 | Baylink-afk | Whereas being able to dial DTMF is. Does 2833 send the high-4? :-) |
| 20:03.55 | *** join/#asterisk jasonjjohnsonjr (~jjohnson@adsl-93-32-130.owb.bellsouth.net) |
| 20:04.15 | [TK]D-Fender | Baylink-afk: Should |
| 20:04.23 | Baylink-afk | So you'd hope. ;-) |
| 20:04.42 | [TK]D-Fender | Baylink-afk: Its active on the PST, so I can't see how they'd not support it |
| 20:04.43 | iscario | true, i didn't knew that in fact. [TK]D-Fender ... I thought it could work, but i think i understant know ;) |
| 20:05.04 | iscario | -k |
| 20:05.08 | Baylink-afk | Well, not really... I don't know of any production use of ABCD on PSTN switches, myself, at least. |
| 20:05.22 | Baylink-afk | It's really rare on end-devices, for that matter. |
| 20:06.01 | *** join/#asterisk Naikrovek (~jjohnson@unaffiliated/naikrovek) |
| 20:06.24 | Baylink-afk | And, since I have this cool new thing called The Google, I looked: yes, 2833 encodes 0-9 ABCD, # * and FLASH |
| 20:09.18 | TehRabbitt | Ok well the phone is looking for the TFTP server and is not finding it :( and then just goes "not in service" |
| 20:09.45 | fenrus | provide the tftp-server address with a dhcp option |
| 20:09.57 | p3nguin | I wouldn't bother with that. |
| 20:10.11 | p3nguin | Provide the dynamic tftp address in the phone manually. |
| 20:10.24 | p3nguin | we're talking about ONE device, here. |
| 20:10.27 | fenrus | :) |
| 20:11.02 | TehRabbitt | Heh... I put the IP for TFTP into the phone and it's "looking for CM Entries" but it fails / goes "not in service" |
| 20:11.12 | TehRabbitt | so i'm thinking it's TFTP that's not running properly |
| 20:11.26 | TehRabbitt | TFTP Timeout |
| 20:11.44 | TehRabbitt | DNS Unknown Host, TFTP Timeout |
| 20:11.49 | TehRabbitt | that's what it shows in the phone's status log |
| 20:11.54 | fenrus | check the daemon log and if the tftpd is running |
| 20:14.09 | TehRabbitt | nope not running |
| 20:14.11 | ChannelZ | do you know what path/filename it's looking to fetch? |
| 20:14.13 | TehRabbitt | hm weird |
| 20:17.53 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 20:19.44 | TehRabbitt | Hmph phone still can't find the TFTP server :( |
| 20:20.20 | fenrus | can you download stuff from it manually ? |
| 20:20.24 | fenrus | with a tftp client |
| 20:20.43 | TehRabbitt | http://pastebin.com/4drBCJr4 |
| 20:20.52 | TehRabbitt | thats the XML file it's supposed to pull |
| 20:20.58 | TehRabbitt | the so-called "CM List" |
| 20:21.11 | TehRabbitt | havent' tried that... lemme try pulling using a tftp client |
| 20:24.01 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 20:24.37 | p3nguin | If the phone doesn't know where the tftpd is, it cannot get that file. Did you specify in the phone's settings where the tftpd is? |
| 20:24.48 | p3nguin | In SIP, you use the dynamic tftp setting. |
| 20:25.07 | TehRabbitt | yep, 192.168.1.70 which is up, responds to ping, and TFTP is running however clients can't connect to TFTP so its gotta be an issue with the TFTP still |
| 20:25.12 | TehRabbitt | Transfering file XMLDefault.cnf.xml from server in ascii mode... |
| 20:25.12 | TehRabbitt | Packet will be sent. len=61, opcode=1 |
| 20:25.12 | TehRabbitt | Packet received. len=36, opcode=5 |
| 20:25.12 | TehRabbitt | Error occurred during the file transfer (Error code = 8): |
| 20:25.12 | TehRabbitt | Unsupported option(s) requested |
| 20:25.57 | p3nguin | How about your /etc/hosts.allow file? |
| 20:26.06 | p3nguin | Did you specify in.tftpd: ALL: allow |
| 20:26.25 | TehRabbitt | I dont think I did... lemme check |
| 20:26.47 | p3nguin | The system's log file should have been griping about it. |
| 20:28.00 | *** join/#asterisk aidinb (~Aidin@71-94-148-218.static.mtpk.ca.charter.com) |
| 20:28.11 | TehRabbitt | May 2 15:49:24 thoth in.tftpd[8048]: cannot bind to local socket: Address already in use |
| 20:28.11 | TehRabbitt | May 2 16:09:01 thoth /USR/SBIN/CRON[8119]: (root) CMD ( [ -x /usr/lib/php5/maxlifetime ] && [ -d /var/lib/php5 ] && find /var/lib/php5/ -type f -cmin +$(/u |
| 20:28.11 | TehRabbitt | sr/lib/php5/maxlifetime) -print0 | xargs -n 200 -r -0 rm) |
| 20:28.16 | TehRabbitt | thats what it shows 0_o |
| 20:29.19 | p3nguin | cannot bind to local socket: Address already in use |
| 20:29.39 | TehRabbitt | yea :-\ how do I know what else is using that address |
| 20:29.39 | p3nguin | So you need to destroy whatever is using your port. |
