| 00:00.04 | Juggie | how's todays cvs |
| 00:01.29 | morris | I have never run asterisk before (beyond installing from cvs on my gentoo box and executing asterisk binary (Was able to see a prompt).. problem im faced with at the moment is with debian based machine (ubuntu) when i run asterisk it complains about no /dev/dsp.. ok so sound isnt installed how vital is that? i guess its for the music on hold?.. the other thing is when doing asterisk -r it gives me the version info then segmentation fault |
| 00:02.15 | rcam | Anyone here use Nufone... Can you currently make an outbound call? |
| 00:02.25 | rcam | jbot seen jerjer |
| 00:02.28 | jbot | jerjer <~JerJer@DSL-226.206-rt-bras.che.centurytel.net> was last seen on IRC in channel #asterisk, 2d 10h 50m 14s ago, saying: 'ManxPower: sounds interesting, but i've still never used it'. |
| 00:03.03 | Darien | hmm, I should post to the mailing list I guess |
| 00:03.20 | ManxPower | Juggie, DO you have a libspeex.so.1? |
| 00:03.38 | ManxPower | If so the the lib loader just isn't finding it. THAT'S easy to fix. |
| 00:03.54 | Juggie | manx, i do now.... i just used apt-get to install it and its happy |
| 00:03.57 | Juggie | now i have another problem |
| 00:03.58 | Juggie | May 17 19:52:43 WARNING[20568]: codec_speex.c:211 speextolin_framein: Out of buffer space |
| 00:03.58 | Juggie | May 17 19:52:43 WARNING[20568]: chan_zap.c:4472 zt_write: Frame too large |
| 00:04.05 | rabelais | http://pastebein.ca/12020 does anyone see what's wrong with my config? I get a 404 when I try to dialout |
| 00:04.35 | ManxPower | rabelais, Well for one thing try pastig the URL. |
| 00:05.27 | rabelais | sorry, typo...http://pastebin.ca/12020 |
| 00:06.46 | Juggie | ManxPower, any idea on the warnings? |
| 00:07.26 | ManxPower | Juggie, nope. I use speex for all IAx stuff. |
| 00:08.52 | bkw_ | Juggie, don't use speex |
| 00:09.02 | bkw_ | he's using xlite |
| 00:09.06 | bkw_ | and xlite pads it |
| 00:09.11 | bkw_ | thats the only time I see that error |
| 00:11.18 | tzafrir | morris, the ubuntu asterisk packages are out of date |
| 00:11.38 | ManxPower | bkw_, So asterisk/contrib/thirdparty/spexxilbcfix_xlite.reg doesn't fix that? |
| 00:12.12 | bkw_ | it might |
| 00:12.14 | tzafrir | morris, use the proper debian packages (rebuild them if you need), or build from source |
| 00:12.15 | bkw_ | but last I checked no |
| 00:12.21 | morris | tzafrir...... shall i use the cvs version |
| 00:12.32 | bkw_ | because they pad it |
| 00:12.38 | tzafrir | morris, if you really like bleeding edge |
| 00:12.39 | bkw_ | irts worth a shot |
| 00:12.45 | tzafrir | HEAD is less expected |
| 00:12.46 | morris | i just want it to work ;/ |
| 00:12.49 | morris | hehe |
| 00:12.55 | bkw_ | what to work? |
| 00:13.00 | bkw_ | Updating ... |
| 00:13.25 | tzafrir | morris, as for sound: do you use OSS or ALSA? |
| 00:13.37 | morris | none atm |
| 00:13.39 | morris | i havent set anything up |
| 00:13.55 | tzafrir | well, it's just a warning |
| 00:14.03 | morris | will music on hold still work ? |
| 00:14.14 | bkw_ | you don't need a sound card for MOH |
| 00:14.22 | morris | sweet |
| 00:14.22 | bkw_ | damn I wish I could smack the person that said that |
| 00:14.31 | bkw_ | I would slap them so hard their momma would feel it |
| 00:14.38 | bkw_ | 40% |
| 00:14.41 | bkw_ | 50% |
| 00:14.45 | znoG | i'd slap their momma |
| 00:14.50 | morris | i am their momma |
| 00:14.56 | bkw_ | *SMACK* |
| 00:15.01 | morris | nice |
| 00:15.07 | bkw_ | 90% |
| 00:15.13 | bkw_ | BOOTING.... |
| 00:15.33 | znoG | bkw_: you know much about dring detection in Zap? |
| 00:15.45 | *** join/#asterisk rube2 (~rube@83-131-70-237.adsl.net.t-com.hr) |
| 00:16.04 | tzafrir | I've tried dring detection recently, but I still get unexpected results. |
| 00:16.29 | *** join/#asterisk dmccollum (~dmccollum@eycb01-00-cntnga-69-164-245-72.atlaga.adelphia.net) |
| 00:16.32 | znoG | you also get a range of patterns for the same distinctive ring number? |
| 00:16.41 | tzafrir | yes |
| 00:16.58 | dmccollum | Evening everyone |
| 00:17.09 | dmccollum | Dishwasha Are you around? |
| 00:17.17 | tzafrir | morris, what version do you have (befor eit segfaults) |
| 00:17.49 | znoG | tzafrir: ah, good to know i'm not the only one then :) |
| 00:18.14 | znoG | tzafrir: it also says it should go to the default context if it doesn't match, but i had to explicitly add a dring pattern 0,0,0 for the main number. You also had to do this? |
| 00:18.48 | tzafrir | znoG, I gave up on it for the moment, and decided to study it later |
| 00:19.37 | morris | root@gentoo1 tikka # asterisk -h |
| 00:19.37 | morris | Asterisk CVS-HEAD-03/24/05-00:47:17, Copyright (C) 2000 - 2005, Digium. |
| 00:19.41 | morris | is the one on my gentoo box |
| 00:19.51 | morris | but i need to set it up at work before i do my home one really |
| 00:20.01 | morris | since at home i have no use for it ;p |
| 00:20.07 | znoG | tzafrir: i, unfortunately, need it so i guess i would have to learn how distinctive ring works and play with the code to get it to work. |
| 00:20.11 | znoG | reliably |
| 00:26.57 | morris | checking whether the C compiler (gcc ) works... no |
| 00:26.59 | morris | erm |
| 00:27.01 | morris | WHY? |
| 00:27.01 | morris | lol |
| 00:28.16 | *** join/#asterisk newbien (~e@147.241.33.65.cfl.res.rr.com) |
| 00:28.37 | *** join/#asterisk DEEZED (deezed@adsl-065-006-189-182.sip.bct.bellsouth.net) |
| 00:28.50 | wisdom | Hrm, has anybody worked extensively with the Sipura spa841's ? (as in a reasonably large deployment of them) |
| 00:29.05 | Mavvie | morris: check your config.log |
| 00:30.55 | morris | aparently, ubuntu doesnt come with a compiler as default |
| 00:31.25 | Darien | cute |
| 00:31.56 | morris | yaw |
| 00:32.21 | morris | Mavvie, dont slap me.. where would one normaly find the config.log |
| 00:32.38 | wisdom | nobody ? |
| 00:32.39 | wisdom | Hmm |
| 00:32.54 | Mavvie | morris: in the same directory as where configure was ran. |
| 00:33.13 | morris | ah, it doesnt have configure |
| 00:33.20 | morris | i just did make |
| 00:33.36 | morris | i downloaded tar.gz from asterisk.org |
| 00:34.20 | *** join/#asterisk ManxPower (~eric@adsl-6-105-205.msy.bellsouth.net) |
| 00:34.20 | Mavvie | morris: the directory editline has one. |
| 00:34.54 | Darien | damn, I can't find anything in the documentation about locales |
| 00:35.16 | morris | ah i see |
| 00:35.24 | morris | thanks Mavvie |
| 00:36.03 | *** join/#asterisk DFT (~dft@CPE0040050149d6-CM00080d77ae83.cpe.net.cable.rogers.com) |
| 00:36.07 | morris | what is termcap |
| 00:36.20 | Mavvie | man termcap |
| 00:36.24 | morris | yea |
| 00:36.26 | morris | good point |
| 00:36.27 | morris | brb ;p |
| 00:36.38 | DFT | hello, how can I monitor in realtime what's going on when I try to connect vi sip softphone? |
| 00:36.47 | Mavvie | it describes what your terminal can do, and how to do it. |
| 00:38.01 | xeet2 | anyone know how to determine the ip address of an iaxy? |
| 00:38.19 | xeet2 | doesn't seem to be grabbing an ip through dhcp |
| 00:39.04 | Juggie | is it maybe set to static |
| 00:39.12 | xeet2 | might be, it used |
| 00:39.17 | xeet2 | its |
| 00:39.52 | Juggie | there a button to reset it |
| 00:44.33 | *** join/#asterisk sandnigg0r (~niggerplz@66-55-197-254.gwi.net) |
| 00:45.57 | ManxPower | If anyone wants to talk about QoS issues, join #qos |
| 00:46.26 | Mavvie | is that related to the asterisk source code and documentation? |
| 00:47.59 | xeet2 | haha |
| 00:49.45 | Darien | in an SQL table or an HTML table? |
| 00:49.45 | *** join/#asterisk pigpen (~mark@fw.seamans.cc) |
| 00:49.51 | Mavvie | an HTML table |
| 00:49.59 | Darien | sure there is |
| 00:50.10 | Mavvie | oh! do tell! do tell! |
| 00:50.26 | Darien | oh |
| 00:50.29 | Darien | I don't know what it is |
| 00:50.30 | Darien | ask in #web |
| 00:50.33 | Mavvie | hehe |
| 00:50.35 | Darien | I just know that it can be done |
| 00:50.59 | morris | how do u mean update a table cell |
| 00:51.21 | morris | if u give me a chance to understand ur problem, i could probebly solve it |
| 00:51.59 | Darien | to change the content of an element in HTML using e.g. Javascript |
| 00:52.17 | Mavvie | if I have this: <span onClick="change()" id="boo">boo</span> |
| 00:52.31 | Mavvie | and want to change the boo between the span tags into bar. |
| 00:52.43 | Mavvie | document.getElementById("boo").value="bar"; <- doesn't work (does't complain neither ;-) |
| 00:52.49 | morris | give me 2 seconds |
| 00:52.54 | morris | ill see what i can do |
| 00:52.57 | Mavvie | okies, thanks for your help (in advance) |
| 00:53.10 | Darien | document.getElementById("boo").innerHTML="pee"; |
| 00:53.11 | Darien | try that |
| 00:53.25 | Mavvie | OOOOOOOOOOH! beautiful! |
| 00:53.33 | morris | bastard ;p |
| 00:53.50 | Darien | Mavvie: test it in Firefox and Safari first to make sure it works |
| 00:54.02 | Darien | it's a generally-accepted IE extension |
| 00:54.06 | Mavvie | It works in mozilla. let me check in IE. |
| 00:57.01 | meppl | gute nacht |
| 00:57.09 | meppl | good night |
| 00:57.50 | Darien | hmm |
| 00:57.56 | Darien | Mavvie: the 'proper' way is thus: document.getElementById("boo").firstChild.data = "foo"; |
| 00:59.15 | Mavvie | aha, that one works too. This is making my day so much better! |
| 00:59.20 | Darien | :D |
| 00:59.21 | Darien | <3 me |
| 01:05.46 | xeet2 | uhm, wow |
| 01:06.00 | xeet2 | never seen browser discussion in here before |
| 01:06.22 | xeet2 | can I ask some windows xp questions? =P |
| 01:08.13 | morris | i will answer one ;p |
| 01:08.20 | morris | coz i suck at everything else |
| 01:08.48 | *** join/#asterisk Moc[Toronto] (~mochouina@142.131.66.185) |
| 01:08.53 | Pete_Largo | html, windows, what's next a new steak-on-the-grill recipe? |
| 01:09.05 | Moc[Toronto] | hail |
| 01:09.07 | morris | lol |
| 01:09.25 | morris | Pete_Largo, thats bad of you.. mentioning food when its 2am |
| 01:09.36 | morris | made me starving hungry in about 5 seconds flat |
| 01:09.40 | Pete_Largo | it's never too late to eat :) |
| 01:09.56 | Pete_Largo | besides, it's only 8pm here in Texas |
| 01:10.59 | morris | hehe |
| 01:11.10 | morris | yea spose its not too late to eat.. unless i want more chins |
| 01:11.38 | morris | dont want* whatever |
| 01:11.41 | Pete_Largo | oooh Chinese! |
| 01:11.48 | Pete_Largo | I love Chinese food |
| 01:11.51 | morris | hmm chinese would be cool |
| 01:11.56 | morris | i would settle for pizza tho |
| 01:12.05 | newmedian | Moc you should take advantage of all the nice Torontonian food. |
| 01:13.02 | Pete_Largo | hey newmedian |
| 01:13.17 | newmedian | hey Pete_Largo |
| 01:13.52 | *** join/#asterisk file[mac] (~jcolp@mctn1-3494.nb.aliant.net) |
| 01:14.42 | morris | argh u really have made me want food |
| 01:14.44 | morris | :( |
| 01:17.19 | xeet2 | hey kram |
| 01:17.25 | file[mac] | my sweet sweet krammy boy! |
| 01:17.48 | kram | hi xeet |
| 01:19.27 | MikeJ[Laptop] | it's KRAM! |
| 01:20.16 | xeet2 | how's life @ digium |
| 01:20.32 | *** join/#asterisk shankland (shankland@82-41-86-32.cable.ubr01.dund.blueyonder.co.uk) |
| 01:20.36 | shankland | Hello |
| 01:21.17 | morris | right im gonna sleep |
| 01:21.18 | morris | good night guys |
| 01:21.23 | morris | thanks for help / chat |
| 01:21.30 | Pete_Largo | Polycomm Soundpoint 400 ??? |
| 01:21.34 | Pete_Largo | on ebay |
| 01:22.09 | shankland | haha |
| 01:22.12 | shankland | morris web:D:P |
| 01:22.25 | Pete_Largo | I know Polycom has 300, 500, 600, but I don't see a 400 on the polycom.com web site |
| 01:23.50 | Pete_Largo | but google shows a ton of links so I guess I _don't_ actually know everything :) |
| 01:24.51 | file[mac] | what's the matter bkw? :( |
| 01:25.37 | morris | lol @ shankland |
| 01:26.02 | morris | i was off to bed ;p |
| 01:26.07 | shankland | lol |
| 01:28.38 | *** join/#asterisk CdtDelta (~CdtDelta@dsl081-225-161.chi1.dsl.speakeasy.net) |
| 01:31.46 | *** join/#asterisk netvulture (0@63.174.172.245) |
| 01:32.17 | nestAr | do the TDM400 cards work better than the X100P's? |
| 01:32.20 | newmedian | wouldn't you prefer a nice steak morris? |
| 01:32.35 | newmedian | or chinese? mmmm. |
| 01:33.50 | shankland | :( |
| 01:33.58 | morris | lol |
| 01:34.02 | morris | bastard ;p |
| 01:34.08 | morris | they keep making me hungry mr shankland |
| 01:34.18 | morris | ive STARVING DAMNIT |
| 01:34.18 | morris | lol |
| 01:34.20 | morris | im& |
| 01:34.22 | morris | im * |
| 01:34.25 | shankland | eat then;) |
| 01:34.30 | shankland | mmmm steak is good |
| 01:34.31 | netvulture | hey all, got a question about what the best way is to intergrate 2 sites each with their own asterisk server when it comes to voicemail? Any ideas? I'd prefer to have voicemail boxes local to the users, but still need to be able to forward cross-site. Thanks in advance! |
| 01:34.37 | shankland | mmm domions pizza is good 2:D |
| 01:34.42 | newmedian | Some nice Thai food... Toronto has such good food choices |
| 01:35.16 | shankland | I have pizza sitting down stairs 2 |
| 01:35.17 | shankland | mmm |
| 01:38.35 | netvulture | any ideas out there? |
| 01:40.00 | netvulture | is that question for me shankland? |
| 01:40.12 | shankland | yes |
| 01:40.15 | *** join/#asterisk AjudaBrazil (~AjudaBraz@empresas.acate.com.br) |
| 01:40.38 | netvulture | well - i'm not a complete expert, but savvy enough to hear the answer and know what you mean |
| 01:41.05 | AjudaBrazil | hello people. i need help please.. in asterisk with digium te100p + E1 PRI in Brazil.... any body can help me please? |
| 01:41.13 | AjudaBrazil | ops, 110p |
| 01:41.18 | netvulture | so i guess that would be a yes - i know it well enough |
| 01:42.36 | PTG123 | anyone in here really good with asterisk looking for part time work, possibly turning i nto to full time.. Tech support and basic install stuff. |
| 01:43.02 | Pete_Largo | define 'really good' |
| 01:43.09 | PTG123 | experience |
| 01:43.12 | PTG123 | er experienced |
| 01:43.21 | Pete_Largo | how much experience? |
| 01:43.40 | PTG123 | Dude i don't know :) |
| 01:43.42 | newmedian | ~docs |
| 01:43.43 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org |
| 01:45.58 | Pete_Largo | Well, really good and experience probably counts me out. |
| 01:50.26 | *** join/#asterisk rvhi (~rv@66.175.65.89) |
| 01:51.41 | *** join/#asterisk Markaci (~Markaci@CPE00508de53d8f-CM00080d924284.cpe.net.cable.rogers.com) |
| 01:54.06 | *** join/#asterisk NewSole (dave@i216-58-44-245.avalonworks.net) |
| 01:55.37 | rvhi | kept getting core dump |
| 01:56.25 | rvhi | this is the bt |
| 01:56.26 | rvhi | #8 0x0807d9c9 in pbx_extension_helper (c=0x839c520, context=0x839c67c "dev1", |
| 01:56.26 | rvhi | <PROTECTED> |
| 01:56.26 | rvhi | <PROTECTED> |
| 01:56.26 | rvhi | #9 0x08077748 in ast_pbx_run (c=0x839c520) at pbx.c:1897 |
| 01:56.26 | rvhi | #10 0x0807e221 in pbx_thread (data=0x130c2) at pbx.c:2120 |
| 01:56.28 | rvhi | #11 0x40026e51 in pthread_start_thread () from /lib/libpthread.so.0 |
| 01:56.32 | Qwell | ~pastebin |
| 01:56.33 | jbot | i heard pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca |
| 01:56.37 | NewSole | sounds like me and my last GF... she kept dumping me |
| 01:56.40 | rvhi | oops, sorry |
| 01:56.54 | rvhi | as you can see, exten is mess up |
| 01:57.02 | rvhi | callerid has the wrong pointer |
| 01:57.56 | Mavvie | opener.documentit's more that "data" in frame 10 is wrong I think |
| 02:00.56 | ManxPower | If anyone wants to talk about QoS issues, join #qos |
| 02:01.19 | Qwell | ManxPower: How much are tickets to ManxPower 2005? |
| 02:01.29 | NewSole | lol |
| 02:02.06 | ManxPower | Qwell, They are only available to people in europe. |
| 02:02.19 | Qwell | So I see |
| 02:02.22 | rvhi | mavvie, do you know how to print out data in frame 10? |
| 02:03.40 | Mavvie | frame 10 and then euhm... www.refcards.com -> GDB cheat sheet |
| 02:04.32 | *** join/#asterisk drbrown (~chatzilla@user-0cdv208.cable.mindspring.com) |
| 02:06.44 | AjudaBrazil | anyone from brazil?? |
| 02:08.33 | newmedian | AjudaBrazil could use some assistance configuring his E1 with Embratel (/Telmex). |
| 02:09.56 | AjudaBrazil | yes ;) |
| 02:12.13 | PTG123 | the movie i just rented today is on hbo right now |
| 02:12.15 | PTG123 | grr |
| 02:12.49 | Qwell | haha, that sucks |
| 02:13.30 | PTG123 | oh well atleast am ripping it to my hard drive :) |
| 02:14.59 | *** join/#asterisk JimVanM (~jimvanm@HSE-Toronto-ppp3489845.sympatico.ca) |
| 02:15.46 | blitzrage | heya JimVanM ! |
| 02:16.03 | blitzrage | someone please help Jim, he's a good guy :) |
| 02:16.23 | JimVanM | LOL! Thanks Leif |
| 02:16.25 | *** join/#asterisk doolph (doolph@200.46.148.35) |
| 02:16.37 | doolph | hi |
| 02:16.41 | doolph | nihao |
| 02:18.19 | JimVanM | Anyone connected to a Sonus via PRI (NI2)? The Sonus is complaining about an information element that it doesn't recognize. |
| 02:18.31 | newmedian | code 46 violation |
| 02:18.34 | JimVanM | (conglvl |
| 02:23.22 | kram | hi jim |
| 02:25.14 | Qwell | kram: afternoon |
| 02:25.45 | JimVanM | heya kram |
| 02:26.18 | kram | afternoon awell |
| 02:26.21 | kram | err qwell |
| 02:26.26 | blitzrage | hello everyone! :) |
| 02:28.11 | iq | hi blitzrage |
| 02:30.48 | *** join/#asterisk JimVanM (~jimvanm@HSE-Toronto-ppp3489845.sympatico.ca) |
| 02:32.07 | *** join/#asterisk xeet2 (~xeet3@pbx1.istx.net) |
| 02:32.23 | xeet2 | the iaxy is officially wonderful |
| 02:33.25 | newmedian | orange or dark? |
| 02:33.33 | xeet2 | orange |
| 02:33.40 | xeet2 | haven't gotten my hands on a dark one yet |
| 02:33.52 | Silik0n | anyone have a really good spot for free (as in free not just royalty free) stuff for MOH? |
| 02:33.55 | xeet2 | do they look as cheesy as they do in the images? |
| 02:34.02 | newmedian | Are the new IAXys on the street yet? |
| 02:34.09 | Qwell | Silik0n: find some indy artist sites |
| 02:34.20 | JimVanM | Do they do name resolution? |
| 02:34.40 | xeet2 | jim: no, but you can use backup servers |
| 02:34.58 | newmedian | Silik0n: have you looked at http://creativecommons.org/audio/ |
| 02:35.17 | Silik0n | Qwell: I've been digging for some but havent found anything that said "good free" that was more then a 90 sec song |
| 02:36.01 | Qwell | Silik0n: You didn't say the songs had to be good. Hell, look at most MoH :p |
| 02:36.25 | Silik0n | hell I should just do like everyone else, load my mp3 collection in there heh |
| 02:36.45 | Silik0n | or does anyone know the answer to that? |
| 02:36.59 | Silik0n | or do I have to drive across town and see if I can scam one? |
| 02:37.13 | Qwell | scam one what? |
| 02:37.28 | Silik0n | misfired to wrong channel |
| 02:38.11 | newmedian | Silik0n: Well, if you like Star Wars then you could always use http://www.panicstruckpro.com/4Marc/Revelations_Soundtrack.zip from Star Wars: Revelations. That's a bit geeky but timely, considering how close the next Star Wars film is. ;) |
| 02:38.26 | Silik0n | yeah |
| 02:38.44 | Silik0n | already got tix for friday showing |
| 02:39.25 | newmedian | Same here. Hope it is better than first two. I remember seeing the first showing of the originals, back in the day. |
| 02:40.35 | Qwell | NewSole: Whats the problem? |
| 02:42.15 | *** join/#asterisk SarahEmm (~sarahemm_@MTL-ppp-158713.qc.sympatico.ca) |
| 02:43.13 | NewSole | been waiting on someone for doing some work and time is critcal.... |
| 02:43.20 | *** part/#asterisk sivana (~sivana@mixdown.ca) |
| 02:43.28 | *** join/#asterisk sivana (~sivana@mixdown.ca) |
| 02:43.36 | Qwell | NewSole: like? |
| 02:43.38 | NewSole | and he said for last few day we will do it after work... and no show |
| 02:43.52 | Qwell | Hmm, sounds like somebody I know |
| 02:44.38 | Qwell | NewSole: What type of work you talking? |
| 02:45.19 | NewSole | getting system stable... clean it up and do up dial codes |
| 02:47.00 | *** join/#asterisk TheEmperor (~TheEmpero@203.114.48.52) |
| 02:47.33 | blitzrage | ariel_: please turn off your auto announce |
| 02:47.52 | Qwell | NewSole: stable how? "dial codes"? |
| 02:48.23 | ManxPower | If anyone wants to talk about QoS issues, join #qos |
| 02:48.31 | NewSole | well it works... but some providers I call I only get 1way audio |
| 02:49.39 | Silik0n | are you having NAT issues? |
| 02:49.52 | NewSole | no its IAX |
| 02:50.03 | Silik0n | really |
| 02:50.07 | Silik0n | thats interesting |
| 02:50.24 | Qwell | which way is the audio? |
| 02:50.39 | NewSole | I can hear them but they can not hear me |
| 02:50.55 | Qwell | all phones do the same thing? |
| 02:51.03 | TheEmperor | hello |
| 02:51.04 | NewSole | yup |
| 02:51.35 | TheEmperor | i've got a pri connected to an e1 card, zapata.conf is configured, but i get this error message when i call in: Extension '21682888' in context 'incoming' from '0321640263' does not exist. Rejecting call on channel 4, span 1 |
| 02:51.43 | file | TheEmperor: read it |
| 02:51.49 | TheEmperor | dialling out is not a problem.. |
| 02:51.53 | file | read the error |
| 02:52.33 | TheEmperor | file:i've got 30 channels on the pri, so i need to configure a context for each number? |
| 02:52.37 | TheEmperor | is that what i am doing wrong? |
| 02:52.47 | file | you are sending calls into the context incoming |
| 02:52.58 | TheEmperor | yes.. |
| 02:53.01 | file | well for the number 21682888, there are no extension entries in that context to handle it |
| 02:53.28 | TheEmperor | i thought that if you put in s,1, then it should handle it? |
| 02:53.33 | file | no |
| 02:53.41 | TheEmperor | or since it is a pri, i need to specify each number and where it goes? |
| 02:53.41 | file | s,1 is only when the dialed number is not known |
| 02:53.45 | TheEmperor | i see... |
| 02:54.17 | file | correct, or use a pattern to match it |
| 02:54.17 | *** join/#asterisk likwid-- (likwid@nc-69-68-74-187.dyn.sprint-hsd.net) |
| 02:54.17 | TheEmperor | what kind of pattern could I use? |
| 02:54.17 | Qwell | 216XXXX? |
| 02:54.17 | Qwell | only you know the answer to that |
| 02:54.19 | file | Qwell: it's 5 Xs actually... |
| 02:54.25 | TheEmperor | ok i get it now.. |
| 02:54.25 | file | if 216 remains constant |
| 02:54.36 | Qwell | 8 digit number? |
| 02:54.43 | file | Qwell: I found it weird too |
| 02:54.45 | Qwell | E1, I know nothing about |
| 02:54.49 | TheEmperor | 8 numbers correct |
| 02:56.49 | file | I don't understand how we can make the error messages any more clearer |
| 03:01.57 | TheEmperor | what i've done now is to specify it as exten => 21682XXX,1,Answer() but it still says that that extesnsion does not exist :( |
| 03:02.13 | *** join/#asterisk Kernel_Core (Raph@47.229.dial-up.xter.net) |
| 03:02.13 | Qwell | _21etc |
| 03:02.18 | Qwell | you need the _ |
| 03:02.28 | Qwell | read the docs to find out why |
| 03:02.38 | TheEmperor | ? |
| 03:03.34 | newmedian | You could always _X. it? |
| 03:03.39 | xeet2 | TheEmperor: instead of 21682XXX, you need _21682XXX, |
| 03:03.46 | TheEmperor | o |
| 03:03.56 | TheEmperor | why is that? |
| 03:04.02 | xeet2 | regular expressions |
| 03:04.10 | blitzrage | thats how pattern matching works |
| 03:04.13 | Kernel_Core | hi all , I am useing Xten SiP soft phone, I connect to asterisk ,(GSM Codec ) DTMF works , but when I want to pass DTMF to another cisco sip through asterisk , it doesn't work ! what is the solution ?! |
| 03:04.18 | blitzrage | _ == pattern matching |
| 03:04.23 | Qwell | I said to read the docs to find out why...come on now |
| 03:04.31 | blitzrage | Qwell: oh.... sorry :D |
| 03:04.32 | TheEmperor | ok.. |
| 03:04.35 | xeet2 | hehe, sorry qwell |
| 03:04.40 | Qwell | nah, not you guys |
| 03:04.55 | Qwell | he asked a dumb question after I explicitly told him what to do, and where to look |
| 03:05.04 | blitzrage | Qwell: yah - gotta learn :) |
| 03:05.19 | blitzrage | welp... bed time, lates |
| 03:06.04 | Kernel_Core | any suggestion? |
| 03:06.50 | MikeJ[Laptop] | Kernel_Core, fix your dtmf settings on your devices |
| 03:07.37 | Kernel_Core | MikeJ[Laptop]: what should I set for cisco !? dtmfmode=rfc2833 is it correct ?! |
| 03:08.25 | MikeJ[Laptop] | check the wiki for the specific device, there are config examples out ther |
| 03:08.43 | Kernel_Core | OK! |
| 03:08.52 | MikeJ[Laptop] | and make sure the device is set to the same thing as asterisk |
| 03:09.10 | MikeJ[Laptop] | if you tell asterisk rfc, but the device is set to inband it will not work well |
| 03:09.43 | Kernel_Core | yea , it seems |
| 03:10.18 | *** join/#asterisk santiago (~santiago@63.245.86.248) |
| 03:11.09 | *** part/#asterisk SarahEmm (~sarahemm_@MTL-ppp-158713.qc.sympatico.ca) |
| 03:12.20 | |Vulture| | znoG: there is a thing in the wiki about detecting and handling it? |
| 03:12.21 | |Vulture| | urg |
| 03:12.27 | |Vulture| | hit the up arrow |
| 03:12.40 | |Vulture| | Anyone ever see a problem where gsm files just stop playing |
| 03:12.45 | |Vulture| | Tones play fine |
| 03:12.50 | |Vulture| | its over Zap and SIP |
| 03:14.13 | *** join/#asterisk Nukemizer (~Nuke@66.237.85.58.ptr.us.xo.net) |
| 03:18.56 | *** join/#asterisk harryvv (~leonardo@S010600a0c93f6f7e.vs.shawcable.net) |
| 03:19.15 | *** join/#asterisk Sedorox (brandon@sedorox.staff.smartserv) |
| 03:19.31 | *** join/#asterisk Inv_arp (junya@adsl-3-248-195.mia.bellsouth.net) |
| 03:20.37 | harryvv | my cell phone carrier just droped its per min rate from 30 cents for each min to 30 cents for 2 min then 5 cents every min after. |
| 03:20.48 | harryvv | thay really want the biz :) |
| 03:23.45 | *** join/#asterisk CdtDelta_PM (~CdtDelta_@dsl081-225-161.chi1.dsl.speakeasy.net) |
| 03:27.41 | NewSole | hmm |
| 03:28.25 | Nukemizer | I am trying to create a dial plan to hangup on particular Caller ID, but can not figure out what syntax I have wrong Can anyone help ? |
| 03:28.38 | Nukemizer | exten => s,1,GotoIf($[${CALLERIDNUM}= 5551212]?Hangup) |
| 03:28.57 | *** join/#asterisk NewSole (dave@i216-58-44-245.avalonworks.net) |
| 03:31.37 | newmedian | Nukemizer, see http://www.voip-info.org/tiki-index.php?page=Asterisk%20config%20extensions.conf for ex-girlfriend example |
| 03:31.48 | Nukemizer | lol |
| 03:34.47 | nestAr | exten => s/_2125551212,1,Hangup |
| 03:35.13 | Nukemizer | oh my.. that easy.. dang i was making that WAY to hard |
| 03:35.17 | nestAr | :) |
| 03:35.21 | Nukemizer | thank you both ! |
| 03:35.37 | nestAr | no problem |
| 03:35.54 | TheEmperor | what does s/ do? |
| 03:38.11 | NewSole | hi folks |
| 03:38.12 | newmedian | s is start, /_blahblah matches the caller ID... see the examples around the ex-girlfriend example, for example. |
| 03:40.46 | *** join/#asterisk Silik0n (~krice@newrso.suspicious.org) |
| 03:47.24 | *** join/#asterisk Silik0n[laptop] (~ken@12-219-156-206.client.mchsi.com) |
| 03:47.29 | *** join/#asterisk matobago (~matobago@einstein.transtelco.com.mx) |
| 03:48.16 | matobago | does anyone knows how could i see who im calling in a cisco 7940/7960 |
| 03:48.18 | matobago | ??? |
| 03:48.36 | loud | mm on the screen ? |
| 03:48.43 | matobago | yeap |
| 03:48.55 | matobago | like a reverse callerid |
| 03:49.06 | Moc[Toronto] | anyone live in toronto ? |
| 03:49.39 | matobago | any ideas? |
| 03:49.42 | newmedian | anyone else? |
| 03:49.42 | Moc[Toronto] | are the train going to strike tomorow ? |
| 03:49.45 | *** join/#asterisk Silik0n (~krice@newrso.suspicious.org) |
| 03:49.49 | Moc[Toronto] | I mean operator |
| 03:49.58 | DFT | I'm in the GTA |
| 03:50.20 | DFT | and I have to drive downtown in the morning too..blah |
| 03:50.35 | newmedian | Well, Moc, do you mean http://www.pulse24.com/News/Top_Story/20050517-003/page.asp |
| 03:51.23 | Godsey | might anyone know how to configure a pap2-na properly to have the message waiting lamp light up on new voicemail? |
| 03:51.39 | doolph | how can I forward an extension to another? |
| 03:51.46 | Godsey | I can make it ring the phone every once in a while, but not light the lamp |
| 03:51.55 | Godsey | the lamp on my polycom phones work just fine tho |
| 03:51.56 | newmedian | Moc, where are you, roughly? From reading that, TTC transit seems to be okay, just Go Transit stopping. |
| 03:53.16 | Moc[Toronto] | Im in front of the Toronto Rail station |
| 03:53.32 | doolph | who want to test with me |
| 03:53.47 | Moc[Toronto] | I didnt saw much ppl |
| 03:54.07 | newmedian | Moc: Are you planning on leaving the city by train, or going outside of the core? The core TTC transit I think is fine, just the trains (and Go trains). First I've heard of it, anyway, I've had my head down. |
| 03:54.21 | Moc[Toronto] | im supose to leave friday by train |
| 03:54.43 | newmedian | Moc: not sure how fast this will be resolved. You may want to look into alternative transit. |
| 03:54.47 | DFT | you'll probably be fine, I have to drive into the core in the morning |
| 03:55.07 | newmedian | Uncertainty. It's what's for breakfast. |
| 03:55.07 | DFT | but newmedian has a point..this is a labour dispute |
| 03:55.31 | DFT | hang tight for a few more minutes..then tune into city.tv |
| 03:55.53 | DFT | they'll have an announcement for sure..if not city.tv cp24 will have it covered |
| 03:55.57 | newmedian | Yes.. after midnight. Cable Pulse 24 might also be a good repetitive news venue to find out the status of that. |
| 03:56.03 | *** join/#asterisk mutilator (WebChat@i.think.napoleon.dynamiteblows.com) |
| 03:57.10 | Moc[Toronto] | I'll stay here ;) |
| 03:57.21 | DFT | Probably a good idea |
| 03:57.25 | Moc[Toronto] | in the hotel all inclusive by buisness.. |
| 03:57.31 | Moc[Toronto] | (the only reason keeping me here) |
| 03:58.22 | NewSole | E-911 is Fun |
| 03:59.03 | mutilator | fun?! |
| 03:59.24 | mutilator | ehm .. improbable at current is more like it |
| 04:00.02 | newmedian | cat /dev/random > myaddress |
| 04:00.04 | Moc[Toronto] | it a crapy old hotel, but atless I got highspeed |
| 04:01.07 | doolph | hey there's any idea to translate all voice/sounds to spanish |
| 04:02.00 | newmedian | Moc if you walk West by about 3-4 blocks, then North 2-4 blocks, you'll be around dance central. Or walk West another 2-3 blocks and North for 10 mins and you'll be in Chinatown. Roughly. |
| 04:08.58 | Moc[Toronto] | no train strike |
| 04:09.26 | Moc[Toronto] | ok well sleep time, |
| 04:09.56 | newmedian | Moc, if you're in for a late night snack (or any time), I can recommend this restaurant in Chinatown: http://www.toronto.com/profile/225473 |
| 04:10.15 | doolph | omg how big is l_ipp_ia32_itanium_p_4_1.tar |
| 04:10.16 | newmedian | Swatow: http://www.google.com/local?q=%2Bswatow+spadina&hl=en&lr=&sa=G&near=toronto,+ontario&sc=1&radius=0&latlng=43723057,-79392486,5129738227890094426 |
| 04:11.05 | Moc[Toronto] | hehe thanks will check it |
| 04:11.42 | newmedian | Moc do you like coffee? |
| 04:12.17 | *** join/#asterisk mes (~mes@70.66.246.248) |
| 04:13.59 | newmedian | I can recommend this place (Casa Acoreana) for Coffee. They have a little Cafe, but the idea is you can buy a large bag (5lb?) of Blue Mountain Blend coffee beans for around $40 CDN (cash only). http://www.google.com/local?q=%2Bcasa+acoreana&hl=en&lr=&near=toronto,+ontario&sc=1&radius=0&latlng=43723057,-79392486,9529158991227841209 |
| 04:14.05 | *** join/#asterisk Qorky (~Pooa@dsl-202-72-146-104.wa.westnet.com.au) |
| 04:15.43 | newmedian | The Casa Acoreana is within a few minutes walking distance from Swatow. :) |
| 04:17.13 | newmedian | damn. Now I want some Singapore Style noodles from Swatow. :( |
| 04:19.29 | *** join/#asterisk [hC] (~hardcore@c-65-34-203-137.hsd1.fl.comcast.net) |
| 04:22.45 | *** part/#asterisk trig (~jb@xob.neospire.net) |
| 04:30.40 | TheEmperor | ,. |
| 04:31.20 | DFT | newmedian: have ever tried Pho 88 on Spadina? |
| 04:36.34 | newmedian | DFT: Don't think I have. You like? |
| 04:36.38 | *** join/#asterisk PTG123 (~PTG123@ip68-106-24-139.ph.ph.cox.net) |
| 04:38.40 | NewSole | ..,,oO|oo,,.. |
| 04:40.01 | PTG123 | __ooOoo____0o0____ooOoo___ |
| 04:40.11 | NewSole | .^,,oO~|~Oo,,^. |
| 04:40.46 | newmedian | you letting bubbles go in the bath again? |
| 04:41.14 | NewSole | na... my chair keeps squeeking |
| 04:42.23 | *** join/#asterisk riksta (~rick@host81-155-216-40.range81-155.btcentralplus.com) |
| 04:43.34 | `Sauron | Pho 75 on Lamar is good, though |
| 04:44.25 | *** join/#asterisk doolph (doolph@200.46.148.35) |
| 04:44.34 | doolph | anyone have installed speex ? |
| 04:45.26 | `Sauron | somehow, I doubt we're talking about the same city, though |
| 04:46.11 | *** join/#asterisk scardinal (~supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 04:46.24 | TheEmperor | someone was telling me how you need serious hardware to conduct a voip business like Quintum, is Asterisk serious hardware for voip? |
| 04:46.41 | newmedian | Toronto. Although Montreal has excellent choices. |
| 04:47.03 | MikeJ[Laptop] | TheEmperor, depends on what you are doing |
| 04:47.10 | MikeJ[Laptop] | and how you do it. |
| 04:47.21 | newmedian | It's not the size, it's how you use it. |
| 04:47.30 | TheEmperor | Yeah, I was just wondering |
| 04:47.39 | TheEmperor | I mean, I have had all but success with Asterisk |
| 04:47.41 | MikeJ[Laptop] | newmedian, is that what people tell you... they lied |
| 04:47.51 | TheEmperor | why would I need Quintum stuff? |
| 04:48.33 | MikeJ[Laptop] | TheEmperor, asterisk is not a proxy... so you can do much higher load on a real proxy, if what you are trying to do is proxy. |
| 04:48.33 | `Sauron | newmedian: Ah. Austin here. |
| 04:49.21 | *** join/#asterisk shido6 (~greg@d57-87-253.home.cgocable.net) |
| 04:49.55 | `Sauron | asterisk isn't hardware |
| 04:50.01 | TheEmperor | MikeJ[Laptop] well not yet anyway.. |
| 04:50.08 | shido6 | erf? |
| 04:50.22 | TheEmperor | if I just wanted to do calling card applications, do I need a proxy? |
| 04:50.27 | MikeJ[Laptop] | it's squisywhere |
| 04:50.34 | newmedian | squishy? |
| 04:50.38 | MikeJ[Laptop] | hehe |
| 04:50.56 | MikeJ[Laptop] | not hard, but not quite soft :D |
| 04:50.56 | newmedian | It's * squishy good |
| 04:51.17 | newmedian | firm? firmware? |
| 04:51.54 | MikeJ[Laptop] | firmware is for hardware, squishyware is for software? |
| 04:52.08 | MikeJ[Laptop] | what's flacidware for ? |
| 04:52.33 | newmedian | nowhere? |
| 04:52.50 | MikeJ[Laptop] | so, anyone here use asterisk.. I just saw it on /. and it looks really cool ;) |
| 04:53.47 | MikeJ[Laptop] | can you really make calls for free on it.... Can somone tell me eveything I need to do.. I don't like to read docs and i figgured I'd get quicker answers just asking anyway |
| 04:53.47 | newmedian | Asterisk? Never heard of it. I thought this channel was about pirates. Av Ast! |
| 04:54.15 | newmedian | :) |
| 04:54.36 | MikeJ[Laptop] | hehe... .this channel cracks me up... so bad I can't handle it usually |
| 04:54.39 | Nukemizer | Jaffa KREE |
| 04:54.44 | newmedian | Maybe a split into #newtoasterisk and #asteriskathome might be in order? |
| 04:55.10 | MikeJ[Laptop] | hey, I love telling newbies to install *@home.. |
| 04:55.17 | MikeJ[Laptop] | I think it is a great learning tool |
| 04:55.20 | *** join/#asterisk hellop (~LeeHarvey@cpe-70-93-40-171.hawaii.res.rr.com) |
| 04:55.30 | hellop | ack |
| 04:55.48 | newmedian | $6.95 Clone card + *@home = ... questions |
| 04:55.48 | MikeJ[Laptop] | and it usually gets them up to speed quicker than asking 10million questions in #asterisk |
