| 00:04.29 | *** join/#asterisk coppice (n=chatzill@102.204.17.210.dyn.pacific.net.hk) |
| 00:04.38 | *** join/#asterisk saftsack (n=oliver@pD9E07DA2.dip.t-dialin.net) |
| 00:08.18 | Siya | couldn't sleep |
| 00:08.22 | Siya | tx fujin_ |
| 00:08.39 | Siya | http://www.voip-info.org/wiki/index.php?page=Asterisk+func+cut |
| 00:08.47 | fujin_ | np |
| 00:11.00 | *** join/#asterisk techie (n=techie@adsl-76-214-9-1.dsl.lsan03.sbcglobal.net) |
| 00:11.02 | *** part/#asterisk techie (n=techie@adsl-76-214-9-1.dsl.lsan03.sbcglobal.net) |
| 00:20.05 | *** join/#asterisk Hadi- (n=Hadi@CPE001310492769-CM001225e00576.cpe.net.cable.rogers.com) |
| 00:20.49 | Hadi- | Hello everyone... im having an issue with DTMF... anyone available? |
| 00:23.08 | saftsack | Hadi-, tell us more please ... :-p |
| 00:23.41 | Hadi- | well I have a SIP PRI |
| 00:23.45 | Hadi- | that is pointing to a Cisco 2800 series router |
| 00:23.58 | Hadi- | the DTMF is set to RFC2833 as default from my provider |
| 00:24.13 | Hadi- | when I set my trunk in asterisk to that dtmf.. its not working |
| 00:24.25 | Hadi- | when I call the DID's... incoming DTMF works fine |
| 00:24.29 | Hadi- | its only outgoing |
| 00:24.41 | Hadi- | that is not working |
| 00:24.47 | JT | you do NOT have a SIP PRI |
| 00:24.51 | JT | please don't say it ever again |
| 00:25.02 | Hadi- | hehe |
| 00:25.07 | Hadi- | what do you mean |
| 00:25.33 | JT | we've been through this haven't we? |
| 00:25.42 | JT | i mean a SIP PRI really ABSOLUTELY DOES NOT EXIST |
| 00:25.48 | macTijn | lol |
| 00:26.31 | macTijn | it's probably sales-speak for "you can do 30 calls simultaneously, it'll cost you roughly 2 mbits/sec of traffic" |
| 00:26.39 | JT | heh |
| 00:26.58 | Hadi- | okay... |
| 00:27.03 | Hadi- | so anyways.. thats my issue |
| 00:27.25 | macTijn | uhuh |
| 00:27.29 | JT | so it's SIP |
| 00:27.31 | JT | and no PRI |
| 00:27.32 | JT | okay |
| 00:27.48 | macTijn | JT: it seems so, Hadi- is talking about RFC2833 |
| 00:27.55 | Hadi- | we do have reinvite enabled (media flow-through <-- cisco dial peer command) |
| 00:28.00 | Hadi- | on the dial peer |
| 00:28.08 | Hadi- | that we are sending calls out to... |
| 00:28.30 | macTijn | RFC2833, how is that again ? |
| 00:28.48 | macTijn | ah, RTP payload |
| 00:28.55 | macTijn | sucks to be you, man. |
| 00:29.07 | macTijn | did you try any other forms of DTMF ? |
| 00:29.37 | *** join/#asterisk ming_zym (i=ming_zym@nat/yahoo/x-af6ad54861a9edc1) |
| 00:30.01 | macTijn | like inband or INFO messages ? |
| 00:30.25 | JT | macTijn: what is wrong with rfc2833 |
| 00:31.05 | Hadi- | well |
| 00:31.24 | Hadi- | rfc2833 is not working for us with g711ulaw |
| 00:31.44 | macTijn | JT: it might not work everywhere |
| 00:31.55 | fujin_ | Hadi-: use inband on outgoing |
| 00:31.58 | fujin_ | makes happytime |
| 00:32.00 | macTijn | uhuh |
| 00:32.04 | macTijn | mostly it does |
| 00:32.44 | JT | macTijn: it works for most people |
| 00:33.01 | JT | RFC2833 is the best choice unless dealing with defective SIP devices |
| 00:33.21 | macTijn | JT: I believe he was talking about Cisco |
| 00:33.30 | JT | Hadi-: perhaps you have a defective provider |
| 00:33.40 | macTijn | and I haven't seen any great work from them regarding signalling |
| 00:34.04 | Hadi- | JT: perhaps MCI is defective :) |
| 00:34.21 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@adsl-68-93-83-7.dsl.hstntx.swbell.net) |
| 00:34.40 | macTijn | Hadi-: I've seen crazier things :) |
| 00:35.20 | JT | Hadi-: probably |
| 00:35.34 | JT | Hadi-: most telcos would prefer to speak a less defective protocol, like H.323 |
| 00:36.22 | obnauticus | Damnit, is ANYONE ehre having problemns with VoipJet and the lack of MD5 string they are giving me |
| 00:36.40 | obnauticus | and does anyone here have a contact number for them, because they aren't replying to my email, and I cannot see the password to auth |
| 00:37.02 | obnauticus | It's relaly pissing me off, I paid them 20$s and have no service because i don;'t know my F*cking secret :\ |
| 00:37.38 | JT | 20$s? |
| 00:37.43 | obnauticus | Ya.. |
| 00:37.45 | obnauticus | I just wanna try it out |
| 00:37.48 | obnauticus | They allow that amount |
| 00:37.51 | JT | what does 20$s mean? |
| 00:37.55 | obnauticus | $20 |
| 00:37.57 | obnauticus | fine :\ |
| 00:38.07 | obnauticus | I'm used to saying twenty dollar's |
| 00:38.12 | obnauticus | It should be written the same way. |
| 00:38.16 | JT | incorrect |
| 00:38.24 | obnauticus | Do you say dollar's twenty? |
| 00:38.35 | JT | it should be written in a standard format that is nice for writing |
| 00:38.41 | JT | written english is not the same as spoken |
| 00:38.46 | obnauticus | We aren't writing it though :\ |
| 00:38.56 | obnauticus | Do you have a contact number for Voipjet? |
| 00:39.02 | JT | if you want to transcribe how you speak, type "twenty dollars" |
| 00:39.10 | JT | no, i've never dealt with them |
| 00:39.13 | JT | we are writing |
| 00:39.18 | JT | this looks like text to me |
| 00:39.18 | obnauticus | Nuts. |
| 00:39.30 | obnauticus | I dunno about you but I am typing. |
| 00:39.58 | JT | typing creates written english on a screen |
| 00:40.07 | fujin_ | o_0 |
| 00:40.20 | obnauticus | Ah yes, but AM i writing? |
| 00:40.25 | JT | i don't think it's that wrong to call it writing |
| 00:40.27 | JT | sure |
| 00:40.32 | obnauticus | on a very technical level, what `action word' am I using? |
| 00:40.36 | obnauticus | (ie. verb) |
| 00:41.03 | fujin_ | You're typing. |
| 00:41.20 | RypPn | pedantry I believe its called |
| 00:42.38 | JT | nothing worse than people typing 383$ etc |
| 00:42.39 | macTijn | aw man |
| 00:42.41 | JT | and $s wtf |
| 00:42.51 | obnauticus | JT, how isn't it logical? |
| 00:42.53 | macTijn | what is this for a discussion on #asterisk ? |
| 00:43.02 | obnauticus | In the English language, if you can READ it most of the time you can speak it. |
| 00:43.08 | macTijn | "$" isn't a drop-in replacement for "dollar" |
| 00:43.15 | obnauticus | I realise that |
| 00:43.19 | obnauticus | it's to label the amount. |
| 00:43.28 | JT | obnauticus: it's illogical to create your own standards for writing english and ignore standards that have been around for hundreds of years |
| 00:43.36 | macTijn | ok |
| 00:43.39 | obnauticus | That is why english sucks though |
| 00:43.41 | obnauticus | :\ |
| 00:43.43 | macTijn | JT: stop flamebaiting :) |
| 00:43.46 | JT | nope |
| 00:43.48 | RypPn | why not, the americans did it with english...lol |
| 00:43.49 | JT | $5.73 |
| 00:43.53 | JT | very logical |
| 00:43.55 | fujin_ | macTijn: this is #asterisk |
| 00:43.56 | JT | RypPn: true |
| 00:44.03 | macTijn | fujin_: so very true... |
| 00:44.03 | fujin_ | not a topical channel, whatever. |
| 00:44.16 | fujin_ | infact, most of the days consist of 'phone' discussions |
| 00:45.07 | dr1 | not being able to use numbers on irc, oh dear |
| 00:45.56 | dr1 | where exactly is the $ appropriate |
| 00:46.15 | dr1 | ill write it down so I personally wont make the mistake |
| 00:46.47 | JT | the symbol preceeds the numeral |
| 00:46.51 | obnauticus | Unlike some Asian languages, English is written how it is spoken. |
| 00:47.11 | obnauticus | If you can read english, you can speak it, and visa versa. |
| 00:47.19 | RypPn | that would depend on the dialect |
| 00:47.19 | obnauticus | How the fuck does Number $s work |
| 00:47.26 | obnauticus | RypPn lol |
| 00:47.31 | obnauticus | i mean |
| 00:47.35 | obnauticus | $ number** |
| 00:47.40 | JT | it's not your job to redefine english |
| 00:47.44 | JT | because it reads properly |
| 00:47.48 | fujin_ | x$s is just wrong |
| 00:47.56 | dr1 | its your job to enforce it though right |
| 00:47.57 | obnauticus | I understand that, but is it logacal? Being a regular computer user you should understand the concept. |
| 00:48.12 | JT | yes, it's highly logical |
| 00:48.14 | fujin_ | the '$' symbol implies that the following numbers are not just numbers, but a representation of $CURRENCY |
| 00:48.15 | dr1 | english teacher in another life? ;0 |
| 00:48.36 | fujin_ | in most cases (i.e.; not perl, php, etc.) |
| 00:48.37 | RypPn | I'd go with fujin_ 's last statement |
| 00:48.43 | dr1 | I dunno, looks more like a PHP variable to me :) |
| 00:48.52 | fujin_ | to which my second comment applied |
| 00:48.54 | RypPn | to raise your expection as to what to expect |
| 00:49.03 | obnauticus | I didn't ever say it was incorrect to write it $20, but it makes no sense. |
| 00:49.04 | dr1 | I just made up my own word, by the way |
| 00:49.12 | obnauticus | How are pounds written? |
| 00:49.13 | JT | it makes perfect sense |
| 00:49.15 | JT | it's neat |
| 00:49.18 | obnauticus | (pounds)(Numeral) ? |
| 00:49.20 | fujin_ | yes |
| 00:49.24 | JT | and works well when you have lots of dollar figures |
| 00:49.26 | RypPn | ?20 |
| 00:49.38 | JT | all the dollar symbols align vertically on a balance sheet |
| 00:49.39 | obnauticus | well I guess if it's an intl. standard it makes sense, since other languages could have come up with the concept |
| 00:49.40 | *** join/#asterisk blitzrage (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 00:49.40 | *** mode/#asterisk [+o blitzrage] by ChanServ |
| 00:49.41 | fujin_ | In japan, it's NUMERAL 'yen' |
| 00:49.51 | obnauticus | That's japan, they also have leet internets. |
| 00:49.54 | fujin_ | I'm not familiar with Chinese, but I assume it'd be the same. |
| 00:49.55 | JT | and you can have more than 2 decimal places of accuracy |
| 00:50.05 | obnauticus | And their average IQ is probably 69pts smarter than the average American IQ |
| 00:50.09 | obnauticus | That is all the proof we need LOL |
| 00:50.17 | fujin_ | Yes, well. America |
| 00:50.19 | fujin_ | =fail |
| 00:50.21 | JT | yes americans have made retarded adjustments to english |
| 00:50.22 | obnauticus | Word. |
| 00:50.24 | RypPn | thats why they eat cats of course |
| 00:50.39 | fujin_ | Dog meat is preferable to Cat |
| 00:50.40 | fujin_ | afaik |
| 00:50.43 | obnauticus | JT my adjustment may be retarded if it's an international standard, but since Japan is doing it, it's fine. |
| 00:50.43 | JT | stupid misconception |
| 00:51.01 | JT | why do people from western countries think all asians eat cats and dogs? |
| 00:51.03 | fujin_ | obnauticus: well, if you want to go around saying x yen, you'll be correct |
| 00:51.10 | obnauticus | lol |
| 00:51.10 | fujin_ | x $s is not |
| 00:51.19 | RypPn | why not? others eat frogs and snails |
| 00:51.28 | obnauticus | When I don't want to get someone a Gift Card for a lot of money I just convert the amount to Lira. |
| 00:51.28 | JT | obnauticus: no, you must use the standard pertaining to a currency |
| 00:51.30 | obnauticus | It works every time. |
| 00:51.31 | *** join/#asterisk Freman (n=freman@brdr-gw-01.benon.com) |
| 00:52.07 | Freman | ok, so I'm a little troubled, when I first boot asterisk and the phones they work fine, after a little while the phones are no longer able to ring external numbers, only internal extensions |
| 00:52.09 | fujin_ | yes, the regional standard defines the numerical/currency notations |
| 00:52.26 | fujin_ | Freman: is asterisk handling the outgoing numbers? |
| 00:52.27 | JT | also, yen normally doesn't have decimals |
| 00:52.37 | fujin_ | Indeed. |
| 00:52.39 | Freman | verboseness shows Call from '' to extension '0405000111' rejected because extension not found. |
| 00:52.41 | JT | in any case, that's its standard |
| 00:52.58 | fujin_ | Freman: are outgoing calls being sent into the correct context? |
| 00:52.58 | Freman | so asterisk is seeing the call |
| 00:53.18 | fujin_ | asterisk is trying to Dial(SIP/0405000111) like, without an outbound connection |
| 00:53.20 | Freman | fujin_, well, they work right off the bat, it's after a while they stop |
| 00:53.27 | Freman | which is why I'm very confused |
| 00:53.27 | fujin_ | you're doing it wrong |
| 00:53.34 | fujin_ | pastebin the context configured for your phones |
| 00:53.52 | JT | Freman: pray tell how you access outside numbers |
| 00:54.04 | Freman | over an iax trunk |
| 00:54.12 | fujin_ | ~pb |
| 00:54.13 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 00:54.15 | fujin_ | less talk, more paste |
| 00:54.35 | dr1 | pastebin, is that english |
| 00:54.57 | JT | it's a proper noun |
| 00:55.00 | fujin_ | Yes. It's a portmanteau |
| 00:55.00 | JT | so yes |
| 00:55.06 | fujin_ | (paste) (bin) |
| 00:55.07 | dr1 | oh ok |
| 00:55.10 | fujin_ | A bin to place pastes in. |
| 00:55.28 | fujin_ | Besides, hackers (read: how to be a hacker) have always had an affinity for creating words which did not previous exist |
| 00:55.30 | obnauticus | What a Failure, I emailed voipjet a week ago, requesting my MD5 hash, they sitll haven't replied, I find it ironic in this case the support that can `always be reached' is FASTsupport@voipjet.com |
| 00:55.32 | fujin_ | see: Internet |
| 00:55.43 | dr1 | so why didnt you just say that, see now this is confusing me, with all this proper use of language going around |
| 00:55.46 | dr1 | :p |
| 00:56.12 | JT | obnauticus: why do you need your MD5 hash? |
| 00:56.21 | fujin_ | I guess, by dictionary definition, that 'pastebin' is *not* english. |
| 00:56.23 | obnauticus | To authenticate on their services. |
| 00:56.27 | JT | err |
| 00:56.29 | obnauticus | They salt an MD5 hash and use it as your password. |
| 00:56.31 | JT | the password will do fine |
| 00:56.35 | obnauticus | No, it doesn't |
| 00:56.37 | obnauticus | it's random |
| 00:56.47 | obnauticus | i tried creating an MD5 hash with my password and putting it in there, still got rejected. |
| 00:56.53 | obnauticus | Their page says it should show up, and it's not |
| 00:56.55 | JT | so what you really want is your password |
| 00:56.55 | Freman | http://pastebin.ca/794320 |
| 00:57.12 | obnauticus | I tried plaintext and putting in my account password in, which didn't work. |
| 00:57.12 | JT | because asterisk takes plaintext passwords, and hashes them for you on auth |
| 00:57.25 | JT | then you will need to ask them for your plaintext password |
| 00:57.28 | obnauticus | It's random, I taked to some dude that was is on Voipjet. |
| 00:57.40 | obnauticus | Ya, I know man...I can't get it because they won't reply via support |
| 00:57.45 | obnauticus | their Whois reccord is private |
| 00:57.48 | JT | whatever it is, you need your random password |
| 00:58.06 | JT | you could always reverse payment |
| 00:58.06 | obnauticus | Authorization code (password): (You should see an MD5 string, if it is blank logout and login again and/or use a recent version of Internet Explorer browser) |
| 00:58.19 | obnauticus | and I tried that btw :\ |
| 00:58.28 | JT | why didn't it work? |
| 00:58.42 | obnauticus | I dunno |
| 00:58.49 | RypPn | page cached? |
| 00:58.50 | obnauticus | its a text string, therefore it's probably their fault. |
| 00:59.00 | obnauticus | ctrl+f5 doesn't show up |
| 00:59.03 | obnauticus | I tried through a proxy too. |
| 00:59.07 | JT | i thought you meant reversed payment |
| 00:59.15 | Freman | http://pastebin.ca/794322 <- trys again, I included default context |
| 01:01.43 | Freman | http://pastebin.ca/794325 <- and again, debug from the call |
| 01:02.20 | Freman | but I can call 101 (an extension) just fine |
| 01:03.25 | *** join/#asterisk zerocod3r (n=Z3R3CoD3@b207d241.dorm.bilkent.edu.tr) |
| 01:03.28 | *** join/#asterisk Szab100 (i=szab100@ip57e52b52-pool.wnex.hu) |
| 01:03.36 | Szab100 | hi |
| 01:03.54 | Szab100 | anyone could help me on chan_misdn? |
| 01:04.13 | fujin_ | obnauticus: view source |
| 01:04.18 | obnauticus | Tried fujin_ |
| 01:04.26 | obnauticus | It's litterally not being displayed on their page, and it's their side. |
| 01:04.28 | obnauticus | I'm 100% sure |
| 01:04.33 | obnauticus | Because Ijust made a new account and it's not showing up. |
| 01:04.48 | Szab100 | i've been working on an asterisk install for a while now.. |
| 01:04.51 | obnauticus | I also tried through a proxy, different computer.... |
| 01:05.10 | fujin_ | Freman: which one of those contexts are your phones configured to use? |
| 01:05.39 | Freman | my phones use dialplan context, which includes default |
| 01:06.40 | Szab100 | i installed misdn as well, but when i call one of the msns from my mobile, there's no ringtone |
| 01:06.59 | Szab100 | asterisk said: |
| 01:07.01 | Szab100 | *CLI> P[ 1] --> !! lib: No free channel! |
| 01:07.01 | Szab100 | P[ 1] --> Didn't find BC so temporarly creating dummy BC (l3id:20004) on |
| 01:07.01 | Szab100 | this port. |
| 01:07.02 | Szab100 | P[ 1] Any Channel Requested, but we have no more!! |
| 01:07.04 | Szab100 | P[ 1] --> we have already send Release_complete |
| 01:07.06 | Szab100 | P[ 1] couldn't handle event |
| 01:07.24 | Freman | as soon as I restart asterisk it works again )c: |
| 01:08.02 | Szab100 | so i dont know what to configure to make asterisk answer an incoming call (route to internal extensions) |
| 01:08.15 | fujin_ | Freman: what version? |
| 01:08.31 | [hC] | So, I have blindxfer set to simply '#' in features.conf - using g729 it seems that (especially when using ATA's/cordless phones) that asterisk interprets people pressing # a lot, randomly |
| 01:08.46 | Szab100 | please help, i think i tried to read all info on the net i found |
| 01:08.49 | [hC] | thats kinda bizarre. packet loss or something generating enough of a disturbance in the codec for asterisk to interpret a # ? |
