| 00:00.02 | jayk2 | ok |
| 00:00.05 | JunK-Y | cjk_: when im using txfax with |caller, im gettting segfault. |
| 00:00.18 | shido6 | ok |
| 00:00.25 | jackfiber | cjk_ I modified Makefile patch but at compilation I got some errorrs which version do u use .1 or .2 ? |
| 00:00.29 | shido6 | jayk2, im gonna help you out here. lemme look through this |
| 00:00.32 | jayk2 | ok |
| 00:00.49 | shido6 | jayk2, |
| 00:00.52 | shido6 | o man |
| 00:01.00 | shido6 | youre killing yourself here |
| 00:01.07 | jayk2 | im not using any of the fwd, iptel, iconnect, iax stuff |
| 00:01.09 | jayk2 | oh/ |
| 00:01.24 | cjk_ | JunK-Y: which version of spandsp |
| 00:01.30 | JunK-Y | pre17 |
| 00:01.41 | cjk_ | jackfiber: 2 pre 16 |
| 00:01.47 | cjk_ | the pre 17 was not working |
| 00:02.06 | JunK-Y | and 16 ? |
| 00:02.10 | cjk_ | works |
| 00:02.27 | JunK-Y | then i'll have to downgrade. |
| 00:02.31 | cjk_ | yup |
| 00:02.32 | jackfiber | cjk_ I used 1 but it doesn't work with latest * 1.x haven't tried 2.x but is it stable? |
| 00:02.46 | cjk_ | jackfiber: i think 1 is outdated |
| 00:03.01 | cjk_ | jackfiber: rxfax is working great |
| 00:03.22 | JunK-Y | rxfax yes. |
| 00:03.36 | jackfiber | cjk_, seems u r right but they noticed 1 is stable 2 isn't, anyway 1 doesn't work properly with latest * |
| 00:04.05 | jackfiber | cjk_, u mean txfax doesn't work on 2? or buggy? |
| 00:04.18 | *** part/#asterisk md` (illuminat@md.boerde.de) |
| 00:05.52 | jackfiber | anyone uses mediaproxy? |
| 00:06.01 | *** join/#asterisk robl^ (~robl@dsl093-025-118.hou1.dsl.speakeasy.net) |
| 00:06.08 | *** join/#asterisk r0d3nt (~RatMan@wsip-24-234-241-84.lv.lv.cox.net) |
| 00:07.18 | *** join/#asterisk chaoscon (~ph33r@chaoscon.user) |
| 00:09.20 | doolph | hello |
| 00:09.36 | cjk_ | jackfiber: it works, but till now only with files i alose recieved from rxfax. |
| 00:09.45 | *** join/#asterisk PTG1234 (~PTG123@ip68-106-24-139.ph.ph.cox.net) |
| 00:09.54 | cjk_ | i was not able to generate a correct file for it and i do not know how to set the headers |
| 00:10.00 | cjk_ | think its not possible |
| 00:10.13 | doolph | any newbie around? |
| 00:10.54 | doolph | only me? |
| 00:13.05 | *** join/#asterisk Kumbang (~ecvs@167.205.24.4) |
| 00:14.37 | doolph | hi |
| 00:14.59 | *** part/#asterisk makhtar (~ageller@mail.bulletinnews.com) |
| 00:18.21 | *** part/#asterisk jackfiber (cico@213.217.38.55) |
| 00:19.14 | *** join/#asterisk tsp (~tyler@S01060080c825173c.vc.shawcable.net) |
| 00:19.39 | tsp | how do I get the dial command enabled? I want to use asterisk as just a plane old sip phone, and got it to register. |
| 00:19.43 | tsp | but no dial command. |
| 00:20.18 | *** join/#asterisk jskcr|lappy (~jskcr@jskcr.user) |
| 00:20.26 | jskcr|lappy | hy all |
| 00:20.28 | doolph | dial command? |
| 00:20.55 | tsp | I want to dial a sip number |
| 00:21.52 | PTG123 | you need to patch it |
| 00:21.54 | PTG123 | to do it tsp |
| 00:22.03 | doolph | patch? |
| 00:22.24 | PTG123 | yes |
| 00:22.38 | *** join/#asterisk Raymond_Cool (gang@CPE00a0cc7896fe-CM0012250080d4.cpe.net.cable.rogers.com) |
| 00:22.43 | PTG123 | http://bugs.digium.com/view.php?id=3651 |
| 00:22.54 | doolph | dial a sip number isnt dial a sip extension? |
| 00:23.20 | tsp | dunno |
| 00:23.36 | doolph | then |
| 00:23.56 | *** join/#asterisk BoRiS (boris@S01060040ca1e5b54.wp.shawcable.net) |
| 00:24.13 | tsp | can i log in to bugs as a guest? |
| 00:24.29 | doolph | anyone want to install AMP? |
| 00:24.50 | PTG123 | not sure just sign up for a quick account |
| 00:25.20 | tsp | i just did |
| 00:25.52 | doolph | for what is spandsp-0.0.2 |
| 00:27.49 | JunK-Y | tsp:no u need an account. |
| 00:28.40 | *** join/#asterisk ariel_ (~Ariel@dsl-20-177.cofs.net) |
| 00:29.25 | tsp | got one |
| 00:29.33 | ariel_ | hello everyone |
| 00:35.32 | Raymond_Cool | any website have a good introduction of telephony system? |
| 00:36.16 | JunK-Y | lo ariel_ |
| 00:36.26 | Raymond_Cool | I even don't know what are those CTI IVR |
| 00:36.27 | tsp | which one of these files applies to asterisk 1.0.7? |
| 00:36.30 | tsp | 1 and 3 don't work |
| 00:36.48 | Qwell | tsp: hand patch it |
| 00:36.59 | Qwell | Is that the dial() change for chan_sip? |
| 00:37.14 | ariel_ | hello JunK-Y |
| 00:37.22 | tsp | dial change yeah |
| 00:37.35 | Qwell | yeah, I just hand patched it, it was only a few lines |
| 00:38.08 | tsp | a few? heh |
| 00:38.17 | tsp | should I go for dial1, 2 or 3? |
| 00:39.21 | Qwell | well, if you look at the Issue History... |
| 00:39.24 | Qwell | 3 was added last |
| 00:40.07 | tsp | oh |
| 00:40.19 | *** join/#asterisk dmccollum (~dmccollum@eycb01-00-cntnga-69-164-245-72.atlaga.adelphia.net) |
| 00:40.28 | dmccollum | Evening everyone |
| 00:40.41 | Qwell | 2 looks familiar to me though |
| 00:41.13 | Qwell | 1 does, actually |
| 00:41.22 | Qwell | I need a diff of these diffs, heh |
| 00:42.33 | tsp | this is totally different from the file |
| 00:42.53 | dmccollum | I have something weird happening. About once every minute when talking to someone on a outside line I hear a quick two or three pitch tone that last for about 1 second. The person on the other end cannot hear it. The call is coming in from BellSouth on a X100p and the phones in the house are Cisco 7960's. Any idea what it might be? |
| 00:42.57 | *** part/#asterisk Kumbang (~ecvs@167.205.24.4) |
| 00:43.08 | Qwell | its a diff of chan_sip.c |
| 00:43.44 | Qwell | ok, I used 3 |
| 00:43.52 | Qwell | Would you like a copy of mine, post-patch? |
| 00:43.59 | dmccollum | Could that be from a IRQ sharing problem? |
| 00:44.50 | Moonwick | dmccollum: the problem goes away if a call doesn't go through the x100p? |
| 00:45.51 | dmccollum | Well that's the only way a call can come in. Wife is talking to a friend atm. I'll try it with her from 7960 to 7960 and see if we hear the tone. |
| 00:47.40 | tsp | thanks. what version is that from? |
| 00:47.44 | Qwell | tsp: 1.0.7 |
| 00:47.50 | Nethab | what'd you do now mavvie |
| 00:48.02 | Mavvie | Nethab: asked for some form of documentation in funcs/ :-) |
| 00:48.07 | tsp | ah |
| 00:48.13 | Nethab | docum... what? what's that |
| 00:48.13 | tsp | there we go |
| 00:48.19 | Mavvie | yes ;-) |
| 00:48.47 | tsp | ok - I don't see a dial command on the console |
| 00:48.50 | tsp | hmm |
| 00:48.58 | Qwell | tsp: It has to be done from the console |
| 00:49.00 | Qwell | erm, heh |
| 00:49.02 | Qwell | from the dialplan |
| 00:49.04 | Nethab | dial is only from in the dialplan |
| 00:49.10 | Qwell | dial from the dialplan is only for stuff in the dialplan... |
| 00:49.11 | Qwell | what he said |
| 00:49.15 | Nethab | if you have chan_oss or chan_alsa, i think you use call |
| 00:49.16 | Qwell | god damnit |
| 00:49.22 | Qwell | dial from the console is onl...etc |
| 00:49.27 | Qwell | I got it wrong twice in a row... :p |
| 00:49.35 | JunK-Y | Nethab: yes via command dial, but used it just once. |
| 00:49.45 | tsp | can't find a call command |
| 00:49.50 | tsp | oh there we go |
| 00:50.31 | *** join/#asterisk Moc (~Moc@modemcable165.109-70-69.mc.videotron.ca) |
| 00:50.42 | Nethab | you can use answer to answer one |
| 00:50.53 | tsp | I'm registered. dial sip/411@proxy01.sipphone.com doesn't work. |
| 00:50.58 | *** join/#asterisk ToyMan (~stuq@user-12lcqur.cable.mindspring.com) |
| 00:51.13 | Qwell | tsp: read the diff for syntax |
| 00:51.49 | Qwell | SIP/<user>@<host>/<exten>, specifically |
| 00:53.22 | Nethab | if sipphone is a peer in sip.conf use it's peer name |
| 00:53.40 | Nethab | for me I use SIP/sipphone/<number> |
| 00:53.48 | NewSole | Anyone Played with Call Transfer |
| 00:53.56 | Nethab | transfer using what |
| 00:54.09 | tsp | keep getting no such extension - sip/proxy01.sipphone.com/411 |
| 00:54.11 | tsp | hmm |
| 00:54.40 | doolph | tsp do you mean call a outside sip server number? |
| 00:54.42 | NewSole | in feachers |
| 00:54.45 | Nethab | tsp, use the peername that's in [] in sip.conf instead of the hostname |
| 00:55.14 | tsp | yeah an outside number |
| 00:55.21 | tsp | my sip.conf has [proxy01.sipphone.com] |
| 00:55.25 | doolph | tsp that's easy man |
| 00:56.04 | tsp | its not working worth a darn though |
| 00:56.17 | doolph | damn and I removed all my backups lol |
| 00:56.31 | doolph | i had it for you |
| 00:56.46 | doolph | you need to wait until I get all reinstalled again |
| 00:56.56 | Nethab | you shouldn't use the whole hostname as the peername in sip.conf |
| 00:57.03 | Nethab | it will confuse the heck out of the dial plan |
| 00:57.08 | Nethab | which is what it's doing |
| 00:57.15 | doolph | 19:57:17 (752.39 KB/s) - `asteriskathome-1.0.iso' saved [569147392/569147392] |
| 00:57.20 | Nethab | cause it doesn't know if you want a domain name lookup or a peer in sip.conf |
| 00:57.35 | Nethab | i use [sipphone] |
| 00:57.42 | *** join/#asterisk cpbell (~cpbell@toronto-HSE-ppp3935611.sympatico.ca) |
| 00:58.11 | cpbell | hi ya'll |
| 00:58.21 | tsp | what's that big iso? |
| 00:58.22 | cpbell | who all is here eh? |
| 00:58.32 | doolph | something that will install anything for you |
| 00:59.07 | tsp | I shortened the name to [sp] not it won't register |
| 00:59.09 | tsp | now |
| 00:59.11 | tsp | type = friend |
| 00:59.25 | *** part/#asterisk cpbell (~cpbell@toronto-HSE-ppp3935611.sympatico.ca) |
| 00:59.59 | Qwell | somebody is thoroughly confused |
| 01:00.26 | dmccollum | Moonwick: It doesn't make the sounds when talking between two internal extensions. |
| 01:00.40 | *** join/#asterisk grolloj (~grolloj@dsl254-116-106.nyc1.dsl.speakeasy.net) |
| 01:00.50 | tsp | oh I should set type=peer |
| 01:00.57 | Moonwick | sounds like it's definitely related to the card... |
| 01:01.08 | Qwell | type=friend is spotty at best |
| 01:01.10 | Moonwick | have you tried moving it to another PCI slot? |
| 01:01.17 | dmccollum | I think I have an IRQ problem. |
| 01:01.18 | Qwell | shouldn't use it, unless you know precisely what you are doing |
| 01:01.33 | dmccollum | I'm shutting it down now to bring up stairs and move the card around. |
| 01:01.46 | Qwell | You have to bring it upstairs to move the card? |
| 01:01.49 | tsp | Nethab: thanks |
| 01:02.00 | dmccollum | No monitor in basement to see what IRQ it picks up. |
| 01:02.49 | Nethab | heh sip is not your friend |
| 01:02.50 | Qwell | no need for a monitor. cat /proc/interrupts |
| 01:04.29 | dmccollum | I know, but it's quicker to see the IRQ at post instead of waiting for the OS to boot. |
| 01:05.11 | tsp | i have register => <number>I don't want to keep editing externip whenever my ip changes - any way around that? |
| 01:05.54 | Nethab | most people use dynamic dns |
| 01:05.58 | tsp | I have "register => <number>:password@proxy01.sipphone.com/number |
| 01:06.01 | Nethab | and use externhost |
| 01:06.05 | tsp | its not registering worth a darn |
| 01:06.16 | tsp | that's in [general] |
| 01:06.26 | Nethab | right |
| 01:06.29 | tsp | nat = yes |
| 01:06.46 | Nethab | nat = yes is where |
| 01:06.51 | tsp | in [sp] |
| 01:06.53 | tsp | for sipphone |
| 01:06.57 | Nethab | you don't need it there |
| 01:07.01 | tsp | I don't? |
| 01:07.04 | Nethab | sipphone is not behind nat you are |
| 01:07.07 | tsp | I'm behind a nat where I can't open any ports |
| 01:07.18 | tsp | oh so I need it in [general]? |
| 01:07.20 | Nethab | that's for if that peer is behind nat |
| 01:07.31 | Nethab | no, because none of those people are behind nat, you are |
| 01:07.39 | tsp | ah |
| 01:07.46 | Nethab | you only need nat = yes for those phones on other peoples networks that have nat too |
| 01:08.10 | tsp | sip show registry shows request sent, and it keeps timing out |
| 01:08.12 | Nethab | not for service providers or the like, as they are on public IP |
| 01:08.25 | tsp | ah |
| 01:08.29 | DrukenHME | bkw_: you around? |
| 01:08.32 | tsp | I got rid of nat=yes |
| 01:08.35 | Nethab | hes in a meeting |
| 01:08.52 | DrukenHME | a meeting.... pfft.. hehe |
| 01:08.56 | *** join/#asterisk r0d3nt|m (~RatMan@wsip-24-234-241-84.lv.lv.cox.net) |
| 01:09.14 | tsp | srvlookup=yes and tos=reliability - not sure waht that is |
| 01:09.47 | Nethab | tos is the QoS bits on the voice packets to tell routers to give them priority |
| 01:10.08 | tsp | hmm |
| 01:10.24 | *** join/#asterisk lyoungz (~lyoung@ool-182d73f5.dyn.optonline.net) |
| 01:11.10 | *** join/#asterisk kimo_sabe (nick@zappa.azrackspace.net) |
| 01:12.45 | dmccollum | Are there any BIOS features that should be on or off like CPU to PCI write buffer when using a x100p? |
| 01:13.15 | Nethab | no idea |
| 01:13.48 | tsp | gtg for a few |
| 01:17.28 | *** join/#asterisk Sato1 (~rauleli@sato1.wizardteam.com) |
| 01:19.32 | *** join/#asterisk Smi|k (~Ling@c-069-063-192-006.sd2.redwire.net) |
| 01:20.09 | *** join/#asterisk dft (~dft@CPE0040050149d6-CM00080d77ae83.cpe.net.cable.rogers.com) |
| 01:20.30 | dft | evenin |
| 01:20.39 | doolph | sup |
| 01:20.57 | Smi|k | hey |
| 01:21.07 | Smi|k | where is that place with polycom 500's for $169? |
| 01:21.09 | dft | just readin up, waiting on my fxo/fxs card to arrive any day soon |
| 01:21.34 | dft | has anyone tried compiling asterisk on freebsd? |
| 01:24.52 | *** join/#asterisk kks (~kks@202.73.10.8) |
| 01:26.28 | doolph | not me |
| 01:26.46 | Nethab | dft, there a few people running it on freebsd, and one on Solaris |
| 01:26.56 | dft | just curious, I'm more familiar with Freebsd than linux |
| 01:27.36 | dft | hmm, worth a shot |
| 01:28.53 | *** join/#asterisk cpbell (~cpbell@toronto-HSE-ppp3935611.sympatico.ca) |
| 01:29.33 | cpbell | I'm Colin... was anyone at VON Canada this year? |
| 01:29.50 | tzanger | I was there but not _there_ |
| 01:29.52 | dft | not I, but nice to meet you Colin |
| 01:30.02 | cpbell | or the meeting at UoT following VON |
| 01:30.08 | cpbell | Mark was there talking. |
| 01:30.14 | cpbell | nice to meet you dft |
| 01:30.26 | cpbell | tzanger, there? like at the meeting afterwards? |
| 01:30.51 | tzanger | cpbell: had dinner with mark |
| 01:30.53 | cpbell | or in toronto during VON? |
| 01:30.59 | tzanger | I was next door doing a server upgrade |
| 01:31.16 | cpbell | tzanger... cool. |
| 01:31.20 | tzanger | yup |
| 01:31.32 | cpbell | then I probably bumped into you. |
| 01:31.36 | file | as you all should be! |
| 01:32.06 | cpbell | or saw you... I was there having dinner with Mark too. |
| 01:32.28 | cpbell | I'm looking for anyone working on the current implementation of Gnophone. |
| 01:32.40 | cpbell | I'm about to start playing with it I think. |
| 01:33.30 | file | hrm, the person isn't in here |
| 01:33.37 | file | 'nor is he online - which is odd |
| 01:34.06 | file | lemme grab him and swing him online |
| 01:34.15 | tzanger | ... |
| 01:34.26 | tzanger | you are jason? |
| 01:34.30 | tzanger | or justin? |
| 01:34.34 | tzanger | I KNOW you're not Moc |
| 01:34.42 | cpbell | from what I hear it needs to be upgraded... it does not support the latest. |
| 01:34.50 | *** join/#asterisk goatmilk (~goatmilk@cpe-024-168-247-080.sc.res.rr.com) |
| 01:34.50 | cpbell | no... I was a friend of maddog's |
| 01:35.07 | *** join/#asterisk darwin35 (~darwin35@24.3.226.147) |
| 01:35.21 | darwin35 | anyone having issues with BroadVoice / |
| 01:35.23 | goatmilk | lo, file shinanigans |
| 01:35.35 | file | darwin35: yes yes and yes |
| 01:35.43 | darwin35 | ok |
| 01:35.44 | file | darwin35: it's been discussed sooooooo many times |
| 01:36.08 | darwin35 | ok |
| 01:36.25 | darwin35 | sorry I need to get a backup then |
| 01:36.39 | file | broadvoice is having problems, 'nuff said |
| 01:37.02 | Nethab | luckily i have dialplan rollover |
| 01:37.08 | *** join/#asterisk NewSole (dave@i216-58-44-245.avalonworks.net) |
| 01:37.13 | Nethab | besides i used up my 100 minutes already this month |
| 01:37.23 | darwin35 | I need to setup one but I dont have a fall back provider |
| 01:37.35 | file | choose Asterlink! |
| 01:37.48 | Nethab | it's usage based right? |
| 01:37.52 | file | yes |
| 01:38.25 | goatmilk | was that it? |
| 01:38.36 | file | goatmilk: Awwwww how sad |
| 01:39.31 | tzanger | darwin35: broadvoice is probably the #1 down VOIP provider that supports asterisk |
| 01:39.36 | goatmilk | what did he want anyways |
| 01:39.40 | tzanger | I could give you two fucking cans with string and have better uptime |
| 01:39.46 | darwin35 | ok I will go look |
| 01:40.04 | Nethab | not with this weather the string would rot |
| 01:40.14 | *** join/#asterisk Rick_Hunter (~rhunter@08-170.008.popsite.net) |
| 01:40.14 | tzanger | Nethab: as I said, I would have better uptime than BV |
| 01:40.20 | nestAr | you get nylon string |
| 01:40.20 | tzanger | wow this liquor is starting ot take hold |
| 01:40.22 | tzanger | excellent |
| 01:40.23 | nestAr | less failure |
| 01:40.29 | nestAr | tzanger: lucky uoi |
| 01:40.31 | tzanger | Nethab: not if I have a lighter |
| 01:40.32 | nestAr | you* |
| 01:40.34 | Nethab | nylon eh, i stopped wearing those years ago |
| 01:40.37 | tzanger | no not lucky me |
| 01:40.45 | nestAr | o? |
| 01:40.50 | tzanger | if I was getting lucky I wouldn't be drinking until my limbs were numb every night |
| 01:40.54 | nestAr | lol |
| 01:41.02 | nestAr | i think it's time for a beer and a shot |
| 01:41.10 | nestAr | cinco de mayo and all |
| 01:41.28 | tzanger | nestAr: it's my bday too |
| 01:41.32 | tzanger | I'm 29 today |
| 01:41.38 | nestAr | nice, happy b-day |
| 01:41.42 | NewSole | BV goes up and down more then a Teenager in a whore house.... |
| 01:41.47 | nestAr | i celebrate mine next sunday, the 15th |
| 01:41.48 | tzanger | nestAr: danke |
| 01:41.52 | NewSole | happy Bd tzanger |
| 01:41.57 | nestAr | i'll be 25 |
| 01:42.02 | tzanger | NewSole: heh I hear it as "up and down more than a bride's nightie" |
| 01:42.11 | tzanger | I wish I were 25 again |
| 01:42.14 | *** join/#asterisk iswm (iswm@iswm.user) |
| 01:42.24 | NewSole | lol I wish I was 40 again |
| 01:42.27 | tzanger | I'd still be married |
| 01:42.30 | tzanger | NewSole: heh |
| 01:42.41 | NewSole | I would still be horny |
| 01:42.49 | tzanger | any chick who knew me that long says I get hotter as I age |
| 01:42.54 | nestAr | i'm getting married july 2nd |
| 01:42.57 | nestAr | :happysad: |
| 01:43.00 | tzanger | congrats nestar |
| 01:43.18 | NewSole | tzanger... I just got shit today |
| 01:43.21 | tzanger | don't fuck up -- when your wife says she wants you but wants the money you bring in by working crazy hours, she wants the former more |
| 01:43.26 | tzanger | trust me |
| 01:43.39 | tzanger | NewSole: what'd you do |
| 01:43.41 | nestAr | fair enough |
| 01:43.42 | tzanger | plug the toilet |
| 01:43.47 | tzanger | speaking of plugs |
| 01:43.51 | tzanger | my fucking shower leaks int ot he kitchen |
| 01:43.54 | tzanger | happy bday to me... ugh |
| 01:44.01 | nestAr | lol |
| 01:44.03 | NewSole | My Provider said I was not using enough Min..... |
| 01:44.13 | tzanger | NewSole: ahh |
| 01:44.21 | tzanger | well shit just dial my echo DID and leave it on |
| 01:44.30 | NewSole | lol |
| 01:44.48 | shido6 | youngstuhs |
| 01:45.00 | NewSole | I could call evey pizza shop in North america and order pizza |
| 01:47.33 | tzanger | shido6: how old is your black ass |
| 01:47.47 | JunK-Y | mouhahah |
| 01:48.00 | tzanger | you didn't sound a day over 25 |
| 01:48.42 | nestAr | :o |
| 01:49.15 | nestAr | damn, i'm out of rum and bourbon |
| 01:49.21 | tzanger | rum sucks |
| 01:50.35 | nestAr | i like some from time to time. |
| 01:50.42 | shepherd | vodka #1 |
| 01:50.53 | goatmilk | shepherd: toliet #2 ? |
| 01:51.14 | doolph | what's the best way to admin cdrs? |
| 01:51.15 | nestAr | lol |
| 01:51.18 | shepherd | heh.. maybe for you :) |
| 01:51.33 | *** join/#asterisk brc_ (~brian@brc.base.supporter.pdpc) |
| 01:51.34 | ariel_ | every time I hear about drinking. I just say lets all have some fun. But I feel we all should view this music vid about it. http://wm.atlrec.com/simple_plan/video/untitled_450.wmv |
| 01:52.30 | *** join/#asterisk sandnigg0r (~niggerplz@66-55-197-254.gwi.net) |
| 01:52.32 | goatmilk | yep, shepherd knows what i am talking about. we've just got to keep our cool quota on irc and pretend that we're all able to down a forty of vodka without seein' doubles |
| 01:53.34 | tzanger | no rye |
| 01:53.36 | tzanger | rye rocks |
| 01:53.39 | tzanger | vodka's not bad |
| 01:53.49 | tzanger | ariel_: wow |
| 01:53.55 | tzanger | AA or for other reasns? |
| 01:55.01 | nestAr | my dad just celebrated 19 years |
| 01:55.02 | ariel_ | other reasons. Saw my best friend get killed in front of me in When I was in the US Airforce in the Philippines by a drunk driver. |
| 01:55.07 | nestAr | of not drinking |
| 01:55.43 | ariel_ | I don't mind people drinking. Just want them to do it with there heads in the right place. |
| 01:56.17 | newl | usually facing down :) |
| 01:56.47 | *** join/#asterisk techie (gus@asterisk.horizonte.us) |
| 01:57.05 | ariel_ | in any case have any of you seen the vid it's a great song by a group in Canada. |
| 01:57.16 | Nethab | i don't think anyones ages is more than 1984 |
| 01:57.35 | kimo_sabe | ariel_: they play Simple Plan down here |
| 01:58.28 | ariel_ | kimo_sabe, great to hear it. |
| 01:58.29 | nestAr | <-- 1980 |
| 01:59.03 | dmccollum | <-- 1970 |
| 01:59.34 | meppl | gute nacht |
| 02:00.06 | ariel_ | kimo_sabe, I also have a 2 year girl so I am starting again. (I also have grand kids). |
| 02:00.14 | tzanger | <-- 1976 |
| 02:00.18 | kimo_sabe | ariel_: neat |
| 02:00.19 | tzanger | what are we saying again" |
| 02:00.21 | DeeJayTwo | ;) |
| 02:00.25 | tzanger | dammit my fingers are going muib |
| 02:00.27 | tzanger | ... |
| 02:00.28 | tzanger | numb |
| 02:00.30 | DeeJayTwo | hi JunK-Y ;) |
| 02:00.38 | newl | keep them outta the booze.. |
| 02:00.46 | dmccollum | tzanger: You'll go blind if you keep going. ;P |
| 02:00.56 | JunK-Y | y0 DeeJayTwo |
| 02:01.00 | Qwell | *go* blind? |
| 02:01.02 | tzanger | dmccollum: no I havent started masturbating |
| 02:01.11 | tzanger | that's usually before I get up in the mornign :p |
| 02:01.32 | ariel_ | what no one has heard of hairy palms..... |
| 02:01.38 | tzanger | hehehe |
| 02:01.43 | tzanger | heard of them? I shave every day |
| 02:01.57 | Nethab | 8675309 |
| 02:02.35 | tzanger | what NPA |
| 02:02.41 | tzanger | I'll call you |
| 02:02.42 | dmccollum | A friend of mine was a bartender at this bar and one morning he comes in to open up and had his Girlfriend with him. They walk in and the owner is passed out on the pool table with his pants around his ankles and the porn channel on the Dish. |
| 02:02.45 | tzanger | who the fuck ar you anyawy |
| 02:02.56 | tzanger | HAHAHAHAHAH |
| 02:02.57 | DeeJayTwo | Does anybody know an esthetic way to change the dial tone when there's a new voicemail waiting or any other condition.. |
| 02:03.02 | DeeJayTwo | ? |
| 02:03.09 | tzanger | DeeJayTwo: stutter dialtone is standard with chan_zap |
| 02:03.23 | DeeJayTwo | define stutter? ;) |
| 02:03.24 | Nethab | stuttering dial tone, bleh |
| 02:03.38 | Nethab | bu bu bu beeeeeee |
| 02:03.44 | DeeJayTwo | oh I see ;) |
| 02:03.47 | Silik0n | its a dial tone with a speech impairment |
| 02:03.53 | tzanger | Silik0n: ehhehe |
| 02:04.01 | tzanger | TO BE YOIURSELF IS ALL THAT YOU CANE |
| 02:04.02 | tzanger | er |
| 02:04.05 | tzanger | THAT YOU CAN BE |
| 02:04.13 | tzanger | Bm D G Em |
| 02:04.16 | Nethab | but what if your a targ |
| 02:04.17 | tzanger | PLAY IT WITH ME |
| 02:05.06 | DeeJayTwo | Bu bu bu beee!! |
| 02:05.13 | Qwell | its more like |
| 02:05.15 | Qwell | bu bu bu buuuuuu |
| 02:05.19 | DeeJayTwo | ohh ;) |
| 02:07.21 | ariel_ | argh now we really have gotten to a low point. sounding out sounds with letters..... |
| 02:07.29 | Tond | is nayone here using MS Visual C++ v6? I have a code which I need compiled, and it won't complie properly on anything else.. |
| 02:07.45 | Tond | I don't have that myself |
| 02:08.05 | *** join/#asterisk cpbell (~cpbell@toronto-HSE-ppp3935611.sympatico.ca) |
| 02:08.14 | cpbell | goatmilk? |
| 02:08.16 | goatmilk | lo |
| 02:08.17 | cpbell | you about? |
| 02:08.19 | goatmilk | yep |
| 02:08.22 | cpbell | nice. |
| 02:08.32 | goatmilk | i thought the email would get your attention |
| 02:08.39 | cpbell | Yes... so I'm Colin. We talked on the phone. |
| 02:08.41 | cpbell | yes. |
| 02:08.48 | *** join/#asterisk outtolunc (~me@adsl-69-110-63-171.dsl.pltn13.pacbell.net) |
| 02:09.13 | *** join/#asterisk Carp1 (carp_xigon@ip-204-97-151-146.modem.logical.net) |
| 02:09.23 | ariel_ | emails about Broad voice are getting to me.... I guess I need to make a filter to delete them. |
| 02:09.30 | Carp1 | I just enabled the manager, and I still cant connect. Any idea's? |
| 02:10.12 | outtolunc | upen up port 5038 |
| 02:10.28 | outtolunc | er open <G> |
| 02:10.40 | Carp1 | I'm not really familiar with linux yet. |
| 02:10.56 | Carp1 | Does the port have to be opened? Or does it open when asterisk tells it to listen? |
| 02:11.17 | ariel_ | Carp1, it's up to your firewall |
| 02:11.30 | outtolunc | if you have iptables running and it's config'd fairly well, yes you will need to punch a hole for it |
| 02:11.34 | Carp1 | I don't have a firewalll setup. |
| 02:11.56 | kimo_sabe | firewalls are overrated anyway |
| 02:12.02 | dmccollum | Praise the lord I finally got every PCI card on it's own IRQ. |
| 02:12.08 | *** join/#asterisk tessier (~treed@222.253.78.85) |
| 02:12.45 | ariel_ | PCI cards on there own IRQ's sometimes it's a matter of moving cards around and playing with the bios's. Fun stuff. |
| 02:13.00 | Carp1 | http://www.xorcom.com/manager_tip.html |
| 02:13.00 | outtolunc | carp1: if you don't have a firewall and /etc/asterisk/manager.conf is enable=yes, and has a user |
| 02:13.04 | Carp1 | I followed that guide. |
| 02:13.18 | outtolunc | then maybe you need to read the manager doc |
| 02:13.19 | Carp1 | It it enabled, and I have a user. |
| 02:13.25 | dmccollum | yea, I've been moving cards around and changing BIOS settings for the past 40 minutes. |
| 02:13.33 | *** join/#asterisk s9510278 (~bigdaddy2@203.210.152.35) |
| 02:13.52 | outtolunc | http://www.digium.com/asterisk_handbook/manager.html |
| 02:14.33 | Carp1 | Nevermind I got it. |
| 02:14.35 | ariel_ | dmccollum, I have setup 3 SC420 in the pass 2 weeks with 2 or 3 TDM400p cards in them. I can understand. |
| 02:14.36 | Carp1 | I was using telenet wrong lol |
| 02:14.40 | Carp1 | Sorry and thanks. |
| 02:14.43 | outtolunc | np |
| 02:15.04 | s9510278 | HI, anyone familiar with Queue and related stuff? |
| 02:15.50 | outtolunc | i'm sure some are.. what is the question |
| 02:17.09 | s9510278 | i make a queue with 1 agent (dynamically added), when the agent's busy with a call, the second caller will listen to music, but after approx. 90sec, the 2nd caller drops out for no reason. Why? |
| 02:17.59 | outtolunc | many reasons, there is probably a timeout set somewhere |
| 02:18.24 | s9510278 | no, timeout is set to 300 sec (5 mins) for both Queue and Dial commands |
| 02:19.04 | s9510278 | can it be that the *Local* channel is configured to hangup after 90 seconds??? |
| 02:19.14 | outtolunc | what options are you using in regards to leavewhenempty/joinwhenempty |
| 02:19.35 | Qwell | Is the 2nd caller hanging up? heh |
| 02:19.37 | outtolunc | it, might if an absolutetimeout was set somewhere |
| 02:19.38 | s9510278 | leaveempty = yes, joinempty=no |
| 02:19.49 | Qwell | The most common solution is often the most obvious |
| 02:19.55 | Qwell | erm |
| 02:19.56 | s9510278 | the 2nd caller is NOT hanging up (it's ME :) ) |
| 02:20.02 | Qwell | I think I got that backwards :p |
| 02:20.36 | outtolunc | you best bet is to turn on full debugging and watch the cli |
| 02:20.38 | s9510278 | I test with both handsets so I know when the handset is hung up :) thanks guys |
| 02:21.01 | *** join/#asterisk |Vulture| (~V@c-69-180-67-228.hsd1.fl.comcast.net) |
| 02:21.01 | outtolunc | you haven't said what cvs you are using |
| 02:21.24 | tzanger | TEH CVS |
| 02:21.28 | outtolunc | (notes: there is an issue with using Local/ channels |
| 02:21.33 | outtolunc | ) |
| 02:21.47 | s9510278 | i'm using Ast@home version 0.8 |
| 02:22.03 | outtolunc | never used it... |
| 02:22.21 | outtolunc | is it based on stable? (i'd assume so) |
| 02:22.38 | s9510278 | well, i'm not sure but i'd think so |
| 02:22.40 | |Vulture| | on a PRI if you have 11B+1D is the 1D channel still channel 24 |
| 02:22.46 | tzanger | ... |
| 02:22.48 | Nethab | @home is v 1.0 now |
| 02:22.51 | tzanger | yes the D is always 24 |
| 02:22.55 | |Vulture| | kk |
| 02:22.58 | tzanger | on T1 PRI |
| 02:23.03 | s9510278 | the CLI gives me this ... |
| 02:23.05 | tzanger | on E1 pri it's 16 I think |
| 02:23.05 | outtolunc | vulture, it depends what part of the world you are in |
| 02:23.14 | s9510278 | -- Started music on hold, class 'default', on UniCall/1-1 |
| 02:23.15 | s9510278 | <PROTECTED> |
| 02:23.15 | s9510278 | <PROTECTED> |
| 02:23.15 | s9510278 | <PROTECTED> |
| 02:23.15 | s9510278 | <PROTECTED> |
| 02:23.15 | s9510278 | <PROTECTED> |
| 02:23.17 | s9510278 | <PROTECTED> |
| 02:23.19 | s9510278 | <PROTECTED> |
| 02:23.21 | s9510278 | <PROTECTED> |
| 02:23.23 | s9510278 | <PROTECTED> |
| 02:23.25 | s9510278 | <PROTECTED> |
| 02:23.27 | s9510278 | <PROTECTED> |
| 02:23.29 | s9510278 | <PROTECTED> |
| 02:23.38 | outtolunc | s9510278: pastebin tends to upset people less <G> |
| 02:23.51 | |Vulture| | outtolunc: usa |
| 02:24.05 | s9510278 | sorry outtolunc, it's my first time using IRC, pardon me |
| 02:24.06 | outtolunc | vulture: then yes 24 |
| 02:24.16 | doolph | hello I have a question about hardware and phone company services |
| 02:24.22 | Qwell | doolph: fire away |
| 02:24.35 | |Vulture| | kk thnx rollover tommorow |
| 02:24.42 | doolph | if I buy a T1/E1 hardware what service should I request? |
| 02:24.44 | *** join/#asterisk Rick_Hunter (~rhunter@08-170.008.popsite.net) |
| 02:24.48 | Nethab | rollover? |
| 02:24.54 | ariel_ | doolph, where are you located? |
| 02:24.59 | doolph | Panama |
| 02:25.21 | Qwell | having a T1/E1 would be a good start |
| 02:25.27 | dmccollum | Well I think I fixed my problem with getting weird tones every minute. Swaped cards around till every card had a unique IRQ and swapped the memory from PC100 to PC133. |
| 02:25.29 | doolph | they have some service called isdn Did |
| 02:25.31 | ariel_ | I think that Panama goes with US standards. So you should be able to get a PRI down there. |
| 02:25.32 | Carp1 | Login by sending the following string: |
| 02:25.32 | Carp1 | “Action: Login\r\n\ |
| 02:25.32 | Carp1 | Username: user_name\r\n\ |
| 02:25.32 | Carp1 | Secret: user_password\r\n\ |
| 02:25.32 | Carp1 | ActionID: 1\r\n\r\n” |
| 02:25.34 | Carp1 | Whoops lol |
| 02:25.39 | Carp1 | DIdnt know it would be so many lines |
| 02:26.51 | outtolunc | haha |
| 02:26.51 | |Vulture| | yea from old service to new PRI |
| 02:26.51 | doolph | so they will give me a service called PRI |
| 02:26.51 | Nethab | isdn PRI pr BRI os good |
| 02:26.51 | Carp1 | anyways, that seems simple, I try it in telnet and it doesnt work |
| 02:26.51 | |Vulture| | going from T1 to PRI |
| 02:26.51 | Nethab | cutover |
| 02:26.51 | outtolunc | carp: do it by hand, and there is a 'walkthru' at the end of that manager doc |
| 02:26.51 | |Vulture| | Nethab: thanx yea thats it |
| 02:26.51 | outtolunc | (i know i wrote it) |
| 02:26.51 | |Vulture| | they activated the Data today... but phone tommorow |
| 02:29.28 | Carp1 | Action: Login <CR> |
| 02:29.40 | *** join/#asterisk iguy (~iguy@dsl093-197-234.mke1.dsl.speakeasy.net) |
| 02:29.44 | Carp1 | I am guessing <CR> is CR’s |
| 02:29.49 | *** join/#asterisk chaoscon (~ph33r@chaoscon.user) |
| 02:29.52 | outtolunc | yeah, means hit the <enter> key |
| 02:30.05 | Carp1 | It's not working via telnet |
| 02:30.11 | outtolunc | sure it does |
| 02:30.11 | Carp1 | adter I do Secret: pass... |
| 02:30.16 | Carp1 | it does nothing |
| 02:30.23 | Carp1 | and I hit enter again and it says auth required |
| 02:30.30 | outtolunc | hit enter TWICE |
| 02:30.54 | Carp1 | I do it like this: |
| 02:31.01 | Carp1 | Action: Login <enter> |
| 02:31.09 | Carp1 | Username: blah |
| 02:31.13 | Carp1 | Secret: blah |
| 02:31.16 | Carp1 | whoops |
| 02:31.21 | Carp1 | both with enter's after them |
| 02:31.32 | outtolunc | you need action: login<enter> username: yourum<enter> secret: yoursecret<enter enter> |
| 02:31.44 | Carp1 | Yeah, I'm doing that |
| 02:31.46 | Carp1 | let me try again |
| 02:31.47 | tzanger | hehehehe I have your login now |
| 02:31.53 | tzanger | yourum:yoursecret |
| 02:31.56 | tzanger | hehehe R00TED |
| 02:31.59 | tzanger | PWN3D |
| 02:32.02 | tzanger | I ROCK |
| 02:32.05 | tzanger | FUCK ALL OF YOU |
| 02:32.10 | tzanger | oh wait |
| 02:32.12 | tzanger | this isn't efnet |
| 02:32.17 | tzanger | dammit this is asterisk |
| 02:32.20 | tzanger | and I"m makikng f aofol |
| 02:32.23 | outtolunc | yourum was shorthand for yourusername |
| 02:32.24 | Carp1 | hmm |
| 02:32.29 | Carp1 | I guess it doesnt work when you use capitals? |
| 02:32.35 | outtolunc | typo'd obviously |
| 02:33.02 | outtolunc | i've never tried caps but it shouldn't matter |
| 02:33.05 | Silik0n | tzanger its still eye are sea so that is still value text |
| 02:33.20 | tzanger | value text |
| 02:33.27 | tzanger | sound slike something walmart would come out with |
| 02:33.28 | Silik0n | and its urmom |
| 02:33.32 | tzanger | LOWER WAGES... EVERYDAY |
| 02:33.55 | Silik0n | E/_i~|~E +v//>E r(_)l[-$ |
| 02:35.19 | tzanger | Silik0n: I thought TCP guaranteed data integrity |
| 02:35.47 | Silik0n | this is a UDP irc client |
| 02:35.55 | tzanger | haha |
| 02:35.55 | tzanger | ha |
| 02:36.01 | tzanger | you almost made me spit my rye |
| 02:36.10 | tzanger | wtf was that anyway |
| 02:36.13 | outtolunc | speaking of rye |
| 02:36.16 | tzanger | 22:37 < Silik0n> E/_i~|~E +v//>E r(_)l[-$ |
| 02:36.17 | outtolunc | (afk) |
| 02:36.23 | tzanger | outtolunc: get me more |
| 02:36.26 | tzanger | i'm almost out |
| 02:36.30 | Silik0n | the best thing I heard today was another vendor claiming a test call was a failure when they rcvd a cancel message out of order with the 200 ok |
| 02:36.32 | tzanger | only enough for one or two more wuadfs |
| 02:36.33 | tzanger | er |
| 02:36.34 | tzanger | quads |
| 02:36.44 | Silik0n | on SIP over UDP no less |
| 02:37.47 | Carp1 | action: mailboxstatus |
| 02:37.47 | Carp1 | mailbox: 100 |
| 02:37.47 | Carp1 | Response: Error |
| 02:37.47 | Carp1 | Message: Missing action in request |
| 02:37.51 | *** join/#asterisk kb1_kanobe (~krisbouti@h24-207-80-55.cst.dccnet.com) |
| 02:37.53 | Carp1 | Why is there a missing action? |
| 02:38.34 | Juggie | carp, you cant send the right signals easy over telnet |
| 02:38.37 | Juggie | why dont you use php |
