| 00:00.34 | pauldy | have to look it may be and that would explain a lot if it was |
| 00:00.51 | pauldy | I thought it was 263 |
| 00:02.30 | Dr_Ray | from what I've read (in the last 10 minutes :) is that h.263 is now a legacy thing, and h.264 is the current one |
| 00:03.11 | pauldy | which I don't think asterisk supports at lest for 1.0.9 |
| 00:04.12 | Dr_Ray | well, I'd not do h264 unless there was a set-top box |
| 00:04.26 | Dr_Ray | but there is an h.264 gpl server |
| 00:04.56 | acidfoo | x264 ? |
| 00:05.04 | pauldy | wonder if there are any issues with the 264 codec |
| 00:05.12 | pauldy | aka mp3 |
| 00:05.17 | pauldy | or lik emp3 |
| 00:05.40 | pauldy | hate to build it and have it become property o digium |
| 00:06.23 | Dr_Ray | x264 |
| 00:06.53 | Dr_Ray | the question is do you even need to bother making asterisk do it |
| 00:07.38 | *** join/#asterisk froguz (n=froguz@174-43-112.adsl.terra.cl) |
| 00:07.56 | froguz | hi |
| 00:08.32 | pooh_ | hi |
| 00:08.36 | acidfoo | Dr_Ray: are you doctor? |
| 00:08.47 | Dr_Ray | no, I'm not a ray either |
| 00:08.49 | pooh_ | nephew of Dr. Phil <grin> |
| 00:08.57 | acidfoo | ;)\ |
| 00:09.00 | Dr_Ray | we are both from texas |
| 00:09.03 | Dr_Ray | but I have hair |
| 00:09.04 | froguz | i need some explanation about configuring my own Do Not Disturb |
| 00:09.09 | froguz | any URI? |
| 00:09.20 | pauldy | www.voip-info.org |
| 00:09.31 | froguz | nop, nothing there |
| 00:09.54 | pooh_ | database put DND CALLERID 1 => stdexten => dbget DND CALLERID => jump n+101 ;-) |
| 00:09.55 | froguz | it just says DND comes embeded in asterisk, but it doesn't work for me |
| 00:10.14 | acidfoo | pauldy: erm, isnt aka mp4 ? |
| 00:10.15 | froguz | also, i wanted to make my own DND with my configuration |
| 00:10.16 | pauldy | thats the site I usedto setup my dnd |
| 00:10.30 | froguz | really?mmm |
| 00:10.43 | pauldy | acidfoo, I meant the licensing issues thats why a retyped like mp3 cause I saw the potential confussion |
| 00:10.53 | acidfoo | ok pauldy ;) |
| 00:11.04 | *** join/#asterisk twisted[home] (n=twisted@asterisk/friend-and-developer/pdpc.professional.twisted) |
| 00:11.04 | *** mode/#asterisk [+o twisted[home]] by ChanServ |
| 00:11.12 | froguz | pauldy: are you using the embeded dnd? |
| 00:11.33 | pauldy | froguz, probably not |
| 00:11.48 | pauldy | don't recognize that |
| 00:12.22 | froguz | aha, i will search deep on voip-info |
| 00:14.06 | pauldy | looks like by default you can simply dial *78 and *79 |
| 00:14.13 | *** join/#asterisk tholo (n=tholo@dsl001-136-136.lax1.dsl.speakeasy.net) |
| 00:14.23 | pauldy | looks much easier than what I did |
| 00:15.59 | *** join/#asterisk Qwell (n=chatzill@dsl001-136-136.lax1.dsl.speakeasy.net) |
| 00:16.50 | froguz | and what did yo did?? |
| 00:17.03 | froguz | jejee |
| 00:18.16 | pauldy | created an extension to enable or disable the DND based on if it was set or not and play the appropriate anouncement |
| 00:21.09 | froguz | that's what i would like to do |
| 00:22.56 | *** join/#asterisk oej (n=Olle@dsl001-136-136.lax1.dsl.speakeasy.net) |
| 00:23.06 | *** join/#asterisk Koshatul (n=evangeli@ip157.net65.ipnetworks.net.au) |
| 00:25.14 | Dr_Ray | maybe I'm just better off using mythtv frontend |
| 00:25.23 | *** join/#asterisk Qwell (n=chatzill@dsl001-136-136.lax1.dsl.speakeasy.net) |
| 00:26.28 | phpboy | Qwell!!!! :D |
| 00:26.35 | Qwell | hi |
| 00:27.09 | hardwire | Qwell: Qwell Qwell Qwell Qwell Qwell Qwell Qwell Qwell |
| 00:27.23 | hardwire | !: ! ! ! ! ! ! ! ! |
| 00:30.11 | lancey | :)))) |
| 00:30.24 | lancey | Dr_Ray |
| 00:30.27 | lancey | speaking of set top boxes |
| 00:30.41 | lancey | have you heard of one capable of being used as a cable modem also? |
| 00:32.10 | *** join/#asterisk marc324 (n=marc3234@206-248-159-56.dsl.teksavvy.com) |
| 00:32.39 | marc324 | @docs |
| 00:32.42 | marc324 | ~docs |
| 00:32.44 | 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 |
| 00:34.53 | *** join/#asterisk tholo (n=tholo@dsl001-136-136.lax1.dsl.speakeasy.net) |
| 00:35.43 | *** join/#asterisk MikeJ[Laptop] (n=ircatjer@d14-69-8-30.try.wideopenwest.com) |
| 00:38.24 | ManxPower | ~mailinglist |
| 00:38.25 | jbot | extra, extra, read all about it, mailinglist is Search Asterisk mailing lists by prepending site:lists.digium.com to your Google search. Browse the mailing list archive at http://lists.digium.com/ |
| 00:44.42 | *** join/#asterisk nomazda (i=nyyankee@user-0c6tnqf.cable.mindspring.com) |
| 00:49.49 | jlewis | is the * wireless available to *con attendes? |
| 00:51.11 | *** join/#asterisk cgcorea (n=cgcorea@63.245.14.194) |
| 00:52.37 | *** join/#asterisk NoRemorse (n=axel@202.161.68.2) |
| 00:52.40 | *** join/#asterisk devonst17_ (n=devonst1@dsl092-032-215.lax1.dsl.speakeasy.net) |
| 00:52.47 | NoRemorse | !seen justinnn |
| 00:53.00 | NoRemorse | no seen bot? |
| 00:53.10 | *** join/#asterisk cgcorea (n=cgcorea@63.245.14.194) |
| 00:55.17 | Qwell | ~seen justinnn |
| 00:55.22 | jbot | justinnn <~dsf@solid.mpa.net.au> was last seen on IRC in channel #asterisk, 237d 17h 36m 49s ago, saying: 'how do i convert a .gsm to .wav ?'. |
| 00:55.29 | Qwell | jlewis: yes |
| 00:56.59 | jlewis | where do we get the key?....I'm barely online on an open ap |
| 00:57.13 | Qwell | got me |
| 01:02.33 | jlewis | <PROTECTED> |
| 01:03.48 | *** join/#asterisk veto (i=mdkuser@cpe-66-69-38-192.satx.res.rr.com) |
| 01:06.47 | NoRemorse | ~seen justinn |
| 01:06.49 | jbot | justinn <~justinm@solid.mpa.net.au> was last seen on IRC in channel #asterisk, 495d 23h 30m 8s ago, saying: 'theres nothing in that dir'. |
| 01:06.49 | NoRemorse | ty |
| 01:06.54 | NoRemorse | ~seen justinnnn |
| 01:06.55 | jbot | justinnnn <~dsf@solid.mpa.net.au> was last seen on IRC in channel #asterisk, 203d 17h 50m 7s ago, saying: 'anyone ???'. |
| 01:07.03 | NoRemorse | not looking good lol |
| 01:07.06 | *** join/#asterisk oej (n=Olle@dsl001-136-136.lax1.dsl.speakeasy.net) |
| 01:07.14 | NoRemorse | ~seen justinnnnn |
| 01:07.15 | jbot | justinnnnn <n=justinnn@61.95.68.85> was last seen on IRC in channel #asterisk, 19h 42m 5s ago, saying: 'hey ppls :)'. |
| 01:07.19 | NoRemorse | thats better |
| 01:07.43 | froguz | pauldy, is that the way you did it? : http://lists.digium.com/pipermail/asterisk-users/2003-July/016824.html |
| 01:14.55 | *** join/#asterisk [Jedi] (n=hhgds4@154.Red-217-127-168.staticIP.rima-tde.net) |
| 01:14.57 | [Jedi] | Hello |
| 01:15.19 | ender | I'm having some difficulty w/ Asterisk 'locking up' somewhat when somebody hangs up in voicemail. I was once told it has something to do w/ using 'switch =>' in my dial plan. |
| 01:17.57 | *** join/#asterisk Qwell (n=chatzill@dsl001-136-136.lax1.dsl.speakeasy.net) |
| 01:18.15 | [Jedi] | How can I put in a variable the result of the execution of another application? |
| 01:25.06 | marc324 | what db is better for asterisk? postgresql or mysql? |
| 01:25.19 | Qwell | marc324: whichever one works best for you... |
| 01:26.08 | marc324 | ne docs on asterisk realtime? |
| 01:26.35 | Qwell | http://www.voip-info.org/tiki-index.php?page=Asterisk+RealTime |
| 01:27.13 | oej | There is a readme |
| 01:27.26 | Qwell | oej: hey |
| 01:27.49 | tholo | use whichever oej doesn't recommend. Just for fun. |
| 01:27.53 | lancey | :) |
| 01:28.11 | lancey`away | bye guyz |
| 01:40.03 | *** join/#asterisk kingtux (n=kingtux@pool-151-196-44-240.balt.east.verizon.net) |
| 01:44.00 | kingtux | anyone chatting |
| 01:44.01 | kingtux | ?? |
| 01:45.32 | *** join/#asterisk Corydon76-home (i=gray@pdpc/supporter/sustaining/Corydon76-home) |
| 01:45.39 | tholo | No. |
| 01:46.57 | *** part/#asterisk kingtux (n=kingtux@pool-151-196-44-240.balt.east.verizon.net) |
| 01:48.45 | *** join/#asterisk DeFi (n=DeFi@ip68-6-40-245.sb.sd.cox.net) |
| 01:55.37 | *** join/#asterisk MikeJ[Laptop] (n=ircatjer@d14-69-8-30.try.wideopenwest.com) |
| 02:00.16 | *** join/#asterisk cio (n=na@adsl-072-149-159-016.sip.bhm.bellsouth.net) |
| 02:00.17 | cio | Anyone here run * on debian sarge with the 2.6 kernel and compile from cvs or the 1.09 source? |
| 02:01.16 | *** join/#asterisk gniretar (n=gniretar@66-227-204-109.dhcp.bycy.mi.charter.com) |
| 02:01.46 | gniretar | can anyone recommend to me a good but inexpensive FXS card? |
| 02:02.01 | cio | X100P, $9.00 on ebay. |
| 02:02.04 | Qwell | gniretar: the digium tdm400p |
| 02:02.10 | Qwell | cio: that is not fxs |
| 02:02.14 | cio | It has one of both - |
| 02:02.20 | Qwell | cio: no, it does not |
| 02:02.23 | cio | You sure? |
| 02:02.25 | Qwell | positive |
| 02:02.46 | cio | k. |
| 02:02.56 | cio | Qwell: You run kernel 2.6 or 2.4? |
| 02:03.03 | Qwell | 2.6 |
| 02:03.12 | cio | Custom or distro built? |
| 02:03.24 | Qwell | custom |
| 02:03.45 | cio | When you compile zaptel, are you doing a 'make linux26' or just 'make'? |
| 02:03.45 | gniretar | all real mean run custom kernels |
| 02:04.04 | JamesDotCom | you dont need to run make linux26 for zaptel |
| 02:04.10 | cio | hrmm... |
| 02:04.11 | gniretar | ya |
| 02:04.17 | JamesDotCom | make; make install will be fine |
| 02:04.19 | gniretar | i thought that make would autodetect or somthing |
| 02:04.21 | Juggie | Qwell, tdm400p can be fxo or fxs depending on config |
| 02:04.25 | Juggie | it should auto detect 2.6 yes |
| 02:04.30 | gniretar | cuz make and make linux26 do the same in my linux26 boxes |
| 02:04.36 | clyrrad | How can I get my * CDR records to record the start and end of a call with its duration, but not record individual extensions that were pressed and all activity that happend during the call? There must be a config file or something that I can use to control the behaviour of the CDRs anyone know where it is? |
| 02:04.37 | cio | My * from 1.09 and cvs just core dumps no matter how I make it on my 2.6 (debian sarge) box. |
| 02:04.40 | Juggie | you may also want to do make udev |
| 02:04.40 | Qwell | Juggie: tdm yeah...not the x100p though |
| 02:04.46 | cio | I thought it was zaptel related. |
| 02:04.49 | Juggie | if your system is running udev |
| 02:04.49 | cio | Ahh! |
| 02:05.05 | cio | That may be it. Would that cause an asterisk core dump? |
| 02:05.10 | cio | Or just errors? |
| 02:05.16 | Juggie | just errors |
| 02:05.22 | gniretar | clyrrad: this i would like to know as well. From what i know it isnt possible. CDR records are CDR records. |
| 02:05.25 | Juggie | do you have a reproducable core dump? |
| 02:05.29 | cio | Yep. |
| 02:05.38 | Juggie | stable or head? |
| 02:05.40 | gniretar | cio: your building from source, yes? |
| 02:06.00 | cio | I just install a base debian system, the kernel source, make a ln -s /usr/src/linux whatever and make zaptel, libpri, asterisk, run asterisk -vvvgc, core dump. Every time. |
| 02:06.08 | clyrrad | gniretar.... but I would think there must be a way, for billing purposes all the extra data that is recorded is simply a waste of DB Processing time and space |
| 02:06.19 | cio | I get the same thing from 1.09 tar gz files from asterisk.org and cvs. |
| 02:06.25 | cio | Something weird with debian. |
| 02:06.29 | gniretar | cio: whenever i build from source it tries to load a module for which ther are no extensions. |
| 02:06.31 | gniretar | and dies |
| 02:06.37 | gniretar | cio: please pastebin the error |
| 02:06.40 | cio | hrmm. |
| 02:06.44 | SwK | cio can you get us a backtrace of the dump? |
| 02:07.02 | cio | I'm reinstalling the os now and will try to compile again when done, about 30 minutes. |
| 02:07.08 | gniretar | clyrrad: Well, i would tell you that even a heavily loaded Asterisk server would barely load a MySQL server at all. |
| 02:07.47 | gniretar | clyrrad: the CDR records, even with the extra info are quite small. MySQL would have to be taking hundreds per second (depending on hardware) for it to bog down. |
| 02:07.51 | cio | I just started using sqlite for my cdr and cdr reports via php, sqlite is great for this purpose. |
| 02:08.12 | gniretar | cio: u wite your own PHP or use asterisk-stat? |
| 02:08.19 | gniretar | cio: I use Debian, i have for years |
| 02:08.27 | gniretar | there is nothing hat would cause this wierd bug |
| 02:08.32 | cio | Write my own. Real men build their own reports.. heh |
| 02:08.40 | gniretar | cie: before you reload please, for out own education, pastebin the error |
| 02:08.45 | gniretar | cio* |
| 02:09.01 | gniretar | cio: lol, well when i'm at work i dont have time to write in-house PHP for everything |
| 02:09.02 | cio | I'm already through the installer, the packages are installing now - sorry - It will pop up again ... |
| 02:09.34 | clyrrad | gniretar.... putting bogged down aside, I agree with your first statement it would be nice to control what gets recorded and what does not get recorded, with out modifying the source, looking for a config file or something of the like that woudl allow us to do that |
| 02:09.42 | cio | gniretar: I agree, using asterisk-stat would probably be easier, but the reports are of such a custom nature, I doubt the customer would have it any other way. Plus it's hourly work. |
| 02:09.47 | *** join/#asterisk Koshatul (n=evangeli@ip157.net65.ipnetworks.net.au) |
| 02:09.53 | gniretar | though, i must admit, in site of my love for Debian i bought a PowerBook insead of a PC |
| 02:10.02 | cio | I did the same thing! |
| 02:10.11 | gniretar | cio: I'm hourly as well but my boss is a Nazi about ineffficiant use of time |
| 02:10.12 | cio | I also have debian installed on my OQO. |
| 02:10.14 | *** join/#asterisk shido6 (n=shido6@d221-68-210.commercial.cgocable.net) |
| 02:10.16 | gniretar | lol |
| 02:10.22 | gniretar | well, i have X11 and Fink on my PowerBook |
| 02:10.27 | gniretar | so i'm on xChat |
| 02:10.29 | gniretar | on my mac |
| 02:10.32 | gniretar | life if good |
| 02:10.34 | gniretar | is* |
| 02:10.35 | JamesDotCom | werd |
| 02:10.36 | cio | heh... |
| 02:10.37 | JamesDotCom | osx4life |
| 02:10.58 | gniretar | though Fink's compile of Ethereal isnt working |
| 02:11.00 | gniretar | which is sad |
| 02:11.05 | gniretar | cuz i love my Ethereal |
| 02:11.05 | JamesDotCom | argh! |
| 02:11.13 | JamesDotCom | i just came across that problem yesterday |
| 02:11.15 | cio | gniretar: They handed me a stack of reports, said they wanted them to pump out of their new phone system, I said 120 hours@75.00/hr, they said great... heh |
| 02:11.48 | gniretar | cio: Man i hate you. I am the one and only tech for a small buissness, CCNP certified and i only make $10 |
| 02:11.57 | marc324 | what is "dsn" ? |
| 02:12.00 | cio | $10 for what? |
| 02:12.09 | gniretar | cio: though i am 17 so i'm not worth as much as someone with experience |
| 02:12.14 | gniretar | $10 per hour! |
| 02:12.20 | cio | You should be able to get more than that. |
| 02:12.23 | cio | Even at 17. |
| 02:12.26 | Qwell | with a ccnp? |
| 02:12.31 | Qwell | way more than $10 |
| 02:12.37 | gniretar | cio: yes but noone wants to put a 17 year-old in chanre og mission critical things |
| 02:12.37 | marc324 | is dsn=database name? |
| 02:12.41 | *** part/#asterisk NoRemorse (n=axel@202.161.68.2) |
| 02:12.44 | cio | I'm 35, have an established business, and pull $125-150/hr for most work, $75 for programming. |
| 02:13.00 | gniretar | Quall, not only hat but i got my CCNP at 16. I am one of the yourgest in the world |
| 02:13.19 | cio | Go into business for yourself as soon as you turn 18. |
| 02:13.31 | gniretar | i was thinking of that |
| 02:13.40 | gniretar | just as consulting or somthing |
| 02:13.52 | gniretar | on the side |
| 02:13.55 | gniretar | and try to grow it |
| 02:13.57 | gniretar | its hard though |
| 02:14.01 | gniretar | to get a customer base |
| 02:14.39 | cio | They are out there, just gotta find somebody that can sell while you produce, make it a symbiotic relationship and you'll both do well. |
| 02:14.47 | gniretar | ya |
| 02:14.48 | cio | That's how I started. |
| 02:14.58 | gniretar | well, its a funny story how my boss got into buissness |
| 02:15.17 | gniretar | he had a 2 man web developing firm |
| 02:15.26 | gniretar | build a guge page to sell somthing |
| 02:15.28 | gniretar | huge* |
| 02:15.36 | gniretar | and got screwed my the guy he was making to for |
| 02:15.52 | gniretar | so he started selling the things for whch the page was mode to sell |
| 02:16.17 | gniretar | and is now driving a Porche and we just got a 54,000 sqft building |
| 02:16.34 | gniretar | i wish i could pull that off someday |
| 02:17.34 | cio | Expensive cars aren't a sign of success. |
| 02:17.50 | gniretar | no, the point was they they could easily afford it |
| 02:18.14 | cio | Afford is relative to want and need. |
| 02:18.30 | cio | I pay cash for non-appreciating assets like cars when I can. |
| 02:19.06 | gniretar | cio: that is a good idea. no sense paying interest on somthing that has no return |
| 02:19.42 | cio | I'd rather drive a 10 year old Yugo than pay interest on an auto. |
| 02:19.51 | cio | When I was in my early 20's, I was up to my ears in debt. |
| 02:19.55 | *** join/#asterisk sgorilla (n=tlp@cpe-24-160-119-179.houston.res.rr.com) |
| 02:19.56 | cio | I got rid of it and change my thinking entirely. |
| 02:20.09 | sgorilla | is there a simple way to call two phone numbers at once |
| 02:20.13 | cio | Took me 5 years to get rid of debt that took me about 5 seconds to incure. |
| 02:20.17 | sgorilla | and just bridge over to the one that picks up first |
| 02:20.20 | JamesDotCom | haha |
| 02:20.34 | JamesDotCom | cio, amen, i'm just finishing paying off my debt |
| 02:20.36 | sgorilla | cio: 5 seconds? |
| 02:20.44 | sgorilla | i am in debt, im just starting to pay it off |
| 02:20.54 | cio | Heh - it felt like that when I was bailing myself out ... |
| 02:21.04 | cio | Never again, though. |
| 02:21.21 | sgorilla | what did you do? |
| 02:22.04 | *** join/#asterisk MrBelvedr (n=tt@ip70-187-237-193.dc.dc.cox.net) |
| 02:22.12 | *** join/#asterisk jaike (n=a@203.131.137.76) |
| 02:22.24 | jaike | any commpartners clients here? |
| 02:22.32 | cio | Got a good job, bought a fat car, got a raise, bought a nice house, then WHAM! In debt up to my ass. |
| 02:22.32 | ender | I'm having some difficulty w/ Asterisk 'locking up' somewhat when somebody hangs up in voicemail. I was once told it has something to do w/ using 'switch =>' in my dial plan. |
| 02:23.18 | gniretar | so i'm thinking that i will ake myself a Asterisk box to use to connect to Vonage then set up an IAX trunk to the Asterisk box at work so i can stop using my vonage line for work calls. |
| 02:24.02 | *** join/#asterisk flenders (n=fserto@61.8.29.101) |
| 02:24.08 | gniretar | and i can use my Vonage line from anywhere with a laptop IAX client (i dont wanna use a Sip sptphone, they are a pain in the a$$) |
| 02:24.17 | gniretar | softphone* |
| 02:24.25 | jaike | any downsides to using sip? compared to iax? |
| 02:24.41 | ender | jaike: in what context? |
| 02:25.03 | jaike | ability to handle jitter |
| 02:25.08 | jaike | call quality wise |
| 02:25.21 | gniretar | jaike: IAX is all arround newer and better |
| 02:25.34 | *** part/#asterisk cgcorea (n=cgcorea@63.245.14.194) |
| 02:25.41 | gniretar | jiake: with IAX you can use it from behind a NAT firewall |
| 02:25.54 | Qwell | You can use sip behind nat too |
| 02:26.06 | gniretar | jiake if you have control connection have have voice, period. None of this stupid poking holed in the firewall or any of that |
| 02:26.07 | jaike | were currently testing a new provider that doesnt offer iax, only sip |
| 02:26.13 | loud | most people who say that are not familiar with nat/pat/ and firewalls at all. |
| 02:26.43 | gniretar | Qwell&loud, i an CCNP, i understand exactly how they work. |
| 02:27.19 | gniretar | Qwell&loud, you must poke holed in the firewall to get it to work. that is something i dont wanna do |
| 02:27.27 | gniretar | holes* |
| 02:28.12 | gniretar | SIP was good for its day, IAX is the future |
| 02:28.59 | jaike | my servers arent behind firewalls...any other downsides with sip? |
| 02:29.07 | gniretar | www.voip-info.org/wikki-IAX+versus+SIP |
| 02:29.21 | gniretar | jaike but are your clients? |
| 02:29.32 | jaike | tnx gniretar..will read on that |
| 02:30.02 | gniretar | jaike: yw |
| 02:30.04 | jaike | were using sip phones..polycom |
| 02:30.55 | cio | I have two IP301's I'll sell ya on the cheap. |
| 02:31.31 | gniretar | jaike: we use aastra phones at work and i like them. |
| 02:31.42 | *** join/#asterisk twisted (n=twisted@asterisk/friend-and-developer/pdpc.professional.twisted) |
| 02:31.42 | *** mode/#asterisk [+o twisted] by ChanServ |
| 02:31.53 | jaike | cio: were looking for like 40+ |
| 02:32.04 | cio | I have two, now you only need 38! :p |
| 02:32.09 | jaike | :) |
| 02:32.12 | cio | $100 each, new condition. |
| 02:32.15 | cio | + shipping |
| 02:32.38 | *** join/#asterisk dabigshiznizzle (n=dabigshi@dsl-cw-65-171-138-87.giant-broadband.com) |
| 02:39.03 | *** join/#asterisk MikeJ[Laptop] (n=ircatjer@d14-69-8-30.try.wideopenwest.com) |
| 02:41.19 | gniretar | JamesDotCom: did you ever resolve the Ethereal problem? |
| 02:42.04 | loud | Does anyone have the 64_bit version of codec_g723.so ? |
| 02:42.53 | clyrrad | Anyone know if its possible to configure what CDR actually record? |
| 02:43.39 | Juggie | lourd, that is a codec which requiresl icensing |
| 02:43.49 | Juggie | and therefor not supported in here as it is not sold by digium |
| 02:43.59 | Juggie | there is however a 64bit version of g729 |
| 02:44.08 | flenders | hi, my asterisk is not handling incoming calls as it should do. |
| 02:44.12 | loud | already have it. |
| 02:44.14 | *** join/#asterisk _daver_ (n=daver@ns1.tmok.com) |
| 02:44.17 | marc324 | what's dsn? |
| 02:44.21 | flenders | I've posted the whole description on patebin |
| 02:44.25 | flenders | http://pastebin.ca/25290 |
| 02:44.30 | flenders | can I have some help? |
| 02:44.36 | flenders | please |
| 02:44.41 | flenders | :o) |
| 02:46.37 | ender | flenders: what is the Zap connected to? |
| 02:46.47 | flenders | no zap |
| 02:46.54 | flenders | only SIP |
| 02:46.56 | ender | flenders: I had your issue w/ all calls because I was using the wrong Zap protocal. |
| 02:47.09 | ender | TRUNK => Zap/g2 |
| 02:47.13 | ender | ghy do you have that? |
| 02:47.22 | flenders | I know... it was already there on the template |
| 02:47.26 | flenders | I forgot to take it off |
| 02:47.33 | flenders | it's not beeing used anywhere |
| 02:47.39 | ender | yeah, take it and the other trunk part off. |
| 02:48.03 | ender | flenders: set debugging level to like 4, dump it to the full log file and then paste that log file up there to see what happens when the call is answered. |
| 02:48.26 | flenders | ok |
| 02:49.12 | marc324 | how do you test if realtime DB connection works? |
| 02:51.20 | gniretar | night all and thanks for all the sage advice |
| 02:53.19 | *** join/#asterisk Koshatul (n=evangeli@ip157.net65.ipnetworks.net.au) |
| 02:54.41 | flenders | ender: http://pastebin.ca/25291 |
| 02:54.48 | flenders | have a look at the end of the post |
| 02:57.17 | flenders | could this be the reason: channel.c: Unable to find a codec translation path from g729 to slin |
| 02:58.26 | file[laptop] | your Asterisk can't transcode between G729 and ULAW |
| 02:58.56 | flenders | file: would this be the reason for me not beeing able to answer calls? |
| 02:59.11 | file[laptop] | probably |
| 02:59.34 | flenders | file: have you had a look at pastebin? |
| 02:59.37 | file[laptop] | yes |
| 02:59.51 | clyrrad | where can you set accountcode and userfield for use with CDRs? |
| 03:00.00 | flenders | should I then remove all allow=g729 from sip.conf? |
| 03:00.04 | file[laptop] | Unable to find a codec translation path from ulaw to g729 |
| 03:00.16 | file[laptop] | all you had to do was read that |
| 03:00.19 | file[laptop] | and yes, try removing g729 |
| 03:00.31 | flenders | file[laptop]: thanks, will try that |
| 03:00.44 | clyrrad | file? |
| 03:00.51 | file[laptop] | eh? |
| 03:00.57 | clyrrad | hahaha |
| 03:01.04 | lancey`away | shit |
| 03:01.11 | clyrrad | where can you set accountcode and userfield for use with CDRs? |
| 03:01.23 | file[laptop] | the dialplan? |
| 03:01.26 | file[laptop] | the .conf? |
| 03:01.29 | lancey | can't sleep :/ |
| 03:01.30 | file[laptop] | take your pick. |
| 03:01.40 | marc324 | is dsn(data source name) the name of the database in postgres? |
| 03:01.45 | clyrrad | Yes... .conf is what im looking for.... which one? |
| 03:02.02 | file[laptop] | depends on the technology... |
| 03:02.09 | file[laptop] | iax2 would be iax.conf, sip would be sip.conf |
| 03:02.11 | clyrrad | IAX2 |
| 03:02.21 | flenders | file[laptop]: thanks mate, it works now... |
| 03:02.38 | clyrrad | so you just say accountcode=foo in [general] or how do you use it properly? |
| 03:02.38 | flenders | I better try to understand these different protocols... |
| 03:03.02 | file[laptop] | clyrrad: set it for the user usually... |
| 03:03.12 | file[laptop] | I'm not going to tell you how to use it, because I can't read your mind |
| 03:03.26 | clyrrad | under the specific context? |
| 03:03.30 | lancey | marc324: dsn is odbc term |
| 03:03.45 | file[laptop] | clyrrad: yes... |
| 03:03.45 | lancey | from my point of view |
| 03:03.50 | file[laptop] | ya know, you should just try this stuff |
| 03:03.50 | lancey | though i don't use postgre |
| 03:04.17 | clyrrad | file thanks..... was i correct in my syntax? |
| 03:04.23 | lancey | clyrrad: you can also set it with application |
| 03:04.24 | file[laptop] | sure why not |
| 03:04.34 | lancey | right in your dialplan |
| 03:04.57 | ender | file[laptop]: was it you I was talking to about the switch => causing * to lockup when hanging up from voicemail? |
| 03:05.01 | *** join/#asterisk franck (n=franck@tikiwiki/franck) |
| 03:05.04 | franck | Hi all |
| 03:05.06 | file[laptop] | ender: yes |
| 03:05.14 | ender | file[laptop]: the problem showed up twice today ): |
| 03:05.23 | ender | file[laptop]: and now I can't dupe it agin. |
| 03:05.25 | clyrrad | lancey... I think I would be best to set it in iax.conf.... but that make me wonder about the security with that.... If you are going to be billing people, and they removed that line from their iax.conf they could make free calls could they not? |
| 03:05.27 | DannyF | lo folks |
| 03:05.30 | DannyF | hi file |
| 03:05.40 | file[laptop] | I had a theory... and I think I can tell you how to get over it |
| 03:05.41 | file[laptop] | DannyF: hi |
| 03:05.42 | lancey | clyrrad |
| 03:05.47 | lancey | if you're billing people |
| 03:05.50 | ender | file[laptop]: I'm all eyes. |
| 03:05.51 | *** join/#asterisk SplasPood (n=sp@brooklyn.paravolve.net) |
| 03:05.54 | lancey | they probably don't have access to that file :) |
| 03:06.06 | franck | I'm in Fiji and I have a tender for supplying (and supporting) a PBX with VoIP for our company (70 people). Anyone interested? |
| 03:06.09 | clyrrad | lancey... they do if their pbx accesses yours |
| 03:06.10 | file[laptop] | ender: I gotta find it |
| 03:06.12 | clyrrad | for VOIP |
| 03:06.18 | ender | file[laptop]: take your time. |
| 03:06.23 | lancey | clyrrad: yes, but it's in your pbx |
| 03:06.29 | lancey | you modify your iax.conf |
| 03:06.38 | lancey | it doesn't matter what they setup in theirs |
| 03:06.38 | franck | or where should I post it? Digium people? |
| 03:06.48 | DannyF | franck, asterisk-biz |
| 03:06.50 | clyrrad | Oh you saying you modify your IAX.CONF for each customer that connects throught you? |
| 03:06.53 | file[laptop] | ender: set iaxcompat=yes in [general] in iax.conf |
| 03:07.02 | franck | DannyF: it is a channel here? |
| 03:07.10 | ender | file[laptop]: ah ok. |
| 03:07.11 | DannyF | yes but mailing list is better |
| 03:07.26 | file[laptop] | ender: it'll cause those switch lookups to go into their own threads, and not block |
| 03:07.33 | franck | DannyF: pointer? |
| 03:07.37 | DannyF | just a sec |
| 03:08.03 | lancey | clyrrad: you are anyways |
| 03:08.09 | lancey | how do you give them access then? |
| 03:08.14 | DannyF | franck, http://www.digium.com/index.php?menu=mailing_list |
| 03:08.20 | franck | DannyF: thx |
| 03:08.25 | DannyF | everything you need and then some ,) |
| 03:09.30 | franck | DannyF: Well I need a PBX too ;) |
| 03:09.32 | clyrrad | lancey.... Well the giving them access part I have not done yet, the security part of it held me back, I actually have not yet configured how remote PBX's and IAX devices will connect to my * box as of yet, so what your saying is I would control all of that in IAX.CONF? is that correct? |
| 03:09.50 | DannyF | franck, plenty of folks on the bix list that can provide turnkey systems |
| 03:09.59 | *** join/#asterisk Ayano (n=Ayano_@adsl-70-245-190-90.dsl.spfdmo.swbell.net) |
| 03:10.03 | DannyF | biz* |
| 03:10.14 | lancey | clyrrad absolutely |
| 03:10.35 | *** part/#asterisk Ayano (n=Ayano_@adsl-70-245-190-90.dsl.spfdmo.swbell.net) |
| 03:10.36 | lancey | what do you think iax.conf is for? |
| 03:11.06 | clyrrad | lancey... that actually makes good sense... so if i had 21 customers i should ahve 21 entries for each customer in my IAX.conf file, and there I can put their accountcode or userid or whatever i need.... |
| 03:11.19 | clyrrad | Well........... i thought it was for outgoing connections to your VOIP provider |
| 03:11.27 | *** join/#asterisk rt (n=markv@c-67-180-32-90.hsd1.ca.comcast.net) |
| 03:11.36 | franck | DannyF: yes thanks for the info... I have posted it... |
| 03:11.43 | DannyF | ok ;) |
| 03:11.52 | rt | anyone here run asterisk on freebsd? |
| 03:11.54 | Katty | mew. |
| 03:11.54 | lancey | clyrrad exactly |
| 03:11.59 | lancey | rt: me does |
| 03:12.03 | lancey | *i do |
| 03:12.08 | lancey | my poor head |
| 03:12.19 | rt | should I install from ports, or download the 1.2 beta and compile that? |
| 03:12.23 | DannyF | lancey ;) |
| 03:12.25 | lancey | clyrrad: it's for outgoing, too |
| 03:12.26 | Katty | mew? |
| 03:12.28 | ender | file[laptop]: that would be very nice. |
| 03:12.31 | clyrrad | lancey... what were you saying exactly to? My first comment, my second one or both? |
| 03:12.33 | lancey | that's what's type=xxxx for |
| 03:12.37 | DannyF | Katty, moo ;) |
| 03:12.48 | *** join/#asterisk Micc (n=dotirc@c-24-19-175-112.hsd1.wa.comcast.net) |
| 03:12.56 | lancey | clyrrad: i'm saying it would be good for you to read the asterisk handbook |
| 03:13.03 | Micc | my asterisk just started being really flaky on me. |
| 03:13.04 | clyrrad | lancey.... So its used for all people connecting to you, and for who you also connect to |
| 03:13.06 | lancey | it will help you very much |
| 03:13.09 | Micc | Its hanging up at random times. |
| 03:13.17 | lancey | cause u seem to miss the main asterisk logic |
| 03:13.19 | clyrrad | lancey... dont worry about that I have that printed and have read it :) :) |
| 03:13.21 | Micc | On Zap or IAX channels. |
| 03:13.23 | lancey | yes, it does |
| 03:13.42 | lancey | rt: i've gone with CVS head |
| 03:13.43 | DannyF | Micc, hows cpu temps etc? |
| 03:13.54 | lancey | and have managed to stay working :) |
| 03:13.54 | rt | any difficulties in getting it to work? |
| 03:14.03 | Micc | DannyF, its fine. |
| 03:14.05 | lancey | rt: since the last half year - hardly not |
| 03:14.09 | rt | dare I hope for "compile/install/run"? |
| 03:14.11 | lancey | it was harder before that :) |
| 03:14.18 | lancey | rt: all you need is gmake and bison |
| 03:14.23 | DannyF | Micc, anything hardware related you could pin it on? |
| 03:14.28 | lancey | and then just gmake in /usr/src/asterisk |
| 03:14.31 | rt | well, i tried it recently with 4.x, which seemed problema6tic. |
| 03:14.31 | lancey | and you're done. |
| 03:14.39 | lancey | rt: use 5.3 and above |
| 03:14.57 | lancey | last night i just built * on 6.0-RC1 too :) |
| 03:15.02 | lancey | works like a charm |
| 03:15.09 | Micc | I keep getting Channel 0/1, span 1 got hangup request. |
| 03:15.14 | clyrrad | lancey.... thanks |
| 03:15.20 | Micc | Or I just get hangup |
| 03:15.20 | lancey | but i don't use any specific hardware, though - no zaptels, no isdns, no modems |
| 03:15.31 | lancey | so i can't say what's the status with that |
| 03:15.38 | ender | file[laptop]: ok, I've put those options in. We'll see what we get now. |
| 03:15.39 | Micc | But its doing it on IAX channels too. |
| 03:15.41 | lancey | clyrrad: hope i helped |
| 03:15.48 | clyrrad | you did thanks :) |
| 03:15.55 | Micc | I'm starting to think I've got a buggy version of asterisk. |
| 03:16.01 | Micc | But it worked for a few days. |
| 03:16.12 | DannyF | Micc naaaah couldnt be ;) |
| 03:16.38 | Micc | I built CVS HEAD on 10/5 |
| 03:16.59 | DannyF | could always try make update |
| 03:17.05 | *** part/#asterisk franck (n=franck@tikiwiki/franck) |
| 03:17.29 | *** join/#asterisk kb1_kanobe (n=jsmith@h24-207-96-50.cst.dccnet.com) |
| 03:18.12 | DannyF | my version is patched beyond recognition atm so cant help... |
| 03:19.37 | lancey | naaah |
| 03:19.41 | lancey | going back to bed |
| 03:19.44 | lancey | byez |
| 03:19.52 | clyrrad | night :P |
| 03:20.42 | rt | do i need to build zaptel (presumably not, since I odn't have one) or libpri? |
| 03:20.51 | ender | file[laptop]: thanks! |
| 03:21.47 | *** join/#asterisk CoolAcid (n=jason@216.99.98.39) |
| 03:21.50 | *** join/#asterisk zoa (n=zoa@host06.alica.hyatthsiagx.com) |
| 03:22.09 | kb1_kanobe | rt: you may need ztdummy if you want to use facilities that require a timing source, such as conferencing. |
| 03:22.22 | ender | file: I'm trying to find some documentation on that option, is there any? |
| 03:22.34 | rt | hmmm. that's in zaptel, I take it? |
| 03:22.39 | kb1_kanobe | yes. |
| 03:22.43 | rt | sorry, I know I'm asking stupid questions... |
| 03:22.49 | kb1_kanobe | nope. :-) |
| 03:23.01 | file | ender: a brief note in the example iax.conf |
| 03:23.43 | ender | file: feel free to /query me w/ the explanation. |
| 03:23.45 | rt | and now... chaos! |
| 03:24.56 | Katty | DannyF: for shame! |
| 03:25.11 | Katty | DannyF: it is /mew/! kthx. |
| 03:25.18 | *** join/#asterisk [hC] (n=hardcore@c-24-127-192-210.hsd1.fl.comcast.net) |
| 03:26.33 | *** part/#asterisk dabigshiznizzle (n=dabigshi@dsl-cw-65-171-138-87.giant-broadband.com) |
| 03:28.24 | file | KATTY! |
| 03:30.19 | Micc | ok. I did make update. we'll see if this helps. |
| 03:30.29 | Micc | bunch of files have been updated in the last 6 days. crazy |
| 03:30.56 | *** join/#asterisk Koshatul (n=evangeli@ip157.net65.ipnetworks.net.au) |
| 03:31.06 | clyrrad | file... I am trying to see where I can set accountcode, from within iax.conf, can not seem to get it working. I am under the conext that handles the outgoing calls from that DID, as well as the imcomming, but where can I set that variable? |
| 03:32.26 | Katty | file: mew. |
| 03:35.31 | Supaplex | m00 |
| 03:36.13 | Katty | file: going into suburban life is depressing :< |
| 03:40.47 | *** join/#asterisk jeffik (n=Jeff@CPE0020ed8494b8-CM0012c999ca4e.cpe.net.cable.rogers.com) |
| 03:41.44 | *** join/#asterisk FuriousGeorge (n=brian@ool-44c5a9b8.dyn.optonline.net) |
| 03:42.21 | FuriousGeorge | anyone ever install one of these door phones that go to FXO? |
| 03:42.36 | FuriousGeorge | all i need is the analog doorphone and a spare fxo? |
| 03:47.56 | marc324 | is "dsn" the database name? please help |
| 03:48.29 | kb1_kanobe | usually stands for 'data source name', but where are you looking? |
| 03:48.45 | marc324 | i have a database in postgresql called "asterisk" |
| 03:49.13 | marc324 | does "dsn" refer to the database name? |
| 03:49.25 | kb1_kanobe | ewww... odbc with asterisk? Haven't done that, sorry. |
| 03:50.40 | *** join/#asterisk bmg505 (n=leon@rndf-146-33-60.telkomadsl.co.za) |
| 03:53.04 | *** join/#asterisk Ariel_ (n=Ariel@dsl-20-177.cofs.net) |
| 03:54.22 | *** join/#asterisk mhnoyes (n=mhnoyes@user-2ivflfd.dialup.mindspring.com) |
| 03:55.14 | *** join/#asterisk shidan (n=shidan@CPE0013107d30c4-CM001371871af0.cpe.net.cable.rogers.com) |
| 03:56.06 | *** join/#asterisk Koshatul (n=evangeli@ip157.net65.ipnetworks.net.au) |
| 04:02.14 | *** join/#asterisk Pr0ph37 (i=user@adsl-067-034-122-207.sip.mia.bellsouth.net) |
| 04:02.52 | *** part/#asterisk Pr0ph37 (i=user@adsl-067-034-122-207.sip.mia.bellsouth.net) |
| 04:02.54 | *** join/#asterisk Brijn (n=bas@S0106004063c0fa1f.vn.shawcable.net) |
| 04:04.31 | Brijn | Good evening.. Q about meetme. Just started using *, so newbie question :) I installed AMP and it create meetme_additional.conf. It contains 8200 |
| 04:04.51 | *** join/#asterisk jdiskywlkr (n=kvirc@ip68-0-86-182.tu.ok.cox.net) |
| 04:05.41 | Brijn | extentions.conf has an entry for meeme's eg exten => _8XXXX,1,Macro(user-callerid) |
| 04:06.05 | *** join/#asterisk Veto_laptop (i=mdkuser@cpe-66-69-38-192.satx.res.rr.com) |
| 04:07.01 | Brijn | I guessed that by dialling 88200 I would get into the conference, but the friendly lady tells me that is not a valid conference.. How to use it?? |
| 04:10.56 | *** join/#asterisk Melmac (n=none@m8c2936d0.tmodns.net) |
| 04:17.56 | Brijn | No Meetme users? |
| 04:20.06 | SkramX | ? |
| 04:21.46 | *** join/#asterisk alphaque (n=alphaque@218.111.204.40) |
| 04:22.43 | *** join/#asterisk Koshatul (n=evangeli@ip157.net65.ipnetworks.net.au) |
| 04:23.03 | Brijn | AkramX: in a default * install, what do you dial to get into a conference (meetme.cong contains 8200 for example) |
| 04:26.18 | *** join/#asterisk KaBewM (n=kabewm@66-215-7-106.dhcp.psdn.ca.charter.com) |
| 04:28.35 | count | anybody had any luck compiling ztdummy on ubuntu? |
| 04:28.42 | count | s/compiling/installing |
| 04:28.44 | count | it compiles fine |
| 04:31.31 | *** join/#asterisk n3u7 (n=neutrin0@CPE000d8802a707-CM0011e6c7edb1.cpe.net.cable.rogers.com) |
| 04:33.00 | n3u7 | greets |
| 04:33.31 | count | hi |
| 04:33.51 | n3u7 | funny count |
| 04:33.54 | n3u7 | ... |
| 04:34.36 | n3u7 | my asterisk difficulity is sub_preempt_count missing from a module |
| 04:34.49 | n3u7 | zaptel |
| 04:35.05 | count | mine is compiling ztdummy |
| 04:35.25 | n3u7 | ah, I have a digium x100p card |
| 04:36.12 | Vco | does it make you feel like more of a man then when you didn't have a x100p? |
| 04:36.26 | count | Yes |
| 04:36.41 | n3u7 | :/ |
| 04:36.42 | marc324 | i get this: SQLGetPrivateProfileString failed with . |