| 04:56.07 | MikeJ[Laptop] | ummm.. the zaptel detection stuff is good on AAH. |
| 04:56.22 | shido6 | Bullshit |
| 04:56.28 | newmedian | I think *@home is useful to show someone a complete "working" system; They can look at the config files, get a sense of things, and when they are ready they can wipe it and install from scratch. |
| 04:56.33 | MikeJ[Laptop] | I said good, not great |
| 04:56.41 | MikeJ[Laptop] | but it picks up an x100p fine |
| 04:56.49 | shido6 | I should paypal those nuts money because asterisk@home has sent me a ton of students |
| 04:56.50 | hellop | When compiling asterisk on CLE266, 2.6 kernel: Seqmentation fault, Leaving directory /usr/src/asterisk/funcs. Any ideas? Reinstall OS? |
| 04:56.53 | newmedian | They can get it going within an hour or so following tutorials on the 'net. |
| 04:57.00 | MikeJ[Laptop] | yeah, that's what I am saying |
| 04:58.01 | `Sauron | AAH?\ |
| 04:58.05 | `Sauron | s/\// |
| 04:58.09 | `Sauron | <PROTECTED> |
| 04:58.10 | hellop | Go get some AAH |
| 04:58.12 | *** part/#asterisk santiago (~santiago@63.245.86.248) |
| 04:58.17 | doolph | <MikeJ[Laptop]> hey, I love telling newbies to install *@home.. |
| 04:58.17 | doolph | <MikeJ[Laptop]> I think it is a great learning tool |
| 04:58.21 | doolph | heh |
| 04:58.32 | MikeJ[Laptop] | I don't use it.... |
| 04:58.38 | doolph | why not |
| 04:58.41 | doolph | it's a good tool |
| 04:59.02 | hellop | A quick and dirty guide to making an Asterisk PBX on Debian GNU/Linux. <- is on the wiki, but link is down. Anyone got it? |
| 04:59.06 | `Sauron | I use * @ elsewhere |
| 04:59.22 | MikeJ[Laptop] | I usually say, install *@home, use it for a few days till you are sick of it's limitations, and you will be past all the basic newbie questions if you spend some real time studying how the dialplan works |
| 04:59.46 | MikeJ[Laptop] | doolph, it's fine if you have not run into the limitations. |
| 04:59.59 | doolph | i hacked it already |
| 05:00.12 | MikeJ[Laptop] | I personally am not a fan of how they do the stuff they do in AGI... |
| 05:00.14 | doolph | its pretty cool admin tool |
| 05:00.16 | hellop | fine fine, I'll get it |
| 05:00.18 | newmedian | I think the logical progression goes: a) (optional) has anyone made a Zap card themselves, b) I've install AAH and can't dial in/out/to my provider/can't hear sound one way, c) I don't understand the docs, d) I've installed from scratch but I can't get x working, e) helping newbies, f) ... well, you get the idea. |
| 05:00.35 | doolph | and it help me understand lot of things that i didnt know |
| 05:00.57 | MikeJ[Laptop] | ok, as far as that goes, doolph prooves my point |
| 05:01.21 | MikeJ[Laptop] | then again he is in #asterisk at this hour so he is questionable anyways |
| 05:01.23 | MikeJ[Laptop] | :D |
| 05:01.35 | newmedian | it's only 01:04 EST here |
| 05:01.54 | doolph | i just recompiled asterisk for speex codec |
| 05:02.06 | doolph | it seems to be good but no hardware support |
| 05:02.08 | newmedian | speex no evil? |
| 05:02.15 | MikeJ[Laptop] | shido6, sent you students cuz AAH screwed them up so bad, or cuz they wanted to figure out how to do more and couldn't make the jump to REAL asterisk? |
| 05:02.42 | doolph | heh i wanted my show translation table complete |
| 05:02.51 | doolph | d |
| 05:03.31 | doolph | there's any softhpone that support g723 or g729 |
| 05:03.48 | newmedian | Xten's eyeBeam has speex |
| 05:03.48 | MikeJ[Laptop] | not free ones |
| 05:04.06 | MikeJ[Laptop] | cuz g729 costs money |
| 05:04.16 | doolph | then g723 |
| 05:04.31 | MikeJ[Laptop] | I thought the paid xten stuff had 729, doesn't it? |
| 05:04.38 | doolph | yes |
| 05:04.44 | doolph | but it cost $50 |
| 05:04.52 | doolph | with that i can buy a sipura of 2 lines |
| 05:04.57 | newmedian | http://www.xten.com/index.php?menu=products&smenu=eyebeam |
| 05:04.59 | MikeJ[Laptop] | yeah, not gunna find a free one legally |
| 05:05.03 | newmedian | I have a paid eyeBeam. |
| 05:05.14 | doolph | send it me |
| 05:05.20 | doolph | i just want to test something |
| 05:05.21 | doolph | heh |
| 05:06.11 | *** join/#asterisk implicit (~implicit@ip68-5-148-1.oc.oc.cox.net) |
| 05:06.14 | doolph | can you |
| 05:08.10 | MikeJ[Laptop] | explicit has left #asterisk |
| 05:08.13 | MikeJ[Laptop] | :) |
| 05:08.56 | Godsey | I'm bummed, I have some cordless phones w/ an answering machine that I don't use |
| 05:09.00 | newmedian | I've not tried all the codecs to verify (use/passthrough), but in the eyeBeam menu my enabled codecs are: G722.2 Wideband, Speex Wideband, DVI4 Wideband, G711 uLaw, G711 aLaw, GSM, G729, EVRC, iLBC, Speex, DVI4, L16 PCM Wideband |
| 05:09.10 | Godsey | the VMWI only turns on with it's own answering machine |
| 05:09.16 | Godsey | not via the SPA-2000 |
| 05:09.25 | *** join/#asterisk ds1 (ds@ool-4352e324.dyn.optonline.net) |
| 05:09.42 | ds1 | how does one pronounce "G.711" .. is it G dot seven eleven? |
| 05:10.20 | doolph | g711 |
| 05:13.09 | *** join/#asterisk rue_mohr (~mohr@d154-20-50-233.bchsia.telus.net) |
| 05:13.29 | *** part/#asterisk rue_mohr (~mohr@d154-20-50-233.bchsia.telus.net) |
| 05:14.27 | *** join/#asterisk syslod (~yurplsl@65.114.15.70) |
| 05:17.32 | *** part/#asterisk ds1 (ds@ool-4352e324.dyn.optonline.net) |
| 05:17.37 | newmedian | So far Xten seems to have all the "eye candy"; Firefly and other Soft Phones seem... clunky. |
| 05:18.38 | *** join/#asterisk spiffs (~crux@user-12lmu4a.cable.mindspring.com) |
| 05:18.44 | spiffs | hi |
| 05:18.49 | spiffs | hi |
| 05:19.01 | syslod | hello |
| 05:19.10 | spiffs | got a question |
| 05:19.48 | spiffs | i'm looking to buy a digium x100p |
| 05:20.15 | spiffs | I'm wondering if combined with asterisk, will this do iax and the ulaw codec? |
| 05:20.29 | Qwell | You don't need an x100p to do iax or ulaw... |
| 05:20.53 | spiffs | what's a cheaper alternative? |
| 05:20.58 | Qwell | no hardware at all |
| 05:21.03 | spiffs | i need to hook this up to my alarm system |
| 05:21.14 | shido6 | to do what |
| 05:21.20 | shido6 | pin # verification? |
| 05:21.27 | spiffs | to send contact id alarm messages over iax |
| 05:21.32 | spiffs | nextalarm.com supports it |
| 05:21.40 | spiffs | i'm using a borrowed s100I right now |
| 05:21.49 | *** join/#asterisk kb1_kanobe (~krisbouti@h24-207-80-55.cst.dccnet.com) |
| 05:22.14 | syslod | Anyone got a QOS script laying around??? |
| 05:22.51 | spiffs | so will an x100p replace it for this purpose? |
| 05:22.58 | newmedian | syslod: that sounds like a ManxPower question, but I think ManxPower is alseep |
| 05:23.33 | *** join/#asterisk spiffs (~crux@user-12lmu4a.cable.mindspring.com) |
| 05:23.51 | h3x | spiffs: I bought a ethernet card for my DSC alarm panel |
| 05:23.52 | kb1_kanobe | syslod: Google for 'lartc howto' |
| 05:24.03 | h3x | it just sends the alarms over the internet |
| 05:24.21 | doolph | anyone here can call me with fwd? |
| 05:24.28 | kb1_kanobe | h3x: what's the part number? I've only seen the x-10 interfaces before... |
| 05:24.38 | spiffs | h3x: right, but i need the alarm messages from the ring and tip pins of the alarm panel to the pc |
| 05:24.40 | h3x | its called T-Link |
| 05:24.53 | h3x | why? |
| 05:25.04 | spiffs | the alarm panel itself doesn't support ip |
| 05:25.18 | h3x | ok well its not really gonna work over voip |
| 05:25.41 | newmedian | T-Link hmm? interesting. I used to work with DSC alarms a long while back. |
| 05:25.45 | spiffs | nextalarm has an iax pbx, and you directly connect to it with a digium |
| 05:25.59 | spiffs | thats my goal |
| 05:26.07 | spiffs | i'm using a s100I right now, works great |
| 05:26.13 | h3x | they do? can that be CE certified? |
| 05:26.26 | spiffs | probably not |
| 05:26.33 | kb1_kanobe | spiffs: perhaps a voice-dialer, such a protalk alarm reporting unit (http://www.barnett-engg.com/Pages/quote.html)? |
| 05:27.38 | spiffs | i want to know if the x100p can do iax with asterisk |
| 05:27.41 | spiffs | to send these messages |
| 05:27.44 | h3x | well id suggest using a standalone adapter like they suggest |
| 05:27.46 | spiffs | those cards are cheap |
| 05:27.49 | h3x | coz if your shit goes down you are screwed |
| 05:28.04 | Qwell | figuratively, and likely literally |
| 05:28.07 | spiffs | i intend to run it on a stable OS on an UPS |
| 05:28.24 | h3x | well you cant use a x100p |
| 05:28.28 | h3x | coz thats the opposite of what you need |
| 05:28.48 | spiffs | whys that? |
| 05:28.58 | h3x | its fxo you need fxs |
| 05:29.30 | Qwell | "cheaper alternative to an x100p", because your familys life is only worth $12 |
| 05:29.42 | h3x | hahaha |
| 05:29.43 | spiffs | heh, its just for my stuff |
| 05:30.10 | spiffs | i'll just buy a s100i then :P |
| 05:30.13 | h3x | anyway |
| 05:30.53 | doolph | how many fwd numbers has |
| 05:31.02 | doolph | 5-6? or 5-7? |
| 05:31.15 | Qwell | 3, 5, 6, 7 in the near future |
| 05:31.49 | doolph | 3? |
| 05:31.49 | Qwell | or more, actually |
| 05:32.01 | Qwell | anywhere from 0 to oo |
| 05:32.12 | doolph | what's your number |
| 05:32.14 | spiffs | whats the cheapest fxs card? |
| 05:32.15 | Qwell | 6 |
| 05:32.25 | doolph | can i call you |
| 05:32.39 | Qwell | sure, you'll just get my voicemail though |
| 05:32.54 | doolph | ummm |
| 05:32.57 | doolph | i want to test |
| 05:33.01 | Qwell | test away |
| 05:33.03 | doolph | G723 >> G711 |
| 05:33.40 | doolph | ummm it has g711 default |
| 05:33.56 | Qwell | What does? |
| 05:34.14 | Qwell | hmm |
| 05:34.26 | doolph | ok |
| 05:34.27 | doolph | it works |
| 05:34.36 | Qwell | "The person at extension blah, blah, blah, is unavailable." is it much of a hack to change it to "The person at this extension"? |
| 05:34.53 | jayk_ | i'm trying to change the outside line number from 9 to 8 |
| 05:35.07 | jayk_ | when i hit 8, i get a weird pulsed tone..does anybody know how i can change this? |
| 05:36.11 | doolph | edit your extension.conf |
| 05:36.29 | doolph | you must have some exten = _8XXXX |
| 05:37.32 | Qwell | jayk_: ignorepat => |
| 05:39.20 | jayk_ | i have ignorepat => 8 |
| 05:41.24 | rvhi | hi, someone helps me. so desparate now, * core dump 3 times today |
| 05:42.53 | *** part/#asterisk spiffs (~crux@user-12lmu4a.cable.mindspring.com) |
| 05:43.38 | kb1_kanobe | rvhi: which version (head or stable?) also, under what conditions? |
| 05:45.02 | *** join/#asterisk nine76 (~t00r@cpe-69-135-184-24.woh.res.rr.com) |
| 05:47.07 | nine76 | if anyone knows a reason why one party would be able to hear the other party,but not be able to speak to them,i would really appreciate help. as the "other party" im trying to test with is vacationing in the philippines,and i dont have much time till he gets broed and wonders off:-/ |
| 05:47.33 | nine76 | same results,using x-pro and a sipura,even tried 3 different asterisk servers |
| 05:48.37 | nine76 | thought i covered all the bases:( |
| 05:49.04 | kb1_kanobe | nine76: one way sip audio usually turns out to be a firewall or nat problem, however I have no experience in that area. |
| 05:50.56 | doolph | nine76 try all clients running the same codec |
| 05:51.15 | doolph | but really it should work even it doesnt |
| 05:51.24 | doolph | it seems a firewall problem |
| 05:52.05 | doolph | check rtp.conf |
| 05:57.04 | Qwell | nat=yes, canreinvite=no ? |
| 05:58.20 | *** join/#asterisk Veesmooth (~veegee@bgp505102bgs.verona01.nj.comcast.net) |
| 05:58.29 | *** part/#asterisk Veesmooth (~veegee@bgp505102bgs.verona01.nj.comcast.net) |
| 05:58.31 | *** join/#asterisk iCEBrkr (icebrkr@chrome.cyberdyne.org) |
| 06:00.23 | nine76 | yes to all the above...I tried calling him after a quick extensions change,and when i call him,it works fine |
| 06:00.38 | nine76 | so,since he actually heard my voice hes giving me a few more minutes:) |
| 06:02.43 | *** join/#asterisk Veesmooth (~veegee@bgp505102bgs.verona01.nj.comcast.net) |
| 06:02.50 | Veesmooth | hey everyone |
| 06:02.55 | Veesmooth | anyone up for a question |
| 06:03.02 | Qwell | I could go either way |
| 06:03.06 | nine76 | checking...yeah as I thought,canreinvite=no,nat=yes,rings fine...gets dtmf fine,alls fine except audio |
| 06:03.41 | Veesmooth | who is good with vnc server? |
| 06:03.51 | Qwell | good with it? it's fairly simple... |
| 06:03.56 | Veesmooth | i know |
| 06:04.00 | Veesmooth | but im having a problem |
| 06:04.07 | *** join/#asterisk bjohnson (~bjohnson@66.11.188.191) |
| 06:04.09 | Veesmooth | im pretty good and know the basic commands and stuff with it |
| 06:04.13 | Qwell | This isn't really the right place for it |
| 06:04.19 | *** part/#asterisk hellop (~LeeHarvey@cpe-70-93-40-171.hawaii.res.rr.com) |
| 06:04.27 | Veesmooth | but |
| 06:04.29 | Veesmooth | check this part out |
| 06:04.33 | Veesmooth | when i log into my unix |
| 06:04.38 | Veesmooth | i get only the terminal screen |
| 06:04.43 | Veesmooth | why cant i see my desktop |
| 06:04.45 | Veesmooth | as normal |
| 06:05.20 | Qwell | because you have to setup your WM |
| 06:05.30 | Veesmooth | hmm |
| 06:05.40 | Qwell | This is covered in every vnc howto |
| 06:05.45 | Veesmooth | the wm |
| 06:05.51 | Veesmooth | is that whats it called |
| 06:05.55 | Qwell | or de |
| 06:06.01 | Qwell | and/or |
| 06:06.17 | Veesmooth | ok, let me go check it out |
| 06:06.19 | Veesmooth | and see |
| 06:08.55 | jayk_ | are there any asterisk contractors here? |
| 06:08.55 | jayk_ | :) |
| 06:08.58 | Qwell | tons |
| 06:09.21 | jayk_ | k |
| 06:11.50 | rvhi | anyone good at gdb? |
| 06:12.10 | rvhi | i have a few core dumps, trying to figure out what's wrong for hours now |
| 06:15.17 | PTG123 | jayk: what do you need? |
| 06:15.22 | PTG123 | rvhi: paste them i'll help you |
| 06:18.13 | *** join/#asterisk three55ml (~three55ml@cpe-66-25-89-85.satx.res.rr.com) |
| 06:19.40 | *** join/#asterisk JoshuaTree (~rob@100.mw.wgl.dcsi.net.au) |
| 06:19.50 | JoshuaTree | hello guys.... |
| 06:20.41 | JoshuaTree | ok can i explain my situation and get any help? |
| 06:21.19 | Inv_arp | JoshuaTree: hmm dont know can you? |
| 06:21.37 | JoshuaTree | well thats the question |
| 06:21.41 | Qwell | no |
| 06:21.46 | riksta | just bloody ask |
| 06:21.53 | Qwell | You only get one question here. You wasted yours. |
| 06:21.59 | Inv_arp | Qwell: lol |
| 06:22.03 | *** join/#asterisk [hC] (~hardcore@c-65-34-203-137.hsd1.fl.comcast.net) |
| 06:22.18 | Inv_arp | reminds me of a simpsons episode |
| 06:22.20 | Qwell | <bkw>NEXT!!</bkw> |
| 06:22.27 | riksta | haha |
| 06:22.39 | Qwell | Inv_arp: yeah... |
| 06:22.46 | [hC] | Is there a way with the Cisco 79XX phones to specify an alternate proxy for each "line" you configure, via TFTP? You can do it manually on the phone's interface, but I cant see a way to specify that line2 should use proxy2... |
| 06:25.15 | jayk_ | PTG123: too much...:) |
| 06:25.28 | JoshuaTree | ok we have an asterisk box at my workplace running slackware 10.1 its behind a firewall, when i connect to my VPN at work to use it i can connect and talk out by it but i'm using GSM and it starts getting all crackly and has a horrible echo in the background. is there anyway of fixing this? |
| 06:26.01 | PTG123 | hah thought you fell asleep |
| 06:32.53 | JoshuaTree | so i guess no one can help |
| 06:32.59 | Inv_arp | JoshuaTree: when you ping any machine thru the vpn whats the latency |
| 06:33.15 | JoshuaTree | to the asterisk box |
| 06:33.17 | JoshuaTree | hold on |
| 06:33.39 | JoshuaTree | 38ms |
| 06:36.02 | Inv_arp | hmm |
| 06:36.42 | [hC] | when you are placing a call, is the call itself going over the vpn too? |
| 06:36.52 | JoshuaTree | yep |
| 06:37.09 | [hC] | I would imagine it has something to do with the fact that the vpn is breaking the packets apart and confusing the buffering methods used in the codec |
| 06:37.22 | [hC] | codecs will try to predict how much to buffer, echo cancel, shit like this.. |
| 06:37.44 | [hC] | and when you run over vpn, if it starts breaking up your packets in to segments that it has no idea whats going on, its going to confuse the codec most likely |
| 06:37.53 | [hC] | just a hunch |
| 06:38.04 | JoshuaTree | hmmm |
| 06:39.20 | [hC] | you might try another codec that doesnt do the same type of predictions |
| 06:39.28 | [hC] | g726 might not be as affected as gsm.. |
| 06:40.04 | JoshuaTree | we were going to try g729 |
| 06:40.28 | [hC] | that would be fine too |
| 06:40.39 | [hC] | each codec has its own way of dealing with certain issues |
| 06:40.42 | [hC] | some are better for packet loss |
| 06:40.45 | [hC] | some are better for low bandwidth |
| 06:40.56 | [hC] | a vpn would make the codec think you are having packet loss most likely |
| 06:42.12 | Inv_arp | JoshuaTree: anyway to test talk to asterisk without going thru vpn? |
| 06:44.44 | *** join/#asterisk newmember (user@S010600d0b76b1f36.cg.shawcable.net) |
| 06:49.07 | doolph | mmm |
| 06:50.17 | *** join/#asterisk jonas (jonas@jonas.probe-networks.host.freenode) |
| 06:50.17 | doolph | why in my ip phone says that the call was connected if the party havent answer the call |
| 06:50.17 | Qwell | connected != answered |
| 06:50.17 | doolph | im calling another sip server |
| 06:50.37 | doolph | yes but in the mysql says answered |
| 06:50.39 | *** part/#asterisk implicit (~implicit@ip68-5-148-1.oc.oc.cox.net) |
| 06:50.50 | doolph | I cant control the calls if it says that |
| 06:52.04 | doolph | argh |
| 06:52.06 | doolph | this wont work |
| 06:52.24 | doolph | the call is not answered and it says that is billed |
| 06:52.53 | Qwell | You're saying your provider is billing you for ringing, when its not answered? |
| 06:53.18 | doolph | ok i have 2 servers |
| 06:53.23 | doolph | 1 with pstn access |
| 06:53.32 | doolph | then from 1 i am calling to that one |
| 06:53.43 | doolph | example |
| 06:53.44 | Qwell | Then it is "answered". |
| 06:53.54 | doolph | yes |
| 06:54.00 | Qwell | So whats the problem? |
| 06:54.06 | doolph | ok |
| 06:54.15 | doolph | but if I dont pickup the phone |
| 06:54.21 | doolph | it shouldnt be answered |
| 06:54.30 | doolph | but in my server 1 logs that is answered |
| 06:54.32 | Qwell | it doesn't work that way |
| 06:54.41 | *** join/#asterisk dg1nsw (~schulte@gate.sympat.de) |
| 06:54.42 | Qwell | Your second server IS answering the call. |
| 06:55.35 | doolph | umm |
| 06:55.39 | doolph | it shouldnt be that way |
| 06:55.43 | Qwell | Why not? |
| 06:55.58 | doolph | even my server 2 |
| 06:56.02 | Qwell | server 1 could give a shit less what server 2 does |
| 06:56.03 | doolph | it says that the call is answered |
| 06:56.33 | Qwell | because it IS answering the call |
| 06:58.03 | doolph | the server is answering |
| 06:58.20 | doolph | but the phone is not picked up |
| 06:58.20 | Qwell | If somebody can explain this in different terms, please feel free... |
| 06:58.21 | MikeJ[Laptop] | doolph, not necessisarily |
| 06:58.41 | doolph | i dont want the server answer the call when it is not answered |
| 06:58.58 | PTG123 | you should get billed if its not answered |
| 06:59.05 | MikeJ[Laptop] | if I call an IAX address on server 2, that goes to a dial out a zap chan, it will not report back answered to server one until it acutally answers on the zap chan |
| 06:59.19 | PTG123 | because they are getting billed |
| 06:59.28 | doolph | yes but i am getting billed |
| 06:59.33 | PTG123 | as you should |
| 06:59.48 | PTG123 | your getting billed for like 6 seconds and complaining? :) |
| 06:59.48 | doolph | then it's wrong bill |
| 07:00.00 | doolph | yes |
| 07:00.07 | doolph | because what if the people dont answer |
| 07:00.11 | doolph | it bills it too |
| 07:00.20 | PTG123 | then you pay you .002 and suck it up |
| 07:00.22 | doolph | even the person dont answer |
| 07:00.33 | PTG123 | if your that tight for money, you better get a better job |
| 07:00.47 | doolph | no |
| 07:00.55 | doolph | im not getting billed for real |
| 07:01.07 | PTG123 | huh |
| 07:01.09 | MikeJ[Laptop] | wow, your giving this guy a hard time.. and he's proabably right |
| 07:01.18 | doolph | look |
| 07:01.21 | doolph | I have 2 server |
| 07:01.26 | doolph | 1 with pstn access |
| 07:01.31 | doolph | 1 sip server |
| 07:01.35 | doolph | ok |
| 07:01.39 | MikeJ[Laptop] | ok |
| 07:01.50 | doolph | now with sip server i am connecting to server 1 through iax2 |
| 07:01.55 | doolph | to have pstn access |
| 07:01.59 | MikeJ[Laptop] | ok |
| 07:02.06 | doolph | and when i try to make a call |
| 07:02.24 | doolph | it is logging and billed within my mysql logs |
| 07:02.33 | doolph | even the call ISNOT answered |
| 07:02.35 | MikeJ[Laptop] | billed to who? |
| 07:02.43 | MikeJ[Laptop] | it's your server |
| 07:02.56 | doolph | yes |
| 07:02.56 | PTG123 | thats because the minute yous tart dialing |
| 07:02.56 | PTG123 | it starts billing |
| 07:02.56 | doolph | yes |
| 07:03.00 | PTG123 | as it should |
| 07:03.11 | doolph | ok |
| 07:03.12 | doolph | now |
| 07:03.13 | MikeJ[Laptop] | no it shouldn't |
| 07:03.17 | doolph | yeah |
| 07:03.18 | doolph | it shouldnt |
| 07:03.27 | doolph | it should starts to bill when answer |
| 07:03.31 | doolph | right |
| 07:03.32 | MikeJ[Laptop] | so this is an issue with how the call time is recorded in CDR |
| 07:03.37 | PTG123 | no b/c you don't get anything different sent when you answer |
| 07:03.45 | PTG123 | mikej: no not if he is going via iax |
| 07:03.47 | PTG123 | as soon as he hits dial |
| 07:03.49 | PTG123 | it starts the timer |
| 07:03.53 | *** join/#asterisk sandnigg0r (~niggerplz@66-55-197-254.gwi.net) |
| 07:03.56 | MikeJ[Laptop] | hmmmmm |
| 07:03.57 | PTG123 | b/c the second box has no idea what the first box is doing |
| 07:04.00 | MikeJ[Laptop] | IAX |
| 07:04.02 | PTG123 | rather its ringing or answered |
| 07:04.05 | Qwell | Thats how providers do it, and honestly...why shouldn't they? |
| 07:04.06 | doolph | PTG123 then what via |
| 07:04.09 | PTG123 | iut has no way to know one way or another |
| 07:04.16 | Qwell | They're being charged for the ringing. I should eat that cost, as a customer. |
| 07:04.22 | PTG123 | doolph, its right, there is no way around it |
| 07:04.29 | Qwell | It's MY call. It's MY fault if the call isn't answered. |
| 07:04.31 | doolph | my customer will get complain |
| 07:04.46 | PTG123 | doolph, then write your own standards and your own system, and your own pbx :) |
| 07:04.56 | PTG123 | or you can tell you rcustomers tough shit : ) |
| 07:05.11 | doolph | no man |
| 07:05.17 | doolph | it should have a better way to fix this |
| 07:05.24 | MikeJ[Laptop] | you can use sip between the boxes |
| 07:05.26 | PTG123 | doolph, you don't understand how it works |
| 07:05.28 | MikeJ[Laptop] | hmmmm |
| 07:05.35 | PTG123 | MikeJ[Laptop], won't make a difference |
| 07:05.36 | doolph | i tried sip |
| 07:05.38 | doolph | and the same shit |
| 07:05.40 | Qwell | PTG123: waste of time. :) |
| 07:05.42 | PTG123 | the ringing you hear, is just audio |
| 07:05.48 | PTG123 | its not like a pstn where its actually not connected |
| 07:06.01 | MikeJ[Laptop] | but it is before the answer, like early audio |
| 07:06.18 | MikeJ[Laptop] | now, on IAX.... |
| 07:06.20 | Qwell | as soon as the packet hits the other server, the call is "answered" |
| 07:06.32 | MikeJ[Laptop] | it's not |
| 07:06.41 | MikeJ[Laptop] | you need a reply |
| 07:06.47 | Qwell | You know what I mean. :) |
| 07:06.49 | doolph | let me try with sip |
| 07:06.49 | MikeJ[Laptop] | and negotiation |
| 07:06.49 | PTG123 | exactly what qwell said |
| 07:06.53 | PTG123 | ok listen :) |
| 07:07.00 | PTG123 | i am someone who actually writest the code |
| 07:07.04 | PTG123 | and understands the protocols |
| 07:07.11 | PTG123 | you can try anything till you are blue in the face |
| 07:07.13 | PTG123 | it ain't gonna change it |
| 07:07.20 | doolph | arghhhhhhhhh |
| 07:07.21 | PTG123 | once the rtp stream is connected, its connected |
| 07:07.24 | PTG123 | PERIOD |
| 07:07.28 | doolph | for real? |
| 07:07.35 | PTG123 | the only thing your box knows, is i got this audio stream connected |
| 07:07.44 | Qwell | doolph: as he said, feel free to write your own protocol |
| 07:07.45 | PTG123 | your using voip now buddy :) |
| 07:07.48 | doolph | how the voip providor does then |
| 07:07.54 | Qwell | EXACTLY the same way |
| 07:07.54 | PTG123 | they bill you for it |
| 07:08.01 | Qwell | I get charged for ringing. And, why shouldn't I? |
| 07:08.03 | PTG123 | rather they answer or not |
| 07:08.04 | Qwell | It's my call |
| 07:08.10 | PTG123 | tell your customers this |
| 07:08.12 | *** join/#asterisk RoyK (~roy@179.80-203-29.nextgentel.com) |
| 07:08.14 | PTG123 | if they want to pay 7c a minute |
| 07:08.15 | PTG123 | they can complain |
| 07:08.23 | PTG123 | and for every call under 20 seconds |
| 07:08.25 | Qwell | I'm perfectly fine with paying 0.02c for ringing |
| 07:08.26 | PTG123 | you will give them free :) |
| 07:08.33 | PTG123 | or you can payu the cheaper rates voip gives you |
| 07:08.35 | PTG123 | and shut the fuck up :) |
| 07:08.41 | doolph | yes |
| 07:08.47 | doolph | that's what i was thinking |
| 07:08.56 | doolph | make a php script that skip all calls under 20 secs |
| 07:09.06 | doolph | but sometimes I will lose |
| 07:09.15 | Qwell | large scale abuse... |
| 07:09.27 | MikeJ[Laptop] | or bill off the cdr on your pstn gateway |
| 07:09.32 | PTG123 | i have a nice dialer app i can use to abuse it |
| 07:09.34 | MikeJ[Laptop] | then no prob |
| 07:09.37 | PTG123 | so let me know when you set up your system :) |
| 07:09.43 | Qwell | PTG123: write something for me, to make a new call every 20 seconds, and bridge old calls? |
| 07:09.52 | doolph | abuse it? |
| 07:09.53 | PTG123 | nah i will just play a 15 second message |
| 07:09.53 | PTG123 | haha |
| 07:10.01 | Qwell | that works, haha |
| 07:10.04 | PTG123 | call me at 1800-BIG-TITS |
| 07:10.20 | PTG123 | Hey big boy, want to talk to real women.. 1-800-BIG-TITS (HANGUP) |
| 07:10.25 | MikeJ[Laptop] | if he uses the cdr off the pstn gateway, he does know when the answer is, and can acurately bill |
| 07:10.35 | PTG123 | MikeJ[Laptop], poor hack |
| 07:10.41 | PTG123 | what if he has 2 gateways |
| 07:10.42 | MikeJ[Laptop] | poor hack how |
| 07:10.53 | PTG123 | plus his ld probably is not on the pstn |
| 07:10.58 | PTG123 | so that would be billed that way |
| 07:11.04 | PTG123 | embrance voip |
| 07:11.07 | doolph | my server is 100000km away from me |
| 07:11.07 | PTG123 | er embrace |
| 07:11.13 | MikeJ[Laptop] | so? |
| 07:11.22 | doolph | so i cant attach a pstn there |
| 07:11.24 | MikeJ[Laptop] | he has 2 servers |
| 07:11.27 | Qwell | your server is on the moon? |
| 07:11.32 | doolph | lol |
| 07:11.36 | Qwell | ..thats pretty fucking sweet |
| 07:11.38 | PTG123 | you know in the time i have wasted with this conversation, i could have made more money then all the extra money he was gonna lose by all his clients because of this |
| 07:11.43 | Qwell | no wonder you're worried about .2c |
| 07:11.47 | MikeJ[Laptop] | your going to have bad satalite lag on that moon connection btw |
| 07:11.52 | PTG123 | damn did they colonize the moon when i wasnt' watching? |
| 07:12.04 | doolph | PTG123 mobile calls are expensive |
| 07:12.07 | MikeJ[Laptop] | maybe you should charge for ringing with those satalite bandwidth cost :) |
| 07:12.08 | PTG123 | MikeJ[Laptop], it opens up a warp bubble first |
| 07:12.18 | doolph | and a not accurasi system is not a good idea |
| 07:12.36 | MikeJ[Laptop] | ok.... |
| 07:12.36 | PTG123 | buddy if youw ant to write your own protocol go for it :) |
| 07:12.48 | MikeJ[Laptop] | PTG123, be nicer |
| 07:12.56 | doolph | ummm |
| 07:12.56 | doolph | ok |
| 07:13.01 | doolph | what else should i install to my server |
| 07:15.24 | Qwell | off to bed |
| 07:15.27 | PTG123 | man anyone else hungry |
| 07:15.33 | Qwell | yeah... |
| 07:15.34 | PTG123 | night qwell :) have fun with the rat race tommorow |
| 07:15.51 | Qwell | PTG123: I get to finish unbreaking a windows server tomorrow...fun stuff. heh |
| 07:15.59 | Qwell | god I hosed that thing... |
| 07:16.18 | Qwell | moving a windows HD to a new machine...NOT a good idea |
| 07:16.57 | denon | it works just fine .. |
| 07:17.03 | denon | you just have to know what you're doing. :) |
| 07:17.16 | denon | most likely, just blow away the enum tree and let it detect the new hardware |
| 07:17.28 | *** join/#asterisk outsidefactor (~blah@203-217-79-71.dyn.iinet.net.au) |
| 07:17.29 | Inv_arp | Qwell: new machine is different chipset? |
| 07:17.40 | *** join/#asterisk cjk (~cjk@80.92.64.103) |
| 07:17.54 | denon | if you dont do that .. things may get ugly. but dont blame windows, you were the one who didn't do it. :) |
| 07:17.58 | PTG123 | hah |
| 07:18.05 | PTG123 | we need to get you outo fthat rat rac at some point :) |
| 07:18.27 | PTG123 | denon: how is the xbox? |
| 07:18.40 | denon | fun stuff |
| 07:18.49 | denon | though I've gotta replace that one controller, and that one adapter |
| 07:19.00 | denon | the right joystick thingy on that one controller was totally trashed .. |
| 07:19.04 | denon | I think you must have dropped a couch on it |
| 07:19.14 | Qwell | Inv_arp: same chipset, actually |
| 07:19.21 | PTG123 | didn't work |
| 07:19.23 | PTG123 | or stuck or what? |
| 07:19.24 | doolph | ok |
| 07:19.25 | Qwell | denon: I tried doing an in-place reinstall, and it hosed it even more |
| 07:19.25 | doolph | im off |
| 07:19.29 | PTG123 | sometimes they g et stuck, and you just push em in |
| 07:19.30 | denon | nah, its broke off inside .. |
| 07:19.37 | PTG123 | weird really |
| 07:19.46 | denon | there's plastic pieces rollin around in there .. |
| 07:19.50 | denon | you can sorta make it work |
| 07:19.56 | PTG123 | i wonder if it happened in transit |
| 07:19.58 | denon | but not really |
| 07:20.01 | denon | eh .. hehe .. I doubt it .. |
| 07:20.10 | PTG123 | i probably played that thing maybe 5 times |
| 07:20.10 | denon | not like the thing is smashed on the outside |
| 07:20.37 | Qwell | anyhow, yeah, bed |
| 07:20.44 | PTG123 | well atleast it works, you upgrade the hard drive? |
| 07:20.52 | denon | yeah |
| 07:21.05 | denon | 350 gig or somethin |
| 07:21.05 | PTG123 | i have one with a broken cdrom i may chip |
| 07:21.09 | PTG123 | and put in an arcade machine |
| 07:21.14 | *** join/#asterisk Zeeek (~icechat5@Zeeek.sustaining.supporter.pdpc) |
| 07:21.15 | PTG123 | with a large hard drive |
| 07:21.42 | PTG123 | i have no time for games :( |
| 07:21.51 | denon | yeah, me neither .. so im finding out |
| 07:21.54 | *** join/#asterisk implicit (~implicit@ip68-5-148-1.oc.oc.cox.net) |
| 07:22.19 | denon | played it 5 times so far .. one I got pissed at the controller, and shut it off .. one messin with that stupid broken cable |
| 07:22.20 | denon | hehe |
| 07:22.32 | PTG123 | hah you have 3 people that play it? :) |
| 07:22.33 | denon | gamers are supposed to be de-stressers heheh |
| 07:22.40 | PTG123 | if you had 2 other controllers, why did the other one piss you off so much? :) |
| 07:22.44 | PTG123 | what color was the broken one? |
| 07:22.57 | denon | yeah .. buddy came over, but we ended up just swapping 2 at a time |
| 07:23.21 | Zeeek | ptg123.com is still available :) |
| 07:23.27 | denon | green powerpad |
| 07:23.30 | PTG123 | weird |
| 07:23.47 | denon | the right joystick thing is basically just loose in there .. |
| 07:23.50 | PTG123 | Zeeek, hah :) i already got my realname.com :) |
| 07:23.55 | denon | if you turn the controller on it's side, it'll roll to the side |
| 07:24.14 | PTG123 | seems kind of odd |
| 07:24.19 | PTG123 | last time i used it we used all 3 controllers |
| 07:24.23 | PTG123 | to play football |
| 07:24.28 | Zeeek | were either of you here yesterday for "nat=yes in [global]" discussion ? |
| 07:24.41 | denon | ooooh, that explains it .. you're not supposed to play with the controllers, you're sposed to play on the TV :) |
| 07:25.11 | PTG123 | hah |
| 07:25.15 | PTG123 | Zeeek, no idea |
| 07:25.49 | Zeeek | it began "you must NEVER put nat=yes in global" |
| 07:26.05 | denon | looks like they're about 25 bucks shipped . . bleh, wonder how much locally |
| 07:26.10 | PTG123 | i don't see why not |
| 07:26.17 | PTG123 | why ever trust the ip in the header |
| 07:26.58 | rabelais | does anyone know if broadvoice allows number transfers out of their network? |
| 07:27.02 | Zeeek | PTG well, someone looked at the source and said the value was ignored but my point is... |
| 07:27.21 | PTG123 | rabelais, you mean porting? |
| 07:27.27 | rabelais | PTG123: yes |
| 07:27.35 | PTG123 | rabelais, i don't see how they would have a choice |
| 07:27.45 | rabelais | PTG123: ? |
| 07:28.28 | Zeeek | I've always had nat=yes in general section. So I commented it out 'for fun", restarted and it didn't change anything |
| 07:28.31 | PTG123 | they don't have to say yes to port a number |
| 07:28.41 | PTG123 | Zeeek, sip in asterisk is broken anyhow |
| 07:28.57 | rabelais | PTG123: they don't? who then has the authority over the line? |
| 07:29.10 | PTG123 | if you prove you own the number, with a bill |
| 07:29.11 | PTG123 | they will port it |
| 07:29.28 | Zeeek | the reason I bring this up though is that this morning the ip of asterisk box changed, and my phones were not getting audio. Putting nat=yes back fixed it. |
| 07:29.31 | rabelais | PTG123: they, as in my new service provider? |
| 07:29.41 | Zeeek | Now, tyhink of how many people ask about the no audio problem? |
| 07:29.58 | PTG123 | Zeeek, i have looked at the source, and nat=yes in general does set the default to nat = yes :) |
| 07:30.04 | PTG123 | Zeeek, so the person yesterday was wrong |
| 07:30.06 | PTG123 | rabelais, yes |
| 07:30.07 | MikeJ[Laptop] | who's andreas sikkema? |
| 07:30.23 | rabelais | how cool |
| 07:30.26 | rabelais | PTG123: thanks |
| 07:30.30 | my007ms | can i used meetme in sip "i don't have any card" |
| 07:30.33 | Zeeek | he said it sets the default but that the default is later ignored at peer creation time |
| 07:30.47 | PTG123 | its atually not |
| 07:30.50 | PTG123 | er actually not |
| 07:30.53 | PTG123 | it copies over all defaults |
| 07:31.01 | PTG123 | so he was a moron :) |
| 07:31.02 | PTG123 | or blind |
| 07:31.06 | Zeeek | now, now |
| 07:31.36 | my007ms | can i used meetme in sip "i don't have any card" ? |
| 07:32.11 | denon | heh .. wireless controllers look kinda slick |
| 07:32.23 | PTG123 | denon, my son has one for his gamecube, works well |
| 07:32.24 | denon | sucks that everything takes batteries though |
| 07:32.29 | denon | no rechargable stuff |
| 07:32.30 | PTG123 | they last forever |
| 07:32.35 | denon | 150h |
| 07:32.38 | denon | assuming you play it all at once |
| 07:32.40 | denon | no on/off |
| 07:32.41 | denon | hehe |
| 07:33.47 | PTG123 | heh |
| 07:33.54 | PTG123 | i haven't changed his batteries yet i don't think |
| 07:34.26 | denon | 2.4ghz .. should work well at my place .. Ive got all my wireless stuff off 2.4 |
| 07:34.43 | denon | the only thing that's still 2.4 is the microwave .. :) phones, wlan, etc is far far away |
| 07:35.21 | denon | man, it's a sad day when Best Buy seems to be the cheapest place to get this stuff |
| 07:36.47 | PTG123 | hah |
| 07:36.50 | PTG123 | frys is cheapest for me |
| 07:37.04 | PTG123 | or ecost.com is where i order a ton of stuff |