| 01:09.12 | Freman | http://pastebin.ca/794329 |
| 01:09.14 | fujin_ | [hC]: it's g729 |
| 01:09.35 | Freman | Asterisk 1.4.14 built by bachbuilder @ octopus.physik.fu-berlin.de on a i686 running Linux on 2007-11-17 22:24:04 UTC |
| 01:09.49 | JT | Szab100: |
| 01:09.51 | JT | ~pb |
| 01:09.51 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 01:09.58 | fujin_ | Freman: ok, and whats the console output of it when it doesn't work? |
| 01:10.03 | fujin_ | paste it side-by-side 794329 |
| 01:10.14 | Szab100 | JT sorry.. |
| 01:10.16 | fujin_ | wait a sec |
| 01:10.18 | fujin_ | I see the problem |
| 01:10.26 | fujin_ | Freman: your phones aren't maintaining registration |
| 01:10.31 | fujin_ | [Nov 23 10:59:48] NOTICE[28585]: chan_sip.c:13753 handle_request_invite: Call from '' to extension '0405060165' rejected because extension not found. |
| 01:10.51 | Freman | but they can call each other via asterisk |
| 01:11.13 | Freman | wouldn't messed up registration also kill that? |
| 01:12.45 | Szab100 | so, any help with misdn?? |
| 01:12.51 | Szab100 | pls.. |
| 01:13.28 | JT | Szab100: when you got that message, how many channels were in use? |
| 01:13.43 | Szab100 | 0 |
| 01:14.03 | fujin_ | Freman: sip show peers |
| 01:14.26 | fujin_ | pb your sip.conf |
| 01:14.38 | Szab100 | JT, the cable from the nt goes directly into the isdn card (configured as te_ptmp) |
| 01:14.46 | Freman | http://pastebin.ca/794334 |
| 01:14.50 | JT | Szab100: are you using the correct signalling type? (ptp or ptmp) |
| 01:15.27 | Szab100 | JT, yes, i think (but i tried all), because there is a box "NT" before the pci card |
| 01:15.33 | fujin_ | Freman: it looks to me like the phones are having issues maintaining registration. What phones? |
| 01:15.35 | Freman | interesting.. one phone I've been constantly dialing out on, still dials out, the one besides it that I only randomly tried is no longer dialing out |
| 01:15.38 | Freman | polycom |
| 01:15.47 | Szab100 | JT, so, i think it's point-to-mpoint |
| 01:16.04 | Freman | 102 can call out to my mobile still, 103 can't, however 103 can call 102 |
| 01:16.08 | Szab100 | JT, its a w6692 based card |
| 01:16.45 | Szab100 | JT, i didn't do any extra configuration in etc/asterisk/ just in misdn.conf |
| 01:16.52 | JT | Szab100: umm, hold up |
| 01:16.54 | obnauticus | http://farm3.static.flickr.com/2248/2053458310_6a30f248b1_o.jpg |
| 01:16.56 | obnauticus | ROFL! |
| 01:17.05 | JT | ALL BRIs MUST have an NT1 |
| 01:17.22 | JT | whether it is PTP or PTMP is up to your telco |
| 01:17.26 | tzanger | I guess I'mnot a geek |
| 01:17.32 | tzanger | I just don't find that type of humor funny |
| 01:17.50 | obnauticus | It's not really even supposed to be stupid.. |
| 01:17.53 | obnauticus | i mean funny* |
| 01:17.55 | obnauticus | it's supposed to be stupid* |
| 01:18.02 | *** join/#asterisk lowlevel (n=Stuart@CPE0017f2e2a3e8-CM000f9f7d6742.cpe.net.cable.rogers.com) |
| 01:18.06 | Szab100 | JT, yes, its a ptmp and i have an nt box the telco gave me |
| 01:18.19 | JT | Szab100: the telco told you it's PTMP? |
| 01:18.43 | Szab100 | JT, yes, and i looked after it, its the default configuration in my country (hungary) |
| 01:18.49 | JT | ok |
| 01:18.51 | Szab100 | JT, ptp links are more expensive |
| 01:19.28 | JT | no idea about the w6692 |
| 01:19.39 | JT | you will need contexts for calls to come in btw |
| 01:20.29 | saftsack | gn8 |
| 01:20.38 | Freman | fujin_, polycom 430's with 2.2 firmware |
| 01:20.40 | Szab100 | JT, "misdn channels" command shows: Chan List: (nil) |
| 01:21.32 | Szab100 | context where? in misdn.conf? |
| 01:21.40 | Szab100 | or somewhere else? |
| 01:23.00 | Szab100 | ok i see i have "context=from-pstn" in my misdn.conf (config copied from a misdn howto) |
| 01:23.37 | Szab100 | please let me know where / how to configure this context to make calls come to pbx |
| 01:24.04 | Szab100 | (because i think i don't have this context now, maybe that's the problem) |
| 01:26.25 | gedoc | is there any tut for *now server-to-server as the digium forum is a mess and i cannot find anythiny clear on google |
| 01:26.49 | JT | Szab100: do you understand how calls flow in asterisk? |
| 01:26.59 | gedoc | i believe so |
| 01:27.07 | gedoc | i have it already setup |
| 01:27.08 | *** join/#asterisk UserReg_CL (n=COB@pc-248-68-47-190.cm.vtr.net) |
| 01:27.19 | UserReg_CL | Holas !!! Buenas noches :) |
| 01:27.26 | gedoc | dual nat with external clients which works |
| 01:27.51 | gedoc | now i want to have 2 remote box talk to each other |
| 01:28.05 | JT | gedoc: who are you talking to? |
| 01:28.20 | gedoc | anyone who will listen |
| 01:28.34 | *** join/#asterisk fnordus (n=dnall@24.84.160.227) |
| 01:28.34 | JT | it looked like you were answering to me |
| 01:28.58 | Szab100 | JT, no :(... it's enough if you point me on a really useful / easy to understand article on this |
| 01:29.12 | gedoc | my bad JT |
| 01:29.28 | JT | Szab100: read all the stuff on contexts and extensions.conf in |
| 01:29.29 | JT | ~thebook |
| 01:29.30 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf |
| 01:29.39 | Freman | I've just lowered the registration time in my phones, and increased the overlap |
| 01:29.46 | Freman | we'll see if that helps |
| 01:29.47 | JT | gedoc: search for asterisk dual servers in voip-info.org |
| 01:29.50 | JT | it's an ok pointer |
| 01:29.58 | Freman | when they get around to rebooting |
| 01:30.00 | UserReg_CL | hi... helpme please... need send one password before call |
| 01:30.05 | Szab100 | JT, okay, i saw this book already but bypassed a little :) |
| 01:30.07 | gedoc | thanks |
| 01:30.08 | gedoc | cheers |
| 01:30.39 | Szab100 | i'll read and try again tomorrow.. maybe you'll see me again when i run into problems.. thanks anyway.. |
| 01:30.49 | JT | just remember don't trust voip-info.org word for word |
| 01:32.22 | Freman | seems to have made the situation worse |
| 01:32.31 | gedoc | as a matter of fact i was reading this before i asked my question |
| 01:33.05 | gedoc | this link http://www.voip-info.org/wiki/view/Asterisk+-+dual+servers mentions serverA and serverB which i am cool with |
| 01:33.48 | gedoc | 1 question though. should serverA contain user=serverB and serverB contain user=serverA? |
| 01:38.14 | webman | does anyone know of any other applications apart from asterisk that use zaptel interface/libraries ? |
| 01:39.48 | tzafrir | yate |
| 01:40.44 | obnauticus | 3rd party applications/modules. |
| 01:40.58 | obnauticus | Those use Zaptel interfaces/libraries! |
| 01:41.05 | *** join/#asterisk saint_ (n=saint@c-69-242-118-124.hsd1.nj.comcast.net) |
| 01:43.58 | JT | doesn't callweaver also use zaptel? |
| 01:44.25 | obnauticus | CallweaveR? |
| 01:44.28 | tzafrir | yes |
| 01:44.49 | webman | before I go and look at zaptel too much, is it possible from agi or dialplan/etc to dial a number (on a zaptel PRI) and have it return the busy/ringing/etc status before the other end answers ? |
| 01:44.50 | JT | googleable |
| 01:45.21 | obnauticus | Cann teh call be processed before answered? |
| 01:46.05 | webman | basically I want to call a number, and if it starts ringing, hang up immediately .... and get the info that is was ringing and we hung up .... |
| 01:46.15 | JT | define processed |
| 01:46.34 | obnauticus | I dunno, can it have like a priority and etc. |
| 01:46.37 | obnauticus | before even being answered? |
| 01:46.51 | JT | you can hangup a call before answering |
| 01:46.55 | obnauticus | ya |
| 01:46.56 | JT | you can listen to early media |
| 01:47.07 | obnauticus | That term sounds funny |
| 01:47.07 | fujin_ | detecting what you "hear" would be the hard part. |
| 01:47.15 | obnauticus | `you can hangup a call before answering' |
| 01:47.24 | JT | well you can |
| 01:47.25 | JT | on a pri |
| 01:47.29 | obnauticus | I know |
| 01:47.31 | obnauticus | it just sounds funny :) |
| 01:47.48 | webman | JT: it is more the pri signalling info I want rather than the audio .... |
| 01:47.59 | JT | like? |
| 01:48.12 | fujin_ | ringing state? |
| 01:48.45 | webman | well, I haven't looked too much yet, but fundamentally, I want to "test" a number to see if it rings, busy, disconnected/invalid number, etc.... |
| 01:49.30 | JT | if it's not available, you'll get a cause code that is not 16. |
| 01:49.32 | tzafrir | obnauticus, sure a call can be processed before answered |
| 01:49.58 | obnauticus | I know it just....I dunno |
| 01:50.07 | Mavvie | finally found out why this silly Patton 2977 card didn't want to call out to the PSTN. |
| 01:50.09 | obnauticus | It makes sense, but it doesn't at the same time to me |
| 01:50.18 | webman | JT: but if it is available, from the dialplan, how would I make it stop ringing before the other end answers? |
| 01:50.24 | Mavvie | By default it sets the caller ID to: resentation: Unknown (127) |
| 01:50.38 | Mavvie | But it should be: Presentation: Presentation allowed of network provided number |
| 01:50.41 | Mavvie | very tricky thing. |
| 01:50.56 | obnauticus | Wow, some dude bought a Supersonic Private jet |
| 01:50.58 | obnauticus | I want :\ |
| 01:51.29 | fujin_ | sweet |
| 01:52.09 | obnauticus | Ya it was sold for 80,000,000 duckets. |
| 01:52.25 | webman | the only method I could think of was to set the dialtime to 1 second, and then check the various variables, but ideally I want less than 1 second for the ring time (or more than 1 sec if that particular destination number requires more time to find out if it is available such as a mobile) |
| 01:54.56 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 01:55.36 | JT | webman: umm, use a timeout? |
| 01:57.34 | *** join/#asterisk webman_ (n=adamg@124.246.8.196.static.nexnet.net.au) |
| 01:57.39 | webman_ | sorry, I was disconnected, any comments on the short ring time etc could you please re-send it for me? |
| 01:58.23 | JT | webman: umm, use a timeout? |
| 02:14.14 | *** join/#asterisk fnordus (n=dnall@24.84.160.227) |
| 02:15.10 | *** join/#asterisk DaveCanoe (n=Dave@belbrrcnas12-3467437087.dial.bell.ca) |
| 02:25.58 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 02:27.12 | UserReg_CL | mmm |
| 02:27.50 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 02:27.50 | *** mode/#asterisk [+o russellb] by ChanServ |
| 02:30.54 | *** join/#asterisk fnordus (n=dnall@24.84.160.227) |
| 02:32.18 | *** join/#asterisk heison (n=heison@CPE000a01d49e6f-CM00111a59bec2.cpe.net.cable.rogers.com) |
| 02:32.50 | *** join/#asterisk techie (n=techie@adsl-76-214-9-1.dsl.lsan03.sbcglobal.net) |
| 02:37.40 | *** join/#asterisk bintut (n=bintut@203.125.63.150) |
| 02:39.50 | bintut | hello all.. |
| 02:41.40 | *** join/#asterisk rhombus (n=sfbosch@dsl-cap-66-18-218-36-cgy.nucleus.com) |
| 02:42.24 | rhombus | When I'm setting up SIP subscriptions, where should the hint lines go in the dialplan? |
| 02:44.13 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 02:45.48 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 02:45.48 | *** mode/#asterisk [+o russellb] by ChanServ |
| 02:46.17 | rhombus | hey, russellb |
| 02:46.31 | *** join/#asterisk PepOSX (n=pepOSX@190.72.147.163) |
| 02:46.36 | russellb | hey |
| 02:46.55 | rhombus | i'm trying to understand how hints and subscriptions work. Do you have a useful document you could direct me to? |
| 02:47.26 | rhombus | I've got a context with some hints in it -- only hints -- and then I point to that context in sip.conf |
| 02:47.48 | rhombus | and on my Polycom sets, I have got buddy watch set to point to those extensions |
| 02:47.59 | rhombus | but I don't see anything in the CLI: sip show subscriptions |
| 02:48.06 | rhombus | a big fat zero :) |
| 02:52.11 | [TK]D-Fender | rhombus, "show hints" |
| 02:52.19 | [TK]D-Fender | rhombus, And pastebin your dialplan |
| 02:52.34 | [TK]D-Fender | rhombus, Along with your phone's sip.conf entry |
| 02:52.36 | [TK]D-Fender | ~pb |
| 02:52.36 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 02:52.39 | rhombus | okay |
| 02:52.44 | rhombus | just a moment |
| 02:59.26 | rhombus | [TK]D-Fender: http://pastebin.ca/794409 |
| 02:59.34 | rhombus | hopefully that is what you need |
| 03:00.17 | [TK]D-Fender | rhombus, You're on * 1.4? |
| 03:00.28 | rhombus | no, 1.2 |
| 03:00.31 | UserReg_CL | mmm |
| 03:00.40 | UserReg_CL | ~book |
| 03:00.40 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf |
| 03:00.49 | [TK]D-Fender | rhombus, do you see a "buddies" soft-key on your phone? |
| 03:01.05 | rhombus | no |
| 03:01.16 | rhombus | I take it that means I need to add something to sip.cfg or the overlay cfg? |
| 03:01.31 | [TK]D-Fender | rhombus, Then your Polycom phone is not configured to allow Presence. |
| 03:01.35 | rhombus | I do see the buttons from the directory on the idle screen |
| 03:01.54 | [TK]D-Fender | rhombus, in sip.cfg look fo "Presence". By default it is deisabled ("0"). |
| 03:02.01 | rhombus | [TK]D-Fender: it would appear not. I wish this were mentioned in the Administrator's Guide |
| 03:02.14 | [TK]D-Fender | rhombus, it is in the guide :) |
| 03:02.20 | rhombus | The "Presence" section of the administrator mentions only the directory entry, though |
| 03:02.26 | [TK]D-Fender | rhombus, its under <features |
| 03:02.35 | [TK]D-Fender | rhombus, look HARDER :) |
| 03:02.47 | [TK]D-Fender | rhombus, So go search for it and enable it. |
| 03:03.25 | rhombus | I see the Presence section, but it doesn't mention a flag in sip.cfg |
| 03:05.18 | bintut | regarding http://www.privatepaste.com/5f1FRNW7U6 |
| 03:07.06 | rhombus | [TK]D-Fender: yep, I've confirmed it. The only place it's actually mentioned is in the Configuration Files section; the Presence section under Features says nothing about the flag |
| 03:07.08 | bintut | Do I really need HylaFax for the 2 Asterisk PBX servers based on diagram illustrated at http://www.privatepaste.com/5f1FRNW7U6 ? Both Asterisk boxes comes with IAXModem installed already but I must admit that I haven't tried to configure it. |
| 03:07.13 | rhombus | but that's okay, I'll go and enable it now :) thank you |
| 03:08.39 | [TK]D-Fender | rhombus, I have the guide ope in front of me. 4.6.1.23 FEATURE |
| 03:09.21 | [TK]D-Fender | ""resencse" is the presence feature including management of buddies and own status" |
| 03:09.35 | rhombus | yes, I see that |
| 03:09.44 | rhombus | does it mention the configuration flag? |
| 03:10.03 | rhombus | (also, the docs say that if it is not defined, the phone uses the primary line for presence detection) |
| 03:10.06 | rhombus | which is what I want |
| 03:10.14 | rhombus | I'm trying it with 1 now |
| 03:10.18 | [TK]D-Fender | rhombus, feature.x.enabled (0 or 1 default). Yeah... you should have looked :) |
| 03:10.36 | rhombus | [TK]D-Fender: on what page is that? |
| 03:10.46 | [TK]D-Fender | rhombus, But thats ok, thats what we're here for |
| 03:11.25 | rhombus | thank you -- can you tell me what page you found that on? |
| 03:11.29 | [TK]D-Fender | rhombus, I jsut gave you the chapter #. on the 1.5 admin guide its p.126. I'd have to spend actual time to give you the others. |
| 03:11.48 | [TK]D-Fender | rhombus, was the first version I grabbed off the shelf :) |
| 03:12.26 | rhombus | 4.6.1.23, right? |
| 03:13.26 | [TK]D-Fender | rhombus, yup |
| 03:13.53 | rhombus | Wow, man -- i finally found it |
| 03:13.59 | rhombus | it's 4.6.1.24 in this one |
| 03:14.17 | rhombus | that's obscure -- why wouldn't they mention that in the "presence" feature section? (Chapter 3)? |
| 03:14.36 | rhombus | The only way to find this is to do a text search, and then you get about 80 hits |
| 03:14.48 | [TK]D-Fender | rhombus, 60 tops :) |
| 03:14.57 | russellb | it's a buzzword |
| 03:15.12 | rhombus | okay, 60 :) |
| 03:15.29 | rhombus | anyway, thank you for helping me, even though I am a lazy moron |
| 03:16.09 | rhombus | :) |
| 03:16.15 | russellb | ouch |
| 03:16.23 | [TK]D-Fender | rhombus, Working? |
| 03:16.36 | rhombus | Not yet, still trying to interpret this command |
| 03:16.41 | [TK]D-Fender | rhombus, indeed 4.6.1.24 in the 2.0 Admin Guide. |
| 03:21.37 | ido | someone fucking fuck fuck |
| 03:21.56 | ido | sorry |
| 03:22.03 | ido | my asterisk box just exploded :( |
| 03:22.30 | russellb | O.O |
| 03:22.38 | bintut | hello..? can anyone tell me if i still need hylafax to run on both asterisk-1.4.x boxes with this kind of scenario? ==> http://www.privatepaste.com/5f1FRNW7U6 |
| 03:22.58 | rhombus | [TK]D-Fender: joy! It is working now. Thank you. |
| 03:25.09 | [TK]D-Fender | rhombus, You're welcome |
| 03:26.23 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 03:27.13 | obnauticus | YES |
| 03:27.23 | obnauticus | I can finally play the melody to funky town using only DTMF tones! |
| 03:32.06 | fujin_ | oh my god |
| 03:32.08 | fujin_ | pastebin the code |
| 03:32.21 | obnauticus | j00 serial?! |
| 03:32.33 | fujin_ | ya pls |
| 03:32.40 | fujin_ | 55 9 9 4 55 9 9 4 |
| 03:32.40 | fujin_ | ? |
| 03:32.50 | *** join/#asterisk fnordus (n=dnall@24.84.160.227) |
| 03:34.06 | [TK]D-Fender | obnauticus, and to think all the time I've wasted learning guitar & piano! |
| 03:34.11 | obnauticus | 66511555669#996556665565 |
| 03:34.30 | obnauticus | [TK]D-Fender, that's funny I play the Piano, Guitar, Bass, Viola, Violain, Violinchello, and French horn! |
| 03:34.33 | obnauticus | What a waste of time! |
| 03:34.41 | obnauticus | I'm actually holding my viola right now |
| 03:34.51 | [TK]D-Fender | obnauticus, violin would actually be the next I'd like to learn... |