| 20:29.43 | p3nguin | lsof -i :69 |
| 20:30.01 | TehRabbitt | in.tftpd 8291 root 4u IPv4 66992 UDP *:tftp |
| 20:30.24 | p3nguin | Stop the tftpd for a few seconds and then restart it. |
| 20:30.52 | p3nguin | You should not see that message again. |
| 20:32.54 | p3nguin | Now you've got me thinking about trying chan_sccp. |
| 20:36.47 | *** join/#asterisk gospch (~gospch@p5088EE9D.dip.t-dialin.net) |
| 20:38.09 | TehRabbitt | its still not working so i'm restarting the machine and seeing if it loads on boot / perhaps something else was using the port or something idk |
| 20:38.22 | p3nguin | Did you see the message again? |
| 20:38.33 | ManxPower | Why don't you ask on the channel for your distro? |
| 20:38.41 | p3nguin | By the way, this isn't Windows, so restarting doesn't "solve" problems. |
| 20:38.44 | TehRabbitt | I did.. they sent me here haha |
| 20:39.17 | TehRabbitt | p3nguin: I know that lol but I'm figuring since i've started and stopped services so many times perhaps one of them I lost track of is still running or idk |
| 20:39.36 | p3nguin | Things don't work like that. |
| 20:40.53 | ManxPower | chances are xinetd is configured to launch a tftp server as well. |
| 20:41.08 | fenrus | inetd's are only problems. |
| 20:41.31 | p3nguin | I'm pretty sure he started it as a daemon. |
| 20:42.03 | ManxPower | imagine what would happen if xinetd also tried to listen on the same port |
| 20:42.14 | TehRabbitt | that's what i think was happening |
| 20:42.30 | TehRabbitt | hence why i commented it out in xinetd and just left it installed as a daemon |
| 20:42.31 | p3nguin | Then don't start it from xinetd or don't start it as a daemon. |
| 20:42.35 | ManxPower | fixing that is a distro question |
| 20:43.12 | p3nguin | And you normally don't comment out things in xinetd to stop them. You have to set it to disabled. |
| 20:43.27 | TehRabbitt | oh 0_o |
| 20:43.52 | TehRabbitt | i'm hopping over to #debian to ask them why TFTP wont work heh |
| 20:48.13 | *** join/#asterisk crazybyte (~crzp@unaffiliated/crazypenguin/x-000001) |
| 20:53.06 | p3nguin | chan-sccp.org sure doesn't make it easy to get the v2 stable package. |
| 20:53.53 | TehRabbitt | p3nguin: what do you mean? |
| 20:55.36 | TehRabbitt | v2 better than v1 i'm guessing? |
| 20:59.46 | *** join/#asterisk Jumpie (n3rdz@ip68-98-31-152.ph.ph.cox.net) |
| 20:59.50 | Jumpie | hoal |
| 20:59.52 | Jumpie | hola |
| 21:01.00 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 21:02.28 | TehRabbitt | Ok, so how do I change the interface that TFTPD is bound to? |
| 21:04.15 | fenrus | probably some listen-statement in the config |
| 21:04.55 | TehRabbitt | well the good news is running netstat it shows SCCP and asterisk are all running / listening but apparently TFTP is assigned to the IP 0.0.0.0:69 which would be causing some issues lol |
| 21:05.31 | p3nguin | 0.0.0.0 is all IP addresses on any interface. |
| 21:05.40 | TehRabbitt | oh :( |
| 21:05.47 | p3nguin | Since you probably only have one interface with one IP address, you're fine. |
| 21:05.48 | TehRabbitt | sooo still dont know why TFTPD wont connect |
| 21:06.01 | p3nguin | How about hosts.allow like I asked? |
| 21:06.39 | fenrus | tftpd-hpa and non inetd-crap, will be up and running in 5 minutes. |
| 21:06.43 | fenrus | now off for bed. |
| 21:07.47 | TehRabbitt | p3nguin I added that line into hosts.allow |
| 21:07.52 | TehRabbitt | fenrus: i'm using tftpd-hpa |
| 21:08.06 | p3nguin | Now show me some useful logs. |
| 21:08.13 | fenrus | add some debugging, and parse the logs |
| 21:09.12 | TehRabbitt | which logs do you want to see? |
| 21:10.06 | fenrus | the daemon logs telling you why your client cant connect |
| 21:11.10 | TehRabbitt | looks like the only thing in daemon.log that refrences tftpd is: |
| 21:11.11 | TehRabbitt | May 2 15:36:59 thoth xinetd[8013]: added service tftp [file=/etc/inetd.conf] [line=34] |
| 21:11.40 | fenrus | grep -i tftp /var/log/* |
| 21:11.40 | p3nguin | So you're still running it through xinetd AND as a stand-alone daemon? |
| 21:11.44 | TehRabbitt | in my inetd.conf file: |
| 21:11.44 | TehRabbitt | #tftp dgram udp wait root /usr/sbin/in.tftpd /usr/sbin/in.tftpd -s /var/lib/tftpboot |
| 21:11.51 | fenrus | restarted inetd ? |
| 21:11.53 | TehRabbitt | i'ts got a # in front of it |
| 21:12.05 | TehRabbitt | i've restarted the whole server, that should restart inted no? |
