| 00:00.21 | [TK]D-Fender | SkramX : Ask a specific question, get a specific answer :) |
| 00:00.52 | SkramX | :) Well.. I havent looked att her product line too extensively yet.. but like Cisco, can I code apps for the phones (such as simple 'websites', etc.)? |
| 00:01.58 | *** join/#asterisk hardwire (n=hardwire@rdbck-4891.wasilla.mtaonline.net) |
| 00:02.00 | hardwire | woo |
| 00:02.16 | [TK]D-Fender | SkramX : IP 6XX series, minimally, yes |
| 00:02.24 | SkramX | any experience with that |
| 00:02.44 | hardwire | does asterisk still need a usb or zaptel timing interface for accurate operation? |
| 00:02.52 | hardwire | looking to move my asterisk install into a vserver context |
| 00:03.31 | rpm | what can i use in linux for transcoding a .gsm or .wav file to g729? |
| 00:03.38 | SkramX | hardwire: you would need to patch the host |
| 00:03.42 | SkramX | 's kernel |
| 00:04.25 | hardwire | SkramX: to what avail? |
| 00:04.39 | [TK]D-Fender | SkramX : Yes, I've got live queue stats on Idle on my 600's, and user directories with click-to-call functionality, and a bit more. |
| 00:05.00 | SkramX | telephreak.org/papers -- I didn't write that but I am an administrator of Telephreak, an organization, and work for a company which does similar Virtual Private Servers |
| 00:05.20 | SkramX | [TK]D-Fender: okay.. do they have a dev kit or what? is it XML silliness like Cisco or WAP/HTML? |
| 00:06.36 | hardwire | SkramX: OpenVPS servers? |
| 00:06.37 | SkramX | or should i google? :) |
| 00:06.42 | SkramX | linux-vserver.org |
| 00:06.55 | SkramX | (software) |
| 00:07.05 | hardwire | what are you rambling about? |
| 00:07.07 | *** join/#asterisk jebba (n=jebba@201-212-163-95.net.prima.net.ar) |
| 00:07.45 | hardwire | SkramX: I would read the papers to find out what kind of patch you are suggesting.. but its access denied |
| 00:07.49 | jebba | beta3 is still listed in /topic ;) |
| 00:08.27 | SkramX | hardwire: hmm lemme look into that |
| 00:09.07 | hardwire | I will totally let you |
| 00:09.11 | hardwire | no worries :) |
| 00:09.28 | hardwire | yipes |
| 00:09.30 | SkramX | http://www.telephreak.org/papers/vpa/ |
| 00:09.32 | hardwire | last I used asterisk was 1.2 |
| 00:09.49 | SkramX | Your screen name looks familiar |
| 00:09.49 | JT | it's still in the 1.2.x series |
| 00:10.05 | hardwire | ztdummy rtc patches |
| 00:10.06 | JT | dude, it's a "nickname" ;) |
| 00:10.23 | SkramX | yeah.. i thought of that right after I hit enter |
| 00:10.27 | hardwire | SkramX: yeh.. I've been here before .. for a solid year probably |
| 00:10.32 | SkramX | ok |
| 00:10.39 | hardwire | SkramX: ztdummy rtc I assume? |
| 00:10.45 | SkramX | think so |
| 00:10.56 | hardwire | otherwise you are telling me to patch vserver patches in :) |
| 00:10.59 | hardwire | which is sorta dun |
| 00:11.05 | SkramX | right |
| 00:11.08 | hardwire | by the good people of debian, inc. |
| 00:11.22 | SkramX | heh |
| 00:11.31 | SkramX | [TK]D-Fender: does it do WAP/XML/HTTP? what? |
| 00:11.57 | hardwire | SkramX: any vserver issues other than timing? |
| 00:12.06 | hardwire | just right off the bat you wanna swing my way :) |
| 00:12.15 | SkramX | eh? |
| 00:12.20 | SkramX | no |
| 00:12.23 | hardwire | groov |
| 00:12.34 | hardwire | I love kids movies |
| 00:12.39 | hardwire | Hello fence! |
| 00:13.53 | jebba | fwiw, i've been running asterisk in a debian vserver for a year now. Works quite well. (actually the guest is fedora, host is deb) |
| 00:15.23 | hardwire | jebba: read this paper? |
| 00:15.42 | hardwire | I have no idea why they are defining /dev/zap devices |
| 00:15.44 | hardwire | http://www.telephreak.org/papers/vpa/ |
| 00:16.01 | SkramX | it may be a bit out of date |
| 00:16.13 | hardwire | date aside.. vservers can't really use devices like that |
| 00:16.20 | hardwire | I could be totally wrong however |
| 00:16.41 | hardwire | could/usually |
| 00:16.52 | jebba | hardwire, i haven't read that paper |
| 00:17.51 | hardwire | jebba: SkramX: multiple vservers per host? |
| 00:17.52 | jebba | i did create /dev/zap/* on the guest & host though. No zap hardware. I don't get why asterisk does timing with ztdummy...... |
| 00:18.05 | *** join/#asterisk ronaldl79 (n=chatzill@75.119.1.39) |
| 00:18.05 | hardwire | otherwise I have no idea how ztdummy would share as a module between contexts gracefully |
| 00:18.15 | hardwire | jebba: because it can |
| 00:18.17 | ronaldl79 | Evening. |
| 00:18.17 | hardwire | :) |
| 00:18.27 | jebba | hardwire, i have a 1.2.14, a 1.4 beta4, some random web/mail servers all in their own vservers. |
| 00:18.38 | hardwire | jebba: ok |
| 00:18.45 | hardwire | which asterisk instance loads first? |
| 00:18.50 | hardwire | typically |
| 00:19.02 | hardwire | I would love to know what the last to load one says about the timing interface |
| 00:19.03 | jebba | would be nice if it didn't requite ztdummy. I can't even remember at the moment if 1.4 yanked that requirement or not. |
| 00:19.12 | hardwire | heh |
| 00:19.18 | ronaldl79 | Just upgraded * to 1.4 Beta 4 (actually, it's a clean install .. damn HD crapped out ln), and decided to try out Asterisk-Gui -- are the URLs malformed? |
| 00:19.33 | jebba | hardwire, i can check if you want. How do you want me to see "what it says"? |
| 00:19.57 | hardwire | jebba: I wonder if the zaptel drivers were written well enough so that multiple applications can easily access them and allocate channels in a polite fashion |
| 00:20.25 | hardwire | jebba: jsut grep for ztdummy or "tim" in the asterisk logs I guess |
| 00:20.56 | *** join/#asterisk DavoFrom818 (n=Vito310@cpe-76-173-56-41.socal.res.rr.com) |
| 00:20.57 | jebba | heh. The other issue is that i send logs (or mostly in the past) to /dev/null (!). ... |
| 00:20.59 | DavoFrom818 | hi guys |
| 00:21.25 | DavoFrom818 | is there like a trixbox solution to a prepaid calling cards system? |
| 00:21.36 | JT | rofl |
| 00:21.48 | jebba | hardwire, the main 1.2.14 installation usually has 5-20 calls running through it. The 1.4 beta very few, maybe one or two at a time. No heavy use of conferencing. |
| 00:22.14 | hardwire | ok |
| 00:23.24 | [TK]D-Fender | SkramX :XHTML. a funny subset |
| 00:23.44 | jebba | hardwire, oh, actually i do have /var/log/asterisk/messages, i was just nixing CDRs. No matches for zt or tim in the logs tho |
| 00:23.56 | SkramX | [TK]D-Fender: hmm.. *shrug* |
| 00:23.59 | hardwire | jebba: danke! |
| 00:24.16 | SkramX | :XHTML != XHTML, [TK]D-Fender? |
| 00:24.37 | SkramX | none of the polycoms are color, are they |
| 00:24.58 | irq | can someone tell me what the big differences are between asterisk 1.4 and 1.2? |
| 00:25.03 | SkramX | oh.. they do video stuff but I bet that's over our price range |
| 00:25.19 | ronaldl79 | irq -- http://www.voip-info.org -- there's a good starting point |
| 00:26.05 | hardwire | irq: hehe.. the answer is .2 |
| 00:26.13 | hardwire | which really isn't that big at all |
| 00:26.18 | *** join/#asterisk znoG (n=servat@60-240-15-141-nsw-pppoe.tpgi.com.au) |
| 00:26.26 | irq | i don't see anything that really answers my question on that page ronald, but thanks (i have used that resource for other things in the past) |
| 00:26.42 | znoG | hi all. is it possible that the 1.4.x version of Asterisk doesn't let you include a context from extensions.conf in the extensions.ael file? |
| 00:28.44 | SkramX | [TK]D-Fender: want to give me a code example? |
| 00:28.52 | ronaldl79 | irq -- Have you tried Asterisk.org? Or viewed the SVN tree? |
| 00:29.22 | ronaldl79 | I just read the latest changelog and it had some interesting tidbits you'd be interested in. |
| 00:29.42 | *** join/#asterisk edwar64896 (n=medwards@72.83.233.220.exetel.com.au) |
| 00:29.49 | irq | it's quite amazing how difficult it can be to find a changelog for software through my web browser, without having to actually check out the code |
| 00:30.08 | irq | besides, i'm only looking for the really big changes, which i'm guessing someone could have summarized in fewer words than this one line i'm typing right now :) |
| 00:30.58 | [TK]D-Fender | SkramX : Just picture basic HTML except where ALL tags need to be closed. IMG for instance. <BR /> can be self enclose. Stuff that normally doesn't does now. |
| 00:30.58 | edwar64896 | 'ello asterisk peoples... anyone done any work with meetme applications - selecting empty conferences and using dynamic conferences? |
| 00:31.33 | jebba | irq, fixes. cleaner. jabber/jingle/t38/foo |
| 00:31.33 | SkramX | ok. i thought you said :XHTML was a subset |
| 00:31.39 | SkramX | I do XHTML strict sites all the time |
| 00:31.50 | irq | thanks jebba! just what i was looking for :) |
| 00:31.55 | SkramX | but you meant "Skram: XHTML", not "Skram :XHTML" |
| 00:32.01 | [TK]D-Fender | SkramX : This won't be a stretch then. But its lacking in several areas. like tables. |
| 00:32.07 | SkramX | that just sort of threw me off. |
| 00:32.23 | SkramX | hmm. [TK]D-Fender: how does the user get to a web server? does polycom document this stuff? |
| 00:32.41 | [TK]D-Fender | SkramX : "Services" button is sorta staring you in the face :) |
| 00:32.57 | SkramX | well, how does the phone know where to get the web server ip/address? |
| 00:33.02 | [TK]D-Fender | SkramX : Fromt here it initially starts off on whatever you define as the home page, and away you go. |
| 00:33.10 | [TK]D-Fender | SkramX : All in the phone setup |
| 00:33.11 | SkramX | ok |
| 00:33.39 | SkramX | hrmm. while I like XHTML more than weird XML.. Cisco's phone is color and seemingly more expandable |
| 00:34.01 | SkramX | I would be developing something for a web-development company. they like flashy stuff.. like colors, heh |
| 00:34.04 | SkramX | you know what I mean? |
| 00:35.02 | *** join/#asterisk ManxPower (n=manxpowe@dpc67142183150.direcpc.com) |
| 00:36.29 | [TK]D-Fender | SkramX : Cisco has Polycom's number on this point, but loses on the phone & cost fronts. |
| 00:36.39 | SkramX | right. hrmm |
| 00:37.11 | SkramX | [TK]D-Fender: can the polycom push and pull or only pull data from the webserver |
| 00:37.24 | SkramX | like.. could i make a form and/or maybe buttons? |
| 00:37.33 | SkramX | *shrug* |
| 00:38.10 | [TK]D-Fender | SkramX : Yeah, they can do forms. |
| 00:38.23 | SkramX | ok.. any documentation how to do so, since it isn't a touch screen |
| 00:38.34 | SkramX | or am I missing something |
| 00:38.44 | [TK]D-Fender | SkramX : But again, I'm sure Cisco is a better deal if thats what you're pushing. Thenk again in the converged world you should be doing this stuff on a PC anyways |
| 00:38.56 | SkramX | right.. |
| 00:39.11 | [TK]D-Fender | SkramX : thats what the cursor keys are for |
| 00:39.20 | SkramX | oh |
| 00:39.29 | SkramX | polycom has the right idea about letting the developer use XHTML though |
| 00:40.24 | [TK]D-Fender | SkramX : Well it doesn't let you manipulate the PHONE in any meaningful way, and lacks colour and a lot of the essentials like tables (which really pisses me off), but you can do stuff with it. |
| 00:40.35 | SkramX | rrrrok |
| 00:40.37 | SkramX | *ok |
| 00:40.43 | SkramX | hrmm; i'll need to think about it. |
| 00:40.54 | SkramX | i dont really want to learn the details of ciscos XML stuff |
| 00:41.20 | resistance | hi guys, i'm having problems with my flash: it works but i have to keep pressing # till it works |
| 00:43.11 | [TK]D-Fender | ok, off for a bit, back later |
| 00:43.32 | JT | flash... hook flash? |
| 00:44.23 | *** part/#asterisk VoipMasta (n=fabio@201.139.139.127.cableonline.com.mx) |
| 00:44.31 | shmaltz | which application module is the manager api? I want to unload it |
| 00:44.43 | Strom_C | resistance: pressing # is not the same as a hookflash |
| 00:45.19 | resistance | doesn't work with hookflash |
| 00:45.25 | resistance | it just hangs up |
| 00:45.33 | Strom_C | zaptel channels? |
| 00:45.40 | resistance | yes |
| 00:45.49 | Strom_C | you do have threewaycalling=yes in the zapata.conf, right? |
| 00:45.52 | JT | resistance: der, # is the key that skips through priorities in asterisk |
| 00:46.17 | JT | by default anyway |
| 00:46.17 | Strom_C | because if you don't, then of course the hookflash won't work |
| 00:46.42 | resistance | 3 way calling = yes |
| 00:46.52 | Strom_C | no |
| 00:47.06 | Strom_C | threewaycalling=yes |
| 00:47.20 | resistance | yes, i was just a bit fancy (ehem) |
| 00:47.38 | Strom_C | not "3 way calling" or "threeway" or "oh cool look what this does = yes" |
| 00:47.48 | Strom_C | this is a precise syntax |
| 00:47.58 | resistance | i should try that oh cool one |
| 00:48.09 | Strom_C | yeah, and then try "lol=very" in iax.conf |
| 00:48.25 | resistance | then everything will start working |
| 00:48.27 | resistance | lol |
| 00:48.41 | Strom_C | oh, and "irritateeveryoneonpoundasterisk=true" |
| 00:49.00 | resistance | or dumbass=me |
| 00:49.03 | resistance | he he |
| 00:49.26 | [hC] | hmm. is there a way to, in your dial plan, force-modify the CDR to change the number that was dialed? I have to dial a number stupidly because of my telco, and i want to modify it to show up sane in my cdr database. |
| 00:51.16 | *** join/#asterisk dahunter3 (n=dahunter@155.sub-75-215-176.myvzw.com) |
| 00:51.26 | *** join/#asterisk Eliran_Itzhak (n=eliran@bzq-82-81-151-215.red.bezeqint.net) |
| 01:02.21 | *** join/#asterisk eltech (i=G00Ds@ool-457c93b6.dyn.optonline.net) |
| 01:04.09 | shmaltz | anybody here ever installed/used astbill before? |
| 01:04.42 | znoG | hi all. is it possible that the 1.4.x version of Asterisk doesn't let you include a context from extensions.conf in the extensions.ael file? |
| 01:04.49 | *** join/#asterisk Strom_M (n=pocketir@m250e36d0.tmodns.net) |
| 01:06.01 | edwar64896 | |
| 01:06.10 | resistance | Strom_c: r u around? |
| 01:06.20 | Strom_C | yeah |
| 01:06.22 | Strom_C | whats up |
| 01:06.46 | hardwire | augh |
| 01:07.13 | resistance | excuse my ignorance on this ok? |
| 01:07.20 | *** join/#asterisk CunningPike (n=CunningP@204.239.8.149) |
| 01:07.23 | Strom_C | hardwire: i'm sorry, that's mot a valid question |
| 01:07.26 | Strom_C | s/mot/not/ |
| 01:08.01 | hardwire | Duck Fuo |
| 01:08.11 | resistance | t = allow called user to transfer using #? correct? |
| 01:08.12 | hardwire | s/D/F/ |
| 01:08.15 | hardwire | hehe |
| 01:08.21 | resistance | go buck a Fuffalo |
| 01:08.29 | hardwire | anypoop |
| 01:08.31 | hardwire | augh |
| 01:08.43 | Strom_C | resistance: let me give you a helpful piece of advice: type 'show application Dial' at the CLI |
| 01:09.14 | znoG | it looks like extensions.conf can't see extensions.ael and vice versa |
| 01:09.14 | Strom_C | resistance: but you really shouldn't be doing inband transfers like that. get your switchhook transfers working instead |
| 01:10.19 | resistance | ok, right now if i do that it just hangs up, and I have threewaycalling = yes |
| 01:10.31 | Strom_C | in zapata.conf for your specific FXS port? |
| 01:10.41 | Strom_C | and you did a reload chan_zap.so, right? |
| 01:10.56 | resistance | well the option has been on forever |
| 01:11.10 | Strom_C | pastebin the file |
| 01:11.26 | resistance | i have it under the [channels] context |
| 01:11.29 | resistance | is that correct? |
| 01:11.30 | Strom_C | also try transfer=yes |
| 01:11.41 | Strom_C | it has to be assigned to your fxs port |
| 01:12.00 | resistance | tranfer=yes is also enabled |
| 01:12.08 | Strom_C | pastebin the file |
| 01:12.12 | Strom_C | it'll be easier than trying to talk you through it |
| 01:12.13 | resistance | ok, how do i assign to fxs ports? |
| 01:12.16 | resistance | ok |
| 01:12.20 | Strom_C | !pb |
| 01:12.21 | Strom_C | er |
| 01:12.23 | Strom_C | ~Pb |
| 01:12.25 | jbot | pb is, like, a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste |
| 01:12.56 | resistance | http://pastebin.ca/284355 |
| 01:13.20 | wunderkin | i wonder who added in the debian one |
| 01:13.29 | Strom_C | and what's in zapata-auto.conf? |
| 01:14.44 | resistance | http://pastebin.ca/284356 |
| 01:15.01 | resistance | strom_C: people make fun of me for that one |
| 01:15.14 | resistance | but i had to..... |
| 01:15.16 | Strom_C | resistance: ? |
| 01:15.21 | *** join/#asterisk python_ (n=tim@68-190-146-91.dhcp.eucl.wi.charter.com) |
| 01:15.57 | resistance | Strom_C: ? |
| 01:16.14 | Strom_C | what do people make fun of you for? |
| 01:16.18 | Qwell | ohm? |
| 01:16.24 | resistance | that .conf |
| 01:16.36 | resistance | i posted it here once before |
| 01:16.48 | resistance | zapata-auto |
| 01:17.12 | Strom_C | try this - in zapata.conf, before you include the other file, try adding "channel => 1-48" |
| 01:17.41 | resistance | ok, thanks, brb |
| 01:18.00 | Strom_C | and nothing says you "had to" autogenerate a conf file :) |
| 01:18.08 | Strom_C | real men make 'em by HAND! |
| 01:18.38 | resistance | i'm a real woman |
| 01:18.42 | resistance | hee hee |
| 01:19.10 | python_ | i am very new to asterisk, i am having some troubles getting 2 soft phones to talk to each other through a asterisk server, the two phones are behind simple linksys nat router, and the server is on the public internet, the phones ascossiate with the asterisk server, and when i make a call it shows ( sip debug ) the call is going to the right extention, but the call is never receaved |
| 01:20.12 | python_ | the asterisk server is run on openbsd, and the clients are on suse 10.2 and windows, but i tryed with 2 windows machines today also using x-lite |
| 01:21.22 | *** join/#asterisk mtgh (n=chatzill@dsl093-001-038.det1.dsl.speakeasy.net) |
| 01:25.54 | python_ | is anyone hear? |
| 01:26.23 | Strom_C | no, sorry, we all died three seconds after you started typing |
| 01:26.28 | Strom_C | food poisoning |
| 01:26.30 | python_ | :) |
| 01:26.55 | JT | Strom_C: eating irc printouts is never healthy |
| 01:27.00 | shmaltz | how do I deflate a bz2 file? |
| 01:27.14 | Strom_C | heh |
| 01:27.16 | Qwell | bunzip2 |
| 01:27.48 | JT | tar -jxvf |
| 01:28.00 | Strom_C | catsex.sh |
| 01:28.04 | JT | if it's a tar.bz2 |
| 01:28.47 | python_ | anyone have anyclue on my issue? my conf files are hear if you want to look http://timholum.com/asterisk.txt |
| 01:29.29 | Strom_C | python_: try adding nat=yes and qualify=yes to the sip.conf entries and see if that works |
| 01:30.11 | Strom_C | and make sure your contexts are correct and whatnot |
| 01:30.18 | *** join/#asterisk l2cache (n=Administ@102.133.202.68.cfl.res.rr.com) |
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| 01:32.27 | *** part/#asterisk VoipMasta (n=fabio@201.139.139.127.cableonline.com.mx) |
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| 01:35.54 | *** join/#asterisk foxxtrot_ (n=craig@c-67-185-0-172.hsd1.wa.comcast.net) |
| 01:38.51 | [hC] | Qwell: you arent still updating chan_skinny are you? |
| 01:38.58 | *** join/#asterisk guptaa (n=user@000-103-068.area3.spcsdns.net) |
| 01:38.59 | python_ | i already had the nat=yes and i tryed it with qualify=yes and no luck :( |
| 01:39.02 | [hC] | Qwell:chan_sccp is making my 7970 eat crap all the time. |
| 01:40.02 | *** join/#asterisk foxxtrot_ (n=craig@c-67-185-0-172.hsd1.wa.comcast.net) |
| 01:40.34 | *** join/#asterisk foxxtrot_ (n=craig@c-67-185-0-172.hsd1.wa.comcast.net) |
| 01:41.00 | python_ | is there a way to send a test call from the asterisk server? |
| 01:41.16 | JT | .call files |
| 01:41.22 | JT | or manager interface |
| 01:41.33 | JT | or console (if you have working sound card) |
| 01:42.03 | python_ | i am logged into the asterisk box and have the asterisk cli up? |
| 01:42.30 | *** join/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do) |
| 01:42.35 | JT | ~thebook |
| 01:42.37 | jbot | thebook is, like, a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 01:42.41 | *** part/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do) |
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| 01:46.57 | *** join/#asterisk jart (n=user@ool-44c05258.dyn.optonline.net) |
| 01:48.03 | jart | If I got a SIP call coming in to Asterisk where the SIP headers showed: From: <sip:+12345678901@123.3.6.83;isup-oli=27>;tag=BLAHBLAH |
| 01:48.16 | jart | how could I extract the isup-oli=27 in my dial plan or whatever? |
| 01:49.11 | Strom_M | jart: which provider is giving you olip data? |
| 01:49.22 | jart | bandwidth which is a level3 reseller |
| 01:49.29 | jart | why? |
| 01:49.35 | Strom_M | sweet |
| 01:49.46 | *** join/#asterisk foxxtrot_ (n=craig@c-67-185-0-172.hsd1.wa.comcast.net) |
| 01:49.46 | Strom_M | i love olip data :) |
| 01:50.00 | Strom_M | you could use the cut application |
| 01:50.08 | *** join/#asterisk servat (n=znog@60-240-204-62.tpgi.com.au) |
| 01:50.09 | jart | oh nifty |
| 01:50.15 | jart | but how do I get that string? |
| 01:50.42 | Strom_M | show application cut |
| 01:51.38 | jart | but that's assuming i can extract that olip data in the first place |
| 01:51.39 | [hC] | jart: how do you like their service? and what does it cost ya? |
| 01:51.53 | [hC] | jart: ive been looking for a level3 reseller as i dont do enough minutes yet to meet commits for them |
| 01:51.54 | *** join/#asterisk foxxtrot_ (n=craig@c-67-185-0-172.hsd1.wa.comcast.net) |
| 01:52.10 | jart | about 1.5 cents to 2 cents a minute |
| 01:52.20 | jart | they're pretty good |
| 01:52.59 | jart | not a lot of people can meet commits for level3, and even if you could they're not a fun company to deal with from what i hear |
| 01:53.19 | guptaa | I would like to create a macro that will place a call (to a cellphone) to notify someone that a voicemail was left. I was planning on using the 'h' extension after the voicemail is left to run a DeadAGI that will create a .call file. Does anyone know a better way to do this? The thing I don't like about the .call file is that I want to use a 'hunt' or 'memoryhunt' from dialparties to call a number of people until one answers the call. |
| 01:53.41 | jart | but Strom_M, i know how to use cut, i just need to /get/ the olip data |
| 01:54.03 | jart | i don't know where asterisk is hiding it, if it even extracts it from the sip data at all |
| 01:54.11 | Strom_M | jart: that header is from an actual call. right? |
| 01:54.18 | jart | yea |
| 01:54.47 | Strom_M | and what did you use to reference it in asterisk? |
| 01:55.10 | jart | i know it's there by packet sniffing |
| 01:55.39 | Strom_M | ah |
| 01:56.28 | *** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 01:56.50 | *** join/#asterisk eltech- (i=G00Ds@ool-457c93b6.dyn.optonline.net) |
| 01:58.16 | *** join/#asterisk jeffgus_ (n=jeffgus@greengables.zimage.com) |
| 01:58.45 | *** join/#asterisk Omer^ (i=Omer@203.81.232.55) |
| 01:59.02 | Strom_M | jart: there is a way |
| 01:59.08 | Strom_M | i just dont remember how |
| 02:00.49 | Strom_M | when i get back from the cafe, ill find out :) |
| 02:01.27 | jart | cool |
| 02:01.53 | jart | if you email it to jtunney@gmail.com i will love you forever :) because i have to bounce |
| 02:02.37 | Strom_M | ok |
| 02:04.26 | *** join/#asterisk Druken (n=jdumais@CPE000854ddcdb1-CM00137189cb0c.cpe.net.cable.rogers.com) |
| 02:05.00 | *** join/#asterisk aao_pwner|oper (n=aao_pwne@c-24-21-116-29.hsd1.mn.comcast.net) |
| 02:06.06 | *** join/#asterisk anthonyl (n=anthonyl@65.4.17.52) |
| 02:06.23 | Druken | anyone have allison saying USA ? |
| 02:06.58 | resistance | strom_c: i tried the channel => 1-48, and i can't start asterisk |
| 02:07.21 | [Outcast] | just take u s a from the photonetics stuff |
| 02:07.28 | Strom_M | resistance: asterisk -cvvvvvg and pastebin the error |
| 02:07.33 | *** join/#asterisk lat1234 (n=lat@61.9.4.58) |
| 02:07.35 | *** join/#asterisk brut- (n=brut@66.173.4.254) |
| 02:07.37 | lat1234 | hello |
| 02:07.42 | Qwell | [Outcast]: why the phonetics? |
| 02:08.10 | lat1234 | anybody here knows how to let asterisk passthru a firewalll... what will be the port to allow if asterisk in sip? |
| 02:08.21 | Druken | i think he ment the letters, but that wouldn't sound right... |
| 02:08.30 | Strom_M | lat: one time only please |
| 02:08.33 | Druken | people don't use U S A, they say USA |
| 02:08.46 | lat1234 | ok |
| 02:08.53 | lat1234 | so anybody |
| 02:08.59 | [Outcast] | if the sound file didn't exist already you could piece one together |
| 02:09.31 | *** join/#asterisk zmef420 (n=zmef420@metarb3-pool4-182.mtco.com) |
| 02:09.40 | Strom_M | or you could see if she says "united states" |
| 02:10.23 | *** join/#asterisk Winkie_ (i=sd@87-194-8-125.bethere.co.uk) |
| 02:13.07 | Druken | nah... need USA |
| 02:13.10 | Druken | company name.... |
| 02:14.08 | wunderkin | and the company name is only usa? sorry that one is taken |
| 02:14.47 | Strom_M | hire allison to say the damned thing |
| 02:15.32 | resistance | strom_M, what information do you want? where do i find it? |
| 02:15.38 | Strom_M | or can "bjorns authentic tacos usa incorporated" not afford the twelve dollars? |
| 02:16.28 | Strom_M | resistance: pastebin the whole thing |
| 02:16.36 | resistance | the full log? |
| 02:16.47 | Druken | Strom_C: no idea... hehehe i was just doing it as a favour for someone |
| 02:16.57 | Strom_M | the colsole output from that command |
| 02:17.07 | resistance | i'm using putty, |
| 02:17.07 | *** join/#asterisk stkn_ (i=nobody@gentoo/developer/pdpc.active.stkn) |
| 02:17.28 | Strom_M | dont you know how to redirect? |
| 02:17.40 | resistance | nope |
| 02:17.50 | Strom_M | christ |
| 02:17.58 | resistance | u said it |
| 02:18.05 | resistance | it's a sad shitsuation |
| 02:18.12 | lat1234 | anyone who knows what port to allow in the firewall to let asterisk-sip passthrough? |
| 02:18.14 | Strom_M | learn linux :) |
| 02:18.21 | resistance | hey man, i'm trying |
| 02:18.29 | resistance | it's a huge learning curve |
| 02:18.36 | hads | google bash redirection |
| 02:18.55 | lat1234 | i know linux... what i want to know are the ports to allow |
| 02:20.03 | Strom_C | lat1234: i was talking to resistance |
| 02:20.06 | hads | If you know Linux then grep -i sip /etc/services |
| 02:20.14 | Strom_C | ok, now that i'm home I can type again :) |
| 02:20.26 | Strom_C | resistance: asterisk -cvvvvvvg > output.txt |
| 02:20.33 | Strom_C | then pastebin the contents of output.txt |
| 02:20.39 | Druken | lat1234: 5060, 10000 - 20000 |
| 02:20.59 | guptaa | grep -i bindport /etc/asterisk/sip.conf |
| 02:21.48 | *** join/#asterisk l2cache (n=Administ@102.133.202.68.cfl.res.rr.com) |
| 02:23.03 | lat1234 | thanks drunken... but it wont work to me |
| 02:23.15 | Strom_C | lat1234: UDP |
| 02:24.27 | resistance | Strom_M: http://pastebin.ca/284469 |
| 02:25.03 | Strom_C | resistance: i don't think that's the full file |
| 02:25.13 | *** join/#asterisk Winkie (i=sd@87-194-8-125.bethere.co.uk) |
| 02:25.17 | lat1234 | yes its udp |
| 02:25.20 | JT | eww trixbox |
| 02:25.25 | lat1234 | but no still no audio... |
| 02:25.32 | JT | no wonder my browser froze for a few seconds loading up all that text |
| 02:25.45 | resistance | well that's what i got from asterisk -cvvvvvg |
| 02:25.54 | Strom_C | holy crap, it /is/ trixbox |
| 02:26.00 | Strom_C | resistance: no wonder you're having trouble :) |
| 02:26.06 | *** join/#asterisk obnauticus (n=aao_pwne@c-24-21-116-29.hsd1.mn.comcast.net) |
| 02:26.12 | resistance | i know that shit sucks |
| 02:26.21 | *** join/#asterisk riddlebox (n=james@75-132-205-166.dhcp.stls.mo.charter.com) |
| 02:26.37 | resistance | but my knowledge is limited and i though this would be the easier routr |
| 02:26.38 | l2cache | I use asterisk at my work, why do people not like trixbox? is it only good for small setups? |
| 02:26.43 | riddlebox | if I get a sipura ata device from broadvoice, is there a way to unlock it so I can use both ports? |
| 02:26.48 | Strom_C | ~trixbox |
| 02:26.58 | jbot | i heard trixbox is NOT supported here! People using it should join #trixbox or #freepbx (FreePBX is the new name of AMP) |
| 02:27.10 | Strom_C | oh dammit |
| 02:27.14 | Strom_C | thats not the one i wanted |
| 02:27.16 | Strom_C | ~freepbx |
| 02:27.18 | jbot | freepbx is probably the Microsoft BOB of PBXes and NOT supported here! People using it should join #freepbx (FreePBX is the new name of AMP) |
| 02:27.23 | Strom_C | there we go :) |
| 02:27.25 | lat1234 | im using isa 2004 ... |
| 02:27.33 | *** join/#asterisk nassy (n=nassy@207-38-197-201.c3-0.wsd-ubr1.qens-wsd.ny.cable.rcn.com) |
| 02:27.47 | lat1234 | there is no audio out... |
| 02:27.47 | Druken | ~amp |
| 02:27.49 | jbot | amp is probably NOT supported here! People using it should join #freepbx (FreePBX is the new name of AMP) |
| 02:29.37 | l2cache | my friend swears by trixbox. Is there a good argument for using straight asterisk over trix? other that the obvious reasons(gui is not for admins) |
| 02:29.51 | Strom_C | trixbox is full of unnecessary garbage |
| 02:30.13 | l2cache | any other points...i appreciate it |
| 02:30.23 | resistance | Strom_C: what i'm trying to do is slowly switch myself over the straight asterisk |
| 02:30.26 | Druken | at least if you build it, when it breaks, you have a slight chance of figuring out wtf went wrong.... |
| 02:31.13 | l2cache | i run straight asterisk at home and at work 400+ extensions...has anyone used trixbox in a large-scale environment |
| 02:31.15 | cjlowe | wtf mate :) |
| 02:31.17 | Strom_C | oh there is no "slowly switch myself" |
| 02:31.30 | Strom_C | just jump in the deep end now while you still have a fighting chance |
| 02:31.36 | JT | cjlowe: yeah true blue, dinky di |
| 02:31.51 | JT | l2cache: only insane people |
| 02:31.52 | cjlowe | JT too right mate |
| 02:31.58 | l2cache | true that |
| 02:32.07 | JT | ripper |
| 02:32.12 | *** join/#asterisk hfb (n=hfb@pool-71-118-252-158.lsanca.dsl-w.verizon.net) |
| 02:32.31 | hads | Weird aussies :) |
| 02:32.32 | resistance | well i'm f'd then |
