00:00.04 | Qwell | shmaltz: sounds like a perfect time to upgrade then |
00:00.12 | Qwell | by the time you're done compiling, he'll be off the phone |
00:00.16 | drray | my bad |
00:00.30 | shmaltz | Qwell, that aint an option for tonigh :( |
00:00.45 | shmaltz | I first have to redo my DP to work nicely with 1.2 |
00:00.59 | shmaltz | anybody know of any solution that works like app_valetparking? |
00:01.06 | *** join/#asterisk Ridgeback (n=Ridgebac@104.243.8.67.cfl.res.rr.com) |
00:01.10 | Qwell | app_valetparking does |
00:01.10 | drumkilla | shmaltz: oej is working on one |
00:01.14 | shmaltz | app_valetparking doesn't compile on 1.2 |
00:01.17 | Goshen | perfect, I changed the name of vm-intro.gsm so it couldn't find it, but it still makes a loud screech, is there any way to remove that? |
00:01.22 | shmaltz | drumkilla, that is great news |
00:01.39 | *** join/#asterisk Sedorox (i=brandon@smartserv/cna/Sedorox) |
00:01.40 | drumkilla | he has a branch in svn for it |
00:01.48 | drray | isn't the screech ADSI? |
00:01.51 | Goshen | 'BEEP' 'comedian mail' |
00:01.51 | drumkilla | i have no clue how far along it is |
00:01.55 | *** join/#asterisk santiago (n=santiago@208.195.215.160) |
00:02.01 | Goshen | adsi? |
00:02.12 | alephcom | Goshen: http://www.nathanpralle.com/software/ast_masterlist.html |
00:02.29 | Ridgeback | hello |
00:02.48 | alephcom | they don't list that file. Sorry. |
00:03.03 | Goshen | it doesn't list it as playing a file when it makes the beep |
00:03.57 | shmaltz | wow, that trucking a55hole is still on the phone |
00:04.16 | drumkilla | shmaltz: restart when convenient is your friend |
00:04.56 | drray | 4554013 |
00:05.02 | drumkilla | shmaltz: sounds like a plan! |
00:05.17 | drumkilla | barge and drop it into a meetme so we can all listen |
00:05.28 | alephcom | Quick question... I have a friend who has slapped together a php interface that interfaces with the -realtime database. It is a gui but is used mostly/only to edit the -realtime database. Is there any interest in something like this? |
00:05.34 | Ridgeback | just curious will my old pre 1.2 config files move right over to 1.2? I read the changes doc, but it didn't seem to say there would be much to change. |
00:05.45 | shmaltz | drumkilla, after all he is calling south america, where lots of these so called suck operations exists (I took the name from the list) :) |
00:05.56 | drumkilla | Ridgeback: they should, yes |
00:05.56 | shmaltz | drumkilla, that is realy a good plan :) |
00:05.59 | shmaltz | :P |
00:06.02 | drumkilla | Ridgeback: you may see some deprecation warnings, though |
00:06.06 | Ridgeback | drumkilla thanks |
00:06.08 | drumkilla | but the same options should still work ... |
00:06.25 | Ridgeback | drumkilla: ok cool ill give it a try on the weekend |
00:06.28 | ManxPower | Ridgeback, In theory they should work just fine. |
00:06.31 | *** join/#asterisk dwmw2 (i=ctrlprox@baythorne.infradead.org) |
00:06.40 | drumkilla | *should* :) |
00:06.50 | shmaltz | Ridgeback, what out for the j option |
00:06.54 | Ridgeback | ManxPower: good deal. was just worried my dundi stuff would get all dorked up |
00:06.54 | drumkilla | there was way too much time in between 1.0 and 1.2 |
00:06.55 | ManxPower | If you are using ENUMLookup and not prefixing the query with a + then that will break |
00:06.58 | shmaltz | that is the only think you realy want to know about |
00:07.00 | drumkilla | so it's quite a massive difference in code |
00:07.24 | ManxPower | drumkilla, and yet, 1.2 still doesn't have a SIP jitterbuffer...... |
00:07.44 | Ridgeback | have you guys played with the IAX encryption yet? |
00:07.49 | drumkilla | I didn't know you could have jitter in SIP signalling |
00:07.57 | drray | SSL tunneling? |
00:08.12 | *** join/#asterisk kokey (n=ubunture@201.153.63.79) |
00:08.19 | *** join/#asterisk dwmw2 (i=ctrlprox@baythorne.infradead.org) |
00:08.23 | Ridgeback | drrayi thought 1.2 was to have AES encryption on IAX: |
00:08.32 | shmaltz | this guy is out of his mind, look at this: |
00:08.33 | shmaltz | http://pastebin.com/453291 |
00:08.56 | drumkilla | robin_sz: ot dpes |
00:08.59 | drumkilla | crap |
00:09.00 | Goshen | I keep getting this after upgrading to 1.2.1 from 1.0.7 |
00:09.01 | Goshen | Dec 7 17:08:00 WARNING[28113]: chan_sip.c:9601 handle_response_register: Got 200 OK on REGISTER that isn't a register |
00:09.02 | drumkilla | Ridgeback: it does |
00:09.05 | drumkilla | I totally messed that up |
00:09.39 | Ridgeback | drumkilla: cool stuff cant wait to try it out |
00:09.55 | ManxPower | Goshen, your extensive search of the mailinglist archives did not help? |
00:10.05 | *** join/#asterisk dwmw2 (i=ctrlprox@baythorne.infradead.org) |
00:10.08 | ManxPower | drumkilla, Please step away from the beer. 8-) |
00:10.15 | drumkilla | ManxPower: none this evening :) |
00:10.35 | darwin_35 | the only issue I have had is this stupid spandsp issue |
00:10.36 | drumkilla | but I was at school all day ... same thing |
00:10.40 | drumkilla | Thanks MstlyHrmls !!! |
00:10.42 | darwin_35 | and I will get it working |
00:10.49 | MstlyHrmls | obviously you need one :-) |
00:10.49 | darwin_35 | grrr |
00:10.53 | drumkilla | no kidding |
00:11.08 | ManxPower | darwin_35, Got the 1.2 version of rx_fax, tx_fax now? |
00:11.08 | drumkilla | oh wait!! |
00:11.13 | drumkilla | I have one left! |
00:11.28 | ManxPower | darwin_35, We all do, dude, we all so. |
00:11.29 | drray | with a happy ending |
00:11.29 | darwin_35 | yes |
00:11.35 | drumkilla | darwin_35: me too ... my back is *killing* me for some reason :( |
00:11.37 | Goshen | ManxPower: Just that the message is coming back too early or late or something like that, but doesn't give a fix |
00:11.48 | darwin_35 | mine is spazming |
00:11.48 | *** join/#asterisk dwmw2 (i=ctrlprox@baythorne.infradead.org) |
00:11.49 | Goshen | ManxPower: It didn't give me that message before the upgrade |
00:11.57 | *** join/#asterisk Flauto (n=zhao@c-24-14-197-214.hsd1.il.comcast.net) |
00:12.10 | ManxPower | Goshen, I don't think the older version cared. |
00:12.28 | darwin_35 | I have to compile spandsp 0.0.2pre21c |
00:12.34 | ManxPower | Goshen, you don't have something silly like pedantic=yes do you? |
00:12.35 | darwin_35 | but I have to patch it first |
00:12.50 | Goshen | ManxPower: in sip.conf? |
00:12.56 | darwin_35 | doing it now |
00:13.26 | ManxPower | darwin_35, I can tar up my asterisk, asterisk-sounds, zaptel, libpri, spandsp, with all the patches to load the apps in asterisk. Oh and NXFaxDetect is included. |
00:13.31 | Goshen | ManxPower: I didn't put predantic=no in sip.conf because I read that the default is no |
00:13.33 | *** join/#asterisk dwmw2 (i=ctrlprox@baythorne.infradead.org) |
00:13.49 | ManxPower | Goshen, the default is no |
00:14.12 | Goshen | no, I don't have predantic=yes in my sip.conf |
00:15.18 | *** join/#asterisk dwmw2 (i=ctrlprox@baythorne.infradead.org) |
00:15.29 | darwin_35 | manx that would rock |
00:16.00 | drumkilla | shmaltz: maybe he left it off hook just to piss you off |
00:16.01 | darwin_35 | no need for zaptel or libpri |
00:16.30 | test34 | Can someone control asterisk from the internet and have asterisk call him and another person with VOIP 3-way to save on long distances ? |
00:16.39 | *** join/#asterisk SPAD (n=chatzill@specdl05.cul.columbia.edu) |
00:16.55 | *** join/#asterisk upsite (n=upsite@wls.swh.uni-halle.de) |
00:17.02 | *** join/#asterisk dwmw2 (i=ctrlprox@baythorne.infradead.org) |
00:18.12 | ManxPower | darwin_35, hold on |
00:18.45 | *** join/#asterisk dwmw2 (i=ctrlprox@baythorne.infradead.org) |
00:18.52 | Goshen | any ideas for getting rid of the Dec 7 17:16:01 WARNING[28113]: chan_sip.c:9601 handle_response_register: Got 200 OK on REGISTER that isn't a register |
00:19.07 | Goshen | Predantic=yes is not set |
00:20.13 | ManxPower | Goshen, file a bug, then someone can tell you why it's not a bug 8-) |
00:20.19 | Goshen | lol |
00:20.31 | Qwell | and lose 2 karma points |
00:20.31 | ManxPower | I wasn't joking. |
00:20.31 | *** join/#asterisk dwmw2 (i=ctrlprox@baythorne.infradead.org) |
00:20.32 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
00:20.36 | Qwell | dwmw2: stop that |
00:20.40 | puzzled | evening all |
00:20.45 | Goshen | well, it was misconfigured a couple hours ago |
00:20.49 | Qwell | or not |
00:20.54 | Goshen | perhaps it is stuff coming back from that... |
00:20.59 | Goshen | some kind of crazy time warp |
00:21.12 | Goshen | who knows, I will deal with it again tomorrow if it keeps it up |
00:21.17 | shmaltz | Drumkilla, he hung up now, he didn't leave it off hook because the system charges him, 1 sec checking............................ |
00:21.59 | alephcom | test34: Short answer, yes. Long answer it might be a bit of work. |
00:22.09 | ManxPower | darwin_35, http://www.fnords.org/~eric/asterisk.tar.bz2 |
00:22.14 | *** join/#asterisk dwmw2 (i=ctrlprox@baythorne.infradead.org) |
00:22.24 | ManxPower | This is the beginning of the -BTEL fork of Asterisk |
00:22.47 | mog_work | ? |
00:22.48 | test34 | alephcom, do you know of anybody that did it and posted a howto on the web ? ;) |
00:22.55 | mog_work | another.... |
00:23.21 | Qwell | test34: seems easy |
00:23.35 | Qwell | write a simple script to drop a call file... |
00:23.54 | mog_work | or use manager |
00:23.59 | *** join/#asterisk dwmw2 (i=ctrlprox@baythorne.infradead.org) |
00:24.05 | test34 | Qwell, the problem is I'm new to asterisk.. |
00:24.11 | tzanger | mog_work: any good at threading code? I have a question |
00:24.13 | Qwell | test34: nothing a bit of reading won't fix |
00:24.19 | mog_work | decent |
00:24.25 | Qwell | mog_work: anybody around that can "fix" dwmw2 here? |
00:24.26 | mog_work | i dont always do it right thouhg.... |
00:24.31 | Qwell | and -dev |
00:24.42 | mog_work | i dont have ops anymore.... |
00:24.48 | alephcom | test34: have you looked in the wiki? I know it's been done but.... |
00:24.51 | tzanger | mog_work: http://pastebin.ca/32897 |
00:25.06 | shmaltz | Durmkilla, that phone call will cost him $37.80 |
00:25.12 | tzanger | mog_work: I'm getting alarms more or less every 100ms as expected, but every so often my alarms come every 15-20ms for a few, then back to 100ms |
00:25.31 | tzanger | mog_work: now I can see it SKIPPING now and again if it couldn't get the lock... but it's not |
00:25.47 | test34 | Qwell, I will look into dropping a call file then |
00:25.52 | *** join/#asterisk dwmw2 (i=ctrlprox@baythorne.infradead.org) |
00:25.54 | test34 | alephcom, didnt find anything yet |
00:26.05 | alephcom | test34: Call files are actually quite handy and easy to use. |
00:26.15 | ManxPower | mog_work, I want to take 1.2.3, put in RTP jitterbuffer, the self timing RTP patch, spandsp, the NV*Detect stuff. |
00:26.28 | *** join/#asterisk Pete_Largo (n=PeteLarg@225-196.35-65.tampabay.res.rr.com) |
00:26.47 | Pete_Largo | tzanger: you around? |
00:26.56 | tzanger | yeah |
00:27.22 | Pete_Largo | it's been working like a champ since that day. Thanks again for all your help :) |
00:27.29 | tzanger | Pete_Largo: no problem at all |
00:27.32 | tzanger | glad it's working :-) |
00:27.37 | *** join/#asterisk dwmw2 (i=ctrlprox@baythorne.infradead.org) |
00:27.45 | Pete_Largo | hehe me too! |
00:28.04 | drumkilla | mog_work: "fix" ? |
00:28.10 | Pete_Largo | that threading problem was making all kinds of trouble |
00:28.14 | Qwell | drumkilla: kb dwmw2, would ya? :p |
00:28.27 | drumkilla | will do when he joins again |
00:28.32 | Qwell | he's been on and off about 15 times now, in the last 15 minutes |
00:28.49 | *** join/#asterisk Zodiacal (i=hehehe@bdsl.66.14.242.199.gte.net) |
00:29.07 | Qwell | drumkilla: -dev too... |
00:29.10 | *** join/#asterisk anthm (n=anthm@000-435-220.area4.spcsdns.net) |
00:29.10 | *** mode/#asterisk [+o anthm] by ChanServ |
00:29.18 | docelm0 | whadup? |
00:29.21 | *** join/#asterisk dwmw2 (i=ctrlprox@baythorne.infradead.org) |
00:29.25 | ManxPower | THERE HE IS! |
00:29.25 | *** mode/#asterisk [+b *!*=ctrlprox@*.infradead.org] by drumkilla |
00:29.26 | *** kick/#asterisk [dwmw2!n=russell@asterisk/developer-and-stable-maintainer/drumkilla] by drumkilla (drumkilla) |
00:29.27 | Pete_Largo | docelmo! hey |
00:29.31 | Zodiacal | Hi, can i use an asterisk box as just a voice mail system? i have an existing office Merdian pbx? |
00:29.42 | drumkilla | got him! |
00:29.43 | Zodiacal | and have the phones allready etc.. |
00:29.45 | darwin_35 | manx you there |
00:29.46 | Qwell | <3 <3 |
00:29.53 | shmaltz | well, after all this I'm happy to report: |
00:29.55 | shmaltz | jumping channel 69 (adit fxs) from span 3 to channel 73 of span 4 (adit fxo), confirmed that the new (for me used for someone else) fxo card I got for $200.00 realy works, with callerid and hangup detection :) |
00:29.57 | Pete_Largo | good job drumkilla! |
00:30.12 | drumkilla | yay |
00:30.40 | drumkilla | I'm drinking a Grolsch right now, hehe |
00:30.49 | Qwell | wtf is that? |
00:30.59 | bweschke | good beer! |
00:31.04 | anthm | shuddup? |
00:31.05 | drumkilla | Dutch |
00:31.12 | bweschke | drumkilla!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!1 |
00:31.12 | anthm | you dont know what grolsch is ? |
00:31.21 | tzanger | t's an ok beer |
00:31.34 | ManxPower | There is no beer better than Guinness. |
00:31.37 | Zodiacal | any ideas? |
00:31.46 | docelm0 | Whadup?!!? |
00:31.47 | darwin_35 | yes there is |
00:31.51 | anthm | ahh the only thing better than a db or os zealot is a beer one!!! |
00:31.55 | puzzled | ManxPower: they really brainwashed you didn't they :) |
00:31.57 | bweschke | agree... unless of course it's a black and tan w/ Bass and guinness. :) |
00:31.58 | bweschke | brilliant! |
00:32.03 | darwin_35 | HappyDog Brew from RockBottom Brewry |
00:32.03 | Qwell | anthm: you missed the former two earlier |
00:32.16 | drumkilla | Qwell: isn't that a daily occurance? |
00:32.16 | docelm0 | Supe Pete? |
00:32.25 | Qwell | drumkilla: well...yeah |
00:32.30 | ManxPower | puzzled, I sampled random Dutch and Belgian beers when I was in Europe. Guinness is better. |
00:32.33 | drumkilla | that's why I tend to ignore this channel :-p |
00:32.34 | anthm | it's ok another one will come around in an hour or 2 |
00:32.42 | drumkilla | bweschke: nice |
00:32.46 | Qwell | anthm: yeah...we all know postgresql sucks |
00:32.50 | Pete_Largo | docelmo, what's with the 0 ? |
00:32.59 | puzzled | ManxPower: sampling is good. I do that too |
00:33.04 | bweschke | qwell: oh no u didn't |
00:33.06 | bweschke | :) |
00:33.21 | drumkilla | I'll tell you what else sucks, Linux ... and *BSD |
00:33.37 | anthm | now your on the right track |
00:33.42 | bweschke | lol |
00:33.46 | ManxPower | Users suck. |
00:33.51 | drumkilla | totally |
00:33.54 | alephcom | lol Amen! |
00:33.54 | bweschke | ding! ding! ding! |
00:33.57 | *** part/#asterisk SPAD (n=chatzill@specdl05.cul.columbia.edu) |
00:34.00 | anthm | it's computers in general and everything associated with them that sucks =D |
00:34.09 | drumkilla | I'm so against users, that I don't even use the software I work on |
00:34.17 | docelm0 | Just to make it a bit different for all my locations.. |
00:34.40 | Pete_Largo | lol, docelm0, just so no-one can find you! |
00:34.48 | shmaltz | anthm, you got my /msg? |
00:35.00 | docelm0 | Well msg both of me.. I will answer sooner or later |
00:35.04 | anthm | which? |
00:35.08 | anthm | was it in an email ? |
00:35.13 | shmaltz | ok, both |
00:35.17 | shmaltz | an email and msg |
00:35.25 | shmaltz | email to your yahoo account |
00:35.28 | Pete_Largo | docelm0 - that reminds me of a poem... |
00:35.44 | docelm0 | ohh lord.. |
00:35.46 | anthm | i am on an cheezy irc that doesnt mak a big deal ouyt of pm i must have missed it |
00:35.47 | Pete_Largo | roses are red, violets are blue, I'm a scizophrenic, and so am I |
00:35.52 | anthm | i do see 1 that says "did you get my message" |
00:36.08 | shmaltz | anthm, exactly that /msg i meant |
00:36.13 | shmaltz | but I'm asking about the email |
00:36.16 | shmaltz | did you get it? |
00:36.22 | shmaltz | from shmaltz@gmail.com |
00:36.22 | anthm | let me look |
00:36.26 | shmaltz | thanks |
00:36.28 | *** join/#asterisk EriSan (n=erisan@81-174-42-85.f5.ngi.it) |
00:36.40 | *** join/#asterisk Godsey (n=admin@pdpc/supporter/sustaining/Godsey) |
00:37.00 | upsite | hey is someone using call pickup via the quickdial-buttons on a snom (220) |
00:37.14 | shmaltz | upsite, I'm sure that yes |
00:37.20 | upsite | hmm |
00:37.29 | upsite | mine is'nt working |
00:37.43 | darwin_35 | Manx I will chat with you tomarrow |
00:37.44 | upsite | its sending a notify instead of an invite |
00:37.46 | darwin_35 | going home |
00:37.50 | *** part/#asterisk darwin_35 (n=darwin35@sta-208-139-193-162.rockynet.com) |
00:37.52 | shmaltz | upsite, do a sip debug in asterisk press that button and pastebin your cli output |
00:38.21 | upsite | ok |
00:38.36 | Qwell | I should hack up the extra softkeys on a 7940/60/70 with chan_sccp |
00:38.56 | shmaltz | Qell, you using sccp? |
00:39.02 | Qwell | shmaltz: yeah, I love it |
00:39.03 | shmaltz | and it works nicely? |
00:39.08 | Qwell | works great |
00:39.14 | shmaltz | why is it better than sip? |
00:39.20 | Qwell | he'll be adding my reload and realtime patches soon... |
00:39.26 | Qwell | shmaltz: sccp is just...sexy |
00:39.38 | Qwell | and runs on 7970s ;] |
00:39.42 | shmaltz | Qwell, that will just not let me do any work :P |
00:39.50 | anthm | does it not work anymore? |
00:39.53 | shmaltz | 7970 doesn't support sip? |
00:39.56 | Qwell | nope |
00:39.58 | ManxPower | puzzled, my version of "sampleing" is to tell the bartender, "give me one I've not had before" |
00:39.59 | shmaltz | anthm, nope not in 1.2 |
00:40.31 | anthm | starbucks is daft they want 6 bux an hr for net access good think i have a sprint card sheesh |
00:40.45 | Qwell | $6/h? damn |
00:40.47 | puzzled | Qwell: which chan_sccp? the one from http://chan-sccp.berlios.de ? |
00:40.52 | Qwell | puzzled: yeah |
00:41.03 | shmaltz | anthm, it's only round $3 here (NJ) |
00:41.04 | Qwell | it really does work very well |
00:41.16 | Qwell | we're gonna go production with it |
00:41.19 | mog_work | hey what happened to that fork up above |
00:41.22 | mog_work | the btel one |
00:41.26 | anthm | is head from this week close enuf to 1.2 for you ? |
00:41.45 | puzzled | Qwell: I heard that one is working really well. yesterday I made an rpm of the latest release so I'm prepared when I get a few 7961's |
00:41.57 | Qwell | rpm? eww |
00:42.06 | Qwell | puzzled: I'd repackage it in a few days.. |
00:42.18 | Qwell | fun new stuff should be coming "Real Soon" now |
00:42.27 | puzzled | ok will track it. thanks |
00:42.37 | shmaltz | anthm, the problem is that I cannot take a chance to run head on this machine, I have to run stable, if HEAD from this week will allow it to compile on 1.2 then it's good enough for me |
00:42.51 | Qwell | shmaltz: I think thats what he was asking |
00:42.56 | anthm | i mean compat wise |
00:43.07 | shmaltz | I have been running HEAD on it until now, and I had too many crashes |
00:43.30 | anthm | what doesnt work about it ? |
00:43.44 | shmaltz | I'll give you the output of astxs -install in a sec |
00:44.08 | anthm | my output is a newline |
00:44.14 | anthm | so you probably want my latest copy |
00:44.52 | shmaltz | here: |
00:44.54 | shmaltz | http://pastebin.com/453349 |
00:45.10 | shmaltz | anthm, I guess so, I have the one from bpxfreeware |
00:45.22 | Qwell | You need to include stdio.h before file.h :p |
00:45.25 | anthm | yah when did you dl it ast |
00:45.25 | anthm | last |
00:45.27 | Qwell | simplest fix ever |
00:45.39 | anthm | id try it fresh |
00:45.50 | anthm | you know you can astxs urls to .c files too |
00:45.52 | shmaltz | today was when I downloaded it last |
00:46.09 | shmaltz | anthm, I know, someone Tony told me about it, a while ago ;) |
00:46.51 | orlok | hmm, wtf. my 7940's red light is staying on |
00:46.53 | drumkilla | anthm: really? that's hot |
00:47.01 | Qwell | orlok: the voicemail light? |
00:47.25 | orlok | i think of it as the ring light |
00:47.29 | orlok | but yes, that would be it |
00:47.44 | Qwell | So...you have voicemail |
00:47.45 | shmaltz | anthm, just donloaded it again, and still the same problem, FYI I did: |
00:47.46 | shmaltz | wegt http://www.pbxfreeware.com/app_valetparking.c |
00:47.58 | *** join/#asterisk hhoffman (n=hhoffman@tor/session/x-2209c9215ad3a296) |
00:48.07 | hhoffman | hi |
00:48.23 | shmaltz | orlok, just change the sip setting mailbox to a mailbox that doesn't have any messages, leave a message in that mailbox, then erease it |
00:48.28 | shmaltz | hi hhoffman |
00:48.29 | anthm | aha |
00:48.37 | anthm | the good one wasnt in the url |
00:48.50 | shmaltz | anthm, so whre is it???????????? |
00:49.01 | anthm | try again |
00:49.39 | anthm | it was a stale copy |
00:50.10 | *** join/#asterisk YoMama (n=tchen@pcp02689850pcs.roylok01.mi.comcast.net) |
00:50.20 | YoMama | hey all |
00:51.01 | YoMama | if i'm calling in thru a sip proxy...why would I not be able to hear the * server's AA, but the * can hear me? |
00:51.11 | shmaltz | anthm, ya da man :) |
00:51.13 | YoMama | which is opposite to what i thought the problem would be |
00:51.16 | orlok | ahh, asterisk wasnt running |
00:51.27 | Zodiacal | anyone know if i can use asterisk as just a voice mail system with my existing merdian pbx and phones? |
00:51.31 | YoMama | 'cause if it were a firewall problem....then i'd be able to hear * |
00:51.42 | YoMama | Zodiacal: yes...theoretically |
00:52.13 | shmaltz | thank you everyone, gtg now |
00:52.15 | anthm | YoMama I think the official name for it is RTP/NAT HELL |
00:52.17 | shmaltz | thank you anthm |
00:52.18 | Zodiacal | yomama or would it be even easier to replace the pbx as well as the voice mail system with an * box? but i would like to use my existing phones.... |
00:52.22 | anthm | np |
00:52.24 | YoMama | Zodiacal: look in asterisk-wiki...there's some examples of it in there |
00:52.35 | Zodiacal | yomama wikipedia? |
00:52.40 | YoMama | anthm: i don't think it's a firewall problem... |
00:52.46 | Zodiacal | or does * have its own wiki |
00:52.50 | Zodiacal | got the url off hand? |
00:52.55 | YoMama | Zodiacal: www.voip-info.org |
00:53.02 | drumkilla | there is a wiki built into asterisk! |
00:53.12 | Zodiacal | yomama Thank You! |
00:53.15 | anthm | someone cant hear someone on rtp it's a port mismatch or the wrong ip or something along those lines |
00:53.40 | Zodiacal | yomama i don't wanta use voip tho.. just existing verizon business plan |
00:57.14 | upsite | shmaltz : http://pastebin.com/453361 |
00:57.43 | YoMama | Zodiacal: u don't haveta |
00:57.59 | upsite | thats happning when i press the function key while it's blinking to show me that a call is ringing on the other phone |
00:57.59 | YoMama | Zodiacal: u could use asterisk only as a voicemail server...how many extensions u got? |
00:58.12 | YoMama | Zodiacal: or i should say..how many voicemail ports u think u need? |
00:58.22 | hhoffman | what is the "_" for before a extension number? I'm trying to find it in the book but can't |
00:58.33 | YoMama | hhoffman: means it's a pattern |
00:58.48 | YoMama | hhoffman: www.asteriskdocs.org...read the TFOT book..it explains it all |
00:59.04 | *** join/#asterisk ldnblk (n=Just@optbom1.uk.access.vodafone.net) |
00:59.20 | Zodiacal | yomama 9 stations |
00:59.24 | Zodiacal | 6 phone lines |
00:59.38 | YoMama | Zodiacal: ah..u could get away with a TDM400 card |
00:59.44 | YoMama | maybe 2-3 ports for voicemail |
00:59.49 | hhoffman | YoMama: thanks... is the TFOT book the same one as the OReilly book? |
00:59.50 | Zodiacal | yomama at least one voice mail port... i think thats what we have now. only one can record voice mail at a time i think |
00:59.53 | Zodiacal | but that systme is so old and its dieing |
00:59.56 | YoMama | get some FXO ports on it |
00:59.58 | Zodiacal | systme = system |
01:00.05 | YoMama | Zodiacal: then get 2 or 3 |
01:00.15 | YoMama | 2-3 FXO |
01:00.23 | Zodiacal | will that work with my phones. my phones have lcd screens and line buttons and function buttons etc.. |
01:00.27 | Zodiacal | i.e. paging.. |
01:00.38 | YoMama | the problem isn't asterisk..it's setting up meridian to properly signal asterisk to use the VM system properly |
01:00.45 | Zodiacal | ic.. |
01:00.50 | Zodiacal | welp i hate the merdian thing |
01:00.55 | Zodiacal | maybe i should just replace it too |
01:00.56 | Zodiacal | ? |
01:01.00 | Zodiacal | will * work with my phones? |
01:01.01 | YoMama | Zodical: the handsets aren't the problem...it's the programming of your PBX |
01:01.18 | Zodiacal | what if i use * as the pbx |
01:01.19 | YoMama | no it won't..unless your system's phones are analog..but the chances of that are almost zero |
01:01.29 | Zodiacal | i belive they are digital |
01:01.39 | Zodiacal | cuz i can't connect a regular phone to the jack in the wall :P |
01:01.41 | YoMama | nope then..those handsets only work wiht your PBX |
01:01.49 | Zodiacal | plus they have station names etc on the lcd |
01:02.03 | YoMama | 9 stations..that's a small one |
01:02.04 | Zodiacal | darn |
01:02.30 | dokhench | zodiacal: you can use them if you put a sip gateway in front.. lemme find the link. |
01:03.07 | YoMama | Zodiacal: if u were to replace the whole thing...do u have more than one ethernet run per station? |
01:03.16 | Zodiacal | yes |
01:03.50 | YoMama | k..then u could replace the system with $150 handsets |
01:03.55 | YoMama | $150 x 9 |
01:03.58 | YoMama | so a little over a grand |
01:04.07 | YoMama | a box that u were already going to run asterisk on |
01:04.15 | *** join/#asterisk m160858 (n=jsaenz@200.89.12.46) |
01:04.16 | YoMama | how many incoming lines do u have? DID? |
01:04.20 | Zodiacal | 6 |
01:04.37 | m160858 | hi, i want to connect one asterisk with other by iax |
01:04.47 | YoMama | Zodiacal: u could either get a T1 card and a channel bank..or two TDM400 cards with FXO ports on it |
01:05.03 | YoMama | of course the TDM400's would be cheaper..but won't offer the same amount of expandability |
01:05.07 | Qwell | or a TDM2400P with 2 FXO modules |
01:05.14 | YoMama | oh right |
01:05.16 | Zodiacal | yomama think it would be cheaper if i just got another startalk voice mail system..? |
01:05.16 | YoMama | yeah..those too |
01:05.17 | Qwell | That's what I'd recommend |
01:05.24 | YoMama | Qwell: i always forget about that card :) |
01:05.27 | dokhench | Zodiacal, you can use this to reuse the existing phones http://www.citel.com/products/handset_gateways/index.asp#CITELlink |
01:05.59 | YoMama | Zodical: then finally...make sure you do a gap analysis between how your current system is used and what * can offer...* does almost everything that a traditional PBX does, but the handsets make all the difference..some do intercom and paging..some don't..it all depends |
01:06.05 | YoMama | anyway |
01:06.05 | Zodiacal | dokhench coolness. do you know how much those cost off hand? |
01:06.15 | *** part/#asterisk m160858 (n=jsaenz@200.89.12.46) |
01:06.29 | *** join/#asterisk nassy (n=nassy@207-38-197-201.c3-0.wsd-ubr1.qens-wsd.ny.cable.rcn.com) |
01:06.37 | YoMama | have fun...i gotta go pay some attention to my fiancee before she gets wise and figures out i'm a lazyass dork |
01:06.43 | YoMama | bbl |
01:06.58 | dokhench | Zodiacal, no idea |
01:08.28 | Zodiacal | dokhench Thank You! i have some stuff to think about now |
01:10.06 | *** join/#asterisk Zach^^ (n=Zachary@65.121.244.130) |
01:10.33 | dokhench | Zodiacal, $3120.00 |
01:10.33 | Zach^^ | hello i have so problems trying to get out bound calls to go using voipjet |
01:10.53 | *** join/#asterisk jtodd (n=jtodd@ti.fox-den.com) |
01:11.17 | Zodiacal | yikes |
01:11.26 | Pete_Largo | Zach^^ are you still using the old server? they changed servers a few weeks ago. problem with their East Coast server I think... |
01:11.47 | Qwell | Zodiacal: look at the tdm2400p |
01:11.47 | Zach^^ | hmm i just setup my accout yesterday |
01:11.56 | Qwell | You'd need 2 FXO modules (which would give you 8 ports) |
01:11.58 | Pete_Largo | oh |
01:12.05 | Pete_Largo | then you are probably using the new server :) |
01:12.13 | Zodiacal | qwell will that work with my phones tho? |
01:12.25 | Qwell | I actually wasn't listening to what you're gonna do |
01:12.26 | Zodiacal | :P |
01:12.28 | Qwell | Zodiacal: what phones? and, phones use fxs anyhow |
01:12.34 | Qwell | if they're digital...no |
01:12.42 | dokhench | Qwell, not his. he has pbx phones. |
01:12.58 | Qwell | how many phones, how much you looking to spend total? |
01:13.20 | dokhench | if it is 15 or so phones, it'd be cheaper to just get new phones. |
01:13.23 | Zodiacal | i was hoping i could spend around 1000, 1500.. like for a * box and some cards.. |
01:13.35 | Pete_Largo | it does support the Nortel Norstar line |
01:13.37 | *** join/#asterisk p1tst0p (n=admin@82-38-106-54.cable.ubr03.donc.blueyonder.co.uk) |
01:13.41 | Qwell | Zodiacal: how many phones? |
01:13.49 | Zodiacal | ~9 stations |
01:13.52 | Zodiacal | 6 lines |
01:14.09 | p1tst0p | would it be possible, to display current channels in use through php, would this require use of the Manager API ? |
01:14.11 | Qwell | one tdm2400p with 2 fxo, and 9 decent IP phones...could put you at about ....1500-2k? |
01:14.11 | Zach^^ | anyone able to reach sixtel support? |
01:14.27 | Qwell | (plus the server, of course) |
01:14.51 | Zodiacal | im getting the idea that its probably best just to replace my existing startalk voice mail. thats the thing thats failing.. |
01:15.01 | Qwell | what |
01:15.05 | Qwell | 'll it cost to replace that? |
01:15.16 | Zodiacal | dunno exactly |
01:15.19 | Qwell | it may make fiscal sense in the short run, but... |
01:15.26 | Qwell | You'd be stuck with a junk pbx. heh |
01:15.34 | Zodiacal | i just wanted to see what * had to offer in features compaired to price... |
01:15.43 | Qwell | Zodiacal: can't beat free |
01:15.56 | Zodiacal | :P but the hardware costs bux |
01:16.09 | Pete_Largo | Zodiacal: what model is your voicemail? |
01:16.23 | Zodiacal | 1 sec |
01:16.50 | Zodiacal | Startalk Mini (NOR-015) ? |
01:16.53 | Zodiacal | is that right? |
01:16.56 | Zodiacal | i just have some manuals for it |
01:17.03 | Zodiacal | it doesn't say much on the box it self.. just startalk |
01:17.15 | Zodiacal | Startalk Flash |
01:18.48 | Pete_Largo | you could probably get one refurbished for around 1500 |
01:18.55 | Pete_Largo | that's a guesstimate |
01:19.01 | dokhench | Zodiacal, the digium hardware is inexpensive compared to some of the stuff out there. pbx can cost $$$. |
01:19.15 | Qwell | Zodiacal: at that price...you could get new hardware with * |
01:20.26 | Zach^^ | i get all circuits are busy no when i try to call out |
01:20.53 | *** join/#asterisk m160858 (n=jsaenz@200.89.12.46) |
01:20.55 | p1tst0p | anyone know if if/how i can display current channels in use through a php interface, would this require use of the Manager API ? |
01:20.59 | m160858 | hi |
01:21.24 | m160858 | excuse, but how i can register one asterisk local with other in the WAN? |
01:21.48 | m160858 | on iax |
01:21.58 | *** join/#asterisk wundaboy (n=asdf@c-67-164-107-68.hsd1.or.comcast.net) |
01:23.09 | m160858 | newly, all are sleeping? :) |
01:23.55 | wundaboy | so, this may sound dumb but when im editing extconfig.conf where do i put my mysql credentials? |
01:24.07 | Qwell | wundaboy: in the mysql config |
01:24.35 | wundaboy | Qwell: which file is that? |
01:24.37 | wundaboy | mysql.conf? |
01:24.55 | wundaboy | oh |
01:24.58 | wundaboy | res_mysql.conf |
01:24.58 | Qwell | I'm just guessing here, but res_config_mysql? |
01:24.59 | wundaboy | mybad |
01:25.02 | p1tst0p | <PROTECTED> |
01:25.04 | p1tst0p | yeh |
01:28.51 | *** join/#asterisk Seldon1975 (n=someone@gatekeeper.radintl.com) |
01:29.00 | Seldon1975 | hi all |
01:29.03 | m160858 | anybody? |
01:29.40 | Seldon1975 | I have my asterisk server up with two Polycom Soundpoint 501s on the network - can someone tell me how to assign them Extension numbers in * config |
01:29.47 | Zach^^ | i get all circuits are busy no when i try to call out using voipjet... anyone help? |
01:31.40 | m160858 | i want to know register one local asterisk with other asterisk in the WAN |
01:31.47 | m160858 | it's possible? |
01:32.20 | alephcom | It definitely should be. |
01:32.30 | nassy | Zach^^: im new to asterisk and i get that also. i have a VoIP provider also. it happens to me when i dont dial the correct number. ie, i forget to dial a 1 infront of the number |
01:32.48 | m160858 | how? |
01:33.01 | alephcom | What are the details of the setup? |
01:33.11 | m160858 | yes |
01:33.22 | Zach^^ | nassy who is your voip provider? |
01:33.40 | nassy | telasip |
01:33.46 | m160858 | i want to connect my ATA's to my asterisk local with ullaw |
01:34.05 | alephcom | That should not be a problem |
01:34.13 | SkramX | anyone connected asterisk with ventrilo |
01:34.16 | m160858 | then, this local asterisk connect to the other asterisk in USA with g729 |
01:34.21 | SkramX | I have a user who avidly wants this.. |
01:34.24 | alephcom | right |
01:34.42 | m160858 | and then that asterisk in USA call with gsm |
01:34.51 | Zach^^ | nassy you have a inbound voip? |
01:34.52 | nassy | SkramX: whats ventrilo, a VoIP provider? |
01:35.03 | m160858 | but, i don't have idea |
01:35.06 | SkramX | a team-spreak time thing |
