| 00:00.52 | *** join/#asterisk Defraz (n=t0tal@fw.fuzecore.com) |
| 00:00.57 | drwelby | Only thing I can think of is that asterisk thinks they are calling in from outside, and therefore using [default] context |
| 00:00.57 | Qwell_ | sounds like what you think is the 7940 isn't |
| 00:01.23 | drwelby | Qwell_: hmm? |
| 00:01.28 | Qwell_ | Strom_C: k, didn't think so |
| 00:08.20 | drwelby | Qwell_: I'm sitting in front of the 7940 and watching the debug on the cli |
| 00:11.22 | *** join/#asterisk LeFallen (n=fallen@mail.stabat.com) |
| 00:13.15 | davevg-btwtech | drwelby: what does the cli command "sip show users" say? |
| 00:13.55 | drwelby | davevg-btwtech: Shows no users, because (I guess) all users are in users.conf, so they are peers |
| 00:16.52 | davevg-btwtech | any errors when you "reload chan_sip.so" ? |
| 00:17.57 | my007ms | once i update from asterisk 1.2 too 1.4 no one can register with his sip phone |
| 00:18.42 | drwelby | davevg-btwtech: no errors, it reloads fine |
| 00:19.45 | davevg-btwtech | drwelby: can you pastebin the full cli output from a failed call? |
| 00:20.01 | *** join/#asterisk mltlnx (n=mltlnx@cpe-72-225-244-190.nyc.res.rr.com) |
| 00:21.24 | drwelby | davevg-btwtech: http://www.pastebin.ca/729450 |
| 00:21.34 | *** join/#asterisk JunK-Y (n=junky@modemcable183.17-83-70.mc.videotron.ca) |
| 00:23.38 | davevg-btwtech | Found no matching peer or user for '192.168.16.37:50300' |
| 00:25.51 | drwelby | davevg-btwtech: I saw that. But I figured it would match by username (100) |
| 00:28.22 | drwelby | But that's a hint. If it doesn't match the user it assumes its an ouside SIP call and sends it to [default] |
| 00:28.38 | drwelby | So asterisk is sending it to the *right* context, just not the one I want. |
| 00:29.37 | davevg-btwtech | can you paste your users.conf (masking the passwords)? I'm not too familiar with using it, but we may see something you aren't |
| 00:29.38 | Strom_C | drwelby: type=peer matches by host, not my username |
| 00:29.51 | Strom_C | s/my u/by u/ |
| 00:30.56 | *** join/#asterisk ming_zym (i=ming_zym@nat/yahoo/x-a0749898e827339b) |
| 00:32.28 | drwelby | Here's users.conf: http://www.pastebin.ca/729459 |
| 00:32.53 | *** join/#asterisk idespinner (n=idespinn@cpe-75-82-69-211.socal.res.rr.com) |
| 00:33.13 | drwelby | Sorry, it's from the GUI |
| 00:33.21 | drwelby | Not my first choice, but I have to make it work |
| 00:35.06 | idespinner | any one got some links/documentation for iax trunking, i have a few but still having trouble with it |
| 00:37.19 | Strom_C | idespinner: what specifically are you trying to find out |
| 00:38.03 | idespinner | i have Asterisk 1.2(BE) and trying to trunk with an asterisk appliance(aa50) to make calls inbetween via 1 IAX trunk |
| 00:38.28 | Strom_C | right, but do you want protocol details or do you want configuration assistance or what? |
| 00:38.35 | Qwell_ | idespinner: trunking=yes |
| 00:38.39 | Qwell_ | that's pretty much it |
| 00:38.42 | idespinner | configuration assistance |
| 00:38.45 | idespinner | mainly |
| 00:38.50 | Strom_C | qwell: i thought it was trunk=yes |
| 00:38.59 | Qwell_ | trunk=yes then |
| 00:39.27 | *** join/#asterisk JRsup1 (n=chatzill@12-207-206-43.client.mchsi.com) |
| 00:39.28 | idespinner | is the trunk=yes necesary? |
| 00:39.35 | Qwell_ | to trunk? |
| 00:39.38 | Qwell_ | ...yes |
| 00:39.41 | idespinner | heh |
| 00:39.43 | Strom_C | idespinner: if you want your calls to use the trunking feature of iax2 |
| 00:39.54 | Strom_C | are you having problems with calls setting up but not trunking? |
| 00:40.00 | JRsup1 | can someone tell me how to get and add chan_mobile to my asterisk? |
| 00:40.04 | idespinner | right now I am |
| 00:40.05 | Strom_C | or are you having trouble with calls not setting up at all in the first place? |
| 00:40.07 | JRsup1 | ls |
| 00:40.14 | Strom_C | ERROR: You're on IRC |
| 00:40.15 | idespinner | "rejected connect attempt" |
| 00:40.27 | Strom_C | idespinner: then that's a setup problem, not a trunking problem |
| 00:40.30 | idespinner | yes |
| 00:40.31 | JRsup1 | ...yeah..wrong window |
| 00:40.49 | Strom_C | but you have BE; you have the AA50...wait till tomorrow and call digium for the support you've already paid for? :) |
| 00:40.54 | idespinner | yea |
| 00:41.00 | idespinner | but the aa50 is here at home |
| 00:41.10 | idespinner | sooner the better |
| 00:41.22 | idespinner | If I dont get it done by today, I will call them |
| 00:41.25 | idespinner | tomorrow |
| 00:41.50 | Strom_C | alright |
| 00:41.59 | Strom_C | why is it rejecting connect attempt? |
| 00:42.15 | idespinner | im guessing im not putting in the correct context in the configuration |
| 00:42.30 | idespinner | in either the Dial or on the recieving end |
| 00:42.47 | JRsup1 | I see a lot of talk about loading chan_mobile.so, but I can't find it |
| 00:43.18 | davevg-btwtech | drwelby: does "sip show peers" show an ip in the host field for your peer 100? is your 7940 registering successfully? |
| 00:43.30 | Strom_C | idespinner: you're going to have to provide at least some info; as experienced as I am with asterisk, i'm not psychic |
| 00:43.49 | idespinner | yea sorry |
| 00:43.49 | drwelby | davevg-btwtech: yes it is |
| 00:44.28 | drwelby | davevg-btwtech: This discussion may be relevant: http://lists.digium.com/pipermail/asterisk-dev/2005-December/017659.html |
| 00:48.53 | drwelby | AHA! port=unsecure fixes it! |
| 00:49.14 | drwelby | davevg-btwtech: Thanks for helping me out!!! |
| 00:50.20 | TrentCreek | you are welcome |
| 00:52.06 | idespinner | so, heres a basic rundown of the relevant config |
| 00:52.07 | idespinner | http://pastebin.com/m6a4a4f3b |
| 00:52.11 | idespinner | for my trunking |
| 00:52.30 | idespinner | the aa50 is behind a NAT |
| 00:53.17 | drwelby | s/port=unsecure u/insecure=port u/ |
| 00:54.44 | *** join/#asterisk russellb_ (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 00:54.44 | *** mode/#asterisk [+o russellb_] by ChanServ |
| 01:02.19 | Defraz | Are there any web based options for controlling a Meetme conference? |
| 01:04.24 | russellb_ | http://sourceforge.net/projects/web-meetme/ |
| 01:05.57 | Defraz | thanks |
| 01:06.13 | russellb_ | i'm not sure if that's the best one ... |
| 01:06.21 | Defraz | Hey it is a start right |
| 01:06.23 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 01:06.26 | russellb_ | yup |
| 01:09.48 | Defraz | poop asterisk 1.4 or later. |
| 01:10.25 | russellb_ | that's what you should be running :) |
| 01:11.00 | Defraz | :) I know just haven't switched yet. |
| 01:11.06 | FireMac | just finished installing asterisk. i want to use it with my voip, but i received an ata that is locked by my service provider. |
| 01:11.09 | Defraz | It runs so good with 1.2 |
| 01:11.15 | FireMac | is there a work around with this? |
| 01:11.18 | FireMac | pls help me |
| 01:11.24 | Defraz | FireMac dang, that sucks. |
| 01:11.36 | FireMac | no work around Defraz? |
| 01:11.37 | Defraz | Well, it would be worth it to buy an unlocked ata. |
| 01:12.07 | FireMac | i have unlocked ata too. but i dont know how to set it up like the locked ata from my service provider |
| 01:12.08 | Defraz | There is always a word around but it would take a long time to explain. Have you plugged it into your network? |
| 01:12.28 | Defraz | so you have it unlocked? |
| 01:12.45 | FireMac | i mean i have another ata that is unlocked |
| 01:13.29 | FireMac | my voip is currently working ok. but i want my asterisk to manage calls from my voip |
| 01:14.15 | *** join/#asterisk ZX81 (n=matt@202.49.106.158) |
| 01:19.07 | FireMac | any idea how to do this Defraz? |
| 01:19.08 | *** join/#asterisk pabs3 (i=daemon@60.242.186.48) |
| 01:24.53 | pabs3 | how would I get 1234 and 4321 from a mysql database to a dial call like Dial(SIP/4321&SIP/1234,20,r) ? (I want to create a class of extensions that is dialed from a specific extension - for the receptionists) |
| 01:25.34 | ZX81 | show application MYSQL |
| 01:25.46 | ZX81 | if you have asterisk addons installed |
| 01:27.02 | *** part/#asterisk lirakis (n=lirakis@cpe-68-175-38-65.nyc.res.rr.com) |
| 01:27.40 | pabs3 | :) |
| 01:29.08 | *** part/#asterisk hellop1 (n=hellop@cpe-66-8-248-9.hawaii.res.rr.com) |
| 01:29.15 | pabs3 | it doesn't seem to be possible to join several variables together with a certain string between them, or use arrays |
| 01:29.34 | jql | You can do whatever you have the patience for |
| 01:32.55 | jql | personally, though, I'd do the string manipulation in mysql (upon scrolling up and reading your Q) |
| 01:33.45 | FireMac | can someone tell me if this is ok setup. locked ata->fxo/fxs card->asterisk server? |
| 01:35.17 | *** join/#asterisk theHub (n=chatzill@ool-43577a99.dyn.optonline.net) |
| 01:35.29 | jql | well, I don't doubt you can do that |
| 01:35.35 | jql | should work fine |
| 01:36.46 | FireMac | that way i can manage my calls from my locked ata to my asterisk |
| 01:36.50 | FireMac | is that correct? |
| 01:41.34 | *** part/#asterisk davevg-btwtech (n=davevg-b@nj-67-76-177-147.sta.embarqhsd.net) |
| 01:42.49 | *** join/#asterisk kFuQ (n=somedude@c-67-185-112-20.hsd1.wa.comcast.net) |
| 01:43.31 | *** join/#asterisk Tond (n=t@CPE0014bf30c190-CM00194747ae5e.cpe.net.cable.rogers.com) |
| 01:44.20 | Tond | Hi I ahve a simple and quick question, I want to dial a number and play back and audio, i get the script to call the number, however after i don't hear anything... |
| 01:44.33 | Tond | I use dial command, and then the next priority is SayDigits(123) |
| 01:44.45 | Tond | but that doens't happen, is there another way of doing this? |
| 01:44.49 | *** join/#asterisk conreyt (n=theron@dsl.76.240.networkiowa.com) |
| 01:45.25 | TrentCreek | Anyone here got AstBill running? |
| 01:45.33 | Strom_C | Tond: because that's not how dial() works... |
| 01:45.38 | russellb_ | pfft, why bill people? :-p |
| 01:46.40 | Tond | Strom_c, so what do you recommend? How should i go about this? |
| 01:47.03 | Strom_C | Tond: what exactly do you want to happen? |
| 01:47.55 | Tond | I want to call a nuber, read a number (caller ID) and give them a few options. ex. press 1 for music on hold, 2 for operator, 3 hang up, etc.. |
| 01:48.00 | Tond | number |
| 01:48.13 | Strom_C | you want the called party to hear these options? |
| 01:48.22 | Tond | Yes |
| 01:48.27 | Strom_C | look into using call files |
| 01:48.31 | conreyt | Hi all. I'm new to asterisk and looking for a good debian/xen DomU setup. any links? |
| 01:48.37 | *** join/#asterisk grimsy (n=chatzill@203-206-220-214.perm.iinet.net.au) |
| 01:48.55 | Tond | Strom_c, that's when i drop a file in a directory in * and it calls automatically right? |
| 01:49.53 | Strom_C | yes |
| 01:50.22 | russellb_ | Strom_C: when is your next trip to HSV? The new training room is _awesome_ |
| 01:51.03 | Tond | Thanks |
| 01:51.03 | Strom_C | russellb_: i dont know |
| 01:51.14 | Strom_C | but i cant wait to see the finished room |
| 01:51.23 | Strom_C | ive been in the unfinished room |
| 01:51.32 | russellb_ | still setup to do, and the projector(s) aren't installed ... |
| 01:51.32 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 01:51.34 | russellb_ | but it's still awesome |
| 01:52.11 | *** join/#asterisk ixx (i=foobar@cpe-70-113-209-33.austin.res.rr.com) |
| 01:52.22 | Strom_C | nice |
| 01:52.24 | Strom_C | take photos! |
| 01:52.27 | Strom_C | show me! |
| 01:52.28 | *** join/#asterisk frenzy (n=frenzy@unaffiliated/frenzy) |
| 01:52.35 | russellb_ | alrighty, will do |
| 01:52.41 | Tond | I am looking into buying SwitchVox, but it doens't offer the functionality to a PSTN extention to be able to receive a notification saying who is calling and giving them option to accept, reject or transfer the call. So i am trhinking of doing that by putting an * box in between |
| 01:53.22 | russellb_ | switchvox is amazing (full disclosure: i work for digium ;) ) |
| 01:53.23 | Tond | I am getting very little pre sales support from SwitchVox, so i decided to look into it myself |
| 01:53.36 | Tond | Ya it is.. |
| 01:54.11 | Tond | and i am told what i am looking for may come later in the upgrade, but as for now i am stuck. I am pushing my company to buy it, ad the only hold up is this issue |
| 01:54.35 | Tond | I thought since Digium bought it, they may help me make an * scrip or macro to perform that task, but to me suprise no one cares,.. lol |
| 01:54.49 | Tond | and that is why i rolled up my sleeves to see what i can do.. |
| 01:55.12 | Tond | I got a sneak view when i was in Astricon in AZ, and i was amazed by it! |
| 01:55.17 | Tond | :) |
| 01:56.36 | *** join/#asterisk IgorG (n=FeedomPa@195.162.32.126) |
| 01:57.16 | Tond | @russellb_> perhaps u can help me get someone at Digium who would help out with this. It will be a very simple Macro to write for an Asterisk guru, but for me, it will take me a while.. :P |
| 01:57.22 | *** join/#asterisk waKKu (n=worth@unaffiliated/wakku) |
| 01:57.53 | russellb_ | Tond: heh, well, there is a custom development department that does custom configuration stuff if you're interested int hat ... |
| 01:58.02 | russellb_ | otherwise, you're in the right place to look for help |
| 01:58.07 | russellb_ | another place to try is the asterisk-users mailing list |
| 01:58.12 | Tond | :) |
| 01:58.34 | *** join/#asterisk docelmo (n=vircuser@c-76-99-153-242.hsd1.de.comcast.net) |
| 01:58.39 | Tond | Ok well, rolling up sleeves it is then.. lol |
| 01:59.09 | Tond | how about using the M() function in the dial command to run a Macro that would stream audio, would that work? |
| 01:59.22 | russellb_ | probably |
| 01:59.26 | russellb_ | i don't use this stuff :-p |
| 02:00.09 | Tond | well, thanks for the help.. |
| 02:00.15 | Tond | ya i just tried it and it worked... :) |
| 02:00.20 | russellb_ | cool |
| 02:00.36 | Tond | tnx again.. ;) |
| 02:00.42 | russellb_ | np |
| 02:09.33 | *** part/#asterisk pabs3 (i=daemon@60.242.186.48) |
| 02:11.19 | JRsup1 | can anyone help me get my asterisk set up to use chan_mobile? |
| 02:11.30 | Defraz | What is chan_mobile. |
| 02:11.31 | Defraz | ? |
| 02:12.31 | JRsup1 | can use bluetooth cell phone as an FXO(?) and bt headsets as an FXS (?) |
| 02:13.27 | Defraz | hmmm wow!! |
| 02:13.36 | Defraz | That would be really cool. |
| 02:13.53 | Defraz | Just slap a dongle on our home computer and whne you are there your phones work. |
| 02:13.58 | Defraz | that would be coool |
| 02:14.32 | *** join/#asterisk famicon (i=redz@c51447ddc.cable.wanadoo.nl) |
| 02:14.43 | JRsup1 | that's what I'm doing...I'm sick of running around the house looking for my cell every time it rings |
| 02:14.47 | JRsup1 | or trying to. |
| 02:15.26 | JRsup1 | I just discovered I may need to get my asterisk from 1.2.x to 1.4.? to even get chan_mobile. But I have no idea how. I'm running a trixbox 2.2 VM |
| 02:15.53 | Defraz | yea you will need to run 1.4 |
| 02:15.59 | Defraz | trixbox 2.3 beta |
| 02:16.09 | *** join/#asterisk _ShrikE (n=ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 02:16.54 | JRsup1 | hmmm...so I need to upgrade both. |
| 02:17.10 | Defraz | yea looks like it |
| 02:18.49 | JRsup1 | is there anyway to upgrade what I have to FreePBX 2.3beta and Asterisk 1.4.? |
| 02:18.59 | JRsup1 | or am I better off re-installing? |
| 02:19.03 | *** join/#asterisk Kumbang (n=kumbang@router-wmi.paume.ITB.ac.id) |
| 02:19.05 | JRsup1 | I have nothing set up right now really |
| 02:21.34 | flenders_ | JRsup1: I would go with a plain asterisk 1.4 |
| 02:22.25 | *** join/#asterisk waKKu (n=worth@unaffiliated/wakku) |
| 02:26.08 | JRsup1 | Ok, I'll give it a shot. |
| 02:26.11 | JRsup1 | thanks! |
| 02:26.44 | russellb_ | chan_mobile doesn't actually work with 1.4 ... |
| 02:26.50 | russellb_ | it's trunk only |
| 02:26.54 | russellb_ | (and 1.6 when released) |
| 02:29.23 | *** join/#asterisk breanna_ (n=irssi@c-24-10-238-92.hsd1.ut.comcast.net) |
| 02:29.57 | Defraz | Bummer that sounds so cool. |
| 02:29.57 | wwalker | under heavy call volume, will asterisk make use of multiple cores? |
| 02:30.04 | russellb_ | yes |
| 02:31.56 | wwalker | russellb_: "yes that sounds cool" or "yes it will use multiple cores? |
| 02:32.12 | file | yes it will use multiple cores |
| 02:32.17 | russellb_ | yes, it will use them |
| 02:32.32 | wwalker | file: russellb_ thanks |
| 02:33.04 | russellb_ | file: bridging thread pool ... we need to look into doing that with your new hotness |
| 02:33.41 | file | russellb_: HRM! |
| 02:34.02 | *** join/#asterisk revcane (n=dng@cpe-76-186-113-159.tx.res.rr.com) |
| 02:34.37 | russellb_ | file: i think that would help performance |
| 02:34.46 | russellb_ | file: but it's certainly tricky ... |
| 02:34.47 | wwalker | what's the highest concurrent bridged call count anyone has actually used with good performance? |
| 02:34.52 | file | it would |
| 02:35.02 | russellb_ | wwalker: many thousands ... |
| 02:36.26 | revcane | does anyone know of a decent walkthrough for setting up freeworlddialup? i've followed several differnt "howtos" but im still having a bit of trouble |
| 02:39.52 | ZX81 | man I am so in favour of AsteriskNow/Asterisk-GUI |
| 02:40.30 | ZX81 | it is so much nicer on config files than TrixBox et al |
| 02:40.43 | russellb_ | ZX81: awesome :) |
| 02:40.52 | *** join/#asterisk techie (n=techie@adsl-76-214-3-9.dsl.lsan03.sbcglobal.net) |
| 02:41.01 | ZX81 | yeah for sure - I'm now recommending it to customers who want a GUI |
| 02:41.26 | wwalker | where I am, they are under the impression that ~800 channels (e.g 400-450 inbound calls, all bridging after short IVR) is the limit and that above that there be dragons, crashes, and bad audio. |
| 02:41.44 | ZX81 | 800 is way too high |
| 02:41.55 | ZX81 | unless media goes elsewhere |
| 02:42.10 | ZX81 | call setup would normally mean a max of around 300 |
| 02:43.27 | russellb_ | it all depends on what you're doing |
| 02:43.29 | russellb_ | and hardware ..... |
| 02:43.40 | russellb_ | i think 800 could be perfectly fine |
| 02:43.45 | ZX81 | really? |
| 02:43.49 | ZX81 | with audio? |
| 02:43.49 | russellb_ | sure |
| 02:44.08 | ZX81 | might be time to do some more load tests! :) |
| 02:44.13 | russellb_ | yeah :) |
| 02:44.24 | russellb_ | and if you do, it might be worth comparing 1.2 to 1.4 |
| 02:44.31 | ZX81 | yeah for sure |
| 02:44.34 | russellb_ | we made some nice performance imrprovements |
| 02:44.38 | russellb_ | and are continuing to do so |
| 02:44.39 | wwalker | call comes in. IVR runs. call Dial to make a bridged call. rinse. repeat. currently at 400 channels, uses about 60% of a cpu according to top. |
| 02:44.41 | ZX81 | cool |
| 02:44.51 | wwalker | 1.4.7.1 |
| 02:45.02 | russellb_ | wwalker: transcoding? |
| 02:45.12 | file | and what technologies on each side? |
| 02:45.49 | russellb_ | file: cans and string to SIP over diet coke |
| 02:45.53 | JRsup1 | sorry, back for just a bit. Someone said chan_mobile DOESN'T work in 1.4.? ???? |
| 02:46.06 | file | russellb_: not diet dr. pepper? |
| 02:46.08 | wwalker | russellb_: no transcoding. dual 3G Xeon, 1 GB RAM, g711 |
| 02:46.17 | JRsup1 | and it's "trunk only"? |
| 02:46.21 | Tond | anyuone with some expereince with .call fiels? I am tring to set the callerid(num) variable in my .call file using Set: ${CALLERID(num)}=123 but it doesn't seem to work. Any ideas why? |
| 02:46.22 | JRsup1 | (don't know what that means) |
| 02:46.22 | russellb_ | JRsup1: correct |
| 02:46.27 | russellb_ | JRsup1: the development tree |
| 02:46.29 | file | wwalker: what technologies? |
| 02:47.41 | JRsup1 | oooo kay... I'm still learning a bit about linux but I'm doing some development in other areas. is there any way to get in on the dev tree for asterisk? I know it will be "bleeding edge" and all, but if it does this one thing it that would be all I need |
| 02:47.44 | wwalker | file, SIP, RTP, GigE direct to a Veraz soft switch network. asterisk 1.4.7.1 with ztdummy |
| 02:47.50 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 02:47.58 | file | is it Packet2Packet bridging? |
| 02:49.10 | wwalker | file: :( my ignorance is showing. incoming SIP call, DIAL called to make outgoing SIP call. Implicit bridging. is there a config that I can use to control the type of bridging? |
| 02:49.37 | file | wwalker: no, Asterisk will use the best it can... and it'll tell you on the CLI |
| 02:50.49 | wwalker | checking |
| 02:52.49 | JRsup1 | well, I'll be back on tomorrow probably...bed now. c u later |
| 02:52.49 | wwalker | file. I see nothing about packet2packet. What is the other possibility? I've looked at "sip show channels" and "core show channels verbose" |
| 02:53.16 | *** join/#asterisk fluffyfluffy (n=sbriggs@71.194.234.148) |
| 02:53.35 | file | it says when the channels are bridged together, it won't tell you in either of those CLI commands |
| 02:54.11 | wwalker | oh. let me look in the full log |
| 02:57.23 | wwalker | file, what can I set in the cli so it will show me the bridging tech as it occurs? I don't see anything in my full log |
| 02:57.45 | russellb_ | core set verbose 5 |
| 02:57.52 | russellb_ | but don't do that at high call volume .. |
| 02:58.18 | file | if you still don't see it doing p2p... then something is preventing it, and if you can find out what is preventing it you can increase your capacity |
| 02:58.47 | russellb_ | file: what are some things that would prevent it? transcoding? what else? |
| 02:58.54 | russellb_ | dtmf mode differences? |
| 02:58.57 | file | russellb_: some Dial() options |
| 02:59.09 | russellb_ | ah |
| 02:59.16 | wwalker | exactly what am I looking for? It may be there and I'm not seeing it. |
| 02:59.44 | file | Packet2Packet bridging <first channel name> and <second channel name> |
| 03:02.45 | wwalker | the calls go in and out the same trunk group, so the DTMF modes should be the same. but I don't grep -i packet .... |
| 03:03.57 | russellb_ | wwalker: what options are you using with Dial() ? |
| 03:07.47 | wwalker | russellb_: dial(sip/${SIPPEER}/${dest},60,gL(${totalsec}:${warn})); |
| 03:08.13 | file | I bet it is the L |
| 03:08.50 | russellb_ | yep |
| 03:08.55 | russellb_ | so it can do the timer checks .. |
| 03:09.26 | fluffyfluffy | Hi all does anyone know why the r and A options on the dial command wouldn't work (dial does work the options don't). I'm running 1.4.9. thx |
| 03:09.28 | wwalker | so it can insert audio into the stream for timeout warnings.... |
| 03:09.54 | russellb_ | right |
| 03:09.57 | wwalker | so asterisk stays at the heart of the RTP streams |
| 03:12.15 | wwalker | given that (which can't change) how many bridged call channel pairs would you guess? and will asterisk properly use simultaneous multiple CPU cores ? or do synchronization issues cause blocking that prevents actual simultaneous thread execution of RTP streams? |
| 03:16.06 | Tond | is there a way to park a call in specific parking space? |
| 03:16.15 | Tond | rather than letting asterisk pick it for me? |
| 03:18.23 | Tond | and also is there a way to access linux command shell from a dial plan? |
| 03:18.44 | Tond | so lets say evey time an extension is called a unix shell command runs |
| 03:20.04 | Defraz | That is a good idea. |
| 03:21.20 | russellb_ | Tond: the System() application |
| 03:22.34 | Tond | tnx |
| 03:25.31 | *** join/#asterisk rmayorga_ (n=rmayorga@unaffiliated/rmayorga) |
| 03:31.40 | *** join/#asterisk gardo (n=gardo@121.97.200.66) |
| 03:35.50 | *** part/#asterisk ming_zym (i=ming_zym@nat/yahoo/x-a0749898e827339b) |
| 03:36.42 | *** join/#asterisk ming_zym (i=ming_zym@nat/yahoo/x-394f7ed81e05989d) |
| 03:37.50 | *** join/#asterisk stkn__ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 03:40.30 | Defraz | Hey all, is anyone using a polycom 501 phone. |
| 03:40.33 | Defraz | ? |
| 03:41.46 | *** join/#asterisk NOT_guru (n=chatzill@209.145.181.91) |
| 03:42.21 | NOT_guru | question: the connection port for AIX2 is 4569 but is it tcp or udp? |
| 03:42.30 | NOT_guru | same question for SIP |
| 03:42.36 | russellb_ | udp only for iax2 |
| 03:42.43 | russellb_ | 5060 udp only for SIP (in asterisk, anyway) |
| 03:42.53 | NOT_guru | wonderfull thank you |
| 03:43.00 | russellb_ | np |
| 03:43.00 | NOT_guru | I am just setting up my FW queues |
| 03:43.07 | NOT_guru | so that helps alot |
| 03:43.12 | NOT_guru | greatly appreciated |
| 03:44.39 | Defraz | is there a way to do sip over TCP? |
| 03:44.43 | Defraz | with Asterisk? |
| 03:45.31 | russellb_ | there is a development branch with a patch for it |
| 03:45.39 | russellb_ | it's there for testing |
| 03:46.05 | russellb_ | but we don't have anyone actively working on it to get it finalized and merged right now ... |
| 03:47.06 | Defraz | Oh cool cool. |
| 03:47.49 | Defraz | I might have to test the patch. |
| 03:52.48 | codefreeze | Anybody ever seen that "Rock" puzzle? It's a 14-sided polyhedron, with 8 hexagonal facets, and 6 square facets, with spin-able tiles you can move around, with red, blue, yellow lines...? |
| 03:54.29 | Defraz | no |
| 03:55.06 | codefreeze | Made by a company in NZ called "Twisted Minds"....? |
| 03:56.15 | codefreeze | a gift from my brother; it's been on my desk for nearly a year. Still haven't solved it. It's driving me nuts! |
| 03:59.15 | codefreeze | I guess it'd be called a 'truncated octahedron', I think |
| 04:02.51 | gremzoid | codefreeze, one of my uncles is a smart cookie (works as an engineer), we buy him puzzles like that every xmas... he solves them within hours or days... |
| 04:03.21 | gremzoid | nothing we've got him yet has stumped him... i might try the one your talking about this xmas :P |
| 04:04.21 | codefreeze | I googled for it... see http://www.twistypuzzles.com/cgi-bin/puzzle.cgi?pid=673 |
| 04:05.25 | gremzoid | nar we got him that once... |
| 04:05.29 | gremzoid | it took him half an hour |
