| 00:00.06 | Moonwick | [hC]: sec, let me play with it on my end |
| 00:00.21 | jhowardPA | obsidian-studios: There's a pair of T1's coming in, but I don't think they're full 24 channels each, and they go into AT&T boxes which provide analog pots to a 66 block. |
| 00:00.22 | obsidian-studios | Sift: yes but with exception of video with audio, 10 is more than plenty for audio I believe a perfect phone lines only uses 64k max |
| 00:00.30 | [hC] | Moonwick: cool, thanks. |
| 00:00.55 | Sift | 64k is the 711 codec right? |
| 00:01.02 | obsidian-studios | ok I would then either get fxo cards like the tdm400, recently I purchased some el cheapo modems that act as fxo ports |
| 00:01.17 | obsidian-studios | <PROTECTED> |
| 00:01.51 | obsidian-studios | Sift: not sure on codecs, I just know that like when you split lines or when pots lines use parts of your T-1 they do it in 64k chunks per line |
| 00:02.03 | jhowardPA | obsidian-studios: I've got an Adtran ATLAS 550, but I don't know what types of connections I would need to use... I'd rather not use fxo cards, as that doesn't seem to leave much capability for network-level failover. |
| 00:02.41 | obsidian-studios | jhowardPA: I am not experienced with it but I believe there is other ways to get the lines to * via your T-1 and avoid pots entirely |
| 00:03.14 | obsidian-studios | I would go that route if possible, I know digium and others sell T-1 cards? Just not sure what you need on the carrier end or * end to get things going |
| 00:03.25 | jhowardPA | obsidian-studios: That's what I'm hoping ;) I need to do it in a way where I can have an Asterisk server die, and have the other one take over - without buying new T1 lines. |
| 00:03.29 | obsidian-studios | jhowardPA: you might want to check back during the day when the gurus are around and active |
| 00:03.29 | shido6 | um |
| 00:03.53 | shido6 | Adtran atlas 550 |
| 00:03.53 | shido6 | ok |
| 00:03.57 | jhowardPA | Cool, thanks for the info... I'm in a pretty foreign territory at the moment. :) |
| 00:03.58 | *** join/#asterisk tugalone (~tugalone@pcp0010303951pcs.avenel01.nj.comcast.net) |
| 00:04.02 | obsidian-studios | jhowardPA: hmm, a bit harder you will need a device to terminate the T-1 and then use an available * server |
| 00:04.13 | obsidian-studios | FYI I would believe your T-1 to fail before * |
| 00:04.25 | obsidian-studios | so not really a need for redundant * servers? |
| 00:04.32 | *** join/#asterisk outtolunc (~me@adsl-69-110-56-125.dsl.pltn13.pacbell.net) |
| 00:04.54 | jhowardPA | If I'm running it on PC hardware, it'll only last 3-5 years, max. That's not long enough. ;) |
| 00:04.59 | obsidian-studios | or failover that's fairly advanced, I believe you can daisy chain * servers, and use multiple * servers, but both are way over my head and skills |
| 00:05.03 | shido6 | err |
| 00:05.22 | shido6 | yeah you can connect mutliple * boxes |
| 00:05.23 | shido6 | together |
| 00:05.27 | shido6 | but what are you trying to do |
| 00:05.30 | obsidian-studios | jhowardPA: with two or more, it will not increase that, each * server will that have life span if that's what you think it to be |
| 00:05.32 | shido6 | I missed the bulk |
| 00:05.39 | obsidian-studios | shido6: failover |
| 00:06.00 | obsidian-studios | shido6: he has T-1 with split pots lines now, wants T-1 termintated at like a router, then using more than one * server |
| 00:06.12 | obsidian-studios | so if one * is gone, use another |
| 00:06.18 | jhowardPA | obsidian-studios: But they won't die at the same time, and when one does, there should be a window of an hour or so to re-image a new one and put it back in place. |
| 00:06.40 | obsidian-studios | jhowardPA: do you really expect the machines to fail like that? Very unlike linux :) |
| 00:07.12 | obsidian-studios | I mean I just took down a cobalt cube 2 that was in use for a client for like 4-5 years without issues, and only took it down due to it being slow and hard to update |
| 00:07.28 | jhowardPA | obsidian-studios: Hardware dies, it's a fact. Disks will eventually not spin. |
| 00:07.42 | obsidian-studios | especially with only a T-1 I doubt you could easily kill a machine, unless it's crappy hardware |
| 00:07.59 | obsidian-studios | FYI, I have done * deployments with Microtel PC's from Walmart :) |
| 00:07.59 | *** join/#asterisk ANonymousUser (~ANonymous@f7bf8b07b6ac798b.node.tor) |
| 00:08.26 | obsidian-studios | all true, but why not have a plan to just replace a * server ever couple of years |
| 00:08.51 | shido6 | did batman ever pull a pizza out of his utility belt? |
| 00:09.03 | syle | no just weed |
| 00:09.03 | obsidian-studios | as opposed to spending extra $ etc on fail over to almost never be needed. Unless you buy crap even a disk with allot of I/O should go at least 3-4 years or longer |
| 00:09.05 | jhowardPA | obsidian-studios: I'm talking about running the phone systems for a 100+ user office, which is strictly dependent on the phone systems to operate. Also, they're cheap, and won't buy anything with an SLA, so I'm tasked with coming up wiuth something that won't break. |
| 00:09.06 | Sedorox | why? if it runs... why bother with it... |
| 00:09.15 | shido6 | syle, dcc me some rizzla's |
| 00:09.16 | obsidian-studios | syle: weed were I want some :) |
| 00:09.21 | syle | will do |
| 00:09.33 | obsidian-studios | syle: so the DEA or FEDS can come take me home ;) |
| 00:10.00 | Sedorox | just get a box.. and get a 11 or two compact flashes and use them as the HDD's |
| 00:10.00 | tainted- | jhowardPA f them |
| 00:10.10 | obsidian-studios | jhowardPA: but your only talking over a T-1, even if you get crazy your standard 2gh machine with decent hd |
| 00:10.12 | syle | --/ctcp dcc send shido6 2fatstogies |
| 00:10.15 | obsidian-studios | should have no problem |
| 00:10.21 | Moonwick | [hC]: try googling for 'native bridge' |
| 00:10.31 | [hC] | Moonwick: Excellent. Thanks for the search term. |
| 00:10.37 | Moonwick | - Attempting native bridge of IAX2/voicepulse-in-01@66.234.228.170:4569/3 and IAX2/NuFone/10 |
| 00:10.42 | Moonwick | that's what'll show up when it tries to do it |
| 00:10.44 | tainted- | 100+ & they're cheap? |
| 00:10.52 | ANonymousUser | Hypothetical question: Let's say I had a patch to the default sip_notify.conf with some information gleaned from the documentation Sipura attempts to restrict to service providers only. Would folks-in-general have an issue with such a patch, anonymously provided, given that it contains no copyrighted content itself (but is based on the content of a distribution-restricted document)? |
| 00:10.52 | tainted- | gaylord fockers |
| 00:10.56 | [hC] | Great. Okay I'll play some more. Thanks for looking into it :) |
| 00:10.58 | obsidian-studios | jhowardPA: seriously to justify the need due to load, you would need much more than a T-1 and more than 100+ users |
| 00:11.05 | jhowardPA | obsidian-studios: a pair of T-1's, and the bandwidth isn't an issue (I dn't think), it's the (lack of) reliability inherent in PC hardware. |
| 00:11.21 | obsidian-studios | <PROTECTED> |
| 00:11.23 | Moonwick | no prob |
| 00:11.30 | tainted- | ANonymousUser url please!! lol |
| 00:11.33 | Moonwick | I'd been meaning to make my system do it anyway |
| 00:11.39 | [hC] | You guys probably wont know this, but... (hehe!) there was a tool i used in the past for windows of all things that was a real time SNMP monitor that would draw mrtg-like graphs (polling every 3 seconds or so) - Anyone know of a utility like this? |
| 00:11.41 | obsidian-studios | I have hd's that literally are bad, a few I got that way and used for years as dns servers under linux |
| 00:11.50 | syle | how much does a T1 line cost these days |
| 00:11.51 | Moonwick | cell phone + lag between my box and two different SIP providers sucks |
| 00:12.04 | obsidian-studios | <PROTECTED> |
| 00:12.35 | ANonymousUser | tainted, I'd need a few minutes to test it, and a few more to put it up somewhere suitably anonymous. |
| 00:12.36 | *** join/#asterisk Dovid (~hirisk@pool-138-89-147-151.mad.east.verizon.net) |
| 00:12.43 | syle | screw cable modems and dsl lets just get T1 lines to our houses |
| 00:12.49 | obsidian-studios | <PROTECTED> |
| 00:12.54 | *** join/#asterisk nDuff (~cduffy@64.128.31.220) |
| 00:13.00 | obsidian-studios | syle: I am tomorrow :) |
| 00:13.06 | syle | no shit |
| 00:13.09 | shido6 | brb |
| 00:13.10 | shido6 | FOOD |
| 00:13.11 | obsidian-studios | can't wait, me first T-1 :) |
| 00:13.16 | tainted- | ANonymousUser which providers does it emulate? |
| 00:13.22 | jhowardPA | obsidian-studios: Dude, I know Linux. I know Linux inside and out. I do pen-testing as a consultant, and I run farms. This isn't Linux, this is a single-point-of-failure that would cost tons of money, and is related to hardware. |
| 00:13.23 | obsidian-studios | $460 a month you are welcome to pay :) |
| 00:13.36 | syle | holy crap |
| 00:13.44 | tainted- | jhowardPA then do failover u linux guru u |
| 00:13.46 | obsidian-studios | jhowardPA: ok but I have linux used as POS in bars without a problem |
| 00:13.52 | ANonymousUser | tainted-, eh? Nothing like that... just some SIP NOTIFY events that aren't publicly documented but which Sipura phones handle. |
| 00:13.58 | obsidian-studios | jhowardPA: flying liquor drunk users etc |
| 00:14.10 | ANonymousUser | tainted-, particularly usefully, the "dump your config" one. |
| 00:14.13 | syle | thats probably cheaper than hosting my damn server at a colocation for 1200 a year lol |
| 00:14.18 | obsidian-studios | jhowardPA: machines being kicked, aside from the occasional touch screen eating it, they work great |
| 00:14.22 | syle | oww a month nm |
| 00:14.34 | obsidian-studios | syle: hech yeah I stopped coloing years ago |
| 00:14.34 | jhowardPA | obsidian-studios: What's the MTBF on your hard drives? |
| 00:14.40 | tainted- | ANonymousUser ohh.. for unlocking ATAs? |
| 00:14.46 | obsidian-studios | been with Covad on SDSL for years |
| 00:14.46 | jhowardPA | obsidian-studios: I need more than that ;) |
| 00:14.54 | obsidian-studios | jhowardPA: MTBF? |
| 00:15.00 | Faithful | Hey guys, do I really have to install X in order to use bluetooth with * ? |
| 00:15.03 | jhowardPA | Mean time between failures. |
| 00:15.13 | jhowardPA | It's a vendor rating. |
| 00:15.18 | obsidian-studios | I have not had a failure aside from my laptop |
| 00:15.24 | ANonymousUser | tainted-, for provisioning them. I frankly don't see why Sipura restricts this stuff. |
| 00:15.31 | obsidian-studios | I think the hard drives in my old Cobalt XTR are way beyond that |
| 00:15.46 | obsidian-studios | I can assure you the ones in like that clients old Cube 2 is way beyond it |
| 00:15.58 | obsidian-studios | in like 10+ years the only hds I have lost have had windows on it |
| 00:16.12 | ANonymousUser | tainted-, anyhow, the format of the dump is the same format that's used for uploading XML-based configs |
| 00:16.17 | obsidian-studios | but it's all about load, I/O etc, partitions, file systems, etc |
| 00:16.34 | ANonymousUser | tainted-, consequently, having the ability to do the dump lets one infer much of the interesting stuff from the Provisioning Guide. |
| 00:17.02 | jhowardPA | tainted-: Not trying to sound megalomaniacal, but I'm just pointing out that Linux isn't the issue - it's my lack of telecom knowledge. If I don't know where the T-1 plugs into something, and what options I have on the network side, I can't do failover. |
| 00:17.45 | obsidian-studios | jhowardPA: also FYI one of my * deployments in the same club with the POS, well it's located in FL, Which means lighting and power spikes surges all the time. Many the walmart Microtel has eaten directly because client has not purchase ups, So far no problem, even though lighting during a hurricane took out old pbx |
| 00:18.06 | obsidian-studios | jhowardPA: you can totally do failover |
| 00:18.08 | ANonymousUser | (not all of it; it also documents ie. how they do certificate signing for their SSL-based provisioning, but this is enough for folks to make Sipura config tools that are a bit smarter than "post to the web interface"). |
| 00:18.17 | obsidian-studios | jhowardPA: you are going to spend allot of $ and time, so the need better be there |
| 00:18.46 | ANonymousUser | o'course, it's possible that Cisco'll decide to stop restricting the docs and make this all public knowledge anyhow. I'd hope they would. |
| 00:18.46 | obsidian-studios | jhowardPA: I mean if it fails you will be happy, when it does not you might kick yourself :) |
| 00:19.42 | obsidian-studios | jhowardPA: do not get me wrong I totally understand, but if you spend like 2-3 times the amount of time and $, it better be something used and needed |
| 00:20.01 | obsidian-studios | I mean I would go ahead and look into load balancing as well, not just failover so you can get a ROI |
| 00:21.11 | syle | what do you use? |
| 00:21.13 | *** join/#asterisk vpp (~noone@host-83-146-50-131.bulldogdsl.com) |
| 00:21.20 | syle | i use to use those cisco arrowpoints |
| 00:21.21 | vpp | hi guys |
| 00:21.24 | jhowardPA | obsidian-studios: It's all about guaranteed uptime. It's not a question of whether or not the work will pay off due to use, it's a matter of whether or not I can guarantee that it'll not fail without warning and possibility of resolution. |
| 00:21.27 | vpp | anyone use asterisk with H323 ? |
| 00:21.52 | MeTaBSD | hi |
| 00:22.00 | obsidian-studios | jhowardPA: it won't fail if you do things right |
| 00:22.00 | MeTaBSD | i need help |
| 00:22.23 | vpp | i need help too.. but the doctor game me some pills so i'm ok today |
| 00:22.24 | vpp | lol |
| 00:22.31 | Moonwick | I need an asterisk failover solution that'll withstand an alien fleet destroying the milky way with a black hole generator. |
| 00:22.33 | obsidian-studios | jhowardPA: there are people out there using * with 100k users providing VOIP service and etc check the list tomorrow during the day and they will be in here |
| 00:22.37 | jhowardPA | obsidian-studios: thanks, but I'm afraid you're overly optimistic about hardware. |
| 00:23.01 | obsidian-studios | jhowardPA: yes, because most times it's how hardware is used that causes failures, ie software :) |
| 00:23.10 | vpp | which H323 should i use? |
| 00:23.15 | shido6 | none of them |
| 00:23.18 | shido6 | dont use h323 |
| 00:23.20 | vpp | oh323? |
| 00:23.22 | obsidian-studios | jhowardPA: you know how many PC clients toss, give to me, or I recycle for them due to software issues |
| 00:23.27 | nDuff | obsidian-studios, yes, but the *other* times, while rare, are still enough to ruin one's day. |
| 00:23.34 | vpp | shido6: but i need h323 |
| 00:23.47 | obsidian-studios | jhowardPA: everyone loves to blame hardware, but so much more goes into the hardware than software, not to mention the live span of this stuff just get's longer day after day |
| 00:23.48 | vpp | sip is not an option |
| 00:24.00 | nDuff | obsidian-studios, especially if the suits are in the middle of a conference call w/ a bunch of investors. |
| 00:24.12 | obsidian-studios | nDuff: I agree, I was just pointing out the $ and time, keeping ROI in mind |
| 00:25.03 | MeTaBSD | i search for exten exten => s,2,Playback(message) ... exten => s,3,cliententeracode and this code is in $var and exten => s,4,DBGet(codecheck=blacklist/${var}) |
| 00:25.03 | Moc | hi |
| 00:25.03 | obsidian-studios | because those same suits may one day say why did this cost so much and never get's used :) |
| 00:25.04 | obsidian-studios | nDuff: double edged sword :) |
| 00:25.18 | nDuff | obsidian-studios, if your suits don't appreciate reliability... well, let's say that while my suits have their failings, they don't include that one. |
| 00:25.33 | obsidian-studios | however as NASA has found many times, even a backup to a backup can fail :) |
| 00:26.07 | jhowardPA | obsidian-studios: I have approval for the money, I need to provide guaranteed uptime. I'm not concerned with the cost, so long as it's not an overt waste based upon my own principles of parsimony. I'll be back, I need to look some stuff up. |
| 00:26.13 | jhowardPA | Thanks again for the info :) |
| 00:26.15 | obsidian-studios | nDuff: they do but everyone has a budget, and I see people daily buying more than they need etc |
| 00:26.19 | Moonwick | nduff is pulling your chain. he doesn't actually have a single suit. |
| 00:26.20 | *** part/#asterisk kielstirling (~kiel@knss.net) |
| 00:26.25 | Moonwick | he's a jeans 'n t-shirt kind of man. |
| 00:26.40 | obsidian-studios | <PROTECTED> |
| 00:26.51 | obsidian-studios | <PROTECTED> |
| 00:26.54 | *** join/#asterisk puowvip (ircuser@thegrey.diamond.org) |
| 00:26.59 | nDuff | Moonwick, heh. Actually, I've switched to shorts -- the new building gets pretty hot after the AC's turned off. |
| 00:27.01 | obsidian-studios | I meant $ call Cisco |
| 00:27.12 | jhowardPA | obsidian-studios: That's too expensive to justify. My tastes won't allow it. |
| 00:27.18 | obsidian-studios | well then :) |
| 00:27.36 | obsidian-studios | <PROTECTED> |
| 00:28.09 | obsidian-studios | <PROTECTED> |
| 00:28.36 | obsidian-studios | <PROTECTED> |
| 00:28.57 | nDuff | jhowardPA, wrt outside * consulting, I've found the Bristol Group (www.bg.com) to be pretty helpful. Their tech-types don't know everything, but they know the common cases and are willing to do research (on their dime, in my experience so far) to learn what they don't. |
| 00:29.32 | syle | [19:22] <obsidian-studios> jhowardPA: there are people out there using * with 100k users providing VOIP service and etc check the list tomorrow during the day and they will be in here |
| 00:29.35 | syle | hmmmm |
| 00:29.37 | syle | question on that |
| 00:29.47 | obsidian-studios | syle: yes I mentioned that a few times |
| 00:29.53 | syle | if your databasing extensions.conf with mysql.... |
| 00:30.07 | syle | how many queries a sec , ie how many voip users can you have |
| 00:30.12 | syle | per box |
| 00:30.18 | obsidian-studios | I got to make some stuff for * and Firebird, mysql is for weenies :) |
| 00:30.48 | syle | your entitled to your opinion i'm sure yahoo and NASA are weenies to but just wondering |
| 00:32.19 | obsidian-studios | syle: not SQL-92 compliant or SQL-99, no store procs, triggers, udf etc stuff real dbs have. However my biggest grip is licensing, Nasa and yahoo have $. For the apps I develop if I based them on mysql I would have to buy a license and so would clients :) |
| 00:32.47 | *** join/#asterisk menger (~menger@static-88.243.240.220.dsl.comindico.com.au) |
| 00:33.17 | Nugget | still no excuse to be using it in the condition it's in today, though. |
| 00:33.25 | Nugget | mysql is embarassingly poor. |
| 00:33.40 | tainted- | Nugget how so |
| 00:33.43 | obsidian-studios | nDuff: Firebird/Interbase could be mysql's grandfather, been around much longer and used in some Army tanks |
| 00:33.58 | tainted- | nDuff if u give me the stored procedures argument, i will smack u |
| 00:33.59 | syle | umm get on topic though, asterisk....high read tables, simple fields, for * mysql is suitable |
| 00:34.14 | Nugget | http://sql-info.de/mysql/gotchas.html embodies my perspective fairly well |
| 00:34.28 | puowvip | power requirements of a fully loaded TDM400P? |
| 00:34.31 | Nugget | mysql does many things the totally wrong way for a database to behave |
| 00:34.41 | Nugget | and it lacks features I'm unwilling to live without |
| 00:34.53 | syle | i guess so |
| 00:34.57 | obsidian-studios | syle: many use * in commerical settings |
| 00:35.00 | syle | if your use to coding a certain way i can see that |
| 00:35.04 | obsidian-studios | syle: mysql is not free for many of those uses |
| 00:35.13 | Nugget | over the past 7 or 8 years I've been working with mysql the developers have shown a pretty abject lack of clue, too. |
| 00:35.24 | obsidian-studios | syle: so it directly relates to *, postgresql would be a better fit license wise |
| 00:35.27 | syle | how is mysql not free for commercial use |
| 00:35.32 | syle | its opensource |
| 00:35.34 | obsidian-studios | syle: go look |
| 00:35.37 | obsidian-studios | syle: hell no |
| 00:35.57 | obsidian-studios | syle: sure the code is out there, but for commercial and just about any non web based app you pay |
| 00:36.01 | puowvip | Okay, I'll ask later. |
| 00:36.04 | nDuff | syle, it's under a GPL, including the libraries for using it; consequently, if you want to link to those libraries from a non-GPL app, you need to buy a commercial license. |
| 00:36.14 | *** join/#asterisk exonic (~exonic@c-24-11-2-241.hsd1.mi.comcast.net) |
| 00:36.15 | Nugget | puowvip: best to just ask digium |
| 00:36.30 | exonic | ANyone know how to do password recovery on a sipura 2000, I feel like an id10t for losing it. |
| 00:36.38 | obsidian-studios | syle: mysql is still a good db though, but it's a bit over popular for it's actual weight in gold :) |
| 00:37.01 | nDuff | exonic, use the voice interface to reset it to factory defaults (***RESET), IIRC. |
| 00:37.03 | syle | i;m looking |
| 00:37.07 | syle | .tar.gz source downloads |
| 00:37.20 | syle | commercial support if you your to stupid to run a db on your own |
| 00:37.29 | obsidian-studios | https://shop.mysql.com/ |
| 00:37.44 | exonic | nDuff, lemme try, Thanks |
| 00:37.49 | Nugget | it's not a matter of support, syle. listen to what they are telling you. |
| 00:38.00 | obsidian-studios | it's not just support you are paying for a license |
| 00:38.07 | *** part/#asterisk Uther_P (~uther_p@66.180.120.83) |
| 00:38.10 | nDuff | syle, http://www.mysql.com/company/legal/licensing/commercial-license.html |
| 00:38.11 | syle | yeah thats just the commercial tech support package dude |
| 00:38.29 | obsidian-studios | <PROTECTED> |
| 00:38.47 | nDuff | syle, quote: "The Commercial License is an agreement with MySQL AB for organizations that do not want to release their application source code." |
| 00:38.50 | exonic | nDuff, can i PM you for a bit more help |
| 00:38.52 | obsidian-studios | I would have to buy a license and so would clients |
| 00:39.00 | Nugget | personally, I'm thrilled that mysql is so restrictively licensed. I believe that it prevents many projects from making the mistake of choosing mysql. |
| 00:39.07 | syle | read that page duff it states if you want to package up mysql with code you develop then of course you need a license hehe |
| 00:39.11 | obsidian-studios | <PROTECTED> |
| 00:39.19 | opus_ | haha nugget |
| 00:39.24 | nDuff | exonic, sure -- though I'm not positive how much help I can be. |
| 00:39.43 | syle | nugget you can get around that |
| 00:39.44 | obsidian-studios | postgresql and firebird you can include and resell to your hearts content :) |
| 00:39.49 | Nugget | mysql is the windows me of databases, and the arguments in its favor are remarkable like those of windows me users. |
| 00:39.59 | syle | prerequsites: install mysql , then install my software hehe |
| 00:40.05 | obsidian-studios | Nugget: man that is the best comparison yet |
| 00:40.10 | Nugget | syle: you still cannot link to the mysql libraries. |
| 00:40.12 | obsidian-studios | Nugget: going to save that one |
| 00:40.17 | denon | anyone know anything about the audiovox XV6600? (cdma pda phone) |
| 00:40.17 | Nugget | so for most people that would not be a solution |
| 00:40.21 | Nugget | s/people/projects/ |
| 00:40.23 | opus_ | almost all database servers have a way you can include the database server for free in your product... ms sql server, sybase i know for sure |
| 00:40.31 | obsidian-studios | syle: you can't package as part of a product though the others you can |
| 00:40.35 | syle | ummm i don;t beleive the libraries are licensed |
| 00:40.35 | nDuff | syle, quote: "If you develop and distribute a commercial application and as part of utilizing your application, the end-user must download a copy of MySQL; for each derivative work, you (or, in some cases, your end-user) need a commercial license for the MySQL server and/or MySQL client libraries." |
| 00:40.47 | nDuff | syle, you're not redistributing it yourself in that case. |
| 00:40.55 | syle | how can you license something as stupid as a connect script library |
| 00:41.16 | Nugget | what is a "connect script library?" |
| 00:41.20 | obsidian-studios | syle: you can license anything, most software patents are on concepts not actual code as well if that is not messed up |
| 00:41.20 | exonic | nDuff, ok, I won't i'll do some searchin, I thought I could perhaps dial ***RESET from my phone, but I just get busy signal |
| 00:41.44 | syle | your /usr/lib/mysql.so |
| 00:42.04 | Nugget | and you believe that is a script? |
| 00:42.07 | *** join/#asterisk tck_mi (~tck@adsl-68-74-22-177.dsl.sfldmi.ameritech.net) |
| 00:42.28 | syle | well in reality you don;t even need it even if they did commercialize it you can code your socket routines to use tcp or /dev/sock socket routines |
| 00:42.32 | obsidian-studios | syle: but aside form licenses, there are real world limitations, no argument supports lack of stored procs, triggers, or udf's |
| 00:42.59 | obsidian-studios | syle: the first time you build a db with them, building db's without get's really hard :) |
| 00:43.01 | nDuff | exonic, it might be 4 *'s rather than 3 -- pretty much, you type the *s and then the phone gives you a voice prompt; after that, you can punch "RESET". |
| 00:43.17 | syle | nugget no, its a dynamic library you can link into your c code, but that is way off topic |
| 00:43.29 | exonic | nDuff, I got it , Thanks a ton |
| 00:43.29 | nDuff | exonic, there's some waiting before the *s and the prompt. |
| 00:43.34 | nDuff | exonic, cool. |
| 00:43.38 | Nugget | I was just confused by you having called it a "connect script library", a term which I have still not been able to parse. |
| 00:43.49 | exonic | **** (4 of 'em did the trick) |
| 00:43.49 | syle | its ok |
| 00:44.09 | Nugget | well, it would help if you would use the same terminology as the rest of us. |
| 00:44.18 | syle | as you you mean |
| 00:44.23 | syle | don;t confuse the 2 :) |
| 00:44.51 | Nugget | I'll leave it up to the rest of the channel to determine which of us is more confused. |
| 00:45.00 | syle | i used that analogy on purpose to make what i was saying understandble to people who don;t know c |
| 00:45.29 | Nugget | "connect script library" is not an analogy. |
| 00:45.33 | syle | if your some elite c programmer i appologise |
| 00:45.36 | obsidian-studios | syle: what's C ? :) |
| 00:46.04 | syle | lol |
| 00:46.04 | Nugget | "connect script library" is as much of an analogy as "your" is a contraction. |
| 00:46.09 | Nugget | to use an analogy |
| 00:46.13 | obsidian-studios | syle: what about wannabies :) we get no credit |
| 00:46.15 | syle | nugget do me a favor |
| 00:46.18 | obsidian-studios | or apologies :) |
| 00:46.19 | syle | go smoke a joint please! |
| 00:46.29 | Nugget | stop saying dumb things please! |
| 00:46.53 | syle | nugget is having one of those condescending days i guess |
| 00:46.57 | syle | or is he like that all the time |
| 00:47.20 | Nugget | it's just becaue I'm old and crotchety :) |
| 00:47.27 | obsidian-studios | man I got my little * setup going but I would really like to see if I can get the caller id info to my analog phones,then I would be happy :) |
| 00:48.54 | obsidian-studios | I am pretty sure the Cisco router is not helping since it does the sip -> analog, but I know nothing about the caller id stuff? Is that sent along with the sip information, or separate? I keep seeing stuff on Cisco's docs about SGCP and MGCP, and * has a mgcp.conf file? But do I use that with sip or instead of sip? |
