| 00:00.33 | hardwire | SCTP kinda rocks |
| 00:00.34 | *** join/#asterisk dlucas (n=root@Broadband-Dynamic-Western84.connect.com.fj) |
| 00:03.38 | rayvd | Thanks. *blush* |
| 00:05.30 | *** join/#asterisk viperdude (n=jon@pc4.discover.adsl.entanet.co.uk) |
| 00:07.30 | *** join/#asterisk danalien (n=danalien@unaffiliated/danalien) |
| 00:08.38 | Igbothom | wow - the new Sun Fire X2100 and X4100 boxes would make great *serious* Asterisk servers - issue with the X2100 is the onboard PCI-E slot, no PCI/PCI-X, unfortunately |
| 00:09.15 | kippi | how easy is it to send texts to and from your phone on asterisk? |
| 00:09.18 | *** join/#asterisk shyru (n=shyru@200.94.130.194) |
| 00:09.34 | shyru | hi!!! |
| 00:10.17 | *** join/#asterisk glomph (n=black@dsl-63-249-23-11.zipcon.net) |
| 00:11.48 | *** join/#asterisk r0d3nt|m (i=nobody@wsip-24-234-241-145.lv.lv.cox.net) |
| 00:14.19 | shyru | yesterday i had a problem with zaptel device on FC3, the probem was for irq option. |
| 00:14.38 | *** join/#asterisk Qwell (n=north@68-190-230-117.dhcp.wsco.ca.charter.com) |
| 00:16.14 | kippi | anyone know if you can send text to a grandstream phone? |
| 00:17.17 | shyru | now i have proble with asterisk :S when i ran safe_asterisk show me this error: Asterisk died with code 1. Automatically restarting Asterisk. |
| 00:17.20 | shyru | any idea?? |
| 00:17.37 | shyru | kippi, i don't know :S |
| 00:17.54 | BigJoe1 | run asterisk from the prompt not safe_asterisk to see. or read the logs. asterisk -vvvgc |
| 00:18.21 | shyru | BigJoe1, ok let me see |
| 00:25.41 | *** part/#asterisk jsaunders (i=js@S01060060971c5817.va.shawcable.net) |
| 00:26.42 | kippi | whats the best billing software out there for asterisk? what are people using |
| 00:27.36 | BigJoe1 | kippi, don't know I am too new to know that much yet. But from what I was reading I just might have to make my own. |
| 00:35.44 | shyru | BigJoe1, thanks! aloT!!!!!!!!! |
| 00:35.51 | shyru | BigJoe1, thanks! a lot!!!!!!!!! |
| 00:36.26 | *** join/#asterisk Nivex (i=kjotte@user-0c8hq5r.cable.mindspring.com) |
| 00:36.32 | shyru | BigJoe1, i had problems with mohmp3 |
| 00:36.32 | shyru | :D |
| 00:36.47 | shyru | BigJoe1, my asterisk that's ok! |
| 00:36.51 | BigJoe1 | great to be of help |
| 00:37.16 | shyru | BigJoe1, :D |
| 00:37.42 | shyru | thaxs dude a lot! |
| 00:38.25 | *** join/#asterisk Wi_Fi (n=OUT@c-24-127-12-85.hsd1.ca.comcast.net) |
| 00:47.05 | *** join/#asterisk mjmac (n=mjmac@pdpc/supporter/active/mjmac) |
| 00:49.38 | *** join/#asterisk infinity1 (i=brendon@solara.netcal.com) |
| 00:49.40 | *** join/#asterisk Ariel_ (n=Ariel@dsl-20-177.cofs.net) |
| 00:49.45 | Ariel_ | hello everyone |
| 00:53.02 | infinity1 | yo! |
| 00:53.08 | infinity1 | its friday! and i'm done with work! |
| 00:53.19 | infinity1 | time to see how fast i can hack the neighborhood wep keys :) |
| 00:53.20 | infinity1 | lol |
| 00:55.15 | kippi | is there a * key to make sure that moh is working? |
| 00:56.13 | X-Rob | kippi - there's an application MusicOnHold |
| 00:57.29 | *** part/#asterisk BigJoe1 (n=BigJoe1@dsl-20-177.cofs.net) |
| 00:57.53 | *** join/#asterisk netsurfer (n=bbjunkie@i-83-67-48-18.freedom2surf.net) |
| 00:57.55 | kippi | how do i do a test with it? |
| 00:59.13 | *** join/#asterisk juice (n=juice@mo-65-40-248-239.dyn.sprint-hsd.net) |
| 01:04.58 | infinity1 | any idea why when i set callerid when dialing from voip -> pots phones has no effect after i upgraded HEAD about a week ago? |
| 01:05.27 | infinity1 | is that a known bug? where can i search for callerid related problems? |
| 01:10.34 | malverian[work] | bugs.digium.com |
| 01:10.58 | infinity1 | malverian[work]: sounds like no one else experiences this problem. any idea what my problem might be? |
| 01:16.56 | Qwell | twisted or hypa7ia around? |
| 01:17.33 | FuriousGeorge | one of my servers just isnt registering with the others |
| 01:18.05 | FuriousGeorge | and i cant seem to pinpoint why |
| 01:18.14 | FuriousGeorge | they just timeout |
| 01:18.22 | FuriousGeorge | the register commands are correct |
| 01:18.27 | FuriousGeorge | other servers can see them |
| 01:21.12 | *** join/#asterisk Error_X (n=Error_X@217-120-7.5001.adsl.tele2.no) |
| 01:21.39 | Error_X | I have installed asterisk@home, but when I'm calling for example: *43 which is the echo test, I get no sound |
| 01:22.38 | twisted[asteria] | Qwell, you rang? |
| 01:23.19 | Qwell | twisted[asteria]: What was that site you said, for those codes to get usefulness out of your cell? heh |
| 01:23.28 | twisted[asteria] | huh? |
| 01:23.41 | twisted[asteria] | i just know the shortcuts for nextel |
| 01:23.44 | Qwell | ahh |
| 01:23.59 | twisted[asteria] | OOOOH |
| 01:24.00 | Qwell | maybe it was hypa7ia that said the site then |
| 01:24.01 | twisted[asteria] | speaking of which |
| 01:24.12 | twisted[asteria] | i still need to hook up my data cable so i can get irc on it |
| 01:30.10 | *** join/#asterisk azzie (i=az@cpe-24-168-17-173.si.res.rr.com) |
| 01:35.31 | FuriousGeorge | iax.conf doesnt take the nat= command does it? |
| 01:36.09 | Qwell | FuriousGeorge: it doesn't need it |
| 01:37.31 | FuriousGeorge | if a register command in iax.conf is correct, why would it timeout? nat was what i was thinking. im really stuck on this one |
| 01:40.39 | *** join/#asterisk desktophero (n=chatzill@ip24-56-30-250.ph.ph.cox.net) |
| 01:41.17 | desktophero | hello all |
| 01:41.18 | drumkilla | what is the cheapest headset I could get for a 7960? :) |
| 01:42.14 | *** join/#asterisk florz (i=nobody@2001:1a50:503c:0:0:0:0:1) |
| 01:43.08 | FuriousGeorge | drumkilla: i know the 3.5mm headsets for pcs can be as cheap as $5 |
| 01:43.23 | FuriousGeorge | shoot $1 on newegg |
| 01:43.44 | drumkilla | yeah, but the 7960 uses a different kind of connection |
| 01:43.51 | FuriousGeorge | the tiny one |
| 01:43.59 | FuriousGeorge | i think its 2.5mm i could be wrong about both |
| 01:44.05 | drumkilla | no ... |
| 01:44.23 | FuriousGeorge | oh the 7960 is a cell phone |
| 01:44.26 | FuriousGeorge | nokia |
| 01:44.28 | drumkilla | it actually uses an RJ11 |
| 01:44.39 | FuriousGeorge | i thought we were talking about ip phones |
| 01:44.42 | FuriousGeorge | rj11? |
| 01:44.54 | drumkilla | yes, the Cisco 7960 is an IP phone |
| 01:45.03 | FuriousGeorge | oh |
| 01:45.08 | FuriousGeorge | isnt there a nokia 7960 too |
| 01:45.33 | FuriousGeorge | riiittteee :) |
| 01:46.16 | FuriousGeorge | is it a real rj-11? like POTS line? |
| 01:46.27 | twisted[asteria] | rj-11 is like, handset |
| 01:46.30 | drumkilla | well, the same plug. |
| 01:46.37 | twisted[asteria] | well actually |
| 01:46.41 | twisted[asteria] | drumkilla, it's rj-8 |
| 01:46.42 | FuriousGeorge | old handsets are rj7 or something arent they? |
| 01:46.57 | twisted[asteria] | rj-11 is a larger connector |
| 01:47.10 | FuriousGeorge | 'tis |
| 01:47.10 | drumkilla | whattteevverrrr |
| 01:47.17 | twisted[asteria] | drumkilla, makes a big difference |
| 01:47.22 | drumkilla | :) |
| 01:47.45 | drumkilla | http://www.ciscoheadsetadapter.com/ |
| 01:47.51 | drumkilla | that's probably what i should get ... |
| 01:47.53 | Aughey | the cisco's have a rj11 jack on the back of the phone |
| 01:47.58 | twisted[asteria] | rj8 |
| 01:48.01 | twisted[asteria] | RJ-8 |
| 01:48.02 | twisted[asteria] | AIYEE |
| 01:48.09 | twisted[asteria] | rj-11 will not fit |
| 01:48.11 | twisted[asteria] | you can try it |
| 01:48.14 | twisted[asteria] | but i bet you'll quit |
| 01:48.29 | drumkilla | OK |
| 01:48.32 | drumkilla | I HEARD YOU |
| 01:48.33 | drumkilla | :) |
| 01:48.39 | twisted[asteria] | i was talking to Aughey |
| 01:48.41 | twisted[asteria] | ninja. |
| 01:48.46 | FuriousGeorge | was just gonna ask if there as some sort of adapter |
| 01:48.50 | FuriousGeorge | *was |
| 01:49.02 | drumkilla | i don't want to pay $20 for it |
| 01:49.03 | drumkilla | ugh |
| 01:49.03 | Aughey | I think they do at least. My irc was just lagged, so the last 30 lines were just sent |
| 01:49.03 | twisted[asteria] | hah |
| 01:50.19 | twisted[asteria] | lag is no excuse |
| 01:50.24 | twisted[asteria] | get a real pipe |
| 01:50.28 | twisted[asteria] | err |
| 01:50.38 | twisted[asteria] | haha |
| 01:50.50 | twisted[asteria] | this coming from me on less than a T |
| 01:52.44 | FuriousGeorge | drumkilla: does the cisco phone actually provide talkback to the headset or something? |
| 01:52.50 | FuriousGeorge | talkbat* |
| 01:53.59 | FuriousGeorge | or is it gonna work more like a standard RCA headset? |
| 01:54.29 | Vco | no.. Zenith |
| 01:54.30 | shyru | somebody know how to set lang??? |
| 01:54.34 | Vco | **rimshot** |
| 01:55.00 | alephcom | shyru: This was talked about on the -users mailing list in the last few days I believe. |
| 01:55.59 | shyru | alephcom, thanks! let me find it |
| 01:56.01 | shyru | :D |
| 02:00.26 | *** join/#asterisk ManxPower (n=ewieling@236.sub-70-197-120.myvzw.com) |
| 02:03.54 | *** join/#asterisk mog_home (n=mogorman@user-24-236-84-48.knology.net) |
| 02:04.25 | *** join/#asterisk mcadory (n=mcadory@208.149.64.28) |
| 02:06.45 | *** join/#asterisk cripito (n=ncripito@c-67-161-130-59.hsd1.co.comcast.net) |
| 02:09.32 | ManxPower | . |
| 02:11.22 | twisted[asteria] | ManxPower |
| 02:11.26 | *** join/#asterisk drumkilla (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 02:11.26 | *** mode/#asterisk [+o drumkilla] by ChanServ |
| 02:11.35 | ManxPower | Hello twisted |
| 02:11.39 | ManxPower | Today just sucked. |
| 02:11.43 | twisted[asteria] | dude |
| 02:11.46 | twisted[asteria] | i've been saying that all day |
| 02:11.52 | mog_home | same here |
| 02:12.00 | mog_home | it just was a crappy day all around |
| 02:12.00 | ManxPower | It started out pretty good. There was coffee and muffins and then doughnuts at the office. |
| 02:12.05 | *** join/#asterisk HolyGod (i=nobody@got.securebinary.com) |
| 02:12.21 | twisted[asteria] | heh, mine started out bad |
| 02:12.27 | shyru | :S |
| 02:12.28 | twisted[asteria] | level 3 just fucked me early. |
| 02:12.31 | shyru | somebody know how to set lang??? |
| 02:12.46 | ManxPower | It went downhill from there. Gulfport is just DEPRESSING after Katrina, nobody moved the servers from the old office from the new office. Bell delivered the Frame Relay as an ETHERNET connection, and I got the wrong Digium card for the motherboard. |
| 02:13.01 | ManxPower | shyru, "show applications like language" |
| 02:13.18 | twisted[asteria] | ManxPower, FRoE? |
| 02:13.23 | twisted[asteria] | wtf |
| 02:13.30 | cripito | :) |
| 02:13.38 | ManxPower | twisted, that's what I said. |
| 02:13.49 | cripito | so looks like a fun day :) more than anything else |
| 02:13.56 | ManxPower | Apparently they were terminating the DLCI in their own box, then handing it to us as ethernet. |
| 02:14.07 | twisted[asteria] | well, we always say that bell smokes more crack than a crack whore with $2m |
| 02:14.23 | cripito | :DDDDDD |
| 02:14.37 | shyru | ManxPower, well i need to change lang for when asterisk ask for password (im using disa password for cellular calls) |
| 02:15.20 | ManxPower | twisted, I could have made it work if I had ordered a TE205P instead of a TE210P |
| 02:15.28 | twisted[asteria] | oops. |
| 02:15.33 | ManxPower | Using DACS to split out the channels in Asterisk to hand off to the Cisco router. |
| 02:15.41 | twisted[asteria] | *nods* |
| 02:16.00 | ManxPower | twisted, Silly me, I assumed that the TE210P was just the 2 port version of the TE110P |
| 02:16.00 | *** join/#asterisk wmandra (n=me@pcp04943183pcs.verona01.nj.comcast.net) |
| 02:16.21 | shyru | ManxPower, when users type for celular calls, asterisk ask for password (asterisk ask for password on english) . The people just speak spanish. |
| 02:16.35 | ManxPower | shyru, did you use SetLanguage? |
| 02:16.53 | ManxPower | as documented in "show applications like language" in the Asterisk CLI? |
| 02:17.24 | ManxPower | twisted, UPS lost the cards TWICE. |
| 02:17.31 | ManxPower | Once during Katrina and once during Rita |
| 02:18.00 | shyru | ManxPower, yes i set that op on extensions.conf, but does not rules |
| 02:18.07 | twisted[asteria] | oh geez. |
| 02:18.29 | ManxPower | now we have to return them. |
| 02:18.41 | twisted[asteria] | what, did the hurricanes like reorganize their warehouses? |
| 02:18.44 | *** join/#asterisk SplasPood (n=sp@brooklyn.paravolve.net) |
| 02:18.49 | twisted[asteria] | or are they just that dense |
| 02:19.16 | ManxPower | twisted, Well, Katrina DID "rearrange" their warehouses. |
| 02:19.20 | ManxPower | I dunnot what Rita did |
| 02:19.38 | twisted[asteria] | nice use of the word rearrange there |
| 02:19.45 | shyru | ManxPower, oki let me check thanks dude! |
| 02:20.35 | wmandra | Just wondering what you guys think of aah?? I've been using asterisk for 2 years now and I don't think it was that difficult to figure out how to configure it. It just seems like some people are too lazy to try and learn * and want an easy way out, then end up posting messages everywhere when they can't get aah to work. But how could they expect to if they don't understand the fundamentals of how * works..... -rant over |
| 02:21.37 | ManxPower | wmandra, I think people are just too lazy and take the easy way out, then up posting messages everywhere when they can't get AAH to work. 8-) |
| 02:22.21 | ManxPower | ~docs |
| 02:22.22 | jbot | it has been said that docs is Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk |
| 02:22.28 | ManxPower | ~mailinglist |
| 02:22.29 | jbot | mailinglist is probably Search Asterisk mailing lists by prepending site:lists.digium.com to your Google search. Browse the mailing list archive at http://lists.digium.com/ |
| 02:22.36 | shyru | :S |
| 02:23.27 | wmandra | manx, it's jsut frustrating cause I usually don't mind helping people out, but it gets to a point where it's just silly. I mean, come on if I can get * running with no major issues, how hard can it be :/ |
| 02:23.52 | ManxPower | wmandra, I just say "I can't help you with Asterisk@Home" |
| 02:23.53 | deezed | is that new asterisk book any good? |
| 02:24.39 | wmandra | deezed, haven't seen it yet, but I'd be interested to hear from others that have... |
| 02:25.11 | deezed | looks pretty good.. i guess i may just buy it |
| 02:25.25 | FuriousGeorge | i got this server. none of my other asterisk servers can connect to it, not can my remote sip client. it always times out. |
| 02:25.30 | twisted[asteria] | deezed, it's good |
| 02:25.33 | FuriousGeorge | in addition, it cant reach anything else |
| 02:25.38 | FuriousGeorge | and i cant figure out why |
| 02:25.44 | deezed | does it have a section on agi? |
| 02:25.49 | FuriousGeorge | however i am able to use nufone to call out |
| 02:26.00 | FuriousGeorge | and i can log local clients into it |
| 02:26.10 | deezed | yep |
| 02:26.28 | twisted[asteria] | deezed, http://www.oreilly.com/catalog/asterisk/toc.html |
| 02:27.17 | wmandra | twisted, has anyone done any kind of write-up on the l3 debacle last night?? I'd be interested in a kind of play by play on how things went awry... |
| 02:27.21 | FuriousGeorge | wtf can this be |
| 02:30.39 | deezed | thanks |
| 02:30.51 | cripito | :) |
| 02:31.29 | cripito | manx did u use realtime in any system? |
| 02:31.43 | JerJer | realtime == bad |
| 02:32.04 | cripito | yeap yeap... but some times is convenian... |
| 02:32.12 | JerJer | no its not |
| 02:32.29 | JerJer | there are less obtrusive ways to achieve a database driven config of asterisk |
| 02:32.40 | cripito | example? |
| 02:32.49 | JerJer | #include |
| 02:32.55 | JerJer | perl script |
| 02:33.03 | cripito | #include include a file |
| 02:33.06 | cripito | right? |
| 02:33.08 | JerJer | original thought |
| 02:33.28 | cripito | and perl scripts... usually can be also as obtrusive as realtime |
| 02:33.42 | JerJer | not 4 sql queries per priority |
| 02:33.49 | JerJer | thats just plan wrong |
| 02:34.05 | cripito | that's extensions realtime ... there i have 2 agree with u |
| 02:34.36 | *** join/#asterisk doughecka (n=Miranda@unaffiliated/doughecka) |
| 02:36.21 | ManxPower | The graph of network jitter looks like a clown on meth on a trampoleen |
| 02:36.33 | JerJer | mmmm meth |
| 02:37.47 | cripito | :)) |
| 02:38.11 | ManxPower | I've done most drugs. Meth just fucks you up SO bad after. |
| 02:39.02 | JerJer | i'll stick to powdered caffeine |
| 02:39.04 | ManxPower | Not good for you. |
| 02:39.32 | ManxPower | JerJer, I stick to coffee flavoured caffeene most of the time. |
| 02:39.32 | JerJer | breathing is not good for you |
| 02:39.42 | ManxPower | Well, at least when it comes to uppers |
| 02:40.08 | cripito | so his brains are usually inmerse in caffeene |
| 02:40.35 | JerJer | ludes man call up the ludes man gets some fuckin ludes |
| 02:40.47 | ManxPower | LOL! |
| 02:40.56 | ManxPower | "Reality is for people that can't handle drugs." |
| 02:41.17 | JerJer | I reject your reality and substitute my own |
| 02:42.10 | JerJer | i need to get No Cure for Cancer on DVD - my vhs copy is just worn out |
| 02:42.28 | cripito | :)))) |
| 02:43.31 | deezed | JerJer is there anyway to convert one of my numbers to sip? |
| 02:43.32 | JerJer | man i picked the wrong week to stop popping amphetamines |
| 02:43.37 | JerJer | is there? |
| 02:43.46 | deezed | nufone numbers |
| 02:44.05 | JerJer | i spoze |
| 02:44.45 | deezed | wait im firewalled. I guess ill just have to keep on trying to get asterisk back up |
| 02:44.50 | JerJer | blah |
| 02:44.54 | JerJer | just register to our proxy |
| 02:45.09 | JerJer | i'll even set you up on the spankin new system, if you are brave :) |
| 02:45.13 | doughecka | PREACH THE WORD! |
| 02:45.26 | doughecka | hmm |
| 02:45.38 | doughecka | system |
| 02:46.07 | JerJer | lord jesus please hear my Pear |
| 02:46.11 | doughecka | hah |
| 02:46.31 | doughecka | 175 hours left on my download |
| 02:47.03 | JerJer | smokin bandwidth there |
| 02:47.06 | deezed | JerJer tell me what to do |
| 02:47.22 | deezed | por favor |
| 02:47.24 | doughecka | well, downloading season 1 of lost |
| 02:47.34 | doughecka | bittorrent thats gone a little slow |
| 02:47.34 | doughecka | :) |
| 02:47.44 | JerJer | deezed: email support@nufone.net with your request and the transaction id of your last payment - cuz email addy's can be spoofed |
| 02:48.13 | JerJer | and since you are running asterisk and want SIP i recommend you utilize our new system |
| 02:48.22 | doughecka | JerJer: whats new about it |
| 02:48.41 | deezed | is there a possibility to have IAX and sip on the same account? |
| 02:48.43 | JerJer | what's not new |
| 02:48.47 | JerJer | deezed: sure |
| 02:49.27 | JerJer | just don't try to send IAX to our SIP proxy |
| 02:49.54 | doughecka | downloaded a gig so far, and I cant watch anything |
| 02:50.17 | JerJer | doughecka: i presume you have not found our new website(s) |
| 02:50.23 | doughecka | nope |
| 02:50.49 | doughecka | nufone.net? |
| 02:51.09 | desktophero | evening everyone - is there a softphone that handles G729 and perhaps is....free...? thanks! |
| 02:51.19 | doughecka | ....................... |
| 02:51.22 | doughecka | maybe.......... |
| 02:51.27 | desktophero | i have been looking around...but have not seen anything |
| 02:51.28 | cripito | that's a good one... |
| 02:51.44 | cripito | firefly 3th party is free? |
| 02:51.48 | doughecka | ...............firefly? |
| 02:51.56 | JerJer | http://staging.nufone.net |
| 02:52.00 | desktophero | i am looking to test the codec, but need to limit the "investment". |
| 02:52.05 | desktophero | firefly .. thanks, I will check that out |
| 02:52.17 | JerJer | don't be cheap just cough up a few bucks |
| 02:52.23 | doughecka | ooh, https! |
| 02:52.48 | cripito | i least i remember run a version with the g729.dll from the voiceage |
| 02:52.48 | doughecka | hmm, call waiting... |
| 02:52.56 | doughecka | interesting idea :D |
| 02:53.09 | doughecka | JerJer: dunno, I didnt notice an error |
| 02:53.11 | Qwell | JerJer: Your cert is broken! :p |
| 02:53.22 | desktophero | JerJer - I am not looking to be cheap just frugal... |
| 02:53.31 | JerJer | same same |
| 02:53.32 | doughecka | JerJer: it says it is VERIFIED! |
| 02:53.48 | cripito | jerjer.... i have 2 said... comparing with the old web... amazing |
| 02:53.59 | doughecka | whats wrong with the ssl |
| 02:54.05 | JerJer | cripito: i have 2 say |
| 02:54.07 | JerJer | :) |
| 02:54.11 | JerJer | thanks |
| 02:54.34 | desktophero | nice look JerJer |
| 02:54.35 | FuriousGeorge | i have three asterisk servers in differing stages of production. all have similar network setups, and use similar configs. one of them, i cant figure out why, cannot register to other boxes, not can other boxes register to it (via iax). strangely, however, its having no problem with my iax provider |
| 02:54.45 | doughecka | ooh |
| 02:55.05 | doughecka | join.nufone.net uses www.nufone.net's ssl cert |
| 02:55.07 | FuriousGeorge | so im at a wall here. anyone have any idea why this is goin on? |
| 02:55.12 | cripito | snif snif the join is the old :( |
| 02:55.31 | desktophero | Furious - what errors - if any - are you getting? |
| 02:55.42 | FuriousGeorge | it just times out. other servers dont even flinch |
| 02:55.43 | doughecka | I dig the traceroute and ping logos |
| 02:56.02 | deezed | sent JerJer |
| 02:56.03 | desktophero | are these all on the same lan? |
| 02:56.17 | JerJer | doughecka: yeah plus browsers don't have CACert's root cert installed, which causes another complaint |
| 02:56.21 | FuriousGeorge | i have remote and local sip clients, they work fine. no they are on separate networks same isp |
| 02:56.34 | doughecka | JerJer: they dont?> |
| 02:56.39 | doughecka | my firefox does |
| 02:56.44 | doughecka | didnt add anything extra |
| 02:56.48 | JerJer | doughecka: i gotta fix both traceroute and ping though :( i would like real-time output of the data |
| 02:56.56 | doughecka | oooh, you mean that 'other' web browser |
| 02:56.58 | wmandra | anyone have any sample configs for multi-homing an asterisk box, say put et0 in the DMZ, and have all the internal sip phones connect via eth1??? |
| 02:57.09 | doughecka | umm, internet something.... internet exploder I think |
| 02:57.14 | desktophero | Furious - hmmm, I have 6 servers (4 on one lan and 2 remote). we had issues with routes at first since the remote servers connect via VPN |
| 02:57.18 | JerJer | doughecka: that's serious news to me - i updated firefox not that long ago and i still get complaints about not knowing the signing authority |
| 02:57.27 | desktophero | can you pastebin your iax.conf, perhaps? |
| 02:57.34 | FuriousGeorge | desktophero: sure |
| 02:57.35 | doughecka | JerJer: well, all I know is that it accepted it just fine |
| 02:57.47 | JerJer | you must have installed their cert at one point |
| 02:58.05 | doughecka | this is a newly formatted box |
| 02:58.07 | doughecka | hmm |
| 02:58.15 | cripito | jerjer still only michigan and 8xx? |
| 02:58.29 | doughecka | unless.... I did copy my firefox config over, maybe its in there |
| 02:58.30 | doughecka | :) |
| 02:59.01 | azzie | JerJer, my firefox does not recognize your ssl cert either, and i defenitely did not mess any ssl settings... all default |
| 02:59.12 | *** join/#asterisk Assid (n=assid@203.115.64.62) |
| 02:59.28 | JerJer | azzie: that is the expected behavior |
| 02:59.30 | doughecka | yea, I must have added it |
| 02:59.39 | doughecka | cool :P |
| 02:59.41 | JerJer | cripito: for the moment at least |
| 02:59.54 | desktophero | Furious - please include the info from a 'good' server and the one that can not connect - if you don't midn |
| 02:59.57 | desktophero | mind |
| 02:59.59 | cripito | :) when u get did from colorado.. plz don't forget me... |
| 03:00.00 | cripito | ;) |
| 03:00.03 | JerJer | i refuse to buy someone else's SIP based DIDs |
| 03:00.18 | JerJer | esp L3 |
| 03:00.35 | doughecka | JerJer: where do you have DIDs? |
| 03:00.39 | cripito | :D i known the feeling ;) |
| 03:01.04 | azzie | JerJer, ah... i see... just curious... why not to buy a "normal" cert? :) |
| 03:01.34 | JerJer | that takes work |
| 03:01.41 | doughecka | and money |
| 03:01.56 | doughecka | and nufone is a non-profit so they dont have much money |
| 03:02.08 | Assid | JerJer: you added more states? |
| 03:02.08 | Assid | non profit? |
| 03:02.13 | JerJer | since when we are non-profit ? |
| 03:02.18 | JerJer | are we |
| 03:02.28 | doughecka | I was joking... :P |
| 03:02.49 | doughecka | thats the joke around my office... "Cant we get non-profit licensing?" |
| 03:03.02 | FuriousGeorge | desktophero: http://pastebin.ca/26237 <---there it is but the problem is not in there |
| 03:03.02 | FuriousGeorge | as far as i can see |
| 03:03.28 | FuriousGeorge | the other two servers, Claudia and Filomena, play fine with eachother using very similar configs |
| 03:03.54 | doughecka | what funny names :) |
| 03:04.02 | FuriousGeorge | basicallly substituting only usernames |
| 03:04.05 | desktophero | which server is having the problems? |
| 03:04.07 | *** join/#asterisk alephcom (n=Miranda@207.34.97.130) |
| 03:04.12 | FuriousGeorge | doughecka: thanks (i guess :) |
| 03:04.33 | FuriousGeorge | i work in a predominantly hispanic neigborhood |
| 03:04.42 | FuriousGeorge | only Filomena |
| 03:05.11 | FuriousGeorge | thats her iax.conf you looked at |
| 03:05.24 | FuriousGeorge | and in there nufone works |
| 03:06.01 | JerJer | mucho grande |
| 03:06.18 | FuriousGeorge | JerJer: asi es, brother |
| 03:07.04 | desktophero | so 'Claudia' connects fine? |
| 03:07.17 | FuriousGeorge | claudia connects to Filomena and vice versa |
| 03:07.25 | desktophero | are they in different LANS/WANS? |
| 03:07.30 | FuriousGeorge | but none of them see or can be seen by juanita |
| 03:07.34 | FuriousGeorge | im sorry |
| 03:07.39 | FuriousGeorge | i told you claudia was acting up |
| 03:07.44 | *** join/#asterisk Jzalae (n=sk@216-220-249-33.midmaine.com) |
| 03:07.55 | FuriousGeorge | juanita cant see or bee seen by claudia and filomena who see eachother fine |
| 03:08.07 | FuriousGeorge | its like a god damn logic problem just to ask the question |
| 03:08.24 | *** join/#asterisk cripito (n=ncripito@c-67-161-130-59.hsd1.co.comcast.net) |
| 03:08.25 | JerJer | smells like net mask or some other routing issue |
| 03:08.46 | desktophero | i don't see a context for juanita - except to register FROM the server you set up in iax.conf. is juanita attempting to connect TO this server? |
| 03:08.56 | wmandra | claudia, juanita, filomena: bad porn names, or names of hurricannes ??? :) |
| 03:09.25 | cripito | voices? |
| 03:10.03 | FuriousGeorge | desktophero: your looking at juanita's iax.conf (shes the invisible blind one) so shes trying to register to and log in to claudia and filomena |
| 03:10.31 | FuriousGeorge | JerJer: netmask where? you mean "outside" of asterisk in linux or on my gateway |
| 03:11.05 | desktophero | are claudia and filomena the one of the servers from your 'register' commands in the iax.conf? |
| 03:11.34 | JerJer | FuriousGeorge: are they talking at the network level - like can you ping and ssh between them? |
| 03:11.35 | desktophero | from juanita, can you ping to the other servers - is IAX the only issue? SIP calls work? |
| 03:11.58 | FuriousGeorge | JerJer: yeah im sshed into all of them now, and they ping eachother and ssh into eachother |
| 03:12.01 | wmandra | jerjer: have you ever setup ast with multiple interfaces? eth0 to sip/iax peers & eth1 to client phones ?? |
| 03:12.13 | FuriousGeorge | its just asterisk on that one server thats handicapped |
| 03:12.25 | JerJer | FuriousGeorge: then it has to be either a firewall or sip/iax.conf is bound to the wrong interface |
| 03:12.31 | JerJer | iptables -L |
| 03:12.33 | FuriousGeorge | 0.0.0.0 |
| 03:13.06 | JerJer | i picked the wrong week to stop sniffing glue |
| 03:13.12 | wmandra | lol |
| 03:13.23 | FuriousGeorge | JerJer: iptables -L on my Gateway box, right? |
| 03:13.39 | desktophero | the two contexts you have for claudia and filomena are for those servers to log into this box...are the server 'names' from the 'register' commands claudia and filomena? are you pinging ip addresses or the server names themselves |
| 03:13.47 | desktophero | are those server names resolving correctly? |
| 03:13.54 | JerJer | wmandra: yes i have setup boxes with more than one nic |
| 03:14.08 | JerJer | FuriousGeorge: on whatever box isn't getting sip/iax communciation |
| 03:14.19 | wmandra | jerjer: did you run into any problems, like cisco ip phones crashing?? |
| 03:14.20 | JerJer | if there is more than one, then all of them |
| 03:14.42 | deezed | JerJer: how many days does it take to provision me to the new server |
| 03:14.46 | JerJer | wmandra: never - my 7960 and 7910 are 100% solid |
| 03:15.00 | JerJer | deezed: i need to actually go check the email and do it |
| 03:15.06 | JerJer | it is a friday night, yo |
| 03:15.16 | deezed | foshizzle |
| 03:15.19 | JerJer | i know shido tain't around |
| 03:15.37 | *** join/#asterisk coppice (n=chatzill@194.143.17.210.dyn.pacific.net.hk) |
| 03:15.44 | FuriousGeorge | JerJer: http://pastebin.ca/26238 <-- the server is 10.0.0.4 behind the network and thats the iptables from the gateway |
| 03:15.44 | *** join/#asterisk _daver_ (n=daver@ns1.tmok.com) |
| 03:16.34 | wmandra | jerjer: hmmmm, for some reason I cant get ast to register via eth0 and my 7960's to register with ast on eth1, both interfaces are on completely different subnets, and iptables is configured correctly..... |