| 04:05.52 | codefreeze | Oh, that makes me feel good. |
| 04:06.24 | gremzoid | don't worry... i'm terrible at such puzzles |
| 04:06.34 | codefreeze | Instead, **I** am going to write a program to find ALL the possible solutions. |
| 04:06.47 | gremzoid | thats cheating isn't it? |
| 04:06.50 | gremzoid | :P |
| 04:06.55 | codefreeze | Yep! |
| 04:14.14 | *** join/#asterisk hmm-home (n=hmm-home@24-119-176-74.cpe.cableone.net) |
| 04:17.14 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 04:17.19 | *** join/#asterisk AJaymn (n=mypocket@71-82-218-158.dhcp.mdsn.wi.charter.com) |
| 04:19.16 | AJaymn | anyone able to add field features to ARI? |
| 04:20.54 | *** join/#asterisk mltlnx (n=mltlnx@cpe-72-225-244-190.nyc.res.rr.com) |
| 04:22.37 | *** join/#asterisk aday (n=aday@61.29.15.163) |
| 04:23.00 | TrentCreek | groovy |
| 04:23.16 | TrentCreek | who here has used AstBill or any other open source billing? |
| 04:24.08 | Nugget | I thought that it was illegal to bill for calls on asterisk because of the GPL. |
| 04:24.54 | TrentCreek | nope |
| 04:25.13 | AJaymn | was that a serious question Nugget? |
| 04:25.43 | Defraz | hahahaha |
| 04:25.44 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 04:25.53 | AJaymn | or.. statement :P |
| 04:30.42 | TrentCreek | I'd like to license my code under the GPL, but I'd also like to make it clear that it can't be used for military and/or commercial uses. Can I do this? |
| 04:30.42 | TrentCreek | No, because those two goals contradict each other. The GNU GPL is designed specifically to prevent the addition of further restrictions. GPLv3 allows a very limited set of them, in section 7, but any other added restriction can be removed by the user. |
| 04:31.18 | TrentCreek | where there we have our answer from gnu.org |
| 04:31.40 | AJaymn | :) nice |
| 04:31.49 | Nugget | No, it's true. I GPL'd my code and then one day RMS showed up and explained that it meant I had to let him crash on my couch when he was in town. |
| 04:31.52 | Nugget | never again. |
| 04:32.19 | Nugget | Now I only use the mozilla license. |
| 04:32.41 | TrentCreek | "'RMS"? Really Mean Sucker? |
| 04:32.47 | AJaymn | lol |
| 04:33.15 | Nugget | the only downside to the mozilla license is that I randomly get drunken late night phone calls from jwz |
| 04:33.30 | wwalker | RMS won't stay at my house. we bathe |
| 04:33.50 | AJaymn | you guys are crazy lol |
| 04:33.57 | wwalker | g'night |
| 04:40.36 | *** join/#asterisk _pepo_ (n=Pepo@190.10.187.20) |
| 04:41.31 | gremzoid | TrentCreek, that would be a cool opensource license tho... one thats 'free' but cannot be used for military/war |
| 04:42.30 | _pepo_ | where do I can download a good book to learn about Asterisk, (context,extension, and so on)? |
| 04:43.49 | orakle | o'reilly has a wicked book on asterisk |
| 04:43.51 | orakle | it explains very well |
| 04:43.52 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 04:43.57 | orakle | not sure where you could download it though |
| 04:44.02 | gremzoid | ~book |
| 04:44.02 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf |
| 04:44.17 | gremzoid | updated last week to ver 2 or something |
| 04:44.22 | TrentCreek | AstBill anyone? |
| 04:44.24 | gremzoid | good place to get started |
| 04:45.03 | _pepo_ | thanks |
| 04:47.51 | *** join/#asterisk mltlnx (n=mltlnx@cpe-72-225-244-190.nyc.res.rr.com) |
| 04:50.59 | *** join/#asterisk marcan (i=1337@host214-134.cvd.fit.edu) |
| 04:51.52 | tzafrir_home | TrentCreek, that extra restriction you have in mind makes the software non-free. Period |
| 04:54.44 | tzafrir_home | "free for non-commercial use" is far cry for being free |
| 04:55.30 | tzafrir_home | It means it cannot be put on, e.g. most linux distributions. As htose are typically sold for profit occasionally (e.g: mail or der CDs) |
| 04:56.35 | gremzoid | just give it away! screw the GPL :P |
| 04:57.01 | tzafrir_home | He does not want to give it away. He wants to make it "free for non-commercial use" |
| 04:57.14 | gremzoid | tzafrir, i'm joking |
| 05:03.50 | TrentCreek | according to GPL it is free for commerical use |
| 05:13.36 | *** join/#asterisk bmg505 (n=leon@196.209.176.188) |
| 05:15.42 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 05:17.24 | *** join/#asterisk flying_Luck (n=melifaro@secured.by.ipfw.ru) |
| 05:17.48 | *** join/#asterisk JunK-Y (n=junky@modemcable183.17-83-70.mc.videotron.ca) |
| 05:18.00 | JunK-Y | any1 running 1.4.12.1 now? |
| 05:18.17 | TrentCreek | i think I am |
| 05:19.45 | JunK-Y | could you try automon and let me if you get a continue dtmf tone on the callee chan? |
| 05:19.56 | *** join/#asterisk munmun (n=mun_mun@202.122.127.252) |
| 05:20.17 | TrentCreek | i dont know what it is...I got my first IAX termination service and sip device connected today |
| 05:24.23 | *** join/#asterisk DaPrivateer (i=Privatee@crimson.66fruit.com) |
| 05:28.07 | *** join/#asterisk Strom_M (n=strom@208.127.172.112) |
| 05:28.35 | *** join/#asterisk Chris-NB (n=chris@ng1.kurtkrenn.com) |
| 05:31.59 | CCFL_Man2 | Strom_M: i finally got a 684A ringer |
| 05:32.08 | Strom_M | yay |
| 05:33.07 | TrentCreek | whats that? Translation: outdoor bell ;-) |
| 05:36.10 | *** join/#asterisk reaxion (n=reaxion@218.189.184.34) |
| 05:36.59 | reaxion | hi folks |
| 05:37.20 | reaxion | is there any way to make Asterisk loop an audio file that it's playing back? |
| 05:37.36 | tzafrir_home | Goto ? GotoIf ? |
| 05:37.49 | tzafrir_home | for a loop |
| 05:38.09 | reaxion | Ah, thanks. I'll dig through the function's info. :) |
| 05:38.24 | gremzoid | gotoif rocks! |
| 05:38.39 | reaxion | Cool. I bet it does exactly what I need. |
| 05:41.08 | tzafrir_home | Sure. If you're used to assmebly programming :-) |
| 05:43.59 | reaxion | What's an assmebly? |
| 05:44.09 | reaxion | :-{) |
| 05:49.58 | *** join/#asterisk munmun (n=mun_mun@202.122.127.252) |
| 05:50.43 | gremzoid | heh |
| 05:51.55 | TrentCreek | a language |
| 05:56.06 | *** join/#asterisk J4zen (n=Jvan4zen@a213-84-139-244.adsl.xs4all.nl) |
| 05:58.10 | *** join/#asterisk xma1 (n=xma2@adsl-70-233-17-150.dsl.chcgil.sbcglobal.net) |
| 05:58.17 | *** join/#asterisk JungleRob (n=junglero@user-24-214-39-183.knology.net) |
| 05:58.37 | xma1 | hello |
| 05:59.07 | JungleRob | hello |
| 05:59.13 | xma1 | I need some asterisk help, cannot make outbound calls |
| 05:59.17 | TrentCreek | jello |
| 05:59.21 | TrentCreek | we know |
| 05:59.24 | xma1 | Gravy with inbound calls |
| 05:59.33 | xma1 | opps |
| 05:59.38 | xma1 | the other way around |
| 06:00.02 | xma1 | anyone up to it? :_) |
| 06:00.14 | TrentCreek | what is the error/defect |
| 06:00.18 | *** part/#asterisk JungleRob (n=junglero@user-24-214-39-183.knology.net) |
| 06:00.47 | xma1 | chan_sip.c:13664 handle_request_invite: Call from '101' to extension '911' rejected because extension not found. |
| 06:03.09 | TrentCreek | did you make any changes? |
| 06:03.33 | xma1 | besides setting it up, all kinds |
| 06:03.52 | xma1 | i'm using a grandstream 4 port fxo |
| 06:04.43 | TrentCreek | if the extension not found then you should be looking in extensions.conf at that object |
| 06:05.01 | remmo | the universe is laughing at us all |
| 06:05.20 | *** join/#asterisk JungleRob (n=jungle@user-24-214-39-183.knology.net) |
| 06:05.38 | xma1 | i have them defined |
| 06:05.54 | xma1 | you wanna take lookie, maybe my syntax is afu |
| 06:06.37 | reaxion | xma, did you reload your extensions? |
| 06:06.45 | reaxion | in CLI, type extensions reload |
| 06:06.54 | xma1 | everytime i make a change i restart asterisk |
| 06:07.17 | reaxion | No need to do that, just go into asterisk using 'asterisk -r' to attach to a console |
| 06:07.47 | reaxion | Your 911 extension, isn't that a risky number? |
| 06:07.50 | xma1 | I run asterisk as a service |
| 06:07.55 | reaxion | Me too |
| 06:08.05 | reaxion | Asterisk has a console running all the time |
| 06:08.12 | xma1 | just for testing... i can't dial to my cellphone either |
| 06:08.17 | reaxion | no need to restart it every time as it will terminate all calls |
| 06:08.22 | *** join/#asterisk santiago (n=santiago@debian/developer/santiago) |
| 06:08.29 | reaxion | How are you dialing out? using the Dial() function? |
| 06:08.36 | xma1 | i dont have any calls running right now it's night time |
| 06:08.39 | reaxion | If so, can you paste your dial function line here? |
| 06:08.58 | xma1 | [to-pstn] |
| 06:08.58 | xma1 | exten => _9NXXXXXX,1,Dial(SIP/${EXTEN:1}@pstn-gateway,30) |
| 06:08.58 | xma1 | exten => _9NXXXXXX,2,Congestion |
| 06:08.58 | xma1 | exten => _91NXXNXXXXXX,1,Dial(sip/${EXTEN:1}@pstn-gateway,30) |
| 06:08.58 | xma1 | exten => _91NXXNXXXXXX,2,Congestion |
| 06:08.59 | xma1 | exten => 911,1,Dial(SIP/${EXTEN}@pstn-gateway,30) |
| 06:09.19 | reaxion | Is the pstn gateway a separate piece of hardware? |
| 06:09.41 | xma1 | yea a grandstream gxw4104 |
| 06:09.43 | *** join/#asterisk tengulre (n=tengulre@222.90.66.10) |
| 06:09.48 | *** join/#asterisk chendy (n=chendy@121.76.132.123) |
| 06:11.08 | *** part/#asterisk JungleRob (n=jungle@user-24-214-39-183.knology.net) |
| 06:11.12 | reaxion | can you show me the line for extension 101? |
| 06:11.14 | idespinner | why does my Asterisk CLI disconnect when i do a reload on Asterisk? any ideas? |
| 06:11.28 | reaxion | How are you reloading idespinner ? |
| 06:11.32 | idespinner | reload |
| 06:11.44 | xma1 | exten => 101,1,Dial(SIP/101,20) |
| 06:11.44 | xma1 | exten => 101,2,Voicemail(u101) |
| 06:11.44 | xma1 | exten => 101,102,Voicemail(b101) |
| 06:11.44 | xma1 | exten => 101,103,Hangup |
| 06:11.52 | reaxion | Are you seeing any errors in /var/log/asterisk/messages? |
| 06:12.15 | idespinner | no errors that I know of, would that cause it? asterisk runs fine |
| 06:13.00 | idespinner | just a minor annoyance. It doesnt occur on other installations |
| 06:13.35 | reaxion | xma, what happens if you try to dial 911 from the asterisk console? |
| 06:13.52 | xma1 | how do i do that? |
| 06:14.20 | reaxion | use the terminal on your asterisk server, and type: asterisk -r |
| 06:14.22 | reaxion | then |
| 06:14.23 | idespinner | wow, thats a cool tip reaxion |
| 06:14.35 | xma1 | i'm using 1 stage dialing.. 911 is the number i'm actually trying to dial out |
| 06:14.37 | idespinner | never knew about that one |
| 06:14.42 | reaxion | What's that idespinner? This is for xma |
| 06:14.46 | idespinner | yea |
| 06:14.51 | reaxion | he didn't know |
| 06:14.52 | reaxion | so |
| 06:14.54 | reaxion | i told him |
| 06:14.56 | reaxion | ... |
| 06:14.57 | reaxion | ? |
| 06:14.58 | idespinner | but i never knew you could dial |
| 06:15.02 | reaxion | Ah :D |
| 06:15.06 | reaxion | Yep |
| 06:15.10 | reaxion | You can even add extensions |
| 06:15.15 | reaxion | and loads of other stuff |
| 06:15.16 | idespinner | from the CLI? |
| 06:15.19 | reaxion | without messing with configs |
| 06:15.19 | reaxion | yep |
| 06:15.32 | xma1 | so whats the command to dial from asterisk |
| 06:15.37 | idespinner | wow |
| 06:15.38 | idespinner | yea |
| 06:15.43 | *** part/#asterisk aday (n=aday@61.29.15.163) |
| 06:15.46 | xma1 | same as in extensions.conf? |
| 06:15.46 | idespinner | dial extension@context |
| 06:16.01 | reaxion | xma1: dial 911@context |
| 06:16.30 | idespinner | markster... |
| 06:16.37 | idespinner | if i dial markster, will he answer? |
| 06:17.20 | reaxion | xma1: the asterisk console is something you can run, then try a dial from your phone and watch what asterisk does |
| 06:17.46 | reaxion | to exit it use the quit command or ctrl-c |
| 06:18.10 | tzafrir_home | xma1, "originate" |
| 06:18.39 | reaxion | what's that, tzafrir? |
| 06:18.41 | tzafrir_home | There's also "dial" , but it uses the local sound card as a phone |
| 06:18.43 | idespinner | what is "originate" ? |
| 06:18.47 | xma1 | I think the 911 call went through .. OOPPS |
| 06:18.50 | xma1 | here is the output |
| 06:19.03 | xma1 | localhost*CLI> dial 911@to-pstn |
| 06:19.03 | xma1 | The 'dial' command is deprecated and will be removed in a future release. Please use 'console dial' instead. |
| 06:19.03 | xma1 | <PROTECTED> |
| 06:19.03 | xma1 | <PROTECTED> |
| 06:19.03 | xma1 | <PROTECTED> |
| 06:19.03 | xma1 | <PROTECTED> |
| 06:19.05 | xma1 | <PROTECTED> |
| 06:19.07 | xma1 | <PROTECTED> |
| 06:19.22 | JT | xma1: stop spamming the channel |
| 06:20.06 | JT | idespinner: what is the exact reload command? |
| 06:20.08 | idespinner | originate is asterisk 1.4 only i take it |
| 06:20.16 | idespinner | jt: reload |
| 06:20.20 | idespinner | asterisk 1.2 |
| 06:20.33 | tzafrir_home | right. For earlier versions I have a script to put a call file |
| 06:20.48 | JT | maybe it restarts asterisk |
| 06:20.54 | idespinner | it seems to be |
| 06:21.03 | idespinner | Disconnected from Asterisk server |
| 06:21.10 | idespinner | Executing last minute cleanups |
| 06:21.19 | idespinner | then linux console |
| 06:21.23 | reaxion | tzafrir, I'd like a link to that too :) |
| 06:23.14 | xma1 | hmm |
| 06:23.34 | tzafrir_home | I'll have to clean it up a bit |
| 06:23.47 | reaxion | xma, so that worked? |
| 06:23.51 | reaxion | It looks like it. |
| 06:24.12 | xma1 | i called my cell using dial from CLI and my cell didn't ring |
| 06:24.33 | reaxion | Probably better to use a test number other than emergency services |
| 06:24.42 | reaxion | what about a landline number? |
| 06:27.39 | xma1 | what is "answered OSS/dsp" |
| 06:28.38 | xma1 | it looks like asterisk dial using cli but the landline does not right |
| 06:28.41 | idespinner | it seems the command "dial" isnt in asterisk 1.4 |
| 06:28.57 | xma1 | it is, i'm using 1.4 |
| 06:29.06 | reaxion | console dial is |
| 06:29.14 | idespinner | ok |
| 06:29.26 | idespinner | it seems the command "dial" isnt in Asterisk Appliance(aa50) 1.4 |
| 06:29.56 | idespinner | i thought the aa50 was 1.4 but they must be different a little bit |
| 06:30.12 | reaxion | Yep, I think so.. try console dial though as that should work |
| 06:30.26 | idespinner | no luck |
| 06:30.31 | idespinner | no "console" |
| 06:30.57 | idespinner | i have originate |
| 06:31.04 | idespinner | is that the same? |
| 06:31.16 | reaxion | pretty much |
| 06:31.42 | reaxion | except for the reasons pointed out further up |
| 06:32.59 | TrentCreek | AstBill anyone? |
| 06:33.13 | reaxion | Is that an offer, TrentCreek? |
| 06:33.17 | idespinner | sure |
| 06:33.21 | idespinner | i'll take one |
| 06:33.25 | reaxion | me too |
| 06:33.50 | idespinner | originate seems to do extension to extension dialing |
| 06:35.19 | TrentCreek | yeah..if anyone can get it workign with current v of Asterisk |
| 06:35.23 | reaxion | Can originate between a channel and a specific application or extension in the dialplan |
| 06:35.57 | idespinner | with only 1 extension on this system, dial would be the best tool |
| 06:36.18 | idespinner | I dont think I can use originate |
| 06:36.46 | reaxion | I think dial is fine, and console dial on 1.4 or the appliance |
| 06:37.12 | idespinner | yea, there is no "console" or "dial" |
| 06:37.21 | idespinner | i have a putty window open atm |
| 06:37.43 | idespinner | atleast not on the appliance |
| 06:38.18 | reaxion | Hmm, strange. |
| 06:39.24 | *** join/#asterisk gardo (n=gardo@121.97.210.126) |
| 06:39.25 | reaxion | you can, of course put a .call file in the outgoing dir and see if that outdials? |
| 06:39.48 | idespinner | not a bad idea |
| 06:40.30 | *** join/#asterisk syneus (n=syneus@syneus.aemcom.net) |
| 06:40.43 | idespinner | the folder is there, so it should work |
| 06:40.45 | reaxion | in /var/spool/asterisk/outgoing |
| 06:40.55 | idespinner | yea |
| 06:40.55 | reaxion | Cool, yep |
| 06:43.42 | idespinner | well, im off to bed |
| 06:43.47 | idespinner | thanks for the help |
| 06:44.46 | reaxion | no probs |
| 06:44.48 | reaxion | :) |
| 06:44.59 | TrentCreek | now what was that about fixing AstBill? ;-) |
| 06:45.04 | reaxion | heh |
| 06:45.11 | TrentCreek | SPAMMER |
| 06:47.50 | reaxion | Can you stop slapping your winkie in the channel please, TrentCreek? |
| 06:49.39 | TrentCreek | heh |
| 06:50.03 | reaxion | I think the problem with 1.4 is that it's such a major rejigging of stuff that most things will need some rewriting to work with it again |
| 06:50.37 | TrentCreek | looking at the install script, it is looking for directories that do not exist |
| 06:50.56 | TrentCreek | and errors |
| 06:51.13 | reaxion | I see your post on the forums |
| 06:51.47 | reaxion | Assuming you're the same Trent :) |
| 06:51.47 | TrentCreek | oh you ar elooking |
| 06:51.52 | TrentCreek | yeppers |
| 06:51.53 | reaxion | Oh yes |
| 06:52.12 | reaxion | Let me just have a peek at the install script. |
| 06:52.31 | TrentCreek | i would hate to change the script and the application not work |
| 06:52.42 | TrentCreek | let me post the errors |
| 06:52.48 | reaxion | Yeah, it's quite likely that'd happen |
| 06:52.59 | reaxion | Send them over in a PM please |
| 06:57.31 | *** join/#asterisk dominic1 (n=dob@213.221.82.242) |
| 06:59.39 | *** part/#asterisk dominic1 (n=dob@213.221.82.242) |
| 07:00.46 | *** join/#asterisk manfish (n=manfish@82-70-235-156.dsl.in-addr.zen.co.uk) |
| 07:01.00 | manfish | good morining |
| 07:01.53 | reaxion | Good moaning to you too |
| 07:02.34 | manfish | anyone have a bit of time to help with a TE120p card? |
| 07:03.10 | reaxion | What's up, manfish |
| 07:03.11 | reaxion | ? |
| 07:03.12 | manfish | just started to get some problems with Primary D-Channel on span 1 down |
| 07:03.25 | manfish | and it started cutting off calls |
| 07:04.17 | manfish | I have attempted to reinstall the zaptel drivers etc |
| 07:04.39 | reaxion | Is your span a T1/E1/fraction? |
| 07:04.46 | manfish | E1 |
| 07:05.50 | manfish | system is CentOS 4.4 kernel 2.6.9-55.0.9.ELsmp, latest versions of asterisk, zaptel |
| 07:06.04 | manfish | I am also running freepbx |
| 07:06.18 | reaxion | can you pm me the output of: cat /proc/zaptel/1 |
| 07:07.07 | *** join/#asterisk [hC] (n=hardcore@S01060016b61c8983.vf.shawcable.net) |
| 07:07.21 | reaxion | and also the output of: [14:55] <TrentCreek> cp: cannot stat `etc/asterisk1.2.6.sample/*': No such file or directory |
| 07:07.22 | reaxion | [14:55] <TrentCreek> cp: cannot stat `etc/asterisk/*': No such file or directory |
| 07:07.22 | reaxion | [14:55] <TrentCreek> cp: cannot stat `themes/*': No such file or directory |
| 07:07.22 | reaxion | [14:55] <TrentCreek> cp: cannot stat `images/*': No such file or directory |
| 07:07.22 | reaxion | [14:55] <TrentCreek> cp: cannot stat `modules/astbill': No such file or directory |
| 07:07.22 | manfish | http://pastebin.com/m2e5261df |
| 07:07.23 | reaxion | [14:55] <TrentCreek> cp: cannot stat `ajax/*.php': No such file or directory |
| 07:07.25 | reaxion | [14:55] <TrentCreek> chmod: cannot access `scripts/backupastbill.sh': No such file or directory |
| 07:07.27 | *** join/#asterisk Nic-I (n=monster@bb220-255-41-253.singnet.com.sg) |
| 07:07.27 | reaxion | Gah. |
| 07:07.29 | reaxion | Sorry |
| 07:07.31 | TrentCreek | hehe |
| 07:07.33 | reaxion | Wrong past :) |
| 07:07.41 | TrentCreek | correct future |
| 07:07.51 | reaxion | handy site that, manfish |
| 07:08.13 | Nic-I | hi .. can someone recommend me a AT 775 P4 mainboard? |
| 07:08.14 | TrentCreek | any relation to mancow? |
| 07:08.16 | reaxion | manfish, you're getting IRQ misses |
| 07:08.46 | *** join/#asterisk [hC] (n=hardcore@S01060016b61c8983.vf.shawcable.net) |
| 07:08.58 | TrentCreek | ECS from Frys.. > $100 with CPU |
| 07:09.25 | Nic-I | built-in LAN card keep having conflict with the digium card |
| 07:09.37 | reaxion | manfish, are you experiencing any audio issues with the calls? |
| 07:10.00 | TrentCreek | ohhh...I do recall the Astrisk book to get a board without stuff built in |
| 07:10.40 | TrentCreek | but a little hard to do now days |
| 07:10.41 | Nic-I | TrentCreek but the problem, i disabled the builtin and replace with a pci-x, it still conflict |
| 07:10.49 | manfish | sorry phone |
| 07:10.58 | reaxion | k |
| 07:11.01 | manfish | no audio issues |
| 07:11.02 | TrentCreek | did you disable then install? |
| 07:11.07 | TrentCreek | or install then disable? |
| 07:11.12 | Nic-I | TrentCreek i did that |
| 07:12.06 | TrentCreek | then you better check the bios. |
| 07:12.21 | Nic-I | TrentCreek check what? |
| 07:12.25 | TrentCreek | though it is disabled, it may still have the IRQ reserved for it |
| 07:12.51 | Nic-I | TrentCreek this board dont have such a function |
| 07:13.05 | manfish | reaxion is there anything I can do to resolve the IRQ problems? |
| 07:13.13 | TrentCreek | better triple check |
| 07:13.15 | reaxion | Looking into it for you, hold a bit |
| 07:13.19 | manfish | thanks |
| 07:13.38 | Nic-I | TrentCreek i did more than triple |
| 07:13.43 | manfish | it only seems to have started happening after doing a update to latest kernel |
| 07:14.06 | manfish | I rolled back to a previous one but the issue remains |
| 07:14.06 | orakle | hey |
| 07:14.08 | Nic-I | i will wanna dump this lousy asus board for another.. what is a good board? |
| 07:14.10 | orakle | anyone using chan_cellphone? |
| 07:14.11 | reaxion | Yeah it may be because your new kernel is assigning shared IRQs |
| 07:14.17 | orakle | i'm trying to get it working, but it thinks my phone is a headset |
| 07:14.26 | orakle | i can't figure out how to get my phone paired.. |
| 07:14.43 | reaxion | manfish: Try this linky out. http://www.voip-info.org/wiki/view/Asterisk+PCI+bus+Troubleshooting |
| 07:15.02 | reaxion | A very good page on fixing misses that |
| 07:15.25 | manfish | thanks will take a look |
| 07:15.29 | reaxion | Let me know how you get on. |
| 07:19.13 | manfish | strnge it is sharing the IRQ with the HP ILO port, I am sure I set it in the BIOS not to. |
| 07:19.32 | reaxion | Very interesting. That is quite likely the cause of your problem |
| 07:19.58 | reaxion | Are you able to reassign the IRQ of the slot? |
| 07:20.18 | reaxion | and also, what's your zttest output? |
| 07:20.28 | manfish | need to reboot and take a look at the bios |
| 07:20.36 | manfish | I am sure I should be able to |
| 07:20.50 | reaxion | Okay. Might be worth a try. |
| 07:20.55 | manfish | its a HP DL360 G4 server |
| 07:21.13 | reaxion | The thing is that Linux ignores some bios settings |
| 07:21.24 | sheppard | some? |
| 07:21.27 | sheppard | all. |
| 07:21.28 | reaxion | :D |
| 07:21.52 | jql | hah |
| 07:21.54 | manfish | is there any way of reventing IRQ sharing in linux? |
| 07:23.01 | jql | there's a problem... irqset or something |
| 07:23.03 | jql | program |
| 07:23.16 | jql | dunno what it's really called or where to get it |
| 07:24.49 | reaxion | It seems Linux respects the IRQs set in the bios unless they are overidden by kernel driver switches |
| 07:25.01 | sheppard | yep |
| 07:25.21 | reaxion | So Linux doesn't ignore the bios then, does it? :P |
| 07:25.36 | sheppard | it respects IRQ settings |
| 07:25.40 | sheppard | but that's about it |
| 07:25.43 | reaxion | Just make sure your BIOS doesn't have "Plug and Play OS" set |
| 07:25.46 | sheppard | everything else it probes itself |
| 07:26.05 | reaxion | Then your IRQs should be fixed, unless Linux has a good root around |
| 07:26.14 | sheppard | and hopefully the bios supports setting specific irq's per port |
| 07:26.22 | reaxion | yep |
| 07:26.25 | reaxion | HP should be ok |
| 07:26.31 | reaxion | but you never know. dell suck at that |
| 07:26.41 | sheppard | dell fails at just about everything |
| 07:26.42 | manfish | ztteest result http://pastebin.com/m71fdf380 |
| 07:26.47 | *** join/#asterisk nfi|ermes (n=ErMeS@217.220.121.62) |
| 07:27.01 | Nic-I | i am having that IRQ conflict.. that's why i wanna change another board |
| 07:27.13 | reaxion | manfish, that looks fine |
| 07:27.18 | manfish | ok |
| 07:27.42 | manfish | I might just disable the ILO card as I am not using it anyway |
| 07:28.00 | reaxion | Yeah, or take it out completely |
| 07:28.14 | manfish | need a soldering iron for that one :-) |
| 07:28.19 | manfish | its on the mobo |
| 07:28.24 | reaxion | :D |
| 07:28.28 | reaxion | Or a hammer |
| 07:28.33 | reaxion | My preferred choice |
| 07:28.38 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 07:28.41 | manfish | ok be back in 10 thanks |
| 07:28.46 | reaxion | no probs |
| 07:38.09 | J4zen | Does anyone use a QuadBRI PCI card in here? |
| 07:38.19 | *** join/#asterisk manfish (n=manfish@82-70-235-156.dsl.in-addr.zen.co.uk) |
| 07:38.26 | *** join/#asterisk yassaccan (n=yassacca@admin189.hgo.se) |
| 07:39.04 | *** join/#asterisk qdk (n=qdk@85.235.253.139) |
| 07:39.40 | manfish | still saying I am getting IRQ misses |
| 07:39.40 | J4zen | or any Junghanns hardware/Bristuff driver operated hardware? |
| 07:40.10 | reaxion | Are you still seeing an IRQ conflict? |
| 07:40.38 | manfish | no |
| 07:40.45 | manfish | it is on its own IRQ |
| 07:41.10 | manfish | I just changed the setting for the ILO card and changed it so the card is on IRQ 7 |
| 07:42.04 | *** join/#asterisk patrickv0x (n=patrick@64.235.249.36) |
| 07:42.07 | patrickv0x | Has anyone been able to register a Cisco 7970 with Asterisk ? If so, could you share your config file with me please? Thanks. |
| 07:42.15 | *** join/#asterisk BBHoss (n=hoss@146.229.178.140) |
| 07:42.26 | BBHoss | hey guys anybody alive right now? |
| 07:42.39 | TrentCreek | no |
| 07:42.43 | BBHoss | good |
| 07:42.43 | patrickv0x | hey BBHoss |
| 07:42.48 | TrentCreek | still trying to get AstBill to function |
| 07:43.15 | BBHoss | how would i get asterisk to accept an incoming call on IAX |
| 07:43.26 | BBHoss | like the digium demo does |
| 07:43.43 | BBHoss | im getting a no authority found atm |
| 07:43.46 | reaxion | we're all dead BBHoss |
| 07:43.48 | BBHoss | but thats expected |