| 00:49.11 | drbrown | Has anyone tried to get spa-841's to work in an intercom setup? |
| 00:49.28 | drbrown | I have been working on it, but am unable to get it to work |
| 00:50.06 | benw | hi, I'm getting chan_iax2.c:2212 create_addr: No such host: 3230 |
| 00:50.30 | *** join/#asterisk Rick_Hunter (~rhunter@02-135.008.popsite.net) |
| 00:51.00 | benw | I have changed iax.conf extensions.conf and sip.conf and it was working. not sure whats broken, any ideas? |
| 00:51.16 | *** join/#asterisk chaoscon (~ph33r@chaoscon.user) |
| 00:51.51 | shido6 | that sux |
| 00:52.02 | shido6 | whats in iax.conf and extensions.conf |
| 00:52.05 | shido6 | pastebin.ca |
| 00:52.21 | shido6 | crap I thought I could stick with xp again... screw it brb |
| 00:52.39 | *** part/#asterisk Nuxi (~nuxi@lin-dsl-static-206-222-194-115.inetnebr.com) |
| 00:52.41 | Moonwick | syle: he's been like that for the 8 years I've known him, so it's nothing new :) |
| 00:52.57 | benw | in the iax.conf I have added register => 3230:username@gw2.austechpartnerships.com |
| 00:53.42 | benw | shido6: i'm trying to connect to another iax server for outbound calls |
| 00:53.51 | *** join/#asterisk jusmon (~tarzan@68-235-252-161.atlsfl.adelphia.net) |
| 00:54.41 | syle | thats ok |
| 00:54.56 | syle | i plan on being the newbie in here anyways when i get my digium card eventually lol |
| 00:55.06 | benw | in the extensions.conf i have exten => _[8-9]XXXXXXX,1,Dial(IAX2/3230/${EXTEN}) |
| 00:55.40 | benw | but when I try to dial 9XXX XXXX i get the above error. any ideas? |
| 00:56.25 | *** join/#asterisk shido6 (~shido@d57-87-253.home.cgocable.net) |
| 00:57.22 | exonic | benw, perhaps try debugging iax on both sides |
| 00:57.27 | shido6 | ? |
| 00:57.36 | shido6 | is there a pastebin I missed during reboot? |
| 00:57.51 | obsidian-studios | no |
| 00:57.54 | JunK-Y | shido6: nah |
| 00:58.03 | benw | exonic: do you mean use tcpdump to see whats happening to the packets? |
| 00:58.34 | exonic | benw, in asterisk do "iax2 debug" |
| 00:58.51 | tainted- | obsidian-studios why obsidian.. are u guys volcanic in nature? or just black and mysterious like shaft |
| 00:58.52 | *** part/#asterisk jhowardPA (~jhoward@12.25.177.120) |
| 00:59.12 | exonic | lol |
| 01:00.17 | obsidian-studios | tainted-: long long story, had another business that I purchased name for, and buddy already had this name when I started business with him, but he went other ways so I kept at it. I did some research once and guess obsidians can come in many tints, black based, one with a purple tint. There was like some indian spiritual properties. I lost the link before I could copy the blurb |
| 01:00.20 | benw | exonic: i switched on the debugging but I get the same error |
| 01:00.37 | *** join/#asterisk hassler (~hassler@cpe-65-31-36-179.woh.res.rr.com) |
| 01:01.04 | obsidian-studios | tainted-: if you want to laugh I was in CA at the time. When I moved back to my home town of Jacksonville FL, there was a game company that used to use that name, less the hyphen:) Nice quinky dink |
| 01:01.19 | hassler | hey folks, is gnophone still active / valid, or is there a better option? |
| 01:01.22 | exonic | benw, what's the eror, I musta missed in |
| 01:01.44 | opus_ | Comedian Chris Tucker was arrested in April 2005 and charged with reckless driving and fleeing to elude after he did not immediately pull over his speeding 2005 Bentley. Tucker, 33, spent about 30 minutes in a McDuffie County lockup before posting cash bond and being released. According to cops, the "Rush Hour" star, an Atlanta native, was doing 109 mph on Interstate 20 when clocked by state troopers. |
| 01:01.44 | tainted- | actually it sounds like a gaming company |
| 01:01.46 | exonic | benw, oh I see it, that's not the right format. Check the wiki on IAX channels |
| 01:02.00 | *** join/#asterisk tzafrir_laptop (~tzafrir@bzq-179-40-134.cust.bezeqint.net) |
| 01:02.09 | opus_ | whopts wrong channel |
| 01:02.31 | obsidian-studios | tainted-: even funnier another buddy of mine loves nwn, he sucked me into it. Obsidian Entertainment is making nwn2 :) |
| 01:02.43 | Optic | moo |
| 01:02.50 | obsidian-studios | tainted-: seems everywhere I go I am surrounded by other obsidians :) |
| 01:03.03 | tainted- | it's okay, i'm surrounded by tainteds |
| 01:03.18 | tainted- | mostly porn related though.. sigh |
| 01:03.25 | benw | exonic: whats not the right format? |
| 01:03.30 | obsidian-studios | man 109 is nothing I drive faster than that daily and I am not kidding :) |
| 01:03.41 | obsidian-studios | love the FL roads :) |
| 01:03.59 | tainted- | well if u were in CA u know the crazy driving style they've got here |
| 01:04.01 | Qwell | please, you guys think you have it bad? Check out what google shows for "Qwell". I'm a lice treatment. |
| 01:04.13 | shido6 | ... |
| 01:04.19 | Qwell | and an RPG currency, or something |
| 01:04.22 | tainted- | i remember grannies flying by in civics when i first came to LA while i was doing 80 |
| 01:04.33 | shido6 | turbo'd ? |
| 01:04.36 | shido6 | hondata |
| 01:04.41 | shido6 | nitrous beasts!! |
| 01:04.45 | Optic | i've pretty much got the go-ahead to do a 30-set PRI-based asterisk setup at work :) |
| 01:04.46 | obsidian-studios | tainted-: it's only bad in CA because you have allot of immigrants and etc from places where driving is limited or not the same, so their skills are non existent. In the state with the most and worse traffic in the US |
| 01:04.47 | Optic | whoot |
| 01:05.09 | shido6 | have fun Optic , stare at the LEDS for 5 minutes for me |
| 01:05.24 | Qwell | obsidian-studios: If you live in CA, it's stupid to not have insurance that covers uninsured motorists. |
| 01:05.29 | Optic | heh |
| 01:05.35 | tainted- | Qwell yea ur nick sucks |
| 01:05.39 | Qwell | tainted-: :p |
| 01:05.44 | *** part/#asterisk hassler (~hassler@cpe-65-31-36-179.woh.res.rr.com) |
| 01:05.45 | obsidian-studios | nDuff: I miss 280 to Santa Cruz at like 2-3 AM doing like 130-140 or hitting the summit at like 80 |
| 01:05.57 | Optic | i'm ordering a bunch of sets tomorrow to try... I need to find "basic" and "executive" phones |
| 01:06.10 | obsidian-studios | Qwell: sore subject I lost on of my favorite cars in HS to a uninsured motorist |
| 01:06.13 | Optic | i'm thinking polycom |
| 01:06.18 | tainted- | Qwell looks like qwest, reminds me of queer or qweeve |
| 01:06.38 | obsidian-studios | Qwell: moved to CA with my mom who worked for Prudential. I had life insurance since I was born :) |
| 01:07.48 | *** join/#asterisk amir_ (~amir@195.226.9.186) |
| 01:08.06 | obsidian-studios | ok I am off for the evening to work on other things etc. I will be back tomorrow to get info on cid, sip, etc |
| 01:08.22 | obsidian-studios | l8r all, * on |
| 01:08.26 | PTG1234 | anyone need a dual xeon :) |
| 01:08.27 | Optic | moo |
| 01:08.33 | tainted- | i had a friend named jon zuhkowski-faust.. he used to tell people that it was pronounced with a silent h.. i was like.. no one can say your f'in name silent h or not |
| 01:08.39 | obsidian-studios | PTG1234: just as I leave, who does not :) |
| 01:08.43 | PTG1234 | hah |
| 01:08.44 | Optic | do you think a P4 2.8 would be okay for 30 sets and PRI? |
| 01:08.46 | PTG1234 | i just built a new one |
| 01:08.49 | *** join/#asterisk meppl (mephisto@pD9E686AE.dip.t-dialin.net) |
| 01:08.50 | PTG1234 | and graphica are choppy |
| 01:08.53 | PTG1234 | and i can't figure out why |
| 01:08.56 | PTG1234 | so i am about ready to just sell it |
| 01:09.03 | meppl | gute nacht |
| 01:09.06 | obsidian-studios | PTG1234: nah I only accept donations :) |
| 01:09.41 | *** part/#asterisk obsidian-studios (~obsidian-@c-66-177-188-197.hsd1.fl.comcast.net) |
| 01:10.47 | *** join/#asterisk BoRiS (boris@S01060040ca1e5b54.wp.shawcable.net) |
| 01:11.37 | Optic | boing |
| 01:12.10 | *** join/#asterisk shido6 (~shido@d57-87-253.home.cgocable.net) |
| 01:12.35 | *** join/#asterisk _Vile (~vile@90.b160.bendtel.net) |
| 01:19.07 | nwhit | ~nwhit |
| 01:19.35 | _Vile | ~jbot |
| 01:19.36 | jbot | somebody said jbot was heading for a crash is assigned nothing and reported nothing. |
| 01:19.46 | _Vile | ~karma _Vile |
| 01:19.46 | jbot | _vile has neutral karma |
| 01:20.06 | tainted- | _Vile-- |
| 01:20.11 | _Vile | ksdfsdfjkh |
| 01:20.14 | Optic | ~cow |
| 01:20.15 | jbot | I am a cow, hear me moo. I eat grass and weigh twice as much as you. |
| 01:20.17 | tainted- | ~karma _Vile |
| 01:20.17 | jbot | _vile has neutral karma |
| 01:20.25 | tainted- | hmmm |
| 01:20.30 | nwhit | urg... i started something |
| 01:20.31 | *** join/#asterisk implicit (~implicit@lgb-cust-66.18.140.106.mpowercom.net) |
| 01:20.36 | JunK-Y | ~karma junky |
| 01:20.36 | jbot | junky has neutral karma |
| 01:20.41 | _Vile | tainted, I will be upset |
| 01:20.42 | implicit | is anyone here using any mediatrix equipment |
| 01:20.43 | implicit | ? |
| 01:20.50 | nwhit | ~karma nwhit |
| 01:20.50 | jbot | nwhit has neutral karma |
| 01:21.00 | implicit | fucking configuration sucks |
| 01:21.01 | nwhit | ~where nwhit |
| 01:21.26 | nwhit | ~madcow |
| 01:21.27 | jbot | moooooooheh hehehehe! |
| 01:21.43 | JunK-Y | implicit: nah |
| 01:21.50 | _Vile | ~asterisk |
| 01:21.51 | jbot | extra, extra, read all about it, asterisk is a PBX (Private Branch eXchange) and telephony toolkit. http://www.asterisk.org |
| 01:21.54 | implicit | JunK-Y, how've you been? |
| 01:21.57 | nwhit | ~ser |
| 01:21.58 | jbot | extra, extra, read all about it, ser is Sip Express Router - see http://www.iptel.org/ser/ |
| 01:22.04 | implicit | ~ser kicks ass |
| 01:22.21 | _Vile | anyone have any external updates outside of the mailing list on the DS-3000? |
| 01:22.27 | nwhit | ~sangoma |
| 01:22.28 | jbot | methinks sangoma is a company that makes PRI cards the way Digium should have done it in the first place.... |
| 01:22.37 | _Vile | hah |
| 01:22.41 | nwhit | yup |
| 01:22.43 | nwhit | i agree |
| 01:22.53 | JunK-Y | implicit: a lot of work. |
| 01:23.12 | _Vile | i'm taking a full week off in july |
| 01:23.14 | implicit | JunK-Y, same here man, lots of cool SER stuff |
| 01:23.35 | JunK-Y | i never touched SER, * is more a priority atm. |
| 01:23.54 | implicit | JunK-Y, well i do ITSP stuff so it's a must |
| 01:24.05 | nwhit | ~snom |
| 01:24.06 | nwhit | ~sipura |
| 01:24.07 | nwhit | ~dell |
| 01:24.08 | jbot | Dude! Are you getting a Dell? |
| 01:24.08 | nwhit | oh well |
| 01:24.10 | _Vile | two days of which is spent flying, but days off |
| 01:24.31 | nwhit | finally |
| 01:24.50 | nwhit | ~help |
| 01:25.25 | _Vile | fiel joo |
| 01:25.38 | file | eep |
| 01:25.43 | _Vile | goodbye |
| 01:26.20 | Sedorox | mm.. that didn't come out as planned.... |
| 01:27.13 | implicit | JunK-Y, i'm doing all the registration, accouonting, billing, routing, vertical services, ... all on SER |
| 01:27.52 | implicit | no asterisk in there now, but i'm going to pop it in to be a vm server and for error messsages etc |
| 01:28.14 | JunK-Y | oky |
| 01:28.17 | _Vile | what are you using for billing/routing? |
| 01:30.52 | Faithful | Hey guys, do I really have to install X in order to use bluetooth with * ? |
| 01:31.42 | nDuff | I'm trying to run Asterisk CVS, and it's failing on boot: "__load_resource: /usr/lib/asterisk/modules/res_features.so: undefined symbol: ast_monitor_stop". ast_monitor_stop is defined in res_monitor,so; how can I specify that res_monitor.so be loaded before res_features.so? |
| 01:32.04 | shido6 | turn it off in modules.conf |
| 01:33.44 | file | or delete /usr/lib/asterisk/modules and do a make install again |
| 01:33.50 | *** join/#asterisk fafnir (~hello@tdds-gw.Moscow.gldn.net) |
| 01:33.56 | file | er wait haha |
| 01:34.20 | file | it should automatically load it in that order |
| 01:34.40 | shido6 | noload => res_features.so in /etc/asterisk/modules.conf |
| 01:34.49 | file | odd though, that it isn't... |
| 01:36.27 | JunK-Y | u cant do noload => res_features.so |
| 01:36.44 | JunK-Y | chan_sip, chan_zap needs it. |
| 01:36.59 | file | yay stuff |
| 01:37.05 | nDuff | shido6, I don't want to disable res_features -- I just want it to be loaded _after_ res_monitor. |
| 01:37.19 | JunK-Y | nDuff: try what we already told ya. |
| 01:37.46 | *** join/#asterisk amir|away (~amir@195.226.9.186) |
| 01:38.48 | nDuff | JunK-Y, I'm a C programmer, I've written code that uses dlopen() on more than one occasion, I understand -why- clearing and rebuilding the modules would sometimes work and consequently I understand under what circumstances it won't. Mine are a member of the latter set. |
| 01:39.11 | file | nDuff: yeah I didn't read enough, my attention is elsewhere |
| 01:39.40 | file | nDuff: but lemme go into the nitty gritty - it *should* automagically load in the correct order, cause if it didn't - loads of other people would be complaining... have you modified modules.conf at all? |
| 01:39.54 | *** join/#asterisk anthm (~anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 01:39.54 | *** mode/#asterisk [+o anthm] by ChanServ |
| 01:40.27 | alt | nDuff: why not just do a load=>res_monitor.so and then load=>res_features.so? will that help? |
| 01:40.46 | nDuff | file, not that I know of. However, I'm working off a Gentoo ebuild modified to work against CVS, so it's possible it twiddled something during the build process. |
| 01:40.58 | file | funky |
| 01:41.53 | nDuff | okay -- I put load=>res_monitor.so before load=>res_features.so, and it's no longer lacking the symbol in question -- now it's looking for adsi_available; so it looks like it really is an ordering issue. |
| 01:42.08 | file | is there autoload => yes ? |
| 01:42.10 | nDuff | (not that either load=> line was there beforehand -- had to add them both) |
| 01:42.11 | alt | nDuff: yay! |
| 01:42.19 | *** join/#asterisk figfig (~jim@adsl-66-218-45-62.dslextreme.com) |
| 01:42.23 | nDuff | file, yes. |
| 01:42.45 | file | nDuff: I think you should grab CVS head and install from there ;) including samples |
| 01:43.06 | figfig | I am going crazy. I have two broadvoice accounts, and one asterisk box. I want to use both broadvoice accounts in one asterisk box. Is this even possible? I can't get it to work. |
| 01:43.20 | file | figfig: sure it's possible |
| 01:43.28 | Optic | that's no problem |
| 01:43.46 | Optic | it's the POWER of the DIALPLAN! |
| 01:43.48 | Optic | muahaha |
| 01:43.50 | figfig | for some reason I can not figure out how to do it. |
| 01:43.52 | file | yay dialplan |
| 01:44.00 | figfig | I have two registrations in sip.conf |
| 01:44.09 | figfig | and two contexts in the dialplan |
| 01:44.09 | nDuff | comparing the modules.conf in /etc/asterisk to modules.conf.sample in CVS, they match (except for the changes I just made). |
| 01:44.17 | figfig | and two sections in sip.conf |
| 01:44.22 | figfig | but it doesn't work |
| 01:44.26 | nDuff | will try doing a stock, completely unmodified build though. |
| 01:45.03 | figfig | If I comment out the stuff for one number in sip.conf it works, and same for the other, but it doesn't work when they are both there. |
| 01:45.22 | figfig | any suggestions? |
| 01:45.45 | figfig | or any information that I can give that would allow someone here to see the problem? |
| 01:46.06 | figfig | Am I correct that I need two registrations, and two sections in sip.conf for each account? |
| 01:46.14 | Optic | fig: you need to understand the dialplan, really |
| 01:46.28 | Optic | you could map a dial prefix to each one, 8 and 9 for example |
| 01:46.34 | Optic | o ryou oculd probably select them at random |
| 01:46.57 | figfig | optic: right now I have a context for each account |
| 01:47.08 | Optic | in sip.conf? |
| 01:47.27 | file | for incoming or outbound? |
| 01:47.38 | figfig | optic: I have two contexts in extensions.conf, and point to each one in the corresponding section of sip.conf |
| 01:47.49 | Optic | that should work |
| 01:48.03 | file | you're pointing to both? |
| 01:48.14 | figfig | optic: when I comment one number from sip.conf, the other works. |
| 01:48.30 | file | cause, like, broadvoice matches based on IP address on inbound calls... so like only one will match and go to the context |
| 01:49.07 | figfig | file: I have a line context=<context> in the sections of sip.conf (ie- one points to the first context, the other to the other). |
| 01:49.42 | file | figfig: you have two entries in sip.conf for inbound calls, correct? |
| 01:49.50 | file | are they both peers? |
| 01:49.51 | figfig | file: so you can't set up two accounts with one asterisk box. |
| 01:50.00 | file | yes, you can |
| 01:50.15 | file | you just don't understand how the call gets into asterisk |
| 01:50.27 | file | asterisk can't separate them because no authentication takes place - we match based on the IP address the call is coming from |
| 01:50.45 | figfig | file I have two lines in sip.conf that start with rigister => |
| 01:50.56 | file | a register line is not how a call gets into the box |
| 01:51.01 | figfig | file: they are both peers |
| 01:51.05 | file | it simply tells broadvoice where to send your calls |
| 01:51.41 | *** join/#asterisk heath__ (~heath@12-215-32-56.client.mchsi.com) |
| 01:51.45 | file | figfig: do this, make one context with both numbers... and set the entries in sip.conf to that context |
| 01:51.57 | file | figfig: if it works, you owe me a can of 7-up |
| 01:52.13 | heath__ | which signalling should i use if i'm connecting a dialog card to a quad card (terminating to sip) |
| 01:52.18 | figfig | file: do I need two [] sections in sip.conf? |
| 01:52.36 | figfig | file: somewhere I need to put both passwords. |
| 01:52.36 | *** join/#asterisk mentat (~mentat@pcp01260498pcs.nhaven01.ct.comcast.net) |
| 01:53.10 | file | figfig: have two entries, but set the contexts to a single context... not two |
| 01:53.21 | figfig | file: one second, I will try. |
| 01:53.25 | file | figfig: asterisk doesn't know which account your call comes in from, so it goes to one of the two... |
| 01:55.51 | figfig | file: now I am not even getting into aterisk (ie- a call is answered by the broadvoice answering machine). |
| 01:56.13 | file | figfig: insecure=very is set in sip.conf for the peer? |
| 01:56.41 | file | the host is set to where they're sending the call from? |
| 01:56.57 | Sift | so with * you basically can sound like a huge company...on incoming calls...based on what you have enabled? |
| 01:57.17 | file | Sift: sure, why not |
| 01:57.22 | Sift | is there places where you can get recording done |
| 01:57.30 | Sift | for professional voice menus |
| 01:57.31 | file | http://thevoice.digium.com/ |
| 01:57.44 | tainted- | i just sent an awesome e-mail telling a bitch interviewer that i just closed a salary deal for 40% more than what they were offering, and that my new employer is looking for more talent if they'd like to pass me their resume. lol |
| 01:58.35 | Nugget | haha |
| 01:58.40 | ariel_ | tainted-, what type of work is it? |
| 01:58.45 | figfig | file: I am trying to figure out why sip show registry isn't even showing the account registered now. One second. |
| 01:59.06 | tainted- | ariel_ c# programming |
| 01:59.25 | ariel_ | tainted-, I see. A programmer. |
| 01:59.27 | Sift | file does that cost? |
| 01:59.33 | file | Sift: yes |
| 01:59.42 | file | Sift: you expect people to freely record stuff for you out of their time? |
| 01:59.46 | file | gah, you already get a free phone system |
| 01:59.52 | file | stop being cheap :P |
| 01:59.59 | Sift | lol |
| 02:00.01 | Sift | true |
| 02:00.31 | Sift | its not that I was being cheap...just that since I havent signed up yet...wasnt sure what was involved |
| 02:01.26 | tainted- | 1 credit = 1 professionaly recorded prompt or twenty words |
| 02:01.32 | tainted- | damn |
| 02:01.43 | tainted- | was gonna ask her to read war & peace |
| 02:01.50 | *** join/#asterisk SuperN (SuperN@100stb35.codetel.net.do) |
| 02:01.51 | figfig | file: I think it now works. Thanks! Can you explain to me why they need to both be in the same context? |
| 02:01.52 | Qwell | tainted-: should only be a few bucks |
| 02:01.56 | SuperN | good night |
| 02:02.18 | file | figfig: because asterisk matches based on IP address and has no idea what account the call belongs to |
| 02:02.45 | SuperN | can somebody help please? |
| 02:03.04 | file | SuperN: can't help if we don't know what to help with |
| 02:03.15 | shido6 | SuperN, ? |
| 02:03.16 | figfig | file: how does it eventually know what extension to send to then (ie- I have a different extension in the registeer => lines, that is the only way it is differentiating). |
| 02:03.20 | shido6 | whats wrong SuperN ? |
| 02:03.26 | file | figfig: magic |
| 02:03.37 | SuperN | file, I need to know if I can use the asterisk with a Netphone KE1020 |
| 02:04.18 | file | figure out what protocol it uses, look at the list on http://www.asterisk.org/ and if it's there - then you can |
| 02:04.43 | tainted- | i'd pay money to hear allison say "g's up -- hoes down" |
| 02:05.11 | SuperN | yes, the NetPhone KE1020A supports most of the asterisk's protocols |
| 02:05.11 | tainted- | or "rub your titties if u love big pappa" |
| 02:05.24 | figfig | file: I just mean, if it can differentiate which account is coming in to to send it to the corresponding extension, why can't it also figure out the account that is coming in to send to the correct context.... |
| 02:05.32 | file | message me and die |
| 02:05.38 | file | er I mean, don't message me |
| 02:05.45 | file | tainted-: DEAD YOU ARE. |
| 02:05.48 | file | Qwell: you too |
| 02:05.50 | Qwell | ;] |
| 02:05.51 | tainted- | lol |
| 02:06.04 | Qwell | at least 5 :p |
| 02:06.06 | file | figfig: just accept it, unless you wanna rewrite it |
| 02:06.34 | puowvip | ninight |
| 02:06.43 | file | SuperN: no. |
| 02:06.43 | Qwell | tainted-: think she'd do a phone sex line? |
| 02:06.58 | figfig | file: I don't want to rewrite it, just understand it (so that I don't have to bug someone here the next time I run into this). |
| 02:07.10 | SuperN | file, no what? |
| 02:07.27 | figfig | file: oh well, it works now. |
| 02:07.28 | file | figfig: most people don't use the / at the end, cause most providers send the number dialed... and asterisk uses that as the extension, try it and see |
| 02:07.42 | file | SuperN: no as in I don't do private messages |
| 02:07.47 | SuperN | ok. |
| 02:08.06 | SuperN | can I use the asterisk just for LAN? |
| 02:08.29 | JunK-Y | LAN? |
| 02:08.37 | figfig | file: perhaps that would be a cleaner way to do what I am doing. |
| 02:08.54 | SuperN | yeah, Junk-Y.. just to call within the LAN |
| 02:08.57 | SuperN | is this posible? |
| 02:09.04 | nDuff | SuperN, of course. |
| 02:09.16 | JunK-Y | natted? |
| 02:09.27 | SuperN | I don't want to have internet access |
| 02:09.29 | figfig | file: I will try, one second. |
| 02:09.32 | JunK-Y | sure. |
| 02:09.33 | SuperN | I just to do calls in the same LAN.. |
| 02:11.10 | nDuff | SuperN, yes, that'll work fine. |
| 02:12.32 | SuperN | I have installed the asterisk.. where can I find basic information about setting up 2 Netphones? |
| 02:16.43 | figfig | file: strangly enough it doesn't work when I set the extension as the phone number in extensions.conf |
| 02:17.47 | nDuff | file, cleared out /etc/asterisk and /usr/lib/asterisk/modules, did "make install" on a pristine copy of CVS HEAD, same issue. |
| 02:19.26 | SuperN | I have installed the asterisk.. where can I find basic information about setting up 2 Netphones? |
| 02:20.16 | bkw_ | READ |
| 02:20.53 | SuperN | jeje.. |
| 02:23.13 | shido6 | he |
| 02:23.14 | shido6 | h |
| 02:23.18 | *** join/#asterisk iq (~iq@70-59-161-163.omah.qwest.net) |
| 02:26.07 | figfig | file: thanks for your info, it saved me more annoyance! |
| 02:29.27 | *** join/#asterisk yxa (~void@203.118.40.42) |
| 02:35.58 | *** join/#asterisk insync (~spam@66-188-104-11.mad.wi.charter.com) |
| 02:39.16 | *** join/#asterisk remmo (~rem@smack.isp.net.au) |
| 02:42.03 | nDuff | well, heck. |
| 02:42.18 | nDuff | I just had to add "load => res_monitor.so", and that was it. |
| 02:42.44 | usam | one simple question, what is the most profitable to do with asterisk? .. Too many feature confuses me where to start ;) |
| 02:45.03 | *** join/#asterisk drbrown (~chatzilla@user-0cdv208.cable.mindspring.com) |
| 02:45.33 | *** join/#asterisk kb1_kanobe (~krisbouti@h24-207-80-55.cst.dccnet.com) |
| 02:46.14 | drbrown | how is everyone this evening? |
| 02:48.56 | kb1_kanobe | all messed up from another day of *, but otherwise fine... |
| 02:49.00 | *** join/#asterisk khb (~root@ool-43516e13.dyn.optonline.net) |
| 02:57.43 | *** part/#asterisk khb (~root@ool-43516e13.dyn.optonline.net) |
| 02:59.39 | kb1_kanobe | Anyone fancy a rousing game of 'find out why my_zt_write() is being intermittantly blocked (EAGAIN) when writing to a t100p card'? |
| 03:02.20 | yxa | what is the best way to configure asterisk to access a database and return a result? |
| 03:03.09 | denon | yxa: look at AGI |
| 03:03.16 | denon | ~google asterisk agi |
| 03:03.42 | yxa | denon thanks |
| 03:04.40 | *** join/#asterisk Specky[W] (~stefan@p50922734.dip0.t-ipconnect.de) |
| 03:04.48 | *** part/#asterisk Specky[W] (~stefan@p50922734.dip0.t-ipconnect.de) |
| 03:05.11 | file | I'll pretend I didn't click that jast-agi link |
| 03:06.27 | iq | Hi, how can I tell the Asterisk version is running on a machine? |
| 03:06.40 | Qwell | show version? |
| 03:06.50 | kb1_kanobe | iq: Perhaps 'show version' in the console? |
| 03:07.22 | iq | kb1_kanobe, Qwell, thank :) |
| 03:08.25 | kb1_kanobe | 'help' is also a useful console command. |
| 03:08.27 | iq | Asterisk CVS-HEAD-04/11/05-06:05:44 |
| 03:08.44 | iq | kb1_kanobe, most useful :) |
| 03:15.36 | *** join/#asterisk santiago (~santiago@63.245.86.199) |
| 03:17.00 | *** join/#asterisk Cresl1n (~matt@216.207.245.23) |
| 03:18.05 | *** join/#asterisk Rick_Hunter (~rhunter@04-089.008.popsite.net) |
| 03:24.10 | Sift | anyone have a * box I can call to see how it sounds? |
| 03:24.24 | kb1_kanobe | calling from what? |
| 03:24.35 | Sift | packet8 |
| 03:25.27 | kb1_kanobe | Sorry, do you mean you want to call an * box on the pstn from packet8 or you want to call someone who has an * box that has VoIP channels to packet8? |
| 03:25.39 | Qwell | sift: one sec, lemme get my FWD number |
| 03:25.49 | Sift | I want to call an asterisk box |
| 03:26.03 | Sift | from my packet8 line |
| 03:26.15 | Qwell | Sift: try dialing 0451527043 |
| 03:26.19 | Qwell | It should go straight to my VM |
| 03:26.36 | Sift | dialing |
| 03:26.54 | Sift | haa |
| 03:27.06 | Sift | and that is just a mailbox on your *? |
| 03:27.10 | Qwell | yeah |
| 03:27.11 | Qwell | try... |
| 03:27.14 | Sift | thats sweet |
| 03:27.16 | Qwell | 0451613 |
| 03:27.21 | Qwell | I think it was 613...its an echo test |
| 03:27.23 | Sift | how did you get the lady ? |
| 03:27.25 | Qwell | actually, no |
| 03:27.31 | Qwell | that won't work... |
| 03:27.41 | Qwell | Sift: All those prompts are in asterisk-sounds |
| 03:28.10 | Sift | hmm man I gotta install this |
| 03:28.16 | *** join/#asterisk bah (048830696@AC925E85.ipt.aol.com) |
| 03:28.39 | Qwell | Sift: Just remember the 0451 if anybody ever gives you a FWD number to call. |
| 03:28.39 | Sift | but is it possible to use packet8 with *? |
| 03:28.43 | Qwell | not easily |
| 03:28.56 | Qwell | it'll go voip>analog>voip |
| 03:29.01 | Sift | hmm |
| 03:29.12 | Sift | so its better for me to get a sipura 3000 |
