| 00:00.05 | *** join/#asterisk bjohnson (~bjohnson@66.11.165.161) |
| 00:02.46 | *** join/#asterisk brc_ (~brian@brc.base.supporter.pdpc) |
| 00:03.18 | *** join/#asterisk mithro (~tim@202.191.111.52) |
| 00:04.01 | *** join/#asterisk ctooley (~ctooley@rrcs-24-153-228-6.sw.biz.rr.com) |
| 00:04.36 | alt_phil | Anyone every install asterisk from the debian apt repositories without building it from source - and have good luck? My boss wants me to rebuild asterisk, I want to go source, he wants to go with the apt repository. Suggestions? |
| 00:05.09 | *** join/#asterisk zotz (~zotz@24.231.32.191) |
| 00:05.59 | JunK-C | why not getting the cvs-head? |
| 00:06.44 | ManxPower | alt_phil: Asterisk is one of those few pieces of software that is best built from source, at least until you become familiar with it. |
| 00:06.56 | ManxPower | Then build your own pacakge |
| 00:07.01 | NewSole | hmmmmm..... |
| 00:07.02 | alt_phil | That's what I want to do. I'm looking for arguments to convince my boss against using the apt repository |
| 00:07.32 | alt_phil | I'd just be more comfortable knowing everything was "so fresh and so clean" |
| 00:07.42 | tzafrir | alt_phil, me: both ways |
| 00:07.42 | alt_phil | But that's not much of an argument :) |
| 00:07.47 | ManxPower | alt_phil: I have never seen anyone come into the channel that was using a packaged asterisk that didn't switch to using source. |
| 00:08.01 | Jerub | alt_phil: go both! apt-get source -b asterisk ! |
| 00:08.05 | tzafrir | If you don't like the binary package from your distro, rebuild it |
| 00:08.10 | ManxPower | There are obviously people out there that use packages, but I just can't see doing it, and I'm a BIG fan of packages. |
| 00:08.42 | tzafrir | ManxPower, what's so wrong with that? |
| 00:08.52 | tzafrir | As opposed to , say, apache and mysql |
| 00:09.06 | NewSole | any one know why I am getting message from zap card..... |
| 00:09.07 | NewSole | <PROTECTED> |
| 00:09.07 | NewSole | Mar 29 19:04:28 WARNING[22993]: chan_sip.c:2020 sip_write: Asked to transmit frame type 4, while native formats is 256 (read/write = 256/256) |
| 00:09.33 | ManxPower | NewSole: "show codecs" |
| 00:10.07 | NewSole | did |
| 00:10.44 | *** join/#asterisk ArkyLady (ArkyLady@h248.76.255.206.cable.htsp.cablelynx.com) |
| 00:11.49 | Jerub | tzafrir: most people I know build apache and mysql from source too - there are compile time options that you need to fiddle with for optimisation and sanity reasons. |
| 00:12.43 | tzafrir | Jerub, wrong excuse: you can always patch the distro's apache |
| 00:13.07 | ManxPower | tzafrir: Asterisk changes fast. |
| 00:13.08 | tzafrir | And a good distro will have apache which is much saner to configure |
| 00:13.17 | *** join/#asterisk hermie (~nick@24.236.167.53.bay.mi.chartermi.net) |
| 00:13.29 | ManxPower | Building a package for each update is silly. |
| 00:13.29 | tzafrir | ManxPower, Asterisk depends on quite a few libs |
| 00:13.52 | ManxPower | tzafrir: There is that too. |
| 00:13.54 | Jerub | tzafrir: yes, but this was RHEL. ;) |
| 00:14.04 | tzafrir | ManxPower, building asterisk again for each computer is silly |
| 00:14.05 | ManxPower | And of course zaptel has to be built from source on each system |
| 00:14.27 | ManxPower | or at least on each kernel version |
| 00:14.31 | tzafrir | ManxPower, not if the distro provides zaptel modules for its kernel |
| 00:14.54 | ManxPower | tzafrir: even so. You should keep Asterisk and Zaptel build from the same version of source. |
| 00:15.15 | tzafrir | That's what Build-Depends is for |
| 00:15.22 | *** join/#asterisk anthm (~anthm@h460825da.area4.spcsdns.net) |
| 00:15.22 | *** mode/#asterisk [+o anthm] by ChanServ |
| 00:15.30 | ManxPower | i.e. zaptel-1.0.3 and asterisk-1.03 and not asterisk-1.0.8 |
| 00:16.18 | ManxPower | I upgrade when I see a fix posted to asterisk-cvs that I feel we need. |
| 00:16.19 | tzafrir | ManxPower, without package management you can only hope that this is the case. Typically you get error messages at run-time and not at install-time |
| 00:16.24 | ManxPower | Not based on version numbers |
| 00:17.03 | ManxPower | tzafrir: A whole lot of people stoped having problems when they used asterisk built from source. |
| 00:17.24 | tzafrir | ManxPower, this is a symptom of a build system that is not good enough |
| 00:17.38 | tzafrir | Asterisk has a relatively compex build system |
| 00:18.18 | tzafrir | It takes some taming. Once the taming is done, the process will be much simpler |
| 00:18.42 | ManxPower | tzafrir: Perhaps the whole asterisk-sounds fisasco soured me on the idea. |
| 00:19.03 | tzafrir | What do you mean? Any pointers? |
| 00:19.39 | *** join/#asterisk cbachman (~chatzilla@victory.ece.northwestern.edu) |
| 00:19.55 | dwmw2 | it would be nice to get asterisk and the zaptel drivers in RHEL |
| 00:20.02 | dwmw2 | or at least Fedora |
| 00:20.13 | dwmw2 | the zaptel drivers should ideally be in the upstream kernel |
| 00:20.15 | ManxPower | There is an asterisk-sounds cvs. It contains extra sounds for Asterisk. Some idiot pacakger took Asterisk, then broke it into asterisk and asterisk-sounds package. The asterisk-sounds were the sounds in the original asterisk tarball. |
| 00:20.24 | tzafrir | dwmw2, if you think it'd be so nice, go ahead and package it |
| 00:20.32 | ManxPower | Imagine the confusion when we told these people they wanted to get asterisk-sounds |
| 00:20.34 | dwmw2 | tzafrir: with pleasure |
| 00:20.41 | alt_phil | Be damn nice to get them in RHEL. I've got out box running on RHEL now, and a backup on a debian system, and both systems have different problems. It's driving me nuts. |
| 00:21.01 | dwmw2 | alt_phil: Do me a favour; file a RFE for it :) |
| 00:23.45 | tzafrir | ManxPower, a good package is not an easy thing to get. To get a package to magically install and upgrade requires some testing and debugging. |
| 00:23.58 | *** join/#asterisk lilo_ (lilo@levin-pdpc.staff.freenode) |
| 00:24.17 | tzafrir | And natuarally some silly mistakes are done in the process. |
| 00:24.42 | ManxPower | tzafrir: Everything I install other than Asterisk is from RPM |
| 00:25.25 | tzafrir | libtiff? h323? |
| 00:25.38 | ManxPower | tzafrir: I don't use either of those two things |
| 00:26.19 | tzafrir | There are so many very specific build requirements, and people waste tons of time reading the docs and compiling with the wrong versions and to the wrong locations. |
| 00:26.20 | ManxPower | I did have to install a libtiff when I played with spandsp way back |
| 00:26.44 | alt_phil | There's a good question to bring up. libtiff - trust a packaged one? (ie: Hope they didn't backport whatever broke the newer versions) |
| 00:27.02 | tzafrir | They still feel they've gained some guru-factor, but is it really necessary just to get the ****** PBX working? |
| 00:27.06 | ManxPower | alt_phil: I'm not up on what version spandsp requires these days. |
| 00:27.43 | alt_phil | I believe 3.5.7 and 3.6.0 are ok, but anything else is pretty much b0rkb0rkb0rk. I could be wrong on the versions, but I'm close. |
| 00:27.46 | tzanger | alt_phil: no don't trust a packaged one |
| 00:27.50 | ManxPower | tzafrir: I rsync everything from a central location. do: cd zaptel; make install; cd ../libpri; make install; cd ../asterisk; make install |
| 00:27.52 | tzanger | even the slackware 3.5.7 one didn't work right |
| 00:27.52 | ManxPower | and that's it |
| 00:28.08 | tzafrir | alt_phil, with a distro package you can normally know that others have built with the same version as you did, so success you can easily share your success/failure stories with others |
| 00:28.12 | dwmw2 | alexns: I'm using the FC3 packaged version with spandsp-0.0.2pre10 |
| 00:28.38 | ManxPower | dwmw2: the latest one fixes problems with some fax machines. |
| 00:28.48 | dwmw2 | latest spandsp? |
| 00:28.49 | tzafrir | Well, current version on Debian Sarge and Sid seems to be good enough |
| 00:28.51 | NewSole | Maxx you have any idea |
| 00:29.00 | dwmw2 | or latest libtiff? |
| 00:29.01 | tzafrir | I haven't yet built pre11, though |
| 00:29.03 | ManxPower | NewSole: about what? |
| 00:29.08 | tzafrir | latest libtiff |
| 00:29.12 | dwmw2 | ah, ok. |
| 00:29.13 | ManxPower | dwmw2: latest spandsp |
| 00:29.40 | ManxPower | dwmw2: It was announced on the mailing list. Go look for yourself. |
| 00:30.06 | ManxPower | ~google site:lists.digium.com *pre11* |
| 00:30.18 | ManxPower | gmm |
| 00:30.39 | tzafrir | How often does google index the list archives? |
| 00:31.03 | ManxPower | ~google site:lists.digium.com 4247B8F3.8040209@coppice.org |
| 00:31.16 | ManxPower | not enoug to index that |
| 00:31.39 | NewSole | ManxPower... any idea why I am getting that from zap card |
| 00:31.51 | ManxPower | The message is dated MAR 28 2005 |
| 00:32.05 | *** join/#asterisk Grooby (~Grooby@12.22.232.212) |
| 00:32.15 | ManxPower | NewSole: Did you look at "show codecs" |
| 00:32.19 | tzafrir | Anyway, what parts exactly of the kernel source does zaptel use for building? |
| 00:32.30 | NewSole | yes... how do I assign a codec to it |
| 00:32.30 | ManxPower | i have no idea. |
| 00:32.47 | ManxPower | NewSole: so what are the codec names for the numbers reported in the error message. |
| 00:32.50 | tzafrir | I wonder if the package that is called "kernel-headers" on Debian should be sufficient to build zaptel |
| 00:33.16 | dwmw2 | tzafrir: there should be a package which contains enough to build kernel modules. It's headers and a few makefiles. |
| 00:33.24 | tzafrir | It basically includes the skeleton of the kernel tree with the makefile and .config and include files |
| 00:33.28 | dwmw2 | kernel-headers may be the header files which are shared by userspace. |
| 00:33.36 | NewSole | 4 = G.711 u-law and 256 = g729 |
| 00:33.40 | dwmw2 | in Fedora the package you want would be kernel-devel |
| 00:33.52 | dwmw2 | and the userspace headers glibc-kernheaders |
| 00:34.15 | dwmw2 | tzafrir: do you have anything in /lib/modules/`uname -r`/build ? |
| 00:34.17 | ManxPower | NewSole: some device is trying to use G729 and you don't have a license. eithe buy a license or make the device not use G729 |
| 00:34.28 | tzafrir | But should that be enough to build zaptel? Or does it require some objects/sources from the source tree? |
| 00:34.38 | NewSole | ahh ol |
| 00:34.39 | dwmw2 | tzafrir: I think that should be enough |
| 00:34.55 | tzafrir | Well, that'll save me some space |
| 00:35.01 | NewSole | ya I forgot... have lic... just did not reg it |
| 00:35.08 | NewSole | thnx |
| 00:35.20 | tzafrir | BTW: is there really such a size difference between the 2.4 modules and 2.6 modules? |
| 00:35.29 | ManxPower | NewSole: It's best not to try to use Asterisk when you've been drinking. |
| 00:35.56 | NewSole | or up for 36 hours |
| 00:36.03 | tzafrir | The 2.4 modules are about 100-150kb (packed). The 2.6 modules take around 1MB |
| 00:36.05 | ManxPower | same effect |
| 00:36.38 | NewSole | hehehe |
| 00:36.40 | tzafrir | (that is: modules built for kernel 2.4 and for kernel 2.6) |
| 00:37.25 | dwmw2 | is that true even when the latter are stripped? |
| 00:37.45 | dwmw2 | <PROTECTED> |
| 00:37.49 | dwmw2 | <PROTECTED> |
| 00:37.53 | alt_phil | Oooo man, I think I may try working on my asterisk machines while drunk. I'm ready to try anything to get rid of these damned HDLC Abort errors. |
| 00:37.57 | tzafrir | dwmw2, I tried stripping the former. But then they refused to load |
| 00:38.05 | dwmw2 | 'strip -g' |
| 00:38.12 | *** join/#asterisk Brixius (~Brixius@rrcs-24-172-13-162.midsouth.biz.rr.com) |
| 00:38.17 | dwmw2 | stripping them completely wouldn't be wonderfully useful |
| 00:38.22 | tzanger | alt_phil: what have you done so far |
| 00:38.42 | *** join/#asterisk hardwire (~hardwire@209.112.194.45) |
| 00:39.36 | *** join/#asterisk lilo_ (lilo@levin-pdpc.staff.freenode) |
| 00:39.38 | alt_phil | First, the new digium card wasn't syncing it's clock, even though it was set right. Digium claims it's a bug in our PRI card. So we used the old card, had an IRQ conflict, fixed that, had our switchtype wrong, fixed that, rebuild the zaptel driver a few times... not sure what else to do yet. |
| 00:39.45 | dwmw2 | out of interest, why build Asterisk with -fsigned-char? Can't we just fix anywhere that makes broken assumptions about 'char' being signed? |
| 00:40.10 | *** join/#asterisk mgth (~chatzilla@dsl093-001-038.det1.dsl.speakeasy.net) |
| 00:40.12 | tzanger | alt_phil: what hardware are you on |
| 00:41.27 | alt_phil | A dell poweredge 400sc with a digium card needing the wct1xxp driver. |
| 00:41.58 | hardwire | as a voip provider the only way to set CID outbound is w/ an SS7 ? |
| 00:42.05 | alt_phil | T100P |
| 00:42.06 | *** join/#asterisk Kumbang (~ecvs@167.205.24.4) |
| 00:42.08 | tzanger | alt_phil: ok |
| 00:42.13 | tzanger | I have several of those working |
| 00:42.39 | tzanger | first things first -- I have always had better success compiling zaptel with MMX enabled and -march=pentium4 (or whatever your CPU is) |
| 00:42.50 | tzanger | actually before the first thing |
| 00:43.01 | tzanger | completely blow away your zaptel directory and check out fresh cvs |
| 00:43.05 | *** join/#asterisk easydone (~notdone@eksel.demon.nl) |
| 00:44.14 | alt_phil | Yeah, P4 2.8g |
| 00:44.55 | Brixius | is anyone besides me having issues with "avoiding iax destroy deadlock" errors locking up * |
| 00:46.39 | *** join/#asterisk mog_home (~mogorman@146.229.184.211) |
| 00:46.55 | tzafrir | dwmw2, thanks, strip -g seems to allow the module to load |
| 00:55.25 | *** join/#asterisk dmccollum (~dmccollum@69-164-245-72.atlaga.adelphia.net) |
| 00:55.33 | *** join/#asterisk _Blue_ (~devnull@dynamictelecom.net) |
| 00:56.06 | *** join/#asterisk DaGrim (DaGrim@dagrim.user) |
| 00:56.11 | _Blue_ | hello... is it possible to use TDMoE equipments like those from CTC Union with asterisk? |
| 00:56.27 | DaGrim | whats a good site that I can check the availability of a DID on for free? |
| 00:56.30 | DaGrim | anybody? =) |
| 00:57.25 | _Blue_ | :( |
| 01:03.11 | PTG1234 | on for free? |
| 01:03.19 | PTG1234 | like to see if a number is available? |
| 01:03.23 | DaGrim | Yea.. |
| 01:03.24 | PTG1234 | you mean 800 or normal? |
| 01:03.28 | DaGrim | 866 |
| 01:03.28 | mgth | dagrim: 800s you can do att.com |
| 01:03.35 | mgth | att.com |
| 01:03.36 | DaGrim | ok.. thanks |
| 01:03.48 | DaGrim | hah.. thats what I thought.. |
| 01:03.51 | DaGrim | hmm |
| 01:04.11 | *** part/#asterisk alt_phil (~alt_phil@abgtr1.abgnetwork.net) |
| 01:05.15 | DaGrim | All 3 of their stations here.. wont shut the hell up about how theyre giving away $10,000 every hour.. anybody that is whatever caller gets it automatically... right? here is the funny part.. I just purchased the DID they have been giving out on there air.. lol |
| 01:05.28 | Faithful | peterS gravity,helix: does the poetry that comes out of either one of them actually have enough structure to be called crappy haiku, though, as opposed to just generic crap? |
| 01:05.47 | DaGrim | And they plan on continuing that 'contest' for another 3 weeks? Obviously its a up their ratings.. |
| 01:07.50 | Jerub | DaGrim: you just bought the number? |
| 01:08.03 | Jerub | DaGrim: so you're going to be talking to the callers? |
| 01:08.07 | DaGrim | YEP |
| 01:08.45 | Jerub | DaGrim: send a fax to their competition, explain what you've done. |
| 01:08.58 | DaGrim | Hmmmm .. good idea |
| 01:09.03 | DaGrim | hehe |
| 01:09.20 | Jerub | DaGrim: a fax will get to the right people the fastest. an email will be ignored, and everyone knows you can't phone a radio station |
| 01:09.20 | webmiko | yea. tell them for enough youll have a message advertising any radio station they want. |
| 01:09.27 | DaGrim | =D |
| 01:11.06 | Jerub | DaGrim: got any calls yet? ;) |
| 01:12.36 | DaGrim | havent quite decided what to do yet |
| 01:12.36 | DaGrim | lol |
| 01:18.02 | Jerub | DaGrim: no, seriously, you've got the number, have you gotten any callers yet? |
| 01:18.15 | Jerub | DaGrim: I don't know how this stuff works, do you have to set it up somehow? |
| 01:18.39 | DaGrim | yea.. like .. I dont have enough bandwith to use it.. at the moment.. |
| 01:21.19 | bkw_ | quick |
| 01:21.20 | bkw_ | http://www.acurrentaffair.com/ |
| 01:21.23 | bkw_ | everyone go there |
| 01:21.26 | bkw_ | vote online |
| 01:21.28 | bkw_ | and call the tollfree |
| 01:21.39 | bkw_ | to vote if jacko is guilty or not |
| 01:22.37 | Jerub | wtf? |
| 01:24.05 | Nivex | bkw_: I think you have us confused with people who care |
| 01:24.25 | *** join/#asterisk kks (~kks@203.115.208.140) |
| 01:24.36 | dmccollum | you're thinking sexual. It's not sexual. it's a loving relationship. We lay in bed watching movies, eat cookies, read playboys and drink warm milk. |
| 01:24.50 | DaGrim | hah |
| 01:27.02 | fugitivo | dmccollum: i know what that feels, boring |
| 01:27.42 | Brixius | bkw_: where are you, that was on tv about an hour or 2 ago |
| 01:27.42 | *** join/#asterisk Sedorox (brandon@Neptune.client.wlmsprt.pa.sed6.net) |
| 01:28.24 | dmccollum | hey if Jackson can be late to court, bkw can be late with the info. |
| 01:28.34 | Brixius | true |
| 01:28.36 | dwmw2 | <PROTECTED> |
| 01:28.36 | dwmw2 | <PROTECTED> |
| 01:28.36 | dwmw2 | <PROTECTED> |
| 01:28.53 | dwmw2 | this was not the desired effect :) |
| 01:29.20 | Brixius | I was assuming that he's in another time zone then me, which would explain it. |
| 01:29.44 | Jerub | astounding |
| 01:31.26 | dmccollum | I heard something interesting the other day on the radio. Jackson started with all the plastic surgery after Brooke Shields told him she just wanted to be friends. |
| 01:33.47 | Jerub | I heard on the radio the other day that americans back assassins tried to kill a political leader in haiti. |
| 01:34.05 | Jerub | *grumble* |
| 01:34.15 | fugitivo | If my line doesn't have the right tension impulse to detect the hungup with my x100p, is any way to solve that problem? |
| 01:34.28 | *** join/#asterisk IQ (~IQ@65-103-166-49.omah.qwest.net) |
| 01:35.30 | Brixius | how do |
| 01:35.45 | Brixius | I checkout an old version of * from cvs |
| 01:36.17 | *** join/#asterisk DEEZED (~Scrilla@adsl-065-006-189-182.sip.bct.bellsouth.net) |
| 01:37.15 | Mavvie | Brixius: cvs update -r. see the cvs man-page for more information |
| 01:37.34 | Brixius | thanks |
| 01:41.19 | *** join/#asterisk zhier (~nick@219.137.39.14) |
| 01:44.02 | *** part/#asterisk Kumbang (~ecvs@167.205.24.4) |
| 01:44.11 | *** join/#asterisk JohnnyD (~passionfr@203-217-21-234.perm.iinet.net.au) |
| 01:44.22 | *** join/#asterisk verge (~jfargen@rrcs-67-78-209-206.se.biz.rr.com) |
| 01:44.46 | Brixius | does anyone know what the ip address is for cvs.digium.com for some reason I can't refrence it by name |
| 01:45.35 | Beirdo | ~dns cvs.digium.com |
| 01:45.43 | Beirdo | hmph |
| 01:45.47 | Beirdo | there ya go |
| 01:45.51 | Brixius | thanks |
| 01:45.52 | Beirdo | ~seen slePP |
| 01:45.54 | jbot | slepp is currently on #asterisk |
| 01:46.00 | *** join/#asterisk iCEBrkr (icebrkr@chrome.cyberdyne.org) |
| 01:46.29 | verge | I just tried to call a # that was busy. There was nothing but dead air. Is there something I can add to my dial plan so that I will receive the busy signal? |
| 01:46.47 | verge | I didn't know it was busy until I tried again using my landlind. |
| 01:47.22 | iCEBrkr | Any reason why I'd be getting this.. |
| 01:47.22 | iCEBrkr | Mar 29 20:44:36 NOTICE[7873]: chan_sip.c:7848 sip_poke_noanswer: Peer '2102' is now UNREACHABLE! |
| 01:47.31 | iCEBrkr | I'm still trying to debug my SPA2k |
| 01:49.10 | *** join/#asterisk sudhir492 (~sudhir@wbar1.wdc2-4-8-141-004.wdc2.dsl-verizon.net) |
| 01:49.19 | JohnnyD | this is linked to a qualify=yes or qualify=<a number> line in sip.conf |
| 01:49.21 | *** join/#asterisk tull (~danka@wwwcache1.livjm.ac.uk) |
| 01:49.22 | tull | hello |
| 01:49.31 | tull | does anyone use actos? |
| 01:49.47 | iCEBrkr | JohnnyD: Everyone tells me this, but that doesn't explain why it's unreachable. |
| 01:49.56 | iCEBrkr | It's on the same switch on the same network in the same house... |
| 01:50.19 | JohnnyD | I use Spa2ks as well, and they stay "reacheable"...hmmmm |
| 01:50.37 | iCEBrkr | I can ping the thing all night and day... |
| 01:50.37 | iCEBrkr | 10 packets transmitted, 10 received, 0% packet loss, time 9009ms |
| 01:50.38 | iCEBrkr | rtt min/avg/max/mdev = 0.768/0.776/0.787/0.031 ms |
| 01:51.17 | iCEBrkr | JohnnyD: It worked fine up until 2 days ago |
| 01:51.20 | marlowe | It can simply be a bad device... |
| 01:51.29 | marlowe | I've had devices simply gone bad. |
| 01:51.33 | *** part/#asterisk Defraz (~t0tal@sonicwall.dcdi.net) |
| 01:51.34 | iCEBrkr | I got another SPA-2k down here ( remote from my asterisk box ) and it's registered and reachable just fine. |
| 01:51.35 | JohnnyD | iCEBrkr: have you checked the value for the qualify statement? |
| 01:51.46 | iCEBrkr | marlowe: I'm leaning towards that. |
| 01:51.48 | marlowe | You say you can ping i but you can you access it via a web control panel / telnet, etc. |
| 01:51.52 | marlowe | Don |
| 01:51.54 | iCEBrkr | marlowe: Since I can't even upgrade the firmware |
| 01:52.09 | marlowe | I'm 99% sure it's not asterisk and it's the phone |
| 01:52.13 | marlowe | Or a bad switch, or hub. |
| 01:52.17 | JohnnyD | seems to be a dead Spa |
| 01:52.17 | marlowe | Even though you can ping it... |
| 01:52.18 | iCEBrkr | marlowe: I can configure it via the web interface as well, yes. |
| 01:52.34 | marlowe | I'd exchange it if it's under warranty. |
| 01:52.37 | marlowe | If it happens after that |
| 01:52.40 | marlowe | You know it's something else. |
| 01:52.45 | JohnnyD | doews it stays registered when you get the "unreacheable" message? |
| 01:53.02 | marlowe | I've driven myself crazy before with all types of phones with common problems.. Usually a bad device |
| 01:53.15 | iCEBrkr | <PROTECTED> |
| 01:53.16 | iCEBrkr | <PROTECTED> |
| 01:53.25 | marlowe | Especially if it'll initially register and then de-register |
| 01:53.32 | marlowe | iCEBrkr: Upgrade the firmware |
| 01:53.39 | JohnnyD | he can't! |
| 01:53.44 | iCEBrkr | it won't |
| 01:53.45 | marlowe | Why not? |
| 01:53.47 | iCEBrkr | dunno |
| 01:53.51 | marlowe | Get a new one |
| 01:53.53 | marlowe | End of story. |
| 01:53.56 | marlowe | Next |
| 01:54.04 | marlowe | :) |
| 01:54.05 | JohnnyD | put under the front (or rear tyre)! ..sorry |
| 01:54.14 | marlowe | tyre? |
| 01:54.16 | iCEBrkr | Yea, that's what I'm thinking |
| 01:54.18 | marlowe | Or tire? |
| 01:54.19 | JohnnyD | of your car |
| 01:54.26 | JohnnyD | lol |
| 01:54.38 | marlowe | I can accept fone instead of phone.. and there instead of their |
| 01:54.41 | marlowe | But tyre?? |
| 01:55.14 | Beirdo | tyre is proper spelling if you are a non-Yank |
| 01:55.20 | JohnnyD | even from OZ? |
| 01:55.23 | Beirdo | it is a possible proper spelling |
| 01:55.38 | marlowe | Is it really? |
| 01:55.41 | marlowe | I've never seen it before |
| 01:55.46 | JohnnyD | strait from teh Oxford dict. |
| 01:55.49 | Jerub | tyre is how I spell it. |
| 01:55.56 | marlowe | lol sorry for making fun of it then :) |
| 01:55.59 | iCEBrkr | Mar 29 20:55:44 WARNING[7873]: chan_sip.c:685 retrans_pkt: Maximum retries exceeded on call 73715f662d4244a26396a0d302119f5b@207.166.196.131 for seqno 102 (Non-critical Request) |
| 01:56.02 | iCEBrkr | See. |
| 01:56.04 | iCEBrkr | I shouldnt' get crap like that |
| 01:56.05 | marlowe | I've really never seen tyre.. Ok :) |
| 01:56.08 | Jerub | marlowe: I suppose you've never been to gaol then ;) |
| 01:56.17 | Beirdo | hehe |
| 01:56.24 | marlowe | iCEBrkr: That's kind of normal once in a while |
| 01:56.25 | Beirdo | let's hope not |
| 01:56.31 | marlowe | Jerub: nope! |
| 01:57.09 | marlowe | According to dictionary.com a tyre is curdled milk |
| 01:57.13 | marlowe | lol |
| 01:57.16 | iCEBrkr | Oh well. |
| 01:57.20 | iCEBrkr | I'm gonna have to consider this thing dead. |
| 01:57.22 | marlowe | amongst other things |
| 01:57.28 | marlowe | iCEBrkr: dont drive yourself crazy |
| 01:57.31 | marlowe | If it's under warranty replace it |
| 01:57.37 | marlowe | Especially since you cant upgrade the firmware |
| 01:57.53 | iCEBrkr | Well, I'm remote, so I can't tell exactly whats going on |
| 01:58.10 | Hmmhesays | are there any good reverse mobile lookups in the us? |
| 01:58.17 | sudhir492 | anyone using h323 here |
| 01:58.24 | marlowe | Hmmhesays: There's... uhh none |
| 01:58.26 | marlowe | that I know of |
| 01:58.30 | marlowe | There is talk of one |
| 01:58.41 | marlowe | I dont think it'll happen for a while to come |
| 01:58.56 | Hmmhesays | yeah |
| 01:58.58 | Hmmhesays | that sucks |
| 01:59.08 | marlowe | Not really.. people don't want it.. (most people) |
| 01:59.16 | marlowe | Think about what would happen if telemarketers got a hold of that |
| 01:59.31 | Hmmhesays | well, it sucks when you need to know the name of a number holder |
| 01:59.33 | sudhir492 | chan h323's documentation points to versions of pwlib and openh323 which are non existent at sourceforge:-( |
| 01:59.39 | marlowe | There is talk about having it.. Not having reverse only forward and not being able to search online |
| 01:59.46 | marlowe | It doesn't suck, it's called privacy |
| 01:59.48 | marlowe | Call them and ask |
| 01:59.49 | Hmmhesays | haha |
| 02:00.00 | Hmmhesays | it's a long story |
| 02:00.06 | marlowe | Give me $150 |
| 02:00.11 | marlowe | I'll give you the name and billing address |
| 02:00.15 | marlowe | ;) |
| 02:00.22 | Hmmhesays | haha I could slip the cellone guy a $50 |
| 02:00.25 | marlowe | Even a months of call log |
| 02:00.35 | Hmmhesays | now that would be impressive |
| 02:00.42 | marlowe | It's easy.. Well I don't do it |
| 02:00.44 | marlowe | my friend does. |
| 02:00.47 | DEEZED | hey is there a way to run caller ID information by a information database to do a reverse loojk up? |
| 02:00.58 | Hmmhesays | that's pretty interesting |
| 02:01.04 | Brixius | google a phone # |
| 02:01.08 | marlowe | DEEZED: Yea, calll verisign and ask to sign up for there sip service.. lol |
| 02:01.24 | Hmmhesays | one would think most service provider sites would be fairly secure |
| 02:01.39 | marlowe | Hmmhesays: I think he knows inside people at all companies |
| 02:01.39 | DEEZED | so i guess no automated way... I saw a company that offered it as a service with their IVR accounts |
| 02:01.47 | marlowe | Nothing to do with the web site |
| 02:01.59 | marlowe | DEEZED: sure, write a script.. Then it'll be automated. |
| 02:02.04 | DEEZED | lmao |
| 02:02.07 | marlowe | I've got a script that does a reverse lookup on anywho |
| 02:02.19 | marlowe | Then if nothing is found it queries my mysql database and prints the city and state |
| 02:02.34 | DEEZED | oh sweet |
| 02:02.47 | Hmmhesays | knowing people helps |
| 02:02.50 | marlowe | Yeah, it comes with perl-agi or asterisk, i forget which |
| 02:02.54 | *** join/#asterisk lilo (lilo@levin-pdpc.staff.freenode) |
| 02:02.57 | marlowe | Most of it - I made some modifications |
| 02:03.20 | JohnnyD | something funny with TDM4xx cards: I have two of those and the "top" command reports about 16% of cpu time spent in irq state. That goes to 32% when I put the second crad in. Is this normal? |
| 02:03.34 | JohnnyD | anyone seen that before? |
| 02:03.36 | marlowe | JohnJacob: not really |
| 02:03.38 | marlowe | err |
| 02:04.06 | marlowe | nope |
| 02:04.24 | *** join/#asterisk di5co (di5co@66.92.235.17) |
| 02:04.26 | JohnnyD | hmmmmmm |
| 02:04.34 | marlowe | This is normal.. |
| 02:04.49 | marlowe | CPU0 states: 0.0% user, 66.0% system, 0.0% nice, 33.0% idle |
| 02:04.49 | marlowe | CPU1 states: 0.0% user, 0.0% system, 0.0% nice, 100.0% idle |
| 02:04.49 | marlowe | CPU2 states: 0.0% user, 0.0% system, 0.0% nice, 100.0% idle |
| 02:04.49 | marlowe | CPU3 states: 0.0% user, 0.0% system, 0.0% nice, 100.0% idle |
| 02:04.55 | marlowe | woahh 33% idle |
| 02:04.55 | marlowe | wtf |
| 02:04.57 | JohnnyD | thanks, i feel better |
| 02:05.10 | marlowe | top is taking 94.1% cpu |
| 02:05.11 | marlowe | wtf |
| 02:05.16 | marlowe | 13598 root 15 0 1116 1116 864 R 94.1 0.1 0:00 top |
| 02:05.26 | Brixius | time to quit top |
| 02:05.32 | Jerub | top lies anyay. |
| 02:05.33 | JohnnyD | obviously |
| 02:05.56 | Jerub | because it only measures every 2 seconds or so, so it reports itself as using lots of cpu. |
| 02:06.04 | marlowe | Very true |
| 02:06.07 | marlowe | but not 94.1% |
| 02:06.09 | marlowe | It's normal now |
| 02:06.18 | Jerub | 94% isn't much |
| 02:06.28 | JohnnyD | it's normally around a couple of % for top |
| 02:06.30 | Jerub | that's just "cpu bound operation for a short time" |
| 02:06.32 | marlowe | top using 94.1%? |
| 02:06.40 | marlowe | No it was nailed at that for like 20 seconds |
| 02:06.40 | marlowe | lol |
| 02:06.49 | Jerub | oh, that's not normal. |
| 02:07.08 | Jerub | have a look at ps while top is running, see if you get similar numbers. |
| 02:07.16 | marlowe | Exactly |
| 02:08.30 | *** join/#asterisk marlowe (~marlowe@marlowe.active.supporter.pdpc) |
| 02:10.55 | marlowe | Ever since I've starting using asterisk's native moh... my cpu usage is very low |
| 02:11.55 | iCEBrkr | native moh? |
| 02:12.00 | marlowe | uh huh |
| 02:12.07 | iCEBrkr | What is this blasphemy you speak of? |
| 02:12.12 | iCEBrkr | :D |
| 02:12.22 | iCEBrkr | Ya don't have to use mpg123 anymore? |
| 02:12.40 | marlowe | no |
| 02:12.55 | marlowe | Wake up |
| 02:13.02 | marlowe | http://www.voip-info.org/tiki-index.php?page=Asterisk%20config%20musiconhold.conf |
| 02:13.14 | marlowe | Look towards the bottom |
| 02:13.41 | iCEBrkr | Nice. |
| 02:13.48 | iCEBrkr | That's gotta me relatively new. |
| 02:13.59 | Brixius | I was thinking of using a net radio station for moh, but does the native moh only running when someone's on hold, or will it be downloading the stream anytime asterisk is running? |
| 02:14.04 | marlowe | Not that new.. I dunno I cant remember |
| 02:14.14 | marlowe | Only when someons on hold |
| 02:14.15 | DEEZED | native moh automatically converts it? |
| 02:14.23 | Brixius | cool |
| 02:14.29 | iCEBrkr | Well.. it's not in 1.0.3 and if it is it wasn't documented when I compiled it |
| 02:14.39 | marlowe | But native wont work for streaming |
| 02:14.54 | iCEBrkr | I need this |
| 02:14.58 | marlowe | Shoutcast Music On Hold |
| 02:14.59 | marlowe | You can have asterisk use a streaming source for on-hold music. |
| 02:14.59 | marlowe | Make a directory and put a 0 size file ending in .mp3. |
| 02:14.59 | marlowe | I called my directory: /var/lib/asterisk/mohmp3-empty |
| 02:14.59 | marlowe | in musiconhold.conf, add a line such as: |
| 02:15.00 | marlowe | default => mp3:/var/lib/asterisk/mohmp3-empty,http://www.waixwave.com:8000/ |
| 02:15.05 | iCEBrkr | now I'm gonna have to tinker with it! |
| 02:15.15 | DEEZED | do you stull need to convert the mp3s to lame -q -p --mp3input -a --preset 8 -m mono in.mp3 8kout.mp3 |
| 02:15.22 | marlowe | DEEZED: no |
| 02:15.22 | iCEBrkr | shoutcast for moh.. that just chews up even MORE bandwidth. |
| 02:15.44 | marlowe | What I like about native moh is that everytime you place someone on hold it starts from the beginning of the file.. |
| 02:15.57 | DEEZED | really? thats great for companies |
| 02:16.02 | DEEZED | can you turn that off though? |
| 02:16.07 | marlowe | And if you have multiple files like I do and use the random feature, I have 20 files with advertisements and info for our company |
| 02:16.25 | marlowe | Ever since native moh + my new moh records.. my phone system rocks |
| 02:16.42 | marlowe | Customers have actually commented on the entire phone system, from the ivr to the moh.. to the functionality |
| 02:16.55 | marlowe | They love that they can reboot there own router by calling us and using the IVr |
| 02:17.15 | TomL | moh? |
| 02:17.20 | TomL | oh |
| 02:17.26 | marlowe | music on hold |
| 02:17.34 | TomL | right |
| 02:18.06 | *** part/#asterisk trig (~jb@xob.neospire.net) |
| 02:18.09 | marlowe | My nxt project which wont be hard is to let customers know there balance and accepted automated payments. |
| 02:18.11 | TomL | how do you reboot routers? |
| 02:18.23 | TomL | oh... wait |
| 02:18.26 | marlowe | TomL: A lot of code, a lot of work.. It'll be released one day. |
| 02:18.32 | TomL | network power switches and SNMP... |
| 02:18.36 | marlowe | no |
| 02:18.41 | TomL | yea, I could do that |
| 02:18.43 | TomL | no? |
| 02:18.43 | marlowe | It depends on the customer |
| 02:18.59 | marlowe | and what equipment they have |
| 02:19.07 | TomL | so yo don't just power cycle? |
| 02:19.11 | DEEZED | whats the cheapest sip phone or ata? |
| 02:19.12 | marlowe | no |
| 02:19.18 | TomL | what about issues that need a power cycle? it happens |
| 02:19.23 | marlowe | Physically login to the router and reboot it.. if that fails.. then power cycle |
| 02:19.44 | TomL | how does your IVR power cycle them? |
| 02:19.49 | iCEBrkr | DEEZED: why? what'cha looking to do? |
| 02:20.02 | DEEZED | something to replace eyebeam (xlite) |
| 02:20.04 | DEEZED | =) |
| 02:20.16 | marlowe | TomL: Custom made by an employee |
| 02:20.24 | *** part/#asterisk JerJer[mobile] (~jj@mail.nufone.net) |
| 02:20.26 | TomL | I mean, physically |
| 02:20.28 | iCEBrkr | DEEZED: Pick up a cheap-o Grandstream BT100 |
| 02:20.40 | x9-max | yea thats a gerat sip phone |
| 02:20.42 | TomL | cycling power requires controlling a relay at the minimum |
| 02:20.45 | x9-max | for the price lol |
| 02:20.49 | iCEBrkr | x9-max: hehe |
| 02:20.49 | marlowe | Correct |
| 02:21.04 | iCEBrkr | x9-max: It works fine.. just feels like it's made with tinker-toys |
| 02:21.04 | marlowe | It's a circuit board that controls a relay |
| 02:21.12 | DEEZED | oh yeah i saw that... whats a good website to buy from? |
| 02:21.18 | marlowe | You connect to the circuit board via IP, web or serial |
| 02:21.26 | x9-max | haha yea i was a lil scared when i unboxed mine thinking it would last a week |
| 02:21.37 | iCEBrkr | DEEZED: You can try voxilla.com |
| 02:21.49 | marlowe | i think i have the specs here if you want to see.. one sec |
| 02:21.52 | DEEZED | cool thanks man |
| 02:21.56 | iCEBrkr | x9-max: I still have and use mine.. But I did pick up a SPA-2k shortly after |
| 02:22.26 | marlowe | Of course my VPN isnt working.. :-/ |
| 02:22.31 | x9-max | ehh, ill wait till mine breaks or i get sick of it and bash it |
| 02:23.23 | marlowe | Designed primarily for power control of POE connected devices. The RPM-LV8 features 8 independently controlled power outputs and 6 short to ground inputs for "A/C on/off" or "door open/closed" type detection. Screw terminals allows for connections to a wide range of cables and accessories. The device is controlled via a COM cable connected to a computer or modem and works with any terminal program supporting VT100 emulation. The easy to u |
| 02:23.32 | iCEBrkr | haha |
| 02:23.54 | marlowe | Wow that's a bit outdated |
| 02:24.49 | marlowe | Ahh found the specs for it |
| 02:24.53 | marlowe | Seems he disspeared though |
| 02:29.45 | *** join/#asterisk viking78 (~Blah@cerberus.franklinamerican.com) |
| 02:29.49 | *** join/#asterisk ZX81 (~ZX81@222-153-20-34.jetstream.xtra.co.nz) |
| 02:29.55 | ZX81 | hi all |
| 02:30.10 | ZX81 | what's up with digium dns today - anyone else having problems? |
| 02:30.19 | ZX81 | working now |
| 02:30.20 | marlowe | not I |
| 02:30.34 | marlowe | then again I havent had to use digium at all today :) |
| 02:30.39 | ZX81 | :) |
| 02:30.44 | ZX81 | you're no help |
| 02:30.45 | ZX81 | :) |
| 02:31.01 | *** join/#asterisk NatRH (~Nat@dargo.trilug.org) |
| 02:31.07 | marlowe | Well you're right |
| 02:31.11 | marlowe | I can't resolve |
| 02:33.26 | Faithful | do zaphfc cards have problems with IRQ sharing? |
| 02:33.51 | Faithful | I will move it before I start if they do |
| 02:37.53 | *** join/#asterisk zotz (~zotz@24.231.32.191) |
| 02:39.27 | kks | Any recommendation where should I implement LCR in ser or asterisk? |
| 02:42.07 | ZX81 | marlowe: :) |
| 02:42.12 | *** join/#asterisk mogorman (~mogorman@dhcp-162.digium.com) |
| 02:42.22 | ZX81 | Faithful: better safe than sorry |
| 02:42.28 | *** join/#asterisk kram (~mark@kram.digium.sponsor.pdpc) |
| 02:42.28 | *** mode/#asterisk [+o kram] by ChanServ |
| 02:42.37 | ZX81 | if they do 1000 interrupts per second then yes |
| 02:42.44 | ZX81 | if not, then probably not |
| 02:45.37 | *** join/#asterisk nix000 (~nix000@66.11.165.188) |
| 02:46.16 | nix000 | anyone know what is the largest install based of digum and/or * ? and how they are configured ? |
| 02:46.39 | *** join/#asterisk pr0m (~pr0metheu@ip-wv-68-187-250-031.charterwv.net) |
| 02:47.29 | Mavvie | nix000: unless somebody overrules me, I have 2 quad PRI cards :-) |
| 02:47.50 | *** join/#asterisk techie (gus@asterisk.horizonte.us) |
| 02:48.35 | nix000 | Mavvie, you would think there is a where its used page on digium pages ! |
| 02:48.47 | Mavvie | we want to keep it secret. |
| 02:50.09 | Brixius | zx81, I had issues earlier today. |
| 02:50.15 | nix000 | Mavvie, have you seen server configurations for high traffic based on digium cards ? |
| 02:50.16 | Brixius | with digium dns that is |
| 02:50.40 | _Vile | Mav, what box? |
| 02:51.17 | nix000 | Mavvie, the sysmaster gateway box seems very nice. How did they do it ? |
| 02:51.18 | Mavvie | not that interesting, it's a pentium 3.4GHz with 1.5Mb of memory. |
| 02:51.23 | Mavvie | Gb of memory. |
| 02:51.54 | Mavvie | hmm... "sip reload" doesn't redo the bind address. |
| 02:51.56 | _Vile | what kinda mobo? |
| 02:52.31 | nix000 | They are using server blades made out of digium vards. |
| 02:52.32 | Mavvie | an Intel one. if you know how to get the information from /proc I'll tell you. |
| 02:52.58 | nix000 | cat /proc/cpuinfo ??? |
| 02:53.13 | Mavvie | that doesn't show the motherboard. |
| 02:53.18 | nix000 | oops that cpu ! |
| 02:53.27 | _Vile | cat /proc/ioports |
| 02:53.37 | _Vile | cat /proc/pci |
| 02:53.39 | _Vile | :) |
| 02:53.51 | *** join/#asterisk brettcar (~brettcar@69.60.121.206) |
| 02:54.52 | brettcar | Hello all, might anyone have any idea why'd I'd get the following error when calling $AGI->stream_file('recorded-voice'); |
| 02:54.55 | brettcar | Use of uninitialized value in numeric eq (==) at /usr/local/share/perl/5.8.4/Asterisk/AGI.pm line 188, <STDIN> line 1. |
| 02:55.16 | brettcar | I looked at the AGI.pm file and I'm pretty clueless as where to look next. I also tried to debug my own code, not sure what is causing it. |
| 02:55.46 | nix000 | Mavvie, i need someone to bounce some * gateway config ideas .. you volunteer ? |
| 02:56.03 | _Vile | brett -> my $variable; |
| 02:56.11 | Mavvie | nix000: no |
| 02:56.20 | nix000 | tow! |
| 02:57.06 | nix000 | anyone ever done a large scale * deploy pleaze raise your hand ! |
| 02:57.14 | brettcar | _Vile: I do have all my local variables with 'my' already though. |
| 02:57.36 | brettcar | _Vile: Oh wait, hold on. |
| 02:58.01 | brettcar | _Vile: Found one without my, adding it didn't help though. |
| 02:58.52 | MajestiK | I've got a weird problem, every time I hang up after a call, my phone rings once. I'm using analog phones on a pap2-na |
| 02:59.24 | *** join/#asterisk tessier (~treed@222.253.65.202) |
| 02:59.41 | bkw_ | FYI folks http://www.acurrentaffair.com/ <-- the jacko poll number there.. is racing thru our asterisk system right now |
| 02:59.54 | bkw_ | or should I say serviced by asterisk |
| 03:00.15 | Qwell | ugh |
| 03:00.27 | Qwell | acurrentaffair == crap :( |
| 03:00.37 | bkw_ | who cares |
| 03:01.18 | Qwell | Thats cool though |
| 03:01.18 | Mavvie | brilliant poll! "Is Michael Jackson Guilty or Not Guilty?" |
| 03:01.18 | TomL | are you getting paid? |
| 03:01.18 | nix000 | i am thinking to deploy multiple (no hdd movingpart) mini-itx based quad PRI setups so that if one of the cards fails i can replace it whithout taking out a whole city. Any idea why that could be wrong |
| 03:01.18 | Qwell | bkw_: How'd you get that info? |
| 03:01.28 | tessier | Mavvie: Because we've all heard all of the evidence and can actually have an informed opinion, right? |
| 03:01.32 | tessier | I don't vote in polls like that. |
| 03:02.07 | *** join/#asterisk Newbie___ (me@60.48.165.86) |
| 03:02.11 | tessier | In polls like "Do you feel Terry Schiavo should live or die?" I can vote because that is asking for something I can have an opinion on. But Jacko? No. |
| 03:02.12 | *** part/#asterisk mog_home (~mogorman@146.229.184.211) |
| 03:02.17 | Mavvie | nix000: smart dialplan and redundant ports. |
| 03:02.23 | Newbie___ | hi, any AGI expert here? |
| 03:02.37 | chap | Ok... "Is michael jackson an odd man?" YES! ;) |
| 03:02.38 | Mavvie | tessier: :- |
| 03:02.39 | Mavvie | tessier: :-) |
| 03:02.41 | Mavvie | that's the one |
| 03:02.52 | Qwell | chap: Give me 5 minutes, and I'll put that poll up. :P |
| 03:04.06 | tessier | "Do you think Michael Jackson might be the sort of person who would might bugger a baby?" YES! |
| 03:04.17 | Damin | Lick me.. |
| 03:04.28 | brettcar | _Vile: Thanks anyways figured it out. |
| 03:04.30 | *** part/#asterisk brettcar (~brettcar@69.60.121.206) |
| 03:05.05 | `Sauron | TEssier: DO you really think that, or are you just following the leader? |
| 03:05.21 | nix000 | Mavvie, we are using areski, and altho i have not played with it yet i think the dial plan is based on the db. as for the redundancy i imagine it is cheaper to keep 2 mini-itx boxes than full blown servers. |
| 03:05.28 | DEEZED | hey if i want all calls to use my iax.cc, how do i modify this line for iax.cc use? (sixtel is the name in iax.conf) exten => _17XXNXXXXXX,1,Dial(IAX2/jjhall@iaxtel/${EXTEN}@iaxtel) |
| 03:05.50 | Damin | dddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddd |
| 03:05.55 | Mavvie | nix000: in theory, with the redundancy, you need as many free ports as you think there will be blow out at once. |
| 03:05.59 | `Sauron | Damin fell asleep |
| 03:06.11 | Newbie___ | hmmmm |
| 03:06.37 | Newbie___ | using agi-egate , how do i make * dial a desire provider like bv ? |
| 03:06.53 | Brixius | is there a features matrix for stable vs head |
| 03:06.55 | Damin | dddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddsaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaa |
| 03:07.29 | Mavvie | nix000: so if you have 8 boxes with 4 PRI each and you expect that two will burn out at once, you need eight spare ports which makes that you need 10 boxes with 3 used PRI ports and one free. |
| 03:07.47 | Mavvie | and if one burns out, you juggle the PRI links to the new boxes. |
| 03:08.23 | Mavvie | `Sauron: falling asleep on the D? I believe. But going to A with just one single S? No way. |
| 03:08.46 | `Sauron | You'd be surprised. |
| 03:09.07 | bjohnson | DEEZED: you're going to have to try a little harder than that |
| 03:09.18 | DEEZED | lmao |
| 03:09.29 | bjohnson | Brixius: nothing that organized |
| 03:09.36 | bjohnson | Brixius: you could start one |
| 03:10.33 | nix000 | Mavvie, exactly that is what i am triying to minimize ... if every mini-itx contains one pri then i only need a much less number of redeundant server/blades |