| 03:16.35 | FuriousGeorge | JerJer: im installing it on the asterisk box now, or "emerging it" |
| 03:16.38 | JerJer | leme hook deezed up so he doesn't have a coronary |
| 03:17.03 | deezed | hehe its cool |
| 03:17.07 | FuriousGeorge | by all means sir hookem up |
| 03:17.15 | cripito | :)) |
| 03:17.19 | FuriousGeorge | forthwith |
| 03:17.26 | JerJer | hereintofor |
| 03:19.09 | JerJer | deezed: hell you coulda did this |
| 03:19.09 | JerJer | :( |
| 03:19.50 | *** join/#asterisk breeves (n=breeves@ppp-70-242-183-158.dsl.lbcktx.swbell.net) |
| 03:20.15 | deezed | your the leet one |
| 03:20.39 | JerJer | adding a device in members.nufone.net does work :) |
| 03:20.44 | breeves | Can someone recommend a good receptionist/switchboard program? |
| 03:21.03 | JerJer | one just needs to be informed of the config deals - which does not work :P |
| 03:21.13 | FuriousGeorge | breeves: ive heard good things about one called FOP or something |
| 03:22.09 | breeves | Flash Operator Panel, I've tried it out, but I wondered if anything else was worth a look. |
| 03:23.17 | deezed | wait.. where did that new login screen come from? |
| 03:23.29 | JerJer | breeves: pencil and paper |
| 03:23.48 | deezed | wow.. i was using the old bare one |
| 03:24.29 | deezed | i am impressed |
| 03:24.34 | JerJer | staging.nufone.net and its associated members site hasn't launched quite yet |
| 03:24.49 | JerJer | quite a lot works on members |
| 03:25.00 | JerJer | but quite a lot don't so ya know |
| 03:25.59 | FuriousGeorge | so i swiched to middies cuz dank was too expensive. im counting on this telephony thing to pay off |
| 03:26.22 | JerJer | homegrown is the only way to go |
| 03:26.31 | FuriousGeorge | roomates |
| 03:26.36 | JerJer | kick em out |
| 03:26.43 | FuriousGeorge | broke-assed scrub roomates |
| 03:26.58 | deezed | i got a spa1001.. i am following the settings. what should i put for unique identifier? (i assume calling party number is cid?) |
| 03:27.23 | JerJer | unique id is a globally unique value you provide to us for authentication |
| 03:27.47 | FuriousGeorge | i'd have to grow a lot to cover the rent |
| 03:27.58 | JerJer | and if you pick something that is already in use you won't like the error message - i will be trapping for that very soon |
| 03:28.22 | JerJer | FuriousGeorge: damn you live in Manhattan or something |
| 03:28.54 | FuriousGeorge | no, but we split a house 4 of us for 2400 |
| 03:29.00 | JerJer | ugh |
| 03:29.11 | JerJer | my house is paid off - thankfully |
| 03:29.23 | JerJer | car and all the toys are as well :) |
| 03:29.25 | cripito | 2400 each? |
| 03:29.37 | FuriousGeorge | if i could afford to live in manhattan i'd be there :) (600 each) |
| 03:29.46 | JerJer | it goes for $3000 a pound up here yo |
| 03:29.48 | cripito | :D aaaaaaaaaaaaaaaaaah |
| 03:30.30 | FuriousGeorge | about the same here, q/p for 1000, anything more than that was always "out of my league" even in college :0 |
| 03:30.59 | alephcom | You guys would like where I live. Payments are $320CDN on an acreage out of town. :-) |
| 03:31.22 | FuriousGeorge | alephcom: yeah, but how far to the beach |
| 03:31.41 | ManxPower | Ya know, if these mobile phone companies want to make revenue from addon services they HAVE TO WORK FIRST |
| 03:31.53 | FuriousGeorge | halleluja |
| 03:32.27 | FuriousGeorge | cuz the LORD helped JONAS-ah, defeat goliath |
| 03:32.39 | ManxPower | I can't download ringtones, I can't sign on to my AIM account from my cell phone, I can't even set up AIM alerts, I get a server error message. |
| 03:33.41 | ManxPower | I thought at first that I was doing something wrong, never having done that before. |
| 03:33.49 | ManxPower | but when you get server error messages, it's their fault |
| 03:33.55 | FuriousGeorge | The LORD helped DAVID-ah, get out da whale. The LORD helped Gilignaaaaaan get off the ISLAND, yeah-as |
| 03:34.25 | ManxPower | FuriousGeorge, Please step away from the bong. |
| 03:34.38 | FuriousGeorge | (coming to america) |
| 03:34.53 | alephcom | FuriousGeorge: too far. :-( |
| 03:34.58 | FuriousGeorge | i thought it was a microphone, man |
| 03:35.13 | FuriousGeorge | alephcom: i was quoting the movie, but you have an open invite nonetheless |
| 03:37.01 | *** join/#asterisk encode (n=encode@li8-128.members.linode.com) |
| 03:38.15 | FuriousGeorge | what movie: "You want a toe? I'll get you a toe by four o'clock... with nail polish." |
| 03:40.19 | FuriousGeorge | register sill whore |
| 03:40.33 | FuriousGeorge | *silly |
| 03:43.43 | encode | anyone here from australia? |
| 03:44.11 | FuriousGeorge | not me, mate |
| 03:45.28 | encode | from what i've read australia has some unique issues with detecting PSTN hangups, just wondering if anyone here has had any experience with configuring asterisk to get around the issues |
| 03:50.51 | *** join/#asterisk bmg505 (n=leon@rndf-146-0-114.telkomadsl.co.za) |
| 03:55.39 | *** join/#asterisk dlucas (n=root@Broadband-Dynamic-Western84.connect.com.fj) |
| 03:55.48 | dlucas | Hello, |
| 03:56.07 | dlucas | wanting help with a system that does not detect incoming call hangup! |
| 04:00.49 | *** part/#asterisk SplasPood (n=sp@brooklyn.paravolve.net) |
| 04:01.48 | FuriousGeorge | encode: now that you mention it, ur the second aussie ive heard that from |
| 04:02.01 | FuriousGeorge | did you try the wiki |
| 04:02.27 | FuriousGeorge | and theres that aussie voip site. i bleieve its called aussievoip.com or .au or something |
| 04:06.29 | *** join/#asterisk Koshatul (n=evangeli@ip157.net65.ipnetworks.net.au) |
| 04:07.51 | *** join/#asterisk omal_ (n=chris@216.215.208.188.nw.nuvox.net) |
| 04:08.37 | omal_ | ok, i'm sure this is something silly |
| 04:08.57 | omal_ | i just want to play around with asterisk on my laptop and try and learn to write some silly AGI scripts |
| 04:09.22 | omal_ | but i'm having an issue: it appears that the asterisk server and my softphones are fighting for control of the soundcard |
| 04:10.58 | omal_ | i'm sure there are people doing exactly what i'm talking about...its probably just a linux soundsystem issue and not an asterisk issue i'm thinking |
| 04:11.06 | *** join/#asterisk Wi_Fi (n=OUT@c-24-127-12-85.hsd1.ca.comcast.net) |
| 04:17.35 | *** join/#asterisk wmandra (n=me@pcp04943183pcs.verona01.nj.comcast.net) |
| 04:18.09 | FuriousGeorge | oops, turns out i did forget to foreward a port, now the other 4 boxes register ok to my server, but its still cant see them |
| 04:18.16 | FuriousGeorge | registrations still timing out |
| 04:18.25 | FuriousGeorge | (iax registrations) |
| 04:20.12 | *** join/#asterisk Corydon76-home (i=green@pdpc/supporter/sustaining/Corydon76-home) |
| 04:21.15 | *** join/#asterisk diamon (n=diamon@c-67-191-83-209.hsd1.fl.comcast.net) |
| 04:24.40 | JerJer | ok cvs -head doesn't compile with the utils.c change tonight |
| 04:24.54 | Qwell | hmm, does an iaxy require dhcp? |
| 04:24.58 | JerJer | no |
| 04:25.02 | Qwell | because friends don't let friends dhcp |
| 04:25.10 | FuriousGeorge | lol |
| 04:25.14 | Qwell | any clue what the default IP is? |
| 04:26.28 | cripito | jerjer i just cvs and is compiling... |
| 04:26.44 | FuriousGeorge | bind addr in iax.conf of 0.0.0.0 means it accepts connections on all interfaces right? |
| 04:26.48 | cripito | or u are talking about a dif utils.c? |
| 04:26.51 | JerJer | cripito: give it a min |
| 04:27.15 | JerJer | yes utils.c blows up on gcc v3.2.2 |
| 04:27.26 | JerJer | doesn't like hp->hp.h_addr_list = hp->buf; |
| 04:27.45 | cripito | i tell u in a min |
| 04:28.10 | cripito | mine did |
| 04:28.15 | FuriousGeorge | u guys think water for a glacier would taste good |
| 04:28.24 | cripito | gcc 3.4.2 |
| 04:28.36 | JerJer | utils.c:195: warning: assignment from incompatible pointer type |
| 04:28.51 | JerJer | make: *** No rule to make target `config_old.o', needed by `asterisk'. Stop. |
| 04:29.07 | JerJer | might be a makefile problem on my end then |
| 04:29.08 | JerJer | hmm |
| 04:29.34 | cripito | i am installing ;) |
| 04:30.08 | JerJer | ahh i love gig-e |
| 04:31.02 | JerJer | and the cvs mirror being 0.124 ms away |
| 04:31.48 | JerJer | ok yeah it was just a makefile fuckup on my part |
| 04:31.56 | JerJer | but the new utils.c fix today caused a warning |
| 04:32.51 | *** part/#asterisk wmandra (n=me@pcp04943183pcs.verona01.nj.comcast.net) |
| 04:32.51 | *** join/#asterisk _Madar (n=tophe@219-84-129-163-adsl-tpe.static.so-net.net.tw) |
| 04:33.04 | cripito | :) |
| 04:34.19 | *** join/#asterisk paxr0 (i=jjhhjkh@216-155-93-60.bk2-dsl.surnet.cl) |
| 04:34.27 | cripito | i am going for a donna.. :P my asterisk is working at home |
| 04:35.46 | paxr0 | how i can block all incomming calls (PSTN in fxo)?, where i can read about this... |
| 04:36.12 | JerJer | exten => s,1,Hangup |
| 04:36.18 | JerJer | in the right context |
| 04:37.07 | paxr0 | hummm in default context , for my outcalling calls im using other context... thanks :) |
| 04:37.37 | FuriousGeorge | JerJer: turns out when i frwrded the port on the box, other boxes were able to authenticate with it. but its still timing out when registering with them |
| 04:38.06 | JerJer | why do you need to port forward? |
| 04:38.18 | JerJer | asterisk really should have a routable ip |
| 04:38.22 | FuriousGeorge | i dunno, but it worked. my servers i behind nat |
| 04:38.37 | JerJer | good luck |
| 04:39.28 | FuriousGeorge | well i dunno if this approaches the same thing, but i use a dynamic ip service and its worked fine with the first two servers, sip and iax remote and local, but this one is having a lot of trouble with something |
| 04:40.04 | FuriousGeorge | plus i got the one at home which makes four, and it works fine with the other two, but that same one cant log into it |
| 04:42.17 | *** join/#asterisk newmember (n=newmembe@S010600a0c93dce87.cg.shawcable.net) |
| 04:43.55 | FuriousGeorge | i can even log a remote sip client from one network to the other. |
| 04:44.28 | *** join/#asterisk wmandra (i=wmandra@pcp04943183pcs.verona01.nj.comcast.net) |
| 04:46.11 | FuriousGeorge | besides i thought one of the perks of IAX was how well its nat friendliness |
| 04:48.50 | FuriousGeorge | shouldnt these registration timeouts show up in the full log? |
| 04:48.59 | FuriousGeorge | full => notice,warning,error,debug,verbose |
| 04:55.48 | Qwell | there...one provisioned iaxy |
| 04:55.53 | Qwell | I had to use dhcp ;/ |
| 04:57.10 | FuriousGeorge | maybe if i reboot the fucker |
| 04:59.48 | cripito | :D |
| 04:59.57 | cripito | what happen? |
| 05:01.10 | *** join/#asterisk iceyp (n=icepick@jail.unix.co.nz) |
| 05:01.43 | *** join/#asterisk babak (n=root@r-72-244-140-113.nycmny83.covad.net) |
| 05:01.51 | babak | hi |
| 05:02.07 | babak | what units does ztmonitor show? |
| 05:02.12 | FuriousGeorge | holy cow that fricken worked |
| 05:02.29 | babak | when i see 14844. what does that mean? what is the unit? |
| 05:02.41 | FuriousGeorge | JerJer: i just needed to reboot, for whatever reason |
| 05:03.10 | cripito | :)) george how many hrs are u being fighting with this? |
| 05:03.24 | iceyp | hey guys, trying to sort out my dialplan for international calls through voipjet with failover to nufone. |
| 05:03.27 | iceyp | is this correct: |
| 05:03.27 | iceyp | exten => _00.,1,SetCallerID(6499125668) |
| 05:03.27 | iceyp | exten => _00.,2,Dial(IAX2/2513@voipjet/011${EXTEN:2}) |
| 05:03.27 | iceyp | exten => _00.,103,SetCallerID(64273040757) |
| 05:03.27 | iceyp | exten => _00.,104,Dial,(IAX2/icepick@NuFone/1${EXTEN:0}) |
| 05:03.28 | iceyp | exten => _00.,205,Hangup |
| 05:03.36 | babak | anyone here familar with ztmonitor? |
| 05:04.15 | cripito | 00.. that's europe ? |
| 05:04.33 | iceyp | nah, i'll be using the dialplan 0064 or 001 or 0061 etc |
| 05:04.37 | iceyp | from new zealand |
| 05:04.40 | alephcom | I would probably just go 2 then 3 etc. I don't think you need to jump to 103 |
| 05:04.49 | alephcom | I could be all wrong but it works for me that way |
| 05:04.50 | iceyp | but is that correct? |
| 05:05.22 | iceyp | do i need to set a ring length? |
| 05:05.28 | iceyp | for each |
| 05:06.13 | cripito | is correct |
| 05:08.59 | FuriousGeorge | cripito: to answer your question. i'd been dealing with that on and off for about 36 hours |
| 05:09.19 | cripito | geez that hurts.... |
| 05:09.34 | FuriousGeorge | thinking "how can i not be able to get something so basic working. how am i gonna install pbx if i cant even register my servers" |
| 05:09.46 | FuriousGeorge | its always bittersweet to solve something with a reboot |
| 05:09.49 | babak | no one to help with ztmonitor? |
| 05:09.52 | babak | :( |
| 05:09.59 | FuriousGeorge | babak: what about it |
| 05:10.16 | babak | i'm wondering what the unit mean. |
| 05:10.40 | babak | the ouput of ztmonitor is some number like 4534. what unit is that? |
| 05:11.03 | babak | maybe a better quetsion is this: |
| 05:11.31 | FuriousGeorge | what i see is a representation of an analog spectrum |
| 05:11.56 | FuriousGeorge | tx= transmitted rx = received and Rx = prescription |
| 05:11.58 | babak | i'm getting a reading from ztmonitor. i want to adjust my gains by -2db to -3db |
| 05:12.22 | babak | should the new number in ztmontor by halved? |
| 05:12.27 | FuriousGeorge | gains can only be adjusted by percentage i think |
| 05:12.40 | babak | or do i halve my gains? |
| 05:12.42 | FuriousGeorge | and i heard you shouldnt do more than 15, or risk hw damage |
| 05:13.02 | *** join/#asterisk SplasPood (n=sp@brooklyn.paravolve.net) |
| 05:13.15 | FuriousGeorge | i did -10 once when i had an echoy handset and it was too low |
| 05:13.16 | babak | that's my problem. gains are in % |
| 05:13.23 | FuriousGeorge | had to use -5 |
| 05:13.38 | FuriousGeorge | what you trying to do? |
| 05:13.45 | babak | get rid of echo |
| 05:13.58 | FuriousGeorge | you got echocancel=yes set in zapata |
| 05:14.04 | babak | done that |
| 05:14.14 | FuriousGeorge | did you build asterisk with mmx support? |
| 05:14.32 | babak | not sure. i didn't build it. |
| 05:14.40 | FuriousGeorge | a@h |
| 05:14.41 | FuriousGeorge | ? |
| 05:15.15 | babak | on thing. i don't have an echo issue at the beginning of th call. |
| 05:15.32 | babak | i get echo in the call and then it goes away and comes back, etc. |
| 05:16.32 | FuriousGeorge | when do you get echo? calling with voip to the pstn? taking calls from the pstn? |
| 05:16.52 | babak | taking call from the pstn and making calls to the pstn |
| 05:17.00 | babak | i have a tdm card |
| 05:17.26 | babak | echotrainging is on and i'm fine with the begninng of the call |
| 05:17.38 | babak | things get screwy during the call. |
| 05:17.56 | babak | it's like * forget the echo trainingg and trys again |
| 05:18.08 | babak | or something happens to the line... |
| 05:18.53 | FuriousGeorge | tried a different handset? |
| 05:19.04 | babak | no. |
| 05:19.09 | babak | could that be it? |
| 05:19.25 | FuriousGeorge | yes it could |
| 05:19.38 | *** join/#asterisk |Vulture| (n=V@c-69-180-67-53.hsd1.fl.comcast.net) |
| 05:19.43 | FuriousGeorge | i had that once |
| 05:19.57 | cripito | is just echo? |
| 05:20.01 | cripito | in what card? |
| 05:20.06 | FuriousGeorge | tdm |
| 05:20.11 | babak | tdm |
| 05:20.29 | |Vulture| | Anyone here using XO for phone service? |
| 05:20.32 | cripito | t1? |
| 05:20.42 | babak | i'm trying to adjust gains using ztmonitor and milliwatt line |
| 05:20.47 | babak | pots |
| 05:20.52 | cripito | mm |
| 05:21.06 | babak | so, i think i've got tthe 14844 number that everyone is talking about... |
| 05:21.23 | babak | ...and now i've heard that i should bring it down by 2db |
| 05:21.46 | FuriousGeorge | actually, you shouldnt have ech when taking calls unless theirs a problem wiring inside, or something |
| 05:21.46 | babak | i just what to know what 2db is in terms of what ztmonitor shows or in % for the gain settting |
| 05:21.58 | FuriousGeorge | how long can the delay be b/w voip * and phone |
| 05:22.20 | iceyp | anyone here use nufone? |
| 05:22.29 | iceyp | what prefix do they requrie to dial international |
| 05:22.37 | FuriousGeorge | babak: just try rxgain=-5.0 and txgain=-5.0, but first try another handset, and wire |
| 05:22.39 | iceyp | i.e. to dial country code 64 what would i require |
| 05:22.57 | babak | ok. thank you all for your help. take care. |
| 05:22.59 | FuriousGeorge | and dont forget to reload zapata service and restart * |
| 05:23.13 | babak | will do |
| 05:23.19 | *** join/#asterisk digime (n=drooth@ip68-111-235-172.sd.sd.cox.net) |
| 05:23.38 | babak | bye |
| 05:24.32 | alephcom | iceyp: I could be wrong but I'm not sure they enable international calls by default. I think you might have to get in touch with them. Not sure but I heard that a while ago. |
| 05:24.48 | FuriousGeorge | iceyp: from here in the us its exten = _01164.,1,dial(blah... |
| 05:26.31 | iceyp | thanks |
| 05:28.29 | JerJer | FuriousGeorge: _01164X. |
| 05:28.52 | FuriousGeorge | iceyp: listen to JerJer |
| 05:31.20 | JerJer | but there is no reason you need to dial 011 |
| 05:31.47 | JerJer | you can format the dial string to something more appropriate for 'normal' dialing rules in your area |
| 05:33.09 | FuriousGeorge | this is also true. i reenabled 7 digit dialing for local numbers with one line of the dialplan |
| 05:33.17 | FuriousGeorge | it was phat |
| 05:33.53 | JerJer | yep |
| 05:34.11 | JerJer | i even have 4 digit dialing on my phone here |
| 05:34.14 | JerJer | cuz im lazy |
| 05:34.32 | JerJer | i prepend the first 7 digits :P |
| 05:34.39 | FuriousGeorge | 4 digit dialing.. arrrrgggghhhhhh |
| 05:34.48 | JerJer | but we have a total of one exchange here so its trivial |
| 05:34.56 | FuriousGeorge | what for numbers with the same exchange? |
| 05:35.00 | JerJer | yeah |
| 05:35.23 | FuriousGeorge | i did separate file for local calling here to 973589 |
| 05:35.28 | FuriousGeorge | 240 lines or so |
| 05:35.33 | FuriousGeorge | 4 area codes |
| 05:35.42 | JerJer | like i dial 4403 for my friend gary - but my dialplan munges it into a normal 1+areacode+number before shitting it out to switch-1 |
| 05:37.17 | FuriousGeorge | i did it for the one area code with a exten => NXXXXXX,1,goto(1973589${EXTEN},1) |
| 05:37.29 | FuriousGeorge | plus or minus a _ |
| 05:40.59 | FuriousGeorge | JerJer: so your local calling, if you were using local calling via POTS, would just be one line |
| 05:41.33 | FuriousGeorge | or how many exchanges would be local to you if you were using bell michigan or whatever, is what im asking |
| 05:42.49 | JerJer | there is only one exchange we are 'local' to where i live |
| 05:43.06 | JerJer | so i have _XXXX |
| 05:43.15 | JerJer | and add in the 1+areacode |
| 05:45.32 | |Vulture| | Anyone here using XO for phone service? |
| 05:46.16 | Inv_arp | <PROTECTED> |
| 05:46.27 | |Vulture| | yea I have been really busy |
| 05:47.02 | |Vulture| | about the sign a contract for 2 XOptions Flex plans and wanted to know if anyone is using them yet |
| 05:47.10 | cripito | bed time |
| 05:47.28 | FuriousGeorge | JerJer: my local extensions http://pastebin.ca/26249 (i fixed the syntax inside the [ ] since this one) |
| 05:47.42 | cripito | enjoy guys |
| 05:48.16 | FuriousGeorge | night |
| 05:49.34 | *** join/#asterisk Tommmo (n=tps@203.62.181.52) |
| 05:49.43 | Tommmo | anyone tried setting up a failover SIP server in Polycom phones? |
| 05:50.54 | *** join/#asterisk santiago (n=santiago@208.195.215.158) |
| 05:52.37 | Tommmo | ive configured 2 asterisk servers in the phone, but when the first server is taken offline, the phone doesnt seem to register with the new one |
| 05:52.48 | Tommmo | any ideas? |
| 05:52.59 | JerJer | you sure it took the second one? |
| 05:53.08 | JerJer | perhaps the reg timeout needs to expire |
| 05:53.21 | Tommmo | thats what i thought may be the case |
| 05:53.36 | Tommmo | i'm trying to get it such that if it can't contact the first server, it will immediately re-register with the second one |
| 05:53.45 | JerJer | yell at polycom |
| 05:53.53 | Tommmo | i see call attempts straight away on the 2nd server, but the debug shows "proxy auth required" |
| 05:53.54 | JerJer | it should act sanely |
| 05:54.07 | JerJer | it then should go thru the auth process |
| 05:55.27 | Tommmo | --- |
| 05:55.27 | Tommmo | Retransmitting #2 (no NAT) to 172.16.100.2:5060: |
| 05:55.27 | Tommmo | SIP/2.0 407 Proxy Authentication Required |
| 05:55.54 | JerJer | then your fone should respond to that with the proper info |
| 05:55.59 | Tommmo | hmm |
| 05:57.10 | Tommmo | http://mrtg.abodigital.net/polycom/reg.txt |
| 05:57.15 | Tommmo | this is the registration information im using |
| 05:57.51 | power1 | Is there a command lime command to check what pci version a motherboard is? |
| 05:57.57 | iceyp | hey jerjer, can you help me for a minute, trying to setup dialplan with a failover to nufone |
| 05:58.33 | iceyp | exten => _00.,1,SetCallerID(6499125668) |
| 05:58.33 | iceyp | exten => _00.,2,Dial(IAX2/2513@voipjet/011${EXTEN:2}) |
| 05:58.33 | iceyp | exten => _00.,3,SetCallerID(6499125668) |
| 05:58.33 | iceyp | exten => _00.,4,Dial(IAX2/icepick@NuFone/011${EXTEN:2}) |
| 05:58.33 | iceyp | exten => _00.,5,Hangup |
| 05:58.47 | iceyp | is this correct? |
| 05:59.18 | e-Hernick | no. |
| 05:59.29 | e-Hernick | Dialing failover isn't easy |
| 06:00.04 | e-Hernick | I haven't managed to get it to work right myself. I use voipjet and voicepulse, with voipjet as my primary account, but sometimes the call doesnt' go through |
| 06:00.19 | e-Hernick | I have yet to find a way to then automatically dial the failover account |
| 06:00.33 | |Vulture| | it depends on how the carrier handles the call |
| 06:00.56 | |Vulture| | if the carrier picks up the call and gives an error, or a hangup it will not proceed |
| 06:01.14 | e-Hernick | well, the voipjet problem is that the call just stalls |
| 06:01.20 | e-Hernick | it doesnt' start ringing |
| 06:01.40 | *** join/#asterisk brookshire (n=matt@216.215.208.188.nw.nuvox.net) |
| 06:01.42 | |Vulture| | yea prolly cause voipjet answers it, but then it errors internally with voipjet |
| 06:01.55 | e-Hernick | so, what's the fix |
| 06:01.56 | |Vulture| | so voipjet has the call open to you, but not to their PRI |
| 06:02.05 | |Vulture| | remove voipjet from your group |
| 06:02.05 | e-Hernick | yeah, that sounds reasonable |
| 06:02.10 | e-Hernick | wtf ? |
| 06:02.14 | e-Hernick | when voipjet works, it works great |
| 06:02.17 | |Vulture| | or set a timeout on answer |
| 06:02.21 | e-Hernick | and it's cheaper than the other providers |
| 06:02.25 | |Vulture| | but that would piss a user off |
| 06:02.36 | e-Hernick | I wanna use voipjet as my primary, but there's the 10% of the time where it fails, which is annoying |
| 06:02.44 | |Vulture| | 10% |
| 06:02.48 | e-Hernick | can I set a timer on maximum time before ringing |
| 06:02.48 | |Vulture| | damn thats pretty high |
| 06:02.57 | e-Hernick | i know, it's too high |
| 06:03.03 | |Vulture| | e-Hernick: depends on what voipjet reports back |
| 06:03.14 | |Vulture| | if it shows that the line is bridged, then there isn't much you can do |
| 06:03.25 | |Vulture| | cause * thinks the call is fully connected |
| 06:03.28 | e-Hernick | can't i detect ringing |
| 06:03.34 | Tommmo | does anyone know what "lineseize" is on a polycom phone? |
| 06:03.50 | |Vulture| | might be able to now I have never messed with that |
| 06:04.05 | |Vulture| | Tommmo: that could be referring to how many active calls a line can have |
| 06:04.33 | |Vulture| | I believe its like 8 or something crazy on the IP-50X |
| 06:04.54 | |Vulture| | Tommmo: oh was that linesize or seize? |
| 06:04.55 | iceyp | yeah |
| 06:05.04 | iceyp | so e-Hernick u didnt get it working? |
| 06:05.10 | iceyp | sorry was on the phone |
| 06:05.19 | Tommmo | seize (not size) |
| 06:05.19 | e-Hernick | iceyp, no, sorry. |
| 06:05.25 | e-Hernick | I wish I had a fix though |
| 06:05.38 | Tommmo | they talk of a line seize period |
| 06:05.40 | |Vulture| | iceyp: you don't need to reset CID once its set it stays until changed |
| 06:05.43 | Tommmo | but can't find anything on it |
| 06:05.45 | iceyp | i read something about a ,g) option to go to next line if it hangs up |
| 06:05.52 | iceyp | sweet |
| 06:06.25 | |Vulture| | they have changed a lot in HEAD on handling progression |
| 06:06.37 | |Vulture| | it use to be n+101 but now there are some new things I got to read up |
| 06:06.43 | iceyp | e-Hernick like this: exten => s,n,Dial(${PROVIDER1}/${ARG1},,g) |
| 06:06.46 | iceyp | http://lists.digium.com/pipermail/asterisk-users/2005-January/082882.html |
| 06:07.11 | e-Hernick | i dunno, but voipjet doesn't hang up on me |
| 06:07.14 | iceyp | just cant work out how to use it |
| 06:07.14 | e-Hernick | it simply stalls |
| 06:07.18 | iceyp | ahh true |
| 06:07.46 | |Vulture| | yea those dial-CANCEL etc. those are new |
| 06:07.48 | iceyp | I may just have a 00 [contry-code] for voipjet and tell users if it fails to use 01 [country-code] for nufone |
| 06:08.00 | *** join/#asterisk hiren (n=hiren@c-67-170-243-17.hsd1.ca.comcast.net) |
| 06:08.10 | |Vulture| | iceyp: where are you located? the states? |
| 06:08.20 | iceyp | |Vulture| new zealand |
| 06:08.27 | e-Hernick | iceyp, that's unfriendly |
| 06:08.43 | |Vulture| | iceyp: ah okay |
| 06:08.55 | iceyp | e-Hernick i know, but if it's the only solution then it needs to be done ;/ |
| 06:09.18 | e-Hernick | you could let the user interrupt dialing with the # key |
| 06:09.41 | e-Hernick | which would jump to the next dialout provider |
| 06:09.44 | |Vulture| | that dialplan off the list is interesting |
| 06:10.02 | *** part/#asterisk Assid (n=assid@203.115.64.62) |
| 06:10.34 | |Vulture| | PRI is your friend ;P |
| 06:14.20 | iceyp | let me know if u get it workking plz |
| 06:31.13 | *** join/#asterisk santiago (n=santiago@208.195.215.158) |
| 06:35.19 | *** join/#asterisk copantl (n=copantl@205.240.200.117) |
| 06:36.00 | copantl | some one from digium? |
| 06:36.53 | X-Rob | on a friday night/saturday? |
| 06:36.59 | X-Rob | unlikely 8) |
| 06:37.24 | copantl | yeah |
| 06:37.24 | drumkilla | from digium and on the clock? no :) |
| 06:37.28 | drumkilla | but what do you need |
| 06:38.07 | copantl | i just wanna know about UIT signal class 7 and asterisk |
| 06:38.21 | X-Rob | if that makes anyone feel any better. |
| 06:38.37 | drumkilla | copantl: I don't think I am familiar with that |
| 06:38.45 | X-Rob | copantl - I think you want ss7box - http://www.ss7box.com/ |
| 06:38.54 | copantl | nop |
| 06:38.57 | drumkilla | oh, ss7 ... |
| 06:39.20 | copantl | is a standard from UIT for interconection between 2 centrals |
| 06:39.26 | drumkilla | ss7 is not currently supported in asterisk. |
| 06:39.33 | copantl | ok |
| 06:39.35 | drumkilla | however, you can expect it to be coming ... |
| 06:40.26 | copantl | the question is .... when this standard comes ... do i have to use PRI ? |
| 06:40.57 | JerJer | i prefer cans and string |
| 06:41.03 | X-Rob | copantl - yes. SS7 runs over E1/T1 and above (eg, DS3) |
| 06:41.12 | X-Rob | copantl -really. go read ss7box.com |
| 06:41.27 | copantl | and i always use pri right? |
| 06:41.36 | copantl | ok |
| 06:41.40 | JerJer | pri is just signaling |
| 06:41.41 | copantl | thanks |
| 06:43.57 | copantl | thanks a lot X-Rob |
| 06:44.13 | FuriousGeorge | hey JerJer your famous. i just read that article your in, and holy shit btw |
| 06:44.26 | JerJer | in ? |
| 06:44.27 | Qwell | FuriousGeorge: link? |
| 06:44.47 | JerJer | copantl: i smell lots of vapor around ss7box.com |
| 06:44.55 | X-Rob | JerJer - it actually exists, apparently. |
| 06:45.01 | X-Rob | (or so I'm told) |
| 06:45.10 | FuriousGeorge | its old, from voxilla, hold on |
| 06:45.24 | JerJer | then they are holding on to the chan_woomera pieces |
| 06:45.27 | *** part/#asterisk opus__ (n=opus@dahphish.org) |
| 06:45.34 | FuriousGeorge | http://voxilla.com/name-News-article-sid-166.html |
| 06:45.39 | JerJer | er woomera that is |
| 06:46.37 | *** join/#asterisk advorak (n=advorak@atypedigital.ischool.washington.edu) |
| 06:47.27 | JerJer | FuriousGeorge: there is always http://www.forbes.com/business/global/2005/0919/025.html |
| 06:49.32 | FuriousGeorge | i can now officially say: i know that asshole in forbes :D |
| 06:49.35 | FuriousGeorge | j/k |
| 06:49.53 | JerJer | get the rag and you can have a dart board |
| 06:50.07 | FuriousGeorge | lol |
| 06:50.14 | JerJer | for some reason they wanted my silly lookin picture in their mag |
| 06:54.35 | FuriousGeorge | yeah, no mug on the website that i can see. forbes is clearly skimping on the effort they put into the webpage |
| 06:57.38 | FuriousGeorge | and yet the linked article "Is Sex Necessary" gets a slideshow. |
| 06:57.51 | JerJer | they want ppl to drop the $75 bucks per copy |
| 06:59.12 | JerJer | well like $5 but it feels like a lot for a magazine full of advertisements |
| 06:59.59 | FuriousGeorge | I prefer NewsWeek for the funnies |
| 07:00.05 | FuriousGeorge | "Perspectives" |
| 07:03.49 | copantl | can i use a linux box and sagoma card for ss7 and interconnect to a asterisk box through PRI/ISDN? |
| 07:12.23 | FuriousGeorge | ~ss7 |
| 07:12.24 | jbot | i heard ss7 is something that * does not support right now..... given time, who knows :-\ |
| 07:12.38 | X-Rob | copantl - read ss7box.com |
| 07:12.43 | X-Rob | it's all documented there. |