01:35.17 | m160858 | you're have some examples |
01:35.19 | nassy | Zach^^: yeah, they are my in and out to pstn |
01:35.32 | Zach^^ | SkramX hey.. i signed up for voipjet and sixtel... and need a little help.. you think u can help? |
01:35.36 | m160858 | i don't know to create in iax.conf |
01:35.42 | SkramX | Zach^^: possibly |
01:35.49 | m160858 | the configuration for this |
01:35.50 | SkramX | Im me or open a support ticket with my company. |
01:35.53 | *** join/#asterisk christo (n=chris@brezhnev.spiration.co.uk) |
01:35.56 | christo | evening all |
01:37.34 | hhoffman | hiya christo |
01:37.40 | Zach^^ | SkramX sent u a msg |
01:37.59 | *** join/#asterisk ctooley (n=ctooley@rrcs-24-227-212-162.sw.biz.rr.com) |
01:38.22 | nassy | SkramX: just curious. is ventrilio useful with asterisk. i thought the audio on the phone was mono. is it useful with mono, or is the audio not mono |
01:39.10 | hhoffman | if you are using "ext => X,n,... using n instead of a number how do you do a n + 101? |
01:39.23 | Qwell | n+101 |
01:39.41 | hhoffman | oh! it that it :-) |
01:39.45 | Qwell | Shouldn't use 101 jumping anymore though |
01:39.46 | Zach^^ | SkramX what is your company? |
01:39.47 | hhoffman | s/is that/ |
01:39.56 | Qwell | just check the return code of the stuff when you need to |
01:40.09 | hhoffman | Qwell: ah, ok. thx |
01:40.16 | Pete_Largo | no more +101??? |
01:40.20 | Pete_Largo | what the hell? |
01:40.30 | Qwell | Pete_Largo: jumping like that isn't recommended anymore |
01:40.47 | Pete_Largo | jesus, next I know it's going to be calle #pound instead of #asterisk |
01:40.47 | Qwell | Check things like ${DIALSTATUS} and do stuff based on what that says, instead of blindly jumping elsewhere |
01:41.10 | Qwell | It just isn't useful anymore |
01:41.12 | hhoffman | Qwell: is this explained somewhere? I'm going off of the OReilly book and it's a bit different there |
01:41.19 | Pete_Largo | sure thing, where can I read up on that? |
01:41.23 | Qwell | changelog? |
01:41.24 | Pete_Largo | duh, "check the wiki" |
01:41.34 | Qwell | It's simple really |
01:41.39 | Qwell | s,1,Dial(blah) |
01:42.00 | Pete_Largo | don't stop |
01:42.06 | Pete_Largo | I'm learning new stuff! |
01:42.32 | Qwell | s,2,GotoIf($[${DIALSTATUS} = BUSY]?5:3) |
01:42.35 | Qwell | or some such |
01:42.58 | Qwell | or, as macro-stdexten does it |
01:43.09 | Qwell | s,2,Goto(s-${DIALSTATUS}) |
01:43.15 | hhoffman | ah, cool... I'll look at that |
01:43.18 | Qwell | s-BUSY,1,do(something) |
01:43.23 | Qwell | s-NOANSWER,1,do(something else) |
01:43.36 | Qwell | FAR more useful than just going to 2,102 |
01:43.39 | Qwell | s,102* |
01:44.19 | Pete_Largo | er, that should, have been jump-no-more-academy |
01:44.26 | hhoffman | yeah, that's awesome |
01:44.43 | Pete_Largo | well, time for me to go watch Veronica Mars in a few minutes :) |
01:48.13 | Zach^^ | :s can anyone help with voipjet? |
01:48.36 | tzanger | yeah, try nufone, asterlink, unlimitel... |
01:49.10 | Zach^^ | tzanger when i try to dial out i get an error all circuits are busy |
01:50.20 | SkramX | Im back |
01:56.56 | *** join/#asterisk insomni (n=insomni@x1-6-00-e0-18-6f-34-ad.k455.webspeed.dk) |
01:58.11 | *** part/#asterisk m160858 (n=jsaenz@200.89.12.46) |
01:58.37 | *** join/#asterisk r0d3nt (i=nobody@wsip-24-234-241-145.lv.lv.cox.net) |
01:58.41 | *** join/#asterisk brc_ (n=Brian@pdpc/supporter/basic/brc) |
01:59.37 | Seldon1975 | hi - my polycom501s assume all phone numbers are 2 digits long - whats up with that? |
01:59.50 | Qwell | the phones have a dialplan...change it |
01:59.52 | Seldon1975 | after 2 digits it tries to dial the number |
02:00.00 | MstlyHrmls | sounds like your dialplan is pooched |
02:00.06 | Seldon1975 | ok so its a phone-specific thing |
02:00.11 | Seldon1975 | not asterisk |
02:00.20 | MstlyHrmls | yeah |
02:01.08 | Seldon1975 | anyone there using Polycom Soundpoint 501s? |
02:01.31 | kuku5 | Don't know what to do if second ROSE component is of type 0x6 < what does this mean ? |
02:02.18 | MstlyHrmls | Seldon1975: yup |
02:02.21 | Cinen | Does anyone know where some good resources are for Dundi configs. I tried #Dundi but nobody is home. |
02:02.42 | Zach^^ | /dms 72.20.32.117 |
02:02.43 | Cinen | I set it up like the wiki said with no luck\ |
02:03.02 | Seldon1975 | MstlyHrmls: can you tell me how to change the dialplan? |
02:03.08 | Seldon1975 | or where to get the guide |
02:03.41 | MstlyHrmls | sure, just give me a sec |
02:03.48 | Seldon1975 | thx |
02:06.01 | MstlyHrmls | Seldon1975: OK, probably the best thing to do is to root around here for the Administrator guide for the Polycoms: http://www.polycom.com/support/voip |
02:06.14 | MstlyHrmls | I think you can only get it for 1.5.3, not the latest release though |
02:06.39 | MstlyHrmls | in there, it says to refer to RFC 3435 section 2.1.5 for the dialplan reference |
02:06.43 | Seldon1975 | ta |
02:06.55 | MstlyHrmls | (which is the MGCP RFC, but polycom uses that dialplan) |
02:07.14 | MstlyHrmls | if you have any questions, I might be able to answer them |
02:07.30 | MstlyHrmls | I'm no expert, but I've fiddled with them a bit :-) |
02:07.59 | *** join/#asterisk PBXtech (i=nik@120.sub-70-218-79.myvzw.com) |
02:08.31 | p1tst0p | can you playback an mp3, over a channel from the CLI ? |
02:08.40 | p1tst0p | or a gsm or whatever |
02:11.00 | PBXtech | redirect the channel |
02:11.13 | tzanger | anyone here do much voicexml stuff? |
02:13.03 | p1tst0p | JunK-Y not really mate, just play a audio file, when someone is running out of credit say... |
02:13.10 | Zach^^ | anyone here help me with voipjet? |
02:13.27 | p1tst0p | PBXtech, how does one redirect form CLI ? |
02:13.44 | JunK-Y | but why from the CLI? |
02:15.51 | *** join/#asterisk tuxinator_linuxM (n=spabin@70-32-106-248.ontrca.adelphia.net) |
02:17.15 | Seldon1975 | mstlyhrmls: I've set up two extensions, 1000 and 1001 in my * sip.conf; do I need to tell the phones what their extensions are? |
02:18.38 | MstlyHrmls | is this part of the dialplan still, or is this on to getting the phones to register? |
02:21.48 | *** join/#asterisk outofjungle (n=outofjun@61.247.249.13) |
02:24.13 | *** join/#asterisk anthm (n=anthm@h46087a75.area4.spcsdns.net) |
02:24.14 | *** mode/#asterisk [+o anthm] by ChanServ |
02:24.31 | Zach^^ | WTF why cant i get past all circuits are busy now |
02:27.23 | MstlyHrmls | Seldon1975: still working on the dialplan, or getting the phones to register? |
02:29.14 | *** join/#asterisk brettnem (n=brettnem@72.29.102.158) |
02:29.53 | Seldon1975 | MstlyHrmls registering the phones now |
02:30.00 | Seldon1975 | MstlyHrmls any ideas |
02:30.14 | Seldon1975 | do the phones need to be told? or will they read the info from * |
02:30.20 | MstlyHrmls | then, yeah, you have to tell the phones what extensions to use :-) |
02:30.24 | Seldon1975 | k |
02:30.27 | Seldon1975 | how? :D |
02:30.45 | MstlyHrmls | the easiest way is by the web interface |
02:31.03 | MstlyHrmls | point Your Favourite browser at the IP of one of the phones |
02:31.38 | Seldon1975 | ok I'm there |
02:32.01 | Seldon1975 | Sip Conf? |
02:32.03 | MstlyHrmls | you'll get a "SoundPoint IP Configuration" page. click on the SIP menu |
02:32.04 | MstlyHrmls | yup |
02:32.28 | Seldon1975 | OK I assume it's 'Local SIP Port'? |
02:32.38 | MstlyHrmls | no |
02:33.09 | MstlyHrmls | sorry, the Lines config is under "Lines", SIP is where you configure the asterisk address |
02:35.27 | Seldon1975 | so from http://phone.ip I go to which of 'Home' 'Core Conf' 'SIP Conf' 'Registration' ? |
02:35.43 | MstlyHrmls | 'Registration' |
02:35.50 | *** join/#asterisk fcr (n=fran@r201-217-143-141.dialup.adsl.anteldata.net.uy) |
02:35.51 | MstlyHrmls | what SW are you running btw? |
02:36.19 | *** join/#asterisk doughecka (n=Miranda@unaffiliated/doughecka) |
02:37.32 | Seldon1975 | i think 1.6 |
02:37.56 | fcr | Hello. I need to record a few telephone lines without hunging up, and I would like to know if this is possible with asterisk and some hardware. |
02:38.06 | Seldon1975 | do you see a 'Lines' section under 'Registration' |
02:38.08 | Seldon1975 | because I dont |
02:38.38 | MstlyHrmls | Seldon1975: I don't think you're using 1.6, because that's what I have and I'm seeing different things on the webconfig than you are :-) |
02:38.46 | Seldon1975 | bootrom of phone is 2.6.1.0003 |
02:38.49 | Seldon1975 | ok |
02:38.52 | Seldon1975 | damn |
02:39.24 | MstlyHrmls | are you at the phones right now? |
02:39.35 | SkramX | fcr: it's possible |
02:40.00 | Seldon1975 | yes |
02:40.14 | Seldon1975 | the phone's App version is 1.4.1.004 |
02:40.18 | fcr | SkramX, ok, so what hardware would i need |
02:40.33 | MstlyHrmls | Seldon1975: yeah, I thought that might be it |
02:40.43 | SkramX | So you want to be in the middle of a call and press a button and it records? |
02:41.03 | MstlyHrmls | Seldon1975: unfortunately, I don't have a phone running 1.4 handy, and I don't remember it well enough to step you through from memory :-7 |
02:41.05 | SkramX | fcr: any sip phone should work, just tell asterisk to start recording when you pickup |
02:41.17 | fcr | In fact i need to record a line all the day. |
02:41.22 | MstlyHrmls | Seldon1975: unless you wanted to set up a provisioning server for the phones, rather than just go through the webconfig |
02:41.58 | Seldon1975 | its ok MstlyHrmls; Im about to upgrade the phones to 1.6 |
02:42.04 | Seldon1975 | then we'll be on the same page :D |
02:42.09 | MstlyHrmls | :-D |
02:42.34 | Zach^^ | can you transfer users to the digital receptionist? or the queues? |
02:43.35 | MstlyHrmls | Seldon1975: the coles notes version is: put the asterisk address in the "Address" box of the "Server 1" section on the SIP page |
02:44.06 | Seldon1975 | ok Cinen is helping me modify my sip.cfg file which I think is the same setting |
02:44.16 | MstlyHrmls | Seldon1975: then put the "1000" or "1001" in the "Address" box of the Line 1 section on the "Lines" page |
02:44.23 | MstlyHrmls | yes, it is |
02:44.38 | Katty | hi lads. |
02:44.59 | MstlyHrmls | Seldon1975: are you creating <mac>.cfg files, or just using the 000000000000.cfg file? |
02:47.26 | Seldon1975 | the former |
02:48.42 | Seldon1975 | we are copying the file and renaming it to each phone's mac address |
02:49.52 | MstlyHrmls | cool |
02:50.15 | MstlyHrmls | Seldon1975: while you're at it, make a copy of the phone1.cfg file for each phone |
02:50.37 | MstlyHrmls | Seldon1975: rename it phone1000.cfg and phone1001.cfg or something |
02:50.44 | Katty | way to not say hi. |
02:50.49 | Katty | bunch of anti social geeks. |
02:51.03 | MstlyHrmls | then you can program the extensions in there, and avoid using the web config entirely |
02:51.24 | Katty | Qwell: k, grown up laddy. |
02:51.32 | Qwell | close enough |
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02:52.31 | docelm0 | hmmmm |
02:53.09 | MstlyHrmls | Katty: hi! if we weren't anti-social geeks, wouldn't we be out having lives instead of being here on IRC? :-) |
02:53.57 | Seldon1975 | MstlyHrmls do you mean that the extension setting is in (mac)-phone.cfg? |
02:54.22 | Katty | ManxPower: uhh. |
02:54.32 | Katty | ManxPower: i'm a female, not a feline. |
02:54.50 | Katty | ManxPower: also, i'm a vegan...so please remove all mouse related food items from general vicinity... kthx. |
02:55.05 | MstlyHrmls | Seldon1975: if you use the web config, that's where the phone will store it; but you can put it into the phone-<extension>.cfg file |
02:55.18 | mishehu | Katty: what about the food I feed my Logitech mouse? |
02:55.26 | Katty | mishehu: yay, mishehu |
02:55.35 | Katty | mishehu: that doesn't count. |
02:56.19 | mishehu | Katty: I hate doing dumb coursework. blah. I have one more assignment for this dumb economics class to finish, and I'm working on it now. :-/ |
02:56.27 | Katty | mishehu: :< |
02:56.52 | file[laptop] | Katty: you're silly |
02:57.03 | Katty | file[laptop]: takes one to know one, dear. |
02:57.03 | mishehu | why not file[desktop] |
02:57.18 | mishehu | anyway, back to this dumb coursework... |
02:57.25 | Katty | mishehu: you could always take a little trip to the paper shredder :> |
02:57.37 | mishehu | Katty: I need the grade, it's a required course. |
02:57.44 | Katty | mishehu: :< |
02:57.54 | Katty | mishehu: you could give all related stuffs to birdy when class is done :> |
02:59.16 | file[laptop] | I _hate_ the seatbelts in the Honda Civic Coupe... hate with a passion |
02:59.26 | Katty | file[laptop]: did they try to strangle you? |
02:59.48 | file[laptop] | Katty: well yes, but that's not it - they aren't attached to the seat so you have to reach back far to get them... almost dislocated my shoulder! |
03:00.03 | file[laptop] | which is bad mmmkthxbi |
03:01.33 | Katty | copykat. |
03:01.42 | file[laptop] | not I. |
03:01.53 | Katty | little red hen. |
03:02.23 | mishehu | Katty: I assure you I was already planning on lining hte birdcage with it. ;-) |
03:02.36 | Katty | mishehu: whoo! |
03:02.45 | Katty | mishehu: post gifs when the shredding begins :> |
03:02.57 | file[laptop] | how was your day KittyKatty? |
03:03.36 | Katty | i went on a service call an hour away to fix a symantec antivirus problem. |
03:03.47 | mishehu | Katty: unfortunately the shredding is never that exciting |
03:03.51 | Katty | and then, when i came back, 10 people pounced me with zomghastobedoneNOWstuff. |
03:04.03 | Katty | and then i studied mcse stuffs. |
03:04.08 | Katty | and went home at 4 |
03:04.17 | hhoffman | so are macro-superdial and macro-stdexten exclusive? |
03:04.18 | Katty | so here i've been, ever since, playing earthbound on snes. |
03:04.49 | Katty | file[laptop]: how has your day been? |
03:05.31 | Katty | oh! and i put my christmas tree up :> |
03:05.38 | Katty | but only one string of lights. |
03:05.41 | Katty | cause i'm lazy. |
03:05.50 | Zach^^ | Katty you us astrisk? |
03:05.55 | Seldon1975 | MstlyHrmls thanks for your help |
03:06.06 | Katty | Zach^^: why else would i be in here? |
03:06.10 | Katty | Zach^^: for the annoying questions? |
03:06.30 | Zach^^ | Katty: yep :p you think you can help a newb |
03:06.30 | Katty | Zach^^: yes, i use asterisk. |
03:06.40 | Katty | Zach^^: oh come on, i'm playing earthbound! |
03:06.55 | Katty | Zach^^: what's your issue? |
03:07.19 | Zach^^ | i have voip jet for youtbound and sixtel for incoming and when i place a outbound call i get all circuits busy |
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03:07.46 | Katty | Zach^^: can't help, sorry. |
03:08.02 | Zach^^ | :s why? |
03:08.34 | Katty | that's now what my setup is like. |
03:08.34 | Katty | s/now/not/ |
03:09.05 | MstlyHrmls | Seldon1975: no problem |
03:09.08 | Zach^^ | can anyonehelp? |
03:09.12 | Katty | but if you have issues with making sugar cookies turn out right, i'm totally the person to ask. |
03:09.29 | jake1932 | yum |
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03:09.39 | Zach^^ | yum make me some |
03:09.42 | puzzled | hmm cookies. reminds me I still have some stroopwafels |
03:09.47 | justinu | my fiance makes kick ass cookies |
03:09.47 | Katty | i'm not a maid, sorry. |
03:10.28 | Katty | sounds painful. |
03:10.34 | Zach^^ | how do i transefer users to a queue? |
03:11.23 | kuku5 | Is there a way to pickup a ringing phone from a phone that is not ringing ? |
03:11.32 | Katty | kuku5: features.conf |
03:11.37 | Katty | kuku5: usually *8 or something |
03:11.43 | kuku5 | wow |
03:11.47 | kuku5 | so I can do taht / |
03:11.54 | Katty | kuku5: yup |
03:12.05 | kuku5 | kickass |
03:12.14 | Katty | nonono |
03:12.17 | Katty | *8 |
03:12.19 | Katty | get it right. |
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03:13.13 | kuku5 | *8 and then the extension ? |
03:13.18 | Katty | kuku5: no. |
03:13.23 | Katty | kuku5: it picks up a random ringing phone |
03:13.32 | Katty | kuku5: i think it's random, don't actually know. |
03:13.47 | Katty | kuku5: you can edit your features.conf to stipulate what you want it to be |
03:14.00 | Katty | kuku5: or 'show features' at the asterisk CLI |
03:14.19 | kuku5 | ok |
03:14.24 | kuku5 | i'll need to look into it |
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03:17.05 | kuku5 | Builtin Feature Default Current |
03:17.05 | kuku5 | --------------- ------- ------- |
03:17.05 | kuku5 | Pickup *8 *8 |
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03:37.29 | syle | why do employees act like kids, test you for how much they can get away with |
03:38.54 | syle | answer: because its fun |
03:38.55 | brookshire | why do you always see how much you can get out of a customer? |
03:39.12 | file[laptop] | brookshire: why are you so hot? |
03:39.13 | file[laptop] | :P |
03:39.14 | file[laptop] | lol |
03:39.17 | brookshire | OMG! |
03:39.19 | brookshire | hehe |
03:39.22 | brookshire | hey file |
03:39.26 | file[laptop] | hi brookshire |
03:39.41 | file[laptop] | still hard at work? |
03:39.49 | file[laptop] | and don't read that with a nasty mind |
03:39.56 | brookshire | drum and bass is so hot |
03:39.58 | brookshire | :) |
03:40.10 | brookshire | it's been a long day :/ |
03:40.17 | file[laptop] | ah... poor you |
03:40.29 | brookshire | long and hard day :( |
03:40.39 | file[laptop] | well tomorrow is another day! |
03:41.04 | brookshire | hopefully a short one |
03:41.12 | file[laptop] | hopefully |
03:42.10 | hhoffman | anyone do dynamic meetme extensions with dynamic pin numbers? |
03:42.33 | file[laptop] | brookshire: I suggest you go home and get some sleep though |
03:43.25 | brookshire | i need to finish my staging server first |
03:43.38 | file[laptop] | bah |
03:43.47 | syle | wtf is a staging server |
03:43.51 | syle | does it dance? |
03:44.03 | brookshire | no.. you have a live server and a staging one :) |
03:44.12 | brookshire | staging is what you do development on |
03:44.14 | syle | you mean development box |
03:44.18 | jake1932 | and a development one |
03:44.31 | jake1932 | there could be 3 |
03:44.35 | Nugget | a staging server is a way for smart admins to accomodate software packages that have crappy or unreliable release cycles. |
03:44.47 | Nugget | it's not the same as a development server |
03:44.47 | *** join/#asterisk mog_home (n=mogorman@user-24-236-84-48.knology.net) |
03:44.54 | brookshire | yup.. |
03:44.56 | *** join/#asterisk perlmonkee (i=oginvu@c-24-22-125-15.hsd1.or.comcast.net) |
03:45.06 | brookshire | like we're trying to safely upgrade to mysql 4.1 from 4.0 |
03:45.13 | syle | good luck |
03:45.14 | perlmonkee | I'm having a problem with Asterisk and SipPhone |
03:45.21 | brookshire | and as quickly as possible with little downtime |
03:45.32 | perlmonkee | I can place outgoing calls through SipPhone, and `sip show registry` says I've registered with SipPhone.com |
03:45.41 | hhoffman | only problem I had with that upgrade was CHARSETs |
03:45.44 | syle | i did that ages ago, moving to 5.x was less painful from 4.1 |
03:45.46 | perlmonkee | I've set up call routing for incoming calls to my extension |
03:45.57 | brookshire | 4.0 to 4.1 is a bitch |
03:46.00 | perlmonkee | but every time I call my SipPhone number (from another number), it jumps straight to SipPhone Voicemail. |
03:46.13 | brookshire | but then again i hate everything about 3.x :( |
03:46.18 | brookshire | about = above |
03:46.20 | syle | if i was you i would just dump all my data and go to 5.x like everyone else |
03:46.23 | perlmonkee | If I register directly to SipPhone with a softphone, everything works. |
03:46.23 | hhoffman | brookshire: what's making it so bad for you? |
03:46.38 | syle | over 1 million people downloaded 5.x in first 3 weeks it became stable are the stats |
03:46.42 | brookshire | hhoffman: when you go from 4.0 to 4.1 the authentication methods change |
03:47.02 | brookshire | so php stops working |
03:47.07 | brookshire | and perl stops working |
03:47.09 | hhoffman | ah, yeah... we upgraded all clients and perl/php |
03:47.12 | brookshire | and etc, etc etc |
03:47.21 | brookshire | we've just been putting it off |
03:47.31 | hhoffman | there's a way around upgrading the clients |
03:47.36 | brookshire | because truthfully.. there is no advatage to goto 4.1 |
03:47.39 | syle | no, you just have to recompile perl DBI again new mysql api and php never had a problem |
03:47.54 | syle | s/again/against |
03:48.07 | brookshire | syle: exactly :) |
03:48.16 | perlmonkee | Can anoyone help with Asterisk+SipPhone.com problems? |
03:48.17 | syle | an extra 10 min not a big deal |
03:48.22 | brookshire | but the problem is... |
03:48.25 | hhoffman | we experienced it in a DBI:: app but couldn't upgrade... there's a way in the FAQ to deal with that |
03:48.43 | brookshire | we have to do all of our servers at the same time, because they depend on each other, lol |
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03:49.08 | syle | if you read the mysql docs you can tell it to be backwards compatible by setting password=old or whatever |
03:49.27 | brookshire | syle: i haven't had time yet |
03:49.29 | syle | i prefer to move on with changes |
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03:51.08 | brookshire | anyways.. better to be safe than sorry |
03:51.19 | brookshire | and have asterisk.org down for a couple of days |
03:51.24 | brookshire | lol |
03:51.52 | Cinen | anyone here had any luck with Dundi? I asked in #Dundi but nobody is home |
03:52.02 | syle | i would just do it right brookshire in the first place |
03:52.10 | syle | export everything go to 5.x |
03:52.28 | syle | then you won;t have to worry about it for 2 years insteead of comming back to this in a year going to 5.x when 5.1 is now stable |
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03:52.53 | brookshire | syle: actually.. i've found 4.1 kind of unstable, until the more recient releases |
03:53.07 | brookshire | receint also |
03:53.08 | brookshire | lolalskdjfasdf |
03:53.45 | brookshire | recent |
03:53.47 | brookshire | THERE |
03:53.56 | syle | not like asterisk.org is taking a million hits a day, don; think you have to worry to much :) |
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03:55.08 | syle | i wonder if NASA and sourceforge are using 4.1 or 5.x |
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03:55.23 | brookshire | well.. more like 200,000 hits |
03:55.26 | brookshire | but yeah |
03:55.29 | brookshire | not a million |
03:55.38 | syle | no shit |
03:55.40 | syle | unique? |
03:55.45 | brookshire | oh... just hits |
03:56.01 | brookshire | 4,000 visits |
03:56.06 | brookshire | i think |
03:56.11 | syle | 4000 unique isn;t bad |
03:56.35 | syle | damn in a few years that will be quite amazing |
03:56.48 | brookshire | i know.. we're working on it :) |
03:57.20 | brookshire | that doesn't include downloads, or svn |
03:57.33 | brookshire | viewcvs |
03:57.40 | syle | you keep track of download stats? |
03:57.49 | orlok | Hmm.. I seem to have lost my outbound call routing |
03:57.54 | brookshire | yup :) |
03:57.59 | syle | hehe nice |
03:58.14 | brookshire | close to 2,000 a day |
03:58.18 | brookshire | just asterisk |
03:58.20 | syle | no shit |
03:58.33 | syle | damn i would never have guess that |
03:59.10 | syle | are you sure 2000 a day? |
03:59.16 | syle | unique ip downloads? |
03:59.24 | brookshire | well it's been that reciently because of 1.2 |
03:59.33 | syle | owww that makes sense |
03:59.41 | brookshire | recently |
03:59.42 | syle | how about on average when no new releases? |
03:59.44 | brookshire | there i go again, lol |
04:00.00 | jake1932 | i before e except when no i |
04:00.09 | brookshire | there is no i in recently |
04:00.16 | file[laptop] | brookshire: you can spell... right? |
04:00.17 | tainted_ | lol |
04:00.27 | brookshire | i keep wanting to add it though |
04:00.28 | brookshire | lol |
04:00.47 | tainted_ | jake1932 what english class did YOU attend |
04:00.59 | jake1932 | none - got everything from IRC |
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04:01.17 | tainted_ | sweet |
04:01.39 | syle | i think i was taught in university by programmers to tell people who had spelling mistake problems with you that you aren't an artsy fartsy, your a science man hehe |
04:01.40 | jake1932 | (and read the wiki, of course) |
04:03.00 | jake1932 | any of you use shorewall with asterisk? |
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04:03.18 | docE | ya? |
04:03.20 | syle | although the wars between arts and sciences were always bad in my schools, science students use to wear shirts that said "friends don't let friends take ARTS" |
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04:04.27 | jake1932 | i'm going to attempt setup tomorrow and just wanted to know that it does work on the same box as asterisk giving asterisk the public ip |
04:06.16 | syle | if you didn;t find that funny, you've never been to university :) well back to coding |
04:06.18 | jake1932 | and btw - arts != english - I know plenty of musicians who can't spell worth a damn |
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04:07.54 | syle | you must realize science people are only obligated to take first year english, and are usually passed even if they fail because their in sciences and need to move on |
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04:08.42 | syle | if your in arts, noone cares about you |
04:09.29 | syle | get a BA and work at a gas station, or become a lawyer hehe |
04:09.48 | Nugget | scientists who fail that first year english are likely to fuck up the difference between "there", "their" and "they're" |
04:09.58 | jake1932 | only in IRC |
04:09.59 | Nugget | obviously disqualifying them from a technical life. |
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04:10.35 | syle | maybe thats why you can barely even understand what doctors write on prescriptions hahaa |
04:11.36 | jake1932 | don't quite understand why I can easily explain the difference between when to use there", "their" and "they're, but when I'm typing fast, it doesn't come out right |
04:12.20 | syle | yeah don't mistake can't spell between types to fast for sure |
04:12.26 | Nugget | I absolutely hate it when people use apostrophes in plurals, and yet from time to time my fingers do it without my involvement. |
04:12.33 | jake1932 | right |
04:12.37 | Nugget | I'm always left staring at the little buggar wondering how the hell it happened. |
04:12.49 | jake1932 | there must be something to it |
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04:13.00 | syle | my problem is using ; instead of ' when typing to fast lol |
04:13.00 | Pete_Largo | that happens to me a lot too |
04:13.02 | Nugget | and I'm totally ruined for typing the word "serve" without throwing an extra "r" on the end. |
04:13.12 | Nugget | not to mention "distribute" without a trailing "d" |
04:13.13 | Pete_Largo | I do that too |
04:13.17 | Pete_Largo | and that |
04:13.35 | jake1932 | there must have been a study done on that |
04:13.36 | Pete_Largo | and the / and \ always get me too |
04:13.45 | Pete_Largo | which goes where |
04:13.55 | syle | yes because of going between creating windows shares and working with unix directories |
04:13.59 | jake1932 | Pete - that's a windows/linux thing |
04:14.02 | Pete_Largo | right |
04:14.04 | syle | always think its / on windows at first lol |
04:14.06 | Pete_Largo | just like ls and dir |
04:14.10 | Pete_Largo | del and rm |
04:14.14 | Nugget | it's a windows/UNIX thing. :P |
04:14.15 | Pete_Largo | happens all the time |
04:14.28 | Nugget | because it happens to me, too, and I never use linux. |
04:14.49 | syle | its a windows\UNIX thing, its a UNIX/windows thing :) |
04:14.53 | Nugget | heh |
04:14.55 | jake1932 | hah |
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04:15.49 | jake1932 | night people |
04:16.10 | syle | i'm just waking up |
04:17.21 | dokhench | have you're self a few shot of jack, then right off to bed again. you'll do the sleep wrap around in no time. =) |
04:17.28 | dokhench | yourself that is. |
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04:17.52 | syle | whats the theory behind that |
04:18.19 | dokhench | the jack will drunk ya up, before you have a chance to fully wake up. =) |
04:18.34 | syle | hehe |
04:18.46 | syle | i should try that :) |
04:19.14 | Nugget | all I have here is a fe bottles of wine and a fifth of wild turkey. |
04:19.17 | Nugget | s/fe/few/ |
04:19.48 | syle | whjen i have wine i;m drunk as a skunk |
04:19.52 | syle | worse than beer |
04:20.09 | Nugget | wine drunk creeps up on you |
04:20.11 | syle | seem to pound back a wine bottle quite fast |
04:20.24 | Nugget | you feel fine and then get up to go pee and the room spins |
04:20.31 | syle | yeah its great |
04:20.37 | syle | think i;ll have some for christmas |
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04:21.23 | syle | red wine that is, never got same effects with white |
04:22.21 | Nugget | I'm not fond of white wine. I only drink it when I'm having cheese fondue or something like that. |
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04:25.09 | syle | fondue hmm haven;t had that since i was in montreal |
04:25.12 | syle | you from canada? |
04:25.32 | *** join/#asterisk tholo (n=tholo@nat.sigmasoft.com) |
04:26.41 | Nugget | no |
04:26.48 | syle | i like sweet red wine's but all the damn restaurants only serve dry pisses me off |
04:27.02 | syle | so liquor store works out great |
04:27.54 | syle | i;ve considered making my own at home, but then i think would become an alcoholic lol |
04:28.05 | Nugget | I brew my own beer. It's fun |
04:28.12 | Nugget | I don't know much about winemaking, though |
04:28.44 | syle | takes less time to sit |
04:28.47 | syle | about a week |
04:28.51 | Nugget | cool |
04:28.56 | syle | beer can take a month to get some good alcohol content |
04:29.12 | syle | well depends how much yeast you add hehee |
04:29.13 | Nugget | right |
04:29.26 | Nugget | it takes a month before it will taste good, for sure. |
04:29.36 | Nugget | the alcohol doesn't take that long to build |
04:30.03 | syle | no but you have 1 month cyles, so you have to know how much you drink in a whole month all the time :) |
04:30.17 | Nugget | nah, if I overproduce I just invite people over. |
04:30.21 | Nugget | total non-problem |
04:30.24 | syle | good man :) |
04:30.33 | syle | where you live again :) |
04:31.14 | Nugget | austin texas :) |
04:31.24 | syle | oww the US kewl |
04:31.32 | *** join/#asterisk MikeJ[Laptop] (n=ircatjer@d14-69-8-30.try.wideopenwest.com) |
04:32.26 | syle | i only had one experience with texas, passing through that airport back to canada from the carribean, customs can really go screw themselves on all their security bush created, i was waiting in line 2 bloody hours, never go through the US again |
04:32.47 | syle | direct flights only for now on :) |
04:32.47 | Nugget | austin is almost entirely unlike the whole rest of the state. |
04:33.34 | syle | think i met a few people from austin in panama |
04:33.49 | syle | all into real estate |
04:33.58 | Nugget | I've lived a lot of places, and this is the one where I never want to leave. |
04:35.25 | syle | yeah i couldn;t handle your heat |
04:35.28 | Jack_Storm | syle: WTF are you talking about? if customs fucked with you, you gave them reason. I came back from Russia to Atlanta, had customs there and here, and they had no problems when me, even thou I was smuggling |
04:35.55 | Jack_Storm | and here is New Orleans |
04:35.57 | syle | dude there was no reason, that was the damn lineup |
04:36.15 | syle | i was stuck in line 2 hours with other connecting flights |
04:36.26 | Nugget | flying through houston or dallas? |
04:36.36 | fugitivo | US inmigration department sux |
04:36.38 | syle | hmmm i think it was houston |
04:36.50 | Nugget | houston is really bad. they have like six hundred lines for us citizens and three for non citizens |
04:37.14 | justinu | that rules |
04:37.47 | syle | well they always fuck with me leaving canada anyways, their reason: you brought a laptop: my mind saying: go fuckyourself everytime |
04:38.06 | Nugget | every time I would go on a business trip with Ivo at my last job (he's dutch) I'd clear customs in 5-10 minutes coming back and he'd take an hour. |
04:38.25 | fugitivo | syle: don't tell them you have linux on it, it's worse (you have linux, right?) |
04:38.29 | syle | if i wanted to hide something on my laptop i;d download off the net after i crossed, no reason to search it |
04:38.42 | syle | yes |
04:38.45 | syle | double booting |
04:39.06 | syle | sorry laptop i take is dual booting XP and freebsd |
04:39.11 | syle | but they only see the XP |
04:39.20 | *** join/#asterisk alephcom (n=alephcom@207.34.97.130) |
04:39.29 | syle | i pretty much use to the routine now |
04:39.31 | Jack_Storm | syle: did you have a .ca customs stamped document showing that the laptop was yours and in .ca prior to your departure? |
04:39.33 | fugitivo | they check your files? |
04:39.43 | syle | start menu->find-> *.* |
04:39.50 | *** join/#asterisk MikeJ[Laptop] (n=ircatjer@d14-69-8-30.try.wideopenwest.com) |
04:39.51 | fugitivo | really? |
04:39.52 | syle | look at all my files |
04:39.54 | syle | ok you can go now |
04:40.07 | syle | yes man |
04:40.16 | fugitivo | they shouldn't look at your files |
04:40.16 | fugitivo | that |
04:40.19 | fugitivo | that's private |
04:40.28 | syle | its customs they can do anything they want |