| 03:11.18 | Brixius | bjohnson: thanks |
| 03:12.19 | DEEZED | bjohnson: can you just tell me what _011. means in exten => _011.,1,Dial(IAX2/****@sixTel/${EXTEN}) |
| 03:12.46 | Mavvie | nix000: but then you have boxes which never are used unless in an emergency, and then you realize there is something wrong with them. |
| 03:13.10 | Silik0n | it means anything dialed that starts w/ 011 |
| 03:13.13 | dave_7 | _011. matches any international number |
| 03:13.27 | nix000 | Mavvie, never considered that .. but its a good point. |
| 03:13.29 | DEEZED | oh sweet thx |
| 03:13.38 | Brixius | DEEZED: it matches anything that starts with 011 |
| 03:13.41 | Silik0n | so like check uut the wiki... dialplan paterns are covered there |
| 03:13.53 | Silik0n | ~wiki |
| 03:13.55 | DEEZED | ive been reading it all day... |
| 03:13.56 | *** join/#asterisk IQ (~iq@65-103-166-49.omah.qwest.net) |
| 03:14.01 | Faithful | guys do i need to run ztcfg everytime before running * ? |
| 03:14.20 | Silik0n | if you dont have modules.conf set to run it when you modprobe the TDM drivers |
| 03:14.46 | *** join/#asterisk jdiskywlkr (~kvirc@ip68-0-90-1.tu.ok.cox.net) |
| 03:14.55 | Silik0n | if you like stop asterisk and want to restart it you only need to run it if you changed the Zap configs |
| 03:15.52 | *** join/#asterisk SPoon_TSX (~SPoon_TSX@24.83.96.211) |
| 03:15.56 | *** join/#asterisk kpfleming (~chatzilla@ip68-3-230-141.ph.ph.cox.net) |
| 03:17.14 | Faithful | Ok, I am doing zaphfc for the 1st time so I am not savvy with zap ! |
| 03:17.41 | Faithful | Imagine running * for 3 months and not knowing zap? |
| 03:18.13 | bjohnson | DEEZED: http://www.voip-info.org/wiki-Asterisk+config+extensions.conf |
| 03:18.42 | DEEZED | yes. i am very familiar with that page. Ill read it through again |
| 03:19.28 | bjohnson | DEEZED: http://www.voip-info.org/wiki-Asterisk+Dialplan+Patterns |
| 03:19.44 | *** join/#asterisk vlan (~iq@65-103-166-49.omah.qwest.net) |
| 03:19.50 | DEEZED | thx |
| 03:20.19 | bjohnson | DEEZED: and eventually you'll ask about ${EXTEN:1} which is covered here: http://www.voip-info.org/wiki-Asterisk+variables |
| 03:20.30 | DEEZED | sweet |
| 03:21.06 | DEEZED | please answer this.. what does _ mean in an extension |
| 03:21.36 | *** part/#asterisk BrianR___ (brianr@c-24-61-206-174.hsd1.ma.comcast.net) |
| 03:21.50 | *** join/#asterisk vlan (~iq@65-103-166-49.omah.qwest.net) |
| 03:21.58 | mithro | anyone here using x100p in australia? |
| 03:22.06 | Qwell | DEEZED: _ means that the extension should accept wildcards |
| 03:22.28 | DEEZED | such as n, x, or z |
| 03:22.29 | bjohnson | DEEZED: means it's the start of a pattern match definition |
| 03:23.34 | SPoon_TSX | Hi there, I just got a hardare phone and whenever I tried to use the phone to call, I got this message from Asterisk: |
| 03:23.50 | SPoon_TSX | Cannot convert G732 to ulaw |
| 03:23.56 | SPoon_TSX | Then the call was dropped. |
| 03:23.58 | Brixius | DEEZED: you need the _ if you are using n or x or . if you want to match a # exactly, ie 911 you don't need the _ |
| 03:23.59 | SPoon_TSX | Any idea? |
| 03:24.37 | IQ | ~list |
| 03:24.39 | jbot | one warez list being sent |
| 03:24.52 | IQ | how do we get list of jbot commands? |
| 03:24.54 | *** part/#asterisk Grooby (~Grooby@12.22.232.212) |
| 03:25.07 | SPoon_TSX | Actually the excat message is: channel.c:1764 ast_set_read_format: Unable to find |
| 03:25.07 | SPoon_TSX | <PROTECTED> |
| 03:25.08 | bjohnson | SPoon_TSX: I don't think * supports g732 |
| 03:25.47 | Brixius | SPoon_TSX, g732 is not a supported codec, use gsm, ulaw, or alaw and see if it works |
| 03:26.37 | SPoon_TSX | Damn, the phone doesn't allow me to choose the codec, what can I do?? |
| 03:26.47 | Brixius | what type of phone is it? |
| 03:26.57 | SPoon_TSX | Sayson 480i |
| 03:27.32 | bjohnson | see if they have a firmware update to get it to support something else |
| 03:27.46 | bjohnson | first time I've heard of something not supporting ulaw |
| 03:28.43 | SPoon_TSX | Or, do I need to enter something to force it to use ulaw in SIP.CONF? |
| 03:28.45 | Faithful | So far zaphfc seems to work MUCH better than i4l |
| 03:29.01 | DEEZED | heh finally got it to work... created an outgoing contest with my iax trunk, and then i created a sip context and did a "include = outgoing" |
| 03:29.12 | DEEZED | got im a noob.. but finally i got outgoing working |
| 03:29.57 | nix000 | is there a way one could hook the digium cards back to back and test it by driving both ends of the call ? |
| 03:30.36 | Shido6 | yes |
| 03:30.39 | Shido6 | nix |
| 03:30.41 | Brixius | I vagly remember being there, the zap configuration is somewhat confusing at first since the file format isn't like the rest of the * config files. |
| 03:31.07 | Brixius | nix, x-over t1 cable |
| 03:31.25 | Shido6 | yep |
| 03:31.31 | nix000 | Shido6, Brixius so you could have asterisk driving both ends. |
| 03:31.46 | nix000 | even on the same card ? |
| 03:31.54 | Shido6 | yes |
| 03:32.07 | Shido6 | what are you up to? |
| 03:32.07 | sivana | woot |
| 03:33.15 | Brixius | Is g.711 ulaw or alaw? |
| 03:33.27 | Brixius | or neither |
| 03:33.43 | nix000 | Shido6, i just want to test my setup before i take it in production. in a real telco env. Mind you i still dont have the cards with me yet ! |
| 03:34.58 | Shido6 | 711 |
| 03:34.59 | Shido6 | is ulaw |
| 03:35.04 | Shido6 | g.711 is ULAW |
| 03:35.09 | Shido6 | g711 is ULA |
| 03:35.12 | Brixius | ok, thanks |
| 03:35.54 | Shido6 | nix000 use greg@nufone.net when u get em or ring us and we'll help you out |
| 03:35.55 | nix000 | Shido6, fwiw i am also looking for someone who did large scale * deployment. for example any idea on the gateway configurations ? |
| 03:36.08 | Brixius | SPoon_TSX: Looks like your phone will do ulaw then if you set it to g.711 |
| 03:36.22 | Shido6 | funny |
| 03:36.29 | SPoon_TSX | Brixius: But how? |
| 03:36.36 | Brixius | "G.711 m/A and G.729A (Annex B) configuration" <-- from there sip tech specs. |
| 03:36.38 | Shido6 | ever heard of NuFone? We off support at $85/hr |
| 03:37.00 | Shido6 | if ya need g729 you can buy licenses from Digium at $10/ds0 |
| 03:37.46 | Brixius | SPoon: are you allowing ulaw and alaw in your sip.conf file |
| 03:37.58 | SPoon_TSX | Brixius: So I need to enforce it on my sip.com. btw, I allow=all. |
| 03:38.56 | Shido6 | disallow=all |
| 03:38.58 | Shido6 | allow=ulaw |
| 03:39.01 | Shido6 | in both the user |
| 03:39.02 | Shido6 | and peer |
| 03:39.18 | Shido6 | disallow=all says HEY DAMNIT ALLOW NOTHING!!!!! |
| 03:39.28 | Shido6 | then allow=ulaw says OK ....... ALLOW ULAW ONLY!!! |
| 03:39.36 | *** part/#asterisk JohnnyD (~passionfr@203-217-21-234.perm.iinet.net.au) |
| 03:39.40 | Shido6 | that forces ulaw so make sure the phone is set to use ulaw |
| 03:39.47 | Shido6 | or you'll get adverse effects :) |
| 03:40.16 | Brixius | SPoon: I havn't found any configuration documentation for your phone, was just looking for the specs. |
| 03:40.36 | Shido6 | what phone? |
| 03:40.44 | fugitivo | is any way to setup voicemail timeout? |
| 03:40.55 | Shido6 | what do you mean by voicemail timeout, fugitivo ? |
| 03:41.53 | fugitivo | i mean, record the voicemail for 20 sec, and hungup |
| 03:42.07 | fugitivo | or do whatever i want to do after the 20 sec |
| 03:42.55 | Shido6 | 20 second limit on voicemail |
| 03:42.56 | Shido6 | ? |
| 03:42.59 | fugitivo | yes |
| 03:43.18 | Shido6 | ; Minimum length of a voicemail message in seconds |
| 03:43.18 | Shido6 | ;minmessage=3 |
| 03:43.29 | Shido6 | ; Maximum length of a voicemail message in seconds |
| 03:43.30 | Shido6 | ;maxmessage=180 |
| 03:43.37 | Shido6 | in /etc/asterisk/voicemail.conf |
| 03:43.49 | fugitivo | thanks, i was looking in the wrong file :) |
| 03:43.55 | Shido6 | do you have multiple companys on your asterisk box? |
| 03:44.11 | fugitivo | no |
| 03:44.15 | Shido6 | good to go |
| 03:44.17 | Shido6 | set it |
| 03:44.20 | Shido6 | and forget it |
| 03:44.24 | Shido6 | yay!!!! |
| 03:44.39 | Shido6 | if its a live system |
| 03:44.46 | Shido6 | unload app_voicemail.so |
| 03:44.47 | Shido6 | then |
| 03:44.51 | Shido6 | load app_voicemail.so |
| 03:44.57 | Shido6 | then the .conf gets reparsed |
| 03:45.34 | fugitivo | thanks! |
| 03:49.02 | Mavvie | "reload app_voicemail.so" will reload the configuration too. |
| 03:49.17 | Mavvie | unload and load actually removed and reloads the whole module :-) |
| 03:49.22 | *** join/#asterisk Juxt (user@sfl-dsl-64-135-113-4-cust.host.net) |
| 03:49.27 | Juxt | good afternoon |
| 03:49.34 | Juxt | i mean good nite :-) |
| 03:50.04 | Juxt | is it possible to send a fax via a termination carrier like livevoip? |
| 03:50.11 | Mavvie | always funny that mantis. |
| 03:50.19 | Mavvie | Juxt: yes. pray :-) |
| 03:50.47 | Juxt | so it's been done :-) |
| 03:50.49 | Brixius | SPoon: Here's what I found on the wiki - http://www.voip-info.org/wiki-Aastra+480i |
| 03:50.50 | sivana | Juxt: less than 10% success rate for faxing through general internet traffic |
| 03:50.57 | Mavvie | Juxt: no. didn't pray hard enough. |
| 03:51.39 | *** join/#asterisk [hC] (~turnerd@c-69-180-109-192.hsd1.fl.comcast.net) |
| 03:51.40 | Juxt | so what's a better way of doing this |
| 03:51.47 | *** join/#asterisk dmccollum (~dmccollum@69-164-245-72.atlaga.adelphia.net) |
| 03:51.55 | sivana | Juxt: don't fax over IP :) |
| 03:52.08 | Juxt | well not really a possibility |
| 03:52.10 | Mavvie | Juxt: http://www.soft-switch.org/foip.html |
| 03:52.13 | [hC] | Ive done fax over ip, it seems usable. maybe i just havent done it enough |
| 03:52.18 | [hC] | i just connected it to a sipura. |
| 03:52.19 | [hC] | heh |
| 03:52.41 | sivana | [hC]: over the general internet? |
| 03:52.56 | sivana | I find that hard to believe, fax me :) |
| 03:53.16 | Mavvie | Juxt: but like others said: get a loooooow latency, a clear channel and it will work. |
| 03:53.17 | [hC] | hey, has anyone come up with a decent solution for wanting a cisco 7960 and a cordless phone registered to the same sip extension? (I have a cisco desk phone, but i also want a cordless phone for when im around the house, heh) |
| 03:53.31 | [hC] | sivana: no, t1 PRI in, then * -> sipura -> fax machine |
| 03:53.33 | Mavvie | [hC]: follow me. |
| 03:53.37 | Shido6 | dood |
| 03:53.45 | Shido6 | you dont "register" them to the same extension |
| 03:53.51 | [hC] | Yes I realize. |
| 03:53.51 | [hC] | heh, |
| 03:53.53 | Shido6 | you register them as different users and peers |
| 03:54.02 | Shido6 | but you can have the same extension dialed ring them both |
| 03:54.06 | Shido6 | and whichever you pick up |
| 03:54.10 | [hC] | Yeah I know |
| 03:54.11 | Shido6 | will take the call |
| 03:54.13 | Shido6 | ok :) |
| 03:54.16 | Juxt | well seems like store and forward might work |
| 03:54.16 | Shido6 | so whats up? |
| 03:54.20 | [hC] | well |
| 03:54.33 | Shido6 | I have my IAXy and one of the lines on my crisco to ring when u dial ext 3000 |
| 03:54.35 | sivana | Juxt: I doubt it |
| 03:54.55 | Mavvie | Juxt: to be honest, we do it here via one of our uplinks. 15ms between us and them and 95+% of them work fine. |
| 03:54.56 | sivana | Juxt: wait, what do you mean store and forward? |
| 03:55.20 | [hC] | There's a headset jack on the 7960, i was curious if anyone had done anything creative with that i guess, doing a dial double sip kinda sucks if one extension isnt registered or something, or if one was forwarded and not the other, or something. I might just configure a group with both sip exts. |
| 03:55.22 | Juxt | sorry wasn't for you |
| 03:55.29 | Juxt | i was mumbling about the fax |
| 03:55.42 | sivana | you really need a good QoS for any kind of useable success rate |
| 03:55.44 | [hC] | What im really curious about is if anyone has combined the 7960 and a plantronics headset lifter in conjunction with a bluetooth headset somehow |
| 03:55.53 | Shido6 | ahhhhh |
| 03:55.56 | sivana | and over the general internet isn't it |
| 03:56.00 | [hC] | maybe if there is some sort of rj11->bluetooth adapter |
| 03:56.06 | Shido6 | there's a bluetooth headset for the crisco |
| 03:56.11 | [hC] | but then the bt headset would have to trigger the lifter |
| 03:56.16 | [hC] | there is? |
| 03:56.22 | [hC] | damn rights! any idea on what its called? |
| 03:56.29 | Shido6 | tehy use it on the "Hit Fox TV Show, 24" ( yes I get paid to advertise 24 ) |
| 03:56.29 | *** join/#asterisk MrBelvedr (~tt@ip68-227-209-110.dc.dc.cox.net) |
| 03:56.40 | [hC] | Wow, me too! |
| 03:56.41 | [hC] | (not really) |
| 03:56.44 | Juxt | good nite |
| 03:56.44 | *** part/#asterisk Juxt (user@sfl-dsl-64-135-113-4-cust.host.net) |
| 03:56.48 | Shido6 | (wakes up) |
| 03:57.25 | [hC] | I wanted to use the same headset i use w/ my cellphone.. |
| 03:57.28 | [hC] | I'll look it up.. |
| 03:59.30 | *** join/#asterisk DannyF (~dannyf@h27n3c1o848.bredband.skanova.com) |
| 04:01.15 | *** join/#asterisk Half_Dome (~jelway@mail.westmarkinc.com) |
| 04:01.32 | Shido6 | i got it, stdby |
| 04:02.17 | Half_Dome | Can * run on SuSe 9.1? |
| 04:02.17 | MrBelvedr | to get the most current stable version would the command be: cvs checkout -r v1-0 zaptel libpri asterisk asterisk-addons asterisk-sounds |
| 04:03.01 | Silik0n | why would it not run on suse? |
| 04:03.23 | Half_Dome | Just figured I'd ask as I'm struggling |
| 04:03.29 | [hC] | hm. I found one from GN Netcom |
| 04:03.48 | [hC] | bad sound quality review.. |
| 04:04.29 | *** join/#asterisk TheSin (~TheSin@iphost-64-56-130-194.edm.wiband.net) |
| 04:04.32 | TheSin | lo all |
| 04:04.57 | Silik0n | Half_Dome: is it fialing to compile or something else? |
| 04:05.00 | TheSin | I'm switching from a cisco 3600 PRI card to a digium wcte11xp |
| 04:05.16 | Shido6 | yeAAAAAAAAyuh! |
| 04:05.20 | TheSin | and once configured and running incoming calls look like this in asterisk |
| 04:05.23 | Shido6 | (Lil John, getting VoIP) |
| 04:05.24 | NatRH | New Voicepulse TOS just arrived in email.... :( |
| 04:05.27 | TheSin | -- Extension '' in context 'incoming' from '!' does not exist. Rejecting call on channel 0/1, span 1 |
| 04:05.36 | TheSin | anyone ever see that or know how to fix it? |
| 04:05.40 | Shido6 | ay to go TheSin |
| 04:05.51 | Shido6 | wwwwww |
| 04:05.52 | Shido6 | WWWWWW |
| 04:05.54 | TheSin | :P |
| 04:06.07 | Silik0n | put your configs in a pastebin and post the url someone will look at it |
| 04:06.10 | [hC] | Shido6: you dont know the model number of the bluetooth headset for the cisco do you? |
| 04:06.13 | x9-max | heh they lowered per min price :) |
| 04:06.34 | NatRH | DID - $11 / month |
| 04:06.39 | x9-max | yea not bad |
| 04:06.43 | TheSin | Silik0n, was that for me? |
| 04:06.51 | Half_Dome | Silik0n: I'm pretty sure I've compiled. I don't know where to go next. I can't find samples. When I run asterisk -c I get error while loading shared libraries and somthing about res_features.so |
| 04:06.57 | Silik0n | TheSinYes |
| 04:06.59 | TheSin | if so which confs do you want just zapata.conf? |
| 04:07.43 | Silik0n | both zaptel and zapata /.confs, anand maybe even the relevant contextfrom ext.cong (with any senative data stripped |
| 04:07.56 | TheSin | k |
| 04:08.00 | TheSin | !pastebot |
| 04:08.37 | Silik0n | pastebin.ca |
| 04:08.41 | pigpen | Anyone online that can tell me what the following errors are: |
| 04:08.42 | pigpen | Mar 29 22:06:48 WARNING[3078]: channel.c:2115 ast_channel_make_compatible: No path to translate from IAX2/NuFone@198.22.67.70:4569/1(2) to SIP/mark-066f(256) |
| 04:08.43 | pigpen | Mar 29 22:06:48 WARNING[3078]: app_dial.c:1006 dial_exec: Had to drop call because I couldn't make IAX2/NuFone@198.22.67.70:4569/1 compatible with SIP/mark-066f |
| 04:08.43 | pigpen | <PROTECTED> |
| 04:08.43 | pigpen | <PROTECTED> |
| 04:08.51 | TheSin | sweet .ca |
| 04:08.52 | Silik0n | code mismatch |
| 04:08.58 | Silik0n | codec even |
| 04:08.58 | TheSin | thanks Silik0n |
| 04:09.02 | Silik0n | damned laptop keyboard |
| 04:09.23 | pigpen | Silik0n: regarding my error? |
| 04:09.28 | Silik0n | pigpenyes |
| 04:09.45 | Silik0n | you dont have a codec thats common to nufone's enabled codec enabled |
| 04:09.45 | pigpen | ok...so what do you think would cause this...it had been working fine... |
| 04:10.02 | Silik0n | and/or you dont have the correct transcoder |
| 04:10.08 | *** join/#asterisk SPoon_TSX (~SPoon_TSX@24.83.96.211) |
| 04:10.19 | pigpen | I have purchased 4 G729 codecs...with 4 polycoms... |
| 04:10.19 | Silik0n | change enabled codecs on your phone? |
| 04:10.27 | pigpen | but, I added 1 sipura... |
| 04:10.42 | Silik0n | sips have g729 |
| 04:10.54 | `Sauron | Hum. |
| 04:11.06 | SPoon_TSX | Hello there, I got some problem with my SIP Phone. I can call out no problem but I got drop call after the first few seconds. And I got the message from the Asterisk as below: |
| 04:11.11 | pigpen | maybe the g729 codecs are in use...and mine cannot connect? |
| 04:11.18 | SPoon_TSX | <PROTECTED> |
| 04:11.23 | TheSin | http://pastebin.ca/8460 |
| 04:11.31 | `Sauron | why does "show translation" show that going TO ilbc from anything costs > 20ms, while going FROM ilbc to anything is 5-9ms? |
| 04:11.33 | SPoon_TSX | My rxgain=2.5/txgain=-1.0 |
| 04:11.36 | Silik0n | well if you are doing 729 to nuphone and you are just passing it thru you dont need a lic |
| 04:11.44 | *** join/#asterisk david (~dcoulson@tawny.nacs.net) |
| 04:11.45 | SPoon_TSX | Any idea? |
| 04:11.46 | david | hello |
| 04:11.53 | pigpen | Silik0n: I am actually doing iax to nufone... |
| 04:12.19 | *** join/#asterisk libpcp (libpcp@210.16.20.5) |
| 04:12.19 | Shido6 | im on a mission |
| 04:12.23 | Silik0n | iax has nothing to do with how the voice is encoded |
| 04:12.23 | libpcp | hi all |
| 04:12.24 | david | I'm seeing signicifant latency with meetme, but not with anything else |
| 04:12.27 | Shido6 | to find out what wireless headsets they use on 24 |
| 04:12.29 | pigpen | but I get this if I try to call voicemail: |
| 04:12.31 | pigpen | Mar 29 22:12:26 NOTICE[3078]: channel.c:1691 ast_set_write_format: Unable to find a path from gsm to g729 |
| 04:12.31 | pigpen | Mar 29 22:12:26 WARNING[3078]: file.c:779 ast_streamfile: Unable to open vm-login (format g729): No such file or directory |
| 04:12.31 | pigpen | Mar 29 22:12:26 WARNING[3078]: app_voicemail.c:3347 vm_execmain: Couldn't stream login file |
| 04:12.31 | pigpen | <PROTECTED> |
| 04:12.33 | david | are there any known latency issues specificly with meetme? |
| 04:12.53 | Shido6 | do u have licenses for g729? |
| 04:12.54 | Shido6 | if not |
| 04:12.58 | Shido6 | its not gonna work |
| 04:13.00 | pigpen | I have 4... |
| 04:13.10 | pigpen | so I guess I will force my phone to G711 |
| 04:13.18 | pigpen | for now until I buy more... |
| 04:13.30 | TheSin | BTW Silik0n I also see... |
| 04:13.31 | TheSin | Mar 29 21:13:21 WARNING[3629]: chan_zap.c:7143 zt_pri_error: PRI: received SETUP message for call that is not a new call, wicked!!! |
| 04:13.42 | TheSin | love the asterisk warn/err msgs :D |
| 04:13.49 | Silik0n | TheSin are you sure you have the right switch type? |
| 04:13.56 | TheSin | I've tried them all |
| 04:14.04 | TheSin | on cisco I used ni1 |
| 04:14.08 | Silik0n | whats the ciscoconfigs like? |
| 04:14.10 | TheSin | but I'm sure it was ni2 |
| 04:14.19 | Silik0n | ok ni1 is ni != ni2 |
| 04:14.19 | TheSin | but cisco doesn't have ni2 |
| 04:14.20 | [hC] | Hm. it seems possible that it could be the plantronics m2500 coupled with the hl10 handset lifter |
| 04:14.26 | SPoon_TSX | Hello everyone. for some reason I got all my incoming call dropped on excatly 3 seconds. |
| 04:14.33 | SPoon_TSX | Any idea? |
| 04:14.37 | Silik0n | ni1is ni1 national on asterisk is ni2 |
| 04:14.47 | TheSin | yup |
| 04:14.52 | TheSin | and I tried both |
| 04:14.57 | TheSin | and dms100 just incase |
| 04:15.02 | TheSin | that is what I used in ONT |
| 04:15.06 | TheSin | I'm in AB now |
| 04:15.36 | Silik0n | beats me...those configs look right unless they are doingsomethingscrewy with the pri |
| 04:16.09 | TheSin | all the ztcfg, zttest and ztmonitor stuff looks great too |
| 04:16.22 | Silik0n | oh did you set a spanmap? |
| 04:16.35 | TheSin | when I dial out I get an error from my telco saying the number I dialed couldn't be completed as dialed |
| 04:16.43 | TheSin | no do I need to? |
| 04:16.54 | Silik0n | try that |
| 04:17.07 | TheSin | what shoudl it be 1,1,1? |
| 04:17.11 | Silik0n | about like 175 of your zapata.conf |
| 04:17.22 | Silik0n | 1,1,1 should work fine or 1,1,0 |
| 04:18.01 | TheSin | Mar 29 21:17:53 WARNING[3732]: loader.c:345 ast_load_resource: chan_zap.so: load_module failed, returning -1 |
| 04:18.02 | Silik0n | or do trunkgroup right above that... spanmapreally shouldntbeneeded |
| 04:18.09 | TheSin | for both 1,1,1 and 1,1,0 |
| 04:18.34 | Silik0n | i dunno it should just work as you had it |
| 04:18.43 | Shido6 | http://www.iguarddirect.com/selection.htm |
| 04:18.50 | Shido6 | not quite a headset |
| 04:19.27 | [hC] | http://wireless.engadget.com/entry/1234000083032441/ |
| 04:19.48 | [hC] | This so far (GN Netcom 6210) is the only one ive found thats bluetooth, but has a cradle so you can connect it to the headset jack of the phone |
| 04:20.15 | Shido6 | 200 bucks for the headset |
| 04:20.26 | TheSin | Silik0n, could it be perms on /dev/zap ? |
| 04:20.47 | [hC] | I dont see anything about headsets on iguarddirect.com, heh |
| 04:20.53 | Silik0n | doubt it |
| 04:21.03 | TheSin | asterisk only has read perms to it |
| 04:21.09 | TheSin | and execute not write |
| 04:21.19 | Shido6 | Silik0n ? |
| 04:21.24 | [hC] | I guess the GN netcom is the only one that is bluetooth with the actual headset jack ability |
| 04:21.30 | [hC] | as opposed to just a plain old headset. |
| 04:21.34 | Silik0n | ? |
| 04:21.39 | Silik0n | Shido6 yes |
| 04:21.51 | Shido6 | the headsets are 200 |
| 04:22.02 | Silik0n | i wasnt talking to you |
| 04:22.11 | Silik0n | i was answering a question from above your comment |
| 04:22.21 | Shido6 | ok |
| 04:22.23 | Shido6 | donut? |
| 04:25.10 | FuriousGeorge | if the box i have asterisk on is doing the QoS stuff i wouldnt need a voip router, right? but i still need a firewall b/w and the net, no? wouldnt it be bad to have my box do QoS, Firewall, and be a pbx? |
| 04:25.23 | `Sauron | A "voip router" ? |
| 04:25.27 | *** join/#asterisk PTG1234 (~PTG123@ip68-106-24-139.ph.ph.cox.net) |
| 04:25.30 | FuriousGeorge | u know what i mean |
| 04:25.51 | `Sauron | I don't. |
| 04:26.14 | `Sauron | but my box is a firewall, router and * server |
| 04:26.25 | FuriousGeorge | sorry, linksys and the bunch make routers with modular rj11 jacks for home use. i assume that it does VoIP QoS, or it would be pretty useless |
| 04:26.30 | *** part/#asterisk Half_Dome (~jelway@mail.westmarkinc.com) |
| 04:26.55 | FuriousGeorge | i giess it wouldnt be useless, b/c most people dont want to buy themselves an FXO |
| 04:27.19 | PTG1234 | yes |
| 04:27.25 | PTG1234 | the qos is gonna be the best on it |
| 04:28.14 | FuriousGeorge | `Sauron: so you box is at the head of the internet. i have a buddy who was telling me my server shouldnt be the firewall b/c "it takes all the crap" |
| 04:28.20 | FuriousGeorge | how do i keep making stuff bold |
| 04:28.22 | FuriousGeorge | lol |
| 04:28.30 | `Sauron | I love how everybody talks about QoS across the internet-at-large, when the whole concept is nothing but a huge oxymoron. |
| 04:28.38 | FuriousGeorge | anyway, this buddy knows much more about *nix than me. so i took his word for it |
| 04:28.56 | *** join/#asterisk Sedorox (brandon@Neptune.client.wlmsprt.pa.sed6.net) |
| 04:29.15 | *** join/#asterisk heison (~heison@CPE000a01d49e6f-CM00111a59bec2.cpe.net.cable.rogers.com) |
| 04:29.20 | FuriousGeorge | `Sauron, im not taling about over the internet at large, im taking about out from my network to the internet. what made you think otherwise |
| 04:30.03 | `Sauron | George: If you leave your network, it has to go somewhere. This place is on the internet-at-large, where you have absolutely no control over any quality of anything... |
| 04:30.06 | Newbie___ | using agi-egate , how do i make * dial a desire provider like bv ? |
| 04:30.34 | `Sauron | Sure, you can fiddle the TOS bits, and sure you can tweak your packet size, but really... There's nothing you can do. |
| 04:30.46 | dmccollum | QoS works well with a MPLS network between offices. QoS also does help a bit over the internet since it put the outgoing packets in priority. |
| 04:30.48 | `Sauron | s/nothing/extremely little. |
| 04:31.08 | `Sauron | dmccollum: And how many ISP's do you know that honor end-user TOS bits? |
| 04:31.16 | FuriousGeorge | `Sauron: i realise that, but surely you would not recommend me run a business IP PBX w/ no QoS |
| 04:31.27 | FuriousGeorge | end user TOS bits. i gotta remember that |
| 04:31.35 | FuriousGeorge | for when i shor around |
| 04:32.06 | pigpen | ok..looks like I need to buy some more codecs...on to the next issue ) |
| 04:32.27 | dmccollum | not many ISP's do anything with the QoS, but you can still help the quality a bit by controlling what goes out on your end. |
| 04:32.35 | `Sauron | Sigh. |
| 04:33.16 | `Sauron | Y'all do whatever you think is best. I'll venture a guess that the first edge router your packets get to, tosses out your TOS bits, and dumps your traffic in with the rest of the bulk traffic. |
| 04:34.29 | `Sauron | QoS on the internet is a joke, unless you own the company that runs the backbone. End of story. |
| 04:34.56 | pigpen | I have deployed an fxo...dialing that number, the fxo (sipura 3000) passes the call to asterisk on extention 99...which passes it to my extention... |
| 04:35.16 | pigpen | once to voice mail...all works fine...voice is recorded... |
| 04:35.30 | pigpen | However, when I pick up the call...no voice either direction.... |
| 04:35.39 | pigpen | ideas? |
| 04:39.47 | Mavvie | `Sauron: own pipes rule :-) |
| 04:40.21 | `Sauron | Mavvie: yup |
| 04:40.26 | Mavvie | owning pipes too, but that's a little bit too low-tech for me :-) |
| 04:40.48 | Silik0n | laying pipe rules |
| 04:40.48 | FuriousGeorge | `Sauron: how much overhead do u get from running your firewall on your asterisk box |
| 04:41.09 | `Sauron | None that I can notice. |
| 04:41.16 | FuriousGeorge | `Sauron: and how can someone as knowledgeable as you not have heard of the VoIP routers. |
| 04:41.26 | FuriousGeorge | or do u just callem something else |
| 04:41.41 | pigpen | sip proxy? |
| 04:41.43 | Silik0n | VoIP routers? is that like a broadband riuter w/ a built in ATA? |
| 04:41.50 | FuriousGeorge | yup |
| 04:42.00 | pigpen | iptables is very light... |
| 04:42.00 | FuriousGeorge | you get one with the proprietary services |
| 04:42.14 | Silik0n | PF rocks tho |
| 04:42.18 | fugitivo | fxs? |
| 04:42.22 | *** join/#asterisk argos73 (~mike@65-85-207-125.client.dsl.net) |
| 04:42.36 | fugitivo | Silik0n: i agree |
| 04:42.44 | pigpen | so my issue above....think it is a sipura issue? |
| 04:42.52 | Silik0n | transparent bridge filtering ++ |
| 04:43.04 | Silik0n | with statfeful inspection and shaping |
| 04:43.08 | FuriousGeorge | my folks got aTT callvantage, and i disconnected the POTS line from their NID and just took some cat 3 from the rj-11 port on the thing and put it in a wall jack, and voila |
| 04:43.27 | Shido6 | voip routers? |
| 04:43.31 | FuriousGeorge | lol |
| 04:43.32 | Silik0n | and normalization and mix nat w/ that on one bridge interface from a 3rd NIC |
| 04:43.37 | fugitivo | and the syntax of pf, is just clear |
| 04:43.46 | Shido6 | i think I have an email from a wholesaler |
| 04:43.49 | Silik0n | yeah it is |
| 04:44.01 | Silik0n | dhartmeier is a genious |
| 04:44.14 | Silik0n | but pf's syntax was lifted from IPF |
| 04:44.19 | fugitivo | if you look a pf.conf file, it's art |
| 04:44.29 | Silik0n | fugitivo you should see mine ;) |
| 04:44.30 | FuriousGeorge | shido6: what do you call a router with an rj11 port on the back which is designed for viop services |
| 04:44.51 | fugitivo | Silik0n: :) |
| 04:44.54 | *** join/#asterisk mick_hastings (~mick_hast@61.194.94.123) |
| 04:45.06 | pigpen | residential equipment? |
| 04:45.37 | FuriousGeorge | not bad, but i still think voip router is more descriptive |
| 04:45.52 | Shido6 | u call it a linksys |
| 04:45.53 | *** join/#asterisk yaboo (~jsirucka@220.245.131.131) |
| 04:45.58 | `Sauron | Umm. |
| 04:46.00 | Silik0n | its more of a router w/ integrated ATA |
| 04:46.04 | `Sauron | It's a router with an FXS port |
| 04:46.13 | `Sauron | as such, it's... yeah |
| 04:46.16 | pigpen | exactly... |
| 04:46.19 | FuriousGeorge | silik0n: still doesnt roll off the toungue just right |
| 04:46.37 | *** join/#asterisk |neuro| (~|neuro|@212.176.51.231) |
| 04:46.48 | fugitivo | i have one of those things, but i call it fxs |
| 04:46.48 | Silik0n | yeah and "NIC Card" does right? |
| 04:46.53 | Silik0n | hah |
| 04:47.12 | mick_hastings | hi folks |
| 04:47.18 | IQ | hi mick_hastings |
| 04:47.27 | Shido6 | do you need one? |
| 04:47.40 | mick_hastings | can anyone tell me how to patch the stable 1.0.7 so I can use forcegreetings in voicemail? |
| 04:47.55 | `Sauron | Sigh |
| 04:47.55 | *** join/#asterisk jskcr|lappy (~jskcr@jskcr.user) |
| 04:47.56 | Shido6 | PAP2-NA |
| 04:48.06 | ManxPower | mick_hastings: read the asterisk-cvs mailing list to see what change was made to it |
| 04:48.07 | Shido6 | I get em for 52.50 |
| 04:48.08 | fugitivo | that one |
| 04:48.10 | Silik0n | people saying "NIC Card" annoys me |
| 04:48.12 | `Sauron | George: Call it whatever you want, I don't care. |
| 04:48.12 | fugitivo | works fine with asterisk |
| 04:48.17 | `Sauron | and yes, I've heard of them |
| 04:48.20 | Shido6 | or 200 for 10,500 |
| 04:48.29 | Silik0n | PAP2s are nice |
| 04:48.41 | fugitivo | great design |
| 04:48.43 | Shido6 | u need one? |
| 04:48.45 | FuriousGeorge | silikon: why? b/c its redundant? what about just NIC |
| 04:48.46 | fugitivo | silver |
| 04:48.48 | Silik0n | i gotta few |
| 04:48.50 | fugitivo | with blue leds |
| 04:48.56 | `Sauron | sexy |
| 04:49.05 | Silik0n | FuriousGeorge because its redundant redundant |
| 04:49.11 | `Sauron | Why is it that anything with blue led's is automatically sexy? |
| 04:49.17 | Silik0n | thats like "PIN Number" |
| 04:49.21 | fugitivo | because blue leds are sexy |
| 04:49.37 | fugitivo | we're tired of red and green leds |
| 04:49.39 | FuriousGeorge | personally, im more of a sucker for gui's w/ x-parency |
| 04:49.43 | FuriousGeorge | liek KDE's |
| 04:49.52 | mick_hastings | Thanks Manx |
| 04:49.55 | mick_hastings | but |
| 04:50.16 | mick_hastings | do you mean modifying the source? |
| 04:50.23 | mick_hastings | manually? |
| 04:50.37 | *** part/#asterisk |neuro| (~|neuro|@212.176.51.231) |
| 04:52.32 | *** join/#asterisk denon (denon@synapse.subneural.net) |
| 04:52.33 | *** mode/#asterisk [+o denon] by ChanServ |
| 04:53.18 | FuriousGeorge | u guys know what i hate more than anything in the world? those autoattendants that respond to voice instead of hitting buttons. is that available to asterisk? |
| 04:53.37 | fugitivo | voicexml? |
| 04:53.44 | pigpen | isn't that festival? |
| 04:53.51 | fugitivo | no, festival is a tts |
| 04:53.55 | FuriousGeorge | pigpen: dont know but sounds fun |
| 04:53.59 | fugitivo | he means speech recognition |
| 04:54.15 | FuriousGeorge | i do |
| 04:54.27 | `Sauron | IF you hate it so much, why do you want it? |
| 04:54.28 | ManxPower | mick_hastings: Of course. Official stable 1.0.x does NOT get new features |
| 04:54.28 | pigpen | shit...don't listen to me...I can't even get this dam spa 3000 working... |
| 04:54.33 | `Sauron | Jealous? :) |
| 04:55.10 | fugitivo | FuriousGeorge: search for sphinx-4, maybe you can do something |
| 04:55.36 | FuriousGeorge | `Sauron: im gonna get my pound of flesh from all those other basterds who have one. Seriously, I dont want one, but they seem to be the new fad so i was wondering if it worked w/ * |
| 04:56.02 | FuriousGeorge | fugitivo: thanks |
| 04:56.04 | pigpen | dam George...you are Furious... |
| 04:56.08 | mick_hastings | ManxPower: I thought that was why they invented patches? Im trying to find the info on asterisk-cvs now |
| 04:56.17 | FuriousGeorge | pigpen: thanks for noticing |
| 04:56.23 | pigpen | :) |
| 04:57.00 | fugitivo | FuriousGeorge: it's a good technology, those who can't see, can surf the web and read mails |
| 04:57.18 | fugitivo | FuriousGeorge: from a phone |
| 04:58.07 | FuriousGeorge | fugitivo: thats certainly a noble cause. i just hate talking to machines. kinda ironic, huh |
| 04:58.19 | FuriousGeorge | of course im not blind, if i were im sure it would be different |
| 04:58.33 | FuriousGeorge | i didnt even start leaving voicemails till i was like 17 |
| 04:58.40 | MrBelvedr | if anyone wants to help me as a paid consultant for a few hours work please pm me |
| 04:59.08 | Sedorox | depends what kinda work :-p |
| 04:59.18 | mick_hastings | OK, now this sounds stoopid but where is asterisk-cvs mailing list? |
| 04:59.21 | pigpen | I don't do windows... |
| 04:59.59 | FuriousGeorge | this is me on the phone with a voice recognition auto attendent: VRAA: "I'm sorry, I didnt get that. Please say..." "NO... OPERATOR... CUSTOMER SUPPORT" |
| 05:00.20 | FuriousGeorge | (then sound of me mashing keypad with hand) |
| 05:01.05 | fugitivo | well, it's not perfect, but it works, i'm waiting for the day where we can talk to computers like jean luc picard talk with the enterprise |
| 05:01.50 | FuriousGeorge | Fugitivo: im sure there will be a time in the not too distant future where they work well enough that i get pissy when they arent around |
| 05:02.19 | FuriousGeorge | OPERATOR! (sorry, flashback) |
| 05:02.26 | fugitivo | well, like all technology, you have to use it to make it better |
| 05:03.19 | FuriousGeorge | fugitivo: like i have a choice ;) ur right, i know |
| 05:03.58 | FuriousGeorge | i was reading somewhere they can judge emotion now |
| 05:05.06 | FuriousGeorge | wonder how... if keypad=mashed then caller=irrate else connect to operator |
| 05:05.46 | pigpen | ok..so does anyone have experience with the Sipura SPA 3000 who can lend me an opinion? |
| 05:06.09 | IQ | pigpen: best ATA ever |
| 05:06.14 | pigpen | cool. |
| 05:06.35 | pigpen | I have it setup to ring an extention...but I get no audio |
| 05:06.45 | pigpen | If I leave a message..audio works fine. |
| 05:06.50 | IQ | pigpen: using with * ? |
| 05:06.56 | pigpen | yeah. |
| 05:07.04 | fugitivo | pigpen: softphone? |
| 05:07.14 | pigpen | no. |
| 05:07.17 | IQ | works great for me... on all my 3 * machines and even SIP Service Providers. |
| 05:07.47 | pigpen | mind looking at my config? |
| 05:08.29 | IQ | sure |
| 05:08.34 | FuriousGeorge | this is somwhat off topic, but has anyone used a bluetooth headset with a computer. |
| 05:09.14 | IQ | pigpen: all you need is SIP Proxy, user name, password |
| 05:09.16 | FuriousGeorge | obviously w/ a softphone |
| 05:11.09 | FuriousGeorge | guess not, well it doesnt sound so hot, sometimes (ive tried two brands and several windows boxes) when it connects, all u get is static, and sometimes they just stop working |
| 05:11.47 | FuriousGeorge | the point is: im looking for an opinion about audio HW to use w/ softphones |
| 05:13.07 | Sedorox | night |
| 05:14.11 | fugitivo | FuriousGeorge: i use a crappy $5 headset with my crappy notebook sound card |
| 05:14.39 | PTG1234 | FuriousGeorge: USB headsets |
| 05:14.46 | fugitivo | FuriousGeorge: and it's wireless, i can move with my laptop and talk with the sip hone |
| 05:14.49 | FuriousGeorge | fugitivo: i was thinking along the limes of wireless |
| 05:14.53 | PTG1234 | FuriousGeorge: i use bluetooth it works ok, but usb works flawlessly |
| 05:15.16 | FuriousGeorge | ptg123: usb wired right? it does work well |
| 05:15.23 | PTG1234 | FuriousGeorge: yes |
| 05:15.34 | PTG1234 | FuriousGeorge: my bluetooth headset all have their qwirks... i have 3 of them |
| 05:16.04 | FuriousGeorge | i wouldnt mind, but the people who need to use themn arent as tech savy as u or i (emphasis on u) |
| 05:16.15 | FuriousGeorge | they dont do well with qwirks |
| 05:16.53 | PTG1234 | Yah so thats why usb is the best way to go |
| 05:16.58 | PTG1234 | alot better then using an internal sound card |
| 05:17.00 | *** join/#asterisk mitmit (~mitmit@CPE000c418b4787-CM001225401fee.cpe.net.cable.rogers.com) |
| 05:17.45 | pigpen | fugitivo: softphone...no trying to setup a spa 3000 as fxo |
| 05:18.12 | mitmit | hi, i am new to asterisk and i am setting up my own home server, any good VOIP termination for long distance suggest? thanks |
| 05:18.26 | DEEZED | i use iax.cc |
| 05:18.41 | dmccollum | I've been very happy with Vonage. Very good quality. |
| 05:18.45 | Silik0n | i use asterlink and nuphone |
| 05:18.58 | pigpen | I use * and nufone |
| 05:19.03 | FuriousGeorge | ptg1234: do they make some good usb handsets with the curly wire for easier swing aroundedness |
| 05:19.45 | mitmit | thanks -> DEEZED...dmccollum...Silikon,,,will check.....tanks again |
| 05:19.56 | PTG1234 | FuriousGeorge: um not that i know of.. plantronics makes some good ones (use the gamer series), logitech makes one decient one, and I got my wife a SennHeiser which is pretty awesome |
| 05:20.07 | mitmit | thanks pigpen |
| 05:21.01 | FuriousGeorge | ptg1234: but did you ever think about that? i maen when have you seen a telephone w/o a curly cord? |
| 05:21.10 | PTG1234 | i hate curly cords |
| 05:21.14 | PTG1234 | they get all twisted :) |
| 05:21.47 | PTG1234 | recently i have come to a conclusion nothing beats a cisco 7960 though :) |
| 05:22.06 | pigpen | like the 7960 better than polycom? |
| 05:22.24 | FuriousGeorge | curly cords have their benefeits. the biggest one being efficent use of slack] |
| 05:24.14 | twisted | PTG1234, leave me out of this |
| 05:24.34 | PTG1234 | oh the 7960 blows away the polycom, i should know i have both :) |
| 05:24.37 | PTG1234 | twisted: why is that? |
| 05:24.57 | *** join/#asterisk riksta (~rick@81-178-227-242.dsl.pipex.com) |
| 05:25.01 | *** join/#asterisk shuric (alexander@alexander.office.inter-telecom.net.ru) |
| 05:26.44 | MrBelvedr | anybody here using the manager api? i am having a problem when placing outgoing calls. Sometimes the calls go through, sometimes they don't |
| 05:27.19 | MrBelvedr | the CLI output does not give any errors |
| 05:27.26 | MrBelvedr | so it is hard to debug what is going on |
| 05:28.34 | MrBelvedr | i have the latest stable version installed |
| 05:28.42 | *** join/#asterisk PoWeRKiLL (~PoWeRKiLL@l02m-4-165.d4.club-internet.fr) |
| 05:28.56 | Silik0n | set logger.conf to give debugon the console |
| 05:28.59 | PoWeRKiLL | morning ! |
| 05:29.06 | pigpen | Evening! |
| 05:29.54 | MrBelvedr | k |
| 05:30.03 | fugitivo | morning? its whisky time for me |
| 05:30.35 | pigpen | past bedtime for me... |
| 05:30.36 | Corydon76-home | Whisky sounds good |
| 05:30.44 | Corydon76-home | As long as it's Jack |
| 05:31.23 | Corydon76-home | Gotta love the local stuff that's brewed in a dry county... |
| 05:31.45 | fugitivo | jhonny walker |
| 05:31.51 | fugitivo | black label |
| 05:32.58 | PTG1234 | nah not black |
| 05:32.59 | PTG1234 | green |
| 05:33.06 | fugitivo | green? |
| 05:33.07 | PTG1234 | now thats the shit |
| 05:33.10 | PTG1234 | Oh yah |
| 05:33.11 | Newbie___ | anyone has any experience with AGI dial plan ? |
| 05:33.12 | PTG1234 | have you had green? |
| 05:33.23 | PTG1234 | its about $500 a bottle |
| 05:33.23 | fugitivo | didn't know there was green |
| 05:33.29 | PTG1234 | Oh.. like 30 years old aged |
| 05:33.33 | PTG1234 | it is sooooo smoooooth |
| 05:33.41 | PTG1234 | not a big fan of johnny walker |
| 05:33.45 | PTG1234 | but green label is the shit |
| 05:33.49 | PTG1234 | its better then blue |
| 05:34.12 | pigpen | good stress reducer? |
| 05:34.19 | FuriousGeorge | PTG1234: Scotch tastes too much like wood to me. Now burbon on the other hand ;) |
| 05:34.26 | FuriousGeorge | thats one expensive phone btw |
| 05:34.29 | PTG1234 | http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&category=13916&item=6166885020&rd=1&ssPageName=WDVW |
| 05:34.38 | PTG1234 | FuriousGeorge: i agree with you in general, but the green is something different |
| 05:34.46 | PTG1234 | FuriousGeorge: i paid $136 for mine |
| 05:34.52 | fugitivo | #asterisk, where whisky drinkers met |
| 05:34.55 | Newbie___ | ~agi |
| 05:34.56 | jbot | [agi] the Asterisk Gateway Interface... similar to CGI for web applications AGI lets you script call control and access databases using your favorite language. AGI wrappers are available for Python (pyst), Perl (astperl?) and other languages |
| 05:35.07 | FuriousGeorge | too bad i aint bartending anymore, ive never tried the green |
| 05:35.11 | PTG1234 | that guy has no idea what he has |
| 05:35.13 | PTG1234 | maybe i should buy it |
| 05:35.27 | FuriousGeorge | canadian whiskey is pretty good too |
| 05:35.37 | PTG1234 | The four bottles are Green, Blue, Gold, and Black. The Black label has been aged for 12 years. The Green has been aged for 15 years and these is the first time I have ever seen the Green label in my life |
| 05:35.44 | PTG1234 | see he doesn't understand what green is :) |
| 05:35.58 | fugitivo | anyone here knows something about zaptel code? |
| 05:35.59 | FuriousGeorge | FOOL |
| 05:35.59 | PTG1234 | he thinks blue has been aged |
| 05:36.02 | PTG1234 | he is wrong about that |
| 05:36.06 | PTG1234 | and has no idea how long green is aged |
| 05:36.11 | PTG1234 | he thinks gold is older |
| 05:36.12 | FuriousGeorge | fugitivo: zaptel? arent we in #whiskey |
| 05:36.53 | fugitivo | FuriousGeorge: it's #asterisk, but it's funnier after the whiskey :) |
| 05:37.28 | FuriousGeorge | lol |
| 05:37.39 | PTG1234 | Mmm |
| 05:37.43 | PTG1234 | man i shoul dbid on that set |
| 05:37.49 | PTG1234 | but with my luck the bottle would be broken before i get it |
| 05:38.44 | FuriousGeorge | ptg1234: i didnt know you could buy booze on ebay |
| 05:39.06 | fugitivo | PTG1234: this is luck, breaking my laptop lcd when closing it with little headphones in the keyboard |
| 05:39.32 | fugitivo | *crack*, and no more lcd :) |
| 05:39.40 | FuriousGeorge | that really burns |
| 05:39.43 | PTG1234 | fugitivo: my lcd broken on my notebook a month ago |
| 05:39.53 | PTG1234 | it was ok though, was an excuse to buy a new one.. which is AWEEESOME :) |
| 05:40.33 | fugitivo | PTG1234: heh, how did you break it? |
| 05:40.40 | PTG1234 | fugitivo: closed it :) |
| 05:40.48 | PTG1234 | fugitivo: i guess enough times and it brokens ome ribbon cable |
| 05:40.51 | PTG1234 | that cost like $300 |
| 05:40.51 | PTG1234 | :) |
| 05:40.56 | fugitivo | ouch |
| 05:40.59 | PTG1234 | stupid design if you ask me |
| 05:41.02 | PTG1234 | ribbon cables dry out |
| 05:41.04 | PTG1234 | then snap |
| 05:41.20 | fugitivo | i don't close it anymore |
| 05:41.30 | *** join/#asterisk t0p (t0p@tech-mgr.chatri.com) |
| 05:41.44 | PTG1234 | hehe |
| 05:41.49 | PTG1234 | its hard to transport if you don't close it |
| 05:41.55 | PTG1234 | its ok i had a good run with it |