| 21:12.11 | fenrus | yes. |
| 21:12.13 | p3nguin | sigh |
| 21:12.41 | fenrus | are you certain that the xinetd read from inetd.conf and not some other file ? |
| 21:12.42 | ManxPower | I thought all the distros switched to xinetd, but I guess not. |
| 21:12.45 | TehRabbitt | hold on for pastebin |
| 21:12.57 | TehRabbitt | http://pastebin.com/wi0HwWuN |
| 21:12.57 | ManxPower | why don't you just stop inetd all totather to test |
| 21:13.03 | ManxPower | together, even |
| 21:13.22 | fenrus | i dont understand why all distros still include that crap |
| 21:13.38 | *** join/#asterisk fofware (~fabian@190.7.25.160) |
| 21:13.47 | fenrus | inetd is obsolete and gives more problems than it solves. |
| 21:13.57 | TehRabbitt | Can't even remove inetd... it won't let you |
| 21:14.05 | *** join/#asterisk jasonjjohnsonjr (~jjohnson@adsl-93-3-96.owb.bellsouth.net) |
| 21:14.19 | fenrus | it sure does, i dont run the crap on any of my debian servers |
| 21:14.22 | fenrus | dpkg --purge |
| 21:14.27 | fenrus | and update-rc.d is the shit |
| 21:14.32 | fenrus | no i really need to leave. |
| 21:14.36 | ManxPower | is there a dpkg --binge too? |
| 21:14.51 | TehRabbitt | fenerus how do I remove it? heh |
| 21:18.17 | TehRabbitt | Ok, so I can confirm TFTP is running but nothing is connecting to it :-\ |
| 21:19.57 | p3nguin | It would have taken less time to just use it the way they wanted you to use it. |
| 21:20.11 | TehRabbitt | who cisco? heh |
| 21:20.15 | TehRabbitt | or tftpd? |
| 21:20.19 | p3nguin | tftpd |
| 21:20.28 | TehRabbitt | well i'm trying to use it the way they want me to |
| 21:20.30 | TehRabbitt | but it wont work |
| 21:20.37 | Jumpie | when you email voicemails to a particular email |
| 21:20.43 | Jumpie | is that supposed to show in asterisk cli? |
| 21:21.13 | TehRabbitt | i'm going to purge tftpd and reinstall it |
| 21:21.13 | Jumpie | im having issues with sendmail |
| 21:21.16 | Jumpie | and not sure if im missing sometin |
| 21:22.14 | Jumpie | i see entries in /var/spool/mail/asterisk that the email was formatted but not sure if it sent..so im trying on a diff extension and not seeing anythign else |
| 21:23.42 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 21:26.43 | *** join/#asterisk boodu (~boodu@175.158.129.128) |
| 21:27.06 | boodu | hi |
| 21:28.11 | ChannelZ | Jumpie: sendmail is a living issue |
| 21:28.54 | *** join/#asterisk jasonjjohnsonjr (~jjohnson@adsl-93-3-96.owb.bellsouth.net) |
| 21:31.48 | Jumpie | ChannelZ yeah :( |
| 21:31.50 | *** join/#asterisk gospch (~gospch@p5088EE9D.dip.t-dialin.net) |
| 21:32.04 | Jumpie | i know you guys hate im using freepbx..but that isnt the issue here...im following http://pbxinaflash.com/forum/showthread.php?t=570 |
| 21:32.14 | Jumpie | can you verify if th at looks to be a sound tutorial? |
| 21:32.33 | Jumpie | because honestly there aer several others that give varying answers as to what should be in what files..and imw orried im not quite right |
| 21:33.44 | *** join/#asterisk gospch (~gospch@p5088EE9D.dip.t-dialin.net) |
| 21:40.21 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 21:43.19 | *** join/#asterisk ChannelZ (~bobm@burner.com) |
| 21:43.27 | *** join/#asterisk kotp (~vgoff@96.2.187.66) |
| 21:45.02 | *** join/#asterisk TimeRider (steve@5ac318fe.bb.sky.com) |
| 21:54.13 | *** join/#asterisk kartik (~koolkarti@117.199.112.254) |
| 21:54.45 | Jumpie | ok |
| 21:54.49 | Jumpie | i found out what's going on |
| 21:54.50 | Jumpie | lol |
| 21:54.55 | *** join/#asterisk baddragon (yiffstar66@unaffiliated/devemo) |
| 21:54.58 | Jumpie | the email WAS going out..but was getting sent to spam in gmail |
| 21:55.07 | Jumpie | its because its going from asterisk@..... to voicemail@.... |
| 21:55.16 | Jumpie | Received-SPF: neutral (google.com: 69.255.192.97 is neither permitted nor denied by best guess record for domain of asterisk@wallacepbx.dyndns.org) client-ip=69.255.192.97; |
| 21:55.32 | Jumpie | so..im sure this is an issue before..is this an actual dns change i need to do? or something i can tweak with sendmail? |
| 22:00.12 | TJNII | Setting the RDNS for your domain really helps when you're running a MX. |
| 22:00.20 | TJNII | Don't know if you can do that with dyndns, though. |
| 22:01.40 | chuckf | Jumpie: how many people are going to be using this asterisk system as it is configured? |