| 02:32.43 | *** join/#asterisk RoyK (n=roy@217-175-39.100710.adsl.tele2.no) |
| 02:32.47 | JT | hads: go say bugger, for me :D |
| 02:33.00 | hads | :) |
| 02:33.04 | resistance | bugger is another word for shitpacker |
| 02:33.12 | resistance | not nice |
| 02:33.25 | hads | resistance: Well it really depends where you are from . |
| 02:33.26 | JT | bugger |
| 02:33.30 | resistance | lol |
| 02:33.34 | resistance | unsucht |
| 02:33.41 | JT | and what decade you are living in, too |
| 02:34.00 | resistance | if i'll wanna pack shit, i'll use a baled |
| 02:34.10 | resistance | ~baler |
| 02:34.14 | l2cache | has anyone experimented/used an asterisk load balancing/high availability setup at all? |
| 02:34.19 | hads | Yes, the Internet being international really seems to catch some people out. |
| 02:35.25 | JT | where some people = americans |
| 02:35.25 | *** join/#asterisk lpmusic (n=dballeng@reddy.d-11.denetron.net) |
| 02:35.26 | hads | :) |
| 02:35.40 | resistance | Strom_C: trixbox is all in the configs, technically i could just adjust the configs and get fresh asterisk, or an i fos? |
| 02:35.44 | resistance | ~am |
| 02:35.47 | jbot | somebody said am was an application manager. Armenia |
| 02:35.59 | JT | resistance: trixbox will overwrite them if done wrong |
| 02:36.24 | JT | and have to ever seen the difference between a dialplan made to do something in trixbox, and a dialplan coded by hand to do the same thing? |
| 02:36.27 | resistance | yup, if i use that web editing thingy, i don't |
| 02:36.27 | Strom_C | resistance: taking trixbox and slimming it down is entirely the wrong approach |
| 02:36.31 | cjlowe | resistance, the _additional files are the ones that trixbox seems to put most of your stuff in... |
| 02:36.34 | JT | the trixbox ones are at least 3 times bigger |
| 02:36.45 | JT | with tonnes of garbage |
| 02:36.47 | cjlowe | resistance, better to build from scratch and copy the relevant settings across :) |
| 02:41.52 | *** join/#asterisk _cleric_ (n=dacleric@p54822EBD.dip0.t-ipconnect.de) |
| 02:43.24 | *** join/#asterisk Avochelm (n=damien@gw-morphett.koalatelecom.com.au) |
| 02:43.28 | lpmusic | has anyone in here worked with queues and announcements (of what queue it is)? Ideally I'd like to make it so the announcement plays "you have a call for: x" then you have to ack the call (#) and then you take it rather than ack the call then hear what it's for |
| 02:49.36 | *** join/#asterisk hoobastooba (n=ckwall@c-71-195-199-149.hsd1.ut.comcast.net) |
| 02:49.53 | *** join/#asterisk bjohnson (n=bjohnson@i209-195-110-28.cia.com) |
| 02:50.02 | orlok | Avochelm: hwy, you guys do wireless internet? |
| 02:50.25 | *** part/#asterisk RoyK (n=roy@217-175-39.100710.adsl.tele2.no) |
| 02:50.42 | Avochelm | orlok, yes, in adelaide and the iron triangle |
| 02:50.53 | orlok | havent heard that term before |
| 02:50.57 | orlok | iron triangle |
| 02:51.12 | orlok | Avochelm: we have used a few wisp's, and in general, they have all sucked :-( |
| 02:51.31 | orlok | cant cope with something like battleship radar testing withou falling over |
| 02:51.43 | orlok | :) |
| 02:52.24 | lat1234 | question --> what port should be opened aside from the dns in order to use domain name instead of ip? |
| 02:52.33 | lat1234 | im using xlite |
| 02:53.01 | lat1234 | if i use ip the registration of xlite is succesfful |
| 02:53.11 | Avochelm | orlok, the iron triangle is the mining areas of SA around port augusta |
| 02:53.19 | lat1234 | if i use domain name the registration fails although dns is already allowed in the firewall |
| 02:55.12 | *** join/#asterisk kyo (n=kio@ool-4577ae5e.dyn.optonline.net) |
| 02:55.58 | *** part/#asterisk l2cache (n=Administ@102.133.202.68.cfl.res.rr.com) |
| 02:57.04 | hoobastooba | I am still trying to figure out why asterisk is taking up 54402 of my ram... Here is what I have in top right now: http://pastebin.ca/284511 |
| 02:57.58 | hoobastooba | i am really chewing up the ram!!!! http://pastebin.ca/284512 |
| 02:58.22 | hoobastooba | I am running Asterisk SVN-branch-1.2-r48272M |
| 02:58.37 | hoobastooba | any help would be greatly appreciated. |
| 02:58.55 | orlok | Avochelm: ahh, thought it would have been something like that |
| 02:59.10 | orlok | Avochelm: Only thing i like from SA is Coopers :) |
| 02:59.30 | orlok | port augusta.. van nats were there a few years ago :) |
| 03:00.12 | Avochelm | :) |
| 03:00.26 | orlok | man |
| 03:00.29 | hoobastooba | here is from ps aux | grep asterisk http://pastebin.ca/284518 |
| 03:00.38 | orlok | my first "real" set of holidays in years starts in a few days |
| 03:00.52 | orlok | i am stitting here crossing my fingers that nothing breaks |
| 03:01.10 | orlok | "Dont touch anything, dont try anything, just sit here and hope..." |
| 03:01.10 | *** join/#asterisk juanjoc (n=juanjoc@248-32-235-201.fibertel.com.ar) |
| 03:03.35 | *** join/#asterisk irq (n=dan@wsip-70-167-112-5.sd.sd.cox.net) |
| 03:04.47 | *** join/#asterisk Osochebol (n=Osochebo@210.245.33.1) |
| 03:10.14 | Osochebol | HI ALL |
| 03:10.29 | Osochebol | S.O help me for Asterisk realtime view database from UNIXODBC? |
| 03:14.58 | hoobastooba | is 1309044 bytes in 202 allocations in file 'chan_sip.c' too much for 8 bridged calls? |
| 03:16.12 | [TK]D-Fender | hoobastooba : 1.3meg seem too much? |
| 03:17.48 | hoobastooba | i just for the life of me cannot figure out why I have to restart my asterisk server once every three days. that is my highest use part of asterisk. |
| 03:17.52 | hoobastooba | I am stumped. |
| 03:17.55 | *** join/#asterisk tracy_ (n=tracy@cpe-024-074-100-250.carolina.res.rr.com) |
| 03:18.13 | hoobastooba | it starts to swap after two days. |
| 03:18.21 | hoobastooba | then calls start sounding horrible. |
| 03:18.53 | hoobastooba | I have been to the list and google quite a few times on this... sorry to keep bringing it back. |
| 03:18.59 | hoobastooba | i cannot figure out what I am doing wrong. |
| 03:20.09 | tracy_ | hi every1. i have had a trixbox running in a VM and working nicely with a few voip providers. i want to take to next level and get a cheap fxo card to connect to pstn. should i just buy a x100p or one of the clone cards? |
| 03:20.51 | tracy_ | and i am going to take it out of a vm, before i get into trouble by from the ppl here |
| 03:20.52 | [TK]D-Fender | tracy_ : How many lines to start? Where do you see this going? |
| 03:21.11 | tracy_ | this is home use with a very small 2 person company |
| 03:21.38 | tracy_ | at the moment i have too many phones in the house |
| 03:22.05 | tracy_ | this phone to call out voip, that one for receiving calls... etc |
| 03:22.36 | hoobastooba | [TK]D-Fender: to me 1.3 meg seems to much |
| 03:22.57 | [TK]D-Fender | tracy_ : With * any phone can do EVERYTHING. |
| 03:23.13 | cjlowe | [TK]D-Fender, them's fightin' words |
| 03:23.14 | [TK]D-Fender | hoobastooba : Does it release it upon termination of the channels? |
| 03:23.19 | hoobastooba | no |
| 03:23.32 | [TK]D-Fender | cjlowe : Where I come from all words is fightin' words! |
| 03:23.34 | hoobastooba | it just keeps growing |
| 03:23.42 | [TK]D-Fender | hoobastooba : What version? |
| 03:23.55 | tracy_ | [TK]D-Fender: my problem is my landline is not plugged into * as i dont have a fxo card |
| 03:24.22 | cjlowe | [TK]D-Fender, in soviet russia, words fight you! |
| 03:24.23 | [TK]D-Fender | tracy_ : Do you need any more devices (extensions) as well? |
| 03:24.31 | JT | tracy_: you cannot buy real X100P cards anymore |
| 03:24.35 | tracy_ | nope |
| 03:24.42 | JT | the ones that are out there are likely to be crap |
| 03:24.44 | tracy_ | JT, i'm looking on ebay atm |
| 03:24.50 | JT | none are real |
| 03:24.59 | JT | unless it's an old second hand one |
| 03:25.08 | JT | almost impossible to verify without seeing it in person |
| 03:25.26 | JT | either get a TDM400P or an ATA like the Sipura SPA-3102 |
| 03:25.39 | hoobastooba | [TK]D-Fender: Asterisk SVN-branch-1.2-r48272M ... and I take it back... it did just release a bunch... |
| 03:25.41 | tracy_ | last i looked those were $$$$ |
| 03:25.48 | hoobastooba | i have more calls than before and now I am at 1.0 |
| 03:26.02 | tracy_ | for home use it seemed an overkill :( |
| 03:26.10 | JT | Sipura ATAs are very good value for money |
| 03:26.25 | JT | if you can't afford them for home/soho office, i don't know what you can afford |
| 03:26.34 | hoobastooba | see, here is my real concern... http://pastebin.ca/284518 |
| 03:26.54 | [TK]D-Fender | tracy_ : so how many analog lines max are you thinking about bringing in in say the next year? |
| 03:26.59 | hoobastooba | as i start to run out of memory and start to swap, I can kill the extra processes and get my memory back. |
| 03:27.06 | [TK]D-Fender | hoobastooba : how old is thatr SVN? |
| 03:27.07 | tracy_ | i'm pretty cheap, run 2 linksys paps that cost $0 |
| 03:27.24 | tracy_ | [TK]D-Fender: only the one i have |
| 03:27.38 | hoobastooba | about 2 weeks |
| 03:27.41 | [TK]D-Fender | tracy_ : Then just get an X100P clone and be done with it. |
| 03:28.03 | tracy_ | ok, gonna try that, tx |
| 03:28.20 | JT | X100P fakes are great if your time is worth nothing |
| 03:28.20 | JT | so yeah, go for it |
| 03:28.21 | [TK]D-Fender | tracy_ : If you're saying your very cheap and don't need more than 1 line and don't have any need for an extra phone, then X100P it is. |
| 03:28.26 | orlok | hmmm |
| 03:28.32 | JT | yeah might get one that mostly works, or you might get a complete dud |
| 03:28.40 | orlok | by voip provider says asterisk people need to fix their code, heh |
| 03:28.46 | [TK]D-Fender | JT : Tony Robins you are not! |
| 03:28.47 | orlok | my voip my, even |
| 03:28.54 | JT | [TK]D-Fender: who? |
| 03:29.07 | [TK]D-Fender | JT : Motivation speaker. massively famous. |
| 03:29.17 | [TK]D-Fender | JT : Get back under that rock! |
| 03:29.20 | JT | i see |
| 03:29.21 | *** join/#asterisk bbsf (n=bill@adsl-75-6-246-163.dsl.pltn13.sbcglobal.net) |
| 03:29.29 | JT | i hate motivational speakers and all similar things |
| 03:29.36 | JT | possibly why i don't know who he is |
| 03:29.39 | orlok | [TK]D-Fender: us aussies are not always infected by your scary american memes! |
| 03:29.41 | [TK]D-Fender | JT : Besides tracy_ here is running Trixbox, so mediocrity is the NORM ;) |
| 03:30.00 | [TK]D-Fender | orlok : .... I'm CANADIAN :D |
| 03:30.04 | orlok | oh |
| 03:30.13 | [TK]D-Fender | eh! |
| 03:30.35 | JT | north american names, then :P |
| 03:30.45 | orlok | hmm |
| 03:31.48 | bbsf | anyone who might help with SIP trouble connecting to Hong Kong HKBN-2b? |
| 03:32.12 | JT | yes i'm sure everyone here knows what HKBN-2b is... |
| 03:33.23 | bbsf | well, if it's not recognized it's unlikely help will be forthcoming :-) (== Hong Kong Broadband Network "2b" service) |
| 03:34.30 | *** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 03:35.49 | *** join/#asterisk ShadowHntr (n=sentinel@wikipedia/Shadowhntr) |
| 03:37.29 | *** mode/#asterisk [+o mog] by ChanServ |
| 03:38.00 | hmmhesays | allison fisher is kind of hot |
| 03:39.23 | [TK]D-Fender | hmmhesays : Pool... a game thats ALL about foreplay... |
| 03:40.02 | [TK]D-Fender | hmmhesays : Lean down over a flat hard surface. Spread your legs. Give long steady strocks with your "stick". |
| 03:40.18 | *** join/#asterisk lowlevel (n=Stuart@CPE000e0c057fad-CM000f9f7d6742.cpe.net.cable.rogers.com) |
| 03:40.32 | [TK]D-Fender | hmmhesays : But remember its not how hard you slam your stick... but rather how you position your balls ;) |
| 03:40.37 | Qwell | trying to put "balls" in a "pocket" |
| 03:43.47 | [TK]D-Fender | hmmhesays : Ewe Laurance = hottie.... |
| 03:44.57 | [TK]D-Fender | Lets not even get started about Jeanette Lee...... |
| 03:44.58 | JT | that sounds similar to Hugh Laurie... |
| 03:44.58 | *** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 03:50.22 | Qwell | ! |
| 03:50.51 | file | get back to work, slacker! |
| 03:52.27 | *** join/#asterisk bmg505 (n=leon@c1-12-15.rndf.isadsl.co.za) |
| 03:52.55 | *** join/#asterisk lee_is_me (n=chatzill@12-227-176-77.client.mchsi.com) |
| 03:59.53 | *** part/#asterisk lee_is_me (n=chatzill@12-227-176-77.client.mchsi.com) |
| 04:00.08 | *** join/#asterisk lee_is_me (n=chatzill@12-227-176-77.client.mchsi.com) |
| 04:05.13 | lee_is_me | hi all |
| 04:06.54 | *** join/#asterisk naftali5 (n=naftali5@ool-44c05121.dyn.optonline.net) |
| 04:08.00 | lee_is_me | I'm having a couple of problems with features.conf that I was hoping I could get some help with |
| 04:08.04 | *** join/#asterisk BigCanOfTuna (n=arustad@dsl-mac-66-18-226-119-cgy.nucleus.com) |
| 04:09.01 | BigCanOfTuna | Can someone point me in the right direction for initiating a call from the server side, like as a result from making a request to a browser....if I remember correctly, it has something to do with creating a file in a certain directory....sorry, mind drawing a blank right now. |
| 04:09.35 | BigCanOfTuna | Doh, just found something on voip-info.org. |
| 04:09.50 | lee_is_me | http://www.voip-info.org/tiki-index.php?page=Asterisk+auto-dial+out |
| 04:10.10 | JT | .call |
| 04:10.34 | *** join/#asterisk BhaalWK (i=bhaal@freenode/unconfirmed/bhaal) |
| 04:11.53 | *** join/#asterisk JVH (n=jvh@CPE-24-163-223-125.kc.res.rr.com) |
| 04:13.36 | JVH | Anyone using Grandstream GXP2000 phones? |
| 04:14.01 | *** join/#asterisk Dr-Linux|home (n=Dreamer@DSL-202-59-73-131.nexlinx.net.pk) |
| 04:16.39 | *** part/#asterisk hoobastooba (n=ckwall@c-71-195-199-149.hsd1.ut.comcast.net) |
| 04:17.05 | lee_is_me | I'm trying to get some help with features.conf. I am having the following problems: |
| 04:17.06 | lee_is_me | 1. If a call comes in and I place that call into callparking and then retrieve that call, the outside call will now have transfer capability when I did not give it transfer in the dial comment (no "T" was given, only "t"). I've tested this from a soft phone and a hard phone. Can anyone suggest a pointer on where to look for the problem? |
| 04:17.07 | lee_is_me | 2. I'm still using asterisk 1.2.4 and am having trouble getting features.conf variables such as blindxfer => *1 to work. No matter what I do, only the # key will trigger a transfer. |
| 04:17.09 | lee_is_me | Any ideas? |
| 04:19.03 | icyfire0573 | You've got to allow transfers in the dialplan to make transfers work. |
| 04:19.07 | [TK]D-Fender | lee_is_me : What kind of phones re you using? |
| 04:20.08 | lee_is_me | Fender: I am using a budgetone 100 |
| 04:20.32 | [TK]D-Fender | lee_is_me : Any other phones? |
| 04:20.40 | lee_is_me | icyfire: But why must an outside caller need to have transfer capabilities? |
| 04:21.16 | lee_is_me | no just that and xlite |
| 04:22.03 | [TK]D-Fender | lee_is_me : Ok, if not for x-lite, the BT has a dedicated transfer feature. Maybe just get a better soft-[hone so you don't have to touch features.conf ever again. |
| 04:23.19 | lee_is_me | LOL, I want to touch features.conf! Otherwise I'd just use trixbox or fonality ;) |
| 04:23.20 | [TK]D-Fender | DTMF driven call-features = ass |
| 04:24.14 | [TK]D-Fender | lee_is_me : No, using features.conf driven options with app_dial is crap. get a real soft-phone that has native transfer capabilities and you won't ahve to worry about crap like this impacting parked calls, etc. |
| 04:25.00 | lee_is_me | OK, I see where you're coming from |
| 04:25.22 | rpm | i like being able to dial #9 and it start playing tt-monkeys to the person i am calling.. |
| 04:25.25 | *** join/#asterisk luke-jr_work (n=luke-jr@2002:1891:f663:0:205:4eff:fe44:18ed) |
| 04:26.09 | [TK]D-Fender | rpm : ok, fine... for NON-basic stuff sure. |
| 04:28.05 | lee_is_me | Fender: At any rate, you're saying that I can only use the # key to transfer because it's built into the BT and maybe soft phone? |
| 04:28.52 | *** join/#asterisk creativx (n=creadure@226.62-97-205.bkkb.no) |
| 04:30.03 | [TK]D-Fender | lee_is_me : No, I'm saying with the BT you don't need "tT" or any of that period, and only X-Lite needs it because they crippled the native SIP transfer feature on it. get another soft-phone and you can ditch "tT" altogether. |
| 04:31.23 | hmmhesays | that was a pretty bad ass match, the dutchess of doom beat out the kim |
| 04:31.40 | lee_is_me | Fender: Ah got you now. I wasn't sure how it related to my original questions. |
| 04:32.51 | lee_is_me | If I'm not including "T" in the dial command to dial the extension, then any idea why when I put caller into parking and retrieve them, they now have xfer ability? |
| 04:33.53 | tzanger | woo |
| 04:34.02 | [TK]D-Fender | lee_is_me : I'm guessing based on who is calling who at the point where the call is picked up. Its a bridginge question with parking. |
| 04:34.07 | tzanger | Quesnel, BC is ... meh |
| 04:34.36 | *** join/#asterisk Faithful1 (n=Faithful@ns.linuxterminal.com) |
| 04:35.16 | lee_is_me | Fender: Just seems odd and not very secure to have an outside caller get xfer ability into my system... |
| 04:35.43 | tzanger | lee_is_me: so don't give it to 'em :-) |
| 04:36.01 | rpm | tzanger: you're from quesnel? |
| 04:36.08 | tzanger | rpm: no, I'm in Quesnel right now |
| 04:36.31 | rpm | thats rough. |
| 04:36.54 | tzanger | rpm: heh |
| 04:38.18 | lee_is_me | Ah well, maybe I'll have better luck on the message boards. Night all. |
| 04:38.23 | *** part/#asterisk lee_is_me (n=chatzill@12-227-176-77.client.mchsi.com) |
| 04:39.18 | tzanger | so he gets a call in, parks the dude and then retrieves him and the caller can transfer? |
| 04:39.31 | JT | aparently |
| 04:41.22 | tzanger | hmm |
| 04:41.25 | *** part/#asterisk JVH (n=jvh@CPE-24-163-223-125.kc.res.rr.com) |
| 04:41.26 | tzanger | I don't think I've ever run into that |
| 04:41.30 | tzanger | I don't really do that though |
| 04:41.33 | tzanger | my phone use is very simple |
| 04:42.02 | tzanger | even though I'm an asterisk guru |
| 04:42.03 | tzanger | heh |
| 04:42.25 | tzanger | it's a zen thing... when you have mastered the *, you will get by with a normal phone |
| 04:42.54 | JT | heh |
| 04:43.04 | JT | i guess transfering is pretty advanced :P |
| 04:43.19 | tzanger | indeed |
| 04:43.34 | *** join/#asterisk pemuda (n=hyde@202.169.224.2) |
| 04:43.39 | pemuda | hello |
| 04:44.29 | pemuda | anybody have a script to store data in mysql to write in sip.conf? |
| 04:44.34 | [TK]D-Fender | The only reason for features.conf transfers is X-Lite. Virtually every other solution has a native transfer option. (short of letting inbound boring PSTN callers think they're extensions) |
| 04:44.52 | *** join/#asterisk _VIle (n=vile@bc182112.bendcable.com) |
| 04:45.16 | [TK]D-Fender | pemuda : Please rephrase. That is unclear |
| 04:45.28 | JT | [TK]D-Fender: what about analogue phone extensions? |
| 04:46.00 | *** join/#asterisk ronaldl79 (n=chatzill@75.119.1.39) |
| 04:46.18 | [TK]D-Fender | JT : Any real FXS interface will let you trigger this stuff on a hook-flash therefore liberating pre-hookflash DTMF for what its intended for |
| 04:46.27 | pemuda | i need to write a new account in sip.conf based on data in some mysql database |
| 04:46.42 | ronaldl79 | Well, #Asterisk-Gui is a lively bunch tonight. More like dead silence. |
| 04:46.52 | JT | [TK]D-Fender: ah hmm |
| 04:46.52 | [TK]D-Fender | pemuda : And what do you want to use as the trigger to do this? |
| 04:46.59 | pemuda | yeah |
| 04:46.59 | JT | ronaldl79: welcome to every dya |
| 04:47.01 | pemuda | rightt |
| 04:47.01 | JT | day |
| 04:47.24 | ronaldl79 | JT: I'm just trying to find out who else is having 404 errors with Asterisk-Gui after logging on. |
| 04:48.04 | pemuda | <[TK]D-Fender>: maybe some web application |
| 04:48.09 | ronaldl79 | I've been searching everywhere for answer and haven't found anything. It doesn't make sense to me, because the config is flawless. |
| 04:49.07 | pemuda | <[TK]D-Fender>:like cgi-bin |
| 04:49.24 | [TK]D-Fender | pemuda : Well so far these actions have nothing to do with *. Your web-app is going to do its thing all on its own |
| 04:50.24 | pemuda | <[TK]D-Fender>: or maybe there is some shell or perl script so i can execute via crontab? |
| 04:50.29 | *** part/#asterisk bbsf (n=bill@adsl-75-6-246-163.dsl.pltn13.sbcglobal.net) |
| 04:51.21 | [TK]D-Fender | pemuda : You can do it any which way you want. this is web programming and has nothing to do with *. |
| 04:51.46 | pemuda | [TK]D-Fender:what is *. ? |
| 04:52.04 | tzanger | there's a bird or mouse in my A/C unit in the hotel room |
| 04:52.15 | BigCanOfTuna | I have a .call file that calls my cell phone, I want it to prompt me for a phone number when I pick up, however, it seems that as soon as it dails Zap/1/xxx it start prompting for the number (before I even have time to answer the phone), is there a command that tells the dial plan to wait until the connection is established? |
| 04:52.27 | *** join/#asterisk eltech (i=G00Ds@ool-457c93b6.dyn.optonline.net) |
| 04:53.22 | [TK]D-Fender | pemuda : ASTERISK. You know. The PBX application who's channel you're in. |
| 04:54.13 | [TK]D-Fender | pemuda : Thats like going to Mercedes Benz and asking how to make a milk-shake. |
| 04:57.45 | [TK]D-Fender | BigCanOfTuna : What kind of PSTN link are you using? |
| 04:57.46 | BigCanOfTuna | Anyone? |
| 04:58.12 | *** join/#asterisk tengulre11 (n=tengulre@221.11.5.180) |
| 04:58.43 | FuriousGeorge | BigCanOfTuna: show application background |
| 04:58.46 | BigCanOfTuna | [TK]D-Fender: Sorry dude, my experience with asterisk isn't great....PSTN link? I know what the PSTN is, but what do you mean by link? |
| 04:59.22 | JT | well i assume your asterisk computer has some method to actually call your phone |
| 04:59.43 | [TK]D-Fender | BigCanOfTuna : what kind of lines? |
| 04:59.48 | FuriousGeorge | BigCanOfTuna: hes just asking how you hook up to the pstn |
| 05:00.16 | BigCanOfTuna | [TK]D-Fender: Well, just regular Telephone lines, to a Wildcat? card in my linux box. |
| 05:00.25 | BigCanOfTuna | [TK]D-Fender: I think that's what you mean. |
| 05:00.44 | FuriousGeorge | yeah, thats an analog link to the pstn, so to speak |
| 05:00.56 | FuriousGeorge | so you need a context for incoming calls |
| 05:01.02 | [TK]D-Fender | BigCanOfTuna : Sorry, but analog lines are considered "answered" as soo as zaptel seizes the line. Lack of call progress is a big reasont o go digital. |
| 05:01.22 | BigCanOfTuna | [TK]D-Fender: Boooo....so does this mean I can't do it? |
| 05:01.47 | BigCanOfTuna | [TK]D-Fender: The best I can do is put a long wait on it? |
| 05:01.51 | [TK]D-Fender | BigCanOfTuna : you can try turning regional call progress indication on but that suually results in randomly dropped calls, etc. |
| 05:02.07 | FuriousGeorge | *callprogressdetection |
| 05:02.16 | FuriousGeorge | and it sure does :) |
| 05:02.16 | [TK]D-Fender | BigCanOfTuna : Just about. Also think what happens when your cell's VOICEMAIL kicks in. This whole idea kinda blows fast... |
| 05:02.47 | BigCanOfTuna | [TK]D-Fender: Cell phone doesn't have voice mail (I'm too cheep) |
| 05:03.13 | tengulre11 | how to using the iaxclient on visual C++ 6.0? |
| 05:03.17 | [TK]D-Fender | BigCanOfTuna : lucky you.... |
| 05:03.33 | BigCanOfTuna | [TK]D-Fender: It's not an option for you? You have to have it? |
| 05:04.05 | tzanger | [TK]D-Fender's right, BigCanOfTuna... analog is a big pain in the ass and any cool hack is only cool in theory |
| 05:04.11 | tzanger | trust us on this |
| 05:04.12 | [TK]D-Fender | BigCanOfTuna : No, I meant that semi-sarcastically... in that not EVERY force in the telecom world is against you on this :) |
| 05:04.14 | tzanger | we've been there and done it |
| 05:04.28 | BigCanOfTuna | [TK]D-Fender: haha...got you. |
| 05:04.45 | [TK]D-Fender | BigCanOfTuna : Only 95% of them :D |
| 05:05.01 | [TK]D-Fender | BigCanOfTuna : Once you start pumping calls out the PSTN, its dial & pray :) |
| 05:05.16 | wunderkin | i thought that there was a dial option 'c' that i saw mentioned on the lists sometimes... but it is not listed.... |
| 05:05.19 | BigCanOfTuna | [TK]D-Fender: Alright, thanks. |
| 05:05.59 | wunderkin | that required you to press # or something when you pick up to consider it answered |
| 05:06.11 | tzanger | I should go get a drink and some desert |
| 05:06.39 | [TK]D-Fender | tzanger : Indeed you should! |
| 05:06.45 | tzanger | [TK]D-Fender: I'm lazy |
| 05:06.47 | tzanger | and tired |
| 05:06.49 | tzanger | and not really hungry |
| 05:06.56 | hads | wunderkin: Yeah, that option is there but doesn't seem to be documented. |
| 05:07.53 | tengulre11 | how to use the iaxclient.dll on Visual C++? |
| 05:08.28 | wunderkin | ummm i wonder if my config is wrong or somethin, on my ip430, i have 2 registrations - 2 diff servers, but both the same username... the calls coming in for the 2nd reg come in on the 1st line... and it has displays the sip address for caller id... :P |
| 05:08.47 | wunderkin | i think it must have a problem matching them now |
| 05:10.07 | tzanger | hello file |
| 05:10.18 | file | how're things? |
| 05:10.57 | JT | BigCanOfTuna: another potential hackish solution is using a wait priority or something |
| 05:11.14 | JT | so after a certain amount of time it then gives you the option |
| 05:11.25 | BigCanOfTuna | JT: Yea, that is what I just did...wait 10 seconds. |
| 05:11.31 | BigCanOfTuna | Worked alright. |
| 05:11.57 | BigCanOfTuna | JT: I'm going to set up a google talk bot that will call me on command...nothing special. |
| 05:12.23 | JT | someone should update the wiki to at least include the c option |
| 05:12.33 | JT | and wunderkin seems to know something about it :P |
| 05:12.44 | wunderkin | it should be in show application :P |
| 05:12.44 | resistance | what do u guys think of running asterisk on debian |
| 05:13.07 | wunderkin | its all file's fault |
| 05:13.29 | JT | resistance: works fine |
| 05:13.41 | resistance | better than centos? |
| 05:13.48 | file | wunderkin: darn right |
| 05:13.49 | wunderkin | how about... running asterisk on..... hmm.. os/2? :P |
| 05:13.55 | JT | it's just a distribution |
| 05:14.09 | JT | centos has a bug that affects asterisk too, so i guess you could say it works better |
| 05:14.14 | WilliamK | wunderkin, that would be just down out right wacky |
| 05:14.21 | wunderkin | woot |
| 05:14.25 | *** join/#asterisk Strom_M (n=strom@netblock-66-159-243-60.dslextreme.com) |
| 05:14.53 | [TK]D-Fender | resistance : Almost any *nix will do. |
| 05:15.07 | JT | except if you want to use hardware |
| 05:15.12 | JT | then usually a form of linux |
| 05:15.15 | JT | is easiest |
| 05:15.25 | JT | (and sometimes the only thing supported) |
| 05:15.57 | resistance | jt: appreciate it |
| 05:19.40 | Strom_M | resistance, finally decide to ditch trixbox? |
| 05:19.42 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 05:25.35 | X-Rob | JT, 'a bug that affects asterisk'? What bug? |
| 05:27.12 | resistance | trixbox fuckin sucks my #$@! |
| 05:27.13 | JT | ~centosbug |
| 05:27.18 | jbot | hmm... centosbug is a problem with the latest Centos kernels (4.2 and 4.3). To fix it, paste everything inside the quotes into a root shell: "sed -i s/rw_lock/rwlock/ /usr/src/kernels/*/include/linux/spinlock.h" |
| 05:27.52 | resistance | i am a very pissed off person right now |
| 05:28.19 | resistance | stron_m, one question, can i get something that will log calls like trixbox, i like that feature |
| 05:28.30 | tengulre11 | anybody using AsteriskNOW ? |
| 05:28.41 | tengulre11 | is that good? |
| 05:29.53 | [TK]D-Fender | resistance : You can use the same GUI for tracking your CDR's without surrendering your entire system over to a GUI. |
| 05:30.57 | JT | [TK]D-Fender: how? |
| 05:31.47 | [TK]D-Fender | JT : What do you mean how? CDR's can be viewed and processed by a large number of GUI;d tools that won't impede how you deploy the rest of your system. |
| 05:32.26 | JT | right |
| 05:32.27 | JT | well |
| 05:32.44 | JT | i thought you meant you could use the trixbox CDR viewer without the rest of it |
| 05:34.23 | [TK]D-Fender | JT : You can. It just reads them out of a MySQL Database. just keep the rest of the GUI away from your files by whatever means you feel like and there you go. |
| 05:34.47 | JT | heh, was that what you were suggesting? |
| 05:34.57 | [TK]D-Fender | JT : its a seperate product, jsut like FOP, and so much else that Trixbox uses. |
| 05:35.05 | JT | ah |
| 05:35.44 | [TK]D-Fender | JT : So-so. more like you could get the exact same viewer as a seperate download and build it in yourself, or get another similar one. There are plenty of them out there. |
| 05:36.13 | file | trixbox be a package of it all together matey! yarrrrrr |
| 05:36.27 | russellb | go to bed |
| 05:36.36 | file | russellb: I'm talking to a friend on my cellular telephone! |
| 05:36.49 | russellb | lame |
| 05:37.03 | russellb | i'm just jealous :-p |
| 05:37.03 | JT | "ARR, want to see what's in my box of TRIX??" |
| 05:37.08 | file | whyfor? |
| 05:37.30 | russellb | you don't know my number! |
| 05:37.38 | Corydon76-home | Are you sure? |
| 05:37.40 | file | russellb: I know 2 of your numbers! |
| 05:37.46 | russellb | Corydon76-home: no :) |
| 05:37.57 | Corydon76-home | :-) |
| 05:38.08 | Corydon76-home | Neither am I. |
| 05:38.08 | russellb | i have ... no number! |
| 05:38.18 | file | I have too many! >_< |
| 05:38.22 | Corydon76-home | 877-4-TILGHMAN |
| 05:38.39 | *** join/#asterisk juanmanuel (n=jmacz@190.24.97.151) |
| 05:38.41 | russellb | 7 |
| 05:38.59 | russellb | i'm *that* old ... I got the number 7 |
| 05:40.04 | Corydon76-home | In Soviet Russia, number calls YOU! |
| 05:40.18 | FuriousGeorge | i got an icq number in the 6 digits i still use from time to time |