| 05:41.56 | PTG1234 | 2 years |
| 05:42.04 | *** join/#asterisk clive- (~pirch@rndf-146-44-91.telkomadsl.co.za) |
| 05:42.12 | PTG1234 | now i got this pentium M 2ghz w/ a 1920x1200 display, bluetooth, suiper fast hard drive, 1gig ram |
| 05:42.17 | PTG1234 | its the speed of a p4 4ghz |
| 05:42.27 | PTG1234 | counter-strike plays great at 1920x1200 :) |
| 05:42.33 | PTG1234 | and very thin |
| 05:42.47 | fugitivo | that's power |
| 05:43.26 | fugitivo | this one is athlon64 3400+ 1400x1050, fast harddrive, 1gb |
| 05:43.55 | PTG1234 | yah sucks the battery life doesn't it :) |
| 05:43.56 | PTG1234 | and thick |
| 05:43.56 | PTG1234 | heh |
| 05:44.05 | fugitivo | 3 hours, it's ok |
| 05:44.06 | PTG1234 | i get a good 6 hours of battery life from mine :) |
| 05:44.11 | fugitivo | 6??? |
| 05:44.15 | PTG1234 | yah battery life and size were my main concerns |
| 05:44.17 | PTG1234 | yah 6 :) |
| 05:44.21 | PTG1234 | pentium M baby |
| 05:44.34 | fugitivo | low power, right? |
| 05:44.37 | PTG1234 | this display is insane |
| 05:44.43 | fugitivo | no noise? |
| 05:44.44 | PTG1234 | yah its a very efficent chip |
| 05:44.46 | PTG1234 | no noise |
| 05:44.47 | PTG1234 | no fans :) |
| 05:44.51 | PTG1234 | it has a fan |
| 05:44.54 | fugitivo | that's great |
| 05:44.55 | PTG1234 | kicked on twice that i can remember |
| 05:44.56 | PTG1234 | heh |
| 05:45.00 | fugitivo | what video card? |
| 05:45.14 | PTG1234 | ati 9700 |
| 05:45.18 | PTG1234 | how about yours? |
| 05:45.23 | fugitivo | 9600 |
| 05:45.35 | fugitivo | that's the noise part, right? |
| 05:45.38 | PTG1234 | no |
| 05:45.41 | PTG1234 | that has no fan i don't think |
| 05:45.45 | PTG1234 | just the main cpu fan |
| 05:45.47 | PTG1234 | if it gets too hot |
| 05:45.57 | fugitivo | i think mine has, when i use 3d, i can hear it |
| 05:46.19 | pigpen | PTG1234: hey...would it be a Dell D800? |
| 05:46.20 | fugitivo | but its amd, and its 64bits, its hot and noisy :) |
| 05:46.40 | PTG1234 | yah |
| 05:46.44 | PTG1234 | i opted away from amd for that |
| 05:46.48 | PTG1234 | plus size and battery sucking :) |
| 05:46.51 | PTG1234 | um nope a sager |
| 05:46.59 | PTG1234 | its the only notebook i could find with such a huge display too |
| 05:47.02 | pigpen | cool..where does the integrated bluetooth adaptor go? |
| 05:47.07 | Faithful | must you have a sound card installed to use music on hold? |
| 05:47.09 | PTG1234 | 1920x1200 on this little screen is just about perfect |
| 05:47.18 | PTG1234 | pigpen: yah bluettoh and 802.11g |
| 05:47.27 | PTG1234 | Faithful: no |
| 05:47.31 | fugitivo | PTG1234: you need to get a sb audigy zs pcmcia |
| 05:47.42 | fugitivo | PTG1234: and you're done |
| 05:47.51 | pigpen | cool..where does the bluetooth adaptor go...as in I have the adaptor for mine...but I don't know where it goes? :) |
| 05:48.10 | PTG1234 | fugitivo: heh i have a dedicated game machine, so i don't really play games on this |
| 05:48.13 | PTG1234 | i just like the idea i can |
| 05:48.27 | PTG1234 | pigpen: its underneath a panel on the bottom |
| 05:48.29 | PTG1234 | you unscrew i think |
| 05:48.35 | PTG1234 | it came installed |
| 05:48.42 | pigpen | 3 panels..ah.. |
| 05:48.57 | Faithful | I can't wait to get my bluetooth working |
| 05:48.57 | pigpen | ok..I will probably need to call dell...I have searched and searched... |
| 05:49.47 | PTG1234 | yah bluetooth works well |
| 05:49.53 | PTG1234 | no decient bluetooth mouse i can find though |
| 05:50.18 | pigpen | kensington has a nice small one...with an off switch.. |
| 05:50.31 | pigpen | PioltMouse Mini Bluetooth |
| 05:50.38 | PTG1234 | i guess i should have said i only like logitechs :) |
| 05:50.42 | pigpen | Model 72414 |
| 05:50.42 | Faithful | anyone got the jabra bt200 working yet? |
| 05:50.45 | PTG1234 | i want that laser mouse |
| 05:50.54 | PTG1234 | i got the perfect wireless mouse, its tiny by logitech |
| 05:50.56 | pigpen | I got a logitech bluetooth too... |
| 05:50.57 | PTG1234 | but needs this dongle |
| 05:51.05 | fugitivo | PTG1234: what os have you intalled? |
| 05:51.06 | PTG1234 | Faithful: i run a bt200 on this thing |
| 05:51.20 | pigpen | ok..one last try on this dam sipura 3000.... |
| 05:51.24 | PTG1234 | fugito: xp, with freebs drunning in vmware, ifacing through humingbird exceed |
| 05:51.42 | fugitivo | PTG1234: pentium m is still 32bit, right? |
| 05:51.43 | Faithful | PTG1234: and you have it set up as an extension? |
| 05:52.23 | PTG1234 | Faithful: extension what do you mean? |
| 05:52.32 | PTG1234 | fugitivo: no idea, it is fast though :) |
| 05:52.40 | Faithful | an extension in * |
| 05:52.41 | PTG1234 | i don't go into that 64bit hype |
| 05:52.53 | fugitivo | linux runs great in 64bit |
| 05:52.56 | pigpen | I am running gentoo linux on mine... |
| 05:53.02 | fugitivo | me too |
| 05:53.07 | pigpen | ok...spa 3000 didn't work...I give up. |
| 05:53.17 | pigpen | fugitivo: you gentoo ? |
| 05:53.32 | fugitivo | i started working with opteron servers and gentoo, then i bought this 64bit laptop just to use gentoo :) |
| 05:53.37 | fugitivo | pigpen: yep |
| 05:53.44 | pigpen | cool...amd I see... |
| 05:53.46 | MrBelvedr | thanks silk |
| 05:53.53 | pigpen | my business partner is a kern dev for gentoo... |
| 05:53.58 | Faithful | PTG1234: what are you using the bt200 for ? |
| 05:54.25 | Shido6 | is it a ferrari, fugitivo |
| 05:54.28 | Shido6 | ? |
| 05:54.38 | fugitivo | Shido6: no, hypersonic ax6 |
| 05:54.40 | PTG1234 | Faithful: softphone |
| 05:55.01 | fugitivo | everything works, except the memory stick reader |
| 05:55.04 | MrBelvedr | i have narrowed it down. If I start up asterisk and originate a call using the Manager API the first call is placed perfectly. Every call after the first one looks like it goes out ok on the CLI, but in reality the phone being called never rings |
| 05:55.15 | PTG1234 | everything works in xp for me, including the camera :) |
| 05:55.18 | Faithful | PTG1234: o, with your pc? |
| 05:55.19 | PTG1234 | it has a video camera in it |
| 05:55.27 | fugitivo | PTG1234: built in? |
| 05:55.35 | PTG1234 | yah |
| 05:55.36 | PTG1234 | let me show you |
| 05:55.57 | PTG1234 | http://www.sagernotebook.com/pages/notebooks/product.cfm?ProductType=3790 |
| 05:56.00 | fugitivo | PTG1234: well, it's obvious everything is going to work in xp, not in linux :) |
| 05:56.03 | PTG1234 | see the little tiny dot at the top of the screen |
| 05:56.09 | PTG1234 | fugitivo: run liux in vmware :) |
| 05:56.12 | PTG1234 | works great |
| 05:56.12 | PTG1234 | heh |
| 05:56.17 | fugitivo | i don't like windows |
| 05:56.20 | fugitivo | i don't use it |
| 05:56.29 | fugitivo | at all |
| 05:56.49 | *** join/#asterisk djin (~djin@gridfox.xs4all.nl) |
| 05:57.20 | fugitivo | you have 1920x1200 with 15.4??? |
| 05:57.21 | PTG1234 | man |
| 05:57.25 | PTG1234 | yah fug |
| 05:57.28 | PTG1234 | heh |
| 05:57.32 | PTG1234 | its awesome |
| 05:57.39 | fugitivo | isn't it too tiny? |
| 05:57.43 | PTG1234 | not for me |
| 05:57.46 | PTG1234 | i would have it smaller if i could |
| 05:58.45 | fugitivo | 4 speakers |
| 05:59.00 | fugitivo | my speakers sucks |
| 05:59.03 | PTG1234 | th e speakers kind of suck if you ask me |
| 05:59.07 | PTG1234 | yah they all do on notebooks |
| 05:59.08 | PTG1234 | :) |
| 05:59.12 | fugitivo | all notebook speakers sucks :) |
| 05:59.37 | PTG1234 | although i couldn't imagine this notebook having anything else |
| 05:59.40 | PTG1234 | it seems to have everything |
| 05:59.56 | fugitivo | what brand is the integrated webcam? |
| 06:00.03 | PTG1234 | no clue |
| 06:00.07 | fugitivo | intel i suppose |
| 06:00.12 | PTG1234 | i can't even find someone with one to do a teleconferene with |
| 06:00.19 | fugitivo | it's not expensive at all |
| 06:00.45 | fugitivo | my logitech webcam for notebook doesn't work stable with linux :( |
| 06:00.57 | PTG1234 | for manuyfacture |
| 06:00.59 | PTG1234 | it says VM |
| 06:01.17 | PTG1234 | i was told just about everything worked with linux on this one |
| 06:01.19 | PTG1234 | and yah |
| 06:01.21 | PTG1234 | $1800 full loaded |
| 06:01.24 | PTG1234 | is what i paid |
| 06:01.26 | PTG1234 | er fully |
| 06:01.33 | fugitivo | it's a great price |
| 06:02.20 | PTG1234 | yah i was happy :) |
| 06:02.25 | PTG1234 | made it ok my old one bit the dust |
| 06:02.33 | fugitivo | i have the same wireless card |
| 06:02.42 | fugitivo | i had some problems with old kernel versions |
| 06:02.55 | fugitivo | but now it works better perfect |
| 06:03.58 | PTG1234 | yah |
| 06:04.04 | PTG1234 | i had some problems with earlier xp drivers |
| 06:04.07 | PTG1234 | with my wireless card |
| 06:04.19 | fugitivo | Video Processor Colling Vents |
| 06:04.28 | fugitivo | check the gallery |
| 06:05.02 | fugitivo | you have 2 fans, cpu and video |
| 06:07.23 | *** join/#asterisk Rick_Hunter (~rhunter@170.206.250.81) |
| 06:08.32 | PTG1234 | http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&category=369&item=6166983969&rd=1 |
| 06:08.37 | PTG1234 | that must be some good stuff |
| 06:08.45 | PTG1234 | fugitivo: yah i believe it they never come on |
| 06:12.17 | *** join/#asterisk dmccollum (~dmccollum@69-164-245-72.atlaga.adelphia.net) |
| 06:15.48 | Newbie___ | is it possible to dial skype from asterisk box ? |
| 06:16.47 | Newbie___ | nevermind that |
| 06:16.51 | clive- | does anyone know what this means: AGI Script astcc.agi completed, returning 0 |
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| 06:25.33 | yaboo | can a users voicemail number be the same as the users extenstion? |
| 06:25.51 | Shido6 | not quite |
| 06:25.57 | yaboo | ok |
| 06:26.04 | Shido6 | you CAN make it the same |
| 06:26.11 | Shido6 | if the user dials his own extension |
| 06:26.14 | Shido6 | from HIS/HER own phone |
| 06:26.16 | yaboo | so e.g. if there extension is 3004, best to do voicemail like 7004? |
| 06:26.30 | Shido6 | extension is 3000 |
| 06:26.38 | yaboo | ok, but not from another number |
| 06:26.44 | Shido6 | and if he or she dials his OWN extension |
| 06:26.53 | Shido6 | he / she can get their voicemail |
| 06:27.00 | Shido6 | and voicemail will know its them and login |
| 06:27.07 | Shido6 | to their own vmail |
| 06:27.11 | Shido6 | with ANI |
| 06:27.19 | yaboo | ok, by the digit password |
| 06:37.23 | firestrm | is it possible to set the gain of each channel of a tdm400 seperately? |
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| 06:48.54 | *** join/#asterisk CaptChris (~Chris@c-67-181-99-1.hsd1.ca.comcast.net) |
| 06:51.22 | CaptChris | hello all |
| 06:52.38 | CaptChris | i'm looking for some help in configuring Asterisk |
| 06:52.58 | CaptChris | is anyone available to help? |
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| 06:56.01 | Shido6 | sure |
| 06:56.03 | Shido6 | whats up? |
| 06:56.08 | CaptChris | i seem to have Asterisk configured partway |
| 06:56.11 | Shido6 | ok |
| 06:56.18 | Shido6 | what do you want to do with asterisk? |
| 06:56.24 | CaptChris | i can connect to the server from a remote machine using X-Lite... |
| 06:56.29 | Shido6 | good |
| 06:56.33 | Shido6 | (drum roll) |
| 06:56.33 | Shido6 | but |
| 06:56.34 | Shido6 | ? |
| 06:56.44 | CaptChris | but can't seem to connect with X-Lite on the local server machine. |
| 06:57.01 | Shido6 | ok |
| 06:57.03 | CaptChris | ok. yes |
| 06:57.05 | Shido6 | when do you want to get started? |
| 06:58.26 | CaptChris | well... that's not exactly what i was hoping for. |
| 06:59.12 | Shido6 | all you want is sip setup |
| 06:59.13 | Shido6 | ? |
| 06:59.28 | Shido6 | or do you want to go through asterisk dialplan logic, voicemail extensions, setting up routes |
| 06:59.36 | Shido6 | for the complete hour |
| 06:59.37 | CaptChris | i can tell you that X-Lite is configured exactly the same on both the remote and local (server) machine. while only the remote X-Lite can connect |
| 06:59.56 | CaptChris | yes, just sip |
| 06:59.56 | Shido6 | ok |
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| 07:00.41 | CaptChris | without income and without a job, i'm not able to pay |
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| 07:06.42 | *** join/#asterisk kFuQ (~somedude@24.17.224.78) [NETSPLIT VICTIM] |
| 07:06.42 | *** mode/#asterisk [+o denon] by irc.freenode.net |
| 07:11.40 | *** join/#asterisk Qwell (~north@24-50-66-194.vnnyca.adelphia.net) [NETSPLIT VICTIM] |
| 07:14.27 | *** join/#asterisk mbranca (~matteo@81.208.92.210) |
| 07:17.55 | *** join/#asterisk djin (~djin@62.58.40.196) |
| 07:23.51 | *** join/#asterisk pif (ldm@82.66.93.83) |
| 07:26.01 | *** join/#asterisk Verliba (~Miranda@250.1-14-84.ripe.coltfrance.com) |
| 07:26.08 | *** join/#asterisk sd-tux (user2267@emasq.stusta.mhn.de) |
| 07:34.11 | *** join/#asterisk Alexi1 (~alexis@www.trim.it) |
| 07:34.20 | Alexi1 | hello all |
| 07:37.09 | *** join/#asterisk soundguy (~soundguy@zeus.blendtek.com.au) |
| 07:46.20 | *** join/#asterisk MuppetMaster (~MuppetMas@a82-92-73-185.adsl.xs4all.nl) |
| 07:48.23 | *** join/#asterisk dg1nsw (~schulte@gate.sympat.de) |
| 07:51.01 | MuppetMaster | Hello. |
| 07:51.19 | zoa | He L L o |
| 07:51.20 | MuppetMaster | Does anyone know the cost of embedding Asterisk into another application? As to what the license fees are that would need to be paid to Digium? |
| 07:51.33 | zoa | contact sales@digium.com for that |
| 07:51.54 | MuppetMaster | Understood. But no details elsewhere? |
| 07:51.59 | zoa | no |
| 07:52.04 | zoa | i have a good idea |
| 07:52.08 | zoa | <PROTECTED> |
| 07:52.17 | zoa | its not too expensive though |
| 07:52.24 | MuppetMaster | ballpark? |
| 07:52.30 | zoa | ballpark ? |
| 07:52.46 | MuppetMaster | you said you have a good idea, so what is the idea you have? |
| 07:52.57 | zoa | ah im not sure i can tell it (NDA etc) |
| 07:53.00 | tessier | How odd. When I dial 3 on this 7960 asterisk sees 33 |
| 07:53.01 | MuppetMaster | ah, ok |
| 07:53.01 | zoa | so just call them |
| 07:53.13 | tessier | Anyone know what would cause that? |
| 07:53.26 | tessier | I have a feeling I've seen this before but can't recall what the deal was... |
| 07:53.27 | zoa | i think ive seen that before |
| 07:53.34 | zoa | dunno what the solution was |
| 07:53.41 | zoa | i think the old grandstreams also had that |
| 07:53.45 | tessier | That is going to make it rather hard to navigate voicemail. |
| 07:53.50 | tessier | This is a 7960G with the latest firmware. |
| 07:54.02 | tessier | Dialing in over PSTN actually |
| 07:54.12 | tessier | Asterisk is answering the line on a T-1. |
| 07:54.27 | tessier | The problem must be on the asterisk on my end...although I would think someone would have noticed this earlier... |
| 07:54.29 | Faithful | You must have bluez to use bluetooth devices with * ?? |
| 07:55.28 | argos73 | gotta love it when GPL hits NDA... :) |
| 07:55.59 | *** join/#asterisk afrosheen (~afro@c-67-166-172-141.hsd1.tx.comcast.net) |
| 07:56.42 | [hC] | tessier: its because someone has dialed 33 in the past, and your 7960 is autocompleting a previously dilaed number |
| 07:56.50 | [hC] | tessier: press 3, then more, then clear, then dial |
| 07:56.56 | tessier | Oops, this is a 7912. |
| 07:56.56 | [hC] | it'll stop it |
| 07:57.07 | tessier | [hC]: Did it with 9999 too |
| 07:57.16 | tessier | Came out to 9999999 |
| 07:57.25 | [hC] | its probably becuase someone DIALED THAT in the past! |
| 07:57.25 | tessier | And this is not during the dialing phase. |
| 07:57.25 | [hC] | :P |
| 07:57.29 | [hC] | oh |
| 07:57.35 | tessier | I am connected listening to the auto-attendant |
| 07:57.41 | tessier | Then I want to hit 9999 to get connected to the voicemail system |
| 07:57.49 | tessier | And instead it tries to get 9999999 |
| 07:57.54 | argos73 | tessier: can also happen if you're doing digital->analog->digital over an iffy link |
| 07:57.59 | tessier | I bet something is screwy with the dtmf on this phone... |
| 07:58.00 | [hC] | well then i would say your keypad has some damage and/or sticks |
| 07:58.08 | tessier | argos73: Indeed we are doing d-a-d |
| 07:58.33 | afrosheen | so you get 3 extra 9's? |
| 07:58.35 | afrosheen | or 4 |
| 07:58.55 | argos73 | if the codec is breaking due to bandwidth issues, it could cause dupe dtmf |
| 07:59.18 | *** join/#asterisk Zeeek (~icechat5@Zeeek.sustaining.supporter.pdpc) |
| 07:59.23 | tessier | I got 3 extra 9's |
| 07:59.29 | tessier | 9999 becomes 9999999 |
| 07:59.40 | tessier | er.... |
| 07:59.44 | tessier | Yeah, that's it |
| 07:59.54 | tessier | <PROTECTED> |
| 08:00.12 | afrosheen | what happens if you dial 99 |
| 08:00.17 | tessier | Was wondering there for a second why it was dialing out Zap/g1 but that makes sense now. Had it been 9999 it would have been a local extension. |
| 08:00.17 | afrosheen | do you get 99999? |
| 08:01.22 | argos73 | if the pattern is exactly reproducable every time, it could be a coding thing. if it varies, I might suspect codec problems garbling the analog DTMF tones |
| 08:01.42 | afrosheen | yeah try to reproduce it with more or less 9's, see if it's consistend |
| 08:01.46 | afrosheen | woops consistent |
| 08:02.20 | [hC] | hey does anyone know of a bluetooth headset that works with the cisco 7960? |
| 08:02.41 | afrosheen | is it a bluetooth phone? |
| 08:02.49 | argos73 | nope |
| 08:02.52 | [hC] | not natively, no, but it has a headset jack. |
| 08:03.06 | afrosheen | just get a plantronics headset then |
| 08:03.20 | [hC] | Im looking at getting a handset lifter and a bluetooth headset that i presume would be just like a plantronics CS60 that has a cradle |
| 08:03.33 | argos73 | could always rig up a radio smack "bluetooth to 2.5mm headset jack" gizmo |
| 08:03.43 | [hC] | except the cs60 isnt bluetooth, and i would prefer if i could use the same headset when i walk away from my phone, for use with my cell phone. |
| 08:03.56 | [hC] | (cause it could also cound for presence detection) |
| 08:04.29 | afrosheen | man that thing is dorky |
| 08:04.31 | [hC] | i did find one vendor who had one, but the reviews were so-so |
| 08:04.40 | afrosheen | it should come with a Borg Eye you wear on the other side of your head |
| 08:04.48 | zoa | bluetooth ? |
| 08:04.51 | zoa | gn netcom one |
| 08:04.53 | afrosheen | the cs60 |
| 08:04.54 | zoa | is prolly a good one |
| 08:04.58 | argos73 | ever seen a review that isn't so-so??? :) |
| 08:04.59 | [hC] | yeah zoa thats the one i saw |
| 08:04.59 | tessier | When I dial 9 9 I get 9999 |
| 08:05.02 | tessier | So that gets me into voicemial |
| 08:05.04 | MuppetMaster | I find the GN Netcom wireless edition crap. |
| 08:05.08 | *** join/#asterisk djin (~djin@62.58.40.196) |
| 08:05.13 | tessier | But if I enter mailbox 1000 I get 11000000 |
| 08:05.14 | [hC] | i heard the GN one sucked big time |
| 08:05.22 | tessier | And password 1234 I get 11223344 |
| 08:05.23 | afrosheen | tessier: every time? |
| 08:05.24 | [hC] | audio quality wise |
| 08:05.27 | zoa | hmm |
| 08:05.30 | zoa | we dont have that one |
| 08:05.30 | tessier | Yep. Every time. Identical. Reproduceable. |
| 08:05.34 | zoa | but we have the normal one |
| 08:05.38 | zoa | and that one sounds very good |
| 08:05.39 | afrosheen | tessier: ok so you're having d-a-d issues then I imagine, dtmf echo |
| 08:05.48 | tessier | ugh |
| 08:05.53 | afrosheen | yeah I know it sucks |
| 08:06.07 | tessier | Actually, the signal is 100% digital the whole way. No POTS involved. |
| 08:06.13 | tessier | er....wait..that's not true. |
| 08:06.13 | afrosheen | hmm |
| 08:06.16 | *** part/#asterisk MuppetMaster (~MuppetMas@a82-92-73-185.adsl.xs4all.nl) |
| 08:06.19 | argos73 | LONG LIVE ECHO! ECHO! Echo! echo! |
| 08:06.22 | tessier | I think it does go into a pots line and then this number forwards to another... |
| 08:06.32 | tessier | Let me try dialing the direct number. This is way too complicated already. :) |
| 08:06.50 | argos73 | heh |
| 08:07.11 | argos73 | whenever you throw analog conversion into the mix, things get funky |
| 08:07.55 | afrosheen | werd |
| 08:08.28 | tessier | Ok, there is no analog anywhere in the connection. |
| 08:08.34 | tessier | And it still echoes |
| 08:08.45 | [hC] | only with that particular phone, or any phone? |
| 08:08.54 | [hC] | maybe you need to change the method in which you send dtmf |
| 08:09.04 | tessier | From this 7912 via SIP over the Internet into an asterisk box then out a DS-3 to the PSTN then in a T-1 on the other end into asterisk. |
| 08:09.05 | [hC] | do you use sip info, inband or rfc2833? |
| 08:09.17 | argos73 | hmm... if I velcro'd my ipaq to my new fridge, would it make it an "Internet-enabled refrigerator"?? things to ponder... |
| 08:09.24 | afrosheen | should be inband I would imagine |
| 08:09.51 | afrosheen | but rfc2833 works from here at home with my polycom 500 |
| 08:09.59 | *** join/#asterisk DT-V (~sjaaknabu@fia254-108-100.dsl.mxposure.nl) |
| 08:10.13 | [hC] | i prefer rfc2833 whenever possible, it seems the most compatible and reliable so far. |
| 08:10.30 | afrosheen | yeah dtmf inband seems like a hack |
| 08:10.46 | afrosheen | but I've seen some clients like kphone, that's the only way they'll handle it |
| 08:11.22 | argos73 | inband = easy way out... unfortunately, it usually sucks... |
| 08:11.29 | tessier | Let me try it on another phone... |
| 08:11.40 | afrosheen | and you MUST use ulaw with inband as well, or another wasteful codec |
| 08:12.13 | bkw_ | you can usually use inband with g726 also |
| 08:12.14 | afrosheen | anyone have a link to the archives with the meetme delay issue solved? |
| 08:12.24 | bkw_ | afrosheen, it wasn't |
| 08:12.29 | afrosheen | uh oh |
| 08:12.38 | afrosheen | I thought the |q to silence the entry tones fixed it |
| 08:13.05 | tessier | I have 2 7912's here and they both have the same issue |
| 08:13.19 | tessier | [hC]: I am checking how we are sending dtmf...the 7912 config sucks. :( |
| 08:13.30 | afrosheen | bkw_: link? |
| 08:13.38 | tessier | dtmf is probably coded into one of these ridicilous hex strinsg |
| 08:13.52 | *** join/#asterisk scanna (~scannachi@81-174-16-211.f5.ngi.it) |
| 08:15.01 | afrosheen | bkw_: this is what I was reading before http://lists.digium.com/pipermail/asterisk-dev/2005-February/009513.html |
| 08:17.30 | bkw_ | no no XML encoded DTMF |
| 08:17.34 | bkw_ | ya ya thats how we need to start doing it |
| 08:17.40 | bkw_ | OH WAIT sip already has that |
| 08:18.01 | bkw_ | afrosheen, it still wasn't fixed |
| 08:18.38 | BoRiS | bkw! |
| 08:18.54 | BoRiS | Your still up? |
| 08:19.09 | *** join/#asterisk ckruetze (~nospam@i3ED63CFB.versanet.de) |
| 08:19.30 | bkw_ | BoRiS, just makin sure the shit is still workin |
| 08:19.54 | bkw_ | we are currently billing out over 500 calls a min. |
| 08:20.03 | bkw_ | the system is purring along |
| 08:20.04 | afrosheen | from 1.0.5 to 1.0.7 and the meetme delay is still alive huh..arg |
| 08:20.15 | bkw_ | afrosheen, duh hehe |
| 08:20.34 | BoRiS | very nice bkw! |
| 08:20.38 | bkw_ | good god it might catch up now |
| 08:20.42 | bkw_ | sheesh |
| 08:20.46 | zoa | yeah bkw |
| 08:20.47 | tessier | bkw_: So is Michael winning or losing?? |
| 08:20.54 | bkw_ | tessier, loosing |
| 08:20.54 | tessier | Enquiring minds want to know! |
| 08:20.57 | tessier | doh |
| 08:21.00 | bkw_ | GUILTY |
| 08:21.10 | bkw_ | I can sure tell when the episode airs |
| 08:21.12 | tessier | Mike's headed for Federal "POUND ME IN THE ASS" Prison! |
| 08:21.19 | bkw_ | we get these 200-400 call spikes |
| 08:21.21 | zoa | we tested ours with 500 calls / second |
| 08:21.22 | zoa | :) |
| 08:21.44 | BoRiS | all on one server bkw and zoa? |
| 08:21.47 | zoa | our billing keeps up |
| 08:21.50 | zoa | the asterisk doesnt |
| 08:21.54 | afrosheen | lol |
| 08:21.54 | bkw_ | BoRiS, ours is on 7 servers |
| 08:21.57 | bkw_ | 4 T's per server |
| 08:21.59 | zoa | ours on 20 |
| 08:22.08 | afrosheen | hmm what's bkw doing right |
| 08:22.09 | bkw_ | well this cluster is 7 |
| 08:22.13 | bkw_ | we have another with 5 in it |
| 08:22.18 | bkw_ | but thats in LA |
| 08:22.26 | bkw_ | plus two TNT's |
| 08:22.30 | bkw_ | one in LA |
| 08:22.31 | bkw_ | one in VA |
| 08:22.31 | BoRiS | what clustering software? |
| 08:22.35 | bkw_ | BoRiS, ZERO |
| 08:22.36 | zoa | asterisk hates more than 50 calls / second |
| 08:22.38 | bkw_ | you can't cluster asterisk |
| 08:22.40 | bkw_ | silly boi |
| 08:22.53 | bkw_ | hey wanna see a picture of zoa? |
| 08:22.54 | tessier | ah-ha |
| 08:22.59 | tessier | I think the dtmf setup on the phone is wrong |
| 08:22.59 | BoRiS | What about that linux high availabilty kernel patch or something |
| 08:23.06 | tessier | AudioMode:0x00000011 |
| 08:23.19 | bkw_ | http://homepage.mac.com/brian.west/PhotoAlbum9.html |
| 08:23.21 | tessier | wiki says it shold be AudioMode:0x00000020 |
| 08:23.25 | zoa | hehe |
| 08:23.31 | afrosheen | so that's it then |
| 08:23.35 | bkw_ | 100_0235 |
| 08:23.37 | tessier | I bet we were trying to use inband over g729 |
| 08:23.45 | afrosheen | lol |
| 08:23.47 | bkw_ | tessier, smack |
| 08:23.49 | bkw_ | bad idea |
| 08:23.50 | zoa | that would not give double results |
| 08:23.51 | afrosheen | just what I was saying earlier :) |
| 08:23.53 | tessier | I know it's a bad idea. |
| 08:23.56 | zoa | it would give no results |
| 08:23.59 | tessier | The people who setup this phone probably did not |
| 08:23.59 | bkw_ | you can do it |
| 08:24.06 | bkw_ | bu tit works about 10% of the timee |
| 08:24.08 | bkw_ | and not all in a row |
| 08:24.09 | zoa | yeah but it wont work reliably |
| 08:24.28 | afrosheen | tessier: is there an audio mode list somewhere |
| 08:24.33 | bkw_ | I'm sitting here in my underwear... just in case you wanna know that |
| 08:24.33 | zoa | im migrating the callcenter to asterisk only |
| 08:24.40 | BoRiS | LOL! |
| 08:24.45 | Zeeek | bkw_ too much information |
| 08:24.47 | afrosheen | zoa: have you seen the latest AMP |
| 08:24.51 | bkw_ | zoa http://asterisk.bkw.org/congrats.gsm |
| 08:24.56 | BoRiS | Mr twisted in those pics? |
| 08:24.56 | zoa | amp is nothing for me |
| 08:25.01 | tessier | bkw_: I IRC completely buck nekkid all the time |
| 08:25.03 | Qwell | bkw_: These images need names, for the uninformed. :p |
| 08:25.08 | tessier | In fact I might be nekkid right now. |
| 08:25.15 | bkw_ | tessier, bet you are |
| 08:25.16 | tessier | Having an 11th digit helps me type faster |
| 08:25.17 | afrosheen | zoa: I know you're too leet for it or whatever but it's supporting queueing now, added a ton of new features |
| 08:25.19 | bkw_ | now keep both hands on the keyboard please |
| 08:25.24 | tessier | heh |
| 08:25.30 | zoa | i will have a look at it again |
| 08:25.35 | tessier | The keyboards a little sticky... |
| 08:25.37 | zoa | maybe we can use it for some customers |
| 08:25.56 | zoa | lets get this trunking for high volumes to work this week! |
| 08:26.01 | zoa | and the sip jitter buffer of course |
| 08:26.41 | bkw_ | its cute to hear zoa say "jitter buffer" |
| 08:26.59 | bkw_ | hahaha |
| 08:27.14 | riksta | why? :P |
| 08:27.25 | afrosheen | pikachu-voice? |
| 08:27.50 | bkw_ | thats hot |
| 08:27.57 | bkw_ | thats sexy |
| 08:28.09 | bkw_ | ok sexy.. |
| 08:28.12 | bkw_ | love ya bitch |
| 08:28.12 | argos73 | time for a cold shower... :) |
| 08:28.21 | BoRiS | he does! |
| 08:28.21 | afrosheen | bkw_: dumb question, where do I find the conf() stuff at |
| 08:28.30 | bkw_ | afrosheen, for? |
| 08:28.30 | BoRiS | :-p |
| 08:28.34 | tessier | hmm...that does not seem to have fixed it. |
| 08:28.39 | bkw_ | BoRiS, call 996 |
| 08:28.40 | afrosheen | bkw_: fixing the delay business in meetme |
| 08:28.53 | afrosheen | conf_play() |
| 08:28.55 | bkw_ | good luck.. love ya bitch |
| 08:28.58 | bkw_ | afrosheen, no clue |
| 08:29.03 | bkw_ | BoRiS, nm |
| 08:29.06 | bkw_ | can't talk |
| 08:29.10 | BoRiS | bkw: ok, give me a few minutes... |
| 08:29.11 | BoRiS | oh |
| 08:30.02 | zoa | especially when i got drunk again |
| 08:30.22 | zoa | than it sometimes even looks a little like english :/ |
| 08:30.26 | zoa | sounds even |
| 08:30.55 | *** join/#asterisk chapeaurouge (~chap@217.31.73.114) |
| 08:31.20 | *** join/#asterisk PoWeRKiLL (~PoWeRKiLL@l02m-4-165.d4.club-internet.fr) |
| 08:32.52 | bkw_ | haha |
| 08:32.58 | bkw_ | zoa drunk is about as funny as me drunk |
| 08:33.38 | *** join/#asterisk iceyp (~icepick@202.150.105.150) |
| 08:33.53 | iceyp | hey guys, how can I get a 7940 to work with asterisk, I cant get a codec to work |
| 08:34.05 | riksta | iceyp: ulaw |
| 08:34.13 | iceyp | riksta is that the only one i can use? |
| 08:34.20 | iceyp | I generally use gsm or ilbc |
| 08:34.21 | riksta | yeah |
| 08:34.35 | riksta | no you have to use ulaw to the asterisk server, then you can use whatever you want, |
| 08:34.46 | iceyp | ok |
| 08:34.53 | iceyp | so g711ulaw |
| 08:35.01 | iceyp | whats g711alaw |
| 08:35.01 | iceyp | ? |
| 08:35.05 | riksta | it doesn't matter if you use ulaw because it's internal |
| 08:35.11 | iceyp | this is a remote 7940 phone |
| 08:35.13 | iceyp | over the net |
| 08:35.18 | riksta | oh, well you have to use ulaw |
| 08:35.27 | iceyp | damn |
| 08:36.07 | iceyp | takes so long to reboot :/ |
| 08:36.08 | riksta | i'm sure there is a possible workaround other than having an * box there |
| 08:36.10 | riksta | but i don't know of any |
| 08:36.17 | iceyp | does the whole configuring vlan when i dont even use vlans |
| 08:37.28 | riksta | who cares? |
| 08:37.41 | bkw_ | ok I need sleep |
| 08:39.00 | riksta | later |
| 08:40.43 | *** join/#asterisk webmiko (~courtney@59.145.145.126) |
| 08:41.35 | Shido6 | bleh |
| 08:41.43 | iceyp | Mar 30 20:41:28 NOTICE[50495]: chan_sip.c:7313 handle_request: Unable to create/find channel |
| 08:41.45 | Shido6 | 7940s arent that hard to setup |
| 08:41.47 | iceyp | what's this mean |
| 08:41.53 | Shido6 | u have a tftp server setup? |
| 08:41.57 | iceyp | Shido6 nah got it working |
| 08:42.02 | Shido6 | ok |
| 08:42.07 | Shido6 | then you're set |
| 08:42.08 | Shido6 | next |
| 08:42.11 | iceyp | upgrading firmware was a bitch to start |
| 08:42.33 | iceyp | weird thing is it doesnt seem to work from here at work, but then again neighther does my x-lite |
| 08:42.37 | iceyp | probably firewalled |
| 08:42.43 | Shido6 | thought you set you got it working? |
| 08:42.50 | iceyp | though it used to work direct to asterisk, now I use ser as the proxy |
| 08:42.56 | Shido6 | why do you use ser? |
| 08:42.59 | Shido6 | you dont need ser |
| 08:43.06 | Shido6 | connect it directly to asterisk |
| 08:43.07 | iceyp | for thousands of users? |
| 08:43.09 | Shido6 | yes |
| 08:43.17 | iceyp | I use ser for serweb |
| 08:43.20 | iceyp | free voip in NZ |
| 08:43.25 | iceyp | simular to freeworlddialup |
| 08:43.40 | Shido6 | we have a few customers on our asterisk systems |
| 08:43.43 | Shido6 | NuFone |
| 08:43.50 | iceyp | yeah i know ;) |
| 08:44.00 | riksta | yeah, i think everyone knows |
| 08:44.01 | riksta | :P |
| 08:44.05 | riksta | it's all i hear from Shido6 :) |
| 08:44.32 | Shido6 | whats the problem |
| 08:44.44 | Shido6 | where is your asterisk box in relation to your 7940 |
| 08:45.09 | iceyp | Atm i'm connecting firect to the asterisk |
| 08:45.19 | *** join/#asterisk zhier (~nick@219.137.39.14) |
| 08:45.21 | *** join/#asterisk djin (~djin@62.58.40.196) |
| 08:45.25 | Shido6 | ok so the * has a public ip? |
| 08:45.30 | argos73 | hmm - wife's alarm clock goes off in 10 minutes - time to get to bed before I get in trouble! :) |
| 08:45.31 | argos73 | later |
| 08:45.45 | riksta | LOL |
| 08:45.50 | Shido6 | iceyp your asterisk box has a public ip? or behind nat? |
| 08:45.56 | iceyp | public |
| 08:45.58 | Shido6 | ok |
| 08:46.03 | Shido6 | and where is your phone? |
| 08:46.03 | *** part/#asterisk argos73 (~mike@65-85-207-101.client.dsl.net) |
| 08:46.06 | Shido6 | public ip or nat? |
| 08:46.09 | iceyp | behind nat |
| 08:46.12 | Shido6 | ok |
| 08:46.22 | Shido6 | turn on nat processing on the phone in the sip<mac>.cnf file |
| 08:46.30 | Shido6 | do you have access to the router on the remote end? |
| 08:46.37 | Shido6 | or whatever is doing the nating |
| 08:46.43 | iceyp | no im at work, corp lan |
| 08:46.50 | iceyp | they dont even know i have my phone here ;P |
| 08:47.00 | Shido6 | if you dont have access to the router |
| 08:47.01 | Zeeek | get back to work! |
| 08:47.03 | Shido6 | theres nothing you can do |
| 08:47.08 | iceyp | ahh ok |
| 08:47.09 | Shido6 | exccept try IAX |
| 08:47.17 | iceyp | I need to forward 5060 to my phone? |
| 08:47.24 | Shido6 | not just 5060 |
| 08:47.31 | iceyp | also udp ports |
| 08:47.31 | riksta | iceyp: RTP ports |
| 08:47.34 | riksta | er |
| 08:47.35 | iceyp | ahh ok |
| 08:47.48 | Shido6 | voip_control_port: "5060" |
| 08:47.48 | Shido6 | start_media_port: "16384" |
| 08:47.48 | Shido6 | end_media_port: "32766" |
| 08:47.58 | zhier | if i want talk with somebody else what should i do after i execute the "answer" application |
| 08:48.06 | iceyp | i need to start packing my stuff in the box |
| 08:48.16 | iceyp | Shido6 i wana chat to you, but will have to be tommorow |
| 08:48.26 | iceyp | girls meeting me at home and she wont let me on the puter |
| 08:48.34 | Shido6 | zhier - |
| 08:48.45 | Shido6 | you use answer if you're pickin up a zap channel or have a did coming into your box |
| 08:49.02 | riksta | iceyp: i used to get that :) |
| 08:49.03 | Shido6 | the next priority is whatever you want asterisk to do to handle the call |
| 08:49.15 | Shido6 | iceyp |
| 08:49.20 | Shido6 | call me at 877-677-9649 |
| 08:49.20 | zhier | but i use a softphone to dial or answer |
| 08:49.28 | Shido6 | or IM me at shido6@msn.com |
| 08:49.45 | webmiko | im curious.. anyone know how many people they have over at digium? |
| 08:49.46 | Shido6 | does your softphone have a user and peer setup in sip.conf, zhier ? |
| 08:49.56 | zhier | yes |
| 08:50.04 | Shido6 | and whats the peer called? |
| 08:50.43 | iceyp | will chat to you tomoz |
| 08:50.46 | zhier | for example, i called "dial 2000@from-sip" |
| 08:50.47 | Shido6 | what is your softphones peer called, zhier? |
| 08:50.52 | Shido6 | err |
| 08:50.53 | Shido6 | no |
| 08:51.02 | Shido6 | what is the softphones peer called in sip.conf |
| 08:51.03 | zhier | what? |
| 08:51.11 | zhier | 2000 |
| 08:51.14 | Shido6 | great |
| 08:51.24 | zhier | and then? |
| 08:51.25 | Shido6 | so if you have exten => s,1,Answer |
| 08:51.44 | Shido6 | exten => s,2,Dial(SIP/2000) |
| 08:51.47 | zhier | no i have exten=>2000,1,Answer |
| 08:51.53 | Shido6 | that works |
| 08:51.54 | Shido6 | then |
| 08:51.55 | Shido6 | try |
| 08:52.02 | Shido6 | exten => 2000,1,Answer |
| 08:52.08 | Shido6 | exten => 2000,2,Playback,transfer |
| 08:52.17 | Shido6 | exten => 2000,3,Dial(SIP/2000|20|r) |
| 08:52.24 | Shido6 | exten => 2000,4,Voicemail(u2000) |
| 08:52.32 | Shido6 | exten => 2000,103,Voicemail(b2000) |
| 08:52.35 | zhier | oh i see |
| 08:52.36 | *** join/#asterisk scoof (~scoof@ipa.bryg.org) |
| 08:52.38 | Shido6 | but you dont need the answer |
| 08:52.39 | zhier | thanks |
| 08:52.41 | Shido6 | really |
| 08:52.49 | Shido6 | and setup /etc/asterisk/vociemail.conf |
| 08:52.55 | Shido6 | so you have a 2000 => 1234 line in there |
| 08:53.05 | zhier | yes. |
| 08:53.05 | Shido6 | you'll understand when you read voicemail.conf |
| 08:53.13 | scoof | any chan_sccp users/developers here? |
| 08:53.15 | Shido6 | and add maybe a exten => 6969,1,VoicemailMain |
| 08:53.16 | Shido6 | in there somewhere |
| 08:53.20 | Shido6 | so get to voicemail |
| 08:53.29 | Shido6 | I would do it a totally different way alltogether actually |
| 08:53.37 | Shido6 | make 1 macro for those 4 or 5 lines |
| 08:53.44 | Shido6 | and make a single line when I want the extension to be called |
| 08:53.52 | Shido6 | so when I add ane xtension |
| 08:53.58 | Shido6 | I just add a single line pointing to the macro |
| 08:54.01 | riksta | anyone in the UK who can do me a deal on bulk minutes and DIDs |
| 08:54.03 | zhier | i have configured the voicemail.conf |
| 08:54.04 | Shido6 | rather than adding 4 or 5 additional exten lines |
| 08:54.39 | Shido6 | here's a freebee |
| 08:56.52 | Shido6 | http://pastebin.ca/8468 |
| 08:57.08 | Shido6 | oops |
| 08:57.12 | Shido6 | let me fix that |
| 08:57.38 | Shido6 | http://pastebin.ca/8469 |
| 08:59.03 | riksta | man that pastebin is always SO slow |
| 08:59.41 | Shido6 | I can connect to your box and configure it if you want... a lot faster... |
| 09:00.57 | *** join/#asterisk Aze` (~aze@host229-162.pool80105.interbusiness.it) |
| 09:00.59 | Aze` | re |
| 09:01.28 | Aze` | Anyone use BRI ISDN in PTP mode ? |
| 09:01.43 | *** join/#asterisk soundguy (~soundguy@zeus.blendtek.com.au) |
| 09:03.33 | *** join/#asterisk Delvar (~irc@83.146.53.34) |
| 09:08.39 | *** join/#asterisk Martohtar (~Martohtar@82.196.218.80) |
| 09:13.23 | *** join/#asterisk Zgarbi (~my@212.58.125.68) |
| 09:13.29 | Zgarbi | re |
| 09:14.58 | Zeeek | re |
| 09:15.54 | Zgarbi | where is 1.0.7 Release to download? |
| 09:16.59 | djin | search and you will find it. |
| 09:17.34 | djin | http://www.voip-info.org/tiki-index.php?page=Asterisk-mirrors |
| 09:17.42 | djin | might be a good start |
| 09:17.51 | djin | or http://www.asterisk.org |
| 09:19.51 | Zgarbi | on asterisk.org 1.0.6 |
| 09:19.56 | *** join/#asterisk Jas_Williams (~Jason@host217-44-216-142.range217-44.btcentralplus.com) |
| 09:20.29 | Zgarbi | is a cvs 1.0.7? |
| 09:21.34 | djin | click the first FTP link or use the mirrors page |
| 09:22.40 | Jas_Williams | Zgarbi: No 1.0.7 is stable |
| 09:23.17 | *** join/#asterisk fishboy1669 (proxyuser@62.69.81.129) |
| 09:23.53 | Zgarbi | ok |
| 09:26.04 | Zgarbi | ok, I found it on german mirror |
| 09:27.56 | *** join/#asterisk chapeaurouge (~chap@217.31.73.114) |
| 09:29.24 | *** join/#asterisk _|ms|_ (~mstremer@p83.129.1.149.tisdip.tiscali.de) |
| 09:29.29 | *** join/#asterisk meppl (~mephisto@pD9542453.dip.t-dialin.net) |
| 09:30.50 | webmiko | with GPL licensing having to apply to asterisk modules. does that mean you cant wrap up anything under the bsd license into a module? |
| 09:32.27 | meppl | guten morgen |
| 09:32.52 | _|ms|_ | talk in english please :) |
| 09:33.57 | zoa | aufmachen! |
| 09:34.25 | _|ms|_ | oder ist heute hier deutsch angesagt? LOL |
| 09:34.35 | meppl | it was a amsg |
| 09:34.43 | meppl | good morning ;) |
| 09:34.44 | zoa | woher geht der bus ? |
| 09:34.55 | zoa | ich bin neu her |
| 09:35.03 | zoa | volltanken bitten |
| 09:35.11 | Pj386 | webmiko: I don't think so |
| 09:35.12 | zoa | Schweinhund! |
| 09:35.21 | meppl | zoa, so, its really an english-speaking channel |
| 09:35.22 | Pj386 | that's what the LGPL is for :) |
| 09:35.23 | riksta | #gpl |
| 09:35.30 | chapeaurouge | lol @] Zoa |
| 09:35.53 | _|ms|_ | if you want to talk in german open a pm... or a new channel |
| 09:36.27 | zoa | Ramm....Stein.... |
| 09:36.46 | zoa | 99 luftballon, und ich liebe, ich liebe dich |
| 09:36.49 | zoa | or something |
| 09:36.50 | zoa | :) |
| 09:37.02 | scoof | zoa: not quite :) |
| 09:37.09 | zoa | fear my leet german |
| 09:38.37 | webmiko | Pj> but asterisk isnt lbpl unfortuantely. |
| 09:38.50 | webmiko | shit. lgpl i mean heh |
| 09:39.33 | tessier | Whenever I see german I think of scheisse videos. |
| 09:39.40 | tessier | The Internet has warped my wittle mind. |
| 09:40.45 | Pj386 | webmiko: oh ok... So why did you say GPL ? :) If it's LGPL then you can link with any license |
| 09:41.34 | *** join/#asterisk sambal (~sambal@gateway.office.flatbox.nl) |
| 09:42.04 | webmiko | Pj> i said gpl because it is gpl. that seems counterproductive to asterisk development. specifically keeping developers from building modules using more flexible licensing... |
| 09:43.51 | newl | Is there something stopping you from creating a GPL'd wrapper that loads a binary library? :) |
| 09:45.03 | *** join/#asterisk Xander77 (~alex@seek-it.demon.co.uk) |
| 09:46.48 | zoa | no but that library itself will not be legal |
| 09:46.56 | zoa | as it has to be linked to asterisk |
| 09:47.32 | webmiko | yea thats what i was thinking. |
| 09:50.50 | zoa | unless you write wrappers for all ast_stuff |
| 09:51.38 | Pj386 | wrappers ? How would you do it _without_ linking to asterisk ? |
| 09:52.07 | Pj386 | The only way I see to escape the GPL licensing is to use AGIs, since they're not linked |
| 09:52.13 | Pj386 | or fastAGIs even |
| 09:52.39 | *** join/#asterisk Faithful (~Faithful@202-6-145-116.ip.adam.com.au) |
| 09:57.12 | *** join/#asterisk Wonka (produziert@wonka.support.madwifi) |
| 09:57.16 | Wonka | re |
| 09:59.44 | zhier | who can help me? i want to talk somebody else by the softphone.and what should i do after i called the "Answer" application |
| 09:59.56 | libpcp | anyone has an experience with SMS in asterisk? |
| 10:00.28 | Zeeek | libpcp what's up? |
| 10:00.46 | Zeeek | cd |
| 10:01.55 | *** join/#asterisk soundguy (~soundguy@zeus.blendtek.com.au) |
| 10:01.59 | *** join/#asterisk mesi (~player@dsl-082-083-063-222.arcor-ip.net) |
| 10:02.16 | mesi | YES!!!!!!! Sombody made a call I routed to +49 700 ... :-D |
| 10:02.43 | Zeeek | libpcp your SMS problem? |
| 10:02.45 | zhier | who can help me? i want to talk somebody else by the softphone. And what should i do after i called the "Answer" application? |
| 10:03.11 | Zeeek | Starter tutorial: |
| 10:03.11 | Zeeek | http://www.onlamp.com/pub/a/onlamp/2003/07/03/asterisk.html |
| 10:03.11 | Zeeek | http://www.onlamp.com/pub/a/onlamp/2004/01/22/asterisk2.html |
| 10:03.11 | Zeeek | http://www.automated.it/guidetoasterisk.htm |
| 10:03.11 | Zeeek | THE reference of the moment: |
| 10:03.11 | Zeeek | http://www.asteriskdocs.org |
| 10:04.56 | zhier | i have seen these docs, and all of them just answer the call |
| 10:05.09 | *** join/#asterisk Gh0sty (~Ghosty@ip-81-11-221-231.dsl.scarlet.be) |
| 10:05.32 | Zeeek | if you saw the docs, what is your exact question? |
| 10:06.55 | zhier | for example i dial 2000 and the the "2000" answers the call but what should i do after the answer |
| 10:08.05 | Zeeek | if you saw the docs.... it's all there unless you are trying to do something special |
| 10:08.16 | *** join/#asterisk shmooz (~nobody@host6411912762.biz.tor.fcibroadband.com) |
| 10:10.57 | zhier | the docs just teach me how to answer, but after the answer, we(caller and callee) can't talk with each other |
| 10:11.22 | Zeeek | Why are you answering someone else's phone? |
| 10:11.37 | Zeeek | you want to call 2000 from another phone? |
| 10:11.50 | Zeeek | post your extensions to pastebin.ca |
| 10:12.04 | Zeeek | if the phones are SIP post the sip.conf stuff too |
| 10:13.00 | zhier | no. i just dial in the CLI command line |
| 10:14.00 | Zeeek | I think you are missing some fundamental understanding of asterisk, like extensions and phones... but I'm leaving for lunch |
| 10:14.45 | *** join/#asterisk zhier (~nick@219.137.39.14) |
| 10:16.47 | mesi | Damn! I have a serious problem. There are several calls "I gave", but of length 0 seconds :-( |
| 10:19.44 | Zgarbi | app_addon_sql_mysql.c:164:36: error: macro "AST_LIST_REMOVE" requires 4 arguments, but only 3 given |
| 10:19.53 | Zgarbi | ? |
| 10:20.08 | riksta | Zgarbi: are you running asterisk cvs version |
| 10:20.30 | Zgarbi | no, trying to compile 1.0.7 |