| 22:02.02 | *** part/#asterisk bminish (~bminish@pdpc/supporter/professional/bminish) |
| 22:02.03 | Jumpie | chuckf well really just 2 |
| 22:02.19 | Jumpie | but i want to be sure i cant have others try to somehow use the dyndns domain in their emails as well? |
| 22:02.34 | Jumpie | TJNII this is really a dns issue? i cant change somet things in sendmail to avoid this behavoir? i mean i understand why gmail is doing this |
| 22:03.46 | chuckf | Jumpie: at TJNII said, its a reverse dns issue and and gmail is catching it as spam. I don't believe that the free dyndns can do rdns |
| 22:03.51 | TJNII | You're hitting a spam filter. You're already down because you're (assumedly) on a home connection. Most major ISPs check the RDNS record, and if it doesn't match you get filtered. |
| 22:04.23 | *** join/#asterisk tzafrir (~tzafrir@212.179.75.202) |
| 22:04.50 | Jumpie | so this has nothing to do with the asterisk user vs "voicemail" from |
| 22:04.53 | Jumpie | ok..i just wanted to verify that |
| 22:05.00 | chuckf | Jumpie: the easy thing to do is just whitelist the domain through your gmail or set sendmail to forward and use your gmail to send the vmails out with your gmail account |
| 22:05.04 | Jumpie | this is really only going to one email account..i was using the jumpie@gmail to test sendmail |
| 22:05.20 | Jumpie | i ust hope...where i send it to doesnt totally block it..if he sets it to not set that email as spam we should be ok |
| 22:16.48 | p3nguin | You could easily configure postfix to relay via your gmail account. I can only assume sendmail is almost as easy. |
| 22:19.16 | Jumpie | p3nguin actually im lookin on dyndn's web site |
| 22:19.23 | Jumpie | they offer a mailhop secure relay option |
| 22:19.40 | p3nguin | They can't do much better than relaying through gmail. |
| 22:19.45 | Jumpie | true |
| 22:19.49 | Jumpie | although i still run into reverse dns issue |
| 22:19.58 | Jumpie | i mean everything works functionally |
| 22:22.13 | TehRabbitt | p3nguin: i'm back... |
| 22:22.15 | TehRabbitt | TFTP is working... |
| 22:22.21 | TehRabbitt | now it just shows up "Registration Rejected" |
| 22:22.26 | TehRabbitt | asterisk isn't accepting the phone :( |
| 22:22.31 | p3nguin | configure it. |
| 22:22.50 | Jumpie | whatp hoen |
| 22:22.50 | TehRabbitt | aight lol |
| 22:22.53 | Jumpie | er phone |
| 22:23.30 | p3nguin | From what I can tell, no configuration needs to be done on the phone. You configure skinny.conf with the info and that's it. |
| 22:24.01 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 22:24.54 | Jumpie | im curious if you can statically assign up addresses to aastra 6731i |
| 22:24.57 | Jumpie | from config file |
| 22:24.58 | Jumpie | and not on the phoen |
| 22:25.07 | Jumpie | dhcp default |
| 22:25.40 | p3nguin | Are you talking about dhcp reservations? |
| 22:26.13 | TehRabbitt | p3nguin: yea I think it's just an issue with the skinny.conf file |
| 22:26.15 | Jumpie | nono |
| 22:26.19 | Jumpie | so that it doesnt use dhcp to obtain |
| 22:26.23 | Jumpie | i wanna statically assign ips to my phones |
| 22:26.34 | Jumpie | its easy in the phone but..i wanna see if i can do it via mac.cfg |
| 22:27.06 | p3nguin | And where does the mac.cfg come from? |
| 22:27.46 | Jumpie | the MAC.cfg is the phones macaddress in caps.cfg |
| 22:27.49 | Jumpie | resides in /tftpboot |
| 22:27.51 | Jumpie | auto configures phones |
| 22:27.57 | Jumpie | per phone config...vs aastra.cfg which is global |
| 22:28.01 | p3nguin | So the file is on a server. |
| 22:28.05 | Jumpie | yea on the pbx |
| 22:28.17 | Jumpie | i just dont remember the syntax on how to statically assign ips to the phone |
| 22:28.25 | Jumpie | if im ATt he phone...which im not cause i 3000 miles away, i cans et it up no problem |
| 22:28.27 | p3nguin | How do you propose to get the file from a server which is on a computer network which speaks IP... without the phone having an IP address? |
| 22:28.28 | Jumpie | :P |
| 22:28.33 | Jumpie | haha no |
| 22:28.36 | Jumpie | let me back up |
| 22:28.47 | Jumpie | all im saying is..i want to statically assign the ip to the phone vs it getting it via dhcp |
| 22:28.55 | Jumpie | which i can do IF im at the phone through the phone ui |
| 22:28.57 | p3nguin | That would be done IN THE PHONE. |
| 22:29.10 | Jumpie | but im sayin im pretty sure within the phone configs in MAC.cfg i can do it also |