04:40.31 | MstlyHrmls | I've never had them do that to me |
04:41.01 | Jack_Storm | syle: I came back from Russia, they didn't want to look at anything, it's all on how you present your self. |
04:41.02 | MstlyHrmls | I've never really had much flack at all |
04:41.21 | Jack_Storm | MstlyHrmls: nod same here |
04:41.29 | MstlyHrmls | perhaps the pre-clearance centres are a little more lax |
04:41.47 | syle | its only certain airports |
04:41.57 | MstlyHrmls | but as long as I'm only going down for meetings, they don't seem to care |
04:42.02 | Jack_Storm | syle: no nothing to do with Bush then? |
04:42.05 | syle | well 1 airport |
04:42.10 | Jack_Storm | s/no/so/ |
04:42.44 | syle | you kidding? |
04:42.53 | syle | how about the iris and fingerprint scanners |
04:43.58 | fugitivo | us inmigration department is a joke |
04:44.04 | syle | convenient to pass through the iris scanner very quick, but thats quite an invasion of privacy |
04:44.14 | Jack_Storm | syle: he wouldn't do that to only certain airports |
04:44.23 | fugitivo | they check your files, that's invasion of privacy |
04:44.23 | syle | only time i want to voluntary give out my fingerprints to any country is if i;m being arrested thankyou |
04:44.44 | Jack_Storm | syle: by your def, your passport is an invasion of privacy. |
04:44.56 | syle | how so? |
04:45.11 | *** join/#asterisk delox99 (n=delox99@206-248-149-59.dsl.teksavvy.com) |
04:45.15 | delox99 | hi all |
04:45.27 | Jack_Storm | syle: does it not contain a likeness of you? your sig? your address and where you have been? |
04:45.45 | syle | its a picture with a my signature, not much different than my drivers license |
04:46.13 | syle | well where you have been is just you know your from here |
04:46.56 | syle | does it really matter? |
04:46.58 | *** join/#asterisk cnet2 (n=jjohn@201.192.107.58) |
04:47.05 | Jack_Storm | syle: and your personal info on your drivers lic is ok, but an eye scan to pass you quickly is an invasion of privacy? |
04:47.10 | syle | if they have your SIN or SSN they got everything they need anyways |
04:47.21 | *** join/#asterisk asterboy (n=Snake@S01060204ee2b6007.ed.shawcable.net) |
04:47.33 | delox99 | in zapata.conf what s the difference between group, callgroup, pickupgroup, and group for channels? |
04:47.41 | syle | jack yes |
04:48.28 | syle | lets say i figure out my fingerprints to some asswipe at some countries border, he decides to frame me for a crime, he has my fingerprints, totally screw me over, like i said invasion of privacy |
04:48.52 | syle | s/figure/gave |
04:49.13 | Jack_Storm | syle: I have your sig off your passport, forge it to a check that was used to pay someone to kill someone, same diff. |
04:50.00 | syle | yes but DNA data is unargueable in court |
04:50.20 | syle | i could argue the signature thing |
04:51.09 | Jack_Storm | syle: and I could argue the iris scan, and the MS backed filesystem and DB cluster. |
04:51.21 | syle | and we all know americans get ahold of all the customs records who pay for it |
04:51.31 | syle | and they get sold on the black market |
04:51.41 | syle | as i said less info the better |
04:52.13 | *** join/#asterisk pigpen2 (n=mark@fw.seamans.cc) |
04:52.18 | syle | you gonna make an arguement that you don;t have corrupt politicians good luck |
04:52.31 | Jack_Storm | syle: and thats why I can buy a HDD in Moscow filled with .ca personal info and photos of people with out passports? |
04:52.39 | pigpen2 | Could someone answer a question regarding the Polycom 601? |
04:53.19 | syle | yes |
04:53.21 | MstlyHrmls | pigpen2: maybe |
04:53.50 | pigpen2 | Ok..with the exten presence (ie: hint) it will monitor 7 exten's.... |
04:53.58 | syle | you know enough wealthy people in american you can get any info you want |
04:54.03 | pigpen2 | is this per phone, or per expansion module? |
04:54.14 | MstlyHrmls | per phone |
04:54.35 | pigpen2 | that is what I thought...thanks..you saved me time of testing it. |
04:54.43 | MstlyHrmls | :-) |
04:54.47 | pigpen2 | any news on it getting expanded? |
04:54.56 | pigpen2 | 7 just doesn't cut it. |
04:55.22 | MstlyHrmls | "in a future release" |
04:55.33 | pigpen2 | so there is rumor... |
04:55.48 | MstlyHrmls | and, no, 7 doesn't cut it |
04:55.54 | pigpen2 | I figured it was 7 due to the fact the 600 did not have the expansion module. |
04:56.21 | asterboy | how much did you pickup your 601 for? |
04:56.24 | pigpen2 | ok..next question (sorry, not polycom related) |
04:56.33 | pigpen2 | 601: $239 US |
04:56.48 | asterboy | good price |
04:57.09 | pigpen2 | Has anyone used the "metermaid" patch for monitoring parked exten's? |
04:57.55 | pigpen2 | ie: http://bugs.digium.com/view.php?id=5779 |
04:58.37 | *** join/#asterisk `Sauron (i=sauron@h-69-3-12-50.hstqtx02.covad.net) |
05:00.32 | pigpen2 | well, seems pretty cool...we are compiling it in now. |
05:00.49 | *** join/#asterisk wundaboy (n=asdf@c-67-164-107-68.hsd1.or.comcast.net) |
05:03.14 | *** part/#asterisk tholo (n=tholo@nat.sigmasoft.com) |
05:05.13 | *** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net) |
05:06.07 | *** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net) |
05:06.35 | denon | you guys seen this? http://denon.cx/christmas |
05:06.39 | denon | someone's got way too much time on their hands |
05:06.56 | *** join/#asterisk rajiv (n=irc@gentoo/developer/rajiv) |
05:08.31 | syle | damn |
05:08.45 | syle | that makes me want to drink beer on that lawn |
05:09.14 | denon | u huh |
05:09.27 | syle | this would be great for an outside bar |
05:10.12 | syle | well if you know anything about bars you know most popular ones are the ones that invested alot of money in lighting shows |
05:10.39 | *** join/#asterisk mog_home (n=mogorman@user-24-236-84-48.knology.net) |
05:11.00 | syle | one owner here retired in 4 years |
05:11.02 | fugitivo | bars or discos? |
05:11.09 | syle | paid 1 million for his lighting in his bar |
05:11.28 | syle | made that back in 1 year, made another 5-10 mill, retired |
05:11.36 | syle | moved back to europe |
05:12.30 | syle | but we;re talking about a 1 million dollar laser system show |
05:12.47 | syle | mirror galore etc |
05:13.44 | syle | smart guy, and i beleive the move to europe was for a tax break on his last year he made money |
05:14.31 | syle | as you know canada, US and australia are the top taxing countries, if you make alot of money in one year its best to move out of those countries for abit if possible |
05:16.33 | syle | and thats why security is bad, gov't is using it as a way to track your offshore bank accounts |
05:17.04 | SkramX | Hi All |
05:19.31 | orlok | syle: i must say, i have never, ever been struck by lighting shows as being something a bar needs |
05:19.34 | orlok | 1. beer |
05:19.41 | orlok | 2. something to rest the beer on |
05:19.51 | orlok | and if you are lucky, 3. somewher to place your butt |
05:19.55 | syle | your wrong its all about atmosphere |
05:19.57 | orlok | walls and a roof are optional |
05:20.17 | orlok | yeah, and its a pub, not a rave or a disco! |
05:20.33 | syle | well depends if you want to make 50k a year or a million i guess |
05:20.51 | orlok | which country you from? |
05:20.54 | syle | well thats not accurate |
05:20.56 | orlok | the states? |
05:21.03 | syle | all bars make a few hundred k a year |
05:21.15 | orlok | cos whereever it is, they have a warped view of pubs compared to the ones here in australia |
05:22.18 | syle | you have to realize there is no outdoors in winter here |
05:22.33 | orlok | which country? |
05:22.40 | syle | canada |
05:22.47 | orlok | ahh |
05:22.51 | orlok | yeah, i been there in winter |
05:22.58 | orlok | in early 1989 |
05:23.07 | orlok | -26.. my face froze up i swear |
05:23.22 | syle | only -26 damn come in january :) |
05:23.39 | syle | -40 at times with windchill :) |
05:24.14 | syle | literally my eyelids can freeze in about 20 sec |
05:24.16 | syle | :) |
05:24.19 | orlok | yeah, it was jan. |
05:24.27 | orlok | on the way back home after xmas in england |
05:24.43 | *** join/#asterisk Smi|k (n=smilk@adsl-66-159-200-157.dslextreme.com) |
05:24.48 | orlok | hmm, i hooked my bosses voip phone up to asterisk |
05:24.56 | orlok | and i've slightly buggered up inbound and outbound calls |
05:25.05 | syle | i stayed in phoenix arizona for awhile when i was working, i couldn;t handle that heat |
05:26.05 | syle | haha and your boss wants to kick your ass to fix it or put it back the way it was right hehe |
05:27.27 | mog_home | anyone ever set up jabber2 lately, i switched from ldap to pam and cant seem to get it to let me log in |
05:27.40 | ManxPower | ~docw |
05:27.45 | ManxPower | ~docs |
05:27.46 | jbot | i heard docs is probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com |
05:27.48 | ManxPower | ~mailinglist |
05:27.49 | jbot | extra, extra, read all about it, mailinglist is Search Asterisk mailing lists by prepending site:lists.digium.com to your Google search. Browse the mailing list archive at http://lists.digium.com/ or search through it at http://www.asteriskguru.com/archives/search.php |
05:27.54 | mog_home | ? |
05:28.35 | syle | orlok you have extensive experience with asterisk and mysql? |
05:29.03 | *** part/#asterisk fcr (n=fran@r201-217-143-141.dialup.adsl.anteldata.net.uy) |
05:29.15 | syle | i'm looking for beta testers today, releasing beta version of module i have been working on for 2 months |
05:29.19 | kuku5 | jbot: put it on the topic |
05:29.23 | kuku5 | what module |
05:29.43 | syle | Sunsaturn |
05:29.55 | kuku5 | eh |
05:30.25 | Smi|k | anyone mind looking over a new page design and giving me a little feedback? |
05:30.51 | syle | sure |
05:31.03 | syle | i;m great at critisizing designs |
05:31.19 | Smi|k | www.mp3yourcar.com vs. http://mp3yourcar.com/default3g.asp |
05:32.19 | *** join/#asterisk MikeJ__ (n=ircatjer@d14-69-8-30.try.wideopenwest.com) |
05:32.44 | syle | just google Sunsaturn asterisk if you want more info |
05:33.20 | asterboy | dam both are good designs |
05:33.38 | syle | you designed that? |
05:33.55 | Smi|k | I edited the first into the 2nd |
05:34.09 | *** join/#asterisk MikeJ[Laptop] (n=ircatjer@d14-69-8-30.try.wideopenwest.com) |
05:34.27 | syle | http://mp3yourcar.com/default3g.asp is better |
05:34.34 | Smi|k | 25% of people who hit the opening page of the first one leave right away |
05:34.47 | Smi|k | dont click anything else, dont use the dropdown menus etc..etc.. |
05:34.54 | Smi|k | 2nd one I dont know if it will be better or not |
05:35.05 | syle | of course it is |
05:35.09 | orlok | syle: no, def. not extensive :) |
05:35.19 | syle | you finally put that damn search stuff out of the way on the right |
05:35.25 | syle | instead of in main body |
05:35.39 | Smi|k | you cant actually buy ANYTHING until you search for your vehicle |
05:35.56 | syle | yes but second design i can see products on the page without scrolling to |
05:36.01 | Smi|k | so the idea behind clicking one of the ipods is the next step you click your car make |
05:36.06 | syle | thats where my attention goes |
05:36.07 | Smi|k | then you click your car model, then you click your car year |
05:36.26 | Smi|k | we dont actually sell ipods, it saying to select your ipod |
05:36.32 | Smi|k | thats just to have poeple click something |
05:37.09 | syle | right but design 2 is better cause then i can see what i can get for my car first |
05:37.26 | Smi|k | its not showing anything you can get for your car unless you select your car |
05:37.43 | Smi|k | we sell parts that connect the ipod (which you already have) to your car (which you already have) |
05:38.17 | syle | then your not taking advantage of marketing |
05:38.30 | syle | you should open reseller account for the actual ipods, and sell the parts |
05:39.02 | Smi|k | assuming I want to sell car adapters |
05:39.35 | syle | more you sell more you make |
05:39.36 | syle | simple |
05:40.53 | syle | i can;t say i like the design then if its for just the car audio hookup |
05:40.55 | orlok | heh, nope :) |
05:41.15 | syle | at least show some pictures of cars, people happy with their new sound system etc |
05:42.08 | syle | or keep your currrent design and take advantage of being an ipod reseller as well |
05:43.10 | *** join/#asterisk tuxinator_linuxM (n=spabin@70-32-106-248.ontrca.adelphia.net) |
05:44.11 | syle | i;ve always been an expert when it comes to design and marketing, just suck with dealing with sales(hate talking with people) but i can get them all there :) |
05:45.01 | syle | last 3 companies i worked for are millionares now :) |
05:45.45 | Smi|k | hrm |
05:45.52 | Smi|k | so how do I sell ipod car connections |
05:46.10 | syle | first you have to consider whats the first thing they see when they come to the page |
05:46.43 | syle | and control the flow from there, i don;t have enough time to give that many tips but gl |
05:46.44 | Smi|k | should be the header sentence |
05:46.59 | Smi|k | " |
05:46.59 | Smi|k | Finally, an iPod Car Kit That Doesn't Force You to Settle for Poor Audio Quality |
05:46.59 | Smi|k | " |
05:47.26 | asterboy | you have to sell the right market. |
05:47.46 | syle | market is different all together |
05:47.50 | Smi|k | the market is people who tried the itrip and hate the fm quality |
05:47.52 | syle | hes talking about just design |
05:48.04 | Qwell | small market |
05:48.16 | Smi|k | oh, I'm not changing design because I'm an artist, I want more people to buy car ipod adapters |
05:48.22 | asterboy | que in the 18 year old girl that I will get because I'm listening to "your" ipod equipment. |
05:48.23 | tuxinator_linuxM | what's an Ipod? |
05:48.35 | tuxinator_linuxM | just kidding |
05:48.40 | tuxinator_linuxM | I don't have one |
05:48.41 | Qwell | "If you buy this, more girls will sleep with you." |
05:48.43 | syle | yes asterboy makes a good point |
05:48.50 | syle | what is your target audience? |
05:48.58 | tuxinator_linuxM | people with money |
05:49.03 | Smi|k | 21-35 male |
05:49.07 | asterboy | young geeks |
05:49.09 | syle | your design should be influenced around that age/group |
05:49.09 | Qwell | "...or boys...we don't judge." |
05:49.51 | Smi|k | products differ based on their vehicle |
05:49.59 | tuxinator_linuxM | I like the "don't suck" idea |
05:50.00 | syle | so 21-35 male, likes: fast cars, kewl gadgets, easy on their pocket books, do some research and tailor |
05:50.01 | Smi|k | so before I can show them products I must find out what kind of car they have |
05:50.14 | Smi|k | or else they see 10 products with 10 diff pics/descriptions and only one of the 10 works for their car |
05:50.30 | *** join/#asterisk alvariux (n=alvaro@201.155.166.186) |
05:50.33 | alvariux | hello |
05:50.36 | *** join/#asterisk r0d3nt|m (i=nobody@wsip-24-234-241-145.lv.lv.cox.net) |
05:50.55 | syle | here is a very big tip: They want to know how it will change their life |
05:51.05 | Smi|k | syle, the ipod is a culture of its own now, so instead of focusing on fast cars I can sell to a massive number of fast car drivers AND toyota prius owners because they can identify with ipod culture |
05:51.34 | alvariux | im buildg asterisk from svn buillds ok but when i try to start it shows me this: loader.c:414 __load_resource: app_db.so: load_module failed, returning 3 |
05:51.41 | syle | will i be kewler having one? will i be popular with the girls? |
05:51.52 | alvariux | somebody knows what is wrong? |
05:51.52 | Smi|k | I tried to hit the changing their life aspect #1 by writing "you dont have to settle for poor quality music" |
05:52.09 | Smi|k | syle, you will enjoy your ipod MORE. |
05:52.13 | asterboy | how about reversing the image? |
05:52.14 | syle | settling is not changing |
05:52.45 | Smi|k | "It is time to enjoy your iPod in the car" |
05:52.46 | Smi|k | how about that? |
05:52.58 | alvariux | can somebody help me |
05:53.01 | syle | yes but what can the IPOD to for me? |
05:53.03 | syle | do |
05:53.14 | asterboy | you show a young geek male with a rust bucket...but his clean ipod gives him the illusion of driving a porche. |
05:53.15 | Smi|k | they already paid $300 for an ipod and loaded it with music |
05:53.28 | Smi|k | everyone coming to my site has an ipod already |
05:53.41 | syle | your thinking wrong, maybe they will buy an ipod and your hookup |
05:53.45 | Qwell | Smi|k: I don't. Otherwise I'd buy one now |
05:54.01 | Smi|k | apple doesnt have online resellers like that |
05:54.01 | syle | you can make profit twice |
05:54.06 | Smi|k | and ipods dont have margins |
05:54.48 | Smi|k | if market research says 99% of the people who come to my site have an ipod already I should base the site around the assumption that the visitor has an ipod already correct? |
05:54.52 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
05:55.15 | syle | smi|k truely all this shit is trial and error and takes experience, you setup your programming scripts to track stats based on each new design you put out and watch the popularity between designs |
05:55.24 | Smi|k | once they enter the vehicle search I find out if they have a 1995 Nissan Ultima or a 2006 Mercedies S500 |
05:55.33 | Smi|k | before they do that I only know they have an ipod |
05:56.15 | syle | only stupid people use the same design all the time |
05:56.23 | Smi|k | hrm, who usually does this, because I am totally lost... you are helping a lot right now, but I mean in the long run I need to do it right |
05:57.22 | syle | i use to run a marketing operation for a company and made about 12 thousand a day |
05:57.31 | Smi|k | website design contest signup link on the right hand side? |
05:58.26 | syle | hehehe |
05:58.27 | Smi|k | every week give away one car adapter to the top front-page-design winner, and monthly the best front-page-designer gets $xxxx to design the rest of the site |
05:58.31 | Smi|k | or does stuff like that not work |
05:58.39 | syle | just try new designs |
05:58.43 | syle | and go from there |
05:58.58 | Smi|k | its the whole chicken-egg problem here... sales from design, money from sales, money for design, design for sales |
05:58.59 | syle | you;ll need to learn alot more about marketing to sell |
05:59.31 | syle | understand basic marketing lingo first |
05:59.45 | syle | CPM CPC branding, co-branding |
06:00.17 | Smi|k | the new logo at the top is trying to create a "brand" for the store, I was told that would help shoppers be more comfortable vs. no brand as before |
06:00.52 | syle | are you the owner or you some guy working on this website? |
06:01.02 | Smi|k | unfortunantely both |
06:01.09 | Smi|k | chicken-egg :) |
06:01.49 | *** join/#asterisk rezEdit (n=rezEdit@zapdos.omnigroup.com) |
06:02.02 | syle | depends how far you want to go with this |
06:02.13 | Smi|k | to the end, I started it in the beginning |
06:03.06 | Smi|k | my passion, my life, its all been based around it for almost 10 years now |
06:03.49 | syle | i don't understand, ipods haven't been out that long |
06:03.55 | Smi|k | http://wired-vig.wired.com/news/print/0,1294,17738,00.html |
06:04.35 | Smi|k | click the name Jeremy, then click "order now" on the upper right link of the page it loads |
06:06.13 | syle | http://www.ipodhacks.com/article.php?sid=185 |
06:06.17 | syle | top google search |
06:06.35 | Smi|k | for what? |
06:06.42 | syle | ipod car adapter |
06:06.46 | Smi|k | try ipod car kit |
06:06.51 | Smi|k | or ipod car |
06:07.40 | Smi|k | or a page I really dont understand as I no longer understand google, try "car kit" |
06:07.40 | syle | hmm nice |
06:07.50 | syle | how many unique hits a day? |
06:08.10 | rezEdit | hey everyone.... I am having a bit of trouble with zaptel.conf and was wondering if anyone had any hints.... I started with a TDM04B card (4 FXO modules) and that was all good. Now I have added a TDM20B (another card with 2 more FXO modules and 2 empty spots) and am not sure how to get it to properly handle this config.... |
06:08.26 | Smi|k | not too sure, marketings never been my thing, I just like mp3's in my car a lot and always have |
06:08.31 | Smi|k | everything else is just there |
06:08.38 | syle | hehe |
06:08.43 | asterboy | can I get a car kit to turn my ipod into a wifi enabled handsfree sip phone? |
06:08.45 | rezEdit | no matter what I add to zaptel.conf, it can not work with more than 4 channels |
06:08.59 | Smi|k | asterboy, you will need a new toaster oven for that |
06:09.04 | asterboy | lol |
06:09.11 | fugitivo | rezEdit: did you edit zapata.conf also? |
06:09.33 | asterboy | sure your edititing the right zaptel.conf file? |
06:09.56 | rezEdit | fugitivo: not yet, but right now I am just running ztcfg so does zapata.conf matter? |
06:10.05 | Smi|k | nevermind on weird result, damn desktop search |
06:10.16 | fugitivo | rezEdit: pastebin your zaptel.conf |
06:10.32 | rezEdit | asterboy: yeah, the output from ztfg changes as I make edits |
06:10.44 | Smi|k | wait, no its not, I dont get it, any experience? |
06:10.57 | asterboy | ok good. |
06:11.22 | rezEdit | fugitivo: it's just basically fxsks=1-4 |
06:11.22 | rezEdit | fxsks=5,6 |
06:11.39 | rezEdit | I have also tried 7,8 instead of 5,6 |
06:11.49 | asterboy | pastebin time |
06:11.53 | rezEdit | and 1,2 with 3-6 |
06:11.57 | rezEdit | ok |
06:12.50 | asterboy | include /proc/interrupts and modules |
06:13.02 | rezEdit | http://pastebin.ca/32931 |
06:13.07 | fugitivo | rezEdit: fxoks=1,2 for your fxs |
06:13.21 | fugitivo | err |
06:13.27 | fugitivo | fxoks=5,6 |
06:14.25 | rezEdit | oh damn |
06:14.37 | fugitivo | ok, does it work? |
06:14.45 | rezEdit | TDM20B - that's FXS modules |
06:14.58 | fugitivo | yes |
06:15.06 | fugitivo | you should use fxoks, not fxsks |
06:15.25 | rezEdit | yeah, what I meant to have was a TDM02B |
06:15.30 | rezEdit | my bad, sorry |
06:15.56 | fugitivo | oh |
06:16.03 | fugitivo | then |
06:16.20 | rezEdit | I already have a TDM04B and wanted 2 more FXO modules, but I guess outr operations person ordered the wrong card. |
06:16.44 | fugitivo | you want fxo or fxs? |
06:16.46 | fugitivo | i'm confused |
06:17.13 | asterboy | Do you want to sell those FXO modules? |
06:17.17 | rezEdit | heh.... I need to bring in 6 analog phone lines so I need 6 FXO modules |
06:17.28 | fugitivo | ok |
06:17.37 | rezEdit | but now I have 1 card with 4 FXO's and 1 card with 2 FXS |
06:17.59 | drray | digium sells the modules seperate |
06:18.00 | fugitivo | do you have a TDM02B or TDM20B? :) |
06:18.11 | asterboy | Oh, you meant to say they miss ordered FXS...you WANT FXO |
06:18.28 | rezEdit | yeah |
06:19.12 | rezEdit | crap. I was hoping to go live with this tomorrorw. |
06:19.21 | fugitivo | sleep time |
06:19.23 | fugitivo | bye |
06:19.29 | asterboy | night |
06:19.32 | rezEdit | fugitivo: g'nite. |
06:19.47 | asterboy | If you burn out R13 and R14, you turn the FXS into an FXO |
06:20.32 | *** join/#asterisk doushanes (n=doushane@c-67-173-1-227.hsd1.il.comcast.net) |
06:25.44 | asterboy | :P |
06:26.05 | drray | asterboy? |
06:26.36 | asterboy | yes |
06:26.57 | drray | that's the only difference between an FXS and FXO module for the TDM400p? |
06:27.21 | asterboy | no...I was just seeing if anyone was paying attention. |
06:27.33 | asterboy | would be nice though. |
06:28.52 | *** join/#asterisk kimosabe (n=kimosabe@201.135.10.173) |
06:29.09 | kimosabe | how can i make my asterisk box grafical |
06:29.25 | drray | you can use FOP |
06:29.26 | ptiggerdine | amp |
06:29.32 | drray | flash operator panel |
06:29.49 | drray | but I don't like amp or fop really as solutions |
06:30.06 | asterboy | The X100P clone cards are actually intel modems...those have the R13 and R19 resistors that can be removed to make the card show as a "genuine |
06:30.12 | drray | yeah |
06:30.13 | asterboy | version |
06:30.22 | drray | or you can modify the zaptel source |
06:30.42 | kimosabe | what is the site for amp |
06:31.16 | drray | let me qualify what I said about amp and fop, I personally don't care for them.. |
06:31.22 | rezEdit | asterboy: heh, you almost had me. I'll have to call viopsupply in the morning. I just checked the order and we did ask for a TDM02B but they sent a TDM20B |
06:31.34 | rezEdit | bastards! |
06:31.55 | asterboy | they killed kenny! |
06:32.05 | drray | I think digium was too clever for their own good in naming those cards that |
06:32.11 | mog_home | what you say |
06:32.16 | mog_home | to clever.... |
06:32.18 | mog_home | indeeeeed |
06:33.04 | *** join/#asterisk testmachine (n=assink@ip237-239-58-62.adsl.versatel.nl) |
06:33.06 | mog_home | hehe i have my new server up and running |
06:33.11 | rezEdit | agreed |
06:33.15 | mog_home | all off my little raq box |
06:33.33 | asterboy | what is the html front end for * ?? |
06:33.43 | ptiggerdine | * = ? |
06:33.52 | drray | * = asterisk |
06:33.59 | mog_home | none |
06:34.08 | ptiggerdine | AMP |
06:34.14 | mog_home | amp is the closest thing to it |
06:34.14 | asterboy | ~* |
06:34.16 | jbot | it has been said that * is asterisk |
06:34.19 | ptiggerdine | asterisk management portal |
06:34.34 | mog_home | config files arent hard |
06:34.36 | mog_home | enjoy em |
06:34.38 | ptiggerdine | ~anyhtinguseful |
06:34.54 | asterboy | ~beer |
06:34.56 | jbot | extra, extra, read all about it, beer is not just for breakfast anymore |
06:34.57 | ptiggerdine | ~AMP |
06:34.58 | jbot | amp is probably a web based interface for configuring Asterisk. See http://amp.coalescentsystems.ca/ |
06:35.00 | drray | I don't think AMP makes asterisk easier |
06:35.07 | mog_home | no it doesnt |
06:35.09 | ptiggerdine | drray, agree |
06:35.09 | mog_home | just harder |
06:35.31 | doushanes | speaking of html front ends for asterisk, anyone try signates asterisk manager? |
06:35.44 | ptiggerdine | and the permission issues sux |
06:35.49 | asterboy | ya, I can't leave my CLI |
06:36.09 | ptiggerdine | comerical propiety shit |
06:36.55 | ptiggerdine | FLOSS!!! |
06:37.52 | asterboy | For those that want AMP up and running fast. Coalescent Systems will install AMP for a small setup fee of $249 USD |
06:37.55 | rezEdit | I started knowing NOTHING about this stuff and am glad I never tried AMP or anything else. |
06:38.01 | rezEdit | dig into the configs |
06:38.08 | rezEdit | you learn more and have more control. |
06:38.11 | asterboy | so where is the download if I don't want them to install it? |
06:38.38 | drray | I see why people want AMP |
06:38.48 | asterboy | why |
06:38.49 | drray | AMP just does not do it for me |
06:39.12 | rezEdit | I wish there was something nice for a front end so that mere mortals could update configs as needed. |
06:39.18 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
06:39.31 | *** join/#asterisk MikeJ__ (n=ircatjer@d14-69-8-30.try.wideopenwest.com) |
06:39.48 | drray | asterboy - because non * people want a way to manage extensions, my boss is already beholden to teh telco guys on the mitel.. I'm not sure he wants to be beholden to a unix guy too |
06:40.04 | ptiggerdine | there is a OSX interface to asterisk that I like ( or looks a lot better than AMP) |
06:40.11 | rezEdit | We have been using a 3com NBX 100 system which has a half-decent web front end.... something anyone on our staff can use when need be |
06:41.01 | rezEdit | ptiggerdin: OS X interface? you mean the sunrise people? |
06:41.22 | drray | I just have not found a GUI that has grabbed me, I've dinked around with making a gkrellm tool for asterisk, and with a ncurses tool, but vi and asterisk -r work best for me |
06:41.37 | asterboy | I have used simple text forms to be sent via email to change many things in linux. |
06:42.36 | asterboy | zsh?!? |
06:42.51 | rezEdit | LOL It's what I learnt on |
06:42.57 | asterboy | haven't heard that word since SCO |
06:43.08 | rezEdit | hah! |
06:43.16 | asterboy | lol |
06:43.54 | rezEdit | it's the standard here at work, and we have a lot of sheel scripts that depend on it so.... I don't have much choice. |
06:44.00 | rezEdit | er shell. |
06:44.32 | asterboy | there are substantial differences. |
06:45.11 | drray | can't you just run the zsh scripts from bash? |
06:45.12 | ptiggerdine | rezEdit, how well does asterisk intergration into the 3com NBX 100/ |
06:45.20 | rezEdit | yeah, as I have had opportunity to discover.... I guess I don't use the shell for enough stuff that it matters |
06:45.48 | rezEdit | ptiggerdin: No integration here. Completely replacing the NBX 100 which is about to die. |
06:45.55 | ptiggerdine | oh ok |
06:46.15 | drray | I'm waiting for my last two mitels to die |
06:46.16 | ptiggerdine | bugger eh, would have being keen to do that. |
06:46.24 | asterboy | I guess you could zsh from bash. |
06:46.29 | asterboy | visa versa |
06:46.33 | bsdfreak | heh |
06:46.37 | rezEdit | asterboy: yup |
06:46.39 | bsdfreak | =] |
06:46.50 | ptiggerdine | something about support for legacy with asterisk makes it fun. |
06:46.57 | drray | yes |
06:47.04 | drray | or at least interesting for an afternoon |
06:47.18 | asterboy | I'm glad I skipped the whole PRI thing. |
06:47.23 | rezEdit | our NBX 100 is in BAAAAAAD shape. We can still make and recieve calls, but while customers can leave voicemail, we can't retrieve it.... it just hangs up on us. The IMAP interface to voicemail has stopped working, and we have not been able to access the web interface for weeks. |
06:47.28 | asterboy | IP phones all the way baby! |
06:47.35 | *** join/#asterisk MikeJ[Laptop] (n=ircatjer@d14-69-8-30.try.wideopenwest.com) |
06:47.39 | *** join/#asterisk AFK1 (n=itsme@203.81.233.76) |
06:47.43 | drray | I hate IP phones |
06:48.14 | rezEdit | The hard disk is dying a slow death, me thinks. |
06:48.22 | ptiggerdine | ah ok |
06:48.26 | asterboy | why so? |
06:48.38 | drray | rezedit - have you thought of imaging the hard drive and doing a transplant? |
06:49.17 | rezEdit | drray: no. The system is like 5 years old, and we wanted to go VoIP anyways. |
06:50.09 | rezEdit | drray: Plus I have no idea how to work with that thing. There is a serial interface, but we have very few PCs here, and I wouldn't know what to do once I got a connection. There is very little documentation out there, since 3com wants to charge for all support. |
06:50.16 | drray | I have a serious hard on for zaptel |
06:50.31 | asterboy | lol |
06:51.26 | asterboy | I just love the concept that you can take your IP office phone with you anywhere in the world and stay connected as though you were right at the office. |
06:51.36 | drray | I also don't have a building setup where voip works |
06:51.42 | drray | I have to use copper pairs |
06:51.50 | asterboy | right...zaptel it is! |
06:52.00 | rezEdit | asterboy: that's another plus. We have a few people off-site and this will save big on long-distance. |
06:52.05 | asterboy | (reference to brain candy |
06:52.19 | asterboy | totally |
06:52.20 | drray | I like how zaptel, has a longer range. is line powered.. |
06:52.32 | asterboy | wait till the wifi phones take off. |
06:52.34 | drray | and just needs a pair |
06:52.34 | *** join/#asterisk mog_home (n=mogorman@user-24-236-84-48.knology.net) |
06:52.52 | asterboy | I do like that too. |
06:52.54 | rezEdit | asterboy: I already have them logging in to the system using X-Lite until I can send them their IP 501's. |
06:52.57 | drray | our building has metal door frams, metal siding and flourescent lighting.. wifi will not work here |
06:53.06 | asterboy | nice. |
06:53.08 | drray | hotel faraday |
06:53.14 | asterboy | how much did you get the 501s for? |
06:53.25 | asterboy | lol, hotel faraday! |
06:53.31 | tainted_ | omg atacomm is such a ripoff when it comes to shipping |
06:53.38 | asterboy | one big capacitor |
06:53.40 | rezEdit | asterboy: not sure.... from voipsupply, but they had a black friday sale. |
06:53.43 | AFK1 | hii all |
06:53.54 | AFK1 | need some information about a server required |
06:53.59 | drray | the building was built in 1993, so they went out of their way to not use cat3 or better cable in the walls |
06:54.00 | rezEdit | asterboy: we had an order in but cancelled it in time and reordered under the sale. |
06:54.00 | AFK1 | for putting up Digium 2400 model cards |
06:54.30 | asterboy | didn't know about the black friday sale...is that every friday? |
06:54.37 | AFK1 | we want to put 24 FXS and 24 FXO in one server |
06:54.50 | rezEdit | asterboy: no, just the Friday after Thanksgiving Thursday |
06:54.50 | AFK1 | can any one suggest what server specs are required |
06:54.50 | AFK1 | ??? |
06:54.51 | asterboy | no problem |
06:55.26 | AFK1 | NEED HELPPPPPPPPPPPPPP guys |
06:55.27 | asterboy | darn, missed that...no big deal...got my IP 500s for $95 delivered. |
06:55.37 | mog_home | 1 ghz |
06:55.44 | mog_home | any machine you can buy these days |
06:55.44 | asterboy | AFK1, 80386 |
06:55.45 | rezEdit | asterboy: Yikes! from where!?!?!? |
06:55.46 | mog_home | go to walmart |
06:55.49 | mog_home | get an emachine |
06:55.54 | mog_home | if you want to be a cheapo |
06:55.56 | asterboy | ebay |
06:55.56 | AFK1 | asterboy :) |
06:55.58 | mog_home | if you want to spend it |
06:56.05 | rezEdit | asterboy: I am sure we paid like $200 a pop. |
06:56.06 | mog_home | get a dell or some server |
06:56.07 | drray | almost any PC will work with with one zaptel card |
06:56.13 | mog_home | exactly |
06:56.29 | asterboy | AFK1, seriously...you can use any new machine...plenty of jam. |
06:57.03 | asterboy | If you can afford the 2400s you can afford a dual core P4 |