| 07:14.43 | FuriousGeorge | X-Rob: did they modify * or just right a driver for the sangoma sw or something? |
| 07:15.49 | X-Rob | it uses chan_woomera |
| 07:15.56 | X-Rob | so it's a plugin I think |
| 07:16.43 | FuriousGeorge | jbot: no, ss7 support can be achieved using a plugin (chan_woomera), according to X-Rob |
| 07:17.17 | *** join/#asterisk outtolunc (n=me@adsl-69-110-28-66.dsl.pltn13.pacbell.net) |
| 07:18.12 | outtolunc | oops, aw hell, i'm here |
| 07:18.50 | JerJer | fun |
| 07:18.58 | JerJer | now the party can get started |
| 07:19.08 | twisted | party? |
| 07:19.09 | twisted | where/ |
| 07:19.43 | outtolunc | party, i wish |
| 07:19.57 | twisted | there's a party in my pants, and only 2 girls cam |
| 07:19.59 | twisted | came |
| 07:20.04 | twisted | i don't get it. |
| 07:20.53 | *** join/#asterisk WJN (n=kvirc@adsl-69-225-202-166.dsl.irvnca.pacbell.net) |
| 07:21.27 | Qwell | You are invited to a pants party! A party with pants... |
| 07:21.39 | X-Rob | Heh. |
| 07:21.44 | X-Rob | Anchorman |
| 07:21.54 | Qwell | X-Rob: :p |
| 07:22.18 | X-Rob | funny, way underrated, movie 8) |
| 07:24.09 | outtolunc | so i hear astricon 2005 as a hit? |
| 07:24.18 | Qwell | was |
| 07:24.24 | outtolunc | nods |
| 07:24.37 | outtolunc | <- toasted, given to typo's |
| 07:25.14 | outtolunc | who has a pics? |
| 07:25.59 | outtolunc | i expect to see file sleeping on a desktop this time <G> |
| 07:26.21 | Qwell | outtolunc: file couldn't make it |
| 07:26.30 | outtolunc | eh? damn |
| 07:26.55 | outtolunc | he looks forward to that |
| 07:26.59 | JerJer | leme guess - he couldn't get time off work? :) |
| 07:29.11 | outtolunc | well sheesh, anyone do up a narative breakdown of it? |
| 07:29.36 | Qwell | this is easy... |
| 07:29.46 | JerJer | lets see...Damin's speaking, Damin's drunk, Damin's passed out under a piano |
| 07:29.51 | Qwell | outtolunc: day 1: interesting talks, party |
| 07:30.01 | Qwell | day 2: less interesting talks, party |
| 07:30.12 | Qwell | day 3: mostly uninteresting talks, FREE BEER party |
| 07:30.23 | Qwell | day 4: hungover talks, party |
| 07:30.30 | Qwell | JerJer: sound about right to you? |
| 07:30.31 | JerJer | 404 |
| 07:31.10 | drumkilla | 404! |
| 07:32.06 | outtolunc | well *someday it'll swing back to san jose |
| 07:32.15 | twisted | ugh |
| 07:32.19 | outtolunc | hehe |
| 07:32.22 | drumkilla | Dallas next year |
| 07:32.23 | twisted | i love getting blazed on a friday night |
| 07:32.25 | JerJer | drumkilla: i missed 404 ;( |
| 07:32.29 | drumkilla | twisted: w00t |
| 07:32.33 | twisted | drumkilla, ;) |
| 07:32.37 | outtolunc | you say that now, but when i offer a bbq <G> |
| 07:32.37 | twisted | it's been a LOOOOOOONG week |
| 07:32.49 | JerJer | i love getting blazed on any day ending in a y |
| 07:32.53 | twisted | JerJer, :P |
| 07:33.21 | outtolunc | saint forgetfullnessday |
| 07:33.50 | JerJer | drumkilla: and i already offically volunteered to deal with sound in Dallas |
| 07:34.12 | JerJer | some of shit i heard this year was horrid |
| 07:34.35 | outtolunc | 'LA' is a horrid word <G> |
| 07:35.10 | JerJer | HelLA |
| 07:35.31 | Qwell | okay, off to bed |
| 07:35.38 | outtolunc | gnight |
| 07:35.53 | twisted | wheee |
| 07:36.00 | twisted | what was wrong this year? |
| 07:36.26 | twisted | aside from the random cellphone interfering with the audio :P |
| 07:36.41 | twisted | and the overdriven mic |
| 07:36.52 | twisted | or the not-so mic'd speakers |
| 07:36.54 | JerJer | how about the wonderful sideways only mic |
| 07:36.58 | twisted | haha, yeah, well. |
| 07:37.19 | twisted | who's equipment was that? the hotel's? |
| 07:37.39 | JerJer | no they bought it cause the hotel wanted a stupid amount to do sound |
| 07:37.45 | twisted | lol |
| 07:37.50 | outtolunc | sideways only mic... last time i heard that was in reference to a dildo |
| 07:37.55 | JerJer | but they bought pretty low end stuff |
| 07:37.56 | twisted | outtolunc, hahahahah |
| 07:37.59 | twisted | JerJer, i could tell |
| 07:38.06 | twisted | JerJer, were the mixers bought too? |
| 07:38.21 | twisted | cuz they looked like mackie's |
| 07:38.23 | JerJer | i brought my own lil console |
| 07:38.27 | twisted | ahh |
| 07:38.37 | drumkilla | i told him i'd go, but i'd be coming monday night ... |
| 07:38.43 | twisted | drumkilla, / |
| 07:38.45 | twisted | er ? |
| 07:38.47 | drumkilla | d'oh |
| 07:38.54 | twisted | haha |
| 07:38.55 | drumkilla | curse of the up arrow |
| 07:39.02 | twisted | i have a mac |
| 07:39.05 | twisted | mac == great mixer |
| 07:39.06 | twisted | :P |
| 07:39.20 | twisted | (with the right audio card, that is) |
| 07:39.26 | JerJer | next year it will be better. I've got a bunch of wireless mic's and consoles and recording gear |
| 07:39.33 | twisted | schweet |
| 07:39.40 | twisted | ugh |
| 07:39.43 | twisted | damn starbucks liqour |
| 07:39.48 | twisted | i'm all hyper and drunk |
| 07:40.22 | twisted | anywahoo |
| 07:40.28 | twisted | i'm gonna watch a movie |
| 07:40.31 | twisted | nite |
| 07:40.39 | outtolunc | gnight |
| 07:41.38 | *** join/#asterisk DBlue (n=blue@Toronto-HSE-ppp3710756.sympatico.ca) |
| 07:43.37 | outtolunc | reminds me, i think i'm down to 2 now |
| 07:44.07 | outtolunc | time to get some more |
| 07:44.27 | power1 | Is it a good idea to do a "yum -y" update on an a@h 1.5 system? |
| 07:44.40 | Vco | sure |
| 07:45.03 | power1 | Vco, that for me? |
| 07:45.05 | outtolunc | goodluck <G |
| 07:45.07 | outtolunc | > |
| 07:45.07 | Vco | sure |
| 07:45.17 | DBlue | Hay guys , i have a question |
| 07:45.30 | outtolunc | we have an answer |
| 07:46.09 | DBlue | i've just started to read asterisk (the future of telephony) and before i get into the book |
| 07:46.16 | *** join/#asterisk damned (n=vpol@damned.vpol.org.ru) |
| 07:46.23 | DBlue | i would like to make sure if i can use asterisk in the following way |
| 07:46.33 | DBlue | - put it in a server |
| 07:46.40 | Vco | call somehwere...and talk |
| 07:47.03 | JerJer | Brilliant |
| 07:47.04 | DBlue | - log in remotly to the server (which has modems) use it to make phone calls |
| 07:47.08 | Vco | maybe even over the internet |
| 07:47.09 | outtolunc | asking a question, *before* you read about it (in detail).. hmmm surely turns of my *want to respond responce |
| 07:47.18 | JerJer | Brilliant |
| 07:48.06 | Vco | agree even |
| 07:48.14 | DBlue | so that the person who is billed for the phone call is the one with the phone line connected to the server |
| 07:48.32 | JerJer | modems don't jive asterisk |
| 07:48.37 | JerJer | you dig? |
| 07:48.46 | Vco | as opposed to some clown that is in no way, shape or form, attached to the server? |
| 07:49.04 | DBlue | so can it be done? |
| 07:49.14 | JerJer | not with modems |
| 07:49.35 | *** join/#asterisk littleball (n=littleba@cm52.epsilon173.maxonline.com.sg) |
| 07:49.46 | Vco | but video cards...thats where the telephony money is at |
| 07:49.55 | DBlue | thats fine i could buy from digium |
| 07:50.02 | JerJer | there ya go |
| 07:50.31 | JerJer | GPU codec processing :) |
| 07:50.42 | Vco | fucken'a |
| 07:50.55 | Vco | talking at 45 FPS |
| 07:50.57 | DBlue | any recommended card for this purpose |
| 07:51.05 | Vco | Nvidia? |
| 07:51.13 | JerJer | DBlue: depends on volume of outbound calls required |
| 07:51.43 | DBlue | no pressure on the server ... its only me wanting to tab into another intercom server remotly |
| 07:52.05 | DBlue | sorry not an intercom server but you get the idea |
| 07:52.26 | JerJer | you can deal with four telephone lines on a TDM400P |
| 07:52.55 | Vco | anything after that you're really smoking cheap crack and should get a T1 capable card proper |
| 07:53.34 | JerJer | mmm cheap crack |
| 07:53.41 | Vco | not just cheap.. |
| 07:53.42 | DBlue | all four plugs connect to normal pots? |
| 07:53.46 | Vco | but cheap ass |
| 07:53.51 | Vco | cheap ass crack |
| 07:54.03 | JerJer | DBlue: if you buy one with sufficient modules, sure |
| 07:54.21 | DBlue | OK, thanks ... |
| 07:54.34 | JerJer | Thank you, drive-thru |
| 07:54.45 | Vco | i mean, above and beyond NTT etc |
| 07:55.37 | JerJer | like i know you can be overwhelmed and you can be underwhelmed but can you ever be just whelmed |
| 07:56.17 | littleball | hello, i want to install one asterisk box for the every office in different country to save the internation calling cost. who can suggest a cheap solution? (each office has two/three phone lines) |
| 07:56.29 | Vco | fuck this, i'm going to get even more plastered and pass out screaming obscenities at my TV |
| 07:56.43 | X-Rob | littleball - I'd suggest installing asterisk. |
| 07:56.53 | JerJer | and then configuring asterisk |
| 07:56.58 | X-Rob | and then using asterisk |
| 07:57.03 | Vco | and then blaming asterisk |
| 07:57.13 | Vco | and then whining about it here |
| 07:57.18 | X-Rob | then opening your wrists and blaming it all on digium. |
| 07:57.23 | littleball | X-Rob, there is small PBX already in each office. I cannot just remove pbx and add other sip phones . |
| 07:57.38 | Vco | bullshit |
| 07:57.40 | JerJer | then hybrid into the pbx, yo |
| 07:57.44 | outtolunc | then getting internet access (with sufficient bandwidth to suffice those calls on *that codec) between those offices |
| 07:58.56 | littleball | i just imaging the following solution: existing office phones-->PX-->asterisk box--->internet----asterisk box in office 2-->PBX-->office phone. |
| 07:59.04 | JerJer | sure |
| 07:59.08 | X-Rob | no probs. |
| 07:59.09 | littleball | PX/PBX. |
| 07:59.28 | JerJer | just use a trunk line on your existing PBX to interconnect with Asterisk |
| 07:59.41 | outtolunc | probs being, proto used, firewalls/switches that *might be in the middles |
| 07:59.42 | littleball | in this way, i need to add a computer for each PBX, right? |
| 07:59.43 | JerJer | which means putting FXS ports into Asterisk |
| 07:59.45 | outtolunc | er -s |
| 08:00.07 | JerJer | littleball: yes |
| 08:00.24 | littleball | JerJer, i am not familiar with PBX. what is the protocol between pBX and asterisk? |
| 08:00.33 | JerJer | phone line jive |
| 08:00.40 | JerJer | FXS -> FXO |
| 08:00.41 | JerJer | or T-1 |
| 08:00.44 | littleball | and how to connect the PBX with asterisk? |
| 08:00.45 | X-Rob | littleball - Stop right there. You should spend some time with voip-info.org |
| 08:00.52 | JerJer | depending on the volume you require |
| 08:00.52 | X-Rob | you're _way_ under-knowleged to even start thinking about this. |
| 08:01.01 | JerJer | word |
| 08:01.10 | outtolunc | *2 |
| 08:01.15 | JerJer | maybe even __WaY__ |
| 08:01.23 | X-Rob | fwor. |
| 08:01.57 | littleball | X-Rob, different people use different ways to ask questions and discussions. |
| 08:02.31 | outtolunc | littleball: but most at least prepare the question |
| 08:02.48 | X-Rob | ~docs |
| 08:02.50 | jbot | it has been said that docs is Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk |
| 08:03.01 | X-Rob | littleball ^^^^ start there. |
| 08:03.09 | X-Rob | if you don't understand something, feel free to ask here and we'll try to explain |
| 08:03.16 | X-Rob | but you don't understand the base concepts. |
| 08:03.33 | X-Rob | and it's like trying to teach a slug particle physicis. |
| 08:03.36 | X-Rob | physics even |
| 08:04.04 | JerJer | or teaching a monkey to spell |
| 08:04.11 | littleball | i already read those. I also setup my own TE4110P systems. My really question is: is it possible to have a cheaper solution instead of installing a computer for each PBX? |
| 08:04.35 | JerJer | littleball: possibly a media gateway |
| 08:04.45 | JerJer | if you are prepared to deal with the bullshit |
| 08:04.52 | JerJer | get a mini-itx based system |
| 08:05.11 | outtolunc | but yet, that still IS another computer |
| 08:05.35 | JerJer | yep just slightly more customized |
| 08:05.51 | outtolunc | (is amazed jerjer didn't mention a wrt54gs<G>) |
| 08:05.54 | littleball | i am talking about cost, outtolunc. For customers, installing a computer for a small PBX is not acceptable. |
| 08:06.44 | pooh_ | jt55ynp;0l[k,‮bhttp://www.mail-archive.com/asterisk-users@lists.digium.com/msg30672.html[i;/\yjmn\0' |
| 08:07.08 | JerJer | looks like a kitty cat just ate pooh |
| 08:07.09 | outtolunc | littleball: if your 'customers' aren't able to cover the 'cost' of a 'computer' as the REPLACEMENT for their existing 'pbx' they are not looking at it the right way |
| 08:07.34 | JerJer | outtolunc: non-trivial to get zaptel pci card in a wrt |
| 08:07.43 | outtolunc | haha |
| 08:07.53 | outtolunc | read up |
| 08:08.37 | outtolunc | if his 'customers' are 'business' clients, they need to look at their existing pbx costs |
| 08:08.40 | littleball | outtolunc, i am talking with an PBX installer who has 1000 customers(PBXes). So, the cost of the solution is very important. |
| 08:08.54 | littleball | outtolunc, you cannot replace the PBX |
| 08:09.05 | outtolunc | excuse me? |
| 08:09.05 | JerJer | Asterisk IS a PBX |
| 08:09.21 | outtolunc | you obviously didn't understand *what you read |
| 08:09.32 | outtolunc | re read it |
| 08:09.39 | littleball | the PBX installer won't work with me if i ask him to replace the existing PBXs. :) |
| 08:09.57 | JerJer | then don't work with him |
| 08:10.01 | JerJer | put him out of business |
| 08:10.03 | littleball | the value is to save the international calling cost. that is all |
| 08:10.24 | JerJer | then hook an asterisk box in to the equation |
| 08:10.39 | littleball | JerJer, :) lots of VoIP companies want to work with them. Because they lock the customers already. |
| 08:10.47 | JerJer | we are not going to hand you a solution here in IRC |
| 08:11.21 | outtolunc | you want me/us to but batteries in our magic wands and make *unknown* pbx systems talk to asterisk (that is virtual, since you don't want to buy a box for it) and expect us to take you seriously |
| 08:11.31 | outtolunc | er but=buy |
| 08:11.41 | outtolunc | er put |
| 08:11.44 | outtolunc | sheesh |
| 08:12.26 | outtolunc | why don't you tell us what THE EXISTING pbx are? |
| 08:12.41 | outtolunc | or are there none? |
| 08:12.59 | outtolunc | and those 1000 *clients are just *home users |
| 08:13.26 | littleball | outtolunc, can you image what are the existing pbx? all PBX models are included . Because almost all companies here are using pbx and now almost all companies have branches in other countries. |
| 08:13.44 | littleball | companies |
| 08:14.38 | outtolunc | littleball: i can imagine alot (imagine doing this shit for over 25 years) so if i ask you to be specific, i'm attempting to help |
| 08:14.50 | outtolunc | if you don't want that help, then fine |
| 08:15.36 | littleball | i am interested in the mini-its. What is it and how is the price? i am googing now |
| 08:15.48 | outtolunc | i don't come here often for AN obvious reason, you just happen to be the flavor of the minute |
| 08:16.23 | littleball | thanks. |
| 08:16.45 | outtolunc | you realise that 'being interested in the mini-itx' systems ARE computers.. right? |
| 08:16.55 | littleball | yes |
| 08:17.05 | outtolunc | nevermind.. |
| 08:17.24 | littleball | but i never use such box. so i am appreciate if you can share me more info about this |
| 08:17.36 | outtolunc | www.mini-itx.com |
| 08:17.44 | outtolunc | get 'a clue' |
| 08:17.49 | littleball | i am on this site already |
| 08:18.37 | littleball | one more question, any one use such mini box for asterisk installation? |
| 08:18.38 | *** join/#asterisk oej (n=Olle@apollo.webway.se) |
| 08:19.07 | outtolunc | my suggestion to you, read the wiki again, read about your *options, then some back, if you can ask a question that shows you put some effort into this, i'll be more than happy to help you |
| 08:19.20 | littleball | ok |
| 08:19.44 | outtolunc | er some=come |
| 08:20.03 | outtolunc | seriously |
| 08:20.22 | encode | i have a question - how can i decrease the length of time / silence, between dialing a number and hearing it ringing in the headset? |
| 08:20.40 | outtolunc | to answer your last question, yes, there are some that even do it on NON i386 hardware |
| 08:20.48 | encode | im using asterisk@home 1.5 |
| 08:22.46 | encode | any ideas as to what configuration options i should be looking at? |
| 08:25.30 | *** join/#asterisk hatems (n=hatems@196.203.15.218) |
| 08:25.36 | hatems | hi all |
| 08:26.15 | FuriousGeorge | this iax trunking i hear about.... does it just kinda happen automagically? i see nothing on the wiki about enabling it or anything |
| 08:27.17 | pooh_ | ]\]=[hy7jm |
| 08:28.28 | FuriousGeorge | pooh_: thats what i was saying |
| 08:32.03 | outtolunc | fg, depending what asterisk version you are running will define if it is automagically run, or not |
| 08:33.04 | outtolunc | simple test, do an 'iax show netstats' or similar, if you see 40ms base, trunking is active |
| 08:34.37 | littleball | outtolunc, mini itx will works. thanks. it is slightly cheaper than normal pc but size is much smaller |
| 08:34.51 | outtolunc | if you want it, adding 'trunk=yes' to a particular iax client will/should force it |
| 08:35.45 | *** join/#asterisk Falle (i=falstaf@voip-forum.se) |
| 08:35.52 | pooh_ | FuriousGeorge: sorry, cat walked over the keyboard |
| 08:35.57 | outtolunc | littleball: do you realize that mini-itx systems (the nice ones) are actually MORE expensive than just plain ol desktop's acting as servers |
| 08:36.20 | outtolunc | ? |
| 08:36.21 | *** join/#asterisk Lurr (n=pr0ph3t@66.243.75.36) |
| 08:36.31 | outtolunc | if not, you need to read more |
| 08:36.40 | X-Rob | usually about 33% more expensive |
| 08:36.50 | outtolunc | ty X |
| 08:37.02 | X-Rob | if you want reliability, you need to go for the via embedded stuff, which is even _more_ expensive. |
| 08:37.14 | littleball | outtolunc, the size cannot be too big. This is also my customers' requirement. Not only the price. |
| 08:37.17 | X-Rob | I pay AU$500 for the POS motherboards that I use |
| 08:37.26 | X-Rob | Point Of Sale |
| 08:37.56 | outtolunc | littleball: i'm not telling to 'do this' or 'do that' i'm telling you that you 'obviously need to read more' |
| 08:38.12 | littleball | of course, i am keep reading |
| 08:38.14 | X-Rob | I would, actually, stronly suggest you find an asterisk consultant in your area. |
| 08:38.15 | outtolunc | k |
| 08:38.29 | X-Rob | pay them for a couple of hours of their time and they can help you out a lot more. |
| 08:39.52 | encode | X-Rob: i take it from your previous comment that you're from australia? |
| 08:40.15 | X-Rob | encode - yeah. |
| 08:40.16 | outtolunc | so, some may say my advise is worthless, i suggest you read more about who i am, and decide for yourself |
| 08:40.57 | X-Rob | outtolunc - uh, I wasn't dismissing your input. I was suggesting that he needs some one-to-one work, and the best way to do that is to hire a consultant. |
| 08:41.23 | outtolunc | X, i wasn't responding to you, you were doing just fine |
| 08:42.05 | littleball | consultant is good. But the cunsultancy business model is not defined properly. Some people can only give you concept which you already understand. For the real problems, they cannot help. |
| 08:42.21 | *** join/#asterisk RoyK (n=roy@55.80-202-161.nextgentel.com) |
| 08:42.36 | outtolunc | and i think that he really does need a consultants help, IF he has a timeframe he needs to hit |
| 08:43.30 | outtolunc | littleball: i already mentioned, go read, come back with a 'real question' and i'd be MORE than willing to answer it |
| 08:44.15 | *** join/#asterisk BladeRunner05 (n=feelme@adsl-ull-97-68.44-151.net24.it) |
| 08:44.34 | littleball | i have no timeframe. I have my own business. I am discovering new business oppotunity. RoyK, that is good. :) |
| 08:45.09 | power1 | Is it a good idea to do a "yum -y" update on an a@h 1.5 system? could it possibly mess up a working system? |
| 08:45.11 | outtolunc | littleball: no you aren't |
| 08:46.08 | outtolunc | meaning, royk can be me, it the worse way <G> |
| 08:46.45 | X-Rob | power1 - yes. yum update. |
| 08:46.50 | X-Rob | always yum update. |
| 08:47.25 | power1 | X-Rob, so i should run a "yum -y" ? |
| 08:47.35 | X-Rob | 'yum update' should do it |
| 08:47.43 | outtolunc | yum it up like there is no tomorrow, that is as long as you understand after kernel xyz, the source isn't available, so you have to compile you own |
| 08:48.12 | X-Rob | outtolunc - most A@H users don't care. |
| 08:48.17 | outtolunc | k |
| 08:48.27 | power1 | X-Rob, outtolunc thanks.... |
| 08:48.39 | outtolunc | just noting that brickwall that *can occur* <G> |
| 08:48.47 | X-Rob | FWIW - openpbx is _not_ using /etc/asterisk (potentically clobbered in an updgrade) for config files |
| 08:49.02 | X-Rob | Damn I'm happy about that |
| 08:49.04 | X-Rob | it's always worried me |
| 08:49.09 | X-Rob | just a low grade niggle. |
| 08:49.13 | X-Rob | but a niggle indeed. |
| 08:49.36 | power1 | can some1 elaborate on this issue..according to digium they say that their 400p card requires a pci 2.2 slot, but that also say minimum specs are p3 500...correct me if im wrong, but p3's dont have version 2.2 pci slots? |
| 08:49.52 | *** join/#asterisk brookshire (n=matt@216.215.208.188.nw.nuvox.net) |
| 08:50.07 | outtolunc | power1, there are t400p and tdm400p cards, which is it |
| 08:50.18 | X-Rob | te400 tdm400 |
| 08:50.37 | power1 | tdm400p |
| 08:50.44 | outtolunc | no one i've met says a te as just t |
| 08:50.57 | outtolunc | point made <G> |
| 08:51.31 | outtolunc | so this is just the 4 port pstn card, what modules? |
| 08:51.52 | X-Rob | power1 - I'm sure I can find a P3/500 with PCI2.2 slots |
| 08:52.04 | power1 | X-Rob, can u? |
| 08:52.09 | X-Rob | but, saying that, they _do_ work in 2.1 slots, but they're not supported. |
| 08:52.18 | *** join/#asterisk brookshire (n=matt@216.215.208.188.nw.nuvox.net) |
| 08:52.28 | outtolunc | i didn't realize they *required the 2.2, it was more a suggestion |
| 08:53.05 | X-Rob | outtolunc - and the reason why I know 'em as TE400's, is that we use E1's over here. Not your wierd 1.544mbit connections! |
| 08:54.07 | power1 | I had a weird problem....I did a fresh install on 2 pc's the 1 an intel original 865 board with a p4 3gig HT cpu and the other a Ibm 845 pwntium 4 1.7 gig. On the ibm the tdm400p does not hang up the channel when a caller hangs up when dialing into a fxo port..on the other pc the same tdm400p card works perfectly.... |
| 08:54.15 | outtolunc | well back then, it was an e400p |
| 08:54.35 | power1 | any ideas...i thought this might be related to a pci version problem...on both pc's the tdm is not sharing any irq's |
| 08:54.41 | outtolunc | <- has been doing the 'asterisk thing' for over 3 years <G> |
| 08:56.01 | outtolunc | meaning: they had a t400p and a e400p (the t1/e1) products, which are not whats he's talkin about |
| 08:56.44 | outtolunc | he's talking about the 4 port (possible) pstn card that takes modules |
| 08:57.38 | power1 | outtolunc, yip thats right the 4 port that takes modules..ive got 2 x fxo modules and 1 x fxs module on the card. |
| 08:58.26 | outtolunc | power1: the first thing i'd say is that did you re-compile after installing the card in the new box, or did you just copy over the compiled code |
| 08:58.39 | brookshire | i sooo want one |
| 08:58.39 | X-Rob | brookshire - the sangoma shark? |
| 08:58.44 | brookshire | no.. tdm2400p |
| 08:58.49 | brookshire | from digium |
| 08:58.54 | X-Rob | oh. What's new with that? |
| 08:58.56 | brookshire | 24 ports |
| 08:58.57 | brookshire | :) |
| 08:59.01 | brookshire | with echo can |
| 08:59.03 | power1 | outtolunc, a frsh install from the boot iso...no copying code over. |
| 08:59.09 | X-Rob | so it's like the shark then |
| 08:59.12 | brookshire | either fxo or fxs |
| 08:59.18 | X-Rob | but probably twenty times as much |
| 08:59.23 | X-Rob | what if you want 4 ports with echo can? |
| 08:59.30 | brookshire | it's modular |
| 08:59.35 | brookshire | 4 ports / module |
| 08:59.40 | brookshire | 6 modules total |
| 08:59.41 | X-Rob | so it's like the shark then |
| 08:59.48 | X-Rob | it _is_ the shark |
| 08:59.59 | X-Rob | that' exactly the same as the sangoma shark |
| 09:00.03 | outtolunc | well i noticed you mentioned the i865 |
| 09:00.15 | power1 | yeah |
| 09:00.17 | outtolunc | did you read the notes from digium? |
| 09:00.18 | brookshire | you mean.. the sangoma shark is exactly like digium's cards ;) |
| 09:00.36 | *** join/#asterisk hatems (n=hatems@196.203.15.218) |
| 09:00.39 | power1 | outtolunc, no, notes related to i865? |
| 09:00.41 | hatems | hi all |
| 09:01.02 | brookshire | sangoma's fxo/fxs stuff won't be out for a while, i hear |
| 09:01.03 | X-Rob | Well, possibly. But I think it's more likely that digium licenced the sangoma stuff |
| 09:01.07 | *** join/#asterisk pa (n=Paolo@unaffiliated/pa) |
| 09:01.08 | hatems | any one know fonality |
| 09:01.14 | brookshire | no.. digium did not licence sangoma |
| 09:01.30 | brookshire | if that were true.. sangoma would be in zaptel |
| 09:01.47 | littleball | power1, is it possible that your asterisk server doesn't detect the hangup? in this case, the channel is not released. |
| 09:01.53 | outtolunc | could have swore i seen notes (on the ML) about the i865 over the years |
| 09:02.01 | X-Rob | but you said the card hasn't been released yet |
| 09:02.08 | power1 | littleball, yeah thats exactly whats happening. |
| 09:02.15 | outtolunc | goto the asterisk ml archive and search |
| 09:02.27 | brookshire | X-Rob: ?? |
| 09:02.34 | brookshire | sangoma's card can only do 16 channels |
| 09:02.36 | littleball | this is normal i think. |
| 09:02.38 | brookshire | 2 channels on a module |
| 09:02.41 | power1 | outtolunc, well its the 865 that is working properly , its the ibm 845 thats giving the hassles. |
| 09:02.50 | brookshire | and you have to buy extra bords for more than 4 ports |
| 09:02.51 | X-Rob | 4 channels, 2 channels per board. |
| 09:02.59 | outtolunc | then search the 845 |
| 09:03.04 | X-Rob | 2 boards per card |
| 09:03.10 | X-Rob | s/card/slot/ |
| 09:03.11 | brookshire | 2400p is one board with 24 channels |
| 09:03.14 | X-Rob | ick |
| 09:03.22 | X-Rob | so it is gunna be expensive. |
| 09:03.30 | brookshire | so you can put like 6 2400ps and get 6x24 :) |
| 09:03.36 | littleball | in one of my conference call system, the central office sometimes canot detect remote hangup, especially, the remote participants are in other coutries |
| 09:03.36 | hatems | im interreseted to bye a complet system for call center does any one recomande me fonality company ar any other compnay ? |
| 09:03.42 | *** join/#asterisk coppice (n=chatzill@24.192.17.210.dyn.pacific.net.hk) |
| 09:03.47 | power1 | littleball, u saying its normal for asterisk not to detect a hangup? |
| 09:03.57 | brookshire | i think sangoma is going to have a hard time with fxs/fxo.. analog ain't easy |
| 09:04.03 | encode | why are there no google results for a digium 2400p? |
| 09:04.07 | littleball | power1, this is not asterisk problem. It could be due to central office |
| 09:04.11 | brookshire | it took digium a long time to fix all there issues :D |
| 09:04.46 | outtolunc | littleball: remote supervision (disconnect) is an issue with various systems, specify which |
| 09:04.47 | brookshire | there = their |
| 09:04.48 | brookshire | lol |
| 09:04.48 | X-Rob | Personally, I'd be working on their echo issues as problem number one 8-( |
| 09:04.50 | coppice | analogue FXO and FXS isn't hard |
| 09:04.55 | power1 | littleball, do u mean by central office as telephone company? |
| 09:05.10 | brookshire | coppice: harder than digital |
| 09:05.26 | littleball | power1, the central office sometimes doesn't detect the disconnection. In my service provider case, if it doesn't detect hangup tone, it will use silence detection. But if the remote participants are oversea, local CO desn't help |
| 09:05.33 | littleball | power1, yes |
| 09:05.35 | coppice | brookshire: rubbish |
| 09:06.18 | X-Rob | Ooh |
| 09:06.32 | X-Rob | It's beer oclock |
| 09:06.40 | power1 | littleball, the strange thing is that on the one a&h box it hangs up... but on the other with the same digium card in it..it does not hang up. thats connected to the same central office. |
| 09:06.45 | littleball | power1, so what i did is ask the participants to click "#" to send the DTMF to the system so that asterisk can hanup the channel |
| 09:07.30 | outtolunc | power1: would that other box by chance have a silly 1 port digium fxo card in it? |
| 09:07.54 | BladeRunner05 | Who has experienced with debian, eicon diva 2bri installation ? |
| 09:08.14 | X-Rob | personally, not being anywhere near the clue level of coppice, I thought it was pretty easy for both. Look at the SPA2000 - it's a $100 piece of hardware that works 10 times better than the TDM400 |
| 09:08.34 | X-Rob | and on the te11x, there's only 4 or 5 cips. |
| 09:08.36 | X-Rob | chips |
| 09:08.56 | outtolunc | X and if you need isdn? |
| 09:09.06 | encode | how does the spa 2000 work 10 times better than the tdm400? |
| 09:09.07 | X-Rob | te110 is a 1 port T1/E1 |
| 09:09.14 | power1 | outtolunc, the 2 pc's are right next to each other..im using the exact same tdm400p card for both boxes....ie" i shutdown the box pull out the digium board and test it in the pc right next to it. in the one machine it hangs up..in the other it does not! |
| 09:09.14 | X-Rob | which only has 4 or 5 chips |
| 09:09.30 | X-Rob | encode - send a fax through one. It barfs. |