| 05:40.22 | juanmanuel | Hi everyone, any one knows where I can find information about the prices for assisting to the next astricon (or the prices of the last one at Dallas). The Astricon site doesn't provides much information on this. |
| 05:40.55 | FuriousGeorge | assisting!=attending my multilingual friend |
| 05:40.55 | file | russellb: are you... 911? |
| 05:41.44 | juanmanuel | FuriousGeorge, yes I mean attending (sorry e.s.l) |
| 05:42.15 | FuriousGeorge | juanmanuel: lo siento, pero no se el precio ;) |
| 05:43.18 | juanmanuel | FuriousGeorge, vale, en todo caso gracias :P |
| 05:43.32 | russellb | file: yes no maybe so |
| 05:43.36 | file | eep |
| 05:43.37 | FuriousGeorge | no hay porque |
| 05:43.41 | juanmanuel | :) |
| 05:43.52 | file | gah emails! go away. |
| 05:44.03 | *** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca) |
| 05:44.16 | russellb | file: fine. not that important |
| 05:46.31 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 05:46.52 | *** join/#asterisk WAudette70 (n=WAudette@c-71-237-146-239.hsd1.or.comcast.net) |
| 05:47.01 | *** part/#asterisk WAudette70 (n=WAudette@c-71-237-146-239.hsd1.or.comcast.net) |
| 05:50.49 | *** join/#asterisk Newbie___ (n=me@211.24.146.11) |
| 05:51.24 | Newbie___ | hi , anyone using a adit 600 |
| 05:52.08 | tzanger | Newbie___: yep |
| 05:53.12 | Newbie___ | tzanger: thank god. if i have a 48 FXS in one box and a router card, do i still need a TDM ? |
| 05:53.21 | tzanger | yep |
| 05:53.44 | tzanger | I haven't used the router card before but unless it does TDMoE and it's the exact same spec as what Zaptel does, you're scrogged |
| 05:54.14 | Newbie___ | hmm , i though by using CMG card, i would not need a TDM card |
| 05:54.46 | tzanger | Newbie___: you need ot research first :-) |
| 05:55.08 | Newbie___ | tzanger: been lookig for days |
| 05:55.48 | *** join/#asterisk ozoneco (n=stanp@CPE-24-27-138-124.neb.res.rr.com) |
| 05:55.49 | Newbie___ | just not sure if a CMG card can handle 48 FXS |
| 05:56.18 | *** join/#asterisk anthonyl (n=anthonyl@65.4.17.52) |
| 05:56.19 | tzanger | again, unless the CMG is giving you SIP or TDMoE, and the latter is the same "brand" as what zteth uses, it's useless |
| 05:56.51 | Newbie___ | tzanger: ok |
| 05:59.31 | *** join/#asterisk swdevv (n=software@VDSL-130-13-100-226.PHNX.QWEST.NET) |
| 06:00.50 | tzanger | hahahaha |
| 06:00.51 | tzanger | http://users.skynet.be/ppc/push_puppet_toy/ |
| 06:00.51 | tzanger | love it |
| 06:01.10 | X-Rob | ~centosbug |
| 06:01.12 | jbot | i heard centosbug is a problem with the latest Centos kernels (4.2 and 4.3). To fix it, paste everything inside the quotes into a root shell: "sed -i s/rw_lock/rwlock/ /usr/src/kernels/*/include/linux/spinlock.h" |
| 06:01.23 | *** join/#asterisk stephane (n=stephane@gw.sortilege.net) |
| 06:02.37 | X-Rob | jbot no, centos bug was a problem with the 2.6.9-42 kernels prior to 2.6.9-42.0.1. If you can't compile zaptel, do a 'yum update', you're running an old kernel. If you HAVE to run an old kernel, the fix is "sed -i s/rw_lock/rwlock/ /usr/src/kernels/*/include/linux/spinlock.h" |
| 06:02.39 | jbot | X-Rob: what are you talking about? |
| 06:02.44 | X-Rob | doh |
| 06:02.51 | X-Rob | jbot no, centosbug was a problem with the 2.6.9-42 kernels prior to 2.6.9-42.0.1. If you can't compile zaptel, do a 'yum update', you're running an old kernel. If you HAVE to run an old kernel, the fix is "sed -i s/rw_lock/rwlock/ /usr/src/kernels/*/include/linux/spinlock.h" |
| 06:02.52 | jbot | X-Rob: I think you lost me on that one |
| 06:03.04 | X-Rob | jbot no, centosbug is a problem with the 2.6.9-42 kernels prior to 2.6.9-42.0.1. If you can't compile zaptel, do a 'yum update', you're running an old kernel. If you HAVE to run an old kernel, the fix is "sed -i s/rw_lock/rwlock/ /usr/src/kernels/*/include/linux/spinlock.h" |
| 06:03.06 | jbot | okay, X-Rob |
| 06:03.09 | X-Rob | There we go. |
| 06:04.27 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 06:04.53 | *** join/#asterisk Ciber311 (n=Ciber311@user-1087e94.cable.mindspring.com) |
| 06:05.03 | *** join/#asterisk irq (n=dan@wsip-70-167-112-5.sd.sd.cox.net) |
| 06:05.37 | Newbie___ | tzanger: yes, a CMG router card can support up to 48 FXS |
| 06:05.48 | JT | X-Rob: dude, in private next time :P |
| 06:05.53 | Newbie___ | and uses MGCP protocol |
| 06:06.18 | tzanger | Newbie___: oh, MGCP |
| 06:06.24 | tzanger | you can probably get away with it then |
| 06:06.43 | X-Rob | MGCP *spit* |
| 06:06.49 | Newbie___ | tzanger: only support FXS at this time |
| 06:09.44 | *** join/#asterisk npalmer (n=nate@208.53.47.148) |
| 06:11.15 | *** join/#asterisk tr2x (n=alvar@80-218-185-55.dclient.hispeed.ch) |
| 06:13.34 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 06:13.50 | Strom_M | well, i think the aastra 480i wins the "bedside phone" contest for now |
| 06:14.02 | JT | no ip650? |
| 06:14.14 | tzanger | bedside phone for me is a Panasonic 5GHz cordless |
| 06:14.17 | Strom_M | dont have one of those |
| 06:14.19 | hads | Huh, what's the critera for a bedside phone? |
| 06:14.38 | tzanger | hads: barowarm-chicka-chicka-barowarm-barowarm ringtones |
| 06:14.43 | hads | My main critera is no lights. |
| 06:14.50 | Strom_M | but the 480i wins on the grounds of having a loud ringer that sounds like a phone ringing and not like synthesizers farting, and a backlit screen |
| 06:14.52 | hads | tzanger: Hahaha |
| 06:15.17 | *** join/#asterisk Splat (n=Splat@220-253-135-77.TAS.netspace.net.au) |
| 06:15.22 | JT | rofl, syntehsizers farting |
| 06:15.30 | JT | do they make for bad wakeup calls? |
| 06:15.44 | Strom_M | JT: that polycom ringer just will not wake me up |
| 06:16.08 | JT | maybe you need a force transducer under the matress |
| 06:16.22 | Strom_M | heh |
| 06:21.50 | *** join/#asterisk [B]abyCool16 (n=zeynoOo_@server.ivinskis.kursenai.lm.lt) |
| 06:22.37 | *** join/#asterisk [B]abyCool16 (n=_kardele@server.ivinskis.kursenai.lm.lt) |
| 06:25.12 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
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| 06:46.55 | *** part/#asterisk Telamon (i=telamon@blk-222-23-213.eastlink.ca) |
| 06:49.30 | *** join/#asterisk xain (n=aadilism@2-237-154-202.wol.net.pk) |
| 06:50.28 | xain | hi |
| 06:53.01 | Strom_M | instant catsex |
| 07:01.26 | *** join/#asterisk mitcheloc (n=mitchelo@titaniumsoft.net) |
| 07:03.05 | *** join/#asterisk dasenjo (n=dasenjo@63.245.86.215) |
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| 07:29.15 | *** join/#asterisk TonyM (n=TonyM@softins.claranet.co.uk) |
| 07:48.03 | *** join/#asterisk af_ (n=af@ip-131-134.sn2.eutelia.it) |
| 07:49.53 | Un1x | 4 NO ONE HERE JOIN THE ASTERISK MAILING LIST, THE EMAILS ADDRESS ARE NOT HMM WHAT U CALL IT BLOCKED IN SORT OF WAY THERE POSTED ON OTHER RELAY SITES THUS YOU GET ALOT OF SPAM FROM PEOPLE HARVESTING FROM GOOGLE. |
| 07:51.42 | hads | quit your red crap. |
| 07:51.54 | hads | and your yelling. |
| 07:51.58 | sevard | thanks. |
| 07:52.12 | sevard | you should be banned for using that sort of text in any channel. |
| 07:53.47 | *** join/#asterisk _omer (i=_omer@202.38.55.125) |
| 07:53.51 | *** part/#asterisk _omer (i=_omer@202.38.55.125) |
| 07:54.28 | *** join/#asterisk DarkWater13 (n=erestu@212.183.201.175) |
| 07:54.57 | DarkWater13 | Hi, someone can help me, with a small question? ( i think it´s easy for you :D ) |
| 07:55.26 | Strom_M | the answer is lol=very |
| 07:55.40 | hads | 42 |
| 07:56.34 | DarkWater13 | thx Strom_M ... are you a good friend ... |
| 07:56.36 | DarkWater13 | omg |
| 07:56.44 | DarkWater13 | Please, it´s an important thing. |
| 07:56.58 | hads | So... ask the question then. |
| 07:57.13 | hads | No one wants to hear you ask about asking a question. |
| 07:57.13 | *** join/#asterisk pfn (n=pfnguyen@netblock-66-245-252-239.dslextreme.com) |
| 07:57.18 | Strom_M | perhaps if you just asked the question instead of writing a lengthy preamble, we'd be able to get on with answering it |
| 07:58.06 | DarkWater13 | ok, sorry. I only want to be educated |
| 07:58.16 | hads | You just have been :) |
| 07:59.00 | DarkWater13 | Good. I have now Asterisk installed on a server. I have 2 zapchannels with a Digium card, and 2 IPPHones. All runs fine, the inbound calls are redirectly to the phones each of zapchannel. |
| 07:59.14 | DarkWater13 | My problem is that i don´t know how to assign a zapchannel to each IPphone |
| 07:59.18 | DarkWater13 | exclusivelly |
| 07:59.27 | DarkWater13 | channel 1, only for ipphone 1 |
| 07:59.30 | DarkWater13 | and channel2, only for ipphone 2 |
| 07:59.36 | DarkWater13 | ( I have Asterisk@Home ) |
| 07:59.40 | hads | You would use a different context |
| 07:59.44 | Strom_M | ~trixbox |
| 07:59.50 | jbot | methinks trixbox is NOT supported here! People using it should join #trixbox or #freepbx (FreePBX is the new name of AMP) |
| 07:59.50 | hads | Ah. |
| 07:59.51 | DarkWater13 | yes, Trixbox, sorry |
| 08:00.04 | hads | Over that way -> |
| 08:00.25 | DarkWater13 | Ok , i understand |
| 08:00.33 | DarkWater13 | Thx for your time |
| 08:00.34 | DarkWater13 | :D |
| 08:00.39 | mitcheloc | have you tried asterisk now instead? |
| 08:01.24 | DarkWater13 | I know that asterisk isn´t equal that trixbox, but i think that you knows that. |
| 08:01.44 | DarkWater13 | I have read all extensions.conf, and I intuit someone .. but .. |
| 08:02.50 | hads | Don't bother trying to read the extensions.conf from trixbox, you won't get anywhere. |
| 08:03.48 | DarkWater13 | oh, and .. have you some idea for me ? Any information will be good |
| 08:04.02 | hads | Yes, we told you already. |
| 08:04.06 | hads | ~trixbox |
| 08:04.08 | jbot | somebody said trixbox was NOT supported here! People using it should join #trixbox or #freepbx (FreePBX is the new name of AMP) |
| 08:04.44 | DarkWater13 | yes, i have readed this, but in trixbox there are nobody :D |
| 08:04.54 | DarkWater13 | Also .. thx for your help. |
| 08:07.51 | *** join/#asterisk ShadowHntr (i=sentinel@wikipedia/Shadowhntr) |
| 08:11.26 | *** join/#asterisk Gunnar (n=gunnar@62.97.242.6) |
| 08:17.15 | *** join/#asterisk Miss-tURk[off] (n=_kiz_@88.224.163.47) |
| 08:26.12 | *** join/#asterisk angryuser (n=Miranda@i03v-213-44-169-43.d4.club-internet.fr) |
| 08:27.13 | angryuser | sorry for offtopic, has anyone here managed to get to work netgear vpn switch with openvpn? |
| 08:28.29 | hads | try #openvpn |
| 08:30.07 | angryuser | thx |
| 08:30.24 | evilbuny | angryuser, wiki.cacert.org |
| 08:30.29 | evilbuny | search for openwrt |
| 08:37.49 | *** join/#asterisk _omer (i=_omer@202.38.55.125) |
| 08:37.50 | _omer | hi |
| 08:38.01 | _omer | anyone know how to delete an IP Alias ? |
| 08:38.42 | *** join/#asterisk kumamoto (n=kumamoto@71-11-200-124.dhcp.ftwo.tx.charter.com) |
| 08:39.06 | hads | Asterisk? |
| 08:39.27 | _omer | no linux...and didnt get answer in ##linux |
| 08:39.57 | *** join/#asterisk vgster (n=vgster@cpc2-ledn1-0-0-cust944.leed.cable.ntl.com) |
| 08:40.38 | hads | So you came here? Tey man ip |
| 08:40.46 | hads | s/Tey/Try/ |
| 08:41.32 | _omer | ifconfig eth0:0 IP_ADDRESS to create an alias |
| 08:41.36 | _omer | but how to delete it |
| 08:43.00 | *** join/#asterisk ghenry (n=ghenry@mail.suretecsystems.com) |
| 08:44.27 | *** join/#asterisk KermitTheFragger (n=ktf@118-197.bbned.dsl.internl.net) |
| 08:50.07 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 08:50.45 | *** join/#asterisk dlynes_laptop (n=dlynes@S0106001346f7843f.vc.shawcable.net) |
| 08:51.25 | *** join/#asterisk Oh-Ya (i=Oh-Ya@ip66-104-51-162.z51-104-66.customer.algx.net) |
| 08:51.40 | *** join/#asterisk Maroder (n=drago@fanatici.net) |
| 08:51.40 | Oh-Ya | can some one tell me which codec got better voice quailty g723 or Ilbc |
| 08:53.07 | *** join/#asterisk badcfe (n=cso@LNeuilly-152-22-86-193.w193-251.abo.wanadoo.fr) |
| 08:53.51 | *** join/#asterisk DavitFrom818 (n=DavitFro@cpe-76-173-56-41.socal.res.rr.com) |
| 08:53.53 | DavitFrom818 | hi |
| 08:54.05 | DavitFrom818 | is there any softphone for windows that supports LPC10 codec? |
| 08:56.51 | *** join/#asterisk moon06 (n=michael@cim06-1-82-228-240-97.fbx.proxad.net) |
| 08:58.56 | dlynes_laptop | Oh-Ya: probably ilbc, but it's got a huge demand on the processor |
| 08:59.16 | dlynes_laptop | DavitFrom818: you want to sound like a robot? |
| 08:59.22 | DavitFrom818 | yes |
| 08:59.26 | *** join/#asterisk frogzoo (n=frogzoo@202.155.165.25) |
| 08:59.29 | DavitFrom818 | its allow=speex? |
| 08:59.40 | dlynes_laptop | DavitFrom818: no...lpc10 is different from speex |
| 08:59.55 | DavitFrom818 | is speex better? |
| 08:59.58 | dlynes_laptop | DavitFrom818: Why not take ulaw from the softphone, and convert it to lpc10 inline, then? |
| 09:00.04 | dlynes_laptop | DavitFrom818: yes |
| 09:00.16 | dlynes_laptop | DavitFrom818: but not for robot sounding codecs |
| 09:00.20 | DavitFrom818 | ok here is my problem |
| 09:00.24 | dlynes_laptop | DavitFrom818: for human sounding codecs |
| 09:00.34 | DavitFrom818 | my user gets charged per meg |
| 09:00.45 | DavitFrom818 | i was hoping i can get 5 minutes of talk time for 1 meg |
| 09:00.49 | *** join/#asterisk jm|work (n=jamiem@dilbert.jamiem.com) |
| 09:00.49 | DavitFrom818 | is that possible? |
| 09:00.50 | Maroder | hi, i have problem with ata grandstream device, when user make a call transfer the device just die and only way is turn off |
| 09:00.59 | dlynes_laptop | DavitFrom818: why not go with g729? |
| 09:01.02 | Maroder | anybody with same problem ? |
| 09:01.03 | DavitFrom818 | i am |
| 09:01.12 | DavitFrom818 | i get 2 1/2 minutes for 1 meg |
| 09:01.22 | DavitFrom818 | i was hoping i can get 5 or more |
| 09:01.32 | dlynes_laptop | DavitFrom818: one second |
| 09:02.49 | monsted | mmm, g711ulaw :) |
| 09:03.18 | DavitFrom818 | lol |
| 09:03.31 | DavitFrom818 | yeah that would give me about half a minute for 3 megs |
| 09:03.32 | DavitFrom818 | lol |
| 09:03.33 | dlynes_laptop | DavitFrom818: g729 is a lower bandwidth codec than ilbc |
| 09:03.42 | *** join/#asterisk badcfe (n=cso@LNeuilly-152-22-86-193.w193-251.abo.wanadoo.fr) |
| 09:03.45 | DavitFrom818 | so should i try speex? |
| 09:03.55 | monsted | we want our users to use as much bandwidth as possible so they'll buy fatter pipes |
| 09:04.09 | dlynes_laptop | DavitFrom818: erm...nvm....ilbc is slightly less bandwidth than g729 |
| 09:04.24 | monsted | speex is quite configurable, but sounds like crap |
| 09:04.42 | hads | monsted: That's a rash generalisation. |
| 09:05.25 | monsted | yes, it is |
| 09:05.27 | monsted | :) |
| 09:05.28 | Oh-Ya | is asterisk come with ilbc codec |
| 09:05.35 | Oh-Ya | or i have to install it seprately |
| 09:05.36 | *** join/#asterisk nortex_work (n=breeves@snapper.titanspecialties.com) |
| 09:05.49 | DavitFrom818 | i installed it |
| 09:05.52 | hads | :) |
| 09:05.58 | DavitFrom818 | ok guys |
| 09:06.02 | DavitFrom818 | im going to hit the sack |
| 09:06.08 | DavitFrom818 | good night thnx for everything |
| 09:06.11 | monsted | don't do that, she might press charges |
| 09:06.15 | Maroder | hei |
| 09:19.03 | Maroder | can somebody help me |
| 09:20.27 | *** join/#asterisk darkskiez (n=mbryars@195-11-205-216.suip.mezzonet.net) |
| 09:22.49 | dlynes_laptop | Maroder: never run into that problem, personally...but then again, I use sipuras, not grandstreams |
| 09:24.34 | Maroder | dlynes_laptop ahamz |
| 09:24.45 | dlynes_laptop | ahamz? |
| 09:25.03 | *** join/#asterisk inspired (n=mikael@85.221.7.59) |
| 09:25.27 | hads | Clearing your throat? |
| 09:25.52 | JT | Maroder: ? |
| 09:26.58 | hads | Wow, an ad just reminded me that's it's only 6 days till christmas. |
| 09:27.10 | Maroder | baad |
| 09:27.12 | Maroder | :)) |
| 09:27.48 | Maroder | JT <Maroder> hi, i have problem with ata grandstream device, when user make a call transfer the device just die and only way |
| 09:27.48 | Maroder | <PROTECTED> |
| 09:27.51 | EmleyMoor | I can't detect distinctive rings if caller ID detection is turned on, even with the patch installed and enabled - I'm on BT in the UK - can anyone advise what I might be able to do, if anything? |
| 09:28.19 | JT | get an isdn line ;) |
| 09:28.21 | EmleyMoor | (all ring cadences appear as 0,0,0) |
| 09:29.06 | EmleyMoor | Without getting an ISDN line |
| 09:29.16 | JT | :( |
| 09:33.40 | EmleyMoor | Are there any workable FoIP options (for *?) yet? |
| 09:34.04 | JT | not as far as i'm aware |
| 09:34.04 | Maroder | JT are you work with grandstream ata |
| 09:34.07 | EmleyMoor | (I still need dring detection for another purpose) |
| 09:34.08 | Maroder | ? |
| 09:34.27 | JT | the only solutions i'm aware of aren't with asterisk, EmleyMoor |
| 09:34.33 | JT | Maroder: no |
| 09:36.54 | *** join/#asterisk RoyK (n=roy@80.239.107.70) |
| 09:37.19 | *** join/#asterisk FreePBX8734 (n=FreePBX8@203-206-208-253.perm.iinet.net.au) |
| 09:38.05 | *** join/#asterisk stkn (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 09:38.41 | EmleyMoor | Anyone else got any clues on Caller ID and dring detection on BT? |
| 09:38.52 | FreePBX8734 | hello....I was wondering if anyone here has connected an Asterisk box to a PABX in the specific designed of having 2 cards, one connecting to in TE mode to a T1/E1 serice and another card T1/E1 card in NT Mode connecting to a PABX? |
| 09:40.01 | FreePBX8734 | or any information on doing so? |
| 09:41.02 | JT | well it's quite possible |
| 09:42.28 | *** join/#asterisk tr2x (n=alvar@147.87.255.14) |
| 09:42.34 | FreePBX8734 | sure...I know this ;-) But looking for cards that do it (from what I can tell Digium don't have an E1 card that talked in NT mode for connecting to the PABX) and also can't find a how-to |
| 09:42.45 | JT | they do |
| 09:43.02 | JT | the mode is set in software |
| 09:43.11 | JT | T1 or E1 is set by a jumper |
| 09:43.56 | FreePBX8734 | talking about NT mode and TE Mode...NT mode is for connnecting to a PABX and TE mode is for connecting to a T1/E1 ISDN service |
| 09:44.13 | FreePBX8734 | once need a card that supports NT mode, most only do TE Mode |
| 09:44.30 | JT | i know very well about the modes |
| 09:44.40 | FreePBX8734 | ok...sorry I don't other than what I've just said |
| 09:44.57 | JT | TE mode = pri_cpe, NT mode = pri_net |
| 09:45.30 | FreePBX8734 | for example of you look at Digium's new 4 Port BRI card is states that it does TE and NT mode....whereas on Digium's information on their E1/T1 cards it doesn't state this |
| 09:45.53 | JT | i think they do it, even if it's stated |
| 09:45.57 | JT | not stated |
| 09:46.54 | FreePBX8734 | ok...is there anyone that has done a how-to/example on the net using Asterisk connecting to a PABX? |
| 09:47.15 | FreePBX8734 | specifically in the design of being in front of the PABX instead of "tacked on to the side" |
| 09:47.38 | JT | there are some mediocre docs online |
| 09:47.43 | JT | i know, gateway configuration |
| 09:47.58 | JT | it's not hard once you have an understanding of asterisk |
| 09:48.26 | FreePBX8734 | well I am pretty ok with Asterisk...but never done this design |
| 09:49.11 | FreePBX8734 | got a number of boxes out there with Digium card in them, using FXS devices and even tacked on to the side of a PABX...but never done it in front of a PABX... |
| 09:49.13 | JT | you basically just have two pri spans |
| 09:49.26 | JT | calls go between them as you desire, some calls can go via voip if you wish |
| 09:50.05 | FreePBX8734 | ok...all sounds simple enough...but was looking forward to reading someone else's how-to first ;-) |
| 09:50.24 | JT | don't use freepbx though :P |
| 09:50.51 | FreePBX8734 | how come? FreePBX using handles most things ok? |
| 09:51.16 | *** join/#asterisk grEvenX (n=even@pc107-130.ktv.no) |
| 09:51.32 | JT | not great for business |
| 09:51.38 | JT | hard to manage in the long term |
| 09:51.45 | JT | it makes horrible dialplans |
| 09:51.46 | *** join/#asterisk xnon (n=xnon@200.8.5.123) |
| 09:52.53 | FreePBX8734 | all true...but quick and easy for a simple small business that is really only interested in adding in some SIP phones and keeping their old PABX all still using their ISDN for outbound...not real routing..all calls go via the ISDN |
| 09:54.11 | JT | you won't find much support for it either |
| 09:55.15 | FreePBX8734 | actually...quite honestly I've found the other way around...massive and really good forums for FreePBX etc. Even Rob helps out a lot himself...not much on plain Asterisk and you have to dig dig dig dig etc. |
| 09:55.57 | inspired | EmleyMoor, you might have luck with OpenPBX.org for fax |
| 09:59.07 | *** join/#asterisk lorinc (n=ang@caracas-4427.adsl.interware.hu) |
| 10:00.01 | JT | FreePBX8734: shrug, up to you, the experienced people here will all recommend against doing PRIs worth of business (or any business) on ast |
| 10:00.54 | FreePBX8734 | sorry JT, you recommand against using PRIs with Asterisk? Not sure what you mean |
| 10:00.55 | FreePBX8734 | ? |
| 10:01.24 | JT | freepbx i mean |
| 10:01.55 | JT | if you're experienced like you say you are, you would've moved on from freepbx |
| 10:03.59 | FreePBX8734 | I used FreePBX for limiting end users in companies to manage their own PBX, like adding an extention easily etc....without it, or similar products, the cost of owning a soft PBX for companies is reduced over a typical plain old PABX |
| 10:04.09 | FreePBX8734 | also....it's the way Digium is going itself anyhow |
| 10:04.20 | FreePBX8734 | so, live with it JT ;-) |
| 10:05.04 | FreePBX8734 | not everyone wishes to waste time and be hardcore...not everyone needs or wants carrier grade Asterisks |
| 10:06.25 | JT | asterisk doesn't have an S on the end |
| 10:06.30 | JT | and is not really carrier grade |
| 10:06.52 | JT | and end users should not be configuring more than their name on their phone and voicemail |
| 10:07.11 | JT | pabxs are beyond the understanding level of most users |
| 10:07.42 | JT | the cost of ownership should reduced due to being able to do lots more for less money |
| 10:08.01 | JT | not because you let noobs screw up your pabx |
| 10:08.44 | sweeper | but then you can charge them $$$ to fix it >.> |
| 10:08.54 | JT | indeed |
| 10:10.03 | JT | digium is doing a gui because there obviously is a market, newbies/lazy people, and because freepbx makes such horrible dialplans |
| 10:10.14 | sweeper | :D |
| 10:10.21 | *** join/#asterisk Aboulafia (n=adlp@shm67-4-82-242-214-214.fbx.proxad.net) |
| 10:11.26 | *** join/#asterisk amir_ (n=amir@80.238.134.134) |
| 10:14.29 | EmleyMoor | Is there a general * support mailing list? |
| 10:14.48 | JT | asterisk-users |
| 10:15.19 | Oh-Ya | JT so you are saying Asterisk gui will be better then Frepbx |
| 10:15.33 | EmleyMoor | OK - will look into joining - maybe someone on there will have an idea for my dring problem |
| 10:15.36 | JT | it may be one day |
| 10:15.49 | JT | but i also say you should avoid guis to build your pbx |
| 10:16.14 | *** join/#asterisk tr2x_ (n=alvar@80-218-185-55.dclient.hispeed.ch) |
| 10:16.17 | Oh-Ya | well I think beigners like me its good to have gui once you know the hole system then you can paly around with cli |
| 10:16.34 | Oh-Ya | play* |
| 10:16.45 | JT | the risk there is that guis can teach you bad habits though |
| 10:17.06 | Oh-Ya | its heard when you are not linux sabby ... to use cli |
| 10:17.34 | JT | well you need to learn *nix of course |
| 10:18.45 | *** join/#asterisk isladelobos (n=user@43.Red-83-38-108.dynamicIP.rima-tde.net) |
| 10:19.25 | *** join/#asterisk CleanerX (n=nix@p54A395AA.dip0.t-ipconnect.de) |
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| 10:22.06 | *** join/#asterisk rcsw (n=richard@mail.shout-telecoms.com) |
| 10:47.55 | *** join/#asterisk dpenev (n=Miranda@sparnex.tea.bg) |
| 10:50.32 | *** join/#asterisk MGSsancho (i=howdy@adsl-67-122-137-49.dsl.pltn13.pacbell.net) |
| 10:52.14 | MGSsancho | ok at work im going to setp asterisk. i want to do this, T1 -> cisco 1700 then have data out ts ethernet port, and have another T1WIC -> an asterisk box |
| 10:52.19 | MGSsancho | is shit possible |
| 10:52.26 | MGSsancho | *this |
| 10:53.31 | MGSsancho | i would like for my internets to go through the 1700. and see if the 1700 can split the channels so the voice goes to an asterisk box with a digium t1 card |
| 10:53.51 | MGSsancho | and i have 1 more question, can i put a T1 card in a PCI-X slot? |
| 10:54.39 | Aboulafia | hi everyone.... |
| 10:55.09 | MGSsancho | morning |
| 10:56.26 | Aboulafia | i would like to offer to myself (for chrismas) a book :) |
| 10:56.37 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 10:56.52 | Aboulafia | i was looking for the oreilly, asterisk, the future of voip... what do you think about it ? |
| 10:57.31 | MGSsancho | havent read the oreilly one et =/ |
| 10:57.34 | MGSsancho | 8yet |
| 10:58.27 | JT | Aboulafia: future of telephony you mean |
| 10:58.37 | Aboulafia | JT: yes ;] |
| 10:58.37 | JT | Aboulafia: it's generally considered the yardstick of asterisk books |
| 10:58.39 | JT | the standard |
| 10:58.46 | FreePBX8734 | JT, you sound like a LINUX person that doesn't agree with GUI's for desktops |
| 10:58.47 | JT | you can read it for free first if you wish |
| 10:59.07 | JT | ~thebook |
| 10:59.08 | jbot | extra, extra, read all about it, thebook is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 10:59.18 | JT | FreePBX8734: GUIs are good for desktops and users |
| 10:59.23 | JT | a PBX is NOT a destop item |
| 10:59.31 | JT | desktop |
| 10:59.33 | FreePBX8734 | true... |
| 10:59.45 | [hC] | hmm .... uh -oh's.... my remote polycom upgrade seems to have requested the boot rom, wrote the boot log, then stopped. |
| 10:59.52 | FreePBX8734 | but...I've still found that there is nothing I cannot do with FreePBX as per doing it from conf files... |
| 10:59.54 | Aboulafia | JT: ok, but I love paper support :) |
| 11:00.10 | JT | yeah paper is easier to read in a lot of ways |
| 11:00.16 | Aboulafia | JT: but i will have a look about it befoire bying |
| 11:00.41 | *** join/#asterisk ghenry (n=ghenry@mail.suretecsystems.com) |
| 11:01.07 | JT | MGSsancho: if the 1700 has 2 T1 interfaces and can act as an add-drop interface, then yes it can be done |
| 11:01.18 | JT | if you have voice and Internet data delivered over the same T1 |
| 11:01.30 | JT | add-drop multiplexer i meant |
| 11:01.49 | MGSsancho | ahh cool. o i just need to but another T1 WIC then |
| 11:01.52 | MGSsancho | *so |
| 11:01.56 | MGSsancho | *buy |
| 11:02.16 | JT | i would check the 1700 can do it |
| 11:02.36 | JT | does your Internet and TDM voice get delivered over the same T1? |
| 11:02.49 | MGSsancho | there are also fxs and fxo cards for the 1700 <__< |
| 11:03.02 | JT | heh |
| 11:03.05 | MGSsancho | it will when i call at&t |
| 11:03.12 | JT | hrm |
| 11:03.30 | JT | i assume there is some financial advantage to doing so, otherwise it wouldn't be worth it |
| 11:03.37 | MGSsancho | boss wants to get of of our 2pots and 1 fax line |
| 11:03.49 | MGSsancho | it would save $157 a month |
| 11:03.56 | JT | umm, will it be proper TDM voice |
| 11:04.01 | JT | or delivered over IP |
| 11:04.34 | MGSsancho | im just curious if we would need to change our buisness phone numbers |
| 11:05.01 | *** join/#asterisk dpenev (n=Miranda@sparnex.tea.bg) |
| 11:05.22 | *** join/#asterisk Ahrimanes (n=ma@81.7.159.2) |
| 11:05.26 | MGSsancho | if its TDM, im assuming we get slower internets and keep the numbers |
| 11:05.27 | Aboulafia | just to say, because, I'm so proud about my last stupid thing that i do : @home (because I've to internet access) I've 2 asterisk, who are runing in the same domU, with one allowed to use my TDM400P :) |
| 11:05.30 | Ahrimanes | hey |
| 11:05.41 | Aboulafia | (and it's seem to wok fine :)) ) |
| 11:05.46 | MGSsancho | lol |
| 11:05.48 | Ahrimanes | anyone using cisco's "piggyback" nat feature with asterisk? |
| 11:06.07 | JT | Aboulafia: domU? |
| 11:06.26 | *** join/#asterisk FWP (n=FWP@unaffiliated/fwp) |
| 11:06.46 | Aboulafia | JT: yep in a Xen virtual machine :) |
| 11:06.54 | JT | ok |
| 11:06.56 | JT | what is domU? |
| 11:07.03 | MGSsancho | JT,if i had it dilivered over IP, i would need to get a third party VoIP and get new phonenumber correct? |
| 11:07.21 | JT | i have no idea what number portability islike in your area |
| 11:07.23 | JT | but possibly |
| 11:07.39 | Aboulafia | in Xen you have the Dom0 who is the hypervisor (who give physical ressouce to the virtual machine) and all the virtual machine called DomU |
| 11:07.49 | JT | ah ok |
| 11:10.59 | *** join/#asterisk r0d3nt (n=RatMan@foster.stonedcoder.org) |
| 11:11.18 | Aboulafia | JT: so it's less expensivive than buying several computer.... |
| 11:11.30 | JT | yes |
| 11:14.59 | Aboulafia | and I prepare two asterisk, because I've two internet access, and with SIP.... but my problem, now, is to make a correct dialplan to allowed everyone to register onto each asterisk, with the same dialplan.... |
| 11:17.25 | *** join/#asterisk RoyK (n=roy@80.239.107.70) |
| 11:26.36 | *** join/#asterisk shtoom (n=godson@202.149.38.14) |
| 11:27.28 | shtoom | Hi can we use asterisk realtime static and realtime dynamic simultaneously? |
| 11:27.46 | *** join/#asterisk _omer (n=omer@203.128.20.84) |
| 11:28.10 | RoyK | wtf is realtime static? |
| 11:28.36 | MGSsancho | i think hes missing 2 camas |
| 11:29.25 | shtoom | RoyK: realtime static is storing *.conf files is in database |
| 11:30.12 | *** join/#asterisk FreePBX3608 (n=FreePBX3@211.26.222.178) |
| 11:30.25 | *** join/#asterisk CrashHD (i=CrashHD@c-67-182-167-222.hsd1.ca.comcast.net) |
| 11:30.52 | *** join/#asterisk jmesquita (n=jmesquit@201.7.117.114) |
| 11:30.52 | shtoom | and dynamic is storing sip peers or user in database not the exact .conf files, that is what i came to know after hours of googling |