| 02:38.44 | tzanger | php sucks ass |
| 02:38.58 | tzanger | I have enough for maybe 5 shots |
| 02:39.02 | Juggie | perl,python etc... whatever then |
| 02:39.05 | tzanger | remember these are chanadian shots |
| 02:39.06 | tzanger | er |
| 02:39.08 | tzanger | canadian |
| 02:39.33 | Carp1 | Yeah, I'll check it out with PHP later. |
| 02:39.34 | Carp1 | Thanks for the help. |
| 02:41.58 | *** join/#asterisk Juxt (~Juxt@sfl-dsl-64-135-113-4-cust.host.net) |
| 02:42.40 | *** part/#asterisk Gand_DJ (~gandalf@ptr-207-54-104-24.ptr.terago.ca) |
| 02:43.25 | *** join/#asterisk iguy (~iguy@dsl093-197-234.mke1.dsl.speakeasy.net) |
| 02:46.44 | *** join/#asterisk anderiv (~anderiv@207-67-87-34.gen.twtelecom.net) |
| 02:46.49 | *** join/#asterisk key2 (~key2@gob75-2-81-56-64-17.fbx.proxad.net) |
| 02:46.51 | key2 | hey |
| 02:46.58 | Juxt | good afternoon |
| 02:47.02 | Juxt | i mean good nite |
| 02:47.04 | *** part/#asterisk JunK-Y (~junky@modemcable174.107-81-70.mc.videotron.ca) |
| 02:47.29 | key2 | I was wondering to know if it would be interesting to make a PCI card that would do the echo cancellation for asterisk... |
| 02:47.39 | Juxt | i have the following issue - i am using cdr-pgsql |
| 02:47.52 | *** join/#asterisk mbishop (~martin@mbishop.user.gentoo) |
| 02:47.55 | Juxt | and my amaflags get set to 3 no matter what the actual setting is in the iax.conf |
| 02:49.14 | mbishop | is it possible with kiax or just asterisk itself to make 3 way calling? like I call someone, hold them, call someone else, unhold and have them bridged? |
| 02:49.19 | kb1_kanobe | key2: there's plenty of gear to do that at the interface level - see Tellabs for examples. |
| 02:49.45 | outtolunc | key2, my thoughts would be, it might be for some, but issues are added bus usage, and a lot of people are building 1U's where the 1 slot would be taken by a digium card |
| 02:50.33 | tzanger | wow |
| 02:50.35 | kb1_kanobe | they either need to fix the software echo can (not impossible) or integrate echo cancellation on the interface cards (coming shortly to a card near you I ear...) |
| 02:50.37 | tzanger | holy fuck I cant'r ead |
| 02:50.53 | tzanger | kb1_kanobe: te410 and 405 have hardare cho can modules now |
| 02:50.54 | kb1_kanobe | s/ear/hear |
| 02:51.25 | kb1_kanobe | that'll help a lot of new people, but it's still not going to get mec2 fixed... |
| 02:51.53 | tzanger | MEC2 works for me |
| 02:52.07 | tzanger | for the most part |
| 02:52.11 | key2 | outtolunc: firewire :) |
| 02:52.12 | outtolunc | <- has been on MEC2 since day 1 (years ago) |
| 02:52.22 | kb1_kanobe | and works for me but only on some of my calls - say, 60%. |
| 02:52.27 | mbishop | 3 way calling with kiax/asterisk? anyone? |
| 02:52.44 | key2 | tzanger: really ? |
| 02:52.47 | outtolunc | key2, when was the last time you seen a 1U with a firewire hanging out of it <G> |
| 02:53.00 | outtolunc | (in a datacenter) |
| 02:53.06 | tzanger | yup |
| 02:53.07 | key2 | outtolunc: never saw one |
| 02:53.14 | kimo_sabe | outtolunc: Xserve :) |
| 02:53.16 | outtolunc | point taken? |
| 02:53.20 | key2 | yeap |
| 02:53.23 | outtolunc | xserve is diff |
| 02:53.30 | outtolunc | (it's sweet) |
| 02:53.39 | outtolunc | it rates extra shit |
| 02:53.45 | key2 | tzanger: so basically how is it ? and can the module echo cancel for all the lines ? |
| 02:53.52 | tzanger | key2: yes |
| 02:53.56 | tzanger | contact digium |
| 02:54.01 | tzanger | IIRC they have tradein |
| 02:54.03 | tzanger | options |
| 02:54.07 | key2 | outtolunc: by the way, if u use SIP, without digium cards, then it might be usefull ? |
| 02:54.24 | *** join/#asterisk newmedian (~crowlther@Quebec-HSE-ppp230300.qc.sympatico.ca) |
| 02:54.37 | tzanger | kb1_kanobe: nope |
| 02:55.11 | tzanger | TE410/405p[ |
| 02:55.13 | tzanger | without the [ |
| 02:55.17 | kb1_kanobe | :-) |
| 02:55.29 | tzanger | <-- reall altoghether too drunk to type |
| 02:56.26 | mbishop | 3 way calling with kiax/asterisk? anyone? |
| 02:59.10 | *** join/#asterisk mansol (~blah@210-9-153-152.msi.net.au) |
| 03:01.27 | Nethab | three way calling can be done |
| 03:01.49 | Nethab | the easiest way is within the phone itself |
| 03:02.56 | Nethab | but if you look up the G dial plan option you can jump to another dialplan entry after your call is answered |
| 03:03.09 | Nethab | and from the other dialplan entry you should be able to dial again |
| 03:03.25 | file | that won't do three way calling though... |
| 03:03.29 | Nethab | ssssh |
| 03:03.32 | Nethab | let him try it |
| 03:03.38 | file | okay then |
| 03:03.48 | mansol | hi everyone - i was wondering if someone may be able to help me, anyone here with an understand of the way asterisk does callerID on pstn? |
| 03:03.59 | file | mansol: that's very vague |
| 03:04.12 | Nethab | asterisk doesn't generate caller id |
| 03:04.20 | Nethab | the pstn fxo interface device does |
| 03:04.47 | file | if you're talking PRIs then it's a facility message or ie... I forget, not a PRI person... |
| 03:05.32 | Nugget | not file[pri]? :) |
| 03:05.38 | file | nope |
| 03:05.55 | file | I know the general idea, enough that I like it, but I don't know the deep stuff |
| 03:05.58 | file | SIP is my forte |
| 03:06.00 | kb1_kanobe | it's a FACILITY message for NI-2, I beleive |
| 03:06.13 | *** part/#asterisk darwin35 (~darwin35@24.3.226.147) |
| 03:06.25 | mansol | We are trying to interface asterisk with a legacy PBX as a voicemail system (the phone system is a Hybrex GDS in Australia). I am having trouble getting the Called number. The format the phone system _should_ be sending is along the lines of C11+STB+STA where STA is the Caller ID, STB is the called number. I am after STB but asterisk only seems to see the last three digits, STA! |
| 03:06.30 | Smi|k | is there a chance that I will get a T1 and it will not be good for voip? |
| 03:07.12 | kb1_kanobe | mansol: is it connected via T1/PRI or is this DTMF signalling? |
| 03:07.13 | Smi|k | mansol, sounds like you need to add a listener delay |
| 03:07.31 | sudhir492 | Smilk: Yes, if you get only 4 digits on the trunk |
| 03:07.44 | Smi|k | 4 digits on the trunk? |
| 03:07.52 | sudhir492 | make sure you ask your telco to give you all 10 digits |
| 03:08.03 | mansol | sorry - should have been more clear, I have a TDM400B with 4 FXO modules connected to the phone system |
| 03:08.11 | Smi|k | this is data T1 |
| 03:08.18 | mansol | TDM400P sorry |
| 03:08.20 | file | mansol: oh... uh... yeah, have fun |
| 03:08.39 | sudhir492 | Smilk: sorry I misunderstood you. Data T1 is fine |
| 03:08.45 | Smi|k | if each voip line is 40kbps then a data t1 should carry 35+ lines tight? |
| 03:08.56 | Smi|k | while a voice T1 will be much less |
| 03:09.02 | Smi|k | why would someone opt for a voice t1? |
| 03:09.09 | *** join/#asterisk Dovid (~hirisk@pool-138-89-23-29.mad.east.verizon.net) |
| 03:09.29 | file | Smi|k: uh because voice is actual phone, while data is just data... |
| 03:09.37 | file | voice is uncompressed, a PRI... |
| 03:09.37 | mansol | file: any hints? I am not sure where to go from here.... |
| 03:09.39 | file | data is like, to the net |
| 03:09.44 | file | mansol: last I checked asterisk didn't support it |
| 03:09.48 | sudhir492 | A data T1 sure will carry over 40 conversations on g729 |
| 03:09.55 | Nethab | because data T1 service level agreement isn't as good as voice T1 |
| 03:09.57 | Smi|k | but isint T1 very stable data connect? |
| 03:10.08 | file | PRIs are still great |
| 03:10.08 | mansol | file: didn't support what? i can get a portion of the callerid string so it must be supporting something? |
| 03:10.32 | file | mansol: it doesn't support getting the DID dialed over a regular phone line to an FXO module, it does support callerid though |
| 03:10.39 | sudhir492 | Smilk: if you have to interface with pstn, you need voice t1 |
| 03:11.08 | Smi|k | isint there plenty of low cost voip providers to interface with pstn? |
| 03:11.16 | file | sure but a PRI is much more reliable |
| 03:11.21 | sudhir492 | smilk: definitely no |
| 03:11.27 | file | (well, that's controversial...) |
| 03:11.28 | sudhir492 | if you keep quality in mind too |
| 03:11.58 | mansol | file: The pbx is sending the DID as part of the callerID, specifically DID+CallerID. I can't see how asterisk can only support reading the last 4 digits of this string and not the last 8? |
| 03:11.59 | sudhir492 | Even a well known company like Primus sucks |
| 03:12.06 | Smi|k | so voip using services like broadvoice and others is not good enough for business use? |
| 03:12.12 | sudhir492 | Their voip termination is BAD |
| 03:12.16 | file | Smi|k: broadvoice has been having problems since Monday fyi |
| 03:12.26 | file | mansol: like I said, have fun :) |
| 03:12.31 | sudhir492 | so I resorted to terminating everything via pstn |
| 03:12.39 | Smi|k | lol |
| 03:12.43 | mansol | file: thanks :) |
| 03:12.44 | Smi|k | I see why everyone is against voip now :) |
| 03:12.49 | tsp | dumb question. where are the asterisk docs? |
| 03:12.54 | file | ~useful asterisk docs |
| 03:12.55 | jbot | it has been said that useful asterisk docs is it has been said that useful asterisk docs is (BOOKMARK THEM!): http://www.digium.com/index.php?menu=documentation (look at the "Unnoficial Links") and http://www.voip-info.org/wiki-Asterisk (the Wiki), and http://www.fnords.org/~eric/asterisk (ManxPower's site), and http://asteriskdocs.org, also, read all files in ... |
| 03:12.56 | Smi|k | what about pstn incoming and voip outgoing? |
| 03:13.02 | Smi|k | with PSTN outgoing as backup? |
| 03:13.24 | tsp | fsbot: more |
| 03:13.31 | file | tsp: that's the good ones. |
| 03:13.37 | tsp | ah |
| 03:13.38 | sudhir492 | for voip, i just trust my own servers :-) |
| 03:13.39 | tsp | thanks |
| 03:13.46 | tsp | trying to get sipphone to work from behind nat but it can't register |
| 03:14.00 | file | tsp: set nat=yes in the entry in sip.conf, and qualify=yes |
| 03:14.10 | Carp1 | How cna I press 9 and get a new dial tone? |
| 03:14.22 | Carp1 | I want to press 9, new tone, then the number i'm calling. |
| 03:14.31 | Nugget | No you don't. |
| 03:14.33 | file | Carp1: depends where you're typing 9, if it's a SIP phone or SIP adapter read the documentation for it - asterisk isn't providing the dialtone |
| 03:14.36 | Nugget | You just think that's what you want. |
| 03:14.54 | file | Nugget: stop telling people what to think, they'll just ask you for help |
| 03:15.02 | Nethab | ignorepattern is useless unless your on zap |
| 03:15.03 | Nugget | eek, you're right. |
| 03:15.03 | newmedian | Jar jar says DISA what he wants? ;) |
| 03:15.07 | tsp | the only files i have in /etc/asterisk are modules.conf and sip.conf. |
| 03:15.12 | tsp | i can't register with type=peer |
| 03:15.30 | file | tsp: there's example configs galore out there, go find them and try them... and read up on the different options |
| 03:15.31 | tsp | keep getting critical request, and timed out errors |
| 03:15.40 | Carp1 | Does asterisk handle the ring tone? |
| 03:15.53 | *** join/#asterisk tessier (~treed@222.253.78.85) |
| 03:15.56 | file | you for sure need nat=yes, and if your NAT setup is strict enough you'll need to map the UDP ports 5060 plus whatever RTP ports sjphone uses |
| 03:16.06 | file | Carp1: that depends ... |
| 03:16.31 | tsp | I can't map ports - I have no control over the router |
| 03:16.38 | tsp | sjphone works just fine over nat |
| 03:16.39 | Carp1 | SIP phonre |
| 03:16.41 | Carp1 | phone* |
| 03:17.01 | file | Carp1: it can generate it, your phone can generate it, and the far remote side can generate it |
| 03:17.23 | Carp1 | I want asterisk to handle it for my SIP |
| 03:17.45 | file | progressinband=yes, but that'll only work if asterisk gets proper indication of how the call is progressing |
| 03:18.09 | file | if you get a "is making progress passing it to" then the indication (ringing/busy/whatever) is being sent as audio to your phone from the remote side and you have no control over that |
| 03:18.21 | tsp | I can't register with sipphone no mater what I do |
| 03:18.24 | Carp1 | cna I just change ring = in indications.conf |
| 03:18.25 | Carp1 | ? |
| 03:18.37 | file | Carp1: go ahead and try, experiment |
| 03:19.03 | file | but all I'm saying is you're insane for doing it and it probably won't work all the time |
| 03:19.27 | Carp1 | Why am I insane? lol |
| 03:19.40 | file | nevermind |
| 03:19.40 | tsp | file: I changed my entry from [sipphone] to [proxy01.sipphone.com] and I now get "is unreachable!" |
| 03:19.49 | file | if it doesn't work, don't come running to me |
| 03:19.52 | *** join/#asterisk CdtDelta_PM (~CdtDelta_@dsl081-225-161.chi1.dsl.speakeasy.net) |
| 03:19.57 | Nethab | sipphone is the provider |
| 03:20.07 | file | tsp: just go find examples of a sipphone configuration... there's probably tons out there |
| 03:20.23 | file | and is your asterisk behind NAT? |
| 03:20.24 | Nethab | are you sure your password is right |
| 03:20.36 | NewSole | tsp |
| 03:20.58 | NewSole | can you http://pastebin.ca info |
| 03:21.11 | file | I have a feeling I know what's going on |
| 03:21.26 | NewSole | so do I want to make sure tho |
| 03:21.43 | file | good, then you can help |
| 03:21.47 | tsp | file: my asterisk is behind a nat |
| 03:21.51 | file | ah that's why |
| 03:21.52 | *** join/#asterisk r0d3nt (~RatMan@wsip-24-234-241-84.lv.lv.cox.net) |
| 03:21.56 | newmedian | All: Here's a thought (comments?): Why don't (we) construct a Help Wizard a la Microsoft Troubleshooter, that asks basic questions in a flowchart fashion to help diagnose the most common SIP/NAT/provider connectivity questions? Then we can direct people to it. |
| 03:22.11 | file | you need to set externip to your external IP, and localnet to your local network settings |
| 03:22.16 | file | newmedian: feel free to |
| 03:22.56 | tsp | i have externip and localnet set |
| 03:22.57 | Sato1 | i got some work started on an eggrop to do that kind of sopport, but requires lot of time grabing the questions, answering, bla bla bla |
| 03:23.07 | newmedian | I just might indeed. |
| 03:23.13 | syle | wcfxo what module is that? |
| 03:24.03 | newmedian | Is there a F/OSS wiki collab style for automated troubleshooting that anyone knows off the top of their head? (Obviously I'll sourceforge/freshmeat/google) |
| 03:24.38 | file | syle: There's a file that outlines what each module is for... |
| 03:25.03 | file | crazy people |
| 03:25.06 | tsp | does it matter if i get "requested contexts couldn't be merged?" |
| 03:25.52 | file | syle: ironically it's called "README" in zaptel |
| 03:26.05 | tsp | ok - its on pastebin |
| 03:26.38 | Nethab | wait, i just noticed, my registers to sipphone are timing out |
| 03:27.08 | file | conspiracy? I think so. |
| 03:27.34 | Nethab | chan_sip.c:8158 handle_response: Got 200 OK on REGISTER that isn't a register |
| 03:27.37 | tsp | is sipphone down? |
| 03:27.44 | Nethab | i get that instead |
| 03:27.46 | *** join/#asterisk CdtDelta_PM (~CdtDelta_@dsl081-225-161.chi1.dsl.speakeasy.net) |
| 03:28.12 | file | they're a little bogged down |
| 03:28.35 | *** join/#asterisk gambolputty (~gambolput@cblmdm65-221-51-80.buckeye-express.com) |
| 03:28.43 | file | Nethab: can you grab me a sip debug of a register attempt? I'm curious to see something |
| 03:29.04 | tsp | I have an fwd account maybe that'll work |
| 03:29.05 | tzanger | wow |
| 03:29.07 | Nethab | yeah i'm waiting for it to retry |
| 03:29.08 | tzanger | I have a large bladder |
| 03:29.11 | tsp | I just wanna test asterisk and use it for a sip client |
| 03:29.12 | mbishop | is it possible with kiax or just asterisk itself to make 3 way calling? like I call someone, hold them, call someone else, unhold and have them bridged? |
| 03:29.16 | tzanger | how many people can pee for 25 seconds straight |
| 03:29.28 | tsp | tzanger: who cares? |
| 03:29.32 | Carp1 | mbishop, you can do a conference. |
| 03:29.32 | tsp | tzanger: start up a wiki about that |
| 03:29.34 | tzanger | tsp: hehe |
| 03:29.34 | file | mbishop: asterisk doesn't do three way calling without putting everyone into a meetme, it's phone dependent... so see if kiax supports it |
| 03:29.38 | HeadachesAbound | I have some code sitting around that would work well for building a web-based troubleshooting app if you want to get the questions / answers together newmedian |
| 03:29.44 | mbishop | file: it doesn't |
| 03:29.49 | mbishop | file: well as far as I can tell |
| 03:30.25 | HeadachesAbound | it was previously used to build a flow-based troubleshooting app. |
| 03:30.33 | Carp1 | Whats a good softphone? |
| 03:30.42 | Carp1 | |
| 03:31.04 | file | X-Lite, Eyebeam, Firefly, SJphone, take your pick |
| 03:31.07 | Nethab | x-lite, sjphone |
| 03:31.08 | Sato1 | x-ten |
| 03:31.19 | Carp1 | Thanks :) |
| 03:31.21 | Nethab | firefly was janky, but i haven't used the new version yet |
| 03:31.25 | tsp | I need a softphone for the console |
| 03:31.29 | tsp | linphone doesn't work worth a darn |
| 03:31.35 | Nethab | it would bridge the voice channels even if i didn't answer |
| 03:31.55 | newmedian | Headaches: Thanks; I've got some ideas fomenting and will do some groundwork as a first step |
| 03:31.56 | Carp1 | x-lite.com? |
| 03:32.01 | Nethab | xten.com |
| 03:32.15 | tsp | xten isn't console |
| 03:32.43 | newmedian | I've got Xten's eyeBeam here, and it's respectable. |
| 03:33.08 | mansol | callerid is nuts! is there anyone with familiarity of the australia caller id system? |
| 03:33.19 | mansol | i am wondering when the caller id gets sent? |
| 03:33.22 | newmedian | For simplicity w/IAX2 Firefly seems to get in and out quickly, and seems more stable than SIP/eyeBeam. |
| 03:33.46 | file | there's some weird flukes with the Xten stuff... |
| 03:33.50 | Nugget | eyeBeam is chock full of features that have no relevance whatsoever when talking to an asterisk server. |
| 03:34.04 | file | Nugget: yet. |
| 03:34.12 | Nugget | I will say this, though. the acoustic echo cancelling in xpro and eyebeam are invaluable for users who want to just use their speakers and a mic to talk. |
| 03:34.28 | newmedian | Nugget: agreed. |
| 03:35.09 | newmedian | I've found that elevating the task priority to High in the Task Manager also increases stability for SIP/IAX2 calls on soft phones. |
| 03:35.59 | tsp | is eyebeam console based? |
| 03:36.07 | Nugget | no, it's windows or osx based. |
| 03:36.13 | tsp | damn |
| 03:36.21 | file | there's no good console based softphones, give up now and keep your sanity |
| 03:36.31 | Nethab | here's the sipphone debug |
| 03:36.50 | Nethab | http://pastebin.ca/11098 |
| 03:37.21 | tsp | thats why I'm using asterisk for a softphone |
| 03:37.22 | file | their system is... interesting |
| 03:38.05 | file | Nethab: they're bogged down with requests I think |
| 03:38.42 | Nethab | i think the 200 is coming out of order and asterisk thinks it's not related to it's request |
| 03:38.46 | file | yes |
| 03:38.57 | *** join/#asterisk knapjack (~root@ip76.qdsl.openaccess.org) |
| 03:39.02 | file | we schedule deletion of that sip register before we get the 200 OK |
| 03:39.13 | file | is your DNS ok? |
| 03:39.17 | Nethab | mine? |
| 03:39.23 | file | yes |
| 03:39.23 | newmedian | Okay; here's another idea. Since there are so "few" providers, now might be a good time for an enterprising individual to make something like an Internet Traffic Report for VOIP providers, re: registration times, latency, etc. |
| 03:39.27 | knapjack | Anyone running asterisk on a VIA C3? |
| 03:39.50 | tzanger | knapjack: don't enable p5 optimizations |
| 03:40.06 | Nethab | it's running same as always, although i did get that dnsmgr thing runnung |
| 03:40.13 | file | k |
| 03:40.14 | Nethab | don't know what it does though |
| 03:40.43 | knapjack | I didn't, I'm using asterisk@home, but I get the same illegal operation error message that you'd see from the optimizations. |
| 03:40.51 | Nethab | are you using 1.0? |
| 03:40.55 | knapjack | Yep |
| 03:41.20 | knapjack | I thought it might be the mpg123 code, but setting that module to noload didn't help. |
| 03:41.28 | kb1_kanobe | knapjack: the earlier c3 does not support the CMOV instruction and gcc assumes it does. You cannot run any binary code that was compiled > 486. |
| 03:41.55 | knapjack | Yep, knew that, but *@home compiles everything for i486. |
| 03:42.14 | knapjack | As far as I can tell. |
| 03:42.26 | kb1_kanobe | hmmm... well, that's the source of the problem with the samuel processors. |
| 03:42.36 | knapjack | Yeah, and this is a Sam II. |
| 03:42.55 | kb1_kanobe | ah. check /proc/cpuinfo for cmov support, then I'll shut up. :-) |
| 03:43.24 | Nethab | Son of Sam |
| 03:43.26 | knapjack | I was hoping someone else had run into the problem already. I thought I would look at the build script and see if I can build for i386, but I'm worried there's some assembler hidden in there somewhere. |
| 03:43.40 | knapjack | The Sam, Sam2 and Ezra are all missing the CMOV |
| 03:43.42 | kb1_kanobe | got any g729 licenses installed? |
| 03:45.12 | file | c'mon lemme hear you say! asterisk rocks! |
| 03:45.14 | file | louder! |
| 03:45.16 | file | I said louder! |
| 03:45.17 | *** join/#asterisk slePP (~slepp@S01060040f48412ad.ed.shawcable.net) |
| 03:45.29 | tsp | asterisk woudl work if I could connect ot sipphone |
| 03:45.31 | Nethab | +volume file -2 |
| 03:45.43 | Silik0n | more like |
| 03:45.46 | Silik0n | +mute file |
| 03:45.49 | file | eep |
| 03:48.21 | knapjack | Uh...rocks. |
| 03:48.25 | knapjack | ;) |
| 03:48.58 | Nethab | ln -s * world |
| 03:51.25 | kb1_kanobe | * rulez - all the overtime and high bloodpressure you can take. |
| 03:53.32 | knapjack | Anyone using the same bluetooth headset for their computer and their cell? |
| 03:54.04 | kimo_sabe | knapjack: I have mine paired to my phone and an asterisk box |
| 03:54.18 | knapjack | Which headset are you using? |
| 03:54.34 | kimo_sabe | knapjack: a motorola cute little something |
| 03:54.55 | kimo_sabe | knapjack: HS820 |
| 03:55.42 | Nethab | is there a blutooth -> rj11 reciever? |
| 03:55.44 | knapjack | I'm thinking of buying one for X-Lite. |
| 03:56.36 | kimo_sabe | Nethab: not that I've seen. Why would you want one? |
| 03:56.49 | *** part/#asterisk s9510278 (~bigdaddy2@203.210.152.35) |
| 03:57.06 | Nethab | well I really want a rj11 headphone jack to 1/4" balanced connector |
| 03:57.26 | knapjack | RJ11? |
| 03:57.39 | Nethab | the headset port on polycom cisco etc |
| 03:57.39 | knapjack | RJ11 is a standard phone jack. |
| 03:57.39 | *** join/#asterisk Zebble (~Zebble@HSE-Ottawa-ppp3492547.sympatico.ca) |
| 03:57.45 | Nethab | is rj11 |
| 03:57.49 | knapjack | Ah |
| 03:57.55 | knapjack | Didn't know that |
| 03:57.59 | Nethab | tip/ring in and tip/ring out |
| 03:58.05 | Nethab | two pair |
| 04:03.28 | Zebble | i'm having the strangest problem. I have a TE410P connected to two Atran TA750's and running CVS HEAD dated 2004-11-12. I upgraded Zaptel today to latest CVS HEAD and I get a constant RED alarm on all channels. If I go back to the 2004-11-12 CVS HEAD, it works perfectly again. |
| 04:03.31 | Zebble | Any ideas? |
| 04:03.49 | *** join/#asterisk shmaltz (~chatzilla@69.28.255.210) |
| 04:04.33 | kb1_kanobe | I've heard mention of red alarm problems with zaptel-head for the last few days. Haven't gone near it myself, for obvious reasons. |
| 04:04.40 | *** join/#asterisk lehel (~lehel@82.79.20.17) |
| 04:04.45 | shmaltz | I'm trying to build an rc.zaptel file for slackware, and my machine hangs if I load zaptel twice and then run ztcfg, so I want to first test to make sure that zaptel isn't loaded yet, how would I do that? |
| 04:05.19 | lehel | shmaltz: lsmod |
| 04:05.47 | shmaltz | lehel, i know, but how would I put this into an if command? |
| 04:06.34 | Juggie | doesnt modprobe only load it once? |
| 04:07.11 | *** join/#asterisk MikeJ[Laptop] (~jayden@pcp02795302pcs.roylok01.mi.comcast.net) |
| 04:07.53 | shmaltz | Juggie, could be, but for some reason if I run this script I wrote twice my box hangs |
| 04:09.16 | tsp | anyone using sipphone successfully now? |
| 04:09.57 | *** join/#asterisk heath__ (~heath@12-215-32-56.client.mchsi.com) |
| 04:10.10 | Nethab | still timing out for me |
| 04:10.22 | heath__ | anyone know why dial won't jump to n+1 if it fails? the wiki says it should |
| 04:10.42 | Nethab | it tries +101 if it's there |
| 04:10.53 | Nethab | the n+1 might be a recent thing |
| 04:11.03 | Nethab | what version you havwe |
| 04:11.22 | *** join/#asterisk jskcr|lappy (~jskcr@jskcr.user) |
| 04:11.35 | heath__ | about 2 week old head |
| 04:12.06 | Nethab | heh, 2 week ol... nevermind |
| 04:13.19 | syle | hmmm shmaltz |
| 04:13.32 | syle | i just added lines that finally work in rc.local |
| 04:14.03 | *** join/#asterisk remmo (~roger@smack.isp.net.au) |
| 04:14.19 | syle | modprobe zaptel;sleep 2; modprobe wctdm; sleep 5; /sbin/ztcfg -v; sleep 2; asterisk |
| 04:14.43 | Juggie | use safe_asterisk |
| 04:15.05 | tzanger | safe_asterisk is for pussies |
| 04:15.10 | syle | safe_asterisk go read it, it don;t load the modules |
| 04:15.11 | tzanger | screen |
| 04:15.17 | Juggie | i never said it did |
| 04:15.31 | syle | probably better using /etc/inittab anyways |
| 04:15.33 | Nethab | load what modules |
| 04:17.05 | syle | i always get a frieken cannot load /dev/zap/ctl message when modprobing wctdm all the time, but it does load looking at lsmod, and asterisk loads fine |
| 04:17.15 | syle | strange |
| 04:18.19 | syle | probably just weirdness with fedora core 3 even though i have the devices added in /etc/udev/rules.d/50-udev.rules |
| 04:18.33 | Juggie | syle, you have to wait a few moments |
| 04:18.37 | Juggie | for udev to create the file |
| 04:18.51 | Juggie | its just not handled properly |
| 04:18.52 | Juggie | but it works |
| 04:19.30 | syle | thats comforting juggie thx |
| 04:22.01 | *** join/#asterisk pdracevich (~bob@smtp.aucklandtax.co.nz) |
| 04:22.42 | pdracevich | moring all, Who lives in Auckland New Zealand and know eveything about Asterisk? |
| 04:22.55 | syle | 0 |
| 04:23.04 | pdracevich | 0 ?? |
| 04:23.05 | Nethab | well no one knows everyrhing including mark |
| 04:23.16 | pdracevich | hummm, ok |
| 04:23.31 | pdracevich | moring all, Who lives in Auckland New Zealand and know alot about Asterisk |
| 04:24.01 | remmo | i live in aust |
| 04:24.12 | remmo | might be the closest you will get |
| 04:24.20 | pdracevich | *sigh* bugger, |
| 04:24.44 | pdracevich | I am looking to hire a person... |
| 04:25.01 | Juggie | NewSole, got 613? |
| 04:25.16 | remmo | good luck, best to outsource |
| 04:25.26 | JamesDotCom | heh, i live in aust and was born in auckland |
| 04:25.28 | JamesDotCom | i'm closer :D |
| 04:25.30 | newmedian | pdracevich: Look at http://www.sipserve.co.nz/ and talk to Hamish |
| 04:25.35 | heath__ | bwahahahaha i figured it out |
| 04:26.42 | heath__ | I had the timeout in the .call file smaller than the Dial timeout, asterisk was cancelling the call |
| 04:26.46 | *** join/#asterisk CdtDelta (~CdtDelta@dsl081-225-161.chi1.dsl.speakeasy.net) |
| 04:27.46 | syle | does uniden make the only ip cordless phone? |
| 04:27.50 | syle | anyone have one of those? |
| 04:33.13 | knapjack | Well, -march=i486 -Os and I get a different error message. |
| 04:33.15 | knapjack | Dang. |
| 04:33.36 | knapjack | Ouch ... error while writing audio data: : Broken pipe |
| 04:34.09 | jayk2 | <PROTECTED> |
| 04:34.14 | knapjack | Looks like it's time to dig the old PII out of the garage. |
| 04:39.48 | *** join/#asterisk sandnigg0r (~niggerplz@66-55-197-254.gwi.net) |
| 04:42.52 | Nethab | i use a PII 450 as my * server |
| 04:43.10 | *** join/#asterisk techie (gus@asterisk.horizonte.us) |
| 04:44.36 | Smi|k | did broadvoice provide call forwarding while their service was down? |
| 04:45.16 | Smi|k | I wonder if I can get reliable business voip by using 2 providers having each forward to the other when down or to capacity for incoming calls? |
| 04:47.48 | Nethab | i set up call forwarding with them |
| 04:47.50 | Nethab | but it didn't work |
| 04:48.04 | Nethab | it's supposed to forward when busy or unavailable |
| 04:48.07 | Nethab | but nothing happened |
| 04:48.09 | Nethab | dead air |
| 04:49.36 | |Vulture| | Smi|k: I think so.. they provided voicemail as well |
| 04:50.00 | Nethab | they say they do, but it didn't work today |
| 04:50.58 | Smi|k | urgh |
| 04:51.06 | Smi|k | maybe regular phone lines are best for incoming |
| 04:51.15 | Smi|k | then a data T1 for outgoing linked to multiple providers |
| 04:51.29 | pdracevich | Who knows about DUNDi? |
| 04:52.12 | hardwire | not me said the me |
| 04:52.34 | NewSole | lol |
| 04:54.38 | *** join/#asterisk iq|laptop (~iq@63-230-47-122.omah.qwest.net) |
| 04:57.28 | tsp | anyone using fwd? |
| 04:57.52 | Nethab | yes |
| 04:57.56 | Nethab | i use them all |
| 04:57.56 | tsp | is it registering? |
| 04:57.58 | Nethab | muhahaha |
| 04:58.43 | Nethab | lemme see |
| 04:59.02 | newmedian | My FWD registered ok an hour ago. |
| 04:59.43 | tsp | i'm trying to use sip |
| 04:59.49 | iswm | I'm talking over FWD right now. Working fine |
| 05:00.14 | Nethab | i register with sipphone and fwd every 15 minutes |
| 05:00.26 | tsp | iswm: why can't I register worth a damn then? |
| 05:00.55 | *** join/#asterisk Inv_arp (junya@adsl-3-244-116.mia.bellsouth.net) |
| 05:00.56 | iswm | tsp, how long ago did you register. |
| 05:01.20 | Smi|k | anyone played with the use of dynamically generated webpages with phone numbers local to the viewer based on IP of viewer VS. 800# |
| 05:01.23 | tsp | I"m just using asterisk as a voip client. |
| 05:01.26 | tsp | no proxy etc. |
| 05:01.35 | Smi|k | to see if it increases the phone calls as people feel the number is a local company |
| 05:01.45 | tsp | peer "fwd" is unreachable! |
| 05:02.15 | tsp | I keep getting reqno 102 errors etc |
| 05:04.12 | newmedian | I use IAX2 to FWD from Asterisk. Is registering fine regularly. |
| 05:04.30 | *** join/#asterisk peted20 (~chatzilla@d30-7.rb.gh.centurytel.net) |
| 05:04.47 | tsp | i'm using a nat here... with no control over it |
| 05:05.03 | Nethab | are you using qualify |
| 05:05.28 | tsp | yeah qualify=yes |
| 05:05.29 | Nethab | don't use qualify, it'll only make things worse |
| 05:05.33 | tsp | why? |
| 05:06.02 | Nethab | cause qualify will convince * they are unreachable if they are just late on their packets returrn trip |
| 05:06.28 | tsp | ah. |
| 05:06.32 | tsp | it still timed out |
| 05:06.41 | tsp | well... gonna try iaxtel |
| 05:06.42 | Smi|k | is iax.cc reliable for incoming calls? |
| 05:06.52 | Nethab | iaxtel is a known problem service |
| 05:06.59 | Nethab | they have too many registered users |
| 05:07.22 | lehel | i'm using zap.. any opinions about zap? |
| 05:07.35 | newmedian | I had to stop using iaxtel because registration kept timing out. |
| 05:07.57 | jskcr|lappy | anyone here for any suggestions of third party software to add to a asterisk live cd Im working on, AMP is already part of it |
| 05:08.17 | tsp | what should I do then? |
| 05:09.31 | newmedian | You can do anything, at www.zombo.com |
| 05:09.53 | tsp | zombo isn't evena page |
| 05:10.10 | jskcr|lappy | anyone? |
| 05:10.33 | tsp | oh its a flash movie and my browser dont support it |
| 05:12.14 | *** join/#asterisk rue_mohr (~mohr@d154-20-50-233.bchsia.telus.net) |
| 05:12.23 | tsp | damnit |
| 05:12.36 | rue_mohr | kb1_kanobe greetings |
| 05:12.43 | newmedian | not important, tsp. |
| 05:13.06 | newmedian | ... ... you're welcome, pdracevich. ... ... |
| 05:13.18 | kb1_kanobe | rue_mohr: evening. hows tricks? |
| 05:14.01 | rue_mohr | I just plugged in my rockwell pci 56K, I think I should be able to get it working as an answering machine... |
| 05:14.03 | rue_mohr | :) |
| 05:14.46 | rue_mohr | kb1_kanobe you playing with phones or budget ? |
| 05:14.49 | kb1_kanobe | You've come to the right place then: apt-get asterisk |
| 05:14.55 | kb1_kanobe | phones, what else. :-) |
| 05:15.09 | tzanger | eww |
| 05:15.14 | tzanger | dbnm |
| 05:15.15 | tzanger | er |
| 05:15.17 | tzanger | debain |
| 05:15.19 | tzanger | ]GNU/fuck you |
| 05:15.41 | kb1_kanobe | tzanger has been enjoying himself this evening, please excuse him. |
| 05:15.44 | jskcr|lappy | also checkout rarewares.org for lame .deb |
| 05:15.59 | rue_mohr | kb1_kanobe tracked down the dropped audio a little more? |
| 05:16.00 | tzanger | hhhhhhhhhhhhhhhhhajfha |
| 05:16.16 | rue_mohr | kb1_kanobe the wonder of /ignore |
| 05:16.22 | kb1_kanobe | rue_mohr: nope. waiting to hear back from digium. it's beyond me. |
| 05:16.42 | kb1_kanobe | rue_mohr: tzanger's fine, just a little drunk. |
| 05:16.43 | rue_mohr | :( |
| 05:16.52 | rue_mohr | hehe |
| 05:17.16 | tsp | why doesn't fwd work worth a shit? |
| 05:17.24 | newmedian | it works fine. |
| 05:17.27 | *** join/#asterisk PTG1234 (~PTG123@ip68-106-24-139.ph.ph.cox.net) |
| 05:17.51 | tsp | i can't use sip.conf to connect to it |
| 05:18.05 | tsp | it keeps saying registration timeout |
| 05:18.26 | syle | anyone stoned? |
| 05:18.43 | jskcr|lappy | drugs are bad! Mkay |
| 05:18.49 | jskcr|lappy | ask Mr Makey |
| 05:18.58 | tsp | damn fwd |
| 05:19.17 | newmedian | Parenthetically, I found recently that some NAT implementations are too anal (perhaps under-implemented?) in ADSL modem/routers like the speedstream line; Converting them to a Modem instead of a Modem+Router and using a different/real router works much better. |
| 05:19.20 | syle | life is a drug |
| 05:19.32 | tsp | I can't receive any incoming connections at all period |
| 05:21.05 | rue_mohr | hmm there's cute... /proc/pci entry for modem has no IO... |
| 05:21.37 | jskcr|lappy | the ds-linux project got linux to boot on a nintendo ds, its on slashdot |
| 05:21.48 | newmedian | oooh, scary. was just reading that. |
| 05:22.10 | newmedian | was thinking Asterisk on DS. hmm. |
| 05:22.32 | jskcr|lappy | lol |
| 05:31.00 | Vco | why not go a step further and get a netbsd and * install going so you can run a pbx on your calculator watch? |
| 05:31.13 | Vco | i medan you *would* already have a dialpad et'all |
| 05:31.16 | jskcr|lappy | already did that |
| 05:32.36 | Vco | every time i hear some "insensitive clod" comment i have this insane urge to bitchslap someone |
| 05:33.01 | pbx123 | hello, can anyone pls help me with this question? - lets say i purchase 5 licenses for codec G729 for now. Will it be possible to purchase additional licenses in the future and be incorporated on the same asterisk box? |
| 05:33.43 | jskcr|lappy | yes |
| 05:34.31 | pbx123 | is that an answer for me? lappy? |
| 05:35.00 | jskcr|lappy | yup |
| 05:35.27 | pbx123 | ok thank you very much dude :) |