| 07:46.22 | TrentCreek | there is a video I saw |
| 07:46.23 | BBHoss | anyone from the huntsville area on here? |
| 07:46.30 | TrentCreek | AL or TX |
| 07:46.46 | BBHoss | AL |
| 07:47.09 | TrentCreek | This video shows you http://revision3.com/systm/asterisk/ |
| 07:47.13 | *** join/#asterisk tcseke (n=chatzill@22-36.adsl.etel.hu) |
| 07:47.19 | TrentCreek | you can downlod it for better view |
| 07:47.37 | TrentCreek | More problems with AstBill..dammit |
| 07:47.42 | *** join/#asterisk xheliox (n=jeff@pdpc/supporter/active/xheliox) |
| 07:47.58 | BBHoss | that will allow me to feed a simple dial string into asterisk and it connect on the other end? |
| 07:49.50 | TrentCreek | it will show you how |
| 07:51.14 | *** part/#asterisk arekm (i=arekm@pld-linux/arekm) |
| 07:54.42 | TrentCreek | Well craps |
| 07:55.04 | patrickv0x | Has anyone been able to register a Cisco 7970 with Asterisk ? If so, could you share your config file with me please? Thanks. |
| 07:55.21 | orakle | I got a 7940 working if that would help you |
| 07:55.26 | orakle | are you trying with SIP or SCCP? |
| 07:55.32 | patrickv0x | using sip |
| 07:55.36 | patrickv0x | 7940/7960 is easy |
| 07:55.39 | patrickv0x | 7970 is driving me nuts |
| 07:55.42 | orakle | oh |
| 07:55.49 | orakle | yeah 7960's the same as the 40 |
| 07:55.54 | orakle | what's different about the 70? |
| 07:56.01 | patrickv0x | 7970 is really dumb |
| 07:56.46 | ApEtc | [A |
| 08:01.34 | *** join/#asterisk michael-i (n=michael-@141.41.40.186) |
| 08:05.14 | *** join/#asterisk KermitTheFragger (n=siepkes@118-197.bbned.dsl.internl.net) |
| 08:07.06 | manfish | reaxion thanks for your help I will monitor the server this morning but at the moment it appears more stable |
| 08:08.47 | manfish | I just need to stop being lazy now and work out how to configre an asterisk server without the aid of freepbx :-) |
| 08:11.05 | manfish | I spoke too soon |
| 08:11.28 | manfish | <PROTECTED> |
| 08:11.32 | TrentCreek | tell your brother mancow hello |
| 08:11.42 | manfish | lol |
| 08:13.11 | *** join/#asterisk vgster (n=vgster@host81-149-46-66.in-addr.btopenworld.com) |
| 08:13.40 | manfish | hi vgster |
| 08:13.54 | vgster | hi |
| 08:14.23 | manfish | hows life and all that stuff |
| 08:14.52 | vgster | average |
| 08:20.24 | TrentCreek | Ast Bill |
| 08:20.26 | TrentCreek | AstBill |
| 08:22.34 | *** join/#asterisk Poincare (n=jefffnod@amp89.ampersant.be) |
| 08:22.55 | TrentCreek | AstBill AstBill |
| 08:27.11 | *** join/#asterisk oej (n=olle@soll4-125.cust.blixtvik.net) |
| 08:27.59 | TrentCreek | AstBill AstBill |
| 08:28.16 | *** join/#asterisk PepOSX (n=pepOSX@190.72.151.217) |
| 08:28.56 | *** join/#asterisk michael-i (n=michael-@141.41.40.186) |
| 08:33.30 | *** join/#asterisk phearless (n=phear@host217-34-75-65.in-addr.btopenworld.com) |
| 08:43.15 | *** join/#asterisk blq (n=Bl@dslb-088-064-148-094.pools.arcor-ip.net) |
| 08:45.08 | *** join/#asterisk ming_zym (i=ming_zym@nat/yahoo/x-912ff277bb3ea2a1) |
| 08:48.34 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 08:49.46 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 08:54.30 | *** join/#asterisk IvanV3835 (n=Miranda@styx.mcn.ru) |
| 08:59.45 | TrentCreek | mp[e |
| 09:17.11 | *** part/#asterisk BBHoss (n=hoss@146.229.178.140) |
| 09:23.13 | *** join/#asterisk Chris-NB (n=chris@ng1.kurtkrenn.com) |
| 09:24.01 | *** join/#asterisk ming_zym (i=ming_zym@nat/yahoo/x-f286570ccc59760e) |
| 09:25.00 | *** join/#asterisk ghenry (n=ghenry@suretecsystems.plus.com) |
| 09:25.12 | *** part/#asterisk ming_zym (i=ming_zym@nat/yahoo/x-f286570ccc59760e) |
| 09:25.38 | *** join/#asterisk ming_zym (i=ming_zym@nat/yahoo/x-b94ca84674d14d1f) |
| 09:30.15 | *** join/#asterisk stkn|work (n=stkn|wor@gentoo/developer/pdpc.active.stkn) |
| 09:37.45 | *** join/#asterisk Schumie (n=Steve@cmarfw01.marlow.spinvox.com) |
| 09:49.59 | Nic-I | hmmm .. my system does not detect the digium card |
| 09:52.09 | *** join/#asterisk IgorG (n=FeedomPa@195.162.32.126) |
| 09:53.00 | xma1 | it just doesn't wanna ring |
| 09:56.43 | tzafrir | Nic-I, "does not detect" as in? |
| 09:57.03 | tzafrir | Do you see the card on lspci? |
| 09:57.20 | *** join/#asterisk epasqu (n=pasqu@host107-107-static.184-82-b.business.telecomitalia.it) |
| 09:58.07 | epasqu | hi all |
| 09:58.28 | epasqu | i have i little question |
| 09:58.46 | epasqu | is normal that one tdm400 with 2 fxs modules |
| 09:58.54 | epasqu | when go in offhook |
| 09:59.07 | epasqu | put my cpu @100% |
| 10:03.42 | Nic-I | tzafrir this is a centos 5 box and it isnt detecting the digium card |
| 10:09.54 | *** join/#asterisk famicon (n=redz@c51447ddc.cable.wanadoo.nl) |
| 10:11.14 | J4zen | Im running into an installation issue when using the BRistuff drivers for my Junghanns QuadBRI PCI card, it all downloads.. installs fine up untill its off to compile ztgsm libraries |
| 10:11.18 | J4zen | it then exists with error code 1 |
| 10:11.29 | J4zen | i remember it was a matter of commenting out one or two lines in a file |
| 10:11.30 | J4zen | but i forgot |
| 10:11.35 | J4zen | http://pastebin.com/d498025d9 for the debug |
| 10:11.41 | J4zen | Any help would be appreciated :) |
| 10:13.34 | Nic-I | how do i see is the card is installed? |
| 10:28.19 | *** join/#asterisk creativx (n=creadure@226.62-97-205.bkkb.no) |
| 10:33.54 | gremzoid | what kind of configuration setting would throw up and error like: chan_iax2.c:5160 register_verify: No registration for peer 'user' (from IPaddr) ? |
| 10:38.41 | J4zen|afk | Simply not having a sip peer registered under that IP/User |
| 10:42.37 | *** join/#asterisk santibiotico (n=santi@ip23498.bcn.altecom.net) |
| 10:42.39 | santibiotico | hi |
| 10:42.55 | gremzoid | weird |
| 10:43.00 | gremzoid | i get that error... however my iax softphone can still make calls |
| 10:43.18 | gremzoid | the softphone thinks it's offline/logged out as well... |
| 10:44.06 | santibiotico | is there any way to provide a dial tone when waiting in waitexten() |
| 10:44.33 | santibiotico | i want the user to dial i.e. '0' and then to be provided a dial tone to dial another no. |
| 10:45.57 | J4zen | Google won't let you down on that one =\ its one of the most common examples |
| 10:46.12 | jql | playtones(dial) or some such |
| 10:46.25 | *** join/#asterisk shtoom (n=shtoom@59.93.112.135) |
| 10:46.57 | J4zen | exten => 0,1,Playback(start-dial-after-dialtone) |
| 10:46.58 | J4zen | exten => 0,2,Dial(SIP/Telstra/{EXTEN}) ; Telstra being the name of the SPA3K trunk. |
| 10:46.58 | J4zen | exten => 0,3,Hangup |
| 10:46.58 | J4zen | Every time when I dial ‘0’, the SPA3K will send it to Gateway 0, which is the PSTN line and present me with a second dial tone. |
| 10:47.02 | J4zen | Excuse my spam. |
| 10:47.24 | shtoom | Hi I am trying our external IVR i've sent a line "A,/var/lib/asterisk/sounds/vm-goodbye\n" on stdout |
| 10:47.29 | shtoom | but nothing happens |
| 10:47.33 | gremzoid | ewww telstra |
| 10:47.36 | sysreq | santibiotico: check out the DISA application, http://www.voip-info.org/wiki-Asterisk+cmd+DISA |
| 10:47.44 | michael-i | missing a "$" there aren't you? |
| 10:48.20 | J4zen | Who cares what he uses, its an accurate example for your own implementation |
| 10:48.31 | shtoom | is there any other specific protocol that should be followed apart from the one delivered with asterisk source for app_externalivr? |
| 10:50.19 | *** join/#asterisk s0ck (n=m@unaffiliated/s0ck) |
| 10:50.28 | santibiotico | J4zen: i don't mean what you said |
| 10:50.43 | santibiotico | J4zen: i don't want an external line nor an external device to provide me a 2nd dial tone |
| 10:52.21 | tzafrir | Nic-I, please pastebin the output of lspci |
| 10:52.42 | santibiotico | J4zen: playtones(dial) is enough |
| 10:52.56 | sysreq | santibiotico: did you check out the DISA() app? |
| 10:53.08 | J4zen | oh you just want to play a tone in any given situation? |
| 10:53.13 | tzafrir | J4zen, you don't need libgsmat for your quadbri card. Simply skip it |
| 10:53.22 | J4zen | So i noticed :) Thanks |
| 10:53.41 | santibiotico | sysreq: yes, but it's enough with playtones(dial). thx anyway |
| 10:53.47 | sysreq | np |
| 10:53.57 | tzafrir | I hope to soon upload a new version to http://updates.xorcom.com/astribank/bristuff that fixes this stupidity, anyway |
| 11:05.41 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 11:09.16 | *** join/#asterisk Schumie (n=Steve@cmarfw01.marlow.spinvox.com) |
| 11:10.04 | *** join/#asterisk MicW (n=michael@dslb-088-074-135-158.pools.arcor-ip.net) |
| 11:10.05 | MicW | hi |
| 11:10.46 | MicW | i cannot register my new asterisk to my provider. when i debug sip, i get "SIP/2.0 400 Missing CSeq Header" as response |
| 11:11.20 | MicW | the sent request contains "CSeq: 109 REGISTER" |
| 11:16.26 | J4zen | Does anyone know what this indicates, its being debugged in CLI on high verbosity whenever i push a button on my SNOM 320 ( ip _.57 ) |
| 11:16.26 | J4zen | chan_sip.c:11784 handle_request: Unknown SIP command 'PUBLISH' from '192.168.1.57' |
| 11:29.36 | *** join/#asterisk Zeeek (n=Heh@pdpc/supporter/active/Zeeek) |
| 11:34.05 | *** join/#asterisk The_Ball (n=The_Ball@122-148-187-181.static.dsl.dodo.com.au) |
| 11:34.19 | *** join/#asterisk pfn (n=pfnguyen@hanhuy.com) |
| 11:41.36 | *** join/#asterisk dominic1 (n=dob@213.221.82.242) |
| 11:41.37 | *** part/#asterisk dominic1 (n=dob@213.221.82.242) |
| 11:43.53 | *** join/#asterisk michael-i (n=michael-@141.41.40.186) |
| 11:48.46 | *** join/#asterisk elixer (i=elixer@65.207.74.18) |
| 11:49.01 | xma1 | hey there |
| 11:49.13 | Zeeek | Yo! |
| 11:49.39 | xma1 | when i dial a number via CLI it strips the prefix 1( based on my dialplan i assume) |
| 11:50.22 | xma1 | exten => _1NXXNXXXXXX,1,Dial(sip/201/${EXTEN:1},30) |
| 11:50.36 | xma1 | what can i do so it wont stip the 1 away |
| 11:51.18 | Zeeek | remove the :1 which tells it to stip the first digit |
| 11:51.50 | The_Ball | Hi, I have just setup "originate" outbound calling with vTiger crm. This works very well, but we use one normal IP phone and one softphone per employee, so that if we are using a laptop with vpn we can still use voip. It seems that the management command "originate" can't accept a extention like: "SIP/83&IAX2/alex" it can only accept single extentions, is there a way around this? |
| 11:53.40 | xma1 | that worked!! |
| 11:54.00 | xma1 | what a dumb ass example by grandstream |
| 11:54.42 | Zeeek | of course it worked. This is how asterisk works |
| 11:54.44 | *** join/#asterisk Mr_Varmit (n=varmit@rrcs-64-183-163-178.west.biz.rr.com) |
| 11:55.02 | Zeeek | read up about the dialplan |
| 11:56.10 | xma1 | yea yea.. second day with only |
| 11:56.17 | xma1 | a first all nighter |
| 11:56.29 | xma1 | gotta call in sick to work |
| 11:56.41 | Zeeek | well, free advice: the absolute first thing to do is learn what the dialplan is and how it works. THe rest is easy |
| 11:57.50 | xma1 | OMG.. for the first time in 2 days i was able to call myself |
| 11:58.07 | Zeeek | handy, that |
| 11:58.14 | elixer | probably a boring conversation though... |
| 11:58.40 | Mr_Varmit | i got a problem on the Voice Menus, the call answears...plays my greeting...i have no clue how to insert the KEYPRESS command on asterisknow |
| 11:59.11 | xma1 | it went like this: "Hello, Hello" |
| 11:59.14 | Mr_Varmit | elixer needs a xylophone |
| 11:59.36 | elixer | do i? |
| 11:59.44 | Mr_Varmit | lolol |
| 12:00.17 | Zeeek | xma1 you have to learn to say "tick tick tick" to check the lag and echo |
| 12:00.27 | Mr_Varmit | how do i get the keypress command in the voice menus, anyone help me on this |
| 12:00.44 | Zeeek | keypress command? |
| 12:00.52 | Mr_Varmit | yes |
| 12:00.56 | Mr_Varmit | wait for keypress |
| 12:01.19 | Zeeek | look for "background" in the literature |
| 12:01.20 | xma1 | let me call myself again |
| 12:01.35 | Zeeek | show applications background |
| 12:01.57 | Mr_Varmit | right now...i have answear the call Play my custom greeting wait 5 secs.... |
| 12:02.02 | Mr_Varmit | hmmm |
| 12:02.06 | Mr_Varmit | applications? |
| 12:02.08 | Mr_Varmit | ill check that |
| 12:02.16 | Zeeek | Mr_Varmit ~google asterisk background |
| 12:02.32 | *** join/#asterisk Alex_WinKiller (n=nowhere@mgm.cornet.am) |
| 12:02.39 | Zeeek | which will return something like this: http://www.voip-info.org/wiki-Asterisk+cmd+Background |
| 12:02.56 | Alex_WinKiller | hi guys, can someone help me with quintum ? pls ? |
| 12:03.03 | xma1 | i didn't really year any echo's |
| 12:03.41 | Mr_Varmit | Zeeek i called my phone..and i press 1 nothign happens...does the cmd background help me with that problem? |
| 12:04.09 | Zeeek | the civil answer to that is read the fscking page I showed you and see |
| 12:05.04 | Mr_Varmit | Zeeek im aware of the background command.....i select it on the drop down...on the right where i ADD, it doesnt say anything about wait for keypress |
| 12:05.21 | Zeeek | Mr_Varmit what drop down? |
| 12:05.46 | Mr_Varmit | im using the asterisknow gui |
| 12:05.56 | Zeeek | In that case, read the topic |
| 12:06.25 | Zeeek | I personally don't know about AsteriskNOW |
| 12:06.36 | Zeeek | but for help on it, there are specific places to ask |
| 12:06.39 | Mr_Varmit | ok..thanks for replying... |
| 12:06.46 | Zeeek | no problem at all |
| 12:06.57 | Mr_Varmit | i do appreciate the help....its just driving me nuts |
| 12:07.02 | Mr_Varmit | lol |
| 12:07.04 | Zeeek | background is the correct answer |
| 12:07.12 | Mr_Varmit | thanks.... |
| 12:07.16 | Zeeek | now go find out how to use it! :) |
| 12:07.40 | Zeeek | ~Zeeek |
| 12:07.40 | jbot | [zeeek] someone who once said "learning asterisk using a GUI is like learning sex through masturbation. You'll never get to the good stuff." |
| 12:07.53 | Zeeek | Proven true once again |
| 12:08.37 | Mr_Varmit | lolol |
| 12:08.51 | Mr_Varmit | it depends on what she looks like while shes doign it |
| 12:08.57 | xma1 | how often is does it happen that dialing a cellphone doesn't work, while dialing a land line works? |
| 12:09.07 | Zeeek | often |
| 12:09.16 | Zeeek | because the signaling is so different |
| 12:09.32 | xma1 | is there a work around |
| 12:09.41 | Zeeek | when you call a cellphone, asterisk thinks it has answered as soon as it starts to ring |
| 12:09.43 | *** join/#asterisk michael-i (n=michael-@141.41.40.186) |
| 12:09.59 | Zeeek | work around is don't call cellphones ;) |
| 12:10.17 | xma1 | i don't have a house phone |
| 12:10.23 | xma1 | or a landline |
| 12:10.54 | xma1 | is somekind of a delay needed or something |
| 12:10.57 | Zeeek | call a payphone |
| 12:11.12 | Zeeek | what happens when you call the cell number? |
| 12:11.53 | xma1 | executes a new stack |
| 12:11.56 | xma1 | says it calls |
| 12:12.02 | xma1 | then this: Remote host can't match request CANCEL to call |
| 12:12.16 | Zeeek | bad. very bad. |
| 12:12.26 | xma1 | wait |
| 12:12.33 | Zeeek | deadly in fact |
| 12:12.53 | xma1 | there is a nobody picked up in 3000ms (my ring duration) |
| 12:13.11 | Zeeek | you gave it 3 seconds to pick up? |
| 12:13.15 | xma1 | 30,000 |
| 12:13.20 | xma1 | 30 seconds |
| 12:13.33 | Zeeek | and it gives up before the 30 secs? |
| 12:13.43 | Zeeek | and you never hear a ring? |
| 12:14.01 | xma1 | do not hear a ring.. |
| 12:14.11 | Zeeek | on your cell I mean |
| 12:14.59 | xma1 | no cell phone right |
| 12:15.45 | xma1 | yes about 30 seconds (counting the way they teach in drivers ed) |
| 12:16.06 | Zeeek | one one thousand two one thousand three one thousand... |
| 12:16.09 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 12:16.19 | xma1 | and i have to console hangup after this failed attempt |
| 12:16.44 | Zeeek | what cell provider is this and what country? |
| 12:17.13 | xma1 | Nextel(USA) |
| 12:17.18 | awk | ~pb |
| 12:17.19 | jbot | A Pastebin is a place to paste your stuff without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 12:17.47 | Zeeek | I see no reason why it shouldn't work. WHat callerID are you sending and how are you calling? (SIP, ZAP?) |
| 12:18.28 | awk | arg, some issues here in digium code, http://paste.debian.net/39088 |
| 12:18.31 | Zeeek | Damn, I thought a pâstebin was a place to keep paste |
| 12:18.36 | awk | really straining the box, anyone had these issues |
| 12:18.39 | Mr_Varmit | Zeeek, thanks....your link helped me.....i got it working.....all i have to figure out now is the how to make it where i can dial more than 7 numbers....in my dialplan |
| 12:18.53 | xma1 | sip (asterisk > grandstream fxo gateway ) |
| 12:19.03 | Zeeek | Mr_Varmit keep reading about dialplans and you'll find out soon enough |
| 12:19.28 | Zeeek | xma1 where is the fxo connected? |
| 12:19.51 | xma1 | BTW internal timing is only needed in internal FXO/FXS devices? |
| 12:20.02 | xma1 | fxo connected to the phone lines |
| 12:20.06 | Zeeek | yes I'd say so |
| 12:20.20 | Zeeek | so, connect a normal phone and call that number and see what happens |
| 12:20.23 | xma1 | the dummy timing driver is what i meant |
| 12:20.30 | Zeeek | you don't need it |
| 12:21.51 | ThoMe | hello |
| 12:21.57 | ThoMe | do u speak german? |
| 12:21.57 | Zeeek | hallo |
| 12:21.59 | ThoMe | a little bit? |
| 12:22.19 | Zeeek | nein mein Here |
| 12:22.24 | Zeeek | as you see |
| 12:22.26 | thewiizle | how can i set a variable from an AGI script? |
| 12:22.30 | thewiizle | do i echo it |
| 12:22.31 | thewiizle | ? |
| 12:22.35 | thewiizle | or print_r? |
| 12:24.23 | xma1 | i can call land lines fine it seems |
| 12:24.50 | ThoMe | how i can reload the voicemail.conf? |
| 12:24.56 | ThoMe | only "reload" ? |
| 12:24.59 | ThoMe | "reload voicemail" ? |
| 12:25.34 | Zeeek | ThoMe I think reload the module (ending in .so) |
| 12:25.51 | ThoMe | Zeeek: hm. i can't find a voicemail modul |
| 12:25.56 | ThoMe | ok.. :) only reload |
| 12:26.17 | Zeeek | well if there's a command reload voicemail that does sound good |
| 12:26.21 | Mr_Varmit | [2-9]11|0T|011xxx.T|[0-1][2-9]xxxxxxxxxx|[2-9]xxxxxxxxxx|[2-9]xxxT |
| 12:26.34 | Mr_Varmit | anyone have a better digitmap? |
| 12:26.58 | Zeeek | digitmaps are geo-dependent |
| 12:26.58 | Mr_Varmit | i cant get it to dial more than 7 numbers |
| 12:28.33 | Mr_Varmit | Zeeek, im not worried about what calls that are made...there is only three of us here...we all foot the bills....we dont let anyone use the phones...so what would allow the 11 numbers to be dialed? |
| 12:28.56 | Mr_Varmit | i have _9. in there now.... |
| 12:29.13 | Mr_Varmit | from what i understand..thats supposed to allow any number correct? |
| 12:29.20 | Zeeek | what is this digitmap for? |
| 12:29.28 | Mr_Varmit | but my digitmap is stopping that i would assume |
| 12:30.00 | Mr_Varmit | ive read that the digitmap is stopping me from having past 7 numbers |
| 12:30.07 | Mr_Varmit | no matter what i put in the dialplan |
| 12:30.15 | Zeeek | what is the digitmap for? Your phone? |
| 12:30.36 | Mr_Varmit | yes sir |
| 12:30.50 | Zeeek | and we are expected to guess what the phone is? |
| 12:31.23 | *** join/#asterisk lirakis (n=lirakis@65.200.191.253) |
| 12:31.27 | Mr_Varmit | im sorry...lolol..its a polycom 500 |
| 12:31.34 | Zeeek | http://bc.whirlpool.net.au/forum-replies-archive.cfm/480674.html |
| 12:31.43 | Mr_Varmit | ive read the polycom admin guide....i cant make heads or tails of it |
| 12:31.49 | Zeeek | go look around for info on polycomm digitmaps, that would be logical |
| 12:32.11 | Zeeek | or wait for TKDefender who will help you on Polycom questions |
| 12:33.31 | Mr_Varmit | ok...thanks....ill read up alittle more....i do apologize for being a pain :) |
| 12:35.52 | Zeeek | we're all used to that here |
| 12:36.15 | Zeeek | you can find material on the phone dialplan though, it's standard |
| 12:36.36 | *** join/#asterisk ming_zym (n=ming_zym@124.254.57.221) |
| 12:37.03 | *** join/#asterisk elixer (i=elixer@65.207.74.18) |
| 12:38.32 | ThoMe | what is "dad" and "oad" ? |
| 12:39.17 | Zeeek | ??? |
| 12:39.23 | Zeeek | use in sentence please |
| 12:40.03 | Mr_Varmit | lolol |
| 12:40.18 | Zeeek | ~dad |
| 12:40.19 | jbot | rumour has it, dad is mom |
| 12:40.21 | Mr_Varmit | omg...i choked on my coffee on that one |
| 12:40.25 | Zeeek | õad |
| 12:40.34 | Zeeek | ~oad |
| 12:41.05 | Mr_Varmit | whats the one from revenge of the nerds......what if cat....really means dog..... |
| 12:41.21 | *** join/#asterisk dominic1 (n=dob@213.221.82.242) |
| 12:42.50 | nexilus | When doing a outgoing/ text file, can i control which of the two "parties" to call first? |
| 12:43.11 | nexilus | so far it allways dials the "Channel" party first, then the Extension party |
| 12:43.17 | nexilus | and i would like it the other way around |
| 12:43.27 | Zeeek | ThoMe look type reload <tab> in the cli - see app_voicemail.so |
| 12:44.40 | nexilus | Any ideas? |
| 12:44.57 | *** part/#asterisk dominic1 (n=dob@213.221.82.242) |
| 12:45.59 | Zeeek | reverse the two? |
| 12:46.11 | Zeeek | use a Local channel |
| 12:46.33 | *** join/#asterisk ManxPower (n=manxpowe@5.sub-75-202-141.myvzw.com) |
| 12:49.24 | xma1 | alrighty |
| 12:49.30 | xma1 | i'm able to call my cell phone |
| 12:49.39 | Zeeek | we're thrilled with that news |
| 12:49.53 | xma1 | I'm thrilled |
| 12:49.54 | *** join/#asterisk theHub (n=theHub@69.177.93.21) |
| 12:50.06 | Zeeek | I'm thrilled that you're thrilled |
| 12:50.08 | xma1 | just takes a while to receive the hang-up signal |
| 12:50.14 | xma1 | Oh thank you Sir! |
| 12:50.35 | Zeeek | yes, something like the "flush twice, it's a long way to Taco Bell" theory |
| 12:51.54 | *** join/#asterisk saftsack (n=saftsack@p54A743F9.dip.t-dialin.net) |
| 12:51.58 | Strom_M | that's nothing |
| 12:52.02 | Strom_M | I can call the correct time |
| 12:52.15 | Strom_M | !!!!!!!!!!!!! |
| 12:52.19 | Zeeek | I can call the correct temperature |
| 12:52.32 | xma1 | is there a device that's a bit 'snappier' than the grandstream. the old pbx that IP-PBX replaced was pretty quick in dialing |
| 12:52.46 | Strom_M | I can call the correct time AND the correct temperature ALL AT THE SAME TIME |
| 12:52.48 | Zeeek | Sipura |
| 12:52.58 | Strom_M | Polycom |
| 12:53.07 | xma1 | 4 port FXO |
| 12:53.10 | Zeeek | I can call the correct time, temperature and have a sexy female voice say dirty things to me |
| 12:53.17 | JT | ~gs |
| 12:53.18 | jbot | GrandSuck phones & gateways are cheap junk which should be avoided with extreme prejudice. |
| 12:53.39 | Zeeek | "starter home, needs fixing" |
| 12:53.41 | *** join/#asterisk michael-i (n=michael-@141.41.40.186) |
| 12:53.50 | xma1 | so whats better.. |
| 12:54.07 | Zeeek | I'd just use a SIP provider |
| 12:54.18 | Strom_M | i'm going back to bed |
| 12:54.28 | Zeeek | call time first |
| 12:54.41 | Zeeek | anyone using Freenum? |
| 12:54.51 | xma1 | this is for a business |
| 12:55.01 | Zeeek | even more important |
| 12:55.05 | Zeeek | ditch the fxo |
| 12:55.19 | xma1 | we got snom phones |
| 12:55.26 | xma1 | they seem pretty nice |
| 12:55.57 | JT | ~phones |
| 12:55.58 | jbot | i guess phones is http://bani.anime.net/phones/. While personal preference will dictate which phone works best for you, general consensus on a rough order of quality and suggestibility is as follows: Polycom (any), Aastra 480i, Aastra 5i Series, Cisco 7940+, Linksys SPA-9XX, Snom, and finally everything else. Do not consider Grandstream phones. Ever. places ... |
| 12:56.45 | ManxPower | xma1: the GS phones do not support dialing other than via a timeout or pressing the Dial button. They suck. |
| 12:57.13 | nexilus | Zeek: sorry for the delay, "use a local channel" ? how do you mean? |
| 12:57.25 | Zeeek | see Local channel |
| 12:57.32 | nexilus | ~Local channel |
| 12:57.38 | *** join/#asterisk vpanayotov (n=kvirc@83.228.51.12) |
| 12:57.42 | nexilus | guess im not authorized ^^ ima google |
| 12:57.42 | Zeeek | http://www.voip-info.org/wiki/index.php?page=Asterisk+Local+channels |
| 12:57.42 | xma1 | the snom300 seems like a solid phone, very nice |
| 12:58.15 | Zeeek | nexilus I'm guessing with a local channel you can control the order (in the dialplan) |
| 12:58.17 | *** join/#asterisk Aeudian (n=somewher@ip67-94-157-98.z157-94-67.customer.algx.net) |
| 12:58.31 | Zeeek | But I'm not the brightest bulb in the bunch |
| 12:59.07 | vpanayotov | hi. Is it possible to use automon feature with sjphone. I believe that the UAC should send SIP INFO but I don't know how |
| 12:59.12 | JT | xma1: it's ok if you haven't seen better |
| 12:59.24 | JT | xma1: certainly not "very nice" though |
| 13:00.16 | ManxPower | xma1: Many of us like Polycom -- I recommend Polycom |
| 13:00.38 | xma1 | it feels very nice.. maybe not in function |
| 13:00.56 | xma1 | and only 99 american dollars |
| 13:01.03 | ManxPower | vpanayotov: you need to get DTMF working before you try automon. We recommend RFC2833 as the DTMF mode on BOTH the phone AND Asterisk |
| 13:01.08 | *** join/#asterisk _ShrikE (n=ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 13:01.22 | JT | xma1: what a waste of money |
| 13:01.39 | vpanayotov | ManxPower: Thank you! I will try this |
| 13:02.08 | JT | xma1: a polycom IP320 is USD$85 |
| 13:02.25 | ManxPower | JT: If the USD/other currency exchange rates keep falling, we'll be PAYING people to buy phones. |
| 13:02.56 | JT | heh |
| 13:03.28 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 13:03.28 | *** mode/#asterisk [+o anthm] by ChanServ |
| 13:06.02 | xma1 | if i get a sip provider here in the us, can i still use asterisk, or not even use it at all? |
| 13:06.14 | *** join/#asterisk Lawbringer (n=Lawbring@212.183.136.192) |
| 13:06.33 | *** join/#asterisk davevg-btwtech (n=davevg-b@nj-67-76-177-147.sta.embarqhsd.net) |
| 13:08.45 | Zeeek | rephrase the question |
| 13:08.58 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 13:11.13 | *** join/#asterisk Koshatul (n=evangeli@ppp188-33.static.internode.on.net) |