| 11:31.27 | *** join/#asterisk Faithful1 (n=Faithful@ns.linuxterminal.com) |
| 11:31.57 | shtoom | http://www.voip-info.org/wiki/view/Asterisk+RealTime |
| 11:32.16 | shtoom | http://www.asteriskguru.com/tutorials/realtime_pgsql.html |
| 11:33.49 | shtoom | the later link says so static realtime and dynamic realtime but to avoid confusion we may simply call it as database static configuration and database realtime configuration for asterisk |
| 11:34.51 | shtoom | Hi can we use asterisk static and realtime database configurations simultaneously? |
| 11:35.33 | shtoom | Are there any known problems by doing so? |
| 11:37.36 | *** join/#asterisk apardo (n=apardo@87.217.147.173) |
| 11:41.34 | *** join/#asterisk kippi (n=pssedoff@untrust-gct.equinoxit.net) |
| 11:41.36 | kippi | hey |
| 11:43.20 | kippi | what is the best way of linking to asterisk servers together? SIP? |
| 11:43.58 | kaldemar | IAX2 |
| 11:46.05 | kaldemar | http://www.voip-info.org/wiki/view/Asterisk+-+dual+servers |
| 11:47.09 | shtoom | Hi can we use asterisk static and realtime database configurations simultaneously? Are there any known problems by doing so? |
| 11:50.14 | *** join/#asterisk qdk (n=qdk@213.150.62.28) |
| 11:52.24 | Ahrimanes | shtoom, is it realtime for some config files and static for others or a mix that you want? |
| 11:57.08 | *** join/#asterisk ghenry (n=ghenry@mail.suretecsystems.com) |
| 11:57.13 | shtoom | Ahrimanes: thanx for the reply yes,i want a mix, because there is no way to use "include" statement (in entensions table ) as well as register a trunk in realtime sip table |
| 11:58.06 | RoyK | shtoom: how can "static" == "realtime"?? "realtime" implies dynamic operation, as in 'config is updated in realtime' |
| 11:59.15 | Ahrimanes | shtoom, i do believe that if you use realtime sip peers, you can still do register's in sip.conf |
| 11:59.35 | shtoom | it is explained here http://www.voip-info.org/wiki/view/Asterisk+RealTime |
| 11:59.45 | *** join/#asterisk gcorbaz_ (n=gcorbaz@250-60.2-85.cust.bluewin.ch) |
| 12:00.57 | *** join/#asterisk gcorbaz_ (n=gcorbaz@250-60.2-85.cust.bluewin.ch) |
| 12:01.01 | kippi | kaldemar: many thanks |
| 12:01.57 | shtoom | Ahrimanes : ya i can do that but i am writing a web based interface for asterisk where i dont want to parse a text file, want to do every thing in database |
| 12:03.13 | Ahrimanes | shtoom, i doubt you can do #include like things in the database.. but do test.. |
| 12:05.20 | *** join/#asterisk stephane (n=stephane@gw.sortilege.net) |
| 12:05.50 | shtoom | Ahrimanes: in realtime database you can't use include, just scroll down to the last part of this page and see that we can use include statement there as there is no column to hold that ! |
| 12:05.56 | shtoom | http://www.asterisk.org/doxygen/trunk/AstARA.html |
| 12:10.20 | *** join/#asterisk tr2x (n=alvar@80-218-185-55.dclient.hispeed.ch) |
| 12:11.54 | shtoom | Hi can we use asterisk static and realtime database configurations simultaneously? Are there any known problems by doing so? |
| 12:14.44 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 12:17.21 | *** join/#asterisk apardo_ (n=apardo@87.217.146.18) |
| 12:17.32 | *** join/#asterisk VibroMax (n=phisto@83.229.70.154) |
| 12:21.35 | Ahrimanes | shtoom, i think it's safe to say by now, that noone here seems to have an answer.. test it, and write a howto :) |
| 12:26.03 | *** join/#asterisk frogzoo (n=frogzoo@202.155.165.25) |
| 12:26.58 | kippi | has anyone got anyideas why when using exten => 478102,1,Dial(sip/1130&sip/1134&sip/1131&sip/1133&sip/1135&sip/1136&sip/1138&sip/1141&sip/1139) why they dnt keep on ringing, my moblie keeps on ringing and asterisk dosn't hang up but the phones stop anyideas? |
| 12:28.22 | shtoom | Ahrimanes: Oh this is my first query on this irc, ya i'll test it if noone is going to answer , anyways thanx for your help |
| 12:28.57 | *** join/#asterisk amir_ (n=amir@80.238.134.134) |
| 12:31.51 | hwt | kippi: try with e.g. a |300 at the end of the appdata. |
| 12:32.05 | hwt | kippi: to force it to dial for 300 seconds. |
| 12:32.20 | *** join/#asterisk voiceintegrity (n=bob@dsl-202-173-131-82.nsw.westnet.com.au) |
| 12:33.31 | *** join/#asterisk DrCron (n=rszasz@c-24-7-33-87.hsd1.ca.comcast.net) |
| 12:36.44 | kippi | hwt: is there a default setting for one min somewhere? |
| 12:38.42 | kippi | hmm that stoped to |
| 12:41.44 | *** join/#asterisk stkn (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 12:53.12 | hwt | kippi: sorry, i have problems understanding you. |
| 12:54.57 | kippi | There must be a default setting somewhere that tells the phones to keep on ringing but not hangup the call |
| 12:55.37 | DrCron | ah, you mean the Dial timeout? |
| 12:55.49 | hwt | kippi: try with |300 after the Dial. like this: Dial(SIP/foo&SIP/bar&...|300) |
| 12:59.54 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 13:00.10 | *** join/#asterisk anthonyl_ (n=anthonyl@adsl-230-22-85.hsv.bellsouth.net) |
| 13:00.35 | *** join/#asterisk juanjoc (n=juanjoc@248-32-235-201.fibertel.com.ar) |
| 13:04.56 | *** join/#asterisk sac|h0p|werk|afk (n=h0p@69.10.147.2) |
| 13:07.07 | *** join/#asterisk stuq (n=Stuart@user-12lcqia.cable.mindspring.com) |
| 13:07.39 | xain | where CDR save in LINUX ??? |
| 13:07.59 | *** join/#asterisk reber (n=reber@cl-157.dub-01.ie.sixxs.net) |
| 13:09.25 | kaldemar | xain: if you're using asterisk: /var/log/asterisk/cdr-csv/Master.csv |
| 13:09.31 | jmesquita | Is anyone using dynamic queue members here? |
| 13:10.01 | *** join/#asterisk tm (n=tm@tm.muc.de) |
| 13:10.04 | tm | hi-de-ho |
| 13:10.07 | jmesquita | Or has anyone used ? |
| 13:11.25 | DrCron | so, asterisk realy wants a hardware timing source it seems. What is the cheapest available one |
| 13:12.00 | xain | <kaldemar> yeah i m using .. asterisk ... and want to delete some record from back hand ... don't want to delete from DB .. can i delete from back hand ... |
| 13:12.20 | tzafrir | DrCron, x100p |
| 13:13.12 | DrCron | nice, can it recieve faxes? |
| 13:13.13 | tm | kann hier wer deutsch? ;) |
| 13:13.18 | DrCron | not needed, just wondering |
| 13:13.57 | hwt | can anyone recommend a mysql-cdr web-frontend? |
| 13:14.05 | hwt | open source/FOSS. |
| 13:14.08 | *** join/#asterisk VibroMax (n=phisto@83.229.70.154) |
| 13:14.20 | *** join/#asterisk bjohnson_ (n=bjohnson@jecinc.tor.istop.com) |
| 13:14.56 | tm | eliXier: hallo |
| 13:22.54 | *** join/#asterisk Zand3r (n=Zand3r@spc1-bolt7-0-0-cust660.bagu.broadband.ntl.com) |
| 13:26.30 | *** join/#asterisk cjlowe (n=cjlowe@c220-237-173-67.lowrp1.vic.optusnet.com.au) |
| 13:28.32 | cjlowe | the "goodbye" recording is so darn enthusiastic... |
| 13:28.37 | *** join/#asterisk jmesquita_ (n=jmesquit@201.7.117.114) |
| 13:29.27 | *** join/#asterisk p0g0 (n=pogo@madwifi/support/p0g0) |
| 13:37.51 | *** join/#asterisk jart` (n=user@ool-44c05258.dyn.optonline.net) |
| 13:41.34 | robl^ | need coffee!!! morning everyone |
| 13:42.46 | *** join/#asterisk sigmounte (n=sigmount@www.sighq.net) |
| 13:44.17 | cjlowe | rofl, morning robl^ |
| 13:45.10 | tzanger | you're telling me |
| 13:45.24 | tzanger | is it 5:45 or 6:45 in British Columbia right now? |
| 13:46.06 | *** join/#asterisk vooduhal (n=vooduhal@tc-proxy2.catt.com) |
| 13:46.26 | vooduhal | Would there be a reason why pressing "*" on a zap channel would cause problems? |
| 13:46.48 | tzanger | I think there's a dial option which lets you hang up with * |
| 13:47.31 | *** join/#asterisk iCEBrkr (i=icebrkr@69.9.167.70) |
| 13:48.14 | robl^ | how is everyone this morning? |
| 13:49.12 | cjlowe | robl^, pretty darn good |
| 13:49.17 | cjlowe | robl^, you? |
| 13:50.26 | tm | anybody here? |
| 13:50.32 | heartones | doing well here u? |
| 13:50.39 | robl^ | cjlowe: I have coffee on my desk. I will be ok, soon |
| 13:50.40 | tm | :-) |
| 13:50.40 | heartones | sure |
| 13:50.46 | tm | heartones: i need help |
| 13:51.10 | cjlowe | robl^, nice =) |
| 13:51.10 | tm | heartones: can u help me with misdn? ;) |
| 13:51.10 | heartones | any one has configured a sipura 3000 here |
| 13:51.10 | cjlowe | robl^, I know the feeling... just hang in for the coffee :) |
| 13:51.32 | cjlowe | hey, quick question... asterisk isn't beeping when we one-touch-record calls, where is one-touch defined? we just want a confirm beep. |
| 13:51.45 | cjlowe | heartones, is that the one with the port in as well as out? |
| 13:51.48 | *** join/#asterisk PupenoR (n=pupeno@2002:c87b:b75a:1:240:f4ff:fe6b:7650) |
| 13:52.13 | heartones | cjlowe : the one with FXO/FXS/RJ |
| 13:52.51 | heartones | I just wanted to know if I have to load a specific driver on zaptel.conf |
| 13:53.49 | heartones | would asterisk recognize it automatically |
| 13:53.58 | cjlowe | heartones, doesn't it act as a SIP device? |
| 13:54.16 | heartones | yes it does |
| 13:54.38 | heartones | I wanted to hook it up to the pstn as well |
| 13:55.20 | cjlowe | heartones, ahh :) only ever used it as an extension, sorry... maybe you define it as a sip trunk, at a guess... but I wouldn't know |
| 13:55.43 | heartones | kwel start protocols and the span, etc.... do I have to set this manually for it |
| 13:56.16 | *** join/#asterisk tzafrir_ (n=tzafrir@62.90.10.53) |
| 13:56.20 | heartones | I haven't hooked it up yet :) |
| 13:58.05 | cjlowe | heartones, I honestly wouldn't know :) it was ages ago, and I just added it as an extension, so I pretty much just copied and pasted everything... |
| 13:58.37 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 13:58.58 | tm | can anybody speak german here? |
| 13:59.15 | heartones | so if I use it as an extention I won't be needing to lood any zaptel driver |
| 13:59.23 | heartones | tm : little |
| 13:59.31 | cjlowe | tm I can, but its been a while |
| 13:59.44 | tm | heartones / cjlowe ah schoen |
| 13:59.48 | cjlowe | heartones, as an extension, I don't think you can route calls through it |
| 13:59.59 | heartones | danke |
| 14:00.17 | tm | ich möchte meinen isdn asterisk server anwählen (it works..) und mich dann weiter verbinden mit einer bestimmten numme,r wie mache ich das? |
| 14:01.00 | cjlowe | I want to dial my asterisk server and then get further connected to a specific number, how do I do that? |
| 14:01.09 | cjlowe | my ISDN asterisk server* |
| 14:01.13 | nibbler_de | tm: du willst via isdn einen anruf annehmen und ihn wieder via isdn rausschicken an eine bestimmte nummer? |
| 14:01.16 | cjlowe | for those who don't understand :) |
| 14:01.29 | tm | nibbler_de: ah hi :-) |
| 14:01.44 | nibbler_de | mahlzeit ;) |
| 14:01.54 | tm | nibbler_de: genau. ich hab zum esten ein: exten => 37061215,1, Dial(misdn/1/08003301000) |
| 14:01.57 | tm | das tut. |
| 14:01.58 | tm | ich moechte abeR: |
| 14:02.09 | *** join/#asterisk EyeCue` (n=eyecue@220-253-130-38.QLD.netspace.net.au) |
| 14:02.18 | nibbler_de | .oO(warum nehmen alle grad diese nummer immer zum testen *gg*) |
| 14:02.26 | tm | nu moecht eich aber 37061215 anrufen, einen pin eingeben bsp. 1234, und dann ein OK hoeren.. wie auch immer und dann |
| 14:02.36 | tm | eine weitere nummer eingeben wohin er mich verbinden soll |
| 14:02.38 | cjlowe | nibbler_de, LOL |
| 14:02.53 | nibbler_de | tm: du willst dir mal disa ansehen ;) |
| 14:03.10 | nibbler_de | voip-info.org/wiki und nach disa gucken |
| 14:03.43 | tm | find hier kein disa http://voip-info.org/wiki/ |
| 14:03.51 | tm | nibbler_de: kann man das ned irgend wie realisieren? |
| 14:03.55 | nibbler_de | klar |
| 14:03.57 | tm | mit meinem normalen aufgesetzt asterisk? |
| 14:04.00 | nibbler_de | mit dem disa command |
| 14:04.18 | tm | ajo |
| 14:04.21 | tm | auch so mit pin und so? |
| 14:04.26 | nibbler_de | ich kann grad nicht cut&pasten, sonst wuert ich dir schnell den teil aus meiner config geben |
| 14:04.50 | tm | nibbler_de: ja klar... nibbler_de lad mal hoch |
| 14:04.54 | tm | nibbler_de: http://tm.muc.de/up/ |
| 14:05.04 | nibbler_de | ich lade dir sicher nicht meine config hoch |
| 14:05.32 | tm | nibbler_de: aeh. nein doch nicht die ganze. |
| 14:05.35 | tm | horn horn.. |
| 14:05.36 | tm | :-) |
| 14:06.24 | nibbler_de | man google ;) der tipp von wegen "disa" sollte dich eigentlich schon in die richtige richtung bringen |
| 14:06.53 | nibbler_de | www.voip-info.org/wiki/index.php?page=Asterisk+cmd+DISA |
| 14:07.03 | robl^ | Ich bin krank, aber ich trinke jetzt Kaffee. Ich verbessere bald. |
| 14:07.11 | nibbler_de | das example2 |
| 14:07.13 | tm | nibbler_de: iss disa schon mit drin? |
| 14:07.19 | tm | oder en modul? |
| 14:07.41 | nibbler_de | modul? aehm - ist halt ne application - und die ist standardmaessig freilich dabei |
| 14:07.45 | tm | aso. |
| 14:07.49 | tm | nibbler_de: iss ja gut! :-) |
| 14:08.10 | cjlowe | wie komisch... dies ist ja ##australia, doch sprechen wir alle auf deutsch |
| 14:08.52 | *** join/#asterisk engrxyz (n=gsdfgdgf@89.222.4.8) |
| 14:08.53 | nibbler_de | cjlowe: je parle francais aussi si tu veux changer la langue ;) |
| 14:09.05 | *** join/#asterisk MGSsancho (i=howdy@adsl-67-122-137-219.dsl.irvnca.pacbell.net) |
| 14:09.12 | engrxyz | anyone? |
| 14:09.22 | cjlowe | engrxyz, mmhmm? |
| 14:09.36 | cjlowe | bah, i'm such an idiot, this isn't ##australia, it's #asterisk |
| 14:09.55 | cjlowe | it's 1am in australia, so I claim tiredness in defense |
| 14:10.02 | tm | nibbler_de: was eigenartig ist das er immer scho abspielt obwohl ich onch ned verbunden bin? |
| 14:10.03 | engrxyz | need some help down here.. i am a totally newbie...how should i get a dialtone from my ip phone |
| 14:10.18 | tm | nibbler_de: also manchmal bekomm ich ein wort ned mit.. |
| 14:10.51 | nibbler_de | tm: wait(1) ist dein freund ;) |
| 14:10.55 | nibbler_de | mach ich hier auch so |
| 14:10.56 | robl^ | Ich werde nicht von den deutschen Sprechern gestört. Ich kann Deutsches gut verstehen, leider ich spreche es nicht gut. |
| 14:11.21 | nibbler_de | robl^: your german is... uhm... improvable. |
| 14:11.23 | cjlowe | robl^, ich habe das selbe Problem |
| 14:11.29 | engrxyz | i installed asterisk but i cannot get a dialtone when i attached my ip phone to it |
| 14:11.43 | *** join/#asterisk dsfr (i=dsfr@pdpc/sponsor/digium/dsfr) |
| 14:11.46 | cjlowe | engrxyz, what do you mean by "attach" your phone? |
| 14:11.52 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 14:11.58 | tm | nibbler_de: geil |
| 14:12.16 | robl^ | I only had 6 weeks of german. ;-) |
| 14:12.17 | djflux | anyone with a 1.4.0-beta4 and an Ekiga softphone want to verify something for me? DTMF within a call doesn't seem to work and I want to make sure it's not just me |
| 14:12.26 | *** join/#asterisk tdonahue-laptop (n=tdonahue@vonmail.vonworldwide.com) |
| 14:12.29 | tm | nibbler_de: darf ich nochmal? ;9 |
| 14:12.34 | nibbler_de | tm: was? |
| 14:12.47 | tm | nibbler_de: das mit dem passwor thab ich nun. um nun weiter zu verbinden geht das dann so? |
| 14:12.50 | tm | Read(Nummer,give-number-sound,20) |
| 14:12.53 | tm | Dial(CAPI/ISDN1:${Number} |
| 14:12.55 | tm | ode so aehnlich? |
| 14:13.16 | nibbler_de | aehm - nimm halt disa |
| 14:13.19 | nibbler_de | brb |
| 14:13.20 | *** join/#asterisk tdonahue_ (n=tdonahue@vonmail.vonworldwide.com) |
| 14:14.11 | *** join/#asterisk juanjoc (n=juanjoc@248-32-235-201.fibertel.com.ar) |
| 14:14.18 | engrxyz | i have a unix box with fresh asterisk installed on it. one of its interface is connected to a which in which my ip phone is connected as well. hints and tips on how to get the dial-tone? |
| 14:15.32 | tm | nibbler_de: wenn du wieder da bist schrei mal kurz ;) |
| 14:16.26 | cjlowe | engrxyz, would recommend you define an extension for the phone and set the details for it in the phone :) |
| 14:16.27 | *** join/#asterisk tdonahue-laptop (n=tdonahue@vonmail.vonworldwide.com) |
| 14:17.02 | engrxyz | cjlowe: this are my extensions.conf entries |
| 14:17.08 | engrxyz | general] |
| 14:17.08 | engrxyz | writeprotect=no |
| 14:17.08 | engrxyz | [hints] |
| 14:17.08 | engrxyz | exten => 14344,hint,SIP/sipura |
| 14:17.08 | engrxyz | [incoming-uri] |
| 14:17.08 | engrxyz | exten => sipura,1,Goto(internal,14344,1) |
| 14:17.19 | DrCron | ~pastebin |
| 14:17.21 | jbot | from memory, pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste |
| 14:17.24 | cjlowe | engrxyz, better not to paste stuff here :) |
| 14:17.31 | engrxyz | ok sorry |
| 14:18.08 | engrxyz | cjlowe: those are my entries |
| 14:18.33 | robl^ | pasting more then 2 lines is usually enoug to have you hung, skinned, boiled, and electrocuted. ;-) |
| 14:18.53 | engrxyz | on the sipura ip phone, i configure the first extention and its says registration fails |
| 14:19.03 | cjlowe | engrxyz, does it say why? |
| 14:19.35 | engrxyz | it just says that registration failed |
| 14:19.55 | cjlowe | engrxyz, what's the phone? |
| 14:20.09 | engrxyz | sipura ip phone |
| 14:20.25 | cjlowe | spa-841 or whatever it is? |
| 14:20.33 | engrxyz | spa-841 |
| 14:20.49 | cjlowe | engrxyz, have one of those on my desk :) |
| 14:21.04 | engrxyz | but this ip phone works with an edgebox here |
| 14:21.08 | tm | nibbler_de: hm ich muss nun was neues definieren oder in der ext*.conf? |
| 14:22.08 | cjlowe | engrxyz, first things first - can you get to the phones web configuration page? |
| 14:22.49 | engrxyz | cjlowe; yes |
| 14:26.22 | *** join/#asterisk badcfe (n=cso@LNeuilly-152-22-86-193.w193-251.abo.wanadoo.fr) |
| 14:26.28 | *** join/#asterisk _BOBWEEVER (n=chatzill@adsl-150-0-187.aby.bellsouth.net) |
| 14:26.37 | tm | aeh |
| 14:26.37 | tm | [mycontext] |
| 14:26.37 | tm | exten => s,1,Read(Nummer,give-number-sound,20) |
| 14:26.38 | tm | exten => s,2,Dial(misdn/1/${Number}) |
| 14:26.38 | tm | exten => s,3,Hangup |
| 14:26.48 | tm | or |
| 14:26.48 | tm | [mycontext] |
| 14:26.48 | tm | exten => 37061215,1,Read(Nummer,give-number-sound,20) |
| 14:26.48 | tm | exten => 37061215,2,Dial(misdn/1/${Number}) |
| 14:26.48 | tm | exten => 37061215,3,Hangup |
| 14:26.51 | tm | ? |
| 14:28.10 | *** join/#asterisk b11d (n=noway@234-200-29-134.hcc.mnscu.edu) |
| 14:28.27 | b11d | Qwell you bastard.. you could have warned me before banning me ;) |
| 14:28.50 | b11d | anyway |
| 14:28.52 | b11d | morning lads |
| 14:29.21 | engrxyz | tm? |
| 14:29.30 | tm | engrxyz: emm |
| 14:29.49 | tm | if u haven... |
| 14:29.50 | tm | have |
| 14:29.51 | tm | .... |
| 14:29.51 | tm | exten => 37061215,5,Authenticate(12345) |
| 14:29.51 | tm | exten => 37061215,6,DISA(no-password|mycontex |
| 14:29.56 | tm | and |
| 14:29.57 | tm | [mycontext] |
| 14:29.57 | tm | exten => Read(Nummer,give-number-sound,20) |
| 14:29.57 | tm | exten => Dial(misdn/1/${Number}) |
| 14:29.57 | tm | exten => Hangup |
| 14:30.06 | tm | then hang up my server after |
| 14:30.13 | tm | insert my code 12345 |
| 14:30.31 | engrxyz | tok |
| 14:30.32 | engrxyz | ok |
| 14:30.45 | tm | but i like a forward to my insert number...? |
| 14:30.54 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 14:31.02 | engrxyz | tm; just a typical thing |
| 14:31.10 | tm | engrxyz: hm. can u help me? |
| 14:31.30 | *** join/#asterisk ima (n=ang@caracas-3558.adsl.interware.hu) |
| 14:31.48 | engrxyz | tm; i have an asterisk in a unix box and i have an ip phone, any hints or tip for my ip phone to have a dialtone |
| 14:32.37 | tzafrir_ | engrxyz, depends on the ip phones. Some of them don't need to be configured to give a dialtone... |
| 14:33.38 | engrxyz | tzafrir_: i tried sipura-spa-841 in an edgebox and it works well. i tried it in asterisknow as well as in asterisk 1.2 and i don't have heard any dialtone at all |
| 14:34.15 | engrxyz | it says registration state failed |
| 14:35.16 | tzafrir_ | engrxyz, now is the time that you try to convince us that you configured things properly. Please post the relevant config bits, and relevant traces that demonstrate that you have configured things properly. |
| 14:35.27 | tzafrir_ | (to the best of your knowledge) |
| 14:35.54 | engrxyz | what do u want to know then |
| 14:36.36 | *** join/#asterisk ast_freak (n=jesse@h69-130-160-57.69-130.unk.tds.net) |
| 14:37.18 | tzafrir_ | never mind. do you have an entry for the device in sip.conf (a section where the type is user or friend)? |
| 14:37.28 | *** join/#asterisk swilliamson (i=swilliam@209.42.110.46) |
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| 14:38.18 | *** mode/#asterisk [+o anthm] by ChanServ |
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| 14:39.53 | *** join/#asterisk Dovid (n=Dovid@ool-43530a83.dyn.optonline.net) |
| 14:40.41 | shtoom | <PROTECTED> |
| 14:40.47 | *** join/#asterisk s1gny|wrk (n=s1gny@p54917662.dip.t-dialin.net) |
| 14:40.50 | Dovid | anyone know where I can get info on the asterisk database |
| 14:40.59 | Dovid | how i can clean it out, remove entries etc. ? |
| 14:41.17 | *** part/#asterisk s1gny|wrk (n=s1gny@p54917662.dip.t-dialin.net) |
| 14:41.17 | b11d | you mean astdb? |
| 14:43.05 | *** join/#asterisk ManxPower (n=manxpowe@dpc67142183150.direcpc.com) |
| 14:43.26 | b11d | bonjour ManxPower |
| 14:43.35 | *** join/#asterisk angryuser (n=Miranda@i03v-213-44-169-43.d4.club-internet.fr) |
| 14:43.36 | engrxyz | tzafrir_, i have entry in sip.conf |
| 14:45.47 | Dovid | boker toker tafrir |
| 14:45.51 | Dovid | tzafrir* |
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| 14:46.06 | tm | nibbler_de: hm |
| 14:46.11 | Dovid | shtoom: what do u mean bu that ? |
| 14:46.17 | Dovid | b11d: yes |
| 14:46.23 | Dovid | i did show applications astdb |
| 14:46.26 | Dovid | correct command ? |
| 14:46.38 | Dovid | been a while since i have played in asterisk - the old memory is getting to me |
| 14:47.00 | resistance | hi, how can i install FOP and the log feature into straight asterisk |
| 14:48.59 | shtoom | Dovid: http://www.voip-info.org/wiki/view/Asterisk+RealTime there two styles of configurations realtime and static i am askign can we use both? |
| 14:49.12 | shtoom | together at a time |
| 14:49.52 | *** join/#asterisk dasenjo (n=dasenjo@63.245.86.215) |
| 14:51.39 | tm | hm |
| 14:51.39 | tm | neuersamba*CLI> misdn reload |
| 14:51.39 | tm | Reloading mISDN Config |
| 14:51.39 | tm | Dec 19 15:38:22 WARNING[30192]: misdn_config.c:940 _build_port_config: misdn.conf: "ports=(null)" (section: TEports) invalid or out of range. Please edit your misdn.conf and then do a "misdn reload". |
| 14:51.43 | tm | neuersamba*CLI> |
| 14:51.52 | tm | [TEports] |
| 14:51.52 | tm | context=eingehend |
| 14:51.52 | tm | ports=2 |
| 14:51.55 | tm | why? |
| 14:52.03 | Dovid | shtoom: i never tried but dont think so |
| 14:52.10 | tm | can anybody help me? |
| 14:52.22 | Dovid | i know u can use the conf. files and real time at the same time |
| 14:52.55 | Dovid | tm: please use pb in the future |
| 14:52.55 | Dovid | ~pb |
| 14:52.59 | jbot | pb is probably a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste |
| 14:52.59 | Dovid | and if some one has an answer they will reply. |
| 14:53.00 | tm | Dovid: hm? |
| 14:53.00 | tm | what? |
| 14:54.39 | b11d | tm.. maybe because you cant set a NULL port? |
| 14:54.53 | *** join/#asterisk shinux__ (n=shinux@196.220.26.77) |
| 14:55.33 | b11d | ooohh.. i cant wait for 1.4 |
| 14:55.49 | tm | b11d: ports=1 works |
| 14:55.50 | tm | 2 not |
| 14:56.12 | b11d | <PROTECTED> |
| 14:57.01 | tm | i have |
| 14:57.01 | tm | Dec 19 15:43:52 WARNING[30642]: misdn_config.c:942 _build_port_config: misdn.conf: "ports=(null)" (section: TEports) invalid or out of range. Please edit your misdn.conf and then do a "misdn reload". |
| 14:57.07 | tm | ;) |
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| 15:00.25 | b11d | yeah |
| 15:00.30 | b11d | lets see that misdn.conf then |
| 15:00.36 | b11d | put it up on pastebin.ca |
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| 15:03.45 | *** part/#asterisk IMG-SD (n=IMG-SD@as2.imperialgroup.ca) |
| 15:04.34 | *** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
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| 15:13.08 | Dovid | trying to remove an entry from astdb and having a problem. can anyone help ? |
| 15:13.27 | Gankhuu | what equipment would I need if I had a T-1 with 6 voice channels and 18 data channels, to split out the voice and data? |
| 15:14.19 | Gankhuu | I need to put the audio into tdm cards and use the data for trunk |
| 15:14.31 | blitzrage | Dovid: you're best to paste the error into a pastebin and not ask for help -- if someone knows the answer, they will help |
| 15:14.32 | b11d | what do you need to remove from astdb? |
| 15:14.39 | b11d | its a bdb db iirc.. |
| 15:14.56 | Dovid | its a syntax error |
| 15:15.04 | *** join/#asterisk supjigatr (n=syslod@152.53.16.10) |
| 15:15.11 | Dovid | i dont know what it means by family and key |
| 15:15.18 | tm | is it okay? |
| 15:15.19 | tm | exten => _.,3,Read(Number,call-forwarding) |
| 15:15.19 | tm | exten => _.,4,Dial(misdn/1/${Number}) |
| 15:15.19 | tm | ? |
| 15:15.53 | blitzrage | Dovid: then you should read some documentation I'd suggest |
| 15:16.05 | blitzrage | A family contains one or more keys, and keys contain a value |
| 15:16.11 | Gankhuu | or even where I could read to figure it out myself? |
| 15:16.12 | b11d | tm : PASTEBIN.CA PASTEBIN.CA PASETEBIN.CA FOR CHRIST SAKES :) |
| 15:16.19 | Dovid | blitsrage: on voip info - it doent explain well |
| 15:16.22 | b11d | how many times must it be said? |
| 15:16.26 | Dovid | ;) |
| 15:16.35 | blitzrage | Dovid: read the book at www.asteriskdocs.org |
| 15:16.36 | *** join/#asterisk JerJer (n=PhatJ@pdpc/supporter/bronze/jerjer) |
| 15:16.39 | Dovid | ok |
| 15:16.42 | tm | b11d: sorry. |
| 15:16.49 | tm | b11d: but is the row ok? |
| 15:16.50 | Dovid | is the book there Aserisk TFOT ? |
| 15:16.53 | *** join/#asterisk klasstek (n=nunyobiz@ip67-154-143-190.z143-154-67.customer.algx.net) |
| 15:16.55 | blitzrage | Dovid: yes |
| 15:17.00 | Dovid | ok |
| 15:17.04 | Dovid | never got that fat |
| 15:17.46 | JerJer | surely someone in here has experienced this: two (or more) sip phones in an office using a 'hosted' service yet their office has NAT. how does one make it so calls between the IP phones keep the RTP local for those calls, but properly sends the RTP thru for pstn calls, without having a box locally dealing with this logic ? |
| 15:18.05 | *** join/#asterisk parag (n=Administ@dxb-b123242.alshamil.net.ae) |
| 15:18.07 | Dovid | JerJEr: I believe ur outa luck |
| 15:18.30 | blitzrage | JerJer: great question........ I've not done it either |
| 15:18.34 | Dovid | JerJer: I had an idea to use a wrt54g and put on ur own firmware |
| 15:18.41 | JerJer | i have to beleve SER should be able to pull this off |
| 15:18.42 | Dovid | and add asterisk |
| 15:18.46 | supjigatr | JerJer: On polys we just put direct DSS buttons. |
| 15:18.47 | JerJer | by using domains |
| 15:19.01 | JerJer | then give each 'office' their own domain |
| 15:19.01 | blitzrage | JerJer: that makes sense to me -- I've just started using SER myself |
| 15:19.06 | supjigatr | JerJer: We have done it with openser. |
| 15:19.06 | robl^ | Dovid: it stores data in a simple structure.. A "family" is like a group of related items. Like a folder or branch on a tree. The key allows you to identify a specific item in a family. For example, a family might be a category.. like "callforwading", the key might be an extension. so family callforwarding / key "240" and a value of "5551212" might mean to forward calls fro extension 240 to 5551212 |
| 15:19.31 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 15:19.38 | JerJer | hmm domains might not be necessary |
| 15:20.17 | JerJer | perhaps for TBD digits (which as to be less than a 'regular' pstn format) could be instructed not to 'fix' the nat addresses |
| 15:20.18 | supjigatr | JerJer: Just watch for NAT checks. You need to ignore and have the phone do a direct connect for local contacts. |
| 15:20.18 | JerJer | hmmmm |
| 15:20.35 | JerJer | supjigatr: |
| 15:20.35 | JerJer | yeah |
| 15:20.49 | ManxPower | and jerjer too |
| 15:20.50 | JerJer | i might have answered my own question by typing it out like this |
| 15:20.56 | JerJer | ManxPower: howdie |
| 15:21.05 | robl^ | hey ManxPower |
| 15:22.27 | *** join/#asterisk reber (n=reber@cl-157.dub-01.ie.sixxs.net) |
| 15:25.39 | *** join/#asterisk yxa (n=yxa@cm127.gamma228.maxonline.com.sg) |
| 15:26.36 | JerJer | supjigatr, thanks for helping too |
| 15:26.41 | supjigatr | np |
| 15:27.22 | *** join/#asterisk BSDTech (n=RNeese@adsl-69-230-170-85.dsl.irvnca.pacbell.net) |
| 15:29.36 | tm | how i can read the dtmf? |
| 15:29.51 | tm | with exten => s,1,Read(Nummer,give-number-sound,20) .. exten => s,2,Dial(misdn/1/${Number}) ? |
| 15:36.28 | *** join/#asterisk mhnoyes (n=mhnoyes@dialup-4.246.6.37.Dial1.SanJose1.Level3.net) |
| 15:37.32 | *** join/#asterisk malverian (n=malveria@gentoo/developer/malverian) |
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| 15:44.34 | *** join/#asterisk Crescendo_ (n=martinda@adsl-072-151-080-148.sip.rmo.bellsouth.net) |
| 15:44.36 | tm | hello? |
| 15:45.31 | *** part/#asterisk michael-i (n=michael-@141.41.40.55) |
| 15:49.22 | *** join/#asterisk zoa (n=d@pirus.securax.be) |
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| 15:56.57 | *** join/#asterisk twen (n=Twen@86.71.90.249) |
| 15:57.03 | twen | hop |
| 15:58.00 | aydiosmio | jump |
| 15:58.02 | aydiosmio | skip |
| 15:58.10 | *** part/#asterisk vooduhal (n=vooduhal@tc-proxy2.catt.com) |
| 15:58.32 | *** join/#asterisk apardo_ (n=apardo@87.217.147.90) |
| 15:58.43 | tm | can anybody help me? |
| 15:58.49 | tm | twen ? aydiosmio ? |
| 15:59.13 | Strom_M | tm, the answer is "cheese=yes" |
| 15:59.14 | twen | tm: about what ? |
| 15:59.15 | aydiosmio | you're beyond help my friend |
| 15:59.20 | twen | :) |
| 15:59.32 | tm | i have |
| 15:59.32 | tm | exten => 37061215,5,Authenticate(54321) |