| 03:29.12 | Sift | ? |
| 03:29.15 | Sift | with broadvoice? |
| 03:29.25 | Qwell | better then packet8? Probably |
| 03:29.31 | Qwell | I hear bad things about broadvoice though... |
| 03:29.34 | Sift | hmmm |
| 03:29.51 | Sift | so who do you recommend? |
| 03:29.57 | Qwell | I like nufone, personally |
| 03:30.15 | Sift | gotta link? |
| 03:30.18 | Qwell | nufone.net |
| 03:30.34 | Sift | heh |
| 03:30.35 | Sift | Due to system upgrades we cannot accept any new accounts at this time. |
| 03:30.35 | Sift | Please be patient. |
| 03:30.36 | Qwell | They aren't accepting customers through the site right now, but you can send an email to greg@nufone.net, and he'll help you |
| 03:30.46 | Qwell | Just make sure to tell him Qwell sent you. :) |
| 03:31.05 | Sift | so what options do they have |
| 03:31.06 | *** join/#asterisk MaggieL (~chatzilla@lata228-01-c210.lata228-c.voicenet.com) |
| 03:31.11 | Qwell | as far as? |
| 03:31.25 | Qwell | caller id, voicemail, etc? |
| 03:31.30 | Sift | do they lock down devices? |
| 03:31.35 | Qwell | Not a chance |
| 03:31.35 | Sift | or can you customize |
| 03:31.45 | Qwell | You just connect your * box straight to them |
| 03:31.51 | Sift | hmm |
| 03:32.01 | Sift | but you need an fxo card right? |
| 03:32.04 | Qwell | nope |
| 03:32.15 | Sift | its strictly IP? |
| 03:32.18 | Qwell | yes |
| 03:32.21 | *** join/#asterisk vinko (~vinkoval@voice.iwobble.com) |
| 03:32.22 | Sift | interesting |
| 03:32.32 | Sift | call quality? |
| 03:32.41 | Qwell | out of 10, I'd give them a 9, honestly |
| 03:32.58 | Sift | no fade or echo? |
| 03:33.00 | Qwell | I was so surprised when I got my phone...I had never really used voip, so I expect a 5 maybe |
| 03:33.14 | Sift | how much a month? |
| 03:33.24 | Qwell | Sift: well, occasionally it will break up, but thats my end lagging for some other reason |
| 03:33.28 | Qwell | 2c/minute outgoing |
| 03:33.54 | Sift | ohh so pay as you go |
| 03:33.54 | Qwell | and you can get a DID in MI for like $8 a month or so, with free incoming, OR, a US48 toll-free DID for $0/month and 2c/minute incoming |
| 03:34.48 | Sift | so 8 for .2 outgoing |
| 03:34.56 | iq | will new version of * have native SQL support ? |
| 03:34.59 | Qwell | right |
| 03:35.10 | Qwell | The only drawback to that, is they only have phonenumbers in MI right now |
| 03:35.18 | Qwell | but, if you don't mind it being in MI, its great |
| 03:35.21 | Sift | right now I pay 20 bucks for nationwide |
| 03:35.32 | Qwell | this is nationwide and Canada for 2c/min |
| 03:35.45 | Qwell | and VERY low rates for other countries. check out the rates.csv on the main page |
| 03:35.56 | Sift | whats your aver bill each mo |
| 03:36.16 | Qwell | only been using them a little over a month, so, 24 cents, technically |
| 03:36.33 | Sift | I gotta have an OR number |
| 03:36.46 | Qwell | Sift: I just went ahead and got an 8xx DID |
| 03:36.50 | Qwell | it works out great for me |
| 03:37.11 | Sift | so calls to VM dont get charged? |
| 03:37.31 | Qwell | They do |
| 03:37.42 | Sift | but calling out doesnt? |
| 03:37.57 | Qwell | calling out from nufone is always 2c/min |
| 03:38.01 | Sift | ahhh |
| 03:38.04 | Qwell | in Canada and the US that is |
| 03:38.24 | *** part/#asterisk JunK-Y (~junky@modemcable174.107-81-70.mc.videotron.ca) |
| 03:38.34 | Qwell | Sift: The only time you get charged for incoming, is if you have a tollfree DID |
| 03:38.43 | Sift | right |
| 03:41.21 | Sift | so * is all command line driven? |
| 03:41.38 | Qwell | there are GUI config utils, but I wouldn't recommend them |
| 03:42.17 | shido6 | dont freak out yet |
| 03:42.24 | shido6 | we have a new service coming out for vmail |
| 03:42.31 | Qwell | shido6: ooo, scoop? |
| 03:42.40 | shido6 | crap |
| 03:42.44 | shido6 | did I say that out loud? |
| 03:42.56 | Sift | a new service? |
| 03:43.12 | file | shido6: shhh be quiet |
| 03:43.23 | Qwell | Sift: shido is who I said to email. if you have any questions, I'd bet he'll answer them |
| 03:44.01 | Sift | ahh greg |
| 03:44.35 | iq | help: on one of my machine I am not getting audio of either side. No one can hear the other person |
| 03:44.42 | Qwell | iq: NAT? |
| 03:45.15 | iq | Qwell: client is on NAT (DSL Modem -> Router -> ATA) |
| 03:45.17 | Qwell | You know...as many times as I've seen NAT issues resolved, I couldn't even begin to tell you where to start. I don't pay attention to NAT issues, because they don't affect me. heh |
| 03:45.29 | Qwell | only one ATA? |
| 03:45.33 | file | forward ports, nat=yes |
| 03:45.35 | file | canreinvite=no |
| 03:45.37 | Qwell | hell, forward 10,000-20,000 to the ATA |
| 03:45.42 | Qwell | ..and 5060 |
| 03:45.45 | file | if asterisk inside set localnet and externip |
| 03:45.48 | Qwell | ok, here's a silly question |
| 03:45.53 | file | there, file's quick NAT lesson |
| 03:45.57 | Qwell | Vonage works behind NAT, without having to forward ports. |
| 03:45.58 | iq | file, nat=yes . already exist otherwise I can't even get the ring to work |
| 03:46.00 | Qwell | Whats up with that? |
| 03:46.11 | file | iq: qualify=yes, have them register and set canreinvite=no |
| 03:46.32 | file | Qwell: in reality you rarely have to forward the ports, it usually just works |
| 03:46.52 | file | like I use Eyebeam, two X-lites, a Cisco, and a PAP2-NA behind my NAT and don't forward any ports... they just work |
| 03:46.52 | Qwell | hmm, I think I recall PTG saying that once the connection is made, it'll stay open for a little while |
| 03:46.58 | Qwell | So if you redo the connection, it'll "just work" |
| 03:47.02 | Qwell | every so often, that is |
| 03:47.11 | Sift | file what do you need that many for :) |
| 03:47.15 | file | yeah... routers keep UDP mapping tables... |
| 03:47.20 | file | Sift: testing |
| 03:47.33 | Qwell | file: well, when you're done testing the cisco... :P |
| 03:47.46 | file | HA |
| 03:47.53 | Sift | is there any way to keep the signal all digital when using like broadvoice/vonage....those types of service? |
| 03:47.55 | Qwell | file: It *IS* an issue when both sides are on NAT though, right? |
| 03:48.02 | Qwell | digital? |
| 03:48.05 | Qwell | like, IP? |
| 03:48.11 | file | Qwell: that's when you use canreinvite=no, well you should use it anyway... but no, not usually an issue |
| 03:48.14 | Sift | yeah so it doesnt go to analog |
| 03:48.24 | Qwell | Sift: Vonage no, broadvoice and such yes |
| 03:48.31 | Qwell | Just get an IP phone, like a cisco |
| 03:48.52 | file | very strict firewalls/NATs are what kill |
| 03:49.00 | Qwell | NAT sucks |
| 03:49.10 | Sift | so this external hardware device just sits on the network and communicates with the * box over IP |
| 03:49.21 | Qwell | Sift: correct |
| 03:49.25 | Sift | and then on out to your WAN interface |
| 03:49.28 | Qwell | yep |
| 03:49.35 | Sift | to comm with BV or whomever |
| 03:49.36 | Qwell | then it gets to them, and will hit the PSTN |
| 03:49.39 | file | I just set everyone on my box up with nat=yes and canreinvite=no, I have yet to encounter a NAT user that has problems with those two... works fine, oh and qualify=yes to get past the sucky D-Link my best friend had |
| 03:49.55 | Qwell | sucky d-link? |
| 03:49.59 | file | D-Link router |
| 03:50.08 | file | it had a low expiry time on the UDP port mappings |
| 03:50.08 | Qwell | yes, I know what a d-link is. :p |
| 03:50.13 | Qwell | or was my question redundant? |
| 03:50.17 | Qwell | ahh |
| 03:50.24 | file | so if you tried to call him when he hadn't placed a call for a bit, you couldn't |
| 03:50.26 | Qwell | how long is "average"? |
| 03:50.28 | *** join/#asterisk kjs3 (~kjs3@c-24-98-102-138.hsd1.ga.comcast.net) |
| 03:50.33 | Qwell | like minutes, or hours? |
| 03:50.35 | file | it was a minute |
| 03:50.46 | Sift | so when they talk about the sipura having 1 pstn port |
| 03:50.46 | file | it was horrible... |
| 03:50.54 | Sift | what does that mean |
| 03:50.56 | Qwell | ~fxo |
| 03:50.57 | jbot | foreign exchange office - type of port you need to connect a POTS (Plain Old Telephone Service) line from your telco to a pbx http://www.digium.com/index.php?menu=fxsvfxo |
| 03:50.59 | Qwell | read that link |
| 03:51.00 | file | so I enabled qualify, and that fixed it :) |
| 03:51.23 | Qwell | Sift: very large different between FXS and FXO |
| 03:51.32 | Qwell | saying "PSTN port" is silly, and mostly meaningly |
| 03:51.35 | Qwell | ...less |
| 03:51.41 | Qwell | meaningly, heh, thats going in the log |
| 03:52.19 | iq | file, qualify=yes and qualify=yes |
| 03:52.19 | iq | canreinvite=no |
| 03:52.29 | file | nat=yes, qualify=yes, canreinvite=no |
| 03:52.32 | file | for peers behind NAT |
| 03:52.45 | iq | sorry about this, didn't meant to paste all here |
| 03:53.15 | iq | file: ya it is done. but still no audio. what you meant by have them register? |
| 03:53.45 | file | iq: and you reloaded? |
| 03:53.48 | Sift | ok interesting |
| 03:54.03 | iq | file: yes 'CLI> reload' |
| 03:54.17 | file | did you know you can do sip reload? it's quicker, ANYWAY |
| 03:54.32 | file | is your asterisk box behind NAT? |
| 03:54.38 | file | like, what is the setup... |
| 03:54.44 | iq | file: no I did not know. thanks :) ... I just have 3 extensions so far |
| 03:54.57 | file | well I meant in the context of where the box is, where the phones are |
| 03:55.06 | iq | file: astersik has internet IP . not behind NAT... all ports open |
| 03:56.33 | file | and just the clients are behind NAT? what type of NAT.... |
| 03:56.52 | *** join/#asterisk _solstice_ (~solstice@dsl-cap-209-5-169-205-cgy.nucleus.com) |
| 03:56.57 | iq | file: Asterisk box is in CA. Client is in Nebraska (at my home) |
| 03:57.37 | iq | ADSL Modem -> Belkin Router -> SPA-3000 |
| 03:57.45 | file | belkin? those are usually fine |
| 03:57.45 | Sift | ok heres a question...could i, from work, use a softphone...communicate through my home * box, and out like I was calling from home? |
| 03:57.52 | file | Sift: sure. |
| 03:58.18 | file | iq: I'd say manually map the rtp ports, 10000-20000 |
| 03:58.29 | Sift | how would that work, just point the address to my home IP? |
| 03:58.38 | file | Sift: sure |
| 03:58.57 | Sift | iq I grew up in lincoln |
| 03:59.01 | iq | file: I connect to about 6 different Asterisk servers using my SPA-2000 from behind my NAT - this is the first ever problem of this kind :) |
| 03:59.11 | iq | Sift, where are you now ? |
| 03:59.18 | Sift | OR |
| 03:59.21 | file | iq: did you set both peers to canreinvite=no? |
| 03:59.24 | Sift | moved here a year ago |
| 03:59.31 | _solstice_ | Does anyone know what the dial plan/config files would be to setup distinctive ring to force fax receive. The problem i am having is that some faxes don't seem to be sending the correct signalling to iniate the faxreceive part on asterisk .. |
| 03:59.33 | iq | file: map ports on server side? |
| 03:59.47 | file | iq: er I meant both peers |
| 04:00.02 | iq | file: all extensions got nat = yes, canreinvite=no and qualify=yes |
| 04:00.35 | file | quite odd |
| 04:00.41 | iq | Sift, I moved from St Louis in September |
| 04:01.11 | Sift | I told myself Id never go back to the midweset |
| 04:01.15 | Sift | west |
| 04:01.29 | iq | Sift, I dont blame you for that :) |
| 04:02.39 | iq | file: if UDP 10k-20k are not properly configured on server side then voice can't travel on RTP, right? |
| 04:02.53 | file | aye |
| 04:03.20 | Sift | so the most traffic you can produce in a single voip call is 6.4K/s? |
| 04:03.45 | file | there's a calculator... |
| 04:04.08 | Qwell | Sift: depends on the codec |
| 04:04.18 | file | thus why a calculator exists |
| 04:04.25 | Qwell | indee |
| 04:04.25 | Sift | link to said calculator? |
| 04:04.26 | Qwell | d |
| 04:04.36 | Qwell | ~google voip codec bandwidth calculator |
| 04:06.37 | Sift | can one control what codec is used when using commercial companies such as broadvoice? |
| 04:06.47 | iq | any easy way of confirming open ports ? |
| 04:06.50 | file | well, the company has to support it |
| 04:06.57 | Qwell | Sift: but, yes |
| 04:07.00 | Qwell | if ^ |
| 04:07.29 | Sift | so you can set it to 711 all day long on *, but if broadvoice doesnt support it...its still gonna sound like crap? |
| 04:07.38 | Qwell | Sift: no, it simply won't work |
| 04:08.42 | Qwell | it'll basically say that no compatible codecs were found |
| 04:08.47 | Sift | is there anytime where you can control it? |
| 04:08.55 | Sift | or are we always dependant |
| 04:08.59 | Qwell | Sift: If the providers uses more then one(most do) |
| 04:08.59 | Sift | on the company |
| 04:10.44 | Sift | yikes |
| 04:10.53 | Sift | packet8 is using 729 |
| 04:10.57 | *** join/#asterisk jeffpc (~jeffpc@ool-44c218a8.dyn.optonline.net) |
| 04:11.07 | Qwell | g729 is supposed to be good, but its not free |
| 04:11.18 | Sift | is there a noticable diff between 711 and 729 |
| 04:11.24 | Sift | isnt 711 higher qual |
| 04:11.47 | jeffpc | Hello. For whatever reason, when I plug my phone line into the LINE jack on my X100P, it keeps the line off-hook. Any suggestions? |
| 04:12.40 | Qwell | isn't line for...line? |
| 04:13.43 | jeffpc | I'm not following |
| 04:13.49 | kb1_kanobe | jeffpc: is the card correctly configured and running? That sounds like it's trying to busy-out the line for some reason. |
| 04:13.55 | Qwell | a line plug, is where you plug in a line |
| 04:14.03 | Qwell | a phone plug, is where you plug in a phone |
| 04:14.12 | Qwell | ~fxofxs |
| 04:14.14 | jbot | extra, extra, read all about it, fxofxs is An FXO port expects to receive dialtone and receive ring voltage. An FXS port expects to provide dialtone and provide ring voltage. |
| 04:14.39 | Qwell | the x100p has a passthrough port, which is where you would plug in a phone |
| 04:15.47 | jeffpc | Qwell: yes, I understand. The wire comes off the pole, and gets split to several phones and my box |
| 04:15.51 | jeffpc | which has the X100P in it |
| 04:16.04 | jeffpc | the cable is connected to the LINE port |
| 04:16.06 | jeffpc | that's it |
| 04:16.15 | Qwell | phone line...right, see, I read that as phone |
| 04:16.27 | Qwell | is it a clone? |
| 04:16.27 | jeffpc | Qwell: oh..sorry |
| 04:16.47 | jeffpc | Qwell: no, not as far as I know (it was given to me) |
| 04:17.24 | Qwell | Did you setup the drivers properly? |
| 04:17.36 | jeffpc | Qwell: I think so :-) |
| 04:17.48 | jeffpc | I'm following http://ourproject.org/docman/view.php/116/150/vm1.html |
| 04:18.41 | vinko | jeffpc: what does the zttool say Is it in Alarm? |
| 04:19.11 | jeffpc | vinko: hrmm |
| 04:19.13 | jeffpc | Red |
| 04:19.44 | vinko | jeffpc: if you unload the modules then reload them.. does it go to "OK" then back to RED? |
| 04:21.42 | iq | If ATA doesn't support one of the available Codecs - can this cause no audio problem? |
| 04:22.31 | jeffpc | vinko: I'm using 2.6 kernel |
| 04:22.39 | jeffpc | and it didn't want me to remove the module |
| 04:22.43 | jeffpc | so I forced it :-) |
| 04:23.06 | *** join/#asterisk drumkilla (~russell@12.21.241.80) |
| 04:23.06 | *** mode/#asterisk [+o drumkilla] by ChanServ |
| 04:23.08 | jeffpc | and now everything zaptel related segfaults :-) |
| 04:23.13 | Qwell | hmm |
| 04:23.27 | Qwell | If you upgrade newt, do you have to recompile zap? |
| 04:24.04 | Qwell | nevermind |
| 04:24.23 | jeffpc | I have to reboot to clean up this mess :-) |
| 04:24.29 | jeffpc | kernel going nuts |
| 04:24.34 | vinko | :) Ok |
| 04:25.01 | Dovid | . |
| 04:25.09 | Dovid | ~seen kevin |
| 04:25.13 | jbot | kevin <dd@85.96.41.206> was last seen on IRC in channel #debian, 53d 19h 58m 45s ago, saying: 'FREE Hosting Reseller www.otomotivshow.com'. |
| 04:31.18 | Sift | so what is better....getting a digium card....or a sipura 3000 |
| 04:31.31 | Qwell | depends |
| 04:31.44 | Qwell | I prefer the digium cards, but other people like the sipuras |
| 04:32.17 | *** join/#asterisk jeffpc (~jeffpc@ool-44c218a8.dyn.optonline.net) |
| 04:32.22 | jeffpc | still red |
| 04:32.51 | jeffpc | also while I was booting |
| 04:32.59 | jeffpc | I had the cable plugged in |
| 04:32.59 | Sift | one is out of site...the other has to sit on a desk |
| 04:33.23 | jeffpc | and the standard dialtone changed to ringtone |
| 04:33.24 | jeffpc | I think |
| 04:33.29 | jeffpc | at least it sounded like it |
| 04:35.31 | vinko | jeffpc: what I have had a problem with.. is that my Alarm was always "RED" and I found that |
| 04:36.19 | vinko | my land line what "Off Hook" some where.. Like a phone had been left of the hook |
| 04:37.03 | vinko | Do you have any line on Regular phone pluged into the same line that might be off hook? |
| 04:37.23 | jeffpc | there are several phones on the same line |
| 04:37.42 | jeffpc | but none of them make the in-use LED blink on the cordless phone :-) |
| 04:37.59 | vinko | If you pick one of them up do you hear dialtone? |
| 04:38.00 | jeffpc | but if I plug the cable into the digium board..bam |
| 04:38.05 | jeffpc | yes |
| 04:38.09 | vinko | ok |
| 04:38.40 | vinko | when you run the ztcfg utility.. does it show that its configured properly? |
| 04:39.19 | jeffpc | yes |
| 04:39.26 | jeffpc | 1 channel |
| 04:39.33 | jeffpc | FSX ks |
| 04:39.55 | jeffpc | FXS |
| 04:41.13 | jeffpc | interesting |
| 04:41.18 | blitzrage | yo yo |
| 04:41.22 | jeffpc | if I pick up a phone |
| 04:41.33 | jeffpc | I hear the dialtone loud and clear |
| 04:41.43 | jeffpc | if I plug the cable into x100p |
| 04:41.53 | jeffpc | the dialtone gets fainter |
| 04:41.58 | *** join/#asterisk isamar (~isamar@p8131-ipadfx21sasajima.aichi.ocn.ne.jp) |
| 04:42.03 | isamar | Hi folks |
| 04:42.07 | vinko | How many phones do you have on that line? |
| 04:42.14 | isamar | anybody using sangoma boards?? |
| 04:42.27 | *** join/#asterisk twisted (~twisted@twisted-professional-pdpc.developer.and.friend.of.asterisk) |
| 04:42.27 | *** mode/#asterisk [+o twisted] by ChanServ |
| 04:42.31 | twisted | did someone call satan? |
| 04:42.36 | Qwell | twisted: yes ma'am |
| 04:42.54 | jeffpc | vinko: around 5 |
| 04:42.58 | twisted | call me ma'am again and find out what steel feels like against your buttocks. |
| 04:42.59 | Qwell | twisted: 39372826 |
| 04:43.31 | Qwell | twisted: are you threatening me in a bizarre way? |
| 04:43.31 | jeffpc | vinko: actually 6 |
| 04:43.33 | vinko | plus the one that your using for the X100P |
| 04:43.43 | jeffpc | vinko: correct |
| 04:44.03 | jeffpc | btw, I'm not using the pass-though |
| 04:44.04 | twisted | Qwell, threats are a matter of perception. |
| 04:44.04 | vinko | Are you in a position to unplug the others.. |
| 04:44.09 | Qwell | twisted: :) |
| 04:44.16 | vinko | so your using just the one that you have for Asterisk? |
| 04:44.52 | vinko | the more phones on a line the more the circuit is degraded.. |
| 04:45.01 | jeffpc | vinko: well...if I were to plan it to happen in the middle of the night |
| 04:45.03 | jeffpc | :-) |
| 04:45.37 | jeffpc | you think it might be that the x100p is expecting better signal? |
| 04:45.49 | Qwell | could be |
| 04:45.53 | Sift | Qwell are you able to config * so that when you press 911 on your phone, it will call the local police, since voip doesnt officially support e911? |
| 04:46.03 | Qwell | Sift: sure, if you know the number |
| 04:46.09 | vinko | I have been told that any more that 4-5 phones on the same line causes problems.. |
| 04:46.12 | vinko | so it might be. |
| 04:46.24 | Qwell | well, since it *IS* a pbx, there should be 0 phones connected |
| 04:46.42 | vinko | QWell: in a perfect world. |
| 04:46.48 | vinko | but the wife still needs to use the phone.. |
| 04:46.55 | Qwell | Thats what FXS ports are for |
| 04:47.08 | jeffpc | Qwell: the phones where here before I even knew of asterisk |
| 04:47.19 | vinko | :) |
| 04:47.32 | jeffpc | and I doubt my parents would want their whole phone system redone :-) |
| 04:47.39 | kb1_kanobe | Sift: you would need a line that terminates on your local telco exchange to get 'normal' 911 integration. Even if you were to dial the call center directly via a VoIP provider they wouldn't get the Computer Aided Dispatch information for your call (ie. the 'Enhanced' bit of 911) |
| 04:48.24 | Sift | correct...but so it was "transparent" to the household...when they dial 911 |
| 04:48.42 | Qwell | Sift: You would need to repeat your address to the operator |
| 04:48.49 | Sift | thats fine |
| 04:48.55 | Sift | they always ask anyway |
| 04:48.57 | Sift | :) |
| 04:49.37 | Sift | is there anyway to block your # with asterisk? |
| 04:49.47 | Qwell | block an incoming #? |
| 04:49.49 | jeffpc | Qwell: actually, in the perfect world, I'd replace all the phones with VoIP equivalents :-) |
| 04:50.32 | Sift | block outgoing |
| 04:50.42 | kb1_kanobe | Sift: I have an analog line at each of my business locations that gets the first 911 call and pumps it out there. Subsequent calls overflow onto my central PRI which has been tagged as a non-geographic number in the database of the Call Center that provides our 911 |
| 04:50.43 | Qwell | Sift: like mask your callerid? |
| 04:50.45 | Sift | caller ID I mean |
| 04:50.55 | Qwell | some providers let you change the CID |
| 04:51.47 | jeffpc | I'll be right back |
| 04:52.08 | *** join/#asterisk cypromis (chuck-the-@62.212.85.27) |
| 04:52.37 | Sift | but you know on a regular POTS line...even tho you block your caller ID, when you dial 800 numbers...it will still show up |
| 04:52.43 | Sift | is it possible to block that with voip |
| 04:53.14 | *** join/#asterisk jeffpc (~jeffpc@ool-44c218a8.dyn.optonline.net) |
| 04:53.15 | mishehu | there is always ANI |
| 04:53.28 | kb1_kanobe | Sift: if the party you're calling is ISDN and running * then there is always callerID data - aka. ANI. |
| 04:53.49 | mishehu | cid is easy to spoof, ani not. |
| 04:54.16 | Sift | but like what if I call to redeem a coupon somewhere...and it is an 800 number...I dont want my number showing up...so they can spam me later with calls |
| 04:54.27 | kb1_kanobe | not gonna happen. |
| 04:54.40 | kb1_kanobe | hiding the number, not the spam :-) |
| 04:54.42 | mishehu | Sift: use a payphone |
| 04:54.47 | Sift | why are 800 numbers immune |
| 04:55.00 | kb1_kanobe | immune? |
| 04:55.13 | *** join/#asterisk Defraz (~t0tal@sonicwall.dcdi.net) |
| 04:55.17 | Sift | why cant one totally block callerid from 800 numbers |
| 04:55.29 | Sift | how is it that they can still read the ani |
| 04:55.34 | mishehu | Sift: because the owner of the 800 has a right to know whose call he's paying for. |
| 04:55.42 | mishehu | Sift: CID != ANI |
| 04:56.09 | Sift | hmm |
| 04:56.09 | mishehu | CID is something that you can effectively change. ANI is supposed to be provided by the carrier, and unless you're a carrier, you can't change it. |
| 04:56.22 | kb1_kanobe | there are two different things involved. CallerID is what appears on the phone, ANI is internal telco information about the call. If you use ISDN then you get the ANI data as well as the callerID so you just discard the clid and replace with the ANI. |
| 04:56.50 | Sift | ok |
| 04:57.07 | vpp | hmm this is quite insane |
| 04:57.09 | Sift | so every call has ani data |
| 04:57.25 | mishehu | Sift: *SIGH* yes. |
| 04:57.26 | kb1_kanobe | however, there are two parts to callerID also - the name and the number. ANI only provides the number, but the call centers don't really care about the name now do they... |
| 04:57.32 | vpp | asterish-oh323 0.7.1 only supports openH323 1.13.5 |
| 04:57.47 | mishehu | kb1_kanobe: name and number as far as telcos go are only valid in certain regions of the world. |
| 04:57.53 | vpp | but centos 4.0 has gcc 3.4.3 which can't compile openh323 1.13.5 |
| 04:58.01 | vpp | and that 'problem' was fixed in a later openH |
| 04:58.03 | kb1_kanobe | mishehu: true. |
| 04:58.06 | mishehu | US/Canada has CIDName and CIDNumber fields, but Israel only has CIDNumber. |
| 04:58.14 | vpp | which u can't use because asterish-oh323 isnt compatible |
| 04:58.19 | vpp | so now i have to downgrade gcc |
| 04:58.22 | vpp | INSANE! |
| 04:58.44 | mishehu | vpp: run a parallel gcc install, or build it on another machine. |
| 04:58.46 | Sift | mishehu go easy on a newbie :) |
| 04:59.03 | mishehu | Sift: being a newbie does not excuse you from asking redundant questions. |
| 04:59.11 | vpp | mishehu: i'm just installing gcc 2.9.something in usr/local/bin and moving that to the front of my path |
| 04:59.22 | mishehu | ick |
| 04:59.26 | mishehu | gcc 2.9. |
| 04:59.37 | mishehu | vpp: what distro? |
| 04:59.40 | vpp | centos |
| 04:59.44 | mishehu | oh right |
| 04:59.49 | mishehu | it didn't click with me. |
| 04:59.55 | vpp | ahh heheh |
| 04:59.59 | mishehu | that and I've never heard of it |
| 05:00.05 | mishehu | heh |
| 05:00.05 | Ciber | hey guys, can i make asterisk read multiple voice prompts with one line? |
| 05:00.11 | vpp | am i right tho.. there isnt any way to use openh1.17.1 with asterisk? |
| 05:00.19 | Ciber | i tried exten => 2001,2,Background(for-tech-support,vm-press,1) |
| 05:00.19 | vpp | its based on redhat enterprise |
| 05:00.21 | mishehu | Ciber: what exactly do you mean? |
| 05:00.30 | icexx | vpp: had same prob |
| 05:00.32 | Ciber | but commas don't work :P |
| 05:00.34 | icexx | w oh323 |
| 05:00.34 | mishehu | Ciber: you mean you want an IVR menu? |
| 05:00.43 | Ciber | well yeah |
| 05:00.53 | Ciber | but as you can see that's 3 sound files |
| 05:00.54 | mishehu | Ciber: Background() only plays one file. |
| 05:01.01 | Ciber | bah :P |
| 05:01.15 | vpp | icexx: what did u do? |
| 05:01.20 | vpp | cos gcc compile just failed! |
| 05:01.24 | Ciber | so i need to make 3 those lines heh |
| 05:01.37 | vpp | i feel silly now debuging that, when i should be fixing the openh issue directly |
| 05:01.47 | kb1_kanobe | Ciber: you can use Festival to generate prompts in real time. It hasn't got much personality but if you use the ARCTIC_HTS voice it's not too bad. |
| 05:02.02 | mishehu | vpp: would it make you feel any better to know that I can't build samba 3.0.14a on Slamd64 with kernel 2.6.11.7 and gcc 3.4.3? |
| 05:02.04 | mishehu | heh |
| 05:02.12 | Ciber | i'm on a mac :P |
| 05:02.18 | Ciber | don't think we have any festival |
| 05:02.19 | vpp | mishehu: a little lol |
| 05:02.27 | mishehu | kb1_kanobe: does festival have VALLEY_GRL mode? |
| 05:02.27 | mishehu | heh |
| 05:02.59 | mishehu | vpp: oh that and my PCIExpress Radeon X800 has no functional drivers for it... so vesa default driver for me... *ick* |
| 05:03.36 | mishehu | I'd like festival to randomly add "like", "way out", and "you know" randomly during the prompt |
| 05:03.36 | mishehu | heh |
| 05:04.27 | kb1_kanobe | mishehu: unfortunately no, but it can be quite interesting getting the misspellings just right to get the pronounciation to work out... |
| 05:04.57 | *** join/#asterisk ariel_ (~Ariel@dsl-20-177.cofs.net) |
| 05:06.20 | mishehu | kb1_kanobe: hahaha |
| 05:06.29 | mishehu | if only I had the free time that it would take to set that up |
| 05:10.14 | *** join/#asterisk ManxPower (~eric@dsl-209-205-172-111.i-55.com) |
| 05:11.34 | *** join/#asterisk vpp (~noone@host-83-146-50-131.bulldogdsl.com) |
| 05:11.43 | vpp | stupid internet gateway died |
| 05:11.47 | vpp | its all happening today! |
| 05:12.10 | vpp | what did i miss |
| 05:12.15 | vpp | any ideas on what i can do? |
| 05:13.04 | vpp | i probably missed the last few messages |
| 05:13.46 | kb1_kanobe | <mishehu> vpp: would it make you feel any better to know that I can't build samba 3.0.14a on Slamd64 with kernel 2.6.11.7 and gcc 3.4.3? |