| 22:29.15 | p3nguin | or by a static dhcp entry. |
| 22:29.19 | Jumpie | yea |
| 22:29.23 | Jumpie | i need to research the syntax |
| 22:29.39 | p3nguin | You're not going to get the mac.cfg from the server if you don't already have an IP address. |
| 22:30.20 | Jumpie | yea i can |
| 22:30.30 | Jumpie | bootp |
| 22:30.42 | Jumpie | as soon as the phone is on the network it sendsa broadcast |
| 22:30.43 | Jumpie | layer2 |
| 22:30.49 | Jumpie | if the mac.cfg already exists |
| 22:30.56 | Jumpie | it finds it |
| 22:31.01 | Jumpie | at this poitn connectivity isnt an issue |
| 22:31.07 | Jumpie | all im saying is now..i have an ip on the phone..but its gotten by dhcp |
| 22:31.12 | Jumpie | i want to statically assign ips to all these phones |
| 22:31.21 | Jumpie | and then label the phone with the ip |
| 22:31.29 | Jumpie | im doign this to dumbproof it for user..i dont really care |
| 22:31.43 | *** join/#asterisk jameswf (~james@unaffiliated/jameswf-home) |
| 22:32.07 | TJNII | Heh, my IP phones are the only things on my network that I don't care what IP they get.... |
| 22:32.17 | Jumpie | strangely, sometimes i ahve to 'nudge' the phones to finding the server |
| 22:32.35 | TehRabbitt | p3nguin: It works... :-D |
| 22:32.36 | Jumpie | TJNII i really dont care but, i want the customer to be able to log into the phones through web ui and change things |
| 22:32.41 | TehRabbitt | shows the time, and the extension # |
| 22:32.42 | TehRabbitt | w00t |
| 22:32.46 | Jumpie | and i dont want to try to step him thorugh how to find the IP and confuse him |
| 22:32.55 | Jumpie | yay :D |
| 22:33.09 | p3nguin | tehrabbitt: 523 hours later... |
| 22:33.23 | TehRabbitt | haha and now I need to figure out how to configure asterisk to actually place a call outbound lmao that should be easy though... |
| 22:33.37 | p3nguin | extensions.conf |
| 22:33.41 | p3nguin | and probably sip.conf. |
| 22:33.41 | TehRabbitt | ah.. |
| 22:34.02 | p3nguin | Are you talking to your ITSP via SIP or IAX2? |
| 22:34.07 | *** join/#asterisk jasonjjohnsonjr (~jjohnson@adsl-93-3-96.owb.bellsouth.net) |
| 22:34.07 | TehRabbitt | so just take my SIP credentials, put them into sip.conf then just configure extensions.conf telling what ext 500 is supposed to do? |
| 22:34.14 | TehRabbitt | ITSP? |
| 22:34.24 | p3nguin | That's pretty much it. |
| 22:34.24 | Jumpie | voip termination provider |
| 22:34.28 | p3nguin | ~itsp |
| 22:34.29 | infobot | [~itsp] An ITSP is an Internet Telephony Service Provider (or VoIP telephone company). They allow you to either SEND calls to the PSTN (this is called termination), RECEIVE calls from the PSTN (called origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs. |
| 22:34.29 | TehRabbitt | Ah, ViaTalk uses SIP |
| 22:34.35 | TehRabbitt | SIP |
| 22:34.49 | TehRabbitt | they give me the SIP proxy, port, and UN/PW for me to connect |
| 22:34.54 | Jumpie | aaw..dyndns wanst $19.95 for outbound mail relay |
| 22:34.55 | Jumpie | sigh |
| 22:35.01 | p3nguin | That goes in sip.conf then. |
| 22:35.11 | p3nguin | gmail is free! |
| 22:35.18 | TehRabbitt | gmail? |
| 22:35.24 | p3nguin | google |
| 22:35.26 | p3nguin | mail |
| 22:35.27 | p3nguin | gmail |
| 22:35.31 | TehRabbitt | for SIP? |
| 22:35.32 | TehRabbitt | i'm confused |
| 22:35.34 | TehRabbitt | OHHH nvm |
| 22:35.37 | p3nguin | NO |
| 22:35.49 | *** join/#asterisk CoderForLife (~Miranda@cpe-174-101-155-51.cinci.res.rr.com) |
| 22:35.52 | TehRabbitt | if only google voice supported SIP now THAT would be nice |
| 22:35.52 | TehRabbitt | heh |
| 22:36.04 | p3nguin | They do... sorta. |
| 22:36.20 | TehRabbitt | how? |
| 22:37.11 | p3nguin | http://lmgtfy.com/?q=google |
| 22:37.26 | p3nguin | http://lmgtfy.com/?q=google+voice+sip |
| 22:37.33 | Nugget | I love that site. |
| 22:37.37 | p3nguin | disregard that first one. |
| 22:38.01 | TehRabbitt | lmao |
| 22:39.36 | *** join/#asterisk Torrieri (~Torrieri@nelug/crew/torrieri) |
| 22:41.30 | TehRabbitt | Sigh call is failing :( |
| 22:41.47 | p3nguin | paste the info into pastebin.com if you want anyone else to look at it and help. |
| 22:42.01 | *** join/#asterisk JT (~j@unaffiliated/jt) |
| 22:42.21 | TehRabbitt | k |
| 22:43.00 | TehRabbitt | http://pastebin.com/H6YLjmJR |
| 22:46.28 | p3nguin | What context is your phone in? |
| 22:46.35 | TehRabbitt | what do you mean? |