| 15:59.32 | tm | exten => 37061215,6,DISA(no-password|weiterleitung) |
| 15:59.40 | tm | and |
| 15:59.41 | tm | [weiterleitung] |
| 15:59.52 | tm | exten => 37061215,3,Read(ziel) |
| 15:59.52 | tm | exten => 37061215,4,Dial(misdn/1/${ziel}) |
| 15:59.57 | tm | but not works :/ |
| 16:00.01 | tm | syntax error? |
| 16:00.17 | Qwell[] | where are your priority 1 and 2? |
| 16:00.30 | twen | in weiterleitung |
| 16:00.38 | Qwell[] | I only see 3,4 |
| 16:00.39 | tm | Qwell[]: |
| 16:00.40 | tm | exten => 37061215,1,DigitTimeout(5) |
| 16:00.40 | tm | exten => 37061215,2,ResponseTimeout(10) |
| 16:01.04 | Qwell[] | so, you're always going to dial 37061215 in DISA? |
| 16:01.13 | Qwell[] | because if not...then it isn't matching anything |
| 16:01.19 | tm | hm |
| 16:01.35 | b11d | . |
| 16:01.54 | Qwell[] | b11d: Be glad the ircops didn't kline you... they have a script that does it ;/ |
| 16:01.57 | Qwell[] | it was for your own good, heh |
| 16:02.04 | b11d | coulda just warned me |
| 16:02.06 | b11d | but its cool.. |
| 16:02.09 | twen | does someone knows where to buy fxo card for a computer ? (from France) :/ I couldn't find yet a reseller, any links ? |
| 16:02.36 | Qwell[] | b11d: If this were efnet, it would've been a bit longer than 20 minutes ;p |
| 16:02.46 | blitzrage | ? |
| 16:02.46 | b11d | :P |
| 16:02.53 | tm | Qwell[]: better? http://tm.muc.de/bla.txt |
| 16:03.11 | b11d | never the less.. I wonder what clients those guys were using to get knocked off like that.. |
| 16:03.29 | Qwell[] | b11d: various, sadly |
| 16:03.29 | b11d | i'd not seen that before |
| 16:03.29 | blitzrage | not client -- server |
| 16:03.56 | tm | Qwell[]: ? |
| 16:03.59 | b11d | haha |
| 16:04.03 | b11d | brb |
| 16:04.29 | Qwell[] | tm: not quite, heh |
| 16:04.46 | Qwell[] | tm: that would only match an empty extension |
| 16:05.11 | *** join/#asterisk Ebola (n=Ebola@host81-152-204-231.range81-152.btcentralplus.com) |
| 16:06.12 | tm | Qwell[]: ok and now? |
| 16:06.19 | tm | http://tm.muc.de/bla.txt refresh? |
| 16:06.31 | Qwell[] | tm: _X. would be far better |
| 16:06.45 | Qwell[] | but, it would be best if you could make it specific for your regions dialing plan |
| 16:07.12 | tm | http://tm.muc.de/bla.txt ? |
| 16:07.13 | tm | refresh |
| 16:07.20 | Qwell[] | meh |
| 16:07.59 | tm | meh? |
| 16:08.05 | Qwell[] | exactly |
| 16:08.25 | Qwell[] | it'll work, but it's not great |
| 16:08.34 | Qwell[] | and I don't know what your region uses for a numbering plan, so... |
| 16:08.53 | Qwell[] | like, in the US, we'd use _NXXNXXXXXX,1,blah |
| 16:09.06 | Qwell[] | well, _1NXXNXXXXXX |
| 16:09.25 | *** join/#asterisk Op3r (i=Op3r@121.97.192.57) |
| 16:09.55 | tm | Qwell[]: hm |
| 16:09.58 | tm | if i have: |
| 16:10.15 | tm | exten => _X.,4,Dial(misdn/1/08003301000) <<its works |
| 16:10.27 | tm | with |
| 16:10.28 | tm | exten => _X.,4,Dial(misdn/1/${ziel}) |
| 16:10.40 | tm | P[ 0] misdn_call: No Extension given! |
| 16:10.42 | tm | Qwell[]: hmm? |
| 16:11.00 | Qwell[] | I don't know - I just work here |
| 16:11.44 | *** join/#asterisk robl^ (n=robl@dsl093-025-218.hou1.dsl.speakeasy.net) |
| 16:13.27 | Corydon-w | tm: did you actually Set(miel=something) ? |
| 16:13.40 | Corydon-w | or ziel, or whatever variable you're using |
| 16:13.48 | Qwell[] | Read(ziel), or something |
| 16:14.22 | tm | Corydon-w: oeh |
| 16:14.40 | tm | Corydon-w: |
| 16:14.41 | tm | exten => _X.,3,Read(ziel) |
| 16:14.41 | tm | exten => _X.,4,Dial(misdn/1/${ziel}) |
| 16:14.42 | *** join/#asterisk robl^ (n=robl@dsl093-025-218.hou1.dsl.speakeasy.net) |
| 16:14.48 | tm | read <<for dtmf or? |
| 16:14.49 | Op3r | what's the php script that trixbox used to allow you to edit the conf's via web based? |
| 16:14.59 | tm | Corydon-w: what u mean? |
| 16:15.03 | b11d | trixbox is in #openpbx |
| 16:15.04 | Qwell[] | Op3r: see topic |
| 16:15.15 | *** join/#asterisk Fraeggl (n=ke@rkom.r-kom.de) |
| 16:15.23 | tm | Fraeggl: tach |
| 16:15.24 | b11d | oh man Fraggle rock was the best |
| 16:15.26 | b11d | I miss boober |
| 16:15.28 | Op3r | oh ok |
| 16:15.31 | Op3r | sorry |
| 16:15.31 | Op3r | :( |
| 16:15.33 | b11d | np |
| 16:15.38 | Corydon-w | tm: so are you entering any digits at that prompt? |
| 16:15.46 | *** join/#asterisk aleswy (n=aleswy@82.159.11.168) |
| 16:15.54 | tm | Corydon-w: aeh with my isdn phone u mean? jo |
| 16:16.15 | tzafrir_ | b11d, was that intentional? (#openpbx)? |
| 16:16.20 | tm | Corydon-w: hm? |
| 16:16.21 | b11d | yes |
| 16:16.28 | *** join/#asterisk topping_ (n=topping@207.47.6.182.static.nextweb.net) |
| 16:16.29 | b11d | am I wrong |
| 16:16.32 | b11d | its freepbx isnt it |
| 16:16.38 | b11d | why do I ALWAYS get those backwards.. |
| 16:16.39 | *** join/#asterisk marv[work] (n=timr@24.214.206.254) |
| 16:16.43 | b11d | oh yeah.. the weed.. |
| 16:16.49 | *** join/#asterisk brodiem (n=brodiem@67.110.68.66.ptr.us.xo.net) |
| 16:16.50 | Corydon-w | tm: might also be that you haven't answered the channel |
| 16:17.12 | tm | hm |
| 16:17.29 | irq | b11d: you've probably seen a lot of these before but i don't think i gave you the url last time, http://zeppelin.stepahead.net/~dan/nikki (definitely nsfw) |
| 16:17.34 | tzafrir_ | Op3r, the script name is asterisk-config or something. You can find it in the Asterisk distribution in the contrib section |
| 16:17.51 | Corydon-w | tm: and it's really kind of strange that you don't just use ${EXTEN} |
| 16:17.57 | Op3r | tzafrir_, oh ok |
| 16:17.58 | Op3r | thanks |
| 16:17.59 | b11d | irq, you always make my day so much brighter.. |
| 16:18.00 | b11d | :) |
| 16:18.04 | irq | he |
| 16:18.06 | irq | hrh |
| 16:18.08 | irq | damn! |
| 16:18.09 | irq | heh |
| 16:18.18 | *** join/#asterisk Strom_C_ (n=strom@netblock-66-159-243-60.dslextreme.com) |
| 16:18.36 | tzafrir_ | Op3r, mind you, it requires permitting the web server to write to Asterisk configuration. Which is not necessarily a smart idea |
| 16:18.38 | *** join/#asterisk jart` (n=user@ool-44c05258.dyn.optonline.net) |
| 16:18.50 | b11d | is this the result of your weekend of trolling usenet? |
| 16:19.02 | Op3r | tzafrir_, im thinking about integrating it to the vicidial admin page though |
| 16:19.04 | tm | Corydon-w: u mean exten => _X.,4,Dial(misdn/1/${EXTEN}) ? |
| 16:19.21 | Corydon-w | tm: yep |
| 16:19.29 | *** join/#asterisk shellsha1k (n=x86@74-135-64-209.dhcp.insightbb.com) |
| 16:19.34 | tm | Corydon-w: i test it. |
| 16:19.45 | tzafrir_ | Op3r, are you sure you want to allow arbitrary editing? If you want to allow arbitrary editing, use winscp and be done with it |
| 16:19.53 | irq | no b11d, i've had those for a long time, it was going to be a project but i ended up getting a real job |
| 16:20.06 | b11d | lol |
| 16:20.42 | tm | Corydon-w: ah, now. its works ;) |
| 16:20.45 | tm | Corydon-w: :-))) |
| 16:20.49 | Op3r | tzafrir_, i am just going to see the code that show's how they edit the confs then just create an interface to add a trunk and sip extensions nothing more |
| 16:20.49 | tzafrir | It puts a saner and more controlled method of authentication. |
| 16:21.00 | Corydon-w | Imagine that |
| 16:21.46 | blitzrage | b11d: don't blame the weed |
| 16:21.52 | tzafrir | Op3r, also consider the simple interface from #asterisk-gui is you want very simple things and can afford 1.4 |
| 16:22.49 | Op3r | tzafrir, still dont want to use 1.4 cos its not yet needed |
| 16:23.15 | tzafrir | Op3r, for the #asterisk-gui it is a requirement. |
| 16:24.02 | b11d | you're right blitzrage.. its not the weeds fault.. |
| 16:24.04 | *** join/#asterisk anthonyl_ (n=anthonyl@65.4.0.174) |
| 16:24.04 | b11d | it never is.. |
| 16:24.04 | b11d | :) |
| 16:25.41 | tm | Corydon-w: hach hach |
| 16:26.50 | tm | Corydon-w: the timeout is long hmm |
| 16:27.10 | tm | for the dtmf |
| 16:27.15 | Corydon-w | ~thebook |
| 16:27.17 | jbot | methinks thebook is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 16:27.23 | Corydon-w | Go read. |
| 16:27.44 | tm | Corydon-w: under? |
| 16:27.46 | tm | the word? |
| 16:27.52 | *** join/#asterisk Defraz (n=t0tal@fw.centrisys.com) |
| 16:28.13 | b11d | wow. |
| 16:29.42 | *** join/#asterisk visba (n=dca[lapt@sta-208-139-193-162.rockynet.com) |
| 16:30.25 | b11d | VISBA!!!!! |
| 16:30.34 | *** join/#asterisk buleeahn (n=buleeahn@rrcs-24-227-212-162.sw.biz.rr.com) |
| 16:30.44 | b11d | BULEEAHN!!!!!!! |
| 16:30.56 | buleeahn | :) |
| 16:30.59 | b11d | its just like old times :) |
| 16:31.33 | tm | Corydon-w: oeh |
| 16:31.44 | tm | exten => _X.,1,DigitTimeout(1) |
| 16:31.44 | tm | exten => _X.,2,ResponseTimeout(2) |
| 16:35.01 | buleeahn | So, does anyone know the difference between 1.2.13 and 1.2.14 ? |
| 16:35.32 | Qwell[] | buleeahn: the release announcement explained the difference |
| 16:35.54 | buleeahn | I missed that then...checked the changelog and readme, I'll go peek at that. |
| 16:36.21 | buleeahn | pfft...yeah, that big obvious notice on the website...that fills in the blanks. |
| 16:36.36 | Op3r | have anyone tried using mysql for sip or iax extensions? |
| 16:37.00 | b11d | lol.. |
| 16:37.04 | b11d | buleeahn.. you rule |
| 16:37.29 | *** join/#asterisk wunderkin (i=kev@ip72-208-3-221.ph.ph.cox.net) |
| 16:40.38 | *** join/#asterisk tr2x (n=alvar@80-218-185-55.dclient.hispeed.ch) |
| 16:40.55 | *** join/#asterisk nukio (n=deep@cpe-66-65-187-187.nyc.res.rr.com) |
| 16:41.49 | *** join/#asterisk CunningPike (n=CunningP@dhcp-10-234.district.north-van.bc.ca) |
| 16:42.50 | *** join/#asterisk [1]Dovid (n=Dovid@pool-72-68-86-245.nwrknj.east.verizon.net) |
| 16:43.57 | *** join/#asterisk [2]Dovid (n=Dovid@ool-43530a83.dyn.optonline.net) |
| 16:46.06 | *** join/#asterisk grantm (n=grantm@207.88.78.2) |
| 16:46.11 | dasenjo | Hi! |
| 16:46.24 | *** part/#asterisk brodiem (n=brodiem@67.110.68.66.ptr.us.xo.net) |
| 16:46.43 | dasenjo | I'm obtaining an error trying to get the callerid on TDM400 FXO: |
| 16:47.02 | dasenjo | Dec 19 11:24:43 ERROR[21795]: callerid.c:276 callerid_feed: fsk_serie made mylen < 0 (-5) |
| 16:47.17 | dasenjo | what can I do? |
| 16:47.22 | *** join/#asterisk brodiem (n=brodiem@rrcs-24-227-171-186.sw.biz.rr.com) |
| 16:49.05 | b11d | weird |
| 16:49.17 | *** join/#asterisk irq (n=dan@wsip-70-168-52-194.sd.sd.cox.net) |
| 16:49.18 | b11d | it sets the caller id to a negative integer.. strange indeed.. |
| 16:49.39 | b11d | try turning callerid off on that particular sip peer? |
| 16:49.41 | b11d | is that possibe |
| 16:51.28 | *** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 16:51.36 | dasenjo | sip peer? I'm getting the error on a zap line |
| 16:52.11 | *** mode/#asterisk [+o mog] by ChanServ |
| 16:54.14 | *** join/#asterisk hfb (n=hfb@pool-71-118-252-158.lsanca.dsl-w.verizon.net) |
| 16:54.26 | resistance | is there something out there that can be used to keep a record of calls with asterisk? |
| 16:55.28 | blitzrage | resistance: its called CDR |
| 16:56.13 | *** join/#asterisk kippi (n=pssedoff@untrust-gct.equinoxit.net) |
| 16:57.46 | resistance | ahhhh |
| 17:07.06 | tm | Qwell[]: hello? |
| 17:07.38 | ai-a[work] | is view/asterisk/trunk/ always the head version ? |
| 17:09.10 | *** join/#asterisk BSDTech (n=RNeese@adsl-69-230-170-85.dsl.irvnca.pacbell.net) |
| 17:12.17 | *** join/#asterisk root (n=root@200-127-42-155.cab.prima.net.ar) |
| 17:12.27 | shellsha1k | is it possible to get fractional PRI's? |
| 17:12.29 | Qwell[] | ~root |
| 17:12.31 | jbot | extra, extra, read all about it, root is not a Good Thing to use when using IRC. Please use a different account. You will probably not be able to speak until change your user account. |
| 17:12.34 | Qwell[] | shellsha1k: sure |
| 17:12.44 | boquita | hello |
| 17:13.31 | shellsha1k | Qwell[]: cool, because POTS lines in my area are crap, and I don't need 23 total channels |
| 17:14.03 | blitzrage | ai-a[work]: its the trunk version (no such thing as HEAD anymore) -- but it is the equivelent of HEAD (newest development version -- not for production) |
| 17:14.09 | bkruse | shellsha1k: ya, call up your local provider and ask, ive seen fract t1 with 12 channels |
| 17:14.17 | shellsha1k | Qwell[]: i've never priced out a voice PRI, is the pricing about the same as a data T1 circuit? |
| 17:14.22 | ai-a[work] | blitzrage: but something to use for the fun of it :) |
| 17:14.22 | b11d | get your d channel on channel 7 |
| 17:14.26 | b11d | for no real reason |
| 17:14.35 | ai-a[work] | blitzrage: sorry, still a cvs user :) |
| 17:14.41 | *** join/#asterisk boqui (n=boqui@200-127-42-155.cab.prima.net.ar) |
| 17:14.43 | blitzrage | ai-a[work]: ahh... asterisk uses SVN now :) |
| 17:14.49 | bkruse | b11d: ha i like it |
| 17:14.50 | blitzrage | thank god |
| 17:14.53 | boqui | hi |
| 17:14.58 | bkruse | yay svn! |
| 17:15.02 | blitzrage | indeed! |
| 17:15.11 | blitzrage | ugh.... back to more dialplan |
| 17:15.12 | shellsha1k | b11d: when then build it out they let you choose what channel the D channel takes up? |
| 17:15.27 | ai-a[work] | we'll convert soon.. |
| 17:15.34 | *** join/#asterisk bnolte (n=bnolte@207.210.228.172) |
| 17:15.34 | blitzrage | maybe I'll get this clustered centrex environment ready for testing soon |
| 17:15.36 | bnolte | http://tinyurl.com/y4tund |
| 17:15.58 | bkruse | blitzrage: yay clusters |
| 17:16.03 | bkruse | rocks? |
| 17:16.13 | bkruse | cluster knoppix ( based off rocks i think ) |
| 17:16.18 | bkruse | beowulf? |
| 17:16.37 | blitzrage | bkruse: nope -- own solution |
| 17:16.42 | b11d | shell.. dont listen to me.. |
| 17:16.47 | bkruse | blitzrage: elaborate plz ;] |
| 17:16.49 | b11d | im kidding about d on ch7 |
| 17:17.11 | blitzrage | bkruse: I'm just using a clustered DB solution, then using Asterisk stuff to cluster with the DB and across the network |
| 17:17.43 | blitzrage | each box is a node in the cluster, and it doesn't matter where you drop the calls into, it routes the call to the appropriate destination |
| 17:18.00 | blitzrage | even if the destination isn't on the box the call was delivered to |
| 17:18.53 | bkruse | blitzrage: awesome, mysql clusters management stuff is Very interesting |
| 17:19.02 | blitzrage | mysql sucks |
| 17:19.07 | blitzrage | I use postgresql |
| 17:19.35 | bkruse | blitzrage: good choice ;] |
| 17:19.38 | boqui | you know a error in te110p where channels is restarted? |
| 17:20.57 | b11d | does anyone know how I can manipulate what the "reject" option does when i hit it on a Polycom 501? |
| 17:21.39 | *** join/#asterisk [hC] (n=hardcore@S0106000fb51cc225.vf.shawcable.net) |
| 17:22.21 | *** join/#asterisk Pelipe (n=Pelipe@dslc-082-082-092-125.pools.arcor-ip.net) |
| 17:22.47 | Pelipe | Hello |
| 17:22.49 | b11d | high |
| 17:22.52 | *** join/#asterisk s1gny|wrk (n=s1gny@p549171E9.dip.t-dialin.net) |
| 17:23.00 | *** part/#asterisk s1gny|wrk (n=s1gny@p549171E9.dip.t-dialin.net) |
| 17:23.06 | Pelipe | can you tell me how to change the language of my aterisk? |
| 17:23.14 | Pelipe | i am german |
| 17:24.06 | boqui | anyone see this error before? pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 |
| 17:24.21 | b11d | only if you can tell me what the hell a specific german phrase is.. all I know is it sounds like "rushus los" and people say it when greeting one anothe.r |
| 17:24.30 | b11d | but its probably not that at all.. |
| 17:24.34 | blitzrage | boqui: does it say [ERROR] beside it, or WARNING, or NOTICE? |
| 17:24.37 | swilliamson | does not sound right |
| 17:24.42 | swilliamson | los is go |
| 17:24.43 | swilliamson | or move |
| 17:25.14 | boqui | NOTICE |
| 17:25.14 | b11d | hrm..i know im spelling it wrong.. |
| 17:25.14 | blitzrage | boqui: then its not an ERROR |
| 17:25.15 | b11d | but its a common phrase |
| 17:25.15 | b11d | ive heard it in movies, etc.. |
| 17:25.15 | boqui | bliztrage: i know |
| 17:25.19 | swilliamson | Pelipe: do you have the german recordings |
| 17:25.25 | Pelipe | nope |
| 17:25.28 | Pelipe | i dont know |
| 17:25.40 | swilliamson | oh where are the recordings stored again... |
| 17:25.41 | Pelipe | how to check? |
| 17:25.41 | swilliamson | sec |
| 17:25.47 | Pelipe | mkay |
| 17:26.03 | *** part/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 17:26.06 | boqui | bliztrage: but this not affect the performance? |
| 17:26.27 | b11d | no, it wont.. |
| 17:26.27 | infernix | "Wass ist los" probably |
| 17:26.31 | b11d | THATS IT! |
| 17:26.40 | b11d | wtf does that mean? |
| 17:26.44 | infernix | it means "what's up" |
| 17:26.48 | Pelipe | ^^ |
| 17:26.49 | infernix | sort of |
| 17:26.50 | Pelipe | its german |
| 17:26.53 | b11d | I thought "los" meant go.. |
| 17:26.53 | b11d | :) |
| 17:26.55 | b11d | ahh |
| 17:26.58 | Pelipe | jep |
| 17:27.01 | b11d | kind of a "lost in translation" thing eh |
| 17:27.20 | b11d | I appreciate you finally putting that one to rest for me.. |
| 17:27.24 | b11d | its bothered me for YEARS |
| 17:27.24 | b11d | :) |
| 17:27.29 | infernix | yw |
| 17:27.32 | swilliamson | " /var/lib/asterisk/sounds |
| 17:27.36 | Pelipe | okay |
| 17:27.38 | swilliamson | look for -de |
| 17:27.40 | Pelipe | i'll check |
| 17:27.41 | swilliamson | in the file name |
| 17:27.48 | swilliamson | *names |
| 17:27.49 | Pelipe | and then ? |
| 17:27.59 | swilliamson | was ist los |
| 17:28.07 | swilliamson | b11d: that is what you are thinking of |
| 17:28.22 | b11d | yeah, like infernix said.. |
| 17:28.24 | boqui | ok, and.. anyone see this error? |
| 17:28.26 | swilliamson | means what's up, or directly translated "what is going" |
| 17:28.28 | swilliamson | missed it |
| 17:28.32 | b11d | thats cool.. |
| 17:28.34 | b11d | I appreciate that |
| 17:28.40 | boqui | Dec 13 02:01:44 WARNING[1622] chan_zap.c: Detected alarm on channel 13: Yellow Alarm |
| 17:28.53 | b11d | it totally makes sense. i'm going to say it now that I know what it means :) |
| 17:29.03 | boqui | before this the channels are reseted |
| 17:29.16 | Pelipe | swilliamson: there is noch folder named sounds in /var/lib/asterisk/ |
| 17:29.35 | swilliamson | b11d: hit the non-vowels hard when you do say it |
| 17:29.48 | swilliamson | Pelipe: humm, what install you using |
| 17:29.59 | swilliamson | Pelipe: linux distro/asterisk version |
| 17:30.00 | Pelipe | 1.2.13 |
| 17:30.07 | Pelipe | debian |
| 17:30.22 | swilliamson | and the asterisk was installed by what method |
| 17:30.40 | Pelipe | i dont know, mom |
| 17:30.44 | b11d | will do |
| 17:31.08 | swilliamson | ha |
| 17:31.31 | Pelipe | via apt-get install |
| 17:32.05 | swilliamson | okay, maybe do a find / -name vm-intro.gsm |
| 17:32.24 | *** join/#asterisk in-pt (n=lokesh@estrela.nortenet.pt) |
| 17:32.29 | boqui | anyone? |
| 17:32.39 | Pelipe | /usr/share/asterisk/sounds/vm-intro.gsm |
| 17:32.44 | swilliamson | boqui: no idea |
| 17:32.51 | swilliamson | boqui: no zap here |
| 17:32.55 | swilliamson | ah |
| 17:33.02 | *** join/#asterisk h0 (n=fakhir@unaffiliated/fakhir) |
| 17:33.13 | swilliamson | Pelipe: then check this page for files: http://www.voip-info.org/wiki/view/Asterisk+sound+files+international#German |
| 17:33.21 | Pelipe | thanks |
| 17:33.30 | swilliamson | kein problem |
| 17:33.40 | *** join/#asterisk Corydon76-home (i=six@c-68-53-162-99.hsd1.tn.comcast.net) |
| 17:33.40 | *** mode/#asterisk [+o Corydon76-home] by ChanServ |
| 17:33.59 | boqui | swilliamson: where can find the solution for this?? |
| 17:34.15 | *** join/#asterisk diegor (n=diegor@85.116.131.5) |
| 17:34.27 | swilliamson | boqui: what does google say when you put in the error in quotes |
| 17:34.31 | *** part/#asterisk diegor (n=diegor@85.116.131.5) |
| 17:34.34 | *** part/#asterisk elvistheslug (n=geoffsum@user-0c6tcca.cable.mindspring.com) |
| 17:34.37 | *** part/#asterisk EmleyMoor (i=ejabberd@hallam.tinsleyviaduct.com) |
| 17:35.56 | *** join/#asterisk jcims (n=jcims@rrcs-24-172-217-2.central.biz.rr.com) |
| 17:36.01 | swilliamson | boqui: other than that check voip-info.org |
| 17:37.17 | boqui | swilliamson: i'm check on this places, asterisk-guru say card broken but i trade with 3 cards and all have same problem |
| 17:38.20 | swilliamson | boqui: humm, could be the kernel module |
| 17:39.03 | swilliamson | maybe it's not being loaded, check a lsmod for zt... stuff |
| 17:39.35 | *** join/#asterisk Crescendo_ (n=martinda@adsl-072-151-080-148.sip.rmo.bellsouth.net) |
| 17:39.46 | boqui | swilliamson: all ok, i can call good but every five minutes the channels is reseting |
| 17:39.49 | boqui | *are |
| 17:39.57 | swilliamson | oh |
| 17:41.21 | *** join/#asterisk irq (n=dan@wsip-70-168-52-194.sd.sd.cox.net) |
| 17:41.21 | boqui | swilliamson: with the last card this time is too large, every... fifty minutes |
| 17:41.21 | *** join/#asterisk jerryoc (n=jerryoc@cpe-75-80-102-22.socal.res.rr.com) |
| 17:41.51 | redondos | boqui: What does your provider say? |
| 17:42.10 | redondos | You're using what, E1? With whom, Telefonica? |
| 17:43.01 | PupenoR | Hello. |
| 17:43.07 | PupenoR | Anyone tried trixbox ? |
| 17:43.36 | redondos | PupenoR: Yes, but this is not a Trixbox support channel. Read the topic, join #freepbx. |
| 17:43.42 | *** join/#asterisk irq (n=dan@wsip-70-168-52-194.sd.sd.cox.net) |
| 17:44.35 | swilliamson | boqui: I have never used a E/T1 with a zap card... sorry |
| 17:45.14 | *** join/#asterisk sahafeez (n=sahafeez@67.109.14.227.ptr.us.xo.net) |
| 17:46.02 | *** join/#asterisk anthonyl (i=anthonyl@nat/digium/x-45ddc7b2a4693baa) |
| 17:47.36 | kippi | where can i find the information about putting the cdr data to mysql? |
| 17:47.37 | Younss | hi |
| 17:47.46 | redondos | kippi: voip-info.org has it. |
| 17:48.13 | kippi | ah |
| 17:48.21 | Younss | can you tell me which is the best material configuration for asterisk to work well ( i mean CPU Speed,RAM...)? |
| 17:48.24 | *** join/#asterisk stephane_ (n=stephane@gw.sortilege.net) |
| 17:48.36 | redondos | Younss: There's no rule of thumb. |
| 17:50.36 | kippi | I have installed Asterisk-Stat and setup my database but it dosn't seem to be drawing the graphis anyideas/ |
| 17:51.09 | ai-a[work] | how do i install the sources for the 2.6.18-1.2849.fc6 kernel ? |
| 17:54.03 | Younss | redondos: why |
| 17:54.06 | Younss | ? |
| 17:54.13 | swilliamson | base hp gl360 g4 is what I use |
| 17:54.17 | swilliamson | works good |
| 17:54.23 | swilliamson | is not cheep |
| 17:54.44 | *** join/#asterisk vAd0r (n=poop@216-201-139-51.res.logixcom.net) |
| 17:54.51 | vAd0r | helo |
| 17:55.05 | *** join/#asterisk sukimono (n=sukimono@202.164.181.222) |
| 17:55.08 | vAd0r | does anyone know how to setup a cisco vg200 to connect to asterisk |
| 17:55.15 | vAd0r | as my trunk |
| 17:56.09 | *** join/#asterisk irq (n=dan@wsip-70-168-52-206.sd.sd.cox.net) |
| 17:56.33 | Younss | swilliamson: how many users do yo have |
| 17:56.51 | swilliamson | 1500 just for voicemail, we hope to support 24 concurrent calls |
| 17:56.54 | swilliamson | via sip |
| 17:56.59 | swilliamson | from cisco call manager |
| 17:57.16 | swilliamson | but I am sure it could do a lot more |
| 17:57.29 | vAd0r | will vg200 work with asterisk |
| 17:57.37 | Younss | i ve |
| 17:57.45 | Younss | i ve more than 50 users |
| 17:58.31 | swilliamson | if you have the cash, and a rack, i would get the hp dl360 cuz it's on the tested list from digium... you can get support and run business edition on it |
| 17:58.57 | swilliamson | Younss: softphone? ata's? zap T1's? what is your config |
| 17:58.58 | Younss | i build my own pc's |
| 17:59.11 | Younss | i'll bay polycom |
| 17:59.27 | Younss | i ve bouth zap & T1 |
| 17:59.43 | b11d | does anyone know how to manipulate the "reject" button on Polycom 501's ? |
| 17:59.52 | swilliamson | well you need to consider # of concurrent calls, and what channels, is there media conversion, from sip/rtp to t1 for example? |
| 18:00.29 | Younss | that's why i m asking about the best configuration |
| 18:00.41 | swilliamson | will you have a T1? |
| 18:00.41 | vAd0r | does anyone know the answer |
| 18:00.43 | vAd0r | ? |
| 18:00.54 | b11d | im using a vg224 |
| 18:01.01 | b11d | cisco vg224 |
| 18:01.06 | swilliamson | vAd0r: no, b11d: no idea |
| 18:01.09 | redondos | Younss: I also build most of the servers I administer, unless the customer wants support from the vendors and that's their problem. |
| 18:01.19 | b11d | im going to have to email polycom.. |
| 18:01.21 | redondos | Younss: what are those 50 users going to do? what kind of simultaneity do you expect? |
| 18:01.38 | swilliamson | Younss: will you have a T1? |
| 18:01.42 | b11d | I dont know what the vg200 is.. |
| 18:01.55 | Younss | yes |
| 18:02.03 | vAd0r | does anyone know how to setup a cisco vg200 to connect to asterisk |
| 18:02.06 | b11d | YES |
| 18:02.08 | b11d | christ |
| 18:02.10 | b11d | read the responses |
| 18:02.21 | vAd0r | b11d who are you talking to |
| 18:02.23 | b11d | YOU |
| 18:02.25 | swilliamson | okay, I would use some server class MB, like from tyan with multiple pci busses, one for the zap T1 card, one for the nic |
| 18:02.28 | vAd0r | what response |
| 18:02.29 | *** join/#asterisk MGSsancho (i=howdy@adsl-68-120-71-172.dsl.irvnca.pacbell.net) |
| 18:02.32 | b11d | ugh |
| 18:02.34 | b11d | look.. |
| 18:02.38 | *** join/#asterisk VJG (i=VijayG@202.131.145.235) |
| 18:02.39 | vAd0r | ok now i see yours |
| 18:02.41 | b11d | the vg-series works with asterisk |
| 18:02.42 | VJG | hello |
| 18:02.45 | vAd0r | can you take a moment to help me |
| 18:02.46 | vAd0r | please |
| 18:02.46 | b11d | im using the vg224 and it runs fine |
| 18:02.49 | b11d | yes |
| 18:02.53 | b11d | because others helped me with mine.. |
| 18:02.54 | Younss | i'll bay tyan |
| 18:02.57 | vAd0r | i have a vg224 at my work |
| 18:02.57 | monsted | i don't think any sort of voice application can stress a pci bus |
| 18:02.58 | b11d | so I will help you with yours |
| 18:02.58 | vAd0r | to cisco |
| 18:02.59 | Younss | i ve here 7 TYAN server |
| 18:03.04 | vAd0r | and a vg200 at my house |
| 18:03.04 | b11d | to cisco? |
| 18:03.06 | VJG | hello |
| 18:03.09 | Younss | they work all well |
| 18:03.12 | vAd0r | cisco call manager at work |
| 18:03.14 | b11d | ok vad0r. quit breaking up your sentences so muhc |
| 18:03.16 | vAd0r | asterisk at home |
| 18:03.35 | b11d | I |
| 18:03.35 | b11d | hate |
| 18:03.36 | b11d | when people |
| 18:03.38 | b11d | talk |
| 18:03.39 | b11d | like this |
| 18:03.39 | groogs | what happens if you have parkext=>70 and parkpos=>70-79 (ie, they overlap)? I'm assuming that's a bad thing(tm) to do? |
| 18:03.44 | vAd0r | i want to plug in my home telephone line to my vg200 and then make it a trunk to asterisk so i can get local calls to my system |
| 18:03.44 | swilliamson | Younss: asterisk is timeing sensitive, so that will help, and modern cpu and normal ammount of ram ~1gb should do you |
| 18:04.11 | b11d | hrm.. never done that.. I only use SIP on the vg224 |
| 18:04.12 | swilliamson | vAd0r: do you even have IOS on there that supports SIP? |
| 18:04.12 | vAd0r | i know the vg200 was setup for my call manager that i had running at my house and working so do i just need to build a trunk for it |
| 18:04.13 | Younss | swilliamson: an amd opteron is good? |
| 18:04.39 | *** join/#asterisk irq_ (n=dan@wsip-70-168-52-206.sd.sd.cox.net) |
| 18:04.45 | b11d | im using opterons in my asterisk servers |
| 18:05.05 | vAd0r | how is yours setup b11d |
| 18:05.08 | swilliamson | Younss: can't say 100% but should be fine... be carful that you checkout the distro and package version of asterisk that you will run to see that it supports the cpu and mb chipset well |
| 18:05.09 | infernix | I wonder if it'll run on my SGI O2... |
| 18:05.18 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
| 18:05.24 | b11d | <analog phones>----[vg224]==(SIP)===[asterisk] |
| 18:05.48 | Younss | redondos: my 50 users hemm... may be 30 will make calls in the internal and 12 make outgoing calls |
| 18:05.56 | vAd0r | how did you link it to your asterisk |
| 18:06.08 | b11d | with SIP as i've illustrated |
| 18:06.19 | swilliamson | maybe try it with a spare you have around, that config sounds fine to me though |
| 18:06.25 | b11d | over a category 5e cable |
| 18:06.37 | vAd0r | i understand that part |
| 18:06.41 | b11d | thats all |
| 18:06.53 | Younss | swilliamson:that what i m afraid that asterisk will not support the hardweare |
| 18:06.56 | vAd0r | i want to setup analog phone -->vg200 sip to asterisk |
| 18:07.05 | b11d | ok.. so whats the issue then? |
| 18:07.13 | vAd0r | what did you do to make your vg talk to asterisk |
| 18:07.19 | b11d | i'll post my config here in a minute.. |
| 18:07.20 | redondos | Younss: Do you want to record these calls? What codec are you going to use? What type of users are they? (SIP?) |
| 18:07.28 | vAd0r | cool thanks alot |
| 18:07.29 | swilliamson | Younss: it should, pay close attention to the PCI cards for T1's as they have specific requirements for bus voltages and such |
| 18:07.32 | *** join/#asterisk wiljacket (n=wilson@cpe-76-173-243-4.socal.res.rr.com) |
| 18:07.33 | b11d | np.. hang on |
| 18:08.10 | redondos | Younss: I think any 2.6GHz system with 1-2GB of memory will be able to handle that with room for expansion, even. But, as usual, the better the hardware the better the results. |
| 18:08.18 | *** join/#asterisk rmayorga (i=rmayorga@168.243.89.17) |
| 18:08.33 | swilliamson | and who wants to upgrade hardware right after you put something in production |
| 18:09.10 | Younss | i don't gess |
| 18:09.12 | Younss | ?? |
| 18:09.23 | swilliamson | I run 19 users voicemail off a crappy p4 celeron desktop no problem |
| 18:09.40 | b11d | vAd0r.. |
| 18:09.40 | b11d | http://pastebin.ca/285214 |
| 18:09.42 | Younss | tyan is good no |
| 18:09.43 | b11d | thats my config.. |
| 18:09.51 | swilliamson | sound is a little sukky though, choppy |