| 09:09.37 | littleball | power1, the same phone line? |
| 09:09.41 | outtolunc | isdn isn't *just T1/E1 level <G> |
| 09:09.53 | power1 | littleball, yes the exact same phone line. |
| 09:10.00 | outtolunc | afk a sec |
| 09:10.01 | X-Rob | but I can send a fax between two SPA2000s, across ethernet, more reliably. |
| 09:10.21 | X-Rob | outtolunc - I'm clueless. |
| 09:10.32 | X-Rob | he's got the clues in this conversation. |
| 09:10.58 | coppice | X-Rob: sending between the SPA2000s is pure luck. sending through a TDM400 card should be utterly reliable |
| 09:11.08 | littleball | power1, after caller hangup the phone, then pick up the phone again to find out whether the line is busy. |
| 09:11.13 | coppice | notice the use of the word "should" |
| 09:11.44 | littleball | power1, you use another phone to call asterisk, right? |
| 09:11.48 | X-Rob | coppice - I'm aware that it's pure luck. but it does work a hell of a lot better than in and out through a tdm400 |
| 09:12.13 | X-Rob | and of course, plenty of caveats with utilisiation and bandwidth blah blah |
| 09:12.38 | power1 | littleball, I am the caller in this case, im calling on another analog line and using my mobile phone. Even after i hang up I get dial tone again. but asterisk keeps the channel open. so if i phone back its engaged. |
| 09:12.45 | X-Rob | I do neither. I have a POTS line come in for faxes. Saves headaches. |
| 09:13.02 | littleball | power1, this is what i met before |
| 09:13.04 | coppice | X-Rob: notice the sarcastic use of the word "should" :-) |
| 09:13.10 | brookshire | i think asterisk will have x.25 support soon |
| 09:13.20 | brookshire | then the t1/e1 cards should support fax |
| 09:13.25 | littleball | the reason is due to your central office |
| 09:13.27 | X-Rob | coppice 8) |
| 09:14.17 | power1 | littleball, yes of course im calling in with another phone on a different pots line. |
| 09:14.18 | littleball | power1, normally (if not international calling), the co will disconnect you after one minute silence time |
| 09:15.05 | littleball | power1, Phone A--->call asterisk. and you hangup Phone A. THen you use phone C call phone A, is phone A busy? |
| 09:15.19 | hatems | im interreseted to bye a complet system for call center does any one recommand me fonality company ar any other compnay ? |
| 09:15.22 | coppice | brookshire: is there much value in x.25 these days, and do you mean on board DSP for FAXing? |
| 09:15.30 | power1 | littleball, yes its busy |
| 09:15.51 | littleball | yes, sure? if yes, this is what i met before |
| 09:16.09 | power1 | littleball, im 100% sure.!! |
| 09:16.55 | littleball | ok. it is CO problem. You have no other choice. Either ask them whether you can upgrade your serivce plan or like what i did "the user type "#" to terminate the call' |
| 09:17.24 | *** join/#asterisk kisu (n=kisu@2001:618:400:0:0:0:da26:a0d9) |
| 09:17.40 | brookshire | coppice: no wait it's T.38 |
| 09:18.08 | littleball | power1, CO could introduce lots of strange problem. In my office, one phone works fine. If plugin another phone line, it has problem. Everything is the same |
| 09:18.17 | littleball | except the phone line |
| 09:18.38 | coppice | brookshire: you mean hardware support for T.38? |
| 09:18.42 | power1 | littleball, ok..but tell me why it works on 1 box and not the other with the same telephone line and using the same tdm400p card..if it is a HO problem surely it would do the same thing on both a@h boxes. |
| 09:18.56 | brookshire | coppice: no software it looks like |
| 09:19.07 | BladeRunner05 | I'm troubling with divactrl who can help me ' |
| 09:19.13 | littleball | power1, the hardware loh. |
| 09:19.48 | brookshire | anyways.. ni ni |
| 09:20.03 | littleball | power1, i don't know what is the exactly the reason. I cannot understand sometimes, i only change the phone line, it get hangup problem. |
| 09:20.07 | X-Rob | coppice - doesn't that piss you off? |
| 09:20.17 | power1 | littleball, exactly..it has to be a hardware issue. ie the tdm400p does not like the motherboard on the one pc. |
| 09:20.28 | *** join/#asterisk Shoragan (n=shoragan@d072.apm.etc.tu-bs.de) |
| 09:20.43 | X-Rob | power1 - I'd be checking that the zaptel.conf and zapata.conf is the same. |
| 09:20.52 | coppice | brookshire: do you mean the HMP stuff from Intel? |
| 09:20.54 | power1 | littleball, in my case im using the same phone lines on the same ports in both tests..the 1 pc works fine...the other does not. |
| 09:21.07 | X-Rob | coppice: * brookshire has quit IRC ("EPIC4-2.2[730] - oblivion : lates") |
| 09:21.24 | power1 | X-Rob, they are both virgin new installs no diff in the configs. |
| 09:21.31 | *** join/#asterisk Dsbeerf (i=Dsbeerf@modemcable121.176-130-66.mc.videotron.ca) |
| 09:21.33 | littleball | power1, just to clarify, i am not asterisk consultant. I am doing project based on asterisk. :) |
| 09:21.40 | X-Rob | so have you _looked_ at the busy detection stuff? |
| 09:21.41 | *** part/#asterisk Dsbeerf (i=Dsbeerf@modemcable121.176-130-66.mc.videotron.ca) |
| 09:21.48 | X-Rob | like, tried turning it on? |
| 09:22.10 | outtolunc | or changing the params |
| 09:22.19 | X-Rob | anyway |
| 09:22.25 | X-Rob | time for scotch and pizza. |
| 09:22.27 | outtolunc | ^1 |
| 09:22.37 | power1 | littleball, thanks for the help..anyways man..its appreciated...I allways seem to land up with these weird and wonderfuls. |
| 09:22.39 | outtolunc | already had pizza |
| 09:22.52 | outtolunc | shit already had whiskey too <G> |
| 09:22.52 | littleball | power1, my pleasure. |
| 09:22.55 | X-Rob | outtolunc - time for scotch then! |
| 09:23.19 | outtolunc | <- irish whiskey kinda guy |
| 09:23.29 | outtolunc | (so close) |
| 09:23.40 | power1 | <grin> BRB im actually going to go 4 a whisky..I need it. |
| 09:24.08 | outtolunc | power1, read more, then drink |
| 09:24.12 | outtolunc | hehe |
| 09:24.31 | outtolunc | or both at the same time, works for me |
| 09:25.29 | outtolunc | drink, napkin, pen/pencil make for a wild time 'out on the town' |
| 09:27.36 | outtolunc | and yet, i still haven't seen a worth while question in here |
| 09:30.06 | *** join/#asterisk wmandra (i=wmandra@pcp04943183pcs.verona01.nj.comcast.net) |
| 09:30.53 | outtolunc | howdy |
| 09:35.50 | outtolunc | strange thing is, when i come here this is all the interaction that seems to happen |
| 09:36.49 | outtolunc | ok, time to leave so things can get back to normal |
| 09:36.57 | *** part/#asterisk outtolunc (n=me@adsl-69-110-28-66.dsl.pltn13.pacbell.net) |
| 09:40.53 | *** join/#asterisk w0w0 (n=w0w0@36.Red-83-50-231.dynamicIP.rima-tde.net) |
| 09:52.09 | *** join/#asterisk VxJasonxV (n=jason@unaffiliated/VxJasonxV) |
| 09:54.23 | VxJasonxV | Does Asterisk support Speex (or otherwise, some OGG format) for voicemails? |
| 09:57.45 | RoyK | VxJasonxV: iirc asterisk supports all [supported] codecs on voicemail |
| 09:57.58 | RoyK | ogg is not supported in asterisk |
| 09:58.40 | *** join/#asterisk fulgas (n=fulgas@a81-84-116-219.cpe.netcabo.pt) |
| 09:59.07 | *** join/#asterisk [Airwolf] (n=airwolf@airwolf.xs4all.nl) |
| 09:59.48 | *** join/#asterisk WJN78 (n=william@adsl-69-225-202-166.dsl.irvnca.pacbell.net) |
| 09:59.56 | lters | any color syntax highlighter that is good in kate/kde? |
| 10:00.31 | lters | for extensions.conf |
| 10:04.50 | *** join/#asterisk rubble (n=netclass@wbs-146-144-28.telkomadsl.co.za) |
| 10:11.03 | *** join/#asterisk zotz (n=zotz@24.231.36.100) |
| 10:12.48 | hatems | bye evry body |
| 10:14.31 | *** join/#asterisk folsson (n=filip@h147n1fls32o985.telia.com) |
| 10:15.17 | littleball | hatems, what? |
| 10:15.20 | *** join/#asterisk shadebob (n=shadebob@ll81-144-114-192-81.ll81.iam.net.ma) |
| 10:15.22 | shadebob | hi |
| 10:15.48 | shadebob | someone had info on Xorcom Asterbank8? |
| 10:46.39 | VxJasonxV | RoyK, apparently Asterisk 1.2 will support ogg vorbis |
| 10:46.45 | VxJasonxV | however, it needs to support Speex :( |
| 10:47.26 | *** join/#asterisk GingerDog (n=david@82-37-168-14.cable.ubr03.wals.blueyonder.co.uk) |
| 10:48.04 | GingerDog | can anyone offer suggestions on getting caller id working with Telewest PSTN (UK) via a X100p clone? |
| 10:48.38 | *** join/#asterisk brookshire (n=matt@216.215.208.188.nw.nuvox.net) |
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| 11:16.57 | *** join/#asterisk Kliment (n=kliment@a80-186-66-67.elisa-laajakaista.fi) |
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| 11:38.57 | *** join/#asterisk Madkiss (i=madkiss@freenode/staff/madkiss) |
| 11:38.58 | Madkiss | Hi all. |
| 11:39.17 | Madkiss | I have multiple users configured in sip.conf and want my extensions.conf now make to ring _all_ phones if a specific number is called |
| 11:39.20 | Madkiss | How to do that? |
| 11:40.25 | e-Hernick | Dial(SIP/phone1&SIP/phone2&SIP/phone3) |
| 11:42.06 | *** join/#asterisk Tangent (n=Arc_Tang@82-40-187-54.cable.ubr06.croy.blueyonder.co.uk) |
| 11:56.30 | *** join/#asterisk Corndawg_ (i=whoisit@71.196.61.13) |
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| 12:22.35 | *** join/#asterisk Hobbes` (n=calvin@59.92.130.125) |
| 12:24.29 | *** join/#asterisk znoG (n=gs@200.115.218.81) |
| 12:29.22 | Hobbes` | anyone know what could cause this error ? Oct 22 18:02:15 NOTICE[27612]: chan_bluetooth.c:2082 try_connect: Connect to device Schattenj..ger failed: Permission denied (errno 13) |
| 12:29.56 | *** join/#asterisk pa (n=Paolo@unaffiliated/pa) |
| 12:30.33 | JamesDotCom | Hobbes`: obviously a problem with permissions, check what file asterisk is trying to access |
| 12:30.38 | JamesDotCom | either in source or with strace |
| 12:31.30 | Hobbes` | I chown'ed /dev/rfcomm0 |
| 12:31.35 | Hobbes` | and now that error has changed |
| 12:36.28 | power1 | whats the defauly user/pass login for sugarcrm on an a@h 1.5 install ? |
| 12:37.32 | *** join/#asterisk mcf3782 (n=mcf3782@adsl-065-012-184-148.sip.asm.bellsouth.net) |
| 12:38.32 | *** join/#asterisk CodeBot (i=stjohn@212.68.221.41.brutele.be) |
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| 13:02.16 | *** join/#asterisk lcars (n=lcars@gentoo/developer/lcars) |
| 13:02.28 | *** join/#asterisk demetrio (n=demetrio@62.173.180.182) |
| 13:02.40 | demetrio | Hi everybody |
| 13:06.50 | e-Hernick | Hi dr. demetrio |
| 13:08.52 | demetrio | I have a question about asterisk AGI: is it possible to detect when a call that has been Dialed over is answered from within an AGI script? |
| 13:15.39 | Aughey | you mean if a remote computer is using agi to handle acall? |
| 13:16.22 | power1 | Hey all, which VOIP Service Providers do you guys reccomend? |
| 13:17.06 | X-Rob | demetrio - no. |
| 13:27.34 | power1 | Hey all, which VOIP Service Providers do you guys reccomend? |
| 13:28.29 | *** part/#asterisk mcf3782 (n=mcf3782@adsl-065-012-184-148.sip.asm.bellsouth.net) |
| 13:29.58 | *** join/#asterisk CodeBot (i=stjohn@212.68.221.41.brutele.be) |
| 13:30.44 | demetrio | So I'll change the question |
| 13:31.20 | demetrio | Is it possible to bridge a call, without using Dial or other application that lock up everything until the call is hung up? |
| 13:31.27 | *** join/#asterisk nassy (n=nassy@207-38-252-103.c3-0.wsd-ubr1.qens-wsd.ny.cable.rcn.com) |
| 13:33.09 | nassy | hi, im new to asterisk. just set it up with a voip provider: telasip. when i try to make a call from x-lite to anyones number i get a message saying all circuits are busy. is that the voip circuits or a configuration error on asterisk? |
| 13:34.41 | *** join/#asterisk znoG (n=gs@200.115.218.81) |
| 13:35.16 | *** part/#asterisk lcars (n=lcars@gentoo/developer/lcars) |
| 13:35.46 | e-Hernick | demetrio, yeah, meetme |
| 13:36.06 | e-Hernick | demetrio, it's a bit of a pain though if you're dealing with a large number of calls |
| 13:36.18 | demetrio | e-Hernick, meetme would be... ? |
| 13:36.54 | e-Hernick | call conference bridge |
| 13:37.06 | e-Hernick | you have your user call the meetme conference |
| 13:37.13 | e-Hernick | and you connect your second call to the conference |
| 13:37.34 | e-Hernick | then you can do async stuff |
| 13:37.41 | demetrio | really doesn't sound clean, but may do the trick |
| 13:37.56 | e-Hernick | it ain't so clean no |
| 13:38.00 | *** join/#asterisk belo__ (n=some@adsl11.ixpert.hu) |
| 13:39.04 | e-Hernick | but it's very flexible |
| 13:39.31 | e-Hernick | your other option |
| 13:39.33 | e-Hernick | is park and connect |
| 13:39.41 | demetrio | Ok, but now that I think about it better I realize that it won't help me anyway |
| 13:39.51 | e-Hernick | why not |
| 13:40.13 | e-Hernick | I think it will help you |
| 13:40.33 | demetrio | to do async I could pass control to an agi script which could fork |
| 13:43.16 | drumkilla | if you just need to run something when the call is answered, there is an option to dial to execute an AGI script when the call is answered |
| 13:43.37 | demetrio | really? |
| 13:43.44 | drumkilla | yep. |
| 13:43.46 | demetrio | that's exactly what I was looking for |
| 13:43.50 | demetrio | thanks a lot! |
| 13:43.53 | drumkilla | :D |
| 13:43.57 | demetrio | I'll check it out |
| 13:44.17 | ManxPower | Wow! |
| 13:44.37 | ManxPower | My cellular internet connection didn't drop overnight. |
| 13:44.41 | drumkilla | wait a second, I partially made that up |
| 13:45.01 | drumkilla | demetrio: it's actually an option that will execute a dialplan macro |
| 13:45.11 | drumkilla | but then you just call AGI from there |
| 13:45.28 | drumkilla | the M() option |
| 13:45.33 | drumkilla | ManxPower: w00t! |
| 13:45.58 | demetrio | well |
| 13:46.12 | demetrio | M(x): Executes the macro (x) upon connect of the call |
| 13:46.26 | demetrio | does "connection" mean "answer" ? |
| 13:46.31 | drumkilla | yes |
| 13:47.05 | demetrio | well then, sounds perfect |
| 13:47.09 | drumkilla | yay! |
| 13:47.23 | ManxPower | Remember any call out an analog FXO port is considered answered as soon as it's done dialing |
| 13:47.56 | drumkilla | ManxPower: there you go, raining on my parade |
| 13:48.06 | drumkilla | :) |
| 13:49.02 | ManxPower | drumkilla, Of course. |
| 13:49.04 | demetrio | sooo... ManxPower, if the kind of call I have to deal with is voip-versus-pstn, will this problem affect me? |
| 13:49.44 | nassy | pc magazine's review of an asterisk based commercial product says: The system lacks some of the more impressive features found among its competitors, such as conference bridging and integration with Outlook. what is conference bridging? |
| 13:49.48 | nassy | http://www.pcmag.com/article2/0,1759,1867604,00.asp |
| 13:50.07 | ManxPower | demetrio, If you mean "I'll be dialing out using a VoIP connecton with a provider", well, almost ALL providers use PRI for the PSTN interface and PRI is not analog |
| 13:50.23 | demetrio | what is integration with Outlook? who the heck would consider integration with Outlook? : |
| 13:51.02 | ManxPower | nassy, Asterisk has "conference bridging" |
| 13:51.18 | ManxPower | demetrio, people that are too lazy to actually dial the phone |
| 13:51.21 | nassy | great. :-) thanks |
| 13:51.29 | ManxPower | nassy, it's called "meetme" |
| 13:51.31 | coppice | Starting with "We were a little skeptical at first about the open-source foundation" tells you what their conclusion will be |
| 13:53.01 | ManxPower | I'm REALLY glad I don't live near the coast anymore. |
| 13:53.29 | drumkilla | why can't they just review asterisk on its own ... :( |
| 13:53.45 | coppice | ManxPower: away from the coast is dangerous - all those tornados :-) |
| 13:54.05 | ManxPower | coppice, Yeah, there is that. |
| 13:54.17 | ManxPower | But I'm SO over hurricanes. |
| 13:55.11 | coppice | drumkilla: why would that be good? comparison is good, if its fair |
| 13:55.56 | ManxPower | coppice, Why should we care about an objective review. No other product wants an objective review. |
| 13:57.11 | coppice | but you either get "the people who pay us the most advertising dollars happen to produce by far the best product" or the car review style. "station wagon X wins because although is gas consumption, load space, smoothness of ride and TCO suck, it has the best 0-60 time". |
| 13:59.47 | nassy | i am slowly replacing a lot of the programs we use at work with open source equivalents. the windows admins prefer to use commercial closed source programs though. (more trustworthy they say) i am hoping to replace our fax server with hylafax and eventually our PBX with asterisk. is it possible to set up a system where asterisk would pass any faxes to hylafax with the DID (and caller id info). DID (T1) -> asterisk box |
| 14:00.31 | nassy | do you get a lot of resistance from windows admins (or others) about using open source? |
| 14:01.18 | nassy | http://www.hylafax.org |
| 14:01.54 | coppice | nassy: see iaxmodem.sourceforge.net |
| 14:02.03 | nassy | thanks |
| 14:02.58 | *** join/#asterisk BladeRunner05 (n=feelme@adsl-ull-159-64.44-151.net24.it) |
| 14:03.35 | nassy | oh my gosh. i cant believe it |
| 14:03.46 | nassy | thats exactly what i wanted |
| 14:04.12 | nassy | is there anything this thing cant do |
| 14:05.31 | coppice | it only runs up to 9600bps right now |
| 14:05.46 | coppice | 14,400bps might be completed soon |
| 14:07.27 | *** join/#asterisk phrog123 (n=francois@ns.menards.ca) |
| 14:08.12 | nassy | thats ok i dont mind that. |
| 14:08.30 | nassy | i think most of our faxes come in at 96 anyway |
| 14:08.33 | phrog123 | How do I PREDICTABLY ensure that * will not answer ZAP/1 - i.e. an FXO port (configured as FXS_KS) |
| 14:08.49 | phrog123 | I want my home pots line to be outbound only from * |
| 14:09.17 | phrog123 | if I do not specify a context into zapata.conf - what context is the default one? |
| 14:11.56 | phrog123 | ne1 here? |
| 14:13.41 | BladeRunner05 | Hi sir, someone has expericed in istalling diva server bri on debian ? |
| 14:17.11 | phrog123 | if I do not specify a context into zapata.conf - what context is the default one? |
| 14:19.19 | *** join/#asterisk n0rf- (n=n0rf@ip152.66.1311D-CUD12K-03.ish.de) |
| 14:23.37 | *** join/#asterisk zyke (n=zakforev@84-45-132-117.no-dns-yet.enta.net) |
| 14:24.52 | phrog123 | if I do not specify a context into zapata.conf - what happens? |
| 14:27.44 | zyke | is meetme supported in realtime? |
| 14:28.24 | ManxPower | phrog123, Bad things happen |
| 14:29.15 | zyke | manx: do u know if meetme is supported in realtime? |
| 14:29.39 | *** join/#asterisk zeedo (n=zeedo@80.68.92.188) |
| 14:30.33 | ManxPower | zyke, I don't see any reason it would not be. |
| 14:30.56 | ManxPower | but honestly, there's not a lot of config options in MeetMe |
| 14:31.09 | *** join/#asterisk ThugDoubt (n=no@216.40.73.165) |
| 14:31.16 | ThugDoubt | hello |
| 14:32.13 | zyke | but you would still need to reload if you can't use realtime |
| 14:32.31 | ThugDoubt | hey i have a question - anyone here to listen? |
| 14:33.48 | ThugDoubt | anyone? |
| 14:34.27 | ManxPower | zyke, Why is reloading a problem? |
| 14:35.19 | *** join/#asterisk twista (i=twista@p5084531E.dip.t-dialin.net) |
| 14:35.52 | twista | re |
| 14:36.00 | ThugDoubt | ok -- well here's my question: We are going live with Asterisk on Friday and I need to get a couple of polycom 501's working from outside of our firewall, I've put eh *box in the dmz and the phone will still not work..... |
| 14:36.40 | file[laptop] | ThugDoubt: externip and localnet needs to be set |
| 14:37.22 | ManxPower | phrog123, don't define the channel iin zapata.conf |
| 14:37.44 | zyke | manx: say you have load balancing across 3 * boxes and if you can't apply new changes from a DB (realtime) you would need to reload each box when you create a new entry in meetme |
| 14:37.46 | ThugDoubt | i set this on the phone config? |
| 14:37.50 | ManxPower | ThugDoubt, nothing like waiting until the last min |
| 14:37.50 | BladeRunner05 | while compiling * 1.0.9 on debian it return this error termap support not found, where I find this ? |
| 14:38.06 | ThugDoubt | ha, well these are not needed right away |
| 14:38.07 | file[laptop] | BladeRunner05: install libncurses5-dev |
| 14:38.15 | BladeRunner05 | thanks file |
| 14:38.15 | twista | does anyone know if its possible to dial a voip-extension from pstn with only one "outside"-number for asterisk? like dialing the 35 for the asterisk box and another 1005 to reach the voip-phone xzy? |
| 14:38.37 | file[laptop] | twista: coming in on the same line you wanna go out on? |
| 14:38.46 | twista | hm? |
| 14:39.05 | file[laptop] | ah |
| 14:39.14 | file[laptop] | DISA... gives you an internal dialtone |
| 14:39.27 | BladeRunner05 | I'm using chan_capi I have to compile first * and after chan_capi or is the same ? |
| 14:39.28 | twista | hm i mean something like: * picks up the call and "asks" for an extension to dial internally |
| 14:39.47 | twista | (or well...it does not have to "ask" but at i guess you get the picture" |
| 14:39.47 | file[laptop] | twista: then use Background, it plays a sound file while waiting for an extension to be input |
| 14:40.14 | twista | hm wait i'll look up that disa-thing |
| 14:41.04 | BladeRunner05 | Another error compiling * /usr/bin/ld cannot find -lssl |
| 14:41.27 | ManxPower | BladeRunner05, read the docs that tell you what external libs Asterisk requires |
| 14:41.42 | twista | BladeRunner05: aptitude install libssl-dev (in case of debian) |
| 14:42.19 | BladeRunner05 | thanks twista |
| 14:42.50 | twista | file[laptop] thanks, thats exactly what i was looking for ;) |
| 14:43.28 | BladeRunner05 | while compiling I see a lot of warings is a problem or is normal ? |
| 14:43.36 | BladeRunner05 | warnings |
| 14:43.49 | nassy | if i set up asterisk with a voip provider (telasip) who is responsible for sending the location info (e911) for emergency calls for each DID. and if me (person running asterisk) does asterisk support it? |
| 14:44.18 | zyke | manx: say you have load balancing across 3 * boxes and if you can't apply new changes from a DB (realtime) you would need to reload each box when you create a new entry in meetme |
| 14:44.43 | file[laptop] | nassy: it's not standard to send any of that info in your calls, you usually give them a list ahead of time... |
| 14:45.27 | *** join/#asterisk mog_home (n=mogorman@user-24-236-84-48.knology.net) |
| 14:46.21 | ThugDoubt | what do I set the externip and localnet to??? |
| 14:46.38 | file[laptop] | here's an example |
| 14:46.43 | file[laptop] | externip=66.199.241.90 |
| 14:46.47 | file[laptop] | localnet=192.168.2.0/255.255.255.0 |
| 14:46.49 | nassy | ok, so the voip provider would need to be notified each time the phone is say (forwarded) to a new physical location, if they dont provide an interface for the user to change. ie, it is the voip provider that is responsible for sending that info to 911 |
| 14:47.19 | file[laptop] | nassy: pretty much |
| 14:47.34 | nassy | thanks |
| 14:47.42 | ThugDoubt | ok so in this example, 66.199.241.90 refers to what? (asterisk box, dns)? |
| 14:47.56 | file[laptop] | asterisk box |
| 14:48.01 | twista | ThugDoubt your "outside-ip" |
| 14:48.27 | file[laptop] | yeah, otherwise like... the outside phone will try to contact your server using your internal IP address for like audio and stuff |
| 14:48.28 | ThugDoubt | can i use something like "office.mycomp.com" or does it have to be an ip? i mean will it resolve? |
| 14:48.38 | *** join/#asterisk lcars (n=lcars@gentoo/developer/lcars) |
| 14:48.55 | ManxPower | ThugDoubt, Asterisk does NOT deal well with DNS. |
| 14:49.03 | ThugDoubt | ok i can use ip |
| 14:49.10 | lcars | is there anyone that has dealt with dtmf sending issues for voicemail when using a pstn <> sip gateway ? |
| 14:49.26 | ThugDoubt | now for localnet, i use the ip pattern of the network the external phones are on right? |
| 14:49.31 | ManxPower | lcars, no, but general DTMF issues are common when you have the wrong DTMF settings |
| 14:49.53 | lcars | tried every setting without luck |
| 14:50.08 | lcars | and I have no control overt pstn <> sip gateway conversion |
| 14:50.18 | *** join/#asterisk SERGEUS (i=sergey@195.112.98.13) |
| 14:50.21 | lcars | using 'info' blocks every dtmf apparently |
| 14:50.48 | file[laptop] | ThugDoubt: localnet is your internal local network, it's how Asterisk figures out whether to put your internal IP address in the SIP messages, or the external one |
| 14:51.32 | SERGEUS | Hi everybody! i want to add some functionality to ExternalIVR() can anybody point me to GOTO() src file? |
| 14:51.50 | BladeRunner05 | someone experienced in compiling chan_capi ? |
| 14:51.54 | twista | file[laptop] is it possible to force rtp-packets getting "proxied" through asterisk? (e.g. like in the situation where asterisk has to alter the codec due to incompabilites between the voip-phones)? |
| 14:52.03 | twista | BladeRunner05 i did so...why? |
| 14:52.11 | file[laptop] | twista: canreinvite=no will force RTP to always go through Asterisk |
| 14:52.13 | *** part/#asterisk lcars (n=lcars@gentoo/developer/lcars) |
| 14:52.18 | *** join/#asterisk Felagund (n=nothing@wnpgmb11dc1-45-172-235.dynamic.mts.net) |
| 14:52.30 | Felagund | morning |
| 14:52.33 | Felagund | I have a question |
| 14:52.36 | twista | oh...hey thx...that's what i've been trying to figure out for a couple of days no |
| 14:52.38 | twista | no = now |
| 14:52.42 | BladeRunner05 | twista: I'm trying to compile chan_capi 0.3.5 but a lot of errors and stop with make: *** [chan_capi.o] Error 1 |
| 14:53.01 | Felagund | is it safe for someone who has never set up asterisk before to set up a company's phone system? |
| 14:53.11 | Felagund | how many things can go wrong with asterisk |
| 14:53.36 | ThugDoubt | thats what im doing |
| 14:53.41 | file[laptop] | Felagund: uh, well - you'd have to do a lot of reading... |
| 14:53.42 | ThugDoubt | it was easy for the most part |
| 14:53.53 | ThugDoubt | but there is a lot of research involved |
| 14:54.01 | twista | BladeRunner05 well i compiled (on debian sarge) the latest asterisk cvs head + the stable chan_capi (from http://www.junghanns.net/en/chan_capi.html) yesterday (and it worked) |
| 14:54.03 | power1 | Hey all, which VOIP Service Providers do you guys reccomend? |
| 14:54.19 | BladeRunner05 | Twista: take a look at this: http://pastebin.com/401924 |
| 14:54.24 | Felagund | I'm not doing it |
| 14:54.30 | Felagund | one of my staff might be |
| 14:54.52 | Felagund | and I don't want to get in shit if it doesn't work and our phone service goes down |
| 14:55.34 | SERGEUS | power1: voipjet |
| 14:55.56 | SERGEUS | power1: lowest prices |
| 14:56.03 | Inv_arp | power1: i use inphonex/inbound voipjet//out |
| 14:56.31 | twista | BladeRunner05 hm i've never seen those errors before (although right now i'm not sure wheather i had to keep kernel sources with capi-stuff or other capi-stuff available for the make process) |
| 14:56.38 | nassy | im new to * and bought the book. the book suggests to have a really reliable power supply otherwise lots of things can go wrong sporadically making it difficult to troubledshoot. it also suggests that using the system with other programs could salso cause problem when those progs compete for cpu time |
| 14:56.49 | Felagund | once it's up and running in asterisk is there a lot that can go wrong |
| 14:56.52 | file[laptop] | I don't understand why people go with the cheapest providers |
| 14:57.03 | twista | well nassy thats quite a general problem ;) |
| 14:57.05 | file[laptop] | yes, they're cheap - but do you really trust them at that price? |
| 14:57.12 | nassy | with = without |
| 14:57.28 | file[laptop] | </rant> |
| 14:57.30 | azzie | file[laptop], they can always blame them on forums later |
| 14:57.46 | BladeRunner05 | Twista: Do u manipulate MakeFile first to compile ? |
| 14:57.56 | twista | no i didnt |
| 14:57.57 | coppice | file: why would trust the expensive ones any more? |
| 14:58.03 | Felagund | hrm |
| 14:58.13 | file[laptop] | because they might use the extra money to do stuff... like technical support |
| 14:58.23 | twista | actually wait...yesterday a manipulated the asterisk makefile to include the mmx-support |
| 14:58.43 | ThugDoubt | what ports do i need to open in my firewall? |
| 14:58.44 | twista | but i didn't alter the chan_capi stuff |
| 14:58.49 | coppice | file: they might consider it better to spend that money on something else, and often do |
| 14:58.52 | ThugDoubt | for SIP polycom 501 |
| 14:59.05 | BladeRunner05 | Ok twista, I'm using * 1.0.9 and chan_capi 0.3.5 could be this a problem ? |
| 14:59.10 | file[laptop] | coppice: so when their service goes kaboom, they get no response, then they finally get a clue? |
| 14:59.40 | twista | BladeRunner05 i had this combination up and running previously |
| 14:59.42 | power1 | SERGEUS, thanks..could u provide url? |
| 14:59.44 | Felagund | hrm |
| 14:59.52 | twista | and i didn't end up with any problems at all |
| 14:59.53 | coppice | file: something like that. price is very little indicator of service quality for any service I ever used |
| 14:59.57 | power1 | Inv_arp, thanks could u provide url. |
| 15:00.01 | Felagund | nobody has any feedback for me? |
| 15:00.07 | SERGEUS | Voipjet provide good quality |
| 15:00.22 | BladeRunner05 | Could be 'cause I haven't installed some source ? |
| 15:00.49 | SERGEUS | besides high price doesn't garant anything except big payments |
| 15:02.42 | azzie | higher-price providers tend to provide more quality service , though it is not correct for all of them |