| 05:13.46 | kb1_kanobe | <mishehu> heh |
| 05:13.49 | Nugget | while you were gone we had a great discussion about gay marriage, abortion, gun control, and mysql. We were able to come to a pretty solid agreement and then we all went out for ice cream. |
| 05:13.55 | kb1_kanobe | <mishehu> vpp: would it make you feel any better to know that I can't build samba 3.0.14a on Slamd64 with kernel 2.6.11.7 and gcc 3.4.3? |
| 05:13.55 | kb1_kanobe | <mishehu> heh |
| 05:14.06 | kb1_kanobe | ah, sorry about that! |
| 05:15.19 | *** join/#asterisk flot (CCCP@213.152.157.68) |
| 05:15.29 | vpp | lol |
| 05:15.48 | vpp | u didnt get round to talkin about world peace then? |
| 05:15.55 | vpp | i was gone for a whole few minutes u know |
| 05:15.59 | vpp | lazy people u |
| 05:16.01 | vpp | lol |
| 05:16.11 | Nugget | heh |
| 05:16.42 | vpp | so anyone have any ideas, or shall i continue fiddling by my self |
| 05:16.48 | vpp | *with this compile |
| 05:17.01 | vpp | just to clarify lol |
| 05:19.12 | *** join/#asterisk wvbroadband (~dgd@pool-141-153-74-28.clrk.east.verizon.net) |
| 05:19.16 | mishehu | vpp: world peace? you're a dreamer! |
| 05:22.11 | *** join/#asterisk scardinal (~supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 05:22.28 | vpp | ok what gcc should i downgrade to? |
| 05:22.47 | isamar | anybody using Sangoma??? |
| 05:24.12 | vpp | ok gonna try 3.3.4 |
| 05:24.21 | isamar | I am getting "segmentation fault" when I start * with Sangoma A101.... |
| 05:25.28 | *** join/#asterisk shido6 (~greg@d57-87-253.home.cgocable.net) |
| 05:28.07 | *** join/#asterisk los415 (~los415@adsl-69-104-179-191.dsl.pltn13.pacbell.net) |
| 05:30.02 | Ciber | anyone know of a gsm encoder for mac? |
| 05:30.35 | vpp | is there any g729A,B,AB codes for openh232 btw? |
| 05:30.42 | vpp | *codecs |
| 05:31.21 | PCadach | vpp: codecs - yes, but not sure about capabilities... |
| 05:31.54 | vpp | i noticed talk about g729, but not any specifics about A,B or AB |
| 05:34.34 | tainted- | Ciber just use sox |
| 05:34.43 | tainted- | Ciber isn't there a mac port of sox? |
| 05:34.53 | *** join/#asterisk BoRiS (boris@S01060040ca1e5b54.wp.shawcable.net) |
| 05:34.55 | Qwell | heh, macs don't use mixers |
| 05:34.59 | Qwell | or, something |
| 05:35.07 | Qwell | one sec, lemme get the exact quote |
| 05:35.53 | Qwell | 2005-03-09 23:24:11 <trey_> cls http://mac.softpedia.com/get/Audio/SoX%20Wrap.shtml |
| 05:35.53 | Qwell | 2005-03-10 00:00:08 <cls> but that doesnt fix the fact that MAC OS X (not sox) has no /dev/audio device |
| 05:35.54 | Qwell | there |
| 05:36.10 | tainted- | lol what |
| 05:36.17 | tainted- | then how do macs do audio |
| 05:36.27 | Qwell | umm, lemme see if I can find the answer, heh |
| 05:36.57 | twisted | they produce electromagnetic waves from the hard drive that eminate through the speakers. |
| 05:37.21 | tainted- | no wonder girls like macs so much |
| 05:37.31 | tainted- | ~brrrrrrrrrr~ |
| 05:37.37 | Qwell | ahh, the answer was to convert to mp3, heh |
| 05:37.39 | tainted- | just listening to some music honey |
| 05:37.41 | icexx | tainted |
| 05:37.42 | Qwell | using lame |
| 05:37.43 | tainted- | ~brrrrrrrrrr~ |
| 05:37.45 | icexx | you live here ;) |
| 05:37.53 | icexx | any time of the day I come , you are here ;) |
| 05:38.11 | Ciber | googling around and still can't find a damn converter |
| 05:38.15 | twisted | it takes two to tango. |
| 05:38.16 | tainted- | where else can u hang out with the net's brighest |
| 05:38.18 | PCadach | vpp: H323 have individual capabilities per each codec and I'm not sure it would happy if you announces G.729AB while remote party wants just G.729... |
| 05:38.55 | tainted- | iCExx what is iconnecthere |
| 05:39.01 | icexx | deltathree |
| 05:39.06 | icexx | vob provider |
| 05:39.17 | vpp | PCadach: yeah thats the thing.. with my current setup i'm having problems with speech path |
| 05:39.19 | tainted- | sip or iax |
| 05:39.21 | icexx | sip |
| 05:39.28 | tainted- | did u fix your problems? |
| 05:39.31 | *** join/#asterisk JerJer[mobile] (~nonyobizn@166.205.56.92) |
| 05:39.55 | icexx | nope ;) it doesn't work w asterisk. gives some error in the mid of the conversation ;) probably something with codecs |
| 05:40.27 | tainted- | u sure it doesn't work with asterisk? or u have some kind of config issue |
| 05:41.05 | *** join/#asterisk r0d3nt (~RatMan@wsip-24-234-241-84.lv.lv.cox.net) |
| 05:41.13 | icexx | i am not sure, but listen, it connects me to the other party, and 2 seconds into the conversation, a lady voice from deltathree tells me call cannot be completed as dialed, error 480 and drops the call |
| 05:41.14 | icexx | ;) |
| 05:41.23 | icexx | w/o asterisk works fine |
| 05:41.55 | tainted- | i think 480 is a server related error |
| 05:41.58 | tainted- | 4xx |
| 05:42.07 | icexx | ye, but not from * side |
| 05:42.16 | icexx | from their side, but why, i can't understand |
| 05:43.06 | tainted- | which codec are u using |
| 05:43.08 | tainted- | gsm? |
| 05:43.14 | icexx | ye |
| 05:43.24 | icexx | you want me to show you sip debug out? |
| 05:43.37 | tainted- | http://lists.digium.com/pipermail/asterisk-users/2003-March/009502.html |
| 05:43.42 | tainted- | did u read that thread? |
| 05:43.49 | tainted- | i think they fixed it when they switched to 711 |
| 05:44.46 | tainted- | there's also a sample config.. |
| 05:44.56 | icexx | lemmi see, didn't notice this one |
| 05:45.10 | icexx | yea, seen this one |
| 05:45.26 | tainted- | did u try 711 ulaw? |
| 05:46.15 | icexx | nope ;)) let me try |
| 05:46.33 | icexx | disallow gsm ? |
| 05:46.47 | tainted- | or put it below allow = ulaw |
| 05:47.25 | icexx | ok |
| 05:47.54 | tainted- | wow OS X tiger looks amazing |
| 05:48.07 | JerJer[mobile] | tainted-: will it run on x86? |
| 05:48.17 | vpp | arrggghh gcc wont build |
| 05:48.34 | tainted- | osx on x86? not likely |
| 05:48.50 | JerJer[mobile] | then what good is it? |
| 05:49.10 | tainted- | i'm not an apple fan by far |
| 05:49.16 | tainted- | but the gui is amazing |
| 05:49.30 | tainted- | they are king of usability |
| 05:51.25 | Qwell | sweet |
| 05:51.35 | Qwell | You ask a friend for a google pen, you end up with a google shirt |
| 05:56.37 | JerJer[mobile] | hell i own goog ipo and I didn't get a pen or shirt |
| 05:56.39 | JerJer[mobile] | bastards |
| 05:57.15 | JerJer[mobile] | just lame ass proxy vote scantron sheets to approve some director morons to which I have no clue who they are |
| 05:58.56 | tainted- | how is google ipo doing |
| 05:59.05 | tainted- | s/ipo/stock/ |
| 05:59.37 | Qwell | JerJer[mobile]: odd, I work for a company, and thats all I get |
| 06:00.05 | JerJer[mobile] | tainted-: i bought in at 85 and last i looked it was like 219 |
| 06:00.52 | tainted- | what timespan |
| 06:01.05 | tainted- | since ipo? |
| 06:01.35 | JerJer[mobile] | yes ipo was 85 |
| 06:01.35 | Qwell | like Aug 04 |
| 06:02.20 | tainted- | crazy |
| 06:05.52 | *** join/#asterisk Rick_Hunter (~rhunter@04-089.008.popsite.net) |
| 06:06.52 | *** join/#asterisk aminorex (~tony@c66.191.69.132.dul.mn.charter.com) |
| 06:13.23 | *** join/#asterisk gres (~serg@81.222.48.242) |
| 06:14.28 | *** join/#asterisk ISMe (me@218.111.156.211) |
| 06:14.32 | ISMe | hi all |
| 06:15.22 | ISMe | i plan to have 3 X SIPURA 2000 behind the same router to connect to *, any recommendation ? |
| 06:15.55 | ISMe | can i use port 5060 - 5065 ? |
| 06:23.20 | *** join/#asterisk lehel (~lehel@82.79.20.17) |
| 06:23.46 | lehel | good whatever u want.. |
| 06:24.07 | lehel | hi tzafrir, ? r u 2day? |
| 06:24.42 | lehel | 'tsup jakepdev |
| 06:28.16 | lehel | everybody is still sleeping maybe |
| 06:30.14 | Qwell | or still reeling from trying to read that |
| 06:30.41 | icexx | tained: have same prob |
| 06:31.00 | tainted- | icexx complain to them |
| 06:31.21 | tainted- | icexx i'm sure ur not first to run into 4xx error w/ asterisk |
| 06:31.33 | tainted- | icexx if they don't respond, change provider |
| 06:31.39 | icexx | ye |
| 06:31.59 | Ciber | four sipuras? isn't that going to get messy? :P |
| 06:32.04 | icexx | m=audio 6550 RTP/AVP 0 96 |
| 06:32.05 | icexx | a=rtpmap:0 pcmu/8000 |
| 06:32.05 | icexx | a=rtpmap:96 telephone-event/8000 |
| 06:32.05 | icexx | a=fmtp:96 0-15 |
| 06:32.05 | icexx | a=ptime:20 |
| 06:32.12 | icexx | 711 right? |
| 06:32.13 | icexx | pcmu? |
| 06:32.34 | tainted- | a=rtpmap:0 pcmu/8000 |
| 06:33.00 | ISMe | Ciber: any suggestion ? |
| 06:33.21 | Ciber | umm |
| 06:33.24 | Ciber | it won't matter? |
| 06:33.39 | tainted- | icexx their live chat is open |
| 06:33.45 | Ciber | the boxes will be in your lan |
| 06:33.51 | Ciber | so using the same port won't matter |
| 06:33.59 | Ciber | since they'll have different private addresses |
| 06:34.25 | Qwell | unless the * box is outside the LAN |
| 06:34.33 | Ciber | yeah that would be a problem |
| 06:34.34 | Qwell | he didn't really specify |
| 06:34.39 | ISMe | * box is outside the LAN |
| 06:34.45 | Qwell | :p |
| 06:34.47 | Ciber | oh |
| 06:34.48 | Ciber | ummm |
| 06:34.50 | Ciber | lol |
| 06:34.54 | ISMe | sipura and * is like 500 miles away |
| 06:35.01 | Ciber | depends how much your router sucks i guess |
| 06:35.02 | ISMe | i mean apart |
| 06:35.33 | Ciber | i remember being able to do like 5060+1 or something to map the same port to different ip's or something on a router i had |
| 06:35.38 | Ciber | but my memory sucks :P |
| 06:35.41 | ISMe | my dlink router can only open 5060 for 1 IP address |
| 06:36.02 | ISMe | sipura uses 5060 and 5061, no problem if 1 sipura is use |
| 06:36.27 | Ciber | can't you setup another * box to manage the 4 sipuras? |
| 06:37.01 | Qwell | That would be ideal, but... |
| 06:37.13 | ISMe | errrr to setup a new * would not be cost effective. server+TDM |
| 06:37.15 | Qwell | but... |
| 06:37.23 | Qwell | as long as the * server is setup right, NAT shouldn't be an issue |
| 06:37.44 | Ciber | not cost effective? |
| 06:37.51 | ISMe | Qwell: we could use IAX to connect 2 boxes |
| 06:38.00 | *** join/#asterisk r0d3nt (~RatMan@wsip-24-234-241-84.lv.lv.cox.net) |
| 06:38.02 | Ciber | get a junk box from salvation army |
| 06:38.05 | Ciber | like 20 dollars |
| 06:38.07 | Qwell | no, I mean, without a second * box |
| 06:38.17 | Ciber | and a 10 dollar 4 port switch |
| 06:38.19 | Ciber | bam |
| 06:38.32 | Qwell | if the sipuras are registering right, and the remote * box is setup right, NAT shouldn't be an issue |
| 06:38.45 | Ciber | that too |
| 06:39.06 | Ciber | is he using sip? |
| 06:39.11 | Ciber | can just use iax :P |
| 06:39.12 | Qwell | sipura... |
| 06:39.17 | Qwell | kinda implies sip |
| 06:39.26 | Ciber | those don't support iax? |
| 06:39.36 | Qwell | no |
| 06:39.46 | Ciber | poor guy :P |
| 06:42.14 | Qwell | hmm |
| 06:42.15 | Qwell | ~nat |
| 06:42.16 | jbot | [nat] Network Address Translation Usable in Asterisk sip.conf file with externip, localnet, and localmask setup properly. See docs. |
| 06:42.42 | Qwell | ~natpeer |
| 06:44.16 | ISMe | so guys, i cant use 5062 and so on for my 2nd sipura then |
| 06:48.13 | Qwell | Shouldn't need to |
| 06:48.53 | Qwell | <file> nat=yes, qualify=yes, canreinvite=no |
| 06:48.53 | Qwell | <file> for peers behind NAT |
| 06:50.37 | ISMe | Qwell: then what port on the router should i open for my 2nd sipura ? |
| 06:56.26 | lehel | can I apt-get from somewhere an Asterisk Graphical Interface? |
| 06:57.28 | *** join/#asterisk Gunnar (~gunnar@34.80-203-209.nextgentel.com) |
| 06:57.53 | *** join/#asterisk RestLessGemini (~umairbari@202.142.189.86) |
| 06:59.17 | ISMe | lehel: http://amp.coalescentsystems.ca/ |
| 07:01.35 | *** join/#asterisk tsipl (~ra@pilot.generation-p.com) |
| 07:02.24 | lehel | 10x ISMe |
| 07:03.55 | *** join/#asterisk Blackvel (~blackvel@dsl-213-023-034-179.arcor-ip.net) |
| 07:04.35 | Blackvel | morning. is it possible to use the Manager API and Originate to dail outbound without calling the "agent" first? |
| 07:04.47 | Blackvel | only connect to the agent laters, when the customer is connected? |
| 07:06.51 | JerJer[mobile] | Blackvel: eh? |
| 07:07.19 | JerJer[mobile] | just orignate it into a context and exten that drops the call into a queue |
| 07:11.37 | rvhi | is there any application to list all available variables? |
| 07:12.18 | *** join/#asterisk shido6 (~shido@d57-87-253.home.cgocable.net) |
| 07:14.22 | Qwell | rvhi: for asterisk? |
| 07:14.32 | Qwell | README.variables |
| 07:19.21 | *** join/#asterisk JerJer[mobile] (~nonyobizn@166.205.60.108) |
| 07:20.04 | *** join/#asterisk scanna (~scannachi@81-174-16-211.f5.ngi.it) |
| 07:21.09 | *** join/#asterisk PTG1234 (PTG123@ip68-106-24-139.ph.ph.cox.net) |
| 07:24.54 | *** join/#asterisk fenlander_ (~neils@82.152.81.57) |
| 07:26.07 | *** join/#asterisk chaoscon (~ph33r@chaoscon.user) |
| 07:27.48 | rvhi | Qwell, i want to see the value of all variables in the dialplan |
| 07:29.01 | *** join/#asterisk n4y (~frodo7@host-ip226-209.crowley.pl) |
| 07:30.35 | Blackvel | JerJer[mobile]: into a queue? but I want to do outbound dialing, not inbound agent routing. Also the J2EE application selects the agent, not Asterisk with queueing itself |
| 07:31.26 | Blackvel | it is not possible to "only" dail an outbound party and on connection join it with an REDIRECT to an agent? |
| 07:32.04 | JerJer[mobile] | fuck j2ee |
| 07:32.08 | JerJer[mobile] | are you crazy? |
| 07:32.37 | Blackvel | that is the spec |
| 07:32.41 | Blackvel | I can not do anything against it |
| 07:32.42 | Blackvel | ;) |
| 07:33.01 | JerJer[mobile] | sure you can - simply don't do that project |
| 07:33.17 | JerJer[mobile] | tell them j2ee is not the answer and to come back to you when they find a clue |
| 07:33.24 | Blackvel | but according to Manager API docs on WIKi, originate can not call one party without calling the agent first |
| 07:33.28 | Blackvel | I wont |
| 07:33.31 | Blackvel | I dont care |
| 07:33.40 | Blackvel | as long as I get this code finished |
| 07:33.41 | JerJer[mobile] | agent? |
| 07:33.42 | JerJer[mobile] | what |
| 07:33.50 | JerJer[mobile] | you can orignate any call you want |
| 07:34.09 | JerJer[mobile] | there is nothing saying you have to use an agent |
| 07:34.16 | *** join/#asterisk tainted- (~identd@65-60-70-243-cust.telepacific.net) |
| 07:34.22 | Blackvel | funny English |
| 07:34.23 | Blackvel | no |
| 07:34.33 | JerJer[mobile] | the wiki needs to die - its full of useless bullshit that people call docuemention |
| 07:34.34 | Blackvel | the system should call an outbound party |
| 07:35.00 | Blackvel | without calling an company telephone first |
| 07:35.05 | *** join/#asterisk Jas_Williams (~jason@host217-43-100-176.range217-43.btcentralplus.com) |
| 07:35.06 | Blackvel | call it agent or whatever |
| 07:35.07 | JerJer[mobile] | then do that |
| 07:35.18 | Blackvel | seems not working |
| 07:35.34 | Blackvel | I have to provide Action: Originate with Channel: + exten |
| 07:35.44 | JerJer[mobile] | yes |
| 07:35.51 | Blackvel | once the company telephone is picked up, originate calls the outbound |
| 07:35.55 | JerJer[mobile] | that is precicely how it works |
| 07:36.04 | Blackvel | I can not leave out exten |
| 07:36.10 | JerJer[mobile] | then you are using the oringate backwads |
| 07:36.13 | Blackvel | and act as an application DIAL command |
| 07:36.14 | JerJer[mobile] | backwards |
| 07:36.26 | Blackvel | you mean I could do this? |
| 07:36.31 | JerJer[mobile] | Channel: Zap/g1/12345 |
| 07:36.44 | Blackvel | but only at a successful connection it is clear what company telephone I can dail |
| 07:36.49 | Blackvel | not when I start the Originate Action |
| 07:36.55 | Blackvel | weird |
| 07:36.57 | JerJer[mobile] | that will cause a call to go out to zap group 1 |
| 07:36.59 | Blackvel | btw, tried that zap thing |
| 07:37.02 | Blackvel | had no success |
| 07:37.15 | JerJer[mobile] | then when it connects it will drop into whateer context and exten you define |
| 07:37.25 | Blackvel | DIAL(ZAP/g1/1234) works in dialplan |
| 07:37.46 | JerJer[mobile] | and it does work in an action: originate command |
| 07:37.47 | JerJer[mobile] | i do it all the time |
| 07:38.05 | Blackvel | I use zaphfc/bristuff |
| 07:38.14 | Blackvel | maybe it does not support this? Could that be? |
| 07:38.14 | JerJer[mobile] | can't help you there |
| 07:38.22 | JerJer[mobile] | i do it on TE410P cards all day long |
| 07:38.27 | Blackvel | can I leave out exten in an originate? |
| 07:38.33 | JerJer[mobile] | no |
| 07:38.39 | JerJer[mobile] | that's not how it works |
| 07:38.51 | omadon | Can anybody help me with zaphfc problem and i4l question |
| 07:38.54 | Blackvel | what if I do not want to define the exten in the dialplan? |
| 07:39.00 | omadon | hello |
| 07:39.02 | JerJer[mobile] | omadon: only if you ask a specific question |
| 07:39.12 | JerJer[mobile] | Blackvel: you have to - that's how asterisk works |
| 07:39.14 | Blackvel | I want to use the originate as an dial, give the technology, ressource + extension directly |
| 07:39.20 | JerJer[mobile] | or you can dump the call into a specifc application |
| 07:39.30 | JerJer[mobile] | THEN DO EXACTLY THAT |
| 07:39.35 | Blackvel | how? |
| 07:39.42 | Blackvel | if I have to provide exten |
| 07:39.44 | Blackvel | it does not work |
| 07:39.53 | Blackvel | or am I wrong? |
| 07:39.54 | JerJer[mobile] | channel: Technology/resource/exten |
| 07:40.03 | Blackvel | right |
| 07:40.05 | JerJer[mobile] | you have to either provide a context and exten and priority |
| 07:40.06 | Blackvel | and for the 2nd |
| 07:40.06 | Blackvel | ? |
| 07:40.07 | JerJer[mobile] | OR |
| 07:40.16 | JerJer[mobile] | an application |
| 07:40.55 | Blackvel | sorry, didn't try originate with application: yet |
| 07:41.05 | Blackvel | I could pass dial(ZAP/g1/1234)? |
| 07:41.18 | Blackvel | application: is a command from show applications, e.g DIAL? |
| 07:41.19 | JerJer[mobile] | maybe, but why? |
| 07:41.36 | Blackvel | because there will be nothing in the dialplan |
| 07:41.56 | JerJer[mobile] | the same exact thing is accompolsihed by using a context and exten that does that same exact thing |
| 07:42.10 | Blackvel | right ok |
| 07:42.11 | JerJer[mobile] | hey a circular sentance |
| 07:42.25 | JerJer[mobile] | you can do the same exact thing by usng a context and exten |
| 07:42.52 | JerJer[mobile] | unless your crazy app has two totally random numbers being called, which you need to bridge |
| 07:43.19 | Blackvel | as I got told, the "agent"/company phone is unknown (2nd part of Originate) |
| 07:43.36 | Qwell | rvhi: yes, README.variables |
| 07:43.36 | Blackvel | only on connect it will be selected and then some bridge should happen |
| 07:43.37 | JerJer[mobile] | and i don't know how u pass arguments using the Application keyword on an action: origniate |
| 07:44.05 | JerJer[mobile] | Blackvel: it has to be known - else how you going to call it? |
| 07:44.22 | JerJer[mobile] | smells like your spec is broken |
| 07:44.55 | Blackvel | show manager originate does not tell too much, true. show application dial tells more. Manager API should be updated ;) |
| 07:45.07 | Blackvel | the outbound caller is known, not the company phone |
| 07:45.14 | Blackvel | only on connect it should be dynamically selected |
| 07:45.28 | Blackvel | looks like I have to tell throw away spec and do something else |
| 07:45.29 | Blackvel | hehe |
| 07:45.53 | Blackvel | that is the point |
| 07:45.58 | JerJer[mobile] | yeah and get rid of j2ee while your at it |
| 07:46.03 | Blackvel | originate calls 2 parties at the same time |
| 07:46.12 | JerJer[mobile] | it has to be known - else how you going to know who to call? |
| 07:46.17 | Blackvel | that "might" be a problem |
| 07:46.24 | *** join/#asterisk Delvar (~irc@83.146.53.34) |
| 07:46.25 | McUnixJr | is there a way to do party line calls on * ? |
| 07:46.33 | JerJer[mobile] | or you have to define how to dynamically select the 'agent' |
| 07:46.48 | JerJer[mobile] | McUnixJr: conference calls? certainly |
| 07:47.05 | McUnixJr | is there a limit on the number of attendees? |
| 07:47.05 | Blackvel | passing one originate and redirect afterwards would be the solution |
| 07:47.25 | JerJer[mobile] | but you have to have source and destinaton channels |
| 07:47.33 | Blackvel | jerjer: yes, that is defined, but not before calling the originate. makes no sense with predictive dailer |
| 07:47.44 | JerJer[mobile] | oh god |
| 07:47.49 | JerJer[mobile] | good luck |
| 07:47.54 | Blackvel | haha |
| 07:47.55 | Blackvel | I see |
| 07:47.59 | JerJer[mobile] | 'm not helping spammers |
| 07:48.06 | Blackvel | I am no spammer |
| 07:48.09 | Blackvel | only a programmer |
| 07:48.14 | JerJer[mobile] | you are helping them |
| 07:48.19 | JerJer[mobile] | so you are equal |
| 07:49.07 | McUnixJr | JerJer[mobile], so it is possible to have a party line functionality using conference call, is there a limit to the number of attendees? |
| 07:49.15 | Blackvel | current part i am programming on (which also uses manager api for dial/hangup) is not. so I have to know manager api yet. dunno about the 2nd project part. |
| 07:49.44 | Blackvel | makes no difference for me to call firstly the agent, then the outbound party |
| 07:49.48 | Blackvel | spamming is spamming |
| 07:49.51 | Blackvel | you can abuse anything |
| 07:50.26 | Blackvel | i do not know anything about that company and what it does, what it is using use. 3 around corner project. I do not know the final client |
| 07:50.29 | Blackvel | so I dont care |
| 07:50.34 | Blackvel | need something for my CV |
| 07:50.45 | Blackvel | ;) |
| 07:51.28 | JerJer[mobile] | McUnixJr: Asterisk doesn't care - so no there is no limit |
| 07:51.39 | JerJer[mobile] | but then there are bandwidth or phone line issues |
| 07:53.12 | Blackvel | hm |
| 07:53.29 | JerJer[mobile] | Blackvel: your spec is broken |
| 07:53.32 | Blackvel | how to make it working to block some events in manager api? i am not interested in registry and peer connected things |
| 07:53.47 | JerJer[mobile] | when you place a phone call there is always a source and a destination |
| 07:54.00 | Blackvel | JerJer[mobile]: maybe for the 2nd part of the project yes. but I need to find this out. not for the 1st part (inbound caller + AGI) |
| 07:54.00 | JerJer[mobile] | the source can be a playback of a prompt or another channel |
| 07:54.01 | JerJer[mobile] | something |
| 07:54.33 | Blackvel | maybe MOH is the solution |
| 07:54.37 | Blackvel | haha |
| 07:54.48 | Blackvel | I am only the pure programmer, please don't judge me |
| 07:54.59 | JerJer[mobile] | so your saying my home telephone is going to ring and all i'm going to hear is MOH? |
| 07:55.06 | JerJer[mobile] | that's gonna generate a lot of revenue |
| 07:55.20 | JerJer[mobile] | Blackvel: you still have the option to tell them to go piss off |
| 07:55.23 | JerJer[mobile] | as the programmer |
| 07:55.38 | Blackvel | dunno, maybe I need to tell, as you said, select the damn agent first, throw away the spec and dail then (2 channels) |
| 07:55.39 | Blackvel | I will see |
| 07:55.40 | JerJer[mobile] | i have told many phone spammers what's up |
| 07:55.55 | Blackvel | anyways, I need to find out more about this manager api :) |
| 07:55.57 | newl | JerJer[mobile]: So by your method of thinking, Ford Motor Company is equal to someone who uses their vehicle to kill someone for making said vechicle. |
| 07:56.15 | JerJer[mobile] | no |
| 07:56.17 | Qwell | newl: No, it would be like being an engineer |
| 07:56.23 | Qwell | and being told to make brakes that don't work |
| 07:56.23 | JerJer[mobile] | ford doesn't make you buy a car |
| 07:56.25 | Qwell | and doing so |
| 07:56.35 | JerJer[mobile] | or make you drink and drive |
| 07:57.02 | Blackvel | I really do not know what this system is doing for the 2nd part. the 1st part is inbound. so fine with me. and for dial/hangup/play MOH I need MAPI anyways, and manager api spec/docs drives me crazy anyways |
| 07:57.08 | *** join/#asterisk Sander4000 (~sanderrar@dslam228-48-166-62.adsl.versatel.nl) |
| 07:57.21 | JerJer[mobile] | Blackvel: mapi ? |
| 07:57.25 | Sander4000 | Hello there |
| 07:57.27 | JerJer[mobile] | not gonna happen |
| 07:57.28 | Blackvel | doc like Application: Application to use is useless |
| 07:57.30 | newl | Qwell: Or the alternative would be that the engineer was hired to do the job of designing the car that worked for the company to use and/or sell as they saw fit. |
| 07:57.36 | Blackvel | manager api = mapi :) |
| 07:57.40 | McUnixJr | JerJer[mobile], Thanks for the information ! |
| 07:57.52 | JerJer[mobile] | Blackvel: um ok |
| 07:57.55 | McUnixJr | next question - what provider is best at the moment? |
| 07:58.04 | JerJer[mobile] | define best |
| 07:58.05 | Qwell | newl: No matter how many people the cars (which were purposely designed to) killed |
| 07:58.14 | Blackvel | hm |
| 07:58.19 | Sander4000 | can anyone tell me what the best linux distro is for setting up the bristuff driver? |
| 07:58.25 | Blackvel | didn't want to break up a ground discussion in this chan |
| 07:58.33 | Blackvel | only about manager api |
| 07:58.41 | Blackvel | so guys, forget what I told |
| 07:58.42 | Blackvel | :) |
| 07:58.51 | newl | Sander4000: Any Linux distro should suit your needs if you'd research. |
| 07:58.59 | newl | Even LFS. |
| 07:59.40 | JerJer[mobile] | espcally LFS |
| 07:59.43 | Sander4000 | hmm i'm trying linux FC2 but i get all sorts of error during installation of bristuff |
| 07:59.55 | newl | Qwell: Would it be any different if we changed the car to a missile that was designed and built for (presume for this example) defense purposes? :) |
| 07:59.56 | JerJer[mobile] | then find the author |
| 08:00.06 | JerJer[mobile] | provded you have read all of the relvant included documenation |
| 08:00.09 | newl | or build from source |
| 08:00.47 | JerJer[mobile] | Sander4000: download slackware or debian - FC2 is a joke |
| 08:01.11 | Qwell | newl: meh, if you're gonna keep using bad analogies, I'm done |
| 08:01.47 | Qwell | Creating X that Y told you to create, even though X is immoral. |
| 08:02.12 | Qwell | s/,/, and doing so,/ |
| 08:02.25 | newl | Qwell: heh okay. |
| 08:02.43 | newl | I just didn't agree with the original "is equal" statement. |
| 08:03.01 | JerJer[mobile] | newl: then say that |
| 08:03.12 | Sander4000 | and denian works better? i have no problem installing tdm400 but i have to get tdm400 with 2 fxo, one te110p, and a quadbri card |
| 08:03.31 | JerJer[mobile] | it is my opinon if you purposely take on a job that supports the scum of the earth, you are the same as well |
| 08:03.59 | JerJer[mobile] | Sander4000: its just a distro - it is the same basic linux kernel |
| 08:04.00 | Qwell | Like working for a politician. heh |
| 08:04.04 | *** join/#asterisk Zeeek (~icechat5@Zeeek.sustaining.supporter.pdpc) |
| 08:04.07 | *** join/#asterisk jetdotnet (~jetdotnet@adsl-64-219-216-41.dsl.hstntx.swbell.net) |
| 08:04.41 | newl | JerJer[mobile]: Fair enough. I believe that everyone is entitled to their opinion. :) |
| 08:05.04 | JerJer[mobile] | i have fired four or five morons doing predictive dialing on our network |
| 08:05.12 | Sander4000 | hm i'll check it out then, configuring asterisk is no problem for me, but linux is a bit new to me |
| 08:05.16 | newl | politics..I usually avoid those conversations. They're almost as bad as distribution and db wars hehe |
| 08:05.24 | JerJer[mobile] | Sander4000: then check out mandrake |
| 08:05.32 | JerJer[mobile] | Sander4000: and do not run X windows |
| 08:05.48 | Qwell | mandrake without X is like... |