| 05:36.36 | pbx123 | is there some kind of "free" alternative for this? instead of purchasing the codec G729? :) |
| 05:37.09 | pbx123 | but with smaller bandwith too |
| 05:37.38 | jskcr|lappy | there is speex |
| 05:38.26 | pbx123 | what is that man? is there a link i can look at? |
| 05:39.29 | jskcr|lappy | http://www.voip-info.org/wiki-Codecs |
| 05:40.07 | *** join/#asterisk jedaustin (~chatzilla@host4.twingeckos.net) |
| 05:40.24 | jedaustin | Sigh.. Broadvoice has been down all day :( |
| 05:41.06 | Moonwick | down for everyone? |
| 05:41.31 | Nethab | did you read the article on slashdot |
| 05:41.40 | jedaustin | Dunno, several people complaining about it on the asterisk mailing list. All BV will say is "We are currently experiencing in-bound call issues with a carrier partner in some areas. We are aware of the issue and our engineers are working to have it resolved as soon as possible." |
| 05:41.45 | Nethab | about the DoS attacks against VoIP companies |
| 05:42.02 | jedaustin | why would someone do that? |
| 05:42.10 | rue_mohr | kb1_kanobe hmm, I have to be in bed in -13 minutes, at what multiple of the speed of light must I travel to make it in time? |
| 05:42.12 | Nethab | don't know, I'm reading an article |
| 05:42.17 | rue_mohr | :) |
| 05:42.50 | pbx123 | jskcr|lappy, what do you recommend that I use? |
| 05:43.55 | jskcr|lappy | well how much b/w ya got? |
| 05:44.13 | pbx123 | 512Kbps DSL |
| 05:44.37 | jskcr|lappy | hmm how many concurrent calls? |
| 05:45.05 | pbx123 | were planning for 24 |
| 05:45.29 | jskcr|lappy | on 512 dsl? |
| 05:45.33 | Nethab | T1 can do 24 simul |
| 05:45.41 | Nethab | 512 dsl won't |
| 05:45.46 | jskcr|lappy | or a pri |
| 05:45.57 | jedaustin | So other than Vonage, what voip companys are worth a damn? |
| 05:45.58 | syle | T1 is 23 isn;t it? 1 line for pri |
| 05:46.12 | jskcr|lappy | 23B + 1D |
| 05:46.13 | Nethab | well, t1 data |
| 05:46.14 | *** join/#asterisk FuriousGeorge (~brian@ool-43516aa2.dyn.optonline.net) |
| 05:46.19 | pbx123 | yeah thats why we want codec :( |
| 05:46.21 | FuriousGeorge | hi all |
| 05:46.26 | jskcr|lappy | hya george |
| 05:46.28 | Nethab | hello furious |
| 05:47.10 | pbx123 | so i have no choice but purchase the G729 huh? lappy? |
| 05:48.16 | alt | pbx123: if I was doing that link, I'd only try and put 15 calls through. |
| 05:48.23 | alt | on g729 |
| 05:48.44 | alt | 5 on G711 |
| 05:49.14 | FuriousGeorge | i was just wondering: that softphone frome xlite suppots IM and video chat. is that p2p? and independant of * ? |
| 05:49.31 | FuriousGeorge | eyebeam, i think it is |
| 05:49.31 | Nethab | it's part of the SIP spec |
| 05:49.34 | alt | FuriousGeorge: SIP by definition can be P2P |
| 05:49.37 | Nethab | instant messaging |
| 05:49.41 | *** part/#asterisk grolloj (~grolloj@dsl254-116-106.nyc1.dsl.speakeasy.net) |
| 05:49.57 | Nethab | my polycom has instant messaging too |
| 05:50.04 | alt | I wish my Cisco did that :( |
| 05:50.16 | alt | I tihnk I'm going to recommend polycom SIP clients. |
| 05:50.36 | Nethab | i just got mine yesterday, i like it |
| 05:50.43 | alt | you use Cisco 7940s? |
| 05:50.44 | Nethab | especially the full duplex speakerphone |
| 05:50.50 | Nethab | polycom |
| 05:51.01 | FuriousGeorge | so its actually all about configuring it with the * box, then. |
| 05:51.02 | alt | have you used 7940s as well? |
| 05:51.14 | Nethab | no, not yet |
| 05:51.19 | alt | I'm looking for a comparison, that's all. |
| 05:51.39 | alt | we're in the market for new phones. our loaners are going to go away :\ |
| 05:52.11 | alt | well, that's cool. Now I have a new reserved extension called 'r' that voicemail can call if the user presses '1' :) |
| 05:52.35 | Nethab | i use *1 |
| 05:52.35 | Nethab | but that's jut me |
| 05:52.35 | alt | from voicemail? |
| 05:52.39 | alt | oh, I see |
| 05:52.41 | Nethab | to call voicemail |
| 05:52.45 | alt | no |
| 05:53.08 | alt | this is when you call someone and get voicemail, it presents the option to press '1' to redirect to another number seperate from the operator. |
| 05:53.15 | Nethab | i have my * setup so when an extension dials *1 it connects directly to their voicemail box |
| 05:53.38 | Nethab | oh i see |
| 05:53.40 | alt | it's for the calling party, not the local user. |
| 05:53.59 | alt | it was a special request. |
| 05:54.29 | alt | I was just going to use *1, but there's a large lag between when you press '*' and when you can then press '1' |
| 05:54.40 | Nethab | how come? |
| 05:54.53 | alt | how come they wanted that? |
| 05:54.58 | alt | or home come there's a lag? |
| 05:55.00 | Nethab | how come there's lag after * |
| 05:55.03 | alt | s/home/how/ |
| 05:55.14 | alt | dunno |
| 05:55.23 | alt | probably because it's on a P3-1GHz machine. |
| 05:56.08 | jedaustin | anyone tried ryantech ? |
| 05:57.05 | alt | well, nice yaking with you guys, but my carcass has to get into bed because it has to wake up at 6am :( |
| 05:57.48 | `Sauron | Wimp |
| 06:04.23 | tzafrir_laptop | what is the recommended sip/iax soft-phone for win32? this needs to be a quiet "non intrusive" program that sits in the tray unless requested |
| 06:04.37 | tzafrir_laptop | (and free. preferably not only beer) |
| 06:04.45 | Mavvie | tralala. |
| 06:05.05 | `Sauron | Mavs |
| 06:05.10 | Mavvie | hi `Sauron |
| 06:05.16 | Mavvie | all okay on that side of the world? |
| 06:05.19 | tzafrir_laptop | It sits in the same room with the server, so codem support isn't an issue |
| 06:05.23 | `Sauron | you a postfix or qmail guy? |
| 06:05.31 | Mavvie | I'm a postfix guy. |
| 06:05.34 | `Sauron | Hum. |
| 06:05.40 | `Sauron | Done much virtual transport stuff? |
| 06:05.59 | Mavvie | Wietse was a teacher at my uni, but since I did EE and not CS I never saw him. |
| 06:06.03 | `Sauron | I had a system running at one point, but changed stuff, and now I want to get it back to work, and so far, it's a no-go. :/ |
| 06:06.08 | `Sauron | Hum, cool. |
| 06:06.15 | Mavvie | what's the problem? |
| 06:06.42 | `Sauron | well, it's prolly a ldap question more than virtual :) |
| 06:06.52 | Mavvie | oh, never played with ldap, only with postgresql. |
| 06:06.58 | `Sauron | where's the docs for which %<whatever> things you can use in the search filter |
| 06:10.20 | Mavvie | (as the backend databae) |
| 06:10.20 | tzafrir_laptop | `Sauron, there's also #postfix, you know |
| 06:10.20 | `Sauron | Yeah, yeah, yeah. |
| 06:10.21 | *** join/#asterisk rue_mohr (~dan@sec1d9.dial.uniserve.ca) |
| 06:10.21 | Mavvie | tzafrir_laptop: and this channel is about asterisk, not about W32 clients. |
| 06:10.21 | tzafrir_laptop | clients for * |
| 06:10.22 | *** join/#asterisk gres (~serg@81.222.48.242) |
| 06:10.22 | tzafrir_laptop | (the server that sits in the same room) |
| 06:10.22 | `Sauron | You could always write your own... |
| 06:10.22 | Mavvie | I have in my main.cf: virtual_maps = pgsql:/etc/postfix-dbmail/pgsql-virtual.cf |
| 06:10.22 | `Sauron | What a novel concept. |
| 06:10.22 | rue_mohr | msg kb1_kanobe dont mean to nag..your going to make it home tonight right? |
| 06:10.23 | rue_mohr | oops |
| 06:10.23 | `Sauron | mavs, grep 'search' pgsql-virtual.cf ? |
| 06:10.23 | Mavvie | there is no search in it, only database definitions and select-statement definitions. |
| 06:10.23 | `Sauron | Hum. |
| 06:15.11 | *** join/#asterisk BoRiS (boris@S01060040ca1e5b54.wp.shawcable.net) |
| 06:15.11 | Mavvie | I think that the man page of ldap_table(5) should give you enough informatio |
| 06:15.11 | Mavvie | information. |
| 06:15.11 | `Sauron | Hum. |
| 06:15.12 | `Sauron | Yup |
| 06:15.12 | `Sauron | Thanks :) |
| 06:15.12 | Mavvie | my pleasure |
| 06:15.42 | *** join/#asterisk wildgoose (~edward@xunil.mail.wildgooses.com) |
| 06:29.23 | *** join/#asterisk Fabe (~spamhere@pD95B0BF7.dip0.t-ipconnect.de) |
| 06:30.16 | pbx123 | if i have a te110p which consist of 24 channels t1 and i purchase 5 g729 codec license to digium for now then later on upgrade more license on the same card in the future, is that possible? |
| 06:30.50 | *** join/#asterisk JohnJacob (~JohnJacob@pcp0011542342pcs.mainf01.in.comcast.net) |
| 06:31.04 | BoRiS | Yes... but you have to contact Digium to *merge* the two licenses together. |
| 06:32.31 | *** join/#asterisk MrEntropy (~entropy@101.ade0506.ade.iprimus.net.au) |
| 06:34.32 | MrEntropy | yo |
| 06:34.54 | BoRiS | Yo |
| 06:35.18 | MrEntropy | how can i make two phones ring and get the call redirected to whichever one picks up first? |
| 06:35.32 | Qwell | dial(sip/phone1&sip/phone2) |
| 06:35.38 | BoRiS | Dial(SIP/1000&SIP/1001) |
| 06:35.53 | BoRiS | Qwell beat me. :-p |
| 06:35.57 | Qwell | indeed |
| 06:36.01 | BoRiS | lol |
| 06:36.02 | MrEntropy | cool, thanks |
| 06:36.26 | |Vulture| | thats like * 101 |
| 06:36.29 | Qwell | Yours was case proper though, so... |
| 06:36.32 | |Vulture| | :P |
| 06:38.28 | *** join/#asterisk TheEmperor (~user@202.187.40.2) |
| 06:38.35 | MrEntropy | |Vulture|: yeah but i'm out of touch with it, i've been doing a hell of a lot more with ser |
| 06:38.59 | *** join/#asterisk three55ml (trilluser@cpe-66-25-89-157.satx.res.rr.com) |
| 06:39.03 | *** join/#asterisk Newbie___ (~me@218.111.224.175) |
| 06:39.11 | Newbie___ | hi guys |
| 06:39.21 | |Vulture| | I am just giving you a hard time, I mean I have been using * for a year and tommorow will be my first TDM FXS module install |
| 06:39.21 | MrEntropy | yo |
| 06:39.34 | Newbie___ | exten => s,1,Dial(Zap/g2/${NEXTEL},30,D(w${ARG1}#)) |
| 06:39.37 | Newbie___ | ops |
| 06:39.46 | |Vulture| | ooo nextel :P: |
| 06:40.03 | Newbie___ | kekek |
| 06:40.11 | Newbie___ | exten => s,1,Dial(Zap/g2/$(TIMEGOLD},30,D(w$1#w${ARG1}#)) |
| 06:40.27 | BoRiS | lol...Changing it so quickly? |
| 06:40.32 | |Vulture| | damn thats a lot of variables on the end there |
| 06:40.32 | Qwell | to protect the innocent |
| 06:40.37 | Newbie___ | i am trying to have * dial a 1# after * dial TIMEGOLD, the above seems wrong |
| 06:40.37 | TheEmperor | Hello |
| 06:40.50 | TheEmperor | got a question |
| 06:40.54 | brc_ | ~data |
| 06:42.11 | jbot | Don't Ask To Ask. Just ASK |
| 06:42.11 | brc_ | ~ |
| 06:42.11 | brc_ | ! |
| 06:42.11 | TheEmperor | 2 asterisk boxes connecting to each other, one is dynamic ip and the other is static |
| 06:42.11 | brc_ | where's the bot? |
| 06:42.11 | TheEmperor | how does the dynamic ip asterisk box find it's extensions? |
| 06:42.11 | Newbie___ | exten => s,1,Dial(Zap/g2/$(TIMEGOLD},30,D(w$1#w${ARG1}#)) <-- tying to have * dial 1# after TIMEGOLD, it doesnt seem to work that way, pls help |
| 06:42.12 | TheEmperor | and does the extensions in the dynamic ip box have to be defined in the static ip box's extensions.conf? |
| 06:42.12 | brc_ | that don't make no sense at all |
| 06:42.13 | Newbie___ | TheEmperor: i use no-ip.com to find dynamic ip |
| 06:42.37 | TheEmperor | Newbie__:I've got the dynamic box registering on the static box |
| 06:42.58 | three55ml | Anyone in here in the west coast in the US? |
| 06:43.23 | Newbie___ | hmmmm, i have the same thing too. i just point to static IP |
| 06:43.49 | Newbie___ | host=dynamic ? |
| 06:44.03 | Nethab | so what's the problem |
| 06:44.31 | TheEmperor | problem is, which ip address do i put in the hard phones? |
| 06:44.38 | Nethab | the dynamic box knows the static box because it's IP never changes, and the static box knows the dynamic box because it registers to it |
| 06:44.47 | TheEmperor | understand that part |
| 06:45.01 | TheEmperor | the problem is that the hardphones can't seem to log in |
| 06:45.03 | Nethab | is the dynamic box behind nat |
| 06:46.58 | TheEmperor | no |
| 06:46.58 | Nethab | do you have a dynamic DNS provider |
| 06:46.58 | TheEmperor | well, the dynamic box is on an adsl connection |
| 06:46.58 | TheEmperor | the ip always changes, so that is why i made it register with the static box |
| 06:46.58 | *** join/#asterisk RestLessGemini (~umairbari@202.142.189.86) |
| 06:46.58 | TheEmperor | the issue now is how to get the voip hardphones working |
| 06:46.58 | Nethab | right, and most people use a dynamic DNS so that a specific domain name will resolve to whatever your IP address is at the time |
| 06:47.02 | rue_mohr | three55ml: 11:44pm |
| 06:47.15 | TheEmperor | so my question is: what ip address is put into the ip phones? |
| 06:47.49 | Nethab | you either have to put a domain name |
| 06:47.58 | Nethab | which points to your dynamic ip |
| 06:47.59 | tzafrir_laptop | TheEmperor, use a dynamic dns service? |
| 06:48.05 | Nethab | or the static IP server |
| 06:48.18 | tzafrir_laptop | dyndns.org worked fine for me |
| 06:48.45 | TheEmperor | If I put in the address of the static ip server, do i have to add all the extensions in on the static box in extensions.conf |
| 06:49.02 | TheEmperor | the thing is, none of my clients have fixed ip, all are on dynamic ip, all want their own * box |
| 06:49.09 | Qwell | Thats why you use dynamic DNS |
| 06:49.25 | Nethab | every person wants their own * box? |
| 06:49.27 | Qwell | The hostname always stays the same, and the DNS server knows the new IP |
| 06:49.41 | *** join/#asterisk udppacket (~udppacket@73.155.39-62.rev.gaoland.net) |
| 06:49.49 | TheEmperor | every company I should say.. |
| 06:49.58 | Nethab | if they have a linksys, or other nat firewall, they can register to the private IP of the local server |
| 06:50.05 | Newbie___ | haha, i fixed my own error. damn typo |
| 06:50.13 | Nethab | and then the server can register to your static server |
| 06:50.37 | TheEmperor | what if they don't have a linksys or nat firewal? |
| 06:50.50 | tzafrir_laptop | TheEmperor, what do they have as a nat router? |
| 06:51.05 | tzafrir_laptop | TheEmperor, or is the * computer connected directly to ADSL? |
| 06:51.11 | TheEmperor | they would be using linksys or dlink switches/routers |
| 06:51.31 | TheEmperor | the * server would be connected directly to ADSL |
| 06:51.58 | TheEmperor | do I need to add all the extensions on the static boxes extensions and sip/iax.conf? |
| 06:52.12 | TheEmperor | as well as on the dynamic boxes extensions/iax/sip.conf? |
| 06:52.37 | tzafrir_laptop | TheEmperor, if the * server knows about IP address changes, then this is the simplest case |
| 06:52.41 | Nethab | you use a switch => otherbox to specify extensions in thata context try the other box |
| 06:53.10 | TheEmperor | tzafrir_laptop: the dynamic server registers on the static server |
| 06:53.11 | remmo | where is the docos for the switch statement? |
| 06:53.11 | tzafrir_laptop | TheEmperor, simply register a dyndns address and update it on address change with one of the availble clients |
| 06:53.33 | Nethab | but the phones would have to be register to that box, so they could control their own extensions, which means it needs a static IP (even a fake one) or a dynamic dns |
| 06:53.41 | *** join/#asterisk jskcr|lappy (~jskcr@jskcr.user) |
| 06:53.47 | TheEmperor | tzafrir_laptop: then which address is put into the ip phones? |
| 06:54.06 | Nethab | local phones should go to the local * box |
| 06:54.19 | Nethab | but that * box has to be locatable by IP or dynamic DNS |
| 06:54.35 | Nethab | has to be locatable by the phones |
| 06:54.36 | TheEmperor | what i have done is to put in the static ip address in the ip phones, the phones are registered but can't seem to logon as their sip/iax.conf file is on the dyanmic box |
| 06:54.47 | jskcr|lappy | wooooooooohooooooooooo :) |
| 06:54.47 | tzafrir_laptop | TheEmperor, sorry, I don't get it. To which * server do phones register? Is it in their lan or over the internet? |
| 06:54.52 | Nethab | which means private subnet (192.168.*.* etc) |
| 06:55.04 | Nethab | that's what he wants to know |
| 06:55.23 | jskcr|lappy | just tested my beta of my live asterisk cd and it works it works!!!!! ITS ALIVE!!! ITS ALIVE!!!! |
| 06:55.27 | tzafrir_laptop | But why separate this to two boxes? |
| 06:55.29 | TheEmperor | tzafrir_laptop: yeah, i have a client who wants his own * box but they are on dynamic ip |
| 06:55.42 | Nethab | because his customers want their own * box to control their own phones |
| 06:55.51 | Nethab | but they don't have static IP's |
| 06:55.56 | TheEmperor | Nethab: you are right, spot on |
| 06:56.07 | Nethab | i pay attention, it's a curse |
| 06:56.08 | tzafrir_laptop | TheEmperor, so what are the relevant clients? |
| 06:56.22 | TheEmperor | tzafrir_laptop: what do you mean? |
| 06:56.25 | tzafrir_laptop | TheEmperor, local phones? Internet phones? VoIP providers? |
| 06:56.42 | TheEmperor | tzafrir_laptop: ip phones |
| 06:57.00 | jskcr|lappy | A compete asterisk system with AMP/apache/mysql on a live cd that can also use a hardisk for persistant data |
| 06:57.11 | tzafrir_laptop | TheEmperor, and the ip phones are on the same LAN? |
| 06:57.48 | Nethab | think multiple businesses with adsl dynamic ip's |
| 06:57.58 | Nethab | they all want ip phones and their own * box |
| 06:58.00 | tzafrir_laptop | jskcr|lappy, did you use any compression? (it slows things) |
| 06:58.27 | tzafrir_laptop | jskcr|lappy, I also wonder if you used unionfs? |
| 06:58.39 | jskcr|lappy | I used a compressed loopback system |
| 06:58.49 | jskcr|lappy | Its a hacked up version of knoppix |
| 06:59.03 | Nethab | most companies that want to hosr their own servers mail, dns, etc, will have a private subnet, and a NAT firewall |
| 06:59.07 | tzafrir_laptop | jsharp, I hope you don't run X |
| 06:59.22 | TheEmperor | so what's the best option |
| 06:59.26 | tzafrir_laptop | jsharp, that was for jskcr|lappy |
| 06:59.30 | Nethab | then they can have a static IP on the inside, and a dynamic IP on the outside |
| 06:59.50 | jskcr|lappy | tzafrir: yes and no to configure it you use X and AMP then you can set it to boot into run level 2 |
| 07:00.03 | tzafrir_laptop | jskcr|lappy, which knoppix? 3.8? |
| 07:00.14 | Nethab | if the phones register to your static server, there's no point in them haveing their own * box at their location |
| 07:00.15 | jskcr|lappy | 3.8.1 with a custom 2.6.11 kernel |
| 07:00.21 | *** join/#asterisk K9DI_BSD_WrkStn (~k9bsd@207-246-185-168.EastVillage.ResNet.wiu.edu) |
| 07:00.26 | TheEmperor | Nethab: I see |
| 07:00.28 | Nethab | because their on your server anyeay |
| 07:00.33 | jskcr|lappy | and the latest version off AMP |
| 07:00.58 | TheEmperor | Nethab: so I guess the best option is no-ip.com or dydns to get a kind of fixed ip address for the phones? |
| 07:01.11 | tzafrir_laptop | jskcr|lappy, the next optimization to do would be to work with a non-spmpressed filesystem |
| 07:01.20 | Nethab | or put their server on a private network behind a firewall |
| 07:01.24 | jskcr|lappy | everything is preconfigured, just start it up and bang |
| 07:01.38 | jskcr|lappy | tzafrir: it would have to be dvd based |
| 07:01.39 | Nethab | that's right |
| 07:01.48 | tzafrir_laptop | jskcr|lappy, it would make the cd creation cycle much faster (mkisfs is fast) and you don't really need more than 640MB |
| 07:02.01 | TheEmperor | does dydns provide an actual ip address? |
| 07:02.12 | Qwell | TheEmperor: no, it gives you a hostname |
| 07:02.23 | tzafrir_laptop | TheEmperor, the idea is that you provide the ip, and they provide the dns name |
| 07:02.25 | Qwell | which resolves to your current IP address |
| 07:02.35 | TheEmperor | Qwell: will this work in softphones/ip phones? |
| 07:02.38 | Nethab | it resolves the domain name to whatever IP you tell it to, which is update everytime you get a new IP |
| 07:02.39 | tzafrir_laptop | TheEmperor, your ISP shold give you an IP |
| 07:03.06 | jskcr|lappy | well the data can be stored on the harddisk, so the compressed fs is not a big issue |
| 07:03.14 | TheEmperor | tzafrir_laptop:problem is that fixed ip addresses are expensive and all my clients are on a dynamic ip who all want their own asterisk box |
| 07:03.32 | tzafrir_laptop | TheEmperor, and that's what dyndns is for. |
| 07:03.44 | TheEmperor | no-ip.com same? |
| 07:03.46 | jskcr|lappy | So for a completey setup asterisk system just pop a cd in with a hard disk and your off |
| 07:04.00 | tzafrir_laptop | TheEmperor, to provide a fixed name for a dynamic IP. But read the docs there carefully |
| 07:04.14 | jskcr|lappy | Im releasing it beginning of next week :) |
| 07:04.41 | jskcr|lappy | tzafrir_laptop: any recomendations on other third party asterisk software to put on there besides AMP? |
| 07:04.45 | tzafrir_laptop | jskcr|lappy, Is there an option to back-up you custom configuration? |
| 07:04.46 | TheEmperor | what if the customer does not have a domain name? |
| 07:04.54 | jskcr|lappy | tzafrir_laptop: yup |
| 07:05.07 | Nethab | having a dynamic IP is fine, if you have a nat firewall |
| 07:05.16 | tzafrir_laptop | jskcr|lappy, not sure. I don't really like AMP's concept of being a complete distribution of asterisk |
| 07:05.37 | jskcr|lappy | tzafrir_laptop: it makes setup alot easier. |
| 07:05.37 | TheEmperor | if they just have switches and routers would that work? |
| 07:05.38 | tzafrir_laptop | jskcr|lappy, I have begun working on amportal debs, http://tzafrir.org.il/rapid/ |
| 07:06.10 | Nethab | if they have a DSL router, they already have a NAT firewall |
| 07:06.38 | Nethab | linksys, dlink, etc come with it |
| 07:06.47 | tzafrir_laptop | Nethab, you can actually connect a single computer to a DSL router without any need for NAT. |
| 07:07.02 | TheEmperor | Nethab: so which ip address would i put in the * box and their phones? |
| 07:07.03 | Nethab | not if they have PPPoE |
| 07:07.12 | jskcr|lappy | tzafrir_laptop: that will be awesome, it would be easier to traverse the 7 levels of hell then to install amp on debian |
| 07:07.24 | TheEmperor | Nethab: and what happens if the phones are not located in the office but in another remote location? |
| 07:07.29 | RaYmAn-Bx | Nethab: sure you can..there are plenty of pppoe clients both for windows and linux (and other os's) |
| 07:07.45 | RaYmAn-Bx | gets a bit harder if it's PPPoA, but it's still possible. |
| 07:07.50 | Nethab | like work from home people? |
| 07:07.53 | TheEmperor | yeah |
| 07:09.26 | Nethab | well they would need a static IP or dynamic DNS name |
| 07:09.26 | Nethab | to connect to |
| 07:09.27 | TheEmperor | is there no way to use my static box's ip? |
| 07:09.27 | tzafrir_laptop | anybody here uses a softphone on windows? (/me rephrases) |
| 07:09.27 | Nethab | not without configuring each phone in your server, which would defeat the need for each place to have a * box |
| 07:09.34 | TheEmperor | that's true... |
| 07:09.46 | Nethab | i used X-Lite on windows, until yesterday |
| 07:09.56 | tzafrir_laptop | I wonder how much extra configurations would those phones need if the * box is already called "sip" |
| 07:10.11 | Nethab | no difference that's just a name to them |
| 07:14.11 | TheEmperor | in the ip phones, can one use a domainname instead of a number to register with an * server? |
| 07:16.05 | Nethab | yes |
| 07:17.59 | MrEntropy | does asterisk have any support for some hardware codec dsp cardS? |
| 07:19.51 | Nethab | yes, the T1 line cards, and FXO FXS cards all have DSP's |
| 07:20.01 | Nethab | and asterisk can use them in encoding |
| 07:21.00 | Qwell | dsp? |
| 07:21.05 | Qwell | digital signal processor? |
| 07:21.08 | MrEntropy | yes |
| 07:21.11 | Nethab | to process the voice codecs |
| 07:21.48 | MrEntropy | i don't think it does dood, when i try to dial out in a codec asterisk doesn't have in its software, it drops the call |
| 07:22.08 | MrEntropy | if the codec was on the dsp (and g729 is pretty popular), the dsp would do the transcoding |
| 07:23.22 | MrEntropy | of course i'm assuming you also meant the x100p fxo's |
| 07:24.53 | opus_ | no but it needs a dsp |
| 07:25.51 | MrEntropy | which is why i ask where i can get a dsp for asterisk? |
| 07:26.20 | MrEntropy | i don't want to have to buy the software codec, too much licencing crap. just want a hardware card |
| 07:26.39 | Qwell | You need the license for g729 |
| 07:27.37 | MrEntropy | not if you buy hardware that can transcode in g729, which is what i want |
| 07:28.20 | Romik | did digium compiled g729 with LPC? |
| 07:29.08 | MrEntropy | if i was to have the g729 software transcoder, then yes i do need a licence because i have the capability of unlimited channels. but when you buy a hardware transcoder it has a physical limit, and you pay for that limit and codec when you buy the card |
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| 07:34.36 | opus_ | its all about the offshore transcoding. |
| 07:37.51 | MrEntropy | opus_ hahaha |
| 07:41.16 | jskcr|lappy | ps -ef | grep find |
| 07:41.21 | jskcr|lappy | oopz |
| 07:43.57 | Nethab | well no the codecs aren't in the dsp |
| 07:44.10 | *** join/#asterisk tafazzi (~Dario@eporroj0-1.customer.nettuno.it) |
| 07:44.17 | Nethab | it uses the dsp to process the codecs it has, offload processing |
| 07:45.26 | Nethab | all the 'hardware' that transcodes g729 has the codec in software and uses a dsp to do the work |
| 07:45.42 | Nethab | the software just happens to be in a ROM or firmware |
| 07:46.01 | Nethab | the dsp is generic, or tuned for audio purposes |
| 07:46.21 | Nethab | if they support multiple codecs, they all use the same dsp |
| 07:48.19 | nDuff | tzafrir_laptop, there's a page on voip-info.org listing available softphones. Go read there. |
| 07:48.25 | cjk_ | hi, how can i create a tiff g3 from a pdf or a ps file? any idea? |
| 07:49.56 | nDuff | cjk_, Ghostscript should be able to do that for you. |
| 07:50.17 | Nethab | PS is the best format to start with |
| 07:51.27 | cjk_ | nDuff: anye hint do a manpage or a command? |
| 07:52.08 | cjk_ | nDuff: becaseu ghoscript is installed, just i could not yet find how to create the tiff file. the other way round is working great |
| 07:53.02 | nDuff | cjk_, RTFM -- particularly, "man gs". You want the output format "tiffg3". |
| 07:53.42 | cjk_ | nDuff: thanks, well you did not need the paste the rtfm part. man gs is aready heplfull |
| 07:54.18 | nDuff | cjk_, I imagine it'd be something like "gs -sOutputFile=foo.tiff -sDEVICE=tiffg3 foo.ps" |
| 07:56.47 | opus_ | __PRETTY_FUNCTION__ wtf |
| 07:57.33 | wvbroadband | ~seen jerjer |
| 07:57.57 | jbot | jerjer <~JerJer@DSL-226.206-rt-bras.che.centurytel.net> was last seen on IRC in channel #asterisk, 8d 2h 33m 47s ago, saying: 'try ulaw or gsm'. |
| 07:57.59 | *** join/#asterisk macTijn (martijn@linda.net.insecure.nl) |
| 07:58.08 | opus_ | nethab - i just tried to compile codec_gsm.c into verilog, but it didn't work |
| 07:58.31 | opus_ | sorry |
| 07:58.54 | Nethab | what? |
| 07:59.41 | opus_ | you know, to get the pure dsp |
| 07:59.58 | Nethab | i still don't know what you mean |
| 08:00.06 | opus_ | verilog is used to design chips |
| 08:00.25 | Nethab | ok, and? |
| 08:01.32 | Nethab | a dsp, especially the kind you get in hard phones, and such, are general purpose, their not meant for one particular codec |
| 08:01.53 | opus_ | ever look inside a cisco router? |
| 08:01.54 | Nethab | but the firmware uses them to decode and encode voice data |
| 08:02.16 | Nethab | physically? no, warraty violation and such |
| 08:02.30 | kajtzu | depends on the router :) |
| 08:02.33 | opus_ | oh.. :) |
| 08:02.55 | kajtzu | Nethab: most cisco routers have user replaceable parts inside (the small ones have dram and flash you can replace) |
| 08:03.03 | kajtzu | and the bigger ones are modular |
| 08:03.58 | Nethab | ok, where is this conversation going, and how did we get here |
| 08:04.30 | Nethab | stop the world i wanna get off |
| 08:06.25 | opus_ | they got fpga's |
| 08:06.26 | *** join/#asterisk scanna (~scannachi@81-174-16-211.f5.ngi.it) |
| 08:06.44 | Nethab | ok |
| 08:06.53 | Nethab | field programmable gate arrays |
| 08:07.44 | Nethab | you use them to make processors |
| 08:08.02 | Nethab | or ASIC's |
| 08:08.15 | Nethab | or DSP's |
| 08:13.40 | opus_ | Cisco 7960 has an fpga |
| 08:18.15 | cjk_ | nDuff: even with ghostscript its not working, thanks however |
| 08:23.26 | Nethab | are you sure |
| 08:24.01 | Nethab | cjk_: have you tried ImageMagick |
| 08:24.28 | nDuff | cjk_, "its not working" is pretty general. Ghostscript is definitely capable of what you're trying to use it for, if you use it correctly. |
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| 08:59.10 | cjk_ | nDuff: are you able to transfrom a pdf 2 tiff anf send it via txfax? |
| 09:00.21 | Donuil | hi to all... just a question... can I route through asterisk a call from pstn to any sip phone served by asterisk? I use a digium card with an fxs channel and and an fxo channel... |
| 09:00.49 | cjk_ | Donuil: yes |
| 09:01.44 | jskcr|lappy | yup thats whats the fxo is for |
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| 09:05.22 | Donuil | cjk_ can you tell me how?I create an pstn-incoming context.. and i use exten => _5XXX,1,Goto(default,${EXTEN:1},1)... is it correct? |
| 09:07.59 | ManxPower | ~docs |
| 09:08.00 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org |
| 09:14.57 | Donuil | I founded several examples for out-coming dialiang but related to dial incoming usually suggest to route dial to only one phone... can I configure my exten in order to choise through digits which of my phone calling from a pstn line? |
| 09:15.07 | nDuff | cjk_, I've used ghostscript to transform from pdf to arbitrary file formats before -- and frankly, as long as you can get it out to *something* you can convert to TIFF (even if you have to use a separate tool like ImageMagick to do so). |
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| 09:21.37 | RoyK | morning |
| 09:22.40 | tengulre | afternoon! |
| 09:23.18 | jskcr|lappy | evenin |
| 09:23.32 | RoyK | I try to connect asterisk to another pbx with a te410, and I get these: PRI ERROR we think we're the CPE, but they think they're the CPE too. How can I differenciate between signalling on different ports on a te410p? |
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| 09:27.27 | cjk_ | nDuff: to transform to tiff is no problem. damn im no stupid. but to transform it to a tiff working with txfax is impossible |
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| 09:31.16 | *** mode/#asterisk [+n] by irc.freenode.net |
| 09:31.16 | *** mode/#asterisk [-n] by kram |
| 09:31.16 | *** mode/#asterisk [+n] by ChanServ |
| 09:31.16 | *** mode/#asterisk [-n] by kram |
| 09:31.16 | *** mode/#asterisk [+n] by ChanServ |
| 09:31.45 | jskcr|lappy | hya kram |
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| 09:45.24 | Donuil | I use in out-coming context exten => _0XXXXXX,1,Dial(Zap/g2/${EXTEN}|20,t) in order to call to pstn-line through my zaptel card... I use in my (local) default context exten => 5551,1,Dial(Sip/mysjphone)... How Can I tel from pstn to my sjphone through asterisk supposed that my pstn number is 12345? |
| 09:46.41 | Donuil | sorry tel=dial :( |
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| 09:57.19 | *** join/#asterisk ellvis (~ellvis@adsl-data-141.84-47-13.telecom.sk) |
| 09:57.22 | ellvis | hi people |
| 10:01.01 | Donuil | Is it correct exten => _123455XXX,1,Dial(default,${EXTEN:5:4},1) ? |
| 10:04.55 | *** part/#asterisk tengulre (~tengulre@61.185.238.166) |
| 10:05.33 | *** join/#asterisk agallo (~agx@ip-37-53.sn1.eutelia.it) |
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| 10:08.44 | Donuil | bye to all |
| 10:08.46 | *** part/#asterisk Donuil (x@adsl-ull-30-65.41-151.net24.it) |
| 10:12.10 | *** part/#asterisk tzafrir_home (~tzafrir@station-131.gadot.org.il) |
| 10:15.57 | agallo | hallo, how to use atxfer with *2? it seems not working |
| 10:16.35 | ManxPower | agallo: CVS-HEAD? |
| 10:16.45 | agallo | ManxPower, no 1.0.7 |
| 10:17.04 | ManxPower | agallo: attended transfer using t/T is not in 1.0.x |
| 10:17.23 | ManxPower | Use your phone's attended transfer if you can |
| 10:17.30 | agallo | ManxPower, changelog say its in 1.0.1 |
| 10:17.57 | agallo | well i tried to use CVS (from yesterday night) but running asterisk i get a Sig11 (debian/testing environment) |
| 10:18.16 | nDuff | cjk_, "impossible"? This is software. ghostscript explicitly supports the relevant tiff variants. I don't believe you that it's impossible -- wait a few weeks for me to get to that TODO list item and I'll have a solution for you. |
| 10:18.24 | ManxPower | agallo: I do not see that in the changelog |
| 10:18.55 | ManxPower | nDuff: What is cjk_ trying to do? |
| 10:21.42 | agallo | ManxPower, check the "Asterisk status" page on wiki |
| 10:22.31 | ManxPower | agallo: Check the ChangeLog file in the Asterisk source. |
| 10:22.53 | nDuff | cjk, use ghostscript to create a tiff file printable via txfax. |
| 10:23.00 | nDuff | erm, |
| 10:23.07 | nDuff | ManxPower, cjk is trying to use ghostscript to create a tiff file printable via txfax. |
| 10:23.18 | ManxPower | agallo: there are 75 pages containing asterisk status. Do you have a URL? |
| 10:23.27 | ManxPower | nDuff: you mean like I'm doing with a script I wrote |
| 10:23.39 | agallo | ManxPower, thanks; do you suggestme to checkout the files from CVS with a specific tag to have an almost working system? |
| 10:24.01 | ManxPower | agallo: Why can't you use the transfer features of the phone you have? |
| 10:24.46 | agallo | ManxPower, http://www.voip-info.org/wiki-Asterisk+status |
| 10:25.41 | agallo | ManxPower, i'm using SJPhone on a W2K client machine; if i click the transfer button it always do a direct transfer; everybody needs supervised transfers *g* |
| 10:25.49 | ManxPower | agallo: I STILL don't see the AXFER mentioned |
| 10:26.21 | agallo | ManxPower, no but that page tell that 1.0.3 has fixed a bug in supervised transfer (so i imagine that this features is in) *g* |
| 10:26.55 | ManxPower | I see "bug fixed with supervised transfer" for 1.0.3, but that's normal supervised transfers, not the t/T transfers (which is an ugly hack for phones too stupid or braind dead to have their own transfer feature) |
| 10:27.26 | ManxPower | agallo: 1.0.x supports supervised transfers using the transfer button on the phone. |
| 10:27.38 | ManxPower | It does not support superfixed transfers using DTMF |
| 10:27.46 | ManxPower | superfixed == supervised |
| 10:27.56 | agallo | ManxPower, ouch sorry i didn't know about such difference *g* |
| 10:28.17 | ManxPower | agallo: I've lobbied for over a year to have it much more clear in the docs |
| 10:29.10 | ManxPower | agallo: spend the money on real phones. |
| 10:29.11 | agallo | ManxPower, lol maybe it isn't |
| 10:29.27 | ManxPower | SIPura phones are about $100 and should support supervised transfers with no problems. |
| 10:29.59 | ManxPower | last I heard the grandstream phones do not support supervised transfers |
| 10:30.18 | ManxPower | agallo: or try X-Pro if you MUST use softphones |