| 15:03.02 | twista | BladeRunner05 do you have asterisk installed in the standard locations? |
| 15:03.06 | file[laptop] | well I work at Asterlink... and the reason we do 2 cents per minute is because we have actual support |
| 15:03.34 | BladeRunner05 | twista: sure if I start it runs fine |
| 15:03.38 | ManxPower | 2 cents - 3 cents / min seems pretty standard |
| 15:03.48 | SERGEUS | power1: www.voipjet.com |
| 15:03.57 | coppice | my telco is the big dumb telco here and it lets me call the US for about 1cent per minute, and the service just works |
| 15:04.01 | file[laptop] | I mean, I answer calls... answer e-mails within 5-10 minutes... and I've even logged onto a client's box, modified their config (since it was their issue - but they were complaining it was ours), and made it work... |
| 15:04.45 | coppice | telephony pricing is all about volume. enough volume lets great service go with great pricing |
| 15:04.47 | twista | BladeRunner05 hmm i guess i can't give any further hints from here...i currently neither have a linux box here to check things nor access to my asterisk installation |
| 15:05.07 | ManxPower | I personally use Teliax and Nufone. |
| 15:05.19 | BladeRunner05 | twista: tnx |
| 15:05.23 | ManxPower | but I don't expect support from NuFone. |
| 15:05.37 | coppice | I wouldn't use VoIP services. too damned expensive compared to the PSTN :-) |
| 15:05.46 | ManxPower | I gotta record my next support call to NuFone 8-) |
| 15:06.25 | SERGEUS | the goal of v |
| 15:06.31 | file[laptop] | <end slight rant> |
| 15:06.44 | ManxPower | coppice, I only use VoIP service providers for 2 things: 1) oh my god our call volume doubled and we can't get lines installed quickly and 2) wow, I have such low volume usage I'm paying like $2/min if you consider monthly fee |
| 15:07.18 | ThugDoubt | file[laptop]: i set externip and localnet and put * in dmz and phone will still not work! |
| 15:07.49 | file[laptop] | ThugDoubt: then do some debugging and figure it out: sip debug, rtp debug, what exactly won't work? |
| 15:08.06 | file[laptop] | you remind me of one of our support agents, they tell me "it's not working" but won't tell me what "it" is |
| 15:08.19 | ThugDoubt | phone uploads LOG files fine and no errors in log files, but the phone does not connect when i try to call an ext |
| 15:08.30 | ThugDoubt | sorry |
| 15:08.33 | file[laptop] | okay so look on the Asterisk box |
| 15:08.35 | file[laptop] | and see what it's saying |
| 15:08.47 | file[laptop] | if you get nothing in sip debug from the phone, then the packets are not getting to the Asterisk box |
| 15:09.02 | ManxPower | ThugDoubt, Phones on the internet and Asterisk behind firewall/NAT is one of the hardest VoIP things to make work |
| 15:09.12 | ThugDoubt | awesome.... |
| 15:09.21 | file[laptop] | that depends... if you know what you're doing, it's okay |
| 15:09.25 | ThugDoubt | im rebooting the phone now and watching the *box |
| 15:09.26 | mmlj4 | hey ManxPower |
| 15:09.39 | mmlj4 | i'm going to gulfport on tuesday |
| 15:09.44 | file[laptop] | it's the end of the PBX as we know it, and my calls are fine! |
| 15:09.55 | ManxPower | mmlj4, Yeah. I think we are going back to Gulfport on sunday. |
| 15:10.01 | ManxPower | Gulfport is depressing |
| 15:10.09 | mmlj4 | i''m sure |
| 15:10.57 | ThugDoubt | phone IP is 192.168.1.3 and i set the localnet = 192.168.1.0/255.255.255.0 |
| 15:11.16 | ThugDoubt | phone is now started and nothing shows up in * |
| 15:11.20 | ManxPower | ThugDoubt, Um, if the phone is on the same network as the server, then you don't need any of that stuff. |
| 15:11.30 | ThugDoubt | its not on the same network |
| 15:11.33 | lters | using festival, what voice sounds best in the us? |
| 15:11.48 | mmlj4 | the festive one |
| 15:11.54 | lters | the default one? |
| 15:11.56 | ThugDoubt | did i set localnet correctly? |
| 15:11.58 | ManxPower | ThugDoubt, localnet is the ASTERISK local network |
| 15:12.03 | ManxPower | not the phone local network |
| 15:12.05 | ThugDoubt | ok wasnt sure |
| 15:12.06 | ThugDoubt | lemme try again |
| 15:12.18 | ManxPower | mmlj4, I'm on the northshore until wed evening |
| 15:12.28 | mmlj4 | at l'hotel? |
| 15:12.35 | mmlj4 | or john's |
| 15:12.40 | ManxPower | stirling stuff, wireing John's house for ethernet, dealing with united med care Slidell and mandeville |
| 15:12.44 | ManxPower | staying at Johns. |
| 15:12.52 | mmlj4 | ah, yeah... |
| 15:13.02 | ManxPower | I won't be staying at a hotel until there are actual room available 8-) |
| 15:13.09 | mmlj4 | oh, UMC, are they moving on that yet? |
| 15:13.15 | lters | mmlj4, so is theren't a better one than the default? |
| 15:13.24 | ManxPower | mmlj4, Yes, T-1 was installed on tued |
| 15:13.27 | ManxPower | Tues |
| 15:13.29 | *** join/#asterisk wunderkin (i=kev@12-219-162-233.client.mchsi.com) |
| 15:13.33 | mmlj4 | lters: i don't know, i was just being silly |
| 15:13.46 | ManxPower | I guess they had someone install conduit from the outside pedistal to the phone room. LOL! |
| 15:13.46 | lters | kll, thanks. |
| 15:14.03 | mmlj4 | ok... there is 1 room that was locked, so I couldn't ID nor punch down those lines |
| 15:14.57 | mmlj4 | and I think a couple of the wallplates need convincing (longer screws, and one instance of being too close to the funiture) |
| 15:15.02 | mmlj4 | minor stuff, in other words |
| 15:16.01 | BladeRunner05 | twista: I have compiled chan_capi but where start asterisk it return: /usr/lib/asterisk/modules/chan_capi.so: undefined symbol: ast_smoother_feed |
| 15:16.14 | mmlj4 | i want to finish that stuff up... when are you there next? I'd like access to the meds room that's locked, i guess I can call JR for that |
| 15:16.32 | ThugDoubt | file[laptop]: Ok, so its not registering, in sip show peers i get |
| 15:16.40 | ThugDoubt | 25/25 (Unspecified) D 255.255.255.255 0 Unmonitored |
| 15:17.33 | ThugDoubt | do i need to do anything with sip.cfg? |
| 15:19.30 | SERGEUS | Can anybody help me to find APP_EXTERNALIVR maintainer? |
| 15:22.03 | file[laptop] | ThugDoubt: set nat=yes |
| 15:22.53 | ThugDoubt | k |
| 15:28.50 | *** join/#asterisk Wi_Fi (n=OUT@c-24-127-12-85.hsd1.ca.comcast.net) |
| 15:30.15 | *** part/#asterisk SERGEUS (i=sergey@195.112.98.13) |
| 15:31.05 | ThugDoubt | <nat nat.ip="" nat.signalPort="" nat.mediaPortStart=""/> |
| 15:31.29 | ThugDoubt | i dont know what to puut for the values here |
| 15:31.56 | ThugDoubt | thanks for all your help btw |
| 15:35.34 | *** join/#asterisk bmg505 (n=leon@rndf-146-0-114.telkomadsl.co.za) |
| 15:36.19 | *** join/#asterisk ReD-MaN (i=redman@dhcp-0-2-b3-9a-4a-5b.cpe.quickclic.net) |
| 15:45.16 | *** join/#asterisk Wi_Fi (n=OUT@c-24-127-12-85.hsd1.ca.comcast.net) |
| 15:54.39 | power1 | Inv_arp, thanks for the reccomendation, ive just signed up for an inphonex pay as u use account, I was just wondering how long they take to activate accounts so that i can start making calls? |
| 15:55.22 | Inv_arp | power1: like an hour for me |
| 15:55.34 | Inv_arp | power1: i use them for inbound tho |
| 15:58.10 | power1 | Inv_arp, I just signed up with a minimal ammount to give them a test, why do u only use them for inbound? |
| 15:59.00 | Inv_arp | because I dont do much outbound |
| 16:02.12 | power1 | Inv_arp, but why do u use voipjet for out, are they cheaper or is quality better? |
| 16:02.57 | Inv_arp | power1: I had them first, then i got inphonex , but they are cheaper for outbound |
| 16:08.33 | power1 | Inv_arp, thanks.... |
| 16:14.12 | VxJasonxV | Everytime I shut down asterisk, I get: Yuck! Error in buffer handling...: Success |
| 16:14.29 | VxJasonxV | Asterisk does cleanly end, but it's bothering me that I see that every single time :P |
| 16:18.07 | *** join/#asterisk RoyK (n=roy@109.80-202-9.nextgentel.com) |
| 16:21.58 | *** join/#asterisk [Airwolf] (n=airwolf@airwolf.xs4all.nl) |
| 16:22.01 | *** join/#asterisk pa (n=Paolo@unaffiliated/pa) |
| 16:27.55 | drumkilla | mog_home: ! |
| 16:28.03 | mog_home | hey it had to be done |
| 16:28.09 | mog_home | drumkilla! |
| 16:28.24 | file | !!! |
| 16:28.45 | *** join/#asterisk hanchi (n=telliott@68-112-44-203.static.sprn.tx.charter.com) |
| 16:29.34 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 16:29.45 | hanchi | I need to build a production * system, I test with Fecora Core 4 and the test box worked, limited dial plan and only two pots lines, for the production system, would Fedora Core 3 be a better choice |
| 16:30.01 | drumkilla | doesn't matter |
| 16:30.12 | drumkilla | but might as well use FC4 |
| 16:30.25 | mog_home | well fc4 people seem to not like |
| 16:30.31 | mog_home | fc3 would be fine too |
| 16:30.44 | hanchi | thanks, i tested with CVS head, is * 1.2.0 the best choice for the production * |
| 16:31.02 | mog_home | depends |
| 16:31.08 | mog_home | many use it in production |
| 16:31.17 | mog_home | but it is not as bug free as 1.0 branch |
| 16:31.54 | hanchi | is for a relatively critical system, a city hall, so 1.0.9 would be the best choice ??? |
| 16:32.13 | hanchi | best for stability would be fc3 and 1.0.9 ??? |
| 16:33.45 | BladeRunner05 | Someone have installed EICON DIVA server BRI ? |
| 16:34.22 | mog_home | yeah id agree with that hanchi |
| 16:34.32 | mog_home | esp if you dont need the newer featuers |
| 16:35.58 | *** join/#asterisk h4mm3r` (n=h4mm3r@85-18-14-10.fastres.net) |
| 16:36.04 | hanchi | I can upgrade without a major hassle later if I need the newer features, right??? |
| 16:36.12 | mog_home | yeah |
| 16:36.15 | mog_home | its not that bad |
| 16:36.21 | hanchi | cool, thanks, that's a plan then |
| 16:39.49 | *** join/#asterisk shyru (n=shyru@200.94.130.194) |
| 16:39.51 | shyru | hi all!!! |
| 16:42.22 | *** join/#asterisk escribzz (n=escribzz@wsip-24-249-174-19.ph.ph.cox.net) |
| 16:43.27 | *** join/#asterisk obsidian-studios (n=obsidian@c-66-177-188-197.hsd1.fl.comcast.net) |
| 16:45.41 | obsidian-studios | looking for info on why vm's stopped working running * 1.0.9-r1 on Gentoo cli output during vm attempt http://pastebin.ca/26288 |
| 16:46.54 | *** join/#asterisk santiago (n=santiago@208.195.215.158) |
| 16:48.10 | fugitivo | obsidian-studios: check disk space and permissions |
| 16:48.12 | *** join/#asterisk mhnoyes (n=mhnoyes@user-2ivfk3e.dialup.mindspring.com) |
| 16:48.38 | obsidian-studios | fugitivo: doh, checked permissions, forgot disk space, and yes it's maxed out, damit ;) |
| 16:48.46 | fugitivo | obsidian-studios: wait, i think i had that error before, if you delete the file msg0000.wav it'll work, maybe it's a bug |
| 16:48.51 | mog_home | eep |
| 16:49.22 | fugitivo | oh, it was disk space? :) |
| 16:50.13 | obsidian-studios | fugitivo: yes, seems logs stopped getting rotated, so they just filled up the /var partition. thankfully my default is to create many partitions so it is only effecting * vm's not the entire system :) |
| 16:50.32 | fugitivo | that's the way |
| 16:56.49 | *** join/#asterisk iq (n=iq@204-26-72-44.omah.qwest.net) |
| 16:59.48 | *** join/#asterisk pa (n=Paolo@unaffiliated/pa) |
| 17:02.02 | tzafrir_laptop | hopefully the following is not too silly: |
| 17:03.08 | lters | utils.c:611 tvfix: warning too large timestamp 1044275167.1017355626 |
| 17:03.36 | lters | running cvs head, and festival, sending the audio to chan_sccp? |
| 17:03.45 | tzafrir_laptop | I'm trying to test my shiny new goiax.com DID. But I can't figure how to call to that number from here (Israel). What do I need to dial after the international prefix (00 here)? the DID is 7812192817 |
| 17:04.14 | lters | shall I call it ;) |
| 17:04.48 | lters | do u have a us outbound voip provider? |
| 17:04.52 | snitt | tzafrir_laptop: maybe a 1? |
| 17:04.58 | snitt | tzafrir_laptop: is it located in the us? |
| 17:05.02 | tzafrir_laptop | I tried '1' |
| 17:05.06 | *** join/#asterisk pussfeller (n=todd@12.150.129.170) |
| 17:05.08 | snitt | 001 781... |
| 17:05.10 | lters | goiax is MA |
| 17:05.23 | *** join/#asterisk dalfry (n=dalfry@ool-435285b1.dyn.optonline.net) |
| 17:05.26 | snitt | ma should be a state |
| 17:05.45 | lters | y |
| 17:05.48 | wunderkin | should be |
| 17:05.57 | snitt | then dial 1 |
| 17:05.57 | tzafrir_laptop | 781001 21192817 ? |
| 17:06.02 | snitt | no :) |
| 17:06.05 | snitt | 001 7812192817 |
| 17:06.40 | tzafrir_laptop | I'm trying to call to that number. Shouldn't there be a prefix, rather than a suffix? |
| 17:06.50 | lters | you should be able to call your self with goiax . |
| 17:07.12 | lters | 1 7812192817 if you use goiax |
| 17:07.25 | wunderkin | right |
| 17:07.27 | tzafrir_laptop | I can afford a number of PSTN phone calls for testing. goiax don't provide outgoing calls anymore |
| 17:07.43 | tzafrir_laptop | And I want to test calls from PSTN |
| 17:07.46 | lters | inbound is free from goiax, right? |
| 17:08.05 | *** join/#asterisk paxr0 (i=jjhhjkh@216-155-93-228.bk2-dsl.surnet.cl) |
| 17:09.05 | *** join/#asterisk pussfeller (n=todd@12.150.129.170) |
| 17:16.20 | *** join/#asterisk phrog123 (n=francois@ns.menards.ca) |
| 17:17.45 | *** join/#asterisk Astinus (i=iBook@freenode/staff/gentoo.astinus) |
| 17:18.55 | paxr0 | how i can force that my receiver answer before im play using Playback?, exten=> s,3,Playback(soundfile,skip) appear not working |
| 17:19.14 | JerJer | s,1,Answer |
| 17:19.16 | IronHelix | exten => s,2,Answer() |
| 17:19.21 | IronHelix | beaten :| |
| 17:19.31 | JerJer | but playback will answer the channel if its not already up |
| 17:20.10 | *** part/#asterisk santiago (n=santiago@208.195.215.158) |
| 17:20.16 | paxr0 | exten => s,2,Answer |
| 17:20.34 | paxr0 | is the same that exten => s,2,Answer() ? |
| 17:20.45 | IronHelix | yup |
| 17:21.06 | paxr0 | :( is not working |
| 17:21.33 | IronHelix | are you sure the channel is up? perhaps if you are using SIP you have RTP port issues? |
| 17:22.53 | paxr0 | yep , im calling and i can hear the sound but the sound starts when is calling, before answer |
| 17:23.18 | paxr0 | exten => s,1,ResponseTimeout,3 |
| 17:23.19 | paxr0 | exten => s,2,Answer |
| 17:23.19 | paxr0 | exten => s,3,Wait(2) |
| 17:23.19 | paxr0 | exten => s,4,Playback(saldia55,noanswer) |
| 17:23.28 | paxr0 | excuse this flood |
| 17:23.45 | IronHelix | no worries |
| 17:23.48 | paxr0 | replace noanswer/skip |
| 17:24.12 | IronHelix | so you call and saldia55 starts playing as soon as it picks up? |
| 17:24.13 | paxr0 | im trying with all options and i havn`t results |
| 17:24.18 | IronHelix | hmmm |
| 17:25.03 | *** join/#asterisk pa (n=Paolo@unaffiliated/pa) |
| 17:25.40 | paxr0 | play before recept answer |
| 17:26.20 | paxr0 | i think skip is not working in my Playback cmd |
| 17:27.11 | IronHelix | dump the subcommand |
| 17:27.17 | IronHelix | just do playback(saldia55) |
| 17:27.25 | IronHelix | noanswer makes it not check if its answered first |
| 17:27.44 | IronHelix | skip dumps to the next priority if the channel isnt answered |
| 17:28.11 | paxr0 | yep i know im tryed much times with/without skip |
| 17:28.16 | IronHelix | with answer and wait(2) (perhaps should be wait(1)) you dont need a sub argument |
| 17:28.36 | IronHelix | dumb q... did you remember to reload? |
| 17:28.43 | IronHelix | (i've forgotton that a few times) |
| 17:29.44 | jake1932 | i found that on some services you need a noise to initiate the connection - could try playback(beep) before the others |
| 17:31.19 | paxr0 | IronHelix: yep i remember |
| 17:31.36 | IronHelix | or try answer() ringing() wait(2) playback(scaldia55) |
| 17:31.50 | jake1932 | right - same idea |
| 17:31.55 | *** join/#asterisk antoniojulio (n=antonioj@unaffiliated/antoniojulio) |
| 17:32.07 | paxr0 | ok thanks im trying... |
| 17:32.16 | IronHelix | i use that on a few things, for sip channels that take a second to stabilize (ipkall) it makes things nice and makes callers not confused |
| 17:33.04 | jake1932 | yeah - that's a good idea |
| 17:33.38 | IronHelix | cuz you know there are a few idiots that will be frightened and confused by a beep and then *two whole seconds* of silence |
| 17:33.43 | *** join/#asterisk santiago (n=santiago@208.195.215.158) |
| 17:33.53 | jake1932 | no doubt |
| 17:33.57 | paxr0 | hummm interesting :) |
| 17:34.08 | jake1932 | people never actually heard the beep - on my setup |
| 17:34.36 | jake1932 | but the silence - yeah - i'm sure a few may have been confused |
| 17:39.04 | *** join/#asterisk fugitivo (n=ajf@209.13.241.66) |
| 17:40.21 | *** join/#asterisk damned (n=vpol@damned.vpol.org.ru) |
| 17:41.36 | *** join/#asterisk kippi (n=kippi@cpc4-hatf3-6-0-cust33.lutn.cable.ntl.com) |
| 17:41.42 | kippi | I am getting this |
| 17:41.43 | kippi | RFC3389: 5 bytes, level 4... |
| 17:41.49 | kippi | any ideas what it is? |
| 17:42.58 | *** join/#asterisk vmlinuz (n=nabudoco@ns1.ensenada.gob.mx) |
| 17:43.00 | *** join/#asterisk realneo (n=flori@p54B0A814.dip0.t-ipconnect.de) |
| 17:43.11 | realneo | hi |
| 17:45.40 | IronHelix | hi |
| 17:46.24 | *** part/#asterisk santiago (n=santiago@208.195.215.158) |
| 17:49.55 | *** join/#asterisk Abbas (i=Abbas@202.147.162.226) |
| 17:53.26 | *** join/#asterisk |Vulture| (n=V@c-69-180-67-53.hsd1.fl.comcast.net) |
| 18:02.59 | |Vulture| | anyone have an example of the new Dialplan variables and progressions? |
| 18:05.36 | blitzrage | dialplan variables should be in the README files in the doc/ dir |
| 18:08.09 | |Vulture| | okay I just wanted to see an example of the new progression like s,n,Dial ... instead of s,1,Dial and the n+101 etc. |
| 18:13.00 | IronHelix | oh |
| 18:13.05 | IronHelix | you want AEL? |
| 18:13.32 | *** join/#asterisk kFuQ (n=somedude@c-67-185-114-199.hsd1.wa.comcast.net) |
| 18:13.38 | coppice | what? japanese smoked ael? delicious |
| 18:13.42 | IronHelix | http://www.voip-info.org/wiki/view/Asterisk+AEL |
| 18:13.44 | IronHelix | lol |
| 18:13.49 | IronHelix | Asterisk Extension Language |
| 18:14.17 | |Vulture| | IronHelix: thank you |
| 18:14.34 | |Vulture| | my dialplan is starting to get outdated its almost 2 years old |
| 18:14.36 | IronHelix | np :) |
| 18:15.03 | |Vulture| | running head looks like a death blow... WARNING: SetVar is depricated... |
| 18:16.56 | IronHelix | hehe |
| 18:16.59 | mog_home | it still goes |
| 18:17.03 | mog_home | but it just says dont |
| 18:17.05 | mog_home | do it |
| 18:17.22 | |Vulture| | yea |
| 18:17.51 | |Vulture| | the dialplan works fine but there are so many new things that are not being used |
| 18:19.16 | drumkilla | SetVar is just Set |
| 18:19.26 | drumkilla | because you can set more than just a variable ... |
| 18:19.42 | drumkilla | like, dialplan functions! |
| 18:19.45 | |Vulture| | and CID |
| 18:19.47 | |Vulture| | ;) |
| 18:19.49 | drumkilla | show functions at the cli |
| 18:19.57 | |Vulture| | yea I corrected the CID Set awhile ago |
| 18:20.13 | |Vulture| | for the new CIDNUM/CIDNAME how it got split... rightfully so |
| 18:20.47 | *** join/#asterisk trym (n=trym@cD9088B17.sdsl.catch.no) |
| 18:21.50 | *** join/#asterisk drkbl (n=csd@86.109.65.180) |
| 18:23.22 | *** join/#asterisk Mortymer (n=gberenyi@catv-d5dea5ba.catv.broadband.hu) |
| 18:27.32 | alephcom | Has anyone here looked at using Yate to translate between SIP and H323? It is supposed to be able to do upwards of 85 simultaneous calls without any real crazy hardware or work for that matter. |
| 18:29.08 | *** join/#asterisk pa (n=Paolo@unaffiliated/pa) |
| 18:29.56 | drkbl | does anybody know about Junghanns.net? I was not able to call, fax or email them for more than a week now. |
| 18:30.14 | drkbl | till last thursday everything was ok. |
| 18:31.15 | *** join/#asterisk wunderkin (i=kev@24.137.148.145) |
| 18:31.58 | *** join/#asterisk Spum (n=Spum@86.112.145.176) |
| 18:32.02 | Spum | 'ello |
| 18:32.15 | wunderkin | hmm no reverse dns, nice |
| 18:32.45 | Spum | Could anyone point me to a guide on what is good hardware to buy for a relatively cheap PBX setup, and also, what's the name of the linux distribution which installs asterisk? |
| 18:33.12 | IronHelix | hi spum |
| 18:33.17 | Spum | Hello |
| 18:33.19 | IronHelix | hardware depends on what you are doing |
| 18:33.22 | ManxPower | alephcom, Um, Asterisk should be able to do that just fine |
| 18:33.39 | Spum | well, the PBX i'm looking to set up is for a small-sized business |
| 18:33.40 | ManxPower | As long as you don't have codec translation |
| 18:34.00 | Spum | Sorry, i'm brand new to PBX |
| 18:34.05 | IronHelix | how many users/extensions, how many lines, and what kind of lines are they? |
| 18:34.32 | Spum | Hmm, extensions as in # of phone lines? |
| 18:34.41 | Spum | Bare in mind, i'm setting this up in the UK, not US |
| 18:34.44 | IronHelix | extensions like how many phones need to be on peoples desks |
| 18:34.54 | IronHelix | ok, that doesnt change much |
| 18:34.56 | Spum | Oh, about 3 |
| 18:35.02 | IronHelix | how many lines? |
| 18:35.04 | Spum | Not that many, really |
| 18:35.04 | Spum | 2 |
| 18:35.19 | Spum | One for the PBX, one for normal phone calls? Or do they need just one |
| 18:35.37 | IronHelix | if you are using analog (pots) lines, you should have at least 2 |
| 18:35.47 | IronHelix | becuase each line can only make or recieve one phone call at at ime |
| 18:35.48 | Spum | Yeah, well that was what we were going to go for |
| 18:35.58 | Spum | really? |
| 18:36.05 | Spum | Oh, duh.. lol |
| 18:36.13 | IronHelix | hehe |
| 18:36.17 | Spum | Hmm, righty |
| 18:36.22 | IronHelix | you can also use a voip provider and get your lines through the internet |
| 18:36.30 | Spum | Oh, dip |
| 18:36.37 | Spum | Any in the UK? Or just a US thing |
| 18:36.44 | IronHelix | that is usually better because it makes having multiple concurrant calls easier, and is usually cheaper than going through a telco |
| 18:36.49 | IronHelix | no there are tons of those everywhere |
| 18:36.57 | Spum | One sec |
| 18:37.01 | IronHelix | voip providers are a dime a dozen |
| 18:37.04 | Spum | Allow me to google it :-) |
| 18:37.10 | IronHelix | neg |
| 18:37.11 | Spum | Or a penny :-P |
| 18:37.13 | IronHelix | www.voip-info.org |
| 18:37.32 | IronHelix | http://www.voip-info.org/wiki-VOIP+Service+Providers |
| 18:37.44 | IronHelix | extensive list of most known voip providers |
| 18:37.47 | IronHelix | sorted by country |
| 18:37.51 | IronHelix | you want the business section |
| 18:38.02 | Spum | well, i found one called "vonage" |
| 18:38.08 | IronHelix | dont use vonage |
| 18:38.14 | Spum | Hmm |
| 18:38.22 | IronHelix | make sure your providers support BYOD (bring your own device), that way instead of making you use their worthless adapter you can just plug the settings into asterisk |
| 18:38.28 | IronHelix | and whammo, you're in |
| 18:38.36 | IronHelix | vonage does not support byod |
| 18:38.49 | Spum | Ah, right |
| 18:38.58 | IronHelix | they make you use their ATA (analog telephony adapter), has ethernet on one end and a phone plug on the other |
| 18:39.04 | IronHelix | it counts as one 'line' |
| 18:39.05 | Spum | See, i'm confused to hell by all these acronyms |
| 18:39.12 | Spum | PTSN |
| 18:39.23 | IronHelix | public switched telephone network, aka the normal phone system |
| 18:39.34 | Spum | Oh right |
| 18:39.45 | IronHelix | whereas if you use another provider that supports byod, you are only limited by the number of current channels they let you use at once, usually 3-4 for buisness service |
| 18:40.01 | Spum | Okay, so anyway |
| 18:40.16 | IronHelix | first, hardware |
| 18:40.19 | Spum | What specification is desirable for Asterisk |
| 18:40.22 | trym | breaking news.. notepad replaces asterisk |
| 18:40.27 | IronHelix | for only 2 or 3 users you need barely anything |
| 18:40.31 | *** join/#asterisk zeedo (n=zeedo@80.68.92.188) |
| 18:40.35 | IronHelix | dig up an old pc out of somebodys basement |
| 18:40.39 | Spum | Nah |
| 18:40.46 | Spum | We're buying one , specifically |
| 18:40.46 | IronHelix | maybe 500mhz, 256mb ram |
| 18:40.53 | Spum | Because it'll house the Website, etc |
| 18:40.57 | IronHelix | ahhh |
| 18:41.13 | IronHelix | then get a decent sized linux server, asterisk will use almost none of it |
| 18:41.22 | Spum | Ahh, neat |
| 18:41.29 | Spum | I'll just build one |
| 18:41.31 | IronHelix | also, what kind of phones will you be using (explained...) |
| 18:41.36 | Spum | thenMake linux from scratch |
| 18:41.49 | Spum | May have been 2 years, but i still know how to do it |
| 18:41.50 | Spum | :-) |
| 18:41.58 | alephcom | ManxPower: The problem is, I will have to do codec translation. My idea was to still have asterisk do the codec translation. |
| 18:42.10 | IronHelix | you can use normal analog phones, and plug them into either ATAs (you can buy an ata and link it to your * server) or you can get a telphone board from DIgium (guys that make asterisk) |
| 18:42.30 | Spum | Hmm, some of these give you 0845 numbers, that's neat |
| 18:42.45 | IronHelix | for normal analog phones you will be paying about $30-$70us per port depending on how you do it |
| 18:42.48 | Spum | Right, anyways |
| 18:42.55 | Spum | £ ;-) |
| 18:43.00 | Spum | Okay, now |
| 18:43.09 | Spum | The office hours are 9am - 5pm |
| 18:43.20 | Spum | Any way you can make it route calls to the people who are in |
| 18:43.29 | pr0m | yes |
| 18:43.30 | IronHelix | yes absolutely |
| 18:43.34 | Spum | And then, if they go out, to make it do some form of answering |
| 18:43.37 | IronHelix | you can also use ip phones, these are telephones that have ethernet ports and connect directly to * (i recommend this), they are more expensive however |
| 18:43.38 | IronHelix | yup |
| 18:43.42 | Spum | Awesome |
| 18:43.46 | IronHelix | you can define complex behavior patterns |
| 18:43.56 | Spum | Fuck those PBX "experts" who said this would cost me £4000 |
| 18:44.06 | IronHelix | this will also be far more flexible |
| 18:44.17 | *** part/#asterisk drkbl (n=csd@86.109.65.180) |
| 18:44.19 | Spum | Hurray for Open Source |
| 18:44.27 | IronHelix | asterisk can do voicemail quite well, as well as voice mail to email for unified messaging |
| 18:44.34 | IronHelix | ie it emails you the voice mail as a wav file |
| 18:44.38 | IronHelix | and you listen to it on the computer |
| 18:44.40 | IronHelix | which is VERY cool |
| 18:45.05 | pr0m | Spum: designing a good dialplan may be more difficult than your think. |
| 18:45.08 | Spum | Holy shit |
| 18:45.20 | Spum | Meh, nah i enjoy coding |
| 18:45.29 | IronHelix | yeah, writing the dialplan is daunting at first |
| 18:45.34 | pr0m | ok |
| 18:45.41 | Spum | I've done C++ and other stuff for over 5 years |
| 18:45.45 | Spum | so it's cool :-) |
| 18:45.48 | IronHelix | dialplan btw is the file that tells asterisk what to do with things, ie when a call comes in it should do XXXX |
| 18:45.53 | Spum | Soon as i get the syntax, it's all over |
| 18:45.54 | Spum | :-D |
| 18:46.02 | Spum | Hmm oh yeah |
| 18:46.05 | IronHelix | then heres your 30 second non-ael introduction |
| 18:46.09 | IronHelix | [context] |
| 18:46.20 | IronHelix | exten => extension,priority,application(arugments) |
| 18:46.30 | Spum | you know when you phone up some places, and they have that "press 1 for, press 2 for" is that what you are talking about? |
| 18:46.34 | IronHelix | context is where the current call is, and it can call any extension in it and any included contexts |
| 18:46.36 | IronHelix | yuppers |
| 18:46.45 | IronHelix | priority is the order in which it does stuff, application is what it does |
| 18:46.59 | IronHelix | understand those three things and you are 95% golden |
| 18:47.13 | Spum | Righty, i noticed the voice extension thing |
| 18:47.27 | Spum | can you make it say funny sentences? ;-) |
| 18:47.33 | IronHelix | sure you can |
| 18:47.34 | pr0m | Spum: yes, but that's just the beginning. asterisk is very configurable... and so concomitantly may be as complex as you make it. |
| 18:47.40 | IronHelix | you can record anything you want it to say |
| 18:47.45 | IronHelix | and use text to speech for anythign else |
| 18:47.47 | Spum | That's sexy :-) |
| 18:48.02 | Spum | Oh yeah, does this have voice recognition? :-P |
| 18:48.08 | pr0m | Spum: or it could be very simple. but that's the rub. to know when enough is enough and make it logically follow for ALL users. |
| 18:48.09 | IronHelix | not easily |
| 18:48.12 | Spum | I worked at a place that had it |
| 18:48.17 | Spum | and english accents broke it |
| 18:48.19 | Spum | :-/ |
| 18:48.48 | IronHelix | hehe |
| 18:48.49 | pr0m | hmmm. i think asterisk will do text-to-speech but not the other way around. |
| 18:48.54 | Spum | ah right |
| 18:49.00 | IronHelix | yeah, it uses festival for tts |
| 18:49.04 | IronHelix | but no voice recognition |
| 18:49.06 | pr0m | of course, there's always room for a new application within asterisk. |
| 18:49.10 | |Vulture| | I hate those IVR systems that want you to talk to them |
| 18:49.11 | Spum | Ah, no problems :-) |
| 18:49.27 | Spum | I love those IVR things |
| 18:49.32 | |Vulture| | I think it is Bellsouth that has it now... awful |
| 18:49.45 | Spum | Say your name is "slobodan grozveny abadar nikolaizo" |
| 18:49.48 | IronHelix | lol |
| 18:49.52 | Spum | It'll blow up |
| 18:50.06 | |Vulture| | I always end up yelling at it until it says "Please hold while we transfer you to a rep" |
| 18:50.11 | IronHelix | lol |
| 18:50.12 | pr0m | or slob for short. |
| 18:50.13 | Spum | One malfunctioned on me |
| 18:50.22 | Spum | It started just saying "error" |
| 18:50.26 | IronHelix | lol |
| 18:50.26 | |Vulture| | lol |