06:57.06 | AFK1 | asterboy: we want to terminate 24 - live PSTN in 24 FXO card and then forward these out of the 24 - FXS card in the same server, in the process we want to record all ongoing calls, basiclaly we r trying to use ASterisk for recording services |
06:57.10 | drray | I need to find some chumps to buy my TDM4xxp cards so I can get a new TDM2400 cared |
06:57.34 | asterboy | AFK1, buy 2 servers exactly the same. |
06:57.36 | j4m3s | what revs? |
06:57.39 | asterboy | share load them. |
06:57.47 | rezEdit | AFK1: They don't lie. I have asterisk running with 1, 4-port card on old Power Mac hardware and it never even begins to get anywhere near coughing while serving 30 phones. |
06:58.13 | j4m3s | drray, what rev tdm? |
06:58.24 | asterboy | ya, the digium cards your buying, preferrably with hardware echo, will do most of the work. |
06:58.28 | *** join/#asterisk krisguy (n=krisguy@h216-170-039-057.adsl.navix.net) |
06:58.31 | AFK1 | two servers, the solution we r putting is for like 48 ports , that means 4 servers in all?? |
06:58.40 | AFK1 | rezEdit :) i m sure they dont |
06:58.42 | drray | I'd need to look, but it was from the dev kit back in the day.. when it included x100p |
06:58.54 | j4m3s | do you still have the x100ps? |
06:58.55 | AFK1 | we will definitely go for the echo cancellation cards, |
06:58.58 | rezEdit | asterboy: Seriously, where did you get your IP501's? |
06:59.04 | drray | j4m3s - yes |
06:59.19 | j4m3s | would you bundle them together? |
06:59.20 | asterboy | seriously...ebay |
06:59.34 | rezEdit | asterboy: ahhh. Ok. |
06:59.39 | AFK1 | asterboy: Do you think the way we r trying to use asterisk for recording calls is good enough? |
06:59.41 | *** join/#asterisk magic_1 (n=maig_1@dsl-165-149-85.telkomadsl.co.za) |
06:59.44 | drray | hell yeah, I only have 1 x100p |
06:59.46 | rezEdit | asterboy: Thanks. |
06:59.47 | asterboy | yes |
06:59.54 | asterboy | no prob |
07:00.06 | magic_1 | lo all |
07:00.19 | drray | AFK1 - that will work |
07:00.29 | AFK1 | i was thinking of putting up 2 x 2400 in server class machine with 1 x Xeom , 1 GB RAM, for recording 24 ports |
07:00.36 | rezEdit | Time for me to head home folks. Have a good night, all (or day as the case may be). |
07:00.39 | asterboy | AFK1, totaly...you only need 1 server...but 2 is nice in case one goes offline...you can transfer cards to the remaining working one. |
07:01.41 | drray | I don't know that you need that much hardware either (cpu wise) |
07:02.01 | asterboy | I concentrate on redundancy. |
07:02.26 | asterboy | When phones go down....everyone gets excitted...when computers go down...they go for a coffee break. |
07:02.51 | magic_1 | hi all -AFK1-would u possibly know how i would integrate the AMP with my existing asterisk box |
07:04.08 | asterboy | from what I have heard here... "rm -rf AMP*" and "asterisk -c" seem to be popular. |
07:04.43 | drray | if he wants to run AMP, let him |
07:04.59 | drray | hell, he could be the one to make it suck less |
07:05.04 | asterboy | lol |
07:05.08 | magic_1 | hehehhe |
07:05.28 | drray | and I'm not saying AMP sucks, just personal preference |
07:05.55 | drray | the orrielly book is the best GUI for asterisk |
07:06.08 | asterboy | not tried it...but after reading the consensus here, I'll be happy with a shell prompt. |
07:06.28 | drray | asterboy - it's like using a cruise missle to remove a stump |
07:06.36 | asterboy | lol |
07:06.40 | drray | it tries to be everything |
07:06.51 | magic_1 | its just that i am rightfully a newbie and,asterisk is not the easiest to get going ,i heard that AMP might be the eassiest to start of with |
07:07.08 | drray | asterisk by itself is the easiest |
07:07.16 | magic_1 | i wanted to use asterisk and not asterisk@home cause i want to do this properly |
07:07.18 | asterboy | ya gotta agree there. |
07:07.19 | magic_1 | heheheheh |
07:07.27 | drray | AMP,FOP and anything else won't work if your base asterisk install is befuct |
07:07.36 | magic_1 | heheheheh] |
07:08.06 | magic_1 | nah my asterisk is working ,just i dont know how to work it |
07:08.13 | asterboy | no...the proper way would be to buy an Asterisk book a developer kit and play around. |
07:08.19 | magic_1 | and i would really like to learn |
07:08.23 | drray | my big beef with FOP is that he uses flash, and flash plus firefox has killed every OS I've ever run it under |
07:08.29 | magic_1 | that is true |
07:08.45 | brookshire | at least it's not java |
07:08.49 | magic_1 | i have heard similar things |
07:08.52 | drray | hey, I like java |
07:08.57 | magic_1 | heheheh |
07:08.58 | brookshire | i'm sorry |
07:08.58 | brookshire | :) |
07:09.14 | brookshire | i like programming in java.. but i hate running it |
07:09.15 | brookshire | lol |
07:09.25 | drray | well, I don't like java, but I like being able to sell applications to linux/osx/windows/bsd/solaris |
07:09.39 | magic_1 | my thing is i am having trouble (as stupid as it is )setting up a simple ext |
07:09.42 | magic_1 | heheheheheh |
07:09.48 | magic_1 | hehehehehe |
07:10.02 | drray | magic - what kind of hardware do you ahve? |
07:10.47 | magic_1 | AMD 64 3500+ 1gig corsair 80 serial ATA and GIGabyte motherboard |
07:10.56 | drray | I'm sorry, what kind of asterisk hardware? |
07:11.12 | magic_1 | also running fedora core 3 64bit with asterisk 1.2 |
07:11.19 | *** join/#asterisk MikeJ[Laptop] (n=ircatjer@d14-69-8-30.try.wideopenwest.com) |
07:11.27 | drray | so you want to setup sip extensions? |
07:11.34 | magic_1 | true |
07:11.52 | magic_1 | cause i dont have zatel hardware i use ztdummy |
07:12.01 | magic_1 | as i am sure that u know |
07:12.10 | drray | sip softphones? |
07:12.24 | magic_1 | no grandstram phone |
07:12.39 | magic_1 | i meant grandstream |
07:12.40 | magic_1 | heheh |
07:12.42 | magic_1 | sorry |
07:12.55 | drray | budgetone was part of my "dev kit" |
07:13.01 | magic_1 | nice |
07:13.08 | asterboy | are they any good? |
07:13.09 | magic_1 | i quite like this phone |
07:13.20 | magic_1 | very have had no hassle to thus far |
07:13.29 | asterboy | price sure is good |
07:13.39 | drray | they are like a $50 gun, they are better than nothing in a pinch, and should be used to get you a better gun |
07:13.52 | drray | I've since upgraded to cisco 7960's |
07:13.59 | magic_1 | my colleuge is running a *@home solution and is having no hassle with it |
07:14.15 | magic_1 | cisco is very nice but here in SA quite expensive |
07:14.32 | drray | yeah, it's expensive here too |
07:14.36 | magic_1 | hehehehe |
07:15.32 | asterboy | polycom seems to balance cisco quality with price. |
07:15.32 | *** join/#asterisk MikeJ[Laptop] (n=ircatjer@d14-69-8-30.try.wideopenwest.com) |
07:15.32 | magic_1 | yeah but it is worth it |
07:18.25 | magic_1 | my problem mainly is ,is that i am a network admin and engineer,i have never worked with asterisk before like this,only the hardware setup ,an have now dicided to get more acquanted with it so there it is all quite new to me |
07:18.26 | drray | http://www.voip-info.org/wiki-Budgetone |
07:18.26 | trixter | is there emphasis on the budget part of budgetone? |
07:18.27 | magic_1 | hehehe |
07:18.32 | drray | the budgetone is not a bad phone |
07:18.39 | drray | for what it is |
07:18.50 | magic_1 | true |
07:18.58 | magic_1 | i quite like it |
07:19.06 | drray | mine has been on the shelf unplugged for a year or so |
07:19.10 | orlok | magic_1: hey, have you heard of Obsidian? |
07:19.15 | magic_1 | yeah |
07:19.31 | magic_1 | use it u wont be sorry |
07:19.53 | magic_1 | just make sure u do all the firmware upgrades |
07:20.02 | orlok | magic_1: mate of mine went over there a few years ago to do some stuff with them |
07:20.10 | magic_1 | nice |
07:20.20 | orlok | yeah, nandos stuff |
07:20.27 | orlok | apparently thye are cool guys |
07:20.30 | magic_1 | hehehehehe |
07:20.44 | orlok | nandos, portugese chicken from south africa :) |
07:21.01 | magic_1 | heheheheheh |
07:21.10 | magic_1 | hows that for u |
07:21.28 | orlok | love nandos, we had some for lunch today :) |
07:21.57 | magic_1 | nandos rocks |
07:25.25 | drray | Yate for windows looks interesting |
07:26.03 | mog_home | ? |
07:26.59 | *** join/#asterisk Assid (n=assid@203.115.64.62) |
07:27.19 | *** join/#asterisk ORiON2012 (n=orion@cpe-70-117-0-232.satx.res.rr.com) |
07:27.26 | magic_1 | so any one got an idea how i can get am installed in asterisk |
07:28.28 | syle | questions like that will get you very hated in this channel |
07:28.43 | magic_1 | heheheheheh thanks for the heads up |
07:29.16 | drray | ahve you looked on the wiki? |
07:29.19 | drray | http://www.voip-info.org/wiki-Asterisk+Management+Portal |
07:29.28 | ORiON2012 | Quick question from a noob. I have voip service from Time Warner. Their system plugs into a POTS jack in my home and "energizes" the other jacks to enable use of my regular phone. Is something like that possible with asterisk? |
07:29.35 | magic_1 | its just i cant get a ext running on my asterisk 1.2 and from what it seems it looks so straight forward |
07:29.48 | Qwell | ORiON2012: get a/some FXS card(s) |
07:29.54 | Qwell | card/port |
07:30.04 | mog_home | QWELL! |
07:30.08 | Qwell | MOG! |
07:30.20 | Qwell | tip: noload => chan_skinny.so |
07:30.24 | Qwell | it helps ;] |
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07:31.17 | mog_home | man i need to make asterisk run with less ram |
07:31.18 | ORiON2012 | Qwell: do I have to hook the phones up directly to the card, or can I use the existing wiring in my home? |
07:31.26 | mog_home | its running pretty fat on my little server |
07:32.50 | brookshire | mog: goto sleep :) |
07:33.31 | trixter | 21M on mine with 9M resident |
07:33.32 | trixter | 1.2 |
07:33.49 | drray | I don't even look at ram |
07:33.51 | trixter | seems high for what its doing, right now there arent even any calls |
07:40.22 | mog_home | naver |
07:40.30 | mog_home | im goin in early tommorrow |
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07:47.40 | Qwell | ORiON2012: sometimes you can use the existing wiring. not always |
07:47.50 | Qwell | depends on how it comes out of the box |
07:48.38 | mog_home | man i need more ram for this box... |
07:49.20 | Qwell | ram is always good |
07:49.40 | mog_home | i have 64mb |
07:49.48 | mog_home | which is good for a 250mhz machine |
07:49.56 | mog_home | just no dma on the box.... |
07:50.07 | mog_home | meh its good enough for what i need it to do... |
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07:53.00 | asterboy | ya I have a 450 with 64Mb ram for my asterisk...doesn't skip a beat. |
07:53.19 | Qwell | man...my 110mhz router has 4x that much.. |
07:53.25 | asterboy | lol |
07:54.02 | asterboy | I'd like to get my hands on one of those PowerMac 9600s |
07:55.52 | asterboy | should put a wanted ad on Google Base |
07:58.32 | drray | ok, what is google base? |
07:58.37 | drray | url? |
07:58.48 | drray | m, |
07:58.50 | drray | er,nm |
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08:01.51 | asterboy | back |
08:02.01 | asterboy | just checking out google base. |
08:02.17 | asterboy | http://base.google.com/base/default |
08:03.53 | *** join/#asterisk Math[laptop] (n=math@modemcable148.4-81-70.mc.videotron.ca) |
08:04.39 | Math[laptop] | I setted up dundi between 2 servers but, calls are failing with "Call rejected by 10.0.0.5: No authority found" |
08:04.53 | Math[laptop] | (using iax2 as tech) |
08:04.54 | asterboy | lol "setted up" |
08:05.08 | Math[laptop] | yeah well... you know.. :P |
08:05.16 | asterboy | :P |
08:05.36 | Math[laptop] | this is without mentionning the fact that asterisk advertised itself as 127.0.0.1 on the dundi net |
08:07.04 | asterboy | have not done dundi yet...sounds like a great directory tool if you can get it working. |
08:07.24 | Math[laptop] | its just the part of dundi that handles authentication didnt register itself into the iax2 user list |
08:07.46 | Math[laptop] | I can always failover by creating a "dundi" user in iax2 with no password, but its kind of a security flaw |
08:08.11 | Math[laptop] | especially when pstn routes are announced |
08:08.30 | asterboy | I'll have to get into that. |
08:08.44 | asterboy | night for now. |
08:08.46 | Math[laptop] | nite |
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08:09.25 | *** join/#asterisk Gunnar (n=gunnar@bkkb-gw.bitcon.no) |
08:09.37 | Math[laptop] | hey MikeJ[Laptop], did u play with dundi on head? |
08:11.16 | perlmonkee | I'm having an Asterisk+SipPhone.com problem. I can place calls out through SipPhone.com, and `sip show registry` says I have registered, but when I (or others) try to call my SipPhone number, it goes straight to SipPhone voicemail, as though I weren't registered. If I have my soft phone register directly with SipPhone.com, I can recieve calls no problem. |
08:11.23 | perlmonkee | Can anyone shed any light on this? |
08:11.38 | perlmonkee | Is there something I have to do other than reguster => user:passwd@proxy01.sipphone.com ? |
08:11.47 | perlmonkee | register* |
08:11.58 | Math[laptop] | uhm yeah |
08:12.03 | Math[laptop] | register => user:pass@host/exten |
08:12.12 | Math[laptop] | context=the_context_you_want |
08:12.19 | Math[laptop] | and define the exten |
08:12.24 | Math[laptop] | oh ok |
08:12.32 | perlmonkee | I've followed the examples on Voip-info.org |
08:13.00 | Math[laptop] | well... whats you dialplan for the incoming call? |
08:13.43 | perlmonkee | just a moment. |
08:14.28 | perlmonkee | exten => ${SIPPHONEUSERID},1,Macro(stdexten,100,SIP/perlmonkee) |
08:15.19 | Math[laptop] | can u pastebin your extensions.conf |
08:15.52 | Math[laptop] | because I think thats not the way to call stdexten, but Im not sure til I use ael with home-made-stuff(tm) |
08:16.33 | perlmonkee | well, here is an example of a working stdexten call: |
08:16.36 | perlmonkee | exten => 100,1,Macro(stdexten,100,SIP/perlmonkee) |
08:16.49 | Math[laptop] | ah |
08:16.56 | perlmonkee | that one works no problem. |
08:17.18 | Math[laptop] | and, when you run asterisk and someone calls you, what do you see on the CLI |
08:17.23 | Math[laptop] | (with a lot of -vvvvvvvvvv) |
08:17.36 | perlmonkee | If someone tries to call my SipPhone number - I see nothing. |
08:17.40 | drray | set verbose 10 |
08:18.00 | Math[laptop] | then try again |
08:18.00 | perlmonkee | okay, set to 10 - trying again. |
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08:18.07 | perlmonkee | (it was at 4) |
08:18.29 | Math[laptop] | k |
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08:19.25 | perlmonkee | I see my outgoing call being setup and then sipphone.com answering it. |
08:19.38 | perlmonkee | and sipphone.com gives me the "This user is currently offline" message. |
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08:20.18 | perlmonkee | nothing about any incoming calls. |
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08:22.33 | Math[laptop] | perlmonkee, whats your number |
08:22.44 | Math[laptop] | maybe your not allowed to receive and send a call at the same time |
08:22.59 | Math[laptop] | lemme call you |
08:23.34 | perlmonkee | 17476278960 |
08:23.52 | delox99 | Math[laptop]: are you behind a router or firewall? |
08:26.25 | perlmonkee | my asterisk box is not. |
08:26.26 | Math[laptop] | working? |
08:26.26 | Math[laptop] | ringing here |
08:26.26 | perlmonkee | I am not recieving any call. |
08:26.26 | Math[laptop] | 1-747-627-8960 |
08:26.26 | perlmonkee | nothing on the console =/ |
08:26.26 | perlmonkee | yup |
08:26.26 | Math[laptop] | what about sip debug peer [peername] |
08:26.26 | perlmonkee | okay, just a moment. |
08:26.29 | perlmonkee | enabled |
08:26.29 | perlmonkee | Destroying call '3d40662a245d05342bc05ddc59af4042@192.168.1.1' |
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08:26.42 | perlmonkee | thats all I've seen since enabling it. |
08:26.49 | *** part/#asterisk doushanes (n=doushane@c-67-173-1-227.hsd1.il.comcast.net) |
08:27.00 | Math[laptop] | k calling |
08:27.45 | perlmonkee | woah - okay - a bunch of crap. |
08:28.19 | Math[laptop] | didnt call ye tlol |
08:28.29 | trixter | gee yahoo is doing voip now to fight google.. wonder if its like sonys 'voip plus' wherei ts only on net calls and doesnt in any way allow a regular phone call (that is all I have seen from yahoo) to further confuse people.. a HIGH percentage of people that are voip aware dont know you can call PSTN numbers with any voip |
08:28.45 | perlmonkee | it was all registration crap |
08:28.55 | Math[laptop] | damn the cat5 wire of my ATA broke |
08:28.56 | Math[laptop] | brb |
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08:31.52 | Math[laptop] | dialing |
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08:33.35 | Math[laptop] | got anything |
08:33.55 | Math[laptop] | answer+hangup? |
08:35.18 | perlmonkee | nothing |
08:35.52 | perlmonkee | just registration crap. |
08:36.02 | perlmonkee | >_< |
08:36.18 | perlmonkee | okay, I'm going to start all over with sipphone - erase everything I have and write it all again. |
08:37.04 | perlmonkee | sip.conf relevant stuff: |
08:37.28 | perlmonkee | regester => number:passwd@proxy01.sipphone.com/perlmonkee |
08:37.47 | perlmonkee | [sipphone] |
08:37.50 | perlmonkee | type=peer |
08:38.05 | perlmonkee | host=proxy01.sipphone.com |
08:38.14 | perlmonkee | canreinvite=no |
08:38.18 | Math[laptop] | ~pastebin |
08:38.19 | jbot | methinks pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca/ |
08:38.23 | perlmonkee | =( |
08:38.23 | perlmonkee | sorry. |
08:38.29 | Math[laptop] | :) |
08:39.02 | Math[laptop] | er... register => user:pass@host/extension |
08:39.09 | Math[laptop] | the exten must be a digit |
08:39.13 | Math[laptop] | or a number |
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08:42.38 | magic_1 | at the host part can i put the host ip instead is that possible |
08:42.51 | perlmonkee | http://pastebin.com/453746 |
08:43.54 | magic_1 | perlmonkee,may i ask u a question |
08:44.13 | perlmonkee | You may. |
08:44.19 | perlmonkee | (how polite) |
08:45.29 | perlmonkee | okay... so I made that change - verbosity is set to 10, debugging for peer sipphone is enabled. |
08:45.41 | perlmonkee | if anybody wants to try a call - I'd appreciate it. |
08:45.59 | magic_1 | thankz |
08:48.31 | perlmonkee | I'm not too great with this myself just yet - but setting up an extension is pretty easy. I'm sure if you ask a more specific question I, or any number of more experienced people here would be happy to try and answer it. |
08:49.03 | magic_1 | thanks i appreciate |
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09:01.30 | niZon | someone tell me why x-lite uses more ram than windows explorer |
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09:03.31 | pif | hi, with what directive can I test an IAX link for latency? |
09:06.42 | drray | pif, i've dialed in to my asterisk box with the milliwatt() app and listened for defects |
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09:09.48 | many | pif: qualify=yes ; iax2 show peers |
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09:13.05 | kks | sorry to jump topic, Is anyone get MYSQL cmd work asterisk realtime? i'm working with Asterisk 1.2.1 and Asterisk-addon 1.2.1 |
09:15.51 | pif | many : thanks, qualify=yes checks for peers every 60s, right? |
09:16.12 | pif | but not the latency to the peer |
09:19.51 | many | it does check the the latency then |
09:20.34 | pif | so qualify=1000 will skip a host over 1000ms latency? |
09:21.08 | many | i think it declares a host lagged when > 1s and unreach when > 2s |
09:21.38 | pif | I'm googling but can't find authoritative info on that |
09:21.45 | many | tx-rico 172.21.253.254 (S) 255.255.255.255 4569 OK (34 ms) |
09:21.54 | many | UTSL :) |
09:22.18 | pif | ? |
09:22.27 | many | use the source luke |
09:22.32 | pif | ah :) |
09:22.39 | many | the only authoritive information you will ever find is in the source. |
09:22.50 | many | you know, the config files whose extension is .c |
09:22.59 | many | :-P |
09:23.12 | pif | dude |
09:23.34 | many | Yea? :) |
09:23.53 | *** part/#asterisk ORiON2012 (n=orion@cpe-70-117-0-232.satx.res.rr.com) |
09:23.57 | pif | how do you test for lagged status before Dial'ing ? |
09:24.14 | many | oh. i dont. |
09:24.32 | pif | or does Dial return a failure? |
09:24.51 | many | no idea, honestly. i didnt ever had calls with high lat. |
09:25.14 | many | the 30ms you see above is my sucky dsl over a vpn |
09:25.21 | many | or rather a vpn over my sucky dsl |
09:26.22 | many | or rather: iax2 over tcp over vpn over pppoe (1000/128kbit) |
09:26.30 | many | err, i give up, you get the idea. :) |
09:26.38 | Qwell | kludge |
09:26.53 | pif | better to let iax2 packets on the wild internet |
09:27.23 | many | the whole fucking IT is a single sucking kludg.e |
09:27.35 | many | sorry. should watch my word. |
09:27.36 | many | s |
09:27.40 | pif | layers of encapsulations mess with error correction of voip protocols |
09:28.13 | many | error correction? in a tcp based protocol? uh. well.. |
09:29.13 | pif | you vpn packets are tcp based (will wait and retry) when the aim of UDP/voip is immediate delivery |
09:29.23 | many | youre wrong |
09:29.26 | many | my vpn runs over udp |
09:29.40 | many | and iirc iax2 isnt udp, but tcp. |
09:29.44 | Qwell | udp |
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09:29.47 | pif | no sir |
09:29.53 | pif | Qwell : yes |
09:30.02 | many | okay, so why did i think that iax2 did not suck, then? |
09:30.05 | many | mhh. |
09:30.14 | pif | udp is good for voice |
09:30.14 | Qwell | sccp is tcp ;] |
09:30.17 | mog_home | iax is the coolest.... |
09:30.26 | Qwell | mog_home: go to bed |
09:30.31 | mog_home | never |
09:30.32 | many | pif: yeah, basically yea. |
09:30.51 | Qwell | I got stuck watching a police chase by my house... |
09:30.54 | many | but udp is a good shoot-in-the-foot with alot of soho routers and stuff. |
09:31.06 | Qwell | went on so long though, that the helicopter ran out of gas, and had to take off. heh |
09:31.23 | mog_home | lol thats funny |
09:32.07 | pif | "out of gas" + "had to take off" = core dumped |
09:33.25 | iDunno | Qwell: surely it had already taken off, hence it running out of gas... ;) |
09:33.58 | iDunno | 9.30am and I'm already slapped, interesting. |
09:34.11 | pif | ok, notice to all west-coasters : herbal tea, jerk off and to bed |
09:34.51 | pif | attaboy |
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09:36.38 | many | dig chicks instead of just jerking off |
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09:38.38 | pif | chicks, you mean all that flesh around the pussy? |
09:39.09 | many | Yah. |
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09:42.13 | mog_home | anyone good with iptables |
09:42.23 | mog_home | i have a really simple firewall i want to set up |
09:42.40 | mog_home | eh never mind |
09:42.45 | mog_home | found info i needed |
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10:15.24 | perlmonkee | Does anyone know of a service that you can have call you to test your incoming call routing? |
10:15.34 | perlmonkee | I know I used one about a week ago, but I can't remember where it was. |
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10:18.41 | shido6 | what do you mean perlmonkee ? |
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10:20.09 | Dandre | Hello all, |
10:20.30 | Dandre | I have a problem with fop and asterisk 1.2 |
10:21.08 | RoyK | fop? |
10:21.13 | RoyK | FoIP? |
10:23.22 | drray | flash |
10:23.26 | mog_home | flash operator panel |
10:23.27 | drray | operator |
10:23.31 | drray | panel |
10:23.55 | perlmonkee | shido6: I'm looking for a form I can put my SIP address in and then some server that processes this form will initiate a call to me. |
10:24.00 | perlmonkee | thus testing my incoming call routing. |
10:27.10 | perlmonkee | I'd still like to find a service like I mentioned though (especially since I used one not very long ago) |
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10:34.45 | Dandre | sorry guys I have been disconnected |
10:35.23 | Dandre | did you seen my previoux post about FOP? |
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10:42.20 | Dandre | Is there any one here? |
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11:08.01 | KriS83 | Hi... could someone have a look at this, and tell me why ${foo} is empty? or what I am doing wrong? |
11:08.05 | KriS83 | http://pastebin.ca/32944 |
11:09.15 | *** join/#asterisk MikeJ[Laptop] (n=ircatjer@d14-69-8-30.try.wideopenwest.com) |
11:09.19 | *** join/#asterisk MassiveBlue (n=Massive@80.243.63.82) |
11:09.34 | MassiveBlue | hi there |
11:09.47 | KriS83 | Hi |
11:10.03 | iDunno | Ah ha! That's what I was looking for, the s option to ChanIsAvail :) |
11:11.35 | *** join/#asterisk MikeJ__ (n=ircatjer@d14-69-8-30.try.wideopenwest.com) |
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11:22.40 | dario77 | hi |
11:23.37 | dario77 | i have a problem concerning asterisk & rtcp, can you help me? |
11:23.51 | *** join/#asterisk MikeJ[Laptop] (n=ircatjer@d14-69-8-30.try.wideopenwest.com) |
11:25.10 | *** join/#asterisk MikeJ[Laptop] (n=ircatjer@d14-69-8-30.try.wideopenwest.com) |
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11:34.55 | *** join/#asterisk netwind (n=pavlo@tech02.off.comlink.ru) |
11:35.13 | netwind | hmm |
11:35.30 | netwind | jerjer doesn't appears here ? |
11:35.50 | netwind | wich openh323 version are recommended for now? |
11:36.13 | netwind | i can't build char_h323 in new asterisk 1.2.1 |
11:36.42 | netwind | 1.0.10 build is mailfunctional |
11:40.19 | shido6 | chan_h323 ? |
11:40.45 | netwind | of course chan_h323 |
11:41.39 | *** join/#asterisk MikeJ[Laptop] (n=ircatjer@d14-69-8-30.try.wideopenwest.com) |
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11:46.31 | *** join/#asterisk zoa (n=kkk@pirus.securax.be) |
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12:03.32 | shido6 | is there a nagios check plugin? |
12:03.46 | shido6 | for mysql |
12:04.27 | *** join/#asterisk coppice (n=chatzill@108.166.17.210.dyn.pacific.net.hk) |
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12:08.38 | zoa | shido there is |
12:08.41 | zoa | ive seen one before |
12:18.05 | *** join/#asterisk mjmac (n=mjmac@pdpc/supporter/active/mjmac) |
12:35.41 | MassiveBlue | Dec 7 18:56:14 WARNING[5135] chan_iax2.c: Firmware file '/var/lib/asterisk/firmware/iax/iaxy.bin' fails checksu |
12:35.41 | MassiveBlue | m |
12:35.44 | MassiveBlue | oops |
12:36.51 | MassiveBlue | i get warnings "Firmware file '/var/lib/asterisk/firmware/iax/iaxy.bin' fails checksum" - anybody knows something about this warning? |
12:37.12 | KriS83 | Hi... could someone have a look at this, and tell me why ${foo} is empty? or what I am doing wrong? |
12:37.14 | KriS83 | http://pastebin.ca/32944 |
12:44.15 | *** join/#asterisk nagl (n=nagl@213.235.241.6) |
12:44.44 | nextime | anyone using astbill? |
12:46.23 | mrtwister | nextime: i use |
12:46.33 | iDunno | hmmm |
12:46.41 | mrtwister | nextime: and areski and sometimes mcc :) |
12:47.31 | iDunno | or rather, how it works. |
12:47.52 | nextime | mrtwister : i'm trying to test it, but i have some trubble creating the database tables, 20 asv* tables are "in use" from mysql using the default .sql to install, is it a common error or so on? |
12:48.29 | iDunno | (hmm, not as I want it to, damn) |
12:48.31 | mrtwister | nextime: you installed mysql 5? |
12:48.36 | nextime | mrtwister : of course yes |
12:48.38 | mrtwister | nextime: and what os you have? |
12:48.46 | nextime | debian sarge, mysql 5.0.16 |
12:48.51 | *** join/#asterisk tld (n=tld@33.84-48-78.nextgentel.com) |
12:48.58 | mrtwister | nextime: hm, i have no issues on debian |
12:49.25 | mrtwister | nextime: on fedora core got error once, but later corrected... do full reinstall of mysql |
12:50.09 | nextime | mrtwister : mysql is working great, i don't think that is the problem ( i have a 18 gigs complex db on the db server.. ) |
12:50.10 | mrtwister | nextime: btw i still not know, imho areski at present is better, and with astbill you cannot use AMP portal |
12:50.36 | nextime | mrtwister : i hate AMP, it don't reflect my need, areski the same |
12:50.40 | mrtwister | nextime: week ago i installed astbill, no problems at all. |
12:51.01 | mrtwister | nextime: try mcc also :) there is no AGI |
12:51.16 | mrtwister | nextime: compiled c program. www.paskambink.lt/mcc/ |
12:51.53 | nextime | i will look mcc |
12:51.59 | mrtwister | nextime: astbill i instaleld and test and maybe will use. good that it integrated to CMS drupal, where i can use other modules and do e-commerce and other things |
12:52.36 | nextime | mcc need postgres at the momenty |
12:53.12 | nextime | i don't want to install another db server while i have a mysql cluster in production |
12:53.46 | mrtwister | nextime: you have to use simple machine for gaming and tests :) |
12:53.51 | mrtwister | nextime: or vmware |
12:54.21 | mrtwister | nextime: anyway, i had no problems with astbill, but there at forum lot of topics, seems product is not finished until end. |
12:54.53 | nextime | mrtwister : yes, but i must test something that i can use in the production environment :) |
12:56.11 | RoyK | zoa: ping |
12:56.21 | nextime | i think that i will return to use my simple mod_python based web administration, i'm tired to test something that don't work at 100% or isn't good for my use, anyway, thanks for your opinions. |
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13:01.48 | *** part/#asterisk chapeaurouge (n=chap@85.201.81.201) |
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13:07.06 | onas | mrtwister :) |
13:07.20 | mrtwister | onas: ? |
13:07.41 | onas | mrtwister entusiastic page www.paskambink |
13:08.58 | sivana | Yahoo Messenger adding computer-to-phone capabilities |
13:10.23 | *** join/#asterisk bintut (n=bintut@202.128.40.243) |
13:11.08 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/sustaining/cpm) |
13:12.09 | *** join/#asterisk gnosys (n=ksford@ip68-9-201-250.ri.ri.cox.net) |
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13:20.02 | *** part/#asterisk cpm (n=Chip@pdpc/supporter/sustaining/cpm) |
13:23.43 | *** join/#asterisk asteriskmonkey (n=phil@69.156.197.242) |
13:24.00 | gnosys | anyone here ever connected a Cisco 7960 phone to the Free World Dialup service? Could you offer some pointers? I just got through connecting to FWD using X-Lite from CounterPath, so my next step is seeing if I can do the same with my hardware SIP client (the Cisco 7960... loaded with SIP firmware, version 7.1). |
13:24.48 | asteriskmonkey | gnosys: you shoul have no problem if you if you use the same settings |
13:26.36 | gnosys | the thing that's confusing me is the differently-named settings in the Cisco. I have the 7960 working with my DHCP and TFTP servers, but when it boots up, it's telling me that it's unprovisioned. In the phone status message center (LCD screen), it's complaining about W351 and W350 (unprovisioned proxy_emergency and unprovisioned proxy_backup, respectively) |
13:27.55 | gnosys | And I don't get a dial tone when I pick up the handset... |
13:28.19 | bintut | what is the best digium card with 4 FXOs you can recommend? i find confusing the digium website. |
13:30.13 | asteriskmonkey | bintut : get a tdm400 with 4 fxo's on |
13:30.26 | asteriskmonkey | gnosys: have you provisioned it yet usiung the tftp? |
13:30.48 | asteriskmonkey | gnosys: open a log window to see if it grabs the config from the tftp |
13:31.06 | *** join/#asterisk Dandre (n=Dandre@was59-3-82-236-48-30.fbx.proxad.net) |
13:31.12 | Dandre | Hello, |
13:31.25 | RoyK | ~seen zoa |
13:32.57 | jbot | zoa is currently on #asterisk (1h 46m 26s). Has said a total of 2 messages. Is idling for 1h 24m 16s |
13:32.58 | gnosys | open a log window... by telnetting into the phone? or is that accessible from the lcd? |
13:32.58 | asteriskmonkey | bintut: http://www.digium.com/index.php?menu=product_detail&category=hardware&product=TDM400P |
13:32.58 | zoa | im here royko |
13:32.58 | RoyK | oki |
13:32.58 | asteriskmonkey | gnosys: where is your tftp program? run the monitor there to see if the phone connects and grabs the file from it |
13:32.58 | RoyK | zoa: i just tried the latest version of the jb |
13:32.59 | Dandre | I am trying to use ALERT_INFO with 1.2. It seems to be different in 1.2 as in 1.0.7 but I can't find the docs on voipinfo :-( |
13:32.59 | zoa | patch3 ? |
13:32.59 | gnosys | done that already. it does load the SIPDefault.cnf and SIPmacaddress.cnf. I guess that maybe I've set something incorrectly in one of those two files... |
13:32.59 | zoa | what happened ? |
13:32.59 | RoyK | zoa: yeah |
13:32.59 | RoyK | zoa: sound is terrible |
13:33.04 | RoyK | zoa: that is, incoming audio on the asterisk server |
13:33.24 | RoyK | zoa: can i have your email address once more? i can email you a monitored file |
13:33.30 | zoa | can you post a message on mantis with the flow ? ( i mean what protocol to what protocol + what options so that slav can have a look at it ? |
13:33.34 | zoa | joachim@securax.be |
13:33.51 | asteriskmonkey | gnosys: most likley a spelling mistake in your config file :) |
13:34.07 | RoyK | zoa: sent |
13:34.31 | zoa | thnx |
13:34.49 | zoa | be sure to post the message on mantis |
13:34.54 | bintut | asteriskmonkey: thanks for the link |
13:35.07 | gnosys | I guess that could be asteriskmonkey... i thought I reviewed it pretty carefully for that. any chance i could get you to look it over for me? maybe send you a file or something? |