| 10:30.21 | agallo | ManxPower, well at the moment i need softphone for Win2k, anybody can suggest me one with that feature? |
| 10:30.37 | ManxPower | agallo: all softphones suck |
| 10:30.51 | agallo | ManxPower, good news *g* |
| 10:31.28 | agallo | ManxPower, well there is something that work or all the VoIP panorama is just a toy? *g* |
| 10:32.03 | ManxPower | agallo: I feel that software phones are just toys. Others disagree with me. |
| 10:32.34 | ManxPower | I have about 80 people using Asterisk at my largest customer. It's not a toy. |
| 10:32.38 | agallo | ManxPower, well i cannot travel around with my notebook and a ethernet phone *g* |
| 10:32.51 | ManxPower | agallo: Why not? |
| 10:33.00 | agallo | ManxPower, i was not tallking about asterisk but about "VoIP Panorama" |
| 10:33.14 | ManxPower | I do not know what "VoIP Panorama" is. |
| 10:33.21 | agallo | ManxPower, lol i've no room for it in my bag :-P |
| 10:33.51 | ManxPower | agallo: then you will have to accept the limitations of a software phone. |
| 10:37.56 | agallo | ManxPower, ok to use the supervised transfer with voip phones (or at least) ata do i need a specific code or it works also with pcmU ? *g* |
| 10:44.35 | agallo | its ok Sipura SPA-841? |
| 10:45.50 | *** join/#asterisk cmk (~cmk_@p54A3BCAA.dip.t-dialin.net) |
| 10:48.28 | nDuff | ManxPower, yup, like that. (post that script on the wiki? I'll prolly be looking to do that myself in a few weeks once my plate clears off) |
| 10:49.10 | *** join/#asterisk Blackvel (~blackvel@dsl-084-057-127-225.arcor-ip.net) |
| 10:49.19 | nDuff | agallo, it doesn't particularly matter what codec you use, no -- the codec impacts how the voice is encoded, but not how the features (like transferring calls) work. |
| 10:51.28 | nDuff | agallo, my site uses SPA-841s; they're excellent phones, except for the speakerphone, which is next to useless. They certainly support supervised transfers. |
| 10:51.52 | *** join/#asterisk ckruetze (HydraIRC@cpc3-cmbg7-5-0-cust100.cmbg.cable.ntl.com) |
| 10:51.55 | nDuff | agallo, that said, we're getting ready to buy a patch of Polycoms for the business-types because they find the (half-duplex) speakerphones on the Sipuras so useless. |
| 10:52.46 | agallo | nDuff, wich model? price? |
| 10:53.09 | nDuff | agallo, SPA-841s. They're the only phones Sipura makes -- or are you asking about the Polycoms? |
| 10:53.54 | *** join/#asterisk langals (~icechat5@196.7.14.183) |
| 10:54.29 | *** join/#asterisk cmk (~cmk_@p54A3BCAA.dip.t-dialin.net) |
| 10:54.58 | agallo | nDuff, yes |
| 10:55.02 | langals | Hi there...I am using Meetme....I am getting a 1 - 2 second delay when using IAX clients...when I was using SIP clients I wasn't getting such a delay...any idea what the problem could be? |
| 10:55.11 | nDuff | agallo, wrt the Polycoms, they're about $300 each for SoundPoint 500s -- at least, that's the price we're getting, but we're local (literally right next door to them until we moved about a month ago) and our people know their people. |
| 10:55.27 | agallo | nDuff, LOL |
| 10:56.06 | nDuff | agallo, for their non-SIP conference phones (which we're connecting via SPA-2100s), we're getting them about $70 each refurb -- but that's a very, very unusually good deal. |
| 10:56.11 | agallo | my ATA has a Call Transfer feature "with consultation" set ON; Feature code "*" Transfer code "#" but i do not know how to use it now *g* |
| 10:56.36 | nDuff | agallo, couldn't tell you myself without checking the manual. |
| 10:57.50 | agallo | they didn't provided it to mee *g* |
| 11:11.09 | *** join/#asterisk zotz (~zotz@24.231.32.109) |
| 11:13.20 | *** join/#asterisk FaithX (~Faithful@202-6-145-116.ip.adam.com.au) |
| 11:13.58 | FaithX | Hey all! |
| 11:14.27 | jskcr|lappy | hya |
| 11:18.01 | *** join/#asterisk dtwilson (~dave@host217-36-121-129.in-addr.btopenworld.com) |
| 11:19.30 | dtwilson | hi all - any of you guys in UK have much experience integrating asterisk as a proxy between isdn30 and norstar systems? |
| 11:20.07 | *** join/#asterisk gambolputty (~gambolput@cblmdm65-221-51-80.buckeye-express.com) |
| 11:21.06 | dtwilson | I've done soem googling and it does seem possible, but am wondering what kind of functionality I'd be left with |
| 11:22.25 | dtwilson | i.e. will I have to forfeit any asterisk pbx functions and leave those to the norstar boxes? |
| 11:23.20 | *** join/#asterisk onkeltimm_ (~chatzilla@213-84-102-203.adsl.xs4all.nl) |
| 11:24.23 | *** join/#asterisk Fabe (~spamhere@pD95B0BF7.dip0.t-ipconnect.de) |
| 11:30.09 | ManxPower | ~docs |
| 11:30.29 | 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 |
| 11:30.29 | ManxPower | ~mailinglist |
| 11:30.41 | jbot | mailinglist is probably Search Asterisk mailing lists by prepending site:lists.digium.com to your Google search. Browse the mailing list archive at http://lists.digium.com/ |
| 11:30.41 | ManxPower | dtwilson: We have telco->Asterisk->nortel |
| 11:30.41 | ManxPower | We require users to get an "outside" line to dial the VoIP extensions |
| 11:31.34 | *** join/#asterisk ennuyeux72 (~ennuyeux7@83.146.53.34) |
| 11:38.26 | langals | ManxPower - hi there....I am directing this to you because you seem to be an expert :-)....I am using Meetme with AIX2 softphones to dial in...when there are 3 people in the conference then the audio is very delayed for 1 participant (about 4 seconds!),...I never got this problem with SIP...any idea what the problem is? |
| 11:43.06 | *** join/#asterisk laotan (~jesse@H38.C18.B96.tor.eicat.ca) |
| 11:47.52 | ManxPower | langals: no. But it's a known issue. The "q" option is supposed to fix the problem (if you are experiencing the problem I'm thinking of) |
| 11:49.08 | Mavvie | somebody here with TDMoIP experience? |
| 11:49.49 | Mavvie | I'm wondering if it has the same full-bandwidth-usage as TDMoE has. |
| 11:49.56 | agallo | ManxPower, thank you it works with my ATA *g* just a little "complicated" to use (i've to put on hold then dial *00 if my collegue wants the call i've to dial #44 and hangup) |
| 11:50.50 | ManxPower | Mavvie: TDMoIP would have similar or more bandwidth usage as IAX2 w/trunking |
| 11:51.08 | ManxPower | (using ulaw/alaw, I would assume) |
| 11:51.49 | Mavvie | Aha, yes I got it here too. |
| 11:52.18 | Mavvie | "TDMoIP bandwidth is not consumed by silent circuits." |
| 11:53.06 | clive- | .mavvie sounds interesting |
| 11:54.16 | Mavvie | with * on it. |
| 11:54.42 | Mavvie | no hassle with all that strange TDMoI stuff. |
| 11:55.26 | langals | ManxPower - where do I put the "q" - is it not something to do with jitterbuffer, etc |
| 11:55.28 | clive- | I heardly ever hear of anyone using TDMoIP |
| 11:56.24 | Mavvie | clive-: I think you will see it more in big bussinesses with large PABXs than here. |
| 11:57.19 | Mavvie | it's like the same companies which have TDM things already to carry SNA and IP and Voice to remote locations. |
| 11:58.47 | ManxPower | langals: when you run MeetMe |
| 12:00.08 | ManxPower | I don't see the advantage of TDMoIP .vs. SIP or other VoIP protocols |
| 12:01.18 | Mavvie | ManxPower: it's a different approach. |
| 12:02.03 | ManxPower | Looks like all the cool topics already have speakers for Astricon EU |
| 12:02.15 | *** join/#asterisk GrahamC (hidden-use@213-131-100-29.onyx.net) |
| 12:02.23 | Mavvie | wake me up when there is an Astricon AU |
| 12:02.56 | ManxPower | Mavvie: I'll wake you up June 15th then |
| 12:03.01 | Mavvie | oops. |
| 12:03.08 | Mavvie | do you have an URL about that? |
| 12:03.40 | ManxPower | http://www.astricon.net/europe/ |
| 12:03.48 | ManxPower | Their adgenda is wrong, however. |
| 12:04.27 | Mavvie | .AU, not .EU |
| 12:06.36 | GrahamC | I'm having a problem getting a Fritz PCI BRI to work using hisax. Attempted to follow Wiki and other docs but am lost. I'm a newbie. Asterisk log shows probs opening and registering /dev/ttyI0, also failure to load module chan_modem.so. Anyone help ? |
| 12:08.00 | *** join/#asterisk tomassia (~elijahcom@ipdial-179-189.tri-isys.com) |
| 12:08.51 | tomassia | CAN ANYONE TELL ME how to fix the unknown RTP codec 72 |
| 12:09.18 | Mavvie | START RECOGNIZING RTP CODEC 72 |
| 12:10.20 | tomassia | how? |
| 12:10.25 | ManxPower | tomassia: You are using X-Lite |
| 12:10.30 | tomassia | yes |
| 12:10.30 | *** join/#asterisk wiz8291 (~dang@kay.arcbox.com) |
| 12:10.54 | wiz8291 | guys |
| 12:10.57 | wiz8291 | how can i listen in on a zap channel to hear whats happening on the line as its rining? |
| 12:11.06 | tomassia | i am using xlite calling thru pstn |
| 12:11.09 | wiz8291 | i.e. i want to try and hear the handshake for call waqiting to check its working |
| 12:11.24 | ManxPower | tomassia: It is apparently a harmless message. |
| 12:11.32 | dtwilson | Manxpower: thanks, I'm intending to initially just use asterisk for siphoning off call data via mysql cdr and a simple agi script for providing customer information screen pops on agent's desktops. The initial deployment won't really be concerned so much with voip - intending to let our agents use their existing hadnsets from the norstar system. Though I'm curious as to whether we'd be able to add asterisk pbx functions piped th |
| 12:11.55 | tomassia | can i just ignore it? |
| 12:12.19 | ManxPower | tomassia: unless it is causing you problems. |
| 12:13.13 | tomassia | is there a way to suppress these messages |
| 12:13.21 | *** join/#asterisk Gh0sty (~Ghosty@ip-81-11-206-236.dsl.scarlet.be) |
| 12:15.18 | ManxPower | tomassia: in X-Lite try setting Transmit Silence = Yes |
| 12:15.30 | ManxPower | As talked about in http://lists.digium.com/pipermail/asterisk-users/2005-February/090097.html |
| 12:15.51 | ManxPower | But you knew that already since you searched the mailing list archive via Google, right? |
| 12:16.02 | tomassia | ok, thanks a lot |
| 12:17.58 | key2 | Is there any DSP card for asterisk for not overloading the CPU when it does echo cancelling ? |
| 12:19.10 | clive- | key2 there is one in development asaik called ipvolution |
| 12:19.27 | clive- | afaik |
| 12:25.22 | *** join/#asterisk bjohnson (~bjohnson@jecinc.tor.istop.com) |
| 12:26.48 | ManxPower | I paid a company to write my Resume/CV. Gads, the damn thing has so many buzzwords I'm afraid it will colapse into a black hole og buzzwordieness, complete with an event horizon of orbiting managers. |
| 12:27.14 | key2 | someone has tryed X-PRO for pocketpc here/ |
| 12:27.15 | key2 | ? |
| 12:27.36 | Mavvie | ManxPower: you're old fashioned. A CV should describe what the people are looking for, not what you can offer. |
| 12:28.37 | ManxPower | Mavvie: THAT'S why I paid a company to write it. I could never put enough buzzwords into a resume. |
| 12:29.04 | ManxPower | I think I'll do a technical resume/cv myself. |
| 12:29.38 | *** join/#asterisk Rick_Hunter (~rhunter@01-186.008.popsite.net) |
| 12:29.43 | ManxPower | "4) I understand that HR and managers have an unhealthy fetish for buzzwords, but as a computer geek, I hate them." |
| 12:33.51 | *** join/#asterisk adjacent (~scott@nc-65-40-81-64.sta.sprint-hsd.net) |
| 12:36.16 | *** join/#asterisk Aze` (~aze@host229-162.pool80105.interbusiness.it) |
| 12:36.16 | *** join/#asterisk MikeJ[Laptop] (~ircatjerr@mi.origenfinancial.com) |
| 12:42.07 | jskcr|lappy | Max: That and they have a love for turing everything into three letter acronyms |
| 12:42.31 | ManxPower | Yeah. |
| 12:42.32 | jskcr|lappy | IE: The blc as related to the tco and our current tpp |
| 12:42.46 | ManxPower | away too |
| 12:43.16 | jskcr|lappy | kram alive? |
| 12:43.31 | ManxPower | he's never alive anymore. |
| 12:43.33 | jskcr|lappy | guess not |
| 12:43.59 | jskcr|lappy | lol |
| 12:43.59 | jskcr|lappy | I'm comming very close to the finnish line on the asterix live cd im working on :) |
| 12:44.35 | newl | And how do the Finns feel about this Asterix proximity? |
| 12:45.31 | dtwilson | I'm sure they're sniggering their socks off - aren't the finns on of the most advanced countries in the world when it comes to telecoms? |
| 12:45.31 | *** join/#asterisk makhtar (~ageller@mail.bulletinnews.com) |
| 12:45.42 | jskcr|lappy | so far this week ive put about 90 hours into building it |
| 12:46.30 | MikeJ[Laptop] | kram was alive and kicking just fine yesterday |
| 12:47.21 | *** part/#asterisk GrahamC (hidden-use@213-131-100-29.onyx.net) |
| 12:47.23 | *** part/#asterisk queuetue (~Scott@h69-21-252-54.69-21.unk.tds.net) |
| 12:48.23 | jskcr|lappy | 10 bucks for a asterisk live cd, knopsterisk can byte my shiney metal ass. My live cd blows theres to shit and its free |
| 12:48.29 | *** join/#asterisk stefanocarlini (~stefano@host-84-222-61-241.cust-adsl.tiscali.it) |
| 12:48.40 | stefanocarlini | hello |
| 12:48.54 | *** join/#asterisk shido6 (~greg@d57-87-253.home.cgocable.net) |
| 12:49.01 | stefanocarlini | Can I do videoconferencing SIP using Asterisk? |
| 12:49.58 | *** join/#asterisk ariel_ (~Ariel@adsl-068-157-125-248.sip.mia.bellsouth.net) |
| 12:50.08 | shido6 | yes |
| 12:50.51 | stefanocarlini | Ok. Do you can tell me when I can find docs or tutorial about it? |
| 12:51.19 | *** join/#asterisk HeadachesAbound (HeadachesA@wsip-68-99-73-32.tu.ok.cox.net) |
| 12:51.43 | shido6 | ~docs |
| 12:51.44 | 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 |
| 12:52.16 | dtwilson | jskcr: is your live CD debian based? |
| 12:52.18 | *** join/#asterisk katrat (~johnr@gir.dragonsbreath.org) |
| 12:52.24 | jskcr|lappy | yup |
| 12:52.39 | jskcr|lappy | with the latest version of AMP |
| 12:52.58 | stefanocarlini | yes I looked at it but I didn't find nothing about videoconferencing |
| 12:53.01 | *** part/#asterisk langals (~icechat5@196.7.14.183) |
| 12:53.08 | dtwilson | can it detect E1 cards? |
| 12:53.28 | jskcr|lappy | with 2.6.11 kernel support and all the zaptel modules |
| 12:53.30 | jskcr|lappy | yup |
| 12:53.55 | shido6 | set up your e1 card manually |
| 12:53.59 | dtwilson | :) - whats it called? |
| 12:54.12 | jskcr|lappy | kasterx |
| 12:54.16 | jskcr|lappy | It will go live on monday |
| 12:54.28 | dtwilson | excellent - I'll prolly have a wee play around with it |
| 12:54.41 | dtwilson | have to get a PRI lab environment going first though |
| 12:55.08 | jskcr|lappy | Im just finishing up the persistant hard disk support. |
| 12:55.10 | dtwilson | my experience has been limited to X101's on POTS |
| 12:55.14 | dtwilson | to date |
| 12:55.40 | jskcr|lappy | That and the partition backup tool for the data. |
| 12:55.52 | *** join/#asterisk Patrick^ (~patrickm@birch4.mountaincable.net) |
| 12:56.47 | dtwilson | what's the footprint size for standard install? roughly |
| 12:57.15 | *** join/#asterisk zotz (~zotz@24.231.32.109) |
| 12:57.15 | shido6 | ? |
| 12:57.22 | jskcr|lappy | it all runs off the cd and config and vmail files are stored on the drive |
| 12:58.03 | dtwilson | ahhh - my bad - didnt register the "live CD" bit there :) |
| 12:58.24 | jskcr|lappy | the config files can also be saved to a floppy. |
| 12:58.38 | jskcr|lappy | server dies drop the cd in another machine with the floppy and your back up |
| 13:00.08 | dtwilson | yeah i can appreciate that - I'm looking for something that will work on small form factor hardware though - i.e. no room for CD or floppy drives |
| 13:00.33 | jskcr|lappy | dtwilson the cf version will come in a few weeks |
| 13:00.41 | dtwilson | :D |
| 13:02.29 | dtwilson | I'll keep an eye out for that then |
| 13:04.29 | jskcr|lappy | dtwilson, what type of small form factor like mini-itx? |
| 13:05.16 | *** join/#asterisk jonas (jonas@jonas.probe-networks.host.freenode) |
| 13:05.55 | dtwilson | jskcr: yeah I was thinking along the lines of mini-itx |
| 13:06.33 | jskcr|lappy | dtwilson: send me nice mini-itx system and ill build ya one :) |
| 13:06.41 | dtwilson | haven't really thought too much about it - just got back from initial client visit yesterday and noticed they have extremely limited space where their PRI coms in |
| 13:07.23 | jskcr|lappy | that is extremely limited space if it can only fit mini-itx |
| 13:07.29 | dtwilson | I'm doing a simple proxy of telco -> asterisk -> norstar to begin with |
| 13:08.32 | dtwilson | well it's not so much the space is that limited - it's just that the line and norstar are on a wall of a fairly well used narrow corridor in a car dealership parts store |
| 13:08.35 | jskcr|lappy | 24 port system? |
| 13:08.53 | dtwilson | yup |
| 13:09.22 | jskcr|lappy | what is it a Nortel CICS? |
| 13:09.27 | jskcr|lappy | or MICS? |
| 13:10.00 | bjohnson | telco -> asterisk -> norstar is not simpler than telco -> asterisk -> phones |
| 13:10.17 | dtwilson | I'm not 100% sure - I didn't take down the details as it was only a prelim meeting and telephony only a tiny fraction of what we are looking at |
| 13:10.26 | jskcr|lappy | how many stations 24? |
| 13:10.29 | *** join/#asterisk lately (~doug@80.4.233.220.exetel.com.au) |
| 13:11.08 | dtwilson | bjohnson: yeah i realise that, but don't want to alienate the client by telling them they'll have to upgrade/change/re-invest in their existing infrastructure |
| 13:11.19 | dtwilson | 24 at this one site |
| 13:11.24 | dtwilson | the have over 20 sites though |
| 13:11.28 | jskcr|lappy | well your in for a suprise then |
| 13:12.07 | jskcr|lappy | That sounds like its a Nortel CICS, which mean 8 lines, 24 stations |
| 13:12.21 | dtwilson | heh - that's why I'm in here atm :) - to find out as much as possible what I could be faced with :) |
| 13:12.41 | lately | I want to know why on one provider using IAX2 I need to send username+password in the dial function whereas another provider using SIP I don't have to. eg Dial(SIP/${EXTEN}@provider) and Dial(IAX2/username:pass@provider/${EXTEN}). I've tried putting the username in the iax.conf bit. Am I doing something wrong? |
| 13:12.51 | jskcr|lappy | that means your capped at 8 lines with the CICS |
| 13:13.16 | *** join/#asterisk mjrece (~mjreeves@pcp05929941pcs.vnburn01.mi.comcast.net) |
| 13:13.39 | bjohnson | lately: yes |
| 13:13.47 | jskcr|lappy | a pri would be wasting 16 lines |
| 13:14.07 | bjohnson | or more |
| 13:14.20 | bjohnson | maybe a partial PRI? |
| 13:14.33 | jskcr|lappy | no the nortel CICS maxes at 8 lines, 24 ports |
| 13:14.38 | bjohnson | unless they want to expand their capability |
| 13:14.56 | bjohnson | yes .. maxes out .. they could be using less |
| 13:15.04 | dtwilson | perhaps I've been misinformed - their "IT manager" isn't extremely literate and was very vague about what exactly they have etc |
| 13:15.14 | bjohnson | which = wasting more than 16 PRI channels |
| 13:15.23 | lately | bjohnson: Hrmm. And? I tried removing the user:pass and putting Dial(IAX2/${EXTEN}@provider) and making sure the username/pass is in the iax.conf. Why can't it work? |
| 13:15.37 | bjohnson | lately: you've got something wrong |
| 13:15.57 | bjohnson | lately: I don't know what .. that wasn't the initial question |
| 13:16.47 | dtwilson | well - for the time being my objectives aren't so much to do with telephony - more to do with CTI - logging all calls with mys_cdr and a simple AGI script to furnish customer details screen popping on internal desktops |
| 13:17.27 | lately | Well. Can someone please explain what the differences are when putting the username password in Dial() and putting it in the iax.conf provider bit? |
| 13:18.10 | jskcr|lappy | dtwilson: be afraid of IT mangers that a vague, they can really fuckup your day lol |
| 13:18.17 | jskcr|lappy | s/a /are :) |
| 13:18.54 | *** join/#asterisk _omer (dfsdf@202.147.174.177) |
| 13:19.17 | dtwilson | jskcr: heh and don't I know it :) |
| 13:19.40 | _omer | any QUINTUM expert ? |
| 13:19.41 | bjohnson | lately: check your iax.conf section and look for things that could screw it up. most providers have an example entry to follow |
| 13:20.21 | *** join/#asterisk znoG (gs@200.115.216.109) |
| 13:20.54 | lately | bjohnson: The provider's doco says to put the user:pass bit in the Dial function. I don't understand why this is necessary. |
| 13:20.57 | jskcr|lappy | probably just a mechanic who works there who knows how to reboot stuff |
| 13:21.02 | dtwilson | lately: I don't think there's any difference as to where you specify the user:pass per se, however I believe keeping it in the iax.conf helps in keeping your configs a bit tidier if anything |
| 13:21.03 | *** join/#asterisk kenshinblade (~maccheron@host37-9.pool80105.interbusiness.it) |
| 13:21.09 | bjohnson | lately: it isn't necessary |
| 13:21.38 | *** join/#asterisk dmccollum (~dmccollum@eycb01-00-cntnga-69-164-245-72.atlaga.adelphia.net) |
| 13:21.44 | jskcr|lappy | HeadachesAbound: Im bound by about 1/2 dozen nda's |
| 13:22.06 | lately | bjohnson: I've copied pretty much identically what I have for the other provider which uses SIP. This is why I ask the question. |
| 13:22.18 | kenshinblade | problem: does the "hint" special priority work on extension associated to IAX2 channels? |
| 13:22.23 | jskcr|lappy | HeadachesAbound: my gf is quite used to me refering as jobs as computer stuff lol |
| 13:22.23 | dtwilson | bjohnson: this particular "IT Manager" manages simple data entry jobs and pretty much nothing else. He outsources every ounce of technical work there is |
| 13:22.24 | HeadachesAbound | jskcr : I'm not bound yet...I've avoided many of those. |
| 13:22.33 | ManxPower | kenshinblade: no |
| 13:22.36 | lately | dtwilson: I agree. This is why I would like it changed. |
| 13:22.49 | jskcr|lappy | HeadachesAbound: I can say the companies are listed on the nyse and nasdec |
| 13:22.58 | bjohnson | lately: so is your question .. can somebody please look at my iax.conf file and tell me why I can't get it to authenticate against my provider? |
| 13:23.12 | ManxPower | I quit the last employer that wanted me to sign an NDA. |
| 13:23.13 | jskcr|lappy | three of them are in the fortune 500 |
| 13:23.13 | HeadachesAbound | jskcr : all of my work is computer stuff. I haven't had a day off in months. |
| 13:23.24 | ManxPower | If I had confidence in the company I would have considered signing it. |
| 13:23.45 | kenshinblade | manxpower: great! do you know some workaround to have some signal from those extensions anyway? |
| 13:23.45 | lately | bjohnson: If someone wouldn't mind helping then yes. :-) |
| 13:23.51 | jskcr|lappy | HeadachesAbound: last jobsite I flew out to I had to have a excort to the bathroom and search entering and exiting the facility |
| 13:23.58 | bjohnson | lately: and you've posted it where? |
| 13:24.01 | ManxPower | kenshinblade: Use the Flash Operator Panel |
| 13:24.12 | lately | bjohnson: Getting it organised |
| 13:24.21 | HeadachesAbound | jskcr: I hate it when they escort me to the bathroom. It is just a little disturbing. |
| 13:24.21 | _omer | any QUINTUM expert ? |
| 13:24.22 | ManxPower | bjohnson: I can if I must. |
| 13:24.28 | *** join/#asterisk frogy (edmund@cm218-255-137-80.hkcable.com.hk) |
| 13:24.32 | dtwilson | jskcr: you mean you work for the mob? ;b |
| 13:24.33 | shido6 | oh lord |
| 13:24.37 | shido6 | quintum, eh? |
| 13:24.44 | jskcr|lappy | no but just as bad |
| 13:24.51 | ManxPower | HeadachesAbound: Ask them to "hold it for you". That should stop the escorting to the bathroom. |
| 13:25.01 | kenshinblade | manxpower: does it achieve to monitor even IAX2 extensions? |
| 13:25.13 | ManxPower | kenshinblade: Perhaps you should look at the software. |
| 13:25.22 | clive- | jsharp is quintm any good? |
| 13:25.37 | HeadachesAbound | ManxPower: I did. They weren't amused. |
| 13:25.42 | ManxPower | LOL! |
| 13:26.00 | jsharp | Its pretty decent. We've got a CMS-960 gateway that is pretty bulletproof. |
| 13:26.07 | ManxPower | It might me nice to work for a company that actually takes security seriously. |
| 13:26.17 | kenshinblade | ManxPower: i will. thanks |
| 13:26.30 | jskcr|lappy | The customers for which Im not bound by a nda which I can mention are compusa/walmart/pepsico/symbol |
| 13:26.59 | clive- | jsharp how does it compare to Cisco |
| 13:27.07 | ManxPower | I like the culture of a small company. I like the money of a large company. |
| 13:27.12 | frogy | Any one used * with Cisco 5300 gateway? Are they inter-workable? |
| 13:27.29 | lately | bjohnson: http://www.balmer.id.au/iax.conf |
| 13:27.57 | jsharp | I'd say they're roughly equivalent as far as quality. The 960 is a little quirkier to set up routing on, especially if you have an external route manager. |
| 13:27.59 | _omer | shido6: eh? or ekh ? ;) |
| 13:28.28 | key2 | when I'm on my local network (192.168.x.x) what is the domain name ? |
| 13:28.56 | clive- | jsharp thanks, ..it may be an option for a project I am working on, although I prefer cisco |
| 13:29.18 | jskcr|lappy | ManxPower: I am not fond of large companies very much, too many conference calls, theres a problem when you hold a conference call to schedule a conference call |
| 13:29.28 | frogy | jsharp: Were u used * with Cisco GW? |
| 13:29.36 | laotan | jskcr|lappy: do you have a url for kasterx? |
| 13:29.45 | jsharp | frogy: I have. |
| 13:29.50 | jskcr|lappy | laotan, it will be up on monday |
| 13:29.54 | laotan | ok |
| 13:30.13 | frogy | So they work together? what codecs did u used? |
| 13:30.31 | jskcr|lappy | Im finishing testing it off and doing the stuipid stuff like pretty graphics for it and a asterisk help system |
| 13:30.35 | jsharp | Both G.711 and G.729 |
| 13:30.47 | HeadachesAbound | jskcr: Ever experienced the conference call to schedule a conference call to schedule a conference call? and yes that was literal. |
| 13:30.48 | frogy | quality ok? |
| 13:30.56 | jsharp | Yeah. |
| 13:30.58 | jskcr|lappy | yes |
| 13:31.20 | jsharp | Lets touch base on that synergistic action item! |
| 13:31.42 | frogy | How easy is it to setup the H323 channel? |
| 13:31.58 | ManxPower | frogy: I'd rather have a root canal. |
| 13:32.15 | ManxPower | oh, you don't mean with Asterisk. Nevermind. |
| 13:32.25 | _omer | unfortunately, pull out a report is really a difficult stuff in asterisk |
| 13:32.38 | jsharp | If you follow the README exactly, its not too painful. If you don't, be prepared to pack a lunch & dinner figuring it out. |
| 13:34.25 | _omer | and where do we get that README ? |
| 13:34.55 | jsharp | asterisk/channels/h323/README |
| 13:35.00 | frogy | A client of mine wants to use * as a soft switch to route IDD calls from one end to the other. Any one tried this? |
| 13:36.12 | frogy | firstly, I want to see if it is wise to suggest him to try this on *. |
| 13:37.40 | jsharp | * should switch the call however you send it. |
| 13:37.40 | frogy | Coz' aegular soft switches out there is very costly. So I wanna see if * is a viable alternative? |
| 13:39.16 | *** join/#asterisk iq (~iq@70-59-163-80.omah.qwest.net) |
| 13:39.24 | jsharp | What is he routing the calls from/to? |
| 13:40.15 | frogy | not specified! perhaps from some place on the internet to another. |
| 13:40.32 | jsharp | I don't see why * wouldn't work. |
| 13:40.56 | _omer | my client want reports....which is difficult to pull out from asterisk.. |
| 13:41.15 | jsharp | You can't pull them out of the CDR data * generates? |
| 13:41.36 | ManxPower | _omer: Incorrect. Asterisk has NO report generation features. If you want reports you'll have to generate them from the CDR files using an external application |
| 13:41.49 | *** join/#asterisk VaradeR0 (~mortality@62.38.196.121) |
| 13:41.58 | frogy | Only one doubt, which was the inter-working with Cisco 5300 GW. Most of the IDD player use it. |
| 13:42.28 | jsharp | It should work. I've used 5300s against * in both H323 and SIP mode. |
| 13:42.39 | frogy | any one do IDD business here? |
| 13:42.47 | shido6 | idd? |
| 13:42.49 | ManxPower | frogy: Define "IDD" |
| 13:43.00 | *** join/#asterisk jwitte (~jwitte_@port-212-202-101-206.static.qsc.de) |
| 13:43.01 | *** join/#asterisk Gand_DJ (~gandalf@ptr-207-54-104-24.ptr.terago.ca) |
| 13:43.03 | jsharp | Int'l direct dialing. |
| 13:43.09 | *** join/#asterisk thieumS (~darkmind@bea75-1-82-234-122-35.fbx.proxad.net) |
| 13:43.14 | frogy | international Direct Dial (IDD) |
| 13:43.24 | shido6 | lord |
| 13:43.35 | ManxPower | What's the difference between IDD and regular dialing? |
| 13:43.56 | ManxPower | Or maybe you mean what we call in the USA "DID" or in Germany "DDI" |
| 13:44.39 | frogy | no difference, only the equipment is matters. |
| 13:44.40 | jwitte | Hi, my machine freezes, when unloading wcte11xp -- anybody has the same experiences? |
| 13:44.48 | ManxPower | jwitte: yes. |
| 13:44.59 | *** join/#asterisk MatsK (~matsk@66.80-202-60.nextgentel.com) |
| 13:45.07 | jwitte | ManxPower: nice, and how did You solve it? |
| 13:45.12 | jwitte | :) |
| 13:45.13 | VaradeR0 | May I ask a hardware related question? (digium rather than asterisk)? |
| 13:45.19 | ManxPower | Digium Support said "Use CVS-HEAD Zaptel". |
| 13:45.23 | VaradeR0 | Is it possible to use digium TE410P on a HP DL380? |
| 13:45.24 | ManxPower | jwitte: I have not resolved it. |
| 13:45.28 | Silik0n | quit unloading it |
| 13:45.41 | jwitte | May I use it with asterisk v1-0-7? |
| 13:45.43 | jsharp | VaradeR0: Does the DL380 have a 3.3V PCI slot? |
| 13:45.48 | jwitte | and libpri 1-0-7 |
| 13:45.48 | bkw_ | use cvs head |
| 13:45.51 | bkw_ | NEXT!!! |
| 13:45.51 | ManxPower | Digium's support about such things is amazingly un-helpful. |
| 13:46.05 | bkw_ | ManxPower, in what way? |
| 13:46.11 | ManxPower | jwitte: I only use 1.0.x |
| 13:46.14 | VaradeR0 | jsharp: I don't have a clue. Thus, I would like to ask if someone has experience |
| 13:46.29 | bkw_ | ManxPower, thats your choice to use a fetureless release such as 1.0.7 |
| 13:46.29 | Silik0n | its like microsoft, they'll giuve you a technically correct answer that doesnt really solve the problem |
| 13:46.35 | ManxPower | bkw_: Unhelpful as in "giving workable advice that fixes the problem" |
| 13:46.37 | thieumS | anyone available to help me with a RTP one-way issue ? |
| 13:46.42 | jwitte | ManxPower: and You never unload Your modules? |
| 13:46.45 | bkw_ | ManxPower, whats unhelpful? |
| 13:46.53 | ManxPower | bkw_: I even asked for the bug # that fixed the problem. He could not give it to me. |
| 13:46.53 | Silik0n | thieumS: get rid of the double nat |
| 13:47.04 | ManxPower | jwitte: if I unload the module I am on site. |
| 13:47.06 | bkw_ | ManxPower, hardware issues aren't in the bug tracker |
| 13:47.27 | jsharp | VaradeR0: The DL380 has 3 3.3volt PCI sockets, so the TE410 should work. |
| 13:47.33 | ManxPower | bkw_: then ANYTHING that would help drumkilla fix the problem in 1.0.x Zaptel would have been helpful |
| 13:47.49 | VaradeR0 | jsharp: thanx a lot |
| 13:47.55 | jwitte | ManxPower: thats nice - but my customers are 400km away :) |
| 13:48.04 | frogy | is the chan_dialogic driver still available, and where to get it? |
| 13:48.18 | ManxPower | bkw_: And it's your choice to run a version of asterisk that changes every week and has no testing before patches are added. |
| 13:48.48 | dmccollum | Anyone have a particular server brand and model that they've found works best with asterisk and digium cards? |
| 13:49.11 | ariel_ | even with the correct drivers the dialogic boards don't work well with asterisk. |
| 13:49.32 | *** join/#asterisk tld (~tld@80.203.70.227) |
| 13:49.47 | frogy | asiel_: so have you tried the driver? |
| 13:49.49 | bkw_ | ManxPower, UM dude we are the ones putting 90% of those patches on it |
| 13:49.53 | ariel_ | dmccollum, I have been using the Dell's SC420 without any issues. In fact I am able to put up to 3 TDM400 boards in there system without issues of irq's. |
| 13:49.56 | tld | Any PostgreSQL users in here? |
| 13:50.00 | bkw_ | so shut the fuck up dude.. you spew crap that isn't based on any FACTS |
| 13:50.09 | ManxPower | Digium Support "I do not have that information easily available to me, nor am I 100% |
| 13:50.17 | ManxPower | certain on this. I had recalled someone saying that issue was resolved recently." |
| 13:50.19 | ariel_ | frogy, there bad drivers due to most of dialogic boards are not full duplex. |
| 13:50.38 | ManxPower | bkw_: How many calls per day do you send via CVS-HEAD |
| 13:50.58 | Silik0n | I send tons of calls via CVS-HEAD daily |
| 13:50.59 | bkw_ | THOUSANDS!!!! |
| 13:50.59 | ManxPower | unmodified, CVS-HEAD |
| 13:51.20 | dmccollum | Areil: Thanks! I know a lot of people prefer HP/Compact over Dell, but I've always had good luck with Dell servers. Can't beat their price either. |
| 13:51.28 | thieumS | Silik0n no NAT performed on both sides |
| 13:51.41 | MikeJ[Laptop] | what is an unmodified cvs-head... |
| 13:51.43 | jwitte | Dells 2850 does not work well with the stable zaotel drivers |
| 13:51.44 | ManxPower | ariel_: I thought I was the only one left using 1.0.x |
| 13:51.52 | Silik0n | personally I think we are way past due for featurefreeze/branch |
| 13:52.01 | ManxPower | Silik0n: I agree. |
| 13:52.05 | ariel_ | ManxPower, I have over 20 installs with stable and keeping them there for now. |
| 13:52.08 | MikeJ[Laptop] | funny, we talked about that yesterday on the call |
| 13:52.14 | Silik0n | jwitte: 2850's work fine w/ head drivers |
| 13:52.22 | dmccollum | I would like to atleast see the Realtime get pused to stable. |
| 13:52.28 | ariel_ | I give my vote for the freeze |
| 13:52.31 | dmccollum | pushed |
| 13:52.31 | ManxPower | ariel_: I have 6 or so servers |
| 13:52.32 | Silik0n | I have dozens of them deployed for various tasks |
| 13:52.36 | jsharp | are there audio archives of the dev meetings available somewhere? |
| 13:52.45 | Silik0n | F@realtime |
| 13:52.57 | MikeJ[Laptop] | content of the 1.2 discussion is on -dev mailing list |
| 13:53.08 | MikeJ[Laptop] | not the audio, just the outcome |
| 13:53.18 | ManxPower | ariel_: Asterisk seems a lot like phpGroupWare in regards to releases. |
| 13:53.30 | ManxPower | And we all know what happened to phpGroupWare, don't we? |
| 13:53.30 | bkw_ | ManxPower, we have alot of local patches for functions but they are all external apps... as it stands its unmodified. |
| 13:53.40 | frogy | what's the max concurrent number of calls, can u guys achieve in your servers? |
| 13:53.42 | ManxPower | bkw_: and the CVS date? |
| 13:53.52 | bkw_ | ManxPower, I got yesterdays on a box too |
| 13:53.54 | MikeJ[Laptop] | but, 1.2 will NEVER come, unless we can get a push to close at least the few major and crash bugs in mantis |
| 13:53.57 | ManxPower | Fraeggl: 24 on my servers. |
| 13:53.57 | bkw_ | two weeks ago on some... |
| 13:54.09 | bkw_ | ManxPower, you continue to spew crap that is not based on any facts |
| 13:54.15 | *** join/#asterisk boch (~as24@200.59.172.98) |
| 13:54.20 | bkw_ | so unless you actually know what the hell you're taking about please SHUT UP |
| 13:54.23 | NewSole | bkw.... |
| 13:54.30 | NewSole | u got a sec |
| 13:54.38 | bkw_ | NewSole, maybe whats up |