| 04:37.12 | *** join/#asterisk ManxPower (n=eric@slip-12-65-54-145.mis.prserv.net) |
| 04:37.29 | n3u7 | well I'm going to read the mailing list that is piling up in my inbox |
| 04:39.42 | *** join/#asterisk Koshatul (n=evangeli@ip157.net65.ipnetworks.net.au) |
| 04:44.21 | *** join/#asterisk tainted_ (n=identd@ppp-71-137-169-240.dsl.irvnca.pacbell.net) |
| 04:44.32 | *** join/#asterisk Corydon76-home (i=grey@pdpc/supporter/sustaining/Corydon76-home) |
| 04:44.55 | *** join/#asterisk newmember (i=user@S010600036d1139fb.cg.shawcable.net) |
| 04:45.04 | wunderkin | fucking theives, damn telcos |
| 04:45.39 | wunderkin | my first bill, overcharging for loop and charging for service before install |
| 04:46.10 | wunderkin | plus charging for taxes on the mailing address and not service address lol |
| 04:46.26 | wunderkin | hilarious |
| 04:46.36 | marc324 | postgresql has a odbc driver.... why use unixodbc? |
| 04:46.52 | ManxPower | wunderkin, do that have a "15 day money back guarntee"? |
| 04:47.02 | *** join/#asterisk Snake-Eyes (n=blog@203.220.55.70) |
| 04:47.41 | wunderkin | it was dormant for 3 weeks, then when i hook it up it doesnt work (verified it worked on the install), the long distance company changed the framing to SF instead of ESF and it took them 3 days of blaming my equipment to get to the bottom of it |
| 04:48.00 | marc324 | can someone help me get asterisk realtime working? |
| 04:49.37 | *** join/#asterisk spootnick (n=irc@CPE-144-133-126-245.nsw.bigpond.net.au) |
| 04:51.16 | Brijn | Q: In the default * install, there is a line exten => _8XXXX,4,GotoIf($[${CALLERIDNUM} = ${EXTEN:1}]?6:5) I guessed dialling 88200 would dump me in the 8200 confrence. It tells me that the conference doesn't exist |
| 04:51.52 | spootnick | Asterisk's voip-info.org Wiki says it's possible to have multiple switches (realtime). have anybody used this already? i tried putting a switch after the other, but the second one never actually takes place |
| 04:54.16 | ender | Ok, I can never keep it straight. |
| 04:54.37 | ender | I have two FXO ports and two FXS ports. I need to plug an analog phone line into one of them for calling 911. Which port do I plug into? |
| 04:56.29 | spootnick | ender: if you're connecting an external POTS line into your * box, then it's FXO |
| 04:56.53 | ender | k |
| 04:57.31 | ender | and if my /etc/zaptel.conf file says fxs=1-2 that means that 1 and 2 are FXO ports getting FXS signalling right? |
| 04:59.21 | spootnick | no, it means the interface you're configuring will be identified as channel 1 and 2 in other config files |
| 04:59.29 | ender | oh ok. |
| 04:59.36 | ender | ah we got it, thanks. |
| 05:03.33 | *** join/#asterisk fyn (i=n4CGoz4m@24.72.11.11) |
| 05:06.06 | count | spootnick: what doyou mean multiple switches in realtime? |
| 05:06.06 | ManxPower | It also means it won't work, since you would need fxsls=1-2 or fxsks=1-2 |
| 05:07.03 | spootnick | count: i mean, in extensions.conf, in a certain point i'm using "switch => Realtime/@family". works fine. but i wanted to include another "switch => Realtime/@otherfamily" below it |
| 05:07.03 | ManxPower | ender, fxo ports get phone lines, fxs ports get phones, fxs ports get dead if you plug a phone line into them and the line rings. |
| 05:07.08 | ManxPower | ~fxofxs |
| 05:07.10 | jbot | methinks fxofxs is An FXO port expects to receive dialtone and receive ring voltage. An FXS port expects to provide dialtone and provide ring voltage. |
| 05:07.33 | ender | ok. |
| 05:07.44 | spootnick | count: there's a need to keep these two families separated, in case you're wondering |
| 05:08.11 | marc324 | ne1 here has succeeded in running asterisk realtime with postgres? |
| 05:08.16 | ManxPower | why not use dundi |
| 05:08.36 | marc324 | is that directed tome? |
| 05:09.56 | count | oh |
| 05:10.01 | count | marc324: yeah! |
| 05:10.06 | *** join/#asterisk docE (n=docE@206.165.75.198) |
| 05:10.08 | count | It runs great :) |
| 05:10.15 | count | spootnick: can they be different contexts? |
| 05:10.27 | docE | anyone in here @ astricon near Damin? |
| 05:10.32 | spootnick | count: hmm, i suppose so |
| 05:10.36 | count | spootnick: I use this: |
| 05:10.38 | count | [1-int] |
| 05:10.38 | count | switch => Realtime/@ |
| 05:10.38 | count | [1-ext] |
| 05:10.39 | count | switch => Realtime/@ |
| 05:10.45 | count | in extensions.conf |
| 05:10.57 | count | 1-int / 1-ext being in the 'context' field in the db |
| 05:14.26 | *** join/#asterisk dan__t (n=dant@ip70-176-120-15.ph.ph.cox.net) |
| 05:14.34 | dan__t | what's up, kids. |
| 05:14.49 | spootnick | count: wouldn't that be the same than doing '[context1] switch => Realtime/@familly' and '[context2] switch => Realtime/@family' ? |
| 05:15.01 | MooingLemur | cat /dev/urandom | perl -pe 's/[^a-z]//g' | dd bs=1 count=200 2>/dev/null | perl -pe 's/(.*)/\(SayText "$1"\)/' | festival |
| 05:15.25 | spootnick | then, to mix them up, include both contexts in a third one |
| 05:15.41 | dan__t | So I'm still new to Asterisk. I've been reading a lot about it, and trying to familiarize myself with the terminology and such. |
| 05:15.49 | dan__t | I guess what I ask now is, where do I start? |
| 05:16.13 | dan__t | What do I do now, which configs do I start with, yada yada. |
| 05:16.24 | dan__t | heheh |
| 05:16.32 | rt | and is going to try to get it working with FWD. |
| 05:16.36 | count | spootnick: possibly? |
| 05:16.42 | rt | that's sort of my first step. |
| 05:16.42 | dan__t | I have a small goal at this point; be able to use a Softphone to make a VoIP call via IAX, with an IAX provider |
| 05:16.48 | spootnick | count: i'm trying it now... let's see |
| 05:17.03 | *** join/#asterisk SwK[Work] (n=SwK@border0hsv.asterisksgi.com) |
| 05:17.39 | *** join/#asterisk SwK (i=zbaxbc@12-219-144-126.client.mchsi.com) |
| 05:17.52 | dan__t | from what I understand, there's a certain hierarchy of configuration files to be edited, as they share one or many common contexts |
| 05:18.17 | count | dan__t: you get * installed? |
| 05:18.20 | dan__t | I di. |
| 05:18.22 | dan__t | did, rather. |
| 05:19.41 | count | heh |
| 05:19.44 | count | is it not anymore? |
| 05:19.44 | spootnick | count: doesn't work |
| 05:20.18 | count | spootnick: with @family ? |
| 05:20.20 | *** join/#asterisk twisted[asteria] (n=twisted@asterisk/friend-and-developer/pdpc.professional.twisted) |
| 05:20.20 | *** mode/#asterisk [+o twisted[asteria]] by ChanServ |
| 05:20.42 | dan__t | No, it's still installed, count. |
| 05:20.42 | dan__t | haha |
| 05:20.48 | spootnick | count: yep. i suppose omitting the family would only force me to have the contexts in the database, so, no difference |
| 05:20.56 | count | can you get your softphone to register with it? |
| 05:21.06 | dan__t | Are you talking with me, count? |
| 05:21.08 | spootnick | what troubles me is that the holy wiki says one can do that. i wonder how... |
| 05:21.10 | count | spootnick: you need the context in the database anyways, for lookups aren't you? |
| 05:21.15 | count | dan__t: yeah, sorry :) |
| 05:21.27 | dan__t | How likely do you think it's going to be for me to pull that off, if I've not configured it at all? |
| 05:21.30 | count | the holy wiki is lacking wrt to realtime info :) |
| 05:21.41 | count | dan__t: oh, you just got a bare default one? :) |
| 05:21.43 | spootnick | count: i see that now |
| 05:22.02 | dan__t | Almost. |
| 05:22.06 | count | spootnick: It kind of sucks you have to touch extensions.conf for anything with realtime, huh? :) |
| 05:22.12 | dan__t | I edited it's prefix, because there are no working recent RPMs for RHEL/CentOS |
| 05:22.15 | count | dan__t: you intent on using IAX and not SIP? |
| 05:22.19 | Vco | omfg |
| 05:22.31 | Vco | this thing would take war driving to a whole new level |
| 05:22.32 | FuriousGeorge | has anyone ever installed a doorphone? |
| 05:22.36 | dan__t | for pbx <-> pbx stuff, yes. |
| 05:22.37 | Vco | http://cgi.ebay.com/ebaymotors/Mobile-Cellular-Cell-Tower-Truck-Antennas-Ham-Radio_W0QQcmdZViewItemQQcategoryZ63739QQitemZ4581983789QQrdZ1#ebayphotohosting |
| 05:22.55 | dan__t | but softphone <-> pbx would be sip |
| 05:24.00 | ender | I just noticed something missing w/ * vs another PBX I've used. |
| 05:24.07 | ender | how the heck do you do a page all? |
| 05:24.43 | dan__t | haha |
| 05:25.27 | ender | not that I'm missing it much, but I just know somebody is going to ask. |
| 05:25.43 | dan__t | So what do you say, count? |
| 05:25.54 | count | dan__t: is your box local? |
| 05:26.33 | dan__t | It is,. |
| 05:27.26 | count | ok |
| 05:27.30 | count | setting up a sip client should be easy |
| 05:27.42 | dan__t | client, yes. |
| 05:27.43 | dan__t | server? |
| 05:27.48 | count | that too :) |
| 05:27.54 | dan__t | That's what I'm worried about. |
| 05:27.58 | dan__t | I have *zero* exposure to this. |
| 05:28.10 | count | ok |
| 05:28.10 | count | in sip.conf |
| 05:28.10 | dan__t | But hey, it looks bad-ass, so.. |
| 05:28.13 | dan__t | I did install the samples. |
| 05:28.17 | count | you need to define a user/peer |
| 05:28.20 | dan__t | ok. |
| 05:28.26 | count | for example: |
| 05:28.31 | count | [dant] |
| 05:28.33 | count | type=peer |
| 05:28.36 | count | secret=somepasswd |
| 05:28.39 | dan__t | you forgot the __'s |
| 05:28.40 | count | host=dynamic |
| 05:28.43 | count | ;P |
| 05:28.46 | dan__t | ok. |
| 05:28.53 | count | context=somecontext |
| 05:28.59 | dan__t | do I need a [general] section in there, per the examples? |
| 05:29.11 | count | Yeah, but the default one should be fine for now |
| 05:29.19 | dan__t | ok |
| 05:29.37 | count | oh, wait |
| 05:29.37 | count | sorry |
| 05:29.40 | dan__t | ? |
| 05:29.41 | fyn | Er ... how do I troubleshoot this iaxy thinger? It says it "Got response back" from provisioning so it's listening on the IP it got from DHCP, I provisioned it with this: http://notpublic.wrong.button.com/iaxy.conf and not only does it not register and light up orange, but apparenly no traffic at all went to <ip> ... is it just mysteriously broken or is there something I can try? |
| 05:29.44 | count | you need to define the context for inboundcalls |
| 05:29.51 | dan__t | can I forget about that now |
| 05:29.53 | count | [general] |
| 05:29.55 | dan__t | just focus on the outbounds |
| 05:29.57 | count | context=inbound-calls |
| 05:29.57 | count | ok |
| 05:30.04 | dan__t | baby steps |
| 05:30.09 | count | heh |
| 05:30.11 | dan__t | ever seen "What About Bob?"? |
| 05:30.14 | count | nope? |
| 05:30.19 | dan__t | hm |
| 05:30.25 | dan__t | Alright then. |
| 05:30.42 | count | so |
| 05:30.47 | count | whats that pastebin site |
| 05:30.52 | dan__t | pastebin.com |
| 05:30.52 | count | so I don't flood irc with example configs |
| 05:32.19 | count | http://pastebin.com/390890 |
| 05:32.22 | count | something simple like that |
| 05:32.34 | count | this'll get you started with sip calls + voicemail eventually |
| 05:33.11 | dan__t | that is all part of sip.conf? |
| 05:33.14 | count | yes |
| 05:33.19 | count | you can throw it in at the bottom |
| 05:33.22 | dan__t | can there exists two identical contexts in the same file? |
| 05:33.28 | dan__t | can there exist, rather. |
| 05:33.29 | count | those aren't contexts |
| 05:33.34 | count | those are sip peers/users |
| 05:33.37 | count | one is a user one is a peer |
| 05:33.38 | dan__t | oh.. |
| 05:33.43 | count | you could define them both at once as a 'friend' |
| 05:33.48 | count | but I've not had good experience that way |
| 05:33.48 | dan__t | hm |
| 05:33.49 | dan__t | ok. |
| 05:33.57 | count | one lets you recieve calls (peer) |
| 05:34.01 | count | the other lets you make them (user) |
| 05:35.04 | dan__t | Oh, I see. |
| 05:35.13 | dan__t | and what dictates that is the type= directive. |
| 05:35.17 | count | yep |
| 05:35.24 | rt | to connect from a soft phone, presumably you'd use some kind of sip: url? |
| 05:35.35 | count | rt: depends on the softphone |
| 05:35.41 | count | with say, xten |
| 05:35.45 | count | you use a DNS record |
| 05:35.47 | dan__t | Which softphone would you recommend |
| 05:35.50 | count | and just a username |
| 05:35.51 | dan__t | that's just clean and simple. |
| 05:35.52 | count | I use xten |
| 05:36.04 | count | I think there's a free version |
| 05:36.13 | count | same as the $$ version, but doesn't use the G729 codec |
| 05:36.17 | Brijn | Yes, there is |
| 05:36.21 | rt | hmmm. |
| 05:36.21 | count | which won't really matter to you for now |
| 05:36.28 | count | G711 is fine for setup and testing |
| 05:36.29 | count | :) |
| 05:37.38 | dan__t | xten is what messed me up last time |
| 05:37.44 | dan__t | I was trying to connect to * |
| 05:37.46 | count | haha |
| 05:37.53 | dan__t | I was like. Oh shit. |
| 05:38.06 | Brijn | It's not bad, and a good docs on voip-info |
| 05:38.15 | count | xten is kinda neat, it takes 3 lines to get my sip account working :) |
| 05:38.18 | count | 3 config entries |
| 05:38.22 | count | user, pass, and domain |
| 05:38.25 | dan__t | ah |
| 05:38.34 | count | not sure how well it'd work if you don't have SRV records in dns for your * though |
| 05:38.50 | Brijn | http://www.asteriskguru.com/xlite.html |
| 05:38.59 | Brijn | count: Just dump in the IP |
| 05:39.09 | count | ok, cool |
| 05:39.11 | dan__t | that's what I had planned. |
| 05:39.13 | count | I've always had the SRV |
| 05:39.16 | count | so it wasn't a deal |
| 05:39.19 | count | but that's good to know |
| 05:39.24 | *** part/#asterisk jaike (n=a@203.131.137.76) |
| 05:39.40 | fyn | How do you update the firmware on an iaxy? |
| 05:40.11 | count | dan__t: let me know when you're ready for step2 |
| 05:42.10 | dan__t | "call not approved", from the * console. |
| 05:42.43 | count | set verbose 15 |
| 05:42.48 | count | on the asterisk cli |
| 05:42.50 | count | and make sure you do |
| 05:42.52 | count | sip reload peers |
| 05:42.57 | count | after you edit sip.conf |
| 05:44.19 | dan__t | same thing, nothing on the console. |
| 05:45.00 | count | what does the xten sya? |
| 05:45.01 | count | say |
| 05:46.51 | Brijn | Try dtmfmode=rtc2833 then "sip reload" |
| 05:47.17 | count | rfc |
| 05:47.33 | dan__t | *CLI> -- Saved useragent "X-Lite release 1103m" for peer dant |
| 05:47.39 | dan__t | that looks kinda promising. |
| 05:47.45 | count | heh |
| 05:47.47 | count | yep |
| 05:47.50 | count | do sip show peers |
| 05:48.01 | dan__t | yum. |
| 05:48.02 | dan__t | er |
| 05:48.12 | dan__t | yup. what the hell, yum, too. |
| 05:48.23 | count | haha |
| 05:48.29 | count | does sip show peer show you ? |
| 05:48.49 | dan__t | Yes. |
| 05:49.13 | count | cool |
| 05:49.14 | count | ok |
| 05:49.16 | count | that's step one |
| 05:49.21 | count | your phone registeres successfully |
| 05:49.30 | dan__t | you sure you don't mind helping me out with this? |
| 05:49.37 | count | naw, I'm watching * compile |
| 05:49.41 | count | over and over |
| 05:49.53 | count | trying to get the beta installed |
| 05:49.58 | count | so I got plenty of time :) |
| 05:50.45 | Brijn | count: Many changes in the beta? |
| 05:50.45 | count | And I remember how incredibly frustrating it was trying to get started |
| 05:50.59 | count | Brijn: yeah, the changelog is kinda long |
| 05:51.02 | dan__t | excellent. |
| 05:51.04 | dan__t | thanks, i appreciate it. |
| 05:51.05 | count | I <3 Realtime |
| 05:51.14 | count | not having to reload to change the dialplan == priceless |
| 05:51.32 | Brijn | Ca imagine in larger installs |
| 05:51.37 | Brijn | can |
| 05:51.38 | count | storing everything, dialplan, sip, voicemail, cdr, etc in postgres == really easy to have multiple frontend asterisk boxes working off the same set of configs |
| 05:51.41 | count | Yeah |
| 05:51.54 | count | I'm prepping for a few thousand sip users |
| 05:52.02 | dan__t | rad |
| 05:52.13 | dan__t | i'm preparing for 1 |
| 05:52.16 | Brijn | Migration from old style PBX or new install |
| 05:52.21 | count | Brijn: new install |
| 05:52.27 | count | F TDM |
| 05:52.31 | count | I hate that shit with a passion |
| 05:52.39 | count | now my nemesis is NAT |
| 05:53.26 | dan__t | So let's discuss step 2 here, count |
| 05:53.26 | dan__t | heheh |
| 05:53.35 | count | ok |
| 05:53.35 | count | NEXT! |
| 05:53.39 | count | what you've done so far |
| 05:53.46 | count | is create a sip user/peer |
| 05:53.48 | dan__t | set up sip client |
| 05:53.50 | count | to allow inbound/outbound calling by your softphone |
| 05:53.51 | Brijn | You do * consulting? @ count |
| 05:53.51 | dan__t | yes. |
| 05:53.55 | count | Brijn: sorta |
| 05:54.13 | count | Brijn: hosting of * boxes |
| 05:54.24 | Vco | he has this great scam where he "helps" you throug a setup in irc channels... |
| 05:54.31 | dan__t | haha |
| 05:54.32 | Brijn | ASP solu for VoiP |
| 05:54.39 | Vco | then sends thugs to collect on the invoice |
| 05:54.56 | count | Brijn: yeah :) |
| 05:54.59 | count | Vco: hahaha |
| 05:55.07 | count | No, some dude tried that to me when Iw as first starting out |
| 05:55.08 | Brijn | :) |
| 05:55.08 | dan__t | sounds great. |
| 05:55.09 | dan__t | haha |
| 05:55.13 | count | he was all like 'Ill help sure! just msg me' |
| 05:55.28 | count | and then I msged him and he goes 'I charge $89.95/hr, send me your paypal info' |
| 05:55.31 | count | wtf |
| 05:55.34 | dan__t | haha |
| 05:55.43 | count | If I wanted to pay, I'd call digium |
| 05:55.45 | count | But I don't |
| 05:55.46 | count | so I'm on irc |
| 05:55.47 | count | haha |
| 05:55.51 | count | right dan__t ?! :) |
| 05:56.01 | dan__t | There you go. |
| 05:56.13 | count | haha |
| 05:56.14 | count | ok |
| 05:56.14 | count | so |
| 05:56.15 | count | step 2 |
| 05:56.18 | dan__t | Yes. |
| 05:56.20 | count | did you read anything about contexts? |
| 05:56.24 | Brijn | I got some good help here yesterday as well. When you start all is strange |
| 05:56.29 | dan__t | A little bit, yes. |
| 05:56.32 | count | Brijn: ain't that the damn truth |
| 05:56.37 | dan__t | The general idea, I understand. |
| 05:56.40 | count | ok |
| 05:56.42 | count | So |
| 05:56.44 | count | in your sip user/peer |
| 05:56.47 | count | you set 'context' |
| 05:56.59 | count | if you just copy/pasted, it was 'sip-internal' |
| 05:57.19 | dan__t | ok |
| 05:57.22 | count | so |
| 05:57.26 | dan__t | I see. |
| 05:57.26 | Brijn | if it in [general] it will inherit down? |
| 05:57.29 | count | now, we go to extensions.conf |
| 05:57.35 | count | Brijn: yeah, thos esetting apply globally |
| 05:57.37 | count | unless overridden, I think |
| 05:57.41 | count | not sure about overrides |
| 05:57.50 | count | but I think it's a 'most specific' type thing |
| 05:57.59 | count | dan__t: we need to create a context |
| 05:58.03 | dan__t | ok. |
| 05:58.10 | count | this would be your starting point for what gets done based on buttons you press :) |
| 05:58.19 | dan__t | heheh |
| 05:58.21 | dan__t | ok. |
| 05:58.50 | count | [sip-internal] |
| 05:58.51 | count | exten => 100, 1, Playback(demo-thanks) |
| 05:59.01 | count | Stick that at the bottom of extensions.conf for now |
| 05:59.27 | dan__t | ok. |
| 05:59.41 | count | oops |
| 05:59.43 | count | add this line under that: |
| 05:59.47 | dan__t | Do I need to reload extensions now or something? |
| 05:59.48 | Vco | so new linksys/cisco skype phone huh.. |
| 05:59.50 | count | exten => 100, 2, Hangup() |
| 05:59.57 | count | yeah |
| 06:00.02 | count | after you edit these files you gotta reload extensions |
| 06:00.07 | Vco | which is kinda gay since it's just a cordless |
| 06:00.12 | Vco | and only works with skype |
| 06:01.03 | dan__t | ok. |
| 06:01.33 | count | now |
| 06:01.35 | count | dial 100 |
| 06:01.38 | count | on your xten |
| 06:01.43 | count | and watch the console |
| 06:01.44 | *** join/#asterisk digime (n=drooth@ip68-111-235-172.sd.sd.cox.net) |
| 06:02.16 | Vco | you set the xten to transmit silence right? |
| 06:02.27 | Brijn | Hahh, worked for me (following this, I started yesterday :) ) |
| 06:02.29 | dan__t | wow |
| 06:02.30 | count | Brijn: :) |
| 06:02.42 | dan__t | very cool. |
| 06:02.45 | dan__t | there's some hot slut on the phone |
| 06:02.46 | dan__t | brb |
| 06:02.49 | count | ahahaha |
| 06:02.49 | dan__t | No, just kidding :( |
| 06:03.28 | dan__t | anyway |
| 06:03.30 | dan__t | yeah that's kinda neat. |
| 06:03.30 | dan__t | heh |
| 06:03.48 | count | ok |
| 06:03.49 | count | So |
| 06:03.52 | count | You've done a few things there |
| 06:03.55 | count | 1) you know xten works |
| 06:03.58 | count | 2) you know asterisk works |
| 06:04.14 | count | so lets look at what that context did |
| 06:04.17 | count | the first line |
| 06:04.26 | count | [sip-internal] is obviously the name of the context |
| 06:04.38 | count | you use that name to reference it in sip.conf and other chan configs (like for IAX) |
| 06:04.55 | dan__t | inside extensions.conf? |
| 06:05.06 | count | well |
| 06:05.11 | count | you create contexts in extensions.conf |
| 06:05.15 | count | and then reference them from the other files |
| 06:05.17 | count | such as sip.conf |
| 06:05.30 | count | so in your sip peer entry for [dant] the line 'context=sip-internal' |
| 06:05.37 | count | references extensions.conf context [sip-internal] |
| 06:05.52 | count | unlike sip.conf, however, you can only have one context of a given name |
| 06:05.58 | count | so you can never define another [sip-internal] in extensions.conf |
| 06:06.22 | dan__t | I see. |
| 06:06.33 | dan__t | Alright. |
| 06:06.43 | Brijn | count: I have [iaxfwd] in iax.conf.. How do you call such an header there, that is not a context? |
| 06:06.44 | dan__t | Would you mind explaiing "dialplan" to me again, please? |
| 06:07.06 | Brijn | since there is also a line context= |
| 06:07.16 | count | dan__t: just a sec |
| 06:07.31 | count | Brijn: thats an iax user/client or whatever |
| 06:07.36 | count | not sure what the protocol term is for iax |
| 06:07.40 | Brijn | ok |
| 06:07.51 | count | but the same thing, your 'context=' line references a context in extensions.conf |
| 06:08.00 | count | ok, dialplans |
| 06:08.01 | Brijn | following that now |
| 06:08.03 | count | this is the fun stuff |
| 06:08.19 | count | I assume you all have worked at an office with a pbx right? |
| 06:08.41 | count | or at least understand the concept of 'extensions' ? |
| 06:08.41 | count | heh |
| 06:08.47 | count | I'm at extension 1000 |
| 06:08.48 | count | or whatever |
| 06:09.05 | count | in extensions.conf, that first field after exten=> is the extension |
| 06:09.08 | count | that's what you dial |
| 06:09.21 | count | the second item, the '1' or '2' |
| 06:09.23 | count | is the priority |
| 06:09.34 | dan__t | yeah i remember that part. |
| 06:09.36 | count | the third item is the action to take |
| 06:09.38 | count | so |
| 06:09.39 | *** join/#asterisk scfrec (i=scfrec@scfrec.compic.ee) |
| 06:09.45 | count | 100,1,Playback(demo-thanks) |
| 06:09.48 | count | 100,2,Hangup() |
| 06:10.04 | count | when you dial 100, asterisk looks in your context for an extension with that number |
| 06:10.14 | scfrec | hello to all. |
| 06:10.18 | count | hi |
| 06:10.26 | count | It then goes to priority '1' in this case |
| 06:10.31 | scfrec | small question - i need to forward all outgoing cals via IAX2 |
| 06:10.36 | count | (there are other special priorities I'm not gonna cover yet) |
| 06:10.44 | scfrec | is any good faq? |
| 06:10.51 | count | voip-info.org ? |
| 06:11.17 | dan__t | hmm |
| 06:11.31 | count | It executes the item in priority 1 |
| 06:11.42 | count | then it looks for priority 2 (assuming priority 1 runs properly) |
| 06:12.10 | Brijn | count: if you have more then one prio 1, will it complain,or execute them in the sequence in the list |
| 06:12.13 | count | scfrec: you could try putting _1XXXXXXXXXX,1,Dial(IAX2/peer) |
| 06:12.15 | count | in your dialplan |
| 06:12.22 | count | Brijn: it'll not work :) |
| 06:12.27 | Brijn | ok |
| 06:12.32 | count | actually |
| 06:12.39 | count | it might do something, bu tI dont think it's deterministic |
| 06:12.43 | count | I think it just barfs |
| 06:12.53 | count | you can use priority 's' |
| 06:12.56 | count | as the start |
| 06:13.00 | Vco | ususally, if there is a ,s, in teh context it will drop you there |
| 06:13.01 | count | and have a bunch of those in a row |
| 06:13.09 | count | Yeah, what Vco said |
| 06:13.43 | count | So, to setup a small dialplan |
| 06:13.47 | count | you could do something like: |
| 06:14.00 | count | [sip-internal] |
| 06:14.00 | count | exten => 1700,1, Dial(SIP/count) |
| 06:14.00 | count | exten => 2000,1, Dial(SIP/support) |
| 06:14.01 | count | exten => 3000,1, Dial(SIP/sales) |
| 06:14.07 | Rowter | something strage I been noticing, at the end of the day I see some DISA on show channel just hanging there, as if they got stuck there.. any ideas? |
| 06:14.38 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 06:15.16 | count | dan__t/Brijn: so far so good? |
| 06:15.18 | count | oh |
| 06:15.20 | count | the SIP/count |
| 06:15.22 | count | in Dial |
| 06:15.28 | dan__t | Yes, I follow |
| 06:15.29 | count | is how you tell a call to go to a specific peer |
| 06:15.38 | count | so I would have a [count] peer defined in sip.conf |
| 06:15.44 | count | and to send a call to him, Dial(SIP/count) |
| 06:15.49 | Brijn | yep! Just tried the welcome/hangup both with prio one (and ext reload'ed), worked ok |
| 06:16.02 | count | what'd it do brijn? |
| 06:16.06 | count | oh, execute them in order? |
| 06:16.08 | dan__t | So... using _1XXXXXXXXXX,1,Dial(IAX2/peer) generally means dial up to 10 numbers, across IAX2 peer named "peer" ? |
| 06:16.09 | *** join/#asterisk gres (n=serg@62.152.85.99) |
| 06:16.41 | Brijn | _1 means starting with 1? |
| 06:16.54 | Vco | ya |
| 06:16.59 | count | Yep! |
| 06:17.11 | Vco | _1800XXXXXXX would be any 800 toll free etc |
| 06:17.24 | dan__t | Would I have to explicitly dial 1? |
| 06:17.27 | count | yes |
| 06:17.29 | dan__t | What if I ust want to dial 555-555-5555 |
| 06:17.53 | Vco | then drop the 1 and have 10 X's |
| 06:17.58 | count | yep |
| 06:18.02 | lancey`away | :)) |
| 06:18.05 | dan__t | oh |
| 06:18.06 | dan__t | haha |
| 06:18.15 | lancey | i *can't* sleep |
| 06:18.16 | lancey | holly shit |
| 06:18.24 | Vco | who's holly? |
| 06:18.29 | count | is she hot? |
| 06:18.31 | lancey | someone lend me a hammer |
| 06:18.39 | Vco | well..she's kinky aparantly |
| 06:18.44 | count | dan__t: it gives you more flexibility when you have a mandatory first digit |
| 06:18.48 | count | kinda like 'dial 9 to get an outside line' |
| 06:18.51 | Vco | ie. ( |
| 06:18.53 | Vco | 9 |
| 06:18.55 | count | not a big deal at home or whatever |
| 06:18.56 | SwK | dont use 10 Xs for north american numbers... use _NXXNXXXXXX |
| 06:18.57 | Vco | Ya, what he said |
| 06:19.10 | Brijn | count: do you have good URL with the "rules" for these numbers? |
| 06:19.21 | count | voip-info.org has a few examples |
| 06:19.25 | count | asteriskguru does too I think |
| 06:19.43 | lancey | :)))) |
| 06:20.02 | Brijn | Ohh, I'll have a look at Guru, didn't find anything at voip-info (but will search again) |
| 06:20.16 | lancey | Brijn anything about what? |
| 06:20.45 | dan__t | hmmmm |
| 06:20.46 | dan__t | brb man |
| 06:20.49 | dan__t | something came up. |
| 06:20.59 | Brijn | A nice overview of the rules for using in number plans, much like a regexp overview for Perl |
| 06:21.20 | lancey | what rules? |
| 06:21.27 | lancey | the number wildcards? |
| 06:21.37 | count | While I'm giving newbie help, does anyone watching have any docs on storing voicemail messages in a database, odbc style? |
| 06:21.44 | count | I know it's possible! |
| 06:21.48 | count | but I can't see how :( |
| 06:21.59 | SwK | starting with _ means its a patern |
| 06:22.03 | Brijn | Related question.. The default config has an entry for meetme, it says: exten => _8X,1,Macro(user-callerid) and then later exten => _8XX,1,Macro(user-callerid) etc etc |
| 06:22.06 | SwK | X means any digit 0 - 9 |
| 06:22.14 | SwK | N means many digit 2 - 9 |
| 06:22.18 | lancey | N means 2-9 |
| 06:22.21 | SwK | and . mean follow up with anything |
| 06:22.24 | *** join/#asterisk jeffgus (n=jeffgus@2002:d856:c704:0:0:0:0:1) |
| 06:22.28 | lancey | yup |
| 06:22.34 | lancey | SwK is absolutely correct |
| 06:22.35 | count | freenode has ipv6 support ? |
| 06:22.36 | count | neeato |
| 06:22.37 | Brijn | I read this as Everting starting with 8 and has 8+one digit, 8+two digit etc |
| 06:22.46 | count | Brijn: yeah |
| 06:22.46 | lancey | and i do believe voip-info.org says this for sure |
| 06:23.04 | Brijn | SwK, ones are normally noy used in American numbers???? |
| 06:23.05 | lancey | Brijn so whats unclear to you? |
| 06:23.17 | Brijn | lancey: it doesn't work :-) |
| 06:23.21 | SwK | so _NX. would match ^[2-9][0-9]$(ANYTHING) |
| 06:23.24 | Brijn | I have a meetme.conf with: |
| 06:23.35 | count | Brijn: do you have ztdummy or zaptel hardware installed? |
| 06:23.49 | Brijn | conf => 8200 |
| 06:23.58 | lancey | Brijn what doesn't work |
| 06:23.59 | Brijn | Uhmmm, euhhhh, huhhh |
| 06:23.59 | SwK | 1s are using in NANP numbers all the time, just not for the first or the 4th digit ever (in a 10 digit number) |
| 06:24.05 | lancey | post here an example |
| 06:24.09 | Brijn | :) Don't think so :) |
| 06:25.12 | SwK | exten => _91NXXNXXXXXX,1,Dial(${EXTEN:1}) would catch a North American number and dial it after stripping the 9 |
| 06:25.40 | Brijn | SwK: Ahh, so area codes are never with a 1, is that because 011 is used for international calling? (I just moved from .nl to .ca, so need to get used to this) |
| 06:25.57 | dan__t | god damnit. |
| 06:26.00 | lancey | Brijn: look closely : http://www.voip-info.org/wiki/index.php?page=Asterisk+Dialplan+Patterns |
| 06:26.05 | dan__t | i'll be back in like 30 mins here, count. |
| 06:26.10 | count | dan__t: I'll be up for a while yet |
| 06:26.11 | dan__t | I appreciate the help, and apologize for this. |
| 06:26.14 | dan__t | Excellent. |
| 06:26.14 | count | it's only 2:30am :) |
| 06:26.14 | SwK | areacodes in NPA (area codes) and NXX's (Exchanges) never have 1 for the first digit |
| 06:26.17 | SwK | or0 |
| 06:26.20 | lancey | it can't be explained more perfect |
| 06:26.32 | Brijn | lancey: Bookmarked!! |
| 06:26.48 | lancey | :) |
| 06:27.18 | count | .nl is netherlands right? |
| 06:27.22 | lancey | it appears to be a good thing i can't sleep :) |
| 06:27.28 | count | sleep is overrated |
| 06:27.32 | SwK | yeah .nl is Netherlands |
| 06:27.34 | count | esp. when there is irc available |
| 06:27.39 | lancey | count so are my eyes and head :) |
| 06:27.42 | count | Why would you move to canuckia from .nl? :) |
| 06:27.50 | Brijn | count: I need a ztdummy for conferences? |
| 06:27.51 | SwK | free healthcare? |
| 06:27.56 | Brijn | count: Mountains!!!!! |
| 06:28.04 | Brijn | SwK: same in .nl |
| 06:28.05 | count | Brijn: depends, do you have zaptel hardware? |
| 06:28.07 | SwK | you need some kind of timing source for meetme |
| 06:28.17 | Brijn | count: No sir |
| 06:28.20 | count | I need ztdummy 'cause I'm pure software |
| 06:28.23 | *** join/#asterisk MikeJ[Laptop] (n=ircatjer@d14-69-8-30.try.wideopenwest.com) |
| 06:28.24 | count | yeah, then you need ztdummy :) |
| 06:28.31 | *** join/#asterisk Snake-Eyes (n=blog@203.220.55.70) |
| 06:28.31 | Brijn | Only the cool boys have that :) |
| 06:28.36 | count | ick |
| 06:28.37 | count | tdm! |
| 06:28.46 | count | all the coolcats are pure voip |
| 06:28.47 | count | :) |
| 06:28.48 | SwK | tdm > SIP/IAX at times |
| 06:29.01 | SwK | count I wanna see you send a 50 page fax over SIP or IAX |
| 06:29.04 | Brijn | tdm == signaling on POTS? |
| 06:29.09 | JamesDotCom | sip > tdm > iax |
| 06:29.16 | lancey | Brijn: TDM = Time Division Multiplexing |
| 06:29.19 | count | SwK: fax == 19th century |
| 06:29.23 | count | here in 2005 we have email |
| 06:29.25 | count | :) |
| 06:29.26 | SwK | tdm is Time Domain Multiplexing which is how calls are encoded on T!s and E1s |
| 06:29.26 | lancey | theres TDM over Ethernet, for e.g. |
| 06:29.28 | Brijn | and pDF |
| 06:29.34 | count | lancey: ew |
| 06:29.41 | lancey | xm |
| 06:29.44 | lancey | *hm |
| 06:30.05 | SwK | count: yeah well here in 2005 corporate america still uses those old crappy fax machines |
| 06:30.06 | lancey | :) |
| 06:30.07 | Brijn | Ah, so my ISDN PRI with 30 channels, is TDM'ed |
| 06:30.11 | count | SwK: T.38 ftw |
| 06:30.21 | count | of course, my t.38 relay drops it onto tdm |
| 06:30.25 | SwK | count last time I looked T.38 doesnt work with asterisk |
| 06:30.25 | count | but faxes aren't cool in any sense |
| 06:30.32 | count | Nope |
| 06:30.33 | count | sadly |
| 06:30.37 | SwK | next |
| 06:31.02 | *** join/#asterisk |Vulture| (n=V@211.119.205.68.cfl.res.rr.com) |
| 06:31.05 | wunderkin | they are working on it now |
| 06:31.10 | SwK | until that changes most companies will want TDM hardware for their asterisk servers |
| 06:31.22 | count | or just don't use asterisk for fax |
| 06:31.26 | |Vulture| | Anyone ever seen this? http://pastebin.ca/25295 |
| 06:31.38 | |Vulture| | I am getting this on inbound calls on that line, through a TDM |
| 06:31.54 | *** join/#asterisk tclineks (n=tclineks@ppp-70-243-238-201.dsl.tpkaks.swbell.net) |
| 06:31.56 | count | neat |
| 06:31.59 | count | no, sorry :) |
| 06:32.14 | lancey | jeez |
| 06:32.28 | lancey | |Vulture| misconfigured zap devices? |
| 06:32.41 | |Vulture| | lancey: the other channels work |
| 06:32.45 | SwK | misconfigured CID type? |
| 06:32.49 | lancey | yup |
| 06:32.54 | tclineks | can someone point me to setting up email notification upon voicemail arrival? |
| 06:33.00 | SwK | what kind of TDM interface? |
| 06:33.13 | |Vulture| | its just a digium TDM with 4 FXS cards |
| 06:33.15 | Brijn | How complex is interfacing to a ISDN PRI if you have not to much experience in that area? |
| 06:33.20 | tclineks | ideally attaching the message (not sure if this is supported out of the box) |
| 06:33.25 | SwK | FXS or FXO? |
| 06:33.26 | |Vulture| | Brijn: very easy |
| 06:33.28 | |Vulture| | FXS |
| 06:33.30 | |Vulture| | urg |
| 06:33.31 | lancey | Brijn it's easy |
| 06:33.31 | |Vulture| | FXO |
| 06:33.33 | SwK | ok |
| 06:33.38 | SwK | probably a blown module |
| 06:33.38 | lancey | |Vulture| FXS & FXO? |
| 06:33.40 | lancey | ISDN!? |
| 06:33.45 | |Vulture| | 4xFXO with FXS signal |
| 06:33.47 | lancey | PRI? |
| 06:33.53 | |Vulture| | lancey: 2 different conversations |
| 06:34.00 | lancey | yup |
| 06:34.01 | lancey | :)))) |
| 06:34.04 | lancey | as i said |
| 06:34.06 | SwK | |vulture| hit it with ztmonitor and see if the noise floor is a lot higher on that channel |
| 06:34.07 | |Vulture| | lancey: I use sangoma for my PRIs |
| 06:34.08 | lancey | i can't sleep |
| 06:34.08 | lancey | ;) |
| 06:34.15 | SwK | |vulture| but it could be a module going bad |
| 06:34.20 | Brijn | Our PBX might be needing replacement in the near future, would be interesting to see if there is a business case for moving to * with all VoiP phones |
| 06:34.20 | lancey | it's getting results :) |
| 06:34.30 | |Vulture| | SwK: I bet your right... good thing I got an extra in that one |
| 06:34.33 | SwK | brijn there is |
| 06:34.38 | |Vulture| | SwK: Ill try that tomorrow |
| 06:34.53 | |Vulture| | Brijn: do you have branch offices? |
| 06:34.58 | *** join/#asterisk newmember (n=newmembe@S010600a0c93dce87.cg.shawcable.net) |
| 06:35.06 | SwK | brijn: how many extension? |
| 06:35.15 | count | haha |
| 06:35.20 | Brijn | Vulture: Not yet, but that will come, and then it's becoming a lot more attractive all of a sudden |
| 06:35.30 | Brijn | SwK: only 40 |
| 06:35.36 | |Vulture| | yea we have 10 offices and the * system has been a lifesaver |
| 06:36.01 | |Vulture| | Brijn: hook yourself up with one of the Flex PRIs from XO and you will have all the LD you need in all your offices :) |
| 06:36.02 | count | unless you like bleeding money for call manager :) |
| 06:36.09 | |Vulture| | count: lol |
| 06:36.17 | Brijn | Vulture: LD??? |
| 06:36.27 | count | Long Distance |
| 06:36.28 | |Vulture| | Long Distance |
| 06:37.07 | |Vulture| | SwK: yea that channel is reading a high level of RX interference |
| 06:38.05 | SwK | brijn: at 40 extensions with all new phones you're only talking about a $20K max for a complete replacement with big PBX like features which you would spend probably $15K alone on the PBX phones |
| 06:38.32 | count | Polycom for the win |
| 06:38.36 | |Vulture| | I vote IP601 |
| 06:38.50 | |Vulture| | or the 501 w/PoE |
| 06:38.50 | count | haha, yeah |
| 06:38.52 | count | that phone rules |
| 06:38.54 | |Vulture| | for 40 extensions |
| 06:39.00 | SwK | that 20K estimate (which is very rought) was with poly's ;) |
| 06:39.01 | Brijn | I might get one of these Polycom phones (IP501?) for at home, seems like a nice phone |
| 06:39.03 | |Vulture| | Netgear switch... |
| 06:39.04 | count | I'm stuck on a 501 with special poe :) |
| 06:39.10 | count | get the 601 |
| 06:39.12 | count | it's nicer |
| 06:39.14 | SwK | i have IP500s and they rock |
| 06:39.14 | count | and not much more |
| 06:39.19 | |Vulture| | I only have 50X |
| 06:39.26 | count | me too |
| 06:39.28 | |Vulture| | I want to get a 601 but its overkill for the offices |
| 06:39.37 | |Vulture| | they just started coming down in price |
| 06:39.46 | count | the xhtml browser screen jibbie makes it really useful in like a call center |
| 06:39.48 | SwK | yeah 601s seem like overkill for anything I need |
| 06:39.49 | |Vulture| | now they are like $100 more than 500s they use to be 2x as much |
| 06:39.50 | count | point it at queue stats and stuff |
| 06:40.10 | count | 501 is $170, 601 is $210/220 I think |
| 06:40.15 | SwK | count: thats why we wrote a call center softphone |
| 06:40.18 | |Vulture| | thats cheap for a 601 |
| 06:40.25 | count | http://www.tritechcoa.com/ has polycom for stupid cheap |
| 06:40.33 | count | (not my site, I just buy from them alot ) |
| 06:40.34 | count | :) |
| 06:40.41 | count | $20 cheaper on the 501 than anyone else |
| 06:40.49 | count | Just remember to ask for the newer firmware |
| 06:40.56 | count | speeds up reboots of the phone considerably! |