| 10:20.38 | riksta | it only works with cvs dude |
| 10:20.40 | Zgarbi | this is addon |
| 10:20.46 | riksta | i know what it is |
| 10:21.40 | Zgarbi | with 1.0.6 I have problem music on hold, while conversation if put one side on hold no music on other side |
| 10:22.13 | Zgarbi | is this fixed in cvs? |
| 10:22.14 | riksta | what version of mpeg123 do you have |
| 10:23.14 | Zgarbi | Version 0.59r (1999/Jun/15). |
| 10:23.35 | riksta | hm should be ok |
| 10:23.49 | puppet | zgarbi: upgrade to 1.0.7 then? |
| 10:23.52 | puppet | ;p |
| 10:24.00 | Zgarbi | it's located in /urs/local/bin is this problem? |
| 10:24.15 | Zgarbi | not iet, prepare |
| 10:24.21 | riksta | do you have two versions |
| 10:24.23 | Zgarbi | already make "make" |
| 10:24.33 | Zgarbi | no |
| 10:24.52 | Zgarbi | but if addons doesnt works then i skip this |
| 10:24.58 | Zgarbi | I need mysql support |
| 10:24.59 | riksta | dude |
| 10:25.01 | riksta | i just told you |
| 10:25.05 | riksta | the addon works with asterisk cvs |
| 10:25.28 | puppet | but why run cvs when there is 1.0.7 ? |
| 10:25.32 | Zgarbi | it was worked with 1.0.6 |
| 10:25.57 | Zgarbi | I download 1.0.7 asterisk and asterisk-addons |
| 10:26.12 | Zgarbi | both for 1.0.7 |
| 10:26.18 | riksta | puppet: it doesn't compile against 1.0.7 |
| 10:26.22 | Zgarbi | but iet not install 1.0.7 |
| 10:26.28 | riksta | are you sure you didnt get the cvs addons? |
| 10:26.35 | Zgarbi | sure |
| 10:26.38 | puppet | riksta: worked here :/ |
| 10:26.49 | riksta | hm |
| 10:27.43 | Zgarbi | http://www.asterisk-support.de/mirror/asterisk-1.07-STABLE/asterisk-addons-1.0.7.tar.gz |
| 10:28.40 | riksta | i just tried it |
| 10:28.42 | riksta | i get the same error |
| 10:28.47 | riksta | and im running 1.0.7 |
| 10:28.57 | riksta | im sure that someone said it needs to be running cvs * |
| 10:30.57 | shmooz | what is the best frontend for asterisk at the moment? |
| 10:30.57 | *** join/#asterisk meppl (~mephisto@p3E9E2B55.dip.t-dialin.net) |
| 10:31.33 | Delvar | SSH |
| 10:32.15 | shmooz | :\ |
| 10:40.02 | Pj386 | I'm gonna test out amp soon, it seems good... A little bit afraid of the "black magic" side, so I have to see how it works internally, but it "looks" nice |
| 10:45.26 | shmooz | Pj386 I'm I'm doing a frontend in php, it adds removes views ,contexts phones voicemail info and queues so far, have to add the dialplan part and stuff... |
| 10:45.44 | mesi | Anybody using fwdOUT? |
| 10:48.08 | *** join/#asterisk sudhir492 (~sudhir@wbar1.wdc2-4-8-141-004.wdc2.dsl-verizon.net) |
| 10:48.11 | sudhir492 | hi all |
| 10:48.40 | mesi | hi sudhir |
| 10:51.06 | *** join/#asterisk bjohnson_ (~bjohnson@66.11.165.161) |
| 10:51.42 | Zgarbi | so better to run on cvs then stable? |
| 10:52.55 | riksta | depends what you mean by better |
| 10:53.24 | Zgarbi | addon |
| 10:53.32 | riksta | wtf |
| 10:54.01 | Zgarbi | either I already install 1.0.7 and it's not working for some extensions |
| 10:54.05 | Zgarbi | strange |
| 10:54.16 | riksta | more liekly to be user configuration error tbh |
| 10:54.21 | Zgarbi | musiconhold didnt works for conversation :( |
| 10:55.32 | riksta | does normal MOH work? |
| 10:55.53 | riksta | add this to your dialplan to test it |
| 10:56.00 | riksta | exten => 555,1,WaitMusicOnHold(120) |
| 10:56.10 | riksta | or something similar |
| 10:56.36 | Zgarbi | I have 2 things |
| 10:56.43 | Zgarbi | one: |
| 10:56.44 | Zgarbi | exten => 1991,1,Answer |
| 10:56.45 | Zgarbi | exten => 1991,2,Wait,1 |
| 10:56.45 | Zgarbi | exten => 1991,3,MusicOnHold(default) |
| 10:56.45 | Zgarbi | exten => 1991,5,Wait,20 |
| 10:56.45 | Zgarbi | exten => 1991,6,Hangup |
| 10:56.49 | Zgarbi | it's works |
| 10:56.58 | Zgarbi | second: |
| 10:56.59 | Zgarbi | exten => 1992,1,Answer |
| 10:56.59 | Zgarbi | exten => 1992,2,Wait,1 |
| 10:56.59 | Zgarbi | exten => 1992,3,MusicOnHold(radio_ge) |
| 10:57.00 | Zgarbi | exten => 1992,5,Wait,20 |
| 10:57.00 | Zgarbi | exten => 1992,6,Hangup |
| 10:57.21 | riksta | that isn't the standard music on hold is it |
| 10:57.28 | Zgarbi | second is a streamed music and it's worked for me in 1.0.6 but non in 1.0.7 |
| 10:57.40 | Pj386 | shmooz: I'd be intersted, tell me if you have a website |
| 10:58.12 | Zgarbi | but no music on hold while conversation hold |
| 10:58.12 | Zgarbi | yes, standard |
| 10:58.43 | Zgarbi | second is connecting, but no voice |
| 10:59.10 | riksta | what do you mean no voice |
| 10:59.24 | riksta | have you checked the stream in xmms or something |
| 10:59.28 | Zgarbi | quite |
| 10:59.38 | riksta | quite? |
| 10:59.44 | riksta | are you on crack? |
| 10:59.54 | Zgarbi | I mean no sound |
| 11:00.44 | riksta | check the steam is working in xmms |
| 11:00.50 | Zgarbi | I retransmit streamed audio... but not anymore |
| 11:00.54 | Zgarbi | ok |
| 11:06.03 | shmooz | Pj386 : I don't have a demo setup right now , I can send you the php code if you want to try it on your server |
| 11:08.35 | *** join/#asterisk JonR800 (jr@pcp05013027pcs.plyntv01.mi.comcast.net) |
| 11:12.21 | *** join/#asterisk nesys (~nesys@81-174-12-111.f5.ngi.it) |
| 11:12.54 | *** join/#asterisk CleanerX (~nix@nat-ph3-wh.rz.uni-karlsruhe.de) |
| 11:23.25 | queuetue | How do I automatically repeat a menu after waiting for a response? |
| 11:23.41 | *** join/#asterisk Fabe_ (~spamhere@pD95B0BF7.dip0.t-ipconnect.de) |
| 11:23.51 | InfraRed | loop? |
| 11:24.26 | queuetue | Are there loops? Where is asterisk's extensions syntax really described? |
| 11:25.21 | Jas_Williams | queuetue: t means time out exten=> t,1,goto(s) should work |
| 11:26.21 | queuetue | Jas_Williams, But s starts with Answer() - is that ignored (or unnecessary?) |
| 11:27.08 | Jas_Williams | It will be ignored it is not necessary if you are doing a background as this answers the line when neccessary |
| 11:28.17 | Jas_Williams | or an exten=> t,1,goto(s,2) will got to priority 2 |
| 11:28.22 | Zeeek | hey Jas_Williams |
| 11:29.46 | Jas_Williams | afternoon Zeeek |
| 11:31.04 | Mavvie | on a packet received with recvfrom(), how can I see *to* which IP address it was send. |
| 11:31.16 | Zeeek | Jas have you ever set up a small asterisk box? Like using no disks, anything like that? |
| 11:33.12 | Jas_Williams | Zeeek: no fraid not BUt i'm sure there was some builds on the users list search that |
| 11:33.42 | Zeeek | I've been watching the list, yeah there's been a little about it. |
| 11:38.00 | scoof | anybody got Cisco 7970s they want to test with Asterisk? |
| 11:38.02 | queuetue | I'm using x-lite's soft phone for linux. outgoing calls work fine, but incoming ones play this horrifying, cacaphonous racket instead of ringing, then make absolutely no sound after connecting. Is this a problem with the telco provider (broadvoice), asterisk config, x-lite config, or screwed up ausio in linux? |
| 11:38.29 | queuetue | Caller id works fineon these calls. |
| 11:39.29 | queuetue | (are hard phones less trouble, or is wierd noise and mysterious connection state just a given with sip phones, soft or hard?) |
| 11:39.38 | *** join/#asterisk PoWeRKiLL (~PoWeRKiLL@l02m-4-165.d4.club-internet.fr) |
| 11:41.00 | *** join/#asterisk Memphili (~Mephisto@dsl-084-058-006-104.arcor-ip.net) |
| 11:41.11 | Memphili | hi |
| 11:42.26 | *** join/#asterisk markak2 (~twist@ndn-165-150-215.telkomadsl.co.za) |
| 11:43.36 | *** join/#asterisk tessier (~treed@222.253.72.192) |
| 11:43.51 | *** join/#asterisk pif (ldm@82.66.93.83) |
| 11:44.11 | markak2 | hi all, i have a small question. my asterisk installation is running perfectly. i am busy adding options to it. the one i am trying to do is to allow a caller to get passed through to one of the other zap channels so that they may dial out on it. this is to allow a cellphone user to call international destinations. ie: cell clls in on zap1 certain extension connects him to zap2 with open pstn line. is this possible. |
| 11:46.37 | *** part/#asterisk Memphili (~Mephisto@dsl-084-058-006-104.arcor-ip.net) |
| 11:47.32 | Zeeek | markak2 yes look for DISA |
| 11:47.53 | Zeeek | or just Dial(ZAP/1) |
| 11:48.03 | Zeeek | or whatever you your line is |
| 11:48.13 | Zeeek | ZAP/2 you said |
| 11:49.13 | mesi | That's bad there is no forum on fwdOUT.net |
| 11:49.23 | Zeeek | yes it is bad |
| 11:49.28 | Zeeek | terrible in fact |
| 11:49.38 | Zeeek | but there is FWD forum |
| 11:53.20 | Jas_Williams | mesi: there its http://forum.fwdout.net/ |
| 11:53.31 | Zeeek | how about that? :) |
| 11:54.19 | Jas_Williams | ;-P |
| 11:54.35 | Zeeek | I tried bellster for about 1 day |
| 11:55.54 | queuetue | How do I play a .gsm in linux (outside of asterisk) |
| 11:56.06 | Zeeek | I hear sox does it |
| 11:56.10 | newl | soxplay |
| 11:56.24 | _|ms|_ | and what do you use for windows? |
| 11:56.42 | Zeeek | quicktime or efax |
| 11:57.08 | queuetue | soxplay? I've got sox, but do not know soxplay. |
| 11:58.01 | queuetue | the "play" command works here (ubuntu-hoary) |
| 12:01.54 | queuetue | How do I insert a playback before an extension is called? (I have ext 1003, 1004, 1005 set up, and I want a "please hold while I connect you" message to play before the extension is called, without inserting it in each extension individually) |
| 12:02.14 | markak2 | thanks all going to use DISA option. |
| 12:02.34 | *** part/#asterisk markak2 (~twist@ndn-165-150-215.telkomadsl.co.za) |
| 12:02.44 | Zeeek | queuetue what does the dial command looklike? |
| 12:03.16 | queuetue | Zeeek, exten => 1060,1,Dial(${SALES}) |
| 12:03.29 | Zeeek | and sales is the multiple extensions? |
| 12:03.40 | *** join/#asterisk riksta (~rick@81-178-231-174.dsl.pipex.com) |
| 12:04.03 | queuetue | Zeeek, I also have a exten => 1001,1,Dial(${MIKE}) , etc.. |
| 12:04.29 | Zeeek | you it tries them one after the other? |
| 12:04.41 | queuetue | Am I setting this up wrong? These are all in an "extensions" section that gets included by [incoming] and [intrnal] |
| 12:04.52 | queuetue | Zeeek, No, it just tries the one you dial, I think... |
| 12:04.59 | Zeeek | depends on what you are actually trying to accomplish |
| 12:05.16 | Zeeek | just answer and playback the short message and then dial |
| 12:05.42 | Zeeek | if you mean how to insert it always, maybe you want to make a macro for this kind of extension |
| 12:05.59 | queuetue | Ok, macros. Off to read more. |
| 12:06.14 | Zeeek | do a search on macro-online |
| 12:06.26 | Zeeek | there's the classic macro that's been floating around everywhere |
| 12:11.36 | *** join/#asterisk caesar2 (caesar@p5497CB77.dip.t-dialin.net) |
| 12:12.14 | caesar2 | hi, good iax soft phone, besides firefly ? firefly crashes at 100% cpu load |
| 12:15.28 | *** join/#asterisk CrudeOil (CrudeOil@dsl-217-155-150-237.zen.co.uk) |
| 12:27.39 | *** join/#asterisk PoWeRKiLL (~PoWeRKiLL@l02m-4-165.d4.club-internet.fr) |
| 12:28.18 | *** join/#asterisk emiddleton (~edward@ZQ236211.ppp.dion.ne.jp) |
| 12:34.35 | Zeeek | ooops macro-oneline ! |
| 12:39.07 | *** join/#asterisk dmccollum (~dmccollum@69-164-245-72.atlaga.adelphia.net) |
| 12:53.17 | *** join/#asterisk jmacz (~jmacz@63.245.86.116) |
| 12:55.56 | *** join/#asterisk verge (~jfargen@rrcs-67-78-209-206.se.biz.rr.com) |
| 12:56.15 | Zeeek | exit |
| 12:56.19 | verge | good morning anyone in asterisk land |
| 12:57.53 | Essobi | woop |
| 12:59.39 | *** join/#asterisk fenlander (~irc@82.152.81.57) |
| 13:07.57 | *** join/#asterisk MikeJ[Laptop] (~icechat5@65.170.43.34) |
| 13:11.44 | *** join/#asterisk Moc (~Moc@modemcable165.109-70-69.mc.videotron.ca) |
| 13:13.34 | *** join/#asterisk sandnigg0r (~crunkuser@66-55-197-254.gwi.net) |
| 13:13.59 | *** join/#asterisk matiasg (~listas_as@200.68.82.225) |
| 13:15.07 | *** join/#asterisk gambolputty (~gambolput@cblmdm65-221-51-80.buckeye-express.com) |
| 13:15.55 | *** join/#asterisk bjohnson (~bjohnson@jecinc.tor.istop.com) |
| 13:17.02 | t0p | verge: are you there? |
| 13:23.35 | marlowe | booya |
| 13:23.40 | marlowe | t0p: Im here. |
| 13:23.43 | marlowe | Am I good enough? |
| 13:23.48 | marlowe | :) |
| 13:23.49 | bjohnson | no |
| 13:23.53 | marlowe | :-0 |
| 13:24.13 | jay-p | This question haunts my mind / Will we survive this night? / We're harboring the meek / Will we survive the sleep? |
| 13:24.27 | marlowe | bjohnson: I'm good enough. |
| 13:24.57 | Zeeek | no |
| 13:25.30 | marlowe | Zeek: Yes! |
| 13:25.37 | marlowe | Don't make me cry. |
| 13:25.39 | Zeeek | not not not ! |
| 13:25.53 | bjohnson | This quesion haunts my mind: will I become like jay-p when I become senile? |
| 13:26.12 | Zeeek | what do you mean, when? |
| 13:26.35 | queuetue | jay-p, What are you talking about? That sounds like high school poetry. :) |
| 13:26.39 | jay-p | bjohnson, any senior citizen who plays pantera is cool in my book |
| 13:26.50 | marlowe | la la la |
| 13:27.04 | marlowe | lilo messed my cloak up |
| 13:27.08 | marlowe | I paid for sustaining |
| 13:27.12 | marlowe | I got active |
| 13:27.14 | marlowe | :'( |
| 13:32.42 | verge | t0p: I am here. |
| 13:40.14 | *** join/#asterisk tessier (~treed@222.253.72.192) |
| 13:44.08 | t0p | marlowe, verge: Sorry, I was away |
| 13:44.17 | t0p | just got some questions to ask |
| 13:45.22 | t0p | do you guys know why a softphone doesn't send asterisk the registry statements |
| 13:47.14 | *** join/#asterisk m654321 (~twist@ndn-165-150-215.telkomadsl.co.za) |
| 13:47.24 | m654321 | zeeek are you there ? |
| 13:48.14 | m654321 | anybody understand the disa command well |
| 13:49.28 | bjohnson | no |
| 13:51.14 | *** join/#asterisk langals (~icechat5@196.7.14.183) |
| 13:51.40 | langals | Hi there...I have a question regarding codecs.... |
| 13:52.30 | langals | I have implemented Meetme... |
| 13:53.08 | langals | Now, I read on voip-info.org that Meetme will use Ulaw by default... |
| 13:53.14 | *** join/#asterisk cjk (~cjk@80.92.64.103) |
| 13:53.30 | verge | t0p-I have zero experience with softphones. |
| 13:53.36 | cjk | hi, does anyone of you guys know a phone supporting ilbc and iax2 |
| 13:54.21 | t0p | verge: that's okay I will keep trying |
| 13:54.28 | *** join/#asterisk kleper (~kleper@200.30.69.177) |
| 13:54.29 | langals | In my sip.conf file I have specified - "disallow=all; all=gsm". Does this mean that both client and meetme server are using GSM...or that client is using GSM and server is transcoding to ULAW |
| 13:54.31 | kleper | hi |
| 13:54.50 | *** join/#asterisk tessier_ (~treed@222.253.72.192) |
| 13:55.04 | *** join/#asterisk dsfr (~dsfr@216.207.244.183) |
| 13:56.40 | queuetue | What's a good supplier for globalstream phones and digium "clone" cards? |
| 13:57.13 | queuetue | (The tech test is entering next phase :) ) |
| 13:57.38 | Faithful | Using Asterisk software and standard PC hardware, one can create a SOHO (Small Office Home Office) telephony environment that includes all the sophisticated features of a high-end business telephone system. |
| 13:57.52 | kleper | i have a server asterisk on my lan , and i have only one public IP, i need can connect to the server asterisk calling to mu public ip? is possible |
| 13:57.58 | Faithful | what a plug!!! |
| 13:58.10 | Faithful | And that's from a reseller |
| 13:58.24 | Faithful | * is SOHO |
| 13:58.52 | kleper | on the wiki of voip i see a option with port forwarding and some header mangling magic??? but i need more info??? |
| 13:59.13 | Faithful | kleper: you got a firewall? |
| 13:59.15 | scoof | kleper: then you should look to the documentation for your NAT device |
| 13:59.38 | kleper | i have a linux firewall |
| 14:00.10 | kleper | so i use iptables but i don't know all the ports tha i have forward |
| 14:00.18 | kleper | *that |
| 14:00.29 | scoof | kleper: What protocols do you want to use? |
| 14:00.32 | Faithful | so it's pretty easy... just use the iptable rules they give you on viop-info.org |
| 14:00.35 | bjohnson | langals: my understanding is that meetme is a p2p type app. So perhaps it will use gsm with the other peers that it connects to? Try it out both ways and update the wiki |
| 14:00.56 | pigpen | Hi all...I am having issues with * and a Sipura SPA 3000 FXO...rings...but no audio.. |
| 14:00.58 | Faithful | just doing iax in? |
| 14:01.04 | kleper | sip and iax |
| 14:01.10 | Faithful | pigpen: codec |
| 14:01.16 | bjohnson | cjk: iax2 phones have only been available in the last few months. You'll have to 1. find some 2. check what codecs they support |
| 14:01.23 | pigpen | Faithful: that was fast. |
| 14:01.31 | Faithful | 4569 udp --> * |
| 14:01.36 | queuetue | sip incoming (through the firewall) or just outgoing? |
| 14:01.50 | pigpen | the spa is on eth1 with no nat... |
| 14:02.00 | pigpen | my phones are on the internet.... |
| 14:02.01 | *** join/#asterisk jakepdev (~jakepdev@pool-68-163-29-219.phil.east.verizon.net) |
| 14:02.05 | pigpen | phones work fine... |
| 14:02.12 | pigpen | spa to voice mail works fine... |
| 14:02.19 | pigpen | but spa to phones no audio... |
| 14:02.46 | bjohnson | kleper: use iax2 and do port forwarding on the iax2 port (I think 5679) |
| 14:03.03 | CrudeOil | hi all |
| 14:03.09 | kleper | ok bjohnson i have iax2 |
| 14:03.14 | pigpen | Faithful: 4569 udp for the spa or the phones? |
| 14:03.15 | Faithful | kleper: http://voip-info.org/tiki-index.php?page=Asterisk%20firewall%20rules |
| 14:03.25 | bjohnson | pigpen: all on one lan? ie no NAT routers being crossed? |
| 14:03.28 | Faithful | pigpen: kleper |
| 14:03.40 | Faithful | I have a mouse |
| 14:03.54 | Faithful | I hear him... |
| 14:03.55 | bjohnson | pigpen: nnm |
| 14:04.14 | pigpen | I have the * box at my colo....with 2 nics...1 public / 1 private. |
| 14:04.18 | pigpen | spa's are on the private |
| 14:04.22 | CrudeOil | when sending a call to a Quintum from * the Quintum only reports 1 codec (g723) and so * will only send g723 even though the Quintum will auto-negotiate any codec |
| 14:04.24 | bjohnson | pigpen: as Faithful said .. it is likely a codec problem |
| 14:04.43 | pigpen | yeah..but shouldnt I get some sort of codec error in asterisk? |
| 14:04.49 | CrudeOil | how can i force * to send any codec even if the Quintum only reports g723 |
| 14:04.55 | jakepdev | anyone think their VOIP connection sounds as good as or better than POTS? - what is the ping time to your provider? |
| 14:05.05 | bjohnson | pigpen: maybe if you have the logs turned up high enough |
| 14:05.13 | bjohnson | pigpen: is it the fxs or fxo port? |
| 14:05.21 | bjohnson | pigpen: fxs right? |
| 14:05.26 | pigpen | fxs turned off...fxo only |
| 14:05.41 | Faithful | jakepdev: were not all lining up to tell you!!! |
| 14:05.46 | bjohnson | you followed the info at the bottom of the wiki page about 3ks? |
| 14:05.59 | bjohnson | pigpen: what do you mean it rings? |
| 14:06.09 | jakepdev | Faithful - I can see that :) |
| 14:06.29 | pigpen | ie, If I call the number the fxo is conn to...I have it forwarding to my extention...it rings...I answer...no audio... |
| 14:06.33 | bjohnson | jakepdev: ping times are normally 30-50 ms |
| 14:06.40 | pigpen | If I let it go to voicemail...all is fine. |
| 14:06.43 | Faithful | Hey guys, I did the zaphfc for ISDN and now I am very happy FYI no echo and reliable |
| 14:06.57 | jakepdev | bj - as good as or better than POTS? |
| 14:07.11 | bjohnson | jakepdev: equivalent to mey ear |
| 14:07.19 | jakepdev | tnx |
| 14:07.22 | kleper | Faithful, thx |
| 14:07.30 | bjohnson | pigpen: pastebin the cli info of a call with set verbose 5 |
| 14:07.33 | Faithful | jakepdev: bjohnson is deaf btw |
| 14:07.39 | jakepdev | hah |
| 14:07.39 | bjohnson | what? |
| 14:07.51 | pigpen | k... |
| 14:08.20 | langals | bjohnson - thanks. If I type "sip show channels" and it indicates "GSM" under format, does this mean that both client and server are sending in GSM? |
| 14:09.03 | bjohnson | it means that .. that call is using gsm |
| 14:09.09 | Jas_Williams | langals: Yes if they wer not using the same codec they would not be able to talk to one another ;-) |
| 14:09.16 | *** join/#asterisk LoRez_ (lorez@lorez.staff.freenode) |
| 14:09.28 | jakepdev | I have 58-60ms and the latency seems to be a little much |
| 14:09.31 | Faithful | jakepdev: I'm using ilbc now and it sound the best it ever has |
| 14:10.01 | bjohnson | jakepdev: find a faster way to connect or a different voip provider |
| 14:10.11 | *** join/#asterisk smeevil (~smeevil@gremesh1.demon.nl) |
| 14:10.25 | jakepdev | ideal should be 30-50? |
| 14:10.27 | scoof | jakepdev: tried tweaking your jitter-buffers? Do you know how big they are now? |
| 14:10.28 | smeevil | goood day all |
| 14:10.47 | bjohnson | ilbc is the best codec in high jitter situations .. but takes more cpu power than some other codecs and is not supported by all devices |
| 14:10.56 | bjohnson | jakepdev: ideal would be 0 |
| 14:11.04 | smeevil | can anyone tell me which module to load , to solve the following problem on chan_sip : chan_sip.so: undefined symbol: ast_park_call |
| 14:11.08 | pigpen | bjohnson: http://www.pastebin.com/264776 |
| 14:11.15 | pigpen | not much info...looks like a normal call. |
| 14:11.19 | scoof | jakepdev: and how have you deemed that 58-60ms is "a little too much"? |
| 14:11.50 | Faithful | jakepdev: the issue is jitter, check your jitter |
| 14:11.54 | Faithful | mdev |
| 14:12.38 | bjohnson | pigpen: that is calling INTO the fxo? |
| 14:12.43 | pigpen | yep. |
| 14:13.02 | pigpen | I can get you access to the spa if it would help... |
| 14:13.20 | *** join/#asterisk af_ (~af@ip-148-227.sn1.eutelia.it) |
| 14:13.34 | bjohnson | nope |
| 14:13.43 | bjohnson | I want the extensions.conf for that though |
| 14:13.53 | pigpen | k. |
| 14:13.58 | pigpen | pastebin? |
| 14:14.24 | bjohnson | yes |
| 14:14.44 | *** join/#asterisk anthm (~anthmct@69.76.83.52) |
| 14:14.44 | *** mode/#asterisk [+o anthm] by ChanServ |
| 14:14.49 | bjohnson | pigpen: is audio one way or not at all between the fxo call and the ip phone? |
| 14:14.49 | langals | Jas_Williams - thanks |
| 14:15.08 | bjohnson | I assume 'mark' is an ip phone |
| 14:15.22 | Hmmhesays | this memory leak is starting to piss me off |
| 14:15.24 | langals | Has anyone out there done any programming with the MS RTC core? |
| 14:15.41 | bjohnson | Hmmhesays: my memory has been leaking for years |
| 14:15.50 | pigpen | bjohnson: http://www.pastebin.com/264779 |
| 14:15.57 | pigpen | bjohnson: yeah...mark is my extention |
| 14:16.02 | langals | I apologise for mentioning microsoft :-) |
| 14:16.04 | Hmmhesays | I left the server with 200mb used last night, now it's at a gig |
| 14:16.25 | bjohnson | pigpen: what context does the fxo come in on |
| 14:16.37 | pigpen | CCNBI-Default |
| 14:17.33 | pigpen | see...I called sipura...they had me set up the fxo to dial into * to ext 99 |
| 14:17.41 | pigpen | then from there I was passing it to my extention... |
| 14:17.47 | bjohnson | pigpen: you've got multiple s extens .. which one do you suppose it will use? |
| 14:17.48 | pigpen | but I figured that was kinda redundant |
| 14:18.03 | pigpen | so I set the spa to pass it to ext 213 |
| 14:18.15 | pigpen | ie: mark 's phone |
| 14:18.26 | *** join/#asterisk Fabe_ (~spamhere@pD95B0BF7.dip0.t-ipconnect.de) |
| 16:03.24 | *** join/#asterisk jbot (ibot@apt.bot.TimRiker.active.supporter.pdpc) |
| 16:03.24 | *** topic/#asterisk is Asterisk: The Open Source PBX || 1.0.7 Released || http://www.theregister.co.uk/2005/03/16/asterisk_open_source_pbx/ || Allison ROCKS my socks!!! |
| 16:03.33 | queuetue | Seriously, is anyone seeing my messages? (If no one wants to answer my questions, that's fine - but if IRC isn't working, I'd like to know. :) ) |
| 16:03.37 | pigpen | I think it is how vonage sends the caller id... |
| 16:03.44 | pigpen | everything comes in with " xxx " |
| 16:03.44 | AgiNamu | queutue, now , everyoe is ignoring you |
| 16:03.53 | AgiNamu | andi cant write wott shit |
| 16:03.54 | Hmmhesays | queutue |
| 16:03.56 | queuetue | AgiNamu, Good toy know. :) |
| 16:04.11 | Hmmhesays | you speak of digium clone cards |
| 16:04.27 | Hmmhesays | if you are talking about a 5 dollar card you find on ebay... don't bother |
| 16:04.33 | Mimmus | how can I know if bug ID 2687 is fixed in current stable version of Asterisk? |
| 16:04.37 | Jas_Williams | langals: The meetme uses a function of the zaptel timer to distribute voice frames in a timely manner to all parties in the meetme |
| 16:04.41 | queuetue | Hmmhesays, wny dont bother? |
| 16:04.43 | pigpen | ManxPower: I guess I will see if I can setup vonage from sending the caller id... |
| 16:04.45 | ManxPower | Mimmus: You read the asterisk-cvs mailing list |
| 16:04.57 | Hmmhesays | cause you'll end up pissing everyone in here off when it creates strange problems for you |
| 16:05.03 | Mimmus | ManxPower: No |
| 16:05.13 | ManxPower | pigpen: You are using Vonage softphone account. |
| 16:05.15 | AgiNamu | thats right |
| 16:05.16 | cbachman | Jas_Williams, I've used mine mostly to call out through VOIP, but I've not noticed any problems with incoming calls that are passed through. |
| 16:05.20 | AgiNamu | you'll run into enough issues as-is. |
| 16:05.20 | ManxPower | Mimmus: Then you will not know the information you are looking for. |
| 16:05.25 | Hmmhesays | now I must find out why my openvpn does not work |
| 16:05.27 | AgiNamu | adding some kinda clone just makes things worse |
| 16:05.38 | pigpen | ManxPower: no...I am passing my ata account directly from my cisco ata to the sipura... |
| 16:05.40 | queuetue | Hmmhesays, So, what is the entry price of attaching a single phone line to asterisk? |
| 16:05.48 | Mimmus | ManxPower: Bugtracker Mantis doesn't work? |
| 16:05.51 | Hmmhesays | roughly 100 bucks if I remember right |
| 16:05.51 | ManxPower | pigpen: that's just sick. |
| 16:06.02 | pigpen | ManxPower: I know...I have a PRI on order... |
| 16:06.03 | queuetue | (If the $5.00 cards don't work) |
| 16:06.20 | Hmmhesays | queutue: the 5 dolla cards MIGHT work ok... that is the key word there MIGHT |
| 16:06.21 | ManxPower | Mimmus: No. That just lists what is submitted. It MAY show if it was commited to CVS, but honestly the correct place to look is the asterisk-cvs mailing list. |
| 16:06.24 | pigpen | but we have used vonage for 2 years...we need time to move over.. |
| 16:06.28 | Jas_Williams | cbachman: I was trying to use it connected to a dect phone and the wife complained about voice quality and I had to agree with her it is not toll quality but I could not explain why .. |
| 16:06.46 | Hmmhesays | queutue: buy a card from digium and you have warranty and support |
| 16:06.54 | queuetue | ...80.00 per phone and 100 bucks per incoming line, and suddentlly, I might as well buy a proprietary pbx. |
| 16:06.59 | Mimmus | ManxPower: mmmmm, bad thing. I'm not a cvs, test-and-compile guru |
| 16:07.09 | Hmmhesays | queuetue: then go buy one if you like your legacy stuff |
| 16:07.16 | tzanger | queuetue: no you don't use an X100P for every incoming line, that's gsick |
| 16:07.17 | cbachman | Jas_Williams, I couldn't either. It's passing directly across the loopback. |
| 16:07.27 | tzanger | queuetue: and if you want a proprietary PBX then go for it |
| 16:07.37 | tzanger | asterisk doesn't just compete on price |
| 16:07.49 | Hmmhesays | asterisk competes on tweakablility factor |
| 16:08.05 | Katty | Hmmhesays: a+ |
| 16:08.13 | fugitivo | i'm looking for some doc or webpage describing all the available codecs for asterisk |
| 16:08.14 | Jas_Williams | cbachman correct but it still seemed to be experiencing drop out so I am not using it at the moment |
| 16:08.20 | Hmmhesays | I got some good ebooks on A+ |
| 16:08.33 | Katty | Hmmhesays: and i've got online classes |
| 16:08.38 | Hmmhesays | Katty: cool |
| 16:08.41 | Katty | mhmm |
| 16:08.50 | Hmmhesays | I never got my A+ probably wouldn't be to hard though |
| 16:09.02 | Katty | yeah, just all these silly facts to memorize |
| 16:09.03 | *** join/#asterisk tessier (~treed@210.245.100.67) |
| 16:09.16 | *** join/#asterisk tessier_ (~treed@210.245.100.67) |
| 16:09.24 | AgiNamu | it's $337 for 4 FXO ports on a TDM |
| 16:09.26 | AgiNamu | card |
| 16:09.46 | tessier | You know... |
| 16:09.46 | Hmmhesays | most geeks have freakiskly good memory when it comes to inane facts |
| 16:09.51 | tessier | These days, I would just buy a Cisco with 4 FXO ports |
| 16:09.52 | eKo1 | How much of an increase in salary am I supposed to get with A+? |
| 16:09.53 | tessier | It's cheaper. |
| 16:10.00 | tessier | And a hell of a lot more reliable. |
| 16:10.01 | Katty | Hmmhesays: just except me, apparently. |
| 16:10.09 | AgiNamu | and for comparable features.... it is rather inexpensive. |
| 16:10.10 | Katty | Hmmhesays: or maybe i've just not been around long enough? |
| 16:10.19 | queuetue | And how well do QOS solution work for voip trunklines? There's no way we can have calls drop to nothing every time someone connects to AOL or whatever...and dropping a T1 in here to make the phones work better doesn't make any sense. |
| 16:10.24 | AgiNamu | AND..... well.... if you dont use analog lines, it's real cheap |
| 16:10.34 | Katty | Hmmhesays: i was sorta newborn with the when the 286 came out |
| 16:10.41 | Katty | ... |
| 16:10.48 | AgiNamu | you were newborn when 286s come out? |
| 16:10.49 | Hmmhesays | Katty: I think I was 2 |
| 16:10.53 | AgiNamu | that was like... a few years ago. |
| 16:10.58 | Hmmhesays | 85? |
| 16:11.01 | Katty | AgiNamu: yes. |
| 16:11.04 | Katty | Hmmhesays: 84 |
| 16:11.06 | *** join/#asterisk heison (~heison@ns.somanetworks.com) |
| 16:11.10 | queuetue | I still have a car that's that old. :) |
| 16:11.14 | tessier | queuetue: Just prioritize the traffic on yor IP connection and you'll be fine. |
| 16:11.14 | AgiNamu | hehe... i was only born a few years before. |
| 16:11.14 | Hmmhesays | when the 286 came out? |
| 16:11.16 | Katty | AgiNamu: i'm only 20. heh |
| 16:11.27 | tessier | queuetue: I use Wondershaper from lartc.org on my cable/dsl lines and it works great. |
| 16:11.30 | Hmmhesays | ok i was 2 when the 286 came out |
| 16:11.31 | Katty | bkw_: YOU OLD GEIZER |
| 16:11.39 | AgiNamu | im only 23 |
| 16:11.42 | queuetue | tessier, what kind of hardware does that require? |
| 16:11.47 | Jas_Williams | Makes me feel old at 38 |
| 16:11.50 | AgiNamu | but i remember upgrading to a 386 |
| 16:11.54 | Katty | Jas_Williams: you just need hugs. |
| 16:11.55 | tessier | queuetue: Just a Linux box to be your router. |
| 16:12.01 | Hmmhesays | i didn't start in with computers until 1999 |
| 16:12.11 | tessier | Hmmhesays: Wow. Newbie! |
| 16:12.12 | Jas_Williams | me/ likes hugs |
| 16:12.29 | tessier | An Apple ][c |
| 16:12.34 | Katty | i started html when i was..uhh...13ish |
| 16:12.35 | Hmmhesays | I got my first computer in 1999 |
| 16:12.39 | queuetue | So, another computer. :) Why do I have a feeling the economy of scale makes no sense to deploy astrisk until you get up into the hundreds of lines. :) |
| 16:12.41 | Hmmhesays | and now here I sit |
| 16:12.44 | Jas_Williams | kit |
| 16:12.57 | tessier | queuetue: That's not true at all. |
| 16:13.04 | queuetue | (Well, unless you have some particularly complex phon control needs.) |
| 16:13.16 | tessier | queuetue: Another computer when you probably already have a router there you can replace. Plus you don't need an expensive computer. A 486 could route a dsl/cablemodem connection. |
| 16:13.19 | Katty | do i really need to know the external speed of all these processors? |
| 16:13.33 | Hmmhesays | they are pretty easy to memorize |
| 16:13.37 | tessier | Katty: Not only that but what is the airspeed velocity of an unladen swallow? |
| 16:13.39 | Katty | i have flashcards! |
| 16:13.44 | Katty | tessier: what do you mean? |
| 16:13.45 | Hmmhesays | african swallow? |
| 16:13.48 | Katty | tessier: african or european? |
| 16:13.50 | AgiNamu | tessier: african or european. |
| 16:13.52 | AgiNamu | damn |
| 16:13.54 | ManxPower | Anyone that uses Asterisk CVS and is not on the asterisk-cvs mailing list is an idiot! |
| 16:13.55 | tzanger | Katty: so do I... 16M, 32M and a couple 512M |
| 16:13.55 | tessier | heh |
| 16:14.05 | Shido6 | boink |
| 16:14.07 | Katty | tzanger: *smirk* |
| 16:14.09 | SPoon_TSX | Hello there, Does anyone having an experience that the hardware phone will dial outgoing call no problem but it will hung up on any incoming call on excatly 3 seconds? |
| 16:14.13 | Katty | OH NOES |
| 16:14.25 | Katty | tessier: butbut, i know my favorite color. |
| 16:14.29 | eKo1 | ManxPower: I resent that. |
| 16:14.34 | queuetue | tessier, Yes, and another box to babysit and maintain. Only now, it's a 486 with a fan and hd/floppy that will fail any day now. Putting substandard equipment in place for mission-critical jobs is dangerous. |
| 16:14.46 | tessier | queuetue: uh...don't use a 486 with a fan and hd/floppy. |
| 16:14.50 | Hmmhesays | haha I'm sick of people asking me if they can run voip on a 64k satellite connection |
| 16:14.54 | tessier | And don't use substandard equipment. Geez. |
| 16:14.57 | Katty | another dumb question! |
| 16:14.57 | Shido6 | spoon |
| 16:14.59 | Shido6 | what the heck |
| 16:15.04 | ManxPower | Delvar: You'll sign up for the asterisk-cvs mailing list when your Asterisk breaks when you update often enough, because you didn't know about a specific change that impacts you |
| 16:15.07 | Katty | Topic: Dynamic Processing |
| 16:15.25 | Katty | does that mean the processor has queues of commands waiting to be run? |
| 16:15.30 | tessier | ManxPower: Unfortunately, that is indeed a sign that asterisk isn't ready to do anything important. :( |
| 16:15.33 | AgiNamu | hmmhesays well, the answer is yes. |
| 16:15.37 | AgiNamu | so tell them yes and thats it. |
| 16:15.38 | tessier | Assuming stable isn't good enough for you. |
| 16:15.40 | Katty | and it skips the 'job' it's on and looks at the next command? |
| 16:15.41 | Hmmhesays | AgiNamu: i know |
| 16:15.42 | AgiNamu | oh, wait, "yes and it sucks" |
| 16:15.45 | ManxPower | tessier: Only that CVS version is not ready for that |
| 16:15.45 | Delvar | ManxPower: i know :P |
| 16:15.46 | tessier | Unfortunately stable has bitten me too. |
| 16:15.51 | tessier | ManxPower: Indeed. |
| 16:16.00 | AgiNamu | no, you see ugys |
| 16:16.04 | Delvar | ManxPower: i just like supprises |
| 16:16.07 | ManxPower | tessier: Yeah, but the GOAL of stable is NOT be bite people. We are not there yet, but we are getting there. |
| 16:16.10 | AgiNamu | the power of open source is that EVERYONE gets to run their OWN test lab! |
| 16:16.14 | Hmmhesays | 750ms is good latency to some of these people |
| 16:16.21 | fugitivo | tessier: lot of people say that about opensource software |
| 16:16.26 | ManxPower | Delvar: Well don't come crying to me when your users burn you as a witch. |
| 16:16.32 | AgiNamu | Why have a big centralized lab that tests for all sorts of conditions?? Just load it up on your own machine and makes sure it works! |
| 16:16.33 | fugitivo | tessier: lot of people use opensource in their companies |
| 16:16.40 | AgiNamu | you even get to develop your own load scripts and everything. |
| 16:16.41 | Delvar | ManxPower: hmmm roast pork |
| 16:16.43 | AgiNamu | it's great! |
| 16:17.03 | *** join/#asterisk loick (~loick@APuteaux-151-1-37-144.w82-124.abo.wanadoo.fr) |
| 16:17.30 | AgiNamu | now, lets start the discussion |
| 16:17.32 | Hmmhesays | is iax2 trunking working good yet? |
| 16:17.49 | AgiNamu | someone suggest that Asterisk be improved, by say, BVTs and a large test lab. |
| 16:17.53 | AgiNamu | then someone respond that OSS is a gift. |
| 16:18.09 | AgiNamu | then someone lament digium not making enough money to spend on it. |
| 16:18.16 | AgiNamu | and so on. |
| 16:18.38 | queuetue | There si a pretty wide gap between 5.00 and 100.00 - I wonder why there ar eno third-party cards filling in the gaps at 10 and 50 bucks... |
| 16:18.51 | AgiNamu | there are lots |
| 16:18.57 | ManxPower | queuetue: Many reasons. |
| 16:19.01 | AgiNamu | But the quality or stabilty isnt guaranteed. |
| 16:19.06 | *** join/#asterisk kimc (~freenode@pcp09643046pcs.wbrmfd01.mi.comcast.net) |
| 16:19.22 | Hmmhesays | ok there is no good reason to use silence suppression |
| 16:19.23 | ManxPower | The cheap FXO cards will be going away soon, since Intel stopped making them. |
| 16:19.38 | fugitivo | ManxPower: no! oh god no! |
| 16:19.41 | AgiNamu | hmmhe wha? |
| 16:19.44 | chap | Not that they worked spectacularly well, anyway. |
| 16:19.50 | ManxPower | fugitivo: Why do you think Digium stopped selling them? |
| 16:19.59 | fugitivo | i'm going to buy 100 now |
| 16:20.01 | *** join/#asterisk Conductor (~thomas@62.8.240.132) |
| 16:20.14 | *** join/#asterisk MajestiK (~MajestiK@S01060800208687ec.ed.shawcable.net) |
| 16:20.19 | Hmmhesays | AgiNamu: can you think of a good reason to use silence supression? |
| 16:20.27 | AgiNamu | sure, less bandwidth. |
| 16:20.30 | fugitivo | ManxPower: don't know, why? |
| 16:20.36 | *** join/#asterisk Juxt (~Juxt@adsl-068-213-216-087.sip.bct.bellsouth.net) |
| 16:20.39 | ManxPower | fugitivo: the X100P was designed for USA usage and doesn't work correctly in many other places. |
| 16:20.40 | AgiNamu | along the same reason I use G&29 |
| 16:20.44 | Hmmhesays | ok, lets assume you're not using dialup |
| 16:20.52 | fugitivo | ManxPower: it works perfect here |
| 16:20.56 | Juxt | hello everyone |
| 16:21.08 | ManxPower | fugitivo: except that it doesn't detect hangup all the time. |
| 16:21.15 | fugitivo | ManxPower: Argentina, i had problems with hangup detection, but its solved after a talk with my provider |
| 16:21.24 | MatsK | ManxPower: Do you still have a copy of the weatherscript that you could share ? |
| 16:21.38 | tessier | fugitivo: I know lots of people say that about open source software but it usually does not bite me as often as it has with * |
| 16:21.38 | fugitivo | ManxPower: what is a replacement for the x100p? |
| 16:21.45 | Hmmhesays | tdm400 |
| 16:21.49 | Hmmhesays | p |
| 16:21.55 | queuetue | I get a charge out of these telco companies where you have to call some damned salesman to get a price - straight to the "no" bin... |
| 16:21.58 | fugitivo | Hmmhesays: how much is that? |
| 16:22.03 | ManxPower | MatsK: I tared up my entire site and donated it to asteriskdocs.org. you can get the tarball at http://www.fnords.org/~eric/asterisk/wffs.tar.gz |
| 16:22.05 | Hmmhesays | www.digium.com |
| 16:22.08 | ManxPower | fugitivo: The TDM400P |
| 16:22.12 | tessier | queuetue: What do you mean? Price on what? |
| 16:22.16 | MatsK | ManxPower: THX |
| 16:22.16 | *** join/#asterisk ell (~ali@66-207-218-199.beanfield.net) |
| 16:22.17 | fugitivo | ManxPower: how much? |
| 16:22.21 | Hmmhesays | www.digium.com |
| 16:22.29 | ManxPower | fugitivo: about US$125 |
| 16:22.29 | queuetue | tessier, the voicetronix stuff. |
| 16:22.31 | tessier | It's pretty standard that you have to ask a salesperson for pricing on things where the price can be variable. |
| 16:22.33 | AgiNamu | $337 for the TDM400 with 4 FXO |
| 16:22.40 | fugitivo | ManxPower: that's expensive |
| 16:22.50 | Hmmhesays | fugitivo: are you drunk? |
| 16:22.56 | ManxPower | fugitivo: it's only expensive if you've never tried upgrading a commercial PBX. |
| 16:22.57 | queuetue | tessier, There is a "price" button and if you click it, it takes you to a mailto for sales. :) |
| 16:23.17 | Hmmhesays | 337 for 4 fxo is not expensive |
| 16:23.20 | AgiNamu | and you can buy a T1 card for like $500 or so |
| 16:23.29 | queuetue | tessier, "The price is avariable" means "we cheat everyone we can"... |
| 16:23.31 | AgiNamu | and a channelbank for like $1000 or so? |
| 16:23.33 | tessier | Saved a fortune. :) |
| 16:23.37 | *** part/#asterisk ell (~ali@66-207-218-199.beanfield.net) |
| 16:23.40 | AgiNamu | so that brings the price per line down |
| 16:24.20 | Juxt | can someone tell me how does vonage manage faxes over ip? |
| 16:24.30 | Conductor | is it possible to send a sip request with java to find out if a peer is busy or dnd`? |
| 16:24.31 | ManxPower | Juxt: they say they do. |
| 16:24.55 | Juxt | well it has to work |
| 16:25.10 | AgiNamu | they use ulaw and just expect things to be perfect? or they use T38? |
| 16:25.15 | fugitivo | ManxPower: if i want only one fxo for my house, that's expensive |
| 16:25.29 | Katty | Hmmhesays: do gamers like intel or amd? |
| 16:25.39 | Juxt | i don't think their ATA boxes do T38 |
| 16:25.47 | tessier | Juxt: To do fax over IP you need a clean connection and no compression. That's all there is to it. |
| 16:25.58 | tessier | Our fax machine at our office is over IP using an ATA-186 and it works great. |
| 16:25.58 | ManxPower | fugitivo: it's much cheaper than any other FXO PCI option |
| 16:25.59 | fugitivo | Katty: amd 64bit |
| 16:26.00 | Hmmhesays | Katty: depends on the gamer |
| 16:26.08 | AgiNamu | *clean* connectiong being the difficult part |
| 16:26.17 | AgiNamu | yea, I'll never buy AMD |
| 16:26.18 | fugitivo | ManxPower: i can buy the x100p clone for $10 |
| 16:26.20 | AgiNamu | even though the P4 design is shit |
| 16:26.29 | Hmmhesays | Most gamer's I run into prefer amd stuff |
| 16:26.30 | tessier | AgiNamu: Why not? I have been quite impressed with AMD so far |
| 16:26.30 | AgiNamu | I'll wait till there's a nice Pentium D out or something. |
| 16:26.34 | Katty | Hmmhesays: the reason i ask is because amd isn't know for the fpu, which as i understand it, handles all 3d rendering. |
| 16:26.41 | AgiNamu | cause i havent got the bad taste out of my head. |
| 16:26.46 | AgiNamu | of their incompatible chips years ago |
| 16:26.48 | Hmmhesays | the fpu handles all of your floating point math |
| 16:26.50 | fugitivo | Katty: amd 64 bit for sure |
| 16:26.50 | ManxPower | fugitivo: no, you are buying an Intel WinModem that is marketed as an X100P clone for US$10 |
| 16:26.59 | Katty | Hmmhesays: which in turn is most 3d rendering |
| 16:27.00 | AgiNamu | katty, all 3D rendering is done on the video card |
| 16:27.05 | fugitivo | ManxPower: so? it works |
| 16:27.07 | Hmmhesays | Katty: very good young jedi |
| 16:27.13 | ManxPower | fugitivo: you are welcome to write drivers for other WinModem cards. |
| 16:27.18 | tessier | AgiNamu: I'm sure they were trying to be compatible, no? |
| 16:27.19 | Katty | AgiNamu: no it's not |
| 16:27.23 | tzanger | I buy badass video cards to use their VRAM for swap :-) |
| 16:27.39 | AgiNamu | ok, almost all 3d functions are done on the video card |
| 16:27.44 | Katty | AgiNamu: k, better. |
| 16:27.44 | tessier | tzanger: Uh... |
| 16:27.52 | Hmmhesays | games however do not take advantage of amd's different instruction set unless the game is specifically written for an amd processor |
| 16:27.52 | tessier | tzanger: Why not just buy more RAM? :) |
| 16:27.54 | AgiNamu | and not often is the CPU the bottleneck for more frames/sec. |
| 16:27.54 | fugitivo | ManxPower: don't you think people should have a cheap option? |
| 16:28.07 | tzanger | tessier: sometimes that's not an option. and CAD$80 for a 256M video card isn't bad |
| 16:28.09 | Katty | Hmmhesays: do video cards have their own processor? |
| 16:28.13 | Hmmhesays | Katty: yes |
| 16:28.15 | Hmmhesays | the GPU |
| 16:28.22 | AgiNamu | and the GPU is powerful as shit |
| 16:28.25 | Katty | i'm so not a gamer. |