| 13:11.32 | Katty | herro. |
| 13:12.07 | *** join/#asterisk luciofm (n=gcc@201.37.84.83) |
| 13:12.13 | xma1 | if i were to go the sip provider route (instead of FXO gateway) i can axe asterisk? |
| 13:12.21 | *** join/#asterisk w9sh (n=w9sh@adsl-068-209-117-205.sip.asm.bellsouth.net) |
| 13:12.36 | luciofm | hi |
| 13:12.56 | luciofm | where can i find the API diferences from 1.2 to 1.4? |
| 13:13.24 | luciofm | i'm writting an app that must run on both versions |
| 13:13.41 | *** join/#asterisk Schumie (n=Steve@cmarfw01.marlow.spinvox.com) |
| 13:16.10 | creativx | Hey did I tell you that I hate html |
| 13:17.09 | xma1 | who offers good sip service in USA |
| 13:17.10 | Zeeek | Hi Katty |
| 13:17.55 | Zeeek | xma1 what region are you in? |
| 13:18.07 | xma1 | midwest |
| 13:18.46 | Zeeek | look at nufone.net |
| 13:20.29 | Zeeek | look at dslreportslook at dslreports.com in their forums on voip |
| 13:20.33 | Mr_Varmit | would there be a reason why i cant save 9am on the time conditions? i can on 10am...but not 9 |
| 13:20.34 | Zeeek | lots of reports |
| 13:21.14 | Katty | hi Zeeek |
| 13:21.23 | Zeeek | bad lag today |
| 13:21.24 | Katty | creativx: why do you hate html? |
| 13:21.32 | Katty | creativx: html is okay, if you use css :< |
| 13:21.50 | Mr_Varmit | it keeps going back to 00 |
| 13:22.04 | Mr_Varmit | i select 10 and it saves |
| 13:22.15 | Mr_Varmit | lol..it wants me to have bankers hours |
| 13:22.17 | Zeeek | no one understands you AsteriskNOW questions |
| 13:22.29 | *** join/#asterisk mordaunt (n=mordaunt@unaffiliated/mordaunt) |
| 13:23.00 | Katty | a co-worker made me cookies :> |
| 13:23.08 | Zeeek | with milk? |
| 13:23.14 | gremzoid | of the out-space kind? |
| 13:23.17 | Katty | i don't drink milk. |
| 13:23.20 | gremzoid | outter* |
| 13:23.21 | Katty | it makes me sick. |
| 13:23.25 | gremzoid | -t |
| 13:23.35 | Zeeek | I know Katty that's why I asked. |
| 13:23.44 | Katty | oh. |
| 13:23.56 | Katty | heh |
| 13:24.36 | luciofm | hi, there is some kind of 1.2 to 1.4 developer guide, or something? or some diferences guide? |
| 13:25.06 | *** join/#asterisk jacq (n=jal@61.12.17.165) |
| 13:25.10 | gremzoid | changes.txt? |
| 13:25.15 | gremzoid | CHANGELOG.TXT? |
| 13:25.31 | lirakis | luciofm: ATFOT |
| 13:25.33 | lirakis | ~thebook |
| 13:25.33 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf |
| 13:25.35 | [TK]D-Fender | UPGRADE.TXT ? |
| 13:26.16 | Katty | ^_- |
| 13:26.16 | jacq | anyone can recommend a Asterisk cluster code/solution... not sure which one is most popular. |
| 13:26.41 | Katty | hi jacq! |
| 13:26.50 | michael-i | luciofm, I would find an application that does something similar to yours that is part of the normal Asterisk distribution, then see how it is implemented in 1.2 vs 1.4 |
| 13:27.13 | iCEBrkr | [TK]D-Fender: Omg.. get a room. |
| 13:27.24 | Katty | iCEBrkr: simmer down now. |
| 13:27.29 | iCEBrkr | Katty: now? not later? |
| 13:27.38 | Katty | iCEBrkr: yes, now please. you're about to boil over. |
| 13:27.43 | iCEBrkr | :( |
| 13:27.49 | [TK]D-Fender | iCEBrkr, we've got a room... but you're all in here crowding us out! |
| 13:27.49 | Katty | fine fine. |
| 13:27.57 | iCEBrkr | oof! |
| 13:28.04 | creativx | Katty: i hate it because i got stuck with some css hell site that i had to customize |
| 13:28.12 | Katty | creativx: ooooh. |
| 13:28.17 | Katty | creativx: you should learn css. |
| 13:28.20 | Katty | creativx: it's teh uber. |
| 13:28.22 | creativx | i know |
| 13:28.23 | iCEBrkr | OMGWTFBBQ! Bioshock FTW.. FYI.. K/THX |
| 13:28.26 | creativx | i like css when it is nice to me |
| 13:28.29 | *** join/#asterisk saftsack (n=saftsack@p54A743F9.dip.t-dialin.net) |
| 13:28.34 | Katty | oh, it was mean css :< |
| 13:28.38 | Katty | that's just No Good(tm) |
| 13:28.41 | creativx | but i dont like css when shit has to work in ie, opera and firecrap |
| 13:28.48 | jacq | asterisk cluster... anyone? |
| 13:28.57 | creativx | and when the entire site uses some inbred css positioning soup |
| 13:28.58 | iCEBrkr | jacq: Sure, I'll take 2. |
| 13:29.03 | Katty | creativx: yeah that's a pain. especially with IE. |
| 13:29.17 | *** join/#asterisk saftsack (n=saftsack@p54A743F9.dip.t-dialin.net) |
| 13:29.34 | jacq | >< |
| 13:30.24 | jacq | my starting info is from voip-info.org... and i'm sure its outdated ! |
| 13:30.26 | Katty | jacq: i'll pass, thanks. |
| 13:31.24 | creativx | Katty: oh yes. im glad i dont work with this normally |
| 13:31.41 | *** join/#asterisk _PauloS_ (n=sony@c906b9f9.virtua.com.br) |
| 13:31.51 | _PauloS_ | Hi folks. |
| 13:31.58 | _PauloS_ | Somebody using unicall? |
| 13:32.38 | _PauloS_ | I can't find a set of spandsp/unicall wich goes well together. |
| 13:33.03 | Katty | creativx: i usually find a nice little template where most of the work is done, then go back and tweak it. |
| 13:33.08 | Katty | creativx: but it's notlike you have that option. |
| 13:33.29 | Katty | `Sauron: herro. |
| 13:34.19 | *** join/#asterisk nighty^ (n=nighty@sushi.rural-networks.com) |
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| 13:36.15 | creativx | Katty: sounds smart.. im glad our crm wraps the IE webbrowser control.. so thats the only thing i have to care about :) |
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| 13:48.46 | *** part/#asterisk Mr_Varmit (n=varmit@rrcs-64-183-163-178.west.biz.rr.com) |
| 13:51.07 | *** part/#asterisk xma1 (n=xma2@adsl-70-233-17-150.dsl.chcgil.sbcglobal.net) |
| 13:51.42 | Aeudian | Good Morning, I am looking for an Operator Panel that is professional looking that is supported by Windows/OS X. I have looked into hudlite but i seem to run across isntallation issues like everyone else does as well as flash operator panel but it really doesn't come across as a professional interface but since it is a web based interface it works on windows/os x. does anyone else have a suggestion on an operator panel wh |
| 13:52.50 | thewiizle | Can someone correct this for me, GotoIf(["${BLOCKED}" = "1"]?6:8) |
| 13:52.57 | thewiizle | If blocked = 1 i want it to go to 6 |
| 13:53.56 | Katty | Aeudian: FOP can look professional if you change the color scheme (= |
| 13:54.14 | Katty | Aeudian: otherwise, I suggest you hire a programmer to write something for you. |
| 13:55.25 | Aeudian | Katty: oh, you can change the color? hmmm |
| 13:55.27 | iCEBrkr | Aeudian: It's unfortunate that most of the apps out there are written by geeks who have no idea how to make an application look 'professional'. |
| 13:56.11 | iCEBrkr | Katty: Last I looked, it really seemed like some 12yr old put it together. |
| 13:56.15 | Katty | Aeudian: Yes. everything. the color of the button, the border, the background. put your company logo up there if you like. |
| 13:56.25 | iCEBrkr | Buttons were flrrrrrig'n huuge. |
| 13:56.42 | Katty | iCEBrkr: mine are tiny to accomodate 50 buttons. |
| 13:56.49 | Katty | iCEBrkr: with a sleek black look, and the company's logo |
| 13:56.59 | Katty | iCEBrkr: actually, i took the logo off. but it was up there. |
| 13:57.08 | Aeudian | Katty: any chance i could see a screenshot of yours just to get an idea? |
| 13:57.10 | iCEBrkr | Who needs a stink'n logo? |
| 13:57.20 | iCEBrkr | more screen clutter |
| 13:57.31 | Katty | sure, sec. |
| 13:57.43 | Aeudian | iCEBrkr: ya i agree with the buttons i was looking into isymphony but its just as bad as hudlite for setup and doesnt support mac |
| 13:58.02 | iCEBrkr | The same holds true with the majority of the softphones out there. |
| 13:58.17 | iCEBrkr | They all have some sort of skinning dealio that in reality sucks donkey nuts. |
| 13:58.29 | iCEBrkr | DU-PLO BUTTONS! |
| 13:58.38 | iCEBrkr | EVEN A 6yr old can understand! |
| 13:58.54 | iCEBrkr | Why does a softphone have to take up 3/4 of the screen? I'm not sure. |
| 13:59.13 | Aeudian | iCEBrkr: your correct right on |
| 13:59.16 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 13:59.59 | iCEBrkr | If I could just wrap my head around this threading thing. I'd have a slick call manager available. |
| 14:00.50 | iCEBrkr | and once the call manager was done, there'd be an operator panel. |
| 14:00.53 | iCEBrkr | :-\ |
| 14:01.29 | grandpapadot | Aeudian: Operator Anywhere - http://heavylogic.com/voip/oa |
| 14:01.35 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 14:01.39 | grandpapadot | It's written in Flex2/php back end. |
| 14:01.45 | *** join/#asterisk deeperror (n=deeperro@69-215-202-195.ded.ameritech.net) |
| 14:02.15 | iCEBrkr | grandpapadot: I'm actually impressed with the vapor-ware screenshot :D |
| 14:02.59 | grandpapadot | iCEBrkr: Not vaporware, it works, very well. |
| 14:03.31 | grandpapadot | It's just for our service mainly, but I'm going to post the source on the wiki |
| 14:03.53 | grandpapadot | It's all XML based, so you just modify the server side php to suite you. |
| 14:03.59 | iCEBrkr | nice |
| 14:05.24 | grandpapadot | Aeudian: Or roll your own... |
| 14:06.20 | Aeudian | grandpapadot: the OA looks nice i just had a server deadline this week |
| 14:06.21 | iCEBrkr | grandpapadot: it's about time someone took notice and designed a clean interface. |
| 14:06.35 | Katty | mmmmclean. |
| 14:06.46 | Katty | speaking of clean, i need to give jager a bath tonight >.< |
| 14:06.51 | Katty | 70lbs of wet dog. fun. |
| 14:06.56 | Aeudian | lol |
| 14:07.00 | grandpapadot | iCEBrkr: Truth be told.. that's just the default Flex2 stuff. You can just drop a CSS in there and change the look and feel however you want. |
| 14:07.13 | iCEBrkr | grandpapadot: Pefect! |
| 14:10.01 | iCEBrkr | I should rewrite my stilly CallerID dealio. |
| 14:10.05 | iCEBrkr | stilly? |
| 14:10.08 | iCEBrkr | silly even. |
| 14:10.26 | *** join/#asterisk Dan0maN_Work (n=dschuh@64.149.174.136) |
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| 14:15.47 | gremzoid | ~pb |
| 14:15.47 | jbot | A Pastebin is a place to paste your stuff without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
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| 14:39.07 | Andretii | Hello, does anyone knows about transfering calls issues after upgrading to 2.0. My polycoms are only allowing me to dial 2 digits out of 3 when i try to do a transfer |
| 14:40.37 | *** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman) |
| 14:41.45 | *** join/#asterisk l2trace99 (n=asd@fl-67-76-209-28.sta.embarqhsd.net) |
| 14:42.42 | TrentCreek | j |
| 14:43.12 | etfonhomey | Andretii,That's probably to do with your dial string on the phones. |
| 14:43.26 | etfonhomey | Andretii, Excuse me, digit map. |
| 14:44.38 | Andretii | So you think that might have change after the upgrade? |
| 14:44.45 | TrentCreek | they cam out with version 2 while i was alseep? |
| 14:45.02 | etfonhomey | Andretii,It's always possible or you lost your own customizations. See dialplan.digitmap in sip.cfg. |
| 14:45.27 | Andretii | I will check |
| 14:45.43 | *** join/#asterisk galeras (n=galeras@201.244.244.170) |
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| 14:47.46 | *** mode/#asterisk [+o blitzrage] by ChanServ |
| 14:49.07 | Andretii | etfonhomey I see no entry for dialplan.digitmap in sip.conf |
| 14:49.53 | etfonhomey | Andretti, I'm talking about sip.cfg that you use to confgure your phones not *'s sip.conf. |
| 14:53.25 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 15:02.20 | Andretii | etfonhomey i found that the new sip.cfg is <dialplan.digitmap="[2-9]11|0T|011xxx.T|[0-1][2-9]xxxxxxxxx|[2-9]xxxxxxxxx|[2-9]xxxT" |
| 15:02.41 | Andretii | and an old version has digimap in front of dialplan.digitmap |
| 15:03.01 | Andretii | think that can be the problem? |
| 15:04.33 | *** join/#asterisk qdk (n=qdk@0x573fe57e.bynqu2.broadband.tele.dk) |
| 15:04.55 | etfonhomey | I can't say. You might check the release notes for the version you are using to see if it mentions it. I have SIP 2.2 and the digitmap settings are in the <digitmap /> XML tag. |
| 15:06.30 | etfonhomey | There are two settings dialplan.digitmap and dialplan.digitmap.timeOut |
| 15:09.30 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 15:13.26 | TrentCreek | Hye..a bit off topic, but I am trying to set up AstBill for Asterisk, and the install docs call for making a symbolic ling from the httpd directory to a directory at another location (/home/astbill/wwwroot) |
| 15:14.21 | TrentCreek | Secuirty is preventing it from happening..though the warning says to use restorecon -v tp allow it, but it still does not |
| 15:14.50 | deeperror | codefreeze: around today? |
| 15:18.39 | codefreeze | deeperror: I am. how goes? |
| 15:19.07 | *** join/#asterisk gremzoid (n=gremzoid@d58-111-173-16.rdl5.qld.optusnet.com.au) |
| 15:19.09 | *** part/#asterisk gremzoid (n=gremzoid@d58-111-173-16.rdl5.qld.optusnet.com.au) |
| 15:19.43 | deeperror | with regards to cdr warnings from last week, I have an isolated call from start to fin if you have a few |
| 15:19.55 | deeperror | it happens on 3 way calling |
| 15:20.11 | deeperror | when the callee drops out |
| 15:21.23 | tzafrir | anybody recommend a good BRI traffic analyzer? Preferebly a free software one |
| 15:21.37 | Andretii | etfonhomey yes, I have dialplan.digitmap.timeOut an it is the same in 2 sip.cfg files |
| 15:21.42 | tzafrir | (yeah, I know wireshark) |
| 15:22.31 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 15:22.41 | ThoMe | tzafrir: hi. benutzt you phpagi-asmanager ? |
| 15:23.05 | deeperror | having an issue with pastebin ha |
| 15:23.33 | tzafrir | ThoMe, sorry, but I failed to parse that... |
| 15:23.47 | ThoMe | tzafrir: hm ok. |
| 15:24.13 | *** join/#asterisk karlh626 (n=kharris@IP-66.249.227.5.indigital.net) |
| 15:24.14 | *** join/#asterisk JRsup1 (n=chatzill@12-207-206-43.client.mchsi.com) |
| 15:25.37 | deeperror | codefreeze: http://pastebin.ca/729884 |
| 15:30.00 | JRsup1 | can anyone tell me how to get chan_mobile added to my asterisk? |
| 15:33.36 | TrentCreek | i dont know |
| 15:35.13 | codefreeze | deeperror: how does Zap/11 get dialtone for the 2nd party? |
| 15:36.39 | codefreeze | does he flash hook or use # or something? |
| 15:37.09 | TrentCreek | everyone fell alseep |
| 15:38.05 | Sci_05 | its to hot outside today to work.... |
| 15:38.20 | codefreeze | JRsup1: you go to asterisk-addons, trunk, and do make; then copy the chan_mobile.so to /usr/lib/asterisk/modules, or wherever they are stored. |
| 15:38.46 | De_Mon | anyone successfully connected an ESI phone to asterisk? |
| 15:38.58 | TrentCreek | we lucky..it rained today and only 73...othereise it would be 90 |
| 15:39.10 | deeperror | codefreeze: agents are using flash. I'm able to reproduce here by just performing a 3 way call it will give those warnings every time |
| 15:39.14 | Sci_05 | I am over by chicago, its going to be 90 today |
| 15:39.27 | JRsup1 | codefreeze: where is asterisk-addons? |
| 15:39.29 | Sci_05 | thank god I have a server room, I can just go chill in there |
| 15:39.44 | ThoMe | iss asterisk 1.2.X comlete removed from the site? |
| 15:39.52 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 15:39.57 | TrentCreek | wow...that is near the artic..should be like 50 by now |
| 15:40.39 | TrentCreek | an i am way down in south texas where it may freeze once for a ffew hours during the winter |
| 15:40.42 | outtolunc | ThoMe, yes it was removed last week |
| 15:41.03 | *** join/#asterisk smash- (n=smash@70.103.90.214) |
| 15:41.06 | deeperror | codefreeze: it gives the warnings once the callee hangs up the call. |
| 15:41.32 | deeperror | from a user standpoint it seems 100% fine. I just get the warnings. |
| 15:41.33 | smash- | Hey does anyone know what requirements a switch must have to function correctly as a voip and ipsec switch? |
| 15:42.05 | smash- | I'm wondering if a 3com 3300 series will work correctly without collison |
| 15:43.39 | codefreeze | deeperror: I'm mainly concerned **how** the 3-ways are formed. Different ways == different code paths in chan_zap... |
| 15:43.51 | codefreeze | and different cdr code. |
| 15:44.11 | deeperror | codefreeze: being performed by pressing flash on the telephone. hook flash. |
| 15:44.39 | codefreeze | JRsup1: you have to check it out separately with svn. Instead of asterisk/trunk, check out asterisk-addons/trunk |
| 15:45.02 | codefreeze | deeperror: ok, looked like it, but thought best to verify. |
| 15:45.18 | deeperror | no prob anything else you need i can make available |
| 15:46.09 | deeperror | it is writing the first outbound call to Master.csv but the second call is not in the log so that might be the effect of these warnings that the second call isn't being written |
| 15:46.32 | JRsup1 | codefreeze: Ok, I'm currently only running a standard trixbox beta distro and I've only used SVN once before for something else. Can you point me to any instructions for setting up SVN for use with this? |
| 15:46.58 | *** join/#asterisk sacitec (n=tobi@189.149.103.123) |
| 15:47.18 | sacitec | good morning |
| 15:47.36 | sacitec | what's the max value of rx and tx values in zapata.conf ? |
| 15:48.08 | sacitec | i have low call volume, and want to fix ti by moving that values |
| 15:48.40 | outtolunc | <PROTECTED> |
| 15:48.40 | outtolunc | <PROTECTED> |
| 15:48.54 | outtolunc | (that is in 1.2 anyways) |
| 15:49.26 | deeperror | sacitec: -100 to 100 |
| 15:51.01 | sacitec | if people that calls me via ZAP and doesn´t hear me, i need to modify tx value, right ? |
| 15:51.21 | deeperror | is it quiet or they don't hear you period? |
| 15:51.45 | sacitec | they say they don't listen enought, it's to low |
| 15:51.57 | *** join/#asterisk saftsack (n=saftsack@pD9E05A3D.dip.t-dialin.net) |
| 15:52.53 | *** join/#asterisk agx (n=AGX@host63-216-static.34-88-b.business.telecomitalia.it) |
| 15:53.14 | deeperror | tx value |
| 15:53.23 | agx | Hi, when compiling a kernel for asterisk + zaptel + mISDN how much is the HZ frequency? 250, 300 or 1000 ? |
| 15:53.38 | ManxPower | agx: 1000 |
| 15:53.40 | deeperror | i use channel banks and had better success changing the gain on the channel itself |
| 15:53.41 | ManxPower | always 1000 |
| 15:53.42 | agx | ManxPower, ty |
| 15:53.49 | sacitec | thanks |
| 15:53.58 | ManxPower | Looks like a crashed a router. *grumble* |
| 15:54.23 | codefreeze | JRsup1: try instructions at http://asterisk.org/developers/get-source |
| 15:54.48 | codefreeze | JRsup1: you'll need svn, or subversion, installed |
| 15:55.01 | *** part/#asterisk agx (n=AGX@host63-216-static.34-88-b.business.telecomitalia.it) |
| 15:55.33 | ManxPower | remind us again why JRsup1 should be running SVN? |
| 15:55.53 | codefreeze | JunK-Y: howdy!!! how goes? Survive Astricon? |
| 15:56.08 | JRsup1 | ManxPower, I'm trying to add the chan_mobile module |
| 15:56.19 | JunK-Y | yes survived, drunk for the next month :) |
| 15:56.36 | ManxPower | Oh. I guess that might be worth the torture and pain of running unreleased code. |
| 15:57.04 | tiav | hello |
| 15:57.12 | ManxPower | JRsup1: I take it you don't have any boxes in a production enviroment. |
| 15:58.15 | mvanbaak | I run SVN in production |
| 15:58.15 | *** join/#asterisk shinao1 (n=shinao1@41.222.65.162) |
| 15:58.20 | JRsup1 | ManxPower: No. This is a VM I'm trying to set up (pass-through of bluetooth device has proven successfull) at home. All I want it to do is take my cell phone and pass it's calls off to the house's POTS |
| 15:58.21 | mvanbaak | branches/1.4/ |
| 15:58.31 | mvanbaak | using svn to stay up-to-date |
| 15:58.48 | mvanbaak | running trunk in production is another story |
| 15:58.54 | tiav | Somebody know if asterisk have postgresql native support ? |
| 16:00.56 | karleeto | how do i reboot a bunch of polycom's again? sip notify chk-config ??? |
| 16:02.36 | ManxPower | karleeto: yes |
| 16:03.19 | karleeto | Unable to find notify type 'chk-config' |
| 16:05.07 | ManxPower | karleeto: then you did not configure it |
| 16:05.07 | *** join/#asterisk halogen8 (i=halogen8@ip68-6-197-105.sd.sd.cox.net) |
| 16:05.10 | patrickv0x | as anyone been able to register a Cisco 7970 with Asterisk using SIP ? If so, could you share your config file with me please? Thanks. |
| 16:05.18 | codefreeze | deeperror: so who drops first out of the 3-way? Zap/11, the first sip callee, or the second sip callee? |
| 16:05.44 | *** join/#asterisk Carlos_Tico (n=grillo_v@c-98-195-31-182.hsd1.tx.comcast.net) |
| 16:05.49 | halogen8 | why in sip registry would a peer show up as unreachable, but in sip peers, the same peer shows as registered? |
| 16:05.54 | karleeto | ahhh its polycom-chk-config |
| 16:06.12 | Carlos_Tico | i have a question people |
| 16:06.25 | Carlos_Tico | how can i install asterisk on my wrt54gs router ? |
| 16:06.39 | deeperror | Zap/11 |
| 16:06.50 | Carlos_Tico | Zap/11 ? |
| 16:06.50 | JunK-Y | Carlos_Tico: there already package for that. |
| 16:06.51 | deeperror | then both sip callee's remain in the call |
| 16:07.00 | codefreeze | very interesting. |
| 16:07.02 | Carlos_Tico | Junk-Y ? |
| 16:07.05 | Carlos_Tico | what do you mean ? |
| 16:07.27 | deeperror | codefreeze: but this occurs on all zap and has from day one with 1.4.11 |
| 16:08.24 | Carlos_Tico | <JunK-Y> Carlos_Tico: there already package for that. -- > where can i find that information i was almost 3 hours googling and nothing |
| 16:08.40 | *** join/#asterisk DrAk0 (n=ljd@nelug/coreteam/luisjose) |
| 16:09.07 | wwalker | Carlos_Tico: http://downloads.openwrt.org/whiterussian/0.9/packages/ |
| 16:09.34 | patrickv0x | Is anyone here using Cisco 7970? |
| 16:09.46 | l2trace99 | bunches of them |
| 16:09.48 | _PauloS_ | I'm using 7960s |
| 16:10.13 | patrickv0x | l2trace99: you're using 7970 bro ? |
| 16:10.23 | *** join/#asterisk [hC] (n=hardcore@S01060016b61c8983.vf.shawcable.net) |
| 16:10.25 | patrickv0x | Paulos: me too; i love 7940/7960 |
| 16:10.43 | l2trace99 | not with sip |
| 16:10.44 | codefreeze | deeperror: all 3 parties zap, and still occurs? hmmm. Lessee. Scenario 5-7 zap 3-ways, different parties hang up first. Time to test out scenario 7 again. |
| 16:10.46 | l2trace99 | tho |
| 16:10.53 | patrickv0x | l2trace99: could you hook me up with your config ? |
| 16:10.57 | _PauloS_ | patrickv0x: dont waste time trying to make them work with callweaver... |
| 16:10.59 | patrickv0x | ahh, you use it with call-manager or sccp ? |
| 16:11.10 | patrickv0x | what is callweaver ? |
| 16:11.10 | l2trace99 | forgot they are on cm |
| 16:11.25 | Carlos_Tico | <wwalker> -- >Thanks pal |
| 16:11.36 | karleeto | ff |
| 16:11.38 | karleeto | fffff |
| 16:11.40 | _PauloS_ | patrickv0x: the asterisk fork from Steeve Underwood of spandsp and unicall fame |
| 16:11.50 | karleeto | for future reference it was |
| 16:12.19 | deeperror | codefreeze: on all zap lines on the channel bank not just 1 specific. And both callee's are reach thru SIP. |
| 16:12.23 | karleeto | sip notify polycom-check-cfg 101 102 103 ... |
| 16:12.36 | ManxPower | l2trace99: SCCP/Skinny protocol does not support registration as far as I know. |
| 16:12.50 | l2trace99 | cm == Call Manager |
| 16:12.55 | l2trace99 | they are pure Cisco |
| 16:13.04 | ManxPower | Call Manager uses SCCP aka Skinny |
| 16:13.09 | l2trace99 | right |
| 16:13.18 | ManxPower | That does not invalidate my statement |
| 16:13.44 | l2trace99 | polar bear are white |
| 16:13.56 | l2trace99 | my statement is all valid |
| 16:14.01 | l2trace99 | but off point |
| 16:14.25 | l2trace99 | i forgot that it was registered on cisco so i cannot off the help he needed |
| 16:15.14 | De_Mon | l2trace99 if SCCP/Skinny do not support registration, how can the phone be registered on cisco? |
| 16:17.39 | *** join/#asterisk |omni| (n=rob@c-67-185-56-106.hsd1.wa.comcast.net) |
| 16:20.13 | JRsup1 | codefreeze: I got the SVN and did 'make' (and make again after it does configureation) and I don't see that it made chan_mobile.so. There's a chan_mobile.c, so I think it should have, but I can't find the built module |
| 16:20.35 | De_Mon | JRsup1 did you run the menuconfig? |
| 16:21.01 | JRsup1 | De_Mon: can't...it says I need ncurses. (which I have...oddly enough) |
| 16:21.32 | codefreeze | JRsup1: Uh, you may need to install libbluetooth modules.... |
| 16:21.53 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 16:22.06 | JRsup1 | codefreeze: standard yum bluez-utils? or is that for asterisk? |
| 16:22.09 | l2trace99 | The phone registers to cisco call manager. It is cisco's term. |
| 16:22.22 | l2trace99 | Their product so i use their terms |
| 16:22.46 | De_Mon | ok, just remember you're not in their channel so the words mean different things in here |
| 16:23.06 | l2trace99 | he was asking about their product |
| 16:23.09 | codefreeze | JRsup1: I forget the list. the linux kernel usually has the bluetooth stuff by default... |
| 16:23.14 | l2trace99 | it still registers |
| 16:23.25 | l2trace99 | just not with in the protocol |
| 16:23.29 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 16:23.50 | l2trace99 | the process of linking of the phone to CM is call registering |
| 16:24.16 | l2trace99 | if you just want to be right |
| 16:24.25 | l2trace99 | your right |
| 16:27.24 | *** join/#asterisk RakMac (n=Raky@d58-104-196-78.dsl.nsw.optusnet.com.au) |
| 16:28.02 | RakMac | hey guys |
| 16:28.17 | RakMac | I seem to have this recurring issue, which makes no sense to me. |
| 16:28.28 | RakMac | I have two machines trunked to each other, one in the US the other in AU |
| 16:29.14 | RakMac | All direct calls from AU extensions to US extensions have the following problem: AU users hear the US voice all scrambled, however the US hear the AU voice perfectly fine. |
| 16:29.29 | RakMac | However, when the AU users, connect to the US conference line, everyone can hear each other fine. |
| 16:29.40 | RakMac | Does anyone know what I could be doing wrong, or what the issue could be? |
| 16:30.13 | codec | different codecs? ;) |