| 06:40.57 | SwK | $200s for a IP50X is not a bad price |
| 06:41.08 | SwK | just get the firmware from freedomphones |
| 06:41.14 | count | they don't have the latest |
| 06:41.19 | count | or didn't last I checked |
| 06:41.21 | count | is it up now? |
| 06:41.23 | |Vulture| | I pay $180ish for the 501 and about 200 for the 501 w/PoE |
| 06:41.25 | SwK | I dunno |
| 06:41.37 | SwK | is there something newer then 1.5.2? |
| 06:41.38 | |Vulture| | didn't freedomphones take the firmware offline? |
| 06:41.38 | lancey | hm |
| 06:41.44 | SwK | no its still there |
| 06:41.52 | SwK | it just got nuked off the wiki for some reason |
| 06:41.54 | |Vulture| | don't think its linked on the wiki |
| 06:41.56 | |Vulture| | yea |
| 06:41.57 | count | 1.5.3 is out |
| 06:42.04 | count | and bootrom 3.0.1 |
| 06:42.17 | |Vulture| | yea but 3.0.1 is really just for the X01 series |
| 06:42.21 | count | Yeah |
| 06:42.23 | count | That's what we use, the 501 |
| 06:42.29 | lancey | anyone of you tried PA168-based IP phones? |
| 06:42.32 | lancey | i'm pretty happy |
| 06:42.36 | lancey | $67 |
| 06:42.40 | lancey | and it does IAX also |
| 06:42.47 | lancey | and almost any codec, including iLBC |
| 06:42.50 | Brijn | lancey: What brand is that? |
| 06:42.51 | lancey | firmware opensourced... |
| 06:42.58 | lancey | Brijn this is a chip solution |
| 06:43.00 | |Vulture| | IAX is nice but IAX2 is gunna be around soon |
| 06:43.07 | lancey | there are many products using it |
| 06:43.10 | SwK | we're already useing IAX2 I thought |
| 06:43.16 | |Vulture| | urg |
| 06:43.19 | |Vulture| | Im tired |
| 06:43.21 | lancey | Brijn: http://www.voip-info.org/tiki-index.php?page=PA168 |
| 06:43.24 | |Vulture| | I duno what I was thinking |
| 06:43.34 | SwK | hence the iax2 command ;) |
| 06:43.44 | lancey | i've been using 4 ATCOM's for 3 months now |
| 06:43.45 | |Vulture| | yea something v2 is coming out |
| 06:43.47 | lancey | now flows at all |
| 06:43.50 | lancey | *no |
| 06:43.50 | count | lancey: does it do g729? |
| 06:43.54 | lancey | count yes |
| 06:44.03 | count | oooh |
| 06:44.05 | count | only $67? |
| 06:44.08 | lancey | yup |
| 06:44.10 | count | it doesn't suck ass like the grandstream does it? |
| 06:44.12 | lancey | and this is price here in Bulgaria |
| 06:44.15 | count | holy shitty speakerphone batman |
| 06:44.16 | |Vulture| | I love 729 for remote phones |
| 06:44.34 | lancey | count: it's nothing fancy, it's $67 at all.. |
| 06:44.52 | count | lancey: nothing fancy is fine |
| 06:44.52 | lancey | it has 2-line display |
| 06:44.56 | count | As long as it works well |
| 06:45.03 | count | The gxp2000's speakerphone was painful |
| 06:45.06 | lancey | comfortable enough |
| 06:45.07 | tclineks | Does anyone have any experience with http://www.ovislink.ca/voip/ATA286.htm a handytone-286 for non-voip phone connectivity |
| 06:45.18 | lancey | speakerphone is not something to be proud of, though |
| 06:45.20 | lancey | but it works |
| 06:45.28 | lancey | AND it has 2 ethernet ports |
| 06:45.33 | lancey | something i've search long for |
| 06:45.38 | count | haha |
| 06:45.44 | lancey | and they are 100 mbits |
| 06:45.49 | Brijn | How is speakerphone on the IP501, being polycom I would expect really good? |
| 06:45.52 | lancey | not 10 like the cisco ATA 188s |
| 06:45.53 | count | *awesome* |
| 06:46.04 | count | ^^^ that ws for Brijn |
| 06:46.05 | count | heh |
| 06:46.09 | Brijn | :) |
| 06:46.20 | lancey | Brijn Polycom phones are G O O D |
| 06:46.28 | lancey | i don't have bigger letters :) |
| 06:46.31 | tclineks | I'm using the aforementioned handytone and http://www.digitnetworks.com/store/product_info.php?cPath=22&products_id=28 and the sound quality is terrible, unusable. |
| 06:46.41 | count | My only bitch about polycom is that they don't offer support (ie, firmware) to non-partners |
| 06:46.46 | Brijn | We had a few analog ones i my previous company, and they were indeed very good |
| 06:46.49 | tclineks | what i get for buying cheap hardware? |
| 06:46.59 | count | and you can't be a partner without being certified by one of their tech partners, who are competitors of most asterisk installers :) |
| 06:47.00 | SwK | if your going to get SIP phones for your Asterisk PBX you can not go wrong with a) Polycom or b) Ciscos |
| 06:47.09 | SwK | Ciscos are nice also |
| 06:47.14 | count | ciscos are $$ though, compartively |
| 06:47.16 | count | but nice, yes |
| 06:47.19 | SwK | not really |
| 06:47.22 | lancey | LinkSys PAP2-NA and normal phones work very good, too |
| 06:47.34 | SwK | PAP2s are cisco ;) |
| 06:47.44 | count | linksys==sipura==cisco now |
| 06:47.47 | count | so much for competition! |
| 06:47.54 | |Vulture| | I don't really like cisco |
| 06:47.58 | SwK | whats competition? |
| 06:48.00 | |Vulture| | I have issues with them on VLANs |
| 06:48.08 | SwK | LinkSys == Sipura in the first place |
| 06:48.09 | |Vulture| | but the new firmware seems to fix that |
| 06:48.52 | SwK | a linksys pap2 == sipura spa-2001 w/ different plastic and skin ont he web ui |
| 06:48.56 | lancey | :) |
| 06:49.01 | lancey | bye guys |
| 06:49.02 | lancey | :) |
| 06:49.17 | |Vulture| | speaking of linksys... I love my WRT54Gs! |
| 06:49.19 | |Vulture| | lol |
| 06:49.52 | Brijn | I have been looking at these uber geile VoiP WiFi phone, very nice, but a bit expensive :( |
| 06:50.14 | count | cisco's/moto have the dualmode 802.11/CDMA phone coming out soon |
| 06:50.16 | |Vulture| | have they gotten any better? |
| 06:50.23 | count | or is it 802.11/GSM |
| 06:50.40 | |Vulture| | the old cisco phone doesn't have a sip image |
| 06:50.43 | |Vulture| | the 802.11 |
| 06:51.27 | count | they mgcp or sccp ? |
| 06:51.33 | |Vulture| | mgcp |
| 06:51.36 | count | yum |
| 06:53.21 | Brijn | Time to leave.. Many thanx for all the help/explanations!!! CY |
| 06:53.53 | count | sigh |
| 06:54.03 | count | asterisk needs to start using a configure --prefix script |
| 06:54.09 | count | this make file mess kills me |
| 06:55.15 | SwK | autoconf would be nice |
| 06:55.56 | count | it makes packaging it a pain in the ass |
| 06:55.57 | count | heh |
| 06:56.11 | count | or, say, not installing configs to /etc |
| 06:56.15 | count | er, /etc/asterisk |
| 06:56.43 | count | so SwK, do you have any experience on storing vmail messages via odbc? |
| 06:56.56 | SwK | i wrote some of those patches |
| 06:57.13 | SwK | so yes |
| 06:57.27 | SwK | but you probably dont wanna know what I really think about it |
| 06:57.27 | count | how in gods name do you enable it?! |
| 06:57.38 | count | haha |
| 06:57.40 | count | what's that? |
| 06:57.47 | SwK | uncomment the define in the makefil |
| 06:57.59 | *** join/#asterisk PakiPenguin (n=uppal@linuxpakistan/admin/pakipenguin) |
| 06:58.01 | PakiPenguin | morning |
| 06:58.07 | count | howdy |
| 06:58.19 | *** join/#asterisk dos000 (n=dos000@CPE00119572fd49-CM00137186e53a.cpe.net.cable.rogers.com) |
| 06:59.01 | SwK | count: look around line 75 in the apps/Makefile |
| 06:59.06 | count | yeah, got that |
| 06:59.07 | tclineks | could someone point me in the path of setting up email notification upon voicemail arrival? |
| 06:59.45 | SwK | tclineks: look at voicemail.conf.sample in asterisk/configs in the source |
| 06:59.56 | count | yeah, 90% of that file is about email notification |
| 07:00.12 | SwK | iirc its like this tho mailboxnum => passcode,Users Name,email@domain.com |
| 07:00.59 | SwK | theres some more options but if you have an MTA setup correctly and it has a standard "sendmail" wrapper (or you are using sendmail) it should "just work" (tm) |
| 07:01.32 | tclineks | SwK i have that much set up |
| 07:01.35 | tclineks | mta is functional |
| 07:01.48 | tclineks | qmail |
| 07:01.50 | count | SwK: so, after building that, set 'odbstorage=dbnamefrom res_odbc'? |
| 07:02.01 | count | does qmail have a sendmail wrapper? |
| 07:02.05 | tclineks | count: yes |
| 07:02.55 | count | haha |
| 07:02.56 | count | whoops |
| 07:03.05 | count | odbc voicemail is bobobusted in the beta tgz |
| 07:04.01 | *** join/#asterisk NoRemorse (n=axel@202.161.68.2) |
| 07:04.21 | NoRemorse | hi all, anyone here familair with openh323 use in asterisk? having trouble getting a call to place |
| 07:04.24 | *** join/#asterisk CleanerX (n=nix@nat-ph3-wh.rz.uni-karlsruhe.de) |
| 07:05.22 | SwK | count: yeah theres some info on it in the wiki I think |
| 07:05.32 | SwK | qmail ships with a sendmail wrapper |
| 07:06.17 | SwK | count: odbc voicemail is just extra over head... it still writes out the files just like regular voicemail storage, then moves it to the DB then deletes the files |
| 07:06.26 | SwK | retrieval works opposite of that |
| 07:06.46 | count | SwK: thats fine |
| 07:06.55 | count | I'm trying to abstract out as much as physically possible from the local disk |
| 07:06.57 | SwK | gets the data from the DB, creates the files in spool/asterisk/voicemail... then plays them back then deletes them |
| 07:07.02 | count | and nfs is dirty |
| 07:07.19 | SwK | nfs works far better then odbc storage |
| 07:07.29 | count | well |
| 07:07.33 | count | considering it won't even compile |
| 07:07.38 | count | I'm inclined to agree :) |
| 07:07.59 | *** join/#asterisk Gunnar (n=gunnar@bkkb-gw.bitcon.no) |
| 07:09.20 | *** join/#asterisk \PsyKo\ (n=xxxxxxxx@golia.caltanet.it) |
| 07:17.12 | *** join/#asterisk dan__t (n=dant@ip70-176-120-15.ph.ph.cox.net) |
| 07:17.15 | dan__t | bah. |
| 07:18.08 | count | hah |
| 07:18.27 | dan__t | power went out at work |
| 07:18.39 | dan__t | GOOD THING THE UPS BATTERIES ARE SHIT |
| 07:18.41 | count | UPS + genset! |
| 07:20.22 | NoRemorse | anyone know where pwlib and openh323 are installed to? |
| 07:20.33 | *** join/#asterisk starman (n=inshift@dsl001-136-136.lax1.dsl.speakeasy.net) |
| 07:20.43 | count | NoRemorse: wherever you told them to go? |
| 07:23.10 | NoRemorse | found em didnt tell em anywhere lol. default is /usr/lcoal/lib |
| 07:23.18 | count | wtf does asterisk insist on having it's config directory outside of it's install directory :( |
| 07:24.02 | NoRemorse | hmm how do I disable all that gtk crap when compiling asterisk please? |
| 07:24.24 | count | gtk crap? |
| 07:24.36 | Vco | ^ what he said |
| 07:24.42 | NoRemorse | pbx_gtkconsole.c:39:21: gtk/gtk.h: No such file or directory |
| 07:24.46 | count | are you building the addons? |
| 07:24.49 | count | or asterisk itself? |
| 07:24.56 | NoRemorse | asterisk, just did a make |
| 07:25.17 | NoRemorse | what ARE the addons btw? |
| 07:25.17 | Vco | uhh.. |
| 07:25.20 | Vco | addons....natuarally |
| 07:25.24 | count | hahaha |
| 07:25.27 | count | 'duh' |
| 07:25.29 | NoRemorse | such as..... |
| 07:25.34 | Vco | sql stuff..mp3..something.. |
| 07:25.39 | NoRemorse | ah ok thanks |
| 07:25.42 | Vco | haven't looked in a while |
| 07:25.45 | dan__t | k back. |
| 07:25.46 | dan__t | anyway |
| 07:25.53 | dan__t | alright, count. I think I'm ready for step 3 |
| 07:25.55 | dan__t | wait no. |
| 07:26.02 | count | haha |
| 07:26.07 | dan__t | In those extensions, can I pause the transfer, so it doesn't happen for like 5 seconds? |
| 07:26.11 | count | yeah |
| 07:26.17 | count | that's the app 'Wait' |
| 07:26.21 | dan__t | oh ok. |
| 07:26.22 | count | 100,2,Wait(5) |
| 07:26.23 | dan__t | Wait(5)? |
| 07:26.26 | dan__t | excellent. |
| 07:26.28 | count | 100,3,Dial(blah) |
| 07:26.29 | dan__t | let me try that. |
| 07:26.47 | count | NoRemorse: your source package is wierd |
| 07:26.53 | count | I have a gtkconsole and stuff |
| 07:27.00 | count | .c's/.h's |
| 07:27.05 | count | but just a make doesn't do anything |
| 07:27.09 | count | are you compiling from an xterm? |
| 07:27.38 | count | oh |
| 07:27.43 | *** join/#asterisk lenne_dk (n=mirc@83.72.129.7.ip.tele2adsl.dk) |
| 07:27.43 | count | it looks like it looks for gtk-config |
| 07:27.48 | count | and if it sees it, it tries to compile that? |
| 07:28.01 | dan__t | hrm, doesn't seem to be working |
| 07:28.06 | dan__t | even after I did a 'reload extensions' |
| 07:28.17 | count | whats the console say |
| 07:28.20 | dan__t | exten => 100, 1, Wait(5) is what I have |
| 07:29.00 | count | I think you need to answer before you can Wait ? |
| 07:29.11 | count | try Answer() as prio 1, and Wait as 2 |
| 07:29.15 | dan__t | executing playback of demo-thanks, hangup |
| 07:29.25 | dan__t | I don't answer |
| 07:29.28 | dan__t | I mean it just answers |
| 07:29.38 | dan__t | There's no directive in there like Answer() |
| 07:29.47 | count | I mean, put one in there |
| 07:29.48 | count | heh |
| 07:29.50 | dan__t | oh. |
| 07:30.51 | dan__t | It goes "Buh bye... Thank you for trying out the Asterisk Open Source PBX" |
| 07:30.55 | NoRemorse | how do I disable gtk in asterisk please? |
| 07:31.15 | dan__t | look at the Makefile? |
| 07:31.15 | dos000 | is it not possible to transcode voice prompts in g729 ahead of time and then just stream the resulting packets ? why does it have to be done in real time |
| 07:31.26 | count | dos000: yes it's possible |
| 07:31.28 | count | easy even |
| 07:31.32 | count | if you a g729 codec |
| 07:31.35 | NoRemorse | dan__t ther eis no occurance of the string gtk anywhere in asterisk dir |
| 07:31.41 | count | NoRemorse: yes there is |
| 07:31.41 | count | :) |
| 07:31.46 | dos000 | count anyone done it .. care to give howto |
| 07:31.51 | NoRemorse | doh bet its caps |
| 07:31.52 | count | in /asterisk/pbx/Makefile |
| 07:31.54 | count | is where it is |
| 07:32.04 | NoRemorse | nope |
| 07:32.13 | dan__t | grep -Ri "gtk" asterisk-source-dir |
| 07:32.14 | NoRemorse | what am I looking for if not gtk? |
| 07:32.28 | count | GTK_FLAGS=`${CROSS_COMPILE_BIN}gtk-config --cflags gthread` |
| 07:32.29 | count | GTK_LIBS=`${CROSS_COMPILE_BIN}gtk-config --libs gthread` |
| 07:32.29 | count | MOC=$(QTDIR)/bin/moc |
| 07:32.42 | count | lines 22-24 of the Makefile |
| 07:32.43 | dan__t | it's gotta be before that. |
| 07:32.43 | count | :) |
| 07:32.54 | count | dos000: um |
| 07:32.58 | count | I did it |
| 07:33.05 | dos000 | count pray tell how |
| 07:33.09 | count | 'sec |
| 07:33.11 | NoRemorse | ah its in pbx dir thanks |
| 07:33.28 | dan__t | Alright, so, my wait doesn't work. |
| 07:33.47 | dan__t | count, http://pastebin.com/390939 |
| 07:34.27 | *** join/#asterisk mithro (n=tim@tagung-233-198.tagung.uni-hamburg.de) |
| 07:34.44 | count | what happens? |
| 07:34.58 | dan__t | I call the extension, and it goes right to the message |
| 07:35.02 | dan__t | demo-thanks |
| 07:35.56 | count | on the * console do you see it execute the wait? |
| 07:35.56 | dan__t | No, I do not. |
| 07:35.56 | NoRemorse | any here use dopenh323 before please? |
| 07:35.56 | dan__t | http://pastebin.com/390940 |
| 07:35.56 | NoRemorse | lol |
| 07:36.06 | count | dan__t: your extensions aren't reloaded I think |
| 07:36.11 | *** join/#asterisk Snake-Eyes (n=blog@203.220.55.70) |
| 07:36.19 | dan__t | Well, look at that error |
| 07:36.28 | dan__t | Spawn extension (sip-internal, 100, 2) exited non-zero on 'SIP/dant-81f4' |
| 07:36.34 | dan__t | that second priority, is the Wait() |
| 07:36.40 | dan__t | is Wait the right function? |
| 07:36.51 | dan__t | (What's the proper term, function or application?) |
| 07:37.18 | count | command |
| 07:37.25 | dan__t | ahh. |
| 07:37.28 | lenne_dk | It appears that the sip reload didn't work? |
| 07:37.45 | dan__t | sip reload? Thought it was reload extensions |
| 07:37.48 | count | no |
| 07:37.59 | count | that exit non zero means you hungup |
| 07:37.59 | lenne_dk | Sorry, yes |
| 07:38.01 | count | didn't you? |
| 07:38.04 | dan__t | Nope. |
| 07:38.06 | dan__t | It hung me up. |
| 07:38.11 | dos000 | count did you read it from somewhere how to do it ? |
| 07:38.13 | count | oh |
| 07:38.17 | count | dos000: yeah, I'm trying to find the link |
| 07:38.23 | dos000 | okay |
| 07:38.32 | count | dos000: you can always use record() from a phone that does g729 :) |
| 07:38.43 | dan__t | Why does it answer with "Bum bye" |
| 07:38.47 | dan__t | "Buh bye", rather. |
| 07:38.48 | dan__t | heh |
| 07:38.56 | lenne_dk | Try "show dialplan" to see of the extensions are loaded correctly |
| 07:39.16 | dos000 | count, but the prompts are in gsm/wav formats |
| 07:39.36 | dan__t | doesn't look like they're all there. |
| 07:39.51 | dan__t | killing * and restarting it altogether, loads it. |
| 07:39.57 | dan__t | per "show dialplan", anyway |
| 07:39.59 | dan__t | let me try it |
| 07:40.01 | count | dan__t: cause that file is the last of a really long demo :) |
| 07:40.08 | count | I just use it cause it's easy to rememeber 'demo-thanks' |
| 07:40.14 | count | if you look in extensions.conf |
| 07:40.18 | count | at the demo extensions |
| 07:40.22 | count | you'll see a huge number of them |
| 07:40.26 | dan__t | Yeah. |
| 07:40.30 | dan__t | But what's that got to do with my, uh, context |
| 07:40.33 | dan__t | context right? |
| 07:40.40 | count | yeah! |
| 07:40.47 | dan__t | heh. |
| 07:40.49 | count | just change your sip context to 'demo' or whatever it is |
| 07:40.55 | count | and then you'll be able to dial/use those numbers |
| 07:40.56 | dan__t | oh |
| 07:40.56 | dan__t | hrm. |
| 07:40.58 | dan__t | yea... |
| 07:41.02 | dan__t | let me try this. |
| 07:41.50 | dos000 | count, i am worried * will refuse to talk g729 if the lib support is not there. |
| 07:42.14 | count | no it won't |
| 07:42.16 | count | I've done it |
| 07:42.29 | lenne_dk | The new grandstream BT100 firmware has a "Special feature" pulldown, which can be "Standard" or "MediaRing". What might it do? |
| 07:42.29 | count | you just gotta make sure you get EVER one of those files |
| 07:42.34 | count | like, beep.gsm :) |
| 07:42.40 | count | lenne_dk: ring tones? |
| 07:42.41 | count | heh |
| 07:43.32 | dan__t | mkay... the context for my user is = demo, but when I reload and call back that extension, I get a fast busy signal |
| 07:43.42 | count | call what extension? |
| 07:43.46 | dan__t | 100 |
| 07:43.48 | dan__t | that one that we made |
| 07:43.51 | dos000 | count is this it ... http://www.voip-info.org/tiki-index.php?page=Asterisk+G.729+pass-thru |
| 07:43.55 | dan__t | i just changed the context from sip-internal to demo |
| 07:44.07 | dan__t | oh shit |
| 07:44.12 | dan__t | because I used a # instead of ; |
| 07:44.35 | dan__t | Nope just kidding. |
| 07:44.38 | dan__t | that wasn't the problem. |
| 07:45.19 | count | the one you made wont work |
| 07:45.23 | count | if you changed your context to demo |
| 07:45.32 | count | because you no longer have access to the sip-internal context you created |
| 07:45.33 | dan__t | why not? |
| 07:45.36 | count | it's kind of like access control |
| 07:45.41 | dan__t | ... |
| 07:45.41 | count | your 'context' is now 'demo' |
| 07:45.47 | count | so the only thing you can dial is in the 'demo' menu |
| 07:46.13 | dan__t | yeah |
| 07:46.24 | dan__t | but my uh, ... extension? ..., is 100 |
| 07:46.32 | count | no :) |
| 07:46.38 | count | you don't have an extension unless you set one up |
| 07:46.46 | count | its not assigned to your phone |
| 07:46.49 | dan__t | We did. |
| 07:46.54 | dan__t | Remember that extension 100 that we made? |
| 07:46.55 | count | but that's in a different context :) |
| 07:46.58 | count | yeah :) |
| 07:47.12 | dan__t | hmm |
| 07:47.14 | count | Here's an example of why that works like that |
| 07:47.18 | count | I have 2 contexts |
| 07:47.22 | count | sip-internal and sip-external |
| 07:47.30 | dan__t | (I only have sip-internal) |
| 07:47.40 | count | you also have dmeo |
| 07:47.41 | count | demo |
| 07:47.43 | dan__t | yes. |
| 07:47.45 | count | I'm just giving an example |
| 07:47.47 | dan__t | ok |
| 07:47.51 | count | instead of demo, I have sip-external |
| 07:48.07 | count | Sip-external is my default context for incoming calls on a public phone number |
| 07:48.12 | count | you can call me over a 1800, for example |
| 07:48.13 | dan__t | ok. |
| 07:48.31 | count | now, I do'nt want you to be able to then dial 15555555555 and make an outbound call using my phone system |
| 07:48.35 | count | so I get billed for it |
| 07:48.38 | count | so |
| 07:48.47 | count | the context, sip-external, doesn't have anything to dial outbound on |
| 07:48.55 | count | it just has a dial(SIP/count) |
| 07:49.04 | count | so it sends the incoming call to my sip phone |
| 07:49.05 | dan__t | hmm |
| 07:49.09 | count | on my sip phone, however |
| 07:49.20 | count | I have an extension _1XXXXXXXXXX |
| 07:49.28 | count | which lets my sip phone, because it's in that context, dial outbound |
| 07:49.35 | dan__t | yes. |
| 07:49.40 | count | so |
| 07:49.42 | dan__t | ok. |
| 07:49.45 | dan__t | I see. |
| 07:49.45 | count | if your phone's context is 'demo' |
| 07:49.46 | dan__t | Kinda. |
| 07:49.53 | count | It's kind of like a menu system |
| 07:50.03 | count | context is the menu of choices in front of you |
| 07:50.06 | count | if you change what your menu is |
| 07:50.11 | count | you don't get to see the old menu |
| 07:50.16 | dan__t | yeah i know |
| 07:50.19 | count | Now, there are commands to jump contexts within a dialplan |
| 07:50.21 | count | goto() |
| 07:50.25 | dan__t | but I thought I called the demo context from my inbound uh... thing. |
| 07:50.32 | count | heh |
| 07:50.42 | count | you called from your outbound peer |
| 07:50.42 | dan__t | (what is thing? you know what i'm talking about) |
| 07:50.49 | count | er |
| 07:50.50 | count | sorry |
| 07:50.52 | count | your inbound user |
| 07:50.52 | lancey`away | !? |
| 07:50.56 | dan__t | yes. |
| 07:50.56 | lancey`away | aaa |
| 07:50.57 | lancey`away | :) |
| 07:50.58 | count | you get calls to your peer |
| 07:51.15 | lancey | peer -> something you make calls TO |
| 07:51.22 | lancey | user -> something you get calls FROM |
| 07:51.28 | lancey | friend -> hermafrodite :) |
| 07:51.43 | dan__t | werd. |
| 07:51.50 | count | friends dont let friends use friends |
| 07:51.52 | count | :) |
| 07:51.57 | lancey | :))) |
| 07:52.20 | dan__t | Alright, interesting. |
| 07:52.25 | dan__t | I think I've had enough for tonight, however. |
| 07:52.25 | dan__t | haha |
| 07:52.29 | count | haw |
| 07:52.32 | count | well |
| 07:52.34 | count | the next step |
| 07:52.40 | dan__t | I do appreciate the help. |
| 07:52.43 | count | is to play around with the 1358713513513580 commands available |
| 07:52.46 | count | and to setup voicemail! |
| 07:52.49 | dan__t | I'll be back tomorrow for sure. |
| 07:52.50 | count | no pbx is worthwhile without voicemail |
| 07:52.52 | dan__t | Yes, definitely. |
| 07:52.53 | count | and mp3 music on hold |
| 07:52.54 | dan__t | haha yes. |
| 07:52.54 | count | haha |
| 07:52.55 | dan__t | haha |
| 07:52.57 | dan__t | excellent. |
| 07:53.09 | count | its only 4am |
| 07:53.11 | dan__t | I have an idea for a product |
| 07:53.14 | dan__t | which is why I want to explore * |
| 07:53.15 | count | you can't be going to sleep :) |
| 07:53.23 | NoRemorse | should I be using oh323.conf or h323.conf with openh323? |
| 07:53.24 | lenne_dk | Or interface with the tempeature sensors in the fridge... |
| 07:53.27 | dan__t | It's 2am here, and I have to be up at the asscrack of nooon. |
| 07:53.29 | dan__t | noon, rather. |
| 07:53.41 | count | haw |
| 07:53.44 | dan__t | I slept like 2 hrs last night |
| 07:53.48 | dan__t | and maybe the same, the night before, |
| 07:53.54 | count | I average about 3-4 hours per night :) |
| 07:54.12 | dan__t | then i'm going back to Phx Thurs for a few days |
| 07:54.23 | dan__t | heh |
| 07:54.30 | dan__t | I hate CO, so I fly back "home" 2-3 times/month |
| 07:54.47 | count | Phx? |
| 07:54.52 | dan__t | Phoenix. |
| 07:54.55 | count | ohh |
| 07:54.59 | count | I was wondering what ph.ph was |
| 07:55.03 | count | thought it might be philly |
| 07:55.04 | dan__t | heheh |
| 07:55.18 | dan__t | I just moved to Denver for some work. |
| 07:55.25 | count | cool |
| 07:55.35 | dan__t | not really |
| 07:55.36 | dan__t | haha |
| 07:55.50 | dan__t | I'll be fscking with * from work |
| 07:56.14 | dan__t | anyway |
| 07:56.15 | count | haw |
| 07:56.16 | dan__t | thanks again for the help. |
| 07:56.18 | count | np |
| 07:56.20 | count | later! |
| 07:56.21 | dan__t | 'nite |
| 07:57.39 | NoRemorse | does anyone have ANY idea what the diff between oh323.conf and h323.conf is? |
| 07:57.40 | *** join/#asterisk Akelavlk (n=jansun@82.119.239.141) |
| 07:57.43 | Akelavlk | Hello, is possible put ISDN2 cable directly to E1/T1 card? |
| 07:57.52 | *** join/#asterisk Snake-Eyes (n=blog@203.220.55.70) |
| 07:58.07 | NoRemorse | there seems to be 2 diff h323 chans available, OH323 and H323 |
| 07:58.15 | Akelavlk | I mean without any other device.. Just plug cable into E1/T1 |
| 07:58.38 | NoRemorse | what country isdn? |
| 07:59.41 | NoRemorse | Akelavlk: what country are u in? |
| 08:00.13 | lancey | NoRemorse: H323 is NuFone's implementation |
| 08:00.35 | NoRemorse | cos in Australia, ISDN is BRI not PRI, so therefore it wouldnt work in E1 card need a BRI card |
| 08:00.49 | NoRemorse | lancey: thank you!!! so no wonde rit aint working lol |
| 08:01.54 | NoRemorse | is the oh323.conf file identical? ie can I just rename h323.conf? |
| 08:02.04 | *** join/#asterisk tobiasWolf (n=konversa@195.162.255.10) |
| 08:03.02 | Akelavlk | NoRemorse, In slovakia. |
| 08:03.38 | Akelavlk | NoRemorse, how is diffrence between BRI and PRI? |
| 08:04.19 | NoRemorse | they are different, not sure exactly, similar to diff between say a phone line and isdn or say a t1 and e1 |
| 08:05.35 | Akelavlk | Hmm, I know that there are really small diffrences also.. But what diffrences :-) |
| 08:05.48 | flenders | guys if I have a message which is played in background when someone rings me, that tells the callee to dial the extension, when my extension is ringing, I can't see the caller id. |
| 08:05.49 | Akelavlk | Anyway, what ISDN hardware are you using? |
| 08:06.01 | wasim | a BRI is 2B+1D, a PRI is 30B+1D |
| 08:06.18 | flenders | is there a way to pass the caller id to the extensions? |
| 08:06.35 | *** join/#asterisk nextime (n=nextime@213-140-6-96.ip.fastwebnet.it) |
| 08:06.45 | wasim | flenders: its passed to the channel, in dialplan its available through a VAR |
| 08:07.09 | *** join/#asterisk darkskiez (n=darkskie@194.247.78.146) |
| 08:08.30 | *** part/#asterisk NoRemorse (n=axel@202.161.68.2) |
| 08:08.49 | lancey | Akelavlk |
| 08:08.53 | lancey | BRI is 2 channels |
| 08:08.58 | lancey | PRI is 24/30 channels |
| 08:08.59 | lancey | basically |
| 08:09.12 | lancey | oh, someone has answered you |
| 08:09.16 | lancey | sorry :) |
| 08:09.30 | lancey | i really had to sleep, though |
| 08:10.53 | Akelavlk | lancey, thanks for answer.. |
| 08:11.30 | Akelavlk | lancey, So simpli we can say that PRI can be connect into E1/T1 right? |
| 08:12.44 | lancey | yes |
| 08:14.13 | Akelavlk | And usually when you buy ISDN from telecom you have BRI type? |
| 08:14.40 | lancey | depends on what you've ordered |
| 08:14.49 | lancey | if you ask for 2 lines, it would be BRI |
| 08:14.59 | lancey | if you ask for 24/30 lines, it would be PRI |
| 08:15.25 | Akelavlk | Aha.. |
| 08:16.48 | Akelavlk | And when I buy some BRI hardware card with 4 BRI ports. It's possible configure any port such as inbound? |
| 08:18.08 | wasim | Akelavlk: yes, each BRI will give you 2 channels, for inbound and outbound (unless restriced by the telco) |
| 08:18.33 | wasim | so a quad-BRI, like from junghanns will give you a total of 8 voice/data channels to work off |
| 08:19.02 | wasim | ~BRI |
| 08:19.03 | jbot | i guess bri is the Basic Rate Interface , an ISDN access interface type composed of two B-channels each at 64 kbps and one D-channel at 16kbps (2B+D). |
| 08:19.09 | wasim | ~PRI |
| 08:19.11 | jbot | methinks pri is Primary Rate Interface, often called T1 or E1 (European Standard). E1 offers 30 ISDN B-Channels a 64kBit/s + 1 D-Channel with 64kBit/s. The T1 has 23 B-Channels + 1 D-Channel. Cards to use with *: T100P, E100P, TE410P, etc. |
| 08:19.28 | wasim | ~docs |
| 08:19.29 | jbot | i heard 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 |
| 08:19.30 | Akelavlk | Aha, so you can configure any of those channels like inbounds. |
| 08:20.08 | *** join/#asterisk shimi (n=shimi@unaffiliated/shimi) |
| 08:21.15 | Akelavlk | Jbot, it's not about asterisk. It's about protocols, standarts etc.. |
| 08:22.45 | shimi | Hi all. I have a weird problem with Asterisk. I installed Asterisk@Home, and did everything through Asterisk Management Portal, so it is probably not a typo on my behalf. The weird thing is that I defined an extension numbered "274". When people from over extensions call that number, what is heard it the "at the time of the tone, the time will be <and the current time>". The other way around (from 274 to other clients), calling works great. It is to b |
| 08:22.46 | shimi | e noted that 274 is a true IP Phone (Grandstream GPX-2000 if it matters), while the other extensions are softphones (like kphone). It looks like the soft/hardphone is not related, because the weird reply is from Asterisk, and it doesn't even _get_ to the phone... Any idea? |
| 08:27.45 | Rowter | any idea what causes this link problem? [app_rxfax.so]Oct 12 03:23:15 WARNING[2335]: loader.c:325 __load_resource: /usr/lib/asterisk/modules/app_rxfax.so: undefined symbol: fax_set_phase_d_handler |
| 08:30.26 | Delvar | do you need fax? |
| 08:30.35 | Delvar | if not no load it |
| 08:31.38 | Rowter | Delvar, I need it am trying to make it work |
| 08:32.08 | Rowter | first it appear error: structure has no member named `verbose' |
| 08:32.16 | Rowter | and I comment it out as some ppl says on google |
| 08:32.31 | Rowter | then it was able to compile but appears that undefined symbol |
| 08:32.49 | Rowter | http://www.soft-switch.org/downloads/spandsp/spandsp-0.0.2pre21/asterisk-1.1.x/ |
| 08:32.56 | *** join/#asterisk Admiral_Snyder (n=aga@G9b8a.g.pppool.de) |
| 08:33.07 | Admiral_Snyder | good morning |
| 08:35.42 | Admiral_Snyder | can one of you help me with my VoIP Router? |
| 08:39.14 | *** join/#asterisk vlrk (n=vlrk@59.93.69.13) |
| 08:39.19 | *** part/#asterisk vlrk (n=vlrk@59.93.69.13) |
| 08:49.51 | Akelavlk | Is there any web site where can I read about PBX hardware? I mean what hardware setup is good for middle size company etc. |
| 08:50.28 | wasim | Akelavlk: depends on the size and requirements |
| 08:50.57 | Akelavlk | I need 30 telephones and I have 1 PRI ISDN link from Telco. |
| 08:51.59 | Akelavlk | 30 ISDN telephones. |
| 08:55.23 | shimi | why ISDN telephones ? |
| 08:56.41 | Akelavlk | Because company has ISDN phones.. It's problem? |
| 08:56.59 | shimi | I don't think Asterisk supports that... |
| 08:57.01 | Delvar | get snom sip hpnes and an asterisk box with te*10p |
| 08:57.09 | shimi | exactly... |
| 08:57.32 | shimi | that's what I am building in these days. a PRI (E1) connected to Digium TE110P with SIP extensions... |
| 08:57.52 | Akelavlk | You mean that asterisk support just analog phones? |
| 08:58.09 | shimi | no, asterisk talks in VoIP protocols, like SIP and IAX2 |
| 08:58.12 | Akelavlk | I think, that doesn't matter if you have soft phone, analog or ISDN.. |
| 08:58.14 | Delvar | if you can afford ISDN phones then you should be able to afford some snom 320's |
| 08:58.38 | Akelavlk | Delvar, snom 320 what is it? |
| 09:00.18 | shimi | http://www.snom.com/snom320_voip_phone.html |
| 09:00.24 | shimi | but... it's not an ISDN phone... |
| 09:00.42 | Delvar | Akelavlk: asterisk supports SIP or IAX the best, as you can jsut plug the SIP phone into teh network and not have dedicated lines or ports int eh server, ie if you have 5 analug phones you need 5 ports in server, that 2x TDM cards :( |
| 09:00.59 | Delvar | the SNOM 320 is a SIP phone, |
| 09:01.07 | Delvar | probably one of the best iv used |
| 09:01.28 | shimi | how much does it cost? :) |
| 09:01.34 | Delvar | depends :) |
| 09:01.48 | shimi | rough estimate ? |
| 09:01.54 | Akelavlk | Delvar,I know. But what if company has already bought phones? And It's usually has. Why they should buy next phones? Because SIP? |
| 09:02.11 | Delvar | oh you already have the phones? |
| 09:02.21 | Akelavlk | Delvar, better solution is use old phones just switch database.. |
| 09:02.22 | shimi | ISDN is kinda obsolete (for the end user...)... |
| 09:02.24 | Delvar | sorry i thought you were GOING to get them :) |
| 09:02.48 | Akelavlk | Yes, company has some old phones.. |
| 09:02.53 | Delvar | hmm |
| 09:03.10 | shimi | cheap VoIP phones cost around $80 per unit... |
| 09:03.17 | Delvar | sell them on ebay and buy snoms?! :) |
| 09:03.23 | Akelavlk | :-) |
| 09:03.30 | Akelavlk | Also good solution..:-) |
| 09:03.38 | Akelavlk | But I need real one.. |
| 09:03.57 | Akelavlk | So question is, what kind of hardware I need.. |
| 09:04.16 | Akelavlk | Some e1/t1 card with two ports? |
| 09:04.19 | shimi | again, if you use asterisk, you need PRI hardware, like Digium TE110P... which supports all kinds of motherboards |
| 09:04.20 | Delvar | i dont know.. iv never used ISDN phones with asterisk |
| 09:04.37 | Akelavlk | And then I need some channel banks? |
| 09:04.54 | Akelavlk | Delvar, What kind of phones do you have? |
| 09:04.54 | Delvar | of some sort yes |
| 09:04.55 | shimi | you need channel banks that can talk with asterisk (probably over SIP) |
| 09:04.58 | Delvar | i have snoms |
| 09:05.09 | Delvar | ISDN to sip ones |
| 09:05.14 | Akelavlk | How many snoms do you have? |
| 09:05.25 | Delvar | here in this office we have about 20 |
| 09:06.06 | Akelavlk | and what link do you have from telco? |
| 09:06.40 | Delvar | well we have 1 PRI and one BRI and 4 analug and then we have our own VOIP (SIP) to voiptalk.org |
| 09:07.14 | Akelavlk | And what kind of hardware do you have? I mean except snoms. |
| 09:09.49 | Delvar | have an asterisk server... standard network gear switches etc... |
| 09:10.42 | Akelavlk | Yes, but you have PRI and BRI, you you need plug this cable to some hardware card. |
| 09:10.52 | iDunno | http://www.dabs.com/productview.aspx?quicklinx=BJ4&refererid=WQ |
| 09:10.53 | Akelavlk | What cards are you using exactly? |
| 09:10.58 | iDunno | ^ anyone used one of those? |
| 09:11.07 | iDunno | and had good experience with it, hopefully ;) |
| 09:12.02 | Akelavlk | Thanks.. |
| 09:12.29 | Akelavlk | What is that? It's one ISDN port? |
| 09:12.46 | *** join/#asterisk tatuman (n=Miranda@joltid-gw.joltid.org) |
| 09:15.38 | Akelavlk | It's one E1/T1 port? It seems like that. |
| 09:15.40 | iDunno | that's just a BRI card. |
| 09:16.44 | Akelavlk | Aha, yes I see.. So I can plug there just one ISDN phone.. Is there version with two ports? |
| 09:16.53 | *** join/#asterisk ful|work (n=fulgas@213.58.130.46) |
| 09:16.57 | ful|work | hey |
| 09:18.03 | *** join/#asterisk folsson (n=filip@h147n1fls32o985.telia.com) |
| 09:19.06 | shimi | There's a good chance that buying the ISDN cards and the computers to connect all the ISDN phones into it, will cost you much more than buying SIP phones :) |
| 09:21.10 | Akelavlk | Hmm, there is just one problem with softphones FAX-ing. |
| 09:24.05 | *** join/#asterisk NoRemorse (n=axel@202.161.68.2) |
| 09:24.11 | shimi | since when are you using a phone in order to fax? |
| 09:24.26 | NoRemorse | tip: dont ever recompile openh323 ince u have it compiled |
| 09:24.29 | shimi | there's an analog adapter to SIP (those are called "ATA"s) |
| 09:25.31 | Akelavlk | Smihi, Company has faxes. |
| 09:26.00 | NoRemorse | Akelavlk u r the guy with isdn2 and a E1 card right? what exactly you trying to do? |
| 09:26.34 | shimi | ISDN faxes ? |
| 09:26.51 | NoRemorse | ISDN is just a gigital phoneline of course u can send faxes |
| 09:26.56 | NoRemorse | *digital |
| 09:27.06 | shimi | I know, my question is if his fax is digital or not |
| 09:27.19 | Akelavlk | NoRemorne.. Yes, may be.. But I have one ISDN30.. |
| 09:27.27 | shimi | if he has an analog fax, he can use an ATA to connect it... |
| 09:27.30 | Akelavlk | And I am looking for hardware.. |
| 09:27.30 | NoRemorse | my guess is its not |
| 09:27.37 | shimi | too bad for him |
| 09:27.48 | shimi | well I'm off |
| 09:28.05 | NoRemorse | well speaking from my own limited experience dont get a digium e1 card, use a cisco 5300 or similar |
| 09:28.25 | Akelavlk | And for testing I need plug there one or two faxes.. |
| 09:28.34 | Akelavlk | At this time I have just TDM400P. |
| 09:28.42 | NoRemorse | standard analogue line fax machine? |
| 09:28.43 | Akelavlk | Why? |
| 09:28.54 | NoRemorse | they dont work over standard sip with codecs you need a special fax codec |
| 09:29.03 | Akelavlk | I thought that Digium has good piece of hardware.. |
| 09:29.13 | NoRemorse | I just had troubles with it thats all. thats not to say you will :) |
| 09:29.42 | Akelavlk | Are you using also FAX-es? |
| 09:29.47 | NoRemorse | nope |
| 09:31.22 | Akelavlk | Wow, CISCO 5300 looks like good piece of hardware.. Is compatible with Asterisk? |
| 09:31.29 | NoRemorse | try it but make sure you use ulaw or alaw codecs, not a compressed one like g729 or gsm |
| 09:31.44 | NoRemorse | it is SIP V2 so yes |
| 09:32.30 | NoRemorse | cisco 5300 is not cheap, if you are on a budget, get the digium card going :) |
| 09:32.52 | Akelavlk | Ok, I am checking that.. |
| 09:33.03 | Akelavlk | 4 T1/E1 is too much.. |
| 09:33.27 | NoRemorse | well 5300 can do at least 8 |
| 09:33.32 | NoRemorse | so forget that hehe |
| 09:33.48 | Akelavlk | Sht.. |
| 09:33.53 | NoRemorse | cost too much |
| 09:33.55 | Akelavlk | It's really too much.. |
| 09:34.00 | NoRemorse | prolly 5 to 10K USD |
| 09:35.07 | Akelavlk | It's possible use E1/T1 and switch calls to FXO ports? |
| 09:35.37 | NoRemorse | of course thats what asterisk is all about |
| 09:35.43 | Akelavlk | Great.. |
| 09:36.16 | Akelavlk | What hardware are you using, BTW? |
| 09:36.44 | NoRemorse | I dont do any hardware telephony interconenct anymore, they are all H323 or SIP |
| 09:36.54 | NoRemorse | so just dell servers :) |
| 09:37.17 | Akelavlk | And you don't have any connection from telco? Just internet? |
| 09:37.19 | NoRemorse | wa susing a cisco 5300 |
| 09:37.24 | NoRemorse | *was using |
| 09:37.45 | NoRemorse | thats right canceled E1's |