| 03:34.55 | fujin_ | obnauticus: the 1 sounds wrong |
| 03:34.59 | obnauticus | Viola is more fun.. |
| 03:35.01 | obnauticus | ya i dunno |
| 03:35.02 | obnauticus | perfect it |
| 03:35.09 | obnauticus | my Cisco phone gave me busy and i couldn't hear it. |
| 03:35.35 | obnauticus | [TK]D-Fender the violin is easy if you know piano |
| 03:35.39 | obnauticus | just technique and shit |
| 03:36.05 | [TK]D-Fender | obnauticus, I fly on the guitar so its just a question of getting a feel for the bow. |
| 03:37.09 | obnauticus | fujin_ sorry, that was the melody to Take on Me |
| 03:37.10 | obnauticus | lol |
| 03:37.15 | obnauticus | .I get the 2 confused, they are both 80 |
| 03:37.17 | obnauticus | 80's crap |
| 03:37.44 | [TK]D-Fender | obnauticus, "crap 80's", not "80's crap" :) |
| 03:37.50 | obnauticus | lol |
| 03:37.52 | obnauticus | good point |
| 03:38.08 | UserReg_CL | mmm |
| 03:38.34 | UserReg_CL | helpme need set password for each user before call |
| 03:38.52 | DarkRift | Anyone familiar with realtime configuration ? I have my users defined in the postgresql database and I'm trying to get the dialplan variable associated to the "mailbox" field but ${MAILBOX} doesn't seem to return anything, and same for {AUTH_MAILBOX}. Anyone can know the right variable or if it can even be accessed via the variable model ? |
| 03:39.07 | DarkRift | -can |
| 03:41.04 | obnauticus | and [TK]D-Fender once you're good you can just practice by holding it like a guitar |
| 03:41.06 | obnauticus | i dunno in your case |
| 03:41.07 | obnauticus | lol |
| 03:41.52 | [TK]D-Fender | file, test complete, alls good, can switch back now. Thanks |
| 03:49.38 | [TK]D-Fender | DarkRift, how do you want to retrieve it? |
| 03:49.48 | [TK]D-Fender | DarkRift, and to do what with it? |
| 03:51.50 | DarkRift | I want to get the value just like ${callerid(num)} for example to be able to enter the VoiceMailMenu to their associated mailbox when they dial * |
| 03:52.48 | [TK]D-Fender | DarkRift, which of course is NOT the same I take it? |
| 03:52.55 | DarkRift | indeed |
| 03:53.17 | [TK]D-Fender | DarkRift, use func_odbc to pull it direct from the table. |
| 03:53.19 | DarkRift | Thus why I wanted to know if there was a ${variable} or function to retreive their info |
| 03:54.49 | DarkRift | It can be used like that ? ${func_odbc()} ? If so do you have an example or I'd have to make a module for that ? |
| 03:54.53 | obnauticus | [TK]D-Fender you wanna hear something funny |
| 03:55.03 | obnauticus | I made 2nd in state playng the freebird guitar solo on my violin |
| 03:55.04 | obnauticus | rofl. |
| 03:55.33 | [TK]D-Fender | obnauticus, as in the 2nd to actually bother trying? :) |
| 03:56.24 | obnauticus | In state? |
| 03:56.24 | obnauticus | no. |
| 03:56.33 | obnauticus | You have to try to get to state |
| 03:56.34 | obnauticus | silly |
| 03:56.36 | obnauticus | espically onthe viola. |
| 03:56.37 | obnauticus | violin* |
| 03:58.20 | *** join/#asterisk watchy (n=watchy@h200.176.255.206.cable.cmdn.cablelynx.com) |
| 03:58.34 | rhombus | [TK]D-Fender: What forms of SIP presence does Asterisk support? |
| 03:58.44 | watchy | I got an * box when I dialin on a Zap channel and hit a channel bank I have no audio, anyone have any idea where to start looking? |
| 03:58.49 | [TK]D-Fender | rhombus, meaning? |
| 03:59.08 | [TK]D-Fender | watchy, describe both sides and all hardware please |
| 03:59.13 | rhombus | Well, there's "busy" "ringing", "dnd" |
| 03:59.45 | watchy | Rhino PRI T1 to Rhino T1 Card to a Rhino Analog channel bank |
| 03:59.51 | [TK]D-Fender | rhombus, * tracks "on phone, ringing, idle, MIA" to my awareness.... not sure of any others. don't think DND |
| 03:59.52 | rhombus | I recall reading that those aren't strictly supported -- case in point; the presence notification on the Polycom shows "DND" when somebody is making or taking a call |
| 04:00.05 | rhombus | hmn. |
| 04:00.16 | [TK]D-Fender | watchy, so bridging PRI to T1 CB direct? |
| 04:00.59 | [TK]D-Fender | rhombus, Oh, the ICON looks like a DND, but it is its version of "on the phone". Press "buddies" for more detail. |
| 04:01.26 | rhombus | [TK]D-Fender: ah, yes -- it says "on the phone" |
| 04:01.41 | DarkRift | Brb, need to restart my computer |
| 04:01.47 | watchy | yea. They dialin hit the Rhino PRI then it goes to an analog phone on the Chennel bank thats connected to the other T1 card |
| 04:02.02 | [TK]D-Fender | rhombus, I know the icon looks like something you'd associate with "DND", but I interpret as "do not enter", AKA "busy" |
| 04:02.21 | [TK]D-Fender | watchy, have you test them independantly? |
| 04:02.31 | rhombus | I suppose i have to set "notifyringing" in sip.conf to see ringing -- right now it just shows "on the phone" |
| 04:03.15 | watchy | tk: apparently if he hits flash on his phone |
| 04:03.20 | watchy | it connects us and givea audio |
| 04:04.38 | watchy | http://pastebin.com/m551ad4fb |
| 04:04.41 | watchy | check that |
| 04:04.44 | watchy | you see anything wierd |
| 04:08.22 | watchy | that is the ugliest agi crap i have ever seen why cant they just use straight * |
| 04:08.22 | watchy | heh |
| 04:09.15 | russellb | ~trixbox |
| 04:09.15 | jbot | [~trixbox] Trixbox is a full linux distro that includes FreePBX and other 3rd party add-ons. It is all this extra stuff which makes trixbox VERY difficult to support, and is not supported in #asterisk. Try asking in #trixbox or on their forums & wiki at http://www.trixbox.org |
| 04:10.16 | watchy | trixbox sucks |
| 04:10.19 | watchy | htis is elastics |
| 04:11.02 | [TK]D-Fender | watchy, Executing [s@macro-dial:10] Dial("Zap/1-1", "ZAP/105|22|trM(auto-blkvm)") in new stack |
| 04:11.19 | [TK]D-Fender | watchy, First "r" = EVIL. 2nd, WTF is that macro called on dial? |
| 04:11.37 | watchy | dunno dude this is elastics |
| 04:11.52 | watchy | this aint my stupid * distro. were using it cuz this guy i work with loves guis |
| 04:12.07 | [TK]D-Fender | rhombus, Yes, Polycom presences does NOT show ringing as "ringing". Aastra's do however |
| 04:12.33 | rhombus | [TK]D-Fender: ah, okay. |
| 04:12.36 | [TK]D-Fender | watchy, Clearly FrrePBX..... |
| 04:12.53 | watchy | yea, you agree freepbx is a pos? |
| 04:13.04 | [TK]D-Fender | watchy, I think "DUH!" sums it up. |
| 04:13.12 | watchy | i need to hire you to come in and show why freepbx is a POS |
| 04:13.12 | rhombus | [TK]D-Fender: I've just noticed that when I set notifyringing=yes, buddy watch stops working |
| 04:13.17 | watchy | and why we should use straight * |
| 04:13.28 | watchy | you think you could do that |
| 04:13.38 | [TK]D-Fender | watchy, I mean its a GUI and as long as you like everything that it does, then one's as good "or bad" as the next more or less |
| 04:13.57 | watchy | yea except when it has a 1 full page screen for a dialin? |
| 04:14.01 | [TK]D-Fender | rhombus, there is no way to Polycom to represent the ringing.... |
| 04:14.08 | watchy | is that really needed |
| 04:14.09 | [TK]D-Fender | rhombus, Sorry... |
| 04:14.28 | watchy | 1 incoming call 1 full screen of junk |
| 04:14.36 | rhombus | [TK]D-Fender: Okay -- I won't bang my head against the wall. I was just curious if it was possible |
| 04:14.38 | watchy | i think it could be done in a few less lines personally |
| 04:14.44 | hardwire | gobblegobble |
| 04:15.10 | *** join/#asterisk DarkRift (i=dark@bas10-montreal02-1177583123.dsl.bell.ca) |
| 04:15.57 | watchy | i'd sure like to pay you though to atleast be in on a conf call with me and a few of my techs to explain why we should dump freepbx |
| 04:16.39 | russellb | ooh, i'll take money to just sit on a conference call |
| 04:17.14 | watchy | well i also want tk to yell at these idiots |
| 04:17.45 | rhombus | watchy: are you having trouble with FreePBX? What sort of trouble? |
| 04:17.51 | [TK]D-Fender | watchy, use of a GUI depends on your business model. |
| 04:17.54 | watchy | its a POS |
| 04:18.05 | rhombus | watchy: that's awfully inspecific |
| 04:18.19 | rhombus | whether it is a POS depends on what you are trying to do |
| 04:18.23 | watchy | well when it dont work perfectly, the guys installing these boxes cant troubleshoot it |
| 04:18.27 | [TK]D-Fender | watchy, Still we don't know if its truely a hardwre problem. make a test removing the 2 things I jsut told you to remove and try again |
| 04:18.36 | watchy | because freepbx is making a 1 page script for 1 did |
| 04:19.02 | watchy | oh i'm not on site nor really give a shit right now |
| 04:19.28 | watchy | the only reason we had to replace it is because someone bought a full "rhino box" and it started fucking kernel panicing lastnight |
| 04:19.32 | watchy | at 8pm |
| 04:19.35 | rhombus | watchy: that's fair enough, but are you breaking things down properly? [TK]D-Fender is right |
| 04:19.47 | watchy | so he decided to rebuild the POS |
| 04:19.51 | watchy | with Elastics |
| 04:19.56 | rhombus | watchy: purchasing the rhino box was your mistake, not using FreePBX |
| 04:20.07 | watchy | oh i agree, it wasnt my mistake thank god |
| 04:20.15 | rhombus | whoevers mistake |
| 04:20.18 | watchy | i would never buy a POS rhino product |
| 04:20.32 | watchy | i'm %100 sangoma PRI cards and intel generic boxes |
| 04:20.40 | watchy | %100 polycom phones |
| 04:20.47 | watchy | i use proven hardware |
| 04:21.04 | rhombus | that's wise where phones are concerned :) |
| 04:21.22 | watchy | blame TK. Hes the reason i use polys |
| 04:21.31 | rhombus | They're good phones |
| 04:21.45 | watchy | i know i wouldnt trade them for free sex with jessica alba |
| 04:21.47 | rhombus | I recently tried some Aastras, and I was pleased with those also |
| 04:21.56 | rhombus | watchy: i think you are lying |
| 04:22.02 | watchy | we put in some aastras at a place and they worked great to |
| 04:22.08 | watchy | jessica alba has herpes |
| 04:22.12 | watchy | proven fact unfortunately |
| 04:22.17 | rhombus | I would trade all the gold in my grandmother's mouth for free sex with Jessica Alba |
| 04:22.27 | watchy | enjoy your new found herpes |
| 04:22.47 | rhombus | maybe I have it already and don't care |
| 04:23.06 | watchy | hahaha i bet you think jessica simpsons hot and pam anderson to |
| 04:23.11 | watchy | both very ugly women |
| 04:23.12 | rhombus | watchy: no |
| 04:23.15 | [TK]D-Fender | watchy> i know i wouldnt trade them for free sex with jessica alba <-- ok .... there is officially something fundamentally wrong with you now :) |
| 04:23.35 | watchy | a very hot chick is Scarlett Johanson |
| 04:23.37 | rhombus | But I wouldn't kick Jessica Alba out of my bed for eating crackers |
| 04:23.38 | watchy | she is a goddess |
| 04:23.49 | [TK]D-Fender | watchy, SJ does nothing for me...... |
| 04:23.56 | watchy | your gay tk, i understand |
| 04:24.10 | hardwire | be nice y'all |
| 04:24.10 | watchy | also Zoey Dasheuel or whatever her name is from Hitchikers guide |
| 04:24.12 | watchy | shes nice |
| 04:24.22 | rhombus | [TK]D-Fender: can I program the function keys on the left side of the Polycom 501s to do anything interesting? |
| 04:24.32 | [TK]D-Fender | watchy, nope.... just SJ's facial features / build = meh. I go for brunettes :) |
| 04:24.43 | [TK]D-Fender | rhombus, which ones? |
| 04:25.02 | watchy | i bet you think Eva longoria or whatevers hot huh |
| 04:25.08 | rhombus | Directories, Services, Call Lists, etc |
| 04:25.33 | rhombus | watchy: I think you're hot |
| 04:25.38 | watchy | thanks lets date |
| 04:26.00 | [TK]D-Fender | watchy, EL? meh.... The one who plays in the new Bionic Woman, and Rose McGowan are better examples |
| 04:26.16 | watchy | is it possible to make a button on a 501 show the local weather? |
| 04:26.20 | rhombus | The one who plays in the old Bionic Woman was pretty fine also |
| 04:26.28 | [TK]D-Fender | rhombus, Services = MicroBrowser. Thats a world of fun right there. |
| 04:26.47 | [TK]D-Fender | rhombus, in her day hell yeah |
| 04:26.57 | webman | sorry to be so off-topic, but anyone know of a small/simple zaptel application which can dial a number, or know how to code such a thing ? |
| 04:26.58 | rhombus | Lindsay Wagner.... hotttt |
| 04:27.07 | watchy | was hot, not anymore |
| 04:27.14 | rhombus | webman: yes, very OT -- this is the "Who's Hot Who's Not" channel |
| 04:27.23 | [TK]D-Fender | webman, "zaptel application" <- 2 words that don't belong together |
| 04:27.29 | [TK]D-Fender | rhombus, it is now :) |
| 04:27.32 | rhombus | watchy: how would you know? Have you seen Lindsay Wagner lately? Do you even know who she is? |
| 04:27.49 | watchy | i'm only 27 but yes, she does those eye comercials i think |
| 04:27.54 | watchy | or those bed comericials |
| 04:28.01 | watchy | shes olddddddd |
| 04:28.47 | watchy | when I visit where you live tk imma buy you some steaks |
| 04:29.10 | [TK]D-Fender | mooooooooooooooo |
| 04:29.14 | *** join/#asterisk Defraz (n=t0tal@c-76-27-13-164.hsd1.ut.comcast.net) |
| 04:29.38 | [TK]D-Fender | watchy, I want Bessie with the bell still 'round her neck :) |
| 04:29.41 | watchy | i knew a guy named defraz from irc back in the day |
| 04:29.46 | watchy | i doubt its the same one |
| 04:30.01 | watchy | you know i'm the same way, its gotta be bleeding or its not eatable |
| 04:30.07 | rhombus | [TK]D-Fender: If the other set turns DND on, should I see that in my Buddies? |
| 04:30.14 | rhombus | because if so, it ain't working :) |
| 04:30.15 | [TK]D-Fender | rhombus, nope. |
| 04:30.19 | rhombus | okay |
| 04:30.31 | rhombus | so it's pretty limited, then, am I right? |
| 04:30.36 | watchy | buddies on poly don't show dnd? |
| 04:30.42 | [TK]D-Fender | rhombus, DND is not something that is passed back to the server. Its more like a local "state" that tells the phone to simply refuse an invite. |
| 04:30.44 | watchy | i never new that |
| 04:30.50 | rhombus | ahhhhhhhhhhh |
| 04:30.53 | watchy | oh i see |
| 04:31.00 | watchy | i learneded something |
| 04:31.53 | watchy | tk: you ever play with a dell blade? |
| 04:31.56 | [TK]D-Fender | like on Poly's you can selective reject contacts or use the reject key. only diff with DND is that you have it set the return reason in sip.cfg. Then again you won't know how' it'll react to the NEXT call so fat load of good that'll do you in the long run |
| 04:32.13 | [TK]D-Fender | watchy, no, but I've played with sharp blades :) |
| 04:32.35 | [TK]D-Fender | watchy, http://gallery.aocomputing.net/index.php?album=2007-03-02+Oni+Forge+Bushi |
| 04:32.36 | watchy | i bought a new dell blade for my house to play with |
| 04:32.50 | rhombus | [TK]D-Fender: well, the implementation on the Polycom phones is old, so... it's good enough for me |
| 04:32.55 | watchy | cant use it for * though except sip/iax |
| 04:32.58 | watchy | no pci slots |
| 04:33.14 | rhombus | [TK]D-Fender: basically this Buddy Watch -- it's the same as BLF, correct? |
| 04:33.22 | watchy | is that a real ninja sword? |
| 04:33.23 | rhombus | I can't see the difference conceptually |
| 04:33.32 | [TK]D-Fender | rhombus, Same game, different name |
| 04:33.59 | rhombus | that's what I figured -- the funny thing is that the Polycom docs treat them as different -- they use BLF for the 601 + sidecar |
| 04:34.10 | [TK]D-Fender | watchy, that is my current katana (primary). I own 1 other, and an iaito (dulled practice blade) |
| 04:34.18 | watchy | is that sword as good as a hanso? |
| 04:34.20 | [TK]D-Fender | rhombus, all the same |
| 04:34.28 | rhombus | [TK]D-Fender: to your knowledge, have the crashing sidecar problems been corrected? |
| 04:34.29 | [TK]D-Fender | watchy, better... mines REAL :) |
| 04:34.37 | watchy | tk: what did you drop US for it? |
| 04:34.42 | [TK]D-Fender | rhombus, I've never experienced that problem... |
| 04:35.26 | obnauticus | http://www.youtube.com/watch?v=KhPmsIsQ-JE ROFL |
| 04:35.27 | [TK]D-Fender | watchy, was $525 CAD almost a year ago. So $480 maybe |
| 04:35.34 | rhombus | [TK]D-Fender: lots of people have reported trouble with them... sidecars in general seem to be problematic. I've heard reports of trouble with the Aastra and Snom sidecars |
| 04:35.57 | [TK]D-Fender | rhombus, Oh yeah... Aastra's lock up hard |
| 04:36.01 | watchy | wow, do you practice the art of sword fighting or whatever its called? |
| 04:36.08 | [TK]D-Fender | rhombus, they are a little shaky these days |
| 04:36.30 | watchy | i never had to many issues with my poly sidecars |
| 04:36.31 | rhombus | Sidecars in general, the Aastra sidecars, or the 5i series Aastra sets? |
| 04:36.33 | [TK]D-Fender | watchy, here : http://video.google.ca/url?docid=-3592341485993959661&esrc=sr2&ev=v&len=1197&q=katori%2Bshinto&srcurl=http%3A%2F%2Fvideo.google.ca%2Fvideoplay%3Fdocid%3D-3592341485993959661&vidurl=%2Fvideoplay%3Fdocid%3D-3592341485993959661%26q%3Dkatori%2Bshinto%26total%3D109%26start%3D0%26num%3D10%26so%3D0%26type%3Dsearch%26plindex%3D1&usg=AL29H21AshWPPfsBXLA23FVBPBNyajFlcw |
| 04:37.04 | [TK]D-Fender | watchy, Oops... shorter link http://video.google.ca/videoplay?docid=-3592341485993959661&q=katori+shinto&total=109&start=0&num=10&so=0&type=search&plindex=1 |
| 04:37.08 | watchy | haha |
| 04:37.18 | [TK]D-Fender | watchy, best sample of my art |
| 04:37.22 | obnauticus | Tip: Click on the lik and copy the url of google videos, instead of just right clicking and hitting copy link location |
| 04:37.42 | watchy | tk: cool |
| 04:37.50 | [TK]D-Fender | skip to 3:00 for the start |
| 04:38.05 | *** join/#asterisk CunningPike (n=CunningP@S010600095b33697f.vc.shawcable.net) |