| 07:37.10 | denon | hmm .. target has this "hip" one for 19.99 |
| 07:40.16 | *** join/#asterisk clive- (~pirch@rndf-146-44-199.telkomadsl.co.za) |
| 07:41.11 | kapejod | morn clive- |
| 07:41.44 | clive- | hi kape:) |
| 07:42.20 | clive- | building up courage to try chan_capi pre0.4.0 on my production box :) |
| 07:42.27 | clive- | is it stable? |
| 07:42.51 | kapejod | it's called PRE1, not? ;) |
| 07:42.59 | kapejod | does that say STABLE-100? ;) |
| 07:44.49 | *** join/#asterisk blint (~blint@adsl-669.mirage.euroweb.hu) |
| 07:44.55 | blint | hi everyone |
| 07:47.07 | blint | has anyone used Asterisk in a pulse-dial environment? |
| 07:48.34 | clive- | Kape...when do you expect stable to be running? |
| 07:48.37 | blint | we are about to build a PBX for our Russian office and I just learnt that there is no tone dialing in Moscow |
| 07:49.11 | *** join/#asterisk cmk (~cmk_@p54A3D31F.dip.t-dialin.net) |
| 07:54.03 | kapejod | clive-: you will be the first person to know ;) |
| 07:55.19 | *** join/#asterisk newmember (~newmember@S010600a0c93dce87.cg.shawcable.net) |
| 08:00.24 | *** join/#asterisk jefrey (~tmnut@203.115.193.176) |
| 08:00.43 | clive- | lol...if I ask everyday, :) |
| 08:01.10 | clive- | how stable is the pre version? |
| 08:03.18 | kapejod | ok with * 1.0.7 |
| 08:03.26 | kapejod | but not with cvs head |
| 08:09.23 | flotox | <PROTECTED> |
| 08:10.13 | *** join/#asterisk jeffgus (~jeffgus@alpha.zimage.com) |
| 08:14.32 | *** join/#asterisk tessier (~treed@203.210.218.171) |
| 08:14.51 | tessier | The whole idea behind pre-pay is that my account stops working when the balance reaches zero, right? |
| 08:14.59 | tessier | Right. |
| 08:15.08 | tessier | Therefore nufone is clueless for letting my account go negative. |
| 08:15.24 | *** join/#asterisk _mwoodj_ (~mwoodj@hyper-eye.digium.sponsor.pdpc) |
| 08:15.30 | tessier | Apparently they have lost way more money than they ever made off of me. |
| 08:15.42 | my007ms | hi everyone |
| 08:16.30 | tessier | Now I am left to wonder whether I should pay or not. I suspect not since it was nufone that didn't hold up its end of the deal. |
| 08:17.36 | h3x | heh |
| 08:18.43 | my007ms | can some one help pleas i firet time read in API but i don't understand it can some one tell me how can i send Action: login |
| 08:19.00 | tessier | I think I have paid them a total of $40. My account is now $72.00 negative! |
| 08:19.24 | clive- | sounds like a deal there tessier |
| 08:19.35 | tessier | Their automated system has been emailing me about it for ages. I got an email from a human about it and explained the above. |
| 08:19.41 | tessier | clive-: No kidding. I highly recommend nufone! |
| 08:19.44 | tessier | Their accounting rocks! |
| 08:20.00 | tessier | I wonder if I slip 'em another $20 if I will be able to make calls again. |
| 08:20.07 | my007ms | where i can i send this |
| 08:20.39 | my007ms | from asterisk command line |
| 08:21.42 | tessier | ssh is gonna drive me nuts. Still asks for my password even though I have the key installed and I am using ssh-agent |
| 08:22.01 | my007ms | can some one answer pleas |
| 08:22.01 | tessier | Some hosts let me log in without pass others don't. I've checked the perms and everything. Ugh |
| 08:22.30 | denon | tessier: just because they give you a little leway, and dont instantly cut off your phone service, doesnt mean you should abuse them |
| 08:22.58 | denon | nufone goes out of their way to be accomodating - dont screw them because of it. |
| 08:23.09 | tessier | denon: I would never intentionally abuse them and was only joking about trying to get more out of them. |
| 08:23.14 | tessier | I have taken their config out of my asterisk box. |
| 08:23.31 | tessier | But they really should instantly cut off service. That is how everyone understands prepay to work. |
| 08:23.36 | my007ms | How to login API from where i can send Action: login ? |
| 08:23.41 | tessier | That is how mobile phones have always worked. |
| 08:23.55 | denon | most businesses dont want to be instantly cut off |
| 08:24.04 | tessier | Businesses should NOT be on pre-pay. |
| 08:24.19 | denon | there's two definitions of pre-pay here .. |
| 08:24.19 | tessier | It is way too easy to spend more than you intended on phone services. This is why people use prepay. |
| 08:24.27 | denon | the one you're referring to is for consumers that telcos cant trust .. |
| 08:24.39 | tessier | If I only budget $20/mo for long distance then I pre-pay the $20 and rest assured I cannot go over budget. |
| 08:24.52 | denon | a business prepay service is a way of getting around red tape, and being able to offer lower rates without billing and colletion hassles |
| 08:24.54 | Zeeek | there should probably be an email reminder or something, but nufone doesn't do things like that. Most offers include an auto top up option of $nn |
| 08:25.03 | denon | Zeeek: nufone does email reminders |
| 08:25.08 | Zeeek | they do? |
| 08:25.11 | denon | they remind you when you get low, and remind you again when you're below |
| 08:25.13 | denon | yes, they do |
| 08:25.19 | Zeeek | well then there's no issue at all |
| 08:25.20 | denon | every night |
| 08:25.27 | denon | I know - that's what I'm saying. |
| 08:25.27 | tessier | Yes, they do. I get them. |
| 08:25.34 | denon | and yet he bitches :) |
| 08:25.38 | tessier | I expect to get emails when I get low. I don't expect to be allowed to go negative. |
| 08:25.59 | denon | well .. read my statement above |
| 08:26.00 | Zeeek | does the reminder include the balance? |
| 08:26.02 | denon | yes |
| 08:26.20 | denon | uh huh |
| 08:26.21 | Zeeek | NEXT! |
| 08:26.23 | tessier | Guilty of what? |
| 08:26.36 | Zeeek | frivolous complaints |
| 08:26.38 | tessier | The balance should hit zero and stay there! |
| 08:26.56 | denon | tessier .. im telling you .. |
| 08:27.03 | tessier | It's really nufone that should be complaining but... |
| 08:27.04 | denon | the prepay is a way to get out of billing hassles |
| 08:27.08 | denon | its not a cut-off |
| 08:27.18 | denon | how come qwest will let you keep your line for 6 months .. |
| 08:27.22 | denon | even though you never pay your bill |
| 08:27.30 | tessier | denon: Because it's in the contract? |
| 08:27.39 | tessier | nufone has zero reason to trust me |
| 08:27.39 | denon | this isnt a disposible cell phone .. |
| 08:27.43 | denon | its a telco service |
| 08:27.49 | tessier | I paypal them $20 and tell them a username and pass I want. |
| 08:27.53 | tessier | Completely anonymously. |
| 08:28.05 | tessier | Why would they extend me credit? |
| 08:28.08 | denon | I think people should appreciate the fact that they dont cut them off the second they forget to refill their account |
| 08:28.15 | tessier | Without any contract or agreement or anything? |
| 08:28.26 | tessier | I do appreciate $30 worth of free phone service! :) |
| 08:28.34 | denon | *rolls eyes* |
| 08:28.37 | tessier | But that's not the point. |
| 08:28.38 | denon | this is a pointless conversation |
| 08:28.43 | tessier | Apparently. |
| 08:28.45 | denon | im just saying, there's more reasons for prepay |
| 08:28.55 | denon | not just consumers with no foresight |
| 08:29.06 | Zeeek | I love prepay. I have no other kind of account |
| 08:29.28 | denon | anyway, im going to bed |
| 08:29.35 | denon | you can stop typing whatever rant you were working on |
| 08:29.38 | denon | :) |
| 08:31.03 | my007ms | can some one help pleas |
| 08:31.32 | my007ms | How to login API from where i can send Action: login ? |
| 08:31.57 | my007ms | i try in asterisk command line |
| 08:32.50 | my007ms | :) sure this is not right coz API allows a programmer to connect to the Asterisk engine |
| 08:33.09 | my007ms | and this is for sure not from asterisk |
| 08:33.24 | Zeeek | <PROTECTED> |
| 08:33.43 | Zeeek | http://www.voip-info.org/wiki-Asterisk+manager+experience |
| 08:34.09 | Zeeek | http://www.voip-info.org/wiki-Asterisk+Manager+Proxy |
| 08:34.43 | my007ms | zeeek :) this is page i read in now but i need just tell me start where i can send Action: login |
| 08:36.46 | my007ms | where i typ this command |
| 08:38.16 | Zeeek | if I understand your question, which is far from certain, you do not type the command. YOu need to establish a dialog with the manager on port ... |
| 08:39.19 | my007ms | look Zeeek now i run asterisk |
| 08:39.32 | Juggie | port 5038 |
| 08:39.36 | Zeeek | that makes it a lot clearer :) |
| 08:40.16 | Zeeek | my007ms what is the manager API in your understanding ? |
| 08:40.16 | my007ms | :) my q is there to typ this in my shell or in asterisk cmd |
| 08:40.27 | kapejod | telnet asteriskserver 5038 |
| 08:40.31 | kapejod | off you go |
| 08:40.44 | my007ms | some thing to make more thing with astrerisk |
| 08:40.46 | Zeeek | ah, that's where he wanted to go... |
| 08:41.43 | my007ms | :) i chat with u now from asterisk |
| 08:41.48 | my007ms | server |
| 08:42.10 | my007ms | i feel shame from that |
| 08:42.26 | Zeeek | what are you wearing right now? |
| 08:42.34 | my007ms | :D |
| 08:44.18 | tessier | Didn't expect to spend an extra $70 with them but oh well. |
| 08:44.24 | my007ms | look now i have open terminal and runing asterisk and i am in asteriks localhost*CLI> |
| 08:44.30 | tessier | If I don't JerJer will tell people I don't pay my bills. Not worth that hassle. |
| 08:48.06 | Zeeek | my007ms the manager is listening on a port for mannds... your wish is his command one could say |
| 08:50.14 | my007ms | in my manger.conf there is port = 5038 |
| 08:51.01 | Zeeek | another answer to your question is that from the CLI, typing ? should show you all the possibilities of the CLI |
| 08:51.54 | my007ms | in what CLI shell or asterisk CLI |
| 08:52.02 | tessier | hrm |
| 08:52.22 | tessier | paypal says "We are currently performing regular maintenance of our security measures. Your account has been randomly selected for this maintenance, and you will now be taken through a series of identity verification pages." |
| 08:52.31 | tessier | And it asks for credit card and bank account numbers. |
| 08:52.33 | Zeeek | that sounds ominous |
| 08:52.37 | tessier | Highly suspicious. |
| 08:52.43 | Zeeek | phishy even |
| 08:52.44 | tessier | But I keyed in the paypal.com domain name myself! |
| 08:52.47 | tessier | Didn't follow a link. |
| 08:53.03 | Zeeek | if yoiu use firefox there is an extension that checks the server all the time |
| 08:53.06 | tessier | And I entered my username and pass and it takes me to this page. I know what fishing looks like. The url looks perfectly real. |
| 08:53.14 | Zeeek | scary! |
| 08:53.15 | tessier | And this is a trusted certificate. |
| 08:53.22 | Zeeek | unlike nufones? |
| 08:53.23 | tessier | SSL encrypted page and everything. |
| 08:53.26 | tessier | Exactly. |
| 08:53.29 | my007ms | this is my manger.conf [general] |
| 08:53.29 | my007ms | enabled = yes |
| 08:53.29 | my007ms | port = 5038 |
| 08:53.29 | my007ms | bindaddr = 0.0.0.0 |
| 08:53.29 | my007ms | [admin] |
| 08:53.30 | tessier | Why don't they have a real cert? |
| 08:53.30 | my007ms | secret = password |
| 08:53.32 | my007ms | deny=0.0.0.0/0.0.0.0 |
| 08:53.34 | my007ms | permit=127.0.0.1/255.255.255.0 |
| 08:53.38 | my007ms | read = system,call,log,verbose,command,agent,user |
| 08:53.40 | my007ms | write = system,call,log,verbose,command,agent,user |
| 08:53.49 | Zeeek | my007ms please |
| 08:54.04 | tessier | It is a real legit verisign key. |
| 08:54.07 | tessier | So they say it's paypal. |
| 08:54.10 | Zeeek | verislime |
| 08:54.45 | Zeeek | I'd write paypâl |
| 08:54.47 | tessier | Nothing you can do with that alone except deposit money |
| 08:55.01 | tessier | ah...I failed the first login and I am logging in from an IP address in Vietnam |
| 08:55.28 | tessier | So they put me through a test. |
| 08:55.53 | tessier | Then it asks me to confirm that it really was me who tried to access the account from this ip |
| 08:55.54 | *** join/#asterisk akaka (~wade@194.193.169.242) |
| 08:55.58 | tessier | I click yes... |
| 08:56.13 | tessier | Wow, I have positive $12.88 in my paypal account. |
| 08:56.21 | tessier | Ah, I sold lilo some FXO cards a while back. |
| 08:56.50 | *** join/#asterisk JerJer[mobile] (~jj@ip68-103-26-140.ks.ok.cox.net) |
| 08:59.01 | *** part/#asterisk akaka (~wade@194.193.169.242) |
| 08:59.57 | *** join/#asterisk wadea (~wade@194.193.169.242) |
| 09:01.35 | my007ms | can someone tell me what API is |
| 09:02.02 | Zeeek | Application Programming Interface ? |
| 09:02.22 | my007ms | to do what ? |
| 09:02.46 | tessier | Always Purge Intestines |
| 09:02.51 | my007ms | allows a programmer to connect to the Asterisk engine and issue commands or read PBX events over a TCP/IP stream. |
| 09:02.59 | tessier | Almost Pickled Instantly |
| 09:03.09 | tessier | Angry Pets Intimidate |
| 09:04.22 | Zeeek | that's fairly clear my007ms |
| 09:04.39 | my007ms | next Q then can u give me example command line |
| 09:05.23 | Zeeek | my next question is, could you go read and search a little bit to get the basic concepts? |
| 09:06.03 | my007ms | ok i will |
| 09:06.17 | Zeeek | thanks |
| 09:06.46 | my007ms | :) thanks Zeeek |
| 09:06.52 | Zeeek | my pleasure! |
| 09:08.33 | *** join/#asterisk Fabe (~spamhere@pD95B0BF7.dip0.t-ipconnect.de) |
| 09:10.29 | *** join/#asterisk tengulre (~tengulre@61.185.238.166) |
| 09:11.20 | JoshuaTree | does anyone know the best codec for going through a VPN, one that deals with packet segmentation better than GSM |
| 09:11.47 | blint | does someone have experience with Aterisk and pulse dialing? I know that it is possible to do dial out in pulse mode on a Zap channel, but can a voice menu be navigated in pulse mode? |
| 09:12.08 | blint | russia doesn't have tone :( |
| 09:12.55 | JoshuaTree | is anyone using g.729? |
| 09:12.59 | JerJer[mobile] | did i join #newbie ? |
| 09:14.15 | Zeeek | heh |
| 09:14.20 | JoshuaTree | right |
| 09:14.31 | Zeeek | JoshuaTree yea |
| 09:14.46 | JoshuaTree | what? |
| 09:14.50 | Zeeek | we have a measly 4 channels of 729 |
| 09:15.14 | enots | blint, u cant |
| 09:15.51 | JoshuaTree | running from an external source? |
| 09:16.04 | Zeeek | you could buy a handheld DMTF tone generator for users |
| 09:16.30 | Zeeek | like the old answering machine boxes |
| 09:16.37 | blint | i am sad to hear |
| 09:16.44 | kapejod | a silver box |
| 09:16.48 | Zeeek | yeah that sucks in this day and age |
| 09:17.01 | Zeeek | but it does add a layer of security :) |
| 09:17.12 | enots | blint, change the country :) |
| 09:17.53 | blint | Zeeek: my concern is actually someone calling us and not able to go to an extension |
| 09:18.14 | Zeeek | there is usually an option to wait for an operator |
| 09:18.20 | Zeeek | in IVR I mean |
| 09:18.26 | Zeeek | "fallthrough" |
| 09:18.31 | enots | yeah |
| 09:18.35 | kapejod | blint: you could spent some time on sphinx (speech recognition) |
| 09:18.46 | kapejod | "say 1 for disconnect" |
| 09:18.52 | Zeeek | or sphincter - tightening the muscle |
| 09:18.59 | blint | wow |
| 09:19.05 | blint | that would be cool |
| 09:19.17 | blint | checking it |
| 09:19.22 | JoshuaTree | we have an asterisk box at my workplace running slackware 10.1 its behind a firewall, when i connect to my VPN at work to use it i can connect and talk out by it but i'm using GSM and it starts getting all crackly and has a horrible echo in the background. is there anyway of fixing this? i'm told there is ways of using jitter correction and stuff but its funny cause we have two Asterisk boxes and one seemed to work fine for a little while and this |
| 09:20.08 | blint | thx guys for the info! |
| 09:20.17 | kapejod | JoshuaTree: a jitterbuffer cannot fix your echo problem. |
| 09:21.39 | tengulre | do u have dialogic or other drivers for asterisk? |
| 09:21.41 | JoshuaTree | yeah but our original asterisk box worked fine until we tried putting on the g729 codec then GSM really started playing up like i could call i could hear them but they couldn't hear me... and so forth |
| 09:21.56 | tengulre | anyone have dialogic or other drivers for asterisk? |
| 09:22.06 | kapejod | tengulre: digium does. |
| 09:22.35 | tengulre | kapejod , but it no free? |
| 09:22.42 | kapejod | i guess not. |
| 09:22.53 | *** join/#asterisk blop (blop@2001:6f8:204:33:bbbb:bbbb:bbbb:bbbb) |
| 09:23.02 | tengulre | but why? |
| 09:23.17 | tengulre | is it open source? |
| 09:23.23 | kapejod | prolly not gpl |
| 09:23.33 | tengulre | oh,god |
| 09:23.47 | kapejod | why mess with dialogic? |
| 09:24.26 | tengulre | I will dead! |
| 09:24.37 | JoshuaTree | and i alway kept getting too lagged messages |
| 09:25.04 | tengulre | because I have a dialogic card! |
| 09:25.22 | kapejod | we all got to die. |
| 09:25.33 | tessier | meesa people gonna die? |
| 09:25.44 | kapejod | but there is ebay for your dialogic card.... |
| 09:26.17 | tengulre | :) |
| 09:26.17 | tengulre | what u mean? |
| 09:26.17 | tengulre | pls detail! |
| 09:26.30 | kapejod | sell it on ebay. |
| 09:28.27 | tengulre | but I ask my boss it's free! |
| 09:28.51 | kapejod | you got a source for free dialogic cards? |
| 09:28.51 | tengulre | but I tell my boss it's free! |
| 09:29.01 | kapejod | (sell them on ebay and be rich...) ;) |
| 09:29.26 | tengulre | what 's ebay? |
| 09:29.36 | kapejod | www.ebay.com |
| 09:29.50 | tengulre | tks |
| 09:29.55 | *** join/#asterisk bzbw (~wlwzhang@24-205-15-110.mpk-eres.charterpipeline.net) |
| 09:30.34 | Zeeek | could you make that multiple choice, please? |
| 09:30.44 | zoa | i pick c3 |
| 09:31.54 | *** join/#asterisk christo (~chris@office.enovi.com) |
| 09:32.50 | tengulre | zoa,what 's c3? |
| 09:32.54 | Zeeek | Forbidden Planet |
| 09:33.06 | tengulre | kapejod, thk u ,good site! |
| 09:33.41 | *** join/#asterisk syle (~blah@wnpgmb02dc1-61-40.dynamic.mts.net) |
| 09:34.10 | tengulre | I want use asterisk in my home , which card is better and cheap? |
| 09:34.47 | syle | better and cheap don't go together |
| 09:35.12 | tengulre | :) |
| 09:35.21 | Zeeek | syle were you in on the nat=yes discussion yesterday here? |
| 09:35.33 | syle | yep |
| 09:35.55 | syle | this is a good one for your home , in fact the one i use |
| 09:35.58 | syle | http://www.neobits.com/do/dtls?pid=8794 |
| 09:36.01 | Zeeek | because this morning my phone had no audio after removing that from general |
| 09:36.25 | Zeeek | and when I added nat=yes back (uncommented it) I had audio |
| 09:36.45 | Zeeek | the server ip address had changed overnight |
| 09:37.13 | Zeeek | so regardless of whatever else, that bit may be why so many have trouble with no audio or one way audio |
| 09:37.53 | syle | you only need it if your ip changes |
| 09:38.13 | syle | pay the extra 5 bucks from your dsl provider for a static ip and you won;t need it |
| 09:38.25 | Zeeek | well that's not the same as sayingt "NEVER put nat=yes in [general]" is it? |
| 09:38.54 | Zeeek | funny you should mention that $5 |
| 09:39.04 | RoyK | Zeeek: wtf? |
| 09:39.06 | Zeeek | we were promised a fixed ip a year ago |
| 09:39.11 | syle | maybe i wasn't part of discussion your talking about, my NAT discussion was about it don't work i read with realtime |
| 09:39.16 | syle | but maybe that has changed |
| 09:39.31 | Zeeek | syle no that wasn't the same one |
| 09:39.52 | Zeeek | someone was looking at the source and we both made changes and looked at sipp show peers |
| 09:40.04 | Zeeek | it didn't seem to change anything |
| 09:40.49 | syle | promised a fixed ip and they didn;t give you one? |
| 09:40.54 | Zeeek | correct |
| 09:40.55 | syle | you with cable or something? |
| 09:41.03 | Zeeek | well, half correct... |
| 09:41.22 | Zeeek | we have two accounts, one home one office |
| 09:41.22 | JerJer[mobile] | Zeeek: does nat=yes in general even do anything? (haven't eyeballed the code lately) |
| 09:41.33 | Zeeek | yes JerJer it definitely does |
| 09:41.45 | JerJer[mobile] | ahh ok cool |
| 09:42.05 | Zeeek | but whoever looked yesterday came to the conclusion that it was later ignored at perr creation time |
| 09:42.06 | syle | if i remember right nat=yes sends your ip address to the server every 30 sec to 3 min |
| 09:42.17 | nazgool | a question about extensions: is there a way to change ${EXTEN} in extensions.conf? in AGI there's a SET EXTENSION command, but when i just try to Setvar(EXTEN=${defaultext}); that doesn't seem to work |
| 09:42.27 | nazgool | any clue what i might be doing wrong? |
| 09:43.13 | nazgool | (i didn't find an equivalent to AGI "SET EXTENSION" for use in extensions.conf) |
| 09:44.06 | syle | idk i think using setvar is stupid to begin with, i use global section to do blah=blah |
| 09:45.44 | nazgool | it's just in one precise context, in one case of a gotoif that i need to set EXTEN, so i can't do that in global |
| 09:47.11 | syle | i'd help ya but i haven;t gotten to doing agi with perl yet, prob by end of week :( it is 5am though i am sure someone can answer that in a few hours though |
| 09:48.48 | syle | i really hate gotoif though hehee |
| 09:48.49 | *** join/#asterisk bjohnson (~bjohnson@66.11.188.191) |
| 09:48.55 | syle | i wish they had done if and else |
| 09:49.11 | JerJer[mobile] | propose the implementation on bugs.digium.com |
| 09:49.24 | JerJer[mobile] | i promise you will get a binary answer, eventually |
| 09:49.30 | nazgool | lol |
| 09:49.33 | syle | lol |
| 09:55.01 | *** join/#asterisk newl (~newlook@203-59-203-5.dyn.iinet.net.au) |
| 09:55.37 | eper-werk | Found a Wildcard: Digium Wildcard TE110P T1/E1 |
| 09:55.44 | JerJer[mobile] | yay |
| 09:56.37 | tengulre | ? |
| 09:56.49 | tengulre | www.ostel.com |
| 09:56.58 | tengulre | www.zrtek.com |
| 09:57.52 | JerJer[mobile] | i've worked with the ostel guys before - they are cool |
| 09:57.56 | *** join/#asterisk _omer (dfsdf@202.147.167.213) |
| 09:58.11 | JerJer[mobile] | i can't jive chiense or whatever that is, sorry |
| 09:58.23 | tengulre | yes |
| 09:59.04 | tengulre | jerjer,do u know david? |
| 09:59.07 | eper-werk | nobody happens to have the zaptel.conf settings for UK isdn30? :) |
| 10:00.16 | JerJer[mobile] | tengulre: hmmm... maybe |
| 10:00.26 | JerJer[mobile] | i think the guy i've worked with more is Mark ? |
| 10:00.40 | JerJer[mobile] | not sure anymore |
| 10:01.00 | tengulre | JerJer,do u have IVR builder? |
| 10:01.14 | JerJer[mobile] | sure |
| 10:01.25 | JerJer[mobile] | its called Asterisk |
| 10:01.26 | tengulre | youself? |
| 10:01.46 | tengulre | is it free? |
| 10:02.11 | JerJer[mobile] | asterisk is freely downloadable, yes |
| 10:02.16 | Zeeek | everything is free if you're free in your mind |
| 10:02.29 | _omer | I need to fetch reports from asterisk....For example: number of dropped, attended calls inbetween two dates....what do I have to do for it?? |
| 10:03.10 | JerJer[mobile] | mine the CDR records |
| 10:03.29 | *** join/#asterisk Clavell (~clavell@suse.satrax.hu) |
| 10:03.33 | Clavell | hi |
| 10:03.35 | *** join/#asterisk truescot (~truescot2@213.201.171.186) |
| 10:03.40 | JerJer[mobile] | hoe |
| 10:04.07 | truescot | hello ppl |
| 10:04.11 | Zeeek | it's off to lunch I go |
| 10:04.49 | _omer | how do I get reports from asterisk? |
| 10:05.03 | JerJer[mobile] | the CDRs |
| 10:05.09 | _omer | yes, |
| 10:05.32 | JerJer[mobile] | yes |
| 10:05.34 | truescot | is there anyone who can help me?, i have managed to get asterisk installed on fedora core 3, with a digium te100p and developers card, all runs ok and with the make samples it recieves calls |
| 10:06.02 | truescot | but i cqan't get my head around zapata.conf f and extensions.conf |
| 10:06.13 | truescot | is there a good explination out there somewhere? |
| 10:06.36 | Zeeek | Starter tutorial: |
| 10:06.36 | Zeeek | http://www.onlamp.com/pub/a/onlamp/2003/07/03/asterisk.html |
| 10:06.36 | Zeeek | http://www.onlamp.com/pub/a/onlamp/2004/01/22/asterisk2.html |
| 10:06.36 | Zeeek | http://www.automated.it/guidetoasterisk.htm |
| 10:06.36 | Zeeek | THE reference of the moment: |
| 10:06.36 | Zeeek | http://www.asteriskdocs.org |
| 10:06.41 | _omer | JerJer: yes CDR .... |
| 10:07.28 | truescot | yea been through all of those, guess i just have to read smarter :) |
| 10:07.37 | Zeeek | The dialplan is the heart of an Asterisk system, as it defines how Asterisk should handle calls. |
| 10:07.37 | Zeeek | http://www.asteriskdocs.org/modules/tinycontent/content/docbook/current_v1/docs-html_one/vm1.html#AEN650 |
| 10:07.53 | *** join/#asterisk RoyK (~roy@179.80-203-29.nextgentel.com) |
| 10:08.29 | Zeeek | truescot and the wiki has five pages of details on dialplans as well |
| 10:08.44 | truescot | yea, jsut being dumb i guess |
| 10:09.01 | truescot | read them all and still confused, have to re-read i guess |
| 10:09.41 | truescot | that last link u gave i havent read yet, looks good, tnx a lot |
| 10:10.09 | _omer | Zeeek: Make me to Thank you too ;) |
| 10:10.21 | *** join/#asterisk RoyK (~roy@179.80-203-29.nextgentel.com) |
| 10:11.03 | *** join/#asterisk blitzrage (~blitzrage@blitzrage.documenter.extraordinaire.of.asterisk) |
| 10:11.19 | JerJer[mobile] | asterisk wins |
| 10:11.40 | RoyK | wonderful. back to xchat :) |
| 10:11.42 | RoyK | tamtitam |
| 10:12.13 | _omer | ok Asterisk wins.....but how to get reports? |
| 10:12.26 | _omer | do I need to setup MYSQL with asterisk or what??? |
| 10:12.32 | JerJer[mobile] | CDR details |
| 10:12.42 | _omer | YES , CDR Details |
| 10:13.05 | tessier | CDR DETAILS! CDR DETAILS! WOooohoooooooo |
| 10:13.14 | _omer | I am not using Asterisk for calls termination but for my VoIP Based Call Center.. |
| 10:13.20 | JerJer[mobile] | you can generate the reports however you want |
| 10:13.29 | JerJer[mobile] | with asterisk you have the ultimate power |
| 10:14.14 | _omer | JerJer: Yes, but how??? |
| 10:14.26 | JerJer[mobile] | anyway you want |
| 10:14.34 | JerJer[mobile] | write a perl script to consume the csv files |
| 10:14.37 | Clavell | i have some problem: my asterisk box behind nat, and my ip phone is opposite side of my nat-box, |
| 10:14.51 | JerJer[mobile] | or a php script to chew on the records in a mysql db |
| 10:14.55 | JerJer[mobile] | its totally up to you |
| 10:15.02 | Clavell | when the ip phone registers to asterisk, asterisk say "registration failed" |
| 10:15.06 | JerJer[mobile] | Clavell: get asterisk a public ip |
| 10:15.23 | JerJer[mobile] | or mess with the extern* options in the config |
| 10:15.35 | _omer | JerJer: yes, but I need the details..... |
| 10:15.56 | Clavell | but if i remove the secret= directive from sip.conf, it's ok....why? |
| 10:16.56 | flotox | How I can change voicemailmain language? |
| 10:17.31 | flotox | Can i make it? |
| 10:17.53 | JerJer[mobile] | _omer: if you want detailed information you are going to have to pay someone (not me) to hold your hand |
| 10:18.23 | *** join/#asterisk sezuan (sezuan@port-212-202-202-204.dynamic.qsc.de) |
| 10:19.00 | sezuan | Does anybody know what the 3 in PCMU/8000/3 means? |
| 10:19.29 | JerJer[mobile] | not I, says me |
| 10:21.04 | JerJer[mobile] | moo |
| 10:21.23 | _omer | JerJer: Is there any document explaining how to connect Asterisk with MYSQL... |
| 10:21.34 | JerJer[mobile] | i'm sure the wiki has something |
| 10:21.44 | JerJer[mobile] | and/or the asterisk documenation project |
| 10:22.31 | JerJer[mobile] | Clavell: registration failed means the shared secret is not right or host=dynamic is not set on the type=peer |
| 10:22.47 | JerJer[mobile] | make sure the authentication realm is 'asterisk' |
| 10:22.59 | JerJer[mobile] | on your SIP ua |
| 10:23.21 | Delvar | _omer: how do you mean connecto ASterisk to MYSQL? for CDR's there is cdr_mysql, for AGI there is php/perl etc.., for user accounts there is realtime in CVS, for dialplan DBGET etc.. there is a mod |
| 10:23.33 | *** join/#asterisk otaku42 (otaku@otaku42.developer.madwifi) |
| 10:23.36 | otaku42 | hi all |
| 10:24.12 | JerJer[mobile] | oh waitress, check please |
| 10:24.20 | otaku42 | are there any documents describing the steps to take in order to tell asterisk not to log to any file, but use syslog (and only syslog) instead? |
| 10:24.39 | JerJer[mobile] | otaku42: see logger.conf |
| 10:24.52 | JerJer[mobile] | logger.conf.sample to be exact |
| 10:25.38 | _omer | Delvar: now I explain you, Calls are coming over the IP to my ASTERISK (SIP) ... and Reps are attending it thru IP Phones....now I need to get the reports.. |
| 10:26.22 | Delvar | so you need access to the CDRS? |
| 10:27.15 | _omer | right now, I dont know anything how it would work....if Asterisk is already saving info somewhere or I need to setup any database... |
| 10:27.49 | Delvar | ok by default it saves the CDS in a flat file as csv, Master.csv |
| 10:27.49 | _omer | My asterisk dont have any Database or CDR Server..... |
| 10:28.00 | _omer | ok.....and where is it located? |
| 10:28.10 | Delvar | you can make it save CDR's to mySQL or even ODBC |
| 10:28.28 | Delvar | hmm i cant rember i think its /var/lib/asterisk |
| 10:28.42 | _omer | ok.... |
| 10:30.13 | _omer | Delvar: is there any document who could explain about that? |
| 10:30.25 | Delvar | _omer: http://www.voip-info.org/wiki-Asterisk+cdr+mysql |
| 10:30.40 | Delvar | step by step guide to setup mysql CDR's |
| 10:31.01 | Delvar | ~googe asterisk mysql CDR |
| 10:31.06 | Delvar | ~google asterisk mysql CDR |
| 10:31.27 | _omer | wow.... |
| 10:32.41 | _omer | thanks Delvar |
| 10:33.06 | Delvar | np |
| 10:38.15 | _omer | Delvar : Your given URL works!!!! ...I read and found Master.csv ..I think fetching report from Master.csv is better than setting up MYSQL ..etc .. |
| 10:38.18 | _omer | :) |
| 10:38.44 | Delvar | cool |
| 10:39.02 | _omer | specially for a newbie like me ;) |
| 10:39.16 | Delvar | just watch out teh file doesnt get over 2gb |
| 10:39.29 | RoyK | Delvar: wot? |
| 10:39.42 | RoyK | Delvar: asterisk doesn't support largefile? |
| 10:39.45 | Delvar | doesnt it mess up when the Master.csv goes voer 2 gig? |
| 10:40.10 | Delvar | never got it that big myself its jsut what i heard |
| 10:40.25 | _omer | how to make Master.csv empty? |
| 10:40.35 | _omer | just edit and delete the text? |
| 10:46.50 | *** join/#asterisk wiz8291 (~dang@freeon.goscomb.net) |
| 10:46.59 | wiz8291 | hi guys, anyone know of any asterisk consultants in London? |
| 10:47.02 | wiz8291 | (UK) |
| 10:47.37 | kapejod | wiz8291: i would know one in ireland |
| 10:47.45 | fenlander | wiz: we are in Cambridge |
| 10:47.47 | wiz8291 | we have a serious echo problem that needs sorting #;( |
| 10:50.33 | clive- | wiz maybe jason could help |
| 10:53.44 | wiz8291 | i can hear everyone perfectly... its just the person who calls in to our system gets their own voice back |
| 10:53.49 | cjk | sorry for asking this basic questions, but what is a gatekeeper used for |
| 10:54.14 | kapejod | cjk: it's like a registrar server (sip) |
| 10:54.28 | kapejod | wiz8291: what phones are you using? |
| 10:54.34 | wiz8291 | sayson 390s |
| 10:54.37 | *** join/#asterisk fantomax1 (~fanto@81.208.114.250) |
| 10:54.39 | fantomax1 | hi all |
| 10:54.47 | wiz8291 | with a rhino channel bank |
| 10:55.13 | *** join/#asterisk gambolputty (~gambolput@cblmdm72-240-241-108.buckeyecom.net) |
| 10:55.16 | kapejod | and zaptel t1 card? |
| 10:55.29 | wiz8291 | TE410P |
| 10:55.43 | wiz8291 | E1 PRI on span 1 and the rhino on span 2 |
| 10:55.44 | cjk | kapejod: ok so its not that important |
| 10:56.11 | kapejod | wiz8291: echocancelwhenbridged=yes in zapata.conf? |
| 10:56.18 | wiz8291 | kapejod: yes |
| 10:56.30 | RoyK | shit. |
| 10:56.34 | kapejod | echotraining=? |
| 10:56.36 | RoyK | another memory leak |
| 10:56.42 | wiz8291 | no echotraining |
| 10:57.17 | wiz8291 | but i have tried it, it makes no difference |
| 10:57.17 | RoyK | frame.c:305 allocates memory unstoppably |
| 10:57.17 | fantomax1 | if I put an extension like this , I'll have any dialed number going out to server_b ? |
| 10:57.17 | fantomax1 | here is the ext |
| 10:58.16 | fantomax1 | _X.,2,Dial(SIP/$(EXTEN:2)@serverb_out) |
| 11:01.54 | RoyK | wtf does ast_frdup do? |
| 11:02.37 | kapejod | wiz8291: did you play with rxgain and txgain? |
| 11:03.10 | wiz8291 | yup, have been playing with those for a week, the echo only goes when they can't hear me anymore |
| 11:03.59 | kapejod | hehe |
| 11:04.29 | *** join/#asterisk tzanger (~tzanger@mixdown.ca) |
| 11:04.34 | *** join/#asterisk sandnigg0r (~niggerplz@66-55-197-254.gwi.net) |
| 11:04.34 | kapejod | are you sure that your te410p is not dropping audio (look for missed irqs)? |
| 11:04.45 | wiz8291 | there are no missed IRQs at all |
| 11:04.59 | *** join/#asterisk EvlHimeko (~himeko@S01060040ca128fc3.ed.shawcable.net) |
| 11:08.57 | *** join/#asterisk Maksim (~max@213.142.207.20) |
| 11:18.03 | *** join/#asterisk meppl (mephisto@p54AAEFDE.dip.t-dialin.net) |
| 11:18.52 | *** join/#asterisk newbien (~e@147.241.33.65.cfl.res.rr.com) |
| 11:18.53 | *** join/#asterisk gres (~serg@81.222.48.242) |
| 11:22.19 | *** join/#asterisk w0lvie (~jtkane@208.51.186.135) |
| 11:23.21 | *** part/#asterisk w0lvie (~jtkane@208.51.186.135) |
| 11:24.02 | *** join/#asterisk bsunil (~bsunil@202.54.37.182) |
| 11:24.37 | bsunil | can any body tell me how to trace a call through asterisk |
| 11:25.57 | RoyK | set verbose 9 |
| 11:26.00 | RoyK | might help |
| 11:26.06 | RoyK | :P |
| 11:35.16 | RoyK | ~sex |
| 11:35.45 | jbot | updatedb; locate; talk; date; cd; strip; look; touch; finger; unzip; uptime; gawk; head; apt-get install condom; mount; fsck; gasp; more; yes; yes; yes; more; umount; apt-get remove --purge condom; make clean; sleep |
| 11:40.27 | Zeeek | when I'm sleepy after all that |
| 11:43.04 | *** join/#asterisk gth (~gth@cC3013249.inet.catch.no) |
| 11:48.18 | RoyK | anyone that can help me sort out a memory leak? |
| 11:49.29 | Zeeek | I forget, what did you want? |
| 11:49.55 | MikeJ[Laptop] | RoyK, what seems to be the problem |
| 11:50.10 | MikeJ[Laptop] | maybe very quick, I need to go to the office in a couple minutes |
| 11:50.28 | *** join/#asterisk jeffik (~Jeff@69.158.19.117) |
| 11:51.05 | *** join/#asterisk ManxPwr (~eric@adsl-6-105-205.msy.bellsouth.net) |
| 11:51.24 | MikeJ[Laptop] | mornning Manx |
| 11:51.29 | RoyK | MikeJ[Laptop]: buf = malloc(len); called from frame.c:305 isn't getting released |
| 11:51.52 | MikeJ[Laptop] | what happens to that buffer? |
| 11:52.00 | MikeJ[Laptop] | what is it for. |
| 11:52.00 | RoyK | this seems to be getting called from channel.c:377 |
| 11:52.11 | RoyK | function ast_frdup |
| 11:52.15 | RoyK | 1.0.7 |
| 11:52.21 | Sato1 | is there any other way to see "iax2 show user <user>" in the cli? |
| 11:52.22 | RoyK | called from ast_queue_frame |
| 11:52.32 | Sato1 | the option is not available actually |
| 11:52.56 | MikeJ[Laptop] | I beleive when the frame is queued, it is freed after beein written, if you don't queue it, you need to do it yourself |
| 11:53.00 | MikeJ[Laptop] | off the top of my head |
| 11:53.20 | MikeJ[Laptop] | but I don't have time to pick through it |
| 11:53.28 | MikeJ[Laptop] | sorry |
| 11:55.59 | RoyK | ManxPower: g'day |
| 11:56.16 | ManxPower | The Register has such funny headlines. "Yahoo! tests! VoIP!" |
| 11:56.20 | ManxPower | hello royk |
| 11:56.56 | Zeeek | ManxPower I have a correction for you about nat=yes in [general] |
| 11:57.03 | Zeeek | never say never |