| 13:54.44 | puowvip | heh |
| 13:54.44 | MikeJ[Laptop] | bkw_, your chipper today :) |
| 13:54.46 | ManxPower | bkw_: The last time I tried CVS-HEAD Asterisk crashed about once a day. Sorry, but I got tired of users trying to lynch me. |
| 13:54.59 | wiz8291 | how can i listen to a zap channel to monitor whats going on before its picked up? i.e. to hear if caller id is being sent... |
| 13:55.08 | bkw_ | ManxPower, you obviously didn't know what you were doing |
| 13:55.08 | NewSole | did u get a chance to look at # 4188 on the bugs |
| 13:55.16 | ManxPower | bkw_: Obviously. |
| 13:55.29 | puowvip | where are they ops when you need 'em |
| 13:55.33 | puowvip | or rather the ops |
| 13:55.40 | bkw_ | puowvip, I AM an op |
| 13:55.43 | *** join/#asterisk L|NUX (linux@202.5.129.56) |
| 13:55.59 | ManxPower | People got tired of waiting for phpGroupWare to actually release a stable, released version and so three forks of the project happened. |
| 13:56.16 | bkw_ | s/phpGroupWare/Asterisk/ |
| 13:56.37 | bkw_ | asterisk is stable as hell if you just switch calls with it... |
| 13:56.40 | puowvip | bkw, ooh. Whoops |
| 13:56.57 | *** join/#asterisk amir_ (~amir@shield.guindehi.ch) |
| 13:56.59 | ManxPower | NewSole: he hasn't gotten laid recently? |
| 13:57.02 | frogy | ManxPower: only 24 calls? what cpu r u using? |
| 13:57.06 | bkw_ | its when you try to do freaky crap its not so great.. and we do freaky crap all the time.. works fine for us |
| 13:57.10 | MikeJ[Laptop] | wow, testy |
| 13:57.16 | ManxPower | frogy: It's not a CPU issue, it's an interface issue. |
| 13:57.31 | wiz8291 | anyone? listen to a zap channel? |
| 13:57.38 | cjk_ | can anyone recommend a version including realtime module which is running quite stable? |
| 13:57.38 | ManxPower | Oh! I had forgotten about the recently added server. That one handles up to about 48 calls. |
| 13:57.43 | frogy | ManxPower: what card r u using them? |
| 13:57.43 | wiz8291 | caller id is not working and i want to see if asterisk is sending out the tones... |
| 13:57.47 | MikeJ[Laptop] | bkw_, but you do freaky crap you write yourself, not in tree freaky crap :) |
| 13:57.57 | ManxPower | frogy: TE405P. |
| 13:58.00 | bkw_ | ManxPower, you must have gotten a bad CVS checkout... you know when we use CVS I run and test it on a box.. when I know the current cvs-head is stable.. I pull it.. test it.. and put it out there. |
| 13:58.02 | jskcr|lappy | bkw iz a super freak :) |
| 13:58.22 | bkw_ | MikeJ[Laptop], we have given alot of that freaky stuff away for free ;) |
| 13:58.32 | MikeJ[Laptop] | hehe, I know |
| 13:58.41 | ManxPower | frogy: your question has no answer without knowing EXACTLY what sort of configuration you will be using and even then it's next to impossible to come up with an accurate number. |
| 13:58.53 | frogy | ManxPower: so u r saiding the 4 port card can only handle 2 ports of traffic, right? |
| 13:59.04 | ManxPower | frogy: no, I'm saying I only have 2 T-1s |
| 13:59.05 | cjk_ | bkw_: do you knwo how to get a version including realtime db suport which is running stable? you seem to run an unstable version as well |
| 13:59.09 | puowvip | I have asterisk to thank for restoring the peace in my home. So there. |
| 13:59.15 | frogy | c u |
| 13:59.19 | frogy | I c |
| 14:00.04 | bkw_ | unstable? |
| 14:00.05 | bkw_ | haha |
| 14:00.15 | bkw_ | cvs-head is more stable on some days than so called "stable" |
| 14:00.18 | ManxPower | frogy: if you are using ULAW/ALAW codec you will be able to handle MANY MANY more calls than if you are using G729. |
| 14:00.21 | bkw_ | don't let the word "stable" fool you |
| 14:00.28 | bkw_ | it doesn't mean that its "stable" in any way |
| 14:00.32 | bkw_ | it means its feature stable |
| 14:00.32 | ManxPower | cjk_: It's more correct to day "CVS-HEAD" and "1.0.x" |
| 14:00.43 | ManxPower | day == say |
| 14:00.51 | frogy | ManxPower: our application is mainly passing calls from one server to another with g729. |
| 14:00.55 | bkw_ | cjk_, well cvs as of now should be fine.. |
| 14:01.01 | bkw_ | it had some issues yesterday |
| 14:01.04 | bkw_ | but that was build related |
| 14:01.05 | tzanger | CVS HEAD isn't unstable |
| 14:01.13 | tzanger | it's just cutting-edge |
| 14:01.18 | cjk_ | bkw_: ok, becaus some months ago cvs-head was really broken |
| 14:01.18 | tzanger | stable isn't necessarily stable |
| 14:01.20 | tzanger | it's feature-froze |
| 14:01.22 | bkw_ | tzanger, but sometimes that edge can cut your head off |
| 14:01.26 | jskcr|lappy | lol |
| 14:01.27 | ManxPower | Correct. 1.0.x does not get new features, it only gets bug fixes. CVS-HEAD gets bug fixes, new features, removed features, and code reorganization. |
| 14:01.30 | tzanger | bkw_: nonsense |
| 14:01.36 | tzanger | if it don't work you back up a day ro so |
| 14:01.37 | FaithX | I seem to be having some dodgey issues... like bad breakup of voice ... silent sections of say 10 seconds long... where as I have talked to other users using the same voip provider who connect directly with a sip phone and they don't seem to suffer the same problems whatsoever |
| 14:01.38 | tzanger | it ain't hard |
| 14:01.45 | bkw_ | tzanger, it depends... but ya you back up a day or so |
| 14:02.04 | bkw_ | but people that dont have the common sense to do that seem to bitch and spread myths that CVS-HEAD is bad |
| 14:02.11 | tzanger | I always copy the last working build to a temp dir before I upgrade |
| 14:02.14 | bkw_ | oh and stable is actually stable. |
| 14:02.14 | tzanger | if it fails I restore quickly |
| 14:02.15 | bkw_ | :P |
| 14:02.16 | jskcr|lappy | in a production enviorment that can mean your head |
| 14:02.17 | dmccollum | How are those of us who aren't CVS ninja's to know what's a stable release of Head? Do you guys have notes or anything that would give a clue? |
| 14:02.28 | ManxPower | bkw_: I watch asterisk-cvs. I see the kinds of stuff that happens to CVS-HEAD. |
| 14:02.28 | tzanger | dmccollum: there is no "stable release of head" |
| 14:02.37 | tzanger | dmccollum: you pays the man and takes your chances |
| 14:02.42 | jskcr|lappy | lol |
| 14:02.57 | NewSole | bkw... |
| 14:03.00 | jskcr|lappy | man im running low on bawls :( |
| 14:03.03 | MikeJ[Laptop] | there is no stable release of stable either |
| 14:03.05 | bkw_ | bkw_, 4185? is an h323 bug |
| 14:03.07 | jskcr|lappy | I need to goto the store and get another case |
| 14:03.11 | *** join/#asterisk jamese (~nospam@81.168.28.93) |
| 14:03.13 | cjk_ | debian has a stable a testing and an unstable version. asterisk shoud have something like that too |
| 14:03.14 | MikeJ[Laptop] | just less changing, not less boroken |
| 14:03.17 | ManxPower | So basically there is no stable release of Asterisk? |
| 14:03.25 | tzanger | cjk_: no it shouldn't |
| 14:03.26 | bkw_ | ManxPower, correct |
| 14:03.31 | tzanger | unless you're willing to manage it |
| 14:03.33 | cjk_ | tzanger: why? |
| 14:03.34 | jskcr|lappy | yes its called the business version. |
| 14:03.41 | *** join/#asterisk shido6 (~greg@d57-87-253.home.cgocable.net) |
| 14:03.41 | bkw_ | ya the CLOSED SRC one |
| 14:03.42 | NewSole | if I donate a routing/accounting/provision model any way we can get bug fix done for 4188 |
| 14:03.44 | ManxPower | Sounds like a pretty sucky PBX to me. |
| 14:03.47 | tzanger | bkw_: uh |
| 14:03.48 | jskcr|lappy | At least I hope thats stable. |
| 14:03.51 | bkw_ | we will never get the src tree the biz edition is built from |
| 14:03.51 | tzanger | ABE is not closed source |
| 14:03.53 | tzanger | they made that VERY clear |
| 14:04.04 | bkw_ | its an internal fork |
| 14:04.08 | bkw_ | the fixes from it are put back in CVS |
| 14:04.11 | NewSole | http://bugs.digium.com/view.php?id=4188 |
| 14:04.16 | dmccollum | hehe, seems like everyone in here has a different definition of "stable" |
| 14:04.19 | bkw_ | but we will never see the exact tree its built from |
| 14:04.25 | tzanger | bkw_: ahh ok |
| 14:04.31 | tzanger | dmccollum: stable == what works for you |
| 14:04.36 | bkw_ | but we do see fixes from it |
| 14:04.38 | bkw_ | so its all good |
| 14:04.40 | ManxPower | dmccollum: I actually *try* not to use the term "stable" when referring to versions of Asterisk. |
| 14:04.47 | tzanger | when HEAD breaks SIP it's still stbale for me since I odn't use SIP |
| 14:05.21 | bkw_ | haha |
| 14:05.42 | bkw_ | sip is broken wether it work or not |
| 14:05.46 | FaithX | anyone got any ideas for me? |
| 14:05.48 | MikeJ[Laptop] | use the word "release" |
| 14:05.59 | cjk_ | hmm, still there should be some notes about the different heads and there problems... or am i just the only one missing them? |
| 14:06.01 | jskcr|lappy | or "It works for me" |
| 14:06.16 | dmccollum | What would be nice is if when bkw or some other devs feel a particular release of Head is fairly stable they sit it aside for people to download with the understanding of course that it may or may not be totally stable for you. |
| 14:06.20 | bkw_ | cjk_, just always run it |
| 14:06.34 | ManxPower | cjk_: Nobody seems to want to document that (other than what's on the asterisk-cvs mailing list) because CVS-HEAD changes ALL THE TIME. |
| 14:06.36 | bkw_ | dmccollum, I might start doing that |
| 14:06.43 | *** join/#asterisk ronn (ronn@host217-46-199-164.in-addr.btopenworld.com) |
| 14:06.47 | MikeJ[Laptop] | cjk, there are notes about problems in head, cvs mailing list and the bugtracker |
| 14:06.55 | ManxPower | Personally, I think the asterisk-cvs mailing list is good enough for documentation of CVS-HEAD |
| 14:07.08 | *** join/#asterisk ChkDigit (~mike@static65-87-228-18.regina.accesscomm.ca) |
| 14:07.11 | bkw_ | I might actually snag a tarball of a known good day |
| 14:07.38 | MikeJ[Laptop] | well, hopefully we can do more frequent "releases" in the future, and then this might not be an issue |
| 14:07.47 | jskcr|lappy | what about asterisk-1.07 |
| 14:07.58 | bkw_ | MikeJ[Laptop], frequent release? |
| 14:08.01 | bkw_ | that just sounds wrong |
| 14:08.02 | cjk_ | so using cvs head is like roulette. might be a good one or not |
| 14:08.04 | bkw_ | you need Depends? |
| 14:08.09 | MikeJ[Laptop] | sorry |
| 14:08.13 | ManxPower | MikeJ[Laptop]: Keep dreaming. The people that work on Asterisk want new features, they don't want to go back and fix minor issues. |
| 14:08.14 | bkw_ | cjk_, yes.. but you test |
| 14:08.26 | bkw_ | backing up a day or two will usually solve all your problems |
| 14:08.44 | jskcr|lappy | This software comes with absolutely no warranties. It might destroy your machine, make you go crazy and eat up your neighbour. Use of this code is entirely at your own risk. Although it is likely to work fine, there is no guarantee whatsoever. |
| 14:08.48 | bkw_ | ManxPower, not true |
| 14:08.49 | ellvis | ehm, my asterisk installation is pretty stable anyway :) |
| 14:08.50 | bkw_ | we fix stuff all the time |
| 14:08.51 | cjk_ | bkw_: ok i will see. because i really need those realtime thing |
| 14:09.01 | bkw_ | and we continue to fix shit that gets changed during commit |
| 14:09.01 | ManxPower | bkw_: Asterisk is not unique in this problem. |
| 14:09.07 | bkw_ | ManxPower, i know |
| 14:09.15 | bkw_ | but known good working patches get fucked up during commit |
| 14:09.20 | ronn | hi guys .. is there a way to implement call transfer on the asterisk side instead of the handset ? |
| 14:09.22 | bkw_ | "with minor changes" |
| 14:09.29 | ManxPower | But really, it's Mark's job to set release schedules, and he has not done so. |
| 14:09.31 | bkw_ | those minor changes have totally broken the new feature before |
| 14:09.59 | ChkDigit | ronn: Use the manager API. |
| 14:10.09 | bkw_ | is it really an API? |
| 14:10.12 | *** join/#asterisk apardo (~apardo@80.26.165.255) |
| 14:10.13 | bkw_ | I think of it as more of a Kludge |
| 14:10.20 | bkw_ | The Manager Kludge! |
| 14:10.27 | ChkDigit | Well, an interface,. at least... |
| 14:10.43 | Moonwick | release schedules are a tool of the capitalist elite for enforcing their will upon the hard-working programmer. |
| 14:10.48 | Moonwick | RMS would never approve. |
| 14:10.50 | HeadachesAbound | Asterisk CVS-HEAD-05/02/05-09:30:13 - has worked fine for the testing I've been doing over the last few days. |
| 14:11.03 | bkw_ | HeadachesAbound, yep |
| 14:11.17 | bkw_ | cvs head doesn't break often... |
| 14:11.34 | bkw_ | but when it does.. its usually in a bad way... but fixed promptly |
| 14:11.37 | ManxPower | bkw_: you have converted to the H323 Dark Side. |
| 14:11.41 | HeadachesAbound | I have yet to experience fatal issues when using CVS-HEAD and I have also yet to use stable, ever. |
| 14:12.16 | ChkDigit | Hey, I hope somebody here can tell me what it means when dmesg reports: "Ouch, part reset, quickly restoring reality (2)" |
| 14:12.41 | Moonwick | it means your server's accidentally entered a black hole |
| 14:12.48 | ManxPower | It's sad when we lose one of our own. The only thing we can do is grieve and move on. |
| 14:12.54 | ronn | ChkDigit: u can use the manager api to do call transfers? from one user to another ? |
| 14:12.59 | jskcr|lappy | I think its a power alarm |
| 14:13.00 | HeadachesAbound | If there is one thing I like about * it is the fact that there are people who do work with it and develop it and do know what's going on. |
| 14:13.17 | bkw_ | ManxPower, what? |
| 14:13.23 | bkw_ | h323 is not bad |
| 14:13.24 | jskcr|lappy | is it a tdm400 thats lost reality? |
| 14:13.24 | bkw_ | or evil |
| 14:13.28 | ChkDigit | ronn: Yes, for a pretty example see www.asternic.org |
| 14:13.29 | HeadachesAbound | of course, i do believe that bkw_ has been consumed by the dark side of the farce and may soon be departing with the dolphins. |
| 14:13.35 | ManxPower | So, does Digium run CVS-HEAD on their office PBX and on IAXtel? |
| 14:13.54 | bkw_ | HeadachesAbound, you got it |
| 14:13.55 | ManxPower | HeadachesAbound: bkw_ doesn't like fish. |
| 14:14.03 | bkw_ | ManxPower, yes digium runs cvs-head |
| 14:14.05 | ChkDigit | Re: loss of reaility, yes, it is a TDM400. |
| 14:14.05 | jskcr|lappy | HeadachesAbound: dolphins? mysql? |
| 14:14.09 | ronn | thanks ChkDigit |
| 14:14.09 | bkw_ | and yes I like fish.. hate pussy. |
| 14:14.19 | bkw_ | ATTENTION |
| 14:14.26 | HeadachesAbound | jskcr: hhgttg |
| 14:14.28 | bkw_ | Anyone wish to try the new ObjSYS h323 driver? |
| 14:14.39 | bkw_ | its all self contained... its very beta |
| 14:14.42 | *** join/#asterisk makhtar (~ageller@mail.bulletinnews.com) |
| 14:14.44 | ManxPower | bkw_: there's a 3rd H232 channel driver? |
| 14:14.45 | ChkDigit | Anything has to be better than OpenH.323 |
| 14:14.47 | bkw_ | It only supports ulaw |
| 14:14.53 | bkw_ | ManxPower, make that the 4th one |
| 14:14.58 | ManxPower | bkw_: does it depend on the OpenH323 libs? |
| 14:14.59 | MikeJ[Laptop] | hehe |
| 14:15.04 | bkw_ | OpenH323 is not bad nor is it broken |
| 14:15.06 | ManxPower | bkw_: chan_h323, chan_oh323, and ?? |
| 14:15.14 | bkw_ | http://www.obj-sys.com/open/ooh323c_driver-0.1.tar.gz |
| 14:15.15 | jskcr|lappy | ChkDigit: I dont think digium even knows whats causing that at this point |
| 14:15.17 | bkw_ | ManxPower, chan_woomera |
| 14:15.33 | ManxPower | bkw_: never heard of it. |
| 14:15.36 | MikeJ[Laptop] | ManxPower, yes, those 2 are both broken, but thats not openh323's fault |
| 14:15.39 | bkw_ | its only existed for a week now |
| 14:15.47 | ManxPower | Which means it's either well documented, or nobody uses it. |
| 14:16.12 | bkw_ | ManxPower, both of these new drivers have existed for less than two weeks |
| 14:16.16 | bkw_ | SO step off bitch |
| 14:16.26 | MikeJ[Laptop] | bkw, how many working h323 channel drivers have you used this week |
| 14:16.29 | ChkDigit | Hum, I'm wondering if my telco has some issues WRT: Ouch, part reset... |
| 14:16.34 | bkw_ | MikeJ[Laptop], exactly 2 |
| 14:16.47 | MikeJ[Laptop] | funny, me too |
| 14:17.24 | bkw_ | i'm actually talking to MikeJ[Laptop] via h323 now |
| 14:17.47 | ManxPower | ChkDigit: that's usually a hardware issue. contact Digium support |
| 14:17.58 | HeadachesAbound | for those of us who were sleeping during class, what is h323? |
| 14:18.07 | ronn | ChkDigit: i saw the link.. the feature i'm lookin for is allow the called party to be able to transfer the call himself .. instead of by operator .. is that possible? |
| 14:18.26 | jskcr|lappy | ChkDigit: heres a nice long thread on it though with lots of theorys http://lists.digium.com/pipermail/asterisk-users/2004-December/076043.html |
| 14:18.35 | ManxPower | HeadachesAbound: It's like SIP, but older, much more complicated and hostile to NAT, but a lot of telecom gear supports H323 as their only VoIP protocol. |
| 14:19.07 | HeadachesAbound | older, complicated, and hostile...sounds like some of my relatives, not sure I'm interested. |
| 14:19.11 | *** join/#asterisk anderiv (~anderiv@207-67-87-34.gen.twtelecom.net) |
| 14:19.34 | jamese | Hi Guys. New to voip/asterisk and looking for a few pointers re H.263 video streaming. Can anyone help? |
| 14:19.39 | ManxPower | HeadachesAbound: Generally the only people that run H323 are either sociopaths or people that have no other choice because of the hardware they have. |
| 14:19.45 | HeadachesAbound | of course, that might also explain the interest that bkw_ has in it...and his recent change of attitude. |
| 14:19.45 | bkw_ | ManxPower, yet again |
| 14:19.51 | bkw_ | speaking about shit you know nothing about |
| 14:20.01 | jskcr|lappy | It would be nice if someone deflaked the t100p, but at this point I doubt it |
| 14:20.03 | ManxPower | I just say what I see on the channel |
| 14:20.04 | bkw_ | PLEASE ASSIST ME BY SHUTTING UP! |
| 14:20.21 | HeadachesAbound | why do you use it bkw? |
| 14:20.24 | bkw_ | see i'm the myth buster |
| 14:20.39 | bkw_ | HeadachesAbound, we are assisting in writting the new driver from ObjSYS |
| 14:20.59 | bkw_ | h323 can work behind nat |
| 14:21.07 | bkw_ | if you have a gatekeeper |
| 14:21.09 | bkw_ | and get this |
| 14:21.11 | bkw_ | IT WORKS |
| 14:21.14 | MikeJ[Laptop] | H323, is fine, asterisks implementations so far have been weak |
| 14:21.17 | bkw_ | its not more complicated than pri |
| 14:21.21 | MikeJ[Laptop] | that is being fixed |
| 14:21.29 | pigpen | Would someone please give their opinion how many 4 port ananlog cards (FXO) from digium I could run in a single P4 Xeon box? |
| 14:21.43 | bkw_ | pigpen, 30 |
| 14:21.51 | ManxPower | pigpen: two |
| 14:22.04 | MikeJ[Laptop] | pigpen, if you need more than a couple, just use pri cards and channel banks |
| 14:22.08 | ManxPower | pigpen: maybe more, but that would totally depend on your motherboard, etc. |
| 14:22.13 | HeadachesAbound | how well does asterisk run on amd boxes? |
| 14:22.24 | ManxPower | next bkw_ will be telling us that you can share IRQs on Digium cards. |
| 14:23.05 | bkw_ | no but you can with sangoma cards |
| 14:23.16 | bkw_ | zero IRQ issues with sangoma cards |
| 14:23.16 | Nuxi | only if you use digium cards via usb 2.0 and have a firewire connection to the telephone. |
| 14:23.17 | bkw_ | NEXT!!! |
| 14:23.21 | ManxPower | bkw_: Are you a Sangoma convert? |
| 14:23.34 | MikeJ[Laptop] | convert? |
| 14:23.37 | bkw_ | ManxPower, use what works.. we have over 70 digium cards |
| 14:23.46 | MikeJ[Laptop] | are they polar opposites and you can't like both |
| 14:23.54 | MikeJ[Laptop] | like linux and windows :D |
| 14:23.59 | HeadachesAbound | do digium cards play well on AMD machines with asterisk? |
| 14:23.59 | *** join/#asterisk Grooby (~Grooby@12.22.232.212) |
| 14:24.07 | bjohnson | sure |
| 14:24.09 | bkw_ | HeadachesAbound, AMD 64bit |
| 14:24.20 | bkw_ | only use AMD Opteron's with digium cards and asterisk |
| 14:24.27 | Nugget | I use both linux and windows and have about the same amount of affection for both. |
| 14:24.29 | bkw_ | if you're doing 32bit use intel |
| 14:24.34 | *** part/#asterisk jamese (~nospam@81.168.28.93) |
| 14:24.37 | MikeJ[Laptop] | :D |
| 14:24.40 | jskcr|lappy | it will be interesting to see how the dual cores run it |
| 14:24.58 | bkw_ | we will know |
| 14:25.04 | ManxPower | We've had good luck with Intel motherboards. |
| 14:25.13 | ChkDigit | Dual core is just SMP, I assume people run SMP *, right?? |
| 14:25.15 | bkw_ | you know how fast it a quad dual core Opteron can compile the linux kernel in? |
| 14:25.26 | bkw_ | it takes 16 seconds to compile the kernel... |
| 14:25.31 | jskcr|lappy | mmmmmmm yum |
| 14:25.35 | Nugget | spiffy |
| 14:25.49 | Nuxi | I bet it could even open Open Office in under a half hour. |
| 14:26.13 | Nuxi | Maybe not, it still has to come off the disk. |
| 14:26.23 | jskcr|lappy | either that or Ill wait for santa with a stinger across from bkw's house on christmas :P |
| 14:26.39 | *** join/#asterisk osiris_of_the_24 (~osiris@c-67-162-163-67.hsd1.mi.comcast.net) |
| 14:26.48 | Pj386 | bkw_: how many Hz ? |
| 14:26.59 | bkw_ | no clue someone was telling me about it. |
| 14:27.34 | HeadachesAbound | how easy is it to make 2 asterisk boxes talk to each other? I'm thinking about setting up a couple of Dell 750s running Quad T Cards in each one and will need them to talk to each other in a quick and efficient manner, is this possible and easy? |
| 14:27.43 | jskcr|lappy | iax |
| 14:27.56 | MikeJ[Laptop] | HeadachesAbound, yes, easy |
| 14:28.11 | MikeJ[Laptop] | very very easy if they are right next to each other |
| 14:28.47 | bkw_ | or 10000000 miles apart |
| 14:28.55 | bkw_ | jitter might get ya at that distance |
| 14:28.57 | jskcr|lappy | pop a extra ethernet card in them with a crossover cable. |
| 14:28.57 | HeadachesAbound | they will be right next to, or on top of, each other. i'm limited to a budget of $4500 for the hardware on this setup so i'm seriously considering the Dell 750s so i can get the performance that the boss wants |
| 14:28.58 | bkw_ | :P |
| 14:29.18 | HeadachesAbound | jskcr: i just thought of that as well. |
| 14:29.30 | ronn | any simple way to enable called parties to transfer calls to other extensions? if the phone doesn't have transfer function? |
| 14:29.34 | jskcr|lappy | Dell 750's? |
| 14:29.43 | jskcr|lappy | get 2 prolients with kickstart cards |
| 14:29.53 | bkw_ | ronn, read "show application dial" |
| 14:29.55 | bkw_ | check features.conf |
| 14:30.04 | bkw_ | ronn, look at T and t for dial |
| 14:30.28 | jskcr|lappy | I fell quite in love with the hp kickstart cards when a stuipid admin fucked up the bootloaded on a server 2000 miles away in a colo |
| 14:30.55 | ronn | bkw .. that would enble call transfer .. but is there a way for the called party to place the call on hold and transfer the call if the handset doen not have a transfer/hold feature ? |
| 14:31.06 | Aze` | which source files used by attend xfer in cvs head ? |
| 14:31.20 | ChkDigit | ronn: Usually flash will do it for analogue phones... |
| 14:31.39 | ManxPower | ronn: NO! T/t enables DTMF based transfers. You don't need them if the phone has it's own transfer ability. |
| 14:31.47 | ronn | ChkDigit: what's flash ? |
| 14:32.11 | *** join/#asterisk loud (~ariel@blaqhat.com) |
| 14:32.20 | ronn | ManxPower: what happens if the phone does not have hold/transfer button? |
| 14:32.23 | ChkDigit | Quick switch-hook. Tap the hangup lever, or lots of phones have a flash button... |
| 14:32.25 | jskcr|lappy | ronn: hitting the hook real fast |
| 14:32.41 | ManxPower | ronn: then you use T/t and use the # based transfer. |
| 14:32.57 | *** join/#asterisk Ridgeback (~Ridgeback@ppp210-241.lns1.adl2.internode.on.net) |
| 14:33.04 | ManxPower | jskcr|lappy: that only works with ATAs and it still requires the device to have a transfer feature. |
| 14:33.25 | Ridgeback | hello! |
| 14:33.30 | ronn | the # based transfer? the called party will dial # and dial the extension to transfer to? |
| 14:33.55 | ManxPower | ronn: check the Wiki for how to use it. |
| 14:34.25 | ronn | ManxPower: is it called # based transfer? |
| 14:34.37 | Ridgeback | anyway to use SendText to send a message to a distant phone, rather than the channel your on? |
| 14:34.51 | ManxPower | ronn: I don't know if there's an official name. I always call it # based transfer. |
| 14:35.27 | *** join/#asterisk grolloj (~chatzilla@slim-eth0.horizonlive.net) |
| 14:36.00 | *** join/#asterisk ckruetze_ (HydraIRC@cpc3-cmbg7-5-0-cust100.cmbg.cable.ntl.com) |
| 14:38.00 | bjohnson | let's call it t-transfers |
| 14:38.14 | bjohnson | or ttttttttransfer |
| 14:39.40 | ChkDigit | Shouldn't that be trrrrrrrrrrrrrransfer! |
| 14:39.51 | ManxPower | bjohnson: I call it "brain-dead phone transfer" |
| 14:39.57 | *** join/#asterisk znoG (gs@200.115.216.109) |
| 14:40.07 | bjohnson | well .. you use a 't' or a 'T' .. not a 'r' |
| 14:40.07 | *** join/#asterisk [TK]D-Fender (~joe@4.67.252.216.dsl1.colba.net) |
| 14:41.20 | Gand_DJ | ne1 been able to figure out how to setup * so you can use the park feature built into eyebeam? (instead of *70 while on call) |
| 14:43.38 | *** join/#asterisk Cresl1n (~matt@216.207.245.23) |
| 14:43.40 | *** join/#asterisk mhnoyes (~mhnoyes@user-2ivfjsj.dialup.mindspring.com) |
| 14:50.13 | ionix | can anyone check http://pastebin.ca/11117 |
| 14:50.21 | ionix | I don't get the .tiff file... |
| 14:50.30 | ionix | ls /var/spool/asterisk/fax/15149402629/1115386775.11.tif |
| 14:50.30 | ionix | ls: /var/spool/asterisk/fax/15149402629/1115386775.11.tif: No such file or directory |
| 14:50.45 | *** join/#asterisk PBXtech (~nik@67.107.241.3.ptr.us.xo.net) |
| 14:51.32 | PBXtech | dudes |
| 14:55.28 | *** join/#asterisk voip4life (~voip4life@ogd50-33.wi.relia.net) |
| 14:57.14 | voip4life | looking for pointers troubleshooting a flashing red alarm on a T100p card, can't find much documentation online... |
| 14:57.47 | voip4life | I think the card might be bad, because I put in the old card and it works great |
| 14:57.51 | ronn | ManxPower: call parking looks like a good replacement for handset that don't have hold/transfer button.. what do you think? |
| 14:58.13 | PBXtech | sounds like a bad card, call Digium |
| 14:58.32 | ManxPower | ronn: you transfer a call to the parking extension |
| 14:58.36 | voip4life | I've opened a ticket with digium, but have yet to hear back |
| 14:58.45 | PBXtech | yea sounds normal :) |
| 14:59.09 | HeadachesAbound | How many calls do you think this could handle...PowerEdge 750, Intel Pentium 4 processor, 2.8GHz, 1MB Cache, 800MHz FSB, 1GB DDR, 400MHz, 160GB 7.2K RPM SATA Hard Drive, Dual Onboard Gigabit NICs, Floppy, CDRom, Quad T Card from Digium, CVS-Head * |
| 14:59.20 | ronn | ManxPower: yes and the person who was meant to get the call transferred picks it up |
| 14:59.33 | PBXtech | like 24*4 |
| 14:59.35 | ManxPower | why not just transfer it directly to the person |
| 14:59.57 | PBXtech | 96 woot |
| 15:00.09 | ronn | ManxPower: the handset doen't have the hold/transfer button |
| 15:00.25 | PBXtech | press # |
| 15:00.35 | ManxPower | ronn: well if you can't transfer the call to a person you can't transfer the call to a parking extension |
| 15:01.15 | ronn | PBXtech: if you press the # in the middle of the call .. you would get a dial tone? |
| 15:01.20 | [TK]D-Fender | HeadachesAbound : Depends if you have to do protocol transcoding, etc. |
| 15:01.24 | ManxPower | ronn: why don't you TRY IT? |
| 15:01.26 | PBXtech | SIP+ # |
| 15:01.32 | PBXtech | ask for a xfer extension |
| 15:01.34 | [TK]D-Fender | HeadachesAbound : What is your goal? |
| 15:01.39 | PBXtech | ZAP + flash |
| 15:01.43 | ManxPower | ronn: what VoIP phone are you using? |
| 15:02.00 | ronn | ManxPower: i will try that .. may be it's the sip adaptor i'm using |
| 15:02.10 | ManxPower | ronn: what VoIP phone are you using? |
| 15:02.15 | PBXtech | should still pass the dtmf |
| 15:02.16 | ronn | it's a sipura |
| 15:02.42 | ManxPower | ronn: the SIPura supports transfers |
| 15:03.07 | ChkDigit | Press the >> button... |
| 15:03.12 | ronn | ManxPower: may be i need to configure it properly in the sip.conf ... the correct dtmf stuff |
| 15:03.17 | ChkDigit | You get more options that way... |
| 15:03.18 | HeadachesAbound | PBXTech : Yes, 96 Channels, Potention for up to 92 Usable Zap/PRI. I'm thinking about having 2 of these connected together with 7 Ts used for outgoing and 1 for incoming. Internal extensions will be SIP Only. |
| 15:03.37 | PBXtech | i wouldnt do that |
| 15:03.51 | PBXtech | but people do |
| 15:03.53 | ManxPower | ronn: the SIPura uses FLASH/RECALL to do Transfers. You didn't read the SIPura documentation, did you? |
| 15:04.13 | tld | Anyone have experience deploying softphone installations? I'm looking for a decent softphone that I can pre-configure with proxy setting (and preferrably username/password) that'll squeeze into a small budget. |
| 15:04.28 | PBXtech | i dont think a single 2.8G would handle 8 T's |
| 15:04.30 | HeadachesAbound | PBXtech: what would you recommend? |
| 15:04.39 | PBXtech | 1 box 1 quad card |
| 15:04.48 | PBXtech | and an IAX link |
| 15:05.00 | HeadachesAbound | That is 1 box 1 quad card, i plan to have 2 boxes with 1 card each and an IAX link...as you said. |
| 15:05.06 | PBXtech | yea |
| 15:05.43 | [TK]D-Fender | HeadachesAbound : if your using T1's for Channel Banked extensions and for incoming lines on PRI then its all TDM. At which point you can load a LOT onto 1 box... |
| 15:05.48 | HeadachesAbound | The original plan was for 1 box and 2 cards, but I've spent a week listening to bkw and ManxPower rant about the evilness of such a thing and decided I don't have energy to deal with that kind of evil at the moment. |
| 15:06.13 | *** join/#asterisk rndmtngnt ([U2FsdGVkX@203.13.74.236) |
| 15:06.14 | ManxPower | HeadachesAbound: Digium claims you can out two 4-port cards in 1 box |
| 15:06.31 | [TK]D-Fender | I was quoted for a 96ext/24 line setup for my head office. They say "no problem" |
| 15:07.06 | [TK]D-Fender | Its all about the transcoding... |
| 15:07.10 | *** join/#asterisk The_X (chris@true.fiberpimp.net) |
| 15:07.28 | HeadachesAbound | What transcoding is there between SIP and Zap/PRI? |
| 15:07.36 | The_X | what should I use to get my CSR folks to login? |
| 15:07.37 | _omer | anybody related to SIP Servers ,,,,,,,, plz msg me |
| 15:07.50 | tzanger | HeadachesAbound: anything to ulaw/alaw |
| 15:07.54 | The_X | AgentLogin is a failure |
| 15:08.02 | The_X | I need something that will just dial their extension |
| 15:08.03 | ronn | ManxPower: need to do some more reading i guess |
| 15:08.26 | [TK]D-Fender | HeadachesAbound : it still has to do a basic amount to translate your RTP to the PRI standard and it also depends what encoding your using on your SIP RTP stream. |
| 15:08.36 | HeadachesAbound | tzanger: SIP is anything and Zap/PRI is ulaw? so if I force SIP to be ulaw then the transcoding is basically gone, right? |
| 15:08.44 | tzanger | correct |
| 15:09.20 | HeadachesAbound | thank you. i will be forcing ulaw on them all then...evil grin with horns holding up the halo |
| 15:09.52 | _omer | anybody knows "free world dialup" ....what software/sip server is required to build such kind of service.. |
| 15:10.15 | The_X | so, anyone using AgentCallbackLogin? |
| 15:12.31 | *** join/#asterisk bonez39 (~aint@drjones.dsl.xmission.com) |
| 15:13.06 | Meaty | Im trying to use the stutter dial tone when message is leaving in voicemail with analog tel. I define de mailbox in zapata.conf. And I dont hear de stutter dial ton. Any know in asterisk need more configuration for stutter tone ? |
| 15:13.15 | *** join/#asterisk anthm (~anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 15:13.15 | *** mode/#asterisk [+o anthm] by ChanServ |
| 15:13.39 | ManxPower | Meaty: mailbox=vm_box@vm_context BEFORE the channel. |
| 15:14.11 | ManxPower | Note vm_context IS NOT the extensions.conf context. |
| 15:15.01 | *** part/#asterisk ManxPower (~eric@dsl-209-16-67-160.datasync.com) |
| 15:17.08 | *** join/#asterisk flot (~flot@user155.hovrino.net) |
| 15:19.45 | flot | hi all. Whether it is possible to adjust * on the notification about the second call on phone (if by phone already talk)? |
| 15:20.03 | Aze` | How send DTMF via * Manager in channel ? |
| 15:20.41 | *** join/#asterisk gambolputty (~gambolput@cblmdm69-45-216-83.buckeye-express.com) |
| 15:22.34 | jakepdev | Aze` - set the dtmf you want to send to a variable and redirect to a context with SendDTMF |
| 15:24.14 | _omer | anybody knows "free world dialup" ....what software/sip server is required to build such kind of service.. |
| 15:24.37 | MikeJ[Laptop] | _omer, are you building a voip termination provider? |
| 15:28.10 | *** join/#asterisk mogorman (~mogorman@207.111.174.1) |
| 15:29.00 | *** join/#asterisk moy (~kvirc@dsl-200-95-105-36.prod-infinitum.com.mx) |
| 15:29.00 | Moonwick | if you're asking questions like that, you really shouldn't be starting any sort of voip business. |
| 15:29.07 | Moonwick | but that's just my opinion. |
| 15:33.26 | voip4life | what is the most number of x100p cards someone has had in 1 machine? 5 or 6 depending on slots available? |
| 15:34.06 | jakepdev | you'd probably run into IRQ hell after 2 or 3 - better to use TDM400 or something like that |
| 15:34.12 | [TK]D-Fender | Why on earth would you want to use X100p's? |
| 15:34.26 | *** join/#asterisk DEEZED (deezed@adsl-065-006-189-182.sip.bct.bellsouth.net) |
| 15:35.15 | voip4life | I'm waiting for my TDM04 card, but until then, I'm needing 6 lines right away. |
| 15:35.15 | [TK]D-Fender | Definately better off on TDM400's |
| 15:35.52 | [TK]D-Fender | How many total lines do you have? |
| 15:36.01 | voip4life | 6 lines total |
| 15:36.39 | _omer | <Moonwick> : well. It's for a student not for businessman :p |
| 15:36.39 | [TK]D-Fender | Hmm. You could always use Sipura SPA-3000's if you're desperate and they could be recycled into your PBX for other things later... |
| 15:37.00 | voip4life | I have long term plans on using t100p card, testing on PRI now, works great, but it will be 6 months or more before we convert to that |
| 15:37.28 | _omer | MikeJ[Laptop] : no I just need to collect information...... |
| 15:37.45 | *** join/#asterisk coolhp (~crap@mtl149-99-190-66.dedicated.sprintdsl.ca) |
| 15:37.48 | voip4life | so in the interim, I was planning on a TDM card + 2 x100p's |
| 15:38.23 | flot | How to adjust * to inform on phone on the second call? |
| 15:38.35 | voip4life | plus I have 6-7 x100's sitting around doing nothing |
| 15:38.46 | DEEZED | is there a test command to test asterisk's access to mysql? |
| 15:39.50 | coolhp | Good day everyone... I was wondering if anyone had ever had this type of errors before : my_zt_write: Write returned -1 (Resource temporarily unavailable)my_zt_write: Write returned -1 (Resource temporarily unavailable)my_zt_write: Write returned -1 (Resource temporarily unavailable)my_zt_write: Write returned -1 (Resource temporarily unavailable)my_zt_write: Write returned -1 (Resource temporarily unavailable) |