| 09:38.09 | Akelavlk | What if you want call to normal tel. number? |
| 09:38.25 | Akelavlk | You meet some gateway, isn't it? |
| 09:38.48 | NoRemorse | you get a sip cts servcie from a provider and send to that |
| 09:38.54 | NoRemorse | so yes a sip gateway |
| 09:39.27 | NoRemorse | btw why exactl;y do you ahve to use FAX to test this? not a normalphone call? has the boss said "the fax still has to work over it" or something? |
| 09:39.29 | Akelavlk | What is it SIP STS service? I mean, I know what SIP is. But I never heard about SIP STS.. |
| 09:39.40 | NoRemorse | CTS = call termination service, a gateway :) |
| 09:39.50 | NoRemorse | commercial gateway service |
| 09:39.56 | Akelavlk | Because, we already test normal phones, and boss wants test also faxes. |
| 09:40.20 | NoRemorse | ok show him this : http://www.voip-info.org/wiki/view/FoIP |
| 09:40.43 | NoRemorse | and explain that fax needs uncompressed VOIP codecs, and uses alot of bandwidth, and only works 50% of the itme u have to resend fax alot |
| 09:40.52 | *** join/#asterisk Assid (n=assid@203.115.64.57) |
| 09:40.53 | NoRemorse | *time |
| 09:41.26 | Akelavlk | Hmm, it's not good.. |
| 09:41.44 | Akelavlk | I need solution which works on 100%.. Ok 90% is enought.. |
| 09:41.51 | NoRemorse | nope, backwards step really, look to fax over IP or t38 relay with a cisco |
| 09:41.54 | Akelavlk | And how much does it cost? SIP CTS? |
| 09:42.04 | *** join/#asterisk wasim (n=wasim@pdpc/supporter/active/wasim) |
| 09:42.08 | NoRemorse | less than an E1 rental :) |
| 09:42.27 | NoRemorse | and you dont have to rent 30 channels even if you are not using them.... |
| 09:42.56 | Akelavlk | Sounds good.. Really good. Where are you from, BTW? |
| 09:43.01 | NoRemorse | Australia |
| 09:43.17 | *** join/#asterisk djin (n=djin@213-132-172-4.multikabel.nl) |
| 09:43.19 | NoRemorse | ask the telco you get the E1 suplied from if they have a "Voip SIP gateway service" |
| 09:43.32 | NoRemorse | *DONT* bother with H323 |
| 09:43.36 | Akelavlk | Hmm, I am from Europe.. May be SIP CTS is something really new.. |
| 09:43.49 | NoRemorse | hehe nah someone there will have it |
| 09:44.05 | NoRemorse | well they'll have H323 anyways.... so just ask |
| 09:44.12 | *** join/#asterisk langals (n=icechat5@196.7.14.183) |
| 09:44.19 | Akelavlk | Ok, I am checking that SIP CTS service.. Looks like pretty good service.. |
| 09:44.30 | NoRemorse | I am talking to abckward telcos in south east asia who have it so Europe should too :) |
| 09:45.09 | NoRemorse | whats yuour telco called? |
| 09:45.16 | NoRemorse | the E1 provider |
| 09:45.48 | Akelavlk | Dial telecom. |
| 09:46.12 | NoRemorse | OMG do they ahve english? |
| 09:47.29 | langals | Hi there...I am having problems with Meetme and delays - I am using meetme with IAX2 softphones. When a delay from a participant occurs, then this remains until the participant hangs up and dials in again, which seems to fix the problem... |
| 09:47.39 | langals | The delays are up to 10 seconds |
| 09:48.10 | langals | I am using jitterbuffer=yes, with typical jitter buffer settings |
| 09:48.20 | Dr_Ray | are you using ztdummy? |
| 09:48.35 | NoRemorse | Akelavlk: http://www.voip-info.org/wiki-VOIP+Service+Providers+Residential says that Dial Telecom Do sip :) |
| 09:48.37 | Akelavlk | NoRemorse, you mean if it's english company? |
| 09:48.37 | langals | Dr_Ray - no - we have a Zaptel card for timing |
| 09:49.37 | Akelavlk | NoRemorse, Wow pretty nice web page.. |
| 09:49.42 | *** join/#asterisk n3u7 (n=neutrin0@CPE000d8802a707-CM0011e6c7edb1.cpe.net.cable.rogers.com) [NETSPLIT VICTIM] |
| 09:49.42 | *** join/#asterisk stkn (i=nobody@gentoo/developer/pdpc.active.stkn) [NETSPLIT VICTIM] |
| 09:49.42 | *** join/#asterisk Rowter (n=SilverDr@201.135.26.195) [NETSPLIT VICTIM] |
| 09:49.42 | *** join/#asterisk asteriskDOTbz (n=logger@64.5.53.45) |
| 09:49.42 | *** join/#asterisk memic (n=memic@chicago089.server4free.de) [NETSPLIT VICTIM] |
| 09:49.50 | Akelavlk | I hope, it's up to date. |
| 09:50.37 | Akelavlk | And what about stability of SIP CTS? I mean, it's stable like E1/T1? |
| 09:51.05 | NoRemorse | is your internet stable? |
| 09:51.24 | Akelavlk | Hmm, It's not 100%... |
| 09:51.25 | lancey | Akelavlk nothing is stable like E1 / T1 |
| 09:51.27 | NoRemorse | keep an ISDN2 for backup :) |
| 09:51.27 | Assid | okay funny thing.. i can hear my self as a robot.. on this box |
| 09:51.28 | Assid | lol |
| 09:51.34 | lancey | especially VoIP |
| 09:51.38 | lancey | *any* kind of it |
| 09:51.42 | NoRemorse | there has to be a tradeoff for thw savings :( |
| 09:51.46 | Akelavlk | NoRemorse :-) |
| 09:51.52 | lancey | as a friend says |
| 09:51.58 | lancey | "voip is voip, keep that in mind" |
| 09:51.59 | lancey | :) |
| 09:52.10 | NoRemorse | OMFG openh323 is taking HOURS to compile |
| 09:52.11 | Assid | time to update cvs |
| 09:52.23 | lancey | NoRemorse yup |
| 09:52.25 | Akelavlk | Yes, I hope one day, VoIP will be more stable than ISDN.. |
| 09:52.36 | NoRemorse | unlikely :) |
| 09:52.43 | wasim | voip is voip, lala la la la, voip is voip, lala la la la (with apologies to the original singer) |
| 09:53.02 | NoRemorse | voip is choip |
| 09:53.12 | wasim | we can use that as a chorus line :) |
| 09:53.19 | lancey | :))))))) |
| 09:53.30 | wasim | or maybe the backing vocals can sing choip, while the lead since voip ... |
| 09:53.52 | Akelavlk | Hmm, Do you want create first VoIP song? |
| 09:54.01 | NoRemorse | rocking robin, choip choip |
| 09:54.48 | NoRemorse | anyone doing a sip conference thingy at astricon this year? |
| 09:56.38 | *** join/#asterisk kiko69 (n=Keith@kauai.sys.pas.earthlink.net) |
| 09:56.47 | NoRemorse | Akelavlk: a more important question, what SPEED is your internet conenction at work? |
| 09:57.01 | NoRemorse | ecxpecially the upstream speed |
| 09:57.10 | Assid | how do you update zaptel when crc_ccitt is using it |
| 09:57.15 | Assid | and you cand rmmod any |
| 09:57.58 | Akelavlk | At work, We have 2 Mbs download and 256 upload.. |
| 09:58.15 | Akelavlk | At home I have 3 Mbs in both ways. ;-) |
| 09:58.26 | Akelavlk | 256 is not enought? |
| 09:58.32 | kiko69 | anyone around to help with realtime unixODBC question? |
| 09:58.54 | Akelavlk | What is good connection when I will choose SIP CTS? |
| 09:59.09 | Assid | besides rebooting;.. is there a way to unload zaptel modules? |
| 09:59.57 | NoRemorse | symetrical, 1MB/1MB minimum |
| 10:00.05 | *** join/#asterisk RoyK (n=roy@80.239.107.70) |
| 10:00.06 | NoRemorse | 256 will only take 3 calls |
| 10:00.10 | Akelavlk | For how many people? |
| 10:00.21 | NoRemorse | 1MB/80K or so |
| 10:00.23 | Akelavlk | But I can use GSM codec, not? |
| 10:00.40 | *** join/#asterisk h4mm3r` (n=h4mm3r@85-18-14-10.fastres.net) |
| 10:00.45 | NoRemorse | so tahts about 13 or 14 calls, go to 2 or 3MBit for 30 calls |
| 10:00.49 | NoRemorse | not for fax :) |
| 10:01.04 | Assid | gsm codec = 56Kbps dialup average |
| 10:01.06 | Assid | per line |
| 10:01.09 | Akelavlk | I know, I know.. Fax is just for testing.. |
| 10:01.56 | Akelavlk | And how many GSM calls can I can get from 256 kbs? |
| 10:02.06 | NoRemorse | ah ok thats fine then. I thought gsm was 12 or 13k? |
| 10:02.34 | Akelavlk | Yes.. 16 Kbs with all things, I guess.. |
| 10:02.44 | Assid | 256 ? 8 calls maybe? |
| 10:03.09 | Akelavlk | What link do you have? |
| 10:03.12 | Assid | me ? |
| 10:03.19 | Akelavlk | Both.. |
| 10:03.26 | Assid | im using asterisk on a dedicated server |
| 10:03.27 | *** join/#asterisk Igbothom (n=HiltonT@static-84.217.240.220.dsl.comindico.com.au) |
| 10:03.40 | Igbothom | g'day all |
| 10:03.44 | NoRemorse | back later |
| 10:03.45 | Akelavlk | Aha.. So no inbound channel? |
| 10:03.46 | *** part/#asterisk NoRemorse (n=axel@202.161.68.2) |
| 10:03.48 | Assid | err... > 100mbit burst.. 1MB/+sec |
| 10:03.55 | Assid | and then re-route calls |
| 10:04.32 | Akelavlk | So you are using asterisk only for internal communication? |
| 10:04.37 | Akelavlk | What link do you have from Telco? |
| 10:05.56 | Assid | they are using 3Mbps.. for some offices |
| 10:06.03 | Assid | and 512kbps for certain others |
| 10:06.24 | Akelavlk | 3Mbs upload and download.. |
| 10:06.26 | Akelavlk | ? |
| 10:06.35 | Assid | 3mbit up and i think 1 down |
| 10:06.48 | Assid | but they are upgrading the plan to the 15mbps |
| 10:07.08 | Akelavlk | 15 Mbs is pretty good speed. |
| 10:07.11 | Assid | yeah |
| 10:07.27 | Assid | the isp actually screwed up and disabled their account by mistake |
| 10:07.36 | Assid | so they are giving some part of it complimentary or something |
| 10:07.46 | Assid | for a much cheaper price |
| 10:09.48 | Akelavlk | That's fine.. It's same in Europe.. But at this time 2 mbs is normal speed of link.. I heard that In Japan they have 150 Mbs.. |
| 10:09.55 | Assid | yeah |
| 10:09.59 | Assid | those guys are psychos |
| 10:10.05 | Assid | ipv6 is main stream there |
| 10:10.17 | Akelavlk | :-) |
| 10:10.36 | Assid | someone i believe came up with ipv8 |
| 10:10.47 | Assid | i saw an article longgggg ago |
| 10:11.04 | Akelavlk | Article about ipv8? |
| 10:11.06 | Assid | they wanted to push v8 instead of v6 and be ready for anything in future |
| 10:11.08 | Akelavlk | Are you sure? |
| 10:11.10 | Assid | yeah |
| 10:11.17 | Assid | didnt make much noise |
| 10:11.21 | Assid | coz.. it fizzed out |
| 10:11.22 | Akelavlk | Unbelievable.. |
| 10:11.40 | Assid | google for ipv8 |
| 10:11.49 | Igbothom | yeah, kinda died out |
| 10:12.29 | Assid | ipv6 apparently is superior |
| 10:12.33 | Akelavlk | http://www.cctec.com/maillists/nanog/historical/9711/msg00138.html |
| 10:12.40 | Assid | was just on it |
| 10:12.40 | Igbothom | it was, from what I remember, kinda unlikely to take off as a protocol because it wasn't that great |
| 10:12.40 | Assid | hhee |
| 10:13.40 | Assid | im waiting for v6 to come main stream |
| 10:13.42 | Assid | would be fun |
| 10:14.20 | *** join/#asterisk veto (i=mdkuser@cpe-66-69-38-192.satx.res.rr.com) |
| 10:14.29 | Akelavlk | Yes, about ipv6 is talking long time.. |
| 10:14.37 | Akelavlk | But still nothing.. |
| 10:14.56 | Assid | certain isps here.. actually give you internal ips.. |
| 10:15.06 | *** join/#asterisk HiltonT (n=HiltonT@static-84.217.240.220.dsl.comindico.com.au) |
| 10:15.14 | Assid | and if you want static.. they charge around 100$ /yr |
| 10:15.22 | Assid | 120 or so |
| 10:16.18 | lancey | hehe |
| 10:16.29 | lancey | i should try that out |
| 10:16.33 | Akelavlk | 120$ per year for static IP? Hmm, it's 70 $ in our country.. |
| 10:16.45 | lancey | Akelavlk it should not cost a penny |
| 10:16.51 | Assid | as i said |
| 10:16.55 | Assid | some isps act stupid |
| 10:16.56 | Assid | hell |
| 10:16.59 | Assid | i get dhcp ip |
| 10:17.05 | Assid | atleast its global |
| 10:17.08 | Akelavlk | Yes, I think so.. They just setup system... |
| 10:17.10 | Assid | and not internal |
| 10:17.16 | lancey | Assid: yup, this is normal |
| 10:17.22 | lancey | but giving private IPs is... |
| 10:17.35 | lancey | i don't know how to say it |
| 10:17.36 | lancey | ;) |
| 10:17.43 | Assid | hehe |
| 10:17.44 | Assid | brb |
| 10:17.44 | JamesDotCom | well |
| 10:17.47 | HiltonT | fcuked |
| 10:17.49 | JamesDotCom | the price of ip's from APNIC |
| 10:17.53 | JamesDotCom | is a little different to ARIN |
| 10:17.55 | Akelavlk | sht :-) Or fck.. |
| 10:17.58 | JamesDotCom | that's the difference |
| 10:18.07 | lancey | if i sell our IPs i would earn more than we earn now :) |
| 10:19.40 | RoyK | morning, morons |
| 10:19.43 | JamesDotCom | you cant own ips in apnic anymore ;( |
| 10:19.44 | Assid | back |
| 10:20.09 | X-Rob | you never could. |
| 10:20.15 | Assid | stupid idiots should start bringing v6 in now |
| 10:20.16 | HiltonT | JamesDotCom; not for quite a number of years now |
| 10:20.21 | Assid | its kinda time to do so |
| 10:20.39 | HiltonT | X-Rob; sure? I know some places who own /24 subnets from APNIC |
| 10:20.48 | X-Rob | you don't own 'em |
| 10:20.49 | JamesDotCom | yeah, you could |
| 10:20.52 | JamesDotCom | years ago |
| 10:20.54 | X-Rob | you have 'em on loan from apnic |
| 10:20.58 | X-Rob | they didn't charge you for 'em |
| 10:21.05 | JamesDotCom | as of several years ago |
| 10:21.06 | HiltonT | and they don't take 'em back |
| 10:21.09 | HiltonT | fair 'nuff |
| 10:21.12 | X-Rob | but they could (and did, quite a few times) take 'em back if they wanted |
| 10:21.18 | JamesDotCom | oh |
| 10:21.23 | JamesDotCom | they *could* take them back |
| 10:21.37 | JamesDotCom | but the owned ones can still be legitimately owned |
| 10:21.40 | JamesDotCom | the trick was |
| 10:21.52 | X-Rob | The did a big cleanup about 5 years ago, looked at all the class C's that weren't globally advertised and tried to contact the owner |
| 10:21.56 | Assid | this sucks |
| 10:21.58 | Assid | i dont know why |
| 10:21.58 | X-Rob | if no reply, they took 'em back |
| 10:22.01 | JamesDotCom | find an allocation to a dead company, re-register their domain name |
| 10:22.11 | Assid | but .. i keep getting funny connections |
| 10:22.19 | JamesDotCom | then create the admin email addresses |
| 10:23.23 | Assid | can someone help me on this thing |
| 10:23.31 | Assid | this box refuses to work the way it should |
| 10:23.43 | X-Rob | assid - 'funny connections' isn't really going to help us diagnose the problem. |
| 10:24.22 | Assid | okay. i called myself.. and then i conferenced it (xten) |
| 10:24.35 | Assid | now when i say something .. it sounds very robotic |
| 10:24.39 | Assid | my other box hwpoever |
| 10:24.55 | Assid | i tried this same thing.. and it works fine.. it echoes.. but doesnt give this crazy crap |
| 10:28.59 | Assid | the echo on another box ialso sounds weird |
| 10:29.27 | [Jedi] | How can I put in a variable the result of the execution of another application? |
| 10:30.19 | *** join/#asterisk razu_ (n=razu@tln-kontor.norby.ee) |
| 10:30.48 | Delvar | ' another application?' ypou mean an AGI script? |
| 10:31.31 | [Jedi] | no |
| 10:31.44 | [Jedi] | I have a TxFax execution, which returns 0 or -1 |
| 10:31.56 | [Jedi] | I'd like to put that result in a variable |
| 10:32.41 | Delvar | duno |
| 10:33.04 | [Jedi] | would like to avoid agi |
| 10:33.04 | [Jedi] | :) |
| 10:33.11 | Delvar | usualy things either go to priority +101 or set a variable like ${DIALSTATUS} |
| 10:34.25 | Assid | hrmm |
| 10:34.32 | Assid | this box isnt seemt o be running right |
| 10:37.34 | *** join/#asterisk Tili (i=Tili@202-133-67-218-dialup.sat.net.pk) |
| 10:39.24 | [Jedi] | Anyone using TxFax here? |
| 10:39.39 | *** part/#asterisk Akelavlk (n=jansun@82.119.239.141) |
| 10:39.56 | lancey | Delvar |
| 10:40.00 | lancey | +101 will go away soon |
| 10:40.12 | HiltonT | anyone running AstLinux 0.2.8? I can't seem to get to its web inmterface (fresh HDD install) |
| 10:40.13 | lancey | so if you develop something, don't count on that |
| 10:40.56 | lancey | there will be an option for old-style +101 behaviour, though |
| 10:41.24 | *** join/#asterisk zotz (n=zotz@24.231.36.100) |
| 10:44.51 | lancey | anyone used cisco ubr924 |
| 10:44.58 | lancey | for voice over h323? |
| 10:48.29 | Delvar | lancey: i know but i was jsut saying thats how things do it |
| 10:49.25 | lancey | okay |
| 10:49.34 | lancey | i just wanted to point out :() |
| 10:51.36 | X-Rob | I think openpbx is starting with a clean slate and with _no_ +101's |
| 10:52.49 | lancey | i'm personally not glad about the fact of * development branching... |
| 10:53.45 | HiltonT | I'm happy to stick (well, *(*just** starting, actually) with * unless there's some ****really**** good reason to look at OpenPBX |
| 10:55.07 | *** join/#asterisk riksta (n=rick@62.6.163.90) |
| 10:56.05 | X-Rob | HiltonT - there isn't a good reason, unless there is one. |
| 10:56.14 | HiltonT | lol, very true |
| 10:56.18 | X-Rob | asterisk == minix, openpbx == linux? |
| 10:57.13 | X-Rob | for most people, stick with asterisk. |
| 10:57.17 | lancey | :) |
| 10:57.33 | X-Rob | a lot of the non-digium developers have moved over to openpbx, because it gives them a lot more freedom. |
| 10:57.43 | X-Rob | but nothing's going to be useful user-wise for a while yet |
| 10:57.47 | X-Rob | eg, today, gnu autoconf works. |
| 10:57.53 | lancey | define "more freedom" |
| 10:58.03 | X-Rob | they can add other GPL things into it |
| 10:58.10 | HiltonT | hrmph - anyone running Astlinux 0.2.8? I canna get to the web interface - fresh HDD install, only extif (edited rc.conf), but no web interface |
| 10:58.12 | X-Rob | like readline, automake etc. |
| 10:58.32 | X-Rob | HiltonT - you asked the same question before but didn't offer any debugging information |
| 10:58.37 | X-Rob | eg 'my web server isn't running' |
| 10:58.44 | X-Rob | or 'how do I find out if my web server is running' |
| 10:58.46 | HiltonT | mini-httpd is running |
| 10:59.02 | *** join/#asterisk nexis (n=nexis@12-219-60-252.client.mchsi.com) |
| 10:59.14 | *** join/#asterisk Laibsch (n=Laibsch@p54B9972E.dip0.t-ipconnect.de) |
| 10:59.20 | X-Rob | uh. that 'automake' was, obvously, meant to be 'autoconf' |
| 10:59.40 | *** join/#asterisk maik (n=maik@bfs.cs.uni-sb.de) |
| 10:59.45 | HiltonT | huh? |
| 10:59.52 | X-Rob | HiltonT - not to you |
| 10:59.57 | HiltonT | :) |
| 11:00.01 | X-Rob | HiltonT - so. You're saying the web interface isn't installed? |
| 11:00.31 | HiltonT | well, it should be, I'd guess. there's SFA (actually, less than that) documentation for Astlinux, which is a bit disappointing |
| 11:00.43 | HiltonT | maybe I should try learning on A@H |
| 11:01.08 | HiltonT | and mini-httpd is listed as running (top) but I canna get to https://pbx |
| 11:01.31 | HiltonT | (and yes, I mean https://ip.of.the.pbx) |
| 11:01.40 | lancey | https? |
| 11:01.51 | HiltonT | nor can I get to http://blah |
| 11:01.55 | lancey | ahm |
| 11:02.18 | HiltonT | were there some docs, maybe that'd help some :) |
| 11:03.04 | HiltonT | there's a UserGuide for 0.2.6, but that's next to useless, really. More like a really, really quickstart guide. :) |
| 11:03.18 | *** join/#asterisk MikeJ[Laptop] (n=ircatjer@d14-69-8-30.try.wideopenwest.com) |
| 11:03.55 | X-Rob | what makes you think there is a web interface? |
| 11:05.22 | HiltonT | the fact that the doc says "I have included a phpconfig-like GUI... the full URL is https://blah" |
| 11:06.06 | X-Rob | I think that's possibly a lie? 1: Are you using mozilla/firefox to try to connect to the web server? |
| 11:06.21 | X-Rob | because IE doesn't tell you if it's 404, host not found, or it's just grumpy |
| 11:07.01 | HiltonT | hhmmm, methinks there's a bug in the doc - if I have only one interface, disabling INTIF makes no sense, as that's the one I'd need running - this is what the docs recommends. Methinks maybe disabling EXTIF and using INTIF would make way more sense |
| 11:07.19 | HiltonT | and EXTIF will be firewalled, I'd imagine |
| 11:07.35 | HiltonT | hence why I cannot get to it!!! |
| 11:09.06 | HiltonT | tho that's what rc.conf suggests to do - as the docs say, but that makes no sense to me |
| 11:09.26 | *** join/#asterisk h4mm3r` (n=h4mm3r@85-18-14-10.fastres.net) |
| 11:10.02 | [Jedi] | Anyone using TxFax here? |
| 11:13.08 | *** join/#asterisk sivana (n=sivana@mixdown.ca) |
| 11:13.17 | tzafrir_laptop | bah, seems like the mail server for cohens.org.il won't be active for today and tommorow :-( |
| 11:19.18 | lancey | anyone here using Topex GSM gateways? |
| 11:21.23 | X-Rob | ? |
| 11:23.44 | *** join/#asterisk RaYmAn-Bx (i=rayman@x1-6-00-40-63-da-39-3f.k191.webspeed.dk) |
| 11:25.21 | *** join/#asterisk PakiPenguin (n=pingu@linuxpakistan/admin/pakipenguin) |
| 11:25.24 | PakiPenguin | hello everyone |
| 11:25.43 | *** join/#asterisk _omer (i=p@203.215.180.250) |
| 11:25.45 | _omer | hi |
| 11:25.52 | PakiPenguin | can anyone point me to the php script that shows the current sip channels? |
| 11:26.07 | lancey | !? |
| 11:26.17 | iDunno | why do you need a php script? |
| 11:26.36 | PakiPenguin | hehe perl / cgi would do? ijust need it off the web internet |
| 11:26.39 | PakiPenguin | interface* |
| 11:26.44 | wasim | exec (asterisk -rx sip show channels) |
| 11:26.53 | PakiPenguin | oh :) |
| 11:27.09 | _omer | does music on hold (MP3 files) takes System RAM ? ? |
| 11:27.11 | iDunno | sip show channels appears to gimme nothing, sip show peers is handy though ;) |
| 11:27.38 | wasim | _omer: no, it borrows it from the neighbours refrigerator |
| 11:27.44 | _omer | lol |
| 11:27.58 | _omer | well I mean to say....I have 20 MP3 in music on hold folder |
| 11:28.06 | wasim | _omer: no, it shouldn't |
| 11:28.07 | _omer | and If I have 1 MP3 .. |
| 11:28.38 | _omer | but having 20 MP3 in music on hold ..I am getting poor quality on music on hold.. |
| 11:29.06 | _omer | than...having 1 MP3 file.......weird..:D |
| 11:29.30 | PakiPenguin | _omer: join all the mp3s into one mp3 file |
| 11:29.32 | PakiPenguin | a big one :D |
| 11:30.12 | wasim | _omer: make 20 copies of the same file and see at what number it gets bad |
| 11:30.30 | PakiPenguin | hehe |
| 11:31.08 | _omer | well....If I delete those MP3 ..I still get music on hold until I stop and restart asterisk.. |
| 11:32.32 | wasim | _omer: thats coz mpg123 has taken it into account |
| 11:33.09 | _omer | hmmm ...or may be Asterisk doesnt like "slipknot" .. ;) let me check Madonna |
| 11:33.18 | PakiPenguin | :p |
| 11:33.22 | PakiPenguin | oh well |
| 11:36.39 | *** join/#asterisk nexis (n=nexis@12-219-60-252.client.mchsi.com) |
| 11:40.57 | *** join/#asterisk Poincare (n=jefffnod@dD5779BD2.access.telenet.be) |
| 11:47.48 | tzafrir_laptop | _omer, don't use mp3s for MOH. convert it to WAV and do the necessary transcoding boforehand |
| 11:48.51 | tzafrir_laptop | s/WAV/wav/ if we stick with asterisk terminology |
| 11:48.54 | _omer | I think u r right.... |
| 11:49.08 | _omer | getting bad quality ..... |
| 11:49.25 | _omer | even sometimes .. music on hold stopped automatically |
| 11:49.43 | lancey | have a look at raw player |
| 11:49.59 | _omer | yeah I have converters...I check it out |
| 11:51.59 | *** join/#asterisk ragunath (n=ragunath@satpool37.fokus.fraunhofer.de) |
| 11:53.10 | ragunath | i created a local Ad-Hoc network and using a sip WLAN phone and my laptop, i masquride packets form the local wireless network to internet , the phone works when i call some one , but i cant call the sip phone , any idea why? |
| 11:53.12 | *** join/#asterisk oden (n=oden@194-237-146-22.customer.telia.com) |
| 11:55.08 | *** join/#asterisk oden (n=oden@194-237-146-22.customer.telia.com) |
| 11:56.18 | *** join/#asterisk pa (n=Paolo@unaffiliated/pa) |
| 12:06.22 | *** join/#asterisk Juul (n=Juul@81.7.147.193) |
| 12:15.50 | *** join/#asterisk e3g (i=ee@u15157627.onlinehome-server.com) |
| 12:15.53 | e3g | hello |
| 12:16.05 | ragunath | hi any idea about the PCI satilite modem with linux driver |
| 12:16.10 | e3g | I just downloaded SOX ... how to install it ? :$ |
| 12:16.22 | e3g | make install does nothing |
| 12:17.37 | e3g | any help ?? |
| 12:17.41 | festr_ | hello, anyone here using bristuff with E1? i've problem with progress: ISDN phone -> PRI E1, there is no ring |
| 12:17.45 | festr_ | any suggestion? |
| 12:17.51 | festr_ | idea? |
| 12:17.51 | festr_ | :) |
| 12:18.26 | e3g | festr_ did you try progressinband=yes ?? or inbandprogress I think |
| 12:18.30 | *** join/#asterisk coppice (n=chatzill@48.201.17.210.dyn.pacific.net.hk) |
| 12:18.37 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
| 12:18.45 | festr_ | e3g: you mead priindication = passthrough |
| 12:18.55 | festr_ | e3g: priindication = inband? |
| 12:19.18 | e3g | no |
| 12:19.33 | e3g | sip.conf [general] ...add |
| 12:19.50 | festr_ | e3g: but i'm talking about bri -> pri |
| 12:19.59 | e3g | no asterisk ? :) |
| 12:20.08 | festr_ | yes asterisk :) but bristuffed :) |
| 12:20.09 | e3g | okey |
| 12:20.14 | e3g | :) |
| 12:20.23 | e3g | I have stucked in sox installation :D |
| 12:20.26 | festr_ | priindication = passthrough |
| 12:21.06 | festr_ | maybe this could help |
| 12:21.06 | festr_ | <PROTECTED> |
| 12:21.06 | festr_ | Oct 12 14:20:44 DEBUG[15368]: chan_zap.c:4544 zt_indicate: Received AST_CONTROL_PROGRESS on Zap/33-1 |
| 12:21.09 | festr_ | Oct 12 14:20:44 DEBUG[15368]: app_dial.c:392 wait_for_answer: Dunno what to do with control type 15 |
| 12:21.12 | festr_ | <PROTECTED> |
| 12:22.09 | festr_ | e3g: what linux distro do you have? |
| 12:22.26 | e3g | RH |
| 12:22.51 | e3g | I am really fedup of these Installation stuff |
| 12:24.23 | tzanger | e3g: huh? |
| 12:24.26 | tzanger | what installation stuff |
| 12:24.42 | e3g | how to do MAKE INSTALL in sox |
| 12:24.47 | e3g | I just downloaded it |
| 12:24.50 | tzafrir_laptop | apt-get install sox |
| 12:24.59 | tzafrir_laptop | or yum install sox |
| 12:25.44 | e3g | I think yum is the BEST INVENTION in linux ;) |
| 12:26.16 | lancey | ;) |
| 12:26.18 | *** part/#asterisk oden (n=oden@194-237-146-22.customer.telia.com) |
| 12:26.39 | *** join/#asterisk limbique (n=limbique@nl-ifw-oss.orcagroup.com) |
| 12:26.45 | limbique | hi |
| 12:26.47 | e3g | thanks tzanger |
| 12:27.00 | e3g | thanks tzafrir |
| 12:27.13 | coppice | e3g: yum is certainly the biggest step towards an easy life with RPM based systems |
| 12:27.16 | limbique | anyone knows how a conference lookslike in asterisk manager connection? |
| 12:27.31 | limbique | is it link A with B |
| 12:27.37 | limbique | and link A with C ? |
| 12:28.06 | tzanger | coppice: agreed |
| 12:28.40 | tzanger | limbique: IIRC it's more like A <--> Conf, B <--> Conf and C <--> Conf |
| 12:28.53 | *** join/#asterisk lehel (n=asd@82.79.20.17) |
| 12:29.13 | limbique | ok, like a conf channel where all connected to |
| 12:29.15 | [Jedi] | coppice, can I ask you a few questions? I'm modifying TxFax but I'm confused about some concepts |
| 12:29.35 | coppice | go ahead |
| 12:30.01 | [Jedi] | I'm trying to do that TxFax notices asterisk if the fax has been really successfully sent or not |
| 12:30.10 | [Jedi] | I'm modifying the phase E handler |
| 12:30.29 | [Jedi] | and registering the "int result" into an asterisk var |
| 12:31.03 | lehel | hello |
| 12:31.04 | [Jedi] | is that conceptually correct? that way, will I be able to infer if the fax has been correctly sent or not? |
| 12:31.19 | coppice | that won't get you very far right now, as the result is not that meaningful. |
| 12:31.28 | [Jedi] | uhm |
| 12:31.30 | [Jedi] | :( |
| 12:32.14 | [Jedi] | I've looked also at t30.h |
| 12:32.21 | [Jedi] | there's an opaque pointer |
| 12:32.24 | coppice | when I have gone through spandsp making the result always meaningful, making txfax report it is the obvious next step :-) Actually 0.0.3 does this, but it is a work in progress |
| 12:33.32 | [Jedi] | by modifying that opaque pointer in the t30 structure to a "status code" in spandsp, will I be able to do what I want? |
| 12:34.09 | [Jedi] | <PROTECTED> |
| 12:34.23 | [Jedi] | or *phase_e_user_data |
| 12:34.33 | *** join/#asterisk mbranca (n=matteo@host-210-mi.espia-net.net) |
| 12:34.56 | [Jedi] | it's a little hack'ish but do you think that way I will be able to put into asterisk the needed info? |
| 12:35.09 | [Jedi] | I'm not a very good C coder |
| 12:35.27 | [Jedi] | but for me it's really important to get the status of the fax |
| 12:36.13 | coppice | spandsp is not correctly setting the value passed to the phase E handler. until that is sorted out, there is nothing you can do. If you look at the source fo 0.0.3 you will see it does set the code in a number of places, but the changes are incomplete |
| 12:36.56 | [Jedi] | uhm |
| 12:36.58 | [Jedi] | :( |
| 12:37.07 | tatuman | hi´ |
| 12:38.06 | tatuman | does anyone know if it is possible to use a dsp card to code/decode a codec with asterisk, in order for example to reduce the cpu load of a meetme conference? |
| 12:38.31 | [Jedi] | coppice: is spandsp 0.0.3 'usable' right now? |
| 12:38.38 | *** join/#asterisk oden (n=oden@194-237-146-22.customer.telia.com) |
| 12:38.40 | [Jedi] | coppice: if not, can I help to finish it? |
| 12:39.01 | lancey | tatuman you don't have a choice but use faster CPUs |
| 12:39.11 | *** part/#asterisk oden (n=oden@194-237-146-22.customer.telia.com) |
| 12:39.17 | lancey | there are no such cards that i'm aware of |
| 12:39.34 | tatuman | lancey: i use a 3.0ghz xeon |
| 12:39.41 | coppice | earlier I said "Actually 0.0.3 does this, but it is a work in progress" |
| 12:39.43 | tatuman | and it is not powerful enough |
| 12:39.46 | lancey | tatuman well add a second CPU |
| 12:39.49 | lancey | or go Opteron |
| 12:40.06 | [Jedi] | tatuman: you may try Aculab's cards, I think they use DSP's |
| 12:40.29 | tatuman | Jedi: for what codecs, do u know? |
| 12:40.30 | lancey | [Jedi] using such card would require a major rework of the codecs, i think |
| 12:40.34 | lancey | in order to utilize |
| 12:40.45 | lancey | i haven't seen such work |
| 12:40.52 | [Jedi] | lancey: they support asterisk |
| 12:40.56 | [Jedi] | lancey: http://www.aculab.com/products/asterisk/ |
| 12:41.16 | lancey | nice |
| 12:41.20 | lancey | lemme see! |
| 12:41.24 | *** join/#asterisk scanna (n=scannach@81-174-16-211.f5.ngi.it) |
| 12:41.29 | tatuman | so how can i improve a meetme conference |
| 12:41.46 | tatuman | right know it can only hold like 50 clients using ilbc |
| 12:41.54 | tatuman | right now * |
| 12:42.01 | [Jedi] | tatuman: these cards are for TDM, not VoIP |
| 12:42.21 | [Jedi] | Aculab has Prosody X cards for VoIP, but I don't think they're supported under asterisk |
| 12:42.46 | lancey | yup |
| 12:42.51 | lancey | i was just gonna say that |
| 12:42.52 | tatuman | hhhmmm, any ideas to improve a meetme conference just with voip clients? |
| 12:43.05 | lancey | and what they process is maybe echo cancellation and such things |
| 12:43.24 | tatuman | i tested ser+sems and it even scales worst |
| 12:45.16 | *** join/#asterisk Ahrimanes (n=aron@hobbes.bsd-dk.dk) |
| 12:45.26 | tatuman | so no one has any ideas? |
| 12:45.38 | HiltonT | the new Digium cards have onboard echo cancellation (BRI cards, that is) |
| 12:45.59 | [Jedi] | Digium have BRIs? |
| 12:46.02 | [Jedi] | ???? |
| 12:46.15 | tatuman | i just wanna have the maximum users possible in a conference |
| 12:46.21 | coppice | tatuman: unless * conferencing has been changed a lot recently, it does these things in a very inefficient way. If everyone uses iLBC there should just be 50 decompressions and 1 compression. I don't think that happens. |
| 12:46.54 | *** join/#asterisk akimbo64 (n=akimbo64@193.251.169.129) |
| 12:47.42 | akimbo64 | Hello world, Can somone help me on compilling ChanSpy app please ? |
| 12:47.55 | tatuman | coppice: there has to be 50 decompressions and 50 compressions because your voice has to be subtracted to the mix of all voices |
| 12:48.21 | HiltonT | TE405P, TE410P are the old cards, the TE406P and TE411P are the new cards from Digium with Echo Cancellation |
| 12:48.29 | tatuman | anyone has done some work with MeetMe? |
| 12:48.37 | lancey | HiltonT |
| 12:48.41 | HiltonT | sorry - PRI |
| 12:48.42 | lancey | they are PRI |
| 12:48.42 | HiltonT | silly me |
| 12:48.44 | [Jedi] | HiltonT: these are PRI interfaces |
| 12:48.44 | HiltonT | oops |
| 12:48.46 | [Jedi] | :) |
| 12:48.47 | lancey | :)) |
| 12:48.52 | HiltonT | lysdecic figners! |
| 12:49.00 | lancey | and mind :R |
| 12:49.01 | Ahrimanes | haha |
| 12:49.21 | lancey | it happens all the time, don't worry, especially to me :) |
| 12:49.27 | HiltonT | nah, no Digium BRI cards (yet) |
| 12:49.30 | [Jedi] | coppice: just compiled 0.0.3 rpm for testing... any known big bug ? |
| 12:49.39 | HiltonT | yeah - I know the diff, but keep typing the wrong one! |
| 12:49.50 | coppice | tatuman: sorry. there has to be more than one comrpession, but not 50. Most people are not contributing at any moment |
| 12:50.26 | lancey | coppice: why not post a feature request on the bug tracker? |
| 12:50.44 | lancey | it might remind someone capable of doing that :) |
| 12:50.58 | tatuman | coppice: but asterisk has not any silence suppression engine |
| 12:51.05 | tatuman | to take the silence out |
| 12:51.09 | *** join/#asterisk nfi|ermes (n=ermsewrk@217.220.121.62) |
| 12:51.12 | *** join/#asterisk venkat (n=venkat@212.159.2.3) |
| 12:51.33 | tatuman | and the clients are always sending silence packets |
| 12:51.42 | coppice | Jedi: if I knew that it wouldn't still be a work in progress, would it? :-) 0.0.3 is gaining T.38 capabilities |
| 12:52.10 | HiltonT | damn, no silence suppression - that coming in 2.0? |
| 12:52.11 | coppice | tatuman: the conferencer suppresses the quiet channels. if it didn't 50 lots of background noise would build into a roar |
| 12:52.18 | lancey | 2.0? |
| 12:52.20 | lancey | :))) |
| 12:52.27 | [Jedi] | coppice: hehe ok |
| 12:52.33 | HiltonT | 1.2.0 - me needs sleep, methinks! |
| 12:52.34 | lehel | how do i remove manually a call forwarding? |
| 12:52.37 | HiltonT | (or more beer) |
| 12:52.39 | [Jedi] | coppice: can I help in any way to get it finished? |
| 12:53.03 | lancey | [Jedi] sure, send him some beer |
| 12:53.04 | lancey | :) |
| 12:53.09 | [Jedi] | apart from beer |
| 12:53.10 | HiltonT | lol |
| 12:53.12 | [Jedi] | =D |
| 12:53.19 | tatuman | coppice: i just tested it, and i doesnt suppresses this packets |
| 12:53.23 | coppice | Jedi: probably not right now. have you seen chan_fax? maybe that would be useful to you |
| 12:53.24 | lancey | well you could send money as well :) |
| 12:53.30 | HiltonT | "Beer O'Clock" just made it into the new Macquarie Dictionary :) |
| 12:53.41 | coppice | tatuman: how did you tell? |
| 12:53.46 | [Jedi] | coppice: no, I haven't |
| 12:54.01 | *** join/#asterisk zyke (n=zakforev@84-45-132-117.no-dns-yet.enta.net) |
| 12:54.34 | [Jedi] | google doesn't kow a lot about it |
| 12:55.05 | lehel | can i remove manually a zap extensions CF? |
| 12:56.17 | [Jedi] | coppice: where can I find more info about it? |
| 12:56.21 | lehel | i re-doit the extensions but it is still there.. |
| 12:56.45 | akimbo64 | anyone has tested chanspy before please ? |
| 12:57.22 | *** join/#asterisk tdonahue (n=tdonahue@64.201.13.50) |
| 12:58.09 | tdonahue | hi all, has anyone else using an asterisk realtime configuration for your voicemail found that your users can't change their voicemail passwords? |
| 12:58.42 | tdonahue | oh this is with CVS-HEAD as of about 2 weeks ago |
| 12:59.04 | *** join/#asterisk Lee__ (n=Lee__@dsl254-122-010.nyc1.dsl.speakeasy.net) |
| 13:00.11 | *** join/#asterisk gambolputty (n=gambolpu@72.240.241.108) |
| 13:01.13 | coppice | Jedi: I can't find where the author put it, but I just put a copy of the latest version I have in http://www.soft-switch.org/downloads |
| 13:01.15 | Katty | mew. |
| 13:01.32 | *** join/#asterisk Laibsch (n=Laibsch@p54B9972E.dip0.t-ipconnect.de) |
| 13:02.01 | [Jedi] | coppice: what if I put a fax-id in the LOCALHEADERINFO in asterisk, and then execute a command in spandsp just after "Success - delivered n pages" messages to stderr, with the LOCALHEADERINFO (fax_get_header_info) as a parameter? |
| 13:02.07 | Katty | mew? |
| 13:02.29 | [Jedi] | (thanks for chan_fax by the way) |
| 13:02.44 | coppice | Jedi: oooo! messy :-) |
| 13:03.25 | *** join/#asterisk toddf (n=toddf@net-66-210-104-252.theshop.net) |
| 13:04.01 | [Jedi] | coppice: absolutely messy and hack'ish but... do you think that way I'll get the notification on correct fax sending? |
| 13:04.19 | [Jedi] | it would only be a temporary solution until 0.0.3 is finished... :/ |
| 13:04.35 | *** join/#asterisk MikeJ[Laptop] (n=ircatjer@mi.origenfinancial.com) |
| 13:04.38 | coppice | Jedi: spandsp has had an almost complete class 1 FAX modem implementation for over a year. Suddenly people have sprung to life doing things with it. Lee Howard has debugged and polished things. He also produced something at iaxclient.sourceforge.net to use it. Someone else make chan_fax to use it |
| 13:05.21 | MikeJ[Laptop] | yay coppice |
| 13:05.46 | [Jedi] | coppice: the fact is that your piece of software is awesome; an E1 fax board costs zillions and a digium E1 board costs nothing |
| 13:06.09 | [Jedi] | coppice: I guess people has noticed the huge advantage spandsp provides |
| 13:07.01 | syle | the documentation on configuring spansp was rather limited, maybe you can post on voip-info.org |
| 13:07.08 | *** join/#asterisk EvilRick (n=Ev1lRick@196-28-86-129.wdsl.co.za) |
| 13:07.17 | [Jedi] | coppice: by the way... what does chan_fax do? create /dev/ttyX devices which are class-1 fax modems???? |
| 13:07.33 | EvilRick | hey guys.. I get "app_dial.c:412 wait_for_answer: Unable to forward voice" when trying to dial out on my PRI line. any ideas? |
| 13:07.40 | EvilRick | I'mm nto sure taht all isconfigured well |
| 13:08.08 | EvilRick | I'm not sure that all is configured well. |
| 13:08.47 | EvilRick | ztcfg shows me all 32 channels of my E1 so I assume the driver works correctly |
| 13:08.51 | coppice | Jedi: basically. They don't have those particular names, though |
| 13:09.16 | [Jedi] | and does it work??? |
| 13:09.31 | [Jedi] | if it works, it's *the solution* for faxing |
| 13:09.40 | [Jedi] | being able to use hylafax or efax with digium boards... |
| 13:09.54 | coppice | Jedi: the linux developers have changed the pseudo terminal system so it no longer offers reliable operation for this kind of thing, but we have to live with it :-( |
| 13:10.27 | [Jedi] | uhm |
| 13:10.33 | [Jedi] | so in 2.6 it's flawed |
| 13:10.45 | coppice | you need the latest spandsp, with all the fixes for T.31 (class 1 FAX modems) which have been fed back to me. (0.0.2pre21). Someone just pointed out an error in that, so pre22 might be coming shortly. |
| 13:10.57 | *** join/#asterisk iCEBrkr (i=icebrkr@rrcs-24-129-130-158.se.biz.rr.com) |
| 13:10.59 | [Jedi] | ok |
| 13:11.09 | EvilRick | I get "Executing Dial("SIP/1012-f42d", "Zap/g1/6346772")" then "-- Called g1/6346772" and immediately after "-- Channel 0/1, span 1 got hangup" |
| 13:11.41 | [Jedi] | going to RPM it :) |
| 13:12.01 | *** join/#asterisk leszq (n=leszq@82.177.97.254) |
| 13:12.05 | leszq | hello :) |