| 16:28.37 | AgiNamu | NVidia's GeForce FX GPU has 3 times the transistors of a P4, for instance. |
| 16:28.39 | Katty | that would explain why there are big heat sinks on video cards ;) |
| 16:28.47 | Hmmhesays | the gpu is powerful because it is designed for a specific task in mind |
| 16:28.53 | AgiNamu | and why some take up two slots, require a molex power connection |
| 16:28.59 | chap | fugitivo: The marketplace should dictate what is available, and what is not. |
| 16:29.00 | Katty | yes, but back to the CPU |
| 16:29.04 | AgiNamu | and why you can easily spend $2000 on a video card. |
| 16:29.06 | Hmmhesays | my voodoo 5500 requires a molex connector |
| 16:29.08 | tessier | Lots of people talk about offloading all kinds of things to the GPU but the time required to code it just isn't worth it. |
| 16:29.17 | DrWho17 | yes it is |
| 16:29.22 | Katty | Hmmhesays: if the cpu using the fpu for numbers that aren't whole... |
| 16:29.24 | tessier | By the time you perfect your GPU code the next generation of cpu's will be faster than using the GPU. |
| 16:29.33 | AgiNamu | tessier, that's not probable. |
| 16:29.36 | tessier | Then they come out with a new GPU chipset and you get to start all over again, always behind the curve. |
| 16:29.45 | Katty | Hmmhesays: and amd's fpu kinda sucks... |
| 16:29.51 | tessier | AgiNamu: That is the very reason why nobody has done it yet. |
| 16:29.59 | AgiNamu | and if you stick with one vendor, say nvidia.... |
| 16:30.03 | tessier | A few people tried and that is exactly what they ran into. |
| 16:30.05 | AgiNamu | it doesnt need a full rewrite every time. |
| 16:30.11 | `Sauron | Katty: Why're you reading about processor design? Just curious. |
| 16:30.25 | Katty | Hmmhesays: what else does the fpu do besides 3d rendering (autocad) stuff? |
| 16:30.27 | tessier | AgiNamu: We don't even have good open source drivers for nvidia. You think someone is gonna write something to make use of the gpu? |
| 16:30.31 | Katty | `Sauron: company wants me a+ certified. |
| 16:30.32 | Hmmhesays | but even by vendor they intruction sets change |
| 16:30.42 | AgiNamu | katty, even in autocad, a lot of the 3d work is offloaded |
| 16:30.47 | AgiNamu | except when doing a final pass |
| 16:30.52 | Katty | AgiNamu: then what does the fpu do? |
| 16:30.53 | `Sauron | Katty: the FPU does all the floating poing math, whether it's cad/3d related or not. |
| 16:30.59 | tzanger | AgiNamu: our drafting guy shave badass CPUs and badass video cards |
| 16:31.05 | AgiNamu | i.e. when doing rendering in 3d studio, mental ray, etc. |
| 16:31.06 | Katty | `Sauron: k, that doesn't tell me much. |
| 16:31.11 | tzanger | it's not like hte old days of using 387emu to gat acad 10 to work on my 386 :-) |
| 16:31.13 | `Sauron | <Katty> Hmmhesays: what else does the fpu do besides 3d rendering (autocad) |
| 16:31.13 | `Sauron | <PROTECTED> |
| 16:31.15 | tessier | If you add 1.0 and 1.0 to get 2.0 that's a job for the fpu. |
| 16:31.21 | `Sauron | Answered your question. :) |
| 16:31.24 | Hmmhesays | Katty: google on the definition of a floating point operation |
| 16:31.28 | Katty | tessier: no. the alu handles whol enumbers. |
| 16:31.28 | Hmmhesays | you'll understand |
| 16:31.29 | AgiNamu | There's 2 kinds. rendering realtime 3d with opengl, direct3d, etc. |
| 16:31.37 | sudhir492 | what port does IAX2 use? |
| 16:31.37 | Katty | Hmmhesays: k |
| 16:31.42 | AgiNamu | 4569 |
| 16:31.47 | tessier | Katty: The ALU handles 1 and 1 to get 2 |
| 16:31.47 | sudhir492 | thx |
| 16:31.47 | AgiNamu | and rendering some file to an image or likewise. |
| 16:31.53 | tessier | 1.0 is a float. :) |
| 16:31.53 | `Sauron | Basically, for a CPU to do integer math, is easy. |
| 16:32.12 | tzanger | a CPU can do any kind of math easily... at least what it's been designed for |
| 16:32.20 | `Sauron | However, floating point calculations would have to be turned into integer calculations, calculated, then turned back into FP |
| 16:32.30 | tzanger | there are DSPs that can throw around floating point numbers faster than any CPU can handle integers |
| 16:32.32 | *** join/#asterisk implicit (~implicit@ip68-5-148-1.oc.oc.cox.net) |
| 16:32.37 | `Sauron | So they created FPU's to offload the floating point calculations to |
| 16:32.39 | Hmmhesays | which isn't so bad if you are doing 1.039/2.017 |
| 16:32.47 | tessier | tzanger: Unfortunately they have the same problem as GPU's in graphics cards |
| 16:32.53 | tzanger | tessier: depends |
| 16:32.59 | tzanger | DSPs are dark, dark juju |
| 16:33.02 | tessier | Nobody uses dsp's in PC's because they take forever to program and by the time you have the new generation of cpu has overtaken your dsp. |
| 16:33.07 | tessier | Yes, there are exceptions |
| 16:33.13 | AgiNamu | yes, there are. :P |
| 16:33.16 | DrWho17 | yes, like sound cards |
| 16:33.18 | tessier | But I have never personally run into anyone who used dsp hardware. They are extremely rare. |
| 16:33.26 | Hmmhesays | unless you are written embedded stuff of course |
| 16:33.27 | DrWho17 | which all use programmable DSP's |
| 16:33.27 | tzanger | the first foray I ever took into fixed-point math was for sound |
| 16:33.28 | tessier | I am talking about a special DSP board to offload calculations onto. |
| 16:33.31 | Hmmhesays | *writing even |
| 16:33.38 | tzanger | I wrote an S3M player |
| 16:33.42 | `Sauron | fpga > dsp's ;) |
| 16:33.42 | tessier | You are right that sound cards have a dsp. As do modems etc. iirc |
| 16:33.43 | AgiNamu | At IPTEL show last year there was a QuadT1 card with DSPs for $2K |
| 16:33.47 | Hmmhesays | say you are writing an ap for say ... and dreamcast |
| 16:33.53 | implicit | `Sauron, ? |
| 16:33.54 | Hmmhesays | good lord I can't type today |
| 16:33.56 | AgiNamu | or maybe it was an 8-T1 |
| 16:34.07 | Hmmhesays | *app, a dreamcast |
| 16:34.09 | tessier | Lack of DSP is probably the biggest fault of the digium T1/E1 cards. |
| 16:34.12 | tzanger | whatever happened to atacomm's tdm card? |
| 16:34.17 | DrWho17 | tessier: sure, just wished Digium had them |
| 16:34.24 | tzanger | tessier: not really.... lack of even a tiny buffer is their single biggest failing |
| 16:34.28 | tessier | Nowadays I'm afraid I always resort to Cisco for that. |
| 16:34.33 | tessier | tzanger: Good point. |
| 16:34.45 | `Sauron | Does anyone know of PRI/T1 cards that have DSP's on them? |
| 16:34.51 | Hmmhesays | haha damn openvpn it won't start |
| 16:34.51 | tessier | tzanger: Your system better be real time :) |
| 16:34.52 | DrWho17 | Sauron: sure |
| 16:34.53 | `Sauron | AgiNamu: Do you know what company had that card? |
| 16:34.54 | tzanger | even a tiny 5 or 10ms buffer would be soooo much nicer |
| 16:34.57 | implicit | `Sauron, just get a media gateway |
| 16:34.58 | AgiNamu | I forget the name :\ |
| 16:35.03 | DrWho17 | (not sure if it would be appropriate to name them) |
| 16:35.10 | DrWho17 | her |
| 16:35.12 | DrWho17 | here |
| 16:35.12 | tessier | DrWho17: Sure it would! |
| 16:35.21 | AgiNamu | of course |
| 16:35.27 | tessier | DrWho17: We're all looking for the best solution. Sometimes digium isn't it. |
| 16:35.29 | `Sauron | implicit: Nope. I need the DSP channelised PRI, so I can do fax stuff with it properly. |
| 16:35.35 | tessier | I bet even Mark would tell you that. |
| 16:35.41 | implicit | DrWho17, why would it not be appropriate? |
| 16:35.41 | DrWho17 | sangoma |
| 16:35.55 | DrWho17 | implicit: Digium sells hardware |
| 16:35.57 | implicit | sangoma doesn'th ave any card like that on the market or completed yet |
| 16:35.57 | tzanger | sangoma makes damn fine TDM hardware |
| 16:36.09 | implicit | DrWho17, so why does that make you digiums bitch ? :) |
| 16:36.17 | implicit | DrWho17, microsoft sells software |
| 16:36.32 | tessier | implicit: He's afraid kram is going hungry. |
| 16:36.36 | implicit | lol |
| 16:36.40 | implicit | :) |
| 16:36.44 | DrWho17 | yea, well I wouldn't promote linux software on a microsoft channel either |
| 16:36.53 | tessier | We're gonna see Sally Struthers standing in front of TV cameras on his front yard with a really sad face.... |
| 16:36.56 | Hmmhesays | I prefer my isa winmodem |
| 16:36.56 | implicit | well this is an asterisk channel |
| 16:36.59 | implicit | not a digium channel |
| 16:37.03 | `Sauron | DrWho: Last I looked, I couldn't figure out if the sangoma cards would show up as serial ports in linux |
| 16:37.05 | sudhir492 | I finally managed to get Asterisk on FC3 !!! |
| 16:37.15 | tzanger | ... |
| 16:37.19 | sudhir492 | including OH323 |
| 16:37.24 | tzanger | ISA *win*modem? There were controllerless modems on ISA? |
| 16:37.25 | tessier | sudhir492: You stud. |
| 16:37.29 | DrWho17 | Sauron: I don't have any, but I mentioned DSP's on the list, and had a couple vendors contact me off list |
| 16:37.30 | tessier | OH323 is heinous. |
| 16:37.36 | tessier | I am trying like hell not to touch h323. |
| 16:37.36 | `Sauron | Hum. |
| 16:37.41 | Hmmhesays | tzanger: i was just throwing words together |
| 16:37.44 | DrWho17 | Sangoma was one, and the salesman said they had DSP's |
| 16:37.51 | tessier | Been burned by it too many times in the past. |
| 16:37.59 | tzanger | sangoma's cards disable the DSP for asterisk use IIRC |
| 16:38.09 | SPoon_TSX | Hello there, Do anyone having any problem on incoming call drop on first 3 seconds of conversation? |
| 16:38.09 | `Sauron | Which is sad |
| 16:38.16 | sudhir492 | tessier: I have no choice but to use H323. Either use chan_h323 or oh323. |
| 16:38.24 | DrWho17 | well, anyway I'm moving to TNT's |
| 16:38.32 | Juxt | so how realistic would it be to send a fax on ulaw at say 9600 baud |
| 16:38.35 | Hmmhesays | sudhir492: You can fix it! you have the technology! |
| 16:38.36 | DrWho17 | so it shouldn't be an issue anymore |
| 16:38.46 | tessier | sudhir492: Unfortunately that is often the case. Other equipment doesn't do sip sometimes. |
| 16:38.54 | cbachman | I have a page from a magazine on my wall with Sally Struthers: Thinking of running your critical apps on NT? Isn't there enough world suffering? |
| 16:39.04 | tzanger | hahaha |
| 16:39.09 | dwmw2 | Juxt: that works for me -- my ISDN is ulaw and receives faxes with spandsp OK |
| 16:39.22 | implicit | juxt: y not t.38 |
| 16:39.24 | tessier | cbachman: Nice! |
| 16:39.35 | Hmmhesays | spandsp works great with ulaw if you have a crossover cable connected between your voip hardware and the asterisk unit |
| 16:39.36 | tzanger | I have a page from a TIME magazine ... it shows a square bicycle wheel and it says "square wheel" under it |
| 16:39.36 | Juxt | cause i'll never find an ata that supports t.38 |
| 16:39.37 | sudhir492 | Hmmhesays: fix what? |
| 16:39.45 | cbachman | tessier, it's an advertisement from Sun :-) |
| 16:39.47 | tzanger | and to the right of it is the exact same tire with "e-squarewheel.com" under it |
| 16:39.47 | implicit | many providers accept it |
| 16:39.50 | Hmmhesays | sudhir492: i was making a reference to an old american television show |
| 16:39.57 | tzanger | and at the bottom of the ad it says "putting a bad idea on the internet doesn't make it better" |
| 16:40.00 | *** join/#asterisk jabular (~jabular@82-32-105-84.cable.ubr02.hawk.blueyonder.co.uk) |
| 16:40.11 | dwmw2 | Hmmhesays: well, I don't have VoIP involved at all in fax reception |
| 16:40.25 | tzanger | and on my desk here at work I have an ad ripped out of a mag that says "tired of all the e-bs?" |
| 16:40.26 | implicit | juxt: yes u can |
| 16:40.26 | Juxt | implicit: you mean many providers accept t.38? |
| 16:40.31 | implicit | yes |
| 16:40.33 | Hmmhesays | I don't have any in production.... just on my test bench |
| 16:40.47 | Juxt | implicit: which ata support t38? |
| 16:40.47 | implicit | many of the ones i use |
| 16:40.48 | tzanger | it's right beside the one that says "what kind of maniac creates something when no one is asking for it?" |
| 16:41.10 | Hmmhesays | who needs more than 64k of memory |
| 16:41.56 | *** join/#asterisk jeffik (~jeffik@CPE00c049565af7-CM0012256ead9e.cpe.net.cable.rogers.com) |
| 16:42.11 | `Sauron | Hum. |
| 16:42.30 | `Sauron | anyone remember the Digi RasFire cards? |
| 16:42.45 | epoch | Hmmhesays: it was 640k :) |
| 16:42.52 | *** join/#asterisk t0pCop (t0pCop@221.128.101.73) |
| 16:42.56 | Hmmhesays | ohwell, my memory fails me |
| 16:43.07 | epoch | you only have 64k apparently |
| 16:43.08 | epoch | ;) |
| 16:43.10 | Juggie | who remembers VLB and computer programs on casette tapes :) |
| 16:43.14 | *** join/#asterisk [Outcast] (~knoppix@209.213.205.178) |
| 16:43.17 | sudhir492 | Hmmhesays: sorry, I missed the punchline |
| 16:43.27 | `Sauron | Well, they were dense digital modem cards |
| 16:43.36 | Hmmhesays | sudhir492: that was the punchline, it was a lame joke |
| 16:43.42 | [Outcast] | does anyone know how to get a sipura to send a disconnect signal to phone? |
| 16:43.49 | epoch | it's not a joke, it's a bill gates quote! :) |
| 16:43.56 | sudhir492 | tessier: are you using chan_h323? |
| 16:44.03 | tessier | sudhir492: Nope. |
| 16:44.04 | `Sauron | 2 or 4 port PRI cards, that turned into /dev/ttyG* in linux |
| 16:44.11 | tessier | I tried to use it a few times. |
| 16:44.25 | `Sauron | and I have to replace 2 of them in the next 9 months, but it's a pain to find a replacement card.. .:/ |
| 16:44.35 | *** join/#asterisk ell (~ali@66-207-218-199.beanfield.net) |
| 16:44.41 | sudhir492 | what happened? |
| 16:44.45 | AgiNamu | i'd like to get T.38 onto the PA168 |
| 16:44.50 | AgiNamu | THAT'd kick ass. |
| 16:45.13 | *** join/#asterisk MuppetMaster (~MuppetMas@a82-92-73-185.adsl.xs4all.nl) |
| 16:45.40 | AgiNamu | i remember tape loaded programs. that sucked. |
| 16:45.49 | *** join/#asterisk Cheng29 (~cheng29@d57-87-253.home.cgocable.net) |
| 16:46.06 | *** join/#asterisk Wazb (Wazb@207.245.215.111) |
| 16:46.10 | Juxt | tape loaded programs rocked :-) |
| 16:46.29 | Wazb | Hi |
| 16:46.30 | zoa | and then transmitting those over radio waves |
| 16:46.31 | zoa | :) |
| 16:47.15 | AgiNamu | haha, i fixed another PA168 IAX2 bug |
| 16:47.16 | AgiNamu | wai wai! |
| 16:47.43 | Wazb | i am new to Asterisk , i am getting problem installing OpenH323 driver in Astersik , can nayone hlep me |
| 16:47.43 | *** join/#asterisk JerJer[mobile] (~nonyobizn@65.173.197.174) |
| 16:48.16 | sudhir492 | Asterisk works well on FC3 with iptables disabled, need to figure out what to change in the default settings |
| 16:49.54 | sudhir492 | JerJer[mobile]: what versions of PWLib and OH323 are needed for chan_h323? The Readme file in h323 points to versions of both which are not there on sourceforge :-( |
| 16:50.17 | file[laptop] | AgiNamu: do you have the source or something? |
| 16:51.18 | tessier | I'm pretty sure all of #asterisk is tired of hearing about how crappy h323 support is. :( |
| 16:51.39 | tessier | If you need h323 asterisk probably isn't the solution to your problem. |
| 16:52.13 | jakepdev | it could be one day |
| 16:52.27 | tessier | Indeed. |
| 16:52.28 | ManxPower | sudhir492: That is the version you need. |
| 16:52.43 | JerJer[mobile] | if people would follow thru on their bug reports H.323 support would be better |
| 16:52.49 | JerJer[mobile] | and proper motivation is another factor |
| 16:52.52 | ManxPower | sudhir492: http://www.nufone.net/downloads/ |
| 16:53.03 | *** join/#asterisk mbranca_home (~matteo@host-84-222-23-239.cust-adsl.tiscali.it) |
| 16:53.07 | bkw_ | JerJer[mobile], and people willing to pay |
| 16:53.11 | bkw_ | and say thanks helps too |
| 16:53.19 | bkw_ | most people go "give me give me give me" and never give back |
| 16:53.24 | jakepdev | thanks bkw |
| 16:53.28 | *** join/#asterisk pluto70 (~me@80.70.179.76) |
| 16:54.01 | scoof | does anybody know if chan_sccp has some mailinglists, or is it all going on on the asterisk-lists? |
| 16:54.50 | JerJer[mobile] | ManxPower: that is only for the highly broken code on -stable |
| 16:54.51 | *** join/#asterisk elric (~kavit@ppp114-10.static.internode.on.net) |
| 16:55.44 | *** part/#asterisk ell (~ali@66-207-218-199.beanfield.net) |
| 16:55.45 | ManxPower | I'll send drumkilla a request to remove chan_h323 from -stable. |
| 16:56.17 | AgiNamu | file: yes. |
| 16:56.22 | ManxPower | since it's broken |
| 16:56.33 | AgiNamu | it wasn't handling a secondary call coming in. |
| 16:56.58 | AgiNamu | instead, it rejected the second call, and then reset itself, killing (but not disconnecting) the current call |
| 16:57.06 | sudhir492 | bkw_: I have not been able to say thanks to you and so many developers of asterisk personally, but I appreciate very much. Thanks a lot. |
| 16:57.13 | bjohnson | bkw_: I want to take this opportunity to thank you for all of your hard work and contributions to the software that I use for free |
| 16:57.50 | file[laptop] | AgiNamu: ah |
| 16:58.22 | *** join/#asterisk [cc]smart (~smart@gw.ptr-62-65-149-158.customer.ch.netstream.com) |
| 16:58.24 | MikeJ[Laptop] | bkw, gimmie gimmie gimmie |
| 16:58.29 | sudhir492 | Everytime I talk to digium folks, and I buy all the cards from Digium, I always express my appreciation. Once again, THANK YOU VERY MUCH!!! |
| 16:58.35 | Juxt | so which ata supports t.38 - i already checked out like 10 of them |
| 16:58.46 | `Sauron | Humm. |
| 16:58.48 | `Sauron | Manx |
| 16:59.06 | *** join/#asterisk mbaron (~mbaron@AVelizy-154-1-42-83.w82-124.abo.wanadoo.fr) |
| 16:59.08 | Wazb | Can anyone help me what is chan_h323 |
| 16:59.13 | Essobi | hah |
| 16:59.21 | jakepdev | wazb - you're funny |
| 16:59.34 | bjohnson | Juxt: I don't think any of them do |
| 17:00.23 | AgiNamu | I'm enamoured with the Pa168 |
| 17:00.27 | `Sauron | For those who heard my earlier PRI/dsp card question.. I'm looking for a digital modem card with 1 or 2 T1 ports, that's linux supported. |
| 17:00.37 | Wazb | i told you guys in beg. i am new to Asterisk |
| 17:00.44 | `Sauron | All my google searching brings me to old postfix-users and hylafax list messages I've written. :( |
| 17:00.45 | AgiNamu | Juxt, why not add T38 support to the PA168? |
| 17:00.47 | sudhir492 | Wazb: what you dont know will not hurt you :-) |
| 17:01.18 | Wazb | Please guide me ... |
| 17:02.07 | jakepdev | Wazb - h323 doesn't work on * reliably |
| 17:02.17 | Juxt | AgiNamu: isn't it closed source? |
| 17:02.35 | sudhir492 | Wazb: if you have no choice, but to use H323, then chan_h323 or oh323 is there to your rescue |
| 17:02.57 | AgiNamu | juxt, no. |
| 17:03.21 | *** join/#asterisk topping (~topping@dhcp024-210-082-196.columbus.rr.com) |
| 17:03.23 | jakepdev | I take issue with "to your rescue" :) |
| 17:03.41 | *** join/#asterisk expousr (~expousr@66.101.10.149) |
| 17:03.42 | sudhir492 | jakepdev: you may have a point! |
| 17:03.46 | *** join/#asterisk santiago (~santiago@63.245.86.116) |
| 17:03.50 | Juxt | Although the code is closed-source, it's pretty easy (compared to most closed-source products) to obtain redacted source for the firmware. |
| 17:04.09 | ManxPower | H323 is NOT for beginners |
| 17:04.12 | AgiNamu | actually, parts of it is open source. |
| 17:04.19 | twisted[work] | H323 is not for anyone :P |
| 17:04.34 | twisted[work] | I would enjoy seeing it go up in a puff of smooke |
| 17:04.36 | twisted[work] | smoke, too. |
| 17:04.39 | Hmmhesays | as much as I don't like h.323, that's not true twisted |
| 17:04.48 | JerJer[mobile] | jakepdev: H.323 DOES work reliably, if you know WTF you are doing |
| 17:04.53 | JerJer[mobile] | and follow the README |
| 17:05.00 | JerJer[mobile] | and run -head |
| 17:05.11 | twisted[work] | Hmmhesays, learn how to decypher emoticons :) |
| 17:05.18 | jakepdev | Jeremy - I've been advised by many who tried and failed |
| 17:05.28 | Hmmhesays | ;) no way twisted |
| 17:05.43 | sudhir492 | jakepdev: I have not had any experience with chan_h323 yet. I have used oh323 which has its warts. |
| 17:06.03 | JerJer[mobile] | jakepdev: tell that to the 20+ systems i've setup with chan_h323 |
| 17:06.17 | Hmmhesays | both chan_h323 and chan_oh323 have worked well for me in the past |
| 17:06.29 | *** join/#asterisk clint_ (~clint@snap.helixsystems.com) |
| 17:06.53 | *** join/#asterisk ctooley (~ctooley@rrcs-24-153-228-6.sw.biz.rr.com) |
| 17:07.11 | sudhir492 | jakepdev: However, after some experince, I have been able to use oh323 with enough reliability. I would guess chan_h323 should be same |
| 17:07.31 | jakepdev | qhat about the reports of lockups after 200 calls? |
| 17:07.45 | ManxPower | Nobody ever said chan_h323 was EASY for a beginner. |
| 17:07.46 | JerJer[mobile] | absolutely cannot duplcate |
| 17:07.51 | jakepdev | incorrect configs? |
| 17:08.08 | sudhir492 | jakepdev: I have not had that problem |
| 17:08.24 | JerJer[mobile] | i even ssh'd into one persons box that was bitching and he very smply did not have the right versions of Open H.323 |
| 17:08.28 | JerJer[mobile] | ala did not follow the readme |
| 17:08.42 | JerJer[mobile] | *version |
| 17:09.04 | Hmmhesays | who follows the readme's it's more interesting if you don't |
| 17:09.18 | Shido6 | wow |
| 17:09.29 | jakepdev | greg - do you agree? |
| 17:09.40 | ManxPower | I'm sure that's pretty common, JerJer[mobile]. Most software does NOT need specific minor versions of the libs it links too. On the other hand, most applications don't link to succh buggy libs. |
| 17:09.44 | Shido6 | those who read the readmes make the 85/hr |
| 17:09.55 | ManxPower | Shido6: Only $85? |
| 17:09.57 | *** part/#asterisk santiago (~santiago@63.245.86.116) |
| 17:09.58 | jakepdev | those who don't pay the $85 an hour |
| 17:10.47 | Shido6 | those who dont pay, read and struggle through it - nothing against that but it helps to have someone assist if you're demoing something or need something withint a specific time frame |
| 17:11.00 | Shido6 | it does have a learning curve |
| 17:11.03 | pigpen | Just found out that you can port your vonage phone numbers away to a different provider... |
| 17:11.06 | *** join/#asterisk robl^ (~robl@dsl093-025-118.hou1.dsl.speakeasy.net) |
| 17:11.22 | Shido6 | but most of the time, it just takes a little reading and people generally dont want to read |
| 17:11.23 | *** join/#asterisk tessier (~treed@210.245.100.67) |
| 17:11.29 | *** part/#asterisk queuetue (~Scott@h69-21-252-54.69-21.unk.tds.net) |
| 17:11.31 | ManxPower | I'm a BIG fan of spending other people's money to fix problems. |
| 17:11.44 | JerJer[mobile] | ManxPower: it iis not the libraries fault this time - H.323 is a whore of a protocol |
| 17:11.44 | tzanger | ManxPower: :-) |
| 17:12.00 | *** part/#asterisk mbaron (~mbaron@AVelizy-154-1-42-83.w82-124.abo.wanadoo.fr) |
| 17:12.17 | *** join/#asterisk zaptel (~just@216.194.173.2) |
| 17:12.17 | ManxPower | JerJer[mobile]: Then why do you require EXACT versions for the libs? |
| 17:12.18 | pigpen | any way to have * ignore caller id info from a particular fxo sip client? |
| 17:12.31 | Shido6 | LOL |
| 17:12.35 | ManxPower | pigpen: callerid= in sip.conf |
| 17:12.37 | Shido6 | quote of the day |
| 17:12.56 | pigpen | hmm...I will give it a shot. |
| 17:13.09 | JerJer[mobile] | ManxPower: all software has dependencies |
| 17:13.31 | ManxPower | JerJer[mobile]: Yeah, but asterisk does not depend on an exact version of openssl. |
| 17:13.42 | JerJer[mobile] | and people kept bitchng that chan_h323 didn't work and 90% of the time they had some crazy old distro inistalled version |
| 17:13.45 | JerJer[mobile] | whch will not work |
| 17:13.56 | ManxPower | Obviously many things change between even minor versions of OpenH323 |
| 17:13.58 | JerJer[mobile] | so we elected to force the makefile to require specific versions |
| 17:14.43 | ManxPower | So the OpenH323 people are either idiots (changing major things in minor releases) or OpenH323 is very buggy. |
| 17:14.56 | *** part/#asterisk nesys (~nesys@81-174-12-111.f5.ngi.it) |
| 17:15.03 | JerJer[mobile] | more like H.323 is a shitty protocol with very very limited interoperability |
| 17:15.23 | ManxPower | JerJer[mobile]: Any way to put the exact openh323/pwlib version in the test that shows in "show modules"? |
| 17:15.40 | *** join/#asterisk Gerrath (~Gerrath@shanev.lifecor.com) |
| 17:15.40 | sudhir492 | JerJer[mobile]: on nufone.net/downloads I see openh323 1.12.2 and pwlib 1.5.2. The Readme in h323 asks to use Open H.323 v1.15.1 and PWLib v1.8.1 which are not there on sourceforge :-( What is the way out? |
| 17:15.41 | ManxPower | test = text |
| 17:15.43 | pigpen | ManxPower: looks like it is a sipura issue |
| 17:16.18 | pigpen | If I call the fxo number...with vonage sending caller id info...it never passes it to * |
| 17:16.27 | *** join/#asterisk outtolunc (~me@adsl-66-218-53-170.dslextreme.com) |
| 17:16.28 | pigpen | If I call it with *67...works fine. |
| 17:16.47 | *** part/#asterisk Gerrath (~Gerrath@shanev.lifecor.com) |
| 17:19.14 | JerJer[mobile] | sudhir492: which is for cvs -head |
| 17:19.17 | JerJer[mobile] | er |
| 17:19.21 | JerJer[mobile] | which is NOT for cvs -head |
| 17:21.35 | *** join/#asterisk Kel (~kel@dsl39.barrvtel.sover.net) |
| 17:21.44 | JerJer[mobile] | looks like they pulled that version for some reason |
| 17:21.46 | JerJer[mobile] | leme update |
| 17:22.37 | sudhir492 | JerJer[mobile]: Will you please tell me a compatible version of Asterisk, OpenH323 and PWlib so that I am able to use chan_h323? I will appreciate that very much! |
| 17:24.23 | *** join/#asterisk Dovid (~hirisk@pool-141-150-21-15.mad.east.verizon.net) |
| 17:24.23 | pigpen | Is there negatives with me hooking up a pri to a cisco 7507 and passing the voip over to the * box? |
| 17:24.25 | *** part/#asterisk Dovid (~hirisk@pool-141-150-21-15.mad.east.verizon.net) |
| 17:24.28 | *** join/#asterisk Dovid (~hirisk@pool-141-150-21-15.mad.east.verizon.net) |
| 17:24.31 | ManxPower | pigpen: yes. |
| 17:24.32 | pigpen | I have a big router...and lots of ports |
| 17:24.38 | pigpen | what would that be? |
| 17:24.46 | ManxPower | pigpen: you have lots of Cisco voice cards in your box? |
| 17:24.56 | ManxPower | pigpen: You can't control the PRI using Asterisk. |
| 17:25.07 | *** join/#asterisk kingcobra (~mwehner@214.35.233.64.transedge.com) |
| 17:25.15 | pigpen | ah..so we would need some voice cards for the 7507 |
| 17:25.24 | ManxPower | pigpen: most people with T-1 ports on their Ciscos do NOT have cards that support voice. |
| 17:25.42 | ManxPower | pigpen: if you want to terminate voice in the Cisco you need Voice cards and DSP cards |
| 17:25.46 | *** join/#asterisk Dovid (~hirisk@pool-141-150-21-15.mad.east.verizon.net) |
| 17:25.53 | tzanger | yeah I'll terminate voice to my AS5248s. :-) |
| 17:25.57 | pigpen | ok..I am pretty sure we don't... |
| 17:26.38 | pigpen | ok..so best bet a digium card? |
| 17:26.43 | ManxPower | pigpen: yes |
| 17:26.46 | pigpen | k |
| 17:26.51 | Dovid | hello |
| 17:26.53 | ManxPower | or Voicetronix or Sagnoma |
| 17:27.12 | Dovid | i am trying to make zaptal and i am getting the following error can anyone help ? |
| 17:27.32 | Dovid | "/usr/include/linux/proc_fs.h:193: warning: `create_proc_read_entry' declared `static' but never defined" |
| 17:28.50 | pigpen | ManxPower: ok..lets say I have dsp...and disadvantages? |
| 17:28.57 | JerJer[mobile] | DIGIUM |
| 17:29.04 | JerJer[mobile] | not supported |
| 17:29.07 | JerJer[mobile] | pigpen: ^ |
| 17:29.21 | pigpen | Digium is not supported?? :) |
| 17:29.25 | JerJer[mobile] | dsp |
| 17:29.26 | pigpen | yeah...true. |
| 17:29.38 | bonez39 | anyone here using Skype? just read about it, curious what others have to say |
| 17:29.42 | JerJer[mobile] | voicetronix is a joke |
| 17:29.57 | JerJer[mobile] | and i've heard some serious horror stories about sagnoma's card |
| 17:30.38 | Hmmhesays | dig deep enough everyone has a horror story |
| 17:31.00 | MuppetMaster | bonez39: Yes. |
| 17:31.10 | Dovid | anyone that can help ? |
| 17:31.11 | JerJer[mobile] | at least Digium stands behind their hardware and asterisk |
| 17:31.14 | *** join/#asterisk johnnyb (~johnnyb@sdsl-38-17-139.tulsaconnect.com) |
| 17:31.17 | Damin | http://www.lightreading.com/document.asp?site=lightreading&doc_id=71020 |
| 17:31.29 | pigpen | I know this kinda goes against the grain..but anyone look into Atacomm's 4 port card? |
| 17:31.34 | Damin | I got quoted.. |
| 17:31.54 | MuppetMaster | bonez39: The strength of Skype is it's ease of use and ability to slice through firewalls (although IAX does that quite nicely as well). |
| 17:31.55 | bonez39 | MuppetMaster: does it work well for you? |
| 17:32.10 | MuppetMaster | bonez39: It does work fine for me, although I don't like having to have a computer around to use the phone. |
| 17:32.24 | MuppetMaster | bonez39: Skype = casual chats, Asterisk = business. |
| 17:32.26 | bonez39 | MuppetMaster: I was reading that skype encrypts all the communication..so I assume the government will soon want to gut it and disable it, if they could... |
| 17:32.33 | johnnyb | Is there a way to get AGI scripts to terminate if the other party hangs up? |
| 17:32.47 | Hmmhesays | blah skype's ability to slice through firewalls is dependant on people using skype with public ip's acting as a proxy |
| 17:32.54 | MuppetMaster | johnnyb: If you are listening for events on the Manager API, then yes. |
| 17:33.04 | bonez39 | MuppetMaster: do you refer to the IAXy device to connect phone to ethernet or cable ? |
| 17:33.10 | MuppetMaster | Hmmhesays: Absolutely, and there are obviously enough of those. |
| 17:33.16 | johnnyb | MuppetMaster: I'm not familiar with the Manager API -- just using Asterisk::AGI |
| 17:33.25 | Hmmhesays | apparently there is |
| 17:33.28 | MuppetMaster | bonez39: I am referring to the IAX2 protocol, of which the IAXy supports. |
| 17:33.35 | bonez39 | ok.... |
| 17:33.48 | Hmmhesays | it's no different than having asterisk act as a proxy to slice through the same firewall |
| 17:34.07 | MuppetMaster | Hmmhesays: Skype is not a silver bullet, there is no magic about slicing through firewalls. But, with folks out there unwittingly sharing their bandwidth/resources, they seem to provide a compelling service. |
| 17:34.26 | Hmmhesays | yeah, it's good thing most folks don't read the documentation |
| 17:34.31 | *** join/#asterisk zerver (~zerver@dsl-200-78-50-248.prod-infinitum.com.mx) |
| 17:34.35 | MuppetMaster | Hmmhesays: And for your typical user, Skype is loads easier and distributed, so the cost of adding additional users for Skype is next to zero. |
| 17:34.38 | Hmmhesays | I admit it... I use skype |
| 17:34.43 | Hmmhesays | and it works well |
| 17:34.45 | MuppetMaster | Hmmhesays: Yes, I make sure I firewall it. |
| 17:34.59 | MuppetMaster | Hmmhesays: Have a look on the forums over at Skype, I try to keep them honest. |
| 17:35.14 | Hmmhesays | I leave mine running on a public ip at night when I leave work |
| 17:35.25 | MuppetMaster | Hmmhesays: What a nice person you are. |
| 17:35.31 | MuppetMaster | Hmmhesays: Bandwidth for the masses. |
| 17:35.50 | Hmmhesays | don't get me wrong, it is good, but there is nothing really specially about it |
| 17:35.56 | Hmmhesays | *really special |
| 17:36.01 | MuppetMaster | johnnyb: Just a moment, here are some details... |
| 17:36.57 | MuppetMaster | johnnyb: Here is a description of the Manager API: http://www.voip-info.org/wiki-Asterisk+manager+API |
| 17:37.14 | Hmmhesays | the manager is pretty slick... I hear there is talk of changing it though |
| 17:37.17 | MuppetMaster | johnnyb: And if you are developing in Java here is a great library: http://asterisk-java.sourceforge.net/ |
| 17:37.30 | *** join/#asterisk [pkh] (hannah@host-84-9-128-193.bulldogdsl.com) |
| 17:37.39 | *** join/#asterisk malcolmd (~malcolmd@malcolmd.digium.sponsor.pdpc) |
| 17:38.00 | MuppetMaster | johnnyb: There are great examples in there on how to listen for events. And if you use the FastAGI with that Java implementation, you may open the Manager API once, listen for events, and stop your AGI transactions based on certain events. |
| 17:38.27 | johnnyb | MuppetMaster: So I would have an AGI, and a manager script which kills my AGI on hangup? |
| 17:38.33 | MuppetMaster | johnnby: I have written some Java apps with it to open MySQL and Jabber once. As with PHP I was having to do it with each launch of an AGI. |
| 17:38.48 | *** part/#asterisk _|ms|_ (~mstremer@p83.129.1.149.tisdip.tiscali.de) |
| 17:38.56 | *** join/#asterisk SkySky (~Miranda@host6614613596.biz.tor.fcibroadband.com) |
| 17:39.42 | MuppetMaster | johnnyb: Depending on your implementation. You would have a FastAGI server, that is invoked when you send it an AGI request from your dialplan. That same FastAGI server (which is a Java app) is listening for all events on the ManagerAPI. When an event (ie - hangup) occurs on the channel that you are processing in your AGI, you could then exit the AGI. |
| 17:40.02 | MuppetMaster | Of course you could do it all in C/C++ as well. |
| 17:40.31 | Hmmhesays | you can do it in any language that supports sockets |
| 17:40.37 | SkySky | hi, i wonder how would the priority goes if i dial to multiple ppl? |
| 17:40.53 | MuppetMaster | SkySky: Depends on how you write the dialplan. |
| 17:41.47 | *** join/#asterisk harryvv (~leonardo@S010600055d210201.vs.shawcable.net) |
| 17:43.27 | SkySky | ie. I have exten => s,1,Dial(SIP/100&SIP/101,15,tr). if all of them are unavailable.. priority goes to 102 right? wat if 1 is disconnected, and one is available? etc.etc.. |
| 17:44.05 | harryvv | sip show peers skysky |
| 17:44.10 | MuppetMaster | SkySky: On your first assumption yes. On the second, it will ring the one that is connected. |
| 17:44.33 | SkySky | and priority would stay in 1 until 15 seconds after then goes to 2 right? |
| 17:44.38 | MuppetMaster | SkySky: And report in the CLI that it failed to create a SIP channel on the first one. |
| 17:44.45 | MuppetMaster | SkySky: Correct. |
| 17:45.24 | SkySky | icic.. thank you very much^^ (because im planning to make a queue for a group, thx for the help!!!) |
| 17:45.53 | kingcobra | i have two pstn lines, currently connected to * via two X100Ps; i'd like to get away from the X100Ps; are two SPA-3000s still a good way to go? or is there better hw out there now? |
| 17:46.31 | [pkh] | off-topic a bit, but does anyone know of a channel where I could get information on using kphone for making outgoing external calls? |
| 17:46.57 | *** join/#asterisk JerJer[mobile] (~nonyobizn@65.173.197.174) |
| 17:47.38 | *** join/#asterisk eXoR` (~exor@xdsl-84-44-146-24.netcologne.de) |
| 17:47.41 | harryvv | google would know |
| 17:48.23 | [pkh] | harryvv: been looking. just thought I'd ask here if someone knew... |
| 17:48.58 | twisted[work] | whoot. |
| 17:49.09 | twisted[work] | if anyone sees manxpower tell him i said thanks, but i fixed the issue. ;) |
| 17:49.11 | *** join/#asterisk lyroy (~lyroy@modemcable007.224-203-24.mc.videotron.ca) |
| 17:49.29 | lyroy | What is the codec I need to use with a Sipura 2000 please |
| 17:49.41 | harryvv | twisted what was it? |
| 17:50.30 | cbachman | kingcobra, for external, probably, for internal/pci digium's card would work too, and be the same price (but not get you two fxs ports) |
| 17:51.29 | *** join/#asterisk Mw3 (mw3@daisy.chains.ch) |
| 17:52.17 | kingcobra | cbachman: thanks; i like the idea of having them be standalone; the digium X100Ps have recently started crashing my linux box; i don't have time at the moment to figure out why; |
| 17:54.27 | MuppetMaster | lyroy: Depends on how you have your Sipura 2K configured. |
| 17:54.32 | *** join/#asterisk Lee__ (~lee@ool-44c26142.dyn.optonline.net) |
| 17:54.38 | MuppetMaster | It supports several codecs, including G711 and G729. |
| 17:55.13 | Lee__ | where does one buy a g729 licence? |
| 17:55.21 | MuppetMaster | Lee__: From Digium on their website. |
| 17:55.35 | Lee__ | or where does one buy phones with speex support |
| 17:55.50 | MuppetMaster | Lee__: Depends on what you need. |
| 17:56.12 | Lee__ | SIP or IAX handsets for a small business PBX |
| 17:56.20 | *** join/#asterisk tekati (~captain@cpe-66-75-215-63.bak.res.rr.com) |
| 17:56.29 | Lee__ | the speex 8kbps quality is too good to ignore |
| 17:56.49 | MuppetMaster | Lee__: You will find more that support SIP, and if you want to use them with something besides Asterisk you will be able to. IAX2 is great, but locks you into Asterisk. |
| 17:57.01 | MuppetMaster | Lee__: True, speex is great and opensource. |
| 17:57.15 | Lee__ | I haven't found any phones that can enc/dec speex |
| 17:57.31 | MuppetMaster | Lee__: No, but plenty that do G729 or G711 |
| 17:59.43 | lyroy | I always have that message when I try to answer an incoming calls: rtp.c:540 ast_rtp_read: Unknown RTP codec 65 received |
| 17:59.54 | lyroy | does someone know what i'm doing wrong |
| 18:00.03 | *** join/#asterisk genoobie (guest@pool-141-149-140-39.buff.east.verizon.net) |
| 18:00.09 | *** join/#asterisk tessier (~treed@210.245.97.9) |
| 18:02.22 | Kel | can anyone in here higher a hitman for me? I'm not exactly asterisk savvy and I'm trying to work with a client who doesn't know what they want and I just want to die now :-\ stupid clients. |
| 18:02.39 | *** join/#asterisk leandro_pt (~leandro@bl6-124-28.dsl.telepac.pt) |
| 18:02.45 | Hmmhesays | stfu and gbtw usually works out pretty well |
| 18:03.19 | MuppetMaster | Business would be so much easier without customers... |
| 18:03.26 | Kel | lol, indeed. |
| 18:04.12 | Kel | I'm a sysadmin, not a voip monkey :-\ So this is just a tinge more red for me, but whatever. It looks/sounds like a cool technology so I look forward to learning it... it would just be nice if I got to do it under less... stupid circumstances. |
| 18:04.27 | Kel | or rather, I'm a sysadmin who's never messed with voip before |
| 18:04.31 | MuppetMaster | Kel: Are you looking to find some specific information? |
| 18:04.52 | MuppetMaster | Kel: It's easy. Much easier to come from *nix to Asterisk than from a telecom background to Asterisk... IMHO |
| 18:05.02 | *** part/#asterisk Juxt (~Juxt@adsl-068-213-216-087.sip.bct.bellsouth.net) |
| 18:06.59 | genoobie | next generation of communication...shouting louder... |
| 18:07.06 | genoobie | HEY |
| 18:07.11 | genoobie | YOU HEAR ME...:) |
| 18:07.55 | Kel | Not yet |
| 18:08.18 | Kel | Right now we're just flattening out what he wants. Sadly I know nothing of the hardware so I'm googling the hell out of every fifth word and acronym he throws at me |
| 18:08.52 | *** join/#asterisk jhiver (~jhiver@AStDenis-103-1-9-19.w81-248.abo.wanadoo.fr) |
| 18:08.55 | goatmilk | genoobie: I SHALL SMITE THEE |
| 18:09.03 | MuppetMaster | Kel: Great place to get your feet wet is http://www.voip-info.org. |
| 18:09.13 | MuppetMaster | That is the VoIP and Asterisk Wiki. |
| 18:09.37 | MuppetMaster | Kel: Also, to get a quick system up and running on an old platform you could give http://asteriskathome.sourceforge.net a shot. |
| 18:10.40 | *** join/#asterisk AgiNamu (~AgiNamu@216.230.151.230) |
| 18:12.04 | genoobie | peace y'all |
| 18:12.10 | *** part/#asterisk genoobie (guest@pool-141-149-140-39.buff.east.verizon.net) |
| 18:12.56 | jhiver | good evening everybody |
| 18:13.16 | kimc | hiya all.. got a little echo problem here.. |
| 18:13.40 | jhiver | ...echo problem here ...problem here ...here... re |
| 18:13.47 | jhiver | okay that was bad, sorry :) |
| 18:13.55 | kimc | SIP phone -> fxo port, getting echo on the SIP side |
| 18:14.14 | kimc | any thoughts ? |
| 18:14.16 | jhiver | so it's a SIP ATA? |
| 18:14.30 | jhiver | or is it a SIP -> Asterisk -> FXO setup? |
| 18:15.01 | kimc | yes jhiver, -> FX0 -> pots phone |
| 18:15.21 | jhiver | Well I have the same setup |
| 18:15.29 | jhiver | with a cheapo grandstream I have echo |
| 18:15.42 | kimc | The SIP phone is Sipura 841 |
| 18:15.48 | jhiver | with a sipura I don't |
| 18:15.56 | *** join/#asterisk bimmerd00d (~Podunk121@68.92.185.130) |
| 18:16.09 | kimc | Huh.. wonder how to test the hybrid null on the FXO port |
| 18:16.14 | jhiver | have u tried using an alternate SIP client, just to see wether it's your FXO or the SIP phone? |
| 18:16.31 | jhiver | like u could get firefly and try with it |
| 18:16.34 | kimc | jhiver: no have not, which client? |
| 18:16.40 | kimc | ok |
| 18:17.03 | kimc | SIP -> Asterisk -> SIP no problem |
| 18:17.11 | Kel | hm |
| 18:17.15 | Kel | I just pissed him off |
| 18:17.20 | Kel | Luckily it's not the client i'm working directly with |
| 18:17.21 | jhiver | ok then it's prolly your FXO... |
| 18:17.23 | Kel | but someone else within my ocmpany |
| 18:17.29 | Kel | "I don't understand how you can in the same sentence acknowledge not knowing anything about this, then argue against the person who does and is taking time out of his day to help us." |
| 18:17.36 | Kel | I have a friend helping me out a little bit |
| 18:17.43 | kimc | The FXO is in a TDM-11B card |
| 18:17.57 | jhiver | Damn, it's supposingly good stuff |
| 18:18.01 | kimc | yeh |
| 18:18.07 | jhiver | Can't you adjust impedance somewhere? |
| 18:18.16 | bimmerd00d | hi, im trying to install asterisk, and upon running make clean; make install in the zaptel directory, i get an error "make: cc: Command Not found" |
| 18:18.17 | jhiver | might be something to do with that... |
| 18:18.19 | implicit | jhiver, :) |
| 18:18.30 | jhiver | ?? |
| 18:18.40 | kimc | I haven't run across a method to adjust the impedance |
| 18:18.50 | jhiver | oh I think there is, lemme see my config |
| 18:19.03 | jhiver | might be in zaptel drivers or zapata.conf or something |
| 18:19.12 | kimc | That'd be great |
| 18:20.01 | kimc | I acts like there is very little isolation between tx/rx on the FXO port |
| 18:20.07 | bimmerd00d | anyone have any ideas why i wouldn't be able to compile zaptel? |
| 18:20.57 | jakepdev | bd - what's the error? |
| 18:21.15 | fugitivo | bimmerd00d: what distribution? you don't have the c compiler installed |
| 18:21.25 | PTG1234 | anyone use a cisco when i dial 1001 it brinsg up automatically 10011 and i can't clear the last 1.. cisco 7960 |
| 18:21.25 | bimmerd00d | sarge |
| 18:21.27 | bimmerd00d | debian |
| 18:21.38 | bimmerd00d | i'm somewhat new to linux too |
| 18:21.50 | fugitivo | bimmerd00d: type this: gcc --version |
| 18:22.02 | goatmilk | bimmerd00d: and using debian? wow, you're a real trooper. |
| 18:22.20 | jhiver | you could read those posts: |
| 18:22.22 | jhiver | http://www.voip-info.org/wiki-Asterisk+zapata+gain+adjustment |
| 18:22.27 | epoch | hrm |
| 18:22.27 | bimmerd00d | goatmilk: it's per the boss' request here at work, he told me to learn as much as i can about it |
| 18:22.35 | kimc | thanks, brb |
| 18:22.36 | Moc____ | anyone know the list if library needed to get spandsp to work with a redhat enterprise 4 ? |
| 18:22.49 | bimmerd00d | fugitivo: it says command not found when i put in gcc --version |
| 18:22.49 | fugitivo | bimmerd00d: did you type that? |
| 18:22.52 | zaptel | PTG1234, sounds like you have 1001 stored in your personal directory |
| 18:22.55 | fugitivo | bimmerd00d: install gcc |
| 18:22.59 | goatmilk | bimmerd00d: boss is a smart man, listen to his advice |
| 18:23.01 | epoch | I remember reading something about Polycom's new 1.4.1 SIP image not supporting reboot via SIP NOTIFY anymore... anyone know anything about that? |