13:35.18 | RoyK | zoa: left channel is me talking from a sip ata (that black one) using a 1024/200 adsl line. the other side is a gsm phone |
13:35.53 | zoa | is the quality only bad with patch3 ? or also with patch 2 and 1 ? |
13:36.16 | asteriskmonkey | bintut: no prob, if your in canada let me know i can give you the number for this distributor :) |
13:36.18 | RoyK | iirc same with the older ones |
13:36.23 | zoa | aha |
13:37.30 | gnosys | the other thing, asteriskmonkey, is that I'm using STUN in X-Lite to go through my natted firewall by linksys to the fwd proxy, and I don't see settings in the 7960 for doing that... |
13:37.50 | zoa | RoyK, can you also try it with the other 2 models ? |
13:39.14 | asteriskmonkey | gnosys: you are behind a firewall or nat i take it then? |
13:40.04 | Dandre | I have found, there must be an underscore before ALERT_NFO! |
13:40.29 | MassiveBlue | i get warnings "Firmware file '/var/lib/asterisk/firmware/iax/iaxy.bin' fails checksum" - who knows something about this warning? |
13:42.09 | asteriskmonkey | sounds like you have a corrupt file |
13:42.20 | asteriskmonkey | download it from cvs |
13:42.55 | asteriskmonkey | asterisk automatically upgrads the iaxy firmwares as soon as iaxys connect to that asterisk server :) sweet |
13:44.31 | *** join/#asterisk kletter-matze (n=kletter-@212.126.219.82) |
13:44.33 | *** join/#asterisk Uther_P (n=uther_p@66.180.120.82) |
13:44.45 | *** part/#asterisk Uther_P (n=uther_p@66.180.120.82) |
13:44.46 | MassiveBlue | asteriskmonkey: i downloaded this file from CVS on 16 Nov 2005 17:46:59 -0000 |
13:44.54 | kletter-matze | <PROTECTED> |
13:45.02 | kletter-matze | if I dial out, I can see at the cli "Executing SetCallerID("SIP/sysadmin-105c", "+49711123440") |
13:45.11 | kletter-matze | is there anything else I have to do? |
13:45.23 | *** join/#asterisk [TK]D-Fender (n=joe@toronto-HSE-ppp4122655.sympatico.ca) |
13:46.03 | bintut | asteriskmonkey: i'm in south east asia.. |
13:47.54 | *** join/#asterisk tengulre (n=tengulre@222.90.92.54) |
13:49.07 | tengulre | hi,all |
13:50.03 | tengulre | anybody active? |
13:51.10 | zoa | im not here |
13:52.35 | RoyK | zoa: will do |
13:53.17 | Cinen | anyone here had any luck with Dundi? I asked in #Dundi but nobody is home |
13:54.16 | *** join/#asterisk Assid (n=assid@203.115.64.62) |
13:56.05 | *** join/#asterisk nesys (n=nesys@2001:1418:1a6:0:20d:93ff:fe28:3ef8) |
13:56.50 | RoyK | wtf? |
13:56.55 | RoyK | same noisy sound without the jb |
13:57.00 | tengulre | anybody building CALL CENTER with asterisk ? |
13:57.58 | asteriskmonkey | massiveblue: try downloading from head |
13:58.11 | mrtwister | tengulre: asterisk@home is small call center :) |
13:58.27 | mrtwister | tengulre: it is possible in other words... but need lot of work |
13:58.38 | asteriskmonkey | tengulre: call centers are easy to build with asterisk :D |
13:58.53 | gnosys | asteriskmonkey: yes, behind a linksys natted firewall. |
13:59.08 | MassiveBlue | asteriskmonkey: okay I'll try |
13:59.11 | mrtwister | how to compile zaptel on debian. i have installed kernel sources, but have to do some operations before starting compilation. where i can find faq? |
13:59.17 | gnosys | should I not expect the 7960 to be able to handle that with FWD then? |
13:59.56 | asteriskmonkey | gnosys: open the corresponding porst then on the router then and make sure nat is mapping the right ports to that device |
14:00.39 | gnosys | So I guess that implies that the 7960 cannot do STUN like the X-Lite client does? |
14:00.43 | asteriskmonkey | mrtwister: use google for compiling files on linux |
14:00.56 | asteriskmonkey | gnosys: nope :) |
14:01.11 | gnosys | OK. I'll try your suggestion then. Thank you! :-) |
14:01.21 | asteriskmonkey | the 7960 is a much older phone.. i havnt seen there latest software though |
14:01.23 | asteriskmonkey | no prob |
14:01.30 | tengulre | asteriskmonkey: how many lines does asterisk support? |
14:01.33 | mrtwister | asteriskmonkey: not found proper doc.. still looking :) |
14:01.56 | mrtwister | tengulre: no limits, depends of 'style' of usage and cpu |
14:02.03 | asteriskmonkey | mrtwister: do you have gcc and all the other things installed required to build things properly? |
14:02.16 | mrtwister | asteriskmonkey: yes. |
14:02.20 | asteriskmonkey | tengulre: as many as you can build a system to handle |
14:03.25 | asteriskmonkey | mrtwister: go to the directory you have the make/source files in and type make clean; make install; |
14:03.47 | asteriskmonkey | mrtwister: or on bsd style boxes one like :) make clean install |
14:03.53 | tengulre | asteriskmonkey: I want to building a 30 analog lines and 10 clients? |
14:04.15 | [TK]D-Fender | tengulre : very easy |
14:04.28 | [TK]D-Fender | tengulre : You mean 30 phone lines (from telco)? |
14:04.29 | asteriskmonkey | tengulre : get an t1/e1 card |
14:04.31 | tengulre | mrtwister: are you use digium cards? |
14:04.46 | asteriskmonkey | get something like the t11op |
14:04.54 | tengulre | [TK]D-Fender: yes! |
14:05.00 | asteriskmonkey | so you want your users to have 30 lines 3 lines each :) nice |
14:05.04 | iCEBrkr | Werd up! |
14:05.12 | tengulre | asteriskmonkey: does it support chinese telcom? |
14:05.14 | mrtwister | asteriskmonkey: i know it. but i have to do some thinks with kernel source to prepare it for usage |
14:05.16 | [TK]D-Fender | tengulre : where are you located? |
14:05.30 | tengulre | [TK]D-Fender: china! |
14:05.32 | asteriskmonkey | tengulre: what the hell is chinese telecom ? :P |
14:05.33 | mrtwister | tengulre: i like sangoma, but digium also ok, works :) |
14:05.36 | tengulre | Xi'an of china |
14:05.45 | asteriskmonkey | ah... should do fine |
14:05.54 | asteriskmonkey | do you use t1 or e1 there/ |
14:06.10 | [TK]D-Fender | Get off of analog lines and get a T1/E1/J1 (whichever is appropriate to your area) digital link and you're set. |
14:06.26 | tengulre | asteriskmonkey: I m worried it's different chinese |
14:06.52 | iCEBrkr | mrtwister: How's the config/setup for the Sangoma cards? |
14:06.55 | [TK]D-Fender | tengulre : I'm sure your area uses one of the 3 major standards. Do you use normal POTS phones on those lines? |
14:07.04 | asteriskmonkey | tungulre : t1 and e1 are standards ask which your telco provides |
14:07.13 | [TK]D-Fender | iCEBrkr : WANCFG = Creemy goodness :) |
14:07.29 | mrtwister | iCEBrkr: ? it is well described in docs, lot of examples online... |
14:07.44 | iCEBrkr | I'm thinking of getting one. |
14:07.53 | iCEBrkr | But I'm familiar with Digium's cards. |
14:07.54 | tengulre | asteriskmonkey: really? |
14:08.22 | iCEBrkr | It's unfortunate that I don't have much time to muck around. |
14:08.43 | asteriskmonkey | tengulre: yep |
14:08.46 | mrtwister | iCEBrkr: almost same sangoma |
14:08.54 | coppice | tenguler: why do you want to use analogue lines? are E1s hard to get in your area? |
14:09.07 | mrtwister | iCEBrkr: only need to install wanpipe, all is on CD, no errors all works :) |
14:09.09 | asteriskmonkey | think he just dosnt know telecom stuff |
14:10.03 | [TK]D-Fender | iCEBrkr : very easy to setup and a very solid experience. |
14:10.14 | tengulre | anybody know who are seller in this channel |
14:10.34 | [TK]D-Fender | tengulre : Not for telephone service in our are I would think. |
14:10.39 | [TK]D-Fender | you* |
14:10.42 | [TK]D-Fender | you* |
14:10.44 | [TK]D-Fender | your* |
14:10.48 | [TK]D-Fender | damn... can't type today! |
14:11.27 | *** join/#asterisk eye69 (i=magnus@upcore.net) |
14:11.39 | docelmo | its cause your dialplan is 20k |
14:12.07 | asteriskmonkey | tengular: i work for the distributor of digium in canada |
14:12.25 | [TK]D-Fender | docelm0 : Ah the bitter face fo jealousy..... |
14:12.33 | [TK]D-Fender | :D |
14:12.34 | tengulre | asteriskmonkey: are you programmer? |
14:12.36 | docelmo | fo.. I dont think so.. |
14:12.38 | Assid | [TK]D-Fender!!! |
14:12.52 | asteriskmonkey | tengulre: yes i program manny things for asterisk and other things too |
14:12.56 | [TK]D-Fender | hey Assid sorry I didnt get a chance to forward that config file... |
14:13.05 | tengulre | asterikmonkey: cool!! :) |
14:13.06 | [TK]D-Fender | 1.6.2 right? |
14:13.16 | Assid | hehe.. np.. i actually came across some weird issue yday.. but solved it |
14:13.18 | Assid | yeah 1.6.2 |
14:13.23 | iCEBrkr | mrtwister: haha, I suppose if I got zaptel crap compiled, configured and working, anything else can't be much more complicated :) |
14:13.47 | Katty | [TK]D-Fender: thx. |
14:13.53 | Katty | [TK]D-Fender: just what i always wanted. |
14:13.59 | tengulre | asteriskmonkey: my boss let me to select a call center in linux platform, so I want to select asterisk? but I 'm a begginer for it! |
14:14.01 | [TK]D-Fender | Assid : is it working now or would you still like my file? |
14:14.01 | mrtwister | iCEBrkr: yes. in general, sangoma will use same settings |
14:14.13 | iCEBrkr | So what's a Sangoma A104 cost? |
14:14.16 | [TK]D-Fender | Katty : "If thine eye offends thee"... |
14:14.17 | iCEBrkr | ballpark |
14:14.23 | Assid | oh that bug i had was for something else.. like using only line 3 for another user |
14:14.23 | tengulre | asteriskmonkey: I want to got many documents of asterisk! |
14:14.26 | [TK]D-Fender | iCEBrkr : Par with Digium |
14:14.29 | asteriskmonkey | tengulre: its not hard to learn |
14:14.31 | iCEBrkr | [TK]D-Fender: Right on |
14:14.34 | asteriskmonkey | lots of help online |
14:14.35 | Assid | apparently i had to register it under line 2 |
14:14.39 | mrtwister | tengulre: i think, only asterisk |
14:14.48 | mrtwister | tengulre: for callcenter |
14:14.49 | [TK]D-Fender | Assid : definately no need for that. |
14:14.58 | [TK]D-Fender | let me package it up for you |
14:15.01 | Katty | [TK]D-Fender: mew? |
14:15.12 | mrtwister | tengulre: what you want to do |
14:15.17 | [TK]D-Fender | there there..... |
14:15.42 | Assid | well h thats what happened .. i had line 1-2 on 1 user.. and line 3 for another user.. but in the web control.. i jhad to register line 3 under line 2 to get it working |
14:15.43 | Katty | [TK]D-Fender: you don't pet normal girls, do you? |
14:15.49 | Assid | does katty bite? |
14:15.58 | [TK]D-Fender | :O |
14:16.01 | docelmo | yes |
14:16.12 | Katty | Assid: why don't you ask her and find out. |
14:16.18 | docelmo | or so I have heard.. |
14:16.34 | docelmo | hehe |
14:16.37 | Assid | Katty: you bite? |
14:16.42 | Katty | Assid: yes. |
14:16.47 | Assid | damn |
14:16.48 | tengulre | mrtwister: I want to building a call center platform, about 30 analog lines and 10 clients! client application use Microsoft windows , when a caller incoming then auto pop a menu on desktop... .. |
14:17.10 | [TK]D-Fender | Assid : how many differnt regs do you really what the phone to have? |
14:17.36 | Assid | only that unit.. needed 2 registrations.. others.. single registration..- 3 lines |
14:17.39 | [TK]D-Fender | Assid : all of mine have 1 reg with multiple line keys attributed to that reg. Subsequent calls just use the next available line key |
14:17.50 | [TK]D-Fender | Why 2 reg? |
14:18.05 | Assid | have a seperate number for that extension.. |
14:18.09 | mrtwister | tengulre: tell more. what you said is present in asterisk and in related applications |
14:18.17 | Assid | direct number |
14:18.42 | [TK]D-Fender | Assid : you mean DID directed to your phone? |
14:19.00 | Assid | yep.. asterisk gets it.. forawrds call to that phone |
14:19.11 | Assid | no ivr.. nothing |
14:19.24 | [TK]D-Fender | Assid : definately no need. All of my workers have DID's and 1 reg. You could clean things up a LOT.... |
14:19.27 | Assid | oh damn,..i was supposed to ask shido to pickup something |
14:19.36 | *** join/#asterisk Feral_Kid (n=Feral@red-corp-200.56.96.178.telnor.net) |
14:19.47 | Assid | [TK]D-Fender: HuH ? |
14:20.56 | [TK]D-Fender | Assid : My phones use 1 reg using 2-6 line keys (non-lines are speed dials, buddy watch, etc). They have an internal ext (for dialing and callerid purposes) but have DID pointed to them from the outside. You don't need a 2nd reg for that. |
14:21.07 | SkramX | Hi all. |
14:21.15 | Assid | right.. |
14:21.42 | [TK]D-Fender | Assid : would allow you to trim a lot of unnecessary stuff from sip.conf and trim your dialplan. |
14:21.43 | Assid | i have a IVR thing.. for all internal extensions.. |
14:28.53 | Feral_Kid | Any of the NuFone folks around? |
14:29.25 | mutilator | Druken: radio is dead! |
14:29.44 | zoa | nufone just left |
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14:42.19 | *** join/#asterisk wundaboy (n=asdf@c-67-164-107-68.hsd1.or.comcast.net) |
14:43.19 | wundaboy | can somone refresh me on the differentce between a friend,user,peer? |
14:46.57 | zoa | wundaboy |
14:46.57 | mrtwister | wundaboy: peer = asterisk --> somewhere; user = client -> asterisk, friend = asterisk <---> gateway |
14:47.36 | wundaboy | so if im setting up my voip provider (voip-pstn), i should set it up as a 'user'? |
14:48.12 | mrtwister | wundaboy: peer or friend |
14:48.17 | wundaboy | oops, i meant to say peer |
14:48.28 | iCEBrkr | beer? |
14:48.29 | wundaboy | if i just send calls, it should be a peer right? |
14:48.35 | mrtwister | wundaboy: but friend ok too, look also to context= |
14:48.46 | wundaboy | and if i send and recieve it should be friend? |
14:49.22 | wundaboy | also, in the iax.conf what does the 'context' directive do? |
14:51.00 | *** join/#asterisk Ariel_ (n=Ariel@adsl-068-157-125-248.sip.mia.bellsouth.net) |
14:51.13 | Ariel_ | Morning everyone |
14:52.16 | [TK]D-Fender | wundaboy : For clients its the context which controls what they can dial, for providers if the context that will receive incoming connections. |
14:54.37 | Katty | hewwo Ariel_ |
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14:54.52 | Ariel_ | Katty, hello, good morning |
14:55.09 | Katty | Ariel_: not good morning |
14:55.12 | Katty | Ariel_: bad morning |
14:55.16 | Katty | Ariel_: temp is dropping. :< |
14:55.26 | Katty | Ariel_: and it's snowing :< |
14:55.27 | Ariel_ | Katty, sorry to hear it. |
14:55.55 | Ariel_ | I am having 2 of the worst weeks that I can remember. But lets not go there. I hope your day gets better. |
14:56.28 | Katty | Ariel_: :>>> |
14:56.39 | Katty | Ariel_: hope your weeks getting better too (= |
14:57.04 | Ariel_ | as soon as we finally do our closing on the house it will. |
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15:09.24 | zoa | anybody from sineapps / freedam here ? |
15:13.44 | *** join/#asterisk vaewyn (i=freeman@mail.parrishmachine.com) |
15:14.09 | vaewyn | Boooyahhh! (or in other words... long time no see... and good morning :} ) |
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15:15.24 | vaewyn | so... anyone know if you can prevent native bridging on ZAP channels? especially on a per call basis? |
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15:24.14 | blop | vaewyn, why would u do so ? |
15:27.11 | vaewyn | blop: our nortel doesn't have the option to allow forwarding of off campus calls to off campus... and when it sees that bridge attempt to take place it slaps it down... and for some reason I would like to avoid the 4500$ for the option :P |
15:27.57 | vaewyn | blop: it works fine for locally generated calls... just not external |
15:28.01 | *** join/#asterisk Seldon1975 (n=someone@gatekeeper.radintl.com) |
15:28.17 | blop | :) |
15:28.39 | vaewyn | and so far I don't see a canreinvite type deal for zapata :} |
15:28.53 | vaewyn | let alone a per call thing |
15:32.06 | *** join/#asterisk shmaltz (n=chatzill@mail.dmaven.com) |
15:32.50 | Seldon1975 | hi.. no matter what i put for 'fxsks' and 'fxoks' in zaptel.conf, I get "ZT_CHANCONFIG failed on channel 1: No such device or address (6)" |
15:33.30 | ManxPower | Seldon1975, and lsmod shows the modules loaded? |
15:34.31 | Seldon1975 | manx - you're right; it was because the driver wasnt loaded |
15:34.31 | vaewyn | long time no see |
15:34.42 | ManxPower | hello vaewyn |
15:35.23 | *** join/#asterisk Tili (n=Tili@202-133-67-86-dialup.sat.net.pk) |
15:35.39 | vaewyn | Hey ManxPower... you don't by any chance know if there is a way to control native zap bridging do you? |
15:35.44 | Seldon1975 | soz' |
15:35.46 | Tili | anybody has any URL for a symbian VoIP client. I have heard of Buzz2Talk but never found it |
15:36.26 | ManxPower | vaewyn, I can't imagine why you would do that. The only difference between a native zap brige and a zap non-bridge is the format of the audio internally to asterisk, as I understand it. |
15:38.26 | vaewyn | ManxPower: Well... When * attempts to native bridge a call that is off campus -> nortel -> * -> nortel -> off campus The nortel slaps it down... where it doesn't slap down a on-campus -> nortel -> * -> nortel -> off-campus.. We are almost positive it is because we don't have the option on the nortel for forwarding external to external |
15:38.59 | ManxPower | vaewyn, Perhaps you are confused about what a Zap native bridge is. |
15:39.01 | vaewyn | So I was trying to find a way for it to not inform nortel that these are the same call |
15:39.20 | ManxPower | vaewyn, Asterisk should not inform the nortel anything. |
15:39.34 | ManxPower | vaewyn, what interface? CT1/CE1 or PRI? |
15:39.36 | *** join/#asterisk Assid (n=assid@203.115.64.62) |
15:39.54 | vaewyn | PRI - esf,b8zs |
15:40.08 | ManxPower | vaewyn, What makes you think Asterisk is infoming the nortel? |
15:40.16 | bintut | is the Digium TDM04B PCI card works on a standard 32bit PCI slots of the ordinary athlon/p4 motherboards? |
15:40.30 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
15:40.31 | ManxPower | bintut, it should |
15:40.56 | bintut | ManxPower: so, it doesn't need to have a pci-x or pci-express slots, right? |
15:41.10 | vaewyn | ManxPower: because I can call both directions... and nortel phone sourced calls it works... outside sourced calls the nortel requests a hangup the second I see the native bridge line |
15:41.12 | ManxPower | bintut, The part number you quoted does not. |
15:41.28 | ManxPower | vaewyn, You have some other problem |
15:41.57 | ManxPower | vaewyn, you might have a native brigde if you put something like "t" as an option on the dial line. |
15:42.26 | ManxPower | ..er...it might not native bridge if you put something like "t" as an option on the dial line. But that's not the problem. |
15:42.31 | vaewyn | ManxPower: well... as far as * is concerned the calls should be identicle... both are coming from the PRI and are going back out the same PRI |
15:42.56 | vaewyn | Nortel is the only one that "knows" the sources |
15:42.59 | ManxPower | vaewyn, no, each call is unique. |
15:43.11 | ManxPower | You have two calls, the call into asterisk and the call out of asterisk |
15:43.29 | bintut | ManxPower: i just want to confirm to you.. the Digium TDM04B PCI card is good enough for 4 FXOs and all other FXS are VoIP phones.. that card is good voice quality, right? |
15:43.30 | vaewyn | Well... lets just say... they both follow an identicly dialplan path... and the nortel requests a hangup |
15:44.09 | ManxPower | bintut, that should work, but a T-1/E-1 would me more reliable. |
15:44.50 | ManxPower | vaewyn, what happens when you put "t" on the Dial line. |
15:45.00 | ManxPower | (t should prevent a zap native bridge) |
15:45.00 | bintut | ManxPower: we only have 3 phone lines from a local PSTN.. the other one is reserve.. do i need a t1-e1 for this? |
15:45.06 | vaewyn | attempting that now |
15:45.18 | [TK]D-Fender | bintut : For that few lines a TDM400 card would be fine |
15:45.25 | ManxPower | bintut, A T-1/E-1 will always be more reliable than analog lines. But analog lines will work. |
15:45.30 | [TK]D-Fender | bintut : Definately no need for T1/E1 |
15:45.30 | bintut | ManxPower: but if you recommend it because of best voice quality output, i should consider it |
15:45.41 | ManxPower | Unfortunatly, in my experience the TDM400Ps crash about once a month. |
15:45.43 | [TK]D-Fender | bintut : Would be expensive.... |
15:45.49 | bintut | yeah |
15:45.53 | bintut | thanks guys.. |
15:45.59 | [TK]D-Fender | You could always use SPA-3000's for your lines... |
15:46.31 | *** join/#asterisk taec (n=phil@eventhorizon.hosting365.ie) |
15:47.00 | *** join/#asterisk marc32422 (n=marc3234@206-248-133-186.dsl.teksavvy.com) |
15:47.26 | bintut | [TK]D-Fender: i'm thinking of buying Sipura SPA-2100 Analog Telephone Adapter instead |
15:47.40 | taec | I've got Asterisk setup with SIP based IP phones and a ZAP module connected to PRI .... if I'm at the asterisk CLI, is there an easy to way to see the CID's of the incoming calls and which extensions/queues they're currently in? |
15:47.54 | ManxPower | taec, no. |
15:48.04 | *** join/#asterisk af_ (n=af@ip-142-84.sn1.eutelia.it) |
15:48.17 | ManxPower | taec, you can see the callerid number on the console "Accepted call from '1234567890'" etc |
15:48.42 | taec | Ok, is there a hard way? I've noticed that AMP's flash panel can do it fairly easy. I know it interacts with the Asterisk Call Manager. Can I interact with that to get the information I'm looking for simply enough? |
15:49.04 | *** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com) |
15:49.04 | *** mode/#asterisk [+o anthm] by ChanServ |
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15:50.22 | taec | AMP's flash panel seems to be able to grab the external CID of the caller fine and even associates it with an extension (e.g. if they're currently on the phone to them) ... so I presume it's possible without a lot of hassle. I simply presumed it could be done through the CLI as well. How would I go about seeing the CLI of callers who are currently active? no way? |
15:50.44 | *** join/#asterisk jjuhlh (n=Irix@nat.kollegienet.dk) |
15:51.25 | ManxPower | taec, you are wrong. |
15:51.34 | bintut | do you think this mobo <http://tinyurl.com/8hd25> is good enough for a simple PBX with a Digium TDM04B PCI card? |
15:51.40 | ManxPower | AMP prolly gets the info via the manager interface. |
15:51.58 | taec | ManxPower: So it is a lot of hassle? or it's not possible? |
15:52.06 | ManxPower | taec, it's a lot of hassle. |
15:52.22 | taec | ManxPower: yes, it does get it via the manager interface ... but does that not provide the same interface as the CLI? |
15:52.30 | ManxPower | You can always put a Noop in the dialplan so you can see the callerid when the call hits the Noop |
15:52.42 | ManxPower | taec, no, the manager interface is different from the CLI. |
15:53.10 | taec | OK, I should probably look into that a little bit more so |
15:53.19 | ManxPower | taec, why do you want to know the callerid info? |
15:53.57 | taec | Small project I thought of, was that an agent here could login to a page, with an extension number and password and if a customer dials in off a known number, their details could be displayed on-screen |
15:54.06 | anthm | However, I like to make cli commands in anticipation of using it from manager which is possible and is flexable because you can use it from the command line and from a remote program |
15:54.40 | taec | antha: I was under the impression that it was only CLI commands you could use through the Manager, I must have been mistaken! |
15:54.44 | marc32422 | ne1 can recommend a switch for connecting asteirsk servers together? |
15:54.53 | [TK]D-Fender | bintut : I'd pick a better board if I were you. An ASUS or GigaByte.... something with a better bios so you can guarantee it getting a seperate IRQ |
15:55.19 | ManxPower | taec, there is a readme or .txt file documenting manager as past of the Asterisk source. |
15:55.32 | anthm | in reality it's the only one you need manager has a way to install a clumsy manager action thing but the Command action is more than enough |
15:55.33 | ManxPower | past == part |
15:55.41 | *** join/#asterisk asterisk99 (n=dunoowhy@modemcable169.194-130-66.mc.videotron.ca) |
15:56.00 | coppice | Its sad. there was a time when ECS made *the* premium motherboards |
15:56.27 | anthm | and when USR made nice modems =D |
15:56.46 | asterisk99 | anyone familiar with installing zaptel on Gentoo? My ztcfg will not run after reboot :( |
15:56.53 | coppice | I remember when an ECS 486 motherboard cost US$3000 :-) |
15:57.11 | *** join/#asterisk citats (n=james@bgp925576bgs.brghtn01.mi.comcast.net) |
15:57.19 | ManxPower | taec, think of it this way: The manager interface gives you access to all CLI commands, but it also has additional information it can provide to the app that uses the manager interface. |
15:57.44 | [TK]D-Fender | coppice : And when swooping pteradactyl's were the greatest treat to humanity! |
15:57.48 | [TK]D-Fender | threat* |
15:58.10 | iCEBrkr | asterisk99: I know what your problem is.. |
15:58.33 | coppice | i guess you've been to that creationist musuem :-) |
15:58.34 | taec | ManxPower, and what you're telling me is that some of the additional information it provides is CLI information? |
15:58.38 | ManxPower | iCEBrkr, the fact that he doesn't understand the Gentoo boot process? |
15:58.38 | iCEBrkr | asterisk99: You're running Genpoo. |
15:58.42 | taec | apologies, CID information |
15:58.47 | ManxPower | taec, Yes. |
15:58.52 | iCEBrkr | ManxPower: :P |
15:59.00 | taec | Much appreciated ManxPower, thanks for your help! |
15:59.04 | asterisk99 | iCEBrkr: hahaha |
16:00.04 | docelmo | Yippie! |
16:00.24 | docelmo | CentOS is the only way to go.. Except Dovecot.. That program F*!?ing SUCKS! |
16:02.31 | iDunno | CentOS is the only way to go if you're clinically insane. |
16:02.43 | iDunno | (or really really like rpm dependency hell) |
16:04.01 | vaewyn | ManxPower: t option doesn't change it... still hangs up |
16:04.02 | docelmo | iDunno, its not that bad.. works very well for me.. |
16:04.17 | *** join/#asterisk DrDeke (i=dekemar@deculator.engin.umich.edu) |
16:04.41 | vaewyn | debian rocks |
16:04.45 | docelmo | The biggest problem I had with 4.2 was Dovecot and getting it to work.. I ended up chucking it and going with washington.edu's ipop3 and imapd |
16:04.48 | vaewyn | IMH(BC)O |
16:05.09 | docelmo | I like gentoo.. but its a pain in the ass to install |
16:05.33 | iDunno | you had problems with Dovecot?! how?! is simple! |
16:06.24 | asterisk99 | docelmo: do you have asterisk running on gentoo? |
16:09.12 | fugitivo | asterisk99: i do |
16:09.22 | fugitivo | docelmo: it's not a pain in the ass |
16:09.32 | docelmo | nope.. |
16:09.53 | *** join/#asterisk kokey (n=ubunture@201.153.63.79) |
16:09.58 | asterisk99 | fugitivo: have any problem getting ztcfg to run after reboot? |
16:09.59 | docelmo | iDunno, dunno.. I installed it and everything configured it and couldnt get my email clients to work with it. |
16:10.06 | fugitivo | asterisk99: no |
16:10.23 | docelmo | I would if I had the patients to install it. |
16:13.52 | coppice | if you were a doctor you might be able to get your patients to install it |
16:14.18 | iDunno | weird. |
16:15.19 | *** join/#asterisk paryl (n=paryl@216-201-177-82.res.logixcom.net) |
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16:17.13 | paryl | i can dial 800 numbers from asterisk, but i can't dial a specific one that a user needs. if i dial it with a cell phone, it works. any idea what could cause behavior like that? |
16:17.45 | paryl | it comes back (after about 6-7 seconds) with "== No one is available to answer at this time (1:0/0/0)" |
16:18.39 | SkramX | paryl: is your dial plan correctly configures? |
16:19.36 | *** join/#asterisk sysdebug (n=sysdebug@200.163.193.247) |
16:20.26 | ManxPower | paryl, what is the value of HANGUPCAUSE after the call ends? |
16:21.00 | paryl | manxpower... i don't know how to get that |
16:21.16 | paryl | SkramX, it's dialed like every other 800 number, and they work |
16:22.01 | ManxPower | paryl, the priority after Dial would be Noop(HANGUPCAUSE=${HANGUPCAUSE{) |
16:22.08 | ManxPower | well, with the correct braces, of cours. |
16:23.29 | vaewyn | ManxPower: figured it out... nortel is catching on via the RDNIS somehow... need to alter that |
16:23.59 | ManxPower | vaewyn, I DID say it was not a native bridge issue 8-) |
16:24.21 | paryl | manxpower hangupcause=34 |
16:24.37 | vaewyn | ManxPower: Reason I thought it was native bridging is because it dies right on that spot |
16:25.04 | paryl | which is normal congestion... but a cell phone will ring it without issues |
16:25.04 | DrDeke | Hey guys; there is a bug in Zaptel/wctdm.c that makes it not decode pulse-dialed digits on FXS ports correctly. It is very simple to fix, you just have to change one value, but it seems like this should really just be fixed by the project administrators. Would someone please tell me the correct way to report this that would get it added in 1.2.2 or whatever? |
16:25.04 | ManxPower | paryl, 34 is "no circuit or channel available". |
16:25.13 | ManxPower | vaewyn, MANY things happen at that spot 8-O |
16:25.29 | ManxPower | DrDeke, bugs.digium.com |
16:25.33 | paryl | the wiki says "#define AST_CAUSE_NORMAL_CIRCUIT_CONGESTION 34 " |
16:25.55 | vaewyn | means it is full or unavailable |
16:26.22 | vaewyn | can also mean number blocks or such |
16:26.30 | jjuhlh | question :) when a PBX is able to handle 250 calls simultaneous, how many many custommers would you expect to have to reach this limit ? |
16:26.33 | vaewyn | kindof generic :} |
16:26.59 | ManxPower | jjuhlh, What is your customer/line ratio? |
16:27.20 | vaewyn | jjuhlh: we run 10/1 ... but universities are not normal situations |
16:27.22 | ManxPower | jjuhlh, There is NO general answer to your question. Each company/industry has different requirements |
16:28.34 | ManxPower | We run closer to 2 agents for every 1 line at some of our offices. In other offices we run 3 employes / for every 1 line |
16:28.46 | ManxPower | Our people spend their life on the phone. |
16:29.44 | paryl | this makes no sense... i dial it on a cell phone and it answers immediately, i dial through asterisk and it gives code 34 |
16:30.09 | jjuhlh | yes I know... I'm trying to make a budget and have to like explain how many customers I would expect to have after a year, so I also have think about how many PBX I have to run .. if you for example have 5000 customers after one year... just a estimate |
16:30.50 | paryl | jjuhlh: we have ~3500 active customers and service them on one t1 with 4 regular agents |
16:31.01 | *** join/#asterisk Seldon1975 (n=someone@gatekeeper.radintl.com) |
16:31.09 | DrDeke | jjulh: It completely depends what you are planning to do with this phone system, what business your customers are in, and so on. |
16:31.15 | paryl | of course, we have rollover queues for busy times, but that's normally enough |
16:31.28 | jjuhlh | okay :) |
16:32.33 | paryl | i have the txgain bumped down... could by some chance that cause the connection error? |
16:32.54 | jjuhlh | paryl, 4 regular agents.. ? |
16:33.43 | jjuhlh | we are talkink about 4 PBX ? |
16:33.47 | jjuhlh | talking |
16:34.04 | paryl | jjuhlh: no, 4 people answering the main line. that's all |
16:34.33 | jjuhlh | okay... and this is only one PBX ? |
16:34.38 | paryl | yes |
16:34.45 | jjuhlh | interresting |
16:34.52 | paryl | 35 stations, most are outgoing calls |
16:35.04 | paryl | customers call into one number, and get routed to queues |
16:35.05 | jjuhlh | yes I could imagine.. :) |
16:35.06 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-179-75-202.cust.bezeqint.net) |
16:35.12 | docelmo | Anyone know of any decient GSM Gateways compatible with asterisk? |
16:35.18 | paryl | but only 4 are logged on to the main queue at any given time |
16:35.40 | fugitivo | docelmo: www.2n.cz |
16:35.53 | docelmo | thanks.. |
16:36.00 | docelmo | I am looking to do some playing |
16:36.50 | fugitivo | it's like 800 euros |
16:37.02 | fugitivo | but it's the only voip gsm gateway i've seen |
16:37.15 | docelmo | Whats 800 euros in USD? |
16:37.24 | DrDeke | off the top of my head, about 1200 |
16:37.26 | DrDeke | err |
16:37.28 | DrDeke | maybe more like 1100 |
16:37.33 | docelmo | crap.. |
16:37.39 | DrDeke | wait no it's only $938 |
16:37.39 | paryl | haha |
16:37.44 | docelmo | I just wanted like a single line to play with. |
16:38.00 | fugitivo | docelmo: you could get a regular gsm gateway, those are cheap :) |
16:38.03 | fugitivo | not voip |
16:38.07 | DrDeke | You can do it much more cheaply, over chan_bluetooth or something for instance. |
16:38.12 | *** join/#asterisk KranZ (n=user@sme.bestline.net) |
16:38.12 | docelmo | My wife would kill me for 1200 buks.. |
16:38.22 | *** part/#asterisk KranZ (n=user@sme.bestline.net) |
16:38.26 | *** join/#asterisk KranZ (n=user@sme.bestline.net) |
16:38.26 | zoa | docelmo there is easier stuff |
16:38.26 | docelmo | Well what I need is a gsm gateway that will do SMS |
16:38.32 | zoa | aha |
16:38.32 | DrDeke | oh |
16:38.36 | zoa | use a gsm gateway :p |
16:38.40 | zoa | euh |
16:38.41 | DrDeke | I would DEFINITELY do that with bluetooth. |
16:38.43 | zoa | an sms gateway |
16:38.45 | DrDeke | Or USB |
16:38.47 | DrDeke | Or anything of that nature |
16:38.50 | fugitivo | uh |
16:38.51 | DrDeke | (If you wanted it on the cheap) |
16:38.52 | fugitivo | only sms? |
16:39.12 | fugitivo | use a webpage :) |
16:39.15 | docelmo | sms/voce |
16:39.18 | docelmo | err voice |
16:39.34 | *** join/#asterisk Dibbler_ (n=Dibbler@snaddy.plus.com) |
16:40.50 | docelmo | But mainly SMS.. Website is fine.. But that kinda defeats the purpose of being able to do SMS.. :) |