| 11:57.29 | wiz8291 | ok, my channel bank has been upgraded |
| 11:57.37 | wiz8291 | now just to wait for a call to see if its fixed the echo! |
| 11:58.31 | RoyK | any idea how I can track this leak? astmm says buf = malloc(len); on line 305 in frame.c is being called excessively and that buffer does not seem to be freed. I beleive it's called from ast_queue_frame() in channel.c:377 |
| 12:02.47 | kapejod | file a bug :) |
| 12:02.51 | Moc[Toronto] | damn night |
| 12:03.08 | *** join/#asterisk afaict (~afaict@201.6.255.86) |
| 12:04.26 | ManxPower | Guys, I've not even finished my first cup of coffee. don't make me think. |
| 12:05.24 | Sato1 | hehehe |
| 12:06.27 | ManxPower | Sometimes I don't realize just how much talent I have. This morning I managed to not get the coffee pot put in the coffee maker quite right. Went out for a smoke, came back and there was coffee EVERYWHERE, except in the actual coffee pot. |
| 12:07.11 | *** join/#asterisk shaZwaz (~chatzilla@203.81.196.167) |
| 12:07.35 | shaZwaz | hi all |
| 12:07.52 | afaict | Can I set a machine with a FXO card connected in PTNS running asterisk and another machine with a FXS card connected with a normal telephone running asterisk ? Can I set asterisk to talk each other ? |
| 12:07.58 | *** join/#asterisk makhtar (~ageller@mail.bulletinnews.com) |
| 12:08.11 | shaZwaz | u mean PSTN ? |
| 12:08.27 | afaict | public telephone network. i dont remember the name. |
| 12:08.54 | shaZwaz | AFAIK its Public Switched Telephone Network |
| 12:09.00 | afaict | shaZwaz, thanks. |
| 12:09.06 | afaict | s/PTNS/PSTN/ |
| 12:09.08 | afaict | :) |
| 12:09.25 | afaict | Can I set a machine with a FXO card connected in PSTN running asterisk and another machine with a FXS card connected with a normal telephone running asterisk ? Can I set asterisk to talk each other ? |
| 12:09.34 | sudhir492 | ManxPower: I can understand your problem. I am waiting for my first cup of coffee too |
| 12:10.03 | afaict | one machine will be in one office and another will be in another office so it will be connected with IPSEC |
| 12:10.11 | shaZwaz | does any one know of a script to block some code prefixes dialed ? |
| 12:10.31 | afaict | will it works ? |
| 12:10.56 | Zeeek | shaZwaz you don't need a script, just use wilcards to trap them |
| 12:11.08 | sudhir492 | Yesterday I had to rush in the morning for an emergency and by 10 o'clock, my eyes were hurting so badly, and my head was exploding. Didnt realize how much caffeine addiction I have developed |
| 12:11.27 | afaict | sudhir492, lol. nutz |
| 12:11.33 | onkeltimm | shaZwaz: exten => _0900.,1,Congestion or st like that |
| 12:11.48 | shaZwaz | hi Zeeek |
| 12:11.56 | Zeeek | hi |
| 12:11.58 | onkeltimm | or whatever you dial for phonesex in your country |
| 12:12.06 | Zeeek | 911 |
| 12:12.14 | shaZwaz | its a list of like 10-15 prefixes |
| 12:12.30 | shaZwaz | ranging from 2-4 digits |
| 12:12.40 | Zeeek | you can either list them like the 900 above or develop a wildcard expression |
| 12:13.01 | Zeeek | pastebin them to us and we'll work on it |
| 12:13.08 | afaict | no one answered my question . :| |
| 12:13.11 | shaZwaz | so pattern matching might not be a good idea |
| 12:13.18 | Zeeek | the greatest minds are avaialable free (I'm busy though) |
| 12:13.23 | ManxPower | Do eurpopean cities have large numbers of crazy (as in should by in a psych ward crazy) homeless people? |
| 12:13.35 | Zeeek | yes |
| 12:13.36 | shaZwaz | ok pasting now |
| 12:13.36 | afaict | Can I set a machine with a FXO card connected in PSTN running asterisk1 and another machine with a FXS card connected with a normal telephone running asterisk2 ? Can I set asterisk1 to talk asterisk2 ? |
| 12:13.44 | Zeeek | why, thinking of bringing mace? |
| 12:13.57 | onkeltimm | ManxPower: you arrived in the netherlands? yes, then ;) |
| 12:13.59 | vaewyn | afaict: yes... using an IAX connection |
| 12:14.06 | ManxPower | Zeeek, no, just wondering. |
| 12:14.11 | *** join/#asterisk bjohnson (~bjohnson@jecinc.tor.istop.com) |
| 12:14.22 | Zeeek | it's harder to tell noiwadays with cellphones |
| 12:14.32 | Zeeek | if they're actually taling to someone I mean |
| 12:14.32 | afaict | vaewyn, thanks |
| 12:15.01 | vaewyn | Zeeek: especially those dorks with the ear clamshell bluetooth earpiece |
| 12:15.05 | ManxPower | Zeeek, I just thought it was a symptom of the USA not having a public health system |
| 12:15.22 | Zeeek | no I think it's all over |
| 12:15.26 | afaict | vaewyn, so if I make a call from asterisk2 I will be able to transfer to asterisk1 and then send call to PSTN ? |
| 12:15.35 | vaewyn | afaict: yep |
| 12:15.42 | afaict | vaewyn, love ya |
| 12:15.45 | Zeeek | I really want a little device that jams cellphones from time to time |
| 12:16.05 | vaewyn | I have one that jams bluetooth... and wifi... kindof fun :P |
| 12:16.39 | vaewyn | They sold it as a 'Gigarange Cordless phone' :P |
| 12:16.49 | Zeeek | hahah |
| 12:16.55 | vaewyn | but it is a great wifi and bluetooth jammer |
| 12:16.59 | onkeltimm | lol |
| 12:17.00 | shaZwaz | Zeeek: http://pastebin.ca/12042 |
| 12:17.24 | ManxPower | one of my big annoyances with big cities is crazy homeless people. |
| 12:17.26 | vaewyn | pop that puppy on and watch the businessmen howl |
| 12:17.34 | afaict | vaewyn, where can I read about Telecom/asterisk ? docs about asterisk explain something about telecom too or only how to set asterisk |
| 12:17.41 | ManxPower | I just notice it more often now that I don't live in a big city anymore |
| 12:18.10 | Zeeek | afaict |
| 12:18.11 | Zeeek | Starter tutorial: |
| 12:18.11 | Zeeek | http://www.onlamp.com/pub/a/onlamp/2003/07/03/asterisk.html |
| 12:18.11 | Zeeek | http://www.onlamp.com/pub/a/onlamp/2004/01/22/asterisk2.html |
| 12:18.12 | Zeeek | http://www.automated.it/guidetoasterisk.htm |
| 12:18.12 | Zeeek | THE reference of the moment: |
| 12:18.12 | Zeeek | http://www.asteriskdocs.org |
| 12:18.17 | ManxPower | ~docs |
| 12:18.30 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org |
| 12:18.30 | ManxPower | ~mailinglist |
| 12:18.35 | jbot | rumour has it, mailinglist is Search Asterisk mailing lists by prepending site:lists.digium.com to your Google search. Browse the mailing list archive at http://lists.digium.com/ |
| 12:18.35 | Zeeek | ~Zeke |
| 12:18.36 | shaZwaz | ~stuff |
| 12:18.37 | jbot | ACTION is stuffing George* |
| 12:18.37 | Zeeek | ~sex |
| 12:18.38 | jbot | updatedb; locate; talk; date; cd; strip; look; touch; finger; unzip; uptime; gawk; head; apt-get install condom; mount; fsck; gasp; more; yes; yes; yes; more; umount; apt-get remove --purge condom; make clean; sleep |
| 12:18.56 | shaZwaz | Zeeek: you had a look at http://pastebin.ca/12042 ? |
| 12:19.04 | Zeeek | not yet I'm very busy |
| 12:19.10 | shaZwaz | oh |
| 12:19.25 | afaict | Zeeek, thanks |
| 12:19.36 | Zeeek | afaict number two will get you started |
| 12:19.42 | Zeeek | ast+PSTN |
| 12:20.29 | Zeeek | shaZwaz taking a quick look: _00[128]X. will get you started |
| 12:20.46 | wiz8291 | seems the channel bank upgrade may have done the trick |
| 12:20.49 | Zeeek | that's I can give you free :) |
| 12:20.56 | *** join/#asterisk [TK]D-Fender (~joe@4.67.252.216.dsl1.colba.net) |
| 12:21.08 | wiz8291 | out of 3 calls, one had a very slight echo... and that was to a cell phone with bad reception :))) |
| 12:21.23 | *** join/#asterisk L|NUX (~linux@202.5.145.54) |
| 12:21.42 | shaZwaz | Zeeek: need to block these codes |
| 12:21.42 | onkeltimm | shaZwaz: these are just a few. just put them at the beginning of your outgoing context. pattern matching might be a Bad Idea, especially if the numbers may change over time |
| 12:22.00 | *** join/#asterisk switch (~switch@61.206.115.5.user.ad.il24.net) |
| 12:22.05 | Zeeek | shaZwaz someone already answered that the first time |
| 12:22.27 | Zeeek | onkeltimm I agree, it was just to show the idea |
| 12:22.30 | *** join/#asterisk adnans (~adnans@noterik2.demon.nl) |
| 12:22.35 | Zeeek | although he does want 001* |
| 12:22.56 | onkeltimm | shaZwaz: like start with: _001.,1,PlayBack(bad_boy) |
| 12:23.01 | onkeltimm | _001.,2,Hangup |
| 12:23.36 | RoyK | ~nickometer L|NUX |
| 12:23.38 | Zeeek | or just congestion to save time :) |
| 12:23.57 | shaZwaz | you put these exten lines in a context and add it to the outgoing context |
| 12:24.01 | onkeltimm | or in your case more like _9001. |
| 12:24.09 | Zeeek | heh they used to sell dial locks for phones here |
| 12:24.29 | vaewyn | shaZwaz: http://pastebin.ca/12043 |
| 12:24.29 | RoyK | Zeeek: dial locks??? |
| 12:24.52 | Zeeek | yeah a little thing to go in the first hole of the old style dial |
| 12:25.02 | RoyK | Zeeek: hehehe |
| 12:25.24 | RoyK | Zeeek: so all you had to do was hammer the pulse dialling yourself? |
| 12:25.32 | Zeeek | it was a national passtime to use the phone at work to call friends |
| 12:25.49 | Zeeek | RoyK yes you could easily with any dexterity |
| 12:26.14 | L|NUX | hmm |
| 12:26.34 | RoyK | Zeeek: or perhaps if you could get those old digital watches from the eighties that had a phone book with DTMF dialling :) |
| 12:27.09 | Zeeek | they that in Moscow for use with asterisk |
| 12:27.16 | shaZwaz | vaewyn: seems to be nice except 021 |
| 12:27.33 | vaewyn | shaZwaz: don't need 021 since 02. matches that as well |
| 12:27.46 | shaZwaz | ah |
| 12:28.04 | vaewyn | if you block 02 then 021 block is redundant :P |
| 12:28.08 | Zeeek | $$$$drinnnnng$$$$ |
| 12:28.11 | shaZwaz | WDITOT |
| 12:30.13 | shaZwaz | vaewyn: thanks a bunch |
| 12:31.29 | vaewyn | shaZwaz: no problem |
| 12:32.07 | shaZwaz | and Zeeek too |
| 12:32.53 | *** join/#asterisk cmk (~cmk_@p54A3D31F.dip.t-dialin.net) |
| 12:34.03 | *** join/#asterisk dennis (~dennis@2002:d5ef:c64b:1:0:0:0:1) |
| 12:35.40 | tzafrir | speaking of dring |
| 12:36.23 | Zeeek | tzafrir you rang? |
| 12:36.37 | tzafrir | I talked with somebody here yesterday about distinctive ring and mentioned it didn't work for me. On a second chek: it does |
| 12:36.52 | Zeeek | yopu mean r2 r3 etc? |
| 12:36.57 | tzafrir | yes |
| 12:37.04 | Zeeek | I use it in 1.0.6 |
| 12:37.20 | tzafrir | the local line here rings differently in local and external call |
| 12:37.28 | Zeeek | same here |
| 12:37.47 | Zeeek | great facility allows one to ignore people who are calling from inside the co. |
| 12:37.56 | Zeeek | or the opposite |
| 12:38.19 | tzafrir | actually, I kind of trust insiders |
| 12:38.35 | tzafrir | outsiders still get the default demo. |
| 12:39.26 | *** join/#asterisk dmccollum (~dmccollum@eycb01-00-cntnga-69-164-245-72.atlaga.adelphia.net) |
| 12:39.53 | dmccollum | morning everyone. |
| 12:39.56 | onkeltimm | i know i am asking this over and over again, maybe i am asking the wrong question: how do i do callwaiting and callgroups with iax clients? please enlighten me... |
| 12:40.09 | *** join/#asterisk harryvv (~leonardo@S010600a0c93f6f7e.vs.shawcable.net) |
| 12:41.15 | onkeltimm | as far as i can see, chan_iax.c does not implement callgroups like chan_sip.c does for example. so callgroups are not possible with iax? |
| 12:42.08 | onkeltimm | web sources tell me that with iax, callwaiting should be implemented by the phone... but there must be a workaround? |
| 12:42.35 | eper-werk | running in circles trying to set the zap cfg's for this isdn30e |
| 12:43.22 | *** join/#asterisk bofh42 (~bofh42@pD9EC195B.dip0.t-ipconnect.de) |
| 12:43.36 | *** join/#asterisk MikeJ[Laptop] (~ircatjerr@mi.origenfinancial.com) |
| 12:43.57 | *** join/#asterisk switch (~switch@61.206.115.5.user.ad.il24.net) |
| 12:44.22 | *** join/#asterisk jwitte (~jwitte_@port-212-202-101-206.static.qsc.de) |
| 12:45.01 | jwitte | Hello - anybody out there having experiences with Asterisk on Dell 2850? |
| 12:45.53 | harryvv | should work fine |
| 12:46.01 | Zeeek | be sure to search the mailing list too |
| 12:46.10 | *** part/#asterisk Clavell (~clavell@suse.satrax.hu) |
| 12:46.13 | jwitte | http://www.digium.com/index.php?menu=compatibility |
| 12:47.06 | MikeJ[Laptop] | RoyK, find the leak? |
| 12:49.19 | wiz8291 | anyone tell me how to add a new page to the voip-info wiki? |
| 12:50.16 | *** join/#asterisk epoch (epoch@octane.breakbeats.org) |
| 12:50.35 | Zeeek | you need to have a login account (free with purchase) |
| 12:50.43 | Zeeek | if you have that, look for edit |
| 12:50.46 | RoyK | MikeJ[Laptop]: no, see http://bugs.digium.com/view.php?id=4318 |
| 12:50.49 | wiz8291 | yes, i have edit |
| 12:50.56 | wiz8291 | but i want to add a page, not edit an existing one |
| 12:51.04 | Zeeek | I believe there are instructions somewhere to add |
| 12:51.11 | wiz8291 | it just says click edit and add a link |
| 12:51.17 | Zeeek | yeah |
| 12:51.21 | wiz8291 | http://www.voip-info.org/tiki-index.php?page=How%20to%20add%20information%20to%20this%20wiki |
| 12:51.21 | RoyK | gotta |
| 12:51.22 | Zeeek | I'm trying to remember |
| 12:51.22 | RoyK | go |
| 12:51.23 | RoyK | later |
| 12:51.49 | Zeeek | bye |
| 12:53.41 | eper-werk | should zap show channels show 'things' even if the isdn cables not plugged in? |
| 12:54.28 | Zeeek | I think it shows what is entered in zapata.conf |
| 12:55.07 | vaewyn | ~seen JerJer |
| 12:55.23 | jbot | jerjer <~JerJer@DSL-226.206-rt-bras.che.centurytel.net> was last seen on IRC in channel #asterisk, 2d 23h 43m 9s ago, saying: 'ManxPower: sounds interesting, but i've still never used it'. |
| 12:55.31 | Zeeek | use the mobile version |
| 12:55.47 | vaewyn | ~seen JerJer[mobile] |
| 12:55.48 | jbot | jerjer[mobile] <~jj@ip68-103-26-140.ks.ok.cox.net> was last seen on IRC in channel #asterisk, 2h 30m 56s ago, saying: 'logger.conf.sample to be exact'. |
| 12:56.04 | Zeeek | ~seen JerJer[mobile] |
| 12:56.05 | jbot | jerjer[mobile] <~jj@ip68-103-26-140.ks.ok.cox.net> was last seen on IRC in channel #asterisk, 2h 31m 13s ago, saying: 'logger.conf.sample to be exact'. |
| 12:58.21 | kapejod | eper-werk: cat /proc/zaptel/* | grep Span |
| 12:58.32 | kapejod | eper-werk: and on the * CLI: pri show span X |
| 12:59.35 | *** join/#asterisk ariel_ (~Ariel@68.157.125.248) |
| 13:00.05 | eper-werk | oh the cards there and working but im like playing "try every different cfg" for it currently |
| 13:01.52 | eper-werk | people keep noticing when i pull the cable out of the old pbx *dum de dum* |
| 13:02.47 | *** join/#asterisk memic (~memic@chicago089.server4free.de) |
| 13:05.25 | *** join/#asterisk Markaci (~Markaci@70.26.125.95) |
| 13:05.36 | *** join/#asterisk morris (~morris@pcworkshop.plus.com) |
| 13:05.40 | vaewyn | Hey... anyone found a way to toggle the MWI light on a nortel when you have * take over the voicemail system? |
| 13:06.32 | *** join/#asterisk greg_work (~greg@d221-73-240.commercial.cgocable.net) |
| 13:11.21 | morris | Anyone free to spoon feed ,e asterisk? |
| 13:11.27 | morris | ,e = me |
| 13:11.28 | morris | i have it running |
| 13:11.38 | morris | setup to a sipgate account, accepting calls thanks to the demo script |
| 13:11.48 | morris | i would like to learn how to add a sip client |
| 13:12.01 | morris | and route calls to and from that and my sipgate accounts |
| 13:12.21 | MikeJ[Laptop] | morris, no |
| 13:12.25 | MikeJ[Laptop] | but: |
| 13:12.28 | MikeJ[Laptop] | ~docs |
| 13:12.29 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org |
| 13:13.01 | morris | any keyword help speed my find up? |
| 13:13.02 | MikeJ[Laptop] | do some reading and what you can't figgure out people will ususally help with |
| 13:13.06 | morris | yea |
| 13:13.11 | morris | ok thats totaly fair ;p |
| 13:13.13 | morris | however.. |
| 13:13.17 | morris | man im tired from all the reading |
| 13:13.17 | morris | lol |
| 13:13.36 | morris | thx, brb |
| 13:13.39 | MikeJ[Laptop] | but prefacing with hey, I did nothing to learn for myself, so who is willing to spend their time teaching me..... |
| 13:13.47 | MikeJ[Laptop] | not so well received usually |
| 13:15.14 | Zeeek | how do I set up and configure asterisk? I'm in a hurry |
| 13:15.18 | morris | lol |
| 13:15.24 | *** join/#asterisk welby (~welby@tollcross.edihost.co.uk) |
| 13:15.28 | *** join/#asterisk Markaci (~Markaci@CPE00508de53d8f-CM00080d924284.cpe.net.cable.rogers.com) |
| 13:15.31 | morris | welby mcroberts |
| 13:15.43 | welby | lo |
| 13:15.44 | *** join/#asterisk Gand_DJ (~gandalf@ptr-207-54-104-24.ptr.terago.ca) |
| 13:17.06 | Zeeek | ???? for the bad advice yesterday ?? |
| 13:17.25 | Zeeek | well it wasn't advice to me |
| 13:17.34 | vaewyn | Zeeek: First... take off your watch.... err wait... asterisk... ohh I read that wrong |
| 13:17.35 | vaewyn | ;P |
| 13:17.36 | Zeeek | so I ignored it |
| 13:17.59 | *** join/#asterisk pointer (pointer@aj.catt.com) |
| 13:18.41 | pointer | I'm overlooking something simple in my cfg.... |
| 13:18.51 | pointer | I'm getting cidname/num set to the cidname |
| 13:18.52 | [TK]D-Fender | Zeeek : "You must learn patience. PATIENCE?! How long's that going to take?!" |
| 13:19.01 | pointer | on incoming calls to a tdm400p |
| 13:19.48 | [TK]D-Fender | pointer : pastebin your zapata & zaptel and I'll take a look |
| 13:20.11 | *** join/#asterisk pussfeller (~todd@t1-rtc-woodlawn.rtcol.com) |
| 13:21.27 | *** part/#asterisk gth (~gth@cC3013249.inet.catch.no) |
| 13:22.47 | Zeeek | your money or your life ? "Take my life, you can't do shit in this town with no money." |
| 13:25.23 | pointer | [TK]D-Fender: k |
| 13:25.31 | *** join/#asterisk pycsusz (~pycsusz@pluto.euronetrt.hu) |
| 13:26.29 | pycsusz | Hi Everybody! Somebody listened something about this warning message: May 18 15:27:32 WARNING[1150]: chan_zap.c:7970 pri_dchannel: PRI Error: We think we're the network, but they think they'rethe network, too. |
| 13:26.40 | pycsusz | ??? |
| 13:26.47 | Zeeek | nice! |
| 13:27.00 | pycsusz | thanx |
| 13:27.37 | jsharp | change from pri_net to pri_cpe in zapata.conf |
| 13:28.07 | pycsusz | that's all? |
| 13:28.08 | *** join/#asterisk Fabe (~spamhere@pD95B0BF7.dip0.t-ipconnect.de) |
| 13:28.14 | jsharp | Aye. |
| 13:28.19 | pycsusz | thanx |
| 13:28.21 | *** join/#asterisk cpatry (~grepmoo@65.39.228.5) |
| 13:28.55 | cpatry | the polycom SIP 500 is deprecated? |
| 13:29.01 | Zeeek | mine is! |
| 13:29.17 | pointer | [TK]D-Fender: http://pastebin.com/286065 |
| 13:30.04 | vaewyn | I think all the X00 are deprecated now... the new ones they are shipping are the 301 501 |
| 13:30.10 | vaewyn | 600 is still 600 I think |
| 13:30.35 | Zeeek | and the difference is? |
| 13:30.40 | Zeeek | 1 ? |
| 13:30.52 | vaewyn | js... let me find the announcement |
| 13:30.54 | onkeltimm | more expensive, i'd say ;) |
| 13:31.09 | vaewyn | nah... same price |
| 13:31.13 | [TK]D-Fender | pointer : add "callerid=asrecieved" |
| 13:31.51 | [TK]D-Fender | Zeeek : they upped the ram on them |
| 13:31.55 | [TK]D-Fender | and bug fixes |
| 13:32.11 | [TK]D-Fender | 600 supposedly remains the same though |
| 13:32.14 | Zeeek | bug fixes would be under firmware tho |
| 13:32.40 | vaewyn | Something else to cause they only offer the x01s in 3 countries for some reason so far |
| 13:32.48 | Zeeek | is there a publicly available book on these phones somewhere? |
| 13:33.07 | Zeeek | config I mean |
| 13:33.21 | cpatry | maybe theres a link based on http://www.polycom.com/products_services/1,,pw-182-10533,FF.html |
| 13:33.26 | [TK]D-Fender | Zeeek : they have a rough book on Polycom.com |
| 13:33.36 | onkeltimm | vaewyn: maybe they did not get FCC/CE in other countries |
| 13:33.38 | pycsusz | jsharp thank you very much for your help it's already working :) |
| 13:33.46 | Zeeek | [TK]D-Fender thx |
| 13:34.13 | *** join/#asterisk Trakk (~Trakk@adsl-8-245-110.mia.bellsouth.net) |
| 13:34.20 | [TK]D-Fender | I'm tossing up between going with Cisco 7960G's and Polycom IP 600's here for our new phone system |
| 13:34.30 | vaewyn | Zeeek: for configuring them this is the ultimate guide: http://www.freedomphones.net/polycom/files/docs/Admin_Guide-SoundPoint_IP_SIP_2004-06-16.pdf for configs of the actual phones (as in what they have) check out their comparison on the web site |
| 13:34.39 | *** join/#asterisk JimVanM (~jimvanm@HSE-Toronto-ppp3489845.sympatico.ca) |
| 13:34.39 | vaewyn | I'd take the polycoms :} |
| 13:34.57 | jsharp | pycsusz: Excelllent. |
| 13:36.02 | [TK]D-Fender | vaewyn : yeah I think we had this discussion earlier, but I got a nasty good price on the 7960G's :/ Makes the choice a LOT harder. |
| 13:36.08 | cpatry | damn, i bought the 500, 2 weeks ago. |
| 13:36.29 | Zeeek | vaewyn thx for that - I was expecting to have to be a dealer to download or something |
| 13:36.50 | [TK]D-Fender | Zeeek : that guide is avail right on Poly's site :) |
| 13:36.54 | vaewyn | Zeeek: nope... just have to have a link to the dealers stuff :P |
| 13:37.08 | pointer | [TK]D-Fender: hrm...I added that and no go...still get cidname for both |
| 13:37.21 | Zeeek | It also doesn't help that every version filename has a different numbering system |
| 13:37.27 | pointer | [TK]D-Fender: exten => s,3,SetCIDNum(${CALLERIDNUM}) |
| 13:37.30 | Zeeek | I prolly have 20 version of the same doc now |
| 13:37.32 | wiz8291 | http://www.voip-info.org/tiki-index.php?page=UK%20Asterisk%20Details |
| 13:38.34 | [TK]D-Fender | pointer : why are you using SetCIDNum like that? |
| 13:38.43 | [TK]D-Fender | pointer : there's no need. |
| 13:40.23 | *** join/#asterisk aminorex (~tony@c66.191.69.132.dul.mn.charter.com) |
| 13:40.29 | *** join/#asterisk newmedian (~crowlther@Quebec-HSE-ppp230300.qc.sympatico.ca) |
| 13:41.27 | welby | wiz8291, if only that was up about a month ago, i'd still have hair |
| 13:41.37 | wiz8291 | welby: sorry ;-) |
| 13:41.49 | welby | top tip, don't trust WLR people (or in our case teleworst) |
| 13:41.51 | wiz8291 | its taken me about that long to get it all working properly lol |
| 13:41.52 | *** join/#asterisk switch (~switch@61.206.115.5.user.ad.il24.net) |
| 13:42.06 | wiz8291 | just fixed the last prob which was echo |
| 13:42.14 | welby | wlr bt isdn30e (also 8 ch), provided by TW, who are about as help ful as a kick in the teeth |
| 13:42.21 | wiz8291 | down to the channel bank it seems... updated the firmware and all is good |
| 13:42.25 | *** join/#asterisk HA (HeadachesA@wsip-68-99-73-32.tu.ok.cox.net) |
| 13:43.06 | *** part/#asterisk Zeeek (~icechat5@Zeeek.sustaining.supporter.pdpc) |
| 13:43.07 | welby | wiz8291, do you find that interantional calls that come in arnt preceded by a 0(or 00) |
| 13:43.30 | welby | and hence screws up your setting of the cidnum |
| 13:43.30 | wiz8291 | welby: national calls aren't preceeded with a 0 |
| 13:43.41 | wiz8291 | not sure about the internationals |
| 13:43.53 | welby | wiz8291, yup i know that, but internationals (afaik) should be |
| 13:44.08 | JimVanM | Has anyone worked with Asterisk in a HA environment? Running CompactPCI or AdvancedTCA stuff? Specifically, having the ability to run redundant CPU cores as available in CO switches or high end PBXs |
| 13:44.24 | wiz8291 | hmmm... anyone want to call a UK asterisk box from abroad? (i won't pick up) |
| 13:44.45 | pointer | [TK]D-Fender: ah, ok |
| 13:44.45 | *** join/#asterisk switch (~switch@61.206.115.5.user.ad.il24.net) |
| 13:45.02 | pointer | [TK]D-Fender: it'll pass it whatever I dial automatically? |
| 13:45.07 | *** part/#asterisk cpatry (~grepmoo@65.39.228.5) |
| 13:46.18 | *** join/#asterisk heison (~heison@216.191.251.226) |
| 13:46.51 | [TK]D-Fender | pointer : not sure I follow your intent. Could you paste you entire extensions.conf for me? I'll give it a quick scan. |
| 13:47.59 | pointer | [TK]D-Fender: I think I got it....thanks ;) |
| 13:48.14 | pointer | [TK]D-Fender: need to go afk to test on a real (ip500) phone |
| 13:48.52 | ariel_ | [TK]D-Fender, so what did you decide about your pick for phones? |
| 13:49.22 | *** join/#asterisk brad- (~brad@brad.developer.gentoo) |
| 13:49.24 | heison | is there any different between the old IAXy and the new ones? |
| 13:49.37 | heison | s/different/difference/ |
| 13:50.49 | ariel_ | heison, other the blue clear case? |
| 13:51.27 | *** part/#asterisk makhtar (~ageller@mail.bulletinnews.com) |
| 13:52.05 | *** join/#asterisk ctooley (~ctooley@pc51.utati.net) |
| 13:53.11 | [TK]D-Fender | ariel_ : I'm still undecided. Probably will suffer a great dea from "buyer's remorse" either way... |
| 13:53.33 | [TK]D-Fender | deal* |
| 13:53.47 | ariel_ | [TK]D-Fender, why don't you get one of each and test them your self? |
| 13:54.13 | [TK]D-Fender | I hate to say it that so far Cisco 7960G's are a CHEAPER solution than the best offer I could get for Polycom 600's :/ |
| 13:54.21 | ariel_ | I don't use the Cisco due to there Lisc issues. Since I like to make sure my customers have all the right lisc. |
| 13:55.17 | ariel_ | [TK]D-Fender, I use more the IP-500 then the IP-600. Which I have been getting from voipsupply fro under 200 each. |
| 13:55.19 | [TK]D-Fender | ariel_ : I verified that with Insight.ca (Can div of US company). I can get 7960's WITH SIP lic for 253CAD. I'm not sure I believe it.... |
| 13:55.47 | ariel_ | [TK]D-Fender, that would be a great deal if you can. |
| 13:55.57 | eper-werk | yay got it working! |
| 13:56.11 | [TK]D-Fender | ariel_ : I've checked out voipsupply and have found a cheaper place (considerably) : http://www.tritechcoa.com/product/126026.html |
| 13:56.13 | ariel_ | the 7960's are being replaced very soon. |
| 13:56.25 | eper-werk | nothing brought a bigger smile to my face when i called the office from my mobile and the asterisk session flooded with stuff¬ |
| 13:56.27 | ctooley | Yeah Polycom IP500's should go for about $180-185 |
| 13:56.33 | [TK]D-Fender | ariel_ : So's the IP 500 :) |
| 13:56.49 | [TK]D-Fender | ctooley : Atacomm sells 500's for 180$ |
| 13:56.55 | vaewyn | Buy 100+ and you can get IP500s down to 150$ or so |
| 13:57.08 | vaewyn | not many people can stomach that though |
| 13:57.28 | tzafrir | any RTFM on how do I write indications configuration for my country? |
| 13:57.49 | *** join/#asterisk TheEmperor (TheEmperor@60.49.109.105) |
| 13:57.52 | ctooley | vaewyn, which is pretty good considering bying IP500's in lots of 500 from Polycom costs more than that (for the reseller) |
| 13:58.21 | *** join/#asterisk anthm (~anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 13:58.21 | *** mode/#asterisk [+o anthm] by ChanServ |
| 13:58.38 | ctooley | And I haven't used the 501 but since it's the same price from distribution as the 500 I'm sure it's worth going that way over the 500's |
| 13:58.39 | vaewyn | ctooley: I know a supplier that got them in lots of 3,000 for the last couple purchases |
| 13:58.39 | ariel_ | [TK]D-Fender, they have some of there phones on there web with many mistakes. Also what is this Sip Auth req? |
| 13:59.08 | vaewyn | Heck... our next purchase will be 1500-1800 of them :P |
| 13:59.31 | onkeltimm | gurus, IAX and callgroup, puh-lease! if you tell me i have to code it i will code it, i will sell my immortal soul, anything! |
| 13:59.44 | ctooley | vaewyn, yeah but if you if you order 3000 of them from Polycom it takes forever. Even Ingram and Techdata are having trouble getting them because Polycom is having supply problems. |
| 14:00.38 | onkeltimm | just give me a go-nogo please |
| 14:01.02 | vaewyn | ctooley: Ingram's crap... not sure on Techdata but Ingram is pure garbage at getting this stuff |
| 14:01.07 | *** join/#asterisk Umaro (~umaro@209.140.74.64) |
| 14:01.21 | vaewyn | They'll claim anything to cover their goofups |
| 14:02.04 | Umaro | Hey guys, having an odd problem where when my sip phone tries to make a call through nufone, it gets accepted, but immediately gets hung up. when I make a call through a call-file to the same Local/ext@context, it works fine |
| 14:02.15 | ctooley | vaewyn, well, I'm not in the reseller business anymore, thank goodness, but they used to be a lot better than the "preferred partner" people that Polycom shoved at us. |
| 14:02.16 | [TK]D-Fender | ariel_ : I'm not sure personally what their "REQ" means. I do intend to find out as well as sourcing locally (I'm in Canada). |
| 14:02.23 | Umaro | any ideas? I've checked all the obvious things, contexts, codecs, etc. |
| 14:04.19 | ariel_ | wow atacomm has finally lowered his prices. |
| 14:04.28 | [TK]D-Fender | ariel_ : I want PoE so I'm looking at IP 600 anyways, not sure I want to fiddle with extra power conversion cords etc. |
| 14:04.36 | ariel_ | He was selling the IP-500 last year to over 240 dollars. |
| 14:05.00 | ariel_ | [TK]D-Fender, you have a poe switch? |
| 14:05.05 | [TK]D-Fender | ariel_ : and because I want the higer end functionality later I think I'd have to go with the 600's anywyas |
| 14:05.15 | ctooley | [TK]D-Fender, you can use IP500's with PoE just fine. |
| 14:05.18 | [TK]D-Fender | ariel_ : All new equipment to be bought. So yes, I will |
| 14:05.30 | vaewyn | The microbrowser in the 600 rocks... I wait for the day they get that in the 500 |
| 14:05.40 | [TK]D-Fender | ctooley : I know that, its just now you have to tack on 40$us to get the adapter cable for Peo |
| 14:06.18 | [TK]D-Fender | vaewyn : can you program the context/screen keys with a properly made browser page? |
| 14:06.23 | ctooley | That seems odd, I use my IP500's with a standard cable |
| 14:06.42 | [TK]D-Fender | ctooley : Everywhere I've read including Poly's site told me different |
| 14:07.00 | [TK]D-Fender | Esp those selling the cable ;) |
| 14:07.22 | ctooley | [TK]D-Fender, you can't use the cable with the power injector built into it, I just used a standard ethernet cable. |
| 14:07.39 | vaewyn | [TK]D-Fender: Yes and no... you hit the services button and it turns into a webbrowser with the arrow keys and the checkmark driving the browsing |
| 14:07.54 | ctooley | [TK]D-Fender, at worst it's just a cable, buy one, and make your own ethernet cables with that pin out |
| 14:08.10 | [TK]D-Fender | vaewyn : was hoping to implement XML-like services here.... |
| 14:08.27 | vaewyn | [TK]D-Fender: not yet... but it is supposedly coming... |
| 14:08.34 | [TK]D-Fender | ctooley : and bridge the power connector? eek..... sounds warranty-voiding |
| 14:08.48 | ctooley | huh? |
| 14:09.08 | [TK]D-Fender | vaewyn : I can always hope. l ike I said the diff in price is stomach turning. Its always easier when there's a dollar diff :) |
| 14:09.10 | ctooley | the PoE cable is just an ethernet cable. |
| 14:09.29 | *** join/#asterisk cmaj (~chris@65-37-6-42.nrp2.roc.ny.frontiernet.net) |
| 14:09.31 | ctooley | the only "special" cable I've ever seen from Polycom is the one that has the power port for the brick |
| 14:09.39 | *** join/#asterisk lehel (~lehel@82.79.20.17) |
| 14:09.43 | lehel | hello |
| 14:09.47 | [TK]D-Fender | ctooley, AFAIK, it has a std pwr connecter and their PoE adapter cable splits from incoming eithernet and Y's into both. |
| 14:10.20 | vaewyn | Nah... is just poe via RJ45 |
| 14:10.23 | ctooley | You mean the cable that lets you plug a brick into the wall instead of using PoE from a Powered swich? |
| 14:10.24 | [TK]D-Fender | ctooley : but they say quite plainly that you need their adapter cable for 802.11af |
| 14:10.27 | vaewyn | the 300 is the only one you have to split |
| 14:10.48 | [TK]D-Fender | vaewyn : well Poly did a good job of NOT expressing it that way... |
| 14:11.38 | vaewyn | The two different cables are because of the cisco polarity versus 802.11af polarity... get one cable... and make a bunch to match it :P |
| 14:11.39 | ctooley | If you you have a powered switch that supports 802.11af, you just need an ethernet cable that's pinned out as a straight through. If you don't have a powered switch and want to use the brick, you use the cable that comes with the phone. |
| 14:12.21 | ctooley | if you have a Cisco proprietary switch, then yes the cable is different but you can make those easy enough, there's dozens of websites with the pinout on the internet |
| 14:12.49 | ctooley | and it's just a matter of a standard ethernet cable with a slightly different pinout. Cat 5e and RJ45's |
| 14:13.50 | Umaro | Hey guys, having an odd problem where when my sip phone tries to make a call through nufone, it gets accepted, but immediately gets hung up. when I make a call through a call-file to the same Local/ext@context, it works fine. I've tried iax2 debug, didn't give me any more clues.. anyone know what the deal could be? |
| 14:13.58 | ctooley | if you're super paranoid, buy one cable from Polycom and use a cable tester to get the pinout |
| 14:14.35 | ctooley | Umara, try "sip debug" |
| 14:15.33 | mishehu | a client is ordering a PRI for use with asterisk. the provider was suggesting 4 digits outpost with a restriction of no 8, 9, or 0 as the 4th digit. they said that some systems can't handle this. I requested 10 digit outpost with no 8, 9, or 0 restriction. it is my understanding that asterisk does not have that limitation. can anybody verify if I am correct? |
| 14:15.45 | [TK]D-Fender | Ok, Poly's official FAQ PDF file states that for 802-11af you need to by an adapter cable. |
| 14:16.36 | jsharp | mishehu: Yes, asterisk can handle either 4 or 10 digit outpulse. |
| 14:17.22 | *** join/#asterisk grolloj (~chatzilla@slim-eth0.horizonlive.net) |
| 14:17.25 | lehel | Your Asterisk modules directory, located at |
| 14:17.25 | lehel | <PROTECTED> |
| 14:17.26 | lehel | <PROTECTED> |
| 14:17.26 | lehel | <PROTECTED> |
| 14:17.28 | mishehu | jsharp: with no 8, 9, or 0 restriction on the last digit, correct? |
| 14:17.32 | jsharp | Correct |
| 14:17.35 | mishehu | good |
| 14:17.46 | lehel | sorry... somebody have any idea? |
| 14:17.56 | *** join/#asterisk _omer (dfsdf@202.147.167.213) |
| 14:17.58 | jsharp | It means you've got version difference. |
| 14:18.10 | vaewyn | heck... asterisk can handle 100 digit ;P |
| 14:18.15 | ctooley | lehel, did you build asterisk from source code? |
| 14:18.15 | vaewyn | but that would be stupid |
| 14:18.17 | lehel | and how can i make all version the same? |
| 14:18.27 | jsharp | You need to shut down *, nuke /usr/lib/asterisk/modules, and "make install" again. |
| 14:18.27 | lehel | no... from cvs :( |
| 14:18.34 | Umaro | ctooley: looks like it's hanging up immediately after it sends my sip phone an 183 session progress message |
| 14:18.36 | ctooley | lehel, that's fine, that's source code |
| 14:18.59 | *** join/#asterisk desync (~Vasquez@84.242.142.202) |
| 14:19.03 | lehel | so?... ../modules ... make install ? |
| 14:19.32 | ctooley | lehel, cd /usr/lib/asterisk/modules and run ls -lrt. The last time you built asterisk is giong to be the last time those files were created so you should be able to tell modules that older. |
| 14:20.27 | lehel | -rw-r--r-- 1 root root 8708 2005-03-21 06:23 app_devstate.so |
| 14:20.27 | lehel | -rwxr-xr-x 1 root root 79954 2005-05-18 17:11 res_musiconhold.so |
| 14:20.34 | lehel | there is a difference |
| 14:20.49 | tzafrir | so app_devstate is from an older build |
| 14:20.54 | ctooley | is app_devstate the only one? |
| 14:21.20 | Umaro | ctooley: any other ideas? :( |
| 14:21.24 | ctooley | if not, DO NOT paste them in here, that will get you banned for flooding. |
| 14:21.24 | lehel | noo... when i'm running asterisk.. tells me first the res_odbc |
| 14:21.44 | ctooley | Umaro: not really |
| 14:21.52 | lehel | and re_odbc.so and res_config_odbc.so... are old |