| 18:23.25 | bimmerd00d | goatmilk: i know debian is not the easiest distro to learn, but who am i to say no |
| 18:23.29 | fugitivo | bimmerd00d: you should learn about apt-get, you need to read a lot |
| 18:23.46 | fugitivo | bimmerd00d: or just go with fedora |
| 18:23.46 | jhiver | ok there's some more talk about impedance on this page: http://www.voip-info.org/wiki-digium |
| 18:23.49 | PTG1234 | no its just in placed calls |
| 18:23.52 | PTG1234 | how do i clear placed calls? |
| 18:23.54 | PTG1234 | i see a keep |
| 18:23.57 | PTG1234 | i don't see a delete :) |
| 18:23.58 | *** join/#asterisk km- (pgrace@brdgw1.rttx.com) |
| 18:24.19 | kimc | jhiver: thanks muchly |
| 18:24.34 | bimmerd00d | fugitivo: i know how to use apt-get |
| 18:24.37 | zaptel | PTG1234: just reseting the phone |
| 18:24.41 | jhiver | although it doesn't say how to set it... |
| 18:24.55 | fugitivo | bimmerd00d: great, install gcc then |
| 18:25.04 | *** join/#asterisk ariel_ (~Ariel@adsl-070-147-214-250.sip.mia.bellsouth.net) |
| 18:26.33 | bimmerd00d | fugitivo: that worked, sweet, thanks |
| 18:27.47 | *** join/#asterisk AchillesHeel (~root@wblv-146-240-09.telkomadsl.co.za) |
| 18:28.00 | AchillesHeel | so does this thing actually work/ |
| 18:28.02 | AchillesHeel | ? |
| 18:28.09 | goatmilk | what thing? |
| 18:28.16 | fugitivo | AchillesHeel: no, nothing to see here |
| 18:28.28 | Katty | mmm, full *pample* |
| 18:28.32 | AchillesHeel | asterisk, mirc, the planet, our minds, world of warcraft, capitalism |
| 18:28.50 | Katty | i &heart; stirfry |
| 18:29.04 | fugitivo | AchillesHeel: you shouldn't use root user |
| 18:29.11 | AchillesHeel | &heart; Hmm which IRC client shows that? defn. not ircII |
| 18:29.13 | goatmilk | Katty: you know that stirfry is a glibc function? |
| 18:29.29 | AchillesHeel | fugi what can u do |
| 18:29.29 | Katty | goatmilk: it's a vegan function too |
| 18:29.41 | fugitivo | AchillesHeel: me nothing, just an advice |
| 18:29.56 | *** join/#asterisk PMantis (~Miranda@66.251.89.34) |
| 18:30.07 | Katty | back to studying :< |
| 18:30.16 | tzanger | studying what |
| 18:30.21 | Katty | ... |
| 18:30.29 | Katty | tzanger: i've already told you twice, surely. |
| 18:30.41 | tzanger | you did? |
| 18:30.46 | Katty | as+ |
| 18:30.50 | Katty | a+ i mean |
| 18:30.50 | tzanger | as+? |
| 18:30.55 | tzanger | a+? |
| 18:30.56 | Hmmhesays | haha |
| 18:30.58 | epoch | ass plus? |
| 18:30.59 | PMantis | What's new from 1.06 to 1.07? |
| 18:31.02 | tzanger | isn't that a mark? |
| 18:31.03 | Katty | epoch: exactly. |
| 18:31.11 | Hmmhesays | does studying include a pillow fight? |
| 18:31.11 | Lee__ | AchillesHeel: it's working farily well here. Quite complicated to configure but not impossible. Been at it for 8 days now and have local/outgoing/incoming calls working. |
| 18:31.15 | Katty | tzanger: it's a comptia certification that says i'm not shit |
| 18:31.31 | Katty | as if people don't realize i'm not shit to start with, heh. |
| 18:31.46 | tzanger | Katty: ah |
| 18:31.48 | Lee__ | A+ means you know how to install RAM without killing yourself, right? |
| 18:31.55 | Katty | Lee__: uh, something like that. |
| 18:31.55 | implicit | pretty much |
| 18:32.02 | kimc | Does an FXO port want echocancel ? |
| 18:32.11 | Hmmhesays | at least they took the dos memory management stuff out of a+ |
| 18:32.15 | implicit | i don't know why anyone would waste their time getting a+ |
| 18:32.20 | Lee__ | I think anyone who got old skool nintendo cartridges to work gets an A+ certification |
| 18:32.26 | Katty | implicit: my company is paying for it. |
| 18:32.31 | kimc | That is a port with an FXO card on it.. |
| 18:32.31 | Katty | implicit: it'll look nice on a resume. |
| 18:32.45 | fugitivo | what is a+ ? |
| 18:32.46 | implicit | Katty, you would have to pay me $5000 bucks to put it on my resume |
| 18:32.47 | Lee__ | Katty: get a CCNA |
| 18:32.51 | AchillesHeel | listen, anyone know ircII? |
| 18:32.56 | Katty | Lee__: dont' tell me what to do |
| 18:33.00 | Katty | Lee__: that anonys me |
| 18:33.08 | Katty | Lee__: better. |
| 18:33.10 | QuickDry | Doesn't A+ get into large scale SCSI implementation, Quorum drives, and other specialized hardware? or is it just generic stuff? |
| 18:33.15 | jhiver | yeah what the hell is a a+ neway? |
| 18:33.16 | goatmilk | how many times have i heard someone do something because it's good for a resume :) |
| 18:33.17 | Beirdo | Katty: I'll tell ya what to do... :) Keep standing up for yourself :) |
| 18:33.20 | implicit | A+ is bullshit as hell |
| 18:33.21 | Hmmhesays | Katty: give my arm a hug |
| 18:33.30 | tzanger | your arm? |
| 18:33.30 | implicit | Hmmhesays, don't go insane |
| 18:33.39 | tzanger | fugitivo: it's a very good mark |
| 18:33.40 | PMantis | jhiver: A certificaten test |
| 18:33.42 | Hmmhesays | yeah the one that feels funny |
| 18:33.45 | implicit | tzanger, hahahaha |
| 18:34.06 | jhiver | blood type! |
| 18:34.22 | jhiver | but why does being A+ look good on a resume? :) <grin> |
| 18:34.43 | Hmmhesays | because the people hiring are often not very smart |
| 18:35.01 | TomL | HR people are fickle |
| 18:35.03 | fugitivo | i don't put certifications on my resume |
| 18:35.18 | TomL | they're just as likely to hire you for liking the same TV show they do as any cert on your resume |
| 18:35.18 | PMantis | jhiver: LOL ok. I'm O- |
| 18:35.30 | Hmmhesays | my resume consists of a picture of me drunk playing guitar around my fire pit |
| 18:35.36 | jhiver | Oooh PMantis DO NOT put that on your CV man! :) |
| 18:35.52 | PMantis | :) Low grade, eh? |
| 18:36.04 | TomL | o neg.. good to have around. Can donate to anyone :) |
| 18:36.10 | jhiver | it looks like MINUS ZERO |
| 18:36.21 | jhiver | like you could not have PLUS ZERO :))) |
| 18:36.41 | TomL | zero doesn't have a sign :P |
| 18:36.44 | jhiver | biiiiig difference :) |
| 18:36.50 | PMantis | lol |
| 18:36.56 | jhiver | yes it does, it can be either +0 or -0 :) |
| 18:37.14 | jhiver | in all cases equals "la tete a toto" |
| 18:37.20 | TomL | but if he's o neg, we can suck him dry at the first need... everyone who needs blood can take O neg transfusions |
| 18:37.30 | PMantis | Is there a spot on the site that outlines what changed between revisions? 1.0.6 != 1.0.7 |
| 18:37.32 | TomL | regardless of their own type |
| 18:37.38 | bjohnson | just make up certifications |
| 18:37.39 | jhiver | whell not unless u resus positive... |
| 18:37.45 | bjohnson | not like they know what they mean anyway |
| 18:37.52 | jhiver | yeah that sounds good, making up certifs |
| 18:38.00 | km- | Hey, anyone ever use a Nortel DMS-800 |
| 18:38.03 | jhiver | I have a SGBC degree |
| 18:38.04 | PMantis | Ok, I'm PMantis certified! :-) |
| 18:38.07 | bjohnson | just make sure you print one up to back up that you have it |
| 18:38.15 | jhiver | Self-Given-Bullshit-Certificate |
| 18:38.16 | TomL | I've seen a DMS-500, does that count? |
| 18:38.33 | TomL | do you mean, certifiable? |
| 18:38.37 | bjohnson | PMaNTIs certified A+ |
| 18:38.40 | km- | outtolunc: at the very least certifiable! |
| 18:38.45 | jhiver | and a PH.d in fly fucking |
| 18:38.51 | jhiver | fine art that |
| 18:38.54 | TomL | a winner is you |
| 18:39.11 | km- | Toml: heh |
| 18:39.20 | km- | unfortunately, I've got problems with the Nortel DMS-800 that XO has in our demarc |
| 18:39.22 | Katty | Hmmhesays: are you projecting your needs onto your arm? |
| 18:39.26 | TomL | km-: wow you lose :p |
| 18:39.35 | km- | and I was wondering if someone here had experience with them in the past |
| 18:39.35 | implicit | km-, :) |
| 18:39.50 | kimc | jhiver: should I have echocancel=on for the FXS card? |
| 18:39.52 | ChkDigit | Has anyone used Mediatrix equipment in an * setup? |
| 18:40.04 | implicit | if someone came to get a job from me and had A+ on their resume |
| 18:40.06 | jhiver | eeeh you might want to try that if you're having echo problems |
| 18:40.09 | implicit | i would think twice about wanting to hire them |
| 18:40.11 | *** join/#asterisk DEEZED (~Scrilla@adsl-065-006-189-182.sip.bct.bellsouth.net) |
| 18:40.12 | kimc | ok |
| 18:40.12 | jhiver | I though it was an FXO??? |
| 18:40.16 | implicit | it would DEFINITELY be a negative point |
| 18:40.22 | Katty | implicit: whine whine whine |
| 18:40.27 | Katty | implicit: stop being so bitter. |
| 18:40.29 | *** join/#asterisk BoRiS (boris@S01060040ca1e5b54.wp.shawcable.net) |
| 18:40.36 | implicit | Katty: ? |
| 18:40.37 | jhiver | naaah not negative... |
| 18:40.45 | implicit | just being honest |
| 18:40.49 | kimc | Right, its FXS card -> pots phone right? |
| 18:40.51 | implicit | it's not like i give a shit about you personally |
| 18:40.52 | implicit | lol |
| 18:40.53 | jhiver | it's ok to display your degrees as long as you don't make a show of them |
| 18:40.57 | Katty | implicit: good. |
| 18:41.02 | AchillesHeel | haha, laugh at me, what server am i on? |
| 18:41.19 | Katty | implicit: for a second there, i thought you were being a typical bitter geek and putting others down for knowing as much as you |
| 18:41.32 | implicit | hahaha |
| 18:41.32 | Katty | implicit: i'm sure glad i was wrong |
| 18:41.34 | implicit | whats up cypromis |
| 18:41.43 | implicit | u still in toronto? |
| 18:41.49 | cypromis | yeah another 5h |
| 18:41.58 | jhiver | Katty: what are you doing on a bloody IRC channel for Asterisk if you're *not* a typical bitter geek? |
| 18:42.00 | Hmmhesays | my gear will be here tomorrow |
| 18:42.01 | implicit | btw, u want to meet in europe in a couple months? |
| 18:42.01 | jhiver | god damn |
| 18:42.10 | cypromis | sure |
| 18:42.10 | jhiver | it's like someone from the AA being in a pub! |
| 18:42.10 | Katty | jhiver: trying to learn, dear, trying to learn. |
| 18:42.22 | Shido6 | back |
| 18:42.24 | implicit | i am going to be visiting a few clients |
| 18:42.25 | cypromis | I'llbe probbly in chicago beginning of august |
| 18:42.30 | cypromis | when ? |
| 18:42.37 | implicit | end of july probably |
| 18:42.41 | jhiver | katty: trying to learn to be a bitter geek? |
| 18:42.42 | implicit | actually |
| 18:42.44 | cypromis | :) |
| 18:42.44 | jhiver | we can help I'm sure |
| 18:42.47 | implicit | i'll be there mid june as well |
| 18:42.51 | cypromis | where about ? |
| 18:42.51 | implicit | i have to make two trips down |
| 18:43.07 | Hmmhesays | heh msn is puking on itself again |
| 18:43.11 | alt_phil | I live in chicago. It's not all that great. |
| 18:43.12 | implicit | london, switzerland and germany are definite so far |
| 18:43.24 | QuickDry | ~docs |
| 18:43.25 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org |
| 18:43.30 | cypromis | so it will be quite close to me :) |
| 18:43.48 | implicit | but i'll be in China for longer if you are up for meeting there |
| 18:43.58 | *** join/#asterisk piesang (~c10u@wblv-146-240-09.telkomadsl.co.za) |
| 18:44.28 | AchillesHeel | ? |
| 18:44.46 | cypromis | dunno yet |
| 18:44.47 | Katty | jhiver: you're wasting my time. |
| 18:44.51 | cypromis | will probably be in china in june |
| 18:44.52 | cypromis | :) |
| 18:44.54 | Katty | jhiver: stop bothering me |
| 18:45.11 | jhiver | katty: it's easy |
| 18:45.29 | jhiver | katty: just click on the little cross on the top right of your IRC window |
| 18:45.47 | Katty | jhiver: umm, i'm screening irssi, there is no x dear. |
| 18:45.59 | tzanger | what are you trying to do Katty? I use screen and irssi |
| 18:46.09 | Katty | tzanger: nothing mister fix it ;) |
| 18:46.09 | jhiver | oooh screen hey... |
| 18:46.23 | ard | wow.... I just switch here, and I see people having arguments? |
| 18:46.23 | tzanger | hahaha |
| 18:46.24 | jhiver | then ctrl + a + d ought to do the trick |
| 18:46.27 | tzanger | when you're this big, they call you mister |
| 18:46.51 | jhiver | bah it seems that katty has about zero sense of humor |
| 18:47.08 | *** part/#asterisk Hmmhesays (negative3k@66.173.103.108) |
| 18:47.08 | jhiver | sorry about being me |
| 18:47.10 | tzanger | katty just likes the sweet sultry sound of my voice |
| 18:47.12 | *** join/#asterisk blitzrage (~blitzrage@blitzrage.documenter.extraordinaire.of.asterisk) |
| 18:47.18 | ard | :-) |
| 18:47.37 | Katty | tzanger: except you don't talk much |
| 18:47.51 | tzanger | well it's hard to talk when about 750ms later you hear your own voice |
| 18:48.05 | jhiver | heh that's quite a bit of a trip |
| 18:48.33 | ard | Heh, 75ms is already too much |
| 18:48.45 | Katty | tzanger: would you rather i not talk at all? |
| 18:48.57 | jhiver | naaah... I do phone calls with 300~400ms round trip and it's kindof ok |
| 18:49.07 | tzanger | it's the talkback that's hard |
| 18:49.14 | tzanger | Katty: I didn't say that |
| 18:49.14 | ard | jhiver : well I meant echo |
| 18:49.23 | jhiver | aah ok |
| 18:49.25 | tzanger | perhaps next time with the headset |
| 18:49.26 | *** join/#asterisk PTG1234 (~PTG123@ip68-106-24-139.ph.ph.cox.net) |
| 18:49.29 | tzanger | so I can't hear myself |
| 18:49.42 | tzanger | * ard hasn't heard echo for a long time |
| 18:49.43 | jhiver | yeah, echo really is annoying |
| 18:49.45 | *** join/#asterisk MatsK (~NNSCRIPT@107.80-202-57.nextgentel.com) |
| 18:49.46 | tzanger | * ard keeps fingers crossed |
| 18:49.54 | jhiver | even worse is trying to sing something when echo is on |
| 18:49.55 | tzanger | ... crossed |
| 18:49.56 | tzanger | crossed |
| 18:49.57 | tzanger | ssed |
| 18:49.58 | tzanger | d |
| 18:49.59 | jhiver | it's really hard :) |
| 18:50.09 | ard | :-) |
| 18:50.12 | ard | Well |
| 18:50.18 | ard | It's more slow then :-) |
| 18:50.26 | tzanger | this is hilarious |
| 18:50.26 | *** part/#asterisk AchillesHeel (~root@wblv-146-240-09.telkomadsl.co.za) |
| 18:50.35 | tzanger | this box only has 512M RAM and I'm already 705M into swap |
| 18:50.58 | blitzrage | more RAM! |
| 18:50.58 | Katty | tzanger: you got feed back because of my speakers, not * |
| 18:51.06 | PMantis | MAN! What are you running?? |
| 18:51.07 | tzanger | Katty: I know |
| 18:51.10 | Katty | tzanger: k |
| 18:51.11 | piesang | gui |
| 18:51.13 | piesang | no doubt |
| 18:51.29 | ard | something with glib? |
| 18:51.30 | piesang | check whats chowing your ram with top |
| 18:51.48 | *** join/#asterisk iceyp (~icepick@202.150.105.150) |
| 18:52.00 | tzanger | who's being glib |
| 18:52.12 | ard | using is the better word |
| 18:52.18 | ard | what is using glib :-) |
| 18:52.22 | tzanger | :-) |
| 18:52.27 | tzanger | ldd will tell you |
| 18:53.10 | jhiver | hey any french-speaking body on the chan who woudln't mind being a guinea pig^h^h^h^h^h alpha tester for me please? |
| 18:53.13 | ard | Not that I want to put down glib. But in some cases I seems to introduce a lot of memory fragmentation |
| 18:53.41 | AgiNamu | hey, someone here the other day was asking about crackign the g729 code |
| 18:53.44 | AgiNamu | are you heren ow? |
| 18:53.47 | ard | I can put on a french channel, and put the phone in front of it :-) |
| 18:53.47 | alt_phil | Sorry jhiver. If you need someone that knows pig-latin, I'll help you out though :) |
| 18:53.56 | AgiNamu | cracking the activation system that is. |
| 18:53.59 | ard | something like multivision 1 :-) |
| 18:54.04 | jhiver | alt_phil that might be enough :) |
| 18:54.05 | *** join/#asterisk viLeR (1000@ip-47-252.telesat.com.co) |
| 18:54.36 | *** join/#asterisk SPoon_TSX (~SPoon_TSX@toronto-HSE-ppp4117414.sympatico.ca) |
| 18:54.48 | *** join/#asterisk mesi (~player@dsl-082-083-048-092.arcor-ip.net) |
| 18:55.44 | SPoon_TSX | Hello everyone, Wondering if anyone knows any possible reason why whenever I pick a incoming call from Asterisk on my phone, the asterisk will just hung my channell up in about 3 seconds? |
| 18:56.35 | *** join/#asterisk asteriskmall (~chris@12-215-210-142.client.mchsi.com) |
| 18:56.47 | AgiNamu | uh well, you might have to provide a few more details.... |
| 18:56.56 | scoof | SPoon_TSX: Have you tried turning up verbosity in asterisk and see what it says? |
| 18:57.13 | asteriskmall | Is there a way to specifiy for an agi script to be executed after the call hangs up? |
| 18:57.32 | SPoon_TSX | I am runing asterisk -vvvcg. Am I v enough? |
| 18:58.25 | SPoon_TSX | asteriskmall: You may want to take a look of EAGI. |
| 18:58.25 | scoof | SPoon_TSX: Using SIP? |
| 18:58.25 | SPoon_TSX | scoof: Yes. |
| 18:58.25 | scoof | SPoon_TSX: sip debug iå <ip> |
| 18:58.25 | SPoon_TSX | scoof: I have my outgoing call okay but not incoming. |
| 18:58.25 | scoof | SPoon_TSX: sip debug ip <ip> |
| 18:58.25 | jhiver | asteriskmall: |
| 18:58.34 | jhiver | you can use DeadAGI and do your Dial inside the DeadAGI |
| 18:58.50 | jhiver | then after your dial do whatever you like as long as it's in the same script |
| 18:59.02 | jhiver | I have written my custom calling card app this way |
| 18:59.12 | *** join/#asterisk PoWeRKiLL (~PoWeRKiLL@l01m-20-133.d4.club-internet.fr) |
| 18:59.47 | asteriskmall | jhiver: sounds like what I need. I'm also interested in your calling card app. |
| 19:00.07 | asteriskmall | can we discuss this more off channel? |
| 19:00.10 | jhiver | I can send you the stuff but it's got a lot of crap that is specific to my system |
| 19:00.14 | jhiver | yeah you can pm me |
| 19:00.36 | AgiNamu | dfamn, there should be an iax2 debug ip command : |
| 19:01.09 | AgiNamu | Hey, anyone think there's any business in providing hosted asterisk programmability? |
| 19:01.29 | AgiNamu | like, say some programmer wants to voice-enable his app. We provide an API for him to run on his server and we send DIDs to him |
| 19:01.48 | AgiNamu | and he just processes them using our cute api |
| 19:01.57 | *** part/#asterisk MuppetMaster (~MuppetMas@a82-92-73-185.adsl.xs4all.nl) |
| 19:02.01 | AgiNamu | no need to learn asterisk or linux or anything outside his programming environment |
| 19:02.20 | caesar2 | what do you think: it it bad to do all the dialing or other stuff in agi : _.,1,agi,dialinoutorsomewhere.agi |
| 19:04.21 | AgiNamu | caesar2 not at all |
| 19:04.22 | sudhir492 | AgiNamu: I am offering hosted pbx to businesses |
| 19:04.23 | *** join/#asterisk jsolares (~jsolares@200.6.219.36) |
| 19:04.26 | AgiNamu | not hosted pbx |
| 19:04.30 | AgiNamu | hosted programmabilioty |
| 19:04.34 | sudhir492 | Have got decent response so far |
| 19:04.40 | AgiNamu | given some Visual Basic developer |
| 19:04.43 | AgiNamu | he wrote some inventory app |
| 19:04.47 | *** join/#asterisk Maxxed (~user@65.67.149.242) |
| 19:04.50 | Maxxed | yo :) |
| 19:04.51 | AgiNamu | and now he wnats to do "call in and check stock" |
| 19:05.14 | AgiNamu | caesar2, in fact, if your dialplan logic is complex, using an AGI or C program might be an EXCELLENT solution. |
| 19:05.25 | AgiNamu | considering that doing anything even minor in extensions.conf is a pita. |
| 19:05.35 | jsolares | hey ppl, anyone know of a good TTS for linux? i'm using festival with asterisk to make a full database driven IVR with tts, but well festival has bad voice quality, specially in spanish |
| 19:05.46 | mesi | Is there any fwdOUT user online? Strangely I have quite short calls in my list. |
| 19:05.54 | caesar2 | i think so to.. but what is extension.conf for anyway... if i only need on line... |
| 19:06.08 | AgiNamu | jsolares hola :). Cepestral seems to be the other "favoured" player. |
| 19:06.17 | *** join/#asterisk _tekati_ (~captain@cpe-66-75-215-63.bak.res.rr.com) |
| 19:06.20 | AgiNamu | I use Windows, so I've got a wide range of SAPI voices to use |
| 19:06.20 | sudhir492 | AgiNamu: I have no idea about hosted programmability. |
| 19:06.36 | jsolares | hey AgiNamu |
| 19:06.55 | eKo1 | AGI is much easier than making your own app. |
| 19:07.03 | AgiNamu | AGI is also a lot less powerful. |
| 19:07.09 | eKo1 | apps are a bitch to debug though. |
| 19:07.14 | mesi | Hey, fwdOUT actually HAS forums :-) |
| 19:07.17 | bjohnson | so .. I can dial my home * machine via pstn through a SPA 3k fxo and use dtmf to control the voicemail app .. but I often have problems using dtmf to control other companies IVR systems. Any tips? |
| 19:08.16 | Shido6 | inband? |
| 19:08.19 | bjohnson | tzanger mentioned that it might be due to the length of the dtmf tones. Anyone know how to control that? |
| 19:08.30 | Maxxed | im having a bit of a problem upgrading my firmware on my 7940, im runing P0S30200.bin and im trying to upgrade to P0S3-05-3-00 and cant geter to go? |
| 19:08.35 | bjohnson | not inband .. avt on the spa and rfc2833 in * |
| 19:08.35 | jsolares | AgiNamu, what database does the VB inventory app use? |
| 19:08.38 | Maxxed | this there something im missing? |
| 19:08.39 | *** join/#asterisk cpatry (~junky@modemcable174.107-81-70.mc.videotron.ca) |
| 19:08.57 | AgiNamu | jsolares, that's just an example. |
| 19:09.10 | AgiNamu | Myabe it uses MySQL and thus loses its data every time they make a mistake |
| 19:09.24 | Maxxed | i see a W220 TFTP Error: Buffer Full under my stat mesgs :\ |
| 19:09.25 | AgiNamu | or maybe it uses Oracle. or maybe it queries a SOAP service. |
| 19:09.31 | harryvv | bjohnson ive seen that as a issue on my system where using dtmf to control the apps was a issue. I dont know what it was but made some changes to correct it. |
| 19:09.33 | jsolares | well if you can conect to it with perl you can make a tts driven ivr to check inventory with AGI |
| 19:09.41 | AgiNamu | i.e., I'd like to offer a general API for programmer to use |
| 19:09.45 | tzanger | bjohnson: in the zaptel source IIRC |
| 19:09.59 | AgiNamu | i was thinking more like VB.NET, C#, C, C++, Perl, PHP, COBOL, Pascal, Brainfuck, etc. etc. |
| 19:10.03 | jsolares | ahh, tough noodle |
| 19:10.16 | *** join/#asterisk jabular (~jabular@82-32-105-84.cable.ubr02.hawk.blueyonder.co.uk) |
| 19:10.39 | scoof | Maxxed: try to upgrade through the different intermediate versions up to the release you want to end at |
| 19:10.44 | jakepdev | AGI - how can you get the CLR to compile for use in Linux? |
| 19:10.49 | bjohnson | tzanger: these fxo are Sipura SPA 3000 |
| 19:10.52 | bjohnson | SIP |
| 19:10.56 | jsolares | hmm well cepstral is not that good |
| 19:11.17 | tzanger | bjohnson: oh |
| 19:11.25 | tzanger | I doubt htat's the problem then |
| 19:11.29 | Maxxed | scoof: i am :\ i cant even load P0S30203.bin,the next one up |
| 19:11.33 | AgiNamu | jakepdev... Mono |
| 19:11.35 | jaiger | AgiNamu, you're looking to write yet another "generic" voice API? |
| 19:11.47 | AgiNamu | no, im looking to wrap AGI pretty much |
| 19:12.00 | jsolares | there's not much in AGI to wrap around :X |
| 19:12.02 | AgiNamu | and provie FastAGI on a hosted basis. |
| 19:12.05 | Maxxed | sombody mentioned a suport matrix, i have to start from the botem and work my way up, and im trying, but i cant get it past the P0S30200.bin |
| 19:12.17 | AgiNamu | sure there is. AGI is just a set of instructions to send over a pipe |
| 19:12.50 | jsolares | meh i liked scansoft realspeak, but the bastards wont give me info on the phone, it's all on the site they say... bah |
| 19:12.50 | scoof | Maxxed: are you at a universal loader version? |
| 19:12.50 | AgiNamu | writeline("say digits bla ") is not comparable to Channel.SayDigits(xxx) |
| 19:12.50 | Maxxed | scoof: universal loader? |
| 19:12.54 | *** part/#asterisk Alexi1 (~alexis@www.trim.it) |
| 19:13.08 | Maxxed | scoof: Boot Load ID PC030301 |
| 19:13.24 | scoof | Maxxed: I'm not sure about the SIP-images, I'm mostly playing around with SCCP |
| 19:13.31 | jsolares | i didnt mean there wasnt anything to wrap around, just that there arent that many different instructions in agi |
| 19:13.40 | AgiNamu | oh , yea, unfortunately. |
| 19:13.44 | bjohnson | any idea what the Sipura settings DTMF Process INFO and DTMF Process AVT would mean? |
| 19:13.46 | scoof | Maxxed: are you already running SIP software on it? |
| 19:13.48 | AgiNamu | AGI could be useful with a few more features. |
| 19:13.54 | AgiNamu | like, control of the CDR |
| 19:13.57 | AgiNamu | setting dst and src and so on |
| 19:14.11 | Maxxed | scoof: yes, im running sip P0S30200.bin |
| 19:14.30 | Maxxed | scoof: trying to get up to P0S30203.bin |
| 19:14.43 | sudhir492 | JerJer[mobile]: are you still there |
| 19:15.37 | harryvv | jsolares, I just tested scansoft very good female voice. |
| 19:15.56 | Maxxed | scoof: when i wint from mccp to sip, it worked like a champ, but i cant get anything passed what i have now :\ |
| 19:16.11 | AgiNamu | anyone here have a PA168 phone to test an IAX2 fix? |
| 19:16.21 | jsolares | harryvv, that they do, but they wont give me price for realspeak telecom since they say it's on the website, but i havent found it on the website |
| 19:16.26 | jsolares | or atleast realspeak solo |
| 19:16.33 | harryvv | I see |
| 19:16.40 | harryvv | Just testd tom..vry clear |
| 19:16.49 | jsolares | PA168? is that the same as the Atcom 320EE? |
| 19:17.05 | AgiNamu | Atcom uses pa168 chips in some of their prodcuts. maybe all |
| 19:17.25 | scoof | Maxxed: have you tried resetting it and then going from sccp to the newer SIP image? |
| 19:17.28 | AgiNamu | if it supports H323, SIP, MGCP, its prolly a Pa168 |
| 19:17.30 | jsolares | i have the 320EE at the other office, it says it supports iax2 but never connects, only wrks with sip |
| 19:17.37 | scoof | Maxxed: their software doesn't seem too robust to me |
| 19:17.41 | Maxxed | scoof: i dont have a sccp image :( |
| 19:17.44 | AgiNamu | I've got new (and customized) firmware |
| 19:17.54 | Maxxed | scoof: woudlnt wana help a fella out by chance would ya ;) |
| 19:17.55 | AgiNamu | sounds like you're running old software |
| 19:17.57 | harryvv | jsolares, did you fill out there sales contact form? |
| 19:18.07 | jsolares | harryvv, yeah, 2 weeks ago |
| 19:18.09 | AgiNamu | give me your email, and I'll send you new IAX2 firwmare. |
| 19:18.16 | jsolares | AgiNamu, jsolares@gmail.com |
| 19:18.17 | harryvv | Did you follow up? |
| 19:18.17 | scoof | Maxxed: no, sorry |
| 19:18.33 | AgiNamu | and the hardware reference design? do you know it? |
| 19:18.44 | scoof | Maxxed: if you do a factory default reset though, you should end with a non-SIP image |
| 19:18.47 | jsolares | they never contacted me, so i called, and said boohoo it's on the webiste go look it up -_-; |
| 19:18.53 | scoof | Maxxed: http://cisco.com/en/US/tech/tk652/tk701/technologies_tech_note09186a0080094584.shtml#issue3 |
| 19:18.56 | Maxxed | scoof: you sure? |
| 19:19.08 | AgiNamu | yuxin, pa168q, 5111phone, tiger, ywh100, pa168v, etc..... |
| 19:19.28 | jsolares | no idea |
| 19:19.42 | Maxxed | scoof: EXCELLENT! ah, man iv been looking high and low for that doc |
| 19:19.45 | AgiNamu | http://www.voip-info.org/wiki-ATCOM |
| 19:19.49 | AgiNamu | says PA168S |
| 19:19.50 | jsolares | i knew i should've put that phone in my backpack to test around since i wasnt going to be there |
| 19:20.10 | AgiNamu | I use it with IAX2, no problem. have a few clients on it too. works great. |
| 19:20.13 | AgiNamu | Phone or ATA |
| 19:20.13 | jsolares | lets see |
| 19:20.44 | AgiNamu | I'll send you firmware built for the PA168S |
| 19:20.59 | jsolares | ok |
| 19:21.44 | SPoon_TSX | scoof: I got the message said @ http://www.pastebin.com/264890 |
| 19:22.04 | *** join/#asterisk Thus0 (~Thus0@dyn-83-152-146-156.ppp.tiscali.fr) |
| 19:22.04 | SPoon_TSX | scoof: Is that means my phone tell the asterisk to hang up? |
| 19:22.18 | Thus0 | Hi |
| 19:22.20 | SPoon_TSX | My Asterisk IP is 192.168.1.118 |
| 19:22.24 | scoof | SPoon_TSX: looks like it, yeah |
| 19:22.25 | harryvv | jsolares, well i looked atn its not advertised. |
| 19:22.47 | harryvv | jsolares, is this mostly a windows product? |
| 19:22.48 | SPoon_TSX | scoof: Damn. |
| 19:22.49 | AgiNamu | harryvv what isnt? |
| 19:23.08 | harryvv | realspeak is not advertised in there store link |
| 19:23.32 | jsolares | it runs on linux, that much i know, not sure if it works on windows |
| 19:23.43 | Elshar | Hey, anyone know offhand how one goes about making a custom init.cfg for the pap2-na's? |
| 19:23.48 | harryvv | I dont see anything about it running on linux. |
| 19:23.58 | Elshar | I downloaded one, but it seems to be a mostly binary file. :/ |
| 19:24.22 | jsolares | RealSpeak Telecom 4.0 is available for Windows, Linux, and Solaris deployments < http://www.scansoft.com/speechworks/realspeak/telecom/ |
| 19:24.33 | harryvv | k |
| 19:24.41 | harryvv | like the voice quality. |
| 19:24.42 | harryvv | :) |
| 19:24.44 | *** join/#asterisk scorpion68 (~chatzilla@HSE-Toronto-ppp186743.sympatico.ca) |
| 19:24.58 | jsolares | so far it's the best i've found |
| 19:25.33 | harryvv | I have a mail office in blaine that gets alot of calls from canada and alot of chinese japanese ect are hard to understand by these people. |
| 19:25.38 | harryvv | This might help. |
| 19:25.56 | bjohnson | Elshar: it IS a binary file |
| 19:26.07 | *** join/#asterisk doug_ (~icechat5@HSE-Toronto-ppp186743.sympatico.ca) |
| 19:26.33 | bjohnson | Elshar: where'd you find your's? |
| 19:26.43 | Elshar | http://corp.deltathree.com/productsandservices/setup/s_pap2_instructions.html |
| 19:26.47 | harryvv | make a dabase when a package comes in thay check it in the db and whena caller wants to know there package is in thay punch in there mail box and speach senthisis in there language says there package is in. no more language barriers |
| 19:26.49 | Elshar | Came up on a google search |
| 19:27.29 | Elshar | I'm assuming that the original sipura stuff probably had some utility to generate the configs then? They touch on it in the user and admin guides, but nothing terribly helpful. |
| 19:28.22 | *** join/#asterisk loud (~ariel@null0.flapping.net) |
| 19:29.28 | Maxxed | thanks again scoof for the URL, that did the trick!! and keep on trucking with the 7970/sccp :D |
| 19:30.45 | Elshar | Hmm |
| 19:31.14 | harryvv | jsolares, I am on the phone with scansoft. |
| 19:31.33 | SPoon_TSX | scoof: Any suggestion that I may be able to work around this problem? |
| 19:31.47 | doug_ | :) |
| 19:32.50 | scoof | SPoon_TSX: no, sorry, I'm not familiar with your phone - could you perhaps do a dump of a full SIP conversation? |
| 19:33.08 | SPoon_TSX | Yes |
| 19:33.09 | harryvv | jsolares, There sales said this product is not something for retail and is fairly complex. I said it would be used as a IVR for a voipserver and passed me onto sales. I guess thay dont want any sort of technical support. |
| 19:33.26 | *** join/#asterisk fugitivo (~ajf@201.255.106.152) |
| 19:33.28 | *** join/#asterisk ariel_ (~Ariel@dsl-20-177.cofs.net) |
| 19:33.33 | harryvv | calls from the public :) |
| 19:33.49 | jsolares | seems so |
| 19:33.56 | ariel_ | good afternoon all |
| 19:34.00 | jsolares | afternoon |
| 19:34.04 | harryvv | That would be my best assumption. |
| 19:34.27 | jsolares | damn them, it's a good product with great voice quality and they dont want to sell them?? |
| 19:35.19 | *** join/#asterisk carbon60 (~adam@gw.techsupport.ca) |
| 19:35.25 | fugitivo | who? what? |
| 19:35.48 | harryvv | jsolares, this is for telecoms so thats probebly why thay dont nessesarly advertise it to the public. |
| 19:36.14 | jsolares | heh i'm a 2 man telecom company! |
| 19:36.20 | jsolares | well we |
| 19:37.00 | carbon60 | Afternoon folks. |
| 19:37.07 | jsolares | afternoon |
| 19:37.17 | AgiNamu | you are util Telgua and Telefonica start blocking SIP :) |
| 19:37.23 | carbon60 | Anyone know *anything* about the Polycom SoundPoint's "presence" and messaging features? |
| 19:37.31 | AgiNamu | and then all of a sudden, IAX seems really attractive |
| 19:38.22 | jsolares | hehe |
| 19:38.44 | SPoon_TSX | scoof: here you go http://www.pastebin.com/264896 |
| 19:38.52 | *** join/#asterisk mstocco (~mstocco@c-65-34-201-194.hsd1.fl.comcast.net) |
| 19:39.51 | bjohnson | Elshar: Sipura authorized resellers can get a program to compile/encrypt the xml config file |
| 19:40.16 | *** join/#asterisk odie_flocon (~Odie@ptr-64-201-182-211.ptr.terago.ca) |
| 19:40.21 | odie_flocon | hey all |
| 19:40.46 | PMantis | SPoon_TSX: This seems like it's an issue: "Transmitting (NAT) to 192.168.1.129:5060:" |
| 19:40.47 | bjohnson | Elshar: the end result is that you will need to config by hand each unit you have |
| 19:40.57 | PMantis | SPoon_TSX: Don't need NAT internally |
| 19:41.24 | SPoon_TSX | PMantis: But how can I turn it off? |
| 19:41.36 | PMantis | SPoon_TSX: sip.conf |
| 19:41.47 | SPoon_TSX | PMantis: I am wondering would it be the port number causing the problem. |
| 19:42.03 | *** join/#asterisk scorpion68 (~chatzilla@HSE-Toronto-ppp186743.sympatico.ca) |
| 19:42.17 | SPoon_TSX | PMantis: I tried to disable the Nat, (nat=no) and (Canreinvite=yes). Still no goal. |
| 19:42.29 | *** join/#asterisk pigpen (~mark@fw.seamans.cc) |
| 19:43.09 | Shido6 | spoon |
| 19:44.16 | Shido6 | 877-677-9649 ask for Greg |
| 19:45.56 | PMantis | SPoon_TSX: I didn't catch what the problem was.. no voice? |
| 19:46.30 | SPoon_TSX | PMantis: The problem is I can pick up the call but it will drop the call in about 3 seconds. But it only happening on incoming call. |
| 19:47.17 | *** join/#asterisk AsteriskNoob (AsteriskNo@207-114-232-10.gen.twtelecom.net) |
| 19:47.25 | AsteriskNoob | afternoon everyone! |
| 19:47.40 | AsteriskNoob | hey, got a question.... |
| 19:47.57 | PMantis | Ahh, I believe it to be a NAT issue then. The request is coming from one address, ACK from to another (the external IP in sip.conf). Since it's not ACK'd, the call is assumed to be broken, and dropped. |
| 19:48.13 | scoof | yeah |
| 19:48.38 | AgiNamu | PMantis, did somebody say IAX2? |
| 19:48.38 | AsteriskNoob | i've got a system set up with 2 X100p's right now and 8 Cisco 7960's, we are going to be rolling to a TE110P and a TDM400 to run a couple analog for the fax machines and those 8 7960's can a P3 733 handle this? |
| 19:48.43 | scoof | I couldn't find his way around all those 200 OK's |
| 19:48.52 | SPoon_TSX | PMantis: Weird. I got my Asterisk and SIP Phone on the same network, there is no firewall in between. How come I will have NAT issue??? |
| 19:48.53 | AsteriskNoob | (7 channels active on the PRI by the way) |
| 19:49.00 | bjohnson | AgiNamu: no I don't think so |
| 19:49.08 | PMantis | SPoon_TSX: I had a typo there, but that's the basics. Perhaps you should pastebit your (edted for passwords) sip.conf file |
| 19:49.34 | AgiNamu | Spoon_tsx: SIP just sucks that much! |
| 19:49.37 | PMantis | SPoon_TSX: Because Asterisk is assuming that it needs to compensate for a NAT issue, when it doesn't have to. |
| 19:49.40 | bjohnson | AsteriskNoob: likely |
| 19:49.50 | odie_flocon | hey, why is there no zaptel.conf when I build off of the newest CVS? |
| 19:49.57 | PMantis | SPoon_TSX: and yeah... SIP sux. LOL |
| 19:50.08 | bjohnson | AsteriskNoob: a few pages on the wiki about system sizing .. but about 8 concurrent calls is nothing |
| 19:50.13 | SPoon_TSX | PMantis: here you go: http://www.pastebin.com/264903 |
| 19:50.25 | SPoon_TSX | PMantis: Too bad, but I have leave with it. Poor me. = ( |
| 19:50.29 | jhiver | I don't know... SIP does suck of course... but it sortofworks though... |
| 19:50.35 | AsteriskNoob | bjohnson: everything i find just discusses having a dual xeon that barely does 100 calls |
| 19:50.54 | bjohnson | AsteriskNoob: didn't sound like you needed 100 calls |
| 19:51.11 | bjohnson | and I saw a web page that listed an xbox handling 2 calls |
| 19:51.15 | jhiver | IAX doesn't suck but the hardware does. _big time_ |
| 19:51.18 | AgiNamu | dammit. cvs diff -u isnot working |
| 19:51.19 | AsteriskNoob | bjohnson: nope, just 7 incoming pri, the ability to use 4 ports on the TDM and the ability to use 10 phones |
| 19:51.23 | bjohnson | and even a pentium 100 with 16M RAM |
| 19:51.27 | AgiNamu | jhiver, the PA168 phones and ATAs are quite nice. |
| 19:51.27 | scoof | jhiver: SIP's a blessing! I'm trying to get chan_sccp to work currently ;) |
| 19:51.31 | AgiNamu | Better than a grandstream |
| 19:51.31 | PMantis | SPoon_TSX: OK, comment out the externip=70.xxxxxx line, and change all nat=yes lines to nat=no... then reload |
| 19:51.32 | AgiNamu | :) |
| 19:51.44 | AgiNamu | the IAXy sucks, that's all. |
| 19:51.49 | jhiver | I mean does anybody know of like a _reliable_ IAX ATA or phone working? |
| 19:51.50 | Shido6 | AsteriskNoob, how many simultaneous calls on that box? |
| 19:51.52 | PMantis | SPoon_TSX: That *should* be all you need to do. |
| 19:51.59 | jhiver | and don't get me started on the IAXy :) |
| 19:52.03 | AsteriskNoob | bjohnson: its also fully possible that I could have all 7 pri active, all 4 TDM active and up to 20 SIP phones talking... conferences and what not |
| 19:52.07 | AgiNamu | jhiver, I do. I'm using one right now. |
| 19:52.11 | AgiNamu | Actually, I'm using several |
| 19:52.18 | bjohnson | jhiver: didn't AgiNamu tell you that he's using some pa168 devices? |
| 19:52.22 | AsteriskNoob | shido6: not many |
| 19:52.22 | PMantis | jhiver: What's wrong with the IAXy? I've been considering one. |
| 19:52.31 | jhiver | It doesn't work :) |
| 19:52.32 | AgiNamu | ULAW only |
| 19:52.36 | AgiNamu | hear they break a lot |
| 19:52.40 | *** join/#asterisk loick (~loick@APuteaux-151-1-37-144.w82-124.abo.wanadoo.fr) |
| 19:52.52 | AgiNamu | the PA168 has GSM, G729, G723, ULAW, ALAW, and soon iLBC. |
| 19:52.54 | bjohnson | config seems to be a pita |
| 19:53.01 | bjohnson | also read they don't support dhcp |
| 19:53.06 | odie_flocon | Does anybody know why is there no zaptel.conf when I build off of the newest CVS? |
| 19:53.10 | *** join/#asterisk gruph (~tomc@tux.ikano.com) |
| 19:53.12 | AgiNamu | dont do DHCP? lol |
| 19:53.17 | bjohnson | which to me seems to limit their mobility for roaming users |
| 19:53.26 | Dovid | anyone know if there are problems between zaptel and cent os 4 kernel 2.6 ? |
| 19:53.27 | AsteriskNoob | I guess i can just setup MRTG to monitor the CPU/Bandwidth/Simultaneous calls on that box and keep an eye on em |
| 19:53.32 | gruph | does anyone here use Asterisk with SER? |
| 19:53.47 | JerJer[mobile] | sure |
| 19:53.57 | bjohnson | JerJer[mobile]: SER sure? |
| 19:54.21 | bjohnson | fer sure? |
| 19:54.53 | *** join/#asterisk scorpion68 (~chatzilla@HSE-Toronto-ppp186743.sympatico.ca) |
| 19:55.21 | gruph | :) I'm trying to get SER to pass a call to Asterisk... Asterisk is trying to auth the call and SER doesn't respond. Any ideas? or is there somewhere I can look for more resources on SER acting as a UAC to Asterisk? |
| 19:55.32 | SPoon_TSX | PMantis: Still the same. Cut off at about 3 seconds. |
| 19:55.37 | *** join/#asterisk emrah (~emrah@195-137-249-174.ovanet.net) |
| 19:55.56 | PMantis | SPoon_TSX: sip debug show a "(NAT)" ? |
| 19:56.02 | emrah | Hello everybody |
| 19:56.26 | fugitivo | I'm using a linksys pap2-na, and when i call a number from a regular phone, it has a delay of 5 sec before it reaches * |
| 19:56.58 | carbon60 | Anyone know *anything* about the Polycom SoundPoint's "presence" feature? |
| 19:57.09 | JerJer[mobile] | fugitivo: lower the timeout or set a proper dialplan |
| 19:57.30 | gruph | fugitivo: you can also hit "#" to force the call through. |
| 19:57.45 | PMantis | SPoon_TSX: What version os * are you using? |
| 19:58.10 | gruph | fugitivo: hitting # is a temporary work around to push it through if you don't want to adjust the dialplan on the ATA just yet. |
| 19:58.14 | emrah | Anyone know about this error with the Asterisk calling card application? (astcc) http://pastebin.ca/8488 |
| 19:58.17 | fugitivo | found it, thanks guys |
| 19:58.33 | SPoon_TSX | PMantis: How can I tell? |
| 19:59.13 | gruph | SPoon_TSX: from the * console, type "show version" |
| 19:59.23 | scoof | emrah: you need to install some perl modules |
| 19:59.54 | SPoon_TSX | Asterisk CVS-HEAD-03/18/05-11:35:47 built by root@synergize.ca on a i686 running Linux |
| 19:59.55 | *** join/#asterisk scott99 (~kilroy@cpe-66-74-191-249.socal.res.rr.com) |
| 19:59.58 | scoof | emrah: http://asterisk.gnuinter.net/ |
| 20:00.17 | *** join/#asterisk Druken (Druken@67.69.139.226) |
| 20:00.29 | Druken | doesn't the latest stable have realtime? |
| 20:00.51 | emrah | thanks a lot scoof |
| 20:01.18 | *** join/#asterisk IQ (~iq@65-103-165-206.omah.qwest.net) |
| 20:02.15 | gruph | Can anyone point me in a direction of sample configs passing calls from SER to Asterisk? |
| 20:02.27 | harryvv | jsolares, I just talked to one of there sales staff at this company. Realspeak Licence is $5,000 and that included tech support. After the first year each port is $455. |
| 20:02.32 | PMantis | SPoon_TSX: If you're having unexplained problems, (sounds like it's configured OK - should work), try moving to a stable release. Try compiling 1.0.7, and use that. |
| 20:03.10 | johnnyb | does anyone here run asterisk as a realtime process? |
| 20:03.12 | *** join/#asterisk toddf (~toddf@net-66-210-104-252.theshop.net) |
| 20:03.27 | gruph | johnnyb: what do you mean by realtime process? just running it? |
| 20:03.40 | JerJer[mobile] | realtime is one big bug |
| 20:03.43 | PMantis | johnnyb: I don't, but would there be an advantage? |
| 20:03.48 | SPoon_TSX | PMantis: I tried to download the stable version but it just doesn't make call at all. |
| 20:03.48 | Shido6 | http://www.pastebin.com/264909 |
| 20:04.40 | AgiNamu | I'm sure a lot of people would love to know how you scale asterisk to 500 boxes, Jerjer. |
| 20:04.41 | AgiNamu | :) |
| 20:04.46 | PMantis | SPoon_TSX: OK, then I'm sorry - I think I've reached the end of my ability to help. There are others much more knowledgeable than I in this channel. |
| 20:04.51 | JerJer[mobile] | Shido6: username is not necessary |
| 20:06.37 | AgiNamu | Spoon, did you clean your install before installing Stable? |
| 20:06.45 | AgiNamu | just overwriting the source and doing a make install wont work |
| 20:07.10 | jsolares | thanks for the info harryvv |
| 20:07.30 | *** join/#asterisk r0d3nt|m (nobody@wsip-24-234-241-84.lv.lv.cox.net) |
| 20:07.44 | bjohnson | Druken: no |
| 20:08.09 | harryvv | jsolares, the company said thay are number one in this industry so that is why the high cost. |
| 20:08.31 | scoof | SPoon_TSX: does the debug have anything to say about why it believes it should be natting if you reboot the phone? |
| 20:09.05 | Druken | bjohnson: ok, thanks |
| 20:09.57 | jsolares | that they are, the quality of the voice is unsurpassed from everyone elses that i've tried |
| 20:10.19 | mstocco | hey all |
| 20:10.25 | *** join/#asterisk Hmmhesays (negative3k@66.173.103.108) |
| 20:10.39 | mstocco | can anyone comment on expedient as an ISP? |
| 20:10.42 | Hmmhesays | well the eye doctor says i'm not going blind |
| 20:11.02 | file[laptop] | Hmmhesays: that's good |
| 20:11.18 | jsolares | what does the eye doctor say? |
| 20:11.25 | jsolares | too much crt? |
| 20:11.27 | Hmmhesays | yeah I think that is a generally accepted feeling |
| 20:11.27 | file[laptop] | I spy with my little eye! |
| 20:11.30 | AgiNamu | Get your CDRs with REAL uniqueIDs (GUIDs) here: http://bugs.digium.com/bug_view_page.php?bug_id=0003780 |
| 20:11.48 | sudhir492 | JerJer[mobile]: What version of PWLib and Openh323 for CVS head for chan_h323 to work? |