| 04:38.42 | watchy | you are an amazing intresting person. i never woulda taken you for a sword swinging dude |
| 04:38.59 | obnauticus | it's chiine! |
| 04:39.25 | obnauticus | http://www.youtube.com/watch?v=KhPmsIsQ-JE my art! |
| 04:39.25 | obnauticus | lol. |
| 04:40.01 | [TK]D-Fender | watchy, Keep on till the iaijutsu part (with the true sword) |
| 04:40.24 | obnauticus | the true sword -- blessed? |
| 04:40.32 | obnauticus | im just kidding |
| 04:40.38 | obnauticus | replace the 2cm thick wooden dowel with a katana |
| 04:40.40 | obnauticus | rofl. |
| 04:40.41 | rhombus | [TK]D-Fender: Sidecars in general, the Aastra sidecars, or the 5i series Aastra sets? |
| 04:40.47 | [TK]D-Fender | obnauticus, all non-solo katas are with wooden weapons. |
| 04:41.02 | [TK]D-Fender | rhombus, Aastras..... they are a little flakey firmware-wise these days |
| 04:41.30 | rhombus | <PROTECTED> |
| 04:41.39 | [TK]D-Fender | rhombus, its being worked on |
| 04:42.18 | rhombus | <PROTECTED> |
| 04:42.31 | [TK]D-Fender | rhombus, dunno... haven't gone myself... |
| 04:43.31 | [TK]D-Fender | obnauticus, 9min in on the video is the start of the iaijutsu. |
| 04:43.42 | [TK]D-Fender | obnauticus, thats what I use my live blades for |
| 04:45.08 | watchy | obnauticus: that guys fin crazy |
| 04:45.24 | obnauticus | I can left hand pizzacato like him (arond 2min) |
| 04:45.30 | obnauticus | I've been playing for 13 years... |
| 04:45.31 | obnauticus | vioal |
| 04:45.32 | obnauticus | viola* |
| 04:46.00 | obnauticus | I can almost play that though |
| 04:46.27 | [TK]D-Fender | watchy, on the 3rd kneeling kata (nuki uchi no ken) whre he jumps stright up in there air is the only accident I ever had (alone). |
| 04:46.47 | watchy | if someone attacked you would you cut them up? |
| 04:46.47 | [TK]D-Fender | watchy, trashed a shirt and left a tiny scar |
| 04:47.19 | watchy | would you feel comfortable taking a life with your blade? |
| 04:47.19 | [TK]D-Fender | watchy, Depends what I'm carrying when attack, how I'm attacked, and how I feel I can/should respond |
| 04:47.34 | watchy | if someone broke into your house |
| 04:48.22 | [TK]D-Fender | watchy, as a matter of self defense yes. Protection of property, well that would depend on a bad outcome of trying to hold him for the authorties |
| 04:48.35 | rhombus | [TK]D-Fender: is parking BLF support in the current versions of Asterisk 1.2? |
| 04:48.36 | *** join/#asterisk jiffe3 (n=jiffe@67.158.2.241) |
| 04:48.45 | [TK]D-Fender | watchy, but in defense of self/others I would pity the waste of life. |
| 04:48.49 | rhombus | I have no idea how I would verify that. |
| 04:49.08 | [TK]D-Fender | rhombus, Don't think so.... it is in 1.4 |
| 04:49.23 | rhombus | okay |
| 04:52.25 | jiffe3 | anyone know if there is a standard or set of standard APIs for communicating and directing a VOIP device? |
| 04:54.57 | [TK]D-Fender | jiffe3, how do you "direct" a "voip device"? |
| 04:55.37 | jiffe3 | send commands such as to dial a specific number |
| 04:55.57 | rhombus | it depends on the device -- there is no standard API |
| 04:56.21 | jiffe3 | alright, that is what I was wondering |
| 04:56.50 | [TK]D-Fender | jiffe3, you don't tell a phone to dial.... |
| 04:57.01 | [TK]D-Fender | jiffe3, you USE a phone to dial. |
| 04:57.11 | rhombus | I tell the bitch to dial |
| 04:57.22 | rhombus | I say "dial, bitch!" |
| 04:57.26 | rhombus | and usually she does. |
| 04:58.00 | [TK]D-Fender | rhombus, I have the ASR on my cellphone do that too ;) |
| 04:58.04 | obnauticus | MAKE ME A SANDWIDTCH! |
| 04:58.07 | *** join/#asterisk Cresl1n (n=matt@c-68-62-219-187.hsd1.al.comcast.net) |
| 04:58.07 | *** mode/#asterisk [+o Cresl1n] by ChanServ |
| 05:00.01 | obnauticus | Is there a speech to text for * yet? |
| 05:00.06 | obnauticus | I wanna make my Phone based IRC cline |
| 05:00.06 | obnauticus | t |
| 05:00.09 | obnauticus | client* |
| 05:00.10 | obnauticus | lolz |
| 05:01.07 | [TK]D-Fender | obnauticus, Lumenvox <- |
| 05:01.28 | *** join/#asterisk lowlevel (n=Stuart@CPE0017f2e2a3e8-CM000f9f7d6742.cpe.net.cable.rogers.com) |
| 05:01.32 | obnauticus | buggy? |
| 05:01.43 | [TK]D-Fender | obnauticus, its translations can't be worse than your typos and "l33t-sp34k" so ... umm.. yeah ;) |
| 05:02.01 | obnauticus | They aren't typeo's, I just have sexual affair's with my enter key. |
| 05:02.27 | [TK]D-Fender | obnauticus, "typeo's" indeed :) |
| 05:02.40 | obnauticus | That's not what i like to call them |
| 05:02.40 | obnauticus | lol |
| 05:02.44 | watchy | h4x0r sp34k |
| 05:04.54 | rhombus | Thank you [TK]D-Fender for your help this evening. Good luck, and try not to cut your nads off. |
| 05:05.23 | [TK]D-Fender | rhombus, only cut myself once :) |
| 05:05.46 | rhombus | it's still good advice |
| 05:05.55 | [TK]D-Fender | rhombus, I need onother to make it symetrical now :) |
| 05:06.02 | rhombus | they don't call them family jewels for nothing. |
| 05:06.06 | *** part/#asterisk rhombus (n=sfbosch@dsl-cap-66-18-218-36-cgy.nucleus.com) |
| 05:07.01 | obnauticus | I'm asian so that litterally disproves his statment |
| 05:47.43 | *** join/#asterisk mcab (n=mb@mostly-harmless.ca) |
| 06:09.31 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@adsl-68-93-83-7.dsl.hstntx.swbell.net) |
| 06:16.37 | *** join/#asterisk Cresl1n (n=matt@c-68-62-219-187.hsd1.al.comcast.net) |
| 06:16.37 | *** mode/#asterisk [+o Cresl1n] by ChanServ |
| 06:18.06 | *** join/#asterisk sergey (n=sergey@91.189.233.71) |
| 06:26.00 | *** join/#asterisk metfan2007 (n=metfan20@189.180.217.155) |
| 06:26.21 | metfan2007 | Hi all!!!!, Is there any way to login an agent to a queue using the CLI??? |
| 06:32.33 | *** join/#asterisk bintut (n=bintut@203.125.63.150) |
| 06:36.22 | *** join/#asterisk usm88 (n=usm88@125.162.42.14) |
| 06:36.53 | *** join/#asterisk \2Legit (n=is2legit@cpe-24-95-70-237.columbus.res.rr.com) |
| 06:36.55 | bintut | is it really need to install hylafax for faxing over iax using iaxmodem on both asterisk boxes? |
| 06:37.10 | \2Legit | anyone here know what program i need to read .REC files? |
| 06:38.09 | *** join/#asterisk _BaconButtie_ (n=bob@host-82-33-206-45.static.telewest.net) |
| 06:56.58 | *** join/#asterisk ming_zym (i=ming_zym@nat/yahoo/x-4a19afdf6c26e840) |
| 07:02.48 | *** join/#asterisk sergee (n=serg@voip1.west-call.com) |
| 07:07.32 | *** join/#asterisk harpal (n=Harpal@124.125.255.223) |
| 07:16.08 | *** join/#asterisk Strom_M (n=strom@ip70-170-60-8.lv.lv.cox.net) |
| 07:17.57 | Siya | what is the location of svn asterisk-sounds |
| 07:18.08 | Siya | most links I find are dead |
| 07:19.24 | Siya | http://svn.digium.com/svn/asterisk-sounds doesn't exist anymore |
| 07:22.31 | _BaconButtie_ | iirc, if you're using 1.4, you don't need to download sounds. compiling will download the sound packs you select in "make menuselect" |
| 07:24.05 | Siya | _BaconButtie_: ahh, ic |
| 07:24.44 | usm88 | hi _BaconButtie_, may i ask ? |
| 07:25.56 | _BaconButtie_ | usm88 .... go ahead |
| 07:26.27 | usm88 | i just installed asterisk 1.2.24 in my laptap |
| 07:26.44 | usm88 | installation all going smoothly |
| 07:27.00 | usm88 | and i can start asterisk server without problem |
| 07:28.06 | usm88 | but when i'm testing it by using xlite to connect it keep saying login timeout |
| 07:28.20 | usm88 | where should i check ? |
| 07:31.34 | *** join/#asterisk dominic1 (n=dob@213.221.82.242) |
| 07:31.38 | *** part/#asterisk dominic1 (n=dob@213.221.82.242) |
| 07:33.56 | *** join/#asterisk zeeesh (n=zeeesh@202.166.161.45) |
| 07:35.18 | *** join/#asterisk Cresl1n (n=matt@c-68-62-219-187.hsd1.al.comcast.net) |
| 07:35.18 | *** mode/#asterisk [+o Cresl1n] by ChanServ |
| 07:37.07 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 07:39.10 | Siya | _BaconButtie_: tx |
| 07:41.39 | *** join/#asterisk anujsingh (n=Administ@59.94.150.91) |
| 07:41.43 | _BaconButtie_ | usm88: you've created an appropriate entry in sip.conf ? what configs have you created so far ? |
| 07:42.45 | anujsingh | is it possible to play recorded sound files of users via softphone? |
| 07:43.03 | usm88 | here is my sip.conf : |
| 07:43.07 | usm88 | [2052001] |
| 07:43.08 | usm88 | context=default |
| 07:43.08 | usm88 | type=friend |
| 07:43.08 | usm88 | username=2052001 |
| 07:43.08 | usm88 | secret=012345 |
| 07:43.08 | usm88 | host=dynamic |
| 07:43.09 | usm88 | dtmfmode=rfc2933 |
| 07:43.13 | usm88 | mailbox=usman@ptgrowthasia.com |
| 07:43.22 | usm88 | and here is the extension.conf : |
| 07:43.27 | *** join/#asterisk bereghelea (n=ddd@host-static-212-0-198-30.moldtelecom.md) |
| 07:43.28 | *** join/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 07:43.39 | bereghelea | hi all |
| 07:44.00 | usm88 | exten => _20XXXXX,1,Dial(SIP/${EXTEN},30,rt) |
| 07:44.01 | anujsingh | how to play recorded conversation of users, stored in /var/spool/asterisk/monitor/DONE |
| 07:44.02 | usm88 | exten => _20XXXXX,3,HangUp |
| 07:44.09 | bereghelea | chan_h323.c: No RTP stream is available for call ip$10.11.12.13:17583/85 (85)[Nov 23 01:00:04] ERROR[3474] rtp.c: Unexpected bind error: Can't assign requested addre |
| 07:44.21 | bereghelea | what can i do with this error |
| 07:44.53 | harpal | Any one look at http://pastebin.ca/794567 and tell me whats problem? |
| 07:45.31 | usm88 | _BaconButtie_, I just modify two conf : sip.conf and extension.conf |
| 07:45.46 | harpal | I cant do conferencing with MeetMe? |
| 07:48.06 | bereghelea | ìëÿ êó÷à ëþäåé â êàíàëå à òèøèíà êàê íà êëàäáèùå |
| 07:54.06 | FlatFoot | harpal: this line exten => 555,1,MeetMe(555,i,1234) |
| 07:54.17 | FlatFoot | should it be exten => 555,3,MeetMe(555,i,1234) |
| 07:54.33 | FlatFoot | looking at the dialplan it does not seem to follow numerically |
| 07:55.39 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 07:56.01 | harpal | FlatFoot, I have done that but still have same problem |
| 07:56.31 | FlatFoot | also don't forget that a lot of commands / functions are now not active in the latest versions |
| 07:58.44 | harpal | FlatFoot, now see http://pastebin.ca/794575 |
| 07:59.01 | harpal | So Is that MeetMe is not working? |
| 07:59.50 | harpal | I have removed other things Just remaining only exten => 555,1,MeetMe(555,i,1234) but still its not making conference |
| 07:59.55 | FlatFoot | looks like it , not had much experience with meetme , what version of * is it ? does MeetMeCount still exist in that format ? |
| 08:00.40 | harpal | I have 1.4.13 |
| 08:00.59 | FlatFoot | that version has had a lot of things changed |
| 08:01.22 | FlatFoot | give a help at cli it should show a list of available things |
| 08:02.21 | harpal | I have done this http://pastebin.ca/794580 removed that MeetMeCount but still error |
| 08:04.10 | FlatFoot | harpal: just found this To be able to run MeetMe, your Asterisk server needs a Zaptel timer. |
| 08:04.18 | FlatFoot | http://www.voip-info.org/wiki-Asterisk+cmd+MeetMe |
| 08:04.45 | *** join/#asterisk reber (n=reber@LNeuilly-152-21-126-73.w193-253.abo.wanadoo.fr) |
| 08:05.04 | harpal | In my server I dont find MeetMe command at cli |
| 08:05.15 | harpal | I have gone through that link |
| 08:05.53 | FlatFoot | sounds like the install is not complete |
| 08:06.07 | FlatFoot | anyway good luck must go and do some work now |
| 08:06.20 | harpal | FlatFoot, but I dont have zaptel card |
| 08:06.29 | harpal | I am just doing with aoftphones |
| 08:07.00 | harpal | So Is it needed to have zaptel card to work MeetMe |
| 08:08.41 | Qwell | harpal: no, install zaptel and load ztdummy |
| 08:10.43 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 08:10.47 | harpal | Qwell, But without hardware Can I install zaptel? because I have tried once but its giving error so I havent installed it again |
| 08:10.55 | Qwell | yes |
| 08:10.57 | *** join/#asterisk ming_zym (i=ming_zym@nat/yahoo/x-8f015237e2a2ecca) |
| 08:11.36 | harpal | After installing asterisk Can i install zaptel? |
| 08:11.57 | *** join/#asterisk rati (n=rati@124.125.255.223) |
| 08:13.35 | harpal | Qwell, or I have to remove asterisk and than First install zaptel and than install asterisk? |
| 08:25.44 | phix | ey, is there any quick way to add a codec into asterisk without recompiling? |
| 08:26.10 | phix | I want to add ilbc back into the Debian Etch package of Asterisk |
| 08:33.18 | *** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 08:34.44 | *** join/#asterisk mirco (n=mirco@pd95b6029.dip0.t-ipconnect.de) |
| 08:38.05 | *** join/#asterisk Aurs (n=Aurs@1ult2p8.ip.hipercom.no) |
| 08:39.58 | *** join/#asterisk yang (i=yang@static-ip-62-75-255-124.inaddr.intergenia.de) |
| 08:40.12 | yang | I have these 2 values in sip.conf, but I cannot make the call http://openpaste.org/en/4016/ |
| 08:41.08 | yang | <PROTECTED> |
| 08:41.09 | yang | <PROTECTED> |
| 08:49.41 | *** join/#asterisk socken23 (n=azrael@ip-213-189-154-029.fix.magnet.ch) |
| 08:50.02 | *** join/#asterisk jetlagmk2 (i=jetlag@70.17.37.23) |
| 08:51.44 | socken23 | Hey my logfile is 2.6GB big :-) How can I prevent debugging messages from being entered in /var/log/asterisk/full ?? |
| 08:52.33 | *** join/#asterisk darkskiez (i=mhb@cpc1-broo3-0-0-cust659.renf.cable.ntl.com) |
| 08:53.19 | obnauticus | ln -s /dev/null /var/log/asterisk/full |
| 08:53.35 | creativx | logger config |
| 08:53.46 | obnauticus | You can use my ghetto method too |
| 08:53.46 | obnauticus | lol. |
| 08:54.23 | creativx | indeed if you dont want any logging at all |
| 08:54.27 | creativx | just ship it over to devnull |
| 08:54.37 | creativx | but since he said prevent debugging messages.. he might want some of em |
| 08:54.37 | socken23 | hehe, no, that's not what I'm looking for :-) |
| 08:54.37 | creativx | :) |
| 08:54.44 | obnauticus | lol. |
| 08:54.53 | obnauticus | Would it not prevent debugging messages? |
| 08:55.02 | creativx | it would prevent any messages |
| 08:55.05 | creativx | :) |
| 08:55.09 | socken23 | it would.. ok, let me ask again: ONLY debug messages |
| 08:55.10 | obnauticus | Which includes debugging messages |
| 08:55.13 | obnauticus | LOL |
| 08:55.18 | creativx | do a grep |
| 08:55.20 | creativx | ;) |
| 08:56.58 | socken23 | okok, and any way to configure logging?? |
| 08:57.05 | creativx | indeed |
| 08:57.18 | obnauticus | lol. |
| 08:57.19 | creativx | try etc asterisk logger.conf |
| 08:58.04 | socken23 | thanks! |
| 08:58.14 | socken23 | finally! LOL |
| 08:58.23 | creativx | its friday |
| 08:58.25 | creativx | :) |
| 08:58.36 | socken23 | hehe, right! better than monday! |
| 08:59.17 | creativx | thats right |
| 08:59.18 | socken23 | ok found it, thanks alot! |
| 08:59.49 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 09:11.46 | *** join/#asterisk shinao1 (n=shinao1@217.20.242.50) |
| 09:12.39 | *** join/#asterisk ming_zym (i=ming_zym@nat/yahoo/x-9330d27dfcfeccb0) |
| 09:14.31 | *** join/#asterisk stkn|work (n=stkn|wor@gentoo/developer/pdpc.active.stkn) |
| 09:16.36 | SwK | *YAWN* |
| 09:21.07 | *** join/#asterisk KermitTheFragger (n=siepkes@118-197.bbned.dsl.internl.net) |
| 09:27.20 | *** join/#asterisk _ys (n=yuri@236-069.nat.mns.ru) |
| 09:38.08 | *** join/#asterisk duckz (n=duckz@85.204.47.228) |
| 09:41.15 | *** join/#asterisk fujin (n=aj@unaffiliated/fujin) |
| 09:51.32 | *** join/#asterisk reber (n=reber@LNeuilly-152-21-126-73.w193-253.abo.wanadoo.fr) |
| 10:06.16 | yang | 09:40 < yang> I have these 2 values in sip.conf, but I cannot make the call http://openpaste.org/en/4016/ |
| 10:06.19 | yang | 09:41 < yang> -- SIP/btctrunk-08191b90 is circuit-busy |
| 10:06.28 | yang | 09:41 < yang> == Everyone is busy/congested at this time (1:0/1/0) |
| 10:09.42 | harpal | yang, Just change that type=peer to friend in both conf, Try that Is it woriking? Have you doen changes in extension.conf? |
| 10:13.38 | *** join/#asterisk ming_zym (i=ming_zym@nat/yahoo/x-deca7d5932797d74) |
| 10:15.23 | yang | harpal: my extensions go like this http://openpaste.org/en/4018/ |
| 10:16.30 | bereghelea | Native bridging SIP - i have no voice |
| 10:16.53 | yang | i thought of doing IAX, instead of SIP, however I don't know the proper configuration |
| 10:16.54 | *** join/#asterisk sergee (n=serg@voip1.west-call.com) |
| 10:17.36 | yang | harpal: it still isnt working |
| 10:18.10 | yang | sip show peers will show the active peer |
| 10:19.59 | harpal | yang, you have context uplink in extension.conf so add context=uplink in both sip user |
| 10:21.17 | harpal | http://openpaste.org/en/4019/ |
| 10:21.48 | *** join/#asterisk Keltus (i=Keltus@about/cooking/nakedchef/beefstew/Keltus) |
| 10:24.23 | *** join/#asterisk rati (n=rati@124.125.255.223) |
| 10:27.12 | yang | harpal: thanks a lot works now |
| 10:27.42 | yang | harpal: when I define host=IP in the sip.conf it means it will only connect from that host and no other.. |
| 10:27.50 | yang | so i left it without password |
| 10:31.43 | *** join/#asterisk ghenry (n=ghenry@suretecsystems.plus.com) |
| 10:31.58 | deever | hi |
| 10:33.14 | deever | while compiling asterisk on a slug (with debian), suddenly i get this strange error: http://rafb.net/p/fnt9Kb15.html |
| 10:33.32 | yang | harpal: look now i got a bunch of errors when i establish a call http://openpaste.org/en/4020/ |
| 10:33.57 | deever | what's wrong? until here, there wasn't any problem with the arch? |
| 10:34.00 | *** join/#asterisk RoyK (n=roy@80.239.107.70) |
| 10:35.55 | badcfe | when i do make menuconfig i see in the Modules menu that nothing is selected; apps, cdr, channels, codecs etcetera. below its written "Depends on: gnu_ld(E)". |
