| 00:00.21 | De_Mon | Yourname` do this, before you set the callerid to something else, add a NoOp(${CALLERID}) then do another one after you change it and see what happens |
| 00:00.34 | De_Mon | it should, change.. |
| 00:00.38 | generalhan | Cyllene: i would still try it in the dialplan like the first example shows just to be sure |
| 00:00.43 | vaughany | is there nothing more i can do with timeout issues apart from making a cron job to check the register status and do a sip reload. not the most effective method, im sure ill miss calls duing this so id realy like to solve this issue properly. http://pastebin.com/d3b898eaa |
| 00:01.16 | Cyllene | ok |
| 00:02.19 | De_Mon | vaughany what does a good registration look like? |
| 00:02.38 | De_Mon | (has no idea how to help with this issue, just curious :) |
| 00:02.39 | Cyllene | generalhan: That did not work either |
| 00:03.11 | De_Mon | Cyllene did you reload the dialplan? |
| 00:03.27 | Cyllene | De_Mon: I restarted asterisk. |
| 00:03.35 | generalhan | Cyllene: well im tapped. ive never done ANY work with one-touch before, so other than the basic research i have done, there isnt much more i can help with |
| 00:03.48 | Cyllene | generalhan: Transfering calls doesn't work either |
| 00:04.13 | De_Mon | what type of device are you using? |
| 00:04.16 | Cyllene | res_features is loaded too |
| 00:04.22 | Cyllene | De_Mon: Softphones (DIAX) |
| 00:05.02 | De_Mon | hrm |
| 00:05.07 | generalhan | and you are trying all these features from the CALLING user's phone right ? not the one that took the call ? |
| 00:05.18 | Cyllene | Correct, generalhan |
| 00:05.44 | Cyllene | Maybe I need to file a bug report? |
| 00:05.47 | generalhan | Cyllene: yea sorry man ... maybe some one with more experience with this feature will come in and be able to help |
| 00:06.05 | De_Mon | Cyllene does the phone let you pick your dtmf mode? |
| 00:06.16 | *** join/#asterisk TheNewAndy (n=TheNewAn@144.131.134.181) |
| 00:06.20 | De_Mon | im not sure if thats an iaxean feature or sip only.. |
| 00:07.15 | Cyllene | De_Mon: No. |
| 00:08.03 | generalhan | Cyllene: thats a good suggestion though to narrow it down more, do you have a set of SIP phones that you can test this same Dial command on ? |
| 00:08.36 | vaughany | U 2007/08/09 10:08:04.378732 192.168.1.5:5060 -> 203.166.6.160:5060 <- is this showing 192.168.1.5 to my peer? |
| 00:08.48 | *** join/#asterisk johann8384 (n=johann83@gateway.myogre.com) |
| 00:09.02 | Cyllene | generalhan: No. |
| 00:09.10 | vaughany | Contact: <sip:6139095XXX@58.110.226.243>. <- or does this specify what to connect on |
| 00:09.12 | Yourname` | De_Mon: Doesn't work. |
| 00:09.27 | lmadsen | Qwell: ping?! |
| 00:09.50 | lmadsen | Qwell[]: maybe here? |
| 00:10.06 | Yourname` | De_Mon: I changed it from exten=>_NXXNXXXXXX,1,Noop(gafachi) to exten=>_NXXNXXXXXX,1,Noop(${CALLERID}) ... nothing. |
| 00:11.29 | De_Mon | what does your callerid= line look like? |
| 00:11.31 | vaughany | should my trunks be sending registrations every second ? |
| 00:11.40 | De_Mon | (sip.conf) |
| 00:11.44 | Yourname` | Lemme open sip.conf |
| 00:12.03 | `Sean | hey is there any specific ways to record certain calls? |
| 00:12.04 | Cyllene | I guess I'll open a bug report. |
| 00:12.13 | `Sean | like say i wanted to rewcord a conversation between me and a freind how could i do it? |
| 00:12.27 | `Sean | anyone even point to a command would be appreciated |
| 00:12.27 | De_Mon | `Sean show application MixMonitor |
| 00:12.28 | Yourname` | De_Mon: This is the callerid line for the extension [100] : callerid="7601001000" <100> |
| 00:12.35 | `Sean | thanks De_Mon |
| 00:12.45 | De_Mon | hrm |
| 00:14.15 | vaughany | can someone help please, i've been trying to figure this out for days |
| 00:14.20 | De_Mon | Yourname` if you call another peer does it show that callerid? |
| 00:14.58 | Yourname` | Good Q, lemme try |
| 00:15.19 | De_Mon | Cyllene what version are you using? |
| 00:15.29 | Cyllene | 1.4.10 |
| 00:15.32 | `Sean | De_Mon it doesn't exactly say how to use it... |
| 00:15.38 | *** join/#asterisk anthm (n=anthm@mbc0736d0.tmodns.net) |
| 00:15.38 | *** mode/#asterisk [+o anthm] by ChanServ |
| 00:15.40 | `Sean | is it a extensions.conf thing? |
| 00:15.59 | `Sean | hah it is indeed :) |
| 00:16.09 | `Sean | MixMonitor(<file>.<ext>[|<options>[|<command>]]) |
| 00:16.25 | Yourname` | De_Mon: Yup! |
| 00:16.29 | Yourname` | De_Mon: It does! |
| 00:17.01 | De_Mon | Yourname` check my sanity and add a NoOp(${CALLERID}) in that extension too |
| 00:17.15 | Yourname` | lol |
| 00:17.18 | Yourname` | One sec. |
| 00:17.29 | Yourname` | And then try an outbound call? |
| 00:17.41 | Yourname` | Hey, wait.. what do you measn add in that extension too? |
| 00:17.42 | De_Mon | no, call the peer again and see if the callerid is set correctly |
| 00:18.04 | `Sean | anyone have some real examples of mixmonitor? |
| 00:18.13 | De_Mon | your dialplan should have an exten => 1234,1,Dial(SIP/1234) or something |
| 00:18.41 | De_Mon | `Sean exten => foo,n,MixMonitor(file.wav) |
| 00:19.13 | `Sean | De_Mon i get that part, im trying to figure out how i can add dates and timestamp |
| 00:19.21 | `Sean | or even instead have it named by phone number |
| 00:19.23 | Yourname` | De_Mon: http://pastebin.ca/650522 This is what it is right now after your mod. |
| 00:19.24 | `Sean | that was dailled.. |
| 00:19.47 | De_Mon | `Sean pretty simple app |
| 00:20.01 | `Sean | like have it save to a file, like 18008008300.time.wav |
| 00:20.13 | `Sean | or time and date perhaps |
| 00:20.32 | De_Mon | `Sean oh, you want uh.. the function that returns the date :P |
| 00:20.53 | De_Mon | Yourname` right, but there the noop returned nothing... |
| 00:20.53 | `Sean | De_Mon yes, so i know when this call was recorded and to wich phone number this call was placed too |
| 00:21.03 | Yourname` | De_Mon: Ok, so what do we do? |
| 00:21.14 | De_Mon | Yourname` you tried dialing another peer and the callerid showed up |
| 00:21.23 | De_Mon | noop the callerid in that part of your dialplan too |
| 00:21.30 | Yourname` | De_Mon: Ahh.. |
| 00:21.42 | x86 | `Sean: look at the channel variables, the call date and time are in there |
| 00:21.57 | De_Mon | `Sean do 'show functions' and look for something that deals with the date |
| 00:22.06 | x86 | `Sean: can't remember the names off the top of my head, but dump all the variables and you'll find it |
| 00:22.08 | De_Mon | then show function <that function< for more details |
| 00:22.12 | Yourname` | De_Mon: Gotcha. |
| 00:22.20 | x86 | `Sean: along with other cool stuff that might be of interest to you |
| 00:22.23 | De_Mon | x86 a channel variable contains time too? |
| 00:22.24 | `Sean | thanks |
| 00:22.36 | x86 | De_Mon: sure, as an epoch, iirc |
| 00:22.53 | `Sean | CALLERID CALLERID(datatype) Gets or sets Caller*ID data on the channel. |
| 00:22.53 | De_Mon | ew |
| 00:23.25 | `Sean | i dont suppose i can use that can i? |
| 00:23.42 | JT | easy |
| 00:23.58 | De_Mon | ${DATETIME}: Current date time in the format: DDMMYYYY-HH:MM:SS This is deprecated in Asterisk 1.2, instead use :${STRFTIME(${EPOCH},,%d%mNaVH:NaVS)}) |
| 00:24.02 | fujin | Could I get some opinions on queues? |
| 00:24.04 | JT | ${STRFTIME(${EPOCH},,%Y%m%d-%H%M%S)} |
| 00:24.12 | *** join/#asterisk Keltus (n=Keltus@about/cooking/nakedchef/beefstew/Keltus) |
| 00:24.18 | fujin | are dynamic queue members via addqueuemember the best way to do it? |
| 00:24.18 | JT | it's amazing what reading documentation can do :P |
| 00:24.25 | De_Mon | fujin they suck, they rock, they are ok |
| 00:24.26 | `Sean | where is the documentation for that? |
| 00:24.32 | fujin | we've currently got a agentcallbacklogin implementation |
| 00:24.47 | De_Mon | `Sean www.voip-info.org is a lifesaver |
| 00:24.54 | `Sean | De_Mon i'm there |
| 00:25.00 | `Sean | can you paste me th elink where you got that? |
| 00:25.04 | Cyllene | Bug filed. #10412 |
| 00:25.11 | JT | year month day is the way to go especially if you're naming files |
| 00:25.13 | fujin | I just found something on the asterisk mailing list regarding adding a membername argument to AddQueueMember, is this in the current revision? |
| 00:25.19 | De_Mon | I searced for 'channel variable' (thats what x86 was talking about) |
| 00:25.34 | De_Mon | `Sean i bet searching for epoch would give you some good results... |
| 00:25.43 | `Sean | http://www.voip-info.org/wiki/view/MixMonitor |
| 00:26.07 | JT | this is how i mixmonitor: |
| 00:26.11 | x86 | `Sean: don't use voip-info.org's search engine... use google |
| 00:26.25 | `Sean | meh there search engine must suck then :/ |
| 00:26.25 | JT | ,MixMonitor(${STRFTIME(${EPOCH},,%Y%m%d-%H%M%S)}-${ARG1}-${UNIQUEID} |
| 00:26.25 | JT | .wav) |
| 00:26.36 | JT | where ARG1 = CALLERID(num) |
| 00:26.37 | x86 | `Sean: for example: "epoch site:voip-info.org" |
| 00:26.49 | De_Mon | x86 voip-info.org's search engine IS google |
| 00:26.51 | JT | that's for outgoing calls |
| 00:26.53 | JT | for incoming: |
| 00:26.56 | JT | ,MixMonitor(incoming/${STRFTIME(${EPOCH},,%Y%m%d-%H%M%S)}- |
| 00:26.56 | JT | ${CALLERID(num)}-M${EXTEN}-${UNIQUEID}.wav) |
| 00:26.59 | x86 | De_Mon: is it? |
| 00:27.07 | De_Mon | 'Search with Google' |
| 00:27.07 | JT | x86: clearly |
| 00:27.09 | De_Mon | lol |
| 00:27.17 | x86 | i always had major problems with it way back in the day when i first started using voip-info.org |
| 00:27.19 | *** join/#asterisk techie (n=techie@c-67-181-184-170.hsd1.ca.comcast.net) |
| 00:27.27 | `Sean | thanks JT your a life saver |
| 00:27.33 | x86 | perhaps it wasn't google-fied back then |
| 00:27.39 | JT | no probs |
| 00:27.40 | De_Mon | shrug |
| 00:27.46 | De_Mon | Yourname` any results? |
| 00:27.48 | antimoof | yay! it looks like my rhel box has a buggy set of includes that prevents zaptel from copiling properly! |
| 00:28.01 | x86 | antimoof: redhat is horrible |
| 00:28.07 | Yourname` | De_Mon: Got sidetracked. Sorry.. two more mins |
| 00:28.15 | antimoof | not my choice, sadly. |
| 00:28.15 | De_Mon | ;) me too |
| 00:28.31 | De_Mon | antimoof are you using some business edition crap? |
| 00:28.50 | x86 | De_Mon: business edition supports gentoo :) |
| 00:29.02 | De_Mon | its still redhat which sucks balls |
| 00:29.02 | `Sean | thanks very very much JT :) |
| 00:29.05 | fujin | how stable is SVN? |
| 00:29.11 | JT | which svn |
| 00:29.17 | antimoof | it's enterprise, which means "it's glossy and overproduced and really kinda shitty, just like the tv show." |
| 00:29.23 | fujin | 1.4 |
| 00:29.36 | `Sean | JT, can i ask what UNIQUEID does ? |
| 00:29.44 | JT | unique call id |
| 00:29.46 | De_Mon | antimoof asterisk has its own distro *now (I think) it's got support options too, I suggest you switch :) |
| 00:30.04 | antimoof | I don't wanna have to be the one to support the base system. |
| 00:30.27 | De_Mon | oh, then go complain to your support ! |
| 00:30.27 | `Sean | Okay :) |
| 00:31.22 | *** part/#asterisk mtaht4 (n=m@66.153.18.42) |
| 00:31.24 | {Sean} | `s suck |
| 00:32.35 | `Sean | JT, i dont mean to be aggrevated, i had to ask, this what would i use for the variable for ARG1, if i wanted the number dailed to be put there? |
| 00:32.37 | `Sean | instead of callerid |
| 00:33.24 | x86 | `Sean: Set(${ARG1} = ${EXTEN}) ? |
| 00:33.24 | JT | sorry |
| 00:33.28 | JT | i meant EXTEN |
| 00:33.30 | JT | not callerid |
| 00:33.44 | `Sean | exten => s,n,MixMonitor(${STRFTIME(${EPOCH},,%Y%m%d-%H%M%S)}-${ARG3}-${UNIQUEID}.wav) |
| 00:33.56 | `Sean | thats what i have, now so what replace ARG1 with EXTEN? |
| 00:34.08 | `Sean | err ARG3 |
| 00:34.13 | De_Mon | ${EXTEN} |
| 00:34.15 | x86 | that wont work on an "s" extension |
| 00:34.32 | JT | umm |
| 00:34.35 | `Sean | this is for outgoing... |
| 00:34.36 | x86 | can't expect ${EXTEN} to contain anything useful on an "s" exten ;) |
| 00:34.38 | `Sean | not for incoming.. |
| 00:34.39 | JT | `Sean: is it in a macro or not? |
| 00:34.44 | `Sean | yes its in a macro |
| 00:34.50 | De_Mon | didn't catch that |
| 00:34.52 | JT | `Sean: i showed you my outgoing line |
| 00:35.06 | JT | you pass it as arguments when calling the macro |
| 00:35.13 | JT | they will be passed in order |
| 00:35.18 | JT | ARG1, ARG2 and so on |
| 00:35.40 | `Sean | So it has to be argument 1 i suppose.. |
| 00:35.49 | `Sean | ,MixMonitor(${STRFTIME(${EPOCH},,%Y%m%d-%H%M%S)}-${ARG1}-${UNIQUEID} |
| 00:35.49 | `Sean | [8:26pm] <JT> .wav) |
| 00:36.01 | `Sean | that was, your outgoing one.. and you stated, that ARG1 = CallerID |
| 00:36.06 | JT | that was my outgoing |
| 00:36.10 | JT | and i meant EXTEN |
| 00:36.19 | *** join/#asterisk anonymouz666 (n=anonymou@200.218.193.6) |
| 00:36.21 | `Sean | OKay :) |
| 00:36.30 | `Sean | So there is no way to name it on the number you dail? |
| 00:36.39 | JT | yes |
| 00:36.40 | JT | EXTEN |
| 00:36.44 | `Sean | Okay |
| 00:36.45 | x86 | `Sean: [outgoing] _XXX.,1,Macro(dialout|${EXTEN}) [macro-dialout] s,1,MixMonitor(${STRFTIME(${EPOCH},,%Y%m%d-%H%M%S)}-${ARG1}-${UNIQUEID}.wav) |
| 00:36.48 | `Sean | sorry :( |
| 00:37.37 | `Sean | x86, http://pastebin.ca/650538 |
| 00:38.10 | JT | my god |
| 00:38.11 | JT | not |
| 00:38.14 | JT | ~thebook |
| 00:38.15 | jbot | well, thebook is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 00:38.17 | JT | read about macros |
| 00:38.54 | x86 | `Sean: why are you starting the recording after the call has terminated? |
| 00:38.54 | JT | ${EXTEN} is the standard dialled number channel variable |
| 00:39.41 | x86 | `Sean: also, re-paste with the context including the part that calls this macro |
| 00:39.55 | `Sean | Okay :) |
| 00:40.15 | `Sean | Okay, will read that aswell one sec let me get you the context :) |
| 00:40.49 | `Sean | x86 http://pastebin.ca/650542 |
| 00:42.02 | x86 | looks good to me |
| 00:42.08 | x86 | what's the problem? |
| 00:42.50 | `Sean | its not recording... |
| 00:43.17 | x86 | right |
| 00:43.24 | `Sean | http://pastebin.ca/650544 |
| 00:43.32 | x86 | 19:38 < x86> `Sean: why are you starting the recording after the call has terminated? |
| 00:43.36 | x86 | did you miss this part? :P |
| 00:43.55 | `Sean | Ahh sorry i didn't get that line :P |
| 00:47.24 | `Sean | thanks alot x86 |
| 00:49.22 | x86 | no problem :) |
| 00:50.21 | riddlebox | is there a way to have a park 701 button, and just press that button when you want to park a call there? |
| 00:56.17 | `Sean | wow that works like a charm thanks very very much JT and X86 |
| 00:56.40 | JT | no probs |
| 00:59.42 | crimethinker | (08:34:04 pm) <jmalicki> i still can't decide if it's entirely a good thing to have engineers design products |
| 01:03.35 | fujin | weee |
| 01:04.43 | flenders | it's hard to keep up with asterisk releases, isn't it? |
| 01:05.15 | fujin | eh |
| 01:05.19 | fujin | going from binary->sourcebuilt |
| 01:05.22 | fujin | so that I can make use of the features |
| 01:05.29 | fujin | and, prepare for the security audit the manager has planned. |
| 01:06.11 | *** join/#asterisk tengulre (n=tengulre@222.90.66.10) |
| 01:07.23 | phix | hey |
| 01:07.32 | phix | how does one setup a g729 licence? |
| 01:07.42 | x86 | contact digium |
| 01:07.57 | x86 | digium.com sells the licenses and gives you instructions on how to add them |
| 01:08.27 | flenders | 9 releases in 2 months... |
| 01:10.04 | TheNewAndy | I'm having some problems detecting DTMF signals when the call is coming from a VOIP line (presumably compressed using something lossy). My asterisk config accepts incoming calls on SIP, and has "dtmfmode=auto" and "relaxdtmf=yes" in sip.conf. Is there anything else I can tweak? |
| 01:10.19 | TheNewAndy | (it works fine when the incoming call is from a normal landline) |
| 01:10.36 | x86 | TheNewAndy: what codec? |
| 01:10.46 | x86 | TheNewAndy: also, try forcing the dtmfmode |
| 01:10.55 | x86 | auto seems to break sometimes |
| 01:11.47 | JT | dtmfmode=rfc2833 |
| 01:11.56 | TheNewAndy | I have allow=ulaw, alaw, gsm and ilbc. I will try removing all but ulaw and see if that works. |
| 01:11.58 | x86 | *nod* |
| 01:14.43 | phix | x86: ok thank you. I am in AU so I contact digium.com.au, I already have a licence (I didn't install it), is there anything speical I need to know adding in a second one? |
| 01:15.45 | x86 | nope |
| 01:15.47 | JT | digium don't have an australian office |
| 01:15.58 | x86 | yeah, you want digium.com |
| 01:16.17 | x86 | not some knock-off aussie company from "down under" |
| 01:16.35 | JT | please don't use that term, it's patronising :P |
| 01:18.24 | fujin | Anyone have an init script for asterisk handy? |
| 01:18.33 | *** join/#asterisk Kumbang (n=kumbang@router-wmi.paume.ITB.ac.id) |
| 01:18.46 | *** join/#asterisk Delvar (n=Delvar@host-83-146-53-46.bulldogdsl.com) |
| 01:23.11 | TheNewAndy | I still can't get dtmf to work. I've set it to only allow ulaw (in my sip.conf), and tried dtmfmode=inband, dtmfmode=rfc2833. I did notice one strange thing though - each time I checked with a softphone and a normal phone (which I believe goes through a VOIP provider with lossy compression), and the softphone (ekiga) worked even with inband, when its preferences claim that it only supports rfc2833. Does this mean my settings in sip.conf are being i |
| 01:23.48 | JT | try to avoid run on sentences, they get terminated by the ircd |
| 01:26.27 | *** join/#asterisk MindTheGap_ (n=iote@bhe201062200012.res-com.wayinternet.com.br) |
| 01:27.24 | *** part/#asterisk TheNewAndy (n=TheNewAn@144.131.134.181) |
| 01:27.33 | *** join/#asterisk TheNewAndy (n=TheNewAn@144.131.134.181) |
| 01:32.07 | *** join/#asterisk Delvar (n=Delvar@host-83-146-53-46.bulldogdsl.com) |
| 01:33.50 | *** join/#asterisk wunderkin (n=wunderki@ip72-223-86-126.ph.ph.cox.net) |
| 01:35.02 | *** join/#asterisk tuxd00d (n=tuxinato@128.187.128.131) |
| 01:40.39 | *** join/#asterisk eatmypiano (n=eatmypia@CPE00195b4be0d6-CM0019477f689a.cpe.net.cable.rogers.com) |
| 01:43.46 | *** part/#asterisk lirakis (n=lirakis@cpe-68-175-38-65.nyc.res.rr.com) |
| 01:46.23 | Sweeper | hey, he was punctuating properly :P |
| 01:46.32 | Sweeper | he just needs a better client.... |
| 01:47.30 | *** join/#asterisk glacid (i=unknown@evool.com) |
| 01:47.52 | glacid | is anyone here using asterisk with a Cisco 7971? |
| 01:48.54 | snuff-work | i have |
| 01:51.56 | fujin | what controls wether the asterisk console has colour or not? |
| 01:52.06 | TheNewAndy | oh, I assumed that it was just annoying. sorry |
| 01:52.09 | JT | wether, eh ;) |
| 01:52.12 | fujin | if I asterisk -r, I don't see any colour, but if I launch manually with asterisk -cvvv I see colour |
| 01:52.15 | fujin | whether |
| 01:52.25 | JT | yeah it's pesky console color |
| 01:52.36 | fujin | what causes it? |
| 01:52.40 | fujin | It'd be super to have color. |
| 01:53.11 | JT | i think you need to use a script |
| 01:53.20 | JT | i've never really bothered |
| 01:53.33 | fujin | a script which replaces what asterisk -r does? |
| 01:53.45 | JT | to start asterisk |
| 01:53.50 | JT | starting it is the important bit |
| 01:53.57 | TheNewAndy | So, I think my dtmf settings in sip.conf are being ignored. |
| 01:54.13 | fujin | JT: I'm starting it with -p -U asterisk |
| 01:54.13 | JT | TheNewAndy: why do you think that? |
| 01:54.17 | glacid | what happened to theoldandy? |
| 01:54.31 | fujin | the only thing I can think of is if the user asterisk somehow doesn't have colour permissions (lol) |
| 01:54.52 | TheNewAndy | I have set dtmfmode=inband, and a client (ekiga) which only sends rtfXXXX protocol stuff still works |
| 01:54.58 | JT | it needs to have a termtype of xterm, among other things |
| 01:55.06 | fujin | I see |
| 01:55.07 | TheNewAndy | But a landline doesn't work at all |
| 01:55.09 | fujin | export TERM=xterm? |
| 01:55.28 | JT | TheNewAndy: i don't think ekiga only does rfc2833 |
| 01:55.34 | TheNewAndy | glacid: the old andy has the username andy |
| 01:55.40 | glacid | ah |
| 01:55.48 | fujin | JT: is TERM=linux not sufficient? that is what the init script is calling. |
| 01:56.09 | glacid | you guys wouldn't happen to know where to get a cisco 7970 firmware without paying cisco lots of money, would you? |
| 01:56.26 | TheNewAndy | JT: well when I make a call from one Ekiga to another and press the buttons, I don't hear anything. So I don't think it does inband. |
| 01:56.34 | JT | fujin: i don't think so |
| 01:57.14 | fujin | perhaps start-stop-daemon is causing it to *not* pass the termtype. |
| 01:57.17 | *** join/#asterisk Omer^ (n=Omer@203.81.208.43) |
| 01:57.28 | fujin | JT: can I print $TERM inside of the asterisk c onsole, somehow? with an extension perhaps? |
| 01:57.34 | TheNewAndy | I also looked at a packet dump, and I can at least say that it sends the rfc2833 events (whether or not that means it doesn't send the tones is another matter) |
| 01:57.45 | Omer^ | Warning: unlink(cache/sessionsFile.txt): Permission denied in /var/www/html/maint/includes/application_top.php on line 31 |
| 01:57.52 | Omer^ | any one have any idea about this |
| 01:58.22 | fujin | ah, env |
| 01:58.22 | flenders | fujin: I have TERM=linux and mine does colours |
| 01:58.46 | fujin | flenders: yes, the init script which calls start-stop-daemon has TERM=linux also, what I'm wondering is if TERM=linux is being passed through start-stop-daemon |
| 01:58.50 | fujin | i assume it's not. |
| 01:58.51 | JT | fujin: ${ENV(var)} |
| 01:58.54 | fujin | because I get color when I launch manually |
| 01:58.55 | fujin | will have a tutu. |
| 01:58.58 | flenders | when you're starting the CLI, just do a 'asterisk -rc' |
| 01:58.59 | JunK-Y | fujin: !echo $TERM |
| 01:59.08 | fujin | or that |
| 01:59.12 | fujin | term=linux. |
| 01:59.15 | fujin | blasted thing. |
| 01:59.40 | fujin | I should just write a new script for init. |
| 01:59.41 | JT | flenders: that does not work |
| 01:59.46 | JT | asterisk -rc |
| 01:59.49 | JT | black and white |
| 01:59.56 | fujin | yeah ^^. |
| 02:00.05 | flenders | works for me |
| 02:00.16 | JT | i think it's too late once the daemon has decided to start in black and white |
| 02:00.18 | fujin | flenders: how is asterisk being launched on your system, from init script? |
| 02:00.22 | flenders | yeah |
| 02:00.30 | fujin | would you pastebin the init script for me? |
| 02:00.35 | flenders | sure |
| 02:00.52 | fujin | I've just transitioned from binary->source package, so, using the Ubuntu init script *which is probably broken* |
| 02:01.33 | fujin | <PROTECTED> |
| 02:01.35 | flenders | fujin: http://pastebin.ca/650641 |
| 02:01.37 | fujin | how about that.. |
| 02:03.04 | [hC] | so, if i want to interface an FXS style voip device (think spa3000) so that i can reach it over IP, and it places an outbound local call via POTS, but instead of POTS, its ISDN, is there a portable device that does this? |
| 02:03.43 | [hC] | by FXS of course i meant FXO |
| 02:04.10 | flenders | fujin: I copied that init script from the debian packages a while ago |
| 02:04.16 | glacid | anyone have any idea where i can get cmterm-7970_7971-sip.8-2-2.cop without a cisco login |
| 02:04.39 | JT | [hC]: are you asking if SIP to BRI gateways exist? |
| 02:04.43 | *** join/#asterisk litage_ (n=nick@70.55.220.203.static.comindico.com.au) |
| 02:04.45 | fujin | It's very different to the ubuntu one, I'll have a toy with it - thanks ! |
| 02:04.56 | litage_ | in a sip packet, which field(s) contain the caller-ID? |
| 02:05.08 | [hC] | JT: BRI is how you would call that? like a european phone line? (I take it they have isdn lines to their house instead of pots, in some places) |
| 02:05.14 | JT | From: i think |
| 02:05.37 | JT | [hC]: different parts of europe are different, and it's usually a choice. |
| 02:05.41 | JT | and BRI is not europe only |
| 02:05.55 | [hC] | JT: this is norway, and all i know about it is they call their lines just plain old ISDN. |
| 02:06.06 | JT | well they really mean BRI |
| 02:06.10 | [hC] | ok |
| 02:06.52 | x86 | yeah, theres no way everyone has PRI running to their house ;) |
| 02:07.21 | litage_ | n/m, it's the "From" tag |
| 02:07.22 | *** part/#asterisk litage_ (n=nick@70.55.220.203.static.comindico.com.au) |
| 02:08.04 | JT | yes, like i already said, litage |
| 02:11.07 | *** join/#asterisk GoldFingaZ (n=wt5@bas13-ottawa23-1088841481.dsl.bell.ca) |
| 02:11.22 | x86 | 38% [===============================================> ] 22,397,968 1.02M/s ETA 00:32 |
| 02:14.12 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 02:14.51 | phix | channel.c:2571 ast_request: No translator path exists for channel type Zap (native 68) to 256 |
| 02:16.45 | phix | app_dial.c:1056 dial_exec_full: Unable to create channel of type 'Zap' (cause 0 - Unknown) |
| 02:17.59 | JT | x86: 6,537,394 3.90M/s |
| 02:18.00 | JT | :D |
| 02:18.12 | flenders | JT: that's your colo box! |
| 02:18.13 | phix | what the hell does that mean? |
| 02:18.25 | JT | phix: codec mismatch |
| 02:18.28 | JT | flenders: lies! |
| 02:18.37 | flenders | :D |
| 02:18.54 | phix | JT: which means? |
| 02:19.13 | JT | phix: it means only allow codecs that you can actually transcode |
| 02:19.31 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 02:19.42 | phix | JT: well I have recently updated asterisk (from source debian package) |
| 02:20.06 | phix | asterisk was using the debian package before that too, so I need to do something special for g729 support? |
| 02:20.29 | JT | do you have g.729 licenses? |
| 02:20.34 | phix | I have one |
| 02:20.38 | phix | I need to get another one |
| 02:22.17 | phix | I have just upgraded, so I havn't changed any configuration files, just asterisk binary and librarbies |
| 02:22.25 | phix | libraries even |
| 02:23.38 | phix | JT: what do you suggest? |
| 02:25.16 | JT | fix your g.729 setup... |
| 02:26.06 | phix | .... |
| 02:26.16 | phix | how? what docs do I read? |
| 02:26.27 | phix | I dont even know what is wrong with it to begin with |
| 02:26.54 | JT | maybe the licenses are not registering properly |
| 02:27.00 | JT | you can call digium support you know |
| 02:27.23 | phix | hmm I need a support ID? |
| 02:28.02 | JT | i have no clue |
| 02:28.03 | phix | what files is the licence in any way? |
| 02:28.11 | JT | do you really need to ask everything |
| 02:28.13 | fujin | where di dyou buy the license from? |
| 02:29.00 | phix | I see the problem, it is telling me to do something in the startup script, when updating asterisk this file was replaced |
| 02:29.19 | *** join/#asterisk l3jj (n=chatzill@dsl093-001-038.det1.dsl.speakeasy.net) |
| 02:30.44 | JT | reading instructions is a good idea :) |
| 02:32.36 | phix | yeah, but it didn't help |
| 02:37.00 | *** join/#asterisk DarkRift (i=dark@bas10-montreal02-1177582280.dsl.bell.ca) |
| 02:37.06 | phix | <PROTECTED> |
| 02:37.09 | phix | ok so it has support for it |
| 02:37.17 | fujin | Can I boot a user, from the console? |
| 02:37.25 | fujin | sip blow_this_user_out_of_the_water xxx |
| 02:41.48 | De_Mon | man. korean's kick all kinds of ass in starcraft |
| 02:42.23 | De_Mon | fujin delete him from sip.conf and reload |
| 02:42.26 | De_Mon | ^_^ |
| 02:42.45 | *** join/#asterisk CrashSys (n=kumba@158-211.187-72.tampabay.res.rr.com) |
| 02:42.52 | JT | you can kill calls with soft hangup |
| 02:42.58 | phix | <PROTECTED> |
| 02:43.03 | phix | ok that is in there |
| 02:43.20 | fujin | thanks |
| 02:43.22 | fujin | hrm |
| 02:43.28 | fujin | that different init script didn't work |
| 02:44.30 | fujin | Still no colour. bloody hell. |
| 02:45.16 | De_Mon | fujin -rc didn't do it? |
| 02:45.56 | fujin | nope |
| 02:46.02 | fujin | term=linux, too. |
| 02:46.17 | De_Mon | oh hrm... come to mention it my console isn't in color either |
| 02:46.21 | fujin | heh. |
| 02:46.24 | fujin | I wonder what it is. |
| 02:46.42 | phix | grrrr |
| 02:49.55 | De_Mon | -c and -r arn't congruent |
| 02:49.55 | jql | 2.4.10 mysteriously stopped giving me a color console, so I wanted to check what was up with you. alas... |
| 02:55.00 | fujin | What the hell is extensions.ael? |
| 02:55.10 | De_Mon | fujin black magic |
| 02:55.17 | CrashSys | Flux Capacitor |
| 02:55.20 | fujin | It looks leet, how does it work. |
| 02:55.23 | CrashSys | Michael J. Fox is inside |
| 02:55.29 | CrashSys | Uranium |
| 02:55.34 | CrashSys | and 1.1 Jigawatts |
| 02:55.37 | fujin | and can I run extensons.conf + extensions.ael at the same time? |
| 02:57.17 | CrashSys | I believe so |
| 02:57.17 | De_Mon | fujin I think so... It will probably complain if you setup the same contexts/priorities |
| 02:57.27 | CrashSys | It's just a different way to write dialplans |
| 02:57.31 | De_Mon | probably just same prilorities |
| 02:57.45 | CrashSys | Same rules apply, just different syntax |
| 02:57.45 | De_Mon | extensions/priorities I mean |
| 02:57.50 | fujin | ohmygodohmygod. |
| 02:57.56 | fujin | jesus christ |
| 02:58.00 | CrashSys | Where? |
| 02:58.02 | fujin | how did I not know about this? |
| 02:58.03 | De_Mon | please clean that up |
| 02:58.17 | De_Mon | I WANT COLOR |
| 02:58.44 | De_Mon | sudo asterisk -c gives color, but starting asterisk then attaching (-r) gives me no love |
| 02:59.11 | antimoof | color? is that some newfangled thing that's halfway between black and white? |
| 02:59.35 | De_Mon | no, its what happens when black and white kill eachother |
| 02:59.45 | De_Mon | they make red and blue and oh my |
| 03:00.29 | De_Mon | hrm, I wonder if my startup script is setting a non-color TERM |
| 03:04.38 | *** join/#asterisk shido6 (n=shido6@d221-68-200.commercial.cgocable.net) |
| 03:06.29 | *** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 03:06.37 | phix | so, can any one help me with g729 codec? |
| 03:06.38 | *** join/#asterisk |dennis| (n=dennis@200.32.233.84) |
| 03:08.26 | red9012 | so I like to issue a ring tone, but only with one command. the ringing() cmd requires an additional wait() cmd so this option is not possible. |
| 03:08.48 | Strom_M | red9012: that's because Ringing() sends the equivalent of the ISDN ALERTING message |
| 03:15.28 | red9012 | ok, then is there a way to issue two commands on a single priority? |
| 03:15.55 | red9012 | ie. ringing() and a wait(3) on a single context/exte/priority. |
| 03:16.02 | JT | red9012: why on earth can't you use an additional priority? that's silly |
| 03:16.04 | Strom_M | red9012: why? |
| 03:16.58 | *** join/#asterisk mtaht4 (n=m@66.153.18.42) |
| 03:17.01 | red9012 | because the context/exten/priority is programmatically built in a loop |
| 03:17.12 | JT | your loop is programatically broken |
| 03:17.36 | JT | if it can't handle another priority, it is broken |
| 03:17.39 | De_Mon | you are creating a dialplan programatically? |
| 03:18.01 | red9012 | all commands are issued on one line, except this. |
| 03:18.10 | red9012 | de_mon-- yes |
| 03:18.18 | Strom_M | huh? |
| 03:18.19 | JT | yes, rethink reality, your view is warped :) |
| 03:18.27 | De_Mon | uh |
| 03:18.28 | De_Mon | ok |
| 03:18.41 | l2trace9999 | anyone know how to pause a dynamically added agent ? |
| 03:19.07 | De_Mon | red9012 pick a new language if the one you are currently using doesn't let you add these two commands as separate priorities |
| 03:20.04 | De_Mon | I wonder, why are you ringing and wait(3)ing in the first place. Almost sounds like your trying to dial without the dial app |
| 03:20.19 | red9012 | my prog runs a whole bunch of stuff, and builds a complete dial plan resulting in some cases 100+ contexts/exten/priority |
| 03:20.32 | JT | and? |
| 03:20.44 | De_Mon | l2trace9999 AddQueueMember |
| 03:20.52 | De_Mon | oh you said pause |
| 03:21.03 | *** join/#asterisk ToyMan (n=Stuart@user-12lcqol.cable.mindspring.com) |
| 03:21.03 | De_Mon | that would be PauseQueueMember |
| 03:21.11 | l2trace9999 | yes |
| 03:21.21 | l2trace9999 | i have a dial plan setup |
| 03:21.34 | l2trace9999 | to call PauseQueueMember |
| 03:21.56 | l2trace9999 | but I am guessing that I am getting the uri to agent wrong |
| 03:22.19 | De_Mon | l2trace9999 do a show queue members to find out what the agent's name is |
| 03:22.38 | l2trace9999 | i did that |
| 03:22.39 | De_Mon | l2trace9999 you can pause them from the cli and let command completion(tab) help too |
| 03:22.53 | l2trace9999 | bah |
| 03:22.57 | TheNewAndy | as a follow up to my dtmf issue about an hour ago, I've got it working. My settings were being overridden. |
| 03:23.00 | l2trace9999 | i think i got it |
| 03:23.12 | JT | TheNewAndy: how did you fix it? |
| 03:23.22 | l2trace9999 | i think i fat fingered my dial plan |
| 03:23.32 | l2trace9999 | i missed a } |
| 03:24.32 | l2trace9999 | those dam coders !!!!!! |
| 03:24.39 | l2trace9999 | they know i have fat fingers |
| 03:24.45 | implicit | 2 |
| 03:24.49 | implicit | 2 |
| 03:24.54 | red9012 | ideally the ringing command should take a time parameter as input. ie ringing(5) for 5secs. |
| 03:24.54 | JT | implicit: huh? |
| 03:25.04 | implicit | mistype |
| 03:25.06 | implicit | whats up |
| 03:25.29 | Strom_M | red9012: no, the ringing() command sends an ALERTING message or equivalent and then moves on |
| 03:25.29 | JT | red9012: ringing is more about signalling than inband audio |
| 03:29.58 | TheNewAndy | JT: when I was changing the dtmfmode settings, they were being changed back to rfc2xxx somewhere else in my dialplan. |
| 03:30.21 | TheNewAndy | So fixing it was just working out where I should actually be changing the settings, and changing it there |
| 03:30.29 | JT | so you were setting it in global but you had something else in the peer definition |
| 03:30.43 | TheNewAndy | yep (and I'm incapable of remembering 4 digit numbers) |
| 03:34.50 | red9012 | the privacy mode of dialI() cmd had dtmf recognition problems. are those fixed? |
| 03:35.39 | JT | wtf is dialI()? |
| 03:35.59 | red9012 | dial() not dialI() |
| 03:36.27 | JT | Dial, okay |
| 03:41.25 | *** join/#asterisk Stormfr (n=Stormfr@sgc91-2-82-237-76-2.fbx.proxad.net) |
| 03:41.47 | De_Mon | fujin use got help figurin this out, safe_asterisk for color |
| 03:42.09 | De_Mon | move 'use' to after the , |
| 03:42.56 | Stormfr | anyone have issue with agi and channel variable not anymore filled ? Noop even in agi return empty hangupcause or dialstatus |
| 03:46.43 | fujin | De_Mon: really? |
| 03:47.29 | JT | that's pretty much what flenders already said, use safe_asterisk, but okay |
| 03:47.39 | fujin | must have missed that |
| 03:47.47 | fujin | safe asterisk doesn't appear to launch correctly on my system |
| 03:47.52 | JT | he pastebinned his init script |
| 03:48.41 | fujin | oh right |
| 03:48.41 | fujin | lol |
| 03:49.34 | JT | it called safe_asterisk |
| 03:52.39 | fujin | christ, I hate it when other deparments are testing things on my phone system. |
| 03:52.42 | fujin | I just want to STOPPP ITTT |
| 03:52.57 | fujin | Is anyone aware of some script fu which will convert extensions.conf to extensions.ael? |
| 03:53.01 | fujin | I'd like to make the change. |
| 03:53.11 | `Sean | ael? |
| 03:53.14 | JT | no |
| 03:53.20 | JT | what's the point, fujin ? |
| 03:54.03 | De_Mon | fujin yeah something in the safe_asterisk script does it |
| 03:54.03 | fujin | it looks like I can do the ridiculous things that I'm being asked to do, better, |
| 03:54.05 | codefreeze | fujin: feeling like being a guinea pig? |
| 03:54.22 | JT | fujin: it's nothing that spectacular |
| 03:54.23 | fujin | definitely |
| 03:54.26 | De_Mon | fujin you can do all those crazy things now! |
| 03:54.39 | De_Mon | fujin all ael does is convert it back to a normal dialplan |
| 03:54.44 | JT | exactly |
| 03:54.53 | De_Mon | jt probably told me that :) |
| 03:55.05 | JT | not sure |
| 03:55.07 | JT | :P |
| 03:55.34 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 03:56.01 | codefreeze | fujin: I'm serious; I've got the start of something that will convert extensions.conf => extensions.ael |
| 03:56.21 | JT | codefreeze: just so asterisk can convert it back for you? |
| 03:56.22 | fujin | codefreeze: yeah |
| 03:56.39 | codefreeze | JT: Silly boy! of course! |
| 03:56.55 | De_Mon | fujin write what you want in ael, then do a show dialplan in asterisk to know how to do it in extensions.conf :) |
| 03:58.34 | *** join/#asterisk Kumbang (n=kumbang@router-wmi.paume.itb.ac.id) |
| 03:58.42 | codefreeze | fujin: I'm working on a branch, http://svn.digium.com/svn/asterisk/team/murf/bug_7638 |
| 03:59.13 | fujin | aarrrh |
| 03:59.34 | codefreeze | fujin: you know how to do checkouts, right? |
| 03:59.39 | fujin | of course |
| 03:59.50 | fujin | but I unfortunately don't have the time at the moment to do so |
| 03:59.59 | fujin | let me get back to you in a few |
| 04:00.13 | fujin | (god oh god, why did I raise my hand when someone asked 'who knows asterisk') |
| 04:00.51 | codefreeze | OK, well, when you get a chance; do a configure; make menuselect; make; make install, then look in the utils dir |
| 04:00.56 | Juggie | codefreeze: pm. |
| 04:01.08 | codefreeze | hey, Juggie! |
| 04:01.29 | De_Mon | fujin hahahaha new project leader? |
| 04:01.37 | red9012 | whats the right term for describing the ring sound heard after dialing a number? |
| 04:01.46 | Strom_M | red9012: alerting tone |
| 04:01.46 | De_Mon | red9012 ringing |
| 04:01.50 | De_Mon | damn! |
| 04:02.01 | Juggie | codefreeze: see my pm. |
| 04:02.42 | red9012 | ringtone? |
| 04:03.20 | Strom_M | red9012: alerting tone |
| 04:03.51 | JT | private message |
| 04:03.54 | fujin | De_Mon: and everything else that goes with it |
| 04:03.55 | Juggie | :) |
| 04:04.03 | fujin | I'm fortunately actually not managing the project |
| 04:04.09 | fujin | just being responsible for every part of asterisk |
| 04:04.13 | fujin | and everything that interfaces with it |
| 04:04.23 | JT | potentially worse ;) |
| 04:04.35 | De_Mon | the voip expert |
| 04:04.54 | fujin | heh, yeah. :\ |
| 04:04.57 | De_Mon | any nortel crap in your future? |
| 04:05.38 | De_Mon | damn, watchin these starcraft pro games is is better than football! |
| 04:07.42 | fujin | safe_asterisk doesn't give me colour |
| 04:08.12 | De_Mon | your using 1.4.10? |
| 04:08.18 | De_Mon | you're |
| 04:08.26 | fujin | yup |
| 04:09.09 | De_Mon | worked for me, gonna try something else |
| 04:09.36 | Corydon76-home | The terminal on which safe_asterisk was started has to support color |
| 04:09.42 | fujin | it does |
| 04:10.04 | Corydon76-home | It wasn't started, for example, from the system startup scripts? |
| 04:10.24 | Corydon76-home | Because that's a good way for safe_asterisk not to know its terminal type |
| 04:10.38 | Corydon76-home | (which is the entire basis for color) |
| 04:10.43 | Cyllene | Hey Corydon76-home. |
| 04:10.52 | Cyllene | Did you see bug #10412? |
| 04:10.56 | fujin | hrm |
| 04:10.59 | fujin | when I run safe_asterisk |
| 04:11.03 | fujin | I get a werid error, check this out |
| 04:11.10 | fujin | /usr/sbin/safe_asterisk: 175: Syntax error: Bad fd number |
| 04:11.10 | Corydon76-home | Cyllene: no, and I probably won't, either |
| 04:11.22 | Cyllene | Eh, ok. |
| 04:11.54 | *** join/#asterisk DrAk0 (n=ljd@nelug/coreteam/luisjose) |
| 04:11.56 | De_Mon | fujin if you run asterisk -c and then connect (asterisk -r) from another terminal i get color too |
| 04:12.07 | De_Mon | s/you/i/ |
| 04:12.12 | De_Mon | damn u jbot |
| 04:12.15 | fujin | yep |
| 04:12.18 | fujin | I can confirm that, De_Mon |
| 04:12.24 | fujin | safe_asterisk doesn't even launch for me |
| 04:12.30 | *** join/#asterisk bbryant (n=12243@c-68-59-20-153.hsd1.sc.comcast.net) |
| 04:12.35 | Corydon76-home | fujin: what platform? |
| 04:12.41 | fujin | ubuntu linux, i686 |
| 04:12.54 | Corydon76-home | fujin: ln -sf /bin/bash /bin/sh |
| 04:13.11 | fujin | aha |
| 04:13.14 | fujin | it's pointing at dash |
| 04:13.27 | fujin | bingo |
| 04:13.28 | fujin | wd Corydon76-home |
| 04:13.29 | fujin | ty |
| 04:13.50 | fujin | COLOOR! |
| 04:13.52 | fujin | awesomeness. |
| 04:13.52 | fujin | <3 |
| 04:13.53 | Corydon76-home | Yeah, I still don't understand what Ubuntu developers were thinking when they did that |
| 04:14.10 | fujin | Ubuntu developers are retarded |
| 04:14.17 | fujin | I'm just dealing with the tools I've been unfortunately given. |
| 04:14.41 | Corydon76-home | If they really wanted to speed up their wretched boot process, they should have changed the top of all their init scripts, not made all their users suffer |
| 04:15.13 | *** join/#asterisk CVirus (n=GoD@196.218.151.52) |
| 04:15.22 | fujin | aye, wretched it is. |
| 04:15.26 | fujin | I much prefer Gentoo's |
| 04:15.54 | CVirus | Gentoo rocks |
| 04:16.09 | De_Mon | rawr my linux distro is better than yours |
| 04:16.39 | fujin | heh |
| 04:16.42 | fujin | what, you're a ubuntu fan? |
| 04:16.42 | Juggie | centos is the way to go :) |
| 04:16.46 | JT | Corydon76-home: what did they do on ubuntu? |
| 04:16.49 | Juggie | codefreeze: see pm again :) |
| 04:16.55 | JT | haha @ centos |
| 04:17.04 | fujin | dont' even joke, lol |
| 04:17.12 | Corydon76-home | JT: Ubuntu developers linked /bin/sh to /bin/dash |
| 04:17.13 | fujin | wouldn't catch me dead on an RPM system |
| 04:17.29 | JT | Corydon76-home: what does /bin/dash do? |
| 04:17.29 | fujin | what the hell is dash? |
| 04:17.31 | Juggie | <3 RPM |
| 04:17.31 | CVirus | LOL @ Corydon76-home |
| 04:17.44 | De_Mon | no ubuntu sucks just because they are using .deb |
| 04:17.50 | JT | ... |
| 04:17.55 | fujin | deb > rpm |
| 04:17.56 | Corydon76-home | dash is a tiny clone of the original Bourne shell |
| 04:17.58 | De_Mon | dash is not a shell (I duno) |
| 04:18.10 | fujin | dash fails at interpreting safe_asterisk |
| 04:18.18 | Corydon76-home | Unfortunately, it's not 100% compatible with bash |
| 04:18.22 | De_Mon | fujin debian > ubuntu |
| 04:18.33 | fujin | heh |
| 04:18.45 | fujin | I'm a junior engineer. Don't get to make decisions like that |
| 04:18.51 | fujin | if it were me, the entire ISP would run gentoo |
| 04:18.56 | fujin | or a bsd |
| 04:19.18 | Strom_M | Corydon76-home: linux nub question: how easy would it be to have safe_asterisk check if it's running in dash, and if so, have it spawn another copy of itself that runs under bash? |
| 04:19.19 | JT | my gawd |
| 04:19.23 | JT | running an isp on gentoo |
| 04:19.25 | JT | CRACK |
| 04:19.27 | fujin | keke. |
| 04:19.42 | antimoof | if it actually requires bash features, then (IMNSHO) it should damn well invoke bash explicitly. |
| 04:20.01 | fujin | Concur. |
| 04:20.05 | Strom_M | yes, because clearly your e-mail server needs the latest hemorrhaging-edge version of postfix |
| 04:20.07 | fujin | It has !#/bin/sh at the top. |
| 04:20.19 | fujin | err. |
| 04:20.23 | fujin | sha-bang, not bang-sha. |
| 04:20.40 | fujin | Perhaps I should post a bug? |
| 04:20.41 | JT | Strom_M: "yes boss, installing that critical security update... as soon as it finishes compiling" |
| 04:20.43 | fujin | anyone have SVN access? |
| 04:20.52 | fujin | JT: Get faster machines! |
| 04:21.12 | JT | heh |
| 04:21.13 | fujin | VMOTION! |
| 04:21.21 | JT | can't wait till the quad core opterons ship |
| 04:21.41 | JT | ... they are superior, especially in fpu |
| 04:21.48 | JT | not to mention memory bandwidth |
| 04:21.56 | JT | and all cores are on the one die |
| 04:22.05 | JT | but xeon is very price competitive |
| 04:22.29 | Qwell | the quad core xeons are nice... |
| 04:22.30 | Juggie | if only AMD would fix their mobile processors |
| 04:22.50 | fujin | theyc ertainly are |
| 04:22.53 | Juggie | i have 3 1U dual, quad core xeon boxes i just got :) |
| 04:22.53 | JT | ec xeons are nice |
| 04:22.57 | JT | qc |
| 04:22.59 | fujin | all of our VM hosts are running em.. $20k each for one of them |
| 04:23.04 | JT | but qc opterons should be even nicer |
| 04:23.11 | Juggie | 8cpu (2x4) 8gb ram each ;) |
| 04:23.20 | fujin | JT: I'd rather have intels VT on them |
| 04:23.27 | JT | Juggie: you mean 2 cpu? |
| 04:23.30 | JT | uhuh |
| 04:23.31 | De_Mon | poor amd |
| 04:24.17 | Juggie | JT, 2 physical cpus x 4 cores |
| 04:24.17 | JT | imho core != cpu |
| 04:24.34 | JT | Juggie: 2 cpus, 4 dies, 8 cores, 8 threads |
| 04:24.40 | Juggie | ok then, 8core 8gb ram, either way, they rock ;) |
| 04:24.56 | JT | yep |
| 04:25.03 | JT | thinking of buying some to play with |
| 04:26.11 | Juggie | they arnt that expensive, and super small |
| 04:26.24 | JT | they are quite cheap |
| 04:26.44 | JT | Juggie: if density is your concern, have you seen the supermicro 1UTwin? |
| 04:27.36 | Juggie | JT, nope.. but density is not really our concern, im just amazed how much goes in the 1U. |
| 04:27.43 | Juggie | they are HP DL360 G5's |
| 04:27.54 | _mm_ | ~phones |
| 04:27.54 | 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. |
| 04:27.57 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
| 04:28.06 | JT | Juggie: the 1UTwin fits 2 X 2 CPU Xeon motherboards in 1 RU |
| 04:28.15 | JT | so that's 4 quad core xeons |
| 04:28.20 | JT | in 2 systems |
| 04:28.22 | Juggie | nice. |
| 04:28.29 | wunderkin | ya i saw that |
| 04:28.35 | Juggie | but not much room for storage etc |
| 04:28.47 | De_Mon | SAN |
| 04:28.48 | fujin | http://img377.imageshack.us/img377/2208/vmhu5.jpg |
| 04:28.49 | fujin | ^^ |
| 04:28.51 | De_Mon | NAS |
| 04:28.53 | JT | err 4 HDDs total is PLENTY |
| 04:28.56 | JT | exactly |
| 04:29.03 | JT | they're obviously designed for a cluster |
| 04:29.09 | JT | RAID1 on each, i'm happy |
| 04:29.19 | De_Mon | thats what EMC is for :) |
| 04:29.46 | De_Mon | we just bought like 5tb storage array last week. 10krpm drool |
| 04:30.06 | JT | how big were the drives? |
| 04:30.27 | fujin | runnign 3tb 15k spin here |
| 04:30.29 | fujin | win. |
| 04:30.32 | De_Mon | don't recall |
| 04:30.33 | fujin | it's so sexy. |
| 04:30.54 | JT | De_Mon: then what's the point of mentioning their rpm? |
| 04:31.17 | Juggie | JT, faster |
| 04:31.31 | De_Mon | uh, fiber channel drives, 10krpm |
| 04:31.36 | JT | Juggie: err, it's only ONE element of the hard drive speed equation. |
| 04:31.42 | JT | De_Mon: size of each drive is...? |
| 04:31.58 | Juggie | JT, yes thats true, but they are still faster. |
| 04:32.10 | JT | Juggie: faster than what? a statement like that means nothing |
| 04:32.20 | JT | "200% better than everything else*" |
| 04:32.24 | wunderkin | faster than snot? |
| 04:32.25 | De_Mon | fast? I duno 350, 500.. somewhere in there |
| 04:32.50 | Juggie | JT, obviously its not going to scale exactally... |
| 04:33.19 | fujin | well, you have 4gbit access over fibre channel to the drivers, the ONLY limiting factor (speeds <4gbit) is the spindle speed |
| 04:33.19 | De_Mon | there was talk about 750gig drives at 7500 and then the 10k's at something else... |
| 04:33.19 | fujin | and you can increase that by striping across the spindles |
| 04:33.19 | Juggie | but there is a performance gain from 7200 - 10000 - 15000 rpm |
| 04:33.34 | JT | Juggie: at a given density, sure. |
| 04:33.45 | JT | fujin: it is not the only factor. |
| 04:33.49 | De_Mon | all I know is it was less 500 or less |
| 04:33.53 | JT | fujin: data density is very important |
| 04:33.56 | Juggie | sure its possible one manafacture may have a 7200pm drive, thats beating a 10000rpm drive. |
| 04:34.08 | JT | Juggie: err it's better than that |
| 04:34.31 | Juggie | but all things being equal given good drives, there is a performance gain to be had.. it depends on your data. |
| 04:34.36 | JT | the seagate 750GB 7200rpm drive is about the same speed as a 146GB 15krpm Raptor |
| 04:34.40 | De_Mon | 72? hoom, it must be getting late my brain is functioning worse than normal |
| 04:34.51 | JT | remember, extra revs == extra power too |
| 04:35.02 | De_Mon | more heat as well |
| 04:35.05 | JT | yes |
| 04:35.34 | Juggie | yes, like anything there is a finate performance gain to be had. |
| 04:35.47 | Juggie | just like dual core != 200% faster |
| 04:35.49 | De_Mon | hell, at this point I wouldn't be surpprised if they were 15krpm. I'll redeem myself after a good nights rest |
| 04:36.01 | JT | platter quantity and platter density are very important if you care about speed |
| 04:36.23 | Juggie | i coudnt agree more. |
| 04:37.50 | *** join/#asterisk bintut (n=bintut@cm18.gamma181.maxonline.com.sg) |
| 04:38.18 | De_Mon | Juggie so duel core != 200% faster :((( |
| 04:38.21 | De_Mon | jk |
| 04:38.34 | JT | back to the 1UTwin, I'd only do it in a cluster with other systems that can act as backup |
| 04:38.55 | JT | i don't trust 1 computer on one PSU, let alone 2 computers on one PSU |
| 04:39.13 | De_Mon | you can't get 2 psu's in 1U? |
| 04:39.27 | JT | De_Mon: not with 2 PCs in that 1U |
| 04:39.27 | De_Mon | big brother after dark... |
| 04:39.44 | De_Mon | JT its 2 pc's on 1 psu right? |
| 04:39.53 | JT | yes |
| 04:39.57 | JT | a 980W PSU |
| 04:40.03 | De_Mon | bwaa |
| 04:40.10 | JT | ? |
| 04:40.19 | De_Mon | I was expecting 800 at the most |
| 04:40.24 | JT | ah |
| 04:40.53 | JT | hopefully it shouldn't draw more than 300-500W during normal operation |
| 04:41.05 | De_Mon | i'm gona ask newegg what therir highest watt psu is |
| 04:41.21 | JT | some servers have over 1kW per module |
| 04:41.21 | nick125 | De_Mon: 1200W IIRC |
| 04:41.37 | fujin | you can get two psu's in 1u, easy |
| 04:41.42 | fujin | our 1950's all have two |
| 04:41.57 | JT | and if you want to talk about PSUs in general, telco PSUs blow everything in IT out of the water |
| 04:42.08 | JT | fujin: not with 2 PCs inside. no room. |
| 04:44.06 | De_Mon | 1200W for $350, just. I never would have imagined anything short of network storage would need something like that |
| 04:44.38 | De_Mon | my world just got bigger |
| 04:44.57 | JT | heh |
| 04:45.16 | JT | De_Mon: i bet you that telco PSUs kill the power density of that |
| 04:45.49 | De_Mon | JT the same size as a pc psu? |
| 04:45.57 | JT | you can get modern telco power supply modules that are half the size of a tissue box and output 3.6kW at -48VDC |
| 04:46.03 | JT | using passive cooling |
| 04:46.06 | JT | no fans... |
| 04:46.16 | JT | sitting at an efficiency of 96-98% |
| 04:47.00 | De_Mon | what makes all the noise in a telco switching room if not power supplies |
| 04:47.16 | fujin | PCs? |
| 04:47.21 | red9012 | how do i generate a busy tone? |
| 04:47.21 | De_Mon | no |
| 04:47.25 | JT | the power supplies are almost never in the same room as the switches in a big exchange |
| 04:47.43 | De_Mon | so the switches themselves don't actually have bult in psus |
| 04:47.50 | JT | cooling systems, servers, access network equipment |
| 04:47.53 | JT | hahah god no |
| 04:48.02 | De_Mon | computers need to do that |
| 04:48.04 | JT | that's the whole principle of -48VDC |
| 04:48.20 | JT | run high current -48VDC through the whole exchange |
| 04:48.34 | JT | link it back to a bank or two of 2V lead acid cells |
| 04:48.41 | JT | each between 500 and 2000Ah |
| 04:48.52 | De_Mon | half my family works in telco and yet I know so little ;) |
| 04:49.00 | JT | and hook in a generator in front of the PSUs |
| 04:49.17 | JT | it makes for exceptional power reliability |
| 04:49.31 | bintut | anyone here peered with fwd? |
| 04:49.49 | De_Mon | wonder if nasa uses the same stuff in the space shuttle |
| 04:50.13 | JT | i doubt the space shuttle has lead acid cells |
| 04:50.17 | bintut | i gave up peering with iax2 so i used sip now. i'm already registered but i can't receive calls from the "call me" link from my.fwd link |
| 04:50.19 | De_Mon | is anyone in big brother 8 doing it? I don't think I can watach this much longer unless someone is gonna do it |
| 04:50.26 | JT | the technology in the space shuttle is shit |
| 04:50.30 | JT | designed in the 70s |
| 04:50.31 | JT | mostly |
| 04:50.51 | De_Mon | hah, I thought the telcos were doing the same |
| 04:51.15 | bintut | anyone peered with fwd here, please call me at 393000000 or 863676 to confirm if you can reach my number.. thanks.. :) |
| 04:51.18 | De_Mon | copper loops no fiber in most places around here |
| 04:51.18 | JT | De_Mon: what's this about big brother? |
| 04:51.25 | De_Mon | the tv show |
| 04:51.40 | JT | what did it have to do with the conversation, that is? |
| 04:51.41 | De_Mon | theres some 'after hours' thing on showtime |
| 04:51.51 | De_Mon | JT its on tv at the same time! |
| 04:52.00 | JT | mmkay |
| 04:52.13 | JT | fibre is mega awesome |
| 04:52.21 | JT | except you can't viably feed power over it |
| 04:52.31 | JT | and it's expensive to terminate at the moment |
| 04:53.30 | fujin | mm, optical processors. |
| 04:53.46 | JT | heh |
| 04:55.51 | *** join/#asterisk techie (n=techie@c-67-181-184-170.hsd1.ca.comcast.net) |
| 04:55.53 | *** part/#asterisk techie (n=techie@c-67-181-184-170.hsd1.ca.comcast.net) |
| 04:55.54 | JT | 1.6Tbit/s+ over a single fibre pair, nothing else comes close, really |
| 04:57.11 | flenders | you need disks that can write at that speed too. |
| 04:58.16 | JT | no, you really don't :) |
| 04:58.28 | fujin | that'd be awesome. |
| 04:58.37 | fujin | eh; anyone know how to disable this warning? [Aug 9 16:52:26] NOTICE[21696]: rtp.c:783 process_rfc3389: Comfort noise support incomplete in Asterisk (RFC 3389). Please turn off on client if possible. Client IP: 192.168.108.210 |
| 04:58.41 | JT | undersea fibre links don't connect to a single server with a hard drive on each end |
| 04:58.44 | fujin | I'm trying to get the cisco dude to disable it |
| 04:59.08 | De_Mon | fujin someone has silence supression on |
| 04:59.09 | *** join/#asterisk AllanLima (n=mediain@unaffiliated/allanlima) |
| 04:59.23 | fujin | phones don't, 5400 doesn't |
| 04:59.27 | fujin | next suggestion? |
| 04:59.45 | JT | fujin: what are you connecting to? |
| 05:00.00 | flenders | fujin: you're using heartbeat with asterisk, arent you? |
| 05:00.03 | fujin | flenders: yes |
| 05:00.09 | fujin | JT: .210 is an AS5400 |
| 05:00.23 | JT | then switch it off in the as5400? ;) |
| 05:00.47 | fujin | yeah |
| 05:00.48 | fujin | lol |
| 05:01.16 | JT | i don't think you can disable the warning in configuration |
| 05:09.45 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 05:10.19 | AllanLima | it is necessary to install asterisk as root? |
| 05:11.09 | AllanLima | or it is possible to install only for an user? |
| 05:11.26 | De_Mon | AllanLima you should be running asterisk as the asterisk user |
| 05:11.43 | AllanLima | hum |
| 05:11.45 | De_Mon | AllanLima running things as root is generaly a bad idea |
| 05:12.18 | AllanLima | but to install it is necessary to be root? |
| 05:12.43 | De_Mon | ... |
| 05:12.53 | De_Mon | do you need root to install to your home directory? no |
| 05:13.07 | De_Mon | do you need it to put files in /usr/bin etc, I HOPE SO |
| 05:13.26 | AllanLima | hum |
| 05:13.59 | De_Mon | you're looking for the install path option for make install (or something) right? |
| 05:14.42 | AllanLima | if i install in a server with some users, only my user goes to execute asterisk? |
| 05:16.18 | De_Mon | AllanLima huh? |
| 05:17.15 | AllanLima | I would like to install it only for 1 user, not for all the others |
| 05:18.23 | De_Mon | ok |
| 05:18.34 | De_Mon | don't let anyone else use it then |
| 05:19.02 | AllanLima | how? |
| 05:20.57 | De_Mon | AllanLima chown and chmod are the commands that change file owner and permissions in linux.. If you have to ask how, you probably shouldn't be messing with it though |
| 05:21.26 | AllanLima | hum |
| 05:21.50 | AllanLima | all right, let me test |
| 05:21.54 | AllanLima | thank you =) |
| 05:26.34 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 05:26.42 | phix | ok problem is nearly fixed |
| 05:27.34 | phix | JT: your assistants was partly helpful, it was a g729 licence issue, I changed some hardware and aparantly the licence usages the hardware to authenticate against or something like that |
| 05:27.48 | JT | correct |
| 05:27.55 | JT | it uses the NICs to authenticate |
| 05:27.57 | phix | kind of a stupid way to do it |
| 05:28.00 | phix | yeah I changed a NIC |
| 05:28.16 | phix | does it only go by MAC address? |
| 05:28.18 | JT | you failed to mention that |
| 05:28.19 | phix | I could clone it |
| 05:28.29 | phix | JT: I didn't think it was relevent |
| 05:28.33 | JT | i think so |
| 05:28.38 | JT | assumptions don't help ;) |
| 05:28.55 | JT | you have to think what changes where made of any sort before and after a problem |
| 05:28.59 | phix | ifconfig eth0 hw ether $OLD_MAC, woot! |
| 05:29.00 | phix | :P |
| 05:29.26 | phix | well I changed the hardware a month ago, although I was using ulaw at the time |
| 05:30.14 | phix | so I didnt notice it until I was trying to get g729 back up again (I recently patched asterisk to add rtp payload 96 tpye support) |
| 05:30.31 | phix | one problem after another! |
| 05:30.51 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 05:38.19 | *** join/#asterisk JoelSolanki (i=Joel@202.160.161.94) |
| 05:45.05 | *** join/#asterisk nuonguy (n=john@c-24-6-175-26.hsd1.ca.comcast.net) |
| 05:46.01 | *** join/#asterisk waptaxi (n=waptaxi@stat-5-160.e-sky.ru) |
| 05:51.48 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 05:57.00 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 05:59.08 | *** join/#asterisk Chris-NB (n=chris@ng1.kurtkrenn.com) |
| 06:13.36 | Snake-eyes | What would be a good way to send a 404 from a agi script ? |
| 06:13.42 | Snake-eyes | The agi script did a look up and found 0 results and thus sends a sip 404 back to the caller |
| 06:14.38 | Snake-eyes | looking for something better than setting variable and having some mirco look at the var and then hangup the call |
| 06:15.10 | Snake-eyes | *macro |
| 06:19.55 | Snake-eyes | nm |
| 06:22.32 | *** join/#asterisk syneus (n=syneus@syneus.aemcom.net) |
| 06:22.56 | *** join/#asterisk af_ (n=getsmart@81-174-47-190.dynamic.ngi.it) |
| 06:24.17 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 06:27.04 | Sweeper | Snake-eyes: use something with a real SIP stack |
| 06:27.07 | Sweeper | aka, not asterisk |
| 06:33.26 | *** join/#asterisk Aurs (n=Aurs@1ult2p8.ip.hipercom.no) |
| 06:34.39 | russellb | yikes |
| 06:35.06 | russellb | blanket statement with absolutely no useful suggestions |
| 06:35.34 | russellb | Snake-eyes: you can't do that, Asterisk has to already accept the call before running the AGI script |
| 06:35.56 | russellb | Snake-eyes: because the all really was handled and acepted |
| 06:36.16 | russellb | the extension *does* exist, so the number (the AGI) gets called ... |
| 06:36.29 | Snake-eyes | russellb, ok, is there any way i can hangup the call and send a 404 ? |
| 06:36.41 | russellb | you can run Hangup |
| 06:36.44 | russellb | but it won't send a 404 |
| 06:36.53 | Snake-eyes | yea, im seeing 503 |
| 06:37.15 | russellb | ah, Congestion |
| 06:38.59 | Snake-eyes | hmm, so once asterisk has picked up its to late ? |
| 06:39.14 | russellb | yeah |
| 06:39.34 | russellb | because the extension existed |
| 06:39.35 | De_Mon | Snake-eyes you want openser |
| 06:39.47 | russellb | probably |
| 06:40.46 | Snake-eyes | i was wanting to do this in asterisk not ser..... |
| 06:40.51 | russellb | you can do Indicate(BUSY) and make it send 486 Busy Here ... |
| 06:41.46 | *** join/#asterisk zeeesh (i=zeeesh@202.166.161.36) |
| 06:41.51 | Snake-eyes | hmm |
| 06:42.28 | russellb | or i guess the app is Busy() |
| 06:42.32 | russellb | whatever. |
| 06:42.44 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 06:43.21 | Snake-eyes | hehe |
| 06:44.32 | *** join/#asterisk vgster (n=vgster@cpc2-ledn1-0-0-cust826.leed.cable.ntl.com) |
| 06:46.37 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 06:47.31 | *** join/#asterisk jarod14 (n=jarod14@212.99.113.131) |
| 06:48.12 | russellb | Snake-eyes: i have a hak for you |
| 06:48.22 | russellb | hack, even |
| 06:49.07 | Snake-eyes | russellb, yes ? |
| 06:49.21 | Sweeper | netcat :D |
| 06:49.31 | russellb | (it's compiling) |
| 06:49.46 | Snake-eyes | :) |
| 06:49.49 | Sweeper | that's cheating! :P |
| 06:50.01 | Snake-eyes | lol |
| 06:50.15 | Sweeper | I wonder what would happen if you used a script with netcat to send a sip404 message |
| 06:50.30 | russellb | have fun writing it |
| 06:51.07 | russellb | well ... i guess most of the info you need is available |
| 06:51.14 | Sweeper | it's just text |
| 06:51.25 | russellb | it's getting the right text :) |
| 06:51.52 | Sweeper | does SIP sequence the message? |
| 06:51.57 | Sweeper | err |
| 06:51.58 | russellb | such as the correct Call ID, and what address and port number to use ... |
| 06:52.06 | russellb | i got what you meant :) |
| 06:52.07 | russellb | yeah |
| 06:52.16 | russellb | well, it has these "timers" |
| 06:52.48 | russellb | to handle retransmissions if it doesn't get a resonse in a certain time |
| 06:52.53 | russellb | SIP is just really weird ... |
| 06:53.32 | *** join/#asterisk Kumbang (n=kumbang@router-wmi.paume.ITB.ac.id) |
| 06:54.23 | denon | expect nothing less of a protocol that was designed to replace h.323 |
| 06:54.26 | *** join/#asterisk tsurko (n=tsurko@77.70.15.52) |
| 07:02.38 | *** join/#asterisk tragidy (i=Ejeet@insecure.us3r.info) |
| 07:03.00 | *** join/#asterisk kv0s (n=kv0s@p4FD21C4D.dip0.t-ipconnect.de) |
| 07:09.39 | *** part/#asterisk TheNewAndy (n=TheNewAn@144.131.134.181) |
| 07:13.21 | kv0s | What a nice day .... 9am ... and it's rainy since yesterday ... :-( |
| 07:13.35 | JT | 1713 |
| 07:14.48 | *** join/#asterisk dominic1 (n=dob@213.221.82.242) |
| 07:15.13 | flenders | cant wait to go home |
| 07:15.15 | flenders | :) |
| 07:18.28 | phix | I have been home all day :) |
| 07:21.57 | *** join/#asterisk glacid (i=unknown@evool.com) |
| 07:22.17 | glacid | does anyone here have access to the cisco firmwares for the 7970/7971? |
| 07:25.22 | thansen|laptop | anyone know of a good app (mplayer, ffmpeg) to convert from gsm -> mp3? |
| 07:30.58 | kaldemar | thansen|laptop: sox for example |
| 07:31.16 | creativx | sox to lame |
| 07:31.43 | creativx | thansen|laptop: but why are you going backwards? |
| 07:32.13 | thansen|laptop | well, I don't want to save the files off as waves to being with for space concerns |
| 07:32.31 | thansen|laptop | but I don't want the end user to have .gsm files for compatability |
| 07:32.49 | thansen|laptop | so, I save them to gsm, then convert to mp3 :( |
| 07:33.06 | *** join/#asterisk menil (n=meni@bzq-179-153-168.static.bezeqint.net) |
| 07:34.41 | JT | thansen|laptop: that does sound like unnecessary quality loss |
| 07:34.54 | JT | thansen|laptop: you should save them as .wav and then convert them to mp3 |
| 07:35.17 | *** join/#asterisk ptiggerdine (n=ptiggerd@203-206-171-23.perm.iinet.net.au) |
| 07:35.20 | thansen|laptop | I'm not sure I want massive wav files getting saved |
| 07:35.29 | thansen|laptop | although I see your point |
| 07:35.37 | JT | hard drives are cheap. |
| 07:35.45 | JT | and what's it matter if you're converting them? |
| 07:36.16 | thansen|laptop | well, potentially hundreds of people could be recording concurrently |
| 07:36.31 | thansen|laptop | I don't want the io overhead (among other things) |
| 07:38.31 | JT | use a ram disk then |
| 07:38.38 | JT | there's going to be overhead no matter what |
| 07:45.34 | *** join/#asterisk steliosk (n=Stelios@ipa226.211.tellas.gr) |
| 07:46.33 | thansen|laptop | anyone know why all of a sudden I would start getting the server rejecting calls? |
| 07:59.45 | *** join/#asterisk Polis_ttt (n=your@194-237-172-225-no48.business.telia.com) |
| 08:02.01 | *** join/#asterisk KermitTheFragger (n=siepkes@217.149.197.118) |
| 08:03.35 | *** join/#asterisk darkskiez (n=mhb@bb-87-81-62-203.ukonline.co.uk) |
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| 08:04.43 | *** part/#asterisk vutamhoan (n=hoavq@222.255.15.252) |
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| 08:06.54 | *** join/#asterisk vutamhoan (n=hoavq@222.255.15.252) |
| 08:07.41 | vutamhoan | My mp3 file is very good quality with soft-phone, but analog and digital line is so bad. Does anyone help? |
| 08:08.14 | vutamhoan | Sometime they're good, sometime bad :( |
| 08:11.22 | *** join/#asterisk Modcuts (n=modcuts@lan.proporta.com) |
| 08:11.37 | *** join/#asterisk qdk (n=qdk@213.150.62.32) |
| 08:14.39 | kv0s | Is there any way, that asterisk isn't handling my msn with my fax on it? I can't receive fax since installed asterisk. |
| 08:30.28 | tzafrir | kv0s, faxes through what device, exactly? |
| 08:31.00 | kv0s | tzafrir: Through my fax? |
| 08:31.41 | *** join/#asterisk Chris-NB (n=chris@89.26.28.10) |
| 08:32.44 | tzafrir | but how is is the fax connected to asterisk? and how is asterisk connected to your provider? |
| 08:32.53 | kv0s | tzafrir: Asterisk runs perfectly, but it doesn't answer calls for my fax line! My faxline (same isdn/ntba) have another ISDN MSN ... |
| 08:33.17 | kv0s | Asterisk <-> bristuffed hfc-s <-> isdn provider NTBA |
| 08:33.41 | kv0s | Faxserver <-> different server from asterisk <-> isdn provider but same NTBA as asterisk |
| 08:33.59 | tzafrir | hmmm... and is asterisk getting all the calls? |
| 08:34.05 | kv0s | I think i must configure my zaptel/zapata.confs so asterisk isn't answering the fax-msn or? |
| 08:34.35 | tzafrir | set it to only accept specific extensions, I guess |
| 08:34.55 | tzafrir | in extensions.conf |
| 08:34.59 | kv0s | tzafrir: asterisk answer all calls! but it shouldn't do it! only two msn's out of three should be answered with asterisk ... |
| 08:35.22 | tzafrir | not something like _X. . Only sour specific MSN |
| 08:35.37 | tzafrir | s/sour/your/ |
| 08:39.30 | *** join/#asterisk logostech (n=edoardo@83.103.64.14) |
| 08:40.14 | logostech | hi to all! |
| 08:44.49 | kv0s | Hm. |
| 08:45.21 | kv0s | Is the definition of msn a thing must made at extensions.conf or zaptata.conf zaptel.conf? |
| 08:52.12 | logostech | can someone help me? i have a problem with the presence in asterisk 1.4.5. |
| 08:53.19 | tzafrir | in extensions.conf |
| 08:53.34 | tzafrir | I can't think of anything in zapata.conf . Surely not in zaptel.conf |
| 08:53.49 | tzafrir | kv0s, --^ |
| 08:56.04 | *** join/#asterisk HaMYaI (n=LAMER@ppp-58.8.212.139.revip2.asianet.co.th) |
| 09:03.09 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 09:07.03 | *** join/#asterisk dharrigan (n=david@host12.williamhill.co.uk) |
| 09:08.36 | *** join/#asterisk neverblue (n=neverblu@unaffiliated/neverblue) |
| 09:19.56 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 09:23.27 | phix | MSN? and asterisk? |
| 09:24.27 | kv0s | phix???? |
| 09:24.35 | kv0s | What is your question? ,-) |
| 09:25.33 | phix | kv0s: MSN supports some type of protocol which can speak to asterisks? |
| 09:26.16 | kv0s | phix: Ahh ... not MSN from Microsoft. Multiple Subscriber Number! It's used with european isdn (bri) |
| 09:26.22 | phix | oh |
| 09:26.25 | phix | haha |
| 09:26.37 | phix | stupid acronyms. |
| 09:26.43 | kv0s | See above .. ,-) |
| 09:26.56 | phix | ok :) |
| 09:27.07 | phix | isn't that stuff expensive? |
| 09:27.38 | phix | how many calls / simutanous channels does isdn (bri) support? |
| 09:28.03 | kv0s | phix: What? ISDN? From where to u came from? At germany isdn supports two channels simultanusly |
| 09:29.12 | kv0s | I use ISDN for incoming calls. Outgoing i use sipgate.de. |
| 09:29.15 | phix | I am from AUS |
| 09:29.25 | kv0s | australiaß |
| 09:29.26 | kv0s | ? |
| 09:29.29 | phix | yes |
| 09:29.40 | kv0s | Tz. Small world with irc .. ,-) |
| 09:29.59 | phix | yep :) |
| 09:30.13 | tzafrir | ISDN BRI is two channels |
| 09:30.38 | tzafrir | what you descript is the ptmp mode of ISDN (used only in BRI) |
| 09:31.24 | kv0s | Any freepbx experts out there? I've defined one outbound route with uses sip trunk ... my sipgate.de - provider - accepts any cid. can i use at freepbx ${EXTEN} or any other variables to switch the incomming caller id to my external sipgate call - in example for forwarding calls. |
| 09:34.34 | *** join/#asterisk syneus (n=syneus@syneus.aemcom.net) |
| 09:34.52 | *** join/#asterisk nfi|ermes (n=ErMeS@217.220.121.62) |
| 09:38.11 | *** join/#asterisk zeeesh (i=zeeesh@202.38.55.121) |
| 09:40.28 | *** join/#asterisk patrick-- (i=alex@eos.openroot.de) |
| 09:40.30 | patrick-- | Hi all |
| 09:40.43 | patrick-- | Is there someone that could guide me through an asterisk setup with Fritz Card ISDN ? |
| 09:42.38 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 09:45.15 | *** join/#asterisk |dennis| (n=dennis@200.32.233.84) |
| 09:48.42 | *** join/#asterisk linuxmaniac (n=maniac@153.Red-88-22-238.staticIP.rima-tde.net) |
| 09:57.09 | patrick-- | noone? |
| 10:08.16 | *** join/#asterisk aikanaro79 (n=noone@89-180-72-198.net.novis.pt) |
| 10:08.56 | aikanaro79 | does anyone know if there are any problems registering a new user to access asterisk forums? I've tried to do that but I never got my email confirmation |
| 10:11.02 | *** join/#asterisk geoff_k (n=geoff_k_@host81-152-90-185.range81-152.btcentralplus.com) |
| 10:13.13 | aikanaro79 | also, is there a way to configure such a thing as a user class (using only SIP channels) so that I don't have to configure every possible user? later I need to be able to ask asterisk for a list of registered users |
| 10:15.52 | *** join/#asterisk Pon`work (n=jamesm@ip-217.146.113.66.merula.net) |
| 10:16.33 | *** join/#asterisk steliosk (n=Stelios@ipa226.211.tellas.gr) |
| 10:24.42 | *** join/#asterisk tr2x (n=alvar@80-218-185-55.dclient.hispeed.ch) |
| 10:31.07 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 10:40.37 | aikanaro79 | if 2 users register with the same username and password can they later be identified separately if a list of users is asked from the server? |
| 10:46.36 | krdian_ | grrrr, my asterisk is getting more and more memory, what happen ? also manager show eventq showing very long event list |
| 10:47.33 | Pon`work | sounds like a memory leak |
| 10:52.08 | krdian_ | Pon`work: is it system problem ? |
| 10:53.01 | Pon`work | noticed any other instability/memoryeating in any other applications running? |
| 10:53.50 | krdian_ | nope |
| 10:54.41 | *** join/#asterisk Pilko (n=pirch@213.80.169.119) |
| 10:54.44 | krdian_ | Pon`work: just asterisk eating memory |
| 10:55.44 | krdian_ | Pon`work: looks like manager doesn't flush eventq |
| 10:58.59 | *** join/#asterisk stkn|work (n=stkn|wor@gentoo/developer/pdpc.active.stkn) |
| 11:05.57 | Pon`work | krdian_: http://bugs.digium.com/view.php?id=9238 |
| 11:09.46 | *** join/#asterisk ccesario (n=ccesario@189-19-9-100.dsl.telesp.net.br) |
| 11:17.29 | *** join/#asterisk kkn088 (n=kikoun@84.4.74.213) |
| 11:17.44 | kv0s | with show channels i can see all active ... can i see bandwidth consumation too? |
| 11:18.43 | *** join/#asterisk |dennis| (n=dennis@200.32.233.84) |
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| 11:35.14 | zeeesh | will anybody check ... http://pastebin.ca/650996 |
| 11:41.35 | *** join/#asterisk grEvenX (n=even@ti500720a080-6875.bb.online.no) |
| 11:43.16 | grEvenX | when I run the Agi(agi://server_path) command in the dialplan to execute a FastAGI, I get "server_path:" prefixed when using verbose from that context |
| 11:43.44 | *** join/#asterisk |dennis| (n=dennis@200.32.236.20) |
| 11:43.59 | grEvenX | any way to prevent that? Since I use the server_path to pass arguments to the FastAGI script, the server_path gets really long, adding too much garbage to the CLI |
| 11:47.20 | *** join/#asterisk MrWup (i=Neil@i-83-67-202-134.freedom2surf.net) |
| 11:47.22 | MrWup | hi |
| 11:47.37 | MrWup | how do i do an extension for _00[followed by any 10 digits] ? |
| 11:47.47 | MrWup | (im trying to do international calling) |
| 11:48.39 | grEvenX | _00XXXXXXXXXX |
| 11:48.39 | grEvenX | ? |
| 11:49.07 | MrWup | sorry what i meant was |
| 11:49.12 | MrWup | any 10 digits, or fewer than 10 |
| 11:49.29 | MrWup | cause i dont know how long the number is, but it shouldnt be more than say 10 after the 00 for international calls |
| 11:50.36 | grEvenX | _00X. |
| 11:53.49 | MrWup | thanks! |
| 11:54.13 | kv0s | Grml. Call over call ... every time i've problems with echo on my lines. |
| 11:55.14 | kv0s | It makes no difference between outgoing trunk over isdn or sip (over the internet). Each time i've echo on my call. |
| 11:55.47 | kv0s | It is possible, that the echo is produced on my asteriskbox or thrugh my bluetooth headset? |
| 11:56.25 | kv0s | bluetooth headset <-> x-lite (windows) <-> sip <-> asterisk <-> isdn bristuffed line <-> called party use isdn too |
| 11:58.26 | creativx | how about you eliminate the bt headset |
| 11:58.29 | creativx | by using a wired one |
| 12:00.04 | *** join/#asterisk lirakis (n=etamme@65.200.191.253) |
| 12:00.40 | *** join/#asterisk implicit_ (n=implicit@210.16.55.38) |
| 12:01.50 | kv0s | creativx: I don't have one at the moment. Is a bt-headset not really good for use with sip-phones? |
| 12:02.22 | *** join/#asterisk saftsack (n=saftsack@pD9E07758.dip.t-dialin.net) |
| 12:02.30 | kv0s | Mhm. |
| 12:02.46 | MrWup | using a headset with x-lite is crap in my experience |
| 12:03.00 | MrWup | i had all sorts of problems with echo, crackles etc with both x-lite and the headset |
| 12:03.21 | MrWup | in terms of echo... you could also make sure that hardware echo cancellation is enabled on your ISDN card |
| 12:03.25 | MrWup | which ISDN card do u have? |
| 12:03.27 | MrWup | digium? |
| 12:03.52 | kv0s | MrWup: Sorry. No. It's a project for testing asterisk. So we used really cheap hfc-s cards. |
| 12:04.10 | kv0s | MrWup: About only 20 Euros eachs card. |
| 12:04.28 | MrWup | oh |
| 12:04.39 | MrWup | well that could well be the source of echo |
| 12:04.40 | kv0s | MrWup: But the echo also occours at the sip-line. So i don't think it has something to do with the isdn-cards. |
| 12:04.57 | MrWup | i had slight echo problems with my digium card and camer across millions of posts about cheaper ones |
| 12:05.12 | MrWup | ok well you could also check codecs.conf |
| 12:05.25 | kv0s | MrWup: So no headset wired or not works well with softphones like x-lite? |
| 12:05.28 | MrWup | might be able to help |
| 12:05.46 | MrWup | bluetooth is shit for headsets... bandwidth too low |
| 12:05.54 | MrWup | maybe wired is better |
| 12:06.03 | MrWup | in terms of soft phones... i have had huge problems with x-lite |
| 12:06.08 | MrWup | its basically just a piece of shit |
| 12:06.12 | *** join/#asterisk basty (n=basty@212.218.65.195) |
| 12:06.12 | basty | Hi |
| 12:06.48 | MrWup | also, you should check whether the cable between your ISDN point on the wall and the isdn card is too long |
| 12:06.50 | kv0s | MrWup: Actually i'll play with sjphone. |
| 12:07.09 | MrWup | i had about 70 metres of cable between the isdn point and the card and i had echo problems with a good digium card |
| 12:07.17 | MrWup | since i reduced that to 0.3 metres the echo has never come back |
| 12:07.29 | kv0s | MrWup: ISDN - perhaps you've right with your tipps, but the echo also occours on my siptrunk outgoing! |
| 12:07.42 | kv0s | MrWup: I'll try non bluetooth-headset. |
| 12:07.51 | kv0s | But which one? |
| 12:08.05 | kv0s | There are so many different on the market ... |
| 12:08.11 | basty | I have a newbie question - in my extensions.conf I have a dialplan like: exten => _X.,1,Dial.... and _00X.,1,Dial.... for national (_X.) I want to use my ISDN to dial out..for international (_00X.) I want to use my VoIP SIP-Account...when I dial 123456 the call get though the ISDN (thats okay) but if I dial 004912345 it doesnt go though my SIP-Account...anyone knows why ? |
| 12:08.18 | MrWup | if you want real quality go for a plantronics headset |
| 12:08.21 | MrWup | we use those at the office |
| 12:08.28 | MrWup | with a tubular microphone |
| 12:08.35 | MrWup | good heavy build quality but very light looking |
| 12:08.42 | MrWup | about £85 |
| 12:08.51 | MrWup | USD 170 |
| 12:09.00 | kv0s | MrWup: With the tube microfone? |
| 12:09.05 | MrWup | yeah |
| 12:09.07 | kv0s | microphone? |
| 12:09.09 | kv0s | Mhm. |
| 12:09.09 | MrWup | vista connecter |
| 12:09.25 | MrWup | u can get connectors for pc as well |
| 12:09.29 | kv0s | Not really cheap headsets .. ,-) |
| 12:09.33 | MrWup | nope |
| 12:09.39 | MrWup | but good enough for high quality office use |
| 12:09.52 | creativx | ive tried plantronics bt headset and x-lite, the problem is usually the bt headset.. x-lite gives me no problems with a wired headset |
| 12:10.02 | kv0s | Have say headsets to connect with pc withouts soundcards? |
| 12:10.38 | cpm | the plantronics usb headset has great audio quality, but seems to introduce latency |
| 12:10.55 | creativx | cpm: our users dont like it any much |
| 12:11.06 | creativx | introduces noises from time to time |
| 12:11.37 | *** join/#asterisk minkus (n=minkus@pool-72-84-46-134.clrkwv.east.verizon.net) |
| 12:12.09 | basty | if you want to use wireless headsets i would recommend a Netcom USB 9330 - bluetooth is crap.. ;-) |
| 12:12.20 | kv0s | creativx: Why? I've a notebook with soundcards on board. But i'll not use the soundcard for speeking - music, ringtones and more should be played out to the soundcard. speeking/calls to the headset ... |
| 12:12.57 | creativx | kv0s: why what? |
| 12:13.08 | creativx | basty: im about to purchase a gn netcom headset yes |
| 12:14.33 | basty | creativx: we used to test the Plantronics CS60 and Netcom 9330 - the design of the netcom is much better...so for right now we are using about 55 Netcom Dect Headsets with X-Tapi Pro...and it works very well. |
| 12:16.20 | kv0s | Mhm. Is DECT the solution? It is better for audio than bluetooth? |
| 12:16.29 | cpm | neither of them make any sense to me. Bad use of wireless technology imo. And I can't stand things that wrap around my ear. Can wear 'em for maybe 20 minutes |
| 12:17.02 | *** join/#asterisk Chris-NB (n=chris@ng1.kurtkrenn.com) |
| 12:17.30 | basty | kv0s: I would say yes...bluetooth is crap..I mean it works for mobile...but NOT on a workstation...I used to test the bluetooth with the plantronics voyager 510 USB...the connection/sound was nasty. |
| 12:17.33 | creativx | basty: well what i needed was something that could be used as a usb headset.. and is wireless, with good audio quality |
| 12:19.04 | *** join/#asterisk guillote_GNU (n=bancaria@host136.200-117-227.telecom.net.ar) |
| 12:19.28 | creativx | and the reason i didnt order the gn9330usb was due to the price |
| 12:19.40 | creativx | but 1 wk after ordering the plantronics they are priced almost the same.. |
| 12:20.04 | *** join/#asterisk Modcuts (n=modcuts@lan.proporta.com) |
| 12:20.05 | basty | yep |
| 12:20.13 | basty | over here its about 230 Euro |
| 12:20.19 | basty | for the netcom.... ;) |
| 12:22.21 | creativx | 257 here |
| 12:22.22 | creativx | hehe |
| 12:22.36 | minkus | anyone who has experience with the polycom 330 know if the speakerphone is loud enough to be heard in a 20 foot x 20 foot room. The context is a classroom where announcements will need to be made. The phone will be wall mount at front of classroom. |
| 12:26.21 | *** join/#asterisk vutamhoan (n=hoavq@58.187.95.140) |
| 12:26.31 | kv0s | Mhm. Does the gn9330usb headset works with x-lite or sjphone? Or other voip-sip-softphones? |
| 12:26.47 | creativx | that was something i was about to find out aswell |
| 12:26.51 | creativx | before I order a 9330 |
| 12:27.28 | creativx | according to counterpaths forums it should work |
| 12:27.44 | *** join/#asterisk |dennis| (n=dennis@200.32.233.84) |
| 12:28.59 | *** join/#asterisk Uatec (n=uatecuk@adsl.ntsols.com) |
| 12:29.01 | Uatec | greetings |
| 12:29.16 | creativx | np |
| 12:29.16 | *** join/#asterisk Paul_UK (i=Paul_UK@78.32.14.83) |
| 12:29.50 | Paul_UK | hey anyone have a snom phone here? when i have a vm.. i hit the button and it isnt *97 more the hostname of my server. is there anyway i can change it to *97.. with no vm.. the button is *97 lol.. very strange |
| 12:31.05 | creativx | JT: man i have alzheimers.. hit me up with your polycom recommendation again :) |
| 12:31.23 | *** join/#asterisk waKKu (n=eurulo@unaffiliated/wakku) |
| 12:31.45 | kv0s | Uf. |
| 12:31.50 | kv0s | 229 euros? |
| 12:32.27 | kv0s | Mhm. If i know after i've buy the headset i've no echo ... but if not? Puhh ... |
| 12:33.04 | *** join/#asterisk lirakis (n=etamme@65.200.191.253) |
| 12:33.14 | basty | kv0s: the Netcom USB 9330 works very well with X-Lite / X-Tapi Pro and the new "Ninja" |
| 12:33.25 | *** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca) |
| 12:33.52 | basty | Paul_UK: what kind of Snom ? 360 ? |
| 12:34.01 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 12:35.13 | basty | Paul_UK: And what do you mean by *97? I mean in the Snom Webinterface you can setup your Voicemail-Main Extension... |
| 12:35.58 | lirakis | .. morning |
| 12:36.12 | kv0s | Mhmm... pay 230 euros or not ... :-/ |
| 12:36.44 | Paul_UK | basty, snom 320.. On the phone, you can tell it that your mailbox is accessible via *97 (im using freepbx). |
| 12:37.11 | *** join/#asterisk twitchnln (n=Administ@70.43.112.117.nw.nuvox.net) |
| 12:37.17 | Paul_UK | basty, for some reason, with no vm.. the button dials *97, with a vm, the button dials "asterisk" lol.. |
| 12:37.19 | twitchnln | morning everyone |
| 12:37.37 | basty | kv0s: Well..if you want to have a good quality..I would spend 230 euros for such a Headset....if you want crap...go ahead and buy a bluetooth one.. ;-) |
| 12:37.45 | Uatec | I save my voicemail files in wav49 format. How can i make mixmonitor save in that same format? |
| 12:37.55 | Uatec | currently it only saves in wav |
| 12:37.58 | Uatec | and wav is biiiiiig |
| 12:38.44 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 12:38.49 | basty | Paul_UK: strange...if you access the webinterface of your snom 320...what does the "Identity" "Mailbox" say ? |
| 12:38.58 | *** join/#asterisk s0ck (n=m@unaffiliated/s0ck) |
| 12:39.00 | twitchnln | got a question, i just got * setup, verified that the config is working, then installed a2billing, now when i click reports in freepbx i see cdr data, but when i click cdr report under a2billing, i get no data found, anyone got any ideas about this? |
| 12:39.25 | kv0s | basty: Mhmm... |
| 12:40.31 | Paul_UK | basty: Mailbox is *97 |
| 12:41.27 | basty | kv0s: I used to setup an Asterisk PBX for a customer. The customer wanted to have X-Tapi Pro and a wireless Headset...we offered the Plantronics Voyager 510USB and the Netcom...The Voyager Bluetooth was around 80 Euros...the Netcom around 230...the customer wanted to save money...and bought the voyager headsets...We installed it...1 Week later..the company was so unhappy..and used to buy the netcom...after that time..i have never heard anything anymore... ; |
| 12:42.11 | basty | Paul_UK: Okay..Mailbox Extension is available with *97 ?...and if you press the "Retrieve" Button on the Snom...it doesnt dial it ? |
| 12:43.15 | basty | Paul_UK: Oh and the range of the Netcom (DECT) is about 100meters....bluetooth is/was 10meters |
| 12:43.19 | kv0s | Is dect for wireless headsets okay? or it has the same problems as bluetooth? |
| 12:43.37 | *** part/#asterisk dominic1 (n=dob@213.221.82.242) |
| 12:44.10 | basty | kv0s: It IS okay...and in my opinion there are NO Problems with dect. No Echo, No Background Noices...nothing negativ.... |
| 12:44.13 | lirakis | twitchnln: go to #freepbx |
| 12:44.14 | kv0s | Mhm. So i think i must order a dect-headset ... |
| 12:44.38 | kv0s | basty: plantronics s60 - for example? |
| 12:44.40 | basty | kv0s: otherwise contact the netcom reseller and order a test-headset for like 1 week....many resellers offer a test-headset for testing... |
| 12:45.19 | zeenix | hmm.. how do i enable the cli to display each extension as asterisk execute it during a call? |
| 12:45.23 | basty | kv0s: I dont know the S60...only the CS60....the CS60 and Netcom 9330 is actually the same beside the name and the design. |
| 12:45.47 | basty | zeenix: asterisk -r ? |
| 12:46.19 | lirakis | zeenix: .. ? im not sure wahat you are asking |
| 12:46.28 | kv0s | sorry - i mean cs60 |
| 12:46.48 | lirakis | zeenix: ... to see call flow on the cli .. just run asterisk -vvvvr (more v's is more verbose) |
| 12:46.50 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 12:47.29 | basty | kv0s: from the hardware/features and quality they are the same....in my opinion the netcom is much better from the design...oh..and you can see the battery status on the station of the netcom....on plantronics you dont see the status. |
| 12:47.47 | basty | zeenix: or asterisk -r and "set debug 10 / set verbose 10" :-) |
| 12:47.56 | zeenix | basty: yes, once inside the cli using `asterisk -r` |
| 12:48.37 | basty | zeenix: okay - so try "set debug 10" and "set verbose 10" - and you should see alot of informations :) |
| 12:48.52 | zeenix | i set the debug to 10 but it doesn't work |
| 12:49.05 | zeenix | ah, verbose too.. |
| 12:49.08 | basty | yup |
| 12:50.53 | zeenix | now it works, now how to enable colors? |
| 12:50.59 | zeenix | in the output i mean |
| 12:52.08 | *** join/#asterisk SuurMyy (n=SuurMyy_@195.238.211.98) |
| 12:53.00 | ber123 | how can you strip a trailing '*' for all calls in a context |
| 12:53.40 | basty | zeenix: colors? iiihrkk... well...If you restart the Asterisk with the "safe_asterisk" script - you will have colors.. :-) |
| 12:54.11 | basty | ber123: incoming calls or outgoing ? |
| 12:54.16 | ber123 | outgoing |
| 12:54.43 | *** part/#asterisk twitchnln (n=Administ@70.43.112.117.nw.nuvox.net) |
| 12:54.44 | Paul_UK | basty: correct, when there is no vm.. the phone dials *97.. when there is a vm, pressing the same button, the phone dials "asterisk" and not *97 |
| 12:55.01 | basty | ber123: exten => _X.,1,Dial(ZAP/g1/*${EXTEN}) ? |
| 12:55.28 | *** join/#asterisk jeremy_g (n=jerms@static-213-115-44-90.sme.bredbandsbolaget.se) |
| 12:55.29 | jeremy_g | hi |
| 12:55.36 | basty | Paul_UK: Thats weirdo..actually "asterisk" in the display means "unknown"... |
| 12:55.56 | ber123 | does that strip the '*' from the end of a dialed #? |
| 12:56.11 | kv0s | I think i must order cs60-usb ... |
| 12:56.41 | Paul_UK | basty, hmm thanks for the clarification. |
| 12:57.09 | basty | Paul_UK: I am sorry...over here it works..and we use 300,320,360 and even 370 without any problems...what firmware you are using ? |
| 12:57.41 | basty | ber123: Hrm..I dont understand your question....if you dial 12345 and send it to this "new" exten - it would dial via zaptel *12345 |
| 12:57.48 | jeremy_g | i am trying to register my asterisk box with my isp's sip proxy. They have provided me with username, password and an authorization user, domain, display name. and port 8891. Now how do i put all this info in our typical register statement in sip.conf? |
| 12:58.12 | Paul_UK | basty: firmware - snom320-SIP 6.5.10 |
| 12:58.54 | ber123 | batsy |
| 12:58.55 | basty | jeremy_g: register => username:password@hostname of sip provider/number |
| 12:59.10 | basty | Paul_UK: hrm..thats actually the latest... |
| 12:59.18 | ber123 | what is happening is i am getting SIP users dialing 1234567* and its causing routing issues for me because i am matching on 10 digits |
| 12:59.23 | Paul_UK | basty, could it be freepbx? |
| 12:59.31 | ber123 | the * at the end adds an 11th so i want to strip it off first |
| 12:59.37 | jeremy_g | whats the authorization user for |
| 12:59.42 | ber123 | and then apply the pattern matching in the context |
| 12:59.45 | jeremy_g | ok i got it, may be its for the voice mail |
| 12:59.59 | creativx | kv0s: so what are you deciding on? gn or plantronics |
| 13:00.01 | basty | Paul_UK: maybe..on Asterisk it works well ;) |
| 13:01.21 | basty | ber123: so you want to cut the 11th digit ? or just the "*" even when a customer dials 123* you want to cut the "*" ? |
| 13:01.43 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 13:01.43 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 13:01.49 | basty | jeremy_g: thats your login/username to your sip-account |
| 13:02.03 | Corydon76-home | exten => _XXXXXXXXXX*,1,Goto(${EXTEN:0:10},1) |
| 13:02.20 | ber123 | corydon |
| 13:02.24 | creativx | ber123: use cut() ? |
| 13:02.34 | Corydon76-home | CUT is not appropriate here |
| 13:02.41 | ber123 | i dont know how long the dial string could be, does that offset just peel the last digit from the end? |
| 13:02.51 | ber123 | i want somethinglike the perl chop() function |
| 13:03.01 | ber123 | if you are familiar with that |
| 13:03.14 | ber123 | just chop the last digit off if it matches * |
| 13:03.16 | Corydon76-home | Yes, but you need to match the length first |
| 13:03.41 | ber123 | would _X.* do any length? |
| 13:03.47 | Corydon76-home | No |
| 13:03.56 | basty | ber123: exten => _X.,1,set(MYDEST=${CUT(EXTEN,*,1)}) |
| 13:04.06 | Corydon76-home | "." is a short-circuit. Nothing matches after the "." |
| 13:04.11 | ber123 | ah i see |
| 13:04.14 | basty | that will "cut" the "*" |
| 13:04.33 | ber123 | basty, if no * is found is it a NOOP? |
| 13:04.49 | basty | well..if there is no "*" it will do nothing :) |
| 13:04.52 | ber123 | ok cool |
| 13:04.55 | Corydon76-home | However, you could do it a slightly different way... |
| 13:06.11 | basty | ber123: oh, and remember to dial ${MYDEST} otherweise you will have back the "*" ;-) |
| 13:06.35 | ber123 | ah yeah |
| 13:07.07 | Corydon76-home | exten => _X.,1,GotoIf($["${EXTEN:-1:1}" = "*"]?${EXTEN:0:$[${LEN(${EXTEN})} - 1]},1) |
| 13:07.26 | ber123 | thanks corydon |
| 13:07.28 | jeremy_g | can i use port=8891 while creating an outgoing account in sip.conf. is port tag valid |
| 13:07.41 | ber123 | currently I have a bunch of defined pattern matches in there |
| 13:07.42 | lmadsen | thats the listening port I think |
| 13:07.47 | *** join/#asterisk floppp (n=flopp@nat-staff.b3g-telecom.com) |
| 13:07.48 | ber123 | so instead of adding this into each of the pattern matches |
| 13:07.54 | ber123 | is there a way to flow all dials in the context through it |
| 13:07.56 | ber123 | in one place |
| 13:08.04 | ber123 | or do i have to put it in another context with a Goto |
| 13:08.13 | *** join/#asterisk JackEStorm (n=no@ip68-225-77-136.no.no.cox.net) |
| 13:08.42 | Corydon76-home | I gave you the logic... you need to apply it where ever it's needed |
| 13:09.08 | ber123 | yeah i understand that but there should be a more elegant way than duping the same code in 50 pattern matches |
| 13:09.20 | lmadsen | Corydon76-home: TFoT2 goes to print next week |
| 13:09.24 | lmadsen | I just got QC2 |
| 13:09.28 | Corydon76-home | lmadsen: woot |
| 13:09.33 | lmadsen | I'm up to chapter 5 in edits |
| 13:09.43 | Corydon76-home | lmadsen: forward me a copy? |
| 13:09.47 | lmadsen | of course! |
| 13:09.58 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 13:10.13 | Corydon76-home | I'm going to have a ton of time this weekend doing mostly nothing |
| 13:10.14 | lmadsen | I'm hoping to have enough time to update all the appendix examples to no longer using priority numbers (using 'n' and labels instead) |
| 13:10.28 | lmadsen | Corydon76-home: fantastic! I'll forward it over now |
| 13:10.36 | Corydon76-home | (where I'm restricted in movement) |
| 13:10.37 | lmadsen | 576 pages btw |
| 13:10.41 | Corydon76-home | Sweet |
| 13:10.46 | lmadsen | 4 pages short of 200 additional pages |
| 13:10.47 | *** join/#asterisk javar (n=javar@69.79.134.24) |
| 13:15.12 | kv0s | Mhm. Okay CS60-USB ordered. Can't waiting for a echo-free speeking on my lines .. ,-) |
| 13:16.01 | *** join/#asterisk ManxPower (n=manxpowe@032-385-595.area5.spcsdns.net) |
| 13:16.17 | basty | kv0s: why did you buy the plantonics and not the netcom ? ;) |
| 13:16.39 | *** join/#asterisk _bobweever_ (n=_bobweev@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 13:17.21 | jeremy_g | * registers with my isp but when i make a call, the isp returns 603 |
| 13:17.34 | jeremy_g | if i register my widnows based sip phone directly with isp proxy, it works |
| 13:17.49 | jeremy_g | i.e. it answers my invites properly ... |
| 13:17.56 | *** join/#asterisk sakic (n=sakic@adsl-227-157-225.clt.bellsouth.net) |
| 13:18.03 | basty | jeremy_g: what doesn "sip show registry" on the asterisk cli say ? |
| 13:18.14 | jeremy_g | basty:it says you are registered with your isp |
| 13:18.25 | jeremy_g | isp.url Registered |
| 13:18.42 | basty | jeremy_g: okay, extensions.conf? rules for the outgoing call? |
| 13:19.09 | jeremy_g | _X.,1,Dial(SIP/myisp) |
| 13:19.33 | basty | jeremy_g: try _X.,1,Dial(SIP/${EXTEN}@myisp) |
| 13:19.36 | jeremy_g | where myisp is properly defined in sip.conf and i know what i am doing as my other account with another isp works fine that way |
| 13:19.45 | *** join/#asterisk doolph (n=doolph@200.115.147.74) |
| 13:19.51 | doolph | hi |
| 13:19.55 | zeenix | basty: what was all that about? :) |
| 13:20.06 | basty | zeenix: ? :) |
| 13:20.12 | jeremy_g | its only this isp with which my sip software if running on linux, sends an invite, the isp will return with 603. |
| 13:20.14 | doolph | there's any script that change all permisions to run asterisk as non root? |
| 13:20.19 | zeenix | basty [n=basty@212.218.65.195] requested CTCP PING from zeenix |
| 13:20.24 | jeremy_g | doolph:sudo |
| 13:20.40 | basty | zeenix: haha - i was checking if you are lagging...because I wrote something..and you answered 3 mins later ;) |
| 13:20.44 | doolph | ? |
| 13:21.27 | basty | jeremy_g: yeah - because you have forgot the use ${EXTEN}....without exten you dont send the dialed number to your sip-provider...doh |
| 13:21.32 | jeremy_g | basty:sorry i was already trying what you suggested. :) |
| 13:21.49 | jeremy_g | offcourse!! |
| 13:21.51 | basty | jeremy_g: and still..it doesnt work ? ;) |
| 13:22.02 | zeenix | basty: nah! i am in a meeting at the same time :) |
| 13:22.02 | jeremy_g | nopes i forgot to tell you that i didnt forget :) |
| 13:22.17 | basty | jeremy_g: what does the asterisk cli say, if you dial a number ? |
| 13:22.27 | basty | zeenix: hehe..okay ;) |
| 13:22.45 | jeremy_g | basty:with this isp it doesnt work, its fine with others. the isp seem to return 603 if my sip phone or proxy is running on linux. |
| 13:22.48 | jeremy_g | :( |
| 13:23.01 | jeremy_g | it says um dialing |
| 13:23.06 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 13:23.12 | jeremy_g | since i do i dial with an r, so i hear on ring |
| 13:23.17 | jeremy_g | and then it says |
| 13:23.44 | jeremy_g | Everyone is busy/congested at this time (1:0/0/1) |
| 13:23.44 | jeremy_g | <PROTECTED> |
| 13:24.04 | lmadsen | jeremy_g: well of course you're going to hear ringing with 'r'... that's the point -- it rings even when a call is not being setup |
| 13:24.14 | lmadsen | 'r' is useless -- don't use it |
| 13:24.36 | lmadsen | and with that... I'm going back to editing |
| 13:26.16 | basty | jeremy_g: you might need more dialing rules to send a call..I mean maybe your sip-provider needs a e164 format ? like 496912345 ? |
| 13:27.37 | kv0s | basty: because same technics, already made perfect experience with plantronics, my distributor has the plantronics for 180 euros! ;-) |
| 13:28.21 | ManxPower | Paste the CLI output of just the actual dial line |
| 13:30.27 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
| 13:31.24 | ManxPower | Unless you don't want my help, of course. |
| 13:33.07 | basty | kv0s: but the design of the plantonics sucks...and you arent able to view the battery status.. |
| 13:33.27 | basty | kv0s: but anyway..have fun with your new dect headset...hehe |
| 13:34.29 | *** join/#asterisk aikanaro79 (n=noone@89-180-72-198.net.novis.pt) |
| 13:35.08 | *** join/#asterisk flujan (n=flujan@200-160-115-020.static.spo.ctbc.com.br) |
| 13:35.21 | aikanaro79 | if I don't know in advance the "number" to call a specific user (neither do I know his/her name) is there a way to configure a dialplan that makes it possible? |
| 13:35.51 | *** join/#asterisk ManxPower (n=manxpowe@015-819-767.area5.spcsdns.net) |
| 13:36.15 | cpm | aikanaro79, let me get this straight, you don't know who are calling, either by name or number, but you want to connect anyway? |
| 13:36.30 | ManxPower | sorry about that |
| 13:36.57 | aikanaro79 | cpm: in advance I don't...I'm supposed to get a listing of registered users and then simply click on the desired user |
| 13:36.57 | doolph | what is spandsp ? |
| 13:37.19 | ManxPower | doolph: It is an audio processing library |
| 13:37.24 | *** join/#asterisk clandmeter (n=Carlo@81.175.82.2) |
| 13:37.31 | doolph | do still asterisk 1.4 needs them? |
| 13:37.56 | doolph | i think its something to send and receive faxes |
| 13:38.03 | ManxPower | doolph: no version of asterisk needs them |
| 13:38.07 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
| 13:38.26 | doolph | really |
| 13:38.28 | ManxPower | well if you want to send/receive faxes using RxFax/TxFax, then you need spandsp |
| 13:38.56 | aikanaro79 | cpm: is it possible? at least to get a listing of registered users from asterisk? |
| 13:39.05 | doolph | but does asterisk needs them to send faxes? |
| 13:39.27 | Uatec | "does asterisk needs" ??? |
| 13:39.32 | Uatec | -s |
| 13:39.44 | ManxPower | doolph: Asterisk does not come with a software fax. |
| 13:39.55 | cpm | aikanaro79, I expect it is possible, but I've never considered it. I don't *want* to hear from someone who doesn't know who they are calling. That's usually cold-calling sales-spam. |
| 13:40.00 | ManxPower | There are many ways to send faxes thru asterisk. SpanDSP is only one of them |
| 13:40.55 | aikanaro79 | cpm, I see your point..my goal is to develop a private server (it'll only be available inside a private LAN) for communication inside a company..hence my questions |
| 13:43.58 | *** join/#asterisk zydrunas (n=Miranda@216.49.228-ip-184.ckt.net) |
| 13:44.35 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
| 13:45.37 | ManxPower | SpanDSP will NEVER be part of Asterisk. |
| 13:47.10 | Pon`work | what does Asterisk use instead? |
| 13:48.59 | ManxPower | Pon`work: Asterisk does not include support for software fax |
| 13:49.14 | ManxPower | jeremy_g: pervert |
| 13:49.40 | aikanaro79 | cpm, if I want to enable SIP INVITE requests I have to contemplate this in my dialplan, am I right? |
| 13:49.43 | jeremy_g | ManxPower:pervert? |
| 13:50.26 | ManxPower | jeremy_g: T.38 seems to have so many compatibility issues that only a masochist would love it. |
| 13:50.43 | jeremy_g | hehe |
| 13:50.51 | jeremy_g | of bounties and open pbxes |
| 13:51.27 | ManxPower | aikanaro79: Asterisk users don't usually think about such things. They set up SIP, they set up their dialplan, it works. |
| 13:51.46 | ManxPower | But I don't quite understand what you mean by "enable SIP INVITE" |
| 13:51.55 | ber123 | span dsp is that stable now? |
| 13:52.12 | ber123 | i ran into so many issues trying to use span dsp and asterfax i gave up and went 2 hylafax with modems |
| 13:52.24 | ManxPower | ber123: we use it for 20+ incoming faxes per day |
| 13:52.30 | ber123 | 99.9% reliable? |
| 13:52.33 | ManxPower | ber123: it has its issues. |
| 13:52.39 | ManxPower | ..;er.. HAD its issues |
| 13:52.45 | cpm | hylafax is the correct approach. Reliable as dirt, more so than a stand alone fax machine |
| 13:52.47 | ber123 | my hyla is 99.9% reliable |
| 13:52.48 | aikanaro79 | ManxPower, I'm trying to come up with a diaplan that makes it possible to have conference calls..I'm still a newbie when it comes to asterisk (and that's a very big part of my problem)...this is the reason I'm asking such questions |
| 13:52.48 | doolph | erm |
| 13:53.02 | ber123 | the limiting factor is my idiocy and the other software running on the box |
| 13:53.05 | ber123 | not the fax software |
| 13:53.07 | ber123 | :) |
| 13:53.09 | aikanaro79 | ManxPower, and using only SIP channels |
| 13:53.32 | doolph | erm i cannot find this file: app_rxfax.c |
| 13:53.42 | ManxPower | cpm: the MAIN fax machine is on an analog POTS line. But many users want their own fax numbers, so we overload their DID with Fax support |
| 13:53.45 | ber123 | hylafax -- spamming faxes to your business since 1993 |
| 13:53.51 | doolph | it says copy it from there but its not there |
| 13:53.56 | ber123 | or whenever it was created |
| 13:54.00 | cpm | ManxPower, I only use pots for fax. It's the only way to be sure |
| 13:54.01 | ber123 | prob before |
| 13:54.11 | *** join/#asterisk mindCrime (n=chatzill@66.83.208.219.nw.nuvox.net) |
| 13:54.24 | cpm | well, that's not strictly true, I use fxs ports |
| 13:54.28 | [TK]D-Fender | aikanaro79: You set up auth accounts, and if you want, a context for "anonymous" connections. * does not require setup for specific SIP response types |
| 13:54.30 | cpm | but that's only on pris |
| 13:54.52 | ManxPower | aikanaro79: we don't use FXS ports. |
| 13:55.01 | ManxPower | IF we need an FXS port, we get a POTS line |
| 13:55.11 | ManxPower | FXO ports are cheap from our CLEC |
| 13:55.20 | _bobweever_ | Pardon if ths has been mentioned, but has anyone used the attractel fax solution? |
| 13:55.24 | ManxPower | ..er.. POTS lines are cheap from our CLEC |
| 13:55.37 | *** join/#asterisk wchalco (n=wchalco@190.81.57.246) |
| 13:56.21 | aikanaro79 | [TK]D-Fender, how can I set up auth accounts considering what I have said?...I don't know in advance who might use this server |
| 13:57.20 | aikanaro79 | ManxPower, sorry but I didn't quite follow what you said about FXS ports |
| 13:57.23 | doolph | anyone can tell me where is app_rxfax.c |
| 13:59.02 | ber123 | i think i found a way to do that cut thing once batsy was talking about |
| 13:59.25 | ber123 | send the call, then timeout, then capture through t extension and cut and redial |
| 13:59.44 | ber123 | find / -name app_rxfax.c |
| 14:00.18 | doolph | i dont have it :/ |
| 14:00.55 | ManxPower | the same place you got spandsp from |
| 14:01.57 | doolph | why they hide it so hard |
| 14:02.37 | doolph | it doesn't come in the .tgz |
| 14:02.44 | doolph | and its not in the download page |
| 14:03.28 | *** join/#asterisk waKKu (n=eurulo@unaffiliated/wakku) |
| 14:04.09 | ber123 | its a secret, grumble grumble |
| 14:04.17 | ber123 | the secret is in the tip of the nose |
| 14:04.37 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 14:04.48 | *** join/#asterisk luke-jr (n=luke-jr@ip72-206-114-236.om.om.cox.net) |
| 14:05.02 | *** join/#asterisk captiancrash (n=jmoore@70.159.118.70) |
| 14:05.21 | luke-jr | I'm trying to register one Asterisk box with another |
| 14:05.30 | luke-jr | but the other one is sending INVAL responses |
| 14:05.30 | captiancrash | when using asterisk-gui, does it modify the config files directly, or create its own configurations? |
| 14:05.34 | luke-jr | any ideas? |
| 14:05.36 | wchalco | hi |
| 14:05.42 | waKKu | luke-jr r u using iax ? |
| 14:05.57 | *** join/#asterisk hohum (n=dcorbe@gate.globecommsystems.com) |
| 14:06.03 | luke-jr | yeah |
| 14:06.23 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 14:06.23 | *** mode/#asterisk [+o anthm] by ChanServ |
| 14:06.44 | waKKu | did u create peer on asterisk ? configure register => user:passwd@master-asterisk on slave ? |
| 14:07.05 | luke-jr | yep |
| 14:07.19 | waKKu | ok... what shows on CLI when trying to register ? |
| 14:07.27 | luke-jr | nothing unless I debug it |
| 14:07.54 | waKKu | on master, iax2 show peers - shows your slave peer ? |
| 14:08.05 | luke-jr | yes |
| 14:08.58 | luke-jr | http://rafb.net/p/riLEUf79.html |
| 14:09.00 | waKKu | luke-jr well.. if when u r trying to register, master CLI doesnt show anything... probably your request isnt coming to master.. firewall or iax port not configured properly |
| 14:09.15 | luke-jr | I can see it all in debug |
| 14:09.22 | [TK]D-Fender | aikanaro79: If you can't know about these connections they can't be auth'd. This is a checken & egg scenario. You can set up to receive guest connections to a specific conection listed under [general]. |
| 14:09.51 | waKKu | luke-jr paste your iax.conf from other end.. |
| 14:09.57 | luke-jr | that's both ends |
| 14:10.18 | waKKu | well ... so is missing your register => line |
| 14:10.43 | aikanaro79 | [TK]D-Fender: if I got you right I can't define a specific friend is that it? but this way can I make calls? |
| 14:10.45 | luke-jr | register => 117-iax:secret@otherend |
| 14:11.07 | waKKu | luke-jr yeah.. |
| 14:11.25 | aikanaro79 | [TK]D-Fender: also, is there any such thing as a user class or is it possible to configure a "look-alike" when it comes to auth? |
| 14:11.42 | [TK]D-Fender | aikanaro79: You are COMPLETELY mixing yourself up! Yes, you can set up AUTHED accounts. You can also SEPARATELY allow misc connections against another context if you WANT TO. |
| 14:12.01 | [TK]D-Fender | aikanaro79: As for this "look-alike" I have no idea what you're talking about. |
| 14:14.43 | aikanaro79 | [TK]D-Fender: I'm sorry...I'll try to be clearer..is it possible to define a SIP channel such that several users use it to register with asterisk but at the same time any one of them is able to call anyone of the others? |
| 14:15.24 | [TK]D-Fender | aikanaro79: No, each must be a unique account on your system |
| 14:15.45 | waKKu | luke-jr if u still need help: http://pastebin.ca/651119 |
| 14:16.36 | ManxPower | aikanaro79: REGISTRATION requires user/password. You cannot have more than 1 device register against the same userid/password. You do NOT have to have devices register, nor do you require a user/pass in sip.conf to accept calls. Asterisk will BY DEFAULT accept calls that have no auth info and no registration info |
| 14:16.37 | aikanaro79 | [TK]D-Fender: but if I have no way of knowing in advance who might register how can I do it? I got as far as configuring a dialplan entirely based on numbers and using each user's IP address as it's own extension (but I'm not sure this'll work) |
| 14:17.04 | ManxPower | aikanaro79: you do not understand what registration is. |
| 14:17.19 | aikanaro79 | ManxPower, thanks for that info...I see now that my problem was exactly that |
| 14:17.46 | [TK]D-Fender | aikanaro79: this can't work. You cannot "register" to * without an account. And thats how it knows where to call you back |
| 14:18.39 | [TK]D-Fender | aikanaro79: You can't just have 1 guy say "Hey, I'm 12345 at IP 1.2.3.4!" |
| 14:18.59 | aikanaro79 | [TK]D-Fender: sorry for asking but why not? |
| 14:19.07 | aikanaro79 | I can get a peer's IP address |
| 14:19.17 | ManxPower | aikanaro79: because that is how Asterisk works. |
| 14:19.23 | *** join/#asterisk merkurie (n=merkurie@192.153.163.44) |
| 14:19.58 | ManxPower | aikanaro79: The ONLY thing registration does is inform the server what IP is associated with which userid/password. |
| 14:20.04 | ManxPower | It does nothing else. |
| 14:20.18 | luke-jr | I wish Asterisk would be more verbose as to *why* it rejects stuff |
| 14:20.19 | aikanaro79 | ok...I've understood that |
| 14:20.29 | merkurie | anyone got any recommendations for a good asterisk book? |
| 14:20.33 | ManxPower | You can Dial(SIP/12.34.56.87) |
| 14:20.45 | aikanaro79 | ManxPower, that's it...I can do it right? |
| 14:21.02 | ManxPower | aikanaro79: of course. |
| 14:21.17 | ManxPower | whatever is in [general] sip.conf is the settings that will be used for those calls. |
| 14:21.18 | aikanaro79 | ManxPower, do you know of a way of getting a list of IP addresses from asterisk? or is it only possible with registered users? |
| 14:21.36 | ManxPower | aikanaro79: a list of IP addresses of WHAT? |
| 14:22.05 | *** join/#asterisk xpot (n=jim@c-71-195-241-115.hsd1.ma.comcast.net) |
| 14:22.17 | luke-jr | Dial(SIP/192.168.1.${EXTEN}); |
| 14:22.17 | aikanaro79 | ManxPower, asterisk will only know of a user if he/she registers is that it? (I thing I see my problem) |
| 14:22.18 | luke-jr | :p |
| 14:23.02 | ManxPower | aikanaro79: it will only know if the device registers or if you configure it with host= in sip.conf section for each device. |
| 14:23.28 | aikanaro79 | ManxPower, and if I "use" registered users I have to configure each and everyone one of them right? |
| 14:23.41 | ManxPower | aikanaro79: correct. |
| 14:23.51 | aikanaro79 | ManxPower, thanks for your patience |
| 14:24.00 | ManxPower | aikanaro79: Asterisk is a PBX, not some free love hippie telecom lovefest where anything goes. |
| 14:24.01 | aikanaro79 | [TK]D-Fender, thanks also for your help |
| 14:24.37 | aikanaro79 | ManxPower, I get it...but I was asked to use it by someone that I'm beginning to suspect that has never truly looked at it...that's the problem |
| 14:25.53 | luke-jr | aikanaro79: could setup * to get accts from SQL, IIRC |
| 14:26.43 | aikanaro79 | luke-jr, but as such, any user that came later would mean a change to the dialplan...and that's not exactly an option :( |
| 14:26.51 | doolph | argh and the softswitch website is down now |
| 14:27.02 | luke-jr | accts can auto modify dialplan |
| 14:27.29 | luke-jr | and you could always do a mapping like extension=username |
| 14:27.39 | luke-jr | so |
| 14:27.53 | aikanaro79 | luke-jr, that I can't use... |
| 14:27.55 | luke-jr | _1XX => Dial(SIP/${EXTEN}); |
| 14:28.22 | aikanaro79 | luke-jr, but that I can use |
| 14:28.28 | Dr-Linux | just wondering if anyone ever luck to setup cisco 7935 conference with asterisk? |
| 14:32.06 | Dr-Linux | Qwell: around? |
| 14:32.09 | Dr-Linux | Qwell[]: ? |
| 14:32.33 | luke-jr | so no ideas on why IAX2 just plain doesn't work? :/ |
| 14:33.03 | *** join/#asterisk ashd (n=ashleydr@user-194-248-151-83.e7even.com) |
| 14:34.15 | *** join/#asterisk HockeyInJune (n=HockeyIn@pool-162-84-187-246.ny5030.east.verizon.net) |
| 14:34.42 | flujan | ManxPower: hi... :D |
| 14:34.59 | flujan | ManxPower: I am still dancing with my hints problem... |
| 14:35.07 | flujan | ManxPower: could you please check it: http://pastebin.com/d676881f3 |
| 14:36.33 | De_Mon | I thought hints wern't needed in 1.4 |
| 14:37.22 | flujan | i really dunno what i did wrong... :( |
| 14:37.23 | *** join/#asterisk masterisk (n=mascool@70.88.122.206) |
| 14:37.45 | *** join/#asterisk Dan0maN_Work (n=dschuh@64.149.174.136) |
| 14:38.13 | *** join/#asterisk dbailey (i=dbailey@nat/digium/x-68e0163c71124df7) |
| 14:38.21 | *** join/#asterisk techie (n=techie@c-67-181-184-170.hsd1.ca.comcast.net) |
| 14:38.24 | jcolp | dbailey: well well... |
| 14:38.57 | dbailey | I figured I might as well know what's going on here. |
| 14:39.01 | flujan | guys, could you please check my dialplan? |
| 14:39.10 | flujan | i dunno what i am doing wrong with it. :( |
| 14:39.22 | *** join/#asterisk Foxygnu (n=FoX@2001:41d0:1:44c8:cafe:cafe:cafe:42) |
| 14:39.36 | ManxPower | flujan: you cannot goto a hint. |
| 14:39.56 | ManxPower | a hint just is, you don't do anything with it in the dialplan other thann confiture it. |
| 14:40.36 | flujan | ManxPower: hum... understood... I can so put a Noop() on the extensions_hints.conf and go to it right? |
| 14:41.01 | *** join/#asterisk rmayorga (i=rmyorg@unaffiliated/rmayorga) |
| 14:41.07 | ManxPower | no. |
| 14:41.09 | luke-jr | flujan: it would be stupid, but probably |
| 14:41.10 | ManxPower | leave it aline |
| 14:41.14 | ManxPower | 'don't use a goto |
| 14:41.18 | ManxPower | why do you want to use a goto? |
| 14:41.24 | luke-jr | if you Goto something, you won't ever come back |
| 14:41.29 | flujan | because the hints lies on another files... |
| 14:41.41 | ManxPower | flujan: then #include otherfile |
| 14:41.49 | ManxPower | you can't goto another file anyway. |
| 14:42.02 | *** join/#asterisk doolph (n=doolph@201.224.81.130) |
| 14:42.16 | Dr-Linux | ManxPower: can i grab your mind? :) |
| 14:42.27 | *** join/#asterisk Cresl1n (i=matt@nat/digium/x-d499a6658ae40417) |
| 14:42.27 | *** mode/#asterisk [+o Cresl1n] by ChanServ |
| 14:42.52 | ManxPower | Dr-Linux: depends on what you want to do with it. |
| 14:43.01 | Dr-Linux | ManxPower: we are having channel name changing issue in same call, we are looking for a solution |
| 14:43.31 | Dr-Linux | ManxPower: our all application depends on channel name, |
| 14:44.18 | ManxPower | Dr-Linux: applications should not rely on the channel name. Give me an example of a name and what it changed to |
| 14:44.50 | *** join/#asterisk [Mr_X] (n=mrx@78-59-18-15.ip.zebra.lt) |
| 14:44.55 | Dr-Linux | so when i call comes and hit's our AGI IVR, so a channel name assigns to this call, but when the caller wants to talk to a Agent, he presses 0 and we dial xxxx extensoin on localhost which follows the queue |
| 14:45.13 | Dr-Linux | so since we dialed xxxx, so it's new call now, with new channel, |
| 14:45.29 | Dr-Linux | can't we do something from IVR to dialplan .. like Goto app or something? |
| 14:45.33 | *** join/#asterisk wunderkin (n=wunderki@ip72-223-86-126.ph.ph.cox.net) |
| 14:45.33 | *** join/#asterisk kv0s (n=kv0s@p4FD21C4D.dip0.t-ipconnect.de) |
| 14:46.04 | Dr-Linux | ManxPower: application should rely on what? |
| 14:46.09 | ManxPower | Dr-Linux: why not just set a variable at the start of the call, then use that to track the call? |
| 14:46.29 | ManxPower | You could also set an account code, then use that to track |
| 14:47.10 | ManxPower | I've never found a need to track a call like that. |
| 14:48.01 | Dr-Linux | ManxPower: that's what we do normally, we set variable in start, but that variable rely on channel, but when channel changed, we lost what we want |
| 14:48.27 | ManxPower | Set(SAVED_CHANNEL=${CHANNEL}) |
| 14:48.34 | ManxPower | or more correctly |
| 14:49.02 | *** part/#asterisk merkurie (n=merkurie@192.153.163.44) |
| 14:49.04 | *** join/#asterisk fakhir (n=Fakhir@unaffiliated/fakhir) |
| 14:49.06 | ManxPower | Set(__SAVED_CHANNEL=${CHANNEL}) The double _ makes the variable be inherited by child channels |
| 14:49.47 | Dr-Linux | hhm... |
| 14:51.22 | Dr-Linux | but technically i makes a new call, when we send caller to an agent from Agi, we dial localhost/protocol/extension |
| 14:51.29 | Dr-Linux | so it create it's new channel |
| 14:51.38 | luke-jr | well then you're being stupid |
| 14:52.00 | Dr-Linux | luke-jr: talking to me? |
| 14:52.02 | luke-jr | yes |
| 14:52.08 | Dr-Linux | ok |
| 14:52.41 | Dr-Linux | so is there anyway to dial dialplan exten from AGI .. directly as Goto or something within a same call? |
| 14:52.42 | ManxPower | __ variables will be set for all channels spawned |
| 14:53.15 | luke-jr | ManxPower: not if he's going through an extra layer of SIP |
| 14:53.17 | Dr-Linux | ManxPower: i see |
| 14:53.24 | luke-jr | Dr-Linux: yeah, it's called a Goto |
| 14:53.37 | Dr-Linux | ohh :S |
| 14:53.45 | ManxPower | luke-jr: that is correct, but not what he is doing, I think. |
| 14:53.54 | Dr-Linux | yeah, i'm going to dialplan from AGI via SIP/localhost |
| 14:54.06 | ManxPower | the call to the agent will still be a new call/channel |
| 14:54.07 | luke-jr | ManxPower: it's what he should be doing, at worst |
| 14:54.27 | ManxPower | luke-jr: all he wants to do is send a call to an agent |
| 14:54.43 | Dr-Linux | luke-jr: is there anyway that i can do without creating new SIP call ? |
| 14:55.27 | luke-jr | Dr-Linux: ... Goto |
| 14:55.35 | ManxPower | Dr-Linux: you can create as many calls as you want. |
| 14:55.59 | ManxPower | luke-jr: if the goto hits a Dial then a new channel will be created. |
| 14:56.11 | Dr-Linux | luke-jr: yes, but i know how Goto works in dialplain, but not sure how to do i.e. agi to dialplan |
| 14:56.18 | luke-jr | ManxPower: but not a new call |
| 14:56.30 | luke-jr | Dr-Linux: the same way you do Dial, obviously |
| 14:56.33 | ManxPower | luke-jr: yes, a new call |
| 14:56.58 | Dr-Linux | well, i'm concerened with channel name |
| 14:56.59 | ManxPower | Dr-Linux: "agi to dialplan"? |
| 14:57.08 | *** join/#asterisk zpertee (n=chatzill@cpe-24-166-81-150.neo.res.rr.com) |
| 14:57.09 | Dr-Linux | ManxPower: that's correct |
| 14:57.11 | ManxPower | Dr-Linux: no matter what you do, the channel name will change |
| 14:57.28 | Dr-Linux | i see |
| 14:57.33 | ManxPower | Dr-Linux: accept this fact and move on to a solution that works with that fact. |
| 14:58.39 | Dr-Linux | ManxPower: so keeping same channel name the solution is >> Set(__SAVED_CHANNEL=${CHANNEL}) ? |
| 14:58.54 | ManxPower | Dr-Linux: yes, as one of the first priorities |
| 14:59.10 | *** join/#asterisk matias_ (n=matias@mail.rack2.com.ar) |
| 14:59.43 | matias_ | where i can download sounds in spanish for asterisk? |
| 14:59.45 | ManxPower | Dr-Linux: that won't keep the same channel name, but will store the old channel name and associate SAVED_CHANNEL with all future child channels. You could name the variable LESBIANS_AGAINST_BUSH for all Asterisk cares. |
| 14:59.48 | *** join/#asterisk kFuQ (n=somedude@c-67-185-123-34.hsd1.wa.comcast.net) |
| 14:59.51 | Dr-Linux | is that channel variables or, names can be any? :S |
| 15:00.19 | Dr-Linux | i see :P |
| 15:01.45 | matias_ | where i can download sounds in spanish for asterisk? |
| 15:02.06 | ManxPower | matias_: From the Digium web site |
| 15:02.09 | *** join/#asterisk ido (n=ido@unaffiliated/ido) |
| 15:02.14 | Dr-Linux | ManxPower: can we send request to DB using socket communicaiton in dialplan? |
| 15:02.26 | ManxPower | Dr-Linux: no. |
| 15:03.48 | ManxPower | you can run any database dialplan apps you compile, of course, but the dialplan doesn't know if it communicates to the database using a socket, named pipe, or mind control rays |
| 16:51.50 | *** join/#asterisk jbot (i=ibot@pdpc/supporter/active/TimRiker/bot/apt) |
| 16:51.50 | *** topic/#asterisk is Asterisk: The Open Source Telephony Application Platform -=- Asterisk 1.2.24 and 1.4.10 released (August 7, 2007) -=- Join #asterisknow or #asterisk-gui for AsteriskNOW and Asterisk-GUI info -=- Join #asterisk-commits to monitor svn changes -=- Join #freepbx for freepbx/#trixbox for trixbox support. -=- 1.2 is in security maintenance mode. No non-security related bug fixes will be applied. |
| 16:52.39 | *** join/#asterisk Barmal (n=info@c-24-30-126-164.hsd1.ga.comcast.net) |
| 16:53.11 | *** join/#asterisk neverblue2 (n=neverblu@unaffiliated/neverblue) |
| 16:54.41 | TechBlazer | Good morning all. I have finally got some things to work. I can make and receive calls as long my phones are plugged into the same router as the Asterisk server. If I take a phone to another network and plug it in the phone will connect to the server and make calls, but there is not sound. Any ideas? |
| 16:55.05 | kv0s | Good Morning? From where do u came from?? uhaa ... good afternoon! ;-) |
| 16:55.07 | Strom_M | TechBlazer: yeah. SIP and NAT don't play well together. |
| 16:55.08 | *** join/#asterisk techie (n=techie@adsl-68-127-122-88.dsl.frsn02.pacbell.net) |
| 16:55.21 | Strom_M | kv0s: it's not even 10 AM in california |
| 16:55.45 | kv0s | TechBlazer: Perhaps no set your default gateway on your asteriskbox? |
| 16:55.54 | *** join/#asterisk datachomper (n=russ@ool-43509aa5.dyn.optonline.net) |
| 16:55.58 | NoNickToPick | 1pm here, EST baby. |
| 16:56.11 | Strom_M | EST? You don't observe daylight saving time? |
| 16:56.15 | kv0s | TechBlazer: Several networks or NAT? |
| 16:56.32 | NoNickToPick | wait, is it later than 1? -=checks his phone=- |
| 16:56.43 | Strom_M | because it's currently almost 12 PM EST; it's currently almost 1 PM EDT |
| 16:56.59 | Barmal | Little bit offtopic but maybe anybody can help. Is there any kind of eq to connect one side into old nortel pbx and the other end into internet asterisk server or ip phone? Kinda like -Line1---->Nortel PBX--->ext2-->something-->internet-->someting--->phone |
| 16:57.06 | Qwell[] | Strom_M: pedantic much? :P |
| 16:57.10 | *** join/#asterisk mog (i=mog@nat/digium/x-496bb2a762c3f686) |
| 16:57.10 | *** mode/#asterisk [+o mog] by ChanServ |
| 16:57.12 | Strom_M | Qwell[]: highly |
| 16:57.19 | NoNickToPick | no, its one, I'm EST. detroit |
| 16:57.27 | Strom_M | xST/xDT mixups are irritating |
| 16:57.44 | Strom_M | NoNickToPick: then you're EDT until daylight saving time ends in the fall |
| 16:58.03 | Strom_M | then you're EST agtain |
| 16:58.06 | Strom_M | again |
| 16:58.08 | NoNickToPick | :P |
| 16:58.18 | Qwell[] | Unless you're Indiana |
| 16:58.23 | Qwell[] | then...well...nobody cares |
| 16:58.25 | Strom_M | hahaha |
| 16:58.26 | TechBlazer | Well I have a router in my office connected to Comcast cable and another one connected to AT&T DSL, and a phone connected to each. |
| 16:58.26 | NoNickToPick | lol |
| 16:58.29 | NoNickToPick | thats awesome |
| 16:58.38 | Qwell[] | it's funny, I'm installing Debian right now, and I just had to pick my timezone |
| 16:58.46 | Qwell[] | "Eastern Indiana" was one of the options. I chuckled |
| 16:59.08 | kv0s | TechBlazer: Mhm. You'll call one sip phone at asterisk 1 from your sipphone 2 at asterisk 2? With two times nat? |
| 16:59.21 | NoNickToPick | Strom_M: hey, this is pretty cool. teliax even sayes they preffer you uplink your Asterisk server to them |
| 16:59.43 | kv0s | 7:39 pm cest |
| 17:00.54 | [TK]D-Fender | TechBlazer: Go read this now : |
| 17:00.56 | [TK]D-Fender | ~sipnat |
| 17:00.56 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 17:01.30 | *** join/#asterisk NightMonkey (n=NightMon@pdpc/supporter/sustaining/NightMonkey) |
| 17:01.46 | *** join/#asterisk Ebola (n=Ebola@host86-144-86-8.range86-144.btcentralplus.com) |
| 17:02.27 | danielxpt | does anyone have any information on debugging sip? I have two asterisk boxes that will not pass the correct informaiton to each other |
| 17:02.29 | TechBlazer | Okay, I'll check it out right now. Thanks for the help kv0s & [TK]D-Fender. |
| 17:02.59 | [TK]D-Fender | danielxpt: PASTEBIN is your friend. Show us something useful. |
| 17:03.01 | [TK]D-Fender | ~pb |
| 17:03.02 | 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 |
| 17:03.58 | kv0s | Hey nice documentation - but i think it make sense to connect the two locations with a vpn/ip-tunnel ... didn't so? |
| 17:04.55 | danielxpt | ok. I put it out on pastebin.com |
| 17:05.03 | danielxpt | http://pastebin.com/m15d04e20 |
| 17:05.20 | danielxpt | this should be with sip debug on. |
| 17:08.26 | [TK]D-Fender | danielxpt: -- Executing Dial("SIP/247-092160c0", "SIP/test/6019832442|300|") in new stack |
| 17:08.27 | [TK]D-Fender | <PROTECTED> |
| 17:08.29 | [TK]D-Fender | <PROTECTED> |
| 17:08.57 | [TK]D-Fender | danielxpt: * is accepting the INCOMING call from your phone and I'm pretty sure your "test" account is screwed up. |
| 17:09.11 | [TK]D-Fender | danielxpt: But communication from your GXP > * is fine. |
| 17:10.10 | danielxpt | [TK]D-Fender: is there a way to bypass knowing the accounts? |
| 17:10.20 | [TK]D-Fender | danielxpt: ...huh? |
| 17:10.23 | *** join/#asterisk Y0da^ (n=jwilson@70.159.118.70) |
| 17:10.33 | *** part/#asterisk Y0da^ (n=jwilson@70.159.118.70) |
| 17:10.48 | [TK]D-Fender | danielxpt: You have no debug info for thie [test] account you setup, and have not shown us the config for it. |
| 17:11.28 | danielxpt | [TK]D-Fender: ok, my goal is to do something similar to a centrex system. these two are attached to a main server. the sip connections are trunked to the main and it pushes to each system |
| 17:11.29 | NoNickToPick | alright guys, have an awesome day, heading out to do my thing of world domination. |
| 17:12.19 | [TK]D-Fender | danielxpt: thats all fine & dandy... your phone is talking jsut fine to *. Its the "going out this [test]" account that did not work, an you have provided nothing to aid in debugging it. |
| 17:12.53 | danielxpt | [TK]D-Fender: ok, so what can I provide to help debug? |
| 17:13.03 | *** join/#asterisk magic_hat (n=geoffdou@h-74-2-87-16.chcgilgm.covad.net) |
| 17:13.28 | magic_hat | hey everyone. Is there an easy way to create an automated company directory in *? |
| 17:13.34 | [TK]D-Fender | danielxpt: Well you haven't got any debug info for the dial attempt out it, haven't shown us the peer is set up right or anything. |
| 17:13.52 | [TK]D-Fender | magic_hat: "show application directory" |
| 17:15.36 | magic_hat | TKD-Fender: i should have been more specific. if a caller presses 8, asterisk plays a recorded list of employees and their extensions. But we have a lot of employees coming and going, so I'm wondering if there's a way to do the same thing without having to do a new voiceover every time someone leaves or gets hired. |
| 17:16.29 | *** join/#asterisk zcionn_ (n=a@58.69.243.203) |
| 17:16.37 | [TK]D-Fender | magic_hat: that app is what you want. get to it! |
| 17:17.41 | danielxpt | [TK]D-Fender: I think I see what was causing it. |
| 17:18.09 | ManxPower | magic_hat: the voiceover is taken from the "record name" from Voicemail |
| 17:18.45 | ManxPower | Try. It. |
| 17:21.38 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 17:23.23 | *** join/#asterisk ZaVoid (n=zavoid@host-24-225-239-34.patmedia.net) |
| 17:24.51 | magic_hat | ManxPower: yeah, pretty cool. |
| 17:25.39 | datachomper | Anybody have an example sip.conf entry for a standard ATA? |
| 17:25.42 | *** join/#asterisk ramindia (n=ramindia@202.63.96.9) |
| 17:26.17 | ramindia | how can make cluster of asterisk with one Mysql Database to handle more calls.. any pointer and documentation |
| 17:26.35 | magic_hat | hrm... it's not matching any of the names that it should be matching. anyone know why that might happen? |
| 17:27.49 | *** join/#asterisk gardo (n=gardo@121.97.245.10) |
| 17:28.32 | kv0s | nice day guys ... it's time to shutdown computers and go home .. ,-) cu |
| 17:28.37 | [TK]D-Fender | datachomper: look at the sample sip.conf * comes with |
| 17:28.41 | magic_hat | all my users are in the default vm context... exten=> s,3, Directory(default[|default[|]]) |
| 17:28.59 | [TK]D-Fender | magic_hat: All those baces are illegal |
| 17:29.07 | [TK]D-Fender | magic_hat: you aren't supposed to type them |
| 17:29.30 | magic_hat | so: Directory(default|default|) |
| 17:29.33 | [TK]D-Fender | exten=> s,3, Directory() ; This is fine too |
| 17:29.48 | [TK]D-Fender | exten=> s,3, Directory(default,default) ; This is fine also |
| 17:29.53 | Strom_M | no |
| 17:30.05 | [TK]D-Fender | Strom_M: oh? |
| 17:30.07 | Strom_M | exten => s,3,Directory(default,default) |
| 17:30.17 | [TK]D-Fender | Well spaces = bad |
| 17:30.24 | Strom_M | no space after the priority and associated comma |
| 17:32.54 | *** join/#asterisk fakhir (n=Fakhir@unaffiliated/fakhir) |
| 17:33.31 | magic_hat | okay, that works. any way to get it to pronounce the name instead of spelling it out? |
| 17:33.48 | Qwell[] | magic_hat: no |
| 17:33.53 | Qwell[] | not with any accuracy |
| 17:34.09 | magic_hat | okay, I got it. gotta get the users to record their names. |
| 17:34.17 | Dr-Linux | any perl guru around? :) |
| 17:34.34 | magic_hat | Dr-Linux: not a guru, but I may be able to help. |
| 17:34.49 | Dr-Linux | cool |
| 17:34.54 | Dr-Linux | magic_hat: can i /msg you? |
| 17:35.03 | magic_hat | sure |
| 17:35.07 | Dr-Linux | thnx |
| 17:37.57 | *** join/#asterisk generalhan (n=asd@ip67-90-64-2.z64-90-67.customer.algx.net) |
| 17:38.22 | magic_hat | TKD-Fender: should I be denying all but gsm with Teliax? |
| 17:38.40 | generalhan | so i love it when the ERROR messages in the CLI show BAD! BAD! BAD! .... lol, so is that bad ? |
| 17:39.02 | generalhan | maybe some one could give me some insight on this: http://generalhan.pastebin.ca/651357 |
| 17:39.06 | [TK]D-Fender | magic_hat: You shouldn't be doing EITHER. your users should be recording their names for the directory. |
| 17:39.12 | *** join/#asterisk Keltus (n=Keltus@about/cooking/nakedchef/beefstew/Keltus) |
| 17:39.49 | [TK]D-Fender | magic_hat: I think you should be making up your own mind about what codec and means you want to use for your ITSP's. |
| 17:40.27 | [TK]D-Fender | magic_hat: By default I'd say, no use ULAW for everything for quality, use SIP for reliability and transportability, but then again, this may be completely contrary to what you want. |
| 17:42.17 | Aces1Up | what are the requirements for an provider so that i can send my own unique caller id? |
| 17:44.01 | blackdark | is that a way to not have to enter the extension # when calling the voicemail ? |
| 17:44.12 | blackdark | I mean when I call the voicemail with my own telephone |
| 17:44.38 | *** part/#asterisk ramindia (n=ramindia@202.63.96.9) |
| 17:44.55 | *** join/#asterisk Jingles (n=dfbarth@39.183.dowl.anc.borealisbroadband.net) |
| 17:45.39 | Netgeeks | nice ERROR messages generalhan! |
| 17:45.55 | generalhan | Netgeeks: oh yeah ?? nice huh ? ! lol |
| 17:46.12 | generalhan | i get those occasionally over the course of a day |
| 17:46.22 | generalhan | i dont really understand whats going on |
| 17:46.22 | Netgeeks | hrm, yes, I would consider the ignoring of a sip ack as a bad bad bad thing |
| 17:46.53 | [TK]D-Fender | blackdark: "show application voicemailmail" <- read the instructions |
| 17:47.12 | generalhan | Netgeeks: what do i do to try and troubleshoot this issue ? |
| 17:47.17 | *** join/#asterisk apardo (n=apardo@28.65.220.87.dynamic.jazztel.es) |
| 17:47.23 | blackdark | [TK]D-Fender, thanks |
| 17:47.25 | Netgeeks | you got a sip ack for a sip transaction, and the channel was locked by something that prevented asterisk from updating the channel with respect to the ack |
| 17:47.48 | Netgeeks | any more than that layman guess is beyond me, chan_sip is a nightmare of code |
| 17:47.58 | generalhan | yay :( |
| 17:48.45 | *** join/#asterisk apardo (n=apardo@28.65.220.87.dynamic.jazztel.es) |
| 17:48.50 | *** part/#asterisk techie (n=techie@adsl-68-127-122-88.dsl.frsn02.pacbell.net) |
| 17:48.51 | Netgeeks | I'd try and replicate it in a somewhat sterile environment, and get full debug/verbose (level4) with sip debug turned on from it happening |
| 17:49.19 | magic_hat | TKD-Fender: I was thinking ulaw might be causing some of my troubles w/ echo, etc. because it's so big. |
| 17:49.28 | Netgeeks | and then I'd post the results on mantis and poke Olle to take a look at it |
| 17:49.50 | generalhan | Netgeeks: thats my issue ... i dont have another environment to test in ... and turning sip debug on, on the CLI at the highest verbosity already and i wouldnt be able to see anything |
| 17:50.09 | Netgeeks | thats why I said a sterile environment |
| 17:50.18 | generalhan | it would be scrolling so much stuff ... and of course this is our most active phone user in the office. he makes like 400 calls a day |
| 17:50.28 | Netgeeks | take your backup system for your production system, and run the test there with very limited test use only |
| 17:51.25 | Netgeeks | but then if this is load related, you wouldn't be able to reproduce it... |
| 17:51.45 | generalhan | my backup system is the system im using ... my old server took a dive. but i have no extra switches lying around to mess with ... so this net is the only one i have ! hows that for redundancy lol |
| 17:52.11 | *** join/#asterisk Mercestes (n=Merceste@rrcs-71-41-157-70.sw.biz.rr.com) |
| 17:52.15 | crimethinker | noatime |
| 17:52.21 | Netgeeks | heh, how is that temporary spare tire on the freeway? ;) |
| 17:52.33 | Netgeeks | do you have alot of free drive space? |
| 17:52.48 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 17:52.52 | generalhan | Netgeeks: i actually lucked out because my backup is actually MUCH newer and MUCH faster than the original ! |
| 17:53.14 | Netgeeks | if you do, turn up all the logging I mentioned to disk, and then when it happens grab that file and turn all the extra chatter off |
| 17:53.40 | neverblue | where is the best place for my php script to grab asterisk information, in the case of an error? |
| 17:54.21 | generalhan | Netgeeks: im glad you mentioned that ... how do i log directly to a file ?? i have a 300GB SCSI disk in this system just for log files to be transfered to it .,.. so i defintely have the space, but not the know-how to do it |
| 17:54.55 | [TK]D-Fender | magic_hat: codecs have nothing to do with echo |
| 17:54.58 | generalhan | or maybe you can point me to a link that explains it |
| 17:55.01 | Netgeeks | you edit /etc/asterisk/logger.conf and set full logging, the sample file that asterisk puts there tells you how |
| 17:55.14 | generalhan | perfet ! ill take a look now ! |
| 17:55.19 | Netgeeks | then reload logger.conf or just reload everything |
| 17:55.41 | Netgeeks | then on cli do your core set verbose 4 and core set debug 4 and sip set debug |
| 17:55.55 | magic_hat | [TK]D-Fender: So where do I start debugging echoey calls? |
| 17:56.01 | Netgeeks | then look away fast before you freak out over the boatload of cli noise |
| 17:56.07 | Netgeeks | wait for it to happen |
| 17:56.18 | Netgeeks | and then you can reset logger.conf back to it's old settings |
| 17:56.54 | neverblue | Netgeeks, was that directed at me? |
| 17:56.59 | *** part/#asterisk Cresl1n (i=matt@nat/digium/x-d499a6658ae40417) |
| 17:57.36 | Netgeeks | neverblue, nope, that was to generalhan |
| 17:58.00 | flujan | guys, how can I check my channels usage using asterisk? |
| 17:58.21 | flujan | I installed a sip trunk... Need to know how many calls are online using this trunk... |
| 17:58.53 | Netgeeks | you need this automated or you just want a method to see? |
| 17:59.09 | flujan | I know that every channel has a call-count variable which is checked agains the call-limit... |
| 17:59.10 | [TK]D-Fender | flujan: "show channels concise" |
| 17:59.19 | flujan | Netgeeks: automated... |
| 18:00.27 | Netgeeks | hrm, you could use a script that conncects to the cli and scrapes the results of sip show channels or show channels concise.... |
| 18:01.06 | Netgeeks | you could connect to manager interface and do something similar, you could stay connected and have a simple state machine that would track that info and such for you |
| 18:02.19 | flujan | hum... I see... will give it a try guys.. thanks for the help. :D |
| 18:02.43 | Netgeeks | one of the tricks I've played in the past is I assign every channel I want to track to a particular group (SetGroup or Set(GROUP()) and then I use UserEvent at the start of each channel to pop the count of the group at the time the channel starts |
| 18:03.08 | Netgeeks | that way you get a running count of channels each time a new channel is started... good for tracking peaks |
| 18:03.34 | Netgeeks | but not wholy accurate because you don't get notice when a channel hangs up until the next new channel event.... |
| 18:07.40 | *** join/#asterisk zydrunas_ (n=zydrunas@24-119-29-130.cpe.cableone.net) |
| 18:08.24 | *** join/#asterisk dharrigan (n=dharriga@82-71-62-76.dsl.in-addr.zen.co.uk) |
| 18:09.48 | blackdark | is there a config somewhere where I could set the proper file permissions for voicemail ? |
| 18:10.15 | blackdark | i'd like to use vmail.cgi to let users manage their voicemail |
| 18:10.29 | blackdark | but files are created like root:root 700 |
| 18:10.43 | blackdark | so the webserver ID can't read nor delete the files |
| 18:11.03 | *** join/#asterisk saftsack (n=saftsack@pD9E07758.dip.t-dialin.net) |
| 18:11.41 | Mercestes | blackdark: root:asterisk 770 would work |
| 18:11.49 | *** join/#asterisk jmesquita (n=jmesquit@200.162.229.225.user.ajato.com.br) |
| 18:11.57 | Mercestes | blackdark: or root:asterisk 760 to be more realistic. |
| 18:12.19 | blackdark | where can I adjust that so every new file will get those permissions ? |
| 18:12.42 | Mercestes | blackdark: In your umask settings under your user profile for created files |
| 18:13.00 | Mercestes | blackdark: But i fyour creating files, why can't you do that on the fly? |
| 18:13.27 | Mercestes | blackdark: chown/chmod work great for me. |
| 18:13.40 | blackdark | I think we misuderstood |
| 18:13.52 | blackdark | I meant when asterisk create a voicemail file |
| 18:14.01 | Mercestes | blackdark: Don't run asterisk as root. |
| 18:16.05 | *** join/#asterisk Paul_UK (n=foo@78.32.14.84) |
| 18:16.22 | *** join/#asterisk shido6 (n=shido6@d221-68-200.commercial.cgocable.net) |
| 18:16.49 | Paul_UK | hey guys, with asterisk 1.2, where does the vmexten directive reside? |
| 18:17.07 | Paul_UK | currently, it defaults to asterisk and my snom phone needs something else in its place |
| 18:17.26 | blackdark | Mercestes, but even if I run asterisk as an another user, problem is still there |
| 18:17.36 | blackdark | the webserver doesn't as the same user as asterisk |
| 18:17.45 | blackdark | so it won't have access to the voicemail files |
| 18:18.18 | Mercestes | blackdark: You can run asterisk as webuser:asterisk?? |
| 18:18.41 | blackdark | Mercestes, I don;t think it's a good idea |
| 18:18.43 | Mercestes | Again, I'm pretty sure the owner and mask is a user profile setting |
| 18:19.00 | Mercestes | Add your webuser to the asterisk group then |
| 18:19.02 | *** join/#asterisk salvatore2 (n=canberk@88.240.162.7) |
| 18:19.54 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 18:20.29 | salvatore2 | hi |
| 18:20.55 | blackdark | ok now asterisk runs as root, but if the owner/mask would be a user profile, when I create a file with that user, it should get that mask/owner |
| 18:20.58 | blackdark | but I don't |
| 18:21.07 | blackdark | so it's not a user profile setting |
| 18:21.13 | Mercestes | ok |
| 18:21.49 | Mercestes | do a touch corydon and check out the owner/mask of the file you create. |
| 18:21.59 | Mercestes | It got those settings from somewhere. |
| 18:22.42 | gardo | blackdark: why not use the chmod or chown inside your dialplan? |
| 18:23.52 | blackdark | gardo, I can do that in the dialplan ? |
| 18:23.56 | gardo | yep |
| 18:24.06 | gardo | you can put unix commands inside your dialplan |
| 18:24.11 | generalhan | when a remote SIP user qaulifies the latency is 488ms ... is that bad? the user is complaining that both ends of the call are very choppy and i cant figure out why |
| 18:24.13 | blackdark | how can I do that ? |
| 18:24.17 | Mercestes | System() |
| 18:24.19 | *** join/#asterisk sudoer (n=jtoy@mail.backchannelmedia.com) |
| 18:24.26 | blackdark | Mercestes, thanks |
| 18:24.28 | *** join/#asterisk ccesario (n=ccesario@189-19-9-100.dsl.telesp.net.br) |
| 18:24.31 | Mercestes | np |
| 18:24.43 | sudoer | how can I get asterisk to server voicemail over http? |
| 18:24.50 | blackdark | vmail.cgi |
| 18:24.51 | salvatore2 | generalhan, 488ms? |
| 18:24.52 | gardo | call out chmod or chown after running the voicemail |
| 18:24.58 | salvatore2 | it's sooo fuckin late |
| 18:25.02 | blackdark | sudoer, working on that right now |
| 18:25.17 | generalhan | salvatore2: "its so late" ? |
| 18:25.31 | sudoer | asterisk doesn't provide a simple way to stream the file or something via the manager ? |
| 18:25.39 | salvatore2 | i meant the latency is too much |
| 18:25.51 | salvatore2 | ~500ms is way too much |
| 18:25.52 | jbot | salvatore2: okay |
| 18:26.03 | generalhan | hmm, wonder why it is that way |
| 18:26.15 | salvatore2 | try pinging the other server |
| 18:26.19 | blackdark | sudoer, if you compiled your asterisk installation, search in the source for vmail.cgi |
| 18:26.19 | salvatore2 | and tell me the ping time |
| 18:26.30 | generalhan | salvatore2: its about the same |
| 18:26.38 | generalhan | either just under or just over that number |
| 18:26.44 | salvatore2 | then there is nothing you can do practically |
| 18:27.02 | Paul_UK | currently, it defaults to asterisk and my snom phone needs something else in its place |
| 18:27.05 | Paul_UK | hey guys, with asterisk 1.2, where does the vmexten directive reside? |
| 18:28.00 | generalhan | salvatore2: its actually a lot less now with ping ... i get the occasional 455ms but 75% is at 50ms - 70ms |
| 18:28.45 | generalhan | gotta be on my end somewhere ... just now all my sip users where that high ... and now they are all down to 60ms |
| 18:29.10 | *** join/#asterisk ToyMan (n=Stuart@fw.hvs.bsdwebsolutions.com) |
| 18:34.20 | *** join/#asterisk mercestes (n=Merceste@rrcs-71-41-157-70.sw.biz.rr.com) |
| 18:34.56 | [TK]D-Fender | Paul_UK: sip.conf jsut like 1.4 |
| 18:36.58 | *** join/#asterisk Mrtaz (n=wcl@h-68-167-99-74.cmbrmaor.covad.net) |
| 18:37.28 | Mrtaz | hey all, I have a question about the zap destroy channel <num> command, after you issue it, how do you get the channel back with out restarting asterisk? |
| 18:39.52 | steliosk | has anyone tested Digium's transcoder card ? |
| 18:40.17 | [TK]D-Fender | Mrtaz: you have to reload chan_zap.so at a minimum if not * as a whole. You should NOT use that means unless absolutely necessary. |
| 18:40.59 | Mrtaz | hmm ok, well then heres the problem that causes me to use it |
| 18:41.04 | x86 | steliosk: digium makes a transcoder offload card? |
| 18:41.20 | [TK]D-Fender | x86: Yes, wake up and smell the toast burning :) |
| 18:41.28 | steliosk | x86: yes http://www.digium.com/en/products/hardware/tc400b.php |
| 18:42.25 | Mrtaz | I have 2 phyiscal fax machines connected to ZAP/1 and ZAP/2, if I attempt to send a fax from either of those, they bridge to the first available zap channel, usually Zap/5 or Zap/6 but cause the line to just hang and if anyone else tries to dial out asterisk reuses that channel and all the caller can hear is a loud static sound |
| 18:42.41 | sudoer | <PROTECTED> |
| 18:42.57 | x86 | [TK]D-Fender: hehe |
| 18:43.20 | mercestes | sudoer: Cronjob to copy them into yoru wwwroot maybe? Or just make the voicemail directory your wwwroot |
| 18:43.25 | [TK]D-Fender | Mrtaz: set absolute tomeouts on your zaptel dials for them. |
| 18:43.32 | Mrtaz | k |
| 18:43.33 | Mrtaz | thanks |
| 18:43.35 | sudoer | oh, I can do that, cool |
| 18:46.56 | *** join/#asterisk zydrunas_ (n=zydrunas@24-119-29-130.cpe.cableone.net) |
| 18:51.31 | wchalco | Hi,.. i need a little help my friends |
| 18:52.04 | wchalco | i wanna know haow i can get a variable from asterisk manager in the dial plan written in extensions.conf |
| 18:52.38 | wchalco | i tried using setvar action in asterisk manager and .. using var= xxx in the originate action.. |
| 18:53.13 | wchalco | i am using perl and php.. to send accions to asterisk manager.. but i cant get this variables in the dialplan... |
| 18:53.36 | *** join/#asterisk Tako-san (n=Tako-san@154.5.212.245) |
| 18:55.24 | dlynes_laptop | wchalco: Set(VARNAME=value) is the new way of doing it in Asterisk 1.2 and higher (SetVar(name=value) was the old way of doing it in Asterisk 1.0) |
| 18:56.09 | dlynes_laptop | Have a great day, peeps |
| 18:56.11 | dlynes_laptop | Gotta run |
| 18:57.35 | wchalco | mm.. dlynes_laptop.. the actions that i need to send are in a php script |
| 18:57.51 | wchalco | Actions like originate and monitor works ok.. |
| 18:58.03 | *** join/#asterisk UnixDog (n=UnixDog@adsl-69-234-186-68.dsl.irvnca.pacbell.net) |
| 18:58.14 | blackdark | anyone could recommend me a howto to create a macro in the dialplan ? |
| 18:58.26 | wchalco | but when i wanna send this variables trougth Set(VARNAME = value)... notrhing happens in mi CLI console. |
| 19:02.29 | UnixDog | [macro-whatever] |
| 19:03.28 | *** join/#asterisk kkn088 (n=kikoun@84.4.74.213) |
| 19:03.44 | *** join/#asterisk Delvar (n=Delvar@host-83-146-53-46.bulldogdsl.com) |
| 19:04.05 | *** join/#asterisk toddejohnson (n=toddejoh@69.220.214.65) |
| 19:04.33 | *** join/#asterisk Lawbringer (n=Lawbring@84-45-215-247.no-dns-yet.enta.net) |
| 19:04.41 | Mrtaz | hmm how would one go about adding absolute timeout to a zap channel that is dialing out? the extension is only 3 lines, would I just plop it on there? |
| 19:05.49 | *** join/#asterisk ToyMan (n=Stuart@fw.hvs.bsdwebsolutions.com) |
| 19:06.25 | [TK]D-Fender | Mrtaz: "show application dial" |
| 19:07.00 | Paul_UK | hey guys, im trying to setup my snom phone to enquire the asterisk server for the vm for its extension, im getting declined. here is the output: http://pastebin.com/m15f1d9 , can anyone help? |
| 19:07.54 | [TK]D-Fender | Paul_UK: Funny I don't SEE anything about that in there. |
| 19:08.42 | hohum | can anyone recommend a GOOD carrier who will speak SIP to an asterisk box and give me a DID? |
| 19:09.19 | Nugget | http://asterlink.com if you need a US toll free DID, http://connect.voicepulse.com if you need a local DID. |
| 19:09.30 | *** join/#asterisk Op3r (n=Op3r@121.97.147.190) |
| 19:09.31 | [TK]D-Fender | .... |
| 19:09.32 | [TK]D-Fender | telnet |
| 19:09.37 | [TK]D-Fender | :/ |
| 19:09.45 | *** part/#asterisk UnixDog (n=UnixDog@adsl-69-234-186-68.dsl.irvnca.pacbell.net) |
| 19:09.47 | [TK]D-Fender | doh |
| 19:09.55 | Paul_UK | TKD, im a total newb.. thats what happens when I push the button. The code im using is exten => asterisk,1,VoiceMailMain?(${CALLERID(num)}@${VMCONTEXT}) and the directive is called app-vmmain-custom which is called from the directive app-main. |
| 19:10.15 | *** join/#asterisk Strom_C (n=strom@adsl-69-105-23-47.dsl.irvnca.pacbell.net) |
| 19:10.33 | blackdark | when a SIP extention is busy with an anohter line, if an another phone tries to call it, isn't supposed to go to the voicemail ? |
| 19:11.43 | [TK]D-Fender | Paul_UK: "thats nice", but its clearly not using that line you just showed us |
| 19:12.32 | [TK]D-Fender | blackdark: First, there is no such thing as a SIP "extensions", and how a call get treated depends on the device handling it. Voicemail is a DIALPLAN app, and your dialplan does what you TELL it to. |
| 19:12.56 | Paul_UK | TKD: sigh.. well im using freepbx.. and im getting the idea its a pos. Its cool tho, cos soon i leave this job and then i can concentrate on learning asterisk cli |
| 19:13.17 | De_Mon | why does zap get a destory channel, but everything else has to use soft hangup? |
| 19:13.29 | [TK]D-Fender | Paul_UK: Guess you're just wasting your time and ours now... |
| 19:13.46 | Paul_UK | yep |
| 19:13.50 | Paul_UK | im going home lol |
| 19:13.58 | [TK]D-Fender | De_Mon: they decided its not cool to destroy non-zap channels :) You can keep on breaking zap all you want for now :) |
| 19:14.27 | De_Mon | I can never remember 'soft hangup' but DESTORY, thats a memorable one |
| 19:15.07 | De_Mon | i tried sip hangup, sip destory... then just started reading thru the bloody help list |
| 19:15.41 | De_Mon | oh, I tried channel hangup too.. |
| 19:16.20 | *** join/#asterisk Mrtaz (n=wcl@h-68-167-99-74.cmbrmaor.covad.net) |
| 19:21.45 | toddejohnson | I am looking for a good wifi phone. Any ideas? |
| 19:22.18 | denon | invent one :) |
| 19:22.25 | denon | there's no such thing as a good one :) |
| 19:22.37 | toddejohnson | denon:I am quickly finding that out |
| 19:22.57 | denon | having said that, the least worst of them seems to be the hitatchi (IP500s or such) for business/robust |
| 19:23.21 | denon | the linksys (330s? the win-based ones) also seem to be good for bells and whistles |
| 19:23.37 | denon | ie: roaming around open access points, and a built-in browser for captured wifi portals |
| 19:24.10 | toddejohnson | ok I promised a client voip over wifi and the utstarcom I got is just not working out. |
| 19:26.39 | De_Mon | I had a linksys wifi phone, it lasted about 3 hours per charge |
| 19:26.49 | denon | toddejohnson: from what I've heard of the utstarcom, it's about the worst phone you can buy |
| 19:26.54 | denon | probably the cheapest, though |
| 19:26.55 | blackdark | what could be wrong in my config when the voicemail doen't the custom user greeting ? |
| 19:27.05 | blackdark | doesn't user |
| 19:27.07 | blackdark | doesn't user |
| 19:27.10 | CtRiX | i'm using samsungs wip 6000 |
| 19:27.17 | blackdark | sorry for that |
| 19:27.26 | De_Mon | blackdark did you record a custom greeting? |
| 19:27.34 | CtRiX | very good but: 1) short batery life and not stun support |
| 19:27.47 | De_Mon | CtRiX how short? |
| 19:27.59 | blackdark | De_Mon, yes |
| 19:28.13 | De_Mon | what is the line you use to call voicemail? |
| 19:28.29 | De_Mon | well, send someone to voicemail (dialplan line) |
| 19:28.33 | blackdark | exten => 5874,2,VoiceMail(5874@default) |
| 19:29.05 | *** join/#asterisk Strom_M (n=strom@adsl-69-105-23-47.dsl.irvnca.pacbell.net) |
| 19:30.09 | blackdark | de_mon : http://pastebin.arslinux.com/11980 |
| 19:32.00 | De_Mon | blackdark pastebin debug logs of the call |
| 19:32.13 | *** join/#asterisk Nockian- (i=nockian@unaffiliated/nockian) |
| 19:32.37 | CtRiX | De_Mon, without recharging, idle, 4 hours. |
| 19:32.47 | CtRiX | 1 hour of call when not idling |
| 19:32.51 | De_Mon | hehe |
| 19:32.58 | toddejohnson | CtRiX: Is that samsung availble in the us yet? |
| 19:33.11 | CtRiX | don't know |
| 19:33.32 | CtRiX | it also has a camera and can send sms (text) and mms |
| 19:33.40 | CtRiX | * don't support those things, though |
| 19:33.51 | De_Mon | thats pretty bad |
| 19:34.33 | De_Mon | I thought priority jumpign was depriciated |
| 19:34.39 | De_Mon | jumping |
| 19:34.44 | Strom_M | deprecated |
| 19:35.00 | De_Mon | yes, that too |
| 19:35.03 | hmmhesays | you can make most apps do it |
| 19:35.14 | hmmhesays | or set it in extensions.conf ? no? |
| 19:35.17 | De_Mon | _now_ what about in 1.6 |
| 19:35.50 | *** join/#asterisk |dennis| (n=dennis@200.32.233.84) |
| 19:36.03 | De_Mon | extensions.conf is for something a little different, but im pretty sure thats depreciated too |
| 19:36.05 | *** join/#asterisk ToyMan (n=Stuart@fw.hvs.bsdwebsolutions.com) |
| 19:36.20 | hmmhesays | whoa I have the exact shirt hyde is wearing on "that 70's show" |
| 19:36.22 | De_Mon | fallthrough or somesuch |
| 19:36.38 | De_Mon | stole in from the prop room did you |
| 19:36.44 | De_Mon | it |
| 19:37.51 | *** join/#asterisk guillote_GNU (n=bancaria@host136.200-117-227.telecom.net.ar) |
| 19:41.51 | De_Mon | blackdark it sounds like it can't find the files |
| 19:41.51 | [TK]D-Fender | blackdark: "show application voicemail" <---- |
| 19:42.29 | [TK]D-Fender | De_Mon: Strike One... |
| 19:42.46 | blackdark | actually I changed a bit my dialplan and made it work |
| 19:42.46 | De_Mon | oh.. he has to specify which greeting to play |
| 19:42.54 | blackdark | just added a b in front of the user name |
| 19:43.05 | blackdark | b for busy |
| 19:43.19 | blackdark | exten => 5874,2,VoiceMail(b5874@default) |
| 19:43.22 | blackdark | like that |
| 19:43.29 | [TK]D-Fender | blackdark: I would suggets putting it in the 2nd parameter... |
| 19:43.36 | [TK]D-Fender | suggest |
| 19:44.39 | *** join/#asterisk anthm (n=anthm@adsl-69-216-26-86.dsl.milwwi.ameritech.net) |
| 19:44.39 | *** mode/#asterisk [+o anthm] by ChanServ |
| 19:44.46 | blackdark | like VoiceMail(5874@default,b) ? |
| 19:46.39 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 19:48.05 | Dan0maN_Work | all: my company is looking to implement this next year, and i've been tasked to find out hardware needs for servers to support my company. we have ~300 extensions on our current pbx, with ~270 stations. anyone have any insight as to how much storage would be needed for voicemail for that many users? |
| 19:48.41 | Strom_M | 200 gigabytes would be massive overkill |
| 19:48.45 | [TK]D-Fender | Dan0maN_Work: With HD's these days, its an AFTERTHOUGHT |
| 19:49.01 | Dan0maN_Work | HD's? |
| 19:49.06 | Strom_M | poutine or similar |
| 19:49.39 | [TK]D-Fender | Strom_M: If by any freak of luck you end up here, I'll buy you one :) |
| 19:49.47 | Strom_M | here == montreal? |
| 19:49.57 | blackdark | umm ok I fixed it with a b as second parameter, but how could make it dynamic ? |
| 19:49.57 | *** part/#asterisk CtRiX (n=CtRiX@aretha.navynet.it) |
| 19:50.11 | [TK]D-Fender | Strom_M: Yup |
| 19:50.16 | Strom_M | [TK]D-Fender: how's next week sound? |
| 19:50.31 | blackdark | so playing the busy message when the line is busy, unavail when the line is not ava ? |
| 19:50.37 | [TK]D-Fender | Strom_M: I'm not leaving town, so if you're serious, gimme a shout. |
| 19:50.43 | Strom_M | alright |
| 19:50.49 | blackdark | who's talling about poutine ? |
| 19:50.55 | [TK]D-Fender | blackdark: You'll have to determine that and do it in the dialplan |
| 19:50.57 | Dan0maN_Work | D-Fender: sorry. not completely familiar here yet. what do you mean by HD's? |
| 19:51.00 | Strom_M | i'll let you know when I have a clearer idea of which day I'll actually be in montreal |
| 19:51.00 | blackdark | s/talling/talking/ |
| 19:51.07 | [TK]D-Fender | Dan0maN_Work: Hard Drives. |
| 19:51.12 | blackdark | jbot, yes |
| 19:51.12 | jbot | You don't say! |
| 19:51.13 | Dan0maN_Work | oh |
| 19:51.13 | Dan0maN_Work | heh |
| 19:51.38 | blackdark | I live close to Montreal |
| 19:51.40 | [TK]D-Fender | ;) |
| 19:51.49 | Strom_M | jbot, maybe |
| 19:51.50 | jbot | it has been said that maybe is maybe, or the opposite of maybe |
| 19:52.03 | blackdark | [TK]D-Fender, ok how could i do that ? |
| 19:52.22 | blackdark | I supposed I would have to catch the even type or so |
| 19:52.28 | [TK]D-Fender | blackdark: I personally dislike having 2 messages for VM.... it enslaves your users... |
| 19:52.29 | Dan0maN_Work | for redundancy, i was looking to build a raid array with iSCSI to 2 redundant servers (if one fails, i could connect to the iSCSI with the second). is that overkill? |
| 19:52.49 | [TK]D-Fender | Dan0maN_Work: RAID 1 works for me... |
| 19:54.07 | *** part/#asterisk bethor (n=Administ@dslb-082-083-032-092.pools.arcor-ip.net) |
| 19:54.14 | blackdark | [TK]D-Fender, yeah it's true |
| 19:54.16 | *** join/#asterisk drwelby (n=mpfister@mail.enplan.com) |
| 19:54.25 | Dan0maN_Work | D-Fender: ok. what do you think about redundant servers. am i going overboard in thinking this out? |
| 19:54.40 | blackdark | umm a poutine for dinner could good |
| 19:55.10 | *** join/#asterisk guillote_GNU (n=bancaria@host136.200-117-227.telecom.net.ar) |
| 19:55.32 | [TK]D-Fender | Dan0maN_Work: depends on the impact of down-time, etc... I personally feel its often easier to just build 2-3 economical servers and keep it around and have someone there to shut down the primary. |
| 19:56.15 | [TK]D-Fender | Dan0maN_Work: This is where ODBC voicemail, etc comes in handy so your DB server can be strong, and you * server very basic. |
| 19:57.21 | Dan0maN_Work | D-Fender: true. ok. thanks for the insight |
| 19:58.35 | blackdark | thanks for you help guys |
| 20:02.18 | neverblue | where is the best place for my php script to grab asterisk information, in the case of an error? |
| 20:02.36 | *** join/#asterisk Ebola (n=Ebola@host86-144-86-8.range86-144.btcentralplus.com) |
| 20:02.43 | [TK]D-Fender | neverblue : What information, grabbed how, and when? |
| 20:03.17 | neverblue | when: error, what: information about the call (maybe something like, why the call failed), and how, using php |
| 20:03.36 | neverblue | too many commas in that |
| 20:03.55 | [TK]D-Fender | neverblue : You're going be have to be very specific on what you classify as an "error", and when you expect to trap it, and then deal with it. |
| 20:04.16 | neverblue | hmm, more specific, ok, so when a call fails |
| 20:04.40 | *** join/#asterisk jebba (n=jebba@220-179-89-200.fibertel.com.ar) |
| 20:05.10 | neverblue | is there an error log, just like there is a call log, generated ? |
| 20:07.34 | neverblue | for example, a Master.cvs is generated by * atm, for logging all calls |
| 20:13.08 | *** join/#asterisk metfan2007 (n=metfan20@189.136.86.34) |
| 20:13.19 | [TK]D-Fender | neverblue : And how do you define a call "failing"? |
| 20:14.46 | metfan2007 | hi all!!! How can "monitor" a call? I mean, no to record it, I want only to listen an existing call for quality reasons... any idea? |
| 20:15.07 | neverblue | Fender what would you define a call as failed? |
| 20:17.21 | [TK]D-Fender | neverblue : You're the one asking on how to track a failure, it'd help if you could define it. |
| 20:17.54 | neverblue | when a call fails, abnormally |
| 20:19.03 | neverblue | say I have a three way call, in the middle of the call, one person drops |
| 20:19.19 | neverblue | causing the other two to lose connection as well |
| 20:19.33 | *** join/#asterisk pagec (n=pagec@cpe-74-73-191-68.nyc.res.rr.com) |
| 20:20.13 | [TK]D-Fender | neverblue : That sounds like it'd require some serious re-writes to sip.conf. Maybe logger.conf might catch something useful... |
| 20:20.25 | [TK]D-Fender | metfan2007: "show application chanspy" |
| 20:21.29 | neverblue | maybe the * server is having load issues at the time of the call, maybe the termination's server fails |
| 20:22.08 | neverblue | its really difficult to troubleshoot errors, unless you are watching the * -r roll by, with +vvvvvvv |
| 20:22.14 | pagec | using AEL I have globals {x=5555551212 y=text} and when i do something like NoOp${GLOBAL(x)}; i get 5555551212 but when i do NoOp${global(y)}; i get an empty string. does AEL support non-numeric globals? |
| 20:27.56 | *** join/#asterisk ToyMan (n=Stuart@72.168.167.241) |
| 20:32.03 | *** join/#asterisk stefmtl (n=stef@stef.istop.com) |
| 20:33.15 | *** join/#asterisk mrdigital (n=MrDigita@pool-72-94-134-205.phlapa.east.verizon.net) |
| 20:33.43 | stefmtl | hello, i always have files like that 5|1186625433.4150.csv in addition to Master.csv , just on one server, I don't understand why ? |
| 20:34.01 | stefmtl | in my cdr-csv log directory |
| 20:36.11 | neverblue | did you look at them? |
| 20:37.58 | *** join/#asterisk ToyMan (n=Stuart@72.168.167.241) |
| 20:39.06 | stefmtl | neverblue : yes each file is one line of CDR |
| 20:46.39 | Mrtaz | anyone have a patch file for NVFaxDetect so that it will work with * 1.4? the one at http://www.voip-info.org/wiki/view/NewmanTelOnAsterisk14 doesnt seem to work |
| 20:46.58 | *** join/#asterisk pnlarsson (n=pnlarsso@c83-248-12-187.bredband.comhem.se) |
| 20:47.53 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 20:48.27 | pnlarsson | Q: How to get 0313011798 out of a var that contains "0313011798"<sip:0313011798@172.18.1.250> in dialplan? I can do it in php etc, but would like to avoid that. |
| 20:48.50 | pnlarsson | The number after sip: and before @ |
| 20:51.09 | Strom_M | how about the CUT() function |
| 20:53.10 | *** join/#asterisk jkimball4 (n=jerrid@ip24-252-32-248.om.om.cox.net) |
| 20:53.14 | l2trace9999 | anyone know of some docuementation on setting up skill based queues ? |
| 20:53.22 | jkimball4 | What can cause an agent to be "Invalid" |
| 20:53.44 | pnlarsson | CUT only takes one arg for the delimeter |
| 20:53.51 | *** join/#asterisk mercestes (n=Merceste@rrcs-71-41-157-70.sw.biz.rr.com) |
| 20:53.54 | Strom_M | pnlarsson: so you cut it twice |
| 20:53.57 | Strom_M | big deal |
| 20:54.49 | pnlarsson | will try |
| 20:55.08 | Netgeeks | Set(X=${CUT(VARNAME,<,1) |
| 20:55.14 | Netgeeks | Set(X=${CUT(VARNAME,<,1)}) |
| 20:55.17 | Netgeeks | :) |
| 20:55.32 | Netgeeks | that will get everything in front |
| 21:00.24 | pnlarsson | exten => _X.,n,Set(X=${CUT(PAssertedIdentity,:,2)}) |
| 21:00.24 | pnlarsson | exten => _X.,n,Noop(${X}) |
| 21:00.24 | pnlarsson | exten => _X.,n,Set(X=${CUT(X,@,1)}) |
| 21:00.24 | pnlarsson | exten => _X.,n,Noop(${X}) |
| 21:00.35 | pnlarsson | Works! Thanks! |
| 21:02.35 | fujin | anyone know if you can #include inside AEL? |
| 21:06.51 | fujin | nevermind, you can |
| 21:06.55 | stefmtl | quit |
| 21:07.00 | fujin | in asterisk 1.4.10, what is the preferred way to query a database? |
| 21:08.10 | *** join/#asterisk JoseBravo (n=jbravo@190.9.74.174) |
| 21:08.23 | JoseBravo | How can I add a custome message for my voice mail? |
| 21:08.44 | [TK]D-Fender | JoseBravo, in Voicemailmain. |
| 21:09.14 | De_Mon | justdave func_odbc |
| 21:09.24 | De_Mon | fujin func_odbc |
| 21:09.34 | *** join/#asterisk frenzy (n=frenzy@unaffiliated/frenzy) |
| 21:09.45 | *** join/#asterisk gammah (n=gammah@70-253-197-131.ded.swbell.net) |
| 21:09.47 | De_Mon | fujin I dont know if thats the way to do it, but yes you can. |
| 21:10.06 | fujin | I'm currently doing a crontab like echo "select * from calleriddisplay" | mysql |
| 21:10.10 | fujin | and outputting that to a config file |
| 21:10.18 | fujin | and it'd be good to make it so that it querys the db directly |
| 21:10.29 | De_Mon | that is a really crappy method |
| 21:10.59 | fujin | apart from the fact that it doesn't continually place load on the DB, yeah. |
| 21:12.50 | *** join/#asterisk zydrunas_ (n=zydrunas@24-119-29-130.cpe.cableone.net) |
| 21:13.23 | *** join/#asterisk sangee (i=ravi@206.191.114.82) |
| 21:14.07 | sangee | anyone tried vxiasterisk (vxml for asterisk)? |
| 21:15.12 | *** join/#asterisk apardo (n=apardo@2001:5c0:9706:0:0:0:0:2) |
| 21:19.31 | mrdigital | whats vxiasterisk |
| 21:19.45 | De_Mon | i think it's (vxml for asterisk) |
| 21:20.05 | mrdigital | whats it do:? |
| 21:20.20 | De_Mon | its some sorta voice xml language |
| 21:20.25 | mrdigital | hmm |
| 21:20.45 | De_Mon | grammer definitions and such (guessing) |
| 21:21.24 | *** join/#asterisk jmesquita (n=jmesquit@200.162.229.225.user.ajato.com.br) |
| 21:23.54 | gammah | hey I was wondering if anyone has a pcap of a cisco sccp/skinny phone negotiating with asterisk? |
| 21:24.05 | Qwell[] | gammah: I do, actually |
| 21:24.08 | gammah | I'm trying to understand the protocol by reading chan_skinny.c -- would like to see the protocol in action |
| 21:24.09 | Qwell[] | oh, with asterisk...hmm |
| 21:24.12 | gammah | nah |
| 21:24.15 | Qwell[] | I have some with ccm |
| 21:24.16 | gammah | with anything really |
| 21:24.29 | Qwell[] | though, I suppose I could grab one with asterisk |
| 21:24.45 | Qwell[] | msg me your email address, I'll see if I can grab something really quick |
| 21:24.53 | gammah | wow ok cool |
| 21:25.18 | Qwell[] | I'll do a register, and a call in both directions |
| 21:25.27 | Qwell[] | and hold/unhold I guess |
| 21:25.40 | fujin | oh my god |
| 21:25.50 | fujin | this is super awesome |
| 21:25.58 | gammah | werd Qwell[] that rules thx |
| 21:26.01 | Qwell[] | fujin: yes, yes it is |
| 21:28.35 | fujin | aha! one macro down |
| 21:29.17 | [TK]D-Fender | AEL = total waste |
| 21:29.21 | fujin | ff |
| 21:29.35 | fujin | It's easily readable, the language makes more sense to *write* |
| 21:29.50 | mvanbaak | [TK]D-Fender: what's wrong with AEL |
| 21:29.53 | *** join/#asterisk fakhir (n=Fakhir@unaffiliated/fakhir) |
| 21:30.12 | *** join/#asterisk mercestes (n=Merceste@rrcs-71-41-157-70.sw.biz.rr.com) |
| 21:30.15 | mercestes | Comcast blows |
| 21:30.40 | [TK]D-Fender | Does nothing you can do in standard parser, adds a level of complexity to your dialpln, makes reading dumps a serious PITA, and has been in the past buggy as well. |
| 21:31.14 | fujin | complexity = win. |
| 21:31.55 | mvanbaak | I like it that you can easily add extra steps to an extension |
| 21:32.08 | mvanbaak | without having to restate the exten all the time |
| 21:32.27 | fujin | it looks awesome so far, just rebuilt my stdexten macro |
| 21:32.37 | fujin | going to do some databasa |
| 21:32.56 | fujin | is odbc the best way to do a simply query from a db? |
| 21:33.20 | mvanbaak | we use an agi for that |
| 21:33.41 | fujin | I need to rewrite my inbound-queue-prioritisation based on callerID |
| 21:33.48 | fujin | funtimes. |
| 21:34.49 | Dan0maN_Work | D-Fender: asking again about the 300 extension / 270 station setup, would dual-core's be overkill? i have no data for how many concurrent internal calls, but from what i understand, it shouldn't matter as they will connect peer to peer unless recording needs to take place. not planning on increasing our external lines, and we only have a T1 to our current pbx, so 24 max. |
| 21:35.39 | ReDNeQ | Dan0maN_Work we have 2 locations with 30+ extensions/users on both ends |
| 21:35.53 | ReDNeQ | at any given time we have only noticed memory use being high.. nothing on processor |
| 21:36.06 | ReDNeQ | for memory we are talking 500-750 megs |
| 21:36.12 | Dan0maN_Work | sorry about these stupid questions. i've been tasked to spec out and justify the budget items next year, and i haven't even been able to get a working test in place yet ;P |
| 21:36.21 | ReDNeQ | the machine is an amd 64 athlon HP machine |
| 21:36.27 | ReDNeQ | nothing fancy |
| 21:37.46 | *** part/#asterisk lirakis (n=etamme@65.200.191.253) |
| 21:37.51 | [TK]D-Fender | Dan0maN_Work, dual core is pretty common these days.... I don't thikn you'll need to go psycho on this really.. |
| 21:38.29 | Dan0maN_Work | ok. thanks to both of you |
| 21:39.05 | *** join/#asterisk cspot (i=cspot@ip68-1-63-100.pn.at.cox.net) |
| 21:41.04 | *** join/#asterisk klictel (n=klictel@atelka.info) |
| 21:41.22 | *** join/#asterisk explidous (n=explidou@rrcs-24-173-134-222.se.biz.rr.com) |
| 21:41.49 | *** join/#asterisk KpoH (n=AID@host-89-41-66-159.moldtelecom.md) |
| 21:42.28 | KpoH | hello all |
| 21:42.56 | KpoH | i have problem with asterisk compilation, I always get this error http://pastebin.org/340 |
| 21:43.10 | *** join/#asterisk galeras (n=galeras@200.31.204.42) |
| 21:43.56 | KpoH | please advice anyone, what the heck it want from me? |
| 21:44.08 | Strom_M | KpoH: try installing the ncurses dev library |
| 21:44.17 | KpoH | i'm on gentoo |
| 21:44.30 | Strom_M | so? |
| 21:44.48 | KpoH | so I already have header files |
| 21:44.52 | KpoH | of ncurses |
| 21:45.07 | Strom_M | *shrug* |
| 21:45.36 | KpoH | I mean i have installed ncurses |
| 21:46.53 | *** join/#asterisk h0 (n=Fakhir@unaffiliated/fakhir) |
| 21:47.01 | fujin | what the hell |
| 21:47.06 | fujin | I'm unable to connect to pastebin.org |
| 21:47.16 | fujin | down/down |
| 21:48.24 | Strom_M | ~pb |
| 21:48.24 | 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 |
| 21:49.49 | *** join/#asterisk DarkRift (i=dark@bas10-montreal02-1177582342.dsl.bell.ca) |
| 21:50.33 | fujin | oh, I know the other ones. I was just saying that I can't see KpoH's paste. |
| 21:50.52 | fujin | KpoH: can you paste it to another site? |
| 21:53.37 | KpoH | fujin: sec |
| 21:54.23 | KpoH | http://pastebin.ca/651633 |
| 21:54.41 | *** join/#asterisk Yourname` (n=chatzill@76-10-128-178.dsl.teksavvy.com) |
| 21:55.13 | KpoH | Strom_M: any other solutions? |
| 21:55.21 | Yourname` | Hello, I notice a few "call failed to go through, reason 0" and i see reasons from 0-9.. I was wondering if there is some documentation somewhere that mentions what these reasons mean |
| 21:56.00 | fujin | nasty |
| 21:56.05 | fujin | is that ~x86, or x86? |
| 21:56.09 | *** join/#asterisk [hC] (n=hardcore@mail.metrobridge.com) |
| 21:56.18 | Op3r | astbill or a2billing? |
| 21:56.24 | fujin | I always used the gentoo-voip overlay, the emakes were better |
| 21:57.57 | KpoH | fujin: it's without overlay, just pure sources form ftp.digium.com |
| 21:58.42 | KpoH | voip overlay update too slowly |
| 21:59.45 | KpoH | fujin: can you say something regard http://pastebin.ca/651633 ? |
| 22:00.11 | *** join/#asterisk KDan (n=KDan@87-194-122-30.bethere.co.uk) |
| 22:01.42 | *** join/#asterisk mindCrime (i=chatzill@nat/redhat/x-97444aa4871cdd44) |
| 22:02.48 | [TK]D-Fender | KpoH, ncurses devel <- go install |
| 22:03.09 | *** part/#asterisk TechBlazer (n=TechBlaz@70.88.27.130) |
| 22:03.59 | KpoH | [TK]D-Fender: I have ncurses installed on system |
| 22:04.15 | [TK]D-Fender | KpoH, DEVEL <- not just the main |
| 22:04.28 | *** join/#asterisk jtoy (n=jtoy@mail.backchannelmedia.com) |
| 22:05.41 | *** join/#asterisk eatmypiano (n=eatmypia@CPE00195b4be0d6-CM0019477f689a.cpe.net.cable.rogers.com) |
| 22:05.49 | KpoH | [TK]D-Fender: on gentoo every package is devel (with all header files and etc staff) |
| 22:06.31 | [TK]D-Fender | KpoH, go to asterisk.org and check the pre-req's again and your system as well |
| 22:07.50 | fujin | <3 to whever designed AEL |
| 22:08.07 | fujin | that's not an ncurses issue |
| 22:08.15 | fujin | KpoH: which package are you building |
| 22:08.52 | KpoH | fujin: not from package, just pickup tar.gz from asterisk.org and trying to compile it |
| 22:09.21 | *** join/#asterisk mindCrime_ (i=chatzill@nat/redhat/x-fcc29b1a50cac0c2) |
| 22:09.27 | fujin | that's what I mean |
| 22:09.29 | fujin | what version |
| 22:09.35 | fujin | 1.4.10? |
| 22:09.36 | KpoH | 1.4.10 |
| 22:09.41 | fujin | well, I built that same package yesterday |
| 22:09.45 | fujin | try re-downloading it |
| 22:09.58 | fujin | I assume you did ./configure && make ? |
| 22:10.21 | KpoH | sure ./configure && make menuselect && make |
| 22:11.31 | *** join/#asterisk b1shop (n=b1shop@dsl081-149-253.chi1.dsl.speakeasy.net) |
| 22:11.39 | fujin | well, that's strange |
| 22:11.46 | fujin | as I said, I built the same package yesterday |
| 22:11.56 | fujin | did you pass any options to ./configure ? |
| 22:12.27 | KpoH | yep, --with-gsm=internal --with-postgres --with-odbc |
| 22:12.35 | fujin | hrm |
| 22:12.40 | fujin | I didn't pass any to mine.. |
| 22:12.48 | fujin | sorry, I don't know. |
| 22:14.39 | salvatore2 | , |
| 22:14.55 | salvatore2 | what has changed since 1.4.8 ? |
| 22:15.01 | fujin | read the changelog |
| 22:15.06 | fujin | It's available on the website. |
| 22:15.14 | salvatore2 | i wanted to hear from you if there is a major bugfix |
| 22:15.18 | fujin | There is. |
| 22:15.22 | salvatore2 | that i might miss while reading |
| 22:15.23 | fujin | I advise you to read the changelog |
| 22:15.29 | salvatore2 | i already did |
| 22:15.31 | fujin | I'd hope not |
| 22:16.35 | *** join/#asterisk neverblue (n=neverblu@unaffiliated/neverblue) |
| 22:16.49 | Netgeeks | on which web site is the changelog posted? |
| 22:18.03 | salvatore2 | okay so i wonder one thing |
| 22:18.11 | salvatore2 | i am using a voip provider to terminate my calls over sip |
| 22:18.13 | KpoH | Netgeeks: on ftp with asterisk |
| 22:18.17 | salvatore2 | they accept my calls and connect the call |
| 22:18.23 | KpoH | ftp.digium.com |
| 22:18.25 | De_Mon | Netgeeks its in the asterisk package itself and at http://svn.digium.com/view/asterisk/ |
| 22:18.30 | salvatore2 | for the first 5 seconds, the call latency is unacceptable |
| 22:18.39 | salvatore2 | after 5 seconds, it becomes very good |
| 22:18.41 | salvatore2 | why is this |
| 22:18.48 | fujin | your voip provider sucks? |
| 22:18.49 | Netgeeks | ah, okay, so it's not available via http on a web site, but either in svn viewer or th e package itself... |
| 22:19.03 | fujin | there's a changelog in the package |
| 22:19.09 | salvatore2 | it sucks for sure but is this because of the reinvite thing? |
| 22:19.21 | KpoH | Netgeeks: http://ftp.digium.com/ |
| 22:19.25 | fujin | -rw-r--r-- 1 root root 462352 2007-08-08 09:01 ChangeLog |
| 22:19.27 | fujin | yup. |
| 22:19.40 | salvatore2 | is it possible to reduce the reinvite time? |
| 22:20.11 | KpoH | reinvite=no :) |
| 22:20.24 | salvatore2 | :) seriously |
| 22:20.40 | fujin | Modify the code |
| 22:20.48 | fujin | reinvite, when turned on |
| 22:20.53 | fujin | probably tries to bridge as soon as possible. |
| 22:21.29 | Yourname` | Hello, I notice a few "call failed to go through, reason 0" and i see reasons from 0-9.. I was wondering if there is some documentation somewhere that mentions what these reasons mean |
| 22:26.16 | *** join/#asterisk glacid (i=unknown@evool.com) |
| 22:27.06 | glacid | hello, i'm a little confused by musiconhold.conf - in some configuration examples [classes], default=> is defined, and in others it is [default] directory=, which is the current syntax for 1.4 |
| 22:27.23 | KpoH | fujin: hehe, what you think? I run ./configure without parameters, and it fails again |
| 22:29.03 | KpoH | with te same error |
| 22:30.07 | tsurko | Hello, can Realtime load data from LDAP directory for the sippeers/spiusers family? |
| 22:30.42 | *** join/#asterisk DrAk0 (n=ljd@nelug/coreteam/luisjose) |
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| 22:39.21 | glacid | <PROTECTED> |
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| 22:42.04 | Op3r | anybody knows any alternative to a2billing and astbill? |
| 22:43.07 | glacid | those are the two main ones |
| 22:43.17 | glacid | it's pretty much that or write your own |
| 22:43.52 | Op3r | astbill quite is sucky in terms of installing it on centos |
| 22:43.58 | Op3r | :( |
| 22:44.25 | Op3r | a2billing is kinda overkill for a gateway with billing setup |
| 22:44.27 | Op3r | :( |
| 22:45.09 | Op3r | plus the fact that astbill forum is inactive :( |
| 22:45.26 | *** join/#asterisk Curi (n=creinero@206.57.107.242) |
| 22:45.56 | Curi | hello, does any one know how to convert wav files to g729? |
| 22:46.15 | Op3r | is g729 an audio codec? |
| 22:46.27 | glacid | yes it is |
| 22:46.41 | Curi | yup |
| 22:46.46 | Op3r | oh god |
| 22:46.46 | glacid | best to just handle it on the fly |
| 22:47.12 | Curi | glacid: i don't want to stress the server if i can avoid it |
| 22:47.21 | glacid | there is a "free" g729 codec, or you can buy one from digium |
| 22:47.33 | glacid | also, they sell a hardware transcoder card that is blazing fast |
| 22:48.14 | *** join/#asterisk whist1 (n=whistler@71-81-67-70.dhcp.stls.mo.charter.com) |
| 22:48.15 | Curi | glacid: I know, but i thought that there might be just a script or program that could encode to g729 |
| 22:48.27 | Op3r | yeah and the hardware transcoder almost cost like an another box |
| 22:48.28 | Op3r | :( |
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| 22:49.18 | Curi | i know there's this GX::Transcoder but i can't make it transcode, it need some plugin i guess |
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| 22:50.03 | whist1 | hi, Im starting up a call center, not very large, about 20 to 40 employees, Im doing the basic IT, and my partner has lined up a normal PBX for us, but since we are paying for it, I was thinking about exploring some open source alternatives, can someone just let me know if it is worth it to look into asterisks as far as being able to handle the call volume, and reliablity? And what is a rough estimate for hardware cost before I go |
| 22:50.36 | Op3r | whist1: a lot of call centers here in the philippines use asterisk |
| 22:50.38 | whist1 | we will have 1 1-800 line over 2 T1s |
| 22:50.47 | Op3r | whist1: is it outbound or inbound? |
| 22:51.07 | whist1 | at first mainly inbound, than transitioning to both |
| 22:51.14 | glacid | with 40 employees, you could handle that with one beefy box and some digium hardware |
| 22:51.18 | *** join/#asterisk breanna_ (n=irssi@c-24-10-238-92.hsd1.ut.comcast.net) |
| 22:51.29 | glacid | or you could use several smaller machines |
| 22:51.49 | breanna_ | Can I use different gain settings for different PRIs in zapata.conf? |
| 22:52.07 | whist1 | whats considered beefy? and we are lookign at about a cost of 2800$ for the PBX that we have picked out, will I be able to beat that? |
| 22:52.38 | whist1 | Im assuming 4 gig ram, 3+ gig proc, scsi HD etc |
| 22:52.38 | glacid | probably, for your setup i bet you could install 3 asterisk machines for that price |
| 22:53.02 | Sweeper | whist1: including phones? |
| 22:53.08 | Sweeper | $2800, that is |
| 22:53.09 | glacid | HD not so important, unless you are recording stuff |
| 22:53.10 | whist1 | sweeper no |
| 22:53.13 | Sweeper | ah, ok |
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| 22:53.14 | *** mode/#asterisk [+o anthm] by ChanServ |
| 22:53.27 | Sweeper | yea, $1k is enough for a decent asterisk box these days |
| 22:53.39 | glacid | mostly you are going to need fast CPUs and some memory, 4 gigs of ram is overkill for asterisk |
| 22:53.42 | whist1 | recording would be nice, Im assuming asterisk has the ablitiy to remotely monitor phone calls? |
| 22:53.51 | glacid | yes it can |
| 22:53.59 | Op3r | you can also barge calls |
| 22:54.01 | glacid | you can monitor channels and also seize them too |
| 22:54.16 | Op3r | if you want you can put VICIDIAL in it |
| 22:54.48 | glacid | just keep in mind that you will need to manage your asterisk system as well, it's pretty much as complicated or as simple as you want to make it |
| 22:55.27 | whist1 | so, I can get away w/ 1k for this machine? thats pretty sweet. the other thing is, we can probably get some used phones cheap for a normal pbx, and I dont know fi the difference in price will make up for it. |
| 22:55.46 | fujin | anyone got any docs for querying a database from the dialplan? |
| 22:55.50 | fujin | in particular, mysql? |
| 22:56.03 | fujin | I can only seem to find information regarding realtime configuration, which isn't really what I want. |
| 22:56.07 | whist1 | glacid: Ill be there 24/7, and since I negotiated a big slice of the proits, it be nice to make myself irreplaceable...hehe |
| 22:56.08 | mrdigital | fujin: pm? |
| 22:56.13 | fujin | sure |
| 22:56.18 | whist1 | *profits |
| 22:56.45 | fujin | more money = more call capacity |
| 22:58.02 | whist1 | would someone be cool enough to give me a quick synopsis of what a 2 T1 w/ 1800 number setup would look like, with abotu 30 cubes, and pc/phone at each cube? |
| 22:58.16 | glacid | whist1: you can get away with like $40 per phone, probably less if you are buying in quantity - but i wouldn't use those cheap phones for a call center |
| 22:58.36 | glacid | ebay is your friend, you can get Cisco 7940s for anywhere between $75 and $125 a piece.. |
| 22:58.38 | fujin | whist1: that's very basic, but no |
| 22:58.50 | whist1 | k |
| 22:59.27 | glacid | also, consider that if you are running voip phones to each desk you will probably want PoE to power them, although it's not absolutely necessary |
| 22:59.56 | glacid | quick side note, if you do use cisco phones, if you try and power them from the external power brick - ear piece will produce a buzz or hum |
| 23:00.04 | glacid | they work best when running off PoE |
| 23:00.21 | glacid | sorry.. "headset" port buzzes - not the handset port |
| 23:00.37 | fujin | I wouldn't go with cisco handsets, if I were you |
| 23:00.42 | fujin | ~phones |
| 23:00.43 | jbot | i heard 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. |
| 23:00.54 | glacid | personally i love them, but polycoms are gaining popularity |
| 23:00.57 | fujin | yup |
| 23:01.04 | fujin | I went with Linksys spa942's here. |
| 23:01.08 | glacid | i wouldn't trade my cisco phones for anything |
| 23:01.17 | glacid | great call quality |
| 23:01.20 | whist1 | ah, Im going to have to go over some numbers, as I think that the costs of the phone will be kill any savings that we will have going with a traditional system and used equipment |
| 23:01.22 | Corydon76-work | I wouldn't use Cisco phones for anything |
| 23:01.43 | Corydon76-work | Not even for a doorstop |
| 23:02.04 | glacid | some of the higher end cisco phones are a PAIN to configure.. like the 7970/7971 |
| 23:02.15 | Op3r | or you can just get a pap2 and put analog phones in it |
| 23:02.16 | Nugget | Apparently Cisco phones make good props when you're shooting office interiors for a television show. |
| 23:02.18 | glacid | but the 7940 has been awesome to me |
| 23:02.34 | Nugget | They're all over the White House and Studio 60 on the Sunset Strip :) |
| 23:02.39 | glacid | nugget: i think they pay for that product placement |
| 23:02.47 | Nugget | of course |
| 23:02.48 | glacid | nugget: they probably donated the whole setup |
| 23:02.54 | mvanbaak | Corydon76-work: the cisco phones work great with the new chan_skinny.c :) |
| 23:02.58 | mvanbaak | but I'm biased |
| 23:03.14 | glacid | there's also chan_sccp2 - but needs a patch to work |
| 23:03.22 | glacid | i mean with the lateste 1.4.9 |
| 23:03.27 | fujin | lol |
| 23:03.31 | fujin | was quite funny, in 24 |
| 23:03.35 | mvanbaak | and chan_sccp2 is dead |
| 23:03.37 | fujin | they were running around trying to find a working phone |
| 23:03.44 | fujin | and Jack picks up a grandstream -> dead |
| 23:03.46 | fujin | a mitel -> dead |
| 23:03.55 | mvanbaak | ~gs |
| 23:03.56 | jbot | GrandSuck phones & gateways are cheap junk which should be avoided with extreme prejudice. |
| 23:03.58 | *** part/#asterisk b1shop (n=b1shop@dsl081-149-253.chi1.dsl.speakeasy.net) |
| 23:04.09 | glacid | i picked up a GS phone for the garage phone, and i'm sorry i did |
| 23:04.32 | Corydon76-work | The only complaint I have with my GS102 is that the speakerphone isn't loud enough |
| 23:04.42 | glacid | there's no local loop, so when you speak you can't hear yourself in the earpiece |
| 23:05.01 | ManxPower | glacid: your statement does NOT make sense |
| 23:05.10 | Corydon76-work | Uh, I'm sorry, why would you want to hear yourself speak? |
| 23:05.24 | ManxPower | Perhaps the term you are looking for is "sidetone" |
| 23:05.25 | glacid | on any normal handset you can hear yourself in the earpiece |
| 23:05.32 | glacid | manxpower: sorry, wrong term then |
| 23:05.37 | ManxPower | Yes, that is the term you are looking for. |
| 23:05.48 | Corydon76-work | Uh, no, that's ECHO, and that's very undesireable |
| 23:05.58 | glacid | no it's not echo |
| 23:06.07 | glacid | i'm not talking about feedback from the pbx |
| 23:06.13 | Corydon76-work | If I can hear myself in the earpiece, that's echo |
| 23:06.36 | *** part/#asterisk Curi (n=creinero@206.57.107.242) |
| 23:06.40 | glacid | if you can hear yourself with some delay, that's echo |
| 23:06.51 | glacid | ManxPower understands what i'm talking about |
| 23:06.56 | Corydon76-work | Uh, either way it's echo. |
| 23:07.01 | mvanbaak | if I can hear myself it's a miracle |
| 23:07.08 | mvanbaak | ;) |
| 23:07.14 | Corydon76-work | One's just a shorter loop than the other |
| 23:08.01 | glacid | anyways, on standard analog lines, on any "normal" phone, it's what we've all grown to expect - but on grandstream it's missing |
| 23:08.20 | ManxPower | glacid: technically sidetone on analog lines IS echo, it's just echo with so little delay it is not annoying |
| 23:08.34 | ManxPower | glacid: IP phones add their own sidetone. |
| 23:08.42 | mvanbaak | the grandstreams lack more then the sidetone ;) |
| 23:08.49 | glacid | yeah that's for sure |
| 23:08.54 | ManxPower | If your GS phone is not doing that, I would consider it either a design flaw or a broken phone |
| 23:08.55 | glacid | like a real display on thier lowest line model |
| 23:09.02 | glacid | it's a design flaw |
| 23:09.14 | glacid | other people have complained about it besides me |
| 23:10.01 | Corydon76-work | Yeah, it's horrible when you can actually hear the other person on the line |
| 23:10.11 | glacid | anyways, most of the rest of my phones are cisco, 1 aastra, and a cordless through an iaxy, and the GS is by far the worst of them |
| 23:10.22 | Corydon76-work | What are phones for, if not for misinterpreting what people say? |
| 23:10.53 | bkruse | is there a way to NOT log an action: originate? |
| 23:11.07 | bkruse | eg so the GUI does not fill the CDR with executecommand requests? |
| 23:11.12 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 23:11.12 | Corydon76-work | bkruse: comment out all lines in logger.conf |
| 23:11.46 | Corydon76-work | bkruse: oh, you might want to try running NoCDR in the dialplan then |
| 23:11.49 | bkruse | Corydon76-work: ha, we could do that......but we have CDR viewing support now, but most of the calls are executecommand calls, and I was wondering if we could get rid of them somehow |
| 23:12.00 | bkruse | Corydon76-work: I tried it, and that gets rid of the call CDR, but not the originate :[ |
| 23:12.15 | Corydon76-work | Sweet |
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| 23:16.49 | breanna_ | Can I use different gain settings for different groups in zapata.conf? |
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| 23:19.25 | salvatore2 | i love asterisk |
| 23:19.33 | ber123 | me 2 |
| 23:19.43 | salvatore2 | but, to be honest, it ruined my life |
| 23:19.56 | salvatore2 | all my life is consisting of phones now |
| 23:20.11 | mvanbaak | salvatore2: it's known to do that |
| 23:20.16 | bkruse | I love it |
| 23:20.23 | mvanbaak | it's a free feature you get for free with it |
| 23:20.30 | klictel | asterisk is a woman |
| 23:20.39 | salvatore2 | i want to disable this feature mvanbaak |
| 23:20.46 | salvatore2 | maybe i should just comment it |
| 23:20.48 | mvanbaak | salvatore2: halt -p |
| 23:20.48 | salvatore2 | ; |
| 23:20.59 | salvatore2 | :) |
| 23:21.32 | salvatore2 | i want to meet with a female asterisk developer |
| 23:21.38 | Sweeper | XD |
| 23:22.34 | salvatore2 | now i am trying to integrate call phones with asterisk |
| 23:22.41 | salvatore2 | over gsm standard |
| 23:22.49 | salvatore2 | cell phones* |
| 23:29.40 | stubert | quit |
| 23:31.10 | mvanbaak | salvatore2: you want asterisk to be part of the gsm network ? you need a gsm gateway or a gsm zaptel card |
| 23:31.22 | fujin | Anyone know how I can do something like Macro(sql(select something from clidcheck where phonenumber=${CALLERID(num)}))? |
| 23:31.25 | mvanbaak | 2n is offering a gsm <=> sip gateway |
| 23:31.44 | mvanbaak | and junghanns is offering pci cards with multiple gsm modems that have a zaptel driver |
| 23:31.47 | mvanbaak | :) |
| 23:32.34 | ber123 | asterisk isnt a woman |
| 23:32.36 | ber123 | cuz its consistant |
| 23:33.05 | mvanbaak | you sure ? |
| 23:34.15 | *** join/#asterisk sharp (n=sharp@pool-72-92-12-134.phlapa.east.verizon.net) |
| 23:35.18 | *** part/#asterisk galeras (n=galeras@200.31.204.42) |
| 23:35.49 | fujin | anyone? MySQL inside a dialplan? |
| 23:36.03 | mvanbaak | fujin: you can use the mysql dialplan function |
| 23:36.29 | fujin | oh christ |
| 23:36.32 | fujin | how did I not know about that |
| 23:37.26 | mvanbaak | one cannot know everything |
| 23:38.18 | JT | odbc would be more advisable |
| 23:38.38 | fujin | yeah? |
| 23:38.39 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 23:38.48 | fujin | shame I can't find any concrete stuff on it |
| 23:38.57 | JT | yes then you wouldn't be tied to one database |
| 23:39.01 | JT | err okay... |
| 23:39.27 | fujin | i *want* to be tied to one database, so that's not really an issue |
| 23:39.30 | jebba | what's the best way to stream radio to asterisk now? I've done the mpg123 0.59r stuff in the past, but it sounds really jittery now with * 1.4.9 |
| 23:39.32 | fujin | is that the only reason I'd use it over mysql()? |
| 23:39.44 | mvanbaak | fujin: yes |
| 23:39.48 | JT | why would you want to be tied to a database? that's silly |
| 23:40.08 | mvanbaak | JT: maybe they are talking to their inhouse application ? |
| 23:40.09 | JT | that and they're really only supporting odbc in asterisk going forward, from what i've heard |
| 23:40.28 | ber123 | you are joking jt? |
| 23:40.28 | JT | mvanbaak: even so, times change, database indepenence is a Goot Thing |
| 23:40.31 | JT | Good |
| 23:40.38 | JT | why would i be joking |
| 23:40.43 | mvanbaak | JT: I hope not. a lot of my stuff is dependant on cdr_psql and res_config_psql |
| 23:40.49 | *** join/#asterisk jkimball4 (n=jerrid@pc006629.mbsc.unomaha.edu) |
| 23:40.52 | ber123 | i have all kinds of database driven applications |
| 23:40.55 | ber123 | its absolutely crucial |
| 23:40.56 | Nugget | Anything that eliminates mysql lock-in is a huge win. :) |
| 23:41.04 | JT | Nugget: indeed |
| 23:41.20 | Nugget | mysql is the windows me of databases. Being able to avoid it is great. |
| 23:41.21 | JT | but database independence is always a good thing to have |
| 23:41.36 | Nugget | indeed |
| 23:41.44 | mvanbaak | odbc adds an extra layer |
| 23:41.52 | JT | oh gno! |
| 23:42.05 | mvanbaak | it does |
| 23:42.11 | JT | it's irrelevant |
| 23:42.16 | ber123 | yes db independence is good |
| 23:42.17 | mvanbaak | I'd rather talk to psql directly |
| 23:42.22 | JT | it's good to have an abstraction layer |
| 23:42.23 | Nugget | there's nothing inherently good or bad about adding a layer. |
| 23:42.28 | JT | exactly |
| 23:42.31 | JT | just rheotric |
| 23:42.43 | JT | rhetoric |
| 23:42.43 | mvanbaak | with odbc I have to install like 4 extra packages, do configuration of those packages and stuff |
| 23:42.48 | JT | terrible |
| 23:42.52 | ber123 | yum :) |
| 23:42.54 | mvanbaak | while with res_config_psql I only have to alter 1 file |
| 23:43.10 | JT | so the reason we should not use odbc is because you are lazy? |
| 23:43.35 | mvanbaak | the reason _I_ am not using odbc is because I'm lazy yeah |
| 23:43.43 | JT | right |
| 23:44.05 | mvanbaak | but that's what I like bout asterisk and most apps on my system |
| 23:44.07 | JT | don't get me wrong, people are lazy, but it's better to be lazy in the best ways |
| 23:44.10 | Nugget | having native database drivers in asterisk is a poor use of Digium's time and energy. |
| 23:44.13 | mvanbaak | you have a choice |
| 23:44.26 | ManxPower | seems like a lot of extra work for being lazy |
| 23:44.27 | mvanbaak | you can go either with native support or go with the odbc layer |
| 23:44.34 | JT | like "imagine how much effort it will be to chnge databases later, screw that! i think i'll use abstraction" |
| 23:45.03 | ber123 | yeah a lot of times i start off a project quick and dirty with mysql and hten move to oracle once it gets traction |
| 23:45.13 | fujin | eh. We (as an ISP) have no immediate or future plans to change our systems-wide databases |
| 23:45.15 | ber123 | if you are doing small stuff all the time it probably doesnt buy you that much |
| 23:45.19 | fujin | therefore, I will use mysql(). |
| 23:45.30 | JT | fujin: yay, a win for illogicity |
| 23:45.35 | fujin | care factor 0 |
| 23:45.35 | Nugget | starting with postgresql instead of mysql will make a future move to oracle a lot simpler. |
| 23:45.39 | fujin | I don't make the decisions :) |
| 23:45.45 | mvanbaak | ber123: gheh, you are talking about a waste of time and yet you are using oracle ;) |
| 23:45.53 | Nugget | they're closer in form and function |
| 23:45.55 | JT | you should take into account the fact that digium only want to support odbc |
| 23:46.00 | ber123 | oracle is good |
| 23:46.03 | Nugget | yes it is. |
| 23:46.06 | JT | that should rate on the lazy factor, i mean care factor |
| 23:46.09 | vutamhoan | I have mp3 quality problem with hardware line (soft-phone sound good) - Can anybody help |
| 23:46.11 | ber123 | oracle for a newbie is bad |
| 23:46.14 | *** join/#asterisk ZaVoid (n=zavoid@c-67-165-25-195.hsd1.ct.comcast.net) |
| 23:46.16 | Nugget | that's true too. |
| 23:46.22 | ber123 | mysql is nice because it doesnt require you to think that much to be ok from the start |
| 23:46.31 | ber123 | oracle you have to do some legwork up front, or at least i do |
| 23:46.32 | Nugget | I'm just saying that the move from pgsql to oracle is a lot smaller and easier than the move from mysql to oracle. |
| 23:46.35 | ManxPower | vutamhoan: you will always have quality problems with music over telephony |
| 23:46.46 | Nugget | and pgsql doesn't require any more thinking than mysql. |
| 23:46.48 | ber123 | oh yeah |
| 23:47.05 | JT | the reason there's so much morons writing garbage sql code is mysql |
| 23:47.11 | mvanbaak | digium is not only supporting odbc |
| 23:47.11 | JT | zomg LAMP |
| 23:47.18 | mvanbaak | they do support sqlite and pgsql as well |
| 23:47.23 | vutamhoan | ManxPower: Yes, mp3 is low volume and quality is bad |
| 23:47.26 | ber123 | man market took a beating today |
| 23:47.32 | ManxPower | vutamhoan: what codec? |
| 23:47.37 | *** join/#asterisk forrestv (n=forrestv@c-76-110-237-46.hsd1.fl.comcast.net) |
| 23:47.51 | vutamhoan | I use mp3 file for my IVR |
| 23:48.03 | vutamhoan | but gsm is perfect |
| 23:48.23 | fujin | anyone actually using MYSQL() for anythign? |
| 23:48.26 | ManxPower | vutamhoan: convert them to 8Khz mono |
| 23:48.28 | Netgeeks | does the odbc stuff in asterisk allow you to do direct sql queries? |
| 23:48.53 | mvanbaak | Netgeeks: it's odbc |
| 23:48.53 | vutamhoan | Ah, thank you |
| 23:49.12 | vutamhoan | I'm going to do it right now, thanks a lot |
| 23:49.45 | mvanbaak | fujin: not me. like I said before we are using agi scripts for stuff like that |
| 23:51.03 | Netgeeks | I understand it's odbc, but what is the asterisk dialplan interface to it? ODBCget? ODBCput? |
| 23:51.41 | Netgeeks | nvm, I think google found something for me to look at |
| 23:51.57 | jebba | vutamhoan, how are you doing mp3 playing? I'm trying to get reasonable sound from a streaming radio station. I've done it in the past oki, but now it's not cooperating. I have 0.59r installed (and tried other versions too). |
| 23:52.37 | jebba | i also see that there's format_mp3 in asterisk-addons, but there's code in main asterisk 1.4.10 with mp3 as well, so i don't know why addons would be necessary. It /is/ playing, it just sounds like stuttery crap ;) |
| 23:53.11 | Netgeeks | hrm the odbc tools available in the dialplan seem crippled as compared to app_addon_mysql |
| 23:56.21 | fujin | what does 'hint' do? |
| 23:56.30 | Netgeeks | fujin: yes, I use MYSQL() alot |
| 23:56.36 | fujin | ah, sweet |
| 23:56.39 | fujin | have you used it inside AEL? |
| 23:56.43 | Netgeeks | nope |
| 23:56.46 | *** join/#asterisk jesselang|laptop (n=jesse@h75-100-164-249.75-100.unk.tds.net) |
| 23:56.49 | fujin | right, thanks |
| 23:56.52 | Netgeeks | heh |
| 23:57.05 | jesselang|laptop | Hello all. |
| 23:57.14 | Netgeeks | I'm not an ael convert yet |
| 23:57.23 | fujin | I'm working on it. half way converted my extensions.conf. |
| 23:57.27 | jesselang|laptop | I'm trying to get the call duration from a AGI... can anyone help me? |
| 23:57.30 | fujin | Looks good so far. Beautiful to read. |
| 23:57.39 | mvanbaak | yup |
| 23:57.53 | mvanbaak | I like |
| 23:57.59 | mvanbaak | I like AEL |
| 23:58.00 | Netgeeks | no doubt, I tried converting but each time I found there were things I could do in old dialplan that I couldn't in ael, so I got annoyed and abandoned the attempt |
| 23:58.06 | jesselang|laptop | I tried "GET VARIABLE CDR(duration)", but that returned 0 everytime. |
| 23:58.40 | Netgeeks | CDR(duration) is only viable after the channel has hunug up |
| 23:58.46 | Netgeeks | hung up even |
| 23:58.55 | mvanbaak | brb, have to get out of my fatboy to get a powersupply for this laptop |
| 23:59.20 | fujin | anyone for some quick regex/awk/sed? |
| 23:59.26 | fujin | I need to get exten=>661,1,Macro(stdexten,661,SIP/661) -> just the extension number |
| 23:59.30 | fujin | '661'. |