| 18:10.03 | b11d | I've obfucscated the passwords and stuff.. |
| 18:10.27 | vAd0r | cool |
| 18:10.36 | groogs | swilliamson: oh, that's a great thing to do, especially if you're bored on a friday afternoon |
| 18:10.36 | vAd0r | did you have to setup a trunk or anything in asterisk |
| 18:10.52 | b11d | i just add each Voice Port peer as an entry in sip.conf |
| 18:10.54 | swilliamson | Younss: If you have the money, I would go with my config, hp gl360 + asterisk business edition, just because it's tested and you can yell at digium if it breaks |
| 18:10.58 | b11d | like that 5454 is a 'sip peer' in sip.conf |
| 18:11.07 | *** join/#asterisk avalone (n=avalone_@dial-029.vl-cen-as3.avtlg.ru) |
| 18:11.17 | b11d | the "session target" IP is the asterisk server itself |
| 18:11.21 | Younss | no problem with money |
| 18:11.22 | b11d | that will be the IP of your 'remote end' |
| 18:11.25 | swilliamson | groogs: ha |
| 18:11.31 | Younss | when it tuch the production |
| 18:12.14 | swilliamson | swilliamson: we spent about 7k CAD for software, support contract and gear |
| 18:12.16 | *** join/#asterisk boqui (n=boqui@200-127-42-155.cab.prima.net.ar) |
| 18:12.23 | *** join/#asterisk sigmounte (n=sigmount@www.sighq.net) |
| 18:12.23 | b11d | talking to yourself again |
| 18:12.25 | vAd0r | so you add it manually in the sip.conf |
| 18:12.28 | b11d | yep |
| 18:12.33 | b11d | that 5454 has a regular entry in sip.conf |
| 18:12.37 | b11d | "sip show peers" shows it as a regular peer |
| 18:12.38 | *** join/#asterisk Ravi1974 (n=I@static-70-19-119-112.ny325.east.verizon.net) |
| 18:12.50 | vAd0r | can you paste that portion of the sip.conf |
| 18:12.53 | b11d | nope |
| 18:13.02 | b11d | its a regular sip entry.. |
| 18:13.03 | vAd0r | i usually use the webpage to do it |
| 18:13.06 | b11d | ugh |
| 18:13.12 | b11d | i'd hit you with a ruler if I could.. |
| 18:13.16 | vAd0r | as i just installed this 2 days ago |
| 18:13.18 | b11d | bad vAd0r.. bad.. |
| 18:13.20 | vAd0r | sry man |
| 18:13.21 | swilliamson | cisco unified messenger for the same number of users was 305k |
| 18:13.26 | b11d | you're learning bad habbits already then :) |
| 18:13.33 | Qwell[] | swilliamson: how many users? |
| 18:13.35 | b11d | take the time, learn the system.. |
| 18:13.38 | b11d | do NOT rush |
| 18:13.44 | Ravi1974 | is there a way to differentiate between carriers? Like cingular or verizon, land line or VOIP or Cellular? |
| 18:13.45 | b11d | or you'll have a nicely fucked phone system.. |
| 18:13.49 | swilliamson | Qwell ~1500 |
| 18:13.51 | *** join/#asterisk distortion (i=distorti@junipero.3sheep.com) |
| 18:14.04 | vAd0r | once thing that sucked when i used the web it didn't seem to actually go into the files |
| 18:14.07 | vAd0r | is that normal |
| 18:14.12 | b11d | I dont know |
| 18:14.18 | b11d | I never use the "web" for configuring asterisk |
| 18:14.30 | b11d | because thats all a bunch of moron bullshit |
| 18:14.33 | swilliamson | b11d: what if I use a web text editor? |
| 18:14.38 | b11d | then you're cool.. |
| 18:14.53 | b11d | still.. i'd probably rip on you a bit :) |
| 18:14.59 | swilliamson | ha |
| 18:15.11 | b11d | nah im totally aware that people should use what works best for htem.. |
| 18:15.18 | b11d | its just that 95% of the people DONT TRY ANYTHING ELSE |
| 18:15.19 | b11d | :) |
| 18:15.38 | b11d | not only that, they cant understand why the "web interface" doesnt have "all the options" |
| 18:15.42 | b11d | that others are talking about.. |
| 18:15.51 | swilliamson | nah, the web is okay, saves time, lets you do stuff without knowing how it works at a lower level, try to do something outside of the user/extension/trunk model and you are not going to get it done |
| 18:15.57 | b11d | weak |
| 18:16.18 | swilliamson | asterisk still needs extensions.conf to be edited to get what you want done |
| 18:16.20 | b11d | well anyway |
| 18:16.35 | b11d | I need to know how to manipulate the "reject" option for inbound calls |
| 18:16.36 | swilliamson | I guess if you could make macros or modules for freepbx |
| 18:16.39 | b11d | on Poly 501's |
| 18:17.12 | swilliamson | b11d: i am agreeing with you I think |
| 18:17.31 | vAd0r | if i edit the sip.conf |
| 18:17.39 | vAd0r | do i need to restart any services or anyting |
| 18:17.45 | b11d | "reload" |
| 18:17.48 | b11d | at the CLI |
| 18:18.03 | vAd0r | so just type reload? |
| 18:18.03 | b11d | yep |
| 18:18.06 | b11d | at the CLI |
| 18:18.10 | b11d | not the command prompt |
| 18:18.13 | b11d | dont type http://reload either |
| 18:18.15 | vAd0r | does that just reload asterisk or the whole thing like shutdown -r now |
| 18:18.21 | b11d | ugh |
| 18:18.22 | vAd0r | im not a moron |
| 18:18.23 | b11d | it reloads!!! |
| 18:18.23 | vAd0r | lol |
| 18:18.37 | b11d | ahh im just breaking your balls.. we all started sometime.. |
| 18:18.39 | swilliamson | or sip reload, to reload the sip config, just the sip config |
| 18:18.39 | vAd0r | im just windows man and novice linux |
| 18:18.46 | swilliamson | does that still work? |
| 18:18.56 | groogs | swilliamson: just dont' forget, the goal of freepbx (i'm assuming thats what you all mean by "web" in this conversation) is not to present every option in asterisk (since that's impossible), but just to provide an interface to configure a pbx. |
| 18:19.35 | swilliamson | yeah, sorta what I was saying, i agree with you |
| 18:19.49 | b11d | I just dont like web interfaces when it comes to configuring software, plain and simple. |
| 18:19.56 | b11d | personal choice, personal opinion. |
| 18:19.59 | swilliamson | b11d is just hardcore |
| 18:20.05 | b11d | thanks for the recognition! |
| 18:20.06 | b11d | :) |
| 18:20.07 | vAd0r | i hate the gay cisco web config pages |
| 18:20.11 | vAd0r | drives me crazy |
| 18:20.14 | b11d | yeah me too |
| 18:20.22 | swilliamson | conf t man |
| 18:20.28 | vAd0r | i can't never figure out how to do anyting |
| 18:20.28 | groogs | it's like comparing visual c++ and c++.. the visual editor can't write ALL the code that it's possible to write in c++, it just does a bunch of common/tedious stuff for you |
| 18:20.35 | b11d | pfft.. I only manipulate cisco config's with a magnet.. |
| 18:20.44 | b11d | hell, im not even using a screen right now.. |
| 18:20.47 | swilliamson | solar flares here |
| 18:20.48 | b11d | there is no monitor attached to this computer. |
| 18:20.57 | b11d | your responses are put out in a series of beeps |
| 18:21.14 | b11d | :) |
| 18:21.27 | b11d | TK.. you know Polycom phones well.. |
| 18:21.30 | groogs | psh, if you were really hardcore, you'd have electrodes connected to all your toes, and read one byte at a time |
| 18:21.31 | vAd0r | lol im using 4 screens |
| 18:21.34 | b11d | is it possible to change how the "reject" option works? |
| 18:21.34 | swilliamson | groogs: the good thing is that abstraction (like freepbx) opens up the product to many more people |
| 18:21.53 | b11d | i do not see anything in the SIP Admin guide.. :( |
| 18:21.58 | swilliamson | hardwire to brainstem |
| 18:22.06 | [TK]D-Fender | b11d: Yup. You can toggle it as "rejet", and "busy", and I think 1 other thing. |
| 18:22.14 | b11d | hrm.. do you know offhand where to set that? |
| 18:22.27 | b11d | like I said, I dont see anything in it's -sip.cfg or -phone.cfg files |
| 18:22.29 | [TK]D-Fender | b11d: For which you might be able to grab a return code to change your processing for. |
| 18:22.41 | [TK]D-Fender | b11d: not offhand. Can maybe check later. |
| 18:22.45 | b11d | thanks |
| 18:22.48 | b11d | i'll keep digging in the mean time |
| 18:22.54 | vAd0r | b11d is this a sip peer in sip.conf |
| 18:22.55 | vAd0r | http://pastebin.ca/285222 |
| 18:23.11 | b11d | christ you've got a lot of crap in there |
| 18:23.11 | b11d | :) |
| 18:23.22 | vAd0r | hey i just copied it |
| 18:23.29 | *** join/#asterisk mercestes (n=merceste@cpe-70-114-201-110.houston.res.rr.com) |
| 18:23.35 | b11d | im pretty sure the callerid statement is improperly formatted |
| 18:23.52 | b11d | hey merbanan |
| 18:23.53 | b11d | DOH |
| 18:23.54 | vAd0r | but that is a sip peer right like you were referring to |
| 18:23.56 | mercestes | is there a good freenode FoP resource? |
| 18:24.00 | b11d | i've got a big shoulder here mercestes |
| 18:24.09 | b11d | its "like" what i meant.. yes.. |
| 18:24.14 | vAd0r | ok |
| 18:24.18 | vAd0r | then sip reload |
| 18:24.22 | vAd0r | roger that |
| 18:24.32 | b11d | just remember that "tammari" is now going to have to be on your vg224.. |
| 18:24.36 | *** join/#asterisk h1 (n=fakhir@ool-44c69453.dyn.optonline.net) |
| 18:24.39 | b11d | you cant name a number a name when it comes to analog phones.. |
| 18:24.40 | vAd0r | lol |
| 18:24.53 | vAd0r | i should be able to setup the vg200 as my voip provider |
| 18:24.55 | [TK]D-Fender | b11d: "call.rejectBusyOnDnd" |
| 18:24.57 | b11d | your [tammari] needs to be changed to the ACTUAL number thatg will be on the vg200 |
| 18:25.00 | mercestes | CallerID is "name" <number> in sip.conf and can be set via Set(CallerID(Name)) and Set(CallerID(Number)) later. and not all carriers transmit both name and number. |
| 18:25.03 | vAd0r | like i do w/ trunks |
| 18:25.09 | b11d | TK.. I saw that.. it didnt seem like it was the right option |
| 18:25.16 | b11d | i'll mess with it though |
| 18:25.37 | b11d | and name & number ONLY works on PRI.. from what I've been told |
| 18:25.46 | b11d | aside from a pure-voip environment that is |
| 18:26.44 | [TK]D-Fender | b11d : have you done some calls and output the SIP reject reason, and the DIALSTATUS? |
| 18:26.47 | b11d | no.. performing that now.. |
| 18:26.56 | mercestes | PRI, with ISDN signalling. |
| 18:27.00 | mercestes | according to TWTC. |
| 18:27.02 | b11d | I thought PRI implied ISDN |
| 18:27.07 | mercestes | it does. |
| 18:27.08 | *** join/#asterisk tehhh (n=pn@client-82-199-205-198.speedy.sellinet.net) |
| 18:27.18 | b11d | then why say PRI, with ISDN signalling? |
| 18:27.19 | b11d | :) |
| 18:27.30 | mercestes | because I'm redundant and repetitivce. |
| 18:27.45 | [TK]D-Fender | b11d: Should throw back a 606 IIRC |
| 18:27.48 | ai-a[work] | how do i add asterisk as a service to start on boot up ? |
| 18:27.49 | b11d | theres nothing wrong with that.. |
| 18:28.02 | b11d | yeah I saw a thread on it returning 6xx messages.. |
| 18:28.45 | b11d | aparently thats not standard ? |
| 18:28.45 | *** join/#asterisk goodcat (n=torgeir@194.54.103.22) |
| 18:28.45 | [TK]D-Fender | ai-a[work]: Depends on your distro and where you want it to start in order |
| 18:28.45 | b11d | ai.. depends on your OS.. |
| 18:28.45 | ai-a[work] | fc6 |
| 18:28.45 | b11d | well thanks TK.. I'll let you know what I end up with :P |
| 18:28.45 | ai-a[work] | just assumed it would install a service. |
| 18:28.46 | ai-a[work] | can write one i guess. |
| 18:28.54 | b11d | haha new schoolers.. |
| 18:28.59 | mercestes | SO...any good FoP resources? I read the crap at asternic and it no worky..:( |
| 18:29.00 | [TK]D-Fender | ai-a[work]: "make config" should install SysV inits for you |
| 18:29.17 | b11d | FoP? |
| 18:29.28 | ai-a[work] | [TK]D-Fender: tahnks. |
| 18:29.32 | mercestes | Flash Operator Panel. |
| 18:29.38 | b11d | oh |
| 18:29.44 | vAd0r | reload does not work on my cli |
| 18:29.53 | b11d | well that doesnt make any sense |
| 18:29.58 | b11d | are you sure you're in the asterisk CLI? |
| 18:30.12 | vAd0r | im at a linux bash |
| 18:30.12 | b11d | ok |
| 18:30.13 | vAd0r | is that what you mean |
| 18:30.13 | b11d | asterisk -r |
| 18:30.13 | b11d | then type reload |
| 18:30.24 | vAd0r | i see |
| 18:30.26 | goodcat | has anyone gotten relatime asterisk->odbc->mysql to work? |
| 18:30.27 | b11d | heh.. linux parties suck |
| 18:30.30 | mercestes | you should specify "asterisk cli" since cli is ambiguous. |
| 18:30.36 | mercestes | lol Linux ftw! |
| 18:30.45 | b11d | well we're in #asterisk and for christ sakes it says CLI right on the prompt.. |
| 18:30.53 | wiljacket | My penny-pinching client needs about 2 dozen phones, and likes the Grandstream GPX-2000 -- does anybody have any experience with those units? |
| 18:30.54 | b11d | it's in context.. |
| 18:30.58 | ai-a[work] | k, asterisk-gui isnt servicing on port 8088, make checkconfig comes back fine,, basicly copied its readme examples. |
| 18:31.01 | b11d | yeah.. they suck.. dont get them |
| 18:31.10 | goodcat | wiljacket: gxp 2000 sucks... |
| 18:31.11 | mercestes | wiljacket: They suck. |
| 18:31.15 | goodcat | grandstream sucks. period. |
| 18:31.21 | b11d | saving the money is NOT a good idea in this case.. |
| 18:31.22 | vAd0r | should i stay at 1.2.3 |
| 18:31.25 | vAd0r | or upgrade ? |
| 18:31.26 | b11d | spend a LITTLE more and get Polycom's |
| 18:31.33 | b11d | 1.2.3 ?? 1.2.14 is out |
| 18:31.34 | goodcat | or thomson |
| 18:31.45 | wiljacket | Oh thank you guys, I knew they looked awful :) |
| 18:31.46 | goodcat | thomson st-2030 |
| 18:31.54 | b11d | yeah they really arent good phones.. |
| 18:31.54 | goodcat | brilliant phone |
| 18:32.04 | vAd0r | asterisk cli shows me at |
| 18:32.05 | b11d | I dont think i've heard of anyone who likes the Grandstream phones.. |
| 18:32.06 | vAd0r | 1.2.13 |
| 18:32.10 | b11d | ok cool |
| 18:32.13 | b11d | you're good then |
| 18:32.17 | b11d | 1.2.14 IS out, but 1.2.13 is good |
| 18:32.21 | vAd0r | what is the 2.0 something |
| 18:32.28 | vAd0r | is that trix upgrade or something? |
| 18:32.35 | b11d | no idea |
| 18:32.38 | b11d | we dont talk about trixbox here |
| 18:32.39 | b11d | :| |
| 18:32.40 | vAd0r | lol |
| 18:32.40 | mercestes | I give them away as Xmas presents when I run out of fruitcake. |
| 18:32.42 | b11d | :) |
| 18:32.54 | vAd0r | so to check my version asterisk -r |
| 18:33.04 | b11d | asterisk -r will bring you to the asterisk CLI |
| 18:33.06 | mercestes | asterisk -V |
| 18:33.09 | goodcat | has anyone gotten relatime asterisk->odbc->mysql to work? |
| 18:33.13 | b11d | it should say like " host*CLI>" |
| 18:33.16 | b11d | type 'reload' there |
| 18:33.19 | mercestes | you need to remove that trixbox crap and get "real" asterisk however. |
| 18:33.42 | vAd0r | can i just leave it on |
| 18:33.51 | vAd0r | what will it matter if i do everything for cli |
| 18:33.55 | vAd0r | from cli |
| 18:33.56 | vAd0r | sry |
| 18:33.57 | goodcat | seems that asterisk realtime doesn't bother to read the table |
| 18:34.09 | goodcat | sip show peers returns 0 users |
| 18:34.10 | swilliamson | goodcat: I have been using the new realtime ldap, unixODBC lacks docs in my oppinion |
| 18:34.10 | b11d | vAd0r.. |
| 18:34.18 | b11d | you CANT do everything from the CLI |
| 18:34.20 | b11d | its not possible. |
| 18:34.24 | b11d | well.. in a way it is.. but not really |
| 18:34.31 | mercestes | goodcat: they won't show up until they reg |
| 18:34.33 | vAd0r | lol you guys are confusing me |
| 18:34.37 | *** join/#asterisk olinux (n=olinux@starbucks.wellspublishing.net) |
| 18:34.41 | goodcat | mercestes: they can't reg either |
| 18:34.42 | vAd0r | mercestes say take trixbox off |
| 18:34.43 | b11d | well you need to edit your config files.. |
| 18:34.43 | mercestes | try sip show peer <peer> |
| 18:34.49 | b11d | trixbox sucks |
| 18:34.53 | vAd0r | you say i can't do everything from cli |
| 18:34.54 | goodcat | mercestes: returns 0 peers |
| 18:34.56 | b11d | wait.. you're using trixbox vAd0r? |
| 18:35.00 | vAd0r | yes |
| 18:35.03 | b11d | ugh |
| 18:35.04 | b11d | bye |
| 18:35.07 | b11d | go to #freepbx then |
| 18:35.08 | goodcat | mercestes: and the odbc connection is ok |
| 18:35.18 | *** join/#asterisk niter3_ (n=niter3@dhcp-0-18-39-71-48-17.cpe.mountaincable.net) |
| 18:35.20 | vAd0r | what the heck difference does it make |
| 18:35.25 | b11d | a big difference |
| 18:35.42 | mercestes | goodcat turn debug on...it could be a db error |
| 18:35.42 | vAd0r | i thought freepbx was just a front end |
| 18:35.42 | b11d | #freepbx is the trixbox support channel |
| 18:35.45 | tm | nibbler_de: huhu nochmal? |
| 18:35.45 | b11d | its specifically for trixbox & openpbx |
| 18:35.58 | niter3_ | Is there a site where I can find ideas that some people have done with their asterisk systems? |
| 18:36.10 | vAd0r | but it has asterisk |
| 18:36.19 | b11d | it doesnt matter |
| 18:36.22 | tm | how i can transfer a call with music on hold? |
| 18:36.23 | b11d | #freepbx is the trixbox support channel |
| 18:36.25 | b11d | not #asterisk |
| 18:36.32 | b11d | #asterisk is for vanialla asterisk |
| 18:36.37 | b11d | vanilla |
| 18:36.37 | b11d | :P |
| 18:36.38 | swilliamson | freepbx configs are they come with trixbox are like spagetti, extension.conf -> extesions_custom.conf where is that context? |
| 18:36.45 | tm | exten => _X.,5,Dial(misdn/1/${EXTEN}) |
| 18:36.46 | tm | i have |
| 18:36.52 | *** join/#asterisk sukimono_ (n=sukimono@202.164.181.222) |
| 18:36.58 | bkruse | http://asterisknow.org |
| 18:37.05 | niter3_ | http://asterisknow.org |
| 18:37.12 | bkruse | yay gui |
| 18:37.15 | b11d | http://asterisknow.org |
| 18:37.18 | mercestes | vAd0r: Let me give you an example. Gentoo installers. We have to download and compile our stage three, manually setup our modules and our kernel support options, compile the kernel, set our hostname, our locale, our localtime, our network, etc. all in text. |
| 18:37.29 | bkruse | b11d: echoing asterisknow.org is the new cool thing to do |
| 18:37.33 | b11d | right on |
| 18:37.34 | b11d | im down |
| 18:37.38 | Younss | thank you all for your help |
| 18:37.45 | bkruse | b11d: sweet |
| 18:37.45 | bkruse | :] |
| 18:37.45 | mercestes | vAd0r: Then they come out with this graphical installer crap where it acts like a palm reader, asks you a few questiosn about your birthday and your sign and stuff, and then automagically edits everything for you. |
| 18:37.47 | b11d | ftw!!! |
| 18:38.03 | bkruse | b11d: exactly |
| 18:38.06 | bkruse | freepbx ftl |
| 18:38.12 | bkruse | mercestes: ha, gui nubs ;] |
| 18:38.15 | b11d | :) |
| 18:38.16 | mercestes | vAd0r: Then a problem happens....and they go to gentoo, and ask what's wrong, and we ask them "what did you put in this file?" They don't know..because they didn't. we don't know what Trixbox is doing to you. Neither do you. |
| 18:38.29 | *** join/#asterisk Shadower (n=Shadower@vc-196-207-32-235.3g.vodacom.co.za) |
| 18:38.30 | niter3_ | Is there a site dedicated for projects that people have done with Asterisk (eg: alarm reminder) |
| 18:38.34 | niter3_ | ? |
| 18:38.36 | Shadower | hi all |
| 18:38.41 | goodcat | mercestes: debug just says registered database handle 'mysql2' dsn->[MySQL-asterisk] |
| 18:38.41 | vAd0r | so you guys use asterisknow version 1.4.0? |
| 18:38.42 | b11d | i just finished porting asterisk to CP/M -- anyone want it? |
| 18:38.49 | niter3_ | bkruse: Nothing for Nagios? You can program one up |
| 18:38.52 | *** join/#asterisk _BOBWEEVER (n=chatzill@adsl-150-0-187.aby.bellsouth.net) |
| 18:38.58 | goodcat | and then res_odbc: Connected to mysql2 [MySQL-asterisk] |
| 18:39.05 | swilliamson | i copy stuff from the gui ast distros to do stuff |
| 18:39.13 | bkruse | niter3_: how many people really use nagios? is it work doing? |
| 18:39.24 | bkruse | worth* |
| 18:39.25 | tzafrir | mercestes, but many others give up in the long and error-prone process of compiling and building and such |
| 18:39.32 | niter3_ | bkruse: Is what worth doing? |
| 18:39.34 | mercestes | goodcat: Did you setup your res_mysql.conf or your res_odbc.conf to point those configs to the proper databases? |
| 18:39.37 | tzafrir | Which is easy to automate for the common case |
| 18:39.41 | bkruse | building a plugin/script for nagios |
| 18:39.58 | niter3_ | bkruse: It's not difficult if that's what you mean. It's only worth doing depending on your situation. |
| 18:40.06 | file | bkruse: moo |
| 18:40.12 | bkruse | file: :D |
| 18:40.13 | mercestes | tzafrir: Gentoo community is *huge*. I don't see many failures there. Same with *./ |
| 18:40.22 | b11d | [TK]D-Fender.. no dice man.. |
| 18:40.22 | tm | file: can u help me? |
| 18:40.24 | bkruse | file: i passed my physics exam !!!!1!11!111one1! |
| 18:40.25 | swilliamson | b11d: can you port ztdummy to openbsd? |
| 18:40.30 | b11d | sure |
| 18:40.30 | swilliamson | please |
| 18:40.36 | tzafrir | mercestes, the answer is simple: you don't see them. They have not made it inside. |
| 18:40.39 | b11d | I heart openbsd |
| 18:40.45 | tm | file: how i can transfer with misdn with music on hold? |
| 18:40.48 | file | bkruse: zomg |
| 18:40.51 | tm | file: exten => _X.,5,Dial(misdn/1/${EXTEN}) <<this is my row |
| 18:40.52 | bkruse | :] |
| 18:40.56 | goodcat | mercestes: sipusers => odbc,asterisk,sip_buddies |
| 18:41.05 | goodcat | mercestes: same for sippeers |
| 18:41.06 | mercestes | tzafrir: ??? |
| 18:41.21 | swilliamson | b11d: and sun4u? yeah it needs to run on sun4u |
| 18:41.28 | mercestes | goodcat: have you tried connecting to and accessing that database and tables with the username and password you setup in odbc? |
| 18:41.33 | b11d | fuck.. that'll only take five mins. |
| 18:41.37 | b11d | what am I supposed to do with the other 55 mins? |
| 18:41.46 | b11d | thats how hardcore I roll |
| 18:41.47 | bkruse | talk in irc |
| 18:41.52 | swilliamson | just bill be for 6 mins and we'll be good |
| 18:41.56 | b11d | cool |
| 18:41.57 | goodcat | mercestes: yes |
| 18:42.01 | bkruse | b11d: i love how people say "back port" and everyone is like ZOMG! |
| 18:42.07 | b11d | lol |
| 18:42.13 | mercestes | b11d: omg? really? OpenBSD is so mean. |
| 18:42.18 | goodcat | mercestes: table created using http://www.voip-info.org/wiki/view/Asterisk+RealTime+Sip |
| 18:42.19 | bkruse | openbsd is so awesome |
| 18:42.22 | b11d | yeah but thats what I love about it |
| 18:42.37 | b11d | Theo can whip me any day |
| 18:42.54 | mercestes | b11d: ....oh...true. |
| 18:42.59 | niter3_ | Hate to ask again, but is there any dedicated sites for projects people have done with their asterisk boxes? |
| 18:43.07 | b11d | you can whip me too mercestes.. if you ask nice |
| 18:43.16 | swilliamson | yup. sun4u , net install , remote managment card in a closet that will be my legacy machine... |
| 18:43.20 | bkruse | niter3_: http://voip-info.org ? |
| 18:43.22 | bkruse | :D |
| 18:43.24 | b11d | :) |
| 18:43.29 | mercestes | niter3_: I think wiki.asterisk.org is probably your closest equivalent. There is also the netdomination VoIP overlay laying around somewhere. |
| 18:43.37 | *** part/#asterisk parag (n=Administ@dxb-b123242.alshamil.net.ae) |
| 18:43.43 | mercestes | niter3_: Ok, ignore my link. Use the real one bkruse gave you. |
| 18:43.44 | swilliamson | niter3_: I will get jumped on but nerdvittles has some flashy stuff, if you can get it to work |
| 18:43.57 | mercestes | b11d: Can I whip you and make you my OpenBSD slave? >.> |
| 18:43.59 | niter3_ | :) |
| 18:44.02 | mercestes | or should I ask in #openbsd? |
| 18:44.12 | file | bkruse: not bad |
| 18:44.13 | swilliamson | they are fans of wget url -C | sh instructions though, |
| 18:44.16 | file | it's cold out, but meh |
| 18:44.29 | swilliamson | hey download some shell script from my website and run it as root. |
| 18:44.46 | b11d | yes |
| 18:44.46 | b11d | you can |
| 18:44.50 | bkruse | swilliamson: ok! |
| 18:45.08 | bkruse | what does #now mapping local port to my box in nigeria mean? |
| 18:45.16 | swilliamson | just like the old rm -fr ... days |
| 18:45.18 | goodcat | mercestes: odbc show says everything's ok.. |
| 18:45.21 | bkruse | mercestes: one day voip-info.org wont be the only central place for information :D |
| 18:45.27 | b11d | one day.. |
| 18:45.32 | b11d | we can only keep the dream alive.. |
| 18:45.42 | mercestes | bkruse: And then it will die...and I will be out of a job. |
| 18:45.47 | b11d | i dont mind voip-info.org -- it just needs a good cleaning & organizing |
| 18:46.08 | tm | file: huhu? |
| 18:46.08 | mercestes | goodcat: Hrm. :/ Sip show peers showing nothing is normal. What's failing is it seems to not be seeding peers. |
| 18:46.13 | bkruse | mercestes: nono, voip-info.org isnt going anywhere, im just saying there will be more than one place |
| 18:46.21 | bkruse | i assume? |
| 18:46.23 | mercestes | goodcat: I'm going to make the blind assumption that you do have data in your sip_buddies table or whatever it was called. |
| 18:46.27 | Shadower | how would I go about transfreing a analog incoming call (FXO) to a softphone? |
| 18:46.37 | *** join/#asterisk apardo_ (n=apardo@87.217.144.3) |
| 18:46.40 | goodcat | mercestes: duh. yes, I do :) |
| 18:47.10 | goodcat | mercestes: but I only filled name,secret,host,context...that should be enough? |
| 18:47.22 | mercestes | goodcat: Well, here is what should happen. * should successfully connect to said database.table. A peer tries to come in and register. * will fail to see it in sip.conf (which will still work, btw), then * will try to find that peer in the odbc sip table. |
| 18:47.30 | b11d | actually.. voip-info.org really does have a fuckload of info on it.. |
| 18:47.31 | goodcat | mercestes: and type of course |
| 18:47.36 | b11d | I wonder how much space that takes up |
| 18:47.38 | *** join/#asterisk MatBoy (n=MatBoy@wiljewelwetenhe.xs4all.nl) |
| 18:47.45 | MatBoy | Hi guys ! |
| 18:47.49 | b11d | HEY!! |
| 18:47.53 | b11d | whats up MatBoy!!? |
| 18:47.58 | MatBoy | hey ! |
| 18:47.59 | mercestes | goodcat: Then it will do what it calls "Seeding" which will load that information into it's internal database, where it will stay until you shut it down, or restart it, or do a sip reload...or it crashes. It wont' show in sip show peers until it's seeded. |
| 18:48.00 | b11d | It's just like old times!! |
| 18:48.18 | PupenoR | How do I see the queries performed by Asterisk when retrieving realtime values ? |
| 18:48.30 | goodcat | mercestes: ok |
| 18:48.31 | MatBoy | I was wondering if it's possible in some way to route real telephonenumbers with asterisk |
| 18:48.36 | b11d | fuck.. I really hate this.. I've got a song stuck in my head and all that repeats is the break.. |
| 18:48.46 | *** join/#asterisk mhnoyes_ (n=mhnoyes@dialup-4.246.6.61.Dial1.SanJose1.Level3.net) |
| 18:48.46 | b11d | I cant remember ANY lyrics except "sorrow" |
| 18:48.52 | mercestes | goodcat: Now, what could be happening is your phones are failing to reach your server. but, if you had a db entry for say, peer bob, and you did a sip show peer bob in the cli, it *should* force bob to be seeded and show it's information on it, and show it as offline. |
| 18:48.59 | b11d | ...' |
| 18:49.01 | b11d | oops |
| 18:49.12 | b11d | hehe |
| 18:49.21 | *** join/#asterisk nand2 (i=94f54f63@gateway/web/cgi-irc/ircatwork.com/x-b06bc7698457aafd) |
| 18:49.22 | MatBoy | better now ? |
| 18:49.23 | b11d | I fear the only way its going out is to overplay the song, which I cant do, because I dont knwo what it is |
| 18:49.28 | swilliamson | ack, who knows how to do apt-get's in the rpm world |
| 18:49.28 | goodcat | mercestes: what I don't get is that I seem to have to specify DATABASE=asterisk in odbc.ini |
| 18:49.29 | mercestes | goodcat: So if sip show peer bob isn't working. Then * can't see your DB. Are you sure you have debug on? Compiled with the debug flag, have debug set to 37257, and you have debug listed in the cli for logger.conf? |
| 18:49.41 | *** join/#asterisk arcy (n=arcanum@ppp40-9.adsl.forthnet.gr) |
| 18:49.42 | niter3_ | bkruse: http://www.mataluis.com//index.php?option=com_content&task=view&id=39&Itemid=1 |
| 18:49.53 | MatBoy | but are here persons that use asterisk as a telephone provider ? |
| 18:50.00 | b11d | yes |
| 18:50.04 | swilliamson | yes |
| 18:50.05 | b11d | im not one of them |
| 18:50.06 | b11d | but yes |
| 18:50.10 | swilliamson | have one in hong kong |
| 18:50.14 | goodcat | mercestes: I'll try sip show peer <peer> |
| 18:50.16 | swilliamson | and germany |
| 18:50.19 | b11d | let me be your hongkong pothead.. |
| 18:50.22 | b11d | send me there |
| 18:50.25 | mercestes | ini? omg? your in windoze? |
| 18:50.34 | MatBoy | how is it done with number-porting when you have to route the number from another telco ? |
| 18:50.36 | goodcat | mercestes: nope, /etc/odbc.ini |
| 18:50.41 | mercestes | oh. |
| 18:50.53 | mercestes | never setup odbc in linux. never seen an ini in linux either. =/ |
| 18:50.59 | b11d | I need you all to call me and play clips of the song you think is stuck in my head |
| 18:51.10 | goodcat | me neither, but odbc IS (or was) windows-stuff |
| 18:51.11 | b11d | and then i'll say if its right or not.. |
| 18:51.12 | b11d | sounds good |
| 18:51.13 | mercestes | Hey, if I did all my flash operator panel setup crap and changed my manager, etc. would I Hvae to restart asterisk to make it work?? |
| 18:51.13 | b11d | :P |
| 18:51.23 | mercestes | goodcat: good point. |
| 18:51.28 | b11d | did you edit any * config files? |
| 18:51.32 | b11d | or just fop stuff? |
| 18:51.34 | swilliamson | +493055555 |
| 18:51.39 | mercestes | manager.conf |
| 18:51.43 | b11d | then reload manager |
| 18:51.43 | mercestes | but I did a reload app_manager.conf |
| 18:51.45 | b11d | ahh |
| 18:51.48 | mercestes | but my fop screen is blank. |
| 18:51.49 | goodcat | mercestes: asterisk shows "peer 111 not found" |
| 18:51.51 | [TK]D-Fender | mercestes: UnixODBC is pretty easy to set up, even from source. |
| 18:51.52 | b11d | you might need to reload.. |
| 18:52.03 | swilliamson | get yourself a job in berlin you call that number |
| 18:52.08 | *** join/#asterisk santiago (n=santiago@debian/developer/santiago) |
| 18:52.08 | mercestes | I did several reloads...my screen is all blank...*cries* I need a FoP guru |
| 18:52.16 | b11d | [TK]D-Fender.. no dice on that BusyOnDND option |