| 10:35.58 | badcfe | is this normal |
| 10:36.07 | badcfe | should i enable them (i think i want them) .... |
| 10:36.07 | badcfe | ? |
| 10:36.53 | badcfe | did it mean that theyr disabled cause gnu_ld wasnt found durint ./configure ? |
| 10:39.34 | badcfe | more precisely, its in the "Module Embedding" menu. hmm, so its not about the normal modules? |
| 10:40.31 | yang | harpal: its calling it, but i dont hear the ringing on the phone |
| 10:41.06 | yang | its becouse of the firewall, i think |
| 10:41.51 | harpal | yang, replace r instead of t in Dial of your extensio.conf |
| 10:42.15 | harpal | Dial(SIP/${EXTEN}@btctrunk,60,r) |
| 10:47.03 | yang | it rings now, and when i pick up on the other side i get error |
| 10:47.17 | yang | probably an issue of our firewall |
| 10:47.51 | harpal | Yang, no I think you have to just change that dial plan in extension.conf |
| 10:47.54 | yang | becouse the SIP PHONE is shown as unreachable |
| 10:48.14 | yang | ip is shown |
| 10:50.01 | yang | -- Got SIP response 503 "Unavailable" back from 212.103.133.6 |
| 10:50.48 | harpal | yang try http://openpaste.org/en/4021/ in your extension.conf |
| 10:51.49 | yang | which line is changed? |
| 10:53.11 | harpal | Whole [uplink] section has changed |
| 10:53.53 | harpal | and than you can call to 100 for btctrunk and 200 for asterisk-jan |
| 10:54.36 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 10:55.02 | *** join/#asterisk zig5000 (n=kirill@qoxta68-gw.Spb.gldn.net) |
| 10:55.08 | zig5000 | hello |
| 10:56.25 | *** join/#asterisk _ys (i=ys@91.151.196.254) |
| 10:57.18 | zig5000 | I have trouble with asterisk. I try to configure asterisk queues for load configuration from mysql. I want configure realtime queues in asterisk. I write to extconfig.conf and asterisk when start print. Binding queues to mysql. But show queues is empty |
| 10:58.57 | yang | harpal: on the other side the context is called BTC and it then routes the call to the third provider |
| 10:59.02 | zig5000 | Why asterisk no load queues configuration ? |
| 11:00.33 | harpal | yang, I tell you this for only two users asterisk-jan and btctrunk. If you are calling to other than context must be same |
| 11:00.59 | *** join/#asterisk UserReg_CL (n=COB@pc-248-68-47-190.cm.vtr.net) |
| 11:01.15 | harpal | I am also a newbie |
| 11:01.20 | UserReg_CL | :) |
| 11:01.21 | yang | ok |
| 11:01.29 | UserReg_CL | I am also newbie |
| 11:01.31 | zig5000 | if i try "realtime queues 1 1". I see my configuration |
| 11:02.05 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 11:08.30 | deever | while compiling asterisk on a slug (with debian), suddenly i get this strange error: http://rafb.net/p/fnt9Kb15.html |
| 11:08.37 | deever | what's wrong? until here, there wasn't any problem with the arch? |
| 11:12.57 | yang | harpal: i solved it, it was a missing codec issue : |
| 11:15.43 | harpal | yang, good |
| 11:17.27 | zig5000 | Is My question very difficult ??? Or queues in mysql nothing use ? |
| 11:19.36 | harpal | Can any one see http://pastebin.ca/794711 and tell me whats problem ? |
| 11:19.49 | harpal | I cant do conference with MeetMe |
| 11:19.52 | *** join/#asterisk qdk (n=qdk@85.235.253.139) |
| 11:20.22 | *** join/#asterisk myiagy (n=myiagy@201-67-138-60.bnut3703.dsl.brasiltelecom.net.br) |
| 11:21.58 | *** join/#asterisk saftsack (n=oliver@pD9E04BE1.dip.t-dialin.net) |
| 11:22.17 | zig5000 | harpal: Show meetme.conf |
| 11:23.46 | harpal | zig5000, http://pastebin.ca/794714 |
| 11:24.13 | *** join/#asterisk Comete (n=Comete@krypton.saint-lo.fr) |
| 11:24.49 | zig5000 | is zaptel module load ? |
| 11:25.52 | harpal | I dont have any card so i have just install driver zaptel. and than install asterisk. So I think no |
| 11:25.57 | zig5000 | harpal: MeetMe not worked if zaptel not load. If you not have digium cards. load ztdummy module |
| 11:26.30 | harpal | how to load ztdummy module? |
| 11:26.36 | tzafrir | zig5000, to check if zaptel timing works, run zttest -v for a while |
| 11:26.38 | zig5000 | modprobe ztdummy |
| 11:27.01 | Comete | hi, i use asterisk 1.2 and my server is configured with an isdn card, all is working fine, except when i want to forward an incoming call to an external number (ex: a mobile phone). I use *72 for this which is working fine with internal numbers but not with external... any idea how i can do this ? |
| 11:27.13 | harpal | zig5000, Gives me error |
| 11:27.20 | harpal | WARNING: Error inserting zaptel (/lib/modules/2.6.18.8-klips/misc/zaptel.ko): Invalid module format |
| 11:27.27 | harpal | FATAL: Error inserting ztdummy (/lib/modules/2.6.18.8-klips/misc/ztdummy.ko): Invalid module format |
| 11:27.37 | zig5000 | recompile zaptel |
| 11:28.23 | zig5000 | cd <path zaptel-<version>> |
| 11:28.26 | zig5000 | make |
| 11:28.35 | zig5000 | ooops |
| 11:28.40 | zig5000 | make |
| 11:29.06 | zig5000 | ooops keyboard....trouble =)))) |
| 11:30.13 | harpal | Should recompile fixes problem? |
| 11:30.40 | harpal | or I have to configure that zaptel.conf and other files? |
| 11:31.20 | *** join/#asterisk linuxmaniac (n=linuxman@65.Red-81-32-73.dynamicIP.rima-tde.net) |
| 11:34.22 | *** join/#asterisk orn (n=orn@85.197.193.24) |
| 11:35.32 | orn | Is there an application I can use to check if an entry exists in astDB and set a variable if so, and if not, set a different one? |
| 11:36.34 | tzafrir | zig5000, |
| 11:36.45 | tzafrir | zig5000, is it a zaptel problem to start with? |
| 11:37.16 | tzafrir | hmm... you seem to be building vs. the wrong kernel headers |
| 11:37.17 | orn | something like Set fwdnum to value of ${DB(CF/${ARG1})} if that exists, otherwise set it to original ${EXTEN} ? |
| 11:37.33 | tzafrir | zig5000, what distro? distro kernel or your custom kernel? |
| 11:40.02 | *** join/#asterisk Delvar (n=Delvar@77.240.56.17) |
| 11:40.30 | *** join/#asterisk d-k-t (n=dt@60.176.197.184) |
| 11:41.49 | harpal | I got this error zaptel: version magic '2.6.18.8-klips SMP mod_unload 486 REGPARM gcc-4.1' should be '2.6.18.8-klips SMP mod_unload 686 REGPARM gcc-4.1' while loading tzdummy |
| 11:41.50 | zig5000 | tzafrir: I don know your problem.... |
| 11:42.53 | tzafrir | zig5000, oops, this is harpal's problem |
| 11:42.55 | harpal | What should I do? I have reinstalled zaptel |
| 11:43.06 | tzafrir | harpal, what distro? distro kernel or your custom kernel? |
| 11:43.45 | harpal | tzrif, I have debian. and my custom kernel 2.6.18.8 |
| 11:44.38 | tzafrir | what is the output of: ./install_prereq tes |
| 11:44.41 | tzafrir | what is the output of: ./install_prereq test |
| 11:45.15 | harpal | Where I do this? In which folder? |
| 11:45.22 | *** join/#asterisk oej_ (n=olle@soll4-125.cust.blixtvik.net) |
| 11:45.36 | tzafrir | in the zaptel source directory |
| 11:46.13 | harpal | test completed sucessfully |
| 11:47.08 | tzafrir | what i the output of: uname -r |
| 11:47.25 | deever | while compiling asterisk on a slug (with debian), suddenly i get this strange error: http://rafb.net/p/fnt9Kb15.html |
| 11:47.33 | harpal | 2.6.18.8-klips |
| 11:47.35 | deever | what's wrong? until here, there wasn't any problem with the arch? |
| 11:48.04 | tzafrir | find /lib/modules/2.6.18.8-klips -name zaptel.ko |
| 11:48.30 | harpal | Yes its showing that |
| 11:48.39 | tzafrir | only one? |
| 11:48.43 | harpal | Yes |
| 11:50.02 | harpal | tzafrir, I dont have any zaptel hardware. I want to run MeetMe so it requires zaptel or ztdummy. but while loading ztdummy I got some error |
| 11:50.03 | tzafrir | deever, the error seems to come from building of the gsm codec |
| 11:50.20 | harpal | zaptel: version magic '2.6.18.8-klips SMP mod_unload 486 REGPARM gcc-4.1' should be '2.6.18.8-klips SMP mod_unload 686 REGPARM gcc-4.1' while loading tzdummy |
| 11:50.25 | tzafrir | which is unnecessary if you build on debian: use libgsm-dev |
| 11:50.54 | tzafrir | maybe you need to re-./configure and use some explicit parameters |
| 11:51.07 | tzafrir | Probably a workaround rather than a fix, though |
| 11:51.25 | harpal | tzafrir, Is that for me? |
| 11:51.37 | tzafrir | no, for deevar |
| 11:51.54 | harpal | Here it seems that it has 486 in file and I have 686 so it doesnt matching that? |
| 11:52.05 | tzafrir | harpal, any chance you have reconfigured the kernel build tree after installing the kernel? |
| 11:52.30 | *** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 11:52.39 | mort_gib | Anyone here?? |
| 11:52.54 | harpal | tzafrir, no actually I have made .deb file for my kernel and than install it. |
| 11:53.18 | tzafrir | and for linux-headers as well? |
| 11:53.36 | harpal | Oh I havent installed linux-header |
| 11:53.46 | harpal | I have just installed linux-image |
| 11:54.03 | harpal | tzafrir, Is it required? |
| 11:54.41 | tzafrir | I just wonder where the link /lib/modules/2.6.18.8-klips/build actually points to? |
| 11:55.22 | *** join/#asterisk waKKu (n=eurulo@unaffiliated/wakku) |
| 11:55.35 | tzafrir | It should point to a configured kernel tree with a matching configuration. Or a partial one |
| 11:56.10 | tzafrir | ~Anyone |
| 11:56.10 | jbot | *** anyone: No such nick/channel - and yes, there probably is someone, somewhere, who knows or runs it; that doesn't mean /I/ do. |
| 11:56.17 | harpal | tzafrir, it points to /usr/src/linux-2.6.18.8 |
| 11:56.37 | harpal | where my kernel source is there |
| 11:56.49 | tzafrir | mort_gib, so to answer your question: Anyone's not here. Now did you actually want to ask anything? |
| 11:57.40 | tzafrir | harpal, and what is there? |
| 11:57.46 | tzafrir | ah, ok |
| 11:58.16 | tzafrir | in short, the error message complains that zaptel was built with a different kernel than the one currently running |
| 11:58.47 | mort_gib | No, still waiting for my update.. Or maybe more friendly aliens |
| 11:58.55 | tzafrir | You can check that for modules in the local directory: |
| 11:59.36 | tzafrir | modinfo ./zaptel.ko | grep ^vermagic |
| 12:00.13 | harpal | tzafrir, Which directory? zaptel source? |
| 12:00.30 | tzafrir | yes |
| 12:00.36 | harpal | vermagic: 2.6.18.8-klips SMP mod_unload 486 REGPARM gcc-4.1 |
| 12:00.55 | harpal | So how to change that to 686? |
| 12:02.34 | tzafrir | This comes from your current kernel source tree |
| 12:03.03 | harpal | no this comes from zaptel souce |
| 12:03.32 | tzafrir | you shouldn't just change that. You should make sure that the configuration there matches the one of your running kernel |
| 12:03.32 | *** join/#asterisk ayrjola (n=ayrjola@gw.aller.fi) |
| 12:04.59 | harpal | on my kernel source tree I have vermagic: 2.6.18.8-klips SMP mod_unload 486 REGPARM gcc-4.1 |
| 12:05.58 | harpal | I have done modinfo ./zaptel.ko | grep ^vermagic at /lib/modules/2.6.18.8-klips/misc, where my zaptel.ko |
| 12:06.21 | tzafrir | right |
| 12:06.47 | tzafrir | But what I means is that the vermagic is information about the kernel source (or headers) |
| 12:07.40 | harpal | both place it showing same than Why its giving error while locading tzdummy. should I install kernel-header? |
| 12:10.48 | tzafrir | harpal, you need the kernel configured just as the one you used to build the kernel |
| 12:10.58 | tzafrir | same .config file |
| 12:14.21 | *** join/#asterisk stimpie (n=stimpie@84-104-5-115.cable.quicknet.nl) |
| 12:16.38 | *** join/#asterisk SuurMyy (n=SuurMyy_@195.238.211.98) |
| 12:20.32 | *** join/#asterisk marcan (i=1337@host214-205.cvd.fit.edu) |
| 12:20.34 | harpal | tzafrir, thanks a lot. its working now. I have installed 486 kernel. |
| 12:21.29 | harpal | I have both 686 and 486 kernel .deb file. so installed 486 and it has loaded module and its not giving error while calling to conference |
| 12:21.40 | *** join/#asterisk Zeeek (n=Heh@pdpc/supporter/active/Zeeek) |
| 12:23.18 | Zeeek | wow |
| 12:24.47 | *** join/#asterisk Strom_M (n=strom@ip70-170-60-8.lv.lv.cox.net) |
| 12:26.45 | *** part/#asterisk Zeeek (n=Heh@pdpc/supporter/active/Zeeek) |
| 12:27.17 | *** join/#asterisk Op3r (n=Op3r@222.127.208.10) |
| 12:31.05 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 12:33.00 | *** join/#asterisk michael-i (n=michael-@141.41.40.55) |
| 12:38.46 | *** join/#asterisk fujin (n=aj@unaffiliated/fujin) |
| 12:40.11 | *** join/#asterisk adorah (n=Michael@87.69.57.246.cable.012.net.il) |
| 12:45.30 | *** join/#asterisk MicW (n=michael@dslb-088-074-163-040.pools.arcor-ip.net) |
| 12:45.39 | MicW | hi |
| 12:46.13 | gedoc | anyone awake? |
| 12:46.45 | MicW | can i show a dialplan variable at the asterisk console? |
| 12:47.37 | gedoc | i have 2 asterisk registered to each other via IAX2 but i cannot forward extensions. anyone got me on this? |
| 12:49.28 | RoyK | MicW: afaik you can't. add a noop(var is ${var}) in the dialplan |
| 12:49.35 | MicW | ok |
| 12:51.15 | MicW | xten => _X.,n,Set(ISDN_CALLS_LINE_1=$[${ISDN_CALLS_LINE_1} + 1]|GLOBAL) |
| 12:51.29 | MicW | i have a problem with this expression |
| 12:51.52 | MicW | i have this in different contexts, one for each phone number |
| 12:52.24 | MicW | could it be that ${ISDN_CALLS_LINE_1} accesses a local, not the global variable? |
| 12:56.48 | harpal | gedoc, I have same problem and some one given me link http://forums.whirlpool.net.au/forum-replies-archive.cfm/549784.html |
| 12:57.23 | harpal | But I didnt understand that. I think it may help you if you understand |
| 12:58.38 | gedoc | thanks |
| 13:00.32 | *** join/#asterisk MrChimpy (n=MrChimpy@212.158.8.163) |
| 13:01.55 | tzafrir | MicW, isn't this exactly SetGroup? |
| 13:03.42 | tzafrir | MicW, also: to show a local var, show that channel |
| 13:04.01 | MicW | ? |
| 13:04.54 | tzafrir | Why exactly do you need to count that number? |
| 13:05.35 | MicW | i have an isdn channel. there are 3 numbers assigned to it. they have different extensions and are differently handled. |
| 13:06.00 | MicW | when where are one or more isdn calls (inbound+outbound) all incoming isdn calls are defelcted to a voip number |
| 13:06.20 | *** join/#asterisk marcan (i=1337@host214-205.cvd.fit.edu) |
| 13:06.23 | MicW | when there are 2 isdn calls, all outbound calls are dialed over voip |
| 13:06.43 | MicW | because the isdn channel can handle only 2 concurent calls |
| 13:06.51 | tzafrir | MicW, this looks to me like Dial(Zap/g3) , or whatever |
| 13:07.04 | MicW | what's that? |
| 13:07.08 | tzafrir | put whatever channels you want in group=3 |
| 13:07.20 | tzafrir | ISDN: zaptel? |
| 13:07.21 | *** join/#asterisk defswork (n=andy@77.44.54.34) |
| 13:07.38 | MicW | i dont'd use zaptel. just chan_capi |
| 13:12.33 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 13:13.27 | MicW | i'm goind to debug this next week. thanks and have a nice weekend |
| 13:13.30 | MicW | bye |
| 13:14.35 | harpal | tzafrir, Can you tell me in link http://forums.whirlpool.net.au/forum-replies-archive.cfm/549784.html , What Dick try to tell after line ADD A NEW IAX TRUNK |
| 13:15.20 | harpal | gedoc, Have you find any solution? |
| 13:16.52 | gedoc | i am working on it |
| 13:17.18 | gedoc | i think i tried this solution yesterday which did not work |
| 13:18.32 | harpal | Have you understand that TRUNK? I dont know that how to make IAX TRUNK |
| 13:23.31 | *** join/#asterisk hi365 (n=hi365@mail.pcgeula.co.il) |
| 13:24.12 | *** join/#asterisk flujan (n=flujan@200-160-115-020.static.spo.ctbc.com.br) |
| 13:24.18 | flujan | hi all. :D |
| 13:25.06 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 13:25.13 | gedoc | harpal: ask the channel how to |
| 13:27.23 | harpal | ok |
| 13:27.24 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
| 13:31.26 | harpal | I have made call between two servers but when user from 1st server call to user of 2nd server than It can call but its showing call from GUEST IAX USER. so what should I do so that it disply me correct number instead of GUEST IAX USER |
| 13:31.50 | *** join/#asterisk anonymouz666 (n=anonymou@201.19.134.26) |
| 13:32.29 | waKKu | do u create a user in both asterisks ? |
| 13:36.05 | waKKu | harpal a example how I do it: http://pastebin.ca/794838 |
| 13:38.11 | *** join/#asterisk shinao1 (n=shinao1@217.20.242.50) |
| 13:38.37 | *** join/#asterisk zig5000 (n=kirill@qoxta68-gw.Spb.gldn.net) |
| 13:38.47 | flujan | harpal: so bad. :( |
| 13:38.47 | harpal | waKKu, I have register only one server. |
| 13:39.30 | zig5000 | Who Can help me? asterisk not read configuration queues from mysql. realtime load queues 1 1. show fine |
| 13:40.33 | *** join/#asterisk agx (n=AGX@88.34.216.63) |
| 13:42.46 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 13:45.40 | *** join/#asterisk _Krieger_ (n=krieger@193.39.118.158) |
| 13:45.52 | _Krieger_ | hey, any docs regarding res_esel? |
| 13:46.17 | *** part/#asterisk myiagy (n=myiagy@201-67-138-60.bnut3703.dsl.brasiltelecom.net.br) |
| 13:52.00 | *** join/#asterisk saftsack (n=oliver@p54A7BA5A.dip.t-dialin.net) |
| 13:52.14 | saftsack | hi how far is t38 implemented in asterisk? |
| 13:53.25 | [TK]D-Fender | saftsack: it isn't |
| 13:53.44 | saftsack | http://bugs.digium.com/view.php?id=5090 whats with that? |
| 13:53.54 | saftsack | i just want to do passthrough ... |
| 13:55.13 | [TK]D-Fender | saftsack: that is not * supporting T.38, that is * realizing when to keep out of the way between 2 endpoints that CAN do it. |
| 13:55.48 | saftsack | i have a stock asterisk 1.4 does it do it as a normal behaviour? |
| 13:58.53 | *** join/#asterisk jm|laptop (n=jm|home@zen.jamiem.com) |