| 08:05.53 | Sander4000 | no just a clean install :) |
| 08:06.01 | newl | Sander4000: or give Linux From Scratch a whirl, you'd learn something. |
| 08:06.09 | JerJer[mobile] | Qwell: mandrake doesn't lke to run without x? |
| 08:06.14 | newl | Qwell: any other Linux? :) |
| 08:06.18 | Qwell | JerJer[mobile]: well, most of the tools are GUI, aren't they? |
| 08:06.30 | Qwell | the mandrake specific tools that is |
| 08:06.35 | JerJer[mobile] | no clue - i've just been told it is a newbie friendly distro |
| 08:06.42 | Qwell | JerJer[mobile]: Its great...with X. ;/ |
| 08:06.49 | JerJer[mobile] | ahh |
| 08:06.54 | newl | nah, all qui tools have a console equiv. |
| 08:06.55 | Qwell | otherwise, you might as well be using anything else |
| 08:06.57 | JerJer[mobile] | and astersk hates X - at least with framebuffers |
| 08:07.25 | JerJer[mobile] | ever since redhat fucked everyone over - i've started rolling my own distro |
| 08:07.27 | newl | sans the package management tool, though with urpmi, it really isn't needed. |
| 08:07.34 | Sander4000 | hmm linux from scratch is a little to much for me i have to get it up and running in 2 weeks |
| 08:07.47 | Juggie | use centos |
| 08:07.49 | Qwell | JerJer[mobile]: yeah...kinda turned me off to RH. I went to Gentoo, and haven't looked back |
| 08:08.16 | Qwell | I was fairly faithful to RH from 6.1 through FC1, and got fed up |
| 08:08.31 | Juggie | Qwell, centos |
| 08:08.45 | Qwell | meh |
| 08:09.00 | Qwell | I like Gentoo much more then I ever liked RH |
| 08:09.42 | newl | I started out on Slackware back in the pre 1 days, moved to RH in the 4.x days, Mandrake at 7.0, LFS for a few months, back to Mandrake. Recently tried FC3 full install, didn't like it still and went back to Mandrake. Have thought of Gentoo a few times but the thought of compiling all the time when I can actually be using my machine never thrilled me much. |
| 08:10.15 | newl | In the end, any distribution is what you make it. The primary difference are the tools they provide and support (if any). |
| 08:10.23 | Qwell | newl: I spend about one night a week (while I'm sleeping) compiling stuff |
| 08:10.33 | Qwell | and if I want something new, I guess |
| 08:11.00 | newl | Qwell: If I didn't like living on the bleeding edge I could probably get away with that too. |
| 08:11.26 | newl | Oddly enough though, I haven't had anything substantial on Cooker break in quite awhile. |
| 08:11.27 | Qwell | Do it every night then. It's the same amount of compiling, and can be done at night regardless |
| 08:12.13 | newl | might look into it again..tax time is coming up and I've been wanting to get another machine to fart around on. |
| 08:14.23 | Qwell | coming up? |
| 08:14.28 | Qwell | its come and gone man :p |
| 08:15.47 | newl | I don't pay the IRS any longer :) |
| 08:16.07 | Qwell | ahh |
| 08:17.21 | *** join/#asterisk RoyK (~roy@143.80-202-166.nextgentel.com) |
| 08:19.32 | jetdotnet | debian |
| 08:21.54 | *** join/#asterisk ramtha (~tk@gw.01063telecom.de) |
| 08:22.23 | ramtha | hi, how can i change the cid signaling from "network providet" to "past" ? |
| 08:22.47 | *** join/#asterisk masonc (~lists@206.48.59.5) |
| 08:24.32 | *** join/#asterisk christo (~chris@office.enovi.com) |
| 08:26.53 | christo | morning all |
| 08:27.23 | masonc | morning teacher |
| 08:27.50 | christo | :) |
| 08:28.32 | masonc | how are you |
| 08:31.53 | *** join/#asterisk shido6 (~shido@d57-87-253.home.cgocable.net) |
| 08:31.56 | masonc | anyone know a lot about rtp and nat |
| 08:31.59 | christo | Fine thank you class, now please turn to page 37. |
| 08:32.24 | masonc | :-) |
| 08:34.42 | *** part/#asterisk nitram (nitram@superblob.com) |
| 08:36.39 | Zeeek | masonic ask and you shall see |
| 08:44.42 | masonc | I'm having problems with Sipura phones on a cable modem, asterisk server on another cable modem |
| 08:45.06 | masonc | two phones work fine, one has one sided conversation |
| 08:45.23 | masonc | only difference is it is on a different ip class |
| 08:45.30 | masonc | but there should be no firewalling |
| 08:45.43 | Jas_Williams | masonc it it nat ? |
| 08:45.48 | masonc | no |
| 08:45.51 | *** join/#asterisk udppacket (~tcpip@73.155.39-62.rev.gaoland.net) |
| 08:45.52 | masonc | all real IPs |
| 08:45.57 | Zeeek | you asked for nat and rtp? |
| 08:46.10 | masonc | well, yes, I have two questions |
| 08:46.14 | Zeeek | oh. |
| 08:46.28 | Zeeek | are the phones on the same subnet |
| 08:46.41 | masonc | no |
| 08:46.48 | Zeeek | there are total two phones? |
| 08:46.49 | masonc | but the networks can see each other |
| 08:46.54 | Zeeek | can you switch them? |
| 08:46.58 | masonc | three phones |
| 08:47.02 | ramtha | masone: du you have the same problems if you activate canreinvite=no? |
| 08:47.05 | Zeeek | two work, one not? |
| 08:47.06 | *** join/#asterisk meppl (mephisto@pD9E686AE.dip.t-dialin.net) |
| 08:47.23 | masonc | yes, two phones working, one registers but has one sided voice |
| 08:47.35 | masonc | I cannot control the IP, it is dhcp from the ISP |
| 08:47.36 | meppl | guten morgen |
| 08:47.41 | Zeeek | and of the =two that wrok, are they together on subnet? |
| 08:47.46 | masonc | yes |
| 08:47.52 | Zeeek | that can be a proble |
| 08:47.53 | masonc | same C class |
| 08:47.54 | Zeeek | m |
| 08:48.06 | masonc | why is it a problem |
| 08:48.22 | Zeeek | I don't know, I've had problems with IAX even |
| 08:48.39 | Zeeek | several clients on same sub |
| 08:48.40 | masonc | routers shouldn't block rtp should they? |
| 08:48.58 | Zeeek | maybe you could try setting the phones to different ranges? |
| 08:49.02 | Zeeek | for grins |
| 08:49.07 | masonc | ranges? |
| 08:49.16 | masonc | you mean rtp ranges? |
| 08:49.23 | ramtha | hi, how can i change the caller id signaling from "network providet" to "pass" ? |
| 08:49.28 | Zeeek | ya you said sipura, I assume you cangive a strating port? |
| 08:49.35 | masonc | yes |
| 08:49.41 | masonc | RTP Port Min: |
| 08:49.45 | Zeeek | just for the heck of it, set one different |
| 08:49.47 | masonc | 16384 |
| 08:49.49 | masonc | ok |
| 08:49.57 | Zeeek | just a wild hair :) |
| 08:50.04 | masonc | I have another problem, very similar |
| 08:50.16 | Zeeek | discount for multiple questions |
| 08:50.19 | Zeeek | go for it |
| 08:50.19 | masonc | phones and asterisk, inside linksys router |
| 08:50.21 | masonc | :-) |
| 08:50.25 | newl | ramtha: SetCIDName(pass) |
| 08:50.33 | Zeeek | ya, I have the exact situation on both ends right nopw |
| 08:50.35 | masonc | put the pbx on the dmz |
| 08:50.45 | Zeeek | I didn't do that |
| 08:50.46 | masonc | tried to connect with phone outside the router |
| 08:50.52 | masonc | one sided voice |
| 08:50.56 | Zeeek | I've not had luck using DMZ and SIP ever |
| 08:51.03 | masonc | would the router block stuff? |
| 08:51.06 | Zeeek | depends on the router I guess |
| 08:51.20 | Zeeek | I don't know but I use port forwarding on mine and it works |
| 08:51.35 | masonc | port forward to? |
| 08:51.48 | Zeeek | just a sec |
| 08:52.10 | Zeeek | I had a URL but lost it. |
| 08:52.34 | masonc | what is outside your router? |
| 08:52.54 | Zeeek | let me try to explain in concise terms |
| 08:53.11 | Zeeek | asterisk -> Linksys WAG54g -->DSL |
| 08:53.16 | *** join/#asterisk tafazzi (~Dario@eporroj0-1.customer.nettuno.it) |
| 08:53.23 | masonc | but what device did you have outside |
| 08:53.39 | Zeeek | ports 5060, 4569 10000-11000 fwd to asterisk ip |
| 08:53.59 | masonc | did you try a sip phone outside or just providers |
| 08:54.07 | *** join/#asterisk darby_t (~tom@host-ip226-209.crowley.pl) |
| 08:54.14 | Zeeek | both but the phones are all behind their own NAT |
| 08:54.44 | Zeeek | all sip.conf entries are canreinvite=no |
| 08:54.53 | masonc | I do that |
| 08:54.58 | masonc | nat = yes |
| 08:55.03 | Zeeek | my son has a BT102 behind NAT |
| 08:55.14 | Zeeek | he only forwards 5060 to the phone and it all works |
| 08:55.31 | newl | same here. |
| 08:55.32 | masonc | connecting to your asterisk pbx |
| 08:55.36 | Blackvel | a WAG? |
| 08:55.37 | Zeeek | on the asterisk behind NAT, you have to use the externip= too |
| 08:55.38 | Blackvel | uh |
| 08:55.50 | Zeeek | Blackvel that's what I got, it works |
| 08:55.52 | Blackvel | you can not install a 3rd party firmware for that? |
| 08:56.01 | masonc | for what |
| 08:56.06 | Zeeek | I don't think so. I have a WRT54g at home |
| 08:56.25 | masonc | I have WRT54G - it can be upgraded |
| 08:56.33 | Blackvel | jepp, WRT, this is why I ask |
| 08:56.37 | Zeeek | and I haven't played with the other firmwares yet but I've downloaded them for when I retire and have lots of time to waste :) |
| 08:56.39 | masonc | the question is, would it make any difference |
| 08:56.45 | Zeeek | no |
| 08:56.49 | *** join/#asterisk Fabe (~spamhere@pD95B0BF7.dip0.t-ipconnect.de) |
| 08:56.50 | Zeeek | prolly not |
| 08:57.16 | Zeeek | let me say this, my setup works fine, I have several people on SIP phones, some with NAT some not, several providers |
| 08:57.20 | Zeeek | it all plays |
| 08:57.57 | Blackvel | wag has internal adsl modem? |
| 08:58.01 | Zeeek | ya |
| 08:58.04 | Blackvel | I guess no 3rd party firmware supports that |
| 08:58.09 | Zeeek | most everything we buy here does |
| 08:58.11 | Blackvel | I installed openwrt linux on my WRT |
| 08:58.38 | Zeeek | I don't think it has anything I need so I haven't messed with it |
| 08:58.41 | masonc | Zeeek, I would be interested to know if turning on the dmz feature broke it all |
| 08:59.12 | *** join/#asterisk Poincare (~jefffnode@dD5779B07.access.telenet.be) |
| 08:59.36 | Zeeek | masonc I've always been suspicious of DMZ. What order is it handled in? |
| 08:59.47 | Zeeek | ports forwarded first then DMZ? |
| 08:59.58 | Zeeek | I have never gotten anything to work with DMZ |
| 09:00.03 | masonc | ok, I am sld |
| 09:00.06 | masonc | ok, I am sold |
| 09:00.11 | Zeeek | to me, the simplest thing would have been to put the phone in DMZ |
| 09:00.19 | Zeeek | so that's what I did, didn't work |
| 09:00.19 | masonc | can't |
| 09:00.29 | Zeeek | no I'm talking about my experience |
| 09:00.33 | masonc | the phone I want to work is on another C class |
| 09:00.43 | Zeeek | I also tried putting asterisk in DMZ and couldn't get that to work |
| 09:01.02 | Zeeek | my own network incompetence, no doubt, but I DID GET fwding to work :) |
| 09:01.19 | masonc | I am testing now |
| 09:01.26 | Zeeek | I also have an Alcatel Speedtouch that freezes during calls |
| 09:01.36 | Zeeek | so I couldn't get that one to work with asterisk either |
| 09:01.50 | *** join/#asterisk c_k (~ck@82-43-178-166.cable.ubr06.newm.blueyonder.co.uk) |
| 09:03.27 | masonc | Did not work, one sided |
| 09:03.53 | Zeeek | what didn't work? |
| 09:03.59 | masonc | Did you set the forwards as UDP or both |
| 09:04.34 | Zeeek | As a newbie I prolly did both, but I think for rtp udp only is fine |
| 09:04.44 | Zeeek | on 5060 I think I did both |
| 09:05.01 | Blackvel | does the manager support turning off special events, e.g registered? |
| 09:05.14 | Zeeek | there may be quirks with the sipura phones, have you checked the mailing list and all that? |
| 09:11.34 | Zeeek | . |
| 09:11.38 | Zeeek | oh? |
| 09:12.49 | masonc | sorry, on the phone |
| 09:14.32 | Zeeek | yes, that is what it's all about... :) |
| 09:15.05 | *** join/#asterisk qwerp (~abc@60.48.87.197) |
| 09:15.09 | qwerp | harlo |
| 09:15.18 | Zeeek | gene |
| 09:15.19 | qwerp | i am having some problem.. |
| 09:15.43 | qwerp | when i make a SIP call from * to another * thru Iptel (SER) server, |
| 09:15.59 | qwerp | every 1 min ++ i got disconnect for no reason, any idea |
| 09:16.00 | qwerp | ? |
| 09:24.53 | *** join/#asterisk mAsH` (~mAsH@ppp-217-133-150-46.cust-adsl.tiscali.it) |
| 09:25.21 | *** join/#asterisk zotz (~zotz@24.231.32.109) |
| 09:30.29 | *** join/#asterisk tandrews (~tandrews@mail.grok.co.za) |
| 09:30.41 | tandrews | hello |
| 09:32.35 | tandrews | Can someone give me a hand ? I want to get hold of the "called number" on an incoming ISDN call.. |
| 09:32.48 | tandrews | I can't find a variable that contains this info |
| 09:33.15 | tandrews | If I use "bri debug span 1" I can see the number buried in the output |
| 09:34.00 | tandrews | I'm just unsure how to get hold if it for routing incoming fax calls to a specific MSN |
| 09:34.18 | Zeeek | http://www.marko.net/asterisk/archives/0206/0375.html |
| 09:35.43 | Zeeek | search for DNID |
| 09:36.11 | Zeeek | <PROTECTED> |
| 09:36.19 | *** join/#asterisk salviadud (~dude@201.133.209.245) |
| 09:36.24 | Zeeek | tandrews you may find answers here ^^^^^^ |
| 09:36.25 | ramtha | newl: with this option i can rewrite the callerid. out switch sees an atribute in the call (callerid network provided) this must be changed to "pass", otherwise, the callerid is not displayed |
| 09:36.36 | ramtha | but i can not find anything about this in the net |
| 09:36.39 | tandrews | thanks Zeeek - looking now |
| 09:36.56 | masonc | Sorry, Zeeek |
| 09:37.08 | Zeeek | about what? I'm not on hold :) |
| 09:37.41 | masonc | true, but I hate to break th eflow |
| 09:38.11 | Zeeek | heh well it'll soon be lunchtime here |
| 09:38.11 | masonc | I don't know if the sipura has flaws but I tried a laptop with sjphone and it had the same problem |
| 09:38.21 | Zeeek | ah that's interesting |
| 09:38.42 | masonc | I have changed the rtp range to 10,000-11,000 |
| 09:38.51 | Zeeek | did you try switching the phone that doesn't work with one thazt does, leaving all the configs the same? |
| 09:39.15 | masonc | it's not a equipment issue |
| 09:39.19 | masonc | I am sure of that |
| 09:39.58 | Zeeek | X-lite has an issue (I know, nothing to do with your prb) with silence suppression that causes one-way audio |
| 09:40.14 | Zeeek | I have no experience with the sipura phones though |
| 09:40.31 | Zeeek | you should post this to the mailing list and see what comes up |
| 09:40.45 | Zeeek | also have you searched the list? |
| 09:40.53 | masonc | which list |
| 09:40.59 | Zeeek | asterisk-users |
| 09:41.22 | masonc | [Asterisk-Users] Sipura SPA-841 and firewall |
| 09:41.26 | masonc | posted today |
| 09:41.30 | masonc | no responses |
| 09:41.32 | *** join/#asterisk guyee (~izomtriko@nextra.nudli.equitas.hu) |
| 09:41.35 | Zeeek | ~google site:lists.digium.com sipura |
| 09:41.49 | Zeeek | a little wide, but... |
| 09:42.09 | RoyK | perhaps jbot should use tinyurl |
| 09:42.12 | Zeeek | looks like nothing there |
| 09:42.37 | *** join/#asterisk tabmoW (tabmow@tabmow.linuxfordummies) |
| 09:42.39 | Zeeek | no, it's good to wake your ass up once in while RoyK |
| 09:43.47 | masonc | If I take the google search down to requiring 841 as a string, there's nothing |
| 09:44.02 | Zeeek | too soon... but I have seen some things |
| 09:44.22 | Zeeek | in fact a few people have them here, maybe they'll be around later |
| 09:44.40 | Zeeek | I almost bought one but decided to try a polycom ip500 |
| 09:44.40 | salviadud | has anyone fitted an xbox with asterisk in here? |
| 09:44.41 | *** join/#asterisk tengulre (~tengulre@61.185.238.166) |
| 09:44.43 | *** join/#asterisk krzee (k@user-0c9h8u5.cable.mindspring.com) |
| 09:45.01 | masonc | I have some polycoms coming in today, I hope, will try them and see if they do the same thing |
| 09:46.18 | Zeeek | they seem a lot more complex to setup |
| 09:46.27 | Zeeek | I hope it doesn't become a time sink |
| 09:46.33 | Zeeek | when I get mine I mean |
| 09:46.49 | *** join/#asterisk porche (~a@dsl81-215-31900.adsl.ttnet.net.tr) |
| 09:46.50 | porche | hi |
| 09:46.51 | tengulre | Zeeek,Hi! |
| 09:47.10 | RoyK | hmmmmm |
| 09:47.20 | masonc | I am provisioning ten of them so I should get the hand pretty quickly |
| 09:47.41 | Zeeek | well come back and give me a hand around May 15th |
| 09:48.01 | masonc | if fedex finds my missing box, I will |
| 09:49.22 | RoyK | people with house alarms might be less than happy to find that their alarm's modem connection to the central works rather badly over VoIP |
| 09:49.43 | Zeeek | voip is poo |
| 09:49.49 | masonc | that's an issue I have to deal with |
| 09:50.02 | zoa | who needs a sip jitter buffer urgently ? :) |
| 09:50.49 | RoyK | WE |
| 09:50.52 | RoyK | NOW |
| 09:51.25 | zoa | its there |
| 09:51.26 | zoa | :) |
| 09:51.31 | RoyK | it is? |
| 09:51.34 | RoyK | in head? |
| 09:51.37 | RoyK | tail? |
| 09:51.47 | zoa | no no |
| 09:51.48 | porche | hi ppl |
| 09:51.50 | zoa | some patches for -head |
| 09:51.58 | zoa | will put em online somewhere today |
| 09:52.00 | porche | got a problem with iax/dtmf detection with nufone |
| 09:52.02 | zoa | brand new patches |
| 09:52.30 | porche | any1 has got similar problems? |
| 09:52.52 | zoa | hmm maybe it will have to wait till tuesday |
| 09:54.19 | tainted- | porche don't use nufone to place or take calls.. that's when it breaks |
| 09:55.06 | RoyK | zoa: nice |
| 09:55.18 | RoyK | does anyone know when the 1.2 feature freeze will be? |
| 09:55.23 | RoyK | someone said february.... |
| 09:55.28 | RoyK | what year? |
| 09:55.30 | Blackvel | does the manager support turning off special events, e.g Event: Registry? |
| 09:57.11 | porche | tainted |
| 09:57.18 | porche | any suggestion for iax provider? |
| 09:57.34 | masonc | Zeeek - look at this: The sip dub lines show this |
| 09:57.36 | masonc | Retransmitting #1 (no NAT) to 10.10.19.5:5060: |
| 09:57.50 | masonc | that phone is the outside one and has nat=yes |
| 09:59.09 | masonc | porche - teliax |
| 09:59.21 | porche | teliax.com? |
| 09:59.46 | porche | do they provide concurrent outgoing? |
| 09:59.52 | masonc | yes |
| 10:00.14 | porche | cool |
| 10:00.19 | porche | let's try |
| 10:00.25 | masonc | great support and call quality |
| 10:00.37 | masonc | try a pay as you go account, only 2c / min |
| 10:01.07 | porche | yes saw |
| 10:01.09 | porche | seems promising |
| 10:01.14 | porche | even for 50 |
| 10:01.20 | porche | they provide 2500 |
| 10:01.21 | porche | mins |
| 10:01.36 | porche | hms |
| 10:01.42 | porche | no reason for it |
| 10:01.45 | porche | well main issue is |
| 10:01.52 | porche | i am fed up with alterign the providers |
| 10:01.59 | porche | i want iax but |
| 10:02.01 | masonc | everyone is |
| 10:02.02 | porche | a stable one |
| 10:03.02 | tainted- | porche u need concurrent outgoing? |
| 10:03.11 | tainted- | porche where are u calling to |
| 10:03.36 | tainted- | porche gafachi is pretty decent pricing |
| 10:04.20 | tainted- | porche pay as u go. don't expect any support |
| 10:07.23 | masonc | why is cheap pricing and no support attractive |
| 10:08.55 | tainted- | ok |
| 10:08.59 | tainted- | and they're stable |
| 10:09.00 | Aze` | Can i recive multiple call on xlite or sipura spa-841 ? |
| 10:09.30 | tainted- | masonc 1) most of the time with gafachi is client problem (99%) |
| 10:09.42 | tainted- | masonc 2) cheap is good |
| 10:09.52 | tainted- | 3) they're stable |
| 10:13.23 | *** join/#asterisk JonasNZ (jbergler@jonasnz.user) |
| 10:13.52 | *** join/#asterisk jonas (jonas@jonas.probe-networks.host.freenode) |
| 10:14.11 | newl | ramtha: so your carrier is overriding the information you're presenting then? |
| 10:15.04 | masonc | Zeeek - I got it to work |
| 10:16.21 | *** join/#asterisk Pkunk (~Pkunkage@mbbs.munnabhai.info) |
| 10:16.48 | Pkunk | is there any FAQ page on how to get disconnect supervision working with TDM400P ? |
| 10:16.49 | *** join/#asterisk Manipura (~chatzilla@dsl-ep-209-115-250-i114-cgy.nucleus.com) |
| 10:17.07 | Manipura | Hello everyone |
| 10:17.58 | masonc | Aze, yes you can |
| 10:18.14 | masonc | on the sipura you have two line appearances so you can take two calls, or four if you upgrade |
| 10:19.07 | Aze` | masonc .. i can recive one call and make a call |
| 10:19.14 | Aze` | but i cant recive 2 call |
| 10:19.24 | Aze` | it's busy |
| 10:19.39 | *** join/#asterisk Mike_TK (~Mike_@bell.yes.net.ua) |
| 10:19.47 | Aze` | is it configuration problem ? how configure it ? |
| 10:21.56 | masonc | you should disable the second line |
| 10:22.13 | porche | masonc, thanks |
| 10:22.16 | masonc | ext2: Line Enable = no |
| 10:22.21 | porche | do you have other alternative? |
| 10:22.27 | masonc | to teliax? |
| 10:22.36 | porche | yep |
| 10:22.46 | masonc | I also use livevoip |
| 10:22.48 | porche | the thing is |
| 10:22.53 | porche | it's ok in general |
| 10:23.01 | porche | but nufone's quality in voice better |
| 10:23.09 | porche | i only have got dtmf problem there |
| 10:23.32 | masonc | I just had a fabulous call to moscow - best quality I have ever heard |
| 10:23.48 | masonc | vi teliax |
| 10:23.49 | porche | over teliax? |
| 10:23.51 | porche | got it |
| 10:23.55 | porche | I was trying inside usa |
| 10:24.00 | porche | nufone was better |
| 10:24.14 | masonc | what codec |
| 10:25.20 | kimc | good morning from Detroit |
| 10:25.52 | porche | how is |
| 10:25.54 | porche | hmms |
| 10:25.55 | porche | ulaw |
| 10:26.26 | kimc | Pkunk: Are you having supervision problems with a TDM400 ? |
| 10:29.45 | newl | kimc: pipe me some WRIF will ya? The radio stations here suck. 8) |
| 10:30.16 | porche | mason, any other codec that can be better than ulaw? |
| 10:30.20 | Pkunk | kimc: well my phoneline supports it |
| 10:30.39 | Pkunk | kimc: but asterisk just sites there doing nothing when the other side drops the phone |
| 10:31.56 | porche | mason which package is the best for livevoip? |
| 10:32.06 | queuetue | Are some x100 "clone cards" better than others? Is there a source to purchase them instead of bidding on eBay? I just need a single FXO for this location... |
| 10:32.35 | kimc | newl: I'll send you a boat load of 'RIF :) |
| 10:32.50 | tengulre | quit |
| 10:33.05 | kimc | Pkunk: Are you looking for polarity reversal or what? |
| 10:33.51 | *** part/#asterisk JonasNZ (jbergler@jonasnz.user) |
| 10:33.58 | newl | kimc: it was good when they were using Real and had a direct link to the streams. Now they're streaming WMA and their hosting company has some really obscure crap to try and keep people away :/ |
| 10:34.07 | Pkunk | kimc: when the other end hangs up i can see a light go on and of in my phone |
| 10:34.44 | kimc | hmm.. lemme see what happens if I hang up a call from my cell phone.. |
| 10:34.57 | Pkunk | kimc: so i'm using kewlstart for my FXO ports |
| 10:35.19 | kimc | I'm using kewlstart too |
| 10:35.22 | Pkunk | fxs_ks , and only when busydetect=yes it is able to detect hangups |
| 10:35.45 | Pkunk | unfortunately busydetect=yes means a LOT of false hangups |
| 10:35.49 | porche | ehu |
| 10:35.56 | porche | masonc are you there? |
| 10:36.01 | kimc | Is there a cli command to turn up the debugging level for zap channels? |
| 10:37.42 | masonc | porche - I used a city plan I think |
| 10:38.11 | porche | mason |
| 10:38.38 | *** join/#asterisk cupis (~paul@theoldbakery.cupis.co.uk) |
| 10:39.15 | kimc | My dialplan is defective -it eventually goes to a fast busy when the other end hangs up |
| 10:39.28 | cupis | Anyone seen: "chan_zap.c:7143 zt_pri_error: PRI: received SETUP message for call that is not a new call, wicked!!!" before? Asterisk 1.0.5 on Debian sarge. |
| 10:39.43 | kimc | no PRI here |
| 10:39.57 | cupis | machine was running fine for the last few weeks - suddenly doesn't want to accept calls |
| 10:40.28 | guyee | Does * support H.323 for peers (not for clients)? |
| 10:40.49 | vpp | hmm why does it say 'unable to connect to remote askerisk' when i added h232 support |
| 10:40.52 | vpp | arghhhh |
| 10:42.11 | *** join/#asterisk lehel (~lehel@82.79.20.17) |
| 10:45.40 | *** join/#asterisk denon (denon@synapse.subneural.net) |
| 10:45.40 | *** mode/#asterisk [+o denon] by ChanServ |
| 10:46.49 | kimc | is anyone able to use iaxtel? |
| 10:47.11 | kimc | that is, is it up and running? |
| 10:47.54 | kimc | I've tried to set up iaxtel and us it to call Digium but it always returns something about |
| 10:48.04 | kimc | 'Noone is here to take the call..' |
| 10:51.40 | Luke-Jr | kimc: I did a while ago |
| 10:51.44 | *** join/#asterisk darby_t (~tom@host-ip226-209.crowley.pl) |
| 10:53.34 | kimc | Maybe I'll try again today |
| 10:53.44 | *** join/#asterisk Zeeek (~icechat5@Zeeek.sustaining.supporter.pdpc) |
| 10:54.15 | kimc | Just noticed yesterday there is a small er 'phone company' around here with local |
| 10:54.24 | kimc | dialtones all over Michigan |
| 10:54.42 | kimc | They claim free connections into iaxtel |
| 10:54.53 | kimc | So I tried it, and it didn't work |
| 10:55.11 | kimc | This is why I'm wondering about iaxtel still working |
| 10:55.20 | kimc | or not |
| 10:56.11 | Zeeek | not |
| 10:57.52 | *** join/#asterisk Comintel (~tom@seek-it.demon.co.uk) |
| 10:58.01 | Comintel | morning/afternoon |
| 10:58.16 | kimc | morning here in EDT :) |
| 10:58.33 | Comintel | im a bit stuck, i cant seem to get # transfers to work |
| 10:58.58 | kimc | Had the same problem until last week's breakthrough.. |
| 10:59.18 | *** join/#asterisk masonc (~lists@206.48.59.5) |
| 10:59.20 | Comintel | got some Cisco 7910 ip phones and i want to transfer calls between them |
| 10:59.30 | *** join/#asterisk cmk (~cmk_@p54A3D93C.dip.t-dialin.net) |
| 10:59.34 | Comintel | cant seem to figgure it out |
| 11:00.16 | kimc | Did you connect to the cli and issue 'sip debug' and try it? |
| 11:00.20 | porche | is free implementation of g.729 fine? |
| 11:01.05 | PatrickDK | did you use tT on the dial command? |
| 11:01.14 | Luke-Jr | porche: depends if you respect software patents, I think |
| 11:01.18 | PatrickDK | and your using ulaw right? |
| 11:01.18 | kimc | from a shell prompt: asterisk -rvvvvvvvgc then get a prompt: *CLI> |
| 11:01.30 | porche | patrick |
| 11:01.42 | porche | I just want to test it, afterwards, sure must buy |
| 11:01.43 | kimc | *CLI>sip debug |
| 11:02.16 | PatrickDK | porche, I'm not talking to you |
| 11:02.33 | porche | sorry meant luke |
| 11:02.58 | Comintel | i had the tT in the dial statemtn exten => 123,1,Dial(SCCP/test,tT) |
| 11:03.08 | Comintel | i think its in the wrong place |
| 11:03.12 | Zeeek | Comintel you're missing a parameter |
| 11:03.12 | PatrickDK | ya |
| 11:03.16 | *** join/#asterisk csg (foobar@i-195-137-6-228.freedom2surf.net) |
| 11:03.17 | Luke-Jr | porche: I'd personally go ahead and use it w/o buying it |
| 11:03.18 | *** join/#asterisk ariel_ (~Ariel@dsl-20-177.cofs.net) |
| 11:03.23 | Zeeek | show application dial |
| 11:03.28 | Comintel | what am missing |
| 11:03.30 | Luke-Jr | porche: you might not be willing/able to risk it, tho |
| 11:03.31 | Zeeek | show application dial |
| 11:04.04 | RoyK | show application dial |
| 11:04.08 | Zeeek | *CLI> show application dial |
| 11:04.14 | Zeeek | *CLI> show applications |
| 11:04.16 | Luke-Jr | show application dial |
| 11:04.17 | Luke-Jr | :) |
| 11:04.21 | Zeeek | *CLI> show RokY |
| 11:04.21 | Comintel | lol |
| 11:04.22 | Comintel | kk |
| 11:04.36 | porche | :) got it, just need to see, how it is compared to ulaw |
| 11:04.52 | Luke-Jr | porche: where'd you get it? |
| 11:05.14 | porche | http://www.readytechnology.co.uk/open/g729/ |
| 11:08.18 | Zeeek | what does anyone know about RFI |
| 11:08.24 | Zeeek | and DSL |
| 11:08.38 | Comintel | so, would it goe something like this? exten => 150,2,Dial(SCCP/cosh2,20,tT) |
| 11:08.41 | Zeeek | no one answered in 10000ms |
| 11:08.51 | Zeeek | try it Comintel |
| 11:09.00 | Luke-Jr | Zeeek: I could get DSL internet if I wanted to |
| 11:09.32 | Zeeek | I have two DSL connections and one has a weird problem just now causing me to put asterisk on a dynamic ip which is a real pain |