| 14:21.57 | *** join/#asterisk file[class] (~jcolp@66.199.241.90) |
| 14:22.04 | lehel | how can i newer them? |
| 14:22.10 | tzafrir | lehel, so they weren't built |
| 14:22.15 | desync | hi everybody. any useful solutions for situation SIPclientA<->NAT1<->Asterisk<->NAT2<->SIPclientB? |
| 14:22.19 | ctooley | lehel, you could shut down asterisk, move /usr/lib/asterisk/modules to /tmp/asterisk-modules-backup and cd /usr/src/asterisk (or wherever asterisk is) and run make install again. |
| 14:22.21 | Umaro | ctooley: dang :( |
| 14:22.25 | tzafrir | asterisk's makefile has a bad habit of silently failing |
| 14:22.50 | ctooley | don't delete the modules in that directory though, just move them somewhere else. |
| 14:22.58 | lehel | k... |
| 14:23.41 | ctooley | desync, yep, STUN... or move the SIP clients outside of the NAT |
| 14:23.50 | ctooley | those are your only real options. |
| 14:23.59 | desync | ctooley:I need an asterisk-side solution...:> |
| 14:24.02 | desync | not client-side:> |
| 14:24.34 | lehel | ctooley: just make install?.. no make clean? |
| 14:24.36 | ctooley | desync, I think STUN would be your only hope |
| 14:24.53 | ctooley | lehel, if you run make clean it's going to build everything all over again. |
| 14:24.57 | *** join/#asterisk dgillson (~dgillson@rrcs-24-97-7-126.nys.biz.rr.com) |
| 14:25.00 | *** join/#asterisk Cheetah (~Akia@62.217.48.108) |
| 14:25.03 | lehel | ook |
| 14:25.06 | Cheetah | wow hello :D |
| 14:25.44 | mishehu | jsharp: any idea why those "other" systems can't handle the outpulse to be 8,9, or 0 on the last? sounds like a silly limitation. (just a question of curiousity, if you know the answer) |
| 14:25.50 | jsharp | I don't see why sip->nat->*->nat->sip wouldn't work if you turned nat=yes and canreinvite=no |
| 14:26.23 | ctooley | jsharp, you run into problems when either NAT has multiple SIP clients behind it. |
| 14:26.54 | jsharp | I'm not sure why that limitation exists. I've honestly never heard of it. |
| 14:27.06 | ctooley | jsharp, let me rephrase that: _I_ ran into problems in that scenario. |
| 14:27.14 | *** part/#asterisk Nuxi (~nuxi@cust-sdsl-204-250-82-202.bzn-co-i1000-01.bridgeband.net) |
| 14:27.20 | newmedian | ctooley, and what did you do to overcome the problems that you ran into in that scenario? |
| 14:27.20 | *** join/#asterisk Nuxi (~nuxi@cust-sdsl-204-250-82-202.bzn-co-i1000-01.bridgeband.net) |
| 14:27.21 | jsharp | You probably had a NAT box that wasn't doing port translation as well. |
| 14:27.37 | syle | there an example anywhere where i can just call my phone with some mp3 message |
| 14:27.42 | HA | is there an easy way to get * to do something like SayMoney(3.24) and have it read it as 3 dollars and 24 cents? |
| 14:27.59 | ctooley | newmedian, I installed an asterisk server inside the NAT and let it be an IAX peer with the asterisk server outside the NAT |
| 14:28.04 | Cheetah | I've got a stupid question: |
| 14:28.06 | syle | think from what i have read i can just copy a file to outgoing dir on the computer |
| 14:28.22 | HA | syle: use a call file that sets up the connection to an extension that plays the mp3. |
| 14:28.22 | lehel | OKAY!!!!! ladies and gentlem my asterisk is WORKING!!! ;)) |
| 14:28.31 | jsharp | yay! |
| 14:28.38 | lehel | thanks to ctooley and tzafrir :) |
| 14:29.02 | ctooley | lehel, np |
| 14:29.03 | Cheetah | In our company we've got a few spare ISDN lines. I thought about using those using VoIP clients to call real phones.. is this possible with asterisk? |
| 14:29.17 | jsharp | Cheetah: US ISDN lines? |
| 14:29.22 | Cheetah | europ |
| 14:29.29 | jsharp | Can do, then. |
| 14:29.29 | Cheetah | EuroISDN |
| 14:29.30 | desync | jsharp |
| 14:29.38 | jsharp | That's me. |
| 14:29.41 | Cheetah | are there tutorials/howtos on that? |
| 14:29.49 | desync | it doesn't work, cause asterik relies on the fact, that NATs are symetric |
| 14:30.00 | desync | so nat=yes works only for symetric NATs |
| 14:30.48 | syle | you know one thing we are missing is ability to hear voicemail messages as they are talking in our voicemail boxes in realtime |
| 14:30.51 | Cheetah | jsharp, is it hard to do? |
| 14:30.52 | newmedian | Cheetah, you may want to check out http://www.junghanns.net/asterisk/page17.html |
| 14:31.01 | ariel_ | is there a setting to turn on qos for the polycom phones switch? |
| 14:31.06 | jsharp | Yeah, what newmedian said...that's where I was going. |
| 14:31.15 | Cheetah | ah thanks :D |
| 14:31.23 | desync | so, is there a way to force media (RTP) to be passed alwasy by the asterisk |
| 14:31.42 | syle | #DEFINE symentric nat |
| 14:31.46 | jsharp | desync: I'm running several SIP phones behind an asymmetric NetBSD ipf nat box and it all works just happily for me. |
| 14:31.48 | Cheetah | oh. i forgot to mention that I have an active Fritz! ISDN card in that box |
| 14:31.58 | ManxPower | desync, "canreinvite=no" on the peer/user/friend sections of sip.conf |
| 14:31.59 | Cheetah | i read that it would be compatible with asterisk |
| 14:32.29 | desync | syle:symteric NAT is NAT, where protocol:port on some host behind NAT is strictly the same on the NAT |
| 14:32.37 | desync | e.g. if host A is behind NAT |
| 14:32.46 | desync | when host A opens UDP:X |
| 14:32.46 | newmedian | Cheetah, a http://www.voip-info.org/tiki-index.php?page=AVM%20FritzBoxFon ? |
| 14:32.58 | desync | same UDP on port X will be opened on the NAT |
| 14:33.06 | desync | erm, I mean used |
| 14:33.08 | desync | :> |
| 14:33.18 | *** join/#asterisk wasim (~wasim@203.81.201.78) |
| 14:33.31 | newmedian | hmm, no. |
| 14:33.34 | desync | ManxPower:I should check this |
| 14:33.34 | Cheetah | no, basically just a simple ISDN controller that can do voice and fax |
| 14:33.40 | ManxPower | desync, that's not what you asked, however. |
| 14:33.54 | Cheetah | it works with hylafax and other voice based apps |
| 14:33.55 | ManxPower | you asked about RTP always passing thru Asterisk. |
| 14:34.05 | *** join/#asterisk Silik0n (~krice@newrso.suspicious.org) |
| 14:34.41 | newmedian | Cheetah: http://www.voip-info.org/tiki-index.php?page=Asterisk%20CAPI%20Channels |
| 14:34.43 | desync | Manx:yup...cause I think this is the only decision for both clients behind NAT |
| 14:34.56 | desync | to use asterisk as somekind of RTP (media) proxy |
| 14:35.27 | Cheetah | ah tahnks |
| 14:35.31 | Cheetah | that was the missing info ;) |
| 14:35.39 | desync | e.g. to establish media "tunnels" between both sides behind two different NATs |
| 14:35.45 | newmedian | Cheetah: np |
| 14:36.01 | jsharp | if you set canreinvite=no, then yes, all RTP and signalling passes through *. |
| 14:36.16 | desync | jsharp:thanx I'll try |
| 14:37.41 | jsharp | And if you have a symmetric NAT, the worst you'll have to do is set the "local sip port" on each phone to something unique. |
| 14:38.26 | desync | I was trying to say, that asterisk connection tracking algo relies on the fact, that NAT clients are always behind symteric NAT |
| 14:38.30 | *** join/#asterisk makhtar (~ageller@mail.bulletinnews.com) |
| 14:38.59 | jsharp | I don't believe so...I may be on crack, though. |
| 14:39.00 | desync | in fact, even that way, it's not working correctly |
| 14:40.43 | jsharp | I can do a "sip show peers" on a box here and it'll show SIP clients registered at the NAT IP address with port 11300 or something like that....not port 5060.\ |
| 14:41.21 | *** join/#asterisk Zeeek (~Zeeek@Zeeek.sustaining.supporter.pdpc) |
| 14:42.36 | *** join/#asterisk lehel (~lehel@82.79.20.17) |
| 14:42.46 | morris | i am confused about adding sip users? |
| 14:42.50 | desync | guys thanx a lot...that works:> |
| 14:43.02 | desync | of course after some patching in channel_sip.c |
| 14:43.04 | desync | ;> |
| 14:43.07 | morris | in xlite it asks for username and auth. user |
| 14:43.16 | morris | where is that defined in asterisk |
| 14:43.23 | jsharp | sip.conf |
| 14:43.29 | lehel | how can i make a backup of all my asterisk configuration? |
| 14:43.33 | Sato1 | does some one has problems with trunk and jitter on a iax2 channel? |
| 14:43.54 | jsharp | backup asterisk config? tar up /etc/asterisk |
| 14:44.11 | Zeeek | don't confuse authorisation user with password |
| 14:44.19 | lehel | that's all jsharp? |
| 14:44.21 | Sato1 | lehel, tar -cjvf /tmp/asterisk-backup.tar.bz2 /etc/asterisk |
| 14:44.33 | lehel | eventually the zaptel.conf.. from /etc/ |
| 14:44.37 | lehel | ? |
| 14:44.38 | lehel | ok |
| 14:45.28 | morris | Zeeek, *shruggs* |
| 14:46.21 | Sato1 | lehel, you can move zaptel.conf to /etc/asterisk, then make a link back to /etc |
| 14:47.00 | dgillson | is someone available to recommend hardware config for multi location install ? |
| 14:47.07 | lehel | done!... thanks |
| 14:47.23 | *** join/#asterisk jpmcallister (~jpmcallis@kapla.escelsa.com.br) |
| 14:47.27 | wasim | lehel: or use ztcfg -c /etc/asterisk/zaptel.conf |
| 14:47.36 | Zeeek | morris is you X-Lite behind NAT? |
| 14:48.18 | _omer | morris: you can use xlite without password.. |
| 14:48.18 | jpmcallister | What is the hardware used to distribute asterisk extensions to analog telephones? |
| 14:48.23 | morris | zeek yes, but not to the asterisk box.. currently while i have it as a testbed i have it on the same lan |
| 14:48.48 | Zeeek | well I'll post a working config for X-Lite on the asterisk side if you need it |
| 14:48.50 | wasim | jpmcallister: tdm fxs cards, or te cards with channel banks |
| 14:48.55 | jsharp | jpmcallister: Either a TDM400P card with FXS ports for 4 ports per card, or a channel bank. |
| 14:48.59 | Zeeek | hi wasim |
| 14:49.02 | morris | that would be greatly appreciated |
| 14:49.12 | jsharp | er, channelbank + TE110P T1 card. |
| 14:49.22 | morris | _omer, thanks |
| 14:49.24 | Sato1 | jpmcallister, tdm |
| 14:49.26 | Sato1 | i think i m lagged |
| 14:49.39 | _omer | np morris |
| 14:49.53 | wasim | re Zeeek |
| 14:49.58 | *** join/#asterisk jsolares (~jsolares@200.30.141.86) |
| 14:50.02 | Zeeek | moris: http://pastebin.ca/12051 |
| 14:50.26 | Zeeek | obviously set nat=yes when nec. |
| 14:50.42 | Zeeek | wasim you've been scare around these parts for ages! |
| 14:50.45 | Zeeek | scarce |
| 14:51.08 | wasim | Zeeek: oui ... life is hard |
| 14:51.21 | Zeeek | I believe you knowing what I know |
| 14:51.39 | newmedian | Didn't bkw mention that X-Lite pads speex, around 8pm EST last night? |
| 14:52.19 | lehel | it is possible if i leave a voice-message (voicemail.conf <configured) to re-listen from the phone? |
| 14:53.21 | _omer | how to get agent's INUSE time? |
| 14:53.21 | *** join/#asterisk johnnyb (~johnnyb@sdsl-38-17-139.tulsaconnect.com) |
| 14:54.25 | jpmcallister | jsharp: I have a about 200 extensions. So I think the best option is channel bank. Who made that kind of hardware? |
| 14:55.03 | lehel | for example i call 2000... the voicemail is configured at 5000 |
| 14:55.14 | morris | *CLI> May 18 15:53:55 NOTICE[11740]: chan_sip.c:7698 handle_request: Registration from '"Morris Webb (Adlestone)" <sip:2000@192.168.1.2>' failed for '192.168.1.4' |
| 14:55.15 | jpmcallister | never mind. I just found in voip-info. tanks |
| 14:55.22 | morris | poopy |
| 14:55.45 | Zeeek | morris pastebin your config |
| 14:55.58 | *** join/#asterisk Jedirl (~fdsafasdf@213.162.200.226) |
| 14:55.59 | Jedirl | Hello |
| 14:56.05 | Chuji | newmedian : http://ibot.rikers.org/%23asterisk/ |
| 14:56.16 | Jedirl | I'm developing a FastAGI application with Java |
| 14:56.18 | lehel | so.. it isn't possible... ?? |
| 14:56.24 | vaewyn | jpmcallister: Check out ebay for cheap channel banks... adit 600 and such |
| 14:56.32 | Jedirl | I'd like to know how to invoke a "SetLanguage" from my AGI controller |
| 14:56.33 | morris | ok |
| 14:56.37 | *** join/#asterisk hypa7ia (~leigh@696d7ee2fbb8a2fe.session.tor) |
| 14:57.00 | newmedian | Chuji? |
| 14:57.14 | ComedianMale | :} |
| 14:57.23 | Jedirl | I don't know what SetLanguage does... if it is an application which I must "execute", or if it is a variable that I must set |
| 14:57.31 | Jedirl | any hint? |
| 14:57.34 | Chuji | newmember : You asked if bkw mentioned it... There is the log |
| 14:57.42 | Chuji | ~botlog |
| 14:57.53 | Zeeek | nice one vaewyn |
| 14:57.56 | newmedian | Chuji: ah. thanks. |
| 14:58.15 | Chuji | jbot: botlog is Can be found at http://ibot.rikers.org/ |
| 14:58.16 | jbot | Chuji: okay |
| 14:58.23 | Chuji | ~botlog |
| 14:58.24 | jbot | well, botlog is Can be found at http://ibot.rikers.org/ |
| 14:58.31 | vaewyn | Zeeek: thought of it... and just had to :P |
| 14:58.36 | *** join/#asterisk inspired (mikael@213.197.167.61) |
| 14:59.11 | Chuji | jbot forget botlog |
| 14:59.11 | jbot | Chuji: i forgot botlog |
| 14:59.18 | vaewyn | I do love it how most phones the DTMF cuts out the 'p' in password.... |
| 14:59.36 | Chuji | jbot botlog is a log of all conversations that can be found at http://ibot.rikers.org/ |
| 14:59.37 | jbot | Chuji: okay |
| 14:59.42 | vaewyn | 'mailbox?' *bleep bleep bloop* 'assword?' |
| 14:59.44 | vaewyn | hehehe |
| 14:59.53 | Cheetah | assword? ;D |
| 15:00.01 | vaewyn | yep... exactly |
| 15:00.15 | drumkilla | we should add a little wait in there |
| 15:00.15 | Cheetah | "please enter your assword" ... "poop" ... "access granted" |
| 15:00.20 | Cheetah | ROFL this so funny |
| 15:00.28 | file[class] | hellllllllo folks |
| 15:00.28 | Cheetah | AIR |
| 15:01.01 | vaewyn | drumkilla: I've got a patch for that ;P had to do it locally because of a prudish norhell lady |
| 15:01.08 | *** join/#asterisk kapejod (~kapejod@p54BCE3D8.dip.t-dialin.net) |
| 15:01.47 | morris | zeek : http://pastebin.ca/12052 |
| 15:01.50 | newmedian | Chuji what things get replaced with PROTECTED ? |
| 15:01.54 | morris | sorry it took so long |
| 15:02.13 | vaewyn | I think it was like a whole usleep(250) or something ;P |
| 15:02.40 | Zeeek | morris that isn't it at all |
| 15:02.58 | Zeeek | your login is related to sip.conf as someone said above |
| 15:03.06 | newmedian | hello file et. al |
| 15:03.10 | Zeeek | and just the few relevant lines wil doo :) |
| 15:03.11 | vaewyn | drumkilla: life treating you well? |
| 15:03.19 | Cheetah | is there a way to hook up user authentication to a mysql database? |
| 15:03.21 | Cheetah | SIP |
| 15:03.29 | vaewyn | realtime |
| 15:03.34 | morris | thanks |
| 15:03.36 | morris | i hadnt noticed |
| 15:03.41 | morris | its hard following chat |
| 15:03.42 | Zeeek | or reel thyme as it is sometimes know |
| 15:03.44 | morris | and reading the wiki |
| 15:03.57 | Zeeek | better get used to it fast |
| 15:04.03 | morris | lol |
| 15:04.08 | vaewyn | I would recommend against it though... mysql == another point of failure |
| 15:04.14 | Zeeek | agree vaewyn |
| 15:04.38 | *** join/#asterisk jeremywhiting (~jeremy@70-56-99-134.slkc.qwest.net) |
| 15:04.39 | Cheetah | vaewyn, well we've got many many clients and don't want to have to edit a text file every time something changes |
| 15:05.05 | vaewyn | Cheetah: then edit the db and run a script to drop out a new sip.conf based on that... then reload |
| 15:05.11 | Zeeek | Cheetah take a bake approach and generate static files from whatever db you like |
| 15:05.23 | Zeeek | "great minds run on like channels" |
| 15:05.28 | *** join/#asterisk W1thdraw (~Withdraw@lund-juri-sr0-vl101-012.lundsstadsnat.net) |
| 15:05.30 | Cheetah | but there is no direct interface, right? |
| 15:05.42 | vaewyn | realtime is direct |
| 15:06.04 | Cheetah | i mean that asterisk connects to the database and receives the data from there |
| 15:06.12 | vaewyn | yep |
| 15:06.19 | vaewyn | is just risky |
| 15:06.26 | Cheetah | ah, that was my question ^^ |
| 15:06.29 | morris | Zeeek, Quite clearly it was a simple case of.. IM A FUCKWIT.. gr.. its fine now its added to the correct config ;p thanks |
| 15:06.33 | Cheetah | i didnt mean to use it :D |
| 15:06.36 | vaewyn | or 'riskier' I should say |
| 15:06.40 | wasim | you could also just add extension in the *dialplan and then save it, thats direct |
| 15:07.04 | Zeeek | ok morris, you moved up a rung ion the endless progression |
| 15:07.10 | file[class] | away I go soon yessssss |
| 15:07.31 | morris | lol gd gd ;p |
| 15:07.39 | vaewyn | So... lets see... one step up is... nimrod? |
| 15:07.51 | Zeeek | 42 |
| 15:07.53 | vaewyn | hey.. same level as me! |
| 15:07.56 | newmedian | tricksy hobbits... wants to steal our precious Asterissskss |
| 15:07.58 | file[class] | yay 42 |
| 15:08.17 | vaewyn | what is six times sev... errr what was that again? |
| 15:08.27 | vaewyn | :} |
| 15:08.49 | Nuxi | Nimrod was a mighty hunter. |
| 15:09.01 | Zeeek | hey Nuxi I been waiting to tell you |
| 15:09.22 | Zeeek | SIZE^H^H^H nat=yes in general : it DOES MATTER |
| 15:09.28 | newmedian | But Emilio Estevez was a Mighty Duck |
| 15:09.50 | Nuxi | Zeeek, for both peers and users? |
| 15:10.20 | Zeeek | Nuxi last night our server changed ip. When I woke up (yes, after coffee) I rebooted the phone (it does DNS on boot only) and guess what? |
| 15:10.27 | Zeeek | NO AUDIO!!!!! |
| 15:10.42 | Zeeek | I uncommented nat=yes and restarted and guess what? |
| 15:10.53 | Zeeek | back to normal |
| 15:11.01 | vaewyn | 'Hello, my name is Inigo Montoya. You killed my father: prepare to die.' |
| 15:11.02 | Zeeek | whatever it does, it does something! |
| 15:11.14 | Zeeek | what *is* that aboput? seen it in NANAE |
| 15:11.15 | file[class] | and away I go to the physics lab for lunch |
| 15:11.16 | file[class] | bbl |
| 15:11.28 | *** join/#asterisk mike^^ (~mike@ip24-252-68-144.no.no.cox.net) |
| 15:11.31 | Nuxi | I think there are some odd subtleties in this because "sip show peers" does not indicate nat, but "sip show channel" does. |
| 15:11.54 | Zeeek | I know and it didn't matter until a server ip change (+ sip reload) |
| 15:11.56 | Nuxi | In the source, for a peer, the general setting is not copied to the structure on creation, but for a user it is. |
| 15:12.01 | newmedian | By the way, I did mention previously about Windows doing negative DNS caching; If a Soft Phone on Windows does a lookup and gets a negative cache, that could also cause problems. |
| 15:12.06 | Zeeek | ahhhhhhhh this was a user |
| 15:12.26 | Zeeek | friend even |
| 15:12.49 | Zeeek | "friend, good!" |
| 15:13.01 | Nuxi | yet the general setting is used when a call is made. |
| 15:13.13 | Zeeek | but if it works.... |
| 15:13.21 | Nuxi | But I don't know enough to declare it as such. |
| 15:13.41 | Zeeek | I just wonder if this isn't part of the answer to all the "no audio behind nat" questions? |
| 15:13.58 | Zeeek | since the general wisdom has stopped showing nat= in general |
| 15:14.10 | Zeeek | maybe there's something to it? |
| 15:14.29 | Nuxi | I suppose I should go over to dev and see if anybody there (that is knowlegeable) cares. |
| 15:14.42 | Zeeek | no, don't wake those huys yet! |
| 15:14.45 | Nuxi | It'd be a one-liner to fix. |
| 15:14.55 | Zeeek | but it'll break my box! |
| 15:15.19 | Zeeek | did I mention I have nat=yes in each entry as well? |
| 15:15.32 | *** part/#asterisk lehel (~lehel@82.79.20.17) |
| 15:15.54 | Nuxi | Also of interest is that what happens when you say nat=some_made_up_word is differnent for users and peers. user default to nating, where as peers default to not nating. |
| 15:16.15 | Nuxi | So, nat=yup works for users, but not for peers. |
| 15:16.25 | Zeeek | I haven't left defaults anywhere (except where there is no nat come tot hink) |
| 15:16.49 | hypa7ia | does anyone know if it's possible to do asterisk -realtime with an off-box db? |
| 15:16.50 | *** join/#asterisk m0f0x (m0f0x@m0f0x.user) |
| 15:16.58 | *** join/#asterisk heath__ (~dudes_@12-215-34-84.client.mchsi.com) |
| 15:17.22 | wasim | hypa7ia: i'd imagine so |
| 15:17.23 | Zeeek | sounds like a recipe for disaster |
| 15:17.32 | vaewyn | hypa7ia: yes... but then 2 points of failure :} mysql and network |
| 15:17.42 | hypa7ia | not using mysql :-) |
| 15:17.43 | Zeeek | and a chance to hang the big one |
| 15:18.01 | vaewyn | ok... DB of choice... and network |
| 15:18.17 | *** join/#asterisk lehel (~lehel@82.79.20.17) |
| 15:18.18 | Nuxi | sqlite and network? no |
| 15:18.22 | mike^^ | anyone from nufone here? I was sent an e-mai lsaying I have negative balanc e but i paypal'd $30 and my balance shows $5 (DID+$5) |
| 15:18.25 | hypa7ia | lol, no |
| 15:18.28 | lehel | hi |
| 15:18.29 | hypa7ia | postregql |
| 15:18.49 | hypa7ia | except, being able to spell and all :-) |
| 15:18.51 | lehel | "../misc/wcfxs.o: insmod wcfxs failed" |
| 15:18.52 | lehel | why? |
| 15:18.53 | vaewyn | Nuxi: hehehe... you can do sqlite on the network.... look at sqlrelay :} |
| 15:19.04 | Nuxi | ^H^H^H^H^H^H^H^H^H unless you use NFS. or sqlrelay. or something. |
| 15:19.06 | vaewyn | Nuxi: makes me laugh every time I see it |
| 15:19.28 | onkeltimm | you are a bunch. i look away for five minutes and you start making jokes in base 13 and honoring the worlds greatest swordfighter... |
| 15:20.03 | vaewyn | This channel is the home of the RoUCs 'Repositories of Useless Crap' :} |
| 15:20.14 | tzanger | heh |
| 15:20.21 | onkeltimm | oh, and i thought that was my company ;) |
| 15:22.23 | lehel | why wctdm?? why not wcfxs? |
| 15:22.31 | onkeltimm | how do you pronounce RoUCs? something like "rogues"? |
| 15:24.40 | hypa7ia | reminds me of ROUS from the princess bride |
| 15:24.47 | hypa7ia | rodents of unusual size |
| 15:26.11 | onkeltimm | hypa7ia: ah, these. |
| 15:27.56 | [TK]D-Fender | hypa7ia : But I don't believe they exist..... |
| 15:28.22 | [TK]D-Fender | "asssssss yooouuuuu wwwiiiisshhh! *thud*" |
| 15:29.30 | hypa7ia | that movie rocks so much :-) |
| 15:29.42 | [TK]D-Fender | It does. |
| 15:30.06 | [TK]D-Fender | Anbd good seeing Cary Elwes in "Ella Enchanted" too... very reminiscent |
| 15:31.11 | blint | bye everyone |
| 15:31.15 | *** part/#asterisk blint (~blint@adsl-669.mirage.euroweb.hu) |
| 15:31.54 | lehel | why not wcfxs, wcfxo?.. why wctdm?.. from when? |
| 15:32.37 | *** join/#asterisk CaNaBiS (~canabis@pcp0011584253pcs.rthfrd01.tn.comcast.net) |
| 15:33.41 | CaNaBiS | a buddy of mine is going to give me a ip phone to use against his Cisco call manager, but I have to be able to register to his switch via IPSEC, whats the best way to handle this? |
| 15:36.59 | onkeltimm | nobody outside the us seems to know the princess bride... and i was a fantasy-crazed kid when it ran in europe... i am so glad that i did a google on enigo when he appeared in the /. poll... this movie's a gem. |
| 15:37.46 | [TK]D-Fender | Well I'm in Canada, so its not just the US... |
| 15:38.18 | CaNaBiS | I know princess bride |
| 15:41.55 | hypa7ia | CaNaBiS, that's not really an asterisk question is it :-) |
| 15:42.07 | hypa7ia | CaNaBiS, you'd be better off asking on the cisco netpro forums |
| 15:43.21 | onkeltimm | hmmm... "Die Braut des Prinzen" ... perhaps i missed it because the translation sucked big time ;) |
| 15:43.49 | *** join/#asterisk km- (~pgrace@brdgw1.rttx.com) |
| 15:43.53 | CaNaBiS | didnt know if you all knew if the Cisco phones could do ipsec connections |
| 15:43.59 | *** join/#asterisk Blackvel (~blackvel@dsl-084-057-126-093.arcor-ip.net) |
| 15:44.24 | km- | hey, does anyone remember what global technology solution's webpage is? it was something like gtsinfo.biz |
| 15:44.34 | km- | ah shoot |
| 15:44.38 | km- | I remembered it |
| 15:44.38 | Qwell | ~google global technology solution |
| 15:44.38 | km- | hehe |
| 15:44.42 | km- | I tried that |
| 15:44.53 | km- | but thankfully |
| 15:44.57 | km- | I remembered it as I said it |
| 15:44.57 | km- | haha |
| 15:45.08 | Nuxi | well, don't keep us in suspense |
| 15:45.11 | km- | anyone else here selling 7960's want to competitively bid |
| 15:45.14 | km- | gtsinc.biz |
| 15:45.22 | shaZwaz | ok guys have fun |
| 15:45.28 | shaZwaz | bye |
| 15:45.43 | *** join/#asterisk W1thdraw (~Withdraw@lund-juri-sr0-vl101-012.lundsstadsnat.net) |
| 15:46.50 | shido6 | km- ? |
| 15:47.38 | km- | shido6: Jer had told me to go to gts for 7960's at one point |
| 15:47.45 | km- | shido6: so thats who I'm going to |
| 15:47.58 | km- | but I was wondering who else might have a budding asterisk business here who'd like to make some money |
| 15:48.05 | shido6 | what? |
| 15:48.25 | *** join/#asterisk cmk (~cmk_@p54A3D31F.dip.t-dialin.net) |
| 15:48.28 | km- | heh, I need one by monday |
| 15:48.33 | syle | same |
| 15:48.41 | Qwell | its already wednesday :D |
| 15:49.02 | shido6 | I have a few |
| 15:49.14 | syle | i;m having a problem locating 7960's at a good deal in canada |
| 15:49.25 | shido6 | I am in canada |
| 15:49.33 | syle | no shit |
| 15:49.36 | syle | PM |
| 15:49.40 | Qwell | shido6: Where about? |
| 15:49.44 | laotan | canucks'R'us |
| 15:49.56 | shido6 | never said I was canadian :) |
| 15:55.44 | CaNaBiS | anyone happen to know if a Sipura will connect up to a Cisco Call Manager |
| 15:56.17 | syle | isn;t cisco call manager a phone that hasn;t been converted to SIP? |
| 15:56.58 | vaewyn | no... call manager is Ciscos idea of a softswitch |
| 15:57.02 | ariel_ | CaNaBiS, call manager is skinnyc SCCP not sip unless you add it on to them. |
| 15:57.37 | CaNaBiS | so if I wanna use an ATA with his switch, then I need to get a Cisco ata? |
| 15:58.58 | ariel_ | CaNaBiS, no you need to add the sip modules there ata is sip as well. |
| 15:59.10 | *** join/#asterisk outtolunc (outtolunc@adsl-66-218-53-170.dslextreme.com) |
| 15:59.35 | CaNaBiS | is this a paid module? |
| 15:59.39 | CaNaBiS | or just something he can add? |
| 15:59.57 | ariel_ | cisco everything is paid |
| 16:00.23 | ariel_ | unless you got it with the unit when you purchesed it. |
| 16:00.36 | CaNaBiS | he's looking it right now |
| 16:00.51 | CaNaBiS | he says he sees that he can add a sip trunk |
| 16:01.16 | nestAr | hrmm.. I was thinking about doing a small office with a TDM card, but they don't want to have to replace their phones, so I'm thinking I'll just get 4 SPA-3000's, that'll take care of the 4 incoming lines into * and 4 office phones.. |
| 16:01.27 | *** part/#asterisk nazgool (~nazgoool@gatekeeper-e0.twc.de) |
| 16:01.29 | nestAr | anyone see a potential problem with that? |
| 16:02.05 | ariel_ | CaNaBiS, depending on the Cisco unit there is a trunk sip and an local sip addon. But I have not worked with them in over a year now. |
| 16:02.21 | CaNaBiS | heh, he's reading in his Cisco help, it reads how to get it to connect with Asterisk |
| 16:03.01 | ariel_ | nestAr, I have setup some soho's with TDM board on the server and sipura for some of the extensions. |
| 16:03.29 | newmember | Why would I want to buy a T1 card with MGCP capability? |
| 16:03.53 | *** join/#asterisk Himeko (~himeko@S01060040ca128fc3.ed.shawcable.net) |
| 16:04.07 | ariel_ | newmember, don't know but what card are you talking about? |
| 16:04.12 | _omer | Cisco Call Manager doesnt support SIP... |
| 16:04.38 | Juggie | doesnt it support sip trunks |
| 16:05.12 | newmember | ariel_: it more of a gateway option AudioCodes Mediant 2000 Single T1 (MGCP) |
| 16:05.17 | nestAr | ariel_: well, i was just looking for a best way to keep the cost down.. if i was to do a TDM card for incoming lines, then they'd need either another TDM card, or new IP phones... |
| 16:05.30 | CaNaBiS | it must, he sees it in his manager |
| 16:05.31 | nestAr | ~$650 in TDM cards |
| 16:05.45 | nestAr | vs $400 for 4 spa-3000's |
| 16:05.52 | ariel_ | nestAr, it depends on your setup. |
| 16:06.01 | CaNaBiS | and the Cisco help file tells how to convert the ip phone to sip and back |
| 16:06.29 | *** part/#asterisk jwitte (~jwitte_@port-212-202-101-206.static.qsc.de) |
| 16:06.53 | nestAr | the only real problem i see is with dialing out.. it might be a little hectic finding a free FXO to dialout with SPA's.. but i think a checkgroup/setgroup could do it.. |
| 16:06.58 | ariel_ | It's lunch time..... |
| 16:07.15 | nestAr | indeed |
| 16:07.36 | ariel_ | nestAr, I like the tdm for dialing out and turnks you can just set them up as dial,zap/g1 |
| 16:08.11 | morris | [2000] |
| 16:08.11 | morris | type=friend ; takes and makes calls |
| 16:08.11 | morris | username=2000 ; Username on device |
| 16:08.11 | morris | secret=tikka |
| 16:08.11 | morris | nat=no ;change sometime |
| 16:08.12 | morris | disallow=all |
| 16:08.16 | morris | allow=gsm |
| 16:08.18 | morris | allow=ilbc |
| 16:08.20 | morris | allow=ulaw |
| 16:08.24 | morris | host=dynamic |
| 16:08.26 | morris | qualify=500 |
| 16:08.28 | morris | context=privileged |
| 16:08.30 | morris | mailbox=2000@sip |
| 16:08.32 | morris | canreinvite=no |
| 16:08.34 | morris | dtmfmode=rfc2833 |
| 16:08.36 | morris | shit sorry |
| 16:08.38 | morris | May 18 17:07:57 NOTICE[13094]: chan_local.c:378 local_alloc: No such extension/context 2000@default creating local channel |
| 16:08.39 | nestAr | ariel_: yep, i do too |
| 16:08.41 | morris | was what i ment to paste and ask you about ;/ |
| 16:08.45 | *** join/#asterisk jakepdev (HydraIRC@ppp28.pm3-2.phi-pt.pa.localnet.com) |
| 16:08.46 | nestAr | morris: lol |
| 16:08.48 | nestAr | it happens |
| 16:08.53 | nestAr | IBpastebincomments |
| 16:08.56 | morris | thanks lol |
| 16:09.26 | ariel_ | morris, password hummmm |
| 16:09.52 | morris | *hides* |
| 16:09.53 | morris | lol |
| 16:09.57 | outtolunc | on the end of allow=gsm |
| 16:10.00 | jakepdev | anyone ever see a linux server crash with no entries in the kernel logs? |
| 16:10.22 | vaewyn | jakepdev: Yeah... but someone pulled the power cord ;P |
| 16:10.26 | morris | lol |
| 16:10.33 | jakepdev | that is my suspicision also |
| 16:10.49 | jakepdev | otherwise I figured it should log |
| 16:11.43 | jakepdev | same thing with asterisk - if I have full debugging on in logger.conf - should show something in the logs relating to the crash? |
| 16:12.11 | jakepdev | last thing i see is a DTMF digit |
| 16:12.58 | morris | hoooorayyyyyyyyyy |
| 16:13.09 | morris | ive actually managed to do something constructive with asterisk |
| 16:13.10 | morris | weee |
| 16:13.41 | Zeeek | now let us pray... |
| 16:14.00 | kapejod | jakepdev: if you have bad hw the box might crash without anything in the logs |
| 16:14.03 | morris | lol |
| 16:14.26 | jakepdev | thanks kapejod |
| 16:14.59 | Jedirl | Anyone developing with Java Asterisk 's FastAGI? |
| 16:15.04 | Zeeek | shido6 what is the 200 problem? |
| 16:15.23 | Zeeek | 2000 I mean |
| 16:15.24 | jakepdev | Jedirl - we're using FastAGI, but not in Jaca |
| 16:16.17 | morris | Thank you everyone for support, without you this would not of been possible.. ;) |
| 16:16.38 | vaewyn | "I'd like to thank the academy...' |
| 16:16.40 | vaewyn | hehehe |
| 16:16.58 | Zeeek | ok guys, cut it out: http://www.lavinia.es/images/CORP_izqu.gif |
| 16:16.59 | Jedirl | jakepdev: which implementation then? |
| 16:17.20 | jakepdev | just using it with a .NET client - i got it to work with VB6 also |
| 16:17.31 | jakepdev | not very difficult at all |
| 16:17.35 | Jedirl | :) |
| 16:17.38 | Jedirl | how does it scale? |
| 16:17.55 | jakepdev | we're running it on 60 ports with no problems now on a Dual Xeon machine |
| 16:18.18 | *** join/#asterisk Gh0sty (~Ghosty@ip-81-11-208-137.dsl.scarlet.be) |
| 16:18.43 | _omer | does Asterisk support Dual Processor? |
| 16:18.45 | Jedirl | jakepdev: have you tried any way of 'failover'? |
| 16:19.10 | jakepdev | I've requested a patch and JunkY has been working on that |
| 16:19.19 | vaewyn | _omer: yes.... duals... quads.... hexes... octs... :} |
| 16:19.26 | jakepdev | his patch works only in HEAD |
| 16:19.29 | Jedirl | :( |
| 16:19.35 | Jedirl | I'm thinking on developing a simple local C proxy |
| 16:19.50 | vaewyn | me drools over a quad dual core opterson board |
| 16:19.50 | Jedirl | which tests for availability of FastAGI servers and redirects connections to one or another |
| 16:19.52 | _omer | WOW!!!!! Asterisk supports that havent been made yet... ;) |
| 16:20.09 | jakepdev | Jedirl - it can - yes |
| 16:20.17 | vaewyn | _omer: they are to... dual core quad opteron... is an oct |
| 16:20.43 | jakepdev | Jedirl - the patch is in Mantis |
| 16:20.51 | Jedirl | what is mantis? |
| 16:20.59 | jakepdev | the bug tracker |
| 16:21.00 | vaewyn | bugs.digium.com |
| 16:21.50 | Jedirl | ok :D |
| 16:22.55 | Cheetah | hello |
| 16:23.08 | Cheetah | is there a way to test a capi call from the CLI of asterisk? |
| 16:23.13 | Cheetah | like "capi info" |
| 16:23.14 | Cheetah | ? |
| 16:25.59 | jakepdev | Cheetah - don't know of one, but couldn't you initiate the call from another device like an IAX or SIP soft phone? |
| 16:26.26 | Cheetah | thats the second think i'm trying to figure out -- how to allow softphones to use SIP to initiate the call :) |
| 16:26.37 | jakepdev | ok |
| 16:26.43 | *** join/#asterisk Grooby (~Grooby@12.22.232.212) |
| 16:27.07 | tzanger | interesting |
| 16:27.13 | jakepdev | do you have your entry in sip.conf already that pertains to your softphone? |
| 16:27.23 | tzanger | May 18 12:24:45 NOTICE[2679]: chan_sip.c:8284 handle_response: Failed to authenticate on REGISTER to '<sip:username@sip.unlimitel.ca>;tag=as70ca57f9' |
| 16:27.24 | Cheetah | no, its all default (just installed it) |
| 16:27.26 | tzanger | May 18 12:24:45 WARNING[2679]: chan_sip.c:8268 handle_response: Got 200 OK on REGISTER that isn't a register |
| 16:27.38 | tzanger | if it failed to authenticate why'd I get a 200 OK back? heh |
| 16:27.57 | Zeeek | I love that message |
| 16:28.15 | jakepdev | Cheetah - download SJPhone |
| 16:28.23 | *** join/#asterisk MikeJ[Laptop] (~ircatjerr@pcp02795302pcs.roylok01.mi.comcast.net) |
| 16:28.24 | Cheetah | jakepdev, already did that ;) |
| 16:28.37 | Cheetah | what does the sip.conf entry has to look like if I want to use CAPI as outgoing gateway to the phone net? |
| 16:28.57 | morris | For greeting/Announcements, can a WAVE file be used instead of mp3? |
| 16:29.38 | jakepdev | Cheetah - sip.conf only pertains to SIP connections, so you don't need to alter it for CAPI |
| 16:29.51 | jakepdev | morris - yes |
| 16:29.58 | morris | sweet, thanks |
| 16:30.11 | Cheetah | okay, lets assume i leave it like it is, what do I have to do to make SIP clients -> CAPI/ISDN work? |
| 16:30.45 | jakepdev | Cheetah - are you able to make test calls from SJPhone to Asterisk? |
| 16:31.17 | Cheetah | no, because I have no idea what the name/password of a registered SIP account on my server is - i assume I have to add one to the sip.conf first, right? |
| 16:31.25 | jakepdev | yes :) |
| 16:32.08 | Cheetah | well, got an example for it? ;) |
| 16:32.13 | morris | sorry to be pain, what ports do i need to forward when NAT'd? i have to leave work now but i want to continue working with asterisk from home ;/ |
| 16:32.28 | *** join/#asterisk MikeJ[Laptop] (~ircatjerr@pcp02795302pcs.roylok01.mi.comcast.net) |
| 16:32.37 | jakepdev | somewhere around here |
| 16:33.22 | jakepdev | cheetah - look here: http://www.voipuser.org/forum_topic_616.html |
| 16:34.36 | morris[away] | right im off home, thanks again guys for help. |
| 16:35.48 | Cheetah | jakepdev, lemme check that ;) |
| 16:37.18 | *** join/#asterisk NewSole (dave@i216-58-44-245.avalonworks.net) |
| 16:39.38 | *** join/#asterisk MikeJ[Laptop] (~ircatjerr@pcp02795302pcs.roylok01.mi.comcast.net) |
| 16:41.06 | Cheetah | hm |