| 13:12.08 | coppice | not flawed exactly. They added a very flexible system to allow the dynamic creation and deletion of psuedo ttys for things like X11 console. At the same time they have deprecated a proper way to handle static pseudo ttys |
| 13:12.26 | [Jedi] | ah |
| 13:12.49 | [Jedi] | but when using 2.4 there's no problem, right? |
| 13:12.55 | tzanger | I thought that udev was to solve all of those problems |
| 13:12.57 | EvilRick | anyone know if a PRI line has 4 or 2 cables? |
| 13:13.09 | tzanger | EvilRick: 1, 4-wire (2 pair) cable |
| 13:13.23 | tzanger | well actually it's an 8-wire (4 pair) cable but only two pairs are used |
| 13:13.31 | coppice | tzanger: udev solves problem, but the new psuedo-tty scheme created this one. |
| 13:13.32 | EvilRick | tzanger: thanks mate |
| 13:14.10 | coppice | the old pseudo-tty scheme is still there, but it is deprecated, and the distros don't build it |
| 13:14.35 | EvilRick | if the pri on the telco side is "uncofigured" will asterisk just hangup after trying to dial? |
| 13:15.06 | coppice | EvilRick: PRIs use one 4 core cable, or 2 co-ax cables |
| 13:15.16 | [Jedi] | coppice: by the way, do you remember last week when I said I hated spandsp because I couldn't make it work? it was working, but as it logs to stderr I didn't see any message with asterisk -c :(((( |
| 13:17.05 | [Jedi] | EvilRick: pri debug span 1 |
| 13:17.40 | [Jedi] | EvilRick: that way you'll get the Q.931 messages you're sending/receiving to/from the remote telco switch |
| 13:19.23 | EvilRick | cool :) |
| 13:22.26 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
| 13:24.18 | [Jedi] | coppice: with spandsp 0.0.3, chan_fax compiles correctly; but with 0.0.2, it doesn't find "T31_CALL_EVENT_TIMEOUT" |
| 13:24.51 | e3g | hi |
| 13:24.52 | coppice | Jedi: I will check that out |
| 13:25.16 | coppice | Jedi: are you using pre21? |
| 13:25.19 | e3g | I have installed SOX .....but I dont have SOXMIX command to mix the in and out channels |
| 13:25.27 | e3g | :-/ |
| 13:25.34 | e3g | -bash: soxmix: command not found |
| 13:26.51 | lehel | where is restored tha fact if i'm doing a CF? |
| 13:27.02 | *** join/#asterisk eziman (i=superop@64.116.231.226) |
| 13:27.19 | [Jedi] | coppice: neither pre20 nor pre21 define it |
| 13:28.14 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 13:28.29 | e3g | any help |
| 13:28.30 | puzzled | hi |
| 13:29.41 | *** join/#asterisk Delvar (n=irc@host-83-146-53-34.bulldogdsl.com) |
| 13:30.32 | lehel | hi puzzled: could you tell me pls where is restored tha fact that i made a CF? |
| 13:30.37 | coppice | Jedi ah! chan_fax has not been aligned with the latest changes to the t.31 code. |
| 13:31.19 | puzzled | lehel: CF? |
| 13:31.24 | lehel | call forwrading |
| 13:32.05 | *** join/#asterisk shido6 (n=shido6@d221-68-210.commercial.cgocable.net) |
| 13:32.29 | e3g | any help regarding SOX ? |
| 13:32.38 | *** join/#asterisk ManxPower (n=eric@slip-12-65-36-138.mis.prserv.net) |
| 13:32.54 | iCEBrkr | e3g: That's kinda odd |
| 13:33.04 | bjohnson | lehel: what is the question? |
| 13:33.22 | iCEBrkr | e3g: Which version of sox? |
| 13:33.48 | *** join/#asterisk MnDBnDr (n=MnDBnDr@198.234.224.6) |
| 13:34.11 | e3g | <PROTECTED> |
| 13:34.21 | iCEBrkr | e3g: Which distro you running? |
| 13:34.40 | e3g | RED HAT ....one of machine have sox-12.17.7 and it works with SOXMIX |
| 13:34.47 | *** join/#asterisk RaYmAn-Bx (i=rayman@x1-6-00-40-63-da-39-3f.k191.webspeed.dk) |
| 13:34.56 | EvilRick | ok I plugged in another pri line and I get the following |
| 13:34.59 | EvilRick | "PRI Error: We think we're the network, but they think they're the network, too" |
| 13:35.01 | iCEBrkr | So did you RPM install it or compile the source? |
| 13:35.31 | e3g | this time RPM ..... last time Compile... |
| 13:35.33 | EvilRick | I change /etc/asterisk/zapata.conf to use CPE signalling and I get |
| 13:35.37 | ManxPower | EvilRick, Well, then make asterisk pri_cpe |
| 13:35.44 | iCEBrkr | e3g: If you built it from source and did a 'make install' you might want to look in /usr/local/bin/ for soxmix |
| 13:35.58 | e3g | yes I did make install...nothing happened |
| 13:36.03 | e3g | let me do again |
| 13:36.05 | EvilRick | "PRI Error: We think we're the CPE, but they think they're the CPE too." |
| 13:36.10 | e3g | I have the source as well |
| 13:36.18 | ManxPower | EvilRick, Somewhere there is a loopback on your PRI |
| 13:36.29 | e3g | [root@localhost sox-12.17.8]# make install |
| 13:36.29 | e3g | make: *** No rule to make target `install'. Stop. |
| 13:36.29 | e3g | [root@localhost sox-12.17.8]# |
| 13:36.52 | EvilRick | ManxPower, is it possible I hooked the cables up incorrectly? |
| 13:36.57 | iCEBrkr | e3g: It's been my luck that anything I install using the source gets put into some /local/ directory. So it's been habit to ./configure --prefix=/usr |
| 13:37.03 | EvilRick | or is it just not configured correctly on their side |
| 13:37.04 | iCEBrkr | e3g: ok, that's just jacked up |
| 13:37.08 | EvilRick | I have 2 pri lines here |
| 13:37.09 | ManxPower | EvilRick, I doubt it or it would not work at all. |
| 13:37.10 | [Jedi] | coppice: then I shouldn't use it I guess... are these changes very deep? |
| 13:37.27 | iCEBrkr | e3g: You DID do a './configure' before 'make' right? |
| 13:37.28 | ManxPower | EvilRick, do you have any kind of loopback connector on any of the ports? |
| 13:37.29 | lancey`away | bye guys |
| 13:37.35 | EvilRick | one was working on an old pabx the otehr taht I am tryiong now was unused |
| 13:37.36 | iCEBrkr | :) |
| 13:37.40 | e3g | no |
| 13:37.47 | EvilRick | no nothing wierd on my side |
| 13:37.47 | iCEBrkr | e3g: Well, there ya go :) |
| 13:38.07 | ManxPower | EvilRick, call up your telco. Tell them you are seeing a loopback on the line. |
| 13:38.08 | e3g | [root@localhost sox-12.17.8]# make ./configure |
| 13:38.08 | e3g | make: Nothing to be done for `configure'. |
| 13:38.13 | iCEBrkr | e3g: :( |
| 13:38.23 | EvilRick | ta thanks |
| 13:38.35 | e3g | eeekkhh |
| 13:38.37 | *** join/#asterisk count (n=adam@corp.alanne.com) |
| 13:39.34 | synthetiq | anyone here use SER? technically it doesnt do any routing lol |
| 13:39.55 | iCEBrkr | [icebrkr@chrome sox-12.17.8]$ ./configure |
| 13:39.56 | iCEBrkr | checking build system type... i686-pc-linux-gnu |
| 13:39.56 | iCEBrkr | checking host system type... i686-pc-linux-gnu |
| 13:40.04 | e3g | oops |
| 13:40.05 | iCEBrkr | e3g: I dunno what ya did over there, but it's working here :P |
| 13:40.08 | e3g | only ./configure |
| 13:40.14 | iCEBrkr | yeah |
| 13:40.22 | e3g | :( |
| 13:40.30 | iCEBrkr | You'll get it. |
| 13:40.32 | e3g | I did make ./configure |
| 13:40.35 | e3g | yes...working now |
| 13:40.48 | iCEBrkr | Once that's done.. Type 'make && make install' |
| 13:41.18 | iCEBrkr | I still think it's going to install into /usr/local/bin tho. |
| 13:41.50 | e3g | done... |
| 13:42.01 | e3g | yep soxmix is working now... |
| 13:42.05 | iCEBrkr | Cool |
| 13:42.07 | lehel | puzzled: call forwarding |
| 13:42.15 | e3g | thanks iCEBrkr |
| 13:42.18 | iCEBrkr | np |
| 13:42.42 | iCEBrkr | No! you CANNOT have my Bud Lite |
| 13:42.56 | e3g | ;-) |
| 13:43.03 | iCEBrkr | e3g: They all say that |
| 13:43.17 | e3g | :D .gonna check Monitor() |
| 13:43.22 | shido6 | Adium is fixed, yay! |
| 13:43.29 | puzzled | lehel: no idea |
| 13:44.03 | MnDBnDr | can anyone help with oh323 extensions? |
| 13:44.07 | shido6 | ? |
| 13:44.17 | shido6 | whats up MnDBnDr ? |
| 13:44.35 | MnDBnDr | I have compile oh323 just fine |
| 13:44.47 | ManxPower | MnDBnDr, Any reason you are not using the NuFone H323 driver or the Objective Systems H323 driver? |
| 13:44.56 | MnDBnDr | not really |
| 13:45.07 | lehel | puzzled: how would you cancel one zap extensions CF set with the phone, not being among of that |
| 13:45.09 | MnDBnDr | is there an advantage to one over the other |
| 13:45.41 | ManxPower | MnDBnDr, They all suck. |
| 13:45.42 | Laibsch | I have tried to set up basic configuration according to http://www.onlamp.com/pub/a/onlamp/2003/07/03/asterisk.html?page=1. X-Lite is the SIP client. I have the strange situation that "sip show peers" shows one of my two users connected, but X-Lite says "Login timed out. Contact network admin". When I try to call a number I get "Call not approved". |
| 13:46.15 | Laibsch | The second client is configured the same and appears to behave the same but it will never show with "sip show peers". |
| 13:46.17 | MnDBnDr | yea, but I have 3 brand new netvision 802.11b phones i would like to use |
| 13:47.08 | ManxPower | MnDBnDr, Interesting. I didn't know there were any 802.11 phones that used H323 |
| 13:47.25 | Laibsch | I have started asterisk with PARAMS="-vvvvvvvd" but I do not see anything meaningful on the console. |
| 13:47.30 | *** join/#asterisk nomazda (i=nyyankee@user-0c6tnqf.cable.mindspring.com) |
| 13:47.33 | MnDBnDr | they are made very well. Although they are discontinued |
| 13:47.35 | *** join/#asterisk kiwnix (n=egarcia@104.red-82-158-159.user.auna.net) |
| 13:47.45 | *** part/#asterisk eziman (i=superop@64.116.231.226) |
| 13:47.59 | MnDBnDr | I got a ericsson webswitch g4 100 gateway with them |
| 13:48.15 | MnDBnDr | I already have * setup for 3 SIP extensions working very well |
| 13:48.20 | ManxPower | MnDBnDr, Are you SURE it runs REAL H323? |
| 13:48.25 | MnDBnDr | Yes |
| 13:48.58 | ManxPower | I know the Nortel "h323" phones don't run real H323 |
| 13:49.14 | *** join/#asterisk Ariel_ (n=Ariel@adsl-068-157-125-248.sip.mia.bellsouth.net) |
| 13:49.18 | MnDBnDr | I can assure you they are pure h323 phones |
| 13:50.01 | MnDBnDr | http://www.adchouston.com/bcm_documents/UK%20English/nvfeatuk.pdf |
| 13:50.06 | ManxPower | MnDBnDr, OK. Just remember that H323 with Asterisk is not easy to set up and not all that many people run it. |
| 13:50.22 | MnDBnDr | I know |
| 13:50.42 | MnDBnDr | I would hate to waste 3 $500 retail phones though |
| 13:51.05 | MnDBnDr | Maybe I will sell the phones with the h323 ericsson gateway |
| 13:51.14 | HiltonT | good idea |
| 13:51.20 | MnDBnDr | they were sold as a package and work very well together |
| 13:51.44 | HiltonT | then get SIP phones for the SIP server :) |
| 13:51.49 | MnDBnDr | yea |
| 13:52.07 | MnDBnDr | I have a Zyxel p2000w v2 phone wthat works great on * |
| 13:52.26 | HiltonT | I'm looking at those |
| 13:52.35 | MnDBnDr | when I take it home, it registers with my * box at work as if I was there |
| 13:52.54 | MnDBnDr | I can tell you the Symbol phones have a better build quality |
| 13:53.02 | tatuman | anyone has Tony redesign of MeetMe app? |
| 13:53.07 | MnDBnDr | they have a cradle with a spare battery. |
| 13:53.18 | MnDBnDr | the Zyxel has a wall wart |
| 13:53.49 | coppice | how heavy are the symbol phones? they look like the usual symbol bricks in pictures |
| 13:53.55 | HiltonT | Symbol - looking for a .au supplier (pref. wholesaler) |
| 13:54.15 | MnDBnDr | not too bad |
| 13:54.27 | MnDBnDr | I would say about twice as heavy as the zyxel |
| 13:54.44 | MnDBnDr | it is about the size of a standard cordless phone handset |
| 13:54.50 | ManxPower | I didn't like wireless before I tried it and now that I've tried it....I don't like wireless. |
| 13:55.05 | MnDBnDr | I could not live without it |
| 13:55.18 | HiltonT | I'm a phonewalker |
| 13:55.23 | MnDBnDr | I have access points all over the building |
| 13:55.31 | denon | ManxPower: wireless headset on a 7960 is a good combo |
| 13:55.32 | ManxPower | I had to replace all my cordless phones when I put in wireless. |
| 13:55.36 | MnDBnDr | it is nice to carry your extension with you everywhere |
| 13:55.49 | ManxPower | My wireless ISP (the only way to get broadband here) goes down more often than Linda Lovelance |
| 13:55.55 | shido6 | wireless headset with a "lifter" |
| 13:55.57 | MnDBnDr | I still use 900Mhz cordless phones |
| 13:56.08 | MnDBnDr | hehe |
| 13:56.15 | denon | I got sick of 900mhz phones .. |
| 13:56.19 | denon | moved to 5.8g |
| 13:56.23 | denon | even though their range is kinda poor |
| 13:56.38 | MnDBnDr | 900Mhz=longer range most of the time |
| 13:56.45 | ManxPower | denon, I moved to 5ghz phones too, not that I can use them much. My one PSTN line is always tied up with a dialup connection |
| 13:56.47 | MnDBnDr | unless you play with amplifiers |
| 13:56.52 | [Jedi] | 900mhz isn't GSM's band? |
| 13:56.53 | denon | MnDBnDr: yeah, but less security, and these days, lower quality gear |
| 13:57.03 | MnDBnDr | yea |
| 13:57.07 | BrianR___ | Does asterisk 1.0.x preserve timestamps when bridging a call between SIP and IAX? |
| 13:57.11 | MnDBnDr | don't care much about security |
| 13:57.17 | [Jedi] | hehe |
| 13:57.27 | MnDBnDr | if someone wants to listen bad enough, they will find a way |
| 13:57.33 | denon | MnDBnDr: yeah, but quality's an issue .. seems like all modern 900mhz is just cheap crap |
| 13:57.42 | MnDBnDr | not too much of a problem in North east Ohio |
| 13:57.48 | denon | MnDBnDr: true, but with 900mhz, even people who dont WANT to listen may very well hear you :) |
| 13:57.49 | RoyK | 900mhz? |
| 13:57.53 | MnDBnDr | hehe |
| 13:57.54 | ManxPower | denon, I've heard about 900Mhz DSS phones, which are supposed to be very good. |
| 13:57.55 | MnDBnDr | true |
| 13:57.55 | denon | ala analog and tdma cell phone days :) |
| 13:58.02 | *** join/#asterisk kFuQ (n=somedude@c-67-185-114-199.hsd1.wa.comcast.net) |
| 13:58.05 | *** join/#asterisk enemy^x (n=null@morpheus.dataguard.no) |
| 13:58.24 | denon | ManxPower: shrugs, I'll ride 5g until 802.11g phones are decent |
| 13:58.27 | denon | or 802.11n |
| 13:58.34 | HiltonT | the 5.8GHz one I liked is only a Vonage unit - base + 5.8 GHz handsets |
| 13:58.36 | HiltonT | :( |
| 13:58.39 | RoyK | wtf is 802.11n? |
| 13:58.50 | ManxPower | Isn't that "WiMax? |
| 13:58.51 | MnDBnDr | The zyxel v2 is not bad at all |
| 13:58.52 | enemy^x | is it possible to get presence (eyebeam sip phone) working with asterisk? I`ve tried adding Hint into my extensions etc. Without seeing any difference. |
| 13:58.55 | denon | RoyK: the unratified predecessor to 802.11g |
| 13:59.15 | HiltonT | predecessor to 802.11g was 802.11b |
| 13:59.38 | coppice | 802.11n is the proposed successor to 80.11g |
| 13:59.41 | MnDBnDr | yea, a pre-n phone would be nice |
| 13:59.49 | Ariel_ | strange talk about wireless phones I have been using a Sipura 2000 with my Panasonic KX-TG2267 2 line phone and works great. 2.4g |
| 13:59.55 | MnDBnDr | HiltonT |
| 14:00.01 | RoyK | mohaha... |
| 14:00.04 | MnDBnDr | what are you looking for from symbol |
| 14:00.07 | RoyK | 540Mbps wifi |
| 14:00.08 | RoyK | :) |
| 14:00.14 | ManxPower | Ariel_, I use a similar setup |
| 14:00.16 | RoyK | or wlan, that is |
| 14:00.18 | denon | er, post-acessor :) |
| 14:00.22 | MnDBnDr | why stop there. 1Gbps |
| 14:00.36 | HiltonT | I'm looking for SIP Wireless phones for here at the office and at client sites |
| 14:00.37 | coppice | I love the boxes which proudly say they are pre-n products. They might as well say "buy this and see it obsolete by Christmas" :-) |
| 14:00.39 | RoyK | MnDBnDr: and 1km range |
| 14:00.42 | Ariel_ | ManxPower, works great I can take the phone anywhere in the office and pickup both of my lines with it. |
| 14:01.02 | ManxPower | Ariel_, *nod* |
| 14:01.07 | MnDBnDr | the only manu I have seen here that is pushing pre-n is belkin |
| 14:01.15 | ManxPower | Ariel_, and I'll bet it works alot better than a wifi phone. |
| 14:01.20 | ManxPower | and much cheaper too! |
| 14:01.25 | RoyK | MnDBnDr: and built-in barbecue |
| 14:01.26 | MnDBnDr | yea |
| 14:01.26 | Ariel_ | ManxPower, yes it dose |
| 14:01.26 | HiltonT | tho, X-Lite on my PDA works fine with my BT earpiece :) |
| 14:01.35 | MnDBnDr | yea RoyK |
| 14:01.35 | Ariel_ | does |
| 14:01.44 | mutilator | xlite on my pda sucked |
| 14:02.06 | HiltonT | works fine here for the tests I did |
| 14:02.07 | Ariel_ | has anyone heard of a person getting a GS 488 and xlite to talk to each other without an asterisk or other service? |
| 14:02.10 | coppice | The Belkin ones have a rather large "buy me, I'm obsolescent" sticker |
| 14:02.14 | ManxPower | I need to find someone that is line-of-site from me that can get DSL |
| 14:02.38 | MnDBnDr | I am not a fan of Belkin wireless products |
| 14:02.58 | ManxPower | AND I need to find a way to spend less than $400 each for 2 portable towers. |
| 14:03.11 | wasim | baloons! |
| 14:03.21 | ManxPower | wasim, Helium is expensive |
| 14:03.22 | coppice | ManxPower: that might depend on their height |
| 14:03.32 | wasim | ManxPower: use the hotair from #asterisk ... |
| 14:03.47 | ManxPower | wasim, you mean asterisk-biz, right? |
| 14:03.56 | newl | telescoping antennas usually aren't too pricy. |
| 14:04.00 | coppice | ManxPower: you could raise hot air baloons by connecting them to the mailing list |
| 14:04.12 | MnDBnDr | ok. My Symbol phone will ring when called from a SIP phone. |
| 14:04.23 | MnDBnDr | I here no audio on my SIP phone |
| 14:04.30 | ManxPower | I need to get 50 - 100 ft into the air. |
| 14:04.36 | MnDBnDr | Have audio on the Symbol phone |
| 14:04.45 | ManxPower | Until now I didn't HATE TREES |
| 14:04.49 | *** join/#asterisk darwin35 (n=darwin35@208.139.193.178) |
| 14:04.49 | MnDBnDr | also, can't call out from the SIP phone |
| 14:04.55 | newl | 30m is easily doable. :) |
| 14:04.57 | HiltonT | trtees are baaaad |
| 14:05.00 | MnDBnDr | I mean h323 phone |
| 14:05.04 | MnDBnDr | SIP is fine |
| 14:05.12 | enemy^x | anyone here have presence working with eyebeam/asterisk? |
| 14:05.13 | ManxPower | kill the trees! kill the trees! |
| 14:05.29 | ManxPower | enemy^x, your extensive searhc of the mailing list archives was not helpful? |
| 14:05.34 | newl | (sung to Megadeths Kill the King) |
| 14:05.36 | darwin35 | Manx what did you want yesterday |
| 14:05.49 | coppice | ManxPower: use the trees. use the trees |
| 14:05.50 | darwin35 | you never responded |
| 14:05.57 | ManxPower | darwin35, To see if I can get some custom jitterbuffer settings for my account. |
| 14:06.13 | darwin35 | I will check with david when he gets in |
| 14:06.13 | ManxPower | darwin35, My ISP goes down more than Linda Lovelance |
| 14:06.21 | darwin35 | heheheh |
| 14:06.36 | darwin35 | and has nails like Freddie Couger |
| 14:06.57 | darwin35 | just slap the bitch and tell her no teeth |
| 14:07.03 | ManxPower | darwin35, and jitter is pretty amazing. I have never seen an ISP with this weird of jitter and latency |
| 14:07.16 | ManxPower | But they are the ONLY broadband company I can get here. |
| 14:07.18 | *** join/#asterisk jimmy_deanPB (n=jhodapp@adsl-70-228-242-126.dsl.ipltin.ameritech.net) |
| 14:07.26 | darwin35 | understand |
| 14:08.09 | *** join/#asterisk bmg505 (n=leon@rndf-146-52-78.telkomadsl.co.za) |
| 14:09.23 | *** join/#asterisk IPmonger (n=ipmonger@pcp0010577106pcs.coatsv01.pa.comcast.net) |
| 14:10.13 | darwin35 | man its only 8 been here an hour and cleared out the ticket q |
| 14:10.20 | darwin35 | 25 tickets |
| 14:10.55 | ManxPower | darwin35, "Customer config issue. Closing ticket!" *tease* |
| 14:11.07 | Katty | mew, exchange :< |
| 14:11.16 | darwin35 | heheh |
| 14:11.43 | ManxPower | We got a ticket yesterday "We are moving the office tomorrow. Reconfig everything so it works." |
| 14:11.50 | darwin35 | man if you knew how many id 10 t or pebcad errors a day we get its to much |
| 14:11.53 | shido6 | nice |
| 14:12.11 | darwin35 | we just moved offices also |
| 14:12.20 | darwin35 | and I have the only desk in it |
| 14:12.24 | ManxPower | shido6, the funny thing is they are moving from one part of the building to another part all on the same LAN |
| 14:12.32 | darwin35 | the boss moves in today |
| 14:13.25 | darwin35 | Manx move here to Denver and get your life back |
| 14:13.52 | ManxPower | Denver: Where everyone is a member of the Mile High Club! |
| 14:13.54 | mutilator | anyone know of any good shotcasts that play classic rock? i have one but it seems like it streams the same thing at the same time every day |
| 14:13.55 | wasim | no, move to PK, we're worst affected than you now, nyah nyah |
| 14:14.07 | ManxPower | darwin35, naw, I want to live in a campground in the mountians |
| 14:14.18 | darwin35 | www.shoutcast.com |
| 14:14.25 | darwin35 | look threw the channel list |
| 14:14.42 | ManxPower | I might be going to visit the place in a week or so |
| 14:14.45 | darwin35 | we have mountains |
| 14:14.51 | darwin35 | and skiing |
| 14:14.54 | mutilator | yea i have been darwin |
| 14:14.54 | wasim | we have mountains |
| 14:14.56 | darwin35 | and snow camping |
| 14:15.01 | wasim | and earthquakes |
| 14:15.04 | ManxPower | darwin35, But you don't have "my" kind of campgrounds |
| 14:15.06 | mutilator | i can't find anything that plays worthwhile stuff |
| 14:15.19 | darwin35 | pass that pvt |
| 14:15.22 | ManxPower | Contrary to popular belief, Hell is not hot, Hell is cold. |
| 14:15.27 | darwin35 | the butt nekid kind |
| 14:15.29 | mutilator | they're playing like.. 38 special |
| 14:15.34 | mutilator | and 50';s music |
| 14:15.51 | ManxPower | mutilator, I listed to KCRW and WDST (both on shoutcast) |
| 14:15.55 | *** join/#asterisk dalabera (n=Dalabera@pmr.pmrtechnologies.com) |
| 14:16.18 | darwin35 | there is no heaven there is no hell you make your life what you make it and the only one you answe to when it is over is yourself |
| 14:16.20 | ManxPower | Besides, Denver is too far from my current customer base |
| 14:16.33 | darwin35 | awww |
| 14:16.34 | darwin35 | ok |
| 14:16.39 | mutilator | well i got radio free colorado which is what i'm listenin to |
| 14:16.48 | mutilator | it's down half the time |
| 14:16.53 | mutilator | but when it works it's a good station |
| 14:16.59 | darwin35 | I like Digitalgunfire |
| 14:17.05 | ManxPower | darwin35, And I doubt Teliax will pay me what I get now 8-) |
| 14:17.06 | darwin35 | they have loads of channels |
| 14:17.22 | darwin35 | prob right |
| 14:17.32 | darwin35 | give us time to grow then we might |
| 14:17.41 | Corydon76-home | Rizzo on list: "I broke FreeBSD on the 64-bit conversion, so I want to break Asterisk to match FreeBSD' brokenness." Me: "Why not just fix FreeBSD, instead?" |
| 14:18.19 | Corydon76-home | Wow, what a concept... |
| 14:18.25 | darwin35 | no fun in what you fix if you dont break it first |
| 14:19.08 | darwin35 | wow our new servers and equipment should be here today |
| 14:20.11 | darwin35 | where then women are stong and the men do the housework |
| 14:20.46 | mutilator | i wouldn't mind that |
| 14:20.55 | mutilator | i could sit and do stuff on the net all day |
| 14:21.04 | mutilator | like i used to |
| 14:21.35 | mutilator | work on project turmoil again |
| 14:21.40 | Dr_Ray | unless it involves raising young children, I'll stay home |
| 14:22.34 | darwin35 | who knows what a Pebcad error is |
| 14:22.39 | *** join/#asterisk brookshire (n=matt@gateway.digium.com) |
| 14:22.54 | Dr_Ray | chair and dispaly? |
| 14:23.10 | Dr_Ray | pebkac |
| 14:23.25 | darwin35 | Problem exist between chair and desk |
| 14:23.42 | Dr_Ray | we call those ID-ten-T errors |
| 14:23.59 | darwin35 | ID 10 T |
| 14:24.02 | Dr_Ray | yes |
| 14:24.16 | ManxPower | darwin35, I'm in Limping Mule Texas...er....Atlanta Texas |
| 14:24.48 | darwin35 | hehe wow |
| 14:24.50 | ManxPower | BTW, does anyone know of a site that I can look up the street address of a specific CO on? |
| 14:25.07 | darwin35 | mapquest |
| 14:25.17 | mutilator | ^ :P |
| 14:25.19 | ManxPower | darwin35, Um, I don't know the address of the CO |
| 14:25.32 | mutilator | what do you know? |
| 14:25.39 | ManxPower | I know the NPA-NXX |
| 14:25.44 | ManxPower | 903-796 |
| 14:26.11 | darwin35 | brb fone |
| 14:26.25 | mutilator | hmm |
| 14:27.40 | RoyK | hm |
| 14:27.58 | RoyK | anyone that knows a sip client that can run on a sony ericsson z800i? |
| 14:28.50 | leszq | Who knows what will the call quality be if second side is in USA, I am in Poland ....between us is about 180 ms ping delay ... upload/download is about 250 kbit/s in each direction ?? I want use SIP |
| 14:29.43 | leszq | Is a sense to try calling like this? |
| 14:31.47 | dalabera | hello guys, does anyone had tried to call the conference number for astricon, the one they publish on their web? |
| 14:32.54 | mutilator | DAYSTARR, LLC DBA DAYSTARR COMMUNICATIONS - MI |
| 14:32.56 | mutilator | hm |
| 14:33.03 | mutilator | doesn't that kill vampires? |
| 14:33.08 | wasim | leszq: sure |
| 14:33.25 | leszq | wasim: did you try it? |
| 14:34.18 | wasim | leszq: try a jittery, lossy link with 500ms latency and barely 30kbps to have fun |
| 14:34.33 | *** join/#asterisk oej (n=Olle@dsl001-136-136.lax1.dsl.speakeasy.net) |
| 14:35.17 | leszq | wasim: thx |
| 14:36.35 | Katty | hmm |
| 14:38.27 | dalabera | hello guys, does anyone had tried to call the conference number for astricon, the one they publish on their web? |
| 14:39.05 | brookshire | i doubt anyone is even up yet |
| 14:39.06 | brookshire | hehe |
| 14:39.16 | brookshire | it's like 7 am |
| 14:39.33 | leszq | or 6.39 pm :) |
| 14:39.42 | dalabera | ok |
| 14:40.16 | ManxPower | My CO is a Ericsson AXE-10 |
| 14:40.23 | dalabera | but they should be 3 hours from EST |
| 14:40.35 | ManxPower | I'd never even heard of that model |
| 14:43.45 | johnm | RoyK: the GPRS network you need to make a sip call would immediately make it more expensive, no? :) |
| 14:43.59 | johnm | RoyK: unless you can ghet it to connect out via IrDA/Bluetooth. |
| 14:44.34 | Katty | hmm |
| 14:44.35 | johnm | I have to say... a bluetooth SIP phone on ytour mobile would be cool. Especially if it auto-associated. |
| 14:44.57 | Katty | johnm: they're quite expensive |
| 14:45.04 | Katty | johnm: and their range isn't all that great... |
| 14:45.09 | Katty | johnm: but yes, it's quite neat. |
| 14:45.51 | johnm | Katty: thats assuming a bluetooth sip phone. a sip-component to a mobile smart-phone is somethign differnet. |
| 14:46.08 | johnm | and bluetooth can easily do 15m, so it will deffo do most of my house :) |
| 14:46.12 | Katty | johnm: bluetooth is what i said. |
| 14:46.15 | johnm | just stick is central. |
| 14:46.20 | RoyK | johnm: UMTS |
| 14:46.26 | johnm | Katty: no you never :) |
| 14:46.40 | Katty | johnm: i see. |
| 14:46.46 | Katty | johnm: i was referencing bluetooth. |
| 14:46.48 | johnm | RoyK: Probably still more expensive.. which sucks. |
| 14:47.07 | Katty | you can turn practically everything into bluetooth now. |
| 14:47.11 | johnm | Katty: I assumed so.. but the expensive part couldn't have been. |
| 14:48.07 | MnDBnDr | can someone tell me how to remove oh323 completely |
| 14:48.21 | Katty | MnDBnDr: a hammer. |
| 14:48.32 | MnDBnDr | that would work |
| 14:48.44 | MnDBnDr | but I need a less messy solution |
| 14:48.49 | MnDBnDr | :) |
| 14:48.51 | Katty | MnDBnDr: a dumpster. |
| 14:49.20 | *** join/#asterisk fugitivo (n=ajf@201.255.102.19) |
| 14:49.35 | MnDBnDr | hehe |
| 14:49.42 | MnDBnDr | I still need my box for SIP |
| 14:49.43 | bjohnson | bluetooth can do up to 100m |
| 14:49.46 | Katty | MnDBnDr: i dunno ;) |
| 14:49.56 | Katty | bjohnson: that's all theory. i've used bluetooth. |
| 14:50.01 | bjohnson | I'm told high end equipment can reach over 1 km |
| 14:50.03 | Katty | bjohnson: and it isn't 100m |
| 14:50.12 | bjohnson | Katty: read about the classes |
| 14:50.23 | *** join/#asterisk veto (i=mdkuser@cpe-66-69-38-192.satx.res.rr.com) |
| 14:50.26 | Katty | bjohnson: i shall not. |
| 14:50.27 | bjohnson | ie clas 1 vs class 2 |
| 14:50.30 | Katty | bjohnson: you read about the classes. |
| 14:50.33 | bjohnson | then don't presume to correct me |
| 14:50.41 | Katty | bjohnson: i'll presume whatever i wish. |
| 14:50.50 | Katty | bjohnson: just like you do |
| 14:53.05 | bjohnson | RoyK: a sip client on a cell phone doesn't seem to have any advantages. Why do you want one? |
| 14:54.55 | wasim | bjohnson: it does when GPRS rates are cheaper than call rates :) |
| 14:55.18 | Ahrimanes | sip client + wifi in cell phone = good |
| 14:56.04 | bjohnson | johnm: a BT enabled cell phone connected to SIP via asterisk and a BT dongle is theoretically possible. I know people have worked on it but I don't know if success was achieved |
| 14:57.43 | Dr_Ray | a bluetooth headset with dongle seems to be a nicer solution |
| 14:57.56 | *** join/#asterisk Uther_P (n=uther_p@66.180.120.82) |
| 14:58.16 | bjohnson | Dr_Ray: depends on the question |
| 14:59.42 | *** join/#asterisk razu (n=razu@tln-kontor.norby.ee) |
| 14:59.44 | bjohnson | Dr_Ray: some home users where interested in using BT enabled cell phones as pstn connections for their home asterisk pbx .. was a good solution for them (depending on their cell phone plan and calling patterns) |
| 15:01.04 | bjohnson | in a work environment it might be less useful, but the possibility of using bt enabled cell phones and a BT mesh sounds enticing |
| 15:01.33 | bjohnson | same idea as WIFI phones but potentially cheaper system |
| 15:02.46 | *** join/#asterisk jets (n=jets@dsl001-136-136.lax1.dsl.speakeasy.net) |
| 15:08.08 | RoyK | bjohnson: for fun, mostly :) |
| 15:08.35 | *** join/#asterisk Lathos42 (n=Lathos42@adsl-68-255-76-52.dsl.lgtpmi.ameritech.net) |
| 15:08.52 | *** join/#asterisk ManxPower (n=eric@slip-12-65-36-138.mis.prserv.net) |
| 15:09.34 | johnm | bjohnson: all you need is networking via BT, rather than a BT module in * |
| 15:09.36 | nexis | ive been poking at google for a while now, and cant seem to get it, can somone point me in the direction of a agi script that will accept a incoming call, hang it up, then call the number back? |
| 15:11.54 | *** join/#asterisk Veto_laptop (i=mdkuser@cpe-66-69-38-192.satx.res.rr.com) |
| 15:12.07 | *** join/#asterisk moy (n=kvirc@dsl-201-128-242-37.prod-infinitum.com.mx) |
| 15:14.19 | Lee__ | nexis: perldoc Asterisk::AGI |
| 15:14.29 | Katty | who manages a network? |
| 15:14.43 | Katty | where they are responsible for backing up servers. |
| 15:14.44 | ManxPower | Katty, I manage many networks |
| 15:14.58 | Katty | ManxPower: what is your favorite Total Backup Solution(tm)? |
| 15:15.07 | Lee__ | what's up with your network Katty? |
| 15:15.13 | Katty | ManxPower: raid, tape drives, external hard drives, etc |
| 15:15.21 | Katty | Lee__: my network is perfectly fine. |
| 15:15.29 | Lee__ | cool |
| 15:15.30 | ManxPower | Katty, Hmm? We are. The users/office managers can't be trusted to do that. |
| 15:15.39 | ManxPower | Katty, They all suck. We use rsync. |
| 15:15.46 | Katty | ManxPower: k |
| 15:15.52 | Katty | anyone /else/ manage a network? |
| 15:16.16 | ManxPower | Katty, rsync to a central server. no off site backups because the bean counters won't spend the money on that. |
| 15:16.24 | Lee__ | rsync + ssh + online backup to disk + offsite tapes |
| 15:16.54 | Katty | ManxPower: so your stuff just sits on a central server with no backup? |
| 15:16.56 | Lee__ | add Samba in that if you have Windows machines in the mix. |
| 15:17.00 | Beirdo | Katty: we use Veritas NetBackup with a huge tape library, but that's likely out of budget scope :) |
| 15:17.06 | Beirdo | and good morning, BTW |
| 15:17.15 | Katty | Beirdo: how many tapes? |
| 15:17.35 | Beirdo | I'd have to ask the guy who does most of the maintenance... |
| 15:17.40 | [Jedi] | we also use NetBackup with a pair of tape libraries |
| 15:17.40 | ManxPower | Katty, no. Each of the servers at the remote sites, and each of the servers at HQ backup to a central server that only holds the backup data. |
| 15:17.45 | Beirdo | I think it's like 8 drives of LTO |
| 15:17.50 | Katty | asterisk -> cron + rsync + samba to windows server -> tape backups, raid, and external hd backup (that is our network) |
| 15:17.55 | [Jedi] | ours are only two LTO drivers each |
| 15:17.56 | Beirdo | and a few hundred tapes in the library |
| 15:17.57 | [Jedi] | =D |
| 15:18.00 | Katty | ManxPower: k |
| 15:18.11 | Katty | Beirdo: so you only have tape backups? have you ever had to restore anything? |
| 15:18.39 | Beirdo | it takes ages, but yes |
| 15:18.46 | Katty | Am i the only one who has crappy luck with tape drives? |
| 15:18.57 | ManxPower | We've never been able to restore anyrhing from tape, using any software for any system. We finally gave up. |
| 15:19.04 | shido6 | LOL |
| 15:19.07 | bon | :))) |
| 15:19.08 | Katty | ManxPower: sounds like us. heh |
| 15:19.14 | shido6 | tar gz to tape screw s/w |
| 15:19.19 | [Jedi] | we've done it many times |
| 15:19.21 | ManxPower | Katty, *nod* |
| 15:19.25 | Katty | i was thinking about maybe this new rev drive thing |
| 15:19.30 | file[laptop] | KATTY |
| 15:19.32 | Beirdo | we had to restore our entire netapp |
| 15:19.32 | Katty | apparently it's like a jazz drive |
| 15:19.37 | Katty | but with 90 gig storage capacity |
| 15:19.39 | Beirdo | believe me, that was NOT fun |
| 15:19.40 | Katty | file[laptop]: FILE |
| 15:19.48 | Katty | file[laptop]: how does your Backup Solution (tm) work? |
| 15:19.54 | Lee__ | hee hee. jazz drive. that one worked out well. |
| 15:19.57 | file[laptop] | I have none. |
| 15:20.05 | Katty | file[laptop]: zomg, k |
| 15:20.09 | shido6 | the only iomega anything I had fun with was "record it" back in the day |
| 15:20.16 | Katty | what about ait? |
| 15:20.17 | shido6 | recoding lectures in class |
| 15:20.31 | Katty | i've a friend who works out at nasa. |
| 15:20.39 | Katty | they're considering this ait 1/2/3 thingy |
| 15:20.45 | Katty | anyone worked with that? |
| 15:20.58 | Katty | oooh, shiny! |
| 15:21.10 | n3u7 | morning asterisk |
| 15:21.24 | Beirdo | buttholes, and their $700/month in overage |
| 15:21.34 | Beirdo | up yours, put me on a new plan NOW |
| 15:21.37 | n3u7 | I am receiving an error with modprobe zaptel |
| 15:21.53 | Beirdo | file[laptop]: Bell has better coverage in some areas :) |
| 15:21.54 | Katty | Also, i'm trying to figure out how to backup an entire windows server without having to take anything offline |
| 15:21.59 | Beirdo | unfortunately |
| 15:22.01 | n3u7 | I have read the last few weeks of the mailling list |
| 15:22.02 | Katty | yet, i can't seem to figure out a way, other than raid1 |
| 15:22.06 | file[laptop] | Beirdo: realllly? Telus roams on Bell sometimes |
| 15:22.12 | n3u7 | and searched extensivly |
| 15:22.16 | Katty | cause their stupid backup Exchange Aware software is shitty |
| 15:22.19 | file[laptop] | not that I would know, considering it's all Aliant here |
| 15:22.24 | *** join/#asterisk _Sam-- (n=sam@ns2.kneedraggers.com) |
| 15:22.27 | Beirdo | the only Telus I saw here was former Clearnet |
| 15:22.29 | Beirdo | blech |
| 15:22.30 | ManxPower | n3u7, I'm sorry to hear that. Have you tried Prozac. |
| 15:22.36 | bon | :)) |
| 15:22.40 | file[laptop] | Beirdo: I like my plan with Telus... |
| 15:22.40 | ManxPower | Or maybe pasting the one line error message? |
| 15:22.41 | Katty | is there anything built into windows which will do some sort of ghost image? |
| 15:22.43 | Katty | like ISO |
| 15:22.44 | Beirdo | oh that's right, you are in NB aren't ya? |
| 15:22.54 | file[laptop] | Beirdo: unlimited incoming, unlimited early evenings/weekends, 100 daytime, $30/mth |
| 15:22.58 | n3u7 | the err is to the extent :Unknown symbol in module |
| 15:23.00 | Beirdo | heh |
| 15:23.07 | Beirdo | that would suck for me |
| 15:23.09 | Katty | file[laptop]: or nextel. |
| 15:23.12 | Katty | file[laptop]: where all incoming is free |
| 15:23.15 | n3u7 | dmesg reveals the symbol to be sub_preempt_count |
| 15:23.17 | Katty | file[laptop]: and there are no roaming charges |
| 15:23.18 | ManxPower | file, 100 mins/month? I use that up in like 3 days. |
| 15:23.19 | *** join/#asterisk mhnoyes (n=mhnoyes@user-2ivfl2a.dialup.mindspring.com) |
| 15:23.21 | file[laptop] | Katty: Canada is different |
| 15:23.21 | Beirdo | I was > 500min three months running |
| 15:23.24 | Katty | file[laptop]: ooh, right. |
| 15:23.27 | Katty | file[laptop]: nevermind |
| 15:23.29 | Beirdo | hence the huge-ass bills |
| 15:23.30 | Katty | file[laptop]: silly canadian |
| 15:23.43 | Beirdo | now I have 700min + free evening/weekend: $75 |
| 15:23.54 | file[laptop] | god people, think! callback system! duh. |
| 15:23.59 | coppice | 700 mins of what? |
| 15:24.05 | Beirdo | cell phone |
| 15:24.06 | Katty | my phone bill is 10 bucks |
| 15:24.14 | n3u7 | ManxPower: :/ |
| 15:24.15 | Katty | all my incoming is free |
| 15:24.18 | Beirdo | cool :)9 |
| 15:24.20 | Katty | i have no roaming or overages |
| 15:24.26 | Katty | all of my direct connect is unlimited |
| 15:24.31 | Katty | (that's the beep beep nextel feature) |
| 15:24.35 | Beirdo | Oh, and roaming in Puerto Rico is a frigging killer too :) |
| 15:24.35 | ManxPower | Katty, that's because Nextel CANNOT roam |
| 15:24.36 | Katty | plus the motorola phones support dual lines |
| 15:24.39 | coppice | Beirdo: wow, that's expensive |
| 15:24.46 | *** join/#asterisk felipex (n=dsfdsf@85-18-136-75.fastres.net) |
| 15:24.47 | darwin35 | Thanks for Calling Tellia how might I direct you connection |
| 15:24.50 | file[laptop] | uh well, they can roam |
| 15:24.52 | Katty | ManxPower: of course. but they have coverage in all major cities |
| 15:25.01 | file[laptop] | you can roam in Canada... and other places |
| 15:25.07 | Beirdo | coppice: hardly when it's $700 in overages with my former plan |
| 15:25.12 | Katty | ManxPower: i have no need to roam. |
| 15:25.16 | n3u7 | guess I;ll be waiting |
| 15:25.25 | n3u7 | I've read the starfish book too |
| 15:25.34 | ManxPower | I moved back to GSM recently. |
| 15:25.38 | _Sam-- | i had been using zaptel for channels, and switched to a non-zap connection. my stuff is working fine, but when i remove zapata.conf, my asterisk wont start.....and i cant figure out how to make it try to stop loading the zaptel module |
| 15:25.39 | coppice | 1800 mins for $12 is more my kind of pricing for cell phones :-\ |
| 15:25.46 | Katty | oh give me a home, where the file doth not roam |
| 15:25.51 | Katty | do do do de de do de do |
| 15:25.54 | n3u7 | I would like to start writing some agi and dialplans but I can't get past the install |
| 15:26.02 | n3u7 | perhaps th mobo is to old |
| 15:26.11 | ManxPower | Prepay so I don't have to worry about early termination fees for the 2 years I have left on the 1 year contract I signed 4 years ago |