| 15:40.24 | coolhp | (sorry about the bad paste... it should have been shorter :-( ) |
| 15:45.53 | sean | is there a command I can run in a dialplan to "print" to a verbose `asterisk -r`? exten => 1,print(${CALLERID}) for example? |
| 15:46.26 | jakepdev | sean - use NoOp |
| 15:47.19 | sean | jakepdev: thank you! |
| 15:47.30 | jakepdev | np |
| 15:47.34 | ionix | can anyone check http://pastebin.ca/11117 |
| 15:47.35 | ionix | I don't get the .tiff file... |
| 15:47.36 | ionix | ls /var/spool/asterisk/fax/15149402629/1115386775.11.tif |
| 15:47.37 | ionix | ls: /var/spool/asterisk/fax/15149402629/1115386775.11.tif: No such file or directory |
| 15:52.14 | sean | next. I'm looking for a good IAX(2) softphone that's free. Suggestions? |
| 15:53.49 | [TK]D-Fender | Firefly |
| 15:53.51 | jakepdev | sean - firefly |
| 15:54.08 | sean | ok, thanks.. that's on my short list |
| 15:54.29 | sean | it seems to be NOT getting callerID, though -- ever happen to you guys? |
| 15:55.01 | jakepdev | i've only used it for outbound at this point |
| 15:56.25 | sean | -- Executing SetCallerID("SIP/<ip>-080fc500", "5149999999") in new stack |
| 15:56.25 | sean | -- Executing Dial("SIP/<ip>-080fc500", "IAX2/sean@sean|30|r") in new stack |
| 15:56.26 | sean | -- Called sean@sean |
| 15:56.37 | sean | but Firefly doesn't show the number |
| 16:01.07 | *** join/#asterisk jdg (~jdg@CA03F988.adsl.mana.pf) |
| 16:03.55 | *** join/#asterisk smash- (~smash@198.107.16.189) |
| 16:04.01 | smash- | hello |
| 16:04.08 | smash- | fellow asterisk's |
| 16:05.18 | sean | yeah. I belong to no software! |
| 16:06.00 | *** join/#asterisk pbx_lover (lethal-gr@host7.citiwave.ondsl.gr) |
| 16:06.03 | pbx_lover | hi all |
| 16:06.09 | smash- | yeah |
| 16:06.15 | smash- | * is very good |
| 16:06.18 | DEEZED | how big of a difference is there between "database support in VM (USE_MYSQL_VM_INTERFACE=1)" and "real time voicemail" |
| 16:06.46 | jakepdev | sean - you try SetCIDNum? |
| 16:06.53 | pbx_lover | accounting purposes maybe? |
| 16:07.17 | HeadachesAbound | Anybody here use the C AGI Interface? |
| 16:07.26 | sean | jakepdev: hmm .. it seems that the CID _NUMBER_ is coming through with a blank _NAME_ (which is normal, for my cell...) |
| 16:07.29 | *** join/#asterisk Nuxi (~nuxi@cust-sdsl-204-250-82-202.bzn-co-i1000-01.bridgeband.net) |
| 16:07.38 | smash- | hey i got a question if i put a number say 1503-555-5555 |
| 16:07.41 | smash- | as an extension |
| 16:07.51 | sean | I guess I'll try to use a conditional to set the name to the number if the name is blank |
| 16:07.51 | smash- | it will reach that extension b4 dialing a real phone number correct? |
| 16:08.02 | jsharp | Yes. |
| 16:08.07 | pbx_lover | has any1 tried asterisk+FAX ? |
| 16:08.09 | smash- | jsharp for me? |
| 16:08.15 | smash- | pbx_lover |
| 16:08.22 | pbx_lover | ? |
| 16:08.32 | smash- | um |
| 16:08.35 | jsharp | smash-: Yes. |
| 16:08.35 | smash- | no t yet |
| 16:08.47 | smash- | <PROTECTED> |
| 16:08.52 | jsharp | Asterisk has no concept of "real phone numbers". It just sees everything as an extension. |
| 16:09.06 | pbx_lover | yeah |
| 16:09.31 | HeadachesAbound | Asterisk is smart in the sense that it is dumb. |
| 16:10.02 | pbx_lover | fax extension? |
| 16:10.03 | *** join/#asterisk cmk (~cmk_@p54A3BCAA.dip.t-dialin.net) |
| 16:11.26 | *** part/#asterisk Nuxi (~nuxi@cust-sdsl-204-250-82-202.bzn-co-i1000-01.bridgeband.net) |
| 16:12.06 | smash- | pbx how do u plan on faxing things |
| 16:12.26 | smash- | <PROTECTED> |
| 16:12.45 | pbx_lover | trying to figure out, mainly i want to receive |
| 16:12.47 | *** join/#asterisk Nuxi (~nuxi@cust-sdsl-204-250-82-202.bzn-co-i1000-01.bridgeband.net) |
| 16:13.14 | pbx_lover | can asterisk receive fax atm? |
| 16:14.27 | *** part/#asterisk Nuxi (~nuxi@cust-sdsl-204-250-82-202.bzn-co-i1000-01.bridgeband.net) |
| 16:14.36 | *** join/#asterisk Nuxi (~nuxi@cust-sdsl-204-250-82-202.bzn-co-i1000-01.bridgeband.net) |
| 16:15.08 | jsharp | app_rxfax sort of works these days. |
| 16:16.55 | smash- | 8883212307 |
| 16:16.55 | *** join/#asterisk greeneggsnham (zack@216.61.130.19) |
| 16:16.55 | smash- | mt |
| 16:16.55 | smash- | sorry |
| 16:16.56 | smash- | j |
| 16:16.58 | smash- | does it come with asterisk |
| 16:17.03 | sean | sweet: |
| 16:17.05 | sean | exten => 210,2,GotoIf($["${CALLERIDNAME}" = "anonymous"]?4) |
| 16:17.05 | sean | exten => 210,3,SetCIDName(${CALLERIDNUM}) |
| 16:17.05 | sean | exten => 210,4,Dial(IAX2/sean@sean,30,r) |
| 16:17.06 | smash- | o gotta dl it |
| 16:17.49 | smash- | sean does that send all anonymous calls to u? |
| 16:18.06 | jakepdev | sean - probably don't need the quotes |
| 16:18.23 | key2 | when u send a call from a FXO card with asterisk, can you change your caller id number ? or would it be the one of ur PSTN line since that's the pbx of the telephon company that handles the CID number? |
| 16:18.28 | sean | m'eh copied it from the voip-info page |
| 16:18.48 | sean | smash-: no, it rewrites calls with name "anonymous" to set the name field to the number. |
| 16:19.29 | *** join/#asterisk outtolunc (outtolunc@adsl-66-218-53-170.dslextreme.com) |
| 16:19.29 | smash- | <PROTECTED> |
| 16:19.29 | smash- | <PROTECTED> |
| 16:19.37 | sean | well.. copied the syntax, I mean |
| 16:20.17 | Juggie | is it possible to embed functions in GotoIF? |
| 16:20.41 | Juggie | eg, GotoIf($["${CALLERIDNAME}" = "anonymous"]?SetVar(anonymous=1)) |
| 16:21.37 | *** join/#asterisk PCadach (~paul@www.east.telecom.kz) |
| 16:21.40 | jakepdev | Juggie - I don't think that is valid |
| 16:21.43 | Corydon-w | Only if SetVar becomes a function |
| 16:21.49 | Corydon-w | which it is not now |
| 16:22.00 | sean | Juggie: no. Just drop to a priority that has the Setvar, and skip it on false |
| 16:22.02 | Corydon-w | And anyway, your formulation is incorrect |
| 16:22.08 | *** join/#asterisk TechJournalist (~chatzilla@d141-133-19.home.cgocable.net) |
| 16:22.18 | Juggie | sean, i know that, but i was just wondering if it was possible to shorten some code :) |
| 16:22.31 | Juggie | i think it might be possible in cvs head, i seem to remember bkw or anthm showing something |
| 16:22.35 | Juggie | but not in stable |
| 16:22.48 | Corydon-w | SetVar(anonymous=$["${CALLERIDNAME}" = "anonymous"]) |
| 16:22.56 | Corydon-w | That works without functions |
| 16:23.13 | Juggie | nice. |
| 16:23.50 | sean | nice indeed. |
| 16:24.12 | smash- | nice code cory |
| 16:24.15 | Corydon-w | All you're really doing is saving the boolean value, to be used at a later date |
| 16:24.20 | smash- | y would u wanna set id anonymous though? |
| 16:24.35 | Juggie | smash-, it has no real world relevance, just an example |
| 16:24.41 | smash- | o |
| 16:24.55 | Juggie | Corydon-w, i wanted to GotoIF on system |
| 16:25.17 | Juggie | and have true run System then just move to the next number, or have false just move to the next number |
| 16:25.19 | Corydon-w | Think like a programmer... because... the dialplan is a program |
| 16:25.28 | Juggie | yes, i'm a programmer :) |
| 16:25.38 | Juggie | i wanted to do something like |
| 16:25.56 | sean | (-; |
| 16:25.56 | Corydon-w | which is why the Dialplan will never be as easy as moving icons around a screen |
| 16:26.11 | Juggie | GotoIF($[${joined} = 1]?System(/var/lib/asterisk/agi-bin/conf/conf-helper.php ${roomnum} ${numcal} out ${roomid})) |
| 16:26.34 | Corydon-w | Juggie: apply 2720 and use GosubIf... |
| 16:27.07 | Corydon-w | GosubIf($[${joined}]?confhelper) |
| 16:27.19 | MikeJ[Laptop] | Corydon-w, is 2720 ready yet? |
| 16:27.31 | MikeJ[Laptop] | I think you said you had stuff still in the works on that? |
| 16:27.42 | Corydon-w | MikeJ[Laptop]: It'll run as-is, but I think I want to add arguments and return values |
| 16:27.59 | Corydon-w | Or not... I dunno yet |
| 16:28.32 | HeadachesAbound | The Dialplan is process-flow logic. The Dialplan will eventually be as easy as moving icons around on a screen...I just haven't gotten to it yet as I really do have to sleep every 24 hours or so. |
| 16:28.52 | Corydon-w | I think we should probably go with the current implementation, rather than trying to rush in arguments and return values and get a broken implementation |
| 16:29.00 | Corydon-w | i.e. we'll add those things later |
| 16:29.04 | Nugget | I'm skeptical. |
| 16:29.35 | Nugget | the only way to make the dialplan a matter of dragging icons is to severely restrict what you can do with it. |
| 16:29.37 | opus_ | dialplan gui? |
| 16:29.42 | MikeJ[Laptop] | ok, post that to bug please |
| 16:29.51 | Nugget | a dialplan gui is about as feasible as, say, making a "perl gui" |
| 16:30.11 | MikeJ[Laptop] | kevin was already pretty much ready to go and it would be nice for 1.2 (have you seen the dev list today)... |
| 16:30.26 | HeadachesAbound | How so Nugget? I mean, Let's analyze what a dial plan does: |
| 16:30.40 | Nuxi | A perl gui, wouldn't that be one of them pesky 4GLs that were popular a decade ago? |
| 16:31.49 | Nuxi | Actually, wouldn't an IDE customized for the dialplan, with syntax checking/suggesting be useful? |
| 16:31.56 | HeadachesAbound | Enter Context To Modify...This context does not exist, would like to create it?...Yes...Would you like to create the default extensions for this context (s, t, i)?... |
| 16:32.20 | outtolunc | would you like fries with that? |
| 16:32.41 | jsharp | Yes, please. |
| 16:32.52 | Nugget | Nuxi: a syntax-aware filter for vim/emacs/ultraedit or whatever your editor of choice is, sure. |
| 16:33.12 | Nugget | but trying to make a gui to write a dialplan is never going to work beyond the most simplistic uses of the dialplan. |
| 16:33.13 | HeadachesAbound | Yes...Creates Default s, t, i extensions as defined in some template...At that point, you create extensions and specify the steps of that extension without really having to know what the syntax is for say the Dial command when you want to dial 9 for an outgoing line, etc. It's really not that difficult of a concept. |
| 16:33.24 | Godsey | anyone taking bets on how long till broadvoice goes out of business? |
| 16:33.36 | sivana | Godsey: why? |
| 16:33.43 | ChkDigit | Dunno, but my line is like a yo-yo. |
| 16:33.44 | Godsey | going on day 2 for outage :) |
| 16:33.48 | sivana | I've had no problems |
| 16:33.55 | sivana | maybe pay your bill :P |
| 16:34.02 | Godsey | most of their customers have GBLX numbers |
| 16:34.02 | HeadachesAbound | give me an example of something that you think won't work in a gui Nugget...just to see if if my brain is wired right. |
| 16:34.12 | Nugget | HeadachesAbound: no. figure it out yourself. |
| 16:34.55 | Godsey | it's not a billing problem :) |
| 16:35.07 | HeadachesAbound | :) Well, if you insist on saying that it won't work but can't back up your claim then I'm stumped by your amazing abilities. |
| 16:35.17 | Nugget | that's fine |
| 16:35.59 | HeadachesAbound | anyone else care to try and prove that the dial plan is too complex for a gui? |
| 16:37.11 | Godsey | there are guis for the dialplan |
| 16:37.12 | Nuxi | I use a gui to edit my dial plan. I fire up Konsole (a gui shell) and then use ssh and then I use nano. |
| 16:37.40 | Nugget | there are guis that accomodate the most simplistic ways you can use the dialplan, yes. |
| 16:38.16 | Nugget | to make a gui that replicates the capabilities of the dialplan is a silly goal, though. at best you'll end up with a gui which is harder to use than simply programming the dialplan natively |
| 16:38.39 | Nugget | perhaps once HeadachesAbound has, you know, actually set up asterisk at least once he'll realize this. |
| 16:38.46 | HeadachesAbound | the gui i envision would allow you to specify the context, extension, and action for that extension in detailed complexity even it is easier to do it natively because that is what the phb will want. |
| 16:39.23 | HeadachesAbound | i have setup * several times over and i could care less about having a gui, but the phb that i answer to would love a gui just to make him think he is in control. |
| 16:39.41 | jsharp | So hack up a "gui" that doesn't do anything. |
| 16:40.02 | Nugget | it's no problem to make a gui to store static information about extensions. it's not feasible to make a gui to replicate the dialplan language. |
| 16:40.20 | opus_ | already been done |
| 16:40.31 | outtolunc | HeadachesAbound: if anything i'd see it as a GUI flowchart, the prob is that with the use of macros you 'could' have thousands of 'possible' link |
| 16:40.32 | outtolunc | s |
| 16:40.43 | HeadachesAbound | replicate the dialplan language? exactly what kind of gui are you thinking about Nugget? |
| 16:40.45 | Nugget | unless by gui you just mean to make a button for each application |
| 16:41.01 | outtolunc | apps=objs |
| 16:41.07 | outtolunc | think flowchart |
| 16:41.19 | HeadachesAbound | outtolunch: i have built several flow-based apps and the thousands of possible links are easy to deal with. |
| 16:41.20 | sivana | anyone set up cvs? |
| 16:41.29 | outtolunc | k |
| 16:41.45 | Nugget | sivana: you mean set up a cvs server? nothing to it. |
| 16:42.03 | outtolunc | i understand there are enough *colors* <G> it's just when the page is filled with lines and you can't see the objects <G> |
| 16:42.14 | sivana | Nugget: ya, I'm confused on where to set the user permissions? |
| 16:42.18 | HeadachesAbound | flowcharts are just logic, much like the dialplan is just logical stops from point a to point b with a possible diversion to point q. |
| 16:42.31 | Nugget | sivana: cvs is ignorant of that. it just uses the filesystem permissions. |
| 16:43.05 | sivana | Nugget: so, I would just "chown, chgrp" to the user using it? |
| 16:43.11 | sivana | on the cvs root |
| 16:43.14 | *** join/#asterisk BullWeivel (~bullweive@12.165.66.3) |
| 16:43.22 | Nugget | users should never be directly manipulating cvsroot |
| 16:43.34 | sivana | Nugget: drwxr-xr-x 3 richard users 20 2005-05-06 02:00 cvsroot |
| 16:43.52 | sivana | like I said, I'm confused :) |
| 16:44.11 | Nugget | what are you trying to accomplish? |
| 16:44.32 | sivana | I want, pserver and 1 commiter |
| 16:44.42 | puowvip | sigh. |
| 16:44.58 | Nugget | why not just use ssh? simpler and safer if you've just got one committer. |
| 16:45.10 | sivana | ya, ssh:user@server ? |
| 16:45.15 | sean | pserver blows. |
| 16:45.21 | *** join/#asterisk vpp (~noone@host-83-146-50-131.bulldogdsl.com) |
| 16:45.27 | vpp | hi! |
| 16:45.41 | Nugget | have the user set CVS_RSH environment variable to "ssh" |
| 16:45.42 | vpp | anyone use SJphone behind nat with asterisk? |
| 16:45.51 | sivana | I'm just trying to set up the repository, I created a folder /usr/local/cvsroot and chown/chgrp to my user |
| 16:45.59 | Nugget | and CVSROOT to [user@]hostname:/path/to/cvsroot |
| 16:46.09 | sivana | ya |
| 16:46.19 | Nugget | it should "just work" at that point |
| 16:46.24 | sivana | ok :) |
| 16:46.25 | sivana | ty |
| 16:46.54 | BullWeivel | so i guess all cards don't support faxing? is it possible to have the * box receive a fax and forward it to the right person ? |
| 16:47.18 | *** join/#asterisk bowman (~rsp@62.217.44.170) |
| 16:48.03 | bowman | heya. I need to get a console channel working, with alsa sound modules from kernel 2.6 - es1371 sound chip - does anyone know how to do that? |
| 16:48.42 | opus_ | take a look a /etc/asterisk/modules.conf |
| 16:48.51 | opus_ | look for alsa |
| 16:49.41 | bowman | yeah I enabled chan_alsa.so, when I start asterisk it says "unable to open /dev/dsp, device temporarily unavailable" |
| 16:49.53 | opus_ | lsof | grep /dev/dsp |
| 16:50.26 | opus_ | perhaps you need to properly setup your sound card |
| 16:50.40 | terrapen | wooo |
| 16:50.42 | Corydon-w | Or perhaps you need to not run X on your Asterisk server |
| 16:50.56 | terrapen | my modification of Apache::Session::Store::DBI is complete and it actually works |
| 16:51.39 | *** join/#asterisk in-side (~Lowgitek@es-217-129-31-172.netvisao.pt) |
| 16:52.07 | jskcr|lappy | The caffeine concentration in the bottle of Race 2005 Energy Blast was assayed at the Chemistry Centre of Western Australia and yielded a level of 10 g/L, which is more than 60 times the concentration of caffeine in cola beverages |
| 16:52.18 | jskcr|lappy | anyone know where to pick that up, it be great for coding |
| 16:52.25 | terrapen | not for me! |
| 16:52.27 | bowman | opus_, the sound card works perfectly with all applications ;) |
| 16:52.33 | in-side | greatings from portugal |
| 16:52.39 | terrapen | i would be shaking like beavis |
| 16:52.54 | jskcr|lappy | 15g/L is considered lethal |
| 16:53.07 | terrapen | in-side: Portugal! Cool! I've always wanted to meet someone from South America! |
| 16:53.14 | jonas | fine, i want some, where do i have to sign? :) |
| 16:53.21 | in-side | terrapen: well you almost right.. |
| 16:53.21 | dtwilson | let's hope the hoards don't start mixing it with vodka a la vodka and red bull |
| 16:53.26 | in-side | I'm from Brazil rotfl |
| 16:53.30 | tzanger | I thought the LD50 was something like 15mg/kg or something |
| 16:53.40 | terrapen | (ok, it was a joke.) |
| 16:53.49 | in-side | yah I know... |
| 16:54.04 | tzanger | nope |
| 16:54.09 | tzanger | oral LD50 is 192mg/kg |
| 16:54.24 | terrapen | what is portugal like, anyway? |
| 16:54.39 | dtwilson | hot and sunny and full of germans i believe |
| 16:54.39 | Corydon-w | bowman: so let me guess... you're running your Asterisk server on your X11 desktop? |
| 16:54.40 | jonas | hmm |
| 16:54.47 | jonas | looks like you can get it only in australia |
| 16:54.48 | jonas | :/ |
| 16:54.53 | in-side | well.. it's kind of... a peace of something not very usefull |
| 16:54.56 | Corydon-w | i.e. you're SHARING your audio device |
| 16:55.02 | kFuQ | Starting May 2nd and going until June 8th, the server located at http://www.hackiis6.com/ will welcome hackers to attack it. If you can deface the web site or capture the "hidden" document, you win an X-box! Read contest rules for what does and doesn't constitute a successful hack. We've tried to be as realistic as possible in what constitutes a successful hack, and in mimicking a basic HTML and ASP.NET web site |
| 16:55.04 | jskcr|lappy | aussies sure do know how to make a caffiene drink |
| 16:55.04 | terrapen | dtwilson, i thought that was the Maldives :) |
| 16:55.10 | BullWeivel | damn... all these cards are expensive!! anyone know of a good cheap card to handle a pots line?? |
| 16:55.12 | kFuQ | rofl |
| 16:55.12 | tzanger | the lowest published lethal dose for a child though is 320mg/kg |
| 16:55.28 | jonas | i wonder if it pops up at thinkgeek at sometime |
| 16:55.44 | in-side | BullWeivel: why you don't use a spa 2000 ? |
| 16:55.47 | dtwilson | terrapen: I think it's any country with warm sun and beaches these days :) |
| 16:56.11 | in-side | Portugal sux believe me |
| 16:56.14 | opus_ | bullweivel -=Digium Wildcard X100P OEM FXO PCI Card Asterisk PBX |
| 16:56.18 | in-side | best thing here are the people |
| 16:56.19 | terrapen | damn strong Euro |
| 16:56.31 | terrapen | i want a weak euro so that i can afford to go to europe again |
| 16:56.33 | BullWeivel | spa 2000 ?? dont see that on the list. |
| 16:56.46 | in-side | BullWeivel: www.sipura.com |
| 16:56.51 | bowman | Corydon-w, hang on a second ;) |
| 16:57.49 | BullWeivel | the digium card appears to me to be going for around $130 ?? |
| 16:58.06 | opus_ | terrapen - work remotely for a company in europe, spend it in usa |
| 16:58.21 | terrapen | and pay dual taxes |
| 16:58.36 | terrapen | the US IRS makes it difficult to work overseas |
| 16:58.37 | dtwilson | terrapen: not if you use an offshore account |
| 16:58.47 | opus_ | oh. there is no sales tax where i am |
| 16:58.49 | dtwilson | ahhh you're in the US - I was presuming UK |
| 16:59.03 | terrapen | dtwilson, well, you have to pay rent somehow :) |
| 16:59.23 | terrapen | and when your income suddenly drops to nothing, yet you live in the same place, the IRS will notice :) |
| 16:59.23 | opus_ | pay in bullion |
| 16:59.24 | opus_ | :) |
| 16:59.40 | ChkDigit | What does it mean when "channel.c:304 ast_best_codec: Don't know any of 0x0 formats" |
| 16:59.49 | BullWeivel | in-side: ahh the sipura stuff is external. interesting idea. Yeah I was thinking of just something internal for a computer. suppose i could even use the $50 linksys voip as well (not sure if they work with * ) |
| 17:00.19 | opus_ | bullweivel -- they have a rom that is flashed to only call vonage/etc |
| 17:00.34 | opus_ | it would be hard unless you were a EE person |
| 17:00.59 | terrapen | i thought those can be reflashed |
| 17:01.16 | BullWeivel | opus_: they make a non flashed one. you have to be a service provider to buy it, which we are...well sorta ;) |
| 17:01.16 | terrapen | err |
| 17:01.19 | terrapen | maybe it was the cisco |
| 17:02.45 | opus_ | yeah just resolder the jtag interface, whip out your latest rom disassembler and you're off. extremely easy |
| 17:02.49 | *** join/#asterisk bowman (~rsp@62.217.44.170) |
| 17:02.50 | BullWeivel | we just opened an office up, and im looking into getting a pbx. but i need to be cheap and what all the options...so im thinking a little cheapo desktop machine and * will be the best solution :) then maybe down the road if we realy like it and are comfortable with it we will start selling it as a pbx solution to others. |
| 17:03.02 | bowman | ok. even if I have no X running, I get the same complaint from chan_alsa.so :) |
| 17:03.33 | BullWeivel | alsa was a bitch on my debian box to get running with the es card you mentioned. |
| 17:03.52 | BullWeivel | your having problems getting it to run in console right? |
| 17:03.56 | dtwilson | BullWeivel: sounds like a cheapo PC, an x101 and some budgetone grandstreams will be your cheapest option |
| 17:04.23 | bowman | BullWeivel, yeah. it tells me "device temp. unavail.", although the device is NOT in use. |
| 17:04.37 | in-side | BullWeivel: sipura devices is a good option... I have some SPA-3000 |
| 17:04.58 | in-side | it has one fxo port and one fxs, and it does well backup on pstn |
| 17:05.12 | in-side | in case of power failure and that |
| 17:05.19 | in-side | it works nicelly with asterisk |
| 17:05.28 | BullWeivel | bowman: try to get a console mp3 player to work first. |
| 17:05.47 | in-side | fxo: receive calls from pots... fxs: make calls from a pots device to voip |
| 17:05.57 | BullWeivel | bowman: also google for it... i remember having a major issue with it...i just cant remember what it was. |
| 17:06.08 | tzanger | in-side: uh |
| 17:06.32 | tzanger | FXO = connects to the Office (telco) FXS = connects to the Station or telephone Set |
| 17:06.42 | *** join/#asterisk kb1_kanobe (~krisbouti@h24-207-80-55.cst.dccnet.com) |
| 17:06.55 | opus_ | with somebody that has flash can they tell me what time they stop selling breakfast at www.bk.com |
| 17:07.20 | BullWeivel | tzanger: so what your saying is that if the desk had a standard POTS line on it... i would use a FXS port? |
| 17:07.25 | BullWeivel | err one port per phone. |
| 17:07.30 | tzanger | no |
| 17:07.34 | BullWeivel | damn |
| 17:07.36 | BullWeivel | pots phone |
| 17:07.38 | BullWeivel | sorry |
| 17:07.39 | BullWeivel | not line |
| 17:07.40 | bowman | funny enough, chan_alsa.so is not even compiled in my source tree, but can be loaded? huh? well. hrm. :D |
| 17:07.51 | tzanger | if the desk has a standard POTS line on it, you'd use an FXO port to connect to it |
| 17:08.00 | tzanger | if the desk has a standard telephone on it, you'd hook it up to an FXS port |
| 17:08.09 | BullWeivel | gotcha...thats what i meant ;) |
| 17:08.30 | in-side | tzanger: exactly what i said.. |
| 17:08.41 | in-side | I try to simplify the things for him |
| 17:08.42 | tzanger | in-side: but you said it in as confusing a manner as possible |
| 17:08.50 | tzanger | FXO/FXS has nothing to do wiht VOIP either |
| 17:09.12 | in-side | english is not my bet language sorry |
| 17:09.16 | BullWeivel | what converts it from analog to voip? the dsp right?? |
| 17:09.53 | in-side | BullWeivel: what that devices do is to connect pots and pstn to your asterisk box |
| 17:09.56 | in-side | for eg. |
| 17:10.04 | in-side | bet = best |
| 17:11.46 | BullWeivel | who is your guys favorite vendor for gear? (other then digium ;) ) |
| 17:12.06 | BullWeivel | digitnetworks.com appears to have the best pricing i have seen so far. |
| 17:12.18 | bowman | BullWeivel: mp3blaster works perfectly... |
| 17:14.06 | BullWeivel | bowman: is that console based? (i forget which console one i used to test it out) |
| 17:15.11 | opus_ | bowman what kernel version |
| 17:15.21 | bowman | Linux howhigh 2.6.8-2-686 #1 Mon Jan 24 03:58:38 EST 2005 i686 GNU/Linux |
| 17:15.37 | BullWeivel | which version of linux? |
| 17:15.41 | BullWeivel | err...flavor. |
| 17:15.42 | *** join/#asterisk holger_ (~holger@dsl-084-060-202-043.arcor-ip.net) |
| 17:15.45 | bowman | BullWeivel: yep |
| 17:15.51 | bowman | BullWeivel: Debian Testing |
| 17:16.20 | BullWeivel | well if a console program is working, then it likely has something to do with the other product your trying to use. |
| 17:16.47 | bowman | you mean asterisk? ;) |
| 17:16.48 | Blackvel | what is prefix 510? US? |
| 17:17.30 | BullWeivel | yeah... and no.. im assuming you compiled asterisk on your box?? likely debian moved where * thinks something is at. |
| 17:17.33 | *** part/#asterisk jdg (~jdg@CA03F988.adsl.mana.pf) |
| 17:19.28 | BullWeivel | im loving this :) Set of 2 - OEM X100P - FXO PCI Card $38.95 i doubt they will work for faxes though :( |
| 17:20.23 | Aze` | I'm trying a CVS HEAD .. (bristuffed) but when try to dial ... i listen double ring.. 1st real and 2th simulate by asterisk.. Any solution ? |
| 17:21.18 | MikeJ[Laptop] | Corydon-w, :) |
| 17:21.41 | *** join/#asterisk Carp1 (carp_xigon@ip-204-97-151-202.modem.logical.net) |
| 17:22.06 | Carp1 | I just setup xlite, I can call from xlite to a SIP hardphone, but I cant call from the hardphone to the soft |
| 17:22.09 | Carp1 | Any idea's? |
| 17:23.57 | Meaty | hmm, do you have created many sip user in sip.conf ? |
| 17:24.14 | Carp1 | I created a user in sip.conf for the softphone, yes/. |
| 17:24.26 | Meaty | are they each on a different user ? |
| 17:24.41 | Carp1 | I have 2 users in there. one for hard phone and one for soft |
| 17:24.44 | Meaty | Creat an other user for Xlite |
| 17:24.46 | Meaty | ok |
| 17:24.46 | Carp1 | [100] and [102] |
| 17:24.56 | Carp1 | 102 is hte user for xlite |
| 17:25.09 | *** join/#asterisk shido6 (~shido@d57-87-253.home.cgocable.net) |
| 17:25.10 | Meaty | do you have monitoring in cli ? |
| 17:25.15 | Meaty | for error message ? |
| 17:25.25 | Carp1 | like WARNINGS and NOTICES? |
| 17:25.31 | Meaty | yes |
| 17:25.34 | bowman | set verbose 1000000000000000 :) |
| 17:25.40 | Meaty | where go the call ? |
| 17:26.19 | Carp1 | Well, I just tried again and it worked :_\ |
| 17:26.21 | *** part/#asterisk ckruetze (HydraIRC@cpc3-cmbg7-5-0-cust100.cmbg.cable.ntl.com) |
| 17:26.24 | Carp1 | but then I tried once more and it was busy again |
| 17:26.40 | sean | isn't 6 the most significant verbosity? |
| 17:26.44 | opus_ | oh god, broadvoice down again |
| 17:27.19 | opus_ | hey |
| 17:27.30 | opus_ | t-mobile changed all of their customers voicemail systems today |
| 17:28.48 | Moonwick | oh? |
| 17:29.48 | DEEZED | does realtime voicemail have to be configured when you compile asterisk? |
| 17:30.09 | *** join/#asterisk Derkommissar (~alberto@66.64.215.7.nw.nuvox.net) |
| 17:30.38 | Derkommissar | there is a dialer, that dials phones directicly from MS outlook. |
| 17:30.47 | Derkommissar | Now is there one fo ximian evolution ? |
| 17:39.15 | *** join/#asterisk Ayano (~erik_leee@ppp-70-244-234-29.dsl.spfdmo.swbell.net) |
| 17:39.17 | *** join/#asterisk convey (~kvirc@63.115.106.66) |
| 17:39.49 | BullWeivel | hrmm... no faxing abilities with any of the digium cards? |
| 17:40.04 | PBXtech | define abilities :) |
| 17:40.09 | Ayano | Has anyone her ever authenticated with primus befor? |
| 17:40.50 | BullWeivel | just being able to send/receive faxes...thats what im talking about :) |
| 17:41.12 | BullWeivel | primus = packet8 right?? |
| 17:41.22 | Ayano | I can recieve, I've never tried sending. |
| 17:41.34 | BullWeivel | Ayano: which card you using? |
| 17:41.58 | Ayano | tdmp400, one fxo, and fxs |
| 17:42.19 | PBXtech | BullWeivel only thing that works is a software DSP setup like spandsp |
| 17:45.09 | Ayano | I'm going to set up a server here and test. |
| 17:45.23 | *** join/#asterisk FarrisG (~jrush@h-68-164-19-170.dllatx37.covad.net) |
| 17:47.46 | *** part/#asterisk Gand_DJ (~gandalf@ptr-207-54-104-24.ptr.terago.ca) |
| 17:59.07 | *** join/#asterisk Zipper_32 (~x@d209-121-36-44.bchsia.telus.net) |
| 17:59.55 | DEEZED | can anyone help me? I compiled the asterisk add ons, and followed the directions for realtime voicemail on the wiki, and I have reloaded asterisk. Asterisk is not using mysql, it is still using the static voicmail.conf |
| 18:00.03 | DEEZED | any ideas on what could be wrong? |
| 18:00.46 | MikeJ[Laptop] | is it using both? db and file conf? |
| 18:02.18 | DEEZED | no |
| 18:02.33 | MikeJ[Laptop] | pastebin conf please |
| 18:05.03 | DEEZED | http://pastebin.ca/11125 |
| 18:05.09 | DEEZED | is the extensions.conf |
| 18:05.22 | DEEZED | i have put mysql settings in extconfig.conf |
| 18:05.58 | *** join/#asterisk lilwookie (~zoidmeste@modemcable215.87-81-70.mc.videotron.ca) |
| 18:06.59 | MikeJ[Laptop] | y, the extconfig is the important one |
| 18:07.11 | *** join/#asterisk BoRiS (boris@S01060040ca1e5b54.wp.shawcable.net) |
| 18:07.20 | Carp1 | I just tried also, It's not working for me either |
| 18:07.25 | Carp1 | its still using voicemail.conf |
| 18:07.31 | lilwookie | hi all I was compiling spandsp's rxfax and am getting an error "t30_stats_t: Undefined" I have added GNU_SOURCE .. been looking in the lists.. any seen this? |
| 18:08.08 | MikeJ[Laptop] | never had an issue, runs great for me |
| 18:08.08 | Carp1 | if you're following the wiki, it doesnt tell you where to put your database login information |
| 18:08.08 | DEEZED | i have |
| 18:08.08 | DEEZED | voicemail => mysql,asterisk,voicemail_users |
| 18:08.14 | DEEZED | in extconfig.conf |
| 18:08.29 | DEEZED | voicemail_users is the table name of the DB names asterisk |
| 18:08.38 | MikeJ[Laptop] | ahh. I have only used the odbc realtime, not mysql.. |
| 18:08.45 | DEEZED | oh ok |
| 18:09.11 | Carp1 | Yeah, but you need a username/password to access tat database/table |
| 18:09.39 | MikeJ[Laptop] | use odbc w/ mysql |
| 18:10.07 | DEEZED | the db settings are in res_mysql.conf |
| 18:10.13 | Carp1 | Ok |
| 18:10.25 | Carp1 | Hmm, I dont have that file |
| 18:10.27 | Carp1 | interesting |
| 18:10.35 | MikeJ[Laptop] | create it |
| 18:10.37 | DEEZED | yeah i just saw it in the wiki |
| 18:10.42 | DEEZED | under realtime |
| 18:10.55 | DEEZED | its in /asterisk_addons/configs/ |
| 18:11.02 | *** join/#asterisk cjk_ (~cjk@80.92.75.4) |
| 18:13.20 | coolhp | Does anyone know of any problems with the current Zaptel CVS head ? |
| 18:15.23 | iq | Hi, need help with dialplan. Can someone check why cant I dial 011 numbers. Thanks for help! http://pastebin.ca/11128 |
| 18:16.33 | Qwell | iq: because you don't have anything in default that matches |
| 18:16.48 | tzanger | coolhp: getting zt_write failures? |
| 18:16.54 | coolhp | yeah |
| 18:17.01 | tzanger | they moved the message from DEBUG to WARN |
| 18:17.04 | tzanger | you've always had them |
| 18:17.06 | tzanger | just didn't know it |
| 18:17.28 | iq | Qwell, could be in callinfcard.conf ? |
| 18:17.29 | coolhp | Hmmm |
| 18:17.41 | coolhp | Than what can I do about it ? |
| 18:17.54 | coolhp | I'm actually getting floods of them. |
| 18:17.55 | Qwell | iq: it could be, but it still needs to be included |
| 18:17.57 | Qwell | afaik |
| 18:19.53 | iq | so if I add this: exten => _011[0-9],1,Macro(prepaid) will this help? if i add it in default context |
| 18:20.07 | tzanger | coolhp: I am not sure; I get them too and atm am at a loss to eliminate it |
| 18:20.15 | *** join/#asterisk mozrat (~mozrat@80.68.89.215) |
| 18:20.25 | coolhp | I'm sorry... that was kind of missleading from me.... let me rephrase that :-) What could generate these zt_write errors ? Could it be a clocking issue ? |
| 18:20.37 | tzanger | coolhp: no |
| 18:20.54 | Qwell | iq: sure |
| 18:20.54 | tzanger | basically the card is not able to accept the data that asterisk wants to write to it |
| 18:20.56 | mozrat | hey guys - do I need a soundcard in my * server for music on hold to work?? |
| 18:21.05 | tzanger | mozrat: no |
| 18:21.17 | *** join/#asterisk dverzolla (~dverzolla@200.186.101.19) |
| 18:21.32 | mozrat | tzanger: :( ok thanks |
| 18:21.49 | MikeJ[Laptop] | iq, what do you want to match after the 011? |
| 18:21.55 | *** join/#asterisk r0d3nt (~RatMan@wsip-24-234-241-84.lv.lv.cox.net) |
| 18:22.16 | coolhp | I've changed my signaling to 5ess instead of national and I get a few less. |
| 18:22.22 | iq | MikeJ[Laptop], anything. Just get this international number going |
| 18:22.26 | coolhp | But I'm still getting a few issues. |
| 18:22.45 | tzanger | coolhp: set your signaling to whatever the PRI is provisioned for |
| 18:22.59 | tzanger | don't just start fucking around; it was working before (i am assuming) |
| 18:23.10 | coolhp | LOL... |
| 18:23.13 | iq | MikeJ[Laptop], because these international numbers are not fixed width. will it work :) ? |
| 18:23.29 | coolhp | I'm actually using a crossover cable between a Dialogic device and my asterisk box. |
| 18:23.44 | tzanger | coolhp: well whatever it was working with before, leave it |
| 18:23.47 | tzanger | that's not he cause of the problem |
| 18:23.55 | coolhp | :-) |
| 18:23.57 | coolhp | k |
| 18:24.02 | tzanger | and I think digium is gonna get absolutely flooded with calls about this change |
| 18:24.11 | coolhp | Yeah... |
| 18:24.12 | coolhp | LOL |
| 18:24.32 | MikeJ[Laptop] | sure . |
| 18:24.51 | iq | MikeJ[Laptop], ? |
| 18:24.56 | tzanger | both of my * boxes are shit microstar boards at the mement |
| 18:25.10 | MikeJ[Laptop] | use . |
| 18:25.38 | iq | MikeJ[Laptop], [0-9].,1, ? |
| 18:25.52 | tzanger | ?? |
| 18:26.06 | tzanger | exten => _011.,1,Dial(FOO/bar/${EXTEN}) |
| 18:26.09 | tzanger | is it that difficult? |
| 18:27.09 | Qwell | tzanger: of course |
| 18:27.14 | iq | MikeJ[Laptop], thanks :) |
| 18:27.17 | tzanger | Qwell: :-) |
| 18:27.21 | iq | Qwell, thanks :) |
| 18:27.40 | iq | tzafrir, when the boss is on your ass and everything stop working then everything is difficult ;) |