| 16:41.50 | Cheetah | "wrong password on authentication for REGISTER for 'testuser'..." |
| 16:42.24 | jakepdev | the password is secret=????? |
| 16:43.04 | Cheetah | i just set it to "test" |
| 16:43.10 | Cheetah | this is the register line: |
| 16:43.36 | Cheetah | register => testuser:testpass@voip.xxxx.org |
| 16:43.48 | vaewyn | ok... this may sound odd... but does anyone else have "RemoveVoicemail" as an application? |
| 16:44.27 | vaewyn | I can't remember if that is one I added or not .... hehehe |
| 16:45.32 | *** join/#asterisk gtigene (~chatzilla@c-67-184-112-58.hsd1.il.comcast.net) |
| 16:45.53 | *** join/#asterisk bkw_ (~bkw_@bkw.developer.and.friend.of.asterisk) |
| 16:45.53 | *** mode/#asterisk [+o bkw_] by ChanServ |
| 16:46.01 | *** join/#asterisk shido6 (~greg@d57-87-253.home.cgocable.net) |
| 16:46.14 | *** part/#asterisk bkw_ (~bkw_@bkw.developer.and.friend.of.asterisk) |
| 16:46.37 | gtigene | How do you eavesdrop on calls in Asterisk (I am the Asterisk administrator)? |
| 16:46.54 | vaewyn | monitor... or barge commands... or... oor... oorr.. :} |
| 16:47.29 | gtigene | vaewyn: We dont want to barge in on calls, just want to hear the conversation. |
| 16:47.35 | *** join/#asterisk mutilator (~animenodv@65.111.201.79) |
| 16:48.33 | *** join/#asterisk bkw_ (~bkw_@bkw.developer.and.friend.of.asterisk) |
| 16:48.33 | *** mode/#asterisk [+o bkw_] by ChanServ |
| 16:49.13 | *** join/#asterisk devel (~devel@wiggum.digitalcoven.com) |
| 16:49.46 | *** join/#asterisk IPmonger (~ipmonger@pcp0010577106pcs.coatsv01.pa.comcast.net) |
| 16:51.26 | onkeltimm | gtigene: http://voip-info.org/wiki-Asterisk+cmd+ChanSpy |
| 16:51.47 | gtigene | onkeltimm: Thank you |
| 16:55.10 | Zeeek | vaewyn not in 1.0.6 |
| 16:56.05 | onkeltimm | vaewyn: not in head as of last week |
| 16:56.10 | onkeltimm | must be yours |
| 16:56.12 | onkeltimm | # |
| 16:57.42 | jakepdev | Cheetah - was on the phone.... You can take out the register line |
| 16:57.43 | vaewyn | Zeeek & onkeltimm: Thanks :} i think I am having a Reagan moment |
| 16:57.50 | *** join/#asterisk Twister (~jason@216.30.232.106) |
| 16:57.54 | onkeltimm | lol |
| 16:58.04 | Cheetah | jakepdev, so it still says that my PW is incorrect |
| 16:58.25 | Twister | hey all..was just wondering if its worth getting this book (http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&category=3521&item=5196784631&rd=1) or should i just get the information from voip-info |
| 16:58.36 | jakepdev | you can take the secret line out of your sip.conf file |
| 16:58.41 | blitzrage | ariel_: please turn off your auto-announce |
| 16:59.42 | Cheetah | i guess I dont have to regsiter with it, right? i just want to do outbound calls |
| 17:00.25 | MikeJ[Laptop] | Twister, that book is bad |
| 17:00.43 | onkeltimm | got great reviews at amazon, tho... |
| 17:00.45 | jakepdev | Cheetah - SJPhone will perform the register with asterisk - not vice versa |
| 17:01.01 | MikeJ[Laptop] | and older |
| 17:01.06 | Cheetah | i know, but do I need to register a client that only does calls, and doesn't receive any? |
| 17:01.26 | *** join/#asterisk PTG123 (~PTG123@ip68-106-24-139.ph.ph.cox.net) |
| 17:01.48 | blitzrage | MikeJ[Laptop]: yah :) |
| 17:02.05 | blitzrage | Twister: that book IS voip-info - but old |
| 17:02.23 | jakepdev | Cheetah - no - oncee SJPhone registers, you can make and receive calls |
| 17:02.59 | *** join/#asterisk jamesewing (~James@pbx2.jsci.net) |
| 17:03.01 | Cheetah | jakepdev, well, i can't register since it always tells me that the password is incorrect. where do I have to add the account for the client that tries to register? |
| 17:03.22 | jakepdev | Cheetah - in sip.conf - there should be a line that says sercret=???? |
| 17:03.26 | jamesewing | \users |
| 17:03.27 | jakepdev | ???? is your pwd |
| 17:03.31 | jamesewing | :) |
| 17:03.32 | jamesewing | sorry |
| 17:03.47 | Cheetah | jakepdev, yeah, i tried to modfiy a block of them to suit my needs |
| 17:03.50 | Cheetah | didn't work too well though |
| 17:04.24 | jakepdev | Cheetah - it will work without that line also |
| 17:04.38 | Cheetah | there is something like [sip_proxy-out] type=peer, secret=testpass, username=testuser, fromuser=yourusername, host=dynamic |
| 17:04.45 | *** join/#asterisk C4-Telecom (~sales@212.145.163.120) |
| 17:04.53 | C4-Telecom | hi all |
| 17:04.58 | jamesewing | hey |
| 17:05.55 | Twister | oh ok thank you blitzrage |
| 17:05.59 | newmedian | Hello C4. Does your Telecom explode? |
| 17:06.01 | Twister | i figured that was the case |
| 17:06.06 | jsharp | Blam! |
| 17:06.17 | C4-Telecom | how many procesing power is needed for * box no transcoding, 8 E1, only switching & billing ??? |
| 17:06.39 | jsharp | All TDM? Or TDM to Voip? |
| 17:06.46 | C4-Telecom | newmeridian> nope.... explosive prices |
| 17:06.47 | jakepdev | Cheetah - [sjphone] type=friend host=dynamic username=sjphone dtmfmode=inband context=incoming |
| 17:06.49 | C4-Telecom | al tdm |
| 17:07.10 | jsharp | A big, fat, single proc 3Ghz machine should handle that. |
| 17:07.44 | C4-Telecom | jsharp> RAM?? |
| 17:07.52 | jsharp | Enough to keep it from swapping. |
| 17:07.56 | jsharp | 512 or so. |
| 17:08.02 | C4-Telecom | perfect |
| 17:08.15 | C4-Telecom | thanks |
| 17:08.54 | Cheetah | jakepdev, "registration from '<sip:sjphone@192.168.0.250:5060>' failed for '192.168.0.11'" |
| 17:10.23 | HA | what is the pci voltage in a dell 750? i need to know whether to get the te410p or the te405p. |
| 17:10.40 | *** join/#asterisk NightHawke (~NightHawk@68-190-111-175.static.sprn.tx.charter.com) |
| 17:10.45 | *** join/#asterisk jief- (~jief@modemcable196.182-80-70.mc.videotron.ca) |
| 17:11.07 | *** join/#asterisk jamesewing (~James@pbx2.jsci.net) |
| 17:11.26 | NightHawke | trying to get a set of TDM cards to mount on a system |
| 17:11.38 | NightHawke | what is the command string to do that? |
| 17:11.48 | harryvv | ha, same as any other pci slot |
| 17:12.03 | NightHawke | ok what is it? |
| 17:12.05 | Twister | ok couple questions about scaling and such, Im going to implement asterisk in 3 offices, going to be doing voip phones with an isdn pri with 15 lines |
| 17:12.11 | harryvv | google pci specs |
| 17:12.15 | *** part/#asterisk IPmonger (~ipmonger@pcp0010577106pcs.coatsv01.pa.comcast.net) |
| 17:12.18 | Twister | probably 40 extensions |
| 17:12.25 | Twister | the locations are gonna share the lines |
| 17:12.39 | Twister | what kinda hardware would i need for this? |
| 17:12.44 | *** join/#asterisk cpatry (~grepmoo@65.39.228.5) |
| 17:12.53 | HA | harryvv: Is it 3.3 or 5.0? I ask because I know that some people here use them. |
| 17:13.10 | wasim | Twister: nothing much, t1 card, use alaw on the network, so no transcoding |
| 17:13.42 | Twister | alaw? (sorry im a bit of a telen00b |
| 17:13.57 | wasim | like, zero compression |
| 17:14.09 | mishehu | anybody had problems running the g729 register file? I downloaded it and gave it execute perms, but it says "cannot execute binary file" error |
| 17:14.18 | Twister | oh ok |
| 17:14.24 | Twister | so the te110p card? |
| 17:14.26 | Twister | would work |
| 17:15.22 | Twister | how powerful of a machine ya think id need, im working with a very small budget as the company is non profit |
| 17:15.40 | Jedirl | in a IVR machine with a 4-E1 card, with the only codec used being alaw, and all the speech files in alaw format, and all the business logic being done externally with FastAGI... A P-3 with 1Ghz, 1Gbyte of RAM and SCSI hard disk how much channels do you think that could handle? |
| 17:15.58 | Jedirl | at a time? |
| 17:18.45 | *** join/#asterisk hellop (~LeeHarvey@cpe-70-93-40-171.hawaii.res.rr.com) |
| 17:19.03 | hellop | Alright! I got Asterisk installed for the second time! |
| 17:19.05 | HA | harryvv: nevermind. Dell support could improve some, but at least the Linux guy sounded like he had a brain. |
| 17:19.25 | HA | jbot knows it all. |
| 17:19.26 | jbot | Of course! |
| 17:20.26 | vaewyn | jbot: who's your daddy? |
| 17:20.27 | jbot | YOU are, Mr Sexy Pants! |
| 17:20.33 | vaewyn | :} |
| 17:20.56 | mishehu | jbot: who's the most evil person in the world? |
| 17:21.56 | mishehu | "I have combined the DNA of the most evil creatures in the universe to make the most evil one ever..." 'Turns out its Man' |
| 17:22.15 | vaewyn | Bill Gates to be exact |
| 17:22.32 | Jedirl | Which text-to-speech software would you recommend for Linux? |
| 17:22.34 | vaewyn | Although... the SCO guys are really edging in on that |
| 17:22.45 | Jedirl | I'm trying cepstra and seems great but I'd like to test more TTS's |
| 17:22.47 | jakepdev | Cheetah - if you do a sip debug at the asterisk CLI - it should show you why the registration has failed |
| 17:22.57 | vaewyn | cepstra and festival |
| 17:22.59 | Cheetah | okay |
| 17:23.09 | Jedirl | festival's quality for non-english is quite bad |
| 17:23.14 | vaewyn | yep |
| 17:23.19 | Jedirl | any more TTS's? |
| 17:23.24 | jakepdev | festival's quality for english is pretty bad also |
| 17:23.50 | Jedirl | jakepdev: festival's quality for english is awesome compared to festival's quality for spanish |
| 17:23.51 | *** join/#asterisk meppl (~mephisto@p54AADD1D.dip.t-dialin.net) |
| 17:23.51 | Jedirl | :) |
| 17:24.04 | vaewyn | reminds me of a joke 'Those guys at NOAA must have tons of cash... They have Stephen Hawking doing their weather reports!' |
| 17:24.08 | Cheetah | "Looking for 192.168.0.250:5060 in default" ... "SIP/2.0 404 Not Found" |
| 17:24.22 | Jedirl | so there aren't any more available TTS's for linux? just cepstra? |
| 17:24.40 | vaewyn | hate to say it... but cepstra and festival are the best... |
| 17:24.49 | Jedirl | :( |
| 17:24.56 | vaewyn | in fact... basically the only options |
| 17:25.15 | Jedirl | then maybe I should do the TTS in a windows machine |
| 17:25.23 | mishehu | ugh. I want to register this codec before goign to lunch |
| 17:25.33 | Jedirl | which TTSs would you consider for Windows? |
| 17:25.36 | jakepdev | Cheetah - in SJPhone - use the username sjphone with no password |
| 17:25.38 | HA | festival does have multisyn voices which are an improvement over the regular festival voices. |
| 17:25.45 | *** join/#asterisk CoffeeIV (~rristroph@mail.airlinksystems.com) |
| 17:25.56 | jakepdev | Jedirl - Speechworks |
| 17:25.59 | HA | build your own tts in windows. |
| 17:25.59 | Cheetah | jakepdev, it tells me that the password field can not be left empty |
| 17:26.03 | vaewyn | HA: true... and I havn't tried those out yet... |
| 17:26.11 | jakepdev | Jedirl - SAPI |
| 17:26.19 | Jedirl | uhm |
| 17:27.08 | HA | we are using them for pre-recorded tts only at the moment. i've got some testing to do, but it may be possible to do some decent live tts using festival_client instead of text2wave or the built in * commands using multisyn voices. |
| 17:27.34 | mishehu | ugh again. konqueror must have done something to the dl of the file... |
| 17:28.02 | jakepdev | Cheetah - in SJPhone - uncheck require password in Profiles | Edit |
| 17:28.19 | *** join/#asterisk naula (~nschmidt@67.154.228.132) |
| 17:28.29 | Cheetah | jakepdev, do I have to enable STUN? |
| 17:28.32 | jakepdev | nope |
| 17:28.40 | naula | Does anyone know of a good beginners guide to get asterisk up and running? |
| 17:29.33 | Cheetah | okay, i did that |
| 17:29.43 | Cheetah | sip debug still tells me 404 not found |
| 17:29.49 | vaewyn | STUN == evil |
| 17:29.51 | vaewyn | :P |
| 17:29.54 | jakepdev | when trying to make a call? |
| 17:30.12 | jakepdev | or just registering? |
| 17:30.17 | Cheetah | naw, when the client is trying to register to the server |
| 17:31.27 | RickTick | Naula: there is an asterisk guide being sold .... on the net ... pretty informative |
| 17:31.55 | HA | if i didn't mention it before, just for future information, the PE750 has both a 3.3 and a 5.5v pci on the riser. |
| 17:32.25 | jakepdev | user name in sjphone is sjphone? |
| 17:32.59 | Cheetah | jakepdev, yeah |
| 17:34.31 | jakepdev | Cheetah - try this: [8001] type=friend host=dynamic username=8001 dtmfmode=inband context=incoming |
| 17:34.35 | *** join/#asterisk MrClean (~seabrook@store-fw.porchlight.ca) |
| 17:34.41 | jakepdev | and change the user name in SjPhone to 8001 |
| 17:35.22 | newmedian | ~docs |
| 17:35.23 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org |
| 17:35.23 | *** join/#asterisk loick (~loick@APuteaux-151-1-2-92.w82-120.abo.wanadoo.fr) |
| 17:35.55 | naula | thank you |
| 17:36.01 | Cheetah | jakepdev, now it worked.. at least the registrationo |
| 17:36.11 | MrClean | Can somebody tell me if channel variable inheritence (with prepended underscores) is broken or non-existent in asterisk ver. 1.0.7? |
| 17:36.40 | Cheetah | jakepdev, now when I try to make a call it gives me "cannot find extension context 'incoming'" |
| 17:36.41 | jakepdev | ok - now you need to add the context [incoming] in your extensions.conf file and |
| 17:36.41 | Zeeek | quick, I need a unix god |
| 17:36.55 | jakepdev | summoning god - please wait |
| 17:37.00 | newmedian | Zeeek: su - |
| 17:37.17 | jakepdev | connection falied - incomatible codec |
| 17:37.31 | Nivex | god: no such user. |
| 17:38.00 | Zeeek | how do I mv all files that do NOT have .tif at the end to *.tif ? |
| 17:38.01 | newmedian | dog loaded and ready; using codec dyslexia |
| 17:38.16 | Zeeek | I know it's simple but I never do shit like this |
| 17:38.29 | HA | what do ya need Zeeek? |
| 17:38.46 | Zeeek | find file that don't end in .tif - add the .tif |
| 17:39.17 | Zeeek | find .... -exec mv {} {}.tif - doubtful |
| 17:39.33 | jakepdev | Cheetah - you can try the following: [incoming] exten => s,1,Playback(goodbye) |
| 17:39.35 | Zeeek | fir $file in |
| 17:39.50 | jakepdev | Cheetah - that shold play the goodbye message to your softphone |
| 17:39.55 | Cheetah | okay |
| 17:39.59 | Cheetah | *cheks* |
| 17:40.47 | Zeeek | I'm already stuck trying to negate the find -name :) |
| 17:42.53 | Cheetah | jakepdev, now it tells me again 404 not found.. but not for the registration, but for the call itself |
| 17:43.00 | newmedian | You could pipe your output of an ls, e.g. ls | grep -iv ".tif" | ..... |
| 17:43.35 | jakepdev | Cheetah - [incoming needs to be on a line by itself |
| 17:43.42 | Cheetah | i know that ;) |
| 17:43.45 | jakepdev | ok |
| 17:43.47 | Cheetah | thats what I did |
| 17:44.09 | jakepdev | what are you dialing? |
| 17:44.22 | Zeeek | newmedian yeah I suppose that's be simple enuf - sounds like the MSDOS days though |
| 17:44.26 | jakepdev | on your softphone? |
| 17:44.43 | Cheetah | 1234 |
| 17:44.58 | jakepdev | ok - then use 1234,1,Playback(goodbye) |
| 17:45.20 | jakepdev | exten => 1234,1,Playback(goodbye) |
| 17:45.28 | Cheetah | still 404 |
| 17:45.38 | jakepdev | what's the 404 say? |
| 17:45.48 | Cheetah | not found :D |
| 17:45.56 | jakepdev | context not found? |
| 17:46.51 | Cheetah | lemme read the log |
| 17:47.01 | Cheetah | ah |
| 17:47.33 | Cheetah | i did the following: exten => 1234,1,Answer exten => 1234,2,Playback(goodbye) exten => 1234,3,Hangup |
| 17:47.42 | *** join/#asterisk harryvv (~leonardo@S010600a0c93f6f7e.vs.shawcable.net) |
| 17:47.55 | Cheetah | but it cant play the file, since its not found :D |
| 17:50.19 | [TK]D-Fender | Cheetah : I always give mine a fully qualified path name. |
| 17:50.33 | Cheetah | i got it to run with the tt-weasels sound ;) |
| 17:50.47 | jakepdev | Cheetah - the sound files are in /var/lib/asterisk/sounds i believe |
| 17:51.03 | Cheetah | jakepdev, now how do I make it use CAPi to call the world? ;) |
| 17:51.39 | jakepdev | make sure you can get audio through first before you start bridging |
| 17:51.49 | Cheetah | yeah, i can hear the sound now :) |
| 17:51.53 | jakepdev | ok |
| 17:51.57 | Cheetah | "weasels have eaten your phone system" |
| 17:52.01 | jakepdev | cool |
| 17:52.02 | Zeeek | that was an irritating waste of time! I never expected a Windows user to rename graphics files dropping the extensions |
| 17:52.07 | *** join/#asterisk Moc[Toronto] (~mochouina@64.235.210.66) |
| 17:52.37 | jakepdev | Cheetah - now instead of Playback, use the Dial cmd with your CAPI parameters |
| 17:52.50 | Cheetah | okay, where do I get info on that? |
| 17:53.37 | jakepdev | http://www.voip-info.org/wiki-Asterisk+How+to+connect+with+CAPI |
| 17:53.54 | harryvv | What dial application can be used to say..dail 50 phone numbers and play a message? example would be if a company got hit by a virus and wanted to leave a vm to everyone to watch out for this such and such virus? |
| 17:54.07 | Cheetah | thanks! |
| 17:54.35 | jakepdev | actually - looks like there may be some notes in the somments on that |
| 17:54.35 | vaewyn | harryvv: Well... if they are in a * voicemail system then just cp the voicemail into each box :} |
| 17:54.49 | vaewyn | harryvv: if not then use .call files |
| 17:54.51 | harryvv | va, okay and if not? |
| 17:54.57 | harryvv | .call files? |
| 17:55.00 | jakepdev | http://www.voip-info.org/wiki-Asterisk+CAPI+Readme |
| 17:55.30 | vaewyn | harryvv: http://www.voip-info.org/wiki-Asterisk+auto-dial+out |
| 17:56.26 | MrClean | How do I get channel variable inheritance to work? Prepending underscores (ie. SetVar(_FOO=foo)) doesn't seem to work. Do I need a specific version of Asterisk? |
| 17:56.28 | jakepdev | Cheetah - I'm trying to figure out how to get ISDN here. you get good pricing? |
| 17:56.31 | harryvv | http://www.asteriskdocs.org/modules/tinycontent/content/docbook/current/docs-html/x1512.html |
| 17:56.36 | harryvv | that to |
| 17:56.36 | harryvv | thanks |
| 17:56.47 | Cheetah | jakepdev, well, for in-germany calls we pay €0.01 /mn |
| 17:57.08 | jakepdev | ok - that's not bad at all |
| 17:59.07 | vaewyn | although... 200+KB/s wireless ain't bad :P |
| 17:59.59 | vaewyn | They are starting to roll out DSL2 around here though... so maybe I will be within it's range |
| 18:00.14 | Dishwasha | verizon is rolling out fiber to the premise in DFW |
| 18:00.15 | vaewyn | I'm sitting at about 33,000ft from the CO by wire |
| 18:00.21 | jakepdev | FIOS |
| 18:00.35 | Cheetah | hm |
| 18:00.36 | Cheetah | okay |
| 18:00.39 | Cheetah | i got it to use CAPI |
| 18:00.54 | jakepdev | great |
| 18:01.07 | Nivex | jakepdev: my parents sat for three weeks only to find out they couldn't get DSL |
| 18:01.10 | Nivex | jakepdev: twice |
| 18:01.11 | Cheetah | i set TRUNK=CAPI and added "exten => 1234,1,DIAL(CAPI/0:134,60,T)" |
| 18:01.14 | vaewyn | hahah... I'll never see fiber to my home unless I get out my shovel and start digging a 7 mile trench |
| 18:01.22 | jakepdev | haha - i don't feel so bad now |
| 18:01.24 | Cheetah | but it tells me that there are no available card right now "all busy" |
| 18:02.27 | vaewyn | got the main chicago-detroit long distance fiber not 1/2 mile from my house... to bad I can't just 'T' into that ;P |
| 18:02.32 | *** join/#asterisk bmd (~bmd@rrcs-66-27-57-228.west.biz.rr.com) |
| 18:03.08 | Cheetah | uhm |
| 18:03.15 | Cheetah | it worked for a moment and I could hear audio |
| 18:03.36 | *** join/#asterisk ronn (ronn@host217-46-199-164.in-addr.btopenworld.com) |
| 18:03.42 | Dishwasha | Oh yeah, well, SBC has been offering me dark fiber service for years |
| 18:03.46 | Cheetah | now it went "segmentation fault" "eror while writing audio data: broken pipe" "Warning, flexible rate not heavily tested" |
| 18:03.49 | naula | ok, Question. I follow the directions here: http://digium.com/index.php?menu=configuration for installing my TDM card with FXS module and the port still is not working. Is there something else I need to do? |
| 18:04.20 | *** join/#asterisk focks (~craig@nsc66.147.95-93.newsouth.net) |
| 18:04.28 | [TK]D-Fender | naula : Does ZTCFG look like its supposed to? |
| 18:04.45 | focks | how do i tell a GrandStream 102 what digits to dial when i hit the voicemail button |
| 18:04.56 | jakepdev | Cheetah - you'll need someone else to help from here - don't know why it would be saying that |
| 18:05.10 | naula | fender: ztcfg comman? |
| 18:05.16 | Cheetah | google groups said that mpg123 is causing this... |
| 18:05.17 | Cheetah | any ideas? |
| 18:05.29 | vaewyn | focks: Voice Mail UserID entry |
| 18:05.30 | ronn | anyone knows how to kill a zombie channel? |
| 18:05.51 | focks | vaewyn so i'd put like *98 exten? |
| 18:05.57 | jakepdev | Cheetah - i doubt it has to do with mpg123 |
| 18:06.01 | Cheetah | hmm |
| 18:06.07 | Cheetah | now a few test calls worked |
| 18:06.08 | naula | fender: nothing comes up when I type ztcf. |
| 18:06.13 | jakepdev | Cheetah - most likely due to capi configs |
| 18:06.22 | Cheetah | hmmm |
| 18:06.38 | vaewyn | focks: whatever you would normally dial yourself |
| 18:06.54 | focks | thanks |
| 18:06.57 | Cheetah | might "capi debug" have caused this? |
| 18:07.11 | jakepdev | don't think so |
| 18:07.22 | naula | ztcfg -vv shows everything is correct though |
| 18:07.43 | *** join/#asterisk rg1 (~rg1@mail.airlinksystems.com) |
| 18:07.52 | ronn | anyone knows how to kill a zombie channel? |
| 18:07.59 | jakepdev | but - i must admit that i've only worked with Analog, PRI, SIP, and IAX |
| 18:08.36 | ronn | SIP/4488-e964<ZOMBIE> |
| 18:09.04 | focks | sip reload? |
| 18:09.16 | Cheetah | jakepdev, thanks a lot for the help! |
| 18:09.20 | Cheetah | it saved me hours ;D |
| 18:09.27 | sudhir492 | ronn: soft hangup SIP/4488-e964<ZOMBIE> |
| 18:09.50 | ronn | sudhir492: thanks |
| 18:09.59 | jakepdev | Cheetah - np - good luck with it |
| 18:10.04 | Cheetah | ;) |
| 18:11.26 | *** join/#asterisk kiokorobert (~kiokorobe@196.200.26.42) |
| 18:11.31 | ronn | sudhir492: can you do the same for a this as well? Zap/pseudo-1462573877 |
| 18:11.38 | rvhi | i got three core dumps yesterday, here are the 'bt' and 'bt full' |
| 18:11.39 | Cheetah | how do it tell asterisk to use the number that has been entered into the field? i tested "exten => .,Dial(CAPI/1:${EXTEN},60,T) |
| 18:11.42 | rvhi | http://pastebin.ca/12063, http://pastebin.ca/12064, http://pastebin.ca/12065 |
| 18:11.46 | Cheetah | that doesnt work, though |
| 18:11.49 | sudhir492 | ronn: yes |
| 18:12.03 | jakepdev | ${EXTEN} is the number dailed |
| 18:12.07 | ronn | thanks |
| 18:12.25 | Cheetah | hm, what about the identifier in the first place? |
| 18:12.29 | Cheetah | i made it "." |
| 18:12.42 | rvhi | going up a few functions, they all have corrupted data at pbx_extension_helper() |
| 18:12.44 | *** join/#asterisk outtolunc (~me@adsl-66-218-53-170.dslextreme.com) |
| 18:12.47 | jakepdev | how may digits? |
| 18:12.59 | sudhir492 | Cheetah: if you are using pattern, then use exten => _.,1,Dial(CAPI/1:${EXTEN},60,T) |
| 18:13.01 | Cheetah | it varies, depending on how many numbers are in the phone number |
| 18:13.03 | rvhi | anyone can help me on this? |
| 18:13.12 | jakepdev | for example - _XXXXXX will trap 6 digits |
| 18:13.16 | jakepdev | don't use ._ |
| 18:13.22 | jakepdev | _. |
| 18:13.29 | jakepdev | it's bad news |
| 18:13.31 | Cheetah | yay |
| 18:13.33 | Cheetah | it workes :D |
| 18:13.41 | Cheetah | now how to remove the lag that I'm experiencing? :) |
| 18:13.46 | Cheetah | i feels there is a few ms lag |
| 18:13.58 | jakepdev | at what point? |
| 18:14.07 | Cheetah | while talking to the other phone |
| 18:14.25 | jakepdev | youre noticing delay? |
| 18:14.33 | CoffeeIV | I put a TE110P in an asterisk@home computer; I removed the 100P's that were there before and ran "genzaptelconf -s" and it put a commented out section in zapata-channels.conf. I added the lines "span=1,1,0,esf,b8zs" and "bchan=1-23" and reloaded configs, but "zap show channels" shows no channels. Is this what I should expect ? |
| 18:14.41 | Cheetah | yeah |
| 18:15.12 | jakepdev | Cheetah - VOIP is translating so there is a delay when coding/decoding |
| 18:15.21 | Cheetah | ahh okay |
| 18:15.31 | Cheetah | its not really horrible, but noticable if you hear both devices |
| 18:15.41 | Cheetah | so... which codec is the best? without too much quality loss? |
| 18:15.43 | jakepdev | yes - i wish there were no delay also |
| 18:16.01 | CoffeeIV | do I have to add lines such as "fxoks=1-23" or similar before I see channels ? |
| 18:16.06 | jakepdev | g729 works well - but it is $10 per channel - g711 works well though and it's free |
| 18:16.26 | jakepdev | use allow=ulaw |
| 18:16.47 | jakepdev | in sip.conf |
| 18:16.51 | jakepdev | disallow=all |
| 18:16.55 | PTG123 | g711 takes 100kbps, g729 uses 8 |
| 18:17.23 | PTG123 | let me see $10 for life, to save 90kbps? :) |
| 18:17.36 | Cheetah | okay, i did that, but its still not as good as Skype :) |
| 18:17.56 | PTG123 | ulaw quality is going to suck if you have traffic on your network |
| 18:17.57 | Cheetah | bandwidth is not the problem.. the clients and server are connected via 100 mbit |
| 18:17.57 | jakepdev | what codec is Skype using? |
| 18:18.03 | *** part/#asterisk bmd (~bmd@rrcs-66-27-57-228.west.biz.rr.com) |
| 18:18.05 | PTG123 | skype uses gsm most likely try it |
| 18:18.05 | Cheetah | their own |
| 18:18.09 | Cheetah | i just compare it ;) |
| 18:18.15 | PTG123 | try gsm |
| 18:18.19 | Cheetah | sec |
| 18:18.22 | Cheetah | allow=gsm? |
| 18:18.26 | PTG123 | its free, and performance is good |
| 18:18.26 | PTG123 | yes |
| 18:19.23 | MrClean | Does channel variable inheritance work in Asterisk 1.0.7? |
| 18:19.42 | Cheetah | okay |
| 18:19.45 | Cheetah | quality is good now ;) |
| 18:19.58 | Cheetah | now I get alot of "WARNING Inband DTMF is not supported on codes GSM" |
| 18:20.05 | PTG123 | yah disable that in your conf |
| 18:20.08 | PTG123 | inband=n |
| 18:20.09 | PTG123 | er no |
| 18:20.10 | PTG123 | i think |
| 18:20.12 | PTG123 | something like that |
| 18:20.14 | Cheetah | what is inbound? |
| 18:20.30 | jakepdev | dtmf=??? |
| 18:20.35 | focks | how can i fix low volume on a TDM400? |
| 18:20.40 | focks | rx and tx gain? |
| 18:20.45 | Cheetah | i have set "dtmfmode=inband" |
| 18:20.54 | jakepdev | inband means dtmf is in with the audio |
| 18:21.02 | PTG123 | you don't want it inband |
| 18:21.02 | km- | use dtmfmode=rfc2833 I think |
| 18:21.43 | Cheetah | okay |
| 18:21.48 | Cheetah | that fixed it :) |
| 18:21.49 | Cheetah | thanks! |
| 18:21.51 | Cheetah | foodises waits |
| 18:22.08 | PTG123 | i recommend getting g729 licenses |
| 18:22.13 | PTG123 | its much better then gsm |
| 18:22.24 | Cheetah | well, maybe once we use it more itensive on this machine |
| 18:22.32 | Cheetah | for now a ISDN->VoIP gateway will do fine :) |
| 18:22.33 | *** join/#asterisk zotz (~zotz@208.196.247.140) |
| 18:22.48 | jakepdev | MrClean - aren't the variables per channel or global? didn't know of any other options |
| 18:23.04 | naula | ok, Question. I follow the directions here: http://digium.com/index.php?menu=configuration for installing my TDM card with FXS module and the port still is not working. Is there something else I need to do? |
| 18:23.27 | jakepdev | naula - digium provides free support on this |
| 18:23.49 | naula | Where? |
| 18:23.55 | MrClean | The wiki says that channel variables can be inherited by other channels created by the main channel if the variable name is prepended with an underscore or two. The page isn't clear on which version(s) of asterisk support this feature. |
| 18:23.56 | jakepdev | 877-LINUXME |
| 18:24.19 | *** part/#asterisk Zeeek (~Zeeek@Zeeek.sustaining.supporter.pdpc) |
| 18:24.58 | km- | hey, does anyone here use asterisk-java and FastAGI? |
| 18:25.04 | km- | I've been playing with it and it is really something |
| 18:25.43 | *** join/#asterisk IPmonger (~ipmonger@pcp0010577106pcs.coatsv01.pa.comcast.net) |
| 18:26.06 | km- | ipmonger: coatesville! |
| 18:26.09 | Nuxi | Anyone playing with java not with fastagi? |
| 18:26.14 | IPmonger | km-: yes! |
| 18:26.19 | km- | ipmonger: howdy from exton |
| 18:26.23 | IPmonger | km-: hello! |
| 18:26.49 | km- | yeah, we run asterisk at our company |
| 18:27.05 | IPmonger | cool |
| 18:27.18 | km- | nuxi: are there java apis that aren't necessarily wedded to agi? |
| 18:27.32 | km- | hehe, ccm == teh suck |
| 18:27.54 | IPmonger | i think my ccm got set up the bomb |
| 18:27.56 | km- | We've got a p3 sittin here with a TE405P running to our legacy NEC system |
| 18:28.43 | km- | nuxi: I need to find the author of asterisk-java in here and pick his brains, there needs to be tons more tutorials on it |
| 18:29.02 | km- | nuxi: I can't for the life of me figure out how to instantiate the getdigits command so that I actually get digits |
| 18:29.06 | *** part/#asterisk cpatry (~grepmoo@65.39.228.5) |
| 18:29.23 | IPmonger | km-: you near "main street" @ exton? |
| 18:29.29 | km- | ipmonger: I was at there for lunch today |
| 18:29.32 | km- | china grill++ |
| 18:29.34 | IPmonger | heh |
| 18:29.47 | IPmonger | we practically live @ desert moon |
| 18:29.50 | IPmonger | and the b&n |
| 18:29.55 | km- | hahahaha |
| 18:30.03 | km- | I'm not a big fan of mexican |
| 18:30.09 | km- | chinese, however, I can deal with |
| 18:30.22 | km- | what company do you guys work for and are you hiring? :P |
| 18:30.26 | *** join/#asterisk jakepdev[work] (HydraIRC@ppp11.pm3-1.phi-pt.pa.localnet.com) |
| 18:30.46 | km- | there we go with another philadelphia suburb |
| 18:31.01 | IPmonger | a clec w/HQ in NC |
| 18:31.13 | km- | haha, yeah, I think your company is quoting us for phone service |
| 18:31.14 | vaewyn | well... if you don't like chinese just wait 15 minutes and eat something else ;P |
| 18:31.20 | km- | vaewyn: rofl |
| 18:31.26 | km- | in one end, out the other |
| 18:31.59 | Corydon-w | IPmonger: cool, can we get IPv6 routing on USLEC? |
| 18:32.05 | IPmonger | not yet |
| 18:32.09 | km- | I wouldnt mind an ipv6 drop myself |
| 18:32.15 | IPmonger | i've been pushing it |
| 18:32.36 | Corydon-w | Damn... we hardly use our USLEC uplink now anyways |
| 18:32.37 | km- | luckily I've got buds in belgium who gre-tunnel me a couple /64s |
| 18:32.40 | IPmonger | if one of our sales reps talks to your company, make sure and mention that you want it |
| 18:33.11 | km- | more ip addresses than I ever know what to do with.... |
| 18:33.33 | Corydon-w | Supposedly sometime this year, USLEC is going to support dynamic allocation of our PRI channels between voice and data |
| 18:33.47 | km- | ooooh! |
| 18:33.57 | km- | now that sounds like a useful technology |
| 18:34.15 | Corydon-w | km-: well, it's already part of Asterisk... ZapRAS |
| 18:34.21 | IPmonger | Corydon-w: are you referring to the Dynamic-T product? |
| 18:34.32 | Corydon-w | IPmonger: have no idea what the product name is |
| 18:34.35 | IPmonger | heh |
| 18:34.45 | km- | I love talking to telco companies and I tell them I want frac PRI service and they're like "umm, you cant do that!" |
| 18:35.22 | km- | we have XO right now |
| 18:35.31 | km- | and they've more or less bent us over the router and had their way with us |
| 18:35.36 | jakepdev[work] | anyone know where to get ISDN for phila. suburbs? |
| 18:35.48 | km- | jakepdev: there are a couple regional ISP's that can help out |
| 18:35.54 | jakepdev[work] | for voice? |
| 18:36.04 | jakepdev[work] | or only data? |
| 18:36.06 | km- | jakepdev: OH. Have you tried calling verizon? |
| 18:36.12 | IPmonger | Corydon-w: Dynamic-T is a VOIP solution |
| 18:36.14 | km- | I'm pretty sure ISDN is available everywhere |
| 18:36.15 | jakepdev[work] | yep |
| 18:36.22 | km- | at least in SEPA |
| 18:36.23 | jakepdev[work] | they don't offer residential ISDN |
| 18:36.41 | km- | oh, just dont tell them its residential, it's for your "home office" |
| 18:36.41 | jakepdev[work] | they stopped last year |
| 18:36.41 | km- | :) |
| 18:36.47 | jakepdev[work] | ok |
| 18:36.51 | IPmonger | jakepdev[work]: what about DCAnet? |
| 18:36.51 | Corydon-w | IPmonger: well, that sucks |
| 18:37.04 | jakepdev[work] | hmm - i'll check DCA net |
| 18:37.21 | IPmonger | jakepdev[work]: I used to use them for DSL - they're based out of Wilmington, but they do ISDN |
| 18:37.23 | vaewyn | ISDN is usually too far over priced in the US to be worth it anyways... especially with PRI and DSL options |
| 18:38.03 | IPmonger | http://www.dca.net/products/internetaccess/isdn.html |
| 18:38.39 | vaewyn | heck... is 1/5th the cost of a voice PRI to get 2B+D ISDN here |
| 18:38.53 | Nugget | http://slacker.com/~nugget/stuff/greeting.wav <-- yay |
| 18:39.33 | vaewyn | Nugget: bwahahaha... niiiiceeee |
| 18:39.55 | jakepdev[work] | if there is such a thing |
| 18:39.56 | vaewyn | hahahaha... not in the US |
| 18:40.05 | vaewyn | germany has it |
| 18:40.19 | jsolares | anyone here has experience connecting an avaya definity to an asterisk box? |
| 18:40.21 | IPmonger | yes, isdn is very expensive here in the US compared to other options |
| 18:40.28 | jakepdev[work] | jsolares - yes |
| 18:40.51 | km- | jsolares: no |
| 18:40.54 | jakepdev[work] | jsolares - what is your app |
| 18:41.01 | km- | oops |
| 18:41.13 | jsolares | jakepdev[work]: with e1's? i'm having a weird issue, according to my asterisk box the d-chan is down, and acording to the definity there's nothing on the other side |
| 18:41.17 | km- | jsolares: I meant, to say "no, but I did it to an NEC system!" |
| 18:41.22 | jsolares | the zttol gives no alarms |
| 18:41.23 | *** join/#asterisk kiokorobert (~kiokorobe@196.200.26.42) |
| 18:41.27 | jsolares | zttool* |
| 18:41.40 | jsolares | km- an enter too soon ;) |
| 18:41.50 | jakepdev[work] | jsolares - we use T1 E&M - couldn't get the E1 config to work |
| 18:41.51 | km- | exactly, overzealous enter-key massaging |
| 18:42.02 | jsolares | well crap |
| 18:42.15 | jsolares | i have the TN2464 card in the definity |
| 18:42.26 | jakepdev[work] | actually - as trunks or as DS1FD stations? |
| 18:42.33 | jsolares | as trunks |
| 18:42.41 | jsolares | i'm considering a swift kick to make it work |
| 18:42.44 | jakepdev[work] | ok - i think we might have got that to work |
| 18:42.59 | jsolares | ohhhh |
| 18:43.38 | jakepdev[work] | does your d channel match avaya to asterisk? |
| 18:43.59 | jsolares | erhm come again? |
| 18:44.21 | jakepdev[work] | what is the d channel set to on the avaya? |
| 18:44.27 | jsolares | i have d channel on 16 on both |
| 18:45.01 | jakepdev[work] | were you able to get T1 working? |
| 18:45.29 | jsolares | i have a E1 only card on my asterisk, dont have T1 |
| 18:45.31 | jakepdev[work] | or is it just an e1 card for your asterisk? |
| 18:45.33 | jakepdev[work] | ok |
| 18:45.34 | *** join/#asterisk L|NUX (~linux@202.5.145.54) |
| 18:46.22 | jakepdev[work] | asterisk should be set to pri_cpe |
| 18:46.24 | *** join/#asterisk bmd (~bmd@rrcs-66-27-57-228.west.biz.rr.com) |
| 18:46.35 | jsolares | not pri_net?? |
| 18:46.48 | *** join/#asterisk anthm (~anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 18:46.48 | *** mode/#asterisk [+o anthm] by ChanServ |
| 18:46.56 | jakepdev[work] | we use it in cpe mode |
| 18:47.05 | *** part/#asterisk bmd (~bmd@rrcs-66-27-57-228.west.biz.rr.com) |
| 18:47.07 | focks | what is a reasonable rxgain to use for a TDM400? |
| 18:47.17 | jsolares | jakepdev[work] and the avaya is set to network? |
| 18:47.25 | jakepdev[work] | right |
| 18:47.40 | jsolares | that i hadnt thought of, lets see |
| 18:49.06 | jsolares | Status: Provisioned, Down, Active |
| 18:49.07 | *** join/#asterisk goldenear (~goldenear@m29.net81-64-14.noos.fr) |