| 16:30.17 | codefreeze | JRsup1: you'll prob. need libbluetooth1-dev bluez-utils, bluez-pin, bluez-pcmcia-support, maybe bluez-hcidump gnome-bluetooth might be fun... |
| 16:30.35 | RakMac | haha, codec - I have both of them trunked through gsm. |
| 16:30.41 | RakMac | so the trunk = gsm. |
| 16:30.47 | codec | mmh |
| 16:31.04 | RakMac | i've tried changing the codec, but having the same issue. |
| 16:31.18 | codec | bandwidth problems maybe? |
| 16:31.26 | RakMac | hmm. |
| 16:31.37 | codec | i had a similar problem |
| 16:31.45 | RakMac | but why would things stuff up during a direct connection, then work perfectly fine seconds later in a conference? |
| 16:31.50 | codec | user a => 512kbit upload |
| 16:31.52 | RakMac | and this is ALL the time, not just once or twice |
| 16:31.53 | codec | user b => 100mbit |
| 16:32.10 | codec | user a was always scrambled |
| 16:32.23 | codec | but he was able to hear b fine |
| 16:32.34 | RakMac | ahhhh |
| 16:32.38 | RakMac | how did you solve the issue? |
| 16:32.47 | codec | fixed it by upgrading the uplink connection.. |
| 16:32.51 | RakMac | haahaha |
| 16:33.34 | RakMac | hmm, how would that explain things being fine in the conference though? |
| 16:33.37 | codec | maybe the problem just disappeared at the same time ;) |
| 16:34.01 | RakMac | it happens with anyone on that damn US machine. |
| 16:34.13 | RakMac | the other thing is, the AU machine is running off a vmware machine. |
| 16:34.22 | RakMac | so i thought that could be the issue - but still doesn't explain why the conference is fine. |
| 16:34.29 | [hC] | it could |
| 16:34.36 | [hC] | conferences get transcoded to slin |
| 16:34.54 | *** join/#asterisk Toerkeium (i=oo@201.216.206.221) |
| 16:34.59 | RakMac | slin? |
| 16:34.59 | [hC] | which also use a separate timing source |
| 16:35.05 | [hC] | signed linear |
| 16:35.06 | Katty | file: |
| 16:35.10 | Katty | file: ping. |
| 16:35.11 | RakMac | ohhhhh |
| 16:35.12 | [hC] | its a codec type like gsm, or ulaw, or g729 |
| 16:35.13 | RakMac | gotcha |
| 16:35.15 | file | Katty: pong. |
| 16:35.21 | Katty | file: how was the 6th :> |
| 16:35.25 | file | Katty: gooooooooooood |
| 16:35.29 | Katty | file: ^_^ |
| 16:35.36 | RakMac | am i able to trunk the two with slin? |
| 16:35.36 | [hC] | vmware is just... not a good idea.. as i understand it anyways |
| 16:35.38 | RakMac | ? |
| 16:35.43 | Katty | file: did you get cake? :> |
| 16:35.53 | file | Katty: yes, my mother gave me a mini birthday cake |
| 16:35.54 | l2trace99 | codec how many AU -> US are going on at the same time |
| 16:35.56 | JunK-Y | [hC]: send me this great recording of astricon. |
| 16:35.58 | [hC] | im sure there are ways to have vmware work well if you put work into it but i wouldnt touch that |
| 16:36.03 | Katty | file: ooh nicey nice :> |
| 16:36.03 | JunK-Y | while we were in the golf car. |
| 16:36.17 | [hC] | JunK-Y: i dont have it on my computer here, but i will get it later today and email it to you ... what is your email address? |
| 16:36.23 | Katty | file: hippo belated birdie two ewe :> |
| 16:36.24 | file | Katty: and how was the 7th? |
| 16:36.24 | codec | l2trace99: i guess you wanted to ask RakMac? |
| 16:36.26 | *** join/#asterisk etfonhomey (n=chatzill@12.169.248.226) |
| 16:36.37 | Katty | file: i got diamonds :> |
| 16:36.40 | RakMac | l2trace99, they are trunked with GSM - and then the user has a choice between using alaw or ilbc. |
| 16:36.43 | file | Katty: ooooooooooh |
| 16:36.45 | Katty | file: and cookies :> |
| 16:36.46 | l2trace99 | sry i am slow today ;) |
| 16:37.00 | Katty | file: they are shiny, and in mine ears. |
| 16:37.02 | RakMac | codec, is it possible to trunk with slin? |
| 16:37.05 | Katty | file: the..diamonds...not the cookies. |
| 16:37.07 | file | Katty: excellent! |
| 16:37.14 | codec | RakMac: ask hC :) |
| 16:37.31 | RakMac | sorry! haha |
| 16:37.33 | [hC] | RakMac: i dont think so, but i wouldt do that either. |
| 16:37.34 | ManxPower | RakMac: I guess you might be able to use IAX2 trunking with SLN, but I can't imagine why you would want to. |
| 16:37.36 | Katty | file: we're all grown up! |
| 16:37.42 | file | Katty: I knowz! |
| 16:37.42 | [hC] | RakMac: try using g726, or taking vmware out of the mix. |
| 16:37.46 | RakMac | gotcha. |
| 16:37.57 | ManxPower | also try turning off IAX2 trunking |
| 16:37.57 | RakMac | yeah i've heard vmware has weird timing issues. |
| 16:38.08 | [hC] | RakMac: the vmware machine is going to be the big question mark here. |
| 16:38.16 | l2trace99 | RakMac what is the utilation of the AU data link ? |
| 16:38.17 | Katty | and now i shall play warcraft. for it is lunch time. |
| 16:38.25 | l2trace99 | is it max ? |
| 16:38.27 | JRsup1 | adding bluetooth libs didn't help, and it looks like bluetooth is built in my kernel. anyone know why my make menuconfig is asking for ncurses when I already have it? |
| 16:38.34 | RakMac | no, definitely not max at all |
| 16:38.44 | l2trace99 | percentage |
| 16:38.46 | RakMac | ManxPower, if i turned iax2 trunking off, what would i use to trunk the two machines? |
| 16:38.56 | RakMac | i'd say, probably about 20% if not less. |
| 16:39.27 | l2trace99 | maybe codec conversion on from the vm |
| 16:40.09 | ManxPower | RakMac: perhaps you do not understand the term "trunking" Trunking is an IAX2 specific feature that takes multiple small audio packets and puts them inside a larger UDP packet, the far end then unwraps the interleaved packets and turns them back into IAX2 audio streams. |
| 16:40.42 | ManxPower | trunking is a method to save bandwidth used by the UDP packet headers. It does nothing else. |
| 16:41.31 | codefreeze | JRsup1: make menuselect? |
| 16:41.31 | RakMac | ahhh! |
| 16:41.53 | JRsup1 | codefreeze: sry, make menuconfig |
| 16:42.08 | *** part/#asterisk Kurin- (n=Kurin@lithium.delete.org) |
| 16:42.26 | ManxPower | RakMac: so remove the trunk=yes option from iax.conf on both sides of the link |
| 16:43.06 | RakMac | thanks very much guys, you're a great help i'm going to try that now. |
| 16:43.33 | RakMac | quick question though - i know vmware could still be messed up - but why would local calls on the AU machine from AU -> AU work perfectly fine. |
| 16:43.33 | RakMac | ? |
| 16:43.43 | ManxPower | RakMac: we can't tell you. |
| 16:43.58 | RakMac | ah that's right, like HC said - vmware big question mark. |
| 16:44.03 | ManxPower | But once you have removed the stuff that is KNOWN TO MESS UP VOIP then you can look at the issue again. |
| 16:44.04 | RakMac | and that's why most people don't use it, doh! |
| 16:44.15 | RakMac | ok sec! |
| 16:44.45 | hmmhesays | bah I just blew up my psu in my workstation |
| 16:46.28 | cpm | ick |
| 16:47.11 | *** join/#asterisk Zion800 (n=None@cpe-76-167-156-224.socal.res.rr.com) |
| 16:47.54 | Zion800 | When making changes to Zaptel.conf and running ztcfg, is it possible to just do a reload chan_zap.so in asterisk? |
| 16:48.13 | [TK]D-Fender | Zion800, For some changes, yes |
| 16:48.36 | Zion800 | well, i have a TDM2400, and the zaptel.conf file got blown away |
| 16:48.49 | ManxPower | Zion800: /etc/asterisk/zaptel.conf is not touched by ztcfg |
| 16:49.02 | Zion800 | i know |
| 16:49.03 | RakMac | ManxPower, i can hear them a lot clearer now |
| 16:49.12 | Zion800 | zaptel.conf is |
| 16:49.15 | RakMac | i might have to play with codecs now, but setting trunk=no definitely did somthing |
| 16:49.16 | RakMac | *something |
| 16:49.32 | hmmhesays | atacomm is now more, now I can't find cheap polycom ip 320's |
| 16:49.55 | ManxPower | hmmhesays: try Gaston Gureau at 504-523-7874 |
| 16:50.11 | ManxPower | tell him Eric W sent you. I don't get comissions from him. |
| 16:50.35 | hmmhesays | hrm, who is he? |
| 16:50.46 | ManxPower | he is the vendor MY customers usually buy from. |
| 16:51.00 | *** join/#asterisk CunningPike (n=CunningP@S010600095b33697f.vc.shawcable.net) |
| 16:51.09 | ManxPower | He runs Avenue Computer Supply |
| 16:52.40 | RakMac | ManxPower, should i also disable the jitterbuffer |
| 16:53.06 | *** join/#asterisk saftsack (n=saftsack@pD9E05A3D.dip.t-dialin.net) |
| 16:53.13 | hmmhesays | ahh I see |
| 16:53.52 | ManxPower | RakMac: it would not hurt. There have been issues with the IAX2 jitter buffer in the past, but I am not aware of any issues with 1.4, but I don't use 1.4 |
| 16:54.05 | *** join/#asterisk waKKu (n=eurulo@unaffiliated/wakku) |
| 16:54.14 | Zion800 | chan_zap.so is not loading in asterisk. in Zapata.conf, I have channels 4,7-10 set up as fxs_ks and channels 12-24 set up as fxo_ks. In /etc/zaptel.conf, i have fxsks=4,6-10 and fxoks=13-24 |
| 16:54.28 | ManxPower | Zion800: and if you do a "load chan_zap.so" what happens"? |
| 16:54.51 | Zion800 | oops...i was doing reload :-) |
| 16:54.55 | *** part/#asterisk halogen8 (i=halogen8@ip68-6-197-105.sd.sd.cox.net) |
| 16:55.10 | ManxPower | if it's not loading then you can't reload it |
| 16:55.19 | Zion800 | didn't dawn on m |
| 16:55.20 | Zion800 | me* |
| 16:57.21 | RakMac | ManxPower, God this is random. Once again, they hear me fine. But i can hear them a little better than before. |
| 16:57.38 | RakMac | Vmware has to be the bottleneck. |
| 16:57.47 | RakMac | I remember hearing of kernel patches etc, to attempt to get it working. |
| 16:58.46 | tzafrir | Zion800, unload chan_zap.so |
| 16:58.50 | ManxPower | RakMac: you are starting out with a complex environment and are simplifying it. That is backwards. |
| 16:58.51 | tzafrir | load chan_zap.so |
| 16:59.09 | Zion800 | tzafrir: thanks, i got it up and running :-) |
| 16:59.21 | RakMac | good point |
| 17:00.01 | tzafrir | reload [chan_zap.so] will not add any new channels. And it seems that currently 'zap restart' has been made impotent |
| 17:00.24 | RakMac | thanks very much though ManxPower |
| 17:00.28 | Zion800 | they need to give it some viagra |
| 17:00.33 | ManxPower | tzafrir that is 1.4 specific, I velieve |
| 17:00.34 | RakMac | really helped explain things, i was just so curious about that conf working. |
| 17:00.53 | RakMac | =) I appreciate all the help, and will never run * on vmware again. Lesson Learnt =D |
| 17:00.53 | ManxPower | RakMac: feel free to send a donation via paypal to eric@fnords.org |
| 17:01.14 | ManxPower | RakMac: once you get to all working they way you want, THEN try VMWare |
| 17:01.27 | RakMac | yeah i'll remember that next time |
| 17:01.29 | RakMac | thanks very much man |
| 17:02.15 | ManxPower | we just lost power. |
| 17:02.15 | ManxPower | I guess I should go figure out why. |
| 17:04.56 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 17:05.23 | santibiotico | exit |
| 17:07.53 | lesouvage | waht is the cure when a reboot on your asterisk server in a data center gives an input/output error (I'm afraid I'm in trouble) |
| 17:08.26 | *** join/#asterisk Aeudian (n=somewher@ip67-94-157-98.z157-94-67.customer.algx.net) |
| 17:08.44 | Aeudian | Hey Katty are you still around? |
| 17:09.40 | ManxPower | lesouvage: paste the actual text of the error message or we can't help you |
| 17:10.06 | lesouvage | ManxPower: -bash: /sbin/reboot: Input/output error |
| 17:10.12 | *** join/#asterisk Sinju (n=sinju@nc-205-240-45-138.sta.embarqhsd.net) |
| 17:10.35 | ManxPower | lesouvage: that means "your harddrive is going bad" |
| 17:11.05 | ManxPower | expect it to totally fail shortly before you finish copying the data off the server. |
| 17:11.46 | lesouvage | ManxPower: I have raid 1 running with brandnew harddrives, that is realy bad news |
| 17:12.12 | ManxPower | lesouvage: it is not an asterisk related error. |
| 17:13.45 | lesouvage | manxpower: this is just to make my stressfull day complete. I understand I'm in trouble |
| 17:14.41 | ManxPower | If you are using RAID, the issue could be some other issue |
| 17:14.51 | *** join/#asterisk miguellinux (n=chatzill@190.12.67.130) |
| 17:14.52 | ManxPower | but since this is not a RAID channel.... |
| 17:16.37 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 17:17.47 | karleeto | i'm experiencing pops and clicks on a commercial job i'm doing, and im freaked OUT! Our asterisk box is a amd64, 1gb ram, 3ware raid card, digium tdm400p (4 fxo modules and 4 analog lines |
| 17:17.51 | karleeto | ( |
| 17:17.52 | karleeto | err ) |
| 17:18.10 | lesouvage | ManxPower: The server is hanging in an internet exchange so I guess I have to make a backup as soon as possible and get the raid checked |
| 17:18.32 | lesouvage | Manxpower: I don't have access to the server myself. |
| 17:20.01 | karleeto | the lines have been been moved around since last time i fxotuned them, could that be the problem? |
| 17:20.09 | rpm | where can i buy a couple cheap ata's? is my best bet to look on ebay for bulk or do you know of any vendors which i can buy 3 or 4? |
| 17:20.37 | karleeto | we plugged and analog phone into the line thats experiencing the problem and it sounded fine |
| 17:21.54 | karleeto | anyone know what the problem could be? is it cause the 3ware card is using too much I/O, or is that module that the line is plugged into bad? |
| 17:22.40 | hmmhesays | is the polycom ip 550 worth the extra money over the ip 501? |
| 17:23.31 | l2trace99 | karleeto is it just the reboot command ? can you execute other commands ? |
| 17:23.42 | [TK]D-Fender | hmmhesays, nope |
| 17:23.44 | l2trace99 | ones that you don't normally use |
| 17:23.52 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 17:24.01 | [TK]D-Fender | hmmhesays, virtually no point in buying an IP550 |
| 17:24.23 | karleeto | any ideas? this is my first BIG job, been doing it for about 6 months now, and i really want things to work at least as good as their old 80s pbx, WTF!! :) |
| 17:24.42 | l2trace99 | karleeto is there anything in dmesg ? |
| 17:25.39 | hmmhesays | except for the backlight and wideband audio right? |
| 17:27.16 | ManxPower | karleeto: hardware RAID cards, as well as onboard Gigabit Ethernet, and Video Graphics are well known to lock interrupts for long amounts of time, causing issues with Zaptel (and other brands) cards. |
| 17:27.57 | ManxPower | obviously you need to check the common issues like IRQ sharing as well |
| 17:28.23 | karleeto | ManxPower: thats what i was worried about.. but why would it only happen on one line and not the other 3 ? |
| 17:28.47 | ManxPower | karleeto: bizarre things happen with you have IRQ issues. |
| 17:30.37 | *** join/#asterisk SirThomas (n=tomc@mail.kendeco.com) |
| 17:31.05 | grandpapadot | Hey [TK]D-Fender: If you don't mind, take a look at http://pastebin.com/m6e95889f and tell me if you see a reason why none of our phones will update their times ... |
| 17:31.30 | *** join/#asterisk huey23 (n=huey23@64.192.209.34) |
| 17:31.34 | grandpapadot | [TK]D-Fender: It started after the latest firmware update. I have SNTP in a file that is parsed before the main sip.cfg |
| 17:31.53 | ManxPower | grandpapadot: for one thing the config allows DHCP to override the NTP settings |
| 17:32.26 | huey23 | [TK]D-Fender: i've got a new one for ya |
| 17:33.30 | grandpapadot | ManxPower: Thanks -- checking now. |
| 17:33.58 | *** join/#asterisk juxhi (n=juxhi@nathan.epi.usf.edu) |
| 17:35.04 | karleeto | ManxPower: but why would it only happen on one line and not the other 3 lines? does each module on that card get a different IRQ> |
| 17:35.12 | karleeto | ? |
| 17:36.20 | *** join/#asterisk jtknapp (n=skip@65-126-63-1.dia.static.qwest.net) |
| 17:36.30 | *** part/#asterisk jtknapp (n=skip@65-126-63-1.dia.static.qwest.net) |
| 17:36.58 | *** part/#asterisk juxhi (n=juxhi@nathan.epi.usf.edu) |
| 17:42.43 | *** part/#asterisk huey23 (n=huey23@64.192.209.34) |
| 17:44.08 | *** join/#asterisk khronos (n=khronos@aquaman.perryinstitute.org) |
| 17:45.51 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 17:51.48 | *** join/#asterisk syzygyBSD (n=chatzill@66.226.228.204.cpe.speedyquick.net) |
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| 17:57.42 | Katty | mew. |
| 17:58.14 | ManxPower | karleeto: no, but maybe the working three lines were serviced before the interrupt timed out. |
| 17:59.24 | ManxPower | karleeto: I've had issues where calls all sounded fine, unless someone was leaving a voicemail message -- then all the audio went bad - but the recording was fine. It was an IRQ issue. |
| 18:00.19 | ManxPower | it is like trying to beat a train to a crossing. If you are delayed by even 1 second you die. |
| 18:00.36 | ManxPower | "but I didn't die the last 10 time I did it!" |
| 18:03.07 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 18:03.26 | *** join/#asterisk friedrich| (i=friedric@trem-servers.com) |
| 18:06.32 | *** join/#asterisk reber (n=reber@LNeuilly-152-21-126-73.w193-253.abo.wanadoo.fr) |
| 18:12.15 | *** join/#asterisk javb (n=javb@190.80.234.104) |
| 18:12.42 | *** join/#asterisk BadHorsie (i=ospinamo@nat/hp/x-febe09c397f1c332) |
| 18:13.05 | javb | hi all, those someone know an Asterisk Presentation in spanish ? something like to sell asterisk? (advantages... bla bla ) |
| 18:13.35 | *** join/#asterisk angom (n=angom@201.143.89.82) |
| 18:14.02 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 18:18.34 | JRsup1 | ok, this is what I'm finally getting from 'make menuconfig' for asterisk-addons/trunk: http://rafb.net/p/4ryA9f27.html |
| 18:18.48 | JRsup1 | do I have to download the asterisk trunk too? |
| 18:18.50 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 18:19.12 | *** join/#asterisk Sinju (n=sinju@nc-205-240-45-138.sta.embarqhsd.net) |
| 18:21.48 | *** join/#asterisk s0lid (n=jlq@210.213.198.98) |
| 18:24.33 | *** join/#asterisk drwelby (n=mpfister@mail.enplan.com) |
| 18:28.11 | drwelby | Is there a global way to get zap to pause before dialing , instead of using w in the dial command? |
| 18:28.35 | *** join/#asterisk mocker (n=user@198.247.173.227) |
| 18:28.44 | deeperror | wait() |
| 18:28.58 | deeperror | wait(1) |
| 18:29.15 | Aeudian | <PROTECTED> |
| 18:31.08 | drwelby | deeperror: I was hoping to get it to pause from Zapata.conf or something |
| 18:31.52 | drwelby | deeperror: But I guess I could just through a wait in my trunk dialing macro. Thanks! |
| 18:32.20 | *** join/#asterisk msetim (n=marcos@200.195.161.164) |
| 18:33.16 | drwelby | jbot: /s through /u throw |
| 18:33.20 | deeperror | drwelby: is it due to proxy authentication required when dialing out on sip? |
| 18:34.11 | drwelby | deeperror: No, it just seems to sometimes cut off the first digit that needs to be dialed |
| 18:36.57 | deeperror | k |
| 18:40.22 | *** join/#asterisk rsd0991 (n=rsd095@c-71-224-187-182.hsd1.pa.comcast.net) |
| 18:40.27 | *** join/#asterisk wishes (n=wishes@60.234.20.178) |
| 18:41.11 | rsd0991 | quick question......i have certain people that calll me, and i want those numbers to ring straight in instead of going to the IVR. how would i do this? |
| 18:42.05 | grandpapadot | In your default context, exten => 2055551234,1,GoTo(default,801,1) -- 801 being your phone's extension |
| 18:43.48 | grandpapadot | Or just ring the peer directly, exten => 2055551234,1,Dial(801,30,r) |
| 18:44.13 | blitzrage | rsd0991: exten => s/4165551212,n,Macro(stduser,SIP,my_phone) |
| 18:44.26 | grandpapadot | Or what he said, lol |
| 18:48.58 | wishes | exten => 850,1,${TRUNK}/number ? |
| 18:49.29 | blitzrage | sure, you could do that |
| 18:50.10 | rsd0991 | not quite sure i follow that one...... |
| 18:50.28 | blitzrage | that one doesn't solve your problem -- I was just saying it is valid syntax... :) |
| 18:51.38 | rsd0991 | ah |
| 18:52.03 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 18:52.33 | *** join/#asterisk kkn088 (n=kikoun@77.204.209.112) |
| 18:53.25 | *** part/#asterisk karlh626 (n=kharris@IP-66.249.227.5.indigital.net) |
| 18:54.03 | [TK]D-Fender | rsd0991, "show application gotoif" , "show function CALLERID" |
| 18:54.39 | [TK]D-Fender | wishes, and you are sort of missing an APP in there.... nice try though.... |
| 18:56.09 | Aeudian | Is anyone familiar with Flash Operator Panel, I am having diffculty with the drag and drop to transfer phoen calls, when i drag and drop nothing occurs |
| 18:56.26 | *** join/#asterisk UnixDog (n=UnixDog@adsl-69-234-210-145.dsl.irvnca.pacbell.net) |
| 18:56.48 | UnixDog | hey guy I need a way to forword caller id as recieved |
| 18:56.59 | UnixDog | it seems thst cid is not passing |
| 18:57.06 | UnixDog | on anonsip calls |
| 18:57.50 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 18:58.17 | UnixDog | is there a quick way to make sure iit passes the cid |
| 18:58.35 | *** join/#asterisk huey23 (n=huey23@64.192.209.34) |
| 18:58.43 | *** join/#asterisk Schumie (i=SteveWri@cpc2-rdng2-0-0-cust382.winn.cable.ntl.com) |
| 18:59.31 | huey23 | any help is appreciated in finding articles about "unable to find boot server, using existing config" for polycom |
| 18:59.36 | rsd0991 | hmmm....couldn't get any of those suggestions to work....i must be missing something |
| 18:59.46 | UnixDog | huey hold |
| 18:59.53 | huey23 | thanks |
| 19:00.20 | mcab | huey23: do you have a boot server? :-) |
| 19:01.04 | huey23 | lol...yea i think but unsure if it's working...i am unable to get the new software to load because it's unable to find it |
| 19:01.05 | [TK]D-Fender | rsd0991, pastebin what you're DOING, and maybe we'll be able to give you a syntax that'll WORK. |
| 19:01.06 | [TK]D-Fender | ~pb |
| 19:01.07 | jbot | A Pastebin is a place to paste your stuff without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 19:01.08 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^^^^^ |
| 19:02.15 | UnixDog | huey is this a trashbox |
| 19:02.21 | UnixDog | aka trixbox |
| 19:02.25 | [TK]D-Fender | huey23, go TEST your bootserver then. |
| 19:02.31 | huey23 | no...* 1.4 |
| 19:02.45 | jarrod | how does one configure the digium appliance such that a polycom supports more than 2 calls |
| 19:03.02 | rsd0991 | http://pastebin.com/m2ece826e |
| 19:03.14 | [TK]D-Fender | jarrod, you don't.. that up to YOU to configure your PHONES to accept multiple calls. |
| 19:03.21 | [TK]D-Fender | jarrod, this is not *'s job |
| 19:03.30 | jarrod | it autoconfigures the polycom |
| 19:03.37 | jarrod | through the webinterface, so i would disagree |
| 19:03.46 | jarrod | and its not asterisk, its the appliance |
| 19:04.07 | [TK]D-Fender | jarrod, And no, its not the appliance... its the **GUI** that's running on it. |
| 19:04.16 | [TK]D-Fender | jarrod, and you are in the WRONG PLACE to be asking about that. |
| 19:04.17 | rsd0991 | pasted |
| 19:04.28 | jarrod | the appliance incorporates asterisk,the gui, and everything else included |
| 19:04.31 | jarrod | so yes, it is |
| 19:04.46 | [TK]D-Fender | jarrod, read the channel topic. GUI's are not supported here |
| 19:04.58 | jarrod | k, thanks |
| 19:05.21 | [TK]D-Fender | rsd0991, Where do calls come in there? |
| 19:06.33 | rsd0991 | the calls come in from my the outside from my sip provider.... |
| 19:07.13 | *** join/#asterisk reber (n=reber@LNeuilly-152-21-126-73.w193-253.abo.wanadoo.fr) |
| 19:07.21 | [TK]D-Fender | rsd0991, perhaps we should clarify this... where is this IVR that answers your calls CURRENTLY? |
| 19:07.42 | *** join/#asterisk nny (n=scott@64.203.239.83.static-pool-4.pool.hargray.net) |
| 19:07.43 | rsd0991 | in the other room... |
| 19:07.56 | grandpapadot | rsd0991: What context are your inbound SIP calls, from your provider, coming in? It's that context in which you need the "exte => 2055551234,1,do something ..." |
| 19:08.10 | [TK]D-Fender | rsd0991, "other room"? |
| 19:08.14 | nny | working on a definitive howto for voip-info.org for dapper server 6.06 .. anyone know what file modprobe uses to load zaptel modules at boot? |
| 19:08.16 | grandpapadot | oh dear |
| 19:08.23 | rsd0991 | i have a context called sip. |
| 19:08.40 | huey23 | [TK]D-Fender: ok...when you said "TEST" the boot server...i was unsure what you meant...but I was able to ping the phone from the server and the phone answered... |
| 19:08.44 | rsd0991 | and i put that 6105551212,1,xxxx in the sip context |
| 19:08.47 | [TK]D-Fender | rsd0991, show me where your calls come IN <------------ |
| 19:09.01 | grandpapadot | And what do you have it doing? what is xxxx? |
| 19:09.02 | [TK]D-Fender | huey23, How do YOU think you'd test it? |
| 19:09.14 | rsd0991 | outideworkd -> asterisk -> internal extensions |
| 19:09.24 | grandpapadot | Paste what xxxxx is .. |
| 19:09.42 | rsd0991 | i have tried Dial(SIP/Phone1,20,Ttrm) |
| 19:09.43 | [TK]D-Fender | rsd0991, You are completely misunderstanding things. I am now about to assume you have NO functional IVR right now..... |
| 19:10.16 | grandpapadot | rsd0991: Paste your complete sip.conf and extensions.conf in the pastbin (masking any passwords of course). |
| 19:10.39 | [TK]D-Fender | rsd0991, show me a WORKING config (including the CONTEXT header) and pastebin an incoming call with SIP DEBUG enabled so we can see that it's processing the way you think & say it does and we'll tell you how to modify it for what you want to do. |
| 19:10.44 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 19:11.33 | huey23 | [TK]D-Fender: check the firewall and also try an FTP client? that might be what I'd try |
| 19:11.54 | [TK]D-Fender | huey23, I think I see the hamster wheel beginning to turn..... |
| 19:12.20 | huey23 | :P |
| 19:12.57 | UnixDog | tk is there a good document forrabbing caller id and forwading it g |
| 19:13.07 | UnixDog | tk is there a good document for grabbing caller id and forwading it |
| 19:13.24 | UnixDog | sorry this laptop touch pad is pissing me off |
| 19:13.44 | grandpapadot | UnixDog: As long as it's set before the dial cmd, should be sent, but your provider may be doing something to change/override. |
| 19:13.58 | wishes | [TK]D-Fender: heh well i have something similar that works - only it diverts to a macro which does the Dial :D |