| 22:46.56 | p3nguin | I mean: The context for your phone is _________. fill in this blank. |
| 22:47.10 | TehRabbitt | Extension 500 |
| 22:47.12 | TehRabbitt | ? |
| 22:47.14 | TehRabbitt | CISCO? |
| 22:47.28 | TehRabbitt | those last 3 lines |
| 22:47.40 | p3nguin | I don't know how else to put it. |
| 22:47.44 | p3nguin | Your phone. |
| 22:47.47 | p3nguin | It is in a context. |
| 22:47.50 | ppc | haha |
| 22:47.56 | p3nguin | What context would that be? |
| 22:48.02 | TehRabbitt | I'm not sure what you mean by context though that's what has me confused |
| 22:48.23 | p3nguin | In skinny.conf, you had to configure an entry for your phone. |
| 22:48.36 | TehRabbitt | ohhh |
| 22:48.42 | p3nguin | The entry for the phone require a setting of "context" in it. |
| 22:48.46 | p3nguin | What value does it have? |
| 22:49.09 | TehRabbitt | ohhh... |
| 22:49.13 | TehRabbitt | default |
| 22:49.29 | p3nguin | Is this your ENTIRE extensions.conf in the pastebin? |
| 22:49.36 | TehRabbitt | Yes |
| 22:49.44 | p3nguin | Then that's why your calls don't work. |
| 22:49.55 | TehRabbitt | oh lol |
| 22:49.59 | p3nguin | You're missing some important things, plus your phone isn't in the proper context. |
| 22:50.12 | TehRabbitt | what context should it be in? |
| 22:50.21 | p3nguin | Probably users based on this paste. |
| 22:51.06 | TehRabbitt | any good examples or a simple tutorial on how I should set up this extensions.conf? |
| 22:51.58 | Jumpie | haha i actually had the first time i ever needed to do ipconfig /flushdns |
| 22:51.59 | p3nguin | http://pastebin.com/X8MqVz7M |
| 22:52.03 | p3nguin | tehrabbitt: ^^ |
| 22:52.08 | Jumpie | made a ch ange to dyndns and my host wasnt resolving right |
| 22:53.24 | TehRabbitt | ok i'm gonna try that and see if it works |
| 22:53.31 | p3nguin | I also do not have SetCalledParty() as a valid application, so check your system to see if you do or do not have it. |
| 22:54.12 | p3nguin | Make sure you run "dialplan reload" after you save your changes. |
| 22:54.28 | TehRabbitt | ok apparently I can make internal calls but I can't dial out still |
| 22:55.08 | p3nguin | You might have to reload the chan_skinny driver after changing it. I can't recall if there is a reload for its settings. |
| 22:55.17 | Jumpie | my friend uses asterisk to handle his door buzzing in and out |
| 22:55.17 | Jumpie | :D |
| 22:55.26 | TehRabbitt | Jumpie: how? lol |
| 22:55.30 | Jumpie | the pin codes are given out to certain people and they are 'hidden 'extensions on an ivr |
| 22:55.51 | Jumpie | i think like 5 bad attempts in a row, rings his number or sends him a text or something |
| 22:56.07 | p3nguin | I see that you are trying to dial out through SIP/VOIP1. Do you have a peer definition in sip.conf for VOIP1 where you have all the settings for your ITSP? |
| 22:56.30 | TehRabbitt | yes I do |
| 22:56.45 | p3nguin | Did you run "sip reload" after saving the changes? |
| 22:56.48 | TehRabbitt | when I dial a # such as 1-555-555-1212 it says "connecting" then juts hangs up |
| 22:56.51 | TehRabbitt | nope |
| 22:56.55 | p3nguin | Better do that. |
| 22:57.02 | TehRabbitt | how do I run that? |
| 22:57.16 | Jumpie | from asterisk cli |
| 22:57.22 | p3nguin | you type in sip reload and press Enter on the asterisk console. |
| 22:57.37 | TehRabbitt | [May 2 18:58:11] WARNING[4657]: acl.c:408 ast_get_ip_or_srv: Unable to lookup 'proxy1. |
| 22:57.43 | TehRabbitt | apparently it can't find my SIP proxy |
| 22:58.21 | *** join/#asterisk eject_ck (~eject_ck@haitek.cosmonova.net.ua) |
| 22:58.21 | p3nguin | Paste your entire sip.conf, removing your passwords. |
| 22:58.53 | TehRabbitt | ok. |
| 23:00.28 | TehRabbitt | http://pastebin.com/vG0sekj9 |
| 23:02.14 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 23:02.47 | TehRabbitt | hm this is the error asterisk throws when I dial a +1(xxx)xxx-xxxx number |
| 23:02.48 | TehRabbitt | [May 2 19:02:59] WARNING[4682]: chan_skinny.c:2461 skinny_ss: Can't match [11] from '1234561414' in context users |
| 23:04.16 | p3nguin | Is your asterisk system connected directly to the internet? |
| 23:04.27 | TehRabbitt | Yes it is.. |
| 23:04.37 | TehRabbitt | I can ping out from the server |
| 23:04.55 | p3nguin | Are you trying to relate those two things? |
| 23:05.07 | p3nguin | Being able to ping out from the server does not make it a direct internet connection. |