| 11:10.12 | Comintel | wee |
| 11:10.13 | Comintel | thanks |
| 11:10.30 | Comintel | boss is happy, lol |
| 11:10.49 | Comintel | the usual person who does the phone sytem is away |
| 11:11.18 | Zeeek | always nice to rise to a new challenge though, right? |
| 11:11.28 | Zeeek | now you'll be up for a raise |
| 11:11.34 | Zeeek | (rise if in UK) |
| 11:12.47 | Comintel | lol |
| 11:12.49 | Comintel | i doubt it |
| 11:13.12 | Comintel | the bitch is going to be re-writing the call circuit |
| 11:13.31 | Comintel | moved office |
| 11:13.53 | Comintel | so haveing to use asterisk and iax to transfer the old phone lines to the new office |
| 11:13.56 | masonc | Zeeek |
| 11:14.06 | Zeeek | yo maso |
| 11:14.18 | masonc | I found the solution |
| 11:14.23 | masonc | symettric NAT |
| 11:14.39 | Zeeek | ah... the old symmetric NAT ploy... |
| 11:14.46 | masonc | bingo |
| 11:14.55 | masonc | Bamm! |
| 11:15.00 | *** join/#asterisk heka (~heka@82.114.68.126) |
| 11:15.03 | Zeeek | and the solution ? |
| 11:15.19 | masonc | that was the solution, I enabled it |
| 11:15.30 | Zeeek | errrr on what, the linksys? |
| 11:15.53 | masonc | no, on the sipura |
| 11:16.04 | Zeeek | aha |
| 11:16.18 | Zeeek | it'd be great if you'd answer your post on the list |
| 11:16.29 | Zeeek | for future generations I mean :) |
| 11:16.40 | *** join/#asterisk Druken (~druken@CPE00119539b9cc-CM000e5cde4ca2.cpe.net.cable.rogers.com) |
| 11:16.46 | masonc | I usually do that - people think I am strange |
| 11:17.04 | Zeeek | Hey, it doesn't cost much time to provide the asnwer |
| 11:17.14 | Zeeek | I really appreciate when folks do it |
| 11:17.51 | masonc | Symmetric RTP: |
| 11:17.56 | masonc | that was the parameter |
| 11:18.09 | vpp | hey.. where can i add the LD_LIBRARY_PATH so it picks it up before asterisk loads? |
| 11:18.13 | Zeeek | I don't recall anything about that on GS phones... |
| 11:18.13 | vpp | i tried /etc/profiles |
| 11:18.15 | vpp | but no go |
| 11:18.24 | Zeeek | that's in a NAT Travesal section or sthing? |
| 11:18.32 | Luke-Jr | vpp: export it |
| 11:18.35 | masonc | I read something about that on them |
| 11:19.17 | vpp | LD_LIBRARY_PATH=$PWLIBDIR/lib:$OPENH323DIR/lib |
| 11:19.17 | vpp | export LD_LIBRARY_PATH |
| 11:19.27 | vpp | in /etc/profiles |
| 11:19.49 | vpp | *i define PWLIB etc too of course.. just didnt wanna flood too much in here |
| 11:20.32 | vpp | 'set' once its booted up shows it there, but asterisk has failed to load (looked in the logs), if i start it up once its all booted it runs fine |
| 11:20.44 | vpp | it gives me a library error, so its gotta be the path |
| 11:21.31 | vpp | hmmmmm maybe cos i use PWLIBDIR=$HOME/pwlib |
| 11:21.46 | vpp | and asterisk runs under a different user account? (pwlib is under root) |
| 11:22.57 | Aze` | Using dialagi .. i cant receve multiple calls on sip device... why ? |
| 11:31.25 | *** join/#asterisk gambolputty (~gambolput@cblmdm65-221-51-80.buckeye-express.com) |
| 11:38.30 | newl | kimc: who'd that carrier be, Telnet? |
| 11:45.37 | queuetue | Are some x100 "clone cards" better than others? Is there a source to purchase them instead of bidding on eBay? I just need a single FXO for this location... |
| 11:45.37 | *** join/#asterisk jwitte (~jwitte_@port-212-202-101-206.static.qsc.de) |
| 11:46.04 | jwitte | Hello, I get "PRI: !! No channel map, no channel, and no ds1? What am I supposed to identify?" when trying to setup my trunk. Any hints? |
| 11:46.11 | Druken | anyone ever had a problem with no sip clients being able to register? |
| 11:47.15 | Zeeek | Druken when our connection goes down, ya |
| 11:49.13 | Druken | aside from that... |
| 11:49.29 | Druken | i have local sip peers and even they aren't connecting... |
| 11:49.32 | Zeeek | suddenly no phones can register |
| 11:49.35 | Druken | i have no firewall... |
| 11:50.00 | Druken | yer the iax2 is working fine |
| 11:50.59 | queuetue | Where can I find a cheap, most-likely-to-work FXO? |
| 11:51.20 | Druken | ebay :) |
| 11:52.01 | queuetue | Druken, a) How do you know if it's the right model, and I'm not just getting ripped off, and b) Aren't some clones better than others? |
| 11:53.02 | Druken | queuetue: well, i've purchased i don't know how many off ebay, i refuse to pay more then 10 bux a card, and i find my "clones" work better than my digium hardware |
| 11:53.14 | queuetue | Druken, That's good to hear. |
| 11:54.04 | Zeeek | anyway since they don't make 'em anymore... |
| 11:54.19 | *** join/#asterisk fabioFVZ (~fabio@213-92-104-168.f5.ngi.it) |
| 11:54.39 | Druken | well, there's that too Zeeek |
| 11:54.46 | queuetue | Whjy don't they make them anymore? Digium can't make them as cheap as the clones? |
| 11:54.51 | Zeeek | Druken have you tried restarting-rebooting? |
| 11:54.57 | Optic | isn't the x100p just a voicemodem with a stable chipset? |
| 11:55.11 | Druken | tried restarting the server many times |
| 11:55.11 | Zeeek | queuetue I think the complication is when digium has to add support |
| 11:55.36 | Zeeek | perhaps they didn't want to cater to a market that wants (and needs) $10 cards |
| 11:55.40 | queuetue | Optic, If that's so, then shouldn't it be possible to write drivers for other voicemodems? |
| 11:55.47 | Optic | yes |
| 11:55.51 | Optic | it probably IS possible |
| 11:55.54 | Zeeek | go ahead |
| 11:55.59 | Optic | but it's hard because there's 1000 diffferent chipsets out there |
| 11:56.01 | Zeeek | the world is waiting etc |
| 11:56.05 | Optic | so it's good to pick one and support it well |
| 11:56.20 | queuetue | Zeeek, If it's true, maybe I will - I'm a kernel contributor. :) |
| 11:56.34 | Zeeek | we all have our cross.. |
| 11:56.38 | Optic | also, documentation is pretty thin I would imagine |
| 11:56.50 | Zeeek | that might be daunting |
| 11:57.29 | Zeeek | I have never written a driver of any kind (except maybe as an excercise years ago) but I'll bet a lot of the wrok is digging around undocumented features etc |
| 11:57.35 | christo | DBget - 'Retrieves a value from the database'. But what database is this referring to? How can I browse it and learn my way around it? |
| 11:57.39 | Zeeek | guessing parameters |
| 11:57.57 | Zeeek | database show (or is it the other order?) |
| 11:58.01 | Druken | christo: the internal asterisk database |
| 11:58.16 | Druken | database show |
| 11:58.18 | Druken | :) |
| 11:58.22 | Zeeek | even if you don't Put anything, asterisk does so you can look at it |
| 11:58.22 | christo | aaah I see. |
| 11:58.29 | christo | thanks |
| 11:58.40 | Zeeek | then just type a couple of useless entries in for fun |
| 11:59.05 | Zeeek | it's worth noting that you can do neat stuff with asterisk -rx "dbput ...." |
| 11:59.14 | *** join/#asterisk stefanocarlini (~stefano@coleman.almaweb.unibo.it) |
| 11:59.22 | Zeeek | in cron |
| 12:01.11 | christo | hmmm |
| 12:01.35 | Zeeek | testing the syntax? |
| 12:02.04 | Druken | hahahaha i'm such a goof |
| 12:02.25 | Zeeek | you had the router turned off? |
| 12:02.33 | Zeeek | cable unplugged? |
| 12:03.07 | Druken | nope... i had forgotten i had ser running on the same machine |
| 12:03.31 | Druken | i was playing around with ser this morning, and forgot to stop the process |
| 12:04.00 | *** join/#asterisk langals (~icechat5@196.7.14.183) |
| 12:05.16 | langals | Hi there...I am trying to get an iax2 softphone working. It will be used from behind a nat, connecting to asterisk on a public ip...port 4569 is opened in the firewall of the nat, but it still does not want to register - anything else I must be aware of? |
| 12:05.34 | ManxPower | ~docs |
| 12:05.35 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org |
| 12:05.42 | tainted- | i just finished some serious coding |
| 12:05.54 | tainted- | and GOD i love crossing stuff off my TODO: list |
| 12:06.00 | Druken | tainted: did you code in 1's and 0's? |
| 12:06.09 | tainted- | i cross it off ~oh~so~slow~ so i can enjoy it |
| 12:06.11 | ManxPower | tainted-, Anything we would be interested in? |
| 12:06.35 | tainted- | billing modules |
| 12:06.38 | tainted- | lol |
| 12:06.41 | tainted- | so probably not |
| 12:06.44 | ManxPower | ah. Nothing i'd be interested in. |
| 12:06.51 | ManxPower | i don't bill for alls |
| 12:06.58 | ManxPower | calls, even |
| 12:07.04 | Zeeek | balls |
| 12:07.11 | Druken | really... ManxPower wana give me an account then? :) |
| 12:07.23 | tainted- | Zeeek god that reminds me of someone from the past |
| 12:07.41 | ManxPower | Druken, only if you become an employee. Then you'll get an LD auth code and our LD carrier will bill you. |
| 12:07.42 | tainted- | but the crossing off of TODO: list item has become nearly religious |
| 12:07.47 | Faithful | Hey guys, do I really have to install X in order to use bluetooth with * ? |
| 12:08.03 | Druken | ManxPower: how much ya gonna pay me? :) |
| 12:08.10 | ManxPower | Faithful, What drunken psycho told you that? |
| 12:08.32 | tainted- | ManxPower what are u interested in? |
| 12:08.39 | ManxPower | Druken, Nothing. People PAY to work for this company. About US$30,000 is what they pay actually. |
| 12:08.40 | tainted- | maybe i've coded something that u'd like |
| 12:08.51 | ManxPower | You would also have to be a licensed real estate agent in the state of Louisiana. |
| 12:09.06 | tainted- | pay to work as real estate agent? |
| 12:09.18 | Druken | well, that's can't be all that hard to become licensed |
| 12:09.19 | ManxPower | tainted-, That's the way the industry works |
| 12:09.24 | tainted- | i'm bamboozled |
| 12:09.31 | masonc | Anyone working with the hospitality industry? |
| 12:09.43 | tainted- | masonc i have |
| 12:09.44 | ManxPower | tainted-, The recover their costs after selling 1 -2 houses. |
| 12:10.21 | tainted- | psssh 30k in 1-2 houses? must be some serious acreage |
| 12:10.21 | masonc | tainted- have you done hotels with asterisk? |
| 12:10.30 | ManxPower | I think they raised the desk fee this year and it's much closer to US$40,000, actually. |
| 12:10.46 | tainted- | no, but i've done internet terminals for hotels |
| 12:11.02 | Zeeek | there are a lot of areas where you have to pay to be an agent or wholesaler or franchisee |
| 12:11.15 | Druken | i don't think i could be a lowlife real estate agent, they rank with car salesman in my books |
| 12:11.15 | tainted- | crazy |
| 12:11.19 | Druken | i'm too honest |
| 12:11.20 | masonc | I'm looking to get some input from people who have used asterisk, we are about to use it in a hotel |
| 12:11.36 | ManxPower | tainted-, When the house is $500,000, the comissions add up fast. |
| 12:11.37 | masonc | thanks, I'm trying to sell some real estate |
| 12:11.48 | ManxPower | We have at least one agent that won't touch a hoise that lists for less than 1 million |
| 12:11.59 | tainted- | horse |
| 12:12.07 | syle | lol |
| 12:12.10 | tainted- | that bastard |
| 12:12.16 | Sander4000 | at last i got my tdm400p to work :) |
| 12:12.17 | syle | yeah people sell a 500k house everyday lol |
| 12:12.20 | tainted- | how could he discriminate like that |
| 12:12.25 | ManxPower | She's the 8'th in the nation for residential realistate sold. |
| 12:12.34 | tainted- | well here in CA 500k will buy u a dump |
| 12:12.34 | Sander4000 | incomplete manuals bah |
| 12:12.37 | masonc | around here you would be hard pressed to find a house for $55K |
| 12:12.37 | syle | typically they sell 100k-200k homes with a 3.4% commission |
| 12:12.40 | syle | kinda gay |
| 12:12.42 | masonc | $500K |
| 12:12.51 | tainted- | masonc where are u at |
| 12:13.04 | tainted- | what |
| 12:13.07 | tainted- | comm is 6% |
| 12:13.16 | syle | 6% |
| 12:13.21 | syle | what a rip off lol |
| 12:13.33 | masonc | www.anguillaguide.com |
| 12:13.39 | tainted- | i hate real estate agents as well |
| 12:13.48 | ManxPower | THAT is one of the major problems with IT at my largest customer. The buggest assholes are also the ones that brink in the most money to the company. |
| 12:13.59 | ManxPower | I can't type today. |
| 12:14.04 | queuetue | House prices are ridiculous all over. Real Estate crash is just about guaranteed. |
| 12:14.05 | tainted- | masonc where the fuck is that? |
| 12:14.19 | syle | lol |
| 12:14.39 | ManxPower | At this company the comissions are split between the company and the agent. |
| 12:14.50 | tainted- | why does anguilla sounds so naughty to me |
| 12:14.55 | masonc | no idea |
| 12:15.01 | Optic | mooo mooo |
| 12:15.02 | syle | well the thing is... |
| 12:15.03 | Druken | i like to see those 4-6 million dollar homes go for like 800k because of a power of sale |
| 12:15.05 | Druken | hehe |
| 12:15.06 | ManxPower | The actual split percentage is determed on the golf course, of course. |
| 12:15.12 | masonc | this is where I am putting the asterisk pbx |
| 12:15.15 | masonc | www.altamer.com |
| 12:15.19 | syle | gas prices etc are going up, meaning more people with those 300k homes are selling them |
| 12:15.46 | syle | so selling market is big right now |
| 12:15.46 | tainted- | lol |
| 12:15.47 | syle | but it will even out |
| 12:15.47 | tainted- | check out the 'ask carl' graphic |
| 12:15.47 | tainted- | LOL |
| 12:16.03 | tainted- | nice lip gloss carl |
| 12:16.13 | ManxPower | syle, this company has a mix of commercial sale/lease/management, and residential sales. |
| 12:16.20 | queuetue | Bell Canada Internet service is the worst I've ever encountered, I think... Random droputs are making reliability very difficult... |
| 12:16.47 | syle | commercial is always good, unfortunately they only come up in a blue moon |
| 12:16.53 | ManxPower | when the commercial market is down, the residential market is almost always up, and the reverse is also true |
| 12:16.57 | tainted- | masonc omg u are in the middle of paradise |
| 12:17.01 | tainted- | masonc i hate u |
| 12:17.05 | masonc | I know |
| 12:17.14 | masonc | lip gloss? |
| 12:17.18 | masonc | that's a beard |
| 12:17.40 | tainted- | his lips are just a little bit too sassy |
| 12:17.44 | ManxPower | I hate you too, masonc |
| 12:17.53 | syle | i don;t know about that manxpower |
| 12:18.04 | syle | so many factors involved |
| 12:18.05 | tainted- | no he's right |
| 12:18.07 | Druken | uhmm.... is that webcam supposed to show us anything? hehe |
| 12:18.09 | masonc | hey, I'm having such a good time I hate myself |
| 12:18.16 | ManxPower | syle, not ALWAYS, of course. |
| 12:18.20 | masonc | ahh, problemo |
| 12:18.20 | queuetue | masonc, Thanks - I was looking everywhere for a silver caviar bowl! |
| 12:18.21 | Druken | it's showing grass moving in the wind... but that's about it :) |
| 12:18.42 | masonc | that's not grass, that's a palm tree |
| 12:18.45 | *** join/#asterisk o_cee (~o_cee@h250n5c1o1095.bredband.skanova.com) |
| 12:18.49 | ManxPower | But when the economy is down, commercial is down. And when the economy is down people want to "nest", i.e. buy houses. |
| 12:18.56 | tainted- | http://www.altamer.com/ezimagecatalogue/catalogue/variations/224-400x500.gif |
| 12:19.08 | tainted- | carl carl carl |
| 12:19.35 | masonc | my wife drew that so she is sending you dirty looks |
| 12:19.39 | ManxPower | The USA Govt progam of propaganda to make people scared is good for the housing market. |
| 12:19.50 | masonc | terrorists are comign for you |
| 12:19.55 | masonc | they are under every bed |
| 12:20.03 | masonc | remind you of anything? |
| 12:20.04 | tainted- | oh u can tell right off the bat a woman created the website |
| 12:20.09 | syle | people buy houses when the interest rates are low period, ecomomy is alot of factors |
| 12:20.16 | tainted- | especially with THAT color scheme |
| 12:20.37 | masonc | comes from the interior designers colour choices |
| 12:20.43 | tainted- | makes me feel like i'm walking through the drapery section of our local discount mart |
| 12:20.43 | syle | as you can see in the last month interest rates have gone up |
| 12:20.45 | Druken | keep waiting for barbie to pop up somewhere |
| 12:20.49 | *** part/#asterisk lehel (~lehel@82.79.20.17) |
| 12:20.50 | syle | direct relation to gas prices |
| 12:21.12 | masonc | we did the site |
| 12:21.15 | syle | but same shit happens every year, and they will go down again and real estate will boom once again hehe |
| 12:21.18 | tainted- | masonc i'm just kidding.. it's a gorgeous site! |
| 12:21.26 | ManxPower | syle, Yes. But when the economy is down the govt tends to lower interest reates too |
| 12:21.26 | masonc | merci |
| 12:21.40 | tainted- | lol |
| 12:21.42 | Fraeggl | hi, does someone know good iax2 / mgcp softphones ? |
| 12:21.47 | tainted- | you're welcome, CARL |
| 12:21.51 | Fraeggl | or any at all :) ? |
| 12:21.59 | ManxPower | Fraeggl, Only in my fantasies. |
| 12:22.03 | masonc | I'll tell Carl you send regards |
| 12:22.08 | Druken | so... masonc, if we help.. we all get a week's free stay right? :) |
| 12:22.17 | masonc | discounted |
| 12:22.22 | Fraeggl | MaxPower, so there are none ? |
| 12:22.31 | tainted- | omg i actually HAVE that issue of architectural digest |
| 12:22.39 | ManxPower | Fraeggl, Alll softphones, not matter the protocol, suck. |
| 12:22.40 | tainted- | crazy |
| 12:22.43 | masonc | we got siz pages |
| 12:22.49 | masonc | we got six pages |
| 12:23.11 | Druken | agreed |
| 12:23.15 | Fraeggl | ManxPower just would want to try these protocols out.. |
| 12:23.17 | Druken | softphones blow ass |
| 12:23.23 | vpp | hmm |
| 12:23.38 | masonc | softphones are great when you are travelling and all you have is your laptop |
| 12:23.46 | masonc | but they are not suitable for serious work |
| 12:23.56 | syle | US gov't did a good job of convincing its people about "911" and terrorist attacks, hell they took billions of tax payers dollars for security lol |
| 12:24.00 | tainted- | masonc wanna see where I was a while back? |
| 12:24.06 | syle | i bet president is in caribean after this |
| 12:24.07 | masonc | sure |
| 12:24.18 | masonc | employees hate soft phones |
| 12:24.18 | ManxPower | And people wonder why I want to leave the USA. |
| 12:24.23 | masonc | always giving problems |
| 12:24.56 | ManxPower | I found out something interesting last inight. |
| 12:24.56 | masonc | too many configuration variables |
| 12:24.56 | masonc | I don't even want to use them |
| 12:24.56 | syle | problem is the middle class make up 90% of the voters, and they are easily brainwashed by television |
| 12:24.56 | Fraeggl | hm.. sflphone knows iax2 |
| 12:24.56 | tainted- | http://maps.google.com/maps?q=Dutch+Harbor,+AK&spn=0.283203,0.403091&hl=en |
| 12:24.56 | ManxPower | Apparently the sales person at the telco we use found my asterisk site (which is closed, but still has an add about me looking for a job in Europe) |
| 12:24.57 | syle | yes most rich people leave US |
| 12:25.16 | Fraeggl | is there any sofphone which can talk mgcp ? |
| 12:25.19 | ManxPower | He forwarded the page off the the person in charge of the non-asterisk systems at the company and our arch enemy. |
| 12:25.23 | masonc | why would most rich people leave the US? |
| 12:25.32 | syle | hell why not throw another billion to "homeland security" next year lol |
| 12:25.41 | syle | people actually allowing this crap |
| 12:25.43 | ManxPower | syle, Fatherland Security, you mean? |
| 12:25.53 | masonc | you have the most "rich friendly" govenment in the world |
| 12:26.39 | masonc | try being rich in france or england |
| 12:26.39 | syle | yeah, you know their fingerprint stuff at borders to try and prevent people from offshoring their money, smart move on gov;t making it look like a big media terrorist attack to take all that money |
| 12:26.40 | ManxPower | ixx, however, have been careful to fully disclose my plans with the MIS director, so he has known I want to move for many months |
| 12:26.58 | masonc | where is dutch harbour? |
| 12:27.16 | tainted- | aleutian islands |
| 12:27.28 | masonc | OMG, scary - cold? |
| 12:27.55 | tainted- | not when it's calm |
| 12:28.03 | masonc | temp? |
| 12:28.19 | tainted- | when it's windy, ur piss will freeze before it hits the snow |
| 12:28.20 | syle | i;ve considered moving the hell out of here a few times as well, i spent some time in different carribean islands this summer, problem is its so damn third world country that unless its a vacation or a retirement its not worth moving lol |
| 12:28.29 | masonc | not for me |
| 12:28.58 | *** join/#asterisk bjohnson (~bjohnson@jecinc.tor.istop.com) |
| 12:29.02 | syle | everything is cheap down there except the real estate |
| 12:29.17 | syle | but you can get around that by buying cheap labor to build a house for you hehe |
| 12:29.25 | tainted- | syle what's your problem with US? |
| 12:29.31 | tainted- | seems like everyone's got beef |
| 12:29.53 | syle | invasion of privacy is my beef |
| 12:30.10 | tainted- | please.. no one is interested in your p0rn |
| 12:30.25 | syle | trust me customs is |
| 12:30.26 | Luke-Jr | tainted-: unjust laws |
| 12:30.30 | tainted- | airport screening is done throughout most of the world |
| 12:30.40 | syle | spent 15 min going through my laptop last trip |
| 12:30.49 | Luke-Jr | tainted-: Patriot Act isn't |
| 12:30.52 | tainted- | try flying around europe |
| 12:30.54 | Luke-Jr | tainted-: Software patents aren't |
| 12:31.04 | *** join/#asterisk insync (~spam@66-188-104-11.mad.wi.charter.com) |
| 12:31.06 | syle | worse? |
| 12:31.06 | ManxPower | Here's a funny story (funny as in sad, not funny as in haha). One of the PHBs at one of my customers was caught by the personel director in his office with a female employee under his desk getting, shall we say, stress relief. |
| 12:31.15 | ManxPower | They fired the womand, but didn't fire the guy. |
| 12:31.31 | syle | why is that sad |
| 12:31.33 | tainted- | much worse.. they've been dealing w/ terrorists on their homeland waaay before US |
| 12:31.37 | syle | walk in there and cheer the dude on lol |
| 12:31.50 | ManxPower | syle, they should have either BOTH been fired, or NEITHER been fired. |
| 12:32.11 | tainted- | well what was his title vs hers |
| 12:32.29 | syle | what amazes me in australia has had all this border technology the US is getting for many years already |
| 12:32.30 | masonc | there's nothing cheap here, and there's no labour |
| 12:32.32 | tainted- | s/title/position/ but position would've opened myself to jokes |
| 12:32.40 | ManxPower | tainted-, that I don't recall, but he IS on the board of directors and she is not. |
| 12:32.49 | tainted- | there u go |
| 12:32.51 | tainted- | case closed |
| 12:33.03 | ManxPower | tainted-, that doesn't make it right. |
| 12:33.18 | *** join/#asterisk kisu (~Snake@218.237.126.163) |
| 12:33.50 | tainted- | should've, could've, would've, but didn't |
| 12:34.05 | queuetue | ManxPower, Why was it a fireable offense at all? Is your company against sex? :) |
| 12:34.23 | syle | worse in what way manx? |
| 12:34.25 | tainted- | talk about executive privileges |
| 12:34.28 | ManxPower | queuetue, There is some rule about it. |
| 12:34.29 | Zeeek | why is there no humor in asterisk messages? I'd like to see " -- Registered to '69.73.19.178', who sees us as "Duuuuuude" |
| 12:34.57 | tainted- | Zeeek post to mantis |
| 12:35.09 | syle | ever landed in texas entering US |
| 12:35.14 | Zeeek | Apr 29 14:12:17 NOTICE[804]: chan_sip.c:6644 handle_response: "If you can't be troubled to look this shit up, deal with it!" |
| 12:35.15 | syle | what a nightmare |
| 12:36.08 | tainted- | DOH!!!!!!!! |
| 12:36.16 | Zeeek | Watch out behind you! |
| 12:36.28 | Zeeek | you may become someone else's stress reliever |
| 12:36.36 | queuetue | Heh. |
| 12:36.53 | tainted- | lol |
| 12:37.00 | Zeeek | by the way, is queuetue French? |
| 12:37.13 | Zeeek | because it translates to "Dick kill" |
| 12:37.23 | Zeeek | painful |
| 12:37.24 | queuetue | No. :0 Why? |
| 12:37.38 | queuetue | queuetue does? |
| 12:37.58 | Zeeek | yes |
| 12:38.09 | Zeeek | queue = a line, a tail or "Johnson" |
| 12:38.26 | Zeeek | depending on your social framework |
| 12:38.35 | tainted- | anyone here been so lazy that after spilling water u just said, "not to humid in here, should just evaporate" |
| 12:38.53 | queuetue | My nick used to be (maybe a decade ago) queue - then it became hard to get whenI siged up, so I switched to queuetwo, which also became difficult, then queuetoo, which was also eventually co-opted. I have not met another queuetue yet. |
| 12:39.29 | tainted- | i had to resort to taintedtainted on aol |
| 12:39.49 | tainted- | Zeeek was prolly zeek at some point |
| 12:40.06 | Faithful | ManxPower: when I go to install bluez it complains about X as a dependancy for the keyring |
| 12:40.07 | tainted- | syle was prolly style |
| 12:40.17 | tainted- | ManxPower was prolly ManPower |
| 12:40.50 | Faithful | Ah he is gone |
| 12:40.51 | tainted- | Faithful how are u installing it |
| 12:41.13 | Faithful | debs |
| 12:41.27 | Zeeek | tainted LOL isn't strong enough for your comment, it killed me, everyone here in the offcie lokked over "wtf?" |
| 12:41.38 | tainted- | are there sources |
| 12:41.46 | Faithful | Yes |
| 12:41.53 | tainted- | just compile from source |
| 12:41.56 | tainted- | Zeeek what do u mean? |
| 12:42.16 | Zeeek | the humidity - don'tknow why, that hit me hard, hysterical laughter |
| 12:42.23 | Zeeek | so out there... |
| 12:42.28 | tainted- | i know!! |
| 12:42.38 | queuetue | Zeeek must be a big fan of evaporation... |
| 12:42.41 | tainted- | i spilt it b/c of your dick kill comment |
| 12:42.56 | tainted- | i jumped up b/c my box sits right under the desk |
| 12:43.02 | Zeeek | there's something cosmic about that - I must remember it when I next am suffering and feeling sorry for myself :) |
| 12:43.30 | tainted- | and then i realized, the case is small enough where if the water were to reach the ground, it'd flow away from the box |
| 12:43.33 | insync | has ne1 come across this situation I have my * server on a public ip at the office, i have 2 phones at a remote home behind the same fw. when calling from phone to phone that are behind the same fw i get no sound |
| 12:44.05 | tainted- | dude after a night of coding i don't even want to move my joints at the shoulder |
| 12:44.47 | Zeeek | what phones, insync |
| 12:44.59 | insync | grandstream |
| 12:45.00 | Zeeek | and why not just shout? |
| 12:45.12 | masonc | :-) |
| 12:45.18 | masonc | so droll |
| 12:45.20 | insync | thats what i told em hehe. |
| 12:45.37 | Zeeek | seriously, I haven't ever had two SIP phones on the same side |
| 12:45.48 | Zeeek | but I have had two IAX devices |
| 12:46.26 | insync | incidently it happens the same when both are doing fwd |
| 12:46.53 | tainted- | insync can u just peer them directly to each other? |
| 12:46.57 | tainted- | some phones let u do that |
| 12:47.42 | insync | i can try i just thought maybe i was missing something in the config so i dont have to mess with each phone |
| 12:48.12 | *** join/#asterisk dmccollum (~dmccollum@eycb01-00-cntnga-69-164-245-72.atlaga.adelphia.net) |
| 12:49.11 | tainted- | do they support IAX? |
| 12:49.16 | insync | yes |
| 12:49.23 | tainted- | b/c what u are doing is essentially double blind NAT |
| 12:49.30 | tainted- | both behind NAT i mean |
| 12:49.36 | masonc | which will never work |
| 12:49.43 | tainted- | IAX might be friendlier |
| 12:49.45 | masonc | without a proxy |
| 12:49.48 | Zeeek | ah, a hairpinning problem maybe? |
| 12:49.58 | tainted- | no it works, but u have to doing some stuff |
| 12:50.19 | tainted- | http://willypick.mindsay.com/?entry=10 |