| 18:50.27 | Spum | :-/ |
| 18:50.39 | |Vulture| | wait until it yells "Blue Screen of Death!!?!" |
| 18:50.40 | Spum | Press 1 for ERROR |
| 18:50.44 | |Vulture| | hahaa |
| 18:50.46 | IronHelix | lol |
| 18:50.46 | Spum | Press 2 for NOT SET |
| 18:50.48 | Spum | seriously |
| 18:50.51 | Spum | it said that |
| 18:50.53 | |Vulture| | thats pretty bad |
| 18:50.58 | Spum | and then i typed my name in on the keys |
| 18:50.58 | |Vulture| | did you press 1? |
| 18:51.07 | IronHelix | lol |
| 18:51.07 | Spum | and it said it in a REALLY high toned voice |
| 18:51.24 | Spum | It almost broke my ear drums :-/ |
| 18:51.25 | |Vulture| | Spum: whos system was that? |
| 18:51.35 | Spum | I dont know, some healthcare concern :-P |
| 18:51.49 | Spum | We value your custom and ERROR |
| 18:51.55 | |Vulture| | hahaha |
| 18:51.55 | IronHelix | hehehe |
| 18:52.00 | Spum | Oh yeah, the other one was |
| 18:52.07 | |Vulture| | ERROR ERROR DOES NOT COMPUTE |
| 18:52.12 | |Vulture| | danger will robbinson |
| 18:52.16 | Spum | The funniest one was, and this is absolutely true |
| 18:52.38 | Spum | We offer a wide range of medical solutions, if we havent got it, ERROR DICTIONARY ITEM NOT FOUND |
| 18:52.40 | Spum | :-P |
| 18:52.49 | IronHelix | lol |
| 18:52.54 | |Vulture| | wow.... |
| 18:52.55 | IronHelix | yeah badly configured pbx's are funny |
| 18:53.01 | Spum | Someone was probably playing football with the machine |
| 18:53.19 | Spum | I hope i dont get access to this thing |
| 18:53.26 | |Vulture| | yea it sounds like they had like 2 machines and the connection to the DB was toasted |
| 18:53.31 | Spum | I'll add in menu items like "Do you like your weight?" |
| 18:53.36 | Spum | "LOSE SOME WEIGHT PORKY" |
| 18:53.40 | IronHelix | hehe |
| 18:53.43 | |Vulture| | haha |
| 18:53.44 | Spum | :-P |
| 18:53.54 | Spum | "Did you enjoy our service?" |
| 18:53.59 | IronHelix | theres also a public directory of Allison Smith recordings |
| 18:54.00 | Spum | "No? What the fuck does that mean?" |
| 18:54.02 | |Vulture| | or make like an IVR with prices for medial services |
| 18:54.11 | IronHelix | someone paid $10 for her to record the lyrics to louie louie |
| 18:54.18 | |Vulture| | Boob Job - $2999.95 |
| 18:54.33 | IronHelix | (allison smith is a girl who records voice prompts for phone systems, most of the asterisk stuff uses her) |
| 18:54.35 | |Vulture| | Stomach Staple - |
| 18:54.56 | |Vulture| | yea and she does it pretty cheap too |
| 18:55.04 | Icemaann | drumkilla: you around? |
| 18:55.16 | Spum | Hehe ; "Are you a terrorist?" [presses 2 for no] "You have selected; YES. Federal marshalls are 5 seconds from your door. thank you for calling" |
| 18:55.41 | Spum | the funniest thing i heard, and this is also true |
| 18:55.42 | Spum | it said |
| 18:55.49 | IronHelix | theres a allison prompt for 'your telephone number has been flagged for retrieval by law enforcement officials' |
| 18:55.59 | Spum | Please select your language; "1 for spanish, 2 for german and 3 for english" |
| 18:56.05 | Spum | So, i pressed 31 by accident |
| 18:56.13 | Spum | and it said "you have selected ; vietnamese" |
| 18:56.16 | |Vulture| | wow... |
| 18:56.17 | IronHelix | lol |
| 18:56.20 | |Vulture| | HANGUP! |
| 18:56.22 | Spum | Hoo Hi Hee |
| 18:56.26 | Spum | Hoo Hoo Ho |
| 18:56.27 | pr0m | hey. i have a silly question... |
| 18:56.29 | Spum | :-P |
| 18:56.34 | Spum | Yes? |
| 18:56.36 | IronHelix | ? |
| 18:56.47 | IronHelix | theres no such thing as stupid questions, only stupid people |
| 18:56.49 | IronHelix | now ask away :) |
| 18:56.51 | pr0m | how many lines can a single telephone cord have? |
| 18:56.52 | |Vulture| | silly like make me laugh silly? |
| 18:56.57 | Spum | kill -HUP stupid-pbxs |
| 18:57.00 | Spum | :- |
| 18:57.06 | Spum | 8-) |
| 18:57.11 | |Vulture| | killall -HUP stupid-pbxs :P |
| 18:57.16 | Spum | :-O |
| 18:57.20 | IronHelix | standard phone cord generally can have two (four wires), although if it has all 6 positions wired it can have 3 |
| 18:57.20 | pr0m | there are four wires but only two needed for a single call. |
| 18:57.24 | Spum | /trumped |
| 18:57.33 | IronHelix | rj-45 connector can have four calls (8 wires) |
| 18:57.41 | pr0m | that's what i thought. |
| 18:57.49 | |Vulture| | thats why voip rox |
| 18:57.50 | Spum | And also, an R2-D2 is a shit robot |
| 18:57.52 | Spum | :-) |
| 18:58.00 | pr0m | do ya'll use zap for asterisk? |
| 18:58.02 | Spum | "What |
| 18:58.21 | IronHelix | however with rj45 its better to (if you can) use it as ethernet and send the calls over it on voip, or even tdmoe |
| 18:58.23 | |Vulture| | zap? |
| 18:58.27 | IronHelix | that will get you at least ten or twenty |
| 18:58.30 | |Vulture| | like as in phone lines? |
| 18:58.40 | IronHelix | zap, zaptel. its the driver used by digium interface boards |
| 18:58.43 | pr0m | ummm. zap driver is for digium analog cards. |
| 18:58.49 | pr0m | yes. |
| 18:58.55 | IronHelix | and yeah, zap is good, we <3 zap channels |
| 18:58.59 | Spum | "What's that R2?" .. "Bleep Bleep" ... "What? You find the policy on wookies very tenebrious and find such struggles trivial?" |
| 18:59.01 | |Vulture| | well zap is also used in sangoma |
| 18:59.24 | e-Hernick | rj-45 can have 125000 calls (8 wires, gigabit ethernet, gsm) |
| 18:59.25 | Spum | Hmm, do you guys have multi-language speech things |
| 18:59.34 | pr0m | so if i want to ring two separate lines on a single telephone cord (4 wire) how do i specify the second line? |
| 18:59.39 | |Vulture| | speaking of which I got a buncha FXO modules, a TDM card, and a TE110P for sale |
| 18:59.44 | IronHelix | pr0m- you need a 2 line phone |
| 18:59.53 | pr0m | like.... zap/1-1 and zap/1-2 ? |
| 18:59.58 | *** join/#asterisk rene- (n=rene-@201.144.60.6) |
| 19:00.02 | pr0m | yes... i figured that much. |
| 19:00.04 | Spum | Aaaaaanyways |
| 19:00.09 | Spum | Back to the topic |
| 19:00.13 | IronHelix | oh ic what you mean, i dunno if you can do that |
| 19:00.16 | rene- | hello, anyone here from astricon? |
| 19:00.19 | IronHelix | dont think you can run two lines through one zap port |
| 19:00.22 | pr0m | i just need to know how to specify in the extensions.conf file. |
| 19:00.23 | |Vulture| | pr0m: if those channels 1 and 2 are FXS ports |
| 19:00.25 | rene- | i meant the company |
| 19:00.27 | Spum | rene-, which planet is that near? |
| 19:00.36 | rene- | heh |
| 19:00.43 | rene- | ipsando |
| 19:00.45 | |Vulture| | pr0m: what do you want to do?/ |
| 19:00.55 | pr0m | two separate ports? that would be two separate physical cords then... not what i wanted to do. |
| 19:01.01 | |Vulture| | damn I was suppose to go to Cancun next month :( |
| 19:01.03 | Spum | Hmm, hey .. guys |
| 19:01.15 | IronHelix | you want to run a two line phone cord into a digium card and specify line 2 on the cord, right? |
| 19:01.16 | |Vulture| | pr0m: what are you trying to do? |
| 19:01.18 | Spum | Now, i've got this list of VOiP things |
| 19:01.29 | *** part/#asterisk pooh_ (n=hfwang@cust.15.241.adsl.cistron.nl) |
| 19:01.36 | IronHelix | to the best of my knowledge that cannot be done, because the digium fxo module only has one line interface. the other pins are ignored |
| 19:01.37 | Spum | Little confused as to what they all do |
| 19:01.38 | Spum | :-/ |
| 19:01.39 | pr0m | i want to assign different rings to different lines over the same single telephone cord from the same single port on my digium card. |
| 19:01.52 | rene- | did anyone here took dcap testing? |
| 19:01.59 | paxr0 | "Channel Status" works in dialplan context? i want to know channel status where Zap is calling |
| 19:02.00 | |Vulture| | pr0m: can't be done |
| 19:02.01 | rene- | at last astricon? |
| 19:02.03 | IronHelix | each digium port can only support one line |
| 19:02.09 | IronHelix | you need to split it into two ports |
| 19:02.22 | |Vulture| | 1 line per FXO port |
| 19:02.28 | IronHelix | you can buy a radio shack adapter that will split a 2 line plug into L1 / L2 / L1+2 |
| 19:02.37 | IronHelix | plug the l1 and l2 plugs into your digium ports |
| 19:02.38 | pr0m | but i'm using fxs on the digium card. |
| 19:02.55 | Spum | Holy crap |
| 19:02.57 | |Vulture| | pr0m: FXS goes from * to analog phones |
| 19:03.04 | pr0m | yes. that's right. |
| 19:03.05 | Spum | I have been sired by the offer of £0 phone calls |
| 19:03.11 | IronHelix | hehe |
| 19:03.17 | IronHelix | cool, isnt it? |
| 19:03.23 | pr0m | i have that port connected to the house wiring... then all phones in the house ring at the same time. |
| 19:03.28 | IronHelix | and spum whats on your list |
| 19:03.36 | |Vulture| | okay.. |
| 19:03.46 | Spum | the one you gave me |
| 19:03.49 | pr0m | but i'd like to pickup a couple 2-line phones so that both my wife and i can use them at the same time for separate calls. |
| 19:03.51 | IronHelix | ohh |
| 19:03.51 | IronHelix | right |
| 19:03.52 | Spum | you phone jokey, you |
| 19:03.55 | IronHelix | yeah thats a big list |
| 19:03.57 | Spum | ;-) |
| 19:04.04 | IronHelix | ahhh |
| 19:04.05 | IronHelix | ok |
| 19:04.08 | Spum | Some of these seem to be related to |
| 19:04.09 | IronHelix | pr0m |
| 19:04.19 | Spum | Phone calls via VOIP |
| 19:04.20 | IronHelix | question 1- prom are all the phone jacks in your house wired for two lines |
| 19:04.23 | |Vulture| | pr0m: oh okay so you want to combine 2 FXS ports into a single phone chord that goes to your house wiring? |
| 19:04.27 | pr0m | incoming calls are routed through voicepulse to * then to the phone jack in the wall. |
| 19:04.41 | IronHelix | question 1- prom are all the phone jacks in your house wired for two lines |
| 19:04.43 | IronHelix | if so |
| 19:04.48 | IronHelix | then buy TWO fxs modules |
| 19:04.57 | Spum | Wow, that IPclouds offer is facking sexy |
| 19:04.58 | pr0m | |Vulture|: no that wouldn't accomplish anything if i understand correctly. |
| 19:05.00 | IronHelix | get the radio shack L1/L2/both splitter |
| 19:05.14 | |Vulture| | Im lost I don't really understand what hes trying to do |
| 19:05.14 | IronHelix | plug your two fxs modules into l1 and l2 |
| 19:05.18 | IronHelix | i understand him |
| 19:05.31 | *** join/#asterisk TUplink (n=Tommy@68.232.92.81) |
| 19:05.33 | Spum | Anyways |
| 19:05.36 | IronHelix | he just wants to use a two line phone to get two zap channels so he can use his house loop separately from his wife |
| 19:05.36 | pr0m | IronHelix: no... not yet. but i'm interested in purchasing some phones with multiple line capability. |
| 19:05.44 | TUplink | my FWD stoped working any ideas? |
| 19:05.46 | IronHelix | not the phones |
| 19:05.49 | IronHelix | the jacks |
| 19:05.54 | Spum | So, tell me about these VoIP thingy things |
| 19:05.54 | IronHelix | not all houses are wired for two lines |
| 19:05.58 | TUplink | fwd for 1800 |
| 19:06.02 | *** join/#asterisk [Airwolf] (n=airwolf@airwolf.xs4all.nl) |
| 19:06.25 | pr0m | IronHelix: yes. i think you understand. |
| 19:06.30 | IronHelix | spum- you want a provider that supports byod, so you can just punch the sip settings into your asterisk server that is connected to the itnernet |
| 19:06.41 | IronHelix | that will trhen get you phone service via the internet, no telco required |
| 19:06.42 | IronHelix | prom |
| 19:06.51 | pr0m | IronHelix: my house has 4 wires throughout. |
| 19:06.53 | IronHelix | first- make sure that all the jacks in your house are wired for two line |
| 19:06.54 | IronHelix | ok good |
| 19:06.56 | IronHelix | excellent |
| 19:07.01 | IronHelix | now do this |
| 19:07.10 | IronHelix | 1. make sure you have two fxs modules in your * box |
| 19:07.14 | Spum | hmm, actually |
| 19:07.20 | obsidian-studios | pr0m: red green, one line, yellow black, second line |
| 19:07.37 | obsidian-studios | wire them to RJ 11 connector plug into 2 fxs ports |
| 19:08.03 | Spum | Wow, this Skype sounds good |
| 19:08.08 | Spum | Works on Linux, right? |
| 19:08.11 | IronHelix | no skype isnt good |
| 19:08.13 | IronHelix | doesnt work with * |
| 19:08.17 | IronHelix | proprietary protocols |
| 19:08.27 | pr0m | ok. so the fxs ports on the digium card will only handle one line each.. correct? |
| 19:08.29 | *** join/#asterisk pa (n=Paolo@unaffiliated/pa) |
| 19:08.34 | Spum | Hm, but for free calls? |
| 19:08.35 | IronHelix | pr0m- 2. buy a radio shack phone line splitter. it will have one port on one end and 3 ports on the other, L1 L2 and L1/L2 |
| 19:08.38 | IronHelix | pr0m- yes |
| 19:08.44 | pr0m | even though there are 4 wires in each port. |
| 19:08.49 | IronHelix | correct |
| 19:08.51 | Spum | Holy cack, there is a linux port! |
| 19:08.52 | Spum | Hurray |
| 19:09.06 | obsidian-studios | pr0m: to my knowledge yes |
| 19:09.07 | pr0m | ok. damn. well. guess that narrows down my options. |
| 19:09.16 | IronHelix | plug the splitter into your wall. use phone cords to connect the L1 and L2 plugs to the two fxs ports on your * box |
| 19:09.34 | IronHelix | so one fxs port will drive Line 1 |
| 19:09.40 | IronHelix | and the other fxs port will drive Line 2 |
| 19:09.55 | IronHelix | does this all make sense? |
| 19:09.58 | pr0m | i think i see where you're going with this... i was just under the wrong assumption about how many lines each fxs port could do. |
| 19:10.04 | Spum | Righty boys |
| 19:10.13 | pr0m | yes. i understand. just wish it wasn't so. ;-) |
| 19:10.14 | Spum | I will see you tommorow or after that |
| 19:10.17 | IronHelix | spum- you dont want to use skype. its a closed system with a proprietary protocol. check out http://www.fwd.pulver.com instead |
| 19:10.21 | IronHelix | hehe |
| 19:10.26 | IronHelix | cya spum |
| 19:10.27 | Spum | Have to go entertain the kids and stuff :-) |
| 19:10.37 | IronHelix | pr0m- yeah, you gotta buy another fxs module, but other than that its all good |
| 19:10.44 | Spum | Ciao |
| 19:10.50 | pr0m | ok. thanks for your help. |
| 19:10.54 | IronHelix | no worries |
| 19:11.02 | IronHelix | alternatively |
| 19:11.29 | IronHelix | consider wiring your house for ethernet, and setting up IP phones connected to an 802.3af powered switch |
| 19:11.32 | Spum | Isn't Skype supposed to be on a "better" protocol? |
| 19:11.42 | IronHelix | what better way to spend $1000? :D |
| 19:11.56 | Spum | Porno? |
| 19:11.58 | Spum | Err, no |
| 19:12.01 | pr0m | IronHelix: yes. i've thought of that. it was my first option.... but more expensive than i had wanted. |
| 19:12.13 | IronHelix | spum- no not really. here in the voip world, open is good. skype is a closed system, only skype clients can interface with skype |
| 19:12.22 | obsidian-studios | does make sense though if a regular phone can do 2 lines per 4 wires, why can't a digium module? Seems they might want to consider adding that support so a 4 port TDM card could handle 8 channels instead of 4 |
| 19:12.29 | Spum | Really? Those Kazaa bastards |
| 19:12.40 | IronHelix | whereas SIP (the more common more open protocol) is an open spec- anybody that wants to write a sip stack easily can, and asterisk did, so did a shitload of other people |
| 19:12.42 | pr0m | obsidian-studios: agreed. |
| 19:12.56 | IronHelix | yeah that would be cool |
| 19:13.10 | IronHelix | i dont think its possible tho with the current generation of cards |
| 19:13.21 | IronHelix | the side of the module that connects to the phone port only has three pins |
| 19:13.21 | Spum | Alright boys |
| 19:13.26 | Spum | see yer's later :-) |
| 19:13.29 | IronHelix | cya spum |
| 19:13.44 | pr0m | the problem with most sip phones is that you can't "share" lines between phones. that is you can't pickup and talk with the same people already on another active line. |
| 19:13.45 | obsidian-studios | IronHelix: no clue but yes they only support 1 now, 2 would be cool |
| 19:14.13 | IronHelix | yeah it would be awesome, maybe for the next generation tdmxx card |
| 19:14.33 | IronHelix | in theory with rj45 connectors you could have four lines per port |
| 19:15.12 | IronHelix | or maybe a pro version with an amphenol connector on it |
| 19:15.17 | IronHelix | that would be sw33t |
| 19:15.37 | pr0m | i thought that i'd seen phones with 4 lines and only a single rj11 port. am i dreaming? |
| 19:15.52 | IronHelix | no, but those are for a key system |
| 19:15.56 | pr0m | or are they on some sort of "digital" proprietary system. |
| 19:15.58 | IronHelix | they require a base station |
| 19:15.59 | Aughey | they usually have a pbx down in the basement somewhere |
| 19:16.00 | IronHelix | yeah exactly |
| 19:16.06 | pr0m | ah. ok. |
| 19:16.22 | IronHelix | without a connection to the 'brain' the phone will become frightened and confused |
| 19:16.22 | pr0m | huh. |
| 19:16.27 | pr0m | hehe. |
| 19:16.56 | Aughey | kind of like people who aren't connected to this irc "brain" |
| 19:17.11 | IronHelix | hehehe |
| 19:17.23 | Aughey | though I think even people her are frightened and confused |
| 19:18.53 | IronHelix | there are lots of frightened and confused people |
| 19:18.57 | IronHelix | most of them are Skype users |
| 19:19.05 | IronHelix | and have AOL accounts |
| 19:23.10 | *** join/#asterisk brc_ (n=Brian@pdpc/supporter/basic/brc) |
| 19:24.56 | *** join/#asterisk damned (n=vpol@damned.vpol.org.ru) |
| 19:26.34 | rene- | routing is hard.... i used to link networks via VPN, when the linux endpoints were also serving as the default gw for hosts in its subnet things worked fine, in my current scenario, it isnt working, i have tried adding static routes to the remote vpn network to clients and to the default gw in the subnet, it isnt working, id appreciate any help |
| 19:28.11 | rene- | i forgot to mention that in my current setting, there is a default gw and there is the linux vpn gateway |
| 19:29.07 | cryzeck | wrong channel? |
| 19:29.32 | shyru | hi dudes!!! |
| 19:40.49 | *** join/#asterisk s3g (n=s3g@85.105.100.100) |
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| 19:47.00 | TUplink | why isnt my FWD iax woring i call a 1800 number with asterisk and Called ******:***********@iax2.fwdnet.net/*18009993355 |
| 19:47.05 | TUplink | but i never picks up |
| 19:47.44 | *** join/#asterisk MGSsancho (n=user@ppp-67-126-240-180.dsl.irvnca.pacbell.net) |
| 19:47.45 | TUplink | any ideas? |
| 19:47.56 | TUplink | but FWD out works |
| 19:48.16 | TUplink | and is use to work |
| 19:48.26 | *** join/#asterisk st3v (n=st3v@netblock-66-218-41-231.dslextreme.com) |
| 19:49.46 | *** join/#asterisk drkbl (n=csd@86.109.65.180) |
| 19:51.42 | st3v | I have my asterisk server running, and I have my extensions connected to a channel bank. I also have a TDM04P with 4 POTS lines connected to it. I am getting dialtones on the extensions, but I need to configure asterisk. Are there any good tutorials on how to configure the extensions, telephone lines, voicemail, etc? I want the extensions to use whichever one of the 4 phones are not in use. |
| 19:54.49 | *** join/#asterisk stkn_ (i=nobody@gentoo/developer/pdpc.active.stkn) |
| 19:56.59 | *** join/#asterisk SuPrSluG (n=SuPrSluG@pool-141-149-242-183.buff.east.verizon.net) |
| 19:57.06 | SuPrSluG | hi all |
| 19:59.13 | SuPrSluG | anyone else having problem w/ broadvoice incoming numbers? |
| 19:59.19 | Inv_arp | ~docs |
| 19:59.21 | jbot | somebody said docs was Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk |
| 20:00.34 | SuPrSluG | no it's been working for some time. but now it's not. i can call out thru them. just wondering if it's bvoice |
| 20:00.57 | *** join/#asterisk zeedo (n=zeedo@80.68.92.188) |
| 20:05.18 | *** join/#asterisk santiago (n=santiago@208.195.215.158) |
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| 20:14.30 | *** join/#asterisk SERGEUS (i=sergey@195.112.98.13) |
| 20:16.20 | SERGEUS | file[laptop]: i've heard, that you using ExternalIVR(), can you help me? |
| 20:16.21 | TUplink | anyone have a fwd number that i can try to call? |
| 20:16.55 | file | eh? I don't even touch that |
| 20:17.07 | rene- | Anyone here from IPSando? |
| 20:17.35 | drumkilla | file: I heard that you have 3 eyes |
| 20:17.41 | SERGEUS | file: hmmm :) strange http://www.asteriskgeeks.com/logs/irclogger_log/asterisk?date=2005-08-17,Wed&text=on |
| 20:18.18 | SERGEUS | oops |
| 20:18.39 | SERGEUS | that was someone else :) |
| 20:20.05 | SERGEUS | it was wwalker |
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| 20:30.10 | brookshire | z0mg hi! |
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| 20:39.40 | lesouvage | . |
| 20:41.28 | r0d3nt|m | So let me get this straight .... Digium cards cannot share IRQ's @ All ??? |
| 20:41.52 | drumkilla | no, that's not true ... |
| 20:41.52 | r0d3nt|m | they are not compliant with PCI irq sharing ??? |
| 20:41.58 | r0d3nt|m | ok... |
| 20:42.00 | *** join/#asterisk pa (n=Paolo@unaffiliated/pa) |
| 20:42.03 | r0d3nt|m | so whats the deal... |
| 20:42.09 | r0d3nt|m | please explain and forgive my stupidness. |
| 20:42.19 | r0d3nt|m | I've been getting lots of conflicting information. |
| 20:43.05 | drumkilla | well, these problems are generally on crappy boards |
| 20:43.18 | r0d3nt|m | like ? |
| 20:43.29 | r0d3nt|m | I only use digiums cards. |
| 20:43.31 | r0d3nt|m | are they crappy ? |
| 20:43.43 | drumkilla | but if there is any reason why interrupts can't be serviced quickly, audio problems can occur |
| 20:43.50 | r0d3nt|m | I have a client complaining of dropped calls... |
| 20:43.53 | X-Rob | r0d3nt|m - the issue is, when you're sharing an interrupt, when the interrupt is triggered, everything that's on that interrupt has to go and look to see what's happened |
| 20:44.01 | r0d3nt|m | digium told me it's cuz I'm sharing an irq with the card. |
| 20:44.03 | X-Rob | this is a problem when you're caring about audio. |
| 20:44.16 | r0d3nt|m | i have no audio problems. |
| 20:44.17 | r0d3nt|m | NONE. |
| 20:44.29 | drumkilla | it may fix your problem, but it may not |
| 20:44.30 | r0d3nt|m | Is there anyway to see why a call is dropped ?? |
| 20:44.42 | r0d3nt|m | the telco says it's the switch. |
| 20:44.52 | drumkilla | PRI? |
| 20:44.55 | X-Rob | what sort of card and connection? |
| 20:44.56 | r0d3nt|m | yes |
| 20:45.11 | r0d3nt|m | latest single span T1/E1/PRI digium card |
| 20:45.12 | drumkilla | get a pri debug and see who's dropping it |
| 20:45.26 | r0d3nt|m | debug the specific channel ? |
| 20:45.33 | X-Rob | 'pri debug span 1' |
| 20:45.34 | drumkilla | well, the span |
| 20:45.35 | r0d3nt|m | ok |
| 20:45.48 | r0d3nt|m | dwgvoip*CLI> pri debug span 1 |
| 20:45.48 | r0d3nt|m | Enabled debugging on span 1 |
| 20:45.49 | r0d3nt|m | thanks |
| 20:45.59 | r0d3nt|m | any other reasons why it would be dropping calls ?? |
| 20:45.59 | X-Rob | r0d3nt|m - usually it's because your timing source is wrong. |
| 20:46.06 | r0d3nt|m | ok |
| 20:46.12 | r0d3nt|m | for remote network time sync |
| 20:46.16 | r0d3nt|m | what is the setting ? |
| 20:46.17 | X-Rob | check to make sure the second field in your zapata.conf is 1 |
| 20:46.18 | r0d3nt|m | 0,1,1 ? |
| 20:46.20 | X-Rob | not 0 |
| 20:46.22 | r0d3nt|m | yes |
| 20:46.23 | r0d3nt|m | it is |
| 20:46.23 | X-Rob | that's correct |
| 20:46.25 | drumkilla | zaptel.conf :) |
| 20:46.31 | X-Rob | uh yeah. wups. |
| 20:46.39 | X-Rob | damn similar names 8) |
| 20:46.51 | r0d3nt|m | i'm using 1,1,2 |
| 20:47.23 | r0d3nt|m | when i use zttool it says it's internal tho' |
| 20:47.25 | r0d3nt|m | is that correct ? |
| 20:47.32 | X-Rob | zttool is incorrect |
| 20:47.38 | r0d3nt|m | internally clocked |
| 20:47.39 | r0d3nt|m | ok |
| 20:47.49 | r0d3nt|m | so it is incorrect ? |
| 20:47.51 | r0d3nt|m | ok.. |
| 20:48.36 | r0d3nt|m | ok so x,1,x is external time sync |
| 20:48.41 | r0d3nt|m | network time source |
| 20:50.03 | r0d3nt|m | # To not use this as a sync source, just use "0" |
| 20:50.13 | r0d3nt|m | that is confusing.. or maybe I'm just a dolt. |
| 20:50.27 | r0d3nt|m | span=1,1,2,esf,b8zs |
| 20:50.28 | X-Rob | no, that's not confusing |
| 20:50.31 | r0d3nt|m | is how it's setup |
| 20:50.34 | r0d3nt|m | is that correct ? |
| 20:50.36 | X-Rob | that looks fine to me. |
| 20:50.45 | X-Rob | drumkilla - any other suggestions? |
| 20:50.49 | r0d3nt|m | we are a considerable distance from the NIU. |
| 20:51.00 | r0d3nt|m | which is why I set it to 2. for distance... |
| 20:52.37 | drumkilla | yes, it is very confusing |
| 20:52.42 | r0d3nt|m | is there anything else i can look @ for dropped calls ?? |
| 20:52.49 | drumkilla | basically, if you want to *provide* timing, you put 0 |
| 20:53.01 | r0d3nt|m | which we don't.... |
| 20:53.03 | r0d3nt|m | right ?? |
| 20:53.06 | drumkilla | otherwise, you put numbers that indicate your preference for a span to *receive* timing from |
| 20:53.26 | r0d3nt|m | if you were using your 1st pri as a sync source for your others ?? |
| 20:53.29 | drumkilla | for a PRI to the PSTN, you just about always receive timing |
| 20:53.44 | r0d3nt|m | definitely |
| 20:53.55 | r0d3nt|m | their clock source by far is more accurate. |
| 20:53.58 | drumkilla | anyway, try to free up the IRQ thing |
| 20:54.01 | drumkilla | might as well |
| 20:54.04 | r0d3nt|m | i tried... |
| 20:54.08 | drumkilla | can you switch slots? |
| 20:54.12 | r0d3nt|m | dell 2850 with dual nics.... |
| 20:54.17 | drumkilla | ah. |
| 20:54.26 | r0d3nt|m | always 1 nick shares a irq with any card on the pci bus. |
| 20:54.29 | r0d3nt|m | nic* |
| 20:54.41 | drumkilla | using both of them? :) |
| 20:54.46 | r0d3nt|m | yes. |
| 20:54.50 | drumkilla | cool |
| 20:54.53 | r0d3nt|m | 1 public, 1 private |
| 20:55.04 | r0d3nt|m | only the phones are on that private network segment. |
| 20:55.13 | drumkilla | another thing, do you have callprogress on? |
| 20:55.22 | drumkilla | wait, that's for analog, isn't it |
| 20:55.23 | r0d3nt|m | basically it does nat so the phones can see the intarweb. |
| 20:55.30 | r0d3nt|m | no |
| 20:55.35 | r0d3nt|m | call progress is off. |
| 20:55.37 | r0d3nt|m | and i'm not sure |
| 20:55.41 | *** join/#asterisk brookshire (n=matt@216.215.208.188.nw.nuvox.net) |
| 20:55.44 | drumkilla | well. i'd want to see the pri debug |
| 20:55.46 | r0d3nt|m | it says it's experimental or something.... |
| 20:55.53 | r0d3nt|m | drumkilla, i'll be glad to show you sir..... |
| 20:56.28 | r0d3nt|m | ; This feature is HIGHLY EXPERIMENTAL and can easily detect false answers, |
| 20:56.28 | r0d3nt|m | ; so don't count on it being very accurate. |
| 20:56.33 | r0d3nt|m | callprogress=no |
| 20:56.40 | drumkilla | cool |
| 20:56.43 | r0d3nt|m | ya. |
| 20:56.52 | r0d3nt|m | and don't ever run cvs head of zaptel |
| 20:56.55 | r0d3nt|m | no matter what chris from digium tells you |
| 20:57.01 | r0d3nt|m | we had 200% more dropped calls with that POS code. |
| 20:57.05 | r0d3nt|m | no offense. |
| 20:57.16 | drumkilla | that's odd |
| 20:57.19 | r0d3nt|m | ya |
| 20:57.22 | r0d3nt|m | tell me about it.... |
| 20:57.39 | drumkilla | well, like i said, forward them your PRI debug and output from the Asterisk CLI |
| 20:57.43 | drumkilla | i've got to run for now ... |
| 20:58.02 | r0d3nt|m | not only did chris tell telepacific that digium has problems with echo on PRi's with it's hardware, he told me to try and CVS head of zaptel and that it might fix the problem... |
| 20:58.08 | r0d3nt|m | ok |
| 20:58.11 | r0d3nt|m | thank you sir... |
| 20:58.38 | r0d3nt|m | <PROTECTED> |
| 20:58.40 | r0d3nt|m | these normal ??? |
| 20:58.45 | r0d3nt|m | i see these all the freak'n time.... |
| 20:59.34 | trym | well.. it says successfully dont it ? ;) |
| 20:59.42 | r0d3nt|m | ya... |
| 20:59.50 | r0d3nt|m | but why the restart ? couple times an hour... |
| 20:59.54 | trym | then it cant be the worst thing in the world |
| 20:59.59 | trym | i wish I had the answer |
| 21:00.15 | Qwell | If it isn't preceeded by "ERROR:", its probably okay |
| 21:00.23 | r0d3nt|m | to all of us, being my techs and my experience with this stuff, the PBX runs perfectly.... |
| 21:00.36 | r0d3nt|m | however the client complains of echo and dropped calls. |
| 21:00.47 | r0d3nt|m | echo calls are reproducable with certain phone numbers |
| 21:00.58 | r0d3nt|m | which would be a symptom of the telco related echo |
| 21:01.05 | r0d3nt|m | the telco says it's the VoIP PBX |
| 21:01.27 | r0d3nt|m | even had some MIT asshole from CarrierAccess analyze a packet dump of a echo conversation |
| 21:01.44 | trym | lol |
| 21:01.45 | r0d3nt|m | and it was his summary that asterisk is creating the echo |
| 21:01.54 | r0d3nt|m | but for some reason, they didn't share the data they analyzed... |
| 21:02.15 | r0d3nt|m | they reconstructed both 1/2's of the conversation and listen to each 1/2 |
| 21:02.23 | trym | i love the fact that someone who attends MIT is automatically an asshole |
| 21:02.28 | r0d3nt|m | said the echo is coming from the asterisk side |
| 21:02.35 | r0d3nt|m | trym, no... |
| 21:02.42 | X-Rob | r0d3nt|m - those restarts are perfectly normal. |
| 21:02.57 | r0d3nt|m | it was his attitude and blatent disregard for my evidence and rebuttle to his analysis.... |
| 21:03.08 | trym | ah I see |
| 21:03.12 | trym | well many MIT people are assholes |
| 21:03.20 | r0d3nt|m | then completely ignored my last 4 quests to compare the data... |
| 21:03.22 | X-Rob | r0d3nt|m - are you using the KB1 echo canceller? |
| 21:03.45 | r0d3nt|m | X-Rob, honestly.. i have no idea... whatever is enable by default... |
| 21:03.52 | *** join/#asterisk pa (n=Paolo@unaffiliated/pa) |
| 21:03.53 | r0d3nt|m | we didnt' make any changes to the default compilation of asterisk |
| 21:03.59 | r0d3nt|m | we set the confs to use echo cans |