| 23:05.30 | TehRabbitt | Oh you mean do I have a router, yes I do... but the old PAP2 behind the router never had an issue with SIP |
| 23:05.39 | p3nguin | It's not being any NAT/firewall/router devices? |
| 23:05.45 | *** join/#asterisk jks (jks@193.189.93.254) |
| 23:05.48 | TehRabbitt | ^ |
| 23:06.02 | p3nguin | That's not a direct internet connection, then, and you need to follow the sipnat guide. |
| 23:06.05 | p3nguin | ~sipnat |
| 23:06.06 | infobot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 23:07.07 | p3nguin | Your sip.conf is lacking necessary configuration. <--- this is what I was trying to say. |
| 23:08.00 | p3nguin | Here's an example of a working sip.conf: http://pastebin.com/m59d17875 |
| 23:09.37 | p3nguin | This example contains an asterisk box behind NAT, with an example of one ITSP using two servers with static configuration, one ITSP using a dynamic configuration, and one phone. |
| 23:10.38 | *** join/#asterisk lesouvage (~lesouvage@82.73.69.76) |
| 23:12.06 | *** join/#asterisk traderz (~traderz@c-67-184-227-156.hsd1.il.comcast.net) |
| 23:12.43 | traderz | anyone here familiar with the protocol application invalid issue on cisco 7960 phones and how to fix it? i can run version 6.x of the software but can't get to 7.x or 8.x .. |
| 23:13.38 | p3nguin | I use SIP 8.11 on 7940/7960 phones. |
| 23:13.44 | p3nguin | No problems that I can tell. |
| 23:14.15 | traderz | p3nguin, is yours a version g or the older unit? |
| 23:14.54 | p3nguin | Why can't they be old an G models at the same time? |
| 23:15.00 | p3nguin | -an |
| 23:15.05 | p3nguin | +and |
| 23:15.18 | TehRabbitt | what is this line for: |
| 23:15.20 | TehRabbitt | register => 105123:pebbles0123@sip.us2.voip.ms/18005551212 |
| 23:15.26 | TehRabbitt | in the example you sent me |
| 23:15.28 | p3nguin | registering to your ITSP |
| 23:15.34 | TehRabbitt | ohh |
| 23:15.42 | TehRabbitt | that might help no? heh |
| 23:15.47 | p3nguin | probably |
| 23:15.54 | traderz | p3nguin, how did you get it to work. mine fails everytime... |
| 23:16.04 | traderz | what do you have in yoru tftp directory? |
| 23:16.11 | TehRabbitt | so what is that 105123 for? |
| 23:16.39 | p3nguin | username |
| 23:18.28 | p3nguin | traderz: OS79XX.TXT, P003-08-11-00.bin, P003-08-11-00.sbn, P0S3-08-11-00.loads, P0S3-08-11-00.sb2, RINGLIST.DAT, SIPDefault.cnf, SIP<mac address>.cnf, dialplan.xml |
| 23:19.27 | p3nguin | tehrabbitt: I'm starting to get the impression that you failed to look at the sample config files AND the provided documentation AND you didn't read The Book. |
| 23:19.40 | p3nguin | ~book |
| 23:19.40 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 23:19.41 | traderz | p3guin, can i see what you have in your SIP* files |
| 23:19.43 | TehRabbitt | http://pastebin.com/twTPkmn3 |
| 23:21.29 | Z_God | when I define different extensions in the same contexts in extensions.conf and extensions.ael will the just be merged? |
| 23:21.49 | p3nguin | traderz: http://www.loligo.com/asterisk/Cisco/79xx/current/SIPDefault.cnf |
| 23:21.53 | Z_God | when I type 'ael reload' I just get 'command failed' is there any way to see why it failed? |
| 23:22.13 | p3nguin | traderz: http://www.loligo.com/asterisk/Cisco/79xx/current/SIP0002B9EB0EF4.cnf |
| 23:22.14 | TehRabbitt | p3nguin: any ideas on how to get the route to work? |
| 23:22.17 | traderz | is that your current config? |
| 23:22.37 | p3nguin | tehrabbitt: Yes. Read the friggin' book and configure everything correctly. |
| 23:22.52 | p3nguin | traderz: No. You don't need MY configs. |
| 23:23.35 | traderz | p3nguin, those configs on that site are from version 4.x days |
| 23:24.03 | p3nguin | Then don't use them if you feel like they're wrong. |
| 23:27.02 | traderz | p3nguin, i was hoping to seee if there is a difference in your configs versus my configs |
| 23:27.06 | Jumpie | if i delete the actual voicemails in the voicemail recording directory |
| 23:27.14 | Jumpie | will that get rid of the voicemail notification on the actual phones? |
| 23:27.24 | p3nguin | Pretend like the files I just gave you are mine. Compare it to your own. |
| 23:27.30 | Jumpie | i'm doing some testing at a locaion that's not physically where the pbx/phones are, and just wanna fast delete all the voicemails |
| 23:28.18 | p3nguin | If you delete the files from /var/spool/asterisk/voicemail/mailboxid/whatever... yes, the MWI will turn off. |