| 13:58.56 | saftsack | [TK]D-Fender, i want to take t38modem. is it possible to let asterisk to a sip to openh323 conversion without destroying the t.38 communication? |
| 13:59.47 | [TK]D-Fender | saftsack: Doesn't sound like passthrough if * has to translate the 2 ends does it? |
| 14:00.19 | flujan | guys is it possible to limit the time of a caller that stays in the queue? |
| 14:00.29 | saftsack | thats true but i thought that openh323 and sip are a higher layer than t.38 is |
| 14:00.38 | flujan | for intance, sometimes people join the queue and stay waiting for more than a hour. |
| 14:00.53 | *** join/#asterisk oej (n=olle@soll4-125.cust.blixtvik.net) |
| 14:01.37 | [TK]D-Fender | saftsack: And seriously, pay attention to the DATE on that link! 09-01-05 10:16 <------------ |
| 14:01.46 | [TK]D-Fender | YEARS ago. |
| 14:02.03 | [TK]D-Fender | saftsack: And learn to use GOOGLE like the rest of us, and the WIKI. |
| 14:02.10 | [TK]D-Fender | saftsack: http://www.voip-info.org/tiki-index.php?page=Asterisk%20T.38 |
| 14:02.14 | deever | tzafrir: thanks for the info... |
| 14:02.50 | deever | tzafrir: ...but how can i bypass this error and continue the build? |
| 14:03.07 | deever | with touching the file that should be generated? |
| 14:03.18 | tzafrir | --with-gsm=system ? |
| 14:03.26 | tzafrir | maybe some variation on that |
| 14:03.46 | tzafrir | ./configure --with-gsm=system # that is |
| 14:04.49 | deever | oh no...i don't want to start from the beginnig...it took so much time to get here! :( |
| 14:05.18 | deever | is touching a good alternative? |
| 14:05.24 | *** join/#asterisk gazzerh (n=garryh@79.173.139.34) |
| 14:05.40 | *** join/#asterisk ramonpeek (n=ramonpee@e135042.upc-e.chello.nl) |
| 14:09.47 | *** join/#asterisk __freedom__lover (n=eduardo@clipper.provale.com.br) |
| 14:09.53 | *** join/#asterisk danalien (n=danalien@unaffiliated/danalien) |
| 14:10.18 | tzafrir | deever, you could just disable codec_gsm :-( |
| 14:11.38 | deever | hehe...no, i'll touch these files when im at home |
| 14:11.44 | deever | :) |
| 14:12.54 | *** join/#asterisk grEvenX (n=even@1ult2p8.ip.hipercom.no) |
| 14:23.11 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 14:23.51 | *** join/#asterisk Zeeek (n=IceChat7@pdpc/supporter/active/Zeeek) |
| 14:25.14 | Zeeek | wunderbar |
| 14:28.57 | *** join/#asterisk shinao1 (n=shinao1@217.20.242.50) |
| 14:29.34 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 14:30.49 | *** join/#asterisk demiv (n=demiv@134.42.128.66.PPPoECali.dynamic.telesat.net.co) |
| 14:31.14 | *** join/#asterisk heison (n=heison@CPE000a01d49e6f-CM00111a59bec2.cpe.net.cable.rogers.com) |
| 14:33.01 | [TK]D-Fender | Zeeek: Snickers |
| 14:33.02 | *** join/#asterisk TUNTAP (n=error@BSN-77-58-164.dsl.siol.net) |
| 14:33.21 | Zeeek | Turkey-flavored? |
| 14:35.42 | Zeeek | how slick is this? A free SIM card with unlimited incoming calls |
| 14:37.32 | Zeeek | Is it just me, or is it cold in here? |
| 14:38.49 | Zeeek | anyone from VentureVoip here? |
| 14:39.56 | harpal | waKKu, I have done same thing what you have given now I cant establish the call? earlier its calling |
| 14:42.46 | *** join/#asterisk scream_ (n=stas@pix.postmet.com) |
| 14:44.30 | *** join/#asterisk Aurs (n=chatzill@1ult2p8.ip.hipercom.no) |
| 14:47.18 | dmz | hey, anyone here tried getting a new account on fwd recently? i've tried 2 new accounts and both can sign into the web page fine but asterisk is getting registration refused. an old one i have is working great, but i'm trying to setup a new friend and it's just not registering. any suggestions? |
| 14:47.43 | Zeeek | contact FWD support? |
| 14:47.51 | Zeeek | they're great |
| 14:47.54 | Zeeek | Ed Guy |
| 14:52.22 | flujan | guys, is there a easy way to add hints to a dialplan without writing the lines 12345,hint,SIP/12345? |
| 14:52.42 | flujan | I have about 400 channels and I need to setup one hit for each... :( |
| 14:53.15 | *** join/#asterisk thugwh0re (i=plutten@pimping.at.manwh0re.se) |
| 14:53.51 | tzafrir | flujan, I'm not sure if users.conf tries to do that. If not, write a little script to do that... |
| 14:54.49 | thugwh0re | I get this error when i'm trying to do "make" for addons-1.2: "Makefile", line 44: Missing dependency operator |
| 14:54.59 | thugwh0re | anyone else have this problem? |
| 14:55.13 | tzafrir | thugwh0re, a line without ':' ? |
| 14:55.39 | thugwh0re | ifeq (${OSARCH},Darwin) |
| 14:55.41 | thugwh0re | is on that line |
| 14:56.00 | tzafrir | thugwh0re, try using gmake |
| 14:56.45 | thugwh0re | now i get other errors but it acctually started making now :) |
| 14:59.21 | tzafrir | IIRC the 1.2 addons require some docs reading to build correctly |
| 15:02.13 | harpal | Server 1: |
| 15:02.13 | harpal | iax.conf |
| 15:02.13 | harpal | [vitoria] |
| 15:02.14 | harpal | username=vitoria |
| 15:02.14 | harpal | secret=secret |
| 15:02.14 | harpal | type=friend |
| 15:02.16 | harpal | host=dynamic |
| 15:02.18 | harpal | context=internacional |
| 15:02.20 | harpal | disallow=all |
| 15:02.22 | harpal | allow=gsm |
| 15:02.24 | harpal | Sorry |
| 15:02.46 | tzanger | haha |
| 15:02.50 | tzanger | victoria/secret |
| 15:02.52 | tzanger | very funny |
| 15:03.00 | bereghelea | underwear |
| 15:03.02 | bereghelea | its nice |
| 15:04.05 | harpal | any one help me? |
| 15:04.12 | harpal | I have made call between two servers but when user from 1st server call to user of 2nd server than It can call but its showing call from GUEST IAX USER. so what should I do so that it disply me correct number instead of GUEST IAX USER |
| 15:05.07 | harpal | some hints given by waKKu but Its not working |
| 15:18.13 | [TK]D-Fender | harpal: do not spam the channel again, use a pastebin |
| 15:18.15 | [TK]D-Fender | ~pb |
| 15:18.16 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 15:18.17 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^^^ |
| 15:19.23 | [TK]D-Fender | flujan: 10 second job in a spreadsheet. |
| 15:20.39 | Zeeek | cat free pr0n > #asterisk |
| 15:20.53 | Zeeek | go ahead, I dare you |
| 15:21.46 | harpal | [TK]D-Fender, sorry actually I am pasting it to other and window open is this so Sorry again |
| 15:28.56 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 15:28.56 | *** mode/#asterisk [+o anthm] by ChanServ |
| 15:33.33 | *** join/#asterisk truz_`24 (n=truz_`24@74-129-166-232.dhcp.insightbb.com) |
| 15:36.14 | Zeeek | slow dayz |
| 15:37.31 | *** join/#asterisk b1ch0 (n=ralabiso@static-200-105-209-46.acelerate.net) |
| 15:39.45 | *** join/#asterisk irule (n=irule@200.53.61.4) |
| 15:40.01 | *** join/#asterisk _ShrikE (n=ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 15:40.33 | *** join/#asterisk _Krieger_ (n=krieger@195.114.133.197) |
| 15:45.23 | *** join/#asterisk agx (n=AGX@88.34.216.63) |
| 15:47.54 | agx | anyone from DE here that can share me a sound file (call-forward or call-forwarding) ? :) |
| 15:54.40 | *** join/#asterisk coppice (n=chatzill@102.204.17.210.dyn.pacific.net.hk) |
| 16:02.52 | *** join/#asterisk ghento (n=ghento@d216-232-170-111.bchsia.telus.net) |
| 16:04.39 | *** join/#asterisk E-bola (i=snowman@cpe-76-179-4-233.maine.res.rr.com) |
| 16:04.42 | E-bola | Hi all |
| 16:05.02 | E-bola | Im trying to compile newest 1.4 on debian etch and get the following error: configure: error: C++ preprocessor "/lib/cpp" fails sanity check |
| 16:05.11 | E-bola | How can i fix it? |
| 16:05.26 | js_ | try apt-get install build-essential |
| 16:05.38 | rob0 | Take /lib/cpp to the Emacs psychiatrist? |
| 16:06.12 | E-bola | js: thanks, i remember using that before, i bet it will fix it |
| 16:06.39 | E-bola | Yep it did, thanks |
| 16:08.10 | *** join/#asterisk jm|laptop (n=jm|home@dilbert.jamiem.com) |
| 16:11.18 | De_Mon | I just noticed that a 'dialplan reload' stops at this line: == Parsing '/etc/asterisk/users.conf': |
| 16:11.43 | De_Mon | now, /etc/asterisk/users.conf doesn't exist but thats as far this goes |
| 16:12.18 | E-bola | If you disable components/apps/modules in asterisk |
| 16:12.22 | E-bola | willt he conf files still be installed? |
| 16:12.23 | *** join/#asterisk saint_ (n=saint@c-69-242-118-124.hsd1.nj.comcast.net) |
| 16:16.31 | *** join/#asterisk ghento (n=ghento@d216-232-170-111.bchsia.telus.net) |
| 16:17.42 | harpal | I have done exten => 555,1,MeetMeCount(555,CONFCOUNT) and than I have done exten => 555,2,GotoIf($[${CONFCOUNT} < 2]?3:100) |
| 16:18.12 | harpal | So I am limiting user to only 1, but it can connect to more than 1 suser why? |
| 16:19.41 | De_Mon | harpal you'll have to pastebin the whole section of dialplan you're using a gotoif we have no idea whats at those priorities.... |
| 16:20.02 | harpal | ok |
| 16:20.29 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 16:21.04 | De_Mon | I've always put ""'s around the two arguments of a $[], duno if its required or not.. Do you get the same result for CONFCOUNT 0 and CONFCOUNT 2? |
| 16:21.06 | harpal | http://pastebin.ca/794993 |
| 16:21.17 | harpal | yes |
| 16:21.47 | De_Mon | do $["${CONFCOUNT}" < "2"] |
| 16:24.18 | harpal | ok thats working |
| 16:24.22 | harpal | thanks a lot |
| 16:24.23 | *** join/#asterisk ManxPower (n=manxpowe@71-8-61-95.dhcp.leds.al.charter.com) |
| 16:24.35 | Zeeek | that's a good trick to remember |
| 16:24.36 | De_Mon | np |
| 16:24.56 | De_Mon | ;exten => h,n,Exec(${IF($["${CUT(CHANNEL,/,1)}" != "Local"]:GoSub(nextnum,del-${CONFNUM},1)}) |
| 16:25.33 | De_Mon | do you see anything wrong with this line? Asterisk crashes whenever it tries to execute :) |
| 16:27.57 | *** join/#asterisk BadPacket (n=John@unaffiliated/badpacket) |
| 16:28.21 | E-bola | haha cool make insall downloads the extra sound packages it self |
| 16:28.26 | E-bola | never seen that before :) |
| 16:29.20 | ManxPower | lyrics are NSFW http://www.youtube.com/watch?v=yRfvSOuwM_4 |
| 16:29.24 | ManxPower | deeps! |
| 16:30.37 | ManxPower | De_Mon: if asterisk crashes with that line, then it's a problem with Asterisk. However, why not use ExecIf? |
| 16:30.44 | Siya | any advice of which svn code of Asterfax I'd best use? |
| 16:31.02 | ManxPower | Siya: I don't recommend using SVN |
| 16:31.14 | Siya | ManxPower: no? |
| 16:31.39 | Siya | dbian pkg download and manual install as stated on the asterfax site? |
| 16:32.55 | ManxPower | no. download the released source |
| 16:33.33 | *** join/#asterisk Navion (n=billp@75-105-41-123.cust.wildblue.net) |
| 16:34.27 | ManxPower | Siya: this is Asterisk, not Asterfax (what IS that anyway)? |
| 16:36.34 | *** part/#asterisk E-bola (i=snowman@cpe-76-179-4-233.maine.res.rr.com) |
| 16:41.10 | *** join/#asterisk CunningPike (n=arodgers@204.239.12.183) |
| 16:42.32 | badcfe | is it possible to pass a channel variable over to the custom cdr format? |
| 16:45.31 | *** join/#asterisk qdk (n=qdk@0x573fe57e.bynqu2.broadband.tele.dk) |
| 16:46.12 | ManxPower | badcfe: I don't see why that would be a problem. |
| 16:46.22 | *** join/#asterisk rpm (n=russell@75.153.47.179) |
| 16:46.37 | Zeeek | http://x2z.eu is open, alive sort of, after all that turkey, and ready to receive you with a glass of Riesling and a voip joke or tow |
| 16:46.45 | De_Mon | ManxPower something strange is going on, I replaced it with another noop and it still woln't extensions reload correctly... Anyways I'm not using ExecIf() because it doesn't handle the ,'s in the GoSub without escaping... |
| 16:46.57 | Zeeek | wait, no it's too erarly |
| 16:47.12 | ManxPower | OK, why not GosubIf then? |
| 16:47.57 | ManxPower | with Exec, you will be execing an empty string if the ${IF...} is false |
| 16:50.55 | *** join/#asterisk honeybeebuzz (n=bee@206-248-138-47.dsl.teksavvy.com) |
| 16:52.09 | De_Mon | heh, because I wasn't thinking straight when I added the If expression I guess |
| 16:54.05 | Siya | ~asterfax |
| 16:54.29 | Siya | hehe |
| 16:54.33 | Siya | ManxPower: http://www.voip-info.org/wiki-Asterisk+fax |
| 16:55.23 | *** join/#asterisk honeybeebuzz (n=bee@206-248-138-47.dsl.teksavvy.com) |
| 16:56.38 | ManxPower | I don't really see how that relates to installing Asterisk from a package or from source. |
| 17:02.41 | *** join/#asterisk flujan (n=flujan@200-160-115-020.static.spo.ctbc.com.br) |
| 17:03.02 | flujan | [TK]D-Fender: yeap... I am doing it with a python script now.. :D |
| 17:04.51 | flujan | guys, which is the difference between sippeers and sipusers on the extconfig.conf |
| 17:05.14 | De_Mon | peers are called and users make the calls or visaversa I can't keep em straight |
| 17:05.34 | Corydon76-dig | De_Mon: that's correct |
| 17:05.35 | ManxPower | in 1.4 peers can make and receive calls. |
| 17:07.20 | duki | not very clear this sip strory |
| 17:08.20 | duki | It seems enough subtil to make people confused. |
| 17:09.11 | *** join/#asterisk bantu (n=Miranda@p54A32456.dip0.t-ipconnect.de) |
| 17:11.33 | *** join/#asterisk snazm (n=snazm@78.146.230.128) |
| 17:12.45 | Siya | ManxPower: I wasn't asking about how to install asterisk |
| 17:13.11 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 17:13.19 | Siya | merely inquired whether someone knew how to best tackle installing asterfax on debian |
| 17:13.26 | Siya | and asterfax != asterisk |
| 17:14.08 | *** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 17:14.23 | *** part/#asterisk hardwire (n=bip@rdbck-6096.palmer.mtaonline.net) |
| 17:15.17 | *** part/#asterisk snazm (n=snazm@78.146.230.128) |
| 17:15.53 | *** join/#asterisk analyysi (n=ayrjola@cs181173201.pp.htv.fi) |
| 17:16.09 | *** join/#asterisk Strom_M (n=strom@ip70-170-60-8.lv.lv.cox.net) |
| 17:17.05 | k0sm|k0 | anyone know if is it possible use asterisk with an embedded database ? storing in text files... or similar... just for minimal addressbook... |
| 17:17.09 | *** join/#asterisk ajohnstone (n=ajohnsto@host81-134-184-212.in-addr.btopenworld.com) |
| 17:20.00 | Zeeek | Join us http://voipusersconference.org |
| 17:20.11 | Zeeek | for asterisk-related talk |
| 17:22.43 | De_Mon | k0sm|k0 you mean astdb? |
| 17:23.37 | *** join/#asterisk honeybeebuzz (n=bee@206-248-138-47.dsl.teksavvy.com) |
| 17:23.38 | k0sm|k0 | De_Mon: perhaps |
| 17:23.50 | *** join/#asterisk Strom_M (n=strom@ip70-170-60-8.lv.lv.cox.net) |
| 17:23.57 | k0sm|k0 | but i don't find anything about |
| 17:26.21 | *** join/#asterisk Navion (n=billp@75-105-41-123.cust.wildblue.net) |
| 17:27.39 | *** join/#asterisk gazzerh (n=garryh@host-212-158-227-149.bulldogdsl.com) |
| 17:27.52 | De_Mon | k0sm|k0 where are you looking? CLI> show function DB |
| 17:28.30 | *** part/#asterisk reber (n=reber@LNeuilly-152-21-126-73.w193-253.abo.wanadoo.fr) |
| 17:28.53 | ManxPower | k0sm|k0: "show application dbput" |
| 17:29.00 | ManxPower | or show function DB |
| 17:29.03 | *** join/#asterisk BrokenNoze (n=root@host81-149-254-218.in-addr.btopenworld.com) |
| 17:29.14 | ManxPower | also "show applications" and "show functions" |
| 17:30.05 | BrokenNoze | Hi. i'm having some serious dificulties with DTMF. with the exception of dtmfmode= setting is there anything else in the asterisk config i could try changing to get this to work? |
| 17:32.18 | De_Mon | BrokenNoze that would be the first thing to change |
| 17:32.45 | k0sm|k0 | De_Mon, ManxPower.. oh ;D |
| 17:33.16 | BrokenNoze | De_Mx: I've tried all settings. nothing. i'm using polycom 601's but with my wifi Hitachi's everything works great |
| 17:33.27 | k0sm|k0 | De_Mon: i'm searching in google, and pdf book... but all tells about sql db |
| 17:34.03 | *** join/#asterisk honeybeebuzz (n=bee@206-248-138-47.dsl.teksavvy.com) |
| 17:35.11 | Navion | Has anyone set up incoming caller ID with Sangoma FXO's? |
| 17:35.11 | k0sm|k0 | ManxPower: i find only dbdel and dbdeltree (from core show applications) |
| 17:35.21 | k0sm|k0 | no dbput or *get... |
| 17:36.30 | *** part/#asterisk honeybeebuzz (n=bee@206-248-138-47.dsl.teksavvy.com) |
| 17:37.47 | k0sm|k0 | ManxPower: ok, now it's set(db...) |
| 17:37.51 | k0sm|k0 | http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+DBput |
| 17:37.54 | k0sm|k0 | and so on.. |
| 17:37.55 | k0sm|k0 | tnx ;D |
| 17:37.55 | De_Mon | Koshatul really? I just did a google search for 'astdb' and the first link is http://www.voip-info.org/wiki/view/Asterisk+database |
| 17:38.01 | De_Mon | oops k0sm|k0 |
| 17:38.11 | *** join/#asterisk vgster (n=vgster@cpc2-ledn1-0-0-cust826.leed.cable.ntl.com) |
| 17:38.28 | De_Mon | granted that page is outdated, but there it is none the less |
| 17:38.46 | tzafrir | Navion, well, it's zaptel |
| 17:38.55 | tzafrir | have you set callerid=recieved ? |
| 17:39.27 | tzafrir | in zapata.conf |
| 17:39.32 | Navion | tzafrir: In zaptel.conf? |
| 17:40.10 | Navion | tzafrir: OK, any arguments about which CID method? |
| 17:40.12 | ManxPower | Navion: zapata.conf and zaptel.conf are two different files. |
| 17:40.27 | ManxPower | Navion: which method depends on what your telco does. |
| 17:40.39 | ManxPower | (which is usually country specific, but can be carrier specific) |
| 17:41.37 | *** join/#asterisk honeybeebuzz (n=bee@206-248-138-47.dsl.teksavvy.com) |
| 17:41.38 | Navion | This is in North America so I'm assuming all the telco's use the same method |
| 17:42.27 | tzafrir | so the defaults should work, indeed |
| 17:42.33 | ManxPower | then you would not set a callerid mentod. |
| 17:42.39 | *** join/#asterisk ZenBSDi (n=zenbsdi@unaffiliated/ZenBSDi) |
| 17:42.44 | tzafrir | if you set callerid=asreceived ;(sp?) |
| 17:43.03 | ZenBSDi | spic |
| 17:43.14 | ManxPower | BTW, I assume by "north america" you mean USA or Canada? Mexico is part of North America (as is Beleze). |