| 19:14.17 | UnixDog | the call is coming in anon sip but they are passing cid |
| 19:14.36 | UnixDog | but when it dials back out its not sending the cid |
| 19:14.47 | UnixDog | so I want to try and force it |
| 19:14.50 | [TK]D-Fender | UnixDog, depends what you're dialing out OF <- |
| 19:14.55 | grandpapadot | Have you done a exten => ?,n,Verbose(${CALLERID(all)}) |
| 19:14.58 | grandpapadot | To see what's actually there? |
| 19:17.26 | blitzrage | Verbose() requires an argument |
| 19:17.29 | blitzrage | Verbose(1|foo) |
| 19:17.31 | blitzrage | fyi |
| 19:18.01 | rsd0991 | pasted: http://pastebin.com/mf9429a7 |
| 19:18.45 | grandpapadot | rsd0991: Instead of Dial(4......) you can just use GoTo |
| 19:19.46 | [TK]D-Fender | ok, I can see some people just aren't listening... |
| 19:19.59 | grandpapadot | rsd0991: Try exten => s/5555551212,1,GoTo(4,1) |
| 19:20.37 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 19:22.14 | *** join/#asterisk bkruse (i=bkruse@nat/digium/x-c6de403cd8e248b5) |
| 19:22.25 | rsd0991 | still gets answered by the ivr instead of ringing the phone |
| 19:23.03 | Katty | [TK]D-Fender: did i tell you what i got for my birthday? |
| 19:23.14 | [TK]D-Fender | Katty, Now all you need is a coal stove! |
| 19:23.19 | [TK]D-Fender | ;) |
| 19:23.22 | rsd0991 | could it be the register line in my sip.conf. |
| 19:23.22 | grandpapadot | rsd0991: Where does your IVR start? |
| 19:23.25 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 19:23.34 | Katty | [TK]D-Fender: ^_- mew? |
| 19:23.36 | Katty | [TK]D-Fender: you do not parse. |
| 19:23.47 | rsd0991 | ? |
| 19:24.03 | [TK]D-Fender | Katty, my "reverse psychology" way of saying "a lump of coal?" |
| 19:24.08 | Katty | [TK]D-Fender: oh. |
| 19:24.09 | Katty | [TK]D-Fender: yes. |
| 19:24.10 | grandpapadot | Where in your extensions.conf does what you're calling your IVR start? i.e., exten => 4,1, etc, where does it start? |
| 19:24.17 | Katty | [TK]D-Fender: yes it was...a lump of coal..in a past life. |
| 19:24.25 | rsd0991 | 6050 |
| 19:24.26 | [TK]D-Fender | Katty, And this time 'round? ;) |
| 19:24.33 | Katty | [TK]D-Fender: tis shiny ^_^ |
| 19:24.43 | [TK]D-Fender | Katty, oooohhh a sparkly?!?!?! |
| 19:24.51 | Katty | [TK]D-Fender: yesss, very sparkly ^_^ |
| 19:25.00 | [TK]D-Fender | Katty, was it delivered standing? ;) |
| 19:25.06 | Katty | [TK]D-Fender: ... |
| 19:25.08 | Katty | [TK]D-Fender: earings. |
| 19:25.23 | [TK]D-Fender | Katty, (insert collective sigh here) |
| 19:25.30 | Katty | >.< |
| 19:25.36 | Katty | it's not been a year yet! |
| 19:25.48 | Katty | and we still have some issues to sort out regardless. |
| 19:25.50 | Katty | be patient! |
| 19:25.58 | [TK]D-Fender | Katty, just checking :p |
| 19:26.17 | Katty | i go talking about past life coal lumps, and you go nuts. |
| 19:26.20 | grandpapadot | Where is your s extension? i.e, you're passing calls to your context called "SIP" (I might change that as it could get confusing) and then inside your SIP context, it will hit s,1 first ... |
| 19:26.31 | grandpapadot | s priority, rather, not extension |
| 19:27.09 | Katty | [TK]D-Fender: he walked in me into the jeweler, and directed me to the case with the diamonds in it... |
| 19:27.12 | Katty | [TK]D-Fender: and said, Pick something. |
| 19:27.23 | Katty | [TK]D-Fender: he didn't know what to get :/ |
| 19:27.36 | Katty | [TK]D-Fender: but i'm still happy. |
| 19:27.39 | Aeudian | Katty: i have a question for you, i have redisgned my flash operators panel, but i am having trouble clicking the phones for drag and drop transfers any idea? |
| 19:27.51 | [TK]D-Fender | "s" is NOT a priority! |
| 19:28.02 | Katty | Aeudian: you don't drag the icon, you drag the bubble thing |
| 19:28.04 | [TK]D-Fender | Geez.... its the blind leading the blind today... |
| 19:28.13 | grandpapadot | [TK]D-Fender: You're welcome to jump in ... |
| 19:28.34 | [TK]D-Fender | grandpapadot, Oh I asked for what was needed already and it falls on deaf ears. |
| 19:28.38 | grandpapadot | rsd0991: use this exten => 5555551212,1,Dial(4,30,r) and exten => s,1,GoTo(6050,1) |
| 19:29.02 | Katty | Aeudian: or if you have more details, i could probably give you better advice (= |
| 19:29.16 | [TK]D-Fender | grandpapadot, And you can't dial "4" like that. "4" is not a **TECH** |
| 19:29.41 | grandpapadot | Sorry, i meant: exten => 5555551212,1,GoTo(4,1) and exten => s,1,GoTo(6050,1) |
| 19:29.53 | grandpapadot | Not Dial (I was cut and pasting from his example) |
| 19:30.08 | wwalker | lksjdf |
| 19:30.12 | Katty | gosh. |
| 19:30.23 | Katty | so much GRRR in this room. |
| 19:30.30 | UnixDog | ok |
| 19:30.34 | nny | so is ztdummy recommended for anything other than testing systems without cards? |
| 19:30.38 | Katty | let's all take a minute to think about kittens. |
| 19:30.45 | Aeudian | Katty, it says under the help you drag the the active phone to a none active phone for transfer, and i can and it works but im having the hardest time clicking the phones to drag them even the arrows are hard to double click on my mouse phyically changes to a hand like theres a link to drag but grabing ahold of it is another story lol |
| 19:30.51 | nny | i am working on this howto and wondering if ztdummy should be called in /etc/modules |
| 19:30.55 | UnixDog | is there a simple way to grab the inbound cid and shove it back out a trunk call |
| 19:30.56 | grandpapadot | rsd0991: Point is, every context start with an s extension, it's the default extension, so it has to exist if you blindly send calls to just the context |
| 19:31.08 | Katty | ^_- |
| 19:31.16 | Katty | Aeudian: i think i'd have to see it real time to understand what you're saying |
| 19:31.18 | UnixDog | I need to make sure it sets the caller id tight now it passes unknown |
| 19:31.51 | UnixDog | grandpapadot: I added the lin eit didnot help with trouble shoothing |
| 19:32.05 | grandpapadot | UnixDog: What did it output? |
| 19:32.29 | UnixDog | nothing on the screen |
| 19:32.37 | UnixDog | but the cdr shows the cid |
| 19:32.50 | grandpapadot | Did you "set verbose 3" or something so it shows? |
| 19:32.57 | [TK]D-Fender | nny, ztdummy should ONLY be used if you don't have cards. |
| 19:33.08 | UnixDog | hold |
| 19:33.18 | nny | [TK]D-Fender: roger that |
| 19:33.23 | [TK]D-Fender | <grandpapadot> rsd0991: Point is, every context start with an s extension, it's the default extension, so it has to exist if you blindly send calls to just the context <- ummm... nope |
| 19:33.26 | [TK]D-Fender | ~stdextens |
| 19:33.26 | jbot | it has been said that stdextens is "s" Standard Extension : Where a call goes to when * does not know the destination of the call. Ex : Calls coming in on FXO ports (no DID), a call coming in from an ITSP that doesn't specify or where it was not set in the REGISTER line, or FXS port goes off-hook and "immediate=yes" in zapata.conf. "s" is also used to make IVRs ... |
| 19:34.00 | Katty | jbot: hi |
| 19:34.00 | jbot | hello, katty |
| 19:34.02 | UnixDog | -- Executing [s@macro-dialout-trunk:1] Verbose("SIP/65.111.175.72-08f662e0", "5|"anonymous" <anonymous>") in new stack |
| 19:34.02 | UnixDog | <PROTECTED> |
| 19:34.02 | UnixDog | <PROTECTED> |
| 19:34.04 | nny | [TK]D-Fender: just checking.. the howto for 6.06 on voip-injfo is kind of in need of an update.. any advice on how I should go about that? I was going to create a new article, and link it from the old one, as I don't want to undo what original guy did |
| 19:34.08 | grandpapadot | Ok, standard, not default, my bad. |
| 19:34.09 | [TK]D-Fender | and when you receive a call via SIP, there IS always an exten. You have to be careful of WHAT exten THEY are going to dial. |
| 19:34.19 | UnixDog | so its not sending the cid back out |
| 19:34.45 | UnixDog | so I need to grab the inbound cid and sove it back out the outbound call |
| 19:34.45 | grandpapadot | You can always set the caller id explicitely before your Dial() command |
| 19:34.48 | [TK]D-Fender | grandpapadot, And by "standard" this implies nothing so much about SIP as about * call handling and speical properties. |
| 19:35.06 | UnixDog | this is just a pass threw server |
| 19:35.21 | [TK]D-Fender | UnixDog, I told you very specifically that it depends what you are are calling **OUT** on..... |
| 19:35.33 | rsd0991 | that did the trick.....thanks everyone.... |
| 19:35.39 | UnixDog | sip |
| 19:35.47 | grandpapadot | rsd0991: adding an 's' extension? |
| 19:35.51 | *** join/#asterisk jtexter3 (n=jtexter3@64.149.49.193) |
| 19:35.53 | [TK]D-Fender | UnixDog, wrong answer |
| 19:36.17 | rsd0991 | yeah.... plus i stripped the /6050 off the end of the register statement in my sip.conf file |
| 19:37.52 | nny | [TK]D-Fender: i have wcfxo and zaptel as the two "usual" modules loaded.. i know wcxusb, linpri etc get used, are there any other modules that are commonly running on all asterisk systems? |
| 19:37.58 | nny | libpri* |
| 19:37.59 | UnixDog | ok call comes in anon sip. theysend us a cid number. we then dial back out a sip trunk to the voip provider that is our end point terminator |
| 19:38.24 | [TK]D-Fender | rsd0991, you missed the POINT of it..... that SPECIFIES the exten your provider will SED your calls to. You should have had "exten => 6050,1,SomethingUsefulHere" |
| 19:38.38 | [TK]D-Fender | UnixDog, Depends on your sip trunk. |
| 19:38.49 | linagee | is there a class action lawsuit against gs yet? |
| 19:38.53 | UnixDog | so I need to grab the cid number they send us and set it on the outbound call |
| 19:39.17 | [TK]D-Fender | nny, wcfxo is only for x100p, the usb one = bs... don't need either. |
| 19:39.18 | UnixDog | why sue gs |
| 19:39.20 | linagee | if people get their day in court for bricked iphones, i want my day in court for a bricked grandstream gxp-2000 for their crappy firmware |
| 19:39.22 | ManxPower | linagee: I doubt there ever will be. If you are so cheap as to buy a GS product, I doubt you have the money for a class action. |
| 19:39.31 | *** join/#asterisk Lawbringer (n=Lawbring@212.183.134.64) |
| 19:39.40 | linagee | ManxPower: but there are many people who the "new firmware" destroyed their product too |
| 19:39.58 | ManxPower | linagee: Only the cheap people that use GS products. |
| 19:40.01 | grandpapadot | grandscream |
| 19:40.05 | [TK]D-Fender | ~gs |
| 19:40.06 | jbot | GrandSuck phones & gateways are cheap junk which should be avoided with extreme prejudice. |
| 19:40.07 | [TK]D-Fender | ^^^^^^^^^^^^^ |
| 19:40.07 | linagee | ManxPower: only cheap people would buy an iphone |
| 19:40.11 | UnixDog | the gxp2000/2020 work great |
| 19:40.11 | mocker | I had someone at my user group recommend grandstream. |
| 19:40.12 | [TK]D-Fender | ~grandstream |
| 19:40.13 | jbot | i guess grandstream is the Yugo of VoIP hardware. Run. Run away now. |
| 19:40.18 | mocker | I cried a little. |
| 19:40.20 | linagee | ManxPower: i bought one better than an iphone that actually has HSDPA unlike the cheap iphone crap |
| 19:40.21 | UnixDog | the 100/101/102 are crap |
| 19:40.26 | [TK]D-Fender | mocker, draw & quarter them |
| 19:40.36 | linagee | [TK]D-Fender: does mocker include mock? |
| 19:41.09 | mocker | What a weird question. |
| 19:41.13 | linagee | mocker: grandstream works great until you upgrade the firmware and it becomes a piece of crap that doesn't even boot 50% of the time. (seriously) |
| 19:41.22 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 19:41.26 | linagee | mocker: and then you can't downgrade the firmware either |
| 19:41.29 | mocker | linagee: I've just heard too many complaints to risk my reputation on them. |
| 19:41.32 | [TK]D-Fender | linagee : would "linagee"'s 'inclusion' of "age" imply any sense of maturity? ;) |
| 19:41.36 | UnixDog | so I take it I am not going to get a answer |
| 19:42.03 | [TK]D-Fender | UnixDog, I've already answered you. |
| 19:42.07 | UnixDog | I cant find a page for 1.4 asterisk on grabbing cid and setting it back out the trunk |
| 19:42.10 | UnixDog | where |
| 19:42.25 | [TK]D-Fender | UnixDog, read up and PAY ATTENTION |
| 19:42.47 | mocker | UnixDog: Just look at callerid variables.. |
| 19:42.48 | UnixDog | I scrool up and see nothing |
| 19:42.50 | linagee | mocker: so what do we have here then? grandstream works great if you're a home user on a budget, buy things that are crappy instead of $5 more for a 10x better model, people who NEVER upgrade firmwares |
| 19:43.06 | [TK]D-Fender | UnixDog, Because what you are asking for DOESN'T EXIST. CID is passed unless the channel you are calling out on either doesn't SUPPORT it, or IGNORES it. |
| 19:43.19 | [TK]D-Fender | <[TK]D-Fender> UnixDog, Depends on your sip trunk. <--------------------------------- |
| 19:43.27 | [TK]D-Fender | UnixDog, I've said ti TWICE already. |
| 19:43.38 | ManxPower | UnixDog: For one thing you should not need to "grab the CID and send it back out the trunk" A call that is dialed should continue to have the same callerid as the incoming call, subject to the "o" (I think) Dial option |
| 19:43.40 | [TK]D-Fender | [TK]D-Fender> UnixDog, I told you very specifically that it depends what you are are calling **OUT** on..... |
| 19:44.06 | nny | [TK]D-Fender: ok.. so i assume as my dmesg shows my card loaded (a tdm02b), that wctdm, etc. is implied when loading zaptel.. what I can't figure out is why my dev box shows the module "in use" by 2 on my stable box, but 0 on my dev box here. |
| 19:44.21 | ManxPower | [TK]D-Fender: I think UnixDog is in a "won't listen" mode today. If that is the case, no amount of help will be effective. |
| 19:44.26 | nny | [TK]D-Fender: i assume it is a misnomer, but trying to be thorough for my howto |
| 19:44.43 | [TK]D-Fender | UnixDog, and "a sip truck" doesn't mean ANYTHING. Maybe they don't care what you send them. Maybe the way you set up your SIP peer focibly overrides the current channels CID. But I am not psychic you know... |
| 19:44.50 | mocker | linagee: I think the grandstreams have more problems than just the firmware. |
| 19:45.05 | mocker | I've heard of just generally poor quality hardware. |
| 19:45.21 | ManxPower | Saying "SIP trunk" is like saying "The internet is down". It makes people think you rode the short bus to school. |
| 19:45.25 | nny | [TK]D-Fender: alternatively, I see you have your hands full with canines,... so I can just piddle around and see what crappens ^^ |
| 19:45.42 | [TK]D-Fender | UnixDog, stop talking about a "sip trunk" like its an entity that only has 1 fixed set of properties. It doesn't. |
| 19:45.57 | ManxPower | My usual response to "the internet is down" is "Even in China?" |
| 19:46.08 | UnixDog | the provider allows us to set the cid |
| 19:46.18 | mocker | ManxPower: Really? My response is usually, 'Break time!' |
| 19:46.23 | mocker | :P |
| 19:46.31 | ManxPower | UnixDog: good. Now you need to figure out WHY the original callerid is not being passed thru to the outbound leg of the call. |
| 19:46.51 | mocker | UnixDog: NoOp should help w/ that. |
| 19:47.06 | ManxPower | mocker: He's not listening anyway. |
| 19:47.07 | Yourname`` | What are the minimum requirements for Asterisk? |
| 19:47.24 | mocker | Yourname``: I'm running it on an NSLU2, if that tells you anything. |
| 19:47.32 | nny | Yourname``: the book has some good guidelines |
| 19:47.38 | [TK]D-Fender | UnixDog, you should know there's no way I trust you set your peer up right..... Also, please not that your anonymous call's CID **NUMBER** says "anonymous". Maybe, just MAYBE they don't LIKE that and are refusing it... (I can't imagine why....) |
| 19:47.40 | ManxPower | Yourname``: My min requirements for Asterisk is a 1 Ghz system, 1 GB of RAM, Sangoma A102 card, and a PRI. What are yours? |
| 19:48.07 | nny | ManxPower: we use the VIA 1 ghz fanless... good stuff |
| 19:48.11 | Yourname`` | Nothing, really. I'm trying to install my home Asterisk on an http://developer.novell.com/yes/62760.htm. |
| 19:48.21 | Yourname`` | Or.. IBM i-Series Thinkpad Type 1161-44U |
| 19:48.49 | Yourname`` | ..which has 65mb RAM. |
| 19:48.56 | Yourname`` | Can't even install CentOS on it, heh |
| 19:49.19 | nny | Yourname``: needs more cowbell |
| 19:50.06 | nny | or ram |
| 19:50.58 | Yourname`` | Yeah.. |
| 19:51.12 | nny | Yourname``: whats the maximum ram that beast supports? |
| 19:51.12 | [TK]D-Fender | Yourname``, I see you've already custom-built "chan_masochism.so".... |
| 19:51.17 | nny | lol |
| 19:51.21 | Yourname`` | lol |
| 19:51.40 | [TK]D-Fender | Yourname``, ... was it good for you? |
| 19:51.41 | patrickv0x | <PROTECTED> |
| 19:51.46 | Yourname`` | LOL yes sir |
| 19:52.00 | Yourname`` | It was just a thought anyway. Was wondering what I wanted to do with this laptop.. :( |
| 19:52.19 | ManxPower | Yourname``: how were you going to plug a Digium card into that machine? |
| 19:52.21 | Yourname`` | Maybe, just maybe I could use Gateway Solo 5300 for it. -> http://tinyurl.com/2djd8a |
| 19:52.24 | [TK]D-Fender | Yourname``, Frisbee!!!! |
| 19:52.50 | nny | so wait they actually make a usb zaptel card,.. oh good lord what fun that thing must be |
| 19:53.02 | ManxPower | nny: they do not make a USB Zaptel card. |
| 19:53.12 | Yourname`` | Yeah, good weather for frisbee too. 32degrees! |
| 19:53.15 | ManxPower | Hence my question. |
| 19:53.18 | mocker | Yourname``: I have asterisk installed on http://www.amazon.com/Linksys-Storage-Link-Drives-NSLU2/dp/B0001FSCZO |
| 19:53.23 | mocker | Don't give up hope. |
| 19:53.25 | nny | ManxPower: thank god |
| 19:53.29 | mocker | Just don't expect to take a bunch of calls either. :)] |
| 19:53.31 | nny | lol |
| 19:53.33 | Yourname`` | ManxPower: I was going to use a Linksys Sipura that [TK]D-Fender suggested long ago! :) |
| 19:53.37 | Aeudian | patrickv0x: i did it a while back with a 7960 cisco acctually had pretty good documentation on it. |
| 19:53.41 | [TK]D-Fender | Yourname``, You can buy a suitable PC for < $100 |
| 19:53.46 | ManxPower | Who in their right mind would have a VoIP-only PBX anyway? |
| 19:53.47 | nny | anyone tried to install on ddwrt hardware? |
| 19:54.06 | [TK]D-Fender | ManxPower, Ummm.... I do :) |
| 19:54.15 | Yourname`` | [TK]D-Fender: I know, just trying with laptops though. Make it wireless and make it sit in the washroom attic or something so that's like a forgotten server. |
| 19:54.20 | nny | i imagine cid and whatnot won't run on the smaller hardware |
| 19:54.21 | [TK]D-Fender | ManxPower, As for as my home is concerned :) |
| 19:54.24 | Yourname`` | ManxPower: I would.. :D |
| 19:54.47 | ManxPower | When I had a home Asterisk server, I didn't even run that VoIP only. |
| 19:54.59 | [TK]D-Fender | Yourname``, If its buried in your attic, doesn't matter WHAT it is.. you still can't SEE it! |
| 19:55.18 | [TK]D-Fender | ManxPower, Yeah, but how reliable is your internet at home? |
| 19:55.23 | Yourname`` | I know, but still.. space is of essence, my friend! |
| 19:55.46 | ManxPower | [TK]D-Fender: at the time, reasonably reliable. |
| 19:55.47 | [TK]D-Fender | Yourname``, then clearly you need to forget all that and buy the AA50! |
| 19:55.50 | nny | Yourname``: mount it on the wall, flip the screen and make it an asterisk phone server and picture frame |
| 19:55.57 | nny | and hang your coat from it |
| 19:56.00 | nny | :) |
| 19:56.11 | Yourname`` | haha |
| 19:56.24 | Yourname`` | [TK]D-Fender: For my HOME? an AA50? haha, that's some digium thing, right? |
| 19:56.32 | [TK]D-Fender | ManxPower, well mines up solid pretty much year-round. I trunk through my office and have no need for anything else. Access to a full PRI w/ free LD from home = joy. |
| 19:56.38 | [TK]D-Fender | Yourname``, Indeed. |
| 19:56.49 | nny | I have to carry the packets from myhouse to the ISP |
| 19:56.53 | Yourname`` | No, thank you. I like doing unconventional things. :D |
| 19:56.55 | nny | up hills, both ways |
| 19:57.02 | [TK]D-Fender | Yourname``, But space is of ESSENCE! You can't afford the bulk of a laptop... the AA50 is MUCh smaller! |
| 19:57.04 | ManxPower | [TK]D-Fender: I liked having real e911 service |
| 19:57.23 | ManxPower | Since I still needed a phone line for DSL and wanted a timing source. |
| 19:57.25 | Yourname`` | [TK]D-Fender: But much more expenseeeve. |
| 19:57.31 | [TK]D-Fender | ManxPower, I have a cell phone and don't have to give a rats ass about anything :) |
| 19:57.47 | ManxPower | My cell phone is less reliable than Asterisk. |
| 19:57.58 | grandpapadot | ManxPower: Welcome to AT&T |
| 19:58.00 | nny | gah.. memory failing.. 64k not enough.. is it "zap show channels"? |
| 19:58.05 | tzafrir | well, a DIY type might prefer http://ucpbx.com/ . Lower price, no support :-) |
| 19:58.10 | ManxPower | grandpapadot: AT&T bought Verizon? |
| 19:58.18 | grandpapadot | lol, they all suck |
| 19:58.25 | [TK]D-Fender | ManxPower, See your rose-coloured glasses have been permanently stained... you should seek employement at http://www.despair.com |
| 19:58.29 | *** join/#asterisk bantu (n=Miranda@p54A31F90.dip0.t-ipconnect.de) |
| 19:58.29 | Yourname`` | Is that Asterisk, tzafrir? |
| 19:58.30 | ManxPower | Verizon is the only company with service at my location |
| 19:58.42 | tzafrir | Yourname``, yes |
| 19:58.47 | mocker | tzafrir: I'm telling ya, NSLU2. :) |
| 19:58.57 | mocker | ~$80 for new in box. |
| 19:59.12 | Yourname`` | I'd think expensive, again. These laptops are just sitting around at home doing nothing.. so I just wanted to put them to good use. |
| 19:59.13 | *** join/#asterisk jgomo3 (n=jgomez@200.72.223.2) |
| 19:59.16 | [TK]D-Fender | ManxPower, CDMA++ :) |
| 19:59.32 | [TK]D-Fender | ManxPower, if only the OpenMoko was supporting it.... |
| 19:59.55 | nny | tzafrir you get astfin working proper on that hardware yet? |
| 20:00.17 | ManxPower | [TK]D-Fender: I will never be on a cell phone contract again. I am STILL fighting with verizon after 2 years |
| 20:00.30 | ManxPower | I do prepay |
| 20:00.49 | Yourname`` | ManxPower: Go Helio. I wish Helio was here in Canada. |
| 20:01.16 | ManxPower | I'm at the point now where I have to request records from the bank to send to the collection company proving that I sent verizon the money -- they lost the original payments and it is not my problem. |
| 20:01.35 | ManxPower | Yourname``: what part of "verizon is the only company with service at my location" was unclear? |
| 20:01.55 | *** join/#asterisk Buhntz (n=bytewalk@port-212-202-42-40.dynamic.qsc.de) |
| 20:02.13 | [TK]D-Fender | ManxPower, Cell co's up here aren't as nasty.... in fact NONE of our organizations are. USA Light! Diet USA! Half the BS! |
| 20:02.13 | Yourname`` | I think Helio piggyback's on Verizon. But handles everything else. :) |
| 20:02.22 | ManxPower | Yes, I understand that you people that live near a city find it hard to undertstand there are places there there is only 1 carrier. |
| 20:02.52 | mocker | ManxPower: There are places outside of cities? |
| 20:03.25 | Yourname`` | LOL |
| 20:04.50 | ManxPower | Cingular (TDMA and GSM), AT&T (before the sale to SBC), Sprint, Nextel, T-Mobile, Boost all have no signal where I live. |
| 20:05.49 | Yourname`` | Hence, Helio. |
| 20:07.08 | ManxPower | Yourname``: they are $15 more per month than verizon for similar plan |
| 20:07.24 | ManxPower | Yourname``: I assume they are a prepay company |
| 20:07.44 | Strom_M | Yourname``: Helio is an MVNO anyway |
| 20:07.50 | Yourname`` | Yes, and awesome too. I'd think they will give you better plans when you call them telling them about your Verizon. |
| 20:08.00 | Yourname`` | Strom_M: MVNO with Verizon.. |
| 20:08.20 | Strom_M | Yourname``: ZZT WRONG |
| 20:08.21 | Strom_M | Sprint |
| 20:08.31 | Yourname`` | WHAT!? |
| 20:08.41 | ManxPower | Almost all MVNOs are on Sprint |
| 20:08.57 | ManxPower | Strom_M: than you for saving me untold headaches |
| 20:09.15 | Strom_M | any time |
| 20:09.15 | *** part/#asterisk huey23 (n=huey23@64.192.209.34) |
| 20:09.33 | Yourname`` | LOL |
| 20:09.39 | ManxPower | Strom_M: do you know of any MVNOs on Verizon. |
| 20:10.05 | Strom_M | "Amp'd Mobile" IIRC |
| 20:10.10 | Strom_M | but it looks like they're out of business now |
| 20:10.13 | Yourname`` | Wasn't AMPd on it? |
| 20:10.26 | Yourname`` | Yeah, lol. They bankrupted in July, I think. |
| 20:10.28 | nny | anyone here know if i want to backup all asterisk files, and be able to tar them "back" to their proper locations, what is missing from tar -cpf asterisk.tar /etc/zaptel.conf /etc/asterisk /var/lib/asterisk ?? |
| 20:10.43 | nny | it works, but untars them all to the dir that i untar from |
| 20:10.47 | file | page plus cellular is one that people like because they use Verizon |
| 20:11.24 | CrazyTux | Hey guys, I use a database for * mailboxes, how would I go about changing the "voicemail notification" email template? |
| 20:11.26 | Yourname`` | I don't like that botox'ed girl on pageplus' website. |
| 20:11.38 | *** join/#asterisk scurb_w (n=scurb@194.218.238.2) |
| 20:11.58 | scurb_w | Hello, anyone know the status of video handling in meetme? |
| 20:12.16 | file | http://www.uglyeric.com/ will activate a phone on them for $5.99, includes 100 mins |
| 20:12.39 | nny | i assumed -P would archive them with the leading /, so they untarred proper, but no love |
| 20:18.00 | karleeto | i'm having programs finding finding documentation to remap some keys on my polycom 501 |
| 20:19.15 | hmmhesays | ok I set iax2 set debug jb |
| 20:19.18 | hmmhesays | and I get this now vvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvv |
| 20:19.19 | hmmhesays | on the cli |
| 20:19.28 | hmmhesays | and a whole lotta choppy voice |
| 20:19.43 | karleeto | i just want to remap 2 keys, one just needs to dial *51,send and the other just needs to add a 0 infrom of what ever extention they typed before they pressed it and the send |