| 18:27.57 | Qwell | only if you're in the wrong field... |
| 18:28.00 | tzanger | iq: true enough, but it's not that big a change and you don't have to take the box down even |
| 18:28.02 | iq | tzanger, that was for you :) |
| 18:28.16 | dverzolla | Anyone know if is possible to authenticate users in sip.conf througt ldap server? |
| 18:28.45 | iq | tzanger, ya, I agree... maybe I'm too dumb |
| 18:29.02 | iq | tzafrir, thanks for embarasing me anyways ;) |
| 18:29.09 | tzanger | iq: hey no worries |
| 18:29.10 | iq | tzafrir, shoot |
| 18:29.14 | dverzolla | some br here? |
| 18:29.20 | tzanger | you can return the favour :-) |
| 18:29.37 | iq | tzanger, lol ... I can't even type your name right |
| 18:31.05 | Carp1 | How can I answer a call from another extension? |
| 18:31.15 | Carp1 | Say I hear the phone in the office next door ringing |
| 18:31.33 | Carp1 | I want to pick up my [phone and dial 8XXX where XXX is the extension that is ringing and answer it |
| 18:31.49 | *** join/#asterisk cmk (~cmk_@p54A3C4BA.dip.t-dialin.net) |
| 18:33.42 | *** join/#asterisk |Vulture| (~V@199.227.253.212) |
| 18:33.54 | |Vulture| | anyone here know what this means: |
| 18:33.55 | |Vulture| | <PROTECTED> |
| 18:34.20 | Nuxi | Sure, it means this: NOTICE[4798]: chan_zap.c:7709 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 |
| 18:34.47 | |Vulture| | wow thanx Nuxi |
| 18:34.56 | tzanger | |Vulture|: it means that you are getting frame slips or other nasties on the PRI and the D channel messgae got corrupted |
| 18:35.40 | |Vulture| | tzanger: is it possible i have the PRI configured wrong? |
| 18:36.17 | |Vulture| | or is it only possible that it is a line problem |
| 18:36.46 | tzanger | |Vulture|: yup |
| 18:37.28 | dverzolla | Carp1, read about pickupexten |
| 18:37.45 | |Vulture| | tzanger: does it sound like I have the span="" set wrong? |
| 18:38.13 | tzanger | maybe |
| 18:38.15 | tzanger | what are you wired to |
| 18:38.43 | *** join/#asterisk ManxPower (~eric@dsl-209-16-67-160.datasync.com) |
| 18:39.23 | |Vulture| | tzanger: it was strange on the "Customer PRI Info" it doesn't say like bz2s it says |
| 18:39.42 | |Vulture| | Pri Group 18; D-Ch Grp 270-0; B-Ch Grp 271-1 |
| 18:39.47 | *** join/#asterisk onkeltimm (~chatzilla@dsl-082-083-014-147.arcor-ip.net) |
| 18:39.58 | |Vulture| | then a buncha PIC/PTC;ISCN;Switch OE info |
| 18:40.21 | Carp1 | Should music play by default when a call is parked? |
| 18:40.38 | tzanger | |Vulture|: whoa |
| 18:40.51 | |Vulture| | tzanger: huh? |
| 18:41.30 | |Vulture| | tzanger: the order from my other PRI provider told me like n2, bs2z, etc. |
| 18:41.52 | ManxPower | |Vulture|: PRI in the USA MUST be ESF/B8ZS, I think. |
| 18:42.04 | tzanger | not must |
| 18:42.06 | tzanger | just almost always is |
| 18:42.19 | |Vulture| | hmm okay |
| 18:42.21 | ManxPower | You're gonna have problems with the D Channel if you don't have a clear channel setup |
| 18:42.45 | |Vulture| | ManxPower: this is my first install... what does that mean? |
| 18:42.56 | ManxPower | |Vulture|: what does what mean? |
| 18:43.03 | |Vulture| | clear channel setup |
| 18:43.21 | ManxPower | |Vulture|: it means the line does not corrupt data sent over it. |
| 18:43.50 | ManxPower | Or more correctly, it means 64K channels instead of 56K channels |
| 18:43.52 | shido6 | where's denon |
| 18:43.54 | |Vulture| | ManxPower: oh it just means that the line from the CO is kinda iffy... it shouldn't lend to any harsh problems right? |
| 18:44.20 | denon | im around |
| 18:44.33 | ManxPower | |Vulture|: some T-1 protocols need to steal bits from the channel for signalling. PRI puts signaling on a seperate channel, so it's not needed. |
| 18:45.00 | ManxPower | |Vulture|: tell them you want ESF/B8ZS PRI set up as National ISDN 2 |
| 18:45.38 | ManxPower | With my provider I just say "gimme a PRI at $street_address" and it magically appears 14 days later. |
| 18:45.45 | ManxPower | I like my provider |
| 18:46.08 | |Vulture| | ManxPower: lol |
| 18:46.15 | denon | <PROTECTED> |
| 18:46.27 | *** join/#asterisk vpp (~noone@host-83-146-50-131.bulldogdsl.com) |
| 18:47.19 | ManxPower | Then when I'm ready to set up my side, I call my sales rep, he logs into the switch and makes any changes I need |
| 18:47.50 | ManxPower | The sales rep knows who to send the order/contract paperwork to for the order as well |
| 18:48.03 | ManxPower | We ARE the 3rd largest client at this CLEC. |
| 18:48.25 | denon | 7-11, pizza hut, ManxPower |
| 18:48.39 | jsharp | Its fun ordering circuits when you have FEMA as a client. They can issue an order to the telco and have them working round the clock to turn the circuit up. |
| 18:48.53 | |Vulture| | lol |
| 18:49.19 | jsharp | We had bellsouth turn up a DS3 in 48 hours over the weekend. |
| 18:49.23 | *** join/#asterisk r0d3nt (~RatMan@wsip-24-234-241-84.lv.lv.cox.net) |
| 18:49.31 | denon | jsharp: wow, fast as a cross-connect |
| 18:49.33 | ManxPower | jsharp: In the Good Old Days Bell did that for normal customers. |
| 18:52.22 | shido6 | denon |
| 18:54.19 | vpp | anyone use TCL scripts with asterisk? |
| 18:54.55 | mozrat | any UK PRI users here? |
| 18:59.48 | vpp | anyone? |
| 19:00.15 | sean | Bueller? |
| 19:00.36 | *** join/#asterisk jjasper (~User@h-66-112-162-139.connactivity.com) |
| 19:00.49 | tzanger | http://news.yahoo.com/s/ap/penguins_chlamydia |
| 19:00.51 | tzanger | hahaha |
| 19:01.56 | jjasper | sorry if I am in wrong group but could someone direct me to a place to request an Asterisk Consultant for a day's work in Boston? |
| 19:02.15 | Qwell | jjasper: the wiki has a list |
| 19:02.25 | jsharp | Who's been boinking Tux without wrappers? |
| 19:02.33 | jjasper | OK thank-you |
| 19:06.59 | *** join/#asterisk trimi` (Pharrel@62.162.232.171) |
| 19:07.00 | *** part/#asterisk jjasper (~User@h-66-112-162-139.connactivity.com) |
| 19:12.08 | shido6 | we had a customer come to us seeking a couple ds3's and we turned them up in 20 minutes |
| 19:12.20 | shido6 | legacy telco's just suck |
| 19:14.28 | *** join/#asterisk L|NUX (~linux@202.5.145.58) |
| 19:15.36 | Carp1 | Anyone use/used call pickup? |
| 19:16.12 | shido6 | . |
| 19:18.12 | *** join/#asterisk durex (~ironman@weber.anpa.org.br) |
| 19:18.53 | puowvip | yep |
| 19:19.32 | Carp1 | I cant get directed pickup to work |
| 19:19.53 | puowvip | sorry, I don't use that |
| 19:19.59 | Carp1 | hmm |
| 19:20.00 | Carp1 | ok |
| 19:20.18 | Carp1 | Someone should write an AnswerExt app |
| 19:20.22 | Carp1 | would If I could lol |
| 19:20.34 | Carp1 | AnswerExt(100) to answer a call ringing on ext 100 |
| 19:20.47 | drumkilla | pickup groups? |
| 19:20.54 | lilwookie | Carp1, what type of channels? |
| 19:21.31 | Carp1 | calls are coming in from IAX2 and I'm using SIP locally |
| 19:21.47 | Carp1 | I want to dial like 8101 to answer a call ringing on extension 101 |
| 19:21.49 | Carp1 | form my phone |
| 19:22.23 | lilwookie | ah, never tried it with sip, if you specify pickup group on zaps you just have to do *8 and it picks up the call |
| 19:22.42 | Carp1 | thats not what I want |
| 19:22.49 | Carp1 | I want to answer a call on a SPECIFIC extension |
| 19:23.30 | lilwookie | ah Ic, I havent done that :/ |
| 19:23.37 | *** join/#asterisk mog_home (~mog_home@user-24-236-83-199.knology.net) |
| 19:25.17 | *** join/#asterisk m0f0x (m0f0x@m0f0x.user) |
| 19:28.54 | ManxPower | Asterisk does not support Directed Call Pickup. |
| 19:29.50 | mozrat | guys in zaptel.conf span=1,0,0,ccs,hdb3,crc4... what do the zeros signify? |
| 19:30.00 | mozrat | 1,0,0 ...1=Span 1 |
| 19:32.48 | ManxPower | mozrat: the first 0 means "the card provides timing to the far end". Not good if the far end is the telco |
| 19:34.04 | *** join/#asterisk tzafrir_home (~chatzilla@bzq-80-42-55.red.bezeqint.net) |
| 19:34.07 | *** join/#asterisk Rez (lorez@lorez.staff.freenode) |
| 19:34.21 | tzanger | mozrat: you need to read |
| 19:34.34 | tzanger | it's clearly explained in zaptel.conf |
| 19:34.46 | tzanger | ManxPower: actually that's not what it means |
| 19:35.17 | tzanger | ManxPower: 0 means "internally clocked" 1 means "line clocked (i.e. sync to the recovered clock)" |
| 19:35.23 | *** join/#asterisk kb1_kanobe (~krisbouti@h24-207-80-55.cst.dccnet.com) |
| 19:35.31 | tzanger | 2 means the same as 1, but only if the span with a clock of '1' is down |
| 19:35.38 | tzanger | 3 means the same but only if 1 and 2 are down |
| 19:35.38 | tzanger | etc |
| 19:36.04 | mozrat | tzanger: I'm happy to read anything, but it's not documented what that zero means in my config file :) |
| 19:36.34 | *** part/#asterisk loud (~ariel@blaqhat.com) |
| 19:37.24 | NewSole | hey tzanger |
| 19:38.50 | bjohnson | carp1: why the heck would you want that anyway? |
| 19:40.32 | sean | bjohnson: "bob"'s phone, across the office is rining, he's out to lunch; you're expecting an important client. |
| 19:41.38 | NewSole | tzanger... u alive |
| 19:41.56 | ChkDigit | Sounds like a job for forward... |
| 19:42.02 | bjohnson | sean: what's preventing a person from answering another person's phone if they don't want them to? |
| 19:42.12 | bjohnson | sean: how about a call forwarding concept |
| 19:42.17 | bjohnson | sean: or voicemail |
| 19:42.35 | sean | bjohnson: I'm just saying that it COULD be useful. |
| 19:42.59 | sean | .. and I've seen it on other PBXs |
| 19:45.12 | *** join/#asterisk Tili (Tili@202-133-65-68-dialup.sat.net.pk) |
| 19:45.16 | ManxPower | Carp1: Asterisk does not support Directed Call Pickup. |
| 19:45.30 | kb1_kanobe | sean: this is configurable as either a 'call pickup group' (a-la a sales team) or a series of 'watched phones' (a-la a secreatary/boss relationship) on the Norstar PBX - is htis waht you're thinking of? |
| 19:45.49 | ManxPower | sean: Yes, it could be useful, and yes, it is a feature of most PBXs |
| 19:45.53 | *** join/#asterisk Sato1 (~rauleli@sato1.wizardteam.com) |
| 19:46.32 | NewSole | it is out already.... app_pickups.c |
| 19:46.37 | tzanger | hey |
| 19:46.41 | ManxPower | NewSole: CVS-HEAD? |
| 19:47.10 | ManxPower | Doesn't app_tailgateparty.c require app_pickups.c? |
| 19:47.15 | NewSole | no personal... I finished it last week |
| 19:47.36 | ManxPower | NewSole: and why is not not on the bug tracker and disclaimed? |
| 19:47.43 | NewSole | its data base driven tho |
| 19:48.09 | NewSole | cause I have not posted it yet... I like to test it first.... |
| 19:48.09 | sean | I don't remember what the PBX was. I didn't care back then. But when I wanted to pick up someone else's phone (while it was ringing, only), I could dial (some_prefix)(ext_num) |
| 19:48.33 | sean | IIRC, it was an Avaya system. |
| 19:48.47 | NewSole | i will post it this weekend... once I fix call transfer problem |
| 19:48.50 | kb1_kanobe | Directed call pickup and, yes, it can come in handy in an office environment. |
| 19:49.17 | ManxPower | Personally I think people should answer their own damn phone. |
| 19:49.29 | *** part/#asterisk bowman (~rsp@62.217.44.170) |
| 19:49.29 | opus_ | why did bob go out to lunch if he's go an important client calling him |
| 19:49.39 | *** join/#asterisk sob0l (~sobol_@uo166.internetdsl.tpnet.pl) |
| 19:49.44 | opus_ | I think they should fire bob |
| 19:49.51 | NewSole | right now its database driven... I am working on flat file but I can release database driven on weekend |
| 19:49.55 | ManxPower | And why doesn't have his calls forwarded to his cell phone if they are so important. |
| 19:50.06 | tzanger | ManxPower: it's very handy |
| 19:50.11 | opus_ | How do you implement 'followme' now? |
| 19:50.11 | tzanger | not everything is quite so black and white |
| 19:50.17 | HeadachesAbound | because bob doesn't have asterisk. |
| 19:50.30 | ManxPower | NewSole: I think there's something like ast_config or something like that that will allow you to use any config method Asterisk supports. |
| 19:50.46 | kb1_kanobe | because bob is a n00b pointy haired boss. |
| 19:50.53 | jsharp | And bob is too busy with his secretary to take the call. |
| 19:51.31 | NewSole | ya well I made it for my system acualy and it connects VIA odbc to database... and allows pickup on different servers linked |
| 19:52.00 | opus_ | no turn but bob |
| 19:52.39 | opus_ | newsole - so how does it work, does it loop through a select statement for a manager function or something? |
| 19:53.34 | NewSole | its linked like a peering system to different server |
| 19:53.50 | NewSole | and in data base stores accounts and extenions... |
| 19:54.02 | opus_ | in what context do you mean peering system.. what does the different server run |
| 19:54.10 | NewSole | and looks in database to see iff account can pick up extenion |
| 19:54.25 | opus_ | for each ring? |
| 19:55.25 | NewSole | its a patch for channel.c in ast_chan sortof like chanspy... but network based |
| 20:05.24 | Tili | which one is the best calling card app. AreskiCC ASTCC ? |
| 20:08.30 | opus_ | whats the latest voip provider thats good |
| 20:08.41 | opus_ | i'm closing my broadvoice account.. |
| 20:08.45 | shido6 | <PROTECTED> |
| 20:09.13 | jsharp | for i in `ls foo*`; do mv $i $i.mp3; done |
| 20:09.40 | shido6 | ok |
| 20:10.32 | bjohnson | btw .. done is important .. it isn't just jsharp saying he's finished |
| 20:12.13 | *** join/#asterisk my007ms (~mywaleed@62.139.173.127) |
| 20:14.12 | NewSole | this is my first time playing with TDM400 cards... any reason why it cuts out to 1 way audio... people can hear me but I can not hear them |
| 20:14.24 | shido6 | mv: when moving multiple files, last argument must be a directory |
| 20:14.24 | shido6 | Try `mv --help' for more information. |
| 20:14.39 | jsharp | You've got a directory in that list of foo* |
| 20:14.46 | shido6 | no |
| 20:14.48 | opus_ | for i in `ls foo*`; do mv $i ${i}.mp3; done |
| 20:15.16 | tzanger | if there are spaces in the files you need to do that |
| 20:15.24 | jsharp | ohyeah. |
| 20:15.32 | opus_ | for i in `ls foo*`; do mv $i '{i}.mp3' done |
| 20:15.35 | opus_ | for i in `ls foo*`; do mv $i '{i}.mp3';done |
| 20:15.43 | opus_ | $i.mp3 isn't a variable |
| 20:15.49 | opus_ | i hope it isnt:) |
| 20:15.56 | opus_ | oh shit that won't work either |
| 20:15.57 | opus_ | hehe |
| 20:17.00 | NewSole | one sec shido6 |
| 20:17.07 | shido6 | 2427 files |
| 20:17.14 | *** join/#asterisk Rick_Hunter (~rhunter@06-084.008.popsite.net) |
| 20:17.18 | tzanger | opus_: that's why I always do ; do echo mv $i $i.mp3; |
| 20:17.19 | tzanger | :-) |
| 20:17.31 | shido6 | im not touching that command |
| 20:17.44 | tzanger | no |
| 20:18.25 | opus_ | mv * /dev/null |
| 20:18.25 | tzanger | I put 'echo' beore the command I want to run |
| 20:18.25 | tzanger | and see the output |
| 20:18.25 | tzanger | opus_: won't work |
| 20:18.25 | *** join/#asterisk blankman (~blankman@c-24-61-108-24.hsd1.nh.comcast.net) |
| 20:18.48 | hardwire | hmm |
| 20:18.48 | hardwire | anybody had problems with wisips or wip-5000's talking w/ hostap? |
| 20:18.48 | opus_ | hmm, i'm way lagged |
| 20:18.58 | tzanger | opus_: you don't ping the entire channel for that |
| 20:19.04 | opus_ | sorry |
| 20:20.04 | blankman | Hey, anyone on right now that knows the how the asterisk code uses the featdmf singaling stuff? I am on the line with the tech from Global Crossing and they are saying that the asterisk instance isn't adhearing to the standard ... any help here? |
| 20:20.25 | shido6 | opus, so what do I use? |
| 20:20.57 | NewSole | I am just testing it now |
| 20:21.02 | *** part/#asterisk mbishop (~martin@mbishop.user.gentoo) |
| 20:21.44 | opus_ | for i in `ls foo*`; do echo mv $i "{i}.mp3" done |
| 20:21.47 | bjohnson | likely for i in `ls foo*`; do mv $i '$i.mp3';done |
| 20:21.52 | opus_ | read that carefully, then, remove the echo |
| 20:22.10 | my007ms | hi all |
| 20:22.13 | opus_ | no, i fucked up with the single quote thats an optimized version of mv * /dev/null |
| 20:22.15 | my007ms | i am sorry can i ask |
| 20:22.17 | my007ms | ? |
| 20:22.32 | Smi|k | is there any way an operator can manually PUSH someone within voice prompts of asterisk to a specific place in the menu tree? |
| 20:22.39 | my007ms | i need help in some thing abut asterisk |
| 20:22.39 | my007ms | > > i make my config file |
| 20:22.39 | my007ms | > > but i can not config x-lite to test it |
| 20:22.39 | my007ms | > > any one can help |
| 20:22.47 | bjohnson | Smi|k: transfer? |
| 20:22.57 | terrapen | interesting photo on foxnews.comD |
| 20:23.05 | terrapen | a car bomb detonator |
| 20:23.11 | NewSole | for i in *foo*; do mv $i $i.mp3; done |
| 20:23.19 | NewSole | just tested it |
| 20:23.24 | terrapen | i wonder what the different components are |
| 20:23.29 | terrapen | besides the FRS radio |
| 20:24.05 | terrapen | you can sure bet that US forces will jam the fuck out of the FRS freqs from now on |
| 20:24.14 | tzanger | http://bugshit.com/health/index.htm |
| 20:24.23 | opus_ | really |
| 20:24.24 | tzanger | " I just want to get the message out there: too much fried chicken causes male lactation" |
| 20:24.38 | my007ms | pleas some one help |
| 20:25.06 | terrapen | my007, there is plenty of documentation on this in the wiki |
| 20:25.07 | terrapen | ~wiki |
| 20:25.16 | jsharp | I read the "fast facts" thing as "Fast Food for Oil" program. |
| 20:25.20 | NewSole | can anyone tell me why I am getting 1 way audio on TDM400P Cards |
| 20:25.35 | jsharp | 1 Quarter Pounder == 1 Bbl of oil. |
| 20:25.45 | my007ms | can u give me just one link |
| 20:25.47 | my007ms | ? |
| 20:26.13 | *** part/#asterisk MKnight (~ATsirogia@S0106000f665a41db.vf.shawcable.net) |
| 20:27.24 | terrapen | my007, voip-info.org |
| 20:28.14 | my007ms | yes but all speek abt config asterisk no one speek abut x-lite |
| 20:28.15 | *** join/#asterisk PBXtech (~nik@67.107.241.3.ptr.us.xo.net) |
| 20:28.27 | my007ms | i know it's sample but for shame i can not |
| 20:29.58 | terrapen | its on that page, i know this |
| 20:30.04 | terrapen | also, try using Firefly |
| 20:30.10 | terrapen | i like it better than X-lite |
| 20:30.37 | terrapen | also, check out the VoicePulse Connect sample configs |
| 20:30.42 | terrapen | they will get you going |
| 20:30.43 | *** join/#asterisk BullWeivel (~bullweive@12.165.66.3) |
| 20:30.58 | my007ms | :) thanks this is answer i was search for |
| 20:31.01 | my007ms | thaks |
| 20:31.18 | BullWeivel | can you easily use a Cisco 7940 VOIP phone with * ?? I am about to buy two FSO cards, and im wondering if i need a phone or not yet. |
| 20:31.18 | Godsey | tzanger: hah what foods are pumped full of synthetic testostrone :) |
| 20:32.04 | Grooby | I heard doing the "monkey dance" around servers prevent them crashing |
| 20:32.11 | Grooby | that's a rituals I do every morning in the server room |
| 20:32.14 | ManxPower | This is my workstation. |
| 20:32.20 | ManxPower | I think Mozilla needs to be restarted. |
| 20:32.39 | BullWeivel | err...i meant FXO doh! lol |
| 20:32.46 | shido6 | hehe |
| 20:32.51 | shido6 | yeah BullWeivel you can use 7940s |
| 20:33.08 | shido6 | if you can afford the 60 |
| 20:33.10 | shido6 | get the 7960 |
| 20:33.15 | ManxPower | ~fxofxs |
| 20:33.16 | jbot | extra, extra, read all about it, fxofxs is An FXO port expects to receive dialtone and receive ring voltage. An FXS port expects to provide dialtone and provide ring voltage. |
| 20:34.00 | BullWeivel | i have a 7940 sitting here :) |
| 20:34.05 | ManxPower | BullWeivel: The Cisco phones ship with the SCCP/Skinny firmware. You don't want that. You want the SIP firmware. |
| 20:34.38 | BullWeivel | hrmm... can i flash them to SIP and then back to Skinny ? |
| 20:34.45 | greeneggsnham | BullWeivel: yeah |
| 20:34.52 | greeneggsnham | why would you want skinny though? |
| 20:35.03 | greeneggsnham | SIP is far superior :D |
| 20:35.04 | BullWeivel | cool.. my main job i maintain the VOIP system...so im going to borrow a phone for a bit :) |
| 20:35.13 | greeneggsnham | ah |
| 20:35.16 | BullWeivel | its a cisco system...so thats why |
| 20:35.35 | greeneggsnham | i don't think Cisco has come out with sip fw for the 7920 though :( |
| 20:35.43 | greeneggsnham | it's sitting around collecting dust |
| 20:35.48 | greeneggsnham | i should probably throw it up on ebay |
| 20:36.22 | BullWeivel | my next question... are digit network cards decent? Thinking of going this route for my FSO for now: Set of 2 - OEM X100P - FXO PCI Card |
| 20:36.36 | ManxPower | greeneggsnham: you could always donate it to someone working on chan_sccp or chan_skinny |
| 20:36.37 | DEEZED | does anyone have realtime voicemail using mysql working? |
| 20:36.39 | BullWeivel | damn i did that again!! FXO FXO FXO |
| 20:37.26 | greeneggsnham | are they in need of sccp hardware? |
| 20:37.44 | shido6 | not unless you're forced to use it by your boxx |
| 20:37.46 | shido6 | boss |
| 20:39.24 | ManxPower | greeneggsnham: they usually are. Contact the maintainers of both channel drivers and see if they currently have that model |
| 20:43.12 | *** join/#asterisk Romik (~romik@1.fix.netvision.net.il) |
| 20:43.35 | *** join/#asterisk MikeJ[Laptop] (~jayden@pcp02795302pcs.roylok01.mi.comcast.net) |
| 20:44.30 | DEEZED | does G.729 run on athlon processors? |
| 20:44.37 | DEEZED | amd* |
| 20:45.01 | CoaxD | um |
| 20:45.12 | CoaxD | codec has zero to do with processor type |
| 20:45.35 | my007ms | ManxPower can u see #asterisk-stable i do what u say :) |
| 20:45.48 | RaYmAn-Bx | CoaxD: considering it's a binary codec (at least from Digium afaik?) it does have something to do with it :P |
| 20:45.56 | DEEZED | i figured.. but when the doc was mentioning intel libraries i had to wonder |
| 20:46.10 | PBXtech | they have athlon versions |
| 20:46.11 | CoaxD | Rayman-BX: Yes. Correct. However, if the particular processor in question is an x86, it'll work |
| 20:46.12 | opus_ | is telasip good? |
| 20:47.44 | RaYmAn-Bx | CoaxD: mm, I do agree, but it does depend on optimizations used etc (but I'm just nitpicking now) |
| 20:48.41 | Smi|k | is there any way an operator can manually PUSH someone within voice prompts of asterisk to a specific place in the menu tree? |
| 20:48.41 | my007ms | hi all |
| 20:48.45 | my007ms | can i speek |
| 20:48.45 | my007ms | ? |
| 20:48.49 | my007ms | :) |
| 20:48.58 | my007ms | ok i have litel problem |
| 20:49.08 | my007ms | i try runing asterisk |
| 20:49.14 | my007ms | and it's run now |
| 20:49.34 | my007ms | but i need to try this by any softphone |
| 20:49.45 | my007ms | i can not config x-lite |
| 20:49.56 | Smi|k | I would like to contract someone to help make this possible, please msg me if you need more details |
| 20:49.59 | my007ms | can some one help |
| 20:50.42 | ChkDigit | Smi|k: Can't that be done with something like Asternic? |
| 20:51.10 | kb1_kanobe | Smi|k: explain a bit further? how would this be used? |
| 20:51.26 | Smi|k | I want to listen to people who call the asterisk system |
| 20:51.41 | Smi|k | if they yell "you stupid machine, let me talk to someone in sales" into the receiver |
| 20:52.08 | Smi|k | I want to click a link on my web browser that transfers that caller to extension X |
| 20:52.15 | ChkDigit | Ok, if you're listening to the call, why the need for the IVR? |
| 20:52.42 | Smi|k | extension X says "you have reached our sales department, for tom press 1, for Jill press 2, etc..etc.. |
| 20:53.29 | opus_ | perhaps you could make an agi script look for a flag file or semaphore in a directory every few seconds and write some web app that creates that file. |
| 20:54.48 | jakepdev | Smi|k - you can use asterisk manager to do that |
| 20:55.00 | jakepdev | no polling needed |
| 20:55.40 | ChkDigit | my007ms: I can't even understand what your problem is. Sorry. |
| 20:56.49 | *** join/#asterisk Zaw (zaw@zaw.subneural.net) |
| 20:56.59 | jakepdev | my007ms - try firefly http://www.voip-info.org/wiki-Firefly |
| 20:57.02 | my007ms | my probelm |
| 20:57.17 | my007ms | yes but this is not work with asterisk |
| 20:57.26 | my007ms | i try to test my server |
| 20:57.26 | jakepdev | that is 100% incorrect |
| 20:57.31 | jakepdev | i use it right now |
| 20:57.32 | my007ms | :) |
| 20:57.44 | jakepdev | configure iax.conf and extensions.conf |
| 20:58.14 | PBXtech | damn it this box keeps crashing! |
| 20:58.25 | my007ms | look now my server run in 62.139.6.108 |
| 20:58.26 | my007ms | ok |
| 20:58.36 | *** join/#asterisk Marlow (~martin@cerberus.bluetree.ie) |
| 20:59.23 | my007ms | <PROTECTED> |
| 20:59.24 | my007ms | ? |
| 20:59.29 | jakepdev | no |
| 20:59.36 | jakepdev | get the 3rd party one |
| 21:00.09 | jakepdev | look at http://www.voip-info.org/wiki-Firefly again |
| 21:00.43 | my007ms | ok i will try |
| 21:00.48 | my007ms | thanks |
| 21:00.49 | jakepdev | i already hit your box |
| 21:00.53 | jakepdev | it answered |
| 21:01.14 | jakepdev | just said invalid user name - all this was done inside firefly |
| 21:01.24 | my007ms | i will tel u username and one password and u try |
| 21:01.42 | jakepdev | ok |
| 21:02.46 | jakepdev | gotta run soon |
| 21:02.49 | jakepdev | you gonna send it? |
| 21:02.52 | PBXtech | is it * default to log the verbose messages? damn thats odd. wonder if this this AMP |
| 21:03.11 | my007ms | secret is 9overthruster7 |
| 21:03.12 | jakepdev | in the CLI? |
| 21:03.17 | my007ms | username is 2000 |
| 21:03.20 | jakepdev | ^^ for PBXtech |
| 21:03.30 | my007ms | :) |
| 21:03.36 | my007ms | i see |
| 21:03.45 | my007ms | it start work |
| 21:03.52 | my007ms | i like this screen |
| 21:04.06 | my007ms | but why it's not start with me |
| 21:04.40 | jakepdev | cause you're probably using the wrong version of Firefly |
| 21:04.47 | jakepdev | you need to use the 3rd party version |
| 21:04.58 | jakepdev | and use IAX |
| 21:05.00 | jakepdev | then it'll work |
| 21:05.09 | my007ms | but i want sip |
| 21:05.17 | jakepdev | why? |
| 21:05.25 | *** join/#asterisk techie (gus@asterisk.horizonte.us) |
| 21:05.25 | jakepdev | IAX is a lot easier |
| 21:05.47 | jakepdev | one port for everything |
| 21:05.48 | my007ms | what is batter in most softphone |
| 21:05.51 | jakepdev | IAX |
| 21:06.01 | jakepdev | use SIP if your phone requires it only |
| 21:06.09 | my007ms | hmm |
| 21:06.26 | my007ms | great u was so helpfull to me |
| 21:06.31 | jakepdev | np |
| 21:06.31 | PatrickDK | iax doesn't support audio rerouting on call transfers though |
| 21:06.38 | *** join/#asterisk mbaron (~mbaron@AVelizy-154-1-20-123.w82-124.abo.wanadoo.fr) |
| 21:06.39 | *** part/#asterisk Zipper_32 (~x@d209-121-36-44.bchsia.telus.net) |
| 21:06.40 | my007ms | i will take this info and start agian |
| 21:06.51 | my007ms | :( |
| 21:07.27 | my007ms | but i need to built good PBX have all thing like transfers call |
| 21:07.32 | Romik | we have incomming call on asterisk that ring 10 extentions, but some people make *63 to forward their calls to mobiles, how we can ring all extentions except that forwarded to external phones? |
| 21:07.34 | PatrickDK | I use sip phone -> asterisk -> iax -> everyone else |
| 21:07.48 | PatrickDK | my007ms, iax does transfer the call |
| 21:08.03 | my007ms | but u was say it's not |
| 21:08.04 | PatrickDK | but once you connect to asterisk using iax, the audio will always go through that asterisk box |
| 21:08.12 | PatrickDK | on sip, if you transfer the call t another box |
| 21:08.24 | PatrickDK | the audio will skip that asteirks box and go right to the other one |
| 21:08.32 | PatrickDK | not using up extra bandwidth |
| 21:08.33 | my007ms | hmm |
| 21:08.47 | my007ms | so IAX is more good |
| 21:08.57 | jakepdev | it really depends on your app |
| 21:09.04 | PatrickDK | if you only have one asterisk box and internet connection, it doesn't matter |
| 21:09.26 | my007ms | yes this is my topolge |
| 21:09.56 | my007ms | one asterisk i just need to make it good PBX for intrnel used |
| 21:10.17 | my007ms | using softphone |
| 21:10.30 | jakepdev | IAX will work just fine for your purposes |
| 21:10.35 | durex | my007ms you`ll use asterisk only for internal use? no connection to POTS or VoIP provider? |
| 21:10.41 | my007ms | so IAX is more good for me if i understand |
| 21:11.01 | *** join/#asterisk [KFC]Jupiter (~C@69-163-201-17.atlsfl.adelphia.net) |
| 21:11.02 | my007ms | yes |
| 21:11.04 | my007ms | can i ? |
| 21:11.08 | [KFC]Jupiter | Hey - I have a question |
| 21:11.18 | jakepdev | my007ms - you can |
| 21:11.23 | durex | my007ms of course |
| 21:11.33 | jakepdev | [KFC]Jupiter - don't need to preface :) just ask |
| 21:11.38 | *** join/#asterisk xeaded (~Nafas@69-88-216-229.thewavz.com) |
| 21:11.41 | my007ms | i test it firest if it work well i will make it my PBX |
| 21:11.47 | durex | my007ms about IAX, will work perfectly in your topology |
| 21:11.48 | [KFC]Jupiter | OK I am downloading the asterisk@home |
| 21:12.17 | my007ms | and what is need for AMP |
| 21:12.18 | my007ms | ? |
| 21:12.35 | durex | my007ms nothing, just install it |
| 21:12.40 | my007ms | :) |
| 21:12.42 | durex | my007ms what OS are u running? |
| 21:12.48 | my007ms | core 3 |
| 21:12.48 | jakepdev | my007ms - you may not need AMP if you are managing it and understand asterisk well enough |
| 21:12.59 | durex | my007ms but AMP requires a lot of other packages, like mysql, apache... |
| 21:13.07 | my007ms | i read in doc to make config file |
| 21:13.13 | durex | my007ms core 3? |
| 21:13.17 | durex | fedora? |
| 21:13.18 | my007ms | yes |
| 21:13.27 | *** join/#asterisk CleanerX (~nix@nat-ph3-wh.rz.uni-karlsruhe.de) |
| 21:13.59 | my007ms | so i can run asterisk without internet conction at all |
| 21:14.01 | my007ms | ? |
| 21:14.06 | jakepdev | yes |
| 21:14.08 | durex | my007ms yes |
| 21:14.24 | jakepdev | but updating is more challenging |
| 21:14.24 | my007ms | but i see in IAX.conf |
| 21:14.50 | my007ms | that i can not |
| 21:14.50 | my007ms | ? |
| 21:15.13 | DEEZED | If i have a asterisk box that uses an IAX providor, and I want to use g.729, is that just 1 channel making it $10? |
| 21:15.22 | jakepdev | you need an internet connection to connect to an IAX server or softphone on the Internet |
| 21:15.40 | jakepdev | DEEZED - that sounds correct - you can call Digium to confirm |
| 21:15.56 | ManxPower | DEEZED: Trust me, get at least 2 channels just in case |
| 21:15.57 | jakepdev | but it's only a single channel |
| 21:16.18 | ManxPower | DeeJayTwo: What if you want to make more than 1 call |
| 21:16.18 | xeaded | I have an asterisk box that has been running solid since May of last year with 3 X100P cards. I had a problem with echo when I was setting it up and had to turn down the gain some to fix it. The lines can be a little quiet at times. Would an FXO gateway fix this? |
| 21:16.19 | my007ms | but what is job of IAX server if i just need to used in lan |
| 21:16.31 | DEEZED | but even though i can handle 20+ concurrent calls, its still one channel right? |
| 21:16.52 | jakepdev | my007ms - don't need it then |
| 21:16.55 | ManxPower | DEEZED: You need 1 G729 license for each G729 call you have happening AT THE SAME TIME. |
| 21:17.05 | durex | my007ms u have to ask to yourself ;-) |
| 21:17.15 | ManxPower | At one point I think voicemail used an extra license, but I don't know if that was fixed or not |
| 21:17.25 | my007ms | and try user in IAX.conf like ? example |
| 21:17.25 | shido6 | :) |
| 21:17.37 | DEEZED | well that is kind of hard to control.. going to call digium |
| 21:17.51 | durex | gnight for all |
| 21:18.00 | durex | it's time to go ;-) |
| 21:18.41 | jakepdev | my007ms - here's a real good page on IAX.conf http://www.voip-info.org/wiki-Asterisk+config+iax.conf |
| 21:19.35 | my007ms | ok last q so i my asterisk can run in my lan without any internet conction |
| 21:19.57 | my007ms | and without any other software |
| 21:20.09 | [KFC]Jupiter | Shido6 are you here? |
| 21:20.24 | my007ms | i mean i don't have to make " iax server " |
| 21:20.28 | jakepdev | my007ms - that is correct |
| 21:20.52 | my007ms | this is great i realy love asterisk |
| 21:21.01 | jakepdev | and it loves you too |
| 21:21.10 | my007ms | :) |
| 21:21.44 | [KFC]Jupiter | Can someone help me? |
| 21:21.55 | *** part/#asterisk Patrick^ (~patrickm@birch4.mountaincable.net) |
| 21:21.56 | jakepdev | [KFC]Jupiter - please |
| 21:22.05 | jakepdev | [KFC]Jupiter - you don't need to ask to ask |
| 21:22.23 | [KFC]Jupiter | I want to use my VOIP with asterisk@home but I don't know how to "physically" connect things together. |
| 21:22.40 | [KFC]Jupiter | The docs don't really explain that area. |
| 21:22.42 | jakepdev | physically if using VOIP, you just need a LAN cable |
| 21:22.45 | shido6 | what hardware do you have? |
| 21:22.49 | shido6 | get a rj45 cable |
| 21:22.52 | shido6 | a rj11 cable |
| 21:22.52 | [KFC]Jupiter | yes |
| 21:22.54 | [KFC]Jupiter | I have all of that |
| 21:22.55 | shido6 | rj11 for your ata |
| 21:22.57 | [KFC]Jupiter | I have an ATA Box |
| 21:23.00 | shido6 | rj45 for your ata as well |
| 21:23.10 | [KFC]Jupiter | Rj11 is a phone line connector, right? |
| 21:23.13 | jakepdev | yes |
| 21:23.13 | shido6 | dont worry the rh45 wont fit in the rj11 jack |
| 21:23.25 | jakepdev | but vice versa will work |
| 21:23.29 | shido6 | rmaen an sandwhich in the way of keys |
| 21:23.41 | jakepdev | my soda can fits in my cd tray |
| 21:23.47 | [KFC]Jupiter | ok so if I have a Computer with a LAN card and an ATA box and a phone how would it be connected? |
| 21:23.57 | shido6 | err |
| 21:23.59 | shido6 | no |
| 21:23.59 | ManxPower | Just before you are careful. If the plug is twisted, it can get stuck in the RJ45 |
| 21:24.13 | ManxPower | I've had that happen. Still have not gotten the damn thing out of the port. |
| 21:24.21 | shido6 | plug the ata in your switch unless it has 2 ports for ethernet and 1 / 2 ports for fxs / handset (analog phone) |
| 21:24.27 | jakepdev | call in the surgeons |
| 21:24.31 | [KFC]Jupiter | My ATA box |
| 21:24.50 | [KFC]Jupiter | Has INTERNET INput and PC OUTPUT (Like a router) and a LINE 1 and LINE 2 Phone Connection |
| 21:25.05 | shido6 | internet IN comes from your switch |
| 21:25.10 | [KFC]Jupiter | Router |
| 21:25.12 | [KFC]Jupiter | Right? |
| 21:25.19 | [KFC]Jupiter | Where ever the internet is comin from? |
| 21:25.40 | shido6 | ruh roh, no switch? yes where the internet is coming from |
| 21:25.52 | [KFC]Jupiter | Well what do you mean Switch?? |
| 21:26.10 | [KFC]Jupiter | Why do I need a switch if I have a router? |
| 21:26.29 | shido6 | well some networks have a router then have a switch connected to to allow multiple puters or ip phones or both to use the single ip from the isp |
| 21:26.40 | [KFC]Jupiter | Well I have a CABLE modem |
| 21:26.43 | [KFC]Jupiter | then a router |
| 21:26.50 | shido6 | so plug the ata into the router |
| 21:26.53 | [KFC]Jupiter | ok |
| 21:26.55 | shido6 | :) |
| 21:26.59 | shido6 | that was easy |