| 15:26.14 | Katty | i had an orange danish for breakfast! |
| 15:26.21 | Katty | it was dreamy. |
| 15:26.25 | file[laptop] | I'm month to month man! |
| 15:26.30 | Katty | like wget is dreamy |
| 15:26.51 | n3u7 | :/ |
| 15:26.53 | Katty | file[laptop]: invent rhug |
| 15:27.04 | ManxPower | n3u7, I still don't see the actual error message. |
| 15:27.07 | file[laptop] | Katty: I'll get right to that |
| 15:27.10 | Katty | file[laptop]: kthx |
| 15:27.43 | n3u7 | ManxPower:it's on a different box |
| 15:28.01 | *** part/#asterisk nexis (n=nexis@12-219-60-252.client.mchsi.com) |
| 15:28.44 | ManxPower | n3u7, It sucks to be you. |
| 15:29.52 | n3u7 | FATAL:Error inserting zaptel(/lib/modules/2.6.10-5-386/misc/zaptel.ko):Unknown symbol in module , or unknown parameter (see dmesg) |
| 15:30.29 | *** join/#asterisk XTR-II (n=xtr@staff-nat.netnation.com) |
| 15:30.30 | n3u7 | dmesg:zaptel: unknown symbol add_preemt_count |
| 15:30.44 | n3u7 | dmesg:zaptel: unknown symbol sub_preemt_count |
| 15:30.51 | n3u7 | dmesg:zaptel: unknown symbol sub_preempt_count |
| 15:30.54 | *** join/#asterisk Seyr (n=Seyr@rrcs-24-227-133-226.sw.biz.rr.com) |
| 15:31.13 | n3u7 | ManxPower:I suppose |
| 15:31.24 | _Sam-- | having the oppposite problem...how do i tell asterisk to stop using zap? |
| 15:31.25 | *** join/#asterisk sangee (n=rkuru@Toronto-HSE-ppp3697289.sympatico.ca) |
| 15:31.52 | Seyr | I am using Realtime and have extension 200 and extension 201 in mySQL table "sipusers". If I try to call 200 from 201, or 201 from 200, it goes to Unavailable. Any ideas? |
| 15:32.25 | Beirdo | oh yeah, and I got just over another year on the 2yr contract, and I'm thinking of moving out of the country. |
| 15:32.40 | Beirdo | I will kick Bell's ass if they try dinging me too hard |
| 15:32.43 | Seyr | both 200 and 201 can dial outside |
| 15:33.36 | *** join/#asterisk stkn (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 15:33.36 | n3u7 | :( |
| 15:34.12 | bon | seyr :) |
| 15:34.13 | bon | no ide |
| 15:35.15 | *** join/#asterisk [virii] (n=virii@68-186-170-246.dhcp.smrt.tn.charter.com) |
| 15:37.50 | *** join/#asterisk oej (n=Olle@dsl001-136-136.lax1.dsl.speakeasy.net) |
| 15:42.48 | Seyr | file, your using Realtime right? |
| 15:43.08 | *** join/#asterisk SarahEmm (n=sarahemm@Toronto-HSE-ppp3682154.sympatico.ca) |
| 15:44.32 | sangee | i don't get ring back tone when i call from cell phone, anyone know how to solve this issue? |
| 15:44.58 | SarahEmm | what are you calling into? |
| 15:45.02 | SarahEmm | a PRI, a voip provider, a POTS line? |
| 15:46.00 | *** join/#asterisk [TK]D-Fender (n=joe@4.67.252.216.dsl1.colba.net) |
| 15:46.18 | sangee | i am calling to asterisk to sipura device |
| 15:46.24 | SarahEmm | err |
| 15:46.28 | SarahEmm | what is the sipura connected to tho? |
| 15:46.33 | SarahEmm | a remote VoIP provider, a POTS line? |
| 15:47.36 | *** join/#asterisk coppice (n=chatzill@48.201.17.210.dyn.pacific.net.hk) |
| 15:47.40 | sangee | sipura registered with asterisk |
| 15:47.54 | Nyvar | are you calling from PSTN to the asterisk box, which then calls the sipura? |
| 15:48.26 | sangee | yes |
| 15:48.56 | Nyvar | ok, so the question is, how is that pstn call getting into your asterisk box? |
| 15:50.11 | sangee | our pstn is cisco, cisco to asterisk as sip |
| 15:50.26 | *** join/#asterisk outtolunc (i=outtolun@adsl-66-218-53-170.dslextreme.com) |
| 15:51.28 | MikeJ[Laptop] | kram! |
| 15:51.31 | [TK]D-Fender | Dammit I'm still getting mangled faxes through my TE405P, PRI, And Rhino Channel Bank. Any new advice for me to help fix this? Per Rhino I've already started varying the gain on their CB, while leaving * at 0.0 |
| 15:51.41 | sivana | SarahEmm: ping |
| 15:51.46 | SarahEmm | cisco to asterisk how? |
| 15:51.49 | SarahEmm | hi sivana |
| 15:51.52 | sivana | hey |
| 15:52.00 | sivana | I need to move you to another switch :) |
| 15:52.06 | Nyvar | could this be an early-media config problem on sangee's cisco? |
| 15:52.08 | SarahEmm | sangee: your PSTN connection goes into a Cisco box, then to *? what kind of PSTN line? |
| 15:52.14 | SarahEmm | sivana: 'k. which? |
| 15:52.19 | sivana | switch-2 |
| 15:52.26 | SarahEmm | and can you change my password at the same time to something a *little* less obvious? ;P |
| 15:52.27 | sivana | you can copy/paste an identical config |
| 15:52.33 | sivana | hehe |
| 15:52.39 | Seyr | Does anyone know why I cannot call a user that has their record stored in SQL? |
| 15:53.19 | Nyvar | seyr, sipfriends? |
| 15:53.37 | Seyr | yeh |
| 15:53.52 | Nyvar | you running cvs head or thereabouts? |
| 15:53.56 | sangee | cisco is getting pstn call then pass to asterisk, then asterisk to sipura |
| 15:54.09 | Seyr | Nyvar: yes, maybe 2 weeks old CVS |
| 15:54.13 | Nyvar | sangee, google 'cisco early media' |
| 15:54.36 | sangee | you think this is the cisco issue? |
| 15:54.45 | *** join/#asterisk cjk (n=cjk@80.92.64.103) |
| 15:54.47 | Nyvar | yes |
| 15:54.51 | sangee | ok |
| 15:55.00 | *** join/#asterisk shido6 (n=curtis@d221-68-210.commercial.cgocable.net) |
| 15:55.01 | sangee | thanks i will search it |
| 15:57.11 | Seyr | Any idea Nyvar? |
| 16:01.20 | darwin35 | kattyy why are you not at astricon |
| 16:01.25 | SarahEmm | hihihihihhi Katty! |
| 16:02.02 | *** join/#asterisk Rubble (n=netclass@dsl001-136-136.lax1.dsl.speakeasy.net) |
| 16:03.15 | Katty | SarahEmm: :>>>> |
| 16:03.19 | fugitivo | anyone know a good ATA with 2fxs, pstn backup and QOS? |
| 16:03.42 | *** join/#asterisk loick (n=loick@ATuileries-151-1-55-19.w83-202.abo.wanadoo.fr) |
| 16:03.55 | Katty | darwin35: it's not my duty to be at astricon, kthx. |
| 16:04.25 | SarahEmm | *huuuugs on katty* |
| 16:04.34 | file | yay |
| 16:04.36 | file | KITRICH! |
| 16:04.39 | SarahEmm | FILE! |
| 16:05.06 | *** join/#asterisk veto (i=mdkuser@cpe-66-69-38-192.satx.res.rr.com) |
| 16:05.26 | [virii] | configure: error: termcap support not found |
| 16:05.26 | [virii] | make: *** [editline/libedit.a] Error 1 |
| 16:05.28 | [virii] | huh? |
| 16:05.51 | SarahEmm | FILE! |
| 16:05.53 | SarahEmm | oops |
| 16:06.06 | file | [virii]: you need to install... termcap support, like ncurses |
| 16:07.35 | *** join/#asterisk RoyK (n=roy@host-81-191-147-248.bluecom.no) |
| 16:08.05 | RoyK | hi |
| 16:08.07 | RoyK | ho |
| 16:08.13 | SarahEmm | hi |
| 16:08.36 | [virii] | ack =P |
| 16:09.28 | *** join/#asterisk malverian[work] (n=pawalls@pawalls.teamgleim.com) |
| 16:09.32 | darwin35 | kram is never around |
| 16:10.04 | *** part/#asterisk darwin35 (n=darwin35@208.139.193.178) |
| 16:10.27 | cjk | anyone an idea how i can simulate a stun connection using netcat? |
| 16:10.55 | *** join/#asterisk zoa (n=zoa@host06.alica.hyatthsiagx.com) |
| 16:11.16 | zoa | eeps |
| 16:11.20 | zoa | heya mark! |
| 16:11.29 | SarahEmm | hihi kram |
| 16:11.33 | kram | heya zoa :) |
| 16:11.35 | kram | hi sarah |
| 16:11.52 | zoa | is the breakfast open yet ? |
| 16:11.55 | *** join/#asterisk e3g (i=ee@u15157627.onlinehome-server.com) |
| 16:12.00 | kram | uhm it's just coffee and stuff |
| 16:12.02 | e3g | hi, |
| 16:12.09 | zoa | k |
| 16:12.28 | e3g | I am trying to use WAV as music on hold . . but dont hear WAV music ..only MP3 works .. |
| 16:12.31 | e3g | ??? |
| 16:12.44 | fugitivo | why you want to use wav? |
| 16:12.47 | RoyK | zoa: mEEP |
| 16:13.00 | RoyK | e3g: using cvs head? |
| 16:13.28 | e3g | because I get bad quality with MP3 ..and sometimes musicon hold stops without any error or warning |
| 16:13.39 | RoyK | e3g: rotfl. bad quality??? |
| 16:13.50 | fugitivo | e3g: you won't get better quality with wav, mp3 is fine |
| 16:14.03 | RoyK | e3g: alaw/ulaw is the best codec there is for asterisk, and that's far worse than mp3 at a good rate |
| 16:14.15 | RoyK | e3g: but using alaw works with cvs head |
| 16:14.17 | *** join/#asterisk r0d3nt (i=nobody@wsip-24-234-241-145.lv.lv.cox.net) |
| 16:14.22 | RoyK | e3g: see native moh |
| 16:14.25 | fugitivo | e3g: try using madplay, not mpg123 |
| 16:14.57 | RoyK | e3g: just transcoding the files to 16bit 8kbps mono and setup native moh |
| 16:15.06 | ManxPower | outtolunc, one that says "Drink Me" |
| 16:15.10 | e3g | native moh? |
| 16:15.21 | RoyK | e3g: but beware of that it's rather ugly. it starts a new playback for each call |
| 16:15.29 | outtolunc | naw, joke, i'ev been asterisk for an * sticker for years <G> |
| 16:15.32 | RoyK | e3g: in cvs head / 1.2beta |
| 16:15.35 | outtolunc | er asking |
| 16:15.37 | e3g | hmmmm.....then NO! |
| 16:15.45 | RoyK | :) |
| 16:15.48 | ManxPower | outtolunc, Digium sells them |
| 16:16.00 | e3g | Asterisk CVS HEAD built by root@localhost.localdomain on a i686 running Linux on 2005-10-11 07:47:38 UTC |
| 16:16.50 | e3g | then why MP3 stops after a bit time? |
| 16:17.04 | e3g | with some breakage |
| 16:17.13 | RoyK | e3g: prolly a bad mp3 file |
| 16:17.29 | RoyK | e3g: we're running quite a lot of queueing without any problem like htat |
| 16:17.30 | RoyK | that |
| 16:17.33 | e3g | no....it works fine in Windows Media Player |
| 16:17.49 | e3g | Royk : would you like to hear my Musiconhold ? |
| 16:17.49 | brookshire | well.. digium doesn't currently sell any asterisk stickers |
| 16:17.54 | brookshire | we've got to make new ones, lol |
| 16:17.56 | RoyK | e3g: not really |
| 16:18.01 | e3g | :) |
| 16:18.51 | e3g | Royk: try it man |
| 16:19.01 | *** join/#asterisk Teeli (n=Tili@202-133-65-33-dialup.sat.net.pk) |
| 16:19.34 | brookshire | hey! |
| 16:20.15 | Seyr | Does anyone know why I cannot call a user that has their record stored in SQL sipusers? |
| 16:24.42 | fugitivo | mysql? |
| 16:25.16 | Seyr | yes |
| 16:26.05 | fugitivo | mysql is evil |
| 16:26.14 | file | not this again |
| 16:27.08 | lancey`away | õìììç |
| 16:27.22 | lancey | SIP seems broken again |
| 16:27.36 | lancey | doesn't recognize incoming hosts if there's a section defined for them |
| 16:27.43 | lancey | and doesn't take any options in there into account |
| 16:28.25 | lancey | anyone experience this? |
| 16:28.29 | *** join/#asterisk SplasPood (i=jwb@ludicrous.paravolve.net) |
| 16:29.46 | *** join/#asterisk kimosabe (n=nat@201.145.2.86) |
| 16:31.32 | kimosabe | how can i congure a selectfone account on my asterisk server ? |
| 16:31.44 | kimosabe | can some one leadme in a how to direction please |
| 16:31.52 | mutilator | -------------> |
| 16:32.03 | SarahEmm | kimosabe: is it SIP? IAX2? |
| 16:33.03 | kimosabe | sip acount |
| 16:33.41 | kimosabe | i have it on a sipura know but i want to integrate it to my asterisk server so that via my sipura i can take that ip acount and use it |
| 16:34.51 | SarahEmm | you have * set up already? |
| 16:35.46 | kimosabe | i just have 10 sipura pointing to my asterisk box nothing fancy |
| 16:36.22 | SarahEmm | okay, so all you're trying to do now is set up a SIP peer for your selectfone account? |
| 16:36.44 | *** join/#asterisk klasstek (n=nunyobiz@sta-206-168-218-202.rockynet.com) |
| 16:36.58 | kimosabe | yes so that my sips can take that vopi trunk and use it |
| 16:37.06 | *** join/#asterisk Hogie (i=daniel@xbox.gamingzen.com) |
| 16:37.12 | Hogie | howdy guys |
| 16:37.21 | leszq | Who knows what will the call quality be if second side is in USA, I am in Poland ....between us is about 180 ms ping delay ... upload/download is about 250 kbit/s in each direction ?? I want use SIP |
| 16:37.52 | LoRez | 180ms delay will suck rocks |
| 16:38.01 | SarahEmm | meep, i need to run.. sorry kimosabe, hopefully someone else can give you a hand.. and read the wiki :) |
| 16:38.02 | SarahEmm | ~rtfw |
| 16:38.03 | jbot | well, rtfw is http://www.voip-info.org, the only place to get any real answers about * and it's many, many uses. |
| 16:38.36 | Hogie | Unable to create channel of type 'IAX2' <-- Im getting that on a dial(iax2/watson@campwisdom/206@intercom) statement... which worked for months, until yesterday for some reason... |
| 16:38.49 | Hogie | and I dont know why:\ |
| 16:39.00 | file | iax2 show peer campwisdom |
| 16:40.05 | Hogie | its showing unspecified for the host.. but when I reload chan_iax2.so, it is registering with it |
| 16:40.29 | file | registering is so the other side can get to you... not the other way around |
| 16:40.52 | *** join/#asterisk wunderkin (i=kev@12-219-162-233.client.mchsi.com) |
| 16:41.11 | Hogie | the only change was that our main t1 went down yesterday, (and I was out sick), so when it stopped working, they tried rebooting both boxes, but they didn't change any configs (I already checked md5's I had of the asterisk dirs)... |
| 16:41.41 | Hogie | and it was out data T, not a PRI |
| 16:45.12 | *** join/#asterisk pa (n=Paolo@unaffiliated/pa) |
| 16:45.14 | *** join/#asterisk jets (n=jets@dsl001-136-136.lax1.dsl.speakeasy.net) |
| 16:48.35 | Hogie | thanks file |
| 16:48.43 | *** join/#asterisk salmandr (n=salmandr@mdsnwigjbas01-pool0-a85.mdsnwigj.tds.net) |
| 16:54.26 | file | :) |
| 17:01.37 | *** join/#asterisk bweschke (n=bweschke@dsl001-136-136.lax1.dsl.speakeasy.net) |
| 17:02.09 | *** join/#asterisk steve___ (n=steve@store-fw.porchlight.ca) |
| 17:04.22 | *** join/#asterisk AsteriskNoob (i=BoredBoz@207-114-232-10.gen.twtelecom.net) |
| 17:06.29 | *** join/#asterisk Tili (n=Tili@202-133-67-30-dialup.sat.net.pk) |
| 17:10.26 | *** join/#asterisk Vlat- (n=vladimir@213.134.0.111) |
| 17:10.28 | Vlat- | hi |
| 17:11.37 | AsteriskNoob | blech |
| 17:11.52 | lancey | http://bugs.digium.com/view.php?id=5433 |
| 17:11.59 | lancey | anyone came across that? |
| 17:12.04 | *** join/#asterisk fordvoice (n=chrisf0r@cpe-69-133-21-43.cinci.res.rr.com) |
| 17:12.10 | hardwire | my ABP rep just called me from astricon to rub it in |
| 17:12.12 | hardwire | that bastard |
| 17:12.26 | AsteriskNoob | ABP rep? |
| 17:12.34 | hardwire | lancey: is your extconfig.conf set to set up sip.conf? |
| 17:12.43 | hardwire | AsteriskNoob: ABP is a voip hardware resale company |
| 17:12.49 | lancey | hardwire? |
| 17:12.57 | hardwire | lancey: check your /etc/asterisk/extconfig.conf |
| 17:12.57 | AsteriskNoob | oh |
| 17:13.09 | lancey | hardwire it sometimes works |
| 17:13.12 | lancey | couldn't be that |
| 17:13.22 | lancey | i believe it's some interoperability issue |
| 17:13.47 | lancey | yes, i have sippeers from db |
| 17:13.55 | Vlat- | sorry gyus, I know you're using asterisk, but I have a ser question. #ser seems to be dead. Is there other channels where I can ask a question about ser+asterisk connection ? |
| 17:13.57 | lancey | how does this change the situation? |
| 17:14.09 | lancey | sip show peer the-sip-peer |
| 17:14.13 | lancey | shows correct setting |
| 17:14.16 | hardwire | ok |
| 17:14.18 | hardwire | bbl |
| 17:16.25 | JamesDotCom | what's the question Vlat |
| 17:17.46 | JamesDotCom | to a degree, yes |
| 17:18.25 | JamesDotCom | wrong channel |
| 17:18.26 | JamesDotCom | ;( |
| 17:18.29 | Vlat- | JamesDotCom: everything works fine, 'till we get the call from outside, let's say from one of our DID numbers. In that case the person at the other end receive fast busy. And we have nothing in asterisk logs |
| 17:18.53 | Vlat- | oh, sorry, i didn't mentioned that asterisk is voicemail backend |
| 17:19.09 | JamesDotCom | well have you looked at ser's logs? |
| 17:19.38 | Vlat- | ser just sending back the proper code, 404, 486 or 408 ever |
| 17:20.16 | Vlat- | so i thing i lost the point right now. why does ser do such thing, if everything is working fine inside our network |
| 17:20.22 | JamesDotCom | where do the did's come from? |
| 17:21.13 | Vlat- | it's ip outside our segment, it's in trusted table of ser, and it's working, so subscribers receive the ring and use their did-s w/o any problem |
| 17:21.51 | Vlat- | voicemail diversion is done by t_on_failure[x] |
| 17:22.00 | JamesDotCom | sorry, too late at night to comprehend |
| 17:22.08 | JamesDotCom | and on that note i should go to bed |
| 17:22.08 | JamesDotCom | haha |
| 17:22.12 | Vlat- | understand you :) |
| 17:22.20 | Vlat- | have a nice rest@ |
| 17:22.21 | Vlat- | ! |
| 17:22.34 | JamesDotCom | haha np ;) |
| 17:22.39 | AsteriskNoob | ser? |
| 17:22.41 | AsteriskNoob | what's ser? |
| 17:23.09 | Vlat- | AsteriskNoob: are you serious ? :) |
| 17:23.25 | AsteriskNoob | all i know its it's a sip proxy |
| 17:23.25 | AsteriskNoob | lol |
| 17:23.35 | Vlat- | it is :) |
| 17:23.38 | AsteriskNoob | why use anything but asterisk? lol |
| 17:23.50 | Vlat- | ehh, really hate questions like this |
| 17:23.51 | AsteriskNoob | i love my asterisk box |
| 17:23.59 | Vlat- | ser is sip PROXY, asterisk is for anything other |
| 17:23.59 | Lee__ | but are you /in love/ with it? |
| 17:24.00 | JamesDotCom | ahaha |
| 17:24.10 | JamesDotCom | <3 ser |
| 17:24.16 | AsteriskNoob | asterisk has a sip proxy built in ;) |
| 17:24.26 | AsteriskNoob | sure lee, i'm in love with my asterisk box |
| 17:24.28 | AsteriskNoob | it roxors |
| 17:24.34 | JamesDotCom | AsteriskNoob: read the sip rfc, rfc 3261 |
| 17:24.34 | Lee__ | ewwww, gross. |
| 17:24.35 | InfraRed | can ser add SIP headers? |
| 17:24.36 | Vlat- | asterisk's sip implementation is something strange for me |
| 17:24.42 | AsteriskNoob | hahahahahaha |
| 17:24.44 | AsteriskNoob | j/k |
| 17:24.47 | JamesDotCom | you'll do yourself a huge favour if you want to learn about sip |
| 17:25.00 | BrianR___ | I made the horrible mistake of rolling out grandstream |
| 17:25.01 | BrianR___ | <PROTECTED> |
| 17:25.05 | Vlat- | i didn't get budgetone to register correctly with it, with ser it worked at the first try |
| 17:25.08 | AsteriskNoob | james, i know SIP, and i havent had ANY problems with sip in asterisk |
| 17:25.16 | JamesDotCom | anyway, 3:24am ;( |
| 17:25.20 | *** join/#asterisk cio (n=na@adsl-072-149-159-016.sip.bhm.bellsouth.net) |
| 17:25.23 | Vlat- | grandstreams are cheap, but now we're moved to pap2-s |
| 17:25.29 | AsteriskNoob | then again i'm not doing any nat or anything |
| 17:25.36 | JamesDotCom | well then you'd understand exactly what ser is for ;) |
| 17:25.48 | JamesDotCom | trust me, there's more to learn about sip than you think |
| 17:25.49 | JamesDotCom | *gone* |
| 17:25.55 | BrianR___ | Vlat-: pap2? isn't that an ATA? |
| 17:26.06 | shido6 | fresh out |
| 17:26.13 | Vlat- | BrianR___: ATA is old and buggy piece of shi...hardware :) |
| 17:26.18 | AsteriskNoob | i'm running 8 cisco 7960's :) |
| 17:26.19 | Vlat- | PAP2 is rather Sipura |
| 17:26.23 | cio | Hi all. Does * tag it's packets as voice? |
| 17:26.30 | shido6 | pap2s sound better than my 7960 |
| 17:26.34 | Vlat- | Cisco bought the hardware last year |
| 17:26.38 | AsteriskNoob | yes cio it sets TOS bits |
| 17:26.48 | nomazda | vlat did you grab any at that latest staples frenzy? |
| 17:26.49 | Vlat- | btw, nobody from vonage here ? |
| 17:26.59 | cio | Any particular switch you guys like? |
| 17:26.59 | denon | doubtful |
| 17:27.00 | AsteriskNoob | oooo-oooooooo oooo-ooo-ooo |
| 17:27.04 | Vlat- | we was able to unlock the latest PAP2 firmware :)) |
| 17:27.07 | BrianR___ | I need POE powered IP phones for offices.. |
| 17:27.11 | nomazda | yea.. sweet deal ;) |
| 17:27.17 | lancey | cio: * can set the ToS bit |
| 17:27.20 | shido6 | how many, BrianR___ ? |
| 17:27.23 | AsteriskNoob | cio, i love my cisco 3524XL-PWR-EN |
| 17:27.36 | Vlat- | good tip for everyone: never buy PAP2-s from Ebay |
| 17:27.43 | Vlat- | usually they're vonage-locked |
| 17:27.48 | lancey | allied telesyn's 8700's also work flawless |
| 17:27.53 | Vlat- | and people tell nothing about it |
| 17:27.54 | BrianR___ | shido6: bought 20 gxp-2000's, if the past few days are any suggestion it probably means they're all going to fail :( |
| 17:28.15 | Vlat- | BrianR___: wait for the new firmware |
| 17:28.15 | lancey | Vlat- there's a PAP2-NA, which is unlocked |
| 17:28.20 | lancey | and sells regularly out here |
| 17:28.26 | lancey | at $65 or so |
| 17:28.27 | BrianR___ | Vlat-: We're seeing phones that get unusually warm, lock up. |
| 17:28.32 | Vlat- | lancey: we thought they're NA-s |
| 17:28.32 | AsteriskNoob | lets see, cisco switch mixed with cisco 7960's words GREAT they sync like they belong together or something ;) |
| 17:28.47 | cio | heh |
| 17:28.51 | Vlat- | so the guy sold this option received some bad reviews |
| 17:29.08 | lancey | well, if he said PAP2 only |
| 17:29.12 | BrianR___ | You can get an actual sipura spa-21xx for cheap money too... It seems they've hidden a lot of the nifty configurables in the linksys branded one.. |
| 17:29.14 | Vlat- | BrianR___: heh, i have a tip, and maybe even a solution |
| 17:29.15 | lancey | it's not his mistake :) |
| 17:29.21 | BrianR___ | Vlat-: tell me more. |
| 17:29.24 | Vlat- | but if you follow it you'll lose the guarantee |
| 17:29.38 | Vlat- | there to stabilizer inside |
| 17:29.41 | Vlat- | LM7805 |
| 17:29.44 | lancey | BrianR___: what's missing? |
| 17:29.47 | Vlat- | have no idea why two |
| 17:30.24 | BrianR___ | lancey: some of the stuff related to auto-changing the jitter buffer settings when doing g711 passthrough of fax, and the t38 stuff. |
| 17:30.37 | Vlat- | try to change them to the same type. it seems they're using a broken type (or the power supply is not for gxp) |
| 17:30.38 | lancey | hmz |
| 17:31.22 | BrianR___ | Vlat-: We're powering these over POE - does that matter? |
| 17:31.44 | Vlat- | BrianR___: what's the current ? |
| 17:31.57 | Vlat- | btw, 7805-s look like this (there's the full pdf): http://pdf.alldatasheet.com/datasheet-pdf/view/85503/ETC/LM7805.html |
| 17:32.12 | BrianR___ | We're also using the .12 firmware on the GXP2000's - not sure if that matters. it was the only way to make the speakerphone usable though. |
| 17:32.24 | Vlat- | just take a look at pdf, search for min/max current/voltage parameters |
| 17:32.47 | Vlat- | then pick up a multimeter and watch the parameters at the POE's endpoint |
| 17:33.16 | Vlat- | then compare two, if they're slight greater than sheet writes, you have to stabilize them somehow |
| 17:35.11 | BrianR___ | Vlat-: *sigh* |
| 17:35.21 | tzafrir_laptop | any gentoo users here? bsd users? if so: do those distros come with libgsm , and if so: to where exactly is it installed? (gsm.h, libgsm.so) |
| 17:35.27 | BrianR___ | Vlat-: That might be a practical fix if I had a handful of problem phones, but it looks like I may have 20 or more to fix. |
| 17:36.04 | lancey | tzafrir_laptop: FreeBSD doesn't come with one included |
| 17:36.07 | Vlat- | BrianR___: maybe you have a trouble not with the phones, instead with POE power supply |
| 17:36.12 | *** join/#asterisk supaigtr (n=yurplsl@152.53.16.10) |
| 17:36.14 | Vlat- | to be honest never liked POE solutions :) |
| 17:36.16 | supaigtr | Hello. |
| 17:36.23 | Vlat- | too much problems with them |
| 17:36.24 | Vlat- | hi |
| 17:36.34 | BrianR___ | Could be 30+ minutes of rework per phone. Engineering would get pissed if I stole like 2 days worth of technician time... :( |
| 17:37.09 | BrianR___ | Vlat-: POE is the only way to deal with availability issues without putting generator power outlets or UPS's at each office... |
| 17:38.57 | cio | Do the Cisco switches come out of the box prioritizing voice or do you have to go build a config? |
| 17:38.58 | Vlat- | BrianR___: unfotently i know :( |
| 17:39.13 | supaigtr | I have having tiny gaps in audio on an PRI-> *IAX2 - >*IAX2 -> polycom. Any ideas? Is jitterbuffer the answer? |
| 17:40.56 | cio | Does asterisk use 802.11p or 802.11Q? |
| 17:41.17 | *** join/#asterisk RoyK (n=roy@host-81-191-147-248.bluecom.no) |
| 17:41.22 | *** part/#asterisk Seyr (n=Seyr@rrcs-24-227-133-226.sw.biz.rr.com) |
| 17:42.21 | *** join/#asterisk lsf_work (n=lsf@morpheus.dataguard.no) |
| 17:42.50 | *** join/#asterisk justinu (n=j2@dsl001-136-136.lax1.dsl.speakeasy.net) |
| 17:44.34 | lsf_work | Anyone know if Presence works ? |
| 17:45.25 | brookshire | cio: those are networking protocols... asterisk can work over them |
| 17:45.39 | brookshire | but asterisk does not work directly with them |
| 17:46.31 | cio | So with the TOS bits already set, switches will automatically prioritize them? |
| 17:47.19 | brookshire | not automatically.. asterisk has no control over that |
| 17:47.24 | *** join/#asterisk iCEBrkr (i=icebrkr@rrcs-24-129-130-158.se.biz.rr.com) |
| 17:48.06 | *** join/#asterisk zoa (n=zoa@host06.alica.hyatthsiagx.com) |
| 17:48.09 | *** join/#asterisk NetSkier (n=ns@ca-redbch-cuda1-c3a-199.stmnca.adelphia.net) |
| 17:48.23 | lancey | cio probably not |
| 17:48.43 | lancey | some switches do come preconfigured with common qos |
| 17:48.54 | lancey | but you'll probably have to manually configure it |
| 17:48.57 | lancey | to run as expected |
| 17:52.12 | *** join/#asterisk zeedo (n=zeedo@80.68.92.188) |
| 17:52.15 | *** join/#asterisk Gunnar (n=gunnar@34.80-203-209.nextgentel.com) |
| 17:53.01 | *** part/#asterisk loick (n=loick@ATuileries-151-1-55-19.w83-202.abo.wanadoo.fr) |
| 17:54.09 | *** join/#asterisk loick (n=loick@ATuileries-151-1-55-19.w83-202.abo.wanadoo.fr) |
| 17:55.01 | cio | Thanks, lancey. |
| 17:55.26 | Vlat- | sometimes i thing about changing our infrastructure completely to asterisk. currently we have 1000+ customers(sip only) and ser+asterisk(media backend) configuration. |
| 17:55.37 | Vlat- | Is it a good idea ? |
| 17:55.41 | Vlat- | thing-think |
| 17:56.10 | Vlat- | the only thing i dislike in ser - is the configuration file :) |
| 17:57.06 | *** join/#asterisk file (n=jcolp@mctnnbsa31w-142166094161.nb.aliant.net) |
| 17:57.52 | zoa | hey file |
| 17:57.55 | file | hi |
| 17:58.20 | Katty | anyone work at nasa branch office in here? |
| 17:58.58 | *** join/#asterisk harryvv (n=none@S010600a0c93f6f7e.vs.shawcable.net) |
| 17:59.30 | Laibsch | I installed Asterisk@Home on a computer. I choose custom installation. Apparently for some reason, the web server was not installed. How can I add that now? Do I have to reinstall? |
| 17:59.46 | file | Laibsch: I'd suggest going to the AAH channel |
| 18:00.32 | Laibsch | Thanks. Which one is that? #aah did not work. |
| 18:00.42 | file | not a clue, I don't use it |
| 18:00.55 | *** join/#asterisk n0where (n=kc@dsl001-136-136.lax1.dsl.speakeasy.net) |
| 18:06.20 | *** join/#asterisk ManxPower (n=eric@slip-12-64-91-14.mis.prserv.net) |
| 18:07.00 | *** join/#asterisk fri (n=fri@port84.ds1-sdb.adsl.cybercity.dk) |
| 18:11.12 | *** join/#asterisk PBXtech (n=nik@dsl001-136-136.lax1.dsl.speakeasy.net) |
| 18:12.11 | *** join/#asterisk Tili (i=Tili@202-133-65-213-dialup.sat.net.pk) |
| 18:12.59 | *** join/#asterisk syzygyBSD (n=chatzill@66.226.228.204.cpe.speedyquick.net) |
| 18:13.49 | *** join/#asterisk funxion (n=nunya@mtnuser.icgws.com) |
| 18:14.19 | synthetiq | how do u restart or flush zap cards with out restarting asterisk, ? |
| 18:14.31 | *** join/#asterisk michael_fc4linux (n=michael@cpe-24-242-43-177.hot.res.rr.com) |
| 18:16.08 | tzanger | dammit jerjer's not here |
| 18:16.13 | tzanger | Katty: ?? |
| 18:16.19 | Katty | tzanger: mew? |
| 18:18.35 | *** join/#asterisk Cresl1n (n=matt@dsl001-136-136.lax1.dsl.speakeasy.net) |
| 18:26.30 | syzygyBSD | I am trying to add a PRI in addition to a channel bank but keep getting ZT_SPANCONFIG failed on span 2: Invalid argument (22) |
| 18:26.46 | *** join/#asterisk generalhan (i=general_@ip67-90-64-2.z64-90-67.customer.algx.net) |
| 18:26.51 | generalhan | whats up everyone ? |
| 18:27.05 | syzygyBSD | http://rafb.net/paste/results/LuX5iy27.html is my config file |
| 18:27.13 | *** join/#asterisk [VIRIi] (n=virii@68-186-170-246.dhcp.smrt.tn.charter.com) |
| 18:27.28 | *** join/#asterisk kimosabe (n=nat@201.145.2.86) |
| 18:27.35 | syzygyBSD | just beating myself with the keyboard trying to figure this out... |
| 18:28.06 | syzygyBSD | whats up with you? |
| 18:28.12 | [VIRIi] | when trying to compile zaptel i get this message. make: *** [linux26] Error 2 |
| 18:28.16 | [VIRIi] | make: *** SUBDIRS=/usr/src/zaptel: No such file or directory. Stop. |
| 18:31.18 | *** part/#asterisk n0where (n=kc@dsl001-136-136.lax1.dsl.speakeasy.net) |
| 18:31.36 | syzygyBSD | [VIRIi]: where are you making it from? ie, cd /usr/src/zaptel && make |
| 18:32.32 | kimosabe | where can i find a config exapmle for configuring delthatree with asterisk |
| 18:33.41 | *** join/#asterisk djin_ib (n=djin_ib@gridfox.xs4all.nl) |
| 18:33.49 | *** join/#asterisk Los415 (n=los415@c-24-126-63-233.hsd1.ca.comcast.net) |
| 18:34.41 | kimosabe | i just purchased an account from selectfone.com how can i configure it on my asterisk box |
| 18:34.45 | kimosabe | a how to ? |
| 18:34.48 | *** join/#asterisk Guggemand (i=irc@tester2.har-tabt.dk) |
| 18:35.09 | syzygyBSD | kimosabe: have you looked at voip-info yet? |
| 18:35.24 | synthetiq | how do u restart or flush zap cards with out restarting asterisk, ? |
| 18:35.49 | kimosabe | im there but i dont see any exapmles or dont know what im looking for |
| 18:35.53 | ManxPower | synthetiq, define "restart or flush" |
| 18:36.27 | ManxPower | you can make changes to /etc/zaptel.conf and then run ztcfg -vvv but that will terminate any active zap calls |
| 18:37.06 | ManxPower | With 1.2beta/CVS-HEAD you can issue a "reload chan_zap.so" at the Asterisk CLI and that will apply most changes you may to /etc/asterisk/zapata.conf |
| 18:37.36 | ManxPower | You can also "unload chan_zap.so" and then "load chan_zap.so" at the Asterisk CLI, but that won't work if there are any active zap calls |
| 18:37.43 | *** join/#asterisk shido6 (n=curtis@d221-68-210.commercial.cgocable.net) |
| 18:37.47 | Lee__ | why would Asterisk use the defined SIP proxy to ring the other end but bypass that proxy when the call is connected and RTP is flowing? |
| 18:37.56 | tainted_ | tzanger you around? |
| 18:38.06 | ManxPower | Lee__, you prolly don't have a canreinvite=no in sip.conf |
| 18:38.22 | Nyvar | lee, is that proxy doing record_route? |
| 18:38.25 | Vlat- | what does canreinvite field actually do ? |
| 18:38.44 | ManxPower | Vlat-, it prevents RTP frpm flowing directly between the two SIP endpoints |
| 18:38.55 | Lee__ | Nyvar: I don't think so, it's just rewriting the host and forwarding the packet along to a termination service. |
| 18:39.08 | tainted_ | anyone interested in a freelance project? |
| 18:39.08 | Vlat- | ManxPower: thanx. so it forces asterisk to proxy the rtp ? |
| 18:39.15 | *** join/#asterisk mud (n=mud@206-248-138-115.dsl.teksavvy.com) |
| 18:39.27 | tainted_ | i need an asterisk/linux/agi guru |
| 18:39.28 | ManxPower | Vlat-, correct. It's needed for situations where NAT is involved. |
| 18:39.42 | Vlat- | ManxPower: thanks again, may be handy on nat solutions |
| 18:39.52 | Nyvar | vlat, yes, canreinvite=no forces the rtp through * |
| 18:40.21 | Lee__ | thanks. I think that may be my problem. |
| 18:40.31 | tzanger | tainted_: yes |
| 18:40.53 | *** join/#asterisk rt (n=markv@c-67-180-32-90.hsd1.ca.comcast.net) |
| 18:41.19 | *** join/#asterisk Piranha- (i=piranha@d198-166-226-58.abhsia.telus.net) |
| 18:41.41 | Katty | tzanger: did you want something? |
| 18:42.34 | Vlat- | tainted_: i'm not AGI guru, but had a little experience with it. What do you actually need ? |
| 18:42.39 | tzanger | Katty: just wondering why there'd be NASA people in here |
| 18:42.44 | tzanger | seems kind of ... low for them |
| 18:43.13 | generalhan | can some one tell me how to terminate a Meetme room ? |
| 18:43.14 | Lee__ | but here's what I don't understand: I don't care if the actual RTP stream goes between endpoints but why are SIP headers getting transmitted directly between the endpoints? Those two things should be sepperate, right? |
| 18:43.44 | ManxPower | Lee__, Do you have NAT? |
| 18:43.55 | kimosabe | does any one have a sip acount configured on there server so there sipuras can take the accouht and dial to the world |
| 18:44.06 | mmlj4 | hey ManxPower |
| 18:44.07 | Lee__ | yes, Asterisk is behind NAT and the proxy isn't. |
| 18:44.17 | mmlj4 | i hear i'm carrying you to gulfport friday |
| 18:44.21 | ManxPower | Lee__, Anytime you have NAT, weird stuff happens |
| 18:44.27 | ManxPower | mmlj4, First I've heard of it. |
| 18:44.30 | Lee__ | spooky |
| 18:44.40 | ManxPower | I know I'm going to gulfport on friday, didn't know how I was getting there. |
| 18:45.04 | tainted_ | are u guys at astricon? |
| 18:45.05 | mmlj4 | i'm your chauffeur and flunky for the day |
| 18:45.06 | Vlat- | usually STUN + a proper forwarding completely solve every NAT problemm |
| 18:45.16 | tainted_ | i wish i could go:( |
| 18:45.16 | Vlat- | becouse it's not a nat anymore |
| 18:45.20 | hypa7ia | tainted_: come next year |
| 18:45.23 | hypa7ia | it's awesome |
| 18:45.29 | ManxPower | tainted_, Trust me, you don't want to go to Gulfport. |
| 18:45.37 | ManxPower | Unless you like looking at destroyed buildings |
| 18:45.58 | mmlj4 | ManxPower: many tourists in NO right now, looking at the destruction |
| 18:46.15 | ManxPower | Vlat-, I don't have NAT problems and I never use STUN or a proxy |
| 18:46.16 | ManxPower | mmlj4, *gag* |
| 18:46.23 | ManxPower | The stench alone is enough to kill you. |
| 18:46.50 | mmlj4 | actually, it's a lot better now |
| 18:47.04 | mmlj4 | went to lakeview this AM |
| 18:47.27 | kimosabe | hey can some one tell me hos to configure my asterisk box to use a pstn so that my clients can access it via sipura |
| 18:47.52 | tainted_ | hypa7ia the sad thing is i'm in the LA area |
| 18:48.17 | tainted_ | hypa7ia ManxPower but i just can't get time off work.. ironic thing is.. i'm stuck here b/c of an asterisk project |
| 18:48.27 | mmlj4 | hey, kimosabe (always wanted to say that), you need hardware to connect to PSTN |
| 18:48.38 | mmlj4 | ;-) |
| 18:49.01 | ManxPower | mmlj4, I need to save up enough money to get out of Limping Mule Texas |
| 18:49.03 | mmlj4 | tainted_: then why can't you attend astricon? |
| 18:49.07 | bjohnson | mmlj4: no you don't |
| 18:49.13 | mmlj4 | heh, i'll bet |
| 18:49.33 | mmlj4 | bjohnson: someone needs hardware |
| 18:49.34 | bjohnson | yep, but he doesn't |
| 18:49.42 | hypa7ia | tainted_: tragic! |
| 18:49.54 | bjohnson | kimosabe: subscribe to a voip provider and they will connect you to the pstn for a fee |
| 18:49.58 | kimosabe | mmlj4 i have asterisk box with 10 sipuras and i have a selectfone account i want the selectfone account to be accesible via sipura but dont know how to configure it |
| 18:50.02 | mmlj4 | he has to rent it, if not use it himself |
| 18:50.25 | kimosabe | im renting it already |
| 18:50.29 | bjohnson | kimosabe: I'm not familiar with slectfone, is it working for you? |
| 18:50.42 | kimosabe | yes its gfreat service and cheap prices |
| 18:50.48 | mmlj4 | kimosabe: i have no idea how to connect * to selectphone..... though once you have that, getting the sipuras to dial out is the easy part, probably |
| 18:51.09 | kimosabe | is there a config exapmle ? |
| 18:51.10 | bjohnson | in exten.conf, accept calls from the sip phone and dial() the number though the selectfone channel |
| 18:51.12 | ManxPower | mmlj4, I've decided to put up flyers. "Do you have DSL? Earn up to $20/month! Learn How by Calling 903-796-1234!" |
| 18:51.26 | mmlj4 | heh |
| 18:51.26 | ManxPower | mmlj4, Then I'll install my own damn wireless uplink to whoever wants $20/month |
| 18:51.32 | tainted_ | ManxPower are you in town? |
| 18:51.37 | ManxPower | tainted_, Which town? |
| 18:51.47 | tainted_ | los angeles area |
| 18:51.48 | ManxPower | tainted_, no |
| 18:52.00 | ManxPower | I have to do an Asterisk install on Friday |
| 18:52.10 | tainted_ | that's too bad |
| 18:52.16 | tainted_ | it would've been nice to meet you |
| 18:53.05 | mmlj4 | but for that killer hurricane, you might have gotten your wish |
| 18:53.06 | tzanger | someone smack jerjer and tell him that nufone's mishandling calls to canada :-) |
| 18:53.07 | bjohnson | kimosabe: thousands of examples. probably none that you can blindly copy thoug |
| 18:53.16 | bjohnson | my son popped my 'h' key off my keyboad and broke the little clips that are supposed to hold it on |
| 18:53.22 | kimosabe | bjohnson lead me in the direction please |
| 18:53.30 | bjohnson | in exten.conf, accept calls from the sip phone and dial() the number though the selectfone channel |
| 18:54.32 | bjohnson | extensions.conf |
| 18:54.35 | bjohnson | I meant |
| 18:59.38 | kimosabe | how can i see if my asterisk box registered with selectfone |
| 18:59.58 | ManxPower | kimosabe, "iax2 show registry" or "sip show registry" |
| 19:00.34 | ManxPower | But registration only affects calls from the remote server TO your local server. |
| 19:00.44 | kimosabe | cool im regfistered with selectfone |
| 19:01.00 | kimosabe | manxpower a readme so that i can take that line via my sipura ? |
| 19:02.07 | kimosabe | in otherwords i want to be able to make calls from that line with my sipuras |
| 19:02.48 | *** join/#asterisk mcf3782 (n=mcf3782@adsl-065-012-184-148.sip.asm.bellsouth.net) |
| 19:05.09 | *** join/#asterisk n4y (n=tmalkut@fw.orasoft.net.pl) |
| 19:07.59 | mcf3782 | I need a little help understanding what I've configured wrong that isn't allowing me to make outbound local calls. |
| 19:08.37 | mcf3782 | In the [globals] section of my extensions.conf I have "OUTBOUNDTRUNK=Zap/1". |
| 19:09.02 | mcf3782 | Here in Atlanta, Georgia, we have to do 10-digit dialing, even for local calls as we have multiple "local" areacodes. |
| 19:09.28 | mcf3782 | So I have a statement like this for local calls. "exten => _91NXXNXXXXXX,1,Dial(${OUTBOUNDTRUNK}/${EXTEN:1})" |
| 19:09.39 | *** join/#asterisk stkn (i=nobody@gentoo/developer/pdpc.active.stkn) |
| 19:09.41 | wunderkin | ok |
| 19:10.09 | mcf3782 | What I think I should be able to do is dial 97705551212 and get an outbound call. |
| 19:10.14 | wunderkin | no |
| 19:10.28 | mcf3782 | ok. |
| 19:10.29 | wunderkin | 917705551212 |
| 19:10.31 | ManxPower | mcf3782, not if your pattern starts with _91 |
| 19:10.43 | mcf3782 | D'OH! |
| 19:10.54 | mcf3782 | thanks. |
| 19:10.57 | mcf3782 | goes to edit |
| 19:11.04 | kimosabe | manx power i have this i have regfistry from voip provider |
| 19:11.12 | kimosabe | i put context but it dont work |
| 19:11.12 | wunderkin | do an extensions reload from the cli when done |
| 19:11.50 | kimosabe | i did stop now and asterisk -vvvgc |
| 19:12.43 | jarrod | you did not |
| 19:13.37 | kimosabe | does it have to be in same context for this to work ? |