| 20:20.55 | karleeto | can anyone help me with this? or point me in the right direction? i've got the admin guide, the use guide, all sip.cfg has is a setting for scrolling.timeout |
| 20:21.17 | siragon | hi all |
| 20:21.24 | Katty | hi. |
| 20:21.36 | hmmhesays | Katty question for you |
| 20:21.45 | hmmhesays | does the ip 501 have a 2.5mm headphone jack on it? |
| 20:21.46 | Katty | k |
| 20:21.49 | Katty | no |
| 20:21.56 | ManxPower | karleeto: sadly, the best docs for that might be the Wiki |
| 20:21.59 | hmmhesays | they have the rj9 right? |
| 20:22.12 | siragon | can any of you point to an URL where I can find info about using Polycom 501 and XML pages (for feeding stock prices and things like that) |
| 20:22.23 | ManxPower | siragon: did you check the Wiki |
| 20:22.26 | Katty | hmmhesays: yes'r |
| 20:22.31 | ManxPower | or the mailing list archives? |
| 20:22.46 | karleeto | ManxPower: it give me all the keys and corresponding numbers, but doesnt really show me how to do it! |
| 20:23.02 | syzygyBSD | Polycom stock prices -> http://finance.google.com/finance?q=NASDAQ%3APLCM |
| 20:23.07 | mocker | http://www.voip-info.org/wiki/view/Polycom+Microbrowser |
| 20:23.08 | ManxPower | karleeto: that is one of the hardest things to do on the polycoms |
| 20:23.17 | siragon | ManxPower: I looked in google but when you search polycom xml and so on there are many many documents about conf files |
| 20:23.18 | Katty | hmmhesays: we bought some bluetooth headsets for ours. |
| 20:23.24 | *** join/#asterisk zerohalo (n=zeroHalo@pool-72-70-79-233.bstnma.east.verizon.net) |
| 20:23.41 | mocker | I have my polycoms showing the weather.. |
| 20:23.49 | Katty | freak. |
| 20:23.50 | mocker | And a company logo, etc.. |
| 20:23.52 | ManxPower | karleeto: also, why not just put a 0 in front of the callerid before the call gets sent to the phone? |
| 20:23.59 | jgomo3 | Hello. Any body here had work with jiaxclient? |
| 20:24.06 | karleeto | mocker: ever customized any keys? |
| 20:24.12 | siragon | mocker: which model of polycom? |
| 20:24.12 | mocker | karleeto: Nope. |
| 20:24.25 | [TK]D-Fender | hmmhesays, you're in IP 320/330 range now. |
| 20:24.26 | Katty | jgomo3: not me. |
| 20:24.28 | siragon | I already have my company logo |
| 20:24.29 | mocker | siragon: 601 |
| 20:24.42 | siragon | mocker: I only have 430 and 501 |
| 20:24.44 | karleeto | ManxPower: 0 before the extension makes the extension auto-answer for intercom/page |
| 20:24.50 | jgomo3 | siragon: are you from Venezuela? |
| 20:24.57 | siragon | jgomo3: yes |
| 20:25.09 | jgomo3 | siragon: me too? greetings |
| 20:25.27 | siragon | jgomo3: greetings! |
| 20:25.28 | jgomo3 | siragon: me too, greetings * |
| 20:25.50 | [TK]D-Fender | siragon, for MicroBrowser info, check the WIKI.. they have some great pages on it. |
| 20:25.53 | [TK]D-Fender | ~wikis |
| 20:25.53 | jbot | i guess wikis is http://www.voip-info.org |
| 20:26.00 | *** join/#asterisk [hC] (n=hardcore@S01060016b61c8983.vf.shawcable.net) |
| 20:26.18 | JRsup1 | Ok, I'm getting nowhere really slowly. Can anyone point me in a good direction for setting up and using asterisk-addons/trunk from SVN? I don't care if I have to reload the machine with whatever. I'd like to keep some sort of web configuration/admin tool for asterisk as well. |
| 20:26.33 | karleeto | [TK]D-Fender: you are the polycom man! help me customize a key |
| 20:26.35 | hmmhesays | Katty Link |
| 20:26.35 | siragon | ok.., I see |
| 20:26.40 | *** join/#asterisk ellisdee (n=mayfield@72.16.242.51) |
| 20:26.54 | karleeto | [TK]D-Fender: when i figure it out, i'll write a doc on the wiki |
| 20:26.57 | [TK]D-Fender | karleeto, like how? |
| 20:28.06 | ManxPower | karleeto: in most places other than the USA/Canada "0" means "outside line" or "toll call" |
| 20:28.38 | hmmhesays | I got some people looking for a decent headset for an ip 501 |
| 20:28.56 | ManxPower | JRsup1: since trunk is the developement version of Asterisk (which is frequently doesn't even compile), not many people use it here. |
| 20:29.00 | [TK]D-Fender | hmmhesays, Plantronics H261 binaural + M22 Amp |
| 20:29.20 | karleeto | [TK]D-Fender: i want to change the services key (#31 according to admin guide) to dial *51,send; thats it |
| 20:29.51 | [TK]D-Fender | karleeto, don't thikn you can do that... |
| 20:29.58 | [TK]D-Fender | karleeto, And why would you disable the MB like that? |
| 20:30.31 | hmmhesays | bout 150 for that rig |
| 20:30.32 | [TK]D-Fender | karleeto, You can't map full sequences. And what is *51 for? |
| 20:30.43 | [TK]D-Fender | hmmhesays, thats life... |
| 20:30.48 | [TK]D-Fender | hmmhesays, Quality costs |
| 20:31.00 | ManxPower | karleeto: the easiest way to to just set one of your line appearances to be a speed dial. |
| 20:31.10 | [TK]D-Fender | hmmhesays, for an inexpensive solution look at IP320/330 + 2.55mm |
| 20:31.21 | [TK]D-Fender | karleeto, indeed as stated by ManxPower |
| 20:31.30 | grandpapadot | That 330 is a great phone for the price. |
| 20:31.33 | karleeto | [TK]D-Fender: ok, how do i set key 31 to dial a speeddial that is *51 ? |
| 20:31.45 | hmmhesays | yeah, they need someting that is amped more, does the m22 get pretty loud? |
| 20:31.53 | [TK]D-Fender | karleeto, What is the point of that? |
| 20:32.03 | karleeto | [TK]D-Fender: *51 is our overhead paging sysyrm |
| 20:32.09 | *** join/#asterisk mitcheloc (n=mitchel@adsl-67-127-235-238.dsl.irvnca.pacbell.net) |
| 20:32.12 | [TK]D-Fender | hmmhesays, the M22 will handle ANYTHING that is remotely sane. |
| 20:32.14 | ManxPower | karleeto: you don't. the first speed dial entry will appear on the first unconfigured line appearance |
| 20:32.34 | [TK]D-Fender | karleeto, I think there was a way for you to map to a speed-dial.... |
| 20:32.53 | [TK]D-Fender | karleeto, I'd have to look it up... but you're better off just using a line appearance. |
| 20:32.59 | [TK]D-Fender | (key) |
| 20:33.01 | ManxPower | karleeto: what you want to do is very difficult -- maybe impossible. My suggestion is something that can be done fairly easi.y. |
| 20:33.02 | karleeto | [TK]D-Fender: the customers dont want to push like 4 keys to intercom where they used to press 1 with their 1980s system |
| 20:33.21 | [TK]D-Fender | karleeto, using a line-key WOULD eb a 1-press solution. |
| 20:33.30 | grandpapadot | karleeto: It's funny how many businesses get stuck on just that key system feature (intercom) |
| 20:33.34 | karleeto | [TK]D-Fender: how do i do that ? |
| 20:33.44 | *** join/#asterisk xlyz (n=xz@host-84-223-120-104.cust-adsl.tiscali.it) |
| 20:33.45 | ManxPower | karleeto: read the admin guide |
| 20:33.49 | [TK]D-Fender | karleeto, funny I sued to do "[Function] 610" on my Norstars..... 4 keys ;) |
| 20:33.53 | ManxPower | karleeto: look for the info on setting speed dial stuff |
| 20:34.00 | karleeto | OK, thanks! |
| 20:34.19 | [TK]D-Fender | karleeto, don't map all your line-keys to call appearances and your directory will spill onto the empty ones in "speed dial" order. |
| 20:35.08 | karleeto | so should i change num line keys to 2, and map the 3rd to intercom ? |
| 20:35.42 | ManxPower | karleeto: no, just leave the last line appearance with no config |
| 20:35.48 | *** part/#asterisk xlyz (n=xz@host-84-223-120-104.cust-adsl.tiscali.it) |
| 20:36.06 | ManxPower | the first entry in the speed dial list will MAGICALLY (they use voip pixies) appear on the line |
| 20:36.57 | hmmhesays | [TK]D-Fender thanks |
| 20:37.02 | hmmhesays | now onto my iax 2 question |
| 20:37.44 | hmmhesays | i turn the iax2 jitter buffer on I get vvvvvvvvvvvvvvvvvLLLLLLLLLLLLLL screaming across my screen |
| 20:38.11 | hmmhesays | and some choppy audio |
| 20:39.01 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 20:39.43 | ManxPower | hmmhesays: I never ever had IAX2 jitter buffer work well for me. |
| 20:40.19 | hmmhesays | yeah, I'm trying everything I can think of to make this work ok, but it seems iax2 is just not working well |
| 20:40.27 | karleeto | ManxPower: ok, so add *51 in 0000000000-directory.conf ? |
| 20:40.27 | hmmhesays | the same scenario with sip works fine |
| 20:40.58 | *** join/#asterisk errr (n=errr@fedora/errr) |
| 20:41.07 | ManxPower | hmmhesays: we switched to SIP (which does not have a jitter buffer) because we had long audio dropouts on IAX2 and built QoS on our network. |
| 20:41.18 | ManxPower | karleeto: try it. |
| 20:41.41 | ManxPower | I should say "SIP (really RTP) does not have a jitter buffer in 1.2 and 1.0" |
| 20:42.50 | hmmhesays | yeah this is terminating to voipjet I think they are having some issues |
| 20:43.23 | ManxPower | hmmhesays: our IAX2 issues were between Asterisk servers on the corporate WAN with QoS between them. |
| 20:43.28 | *** join/#asterisk sporecook (n=wdjdj@72.37.252.50) |
| 20:44.44 | ManxPower | After 6 months I finally gave up on IAX2. The customer was ready to go back to their owl PBX |
| 20:44.49 | ManxPower | and their old PBX too. |
| 20:45.03 | russellb | iax2 in 1.4 should be many many times better |
| 20:45.15 | *** join/#asterisk ToTo (n=ToTo@host75-142-dynamic.8-87-r.retail.telecomitalia.it) |
| 20:45.39 | ManxPower | russellb: Perhaps. IAX2 in 1.2 was supposed to be good too. |
| 20:46.17 | ManxPower | And I doubt hmmhesays can force voipjet to upgrade to the latest 1.4.x |
| 20:46.32 | hmmhesays | no being they built their network on iax2 |
| 20:47.24 | *** part/#asterisk deeperror (n=deeperro@69-215-202-195.ded.ameritech.net) |
| 20:47.42 | hmmhesays | and that is all they use |
| 20:49.00 | *** join/#asterisk eBody (n=icechat5@207.71.51.162) |
| 20:49.35 | eBody | anyone ever find out what the following means in the asterisk logs: "Stopping retransmissions on..." |
| 20:50.05 | smash- | Hey anyone here heard of Fonality? |
| 20:50.20 | [hC] | fowho? |
| 20:50.21 | eBody | yeah |
| 20:50.22 | lirakis | smash: are you serious? |
| 20:50.22 | [hC] | :) |
| 20:50.25 | mocker | smash-: No, are they involved in Asterisk? |
| 20:50.26 | mocker | :) |
| 20:50.32 | eBody | lol |
| 20:50.50 | smash- | We'll yeah im serious im just looking for possible customer feedback |
| 20:51.03 | smash- | and yes they are involved with asterisk |
| 20:51.21 | russellb | actually, they're not involved in asterisk at all |
| 20:51.29 | putnopvut | uh oh... |
| 20:51.37 | russellb | not in any sort of real development anyway |
| 20:51.43 | russellb | just packaging and distribution |
| 20:51.52 | mocker | Ohsnap, we've got a developer in on it. |
| 20:52.14 | mocker | :P |
| 20:52.17 | smash- | huh? |
| 20:52.36 | mocker | smash-: They make TrixBox, which is a GUI for Asterisk. |
| 20:52.39 | eBody | smash what r u trying to do? |
| 20:52.48 | russellb | mocker: trixbox isn't a gui, either |
| 20:52.55 | russellb | mocker: the gui is another thing they just distribute, and don't make |
| 20:53.28 | smash- | I just order'd a phone system through fonality and i was just a little curious about what freedom they give with asterisk |
| 20:53.30 | eBody | dang if we're being all technical: trixbox is the total package which includes freePBX this is a GUI for asterisk |
| 20:53.36 | smash- | i guess its asterisk on centos |
| 20:53.40 | russellb | eBody: there we go ;) |
| 20:53.49 | ManxPower | Our CLEC merged with another CLEC and are now known a "XFone", which is a good name if you consider their customer service after the merger. |
| 20:54.05 | russellb | i just don't like people getting credit they don't deserve, IMO |
| 20:54.08 | file | let's get technical technical! |
| 20:54.19 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 20:54.23 | ManxPower | smash-: you don't want to ask for help about Trixbox here. |
| 20:54.44 | eBody | i know there's a freepbx channel, is there a trixbox? |
| 20:55.08 | elixer | yes, there is |
| 20:55.08 | ManxPower | %50 bad packets on the MLPPP main internet connection thru the CLEC and they send out a channel bank guy to fix it. |
| 20:55.48 | eBody | anyone use Grandstream phones? |
| 20:56.14 | ManxPower | eBody: nobody sane |
| 20:56.15 | grandpapadot | ~gs |
| 20:56.16 | jbot | GrandSuck phones & gateways are cheap junk which should be avoided with extreme prejudice. |
| 20:56.27 | eBody | dangit! why not ManxPower |
| 20:56.33 | grandpapadot | Ha! I won. |
| 20:57.23 | ManxPower | It sure would be nice if Digium simply said "Grandstream products are not supported by Asterisk." |
| 20:57.45 | eBody | that's all jack'd up. lol. but grandstream sure does put a big asterisk logo on all their media |
| 20:57.49 | `Sauron | Are they not supported? |
| 20:57.58 | `Sauron | Or are they just significantly pain in the ass to support? |
| 20:58.10 | eBody | i wasn't having a problem with our gxp-2000 until up upgraded our service. |
| 20:58.12 | grandpapadot | They are just junk... |
| 20:58.19 | grandpapadot | We call them grandscream |
| 20:58.30 | [TK]D-Fender | ~gs |
| 20:58.31 | jbot | GrandSuck phones & gateways are cheap junk which should be avoided with extreme prejudice. |
| 20:58.32 | [TK]D-Fender | ^^^^^^^^^^^^^^^ |
| 20:58.37 | *** part/#asterisk lirakis (n=lirakis@65.200.191.253) |
| 21:01.58 | ManxPower | `Sauron: Older grandstream products sucked from both a hardware and software standpoint (their phone that supported CallerID only supported callerid NUMBER, not name). Their new products are not as bad for hardware, but they can't make a stable firmware no matter how hard they try. |
| 21:02.29 | ManxPower | many people that have used higher end products say the GS sound quality is terrible. |
| 21:02.41 | nny | so next question from me |
| 21:02.53 | nny | is running asterisk as root "BAD!" or "Not recommended"? |
| 21:03.04 | [TK]D-Fender | nny, "not recommended" |
| 21:03.06 | grandpapadot | nny: Running any daemon as root is bad. |
| 21:03.10 | nny | yeah knew that |
| 21:03.25 | nny | i'll look into setting it up as it's own user |
| 21:03.37 | ManxPower | nny: it is perfectly fine as long as you understand that if someone finds a bug in Asterisk that allows them remote access, they will have whatever permissions the user asterisk is running as has. |
| 21:03.41 | grandpapadot | nny: let me guess, you compiled from source and then ran it for the first time as root without specifying a username/pass and now all your permissions are messed up and it will only run as root? |
| 21:03.53 | nny | grandpapadot: nope not even close, here's a cookie |
| 21:04.01 | jcanfield | Realistically...what is the physical/software limit to the number of T1's on a single * box? |
| 21:04.03 | grandpapadot | lol |
| 21:04.28 | nny | ManxPower: yeah i do. grandpapadot, nah i am writing a howto for ubuntu 6.06 server and defining basic building instructions for our company |
| 21:04.33 | ManxPower | jcanfield: Digium had 8 T-1's in the same box using older cards. |
| 21:05.08 | nny | by default, when compiling, it just assumes root.. most of these boxes don't even have gateway IPs on them, and require a login via VPN to get to them |
| 21:05.16 | ManxPower | Even with the newer cards I would not put more than 8 t-1/e-1s in a box. |
| 21:05.54 | nny | but i understand the risk... might as well run it as asterisk and do the white thing |
| 21:06.01 | ManxPower | jcanfield: of course this is assuming no transcoding |
| 21:06.17 | pigpen | anyone use digium cards with vmware esx server? |
| 21:06.34 | ManxPower | pigpen: only people that don't know what they are doing |
| 21:06.42 | jcanfield | ManxPower: So I shouldn't expect problems getting a quad card and using three T1's? |
| 21:06.42 | grandpapadot | This is a good how-to and will most likely work with 6.06 -> http://www.itinfusion.ca/asterisk/howto-installing-asterisk-on-debian-etch/ |
| 21:07.06 | ManxPower | jcanfield: not unless you have a very slow system or are using HPEC or G729 |
| 21:07.08 | pigpen | ManxPower, yeah...kinda what I was thinking... |
| 21:07.19 | nny | grandpapadot: not bad, but still has some errors |
| 21:07.43 | jcanfield | ManxPower: Nope, will be a new quad core machine. |
| 21:07.46 | nny | grandpapadot: going to add my own to voip-info.org, with some changes |
| 21:07.59 | tzafrir | grandpapadot, I'm not sure I'd recommend 6.06 . It hadn't worked that well before. A newer ubuntu might work better |
| 21:08.13 | nny | tzafrir works well here |
| 21:08.24 | grandpapadot | tzafrir: We run a 16 server distributed cluster with 6.06 and asterisk 1.2, runs great... |
| 21:08.30 | nny | tzafrir I run my office phones on a 6.06 |
| 21:08.50 | tzafrir | I encountered all sorts of things that just don't work. |
| 21:09.04 | tzafrir | specifically with kernel modules building |
| 21:09.16 | nny | tzafrir i'll send you a link when I am done |
| 21:09.20 | tzafrir | But I mostly work with Debian anyway |
| 21:09.35 | grandpapadot | tzafrir: hrm.. I don't recall we had any problems. We are using the K7 kernel (Opterons) but other that that pretty straight forward. Oh, and using the server build, not desktop. |
| 21:09.51 | nny | this mpg123 part seems deprecated, but I have been smoking crack, let me peruse |
| 21:09.58 | grandpapadot | tzafrir: I prefer Debian mysqlf but Ubuntu had the Long Term Support option. |
| 21:10.22 | nny | yup |
| 21:10.25 | jcanfield | ManxPower: What quad-card would you recommend? |
| 21:10.28 | grandpapadot | mysqlf = myself ... geez I've typed mysql so many times I can't even spell myself without automatically hitting q |
| 21:10.40 | tzafrir | grandpapadot, LTS applies only for packages not in universe |
| 21:10.43 | ManxPower | jcanfield: Digium or Sangoma. We use Sangome |
| 21:11.09 | ManxPower | But if Digium had fixed their IRQ latency problems sooner we would never have switched to Sangoma |
| 21:11.28 | critch | is there some mechanism besides kill(pid, SIGHUP) reaping agi apps when a user hangs up? |
| 21:11.57 | jcanfield | ManxPower: Thanks. |
| 21:12.11 | ManxPower | critch: I just register a callback in my asterisk-perl using scripts |
| 21:12.39 | critch | Well, I have done that, and it is still trying to kill my app. Killing it early enough I can't get at my log data |
| 21:13.07 | ManxPower | um, catch SIGHUP in your app |
| 21:13.19 | critch | I even registered __DIE__ so I could stop it from dieing. |
| 21:13.20 | tzafrir | asterisk, zaptel, etc. are in universe |
| 21:13.26 | ManxPower | this is basic signal handling under Unix |
| 21:13.39 | ManxPower | critch: what language are you using for your AGI? |
| 21:13.45 | critch | ManxPower: yes, but my problem seems to be going further than catching that one signal |
| 21:13.46 | grandpapadot | tzafrir: We have a bunch of changes so we use the source tarballs ... |
| 21:13.48 | critch | ManxPower: perl |
| 21:13.56 | ManxPower | critch: and you are using asterisk-perl? |
| 21:14.08 | sporecook | Does anyone know if the the Asterisk 1.4 GUI functional at this point? |
| 21:14.19 | critch | no, older app that we had around before anyone had any libs for it |
| 21:14.24 | Nugget | depends on how loosely you define "functional" I suppose. |
| 21:15.24 | Nugget | "functional" in the sense of "can be used to delegate routine maintenance of a working asterisk install" is a gaping chasm of utility away from "will allow you to avoid learning how to construct config files in an editor" |
| 21:15.25 | ManxPower | $AGI->setcallback(\&mycallback); is what I use. |
| 21:16.13 | grandpapadot | Right, but the 1.1.1 firmware for the 12z has the "Virtual Cyclic Ring" functionality built int. |
| 21:16.20 | grandpapadot | ugh, wrong channel, sorry. |
| 21:16.23 | sporecook | Nugget, LOL |
| 21:16.52 | ManxPower | critch: you might want to look at how asterisk-perl does the setcallback function |
| 21:16.59 | sporecook | thought so, back vi |
| 21:17.01 | blitzrage | functional in terms of it loads something to the screen? yes. functional in terms of what you want to do with it? depends what you need -- try it! |
| 21:17.34 | ManxPower | just don't ask questions about it here -- there are dedicated channels for that sort of thing |
| 21:17.36 | critch | ManxPower: as you mentioned, it is simple unix signal handling, millions of examples on how to catch the signal out there. Still I am not getting the signal. |
| 21:17.51 | ManxPower | critch: what verison of Asterisk? |
| 21:18.10 | critch | ManxPower: Asterisk SVN-branch-1.4-r79049 |
| 21:18.51 | ManxPower | critch: the bug I was thinking of was fixed early in 1.2.x |
| 21:19.17 | ManxPower | critch: have you considered trying a released version of Asterisk? |
| 21:19.45 | critch | isn't the 1-4 branch essentially a release that I just happened to use svn to download? |
| 21:20.00 | ManxPower | no |
| 21:20.21 | ManxPower | it is what will eventually become the next 1.4 release. |
| 21:20.36 | mvanbaak | it is better then the latest release though |
| 21:20.46 | blitzrage | svn co http://svn.digium.com/svn/asterisk/tags/1.4.12 would be a release |
| 21:21.04 | ManxPower | mvanbaak: you must be a developer. |
| 21:21.21 | ManxPower | they always think newer == better |
| 21:21.38 | Nugget | nah, I think that's just your typical linux user mindset. |
| 21:21.40 | mvanbaak | ManxPower: the 1.4 branch only get bugfixes |
| 21:21.55 | mvanbaak | Nugget: I'm not a linux user |
| 21:22.01 | Nugget | yay :) |
| 21:22.03 | ManxPower | mvanbaak: Ah! You must be a NEW developer. |
| 21:22.22 | mvanbaak | OpenBSD > * |
| 21:22.24 | ManxPower | Bugfixes quite often create new bugs. |
| 21:22.28 | grandpapadot | BSD rocks. |
| 21:22.41 | Nugget | if there was a way to compile linux from linus' emacs buffer files I swear some people would want to do it. |
| 21:23.10 | grandpapadot | lol |
| 21:23.17 | [TK]D-Fender | rocks SINK |
| 21:23.24 | grandpapadot | >shudder< emacs |
| 21:23.31 | ManxPower | I've been bitten by AT LEAST TWO crash type bugs that were introduced in the "bug fix only" releases of Asterisk |
| 21:23.46 | file | it's an imperfect world |
| 21:23.57 | grandpapadot | Later all. |
| 21:23.58 | ManxPower | If BSD rocks and rocks sink, then does that mean BSD sinks? |
| 21:24.25 | ManxPower | file: I agree. I just take issue with people that think 1.4 SVN IS a perfect world. |
| 21:24.44 | mvanbaak | I never said it's perfect |
| 21:25.10 | mvanbaak | I said it's better then a release because it get more bugfixes then it creates bugs |
| 21:25.16 | ManxPower | mvanbaak: you implied that is is always better to run 1.4 SVN instead of a 1.4 release. -- at least that how I took it. |
| 21:25.42 | mvanbaak | it all depends on your setup and what bugs you encounter |
| 21:26.01 | ManxPower | at least with a 1.4.x release you usually know what the bugs are |
| 21:26.12 | mvanbaak | heck, we still have machines running 1.0.X because they are stable and none of the bugs in there affect those machines |
| 21:26.41 | ManxPower | mvanbaak: When I experienced the bugs introduced in 1.2.x bug fixes only I was doing cutting edge stuff like using Macros and call parking. |
| 21:27.06 | Nugget | heh |
| 21:27.08 | [TK]D-Fender | ManxPower, parking has been around for a LONG time before 1.2 |
| 21:27.15 | ManxPower | mvanbaak: in that case you are not one of "those" developers. |
| 21:27.19 | mvanbaak | strongbad*CLI> show version |
| 21:27.20 | mvanbaak | Asterisk 1.0.9-BRIstuffed-0.2.0-RC8o built by root@strongbad on a i686 running Linux |
| 21:27.22 | [TK]D-Fender | ManxPower, And macro's? Thats 1.0 stuff as well IIRC |
| 21:27.26 | UnixDog | [Oct 8 16:26:32] ERROR[11874]: chan_sip.c:12065 handle_response_invite: Got error on T.38 re-invite. Bad configuration. Peer needs to have T.38 disabled. |
| 21:27.30 | ManxPower | [TK]D-Fender: you missed my invisible <sarcasm> tag |
| 21:27.31 | mvanbaak | strongbad*CLI> show uptime |
| 21:27.31 | mvanbaak | System uptime: 5 weeks, 5 days, 11 hours, 27 minutes, 30 seconds |
| 21:27.50 | UnixDog | t.38 pass threw not working |
| 21:27.57 | mvanbaak | this machine handles roughly 1000 calls a day |
| 21:27.59 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 21:28.27 | ManxPower | mvanbaak: if you were one of "those developers" you would be running TRUNK SVN. |
| 21:28.58 | ManxPower | UnixDog: you are not doing passthru, you are doing reinvites. |
| 21:29.01 | ManxPower | try turning them off. |
| 21:29.48 | ManxPower | mvanbaak: I'll try to stop ranting. It is one of my pet peeves |
| 21:30.18 | mvanbaak | ManxPower: I _DO_ run svn trunk on some production machines |
| 21:30.31 | mvanbaak | like I said, it all depends on the setup |
| 21:30.37 | ManxPower | pervert |
| 21:30.44 | mvanbaak | good movie |
| 21:30.56 | file | UnixDog: Asterisk tried to send a reinvite with the T.38 information to the device, but the device responded back that it wasn't capable of doing it |
| 21:31.00 | UnixDog | ok canreinvite=no |
| 21:31.12 | ManxPower | Your users must be much more understanding and/or you have a a much larger IT staff. |
| 21:31.14 | file | it could be that the device in question is not configured properly |
| 21:31.31 | ManxPower | My users grab their torches and gather a mob if the PBX drops a call. |
| 21:31.34 | mvanbaak | ManxPower: the trunk boxen are only there to support skinny devices the correct way |
| 21:31.40 | nny | If i grep ps -Aux, i get asterisk running as asterisk, this seems proper, but one is also running as root, my normal box only shows one process as asterisk.. I created the user and group, and edited init script to say USER=asterisk/ group etc. any advice? |
| 21:31.44 | mvanbaak | local machines only handling local traffic |
| 21:31.51 | mvanbaak | all other traffic goes out using IAX |
| 21:31.59 | mvanbaak | the IVR stuff etc is on 1.2 boxen |
| 21:32.22 | mvanbaak | like I said: right tool for the job |
| 21:32.22 | ManxPower | on the other hand my users could not write a decent error report if their life depended on it. |
| 21:34.11 | ManxPower | "All my calls dropped yesterday!" "Can you give me an approximate time and the phone number?" "No." |