| 18:49.09 | jsolares | bleh |
| 18:49.20 | jakepdev[work] | ztcfg -vv? |
| 18:49.42 | jsolares | span=1,1,0,ccs,hdb3,crc4,yellow |
| 18:50.46 | jakepdev[work] | jsolares - I think this config worked for me: http://www.voip-info.org/wiki-Asterisk+Avaya |
| 18:51.09 | jakepdev[work] | (i know it's net) |
| 18:51.36 | goldenear | hi there. I need some advice : what is IYO the best billing/rating (open source) software for asterisk ? |
| 18:52.46 | jsolares | ok a question then, does it have to be AFTER connecting them with the crossover cable? |
| 18:53.35 | goldenear | my asterisk server only manage IAX : all the clients are IAX2 only and I have an iax trunk link to a provider for DID or dialout. |
| 18:54.28 | *** join/#asterisk kiokorobert (~kiokorobe@196.200.26.42) |
| 18:56.21 | jakepdev[work] | we connected it using a standard PRI crossover cable |
| 18:57.11 | jsolares | 1-2,4-5 to 4-5,1-2 if i'm not mistaken |
| 18:57.21 | jsolares | the only problem i seem to have is that the dchan is not up |
| 18:57.27 | *** join/#asterisk darby_t (~tom@dnm246.neoplus.adsl.tpnet.pl) |
| 18:57.31 | jsolares | i change anything in zaptel.conf and i get alarms |
| 18:57.33 | jakepdev[work] | (1-4, 2-5) |
| 18:57.51 | goldenear | does anybody know TRABAS ? |
| 18:57.56 | jakepdev[work] | this one: http://www.voip-info.org/wiki-crossover+T1+cable |
| 18:58.06 | *** join/#asterisk tzafrir_laptop (~tzafrir@bzq-179-40-134.cust.bezeqint.net) |
| 18:59.38 | doolph | anyone has a good billing software that work with sip trunks, but the billing must be accuracy |
| 18:59.48 | jsolares | yep that one, confirmed |
| 19:01.11 | goldenear | doolph, I won't deal with sip, only IAX |
| 19:01.27 | doolph | well it's same |
| 19:01.45 | *** join/#asterisk meppl (mephisto@p54AAEFDE.dip.t-dialin.net) |
| 19:01.45 | doolph | my problem that the call is being connected while it is still ringing |
| 19:01.46 | rg1 | I have a TE110P Digium T1 card - trying to get it "fired up" - no lights on the back - asterisk running - can anyone help me with config? |
| 19:01.55 | jsolares | jakepdev[work]: thanks for the help, i'll be checking it after lunch |
| 19:02.01 | jakepdev[work] | ok - gl |
| 19:02.04 | HA | anybody know where i can get a good and cheap demultiplexer? i think thats what i need to break up our ds3 into seperate t1s |
| 19:02.26 | rg1 | Note: re TE110P - I have a good loopback connector in back - still no lights at all - anyone else using this card? |
| 19:02.36 | doolph | HA you need a good router, like qos |
| 19:02.48 | *** join/#asterisk theHub (~karl@69.177.93.36) |
| 19:03.23 | goldenear | doolph, you may have use the "duration" field of the CDR instead of "billsec" |
| 19:04.14 | goldenear | doolph, what billing system do you use ? |
| 19:04.15 | sudhir492 | HA: what is your budget? |
| 19:05.03 | sudhir492 | rg1: yes I use te110 both in US as well as south america (euro isdn). works well |
| 19:05.50 | rg1 | sudhir492 - can you help me with the .conf for that? |
| 19:05.55 | doolph | goldenear the one from asterisk with mysql |
| 19:06.10 | sudhir492 | rg1: what is your zaptel.conf? strip out the comment and post it on pastebin |
| 19:06.21 | HA | sudhir492: would like to keep it around $2500 |
| 19:06.32 | doolph | my duration is same that billsec |
| 19:06.37 | sudhir492 | HA: you should not have problem with that |
| 19:06.49 | sudhir492 | you can demux cheaper than that |
| 19:07.18 | rg1 | sudhir492 - you mean "zapata.conf"? |
| 19:07.31 | HA | any recommendations? I have no idea where to begin with this and I have to get a quote put together today for the boss. Just found out that our DS3 will be a bnc term and we need to break that up into Cat5 T1 Connections. |
| 19:07.33 | sudhir492 | rg1: /etc/zaptel.conf first |
| 19:07.49 | rg1 | k |
| 19:08.01 | sudhir492 | zapata.conf later |
| 19:08.02 | doolph | goldenear what bill soft do you use? |
| 19:09.15 | goldenear | I haven't done my choise yet ... |
| 19:09.45 | sudhir492 | I wrote my own billing software :-) |
| 19:09.55 | goldenear | that's why I need some advice of people using asterisk and a billing software |
| 19:10.34 | HA | any recommendations on the best way to break a ds3 into t channels? specific hardware that is known to play well with asterisk is best. |
| 19:10.36 | sudhir492 | goldenear: what are you trying to do with asterisk. what are your billing needs |
| 19:10.39 | *** join/#asterisk ChkDigit (~mike@static65-87-226-124.regina.accesscomm.ca) |
| 19:10.39 | rg1 | sudhir492 - http://pastebin.com/286235 |
| 19:10.57 | goldenear | sudhir492, is it open source or does it only feet your personal needs ? |
| 19:11.29 | sudhir492 | I plan to release it opensource, not done yet |
| 19:11.39 | rg1 | sudhir492 - does that pasted info mean that it did not recognize my card? |
| 19:11.39 | goldenear | I simple want to use asterisk as a softswitch for iax only clients |
| 19:11.40 | sudhir492 | some gui work still needes to be done |
| 19:12.29 | sudhir492 | rg1: where is your span line? |
| 19:12.41 | rg1 | its commented out |
| 19:12.50 | sudhir492 | then it will not work :-) |
| 19:13.02 | goldenear | I have a provider for dial out and I need to bill my clients for using the dial out |
| 19:13.17 | rg1 | actually i guess i have it in zapa-channels.conf - but does it need to be in this file as well? |
| 19:13.38 | sudhir492 | rg1: use the following (only 4 lines, hence I am taking the liberty to post right here) |
| 19:13.40 | sudhir492 | span=1,1,0,ccs,hdb3 |
| 19:13.40 | sudhir492 | bchan=1-15,17-31 |
| 19:13.40 | sudhir492 | dchan=16 |
| 19:13.40 | sudhir492 | loadzone=us |
| 19:13.40 | sudhir492 | defaultzone=us |
| 19:13.53 | rg1 | in zaptel.conf, right? |
| 19:13.57 | sudhir492 | yep |
| 19:14.20 | rg1 | does this card have 31 channels? |
| 19:14.41 | sudhir492 | where are you using? in US or somewhere else? |
| 19:14.47 | rg1 | US |
| 19:14.51 | sudhir492 | sorry. |
| 19:14.57 | sudhir492 | here it is for us |
| 19:15.23 | sudhir492 | #span=1,0,0,esf,b8zs |
| 19:15.24 | sudhir492 | #clear=1-23 |
| 19:15.24 | sudhir492 | #dchan=24 |
| 19:15.24 | sudhir492 | #loadzone=us |
| 19:15.24 | sudhir492 | #defaultzone=us |
| 19:15.35 | sudhir492 | take out the comment cahr |
| 19:15.37 | sudhir492 | char |
| 19:15.46 | rg1 | k |
| 19:15.56 | rg1 | do i also put thse lines in zapata-channels.conf? |
| 19:18.40 | sudhir492 | actually, you can make span=1,1,0,esf,b8zs too depending on your timing need |
| 19:19.13 | rg1 | we want the asterisk card to furnish the timing - so 1,1,0,esf,b8zs ? |
| 19:19.16 | *** join/#asterisk bmd (~bmd@rrcs-66-27-57-228.west.biz.rr.com) |
| 19:19.29 | sudhir492 | HA: you can try Adtran M13 DS3 mux on ebay |
| 19:19.51 | *** join/#asterisk PTG123 (~PTG123@ip68-106-24-139.ph.ph.cox.net) |
| 19:19.59 | rg1 | sudhir492 - you show clear=1-23 - shouldn't that be bchan=1-23? |
| 19:20.05 | HA | thx sudhir. |
| 19:20.06 | sudhir492 | you can get cheap sometimes, close to $1000 |
| 19:20.12 | mike^^ | unsupported media type with ich? |
| 19:21.00 | sudhir492 | rg1: you are right. This is from my production box: |
| 19:21.01 | sudhir492 | span=1,1,0,esf,b8zs |
| 19:21.01 | sudhir492 | bchan=1-23 |
| 19:21.01 | sudhir492 | dchan=24 |
| 19:21.01 | sudhir492 | loadzone = us |
| 19:21.02 | sudhir492 | defaultzone=us |
| 19:21.40 | rg1 | sudhir492 - will re-read configs suffice? or do i need to do a restart? |
| 19:22.01 | *** join/#asterisk jmacz (~jmacz@63.245.86.165) |
| 19:22.23 | sudhir492 | I am posting zapata.conf for you on pastebin |
| 19:22.29 | rg1 | sudhir492 - just FYI, we are hooking up our asterisk box to an ADIT 600 |
| 19:22.30 | rg1 | thx |
| 19:23.36 | sudhir492 | http://pastebin.ca/12077 |
| 19:23.38 | rg1 | how do i find that on pastebin sudhir492? |
| 19:23.39 | rg1 | oh |
| 19:23.40 | *** join/#asterisk Dutts (~dutts@81.168.70.41) |
| 19:23.41 | rg1 | :) |
| 19:23.46 | *** join/#asterisk mozrat (~mozrat@80.68.89.215) |
| 19:23.50 | sudhir492 | stop asterisk. then do the following: |
| 19:24.31 | mozrat | Evening guys - any UK PRI users with callerid enabled here? BT tells me they've now enabled it on my line but I can't see any numbers coming through :( |
| 19:24.34 | Dutts | anyone here use their rtp.conf to limit the number of ports they need to open on their firewall? |
| 19:25.12 | sudhir492 | modprobe -r wcte11xp |
| 19:25.12 | sudhir492 | modprobe zaptel |
| 19:25.12 | sudhir492 | modprobe wcte11xp |
| 19:25.12 | sudhir492 | ztcfg -vvv |
| 19:25.12 | sudhir492 | asterisk -vvvgc |
| 19:25.24 | HA | sudhir492: something like this work ? http://www.tritechcoa.com/product/291951.html |
| 19:25.38 | *** join/#asterisk jwitte (~jwitte_su@firefly.alpha-lab.net) |
| 19:25.44 | sudhir492 | rg1: I gave you the url: http://pastebin.ca/12077 |
| 19:26.30 | rg1 | sudhir492 - i see you are using on line#31 "signalling=pri_cpe" - I'm not hooking this up to a PRI - it's getting directly connected into the ADIT 600 TDM T1-1 port |
| 19:27.02 | Dishwasha | I currently have a 7960 phone with 6 lines and have turned off Call Waiting in the phone. I am using the ChanIsAvail app to determine which of the 6 SIP lines are available, but when I am on the first line, ChanIsAvail still shows that line as being available. Of course when it dials it reports the line is busy. Any ideas on why it is showing the channel available or a better way to check... |
| 19:27.04 | Dishwasha | ...if the line is available? |
| 19:27.09 | rg1 | sudhir492 - yes, got it and was viewing line#31 |
| 19:27.33 | sudhir492 | rg1, in that case your signalling will be different |
| 19:27.50 | sudhir492 | and you will have 24 channels too |
| 19:27.52 | rg1 | yes, what should it be you think?> |
| 19:27.59 | rg1 | yes, i already put bchan=1-24 |
| 19:28.12 | sudhir492 | they are not bchan |
| 19:28.22 | rg1 | oh |
| 19:28.24 | sudhir492 | Let me check, I have a box that connects to T1 |
| 19:28.31 | rg1 | thx......... |
| 19:28.54 | Dutts | guys.... anyone? |
| 19:29.02 | doolph | yes |
| 19:29.03 | doolph | ? |
| 19:29.03 | sudhir492 | signalling=em_w |
| 19:29.07 | sudhir492 | channel => 1-24 |
| 19:29.25 | rg1 | k |
| 19:29.41 | sudhir492 | everything else should be the same |
| 19:30.29 | rg1 | k |
| 19:31.04 | sudhir492 | rg1: if you are near the machine, wait for the light turn green before starting asterisk |
| 19:32.07 | rg1 | so stop asterisk now? |
| 19:32.46 | sudhir492 | HA: yes that will work |
| 19:33.32 | *** part/#asterisk johnnyb (~johnnyb@sdsl-38-17-139.tulsaconnect.com) |
| 19:33.44 | rg1 | sudhir492 - i'm pretty confused - I have the TE110P card in the machine - no lights on back |
| 19:34.01 | shido6 | ee |
| 19:34.05 | shido6 | whats ztcfg -vv say? |
| 19:35.48 | mozrat | rg1 T1 or E1? |
| 19:35.56 | *** join/#asterisk Ayano (~erik_leee@adsl-70-245-190-94.dsl.spfdmo.swbell.net) |
| 19:36.02 | sudhir492 | unload the module and load it again, like I told you before. If the T1 is connected to the card, it will turn yellow and then green, if T1 is not connected or is down, the light on the card will blink red |
| 19:36.31 | rg1 | ok |
| 19:36.43 | rg1 | mozrat: T1 |
| 19:36.53 | sudhir492 | with te110p card, make sure you load zaptel first and then wcte11xp |
| 19:37.19 | mozrat | rg1, OK - I had a TE110P card this month and didn't set the jumper on the card to E1 - T1 is the default setting |
| 19:37.27 | sudhir492 | did you check the jumper setting on the card? make sure it is for T1 (I guess that is no jumper) |
| 19:38.23 | rg1 | is there a juimper? we are going to see.... |
| 19:39.10 | sudhir492 | rg1: you have everything you need now :-) good luck. |
| 19:40.56 | *** join/#asterisk n4y (~tmalkut@fw.orasoft.net.pl) |
| 19:43.58 | *** join/#asterisk viperdude_uk (~viperdude@62.249.220.204) |
| 19:48.11 | jakepdev[work] | anyone using the SPA-3000? |
| 19:48.57 | jakepdev[work] | i'm noticing some strange artifact when it's trying to do the echo cancel - i mostly hear this when someone is talking while i'm talking |
| 19:49.19 | *** part/#asterisk IPmonger (~ipmonger@pcp0010577106pcs.coatsv01.pa.comcast.net) |
| 19:50.30 | *** join/#asterisk bkw_ (~bkw_@bkw.developer.and.friend.of.asterisk) |
| 19:50.30 | *** mode/#asterisk [+o bkw_] by ChanServ |
| 19:51.08 | naula | another question for you wonderful people. I setup my TDM w/ FXS card and can get a dialtone. However the second I dial a number asterisk hangs up the call |
| 19:51.11 | *** join/#asterisk tuxinator_linuxM (~spabin@ip68-109-146-168.ph.ph.cox.net) |
| 19:51.23 | naula | any idea what could be wrong |
| 19:52.54 | [TK]D-Fender | Could be your dial plan.... |
| 19:53.00 | [TK]D-Fender | What are you using for FXO? |
| 19:53.09 | ariel_ | naula, you do not have it set to the right context |
| 19:53.42 | naula | I dont have an FXO card, im trying to dial out through my sip provider |
| 19:53.47 | naula | one moment, let me check on the context |
| 19:54.01 | Dishwasha | You know, I really wish they hadn't depricated incominglimit and outgoinglimit. GetGroup/SetGroup doesn't really work in a Dial group scenario |
| 19:54.27 | *** join/#asterisk DARP (~diegoramo@200.71.33.201) |
| 19:54.31 | shido6 | then |
| 19:54.31 | DARP | hi |
| 19:54.34 | shido6 | find the old code |
| 19:54.40 | shido6 | and shove it in your version |
| 19:54.42 | shido6 | :) |
| 19:54.43 | DARP | need some little help |
| 19:55.37 | *** join/#asterisk Dunkirk (~david@pcp0010184333pcs.columbus.in.indy.comcast.net) |
| 19:56.00 | naula | What should the context be on the FXS card? I dont need incoming calls, just outgoing. |
| 19:56.10 | Dishwasha | shido6: Well, maybe there's an alternative that can accomplish what I want. I want a Dial group where it dials 2 extens, plus 1 of 6 extensions. I want it to pick 1 of the 6 extensions based on whether it is currently in use, so if the first extension is in use have it use the second extension, etc. |
| 19:56.38 | *** join/#asterisk bkw_ (~bkw_@bkw.developer.and.friend.of.asterisk) |
| 19:56.38 | *** mode/#asterisk [+o bkw_] by ChanServ |
| 19:56.47 | Dishwasha | Unfortunately in my SIP scenario, ChanIsAvail reports a line is available even if it is in use an cannot accept any more incoming calls |
| 19:57.07 | Dishwasha | I'm afraid I will have to AGI this |
| 19:57.50 | *** join/#asterisk IPmonger (~ipmonger@pcp0010577106pcs.coatsv01.pa.comcast.net) |
| 19:58.19 | *** join/#asterisk bah (048830696@AC866751.ipt.aol.com) |
| 19:58.58 | Dunkirk | I'm trying to do what I thought would be the MOST simple thing to do with *: I bought a TDM411, and I'm trying to hookup my phone line to the FXO port, and my phone to the FXS port. |
| 19:59.36 | Dunkirk | I've got all the modules loaded and the software is running, but the docs stop short of actually telling me how to configure my physical phone to receive calls and dial out. |
| 19:59.38 | shido6 | ok, so whats wrong Dunkirk ? |
| 19:59.42 | Dunkirk | Can anyone help me? |
| 19:59.46 | shido6 | sure |
| 19:59.50 | shido6 | what do you want it to do? |
| 19:59.52 | shido6 | spell it out |
| 20:00.05 | shido6 | "when a call comes in, I want it to answer, "thank you for calling ...." |
| 20:00.12 | rg1 | sudhir492 - ok, so far i have green lights on my TE110P and ADIT-600 connection |
| 20:00.21 | shido6 | then I want it ring my cell phone and ring my fxs , too and dump into voicemail if no one picks up after 5 rings |
| 20:00.27 | shido6 | I need that kind of detail, Dunkirk |
| 20:00.43 | Dunkirk | shido6: I just want an incoming call to ring my physical phone, and to give me dial tone to dial out when I pick up the handset. |
| 20:00.53 | Dunkirk | I just want Asterisk to be invisible for the moment. |
| 20:00.58 | Dunkirk | After that, I can get fancy. |
| 20:01.16 | shido6 | ok well do you have your house line connected to your card |
| 20:01.19 | [TK]D-Fender | Dunkirk : you mean for an extension to get an IMMEDIATE live external dialtone? |
| 20:01.19 | Dunkirk | Yes. |
| 20:01.20 | shido6 | and a handset connected to your card? |
| 20:01.27 | Dunkirk | Yes. |
| 20:01.34 | Dunkirk | [TK]D-Fender, Yes. |
| 20:01.34 | shido6 | show me ztcfg -vv at pastebin.ca |
| 20:01.45 | Dunkirk | pastebin.ca? |
| 20:01.48 | shido6 | yes |
| 20:01.51 | shido6 | http://pastebin.ca |
| 20:01.55 | *** part/#asterisk darby_t (~tom@dnm246.neoplus.adsl.tpnet.pl) |
| 20:02.28 | Dishwasha | Dunkirk: You probably want Dial(line,time|r) the |r gives a ring to the person calling in |
| 20:02.33 | [TK]D-Fender | I guess you could just to a dialplan entry like "exten => _.,Dial(Zap/1/${exten})" but that defeats the idea of having a PBX sorta... |
| 20:02.33 | *** join/#asterisk darby_t (~tom@dnm246.neoplus.adsl.tpnet.pl) |
| 20:02.52 | Dunkirk | Sorry, screwed up the first paste. ;-) |
| 20:03.20 | Dunkirk | [TK]D-Fender, I just want to learn at this point. Ground up. |
| 20:03.34 | Dunkirk | [TK]D-Fender, Nice nic, BTW. |
| 20:04.34 | *** join/#asterisk leenuxg33k (~bpeck@66-189-118-133.dhcp.oxfr.ma.charter.com) |
| 20:05.56 | [TK]D-Fender | What are you using for FXS/FXO? |
| 20:05.59 | Dunkirk | [TK]D-Fender: So that would be my *only* line in extensions.conf? |
| 20:06.09 | Dunkirk | [TK]D-Fender: A TDM411P. |
| 20:06.18 | Dunkirk | The "dev kit" card. |
| 20:06.25 | *** join/#asterisk mike^^ (~mike@ip24-252-68-144.no.no.cox.net) |
| 20:06.27 | [TK]D-Fender | No, far from, but the only basic part for dialing out. |
| 20:07.20 | Dunkirk | [TK]D-Fender: So can you explain "exten => _.,Dial(Zap/1/${exten})" to me? |
| 20:07.33 | Dunkirk | And, using my card, that should be "Zap/4", right? |
| 20:08.57 | Dunkirk | shido6: http://pastebin.ca/12089 |
| 20:09.44 | Dunkirk | [TK]D-Fender: I tried that line, and I can get dial tone and dial a number, but then I just get dial tone again. |
| 20:10.11 | *** join/#asterisk kryme (~kryme@66-211-192-4.velocity.net) |
| 20:10.15 | Dunkirk | My total extensions.conf file is here: http://pastebin.ca/12094 |
| 20:10.39 | kryme | OK. I've come to the conclusion that sixTel sucks. Does anyone else have a provider that they'd recommend? |
| 20:12.01 | viperdude_uk | hi, i am total newbie to asterisk but not to linux. I have been looking for a tutorial on google all afternoon to show me how to do the following. I just want to setup up 2 SIP extensions that can ring each other on my LAN. Does anyone know a basic tutorial that shows how to do this? |
| 20:12.47 | mozrat | viperdude_uk: the asterisk handbook would go some small way |
| 20:12.54 | mozrat | or you could get asterisk@home |
| 20:13.06 | viperdude_uk | asterisk@home? |
| 20:13.32 | mozrat | asteriskathome.sf.net |
| 20:13.37 | mozrat | or xorcom.com |
| 20:13.43 | mozrat | I keep meaning to find out which is better |
| 20:13.52 | mozrat | they are "out of the box" deployments |
| 20:13.52 | viperdude_uk | ok i will check them out thanks |
| 20:15.09 | kryme | I personally prefer Asterisk@Home |
| 20:15.28 | mutilator | woot |
| 20:15.32 | mutilator | state tax return came in |
| 20:15.34 | mutilator | 05/16/2005 ACH CREDIT SOM INCOME TAX PAYMENTS 050516 $5.00 |
| 20:15.37 | rg1 | back here agian seeking assistance - Trying to setup my channels on Asterisk - have TE110P card connected to an ADIT 600. Have green lights all the way around at this point with these 4 lines in zaptel.conf: |
| 20:15.43 | rg1 | span=1,1,0,esf,b8zs |
| 20:15.44 | rg1 | bchan=1-24 |
| 20:15.44 | rg1 | loadzone=us |
| 20:15.44 | rg1 | defaultzone=us |
| 20:15.58 | mozrat | rg1, was it the jumpers? |
| 20:16.00 | rg1 | need help on setting up the channels |
| 20:16.08 | rg1 | mozrat - nope, no jumpers |
| 20:16.13 | rg1 | i think the above lines |
| 20:16.14 | rvhi | hi, my * kept crashing, is there any knob to turn on in Makefile and make it easier troubleshooting? |
| 20:16.25 | rg1 | mozrat, can you help me set up my channels? |
| 20:16.38 | *** join/#asterisk W1thdraw (~Withdraw@lund-juri-sr0-vl101-012.lundsstadsnat.net) |
| 20:17.08 | mozrat | rg1... I can try :) |
| 20:17.10 | shido6 | what is the g729 freq for iax trunking |
| 20:17.33 | mozrat | rg1, are you going to lump them together into one group? |
| 20:17.38 | rg1 | mozrat - super - if I knew even the files I'll be dealing with, that would be a great start |
| 20:17.47 | rg1 | nope - here's the setup mozrat: |
| 20:18.12 | doolph | rvhi check your modules |
| 20:18.18 | rg1 | 1. Channels 1-3 will be connected to loopstart interfaces from phone co |
| 20:18.31 | rg1 | 2. Channels 4-6 will be connected to groundstart interfaces from phone co |
| 20:18.57 | rg1 | 3. Channels 9-16 will be connected to analog handsets |
| 20:19.35 | mozrat | so what do you have in /etc/zaptel.conf ? |
| 20:19.46 | [TK]D-Fender | Dunkirk : That line I gave you was a sample since I don't know your actual setup, but your extensions.conf could be really abbreviated for what you are describing. |
| 20:19.52 | rg1 | span=1,1,0,esf,b8zs |
| 20:19.52 | rg1 | <PROTECTED> |
| 20:19.52 | rg1 | <PROTECTED> |
| 20:19.52 | rg1 | <PROTECTED> |
| 20:21.06 | mozrat | rg1, ok so now in /etc/asterisk/zapata.conf |
| 20:21.12 | mozrat | you can define your groups |
| 20:21.18 | rg1 | going there.... |
| 20:21.48 | mozrat | see, I'm don't know all that much about this - you may need to define some groupings in /etc/zapata.conf... anyone? |
| 20:21.49 | rg1 | i'm there |
| 20:22.09 | *** join/#asterisk lesouvage (~lesouvage@cc341200-a.assen1.dr.home.nl) |
| 20:22.42 | mozrat | wait one sec, rg1 |
| 20:22.49 | rg1 | mozrat - ok |
| 20:23.05 | *** join/#asterisk darby_d (~tom@dnr33.neoplus.adsl.tpnet.pl) |
| 20:23.42 | Dunkirk | [TK]D-Fender: Well, like I said, I just have one FXO port, 1 FXS port, my telephone line, and a handset. I know I've got it wired correctly, I just want Asterisk to act like it's not there: push everything coming in to the handset, and let the handset dial out just like normal. |
| 20:24.08 | Dunkirk | [TK]D-Fender: I would have *thought* that this would have been the starting point for every example, to make sure it all works, but I was oh so wrong. |
| 20:24.30 | mozrat | rg1, in zapata.conf you need to define your ISDN switchtype... |
| 20:24.49 | mozrat | ...pridialplan... |
| 20:24.58 | rg1 | its not that |
| 20:25.16 | lesouvage | does anybody knows a not to expensive place to stay in madrid during astricon |
| 20:25.30 | mozrat | rg1, its the groupings? |
| 20:25.42 | shido6 | ok im back |
| 20:26.40 | lesouvage | Dunkirk: try xorcom rapid asterisk. It's completely preconfigured and I'm sure it will work. |
| 20:26.51 | *** part/#asterisk bkw_ (~bkw_@bkw.developer.and.friend.of.asterisk) |
| 20:26.54 | mozrat | brb |
| 20:26.57 | *** part/#asterisk mozrat (~mozrat@80.68.89.215) |
| 20:27.23 | *** join/#asterisk bkw_ (~bkw_@bkw.developer.and.friend.of.asterisk) |
| 20:27.23 | *** mode/#asterisk [+o bkw_] by ChanServ |
| 20:28.08 | Dunkirk | lesouvage: Thanks for the tip. I'm downloading now. Maybe I can extract the configs and apply it to my Gentoo setup. |
| 20:28.24 | *** join/#asterisk r0d3nt (~RatMan@wsip-24-234-241-84.lv.lv.cox.net) |
| 20:29.01 | *** join/#asterisk pcm (~pcm@user-69-73-0-22.knology.net) |
| 20:29.16 | *** part/#asterisk km- (~pgrace@brdgw1.rttx.com) |
| 20:29.24 | pcm | anyone uses asterisk with SIP and the call sounds good after a few minutes ? |
| 20:30.03 | kapejod | yes |
| 20:30.15 | pcm | kapejod: what version do you use ? |
| 20:30.31 | pcm | kapejod: did you experience something like that in any 1.0.x versions ? |
| 20:30.45 | lesouvage | Dunkirk: if you try rapid don't forget to enable sip in the rapid-menu by typing yes. |
| 20:31.02 | [TK]D-Fender | Dunkirk : what you want to do is set it up with some "test" extensions to lead to voicemail, another one for a playback test, and so on. The defaul ones in the samples file is too convoluted for new users I find. If I'm on later I'll pass you some samples |
| 20:31.10 | kapejod | pcm: we use cvs head and * 1.0.7 |
| 20:31.11 | [TK]D-Fender | but for now I've got to go. |
| 20:31.52 | lesouvage | Is noboddy going to madrid? |
| 20:31.56 | pcm | kapejod: i used broadvoice with 1.0.6 and the call got broken after a few minutes |
| 20:32.12 | pcm | kapejod: i mean the voice just chokes itself ... |
| 20:33.58 | kapejod | pcm: ever complained to broadvoice? ;) |
| 20:33.59 | mike^^ | rtp.c:377 ast_rtcp_read: Got RTCP report of 68 bytes |
| 20:34.00 | mike^^ | heh |
| 20:34.04 | mike^^ | i cant get any calls to go through now |
| 20:34.14 | mike^^ | it dials but then nothing happens |
| 20:34.21 | mike^^ | anyone ever had that prob? |
| 20:34.23 | kapejod | lesouvage: i think madrid in general attracts lots of tourists. |
| 20:35.28 | Blackvel | ah kapejod there...starting ad about his bristuff cvs head version :) |
| 20:35.36 | mike^^ | <PROTECTED> |
| 20:35.36 | mike^^ | May 18 15:34:39 DEBUG[16145]: channel.c:2733 ast_channel_bridge: Bridge stops bridging channels SIP/2200-207b and SIP/iconnect-054a |
| 20:35.37 | mike^^ | weird |
| 20:35.46 | kapejod | Blackvel: nope |
| 20:36.15 | lesouvage | kapejod: there is an asterisk conference taking place in june. Hotel Auditorium is an expensive place to stay so I was just wondering if anybody finds a cheaper place to stay in Madrid during the conference. |
| 20:37.07 | file | there's also Cluecon in August |
| 20:37.18 | *** join/#asterisk RoyK (~roy@179.80-203-29.nextgentel.com) |
| 20:37.23 | file | then come and get some! |
| 20:37.47 | kapejod | what could ulaw clue help me? ;) |
| 20:37.48 | hypa7ia | lesouvage, there is probably a youth hostel in madrid |
| 20:37.56 | kapejod | lesouvage: or a bridge |
| 20:38.09 | hypa7ia | lol |
| 20:39.08 | Blackvel | well you could ask file to pay |
| 20:39.23 | Blackvel | he wants us to be be there |
| 20:39.23 | Blackvel | :) |
| 20:39.36 | *** part/#asterisk sezuan (sezuan@port-212-202-202-204.dynamic.qsc.de) |
| 20:39.45 | kapejod | file: you got astricon shares? ;) |
| 20:40.18 | file | eh? no :P |
| 20:40.30 | file | but you should go to Cluecon, it will deepen your understanding of Asterisk |
| 20:41.02 | mishehu | it's a shame there's no Astricon - Middle East going on next month |
| 20:41.07 | mishehu | as I'll be in the mideast |
| 20:41.27 | kapejod | file: i would doubt that, but in general it might be a good idea for * users. :) |
| 20:41.33 | file | it will! |
| 20:41.38 | anthm | mishehu you said you were gonna come if i changed the date =| |
| 20:41.50 | mishehu | anthm: I am going to the one in chicago |
| 20:41.55 | kapejod | file: seen bristuff yet? :) |
| 20:42.02 | mishehu | anthm: in august |
| 20:42.11 | anthm | otherwise known as cluecon ! |
| 20:42.35 | anthm | registration is open |
| 20:42.35 | mishehu | but I was talking about a hypothetical astricon in the mideast... namely, in Israel. |
| 20:42.43 | mishehu | anthm: oh, have to register? |
| 20:43.02 | mishehu | ah topic |
| 20:43.06 | mishehu | didn't read that for a while now |
| 20:43.09 | anthm | well it helps to get you in if you pay for it and reserve a spot => |
| 20:44.24 | mishehu | ah in elk grove too |
| 20:44.24 | hypa7ia | wow, that looks like a cool con |
| 20:44.32 | mishehu | quite convenient for me. |
| 20:44.34 | mishehu | heh |
| 20:44.49 | anthm | no real offense , i'm not into "PC" so I can't resist , the one in isreal could be astrikkkkkaahhhhhn like a jewish guy coughing =p |
| 20:45.08 | vaewyn | hehehe... bummer you pay for hotel when I live so close :} |
| 20:45.09 | mishehu | anthm: what if I don't need the hotel rooms? |
| 20:45.16 | mishehu | I don't seem to get that option on the register |
| 20:45.19 | anthm | you got a sleeping bag? |
| 20:45.21 | PTG123 | yah or don't want to stay at a best western |
| 20:45.39 | hypa7ia | i lived in a best western for a year |
| 20:45.46 | anthm | you dont want to stay in the hotel where the thing is for a dirt cheap rate that comes with free lunch ? |
| 20:45.49 | PTG123 | i don't like to stay at anything less then 4 stars :) |
| 20:45.52 | hypa7ia | it was converted into a univeristy residence :-) |
| 20:46.26 | anthm | it's kind of a inclusive package |
| 20:46.27 | mishehu | anthm: I have a business to run at the same time as teh conference ;-) and I'm only about 10 to 15 minutes away from elk grove |
| 20:46.44 | anthm | well we can probably swing it |
| 20:46.49 | anthm | ask bkw |
| 20:46.57 | vaewyn | 2 stars... 3 stars... 4 stars... screw it... we want to know important stuff... like how many data T1s do they have to feed us? :} |
| 20:47.27 | mishehu | I mean, I don't mind paying to register, but not $650 to 800 for a hotel room I don't need. |
| 20:48.00 | mishehu | vaewyn: where are you? |
| 20:48.20 | vaewyn | mishehu: SW lower Michigan... about 2hrs out if traffic is really bad |
| 20:48.21 | hypa7ia | mishehu, of that probably only $300 is the hotel |
| 20:48.33 | devel | i just updated to the latest cvs (friday), and now dtmf (rfc2833) on my grandstream devices (bt and ht) doesn't work to asterisk (vm, transfer) but works extermally, and works fine for other devices (polycom, sipura, snom). is it just me? |
| 20:48.34 | tzanger | what was the problem with this again? |
| 20:48.35 | tzanger | exten => s,n,GotoIf($[${ARG2} != ""],s,gotac) |
| 20:48.38 | tzanger | I need spaces? |
| 20:48.56 | mishehu | hypa7ia: still, why would I want to pay $300 when I am so close to home? |
| 20:48.58 | anthm | i think we can |
| 20:49.15 | anthm | pull out a discount for no hotel |
| 20:49.27 | mishehu | anthm: that'd be perfect then. |
| 20:49.48 | PTG123 | yah i thik that would be smart |
| 20:49.51 | vaewyn | give bkw_ a foot rubdown and you're probably in even if you don't pay ;P |
| 20:50.32 | vaewyn | oops... shouldn't have said that... now everyone will be freeloading :} |
| 20:50.36 | *** join/#asterisk vvo (~michal@byz182.neoplus.adsl.tpnet.pl) |
| 20:50.40 | mishehu | hypa7ia: maybe apply that same foot rubdown technique to ms. boss... |
| 20:50.56 | mishehu | vaewyn: nah, I'll pass on giving brian a footrub. |
| 20:51.11 | hypa7ia | that would almost certainly get me fired |
| 20:51.17 | PTG123 | anthm, whats the web address for it again |
| 20:51.26 | anthm | www.cluecon.com |
| 20:52.32 | PTG123 | wonder how long it would take to fly myself |
| 20:52.33 | mishehu | and a Star Trek asterisk convention would be AstriKAAAAAAAAAAAHHHHHHHHNNNNNNNNN!! |
| 20:52.45 | mishehu | or was that khan? |
| 20:53.31 | hypa7ia | astrikhaaaaaaaaaaaaaan! |
| 20:53.38 | mishehu | yah that's it. |
| 20:53.52 | hypa7ia | eggcellent smithers |
| 20:54.18 | mishehu | captain kirk at the helm of asterisk |
| 20:54.28 | hypa7ia | captain kram? |
| 20:54.35 | vaewyn | just don't let him sing... |
| 20:54.51 | mishehu | nah nah, commander kram, reminiscent of commander keen |
| 20:54.59 | vaewyn | hypa7ia: bwahahaha |
| 20:55.16 | hypa7ia | that song is... roflcopter |
| 20:55.19 | vaewyn | hypa7ia: That was Spock... but still hilarious you know that reference |
| 20:55.23 | anthm | mishehu, msg bkw for details |
| 20:55.33 | blitzrage | kram: yo |
| 20:55.58 | blitzrage | commander keen rocked |
| 20:56.01 | vaewyn | hwy... none of this shows bkw_ as speaking... :} |
| 20:56.08 | vaewyn | wimp |
| 20:56.12 | hypa7ia | good job HA :-) |
| 20:56.30 | hypa7ia | jill of the jungle >> commander keen |
| 20:56.48 | vaewyn | 'watch out for that tree!' |
| 20:56.56 | vaewyn | ohh wait.. that was George |
| 20:56.57 | vaewyn | :P |
| 20:56.58 | mishehu | hypa7ia: you just like jill of the jungle because she was showing leg |
| 20:57.18 | hypa7ia | nah man. she was my hero when i was a kid |
| 20:57.25 | hypa7ia | cept for the stupid toggle puzzle |
| 20:57.41 | hypa7ia | blasted toggles. |
| 20:59.38 | HA | includes 35 IP500s, 45 Plantronics Headsets, 2 PE750s, 2 410p cards, Some KVM equipment, a couple of UPS backups, 1 ADTRAN MX2800 M13 - Multiplexor, and a pc with 1TB of data storage for recording all the calls to. |
| 21:00.19 | vaewyn | HA: not too shabby :} |
| 21:00.29 | HA | needless to say, we get free shipping on everything we plan to order from tritech since we will be spending about $12000 there. |
| 21:00.32 | hypa7ia | yeah, that's a sweet setup :"-) |
| 21:00.49 | bkw_ | vaewyn, i'll be there |
| 21:00.55 | bkw_ | what do you want to hear me speak about? |
| 21:01.10 | HA | it will be sweet if it all works. i really need some rest and i won't be getting any for the next 2 months at least. |
| 21:01.58 | HA | if it doesn't work...i'm gonna blame it all on bkw_ and leave his contact details when i disappear. |
| 21:02.03 | vaewyn | bkw_: SIP failover ;P (just kidding... I know...) just couldn't believe you would be such a big part of this and stay silent the next time... you at least have to be the Q&A moderator so you can yell 'NEXT!!!' |
| 21:02.11 | *** part/#asterisk jamesewing (~James@pbx2.jsci.net) |
| 21:02.13 | vaewyn | s/next/whole |
| 21:02.47 | harryvv | ha what are you ordering from them |
| 21:03.04 | harryvv | never mind |
| 21:03.12 | vaewyn | Hmm... wonder if I can get work to pay for this... they may not like it since they already payed for VON :{ |
| 21:04.22 | harryvv | ha you work for some kind of call center? |
| 21:04.37 | *** join/#asterisk Blissex (~Blissex@82-69-39-138.dsl.in-addr.zen.co.uk) |
| 21:04.39 | ha | figures that the nicks i feel like at the moment are all 0wn3d by someone else. |
| 21:05.07 | HA | don't hate me, it's just a job. collections. and the people we call, well, they deserve it. |
| 21:05.45 | vaewyn | HA: is ok... I bet our university has sent tons of people your way :P (or a company like yours) |
| 21:06.17 | vaewyn | '26000$ back rent on a dorm room... ?!?!?!?' COLLECTIONS! |
| 21:06.42 | HA | our collectors are actually quite nice about it and most of the people we call realize that it is something they should have taken care of. |
| 21:06.57 | harryvv | ha yea im sure there is alot of nasty people that balk at paying. |
| 21:07.10 | HA | what uni vaewyn? i'll tell the sales guy to give em a call. |
| 21:07.37 | vaewyn | HA: hehehe... Andrews University... SW Lower Michigan |
| 21:07.56 | harryvv | I guess its one of Krams rare moments of rest and he comes here ;) |
| 21:07.56 | HA | that's why the * box will record it all and why we are going to have a Terrabyte of storage dedicated to the recorded calls. |
| 21:08.20 | vaewyn | HA: very good idea |
| 21:08.24 | *** join/#asterisk pfn (500@netblock-66-245-252-239.dslextreme.com) |
| 21:08.31 | harryvv | Ha what dial conf allows that. I think I have seen it before. |
| 21:08.43 | hypa7ia | laters all |
| 21:10.09 | vaewyn | just a monitor command |
| 21:10.20 | harryvv | yea |
| 21:12.43 | rvhi | anyone uses valgrind before? |
| 21:12.56 | rg1 | OK, newbie here - have asterisk with TE110P T1 card - need assistance in setting up just 1 channel for a loopstart signalling |
| 21:14.39 | mishehu | my mind is fried today, but I thought that the loopstart/kewlstart/etc were for analog lines |
| 21:19.15 | *** join/#asterisk kl_pl_01 (kl_pl_01@82.160.252.17) |
| 21:23.52 | *** join/#asterisk cjk (~cjk@80.92.75.120) |
| 21:24.15 | cjk | hi, how long does digium needs to give us the g729 licenses |
| 21:24.19 | cjk | sould be realtime |
| 21:24.59 | jsolares | it takes a year |
| 21:25.01 | file | a day or two |
| 21:25.06 | jsolares | they do a research of you and your company |
| 21:25.11 | jsolares | pfft |
| 21:25.20 | mishehu | cjk: I ordered mine yesterday, and this morning I received them. |