| 20:11.48 | Hmmhesays | well it started off with "dear penthouse I never though it would happen to me......." |
| 20:11.58 | jsolares | hehehe |
| 20:11.59 | sudhir492 | JerJer[mobile]: where to get them from |
| 20:12.11 | Hmmhesays | sudhir492 read the readme |
| 20:12.26 | Hmmhesays | er.. wait, she said I need to change my contacts more often |
| 20:12.28 | SPoon_TSX | scoof: nope. |
| 20:13.00 | sudhir492 | Hmmhesays: I just read and I cannot find those versions on sourceforge, neither at nufone.net/downloads |
| 20:13.35 | Hmmhesays | www.openh323.org/bin |
| 20:14.53 | Hmmhesays | your welcome |
| 20:14.56 | file[laptop] | lol |
| 20:15.04 | file[laptop] | a commercial or 'newphone' just came on... thought they were talking about nufone |
| 20:15.17 | file[laptop] | er for |
| 20:16.08 | JerJer[mobile] | funny |
| 20:16.19 | JerJer[mobile] | there is inFone |
| 20:16.22 | JerJer[mobile] | as well |
| 20:16.29 | *** join/#asterisk vidia22 ([U2FsdGVkX@vidiamob4.vidiacom.com) |
| 20:16.41 | file[laptop] | JerJer[mobile]: it almost freaked me out |
| 20:16.43 | Maxxed | proxy_register: 1 is what makes my 7940 ipphone register its self to asterisk right? |
| 20:17.06 | JerJer[mobile] | sudhir492: grab the latest release code from sourceforge and by the time you get it compiled i'll have cvs -head updated |
| 20:17.06 | Hmmhesays | upfone downfone? |
| 20:17.17 | scott99 | Anyone have a recommendation for a company that I can pay to take my SIP traffic if I wanted about 10000 minutes of talk time? |
| 20:17.20 | jsolares | oldFone |
| 20:17.25 | AgiNamu | OldPhone: When touch tone just don't cut it. |
| 20:17.30 | Hmmhesays | youngfone? |
| 20:17.30 | scoof | SPoon_TSX: there seems to be some complexity in the selection of NAT in chan_sip.c |
| 20:17.39 | Hmmhesays | is bjohnson from boston? |
| 20:17.39 | scoof | SPoon_TSX: I'm trying to wrap my head around it |
| 20:17.41 | JerJer[mobile] | deadFone |
| 20:17.48 | AgiNamu | F***Fone |
| 20:17.49 | JerJer[mobile] | won'tworkFone |
| 20:17.57 | AgiNamu | thisisreallystupidPhone |
| 20:18.01 | JerJer[mobile] | jitterFone |
| 20:18.13 | JerJer[mobile] | phuckFone |
| 20:18.14 | JerJer[mobile] | :) |
| 20:18.18 | *** join/#asterisk Syrus_ (~pascal@tahiti.mpl.rullier.net) |
| 20:18.20 | jsolares | hehe |
| 20:18.23 | SPoon_TSX | scoof:thsnks. |
| 20:18.23 | sudhir492 | JerJer[mobile]: thannks. |
| 20:18.28 | vidia22 | newbi question - how can I tell if a fxo card is installed properly and being "used " by the system? |
| 20:18.28 | AgiNamu | phuck phone? is that for hockey scores? |
| 20:18.38 | Hmmhesays | vidia22: look at the pretty lights |
| 20:18.44 | MikeJ[Laptop] | who long does it take to get a did from nufone.... |
| 20:18.45 | AgiNamu | newBI question? |
| 20:18.47 | bjohnson | vidia22: it answers calls and can make calls |
| 20:18.55 | Hmmhesays | there are four lights! |
| 20:18.58 | jsolares | MikeJ[Laptop], months if it's a michigan did :X |
| 20:19.01 | bjohnson | MikeJ[Laptop]: at LEAST 4 days |
| 20:19.17 | sudhir492 | I already downloaded the latest release last night :-) Off to compiling now... Talk to you in 15 mins! |
| 20:19.22 | Maxxed | hey, proxy_register: 1 is what makes my 7940 ipphone register its self to asterisk right? |
| 20:19.25 | MikeJ[Laptop] | at least weeks... I'll let you know if it turns into months. |
| 20:19.36 | bjohnson | sudhir492: for h323? famous last words |
| 20:19.40 | MikeJ[Laptop] | you'd think they would mention that.... |
| 20:19.48 | vidia22 | there are no lights - and it doesnt anse\wer or make calls - but I think my config is screwed as well... |
| 20:19.56 | bjohnson | MikeJ[Laptop]: fours weeks for a DID from sixtel for me |
| 20:20.05 | Hmmhesays | haha, well.... you missed my obscure star trek reference |
| 20:20.21 | jhiver | naaah we know |
| 20:20.32 | jhiver | when jl picard is being tortured and stuff |
| 20:20.39 | jhiver | man that's so cheezy :) |
| 20:20.40 | MikeJ[Laptop] | I was told 24 hrs, then the next day 48, then a couple of days later, it could take a while... we'll let you know... I frankly gave up. |
| 20:20.58 | bjohnson | jhiver: wsn't he being forced to watch Star Trek movies? |
| 20:21.02 | jsolares | the 1800 did i got from nufone was instantenously |
| 20:21.03 | *** join/#asterisk IronHelix (~irc@ool-182c3fe9.dyn.optonline.net) |
| 20:21.06 | AgiNamu | "it doesnt answer or make calls" <-- good indication its not working |
| 20:21.18 | Maxxed | ah, it is :) |
| 20:21.19 | AgiNamu | nufone provisions fast for me. |
| 20:21.22 | bjohnson | MikeJ[Laptop]: good for you |
| 20:21.25 | AgiNamu | So does RNK |
| 20:21.35 | sudhir492 | bjohnson : Shhhh. With JerJer around you are not saying anything bad about h323 :-) |
| 20:21.37 | bjohnson | AgiNamu: we don't all rank so highly as you |
| 20:21.42 | jhiver | mind you star trek NG was _so much better_ than the crappy 'enterprise' episodes :) |
| 20:21.43 | AgiNamu | RNK kicks ass actually |
| 20:21.44 | harryvv | bjohnson, wow 4 weeks? |
| 20:21.53 | bjohnson | harryvv: yes |
| 20:22.00 | JerJer[mobile] | MikeJ[Laptop]: i personally emailed you the configs for your royal oak number |
| 20:22.01 | harryvv | wondr why the wait |
| 20:22.14 | harryvv | or is it just not enough manpower to handle the workload. |
| 20:22.19 | bjohnson | I'll show you a royal oak ... |
| 20:22.22 | MikeJ[Laptop] | not to me. |
| 20:22.27 | jhiver | one thing I like about NuFone is that instead of cutting me off they let the balance go negative and sent me an email |
| 20:22.32 | JerJer[mobile] | whatever email address is assocated to your account then |
| 20:22.38 | jhiver | I though this is cool not to just cut you off |
| 20:22.42 | MikeJ[Laptop] | nope. |
| 20:22.45 | JerJer[mobile] | yep |
| 20:22.48 | bjohnson | jhiver: I actually prefer that |
| 20:22.50 | harryvv | bjohnson, thats not good if a customer wants to have a did in the states and cannot wait. |
| 20:22.51 | bjohnson | nope |
| 20:22.52 | MikeJ[Laptop] | try again? |
| 20:22.52 | bjohnson | yep |
| 20:23.05 | *** join/#asterisk gtigene (~gnadenx@c-67-184-112-58.hsd1.il.comcast.net) |
| 20:23.15 | jhiver | bjohnson you prefer what ? |
| 20:23.18 | bjohnson | harryvv: I don't give a damn about the states |
| 20:23.20 | jhiver | being cut off? |
| 20:23.24 | file[laptop] | any Canadians around? |
| 20:23.27 | bjohnson | jhiver: being cut off |
| 20:23.28 | harryvv | bjohnson, :) |
| 20:23.35 | AgiNamu | jhiver, t's also a fuckload easier to let you ride a bit |
| 20:23.36 | tzanger | yup |
| 20:23.39 | tzanger | <-- canadian |
| 20:23.40 | AgiNamu | rather then enforce a zero-balance limit. |
| 20:23.41 | JerJer[mobile] | jhiver: the problem is not everyone likes to pay their bills - so that behaviour has to change |
| 20:23.42 | bjohnson | jhiver: err .. referring to voip accounts |
| 20:23.53 | tzanger | speaking of balance |
| 20:23.57 | tzanger | I should check my nufone balance |
| 20:23.59 | bjohnson | file[laptop]: we're all here |
| 20:24.02 | mstocco | <-- also a Canadian |
| 20:24.09 | jhiver | like you're in this super important deal on the phone that's gonna make u filthy rich and BANG! cut off |
| 20:24.12 | file[laptop] | bjohnson: know who I would call to correct the name/address on my taxes? |
| 20:24.13 | jhiver | great... |
| 20:24.16 | file[laptop] | been wrong for two years... |
| 20:24.19 | tzanger | still +ve |
| 20:24.26 | bjohnson | file[laptop]: revenue canada |
| 20:24.30 | file[laptop] | I'm thinking Customs & Revenue, but their site is like ... kaput |
| 20:24.32 | jhiver | It's not *too* hard to enforce, a zero balance limit |
| 20:24.38 | jhiver | I don't see what's the problem |
| 20:24.44 | bjohnson | file[laptop]: they have a web site with all sorts of contact info |
| 20:24.47 | AgiNamu | no, but an order of magnitude harder than just delay billing the CDRs |
| 20:25.01 | jhiver | it's like: |
| 20:25.02 | AgiNamu | jhiver, cause when a call comes in, you gotta go query the balance, rate it, etc. |
| 20:25.05 | tzanger | sweet |
| 20:25.10 | AgiNamu | As well as keep track of simulatenous calls, etc. |
| 20:25.15 | tzanger | 2160s call, $1.32 |
| 20:25.16 | jhiver | - freeze some money - phone - recredit unused money |
| 20:25.20 | harryvv | file[laptop], what city are you in |
| 20:25.22 | jhiver | that's what I do |
| 20:25.25 | file[laptop] | harryvv: Moncton |
| 20:25.26 | jhiver | easy! |
| 20:25.28 | harryvv | K |
| 20:25.35 | AgiNamu | sure, but it's a lot easier to just "phone" :) |
| 20:25.46 | jhiver | lol ok :) |
| 20:25.46 | harryvv | Dont know what the number there for cra is but its online. |
| 20:26.13 | file[laptop] | trying to get it, but the cra site is not working for me |
| 20:26.35 | scoof | SPoon_TSX: this problem is when you're dialing the phone, right? |
| 20:26.36 | jhiver | Anyway I think *thumbs up* for the current nufone 'don't cut off' policy |
| 20:26.43 | jhiver | if it changes well that's too bad |
| 20:27.08 | jhiver | off course i immediately added some money on my account :) |
| 20:27.28 | JerJer[mobile] | there will be a small amount of credit provided, until it is abused |
| 20:27.45 | jhiver | ah cool |
| 20:27.55 | harryvv | cra in surrey is expanding and upgrading everything. money is cheap these days :) |
| 20:27.55 | Maxxed | ah yes, success! :) |
| 20:28.00 | jhiver | well if you let go of say 2 dollars top |
| 20:28.09 | SPoon_TSX | scoof: Nope, it is from PSTN -> Asterisk -> SIP Phone. |
| 20:28.17 | jhiver | i don't know, it depends on the size of payments i guess |
| 20:28.23 | scoof | SPoon_TSX: ok, that was what I meant ;) |
| 20:28.29 | jhiver | maybe it should be like minus 5% of last payment |
| 20:28.48 | scoof | SPoon_TSX: I'd really like a sip-debug from register all the way through a failed call |
| 20:28.54 | jhiver | so if you paid 10$ you can go 50 cents in the red |
| 20:29.02 | scoof | SPoon_TSX: I think I have an idea where it goes wrong |
| 20:29.17 | SPoon_TSX | scoof: How can I capture those information for you? |
| 20:29.25 | jhiver | although I went like minus 1.5$ and I paid only like 10 or 20 bucks originally |
| 20:29.31 | bjohnson | file[laptop]: http://www.ccra-adrc.gc.ca/ |
| 20:29.37 | scoof | SPoon_TSX: by booting the phone after enabling sip debug ip <ip> |
| 20:29.46 | scoof | SPoon_TSX: and then performing a call |
| 20:29.53 | file[laptop] | bjohnson: yes I know it... |
| 20:29.55 | SPoon_TSX | ok. one second. |
| 20:29.56 | file[laptop] | it's just... incredibly slow |
| 20:29.57 | bjohnson | oh |
| 20:30.12 | scoof | SPoon_TSX: depending on the via-lines in a sip request, Asterix may choose to enable NAT |
| 20:30.17 | scoof | Asterisk |
| 20:30.20 | Maxxed | hey, im curious, when i get a call, i have caller id tell me if my extention was direct dialed or its a cue call, well under the caller id msg, it tells me what i want then under that astterisk, how to i take that asterisk out from under my caller id |
| 20:30.24 | Maxxed | i use, exten => s,1,SetCallerID(Direct Dial) |
| 20:30.28 | bjohnson | file[laptop]: try this 1 800 387-1193 |
| 20:30.43 | file[laptop] | busy |
| 20:31.05 | file[laptop] | it's "ignore file and not let him get to the CRA via website or phone" day |
| 20:31.06 | bjohnson | file[laptop]: Individual income tax enquiries: 1-800-959-8281 |
| 20:31.12 | *** join/#asterisk bah (048830696@AC93E902.ipt.aol.com) |
| 20:31.19 | tzanger | file[laptop]: damn, I always forget that day |
| 20:31.26 | file[laptop] | that number is busy as well |
| 20:31.27 | bjohnson | the other one was a joke .. |
| 20:31.38 | bjohnson | (family benefits) |
| 20:32.10 | bjohnson | file[laptop]: better keep trying .. it's almost quitting time for them |
| 20:33.23 | AgiNamu | shit... fark.com is down. |
| 20:33.23 | *** join/#asterisk ell (~ali@66-207-218-199.beanfield.net) |
| 20:33.26 | AgiNamu | how will i waste time |
| 20:33.31 | harryvv | I dont understand this country as far as its taxes are concerned. I get back 500 -700 for a normal working job in washing almost every yeare yet I had to pay taxes owed here. Thats a first time in my life had to pay any taxes owed. |
| 20:33.46 | file[laptop] | you had taxes owed? how odd |
| 20:33.57 | harryvv | yea had to pay 57 dollars |
| 20:34.52 | harryvv | I am from the states and never in my life owed any taxes to the irs. BTW everythign in bc is expensive ;) I even pay interest on my car insurance. If a person cannot pay one years of car insurance up front thay finance it for you. |
| 20:34.57 | SPoon_TSX | scoof: Full dump here > http://www.pastebin.com/264928 |
| 20:35.18 | jsolares | AgiNamu, ISR? IVA? :O |
| 20:35.25 | AgiNamu | nope |
| 20:35.27 | AgiNamu | fuck that shit |
| 20:35.35 | AgiNamu | i pay IVA cause sometimes they include it |
| 20:35.44 | jsolares | sometimes... |
| 20:35.48 | AgiNamu | but in those cases, i sell my invoices to someone else. |
| 20:35.59 | AgiNamu | guatemala's tax system needs to be killed. |
| 20:36.08 | AgiNamu | for those that dont know |
| 20:36.11 | jsolares | it's not as bad as other countries |
| 20:36.22 | AgiNamu | it makes the assumption that the government has a super database that somehow gets every single invoice in it |
| 20:36.47 | AgiNamu | to legally sell anything, you have to get a registered invoice design approved, printed by a registered approved printing company, and then have approved serial numbers on them. |
| 20:36.58 | AgiNamu | the idea being that you can't just make up an invoice. |
| 20:37.08 | AgiNamu | Of course, liek most things the government comes up with, it just doesnt work :P |
| 20:37.20 | AgiNamu | yea, I hear italy has a fucked up system too :P |
| 20:37.42 | scoof | SPoon_TSX: line 70: no nat yet |
| 20:37.44 | jsolares | hehe |
| 20:37.48 | jhiver | Yeah sounds like screwed |
| 20:37.56 | jsolares | we're in paradise with the IVA/ISR compared to many countries |
| 20:38.03 | jsolares | they're easy to evade :X |
| 20:38.07 | jhiver | how about you're buying something from some other country... => screwed |
| 20:38.09 | harryvv | Untill a year ago anyone Here in BC who made over 45,000 per year had to pay 50% income tax. |
| 20:38.19 | AgiNamu | yea, that's true. it's so ineffective, so long you dont report, you're ok :) |
| 20:38.21 | scoof | SPoon_TSX: there's no NAT in that dump at all! |
| 20:38.27 | AgiNamu | plus , im canadian, and not a resident |
| 20:38.34 | AgiNamu | i dont make any money in canada, so i dont pay canadian taxes. |
| 20:38.46 | scoof | SPoon_TSX: is that a dump of a failed call? |
| 20:38.47 | AgiNamu | poor americans have to report everything they do to big brother, no matter where they live. |
| 20:38.54 | harryvv | agi, you are canadian living in south america? |
| 20:38.59 | jsolares | are you sure you dont need to? |
| 20:39.34 | SPoon_TSX | scoof: Yes. |
| 20:39.36 | AgiNamu | yep |
| 20:39.42 | AgiNamu | and yes jsolares. i dont need to report. |
| 20:39.43 | jhiver | harryvv is that 50% above the 45k or 50% for the full wage? |
| 20:39.45 | scoof | SPoon_TSX: can you get me a tcpdump of this? |
| 20:40.01 | AgiNamu | esp. since i dont do business in guatemala either :P |
| 20:40.09 | harryvv | jhiver not exactly sure. |
| 20:40.09 | jsolares | hehehehe |
| 20:40.17 | SPoon_TSX | scoof: I am wondering may be I should install the stable version instead of the most update CVS version and try. What do you think? |
| 20:40.35 | AgiNamu | SPoon, good idea. |
| 20:40.35 | jhiver | cause if it was for the full wage it would suck earning 46k :) |
| 20:40.40 | AgiNamu | install 1.0.6 |
| 20:40.49 | scoof | SPoon_TSX: but then we won't find the (potential) bug |
| 20:40.51 | AgiNamu | my friend pays tax down here |
| 20:40.53 | AgiNamu | poor guy |
| 20:41.04 | AgiNamu | but he makes a boatload of money (relative) so |
| 20:41.15 | AgiNamu | i guess he's ok with it. |
| 20:41.28 | jsolares | hehe |
| 20:41.51 | AgiNamu | im officially bored as of .... no. |
| 20:41.52 | AgiNamu | now. |
| 20:41.55 | harryvv | jhiver I was wrong on those rates. |
| 20:42.04 | bjohnson | AgiNamu: go to honduras and get me some cigars |
| 20:42.07 | AgiNamu | jhiver, whats the dialcode for reunion |
| 20:42.13 | jhiver | 262 |
| 20:42.13 | jsolares | go find me a better tts than cepstral |
| 20:42.14 | AgiNamu | honduras? why not cuban |
| 20:42.17 | jhiver | why? |
| 20:42.19 | harryvv | Anything over $103,000 in earned income was 43.7% |
| 20:42.23 | bjohnson | I prefer honduran |
| 20:42.25 | AgiNamu | jsolares: sure, windows has SAPI |
| 20:42.31 | SPoon_TSX | Just wondering how can I download the stable release 1.0.7 off the internet? |
| 20:42.31 | AgiNamu | just wondering |
| 20:42.35 | AgiNamu | wanna see what my rate is |
| 20:42.37 | jsolares | eww |
| 20:42.38 | bjohnson | plus .. honduras is right next to guatemala |
| 20:42.47 | AgiNamu | reunion island? |
| 20:42.49 | jhiver | yeah lemme know what your rate is |
| 20:42.50 | jhiver | yeah |
| 20:43.08 | AgiNamu | our residential rate is 9.7cents |
| 20:43.16 | SPoon_TSX | What commend should I use for the download? |
| 20:43.18 | jsolares | ouch |
| 20:43.21 | rvhi | anyone uses a pap2 with credit card machine? |
| 20:43.22 | jhiver | is that the price you buy or the price you pay? |
| 20:43.28 | AgiNamu | thats the price I sell at. |
| 20:43.29 | rvhi | it fails 4 out of 5 times |
| 20:43.32 | jhiver | ah ok |
| 20:43.35 | AgiNamu | for my residential customers. |
| 20:43.40 | jhiver | and how much do you buy it for? |
| 20:43.45 | AgiNamu | less |
| 20:43.48 | jsolares | hehehe |
| 20:43.49 | jhiver | of course :) |
| 20:44.03 | AgiNamu | what are good prices? |
| 20:44.07 | jsolares | i buy at 1.3cents i think |
| 20:44.14 | AgiNamu | to Reunion Island??/ |
| 20:44.16 | jhiver | it would have been useful for me to compare the price you buy it for and how much I can get it for over here |
| 20:44.24 | *** part/#asterisk kingcobra (~mwehner@214.35.233.64.transedge.com) |
| 20:44.26 | jsolares | what who where? |
| 20:44.33 | jsolares | i think i missed a sentence somewhere |
| 20:44.39 | jsolares | wtf is reunion island? |
| 20:44.39 | AgiNamu | were talking about termination to "Reunion Island" |
| 20:44.40 | harryvv | I need to get a web site up that takes prepayments for call billing. |
| 20:44.47 | *** join/#asterisk srineer (~srineer@209.50.133.4) |
| 20:44.50 | AgiNamu | some place with medium rates |
| 20:44.52 | jsolares | yeah i definetely missed that sentence |
| 20:45.23 | jhiver | harryvv I could do that for you for a fee - pm me if you wish |
| 20:45.47 | AgiNamu | well, with say, 1 million minutes, I can sell at $0.0795 |
| 20:45.54 | *** join/#asterisk FuriousGeorge (~brian@ool-43516ebb.dyn.optonline.net) |
| 20:46.00 | AgiNamu | but thats not making much money for me |
| 20:46.40 | AgiNamu | of course, we've spent over $$,$$$ getting termination contract |
| 20:46.42 | jhiver | I can buy local calls @ 2 euro cents per minute - so I could sell @ 4-5 I guess... |
| 20:46.44 | FuriousGeorge | hey all. i have a small office i wanted to do an * test run on. I have a 1ghz celeron CPU (100 mhz fsb). will that have enough muscle for three concurrent conversations |
| 20:46.50 | jsolares | yeah i get reunion island proper for 0.0778 from voipjet |
| 20:46.54 | AgiNamu | jhiver... intersted in doing some bypass business?? :) |
| 20:47.04 | jhiver | if you have some volume, yes! |
| 20:47.07 | AgiNamu | 7.7 wow |
| 20:47.23 | AgiNamu | well, with a price difference of 3-4 cents, we can find volume! |
| 20:47.26 | JerJer[mobile] | until they run out of cash, since they are selling for a loss |
| 20:47.28 | AgiNamu | i dont even know where the hell it is. |
| 20:47.38 | jsolares | just dont call cellulars, 0.2921 |
| 20:48.13 | AgiNamu | furios, yes. |
| 20:48.30 | scoof | SPoon_TSX: how about that tcpdump before you go to stable? |
| 20:48.40 | johnnyb | Has anyone here had a specific TDM module have echo, but not the others? |
| 20:49.19 | AgiNamu | voipjet has loow rates. |
| 20:49.25 | AgiNamu | jsolares, you interested in doing bypass ? :) |
| 20:49.42 | *** join/#asterisk RomanTorres (~root@200.106.49.195) |
| 20:49.45 | AgiNamu | in GT, I can get a PRI and make local calls for 2 cents. 4 cents for cell. |
| 20:49.49 | jsolares | i dont want to go to jail :P |
| 20:50.05 | AgiNamu | and the cheapest price internationally, is like 8.5 cents or so. |
| 20:50.11 | AgiNamu | with a lot of volume |
| 20:50.23 | AgiNamu | so even selling at 7 cents... i can make a ton |
| 20:50.38 | AgiNamu | you know anyone arrested for that? |
| 20:50.44 | AgiNamu | i think telgua just says that kinda shit. |
| 20:50.54 | RomanTorres | Hi everybody, greetings from Mexico City, my name is Roman Torres, I am working with Asterisk for Call Centers and E1 links. |
| 20:51.01 | AgiNamu | cause telgua is full of shit and likes charging, get this, $1/minute to USA |
| 20:51.02 | *** join/#asterisk Wazb (Wazb@207.245.215.111) |
| 20:51.13 | jsolares | well the head at avaya was looking into doing it, beacuse of the $$$$, but a friend says you can get killed for that shit |
| 20:51.14 | AgiNamu | Hello Roman, I am AgiNamu. |
| 20:51.23 | RomanTorres | Hi AgiNamu |
| 20:51.33 | AgiNamu | I talked to someone who had done it. they said telgua just cut their lines. |
| 20:51.38 | FuriousGeorge | allow me to rephrase: how much processor is necessary to hold three conversations on an * server |
| 20:51.38 | *** join/#asterisk zotz (~zotz@24.231.32.191) |
| 20:51.47 | AgiNamu | furious, depends what you're doing |
| 20:51.57 | AgiNamu | if you're transcoding iLBC to G729, a lot more than just doing passthru |
| 20:52.07 | FuriousGeorge | mostly forewarding calls to another location |
| 20:52.11 | jsolares | legally you have to pay them termination fees or something, and that drives up the price too much (from what i've understood) |
| 20:52.15 | emrah | I'm sorry to disturb again, but I have some problems with the AstCC programme. When I try to create the databe, it says failed, but no reason... My information are correct... How should I do? Do you have any guide for this programme? |
| 20:52.40 | jsolares | emrah, how familiar with mysql/linux are you? |
| 20:52.58 | *** join/#asterisk gruph (~tomc@tux.ikano.com) |
| 20:53.16 | RomanTorres | We have had for two months already a double xeon HP 330 server with 8 E1 (240 channels) with 240 incoming calls without problems. |
| 20:53.21 | emrah | I can't answer that question jsolares |
| 20:53.36 | jsolares | i can't answer your question |
| 20:53.40 | emrah | I can check the logs |
| 20:53.42 | gruph | does anyone know how to authenticate a sip client based on their source address instead of a username/password? |
| 20:54.02 | emrah | But I'm just wandering if it's possible to have a README or someting like that to explain how to do. |
| 20:54.02 | jsolares | i had problems with astcc and it's database so i went and created it myself using the cgi source as base |
| 20:54.11 | AgiNamu | furious, then a 1GHz Celeron can do a TON more than 3 calls |
| 20:54.15 | Hmmhesays | i'm having a mental block, what is the command to send a dtmf digit |
| 20:54.16 | AgiNamu | if its just forwarding iax or sip traffic. |
| 20:54.42 | srineer | AgiNamu, how about say 30 calls? |
| 20:54.45 | emrah | gruph: I think it's possible by only specifiing the host=IP without a username and password in sip.conf. |
| 20:54.48 | AgiNamu | definately |
| 20:54.55 | AgiNamu | www.astertest.com |
| 20:54.57 | srineer | AgiNamu, how about say 130 calls? |
| 20:55.12 | RomanTorres | gruph, check on the defaultip parameter, asterisk will try first to send te call to that address even if yur phone is not registered. |
| 20:55.14 | jsolares | try it |
| 20:55.27 | AgiNamu | probably. Astertest has details. |
| 20:55.54 | srineer | cool |
| 20:56.14 | gruph | RomanTorres: it's actually the other way.. I don't want Asterisk to challenge the client if it's from a certain IP address... when it registers or places a call (SIP INVITE) |
| 20:56.36 | sudhir492 | FuriousGeorge: No matter what you are doing, 1GHz celeron is more than enough to hold 3 conversation |
| 20:56.53 | FuriousGeorge | AgiNamu: i have a dsl connection (300/80 KB/S down/up) and thinking about getting another. the box would mostly foreward calls to another location |
| 20:57.11 | AgiNamu | if its just passing them off, then it's more , way more, than enough |
| 20:57.20 | FuriousGeorge | sudhir492: will the bandwidth i just described cerate a bottleneck? |
| 20:57.36 | FuriousGeorge | with less compressed codeecs |
| 20:57.56 | jsolares | why not use g729 |
| 20:57.57 | *** join/#asterisk nel (~oeo@199.75.106.33) |
| 20:58.05 | sudhir492 | You are talking 80Kbps, not 80KBps (which happens to close to 384Kbps) correct |
| 20:58.15 | *** join/#asterisk grendal_prime (~grendal@son-216-86-177-153.static.mlode.com) |
| 20:58.20 | AgiNamu | G729 is nice. |
| 20:58.28 | FuriousGeorge | having never really listened to different codecs im not sure how much actual difference there is in soundquality. but thats my nconcern |
| 20:58.29 | epoch | ...for me to poop on! |
| 20:58.31 | AgiNamu | if anyone wants to play with the g729 codec... check out my site. |
| 20:58.37 | grendal_prime | ok i just want a SIP SERVER so i can kphone a couple of computers...is that all that difficult? |
| 20:58.40 | AgiNamu | you can probably learn enough to make your own patch |
| 20:58.44 | AgiNamu | and activate a ton of licenses. |
| 20:59.01 | jsolares | i'm using intel's code :X |
| 20:59.14 | sudhir492 | FuriousGeorge: where are you forwarding your calls to? Another Asterisk? |
| 20:59.16 | AgiNamu | im using digiums, with licenses. |
| 20:59.18 | nel | anybody had problems with echo and pstn lines? |
| 20:59.21 | AgiNamu | but I also have digiums, without liecnes. |
| 20:59.23 | *** join/#asterisk Sedorox (brandon@Neptune.client.wlmsprt.pa.sed6.net) |
| 20:59.31 | AgiNamu | It's a single BIT patch :) |
| 20:59.33 | *** join/#asterisk toddf (~toddf@net-66-210-104-252.theshop.net) |
| 20:59.48 | FuriousGeorge | sudhir492: nit yet but some day |
| 20:59.53 | FuriousGeorge | *not yet |
| 20:59.55 | grendal_prime | is there anyone who can help me with basic config of a SIP server? |
| 20:59.59 | AgiNamu | so, theoretically, a gamma ray could strike your disk, and activate your codec_g729.so for tons of licenses. |
| 21:00.11 | FuriousGeorge | to PSTN lines at first |
| 21:00.14 | ddum | Damn... just realized there is no way for me (it seem) to do * with CID... GAH. |
| 21:00.14 | sudhir492 | FuriousGeorge: Tell me exactly how you plan to use your asterisk? |
| 21:00.24 | RomanTorres | gruph, that is a strange problem, may I ask the objetive for not letting certain ip addresses to register? because that can be easily done with the linux iptables command |
| 21:00.33 | FuriousGeorge | the box would mostly foreward calls to another location |
| 21:00.40 | FuriousGeorge | its for an office no one is ever at |
| 21:00.45 | FuriousGeorge | after 5 rings foreward |
| 21:00.56 | nel | any ideas on how to troubleshoot zaptel card for echo? |
| 21:01.04 | sudhir492 | JerJer[mobile]: I am done compiling PWLib and Openh323 :-) |
| 21:01.10 | grendal_prime | RomanTorres, maybe you do want individuals to connect to other services |
| 21:01.11 | nel | the other party don't hear the echo, just me calling |
| 21:01.16 | RomanTorres | nel, check the rxgain and txgain parameters on zaptel.conf |
| 21:01.22 | grendal_prime | just not voip..or whatever.. |
| 21:01.48 | gruph | RomanTorres: I'm having an ATA place a call to a SER server, and having the SER server proxy to an Asterisk server to go out to the PSTN. |
| 21:02.12 | gruph | RomanTorres: I can't get SER to respond to an authorization request from Asterisk. |
| 21:02.17 | grendal_prime | you knock them all out with iptables..and well it might just be easyer to use the asterisk app to limit that sort of traffic |
| 21:02.59 | *** join/#asterisk avish (~avishnev@ool-4573cda8.dyn.optonline.net) |
| 21:03.17 | *** join/#asterisk Blissex (~Blissex@82-69-39-138.dsl.in-addr.zen.co.uk) |
| 21:03.58 | RomanTorres | gruph, you can check what is going on if you do this command on the asterisk box: tethereal -p -n port 5060, if the problem is the authorization request maybe it will show you more information |
| 21:04.09 | nel | romantorres: how do those rx and tx parameters affect echo? |
| 21:04.36 | FuriousGeorge | sudhir492: did u see my respone to your question, above |
| 21:04.38 | RomanTorres | nel, when you put a negative value on the rxgain and txgain parameters echo is diminished |
| 21:05.07 | gruph | RomanTorres: I'm actually using tcpdump printing out the raw data watching the SIP traffic. that's how I knew that SER wasn't responding to the auth request. |
| 21:05.52 | avish | anyone has experience with AGI/EAGI? |
| 21:06.04 | RomanTorres | gruph, thethereal shows you a lot more info than tcpdump, since tcpdump only shows ip addresses and ports, tethereal show you more information about sip requests and answers |
| 21:06.13 | Hmmhesays | avish: some love it some hate it, some don't care |
| 21:06.49 | avish | i guess i was wondering how to write an application for asterisk without spawning a new process every call |
| 21:06.56 | sudhir492 | FuriousGeorge: Calls coming on PSTN line to Asterisk box, couple of SIP phones in the office, incoming calls first routed to one of the SIP extensions, if the no one picks up then the call is forwarded to someone else through another PSTN line, correct ? |
| 21:07.00 | gruph | RomanTorres: if you run "tcpdump -s 0 -e -w - udp and port 5060" it'll print the contents of the packets out to the command line. |
| 21:07.08 | RomanTorres | gruph, you are right. |
| 21:07.40 | sudhir492 | FuriousGeorge: Need a complete picture to better help you |
| 21:07.52 | gruph | RomanTorres: thanks for the suggestion though. I appreciate it! |
| 21:08.30 | FuriousGeorge | sudhir492: sort of. calls ring on two analog extensions, if no answer, foreward to another PSTN line via SIP (or maybe another pots, depends which is cheaper) |
| 21:09.09 | emrah | jsolares: Can you please have a look at that? (sorry to disturb you...) http://pastebin.ca/8492 |
| 21:09.11 | FuriousGeorge | obviously, if i do pots i dont need too much bandwidth |
| 21:09.12 | RomanTorres | gruph, are you using the "canreinvite=no" parameter? since if you dont have it maybe asterisk tries to contact the sip client direclty instead on the SER |
| 21:09.17 | sudhir492 | FuriousGeorge: Whatever you are doing, you have enough muscles in your processors. |
| 21:09.40 | FuriousGeorge | thanks, thats really what i was hoping to hear |
| 21:09.54 | gruph | RomanTorres: no I don't... let me give that a try..... |
| 21:10.06 | sudhir492 | FuriousGeorge: Assume around 32Kbps bandwidth consumption per G729 SIP channel, you can figure the rest |
| 21:10.33 | tzanger | ... 32kbps for g729? |
| 21:10.52 | emrah | jsolares: ? |
| 21:10.54 | tzanger | the payload is only 8kbps, why would overhead be quadrupling that? |
| 21:11.04 | tzanger | hell my gsm wirespeed is about 40kbps |
| 21:11.51 | emrah | Anyone can help me please? (sorry to disturb you. ) |
| 21:11.55 | *** join/#asterisk lImbus (lImbus@104-174.244.81.adsl.skynet.be) |
| 21:11.56 | nel | is it normal to have echo on your side using FXO card ? |
| 21:12.53 | FuriousGeorge | i got one more quick theoretical question, then im gonna go practice: if i have two internet connections "bridged" and one goes down is it transparent to asterisk? how much configuration is involved on the asterisk side for that failsafe? |
| 21:13.07 | *** join/#asterisk r0d3nt|m (nobody@wsip-24-234-241-84.lv.lv.cox.net) |
| 21:13.38 | scoof | FuriousGeorge: impossible to tell from that description |
| 21:13.59 | jsolares | emrah, use the latter create table cdrs |
| 21:14.04 | jsolares | the one with callstart |
| 21:14.54 | emrah | sorry? |
| 21:15.06 | FuriousGeorge | scoof: i lack the vocab to elaborate much better: howabout this two modems go into my asterisk box which does all the routing and firewall stuff (a third NIC sends connection to LAN) |
| 21:15.20 | jsolares | emrah, use those create tables |
| 21:15.27 | jsolares | but only use the second for the table cdrs |
| 21:15.35 | FuriousGeorge | ive never set that up on a linux box before, but assuming i did, is it transparent to asterisk if one connection goes down |
| 21:15.35 | jsolares | the one with callstart in it |
| 21:15.38 | nestAr | hrmmm |
| 21:15.46 | SPoon_TSX | May I know how to downlaod the stable release off the internet? |
| 21:15.47 | scoof | FuriousGeorge: you would need some routing mechanism or VRRP-style setup at your provider to fail over there |
| 21:15.55 | nestAr | well.. i've run into a problem with my CheckGroup/SetGroup logic |
| 21:16.13 | scoof | FuriousGeorge: there's a multitude of ways to set that up |
| 21:17.07 | FuriousGeorge | scooif: in windows you can right click on your NIC's icon and bridge the connection, it works as ive described. which of the multitude of ways makes that happen for linux |
| 21:17.07 | scoof | FuriousGeorge: but Asterisk doesn't need be involved in any of them, they're operating-system- or router-specific |
| 21:17.13 | *** join/#asterisk subtract (~subtract@ottawa-hs-209-217-119-73.d-ip.magma.ca) |
| 21:17.32 | FuriousGeorge | scoof: thats my biggest concern. good to know |
| 21:17.45 | *** join/#asterisk RomanTorres (~root@200.106.49.195) |
| 21:17.59 | scoof | FuriousGeorge: modems aren' |
| 21:18.09 | scoof | FuriousGeorge: modems aren't ethernet nics |
| 21:18.14 | FuriousGeorge | i know this |
| 21:18.19 | scoof | FuriousGeorge: what you're describing is probably LAG |
| 21:18.28 | FuriousGeorge | i assumed you understood that the two modems were going into nics |
| 21:18.38 | FuriousGeorge | lag? |
| 21:19.20 | scoof | FuriousGeorge: 802.3ad Link Aggregation |
| 21:19.39 | scoof | FuriousGeorge: and since that's an ethernet-standard, your "modems" doesn't really belong in that design |
| 21:20.41 | *** join/#asterisk Sedorox (brandon@Neptune.client.wlmsprt.pa.sed6.net) |
| 21:21.07 | FuriousGeorge | i know what your saying about this. but i said "two modems going into my asterisk box" as an easy way for me to describe having "two dsl connections from going into eth0 and eth1 respectively..." etc |
| 21:21.38 | scoof | FuriousGeorge: Link Aggregation wouldn't be what you're looking for in that case |
| 21:22.25 | *** join/#asterisk toddf (~toddf@net-66-210-104-252.theshop.net) |
| 21:22.27 | FuriousGeorge | hmm, my goal (in theory) is two have two differnt broadband ISP's. if both are working, combine the bandwidth, one goes down, use only one |
| 21:22.30 | scoof | FuriousGeorge: regardless of the choice of operating system |
| 21:22.46 | *** join/#asterisk evolutionxtinct (~kvirc@198.107.22.13) |
| 21:22.53 | Blissex | FuriousGeorge: that will cost you serious money... |
| 21:23.01 | scoof | Blissex: not really |
| 21:23.06 | FuriousGeorge | around here, a dsl connection is 35 / mo |
| 21:23.22 | Blissex | scoof: FuriousGeorge: go ahead, make my day ;-) |
| 21:23.27 | scoof | FuriousGeorge: a DSL connection to what? The internet or a layer2 point-to-point circuit? |
| 21:23.28 | FuriousGeorge | 300 KB/s down and 80 KB/s up |
| 21:23.53 | FuriousGeorge | scoof: the former since i dont know what the latter is |
| 21:24.04 | AgiNamu | around here, a real ADSL connection is $230 a month. 512K |
| 21:24.09 | *** join/#asterisk SagoDan (~dprotich@nat-pool02.sagonet.com) |
| 21:24.09 | FuriousGeorge | maybe its the latter since you said "point to point" and it uses PPPoE |
| 21:24.18 | scoof | AgiNamu: let's not take that detour again ;) |
| 21:24.25 | FuriousGeorge | AgiNamu: where? |
| 21:24.28 | AgiNamu | guatemala |
| 21:24.33 | FuriousGeorge | hay dios mio |
| 21:24.45 | RomanTorres | FuriousGeorge: to link two different default routes at the same time and do traffic balancing you need to patch the linux kernel with this: http://www.ssi.bg/~ja/ |
| 21:24.48 | scoof | AgiNamu: working for a CLEC, that's a creature that needs no further flogging ;) |
| 21:25.03 | Blissex | RomanTorres: the problem is doing that on ADSL is impossible... |
| 21:25.27 | Blissex | RomanTorres: unless both ADSL ISPs cooperate, and then they charge serious money. |
| 21:25.28 | tzanger | RomanTorres: both sides need to support that |
| 21:25.32 | RomanTorres | Blissex, I have a system with 4 ppoe links over DSL at the same time... |
| 21:25.58 | FuriousGeorge | blissex: really, impossible? forget adsl, what about two residential cable connections |
| 21:26.03 | tzanger | RomanTorres: back in the day I used four 33k6 modems for an uplink for an ISP :-) |
| 21:26.03 | Blissex | RomanTorres: from the same ISP yes.... |
| 21:26.07 | Blissex | FuriousGeorge: same. |
| 21:26.10 | scoof | Blissex: if you could survive with a simple failover and don't need hot standby, that can be done by simple post-routing NAT |
| 21:26.11 | tzanger | four 33k6 modems and a satellite feed to be exact |
| 21:26.22 | tzanger | any lowlatency traffic went through the modems, everything else over satellite |
| 21:26.37 | *** join/#asterisk mbaron (~mbaron@AVelizy-154-1-42-83.w82-124.abo.wanadoo.fr) |
| 21:26.49 | *** part/#asterisk mbaron (~mbaron@AVelizy-154-1-42-83.w82-124.abo.wanadoo.fr) |
| 21:26.52 | Blissex | scoof: think carefully about what FuriousGeorge said: «both are working, combine the bandwidth» |
| 21:27.19 | FuriousGeorge | ...if not use only one... |
| 21:27.26 | scoof | Blissex: you would never loadbalance pr packet anyway, that would introduce huge amounts of jitter |
| 21:27.37 | Blissex | scoof: the problem is that most probably his two ADSL endpoints are in different and not portable IP subranges... |
| 21:27.49 | RomanTorres | Blissex: One is a Satellite link, two DSL with the same provider, another with other provider. The kernel patches makes the linux system do the balancing without any cooperation from the ISP. |
| 21:27.56 | scoof | Blissex: and that's where post-routing NAT comes in to play |
| 21:28.03 | Blissex | RomanTorres: that load balancing is impossible. |
| 21:28.06 | FuriousGeorge | holy crap my head is spinning. the jist im getting is that ISPs dont support what i want to do as far sas the "combining the bandwidth" part |
| 21:28.33 | Blissex | RomanTorres: unless the ISPs give you a portable globally routed set of addresses, for which they charge serious money. |
| 21:28.33 | scoof | Blissex: that's not true; but it's not easy to accomplish in a "clean" way |
| 21:28.42 | scoof | Blissex: again untrue |
| 21:28.53 | Blissex | scoof: as I said, make my day :-) |
| 21:28.54 | FuriousGeorge | im going to "fstab" myself |
| 21:29.12 | RomanTorres | Blissex: check this url: http://www.ssi.bg/~ja/nano.txt |
| 21:29.31 | scoof | Blissex: you're judging way too fast |
| 21:29.40 | Blissex | RomanTorres: its pointless, it is just impossible. Think of what happens to return packets... |
| 21:30.14 | Blissex | scoof: it is one of those things that just cannot work, again unless you pay one or both ISPs to do it. And they charge serious monye for it. |
| 21:30.18 | tzanger | Blissex: well you kind of can, that's why :-) |
| 21:30.25 | SagoDan | I'm having some issues with DID or inbound # from a IAX provider do i have to create an extension in order for it to work ? |
| 21:30.39 | FuriousGeorge | sort of like how ISP's dont really honer QoS from users? |
| 21:30.48 | hardwire | Blissex: you can sort of do that |
| 21:30.52 | Blissex | FuriousGeorge: no, it is really a baseline routing issue. |
| 21:31.04 | SagoDan | if i have the extension created it calls in but then goes to voicemail because that extension isn't live; however if i do not have the extension it'll goto a fast busy signal |
| 21:31.15 | nestAr | gah |
| 21:31.27 | scoof | Blissex: I'd use two GRE tunnels to have some link-failure detection, and route my own set of RFC1918-addresses on those links. I'd do NAT of the two GRE tunnels so that the return traffic would be routed properly the other way, and OSPF and ECMP on the links would balance the traffic. |
| 21:31.32 | FuriousGeorge | what are we saying in practice? that if we send data that is contigious through two disparate connections we cannot guarentee that it will arrive in the correct order w/o cooperation from the ISP |
| 21:31.35 | nestAr | why can't polycom just give me an option to disable call waiting!?!?! |
| 21:31.38 | *** join/#asterisk pluto70 (~me@62.72.83.12) |
| 21:31.46 | nel | is there any program or way to calculate the echo cancellation? |
| 21:31.48 | *** join/#asterisk lesouvage (~lesouvage@cc341200-a.assen1.dr.home.nl) |
| 21:31.57 | tzanger | nel: "calculate the echo cancellation" ? |
| 21:32.05 | hardwire | nestAr: isn't their a Dial parameter that deals w/ that? |
| 21:32.15 | scoof | FuriousGeorge: you should NEVER load-balance per packet over these, for that exact reason |
| 21:32.21 | RomanTorres | Blissex: Just chech that url, http://www.ssi.bg/~ja/nano.txt, it shows you a clean way to have up to 8 different providers with load balancing. |
| 21:32.22 | Blissex | FuriousGeorge: if you ahve two ADSL lines, one has address IP1 and the other address IP2. IP1 belongs to the address allocation of ISP1, and IP2 to the address allocation of ISP2. Unless you pay serious money, packets for or from IP1 will never travel via ISP2, and viceversa. |
| 21:32.28 | nel | tzanger: I'm using a digium fxo card to connect asterisk to an analog phone line on an old centrex and I'm hearing myself when I speak |
| 21:32.39 | Blissex | scoof: but having OSPF support costs serious money. |
| 21:32.40 | *** part/#asterisk srineer (~srineer@209.50.133.4) |
| 21:32.43 | tzanger | nel: what country are you from |
| 21:32.46 | nel | people hear me with no eacho |
| 21:32.49 | scoof | FuriousGeorge: once a session is established on one you'd want to keep it there to not introduce jitter |
| 21:32.49 | nel | argentina |
| 21:32.51 | nestAr | hardwire: SetGroup and CheckGroup works.. but works too well. |
| 21:32.58 | FuriousGeorge | i would |
| 21:32.59 | scoof | Blissex: no, OSPF is free with Zebra |
| 21:32.59 | nel | I'm in US though |
| 21:32.59 | hardwire | too well :) |
| 21:33.13 | tzanger | nel: is your nation's telephone line impedance the same as north america? (600 ohms I think) ? |
| 21:33.27 | nestAr | hardwire: because the SetGroup is set to the Zap channel, when I transfer a call to someone else in the building, i don't get anymore calls until that person hangs up. |
| 21:33.27 | hardwire | Blissex: I typically recommend colocating at the DSL pop if you are doing that |
| 21:33.33 | hardwire | if those are dsl lines from the same pop |
| 21:33.56 | scoof | hardwire: that's not always an option |
| 21:34.01 | Blissex | scoof: OSPF costs money because the _ISP_ has to let you do it. |
| 21:34.28 | Blissex | scoof: ISPs charge money for globally routable, portable IP address ranges... |
| 21:34.30 | nel | tzanger: I'm in US , and I'm using a centrex line from verizon |
| 21:34.36 | scoof | Blissex: not to run OSPF on your own GRE tunnels |
| 21:34.37 | tzanger | nel: ok |
| 21:34.39 | hardwire | scoof: newp |
| 21:34.49 | hardwire | heya.. so whats the issue? |
| 21:34.56 | Blissex | scoof: but the problem is, what is the address of the endpoint of the GRE tunnel... |