16:42.27 | zoa | behave mr! |
16:42.31 | vaewyn | hehehe.. |
16:42.32 | *** join/#asterisk Igbothom_III (n=HiltonT@static-84.217.240.220.dsl.comindico.com.au) |
16:42.38 | vaewyn | long time no see (on my part :{ ) |
16:42.52 | zoa | yes |
16:43.18 | vaewyn | man... I was gone so long you guys actually got head stable enough to release as something |
16:43.25 | zoa | :) |
16:44.16 | *** join/#asterisk znoG (n=gs@OL101-122.fibertel.com.ar) |
16:46.14 | jjuhlh | anyone have a good suggestion to a Switch/Router capable of redudant...? |
16:47.02 | docelmo | redudant what? |
16:47.25 | asteriskmonkey | cisco catalysy 10000 is nice |
16:47.44 | docelmo | well took my idea.. :( |
16:47.54 | jjuhlh | sorry redundancy.... I'm danish hehe |
16:48.07 | *** join/#asterisk Kokey (n=Kokey@201.153.63.79) |
16:48.31 | [TK]D-Fender | jjuhlh : redundant what? |
16:48.46 | fugitivo | is any way to see for example, what modules has a card without having a complete /etc/zaptel.conf? |
16:49.51 | *** join/#asterisk Samoied (n=Samoied@popeye.opens.com.br) |
16:50.53 | jjuhlh | [TK]D-Fender, was that a joke..hehe ;) |
16:51.23 | vaewyn | How on earth does this Nortel know it is a forwarded call when I wipe the RDNIS and PRIREDIRECTREASON before I send the call over the PRI? |
16:51.30 | vaewyn | grrr |
16:51.35 | ManxPower | fugitivo, ztcfg only shows what you configured, not what modules are ACTUALLY on the card. |
16:51.39 | vaewyn | NORHELL SUCKS! |
16:51.41 | [TK]D-Fender | jjuhlh : no, redundant wht? power? wan conenction? Some other proxy service? BGP? |
16:51.49 | ManxPower | when you modprobe the card driver, it should print the info into syslog or dmesg |
16:52.46 | jjuhlh | [TK]D-Fender, it was just about a Switch/Router |
16:53.05 | fugitivo | ManxPower: if you modprobe with no config in zaptel.conf, will it display the info i want? |
16:53.43 | ManxPower | fugitivo, with or without a config, the info will still be in syslog/dmesg |
16:54.23 | fugitivo | ManxPower: i know it'll display that it found a card, but i'm not sure about the number of channels, i don't have a card right now to test it |
16:54.41 | *** join/#asterisk veepster_ (n=veepster@vbn.0012297.lodgenet.net) |
16:54.42 | ManxPower | fugitivo, then ask again when you have a card to test. |
16:54.49 | veepster_ | <veepster_> to test asterisk, do I need any additional hardware or will a basic linux server do? |
16:54.57 | ManxPower | fugitivo, it will display the channel number for each module |
16:55.12 | fugitivo | ok, thanks |
16:55.29 | vaewyn | veepster_: for softphones or network (SIP) phones you just need the server |
16:55.35 | postel | How can i configure ringing a specific number when the handset is lifted in *? in cisco is a single command, cisco calls it PLAR (Private Line Auto Ringdown) |
16:56.07 | postel | (think emergency phone) although my scenario is different |
16:56.24 | ManxPower | postel, on a zap card? |
16:56.47 | postel | ManxPower: nope, sip client |
16:56.57 | ManxPower | postel, it's configured in the SIP client then. |
16:57.12 | ManxPower | SIP devices collect digits and THEN send the call to their server. |
16:57.29 | ManxPower | postel, Cisco and SIPura support PLAR-like features. |
16:57.30 | iDunno | as * doesn't know anything about the phone being picked up other than what the phone tells it. |
16:58.32 | postel | i see, so its not possible, since * would get nothing until i DIAL something out |
16:59.22 | [TK]D-Fender | postel : its the phone's joib to do the dial though. So if your phone has an option for an "immediate" call to a specific # it would dail that as soon as it goes off-hook |
17:00.21 | veepster_ | vaewyn, can you point me towards a link where I can learn more? I want to test asterisk, Im not entirely sure what that entails yet |
17:02.33 | postel | [TK]D-Fender: the feature set of the phone is not in question here, the phone may be able to PLAR without even * being there, i was thinking if the status of the handset on/off could be monitored and trigger an action. It seems thats only possible in ZAPtel and not SIP |
17:02.50 | *** join/#asterisk SpaceBass (n=sp@static-71-251-230-2.rcmdva.fios.verizon.net) |
17:03.12 | ManxPower | postel, what specific device do you have? |
17:03.43 | [TK]D-Fender | No, not in SIP. The phone would have to send some sort of signal... In SIP the phone is "kink", maybe MGCP would be able to do something like that... |
17:03.55 | [TK]D-Fender | king* |
17:04.54 | postel | ManxPower: DECT Panasonic phones on ATAs, some cisco VICs, some 7960s and some analogs on TDMs with FXS modules, why? |
17:04.59 | ManxPower | You could also buy a SIP device that has PLAR functionality |
17:05.08 | ManxPower | postel, WHAT ATAs? |
17:05.16 | postel | ManxPower: 186 ciscos |
17:05.18 | file[desk] | darn so cold |
17:05.34 | ManxPower | Oh! EASY. The docs for the Cisco ATAs talk about PLAR stuff. |
17:05.41 | postel | they do? |
17:05.48 | ManxPower | file[desk], You are in CANADA, of course it's COLD. |
17:05.49 | postel | the Admin Guide? |
17:05.59 | ManxPower | postel, That is correct, grasshopper. |
17:06.15 | ManxPower | postel, I used to use Cisco ATAs until SIPura came out with devices that are SO much better. |
17:06.32 | postel | I've been over the Admin Guide, that was sometime ago though |
17:06.43 | file[desk] | the Sipura stuff is beautiful, same for their SIP implementation |
17:06.51 | postel | when i had trouble with the SS bits |
17:07.22 | ManxPower | postel, I even used the PLAR (they may not use that term) with the Cisco ATAs, but that was like 2 years ago. |
17:09.08 | postel | a quick google search didnt give me much, the good news is i have access to gold partners so if the feature is there they should know bout it. |
17:09.47 | postel | thanks anyways, i'll dig a bit more |
17:10.35 | KranZ | [TK]D-Fender: are the polycom 301s decent phones? |
17:16.47 | ManxPower | postel, I think Cisco calls it "hotline" and "warmline" |
17:17.50 | postel | yeah, thanks, i found it |
17:18.00 | postel | its available from release 2.14 |
17:29.09 | *** join/#asterisk jpablo (n=jpablo@200.94.130.194) |
17:29.14 | *** join/#asterisk Mother (n=mother@93.Red-80-32-127.staticIP.rima-tde.net) |
17:29.55 | jpablo | hi, is auth=plaintext still in asterisk 1.2 ? |
17:34.29 | Corydon-w | jpablo: for what protocol? |
17:34.37 | [TK]D-Fender | KranZ : Poly 301 = wase of time. |
17:34.41 | [TK]D-Fender | waste. |
17:34.56 | [TK]D-Fender | Too low-end for the money. 501+ = good |
17:35.11 | [TK]D-Fender | otherwise look at the SPA-941 or so. |
17:35.30 | Corydon-w | The 301 certainly is a good phone... but I wouldn't recommend that you mix phones at an install. People get phone envy |
17:35.55 | vaewyn | Yeah... spend the couple extra bucks for the 500+ |
17:35.58 | [TK]D-Fender | Good yes, worth the money and setup for the features? no |
17:36.31 | fugitivo | anyone using polycom 501? |
17:36.34 | Corydon-w | The setup for either the 301 or the 501 is the same |
17:36.54 | Corydon-w | Same config files, same FTP method |
17:37.01 | [TK]D-Fender | 2 line phone? ick. SPA-941 gives 4 calls, SPEAKERPHONE< and more. |
17:37.13 | Corydon-w | and if you learn to automate the process, it's even easier |
17:37.30 | [TK]D-Fender | fugitivo : I've used 501's, but run 600's and 601's here. I run all Sipura at home. |
17:38.00 | fugitivo | [TK]D-Fender: how much is the 601? |
17:38.10 | [TK]D-Fender | Corydon-w : if you're deploying other polycom models I'd say sure, but if the 30x is as high as you go, I wouldn't bother. |
17:38.12 | jpablo | Corydon-w, for sip, sorry for the delay |
17:38.24 | fugitivo | [TK]D-Fender: can you run xml apps on it? |
17:38.24 | vaewyn | Will say.. 600 only real advantage I like is the microbrowser... really wish the 500s had that... although they might by now... havn't checked firmware in 6+ months |
17:38.25 | jpablo | Corydon-w, im trying to setup grandstream early dial |
17:39.08 | Corydon-w | jpablo: uh, why? |
17:39.32 | [TK]D-Fender | fugitivo : 250$USD and I use the microbrowser on mine for all sorts of stuff... |
17:39.53 | fugitivo | [TK]D-Fender: that's the 601, the 501 doesn't have microbrowser, right? |
17:40.10 | vaewyn | the microbrowser rocks... weather tickers... stock tickers... al sorts of fun you can do with it... ours we made into timeclocks :} |
17:40.16 | [TK]D-Fender | vaewyn : I also like having 6 line keys. I use 3 for 1 reg normally, and the others for buddy-watch (presence) and speed-dials |
17:40.18 | jpablo | Corydon-w, grandstream dosn't like md5 auth |
17:40.33 | [TK]D-Fender | fugitivo : 60x = MB, 50x = nope |
17:40.47 | jpablo | Corydon-w, i need plaintext auth |
17:40.54 | Corydon-w | Should work fine |
17:40.56 | [TK]D-Fender | vaewyn : timeclocks? as in to punch-in in the morning? |
17:41.04 | vaewyn | [TK]D-Fender: That's the other thing... does presence work in asterisk now? I have been gone a while |
17:41.17 | Corydon-w | jpablo: are you just asking, or do you actually have an issue? |
17:41.25 | [TK]D-Fender | vaewyn : works real nice on my 600's and 601's (with attendand modules :D) |
17:41.26 | vaewyn | [TK]D-Fender: yep :} it drops them on a page and they fill in their ID/pass |
17:41.34 | *** join/#asterisk sahafeez (n=sahafeez@67.109.14.227.ptr.us.xo.net) |
17:41.36 | [TK]D-Fender | vaewyn : neat-o |
17:41.37 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
17:41.52 | [TK]D-Fender | vaewyn : I made one that does a presence detect for the phone company :D |
17:41.55 | vaewyn | [TK]D-Fender: * handles the presence? or external program? |
17:42.01 | vaewyn | bwahaha |
17:42.02 | [TK]D-Fender | s/phone/whole |
17:42.04 | vaewyn | nice |
17:42.05 | [TK]D-Fender | * |
17:42.25 | vaewyn | cool |
17:42.29 | [TK]D-Fender | so I get "phones in use" and a line by line list of everyone, name & number |
17:42.43 | vaewyn | sweeeet |
17:42.46 | [TK]D-Fender | and I use the MB in dle mode to show live queue stats for my CSR's :) |
17:43.01 | [TK]D-Fender | ilde* on 10 sec interval |
17:43.04 | vaewyn | I really need to get back into the * swing... being tasked with other crud has been a real drag |
17:43.14 | jpablo | Corydon-w, i actually have an issue, and in the wiki it saids to use plaintext auth, but it is giving me a warning in asterisk 1.2 |
17:43.15 | [TK]D-Fender | including our company logo |
17:43.46 | vaewyn | [TK]D-Fender: yeah... we use idle mode for current weather and MOTD type stuff |
17:44.06 | Corydon-w | What's the warning? |
17:44.12 | *** join/#asterisk ctooley (n=ctooley@rrcs-24-227-212-162.sw.biz.rr.com) |
17:44.34 | [TK]D-Fender | vaewyn : I'm going to make a MB page so they can subscribe to Idle services :) And maybe use that to implement "messaging" |
17:44.43 | jpablo | Corydon-w, Dec 8 11:40:56 WARNING[27542]: chan_sip.c:11749 add_realm_authentication: Format for authentication entry is user[:secret]@realm at line 1020 |
17:44.55 | jpablo | Corydon-w, it looks like auth= changed it's meaning |
17:44.58 | vaewyn | [TK]D-Fender: heh... not a bad idea :} |
17:45.13 | [TK]D-Fender | abstraction is the key to success.... |
17:45.18 | ctooley | just sent out details of our Asterisk Bounty Pool fund raiser to the asterisk-user mailing list. I'll be putting up a site soon with a running total of the money that's been donated and details on which bounties have been offered and how to claim them. |
17:46.06 | Corydon-w | jpablo: plaintext is the default |
17:46.08 | *** join/#asterisk xheliox (n=jeff@user-0c6s3h2.cable.mindspring.com) |
17:46.19 | vaewyn | ok... how the heck does the norhell know I am fowarding the call when I wipe RDNIS and PRIREDIRECTREASON prior to making the outgoing call |
17:46.49 | [TK]D-Fender | vaewyn : You doing a "Dial" to do your redirect? |
17:47.31 | jpablo | Corydon-w, i don t think so, im seeing the packages and i see auth=md5 in the sip packages |
17:47.56 | Corydon-w | jpablo: how about you just comment it out? |
17:48.04 | vaewyn | [TK]D-Fender: What we have is that when people call in... if their phone is unavailable it attempts to dial a fallback number... the norhell sees this outgoing call and smacks it down because it is originating originally from off campus... and somehow it is figuring that out |
17:48.23 | jpablo | Corydon-w, i get the same |
17:48.35 | Corydon-w | jpablo: did you restart? |
17:48.39 | vaewyn | [TK]D-Fender: Nothing like Norhell and their "options" |
17:48.59 | jpablo | Corydon-w, grandstream getting confused with the 407 messages |
17:49.06 | jpablo | Corydon-w, yes, asterisk & the phone |
17:49.25 | [TK]D-Fender | vaewyn : Cheap hack - Make 2 SIP accounts on your * and have your * register to itself. then place a sip call from 1 account to the other (effectively internal redirect) and it should count the incoming channel as SIP. that way RDNIS = HISTORY. |
17:49.40 | Corydon-w | jpablo: well, someone with a bit more experience than you is going to need remote access to look at the machine |
17:50.01 | jpablo | Corydon-w, what do you know about my experience ? |
17:50.13 | [TK]D-Fender | vaewyn : And because it'll never actually leave the network interface you get NO overhead and G.711u leaves you with no transcoding. |
17:50.15 | Corydon-w | jpablo: Grandstream phones work fine with 1.2 |
17:50.17 | vaewyn | [TK]D-Fender: bwahaha... I might fall to that idea if I can't figure it out another way... Just wish this Norhell didn't think it was so smart |
17:50.27 | jpablo | Corydon-w, including the Early Dial feature? |
17:50.37 | [TK]D-Fender | vaewyn : Nortel = &*^#@$ stupid appliances... |
17:50.41 | Corydon-w | jpablo: why are you using early dial? |
17:50.51 | [TK]D-Fender | SOOO glad I got rid of my 8x24... |
17:50.54 | jpablo | Corydon-w, cause that's a nice feature that i want ... |
17:50.59 | Corydon-w | Because you think it's a whiz-bang cool feature? |
17:51.02 | Corydon-w | Turn it off |
17:51.06 | [TK]D-Fender | I am the king of cheap hacks :) |
17:51.20 | [TK]D-Fender | Just never ask me to do it the "right" way :D |
17:51.21 | vaewyn | [TK]D-Fender: I would ditch ours... but is Option 11C running a @#$@$load of digital phones |
17:51.26 | jpablo | Corydon-w, i know the phones work without early dial, im trying to make early dial work, that's the point of my questions ... |
17:51.32 | [TK]D-Fender | 11C? |
17:51.38 | jpablo | Corydon-w, try to make early dial work |
17:51.40 | Corydon-w | jpablo: ask someone else |
17:51.58 | vaewyn | [TK]D-Fender: Up to 10000 TNs |
17:52.04 | [TK]D-Fender | ? |
17:52.11 | Corydon-w | I can help make it work, but if you insist on using something that it doesn't work with, I'm not going to be any help |
17:52.11 | vaewyn | [TK]D-Fender: currently using 4800 or so |
17:52.13 | [TK]D-Fender | breif explin plz... |
17:52.21 | [TK]D-Fender | TN's (digital sets? |
17:52.28 | vaewyn | [TK]D-Fender: basically |
17:52.33 | [TK]D-Fender | OMG. What for? |
17:52.41 | vaewyn | University |
17:52.52 | [TK]D-Fender | 4800 phones?!?!?!?! |
17:53.05 | vaewyn | handles combo of analog and digital units... 1200 digital... rest are analog |
17:53.10 | vaewyn | Yep |
17:53.14 | [TK]D-Fender | OMG. |
17:53.15 | ctooley | vaewyn, which University? |
17:53.21 | [TK]D-Fender | Ok maybe too much to convert all to * |
17:53.21 | jpablo | Corydon-w, my question was simple, how to enable auth=plaintext, that's it |
17:53.35 | jpablo | Corydon-w, i know what's the problem, the asterisk 407 messages |
17:53.36 | vaewyn | ctooley: Andrews University... private one in SW Michigan |
17:53.41 | [TK]D-Fender | And I'm sure no budget to do so either... |
17:54.03 | ctooley | UTexas uses a Nortel as well, I'm pretty sure they're around 10 or 15 K phones |
17:54.16 | vaewyn | [TK]D-Fender: actually... the Norhell is out of service plan next year... and they want 2.5Mil to get it back on with current gear so... :} |
17:54.22 | Corydon-w | plaintext auth is the default |
17:54.35 | [TK]D-Fender | vaewyn : Citel <---- |
17:54.44 | [TK]D-Fender | SER & * |
17:54.53 | ctooley | vaewyn, 2.5 Mil might not buy you enough phones and network to replace it though. |
17:54.58 | *** part/#asterisk Dandre (n=Dandre@was59-3-82-236-48-30.fbx.proxad.net) |
17:55.03 | vaewyn | [TK]D-Fender: If I can prove * can handle it then we are going to go off plan and use the Norhell as a channel bank :P |
17:55.47 | ctooley | ah. Slowly migrate people to newer devices when the old ones finally die and keep the channel back for all those analog lines. |
17:55.48 | tzanger | vaewyn: already in the works :-) |
17:55.52 | vaewyn | ctooley: If we stick to analog with channel banks... and only use SIP for replacing the digitals we can come in under that |
17:55.58 | jpablo | Corydon-w, ok, thanks. |
17:56.08 | tzanger | citel has a norstar phone driver ... turns them all into sip phones |
17:56.36 | vaewyn | tzanger: Umm... so how do they break out that many ISDN lines? |
17:56.49 | vaewyn | tzanger: or are you talking the nortel network digitals? |
17:56.59 | tzanger | vaewyn: ? it's a D50 to the desk sets |
17:57.03 | tzanger | then ethernet for SIP |
17:57.22 | vaewyn | hmm... don't know my norhell... D50? |
17:57.33 | tzanger | AMP D50 (25 pairs) |
17:57.41 | [TK]D-Fender | 1200 digital = 50 Citel ($2700) = $135,000. the other 3600 = 150 Audiocodes ($2300) = $345,000 ------ $480,000to convert the sets |
17:58.04 | vaewyn | tzanger: ahh... 25 pair... hehe... |
17:58.05 | tzanger | [TK]D-Fender: as opposed to what |
17:58.21 | vaewyn | tzanger: so it drives the fake ISDN crud? |
17:58.25 | ctooley | putting that many extra devices on the network though would likely make the network guys scream upgrade |
17:58.26 | tzanger | vaewyn: correct |
17:58.34 | [TK]D-Fender | tzanger : as opposed to the 2.5M they were asking |
17:58.42 | vaewyn | network is already sized for this and more |
17:58.44 | tzanger | ctooley: you don't put it on the data network |
17:58.50 | tzanger | you put the 50 citel on their own switches |
17:58.51 | vaewyn | 8 pair fiber to all buildings |
17:58.56 | [TK]D-Fender | PRIVATE GB LAN <- |
17:59.06 | tzanger | nice |
17:59.18 | vaewyn | Only real problem we have is power... We need more generators and UPS |
17:59.18 | tzanger | I have a telebridge hooked up to my norstar |
17:59.20 | tzanger | it's kind of neat |
17:59.40 | vaewyn | main IT building is the only 24/7/365 power so |
17:59.49 | ctooley | yeah, those aren't free either |
17:59.51 | vaewyn | rest are at the whim of nature |
18:00.09 | ctooley | environmental costs are probably going to be higher than replacing the handsets |
18:00.18 | vaewyn | luckily they are looking at that anyway for security systems and such so :P |
18:00.33 | [TK]D-Fender | Clearly you would use SER to handle the SIP accounts and * as an app server. |
18:00.59 | vaewyn | Yeah... we sized the * boxes and phones and channel banks... and about 1.5Mil... Generators and UPS another 1 Mil+ |
18:01.15 | vaewyn | hell no... * it all :} |
18:01.15 | ctooley | Between design, implementation, parts and configuration, 2.5 mil would be a tight budget |
18:01.32 | [TK]D-Fender | vaewyn : My idea came out to $480k and lets you keep your phones :) |
18:01.36 | ctooley | Might be possible, but it would be tight. |
18:02.01 | [TK]D-Fender | yet still completely eliminates Nortel (except for the digiat sets. |
18:02.04 | ctooley | [TK]D-Fender, the Citel's use network for interconnect or the phone lines? |
18:02.14 | vaewyn | But... the nice thing is... we know that in 4 -7 years we will have to pay for another frame to move the norhell to VoiP anyways... and that they are already quoting us another 7+mil... so 2.5+ now... and save 7 later is looking very nice to the VP types |
18:02.44 | [TK]D-Fender | Citels offer Norstar Digital on Amphenol and SIP out. |
18:02.46 | vaewyn | [TK]D-Fender: citel does seem tempting... will have to look into that |
18:02.58 | *** join/#asterisk brettnem (n=brettnem@72.29.102.158) |
18:03.25 | vaewyn | wonder what I can get for a bulk buy on those :P |
18:03.26 | [TK]D-Fender | vaewyn : but oh boy are you going to need a good redundant SIP router and PRI gateway. |
18:03.39 | [TK]D-Fender | vaewyn : for 50 I'm sure you could :) |
18:03.39 | SkramX | I cant find any of those free TTS websites.. anyone know? |
18:03.42 | ctooley | <PROTECTED> |
18:04.05 | [TK]D-Fender | ctooley : depending on what you consider cause for replacement (defect or desire) |
18:04.21 | vaewyn | ctooley: I am liking the quality on the Poly 500/600... Is comparable with the norhells |
18:04.26 | [TK]D-Fender | ctooley : but the advantage is no WIRING change. and no change of the handesets themselves. |
18:04.44 | tzanger | [TK]D-Fender: if you go the citel route throw a couple extra in there I will buy them off you |
18:04.48 | [TK]D-Fender | vaewyn : Polycom quality >>> Nortel |
18:04.51 | vaewyn | [TK]D-Fender: any idea how far those will power a line? We have a couple 2000+ft runs |
18:04.52 | ctooley | vaewyn, yes, but the Polycom's are certainly not as sturdily built as the old Nortels |
18:05.12 | [TK]D-Fender | tzanger : I'm not going the Citel route, I'm suggesting it to vaewyn |
18:05.30 | [TK]D-Fender | ctooley : I suppose in the abuse-taking scenario, yeah.... |
18:05.44 | vaewyn | The poly's I have are what I would consider on par with the norhell's in abuse |
18:05.47 | [TK]D-Fender | vaewyn : got to their site and read up. |
18:06.01 | jpablo | fucking early dial, that's a great feature |
18:06.08 | vaewyn | [TK]D-Fender: am there now :P |
18:06.13 | [TK]D-Fender | vaewyn : I'm not really sure... I investigated it as an option for us here. We're an all-Polycom outfit now :D |
18:06.16 | jpablo | why it doesn't work |
18:06.22 | ctooley | [TK]D-Fender, the problem with the Poly's is that display. It lasts quite well until something is spilled on the phone, and then cleaning is impossible. |
18:06.27 | asteriskmonkey | go with that aastra phones there sick :) |
18:06.31 | asteriskmonkey | they have poe aswell |
18:06.45 | Cinen | anyone here had any luck with Dundi? I asked in #Dundi but nobody is home |
18:07.08 | [TK]D-Fender | ctooley : Yeah thats the price you pay with anything witha nice LCD... |
18:07.37 | *** join/#asterisk Meaty (n=cp_simbu@office.abi.ca) |
18:07.58 | ctooley | [TK]D-Fender, yeah, and the display does nothing but confuse my users. The very idea of soft keys confuses them. |
18:08.15 | asteriskmonkey | lol |
18:08.23 | MstlyHrmls | ... |
18:08.25 | [TK]D-Fender | ctooley : font forget the Norstar's use soft-keys too, and they are not as easily defined (borders, etrc) |
18:08.38 | [TK]D-Fender | and fewer characters. |
18:09.24 | [TK]D-Fender | FUGLY <- Anybody who can't figure out the display by just loking at it for a sec is pretty technologically challenged IMO.... |
18:10.03 | ctooley | Most people are pretty technologically challenged |
18:10.24 | [TK]D-Fender | add features = add complexity. more descriptive interface = less complexity. Theres a balance to be struck in there somewhere. |
18:10.38 | [TK]D-Fender | ctooley : I agree.... we should cull the herd a bit ;) |
18:10.56 | ctooley | One thing that is really irritating about the Polycoms is that the soft keys move |
18:11.11 | ctooley | For instance the "Send" option moves depending on the status of the handset. |
18:11.31 | [TK]D-Fender | ctooley : yeah there are some softkey's that when you go from option to option don't overlay in the most logical order. but there are worse. |
18:12.09 | ctooley | If the handset is down and you just start dialing "Dial" is the second one, when you pick up the handset and start dialing "Send" is the first one. Not only are they named differently, they're in different places. |
18:12.13 | [TK]D-Fender | What a thing it would be to help design a "superior" phone.... |
18:12.36 | ctooley | Or just find a phone manufacturer that had a real UI designer working for them. |
18:12.37 | [TK]D-Fender | ctooley : right on the money... |
18:12.59 | asteriskmonkey | lets make a touch screen phones nothing but a big lcd and handset |
18:13.20 | ctooley | I'm definitely not a UI expert, but even I can tell moving button locations an renaming them for the same functionality is a bad idea, |
18:13.21 | *** join/#asterisk destructure (n=irish@rrcs-24-173-126-174.se.biz.rr.com) |
18:13.23 | [TK]D-Fender | asteriskmonkey : umm.... NO. Too fragile. |
18:13.30 | ctooley | asterboy, that sounds like a disaster in the making |
18:13.34 | vaewyn | Heck yeah... who needs these @#$#@$ hard buttons... give me a point-of-sale type interface :P |
18:13.49 | *** join/#asterisk heison (n=heison@ns.somanetworks.com) |
18:14.17 | [TK]D-Fender | I like Poly's use of soft keys. Not exactly their location at all times, but the fact and reasons behind it. it just needs a little tweaking... |
18:14.42 | asteriskmonkey | dude a touch screen lcd phone would rock you could make so many cool looking button combos and dial pads that way |
18:14.52 | asteriskmonkey | and have em play like movies when your bored |
18:14.53 | fenlander | asteriskmonkey: like the AT&T broadband phone from their research lab? |
18:15.06 | [TK]D-Fender | Means you can make a "generic" ohone whose abilities are variable. only the keypad and direct audio controls (volume, mute, spkr, etc) should be "hard" |
18:15.09 | asteriskmonkey | fenlander: never seen that one ? have a link? |
18:15.20 | *** join/#asterisk IOscanner (n=IOscanne@c-24-0-183-125.hsd1.tx.comcast.net) |
18:15.21 | vaewyn | Heck... a desktop unit like my iPaq PDA running iaxcomm :} |
18:15.45 | [TK]D-Fender | vaewyn : My cell is toast and I refuse to pay for anything but a hybrid VoIP one :) |
18:15.54 | IOscanner | anhyone have problems with openvox 4 port fxo cards? |
18:16.05 | fenlander | http://www.cl.cam.ac.uk/Research/DTG/attarchive/bphone/ |
18:16.09 | vaewyn | [TK]D-Fender: heh :} can get them from Japan... but good luck with any US vendor :{ |
18:16.37 | vaewyn | fenlander: exactly! I had forgotten about that one |
18:16.38 | [TK]D-Fender | vaewyn : wah! I might be willing to import if they're in english |
18:16.41 | vaewyn | but that is the right idea |
18:17.03 | vaewyn | [TK]D-Fender: I have seen them in at least 'engrish' mode |
18:17.10 | [TK]D-Fender | EEK |
18:17.19 | [TK]D-Fender | I'm willingt to pay for a nice one... |
18:17.52 | vaewyn | one of my co-workers from japan had it... was quite cool... |
18:18.53 | [TK]D-Fender | 1h20m = CentOS 4.2 download :) 2h30m +/- = Helix 1.7 |
18:19.20 | jpablo | stupid granstream, why they don't have a dialplan in the phone like sipura |
18:19.31 | vaewyn | 7min = Debian install from local mirror + 1 min asterisk install |
18:19.32 | vaewyn | hehehe |
18:20.01 | SpaceBass | I have 2 of the iPicasso touch screen phones... but they don't seem to have firmware |
18:20.03 | [TK]D-Fender | vaewyn : CH34T4|2! |
18:20.08 | vaewyn | hehehe |
18:20.35 | vaewyn | [TK]D-Fender: I have to cheat... 50+ machines to keep in sync :P |
18:20.35 | [TK]D-Fender | I need to learn Debian and RH... |
18:21.02 | vaewyn | dpkg --get-selections | ssh newmachine dpkg --set-selections |
18:21.03 | vaewyn | :} |
18:23.15 | *** join/#asterisk Mitsch (n=mh@u-121-144.adsl.univie.ac.at) |
18:23.36 | KriS83 | Is it possible to set any CIDNumber? I mean can CIDNumber be anything? |
18:23.50 | SkramX | KriS83: It cant be a letter. |
18:23.54 | SkramX | LOL |
18:23.57 | KriS83 | ok |
18:24.03 | KriS83 | cos it's a number :) |
18:24.14 | KriS83 | No but I mean can I set any number? |
18:24.14 | SkramX | CIDName can be letters... |
18:24.19 | SkramX | Yes.. |
18:24.22 | vaewyn | KriS83: anything numeric... and yes... all the pranks have been done... like 8675309 |
18:24.28 | SpaceBass | i wish broadvoice allowed setting the CID |
18:24.37 | SkramX | SpaceBass: I heard they did. |
18:24.44 | vaewyn | NuFone does |
18:24.47 | SpaceBass | SkramX reaaaaalllly? |
18:24.47 | KriS83 | hmm |
18:24.47 | SkramX | I have a client who also works at BV, I will ask him |
18:25.02 | SpaceBass | [TK]D-Fender lol |
18:25.18 | vaewyn | [TK]D-Fender: hehehe.... I have even better... 1-700-867-5309 is mine :} |
18:25.20 | [TK]D-Fender | sut |
18:25.28 | [TK]D-Fender | SLUT |
18:25.29 | vaewyn | I need to hook it up to something :P |
18:25.36 | *** part/#asterisk Mitsch (n=mh@u-121-144.adsl.univie.ac.at) |
18:25.43 | SkramX | SpaceBass: I just text messaged the guy who works there.. have you asked their Customer Support/ |
18:25.53 | *** join/#asterisk tdonahue (n=tdonahue@208.51.101.201) |
18:26.03 | SpaceBass | SkramX I have not asked lately |
18:26.19 | *** join/#asterisk SplasPood (i=jwb@ludicrous.paravolve.net) |
18:26.21 | KriS83 | The thing is I have a number lets say 0694483817 this is my ISDN Phonenumber... but then I also have a so called speical number 01803XXXXX which is forwarded to my 069XXX number.. I can't set the 01803 number as CIDNumber :( |
18:27.16 | SkramX | SpaceBass: I was mistaken. |
18:27.28 | SpaceBass | SkramX just confirmed the same with a quick google |
18:27.34 | SpaceBass | but NuFone allows setcallerid? |
18:27.45 | SkramX | THey do not allow it. They set it in the background and do an ANI fail, so if they let poeople spoof charged numbers.. hehe |
18:27.53 | SkramX | SpaceBass: Yes, so does Iax.cc |
18:28.18 | SkramX | I wonder if viatalk allows it |
18:28.20 | SpaceBass | dont get me wrong... I'm purely going to prank potential here! |
18:28.39 | SpaceBass | actually it would save me the porting process with BV too... but thats a more practical reason |
18:28.53 | zoa | i called my friends from callerid 2004 on new years eve 2003 |
18:29.05 | zoa | and gave em a prank call with the poke speaking |
18:29.26 | zoa | then made a mistake and had 120 channels call all my friends all night |
18:29.40 | vaewyn | "mistake" |
18:29.41 | vaewyn | hehehe |
18:29.46 | many | haha |
18:29.58 | zoa | if they hung up before it was over (and the recording was 2 minutes) |
18:30.00 | zoa | it retried |
18:30.08 | zoa | that was fun |
18:30.11 | *** join/#asterisk Stealthmethod (n=123@72.242.62.209) |
18:30.20 | *** join/#asterisk SplasPood (i=jwb@ludicrous.paravolve.net) |
18:30.23 | SpaceBass | anyone else use iax.cc ? how reliable are they? |
18:30.27 | zoa | also it called people 20 times at the same time |
18:30.40 | zoa | leaving 19 messages on the voicemail at a time |
18:31.12 | KriS83 | Ok I have another question then: -- Accepting overlap voice call from '06941903117' to '<unspecified>' on channel 0/2, span 7 <- the number shown, which "$var" is this? Cos thats the number I want to be forwarding. |
18:31.43 | *** join/#asterisk Sobakai (n=jmwoodga@45.e6.d12c.cidr.airmail.net) |
18:32.16 | bsdfreak | iax.cc is decent, but their support sucks. |
18:32.21 | bsdfreak | it's practically non-existent |
18:32.22 | tzanger | no, iax.cc is NOT decent |
18:32.28 | bsdfreak | heh |
18:32.30 | tzanger | stay FAR AWAY from sixtel/iax.cc |
18:32.36 | SpaceBass | really? |
18:32.39 | tzanger | their service does work to an extent yes |
18:32.57 | tzanger | but getting anything updated/changed/troubleshot is a royal pain |
18:33.01 | tzanger | as is getting service terminated |
18:33.04 | bsdfreak | that's what i just said. |
18:33.05 | tzanger | STAY FAR AWAY |
18:33.05 | bsdfreak | heh |
18:33.07 | bsdfreak | support is teh sux |
18:33.16 | SpaceBass | im happy with BV |
18:33.16 | SkramX | I use iax.cc for personal stuff, not Business. |
18:33.28 | tzanger | fuck that they get none of my business |
18:33.29 | bsdfreak | i'm happy with asterlink |
18:33.34 | bsdfreak | very good support |
18:33.35 | SkramX | I got a 25 dollar credit to my account with iax.cc |
18:33.41 | SkramX | Yes, AsterLink is great. |
18:33.47 | bsdfreak | nufone was nice until they had a 3 day outage and wouldn't even tell me wtf was happening |
18:33.50 | bsdfreak | claimed everything was fine |
18:33.53 | tzanger | I'd be happy with asterlink if I could figure out why qualify is so shitty with them |
18:33.54 | bsdfreak | and NOONE was getting incoming calls |
18:33.57 | justinu | if only asterlink had DIDs |
18:34.00 | *** join/#asterisk bkw__ (n=brian@68.32.112.142) |
18:34.03 | tzanger | nufone is rock-solid but it's good to have backups |
18:34.05 | bsdfreak | asterlink does have DIDs |
18:34.06 | SkramX | justinu: I agree. |
18:34.08 | justinu | yes, they were talking about you |
18:34.12 | justinu | local DIDs |
18:34.13 | tzanger | yes asterlink does have dids and as I siad, they work quite well |
18:34.14 | bsdfreak | tz: nufone is hardly rock solid LOL |
18:34.17 | SkramX | Well, just 800's |
18:34.18 | bsdfreak | they go out at least once a month |
18:34.21 | tzanger | bsdfreak: for termination or origination? |
18:34.34 | bsdfreak | from what i've heard, both. |
18:34.43 | bsdfreak | and from what i've experienced, termination. |
18:34.50 | tzanger | bsdfreak: I've been using them as my primary termination for 2.5 years and they ahve not gone offline for termination ONCE |
18:34.54 | vaewyn | bsdfreak: I have never had them go out... I have had a router between here and there go south... but once BGP fixed that we wre good |
18:35.00 | SkramX | I would like Asterlink to do LOCAL DID's (origination) |
18:35.09 | tzanger | I don't use them for origination(I don't use anyone ofr origination) |
18:35.10 | justinu | asterlink is the only prepaid origination i tried that doesn't sound like crap |
18:35.22 | bkw__ | let me get our network 100 times more redundant and renumbered into our own space from ARIN |