| 19:13.46 | mcf3782 | ok. I corrected that to now look like "exten => _9NXXNXXXXXX,1,Dial(${OUTBOUNDTRUNK}/${EXTEN:1})" |
| 19:14.41 | mcf3782 | if I do "dial 7705551212", the message that's returned is "no such extension '97705551212". |
| 19:15.10 | mcf3782 | I'm using real local numbers in place of the 5551212, just trying to keep it simple on here. |
| 19:15.28 | wunderkin | mcf3782 and you did an extensions reload? |
| 19:15.34 | mcf3782 | yep |
| 19:15.41 | wunderkin | must not be in the same context |
| 19:16.07 | mcf3782 | ok. I'll look closer at that. I thought I had it in the proper place. |
| 19:16.18 | wunderkin | check what context your phone is in |
| 19:16.19 | kimosabe | why doesnt it unhang my line |
| 19:17.12 | mcf3782 | I'm issuing that 'dial' command from the asterisk console, not a phone. Trying to test one step at a time. :) |
| 19:17.20 | *** join/#asterisk DaPrivateer (i=Privatee@CRIMSON.OFF-HOURS.COM) |
| 19:18.26 | wunderkin | not sure that way, may be using default dunno.. try from a phone |
| 19:19.07 | outtolunc | they ever gonna stream astricon? |
| 19:19.13 | *** join/#asterisk pa (n=Paolo@unaffiliated/pa) |
| 19:19.19 | wunderkin | dont know, the dev conference was |
| 19:19.40 | wunderkin | rest it probably not |
| 19:19.48 | outtolunc | ah |
| 19:19.50 | outtolunc | ty |
| 19:20.05 | wunderkin | i would understand the point of doing the dev conference so other developers that couldnt make it could at least help with the dev |
| 19:20.09 | ian_k | mcf3782: You prob don't have it in the "local" context |
| 19:20.16 | *** join/#asterisk Rowter (n=SilverDr@201.135.26.195) |
| 19:20.25 | mcf3782 | wunderkin: you were right. it wasn't in the 'local' context. I'd missed the fact that another context started just out of my edit buffer. *sigh* |
| 19:20.34 | mcf3782 | ian_k yep. |
| 19:20.35 | wunderkin | ok |
| 19:20.39 | mcf3782 | Thanks guys! |
| 19:21.03 | mcf3782 | sometimes all it takes is another pair of eyes, even if they're nowhere near me. :) |
| 19:22.11 | harryvv | in most cases iax devices will go though most firewalls? |
| 19:22.30 | harryvv | hard to demo a phone in a company when there fw is blocking sip ports. |
| 19:23.01 | wunderkin | port 4569 is all thats used i think.. udp? |
| 19:23.08 | harryvv | yes |
| 19:23.16 | *** join/#asterisk jeffgus (n=jeffgus@2002:d856:c704:0:0:0:0:1) |
| 19:23.16 | harryvv | But asking people here from experaince. |
| 19:23.26 | kimosabe | if i registered a device and my asterisk box says regfistered |
| 19:23.40 | wunderkin | well if thats not blocked it should be ok :P |
| 19:23.45 | Vlat- | harryvv: once we encapsulated a sip signalling to tcp 25 :) |
| 19:23.55 | Vlat- | smtp. it was the only open port |
| 19:24.10 | wunderkin | no interweb huh |
| 19:24.29 | jarrod | you would have to be a real momo to do that |
| 19:24.42 | wunderkin | they're out there |
| 19:25.03 | harryvv | vlat so you used the mail port then. |
| 19:25.22 | Vlat- | harryvv: there was no other solution, 'till we reach their admin |
| 19:25.44 | harryvv | okay thats easy enough to change in asterisk |
| 19:26.11 | harryvv | what about port 80 |
| 19:26.19 | Vlat- | would work either |
| 19:26.39 | Vlat- | in fact any port would do the trick, but it was only signalling (sip) |
| 19:26.40 | *** join/#asterisk [ViRIi] (n=virii@68-186-170-246.dhcp.smrt.tn.charter.com) |
| 19:26.49 | harryvv | I wonder if astrisk would except more then one port |
| 19:26.59 | Vlat- | since we're never had of use IAX, we need minimum two |
| 19:27.15 | harryvv | I see |
| 19:27.29 | Vlat- | btw, tell me guys, how IAX is good and what it's a good for ? |
| 19:27.42 | Vlat- | i know that it encapsulate sig+rtp to 1 port |
| 19:27.48 | [ViRIi] | i cant compile zaptel for some odd reason |
| 19:28.04 | Vlat- | back a disadvantage of this could be a lost of packet syncro in case of timing-failure, not ? |
| 19:28.14 | kimosabe | this is what i have |
| 19:28.18 | [ViRIi] | /bin/sh: line 0: [: argument expected |
| 19:28.19 | [ViRIi] | make -C SUBDIRS=/usr/src/zaptel modules |
| 19:28.19 | [ViRIi] | make: *** SUBDIRS=/usr/src/zaptel: No such file or directory. Stop. |
| 19:28.19 | [ViRIi] | make: *** [linux26] Error 2 |
| 19:28.22 | kimosabe | can some one help |
| 19:28.43 | jarrod | kimosabe: Eat more peanut butter |
| 19:29.06 | *** join/#asterisk loick (n=loick@ATuileries-151-1-42-82.w82-123.abo.wanadoo.fr) |
| 19:29.15 | harryvv | vlat, dont know |
| 19:30.24 | *** join/#asterisk xunil (i=xunil@66.194.40.30) |
| 19:31.39 | kimosabe | jarrod i just need helpman not all the penut butter dude |
| 19:34.46 | *** join/#asterisk CANO1982 (n=alejandr@201.255.38.171) |
| 19:34.53 | *** join/#asterisk palad1n (n=eoin@ip247.217.23.209.susc.suscom.net) |
| 19:35.31 | *** join/#asterisk n4y (n=tmalkut@fw.orasoft.net.pl) |
| 19:36.10 | *** join/#asterisk IOscanner (n=IOscanne@c-67-166-249-43.hsd1.tx.comcast.net) |
| 19:36.39 | *** join/#asterisk fugitivo (n=ajf@201.255.102.19) |
| 19:36.56 | syzygyBSD | mmm peanut butter |
| 19:38.41 | Nugget | http://devtoe.blogspot.com/2005/10/oracle-buys-innodb-will-fork-save.html <-- ha ha |
| 19:38.55 | kimosabe | can i get a hand check this out http://pastebin.com/391551 |
| 19:38.56 | *** part/#asterisk palad1n (n=eoin@ip247.217.23.209.susc.suscom.net) |
| 19:39.54 | CANO1982 | Im having some problems with the TDN400p board |
| 19:40.11 | CANO1982 | maybe some related to gains or echo cancellation |
| 19:40.14 | *** join/#asterisk heath__ (n=root@12-215-32-56.client.mchsi.com) |
| 19:40.23 | harryvv | Where can I find a wholesaler that will advertise calls to india at less then 17 cents CDN? |
| 19:40.35 | CANO1982 | may I ask here for help? |
| 19:40.35 | heath__ | anyone know off the top of their head where the DOCROOT is in AAH? |
| 19:41.01 | *** join/#asterisk paryl (n=paryl@209.236.78.59) |
| 19:41.21 | paryl | how can i turn off silence suppression in asterisk? |
| 19:41.37 | file | Asterisk doesn't support silence suppression |
| 19:41.43 | file | thus it's rather hard to turn something off that doesn't exist |
| 19:42.05 | ManxPower | paryl, Asterisk does not support Silence supression, therefore it is not logical to want to turn it off. |
| 19:42.30 | paryl | ok... well, riddle me this: |
| 19:42.46 | CANO1982 | I have 2 asterisk servers running, each one with a tdm400p board |
| 19:42.56 | paryl | i'm using a gxp-2000 and i have silence suppression turned off on it, but the sound cuts off whenever i'm not talking |
| 19:43.02 | paryl | is it the phone or asterisk?> |
| 19:43.07 | *** join/#asterisk obiyoda (n=chatzill@24-119-167-174.cpe.cableone.net) |
| 19:43.27 | ManxPower | harryvv, How about 0.159 |
| 19:43.28 | paryl | (the sound cuts off to the person i'm calling) |
| 19:43.47 | ManxPower | paryl, then it's not turned off on the client |
| 19:43.58 | ManxPower | Since asterisk will stop sending audio if it's not receiving audio |
| 19:44.03 | obiyoda | is there a way in asterisk@home to have the digital receptionist forward calls to a queue if no options are selected? |
| 19:44.31 | harryvv | Manx, is that in Canadian dollars? |
| 19:44.36 | ManxPower | paryl, you would be able to confirm this with something like tcpdump running on the asterisk box |
| 19:44.45 | nomazda | has anyone tried vbuzzer w/ asterisk? |
| 19:44.48 | ManxPower | harryvv, no, american dollars. Sorry about that. |
| 19:45.22 | harryvv | ManxPower yea, then its going to be difficult to compete against Telus. |
| 19:45.48 | CANO1982 | an each onehas an analog phone plugged to it |
| 19:46.08 | paryl | manx: how do i do that? |
| 19:46.20 | harryvv | we have a huge east indian population here and several east indians have asked if i can beat 17 cents cdn. |
| 19:46.33 | CANO1982 | everithings work fine (Zap, SIP and IAX) except qhe I try to make a call between two analog phones |
| 19:46.34 | ManxPower | paryl, "man tcpdump" |
| 19:46.35 | *** join/#asterisk n4y (n=tmalkut@fw.orasoft.net.pl) |
| 19:46.59 | tainted_ | harryvv what's the going rate? |
| 19:47.01 | paryl | yeah... i'm talking about confirming it |
| 19:47.12 | paryl | what do i look for that identifies that it's turned on? |
| 19:47.15 | file | ah crap... |
| 19:47.20 | ManxPower | paryl, tcpdump host theipofyourclient |
| 19:47.22 | CANO1982 | the sound get loud (very), seems overdriven |
| 19:47.31 | file | ah there we go, better |
| 19:47.41 | CANO1982 | any idea? |
| 19:47.44 | ManxPower | CANO1982, reset your txgain and rxgain to be -0 |
| 19:47.46 | ManxPower | ..e.r.. 0 |
| 19:48.23 | ManxPower | It looks like I'm not going to New Orleans tomorrow after all. |
| 19:48.38 | CANO1982 | yes, its 0 on both servers (sorry about my enlgish) |
| 19:49.16 | *** join/#asterisk zoa (n=zoa@host06.alica.hyatthsiagx.com) |
| 19:49.52 | CANO1982 | I?ve tried lewering it (on both sides) but nothing changes after about 9 dB |
| 19:50.28 | ManxPower | 6753 packets transmitted, 0 received, +6664 errors, 100% packet loss, time 6818831ms |
| 19:50.36 | *** join/#asterisk darby_t (n=tom@dnt56.neoplus.adsl.tpnet.pl) |
| 19:50.51 | *** join/#asterisk Moc (n=mochouin@modemcable111.229-203-24.mc.videotron.ca) |
| 19:51.07 | [ViRIi] | ls |
| 19:51.24 | [ViRIi] | anyone know why zaptel wont compile on ubuntu?> |
| 19:51.38 | CANO1982 | It can stablish perfect calls between SIP and analog phones,and IAX and analog phones |
| 19:52.23 | ManxPower | CANO1982, I have never heard of the problem you describe. |
| 19:52.27 | [ViRIi] | ls |
| 19:52.31 | *** join/#asterisk McLazarus (n=mclazaru@pcp0010896371pcs.wilog301.pa.comcast.net) |
| 19:52.50 | CANO1982 | its anoying |
| 19:53.17 | paryl | yeah, so manx... again, what is tcpdump supposed to tell me? i see nothing coming across other than regular ol' packets |
| 19:53.20 | *** join/#asterisk stkn_ (i=nobody@gentoo/developer/pdpc.active.stkn) |
| 19:53.45 | CANO1982 | ViRIi, whats the error on the compilation? |
| 19:54.21 | ManxPower | paryl, the packets should stop when you stop talking. That means the SIP device still has silence supression enabled. |
| 19:54.51 | ManxPower | paryl, of course, if you have AGREESIVE ECHO CANCEL enabled in zapata that would cause a similar issue. |
| 19:55.15 | Katty | dum de dum |
| 19:55.19 | *** join/#asterisk Assid (n=assid@203.115.64.57) |
| 19:56.35 | bjohnson | harryvv: ask unlimitel.ca |
| 19:56.39 | paryl | is that what echocancel=yes is? |
| 19:56.54 | ManxPower | paryl, no, it's a compile time option |
| 19:56.55 | CANO1982 | [ViRIi] wich error? |
| 19:57.06 | paryl | aha |
| 19:57.27 | mcf3782 | heh |
| 19:58.32 | ManxPower | I never hated trees before I moved to Limping Mule Texas |
| 19:58.38 | bjohnson | ManxPower: try a sat dish as a reflector |
| 19:58.47 | mutilator | doesn't work manx |
| 19:58.52 | mutilator | trees move too much |
| 19:58.54 | ManxPower | bjohnson, I have buildings and trees between me and the rest of the workd. |
| 19:59.13 | bjohnson | depends how many |
| 19:59.18 | mutilator | just invest ~$700 in a tower |
| 19:59.22 | ManxPower | The only AP I can connect to is a WiSIP which is down 1/2 the time |
| 19:59.24 | mutilator | $100/10ft |
| 19:59.29 | bjohnson | and stick a windmill on top |
| 19:59.29 | Katty | hmm |
| 19:59.34 | harryvv | Manx, putting up a wvoip service? |
| 19:59.56 | ManxPower | harryvv, Hell no. Not even I like that much pain. |
| 19:59.57 | harryvv | Manx, can out put up a tower? |
| 20:00.18 | bjohnson | I thought you were looking at TO. How did Limping Mule win out? |
| 20:00.21 | harryvv | Manx, contact some hams in the area thay may help |
| 20:00.25 | ManxPower | harryvv, I don't want to invest that much into something I'm only going to use for a couple of months |
| 20:00.30 | bjohnson | How did Limping Mule get anything other than a bullet |
| 20:00.46 | netsurfer | hi harryvv - u got prv |
| 20:00.49 | ManxPower | bjohnson, A friend is from here, and his family arranged for a house to rent, furniture, etc for us after Katrina |
| 20:01.10 | ManxPower | And I'm trying to save up enough money to buy a car and an RV trailer. |
| 20:02.25 | harryvv | Manx katrina affected you ? |
| 20:02.25 | ManxPower | So a $700 tower is out of the question. |
| 20:02.47 | Katty | beep! (x2) |
| 20:02.48 | paryl | manx: sorry to be an idiot, but as far as i can see, any mention of echocancel in the source code is set to 0 |
| 20:02.53 | ManxPower | harryvv, Katrina put 3ft of water in the house I was renting and destroyed (and I mean DESTROYED( %90 of the town I lived in. |
| 20:03.18 | paryl | i compiled is straight from cvs, but you did hit it, this only happens when i'm talking across the tdm400p |
| 20:03.19 | harryvv | wow, did not know. |
| 20:03.44 | harryvv | was this from the levey break or the storm surge. |
| 20:03.44 | ManxPower | hardwire, The two frinds I'm living with were New Orleans Tour Guides, so there's nothing left there for them either. |
| 20:03.52 | Vlat- | ManxPower: maybe insurance ? |
| 20:03.55 | ManxPower | harryvv, storm surge. |
| 20:04.06 | ManxPower | Vlat-, no insurance since I was not in a flood zone. |
| 20:04.11 | Vlat- | damn |
| 20:04.14 | Vlat- | fscking hard |
| 20:04.24 | harryvv | Manx, never knew. Where were you staying during katrina? |
| 20:04.29 | ManxPower | The ocean rose by 30ft where I lived. And that's before you count the huge waves. |
| 20:04.53 | *** join/#asterisk iCEBrkr (i=icebrkr@rrcs-24-129-130-158.se.biz.rr.com) |
| 20:05.01 | ManxPower | harryvv, I was staying with people in Jackson MS (several hours north of the MS gulf coast where I lived) |
| 20:05.15 | harryvv | I see |
| 20:05.21 | Vlat- | heh, today a subsciber forced me to order him a DID "somewhere in Texas". Me (dumb) ordered that DID. |
| 20:05.33 | Vlat- | It TEXAS city! |
| 20:05.35 | Vlat- | and he meant the state |
| 20:05.46 | Vlat- | so, anyone needs a DID in Texas city ? |
| 20:05.47 | Vlat- | :))) |
| 20:05.51 | harryvv | Manx, Eventually the Federal goverment will cut you a check. |
| 20:05.53 | *** join/#asterisk numbone (n=numBone@c-24-129-204-233.hsd1.fl.comcast.net) |
| 20:05.55 | ManxPower | On the bright side, one of the things that kept me from moving and living other places was what to do with my furnatire. That's not a problem any more. |
| 20:06.01 | Vlat- | will give it out for free |
| 20:06.46 | *** join/#asterisk vexorg (n=vexorg@CPE000021ded913-CM001225419164.cpe.net.cable.rogers.com) |
| 20:06.47 | ManxPower | Vlat-, no thanks, but if you get me DSL I'll do just about anything for you. |
| 20:06.57 | *** part/#asterisk numbone (n=numBone@c-24-129-204-233.hsd1.fl.comcast.net) |
| 20:07.04 | Vlat- | ManxPower: not my area, sorry :( |
| 20:07.05 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
| 20:07.06 | *** part/#asterisk heath__ (n=root@12-215-32-56.client.mchsi.com) |
| 20:07.32 | Vlat- | ManxPower: get closer to the Europe and i'll give you 7mbit DSL for a damping price of $20 |
| 20:07.35 | ManxPower | I was north of New Orleans working when my friends got the place in Limping Mule Texas. Nobody checked to find out the place is 20,000ft from the CO |
| 20:07.47 | ManxPower | Vlat-, offer me a job and handle the paperwork. |
| 20:08.11 | ManxPower | I think the post office calls this place "Atlanta Texas" |
| 20:10.23 | Vlat- | ManxPower: would be happy to do it really, but we have a full staff currently, sorry |
| 20:10.43 | Vlat- | and i'm afraid US payments and EU payments are "little" different :) |
| 20:11.08 | ManxPower | Vlat-, People here will tell you that when I'm not whining about my ISP, I do a lot of Asterisk, Telecom, and WAN related stuff. |
| 20:12.07 | Vlat- | ManxPower: i think i'll be here ocassionly from now, so if i find something for you, i'll drop you a line. Is it ok for you ? |
| 20:12.27 | ManxPower | Vlat-, I am most interested in Belgium and Netherlands |
| 20:12.33 | bjohnson | don't you want to know where in europe he is? |
| 20:12.34 | ManxPower | Vlat-, eric@fnords.org |
| 20:12.37 | *** part/#asterisk Juul (n=Juul@81.7.147.193) |
| 20:12.56 | Vlat- | if you consider Belgium, you HAVE to visit www.voxbone.com |
| 20:13.07 | Vlat- | great ITSP, we're buing DID-s from them |
| 20:14.00 | Vlat- | when i say "great ITSP" it means i don't even cosider them as our competitors, these guys are really great |
| 20:14.06 | *** join/#asterisk clyrrad (n=ddd@CPE001195f553c7-CM0011aea484a4.cpe.net.cable.rogers.com) |
| 20:14.28 | Vlat- | we changed about 10 did providers in the past years, voxbone is the only really working 24/7+support :) |
| 20:15.14 | clyrrad | Why do my MySQL * CDR records not show userfield or account code in each entry? Only whenthe call starts. For all other actions during the call, the actions are recorded with out this information. Why is that? |
| 20:17.30 | Vlat- | maybe due transfer call ability ? |
| 20:17.41 | Vlat- | but maybe i'm wrong |
| 20:18.03 | clyrrad | what do u mean by transfer call ability? |
| 20:18.43 | Vlat- | clyrrad: you're able to transfer the call to the 3rd party. and the accounting information would be changed in this case |
| 20:19.06 | ManxPower | clyrrad, In this setup Server A -> Server B -> Server C, and IAX2 call can bypass Server B and so the CDRs can be wrong. |
| 20:19.12 | Vlat- | we're using SER, and the things working a bit different, but we had the problem similar like you said due the call transfers |
| 20:19.28 | Vlat- | also there're Hold call |
| 20:19.34 | Vlat- | even worse |
| 20:20.00 | *** join/#asterisk mutilator (n=animenod@65.111.201.79) |
| 20:20.17 | Vlat- | in other words, call-id info will be stampted only after the call was made |
| 20:20.28 | clyrrad | Right now, its just a matter of when a call comes in, its recorded, and we are only forwarding the call to an internal extension, this is when the information gets lost. |
| 20:20.33 | Vlat- | when asterisk/ser 100% know all the information about session |
| 20:20.43 | clyrrad | What is the work around for this if thats the case? How do you bill your customers? |
| 20:21.04 | Vlat- | we wrote the own billing system |
| 20:21.17 | Vlat- | collector script starting from the cron every minute |
| 20:21.22 | clyrrad | but the billing system had to use the CDR's did it not? |
| 20:21.42 | Vlat- | yes, but ONLY if there is Call start and Call End |
| 20:21.46 | Vlat- | so if the call ended |
| 20:21.48 | *** part/#asterisk CANO1982 (n=alejandr@201.255.38.171) |
| 20:22.13 | Vlat- | the thing you're talking about is similar to RADIUS interimization for me |
| 20:22.17 | *** join/#asterisk enemy^x (i=lkqw@212.62.250.98) |
| 20:22.40 | Vlat- | when you trigger asterisk pool every given amount of time by RADIUS server for the connection data |
| 20:22.52 | enemy^x | Im trying to figure out how to use hint to establish the presence within eyebeam and asterisk, could anyone show me an example extensions.conf where this actually works? Im running 1.0.9 |
| 20:22.54 | Vlat- | and i'm afraid it can be done only with the radius |
| 20:22.56 | clyrrad | Ok so how did you know when a call was started and ended for a particual user? |
| 20:23.31 | Vlat- | clyrrad: it's simple. There's CDR entry with non-zero start-time and end-time in your CDR file |
| 20:23.36 | Vlat- | or database |
| 20:24.09 | clyrrad | yes, I am looking in the database right now, I see what you are refering to, but what im getting at is how do you know what non zero start and end time belong to what user |
| 20:24.10 | Vlat- | in other case the call is still in process or the things REALLY going wrong |
| 20:24.27 | Vlat- | it's even easier |
| 20:24.59 | Vlat- | there's the CallID field (don't know it's exact name in asterisk) |
| 20:25.23 | Vlat- | CallID is unique for each call |
| 20:25.45 | clyrrad | there is src or uniqueid if thats what you mean,but that is the one i was saying keeps loosing its information |
| 20:25.46 | Vlat- | So SELECT .... FROM ... WHERE EndTime NOT NULL GROUP BY CallID |
| 20:25.52 | Vlat- | it's a pseudo-code of course |
| 20:25.55 | clyrrad | perhaps its where im setting it.... where did you set these values? |
| 20:25.59 | Vlat- | sorry, but i must run out right now |
| 20:26.06 | Vlat- | will be back about 01:00 CET |
| 20:26.13 | clyrrad | okay, thanks |
| 20:27.33 | Ariel_ | hello shido6 are you around? need to talk with you please. |
| 20:32.54 | *** join/#asterisk _DAW (n=bob@adsl-150-43-153.msy.bellsouth.net) |
| 20:33.22 | _DAW | Can someone help me with an issue I am having with IAX and realtime? I am getting an error Auto-congesting call due to slow response |
| 20:33.54 | _DAW | everything worked fine before I tried it with realtime |
| 20:33.57 | *** join/#asterisk n0where (n=kc@dsl001-136-136.lax1.dsl.speakeasy.net) |
| 20:35.37 | jarrod | realtime is the bomb |
| 20:37.52 | *** join/#asterisk Damin_PDA (n=pocketir@52.sub-70-209-161.myvzw.com) |
| 20:38.30 | *** join/#asterisk IPmonger (n=ipmonger@pcp0010577106pcs.coatsv01.pa.comcast.net) |
| 20:38.40 | harryvv | Do anything to trade this gloom and doom gun metal grey skys. |
| 20:38.44 | jarrod | i like realtime cluster setup |
| 20:38.49 | fugitivo | mysql is evil |
| 20:38.52 | harryvv | what does realtime do |
| 20:38.58 | jarrod | i load everything in SQL |
| 20:39.03 | jarrod | it works great |
| 20:39.04 | harryvv | we are talking about rtp? |
| 20:39.05 | jarrod | as been for a long while |
| 20:39.09 | jarrod | no |
| 20:39.12 | fugitivo | sql is a bad idea for a stable system |
| 20:39.38 | jarrod | not at all |
| 20:39.46 | harryvv | jarros, no what |
| 20:39.51 | *** join/#asterisk bweschke (n=bweschke@dsl001-136-136.lax1.dsl.speakeasy.net) |
| 20:39.57 | jarrod | as long as you keep it local |
| 20:39.58 | Nugget | fugitivo: did you see http://devtoe.blogspot.com/2005/10/oracle-buys-innodb-will-fork-save.html ? |
| 20:40.01 | Nugget | it's brilliant. |
| 20:40.14 | Nugget | oracle is curb stomping mysql. |
| 20:40.26 | fugitivo | another service, another problem |
| 20:40.37 | Supaplex | <3 Pg |
| 20:40.44 | ender | not to mention that MySQL is an addon, not mainstream. |
| 20:40.48 | ender | another patch to maintain. |
| 20:41.07 | *** join/#asterisk MGSsancho (n=user@adsl-67-125-157-68.dsl.irvnca.pacbell.net) |
| 20:41.22 | fugitivo | flat files are perfect |
| 20:41.38 | fugitivo | Nugget: let's see :) |
| 20:41.41 | Nugget | the biggest issue for me is that the dialplan is code, not data, and doesn't belong in a database. |
| 20:41.51 | *** join/#asterisk supaigtr (n=yurplsl@152.53.16.10) |
| 20:41.59 | harryvv | has anyone worked with asterisk and those radio based cards for repeaters? |
| 20:42.04 | supaigtr | Hello. |
| 20:42.16 | fugitivo | xml config files would be nice |
| 20:42.40 | Damin_PDA | no...xml is bad bad bad... |
| 20:42.43 | Nugget | ew, not xml. |
| 20:42.49 | harryvv | nokia is going to be putting out a wimax phone in about two years. |
| 20:42.53 | Nugget | xml config files are a total hemorrhoid. |
| 20:43.01 | Nugget | they're unwieldy and difficult to work with |
| 20:43.07 | mcf3782 | I agree |
| 20:43.13 | fugitivo | why? |
| 20:43.17 | supaigtr | Anyone know how to debug IAX2 dropping audio in one direction? |
| 20:43.44 | fugitivo | it's easy to write a code to read a xml config file |
| 20:43.48 | Nugget | fugitivo: have you ever used an app which uses xml for config files? it's horrible. |
| 20:44.06 | Damin_PDA | supaightr stable to head? |
| 20:44.15 | ender | fugitivo: hand editing XML files is really shitty |
| 20:44.24 | Nugget | the files are difficult to edit by hand. I'm sure it would be ok if you never expected people to edit them without using a magic app. |
| 20:44.43 | ender | then you have to worry about your magic app doing the 'right thing'. |
| 20:44.46 | ender | which is a joke. |
| 20:44.50 | Nugget | it's hard to spot errors in xml, it's difficult to copy/paste blocks or edit with a powerful editor. |
| 20:45.02 | supaigtr | Two week old head. |
| 20:45.14 | ender | making files xml for the sake of making it easy for the program to read the files is rediculous. |
| 20:45.18 | fugitivo | yes, hand editing xml is a pain in the ass, but it's easy to write a program to read it and modify it |
| 20:45.26 | supaigtr | Damin_PDA: Its become a serious problem. PRI - IAX - IAX - Poly. |
| 20:45.27 | *** join/#asterisk terrapen (n=cjs@fw-01.satx.bikeworld.net) |
| 20:45.32 | Damin_PDA | head to head? |
| 20:45.47 | ender | fugitivo: then you have to rely on the assumptions the program will make when writing out your config. |
| 20:45.57 | supaigtr | I have two * boxes. One with PRI and the other fed by IAX. |
| 20:46.10 | terrapen | sigh... can anybody tell me why I see this: |
| 20:46.10 | terrapen | *CLI> Oct 12 10:54:18 NOTICE[1985]: chan_iax2.c:5468 socket_read: Rejected connect attempt from 208.139.204.232, request '2107646738@abcxyz' does not exist |
| 20:46.13 | ender | supaigtr: sounds like my setup. |
| 20:46.16 | Damin_PDA | both head |
| 20:46.27 | terrapen | I have my teliax entry in iax.conf set to context=inbound |
| 20:46.30 | supaigtr | Same version of head but its about 2 weeks old. |
| 20:46.40 | fugitivo | ender: that isn't a problem |
| 20:46.41 | terrapen | and there is an extension 2107646738 in contect inbound |
| 20:46.46 | Damin_PDA | identify potpal |
| 20:46.47 | terrapen | err context |
| 20:47.00 | Damin_PDA | timestamps enabled? |
| 20:47.20 | terrapen | I'm not sure why Asterisk is trying to put incoming calls into the guest context |
| 20:47.25 | ender | fugitivo: yes it is. IF you've had to deal w/ those types of things youd realize that too. THe beauty of Asterisk is that it is so open and configurable. If you ahve to rely on some program to configure it, value suddenly goes way down. |
| 20:47.37 | terrapen | when I've instructed it to put them into the "inbound" context |
| 20:47.38 | ender | terrapen: your iax isn't registering correctly. |
| 20:47.59 | terrapen | oh. |
| 20:47.59 | terrapen | <PROTECTED> |
| 20:47.59 | terrapen | Oct 12 10:54:08 NOTICE[1985]: iax2-provision.c:496 iax_provision_reload: No IAX provisioning configuration found, IAX provisioning disabled. |
| 20:48.06 | terrapen | maybe that has something to do with it? |
| 20:48.10 | ender | terrapen: thats different. |
| 20:48.18 | terrapen | <PROTECTED> |
| 20:48.23 | fugitivo | ender: i love plain text files, but i'm not thinking about me only |
| 20:48.25 | terrapen | looks like its registering... |
| 20:48.50 | terrapen | I wonder if it has to do with the fact that I have multiple installations of asterisk using Teliax from the same source IP (a NAT'ing router) |
| 20:49.01 | ender | terrapen: debug iax when making the call. |
| 20:49.42 | ender | fugitivo: *shrug* I've hated xml whenver I've ran against it. |
| 20:50.29 | terrapen | gnnn |
| 20:51.00 | mcf3782 | I'm sure XML has it's uses and benefits. But turning the Asterisk config files into XML is not, in my humble opinion; a good use of XML or a good thing for Asterisk on the whole. But just my opinion. |
| 20:51.04 | terrapen | ender: http://pastebin.com/391620 |
| 20:51.09 | supaigtr | Damin_PDA: I'm not familar with timestamps. |
| 20:51.20 | terrapen | strangest thing is, I *have* an 'inbound' context |
| 20:51.26 | supaigtr | It that part of the jitterbuffer? |
| 20:51.59 | terrapen | <PROTECTED> |
| 20:52.01 | terrapen | etc... |
| 20:52.05 | Damin_PDA | supaightr: well,disable them on both sides and disable plc + jitterbuffer...then retest.. |
| 20:52.07 | ManxPower | terracon, IAX Provisioning is for the IAXy hardware device. |
| 20:52.16 | terrapen | I have an IAXy |
| 20:52.28 | terrapen | maybe this is part of the problem? :) |
| 20:52.50 | mmlj4 | i'm going to get one if I can't make my sipura ATA connect to my house while away |
| 20:53.27 | supaigtr | Damin_PDA: I had jitterbuffer=no and got the same results. Not sure what plc and timestamps are. |
| 20:54.17 | Damin_PDA | supaight: disable them.. |
| 20:54.34 | Damin_PDA | .. |
| 20:54.40 | Damin_PDA | read docs... |
| 20:54.54 | ManxPower | terrapen, It should not cause an issue with Teliax |
| 20:54.55 | supaigtr | Ok. Where are docs for plc etc? |
| 20:55.05 | ManxPower | mmlj4, Looks like we won't be going to Gulfport on Friday |
| 20:55.14 | Damin_PDA | iax.conf machine... |
| 20:55.21 | ManxPower | supaigtr, as far as I can tell there are no docs for PLC |
| 20:55.23 | Damin_PDA | errr...file.. |
| 20:55.30 | *** join/#asterisk folsson (n=filip@lund-meje-sr0-vl101-249.perspektivbredband.net) |
| 20:55.35 | terrapen | Manx, these are a modified version of iax.conf/extensions.conf that I have been using with Teliax for quite awhile |
| 20:55.46 | supaigtr | ManxPower: Thats what I thought. Man this is really a problem. |
| 20:55.49 | terrapen | modified with new username/secret/host information of course |
| 20:56.22 | mcf3782 | Out of here for a while. Thanks again to those who helped with my outbound dialing issue earlier!! :) |
| 20:56.23 | *** join/#asterisk jets (n=jets@dsl001-136-136.lax1.dsl.speakeasy.net) |
| 20:56.35 | *** part/#asterisk mcf3782 (n=mcf3782@adsl-065-012-184-148.sip.asm.bellsouth.net) |
| 20:57.35 | *** join/#asterisk iCEBrkr (i=icebrkr@rrcs-24-129-130-158.se.biz.rr.com) |
| 20:57.37 | iCEBrkr | grrrrrrrr |
| 20:57.40 | iCEBrkr | Frick'n Roadrunner |
| 20:57.41 | supaigtr | Anyone else experienced the muting problem? It only seems to always be in one direction. |
| 20:58.11 | malverian[work] | Hahah... anyone think that I might get severe echo problems if my span for my PRI is set to not be used as primary sync source? :-P |
| 20:58.16 | terrapen | this suuuuuhhhuhhhhhuhhhhhhcks |
| 20:58.25 | X-Rob | malverian[work] - nope |
| 20:58.36 | malverian[work] | X-Rob, Are you serious? |
| 20:58.42 | X-Rob | that'll result in frame slips, and crackles |
| 20:58.43 | ManxPower | supaigtr, stop using CVS-HEAD/1.2beta then |
| 20:58.48 | X-Rob | won't have anything to do with echo |
| 20:59.01 | malverian[work] | X-Rob, Ah. So faxing might not work so well ;) |
| 20:59.08 | X-Rob | faxing won't work well at all |
| 20:59.10 | ManxPower | X-Rob, frameslips can confuse Asterisk's really finicky echocan code |
| 20:59.15 | supaigtr | ManxPower: Then I have lots of other problems. |
| 20:59.38 | supaigtr | I'll drive back and work on it some more. Thanks all. |
| 20:59.40 | *** join/#asterisk burton (i=mimx@w201.ljudmila.org) |
| 20:59.53 | malverian[work] | ManxPower, So in your opinion, it could affect echo. I have a feeling that's been my primary issue. I can't believe I had it turned off. |
| 21:00.12 | malverian[work] | ManxPower, I've had to use "Aggressive echo supression" to even have the calls be usable. Hopefully this will help. |
| 21:00.22 | X-Rob | malverian[work] - use KB1 echo can, echotraining=800 echocancelwhenbridged=yes echocancel=yes |
| 21:00.38 | X-Rob | you using E1 or T1? |
| 21:01.01 | *** join/#asterisk svenna (n=svenna@p548D26E5.dip0.t-ipconnect.de) |
| 21:02.09 | *** join/#asterisk Assid (n=assid@203.115.64.57) |
| 21:02.47 | ManxPower | ~mailinglist |
| 21:02.48 | jbot | well, mailinglist is Search Asterisk mailing lists by prepending site:lists.digium.com to your Google search. Browse the mailing list archive at http://lists.digium.com/ |
| 21:02.58 | terrapen | manx: |
| 21:02.58 | terrapen | *CLI> iax2 show registry |
| 21:02.58 | terrapen | Host Username Perceived Refresh State |
| 21:02.58 | terrapen | 208.139.204.232:4569 cnc 209.142.99.217:57229 60 Registered |
| 21:03.08 | terrapen | so it is apparently registering |
| 21:03.19 | terrapen | and I do have the correct context specified in iax.conf |
| 21:03.26 | jlewis | 17 |
| 21:03.29 | jlewis | oops |
| 21:03.31 | terrapen | and that context does exist in extensions.conf and is apparently proper |
| 21:05.15 | *** join/#asterisk LostOl (n=lostonli@wsip-68-15-227-143.om.om.cox.net) |
| 21:05.19 | malverian[work] | X-Rob, t1 |
| 21:05.53 | LostOl | I'm sorry to intrude, but would anyone be able to offer a pointer on how to get an analog phone to ring? (http://sourceforge.net/forum/forum.php?thread_id=1366790&forum_id=420324) |
| 21:06.22 | LostOl | I'm down to scanning .c files, which I think will only complicate my situation =\ |
| 21:06.57 | X-Rob | malverian[work] - are you using CVS HEAD? |
| 21:07.01 | LostOl | yup |
| 21:07.09 | LostOl | oops |
| 21:07.15 | X-Rob | LostOl - uh. You should be on #amportal |
| 21:07.19 | X-Rob | speak to you there. |
| 21:07.27 | LostOl | thanks X |
| 21:09.01 | clyrrad | does anyone know of a good doc that shows that parameters can be used in cdr.conf? |
| 21:09.04 | ManxPower | terrapen, You are using Teliax, right? |
| 21:09.09 | clyrrad | what* |
| 21:09.14 | terrapen | manx, yup |
| 21:09.22 | ManxPower | As I understand it, all DIDs under the same user ID will come in as that userID |
| 21:09.48 | ManxPower | i.e. if I have 5 DIDs in Teliax account ericfnords then I need iax.conf [ericfnords] section |
| 21:09.49 | terrapen | Manx, I'm wondering if this has something to do with using two seperate Asterisk servers (each connecting to their own Teliax account) from behind a signle NAT server |
| 21:10.03 | terrapen | ok |
| 21:10.06 | ManxPower | terrapen, We have done that and it has worked fine. |
| 21:10.07 | terrapen | lemme try that |
| 21:10.23 | ManxPower | the nat router will handle the port translations to make sure the outside device sees two connections |
| 21:10.25 | terrapen | manx, i have a [teliax] section in iax.conf on the other box and it works just fine |
| 21:10.36 | terrapen | (the other box uses teliax account name "bikeworld") |
| 21:10.55 | *** join/#asterisk zotz (n=zotz@24.231.36.100) |
| 21:11.31 | terrapen | ok, renaming the entry to our teliax username does not fix the problem :( |
| 21:13.52 | terrapen | a-ha |
| 21:13.54 | terrapen | interesting. |
| 21:14.09 | terrapen | when I remove the [guest] entry in iax.conf, I get the following message: |
| 21:14.15 | terrapen | Oct 12 11:23:34 NOTICE[1985]: chan_iax2.c:5749 socket_read: Host 208.139.204.232 failed to authenticate as 101 |
| 21:14.16 | *** join/#asterisk FuriousGeorge (n=brian@ool-44c5a9b8.dyn.optonline.net) |
| 21:14.19 | FuriousGeorge | hey all |
| 21:14.23 | terrapen | 101 is the next context that follows [teliax] |
| 21:14.45 | terrapen | err I mean entry, not context |
| 21:14.58 | terrapen | so there must be some kind of syntactical error in the [teliax] entry |
| 21:15.51 | FuriousGeorge | most voip providers say they allow "unlimited" inbound calling, but for pbx purposes are callers gonna get a busy signal if im on the line with an inbound call |
| 21:15.52 | *** join/#asterisk kFuQ (n=somedude@c-67-185-114-199.hsd1.wa.comcast.net) |
| 21:17.32 | terrapen | ok it works when I remove both the [guest] entry and the entry for the [iaxy] |
| 21:17.37 | terrapen | so, so strange |
| 21:18.58 | groogs | FuriousGeorge: 'unlimited' most likely refers to the # of minutes. # of concurrent calls is what you're asking about. most providers that do a flat per-month fee with "unlimited" anything probably only allow one concurrent call. |
| 21:20.11 | groogs | FuriousGeorge: look for someone that provides 'wholesale' or per-minute service.. basically, they give you a DID, charge you $2-5 a month, and 1 - 2 cents per min, for as many concurrent calls as you want |
| 21:20.55 | groogs | no voicemail, call waiting, etc (since 1. if you're using asterisk, you have all that anyways, and 2. call waiting is pointless when you can get concurrent calls) |
| 21:21.27 | FuriousGeorge | groogs: so as far as inbound calling goes, when you wanna get it closer to wholesale youre gonna be paying by the minute |
| 21:21.41 | groogs | yeah. but if you work it out ,its probably cheaper anyway |
| 21:21.47 | *** join/#asterisk supaigtr (n=yurplsl@152.53.17.1) |
| 21:21.52 | *** join/#asterisk wrmem (n=monnin@vpn82-7e-92-d6.near.uiuc.edu) |
| 21:22.04 | Damin_PDA | fixed? |
| 21:22.34 | supaigtr | :( jitt=off plc off timestamps off and I still got the same problem. |
| 21:22.43 | groogs | ie, 19.99 a month compared to $2.50/mo + $0.015/min means you need to make over 1166 minutes to get your moneys worth |
| 21:23.03 | *** join/#asterisk Rubble (n=netclass@206.165.75.199) |
| 21:23.07 | Assid | who gives .015/min? |
| 21:23.14 | Assid | only voipjet right? |
| 21:23.20 | groogs | nah, lots of places |
| 21:23.21 | FuriousGeorge | groogs: which is less than one hour of inbound calling per day |
| 21:23.21 | supaigtr | Damin_PDA: U using head and have 2 boxen with IAX ? |
| 21:23.23 | groogs | btw hi Assid |
| 21:23.35 | Damin_PDA | supa: back off to earlier cvs? try doing packet capture.. |
| 21:23.37 | Assid | oh... |
| 21:23.42 | Assid | hey dude.. how are you |
| 21:23.47 | *** join/#asterisk bjohnson (n=bjohnson@i216-58-62-82.cybersurf.com) |
| 21:23.51 | groogs | you are Assid from dal right? |
| 21:23.53 | Assid | when'd you end up on the * bandwagon? |
| 21:23.56 | Assid | yeah |
| 21:24.09 | supaigtr | Damin_PDA: What am I looking for on the capture? Things look normal to me and it happens intermittantly. |
| 21:24.13 | groogs | cool. i'm good. meh, been doing it for a while. our office has run on it since january |
| 21:24.24 | Assid | niiice |
| 21:24.29 | groogs | i'm one of the devs for AMP too |
| 21:24.42 | Assid | oh you are? |
| 21:24.45 | Assid | hrmm |
| 21:24.54 | Assid | didnt know that |
| 21:24.59 | Assid | whats amp again? |
| 21:25.10 | Assid | ~amp |
| 21:25.12 | jbot | from memory, amp is an Audio MPEG Player. [non-free], or http://amp.coalescentsystems.ca/ |
| 21:25.12 | groogs | gui interface |
| 21:25.24 | groogs | audio mpeg? ok then |
| 21:25.27 | X-Rob | heh |
| 21:25.31 | FuriousGeorge | groogs: the best solution for your "primary" inbound line may be a flat free unlimited, then you could have it failback to the per minute lines |
| 21:25.50 | groogs | FuriousGeorge: well, you also won't get concurrent calls etc with that |
| 21:25.51 | X-Rob | jbot: no, amp is a web based interface for configuring Asterisk. See http://amp.coalescentsystems.ca/ |
| 21:25.52 | jbot | okay, X-Rob |
| 21:25.56 | X-Rob | ~amp |
| 21:25.57 | jbot | amp is probably a web based interface for configuring Asterisk. See http://amp.coalescentsystems.ca/ |
| 21:25.57 | groogs | but i dunno, some providers may enable that for you |
| 21:26.07 | groogs | FuriousGeorge: you can get better rates than 1.5c/min too |