| 21:34.25 | zerohalo | Bahahaha... Sounds familiar. |
| 21:34.57 | ManxPower | I got a report of "garbled pages" the other day. turns out they were trying to use app_page with 40 destinations. |
| 21:35.18 | ManxPower | which creates a meetme conference and then dumps 40 calls into it within a second. |
| 21:35.27 | UnixDog | man clients are a pain |
| 21:35.40 | [hC] | ive got page working with over 40 people |
| 21:35.42 | [hC] | somehow. |
| 21:35.53 | [hC] | however it makes it so the first 1-2 seconds of audio are sometimes missed |
| 21:36.06 | ManxPower | UnixDog: not all of them. Just the ones without a single person that knows Microsoft is a company name, not a wordprocessor name. |
| 21:36.22 | [TK]D-Fender | [hC], call setup delay. |
| 21:36.26 | UnixDog | lol |
| 21:36.28 | ManxPower | "My microsoft is printing bad pages" |
| 21:36.29 | [hC] | [TK]D-Fender: yep. |
| 21:36.54 | UnixDog | I only use BSD and dealing with MS people is a pain in the arsss |
| 21:37.17 | [TK]D-Fender | Dyslexics are peepul too..... |
| 21:37.58 | UnixDog | I am a ser/cisco callmanager person being thrusted into the asterisk world |
| 21:38.14 | UnixDog | they are swaping out the cisco cm boxes for asterisk |
| 21:38.15 | nny | anyone got a sec to help me with running * as non-root? |
| 21:38.29 | UnixDog | make a users asterisk:asterisk |
| 21:38.32 | nny | added user/group, chmod/chown all files as listed |
| 21:38.44 | UnixDog | nny you short cut me |
| 21:38.47 | UnixDog | lol |
| 21:38.50 | nny | changed init script to run as asterisk |
| 21:38.51 | nny | lol |
| 21:39.00 | nny | set asterisk.conf to /var/run/asterisk/ |
| 21:39.08 | nny | makefile seems to already have it set |
| 21:39.23 | nny | although it looked.. not quite right.. using 1.4.12 |
| 21:39.48 | nny | mkdir /var/run/asterisk, but asterisk.ctl etc doesn't exist in it |
| 21:40.00 | nny | which i assume means that i need to recompile and change makefile |
| 21:41.05 | *** join/#asterisk sporecook (n=wdjdj@72.37.252.50) |
| 21:41.16 | nny | ha |
| 21:41.17 | nny | -_- |
| 21:41.28 | nny | forgot /etc/asterisk in my perms "Script" |
| 21:43.04 | nny | meh maybe not, still no /var/run/asterisk/asterisk.ctl |
| 21:43.13 | *** join/#asterisk sporecook (n=wdjdj@72.37.252.50) |
| 21:44.03 | [hC] | asterisk creates asterisk.ctl when it runs if it does not exist, im pretty sure. |
| 21:44.18 | [hC] | as long as your asterisk user has permissions to create items in /var/run/asterisk |
| 21:44.29 | [hC] | i may be wrong |
| 21:45.19 | nny | i think the issue is the makefile is still using the stock locale |
| 21:45.30 | nny | lots of else ifs in there for SunOS etc |
| 21:45.34 | nny | gonna recompile |
| 21:49.01 | *** join/#asterisk ectospasm (n=ectospas@c-68-62-174-102.hsd1.al.comcast.net) |
| 21:49.15 | *** join/#asterisk tripps (n=ss@66.60.235.100) |
| 21:55.07 | *** part/#asterisk jgomo3 (n=jgomez@200.72.223.2) |
| 21:59.03 | *** join/#asterisk NirS (n=chatzill@87.68.157.27) |
| 21:59.23 | NirS | bork bork bork |
| 22:02.50 | drwelby | Is this correct => exten=s,2,Dial(w${ARG1}) for using 'w' to get a pause? |
| 22:03.04 | NirS | no, that seems wrong |
| 22:03.26 | NirS | is ${ARG1} in the format technology/target ? |
| 22:03.57 | *** join/#asterisk iratik (n=itariki@adsl-71-144-123-210.dsl.spfdmo.sbcglobal.net) |
| 22:03.58 | nny | lol |
| 22:04.07 | nny | you guys would love my asterisk uninstall script |
| 22:04.09 | drwelby | ${ARG1} is a phone number that is going to be dialed |
| 22:04.14 | iratik | where is a good support channel for a horrible piece-of-crap service like skype |
| 22:04.14 | nny | rm -rf and lots of it |
| 22:04.50 | NirS | drw, if you're dialing a phone, you must enter the technology, the 'w' paramater has to come in the write place |
| 22:05.02 | NirS | so I would assume the exten should look like this |
| 22:05.10 | drwelby | Oh, I get it |
| 22:05.20 | NirS | exten => s,2,Dail(${ARG1},120,w) |
| 22:05.28 | NirS | where 120 is a waiting time for pickup |
| 22:06.07 | drwelby | ARG1 must contain all of the parameters, not just the phone number |
| 22:06.15 | [hC] | uhh, it doesnt HAVE to |
| 22:06.26 | NirS | no it doesn't |
| 22:06.28 | [hC] | in his example, it needs to contain just the phone number. |
| 22:06.37 | NirS | I suggest that ARG1 will hold only the technology and target |
| 22:06.48 | NirS | and leave the rest to the dialplan or via additional variables |
| 22:06.52 | drwelby | But it probably isn't just the number, so tacking a 'w' in front is not going to work |
| 22:07.04 | NirS | using a single variable for everything will make your dial-plan highly unreadable |
| 22:07.33 | [TK]D-Fender | NirS, You cannot insert a pause int a dial with that ARG1 you are showing. |
| 22:08.13 | [TK]D-Fender | NirS, "w" has to be inserted BEFORE the number part of an analog zaptel dial number. |
| 22:08.52 | NirS | ok, so I don't remember the entire asterisk application by heart :( |
| 22:09.01 | [TK]D-Fender | NirS, And of course you're expecting us to trust that ${ARG1} is even filled with something SANE. |
| 22:09.18 | [TK]D-Fender | NirS, "show application dial" - Oh and excuses won't get you very far.... |
| 22:09.27 | drwelby | NirS isn't the OP |
| 22:09.30 | NirS | Fender, relax dude |
| 22:09.36 | [TK]D-Fender | NirS, :p |
| 22:09.59 | *** part/#asterisk nny (n=scott@64.203.239.83.static-pool-4.pool.hargray.net) |
| 22:10.07 | [TK]D-Fender | NirS, Just seriously avoid the cop-outs... its not like you don't have 100 examples and documentation at your fingertips... |
| 22:10.31 | Strom_M | [TK]D-Fender: ffs man |
| 22:10.37 | [TK]D-Fender | :D |
| 22:10.40 | NirS | Fender, its' midnight over here, and to be honest, I don't remember everything by heart |
| 22:10.58 | Strom_M | [TK]D-Fender: go get irritated about Ontario or something |
| 22:10.59 | [TK]D-Fender | NirS, ..."show application dial" <- see? you can even cut & paste it! |
| 22:11.05 | *** join/#asterisk anthm (n=anthm@mbd0736d0.tmodns.net) |
| 22:11.05 | *** mode/#asterisk [+o anthm] by ChanServ |
| 22:11.27 | [TK]D-Fender | Strom_M, As long as GWB is around I'll never have to look closer for a good target ;) |
| 22:11.29 | NirS | ok, so I didn't look at the manual before typing |
| 22:12.30 | NirS | big f**** deal |
| 22:12.31 | drwelby | So really I should be inserting a 'w' here: exten=_011XXXXXXX!,1,Macro(trunkdial,${trunk_1}/w${EXTEN}) |
| 22:13.14 | Strom_M | NirS: it's ok, [TK]D-Fender just suffers from the affliction of being a stereotypically arrogant and irascible Quebecer |
| 22:13.50 | *** join/#asterisk MrTelephone (n=test@66-78-127-124.access.ripnet.com) |
| 22:13.52 | MrTelephone | hey |
| 22:13.53 | [TK]D-Fender | drwelby, We are still left to assume the contents of ${trunk_1}....... |
| 22:13.53 | Strom_M | we forgive him for it |
| 22:14.08 | Strom_M | it's not his fault :) |
| 22:14.23 | linagee | Strom_M: do you know of a B&M that you can get hard phone (polycom, not gs, etc.) at? |
| 22:14.29 | MrTelephone | does anyone know if polycom 501 send a vendor-identifier string in the dhcp request? |
| 22:14.36 | NirS | Strom, I don't mind that, I'm simply a little too old for that, both in age and with Asterisk - I think I'm one of the oldest Asterisk implementors and integrators in here |
| 22:14.41 | [TK]D-Fender | Strom_M is just jealous becuase he's only MONOtypical and deams of multi-channel greatness :p |
| 22:14.41 | drwelby | [TK]D-Fender: trunk_1=Zap/G1 |
| 22:14.42 | linagee | Strom_M: s/phone/phones/ |
| 22:14.42 | NirS | been doing Asterisk since late 2002 |
| 22:14.49 | Strom_M | linagee: not AFAIK |
| 22:15.17 | [TK]D-Fender | drwelby, that would be the right place then, assuing that zap group contains analog channels |
| 22:15.29 | linagee | Strom_M: gee. that sucks. i talked to someone at fry's who looks like they might be a smart employee trapped amongst stupid ones. i asked about asterisk and they were like, "oh yes! i know what that is. no, none of this crap works with it." |
| 22:15.32 | drwelby | [TK]D-Fender: yes, it does |
| 22:15.50 | Strom_M | linagee: polycom sets are not a mass-market consumer good |
| 22:15.52 | MrTelephone | your your own best engineer |
| 22:16.01 | MrTelephone | eh |
| 22:16.05 | MrTelephone | does anyone know if polycom 501 send a vendor-identifier string in the dhcp request? |
| 22:16.18 | TrentCreek | we should be asking yOu that Mr Telephone |
| 22:16.27 | Strom_M | MrTelephone: i dunno. run tcpdump? |
| 22:16.34 | [TK]D-Fender | MrTelephone, I believe thats a feature in the newer firmwares. |
| 22:16.55 | MrTelephone | noone classes phones differently with dhcpd? |
| 22:17.15 | [TK]D-Fender | MrTelephone, noone I've heard of. |
| 22:17.20 | MrTelephone | they should make a ata that has vpn software built in |
| 22:17.23 | NirS | Fender, suck on this one, that parameters doesn't even appear in the 'show application dial' |
| 22:17.36 | MrTelephone | anyone have a cisco 7920 for sale? |
| 22:17.40 | Strom_M | MrTelephone: on the one occasion I put the Polycoms on a different subnet, I did it by MAC address |
| 22:18.10 | Strom_M | MrTelephone: i dont have a 501, but i can try with a 320... |
| 22:18.25 | Strom_M | linagee: |
| 22:18.28 | Strom_M | ~wglwat |
| 22:18.28 | jbot | extra, extra, read all about it, wglwat is well, good luck with all that |
| 22:18.28 | [TK]D-Fender | NirS> exten => s,2,Dail(${ARG1},120,w) <- this clearly has "w" as a recording parameter.... |
| 22:18.41 | MrTelephone | strom, if you wouldn't mind that would be cool.. I tried to search but I got nothing |
| 22:18.48 | NirS | yes, I didn't read the entire question |
| 22:19.10 | NirS | to be honest, I'm not familiar with a "pause" mechanism in the dial command |
| 22:19.13 | [TK]D-Fender | NirS, well I can only judge based on what I see now can't I? Peace. |
| 22:19.16 | NirS | unless it is an undocumented one |
| 22:19.25 | MrTelephone | pause is wait() |
| 22:19.31 | *** part/#asterisk zerohalo (n=zeroHalo@pool-72-70-79-233.bstnma.east.verizon.net) |
| 22:19.33 | [TK]D-Fender | NirS, No, its all over the WIKI, and pretty sure somewhere in the book |
| 22:19.35 | NirS | MT, that is true |
| 22:19.40 | NirS | hold on |
| 22:19.46 | NirS | got the book with me |
| 22:19.56 | MrTelephone | book worm |
| 22:20.12 | drwelby | wait() works too |
| 22:20.16 | [TK]D-Fender | NirS, Dial(ZAP/[group or channel#]/[bunch of w's]12345678,20,t) |
| 22:20.31 | [TK]D-Fender | NirS, So right before your number after the 2nd slash |
| 22:20.31 | drwelby | but I want zap to wait before it dials |
| 22:20.38 | [TK]D-Fender | drwelby, see above |
| 22:20.57 | drwelby | [TK]D-Fender: yeah, figured out the w part, thanks |
| 22:20.57 | [TK]D-Fender | drwelby, if it occurs as I jsut described it will. 0.5s per "w" |
| 22:21.12 | NirS | well, I'm not familiar with it, I guess you learn something new every day |
| 22:21.20 | [TK]D-Fender | NirS, I try.... |
| 22:21.49 | [TK]D-Fender | NirS, useful in places that don't feed you dialtone and start listening as fast as zaptel trys to throw DTMF at them... |
| 22:22.03 | NirS | no problem in trying |
| 22:22.13 | NirS | just try being a little more civil about it |
| 22:22.34 | MrTelephone | anyone setup call me services on their asterisk serveR? |
| 22:23.04 | [TK]D-Fender | MrTelephone, elaborate please.... |
| 22:23.11 | NirS | here's a small question that I never quite figured why people do |
| 22:23.27 | NirS | why would people run AGI's in the dial-plan with DeadAGI, instead of AGI |
| 22:23.37 | MrTelephone | 1-800-??? -> local did -> enter key code -> dials pre determined number |
| 22:23.43 | MrTelephone | a call home card? |
| 22:23.50 | NirS | I was always under the impression that DeadAGI is meant to run only with the 'h' extension |
| 22:24.08 | [TK]D-Fender | NirS, in places where the call is dead and AGI con't interact properly. I don't the exact reaction it makes to either personally however |
| 22:24.09 | *** join/#asterisk Gamercjm (n=chris@pool-71-254-185-233.lsanca.fios.verizon.net) |
| 22:24.31 | [TK]D-Fender | MrTelephone, Well.... ic an call my home through 1-800# and get to DISA.... |
| 22:24.41 | MrTelephone | DISA? |
| 22:24.48 | [TK]D-Fender | MrTelephone, "show application DISA" |
| 22:25.07 | [TK]D-Fender | MrTelephone, lets you dial out a context, and gives you DIALTONE while waiting for your input |
| 22:25.07 | Gamercjm | how can I set a caller id to say Private or blocked? if I just have an empty caller id it says "unknown" on a landline and '0' on my cell |
| 22:25.28 | [TK]D-Fender | Gamercjm, depends what kind of interface you are dialing OUT on. |
| 22:26.22 | Gamercjm | what do you mean? im trying set callerid then use DIAL() |
| 22:26.32 | Gamercjm | in a context |
| 22:26.40 | MrTelephone | where does authenticate get its passwords from , voicemail? |
| 22:27.10 | [TK]D-Fender | Gamercjm, like I said it depends what you are dialing out **ON** |
| 22:27.46 | [TK]D-Fender | MrTelephone, Authenticate can get from a text file, or a single valid value passed as a parameter. "vmauthenticate" auths against a mailbox. |
| 22:28.03 | MrTelephone | nice |
| 22:28.07 | MrTelephone | looking at it now |
| 22:28.10 | [TK]D-Fender | MrTelephone, it'll return which box it matched if successful. |
| 22:28.30 | MrTelephone | people are getting charged big on collect calls |
| 22:28.34 | [hC] | so, i just converted a box to load asterisk non-root... the recommendation was to recompile, changing the /var/run path to /var/run/asterisk -- I just modified asterisk.conf and it started fine... |
| 22:28.35 | Strom_C | MrTelephone: as best i can tell from looking at these packets, the only way to identify that it's a polycom set is by the MAC address |
| 22:28.43 | [hC] | Am i missing something, or was that an unnecessary step? |
| 22:28.44 | MrTelephone | so I want them to be able to give their friends with no longdistance the 1-800 number |
| 22:29.01 | MrTelephone | strom_c, no vendor-ident? |
| 22:29.03 | MrTelephone | dammit |
| 22:29.12 | MrTelephone | i guess u can class them by mac then |
| 22:29.29 | MrTelephone | even windows xp sends identifier |
| 22:29.35 | Strom_C | MrTelephone: oh wait |
| 22:29.40 | Strom_C | here's a vendor class ID |
| 22:30.14 | Strom_C | it's in the second DISCOVER |
| 22:30.27 | MrTelephone | Soundpoint.19239213012391203? |
| 22:30.29 | MrTelephone | haha |
| 22:30.34 | MrTelephone | just a stab in the dark |
| 22:31.11 | [TK]D-Fender | MrTelephone, WinXP's is to aid in selecting an appropriate exploit ;) |
| 22:31.12 | Strom_C | nope |
| 22:31.20 | MrTelephone | :) |
| 22:31.23 | MrTelephone | nothing strom? |
| 22:31.23 | Strom_C | MrTelephone: telephone make, model, firmware loads |
| 22:31.43 | Strom_C | build dates for those firmware loads |
| 22:32.04 | MrTelephone | i want to match substring |
| 22:32.09 | MrTelephone | is the word soundpoint in there? |
| 22:32.33 | Strom_C | yes |
| 22:32.38 | Strom_C | SoundPoint |
| 22:32.40 | [hC] | if the models in there, i would imagine so since the model is soundpoint ip### |
| 22:32.57 | MrTelephone | nice |
| 22:33.03 | Strom_C | "Polycom SoundPointIP-SPIP_320" |
| 22:33.11 | MrTelephone | nice |
| 22:33.25 | MrTelephone | SPIP_501 then |
| 22:34.39 | Strom_C | MrTelephone: you might want to do a reality check on your end |
| 22:34.45 | Strom_C | tcpdump |
| 22:34.47 | Strom_C | wireshark |
| 22:34.50 | Strom_C | something :) |
| 22:35.02 | MrTelephone | just trying to get presetup on holidays here |
| 22:35.06 | MrTelephone | set vendor-string = option vendor-class-identifier; |
| 22:35.11 | MrTelephone | if u put that in ur dhcpd.conf |
| 22:35.17 | MrTelephone | it will put the vendor-class in dhcpd.leases |
| 22:39.47 | Yourname`` | Ok, here's my good question. I have an analog phone line (Bell Canada), and a laptop with Asterisk on it. I, obviously, cannot add any PCI cards to the laptop or anything. So, would the Linksys SPA3102 suffice? |
| 22:40.13 | Yourname`` | TK defender helped me out with this long ago, but I'm finally about to get one of these.. so one last head nod is what I need. |
| 22:40.34 | Strom_C | Yourname``: probably |
| 22:40.54 | MrTelephone | strom thanks for the lookup |
| 22:42.06 | Yourname`` | Strom_C: What will the question have to sound like to evoke a "Definitely!" from you? |
| 22:42.19 | Strom_C | Yourname``: I'll have to have used a SPA-3102 |
| 22:42.34 | *** join/#asterisk saftsack (n=saftsack@pD9E05A3D.dip.t-dialin.net) |
| 22:42.41 | Strom_C | and also, you probably want to run asterisk on a real server, not a laptop |
| 22:42.46 | Strom_C | etc etc etc |
| 22:43.36 | Yourname`` | Yeah, trying to see if I can get a working barebones Asterisk for home. |
| 22:44.26 | *** join/#asterisk brut- (n=brut@66.173.4.254) |
| 22:44.32 | drwelby | To set blind transfer to ##, is the syntax blindxfer=## or blindxfer=>## |
| 22:44.49 | blitzrage | probably doesn't matter, but check the syntax in the sample file |
| 22:46.07 | UnixDog | ok got the cid working and passing |
| 22:46.14 | [hC] | I use a spa 3102 on one site |
| 22:46.16 | [hC] | it works fine |
| 22:46.31 | MrTelephone | what do you guys know about allow and deny members in dhcpd. do you have to deny each class if you allow 1? |
| 22:46.32 | UnixDog | but now to play with t38 pass |
| 22:46.36 | Yourname`` | ah |
| 22:46.39 | UnixDog | its set |
| 22:46.47 | UnixDog | we will see if it works |
| 22:47.21 | Yourname`` | How about communication is wireless from the laptop? I'd need a wireless enabled SPA then? |
| 22:51.51 | *** join/#asterisk tristanbob (n=tristan@oalug/member/tristanbob) |
| 22:53.23 | *** join/#asterisk brut- (n=brut@66.173.4.254) |
| 22:54.25 | *** join/#asterisk FireMac (n=firemac@CPE000d88ae88b9-CM0011ae8bb0ee.cpe.net.cable.rogers.com) |
| 22:54.49 | [hC] | Yourname``: dont go to that much trouble to test it... |
| 22:55.08 | [hC] | Yourname``: you're setting yourself up for a test case that wont be anything like your real world scenario |
| 22:55.11 | Yourname`` | Would love to keep the laptop away from the public eye, and the wires going to it. |
| 22:55.24 | [hC] | Yourname``: the spa3102 connects to your network independently over ethernet.. just plug it into a switch |
| 22:55.53 | UnixDog | you can use a linksys wbp54g |
| 22:55.56 | [hC] | Yourname``: i really really wouldnt run asterisk on a laptop. i mean you can but why not just put together an old machine for a couple hundred bucks? |
| 22:56.02 | Yourname`` | It would actually. I have a powersocket in my bathroom attic, and I have a couple laptop servers serving http, etc. |
| 22:56.04 | UnixDog | and have your ata be wireless |
| 22:56.53 | Yourname`` | [hC]: I understand. I also have a couple other boxes I can run it on. But, laptop why not? :) |
| 22:57.05 | Yourname`` | UnixDog: That's what I was thinking, the SPA3102 doesn't seem wireless. |
| 22:57.46 | [hC] | Yourname``: suit yourself. |
| 22:57.58 | UnixDog | I use a wpb54g and my gxp2000 and I carry it to where I need it plug it in and go |
| 22:59.09 | Yourname`` | Neat. |
| 22:59.13 | UnixDog | I have even put it on a battery and the wifi and gone to the pool next door and still gotten calls |
| 22:59.17 | *** join/#asterisk BockBilbo (n=BockBilb@eu85-84-62-227.clientes.euskaltel.es) |
| 22:59.22 | BockBilbo | hello |
| 22:59.37 | UnixDog | the wpb54g is only 35 bucks |
| 22:59.42 | UnixDog | so worth the money |
| 23:02.57 | BockBilbo | UnixDog is the wpb54g compatible with the spa3102? |
| 23:03.11 | UnixDog | yes it shoud work |
| 23:03.27 | UnixDog | it plugs into the rj45 |
| 23:03.37 | UnixDog | and the 5v power plug |
| 23:03.43 | UnixDog | so it should work |
| 23:03.55 | UnixDog | it passes the power threw |
| 23:04.00 | BockBilbo | i was thinking on getting a buffalo router and using dd-wrt for making a wireless bridg |
| 23:04.04 | BockBilbo | *bridge |
| 23:06.33 | UnixDog | they are both made by linksys so they should work together |
| 23:08.03 | *** join/#asterisk Agnt_0rnge (n=chatzill@mail.enplan.com) |
| 23:08.18 | Agnt_0rnge | test |
| 23:09.04 | Strom_C | test result: utter failure |
| 23:09.15 | Agnt_0rnge | thanks |
| 23:10.39 | *** join/#asterisk adeel (n=adeel@c-24-7-132-155.hsd1.ca.comcast.net) |
| 23:12.45 | adeel | i seem to be getting a flood of 'Incoming call: got sip response 500 "internal server error" back from 192.168.1.117" (polycom phone) |
| 23:13.01 | russellb | adeel: reboot the phone |
| 23:13.04 | adeel | although that same message is coming in for every polycom phone i have registered |
| 23:13.14 | adeel | russellb, is there a way to remotely reboot the phones? |
| 23:13.16 | *** join/#asterisk GreggB (n=GreggB@66.206.86.107) |
| 23:13.25 | russellb | adeel: yeah, i think so, using the "sip notify" CLI command ... |
| 23:13.40 | adeel | hmm...i've tried that before but i never know if it works or not |
| 23:14.00 | JunK-Y | adeel: look at the phone and see if it reboots or not |
| 23:14.17 | JunK-Y | and see if it becomes unreachable for a moment for *. |
| 23:14.41 | adeel | JunK-Y, i'm not physically at the location where the phones are installed |
| 23:23.30 | *** join/#asterisk remmo (n=junk@203.32.47.250) |
| 23:23.56 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 23:25.00 | TrentCreek | Anyone here know about SELinux??? |
| 23:25.49 | *** join/#asterisk sergey (n=sergey@sergey.iks.ru) |
| 23:27.51 | *** join/#asterisk CVirus (n=GoD@196.205.193.120) |
| 23:29.55 | adeel | how do i set the polycom phones to not do digitmapping and let asterisk do it? |
| 23:30.22 | *** join/#asterisk techie (n=techie@adsl-76-214-30-87.dsl.lsan03.sbcglobal.net) |
| 23:34.18 | blitzrage | adeel: you have to set the dialplan for the phone in the configuration file for the phone |
| 23:34.39 | blitzrage | you can check the polycom docs for the patterns allowed and where to find it |
| 23:35.49 | *** join/#asterisk craigk (n=ckowald@58.174.122.198) |
| 23:35.57 | adeel | blitzrage, that i know, but i was wondering if i could just remove the polycom digitmapping all together...it seems kind of redundant to have asterisk and polycom both do it |
| 23:36.09 | blitzrage | yes you can - that's what I was saying |
| 23:36.17 | blitzrage | set the dialplan in the polycom to something that won't match |
| 23:36.30 | blitzrage | or to something that matches more loosely than asterisk |
| 23:36.58 | BockBilbo | i have asterisk 1.4.11 instaled on ubuntu gutsy from binaries and need to compile and install the addons to support mysql cdr. I have al ready downloaded the addons from the svn, configured and compiled them. Where does the cdr_mysql.so file supposed to be installed when i do make install? i think that the installation is not beeing succesfull.... |
| 23:37.17 | blitzrage | /usr/lib/asterisk/modules |
| 23:37.52 | BockBilbo | blitzrage ok, so just as i though the file is not beeing installed |
| 23:38.03 | blitzrage | is it actually compiled? |
| 23:38.09 | blitzrage | you can just copy it into the dir |
| 23:38.21 | BockBilbo | well, i dont get any error while building the addons |
| 23:38.30 | BockBilbo | but dont see the file anywhere inside the directory |
| 23:38.33 | BockBilbo | :S |
| 23:38.55 | blitzrage | maybe you don't have the right dependencies installed if it's not building the .so file |
| 23:39.37 | BockBilbo | well |
| 23:39.47 | BockBilbo | i get no error when running the configure script |
| 23:39.53 | blitzrage | that doesn't mean anything |
| 23:39.54 | BockBilbo | so i should have all the dependencies |
| 23:39.56 | BockBilbo | mmm |
| 23:39.59 | BockBilbo | ok |
| 23:40.05 | blitzrage | just because it doesn't error doesn't mean it found everything |
| 23:40.21 | blitzrage | it very well could have not found it -- that's not an error |
| 23:40.22 | BockBilbo | checking for asterisk.h... no |
| 23:40.29 | BockBilbo | i bet thats the problem |
| 23:42.44 | *** join/#asterisk Dalbaech (n=Dalbaech@c-98-200-244-16.hsd1.tx.comcast.net) |
| 23:42.46 | Dalbaech | hey guys |
| 23:44.09 | hmmhesays | what a pain in the ass day |
| 23:48.36 | blitzrage | amen |
| 23:48.39 | adeel | amen |
| 23:49.59 | hmmhesays | I just sent of a quote, these guys said top of the line, but they won't like it |
| 23:51.45 | Nugget | We want top of the line, but our budget only allows for grandstream phones hooked up to trixbox running on and old emachines desktop that we found in the supplies cabinet. We'll be usng clone x100p cards, but it should be a problem because we plan to pirate the g729 codec using that ibm public code. so can you help us get realtime mysql working? |
| 23:52.21 | Nugget | it shouldn't take you more than a day or so to get us all up and running, I figure. |
| 23:52.48 | Nugget | talk to our receptionist, she can help you design the dialplan we need for all 12 departments |
| 23:53.16 | hmmhesays | LOL |
| 23:53.24 | hmmhesays | don't make me get the giant trout out |
| 23:55.03 | BockBilbo | blitzrage after compiling, on which directory of the main compiling folder are the binaries supposed to be? |
| 23:55.27 | *** join/#asterisk bkruse_home (n=bkruse@69.73.127.92) |
| 23:55.52 | blitzrage | huh? |
| 23:56.13 | BockBilbo | :S |
| 23:56.26 | BockBilbo | nevermind |
| 23:56.33 | BockBilbo | i guess im too tired to do this tonight |
| 23:56.43 | *** join/#asterisk DarkRift (i=dark@bas10-montreal02-1177581883.dsl.bell.ca) |
| 23:56.51 | JT | :S, worst emocticon ever |
| 23:57.11 | blitzrage | JT, would be the worst emoticon evar |
| 23:57.25 | JT | sure |
| 23:57.41 | BockBilbo | lol |
| 23:57.48 | *** join/#asterisk MaliutaWrk (n=nikolai@fw.hitwise.com) |
| 23:58.00 | Nugget | heh |
| 23:59.01 | *** join/#asterisk livingtm (n=livingtm@cpe-72-226-64-71.nycap.res.rr.com) |
| 23:59.10 | blitzrage | ok, gym time... been sitting here too long |
| 23:59.44 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |