| 00:02.44 | jks | anyone know of an API for the jabra pc suite? |
| 00:03.00 | jks | (looking to get the soft buttons working with asterisk) |
| 00:03.13 | BenB | hey... I'd like to install asterisk for me at home. I'm a sw developer myself, but I don't really care to learn much about it (no time), and I have no idea about the concepts used... is there some easily installed GUI (including webbased) which allows me to click my configuration together? I use ubuntu. I tried asteriskNow and Freepbx/trixbox, but neither of them really guide me with the installation (admin password, "dial plan", |
| 00:03.13 | BenB | SIP registration etc.pp.) |
| 00:03.45 | BenB | is there a nice GUI which is packaged for ubuntu which allows me to use asterisk without learning for hours? |
| 00:03.54 | *** join/#asterisk arpu (n=arpu@chello062178159144.10.14.univie.teleweb.at) |
| 00:03.57 | jks | considered hiring a consultant? ;-) |
| 00:04.10 | BenB | jks: "at home" |
| 00:04.40 | Chainsaw | You hire electricians at home, don't you? |
| 00:05.19 | BenB | Chainsaw: actually, no, I don't. I didn't ask for consultants, I asked for a GUI. |
| 00:05.22 | Chainsaw | You want Asterisk but you don't want to do the work. Either you want a GUI tool like AsteriskNow, or a consultant so you can just say "make it so". |
| 00:05.43 | BenB | Chainsaw: I want a GUI, but more GUI than AsteriskNOW |
| 00:06.17 | BenB | for starters, I can't even log in to the astNOw GUI, because nothing told me the admin password. |
| 00:06.31 | BenB | that's the kind of thing I expect to be guided through. |
| 00:06.41 | Chainsaw | Try trixbox in that case. |
| 00:06.48 | BenB | Chainsaw: I did, same thing. |
| 00:06.51 | Chainsaw | (It even used to be called Asterisk@home) |
| 00:07.12 | Chainsaw | They sell ready-made boxes if you want. |
| 00:07.54 | BenB | I don't want. I want a free software GUI which does not require me to get a telephone technician expert |
| 00:08.15 | Chainsaw | They exist, I just named two. |
| 00:08.36 | BenB | Chainsaw: and I told one problem I have with them. |
| 00:08.48 | Chainsaw | Which can be solved by reading the provided documentation. |
| 00:09.09 | BenB | Chainsaw: which fails my requirement: not needing to read manuals. |
| 00:09.31 | BenB | (BTW, I can't find the documentation either) |
| 00:09.38 | Chainsaw | Then you win BenB. Asterisk isn't suitable for you in any way, shape or form. |
| 00:10.17 | BenB | too bad. it seems to be a nice software. I'd like to make secure phone calls between two of my sites, without having to spend a week or day on it. |
| 00:10.29 | BenB | A proper GUI would really do good. |
| 00:10.42 | Chainsaw | If you write one, there's probably money in it. |
| 00:10.45 | ivanvujisic | BenB: There is app Visual Dialplan for *, google is your friend |
| 00:12.10 | BenB | wonders why Firefox trunk Linux crashes, and remembers that he enabled Flash for Asterisk OpPanel |
| 00:12.34 | lanning | BenB: http://www.digium.com/en/products/appliance/ |
| 00:13.04 | BenB | lanning: costs more than $50? |
| 00:13.19 | lanning | the labor comes from somewhere... |
| 00:13.35 | lanning | free = some assembly required |
| 00:13.46 | BenB | yup, from people like me who write free software, for fun and free |
| 00:14.20 | srf21c | BenB: have you tried PBX in a flash? |
| 00:15.38 | srf21c | BenB: Also, if you want to pony up some money, the switchvox product is super easy to use. |
| 00:15.58 | BenB | srf21c: looking at pbxinaflash.net |
| 00:16.03 | srf21c | BenB: Nerdvittles.com is a great resource for newbs getting an asterisk box going. |
| 00:16.29 | srf21c | Tons of articles there. although I'm afraid it might be tough to get asterisk going without doing a little reading along the way. |
| 00:19.10 | BenB | srf21c: I'm most scared about all the concepts I don't know (and don't want to learn), like "dial plan", "trunk" etc. |
| 00:21.09 | BenB | all I know is that I got a few SIP accounts with phone numbers, a few SIP hardware phones, and want 2 asterisks at 2 sites of mine, connected via VPN. |
| 00:21.34 | lanning | unfortunately, you will have to learn how to design a "dial plan" |
| 00:21.37 | BenB | should be as easy as point and click, as far as I'm concerned :) |
| 00:22.33 | BenB | is there maybe some other free software which is better suited for my needs? |
| 00:23.04 | lanning | no matter what you get, you will still need to design a dial plan. |
| 00:23.18 | *** join/#asterisk JueLopi (n=retrewtr@modemcable174.78-57-74.mc.videotron.ca) |
| 00:23.35 | JueLopi | I don't know if someone have an answer but I would like to know if there's a way to have a sort of gateway that will have a sip soft phone as an input and connect to a iax2 termination? |
| 00:24.00 | srf21c | BenB: Hard for me to say, I'm more a command line type of guy. Haven't messed with the GUI versions other than Switchvox, which will set you back $800 or more. |
| 00:24.22 | BenB | ok, thanks all for your help |
| 00:24.30 | srf21c | BenB: anytime. |
| 00:24.31 | lanning | you can download switchvox free edition |
| 00:24.32 | srf21c | good luck. |
| 00:24.38 | BenB | I'll check out the stuff you pointed me at |
| 00:24.52 | srf21c | lanning: really? cool, wasn't aware of the free version. |
| 00:25.20 | srf21c | JueLopi: I am running a setup that is probably very similar to what you describe. |
| 00:25.26 | lanning | http://www.digium.com/en/products/switchvox/free-trial.php |
| 00:25.53 | lanning | trial is not time limited, just feature/size limited. |
| 00:25.58 | srf21c | I have IP SIP hard phones, connecting to a colocation server running asterisk, which connects to my service provider via IAX. |
| 00:26.53 | JueLopi | srf21c: What kind of software are you using? |
| 00:27.21 | srf21c | JueLopi: OpenBSD 4.4 w/Asterisk 1.4.21 |
| 00:27.32 | srf21c | and Snom IP phones. |
| 00:27.37 | srf21c | which run linux. |
| 00:27.48 | JueLopi | srf21c: Asterisk can do that? |
| 00:28.10 | *** join/#asterisk arpu (n=arpu@chello062178159144.10.14.univie.teleweb.at) |
| 00:28.12 | srf21c | asterisk can connect many different communication technologies together, yes. |
| 00:29.15 | Dovid | on sending a Fax asterisk does not re-invite T.38. on incoming when I get a fax Asterisk gets the re-invite and it works. anything that I need to do so that asterisk will request t.38 ? |
| 00:29.17 | JueLopi | srf21c: Great! I'm going to install it to see what it is. Look a pretty big piece of software |
| 00:29.48 | srf21c | JueLopi: yes, it's extensive. |
| 00:30.27 | JueLopi | srf21c: How hard is it to configure? |
| 00:30.48 | *** join/#asterisk MaliutaLap (n=biteme@203.171.195.204) |
| 00:30.59 | *** join/#asterisk beave (n=beave@DCC.SEND.startkeylogger.000.telephreak.org) |
| 00:32.22 | *** join/#asterisk layne (n=layne@ool-44c0048f.dyn.optonline.net) |
| 00:37.05 | srf21c | JueLopi: It can be challenging for a new user. |
| 00:37.23 | srf21c | I would recommend trying out trixbox or PBX in a flash for a new user. |
| 00:37.37 | srf21c | Unless you are comfortable at the command line and have sys admin experience. |
| 00:37.59 | ivanvujisic | JueLopi: http://www.voip-info.org/wiki/view/Asterisk+config+iax.conf |
| 00:38.10 | ivanvujisic | not too hard |
| 00:38.59 | JueLopi | srf21c: I'm confortable with linux. The only problem is all the terminology |
| 00:39.11 | *** join/#asterisk bbryant (n=brett@c-68-59-21-152.hsd1.sc.comcast.net) |
| 00:39.12 | srf21c | JueLopi: I guess what I meant to say is that setting up a real working dialplan can be challenging. the iax.conf and sip.conf files are relatively easy. |
| 00:39.45 | ivanvujisic | JueLopi: trixbox is to hard if you want to learn * |
| 00:39.59 | srf21c | Download the free PDFMake sure to download the free O'Reilly book http://www.google.com/url?sa=t&source=web&ct=res&cd=1&url=http%3A%2F%2Fdownloads.oreilly.com%2Fbooks%2F9780596510480.pdf&ei=xOk2SvSzB4yoswOTzKjZBA&usg=AFQjCNHH-gNcCSeBwE6DwUNG76JrV64qaw&sig2=uxvuo1kIDQb4wNpx6d4Onw |
| 00:40.18 | srf21c | oops. Try this: downloads.oreilly.com/books/9780596510480.pdf |
| 00:40.29 | srf21c | that's will be a good starting point. |
| 00:41.28 | ivanvujisic | no, starting point is /etc/asterisk/extensions.conf and sip.conf |
| 00:41.56 | ivanvujisic | there is description for meny dalplans |
| 00:43.03 | JueLopi | Thanks for the link! I'll do a quick check of trixbox and asterisk |
| 00:44.34 | Dovid | on sending a Fax asterisk does not re-invite T.38. on incoming when I get a fax Asterisk gets the re-invite and it works. anything that I need to do so that asterisk will request t.38 ? |
| 00:52.43 | srf21c | Dovid: No experience with T.38 here, sorry. |
| 00:54.28 | Dovid | ok. thanks |
| 00:57.24 | srf21c | I'm having the wierd problem whereby when I set the Dial timeout of an incoming call to 25 seconds or more, the call will bail with a == Spawn extension (inbound-context, <number>, 2) exited non-zero on 'IAX2/trunk-15380' -- Hungup 'IAX2/teliaxtrunk-15380' |
| 00:57.36 | srf21c | instead of going to voicemail like it should. |
| 00:57.47 | srf21c | Then it will ring another 25 seconds, and *then* go to voicemail. |
| 00:58.04 | srf21c | If I set the dial timeout to 24 seconds, the problem goes away. |
| 00:58.16 | srf21c | What the eff is going on? |
| 00:59.06 | *** join/#asterisk defsdoor (n=andy@defsdoor.gotadsl.co.uk) |
| 01:00.13 | srf21c | dialplan snippet http://pastebin.ca/1461712 |
| 01:00.47 | srf21c | SIP busy signal sends the call immediately to voicemail as it should. |
| 01:03.09 | drmessano | Am I missing something about using allow lines with & in them? |
| 01:03.19 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-9fc9926e96718235) |
| 01:03.26 | drmessano | allow=ulaw&alaw&speex&zomgwtf |
| 01:05.50 | JueLopi | srf21c, ivanvujisic: Thanks for your help! Now going to read a bit |
| 01:06.58 | *** join/#asterisk ccoenen (n=ccoenen@p5DCF3E13.dip.t-dialin.net) |
| 01:07.37 | ccoenen | I have a peculiar problem with my asterisk installation |
| 01:07.40 | *** join/#asterisk arpu (n=arpu@chello062178159144.10.14.univie.teleweb.at) |
| 01:07.49 | ccoenen | maybe anyone of you has an idea... |
| 01:07.59 | ccoenen | i can't make internal calls. |
| 01:08.31 | ccoenen | they always go outside via ISDN and then fail as number incomplete |
| 01:08.46 | *** join/#asterisk DrAk0 (n=luisjose@nelug/coreteam/luisjose) |
| 01:09.26 | *** join/#asterisk Deeewayne (n=dwayne@76.29.245.9) |
| 01:09.26 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 01:09.28 | ccoenen | my problem is, that i am still fairly new to asterisk, and i don't even really know where i should start |
| 01:10.35 | srf21c | ccesario_: shoot. |
| 01:11.13 | srf21c | ccoenen: start with the most basic dialplan possible.... two internal extensions. |
| 01:11.18 | srf21c | and then build from there. |
| 01:11.59 | srf21c | Check out the Asterisk TFOT Book PDF download. downloads.oreilly.com/books/9780596510480.pdf |
| 01:12.18 | ccoenen | thanks, i'll have a look |
| 01:12.20 | srf21c | make sure to back up your existing dialplan of course. |
| 01:12.51 | ccoenen | that dialplan has been generated by FreePbx/Trixbox |
| 01:13.16 | ccoenen | but the web-interface is not really helping in determining the problem, either |
| 01:13.29 | srf21c | also check out http://saunderslog.com/2006/07/08/2604/ |
| 01:14.17 | srf21c | ccoenen: yes, web interface can hide a lot of under the hood action that makes it bewildering to troubleshoot when it doesn't work. |
| 01:14.41 | srf21c | also bare bones voip example: http://www.onlamp.com/pub/a/onlamp/2003/07/03/asterisk.html |
| 01:14.58 | srf21c | A bit dated, but the basic concepts are the same . |
| 01:15.28 | *** join/#asterisk Octothorpe (n=Octothor@pdpc/supporter/professional/octothorpe) |
| 01:15.41 | ccoenen | ah the systm video! That's really good to get started |
| 01:16.13 | srf21c | not bad, John Todd does a pretty good job of explaining things. |
| 01:16.22 | srf21c | at least he speaks slow a measured. ;) |
| 01:17.42 | BenB | srf21c: the oreilly-book you posted is good, thank! |
| 01:18.06 | srf21c | BenB: cool, glad it's helping out. |
| 01:18.33 | srf21c | pretty awesome of them to make it available for free in PDF, IMHO. |
| 01:19.57 | *** part/#asterisk ruben23 (n=AGENT@124.107.3.178) |
| 01:20.52 | BenB | srf21c: indeed. |
| 01:21.09 | BenB | srf21c: I'm trying out switchvox atm, but again am stuck at the admin password. |
| 01:21.44 | ccoenen | hmm from what i read, if all my SIP phones have the context from-internal, then this should take precedence over dialing out via the landline? |
| 01:21.51 | BenB | the console menu (which comes up after install) has an "Set Admin Password", I set one, I go to the webbased admin tool, enter "admin" as username and the password, and it rejects me. |
| 01:23.38 | *** join/#asterisk Iamnach0 (i=Iamnacho@ip174-70-137-120.ks.ks.cox.net) |
| 01:23.50 | srf21c | BenB: hmm. that's what the docs say. |
| 01:24.09 | srf21c | admin/admin is the default user/pass. |
| 01:24.13 | srf21c | did you enter something else? |
| 01:24.28 | KyleK | hey if i was going to go from 1.4 to 1.6 with a very simple sip only setup, what version should I grab and wrestle with? 1.6.1.1, 1.6.2.0-beta3 or svn? |
| 01:24.35 | srf21c | ccesario_: well, hard to say withou seeing your dialplan. |
| 01:24.45 | BenB | srf21c: I did what I said above |
| 01:24.47 | srf21c | Can you post to pastebin.ca |
| 01:24.52 | BenB | the console menu (which comes up after install) has an "Set Admin Password", I set one, I go to the webbased admin tool, enter "admin" as username and the password, and it rejects me. |
| 01:25.28 | BenB | I just rebooted, in case asterisk doesn't notice the password change, but same problem. admin/admin doesn't work either |
| 01:25.33 | srf21c | BenB: The way it reads to me, you set a password...but not necessarily a password with the characters "admin" |
| 01:25.52 | srf21c | did you set "admin" as the password when you were given the choice? |
| 01:25.58 | BenB | srf21c: no, of couse I don't use "admin" as password |
| 01:26.10 | srf21c | BenB: well try that and see what happens. |
| 01:26.12 | BenB | do I look like a moron? :) |
| 01:26.17 | srf21c | Just to keep things simple. |
| 01:26.26 | srf21c | well no, just trying to eliminate possibilities. |
| 01:27.26 | BenB | FWIW, I tried to log in with user:"admin", password: "admin" now, and it doesn't work either. |
| 01:27.37 | BenB | (I refuse to *set* to password to "admin", though) |
| 01:28.06 | BenB | username: "admin", password: (what I entered at the console menu) doesn't work either |
| 01:29.41 | BenB | it's forcing https, with a broken certificate, BTW. it all looks fairly buggy and rough to me. |
| 01:30.15 | BenB | (the other distros were even worse, not even offering to set the admin password) |
| 01:31.49 | rob0 | I'm not a moron, but I play one on TV. |
| 01:32.30 | blaxthos | question about time conditions |
| 01:33.02 | blaxthos | anyone know if it's possible to give a specific CID inbound two different days of the week allowed for inbound, otherwise just ring forever ? |
| 01:33.07 | blaxthos | i can do it with one time condition |
| 01:33.11 | *** join/#asterisk imcdona (n=imcdona@c-24-19-203-112.hsd1.wa.comcast.net) |
| 01:33.20 | blaxthos | but i need to specify two non-consecutive day/times to allow in |
| 01:33.23 | blaxthos | otherwise not |
| 01:33.55 | srf21c | ~CID |
| 01:33.55 | infobot | somebody said cid was CallerID, or a TCP client/server Caller-ID system, including server and Tk GUI client.. URL: http://www.tummy.com/cid |
| 01:34.09 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 01:34.56 | srf21c | BenB: try setting the password to admin just this once, in order to trouble shoot. |
| 01:35.03 | *** join/#asterisk Kumbang (n=unknown@125.163.83.153) |
| 01:35.41 | srf21c | blaxthos: don't know, although that'd be an interesting application. |
| 01:35.57 | riddlebox | is there any command to type in the cli, to see which stations are in use at that time? |
| 01:36.00 | BenB | srf21c: what is this supposed to gain? if I set the pw to "abc" and can't log in with username "admin", pw "abc", I won't be able to set to "admin" and log in with pw "admin" |
| 01:37.21 | *** join/#asterisk missinglink (n=missingl@ppp166-229.static.internode.on.net) |
| 01:40.51 | srf21c | riddlebox: have you tried "sip show peers" |
| 01:40.59 | srf21c | not sure if that show real time use. |
| 01:41.22 | *** join/#asterisk andresmujica (n=andresmu@ubuntu/member/andresmujica) |
| 01:41.27 | srf21c | BenB: Just trying to see if somethin might be broken with password setting. |
| 01:41.40 | srf21c | when it doubt, use all defaults, is my take. |
| 01:41.45 | *** join/#asterisk jicksta (n=jicksta@67.164.0.78) |
| 01:41.51 | *** part/#asterisk andresmujica (n=andresmu@ubuntu/member/andresmujica) |
| 01:42.56 | riddlebox | srf21c, yeah, but it doesnt show if the peer is in use |
| 01:43.02 | srf21c | riddlebox: aye. |
| 01:44.43 | BenB | srf21c: I have a guess: it only allows authentication from localhost, or something. |
| 01:44.58 | srf21c | BenB: that might be it, yes. |
| 01:45.09 | BenB | and comes back with a wrong error message "Username or password incorrect" |
| 01:45.19 | srf21c | I'm just taking shots in the dark, since I've never used any of these producks. |
| 01:45.33 | blaxthos | lnk ? |
| 01:45.43 | BenB | srf21c: that'd be a misconfiguration, though, as I'll hardly run the webbrowser on the asterisk server console. |
| 01:46.06 | BenB | srf21c: ah, ok, you seemed so cheery about switchvox that I thought you know it. |
| 01:46.13 | blaxthos | my trixbox is a headless VM in my vista x64 desktop |
| 01:46.14 | blaxthos | works great |
| 01:46.29 | blaxthos | except for that timecode routing problem :( |
| 01:47.41 | blaxthos | oh snap |
| 01:47.50 | blaxthos | you can put more than one time period |
| 01:47.52 | blaxthos | that's awesome |
| 01:56.56 | KyleK | I guess the imap voicemail stuff is written with the assumption that the imap server is local? |
| 02:02.49 | leifmadsen | KyleK: I think you can specify it... |
| 02:07.37 | e0n` | man this sucks |
| 02:08.13 | e0n` | So polycom phone I got, used of course, got it all factory reset passwords and all, had an extension of 223, i have cleared out everything I could find on this thing and it's still passing the digest of 223 |
| 02:09.37 | Talkradio | maybe the builtin dial plan on the phone |
| 02:09.42 | e0n` | hmm |
| 02:10.07 | e0n` | it is the digest authentication i know that much |
| 02:10.23 | *** join/#asterisk andresmujica (n=andresmu@ubuntu/member/andresmujica) |
| 02:11.07 | *** join/#asterisk Octothorpe (n=Octothor@pdpc/supporter/professional/octothorpe) |
| 02:12.00 | seanbright | i hate that guy. |
| 02:12.32 | *** part/#asterisk andresmujica (n=andresmu@ubuntu/member/andresmujica) |
| 02:12.50 | e0n` | ? |
| 02:15.27 | rob0 | Sounds like it's mutual! Or, an inside joke. |
| 02:16.32 | *** part/#asterisk phr3ak (n=nnnnphr3@gnet.hu) |
| 02:16.55 | *** part/#asterisk srf21c (n=seth@ip98-165-60-42.ph.ph.cox.net) |
| 02:21.07 | *** join/#asterisk lanning (n=lanning@173.8.187.197) |
| 02:21.28 | ccoenen | aww man |
| 02:21.34 | *** part/#asterisk lanning (n=lanning@173.8.187.197) |
| 02:21.34 | ccoenen | i found the error |
| 02:22.31 | *** join/#asterisk maxagaz (n=maxagaz@soho2.i-xanadu.com) |
| 02:24.10 | *** join/#asterisk blkry (n=blkry@97.95.233.232) |
| 02:45.32 | e0n` | Anyone have a polycom soundstation ip 4000 |
| 02:45.50 | e0n` | or similar |
| 02:45.55 | e0n` | ip 5000/6000 |
| 02:50.33 | *** join/#asterisk moy (n=moy@CPE001cdfec4cee-CM00080dab8485.cpe.net.cable.rogers.com) |
| 02:51.21 | blkry | 6000 |
| 02:52.08 | blkry | e0n': 6000 |
| 02:56.35 | *** join/#asterisk kamanashisroy (n=kamanash@202.56.7.214) |
| 03:00.51 | *** join/#asterisk gunter (n=user@87.127.97.39) |
| 03:04.57 | *** part/#asterisk ccoenen (n=ccoenen@p5DCF3E13.dip.t-dialin.net) |
| 03:07.48 | *** join/#asterisk sah-work (n=Bawbatos@adsl-75-63-18-30.dsl.pltn13.sbcglobal.net) |
| 03:12.58 | *** join/#asterisk mintos (n=mvaliyav@nat/redhat-in/x-4dc855b36748d4b3) |
| 03:20.06 | *** join/#asterisk mindCrime (n=chatzill@cpe-076-182-103-085.nc.res.rr.com) |
| 03:31.48 | *** join/#asterisk BeeBuu (n=beebuu@218.13.68.163) |
| 03:35.04 | BeeBuu | "originate zap/128 application playback demo-instruct" got nothing,why? |
| 03:39.10 | [TK]D-Fender | BeeBuu: Because. |
| 03:40.01 | BeeBuu | what? |
| 03:40.05 | BeeBuu | cause what? |
| 03:40.16 | BeeBuu | [TK]D-Fender: please |
| 03:41.00 | [TK]D-Fender | BeeBuu: You're showing us nothing. |
| 03:41.11 | [TK]D-Fender | BeeBuu: Show us the attempt at CLI with full core and verbsoe debug, show us your zaptel configs. |
| 03:41.12 | BeeBuu | and the channel 128 is meeting, and i run that command under CLI |
| 03:41.28 | [TK]D-Fender | BeeBuu: "meeting"?! |
| 03:41.50 | BeeBuu | yes, in a meet room |
| 03:42.07 | [TK]D-Fender | BeeBuu: I don't follow you... |
| 03:43.07 | [TK]D-Fender | BeeBuu: it CALLS that device, and if it answers THEN it will do the rest of what you tell it |
| 03:43.23 | [TK]D-Fender | BeeBuu: Go provide the complete debug for your attempt |
| 03:43.45 | BeeBuu | i want to play some sound to a member of a meet room |
| 03:43.55 | BeeBuu | please wait |
| 03:43.58 | BeeBuu | ~paste |
| 03:43.58 | infobot | paste is, like, http://rafb.net/paste/, or see also pb, or http://bin.cakephp.org/ |
| 03:50.55 | BeeBuu | is this problem? "chan_zap.c: Failed to read gains: Invalid argument" ? |
| 03:52.14 | [TK]D-Fender | BeeBuu: Maybe something |
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| 04:10.19 | KyleK | hrm i finally got chan_mobile to dial but didn't get audio |
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| 05:05.15 | [TK]D-Fender | checkout time, later all |
| 05:05.49 | e0n` | hmm |
| 05:06.06 | e0n` | Ok a Polycom how can I flush the SIP config completely off the phone and clear this thing 100% no config to factory preset |
| 05:06.16 | e0n` | i did the factory reset but the SIP options were still there |
| 05:07.17 | ricko73 | format the phone |
| 05:07.42 | e0n` | ricko73: hmm, time for google |
| 05:07.43 | e0n` | lol |
| 05:07.59 | ricko73 | yeah , look for the e4voip link |
| 05:08.08 | ricko73 | if you type in 'how to format polyom' it's one of the top links |
| 05:08.14 | ricko73 | http://www.8774e4voip.com/blog/2008/03/how-to-format-polycom-ip-phone.html |
| 05:08.19 | ricko73 | #lazyweb |
| 05:08.25 | e0n` | got it |
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| 05:11.41 | AlmightyOatmeal | say i want someone to hit an extension, but i want it to ring to multiple phones and/or users... how would i go about doing that? |
| 05:12.15 | ricko73 | Dial(SIP/101&SIP/102&ZAP/4&IAX/201) |
| 05:12.21 | e0n` | ricko73 beat me to it |
| 05:12.24 | Jenna | Any can suggest which path to take in setting up heavy duty VoIP service. i.e. asterisk + kamelio ooorrr asterisk + Openser ? |
| 05:13.21 | AlmightyOatmeal | ricko73: and the first person to answer gets the call? |
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| 05:13.29 | AlmightyOatmeal | thats hot! :D |
| 05:13.44 | AlmightyOatmeal | thanks :D |
| 05:13.47 | e0n` | AlmightyOatmeal: yeah, it's extremely simple :) |
| 05:13.49 | e0n` | Makes life easy |
| 05:13.56 | AlmightyOatmeal | lol it sure does |
| 05:16.19 | e0n` | heh |
| 05:16.44 | e0n` | all my new polycom desk phones work like a champ my old cisco ip 7960's just don't work for what they're worth |
| 05:17.05 | kaldemar | Jenna: openser doesn't really exist anymore. it was forked into kamailio and opensips. they're probably not so different (yet) by implementation, you better get to know each project and decide yourself. |
| 05:18.14 | AlmightyOatmeal | heh, i think i @#%$&'d my cisco 7911G ip phone |
| 05:18.26 | Jenna | yeah that much I did gather. I just wanted to know about ur experiences etc.. i.e. which one is stabler, scalable, easy of management etc.. |
| 05:18.32 | AlmightyOatmeal | can't find a clear way to get it working plus i reset it to defaults and i dont think its pulling a new firmware |
| 05:19.01 | AlmightyOatmeal | e0n`: you wouldn't happen to be able to get cisco ip phone firmware's off ciscos site would you? my user account doesn't have the privvies to |
| 05:23.22 | AlmightyOatmeal | wants to find a cheap voip phone or two :'( |
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| 05:37.20 | b14ck | hi all |
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| 07:25.26 | BeeBuu | i had ran "testv=1" command before running asterisk, but ${ENV(testv)} got nothing~~~~,why? |
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| 07:45.34 | redax | hi, |
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| 08:12.52 | WeazelON | can anyone explain to me how do i check why my asterisk is crashing every few minutes ? |
| 08:14.32 | joobie | check ur log files |
| 08:14.38 | joobie | bound to have something in there |
| 08:14.59 | WeazelON | i'm trying the /var/log/asterisk/full not sure what exactly i need to look for |
| 08:15.08 | WeazelON | i mean how do i know how to find the crash point in the logs |
| 08:15.28 | joobie | go into /var/log/asterisk |
| 08:15.37 | joobie | and do.. tail -f * & |
| 08:15.43 | joobie | then get it to crash |
| 08:15.46 | joobie | and watch the pretty colors |
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| 08:17.25 | WeazelON | thanks |
| 08:17.56 | WeazelON | it shows me all the logs at the same tail? |
| 08:18.45 | joobie | ya |
| 08:18.49 | joobie | then press enter a few times |
| 08:18.54 | joobie | and maek asterisk crash |
| 08:19.19 | WeazelON | how do i stop it btw ? |
| 08:20.25 | joobie | type fg |
| 08:20.28 | joobie | then press ctrl-c |
| 08:20.41 | WeazelON | awesome thanks |
| 08:20.55 | joobie | no worries |
| 08:21.17 | WeazelON | is there a specific line that is echoed right when asterisk crashes ? |
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| 08:24.18 | joobie | i dont know |
| 08:24.21 | joobie | make it crash |
| 08:24.27 | joobie | then see what logs spit out as it crashes |
| 08:24.32 | joobie | it will give you info as to what is going on |
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| 09:01.36 | genin | mornin folks |
| 09:03.49 | genin | anyone know how may simulatneous calls are recommended when using adsl? |
| 09:03.55 | genin | using g729 |
| 09:06.02 | genin | ouch i just found an article called ADSL - the voip killer :/ |
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| 09:51.00 | Router222 | hi all |
| 09:51.30 | Router222 | kindly ihave a serious bug with asterisk fax ,they said that i should refer to digium support |
| 09:51.55 | Router222 | any one nows there address , |
| 09:52.00 | Router222 | mail address |
| 09:52.23 | AlmightyOatmeal | check their website |
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| 09:56.36 | Router222 | i check is prepaidsupport |
| 09:59.07 | Router222 | it's |
| 10:08.07 | yang | Router222: you can explain the bug here, someone might help you |
| 10:12.18 | Router222 | yang https://issues.asterisk.org/view.php?id=15328 |
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| 10:14.08 | Router222 | russell said Please contact Digium technical support. |
| 10:14.15 | Router222 | is it free ? |
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| 10:44.57 | gr0mit | genin, depends on your adsl znd the codec |
| 10:45.59 | genin | g729 |
| 10:46.20 | genin | but i just read it isnt actually 8kb for a 729 call but can be closer to 39kb with the rtp overhead etc |
| 10:47.07 | genin | and when i test the adsl connect i see near 500 kb up BUT when i force 200kb up on my network and then use an online voip quality test |
| 10:47.11 | genin | it says there is too much jitter |
| 10:47.30 | genin | mayeb i should use dd-wrt and a linksys to prioritize the traffic for my gateway |
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| 10:52.29 | razu | morning ... can tell me what is causing such alarm on SS7 link using libss7 ? >>> WARNING[7932]: chan_dahdi.c:9897 ss7_linkset: GRS on unconfigured CIC 1 |
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| 11:12.44 | WeazelON | does anyone familier with this error in 1.4 ? --- > rc_avpair_new: unknown attribute 1490026597 |
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| 12:03.42 | madduck | is there a command to show details about existing calls bridged through the asterisk? |
| 12:07.52 | madduck | core show channelstats and core show channel |
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| 12:28.28 | plundra | Hmm, what would an Answer(); Dial(...); do different then just an Dial(...)? (Dial is supposed to Answer() on the call who initiates the call, when the other channel picks up, right?) |
| 12:29.00 | plundra | Because in some situations, I get no audio through what so ever, without doing an Answer() first. |
| 12:30.19 | plundra | Both with: [SIP w/outgoing proxy + NAT] -> [IAX2 behind NAT], and [SIP w/outgoing proxy + NAT] -> [SIP w/outgoing proxy + NAT] |
| 12:31.31 | *** join/#asterisk guax (n=guaxinim@unaffiliated/guaxinim) |
| 12:35.21 | [TK]D-Fender | plundra: Perhaps its changing the reinvite status as * answering the call should force it to initially exchange RTP with the calling device first as opposed to only negociating it upon remote answer. |
| 12:35.49 | [TK]D-Fender | plundra: then again NAT setting may be incorrect anywhay and be an issue here |
| 12:36.08 | plundra | [TK]D-Fender: I tried changing the global canreinvite=no in sip.conf, with the same results. |
| 12:36.57 | [TK]D-Fender | plundra: Describe *'s networking, and that of your remote client |
| 12:37.13 | plundra | Calling out via my provider is no problem at all, but then there is no NAT involved of course. (No Answer needed before I Dial out) |
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| 12:38.43 | plundra | [TK]D-Fender: In all cases there is an OpenBSD-box nating the clients. And with the SIP-clients, I've got siproxd running, which both clients use. |
| 12:39.09 | [TK]D-Fender | plundra: You should not need a SIP proxy for remote devices. |
| 12:39.56 | plundra | [TK]D-Fender: Ever since I've used it, I have had no problems calling in etc. so it feels like they are helping :) |
| 12:40.20 | plundra | And it makes sense that the Via header is added. |
| 12:40.37 | [TK]D-Fender | plundra: Well things aren't always working so you should be removing the unnecessary bits |
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| 12:41.46 | plundra | As soon as I remove the outgoing proxies, things break, so surely it can't be unnecessary? |
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| 12:42.13 | plundra | And I have no "sometimes"-behaviour currently, which is good :-P |
| 12:42.31 | plundra | (I got through to the remote clients behind nat SOMETIMES, when not using the proxy) |
| 12:43.51 | [TK]D-Fender | plundra: Maybe it breaks because you didn't set the rest of it up right |
| 12:44.42 | plundra | [TK]D-Fender: Yes, that is very possible :-) |
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| 13:01.26 | mcargile | One of my clients installed the g729 codec on their server but the build does not look correct for their processor. They have 2 quad core 2.6 Ghz Xeons (family 6 model 23) and it had them install the pentium3m_32 build. |
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| 13:15.23 | Katty | :> |
| 13:17.03 | [TK]D-Fender | Katty: Mew. |
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| 13:18.47 | JayTee52 | mornin Katty *hugs* |
| 13:19.14 | fbnts | hi, in my dialplan I am calling an AGI script but when called the CLI is outputting: ERROR[8712]: utils.c:966 ast_carefulwrite: write() returned error: Broken pipe |
| 13:19.21 | JayTee52 | darn, I left myself logged in at home |
| 13:19.30 | Katty | hugs [TK]D-Fender and JayTee52 |
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| 13:20.00 | Katty | i can't believe this morning. i had an apt scheduled that i was dreading. absolutely dreading. so all last night was dread, and all this morning was dread. |
| 13:20.10 | Katty | got in this morning, they canceled. |
| 13:21.01 | Katty | also my favorite eyeshadow brush is missing. |
| 13:21.07 | Katty | :< |
| 13:21.10 | Katty | blames pippin. |
| 13:21.54 | fbnts | from googling it suggests that my script is cutting the connection before asterisk was expecting it to, however I have checked the script and it all appears ok. Any ideas? |
| 13:24.38 | Katty | it's too early for asterisk stuff. |
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| 13:26.14 | [TK]D-Fender | fbnts: SHOW US. |
| 13:26.57 | lucidsmog | What seems wrong with this line, in Asterisk 1.6.0.9. I want to prepend a '1' onto every incoming number which doesn't already have it (as my SIP inbound provider doesn't put one): Set(CALLERID(num)=${IF(${LEN(${CALLERID(num)})} == 11 ? ${CALLERID(num)}: 1${CALLERID(num)})}) |
| 13:27.18 | eppigy | hi |
| 13:27.29 | [TK]D-Fender | eppigy: YOU ARE DAVE |
| 13:27.30 | Katty | uses eppigy's shoulder as a pillow. |
| 13:28.55 | [TK]D-Fender | lucidsmog: extra whitespace = bad, "==" is not the "equals" operator, and you did not use the proper format for an "Asterisk Expression" inside your "IF" |
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| 13:31.07 | lucidsmog | [TK]D-Fender: Thank you for the information. I'll go read the documentation on Asterisk Expressions again. (is the semi-official place for this stuff really voip-info.org? I keep finding errors in what is there; I ought to correct a few) |
| 13:31.43 | [TK]D-Fender | lucidsmog: Variabel & expression basics are a constant, or you can refer to the CHANNELVARIABLES doc that came with your source tarball |
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| 13:33.36 | lucidsmog | [TK]D-Fender: So the 'IF' function ins't even necessary if you use the $[] syntax it seems? |
| 13:34.42 | [TK]D-Fender | lucidsmog: An expression evaluates math or does a boolean comparison. This is required, as well as a variable ACTION to take place based on that comparison, so yes, an IF is called for here. |
| 13:35.53 | fbnts | Fender: I have commented out the whole PHP script so it doesn't do anything but when I call I still get: [Jun 16 14:37:48] ERROR[8783]: utils.c:966 ast_carefulwrite: write() returned error: Broken pipe about 20 times |
| 13:36.14 | fbnts | do you want me to get the extension call plan or the PHP code? |
| 13:37.28 | Katty | [TK]D-Fender: too early for this. |
| 13:38.29 | [TK]D-Fender | fbnts: this is clearly a problem once you get to your AGI.... |
| 13:40.42 | lucidsmog | [TK]D-Fender: Seems to have worked. Thanks for your time dealing with a newbie question/mistake. |
| 13:41.58 | [TK]D-Fender | lucidsmog: Quite welcome, and glad you seem to have fixed it all in one pass. the read-over will do you good. |
| 13:42.09 | fbnts | Thanks Fender - It was the #! at the top of the script |
| 13:42.21 | fbnts | was calling PHP with an invalid argument |
| 13:45.45 | lucidsmog | [TK]D-Fender: I must admit I can find no file called CHANNELVARIABLES in my source tarball, or that string in any of the files in the tarball. Any other hints? |
| 13:46.59 | beek | lucidsmog: It's in the doc/tex subdirectory |
| 13:47.52 | beek | lucidsmog: It becomes part of tex/asterisk.pdf |
| 13:48.54 | lucidsmog | beek: Ahh, case sensitivity and POSIX ;) Thanks! |
| 13:49.24 | beek | lucidsmog: Definitely print the asterisk.pdf doc... it contains all of the goodies related to your version. |
| 13:49.28 | *** join/#asterisk TpK (n=net@213.190.214.69) |
| 13:50.04 | *** join/#asterisk pmhaddad-lappy (n=Phil@adsl-99-169-190-209.dsl.applwi.sbcglobal.net) |
| 13:51.23 | *** join/#asterisk jplank (n=GBove@cpe-075-181-097-208.carolina.res.rr.com) |
| 13:51.49 | lucidsmog | beek: Oh wow, this is quite helpful. This is the document I have been looking for! |
| 13:54.00 | *** part/#asterisk hesco (n=hesco@24.99.160.121) |
| 13:56.50 | eppigy | Katty: :> |
| 13:57.33 | eppigy | I am hungry |
| 13:57.39 | Katty | imagine that. |
| 13:57.41 | eppigy | I need to start eating breakfast |
| 13:58.49 | Katty | stash granola bars at work. |
| 13:59.01 | eppigy | yes that is also a very good option |
| 13:59.56 | *** join/#asterisk jplank (n=GBove@cpe-075-181-097-208.carolina.res.rr.com) |
| 14:02.08 | eppigy | i am poorly organized |
| 14:03.45 | *** join/#asterisk lirakis_ (n=lirakis@65.200.191.241) |
| 14:04.28 | Katty | you need a girlfriend. |
| 14:04.58 | Katty | to stash granola bars in your pockets before you leave for work. |
| 14:05.47 | *** join/#asterisk IBC_jkenney (n=jkenney@99.23.50.73) |
| 14:06.05 | lirakis_ | does asterisk support multiple choice redirect messages? I've set promiscredir on my peer and it follows the first choice in the redirect it recieves, but it seems to continue to the next priority if recieves a non "ANSWER" status |
| 14:06.29 | IBC_jkenney | using vicidial and getting this error utils.c:966 ast_carefulwrite: write() returned error: Broken pipe is anyone else getting it? |
| 14:07.48 | *** join/#asterisk putnopvut (n=putnopvu@asterisk/master-of-queues/mmichelson) |
| 14:07.48 | *** mode/#asterisk [+o putnopvut] by ChanServ |
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| 14:13.34 | *** join/#asterisk Whitor (n=Whitor@64.128.237.124) |
| 14:19.08 | eppigy | Katty: that is for certain |
| 14:19.26 | eppigy | or pack left overs for me to eat at 10am |
| 14:23.25 | *** join/#asterisk Deeewayne (n=dwayne@75.76.254.162) |
| 14:23.25 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
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| 14:23.31 | *** join/#asterisk csiadmin (n=csiadmin@host217-41-18-3.in-addr.btopenworld.com) |
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| 14:27.03 | *** join/#asterisk obscure1 (n=carnage@S0106001a92bcc9ca.ed.shawcable.net) |
| 14:27.48 | obscure1 | hey guys, any idea why i would keep getting a "handle_response_register: Failed to authenticate on REGISTER to..." message? |
| 14:30.01 | *** join/#asterisk KavanS (n=KavanS@static-71-117-242-28.ptldor.dsl-w.verizon.net) |
| 14:30.21 | *** join/#asterisk Failrar (n=Failrar@tunnel1088.ipv6.xs4all.nl) |
| 14:30.41 | obscure1 | nm, forgot to pay my bill lol |
| 14:34.25 | *** part/#asterisk obscure1 (n=carnage@S0106001a92bcc9ca.ed.shawcable.net) |
| 14:34.38 | *** part/#asterisk Vicky2 (n=chetan@61.17.196.153) |
| 14:40.46 | meingbg | ö |
| 14:40.52 | *** part/#asterisk meingbg (n=user@173-45-238-108.slicehost.net) |
| 14:40.58 | *** join/#asterisk Sajam (n=chatzill@77.42.201.226) |
| 14:41.37 | Katty | hmm |
| 14:41.58 | leifmadsen | Katty: YOU! |
| 14:42.02 | Katty | me? |
| 14:42.06 | leifmadsen | yep! |
| 14:42.16 | Katty | k |
| 14:42.21 | leifmadsen | hi :) |
| 14:42.26 | Katty | hi :> |
| 14:42.30 | Katty | hugs leifmadsen |
| 14:42.54 | leifmadsen | re-hugs Katty |
| 14:43.30 | *** part/#asterisk TpK (n=net@213.190.214.69) |
| 14:43.59 | *** join/#asterisk moy (n=moy@74.12.123.90) |
| 14:44.14 | Katty | eppigy: lunch? |
| 14:44.53 | *** join/#asterisk Defraz (n=T0tal@fw-poky.fuzecore.com) |
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| 14:46.11 | *** join/#asterisk michael-i (n=michael-@141.41.40.153) |
| 14:47.59 | michael-i | hi all, I'm running 1.4.25 on a custom embedded distro. My problem is that Asterisk does not give me back the console after I execute it. Has anyone seen that before? |
| 14:48.33 | michael-i | This is a VERY stripped down system (no syslogd, minimal users/groups, etc...) so anything is possible but perhaps someone has seen this. |
| 14:49.33 | *** join/#asterisk |Cybex| (n=John@80.100.126.176) |
| 14:50.48 | russellb | michael-i: how are you executing asterisk? |
| 14:50.51 | russellb | what command? |
| 14:51.16 | *** join/#asterisk jpeeler (n=jpeeler@asterisk/digium-software-dev/jpeeler) |
| 14:52.21 | michael-i | from my php script: /usr/sbin/asterisk |
| 14:53.55 | michael-i | I added a bunch of -v flags to see how far it was getting. It turns out that it does completely load but isn't able to give back the console for some reason |
| 14:54.15 | russellb | Well, the -v option will do that |
| 14:54.23 | russellb | make sure you do _not_ give v's |
| 14:55.31 | russellb | asterisk with no args should fork away |
| 14:55.31 | *** join/#asterisk xrmx__ (n=rm@host83-249-dynamic.8-87-r.retail.telecomitalia.it) |
| 14:55.31 | csiadmin | michael-i, is it started in the background and therefore won't run in the foreground? |
| 14:56.28 | michael-i | executing it with no args (as my script attempts to do) leaves me with a console waiting for it to finish execution. Running PS on another tty shows about a dozen asterisks running |
| 14:57.39 | michael-i | It is also entirely possible that my environment is completely insane...I'm setting this up from scratch. |
| 15:00.28 | csiadmin | michael-i, did you run 'make config' after you installed asterisk? |
| 15:00.35 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 15:00.44 | JayTee52 | wish this damn rain would let up |
| 15:01.15 | killfill | hey, all my agents are in state pause. |
| 15:01.18 | killfill | How to i unpause them? |
| 15:01.41 | michael-i | csiadmin: my scripts also generate a sane set of default configs (hopefully). Those scripts are pretty much platform independent and should still be working |
| 15:04.31 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 15:06.38 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 15:06.47 | csiadmin | michael-i, have you tried to run it with the -r option? |
| 15:07.23 | csiadmin | michael-i, or else edit logger.conf and set verbose to /var/log/asterisk/messages |
| 15:10.05 | michael-i | a ha... |
| 15:10.36 | michael-i | csiadmin: my asterisk database is not present and cannot be created because of a ro file system |
| 15:10.48 | michael-i | time to move that to a little more sensical location |
| 15:11.47 | csiadmin | michael-i, that'll do it |
| 15:12.11 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 15:12.53 | killfill | all my agents are in pause: Agent/6003 (paused) (Not in use) has taken no calls yet |
| 15:12.57 | killfill | how do i unpause them?... |
| 15:15.43 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 15:16.47 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
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| 15:20.04 | *** part/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 15:20.24 | *** join/#asterisk icyValk77 (n=icyValk7@host217-43-31-121.range217-43.btcentralplus.com) |
| 15:21.46 | michael-i | can I override just the database's location? astdbdir used to be an option but I don't see it in 1.4.25.1 |
| 15:24.43 | *** join/#asterisk mltlnx (n=mltlnx@asams.mserve.com) |
| 15:25.12 | batphone | dumb question: does not having /dev/dsp (i.e., no sound card) affect asterisk's functionality? |
| 15:25.25 | mltlnx | hello, How can I set the channel variable for the called channel? I am able to set the calling channels variables |
| 15:25.47 | mltlnx | batphone: nope |
| 15:26.24 | JayTee52 | michael-i, the default directory for the astdb is /var/lib/asterisk and is set in asterisk.conf using the astdatadir= line in the [general] section |
| 15:26.41 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 15:27.26 | batphone | mltlnx: i see lots of references to "DSP" in TFOT book, especially regarding transcoding |
| 15:27.47 | batphone | mltlnx: i take it that they are referring to some software running on the CPU instead of a physical DSP? |
| 15:28.21 | batphone | "Generally speaking, the more compression |
| 15:28.23 | batphone | that's required, the more workthe DSP must do to code or decode the signal." |
| 15:28.27 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 15:28.30 | batphone | ;/ |
| 15:28.50 | mltlnx | i believe digium makes a transcoding card |
| 15:28.54 | michael-i | JayTee52: thanks! I was looking for the wrong name |
| 15:29.46 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 15:30.34 | *** join/#asterisk jicksta (n=jicksta@67.164.0.78) |
| 15:31.20 | michael-i | JayTee52: actually it looks like the database under astvarlibdir (make_defaults.h:18) |
| 15:34.31 | *** join/#asterisk bijit (i=1000@190.241.15.48) |
| 15:35.21 | *** join/#asterisk InfoNutz (n=infonutz@204.50.209.225) |
| 15:35.48 | eppigy | Katty: LETS GO |
| 15:36.16 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 15:37.01 | *** join/#asterisk [jmc] (n=[jmc]@93-45-238-238.ip104.fastwebnet.it) |
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| 15:39.56 | *** join/#asterisk docelmo (n=chatzill@65.114.160.138) |
| 15:40.14 | docelmo | Anyone know if you can get DID's from Afghanistan? |
| 15:40.20 | docelmo | If so.. Where? |
| 15:42.26 | JayTee52 | www.talibantelecom.com? |
| 15:43.30 | docelmo | You are kidding right? |
| 15:43.56 | JayTee52 | um.....yes. |
| 15:44.08 | KavanS | lol |
| 15:45.49 | docelmo | sigh.. I hate dealing with the middle east.. They are a pain in the arse to get anything done |
| 15:48.55 | michael-i | hmmm....it's still not giving me back my console |
| 15:49.14 | michael-i | (...and at 5:50pm I'm really not caring anymore!) |
| 15:49.49 | *** join/#asterisk ingenius (n=alektro@186.136.6.218) |
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| 15:50.49 | *** join/#asterisk deadpigeon (n=deadpige@office.xpressamerica.net) |
| 15:51.27 | russellb | michael-i: I don't know what to tell you, unless the Asterisk build system determined that there was not a working fork() available |
| 15:51.32 | russellb | you can look in include/asterisk/autoconfig.h |
| 15:51.36 | russellb | for something like HAVE_WORKING_FORK |
| 15:51.59 | *** join/#asterisk rue_mohr (n=rue@24.207.122.10) |
| 15:52.31 | *** join/#asterisk trentcreek (n=kvirc@129.113.113.34) |
| 15:52.50 | rue_mohr | so the svn version of hte dahdi drivers was only allowing audio for outgoing calls, incomming calls (from co) had no audio at all either way |
| 15:53.06 | rue_mohr | thats why I dropped to the lastest stable dahdi drivers |
| 15:53.15 | *** join/#asterisk ReDNeQ (n=ReDNeQ@70.114.229.58) |
| 15:53.26 | rue_mohr | and the dahdi_monitor in the latest stable is 'broken' |
| 15:53.50 | michael-i | russellb, hmmm : http://pastebin.ca/1462286 |
| 15:55.45 | *** join/#asterisk ingenius (n=alektro@186.136.6.218) |
| 15:55.48 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 15:55.59 | russellb | michael-i: ah, uclibc probably? |
| 15:56.25 | michael-i | russellb: yessir...I'm looking for any flags which may not be set correctly |
| 15:57.09 | russellb | it looks like we will not even attempt to daemonize without HAVE_WORKING_FORK |
| 15:57.17 | russellb | and your output shows that as not defined |
| 15:57.20 | russellb | Sooooooo! |
| 15:57.25 | russellb | You're seeing expected behavior unfortunately |
| 15:57.56 | michael-i | well that would explain it :) now to top off my day, I'll go poke around here to see why that's being set |
| 15:58.02 | russellb | However, you can try hacking the source. |
| 15:58.37 | russellb | Edit main/asterisk.c ... find the line ... if (daemon(1, 0) < 0) { |
| 15:58.48 | russellb | a few lines above, you'll see #if HAVE_WORKING_FORK |
| 15:58.57 | russellb | change that to HAVE_WORKING_VFORK (since you have that) |
| 15:59.13 | russellb | It's sort of an odd check there, anyway, since we're using the daemon() system call there, not fork() or vfork() |
| 15:59.41 | michael-i | I'll try to do-it-right(tm) for now since this is a cross-compiling environment. Hacks will probably come back to haunt me! |
| 15:59.57 | russellb | Well, that may actually be the right thing, though ... |
| 16:00.08 | russellb | I'm pretty sure we have all this WORKING_FORK crap in there because of uclibc |
| 16:00.37 | michael-i | Thanks a ton for pointing me in the right direction. But my uclibc does have a working fork :( It's quite angry about the non-recognition |
| 16:00.42 | russellb | the question is, does HAVE_WORKING_FORK not being defined (even though HAVE_WORKING_VFORK is there) actually imply that daemon() isn't going to work |
| 16:00.50 | russellb | heh. |
| 16:00.57 | russellb | Well, it has vfork(), anyway |
| 16:01.22 | mltlnx | how can i set the channel variable parkinglot on the called channel? |
| 16:01.36 | michael-i | when the blackfin port of my project starts in a few weeks, I can see how cool that vfork() is :) |
| 16:01.55 | russellb | ah yes ... |
| 16:02.28 | russellb | michael-i: anyway, good luck to you. If you _really_ want to get dirty, see autoconf/ast_func_fork.m4 |
| 16:02.31 | russellb | that's where we're checking for it |
| 16:03.02 | michael-i | russellb: thanks! I'll take a peek there... |
| 16:03.02 | mltlnx | russellb: Do have a multi parkinglot example for the dialplan? |
| 16:03.32 | russellb | michael-i: just to make sure my memory isn't crazy, you do askozia, right? |
| 16:03.50 | russellb | mltlnx: i do not have an example. multiple parking lots are supported in 1.6, but that's the only info i have |
| 16:04.23 | *** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun) |
| 16:04.28 | mltlnx | ok thanks....I just cant get the the parkinglot chan variable noticed. |
| 16:05.53 | *** join/#asterisk iratik (n=itariki@209.248.216.146.nw.nuvox.net) |
| 16:06.12 | *** join/#asterisk marv[work] (n=timr@router.asteriasgi.com) |
| 16:06.13 | michael-i | russellb: yeah, that's me....I was trying to save face by not stating that here in asterisk-users while I ask questions ;) |
| 16:06.31 | iratik | Hi... Is there a way to trigger a reregister for a specific entry in sip show registry ? |
| 16:06.32 | *** join/#asterisk CunningPike (n=CunningP@204.239.10.119) |
| 16:06.42 | iratik | Besides reloading which would reregister everything in the registry |
| 16:06.46 | russellb | michael-i: lol, well i called you out! |
| 16:06.55 | russellb | michael-i: good luck, take it easy .. |
| 16:07.07 | michael-i | noooooooooo! |
| 16:07.19 | michael-i | will do :) I like being 7+hrs ahead, my day's done! |
| 16:10.01 | *** join/#asterisk ariel_ (i=3fd6eca9@gateway/web/ajax/mibbit.com/x-9cc69bd913458e25) |
| 16:11.28 | iratik | So ... at somepoint (june 09th) the sip.conf needed for magicjack changed .... anyone know what changes they made to how we need to register to magicjack talk4free servers ? |
| 16:12.16 | kaldemar | how about asking magicjack? |
| 16:12.30 | Katty | eppigy: i'm plotting BBQ for lunch |
| 16:13.20 | eppigy | DUDDE |
| 16:13.23 | eppigy | SO AM I |
| 16:13.26 | eppigy | HICKORY HOUSE |
| 16:13.32 | eppigy | that is uncanny |
| 16:14.20 | Katty | eppigy: i'm going to the Branding Iron |
| 16:14.37 | eppigy | :D |
| 16:14.56 | *** join/#asterisk DarKnesS_WolF (n=wolf@unaffiliated/sherif) |
| 16:15.08 | ariel_ | is just sitting at his desk eating the sandwich he made last night.... and a banana. |
| 16:15.33 | eppigy | overnight sammich? |
| 16:16.53 | ariel_ | I am poor and don't have enough cash to go out for lunch |
| 16:18.28 | *** join/#asterisk Firass-z0r (n=asadf@c-67-201-205-34.reshall.wwu.edu) |
| 16:19.19 | drfreeze | Is it possible to have two phones with the same extension, but on different networks, to be controlled by the same asterisk box? |
| 16:19.44 | InfoNutz | hello all!! anyone have an idea what would be the best way to setup the ivr to promt a user dialing zero that the service is unavailable? |
| 16:20.59 | [TK]D-Fender | drfreeze: You can have the same extension dial 2 DEVICES, but you cannot have 2 devices register to the same SIP account |
| 16:21.30 | [TK]D-Fender | InfoNutz: exten => 0,1,Playback(GTFO) |
| 16:21.40 | ariel_ | rofl |
| 16:21.49 | Katty | eppigy: time to goooooooooooooooooooooooo |
| 16:22.08 | *** join/#asterisk qdk (n=qdk@0x573d8dd1.bynqu1.dynamic.dsl.tele.dk) |
| 16:22.10 | InfoNutz | you for real? |
| 16:22.16 | michael-i | How can I check which "host" string was used to compile asterisk with? |
| 16:22.44 | eppigy | YES |
| 16:22.54 | Katty | eppigy: bbq sauce. |
| 16:23.01 | Katty | eppigy: pulled pork sammich yumminess |
| 16:23.11 | [TK]D-Fender | InfoNutz: Taht is perfectly valid... assuming you have a recording named "GTFO" with an appropriate extension in a format * can play |
| 16:23.16 | iratik | How is the "Contact: " part of the sip register packet configured from sip.conf ? |
| 16:23.18 | Katty | eppigy: do you have a bib? |
| 16:23.22 | batphone | hmm |
| 16:23.32 | batphone | newer kernels fail the RTC support test |
| 16:23.33 | InfoNutz | lol k cause the GTFO was throwing me off. Thanks Fender |
| 16:23.42 | batphone | RTC is now used as /dev/rtcN |
| 16:23.55 | batphone | and is not natively configured as CONFIG_RTC=yes anymore |
| 16:23.56 | eppigy | Katty: I do not but I didnt wear an expensive shirt today |
| 16:24.01 | [TK]D-Fender | InfoNutz: I'd like to think you know how the Playback() app works already.... |
| 16:24.03 | batphone | its CONFIG_RTC_BLAH depending on the type |
| 16:25.52 | *** join/#asterisk srf21c (n=seth@ip98-165-60-42.ph.ph.cox.net) |
| 16:26.56 | srf21c | anybody happen to know what the maximum level of verboseness is for the asterisk console? |
| 16:27.16 | srf21c | I've set mine to 22, but not sure if it makes much difference that say, 10. |
| 16:27.22 | [TK]D-Fender | srf21c: 4 IIRC, but I like 10. It feels "important" |
| 16:28.10 | [TK]D-Fender | BBIAB, lunch time... |
| 16:28.28 | srf21c | [TK]D-Fender: thx. You wouldn't happen to know where this particular limit is documented, would you? |
| 16:29.00 | drfreeze | I've been searching out wireless phones for use in an office setting. I have found the Polycom SpectraLink and the D-Link DPH-540 (looks like a cell phone) |
| 16:29.02 | batphone | -vvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvv is usually what i set mine to |
| 16:29.18 | drfreeze | Does anyone have a wireless voip phone they use and like? |
| 16:29.30 | batphone | drfreeze: we have some spectralinks that work well |
| 16:29.41 | drfreeze | I'm guessing the SpectraLink's will auto provision like the soundpoints |
| 16:29.44 | batphone | drfreeze: just be sure to order the one that does not require the stupid polycom server |
| 16:29.46 | JayTee52 | srf21c, if you set it to anything higher than 6 it usually doesn't change anything unless you go to 99 and from what I understand then it will show mysql debug info |
| 16:30.13 | drfreeze | batphone: I see a lot of different versions of spectralinks - I haven't dug in to find out what the differences are |
| 16:30.24 | drfreeze | They can get expensive, but go as low as $259 |
| 16:30.31 | srf21c | drfreeze: I had my eye on the Nokia 6300i, but they are impossible to find. |
| 16:31.04 | drfreeze | srf21c: hmmm |
| 16:31.08 | Chainsaw | drfreeze: Do they handle WPA2 PSK (CCMP)? |
| 16:31.23 | drfreeze | batphone: you know which ones require the server? |
| 16:31.26 | Chainsaw | drfreeze: That's one thing to look out for, when we wanted VoIP handsets it all fell down on WiFi security. |
| 16:31.29 | srf21c | batphone: re: mega v's....seriously? |
| 16:31.31 | ariel_ | drfreeze: we use the spectralink 8020 and 8030's and polycom just released a sip update that no longer needs there svp box |
| 16:31.46 | ariel_ | but there not fully autoprovision phone |
| 16:32.19 | drfreeze | Chainsaw: http://www.voipsupply.com/spectralink-8002-scb |
| 16:32.22 | srf21c | drfreeze: I missed the part about "an office setting" I was looking at Cell phones with wifi and sip clients for home office use. |
| 16:32.43 | Chainsaw | drfreeze: 802.11B |
| 16:33.06 | srf21c | JayTee52: interesting, how did you figure out that level 6 verbosity was the limit? |
| 16:33.08 | drfreeze | what does that imply? |
| 16:33.09 | Chainsaw | drfreeze: I'm concerned about support of even WEP, as it's not mentioned I wouldn't assume it does. |
| 16:33.23 | Chainsaw | drfreeze: 802.11B is the older 11mbit 2.4GHz WiFi standard. |
| 16:33.26 | *** join/#asterisk korihor (n=korihor@190.72.254.245) |
| 16:33.48 | Chainsaw | drfreeze: Likely WEP 128-bit was the highest available WiFi security measure at the time. |
| 16:34.03 | drfreeze | Chainsaw: what, this doesn't make you feel warm and fuzzy: Using the Wi-Fi Allianceâs WMM QoS standard, the SpectraLink 8002 Wireless Telephone interoperates with most consumer-grade and SMB access point infrastructure devices, alleviating the need to install and maintain additional hardware while still providing enterprise-level security and quality voice |
| 16:34.15 | srf21c | JayTee52: I'm having a hard time finding out where this documented. It's not in the man page for the OpenBSD port of asterisk. |
| 16:34.32 | srf21c | google didn't seem to turn up much either. |
| 16:34.34 | drfreeze | 'enterprise-level security'. You can now shut your brain off and open your checkbook. ;) |
| 16:34.46 | srf21c | drfreeze: lulz |
| 16:34.54 | JayTee52 | srf21c, it's not the "limit" it's just that there isn't much that shows up above that level. If I want full verbose I usually never go above 10. I've never seen anything "new" show up and I'm just passing on what I learned in class. |
| 16:35.07 | srf21c | ah, ok. |
| 16:36.14 | Chainsaw | drfreeze: But yes, this WI FI <whisper>802.11</whisper> problem is common. |
| 16:36.26 | Chainsaw | s/802.11/802.11b/ |
| 16:37.01 | drfreeze | ariel_: not seeing the 8020 or 8030, just the 8002 |
| 16:37.22 | ariel_ | the 8020 and 8030 are the more expensive ones |
| 16:37.31 | ariel_ | the 8030 even has push to talk |
| 16:37.35 | drfreeze | the k series for cisco? |
| 16:38.13 | drfreeze | gotta go eat |
| 16:39.55 | Chainsaw | Push to talk is nice, but 802.11b is still a dealbreaker. |
| 16:40.17 | ariel_ | drfreeze: http://www.voipsupply.com/spectralink-wtb151 |
| 16:40.23 | batphone | drfreeze: hang on a sec |
| 16:40.26 | Chainsaw | I'd much prefer 802.11a; the 2.4GHz spectrum is generally way too crowded where we are. |
| 16:40.47 | Chainsaw | Yes. Something like that. |
| 16:40.58 | *** join/#asterisk Maliuta (n=scooby@kiev.lusan.id.au) |
| 16:43.02 | rue_mohr | so if you who have been following might be amused to hear, the phone system here with all the lines for all the businesses has been having fun cause of their dial out on any available line, cause prople are using call display info to return calls, which means now all the incomming calls are all mixed up on all the lines |
| 16:43.16 | rue_mohr | which was simply a matter of fate and time |
| 16:43.20 | rue_mohr | gtg |
| 16:43.25 | Chainsaw | Shame about the "must be sold with expensive proprietary QoS box". |
| 16:44.06 | ariel_ | mixed up lines? |
| 16:44.33 | batphone | drfreeze: we are using the spectralink 8002 |
| 16:44.40 | batphone | drfreeze: works like a champ. roams and everything. |
| 16:44.46 | batphone | drfreeze: aes encryption |
| 16:45.10 | ariel_ | that is the best part of the spectralink's there able to move between access points without dropping your calls. |
| 16:45.29 | batphone | ariel_: as long as your APs are configured on different channels |
| 16:45.46 | *** join/#asterisk jeffik (n=jeffik@69-196-165-181.dsl.teksavvy.com) |
| 16:46.15 | ariel_ | hehe, we set them up for curise lines deck phones and they transfer between 5 to 10 access points. So yes really on how you set the access points up as. |
| 16:47.25 | *** join/#asterisk csiadmin (n=csiadmin@host217-41-18-3.in-addr.btopenworld.com) |
| 16:47.43 | csiadmin | hi all |
| 16:48.16 | batphone | so these TDM cards can get fried if you plug the wrong FX port into the telco? |
| 16:48.34 | ariel_ | batphone: it could happen |
| 16:48.39 | batphone | one would think there would be some hardware logic thrown at that ;[ |
| 16:48.47 | tzafrir_laptop | batphone, which TDM cards? |
| 16:48.56 | batphone | i have seen the copper lines curl up on them before |
| 16:49.00 | batphone | i was just hoping for some change |
| 16:49.03 | tzafrir_laptop | (if they are, they are sub-standard. by definition) |
| 16:49.05 | batphone | tdm400p |
| 16:49.31 | csiadmin | can anyone know how to do an OR conditional in a GoSubIf? |
| 16:49.34 | batphone | im using my old tdm card stocks to build some voip systems for my new employer |
| 16:49.46 | batphone | i built about 120 asterisk pbxs in Houston in 2005 |
| 16:49.51 | batphone | kinda fun now that i get to go back at it ;D |
| 16:50.00 | tzafrir_laptop | that said, it is pointless to plug an FXS port to the telco |
| 16:50.01 | batphone | rusty though so forgive me if i start asking dumb questions |
| 16:50.45 | batphone | anyone have luck getting asterisk to work in a production env. on a dell 1750? |
| 16:51.01 | batphone | i am having to purchase some 12v dc power dongles for the TDM cards |
| 16:51.28 | batphone | right now the dev box's TDM card is being powered by an external ATX PS. |
| 16:51.32 | batphone | funny looking ;) |
| 16:52.20 | ariel_ | the fxo don't need the power cable but the fxs do. |
| 16:54.33 | batphone | ariel_: i never could get zaptel to actually start unless the card had power, regardless of what was in it |
| 16:54.49 | batphone | i can see where fxs wouldnt need the power, but still |
| 16:58.38 | *** join/#asterisk jaguiar (n=jaguiar@201.114.42.25) |
| 17:02.57 | *** part/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
| 17:03.02 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
| 17:06.59 | *** join/#asterisk abatista (i=3fd6eca9@gateway/web/ajax/mibbit.com/x-f57b5fb818f6e51f) |
| 17:10.03 | *** join/#asterisk nny_2 (n=scott@64.203.244.146) |
| 17:10.10 | nny_2 | <PROTECTED> |
| 17:10.16 | nny_2 | how could I have screwed that up heh |
| 17:11.17 | nny_2 | just updating to latest, but not sure what crappened there heh |
| 17:17.50 | *** join/#asterisk Faithful (n=Faithful@ns.linuxterminal.com) |
| 17:22.06 | iratik | Anyone using magicjack as their itsp here? |
| 17:22.29 | eppigy | what |
| 17:23.10 | srf21c | iratik: negative. |
| 17:23.28 | iratik | rats .... they are always finding a way to keep asterisk users from using their service |
| 17:23.51 | srf21c | iratik: I don't imagine that asterisk is part of the business model. |
| 17:24.18 | iratik | doesn't matter ... i'd feel bad if they were really doing advertising through their client software but they arent ... besides low rates to india |
| 17:24.37 | nny_2 | didn't work |
| 17:24.40 | nny_2 | ast_func_write: Function Timeout not registered |
| 17:24.42 | nny_2 | w-t-f |
| 17:24.52 | nny_2 | fresh comile |
| 17:24.56 | nny_2 | er compile* |
| 17:25.12 | *** join/#asterisk MaliutaLap (n=biteme@kiev.lusan.id.au) |
| 17:25.34 | [TK]D-Fender | iratik: Yes, because using 3rd party stuff with their server VIOLATES their terms of service and I hope you don't care about them kicking you off for trying, but it'd be in the "get what you deserve" category |
| 17:25.47 | [TK]D-Fender | nny_2: real pastebin would help. |
| 17:25.48 | Katty | eppigy: oh the pain. |
| 17:25.56 | nny_2 | [TK]D-Fender: |
| 17:25.57 | nny_2 | k |
| 17:26.02 | eppigy | Katty: :D |
| 17:26.14 | Katty | what did you have |
| 17:26.41 | eppigy | i had bbq beef plate |
| 17:26.43 | nny_2 | [TK]D-Fender: http://pastebin.com/m58eefe46 |
| 17:26.48 | eppigy | with green beens and tatoe salad |
| 17:27.04 | Katty | just loose pulled pork? |
| 17:27.05 | nny_2 | [TK]D-Fender: i just did a recompile of 1.4.25, the second error is from adding load => pbx_functions.so in modules.conf |
| 17:27.08 | Katty | err beef |
| 17:27.10 | [TK]D-Fender | nny_2: because functiosn are CASE SENSITIVE |
| 17:27.19 | [TK]D-Fender | nny_2: TIMEOUT() |
| 17:27.27 | [TK]D-Fender | reaches for his ClueBat (tm) |
| 17:27.39 | eppigy | Katty: sliced |
| 17:27.39 | nny_2 | ducks |
| 17:27.39 | Katty | distracts [TK]D-Fender with a cookie. |
| 17:27.41 | eppigy | ^___^ |
| 17:27.46 | Katty | eppigy: nummies. |
| 17:27.49 | eppigy | yesh |
| 17:27.54 | Katty | eppigy: i had pulled pork sammich, with salad, and bacon cheddary fries. |
| 17:27.54 | nny_2 | hey that was old code, I thought it was a proper noun :P |
| 17:28.07 | eppigy | 8[] |
| 17:28.08 | Katty | but the cheddar tasted funny. so i only had a few fries around the corners with minimal cheese |
| 17:28.14 | eppigy | o |
| 17:28.16 | eppigy | :[ |
| 17:28.17 | Katty | salad was yummy |
| 17:28.26 | Katty | :> |
| 17:28.28 | eppigy | yeah I love pulled pork sammiches |
| 17:28.40 | Katty | you know what else i like |
| 17:28.43 | Katty | my mac 182 brush |
| 17:28.45 | Katty | WHICH PIPPIN STOLE |
| 17:28.47 | Katty | cries. |
| 17:28.53 | eppigy | D: |
| 17:29.13 | nny_2 | [TK]D-Fender: thanks :P |
| 17:29.23 | nny_2 | [TK]D-Fender: you have earned your help a moron badge, grats! |
| 17:30.29 | srf21c | this talk of food is making concentration difficult... |
| 17:30.39 | *** join/#asterisk jameswf (n=james@unaffiliated/jameswf-home) |
| 17:31.16 | Katty | srf21c: we can talk about makeup instead, if you like. |
| 17:31.25 | srf21c | instead of bits and bytes, I'm thinking BBQ and bites. |
| 17:31.44 | KyleK | nny_2: haha he could cover himself in such badges if he wanted to |
| 17:31.49 | Katty | smokey eye technique. |
| 17:32.24 | Katty | eppigy: i did a very cool smokey eye today |
| 17:32.37 | Katty | eppigy: smokey pink and charcoal look. |
| 17:32.55 | eppigy | oh nice |
| 17:32.59 | nny_2 | KyleK: hehehe |
| 17:33.00 | eppigy | PICS |
| 17:33.07 | Katty | uhmmmmmmmmmm |
| 17:33.11 | Katty | hmm |
| 17:33.15 | *** join/#asterisk mltlnx (n=mltlnx@rrcs-208-105-83-114.nyc.biz.rr.com) |
| 17:33.16 | eppigy | hrmmmmmmmmmmmmmmmm |
| 17:33.32 | Katty | locates camera, brb |
| 17:33.37 | eppigy | bee arr bee |
| 17:34.43 | Katty | the focus sucks. |
| 17:34.57 | KyleK | theres no flower mode? |
| 17:35.13 | KyleK | try yelling flower power before you take the photo |
| 17:35.19 | Katty | it's kinda hard to takea picture of your EYE |
| 17:35.32 | Katty | http://farm3.static.flickr.com/2244/2264087263_ccde78bc6e.jpg <- that's what i copied tho |
| 17:35.46 | KyleK | ohhh |
| 17:36.01 | Katty | i don't have as much black under the waterline |
| 17:36.46 | [TK]D-Fender | Katty: .... Stop channeling Tammy-Fae Baker..... |
| 17:37.29 | *** join/#asterisk tobias (n=tobias@216.27.28.176) |
| 17:37.38 | Katty | [TK]D-Fender: you keep talking, and i'm going to do your hair like hers. |
| 17:37.54 | comradeb14ck | hi all |
| 17:38.08 | [TK]D-Fender | Katty: Consider me duly terrified ;) |
| 17:38.15 | [TK]D-Fender | Katty: And don't ask what's due :p |
| 17:39.17 | [TK]D-Fender | goes off to hone Mr. Pointy |
| 17:39.39 | *** join/#asterisk evildan7 (n=werdan7@freenode/staff/wikimedia.werdan7) |
| 17:40.07 | *** join/#asterisk theHub (n=theHub@69.177.93.21) |
| 17:40.08 | Katty | eppigy should be my makeup artist. |
| 17:40.25 | eppigy | that is pretty |
| 17:40.30 | eppigy | I am not good with stuff liek that |
| 17:40.37 | *** join/#asterisk grEvenX (n=even@cB78D5AC1.dhcp.bluecom.no) |
| 17:40.41 | eppigy | I am very left brained |
| 17:40.42 | Katty | i can't imagine why not. |
| 17:40.59 | Katty | you could start with something simpler, and more manly. |
| 17:41.04 | Katty | like paint swatches. |
| 17:41.56 | eppigy | D: |
| 17:42.12 | *** join/#asterisk buttons840 (n=buttons8@c-24-10-149-58.hsd1.ut.comcast.net) |
| 17:42.54 | Katty | eppigy: conviently, most have 3 shades. like eyeshadow trios. |
| 17:43.15 | buttons840 | I'm reading the book, and I have some questions about codecs and protocals? Can I use any codec for any purpose? For example, can I use any codec to make a call over the pstn? |
| 17:43.46 | eppigy | negative |
| 17:43.47 | Katty | some codecs require licensing. |
| 17:43.56 | Katty | some codecs are not supported. |
| 17:44.09 | Katty | infobot: codecs? |
| 17:44.09 | infobot | it has been said that codecs is http://snipurl.com/wiki_codecs. If you have audio/codec problems, first try to 'disallow=all' and 'allow=ulaw' and see if that works. Anyone that tells you to use 'allow=all' is an idiot as it usually causes audio problems, or Number/Name: 1/g723, 2/gsm, 4/ulaw, 8/alaw, 16/g726, 32/adpcm, 64/slin, 128/lpc10, 256/g729, 512/speex, 1024/ilibc |
| 17:47.01 | Katty | eppigy: http://www.sherwin.com/visualizer/ |
| 17:47.12 | buttons840 | I'll look into the wiki and come back if i have more q's, thanks. |
| 17:47.41 | Katty | eppigy: after it loads, click Explore Colors at top right, then interior |
| 17:48.16 | iratik | In the authorization header of the sip register packet ... I need to change the uri value without changing the register string? e.g. register => someuser:someepass@proxy.somedomain.com ... but i need the uri: part of the register packet to just say uri:somedomain.com, i can do this if i change the register string ... but then its not contacting the proxy - though the uri part will then be correct ... how can i do this? |
| 17:48.40 | [jmc] | hi guys, I have a question for you |
| 17:48.41 | Katty | eppigy: like a baby pink |
| 17:48.47 | *** join/#asterisk kalib (n=kalib@200.253.26.151) |
| 17:48.50 | [jmc] | well, it maybe just a curiosity |
| 17:48.58 | [jmc] | since Asterisk seems to be working fine |
| 17:49.01 | Katty | eppigy: and TADA |
| 17:49.02 | kalib | Hi guys. in my asterisk CLI I did type: sip debug |
| 17:49.08 | kalib | how can I disable the sip debug? |
| 17:49.08 | [jmc] | but I'm using Twinkle |
| 17:49.19 | [jmc] | kalib: sip set debug off |
| 17:49.23 | eppigy | Katty: this is intense |
| 17:49.25 | kalib | thanks. ;] |
| 17:49.27 | [jmc] | I was saying |
| 17:49.36 | [jmc] | Twinkle connects to the Asterisk account |
| 17:49.39 | *** part/#asterisk srf21c (n=seth@ip98-165-60-42.ph.ph.cox.net) |
| 17:49.42 | [jmc] | but in the account list |
| 17:49.48 | [jmc] | it shows a red error sign |
| 17:49.51 | [jmc] | and it says |
| 17:50.04 | [jmc] | âAvailability: 501 Method Not Implementedâ |
| 17:50.20 | [jmc] | does someone know what can be the cause, or if it normal? |
| 17:50.25 | iratik | shudders |
| 17:50.38 | buttons840 | I must be confused, is the compressed codec data sent out using my home telephone line? or is it decompressed on my end and then sent? what codecs (if any) are supported by my telephone line? |
| 17:50.51 | *** join/#asterisk Faithful (n=Faithful@ns.linuxterminal.com) |
| 17:50.54 | [jmc] | oh, sorry I wrote it wrong |
| 17:51.06 | [jmc] | it says âAvailability: falied to publish (501 Method Not Implemented)â |
| 17:54.01 | Katty | eppigy: 6572 Ruby Shade |
| 17:54.04 | *** join/#asterisk seanbright (n=sean@asterisk/contributor-and-bug-marshal/seanbright) |
| 17:54.29 | Katty | hi sean |
| 17:54.43 | *** join/#asterisk arpu (n=arpu@chello062178159144.10.14.univie.teleweb.at) |
| 17:55.40 | *** join/#asterisk DrAk0 (n=luisjose@nelug/coreteam/luisjose) |
| 17:56.32 | eppigy | I like the camping chair living room |
| 17:58.03 | Katty | apply that 6572 to it |
| 17:58.33 | [TK]D-Fender | buttons840: "telephone lines" don't ahve a codec per-se. Once converted from analog to digital at the telco switch it is compantded to G.711 typically. |
| 17:58.46 | [TK]D-Fender | [jmc]: Depends what that is in response TO |
| 17:59.04 | *** join/#asterisk Zathara (n=georgesi@200.217.64.118) |
| 17:59.06 | eppigy | jeepers |
| 17:59.13 | *** part/#asterisk Zathara (n=georgesi@200.217.64.118) |
| 17:59.42 | buttons840 | [TK]D-Fender, so I could use any codec, and it will be decompressed before it is sent over the pstn? |
| 17:59.58 | buttons840 | and codecs are only used for digital connections, like voip? |
| 18:00.05 | [TK]D-Fender | buttons840: Correct |
| 18:00.07 | Katty | eppigy: what, you don't like it? :P |
| 18:00.11 | buttons840 | ok, thanks |
| 18:00.28 | [TK]D-Fender | buttons840: It is advised to use G.711 though if you can afford it |
| 18:00.48 | buttons840 | afford? meaning i have enough bandwidth? |
| 18:01.47 | [TK]D-Fender | buttons840: yes |
| 18:03.05 | *** join/#asterisk [intra]lanman (n=lanman@freeswitch/developer/intralanman) |
| 18:05.48 | *** join/#asterisk MaliutaLap (n=biteme@203.171.192.87) |
| 18:06.59 | *** join/#asterisk icyValk77 (n=icyValk7@host217-43-31-121.range217-43.btcentralplus.com) |
| 18:07.08 | eppigy | Katty: I mean |
| 18:07.13 | eppigy | i dont know if not like |
| 18:07.18 | eppigy | is the right phrase |
| 18:08.13 | [jmc] | [TK]D-Fender: in response to this: |
| 18:08.16 | [jmc] | http://pastebin.com/m717b9373 |
| 18:09.27 | *** join/#asterisk hi365 (n=hi365@94.159.176.127) |
| 18:12.40 | *** join/#asterisk profXavier (n=MyNick@unaffiliated/neverblue) |
| 18:12.52 | [TK]D-Fender | [jmc]: Looks like a presence notification. Indeed * does not care about that... you can safely ignore it knowing this |
| 18:12.54 | profXavier | guys, im trying to setup a Polycom IP601 phone |
| 18:13.05 | profXavier | it really hangs, almost 5 minutes, on startup |
| 18:13.12 | [jmc] | ok |
| 18:13.21 | profXavier | can I do anything to get the phone to boot, working, quicker ? |
| 18:15.11 | *** join/#asterisk andresmujica (n=andresmu@ubuntu/member/andresmujica) |
| 18:15.18 | [TK]D-Fender | profXavier: You mean jsut slow to boot? |
| 18:15.23 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 18:15.31 | [TK]D-Fender | profXavier: Not that it hangs as in "crashing"? |
| 18:15.40 | profXavier | Fender, slow to boot |
| 18:16.14 | profXavier | i am trying to setup tftpboot on a server, so I can update firmware, configure the phone |
| 18:16.18 | profXavier | its not working too hot.. |
| 18:16.35 | [TK]D-Fender | profXavier: Use the latest firmware and provision it to use the model-specific firmware and not the composite "sip.ld" |
| 18:16.38 | *** join/#asterisk ruben23 (n=AGENT@124.107.3.178) |
| 18:16.39 | profXavier | but the wait between boots is ridiculious |
| 18:16.42 | ariel_ | not working too hot? Hum polycoms take for ever to boot |
| 18:16.56 | [TK]D-Fender | ariel_: Mine take 2 mins tops... |
| 18:17.00 | profXavier | ariel_: sometimes over 5 minutes ? |
| 18:17.13 | profXavier | i see its using the sip.ld |
| 18:17.14 | [TK]D-Fender | ariel_: then again... if you do it right the first time you only have to boot them once anyway ;) |
| 18:17.17 | ariel_ | [TK]D-Fender: mine as well about 2 minutes or less |
| 18:17.18 | profXavier | where do I change that ? |
| 18:17.29 | [TK]D-Fender | profXavier: in the ,mac>.cfg |
| 18:17.36 | [TK]D-Fender | profXavier: in the <mac>.cfg |
| 18:17.44 | ariel_ | but still seems like a life time when your waiting for them... and talk about updating them when you move to a new firmware... |
| 18:17.47 | [TK]D-Fender | profXavier: Thats in the admin guide |
| 18:18.25 | [jmc] | another question... |
| 18:18.42 | [TK]D-Fender | ariel_: I suppose if you're changing them that often. I tend to batch mine so I just down the switch and they all update at once. 5 minutes every couple of months hardly matters to me... |
| 18:18.50 | profXavier | ok, spoonfeed please |
| 18:19.05 | profXavier | in the <mac>.cfg, I will see a reference to sip.ld |
| 18:19.16 | profXavier | and I need to change that to something else, or just remove ? |
| 18:19.18 | [jmc] | when I call myself on Twiknle (yes, I do that, don't ask me why, I need to do that) |
| 18:19.25 | [jmc] | the phone rings |
| 18:19.28 | [jmc] | but when I answer |
| 18:19.33 | [jmc] | it hangs up |
| 18:19.39 | [jmc] | here's my extensions.conf: http://pastebin.com/m182a4315 |
| 18:19.43 | [TK]D-Fender | [jmc]: Probably trying to reinvite to itself... |
| 18:19.48 | ariel_ | [TK]D-Fender: it only seems slow when your doing the first 1 or 2 in the lab to make sure they don't work....but we do them over night as well. |
| 18:19.50 | [jmc] | and the debug is: |
| 18:19.53 | [TK]D-Fender | [jmc]: You can't talk to yourself on the same device.. |
| 18:20.00 | [jmc] | <PROTECTED> |
| 18:20.02 | [jmc] | <PROTECTED> |
| 18:20.04 | [jmc] | <PROTECTED> |
| 18:20.06 | [jmc] | <PROTECTED> |
| 18:20.20 | [TK]D-Fender | [jmc]: PASTEBIN... please stop spamming |
| 18:20.24 | [jmc] | [TK]D-Fender: I don't know, it seemed to work earlier |
| 18:20.36 | [TK]D-Fender | [jmc]: I highly recommend you come up with another test |
| 18:20.37 | [jmc] | I'm not "spamming" |
| 18:20.46 | profXavier | oh oh |
| 18:20.51 | [jmc] | maybe "flooding" would be a better term :D |
| 18:21.26 | [TK]D-Fender | [jmc]: Yes is would. So don't do that EITHER :p |
| 18:21.39 | [jmc] | lol |
| 18:21.46 | [jmc] | [TK]D-Fender: don't get me wrong |
| 18:22.11 | [jmc] | I thought that pastebin-ning *four* lines would be even less comfortable for you |
| 18:22.27 | [jmc] | to open up a new page every time |
| 18:22.42 | [jmc] | I'll stop it if it's disturbing. :) |
| 18:22.44 | profXavier | APP_FILE_PATH="sip.ld" |
| 18:22.58 | profXavier | so I remove that entirely? replace it? |
| 18:22.59 | [TK]D-Fender | [jmc]: No. I read pastebins. If fact if someone has an issue and were nice enough to put it in one and I wasn't even following that conversation I'd look anyway and be more inclined to help |
| 18:23.05 | jjshoe | sssh fonality is coming |
| 18:23.12 | [TK]D-Fender | profXavier: Looks like a clear "replace" |
| 18:23.29 | profXavier | with ? |
| 18:23.40 | profXavier | oh |
| 18:23.41 | profXavier | i see |
| 18:23.43 | [TK]D-Fender | profXavier: the firmware for your specific model. |
| 18:23.44 | profXavier | gotcha |
| 18:23.50 | profXavier | sorry, for my ignorance |
| 18:23.59 | [jmc] | [TK]D-Fender: then I'll do that, I'm sorry ;) |
| 18:24.03 | ruben23 | hi anyone can suggest how do i execute this setting on my dial plan so that my voip trunk would work http://pastebin.com/m2c2602f2 |
| 18:24.07 | [TK]D-Fender | jjshoe: b14ck is alrady here ;) |
| 18:24.11 | [TK]D-Fender | already* |
| 18:24.51 | *** join/#asterisk saxa (n=sasa@host242-95-static.223-217-b.business.telecomitalia.it) |
| 18:24.54 | [TK]D-Fender | ruben23: exten => _Z.,1,Dial(SIP/SIPtrunk/9900131501${EXTEN}) |
| 18:25.03 | [TK]D-Fender | ruben23: And fix the "}" at the end of it |
| 18:25.26 | [TK]D-Fender | (the sip.conf entry) |
| 18:25.26 | jjshoe | [TK]D-Fender I was reffering to samy :P the one from fonality that's actually someone worthwhile watching |
| 18:25.37 | jjshoe | if history has anything to teach us, the ce guy won't be around for much more then a year |
| 18:26.05 | saxa | hi ppl |
| 18:26.10 | [TK]D-Fender | jjshoe: I admire your optimism :) |
| 18:26.49 | ruben23 | <PROTECTED> |
| 18:26.50 | saxa | is there any site out there, where are some templates of the extensions.conf file, something like a menu |
| 18:27.04 | [jmc] | saxa: oh yes there is |
| 18:27.08 | saxa | or do i need to do one myself ? |
| 18:27.24 | [TK]D-Fender | ruben23: Fix the peer name, you have bad braces. |
| 18:27.32 | saxa | [jmc]: coll, may i get the link ? |
| 18:27.37 | profXavier | thanks again Fender |
| 18:27.40 | ariel_ | saxa: have you seen the sample files |
| 18:27.42 | [TK]D-Fender | saxa: Yes, you have to do it yourself |
| 18:27.44 | [jmc] | saxa: oh actually I read the wrong thing |
| 18:27.54 | [TK]D-Fender | prooGone though a reboot cycle with it? |
| 18:27.55 | saxa | :) |
| 18:27.56 | saxa | ok |
| 18:28.00 | [TK]D-Fender | profXavier: Gone though a reboot cycle with it? |
| 18:28.01 | [jmc] | saxa: http://www.voip-info.org/wiki/view/IVR |
| 18:28.10 | [jmc] | I did never search for templates |
| 18:28.18 | [jmc] | but you can find good examples |
| 18:28.19 | JayTee52 | saxa, are you aware of the book? |
| 18:28.24 | [jmc] | and start making you menu from there |
| 18:28.26 | [TK]D-Fender | saxa: for that link keep in mind the WIKI has a lot of 1.0 code that needs adjusting. |
| 18:28.30 | saxa | would be good to have some templates in various languages |
| 18:28.34 | JayTee52 | ~book |
| 18:28.34 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 18:28.49 | saxa | JayTee52: yes, i'm reading the book |
| 18:29.13 | saxa | actually the book is also 1.4 |
| 18:29.23 | JayTee52 | I think the book was released in a Spanish version but not sure if there's a PDF available. |
| 18:29.34 | saxa | i mean some thing doesnt work exactly the same way in * 1.6 |
| 18:30.00 | saxa | it a nice book, well written, |
| 18:30.07 | JayTee52 | true and the authors are hard at work (kicks leif to wake him up from his nap) on a 1.6 version |
| 18:30.46 | saxa | yes, i just tought maybe there is a site with varuos configs already pre-configured |
| 18:31.03 | saxa | anyway, will construct it by myself |
| 18:31.24 | JayTee52 | saxa, as far as other language examples of dialplan examples or configs I'd say your best bet is finding a local Asterisk group in your country if one exists. |
| 18:31.24 | leifmadsen | JayTee52: heh... ya ya ya :) |
| 18:31.37 | JayTee52 | :-) |
| 18:31.54 | JayTee52 | saxa, what language in particular? |
| 18:33.36 | *** join/#asterisk Octothorpe (n=Octothor@pdpc/supporter/professional/octothorpe) |
| 18:36.26 | JayTee52 | I now have a copy of MS Press Microsoft System Center Configuration Manager 2007 which I now have to read to get up to speed on to make sure that the MCSE (Must Call Someone Else) that was directed to deploy it did a proper job because he screws up everything else. |
| 18:37.11 | JayTee52 | fortunately I already knew SMS 2003 and took a class in it and used it alot so I don't need a large dose (fatal) of sleeping pills or a gun :-) |
| 18:37.41 | JayTee52 | but the pills do seem tempting |
| 18:39.16 | [TK]D-Fender | hands JayTee52 a bottle of Troika |
| 18:39.20 | *** join/#asterisk propellerhead (n=yogurt2u@host44.190-230-217.telecom.net.ar) |
| 18:39.24 | [TK]D-Fender | JayTee52: A little something to wash that down with ;) |
| 18:39.59 | JayTee52 | good russian vodka. hard to come by here. better than Stoli |
| 18:40.34 | drfreeze | Hello, Got a situation where a caller gets 1 ring and then gets disconnected |
| 18:41.04 | [TK]D-Fender | JayTee52: Smirnoff = uneducated choice :) |
| 18:41.21 | drfreeze | http://pastie.textmate.org/private/cyjpzawd3nuqg3wefumphw |
| 18:41.29 | *** join/#asterisk arpu (n=arpu@chello062178159144.10.14.univie.teleweb.at) |
| 18:41.35 | drfreeze | It's a digital PRI setup |
| 18:41.44 | JayTee52 | [TK]D-Fender, Smirnoff is just one level above Gordon's and both of those are made from squeezing the socks of cadavers in Russian morgues. |
| 18:43.20 | iratik | The sip registration header on the "From" line needs to say "something" <sip:balnk@domain.com>;tag=sometag How do you configure sip.conf to do the "something" part ? |
| 18:44.36 | *** join/#asterisk tfrew (n=tfrew@office.neteasyinc.com) |
| 18:45.02 | [TK]D-Fender | drLooks like a forwarding issue |
| 18:45.06 | [TK]D-Fender | drfreeze: Looks like a forwarding issue |
| 18:45.31 | JayTee52 | drfreeze, would appear one or more of the SIP phones is set to Do Not Disturb or forwarded OR you have a call limit set and it's exceeded it. |
| 18:45.44 | [TK]D-Fender | JayTee52: No, its a forward... |
| 18:45.53 | [TK]D-Fender | JayTee52: otherwise ti'd be a straight reject |
| 18:46.00 | drfreeze | [TK]D-Fender: yes, it's national set-your-phone-to-forwarding-or-do-not-disturbt day |
| 18:46.10 | JayTee52 | [TK]D-Fender, yeah, just reread through it. SIP/721 |
| 18:46.22 | [TK]D-Fender | drfreeze: good... under 24h to auto-resolution ;) |
| 18:46.22 | Katty | bored. |
| 18:46.33 | iratik | <PROTECTED> |
| 18:46.42 | [TK]D-Fender | JayTeethe issue there is 721 was in the initial call list as well |
| 18:47.04 | [TK]D-Fender | iratik: "core show applications like sip" |
| 18:47.41 | iratik | SIPAddHeader can it modify sip headers? |
| 18:47.41 | *** join/#asterisk lanning (n=lanning@nat/yahoo/x-8cb30392865ca404) |
| 18:48.15 | iratik | it cannot modify sip headers |
| 18:48.16 | [TK]D-Fender | iratik: No. |
| 18:48.33 | [TK]D-Fender | iratik: Ah I missed that. No, what you see is what you get |
| 18:48.55 | [TK]D-Fender | iratik: Anything like this will require modding chan_sip.c |
| 18:49.16 | iratik | Or writing a proxy |
| 18:49.58 | [TK]D-Fender | iratik: Well yeah if you want to go the MIM route |
| 18:52.15 | drfreeze | JayTee52: [TK]D-Fender thanks |
| 18:56.06 | mgroman | !ohmy |
| 18:59.01 | [TK]D-Fender | !ohhenry |
| 19:01.17 | profXavier | when the phone is parked, I can dial just fine, but when i have a dialtone, and I attempt to dial, what do I need to change in the dialplan, for the phone to work the same as parked ? |
| 19:02.00 | *** part/#asterisk icyValk77 (n=icyValk7@host217-43-31-121.range217-43.btcentralplus.com) |
| 19:04.21 | [TK]D-Fender | profXavier: "phone is parked"? You mean dialing on-hook vs off-hook? |
| 19:06.45 | profXavier | correct |
| 19:08.33 | [TK]D-Fender | profXavier: On-hook you can whatever you want. Off-hook you are controlled byt he phone's local dialplan which follows the MGCP RFC and is well documented in the admin guide |
| 19:08.48 | profXavier | i think its the ordering of my context |
| 19:09.10 | profXavier | if I pass a 9+7 digit # |
| 19:09.20 | profXavier | vs. 9 + 10 digit # |
| 19:09.46 | profXavier | it might match in the context, but will the order matter ? |
| 19:09.58 | *** join/#asterisk ReDNeQ (n=ReDNeQ@70.114.229.58) |
| 19:10.06 | profXavier | [2-9]11|0T|100|101|011xxx.T|9011xxx.T|1[2-9]xxxxxxxxx|91[2-9]xxxxxxxxx|9250xxxxxxx|9[2-9]xxxxxx|*xx|[8]xxx|[2-7]xx |
| 19:10.28 | profXavier | when I dial 92504247896 |
| 19:10.40 | profXavier | it dials just 9+ the first 7 digits |
| 19:12.19 | profXavier | *this is when the headset is picked up, and I have a dialtone* |
| 19:14.10 | [TK]D-Fender | profXavier: Well you do have a 9250xxxxxxx pattern so you might think that it'd force you to wait. perhaps the order does matter |
| 19:14.41 | *** join/#asterisk hi365 (n=hi365@94.159.177.112) |
| 19:17.43 | KyleK | whats the T mean |
| 19:17.51 | carrar | 2504247896 will match 9[2-9]xxxxxx |
| 19:17.57 | carrar | 92504247896 will match 9[2-9]xxxxxx |
| 19:18.17 | *** join/#asterisk unasi7 (n=unasi7@85.4.50.131) |
| 19:18.49 | *** join/#asterisk brunner (n=chris@68-119-87-106.dhcp.mtgm.al.charter.com) |
| 19:20.24 | brunner | does Caller ID come through on a BRI faster than it does on a POTS line? |
| 19:22.58 | *** part/#asterisk green-monkey (n=ericshel@70.102.50.18) |
| 19:23.13 | *** join/#asterisk nfi|ermes (n=ErMeS@217.220.121.62) |
| 19:24.52 | KyleK | that depends on how a bri works, does it have a separate data channel? |
| 19:26.33 | *** join/#asterisk Pwn-BoFH (n=Coto@pc-86-7-239-201.cm.vtr.net) |
| 19:29.00 | [TK]D-Fender | brunner: Yes |
| 19:29.27 | [TK]D-Fender | brunner: BRI = ISDN = D-Channel signalling and thus instant. Analog = FSK inter-ring delay |
| 19:29.55 | [TK]D-Fender | (depending on regional style. DTMF CID = worse) |
| 19:33.35 | batphone | hmm |
| 19:33.48 | batphone | rsa keys are not shown when i type 'reload res_crypto.so' |
| 19:33.54 | *** part/#asterisk juanIMP (n=juan@200.71.41.254) |
| 19:34.00 | batphone | why this is? |
| 19:34.36 | saxa | JayTee52: i need brazilian language, russian, english and italian |
| 19:34.37 | *** join/#asterisk |Cybex| (n=John@80.100.126.176) |
| 19:37.20 | *** join/#asterisk mltlnx (n=mltlnx@asams.mserve.com) |
| 19:37.37 | JayTee52 | saxa, I know you won't find dialplan examples on the WIKI in those languages. As for the book you might check O'Reilly's website to see what other languages it's been translated to. |
| 19:39.17 | KyleK | dialplan in different languages? as in Zifferblatt(SIP/.....? |
| 19:41.00 | *** join/#asterisk _Sam-- (n=sam@unaffiliated/sam--/x-573746) |
| 19:41.15 | _Sam-- | what controls which users can issue the command from a unix shell 'asterisk -rx' ? |
| 19:41.27 | saxa | JayTee52: ok, i maybe rcord my own voice in that case :) |
| 19:41.28 | _Sam-- | i can run it fine as root, but want to run something as the same user as our webserver to get some queue statuses |
| 19:41.32 | [TK]D-Fender | _Sam--: rights to the binary |
| 19:41.38 | _Sam-- | the binary is chmod fine |
| 19:41.52 | _Sam-- | -rwxr-xr-x 1 root root 10615030 2008-08-08 10:50 /usr/sbin/asterisk |
| 19:42.12 | JayTee52 | saxa, record? you mean voice prompt files? |
| 19:42.16 | [jmc] | _Sam--: anyone can launch that |
| 19:42.24 | [TK]D-Fender | _Sam--: May have to tweak the rightst o the PID, configs, etc |
| 19:42.31 | [jmc] | if /usr/sbin is in the PATH |
| 19:42.33 | _Sam-- | only the user root can get results returned for the command asterisk -rx 'show queues' |
| 19:42.46 | [jmc] | yes |
| 19:42.47 | _Sam-- | even when the full path is in the command. |
| 19:42.56 | saxa | JayTee52: yes |
| 19:43.04 | [jmc] | that's because of effective rights |
| 19:43.14 | *** join/#asterisk Sajam (n=chatzill@77.42.192.144) |
| 19:43.21 | [jmc] | a program is launched with the rights of the launched user |
| 19:43.23 | [jmc] | * launcher |
| 19:43.23 | _Sam-- | [jmc] : i dont understand your point -- the binary is exectuable by anyone. |
| 19:43.32 | [jmc] | _Sam--: wait... |
| 19:43.34 | _Sam-- | but the command isnt returning restuls for anyone but root. |
| 19:44.13 | _Sam-- | [TK]D-Fender : could you tell me any tips or clues wher ei would look first? |
| 19:44.26 | _Sam-- | i really dont want to use sockets and the manager interface. |
| 19:44.43 | [jmc] | _Sam--: is it returning errors either? |
| 19:45.05 | [jmc] | if user X can't access something, neither 'asterisk' launched by X will |
| 19:45.20 | _Sam-- | it does return an error, something along the lines of cant connect to /var/run/asterisk.ctl, even though asterisk is running and that ctl file exists. |
| 19:45.46 | [jmc] | ooh ç= |
| 19:45.49 | [jmc] | :) |
| 19:45.55 | [jmc] | check the permissions on /var/run |
| 19:46.06 | [jmc] | and on asterisk.ctl as well |
| 19:46.19 | [TK]D-Fender | _Sam--: I told you... rights to the PID as well |
| 19:46.22 | eppigy | Katty: hungry |
| 19:46.33 | [jmc] | yeah, they're called effective rights |
| 19:46.50 | _Sam-- | <PROTECTED> |
| 19:47.07 | [jmc] | what about asterisk.ctl? |
| 19:47.07 | _Sam-- | aha. |
| 19:47.20 | [jmc] | as a last resort, you could setgid the binary |
| 19:47.21 | _Sam-- | this is likely the cause? -rw-r--r-- 1 root root 5 2009-06-02 17:04 asterisk.pid |
| 19:47.35 | [jmc] | no, asterisk.ctl? |
| 19:47.41 | _Sam-- | asterisk.ctl is ther,e and fine. |
| 19:47.50 | _Sam-- | srwxr-xr-x 1 root root 0 2009-06-02 17:04 asterisk.ctl |
| 19:47.55 | JayTee52 | saxa, sorry but afaik the sound prompt files are available only in english, spanish and french |
| 19:48.00 | [jmc] | _Sam: see |
| 19:48.07 | [jmc] | users can't write to it |
| 19:48.14 | JayTee52 | http://downloads.asterisk.org/pub/telephony/sounds/ |
| 19:48.18 | _Sam-- | why would you need to write to it to get the queue status? |
| 19:48.18 | [jmc] | they can only read and (wtf?) execute it |
| 19:48.32 | *** join/#asterisk juanIMP (n=juan@200.71.41.254) |
| 19:48.45 | saxa | JayTee52: so I say, I will record my own ones. |
| 19:48.49 | [jmc] | to communicate with asterisk I guess? |
| 19:49.20 | _Sam-- | what are the proper perms on the ctl file? |
| 19:49.20 | JayTee52 | saxa, guess so. |
| 19:49.26 | _Sam-- | and thank you, both. |
| 19:49.42 | [jmc] | _Sam: I think the ones you need most |
| 19:49.53 | [jmc] | if I were in you, I'd add a new "asterisk" group |
| 19:49.53 | [TK]D-Fender | saxa: the WIKI lists many other packs already created in other languages and accents |
| 19:50.01 | [jmc] | which has access to the control files |
| 19:50.07 | _Sam-- | thanks. |
| 19:50.08 | [jmc] | and only add the interested users to it |
| 19:50.12 | [jmc] | :) |
| 19:50.14 | [jmc] | you're welcome |
| 19:50.25 | [TK]D-Fender | saxa: I've seen Canadian vs Parisian French, UK & AU English vs USA |
| 19:50.33 | saxa | [TK]D-Fender: thx, will look if something suits me |
| 19:50.43 | _Sam-- | thanks again, and even yet again. |
| 19:50.51 | _Sam-- | sorry for being confrontational -- you were 100% right. |
| 19:50.55 | _Sam-- | working fine. |
| 19:51.35 | [jmc] | nice to hear that, Sam ^^ |
| 19:54.18 | _Sam-- | good day. |
| 20:01.32 | JayTee52 | Isn't Canadian French just Parisian French minus the aloof intonation? |
| 20:02.17 | beek | Afternoon JayTee52 |
| 20:02.36 | *** join/#asterisk srf21c (n=seth@ip98-165-60-42.ph.ph.cox.net) |
| 20:03.31 | [TK]D-Fender | JayTee52: terminolgy is different as well, "diaise" vs "carre" for #. Once you learn one, the other stands out like a sore thumb |
| 20:04.57 | *** part/#asterisk nny_2 (n=scott@64.203.244.146) |
| 20:10.10 | *** join/#asterisk grEvenX (n=even@cB78D5AC1.dhcp.bluecom.no) |
| 20:12.38 | *** join/#asterisk [intra]lanman (n=lanman@freeswitch/developer/intralanman) |
| 20:16.03 | *** join/#asterisk scalex000 (n=chatzill@215puntacana02.codetel.net.do) |
| 20:16.11 | scalex000 | hello |
| 20:16.16 | scalex000 | I need suggestion |
| 20:16.20 | *** join/#asterisk bmoraca (n=chatzill@66.242.174.254) |
| 20:17.16 | scalex000 | help? |
| 20:17.35 | srf21c | ok... |
| 20:17.42 | srf21c | can you be more specific please? |
| 20:17.43 | Aiatek | <scalex000> why dont you ask? |
| 20:17.49 | tfrew | scalex200: sounds like you ned a shrink |
| 20:17.49 | scalex000 | thank you |
| 20:18.04 | scalex000 | nop |
| 20:18.36 | Deeewayne | O.O |
| 20:18.37 | scalex000 | well I need to install a PBX so, I dont know what kind ot switch a need can you recommend some one, |
| 20:18.48 | scalex000 | I will going to install a VOip |
| 20:19.03 | tfrew | dell 24 port web managed switchs work |
| 20:19.23 | tfrew | you can setup qos on voip, and put the sip phones on thier own vlan |
| 20:19.32 | srf21c | scalex000: you might want to consider a switch that has PoE so you can power your IP phones via the network cable. |
| 20:19.45 | tfrew | double the cost to get that feature |
| 20:20.04 | tfrew | but yes, that is good, and run the switch on a 1500va ups |
| 20:20.24 | scalex000 | ok |
| 20:20.38 | tfrew | how many phones are you going to get? |
| 20:20.45 | tfrew | or will need to expand to? |
| 20:20.58 | scalex000 | well from begin about 8 |
| 20:21.10 | *** join/#asterisk joako (n=joako@opensuse/member/joak0) |
| 20:21.21 | scalex000 | station |
| 20:21.31 | tfrew | http://configure.us.dell.com/dellstore/config.aspx?c=us&cs=04&l=en&oc=bccw6k1&s=bsd |
| 20:21.38 | tfrew | that's cheap for the features |
| 20:22.23 | scalex000 | ok |
| 20:22.27 | scalex000 | thanks a lot |
| 20:22.27 | tfrew | down to http://www.dell.com/us/en/business/enterprise/switch-powerconnect-2816/pd.aspx?refid=switch-powerconnect-2816&s=bsd&cs=04 |
| 20:22.30 | tfrew | for 16 port |
| 20:22.35 | tfrew | no poe |
| 20:22.56 | tfrew | but you can still do vlan's on that 16 port model and tag your sip packets |
| 20:22.59 | batphone | anyone using RSA authentication on their IAX trunks? |
| 20:23.33 | scalex000 | the function of PoE this depend of what kind of phone I use or every IP phone have this option |
| 20:23.48 | JayTee52 | beek, afternoon! sorry, but I was AFK |
| 20:24.41 | tfrew | if it sais its supported on the the box.... |
| 20:24.49 | tfrew | polycoms and grandstreams come to mind |
| 20:24.59 | tfrew | $120-$300 per phone depending on features |
| 20:26.10 | *** join/#asterisk Ritzerisk (n=Ritztech@nv-65-40-156-46.sta.embarqhsd.net) |
| 20:27.07 | scalex000 | :( |
| 20:27.10 | scalex000 | ok |
| 20:27.15 | scalex000 | thanks anyway |
| 20:28.18 | mltlnx | What would be the correct way to configure variables on the CALLED channel? |
| 20:29.31 | *** join/#asterisk kuku1 (n=ingo@c-67-165-174-85.hsd1.il.comcast.net) |
| 20:30.36 | srf21c | batphone: not using RSA on IAX trunks, but was curious if you've used encryption w/IAX trunks |
| 20:32.27 | kuku1 | I have a queue, and wish to dial an outside extension ( cell phone via sip ), so in 1.2 I created an entry in that same context, and it worked, but in 1.6, I get an error. [Jun 16 15:32:18] WARNING[13877]: chan_sip.c:4526 create_addr: No such host: 111 |
| 20:35.10 | profXavier | sorry, went for lunch |
| 20:35.16 | profXavier | so you think the order doesnt matter |
| 20:35.20 | profXavier | hmmm |
| 20:35.32 | bmoraca | has anyone used the Linksys SPA8000 media gateway? Any input on pros/cons/stability? |
| 20:35.59 | profXavier | so I must have to add something else to the context (digimap) |
| 20:36.31 | batphone | srf21c: i have used openvpn SSL tunnels for it in the past |
| 20:36.35 | batphone | srf21c: not much to it |
| 20:36.59 | srf21c | do you know any ITSPs that support OpenVPN tunnels? |
| 20:37.05 | srf21c | Right now I'm using teliax. |
| 20:37.12 | profXavier | carrar, you replied too, I would love to hear anything you would like to add |
| 20:37.33 | srf21c | Built-in encryption support for IAX at present seems poorly documented and not well tested. |
| 20:38.00 | batphone | srf21c: doubtful that Teliax does but I think CBeyond will set up an IPSec VPN for you |
| 20:38.18 | batphone | srf21c: but they install a cisco IAD for CPE |
| 20:39.34 | srf21c | ~iad |
| 20:40.05 | srf21c | ok, thanks. |
| 20:42.44 | *** join/#asterisk shido6 (n=shido6@96-28-35-120.dhcp.insightbb.com) |
| 20:50.45 | JayTee52 | quittin time, bbiab |
| 20:57.48 | *** join/#asterisk Octothorpe (n=Octothor@pdpc/supporter/professional/octothorpe) |
| 20:59.15 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-179-75-202.static.bezeqint.net) |
| 21:01.02 | [TK]D-Fender | ditto |
| 21:03.06 | Katty | eppigy: ATTENTION |
| 21:03.10 | Katty | eppigy: SERIOUS QUESTION |
| 21:03.13 | Katty | eppigy: SRSLY |
| 21:04.13 | eppigy | HI |
| 21:04.25 | eppigy | Katty: HY |
| 21:04.31 | Katty | eppigy: do i want a KFC honey bbq snacker with green beans and corn. OR |
| 21:04.42 | Katty | eppigy: so i want a 6 inch sub, of some sort. |
| 21:04.47 | Katty | eppigy: from subway |
| 21:07.25 | drmessano | thinks you can't ever go wrong with 6 inches |
| 21:08.40 | Katty | eppigy: or possibly the bob evans heratige chef salad. |
| 21:08.52 | Katty | eppigy: which also sounds absolutely amazing right now |
| 21:11.27 | *** join/#asterisk jaguiar (n=jaguiar@201.114.67.164) |
| 21:13.33 | comradeb14ck | hey--im using asteirsk 1.6 AGI. it looks as if CHANNEL STATUS "" and CHANNEL STATUS both work |
| 21:13.40 | comradeb14ck | in 1.6 are the ""s optional? |
| 21:17.13 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 21:19.27 | *** join/#asterisk Blackvel (n=blackvel@dslb-084-057-082-166.pools.arcor-ip.net) |
| 21:20.32 | *** join/#asterisk gazzerh (n=garryh@93-97-187-150.zone5.bethere.co.uk) |
| 21:20.34 | vAd0r | how do i use sercure sip |
| 21:20.38 | Katty | eppigy: IT HAS BEEN DECIDED |
| 21:20.57 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.218.194) |
| 21:21.22 | Katty | YOU |
| 21:21.23 | Katty | GET OUT |
| 21:21.35 | [TK]D-Fender | Katty: Already did... I'm back now :p |
| 21:21.40 | Katty | oh. |
| 21:21.41 | Katty | well. |
| 21:21.47 | Katty | i guess that is acceptable. |
| 21:21.55 | [TK]D-Fender | Katty: No, not well.. but not sick ;) |
| 21:22.53 | carrar | This looks orange, wonder if Digium makes it: http://www.dapperstache.com/picotheday/laptop_privacy |
| 21:23.21 | saxa | ok |
| 21:23.26 | saxa | if I see the following |
| 21:23.29 | saxa | quadserv*CLI> iax2 show peers |
| 21:23.29 | saxa | Name/Username Host Mask Port Status |
| 21:23.32 | saxa | brastrak/brastr (Unspecified) (D) 255.255.255.255 0 Unmonitored |
| 21:23.35 | saxa | 1 iax2 peers [0 online, 0 offline, 1 unmonitored] |
| 21:23.44 | saxa | it means that the other box is connected or not ? |
| 21:24.25 | saxa | I have 2 boxes, one in brazil, another one in italy |
| 21:24.55 | saxa | so the call from Brazil were working, but now if I try to call it says that the it box is congested |
| 21:25.06 | saxa | i dont know why, maybe a router problem ? |
| 21:25.18 | saxa | i have port 4569 open |
| 21:25.24 | [TK]D-Fender | saxa: Means you didn't specify a host, and the other side didn't register and * has nowhere to contect to reach them |
| 21:25.46 | Katty | [TK]D-Fender: tummyache |
| 21:25.55 | saxa | I have the brazil box on a dynamic ip |
| 21:26.06 | saxa | and the it box on a static |
| 21:26.25 | saxa | I put the italy ip in the iax.conf on the brazil box |
| 21:26.34 | saxa | so it were connecting before |
| 21:26.52 | *** join/#asterisk ghenry (n=ghenry@pdpc/supporter/monthlybyte/ghenry) |
| 21:28.49 | profXavier | help ? |
| 21:29.02 | Blackvel | Hi guys, I have patton gw and snom 370 |
| 21:29.04 | profXavier | trying to setup my digitmap on my Polycom phone |
| 21:29.13 | KyleK | help me obiwan kenobi, you're my only hope |
| 21:29.13 | saxa | [TK]D-Fender: this is what i see in the console |
| 21:29.16 | saxa | http://pastebin.com/m1ce7e02e |
| 21:29.29 | Blackvel | when I use the DIAL command, how shall I check for busy status? Shall ) use Playback(busy) or use the Busy command? |
| 21:29.38 | profXavier | when the call is hung up, I can dial 9+10 digit numbers |
| 21:30.03 | profXavier | but when the receiver is off, I cannot, it just dials when I hit 9+7 digits |
| 21:30.11 | Blackvel | once when I was playing around with zaptel/bristuff, BUSY cmd could not be used and a special ZAP variable had to be set |
| 21:30.28 | Blackvel | how do you guys handle busy state on snoms to play a busy tone signal to the caller? |
| 21:31.32 | [TK]D-Fender | saxa: You should not be putting auth into your DIAL statement, and that does not tell my why you set up a peer and aren't REGISTERING to it like you're supposed to |
| 21:32.15 | saxa | ok, I have the auth also in the iax.conf |
| 21:32.28 | [TK]D-Fender | profXavier: Did you try changing the order? |
| 21:32.54 | saxa | register => brastrak:password@italybox.com |
| 21:32.56 | [TK]D-Fender | BlackWaht you show is not checking for status, its playing indication back to the caller. |
| 21:33.07 | [TK]D-Fender | saxa: It isn't registered |
| 21:33.20 | [TK]D-Fender | saxa: [17:23]<saxa>brastrak/brastr (Unspecified) (D) 255.255.255.255 0 Unmonitored <----- not registered |
| 21:33.28 | [TK]D-Fender | (unspecified) |
| 21:34.01 | saxa | ok |
| 21:34.14 | ariel_ | it's that time, time to go home.... good night folks |
| 21:34.26 | profXavier | Fender, yes, hasn't worked |
| 21:34.30 | Blackvel | cu ariel |
| 21:34.39 | saxa | so this means the br box doesnt get in touch with the it box correct ? |
| 21:34.45 | profXavier | i can dial 912503425876 |
| 21:34.55 | profXavier | but not 92503425876 |
| 21:35.39 | [TK]D-Fender | saxa: It means the other side has not registered against that peer entry |
| 21:35.51 | profXavier | !book |
| 21:36.01 | profXavier | i need to review the contexts, I guess |
| 21:36.01 | [TK]D-Fender | ~book |
| 21:36.01 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 21:36.03 | profXavier | thanks |
| 21:36.15 | [TK]D-Fender | profXavier: No, your problem is a Polycom problem, not an * one. |
| 21:37.19 | profXavier | oh |
| 21:37.36 | profXavier | i thought the contexts where closely intertwined |
| 21:37.48 | [TK]D-Fender | profXavier: You did say that it was cutting you from dialing the full number therefore its teh phone that needs adjustment |
| 21:38.13 | [TK]D-Fender | profXavier: * only comes into play once the phone decides to hand off the * and that is what is cutting you off short |
| 21:38.16 | profXavier | right, but doesnt the logic//syntax of the Polycom match that of * ? |
| 21:38.18 | tfrew | profXavier: look at the polycom rpm from the trixbox distro |
| 21:38.23 | saxa | [TK]D-Fender: so my register => line should have only the host in it ? without the user:password@host part ? |
| 21:38.30 | tfrew | the xml file has the polycom dialplan set correctly |
| 21:38.38 | [TK]D-Fender | profXavier: No. Polycom follows MGCP RFC, * is its own |
| 21:38.42 | profXavier | oh |
| 21:38.47 | profXavier | well, that helps to know |
| 21:39.25 | [pnp]tomas | Is anyone using Fax For Asterisk to do outbound faxing from workstations? |
| 21:39.33 | [TK]D-Fender | profXavier: I did tell you that a few hours ago ;) |
| 21:39.44 | *** join/#asterisk chandoo (n=chandoo@ool-4353b978.dyn.optonline.net) |
| 21:39.48 | profXavier | sorry tfrew, I am not sure I understand what your talking about |
| 21:40.01 | profXavier | Fender, sometimes I chose not to listen to you :D |
| 21:40.16 | [TK]D-Fender | profXavier: No reason not to have today... |
| 21:40.36 | tfrew | profXavier: that's an understatement |
| 21:41.45 | j_kroon | when using asterisk -rx "file convert ..." I seem to be getting my shell back before asterisk has finished the transcoding - is this intended behaviour? |
| 21:42.18 | Blackvel | [TK]D-Fender: do you use then s,n,GotoIf($["${DIALSTATUS}" = "BUSY"]?.... , in combination of ....Playback(Busy) even? wiki suggests to add 3rd s,n,Busy (after Playback(Busy) |
| 21:42.34 | [TK]D-Fender | j_kroon: * likely spawns that as a secondary process so that it doesn't jam things up |
| 21:42.52 | Blackvel | now I can remember pri stuff. I set PRI_CAUSE before with bristuff (which I dropped for patton media gateway) |
| 21:43.17 | [TK]D-Fender | Blackvel: You're asking that on the assumption that I know what the rest of your exten looks like, does, or should do. |
| 21:43.35 | *** join/#asterisk tobias (n=tobias@user-0ce2hp1.cable.mindspring.com) |
| 21:44.16 | [TK]D-Fender | Blackvel: You first asked about checking status. So if you want to react accordingly, thats a separate thing |
| 21:44.30 | *** join/#asterisk jmatthies (n=talaena@168.103.67.196) |
| 21:44.33 | j_kroon | [TK]D-Fender, doesn't look like it from res/res_convert.c |
| 21:45.02 | j_kroon | at least, not for 1.6.1.1 anyway, I'm quickly checking older (1.6.0.9) version of code too. |
| 21:45.12 | [TK]D-Fender | j_kroon: I don't know the specifics, was offering a plausible reasoning for it... |
| 21:45.47 | j_kroon | [TK]D-Fender, ta, your reasoning makes sense, but doesn't match my understanding of the code (my understanding is probably incorrect) |
| 21:46.23 | [TK]D-Fender | j_kroon: Who can tell with that spaghetti ;) |
| 21:46.41 | [TK]D-Fender | goes for more meatballs & parmesan |
| 21:46.54 | *** join/#asterisk freenose (n=freenose@204.97.199.7) |
| 21:47.03 | saxa | [TK]D-Fender: where are you from ? |
| 21:47.08 | j_kroon | well, looking just at res/res_convert.c the handle_cli_file_convert call doesn't look to be dealing with threads in any way. |
| 21:47.22 | saxa | likes too meatballs |
| 21:47.52 | srf21c | gnaws on a slice of pumpernickel bread |
| 21:48.09 | [jmc] | try lasagne :P |
| 21:49.03 | saxa | parmigiana is even better for my taste :) |
| 21:49.05 | [TK]D-Fender | saxa: Sol3, nothern hemisphere, close to the water. and you? |
| 21:49.26 | saxa | slovenija, but leave in Italian city Gorizia |
| 21:49.44 | [jmc] | Italy ftw |
| 21:49.59 | [jmc] | [TK]: lol for the Sol3 :P |
| 21:50.11 | saxa | anyway I need to review the book example, why this thing stopped to connect |
| 21:50.24 | batphone | Texas rules the planet. |
| 21:50.28 | batphone | FYI |
| 21:50.30 | saxa | :) |
| 21:50.45 | saxa | texas music is ok :) |
| 21:51.44 | [jmc] | saxa: "Texas" like the state, not the band :D |
| 21:54.25 | saxa | [jmc]: heheh |
| 21:54.38 | saxa | anyway, let me understand one thing |
| 21:55.00 | [jmc] | about Texas? |
| 21:55.16 | saxa | in the iax.conf on my brazilian box, I need to specify the [context] of the Italian box ? |
| 21:55.26 | tompaw | Hi. |
| 21:55.35 | tompaw | Is there a way to debug dialplan application? |
| 21:55.40 | saxa | I mean, I have a register => statement in it |
| 21:55.56 | saxa | then I have a [context] |
| 21:56.13 | saxa | which should have data of the italian box correct ? |
| 21:56.35 | [jmc] | well, the 'register', mainly |
| 21:56.50 | profXavier | tfrew: how do you add trixbox into the discussion? |
| 21:56.53 | [jmc] | no wait |
| 21:57.02 | [jmc] | saxa: can you show the configuration file? |
| 21:57.12 | [jmc] | I'm not sure about that |
| 21:58.28 | tfrew | not trixbox itself, the rpm package they provide for polycom sip phones |
| 21:58.40 | tfrew | i was refering to a "working example" of the xml files |
| 21:58.47 | profXavier | um |
| 21:58.52 | profXavier | sure, ok |
| 21:58.58 | profXavier | thanks for your input/help |
| 21:59.06 | tfrew | your not welcome |
| 21:59.08 | tfrew | have a cupcake |
| 21:59.32 | tfrew | and a glass of chloroform |
| 22:00.03 | [TK]D-Fender | profXavier: Wow... and you thought I used to fly off on a bender ;) |
| 22:00.15 | j_kroon | ugly. there has to be an easier (better) way than to do while ls -l /proc/$(</var/run/asterisk/asterisk.pid)/fd/ | grep -q "${filename}$"; do sleep 0.1; done |
| 22:01.34 | [jmc] | j_kroon: what are you trying to do? |
| 22:02.07 | *** join/#asterisk lucasb (n=bussey@office.telifon.com) |
| 22:03.04 | Blackvel | I am bit confused about Dial and & syntax. I use it to call snom 370 and snom m3 (the 2nd with Local and & syntax. |
| 22:03.20 | Blackvel | what happens: when snom 370 is busy, it says, busy and stops ringing (hangup) |
| 22:03.40 | Blackvel | but the 2nd channel keeps starts ringing |
| 22:05.01 | Blackvel | checking with ,GotoIf($["${DIALSTATUS}" = "BUSY" seems not to help to stop ringing the 2nd device |
| 22:05.15 | saxa | [jmc]: of course I can |
| 22:05.21 | saxa | let me paste it |
| 22:05.23 | Blackvel | when i am talking on my desktop phone snom 370, i dont need to have my mobile M3 ring |
| 22:05.55 | j_kroon | [jmc], issue an asterisk -rx "file convert infile outfile" and then wait for outfile to be complete. |
| 22:05.55 | tfrew | gives profXavier a hug |
| 22:06.03 | *** part/#asterisk tfrew (n=tfrew@office.neteasyinc.com) |
| 22:06.17 | [TK]D-Fender | Ok, off to martial arts, back in a few.... |
| 22:07.36 | *** join/#asterisk rednul (n=rednul@host-98-127-11-104.bln-mt.client.bresnan.net) |
| 22:07.37 | saxa | [jmc]: http://pastebin.com/m64fd9d0 |
| 22:07.57 | saxa | [TK]D-Fender: enjoy |
| 22:08.33 | rednul | could someone recommend a way of adjusting the volume/gain of an audio stream (for use as hold music?) it looks like mpg123 use to support that, but not anymore... |
| 22:09.27 | saxa | [jmc]: this is on my brazilian box |
| 22:09.51 | saxa | on the italian one I have only few lines which I will paste now, please wait |
| 22:10.02 | tompaw | No application 'MYSQL' for extension... :-( |
| 22:10.12 | tompaw | Hm... doesn't it get automatically installed with asterisk-addons? |
| 22:11.31 | saxa | [jmc]: http://pastebin.com/m706f6b4a <- this is on my italian box |
| 22:12.00 | kuku1 | I have a queue, and wish to dial an outside extension ( cell phone via sip ), so in 1.2 I created an entry in that same context, and it worked, but in 1.6, I get an error. [Jun 16 15:32:18] WARNING[13877]: chan_sip.c:4526 create_addr: No such host: 111 |
| 22:12.20 | kuku1 | [TK]D-Fender: are you around ? |
| 22:12.31 | [jmc] | kuku1: no he's not |
| 22:12.45 | pfn | ugh, why doesn't app_voicemail delete empty/silent messages by default, annoying |
| 22:12.53 | [jmc] | saxa: let me see that |
| 22:13.14 | tompaw | Guys? How do I install MYSQL application? I have *-addons installed. |
| 22:13.23 | KyleK | pfn: nobody has coded up a good way to identify the empty/silent messages? |
| 22:15.16 | [jmc] | saxa: nothing, I'm not able to help you... sorry :) |
| 22:15.20 | saxa | [jmc]: strangely that thing was working |
| 22:15.51 | saxa | [jmc]: its ok, I will re-read the chapter in the book |
| 22:16.59 | pfn | KyleK, well, I wrote a patch for it a long time ago, but I didn't forward-fit it to 1.0 at the time |
| 22:17.03 | pfn | KyleK, https://issues.asterisk.org/view.php?id=2264 |
| 22:17.10 | pfn | I swear, I need to turn off voicemail |
| 22:17.20 | pfn | because I get too many silent messages making it worthless |
| 22:17.55 | *** join/#asterisk profXavier (n=MyNick@unaffiliated/neverblue) |
| 22:18.52 | KyleK | are you using the patch yourself? |
| 22:19.45 | pfn | KyleK, my patch was pre-1.0, 2004 |
| 22:19.53 | pfn | I've been using 1.4 recently |
| 22:20.04 | pfn | and I haven't had time to update the patch, previously, yes, I used that patch |
| 22:20.16 | pfn | needs to get around to re-implementing it |
| 22:20.26 | pfn | I don't see how anyone can stand using app_voicemail in its default state |
| 22:20.33 | pfn | gets a lot of assholes calling... |
| 22:20.54 | pfn | morons sit on the phone thinking that a human is screening their call or something and then hang up after some amount of time |
| 22:20.56 | KyleK | oic |
| 22:21.23 | KyleK | whats your voicemail greeting? |
| 22:21.25 | tompaw | app_addon_sql_mysql_.so is there |
| 22:21.35 | tompaw | why isn't the MYSQL application available? |
| 22:21.41 | tompaw | do I have to enable it somehow in the confs? |
| 22:21.49 | pfn | KyleK, just the default, iirc |
| 22:22.01 | KyleK | pfn: change it to "please breathe heavily into the phone so I can justify not listening to my voicemail you dork" |
| 22:23.05 | *** join/#asterisk jeffgus (n=jeffgus@2002:ad33:b504:0:0:0:0:1) |
| 22:23.43 | tompaw | a ha! |
| 22:23.44 | tompaw | Module 'app_addon_sql_mysql.so' was not compiled with the same compile-time options as this v |
| 22:23.47 | tompaw | ersion of Asterisk. |
| 22:24.16 | tompaw | does Asterisk store its compile-time options anywhere? |
| 22:24.27 | pfn | KyleK, not really practically |
| 22:25.12 | russellb | tompaw: yes. include/asterisk/buildopts.h |
| 22:25.23 | *** join/#asterisk Octothorpe (n=Octothor@pdpc/supporter/professional/octothorpe) |
| 22:25.43 | tompaw | russellb: thank you. just to be 100% sure - what version of addons should I use with 1.6.0.6? |
| 22:25.55 | tompaw | I'm using 1.6.0 at the moment |
| 22:26.24 | russellb | there is a 1.6.0.2 out |
| 22:26.30 | russellb | i always say latest with the latest |
| 22:26.45 | tompaw | ok, will give it a try, thank you. |
| 22:26.50 | KyleK | pfn: if people were leaving me blank messages I'd either sass them with the greeting or call them up and ask about it |
| 22:27.40 | *** join/#asterisk jtodd (i=e92254he@ns.fox-den.com) |
| 22:27.40 | *** mode/#asterisk [+o jtodd] by ChanServ |
| 22:28.28 | pfn | KyleK, they're people I don't care to talk to if they can't be bothered to leave a message |
| 22:28.47 | KyleK | so sass them in the greeting? |
| 22:28.56 | pfn | then it affects legitimate callers |
| 22:29.24 | pfn | the "correct" solution is just to drop empty messages |
| 22:29.32 | pfn | but * doesn't do it for whatever reason |
| 22:29.50 | talirk81 | From AGI im sending "EXEC SET VARIABLE __CallID 25" but that variable is not showing up when i use ${__CallID} or ${CallID} any ideas what im doing wrong? |
| 22:30.01 | KyleK | argh, i'm having nothing but trouble with alsa today |
| 22:30.26 | Blackvel | exten => s,1,Dial(SIP/200&SIP/201&LOCAL/90015300) |
| 22:30.31 | Blackvel | i am using something like that |
| 22:30.40 | seanbright | talirk81: not sure what your specific problem is, but you never access variables with an underscore in the front |
| 22:30.42 | Blackvel | to make the 2nd tech call delayed |
| 22:31.04 | seanbright | talirk81: the underscores "go away" after you call Set() |
| 22:31.15 | talirk81 | ok well i tried both ways |
| 22:31.17 | seanbright | right |
| 22:31.17 | kuku1 | [TK]D-Fender: are you around ? |
| 22:31.19 | kuku1 | I have a queue, and wish to dial an outside extension ( cell phone via sip ), so in 1.2 I created an entry in that same context, and it worked, but in 1.6, I get an error. [Jun 16 15:32:18] WARNING[13877]: chan_sip.c:4526 create_addr: No such host: 111 |
| 22:31.27 | seanbright | just letting you know that the underscore version will never work |
| 22:31.32 | talirk81 | but it seems the EXEC SET VARIABLE isnt really setting it. |
| 22:31.42 | *** join/#asterisk jjshoe (n=jjshoe@h69-129-142-83.mdsnwi.tisp.static.tds.net) |
| 22:31.45 | Blackvel | when the 1st tech is busy, dial does not stop the 2nd. i really want to abort the whole dial and go to BUSY handling |
| 22:32.02 | seanbright | talirk81: why are you using EXEC? |
| 22:32.05 | *** join/#asterisk Aiatek (n=munoz@190.94.60.219) |
| 22:32.07 | talirk81 | its in an AGI |
| 22:32.10 | seanbright | talirk81: you should just call SET VARIABLE |
| 22:32.15 | seanbright | (without the EXEC) |
| 22:32.21 | Blackvel | instead of it dails the 2nd phone which is useless when i am talking on the 1st phon |
| 22:32.23 | Blackvel | phone |
| 22:32.40 | KyleK | why bother calling the 2nd phone then? |
| 22:33.17 | Blackvel | is there any way to tell the DIAL command to hangup both call legs and switch to DIALSTATUS BUSY handling`? |
| 22:33.31 | Blackvel | KyleK: it is a delayed calling of m3 |
| 22:33.41 | Blackvel | if i pickup on the 1st phone, it stops |
| 22:33.59 | Blackvel | if i dont pickup, snom m3 will ring after 10-15 seconds (when i am not on my desk) |
| 22:34.04 | Blackvel | at my desk |
| 22:34.32 | talirk81 | seanbright, thanks that did it, wierd that EXEC wouldnt also do it, had to make a new myAGI::SetVariable() function in my php classes |
| 22:34.39 | Blackvel | i even wonder that it hangs up the 2nd tech |
| 22:34.44 | KyleK | why dont you dial the first phone for 10-15 seconds, then if its busy go to somewhere else, but if its just ignored rin the snom |
| 22:34.46 | Blackvel | when i pickup the first |
| 22:34.55 | seanbright | talirk81: EXEC calls applications |
| 22:34.59 | seanbright | talirk81: so you'd have to do: |
| 22:35.14 | seanbright | talirk81: EXEC Set __CallId 25 |
| 22:35.15 | seanbright | i think |
| 22:35.22 | KyleK | like 1,Dial(me,15) 2,Dial(me2,10) 101,Dial(someoneelse) |
| 22:36.16 | *** join/#asterisk rue_mohr (n=rue@24.207.122.10) |
| 22:36.16 | seanbright | talirk81: but best to just use SET VARIABLE anyway |
| 22:37.06 | Blackvel | KyleK: good question. I think I did that in october 2008 just because I wanted to prevent displaying too many "missed calls" |
| 22:37.34 | Blackvel | when both phones ring (they will after 10 secs) and I pickup any of them, no missed calls will be displayed (I think) |
| 22:37.58 | KyleK | displaying missed calls where? on these phones? |
| 22:38.24 | Blackvel | yepp |
| 22:38.33 | Blackvel | well i have to check |
| 22:38.48 | Blackvel | again |
| 22:38.51 | Blackvel | long time ago |
| 22:41.24 | pfn | ls |
| 22:43.12 | Blackvel | KyleK: funny I can not tell the benefit anymore |
| 22:43.35 | Blackvel | the calls always show on 1st snom as missed (no matter if I pick the call on the 2nd) |
| 22:44.07 | Blackvel | on the 2nd it only shows as missed when I do not pickup the call within ~10 secs on the 1st (as the 2nd starts ringing after 10) |
| 22:44.41 | Blackvel | so there seems not to be much difference i use 2x dial in sequence |
| 22:44.49 | Blackvel | if I |
| 22:45.04 | *** join/#asterisk aliverius (n=aliveriu@athedsl-386219.home.otenet.gr) |
| 22:45.36 | Blackvel | the negativ effect stays |
| 22:45.49 | Blackvel | If I choose to add xlite (3rd) |
| 22:46.08 | Blackvel | I want to get BUSY for the whole dial if any of the three techs run into BUSY |
| 22:46.54 | Blackvel | I just think I used dial with the & techs as I had three before (and removed xlite in the meantime) |
| 22:47.04 | Blackvel | but probably want to add it back at some time |
| 22:47.19 | *** join/#asterisk Iamnach0 (i=Iamnacho@ip174-70-137-120.ks.ks.cox.net) |
| 22:47.34 | [jmc] | see you tomorrow guys |
| 22:47.36 | pfn | looks at how to re-integrate his silent voicemail patch |
| 22:47.41 | [jmc] | happy * |
| 22:47.45 | *** part/#asterisk [jmc] (n=[jmc]@93-45-238-238.ip104.fastwebnet.it) |
| 22:47.47 | *** join/#asterisk mltlnx (n=mltlnx@cpe-68-175-38-221.nyc.res.rr.com) |
| 22:48.29 | *** join/#asterisk dacs (n=chatzill@unaffiliated/dacs) |
| 22:48.37 | dacs | howdy folks |
| 22:49.11 | dacs | i have a question, am sure you guys are familiar with magic jack? |
| 22:49.37 | *** join/#asterisk joobie (n=joobie@mx01.anric.com.au) |
| 22:50.10 | pfn | sounds neat, a usb-based ata |
| 22:50.14 | pfn | + service |
| 22:50.31 | dacs | it connect thru usb and you can use it to call out! is there is anyway can config my * to use that |
| 22:51.36 | dacs | i think it is only a 1 channel thu |
| 22:53.01 | pfn | did app_voicemail change from 1.4 to 1.6? |
| 22:53.18 | pfn | I don't want to have to re-submit a patch for 1.6 since I'm running 1.4... |
| 22:53.58 | Blackvel | found a reasonable explanation: answer "wins", busy on 1st "wins" over others: http://www.mail-archive.com/asterisk-users@lists.digium.com/msg200414.html |
| 22:54.12 | Blackvel | seems that it didn't exist before and that this features is not SLA |
| 22:54.23 | Blackvel | will search around if its in 1.4/1.6 |
| 22:55.29 | *** join/#asterisk mltlnx (n=mltlnx@cpe-68-175-38-221.nyc.res.rr.com) |
| 22:56.19 | KyleK | i guess i could diff my 1.4 and 1.6 copys |
| 22:56.20 | *** join/#asterisk andresmujica (n=andresmu@ubuntu/member/andresmujica) |
| 22:56.46 | KyleK | yea its different |
| 22:57.33 | pfn | KyleK, how different is it? |
| 22:59.02 | *** join/#asterisk bijit (i=1000@190.241.15.48) |
| 22:59.32 | KyleK | they renamed some variables like message_exists to msg_exists, not sure which is which as the diff is 8809 lines |
| 22:59.41 | pfn | ugh, too big |
| 22:59.42 | pfn | heh |
| 22:59.57 | bijit | is it possible to connect a LG Nortel IPLDK 300/300E with asterisk? |
| 23:00.05 | bijit | anyone done this before? |
| 23:00.41 | Blackvel | kylek: is there any way to check if a call is ongoing and a device (2nd dial) is busy? |
| 23:01.00 | Blackvel | if I am going your way with two sequential DIAL commands it stays the same |
| 23:01.08 | KyleK | i dunno im kind of noobtastic here ;) |
| 23:01.28 | KyleK | well doesn't a busy increment it by 100? so 1, 101 in the uh |
| 23:01.34 | KyleK | priorities? |
| 23:01.35 | Blackvel | if I use the 2nd phone for a ongoing call, the first dial to 1st snom phone is free |
| 23:01.37 | Aiatek | <bijit> which protocol support? |
| 23:01.57 | Blackvel | so it rings my first phone even i am busy on the 2nd |
| 23:02.05 | Blackvel | not a very optimal solution either |
| 23:02.09 | KyleK | Blackvel: so what you really want is a function to check the status of your phones |
| 23:02.34 | Blackvel | is there something like a callgroup? |
| 23:02.53 | Blackvel | if all phones in the group are not busy let it ring |
| 23:03.04 | Blackvel | if any of them is busy signal BUSY? |
| 23:03.21 | bijit | Aiatek: I am looking @ the nortel page to see if it supports sip |
| 23:04.08 | Aiatek | bijit you can do it with pri too |
| 23:04.30 | kuku1 | where can I find info on queues and version 1.6, seems a lot has changed since 1.2 and I can't get the right syntax/functions. Please provide a link to some docs.... |
| 23:05.16 | Aiatek | bijit or with fxs/fxo ports it depends what you really want |
| 23:05.32 | bijit | Aiatek: I am looking for all the possible ways..and asterisk will connect to it remotely..(VPN, Dedicated Line) |
| 23:05.33 | *** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 23:06.46 | dacs | so can i use * with magic jack to dial out ? did anyone tried that |
| 23:08.34 | dacs | bijit: 5E support SIP :) |
| 23:09.56 | bijit | dacs: 300E/300? |
| 23:10.45 | dacs | 300 |
| 23:11.18 | Aiatek | <bijit> you need at least firmware |
| 23:11.18 | Aiatek | 3.7 |
| 23:11.27 | bijit | so I can connect 300 with * via sip? |
| 23:11.30 | dacs | bijit: but if you can afford to buy a 5ESS , then you don't belong here :) |
| 23:12.07 | pfn | KyleK, are you running 1.4 or 1.6? |
| 23:12.12 | Aiatek | it supports sip in tha version of firmware for ipLDK-20/50/100/300 |
| 23:12.20 | pfn | KyleK, if 1.4, would you like to try out my silent vm removal patch? |
| 23:12.23 | Aiatek | and h323 too |
| 23:12.40 | bijit | dacs: its a client that has multiple 300 and now want * and the want to interconnect |
| 23:13.00 | bijit | guess they couldn't afford 5ESS |
| 23:13.07 | Aiatek | <bijit> check the firmware version |
| 23:13.15 | bijit | Aiatek: sure |
| 23:14.38 | Aiatek | right now im working in integration with asterisk and 3com nbx5000 |
| 23:16.31 | KyleK | pfn: I'm actually swapping back and forth, I had hacked 1.4's app_voicemail to email me an mp3, but today I'm using chan_mobile on 1.6 |
| 23:16.42 | KyleK | so hopefully nothing important comes in for voicemail ;) |
| 23:17.09 | pfn | KyleK, didn't like the .wav it emailed you? |
| 23:17.50 | KyleK | cant play them in gmail |
| 23:18.14 | pfn | eh? |
| 23:18.27 | pfn | oh, gmail has a flash component for playing mp3? |
| 23:18.27 | pfn | heh |
| 23:18.32 | KyleK | if i email myself a .mp3 i can use the flash player thing, yea |
| 23:18.48 | KyleK | 11.025 khz btw |
| 23:19.13 | pfn | KyleK, couldn't you specify format in voicemail.conf? |
| 23:19.33 | KyleK | not in the version of 1.4 i was using |
| 23:19.54 | *** join/#asterisk sah-work (n=Bawbatos@65-119-47-100.dia.static.qwest.net) |
| 23:20.22 | pfn | what'd it do you said format=mp3? |
| 23:21.25 | sah-work | question. we have an asterisk box with a pri however we are thinking we would like to outsource the phone system and all that goes with it. i need to have access for our current sip phones as well as 800 numbers and numbers in the uk. anyone have any recomendations on providers |
| 23:21.38 | pfn | hmm, chan_mobile sounds interesting, too bad my * box since in a cabinet in the garage |
| 23:21.56 | KyleK | pfn: it fell back on whatever the default is/was |
| 23:22.11 | pfn | KyleK, funky |
| 23:22.29 | KyleK | i'll try with 1.6 really quickly |
| 23:23.23 | lanning | pfn, you could run a second asterisk instance on a quiet machine in the house... |
| 23:23.39 | *** join/#asterisk chandoo (n=chandoo@ool-4353b978.dyn.optonline.net) |
| 23:23.41 | pfn | lanning, nah |
| 23:23.46 | pfn | http://paste.hanhuy.com/asterisk-voicemail-silence-drop |
| 23:23.56 | pfn | there's my diff |
| 23:24.04 | pfn | seems too easy... |
| 23:24.13 | pfn | twiddles thumbs while * rebuilds |
| 23:24.37 | KyleK | hrm my voicemail is configured wrong for 1.6, i'll have to look later |
| 23:26.25 | pfn | tries out silent vm drop |
| 23:26.36 | KyleK | [Jun 16 16:28:13] WARNING[24430]: file.c:1131 ast_writefile: No such format 'mp3' |
| 23:26.39 | KyleK | [Jun 16 16:28:13] WARNING[24430]: app.c:809 __ast_play_and_record: Error creating writestream '/home/asterisk/var/spool/asterisk/voicemail/condo/1000/tmp/KN4Cxc', format 'mp3' |
| 23:26.47 | KyleK | i dont think i've seen that error before |
| 23:27.01 | KyleK | so you might be on to somethng |
| 23:27.12 | *** join/#asterisk hi365 (n=hi365@94.159.177.112) |
| 23:28.00 | pfn | probably just updated to not silently fall back to default |
| 23:28.20 | pfn | there's no format_mp3 ... |
| 23:28.33 | pfn | google's pretty stupid if it only supports mp3 and not wav though |
| 23:28.58 | KyleK | well they'd have to encode wav -> mp3 on the fly |
| 23:29.17 | pfn | why? the control should just support wav |
| 23:29.27 | pfn | or if not, the control should support transcoding it on the fly |
| 23:29.42 | KyleK | well its using the flash thing, not requesting a real control |
| 23:29.53 | pfn | well, "the control" = flash in this case |
| 23:30.00 | pfn | and flash should readily support most wav formats |
| 23:30.40 | KyleK | well even then, the khz isn't right |
| 23:30.50 | *** join/#asterisk DarkRift (n=dark@65.92.167.15) |
| 23:32.04 | pfn | that should be handled by a player naturally |
| 23:33.36 | pfn | twiddles thumbs while the 1.6.2 branch syncs down |
| 23:35.40 | leifmadsen | yay 1.6.2 testing! :) |
| 23:36.12 | pfn | testing? I just want to make sure my patch merges cleanly :p |
| 23:36.48 | leifmadsen | \o/ yay patches! |
| 23:37.00 | pfn | https://issues.asterisk.org/view.php?id=2264 |
| 23:37.04 | pfn | I'm reviving this old ass bug |
| 23:38.06 | KyleK | astspooldir => /var/spool/asterisk <-- is this as close as I get to specifying where the spool/voicemail dir is? |
| 23:39.16 | leifmadsen | I have a script that is running on a dir containing (originally) 63GB worth of recordings -- I'm deleting everything older than 90 days. It has been running since about 11:30am (it is now 7:39pm) |
| 23:39.25 | leifmadsen | I had no idea it was going to take several HOURS |
| 23:39.35 | KyleK | must be a lot of files |
| 23:39.38 | leifmadsen | currently at 36GB |
| 23:39.43 | leifmadsen | ya -- a lot of small files it seems |
| 23:39.53 | pfn | damn, it doesn't merge cleanly into 1.6 :( |
| 23:40.00 | leifmadsen | pfn: not surprised :) |
| 23:44.26 | KyleK | I'm not sure if its good or bad that I can find answers to what im looking for by looking through thousands of lines of code faster than i can google up something |
| 23:46.28 | pfn | depends on what you're looking for |
| 23:46.35 | pfn | it's good that you're skilled enough... |
| 23:47.15 | KyleK | snprintf(VM_SPOOL_DIR, sizeof(VM_SPOOL_DIR), "%s/voicemail/", ast_config_AST_SPOOL_DIR); |
| 23:47.18 | KyleK | true |
| 23:47.21 | *** join/#asterisk layne (n=layne@ool-44c0048f.dyn.optonline.net) |
| 23:47.37 | dacs | i have a question, since x-lite has 2 lines, why i can't add another account in x-lite?! |
| 23:48.21 | KyleK | its the free version |
| 23:48.35 | dacs | errr |
| 23:48.41 | pfn | leifmadsen, looks like it was minor things that made it not patch cleanly |
| 23:49.00 | KyleK | dacs: they gotta get people to give them money somehow ;) |
| 23:49.12 | dacs | KyleK: i am using 3.0 47546 |
| 23:49.56 | drmessano | dacs: Always worked for me |
| 23:50.21 | dacs | drmessano: you have 2 account on the free version |
| 23:51.05 | pfn | KyleK, so... can you try it out my patch against 1.6? :) |
| 23:51.37 | KyleK | hrm my copy of x-lite only allows 1 sip registration |
| 23:51.49 | pfn | http://paste.hanhuy.com/asterisk1.6-silence-patch |
| 23:52.24 | layne | the new version of x-lite restricts free users to 1 acct |
| 23:52.54 | drmessano | it does? |
| 23:53.04 | dacs | layne: its okay, i am booting my laptop and i will use Eikga |
| 23:53.05 | KyleK | i knew i should have kept the copy i downloaded like 5 years ago ;) |
| 23:53.10 | mmlj4 | who uses proprietary crap? go get something else, man |
| 23:54.09 | dacs | mmlj4: yeah, as soon as i get better in *. i finally got the hard copy of the book |
| 23:54.16 | dacs | step-by-step |
| 23:54.39 | dacs | :) |
| 23:56.23 | layne | drmessano, yes it does |
| 23:56.34 | layne | the old one let you have a few iirc |
| 23:57.25 | dacs | [Jun 15 23:59:09] NOTICE[3109]: chan_sip.c:14703 handle_request_invite: Call from '2000' to extension '1000' rejected because extension not found. |
| 23:57.37 | dacs | ^^ but i have extention 2000 |
| 23:58.18 | drmessano | extension 1000 is not found |
| 23:58.28 | drmessano | from _ to |
| 23:58.51 | dacs | drmessano: its there |
| 23:59.12 | dacs | am not arguing , but based on my understanding |