| 21:04.07 | X-Rob | so, are you at least using CVS? |
| 21:04.12 | r0d3nt|m | NO. |
| 21:04.16 | X-Rob | Fix that. |
| 21:04.19 | r0d3nt|m | we tried CVS head and it was 200% WORSE. |
| 21:04.25 | X-Rob | That's incorrect. |
| 21:04.41 | X-Rob | If you're not using CVS you will have severe echo problems. |
| 21:04.42 | Qwell | it was 3 times as bad with HEAD? |
| 21:05.04 | X-Rob | you will still have echo problems with CVS, but it won't be anywhere near as bad. |
| 21:05.06 | r0d3nt|m | X-Rob, we only have echo on certain phone #'s |
| 21:05.09 | r0d3nt|m | it's not within asterisk. |
| 21:05.23 | X-Rob | r0d3nt|m - it's the way echo works. |
| 21:05.25 | X-Rob | trust me on this |
| 21:05.28 | X-Rob | I live in rural australia |
| 21:05.38 | X-Rob | we have _awful_ lines out here. I'm the fucking GOD on echo. |
| 21:05.43 | r0d3nt|m | we can reproduce echo on certain phone numbers... we never hear echo internally or terminated elsewhere.... |
| 21:05.51 | r0d3nt|m | echocancel=128 |
| 21:05.56 | r0d3nt|m | echocancelwhenbridged=yes |
| 21:06.05 | r0d3nt|m | echotraining=yes |
| 21:06.27 | r0d3nt|m | jitterbuffers=4 |
| 21:06.42 | r0d3nt|m | switchtype = national |
| 21:06.42 | r0d3nt|m | signalling = pri_cpe |
| 21:06.42 | r0d3nt|m | group = 1 |
| 21:06.42 | r0d3nt|m | context = default |
| 21:06.42 | r0d3nt|m | channel => 1-23 |
| 21:06.45 | X-Rob | r0d3nt|m - uh. You're not downloading CVS. |
| 21:06.50 | X-Rob | Why are you not downloading CVS? |
| 21:07.04 | r0d3nt|m | chris from digium suggested we try CVS head of zaptel |
| 21:07.05 | X-Rob | you're dumping stuff here when it should be |
| 21:07.08 | X-Rob | ~pb |
| 21:07.10 | jbot | rumour has it, pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca/ |
| 21:07.25 | r0d3nt|m | you call that a flood ? |
| 21:07.26 | X-Rob | r0d3nt|m - here's your clue. I'm giving it to you again. You paying attention? |
| 21:07.27 | X-Rob | USE CVS. |
| 21:07.33 | X-Rob | You get it this time? |
| 21:07.38 | r0d3nt|m | ok.. i'm not an idiot. |
| 21:07.45 | Qwell | famous last words |
| 21:07.46 | r0d3nt|m | please don't talk to me as such. |
| 21:08.01 | Qwell | That, and "trust me, I know what I'm doing" |
| 21:08.06 | X-Rob | Qwell - heh. |
| 21:08.09 | r0d3nt|m | No, i didn't say that.... |
| 21:09.05 | r0d3nt|m | X-Rob, perhaps you arent listening.. Chris from digium suggested CVS head.... |
| 21:09.07 | r0d3nt|m | we tried it.. |
| 21:09.12 | r0d3nt|m | the client reported 200% more echo |
| 21:09.15 | r0d3nt|m | excuse me |
| 21:09.17 | r0d3nt|m | dropped calls. |
| 21:09.19 | X-Rob | that means you ddin't set it up right. |
| 21:09.20 | Qwell | r0d3nt|m: Can I give you a tip? Well, I'm going to anyways. When somebody is trying to help you, you do as they ask, or they won't help |
| 21:09.46 | r0d3nt|m | Qwell, i've tried all the suggestions so far. |
| 21:09.51 | Qwell | With HEAD? |
| 21:09.54 | r0d3nt|m | with results that were worse. |
| 21:10.06 | X-Rob | r0d3nt|m - you didn't set it up right. |
| 21:10.08 | Qwell | Getting HEAD was only step one of X-Rob's plan... |
| 21:10.14 | Qwell | you need to get HEAD before you can proceed to step 3 |
| 21:10.15 | r0d3nt|m | X-Rob, which did i not setup correctly ?? |
| 21:10.21 | r0d3nt|m | i used the same settings for the PRI |
| 21:10.22 | Qwell | (step 2 doesn't exist it seems) |
| 21:10.29 | X-Rob | Step 3 is me saying 'here are all the bits that minimise echo' |
| 21:11.56 | r0d3nt|m | X-Rob, so how am i not setting it up correctly ?? |
| 21:12.07 | r0d3nt|m | besides not using CVS head. |
| 21:12.29 | X-Rob | r0d3nt|m - you have to use the KB1 echo can. |
| 21:12.33 | X-Rob | which is only in CVS HEAD. |
| 21:12.45 | r0d3nt|m | enabled by default ? when that is compiled and installed ? |
| 21:12.49 | X-Rob | KB1 echo can actually works. |
| 21:13.06 | X-Rob | Depends when you tried CVS, it's only recently become the default. |
| 21:13.12 | r0d3nt|m | how recent ? |
| 21:13.14 | r0d3nt|m | weeks ? |
| 21:13.18 | r0d3nt|m | or days ? |
| 21:13.20 | X-Rob | weeks. |
| 21:13.27 | r0d3nt|m | i probably tried it. |
| 21:13.29 | r0d3nt|m | i'll try it again. |
| 21:13.49 | X-Rob | put your /etc/asterisk/zapata.conf on pastebin |
| 21:13.50 | X-Rob | ~pb |
| 21:13.51 | jbot | extra, extra, read all about it, pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca/ |
| 21:14.19 | r0d3nt|m | X-Rob, i'd rather not be spoken down to and mocked. thanks for the help. |
| 21:14.28 | X-Rob | r0d3nt|m - well, next time try listening. |
| 21:14.32 | r0d3nt|m | i've been |
| 21:14.38 | r0d3nt|m | which part did I miss ? |
| 21:14.41 | X-Rob | rather than ignoring what people say. Ignoring people implies ignorance. |
| 21:14.45 | *** join/#asterisk hanchi (n=telliott@015-834-723.area5.spcsdns.net) |
| 21:14.49 | r0d3nt|m | which did i ignore ? |
| 21:15.07 | r0d3nt|m | perhaps you missed the 4 times i said I tried CVS head and cvs of zaptel. |
| 21:15.17 | r0d3nt|m | in response to your magical information. |
| 21:15.22 | X-Rob | put your /etc/asterisk/zapata.conf on pastebin |
| 21:15.24 | X-Rob | ~pb |
| 21:15.26 | jbot | from memory, pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca/ |
| 21:15.29 | X-Rob | (see how you're ignoring me again?) |
| 21:15.57 | hanchi | trying to install mpg123 for * on dual xeon 2.8 gig machine with fc3, can't get make linux to work, there is no onboard audio on this server, help |
| 21:16.20 | X-Rob | hanchi - that shouldn't be relevant. |
| 21:16.29 | X-Rob | what erorr are you getting? Put it on pastebin. |
| 21:16.30 | X-Rob | ~pb |
| 21:16.31 | jbot | well, pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca/ |
| 21:19.25 | r0d3nt|m | http://pastebin.ca/26318 |
| 21:21.36 | r0d3nt|m | X-Rob, any more thoughts ? |
| 21:22.32 | X-Rob | one significant problem - you've got busydetect set to yes. I'm just tidying it up now. |
| 21:23.00 | Qwell | s/busydetect/randomlyhangupcalls/ |
| 21:23.32 | Qwell | r0d3nt|m: before X-Rob asks...are you running HEAD now? |
| 21:23.38 | r0d3nt|m | NO. |
| 21:23.39 | r0d3nt|m | ; On trunk interfaces (FXS) and E&M interfaces (E&M, Wink, Feature Group D |
| 21:23.39 | r0d3nt|m | ; etc, it can be useful to perform busy detection either in an effort to |
| 21:23.39 | r0d3nt|m | ; detect hangup or for detecting busies |
| 21:24.30 | r0d3nt|m | so wouldn't that imply that setting only effects FXS and type interfaces ?? |
| 21:24.34 | r0d3nt|m | and not PRI's and T1's ? |
| 21:25.00 | X-Rob | no |
| 21:25.10 | X-Rob | it means it turns it on for that channel |
| 21:25.19 | r0d3nt|m | a FXs channel ? |
| 21:25.25 | X-Rob | for that channel |
| 21:25.28 | r0d3nt|m | of which i have none. |
| 21:25.29 | X-Rob | whatever that channel may be. |
| 21:26.12 | X-Rob | you're turning it on for whatever you specify in the trailing chan=> group. |
| 21:26.29 | X-Rob | ok. Next problem. You have your TXGAIN turned down to -9?? |
| 21:26.36 | r0d3nt|m | per the telco |
| 21:26.49 | X-Rob | Hurm. |
| 21:26.51 | r0d3nt|m | Telepacific's magical gain settings |
| 21:26.55 | r0d3nt|m | to fix the problem |
| 21:26.59 | r0d3nt|m | which made no different. |
| 21:27.05 | X-Rob | yeah. Don't do that. |
| 21:27.15 | X-Rob | Leave the gain as close to 0 as you can |
| 21:27.23 | X-Rob | (again, this oonly works with KB1) |
| 21:27.28 | r0d3nt|m | alright |
| 21:27.33 | r0d3nt|m | i'll set to -3 and -3 |
| 21:27.51 | X-Rob | http://pastebin.ca/26320 |
| 21:27.57 | X-Rob | leave it at zero zero. |
| 21:28.06 | X-Rob | -3 is halving the signal strength |
| 21:28.16 | X-Rob | if you use that conf with cvs head, that'll do the best it can. |
| 21:28.47 | X-Rob | if you still have echo, you can drop txgain down, but after 1.5 it usually starts to get worse again |
| 21:28.47 | r0d3nt|m | anything else changed ? |
| 21:28.57 | X-Rob | yep. |
| 21:28.57 | r0d3nt|m | -1.5 ? |
| 21:29.09 | X-Rob | after -1.5, yes, sorry. |
| 21:29.35 | X-Rob | I commented the bits I changed. |
| 21:29.49 | r0d3nt|m | in what incriments should i adjust the gain ? |
| 21:29.51 | r0d3nt|m | .1 ? |
| 21:29.57 | r0d3nt|m | if it's adjusted ? |
| 21:30.03 | X-Rob | I usually go .5, and if that doesn't work, 1, and then 1.5 |
| 21:30.11 | X-Rob | it's not rocket science. |
| 21:30.16 | r0d3nt|m | <PROTECTED> |
| 21:30.18 | r0d3nt|m | no shit. |
| 21:30.25 | X-Rob | You _will_ still have echo problems |
| 21:30.31 | X-Rob | but they're now fixable |
| 21:30.41 | X-Rob | turn down people's microphones on their handsets. |
| 21:30.55 | r0d3nt|m | i have. |
| 21:31.01 | r0d3nt|m | well the speaker |
| 21:31.05 | X-Rob | then cvs head with that will probably help. |
| 21:31.14 | r0d3nt|m | cisco 7940's don't have a microphone adjustment if i'm not mistaken. |
| 21:33.50 | hypa7ia | shouldn't take rocket appliances |
| 21:33.55 | hypa7ia | </trailer_park_boys> |
| 21:34.02 | r0d3nt|m | =( |
| 21:34.07 | *** join/#asterisk brookshire (n=matt@216.215.208.188.nw.nuvox.net) |
| 21:34.47 | r0d3nt|m | i'll load cvs head now... |
| 21:35.09 | r0d3nt|m | the machine is rebooting to unload everything.... |
| 21:35.16 | r0d3nt|m | unloading the modules locks up a dell 2850. |
| 21:35.27 | r0d3nt|m | so i have to reboot to unload the modules... |
| 21:35.36 | r0d3nt|m | then recompile, and load them again.... |
| 21:35.37 | *** join/#asterisk hanchi (n=telliott@015-834-723.area5.spcsdns.net) |
| 21:35.55 | r0d3nt|m | oh crap. 6 lines in the last 3 minutes.. i'm flooding. sorry. |
| 21:36.23 | r0d3nt|m | X-Rob, thanks for the information. i'll give it a go and share the results. |
| 21:36.40 | *** join/#asterisk trym (n=trym@cD9088B17.sdsl.catch.no) |
| 21:40.42 | hanchi | charter went down, gotta just love the folks at charter |
| 21:41.05 | *** join/#asterisk shido6 (n=curtis@d221-68-210.commercial.cgocable.net) |
| 21:41.08 | hanchi | the * box i'm building is not on the net yet to export the error, but the final error was make[2]: *** [decode_i386.o] Error 1 |
| 21:42.07 | hanchi | there are a bunch of decode errors with suffix or operands invalid for 'push' or 'pop' |
| 21:43.23 | shido6 | use http://pastebin.ca |
| 21:43.38 | r0d3nt|m | ~pb |
| 21:43.40 | jbot | rumour has it, pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca/ |
| 21:46.09 | r0d3nt|m | hypa7ia, hi |
| 21:49.17 | *** join/#asterisk pa (n=Paolo@unaffiliated/pa) |
| 21:50.23 | hypa7ia | hola r0d3nt|m |
| 21:50.27 | hypa7ia | making any progress? |
| 21:50.35 | r0d3nt|m | not really. |
| 21:50.39 | r0d3nt|m | almost done... |
| 21:50.39 | hypa7ia | :-s |
| 21:50.46 | hypa7ia | i'm working on to2600.org |
| 21:50.51 | hypa7ia | moving it off windows :-p |
| 21:50.56 | r0d3nt|m | uh.. no... |
| 21:51.09 | r0d3nt|m | asterisk won't run in vmware with hardware. |
| 21:51.09 | r0d3nt|m | =) |
| 21:51.11 | brookshire | heh.. |
| 21:51.22 | r0d3nt|m | oh |
| 21:51.25 | r0d3nt|m | moving it off of windows ?! |
| 21:51.27 | r0d3nt|m | HAHA |
| 21:51.28 | r0d3nt|m | ya... |
| 21:51.29 | hypa7ia | lol, yeah, we might run an asterisk on this one |
| 21:51.35 | hypa7ia | it's a debian VM on XEN |
| 21:51.57 | r0d3nt|m | most everyone thinks i'm and idiot already... lets not make them think I'm running asterisk on windows. |
| 21:51.58 | r0d3nt|m | =) |
| 21:52.04 | hypa7ia | lol |
| 21:52.12 | hypa7ia | the current site runs off a win2k3 box |
| 21:52.13 | r0d3nt|m | s/and/an/ |
| 21:52.19 | r0d3nt|m | which ? |
| 21:52.22 | r0d3nt|m | digium.com ? |
| 21:52.24 | r0d3nt|m | asterisk.org ? |
| 21:52.27 | hypa7ia | to2600.org |
| 21:52.29 | hypa7ia | :-p |
| 21:52.29 | r0d3nt|m | i never looked... |
| 21:52.30 | r0d3nt|m | OH... |
| 21:52.32 | r0d3nt|m | well |
| 21:52.34 | r0d3nt|m | they are securable... |
| 21:52.53 | hypa7ia | yeah, but the guy running it is full of crap |
| 21:52.59 | hypa7ia | so it's not secure |
| 21:53.07 | hypa7ia | and more importantly he's not doing anything with it |
| 21:53.21 | r0d3nt|m | so hack it |
| 21:53.22 | r0d3nt|m | Connected to Asterisk CVS HEAD currently running on dwgvoip (pid = 20201) |
| 21:53.32 | r0d3nt|m | X-Rob, CVS HEAD YOUR MIGHTYNESS. |
| 21:53.36 | r0d3nt|m | anything I should test ?? |
| 21:53.45 | r0d3nt|m | i'll go call the known reproducable echo call. |
| 21:53.46 | Qwell | r0d3nt|m: calling |
| 21:53.51 | r0d3nt|m | calling ? |
| 21:53.55 | Qwell | test calling |
| 21:53.56 | r0d3nt|m | don't call my client please. |
| 21:53.57 | brookshire | speaking of 2600 |
| 21:53.57 | brookshire | lol |
| 21:54.11 | hypa7ia | oh that's this weekend isn't it :-) |
| 21:54.16 | hypa7ia | to2600 needs to run a con |
| 21:54.20 | hypa7ia | that would rock |
| 21:54.23 | r0d3nt|m | i knew i should've edited the phone number out of that conf. |
| 21:54.32 | hypa7ia | hehe oops |
| 21:54.38 | r0d3nt|m | but nooooooooooooooooooooooooooooooooooooooooo. |
| 21:54.41 | r0d3nt|m | that is what i get for trusting. |
| 21:54.55 | Qwell | wtf are you talking about? |
| 21:55.01 | r0d3nt|m | nm. |
| 21:55.02 | brookshire | hypa7ia: start opne |
| 21:55.03 | brookshire | one |
| 21:55.11 | hypa7ia | brookshire: once i have the webserver running |
| 21:55.14 | hypa7ia | small steps :-p |
| 21:55.15 | Qwell | hypa7ia: and make it in anaheim :D |
| 21:55.29 | hypa7ia | Qwell: that wounldn't make much sense... |
| 21:55.30 | hypa7ia | :) |
| 21:55.46 | X-Rob | little hi, little ho. |
| 21:55.57 | hanchi | compete error at http://pastebin.ca/26323, sorry for the flood |
| 21:56.11 | X-Rob | eew. |
| 21:56.15 | X-Rob | ok, that's bad. |
| 21:57.26 | r0d3nt|m | X-Rob, anything else i need to do ?? |
| 21:57.51 | X-Rob | r0d3nt|m - What have you done so far? |
| 21:58.03 | r0d3nt|m | installed your conf and installed CVS HEAD. |
| 21:58.08 | X-Rob | test and play |
| 21:58.13 | X-Rob | that's about the best I can make it |
| 21:58.24 | X-Rob | someone else may have better luck, but that's all I can suggest. |
| 21:58.25 | r0d3nt|m | is the best for asterisk/digium ? |
| 21:58.34 | r0d3nt|m | s/the/that/ |
| 21:58.43 | r0d3nt|m | actually + that the |
| 21:58.53 | r0d3nt|m | is that the best for asterisk/digium ? |
| 21:59.05 | X-Rob | hanchi - apparently '-fPIC' triggers that, it's a bug in the compiler apparently. |
| 21:59.29 | X-Rob | on 64bit stuff. |
| 21:59.46 | X-Rob | edit the makefile, see if you can find that and take it out. |
| 21:59.54 | hanchi | is there anyway to install mpg123 on 64bit ? |
| 21:59.59 | X-Rob | edit the makefile, see if you can find that and take it out. |
| 22:00.02 | r0d3nt|m | omfg. |
| 22:00.09 | r0d3nt|m | echo locally. |
| 22:00.33 | X-Rob | r0d3nt|m - bugger. |
| 22:00.44 | r0d3nt|m | i can't even call VM without echo now. |
| 22:00.52 | X-Rob | ok, your phone is broken then. |
| 22:00.57 | r0d3nt|m | 7940G |
| 22:01.00 | r0d3nt|m | 36 of them ? |
| 22:01.01 | X-Rob | vm has nothing to do with zaptel. |
| 22:01.07 | r0d3nt|m | DUH! |
| 22:01.17 | r0d3nt|m | ok captain obvious. |
| 22:01.30 | r0d3nt|m | before we had no echo locally |
| 22:01.31 | r0d3nt|m | none. |
| 22:01.39 | r0d3nt|m | to vm, to ext to ext... |
| 22:01.41 | r0d3nt|m | remote phones |
| 22:01.44 | X-Rob | you shouldn't have echo locally |
| 22:01.44 | r0d3nt|m | local phones |
| 22:01.53 | X-Rob | the only way you can have echo locally is if the phone is broken. |
| 22:01.56 | X-Rob | reboot the phone. |
| 22:02.03 | r0d3nt|m | alrighty. |
| 22:02.14 | r0d3nt|m | sip flash 7.5 |
| 22:02.21 | X-Rob | no idea about cisco phones |
| 22:02.25 | X-Rob | they're a pain in the arse. |
| 22:02.58 | r0d3nt|m | ya.. they're only like the industry standard... |
| 22:03.05 | X-Rob | or possibly not. |
| 22:03.12 | r0d3nt|m | tftp is such a pain sure.... |
| 22:03.13 | X-Rob | they're the cisco standard. |
| 22:03.17 | r0d3nt|m | it's so trivial. |
| 22:03.22 | Qwell | bad pun |
| 22:04.06 | Qwell | r0d3nt|m: Did you intend to make that horrid pun? |
| 22:04.13 | *** join/#asterisk dca (n=dca@c-67-177-236-87.hsd1.co.comcast.net) |
| 22:04.20 | r0d3nt|m | yes i did... |
| 22:04.34 | r0d3nt|m | there isn't much a pain in the ass about cisco phones... |
| 22:04.47 | X-Rob | except for, perhaps, random echo. |
| 22:05.01 | r0d3nt|m | there is no evidence that suggests it's the phone. |
| 22:05.09 | X-Rob | yes there is. |
| 22:05.12 | r0d3nt|m | where ? |
| 22:05.22 | r0d3nt|m | before cvs head, there was no echo locally |
| 22:05.23 | X-Rob | you're getting echo from voicemail |
| 22:05.26 | r0d3nt|m | to Vm or EXT/EXT |
| 22:05.34 | r0d3nt|m | we had none until now. |
| 22:05.36 | X-Rob | do a TCP dump. |
| 22:05.57 | X-Rob | see if you get echo whilst you're leaving a message in voicemail |
| 22:06.00 | X-Rob | that's a good test |
| 22:06.08 | X-Rob | because asterisk is not sending _any_ data then. |
| 22:06.10 | hanchi | we have 33 7960G Cisco phones, and they developed local echo, changed back to sip 7.4 and most went away, our 12 polycom 601 and four snom 360 had no echo when the cisco's did |
| 22:06.47 | X-Rob | hanchi - love the snoms. |
| 22:07.12 | *** join/#asterisk niZon (n=ilt@S0106deadbeefbeef.wp.shawcable.net) |
| 22:07.33 | r0d3nt|m | actually, we're using 7.3, i saw on the reboot... |
| 22:07.44 | hanchi | yep, but the polycom 601 with the sidecars was our only option for a sidecar, besides 7960 w/7914 on sccp, and i was not willing to try sccp on a critical systme, i like my job |
| 22:07.52 | r0d3nt|m | X-Rob, dude... distortion and mass ammount of echo are something we had none of before.... |
| 22:07.56 | *** join/#asterisk zotz (n=zotz@24.231.36.100) |
| 22:08.28 | hanchi | the polycoms work great |
| 22:08.41 | X-Rob | r0d3nt|m - see if you've got echo whilst you're leaving a voicemail message. |
| 22:10.16 | r0d3nt|m | no echo to VM while recording. |
| 22:10.25 | r0d3nt|m | only echo thru the PRI now |
| 22:10.27 | r0d3nt|m | ALWAYS |
| 22:10.29 | r0d3nt|m | distorted |
| 22:10.33 | r0d3nt|m | clipping |
| 22:10.41 | X-Rob | well |
| 22:10.41 | r0d3nt|m | basically farked. |
| 22:10.45 | r0d3nt|m | my client is calling me. |
| 22:10.54 | X-Rob | maybe there is a 10db gain on the txline |
| 22:11.01 | X-Rob | set txgain to =-9 again |
| 22:11.05 | X-Rob | see if that makes a difference 8-( |
| 22:11.47 | r0d3nt|m | i'm going to -3 / -3 and i'll test again |
| 22:12.02 | r0d3nt|m | do the modules need to be reloaded ? ?or will ztcfg make the change ? |
| 22:12.28 | X-Rob | 'zaptel reload' does it |
| 22:12.36 | X-Rob | ztcfg doesn't have anything to do with it. |
| 22:12.37 | r0d3nt|m | that unloads and reload the modules |
| 22:12.38 | drumkilla | reload chan_zap.so ... |
| 22:12.46 | r0d3nt|m | which locks up a dell 2850 |
| 22:12.55 | r0d3nt|m | drumkilla, thank you |
| 22:13.05 | drumkilla | yw |
| 22:13.08 | r0d3nt|m | i'll test again. |
| 22:13.10 | r0d3nt|m | brb. |
| 22:13.16 | hanchi | X-Rob, there is no -fPIC or -fpic in the Makefile, is that what i wanted to remove from the Makefile ? |
| 22:13.28 | X-Rob | hanchi - that's the hint that I found on a website. |
| 22:13.37 | X-Rob | you might have to use madplay |
| 22:14.11 | *** join/#asterisk trym (n=trym@cD9088B17.sdsl.catch.no) |
| 22:14.40 | r0d3nt|m | ok |
| 22:14.48 | r0d3nt|m | no more clipping or distortion |
| 22:14.51 | r0d3nt|m | but still echo's |
| 22:15.00 | r0d3nt|m | thru the PRI |
| 22:15.05 | r0d3nt|m | to the reproducable # |
| 22:15.17 | r0d3nt|m | i'm on telepacific, they're on mpower... |
| 22:15.30 | r0d3nt|m | they're 2 lines of POTS, and we all know i'm using a PRI 23+D |
| 22:15.55 | *** join/#asterisk nassy (n=nassy@207-38-252-103.c3-0.wsd-ubr1.qens-wsd.ny.cable.rcn.com) |
| 22:16.27 | Qwell | it only gives echo on one number? |
| 22:16.55 | X-Rob | turn the gain down to -9. which the telco suggested. |
| 22:17.00 | X-Rob | txgain and rxgain |
| 22:17.09 | X-Rob | (that's probably excessive, but it's worth a shot) |
| 22:17.29 | hardwire | heh |
| 22:17.38 | hardwire | testing a new DSL for just ulaw.. |
| 22:17.40 | hardwire | so weird |
| 22:17.54 | hardwire | feel the need to compress and trunk.. grr.. |
| 22:18.10 | X-Rob | heh |
| 22:18.20 | hardwire | X-Rob: so you do/do not have snom speakerphone woes? |
| 22:18.28 | hardwire | btw.. It ook a 360 apart.. |
| 22:18.32 | hardwire | its so damn simple |
| 22:18.34 | r0d3nt|m | Qwell, several numbers, that are reproducable... |
| 22:18.40 | hardwire | like bigger gumstix w/ a dsp in it |
| 22:18.57 | *** join/#asterisk maruz (n=maumar@host177-128.pool80104.interbusiness.it) |
| 22:20.18 | X-Rob | hardwire - yep. same with the GXP2000's - there's nothing to 'em these days. |
| 22:20.22 | r0d3nt|m | Qwell, the client reports random echo and random dropped calls.. |
| 22:20.37 | X-Rob | random dropped calls == busydetect=yes |
| 22:20.43 | X-Rob | random echo == hard. |
| 22:20.46 | r0d3nt|m | alrighty |
| 22:20.52 | r0d3nt|m | hard ?? |
| 22:20.56 | X-Rob | difficult. |
| 22:20.57 | r0d3nt|m | hard to diagnose ? |
| 22:21.01 | r0d3nt|m | aye... |
| 22:21.15 | X-Rob | A lot of yanks get a telabs echo canceller |
| 22:21.24 | X-Rob | apparently you can buy 'em cheap on ebay |
| 22:21.36 | r0d3nt|m | the telco tried installing a CPE side echo can |
| 22:21.40 | r0d3nt|m | it made no difference... |
| 22:22.01 | X-Rob | possibly a local echo issue. Like you were having xtn->xtn earlier. |
| 22:22.10 | X-Rob | a problem in the phone |
| 22:22.14 | r0d3nt|m | we had none before. |
| 22:22.18 | r0d3nt|m | like never.... |
| 22:22.19 | X-Rob | convince the customer to get a couple of snom 360's or 320s |
| 22:22.29 | r0d3nt|m | why ?? |
| 22:22.34 | X-Rob | because they work really well. |
| 22:22.35 | r0d3nt|m | we never had any echo locally before... |
| 22:24.58 | hardwire | X-Rob: it would be so nice to have a working speakerphone |
| 22:24.59 | hardwire | heh |
| 22:25.26 | X-Rob | hardwire - It is a bit crappy, yes. 8-\ |
| 22:25.36 | X-Rob | Their internal echo canceller is a bit agressive. |
| 22:25.42 | hardwire | yeh it is |
| 22:25.58 | hardwire | I issued bugs w/ them.. they seemed to think it worked fine in their lab |
| 22:26.01 | X-Rob | personally, I'd prefer a good echo can and crap speakerphone. |
| 22:26.10 | X-Rob | it really depends on the surroundings |
| 22:26.11 | hardwire | which probably has nothing on the walls and is 18 feet hight. |
| 22:26.19 | X-Rob | all the ones I have in quiet offices work really really well |
| 22:26.31 | X-Rob | as soon as there's background noise, they're crap. |
| 22:26.32 | hardwire | users are just complaining about the tunnel sound to it |
| 22:26.44 | hardwire | and.. all I can say to them .. time after time is pick up the fucking handset. |
| 22:26.47 | hardwire | whats with you? |
| 22:26.49 | X-Rob | heh |
| 22:26.51 | hardwire | is speakerphone that damn important? |
| 22:26.56 | *** join/#asterisk magister1 (n=peter@a194-109-161-140.adsl.xs4all.nl) |
| 22:26.56 | hypa7ia | yes |
| 22:27.01 | hypa7ia | :-) |
| 22:27.02 | hardwire | every day.. "you know the speakerphone isn't working well right?" |
| 22:27.11 | hardwire | yeh.. like I said.. its not going to just change one day.. |
| 22:27.14 | *** join/#asterisk |Vulture| (n=V@c-69-180-67-53.hsd1.fl.comcast.net) |
| 22:27.17 | hardwire | "well you should work on it" |
| 22:27.20 | hardwire | as if I am snom. |
| 22:27.26 | hardwire | pisses me the twinkie off |
| 22:27.34 | magister1 | sorry to barge in |
| 22:27.40 | X-Rob | the twinkie! |
| 22:27.48 | magister1 | i have a problem |
| 22:27.49 | X-Rob | You got blf's and everything working? |
| 22:28.02 | hardwire | X-Rob: ? |
| 22:28.04 | X-Rob | magister1 - just say what your problem is. if someone can help you they will. |
| 22:28.09 | X-Rob | busy lamp fields |
| 22:28.14 | X-Rob | eg so you can see who's on the phone |
| 22:28.17 | hardwire | ah |
| 22:28.18 | *** join/#asterisk iq (n=iq@204-26-72-44.omah.qwest.net) |
| 22:28.22 | hardwire | X-Rob: semi sorta |
| 22:28.28 | X-Rob | can't do pickups yet |
| 22:28.28 | magister1 | does anyone know about zaptel compilation errors? |
| 22:28.32 | hardwire | whenever the snoms reregister the subscribes.. it works |
| 22:28.37 | hardwire | but thats only every 15 minutes |
| 22:28.39 | X-Rob | magister1 - post the error to pastebin |
| 22:28.40 | X-Rob | ~pb |
| 22:28.46 | jbot | from memory, pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca/ |
| 22:28.53 | hardwire | X-Rob: are you using some sip patchery on chan_sip? |
| 22:28.59 | X-Rob | hardwire - nup. |
| 22:29.01 | hardwire | hmm |
| 22:29.05 | hardwire | and you have blf? |
| 22:29.07 | X-Rob | yeah |
| 22:29.14 | hardwire | ok yoda. |
| 22:29.15 | X-Rob | uh |
| 22:29.18 | |Vulture| | Anyone here using AEL for *? |
| 22:29.19 | hardwire | give me the details. |
| 22:29.57 | X-Rob | hardwire - just checked it, working fine |
| 22:29.59 | hypa7ia | i thought blf was in HEAD |
| 22:30.03 | hardwire | X-Rob: ugh |
| 22:30.07 | X-Rob | however |
| 22:30.17 | hardwire | X-Rob: would you mind handing me a config for your phones? plus sip.conf? |
| 22:30.20 | X-Rob | I _did_ notice at a clients place that a blf light was off when it should have been one |
| 22:30.32 | hardwire | X-Rob: remote pbx? |
| 22:30.47 | *** join/#asterisk Rez (i=lorez@freenode/staff/lorez) |
| 22:30.53 | nassy | if i use a hardware ip phone with asterisk, how do i program the buttons on the phone. is that done on each phone (about 150 for us), or is that done on asterisk |
| 22:31.04 | X-Rob | hardwire - there's nothing complex. fkey3: dest <sip:321@202.173.178.224;user=phone> |
| 22:31.15 | hardwire | X-Rob: 1.2.0 1.0.9 and CVS |
| 22:31.21 | hardwire | 1.2.0 made lights happen all over the place |
| 22:31.24 | r0d3nt|m | nassy, done on the phone, most likely thru tftp. |
| 22:31.25 | hardwire | CVS made less lights happen. |
| 22:31.34 | hypa7ia | nassy: which kind of phones? |
| 22:31.42 | hardwire | but nothing in sync with real life |
| 22:32.06 | nassy | ok, thanks. im considering cisco but they dont seem to have a good rep with asterisk and firmware upgrades |
| 22:32.07 | hardwire | every 15 minutes or so the snoms just up and reregister everything.. causing the asterisk server to slow alittle from all the registrations :) |
| 22:32.18 | hardwire | and then .. if a phone is off-hook and in use it will show up on the f-key |
| 22:32.23 | hardwire | but other than that.. tis not instant. |
| 22:32.24 | nassy | we currently use toshiba. |
| 22:32.29 | magister1 | i have copied as much as i could in http://pastebin.ca/26328 |
| 22:32.52 | hardwire | X-Rob: I would die for a snom config. |
| 22:32.57 | X-Rob | hardwired - Hurm. There were some sip modifications put into HEAD recently. |
| 22:33.14 | hardwire | X-Rob: but it still should have worked like.. 1 month ago . |
| 22:33.24 | X-Rob | magister1 - what happens when you do what it says to do? |
| 22:33.26 | hardwire | I seriously think I saw too many snom + asterisk howtos.. and overdid it. |
| 22:33.26 | antoniojulio | if the echodemo-test is not heard(in softphone), is that the sound card is not full-duplex(in server asterisk)? |
| 22:33.55 | X-Rob | hardwire - it did work a month ago. |
| 22:34.33 | magister1 | sorry haven't tried that one ***blush*** |
| 22:34.36 | nassy | hypa7ia, ok, thanks. im considering cisco but they dont seem to have a good rep with asterisk and firmware upgrades. we currently use a toshiba strata system. would be nice if i could get those phones to work with asterisk. |
| 22:34.44 | hardwire | X-Rob: seriously.. just dump url/settings from a snom and pastebin it to me or somethin :) |
| 22:34.50 | hardwire | I will sort the keys and do a diff. |
| 22:35.06 | magister1 | i thoucht it would compile just out of the box |
| 22:35.06 | hypa7ia | nassy: if there's a pri on the toshiba you could very well... or use them with a channel bank.... |
| 22:35.11 | hardwire | also I may need to have notifysubscribe set differently. |
| 22:35.20 | hardwire | but atleast it works.. just every 15 minutes it updates |
| 22:35.34 | hypa7ia | nassy: as for issues with cisco, they are more administrivia than anything, the phones themselves are awesome once you get the licensing and firmware sorted out |
| 22:35.42 | X-Rob | hardwire - http://202.173.178.224/snom/stds.html |
| 22:35.59 | hardwire | X-Rob: let me seee now. |
| 22:36.01 | hardwire | mwa ha ha ha |
| 22:36.02 | nassy | we have a T1 pri going into to the toshiba strata. i was referring to the toshiba ip phone with asterisk. |
| 22:36.43 | X-Rob | hardwire - wget it |
| 22:36.56 | hardwire | ok |
| 22:37.20 | hypa7ia | nassy: gotcha, are they sIP? |
| 22:37.51 | nassy | hypa7ia: thats good to know. our windows computer consultant doesnt like the cisco phones. not clear why but he has my bosses ear |
| 22:38.13 | magister1 | i have 2 folders for linux |
| 22:38.14 | nassy | no i think they use h.3 something with MGBP |
| 22:38.19 | nassy | something like that |
| 22:38.26 | hypa7ia | nassy: "doesn't like...." he should have a better reason than that lol |
| 22:38.34 | nassy | hehe |
| 22:38.38 | hypa7ia | nassy: they are prolly either h.323 or MGCP |
| 22:38.44 | hypa7ia | neither of which is optimal |
| 22:38.55 | hypa7ia | you should see if there is sip firmware for them, maybe |
| 22:39.18 | nassy | yeah, when i did a packet sniff a while back when we had a problem i saw both show up in the data |
| 22:39.24 | magister1 | one called ~linux and another linux 2.4.21-99 |
| 22:39.40 | magister1 | which one should i use |
| 22:39.43 | nassy | ill ask but it is hard getting info from our phone vendor which is why id like to switch to asterisk |
| 22:39.51 | hypa7ia | nassy: weird... it may be that the phones are talking one protocol and the pbx is talking another to the gateway |