18:35.30 | bkw__ | then more things will come down the pipe |
18:35.37 | bsdfreak | sup bkw :) |
18:35.38 | SkramX | bkw__: good deal. |
18:35.41 | tzanger | yup as I said my *only* "problem" with asterlink is that my iax2 qualify to any of their switches bounces around at least 10 times a day |
18:35.45 | SkramX | But I bet it will take a while. |
18:35.50 | tzanger | when calls go through they go through perfectly |
18:36.01 | justinu | i've been using SIP w/ astelink |
18:36.04 | jpablo | grrr, stupid GS, they fixed the early dial issue GXP-2000 but no in BTs :( |
18:36.07 | vaewyn | hey bkw... how's it been? |
18:36.12 | bkw__ | tzanger, quality has bugs in it yo |
18:36.21 | bkw__ | vaewyn, yo yo yo |
18:36.24 | bkw__ | doh |
18:36.25 | bkw__ | qualify |
18:36.26 | bkw__ | damn keyboard |
18:36.27 | bsdfreak | hehe |
18:36.42 | vaewyn | quality has bugs in it also these days :P |
18:36.43 | *** join/#asterisk Cresl1n (n=matt@gateway.digium.com) |
18:36.46 | tzanger | bkw__: I would tend to agree if it wasn't *just* asterlink I was having trouble with :-) |
18:37.10 | tzanger | actually I think I am in the process of uncovering an insidious little IAX2 call progress bug |
18:37.59 | tzanger | Office* --iax2--> Colo* --iax2--> termination (asterlink, nufone) |
18:37.59 | vaewyn | Ok... there has got to be a way to make this call clean of any "forwarding" or "redirect" telltales before I send it to the norhell |
18:38.16 | tzanger | office*/colo* use RFC1918 over private link |
18:38.27 | tzanger | office* CDRs show every call, whether it answered or not |
18:38.36 | tzanger | colo* CDRs only show calls that went through |
18:38.53 | tzanger | and any calls that did NOT go through seem to have 1-way audio (I can hear them but they can't hear me) |
18:38.59 | tzanger | call back it may work, it may not |
18:39.05 | tzanger | I have verified that IAX2 packets are flowing |
18:39.24 | tzanger | it's like the far side is eithe rnot sending an IAX2 call completion IE or I'm not getting it |
18:39.28 | *** join/#asterisk toddf (n=toddf@ns0.fries.net) |
18:39.37 | bkw__ | tzanger, we haven't updated our side to the code that doesn't have that bug.. I suspect thats why |
18:39.48 | bkw__ | it goes away for 10 seconds and comes back |
18:39.55 | tzanger | bkw__: what's that?? |
18:39.56 | bkw__ | I just turn that crap off :P |
18:40.06 | bkw__ | their was a timing bug in those qualify packets or something |
18:40.10 | tzanger | bkw__: ahh |
18:40.13 | bkw__ | it was a few months ago it was fixed |
18:40.27 | tzanger | interesting |
18:40.28 | anthm | in exchange for breaking 10 other things =D |
18:40.46 | anthm | so alas we cannot inherit the fix |
18:41.02 | bkw__ | ya we have to plan plan plan upgrades so we don't break stuffs |
18:41.13 | bkw__ | I usually roll out one drone with the upgrade... let it sit |
18:41.17 | bkw__ | and see if anything breaks |
18:41.27 | *** join/#asterisk earlt (n=earlt@200.62.22.11) |
18:41.33 | tzanger | anthm: :-) |
18:42.16 | anthm | tzanger, how's your select hell comming along? |
18:42.37 | ctooley | bkw_, is that what happened yesterday? |
18:42.45 | vaewyn | ok... anyone know how to make a call coming in loose all the RDNIS/PRIREDIRECTRESON...etc.. stuff? |
18:42.55 | vaewyn | I want it virgin when it hits the PRI again |
18:43.06 | tzanger | anthm: poorly |
18:43.18 | tzanger | anthm: every time I think I get it working well something ocmes up and bites me |
18:43.20 | vaewyn | norhell is repressing me |
18:43.33 | [TK]D-Fender | vaewyn : Submit to my hack! ;) |
18:43.38 | *** join/#asterisk MikeJ[Laptop] (n=ircatjer@adsl-69-208-116-100.dsl.sfldmi.ameritech.net) |
18:43.50 | tzanger | coppice says not to use sched_yield unless you *need* to give up CPU when you normally wouldn't. poll/cond_wait/sleep all yield just fine and far more effectively |
18:44.08 | tzanger | so I took that out, but now I'm having other odd little issues I'm debugging |
18:44.09 | vaewyn | [TK]D-Fender: heh... I might have to |
18:44.11 | tzanger | vaewyn: what's your specific issues? |
18:44.59 | vaewyn | tzanger: norhell sees that it is a call that it generated from off campus... and then says "dude! I don't have th option installed to let offsite call offsite" *smack* call dies |
18:45.15 | tzanger | vaewyn: well get the option then :-) |
18:45.19 | vaewyn | every other call in any manner is fine... |
18:45.35 | vaewyn | grrr... that's 4500$ + 120$/month I don't want to spend |
18:45.38 | tzanger | yeah |
18:45.45 | tzanger | you don't have DISA? |
18:46.01 | vaewyn | Nope... no need for it so... |
18:46.12 | tzanger | call in on a specific DISA DID ... at second dialtone dial out... |
18:46.12 | *** join/#asterisk Darwin35 (n=Darwin@sta-208-139-193-162.rockynet.com) |
18:46.20 | *** join/#asterisk xianlp (n=xian_1@M1116P028.adsl.highway.telekom.at) |
18:46.21 | Darwin35 | good morning |
18:46.29 | jpablo | why most of the docs you find about asterisk are from years ago and mention stuff that is no longer relevant :( |
18:46.33 | vaewyn | and no matter what I set RDNI and such to ... it still sees it as a forwarded call... has to be a way to scrub that |
18:47.04 | tzanger | vaewyn: you can't run custom libpri that just strips out the RDNIS IE? |
18:47.42 | vaewyn | tzanger: only reason we are running into this is we are having the * box send calls to failed phones to a fallback number... and when it sees offcampus->norhell->*->norhell->offcampus it goes "BAD MAN!" *smack* |
18:47.52 | tzanger | [main] loop took 14 ms |
18:47.52 | tzanger | [main] loop took 10 ms |
18:47.52 | tzanger | [main] loop took 0 ms |
18:47.54 | tzanger | [main] loop took 0 ms |
18:48.08 | tzanger | anthm: that's my big problem there ... why the blue fuck is my main loop taking 0 ms without poll() getting hit?! |
18:48.11 | tzanger | it's literally |
18:48.27 | vaewyn | tzanger: might have to try that... should be a way to tell * to wipe that data though |
18:48.46 | tzanger | do { poll() if(poll_result) { } gettimeofday() if(time > alarm1) { } if(time > alarm2) { } } while (!done) |
18:48.53 | anthm | is it suppoerd to be a timer ? |
18:48.57 | anthm | same every time? |
18:49.04 | tzanger | vaewyn: no there is no *-endorsed way to mangle IEs |
18:49.33 | vaewyn | tzanger: YOu would think though it would let me mangle the RDNIS in * and send the new value |
18:49.40 | tzanger | anthm: I just pasted the loop... it calls poll(wait up to 7ms) every iteration, then compares the time against a bunch of alarms |
18:49.45 | vaewyn | kindof odd though |
18:49.45 | tzanger | vaewyn: nope |
18:50.00 | tzanger | vaewyn: since RDNIS is a PRI term and * has no specific PRI channel |
18:50.02 | vaewyn | should make that a forced read only variable then |
18:50.14 | anthm | so when nothing is happening it should be 7 every time |
18:50.15 | tzanger | vaewyn: I'd just mangle libpri at this point to see if that fixes your problem |
18:50.27 | tzanger | it should be AT LEAST 7 yes |
18:50.44 | bkw__ | mangle or strangle? |
18:50.45 | tzanger | but add in the if()s and the gettimeofday() and you're usually up around 9-10ms |
18:50.50 | tzanger | bkw__: :-) |
18:51.32 | vaewyn | hmm... also I am using Dial to send this call out... why does it treat it as a forward and not as a new clall? |
18:51.35 | vaewyn | call even |
18:52.00 | Cresl1n | tzanger: gettimeofday probably hits it pretty big. The kernel usually uses that opportunity to schedule() |
18:52.03 | vaewyn | that has to be a * thing... cause the PRI has no clue those 2 channels are talking to each other |
18:52.16 | tzanger | vaewyn: I woudl try and figure out why/what the IEs are getting set to |
18:52.31 | anthm | are you calcing your own ms elapsed ? |
18:52.32 | tzanger | Cresl1n: yep, and that's fine... but why is my main loop not waiting at least 7ms due to the poll? |
18:52.37 | vaewyn | Hmm... I know how to fix this... |
18:52.39 | tzanger | anthm: I'm using your awesome code fragment |
18:52.54 | file[desk] | brookshire: poke |
18:53.02 | *** join/#asterisk sofh (n=ok@203.101.182.121) |
18:53.15 | sofh | hello all |
18:53.34 | sofh | can somebody give me idea , how to integerate a usb CDMA phone with asterisk ? |
18:53.42 | anthm | maybe it's not awesome enuf if it doesnt work lol |
18:53.57 | marcus2 | so whats this about meetme coming onto the line and saying something every 10 minutes? |
18:54.35 | paryl | will ztcfg kill all calls in progress? |
18:55.08 | sofh | hey Guys...any idea abuot some USB phone ? like CDMA ? |
18:55.12 | paryl | i need to initialize a channel on my channel bank |
18:56.02 | tzafrir_laptop | sofh, do you know of any USB phone with some sort of Linux interface? Why USB? |
18:56.24 | sofh | its configured fine on linux as a modem |
18:56.25 | tzafrir_laptop | USB CDMA phone, of course |
18:56.26 | sofh | i can do |
18:56.33 | sofh | i can do ATD comm via minicom |
18:56.42 | *** join/#asterisk MikeJ[Laptop] (n=ircatjer@adsl-69-209-166-207.dsl.sfldmi.ameritech.net) |
18:56.43 | tzanger | http://pastebin.ca/32998 |
18:56.47 | sofh | so now should configure it via modem.conf or what ? |
18:56.48 | tzanger | that's the mainloop code |
18:56.56 | tzanger | how the fuck it's getting 0ms every now and again is beyond me |
18:57.20 | tzafrir_laptop | paryl, ztcfg generally shouldn't if there are no relevant config changes |
18:57.50 | sofh | tzafrir_laptop! can you help me a little ? |
18:57.51 | tzafrir_laptop | an ATA modem is not something asterisk can use |
18:58.01 | sofh | it is not an ATA modem... |
18:58.13 | sofh | i just tried to access it jst like an ISDN adaptor |
18:58.25 | sofh | basicaly its a wireless CDMA phone |
18:58.39 | Seldon1975 | hmm I have two Polycom 301 SIP phones on the network with my * server. When I dial extension 1000 from 1001 the call pops up in the * console but the phone at 1000 doesnt ring |
18:58.43 | paryl | actually... i added a channel in zapata.conf, but haven't changed zaptel.conf, so i guess i don't even need to run ztcfg |
18:58.56 | paryl | but after adding it, i try to dial the channel and i get "Unable to create channel of type 'Zap'" |
18:58.59 | tzafrir_laptop | sofh, blutooth headphone or something? |
18:59.01 | tzanger | anthm: any ideas? |
18:59.09 | sofh | no its not blue tooth.. |
18:59.11 | Seldon1975 | I have configures [1000] and [1001] in sip.conf |
18:59.27 | sofh | its CDMA phone and using CDMA technology...bluetooth isn't that device |
18:59.33 | tzafrir_laptop | ztcfg applies configuration from zaptel.conf |
18:59.38 | sofh | it works over wll |
18:59.44 | tzanger | sofh: you have a CDMA USB phone? |
18:59.52 | sofh | yes tzanger |
18:59.58 | tzanger | sofh: got a link? |
19:00.11 | sofh | tzanger: link for ? |
19:00.14 | tzanger | the phone :-) |
19:00.15 | *** join/#asterisk Gunnar (n=gunnar@34.80-203-209.nextgentel.com) |
19:00.31 | sofh | my phone is connected to * box via usb port |
19:00.40 | tzanger | ohhhhhhhhhhh sorry I understand now |
19:00.44 | vaewyn | If I remember right... over the USB link you can only use those as modems... it doesn't have audio paths |
19:00.47 | sofh | now just not getting to play it in modem.conf or where |
19:00.59 | tzanger | vaewyn: correct, it's just a modem (data only) device. |
19:01.14 | tzanger | at least anything I've found |
19:01.25 | sofh | means there is no way to use a CDMA phone with * ? |
19:01.32 | tzanger | sofh: not that I'm aware of, no |
19:01.32 | tzafrir_laptop | "usb" is as good as "pci". It is a nice pipe to pass data through, but won't give you much more without a more specific driver |
19:01.39 | tzanger | not even as an SMS portal |
19:02.01 | sofh | hmm....though the correct hardware is installed in linux |
19:02.11 | sofh | like i can dial via minicom |
19:02.35 | vaewyn | You can use it like a data modem... but it is not a "voice" modem |
19:02.40 | tzanger | sofh: again, you're using it as a data device there, not as a phone |
19:02.53 | sofh | :( |
19:03.03 | sofh | i wish i could use it just like X100P |
19:03.11 | tzanger | sofh: unfortunately not |
19:03.16 | paryl | i added a channel to zapata.conf... shouldn't reload in the console make the changes? |
19:03.17 | tzanger | sofh: I know what you're up to and it would rock |
19:03.20 | anthm | yes |
19:03.38 | sofh | yeah tzanger..but not getting a true way.. |
19:03.43 | anthm | when the poll has an event |
19:03.50 | tzafrir_laptop | an X100P still has a driver that knows very speicfic details about its controller's registers |
19:03.50 | anthm | you must clear it |
19:03.54 | tzanger | anthm: I do |
19:03.56 | anthm | or it will return 0 |
19:03.58 | sofh | when asterisk is dialing even via bluetooth to a cell phone..then why don't a CDMA directly connected to itself ? |
19:04.06 | tzanger | pollfds[p_dn].revents &= ~(POLLIN | POLLPRI); |
19:04.27 | sofh | suppose we assume it as ISDN BIR , still it can't be use ? |
19:04.51 | tzafrir_laptop | sofh, does it provide an interface of an ISDN card? |
19:05.01 | sofh | offcourse not :( |
19:06.08 | sofh | by ISDN i mean just to configure it in modem.conf |
19:06.09 | paryl | so i changed zapata.conf, did a reload, but "zap show channels" doesn't show the channel i added |
19:06.33 | paryl | how can i get it to reload? (without taking asterisk down |
19:07.16 | tzafrir_laptop | ztcfg applies changes to zap channels (kernel). zapata.conf is the configuration of asterisk's chan_zap.so module. |
19:07.33 | [TK]D-Fender | paryl : gotta take it down... its the zaptel driver that needs to be reloaded and * has to be out of the way for that... |
19:07.44 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
19:07.44 | tzafrir_laptop | on asterisk 1.2 a reload should reload chan_zap |
19:08.07 | tzanger | anthm: I do clear it... |
19:08.13 | tzafrir_laptop | though that code is rather untested |
19:08.29 | sofh | tzanger: if i'd an ISDN bri adaptor..internal or external what ever..then how to configure * to use it for outgoing channel ? |
19:08.37 | anthm | you read from the fd |
19:08.40 | anthm | as much as you can |
19:09.20 | tzanger | anthm: I do |
19:09.34 | tzanger | I flag condition in the dn thread actions bitmap |
19:09.36 | tzanger | and it does that |
19:09.47 | tzanger | oh I think I see what you are saying |
19:10.00 | tzanger | if that thread doesn't wake up right away I could spin in the main loop with poll() returning right away |
19:10.03 | tzanger | er no |
19:10.06 | tzanger | er yes |
19:10.07 | tzanger | heh |
19:10.54 | tzanger | since I didn't read yet but I cleared the event, it'll come right back |
19:11.29 | tzanger | let me remove the dn thread entirely to see if that makes the spinning go away |
19:11.37 | tzanger | the only thing left will be the stdin poll() and the timer stuff |
19:11.41 | *** join/#asterisk shido6 (n=shido6@d221-68-216.commercial.cgocable.net) |
19:11.43 | anthm | the only thing that takes away a poll event |
19:11.52 | anthm | is reading the fd dry |
19:12.01 | paryl | [TK]D-Fender: d'oh, and, thanks. |
19:12.34 | tzanger | anthm: yes I realize that :-) I understand why poll() may be returning immediately now though.. I'm removing the dn stuff entirely to see if that makes the spinning go away |
19:12.35 | anthm | http://www.sofaswitch.org/eg/lame.c |
19:13.41 | tzanger | yup that does make it go away |
19:16.26 | tzanger | wow thanks ofr the insight :-) |
19:16.52 | *** join/#asterisk elg (n=fugalh@falcon.fugal.net) |
19:17.12 | tzanger | holy fuck my four-layer board's here already |
19:18.01 | elg | any idea if anyone's working on V.150.1 (MoIP)? |
19:18.41 | zoa | what is MOIP ? |
19:18.54 | brad_mssw | modem over ip |
19:18.58 | zoa | aha |
19:19.05 | zoa | that thing works ? |
19:19.09 | elg | demodulate, ip, remodulate |
19:19.09 | brad_mssw | and yes, it's tied in with the T.38 support that's being worked on |
19:19.18 | brad_mssw | don't expect it for a little while though |
19:19.28 | zoa | we are working on t.38 |
19:19.43 | zoa | not the same stack as coppice's |
19:20.43 | brad_mssw | think v.150.1 might work via pass-thru with these patches : http://bugs.digium.com/view.php?id=5090 not sure though |
19:21.03 | brad_mssw | though I don't know of any ata that supports it :/ |
19:22.33 | *** join/#asterisk heison (n=heison@ns.somanetworks.com) |
19:22.49 | SkramX | how nice.. nufone web management is down |
19:23.02 | *** join/#asterisk r0d3nt (n=RatMan@tinfoilhat.net) |
19:26.27 | paryl | can Background play wav files? |
19:26.31 | tzanger | sweeeeeeeeeeet: http://www.mixdown.ca/~andrew/dump/ipg1.jpg |
19:28.30 | *** join/#asterisk bzbw (n=wlwzhang@ip67-153-142-109.z142-153-67.customer.algx.net) |
19:28.52 | *** join/#asterisk dca_ (n=dca@sta-208-139-193-162.rockynet.com) |
19:29.55 | bzbw | hi, is there a way to have * to send the voicemail via email but not saving it in the server for a particular extension? Wiki does not seem to have such info |
19:32.48 | *** join/#asterisk bwzb (i=bwz@ip67-153-142-109.z142-153-67.customer.algx.net) |
19:33.38 | bwzb | hi, anyone knows whether I can enable the voicemail to be sent via EMAIL ONLY for a select extension? WIKI does not seem to have such info |
19:33.52 | sivana | bwzb: yes |
19:33.53 | SkramX | what do you mean ONLY |
19:33.58 | SkramX | like its not saved on the server? |
19:34.37 | bwzb | I have a catch all extension that have way TOO many voicemails, and it is taking the server space quickly |
19:34.38 | Cinen | it must store it on the server for a few minutes. It can delete right after it emails it out |
19:34.59 | bwzb | Thanks Cinen. How do you set up that way? |
19:35.05 | Cinen | will delete=yes in voicemail.conf noit work for you |
19:35.16 | sivana | bwzb: mailbox => password,Catch-All,someemail@mycompany.com,,|attach=yes|delete=no |
19:35.30 | sivana | except.. make it delete=yes :) |
19:35.47 | Cinen | yea like ivana said |
19:35.48 | bwzb | Cinen: I did not try delete=yes, I will try that now! |
19:35.59 | bwzb | thanks you all! |
19:36.01 | Cinen | np |
19:36.04 | Seldon1975 | are the samples that * creates when you do 'make samples' available thru SVN? |
19:37.45 | *** join/#asterisk santiago (n=santiago@208.195.215.160) |
19:37.52 | Seldon1975 | does anyone kinow? |
19:38.04 | *** join/#asterisk MikeJ[Laptop] (n=ircatjer@adsl-69-209-166-207.dsl.sfldmi.ameritech.net) |
19:38.44 | bwzb | Cinen: thanks, you know the link? |
19:39.11 | Cinen | I do not off the top of my head. Just google it |
19:39.50 | Cinen | I can send you the file if you want |
19:40.41 | bwzb | thx! |
19:41.55 | *** join/#asterisk Micc (n=dotirc@c-24-16-228-130.hsd1.wa.comcast.net) |
19:42.03 | Seldon1975 | the sample extensions.conf has extension 500 set up |
19:42.10 | Seldon1975 | what should I hear when I dial this? |
19:42.20 | Micc | Why is it that I can't use G729a for both lines on my linksys pap2-na? |
19:42.37 | Micc | I have 5 licenses. |
19:43.21 | sivana | Micc: only 1 at a time |
19:43.26 | Micc | sivana, why? |
19:43.29 | sivana | Micc: by design |
19:43.37 | Micc | Is it the linksys that is the problem? |
19:43.56 | sivana | by design on Linksys' side... they didn't put a big enough processor to handle 2 |
19:43.59 | Micc | Is there another adapter that works both lines? |
19:44.25 | sivana | no idea |
19:44.30 | Micc | oh thats lame. I need a better adapter then. |
19:44.40 | justinu | the spa's can probably do it |
19:44.43 | iCEBrkr | Zzzzzzzzzzzz |
19:44.44 | Micc | whats the next best codec to use that it can handle? |
19:45.03 | sivana | I like ulaw myself |
19:47.28 | Micc | ulaw has jitter for me on a cable modem. |
19:47.59 | *** join/#asterisk raptorrat (n=ucs_rat@ab1-1-26.shsu.edu) |
19:48.32 | *** part/#asterisk raptorrat (n=ucs_rat@ab1-1-26.shsu.edu) |
19:48.42 | *** join/#asterisk waddy (n=waddy@83.218.4.231) |
19:49.49 | waddy | when i try to compile asterisk-addons - i get this error |
19:49.50 | waddy | ../asterisk: Not a directory |
19:49.50 | waddy | make: *** [app_saycountpl.o] Error 1 |
19:49.56 | waddy | anyone can help? |
19:50.33 | *** join/#asterisk arguile (i=user224@66.38.201.234) |
19:50.50 | *** join/#asterisk seant (n=seant@67.105.255.132.ptr.us.xo.net) |
19:51.04 | denon | waddy: did you get asterisk's souce, and put it in the same directory as asterisk-addons? |
19:51.05 | denon | (no) |
19:51.13 | paryl | is there a way to respond to dtmf tones while musiconhold is playing? (like a "if you know your party's ext, you may dial at any time" type thing) |
19:51.38 | waddy | ahh ok ill try that thanks |
19:51.47 | denon | follow the instructions on the download page |
19:51.53 | denon | as you're probably missing numerous other steps :) |
19:52.03 | *** join/#asterisk trig_hm (i=jason@home.monkeypr0n.org) |
19:53.46 | seant | Quick Question: Does asterisk support the ability to designate multiple recipients for a voicemail on the fly? ex. Dial multiple extensions who should receive the following voicemail. |
19:55.13 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
19:57.46 | KranZ | seant: yes |
19:58.56 | KranZ | seant: Voicemail(1234&1235) |
19:59.29 | *** join/#asterisk greekman (n=alex@host254.209.113.199.conversent.net) |
20:00.37 | greekman | can anyone help with manipulating queue entries in the cdr? |
20:04.11 | *** join/#asterisk vexorg (n=vexorg@CPE000021ded913-CM001225419164.cpe.net.cable.rogers.com) |
20:11.36 | [TK]D-Fender | greekman : what do you have in mind? |
20:12.49 | [TK]D-Fender | paryl : when is this MOH playing? |
20:13.03 | *** join/#asterisk eKo1 (n=bernd@metrored-gw.tropicohn.com) |
20:13.25 | *** join/#asterisk MikeJ[Laptop] (n=ircatjer@d14-69-8-30.try.wideopenwest.com) |
20:16.58 | *** join/#asterisk paljas (n=paljas@tuxtown.xs4all.nl) |
20:18.53 | *** join/#asterisk paljas (n=paljas@tuxtown.xs4all.nl) |
20:23.29 | paryl | [TK]D-Fender: i play a background recording and go right to moh |
20:23.31 | paryl | as soon as it's done playing, you can't dial any digits |
20:26.06 | Seldon1975 | ive set up extension=>1111 to playback tt-monkeys and when I dial is the Asterisk console says "Playing 'tt-monkeys'" but I dont hear any monkeys! |
20:26.20 | Seldon1975 | where should this sound file be on the asterisk server? |
20:27.03 | *** join/#asterisk paryl (n=paryl@216-201-177-82.res.logixcom.net) |
20:27.21 | paryl | hrmm... lost my connection. Fender: did you respond? |
20:27.56 | greekman | D-Fender: i have a unique situation where I build call files based on an incoming click-to-call. I send the call first to a queue, when an agent picks up, the other leg to the requesting party gets done. The problem is when I look at my cdr, 2 entries are made... the first is the que entry that shows the agent it connects the call to in dstchannel, the second is the actual call with no agent info init, but it has the actual duration of |
20:28.34 | [TK]D-Fender | greekman : ok, out of my league but maybe someone else can pick up on that. |
20:28.41 | greekman | lol |
20:28.47 | greekman | thanks anyway |
20:29.02 | [TK]D-Fender | paryl : WHY are they getting MOH and what kind of options are they supposed to be able to enter? |
20:29.07 | *** join/#asterisk kimosabe (n=kimosabe@201.135.10.173) |
20:29.26 | [TK]D-Fender | paryl : You mean like you want background music while they make up their mind? |
20:29.42 | ctooley | greekman, which click to call provider are you with? |
20:29.47 | [TK]D-Fender | greekman : Good description though, just wait around a bit for an answer on that... |
20:30.28 | ctooley | greekman, I write my own cdrs for this reason. Doing an Originate through the Manager interface instead of using call files might work. |
20:32.41 | kimosabe | i am trying 2 use asterisk as sip client i have 2 sipuras and on my asterisk box i configured one icoinnect acount the servers says its conected but when i dial it says everyone is busy |
20:33.29 | kimosabe | i want the sipuras to connect 2 asterisk box and dial out threw icconect |
20:34.18 | kimosabe | can some one help me out a bit |
20:36.02 | paryl | D-Fender: actually, i WANT them to be thrown into a queue and allow them to dial an extension if they have to hold... but i don't know if that's possible |
20:36.13 | Seldon1975 | <PROTECTED> |
20:36.19 | Seldon1975 | any ideas |
20:36.48 | [TK]D-Fender | Seldon1975 : You need to "Answer" first. <- |
20:37.10 | Seldon1975 | oh |
20:37.12 | bwzb | hi, after building 1.2.1, I can not restart my *, anyone know what seems to be the error? I remember someone mentioned about changing a header or something |
20:37.18 | Seldon1975 | how do I do that Fender? |
20:37.21 | [TK]D-Fender | paryl : ok, thats already easy to do (a dial-out menu while in queue) |
20:37.33 | bwzb | the error is like this: [cdr_addon_mysql.so]Dec 8 12:29:22 WARNING[25951]: loader.c:325 __load_resource: /usr/lib/asterisk/modules/cdr_addon_mysql.so: undefined symbol: ast_load |
20:37.34 | bwzb | Dec 8 12:29:22 WARNING[25951]: loader.c:554 load_modules: Loading module cdr_addon_mysql.so failed! |
20:37.34 | [TK]D-Fender | Seldon1975 : Do an Answer before your playback! |
20:37.48 | Seldon1975 | ok thanks |
20:38.11 | bwzb | the last error: Ouch ... error while writing audio data: : Broken pipe |
20:40.45 | bwzb | maybe 1.2.1 change something in dynamically linking the modules: __load_resource: /usr/lib/asterisk/modules/cdr_addon_mysql.so: undefined symbol: ast_load |
20:43.19 | bwzb | anyone here can help me??? |
20:45.05 | Seldon1975 | Fender, i put the Anser in - still no monkeys :( |
20:45.11 | *** join/#asterisk QbY (n=QbY@adsl-068-209-210-253.sip.cha.bellsouth.net) |
20:45.20 | Seldon1975 | the * console shows the Answer and the PLayback |
20:45.24 | QbY | <PROTECTED> |
20:46.06 | ctooley | bwzb, you didn't build the asterisk-addons package for version 1.2.1 |
20:46.09 | *** part/#asterisk elg (n=fugalh@falcon.fugal.net) |
20:47.39 | paryl | D-Fender: how do you do a dialout menu? i know how to do '0' for an operator, but i thought that was just built-in |
20:48.17 | *** join/#asterisk The_LightSide (n=lightsid@wbs-196-2-122-41.wbs.co.za) |
20:48.24 | paryl | but i don't know how to control the scripting of 'play this message, then music, then this message, etc' |
20:49.06 | The_LightSide | hi all, where could i find info on how to setup AAH2.1 with the built in bristuff? |
20:51.01 | The_LightSide | hello? |
20:51.29 | greekman | ctooley: I made my own click-to call |
20:51.33 | *** join/#asterisk kubejm (i=kubek@AMontpellier-151-1-10-135.w83-205.abo.wanadoo.fr) |
20:51.48 | *** part/#asterisk kubejm (i=kubek@AMontpellier-151-1-10-135.w83-205.abo.wanadoo.fr) |
20:53.11 | paryl | D-Fender: the bigest thing i'd like to do that i can't seem to figure out, is i'd like the queue to dial any available agents immediately, and only play a 'all reps are busy' if the customer has to be thrown into MOH |
20:54.17 | Seldon1975 | when I dial an outside line I get: |
20:54.36 | bwzb | ctooley: do I need to build asterisk-addons? |
20:54.55 | Seldon1975 | NOTICE[3292]: pbx.c: 1731 pbx_extension_helper: Cannot find extension context 'default' |
20:55.02 | Seldon1975 | anyone know what this means? |
20:55.09 | ctooley | you do if you want cdr_ADDON_mysql.so |
20:55.32 | *** join/#asterisk klictel (n=klictel@207.107.208.137) |
20:56.56 | The_LightSide | where could i find info on how to setup AAH2.1 with the built in bristuff? |
20:57.30 | ctooley | The_LightSide, repeating your self will not get you answered. If someone knew, they'd answer you |
20:57.32 | *** part/#asterisk Samoied (n=Samoied@popeye.opens.com.br) |
20:57.36 | KranZ | Seldon1975: means you dont have a default context |
20:57.57 | The_LightSide | ok.... |
20:58.02 | The_LightSide | ta |
20:58.24 | KranZ | @home blah |
20:58.32 | *** join/#asterisk peter_l (n=ploeppky@store-fw.porchlight.ca) |
20:58.37 | KranZ | they should drop that |
20:59.02 | *** join/#asterisk mountie (n=mountie@CPEdeaddeaddead-CM000a739acaa4.cpe.net.cable.rogers.com) |
21:01.00 | mutilator | DOH! |
21:01.02 | mutilator | FrSIRT.com RSS Feeds: |
21:01.02 | mutilator | 2005-12-08 This XML feed has been disabled. FrSIRT security advisories are available on FrSIRT.COM |
21:01.27 | [TK]D-Fender | paryl : Sorry, they get MOH until the call is answered... |
21:03.16 | KranZ | mutilator: they prolly need AD revenue |
21:05.15 | bsdfreak | poop dawg |
21:05.36 | *** join/#asterisk Stealthmethod (n=123@72.242.62.209) |
21:05.43 | *** part/#asterisk Stealthmethod (n=123@72.242.62.209) |
21:07.26 | paryl | ok, how about this... when a call in a queue starts ringing to an extension, will it continue to ring that extension if a timeout occurs? |
21:07.40 | paryl | i.e. could i just set the first queue to timeout in 5 seconds |
21:07.40 | KranZ | if you want |
21:07.42 | paryl | ? |
21:08.12 | zoa | hey hp |
21:08.13 | zoa | ho |
21:08.37 | file[desk] | zoa: zoa zoa zoa |
21:09.36 | *** join/#asterisk flashbac1 (i=flashbac@68-235-251-168.atlsfl.adelphia.net) |
21:09.43 | flashbac1 | hello! |
21:09.47 | *** join/#asterisk supaigtr (n=yurplsl@152.53.16.10) |
21:09.50 | flashbac1 | any realtime gurus here? |
21:10.08 | supaigtr | Anyone seen the adtran Netvanta 7100? |
21:10.26 | zoa | file file file |
21:11.32 | zoa | go poke yourself you dirty bastard! |
21:11.41 | file[desk] | :p |
21:12.06 | *** join/#asterisk saftsack (n=saftsack@p54A7FE77.dip.t-dialin.net) |
21:12.11 | *** join/#asterisk nagl (n=nagl@213.235.241.6) |
21:12.12 | [TK]D-Fender | paryl : sort of against the idea of a queue.... |
21:12.29 | saftsack | hi |
21:12.29 | flashbac1 | hey guys, i keep getting this error with realtime: |
21:12.44 | flashbac1 | Dec 8 16:04:37 WARNING[1817]: res_odbc.c:171 odbc_smart_execute: SQL Execute returned an error -1: 42000: [Sybase][ODBC Driver][Adaptive Server Enterprise]Implicit conversion from datatype 'CHAR' to 'INT' is not allowed. Use the CONVERT function to run this query. |
21:13.07 | flashbac1 | i think this might be a bug... |
21:13.15 | *** part/#asterisk QbY (n=QbY@adsl-068-209-210-253.sip.cha.bellsouth.net) |
21:13.17 | zoa | it looks like a bug |
21:13.23 | saftsack | i defined in my extensions.conf that if someone called the number xyz asterisk answers. and then the caller can call after hearing a voicemessage a special telephone. but now theres a problem |
21:13.24 | zoa | or you fucked up your table |
21:13.25 | bsdfreak | bega |
21:13.49 | saftsack | if the caller hangs up, the other telephone will ring and ring and ring ... |
21:13.50 | *** join/#asterisk n4y (n=tmalkut@fw.orasoft.net.pl) |
21:13.54 | saftsack | it wont stop ringing |
21:14.01 | saftsack | someone can help? |
21:14.23 | KranZ | supaigtr: that thing is a monster |
21:14.30 | *** join/#asterisk Little-L (n=daniel@0x50a471db.svgnxx1.adsl-dhcp.tele.dk) |
21:17.28 | supaigtr | Monster??? |
21:17.39 | Jesster | anyone successful making custom ring tones for Cisco 7940/7960? i've followed what i've read and still no luck. |
21:17.40 | KranZ | yes |
21:17.51 | KranZ | it does about everything |
21:18.01 | supaigtr | Monster as in penis or breasts or monster as in problem? |
21:18.05 | Darwin35 | what is everything |
21:18.13 | *** join/#asterisk A-Tuin (n=a-tuin@steves.ip.v4.me.uk) |
21:18.14 | Darwin35 | do you have call back if busy |
21:18.16 | paryl | [TK]D-Fender: my thought is, if they come into the queue and the call isn't answered immediately, they will be bumped out to an 'all reps are busy' message, and then put back into the queue |
21:18.16 | supaigtr | KranZ: Supports polycom? |
21:18.21 | Darwin35 | do you have fallow me working |
21:18.35 | KranZ | supaigtr: it supports sip |
21:18.50 | Darwin35 | do you have it call oyu vibrator on tech calls |
21:19.05 | *** join/#asterisk dudes (n=dudes@12-215-32-62.client.mchsi.com) |
21:19.07 | supaigtr | KranZ: Any provisioning support for phones? |
21:19.16 | supaigtr | KranZ: is it asterisk based or other? |
21:19.20 | [TK]D-Fender | paryl : there is already a "thanks for holding" message that you can configure, including its frequency. I might suggest playing a message BEFORE the call enters the queue so they know they might be holding for a while |
21:19.21 | Darwin35 | I still need call back wen busy |
21:19.25 | Darwin35 | grr |
21:21.56 | *** join/#asterisk Assid (i=assid@59.183.2.83) |
21:23.28 | *** join/#asterisk Romik_ (n=romik_@1.fix.netvision.net.il) |
21:24.03 | DrDeke | Hey guys, is there any way in an AGI script, to receive DTMF digits without the # meaning "stop receiving" ?? |
21:24.17 | DrDeke | (I would like to receive the # into my application just like any other digit) |
21:24.21 | [TK]D-Fender | DrDeke : Tons |
21:24.50 | [TK]D-Fender | Look at the AGI command list. Should be pretty obvious.... |
21:25.11 | bwzb | anyone know what is the issue with these warning message: |
21:25.12 | bwzb | Dec 8 13:24:42 WARNING[32300]: format_wav_gsm.c:243 update_header: Unable to find our position |
21:25.12 | bwzb | Dec 8 13:24:42 WARNING[32300]: format_wav.c:247 update_header: Unable to find our position |
21:25.32 | *** join/#asterisk devel (n=devel@wiggum.digitalcoven.com) |
21:26.36 | *** join/#asterisk darby_t (i=darby_t@dkr19.neoplus.adsl.tpnet.pl) |
21:26.38 | bwzb | looks like everyone is on vacation now |
21:27.20 | *** join/#asterisk dtev001 (n=mikeh@cpe-24-168-18-30.si.res.rr.com) |
21:27.43 | dtev001 | hey there... when you create a .call file, what context does asterisk use to dial out the call |
21:28.03 | DrDeke | TKD: I am looking at the AGI command list, but get_data appears to be #-terminated and wait_for_digit only takes one digit... |