| 12:50.42 | tainted- | "Yes, it's possible in spite of what ManxPower and others have said on the #asterisk channel!" lol |
| 12:50.45 | tainted- | from the site |
| 12:50.56 | Zeeek | absolutely |
| 12:51.24 | masonc | it's not double NAT if you have port forwarding |
| 12:51.44 | insync | my * is on a public ip |
| 12:51.48 | Zeeek | where did you find that? |
| 12:51.53 | insync | no nat server side |
| 12:52.06 | tainted- | insync well then it should be fine |
| 12:52.11 | tainted- | just fwd some ports |
| 12:52.18 | tainted- | 5060 for SIP |
| 12:52.21 | tainted- | 4569 for IAX |
| 12:52.30 | tainted- | and some RTP ports i think |
| 12:52.38 | *** part/#asterisk stefanocarlini (~stefano@coleman.almaweb.unibo.it) |
| 12:52.58 | tainted- | hmm wait |
| 12:53.00 | insync | forward them where if i want 2 devices |
| 12:53.01 | tainted- | he cheated |
| 12:53.07 | *** join/#asterisk fantomax1 (~fanto@81.208.114.250) |
| 12:53.10 | tainted- | his asterisk box IS the router |
| 12:53.16 | fantomax1 | hi all |
| 12:53.23 | tainted- | oh wait no, it's behind a linksys router |
| 12:53.32 | tainted- | insync just follow that guide |
| 12:53.34 | tainted- | it should work |
| 12:55.02 | insync | ah if i just use his clientside config it may work |
| 12:55.08 | Zeeek | that site was done by a one year old! |
| 12:56.45 | vpp | hmm where do u (if u can) set the packet size for the codecs in asterisk? |
| 12:58.14 | tainted- | good question |
| 12:59.00 | *** join/#asterisk Romik (~romik@1.fix.netvision.net.il) |
| 12:59.03 | Zeeek | isn't that done in the client? |
| 12:59.05 | tainted- | ok this water is really interfering with my typing |
| 12:59.16 | Zeeek | I thought it would evaporate? |
| 12:59.17 | tainted- | i'm leaving |
| 12:59.34 | tainted- | well i'm tired of the guilt trip it gives me |
| 12:59.44 | tainted- | I'M NOT GOING TO WIPE U |
| 12:59.53 | tainted- | sigh |
| 13:00.09 | tainted- | fear my high school level understanding of physics |
| 13:00.32 | *** join/#asterisk zotz (~zotz@24.231.32.109) |
| 13:00.42 | insync | thanx cya |
| 13:03.54 | Aze` | How Can i check if a sip device is busy without dial it ? |
| 13:04.10 | Zeeek | channelstatus? |
| 13:04.26 | *** part/#asterisk kisu (~Snake@218.237.126.163) |
| 13:05.12 | Sander4000 | does anyone know if an quadbri card should be detected at boot with linux fedora core 2? |
| 13:05.37 | Sander4000 | i think i'm going nuts :( |
| 13:06.19 | Sander4000 | every card gets detected but not the isdn quadbri |
| 13:07.22 | cypromis | do a lspci |
| 13:07.44 | Sander4000 | and the i can see if my comp sees the card? |
| 13:08.59 | Sander4000 | ah now i see 2 cards present :) now to put the isdn back in my system thanks! |
| 13:09.30 | *** join/#asterisk gpearson (~Graham@lrt2.niesc.k12.in.us) |
| 13:09.45 | cypromis | np |
| 13:12.51 | tzanger | morning |
| 13:13.24 | *** join/#asterisk negativecreep (~yama@202.147.174.98) |
| 13:13.28 | negativecreep | hi all |
| 13:13.52 | negativecreep | ia m using CVS HEAD and want to know if I need to patch it for ast_data support or not? |
| 13:14.50 | *** join/#asterisk vaewynAFK (freeman@mail.parrishmachine.com) |
| 13:15.19 | vaewyn | so... how bad is -head today? |
| 13:15.28 | Blackvel | haha |
| 13:15.29 | Blackvel | :) |
| 13:15.38 | Blackvel | maybe you should try tomorrow? :) |
| 13:15.38 | negativecreep | guys got any idea about ast_data |
| 13:15.40 | vaewyn | today seems like a good day to die...err... upgrade the test box |
| 13:15.40 | negativecreep | ?? |
| 13:16.03 | Blackvel | vaewyn: bad day today, one day before weekend |
| 13:16.04 | vaewyn | Blackvel: that bad? :} which parts are screwed up? |
| 13:16.18 | Blackvel | if you make an error, you have to work the full weekend |
| 13:16.30 | *** join/#asterisk Gand_DJ (~gandalf@ptr-207-54-104-24.ptr.terago.ca) |
| 13:16.34 | vaewyn | Blackvel: this is just a test box... not the prod onees :P |
| 13:16.57 | Blackvel | depends how often you have to test something |
| 13:17.08 | Blackvel | the test box may be needed earlier as you might think now :) |
| 13:17.15 | Blackvel | but |
| 13:17.26 | Blackvel | why do you guys even upgrade? what benefit do you expect from the upgrade? |
| 13:17.29 | vaewyn | hehehe... This test box I am the only one with access to it so :} |
| 13:17.37 | negativecreep | Blackvel: CVS HEAD has support for ast_data or do I need to patch it? |
| 13:17.42 | vaewyn | I need to upgrade so I can actually submit a couple patches |
| 13:17.44 | Blackvel | ok, I guess you won't have a problem so |
| 13:17.52 | Blackvel | negativecreep: dunno, I use vi |
| 13:18.03 | Sander4000 | is this normal ? ZT_CHANCONFIG failed on channel 32: No such device or address (6) when i first modprobe the wcte11xp it gives this error |
| 13:18.07 | negativecreep | Blackvel: ????? |
| 13:18.43 | Blackvel | negativecreep: why ast_data? wasn't that the thing which loads your extension from database? maybe ast@home does it all you want? |
| 13:18.46 | Sander4000 | and then when i modprobe the wctdm the light will go blinking on the wcte11xp |
| 13:19.16 | Sander4000 | so it seems to work |
| 13:20.08 | negativecreep | Blackvel: i want to load sip and all other configs from the mysql database..ast_data seems to be the solution...aint it? |
| 13:22.37 | Sander4000 | what is the best option for an quadbri ?? isdn4linux bristuff or mISDN ?? i only have to use it in te mode or is the isdn4linux only for single isdn cards |
| 13:22.39 | *** join/#asterisk MaggieL (~chatzilla@lata228-02-c130.lata228-c.voicenet.com) |
| 13:23.01 | Blackvel | maybe, or ast@home |
| 13:25.45 | negativecreep | Blackvel: ast@home seems to be a different thing. |
| 13:28.15 | ionix | ast@home is a management software |
| 13:28.28 | ionix | i.e a package that contains asterisk + GUIs |
| 13:30.50 | negativecreep | ionix: i need to store all sip/iax definitions in the database so that they are pulled from db on runtime. |
| 13:31.34 | negativecreep | i know its possible but cant seem to make it work with ast_data |
| 13:34.08 | *** part/#asterisk n4y (~frodo7@host-ip226-209.crowley.pl) |
| 13:35.12 | *** join/#asterisk iq (~iq@204-26-74-173.omah.qwest.net) |
| 13:37.19 | *** join/#asterisk fugitivo (~ajf@201.255.101.121) |
| 13:39.52 | *** join/#asterisk adjacent_ (~scott@office.bftwave.com) |
| 13:41.40 | Aze` | Sipura spa-841, when all (2) channels are busy i cant trasfert call with xfer (annunced trasfert)... how resolve ? |
| 13:41.43 | *** join/#asterisk felipex (~dsfdsf@host162-91.pool8533.interbusiness.it) |
| 13:42.05 | felipex | is there a way to see the number of the last call? |
| 13:44.21 | vaewyn | tail -1 /var/log/asterisk/cdr-csv/Master.csv :} |
| 13:45.42 | Optic | smooos |
| 13:47.04 | Signuts | Hello everyone |
| 13:48.04 | *** join/#asterisk wildgoose (~edward@xunil.mail.wildgooses.com) |
| 13:48.10 | *** join/#asterisk cbachman (~chatzilla@victory.ece.northwestern.edu) |
| 13:48.21 | Signuts | is there a way to have access to a dialplan in asterisk using /var/spool/asterisk/spool/outgoing/ . Specifically when the Channel: in the .call file isn't answered, I would like to take action, but I can't seem to find a way. |
| 13:55.08 | *** part/#asterisk Optic (dfraser@H31.C18.B96.tor.eicat.ca) |
| 13:56.37 | Zeeek | how do you say "not a single quote mark" in RegEx ? |
| 13:58.00 | zoa | ^'{1} or so |
| 13:58.04 | zoa | euh |
| 13:58.16 | zoa | [^']{1} |
| 13:58.22 | zoa | euhm |
| 13:58.30 | zoa | no still now what you would want |
| 13:59.18 | Zeeek | actually I meant double quote. I have a file that needs certain lines to be joined: looking to replace a line ending in " followed by a line NOT beginning with " |
| 14:00.16 | Zeeek | wait i think I got sumtin' |
| 14:00.46 | Zeeek | woot I did it |
| 14:02.37 | *** join/#asterisk Grooby (~Grooby@12.22.232.212) |
| 14:08.08 | Signuts | I'm creating a .call file in /var/spool/asterisk/outgoing and wondered |
| 14:08.08 | Signuts | if there was a way to detect if the Channel: in the call file did not |
| 14:08.08 | Signuts | answer (from within a dialplan). |
| 14:11.21 | Faithful | I just bought http://www.zyxel.com/product/model.php?indexcate=1092126124&indexcate1=&indexFlagvalue=1075687935 and as a dual port ATA adapter they are excellent. Not as many bells and whistles as the grandstream stuff but much better quality... and the features that count. |
| 14:12.31 | Faithful | And cheaper than the single port grandstream ATA |
| 14:12.42 | drumkilla | how much? |
| 14:13.00 | Faithful | AU$110 |
| 14:13.06 | ionix | expensive |
| 14:13.17 | Faithful | ? |
| 14:13.33 | masonc | I want to use a seperate t.38 ATA for faxing, any ideas on what to buy? and who can provide temrination? |
| 14:13.35 | *** join/#asterisk o_cee (~o_cee@h250n5c1o1095.bredband.skanova.com) |
| 14:13.38 | ionix | 86$ us |
| 14:13.45 | drumkilla | you beat me |
| 14:13.51 | drumkilla | I was about to type it |
| 14:13.56 | ionix | I can get a Linksys dual port ATA + router not locked for 70$ us |
| 14:14.14 | Faithful | Oh.. |
| 14:14.20 | Faithful | Hmmm |
| 14:14.36 | Faithful | I would much rather Linksys |
| 14:14.54 | ionix | gotta go. see you |
| 14:15.40 | *** join/#asterisk vinko (~vinkoval@63.170.64.37) |
| 14:17.42 | *** part/#asterisk Zeeek (~icechat5@Zeeek.sustaining.supporter.pdpc) |
| 14:24.31 | Signuts | anyone want to help me with a .call file problem. I am getting "WARNING[4837]: chan_sip.c:694 retrans_pkt: Maximum retries exceeded on call 0820d86a344b2a30073a5f9171902 (Critical Request)" |
| 14:24.42 | Signuts | and would like to trap this in the dialplan |
| 14:27.47 | *** join/#asterisk santiago (~santiago@63.245.86.199) |
| 14:29.57 | Signuts | anybody? |
| 14:30.05 | Moc | hi mark |
| 14:30.11 | vaewyn | Signuts: that means you don't have network to the device... |
| 14:30.58 | *** part/#asterisk MaggieL (~chatzilla@lata228-02-c130.lata228-c.voicenet.com) |
| 14:30.59 | Signuts | vaewyn, not always, it's a SIP channel, which means the user is just not registered. |
| 14:31.32 | *** join/#asterisk Lusid (~root@69.25.178.6) |
| 14:31.58 | Lusid | Anyone know if there are any softphones besides GnoPhone that support the SendURL function? |
| 14:32.40 | vaewyn | No... it means they lost a connection from a previous registration... If they havn't registered you won't get that error |
| 14:32.48 | vaewyn | qualify would fix that |
| 14:33.10 | *** join/#asterisk lilwookie (~zoidmeste@modemcable215.87-81-70.mc.videotron.ca) |
| 14:33.25 | lilwookie | Goodmoring all :) |
| 14:34.18 | *** join/#asterisk jsolares (~jsolares@200.30.141.85) |
| 14:34.29 | porche | q: which way is best to develop, over dial plan or over agi? |
| 14:34.35 | lilwookie | Quick question, I am using Realtime IAX and it seems to be working, I can make/take calls etc. But when I do IAX2 SHOW PEERS the DB channels dont show up.. this normal? |
| 14:34.54 | Signuts | vaewyn, i'm creating the call from a .call file (placing a tmpfile in /var/spool/asterisk/outgoing) If the channel: field is nto answered, I never get to my context/extension/priority and therefore dont' have the ability to tell the user waiting that I couldn't complete the call. My only option is to have a timeout +5 seconds or so greater than the WaitTime: field in the .call file. It's not a very clean method. |
| 14:35.47 | *** join/#asterisk trig_hm (~jb@home.monkeypr0n.org) |
| 14:36.06 | Signuts | porche, I always base that on how complex the application is. |
| 14:36.15 | Signuts | more complex programs I put into an AGI |
| 14:36.26 | porche | well it's complex |
| 14:36.30 | porche | but the problem is |
| 14:36.38 | porche | there is a problem in agi |
| 14:36.48 | porche | it runs the agi program |
| 14:36.55 | porche | and control goes to agi from then on |
| 14:37.04 | porche | butfor example I have record in one place |
| 14:37.14 | porche | if user hangs up just there |
| 14:37.21 | porche | I have some zombies around |
| 14:37.28 | porche | I do have h, t |
| 14:37.33 | porche | all possible detection |
| 14:37.34 | porche | s |
| 14:37.35 | jsolares | use the callback |
| 14:37.42 | jsolares | use the channel status codes |
| 14:37.45 | *** join/#asterisk convey (~van@206.137.18.56) |
| 14:38.13 | porche | hm, true, |
| 14:38.28 | porche | it's the agi's responsibility to detect it sometimes I think |
| 14:38.33 | porche | got it |
| 14:38.34 | porche | tnx |
| 14:38.42 | Lusid | So, GnoPhone is the only softphone that supports popping a web url? |
| 14:38.53 | porche | to summarize, agi is better i think for the complex ones |
| 14:38.59 | jsolares | it's not, it should be tho |
| 14:39.23 | *** join/#asterisk dikini (~vlado@mec0028.engin.cf.ac.uk) |
| 14:39.26 | *** join/#asterisk uzd (optimist@eurocompton.net) |
| 14:39.31 | uzd | hi |
| 14:39.34 | Sander4000 | ZT_SPANCONFIG failed on span 1: No such device or address (6) i get this error whe doing ztcfg after loading qozap.ko can anyone help me? |
| 14:39.44 | jsolares | but then again, having proper detection only adds a couple of lines at most |
| 14:39.54 | *** join/#asterisk Gh0sty (~Ghosty@ip-81-11-215-197.dsl.scarlet.be) |
| 14:40.05 | *** join/#asterisk jeffik (jefik@69.158.2.124) |
| 14:40.15 | uzd | would someone explain the purpose of the CONSOLE global var in the extensions.conf-dist? |
| 14:40.23 | uzd | i can't get my Dial() macro working |
| 14:40.40 | uzd | not *my*, but the Dial() macro, rather |
| 14:40.41 | porche | got it js |
| 14:40.44 | *** join/#asterisk xAD^nFL (~xAD_nFL@host144-199.pool8290.interbusiness.it) |
| 14:40.48 | porche | I think I must redesign the agi code |
| 14:40.57 | porche | for faster work, i did it with php |
| 14:41.03 | porche | need to at least move to perl |
| 14:41.10 | jsolares | i'm using perl |
| 14:41.38 | uzd | you guys should shift to python :) |
| 14:41.39 | porche | well, I do have several libs in php, ready to go, it was quicki |
| 14:41.45 | jsolares | i'll be moving some to C sometime soon |
| 14:41.47 | porche | uzd, i hate it |
| 14:41.52 | uzd | why? |
| 14:41.54 | jsolares | python is weird |
| 14:42.02 | uzd | well, it's object oriented.. |
| 14:42.06 | porche | yes, C seems more reasonable, but time consuming |
| 14:42.16 | Signuts | I use python all the time, things it never has... can't OOP have myvar++ or myvar += 5, man! |
| 14:42.19 | uzd | that's the biggest obstacle if you're not an OOP person |
| 14:42.22 | porche | most of the time I dont have so much time, need to prototype very quick |
| 14:42.24 | uzd | but it's worth learning. |
| 14:42.25 | jsolares | i'm good with the C*'s , perl, php, java (i hate it tho)... but python... it's just weird |
| 14:42.37 | uzd | isn't php5 oop? |
| 14:42.46 | porche | i have java also |
| 14:42.56 | kimc | how can I ring a zap channel and a sip channel from a DID? |
| 14:43.01 | jsolares | uzd, i think so |
| 14:43.08 | kimc | this doesn't work: |
| 14:43.09 | kimc | exten => s,3,Dial,Zap/1|20&SIP/302,21,tr |
| 14:43.10 | porche | done a project with it, damn, it has got bugs in tcp/ip code at that time |
| 14:43.10 | jsolares | i'm mostly in php4 land tho |
| 14:43.18 | porche | yes php5 is oop |
| 14:44.14 | porche | y php4 more that enough usually |
| 14:44.41 | lilwookie | is it normal that Realtime IAX users dont show up in CLI's IAX2 SHOW PEERS ? |
| 14:49.24 | *** join/#asterisk o_cee (~o_cee@h250n5c1o1095.bredband.skanova.com) |
| 14:49.46 | *** join/#asterisk Goshen (~Goshen@70-57-80-147.slkc.qwest.net) |
| 14:51.52 | langals | hi there...I am using a client which has VAD enabled and it cannot be turned off....when going through a NAT it only keeps the address mapping for a few seconds when no audio from the client....thinking a solution might be to send an empty audio packet every second or so...does anyone know if this would be possible? |
| 14:54.40 | Essobi | Well, umm. |
| 14:54.48 | Essobi | That's called VXD. |
| 14:55.16 | Essobi | It is SIP? |
| 14:55.35 | Essobi | I know there's been some VAD/VXD work done.. |
| 14:57.38 | *** join/#asterisk SuPrSluG (~SuPrSluG@pool-129-44-142-202.buff.east.verizon.net) |
| 14:59.53 | mutilator | so turn off vad for the server end? |
| 15:01.09 | *** part/#asterisk dikini (~vlado@mec0028.engin.cf.ac.uk) |
| 15:01.12 | *** join/#asterisk tessier (~treed@210.245.99.126) |
| 15:01.24 | uzd | ok. i have outbound working fine.. inbound goes straight to voicemail. any advice on why this would be? |
| 15:01.55 | uzd | i'm peered up propery |
| 15:01.57 | uzd | properly |
| 15:01.58 | SuPrSluG | uzd:dialplan |
| 15:02.20 | uzd | well |
| 15:02.32 | SuPrSluG | uzd:show incoming @ pastebin.ca |
| 15:02.40 | uzd | let me ask this.. in the extensions.conf-dist, is there a default context for incoming from pstn? |
| 15:02.42 | uzd | i couldn't find it |
| 15:02.47 | uzd | and have no idea how to create one |
| 15:03.05 | SuPrSluG | uzd:defined in zapata.conf |
| 15:03.40 | uzd | the dialplan for incoming is in zapata.conf? |
| 15:03.58 | SuPrSluG | uzd:the context for incoming is |
| 15:04.29 | SuPrSluG | uzd:where it's pointed to is where it goes |
| 15:04.54 | christo | could somebody please explain what on earth this macro is doing? http://pastebin.ca/10559 |
| 15:04.54 | uzd | what's the name? |
| 15:05.54 | langals | mutilator - but it is on the client end - if there is no audio from the user, then the client does not send any packets |
| 15:05.55 | SuPrSluG | uzd:context = ? in zapata.conf |
| 15:06.17 | uzd | SuPrSluG, you're speaking to abbreviated |
| 15:06.21 | uzd | i can't figure out what you're saying |
| 15:06.22 | uzd | heh |
| 15:06.30 | Chotaire | yeah, lots of ficken. |
| 15:06.35 | uzd | context = default |
| 15:06.42 | SuPrSluG | uzd:you tell it what context to go to . mine = inbound |
| 15:06.57 | uzd | ok, what does your inbound context look like though? |
| 15:08.28 | *** join/#asterisk ManxPower (~eric@stirprop-S0-0-0-26.ndcr2.datasync.net) |
| 15:09.20 | stoyan | can I place a test call to one of the sip channels from the CLI? |
| 15:09.34 | porche | mason |
| 15:09.39 | porche | teliax is crazy :) |
| 15:09.52 | porche | they disabled my account, really weird |
| 15:12.52 | *** part/#asterisk kimc (~freenode@pcp09643046pcs.wbrmfd01.mi.comcast.net) |
| 15:12.55 | stoyan | can you tell me what this warning message means: chan_iax2.c:5631 set_config: Ignoring port for now |
| 15:13.00 | uzd | SuPrSluG, you still here? |
| 15:13.18 | SuPrSluG | uzd:yep. uno momento. |
| 15:13.32 | uzd | bueno |
| 15:13.34 | *** join/#asterisk kimc (~freenode@pcp09643046pcs.wbrmfd01.mi.comcast.net) |
| 15:13.58 | SuPrSluG | uzd:http://pastebin.ca/10562 |
| 15:14.10 | uzd | mine is here: http://pastebin.ca/10563 |
| 15:14.32 | *** join/#asterisk ckruetze (HydraIRC@cpc3-cmbg7-5-0-cust100.cmbg.cable.ntl.com) |
| 15:15.13 | *** join/#asterisk falz (~falz@proxy.supranet.net) |
| 15:15.30 | *** join/#asterisk o_cee (~o_cee@h250n5c1o1095.bredband.skanova.com) |
| 15:15.34 | xAD^nFL | yo all, i have some and weird problems with Asterisk Stable 1.0.7 + Eicon Diva Server 2-Bri + EPIA M6000 + JugCAPI 0.3.5 + Kernel 2.6.10 Capi/Eicon Module, when i made a call over Asterisk (console) or over SIP Phone and the other side pickup the phone.. Asterisk Crash ;-( , i have here the GDB output message, please help me, many thanks |
| 15:16.33 | uzd | what the hell |
| 15:16.37 | uzd | i don't understand |
| 15:16.52 | xAD^nFL | eheh ..yep i have miss |
| 15:16.53 | SuPrSluG | uzd:ok. you're using macro-standard extension. problem = to use that the dialling party must dial an extension. I use that for internal #'s |
| 15:16.58 | xAD^nFL | i made a call over CAPI |
| 15:17.07 | *** part/#asterisk ckruetze (HydraIRC@cpc3-cmbg7-5-0-cust100.cmbg.cable.ntl.com) |
| 15:17.08 | uzd | SuPrSluG, i replaced mine with yours and changed the extension |
| 15:17.32 | uzd | same deal... incoming calls to right to voicemail |
| 15:17.51 | SuPrSluG | uzd:u reloaded? |
| 15:17.58 | uzd | yeah |
| 15:18.44 | SuPrSluG | uzd:context in zapata.conf=from-pstn? |
| 15:18.53 | uzd | yep |
| 15:19.28 | uzd | so close, yet so far :) |
| 15:20.22 | SuPrSluG | uzd:go to CLI> and set verbose 4. then dial in and paste what happens. I should be able to figure it out then. |
| 15:20.31 | dmccollum | Any suggestions on a good distributor for Digium cards? I've been talking with NextUSA out of Greenville, SC. They're price for a TDM11B is $173.00. Not sure if that's considered a good reseller price or not. |
| 15:20.54 | hohum | this is kind of OT |
| 15:20.54 | hohum | but |
| 15:20.57 | dmccollum | Sorry the name is NETXUSA not nextusa. |
| 15:21.31 | hohum | if there are any SIP experts around, what SIP message contains what RTP Proxy I should use on inbound calls <IE my vendor sent the INVITE> |
| 15:21.34 | *** join/#asterisk cmk (~cmk_@p54A3D93C.dip.t-dialin.net) |
| 15:21.52 | *** join/#asterisk odie_flocon (~Odie@ptr-64-201-182-211.ptr.terago.ca) |
| 15:22.20 | odie_flocon | has anybody tried to install * on mdk 10.2???? |
| 15:22.31 | uzd | SuPrSluG: http://pastebin.ca/10564 |
| 15:22.46 | SuPrSluG | uzd:k |
| 15:24.50 | *** join/#asterisk Hogie (daniel@alpha.dfwservers.net) |
| 15:25.03 | *** join/#asterisk astoria (~haydenth@66.235.201.217) |
| 15:25.08 | Hogie | does anybody run * with any digium cards on a Dell PE SC420 by chance? |
| 15:25.13 | SuPrSluG | uzd:r u dialling into BV or a real pstn # |
| 15:25.23 | hohum | where would I be pulling the address to send the RTP stream to? From: or Via:? |
| 15:25.30 | astoria | Hey, can someone confirm for me that the TE110P can do NI-1 Signaling? Cant find anything on google. |
| 15:25.31 | uzd | a real pstn |
| 15:25.34 | hohum | or something else? |
| 15:25.49 | uzd | I have a number from broadvoice |
| 15:25.56 | uzd | I'm dialing into that number from my home pstn |
| 15:26.09 | *** join/#asterisk jief- (~jief@modemcable196.182-80-70.mc.videotron.ca) |
| 15:26.13 | jief- | hello |
| 15:26.18 | hohum | or contact perhaps? |
| 15:26.35 | SuPrSluG | uzd:turn off sip debug w / sip no debug. i want to see if the zap channel switch is triggered. ok |
| 15:26.44 | uzd | ok |
| 15:27.52 | uzd | SuPrSluG, it didn't display anything |
| 15:28.46 | SuPrSluG | uzd:it should look like http://pastebin.ca/10565 |
| 15:29.33 | SuPrSluG | uzd:does CLI> show zap channels have a card there? |
| 15:29.46 | uzd | a card? |
| 15:29.47 | uzd | no |
| 15:29.51 | uzd | this is how i'm setup |
| 15:29.52 | SuPrSluG | channel |
| 15:30.20 | uzd | nothing |
| 15:30.22 | uzd | ok |
| 15:30.26 | stoyan | can you tell me what this warning message means: chan_iax2.c:5631 set_config: Ignoring port for now |
| 15:30.34 | uzd | i think we're confusing things |
| 15:30.51 | uzd | <PROTECTED> |
| 15:30.52 | uzd | <PROTECTED> |
| 15:32.48 | *** join/#asterisk Yellow_FUzzy (yellow@c211-31-41-9.wavrl1.nsw.optusnet.com.au) |
| 15:32.51 | SuPrSluG | uzd:try a lspci and see if your x100p card is recognized |
| 15:33.04 | uzd | i don't have a card |
| 15:33.11 | falz | anyone using 79xx cisco's have a way to make the caller ID when you "transfer" via soft buttons be the original caller, instead of the party that initiates the xfer? |
| 15:33.16 | uzd | sorry, I think I was confused earlier |
| 15:33.28 | uzd | perhaps I am dialing into BV |
| 15:33.33 | SuPrSluG | uzd:u need that to call in from the pstn. |
| 15:33.43 | stoyan | can I place a test call to one of the sip channels from the CLI? |
| 15:33.53 | uzd | SuPrSluG, I'm dialing into bv |
| 15:33.56 | *** join/#asterisk VPhantom (~lis@modemcable084.168-200-24.mc.videotron.ca) |
| 15:34.05 | hohum | What header of a SIP message tells me where to send the RTP stream? |
| 15:35.09 | uzd | i had this config working fine with asterisk@home but I reinstalled :) |
| 15:36.25 | uzd | SuPrSluG , you still around? |
| 15:36.38 | uzd | or did I drive you to the brink of insanity? |
| 15:39.08 | *** join/#asterisk christo (~chris@office.enovi.com) |
| 15:39.18 | SuPrSluG | uzd: u need to start here. http://voip-info.org/wiki-Asterisk+settings+Broadvoice if u have a did(pstn # w/ them. |
| 15:39.31 | VPhantom | Any Dialogic VFX users around? I seek advice on which hardware to purchase. |
| 15:39.41 | *** join/#asterisk Veto (mdkuser@cpe-66-69-38-192.satx.res.rr.com) |
| 15:39.46 | uzd | SuPrSluG, I read all that and have those settings defined |
| 15:39.49 | hohum | what header in a SIP message tells me where to send the RTP stream |
| 15:40.35 | uzd | hohum, i bet the RFC will tell you |
| 15:41.18 | uzd | I have a did |
| 15:41.31 | uzd | i'm getting the terminology f'd up |
| 15:41.32 | uzd | heh |
| 15:42.13 | uzd | anyways... thanks for your help |
| 15:43.00 | SuPrSluG | uzd:CLI> sip show registry |
| 15:43.15 | SuPrSluG | uzd:bv there? |
| 15:43.18 | uzd | Host Username Refresh State |
| 15:43.19 | uzd | sip.broadvoice.com:5060 5618922172@s 16 Registered |
| 15:44.08 | SuPrSluG | uzd:also add insecure=very to ur sip.conf settings for bv |
| 15:44.21 | uzd | i have it. |
| 15:47.15 | uzd | what is zapata? |
| 15:47.30 | Hogie | does anybody run * with any digium cards on a Dell PE SC420 by chance? |
| 15:51.04 | astoria | nope, but you'll probably run into IRQ problems |
| 15:51.18 | uzd | i think this may be related to the fact i'm running on fbsd 4.x.. zapata has some issues on that version. |
| 15:51.23 | astoria | i've seen some people discuss this on the mailing list |
| 15:51.31 | uzd | and i had to hack the code slightly to get it to work. |
| 15:51.48 | astoria | time to go home yipee! |
| 15:52.23 | SuPrSluG | uzd:can u make outbound calls thru bv? |
| 15:52.53 | uzd | yep |
| 15:55.43 | SuPrSluG | uzd:can u make outbound calls thru bv |
| 15:55.51 | uzd | SuPrSluG, yes I can |
| 15:56.37 | SuPrSluG | uzd:try here ? http://edvina.net/broadvoice/ i don't use bv. seem there are some issues. |
| 15:56.39 | lilwookie | anyone using Realtime IAXpeers? |
| 15:57.51 | uzd | SuPrSluG, yeah, that's very old. the problems are non-existent now |
| 15:58.01 | uzd | that was <1.01 |
| 15:59.11 | mike-ff | apparently not completely non-existent ;) |
| 15:59.22 | mike-ff | ok, I have nothing useful to add, I'll be quiet now |
| 15:59.46 | uzd | these issues aren't related to bv per se. but rather my ignorance and lack of configuration skills |
| 15:59.52 | bkw_ | Broadvoice doesn't have ANY issues |
| 15:59.57 | bkw_ | with cvs-head |
| 15:59.58 | bkw_ | NEXT!!! |
| 16:00.01 | bkw_ | move along |
| 16:00.04 | bkw_ | :P |
| 16:00.22 | uzd | bkw_, you using bv? |
| 16:01.56 | tzanger | hahaha |
| 16:02.01 | tzanger | broadvoice just has issues |
| 16:02.04 | tzanger | it has nothing to do with asteirsk |
| 16:02.16 | uzd | who said it did? |
| 16:03.20 | tzanger | just look at the mailing list archives |
| 16:03.30 | tzanger | every fucking week "is broadvoice down" "having bv issues" "can you get to broadvoice" |
| 16:03.33 | uzd | that was from several years ago |
| 16:03.35 | uzd | oh |
| 16:03.36 | tzanger | jesus christ people, stop using it |
| 16:03.36 | uzd | well |
| 16:03.44 | tzanger | uzd: several years ago? This is THIS WEEK |
| 16:03.46 | tzanger | and last week |
| 16:03.48 | tzanger | and the week before |
| 16:03.50 | tzanger | and the week before that |
| 16:03.59 | uzd | sorry, I thought you were referring to the asterisk patch |
| 16:04.00 | tzanger | broadvoice is NOT a shining example of a stable VOIP company |
| 16:04.14 | uzd | sure, but they offer the best LD rates to a lot of places |
| 16:04.20 | tzanger | nufone, OTOH, is. They're just permanently "not officially open" :-) |
| 16:04.29 | tzanger | uzd: do you buy everyting based SOELY on price |
| 16:04.37 | tzanger | how good is something when it's not there 1/2 the time you need it? |