| 21:34.58 | hardwire | you wanna split rtp streams against two uplinks? |
| 21:35.01 | scoof | Blissex: I know how globally routed unicast address space works |
| 21:35.05 | tzanger | nel: basically make sure your gains are set to 0.0 and start out with echocancel=64 |
| 21:35.12 | scoof | Blissex: that's the single public address your ISP assigns to you |
| 21:35.19 | tzanger | you need to stop and start (not just reload) asterisk with every change |
| 21:35.22 | tzanger | it takes some tuning |
| 21:35.23 | nel | tzanger: if 64 doesn't work, what is next? |
| 21:35.24 | nel | :P |
| 21:35.30 | tzanger | the wiki has info on how to do this nicely |
| 21:35.31 | Blissex | scoof: the single public address the ISP assigns you is not _portable_. |
| 21:35.37 | tzanger | I don't think the fxotune utility works with the x100P |
| 21:35.43 | tzanger | nel: try 128, then 32 |
| 21:35.57 | *** join/#asterisk NewSole (david@i216-58-44-245.avalonworks.net) |
| 21:35.58 | scoof | Blissex: it doesn't have to be, you run a single GRE instance for each of your DSL-connections |
| 21:36.15 | nel | how do I update changes, using ztvcfg -v ? |
| 21:36.16 | Blissex | scoof: and is surely not globally routable -- it is only routes by that ISP. Again, it costs money to have an ISP advertise a portable address space. |
| 21:36.16 | reallost1 | When an agi script finishes, what would keep it from returning control to asterisk dial plan? |
| 21:36.25 | tzanger | nel: no that is only for /etc/zapata.conf changes |
| 21:36.31 | tzanger | stop and start asterisk (not just reload) |
| 21:36.38 | Blissex | scoof: you are talking tech mumbo jumbo... Whether GRE is involved or not the problem is a simpl _routing_ issue. |
| 21:36.56 | tzanger | scoof: please step away from the keybaord and listen to Blissex, he is not bullshitting you |
| 21:37.09 | scoof | Blissex: you *don't* use global address space, you run your *own* private address space over a virtual infrastructure |
| 21:37.10 | tzanger | if you want true multirouting you need expensive multihomed IP space. No amount of software magic is going to get around that |
| 21:37.31 | *** join/#asterisk Wazb (Wazb@207.245.215.111) |
| 21:37.35 | Wazb | hi all |
| 21:37.44 | tzanger | scoof: and how do you propose to have this virtual space on two separate ISPs? |
| 21:37.48 | Blissex | scoof: all the routes for a residential IP address are advertised by a single ISP. This means that no other ISP will route them, unless you pay them, and that cost serious money. |
| 21:37.49 | scoof | tzanger: if you want true multihoming to the internet as a whole, yes, not to multihome to another part of your own network over the internet as an infrastructure |
| 21:37.57 | Wazb | how can i register H323 phone with asterisk? |
| 21:38.03 | tzanger | scoof: only if you have equipment at the common point of both networks |
| 21:38.10 | tzanger | scoof: and if that's the case you may as well put your * box there |
| 21:38.21 | RomanTorres | Tzanger: I have a Linux box with 4 links, without paying anything extra to the ISPs. Everything you need is to have a patched linux kernel with the nano extensions. |
| 21:38.25 | scoof | tzanger: and that's what he does, he just doesn't have the phone lines there |
| 21:38.26 | tzanger | scoof: and save yourself all the fucking hassle of pissing about with "virtual ip space" to begin with |
| 21:38.38 | tzanger | RomanTorres: and I will bet you dollars to donuts that you are not truly load balancing |
| 21:38.47 | tzanger | RomanTorres: you're load balancing your OUTGOING traffic, not your incoming traffic |
| 21:38.58 | tzanger | the incoming traffic follows whatever path the outgoing request went through |
| 21:39.01 | scoof | Blissex: I know, I've built such networks |
| 21:39.12 | tzanger | scoof: ok so you know what you're doing, go do it |
| 21:39.41 | RomanTorres | Tzanger: I do balance both, obvously, since any request that by chance gets on a line, is answered on the same line |
| 21:39.49 | tzanger | RomanTorres: exactly |
| 21:39.53 | tzanger | it's not true load balancing |
| 21:40.13 | tzanger | if you make a request that asks for a metric buttload of data in response, that entire metric buttload is coming in over that ONE link, not spread over the 4 |
| 21:40.24 | tzanger | (imperial buttloads might be different, I'm in Canada and don't know <g>) |
| 21:40.32 | FuriousGeorge | ok, waht about something mroe simple. two isp's. use one for and one for surfing and everything else. if isp goes down how can i tell linux to switch automagically? what do i need to go look up? |
| 21:40.47 | Blissex | FuriousGeorge: that's not difficult. |
| 21:40.52 | tzanger | FuriousGeorge: some ping tests and some astdb stuff. piece of cake |
| 21:40.57 | scoof | tzanger: you wouldn't want to balance it per packet anyway, VoIP doesn't like the jitter |
| 21:41.01 | FuriousGeorge | dont know what i did there. should read "use one for * and one for everything else" |
| 21:41.08 | RomanTorres | tzanger: well we can argue about that, but since for example I can tell the linux box to send for each 16 requests, to divide 4 requests on each line, well I have a pretty decent balancing. |
| 21:41.10 | tzanger | scoof: correct |
| 21:41.13 | nel | tzanger: I also compiled zaptel drivers with the aggresive echo suppresion using mark2 |
| 21:41.15 | sivana | bgp4 |
| 21:41.17 | nel | that didn't help |
| 21:41.17 | Blissex | FuriousGeorge: all you need to do, if you want to do it manually, is just bring down the interface that failed. |
| 21:41.23 | tzanger | RomanTorres: and you're doing that? for all protocols? bullshit. |
| 21:41.29 | tzanger | nel: NO |
| 21:41.33 | tzanger | nel: turn off agressive |
| 21:41.39 | tzanger | it's worse than regular old MARK2 |
| 21:41.45 | FuriousGeorge | blissex: no one is ever in this office, i would not want to have to do it manually |
| 21:41.51 | Blissex | RomanTorres: yes, that's pretty decent _connection_ balancing, but it does not have not have much failover. |
| 21:41.54 | FuriousGeorge | shoot if not i could switch the cat5 |
| 21:41.58 | scoof | sivana: you seem to be mimicking Yakov ;) |
| 21:41.59 | tzanger | compile with MMX support (zconfig.h), and compile with CFLAGS+=-march=yourprocessorhere |
| 21:42.24 | Blissex | FuriousGeorge: then as tzanger said, a little script with a 'ping' might be all you need. |
| 21:42.50 | *** join/#asterisk outsidefactor (~blah@203-206-247-72.dyn.iinet.net.au) |
| 21:43.23 | FuriousGeorge | blissex: always wanted an excuse to learn me some scripting. it would be running in the BR, pinging every minute or so on the asterisk ISP, if it went down then switch right |
| 21:43.44 | Blissex | FuriousGeorge: yes, that would be pretty like it. |
| 21:44.18 | FuriousGeorge | last ?: whats involved in the switch part. how do i make isp1=isp2 or eth0 = eth1 as the case may be |
| 21:44.25 | sivana | hehe |
| 21:44.30 | sivana | bgp4 |
| 21:44.33 | Blissex | FuriousGeorge: its not right, but something like: while sleep 10; do if ! ping -q ...; ....; fi; done |
| 21:44.47 | Blissex | FuriousGeorge: you dont need to switch anything... |
| 21:44.56 | Blissex | FuriousGeorge: ecept the default route perhaps. |
| 21:45.10 | FuriousGeorge | i think im gonna cross this bridge after i build it |
| 21:45.15 | Blissex | FuriousGeorge: or if you use 'nexhtop' routing, disable that on one of the hops. |
| 21:45.31 | RomanTorres | Blissex: If you do the ip route thing with the nano extensions, and at the same time you use dns multipble addresses for each of your external addresses, you can have a very good sip server balancing . |
| 21:46.16 | Blissex | RomanTorres: it will be a good per connection balancing -- splitting N connections across M lines. |
| 21:46.59 | tzanger | Blissex: agreed, but that's not data load balancing |
| 21:47.05 | *** join/#asterisk pr0m (~pr0metheu@ip-wv-68-187-250-031.charterwv.net) |
| 21:47.11 | tzanger | which is what I understood the asker really wanted |
| 21:47.13 | RomanTorres | Blissex: Exactly. If you want to have the same connection on several links at the same time it would be a real problem. |
| 21:47.33 | Blissex | tzanger: or for that matter, any degree of transparent failover. |
| 21:47.35 | *** join/#asterisk Sedorox (brandon@Neptune.client.wlmsprt.pa.sed6.net) |
| 21:48.07 | tzanger | Blissex: yes... it'll failover just fine but if a link that you were using falls down you'll have to reconnect |
| 21:48.43 | Blissex | tzanger: that's not very ''transparent'' as we both agree... |
| 21:48.56 | *** join/#asterisk angler_ (~angler@suid.digium.com) |
| 21:49.04 | Blissex | tzanger: ...at least to the two people on that call :-) |
| 21:49.30 | sivana | I don't think it can be across two different subnects |
| 21:49.34 | sivana | subnets |
| 21:50.17 | jhiver | You can do data load balancing with OpenVPN + ethernet bridging + routing |
| 21:50.19 | Blissex | sivana: that's why one needs a _portable_ address range... |
| 21:50.29 | sivana | Blissex: correct, and bgp4 |
| 21:50.51 | jhiver | create an tap (ethernet) OpenVPN tunnel for each link |
| 21:50.55 | Blissex | sivana: more precisely, BGP4 support from the ISPs involved, and that costs money... |
| 21:50.55 | scoof | jhiver: incidentally, a variation of what I just said earlier :) |
| 21:51.03 | jhiver | then bridge all the tunnels together |
| 21:51.13 | jhiver | then route each port through a separate connection |
| 21:51.23 | jhiver | the links must be same capacity |
| 21:51.40 | jhiver | and you need to do the routing on both ends of the link, so you need static ip adresses |
| 21:51.42 | sivana | I'm talking if you want complete transparency without interruption |
| 21:51.43 | tzanger | jhiver: yes, if you have equipment at the common point of where the multiple links come together... but you may as well get multihomed IP space, I bet it'd be cheaper and it would certainly be more reliable |
| 21:51.57 | tzanger | jhiver: and less laggy (no encryption or x86 routing decisions) |
| 21:52.02 | jhiver | it depends what you want to do |
| 21:52.11 | scoof | sivana: BGP global convergence time is too high to get that transparency anyway |
| 21:52.25 | tzanger | scoof: we never said that wouldn't work, just that it's not as feasable as just getting multihomed IP space and paying for it |
| 21:52.45 | scoof | tzanger: I beg to differ |
| 21:52.49 | SagoDan | anyone work with DID #'s ?? |
| 21:53.02 | tzanger | scoof: beg all you want. Set up both and benchmark it. |
| 21:53.06 | jhiver | tzanger: sometimes you don't have that option |
| 21:53.18 | scoof | 23:30 < Blissex> scoof: it is one of those things that just cannot work, again unless you pay one or both ISPs to do it. And they charge serious monye for it. |
| 21:53.59 | *** join/#asterisk dmccollum (~dmccollum@69-164-245-72.atlaga.adelphia.net) |
| 21:53.59 | tzanger | your method = colocation + (usually) consumer grade equipment. standard method = no colocation, just expensive IP space + utilizing the provider's already tested and monitored routers |
| 21:54.00 | Blissex | scoof: and that's exactly true, still. |
| 21:54.20 | jhiver | over here it's 150EUR for a 1024/256 DSL but 1500 for a 512k symetrical (monthly) |
| 21:54.30 | Blissex | tzanger: but colocation does use expensive IP space... |
| 21:54.32 | FuriousGeorge | tzanger: why are the two identical isp's as described above not as feasable as a multihomed ip? |
| 21:54.44 | jhiver | so 3 DSL + OpenVPN + bonding + multiling = 3 times cheaper |
| 21:54.49 | tzanger | Blissex: ok ok technically speaking sure but you're getting more than just ip space for the price :-) |
| 21:54.53 | jhiver | multilink I meant |
| 21:54.57 | tzanger | jhiver: uh |
| 21:54.57 | scoof | tzanger: no colocation in my method, no. |
| 21:55.07 | Blissex | tzanger: the problem statement involves just residential ISP lines... |
| 21:55.11 | QuickDry | SagoDan: Generic DID question, or Asterisk Specific? |
| 21:55.15 | tzanger | jhiver: you're forgetting that you need to colocate at the point where all three ISPs bandwidth comes together |
| 21:55.32 | tzanger | scoof: is this point to point or point to multipoint? |
| 21:55.35 | jhiver | that's like 50EUR / month for dedicated server hosting |
| 21:55.42 | scoof | tzanger: point to point |
| 21:55.46 | FuriousGeorge | blissex: if i may (having made the problem statement) its only a bit more to get a static ip |
| 21:55.52 | scoof | tzanger: with redundancy at one end |
| 21:56.00 | tzanger | jhiver: dedicated server hosting at a datacenter that all three ISP's bandwidth goes through. |
| 21:56.07 | tzanger | scoof: point to point then you can do it, sure |
| 21:56.16 | tzanger | you have N connections at each end (price: 2*N) |
| 21:56.20 | jhiver | tzanger: why? |
| 21:56.24 | Blissex | FuriousGeorge: its not just static IP, the static IP must be in so-called portable (multi-homable) IP address space. |
| 21:56.35 | jhiver | there's enough bandwith on my server, doesn't seem to be a pb |
| 21:56.44 | tzanger | jhiver: how do you expect to load balance the data (I'm not talking connection load balancing, yo udon't need bridging or openvpn for that) |
| 21:56.49 | jhiver | I've tried this solution with 2 connections from the same provider |
| 21:56.58 | jhiver | I told you |
| 21:57.04 | tzanger | oh |
| 21:57.08 | tzanger | same provider, my mistake |
| 21:57.10 | jhiver | 1 - you make 3 virtual ethernet using OpenVPN |
| 21:57.15 | FuriousGeorge | so then what does windows do when you "bridge connection" just the failover part? |
| 21:57.19 | *** join/#asterisk vidia22 ([U2FsdGVkX@vidiamob4.vidiacom.com) |
| 21:57.20 | Hmmhesays | openvpn is nice |
| 21:57.20 | jhiver | 2 - you route them through your different ISP routes |
| 21:57.21 | Hmmhesays | very nice |
| 21:57.23 | Blissex | FuriousGeorge: that is expensive because it is ''business grade''. |
| 21:57.33 | tzanger | I prefer openswan myself |
| 21:57.33 | jhiver | 3 - you bond them back onto one ethernet |
| 21:57.36 | scoof | FuriousGeorge: nope, that's a protocol between windows and the switch it's connected to |
| 21:57.42 | FuriousGeorge | blissex: round here its 70/mo for business dsl w/ static ip |
| 21:57.47 | FuriousGeorge | not sure about the mukltihomed part |
| 21:58.01 | FuriousGeorge | and the whole routing end of it seems more complicated then *. i know it uses PPP |
| 21:58.16 | jhiver | It's better to have the same provider... less jitter = less increased latency |
| 21:58.23 | vidia22 | can anyone give me some advice on how to get a x100p to work? |
| 21:58.25 | Blissex | FuriousGeorge: the multihomed/portable bit is expensive... Not erribly, just a lot more than ADSL.... |
| 21:58.32 | vidia22 | any help would be appreciated:) |
| 21:58.36 | jhiver | and yes i just *love* openvpn |
| 21:58.49 | tzanger | jhiver: except when the provider has connectivity issues :-) |
| 21:58.53 | Blissex | FuriousGeorge: if you have ''portable'' IP addresses the routing is very very simple; it is all automagic. |
| 21:59.00 | nestAr | vidia22: have you checked the wiki? zaptel drivers is pretty much all you need. |
| 21:59.11 | jhiver | well when your bandwith is bust there's not much you can do at any rate :( |
| 21:59.26 | nel | tzanger: I have tried with the different values, I don't see any change:( any other idea? |
| 21:59.55 | FuriousGeorge | blissex: and that is for the load balancing? so if i called my isp and they said our "static IPs are portable", then (if it werent too complicated) i could do the load balancing mentioned aboce |
| 22:00.00 | tzanger | nel: how much time do you have... there's a lot of things but they involve equipment and time |
| 22:00.00 | FuriousGeorge | *above |
| 22:00.16 | FuriousGeorge | (and thats a big if |
| 22:00.26 | jhiver | now some cool providers do DSL bonding... that's cool |
| 22:00.40 | jhiver | order X DSLs, have X times the bandwith |
| 22:00.45 | vidia22 | nestAr - I have and have zapel running... |
| 22:00.47 | jhiver | too bad they don't do it where I live :( |
| 22:00.55 | Blissex | FuriousGeorge: not only their ADSL static IPs will not be portable, but even if they were you need to pay the other ISP too to route them... Then things become easy. |
| 22:00.56 | FuriousGeorge | jhiver who does it where? |
| 22:01.08 | jhiver | Nildram does it in the UK |
| 22:01.20 | nestAr | vidia22: you edit your /etc/zaptel.conf and /etc/asterisk/zapata.conf ? |
| 22:01.28 | Blissex | jhiver: and Easynet too IIRC, and possibly also Clara... |
| 22:01.33 | FuriousGeorge | blissex: i get it. if i had the multihomed watchmacallit then its automagic, but expensive |
| 22:01.45 | vidia22 | nastat - yes - I have a very basic config that I think is good |
| 22:01.55 | jhiver | So I have to stick with the 'poor's man bonding' I devised :) |
| 22:02.00 | vidia22 | nastar - I am wondering if I have bad hardware...??? |
| 22:02.02 | NewSole | Question is there a doc to setup ZapRAS |
| 22:02.04 | FuriousGeorge | jhiver: doubt they have relay stations going "accross the pond" |
| 22:02.04 | Blissex | FuriousGeorge: yes... Not _terribly_ expensive, but still fairly expensive. Like perhaps 10 times more than ADSL alone. |
| 22:02.12 | nestAr | vidia22: does the module load? |
| 22:02.19 | vidia22 | zaptel does yes |
| 22:02.24 | nel | tzanger: I have time |
| 22:02.33 | nestAr | vidia22: IE: modprobe wcfxo |
| 22:02.43 | nestAr | then run ztcfg |
| 22:02.47 | tzanger | nel: then go to voip-info.org and read up... there is a lot of info on there on getting rid of the cho |
| 22:02.47 | nestAr | then run asterisk |
| 22:03.07 | vidia22 | ok - I have run modprobe but not ztcfg... |
| 22:03.09 | Hmmhesays | cho ? margaret cho? |
| 22:03.16 | vidia22 | I should kill asterisk first? |
| 22:03.20 | Hmmhesays | ba dum ching! |
| 22:03.23 | nestAr | i would |
| 22:03.33 | nel | thanks I have read some there |
| 22:03.36 | vidia22 | ok - Ill try - THANKS:) |
| 22:04.02 | nestAr | check the wiki |
| 22:04.03 | nestAr | http://www.voip-info.org/wiki-Asterisk+Hardware |
| 22:04.15 | nestAr | it's been a while since i've messed with a X100P |
| 22:04.29 | nestAr | i have one at home, but i don't use it because caller id doesn't seem to work |
| 22:05.49 | *** join/#asterisk implicit (~implicit@ip68-7-149-247.sd.sd.cox.net) |
| 22:06.04 | JerJer[mobile] | someone remind me...do channel banks have a male or female amphinal connector on them? |
| 22:06.18 | Corydon-w | It depends |
| 22:06.24 | JerJer[mobile] | ta 750 |
| 22:06.31 | Corydon-w | male |
| 22:06.34 | JerJer[mobile] | ok |
| 22:06.53 | nestAr | i've figured out how to fix this SetGroup problem... |
| 22:07.03 | nestAr | buy phones that you can disable callwaiting on |
| 22:07.15 | nestAr | anyone got a suggestion? |
| 22:07.50 | *** join/#asterisk anthm (~anthm@000-435-904.area4.spcsdns.net) |
| 22:07.50 | *** mode/#asterisk [+o anthm] by ChanServ |
| 22:08.24 | JerJer[mobile] | thanks |
| 22:10.24 | tzanger | JerJer[mobile]: I've always seen female |
| 22:10.27 | RomanTorres | Has anyone here messed up with Unicall (E1 MFC/R2)? I have a problem with the dtmf tones not working. |
| 22:10.32 | tzanger | Adit600, AB1/2 |
| 22:12.54 | vidia22 | nestAR: same results... nothing... |
| 22:13.18 | vidia22 | nestAR: is there a way to see results from modprobe? or any way to see output form card? |
| 22:14.49 | nestAr | dmesg |
| 22:15.06 | vidia22 | nestAR: I am running kudzu - should I let it try to config the card? |
| 22:15.06 | nestAr | or grep wcfxo /var/log/messages |
| 22:15.14 | nestAr | no idea man |
| 22:15.30 | vidia22 | thanks again:) |
| 22:15.41 | nestAr | sorry |
| 22:15.42 | nestAr | :) |
| 22:15.56 | nestAr | you can also try doing a ztcfg -v |
| 22:16.01 | nestAr | for verbose output from that part |
| 22:16.26 | Hmmhesays | heh, ser can be a pain |
| 22:18.17 | vidia22 | nestAr - my grep of the log shows that wcfxo is found (DAA mode is FCC) |
| 22:18.30 | nestAr | well, that's promising |
| 22:18.40 | vidia22 | running ztconfig however with the verbose on shows 0 cards configured... |
| 22:19.15 | vidia22 | ???? WTF :) |
| 22:19.56 | nestAr | my /etc/zaptel.conf is very simple |
| 22:19.56 | nestAr | loadzone=us |
| 22:19.57 | nestAr | defaultzone=us |
| 22:19.57 | nestAr | fxsks=1 |
| 22:20.34 | *** join/#asterisk bprice20 (~brandon@Dynamic-216.120.224.151.hrnoc.net) |
| 22:20.34 | vidia22 | yes - I have that ... exactly... |
| 22:20.52 | bprice20 | has realtime been rolled into asterisk stable yet |
| 22:20.56 | nestAr | ztcfg -vvv |
| 22:21.05 | tzanger | I hope never |
| 22:21.23 | bprice20 | I'm using cvs from a month or 2 ago but I want to move to stable for production |
| 22:21.35 | vidia22 | same amount of verbose.... still says 0 channels... |
| 22:21.47 | bprice20 | ftp.digium.com is down or i'd take a look myself |
| 22:21.57 | vidia22 | bad card?? bad pci slot??? |
| 22:22.00 | harryvv | channel =1 |
| 22:22.28 | nestAr | i get... |
| 22:22.28 | nestAr | Channel map: |
| 22:22.29 | nestAr | Channel 01: FXS Kewlstart (Default) (Slaves: 01) |
| 22:22.29 | nestAr | 1 channels configured. |
| 22:23.35 | bprice20 | took a look at the wiki apparently not |
| 22:24.02 | anthm | stable is for horses ! |
| 22:24.07 | SagoDan | QuickDry: its kinda generic question i guess specific to asterisk though... I'm trying to get the DID to work however i wasn't able to with out an extension all i would get would be a busy signal. |
| 22:24.15 | *** join/#asterisk fugitivo (~ajf@201.255.106.152) |
| 22:24.19 | fugitivo | hi |
| 22:26.29 | *** join/#asterisk Los415 (~los415@c-24-126-63-233.hsd1.ca.comcast.net) |
| 22:27.05 | bannerman | The bunny, the bunny, oh I love the bunny. I don't love my mom, or my dad, just the bunny... |
| 22:27.22 | marlowe | bunny's are cool |
| 22:27.28 | Wonka | and tasty |
| 22:27.36 | bannerman | very tasty when made of chocolate |
| 22:30.39 | bannerman | It's so odd. If I add members to my queue by their agent number (agent/101) they aren't able to transfer inbound calls. If I add members directly (SIP/102) they are |
| 22:30.55 | bannerman | it truncates the last digit of the extension that they try to dial |
| 22:30.58 | *** join/#asterisk madounet (~mad|net@juvenal-3-82-226-155-19.fbx.proxad.net) |
| 22:31.16 | *** join/#asterisk facek_ (faceoff@devel.acdbddh.eu.org) |
| 22:31.20 | facek_ | czehello |
| 22:32.07 | *** join/#asterisk cjk (~cjk@80.92.75.232) |
| 22:32.30 | cjk | hi, anyone here who knows how to fix a broken firmware upgraded ATCOM phone |
| 22:32.45 | bannerman | I suggest a large hammer. |
| 22:33.09 | cjk | bannerman, i tried but still is not working |
| 22:33.40 | bannerman | wish I could help, I don't know anything about atcom phones :-/ |
| 22:34.12 | bannerman | would it help if I made more bad jokes? |
| 22:34.53 | cjk | yeah maybe |
| 22:34.59 | cftbl | i think it would |
| 22:35.01 | cjk | but does not matter i will call them |
| 22:35.08 | *** join/#asterisk facek_ (faceoff@devel.acdbddh.eu.org) |
| 22:35.33 | cjk | but their phones are great. iax support is working and ilbc support will be released soon |
| 22:35.37 | cjk | and they are damn cheap |
| 22:35.48 | bannerman | Ok. I'll think up some new ones in case you don't have any luck there. You never know, someone might come around that actually knows about phones too. |
| 22:35.51 | bannerman | I'll have to look into those |
| 22:36.14 | bannerman | these ariavoice phones are ok, but I have a feeling some of my random issues are coming from them just being cheap |
| 22:36.57 | *** kick/#asterisk [AgiNamu!~mark@kram.digium.sponsor.pdpc] by kram (kram) |
| 22:37.35 | *** join/#asterisk brimstone (me@146.229.188.198) |
| 22:37.52 | sivana | heh |
| 22:38.00 | sivana | bye bye |
| 22:38.05 | brimstone | does anyone have an example of a working "Action: Status" command via the manager API? |
| 22:38.39 | nestAr | brimstone: i may |
| 22:38.42 | nestAr | let me look |
| 22:38.48 | brimstone | awesome, thanks |
| 22:39.03 | Hmmhesays | brimstone |
| 22:39.05 | Hmmhesays | wiki |
| 22:39.06 | Hmmhesays | go there |
| 22:39.13 | brimstone | i didn't see one on the wiki |
| 22:39.49 | Hmmhesays | oh |
| 22:40.11 | nestAr | nah, i don't have one |
| 22:40.12 | nestAr | sorryt |
| 22:40.18 | brimstone | the problem is that i don't know what to put for the Channel: parameter |
| 22:40.32 | Hmmhesays | channel name |
| 22:40.34 | brimstone | Zap/2, Zap/2-1 and 2 don't work |
| 22:41.25 | *** part/#asterisk IQ (~iq@65-103-165-206.omah.qwest.net) |
| 22:41.41 | Hmmhesays | i only use the status on sip channels |
| 22:42.27 | brimstone | how would you name the zap channels then? |
| 22:42.34 | brimstone | or get the name of the zap channels? |
| 22:42.43 | brimstone | ZapShowChannels works, but lists all the channels |
| 22:42.48 | brimstone | and i just need to know about one |
| 22:42.52 | Hmmhesays | from the manager? |
| 22:42.56 | brimstone | yup |
| 22:43.05 | Hmmhesays | i don't know of any good way, maybe sort by callerid? |
| 22:43.13 | *** part/#asterisk Jerub (~gideon@jerub.user) |
| 22:43.32 | Hmmhesays | brimstone.... take a look at FOP server.pl |
| 22:43.33 | nestAr | eff work |
| 22:43.34 | brimstone | i'm tring to determine which FXOs are not busy with a call |
| 22:43.35 | Hmmhesays | it's written in perl |
| 22:43.36 | nestAr | i'm going hom. |
| 22:43.44 | brimstone | where is FOP server.pl ? |
| 22:43.54 | Hmmhesays | www.asternic.org i think |
| 22:44.24 | Hmmhesays | nicolas might be able to help you out |
| 22:44.34 | brimstone | ok, thanks |
| 22:44.39 | brimstone | i'll poke around some more with it |
| 22:44.54 | Hmmhesays | fop can monitor zap channels |
| 22:45.28 | *** join/#asterisk r0d3nt (nobody@wsip-24-234-241-84.lv.lv.cox.net) |
| 22:46.59 | *** join/#asterisk VirTERM (~VirTERM@204.225.113.90) |
| 22:47.20 | *** join/#asterisk TheSin (~TheSin@iphost-64-56-130-194.edm.wiband.net) |
| 22:48.01 | fugitivo | if I modify zapata.conf, what do I need to reload? |
| 22:48.05 | Wazb | hi again |
| 22:48.22 | TheSin | fugitivo, asterisk for sure |
| 22:48.25 | Wazb | i need to configure sip proxy server , please help me in setup |
| 22:48.38 | fugitivo | just a "reload" ? |
| 22:48.40 | Hmmhesays | you won't get help like that wazb |
| 22:48.50 | Hmmhesays | you'll get a whole lot of RTFM |
| 22:48.50 | *** join/#asterisk Lee__ (~lee@ool-44c26ebc.dyn.optonline.net) |
| 22:49.02 | facek_ | i am looking for .net component for build iax or sip softphone, abybody know sth about it? |
| 22:49.14 | TheSin | fugitivo, my exp with zapata is you need to restart it |
| 22:49.15 | Lee__ | Is caller ID determined on a per extension basis? |
| 22:49.19 | Wazb | then what i need to do |
| 22:49.32 | fugitivo | TheSin: thanks |
| 22:49.34 | Hmmhesays | read the wiki |
| 22:49.38 | Hmmhesays | ~docs |
| 22:49.39 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org |
| 22:49.47 | Hmmhesays | that is where you start |
| 22:49.56 | *** join/#asterisk IQ (~iq@65-103-165-206.omah.qwest.net) |
| 22:50.50 | TheSin | anyone ever see something like this? |
| 22:50.50 | TheSin | Extension '' in context 'incoming' from '!' does not exist |
| 22:50.50 | TheSin | PRI: received SETUP message for call that is not a new call, wicked!!! |
| 22:51.02 | TheSin | on a digium TE110P |
| 22:52.08 | Wazb | thanks |
| 22:52.36 | Hmmhesays | np |
| 22:53.03 | TheSin | with pri debug span 1 on |
| 22:53.39 | TheSin | I see the Dialed and Dialing number are set and right |
| 22:54.25 | TheSin | why would asterisk get Extension '' |
| 22:54.25 | ctooley | 2005-03-30 16:53:46 WARNING[1846]: chan_iax2.c:6007 socket_read: Received mini frame before first full voice frame |
| 22:54.26 | ctooley | That mean anything to anyone |
| 22:56.46 | QuickDry | Is there a manual anywhere that covers most of the popular features and reads more like a book? Unless I just suck at using the wiki, it seems like it jumps all over the place. Looking for something a little more concise. |
| 22:57.05 | VirTERM | anyone runs iax channels from RealTime? |
| 22:57.25 | emrah | I' having a new problem with the AstCC programme. When I call the AGI interface, it asks for an account number with 12 digits... What's that? My card numbers have 8 digits... |
| 22:57.35 | ctooley | VirTERM, I do |
| 22:57.52 | ctooley | QuickDry, asteriskdocs.org |
| 22:57.55 | VirTERM | can you share your table structure :)? |
| 22:58.14 | ctooley | kram: You see that error I posted I'm geting from chan_iax2? |
| 22:58.17 | QuickDry | ok I'll try there, I was finding myself getting a little overwhelmed with features.... |
| 22:58.31 | ctooley | QuickDry, there's a intro book there |
| 22:58.42 | QuickDry | it might just be my ADD kicking in though.... |
| 22:58.45 | VirTERM | I am having problem with codecs negotiations |
| 22:58.51 | QuickDry | thanks ctooley. |
| 22:59.01 | QuickDry | night all. |
| 22:59.19 | ctooley | VirTERM, don't know, we don't use codecs. we use _a_ codec |
| 22:59.37 | VirTERM | even with one ; same issue |
| 22:59.48 | ctooley | plural means decisions, decisions take time, no time for decisions |
| 23:00.01 | VirTERM | how did you name the fields in the table? is there anything special? |
| 23:00.17 | ctooley | there's an entry in the Wiki |
| 23:00.29 | VirTERM | yeah, but nothing there |
| 23:00.46 | VirTERM | I have voicemail running no problem..no got stack on iax |
| 23:01.17 | VirTERM | do you actually have to remove the iax.conf if you use realtime? |
| 23:02.31 | ctooley | VirTERM, nope |
| 23:02.36 | VirTERM | emrah: this is just a voice prompt asking for 12 digits, try your 8 digit number... |
| 23:02.41 | ctooley | it's just for users and peers |
| 23:02.51 | VirTERM | ok, this is what I've done |
| 23:03.16 | VirTERM | obviously show iax2 peers doesn't show anything |
| 23:03.34 | ctooley | VirTERM, BTW if you are using the latest CVS HEAD (which if you're using Realtime I highly recommend) it uses iaxpeers and iaxusers and iaxfriends is deprecated |
| 23:03.41 | ctooley | VirTERM, that is correct |
| 23:03.41 | VirTERM | but if I try to make a call there is a problem with codec negotiation even if both ends are configured to use only ulaw |
| 23:04.13 | VirTERM | ctooley: yes I know, but it's backwards compatible and just gives you a warning |
| 23:04.38 | VirTERM | I am pointing to iaxpeers and iaxusers (one table) |
| 23:04.41 | ctooley | it is for now. |
| 23:05.02 | fugitivo | anyone using sphinx? |
| 23:06.06 | VirTERM | ctooley: so basically you just have [general] statement in your iax.conf? |
| 23:06.17 | ctooley | VirTERM, yep |
| 23:06.30 | ctooley | VirTERM, ours are actually defined as friends though. |
| 23:06.31 | VirTERM | would you mind doing mysqlshow on your iax tables? |
| 23:06.49 | VirTERM | same here.. |
| 23:06.57 | Dovid | i am trying to install zaptel, it is telling me that i need to install all of the kernal source files, which i did. i am still getting an error that i dont have all the kernal source fiels. any suggestions ? |
| 23:07.52 | VirTERM | when you specify codec, do you say "ulaw" or 4? |
| 23:08.00 | fugitivo | Dovid: ln -s /usr/src/linux-x.x.x /usr/src/linux |
| 23:08.26 | FuriousGeorge | i jsut got incomming calls working the other day. there is one particular line in extensions.conf where i define the rules for outgoing calls. i want to give people in this context the ability to dial any number. i got it working, then i changed the exten=>(number) to exten=>_. (to get any amount of digits, i thought) |
| 23:08.34 | FuriousGeorge | well that didnt work and now i cant get it working again |
| 23:08.41 | FuriousGeorge | whats wrong with this extension exten => _1NXXNXXXXXX,dial(${OUTGOING}/${EXTEN},30,r) |
| 23:09.00 | FuriousGeorge | *i mean to say LOUTGOING CALLS |
| 23:09.15 | fugitivo | FuriousGeorge: try _1.,dial |
| 23:09.59 | VirTERM | ..and don't forget about the priority... |
| 23:10.24 | FuriousGeorge | fugitivo: nope client says "404 not found" |
| 23:10.28 | reallost1 | grrr... DTMF collection in agi |
| 23:11.06 | VirTERM | exten => _1NXXNXXXXXX,1,dial(${OUTGOING}/${EXTEN},30,r) |
| 23:11.16 | FuriousGeorge | d'oh |
| 23:11.25 | fugitivo | right, the priority : |
| 23:11.25 | fugitivo | ) |
| 23:11.27 | *** join/#asterisk riquisim0 (~riquisimo@63.245.8.94) |
| 23:11.51 | riquisim0 | hi |
| 23:12.10 | VirTERM | ctooley: I am really confused with this IAX from realtime... |
| 23:12.12 | fugitivo | FuriousGeorge: you can replace the NX with . if you want any amount of digits |
| 23:13.57 | ctooley | I was too. |
| 23:14.02 | VirTERM | heh |
| 23:14.36 | ctooley | VirTERM, I'll do a show create table for iax and post it, gimme a minute |
| 23:14.44 | VirTERM | super |
| 23:16.04 | FuriousGeorge | fugitivo, thats exactly what i was trying to do, but i deleted the priority too w/o realizing, and i was starting to get so pissed off. i thought * was defying logic to spite me |
| 23:17.51 | lesouvage | I have this "exten => 8,1,Dial(SIP/202)" as a menu option and the cli "Executing Dial("Zap/1-1", "SIP/202") in new stack" It's a line out of the manual. What is going wrong? |
| 23:18.11 | lesouvage | The menu itself works fine, just this line fails. |
| 23:18.58 | VirTERM | well, is your 202 sip device ringing? |
| 23:19.26 | VirTERM | you didn't say what's the problem |
| 23:20.11 | InfraRed | 1.0.7 released? |
| 23:20.22 | InfraRed | on www.asterisk.org i only see 1.0.6 |
| 23:21.00 | lesouvage | It's the number for listening to the mail. The voicemenu doesn't show up (pleas enter mailbox number etc.) |
| 23:21.01 | mstocco | InfraRed: it is there |
| 23:21.23 | VirTERM | you are trying to call the SIP device.... |
| 23:21.32 | *** join/#asterisk pedxing (~batherton@h24-68-208-230.sbm.shawcable.net) |
| 23:21.43 | InfraRed | http://www.asterisk.org/index.php?menu=download |
| 23:21.51 | InfraRed | the tgz is 1.0.6 |
| 23:22.07 | mstocco | InfraRed: ftp://ftp.asterisk.org/pub/asterisk/ |
| 23:22.10 | VirTERM | exten => 8,1,VoicemailMain(s${CALLERIDNUM}) |
| 23:22.18 | VirTERM | try this |
| 23:22.20 | InfraRed | yab using the ftp ;) |
| 23:22.48 | *** join/#asterisk Luhiwu (~marsosa@200.63.89.240) |
| 23:23.12 | VirTERM | or just like this exten => 8,1,VoicemailMain |
| 23:23.16 | mstocco | InfraRed: point your browser there |
| 23:23.59 | VirTERM | you need to call Voicemail application not a device... |
| 23:24.21 | InfraRed | its ok i got it from the ftp |
| 23:24.28 | InfraRed | i emailed the webmaster about the out of date page |
| 23:24.48 | TheSin | anyone here ever get support from digium? Just wondering I thought their support was suppose to be great |
| 23:25.44 | loud | they can troubleshoot anything |
| 23:25.51 | mstocco | InfraRed: ahh ok I see what you are looking at |
| 23:25.51 | loud | plus they have a nice rma support. |
| 23:26.30 | bannerman | anyone have some sample iptables QOS stuff for prioritizing voice? |
| 23:26.45 | lesouvage | VirTerm: thanks it's working now. |
| 23:26.50 | VirTERM | :) |
| 23:26.58 | lesouvage | with your last suggestion |
| 23:27.43 | *** part/#asterisk Lee__ (~lee@ool-44c26ebc.dyn.optonline.net) |
| 23:27.45 | VirTERM | the first line you could use for people calling from inside (SIP) if you want to avoid need for authentication. It will use their cid to authenticate |
| 23:27.50 | TheSin | loud, does it normally take 3 days though and not one thing to try yet |
| 23:28.05 | dmccollum | iptables -t nat -I PREROUTING 1 -i eth2 -p udp -s <IP ADDRESS of Asterisk server> --dport 80 -j ACCEPT |
| 23:28.24 | dmccollum | iptables -t nat -I PREROUTING 1 -i eth2 -p udp -s <IP ADDRESS of Asterisk server> --dport 5060:5065 -j ACCEPT |
| 23:28.24 | *** join/#asterisk ivesti (ivesti@ppp-68-251-35-227.dsl.chcgil.ameritech.net) |
| 23:28.27 | VirTERM | port 80?!? |
| 23:28.37 | dmccollum | typed wrong port on first one. |
| 23:28.50 | ivesti | hello, does anyone know how to connect a sipura spa-3000 with a dock-n-talk device? |
| 23:28.52 | dmccollum | then do the same for ports 10000:20000 |
| 23:29.04 | VirTERM | this won't help you unless you run rtp traffic on these ports... |
| 23:29.13 | VirTERM | ok, I see now |
| 23:29.15 | loud | TheSin, have you called ? |
| 23:29.23 | loud | or just email |
| 23:29.35 | dmccollum | then do the same for ports 53 123 and 69 |
| 23:30.04 | VirTERM | it will however work very well with iAX (udp 4569) |
| 23:30.06 | *** join/#asterisk mw` (id1864@p5480C9F8.dip.t-dialin.net) |
| 23:30.18 | mw` | hi |
| 23:30.22 | VirTERM | but not in realtime :) |
| 23:30.47 | ivesti | hello, does anyone know how to connect a sipura spa-3000 with a dock-n-talk device? |
| 23:30.49 | TheSin | loud, ya but all they do is want ssh access |
| 23:30.49 | Luhiwu | anyone here uses Digium FXO cards? i'm having some problems, it doesn't recognize my dtmf tones |
| 23:30.53 | TheSin | don't even listen |
| 23:30.55 | mw` | anyone experience problems with chan_capi and kernel 2.6.11.x ? |
| 23:31.02 | riquisim0 | is there a way to make several Call Groups within 1 Asterisk PBX Server, where each call group has 1 telephone number assigned for use with PSTN, and also internal extension numbers for each call group, but that those call groups aren't authorized to call extension numbers that belong to other call groups? |
| 23:31.02 | TheSin | so then they tell me to email |
| 23:31.09 | VirTERM | relaxdtmf=yes |
| 23:31.12 | TheSin | and I've done that twice |
| 23:31.47 | Luhiwu | VirTERM, i've already tried that, with no success. If i plug the line to a Cisco's FXO port it works fine, but not in the zap card. |
| 23:32.27 | VirTERM | are you dialing from the cellphone? can you reproduce it everytime? |
| 23:33.18 | riquisim0 | like Centrex services |
| 23:33.32 | VirTERM | I am assuming you are talking about incoming calls |
| 23:33.41 | Luhiwu | i can reproduce everytime, and i'm using a fixed wireless terminal, i call to a cellphone, not from a cell |
| 23:33.42 | loud | TheSin, well if you can't figure it out, give them access, you can sudo them .. or hire someone here. |
| 23:34.21 | TheSin | loud, that means I need to portforward on my FW |
| 23:34.22 | VirTERM | fixed wireless terminal? how do you register with asterisk? sip? |
| 23:34.32 | TheSin | and they can't login to my system during the day cause it's live |
| 23:34.39 | TheSin | and they close before we do |
| 23:34.43 | VirTERM | what's your dtmfmode in sip.conf |
| 23:34.43 | TheSin | so it's just not possible |
| 23:34.53 | Luhiwu | VirTERM, it has a rj11 plug, i've connected it to the fxo card |
| 23:35.13 | ivesti | can wnyone help me configure a sipura spa-3000 with a dock-n-talk device? |
| 23:35.14 | Luhiwu | it is analog, none of those gsm voip cards:) |
| 23:35.48 | VirTERM | and then you are going our to PSTN through another card? |
| 23:35.53 | VirTERM | I am lost :) |
| 23:36.14 | Luhiwu | VirTERM: i call from a fixed line to a cell phone connected to the FXO card in my asterisk |
| 23:36.28 | Luhiwu | when i play some dtmf tones the * doesn't get them |
| 23:36.31 | ivesti | can anyone help me configure a sipura spa-3000 with a dock-n-talk device? |
| 23:37.54 | VirTERM | what about if you do not use the FXO card? can you use dtmf while talking to *? |
| 23:38.48 | VirTERM | so, you use zaptel to interface with GSM netowrk? |
| 23:38.56 | VirTERM | correct? |
| 23:38.59 | Luhiwu | VirTERM, imagine that the cellphone has a FXS interface on it, i plug that interface into the zaptel FXO card |
| 23:39.13 | Luhiwu | i'm not using gsm, just plain analog cellphones and interfaces |
| 23:39.15 | VirTERM | understood |
| 23:39.31 | VirTERM | now, how are you calling it? how do you originate the call? |
| 23:39.56 | Luhiwu | from a plain old telephone, i do call the cellphone, it rings the zaptel card and i hear the welcome message |
| 23:40.14 | Luhiwu | but then i press some digits and the * doesn't seems to receive them |
| 23:40.34 | VirTERM | now i got it :) |
| 23:40.48 | VirTERM | analog->gsm->zaptel->asterisk ? |
| 23:41.05 | Luhiwu | analogFixed->analogCellphone->zaptel->asterisk |
| 23:42.01 | VirTERM | is there anywhere on your cellphone or terminal adapter config to allow/disallow dtmf tones? |
| 23:42.35 | *** join/#asterisk bjohnson (~bjohnson@66.11.165.161) |
| 23:42.50 | VirTERM | if you plug analog line into zaptel and call it from an anlog phone, can you use dtmf? |
| 23:42.51 | Luhiwu | no, there isn't. It is just a special cable with some chips that plugs into the startac and gives an rj11 interface, nothing configurable there |
| 23:43.07 | Luhiwu | yes, i can use dtmf. |
| 23:43.34 | reallost1 | $digit = $AGI->stream_file('/var/lib/asterisk/sounds/ftfla','89') || die "Couldn't play file $ARGV[1]"; |
| 23:43.34 | reallost1 | print STDERR "---Digit Pressed = $digit \n"; |
| 23:43.46 | VirTERM | how about if you call this line from the startec? can you then use dtmf? |
| 23:44.37 | reallost1 | anyone doing digit collection from agi? |
| 23:44.58 | Luhiwu | VirTERM, i can't call from the startac, i just can call to the startac and from zaptel to the cellphone network |
| 23:45.17 | VirTERM | I mean analogCell->analogFixed->zaptel->asterisk |
| 23:45.30 | *** join/#asterisk meanphil (~pmurray@222-152-246-166.jetstream.xtra.co.nz) |
| 23:46.00 | Luhiwu | VirTERM, i can't do that test right now, but i've called digitalCell->analogFixed->zaptel->asterisk without problems |
| 23:46.01 | *** join/#asterisk PoWeRKiLL (~PoWeRKiLL@l01m-20-133.d4.club-internet.fr) |
| 23:46.12 | Luhiwu | i'll try to do that later tonight |
| 23:46.37 | VirTERM | I am not sure how the dtmf tones are passed on the analog cell network, but it must be very unreliable.... |
| 23:47.08 | Luhiwu | they should be passed inband, i've used it many times to do phonebanking... |
| 23:47.19 | meanphil | Does anyone here succesfully use Asterisk+Festival? Whenever I try to use it, Asterisk just starts chewing 100% cpu |
| 23:47.40 | meanphil | it never says anything and I have to kill -9 asterisk |
| 23:47.44 | meanphil | can't figure out what's wrong with it |
| 23:47.47 | VirTERM | I would suggest to play with levels |
| 23:48.01 | VirTERM | increase the levels on zapata (I mean sensitivity) |
| 23:48.27 | RomanTorres | meanphil: check on modules.conf you don't load chan_oss.so : unload=chan_oss.so |
| 23:48.53 | meanphil | I have, noload => chan_oss.so |
| 23:48.56 | meanphil | and the same for alsa |
| 23:49.10 | meanphil | is that the same? |
| 23:50.11 | RomanTorres | meanphil: yes, other than that , is your asterisk system taking 100% of the load all the time , or only when you convert text with festival? |
| 23:50.38 | meanphil | only when I use festival |
| 23:50.49 | meanphil | whether it's from extensions.conf, or from an AGI script |
| 23:51.19 | RomanTorres | meanphil: check this: http://lists.digium.com/pipermail/asterisk-users/2004-March/040819.html |
| 23:51.27 | *** join/#asterisk ariel_ (~Ariel@dsl-20-177.cofs.net) |
| 23:52.04 | meanphil | ok thanks RomanTorres |
| 23:52.09 | meanphil | I'll try the CVS version |
| 23:53.31 | RomanTorres | meanphil: ok, good luck |
| 23:54.42 | Luhiwu | VirTERM, thanks for your suggestion |
| 23:55.57 | emrah | Anyone have experience with astc here, please? |
| 23:59.59 | *** join/#asterisk robl^ (~robl@dsl093-025-118.hou1.dsl.speakeasy.net) |