| 22:40.06 | hypa7ia | nassy: search the interwebs too :-) |
| 22:40.17 | hypa7ia | what model(s) are the toshiba phones? |
| 22:40.53 | nassy | i dont remember off hand, let me see if i can find it in my notes. |
| 22:41.07 | hypa7ia | coolcool |
| 22:41.49 | hardwire | anybody ever set up multi-homing in debian? |
| 22:43.41 | hypa7ia | the person sitting next to me has |
| 22:43.46 | hypa7ia | and says you shouldn't |
| 22:43.47 | hypa7ia | :-) |
| 22:44.09 | *** part/#asterisk antoniojulio (n=antonioj@unaffiliated/antoniojulio) |
| 22:44.11 | hardwire | well get him on here :) |
| 22:44.19 | hypa7ia | he says it worked okay but you should consider investing in hardware that's dedicated to doing that kind of thing |
| 22:44.29 | hypa7ia | someone else has stolen his laptop |
| 22:44.31 | hypa7ia | :-) |
| 22:44.48 | hypa7ia | "flapping your bgp is not good" |
| 22:44.48 | r0d3nt|m | thats what he gets for multi-homing debian. |
| 22:44.52 | hypa7ia | lol |
| 22:45.00 | hardwire | heh |
| 22:45.06 | hypa7ia | at the caffe we're at, not actually stolen :-) |
| 22:45.12 | hardwire | I read the advanced routing howtos on it |
| 22:45.15 | hardwire | and it worked just fine |
| 22:45.25 | hypa7ia | s/stolen/borrowed by a cute girl/ |
| 22:45.34 | hardwire | I was hoping somebody set up ifupdown scripts for it |
| 22:45.50 | hypa7ia | that's so wrong |
| 22:45.58 | hardwire | naw |
| 22:46.08 | hypa7ia | drew says to check out zebra |
| 22:46.10 | r0d3nt|m | hypa7ia, they are not to be trusted. |
| 22:46.18 | hardwire | hypa7ia: I use quagga. which is zebra. |
| 22:46.22 | hypa7ia | r0d3nt|m: i know eh |
| 22:46.25 | hardwire | I fail to see how that can set up a multi-homed setuop |
| 22:46.53 | hypa7ia | can you be more specific about what you mean by multi-homed? |
| 22:46.59 | hardwire | two gateways. |
| 22:47.09 | nassy | hypa7ia: http://www.toshiba.com/taistsd/products/tel_ip_telephony.jsp the 10 button one |
| 22:47.14 | hardwire | one machine .. 3 nics 1 private 2 on different public networks. |
| 22:47.23 | hypa7ia | that's not multi-homed.... that's just multi-linked :-) |
| 22:48.02 | hardwire | err |
| 22:48.25 | hypa7ia | nassy: that site doesn't work so well in firefox |
| 22:48.34 | r0d3nt|m | that is ztspeed for ?? |
| 22:48.37 | r0d3nt|m | what* |
| 22:48.54 | hardwire | http://lartc.org/howto/lartc.rpdb.multiple-links.html |
| 22:48.55 | hardwire | that |
| 22:48.55 | hardwire | :) |
| 22:49.00 | hardwire | so multi-link |
| 22:49.07 | hardwire | multi-home? now whats that mean? |
| 22:49.13 | r0d3nt|m | i use zttool and zttest |
| 22:49.15 | hypa7ia | ahhh that's what we were trying to do at linuxcaffe |
| 22:49.18 | r0d3nt|m | never used ztspeed |
| 22:49.29 | hypa7ia | we ended up getting a little $200CAD linksys box with two wan ports :-p |
| 22:49.34 | hardwire | heh |
| 22:49.37 | nassy | hypa7ia: yeah i know. im using safari and had a hard time getting there too. their whole system is not very platform agnostic. |
| 22:49.38 | hardwire | hypa7ia: one problem |
| 22:49.42 | X-Rob | r0d3nt|m - read the first 5 or so lines in the file |
| 22:49.45 | hardwire | each wan uplink is a /30 |
| 22:49.46 | hypa7ia | you have 3, i know |
| 22:49.47 | X-Rob | ztspeed.c |
| 22:49.48 | hardwire | with a /24 coming into it |
| 22:50.02 | hardwire | so I need to route through these as well.. and keep them seperated. |
| 22:50.09 | r0d3nt|m | <PROTECTED> |
| 22:50.09 | r0d3nt|m | <PROTECTED> |
| 22:50.09 | r0d3nt|m | <PROTECTED> |
| 22:50.09 | r0d3nt|m | <PROTECTED> |
| 22:50.16 | X-Rob | ~pb |
| 22:50.18 | jbot | i heard pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca/ |
| 22:50.20 | X-Rob | please don't flood. |
| 22:50.31 | *** join/#asterisk jontow (i=jontow@bsd.adminforrent.com) |
| 22:50.33 | r0d3nt|m | 4 lines is flooding ? |
| 22:50.40 | hardwire | r0d3nt|m: today it is |
| 22:50.40 | hypa7ia | X-Rob: grow up |
| 22:50.52 | X-Rob | hypa7ia - wtf? |
| 22:51.09 | X-Rob | I could have pasted it |
| 22:51.14 | hypa7ia | 4 lines dude |
| 22:51.21 | X-Rob | but it's rude to paste shit that everyone has and can look at themselves. |
| 22:51.36 | hypa7ia | whatever. |
| 22:51.37 | r0d3nt|m | obviously I'm just too much of a loser for this channel. I won't make par. |
| 22:51.38 | fugitivo | hypa7ia: my bitchx console is 7 lines, so it's a big flood for me |
| 22:51.43 | X-Rob | shall I paste 4 random lines from chan_sip.c? |
| 22:51.47 | X-Rob | what use will that be to you? |
| 22:51.50 | *** part/#asterisk r0d3nt|m (i=nobody@wsip-24-234-241-145.lv.lv.cox.net) |
| 22:52.25 | hardwire | well |
| 22:52.33 | hardwire | I see the focus went away from me. |
| 22:52.37 | hardwire | so I would like to bring it back to me now. |
| 22:52.41 | hardwire | so.. me me me me me |
| 22:52.42 | hypa7ia | hardwire: good plan :-) |
| 22:52.45 | hypa7ia | omg 4 lines |
| 22:52.46 | fugitivo | hardwire: flood the channel and you'll get the focus |
| 22:53.26 | hardwire | X-Rob: I got a 4 line flooder in debian-kernel the other day.. and they ass-raped me.. |
| 22:53.55 | hardwire | I was thinking about this the other day however.. IRC flooding.. I really think instead of encouraging pastebin.. encouraging in-client tinyfication tools instead. |
| 22:54.16 | hardwire | because pastebin is not well known.. and slow as a bitch. |
| 22:54.19 | magister1 | ok i have done make cloneconfig && make dep' in /usr/src/linux/ |
| 22:54.20 | hypa7ia | that would be cool |
| 22:54.25 | hardwire | unless you think voip-info.org needs a pastebin |
| 22:54.30 | hardwire | then it may be more usefull |
| 22:54.40 | magister1 | where do i find ztdummy? |
| 22:54.44 | X-Rob | pastebin.ca is the better (eg, faster) one |
| 22:54.47 | hypa7ia | hardwire: v-i is slow as is :-/ |
| 22:55.02 | hardwire | hypa7ia: I personally think having a key:value type scenario for floods would rock.. |
| 22:55.12 | hardwire | vew a flood in its own window.. or expand it like a tree view |
| 22:55.19 | hypa7ia | that'd be neat |
| 22:55.27 | hypa7ia | but too may people are set in their irc-ways |
| 22:55.33 | hardwire | because I doubt it causes the network much strife. |
| 22:55.46 | hardwire | hypa7ia: well.. irc-ng needs be born then. |
| 22:56.02 | X-Rob | off you go. |
| 22:56.05 | fugitivo | hardwire: what do you mean? |
| 22:56.11 | hardwire | however I can see how one flood of 4 lines.. times the 2371 people in here.. makes a lot of traffic happen |
| 22:56.19 | hardwire | X-Rob: hey.. I am an idealist non-programmer. |
| 22:56.26 | magister1 | sorry found it thru lsmod |
| 22:56.28 | hardwire | 237 :) |
| 22:57.01 | hardwire | X-Rob: mofo.. I can't find a major difference. |
| 22:57.13 | X-Rob | there's nothing difficult there. |
| 22:57.38 | X-Rob | oh, if you're running 4.3, make sure you turn the mic volume down to 3, otherwise it overwhelms KB1 8) |
| 22:57.59 | hardwire | KB1 ? |
| 22:58.06 | hardwire | X-Rob: shit.. can you adjust the ring volume? |
| 22:58.12 | hardwire | my settings have no effect. |
| 22:58.18 | hardwire | it just effects overall speakerphone volume. |
| 22:59.01 | X-Rob | hardwire - hurm. Dunno. Still got to sort out the rings. |
| 22:59.15 | hardwire | X-Rob: broken open a snom 360 yet? |
| 22:59.42 | hardwire | the speaker element is pretty radio shack |
| 23:00.09 | hardwire | I have a feeling if I foam insulated the speaker.. the echocan may work better |
| 23:00.18 | hardwire | as well as tried to isolate the mic from the casing a bit more |
| 23:00.30 | hardwire | they did that by using glue :) |
| 23:00.36 | X-Rob | Heh |
| 23:00.47 | hardwire | ok |
| 23:00.56 | hardwire | I am going to update my production to latest CVS HEAD |
| 23:01.03 | hardwire | all because you mentioned the busy flags. |
| 23:01.04 | magister1 | ok thanx all got 2 go |
| 23:01.06 | hardwire | and now I have to know. |
| 23:01.18 | hardwire | oh shit.. I forgot about the configuration changes. |
| 23:01.24 | X-Rob | there shouldn't be any |
| 23:01.30 | hardwire | I saw some hit HEAD recently |
| 23:01.40 | hardwire | I just need to look at CVS log of each config I use I guess |
| 23:01.59 | X-Rob | there was a bug where the +101 stuff wasn't working for a couple of days |
| 23:02.01 | X-Rob | but that's sorted now. |
| 23:02.06 | hardwire | yay |
| 23:02.09 | hardwire | that would have sucked |
| 23:02.57 | hardwire | g726 should be fun |
| 23:04.38 | *** join/#asterisk MatsK (n=mk@55.80-203-80.nextgentel.com) |
| 23:05.18 | trelane | anyone gotten asterisk to communicate with a vic-2fxs? |
| 23:05.28 | hypa7ia | hehe, running what ios on the router? |
| 23:05.53 | hardwire | ? |
| 23:05.55 | trelane | 12.3 |
| 23:06.01 | hardwire | ah |
| 23:06.08 | hypa7ia | trelane: which featureset though |
| 23:06.18 | hypa7ia | you need the enterprise sp services i think |
| 23:06.25 | trelane | c1700-sv8y7-mz.123-15.bin |
| 23:06.38 | trelane | ip voice and some other stuff |
| 23:06.44 | hypa7ia | ahh you should be good then |
| 23:06.52 | trelane | right this has the full voice featureset |
| 23:06.58 | trelane | what it doesn't support is sip register |
| 23:07.08 | hypa7ia | yup, just need to get an h.323 or mgcp trunk going |
| 23:07.22 | trelane | any online examples? |
| 23:07.47 | hypa7ia | hardwire: my friend got his laptop back, he's hopping onto freenode now |
| 23:08.03 | hypa7ia | trelane: might want to try the cisco netpro forums |
| 23:08.10 | trelane | noted |
| 23:08.29 | *** join/#asterisk ahammond (n=ahammond@66.11.182.39) |
| 23:08.50 | ahammond | strange, no dialtone... |
| 23:09.26 | hypa7ia | trelane: and froma cursory google there are lots of results on asterisk-users |
| 23:10.35 | hardwire | hypa7ia: ok |
| 23:10.48 | ahammond | hardwire: you were asking about multi-link stuff? |
| 23:11.23 | hardwire | yeh |
| 23:11.30 | hardwire | I need to learn it to a serious degree |
| 23:11.47 | hardwire | sans the buying of hardware part :) |
| 23:12.02 | ahammond | hardwire: well, please describe the problem you're trying to solve |
| 23:12.27 | hardwire | ahammond: setting up multi-link using just debian.. possibly zebra (for setting up the routing tables) |
| 23:12.32 | hardwire | without hacking up init.d scripts |
| 23:13.02 | ahammond | hardwire: ok, that's some technical stuff you're trying to implement. what's the "business problem" you're trying to solve? |
| 23:14.20 | hardwire | ahammond: multi-linking and tagging default routes for different networks. |
| 23:14.29 | *** join/#asterisk niZx (n=ilt@S0106deadbeefbeef.wp.shawcable.net) |
| 23:14.38 | Felagund | I have a question |
| 23:14.44 | hardwire | Felagund: most do. |
| 23:15.06 | Felagund | I have an IT staff member who talked my boss into setting up asterisk for our company |
| 23:15.17 | ahammond | hardwire: again, a technical description. what's the actual problem? |
| 23:15.29 | hardwire | ahammond: I don't think your gonna get it from me how you want to hear it. |
| 23:15.31 | Felagund | how many things can go wrong with it |
| 23:15.44 | hardwire | Felagund: you could lose millions of dollars doing it. |
| 23:15.46 | Felagund | b/c this IT staff member hasn't really set it up before |
| 23:15.47 | hardwire | or more. |
| 23:15.51 | ahammond | hardwire: ok. well, assuming you want to do multiple links and tagged rr routing... :) |
| 23:16.04 | Felagund | and he only knows the basics of linux |
| 23:16.11 | hypa7ia | Felagund: urk |
| 23:16.14 | ahammond | your requirement is load balancing, right? |
| 23:16.17 | hardwire | ahammond: no |
| 23:16.23 | ahammond | what then? |
| 23:16.28 | Felagund | the whole thing makes me nervous becuase it's our phones |
| 23:16.30 | fugitivo | Felagund: let him play and learn, don't use a production system |
| 23:16.31 | hardwire | one dsl is set aside for voip trunking |
| 23:16.38 | Felagund | I wanted to outsource the asterisk setup to a local linux company |
| 23:16.41 | hardwire | the other for regular use. |
| 23:16.46 | hypa7ia | Felagund: that's exactly what you should do |
| 23:16.48 | hardwire | however the server needs to be visable to both uplinks |
| 23:16.51 | hardwire | so |
| 23:16.52 | Felagund | and get a support contract from them |
| 23:16.53 | hypa7ia | or a company that specializes in asterisk |
| 23:16.54 | hardwire | its not a big problem :) |
| 23:16.58 | hypa7ia | Felagund: what's local for you? |
| 23:17.03 | Felagund | the local linux company specializes in asterisk |
| 23:17.11 | Felagund | but my boss is listening to the entry level IT guy who says it's easy |
| 23:17.12 | hypa7ia | then you're golden |
| 23:17.17 | ahammond | hardwire: ok, needs to be visible from internal networks or external networks? |
| 23:17.40 | Felagund | and thinks that if we can do it in house it'll be cheaper |
| 23:18.09 | hardwire | ahammond: gimme a sec.. I am configuring it at the moment. |
| 23:18.15 | hypa7ia | Felagund: until it breaks :-( |
| 23:18.17 | Felagund | I want to move to voip.. however I'm terrified of something going wrong with the server that the IT guy won't know how to fix |
| 23:18.19 | hypa7ia | because it might |
| 23:18.21 | Felagund | and then our phones going down |
| 23:18.39 | Felagund | and us not being able to get support from anybody |
| 23:19.00 | hypa7ia | well, even if you set it up yourself, if you can give root access you can get support for it. |
| 23:19.09 | hypa7ia | from digium or others |
| 23:19.18 | Felagund | digium gives support? the guy told me they didn't |
| 23:19.23 | hypa7ia | but it'll cost, obviously |
| 23:19.28 | hypa7ia | haha that's silly |
| 23:19.32 | hypa7ia | ofcourse they support it |
| 23:19.46 | hypa7ia | i think the current rate if you just call them up is $175/hr |
| 23:19.56 | ahammond | hardwire: if the server needs to be "visible" from the outside and may only have _one_ IP by which it is addressed then you're back to multi-homing. |
| 23:20.04 | hardwire | ahammond: brb.. |
| 23:20.15 | hardwire | if this is just a symantics of speech issue.. heh.. bugger off |
| 23:20.39 | hardwire | I did ask what the difference was.. so maybe you are just enlightening me? |
| 23:20.44 | X-Rob | Felagund - asterisk is _not_ easy. It's understandable, but it's _not_ 'easy' |
| 23:20.46 | hardwire | just.. slowly. |
| 23:20.46 | ahammond | hardwire: no, there are some major differences between multi-links and mutli-homed |
| 23:20.53 | Felagund | they think it is easy |
| 23:20.55 | ahammond | hardwire: ok, multi-homed is pretty simple. |
| 23:20.57 | X-Rob | And if the IT guy is talking about asterisk@home, kick him in the nuts. |
| 23:21.06 | hypa7ia | oh cripes yes |
| 23:21.08 | Felagund | in fact they're planning to set up asterisk for two outside companies in 4 months |
| 23:21.12 | ahammond | hardwire: er... sorry s/homed/linked/ |
| 23:21.15 | hardwire | ahammond: in the end.. two isp's.. two /24's per two seperate /30's |
| 23:21.19 | Felagund | the IT guy says it is easy |
| 23:21.26 | hardwire | including another isp. |
| 23:21.28 | X-Rob | the problem is, A@H _is_ easy. |
| 23:21.29 | ahammond | hardwire: ok, so far it's only mult-linked |
| 23:21.31 | X-Rob | It's astoundingly easy. |
| 23:21.31 | Felagund | of course the IT guy also thinks it is easy for us to develop our own wireless protocol |
| 23:21.32 | X-Rob | until it break |
| 23:21.33 | X-Rob | s |
| 23:21.35 | hardwire | where port 80 will be mapped for internal requests |
| 23:21.50 | hardwire | ahammond: but nothing at the ISP level.. or multi ISP level |
| 23:22.00 | X-Rob | and because someone without a clue has set it up, they have no idea how to fix it. Asterisk is an arcane and deeply complex bit of software. |
| 23:22.02 | ahammond | hardwire: ok, then it is _not_ multi-homed. :) |
| 23:22.15 | hypa7ia | Felagund: how has this person not been fired yet :s |
| 23:22.19 | fugitivo | Felagund: everything is easy if you have de knowledge |
| 23:22.24 | hypa7ia | and/or kicked in the nuts / ovaries |
| 23:22.26 | hardwire | ahammond: if I had two isps.. that wanted to route me the same networks.. then it would be. |
| 23:22.26 | Felagund | because his dad is friends with the boss |
| 23:22.29 | Felagund | so the boss listens to him |
| 23:22.31 | hypa7ia | ahh gotcha |
| 23:22.33 | Felagund | basically I promoted asterisk at work |
| 23:22.33 | hardwire | but its not a redundancy or load balance thing. |
| 23:22.38 | Felagund | and said lets move to voip |
| 23:22.42 | ahammond | hardwire: with multi-homed, you need to get an ASN assigned, and IP space, then publish BGP records |
| 23:22.46 | Felagund | how great asterisk is blah blah blah |
| 23:22.52 | hardwire | ahammond: I read that just a few seconds ago |
| 23:22.58 | Felagund | this guy popped up and said he can set it up instead of us needing to outsource |
| 23:23.00 | ahammond | hardwire: yeah, so that's pretty involved. :) |
| 23:23.01 | hardwire | Felagund: you did all this w/o research? |
| 23:23.04 | hardwire | thats how you lose jobs :) |
| 23:23.07 | ahammond | hardwire: what you're looking at is more doable |
| 23:23.10 | hypa7ia | Felagund: if you don't have the expertise, then don't try it yourself... remember that a bad experience with it will turn people off it for a long time |
| 23:23.18 | Felagund | I wanted to outsource the bloody thing |
| 23:23.25 | hardwire | ahammond: well.. hopefully quagag will set up the tables correctly.. /me checks |
| 23:23.33 | hypa7ia | hardwire: sounds mroe like this other dude wanted to stick his fingers in the pie |
| 23:23.35 | Felagund | I didn't want to set it up myself.. from the beginning I hadn't planned on it |
| 23:23.36 | hardwire | I don't like having to add the tables to rt_tables |
| 23:23.38 | Felagund | well |
| 23:23.41 | ahammond | hardwire: nah, you don't need to get that complicated. :) |
| 23:24.00 | Felagund | basically in another 4 months this guy is going to be setting up a phone system for like 300 people in 3 different locations |
| 23:24.04 | Felagund | a few over a satellite link |
| 23:24.09 | hardwire | ahammond: I don't want to just do a split-access setup w/o doing it aroudn something that is pretty standard |
| 23:24.22 | hypa7ia | haha voip + satellite != a good idea |
| 23:24.31 | fugitivo | satellite?? |
| 23:24.36 | hardwire | I hate having to go into a site and reverse everything the sysadmin did because he could do it faster his own very undocumented way. |
| 23:24.36 | hypa7ia | Felagund: so clueless dude wants to do all of this? |
| 23:24.36 | ahammond | hardwire: ok, that's a kinda complicated setup. |
| 23:24.38 | hardwire | hypa7ia: I do it |
| 23:24.46 | Felagund | clueless dude told boss that we could do it |
| 23:24.46 | hardwire | and I like it |
| 23:24.49 | ahammond | hardwire: you may find it easier to just put a router between the two networks. |
| 23:24.58 | Felagund | boss went and also sold this idea two two of our customers who are over a satellite link |
| 23:25.10 | hardwire | ahammond: I may if this fails |
| 23:25.13 | nassy | Felagund: are you switching to voip completely? do you guys receive or send faxes? |
| 23:25.22 | hardwire | I would rather use the two 1u itx routers I built however :) |
| 23:25.26 | Felagund | yes, we do a ton of faxing |
| 23:25.27 | ahammond | hardwire: well, there's the whole issue of maintenance once you've got this done... |
| 23:25.30 | Felagund | wanted to use asterisk to receive the faxes |
| 23:25.34 | Felagund | as PDFs |
| 23:25.38 | hardwire | but yeh.. if I were worried about reverseing a sysadmin.. I wouldn't be doing this kind of routing in linux. |
| 23:25.41 | Felagund | and keep an analong line for sending faxes |
| 23:25.44 | ahammond | hardwire: so you've got a pair of routers already? |
| 23:25.53 | hardwire | ahammond: two VIA C3's |
| 23:25.58 | hardwire | with 6 10/100's |
| 23:25.59 | Felagund | I'm annoyed |
| 23:26.07 | Felagund | becuase if this thing blows up I'm going to get blamed for it |
| 23:26.09 | hardwire | that becomes implemented later. |
| 23:26.11 | ahammond | hardwire: that's a good enough setup |
| 23:26.26 | hardwire | going to use HA inbetween the two. |
| 23:26.36 | ahammond | keepalived? |
| 23:26.39 | X-Rob | Felagund - Voip and satellite _do not mix_ |
| 23:26.40 | hardwire | but I found that I could probably use just routing to work well enough |
| 23:26.44 | hypa7ia | Felagund: you probably want to be getting digium or another consulting firm involved relatively soon |
| 23:26.46 | hardwire | ahammond: no. heartbeat |
| 23:26.50 | X-Rob | I don't know how much more I can emphasise that. |
| 23:27.01 | hardwire | X-Rob: thats cause you smoke shit |
| 23:27.05 | Felagund | hypa7ia we are a consulting firm.. and these guys are hard set on doing it themselves |
| 23:27.08 | X-Rob | Not for years. |
| 23:27.10 | hardwire | then again it kinda depends on what kind of sat uplink |
| 23:27.14 | ahammond | hardwire: check out keepalived. it's a very elegant implementation of VRRP |
| 23:27.15 | hardwire | we have a 512mbps sync uplink |
| 23:27.15 | X-Rob | hardwire - satellite adds 200mec |
| 23:27.21 | hardwire | X-Rob: no.. it adds 600ms. |
| 23:27.24 | ahammond | hardwire: although heartbeat is pretty good too |
| 23:27.26 | hardwire | and we do it just fione |
| 23:27.28 | Felagund | it's T1 equivalent low latency satellite service |
| 23:27.30 | X-Rob | ick |
| 23:27.38 | hardwire | infact.. we do it just as well as the islands primary telco |
| 23:27.41 | nassy | Felagund: how about setting it up in a small remote office and seeing how it goes |
| 23:27.41 | X-Rob | even worse. |
| 23:27.42 | hardwire | since they have to use sat as well |
| 23:27.47 | Felagund | nassy |
| 23:28.02 | Felagund | I'd rather this guy had at least a year of experience with asterisk before setting up any production systems by himself |
| 23:28.06 | hardwire | ahammond: I have just recentl;y heard of VRRP |
| 23:28.24 | hypa7ia | nassy: i couldn;t tell if those phones do sip or not from the site... but h.323 is not really a road you want to go down |
| 23:28.26 | nassy | Felagund: then at least if it goes down it would not be so bad. |
| 23:28.35 | X-Rob | hardwire - ok. Voip and satellite do not mix unless there's no other possible alternative. |
| 23:28.44 | Felagund | nassy well I want him to know how to fix problems with it |
| 23:28.44 | X-Rob | is that better? 8) |
| 23:28.48 | hardwire | X-Rob: but... they still mix :) |
| 23:28.50 | Felagund | the other issue is |
| 23:28.57 | hardwire | just not in a public network .. as available telco routes. |
| 23:28.58 | Felagund | this rural site that is over the satellite link |
| 23:29.03 | Felagund | they can't get ISDN there |
| 23:29.04 | hardwire | they only work as endpoints. |
| 23:29.13 | Felagund | so the only way to connect the asterisk box locally is analog lines |
| 23:29.20 | hardwire | Felagund: wheres the site? |
| 23:29.21 | hypa7ia | Felagund: that's okay |
| 23:29.29 | Felagund | I've heard there are echo issues with that |
| 23:29.37 | hardwire | not if you are smart :) |
| 23:29.50 | hypa7ia | not really... there are echo issues possible with PRI too... you just need to deal witht hem |
| 23:29.54 | *** join/#asterisk philm (n=a@r43h15.res.gatech.edu) |
| 23:29.54 | Felagund | what if you're like this guy who is setting stuff up for the first time |
| 23:30.12 | Felagund | are they easy to fix or do you have to know what you're doing |
| 23:30.15 | fugitivo | Felagund: you'll maybe have echo problems if the analog line is too old or bad wire |
| 23:30.21 | X-Rob | we'll be seeing him in here in the next couple of weeks, I guess. |
| 23:30.25 | hypa7ia | well, itmight work, or it might not, but there is support is avalable |
| 23:30.50 | nassy | Felagund: thats what i plan to do. install it in an office and after about 6 months to a year consider replacing our toshiba box. i need to know how to fix problems fast for the main office and add features fast. problem is the company is growing and just out grew the current phone system so we keep buying more and more things for it making it more difficult to switch. |
| 23:31.05 | nassy | hypa7ia: :-( thanks. |
| 23:31.33 | *** join/#asterisk folsson (n=filip@lund-meje-sr0-vl101-249.perspektivbredband.net) |
| 23:32.05 | hypa7ia | nassy: have you gotten your hands on a copy of the starfish book? |
| 23:32.47 | nassy | yep, read through it once thus far. need to reread for the concepts to sink in. still struggling with some concepts like TDM etc |
| 23:33.30 | hypa7ia | yeah... if you're coming from a data background this voice stuff is all wacky |
| 23:33.44 | hypa7ia | better to come froma knowing-nothing background and have to learn it all |
| 23:33.48 | hypa7ia | i mean... not |
| 23:33.49 | hypa7ia | :-) |
| 23:33.53 | *** join/#asterisk Guggemand (i=Guggeman@tester2.har-tabt.dk) |
| 23:34.00 | nassy | :-) |
| 23:36.52 | *** join/#asterisk CodeBot (i=stjohn@212.68.221.41.brutele.be) |
| 23:38.06 | nassy | there is a wiki that is helping a lot |
| 23:38.12 | hardwire | ok |
| 23:38.18 | hardwire | quagga can't handle tables like it says it can |
| 23:38.23 | hypa7ia | nassy: the voip-info one? |
| 23:38.47 | hanchi | X-Rob, mpg123-0.59s-513.x86_64.rpm worked for the 64 bit machine |
| 23:39.01 | nassy | that one yeah, but for the terms i dont understand there is a wiki i usually get to via google. i should really bookmark it |
| 23:39.12 | hanchi | i'll find out real soon if it works with * |
| 23:39.16 | ahammond | hardwire: well, you don't need a full on routing solution for simple multi-linked. :) |
| 23:39.28 | hardwire | I do.. :) |
| 23:39.37 | hardwire | cause I don't want to add all this cruft to /etc/network/interfaces |
| 23:39.40 | hardwire | in order |
| 23:40.00 | hypa7ia | nassy: there's a pretty good selection of stuff on old-school telecom on wikipedia |
| 23:40.06 | ahammond | hardwire: well... you'll find that running a routing solution is actually more work than just static routing. |
| 23:40.06 | drumkilla | hypa7ia: !!!!!!!!!!!!! |
| 23:40.13 | nassy | yeah, thats the one |
| 23:40.21 | hardwire | ahammond: I know.. but this is primarily just static routing |
| 23:40.27 | hypa7ia | wikipedia == love |
| 23:40.27 | hardwire | therefore I am using just zebrad |
| 23:40.31 | hypa7ia | drumkilla: !!!!!!! |
| 23:40.41 | ahammond | hardwire: that's like squashing an ant with a sledgehammer. |
| 23:41.04 | ahammond | you've also added all the extra effort of maintaining zebra to your setup. :) |
| 23:41.30 | hardwire | ahammond: for now.. |
| 23:41.30 | drumkilla | I'm out of here! |
| 23:41.35 | hypa7ia | awwww |
| 23:41.40 | hardwire | ahammond: production.. this will all be on the servers.. doing something completely different :) |
| 23:41.43 | hardwire | so I suppose I shouldn't worry. |
| 23:42.12 | *** join/#asterisk CodeBot (i=stjohn@212.68.221.41.brutele.be) |
| 23:42.30 | hypa7ia | ahammond in indeed correct |
| 23:42.38 | hypa7ia | s/in/is |
| 23:44.55 | *** part/#asterisk obsidian-studios (n=obsidian@c-66-177-188-197.hsd1.fl.comcast.net) |
| 23:45.50 | *** join/#asterisk UndiFineD (n=keijon@213.160.199.45) |
| 23:47.02 | UndiFineD | I was wondering about the stresstest |
| 23:47.17 | UndiFineD | what is the best asterisk can do per hour ? |
| 23:47.18 | *** join/#asterisk CodeBot (i=stjohn@212.68.221.41.brutele.be) |
| 23:49.30 | X-Rob | UndiFineD - 4.3 millfops. |
| 23:50.22 | UndiFineD | I meant in phone connections |
| 23:50.49 | *** join/#asterisk dos000 (n=dos000@i216-58-64-83.cybersurf.com) |
| 23:51.05 | hypa7ia | nassy: newton's telecom dictionary |
| 23:51.05 | dos000 | howdy |
| 23:51.09 | hypa7ia | check that out |
| 23:51.32 | nassy | thanks |
| 23:51.55 | hardwire | ahammond: so its working |
| 23:52.05 | ahammond | hardwire: cool |
| 23:52.07 | hardwire | even have a dest ip routed via the non default link |
| 23:52.09 | *** join/#asterisk dstr (n=d@c-e73570d5.020-104-73746f25.cust.bredbandsbolaget.se) |
| 23:52.25 | ahammond | hardwire: make sure you're careful not to advertise routes you can't route. :) |
| 23:52.57 | hardwire | not really there yet. |
| 23:53.05 | hardwire | but I have done dynamic routing |
| 23:53.18 | dos000 | i have been trying for a while to figure what i need to connect a modem to a digium card whithout passing by a CO. or if i can even get two cards back 2 back and pass modem stuff on top. |
| 23:53.50 | dos000 | i meant pass v22 like protocol on top |
| 23:53.52 | UndiFineD | ok let me ask in another way, can asterisk do 100.000+ IVR connections per hour ? |
| 23:53.58 | dstr | i've just set up asterisk for the first time, and everything seems to be working - except that there is no sound when i call 1000 for the demo. |
| 23:54.48 | dos000 | dstr, check the logs |
| 23:56.19 | hardwire | now to see if the config passes the reboot test. |
| 23:56.26 | *** join/#asterisk jerlique (n=jerlique@lnk59.adl3.adsl.esc.net.au) |
| 23:57.15 | dstr | dos000: cant find anything in the logs. the console says: |
| 23:57.16 | dstr | <PROTECTED> |
| 23:57.16 | dstr | <PROTECTED> |
| 23:57.31 | hardwire | dstr: you are on shakey ground now man. |
| 23:57.44 | hardwire | thats 3 lines.. plus a bit of a line wrap. |
| 23:57.55 | hardwire | don't you know to use pastebin? |
| 23:58.02 | dstr | thats four lines... |
| 23:58.05 | dstr | five... |
| 23:58.07 | hardwire | heh |
| 23:58.18 | dstr | :) |
| 23:59.35 | twisted | heh |
| 23:59.42 | twisted | Hurricane wilma and tropical storm alpha |
| 23:59.50 | twisted | looks like we've rolled through all of the names for the year |