| 17:43.19 | ZenBSDi | SPelling Is Correct =p |
| 17:44.21 | Navion | Not sure about Beleze but I think US, Canada and Mexico all use the same telco standards. |
| 17:44.25 | De_Mon | what the hell is going on |
| 17:44.25 | *** part/#asterisk honeybeebuzz (n=bee@206-248-138-47.dsl.teksavvy.com) |
| 17:44.31 | *** join/#asterisk honeybeebuzz (n=bee@206-248-138-47.dsl.teksavvy.com) |
| 17:44.34 | *** part/#asterisk honeybeebuzz (n=bee@206-248-138-47.dsl.teksavvy.com) |
| 17:44.40 | ManxPower | Navion: Mexico uses E-1s, I believe. |
| 17:45.16 | De_Mon | are you required to have Hangup in the h extension? |
| 17:45.45 | Navion | Really... I am a little supprised by that. |
| 17:45.54 | ZenBSDi | depends. de_mon |
| 17:46.19 | ManxPower | according to one page, only about half of mexico use FSK for callerid |
| 17:48.19 | ManxPower | De_Mon: no |
| 17:48.23 | ZenBSDi | de_mon, for example.. in my to-pstn section.. I don't list a hangup.. thats just dial out.. but in my internal context .. I put a hangup just to make sure after someone leaves a message it ends |
| 17:49.12 | ManxPower | once you get to exten h the call is alread hungup. It's like asking if you should turn on your car alarm after it was stolen. |
| 17:49.42 | ZenBSDi | for a high call volume setup.. I recommend using hangups in certain areas just so you can keep the traffic going.. but for lows.. don't worry about it |
| 17:50.58 | *** join/#asterisk skirmisha (n=viki@87-126-55-7.btc-net.bg) |
| 17:51.17 | skirmisha | guys, any ideas if sip is working with dundi? |
| 17:51.44 | ManxPower | skirmisha: your search of the wiki did not turn up any info? |
| 17:52.11 | ZenBSDi | http://www.voip-info.org/wiki/view/DUNDi+Enterprise+Configuration+SIP |
| 17:52.18 | ZenBSDi | there ya go skirmisha |
| 17:52.34 | skirmisha | thanks a lot |
| 17:53.15 | ZenBSDi | where's that lame wanna be * master [TK]D-Fender ;) |
| 17:53.31 | [TK]D-Fender | ~cluebat ZenBSDi |
| 17:53.32 | jbot | ACTION pulls out a ClueBat (tm) and thwaps ZenBSDi. |
| 17:53.36 | [TK]D-Fender | uNF! |
| 17:53.42 | De_Mon | Yeah thats what I thought, in exten => when your done its redundant to Hangup() |
| 17:54.07 | ZenBSDi | sup bro |
| 17:54.19 | [TK]D-Fender | Stuffed from off pizza party |
| 17:54.22 | De_Mon | I've somehow created a context that screwes up the extensions.conf parser and adding as the last priority prevents the problem |
| 17:54.36 | De_Mon | adding hangup |
| 17:54.57 | ManxPower | De_Mon: you prolly did something stupid with wildcards. |
| 17:55.08 | De_Mon | I'm trying to reproduce, and #including the file also prevents the problem |
| 17:55.10 | *** join/#asterisk honeybeebuzz (n=bee@206-248-138-47.dsl.teksavvy.com) |
| 17:55.11 | ManxPower | like using exten => _. for example |
| 17:55.18 | De_Mon | no, nothing like that |
| 17:55.31 | *** part/#asterisk honeybeebuzz (n=bee@206-248-138-47.dsl.teksavvy.com) |
| 17:55.39 | ZenBSDi | De_Mon, the same goes for my internal context.. I need a hangup or else I end up with the line still being tied up after someone leaves a message in the voicemail for the extension |
| 17:56.35 | *** join/#asterisk honeybeebuzz (n=bee@206-248-138-47.dsl.teksavvy.com) |
| 17:56.38 | ManxPower | ZenBSDi: you do that in exten => h ? |
| 17:57.08 | ManxPower | hangups are VERY useful outside of exten => h, that is for sure. |
| 17:57.16 | ManxPower | Well, app_hangup, at least |
| 17:57.23 | *** part/#asterisk honeybeebuzz (n=bee@206-248-138-47.dsl.teksavvy.com) |
| 18:03.20 | *** part/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 18:09.18 | [TK]D-Fender | ManxPower: they are pretty much useless IN "h". "h" only gets called when the channel is ALREADY about to completely go away... |
| 18:11.24 | De_Mon | This is weird, I can't reproduce the trouble in another context |
| 18:11.26 | De_Mon | http://pastebin.com/d13059d4a |
| 18:12.22 | De_Mon | if I uncomment line 50, the dialplan loads cleanly. if I comment out line 61, the dialplan loads cleanly. |
| 18:12.37 | De_Mon | If I comment out line 30 the dialplan loads cleanly... |
| 18:12.54 | De_Mon | or line 28 |
| 18:14.42 | ManxPower | define "leads cleanly" |
| 18:14.50 | ManxPower | loads cleanly, that is |
| 18:15.13 | De_Mon | the last line of extensions reload is: == Parsing '/etc/asterisk/users.conf': Found |
| 18:15.27 | De_Mon | When its not "clean" it stops at users.conf': |
| 18:15.38 | ManxPower | why do you have a users.conf? |
| 18:15.48 | De_Mon | no "Found" and hitting line 30 crashes asterisk |
| 18:16.05 | ManxPower | De_Mon: Then you need to report a BUG. |
| 18:17.01 | ManxPower | you can waste your time by running around flapping your arms and saying "I'm a chicken". That doesn't make you a chicken. You can spend all day trying to figure out what your problem is, but we already know what the problem is -- a bug in Asterisk. |
| 18:17.04 | De_Mon | I know, but I cant figure out how to reproduce it |
| 18:17.29 | ManxPower | changing extensions.conf and issuing a "extensions reload" should not crash Asterisk EVER |
| 18:18.01 | De_Mon | reloading doesn't, trying to execute a certain priority in the dialplan does. |
| 18:18.13 | ManxPower | De_Mon: what part of "don't use the Hangup() application in exten => h did you not understand? |
| 18:18.17 | De_Mon | its all very strange and confusing |
| 18:18.35 | coppice | Oh, if only we hadn't figured out how to reproduce, we could have had such a quiet life :-) |
| 18:18.36 | ManxPower | De_Mon: what version of Asterisk? |
| 18:18.42 | De_Mon | 1.4.13 |
| 18:19.21 | Navion | ManxPower: Should it be callerid=asreceived (right now it's set to yes). |
| 18:19.53 | ManxPower | Navion: callerid=yes is probably not a valid setting. |
| 18:20.04 | ManxPower | usecallerid=yes is valid, of course. |
| 18:21.01 | Navion | ManxPower: Sorry, right usecallerid=yes is the current setting. Should I change that? |
| 18:21.39 | De_Mon | it may not be crashing * but just the CLI its all very strange and requires more testing :( |
| 18:21.48 | ManxPower | De_Mon: you realize that Hangup() runs exten => h, right, which hangs up the call, which runs exten =>h, which runs hangup, which runs exten =>/ |
| 18:21.57 | _ShrikE | Is it possible to use custom cdr fields with mysql or odbc? |
| 18:22.35 | ManxPower | De_Mon: can you be ANY more vague please? |
| 18:22.49 | tzanger | ManxPower: oh come on |
| 18:22.51 | tzanger | that wasn't THAT vague |
| 18:22.57 | tzanger | he at least told you it was crashing |
| 18:23.26 | ManxPower | tzanger: Actually now he's saying it's not crashing, now he's saying "the CLI its all very strange" |
| 18:23.35 | tzanger | heh |
| 18:24.40 | ManxPower | De_Mon: you realize that exten => h is only searched in the SAME context as the call, and no other include => context, right? |
| 18:25.08 | [TK]D-Fender | ManxPower: last I checked "Hangup" is explicit. |
| 18:25.48 | ManxPower | Hangup() is rated TV-MA |
| 18:25.49 | ManxPower | 9-_ |
| 18:25.52 | De_Mon | Calling Hangup() within exten h, shouldn't do anything. |
| 18:26.16 | ManxPower | De_Mon: You have your answer. |
| 18:26.34 | De_Mon | What is my question? |
| 18:28.39 | ManxPower | why is it crashing |
| 18:29.10 | De_Mon | I don't have an answer to that, some workarounds yes, but no answer |
| 18:29.19 | ZenBSDi | Hangup() doesn't have an extension, only a killer blade for termination =p |
| 18:29.35 | ZenBSDi | come on folks.. Hangup() fact generator time!~ |
| 18:30.36 | ZenBSDi | Hangup() likes to run after it hears my girlfriend use 5 letters .. b l o w j. |
| 18:30.52 | ManxPower | De_Mon: it is crashing because of a bug in asterisk. exten => h,n,Hangup is triggering that bug. |
| 18:31.41 | [TK]D-Fender | De_Mon>Calling Hangup() within exten h, shouldn't do anything. <- yes... it should terminate the call immediately. |
| 18:34.11 | ManxPower | BTW, does anyone have any sources I can site with regards to Cable TV companies not being allowed to force a person that only wants internet service (and not television service) to have TV service as well? |
| 18:35.07 | ZenBSDi | sadly, you can't do anything about that. |
| 18:35.18 | ZenBSDi | it's their company in that regard. |
| 18:35.38 | ManxPower | ZenBSDi: they are also a govt granted monopoly |
| 18:36.07 | ZenBSDi | if you look closely at the deal.. it's TV with optional internet .. so the company can argue TV is the main thing.. internet is the option and he's paying for the option |
| 18:36.13 | ManxPower | The FCC recently told them they can't have exclusive deals with apartment buildings for cable service. |
| 18:36.43 | ZenBSDi | you could contact the better business bureau |
| 18:36.48 | ZenBSDi | see what they say |
| 18:37.14 | [TK]D-Fender | ZenBSDi: "Go to the Passable-yes-still-legal Business Bureau" ;) |
| 18:38.30 | Navion | ManxPower: That's a real mixed bag. Cable companies are a franchise granted by local government and get their rules set area by area. They are under the FCC for technical standards but the rest is left to the locals. I'm not aware of any litigation that would have a blanket effect. The FCC can't really regulate a closed intra-state system. Unless there is some emission into the air above a set level the FCC isn't involved. |
| 18:39.20 | *** join/#asterisk techie (n=techie@adsl-76-214-9-1.dsl.lsan03.sbcglobal.net) |
| 18:39.20 | ManxPower | Navion: Um, the feds regulate any intra-state thing anytime they want to. |
| 18:39.50 | ManxPower | The whole "states rights" thing has not been valid for a hundred years at least. |
| 18:40.45 | ManxPower | The FCC has ALREADY invalidated any exclusivity contracts the cable companies may have with building owners (apartment buildings, specifically) |
| 18:41.32 | *** part/#asterisk Zeeek (n=IceChat7@pdpc/supporter/active/Zeeek) |
| 18:41.54 | Navion | ManxPower: That'll get challenged. Sounds like they overstepped. |
| 18:42.20 | ManxPower | Navion: You mean like the challanges to FCC regulation of local phone service? |
| 18:43.23 | ManxPower | Also, IIRC, internet service was ruled as "interstate" years ago. |
| 18:44.08 | *** join/#asterisk _ys (n=yuri@80.70.236.69) |
| 18:44.52 | Navion | ManxPower: Local phone service that connectects to inter-state systems are fare FCC turf. We (when I worked at a common carrier) use to connect circuits to inter-state circuits through a switch so we could use the less costly FCC tarrifs even if the inter-state circuit never got used. |
| 18:45.26 | ManxPower | *nod* Sounds much like local internet service connecting to an interstate network |
| 18:46.05 | ManxPower | But this still does not give me a source to cite. Thanks anyway |
| 18:47.29 | De_Mon | I quit. I touched /etc/asterisk/users.conf when i first found the error. now, removing that file gets rid of it too. stupid app |
| 18:47.50 | ManxPower | De_Mon: why were you using users.conf? |
| 18:47.54 | Navion | ManxPower: That's true. Have you tried www.fcc.gov? I have found that lately the federal web sites search engines have gotten pretty good. |
| 18:49.05 | ManxPower | nope, didn't try it. Last time I tried it, the search SUCKED |
| 18:50.12 | Navion | I know the FAA serches are becoming very helpful. I suspect the FCC ones are as well. |
| 18:50.55 | De_Mon | ManxPower Because the dialplan was trying to parse it and never returned "Not Found" or "Found" so I figured i'd create it to try the "Found" path and it didn't help. now I'm using the same exact dialplan that caused the original investigation and its not doing it! |
| 18:51.01 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 18:52.16 | De_Mon | I've copied the entire dialplan to a test system, and I can't reproduce it any which way... |
| 18:52.25 | Navion | ManxPower: Back to my origonal question this morning, I have usecallerid=yes and callerid=asreceived. Should there be anything else to set? Still not getting incomming caller ID information. |
| 18:52.52 | ManxPower | Navion: that should be all that is required. How are you connecting to the telco? |
| 18:53.32 | Navion | Sangoma FXO cards to a 2W co line. |
| 18:53.35 | *** join/#asterisk DrAk0 (n=ljd@nelug/coreteam/luisjose) |
| 18:57.45 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 18:57.56 | ManxPower | Navion: do you have callerid service on the line? |
| 18:58.06 | ManxPower | if you plug in a normal analog phone does callerid work? |
| 19:00.00 | Navion | ManxPower: Pretty sure I do, when I had a Panasonic cordless phone connected to the line it displayed the calling party information. |
| 19:00.38 | ManxPower | Navion: what file are you putting these settings in? |
| 19:02.18 | ManxPower | what exact file and exact path? |
| 19:02.22 | Navion | ManxPower: /etc/asterisk/zapata.conf |
| 19:02.46 | ManxPower | and are those settings under the [channels] section and BEFORE the channel= lines? |
| 19:03.02 | ManxPower | well channel => line |
| 19:04.03 | *** join/#asterisk angom (n=angom@200.38.31.239.dsl.dyn.telnor.net) |
| 19:05.49 | Navion | ManxPower: It's under the [channels] section but I don't see channel = line |
| 19:06.53 | ManxPower | if you don't have a channel => line then you do not have asterisk configured correct.y put a copy of zapata.conf on pastebin.ca |
| 19:07.44 | *** join/#asterisk [hC] (n=hardcore@66.119.167.162) |
| 19:08.25 | [TK]D-Fender | ManxPower: ... lack of that line, and having working channels should set off a rather particular alarm with you ;) |
| 19:08.48 | ManxPower | [TK]D-Fender: hence wanting to see the file. |
| 19:09.06 | ManxPower | I think I smell the stench of a GUI |
| 19:09.59 | ManxPower | [TK]D-Fender: I hate helping slow responders. |
| 19:10.01 | Navion | ManxPower: ?? pastebin.ca ?? Website? FTP server? |
| 19:10.05 | ManxPower | ~pb |
| 19:10.06 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 19:11.27 | Navion | Seems to be waiting a long time to connect to pastebin.ca |
| 19:11.28 | *** join/#asterisk Entr4nced (n=IMG001@adsl-70-238-245-18.dsl.bcvloh.sbcglobal.net) |
| 19:11.51 | [TK]D-Fender | Navion: then pick ANOTHER ONE |
| 19:12.35 | ManxPower | [TK]D-Fender: he's going to regret being slow when I suddenly say "I have to leave to do errands" and he doesn't have a fix. |
| 19:13.18 | *** join/#asterisk masus (n=burdan@88.248.73.2) |
| 19:13.26 | masus | "stale nonce received from" anyone an idea |
| 19:13.27 | masus | ? |
| 19:13.49 | ManxPower | masus: you can ignore that unless there are many of them |
| 19:14.10 | masus | there are many |
| 19:14.23 | Navion | www.pastebin.com |
| 19:14.23 | masus | 30 -40 next to next |
| 19:14.26 | ManxPower | [TK]D-Fender: I had to go to the #exim channel the other day. totally dead |
| 19:14.42 | [TK]D-Fender | Navion: the PRECISE link please....... |
| 19:14.42 | masus | always the same |
| 19:14.51 | ManxPower | Navion: how about the ACTUAL URL |
| 19:15.19 | masus | ManxPower: do u have an idea how to fix this |
| 19:15.25 | Navion | http://www.pastebin.com |
| 19:15.26 | ManxPower | masus: no |
| 19:15.42 | masus | ok thanks |
| 19:15.49 | ManxPower | Navion: the actual URL would look something like http://pastebin.com/m6672bcb0 |
| 19:16.04 | tzafrir | Navion, like this: http://pastebin.com/m108e533 |
| 19:16.14 | Navion | http://pastebin.com/d69c64f9d |
| 19:16.19 | Navion | Sorry... |
| 19:16.31 | [TK]D-Fender | #include zapata_additional.conf |
| 19:16.33 | [TK]D-Fender | yay! |
| 19:17.02 | Navion | Nothing in zapata_additional.conf except 2 comment lines. |
| 19:17.15 | ManxPower | What the hell is this? zapata-auto.conf? |
| 19:17.19 | tzafrir | not to mention zapata-auto.conf, where the channel is actually likely to be |
| 19:18.00 | ManxPower | Navion: how many ports on your card, and are they all FXO ports? |
| 19:18.01 | tzafrir | ManxPower, trixbox. But this specific bit is something it doesn't setup through the web interface |
| 19:18.24 | ManxPower | tzafrir: well SOMETHING put in #include |
| 19:18.40 | tzafrir | Navion, /etc/asterisk/zapata-auto.conf |
| 19:18.59 | tzafrir | Anything there? |
| 19:19.08 | Navion | http://pastebin.com/m7cb141d0 |
| 19:19.32 | Navion | 4 ports all FXO only 2 have lines attached. |
| 19:20.19 | ManxPower | then use this zapata.conf: http://pastebin.com/m4acdaa56 |
| 19:20.35 | ManxPower | if you use Dial(Zap/g0 then change group=1 to group=0 |
| 19:21.18 | ManxPower | then stop and start asterisk. |
| 19:21.22 | ManxPower | a reload won't do it. |
| 19:23.37 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 19:23.44 | Navion | ManxPower: So that's adding a line "channel => 1-2" |
| 19:24.04 | ManxPower | Navion: and removing a bunch of lines. |
| 19:24.15 | ManxPower | and removing the #include |
| 19:25.02 | Navion | The includes include some Sangoma specific stuff too. |
| 19:25.15 | ManxPower | Navion: no it does not. |
| 19:25.25 | ManxPower | the files SAY they are sangoma specific, they are not. |
| 19:25.41 | ManxPower | You can argue all day if you want, but I won't. |
| 19:27.08 | [TK]D-Fender | either way the file was fine... all the parms in the right order... should have worked... |
| 19:28.28 | ManxPower | Navion: what telephone company are providing the lines? |
| 19:28.54 | ManxPower | and I assume you are able to dial OUT of the line? Also what SIP phone are you using?" |
| 19:29.49 | anonymouz666 | is there much speed difference between accessing databases through ODBC (func_odbc), than accessing directly through, let's say, app_mysql? |