| 21:26.26 | X-Rob | groogs - You're Greg MacLellan? |
| 21:26.33 | groogs | yes |
| 21:26.43 | FuriousGeorge | groogs: yeah because it would fail back to the cheaper lines. iow, im saying the you can get an unlimeted inbound @ $8.00/mo or two, and the next in line for failback could be the per minute stuff your talking about |
| 21:26.46 | Assid | hes the guy i kick around in #php :P |
| 21:27.10 | X-Rob | groogs - aaaah. |
| 21:27.17 | syzygyBSD | does anyone know of a reason a T100P and a TE110P wouldn't work in the same system? Do i need updated drivers for the TE110P? |
| 21:27.17 | groogs | FuriousGeorge: oh yeah, $8/mo would be good. never seen those rates though |
| 21:27.19 | FuriousGeorge | since that last line would only be called when the first two are busy, so rarely |
| 21:27.50 | FuriousGeorge | broadvox, which brings me to my next question: anyone use broadvox for a did :) |
| 21:27.53 | Assid | okay time for yours truly to hit the sac |
| 21:28.08 | *** join/#asterisk bweschke (n=bweschke@dsl001-136-136.lax1.dsl.speakeasy.net) |
| 21:28.17 | *** join/#asterisk fordvoice (n=chrisf0r@cpe-69-133-21-43.cinci.res.rr.com) |
| 21:28.23 | groogs | FuriousGeorge: a lot of the voip things aren't really worth it though, just to use for 'unlimited inbound'. just get a POTS line. reliable, unlimited inbound, and fairly cheap (i think we pay $34cdn for business lines) |
| 21:29.04 | FuriousGeorge | groogs: we have one, i just dont want to get another, i want something for our pots line to failback to, besides another 25/mo pots line |
| 21:29.04 | Assid | isnt incoming on pots for 1 simultanous line? |
| 21:29.08 | tzafrir_laptop | which providers besides FWD and iaxtel use RSA keys? I'd like to add them to my package |
| 21:29.13 | Assid | unless you use multiple lines |
| 21:29.17 | groogs | FuriousGeorge: here, i have 3 POTS lines, and use voip only as backup and outgoing. my 800# and primary number goes to our first POTS line, which hunts to #2 and #3, and #3 hunts to our voip DID line |
| 21:29.34 | groogs | FuriousGeorge: if our internet goes down, it just means the 4th person calling gets a busy signal |
| 21:30.06 | Assid | id have it the other way around |
| 21:30.12 | groogs | FuriousGeorge: for outgoing local calls, it uses up to two POTS lines (keeping one free for inbound), and then rolls over to voip.. and for long distance, it uses voip. and of course, if the internet is down, it just falls back to POTS |
| 21:30.14 | Assid | first.. voip.. then pots |
| 21:30.25 | groogs | POTS is free to call outbound locally |
| 21:30.29 | Assid | oh? |
| 21:30.32 | Assid | hrmm |
| 21:30.47 | FuriousGeorge | groogs: im gonna send you a small msg if thats ok |
| 21:30.54 | groogs | sure |
| 21:31.08 | Assid | alrite catch you guys later |
| 21:31.12 | Assid | beddy bye time |
| 21:31.23 | groogs | bye Assid |
| 21:31.43 | malverian[work] | X-Rob, Yeah, I am using CVS. Though until just now it was from about 10 days ago. |
| 21:31.53 | malverian[work] | X-Rob, Sorry for the slow response, had a meeting to attend. |
| 21:31.55 | *** join/#asterisk Poincare (n=jefffnod@dD5779BD2.access.telenet.be) |
| 21:32.25 | _DAW | hrmmm.. has anyone else gotten Auto-congesting call due to slow response when dialing out of IAX in 1.2.0 beta 1? |
| 21:32.27 | *** join/#asterisk kFuQ (n=somedude@c-67-185-114-199.hsd1.wa.comcast.net) |
| 21:32.31 | X-Rob | malverian[work] - use KB1 echo can, echotraining=800 echocancelwhenbridged=yes echocancel=yes |
| 21:32.37 | _DAW | I thought it was realtime, but it happens in my flatfiles as well |
| 21:32.48 | X-Rob | if you still have echo, try setting 'txgain=-1.0' in zaptel.conf |
| 21:32.48 | PupenoL | Have any of you experience random volume problems on zap incomming calls ? some calls are almost in-audible. |
| 21:33.05 | tzafrir_laptop | so far only other provider I found that uses RSA keys: http://www.junctionnetworks.com/Asterisk-config.htm |
| 21:33.34 | malverian[work] | X-Rob, Roger. I used a miliwatt line to get my gains set up correctly. |
| 21:33.43 | malverian[work] | But i may have to do it again now that I've changed my sync source. |
| 21:33.45 | malverian[work] | (who knows) |
| 21:34.42 | X-Rob | malverian[work] - didn't you say you're on a PRI? |
| 21:34.50 | malverian[work] | X-Rob, Yes. |
| 21:34.51 | *** join/#asterisk pattieja (n=pattieja@adsl-69-153-174-41.dsl.stlsmo.swbell.net) |
| 21:34.56 | X-Rob | txgain=0, rxgain=0 |
| 21:35.10 | X-Rob | miliwatt testing is for analog POTS, not digital |
| 21:35.19 | malverian[work] | X-Rob, I'm using 0.2 and 0.205 |
| 21:35.23 | malverian[work] | X-Rob, But okay, I'll try that. |
| 21:35.25 | X-Rob | ah ok |
| 21:35.25 | *** join/#asterisk E-Lore (n=not@dslb-084-058-050-131.pools.arcor-ip.net) |
| 21:35.30 | X-Rob | 0.2 is bugger all |
| 21:35.35 | E-Lore | good evening! |
| 21:36.17 | malverian[work] | X-Rob, Unless this sync-source thing is going to make major help with my echo training, I'll probably have to turn echo supression back on (aggressive mark2) |
| 21:37.08 | malverian[work] | X-Rob, Also, I'm using a ~80ft cat5 cable from my CSU to my WCTE110P, so hopefully that's not causing problems. |
| 21:37.18 | *** join/#asterisk addi (n=none@c-67-166-96-35.hsd1.ut.comcast.net) |
| 21:37.31 | X-Rob | malverian[work] - have you checked those settings I gave you? |
| 21:37.32 | malverian[work] | X-Rob, And our CO told us that we have 0db, but the guy may not have known what he was talking about. |
| 21:37.34 | X-Rob | malverian[work] - use KB1 echo can, echotraining=800 echocancelwhenbridged=yes echocancel=yes |
| 21:37.45 | malverian[work] | X-Rob, Yes, I am using that. I've used that before and it didn't help. |
| 21:37.56 | X-Rob | malverian[work] - that's correct. You're digital. You don't have any audio signal gain/loss |
| 21:38.02 | E-Lore | ok...seems like I have lots to learn :) |
| 21:38.11 | malverian[work] | X-Rob, I was using that (it's basically the default except for echotraining=400 instead of 800) |
| 21:38.24 | X-Rob | 400 is the default |
| 21:38.28 | malverian[work] | X-Rob, But the KB1 wan't cutting it.. we still had severe echo (we're doing SIP clients for most of our phones) |
| 21:38.32 | X-Rob | I found 800 was better for international calls |
| 21:38.34 | malverian[work] | X-Rob, Right, read my sentence, that's what I said :-P |
| 21:38.51 | malverian[work] | X-Rob, My wording was odd I suppose. |
| 21:38.56 | X-Rob | echocancelwhenbridged=no is the default |
| 21:39.05 | malverian[work] | X-Rob, True, but I had it set to yes. |
| 21:39.12 | addi | anybody know of a good ata that is based on the PA1688 chip? I have echo with atcom's ata. |
| 21:39.26 | malverian[work] | X-Rob, I was using exactly what you said originally, except I was using echotraining=yes (400) instead of 800 |
| 21:39.27 | X-Rob | ok - are you using GXP2000's for your phones? |
| 21:39.31 | malverian[work] | X-Rob, But we had terrible echo. |
| 21:39.36 | malverian[work] | X-Rob, No, SNOM 320 |
| 21:39.40 | X-Rob | shit |
| 21:39.42 | malverian[work] | X-Rob, And a few soft phones. |
| 21:39.50 | X-Rob | never had echo issues with 320's |
| 21:40.00 | X-Rob | GXP's and PA1688's have self-generated echo issues |
| 21:40.06 | *** join/#asterisk yogurt2ungue (n=yogurt2u@44-170-114-200.fibertel.com.ar) |
| 21:40.15 | X-Rob | hrmmmmmmmm. |
| 21:40.24 | malverian[work] | X-Rob, Yes, but like I said, my sync span was set to span=1,0,0 instead of 1,1,0 (I didn't have any span set to be sync source) |
| 21:40.32 | X-Rob | well I truly don't know |
| 21:40.38 | malverian[work] | X-Rob, So if what ManxPower says is true, that could have screwed up my echo training. |
| 21:40.45 | X-Rob | I doubt your timing would be causing 'significant' echo issues |
| 21:40.57 | malverian[work] | X-Rob, If that doesn't fix it though, I'm going to have to turn echo supression back on. |
| 21:41.00 | E-Lore | I have a very very basic question..I searched the web for some time now and I'm still missing a proper List of ISDN adapters with hfc-s chipsets |
| 21:41.08 | E-Lore | is there something you know of? |
| 21:41.13 | X-Rob | malverian[work] - your users will hate that |
| 21:41.31 | malverian[work] | X-Rob, I know. But if it's the only option... |
| 21:41.44 | malverian[work] | X-Rob, It's also possible we aren't really 0db |
| 21:41.56 | malverian[work] | X-Rob, Maybe our switch is set at -7 or -15 or something. |
| 21:41.58 | *** join/#asterisk Ariel_ (n=Ariel@dsl-20-177.cofs.net) |
| 21:41.59 | X-Rob | If you still have issues - try turning down txgain |
| 21:42.22 | *** join/#asterisk Ayano (n=Ayano_@adsl-70-245-190-90.dsl.spfdmo.swbell.net) |
| 21:42.25 | malverian[work] | X-Rob, Also.. we can cut off some excess from the cat5 cable.. so maybe that will clean up any potential attenuation issues. |
| 21:42.42 | malverian[work] | X-Rob, We can probably clear off about 30ft of it ;) |
| 21:43.03 | X-Rob | malverian[work] - feel free, but it won't do anything. |
| 21:43.07 | Ayano | are there any codecs that will give you about 32k or less a call? |
| 21:43.15 | X-Rob | Ayano - ilbc I think? |
| 21:43.37 | ian_k | g729 |
| 21:44.01 | malverian[work] | X-Rob, Do you use shielded cable for your line from the switch to your server? |
| 21:44.29 | X-Rob | malverian[work] - nope. Standard cat5 |
| 21:44.33 | Ayano | ian_k; is g729 the one that you have to have a license for or does it just work out of the box on asterisk? |
| 21:44.36 | *** join/#asterisk RoyK (n=roy@55.80-202-161.nextgentel.com) |
| 21:44.47 | *** join/#asterisk pigpen (n=mark@fw.seamans.cc) |
| 21:44.50 | malverian[work] | X-Rob, Well we'll see how things go tonight. Maybe these settings will work decently. |
| 21:46.00 | tzafrir_laptop | I see that in apps/Makefile app_sql_postgresql.so is remmed-out and listed under obsolete. What replaces it? |
| 21:46.01 | ian_k | Ayano: you nede a license for it.. $10/channel |
| 21:46.09 | E-Lore | umm..if you there isnt any such list just tell me so... |
| 21:46.23 | tzafrir_laptop | Anything that retains the same syntax? |
| 21:46.30 | Ayano | how big is g711? |
| 21:46.36 | jarrod | 64k wi overhead ~80 |
| 21:46.44 | ian_k | 711 = 96k |
| 21:46.50 | Ayano | wow. |
| 21:47.16 | Ayano | Does anyone hapen do know what the default on AAH is? |
| 21:47.48 | tzafrir_laptop | Ayano, it's quite easy to see when you make a call |
| 21:48.05 | tzafrir_laptop | try 'sip show channels' and such |
| 21:48.41 | Ayano | how would I show the codec that a specific phone is using? |
| 21:49.11 | tzafrir_laptop | The name of the channel has the name of the peer as a prefix |
| 21:49.26 | ian_k | Ayano - you have to initate a call first.. |
| 21:49.35 | Ayano | k thanks |
| 21:49.59 | ian_k | Ayano - channels can have a list of many that they could use, but one isn't chosen until it negotiates with the other end |
| 21:50.19 | E-Lore | are there any good manuals / faqs out there that might help to get me started? |
| 21:50.40 | *** join/#asterisk gniretar (n=gniretar@66-227-204-109.dhcp.bycy.mi.charter.com) |
| 21:51.03 | tzafrir_laptop | ~voip-info |
| 21:51.04 | jbot | voip-info is probably the Voice Over IP wiki. It is a community resource which will answer all of your questions, from Asterisk to ZTDummy. You can find it over at http://www.voip-info.org - well worth bookmarking |
| 21:51.07 | tzafrir_laptop | ~docs |
| 21:51.09 | 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 |
| 21:51.24 | E-Lore | thats one of the pages I found..the voip wiki |
| 21:51.37 | gniretar | voip wikki kicks major a$$ |
| 21:51.41 | Ayano | ulaw is g711? |
| 21:51.53 | E-Lore | ok..so at least I got that one right :) |
| 21:52.18 | tzafrir_laptop | Ayano, yes |
| 21:52.23 | E-Lore | I just need the adapters then...ebay it is..and hopefully some good pictures of the chips... |
| 21:52.25 | E-Lore | thank you! |
| 21:52.42 | *** join/#asterisk viLeR (i=1000@66.128.47.232) |
| 21:53.06 | X-Rob | Ayano - ulaw is g711u, alaw is g711a |
| 21:53.17 | X-Rob | note that the entire world uses g711a, except for (as usual) the US. |
| 21:53.18 | AsteriskNoob | x-rob which do you like best? |
| 21:53.32 | Ayano | so ulaw g711u is about 96k? |
| 21:53.40 | X-Rob | there's no real difference, use whatever your country uses. |
| 21:53.54 | AsteriskNoob | ISDN switches are all ULAW arent they? even on E1? |
| 21:54.06 | AsteriskNoob | you're saying that E1's and such use A? |
| 21:54.06 | X-Rob | No. |
| 21:54.11 | X-Rob | ISDN is G711u or a. |
| 21:54.36 | X-Rob | If you're in 'the world' you use alaw. If you're in the us you use G711u ulaw |
| 21:54.37 | AsteriskNoob | ok but H.263 is universal right? of course at that point its just data |
| 21:55.33 | *** part/#asterisk Ayano (n=Ayano_@adsl-70-245-190-90.dsl.spfdmo.swbell.net) |
| 21:56.23 | FuriousGeorge | how reliable is asterisk fax detection from a voip did (ulaw) |
| 21:59.40 | Damin_PDA | identify potpal |
| 21:59.40 | Damin_PDA | not |
| 22:00.03 | shido6 | Ive had good luck |
| 22:00.05 | shido6 | with it |
| 22:01.25 | *** join/#asterisk Nukemizer (n=Nuke@67.137.28.163) |
| 22:02.08 | FuriousGeorge | shido6: with fax detection? |
| 22:04.46 | shido6 | yes |
| 22:05.00 | *** part/#asterisk wrmem (n=monnin@vpn82-7e-92-d6.near.uiuc.edu) |
| 22:05.05 | shido6 | worst I had was slowing down to 9600 and resending pages |
| 22:05.09 | shido6 | but we're fixing that. |
| 22:05.17 | FuriousGeorge | cool |
| 22:05.51 | shido6 | stay tuned |
| 22:06.07 | *** join/#asterisk A-Tuin (n=a-tuin@steves.ip.v4.me.uk) |
| 22:08.14 | FuriousGeorge | still working on getting it detected, for whatever reason the "fax" priority in extensions.conf combined with faxdetection=incoming in zapata.conf isnt doing it for me |
| 22:08.42 | FuriousGeorge | oh yeah, and i answer and wait in the s for incoming contaxt, but nothing |
| 22:11.47 | X-Rob | FuriousGeorge - doesn't that give you a hint? |
| 22:11.54 | X-Rob | which file are you putting fax detection in? |
| 22:12.02 | X-Rob | what _configuration_ file... |
| 22:12.13 | FuriousGeorge | X-Rob: well the faxes come in over pots |
| 22:12.29 | X-Rob | you said it was over voip. |
| 22:12.52 | X-Rob | shido6 - T38's going into openpbx. |
| 22:13.03 | FuriousGeorge | X-Rob: im thinking about voip |
| 22:13.12 | FuriousGeorge | X-Rob: b/c im using pots :) |
| 22:13.18 | *** join/#asterisk marc324 (n=marc3234@206-248-159-56.dsl.teksavvy.com) |
| 22:13.23 | X-Rob | my point being that you can't do fax detection on sip or iax. |
| 22:13.38 | X-Rob | only on zaptel |
| 22:15.17 | FuriousGeorge | X-Rob: really, thats good to know |
| 22:15.38 | *** join/#asterisk wolfson (n=hehe@usr-kdh-208-6-58-26.beachlink.com) |
| 22:16.00 | *** join/#asterisk _native_ (n=intuit@cpe-66-87-4-181.ut.sprintbbd.net) |
| 22:16.22 | groogs | why would you need fax detection on voip? just use a DID |
| 22:16.40 | FuriousGeorge | X-Rob: but, it should be working with faxdetect=incoming, and the fax priority in the appropriate context |
| 22:16.59 | X-Rob | FuriousGeorge - what file is the faxdetection=incoming set up in? |
| 22:17.18 | FuriousGeorge | <PROTECTED> |
| 22:17.25 | X-Rob | yup. |
| 22:17.41 | X-Rob | is it in sip.conf or iax.conf? |
| 22:17.45 | FuriousGeorge | so it /should/ be working, right? |
| 22:17.46 | X-Rob | (hint: no) |
| 22:17.46 | *** join/#asterisk nagl (n=nagl@213.235.241.6) |
| 22:17.55 | FuriousGeorge | X-Rob: no, but you just said i couldnt do that anyway |
| 22:18.23 | X-Rob | IF you have a fax call coming in on a zap channel, and you have faxdetect=incoming turned on, and you do an answer, wait(2) then it will jump to the 'fax' extension |
| 22:18.31 | X-Rob | if it detects a fax |
| 22:19.18 | FuriousGeorge | fax extension priority 1 right |
| 22:20.14 | X-Rob | Uh yes. |
| 22:20.18 | X-Rob | [default] |
| 22:20.25 | X-Rob | exten => s,1,Answer |
| 22:20.30 | X-Rob | exten => s,2,Wait(2) |
| 22:20.34 | FuriousGeorge | cuz priority one on s is answer, then wait(2), then dial(3) but faxdetect is on so it should go to fax,1, if it detects a fax |
| 22:20.37 | X-Rob | exten => s,3,Gosomewhereelse |
| 22:20.47 | X-Rob | exten => fax,1,RxFax |
| 22:21.15 | FuriousGeorge | X-Rob: so we've ruled out human error and we can file a bug report? |
| 22:21.24 | X-Rob | what's not working? |
| 22:21.41 | FuriousGeorge | sorry, exten fax,1,hangup isnt working |
| 22:21.50 | FuriousGeorge | it still does s,3,dial |
| 22:21.59 | *** part/#asterisk _native_ (n=intuit@cpe-66-87-4-181.ut.sprintbbd.net) |
| 22:22.27 | FuriousGeorge | im testing my fax detection before i do anything serious with is so im sending faxes to asterisk to see it ill will detect and hangup, or call my users |
| 22:22.30 | FuriousGeorge | see what i mean |
| 22:22.55 | X-Rob | Fax detection works everywhere I've set it up |
| 22:23.05 | *** join/#asterisk Sedorox (n=Brandon@smartserv/cna/Sedorox) |
| 22:23.26 | X-Rob | My money is that you've stuffed up your zapata.conf |
| 22:23.28 | FuriousGeorge | i restarted zaptel service and asterisk too |
| 22:23.58 | X-Rob | ~pb |
| 22:23.59 | jbot | from memory, pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca/ |
| 22:25.54 | FuriousGeorge | X-Rob: http://pastebin.ca/25347 |
| 22:26.09 | FuriousGeorge | hoping ur right |
| 22:26.41 | X-Rob | Hrm 8-( |
| 22:26.48 | ender | has anybody configured IP-501 phones w/ more than one SIP account? I want to dedicate 2 buttons to one extension, and one button to another. |
| 22:26.53 | ender | so far, only the first two are showing up. |
| 22:27.21 | X-Rob | Your spacing is counter-intuative |
| 22:27.41 | X-Rob | you've got group=0 underneath channel=4, but it refers to channels 1 and 2 |
| 22:27.51 | X-Rob | but apart from that, it's ok |
| 22:28.11 | X-Rob | and you can trim most of it down |
| 22:29.24 | FuriousGeorge | X-Rob: yeah, i rmad the tdm and got it back with the modules switched |
| 22:29.38 | X-Rob | http://pastebin.ca/25350 |
| 22:29.41 | X-Rob | that was me tidying it up |
| 22:29.43 | FuriousGeorge | http://pastebin.ca/25349 that one includes the part of extensions.conf, but i cant see how anythings wrong with that |
| 22:30.38 | X-Rob | Dunno |
| 22:30.43 | X-Rob | that looks fine |
| 22:31.15 | FuriousGeorge | well at least my zapata.conf is cleaned up |
| 22:31.55 | FuriousGeorge | X-Rob: what would you do if you were me and didnt know from debugging? mailing list? |
| 22:33.32 | X-Rob | I'd be listening to the call, making sure hte fax is sending the correct tones, puting a noop in place of the hangup, looking at /var/log/asterisk/full etc |
| 22:34.18 | FuriousGeorge | ill see how that does me |
| 22:34.41 | FuriousGeorge | the fax must be sending the correct tones though, it is the one we use all the time, its outside of * right now |
| 22:34.50 | FuriousGeorge | ill check the other things |
| 22:35.04 | FuriousGeorge | (im not in the building right now) |
| 22:37.41 | ender | does anybody have experience w/ IP-501 phones and multiple SIP extensions on them? |
| 22:39.13 | FuriousGeorge | X-Rob: err, i have no /var/log/asterisk/full |
| 22:39.33 | X-Rob | edit /etc/asteisk/logger.conf then |
| 22:40.13 | *** join/#asterisk bweschke (n=bweschke@dsl001-136-136.lax1.dsl.speakeasy.net) |
| 22:40.31 | supaigtr | What are the steps needed to roll back off head to stable? |
| 22:41.35 | FuriousGeorge | emerge asterisk for me |
| 22:47.04 | malverian[work] | Hmm.. if I have a PRI (18 channels) and then two TDM cards (quad span) should I set up a separate span for each of the cards? And if so.. what kind of LBO do you use for a standard analog line? |
| 22:47.05 | *** join/#asterisk darkskiez (n=darkskie@host86-138-169-183.range86-138.btcentralplus.com) |
| 22:47.28 | *** part/#asterisk Uther_P (n=uther_p@66.180.120.82) |
| 22:47.42 | X-Rob | supaigtr - uh. download stable, install it? |
| 22:48.01 | X-Rob | and it's not CALLED stable |
| 22:48.05 | X-Rob | it's called 1.0.9 |
| 22:48.12 | *** join/#asterisk pa (n=Paolo@unaffiliated/pa) |
| 22:48.47 | tzafrir_laptop | FuriousGeorge, have a look at logger.conf . Optionally 'logger reload' |
| 22:48.49 | Nugget | sure, as long as the "it" you mean is something else. |
| 22:48.56 | Nugget | there *is* a stable branch in cvs. |
| 22:48.58 | Vco | god not this "STABLE" bullshit again |
| 22:49.00 | malverian[work] | Nevermind. Apparently you don't. |
| 22:49.21 | tzafrir_laptop | s/STABLE/v1-0/ |
| 22:49.37 | tzafrir_laptop | so we won't argue over sematics |
| 22:50.25 | tzafrir_laptop | Debian is the best distro |
| 22:50.30 | Nugget | Linux is poo. |
| 22:50.38 | *** join/#asterisk epablo (n=epablo@WLL-24-pppoe197.t-net.net.ve) |
| 22:50.45 | supaigtr | X-Rob: checkout v1.0 is correct? |
| 22:50.52 | Nugget | supaigtr: no |
| 22:51.00 | Nugget | cvs checkout -r v1-0 asterisk |
| 22:51.12 | marc324 | how do you force asterisk realtime to use the db instead of the configuration files? |
| 22:51.20 | generalhan | anyone in here worked a lot with the FOP ??? |
| 22:51.24 | supaigtr | Yea. Thats it. So no del mods or anything it'll do it itself right? |
| 22:51.35 | tzafrir_laptop | generalhan, what's your problem? |
| 22:51.50 | malverian[work] | I almost answered yes... thinking "fop" the formatted object parser ;) |
| 22:51.54 | generalhan | not really a problem, just a question ... |
| 22:52.19 | generalhan | is there a way that i can make the FOP show me how many people are in a specific que .. rather who they are. |
| 22:52.38 | malverian[work] | Is there any way for a SIP phone to know that the asterisk server is being restarted? |
| 22:52.40 | tzafrir_laptop | well, I never worked with queues... |
| 22:52.52 | generalhan | as it is riught now it will tell me there are 5 people in the queue, but i want that button to show in 5 places with everyone that is on it |
| 22:53.02 | generalhan | ok lemme ask you onw more question then... |
| 22:53.10 | supaigtr | Nugget: Does v1-0 show up as CVS HEAD for version? |
| 22:53.14 | Nugget | no |
| 22:53.27 | supaigtr | I'm doing something wrong then. |
| 22:53.34 | Nugget | just delete your tree and check it out clean |
| 22:53.42 | FuriousGeorge | tzafrir_laptop: logging enabled |
| 22:53.45 | generalhan | is there a way to set up multiple FOPs to show different things, like i can only fit 100 buttons on there but if i have 100 sales people i want to be able to just pull up sales, or just pull up admin ... |
| 22:54.10 | supaigtr | I did, created a directory asteriskstable and did cvs checkout -r v1-0 asterisk zaptel libpri |
| 22:54.42 | Nugget | ok. sounds good. |
| 22:54.43 | tzafrir_laptop | FuriousGeorge, it's not just "enables/disabled". by default there is no writing to "full" (that line is remmed-out in the sample config) |
| 22:55.07 | generalhan | i want to go to my webserver on my linux box and from the homepage be able to click on "sales", "receptionists" "admin" and have it bring up a seperate FOP for each of those. can i get that done without having to run 3 different perl scripts ? |
| 22:55.34 | supaigtr | Nugget: Hmm Slackware up 2.6 from 2.4 breaks zaptel makefile. |
| 22:55.34 | Nugget | generalhan: yes, the latest versions of the flash operator panel can do that. |
| 22:56.00 | tzafrir_laptop | generalhan, I believe there is something about "doamins", but I'm not sure |
| 22:56.04 | generalhan | now is this the stand alone FOP like through asternic .. or the FOP through AMP ? |
| 22:56.15 | FuriousGeorge | tzafrir_laptop: yeah, i uncomented it and now i have the file |
| 22:56.15 | Nugget | I'm speaking of the asternic one. |
| 22:56.19 | generalhan | really |
| 22:56.24 | FuriousGeorge | it still doesnt tell me anything about the fax though |
| 22:56.26 | Nugget | I've never even seen AMP. |
| 22:56.32 | generalhan | i think i have the newest release i should find out how to do it. |
| 22:57.05 | generalhan | so that like instead of going to 192.168.0.2/html i could have 192.168.0.2/sales and 192.168.0.46/admin ?? |
| 22:57.19 | Nugget | I don't know how the urls work out. |
| 22:57.24 | Nugget | I suggest you experiment and see. :) |
| 22:57.35 | generalhan | ok thanks ... ill see if i can find some documentation on it ! |
| 22:57.48 | groogs | generalhan: if you can bind op_server to specific IPs, you should be able to create multiple instances of it |
| 22:57.54 | tzafrir_laptop | groogs, I actually want to see it implemented in an instant messanger |
| 22:58.20 | tzafrir_laptop | A web browser is not the right application for ir |
| 22:58.27 | tzafrir_laptop | for it |
| 22:58.28 | groogs | yeah it would be neat if there was a jabber interface to it somehow... |
| 22:58.48 | X-Rob | jabber would be easy |
| 22:58.55 | generalhan | well thats just it, i know that i can manipulate op_server to go to a specific site, but in that instance i would have to run 3 different perl scripts for each one i create |
| 22:58.57 | groogs | actually.. |
| 22:59.01 | groogs | thats actually a very good idea |
| 22:59.04 | X-Rob | Cost you US$500 and I could do it on the weekend if you want. |
| 22:59.32 | groogs | tzafrir_laptop: issue would be transfering calls.. |
| 22:59.45 | groogs | though, personally i dont really use that feature much anyway |
| 23:00.03 | groogs | in fact, i think i tested it once and haven't used it since |
| 23:00.36 | groogs | X-Rob: how about you write one and donate it to AMP :p |
| 23:00.53 | *** join/#asterisk sd-tux (i=sd@2001:4ca0:0:fe00:0:0:a96:3f18) |
| 23:01.12 | clyrrad | When using NoCDR, where should it be placed? Directly above the Hangup? |
| 23:01.18 | X-Rob | groogs - possibly. I'm more interested in openpbx at the moment tho. |
| 23:01.30 | X-Rob | they're doing funky stuff with replacing extension.conf |
| 23:01.41 | X-Rob | using sqlite |
| 23:01.43 | groogs | yeah thats why i suggested jabber :p |
| 23:01.46 | FuriousGeorge | X-Rob: i got the full log going but it makes no mention of "fax" anywhere on incoming calls |
| 23:01.47 | groogs | i've been following |
| 23:02.07 | X-Rob | FuriousGeorge - and you're definately bringing these fax calls in via the tdm400 card, right? |
| 23:02.30 | FuriousGeorge | X-Rob: absolutely, its the only fxo i have |
| 23:02.46 | X-Rob | then faxdetect isn't working |
| 23:02.51 | FuriousGeorge | lol |
| 23:02.52 | FuriousGeorge | really |
| 23:02.53 | X-Rob | 8) |
| 23:03.01 | X-Rob | you using 1.2? |
| 23:03.05 | FuriousGeorge | 1.0.9 |
| 23:03.14 | X-Rob | Hurm. Try 1.2, can't hurt. |
| 23:03.23 | clyrrad | Anyone know where to use NoCDR()? Does it go directly above the Hangup or should it be somewhere towards the top of the incomming call? |
| 23:03.27 | X-Rob | (uh, it can, if you're using ResetCDR/NoCDR) |
| 23:03.35 | X-Rob | it's buggy and causes crashes in 1.2 |
| 23:03.54 | FuriousGeorge | X-Rob: u shure it cant hurt, this is a business i wouldnt want unexpected wierdness |
| 23:04.03 | clyrrad | Ah... I am not using ResetCDR, I did not know i had to use both to make it work |
| 23:05.46 | FuriousGeorge | X-Rob: was fax detection changed/improved in 1.2 or something. id rather not use a beta "for real" anyway |
| 23:06.05 | clyrrad | X-Rob, strange even with both of those lines the CDR is getting written |
| 23:06.06 | X-Rob | I use 1.2 in production everywhere |
| 23:06.23 | FuriousGeorge | yeah, but you know from debugging |
| 23:06.31 | clyrrad | exten => s,13,ResetCDR() |
| 23:06.31 | clyrrad | exten => s,14,NoCDR() |
| 23:06.32 | clyrrad | exten => s,15,Hangup |
| 23:06.35 | clyrrad | any ideas? |
| 23:06.50 | X-Rob | clyrrad - Try putting it in the h extension |
| 23:06.57 | X-Rob | s,13,Hangup |
| 23:07.03 | X-Rob | h,1,NoCDR() |
| 23:07.07 | X-Rob | h,2,ResetCDR() |
| 23:07.26 | clyrrad | NoCDR goes before the ResetCDR ? |
| 23:07.50 | X-Rob | if you don't want to save it |
| 23:08.03 | clyrrad | ok trying that now |
| 23:09.56 | clyrrad | hrmmm its still writing it |
| 23:10.17 | shido6 | where does centos hide its iptables/firewall data |
| 23:10.19 | shido6 | ? |
| 23:11.06 | *** join/#asterisk RoyK (n=roy@55.80-202-161.nextgentel.com) |
| 23:12.15 | *** part/#asterisk darkskiez (n=darkskie@host86-138-169-183.range86-138.btcentralplus.com) |
| 23:20.33 | *** join/#asterisk kingtux (n=kingtux@pool-151-196-44-240.balt.east.verizon.net) |
| 23:20.54 | kingtux | What the correct way to fully remove asterisk from my system |
| 23:20.56 | kingtux | ?? |
| 23:21.24 | tzafrir_laptop | kingtux, apt-get remove --purge asterisk zaptel libpri |
| 23:21.42 | SwK[Work] | hey where is blitzrage |
| 23:22.20 | kingtux | i didn't install binaries |
| 23:22.27 | kingtux | i build from source |
| 23:23.05 | tzafrir_laptop | did you actually install anything? /usr/lib/asterisk/modules ? /var/lib/asterisk ? /var/log/asterisk ? |
| 23:23.26 | kingtux | yes |
| 23:23.33 | tzafrir_laptop | And of course: /etc/asterisk |
| 23:23.43 | kingtux | yup |
| 23:24.11 | syzygyBSD | lol |
| 23:24.27 | syzygyBSD | doesn't iomega have something called the lifedrive? |
| 23:24.41 | syzygyBSD | doh, palm... |
| 23:24.44 | RoyK | syzygyBSD: see /. |
| 23:24.55 | generalhan | i know this isnt exactly asterisk related ... but: can some one tell me how to give my op_panel_user write permissions to the var/www/html directory ? |
| 23:25.04 | syzygyBSD | oh.. i saw it 5 hours ago when it was first on there... |
| 23:25.04 | tzafrir_laptop | Is it just me, or generating the docs takes longer than the actual build of Asterisk? |
| 23:25.21 | generalhan | i know its some chmod command but im not real familar with the process |
| 23:25.24 | RoyK | tzafrir_laptop: docs? |
| 23:25.35 | syzygyBSD | generalhan: who is the owner of the files? |
| 23:25.41 | tzafrir_laptop | api documentation with doxygen |
| 23:25.49 | generalhan | root |
| 23:25.50 | syzygyBSD | does anyone else have to write to them? |
| 23:25.52 | kingtux | So how do i uninstall astesisk?? |
| 23:25.54 | generalhan | nope |
| 23:26.01 | RoyK | tzafrir_laptop: as in domestic ornitoric calculated lists? |
| 23:26.04 | shido6 | brb |
| 23:26.06 | shido6 | food |
| 23:26.12 | generalhan | its just so i can run the perl script for my FOP |
| 23:26.30 | syzygyBSD | chown -R op_panel_user /var/www/html |
| 23:26.49 | RoyK | userdel -r syzygyBSD |
| 23:26.58 | tzafrir_laptop | kingtux, rm -rf /the/unnedded/directories is known to improve the free disk-space count |
| 23:27.05 | syzygyBSD | uh.. recursive userdel? |
| 23:27.41 | tzafrir_laptop | syzygyBSD, op-panel only needs to be able to write to variables.txt in that directory |
| 23:27.44 | RoyK | kingtux: uninstalling asterisk is rarely nessecary |
| 23:27.53 | tzafrir_laptop | You only need to chown that file |
| 23:28.06 | *** join/#asterisk Greyh0und (n=Grey@85.11.48.49) |
| 23:28.12 | RoyK | kingtux: you don't have the windows' growing registry syndrom in unices |
| 23:28.25 | Greyh0und | What do i need to get .call-files to work? |
| 23:28.28 | RoyK | kingtux: just don't start it |
| 23:28.36 | kingtux | well i'm trying to start scratch |
| 23:28.42 | kingtux | start from scratch |
| 23:28.42 | RoyK | Greyh0und: copy them to the right place for a start |
| 23:28.53 | Greyh0und | well they shall be |
| 23:28.54 | kingtux | and I want to remove everything and start from new |
| 23:28.56 | syzygyBSD | tzafrir_laptop: oh, i didn't know what he was really trying to do, just what he asked for.. tried to answer that question.. |
| 23:29.13 | *** join/#asterisk syle2 (n=blag@unaffiliated/syle) |
| 23:29.26 | Greyh0und | RoyK: i got them in "/var/spool/asterisk/outgoing" and i think that is correct |
| 23:29.51 | RoyK | kingtux: just reinstall. remove /usr/lib/asterisk/modeules/* and /etc/asterisk/* and /var/lib/asterisk/ (the latter with a -r) and you're on your way |
| 23:30.08 | RoyK | Greyh0und: then the rest is rtfm |
| 23:30.39 | Greyh0und | RoyK: i have done that. No modules that is needed? |
| 23:30.55 | RoyK | Greyh0und: remove them before reinstalling |
| 23:31.18 | tzafrir_laptop | RoyK, if you're after some quick fixes, doxygen seems to give a few simple-tocorrect warnings. At least the with beta1. Things like documented parameters list out of sync wit hthe real one |
| 23:31.54 | *** join/#asterisk brookshire[home] (n=matt@esbrooks3.traveller.com) |
| 23:34.37 | syle2 | your suppose to rewrite latest cvs to be approved for submittions i thought |
| 23:35.17 | RoyK | no. i need the 'stable' to contain a few more things |
| 23:37.15 | epablo | Hi people.. Anyone here worked with EAGI in perl? |
| 23:37.29 | syle2 | i thought chan_sip already contained that |
| 23:37.40 | syle2 | i;ve only played with iaxfriends myself, seems to work fine there |
| 23:38.11 | nomazda | anyone ever try vbuzzer on asterisk? |
| 23:38.31 | generalhan | WTH? so i made an entire new directory with the op_server.pl and i pointed it to a different directory than the one i have already built. yet when i run the perl script it still has the same alyout as the one i did before |
| 23:39.16 | RoyK | syle: not mysql sipfriends |
| 23:39.24 | RoyK | syle: only cvs head |
| 23:40.29 | syle2 | idk , i use latest cvs head myself, all works great :) |
| 23:41.23 | kingtux | I have an old pctel modem can I use it for timming |
| 23:41.31 | epablo | How do I stop MOH on an extension? |
| 23:42.02 | RoyK | syle2: yeah, but you don't have a database backend with a few thousand customers wanting their system to work 24x7, right? |
| 23:42.12 | epablo | Got an AGI where I put someone on MOH a need to stop it and let the AGI keep rolling |
| 23:42.29 | epablo | or at least make it jump to the next priority |
| 23:42.31 | RoyK | epablo: show application moh says it can't be stopped |
| 23:43.30 | RoyK | s/moh/musiconhold/ |
| 23:43.32 | RoyK | whatever |
| 23:43.36 | epablo | RoyK: Have idea of a sub app? |
| 23:43.37 | syle2 | no i don't royk, but even if I did i would run latest cvs, and code process to switch to stable if anything bad happened |
| 23:44.10 | syle2 | but do you have an actual PRI or are you just acting as a proxy |
| 23:44.37 | epablo | I made it stop using ChanGrab, but then I loose control there.. I need to put someone on hold.. then be able to move him to an IVR or bridge a call |
| 23:45.44 | Vlat- | hi |
| 23:45.49 | *** join/#asterisk Rez (i=lorez@freenode/staff/lorez) |
| 23:45.58 | Vlat- | i'm back earlier than i was promise |
| 23:46.01 | Vlat- | 1:45 :) |
| 23:46.23 | Vlat- | i'm till 6:00 CET |
| 23:46.46 | Vlat- | so if any questions about SER+ASTERISK at carrier grade, just welcome |
| 23:47.11 | syle2 | umm who uses asterisk at carrier grade |
| 23:47.14 | Vlat- | (our technicans solved the prob. a spoke at about 18:00) |
| 23:47.20 | epablo | Vlat.. I tried to make such a config.. but it really didn't workout |
| 23:47.25 | syle2 | you mean lucent TNT's etc |
| 23:47.38 | RoyK | http://www.livejournal.com/community/girl_gamers/1871859.html#cutid1 |
| 23:47.41 | Vlat- | syle2: we're using it as media-backend |
| 23:48.10 | Vlat- | epablo: "such a config"... maybe couple of steps earlier ? |
| 23:48.11 | epablo | I didn't find a way of killing the call when the user ran out of funds |
| 23:48.12 | *** join/#asterisk iq (n=iq@207-224-109-4.omah.qwest.net) |
| 23:48.16 | kingtux | do you guys use amp? |
| 23:48.27 | syle2 | well i wouldn;t mind a SER working config if you got one |
| 23:48.34 | syle2 | they seem hard to come by |
| 23:48.52 | Vlat- | the ser working config is on onsip.org |
| 23:49.11 | RoyK | syle2: code stabilise, but is not going back into 'stable' |
| 23:49.36 | epablo | Vlat-: I think I didn't really get the SER paradigm |
| 23:49.51 | syle2 | never looked much at that site, they want you to register just to read articles |
| 23:49.58 | Vlat- | epablo: usually it take a month or two to get in to ser.cfg |
| 23:50.19 | Vlat- | i can imagine it to C source code |
| 23:50.59 | epablo | Vlat-: Yeah.. I sort of got the hang of it, but I needed a prepaid solution, and couldn't find the modules to make it work.. So I ditched it |
| 23:51.14 | Vlat- | damned god :) the competitor admins tried to drunk me in the last 4 hours |
| 23:51.29 | Vlat- | a get back to the office, and what do i see? |
| 23:51.47 | Vlat- | a 2 keg a beer at my table |
| 23:51.56 | *** join/#asterisk MikeJ[Laptop] (n=ircatjer@d14-69-8-30.try.wideopenwest.com) |
| 23:52.02 | *** part/#asterisk kingtux (n=kingtux@pool-151-196-44-240.balt.east.verizon.net) |
| 23:52.03 | Vlat- | "competitor admins" i meant |
| 23:52.09 | RoyK | http://www.livejournal.com/community/girl_gamers/1871859.html#cutid1 |
| 23:52.14 | RoyK | oh sorry |
| 23:52.26 | syle2 | what is a competitor admin |
| 23:52.49 | Vlat- | syle2: it's admin of other voip comapany |
| 23:53.38 | Vlat- | and it was the joke |
| 23:54.34 | syle2 | well I haven;t found a use for SER yet, but my understanding is that it would just act as a proxy either a) proxing the sip connection to level3 or wherever you get it from b) load balance between different asterisk servers....am i way off? |
| 23:54.53 | InfraRed | http://photos.subhi.com/c724474.html |
| 23:54.57 | InfraRed | heh |
| 23:55.31 | Vlat- | syle2: ser is a proxy. nothing more |
| 23:55.46 | Vlat- | syle2: you proxing the signalling with it |
| 23:56.27 | Vlat- | (can proxy RTP too, but why) |
| 23:56.46 | syle2 | idk if signalling is a good word, you mean the sip conversations back and forth, SER would only need to dish to asterisk if it needs to carry the RTP stream correct |
| 23:57.17 | Vlat- | syle2: hm, didn't got this |
| 23:57.29 | syle2 | SER can proxy RTP? |
| 23:57.35 | Vlat- | currently we're using SER as primary gw |
| 23:57.44 | Vlat- | asterisk as the media back-end |
| 23:58.05 | Vlat- | if the customer has the fscking NAT...we can solve it with SER |
| 23:58.27 | Vlat- | (with no additional line to config, it just works) |
| 23:58.40 | syle2 | i don;t see how NAT applies, as long as customer registers you solved the nat issue |
| 23:59.05 | Vlat- | syle2: reverse lookup + storing of the result to lookup table |
| 23:59.06 | epablo | If you have a media-proxy.. it just works.. like in * |
| 23:59.29 | epablo | but if both users are behind nat, you have to use the nat helper |
| 23:59.38 | Vlat- | no |
| 23:59.46 | syle2 | why don;t you just say store the ip address..much shorter vlat hehe |
| 23:59.50 | Vlat- | i just to have propertly set up these routers |
| 23:59.54 | Vlat- | to forward rtp |
| 23:59.59 | Vlat- | and signalling |