| 23:28.26 | Jumpie | perfect, thanks |
| 23:28.33 | Jumpie | and figured out an easy way to send multiple emails :) |
| 23:28.43 | Jumpie | setting up an alias and then sending to thatalias@localhost |
| 23:28.45 | Jumpie | ;P |
| 23:32.11 | *** join/#asterisk nix8n82 (~chatzilla@63.162.27.14) |
| 23:32.30 | traderz | p3guin, do you use dhcp for your phone or hard code ip? |
| 23:32.30 | *** join/#asterisk chuckf (~chuckf@ubuntu/member/chuckf) |
| 23:36.41 | traderz | anyone here familiar with the protocol application invalid issue on cisco 7960 phones and how to fix it? i can run version 6.x of the software but can't get to 7.x or 8.x .. once i try to upgrade from 6.x to 7.x i get protocol application invalid and its stuck unless i go back to verson 6.x |
| 23:40.02 | p3nguin | hahaha |
| 23:40.37 | p3nguin | Why can't you follow the same guides that everyone else uses, use the same sample configs that everyone else uses, and fix it yourself? |
| 23:40.56 | traderz | cuz i have been following them and it's not working after hours of time.. |
| 23:41.08 | carrar | Because you are here so that they do not have to research |
| 23:41.13 | Jumpie | so p3nguin..you think if i used gmail as mail relay instead of comcast..i may have better luck? |
| 23:41.23 | Jumpie | my problem is...even using a different relay is still going to have a reverse dns issue |
| 23:41.44 | p3nguin | Put the files into the tftp root, removing all others, boot the phone. |
| 23:42.16 | p3nguin | jumpie: You use those services as a relay so that you don't have to worry about YOUR OWN reverse DNS. |
| 23:42.18 | carrar | doesn't comcast buisness let you send Email? |
| 23:42.37 | p3nguin | Let them handle DNS. That's what the relays are for. |
| 23:43.23 | p3nguin | The relays won't give a shit what your DNS looks like... they'll send (as long as you are allowed to relay through them). |
| 23:43.58 | p3nguin | Then THAT server talks to other servers, and the other servers look at the DNS of THAT server that just relayed for you. |
| 23:44.23 | Jumpie | p3nguin..im pretty sure thast what i did |
| 23:44.26 | Jumpie | im using comcast's relay |
| 23:44.38 | Jumpie | and yet gmail is bitching about the email but it accepts it ins pam |
| 23:44.38 | p3nguin | And what's the current issue? |
| 23:44.41 | Jumpie | yahoo doesnt even take it |
| 23:44.54 | Jumpie | because the domain im supposedly sending from doesnt match the reverse |
| 23:45.05 | Jumpie | oh..so you sayhing if i dont try to masq |
| 23:45.10 | Jumpie | and just let it 'be from comcast' ? |
| 23:46.06 | p3nguin | If you relay through gmail, you actually authenticate your server like a client to gmail. |
| 23:47.36 | Jumpie | no kidding |
| 23:47.46 | Jumpie | well...take a lookt at this..and can you tell me if this is a yahoo issue, or my server |
| 23:48.45 | TJNII | Yahoo is bad if you're running your own server |
| 23:48.51 | Jumpie | yea it seems to be not likcing |
| 23:48.57 | p3nguin | I would need to see the maillog. |
| 23:49.04 | TJNII | They regularly delay my mail for at least an hour as an "anti-spam" measure. |
| 23:49.32 | Jumpie | http://jumpie.pastebin.com/P7pFswBC |
| 23:49.48 | Jumpie | is th at yahoo rejecting me? or is that something actually wrong with their mail server at the moment |
| 23:50.08 | p3nguin | greylisting isn't new technology. |
| 23:50.29 | Jumpie | if i use gmail as my relay, my concern is will i resolve this yahoo issue |
| 23:50.56 | p3nguin | stat=Service unavailable ... no other codes, such as 450 or 500? |
| 23:50.57 | Jumpie | and lookin at that log entry..im not 100% sure what it means by service unavaialble..i'd figure it'd say something about some authentication problem |
| 23:51.06 | Jumpie | hmm..lemme look more |
| 23:51.47 | Jumpie | nope |
| 23:51.50 | Jumpie | however.... |
| 23:51.54 | Jumpie | on the gmail oen that does go through |
| 23:51.59 | Jumpie | it says dsn 2.0.0 |
| 23:52.08 | Jumpie | yahoo says dsn 5.0.0 not sure what that is offhand |
| 23:52.37 | Jumpie | but no..no ther codes |
| 23:55.19 | p3nguin | hmm |
| 23:56.09 | Jumpie | somebody mentioned going right through gmail instead |
| 23:56.22 | Jumpie | it may be better |
| 23:56.25 | Jumpie | as relay |
| 23:58.01 | p3nguin | Yeah, I mentioned it... because most receiving servers will accept mail from gmail. |
| 23:58.10 | Jumpie | aah |