| 18:52.31 | in-pt | mercestes: i had setup fop |
| 18:52.38 | in-pt | whats your problem |
| 18:52.54 | *** part/#asterisk santiago (n=santiago@debian/developer/santiago) |
| 18:52.57 | mercestes | in-pt: Well, my screen is all blank. |
| 18:52.58 | *** join/#asterisk nand2 (i=94f54f63@gateway/web/cgi-irc/ircatwork.com/x-2a815308b0135bca) |
| 18:52.59 | [TK]D-Fender | b11d: Like I said, I'm betting you'll need to dump the SIP reason and teh DIALSTATUS to be sure (if eve) as to why the call wassn't answered |
| 18:53.24 | *** join/#asterisk keyhack (n=keyhack@68.236.93.224) |
| 18:53.25 | mercestes | lol. And before it was showing three big squares, pink, green, and some other color, with some Wcgt535rt text in them...but..now I'm up to just a blank screen. |
| 18:53.40 | mercestes | goodcat: * is likely not seeing the db then. |
| 18:54.09 | mercestes | goodcat: Did you see all my steps to turn debug on in *? |
| 18:54.15 | b11d | TK.. i think we're not on the same level here.. |
| 18:54.20 | b11d | the problem isnt with calls not being answered |
| 18:54.33 | b11d | its when I hit the "reject" button, its trying to shove the call to a voicemail box that doesnt exist.. |
| 18:54.51 | in-pt | mercestes: which version of fop you are running |
| 18:54.52 | b11d | and thats "reject" when an inbound call is currently being received.. |
| 18:55.09 | mercestes | I dl'd it from asternic like 3 days ago. |
| 18:55.15 | b11d | 3 days? thats tomorrow.. |
| 18:55.49 | in-pt | ok so its the latest.. well i was also having that problem ..but i solved it long time back |
| 18:55.55 | in-pt | i am thinking what i did for that |
| 18:56.25 | mercestes | in-pt: Awesome. I believe in you! You can do it! |
| 18:56.30 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 18:56.36 | keyhack | Does anyone know of a VoIP provider that offers unlimited phone calling for non-residential? heh |
| 18:56.36 | b11d | You CAN do it! |
| 18:56.46 | goodcat | mercestes: allright, I have debug=>debug in logger.conf |
| 18:56.49 | b11d | yeah.. Uncle Franks Illegal Telco |
| 18:56.54 | b11d | www.ufrankstelco.org |
| 18:57.00 | *** join/#asterisk tsurko_ (n=tsurko@vpn-pppoe-213-240-242-237.megalan.bg) |
| 18:57.01 | mercestes | goodcat: Nah, put it under CLI. :) |
| 18:57.15 | mercestes | goodcat: It should be showing you info down to the individual mysql queries. |
| 18:57.47 | goodcat | mercestes: sorry..I don't follow |
| 18:58.09 | goodcat | mercestes: like asterisk -cvvvvvvvvvvvv ? |
| 18:58.17 | b11d | ok TK.. |
| 18:58.18 | mercestes | logger.conf: console => notice,warning,error,debug |
| 18:58.18 | swilliamson | keyhack: < 1ct a min is pretty much free |
| 18:58.20 | b11d | you can stop laughing at me |
| 18:58.20 | in-pt | mercestes: whats the flash dir you had setup in op_server.cfg file |
| 18:58.24 | b11d | I fixed it .. |
| 18:58.27 | mercestes | instead of the default console => notice,warning,error |
| 18:58.28 | b11d | god damn error in extensions.conf |
| 18:58.32 | goodcat | I see |
| 18:58.35 | b11d | not with the poly at all. (big surprise) |
| 18:58.37 | keyhack | swilliamson: who offers < 1 ct/min? |
| 18:58.54 | *** join/#asterisk juanjoc (n=juanjoc@201.216.212.113) |
| 18:59.04 | [TK]D-Fender | b11d: Illegal VM box? Sounds like a dialplan error then, if not a voicemail.conf one |
| 18:59.04 | docelmo | I offer term but not < 1c/min w/o volume |
| 18:59.19 | mercestes | in-pt: flash_dir=/var/www/localhost/htdocs/panel |
| 18:59.40 | *** part/#asterisk dasenjo (n=dasenjo@63.245.86.215) |
| 18:59.54 | b11d | TK.. read my messages above |
| 18:59.54 | swilliamson | can be had from regional telco's here in canada without volume, check voip-info.org... service providers |
| 19:00.00 | b11d | it WAS the extensions.conf |
| 19:00.02 | b11d | all along :) |
| 19:00.47 | in-pt | but in panel folder do you have operator_panel.swf file |
| 19:00.47 | b11d | thanks for your help, as usual.. |
| 19:00.47 | mercestes | Yea. |
| 19:00.48 | keyhack | swilliamson: yeah I was looking at the few they list on there |
| 19:00.50 | b11d | hey Mercestes.. im moving like 3 blocks from you |
| 19:01.00 | mercestes | b11d: ?? In tx? |
| 19:01.04 | b11d | haha yeah |
| 19:01.08 | b11d | ok no.. |
| 19:01.11 | b11d | im not.. |
| 19:01.12 | mercestes | b11d: ....aww.... |
| 19:01.14 | b11d | sorry man. |
| 19:01.19 | mercestes | I wanna cry now |
| 19:01.19 | goodcat | mercestes: right, done.. console still normal |
| 19:01.26 | b11d | i'd stalk you if I had the money right now, but I dont.. |
| 19:01.29 | mercestes | I wanted to whip you and make you my open bsd slave. |
| 19:01.36 | b11d | you will be able to sooner or later :0 |
| 19:01.39 | mercestes | goodcat: type set debug 99 |
| 19:01.58 | mercestes | in-pt: Let me try something just in case. |
| 19:02.02 | docelmo | You know the debug doesnt go > 5 right? |
| 19:02.03 | goodcat | mercestes: ok |
| 19:02.12 | b11d | debug doesnt go > 5?? |
| 19:02.13 | goodcat | mercestes: and then reload? |
| 19:02.18 | swilliamson | keyhack: good thing about voip term, you can buy from anywhere, lag and voice quality suffer tho |
| 19:02.33 | keyhack | swilliamson: right |
| 19:02.34 | docelmo | The most verbose debug cant get is 5.. Anything higer is pointless |
| 19:02.35 | mercestes | in-pt: Yea, ok, I wanted to make sure I could view the demo...lol I can. |
| 19:02.37 | b11d | I can set "set debug 10" and it says it does.. |
| 19:02.45 | b11d | oh |
| 19:02.49 | b11d | now i see I can enter any value |
| 19:02.49 | b11d | :| |
| 19:02.50 | *** join/#asterisk irq (n=dan@wsip-70-168-52-194.sd.sd.cox.net) |
| 19:02.56 | mercestes | b11d: I know, but it makes me happy. |
| 19:02.58 | b11d | irq.. to save the party.. |
| 19:03.03 | mercestes | goodcat: shouldn't have to reload. |
| 19:03.06 | keyhack | swilliamson: I was thinking of a software service that would place outbound calls but, even still, 1ct/min is expensive investment |
| 19:03.07 | mercestes | goodcat: set verbose 99 |
| 19:03.11 | docelmo | ahh well back to setting shit up.. |
| 19:03.21 | in-pt | mercestes: i dont remembers now :( what i did |
| 19:03.43 | mercestes | in-pt: Could it have been a cold restart maybe? |
| 19:04.43 | goodcat | mercestes: chan_sip.c:6648 register_verify: SIP REGISTER attempt failed for (null) : Bad digest user |
| 19:04.51 | goodcat | that's about the only extra info I get |
| 19:05.00 | mercestes | keyhack: Your best bet for unlimited local termination would be fxs to analog lines. But....about hte time you start terminating 70k minutes to your $20 a month AT&T line, questions will be posed I am certain. |
| 19:05.17 | mercestes | goodcat: Hrm. SO yea, not seeing any information. |
| 19:05.23 | keyhack | mercestes: hahaha |
| 19:05.30 | mercestes | goodcat: do a sip show peer 111 and it should show you a SQL query. |
| 19:05.49 | swilliamson | keyhack: oh, you one of those dudes who builds those predictive outbound dialer systems that call me all the f'n time? |
| 19:06.01 | goodcat | mercestes: it doesn't..it just says "peer 111 not found" |
| 19:06.04 | goodcat | no queries |
| 19:06.11 | swilliamson | sometimes they don't even trasfer the call to an agent... |
| 19:06.26 | keyhack | swilliamson: haha no, its a free service people can sign up for, to receive notifications |
| 19:06.29 | mercestes | goodcat: :/ Ok, let me ask some philosophical background questions. Why are you setting up realtime for sip.conf?? |
| 19:06.40 | b11d | maybe he wants both |
| 19:06.48 | b11d | or maybe I SHOULD SHUT UP |
| 19:06.51 | goodcat | mercestes: to have realtime sip peers? |
| 19:06.54 | *** join/#asterisk shido6 (n=shido6@d221-68-200.commercial.cgocable.net) |
| 19:07.05 | mercestes | goodcat: to what end? |
| 19:07.20 | goodcat | mercestes: to be able to store peers in mysql db |
| 19:07.28 | swilliamson | keyhack: oh, well if it is a big investment, than commit to a volume and you will get a better deal than even that |
| 19:07.48 | goodcat | without using the shoddy "config file in a database" workaround for 1.07 |
| 19:07.49 | keyhack | swilliamson: what do you mean? I'm new to phone networks |
| 19:07.52 | mercestes | goodcat: real time sip peers is in general, bad, troublesome, problematic, and buggy. The only real benefit to it is to allow users to modify their own dialplan via a GUI web interface. |
| 19:07.56 | swilliamson | PSTN, costs. hell internet costs... how much does 120k fibre run cost to install? |
| 19:08.43 | goodcat | mercestes: well..it's obviously buggy :p |
| 19:08.45 | swilliamson | keyhack: there are people in this channel that will sell you call termination (outbound) and if you contract with them to buy x mins a month, they give you a rate of y |
| 19:08.52 | mercestes | goodcat: sip.conf is a far better solution. The primary headache is that a sip reload in sip realtime is catastrophic to a large scale system, because it shuts all phones down for atleast 1 phone call, or up to one hour, whichever happens first. |
| 19:08.53 | goodcat | mercestes: or at least lacking docs |
| 19:09.21 | mercestes | goodcat: Now, if you wanna keep bugging with it we can...lol. but, I think you'll be far more satisfied with sip.conf because it's changes and effects are immediate and predictable. |
| 19:09.27 | goodcat | mercestes: what? reload doesn't cut active channels |
| 19:09.33 | mercestes | there is nothing "realtime" about it. |
| 19:09.42 | keyhack | swilliamson: hmm, 70k+ mins @ $30 a month? :-p |
| 19:09.51 | mercestes | goodcat: it doesn't cut active channels, but it does remove every phones registration which will deny it atleast one call. |
| 19:09.53 | swilliamson | keyhack: but why compete with yahoo with < $1bn |
| 19:10.06 | mercestes | goodcat: the good news is, when it fails that first call it will seed the peer and will accept the second call. |
| 19:10.18 | swilliamson | keyhack: because violates TOS, they pay rates/min too |
| 19:10.20 | mercestes | goodcat: now, why it doesn't seed first them pass the call I don't know. I'm not a developer. |
| 19:10.23 | goodcat | mercestes: we could try right now.. just a moment |
| 19:10.28 | b11d | yeah.. dont fuck with The Original Series |
| 19:10.44 | goodcat | mercestes: I have a phone right here, registered |
| 19:10.57 | swilliamson | keyhack: sounds like you are trying a bit of arbitrage |
| 19:11.20 | goodcat | mercestes: doing reload... |
| 19:11.23 | goodcat | mercestes: can still cal |
| 19:11.23 | goodcat | l |
| 19:11.28 | keyhack | swilliamson: just trying to see how cost-effective it is to give it away for free when it costs so much to run the service, heh |
| 19:11.31 | mercestes | goodcat: It can still call out...you can't call it tho. |
| 19:11.40 | mercestes | goodcat; ...is that one phone registered via sip realtime? |
| 19:11.46 | goodcat | goodcat: no |
| 19:11.50 | goodcat | oh |
| 19:11.53 | swilliamson | keyhack: it's not free, they take a risk that you will not use it that much. |
| 19:11.56 | goodcat | I see |
| 19:12.00 | mercestes | goodcat: oh, then it won't be effected by that particular feature. |
| 19:12.04 | goodcat | I see |
| 19:12.12 | mercestes | sip.conf still works, and still is loaded immediately at sip reload. |
| 19:12.16 | swilliamson | read about broadcomm on voxilla forums |
| 19:12.18 | goodcat | thought you were talking about sip registrations in general |
| 19:12.21 | mercestes | sip.conf also overrides yoru database. |
| 19:12.26 | keyhack | swilliamson: I am talking about me here, I'm giving my service away for free to my customers and paying per minute to call them, when I'm not making that much money out of it |
| 19:12.28 | mercestes | nah, just the ones in SQL. |
| 19:12.40 | goodcat | ok |
| 19:13.20 | swilliamson | keyhack: then why do it |
| 19:13.25 | mercestes | keyhack: Most of the ones giving the service away for free have some alterior motive/service that their free service is promoting. |
| 19:13.54 | *** join/#asterisk DaeJeon-Newbie (i=Singh@124.62.150.38) |
| 19:13.56 | keyhack | because there is potential to make money, whether or not that in the end means i profit is a different story lol |
| 19:14.09 | DaeJeon-Newbie | hello asterisk guys |
| 19:14.18 | mercestes | keyhack: Here is a shocker. Your local phone company still pays $$$ for your free local calls. |
| 19:14.26 | mercestes | keyhack: There are no "toll free" calls. |
| 19:14.34 | goodcat | hehehe |
| 19:14.46 | mercestes | keyhack: Every phone call, from the ringing, to the busy signal, to the nifty sleep inspiring static, is costing *someone* money. |
| 19:14.51 | goodcat | that's why companys almost never get those deals |
| 19:14.59 | bkruse | keyhack: nothing is free! |
| 19:15.06 | bkruse | keyhack: with the exception of * |
| 19:15.09 | keyhack | yeah, as long as that *someone* isn't me |
| 19:15.13 | mercestes | keyhack: Sometimes.......just sometimes.....it costs people money for calls that don't even exist. mostly with AT&T. |
| 19:15.17 | irq | someone said my nick |
| 19:15.17 | irq | who! |
| 19:15.44 | mercestes | keyhack: If you wish to offer any form of telco service, that someone will almost invariably be you. |
| 19:15.59 | mercestes | keyhack: Unless you find some way to abuse an existing system which is short term at the best. |
| 19:16.11 | *** join/#asterisk Winkie (i=sd@87-194-8-125.bethere.co.uk) |
| 19:16.35 | keyhack | yeah |
| 19:16.50 | b11d | I did |
| 19:16.56 | b11d | awhile back |
| 19:16.56 | mercestes | you could, theorhetically, "team up" with other VoIP terminated switches out there in the world and offer local "in net" call termination to whatever peers are connected to their switch...or even build your own VoIP network nationwide... |
| 19:17.02 | swilliamson | bkruse: * costs time, but now I get to bill for it |
| 19:17.18 | mercestes | but...I think your chances of finding a genie to be slightly higher. |
| 19:17.28 | keyhack | inbound is a different story though? I saw many providers offering unlimited in |
| 19:17.52 | mercestes | keyhack: inbound still costs them money. |
| 19:18.03 | b11d | I dunno.. I ate these mushrooms last week and there were genies all around me.. |
| 19:18.11 | b11d | took a ride on a magic carpet and everything |
| 19:18.25 | *** join/#asterisk X-Gen (n=X-Gen@dsl-242-18-26.telkomadsl.co.za) |
| 19:18.28 | mercestes | you don't know what....we will find...why don't you come with me little girl? |
| 19:18.34 | DaeJeon-Newbie | I want to buy IP-PHONEs, which run on an asterisk with 0 problem. anyone recommends? |
| 19:18.34 | b11d | great song |
| 19:18.39 | goodcat | hmm..is there any authoritative source (except the source code) for config file options (i.e. sip.conf, extensions.conf)? |
| 19:18.40 | b11d | a lousy can was all i found.. |
| 19:18.41 | mercestes | I have a techno version of it. |
| 19:18.54 | b11d | i had a chemical bro's remix |
| 19:18.56 | mercestes | goodcat: #asterisk and voip-info.org |
| 19:18.56 | b11d | it was good |
| 19:19.01 | kippi | anyone using asterisk gui ? |
| 19:19.07 | goodcat | mercestes: ok |
| 19:19.13 | swilliamson | cisco 7940g work great for me |
| 19:19.15 | mercestes | kippi: #freepbx does. ask them. |
| 19:19.28 | DaeJeon-Newbie | I want to buy IP-PHONEs, which run on an asterisk with 0 problem. anyone recommends? |
| 19:19.34 | b11d | haha |
| 19:19.37 | swilliamson | hey cisco thanks for charging me 450$ for an okay phone |
| 19:19.46 | goodcat | mercestes: so there're no "specs" released? |
| 19:19.48 | b11d | Polycom phones are the phones of choice.. |
| 19:20.00 | danp | seconded |
| 19:20.07 | danp | i am very impressed with these 601's |
| 19:20.09 | b11d | they offer the best mix of price/reliability and config options |
| 19:20.11 | mercestes | goodcat: the problem with realtime...is it's so stinking simple it's nearly impossible to setup. If there were something to "break" then ..there would be more documentation...and a direction to go in. |
| 19:20.15 | danp | i've used grandstreams before and they sucked |
| 19:20.22 | b11d | everyone hates the grandstreams |
| 19:20.28 | goodcat | b11d: not if you're a normal end user who doesn't have "special access" to upgrades, manuals, etc. |
| 19:20.29 | b11d | I know of no one who has praised the grandstream line of phones |
| 19:20.30 | mercestes | goodcat: The problem is, realtime is already "there" and if you setup the dbase and flip a switch in a config file, it magically works. |
| 19:20.47 | b11d | goodcat? |
| 19:20.49 | danp | goodcat: huh? the manuals are openly available |
| 19:20.54 | mercestes | goodcat: You can get one revision backwards, which, at this time, is not bad at all. |
| 19:21.03 | *** join/#asterisk [hC] (n=hardcore@mail.metrobridge.com) |
| 19:21.04 | DaeJeon-Newbie | I want to buy IP-PHONEs, which run on an asterisk with 0 problem. anyone recommends? |
| 19:21.08 | mercestes | goodcat: And if you *ever* need polycom manuls, just let me know. |
| 19:21.09 | b11d | holy shit |
| 19:21.15 | b11d | do you not read anythign in here DaeJeon-Newbie? |
| 19:21.20 | b11d | what did we just say about the polycoms? |
| 19:21.29 | danp | you can download the polycom manuals from the website |
| 19:21.29 | in-pt | DaeJeon-Newbie: cisco 7960 |
| 19:21.40 | mercestes | DaeJeon-Newbie: NO! Polycoms. definately polycoms. |
| 19:21.43 | *** join/#asterisk zmef420 (n=zmef420@metarb3-pool4-182.mtco.com) |
| 19:21.48 | DaeJeon-Newbie | in-pt: it is very expensive |
| 19:21.49 | b11d | 420.. yeahhhhhhh |
| 19:21.50 | b11d | im down |
| 19:22.02 | mercestes | goodcat: make sure you compiled asterisk with the debug option. what linux are you in? |
| 19:22.04 | DaeJeon-Newbie | cheap: 100$ |
| 19:22.04 | goodcat | mercestes: so why do they keep newest versions from end users? |
| 19:22.05 | in-pt | yes: but it is ultimate |
| 19:22.07 | b11d | DaeJeon-Newbie.. buy Polycom 501' |
| 19:22.08 | b11d | s |
| 19:22.09 | goodcat | mercestes: debian |
| 19:22.25 | mercestes | goodcat: They keep the latest version for certified users (it's easy to get certified)< and one revision backwards to everyone. |
| 19:22.26 | b11d | why the fuck do I want end users upgrading SIP? |
| 19:22.28 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 19:22.31 | b11d | I'll handle that, thank you |
| 19:22.33 | mercestes | goodcat: And the one revision backwards at this time is still very good. |
| 19:22.48 | goodcat | hehe.. |
| 19:22.54 | goodcat | well. anything's better than GS |
| 19:22.58 | mercestes | goodcat: gah, you couldn't have said gentoo could you? Do you have portage in Debian? |
| 19:22.59 | b11d | no |
| 19:23.03 | b11d | softphones are not better |
| 19:23.05 | mercestes | goodcat: Or are you a compile from source guy? |
| 19:23.08 | b11d | xlite sucks worse than gs.. |
| 19:23.14 | goodcat | mercestes: I have apt |
| 19:23.28 | goodcat | b11d: you can't compare soft phone and hard phone.. |
| 19:23.33 | b11d | I think I just did |
| 19:23.36 | b11d | but i hear ya |
| 19:23.36 | goodcat | hehe |
| 19:23.50 | mercestes | goodcat: ah yes. can you pass compile flag options to apt-get? |
| 19:24.01 | goodcat | well, I tried x-lite on mac os today..it crashes when you hang up! lol |
| 19:24.02 | b11d | doesnt apt just install binaries? |
| 19:24.07 | DaeJeon-Newbie | "b11d: softphones are not better' why? |
| 19:24.15 | mercestes | b11d: Probably. :/ |
| 19:24.17 | b11d | heh.. ALL softphones suck horribly |
| 19:24.23 | b11d | its the consensus of #asterisk |
| 19:24.25 | danp | goodcat: http://rubyurl.com/0wX -- there are all the docs for the 601s...what were you talking about? |
| 19:24.32 | in-pt | sjphone is good..anyone had problem with that ? |
| 19:24.41 | b11d | cant claim to have used it |
| 19:24.46 | b11d | or heard of it |
| 19:24.54 | goodcat | sjphone is ok |
| 19:24.59 | DaeJeon-Newbie | b11d: expresstalk |
| 19:25.03 | mercestes | in-pt: I liked xlite over sjphone. |
| 19:25.17 | DaeJeon-Newbie | I never had prob |
| 19:25.20 | mercestes | in-pt: Mostly in the eye-candy department. |
| 19:25.24 | b11d | anyway DaeJeon-Newbie.. you should go with Polycom 501s or 601's |
| 19:25.30 | swilliamson | what do people say about ata's? |
| 19:25.32 | b11d | if you're just getting started and want a good reliable high quality phone |
| 19:25.37 | *** join/#asterisk Tili (n=tili@172.Red-88-0-147.dynamicIP.rima-tde.net) |
| 19:25.37 | b11d | ATA's are alright.. |
| 19:25.38 | in-pt | but xlite had some issues..it works well with asterisk but create problems with ser |
| 19:25.40 | b11d | when you need them |
| 19:25.42 | mercestes | ATA's blow. |
| 19:25.47 | b11d | when you need them, they are great |
| 19:25.55 | mercestes | in-pt: Ah, ser. :) You know Clona? |
| 19:26.01 | swilliamson | i hate my ata's, waiting a whole ring for callid |
| 19:26.02 | in-pt | yes of course |
| 19:26.05 | b11d | I'll take FAXing across ATA's before e-faxing(at this point) |
| 19:26.08 | in-pt | she is mastermind |
| 19:26.11 | mercestes | b11d: they blow... |
| 19:26.18 | mercestes | in-pt: clona is a sexy man..:D |
| 19:26.20 | b11d | my cisco vg224 rocks.. |
| 19:26.22 | b11d | NO issues |
| 19:26.24 | b11d | quick |
| 19:26.26 | mercestes | in-pt: I made the same mistake once.....but then he sent me pics. |
| 19:26.38 | mercestes | b11d: multimode T1: with pri channels.... |
| 19:26.43 | swilliamson | sipura 3000 here, echo;s like mad |
| 19:26.44 | in-pt | mercestes: ohh jessus ..the name predicts as if it is a female name |
| 19:26.49 | goodcat | mercestes: you can compile from source with apt, but I don't have a reason to do that |
| 19:26.55 | in-pt | thank god i havent said that to him |
| 19:26.56 | mercestes | in-pt: I know! I so thought he was a girl atfirst too...lol |
| 19:27.02 | mercestes | in-pt: I was hitting on him. |
| 19:27.26 | in-pt | mercestes: thats great |
| 19:27.28 | mercestes | in-pt: Then he's like.."want my pic?" and I was like, "yea!" he was all bald...making a kissy face at me. lol. It was great. |
| 19:27.35 | b11d | lol |
| 19:27.48 | mercestes | it was like prison all over again. |
| 19:27.53 | b11d | ahh prison.. |
| 19:27.55 | in-pt | great: i escaped that situation..hehehehe |
| 19:27.56 | b11d | the good old days |
| 19:28.07 | mercestes | yea...were men were men and boys were girls. |
| 19:28.11 | mercestes | anyways. |
| 19:28.22 | b11d | yeppers |
| 19:28.23 | *** join/#asterisk kamuix (i=kamuix@232.107.94.80.dsl.libello.mc) |
| 19:28.36 | mercestes | goodcat: Try downloading the asterisk sources and compiling them with the debug flag. Google debugging asterisk. |
| 19:28.44 | b11d | agreed |
| 19:28.51 | b11d | everyone should know how to install * from source. |
| 19:28.52 | mercestes | goodcat: You have to compile in the frame pointers or some technical developer term like that. |
| 19:28.59 | b11d | if you're going to run it halfway seriously anyway |
| 19:29.09 | mercestes | b11d: I know how to install asterisk from source. emerge -av asterisk and grab a sammich |
| 19:29.21 | b11d | ... |
| 19:29.34 | mercestes | your so jealous. |
| 19:29.34 | mercestes | :D |
| 19:29.39 | b11d | yeah, I am :)( |
| 19:29.40 | mercestes | gentoo ftw |
| 19:30.00 | mercestes | USE="pri debug zaptel t38" emerge -v asterisk and voila. |
| 19:30.02 | b11d | eff tee doubleyou |
| 19:30.04 | mercestes | does my samples.... |
| 19:30.18 | mercestes | it even works "out of hte box" without me touching anything. not useful..but in a working state. |
| 19:30.32 | b11d | im moving to tx tomorrow then |
| 19:30.42 | b11d | I was in houston a few months back.. |
| 19:30.52 | mercestes | I can *even* emerge -av =net-misc/asterisk-1.0.9 if I want and get a specific version. |
| 19:30.56 | swilliamson | the stock ast source for 1.4 beta builds good too, even checks your enviornment and stuffs |
| 19:30.59 | b11d | what an ungodly spread out city that is.. |
| 19:31.13 | mercestes | Houston is nice.....very inefficient, but nice. |
| 19:31.18 | b11d | yeah I enjoyed it |
| 19:31.32 | *** join/#asterisk djflux (n=djflux@mm.shermfin.com) |
| 19:31.35 | mercestes | should come live under my bed...we can take over the city with cheap VoIP service. |
| 19:31.40 | b11d | woah |
| 19:31.41 | b11d | everyone stoo |
| 19:31.42 | b11d | p |
| 19:31.44 | b11d | wheres that mix djflux? |
| 19:31.45 | *** join/#asterisk RoyK (n=roy@ti211310a080-2073.bb.online.no) |
| 19:31.52 | b11d | mercestes.. sounds like a plan |
| 19:31.58 | *** join/#asterisk domingues (i=domingue@200-170-201-152.core01.spo.ifx.net.br) |
| 19:32.04 | b11d | djflux.. dont leave me hanging |
| 19:32.11 | b11d | I need to hear some DrAirRider |
| 19:32.33 | djflux | HAHA ... haven't had the time to put it together ... I had to watch my bengals get their a** wiped on MNF last night :( |
| 19:32.39 | b11d | :P |
| 19:32.50 | djflux | and then the day job calls |
| 19:32.50 | b11d | use the loss as inspiration |
| 19:32.57 | b11d | if you had been able to play your mix to them.. they would have won |
| 19:33.02 | djflux | LOL |
| 19:33.04 | b11d | they'll change their name to the "Benfluxz" |
| 19:33.08 | djflux | I think so |
| 19:33.09 | b11d | when they hear it |
| 19:33.15 | hardwire | anybody here ever had digium hardware hang up.. but the long distance carrier never showed a hangup? |
| 19:33.19 | mercestes | ok, well I have to go torture my wife. I'm home sick and she's all "torture meeeee!" because she's bored. |
| 19:33.25 | mercestes | goodcat: Look up compiling with debug optins. |
| 19:33.25 | b11d | haha |
| 19:33.30 | b11d | enjoy the anal |
| 19:33.34 | mercestes | b11d: Check out priceline.com for cheap tickets. |
| 19:33.36 | domingues | Hello All, does everybody already worked with BroadVoice Account in Asterisk, I getting problems to activate a account, I made all config exatly in BV Site, but When I dial the Number I got ringing and the called answer, but it is stay in ringing |
| 19:33.37 | hardwire | its happened twice with AT&T resulting in multiple bills of over $20,000 |
| 19:33.44 | mercestes | in-pt: Let me know if you manage to figure out how you fixed FoP... |
| 19:33.45 | b11d | I'm in with Northwest Airlines.. can fly anywhere for free.. |
| 19:33.48 | b11d | so no worries |
| 19:34.03 | mercestes | i'm going to take a break and come back later. I might go back to trying to get bugzilla working. |
| 19:34.08 | b11d | ttyl |
| 19:34.11 | mercestes | l8s |
| 19:34.13 | b11d | let me bug marshall for you when you get that done |
| 19:34.13 | in-pt | its more than 6 months and the version is also different..i am not sure |
| 19:34.20 | in-pt | i am sorry |
| 19:34.33 | b11d | well im going to head out for the day |
| 19:34.34 | b11d | cya chaps |
| 19:34.52 | swilliamson | bye |
| 19:34.58 | domingues | I have BoradVoice, I Dial a number, The order end answer the call, but the call still stay in ringing in ASterisk... |
| 19:35.07 | domingues | any Idea |
| 19:35.16 | goodcat | how do I see which use-flags asterisk is compiled with? |
| 19:35.35 | swilliamson | cat Makefile |
| 19:35.42 | b11d|bbl | cat Makefile | more |
| 19:35.42 | b11d|bbl | :P |
| 19:35.55 | swilliamson | ha |
| 19:35.56 | goodcat | no, without the makefile |
| 19:36.15 | b11d|bbl | ok.. |
| 19:36.18 | b11d|bbl | cat makefile > more |
| 19:36.19 | b11d|bbl | :P |
| 19:36.33 | b11d|bbl | ok im bbl |
| 19:36.37 | b11d|bbl | FTW!!! |
| 19:37.26 | codefreeze | domingues: sounds suspiciously like a prob with devicestate... I'm assuming this is SIP. Try call_limit...? |
| 19:37.49 | codefreeze | domingues: 1.2 or 1.4 version of asterisk? |
| 19:38.06 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-88-153-52-215.red.bezeqint.net) |
| 19:38.18 | domingues | <codefreeze> I tryed with both 1.2 and 1.4 Beta |
| 19:38.43 | domingues | <codefreeze> let me try call-limit... |
| 19:41.17 | *** join/#asterisk TripleFFFF (n=TripleFF@145-27.mc.cite.net) |
| 19:41.29 | TripleFFFF | anyone know why a 7960 would ALWAYS look for a sbn file ? P0S3-07-2-00.sbn |
| 19:41.38 | TripleFFFF | i think sbn is P00 no P0s |
| 19:41.39 | domingues | codefreeze, I set call-limit to 100, but the same problem, the order side answer the call, but to asterisk it still stay ringing |
| 19:42.11 | domingues | * goodcat has quit IRC ("leaving") |
| 19:42.11 | domingues | * TripleFFFF has joined #asterisk |
| 19:42.11 | domingues | <TripleFFFF> anyone know why a 7960 would ALWAYS look for a sbn file ? P0S3-07-2-00.sbn |
| 19:42.11 | domingues | <domingues> codefre |
| 19:42.58 | TripleFFFF | yeah |
| 19:43.18 | codefreeze | domingues: Then it is not a devicestate issue... I think I saw a bug with a similar prob on bugs.digium.com; you might check there |
| 19:43.19 | TripleFFFF | wahts the repeaing mode ? |
| 19:43.33 | domingues | In the BV website they say to use inband as dtmfmode, but I set to inband I get Bad Request from BV SIP SERVER, if I set to rfc2833, I dont get Bad Request, but I got the problem |
| 19:44.22 | *** part/#asterisk TripleFFFF (n=TripleFF@145-27.mc.cite.net) |
| 19:44.29 | swilliamson | BV is broken |
| 19:44.34 | swilliamson | in so many ways |
| 19:45.06 | domingues | swilliamson do you have a sample of SIP.conf to use with BV |
| 19:46.02 | swilliamson | you try the one from voip-info? i gave up on bv two years ago |
| 19:46.11 | swilliamson | try the voxilla forums |
| 19:46.24 | swilliamson | i have no more config |
| 19:46.44 | *** join/#asterisk Strom_C_ (n=strom@70.141.71.195) |
| 19:46.52 | domingues | let me see in voip-info |
| 19:47.24 | tm | so kann hieer wer deutsch? ;) |
| 19:48.37 | *** join/#asterisk sjobeck (n=sjobeck@adsl-074-245-224-237.sip.bct.bellsouth.net) |
| 19:48.56 | swilliamson | ja |
| 19:48.58 | swilliamson | warum |
| 19:49.25 | *** join/#asterisk sjobeck (n=sjobeck@adsl-074-245-224-237.sip.bct.bellsouth.net) |
| 19:49.25 | swilliamson | nicht muttersprachlich |