| 21:28.17 | bkw_ | vaewynAFK, he ran |
| 21:30.04 | bkw_ | i got 100000 people msging me at once |
| 21:30.05 | bkw_ | haha |
| 21:30.05 | cjk | ok thank you guys |
| 21:30.21 | MikeJ[Laptop] | bkw, bkw.. |
| 21:30.30 | MikeJ[Laptop] | can you tell me how asterisk works ? |
| 21:30.48 | HA | \/msg jbot help |
| 21:31.05 | HA | \/msg bkw what is the answer to life and can * tell me? |
| 21:31.25 | HA | \/msg bkw will you help us newbies discover the joys of using *? |
| 21:31.35 | HA | \/kill HA Very annoying! |
| 21:31.43 | HA | that about right? |
| 21:32.13 | jsolares | ppl when connecting 2 asterisk with e1/t1 does one need to be pri_net and the other pri_cpe? or pri_net on both? |
| 21:32.39 | *** join/#asterisk doolph (doolph@200.46.148.35) |
| 21:32.51 | doolph | hi |
| 21:34.44 | *** join/#asterisk toot (chris@212.20.250.187) |
| 21:35.16 | HA | the professionals don't know what to make of me. for some reason they just don't understand transdimensional time travel. |
| 21:36.11 | MrClean | Is it possible to undef a global variable? ie. given SetGlobalVar(foo=bar), is it possible to explicitly clean up ${foo} when it's no longer required? |
| 21:36.33 | HA | does asterisk understand transdimensional coordinates in relation to IAXy packet communication? |
| 21:36.47 | *** join/#asterisk asteriskn00b (user223@adsl-68-91-7-226.dsl.tulsok.swbell.net) |
| 21:37.51 | sudhir492 | rg1: are you still struggling with your T1 card ? |
| 21:38.01 | *** join/#asterisk bjohnson (~bjohnson@66.11.188.191) |
| 21:38.37 | sudhir492 | earlier I forgot to tell you that you need to modify zaptel.conf too if you are using T1 instead of PRI |
| 21:38.52 | rg1 | a bit |
| 21:39.02 | sudhir492 | it should read e&m=1-24 instead of bchan=1-24 |
| 21:39.08 | rg1 | ah |
| 21:39.10 | rg1 | let me do that |
| 21:39.17 | rg1 | and do i need to reboot the machine after that change? |
| 21:40.08 | HA | well, i'm off to see the wizard now. just remember that it is all bkw_'s fault. And thanks for the help on the multiplexer sudhir. |
| 21:40.20 | jsolares | rg1: ztcfg -vv |
| 21:40.48 | asteriskn00b | evening all, new asterisk install on centos 3 Whenever I try to start asterisk, it randomly dumps and will not load, I have tried asterisk -vv but I get no "error" messeges. Any ideas on what to look for? |
| 21:40.58 | rg1 | jsolares: |
| 21:41.00 | rg1 | ztcfg: invalid option -- w |
| 21:41.00 | rg1 | Usage: ztcfg [options] |
| 21:41.00 | rg1 | <PROTECTED> |
| 21:41.00 | rg1 | <PROTECTED> |
| 21:41.00 | rg1 | <PROTECTED> |
| 21:41.01 | rg1 | <PROTECTED> |
| 21:41.03 | rg1 | <PROTECTED> |
| 21:41.05 | rg1 | <PROTECTED> |
| 21:41.06 | *** part/#asterisk grolloj (~chatzilla@slim-eth0.horizonlive.net) |
| 21:41.16 | ariel_ | pastebin pastebin please |
| 21:41.18 | jsolares | it's two v's hehe |
| 21:41.23 | rg1 | whoops |
| 21:41.25 | rg1 | thx :) |
| 21:41.52 | sudhir492 | rg1: you dont have to reboot the machine. We are not running windows, are we :-) Just unload and load the driver |
| 21:42.07 | rg1 | boy thats the truth |
| 21:42.11 | toot | hey - any tips on how i can get the music on hold to be louder plz? i'm using default => mp3:/var/lib/asterisk/mohmp3,-z but its still very quiet? |
| 21:42.38 | sudhir492 | toot: record at a higher volume :-) |
| 21:42.39 | rg1 | ok, what that did was made it so that when i dialed into the asterisk on a phone# it now rings - it was giving me a busy signal |
| 21:43.02 | toot | its just off an mp3 - and it's cd quality ;) :) |
| 21:43.51 | outtolunc | http://www.voip-info.org/tiki-index.php?page=Asterisk%20config%20musiconhold.conf |
| 21:43.52 | sudhir492 | rg1: means you are in business :-) beer time ! |
| 21:43.57 | rg1 | sudhir492 - ok, now i need to somehow get a trunk line configured into asterisk to access that outside line - some sort of channel thing |
| 21:44.19 | rg1 | can you help me do that? Not beer time yet - just sugar-water.....but help me and we'll make it Miller time |
| 21:44.28 | jsolares | ehehe |
| 21:44.52 | rg1 | when I'm in CLI mode, i do "show channels" and it gives me this: |
| 21:45.02 | jsolares | pastebin pastebin |
| 21:45.06 | rg1 | asterisk1*CLI> show channels |
| 21:45.06 | rg1 | <PROTECTED> |
| 21:45.06 | rg1 | 0 active channel(s) |
| 21:45.06 | sudhir492 | didnt I already gave you zapata.conf file, (replace signaling=pri_cpe with signaling=em_w) |
| 21:45.08 | rg1 | sorry |
| 21:45.11 | rg1 | will do that now |
| 21:45.14 | jsolares | oh no channels |
| 21:45.22 | alt | E&M? eeeewww ;-) |
| 21:45.36 | sudhir492 | ztcfg -vvv, before running asterisk |
| 21:45.57 | jsolares | meh, i'll go give the definity and the asterisk box a kick and see if the e1 works then |
| 21:45.57 | *** join/#asterisk Ayano (~Ayano@adsl-70-245-190-94.dsl.spfdmo.swbell.net) |
| 21:46.49 | sudhir492 | try ... Dial(Zap/g1/....) path |
| 21:47.00 | sudhir492 | good luck |
| 21:48.28 | rg1 | hey sudhir492 - |
| 21:48.37 | rg1 | for signalling you want me to use em_w? |
| 21:48.38 | *** join/#asterisk cursor (~kevin@andromeda.office.cursor.biz) |
| 21:48.56 | toot | hmm still can't get it louder - but cheers for the link outtolunc |
| 21:49.39 | sudhir492 | rg1: yes, if you are using T1 trunk from telco |
| 21:49.54 | *** part/#asterisk vphirke (~vphirke@66.9.62.66) |
| 21:50.09 | outtolunc | toot: on that page loud => mp3:/var/lib/asterisk/mohmp3 |
| 21:50.12 | sudhir492 | rg1: actually, you can ask your telco to confirm what kind of signalling they expect |
| 21:50.16 | outtolunc | note the LOUD |
| 21:50.25 | *** join/#asterisk santiago (~santiago@63.245.86.165) |
| 21:50.43 | *** join/#asterisk sandnigg0r (~niggerplz@66-55-197-254.gwi.net) |
| 21:50.54 | toot | yep but loud is just the class - its the mp3 or quietmp3 that defines the volume |
| 21:51.04 | doolph | who got installed a menu that ask you what extension want to transfer |
| 21:51.22 | toot | i'm using default => mp3:/var/lib/asterisk/mohmp3 so its already using the 'non quiet' version |
| 21:51.34 | outtolunc | true |
| 21:51.37 | toot | however its still not loud enough, granted its louder than using quietmp3 :) |
| 21:51.50 | cursor | Try a louder mp3 :-) |
| 21:52.08 | toot | na tis not the mp3 - i'm a tad confused on this one. but hmm |
| 21:52.11 | harryvv | mm my music on hold is not working. |
| 21:52.17 | toot | no odds really :) |
| 21:52.19 | sudhir492 | rg1: provided e&m wink does not work. In USA that is the most common |
| 21:52.35 | *** join/#asterisk SarahEmm (~sarahemm_@MTL-ppp-158713.qc.sympatico.ca) |
| 21:52.50 | rg1 | sudhir492 - i'm not using a T1 trucn from telco |
| 21:52.59 | rg1 | i'm using a loopstart pots line |
| 21:53.18 | santiago | hi, i have a problem with some sipuras 2000, after some time, suddenly the phone doesn't work, but the extension is registerd and when it is called, a extX is ringing message is displayed in the CLI. anyone has had this problem? |
| 21:53.34 | harryvv | I think my wife would love it putting U2 as music on hold ;) |
| 21:53.36 | sudhir492 | rg1: Then how are you interfacing with te110 card? |
| 21:53.53 | rg1 | through an ADIT 600 |
| 21:53.53 | *** join/#asterisk loud (ariel@omfg.wtf.no) |
| 21:54.11 | toot | gone for the Stone Roses myself. naturally i'll get some legal music before i go live. ahem |
| 21:54.16 | rg1 | what is the signalling for loopstart/POTS? |
| 21:54.18 | doolph | who got installed a menu that ask you what extension want to transfer |
| 21:54.34 | sudhir492 | e&m will still work. channel bank is properly configured too |
| 21:54.55 | sudhir492 | make sure that channel bank is properly configured too |
| 21:55.49 | sudhir492 | on a trunk interface, you dont do loopstart. |
| 21:56.12 | sudhir492 | good luck |
| 21:56.16 | jsolares | i give up |
| 21:56.26 | outtolunc | toot: mpg123 -g n set audio hardware output gain |
| 21:56.59 | rg1 | sudhir492 - well, the one line is actually NOT a trunk line |
| 21:57.16 | Dishwasha | stupid question, how do I set a variable to the return value of a function? |
| 21:57.17 | *** join/#asterisk Jas_Williams (~jas_willi@host217-43-100-176.range217-43.btcentralplus.com) |
| 21:57.33 | rg1 | just for the heck of it, do you know what the signallaing is for loopstart? |
| 21:58.11 | Dishwasha | like if ChanIsAvail returns a 0, I want to assign 0 to foo |
| 21:58.43 | jsolares | fxs loop start = fxsls; fxo loop start = fxols |
| 21:59.21 | *** join/#asterisk darkskiez (~mhb@host-84-9-89-165.bulldogdsl.com) |
| 21:59.41 | toot | is there an asterisk gui call manage app for windows about anywhere? |
| 21:59.56 | toot | google gives me lots of pages talking about one but .. |
| 22:01.04 | outtolunc | http://www.voip-info.org/tiki-index.php?page=Asterisk%20GUI |
| 22:01.20 | rg1 | jsolares: what would signalling=fxo_ls do for/to me? |
| 22:01.27 | toot | just got to that site. aye :) |
| 22:01.46 | *** join/#asterisk jeremywhiting (~jeremy@70-56-99-134.slkc.qwest.net) |
| 22:01.55 | Dishwasha | seriously, when I read http://www.voip-info.org/wiki-Asterisk+variables all it talks about is assigning direct values to a variable. Is there some global return variable I can use to find out the return value? |
| 22:02.48 | jsolares | rg1: no idea... |
| 22:04.09 | *** join/#asterisk Dishwasha (~chatzilla@208.251.32.70) |
| 22:04.15 | Dishwasha | Wow, that was weird |
| 22:04.24 | Nuxi | Dishwasha, you may be forced to use agi. |
| 22:04.27 | Dishwasha | Did anybody by chance answer my question while I was gone? |
| 22:04.40 | Dishwasha | Nuxi: Really?? |
| 22:04.44 | jsolares | yes but i forgot the answer |
| 22:04.49 | Dishwasha | jsolares: hardy har |
| 22:05.11 | Dishwasha | You'd think you could use return values in a dialplan |
| 22:05.46 | Nuxi | You might be able to use the if construct. |
| 22:06.51 | Dishwasha | If I use gotoif, gotoif expects 0 for false and anything else for true, but many functions return -1, so it would be misleading if I tested for true |
| 22:07.28 | cursor | Which command do you want to trap the rc from? |
| 22:08.08 | Dishwasha | CheckAvailChan and CheckGroup |
| 22:10.06 | *** join/#asterisk |Vulture| (~Vulture@c-69-180-67-228.hsd1.fl.comcast.net) |
| 22:10.13 | |Vulture| | anyone using a PRI --> spandsp? |
| 22:10.44 | Dishwasha | so how would I go about "trapping" an rc? |
| 22:12.40 | cursor | I've never needed to do that from a dialplan |
| 22:12.46 | *** join/#asterisk lancey (Shady@support.net1.cc) |
| 22:12.48 | lancey | hi guys |
| 22:13.00 | lancey | anyone have a safe_asterisk script for FreeBSD? |
| 22:13.06 | Dishwasha | I'm doing something special : ) |
| 22:13.35 | *** join/#asterisk tangel (tangel@64.135.81.8) |
| 22:13.36 | darkskiez | anybody got a PRI line in the UK? i'm confused about "select services" and their charges. |
| 22:13.53 | tangel | when * picks up my house line on an incoming call the caller receives silence until either pick-up or VM |
| 22:14.03 | tangel | is there a command to place in extensions so that the ring is heard by the caller? |
| 22:14.13 | |Vulture| | tangle: add ,r to your Dial |
| 22:14.18 | Jas_Williams | tangel r |
| 22:14.35 | darkskiez | tangel: what line type is it? |
| 22:14.43 | rg1 | jsolares - also, when I try to add a trunk on the web setup, it asks me for a Zap Identifier (trunk name) - and when I hover over that heading, it says |
| 22:14.59 | jsolares | web setup... ewww |
| 22:15.11 | tangel | darkskiez, zap |
| 22:15.41 | tangel | ,r would go on the Dial() line at the end? |
| 22:15.51 | darkskiez | tangel: change your extensions.conf to not Answer the phone first. |
| 22:15.53 | Nuxi | Dishwasha, as I said, agi. It seems really odd that you can't get a return value, but ... |
| 22:16.16 | darkskiez | tangel: otherwise the callers will be billed for unanswered calls etc. |
| 22:16.32 | tangel | darkskiez, do what now? |
| 22:16.39 | Dishwasha | Nuxi: Np, I think I can do it with n+101 stuff |
| 22:16.39 | tangel | i have privacy manager enabled |
| 22:16.59 | tangel | err. i guess zapateller is what really answers the call |
| 22:17.08 | darkskiez | tangle: Ah, I see, so its on those calls, then add r to the dial, yes. |
| 22:17.18 | tangel | like this: exten => s,4,Dial(${RIALTOALL},15,Ttmr) ? |
| 22:17.30 | SarahEmm | anyone know what the AGI 'TDD MODE' command actually does? (http://www.voip-info.org/wiki-tdd+mode) |
| 22:19.01 | *** join/#asterisk bonez39 (~aint@drjones.dsl.xmission.com) |
| 22:19.54 | darkskiez | SarahEmm: for deaf folks |
| 22:19.57 | Nuxi | TDD sends text to channels supporting it |
| 22:21.38 | tangel | where does the r go? |
| 22:21.56 | darkskiez | tangel: that was right |
| 22:22.03 | tangel | it still doesn't work |
| 22:22.33 | darkskiez | extensions reload? |
| 22:22.39 | tangel | i restarted * |
| 22:22.59 | Nuxi | reboot the windows box next to the * box. |
| 22:23.03 | tangel | what i really would like is * not to pick up the analog call unless someone answers a ringing extension |
| 22:23.11 | tangel | hehe.. good idea nuxi |
| 22:23.20 | tangel | i'm sure my outsources would've recommended exactly that |
| 22:23.27 | tangel | outsourcers |
| 22:23.32 | darkskiez | tangel: dont use zapateller and privacy manager then. |
| 22:23.47 | tangel | it seems like such a good idea though =| |
| 22:23.49 | tangel | i'll try without |
| 22:24.16 | darkskiez | I'm amazed zapateller works if you havent Answered |
| 22:24.16 | |Vulture| | any spandsp users here? |
| 22:24.51 | Jas_Williams | ~seen cursor |
| 22:24.53 | jbot | cursor is currently on #asterisk (36m 16s). Has said a total of 3 messages. Is idling for 12m 13s |
| 22:24.53 | tangel | is there a pastebot? |
| 22:25.01 | Jas_Williams | ;-0 |
| 22:25.18 | darkskiez | ~paste |
| 22:25.19 | jbot | paste is probably see http://paste.husk.org |
| 22:25.27 | Jas_Williams | ~pastebin |
| 22:25.28 | jbot | hmm... pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca |
| 22:25.30 | cursor | "is probably see" ? |
| 22:25.49 | SarahEmm | nuxi: okay.. what kind of channels support it though? |
| 22:26.10 | tangel | http://paste.husk.org/3271 |
| 22:26.21 | Jas_Williams | cursor, Vulture wants help on spandsp |
| 22:26.24 | SarahEmm | darkskiez: hehee. i'm aware of what a TTY is, just wondering what the * function does, to see if it's something i'd need to hack more code into to make useful, or if it's working now and i just can't find any info on it |
| 22:26.44 | |Vulture| | hmm I think its working with HEAD zaptel drivers now |
| 22:26.47 | |Vulture| | doing another test |
| 22:26.57 | |Vulture| | the PRI+spandsp is iffy |
| 22:26.58 | cursor | Jas: I don't use that, so I'm probably no help |
| 22:27.20 | Jas_Williams | k |
| 22:27.43 | tangel | darkskiez, does that make sense? |
| 22:28.02 | Jas_Williams | |Vulture|, Its normally sync problems that breaks spandsp check your sync sources |
| 22:28.19 | Nuxi | SarahEmm, I don't really know. But you can tell by the return values if it is currently supported on the channel you are on. |
| 22:28.22 | lancey | bye |
| 22:28.27 | darkskiez | tangel: not to me. |
| 22:28.52 | tangel | what would you expect it to look like? |
| 22:29.11 | darkskiez | sorry, something else was in my head |
| 22:29.14 | tangel | i really don't want to pick up the call until either someone answers a ringing extension or it times-out to voicemail |
| 22:29.37 | tangel | i would love for the regular analog lines/phones in the house to continue ringing in the meantime |
| 22:29.43 | darkskiez | dunno if zapateller answers the call if answer is set and there is a callerid. |
| 22:30.07 | doolph | what is Queues |
| 22:30.20 | darkskiez | tangel: perhaps you need a wait(2) in there first to allow the callerid to be received. |
| 22:30.28 | SarahEmm | nuxi: okay. to me based on the lack of anything other than that function it looked like it was something to be implemented later, that didn't exist yet.. i don't see any way to transmit/receive text from a channel... |
| 22:30.47 | tangel | caller id seems to be working.. and privacymanager only kicks in when it should |
| 22:30.54 | *** join/#asterisk loud (~ariel@omfg.wtf.no) |
| 22:30.57 | SarahEmm | it'd be easier to do it natively in * than transfer a call to a TTY modem and do it that way. |
| 22:31.05 | Nuxi | Most channels do not support sending text. I think iax does, but I'm not sure. |
| 22:31.15 | SarahEmm | okay |
| 22:31.20 | SarahEmm | i'm talking about sending to/from the PSTN.. |
| 22:31.24 | SarahEmm | over POTS |
| 22:31.27 | SarahEmm | so it's unlikely :) |
| 22:31.38 | *** join/#asterisk Jedirl (ircap@154.Red-217-127-168.pooles.rima-tde.net) |
| 22:31.39 | Jedirl | Hello |
| 22:31.43 | Cheetah | hello :D |
| 22:31.43 | darkskiez | tangel: privacy manager may have the callerid by the time zapateller has executed. |
| 22:32.05 | darkskiez | tangel: set verbose to a highish number and paste the log |
| 22:32.21 | tangel | i expect zapateller would play the tones if it didn't have caller id in the first step |
| 22:32.29 | Jedirl | could anyone help me to size my platform? as it involves no transcoding, I'm a bit lost on sizing it because I'm quite new to asterisk |
| 22:33.00 | Jas_Williams | tangel, do a NoOp(${CALLERID}) before the Zapateller to fid out if caller ID is there at that time |
| 22:33.37 | Jedirl | a P-III 1Ghz with 1Gbyte of RAM and SCSI storage, with a 4-E1 interface card, how many simultaneous channels could handle in a IVR and switching evironment, provided there's no transcoding and everything would be a-law? |
| 22:34.04 | |Vulture| | Jedirl: nice setup... too bad the proc isn't a bit more |
| 22:34.05 | Jedirl | could I handle the full 4-E1s with such a machine? |
| 22:34.17 | |Vulture| | Jedirl: you dropping it to any phones or just IVR? |
| 22:34.32 | |Vulture| | well.. what I mean is will there be any bridged calls |
| 22:34.33 | Jedirl | |Vulture|: prepay application |
| 22:34.40 | Jedirl | first there will be an IVR |
| 22:34.47 | darkskiez | Jedirl: could be iffy from what i've read. if you can afford the card, you should be able to afford a better machine. |
| 22:34.50 | |Vulture| | ah okay they it will bridge to another ZAP |
| 22:34.50 | *** join/#asterisk mattbridges (~mattb@mattbridges.plus.com) |
| 22:35.03 | Jedirl | which handles authentication and credit checking and then it will bridge the call to another channel |
| 22:35.33 | |Vulture| | should work since its all zap |
| 22:35.35 | doolph | how can I playback sounds |
| 22:35.37 | |Vulture| | but could be iffy |
| 22:35.42 | mattbridges | Hello all, can anyone help me to get caller ID working with my X100P in the UK? I've patched it but still can't get it working. |
| 22:35.47 | Jedirl | darkskiez: of course, but it's a rock-solid server and i know it's stable, it's proven, so if it could handle the load I trust it more than I would trust a 800$ 3'2Ghz P4 machine |
| 22:35.47 | |Vulture| | Background();Playback() |
| 22:35.58 | doolph | from cli |
| 22:35.59 | doolph | ? |
| 22:36.07 | doolph | i want to test some sound files |
| 22:36.21 | Jedirl | |Vulture| so you think it should handle the full 4 E1s without problems? thankyou very much :D |
| 22:36.24 | |Vulture| | Jedirl: I would say it should as long as you don't bring any sip peers online |
| 22:36.33 | Jedirl | no, no, never, no VoIP involved |
| 22:36.37 | Jedirl | just plain a-law |
| 22:36.50 | |Vulture| | yea since its just switching its pretty low intensive on the cpu |
| 22:37.04 | Cheetah | i have a question regarding codecs... right now i was testing the voip stuff using the gsm codec. however this one has a very crappy quality compared to normal phone quality. can somone suggest some alternative codes? bandwith is not a problem |
| 22:37.04 | Jedirl | if there was any VoIP, I have a hardware-based VoIP gateway which handles 6 E1s |
| 22:37.47 | Jedirl | Cheetah: I love *-law, but 64kbps may be too much :D |
| 22:37.53 | |Vulture| | damn... |
| 22:38.13 | Cheetah | Jedirl, too much for what? we are just running an ISDN gateway ;) |
| 22:38.52 | Jedirl | too much bandwidth for some kind of internet users |
| 22:39.20 | |Vulture| | I use ilibc |
| 22:39.21 | Cheetah | well, all traffic goes over 100 base t ethernet |
| 22:39.24 | |Vulture| | sounds great to me |
| 22:39.28 | |Vulture| | much better than gsm |
| 22:39.39 | Jedirl | gsm over IP sounds very bad |
| 22:39.47 | |Vulture| | agreed |
| 22:39.48 | Jedirl | I guess GSM in mobile phones is 'enhaced' by many filters |
| 22:39.55 | Jedirl | because it doesn't sound that bad |
| 22:40.00 | darkskiez | Jedirl: very few mobile phones use that gsm codec |
| 22:40.07 | |Vulture| | yea I think old ones do |
| 22:40.12 | darkskiez | they use an enhanced one |
| 22:40.15 | Jedirl | oh |
| 22:40.16 | |Vulture| | the new phones sound much better |
| 22:40.26 | Jedirl | hehe |
| 22:40.28 | Jedirl | ok |
| 22:40.30 | darkskiez | |Vulture|: yeh its quite dated afaik. |
| 22:40.40 | Cheetah | so, what do you suggest to use as codec if you have 100Mbit uplink to the server and require best quality and relatively high compatibility with voip devices? |
| 22:40.56 | Jedirl | Cheetah: *-law |
| 22:41.06 | |Vulture| | my friend was using a 4 year old phone just upgraded and I can hear him perfect now, before it sucked sounded like bad gsm |
| 22:41.13 | Cheetah | allow=alaw? |
| 22:41.17 | Cheetah | allow=ulaw? |
| 22:41.24 | Jedirl | any of them |
| 22:41.32 | Cheetah | where is the difference? |
| 22:41.40 | |Vulture| | ulaw=US, alaw=UK.... basically correct? |
| 22:41.46 | Jedirl | alaw europe, ulaw us |
| 22:41.51 | Cheetah | ah.. okay ;) |
| 22:41.59 | Jedirl | like T1 and E1 |
| 22:42.00 | Jedirl | :D |
| 22:42.00 | |Vulture| | but it doesn't matter sounds the same |
| 22:42.10 | |Vulture| | yea its when it goes over a D-Chan |
| 22:42.22 | Dishwasha | Is SetGroup/CheckGroup independant or shared between contexts? |
| 22:42.33 | |Vulture| | man... KVMoIP is a great invention |
| 22:42.47 | Jedirl | yup, absolutely great |
| 22:42.58 | Jedirl | but I still preffer sun machines with a serial console |
| 22:43.01 | |Vulture| | been using StarTech... cheap and affective |
| 22:43.18 | *** join/#asterisk loick (~loick@APuteaux-151-1-29-223.w82-124.abo.wanadoo.fr) |
| 22:43.26 | |Vulture| | apparently serial on linux can cause problems with the PRI cards from digium |
| 22:43.32 | Jedirl | really??? |
| 22:43.42 | Jedirl | uhm |
| 22:43.43 | |Vulture| | yea there is a whole list |
| 22:43.46 | |Vulture| | run zttest and check it |
| 22:43.47 | Jedirl | hehe |
| 22:43.54 | Jedirl | what's zttest? |
| 22:44.04 | |Vulture| | you want nothing under 99.85 over 5min |
| 22:44.09 | |Vulture| | its in zaptel dir |
| 22:44.14 | |Vulture| | when you compile |
| 22:44.24 | |Vulture| | it tests PCI communications with the card |
| 22:44.24 | Jedirl | I haven't ever installed a zaptel hardware device |
| 22:44.38 | |Vulture| | if its bellow that it can cause drops, pops, and ugly things |
| 22:45.07 | Jedirl | uhm |
| 22:45.21 | Jedirl | interesting |
| 22:45.40 | |Vulture| | but if my Dell SC420s can support TE110P cards... I would think any mobo can lol |
| 22:45.41 | tangel | it definitely seems like zapateller is broken |
| 22:45.46 | tangel | it picks up the call even if there's callerid |
| 22:45.48 | Jedirl | hehe |
| 22:45.58 | |Vulture| | tangel: HEAD or 1-0 or stable? |
| 22:46.00 | Jedirl | Dell+Digium = problems I've herad, right? |
| 22:46.08 | tangel | 1.0.5 |
| 22:46.16 | tangel | i haven't updated * in a very long time |
| 22:46.18 | |Vulture| | Jedirl: nothing I haven't been able to patch for |
| 22:46.23 | Jedirl | :) |
| 22:46.25 | Jedirl | great |
| 22:46.36 | Jedirl | By the way, any AGI or FastAGI developer here? |
| 22:46.37 | |Vulture| | Jedirl: I run only dell hardware so.. |
| 22:46.47 | Jedirl | I have a question |
| 22:46.49 | Jedirl | :) |
| 22:47.24 | |Vulture| | best way to get help here is to just ask :) |
| 22:47.24 | Jedirl | I don't know - if I want to do SetLanguage from my AGI or FastAGI script, what should I do? set a variable? or execute SETLANGUAGE? or what? |
| 22:47.47 | tangel | how cani write a dialplan such that if there's no callerid, then execute zapateller? |
| 22:48.26 | |Vulture| | tangel: GotoIf() |
| 22:48.44 | tangel | i see examples for specific numbers, how can i match any caller id? |
| 22:49.22 | *** join/#asterisk newmedian (~crowlther@Quebec-HSE-ppp230300.qc.sympatico.ca) |
| 22:49.22 | |Vulture| | tangel: well if the Callerid="" |
| 22:49.23 | |Vulture| | then... |
| 22:49.46 | tangel | good point.. ah.. example 3 on voip-info looks good |
| 22:49.50 | *** join/#asterisk KernelSanders (~brian@ool-43516aa2.dyn.optonline.net) |
| 22:50.06 | tangel | do those out-of-order tones really work or am i wasting time on zapateller for nothing? =] |
| 22:50.22 | rg1 | help - in zaptel.conf - what is the loopstart symbol vs. say "e&m" |
| 22:50.51 | SarahEmm | that's the type of signalling used |
| 22:51.03 | rg1 | yes, but what is it for "loopstart"? |
| 22:51.07 | SarahEmm | e&m is usually used on links between PBXs and such |
| 22:51.10 | SarahEmm | oh. |
| 22:51.13 | SarahEmm | one sec :) |
| 22:51.18 | rg1 | thanks........ |
| 22:51.28 | Nuxi | jedirl, you would exec setlanguage from an agi. |
| 22:52.08 | Jedirl | Nuxi: how can I do it? |
| 22:52.11 | Nuxi | which language toolkit are you using? |
| 22:52.19 | Jedirl | uhm |
| 22:52.21 | rg1 | any luck there sarahemm? |
| 22:52.26 | Jedirl | language toolkit? |
| 22:52.28 | SarahEmm | looks like fxsls or fxols |
| 22:52.47 | SarahEmm | is that what you were looking for? i'm slightly confused by the q :) |
| 22:52.54 | Jedirl | I just want to play es/, fr/ or en/ sound files on the IVR depending on the user's input in the IVR |
| 22:52.55 | SarahEmm | (you may also want ks for kewlstart which is ls with disconnect supervision) |
| 22:53.17 | Jedirl | what's a language toolkit? |
| 22:53.25 | Nuxi | jedirl, if you are not using a toolkit it would look like this: EXEC SETLANGUAGE language |
| 22:53.35 | Jedirl | ok, great |
| 22:53.40 | Jedirl | thanks, Nuxi |
| 22:53.47 | Jedirl | by the way, what's a language toolkit? |
| 22:53.55 | Nuxi | are you using a perl php java python or c library? |
| 22:54.12 | Jedirl | Java |
| 22:54.16 | Jedirl | FastAGI with Java |
| 22:55.14 | Jedirl | the one in http://asterisk-java.sourceforge.net/ |
| 22:55.14 | Nuxi | I'm not familiear with the java stuff, but there should be an interface to the exec command. |
| 22:55.21 | Jedirl | yes, there is |
| 22:56.10 | Jedirl | Other implementations of AGI have anything like a 'language toolkit' for developing multi-language IVRs? |
| 22:57.18 | Nuxi | What I meant by language toolkit was a library in some programming language. I guess that was pretty easy to get confused. |
| 22:57.30 | Jedirl | ok ok :D |
| 23:03.00 | *** join/#asterisk crash3m (crash3m@crash3m.user) |
| 23:03.16 | crash3m | anyone know how to reset the user/admin passwords on an IP500, when you dont know the current user/admin passwords? |
| 23:11.58 | SarahEmm | hmm, chan_zap evidently supports TDD mode... |
| 23:13.25 | *** join/#asterisk iq (~iq@70-59-166-10.omah.qwest.net) |
| 23:21.00 | *** join/#asterisk ariel_ (~Ariel@dsl-20-177.cofs.net) |
| 23:21.28 | ariel_ | Hello everyone |
| 23:21.48 | santiago | hi, i have a problem with some sipuras 2000, after some time, suddenly the phone doesn't work, but the extension is registerd and when it is called, a extX is ringing message is displayed in the CLI. anyone has had this problem? |
| 23:23.09 | ariel_ | is the sipura behind a nat? |
| 23:23.17 | Dishwasha | BTW guys, I just implemented a dial plan that will dial any of 6 lines on my cisco 7960 phone based on whether a call has already been placed on that line within a Dial group |
| 23:24.12 | Dishwasha | So far it's only customized for 1 phone with set extensions, maybe sometime I'll virtualize it more so it can be expanded across multiple phones with variable numbers of lines |
| 23:25.37 | *** join/#asterisk morale (russell@secure.deadbolt.ca) |
| 23:26.08 | morale | anyone live in alberta, can you recommend a decent VoIP provider? |
| 23:26.33 | |Vulture| | Dishwasha: kinda like a virtual key-system? |
| 23:26.49 | Dishwasha | |Vulture|: not familiar with the terminology |
| 23:27.05 | |Vulture| | nortel norstar system for example |
| 23:27.17 | Dishwasha | Not familiar with that either |
| 23:27.20 | |Vulture| | where all users can see the total lines on the system on their phone and their current states |
| 23:27.49 | |Vulture| | well what you just explained is a great dialplan for a small office wanting to replace their current system |
| 23:28.42 | *** join/#asterisk Cardoe (~Cardoe@Cardoe.developer.gentoo) |
| 23:28.46 | Dishwasha | It's more of a hack for SIP phones that don't report SIP availability properly (may be an * bug too) |
| 23:29.10 | morale | i need to find a SIP provider which i can have the login/password |
| 23:29.39 | Dishwasha | Or, a dial plan that allows you to dial only 1 line out of 6 on a phone based on what's in use without dialing every available extension on the phone |
| 23:29.48 | Dishwasha | S/Or/And |
| 23:29.55 | Cardoe | Having a firewall/SIP issue |
| 23:30.01 | Cardoe | I have my asterisk box behind a firewall |
| 23:30.09 | Cardoe | I'm forwarding tcp/udp 20000-21000 to it |
| 23:30.14 | Cardoe | rtp.conf lists 20000-21000 |
| 23:30.21 | Cardoe | I'm also forwarding 5060 and 5061 to it |
| 23:30.23 | |Vulture| | the 7960 should report |
| 23:30.31 | Cardoe | my client is behind a firewall as well |
| 23:30.39 | Cardoe | forwarding 5060 to the client |
| 23:30.39 | |Vulture| | because Flash Panel sees their states |
| 23:30.42 | Cardoe | and I still can't connect |
| 23:30.48 | Cardoe | anything else I'm missing? |
| 23:30.53 | Dishwasha | |Vulture|: If I have line 1 in use and call waiting disabled, ChanIsAvail reports 102 as available |
| 23:31.21 | Dishwasha | er, 102 is my extension for line 1, oops |
| 23:32.34 | Dishwasha | and then I get a busy on line 102 when I do a Dial |
| 23:32.52 | Dishwasha | I'm using 1.0.7 though, could be a new bug in * |
| 23:33.10 | |Vulture| | strange |
| 23:33.21 | Dishwasha | verily |
| 23:34.07 | *** part/#asterisk makhtar (~ageller@mail.bulletinnews.com) |
| 23:35.23 | *** join/#asterisk rkioko (~rkioko@196.200.25.253) |
| 23:36.33 | *** join/#asterisk pdracevich (~bob@smtp.aucklandtax.co.nz) |
| 23:36.56 | newmedian | Anyone in NZ? What's the status of the emergency at present? |
| 23:37.09 | pdracevich | hi all, I have just update my test asterisk box to the latest CVS-head and some of my extensions now no longer work? |
| 23:37.19 | crash3m | newmedian: the emergency? |
| 23:37.22 | crash3m | what emergency? |
| 23:37.26 | pdracevich | exten => 706,1,Dial(SIP/21311919@x.x.x.x) |
| 23:37.32 | pdracevich | HELP please |
| 23:38.41 | newmedian | crash3m: take a look at the front page of the NZ Herald using www.pressdisplay.com (select newspaper by Country, New Zealand). Flooding, houses washed away, cars floating away... just reading now. |
| 23:39.15 | *** join/#asterisk duke149 (~ca07b782@cgiirc-gateway.ictoadd.com) |
| 23:39.17 | crash3m | damn |
| 23:39.31 | *** join/#asterisk GyrosGeier (~geier@port-212-202-79-27.dynamic.qsc.de) |
| 23:39.33 | duke149 | peoples |
| 23:39.36 | GyrosGeier | hi |
| 23:40.27 | duke149 | got a quick question for anyone free to help. dont need to know how just if its possible.. |
| 23:41.06 | duke149 | is it possible to use Asterisk with our existing phone system and link them in so the users cant tell which one they are actually going through? |
| 23:42.01 | |Vulture| | yup |
| 23:42.17 | |Vulture| | like you want to use * for LD? |
| 23:42.49 | pdracevich | Any one here from Auckland? |
| 23:42.50 | |Vulture| | newmedian: by what? |
| 23:43.50 | newmedian | |Vulture| take a look at the front page of the NZ Herald using www.pressdisplay.com (select newspaper by Country, New Zealand). Flooding, houses washed away, cars floating away... |
| 23:43.53 | duke149 | basically our current phone system is old and crappy. we need to add more phones to it but its like very expensive for the add on cards. so i was hoping that I could add an Asterisk server to the side. have all the new phones go through Asterisk then link into the old system |
| 23:45.39 | newmedian | Someone could phone Hamish from sipserve.co.nz; I believe sipserve is in New Lynn, Auckland |
| 23:45.42 | NewSole | hello world |
| 23:46.44 | |Vulture| | newmedian: wow I have been so busy I haven't turned on the news |
| 23:47.21 | duke149 | You all from New Zealand? |
| 23:47.46 | *** join/#asterisk abatista (~Ariel@dsl-20-177.cofs.net) |
| 23:47.46 | newmedian | We're from all over. |
| 23:48.09 | duke149 | heh |
| 23:48.18 | newmedian | A fair number of Canadians, Americans, UK, NZ, other countries ... it's a global channel |
| 23:48.36 | duke149 | yeah |
| 23:49.14 | ariel_ | argh network problems today..... |
| 23:49.34 | duke149 | anyone ever used any Wireless VoIP handsets? |
| 23:49.58 | pdracevich | I have duke149 |
| 23:50.21 | duke149 | whats it like pdracevich. Are the handsets you used sturdy? |
| 23:50.27 | newmedian | pdracevich, aren't they head heating headache inducing things that haven't quite got the technology right yet? |
| 23:50.52 | duke149 | we have wireless ericson system here but the dudes in the warehouse drop them all the time. they break after 2-3 drops and they cost like $500 to replace/repair |
| 23:50.56 | pdracevich | I have used to the Zyxel pw2000 (they make good paperweights now) |
| 23:51.01 | duke149 | lol |
| 23:51.10 | newmedian | Better to do ATA + cordless, no? |
| 23:51.22 | pdracevich | The best one I have used is the FT177 handset |
| 23:51.47 | duke149 | whats the distance of them tho? they need to go arround 500 meters and thats with 5 base stations on our site. |
| 23:51.54 | Qwell | newmedian: cheaper, certainly |
| 23:52.09 | *** part/#asterisk santiago (~santiago@63.245.86.165) |
| 23:52.47 | duke149 | this is another reason im looking at Asterisk for wireless voip.. |
| 23:52.53 | Qwell | When one wants to get docs on a commercial PBX (Avaya for instance), where would one go? |
| 23:53.05 | pdracevich | They can do that, I have had them up around 700meters |
| 23:53.26 | duke149 | cool |
| 23:54.37 | pdracevich | whats the other eason Duke149? |
| 23:55.14 | duke149 | we need more phones. we have lots of people sharing 1 ext. like 3 phones on one ext. |
| 23:55.57 | *** part/#asterisk SarahEmm (~sarahemm_@MTL-ppp-158713.qc.sympatico.ca) |
| 23:57.47 | duke149 | we are soon to be rolling out wireless accross the whole site for our barcode system so it would be a good opportunity to get wireless voip going also. |
| 23:58.09 | *** join/#asterisk justnulling2 (justnullin@ool-18bab443.dyn.optonline.net) |
| 23:58.38 | justnulling2 | where can i get latest sip firmware for cisco 7960? |
| 23:58.44 | crash3m | from cisco |
| 23:58.48 | Qwell | Do you really want a large office of assholes on wireless phones walking around? :) |
| 23:59.01 | crash3m | Qwell: unwired assholes? no thanks |
| 23:59.08 | Qwell | crash3m: indeed |
| 23:59.18 | duke149 | lol |
| 23:59.50 | duke149 | well they are already unwired the current system is just crap |
| 23:59.52 | newmedian | Of course there's always something like this http://www.jdteck.com/product/phprepeater.htm but then that's not Asterisk or VOIP. |