| 21:27.01 | shido6 | "click" |
| 21:27.01 | [KFC]Jupiter | then the PC output to pc input? |
| 21:27.12 | shido6 | hehe, yeah you got it |
| 21:27.14 | *** join/#asterisk darby_t (~tom@dmt146.neoplus.adsl.tpnet.pl) |
| 21:27.28 | shido6 | hate the terminology tho :) |
| 21:27.29 | [KFC]Jupiter | Well so then the PBX pc will work with the VOIP Phone |
| 21:27.43 | my007ms | now i have core 3 with 2.6 kernel is this problen with asterisk i was read it's run in 2.4 |
| 21:27.46 | [KFC]Jupiter | It will communicate through the ATA box it is connected through? |
| 21:27.54 | shido6 | the PBX should be connected to the router, too |
| 21:28.06 | [KFC]Jupiter | So I need 2 network cards? |
| 21:28.09 | syle | manxpower i think you smoke alot of weed to get a rj11 stuck in a rj45 hehe |
| 21:28.10 | shido6 | use the ata like you would any other ethernet device |
| 21:28.12 | shido6 | no |
| 21:28.13 | ManxPower | Deflowering Asterisk virgins again I see, Shido. |
| 21:28.26 | ManxPower | syle: no comment |
| 21:28.33 | shido6 | I need some napkins or something there's blood all over the place, ManxPower |
| 21:28.39 | ManxPower | shido6: eewww |
| 21:29.01 | my007ms | now i have core 3 with 2.6 kernel is this problen with asterisk i was read it's run in 2.4 |
| 21:29.10 | shido6 | [KFC]Jupiter, relax... take a deep breath... the pbx/pc will connect to your router in 1 port |
| 21:29.16 | ManxPower | my007ms: Asterisk run on 2.4 or 2.6 |
| 21:29.17 | shido6 | your ata will connect to the SAME router in another port |
| 21:29.30 | shido6 | the cool thing about a switch is that you only have a few ports on your router, right? |
| 21:29.39 | [KFC]Jupiter | Well |
| 21:29.45 | [KFC]Jupiter | I am on a LAN/WiFi network |
| 21:29.46 | shido6 | you can add a switch like a 4 port, 8 port, 16 port, etc to add more ports to your router |
| 21:29.48 | [KFC]Jupiter | yeah |
| 21:29.49 | [KFC]Jupiter | but |
| 21:29.52 | my007ms | do i need kernel src |
| 21:29.52 | [KFC]Jupiter | Here is my setup |
| 21:29.58 | syle | my007ms: http://voip-info.org/wiki-Asterisk+Linux+Fedora |
| 21:30.23 | shido6 | shoulda dropped an egg in my ramen |
| 21:30.36 | [KFC]Jupiter | CABLE INTERNET > LINKSYS ROUTER > LINKSYS WIRELESS GATEWAY > WINKSYS WIRELESS REPEATER > WIFI CARD > PC |
| 21:30.41 | my007ms | :) |
| 21:30.55 | shido6 | my eyes are bleeding, thats called shouting, chill with the caps |
| 21:31.12 | [KFC]Jupiter | Well I did that to show the "map" of my LAN |
| 21:31.16 | *** join/#asterisk MatsK (~matsk@66.80-202-60.nextgentel.com) |
| 21:31.17 | *** join/#asterisk Blissex (~Blissex@82-69-39-138.dsl.in-addr.zen.co.uk) |
| 21:31.21 | shido6 | great map |
| 21:31.25 | [KFC]Jupiter | I though it would be easier to read that wya. |
| 21:31.33 | shido6 | where can I get a winksys? |
| 21:31.37 | [KFC]Jupiter | Linksys |
| 21:31.41 | [KFC]Jupiter | lol |
| 21:31.42 | shido6 | just messin |
| 21:31.42 | Smi|k | anyone have experience with editing of code for Asternci |
| 21:31.46 | Smi|k | Asternic |
| 21:32.11 | shido6 | i pray you dont need to have anyone register to your asterisk box from the otherside of your nat |
| 21:32.13 | [KFC]Jupiter | ok so now I have a "free" LAN card in my PC that I use to hook up my ATA box with a crossover cable |
| 21:32.14 | bkw_ | what the hell is asternic? |
| 21:32.28 | [KFC]Jupiter | No this is for just my VOIP phone. |
| 21:32.36 | Smi|k | I'm sure someone has more experience than bkw :) |
| 21:32.50 | file | bkw is sexy like a coconut! |
| 21:33.00 | Smi|k | http://www.asternic.org/ |
| 21:33.12 | *** join/#asterisk [Latre] (~latre@148.233.19.133) |
| 21:33.15 | [KFC]Jupiter | shido6 : I am going to make a visual drawing of my LAN and explain the details in it so it will help you understand much better! |
| 21:33.19 | *** part/#asterisk Grooby (~Grooby@12.22.232.212) |
| 21:33.40 | [Latre] | some knows what is this error: [cdr_custom.so]May 6 12:33:14 WARNING[3811]: loader.c:258 ast_load_resource: /usr/lib/asterisk/modules/cdr_custom.so: undefined symbol: ast_config_load |
| 21:33.54 | [Latre] | <PROTECTED> |
| 21:35.02 | ManxPower | [Latre]: noload => cdr_custom.so in /etc/asterisk/modules.conf |
| 21:35.39 | ManxPower | file: That I agree with. |
| 21:36.48 | ManxPower | bkw_: asternic.org is where Flash Operator Panel lives. |
| 21:36.50 | bkw_ | ast_config_load not found? |
| 21:37.01 | bkw_ | something else is totally wrong |
| 21:37.04 | bkw_ | has to be |
| 21:37.08 | bkw_ | for that to not be found |
| 21:37.15 | bkw_ | sounds like he moved from head to stable |
| 21:37.16 | bkw_ | maybe? |
| 21:37.47 | ManxPower | bkw_: if so, he didn't follow the instructions at the end of "make install" for 1.0.x |
| 21:37.49 | [Latre] | now is this error /usr/lib/asterisk/modules/app_chanspy.so: undefined symbol: ast_bridged_channel |
| 21:38.15 | *** join/#asterisk pbx123 (~aponph@63.81.136.190) |
| 21:38.21 | bkw_ | ManxPower, and you think people actually read? |
| 21:38.24 | shido6 | never lend anything you want back |
| 21:38.52 | ManxPower | shido6: Can I borrod $100? |
| 21:39.02 | ManxPower | bkw_: the ones that don't go away pretty fast. |
| 21:39.07 | outtolunc | reminds me i need to get my sander back |
| 21:39.14 | bkw_ | sander? |
| 21:39.15 | ManxPower | or the get on my /ignore list, I never know which. 8-) |
| 21:39.16 | bkw_ | wtf |
| 21:39.26 | outtolunc | yeah, let a buddy use it a year ago |
| 21:39.29 | ManxPower | A geek with power tools? |
| 21:39.37 | outtolunc | nods |
| 21:39.41 | bkw_ | ManxPower, he uses it for the vibrations |
| 21:39.43 | *** join/#asterisk Nethab (~chatzilla@adsl-67-113-141-170.dsl.sntc01.pacbell.net) |
| 21:39.46 | ManxPower | power tools are for lesbians! |
| 21:39.49 | file | in a perfect world, we'd all have eachother on our ignore list |
| 21:39.51 | outtolunc | haha |
| 21:39.57 | ManxPower | file: you got that right! |
| 21:40.09 | ManxPower | Hmm..that didn't sound right. |
| 21:40.10 | pbx123 | hi everyone |
| 21:40.20 | pbx123 | i was able to setup SIP |
| 21:40.21 | ManxPower | bkw_: put me on your /ignore list please. |
| 21:40.30 | pbx123 | however, |
| 21:40.43 | pbx123 | my asterisk is unidirectional, |
| 21:40.44 | bkw_ | really? |
| 21:40.48 | bkw_ | I need your permission to do that? |
| 21:40.51 | bkw_ | flippin moron |
| 21:40.53 | pbx123 | other party can't hear me.. |
| 21:40.56 | pbx123 | pls help |
| 21:41.09 | ManxPower | Much better. |
| 21:41.16 | my007ms | what is make ppl pay for sip service while thy can make it free |
| 21:41.23 | file | pbx123: lemme guess, NAT? |
| 21:41.53 | Nethab | cause it's hard, they'd rather pay someone to do it for them |
| 21:42.44 | my007ms | what is hard make to box asteris speek conect |
| 21:42.48 | Smi|k | not in my case, I dont know how to make it free |
| 21:42.57 | Smi|k | can someone explain how I can get a bank of incoming #'s in USA? |
| 21:43.36 | *** join/#asterisk r0d3nt (~RatMan@wsip-24-234-241-84.lv.lv.cox.net) |
| 21:43.52 | *** join/#asterisk apardo (~apardo@80.26.165.255) |
| 21:44.00 | [KFC]Jupiter | shido6: my ATA box has SIP |
| 21:44.00 | Marlow | my007ms: work = money |
| 21:44.00 | *** part/#asterisk mbaron (~mbaron@AVelizy-154-1-20-123.w82-124.abo.wanadoo.fr) |
| 21:44.07 | syle | you make it free by you and your friends sip peering off same server |
| 21:44.17 | [KFC]Jupiter | Power = Money |
| 21:44.21 | Marlow | my007ms: nothing is free .. either you spend the work and time .. or the money |
| 21:44.30 | my007ms | :) |
| 21:44.40 | Marlow | my007ms: and while you spend the work and time you don't earn money :) |
| 21:45.07 | my007ms | what money u get from here |
| 21:45.08 | my007ms | ? |
| 21:45.23 | my007ms | u spend now time where is the money come from |
| 21:45.24 | my007ms | ? |
| 21:45.38 | *** join/#asterisk jeffgus (~jeffgus@alpha.zimage.com) |
| 21:45.39 | Nethab | if you spend your time, your not making money |
| 21:45.54 | Nethab | so time, equals not making money |
| 21:46.09 | Marlow | means time = money |
| 21:46.24 | Marlow | and that is the SIP service that isn't free, no matter how you put it .. |
| 21:46.35 | my007ms | enjoy = life that it i think |
| 21:46.43 | my007ms | u enjoy u do any thing |
| 21:46.57 | Marlow | surely .. but that is personal preference . |
| 21:47.23 | Marlow | my007ms: there are enough people, that want to make money on that technology, don't have the interest in it though .. |
| 21:47.27 | my007ms | and u enjoy here |
| 21:47.32 | Marlow | my007ms: they instead pay for it |
| 21:48.31 | *** join/#asterisk MrClean (~seabrook@store-fw.porchlight.ca) |
| 21:49.25 | [Latre] | bkw_: you right i downgrade and not delete modules......sorry....stupid question! |
| 21:50.11 | ManxPower | [Latre]: next time READ the message on your screen when "make install" finishes |
| 21:52.33 | [Latre] | i dont remember install asterisk in that machine......so....didnt read anything..........only execute that command |
| 21:52.36 | [Latre] | :S |
| 21:52.37 | [Latre] | :P |
| 21:53.39 | shido6 | anyone own an xbox? |
| 21:53.50 | MrClean | Problem: agents *sometimes* need to aknowledge incoming calls with the # before they are bridged. It seems happen for maybe 1/10 of all incoming calls (though it's essentially random). Does anyone know why this might be happening? |
| 21:54.23 | *** join/#asterisk likwid-- (likwid@nc-67-77-248-114.dyn.sprint-hsd.net) |
| 21:54.26 | *** part/#asterisk Cresl1n (~matt@216.207.245.23) |
| 21:56.03 | Marlow | MrClean: did you "ackcall=no" ? |
| 21:56.22 | MrClean | Yes |
| 21:56.48 | Marlow | and what version of asterisk ? |
| 21:57.05 | *** part/#asterisk lilwookie (~zoidmeste@modemcable215.87-81-70.mc.videotron.ca) |
| 21:57.38 | Marlow | oh .. wait .. # on incoming calls? |
| 21:57.56 | MrClean | Version, 1.0.7. Yeah, incoming calls. With AgentCallbackLogin. |
| 21:57.58 | Marlow | that's not even a feature :) |
| 21:58.52 | [KFC]Jupiter | Shido6: see me |
| 22:03.55 | Nethab | no your invisib;e |
| 22:05.19 | my007ms | i find webmin file but it have probelm any one know the auther of this |
| 22:05.27 | my007ms | for asterisk |
| 22:05.53 | file | it's not supported, hasn't been updated in many years, give up now and keep your sanity |
| 22:06.21 | *** join/#asterisk Goshen (~Goshen@70-57-80-147.slkc.qwest.net) |
| 22:06.50 | Nethab | i keep mine in a drawer |
| 22:10.04 | Nethab | you went in my drawers |
| 22:10.17 | Nethab | that's somehow disturbing, and rewarding at the same time |
| 22:10.20 | file | you better believe it |
| 22:10.23 | *** join/#asterisk meppl (mephisto@p54AAF729.dip.t-dialin.net) |
| 22:13.16 | *** join/#asterisk BullWeivez (~BullWeive@12.165.66.3) |
| 22:14.10 | BullWeivez | anyone seen anything that runs asterisks....but when you mount it on the wall it actualy looks like a PBX for an office :) just curious...got pics ?? |
| 22:16.07 | BullWeivez | looks like from my research www.digitnetworks.com has the best pricing around for * compatible gear. anyone know of anywhere else that has better pricing? |
| 22:16.13 | Marlow | BullWeivez: take a ITX board in a plastic case :) |
| 22:17.22 | Nethab | put your right foot in, put your right foot out, put your right foot in, and hammer it to the wall |
| 22:17.28 | *** join/#asterisk PBXtech (~nik@67.107.241.3.ptr.us.xo.net) |
| 22:17.46 | *** part/#asterisk grolloj (~chatzilla@slim-eth0.horizonlive.net) |
| 22:17.49 | Nethab | what does a "PBX for an office" look like anyway |
| 22:17.53 | PBXtech | whats the shell command to tell me what version of firmware is on a digium card? |
| 22:18.00 | PBXtech | what is PBX for an office? |
| 22:18.26 | Nethab | he wanted to know if there were any computers that looked like a PBX for an office that runs astersik |
| 22:18.31 | BullWeivez | PBX for an office...usualy mounted on a wall and looks like a big plastic case with bunches of cables plugging into it :) |
| 22:18.39 | PBXtech | hah |
| 22:18.52 | Nethab | rack mount my friend, the only way to go |
| 22:19.01 | PBXtech | now days for sure |
| 22:19.17 | BullWeivez | just thinking of office solutions... if i realy like what im about to start doing with * then I might start trying to sell it to local companies. |
| 22:19.30 | Nethab | the idea is, the computer doesn't need all those wires coming out, it's all over ethernet |
| 22:19.33 | BullWeivez | but most of them are use to seeing a pbx looking box mounted on their office wall. |
| 22:19.38 | PBXtech | you and everyone else :) |
| 22:19.49 | PBXtech | [BullWeivel]: says who? |
| 22:19.57 | PBXtech | :) |
| 22:20.09 | Marlow | most often that box is hidden away in the server room |
| 22:20.16 | BullWeivez | i just dont want to walk up with a computer... a POE switch... and run with it from there ;) |
| 22:20.20 | Marlow | andy anyhow ... who wants that thing on the wall ? |
| 22:20.25 | Nethab | or the janitors closet, *cough* bob |
| 22:20.28 | PBXtech | ide rather have a 1U HP server than a fat box on the wall |
| 22:20.30 | BullWeivez | server room? what server room... im talking small offices. |
| 22:20.42 | Marlow | nethab: you haven't seen the wiring room in this building ... |
| 22:20.46 | BullWeivez | exactly... janitors closet. the womens bathroom... etc. |
| 22:20.55 | Marlow | nethab: 6 generations of cabling AND the janitors stuff |
| 22:21.01 | PBXtech | most PoE are rack mounted also |
| 22:21.03 | Nethab | the womens bathroom lol, arent' they on the phone enough |
| 22:21.12 | BullWeivez | hehehe |
| 22:21.19 | PBXtech | plus think of the service potential |
| 22:21.25 | Marlow | scary stuff |
| 22:21.29 | PBXtech | heh |
| 22:21.46 | PBXtech | whats the shell command to tell me what version of firmware is on a digium card? |
| 22:22.00 | Nethab | zt something probably |
| 22:22.00 | *** join/#asterisk [KFC]Jupiter (~C@69-163-201-17.atlsfl.adelphia.net) |
| 22:22.01 | ManxPower | PBXtech: Digium cards don't have any firmware |
| 22:22.02 | BullWeivez | im going to replace the panasonic PBX we inherited with an * box.... so i was just wondering :) |
| 22:22.03 | [KFC]Jupiter | Hello |
| 22:22.10 | Nethab | hi |
| 22:22.18 | PBXtech | not upgradable firmware, but there is firmware |
| 22:22.30 | [KFC]Jupiter | shido6: I was downloading Internet Download Manager to download my PBX since my pc timed out; ( |
| 22:22.34 | BullWeivez | it got my mind thinking...hey what are folks going to think when i take a POE switch and computer... |
| 22:22.37 | Nethab | you want to know what version of the board you have |
| 22:23.01 | PBXtech | [BullWeivel]: thank you for saving them a ton of money |
| 22:23.12 | Nethab | their going to think, wow welcome to the 21st centrery |
| 22:23.41 | *** join/#asterisk netofsickcoder (~netofsick@200.121.129.178) |
| 22:23.46 | PBXtech | 1U Server 2U PoE switch, looking lovly |
| 22:24.07 | PBXtech | anyway, beating a dead horse |
| 22:24.12 | Nethab | with a stick |
| 22:26.55 | BullWeivez | curious... what you guys are using for PoE switches? |
| 22:27.32 | PBXtech | i sell Zultys so ... Zultys |
| 22:27.36 | PBXtech | :) |
| 22:28.36 | my007ms | bye all |
| 22:30.34 | Nethab | bye |
| 22:30.37 | *** part/#asterisk my007ms (~mywaleed@62.139.173.127) |
| 22:30.38 | *** join/#asterisk zotz (~zotz@24.231.32.109) |
| 22:30.51 | Nethab | i don't have a PoE switch |
| 22:30.57 | Nethab | so i couldn't recomend one |
| 22:32.16 | PBXtech | zultys sell a 2 port PoE which is kinda cool for the newbie |
| 22:32.59 | PBXtech | and a killer phone, that i like more than cisco. not as pretty but bluetooth in a IP phone... hell ya |
| 22:33.01 | PBXtech | anyway |
| 22:33.36 | *** join/#asterisk cc (~cc@byte.fedora) |
| 22:34.23 | Marlow | pbxtech: bluetooth in a ip phone ? |
| 22:34.29 | Marlow | pbxtech: what phone is that ? |
| 22:35.05 | PBXtech | Zultys 4x5 works great SIP phone |
| 22:35.24 | Marlow | but bluetooth ? for what ? headset ? |
| 22:35.32 | PBXtech | yup |
| 22:35.40 | Marlow | sounds interesting .. |
| 22:36.38 | Marlow | quite expensive though |
| 22:37.19 | PBXtech | yea it is. phone is high end for sure. that phone will actually establish a VPN/IPsec connection to the corp office and has firewall/nat/dhcp built into the phone. |
| 22:37.21 | PBXtech | its amazing |
| 22:37.26 | *** join/#asterisk BoRiS (boris@S01060040ca1e5b54.wp.shawcable.net) |
| 22:37.58 | PBXtech | but amazing is costly heh |
| 22:38.13 | Marlow | pbxtech: fun part ... the vigor 2600 router can do the same .. if necessary .. i mean ipsec |
| 22:38.34 | PBXtech | router not phone |
| 22:38.35 | Marlow | pbxtech: and that's also for the sip port |
| 22:38.51 | Marlow | pbxtech: the vigor 2600 has a build in 2 port ata |
| 22:38.58 | PBXtech | oh thats cool |
| 22:39.00 | Marlow | pbxtech: sip |
| 22:39.14 | PBXtech | how much are they |
| 22:39.21 | Marlow | pbxtech: and can enforce wlan clients to use vpn as do lan-to-lan vpn tunnels |
| 22:39.34 | Marlow | pbxtech: some around 300$ |
| 22:39.44 | Marlow | pbxtech: for the VGi |
| 22:40.04 | Marlow | pbxtech: that is voip + 802.11g access point + build in adsl modem |
| 22:40.04 | *** part/#asterisk BullWeivez (~BullWeive@12.165.66.3) |
| 22:40.06 | PBXtech | cool, im going to look at them. i like the whole plug and play deal like that |
| 22:40.13 | MikeJ[Laptop] | is anyone interested in T.38? |
| 22:40.19 | PBXtech | who isnt |
| 22:40.20 | Marlow | pbxtech: and optionally build in backup isdn dialup |
| 22:40.45 | MikeJ[Laptop] | the bounty is on a bit of a roll.. anyone interested in putting their money where their mouth is? |
| 22:41.15 | PBXtech | you got the solution mike |
| 22:41.15 | BoRiS | someone is working on it buy very slowly has not released any code on it |
| 22:41.18 | Nethab | if i was familiar with T.38... |
| 22:41.25 | PBXtech | fax over IP |
| 22:41.36 | MikeJ[Laptop] | http://www.voip-info.org/tiki-index.php?page=Asterisk+T.38+Bounty |
| 22:41.44 | PBXtech | spandsp guy i think was working on it |
| 22:41.49 | BoRiS | yes |
| 22:41.50 | MikeJ[Laptop] | no, but I know there is at least one in development |
| 22:41.54 | BoRiS | coppice |
| 22:41.57 | MikeJ[Laptop] | yep |
| 22:42.10 | MikeJ[Laptop] | money has ways of making things happen |
| 22:42.24 | BoRiS | not for him |
| 22:42.42 | MikeJ[Laptop] | hehe |
| 22:42.46 | MikeJ[Laptop] | your not helping here |
| 22:43.00 | *** join/#asterisk harryvv (~leonardo@S010600055d210201.vs.shawcable.net) |
| 22:43.14 | BoRiS | sorry |
| 22:43.19 | BoRiS | hehe |
| 22:43.37 | Nethab | did anyone else get the new sipura firmware to connect a 33.6 modem connection |
| 22:43.37 | harryvv | Say there is a case where two people use the same phone line and thay want to seperate incomming calls by distinctive ring. Can this be done? |
| 22:43.51 | Nethab | harryvv: yes |
| 22:44.16 | Nethab | depends on the phone or FXS port though |
| 22:44.32 | harryvv | its a spira ata 1000 |
| 22:44.36 | harryvv | but |
| 22:45.07 | harryvv | that phone is shared by two people. I want it to ring a distinctive ring based on what extention the incomming caller choses. |
| 22:46.08 | Nethab | both extensions ring the same phone |
| 22:46.19 | harryvv | yes |
| 22:46.37 | harryvv | untill the day we can get a second ata or ip phone this will do. |
| 22:46.49 | BoRiS | if you know the callerid of the person or have an ivr ask which people to phone... you could do it with distinct ring |
| 22:47.14 | Nethab | he has two seperate exten => which dial the same SIP/peer |
| 22:47.46 | Nethab | he wants to know if there's something he can put in the dial command or a channel variable to indicate a specif ring to the sipura |
| 22:47.50 | harryvv | yea two seperate that dials the same phone. now need distinctive rining so i wont have to answer my wifes calls all the time..and its alot. |
| 22:49.09 | Nethab | i think you can do something with calleridname that could make the sipura give a different ring |
| 22:50.13 | PBXtech | so whats the deal with this new T1 w/echo cancellation |
| 22:50.22 | harryvv | um, with possibly 50 numbers...?? i would rather give the caller the choice to select which extention to push rather then me make a distinctive ring for every number that comes in. |
| 22:50.24 | PBXtech | when is it available for sale any idea? |
| 22:51.09 | harryvv | actually i think this would work |
| 22:51.12 | harryvv | exten => 1,1,Dial,Zap/28 ; Ring Zap/28 normally |
| 22:51.12 | harryvv | exten => 2,1,Dial,Zap/28r1 ; Ring Zap/28 with ring #1 |
| 22:51.12 | harryvv | exten => 3,1,Dial,Zap/28r2 ; Ring Zap/28 with ring #2 |
| 22:52.17 | BoRiS | Your dialing a Zap channel, not a SIP channel? |
| 22:52.18 | *** join/#asterisk Romik (~romik@1.fix.netvision.net.il) |
| 22:52.45 | harryvv | sorry that was cut from another site. yes its sip so i know that would need to be modified. |
| 22:52.47 | Nethab | no no, add to their callerid name a * or a space that triggers sipura's distinctive ring detection |
| 22:52.59 | Nethab | not on number on name |
| 22:53.02 | harryvv | thats to much work |
| 22:53.19 | harryvv | we both get alot of calls. she gets alot of calls from one time callers. |
| 22:53.42 | BoRiS | Nethab: I installed the new firmware but all the firmware notes said is that for any fax/modem connection, that jitterbuffer and stuff will be turned off and automatically switch to G711u (for pass-thru mode) |
| 22:53.49 | Nethab | don't detect individual numbers, specifically ADD to the callerid a distinctive string |
| 22:54.22 | harryvv | mm not sure what you mean Nethab |
| 22:55.23 | Nethab | so when a call comes in and dials exten 1 SetCallerIDName(${CALLERIDNAME} + "*") |
| 22:55.40 | Nethab | and on sipura if callerid ends in * use different ring |
| 22:55.48 | NewSole | can anyone tell me why I am getting 1 way audio on TDM400P Cards Using FXS ports |
| 22:56.17 | BoRiS | actually... I thoght to use distinct ring... You had to had a "exten => 100,1,SetVar(ALERT_INFO=1)" or something very similar to that |
| 22:56.25 | Nethab | does sipura support that? |
| 22:56.39 | BoRiS | It does for the 841 |
| 22:56.51 | BoRiS | ALERT_INFO=Classic-1 |
| 22:56.51 | Nethab | he has a SPA 1000 |
| 22:56.55 | BoRiS | hmmm |
| 22:57.09 | Romik | somebody know the livevoip.com down? |
| 22:57.33 | Romik | on all calls i receive "unavalible" error |
| 22:58.30 | Nethab | no idea |
| 22:58.39 | BoRiS | Dont use livevoip anymore |
| 22:58.56 | Romik | boris: any good replacement? |
| 22:58.56 | Nethab | heh i use broadvoice |
| 22:59.01 | Nethab | hehehehe |
| 22:59.10 | *** join/#asterisk dan2 (dan@dan2.active.supporter.pdpc) |
| 22:59.46 | BoRiS | Me (soon) :) |
| 23:00.33 | Romik | boris: you start advertising there?;) |
| 23:00.58 | Romik | boris: now i routing all calls via voipjet |
| 23:00.58 | harryvv | so you cannot set distinctive ring in extensions.conf? |
| 23:00.58 | PBXtech | just have a second IP provider in your dialsting eh |
| 23:01.49 | Nethab | harryvv: BoRis says you can set ALERT_INFO=1 to trigger a disctinctive ring |
| 23:02.05 | harryvv | okay and what does that do? |
| 23:02.12 | Nethab | he said that works on the sipura 841 |
| 23:02.37 | harryvv | mmm well this is for a ata 1000 by sipura. |
| 23:02.41 | Nethab | it sets a flag on the call to make the call Higher priority, which just means use different ring |
| 23:02.52 | harryvv | I see |
| 23:02.57 | Nethab | but if sipura did it once maybe they do it on all boxes |
| 23:03.04 | Nethab | wouldn't hurt to try |
| 23:03.05 | harryvv | yea |
| 23:03.56 | Nethab | i have a 3000 i wonder if it's in the manual... |
| 23:04.48 | harryvv | did a google on ALERT_INFO=1 and came up only with cisco |
| 23:05.08 | [KFC]Jupiter | hey, anyone know where any drivers for linux linksys-wusb54g |
| 23:05.08 | Nethab | in the manual for the 1000 2000 3000 it has a whole section on distinctive ringing |
| 23:05.11 | Nethab | i just cheked |
| 23:05.47 | *** join/#asterisk santiago (~santiago@63.245.86.187) |
| 23:06.22 | *** part/#asterisk Marlow (~martin@cerberus.bluetree.ie) |
| 23:06.57 | harryvv | yes but..its only per phone number..what if I wanted to devide 100 numbers up between the both of us? what if either of us get a one time phone call? would not work..instead would rather hav the extention context set the distinctive rung for that extention when pressed by the calling party. |
| 23:07.00 | Nethab | you just create an Alert_info header in the sip message |
| 23:07.09 | Nethab | no it's not keep reading |
| 23:07.15 | Nethab | page 29 |
| 23:07.39 | [KFC]Jupiter | <PROTECTED> |
| 23:07.46 | *** join/#asterisk JunK-Y (~junky@modemcable174.107-81-70.mc.videotron.ca) |
| 23:07.53 | harryvv | This would also be usefull for a new hireee at a work site that shares the same office but a phone is yet not avaible to install. |
| 23:08.18 | Nethab | you can use alert_info header |
| 23:08.26 | Nethab | set it to 1 or 2 or 3 |
| 23:09.33 | [KFC]Jupiter | Anyone who can help: Drivers for linux on linksys-wusb54g |
| 23:09.35 | Nethab | that's the SetVar(ALERT_INFO=1) before the dial |
| 23:09.56 | [KFC]Jupiter | ndiswrapper? |
| 23:11.37 | *** join/#asterisk nextime (~nextime@213-140-6-96.fastres.net) |
| 23:12.57 | harryvv | mmm i wonder if its possible to make evolution make sceduled calls for special apointments :) |
| 23:13.13 | *** join/#asterisk iq_ (~iq@70-59-163-80.omah.qwest.net) |
| 23:13.15 | [KFC]Jupiter | Yeah--- |
| 23:13.40 | harryvv | I dont like to drag around a palm pilot and that would be handy if in the office or home. |
| 23:14.13 | harryvv | actually, that would also be helpfull for a hotel.. make it call people in the morning on request :) |
| 23:14.19 | Nethab | look up auto-dial |
| 23:14.19 | harryvv | wake up call. |
| 23:14.40 | Nethab | just create a .call file in a directory with parameters for number to dial, and what to say when it connects |
| 23:14.43 | [KFC]Jupiter | yeah my school calls their students registered number when they are absent with a prerecorded message |
| 23:14.59 | *** join/#asterisk Uberbot (qqkebkh@pcp01879960pcs.sandia01.nm.comcast.net) |
| 23:15.26 | [KFC]Jupiter | I am afraid that I can't use my Linksys WUSB56G USB Adapter with this Lunix build! |
| 23:15.33 | [KFC]Jupiter | Linux* |
| 23:15.50 | Nethab | Lunix, the OS for crazy people |
| 23:16.04 | Vco | preeeeeeeeeeeeeeetty lame |
| 23:16.08 | Vco | ;) |
| 23:16.09 | [KFC]Jupiter | Linux |
| 23:16.10 | Nugget | Linux is poo. |
| 23:16.17 | harryvv | like reminder events for 200 people and there phone numbers. We got those at microsoft when a virus hit the exchange servers. Was really funny. vm email and some times a knock on the door..Dont open this email if it says blaa blaa... |
| 23:16.17 | harryvv | :) |
| 23:16.40 | [KFC]Jupiter | Nethab - any ideas on how to get WUSB56G working? |
| 23:16.40 | Uberbot | Can anyone help me track down an audio problem? |
| 23:16.48 | Vco | wicked..so every time there's a virus alert you can DoS your pbx |
| 23:16.53 | Nugget | clean the wax out of your ears. |
| 23:16.59 | Uberbot | <snicker> |
| 23:17.05 | harryvv | :) |
| 23:17.17 | Nethab | no idea, my USB devices Just Work |
| 23:17.18 | Uberbot | I get nothing bug garbage at the other end. |
| 23:17.31 | harryvv | BoRiS: you still here |
| 23:17.42 | [KFC]Jupiter | ok well I haven't tried it, but I hope it will since It needs to connect to a WiFI network |
| 23:17.51 | harryvv | BTW |
| 23:17.59 | harryvv | All of Canadian goverment is going Voip |
| 23:18.11 | [KFC]Jupiter | Thats nice for them |
| 23:18.13 | *** join/#asterisk valueape (~valueape@c-24-98-250-188.hsd1.ga.comcast.net) |
| 23:18.15 | Vco | i'm going to assume you mean federal. |
| 23:18.19 | harryvv | Telus and cisco just implemented a total voip system at vancouver airport. |
| 23:18.24 | harryvv | yes federal |
| 23:18.44 | valueape | Hello everyone, first time here - impressed with the number of folks. |
| 23:18.45 | [KFC]Jupiter | Yeah most new buildings have VoIP and network telecommunications hooked up |
| 23:18.47 | Vco | all cisco at the new TO terminal too |
| 23:19.09 | Vco | "newish" terminal |
| 23:19.11 | Vco | rather |
| 23:19.55 | Vco | looks like i get to do some * testing at the lake this summer |
| 23:20.25 | [KFC]Jupiter | ? |
| 23:20.26 | Vco | 10mb wifi for $49/mo |
| 23:20.39 | Vco | haul back into town |
| 23:20.48 | valueape | I'm about to install Asterisk for the first time. Just got 4 Polycom 500s and a 4-port Digium card. Installing Fedora 3. |
| 23:22.26 | Elshar | Congrats :D |
| 23:22.53 | valueape | Thanks! All those damn slashdot articles got me pumped to give it a try. |
| 23:23.02 | harryvv | which ones |
| 23:23.14 | tld | When I'm defining sip phones, is it a problem to use names like 'tld', or do I have to use numbers? |
| 23:23.31 | tld | can I use names, and just map numbers to them by way of extensions.conf? |
| 23:23.45 | valueape | hrm. The first was just a general article and the second about how companies are starting to build appliances based on *. |
| 23:25.06 | valueape | The comments seemed positive...and it seemed like a fun project (I'm a partner at a software development shop in Atlanta) |
| 23:25.37 | Romik | LIVEVOIP message: We currently have a switch outage in LA and technical staff are working to repair it. We have no ETA at this time but people are onsite. |
| 23:25.38 | Uberbot | I have an extension called "time" |
| 23:25.42 | Uberbot | You know what it does.... |
| 23:25.56 | Nethab | times out |
| 23:26.06 | Uberbot | There's one in every crowd.... |
| 23:26.08 | valueape | Anyone have any thoughts on the best free * GUI? |
| 23:26.26 | harryvv | mabey livevoip forgot what the word redundency and mirror means? |
| 23:26.31 | Nethab | define GUI, there's configuration GUI's monitoring gui's |
| 23:26.42 | valueape | Configuration GUIs. |
| 23:27.09 | valueape | There is one on Source Forge called "Asterisk GUI" - just curious what you people think. |
| 23:27.51 | Nugget | there does not exist a gui which will let you avoid learning all the config files. |
| 23:28.07 | Nugget | if you want something to automate or delegate the more tedious aspects of running a working server, go for it. |
| 23:28.37 | Nugget | but there is nothing and probably never will be that can serve as an alternative to reading the docs, learning and editing the config files. |
| 23:28.48 | valueape | I |
| 23:29.03 | valueape | I'm not looking for a GUI to do that. |
| 23:29.20 | Nugget | then there are several options which you might be happy with |
| 23:29.38 | Nugget | I don't have any recommendations, though, I've never used any of them. :) |
| 23:29.49 | valueape | But to simply configuration. hehehe..I'm not one of those QUI cures all ills people :) |
| 23:31.56 | *** join/#asterisk facek__ (faceoff@devel.acdbddh.eu.org) |
| 23:31.58 | facek__ | elou |
| 23:32.10 | facek__ | where is the ast_verbose funcion declarated? |
| 23:34.44 | JunK-Y | grep it ? |
| 23:34.50 | JunK-Y | its in logger.c |
| 23:34.51 | *** join/#asterisk NewSole (dave@i216-58-44-245.avalonworks.net) |
| 23:34.56 | facek__ | thanks |
| 23:37.19 | harryvv | man my wifes work gets constant bussy signals. |
| 23:37.20 | harryvv | :) |
| 23:37.34 | harryvv | one line for a dentist office does not cut it. |
| 23:37.41 | hardwire | bleh |
| 23:39.43 | harryvv | anyone here run the spa 1000-3000 |
| 23:39.58 | shido6 | yep |
| 23:40.06 | *** join/#asterisk HeadachesAbound (~asterisk-@adsl-70-244-228-14.dsl.tulsok.swbell.net) |
| 23:40.09 | harryvv | well, Im asuming you do ;) |
| 23:40.14 | shido6 | :) |
| 23:42.40 | puowvip | http://www.atarimagazines.com/whizkids/thumbs.php?issue=drugs |
| 23:42.56 | *** join/#asterisk whmok (~acidBurn@218.208.199.76) |
| 23:43.31 | *** join/#asterisk tsp (~tyler@S01060080c825173c.vc.shawcable.net) |
| 23:43.38 | tsp | how do i dial an iax number using console? |
| 23:43.52 | Nethab | i have a spa 3000 |
| 23:44.12 | tsp | Nethab: how do you dial using fwd through iax2? I don't have an extensions.conf |
| 23:44.28 | Nethab | you don't have ANY extensions.conf? |
| 23:45.11 | tsp | Nethab: no |
| 23:45.18 | tsp | Nethab: only files I have are modules.conf and iax2.conf |
| 23:45.27 | tsp | Nethab: I just want to use asterisk as a softphone |
| 23:45.51 | Nethab | you have to have more files than that to get * working |
| 23:46.02 | Nethab | extensions.conf controls what things you can dial |
| 23:46.13 | Nethab | without it you can only dial by IP address |
| 23:46.17 | tsp | I do? what other files? |
| 23:46.25 | Nethab | extensions.conf for start |
| 23:46.27 | tsp | won't dial iax2/411/fwd work? |
| 23:46.31 | tsp | or something along those lines |
| 23:47.02 | Nethab | no, because that won't trigger a dial |
| 23:47.08 | Nethab | that can only be done in extensions.conf |
| 23:47.41 | tsp | oh |
| 23:47.43 | tsp | bah! |
| 23:47.50 | tsp | i just want ot type dial number and have that number dial |
| 23:47.59 | Nethab | that string only works inside a dial. outside that's not a valid URL to call |
| 23:48.21 | Nethab | the only thing you can call without extensions.conf is straight IP addresses |
| 23:48.37 | Nethab | like 411@iax2.fwdnet.net |
| 23:49.06 | Nethab | but if you have a sip phone, you can only call sip urls |
| 23:49.12 | Nethab | and iax phones only can call iax url |
| 23:49.14 | shido6 | then do it in the dialplan tsp |
| 23:49.15 | shido6 | yeesh |
| 23:49.29 | Nethab | it's asterisk that does the SIP to IAX oconversion |
| 23:49.35 | Nethab | he doesn't have a dialplan |
| 23:49.40 | Nethab | he doesn't have an extensions.conf |
| 23:49.45 | tsp | ok... dial 411@iax2.fwdnet.net doesn't work |
| 23:49.56 | tsp | no such extension |
| 23:50.08 | Nethab | is that the hostname in iax.conf |
| 23:50.14 | *** join/#asterisk bjohnson (~bjohnson@ip206-172.dsl.istop.com) |
| 23:50.15 | Nethab | i don't know it from memory |
| 23:50.36 | tsp | yeah |
| 23:50.38 | tsp | its the hostname |
| 23:51.04 | Nethab | even so, if you have a register statement in iax.conf it always goes to the dialplan in extensions.conf |
| 23:51.30 | tsp | oh |
| 23:51.48 | tsp | all i want is a softphone... |
| 23:51.53 | tsp | but I have no x support |
| 23:52.34 | Nethab | you want asterisk to be just a softphone |
| 23:52.57 | Nethab | you still need extensions.conf |
| 23:53.02 | Nethab | it says what you can dial |
| 23:53.07 | Nethab | and who can call you back |
| 23:53.15 | tsp | oh |
| 23:53.45 | Nethab | softphones have dialplans too |
| 23:53.46 | harryvv | Whats a good site that sells nice quality voiceovers? |
| 23:53.59 | Nethab | they say if you dial *1 and a number go to fwd |
| 23:54.14 | Nethab | and *2 then number go to sipphone |
| 23:54.23 | tsp | oh |
| 23:54.25 | Nethab | voiceovers? |
| 23:54.30 | Nethab | like Allison? |
| 23:55.25 | harryvv | yes |
| 23:55.47 | harryvv | if thats her name that makes the anouncments in comedian vm |
| 23:56.56 | nestAr | http://www.digium.com/index.php?menu=thevoice |