21:28.14 | DrDeke | (actually it doesn't appear to be #-terminated, I tried it and it is) |
21:28.50 | *** join/#asterisk _ozstealth_ (n=ozstealt@64.141.48.2) |
21:28.59 | *** join/#asterisk jpablo (n=jpablo@200.94.130.194) |
21:29.19 | jpablo | hi, has anyone upgraded to the 1.0.7.11 (beta) firmware in a budgetone ? |
21:29.38 | jpablo | it is trying to download bt-100.bin that isn't included in the firmware zip |
21:29.53 | _ozstealth_ | Hi There, |
21:30.22 | DrDeke | TKD: Any other ideas? |
21:30.25 | _ozstealth_ | I have a question relating to making outbound calls with a recorded (or tts) message - is this the right place to be asking? |
21:30.35 | *** join/#asterisk Oryn (i=oryn@falcore.fsck.tv) |
21:30.36 | DrDeke | oz: Yeah, pretty much. |
21:30.49 | _ozstealth_ | DrD |
21:31.00 | _ozstealth_ | can you provide any advice? |
21:31.18 | DrDeke | What you probably want to look into is a "callfile" |
21:31.18 | DrDeke | j/sec |
21:31.25 | _ozstealth_ | k |
21:31.37 | DrDeke | http://www.voip-info.org/tiki-index.php?page=Asterisk+auto-dial+out |
21:31.39 | Oryn | anyone running the latest bristuff ? I cant get * to reload |
21:32.35 | Jesster | anyone successful making custom ring tones for Cisco 7940/7960? i've followed what i've read and still no luck. |
21:33.03 | *** join/#asterisk Stealthmethod (n=123@72.242.62.209) |
21:34.10 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
21:36.45 | *** join/#asterisk tuxinator_linuxM (n=spabin@70-32-106-248.ontrca.adelphia.net) |
21:36.46 | *** join/#asterisk Micc (n=dotirc@c-24-16-228-130.hsd1.wa.comcast.net) |
21:37.20 | Micc | why does my callerid show me the number instead of the name when I do set CALLERID(name)=something |
21:41.59 | *** join/#asterisk southtel (n=slester@c-24-30-2-230.hsd1.ga.comcast.net) |
21:44.52 | KranZ | Micc: paste your code for that |
21:45.01 | *** join/#asterisk Flixor- (n=Flixor@ip5457002f.direct-adsl.nl) |
21:45.03 | KranZ | the line |
21:45.06 | Flixor- | hi everybody |
21:45.34 | _ozstealth_ | DrD: Thanks for that - that page and one that links from it look awesome. Thanks for your help |
21:45.39 | DrDeke | Cool, good luck. |
21:45.41 | Micc | exten => s,1,Set(CALLERID(name)=GlycoSystem) |
21:45.45 | DrDeke | Feel free to ask in here if you have any more questions. |
21:46.45 | southtel | Anyone know of any reason why a DeadAGI+phpagi app would end immediately when the user hangs up? |
21:46.56 | KranZ | Micc: http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Set |
21:46.58 | Micc | I'm guessing that the PRI provider doesn't allow us to change the name. |
21:47.22 | Flixor- | guys is it possible to call on linux via the voipbuster network |
21:47.35 | KranZ | your telco does a "DIP" in the National CNAM database for caller name information |
21:47.37 | Flixor- | with a client |
21:48.13 | KranZ | Micc: so to change the name, you need access to the cnam db, which you wont get unless your a telco |
21:48.21 | KranZ | Micc: so ask the telco to change the name |
21:49.53 | Micc | KranZ, maybe the name was dropped when we transfered the number. |
21:50.02 | azzie | guys... 16bit PCM for SoX is .ul or .uw ? |
21:51.11 | flashbac1 | can someone help me with a realtime problem? |
21:51.20 | KranZ | Micc: yeah, ask whoever you ported over to to update the cnam info |
21:51.44 | Beirdo | azzie: ul is unsigned long (32 bit), uw is unsigned word (16 bit), ub is unsigned byte (8 bit) if my memory serves |
21:51.54 | Flixor- | KranZ, do you know i could use my voipbuster account on linux |
21:52.05 | Beirdo | and you can use sl, sw, sb too for signed |
21:52.11 | Beirdo | that's my memory of it |
21:52.20 | KranZ | Micc: i think its silly that they have to lookup caller name info and it cant just get passed along with the ani |
21:52.37 | KranZ | Flixor-: voipbuster? |
21:52.46 | Flixor- | yes |
21:52.53 | KranZ | to do what |
21:52.58 | Flixor- | to call |
21:53.07 | Flixor- | check out www.voipbuster.com |
21:53.13 | Flixor- | they have only a windows client |
21:53.30 | Flixor- | but then somebody said to me that you could call under linux allso because its using the aix protocol |
21:53.42 | KranZ | you man iax? |
21:53.44 | KranZ | er mean |
21:53.48 | Flixor- | ehm sorry |
21:53.50 | Flixor- | yes iax |
21:54.09 | KranZ | im sure there's a softiax client out there somewhere |
21:54.09 | *** join/#asterisk Assid (i=assid@59.183.2.83) |
21:54.21 | DrDeke | yeah i don't think too manypeople run asterisk on AIX machines ;) |
21:54.52 | KranZ | or you can setup * to route to voipbuster and use a linux sip client to connect to * |
21:55.04 | *** part/#asterisk southtel (n=slester@c-24-30-2-230.hsd1.ga.comcast.net) |
21:55.27 | KranZ | [linux sip client] <-sip-> * <-iax-> voipbuster |
21:55.38 | *** join/#asterisk g__ (n=g@itd01fw-fibre.itdepartment.com) |
21:56.06 | Flixor- | ehm |
21:56.10 | waddy | need help with an error when compiling chan capi |
21:56.11 | waddy | make: *** [chan_capi.o] Error 1 |
21:56.18 | Micc | Kranz, when I set the caller id number should it include the 1 and can it contain dashes? |
21:56.25 | synthetiq | how can u limit the number of concurrent calls a line can have? |
21:56.25 | Flixor- | well i have kiax installed KranZ |
21:56.40 | KranZ | Micc: only 10 digits |
21:56.49 | KranZ | Flixor-: have you tried it? |
21:56.54 | *** join/#asterisk pryk (n=tmalkut@fw.orasoft.net.pl) |
21:57.07 | *** part/#asterisk _ozstealth_ (n=ozstealt@64.141.48.2) |
21:57.13 | Flixor- | ehm yes i seems that i can login to the server |
21:57.14 | *** join/#asterisk ]expic (i=xuy@h160-60.uni.net.ua) |
21:57.22 | Flixor- | but then again when i call to myself it doesnt work |
21:57.25 | KranZ | Micc: literally 10 digits |
21:57.36 | ]expic | does anybody know how to get calling SIP Ip address in AGI variable? |
21:57.43 | KranZ | does the windows version work? |
21:58.04 | DrDeke | I find that some mobile phone operators in the USA prepend a + to their incoming callerID values, which of course forms an invalid number unless you include the "1" |
21:58.17 | ]expic | does asterisk set SIP client IP to some special variable? |
21:58.19 | DrDeke | (On the other hand, I have not yet tried including the "1" and seeing if that works on a landline callerid box) |
21:59.06 | Flixor- | yes the windows version does work |
21:59.49 | KranZ | Flixor-: dunno, you'll have to start debugging your linux setup |
22:00.12 | ]expic | i need to track client IP addresses in PHP AGI script |
22:00.12 | g__ | Hi, I'm having a problem where Agents answering calls from a queue can't transfer those calls to another extension. around 50% of the time they get dropped. I've searched the mailing lists, and I've found several people mentioning a similar problem, but I haven't found any solutions. Any ideas? |
22:00.14 | ]expic | plz help me |
22:00.51 | KranZ | g__: do you have the latest * version? |
22:01.12 | g__ | The latest old stable (1.0.9) |
22:01.31 | KranZ | g__: do they know how to transfer? |
22:01.33 | g__ | The agent code is a bit newer though.. are there reported problems with the 1.0.x branch? |
22:01.54 | g__ | Yes: we just tried 5 times in a row.. on the 5th one it failed. |
22:01.55 | *** join/#asterisk southtel (n=slester@c-24-30-2-230.hsd1.ga.comcast.net) |
22:02.06 | KranZ | did it spit out an error? |
22:02.20 | g__ | No, there's nothing on the console. |
22:02.34 | g__ | The caller remains "off hook" but without hold music. |
22:02.34 | KranZ | who hungup 1st |
22:03.18 | *** join/#asterisk nagl (n=nagl@213.235.241.6) |
22:03.18 | g__ | Neither.. it was an attended transfer: A joins the queue, B answers, B talks to C brief, then hits 'transfer'. |
22:03.26 | KranZ | are you using flash to transfer? |
22:03.42 | g__ | These are hardware Polycom Sip phones. |
22:04.20 | KranZ | hmm... |
22:04.40 | KranZ | g__: [TK]D-Fender knows alot about polycom phones but he's usually on in the morning |
22:04.47 | KranZ | g__: you should try asking him |
22:04.50 | DrDeke | TKD was in here about 30 minutes ago |
22:04.52 | Druken | g__: i had this problem a while ago, i don't think i ever fixed it... (don't remember) make sure your contexts are correct :) |
22:05.04 | Druken | remember you can't transfer to an extension that doesn't exsist |
22:05.05 | g__ | [transfer] extention [send] .. etc. |
22:05.29 | g__ | KranZ: I'll ask him first thing tomorrow.. thanks for the suggestion. |
22:05.35 | KranZ | np |
22:06.20 | g__ | Druken: the contexts look ok. The 'attended transfers' work well: B gets to talk to C without any problems. |
22:06.50 | g__ | It's just the Zap/1-1 line that gets "lost" in the system as soon as the agent hits "transfer". |
22:07.08 | waddy | anyone can help me with chan_capi 0.3.5 ? |
22:07.18 | g__ | Still, I appreciate you checking the obvious.. |
22:07.33 | waddy | i get an error on make |
22:07.43 | waddy | chan_capi.c:4920: warning: implicit declaration of function `capi20_release' |
22:07.43 | waddy | chan_capi.c:4933: dereferencing pointer to incomplete type |
22:07.43 | waddy | chan_capi.c:4936: dereferencing pointer to incomplete type |
22:07.43 | waddy | make: *** [chan_capi.o] Error 1 |
22:07.54 | waddy | Its on RHEL3 |
22:08.29 | southtel | Could someone help me clear up some quiestions about AGI? |
22:08.32 | waddy | its doing my headin ive tried everything |
22:08.55 | southtel | Specifically, I'd like to know just what DeadAGI is used for. |
22:09.19 | g__ | Anyways, thanks for the suggestion Druken and KranZ! |
22:10.25 | ]expic | does asterisk set SIP client IP to some special variable? |
22:11.46 | *** join/#asterisk toddf (n=toddf@ns0.fries.net) |
22:12.56 | KranZ | g__: np, good luck |
22:12.57 | southtel | expic: I'm pretty sure that there isn't. |
22:13.26 | g__ | Thanks KranZ. What do you use Asterisk for? |
22:13.41 | ]expic | southtel: how it's easy to get client IP from AGI script if this IP is dynamic? |
22:14.41 | *** join/#asterisk Dead-Bum (n=Satan@tor/session/x-6ac3df61d4807d5e) |
22:14.41 | southtel | expic: I'd guess it's possible, but it may take some fenagling. |
22:15.32 | southtel | expic: These SIP users, are they calling in? What are you trying to do? |
22:15.57 | ]expic | southtel: they are calling in |
22:16.04 | ]expic | southtel: i do prepaid service for them |
22:16.20 | ]expic | southtel: now i want to create additional field with client IP in CDRS |
22:16.33 | ]expic | southtel: but cannot find any predefined channel variable with client IP |
22:17.00 | *** join/#asterisk Dead-Bum (n=Satan@tor/session/x-7433e025c63f1248) |
22:17.15 | Dead-Bum | I can't seem to find a clear guide on getting asterisk to register through openSER (or regular ser for that matter) anyone got a good link? |
22:17.21 | ]expic | southtel: regarding deadagi, it's used when the channel is hanged up |
22:17.51 | KranZ | g__: pstn gateway, ivr, call-forwarding, directory, emergency voicemail, time of day routing |
22:18.29 | KranZ | g__: i use SER as my SIP registrant/proxy |
22:18.41 | southtel | expic: They're calling in over SIP or over the PSTN? |
22:18.54 | g__ | How's SER treating you? I've been considering SER for a while now.. |
22:19.12 | KranZ | g__: good, it's quite stable |
22:19.16 | *** join/#asterisk RoyK (n=roy@55.80-202-161.nextgentel.com) |
22:19.28 | KranZ | g__: i like the dialplan configuration better b/c its more programming oriented |
22:19.29 | southtel | expic: I thought that DeadAGI also allowed the agi to continue on a channel once it's hung up. Is that not so? |
22:19.39 | KranZ | g__: and you can restart it w/o dropping calls |
22:19.43 | g__ | Does it take a lot of time to keep the configuration in sync in asterisk? |
22:20.00 | *** part/#asterisk ctooley (n=ctooley@rrcs-24-227-212-162.sw.biz.rr.com) |
22:20.05 | KranZ | g__: my SER conf is pretty static |
22:20.10 | g__ | (I understand Asterisk still needs to handle voicemail if you're using SER.) |
22:20.16 | KranZ | yeah |
22:20.31 | ]expic | southtel: don't think so, because it starts on h extension, when call is hanged up |
22:20.37 | g__ | Does it have to change every time you add or modify a sip client? |
22:21.07 | KranZ | g__: nope, u just add them to the database |
22:21.17 | KranZ | (db on SER box) |
22:21.35 | KranZ | ...and of course make a voicemail box on * |
22:21.37 | g__ | Berkley DB or mysql/postrgres? |
22:21.42 | KranZ | mysql |
22:21.50 | g__ | Intersting.. |
22:22.08 | KranZ | you should start learning it now, the sooner the better |
22:22.10 | g__ | thanks for the rundown, I'll certainly have to look at it. |
22:22.15 | southtel | expic: if your users are connecting via SIP to initiate calls, then you could look at the channel. |
22:22.22 | g__ | It improved the stability of your system? |
22:22.33 | KranZ | it takes a load off * |
22:22.52 | southtel | expic: the channel will have the SIP id, which you probably could use to then do a lookup. |
22:22.55 | KranZ | and allows * to be on a private network |
22:23.14 | g__ | We have several PBXes at many sites, so SER has been on my list of things to look at for a while. |
22:23.20 | KranZ | DrDeke: what lang agi? |
22:23.30 | g__ | Nugget: is that SER-related tong mutilation? |
22:23.33 | KranZ | so you do office * setups? |
22:23.39 | bsdfreak | tao |
22:23.51 | DrDeke | KranZ: Perl. Actually, though, the AGI portion of the system works fine; I am just having trouble with the perl script I am using to process the output of the AGI script :) |
22:24.12 | Nugget | no, mysql-induced. |
22:24.16 | DrDeke | What I am working on is not that hard, but I keep screwing it up ;). I really should just do it tomorrow, but I want to finish it bfore I go home for the day. |
22:24.16 | DrDeke | before* |
22:24.25 | g__ | I have perl scripts to handle individual sites, but inter-site configuration is still all manual. |
22:24.25 | KranZ | DrDeke: i need to get into that, you do any db backend stuff with agi (non * db)? |
22:24.37 | *** join/#asterisk Lurr (n=pr0ph3t@m615e36d0.tmodns.net) |
22:24.48 | *** part/#asterisk Lurr (n=pr0ph3t@m615e36d0.tmodns.net) |
22:24.52 | g__ | Nugget: do you have a choice with SER? (I heard rumour * doesn't support postgre.) |
22:24.53 | DrDeke | KranZ: Not yet, but most programming/scripting languages (particularly perl) have very easy interfaces to various databases. |
22:24.58 | Nugget | dunno |
22:25.01 | ]expic | southern: yes i have sip id |
22:25.16 | Nugget | you can use postgresql for realtime in asterisk. at least, some people do. |
22:25.27 | Nugget | I dunno how solid the support is. I've never looked at realtime. |
22:25.43 | *** join/#asterisk dalabera (n=dalabera@146.82.190.164) |
22:25.50 | dalabera | hello everyone |
22:26.39 | dalabera | installing 1.2.1... Is make linux26 removed from the install process? |
22:26.43 | southtel | expic: what flavor of agi are you using? |
22:26.57 | SwK[Work] | damn it batman |
22:27.56 | KranZ | dalabera: are you in linux 2.6? |
22:27.56 | ]expic | southtel: http://apollo.bcwireless.net/~matthewa/phpagi/ |
22:28.17 | dalabera | yes |
22:28.23 | g__ | Mysql might suck as a relational database.. but maybe it's not so bad as a directory service. |
22:28.34 | *** join/#asterisk vexorg (n=vexorg@CPE000021ded913-CM001225419164.cpe.net.cable.rogers.com) |
22:28.37 | *** join/#asterisk kokey (n=ubunture@201.153.63.79) |
22:28.41 | KranZ | dalabera: so why not just use "make" |
22:28.53 | southtel | expic: in that case, you can probably use phpagi-asmanager |
22:28.53 | waddy | anyone can help me with chan_capi 0.3.5 ? |
22:28.54 | Oryn | is anyone using sipdiscount.com? I cant get any audio via my sip phones, but my isdn phones work fine |
22:28.58 | waddy | i get an error on make |
22:29.00 | waddy | make: *** [chan_capi.o] Error 1 |
22:29.24 | southtel | expic: You'll want to do a SIPshowpeer, and then parse the results. |
22:31.19 | dalabera | Kranz How long you being using Asterisk? |
22:31.26 | SwK[Work] | anyone know a good bug in chan_sip where on a reload chan_sip.so chan sip just hangs with no indication as to why |
22:31.50 | KranZ | dalabera: almost 2 years |
22:32.23 | dalabera | good |
22:32.42 | KranZ | i wish it were more |
22:33.05 | ]expic | <southtel> yes thank you |
22:33.14 | dalabera | then you should knew that there used to be a make linux26 especifically for linux 2.6 kernel |
22:33.48 | *** join/#asterisk Koshatul (n=evangeli@ip157-65-132.cust.bit.net.au) |
22:34.04 | KranZ | dalabera: i might have run across it, but never needed it |
22:35.02 | ]expic | southtel: actually i can easyly do that executing asterisk -r -x and parsing results, don't think i need to use this library just for getting IP |
22:35.07 | southtel | expic: no problem. That's not ideal, but it works. |
22:35.33 | *** join/#asterisk toddf (n=toddf@ns0.fries.net) |
22:35.35 | southtel | expic: Much better! Much less overhead as well. |
22:36.23 | DrDeke | shit. |
22:36.49 | DrDeke | Well, I think it's officially time to quit working for the day, as I just overwrote some unbackedup work of mine while making a mistake with I/O redirection :X :X :X |
22:37.32 | *** join/#asterisk tainted- (n=somewher@mail.k2usa.com) |
22:38.47 | DrDeke | Yeah, I know, but I only wrote this a few hours ago |
22:39.11 | KranZ | heh |
22:39.19 | DrDeke | It won't be particularly hard to replace, it's just annoying when you do something that stupid. |
22:39.58 | *** join/#asterisk MikeJ[Laptop] (n=ircatjer@d14-69-8-30.try.wideopenwest.com) |
22:40.14 | test34 | What do I need to connect a VoIP adapter to the computer so I can use asterisk ? |
22:40.27 | sivana | heh |
22:40.31 | ManxPower | test34, money. |
22:40.34 | KranZ | a usb cable perhaps |
22:40.35 | DrDeke | HA! |
22:40.36 | sivana | VoIP adapter to a computer |
22:40.47 | ManxPower | test34, Buy a SIPura box. |
22:40.51 | sivana | test34: why do you want to connect it to your computer? |
22:40.53 | DrDeke | yeah. money talks, especially when it comes to telephone systems ;) |
22:41.02 | KranZ | no pun intended |
22:41.24 | test34 | so I can use it as FXS ? |
22:41.33 | KranZ | www.sipura.com |
22:41.34 | KranZ | yes |
22:41.50 | DrDeke | Only some sipura units have an FXS port though. |
22:41.55 | DrDeke | You would need to make sure you get one that does. |
22:41.58 | sivana | test34: you have a computer with asterisk AND an adapter? |
22:42.03 | *** join/#asterisk MikeJ[Laptop] (n=ircatjer@d14-69-8-30.try.wideopenwest.com) |
22:42.09 | DrDeke | Sorry, it's only some that have fxO ports |
22:42.19 | KranZ | the spa2100 is what i recommend |
22:42.36 | test34 | sivana, I only have asterisk right now, but I will get an adapter with the cable company when they come install it |
22:42.45 | KranZ | heh |
22:42.48 | KranZ | roadrunner |
22:42.50 | sivana | what kind of adapter? |
22:42.51 | KranZ | ? |
22:43.28 | synthetiq | how can u limit the number of concurrent calls a line can have? |
22:43.33 | KranZ | test34: most likely you wont be able to do anything with it except plug a coax cable into it, your ethernet and your phone |
22:43.36 | sivana | synthetiq: groups |
22:43.47 | test34 | sivana, I'm not sure |
22:44.02 | KranZ | test34: what cable company? |
22:44.11 | test34 | KranZ, brighthouse networks |
22:44.20 | *** join/#asterisk MikeJ__ (n=ircatjer@d14-69-8-30.try.wideopenwest.com) |
22:44.33 | synthetiq | is that a coman sivana |
22:44.36 | sivana | synthetiq: look on the wiki... it's with group count or something |
22:45.06 | *** part/#asterisk g__ (n=g@itd01fw-fibre.itdepartment.com) |
22:45.09 | test34 | they will plug my phone line to the adapter so I can use the regular phone plugs |
22:45.15 | *** join/#asterisk saftsack (n=saftsack@p54A7FFDA.dip.t-dialin.net) |
22:45.19 | saftsack | hi |
22:45.38 | *** join/#asterisk Meaty (n=cp_simbu@office.abi.ca) |
22:45.38 | KranZ | synthetiq: you could set and increment a global variable, then de-increment it when the call hangs up |
22:45.47 | KranZ | synthetiq: add some gotoifs |
22:45.49 | sivana | KranZ: it's even easier than that |
22:45.52 | sivana | I'm looking it up |
22:46.06 | KranZ | sivana: there's a line in sip.conf that'll do it for sip |
22:46.35 | sivana | SetGroup |
22:46.48 | sivana | the sip parameters are deprecated |
22:46.48 | *** part/#asterisk dca_ (n=dca@sta-208-139-193-162.rockynet.com) |
22:46.49 | sivana | http://www.voip-info.org/wiki/view/Asterisk+cmd+SetGroup |
22:47.18 | sivana | synthetiq: look there.. it even has an example |
22:47.35 | KranZ | sivana: good find, i hadnt messed with that (nor needed). |
22:50.30 | ManxPower | KranZ, he should be aware of race conditions, however. |
22:51.36 | *** join/#asterisk InfideNino (i=InfideNi@164-233.surfsnel.dsl.internl.net) |
22:52.18 | *** join/#asterisk MikeJ[Laptop] (n=ircatjer@d14-69-8-30.try.wideopenwest.com) |
22:52.18 | *** join/#asterisk drray (i=drray@dsl254-011-243.sea1.dsl.speakeasy.net) |
22:52.28 | InfideNino | hi all |
22:52.33 | InfideNino | can somebody help me? |
22:52.45 | InfideNino | i need cisco firmware for a 7920... |
22:53.01 | ManxPower | InfideNino, Call Cisco. What firmware do you need? |
22:53.05 | sivana | InfideNino: now that would be illegal |
22:53.10 | ManxPower | SIP, SCCP/Skinny, MGCP, or H323? |
22:53.24 | InfideNino | yes i know you need a contract, but i'm just a home user (poor student...) |
22:53.35 | InfideNino | well if you have sip that would be amazing |
22:53.41 | InfideNino | but i was looking for sccp |
22:53.42 | sivana | everyone's a poor student |
22:53.55 | *** join/#asterisk mog_work (n=mogorman@gateway.digium.com) |
22:54.13 | InfideNino | how much is a cisco contract anyway? |
22:54.21 | sivana | so you didn't even call :p |
22:54.21 | drray | 88 is waht I paid |
22:54.36 | drray | 8 for the phone, and 80 for the general contract |
22:54.45 | sivana | heh |
22:54.55 | *** part/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
22:54.55 | InfideNino | well, how can i call them if my phone isn't working with the old firmware that it was shipped with ;-) |
22:54.57 | test34 | sivana, I will be getting a digital phone/modem combo |
22:55.02 | sivana | the firmware is free.. it's 88 million for the CD it came on |
22:55.09 | *** join/#asterisk MikeJ[Laptop] (n=ircatjer@d14-69-8-30.try.wideopenwest.com) |
22:55.14 | ManxPower | InfideNino, A support contract ($10/year) MIGHT get you access to the firmware, but it would not technically be a legal download |
22:55.20 | *** part/#asterisk DrDeke (i=dekemar@deculator.engin.umich.edu) |
22:55.22 | test34 | err.. digital phone (voip) adapter |
22:55.44 | InfideNino | so if it's not a legal download anyway, one of you guys might as well just share it with me |
22:56.02 | ManxPower | test34, Which specific make/model? |
22:56.19 | InfideNino | i guess we have a cisco contract at work, but no voip; could i use that? |
22:56.20 | test34 | I'm on hold, I should know soon |
22:56.31 | test34 | (on the phone with them) |
22:56.31 | ManxPower | test34, companies like AT&T, Vonage, etc LOCK the adapter to the service so you can't use it with anything else. |
22:56.40 | ManxPower | test34, well come back when you know. |
22:56.45 | sivana | InfideNino: if you can access TAC, you can download it |
22:56.47 | test34 | ok |
22:56.57 | InfideNino | what's TAC? |
22:56.59 | KranZ | i need to start thinking of new year's plans |
22:57.03 | ManxPower | test34, You are not getting it via a service provider, are you? |
22:57.12 | sivana | InfideNino: cisco's download area |
22:57.29 | test34 | ManxPower, I get it from brighthouse networks.. I shouldnt ? |
22:57.50 | ManxPower | test34, never heard of them, but as I said most companies LOCK their device to their service. |
22:57.59 | KranZ | test34: whatever they give you will be locked and only usable with their system |
22:58.18 | ManxPower | Really - JUST BUY A SIPURA |
22:58.20 | test34 | it is free anyways |
22:58.35 | KranZ | ManxPower: but that will only get him to his * box |
22:58.37 | test34 | I might ask for a sipura for christmas |
22:58.43 | ManxPower | test34, if it's free then it will be locked and you'll have to sign a contract (1 year is common) |
22:58.46 | KranZ | ManxPower: which prolly goes nowhere |
22:59.09 | ManxPower | KranZ, Unless the ATA has an FXO port it's not going to do him any good anyway. |
22:59.11 | test34 | there is no contract.. but they might want it back when I discontinue the service |
22:59.18 | KranZ | true |
22:59.24 | *** join/#asterisk Darwin35 (n=Darwin@sta-208-139-193-162.rockynet.com) |
22:59.37 | ManxPower | test34, You won't be able to get any help if you use a device nobody else has. |
22:59.57 | *** join/#asterisk MikeJ__ (n=ircatjer@d14-69-8-30.try.wideopenwest.com) |
23:00.55 | ManxPower | A SIPura is like $60 or so. |
23:01.05 | KranZ | fyi: brighthouse networks resells roadrunner |
23:01.22 | ManxPower | If you cant afford $60 then you should be using a SoftPhone (ALL SOFTPHONES SUCK!) and not a hardware device. |
23:03.32 | ManxPower | Cisco needs to start putting a USB interface on their routers. Then you could plug in a flashdrive for the OS |
23:03.44 | test34 | KranZ, yeah I have roadrunner for internet cable |
23:04.27 | test34 | they cant tell me what modem/adapter I will get , they say they have many brands/models....... |
23:06.05 | ManxPower | test34, Whatever they send you it won't work with Asterisk. |
23:07.10 | test34 | ManxPower, ok thanks |
23:09.00 | KranZ | anyone tested the SPA-941? |
23:09.20 | znoG | does anyone know how to make asterisk wait about 3 seconds before detecting the distinctive ring pattern? |
23:09.29 | ManxPower | KranZ, several have. They talk about their experiences on the mailing lists. |
23:09.48 | ManxPower | ~mailinglist |
23:09.50 | jbot | well, mailinglist is Search Asterisk mailing lists by prepending site:lists.digium.com to your Google search. Browse the mailing list archive at http://lists.digium.com/ or search through it at http://www.asteriskguru.com/archives/search.php |
23:10.41 | KranZ | im gonna try building an office pbx setup with a mini-itx mobo acting as *box and internet gateway.... looking into ip phones for the testbed |
23:11.21 | ManxPower | Has everyone here read the SECURITY file in the Asterisk source? If not, you should do so right now. |
23:14.30 | ManxPower | Kran I recommend the following for a testbed: Polycom Soundpoint IP 50x, SIPura SPA-941, SIPura SPA-2100, Digium TDM11B |
23:15.05 | ManxPower | If you have the money get a TE20xP and an Adtran TA750 or TS850 off of eBay |
23:15.07 | KranZ | i'd like to try the polycom and spa941 |
23:15.22 | KranZ | i'd build the box to connect to SER |
23:15.59 | ManxPower | The list, in my opinion, gives you the most exposure to decent products without spending a fortune testing every product out there. |
23:16.35 | ManxPower | We wasted money on GrandStream, Cisco, Zultsys, etc. |
23:18.40 | drray | problem is phone tastes are subjective |
23:19.02 | KranZ | true, i have a few spa-2100's deployed and only need t1 interfaces |
23:19.10 | KranZ | havent messed with any ip phones tho |
23:19.13 | drray | you'll not get my ciscos from me |
23:19.39 | KranZ | cisco is everywhere which makes the hw cheap, but then you get stuck with licensing which sucks ass |
23:19.50 | Nugget | I like my ciscos well enough, but I don't think they're worth what they cost. |
23:20.12 | KranZ | im sure they're sweet when you have em runnin with all the features |
23:20.23 | *** join/#asterisk PakiPenguin (n=pingu@linuxpakistan/admin/pakipenguin) |
23:20.24 | KranZ | of course, you'll be short an arm and a leg |
23:20.41 | PakiPenguin | hello everyone |
23:20.44 | PakiPenguin | i have a fake ring problem |
23:20.45 | PakiPenguin | i mean when i call some number , after dialing the number , while the call is still being made ( in progress of being sent to the voip company ) , i get this ring ring which is basically my system generated , i want the tone to start when the call actually starts ringing at the number that i called |
23:20.45 | waddy | anyone can recommend a pre-built asterisk system to buy? |
23:20.55 | KranZ | cut-n-paste |
23:21.09 | KranZ | waddy: now why would you go and do that? |
23:21.29 | PakiPenguin | this wasnt in 1.0.9 , i started getting it in 1.2 just now , as i upgraded , can anyone help me? |
23:21.31 | waddy | i want a nice web interface, stable and features |
23:21.56 | KranZ | everyone wants a nice web interface |
23:22.01 | drray | not me |
23:22.10 | KranZ | i'd like to get one goin for end users |
23:22.13 | KranZ | a dashboard |
23:22.16 | waddy | yer |
23:22.24 | waddy | sipX looks ok |
23:22.54 | KranZ | check VMs, change features, check call records... |
23:23.24 | KranZ | i hate web design |
23:23.50 | drray | problem is like the phones, a web interface would be subjective as well |
23:25.54 | drray | I've written some simple tools for managing asterisk, mainly for bossman so I can keephim out of vi |
23:26.56 | *** join/#asterisk implicit (n=implicit@ip70-181-114-97.oc.oc.cox.net) |
23:27.19 | implicit | anyone have any grandstream equipment on firmware 1.0.7.11? |
23:27.20 | ]expic | drray: did you try AMP? |
23:27.37 | drray | I did, and I don't like AMP |
23:27.53 | ]expic | drray: rather good interface |
23:28.28 | KranZ | Nugget: for a dashboard? |
23:28.32 | KranZ | dammit |
23:28.35 | Nugget | nope. |
23:28.37 | KranZ | now how will i know |
23:28.39 | KranZ | oh |
23:28.44 | Nugget | I'm happy with vi to manage my asterisk. :) |
23:28.51 | file[laptop] | fatality! |
23:29.12 | drray | I'm happy too, it's just when bossman gets ideas |
23:29.18 | file[laptop] | Nugget: do you come with sweet and sour sauce? |
23:29.31 | nvrs | well who knows best |
23:29.32 | file[laptop] | and yes, I'm running out of stuff to do with your nick |
23:29.35 | nvrs | you or your boss |
23:29.35 | Nugget | nope, honey mustard. |
23:29.42 | nvrs | you got to shut him down |
23:30.35 | KranZ | the nugget was $5.50 |
23:30.52 | file[laptop] | whyfor must you bill me |
23:31.01 | PakiPenguin | hmmm.. anyone? |
23:31.07 | *** join/#asterisk tessier (n=treed@rrcs-67-53-110-66.west.biz.rr.com) |
23:31.19 | *** join/#asterisk De_Mon (n=de_mon@fl-69-69-147-198.dyn.sprint-hsd.net) |
23:31.49 | KranZ | PakiPenguin: i get that sometimes on a sipura on a cvs head from 2 months ago |
23:31.53 | KranZ | double ring |
23:33.13 | PakiPenguin | KranZ: it never happened on 1.0.9 stable |
23:33.56 | *** join/#asterisk zemmad (n=root@208.0.230.116) |
23:34.40 | zemmad | how well does asterisk work with eyeBeam?? |
23:36.49 | *** join/#asterisk heison (n=heison@CPE000a01d49e6f-CM00111a59bec2.cpe.net.cable.rogers.com) |
23:38.02 | KranZ | PakiPenguin: it probably did |
23:38.38 | *** join/#asterisk Rez (i=lorez@freenode/staff/lorez) |
23:39.52 | PakiPenguin | KranZ: thanks to implicit , if you set |
23:40.01 | PakiPenguin | progressinband=no , it wont happen |
23:40.02 | PakiPenguin | :) |
23:41.27 | p1tst0p | if it want to turn on one touch recording for inbound calls.. where would i put the Ww options ? |
23:42.35 | brettnem | check out phonecall to manage asterisk |
23:42.39 | brettnem | ~phonecall |
23:43.36 | brettnem | jbot, phonecall is a web gui for Asterisk management and can be found at http://www.vecsector.com/phonecall |
23:43.38 | jbot | brettnem: okay |
23:43.43 | brettnem | ~phonecall |
23:43.44 | jbot | i guess phonecall is a web gui for Asterisk management and can be found at http://www.vecsector.com/phonecall |
23:49.42 | *** join/#asterisk harryvv (n=none@S010600a0c93f6f7e.vs.shawcable.net) |
23:49.57 | *** join/#asterisk zmauve (n=mikael@c-9131e253.57-1-64736c10.cust.bredbandsbolaget.se) |
23:50.20 | zmauve | Hi, I just got my asterisk system up and running :) it works great! |
23:50.32 | mog_work | yay |
23:50.56 | zmauve | I will propose we run it at work on monday :) |
23:51.22 | zmauve | (we have no telephony solution at work) |
23:51.54 | harryvv | you mean no pbx |
23:52.03 | zmauve | yep, we have no pbx |
23:53.13 | zmauve | btw, in my dialplan I now have: [incoming] exten => s,1,Answer() exten => s,n,Playback(demo-congrats) exten => s,n,Hangup() |
23:53.16 | docelm0 | ya buddy! |
23:53.32 | Corydon-w | Qwell: please try latest 1.2 |
23:54.25 | zmauve | I tried to add exten => 10,1,Answer() and more so that I could dial my own phone number with an added 10 to it, but I still always get to the "s" extension; am I doing something impossible? |
23:54.36 | nextime | anyone using mcc on SVN-trunk-r7230 ? |
23:54.36 | harryvv | I wonder if newer digium pstn cards have eliminiated that click when the pbx picks up |
23:54.57 | zmauve | harryvv, I hear no clicks, and I got my new card yesterday :) |
23:55.11 | harryvv | zmauve, what card |
23:55.25 | zmauve | tdm400p with on fxs and one fxo port |
23:55.36 | zmauve | lspci says TigerJet something |
23:55.44 | harryvv | yea popular card. I have the much older x100p card. |
23:55.47 | znoG | anyone here use distinctive ring with asterisk? |
23:55.57 | harryvv | znog I have |
23:56.19 | zmauve | harryvv, you don't happen to know If what I am describing above is impossible? |
23:57.01 | harryvv | s,2 s,3 not s,n |
23:57.12 | KranZ | 6pm, time to go home |
23:57.20 | znoG | harryvv: are you able to show me your /etc/asterisk/zapata.conf? |
23:57.23 | KranZ | bbt |
23:58.00 | znoG | harryvv: i just want to check whether I've configured my zapata.conf correctly |
23:58.02 | zmauve | harryvv, the asterisk book says you can use s,1 + s,n with asterisk 1.2.x; am I misunderstanding something? |
23:59.46 | harryvv | best is to look at voip-info.org |