| 19:30.12 | ManxPower | anonymouz666: that depends on how many queries per second you will do. |
| 19:30.27 | ManxPower | I gotta run and do some errands. best of luck everyone |
| 19:30.55 | Navion | Thanks |
| 19:30.58 | *** join/#asterisk techie (n=techie@adsl-76-214-9-1.dsl.lsan03.sbcglobal.net) |
| 19:31.45 | anonymouz666 | Queries per second avg: 9.516 |
| 19:31.47 | anonymouz666 | just that. |
| 19:31.52 | anonymouz666 | he he. |
| 19:31.56 | *** join/#asterisk adr3nalin3 (n=andrewf@ip68-13-89-102.om.om.cox.net) |
| 19:33.00 | anonymouz666 | I was just wondering how could be the impact adding another application layer to db access. |
| 19:33.18 | Navion | Well... That didn't work. |
| 19:39.27 | tzafrir | Navion, do you have an analog phone that shows callerid? |
| 19:41.00 | *** join/#asterisk shinao1 (n=shinao1@217.20.242.50) |
| 19:43.58 | *** join/#asterisk shinao1 (n=shinao1@217.20.242.50) |
| 19:56.42 | *** join/#asterisk hmmhesays (n=Neg@24-119-176-74.cpe.cableone.net) |
| 19:57.01 | [hC] | this client is going to make me go insane. I'm getting claims that on the new polycom 2.2.0 firmware that "after a while the phone stops ringing audibly and just blinks a line indicator" |
| 19:57.05 | [hC] | anyone heard of this? |
| 19:58.16 | *** join/#asterisk asdx (n=diego@adsl-128-20.click.com.py) |
| 19:58.29 | hmmhesays | i have not |
| 20:01.03 | rpm | [hC], we use all polycom here with fw 2.2.0 and i have never heard/seen that (300, 301, 501, 601, 650's and 4000's for conferencing) |
| 20:02.03 | [hC] | rpm: yeah, i havent either. very strange. this guy is also a bit of a nutcase. :) |
| 20:02.51 | rpm | you got those two eh? :) |
| 20:02.57 | rpm | blah, too* |
| 20:04.14 | [hC] | oh yeah. Im pretty sure its universal across the board |
| 20:05.58 | *** join/#asterisk curtn (n=curtis@cl-451.trn-01.it.sixxs.net) |
| 20:13.34 | curtn | i got a SIP problem with "Forbidden - wrong password on authentication for INVITE to" |
| 20:14.01 | curtn | what does that mean ? |
| 20:15.30 | rpm | curtn, probably means you supplied the wrong username/password for the proxy |
| 20:16.12 | rpm | if you get a 401 or 407 back, you have to ack it, that ack contains the wrong username:realm:password combination |
| 20:19.59 | curtn | rpm: it tries to connect at sip:MyPhone@MyInternetIP |
| 20:20.25 | curtn | rpm: why not sip:MyPhone@MyLocalAsterisk ? |
| 20:21.07 | curtn | it seems to be a NAT problem.. somewhere.. |
| 20:21.50 | curtn | rpm: incoming calls are ok, on the same line ("friend") |
| 20:24.01 | *** join/#asterisk scream_ (n=stas@87.110.230.26) |
| 20:26.12 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 20:47.32 | *** join/#asterisk keith4 (n=keith@c-76-24-188-68.hsd1.ma.comcast.net) |
| 20:49.12 | keith4 | what's that asterisk sip nat factoid? |
| 20:54.22 | *** join/#asterisk tzafrir (n=tzafrir@62.90.10.53) |
| 20:55.16 | De_Mon | tzafrir you around? |
| 20:55.22 | tzafrir | yes |
| 20:56.23 | *** join/#asterisk mihinomenest (i=7d7Y@66.255.220.17) |
| 20:56.27 | De_Mon | I noticed yesterday that bristuff addes extra applications n' stuff, but don't see any mention of them in the debian readme or anywhere else in the package is it there somewhere? |
| 20:56.55 | De_Mon | short of downloading the source and looking at debian/patches |
| 20:57.30 | De_Mon | (found out func_devstate isn't a stock feature too ;) |
| 20:57.30 | tzafrir | I think that looking at the patches is the simples thing. patch*es* |
| 20:57.56 | tzafrir | http://svn.debian.org/wsvn/pkg-voip/asterisk/trunk/debian/patches/bristuff/ |
| 20:58.08 | tzafrir | (that's a lousy svn browser, I know) |
| 20:58.37 | tzafrir | look also at the file "series" in the directory above it |
| 20:59.42 | De_Mon | err, I don't see any "files" in this browser |
| 20:59.44 | tzafrir | func_devstate is a backport from trunk, taken from svncomunity.digium.com |
| 21:00.05 | De_Mon | nevermind I see them |
| 21:00.08 | De_Mon | silly icons |
| 21:00.13 | tzafrir | click on "bristuff" |
| 21:00.22 | tzafrir | lousy SVN browser |
| 21:01.02 | tzafrir | the names and the comments in the series file should give you a good idea |
| 21:01.30 | *** join/#asterisk ThatKidKel (n=Kelvin@lacasa.thatkidkel.net) |
| 21:01.43 | ThatKidKel | Does anyone know if Cisco's 7920 (Cordless) has a SIP load? |
| 21:02.12 | De_Mon | cool, thanks |
| 21:04.55 | *** join/#asterisk slowshutt (n=d@196.211.34.2) |
| 21:05.02 | slowshutt | hi guys |
| 21:06.34 | slowshutt | i am kinda new to asterisk, i understand dial plan concepts, now want to focus on agi what programing language is the best to start with if you new to programming? |
| 21:08.00 | slowshutt | looks like friday is a bad time for this channel? |
| 21:08.11 | slowshutt | google here i come |
| 21:08.14 | *** part/#asterisk slowshutt (n=d@196.211.34.2) |
| 21:14.08 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 21:14.10 | *** join/#asterisk _ShrikE (n=ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 21:14.11 | *** join/#asterisk Greek-Boy (n=email@41.221.58.5) |
| 21:15.10 | *** join/#asterisk anonymouz666 (n=anonymou@200.218.193.6) |
| 21:23.06 | *** join/#asterisk Yourname`` (n=Miranda@unaffiliated/yourname/x-837320) |
| 21:23.33 | Yourname`` | Hi. Does anyone have a reference I could use that would tell me about how to monitor queue statistics via manager interface? |
| 21:25.21 | *** join/#asterisk Laureano (n=Laureano@host172038.metrored.net.ar) |
| 21:25.32 | *** part/#asterisk Laureano (n=Laureano@host172038.metrored.net.ar) |
| 21:26.13 | *** join/#asterisk RoyK (n=roy@ti211310a080-7605.bb.online.no) |
| 21:26.24 | De_Mon | ~book |
| 21:26.25 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf |
| 21:26.25 | De_Mon | maybe |
| 21:32.51 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@adsl-68-93-83-7.dsl.hstntx.swbell.net) |
| 21:34.46 | *** join/#asterisk mkl1525 (n=qwertz@p5098c328.dip0.t-ipconnect.de) |
| 21:38.01 | *** join/#asterisk Entr4nced (n=IMG001@adsl-70-238-245-18.dsl.bcvloh.sbcglobal.net) |
| 21:40.18 | mkl1525 | Hi, I'd like to use the automon feature for recording calls + the "Record" button on my snom360 in combination with MixMonitor is this |
| 21:40.22 | mkl1525 | possible? |
| 21:44.12 | De_Mon | huh what? |
| 21:44.34 | De_Mon | you want the record button on your snom360 to "do something" |
| 21:44.56 | De_Mon | that part is possible. now what exactly are you wanting it to do? |
| 21:49.26 | ManxPower | mkl1525: What does the record button cause the phone to send to the server? |
| 21:49.50 | *** join/#asterisk ||seb|| (n=seb@ppp-63-204-112-33.dialup.sndg02.pacbell.net) |
| 21:49.52 | mkl1525 | pressing "Record" button on my snom should record the call - that works already, but I get 2 files one for in and one for out, but I'd like to have a single file that includes in and out channel. best would be some way to mail this file after recording has finished to the mail address that is enabled in voicemail.conf - any ideas how to do this? |
| 21:50.20 | ManxPower | mkl1525: replace occurances of Monitor in your dialplan with MixMonitor |
| 21:50.31 | ||seb|| | help! trying to do voip over wifi instead of wire....doesn't work over wifi...any ideas? |
| 21:50.40 | *** join/#asterisk ghento (n=ghento@d216-232-170-111.bchsia.telus.net) |
| 21:51.22 | ManxPower | ||seb||: there is no difference between ethernet and wifi from a VoIP perspective |
| 21:51.49 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 21:51.57 | ||seb|| | ManxPower: i agree....i'm in a new location with free wifi....dunno why cals don't go through here |
| 21:52.21 | rob0 | So you don't control the router? |
| 21:52.44 | ||seb|| | rob0: what do you mean? |
| 21:53.04 | ||seb|| | rob0: oh...no the router at this RV park i do not control |
| 21:53.43 | ManxPower | ||seb||: You need to learn about RTP, SIP, and NAT to be able to troubleshoot the issue. |
| 21:54.06 | *** join/#asterisk WindBack (n=jorge@host227.190-31-204.telecom.net.ar) |
| 21:54.21 | ||seb|| | ManxPower: all i know is the exact same set up worked when i had wire at other RV park |
| 21:55.12 | *** join/#asterisk zerocod3r (n=Z3R3CoD3@b207d125.dorm.bilkent.edu.tr) |
| 21:55.17 | ||seb|| | ManxPower: would this make a difference?....when i was on wire there was one less NAT level.....on this free wifi there is a 2nd level of NAT |
| 21:56.10 | ManxPower | ||seb||: you don't know ANYTHING about your "setup" or about the previous "setup". |
| 21:56.34 | mkl1525 | ManxPowe, there's no monitor app in my dialplan using the features.conf of * with its automon feature |
| 21:56.34 | ManxPower | what brand of router at each location? What ports were blocked at each location? |
| 21:57.10 | ManxPower | mkl1525: then replace all occurances of Monitor in your features.conf with mixmonitor. This is not rocket science. |
| 21:57.28 | ManxPower | ||seb||: is it symetrical or cone NAT? |
| 21:57.39 | ManxPower | what packets ARE getting to your Asterisk server from the SIP client"? |
| 21:58.27 | ManxPower | Heck, do you even know what PROVIDER the RV park is using? |
| 21:58.32 | [TK]D-Fender | Nested NAT = FUBAR'd |
| 21:58.47 | ||seb|| | ManxPower: no |
| 21:58.47 | ManxPower | [TK]D-Fender: I have personally done SIP over nested NAT |
| 21:59.01 | [TK]D-Fender | ManxPower, yes, but you have to have control over forwarding on both levels. |
| 21:59.08 | ManxPower | ||seb||: So you basically know *nothing* except that it worked at one location and doesn't at another. |
| 21:59.15 | [TK]D-Fender | ManxPower, messy & risky at a bare minimum |
| 21:59.19 | ManxPower | [TK]D-Fender: I only had control over 1 of the 3 NATs |
| 21:59.28 | [TK]D-Fender | ManxPower, interesting... |
| 21:59.53 | mkl1525 | ManxPower, there aren't any monitor commands in features.conf, "grep -ri monitor /etc/asterisk" just show something in agents.conf, queues.conf and rtp.conf - all things I don't use |
| 22:00.07 | ManxPower | mkl1525: then MAKE one. |
| 22:00.15 | ManxPower | all "automon" does is run monitor |
| 22:00.31 | ManxPower | mkl1525: have you read up on the wiki about monitor and mixmonitor and automon? |
| 22:02.40 | ManxPower | although really "show application monitor" and "show application mixmonitor" is a better place to start. |
| 22:03.55 | ManxPower | *sniff* *sniff* I smell a GUI |
| 22:04.17 | rob0 | I thought it was flatulence. |
| 22:04.35 | ManxPower | rob0: it is a similar smell. |
| 22:07.19 | mkl1525 | ManxPower thanks for the hints, in features.conf automon is set to *1 so I'd have to set the monitor options in extensions.conf for extension *1 - right? |
| 22:07.42 | ManxPower | mkl1525: that sounds correct. |
| 22:07.57 | ManxPower | turn off automon for *1, use *1 to call mixmonitor or somesuch |
| 22:08.10 | mkl1525 | thanks again :) |
| 22:16.05 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 22:22.49 | *** join/#asterisk Navion (n=billp@75-105-41-123.cust.wildblue.net) |
| 22:23.23 | Navion | Anyone know of a good way to be able to direct select a single line out of a group? |
| 22:24.20 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 22:29.05 | *** join/#asterisk dlynes (n=chatzill@s209-121-50-177.bc.hsia.telus.net) |
| 22:29.25 | *** join/#asterisk keith4 (n=keith@c-76-24-188-68.hsd1.ma.comcast.net) |
| 22:29.33 | *** part/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 22:31.20 | tzafrir | Navion, just use the specific channel number? |
| 22:33.09 | Navion | From a phone? |
| 22:33.50 | Navion | Or set up a bunch of extension numbers that point to a channel? |
| 22:34.58 | *** join/#asterisk bkw_ (n=brian@adsl-64-149-41-151.dsl.tul2ok.sbcglobal.net) |
| 22:35.31 | tzafrir | If this is an analog line - you really can't tell "easily". So you have two options: |
| 22:35.44 | tzafrir | 1. send both to different contexts |
| 22:36.06 | tzafrir | 2. extract the zaptel channel number from CHANNEL |
| 22:37.13 | Navion | It's not a problem now but I'm looking ahead to trying to debug the system when one line out of group gets noisy (or something like that). |
| 22:38.20 | keith4 | where is a good asterisk + sip + NAT howto? |
| 22:38.40 | *** join/#asterisk scream_ (n=stas@87.110.230.26) |
| 22:39.40 | anonymouz666 | ~nat |
| 22:39.41 | jbot | well, nat is Network Address Translation Usable in Asterisk sip.conf file with externip, localnet, and localmask setup properly. See docs. |
| 22:39.53 | anonymouz666 | ~asterisk+nat |
| 22:40.00 | anonymouz666 | wrong one ;) |
| 22:40.08 | anonymouz666 | ~sip+nat |
| 22:40.08 | jbot | methinks sip+nat is http://www.asteriskguru.com/natut.php <--- Zoa's tutorial |
| 22:40.25 | keith4 | yah, that's the one! |
| 22:40.27 | keith4 | thanks |
| 22:46.56 | *** join/#asterisk craigk (n=ckowald@58.174.122.198) |
| 22:48.08 | Navion | ~voicemail |
| 22:48.37 | Navion | ~vm |
| 22:48.37 | jbot | Emacs-based mail reader. URL: http://www.wonderworks.com/vm/ |
| 22:50.41 | Navion | Is it possible to set up extensions that have no hardware? What I'm thinking here is guest voicemail boxes. |
| 22:51.24 | rob0 | why not? |
| 22:51.35 | *** join/#asterisk RoyK (n=roy@ip-247-8-149-91.dialup.ice.no) |
| 22:57.11 | *** part/#asterisk bkw_ (n=brian@adsl-64-149-41-151.dsl.tul2ok.sbcglobal.net) |
| 22:57.13 | [TK]D-Fender | Navion, And extension is a NUMBER in your dialplan. Nothing more. |
| 22:58.24 | De_Mon | heh, i was just pondering the number of extensions I have that arn't even that |
| 22:58.29 | Navion | So, just tell asterisk it's a SIP phone but never connect it? |
| 22:58.50 | De_Mon | a voicemailbox doesn't require a phone just create the mailbox |
| 22:59.41 | rob0 | No you do NOT tell asterisk it's a SIP phone. |
| 23:00.44 | Navion | But if asterisk doesn't know about the extension number, there's no way to get a call into the right voicemail box. |
| 23:01.21 | anonymouz666 | [TK]D-Fender: freeswitch can use ${var_names} isn't that amazing? |
| 23:03.02 | *** join/#asterisk WilliamK (n=noc@static-71-170-144-3.dllstx.fios.verizon.net) |
| 23:03.09 | Greek-Boy | anonymouz666: I'm guessing they used asterisk code to implement that? |
| 23:03.56 | [TK]D-Fender | Navion, You are failing to follow. An extension is a number you DIAL It doesn't matter what it DOES. I can have 1 SIP phone and 1 MILLION different extensions in my DIALPLAN (extensions.conf) that each do something different. |
| 23:03.59 | anonymouz666 | everything related to freeswitch dev should be asked to freeswitch devs - not me. |
| 23:04.22 | [TK]D-Fender | Navion, NEVER think of a DEVICE (like a SIP phone or anything else) as an EXTENSION. |
| 23:12.39 | *** join/#asterisk hacim (n=micah@debian/developer/micah) |
| 23:13.10 | hacim | i notice that http://www.voip-info.org/wiki/view/Asterisk+encryption is dated 2005, does anyone know if encrypted asterisk, SRTP etc. has gained any more traction since then? |
| 23:14.22 | ManxPower | hacim: I don't think so. |
| 23:15.07 | [TK]D-Fender | hacim, there is a patch out now, and it will be in 1.6 |
| 23:16.34 | hacim | [TK]D-Fender: bug 5413? |
| 23:16.47 | [TK]D-Fender | hacim, I don't know the #'s. Just go look |
| 23:16.53 | *** join/#asterisk RoyK (n=roy@91.149.31.118) |
| 23:16.58 | *** join/#asterisk EnigmaCurry (n=user@67.166.72.245) |
| 23:23.05 | *** join/#asterisk ajohnstone (n=ajohnsto@cpc3-walt3-0-0-cust880.popl.cable.ntl.com) |
| 23:27.08 | *** join/#asterisk CpuID (n=cpuid@gentoo/contributor/cpuid) |
| 23:27.19 | *** part/#asterisk CpuID (n=cpuid@gentoo/contributor/cpuid) |
| 23:28.27 | *** part/#asterisk ThatKidKel (n=Kelvin@lacasa.thatkidkel.net) |
| 23:35.05 | bantu | hi folks. i have successfully compiled and installed cdr_addon_mysql.so. CDR status doesn't show MYSQL as a registered backend. What can i do? |
| 23:36.03 | codefreeze | bantu: find out why--- look thru the console messages when asterisk started... |
| 23:39.37 | km- | Navion: what they're trying to say (re: extensions) is that extensions are a logical item in asterisk that can connect to any of a number of different items. by creating an extension in extensions.conf, eg, exten => 25,1,Voicemail(u25), you can make it so "extension 25" dials voicemail box 25. Never even have to involve sip into it. |
| 23:43.13 | *** join/#asterisk techie (n=techie@adsl-76-214-9-1.dsl.lsan03.sbcglobal.net) |
| 23:44.36 | [TK]D-Fender | Navion, exten => 666,1,Hangup ; I rudely hangup on callers! |
| 23:46.42 | Greek-Boy | [TK]D-Fender you mean I will be able to dial that extension during any call without dtmf rules? |
| 23:47.49 | [TK]D-Fender | Greek-Boy, Did I say HOW you get to dial it? NO. |
| 23:48.50 | km- | [TK]D-Fender: how long did it take you to cultivate the ass persona? :P |
| 23:49.14 | Greek-Boy | lol ok |
| 23:49.15 | Greek-Boy | :) |
| 23:49.19 | [TK]D-Fender | km-, the tilling is the hard part :) |
| 23:49.36 | km- | haha, I'll keep working at it. |
| 23:49.42 | bantu | codefreeze: Seams to load just fine :-( << cdr_addon_mysql.so => (MySQL CDR Backend) >> |
| 23:49.45 | km- | I'm still working on brian's "NEXT!!!!$$$!!$!$!!" |
| 23:50.28 | Greek-Boy | I think one day chan_ss7 will have the potential for production use. one day... Lots of development needed. |
| 23:50.29 | [TK]D-Fender | <beavis> hheheheehhe he said "backend" following Brian </beavis> |
| 23:50.41 | km- | hahahahahaha |
| 23:50.48 | km- | I'm sure brian would appreciate that |
| 23:51.12 | km- | still waiting for him to get tired of freeswitch and come back to meee |
| 23:57.48 | *** join/#asterisk felix_da_catz (n=felix@c-76-31-59-80.hsd1.tx.comcast.net) |