| 19:50.30 | *** part/#asterisk Ravi1974 (n=I@static-70-19-119-112.ny325.east.verizon.net) |
| 19:50.55 | *** join/#asterisk naitram (n=danny@216.77.58.40) |
| 19:51.17 | naitram | anyone have any success using Aastra 480i phones |
| 19:52.35 | tm | swilliamson: hi |
| 19:52.43 | tm | swilliamson: ich moechte ein transfer einleiten mit musconhold |
| 19:52.57 | tm | exten => 37061215,5,Dial(mISDN/1/31772818) ist das so korrekt? |
| 19:53.00 | tm | oder was fehlt mir da? |
| 19:53.26 | swilliamson | glaub, ,Dial(mISDN/1/31772818,m) moment |
| 19:53.37 | tm | ok |
| 19:53.58 | wunderkin | no that isn't right either |
| 19:54.19 | swilliamson | Dial(mISDN/1/31772818,30,m) |
| 19:54.23 | swilliamson | timeout failt |
| 19:54.31 | swilliamson | http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial |
| 19:55.03 | tm | swilliamson: ah sooo einfach |
| 19:55.54 | swilliamson | es ist fast immer einfach |
| 19:56.55 | [TK]D-Fender | naitram: Plenty of people |
| 19:57.38 | *** join/#asterisk e-milio (n=emilio@pmr.pmrtechnologies.com) |
| 19:58.37 | *** join/#asterisk root (n=root@ns1.compuvox.com.br) |
| 19:58.47 | *** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman) |
| 20:01.00 | *** join/#asterisk SuPrSluG (n=SuPrSluG@pool-71-241-163-59.buff.east.verizon.net) |
| 20:01.15 | *** join/#asterisk jmesquita (n=jmesquit@201.7.117.114) |
| 20:01.38 | tm | swilliamson|gone: darf nochmal? ;) |
| 20:05.27 | tm | swilliamson|gone: noch da? ;) |
| 20:06.08 | *** join/#asterisk Dr-Linux|home (n=Dreamer@DSL-202-59-73-131.nexlinx.net.pk) |
| 20:06.45 | naitram | [TK]D-Fender: I cant get the 480i to connect to the tftp server to get its configuration file, any experience? |
| 20:06.53 | Dr-Linux|home | anybody tried LumenVox recognition with asterisk? |
| 20:07.22 | *** join/#asterisk ambriento (n=ambrient@201-95-105-83.dsl.telesp.net.br) |
| 20:07.36 | [TK]D-Fender | naitram: Make sure your TFTP server is running and test it from a PC client. Then make sure your phone is pointing at it. Then make sure your filenames are correct |
| 20:07.55 | robl^ | and make sure you have it turned on |
| 20:07.59 | holmier | just god dman it make sure! |
| 20:08.00 | holmier | ;) |
| 20:10.17 | swilliamson|gone | wireshark, sniff those packets |
| 20:11.12 | swilliamson | <td> weider da |
| 20:15.55 | Dr-Linux|home | anybody tried LumenVox recognition with asterisk? |
| 20:17.01 | *** join/#asterisk [hC] (n=hardcore@mail.metrobridge.com) |
| 20:17.26 | *** join/#asterisk linagee (n=linagee@unaffiliated/linagee) |
| 20:17.46 | linagee | does anyone know of a voip service that will let me port over my number and has IAX and is reliable? |
| 20:17.55 | linagee | s/IAX/IAX2/ |
| 20:17.57 | DaeJeon-Newbie | I want going to buy analog cards http://www.digitnetworks.com/X100P_FXO_PCI_Card_p/x100p.htm |
| 20:17.58 | *** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 20:18.11 | DaeJeon-Newbie | any issue with this card? |
| 20:18.39 | *** join/#asterisk Dovid (n=Dovid@ool-43530a83.dyn.optonline.net) |
| 20:18.42 | blitzrage | anyone happen to use ChanIsAvail() on an IAX2 channel in Asterisk 1.2.12.1? For some reason I have a peer registered, and it works if its SIP, but not if its IAX2 |
| 20:19.17 | DaeJeon-Newbie | this guy claims that X100P different than "clone" cards |
| 20:19.31 | DaeJeon-Newbie | plz have a look |
| 20:19.45 | DaeJeon-Newbie | <PROTECTED> |
| 20:19.58 | DaeJeon-Newbie | any comments? |
| 20:20.04 | DaeJeon-Newbie | good to go? |
| 20:20.21 | Strom_C_ | DaeJeon-Newbie, save yourself a lot of headache and just don't touch anything that claims to be x100p, clone or not |
| 20:20.27 | naitram | [TK]D-Fender: as far as I can tell the aastra phone never attempts anything, i have ethereal sniffer and I never get anything when it boots up. All the settings look right |
| 20:20.35 | robl^ | X100P (even clones) are not really useful for production.. problemmatic. |
| 20:21.21 | robl^ | I have a Digium X101P and it sits in a box (cardboard) |
| 20:21.45 | linagee | robl^: you have a server running inside a cardboard box? heh |
| 20:21.48 | Dovid | anyone know polycoms? |
| 20:21.57 | blitzrage | yep, lots of people I imagine :) |
| 20:21.58 | naitram | [TK]D-Fender: also i can grab the file from my windows pc using tftp |
| 20:22.08 | *** part/#asterisk hardwire (n=hardwire@rdbck-4891.wasilla.mtaonline.net) |
| 20:22.25 | blitzrage | Dovid: just ask a question, it works better |
| 20:22.28 | robl^ | linagee: it's my imaginary backup server! ;-) |
| 20:22.39 | Dovid | i want to upgrade to 1.6.7 from 1.6.6, i made lots of changes to sip.cfg - my question is if I have to upgrade that file to or can i just upgrade sip.id ? |
| 20:22.47 | linagee | robl^: hehehe. better to have a backup server running inside a cardboard box than no backup server. ;) |
| 20:22.48 | DaeJeon-Newbie | robl^: what can I buy then? I need one FXO port |
| 20:22.51 | blitzrage | just upgrade the sip.ld |
| 20:22.56 | Dovid | thx |
| 20:23.14 | linagee | robl^: do you have some sort of failover software running? |
| 20:23.23 | robl^ | DaeJeon-Newbie: you can get a single port (upgradable) TDM card from Digium |
| 20:23.47 | linagee | shellsha1k: around? |
| 20:23.51 | DaeJeon-Newbie | robl model number? |
| 20:24.20 | linagee | shellsha1k: you have a bug on your website. also, what do you charge for LNP? |
| 20:24.30 | [TK]D-Fender | Dovid: you can keep your existing sip.cfg, it'll be fine |
| 20:24.44 | DaeJeon-Newbie | robl^: model number? |
| 20:24.56 | Dovid | TK: so all i need to upload is sip.id ? do i need to update sip.ver ? |
| 20:25.00 | linagee | DaeJeon-Newbie: cardboard box model number XJ-281 |
| 20:25.27 | robl^ | DaeJeon-Newbie: I am looking. Digium has changed their site.. |
| 20:25.51 | DaeJeon-Newbie | robl^: no way to buy now? |
| 20:25.54 | robl^ | Does Digium still sell the 1 FXO / 1 FXS developer kit??? |
| 20:26.16 | robl^ | DaeJeon-Newbie: I am trying to track it. the site has been reorganized since I bought mine |
| 20:26.21 | Strom_M | robl^, it's called the TDM11B |
| 20:26.40 | linagee | Strom_M: what's the diff, Strom_M, Strom_C? |
| 20:26.47 | robl^ | Strom_M: right.. but.. I mean they used to sell a discounted version for developerss.. |
| 20:26.55 | Strom_M | _M is "mobile" |
| 20:26.59 | Strom_M | _C is "at the office" |
| 20:27.06 | [TK]D-Fender | Dovid: LD & VER |
| 20:27.12 | Dovid | oops |
| 20:27.15 | *** join/#asterisk RoyK (n=roy@217-175-39.100710.adsl.tele2.no) |
| 20:27.17 | Dovid | just did ld. |
| 20:27.32 | danp | what's the single FXO SIP convereter of choice |
| 20:27.36 | linagee | Strom_M: what a funny ip. your isp needs to do reverse dns |
| 20:27.37 | danp | err, single FXS |
| 20:27.46 | Strom_M | linagee, I'm at my client's site rightnow |
| 20:27.47 | Dovid | danp: matter of opinion |
| 20:27.53 | Dovid | many people like sipura |
| 20:27.58 | linagee | Strom_M: oic. i thought mobile=EVDO or something |
| 20:28.03 | Strom_M | no |
| 20:28.08 | [TK]D-Fender | danp: Single ports FXS is a complete waste. Spend the extra 10$ ad get a 2-port. SPA-2002 is a good choice. |
| 20:28.09 | Strom_M | mobile == not at the office |
| 20:28.43 | robl^ | DaeJeon-Newbie: call digium ask for a TDM400P with a single FXO |
| 20:28.53 | *** part/#asterisk SkramX (n=mark@70.86.176.2) |
| 20:29.03 | linagee | robl^: that sounds expensive. why not get a TDM400P clone instead? ;) |
| 20:29.28 | danp | what are the other main options? |
| 20:29.41 | robl^ | linagee: I use NO analog cards ;-) all digital is cheaper. |
| 20:30.06 | linagee | robl^: might as well use no analog lines either. go with a T1. ;) |
| 20:30.29 | [TK]D-Fender | danp: Linksys/Sipura SPA series is decent, and Meditrix ATA's a really top-end, but cost a bit more and are more cryptic to learn at the start. |
| 20:30.46 | [TK]D-Fender | danp: There are no others I recommend for this level. |
| 20:30.49 | *** part/#asterisk naitram (n=danny@216.77.58.40) |
| 20:30.59 | *** join/#asterisk daysmen3 (n=primus@host86-138-237-97.range86-138.btcentralplus.com) |
| 20:31.19 | danp | cool, thanks. i'm just trying to find options for plugging a cordless phone or similar into |
| 20:31.21 | robl^ | linagee: I have low volume. I just have 2 IAX trunks and about a dozen SIP phones... works for our needs.. |
| 20:32.09 | linagee | robl^: "all of our circuits are busy now. if you'd like more free channels, please chip in to the robl^ fund" |
| 20:33.17 | [TK]D-Fender | danp: If you're talking in-lan use, then SPA-2002 it is. |
| 20:33.28 | danp | i'll check it out |
| 20:33.46 | robl^ | linagee: hah! I have had up to 15 concurrent calls on the system via the IAX trunks.. but that is VERY rare. never had all circuits buys.. again its for use with a group of volunteers / non-profit (NO money) |
| 20:33.54 | hads | The 2102 has replaced it |
| 20:34.43 | danp | can each port have its own registration? |
| 20:34.47 | danp | i would assume so |
| 20:34.50 | hads | Yes |
| 20:35.37 | [TK]D-Fender | hads: Should have, the 2002 replaced the 2000, and the 2102 replaced the 2100. The 2100 series ATA's have a built-in router which you do NOT want since SIP only worked on the WAN port. Avoid. |
| 20:35.47 | [TK]D-Fender | danp: Yup |
| 20:36.13 | *** join/#asterisk eltech (i=G00Ds@ool-457c93b6.dyn.optonline.net) |
| 20:36.21 | hads | [TK]D-Fender: Ah sorry, my bad. We don't get the 2002 over here, just the PAP2T |
| 20:36.41 | hads | Quite right about the WAN, I was just typing that out myself. |
| 20:36.50 | [TK]D-Fender | hads: PAP2 is similar, but I believe cut-back somewhere... just can't confirm where. where is "here"? |
| 20:37.01 | hads | .nz |
| 20:37.20 | Druken | do they even make the PAP2 anymore? |
| 20:37.35 | hads | The PAP2T replaced it. Compared to the 2102 the PAP2T doesn't do T38. |
| 20:37.49 | *** join/#asterisk dasenjo (n=dasenjo@190.24.179.198) |
| 20:40.38 | hads | I think the PAP2T may have replaced the 2002? |
| 20:41.49 | hads | AFAIK they only have the PAP2T, SPA2102 and SPA3102 in that series now. |
| 20:43.55 | *** join/#asterisk sjobeck_ (n=sjobeck@adsl-074-245-224-237.sip.bct.bellsouth.net) |
| 20:45.21 | Dovid | i am having problems with MWI on a polycom 601, does anyone know what i have to edit in the cfg files ? |
| 20:46.21 | naftali5 | Dovid, what's up |
| 20:46.25 | *** join/#asterisk tr2x (n=alvar@80-218-185-55.dclient.hispeed.ch) |
| 20:46.32 | Dovid | i cant get MWI to work on the phone |
| 20:46.55 | Dovid | what do i have to edit so the light blinks when there is a VM |
| 20:47.03 | danp | Dovid: using realtime? |
| 20:47.05 | blitzrage | I don't want to meet your mom... I just want |
| 20:47.06 | Dovid | yes |
| 20:47.17 | [TK]D-Fender | blitzrage: ! ! ! |
| 20:47.21 | danp | Dovid: you need rtcachefriends=yes in your sip.conf [general] section |
| 20:47.21 | [hC] | doh.. pap2-na's cant do vlans |
| 20:47.24 | blitzrage | mwahahahahaha |
| 20:47.24 | [hC] | tats so weak. |
| 20:47.29 | Dovid | and the phone will get it ? |
| 20:47.33 | danp | yep |
| 20:47.35 | Dovid | thx |
| 20:47.41 | [TK]D-Fender | hads: The SPA-2002 is still current and available. |
| 20:47.43 | danp | if you have the mailbox set for the user |
| 20:47.55 | Dovid | i do |
| 20:47.57 | *** part/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 20:48.09 | [TK]D-Fender | Dovid: Pastebin your entry |
| 20:48.11 | danp | then once you add rtcachefriends and reload they should just work |
| 20:48.55 | *** join/#asterisk ShadowHntr (i=sentinel@wikipedia/Shadowhntr) |
| 20:51.53 | *** join/#asterisk vader-- (n=me@c-71-226-201-15.hsd1.nj.comcast.net) |
| 20:52.15 | *** join/#asterisk s34n (n=s34n@ip-206-159-190-125.mvdsl.com) |
| 20:53.16 | s34n | anybody using snom360 phones? |
| 20:53.34 | Dovid | danp: do i have to set anything in the phone at alll ? |
| 20:53.37 | *** join/#asterisk luchshiy (n=anonymou@d212-53-104-193.cust.tele2.ch) |
| 20:54.27 | Dovid | danp: anything under message center for the polycom 601 ? |
| 20:54.50 | danp | Dovid: i'm pretty sure i didn't...if you have the equivalent of mailbox=123 for that sip user it should just work |
| 20:54.59 | danp | no |
| 20:55.10 | [hC] | anyone know if its possible for the linksys pap2 to do vlans? |
| 20:55.13 | Dovid | danp: its mailbox=123@companyname |
| 20:55.14 | [hC] | Its not there by default |
| 20:55.43 | danp | might turn up your asterisk verbosity and see if there are any errors when the phone tries to subscribe |
| 20:56.04 | Dovid | danp: rebooted the phone and its working |
| 20:56.17 | danp | oh good. it probably would have started working after a while |
| 20:57.25 | *** join/#asterisk groogs (n=greg@d38-54-164.commercial1.cgocable.net) |
| 20:57.36 | robl^ | s34n: I used Snom360s until about 3-4 months ago. switched to Aastra |
| 20:57.56 | s34n | robl^: did you use setting_files? |
| 20:58.34 | robl^ | s34n: I used the config files. nothing was set in the web.. |
| 20:59.00 | Dovid | danp: once i have u here - client is complaining that when they are on the phone if they get a second call then they only hear one beep |
| 20:59.12 | Dovid | they want the phone to beep more often - how do i change that ? |
| 21:00.36 | s34n | http://www.voip-info.org/wiki/view/snom+mass+deployment seems to be broken |
| 21:02.12 | robl^ | s34n: hrmm. prolly a link from the OLD wiki |
| 21:02.14 | Rawplayer | is there any alternative for festival on freebsd? |
| 21:02.27 | danp | Dovid: hmm, no idea about that...let me look at the admin guide real quick |
| 21:02.53 | *** join/#asterisk Miss-tURk[off] (n=YaLanci@85.107.171.161) |
| 21:03.28 | Dovid | okies. the guide is a lil complicated for me - i have made lots of mods to the cfg files but it makes my head spin |
| 21:03.43 | hads | [TK]D-Fender: Two random sites; http://www.voiplink.com/SPA_2002_p/linksys-spa-2002.htm http://www.888voipstore.com/linksys-sipura-spa-2002-pr-16186.html |
| 21:04.05 | hads | They both say that it's been replaced by the PAP2T |
| 21:04.20 | s34n | robl^: I'm looking for more detail on the setting_file contents for snom mass deployment |
| 21:04.50 | robl^ | s34n: http://www.snom.com/wiki/index.php/Mass_deployment <-- try that |
| 21:05.20 | *** join/#asterisk seele_ (n=seelen@dns.datawareltda.com) |
| 21:06.18 | s34n | robl^: I've been there, but it doesn't really tell you the setting names, etc. |
| 21:06.38 | s34n | robl^: some settings reference would be nice. |
| 21:07.39 | robl^ | s34n: http://www.snom.com/wiki/index.php/Web_Interface/Settings <-- that should have the names |
| 21:08.03 | *** join/#asterisk razu_ (n=razu@87-119-182-130.tll.elisa.ee) |
| 21:08.08 | s34n | robl^: thanks! |
| 21:08.33 | hads | Doesn't the phones web interface itself have a settings file dump |
| 21:08.50 | robl^ | s34n: you can set up one phone in the web interface.. then there is a link on the web interface to allow you to view a settings file.. you can use that as a template for all the phones |
| 21:09.04 | *** join/#asterisk ClydeGoffe (n=cgoffe@ool-457d3e0d.dyn.optonline.net) |
| 21:09.07 | hads | Yeah. Thought so. |
| 21:09.08 | s34n | k |
| 21:10.31 | [TK]D-Fender | hads: Nope. Outright misinformation. |
| 21:10.34 | ClydeGoffe | Hey all |
| 21:10.52 | seele_ | please help I need to split 2 different groups of extensions ... how can I configure the context? |
| 21:10.52 | ClydeGoffe | hoping someone can help me decipher a warning i'm getting in the asterisk's logs |
| 21:10.56 | ClydeGoffe | WARNING[32000] interface.c: Junk at the beginning of frame 41504554 |
| 21:11.04 | ClydeGoffe | multiple times |
| 21:11.13 | ClydeGoffe | any idea what that means and what could be causing that |
| 21:11.33 | robl^ | I like(d) Snom.. but they kept breaking the firmware. every version had some pretty annoying bugs.. I ended up ditching them all. still ahve 3 Snom 360s sitting on a shelf after I retired them |
| 21:13.37 | hads | [TK]D-Fender: OK, good to know. It's accurate for the situation here, the current models are the PAP2T, SPA2102 and SPA3102. |
| 21:14.04 | hads | [TK]D-Fender: At least that's all that Linksys NZ are supplying. |
| 21:14.40 | hads | robl^: Intersting, what firmware version did you get up to with them? |
| 21:15.10 | robl^ | hads: 5.x like .54, I think |
| 21:15.20 | robl^ | 5.4, even |
| 21:15.31 | seele_ | how can I create anew context |
| 21:15.34 | seele_ | ? |
| 21:15.45 | hads | OK. I think 6.x is more stable. The snom 360 would be nicer if the screen was slightly higher res IMHO. |
| 21:15.46 | bkruse | [zomgseeleisnub] |
| 21:15.46 | bkruse | exten => whatever,1,Answer() |
| 21:15.55 | *** join/#asterisk dlynes_laptop (n=dlynes@S0106001346f7843f.vc.shawcable.net) |
| 21:17.00 | robl^ | hads: had issues for a while with dropping calls on phones.. had another with a phone rebooting constantly. had one where if you were transfering a call and a call came in at the same time.. you lost both calls |
| 21:17.17 | seele_ | bkruse, I can make same number extensions in different context? |
| 21:17.30 | bkruse | seele_: yes |
| 21:17.34 | hads | Fair enough dropping them then :) The pixels on the 360 are too big compared to the cheapy Linksys sitting next to me. |
| 21:17.42 | bkruse | bkruse: [zomgseeleisnub] |
| 21:17.42 | bkruse | bkruse: exten => 1,1,Answer() |
| 21:17.51 | seele_ | bkruse, sorry for my ignorance ... how ... example please |
| 21:18.30 | robl^ | hads: the rez was low.. but it was useable. I wanted them for a phone.. not to play video games. ;-) |
| 21:18.42 | hads | heh |
| 21:18.56 | hads | Well that's all I do with my phones. |
| 21:19.31 | robl^ | using Aastra 480i and 9133is now. rock solid. not as many bells and whistsles.. but solid. |
| 21:19.35 | bkruse | seele_: make something like this |
| 21:19.35 | bkruse | [zomgseeleisnub] |
| 21:19.35 | bkruse | exten => 10,1,Answer() |
| 21:19.35 | bkruse | exten => 10,n,Noop(YAY!!!) |
| 21:19.35 | bkruse | exten => 10,n,Playback(tt-allbusy) |
| 21:20.02 | seele_ | extensions.conf ? |
| 21:20.33 | bkruse | yes |
| 21:20.40 | bkruse | seele_: http://voip-info.org |
| 21:21.03 | robl^ | "Use the wiki-source, Luke!" |
| 21:21.26 | Nivex | "I'd rather kiss a wiki!" |
| 21:21.32 | Nivex | "That can be arrranged!" |
| 21:23.00 | *** join/#asterisk drfreeze (n=Jim@www.freeze.org) |
| 21:23.03 | drfreeze | Hello |
| 21:23.45 | drfreeze | I've got a couple of Polycom501 phones that seem to ignore their GSM setting and the clocks are off by 6hrs. Anyone have an idea as to why? |
| 21:24.18 | bkruse | drfreeze: no idea, can 501's do ntp? |
| 21:24.21 | dlynes_laptop | drfreeze: wrong time zone? |
| 21:24.34 | Corydon-w | Polycom doesn't support the GSM codec and you forgot to set an NTP server in your DHCP |
| 21:24.57 | Corydon-w | and/or you forgot to set that in the config |
| 21:25.00 | drfreeze | dlynes_laptop: time zones seem to be ignored |
| 21:25.19 | dlynes_laptop | drfreeze: see Corydon-w's suggestion regarding an ntp server |
| 21:25.49 | drfreeze | bkruse: I think they all use the same ntp |
| 21:26.00 | dlynes_laptop | Corydon-w: you can't specify an ntp server, manually? it only grabs it from the dhcp option clause? |
| 21:26.10 | Corydon-w | Sure you can |
| 21:26.25 | danp | Dovid: hmm, i don't see anything about making it beep more than once |
| 21:26.25 | dlynes_laptop | Corydon-w: ah...would just seem kinda silly if it couldn't :) |
| 21:26.31 | bkruse | web interface ;] |
| 21:27.00 | danp | i use the phone configs to set the GMT offset |
| 21:27.03 | dlynes_laptop | web interfaces are highly overrated |
| 21:27.10 | danp | that way there's only one extra option in DHCP |
| 21:27.11 | bkruse | dlynes_laptop: agreed! |
| 21:27.13 | bkruse | links http:// :] |
| 21:27.19 | [TK]D-Fender | You can set NTP in either DHCP or in the phones direct configs overriding DHCP. |
| 21:27.35 | danp | these polycoms are great...i haven't even touched the web interface yet |
| 21:27.41 | dlynes_laptop | I wish all phones would allow you to all the configuration from the phone's dialpad |
| 21:27.44 | [TK]D-Fender | danp: No should you ever |
| 21:27.47 | [TK]D-Fender | nor* |
| 21:27.54 | dlynes_laptop | s/to all/to do all/ |
| 21:27.58 | danp | i have a rails app that generates configs |
| 21:31.14 | *** join/#asterisk shido6 (n=shido6@d221-68-200.commercial.cgocable.net) |
| 21:31.19 | *** join/#asterisk luke-jr_work (n=luke-jr@fl-71-53-155-1.dhcp.embarqhsd.net) |
| 21:31.44 | luke-jr_work | any ideas on determining why Asterisk has decided a SIP client is Unauthorized? |
| 21:32.03 | *** join/#asterisk obnauticus (n=aao_pwne@c-24-21-116-29.hsd1.wa.comcast.net) |
| 21:32.05 | luke-jr_work | it's a PAP2 that has moved from the LAN to a WAN location |
| 21:32.20 | luke-jr_work | so credentials are the same and all |
| 21:32.34 | CunningPike | luke-jr_work: Pastebin your sip.conf |
| 21:32.36 | CunningPike | ~pb |
| 21:32.37 | jbot | [pb] a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste |
| 21:34.22 | dlynes_laptop | luke-jr_work: chances are it's set to a static ip, and it's coming from a different static ip now |
| 21:35.44 | CunningPike | Hey, dlynes_laptop |
| 21:35.58 | dlynes_laptop | heya anthony |
| 21:35.58 | dlynes_laptop | i |
| 21:36.03 | dlynes_laptop | oops...not on mud |
| 21:36.09 | luke-jr_work | dlynes_laptop: nope |
| 21:36.36 | dlynes_laptop | luke-jr_work: well, we're not going to be able to help you without a pastebin of your sip.conf file, either |
| 21:36.45 | luke-jr_work | dlynes_laptop: working on it |
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| 21:37.06 | dlynes_laptop | CunningPike: i'm getting too much sleep lately |
| 21:37.16 | CunningPike | Lucky you |
| 21:37.24 | dlynes_laptop | CunningPike: averaging about 3-1/2 hours every night lately |
| 21:37.33 | CunningPike | dlynes_laptop: Ugh |
| 21:37.45 | dlynes_laptop | yeah, tell me about it |
| 21:37.53 | CunningPike | dlynes_laptop: Should I ask why, or does it involve Oriental ladies? |
| 21:37.56 | drfreeze | Anyone know the default password for the webinterface to a polycom phone? |
| 21:37.56 | dlynes_laptop | Getting faxing working |
| 21:38.03 | dlynes_laptop | Getting monitoring working |
| 21:38.05 | CunningPike | dlynes_laptop: Ewwww |
| 21:38.11 | CunningPike | ~wglwat |
| 21:38.13 | jbot | somebody said wglwat was well, good luck with all that |
| 21:38.13 | dlynes_laptop | Getting a foreign exchange server written |
| 21:38.25 | dlynes_laptop | basically way too much crap |
| 21:38.26 | robl^ | faxing on a pbx has NEVER made any sense to me. |
| 21:38.40 | bkruse | robl^: when i see your name i think rofl |
| 21:38.42 | dlynes_laptop | robl^: We're not trying to fax on a pbx |
| 21:38.46 | CunningPike | drfreeze: Have you tried 456? |
| 21:39.14 | dlynes_laptop | robl^: we're trying to do it from a softswitch |
| 21:39.17 | drfreeze | CunningPike: that is for the phone. I remember something like 'polycom' for the phone, but it is not working |
| 21:39.28 | drfreeze | Also, I am assuming the username is admin, but that may be wrong too |
| 21:39.34 | CunningPike | drfreeze: Worth a shot :) |
| 21:39.40 | drfreeze | sure |
| 21:39.42 | dlynes_laptop | robl^: i.e. it's only used for routing calls, fax blasting advertisements, and receiving fax for fax 2 email |
| 21:39.53 | drfreeze | failed |
| 21:40.40 | CunningPike | drfreeze: Try 'Polycom' and '456' |
| 21:40.42 | robl^ | dlynes_laptop: ahh! |
| 21:40.56 | luke-jr_work | http://rafb.net/paste/results/k15Aew26.html |
| 21:41.21 | dlynes_laptop | robl^: on our pbxes, we only care about being able to fax through, and receive faxes which will get emailed |
| 21:41.23 | drfreeze | CunningPike: Hey, that worked. Thx |
| 21:41.41 | CunningPike | drfreeze: Try reading the manual, too ;) |
| 21:42.00 | CunningPike | luke-jr_work: If you remove the defaultip entry, does it work? |
| 21:42.56 | drfreeze | CunningPike: Do you know why the 501's would ignore the time offset when configured via the phone, but not when configured via the web interface? |
| 21:42.59 | luke-jr_work | nope |
| 21:43.25 | CunningPike | drfreeze: Hmm - not sure. We use a provisioning server |
| 21:44.28 | dlynes_laptop | drfreeze: the keypad interfaces and web interfaces for most voip phones are flaky at best |
| 21:44.28 | *** part/#asterisk allankardec (n=root@ns1.compuvox.com.br) |
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| 21:50.28 | luke-jr_work | :/ |
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| 21:51.33 | luke-jr_work | ... |
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| 21:52.53 | drfreeze | dlynes_laptop: do you know of a way to get the current config of a phone in XML format? |
| 21:53.30 | drfreeze | In other words, I would like to modify the current config and setup the tftp server with that file instead of a completely new file |
| 21:53.53 | dlynes_laptop | drfreeze: for a polycom? no |
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| 21:53.58 | dlynes_laptop | drfreeze: I don't use polycom |
| 21:54.55 | drfreeze | :) |
| 22:00.06 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 22:01.47 | CunningPike | drfreeze: There isn't a way to do that - although lots of people would like that |
| 22:01.56 | *** join/#asterisk luke-jr_work (n=luke-jr@fl-71-53-155-1.dhcp.embarqhsd.net) |
| 22:02.30 | drfreeze | :( |
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| 22:09.51 | sweeper | link to voice compression protocol comparison matrix? |
| 22:10.06 | sweeper | err |
| 22:10.08 | sweeper | codec |
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| 22:22.21 | CunningPike | drfreeze: My advice is to start with the default files and make the changes you need from there |
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| 22:33.09 | robin_sz | sigh ... fscking BT |
| 22:33.22 | robin_sz | what a totally useless company |
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| 22:48.27 | sweeper | The One Book is tasty ,3 |
| 22:48.29 | sweeper | <3 |
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| 22:58.52 | dlynes_laptop | CunningPike: you can do that with aastra, though:) |
| 22:59.08 | dlynes_laptop | anyways...gotta run |
| 22:59.15 | dlynes_laptop | freaking octel voicemail sucks so bad |
| 23:01.04 | _BOBWEEVER | Is zap show status the best way to see errors on T110P? I am measuring BPV errors on the bert tester but am seeing nothing on the * box? Am I not looking in the correct place? |
| 23:04.59 | *** part/#asterisk BSDTech (n=RNeese@adsl-69-230-170-85.dsl.irvnca.pacbell.net) |
| 23:06.08 | luke-jr_work | any ideas on determining why Asterisk has decided a SIP client is Unauthorized? |
| 23:06.35 | *** join/#asterisk hachi (i=hachi@shego.kuiki.net) |
| 23:06.38 | luke-jr_work | no credentials have changed on either client or server (username, password, etc) |
| 23:08.16 | hachi | hey, I've been patching asterisk's rtp.c myself to fix a RFC2833 packet ordering problem for about 6 months now. Is it in scope to be fixed any time? |
| 23:08.50 | orlok | heh |
| 23:08.56 | *** join/#asterisk jeffik (n=Jeff@pool-71-101-197-41.tampfl.dsl-w.verizon.net) |
| 23:09.03 | orlok | hachi: yeah, it will be fixed after they unfuck the sip header handling |
| 23:09.07 | *** join/#asterisk sigmounte (n=sigmount@www.sighq.net) |
| 23:09.08 | orlok | (yeah, right.) |
| 23:09.13 | file | 1.4 and trunk's handling/producing of RFC2833 is vastly different then 1.2 |
| 23:10.30 | hachi | I'm just tired of patching it when a new version of 1.2 is released, and now I just found out that I have to start building it for 64bit platforms as well |
| 23:10.35 | hachi | so my work just tripled |
| 23:10.41 | file | what's the patch? |
| 23:12.26 | *** join/#asterisk timothy11 (n=chatzill@82-39-188-21.cable.ubr03.benw.blueyonder.co.uk) |
| 23:13.44 | hachi | the patch I have basically ignores packets that step backwards in time |
| 23:14.08 | hachi | like... I'd get start-9 start-9 end-9 start-9 end-9 end-9 for DTMF tones |
| 23:14.12 | hachi | and asterisk would read that as 99 |
| 23:14.17 | hachi | when it's actually just one 9 |
| 23:14.20 | file | yay out of order |
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| 23:17.41 | hachi | if I file a bug regarding this against 1.2, is it just going to get thrown away because 1.4 is 'out soon' ? |
| 23:18.04 | file | lemme look at the RFC2833 in 1.2 |
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| 23:19.06 | hachi | http://hachi.kuiki.net/stuff/rtp.diff |
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| 23:22.05 | *** join/#asterisk wwalker (n=wwalker@pdpc/supporter/sustaining/wwalker) |
| 23:22.56 | wwalker | OK, so can anyone give me a reason to use a Netrake nCite instead of a good server running OpenSER? |
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| 23:25.48 | PutLinuxInIt | i don't know what Netrake nCite is :-( |
| 23:26.37 | file | hachi: post it as a bug and I will see if I can come up with a solution based off the stuff I learned from doing the 1.4/trunk implementation |
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| 23:29.04 | hachi | file: k... uh, where do you do that? |
| 23:29.10 | file | bugs.digium.com |
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| 23:40.09 | wwalker | PutLinuxInIt: it's a "session border controller" much like a big SIP proxy server |
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| 23:43.56 | PutLinuxInIt | thanks wwalker |
| 23:44.36 | PutLinuxInIt | <<----- needs help on setting up a pbx for less than 10 users with asterisk. |
| 23:46.51 | Aboulafia | what kind of help ? (I'm not usefull about tuning... |
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| 23:52.23 | hachi | reported, bug # 8628 |
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| 23:53.32 | Strom_C | PutLinuxInIt: what kind of help do you need? |
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