| 16:04.47 | tzanger | is 1/3 of a cent really worth the pain and aggravation? |
| 16:04.49 | tzanger | seriously |
| 16:04.51 | uzd | why would y ou assume everthing? |
| 16:04.57 | tzanger | I'm not assuming anything |
| 16:05.06 | uzd | I'm using VOIP to save money on international long distance |
| 16:05.20 | uzd | broadvoice was the least expensive and I'm not bound to a contact |
| 16:05.24 | tzanger | uzd: and as I said, is the 1/3 of a cent difference between BV and the alternatives worth the hassle? |
| 16:05.27 | uzd | why wouldn't I drop $25 and seee? |
| 16:05.35 | uzd | if it works I use it, if it doesn't I switch |
| 16:05.36 | uzd | big deal |
| 16:05.37 | *** join/#asterisk sudhir492 (~sudhir@4.7.57.152) |
| 16:05.43 | tzanger | sounds like a great plan |
| 16:05.44 | Gand_DJ | Have you tried calling them for help? |
| 16:05.56 | uzd | what's wrong with the plan? |
| 16:05.59 | *** join/#asterisk MikeJ[Laptop] (~ircatjerr@mi.origenfinancial.com) |
| 16:06.01 | tzanger | but 5 minutes of searching on the list would reveeal that they're probably the #1 worst VOIP company in general use |
| 16:06.26 | sudhir492 | when I do database show, I see something like line=92m1z05n at the end of some entries |
| 16:06.32 | uzd | I've been using it for weeks and I haven't had any issues. I just started having issues when I started using it on freebsd 4 instead of 5. |
| 16:06.32 | sudhir492 | what does that mean |
| 16:07.14 | sudhir492 | tzanger: I joined in late. which company are you talking about? |
| 16:07.26 | uzd | he's talking about broadvoice. |
| 16:07.28 | christo | I have a dialplan with a mish-mash of different users. Some have 0870 numbers redirected to SIP phones, some just have SIP phones, some have SIP phones redirected on timeout to mobiles, others have SIP phones dropping to voicemail. There are about 30 users in all, all with different things. Is there any way to put a database behind this config and allow it to be managed thru a web console? |
| 16:07.58 | tzanger | uzd: you're part of the quiet minority then I think :-) |
| 16:08.02 | tzanger | uzd: if it works for you, great |
| 16:08.11 | tzanger | if not, well there are many others in that choir :-) |
| 16:08.23 | uzd | oh and tza, 1/3 a cent difference? they offer $25 unlimited calls.. nobody else does anything close to that. |
| 16:08.25 | *** join/#asterisk Godsey (lanny@goofball.md5.com) |
| 16:08.34 | tzanger | uzd: $25/mo is not unlimited |
| 16:08.45 | uzd | ? |
| 16:08.47 | tzanger | uzd: try pushing 4000 minutes a month through them and find out how unlimited 'unlimited' really is |
| 16:08.56 | Godsey | now that cisco is buying sipura, there is probably no chance of getting sipura supporting iax right? :) |
| 16:09.15 | tzanger | Godsey: no, but sipura will support skinny now :-) |
| 16:09.28 | uzd | tzan, I think I've already used about 3000 this month |
| 16:09.31 | *** join/#asterisk shido6 (~shido@d57-87-253.home.cgocable.net) |
| 16:09.54 | Godsey | I got here late, unlimited on what network? :) |
| 16:09.57 | tzanger | uzd: I'd be VERY surprised if they let you keep this up :-) |
| 16:10.17 | uzd | well.. no contract so if they don't, i bolt. |
| 16:10.39 | Sander4000 | can someone help me with a quadbri with bristuff ? |
| 16:10.41 | Godsey | uzd: what company? |
| 16:10.43 | shido6 | eruocompton? |
| 16:10.47 | shido6 | eurocompton |
| 16:10.53 | shido6 | that sounds funny |
| 16:10.58 | shido6 | like eurohood |
| 16:11.03 | Godsey | we're thinking of doing long distance for our customers :) |
| 16:11.08 | sudhir492 | tzanger: Any VoIP carrier will go bust if they really started giving "unlimited" unlimited. IMO, unlimited is just a euphemism for couple of thousand minutes |
| 16:11.15 | Godsey | we have unlimited long distance on our voice pris |
| 16:11.38 | sudhir492 | In that sense, it can even be called misleading |
| 16:11.55 | Godsey | not really, every industry does it |
| 16:12.02 | shido6 | "unlimited" is a brainwashing technique used by telcos to suck ppl into thinking they arent going to get fucked over |
| 16:12.12 | tzanger | Godsey: ? |
| 16:12.27 | Godsey | uses unlimited with exceptions :) |
| 16:12.36 | Gand_DJ | That's when you read the TOS, and if it don't give a limit... you sue :) |
| 16:12.36 | tzanger | unlimited long distance on your PRI? as in "unlimited calls at $x/min |
| 16:12.42 | shido6 | when in truth all "unlimted" is saying is "elbows on the table, sir." |
| 16:12.45 | newl | There's always fair use, AUP, ToS fine print. :) |
| 16:12.51 | SuPrSluG | uzd:they |
| 16:12.52 | sudhir492 | They all base unlimited on some statistical usage. |
| 16:12.53 | Godsey | tzanger: no, unlimited calls, no fee for call minutes |
| 16:12.55 | SuPrSluG | uzd:they |
| 16:12.58 | SuPrSluG | uzd:they |
| 16:13.11 | SuPrSluG | uzd:they |
| 16:13.22 | SuPrSluG | uzd:they |
| 16:13.24 | Godsey | we've used them in the past as modem call back lines |
| 16:13.29 | shido6 | if its true unlimited, then open up the flood gates and see what they say after a month |
| 16:13.42 | Godsey | it's been this way since 97 :) |
| 16:13.47 | Godsey | AT&T has always offered it |
| 16:14.12 | Godsey | it's $1800/pri |
| 16:14.46 | SuPrSluG | uzd:they're right. last week alone here many a bv user. i use nufone and have never had an issue other than the initial setup. :-) |
| 16:14.46 | Godsey | where our dial in only pris are around 271 |
| 16:14.57 | sudhir492 | Godsey, you are getting screwed if you are paying $1800 for any PRI, even if it is unlimited :-) |
| 16:15.15 | sudhir492 | How many minutes do you use on that PRI? |
| 16:15.24 | SuPrSluG | seems i'm stuttering today!! |
| 16:15.27 | Godsey | about 300/month right now :) |
| 16:15.30 | Godsey | we're an isp |
| 16:15.37 | *** join/#asterisk FarrisG (~jrush@h-68-164-19-170.dllatx37.covad.net) |
| 16:15.37 | Godsey | we receive calls almost exclusivly |
| 16:16.13 | FarrisG | Are there any other decent Soft phones besides X-ten's stuff? |
| 16:16.40 | Godsey | 1800 including all tax if that matters :) |
| 16:17.00 | sudhir492 | On a very busy PRI, one is able to send typically 250,000 minutes per month |
| 16:17.37 | newl | FarrisG: firefly, kiax, FWD's client, linphone to name a few, all fully capable. |
| 16:17.42 | sudhir492 | Theoritically, one can send 4 times as much, but practically it is difficult to send more that quarter million minutes /mont |
| 16:18.26 | uzd | SuPrSluG ? |
| 16:18.52 | uzd | oh.. well regardless of other peoples experiences |
| 16:18.54 | uzd | i've had a great on |
| 16:18.56 | uzd | on |
| 16:18.58 | uzd | one |
| 16:18.59 | Godsey | I guess we could suppliment our pris with l3 or something for completing calls out at a higher rate for peak |
| 16:19.08 | Godsey | and utilize 100% of our cheap pri :) |
| 16:19.09 | tzanger | yeah incoming only PRIs are less expensive than two-way and LD-only PRIs are cheaper than free local PRIs |
| 16:19.26 | Godsey | they are ld-only |
| 16:19.35 | Godsey | we cheat tho :P |
| 16:19.48 | tzanger | ld-only won't be flat-fee then, at least none that I've ever seen |
| 16:20.03 | Godsey | we selected our local calling area as a small area where we have no customers :) |
| 16:20.44 | Godsey | sprint used to have it, not sure if they do now |
| 16:20.53 | Godsey | it was combined voice/data I think it was called ion |
| 16:22.28 | *** part/#asterisk jwitte (~jwitte_@port-212-202-101-206.static.qsc.de) |
| 16:23.13 | *** join/#asterisk outtolunc (outtolunc@adsl-66-218-53-170.dslextreme.com) |
| 16:26.41 | *** join/#asterisk lilwookie (~zoidmeste@modemcable215.87-81-70.mc.videotron.ca) |
| 16:26.57 | Gand_DJ | I got signed up with Selectcom for toll-free services. Hopefully these guys are good. :) |
| 16:28.55 | *** join/#asterisk PCadach (~paul@www.east.telecom.kz) |
| 16:32.24 | *** join/#asterisk Derkommissar (~alberto@66.64.215.7.nw.nuvox.net) |
| 16:33.16 | Derkommissar | is there a way, when the queue does calls the agents, for it to verify first if the agent wants the call. Like. Press 1 to accept this call or 2 to ignore. |
| 16:33.45 | *** join/#asterisk Moc____ (~mochouina@h66-201-214-109.gtconnect.net) |
| 16:34.30 | Sander4000 | can someone help me with a quadbri with bristuff ? |
| 16:37.35 | *** join/#asterisk oden (~oden@194-237-146-22.customer.telia.com) |
| 16:39.09 | shido6 | . |
| 16:39.19 | stoyan | 'ignorepat => 9' doesn't work for me :( I always loose the dial tone after I dial the first digit, even if it's '9'. Any ideas? |
| 16:40.52 | shido6 | dont need it |
| 16:40.57 | lilwookie | stoyan, ignorepat needs to be in the contect the dialing channel is in |
| 16:40.58 | shido6 | ${EXTEN:1} |
| 16:41.28 | shido6 | 9N1.,1,Dial(IAX2/user@peer/${EXTEN:1}) takes out the first digit |
| 16:42.04 | shido6 | exten => _9N1.,1,Dial(IAX2/user@peer/${EXTEN:1}) takes out the first digit |
| 16:42.04 | *** join/#asterisk NickOliveri (~reason731@c-67-165-19-168.hsd1.ct.comcast.net) |
| 16:42.06 | stoyan | shido - it works/dials correctly, only that I don't get dialtone after I dial 9 |
| 16:43.02 | stoyan | lilwookie: yes, it's in the context with the longdistance extentions. and this context is imported into the context my channel is in |
| 16:43.54 | lilwookie | stoyan, in my experience it needs to be in the context of the channel not in an Included context |
| 16:45.19 | shido6 | you want dialtone after 9? |
| 16:45.25 | shido6 | why not dial the complete number? |
| 16:45.41 | shido6 | you should get dialtone when u pick up the phone |
| 16:45.45 | shido6 | whats your setup? |
| 16:46.17 | shido6 | and what do you want ( after stating your setup , may want to paste your zapata.conf , zaptel.conf, extensions.conf to http://pastebin.ca) |
| 16:46.26 | shido6 | brb - my web is down |
| 16:47.51 | stoyan | shido6: it's not that big deal. I can live without a dialtone after the 9, but I was wondering why it doesn't work. |
| 16:48.29 | Gand_DJ | Heh, the phone number for the CRA is busy |
| 16:48.44 | Gand_DJ | appears all pri's are in use |
| 16:48.46 | Gand_DJ | lol |
| 16:49.58 | Gand_DJ | just got in... their hold music is choppy as hell |
| 16:50.11 | Gand_DJ | Canada Revenue Agency needs better pbx equipment...lol |
| 16:51.09 | Syncros | they prefer to spend money on golf bals |
| 16:53.52 | pigpen | You know...I think I am having dtmf issues with my asterisk...when I call in with certian phones...it doesn't recognize the dtmf correctly. I have a incomming PRI on a digium card... |
| 16:58.09 | *** join/#asterisk McUnixJr (~mcmer@McUnixJr.gold.supporter.pdpc) |
| 17:01.12 | *** join/#asterisk shido6 (~greg@d57-87-253.home.cgocable.net) |
| 17:03.50 | stoyan | shido6: http://pastebin.ca/10572 |
| 17:04.03 | stoyan | shido6: it's for the ignorepat thingy :) |
| 17:08.47 | shido6 | do you have any zaptel gear? |
| 17:08.57 | shido6 | or a NuFone account for outbound or a inbound 8xx? |
| 17:10.51 | shido6 | http://pastebin.ca/10573 |
| 17:11.00 | shido6 | stoyan ? |
| 17:12.05 | stoyan | shido6: I have a x100p and iax2 connections for outbound calls |
| 17:12.15 | shido6 | great |
| 17:12.18 | shido6 | show me zaptel.conf |
| 17:12.20 | shido6 | and zapata.conf |
| 17:12.23 | stoyan | all of them work perfect except that i loose the dialtone after '9' |
| 17:12.30 | stoyan | just a sec |
| 17:12.32 | shido6 | you're supposed to |
| 17:12.41 | shido6 | so finish dialing |
| 17:12.55 | shido6 | thats what your dialplan tells it to do |
| 17:12.59 | stoyan | yes, but what is ignorepat used for then |
| 17:13.01 | shido6 | its waiting for the rest of the numbers |
| 17:13.06 | shido6 | its deprecated |
| 17:13.14 | *** join/#asterisk riksta (rick@212.85.228.176) |
| 17:13.29 | stoyan | so I shouldn't be using ignorepat? |
| 17:13.43 | shido6 | do you WANT to still hear a dialtone after you press 9? |
| 17:13.48 | stoyan | yes |
| 17:13.52 | shido6 | or do you want to make the call after you finish dialing the numbers |
| 17:14.04 | shido6 | so if the number is 555-1212 you dial 9555-1212 |
| 17:14.07 | shido6 | and the call goes out |
| 17:14.18 | stoyan | yes |
| 17:15.12 | *** join/#asterisk gambolputty (~gambolput@cblmdm65-221-51-80.buckeye-express.com) |
| 17:17.13 | stoyan | shido6: zapata and zptel configs uploaded [channels] |
| 17:17.13 | stoyan | language=en |
| 17:17.13 | stoyan | context=sbc |
| 17:17.13 | stoyan | switchtype=national |
| 17:17.13 | stoyan | signalling=fxs_ks |
| 17:17.13 | stoyan | usecallerid=yes |
| 17:17.15 | stoyan | hidecallerid=no |
| 17:17.17 | stoyan | callwaiting=yes |
| 17:17.19 | stoyan | usecallingpres=yes |
| 17:17.21 | stoyan | callwaitingcallerid=yes |
| 17:17.23 | stoyan | threewaycalling=yes |
| 17:17.25 | stoyan | transfer=yes |
| 17:17.27 | stoyan | cancallforward=yes |
| 17:17.29 | stoyan | callreturn=yes |
| 17:17.31 | stoyan | mailbox=18585862133 |
| 17:17.33 | stoyan | echocancel=yes |
| 17:17.35 | stoyan | echocancelwhenbridged=yes |
| 17:17.37 | stoyan | rxgain=0.0 |
| 17:17.39 | stoyan | txgain=0.0 |
| 17:17.41 | stoyan | group=1 |
| 17:17.43 | stoyan | callgroup=1 |
| 17:17.45 | stoyan | pickupgroup=1 |
| 17:17.47 | stoyan | immediate=no |
| 17:17.49 | stoyan | callerid=asreceived |
| 17:17.51 | stoyan | busydetect=yes |
| 17:17.53 | stoyan | busycount=4 |
| 17:17.55 | stoyan | channel => 1 |
| 17:17.57 | stoyan | ooops |
| 17:17.59 | stoyan | sorry |
| 17:18.01 | stoyan | shido6 : http://pastebin.ca/10574 |
| 17:18.20 | MikeJ[Laptop] | ~patebin is your friend |
| 17:18.21 | jbot | MikeJ[Laptop]: okay |
| 17:18.27 | MikeJ[Laptop] | oops |
| 17:18.32 | iq | what happened :O |
| 17:19.02 | MikeJ[Laptop] | ~pastebin is a place to paste your configs without flooding the channel at http://pastebin.ca |
| 17:19.03 | jbot | ...but pastebin is already something else... |
| 17:19.18 | MikeJ[Laptop] | ~pastebin |
| 17:19.19 | jbot | extra, extra, read all about it, pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca |
| 17:19.34 | MikeJ[Laptop] | hehe... patebin, oops |
| 17:26.41 | *** join/#asterisk dpryo (hn@donatello.nesland.net) |
| 17:32.52 | *** join/#asterisk MasterYoda (~mnicholso@207.111.174.1) |
| 17:34.09 | mishehu | hmm... why would ztmonitor be complaining about /dev/dsp ? |
| 17:34.32 | *** join/#asterisk MinorKing (~nschmidt@67.154.228.132) |
| 17:34.46 | mishehu | I don't have such a device, and have prevented the oss module from loading in asterisk |
| 17:34.56 | denon | so has anyone rigged up a cdma device or a bluetooth device to tie asterisk to a cell network? |
| 17:34.57 | MinorKing | Can anyone recomend a decent GUI that can be used for things like creating extentions and basic management features? |
| 17:35.07 | denon | I know there was some gsm stuff out there, havent heard much talk of cdma though |
| 17:35.12 | *** join/#asterisk guyee (~izomtriko@nextra.nudli.equitas.hu) |
| 17:35.43 | iq | Okay, the Asterisk I'm trying to connect does not support any of the coded supported by my ATA. What would happen? No Audio ? |
| 17:36.12 | guyee | hi, can NE1 tell me what 'regexten' is used for? |
| 17:36.33 | iq | guyee, what is 'NE1' ? |
| 17:36.38 | guyee | anyone :) |
| 17:36.43 | *** join/#asterisk systest (~systest@c-66-30-196-67.hsd1.ma.comcast.net) |
| 17:37.02 | *** join/#asterisk trash0r (trasher@dsl-084-058-030-210.arcor-ip.net) |
| 17:37.15 | trash0r | hi ;) |
| 17:39.26 | *** part/#asterisk santiago (~santiago@63.245.86.199) |
| 17:39.50 | iq | hi |
| 17:41.11 | *** join/#asterisk PTG123 (~PTG123@ip68-106-24-139.ph.ph.cox.net) |
| 17:42.01 | trash0r | can anybody check this please.. http://rafb.net/paste/results/pyUqvC99.html - i can't get it working :/ |
| 17:42.15 | *** join/#asterisk cianhughes (~cian@g5.cian.ws) |
| 17:44.44 | shido6 | no audio |
| 17:44.46 | shido6 | is nat |
| 17:45.05 | trash0r | ? |
| 17:45.31 | shido6 | trash0r |
| 17:45.32 | shido6 | um |
| 17:45.41 | shido6 | specify a codec in the general stanza of your sip.conf |
| 17:45.45 | shido6 | not just allow=all |
| 17:45.50 | shido6 | disallow=all |
| 17:45.52 | shido6 | allow=ulaw |
| 17:45.54 | shido6 | reload |
| 17:46.59 | trash0r | ok i'll try |
| 17:47.39 | trash0r | still the same problem |
| 17:47.45 | jief- | hey guys. my Gnet sip phones aren't registring with *. I posted my sip.conf and extensions.conf file here: http://www.pastebin.com/277616 Im very new to this whole thing. and i have no idea what to do now. when i try to ring an extension, after a while i get a busy signal |
| 17:47.58 | jief- | also, in the phone 'logs', i always have time outs |
| 17:48.24 | trash0r | it's ringing, but as soon as i accept the call it disconnects :/ |
| 17:48.34 | *** join/#asterisk Elshar (~Elshar@I.wants.your.cheekan.org) |
| 17:48.40 | shido6 | help is on the way |
| 17:48.58 | *** part/#asterisk critch (critch@steven.basesys.com) |
| 17:48.59 | shido6 | anothew newbie mistake |
| 17:49.32 | trash0r | shido6: btw, phoner shows "protocol error, layer 2" on disconnect |
| 17:51.01 | *** part/#asterisk MasterYoda (~mnicholso@207.111.174.1) |
| 17:51.55 | shido6 | stop using friends |
| 17:52.02 | shido6 | a friend is supposed to be a user and peer |
| 17:52.08 | shido6 | break the friend out to a user |
| 17:52.10 | shido6 | and a peer |
| 17:53.30 | *** part/#asterisk systest (~systest@c-66-30-196-67.hsd1.ma.comcast.net) |
| 17:56.29 | trash0r | shido6: are you talking to me? |
| 17:56.40 | *** join/#asterisk jackfiber (cico@82.99.197.169) |
| 17:57.40 | SuPrSluG | jief:http://www.pastebin.com/277622 |
| 17:57.49 | jackfiber | hey I got real trouble with grandstream handytone, while handytone is behind the nat and * is on the Internet, anyone has set this successfully? |
| 17:58.28 | shido6 | hold on |
| 17:58.35 | *** join/#asterisk pussfeller (~todd@t1-rtc-woodlawn.rtcol.com) |
| 17:59.55 | shido6 | where was I |
| 17:59.58 | shido6 | oh trash0r |
| 18:00.14 | jief- | SuPrSluG: thanks, im gonna try this |
| 18:00.31 | trash0r | shido6: mh? |
| 18:00.35 | shido6 | trash0r you have no password set for your phones? |
| 18:01.08 | trash0r | hmm.. isnt it the "secret" thingie? |
| 18:01.16 | shido6 | yes but for 501 |
| 18:01.19 | shido6 | you have no "Secret" |
| 18:01.25 | shido6 | secret=password |
| 18:01.34 | trash0r | eh |
| 18:01.34 | shido6 | and the password is what you should set for your individual phones |
| 18:01.42 | shido6 | and it shouldnt be the username |
| 18:01.43 | *** join/#asterisk Ayano (~erik_leee@ppp-70-244-234-29.dsl.spfdmo.swbell.net) |
| 18:01.44 | trash0r | i am not jief- |
| 18:01.45 | trash0r | ,p |
| 18:01.51 | shido6 | heh |
| 18:01.53 | shido6 | whoop |
| 18:01.54 | shido6 | s |
| 18:03.14 | shido6 | trash0r show me your new sip.conf |
| 18:03.23 | shido6 | do a sip debug at the CLI |
| 18:03.23 | bkw_ | LIES LIES LIES |
| 18:03.25 | bkw_ | ALLL LIES |
| 18:03.26 | shido6 | and a sip show peers |
| 18:03.56 | shido6 | not sms is it, bkw? ;) |
| 18:04.19 | trash0r | 11/11 84.58.30.210 D N 255.255.255.255 1743 OK (150 ms) |
| 18:04.19 | trash0r | 10/10 84.58.30.210 D N 255.255.255.255 5060 OK (197 ms) |
| 18:04.25 | zoa | hehe |
| 18:04.28 | zoa | brian |
| 18:04.31 | zoa | send em my greetings |
| 18:04.36 | zoa | in the email |
| 18:04.44 | bkw_ | 4988 active channel(s) |
| 18:04.55 | shido6 | really... |
| 18:04.56 | jackfiber | anyone has Handytone or any other grandstream phones? |
| 18:05.03 | bkw_ | them dumb bastards |
| 18:05.04 | zoa | yes i do |
| 18:05.05 | bkw_ | I swear |
| 18:05.06 | bkw_ | TO GOD |
| 18:05.08 | shido6 | first ip phone was a budgetone, jackfiber |
| 18:05.08 | zoa | :) |
| 18:05.10 | bkw_ | I WANNA KILL |
| 18:05.13 | zoa | its really stupid |
| 18:05.15 | zoa | told ya so |
| 18:05.23 | trash0r | shido6: http://rafb.net/paste/results/UrYK7J54.html <- this is my config now |
| 18:05.25 | zoa | just didnt have an hour to test it |
| 18:05.40 | zoa | but got mad when i saw the last thing on the mailinglist |
| 18:05.52 | jackfiber | shido6, I wanna to get help on grandstream doesn't matter which product because all are somewhat similar |
| 18:06.54 | nwhit | does anyone here use cisco phones? |
| 18:07.43 | nwhit | i am having problems with them hanging up the person after a certain amount of time if you put a person on hold |
| 18:08.02 | nwhit | and they hang up after some time if the other side puts the person on hold |
| 18:08.08 | pigpen | You know...I think I am having dtmf issues with my asterisk...when I call in with certian phones...it doesn't recognize the dtmf correctly. I have a incomming PRI on a digium card... |
| 18:08.34 | jackfiber | anyone with grandstream phone? |
| 18:09.15 | jackfiber | SHIDO6, do u have budge tone? |
| 18:09.57 | SuPrSluG | jackfiber:i have 1 |
| 18:10.12 | SuPrSluG | jackfiber:a 102 |
| 18:10.15 | zoa | jackfiber: read this: http://www.asteriskguru.com/natut.php |
| 18:10.25 | trash0r | shido6: the debug shows some stuff now: http://rafb.net/paste/results/XlskgO67.html |
| 18:10.27 | zoa | read on stun |
| 18:10.30 | zoa | and the sip.conf options |
| 18:11.13 | jief- | SuPrSluG: i tried your config, i still get a busy signal and ring on the other phone |
| 18:12.57 | SuPrSluG | jief:do they register now? |
| 18:13.32 | SuPrSluG | jief:CLI>sip show peers? |
| 18:14.07 | convey | anyone have experience with robbed bit T1 connections? (NON PRI) |
| 18:14.19 | SuPrSluG | jief:CLI>did u reload? |
| 18:14.20 | jackfiber | THANKS ALOT ZOA |
| 18:14.20 | shido6 | http://rafb.net/paste/results/pNeMYT94.html |
| 18:14.28 | shido6 | trash0r http://rafb.net/paste/results/pNeMYT94.html |
| 18:14.50 | Corydon-w | convey: yeah, they're pretty much the same as connections to channel banks |
| 18:14.56 | jackfiber | ZOA, seems my problem is the Handutone (phone) is behind a symmetric nat so NAT=route can help |
| 18:15.12 | convey | Corydon-w: what signalling do you use? |
| 18:15.25 | shido6 | next |
| 18:15.26 | jief- | SuPrSluG: mind if i paste something in pv? |
| 18:15.28 | Corydon-w | fxs_ks or fxo_ks, depending upon the side you're on |
| 18:15.29 | *** join/#asterisk likwid-- (likwid@nc-69-34-145-174.dyn.sprint-hsd.net) |
| 18:15.29 | convey | Corydon-w: I am trying fxsks and it is haning the channels |
| 18:15.31 | shido6 | jackfiber, whats wrong |
| 18:15.32 | shido6 | ? |
| 18:15.41 | convey | Corydon-w: Hanging |
| 18:15.47 | SuPrSluG | jief:ok |
| 18:15.56 | Corydon-w | convey: are you sure the telco isn't using some other signalling, such as e&m? |
| 18:16.28 | convey | Corydon-w: I am connecting zap channels to a T1 card on my phone system. Using Asterisk as a Voip -> TDM gateway. |
| 18:16.45 | Corydon-w | convey: for that matter, is the line even esf/b8zs or is it d4/ami? |
| 18:17.04 | convey | Corydon-w: it is esf/b8zs |
| 18:17.13 | outtolunc | i think convey needs to start over, slowly <G> |
| 18:17.24 | Corydon-w | convey: well, you need to ask your telco what signalling they're using |
| 18:18.00 | Corydon-w | You could try trial and error, but I wouldn't recommend it |
| 18:18.34 | Corydon-w | convey: also, does your zaptel.conf match what's in your zapata.conf ? |
| 18:18.34 | convey | Corydon-w: I am connecting my asterisk box into my PBX. I am using asterisk as the carrier. |
| 18:18.42 | cianhughes | anyone using asterisk with an ISDN Bri, just need to do a simple PBX setup & am not sure which ISDN PCI card to buy |
| 18:19.01 | Corydon-w | convey: then check your configs |
| 18:19.06 | jackfiber | shido6> I got a HT behind nat and asterisk on Internet, what option do u use to make valid connection anything I use has an issue in example NAT=yes couse no audio at one line no nat causes no registration port forwarding causes no nAT and no audio |
| 18:19.07 | *** join/#asterisk Blissex (~Blissex@82-69-39-138.dsl.in-addr.zen.co.uk) |
| 18:19.41 | *** join/#asterisk Lee__ (~lee@ool-18bb881a.dyn.optonline.net) |
| 18:19.41 | Nivex | jackfiber: you may need reinvite=no and canreinvite=no |
| 18:19.56 | jackfiber | I set both |
| 18:19.58 | Nivex | jackfiber: to keep Asterisk in the loop instead of having it try to hand off |
| 18:20.02 | Nivex | jackfiber: oh, well phooey |
| 18:20.03 | Nivex | :) |
| 18:20.12 | jackfiber | I think one is valid (can reinvite) and reinvite has been removed from asterisk code |
| 18:22.03 | jackfiber | can anyone give me his handytone config that works for him? |
| 18:22.15 | jackfiber | or any grandstream budgetone? |
| 18:22.47 | shido6 | heh |
| 18:22.50 | jackfiber | I need: RTP port, NAt setting and portforwarding if applicable on grandstream side |
| 18:23.02 | shido6 | where is the gstream on your net with regards to your * box |
| 18:23.07 | shido6 | nat? public ip, ? |
| 18:23.29 | jackfiber | grandstream is behind NAT and * on public IP on public Internet |
| 18:24.54 | jackfiber | I need to know a working setting I got tired as I teste most possible settings and got one failure at a time sometimes audio sometimes no ring |
| 18:25.31 | jackfiber | r u using grandstream behind NAT with * on the public? |
| 18:25.51 | jackfiber | xlite works fine !!!!!! but not grandstream !!!!!! |
| 18:25.59 | shido6 | if xlite works |
| 18:26.08 | shido6 | then its just a setting you're screwing up in the grandstream |
| 18:26.09 | *** join/#asterisk obsidian-studios (~obsidian-@c-66-177-188-197.hsd1.fl.comcast.net) |
| 18:26.18 | obsidian-studios | greetings all |
| 18:26.25 | shido6 | let me go get mine and plug it in |
| 18:27.04 | jackfiber | THANK U VERY MUCH SHIDO6 U R GREAT |
| 18:28.14 | shido6 | this is what we do, jackfiber :) |
| 18:29.09 | obsidian-studios | still trying to get caller id working on analog phones via my cisco's ubr924 fxs ports. Seems I have two choices, SIP or SGCP/MGCP? I assume that caller id info is part of the sip protocol? I tried messing with mgcp but got funky results mainly on the routers end. Since router only supports SGCP? |
| 18:30.15 | shido6 | ok |
| 18:30.16 | bjohnson | obsidian-studios: CID can be sent through SIP .. how/if it works comes down to configuration |
| 18:30.28 | shido6 | I bet you used an outbound proxy in your settings, didnt you jackfiber |
| 18:30.34 | obsidian-studios | so far sip works great, but no caller id info. Also the analog phones do not stop ringing when say the call has rung to many times and sent to voice mail? |
| 18:30.39 | shido6 | asterisk isnt an outbound proxy to your phone |
| 18:30.42 | shido6 | its a sip server |
| 18:30.49 | shido6 | so leave outbound proxy blank |
| 18:31.01 | shido6 | fill in sip server, - crap u know what.... let me get a screenpaste for you |
| 18:31.04 | obsidian-studios | <PROTECTED> |
| 18:31.08 | jackfiber | shido6> NO |
| 18:31.22 | Qwell | jackfiber: drop the caps |
| 18:31.23 | jackfiber | xlite works with no nat set on * side |
| 18:31.41 | *** join/#asterisk los415 (~los415@64.201.104.186) |
| 18:31.41 | obsidian-studios | <PROTECTED> |
| 18:31.41 | shido6 | jackfiber, msg me with your settings (Caps is like SCREAMING) |
| 18:31.44 | *** join/#asterisk coppice (~chatzilla@60.195.17.210.dyn.pacific.net.hk) |
| 18:35.08 | bjohnson | obsidian-studios: I don't know cisco systems. CID will work with SIP |
| 18:35.08 | bjohnson | (in general) |
| 18:35.13 | *** join/#asterisk PaulTech (PaulTech@65.5.68.12) |
| 18:35.16 | PaulTech | Hello all |
| 18:35.31 | obsidian-studios | bjohnson: what I figured I think the IOS does not support it via sip |
| 18:35.51 | obsidian-studios | bjohnson: it's a eol platform, with limited resources so I can't use a 12.2 or 12.3 IOS |
| 18:35.52 | PaulTech | Anyone wanna make a few bucks giving me a hand |
| 18:36.13 | Qwell | PaulTech: ask away |
| 18:36.38 | obsidian-studios | bjohnson: took me a while to find a working 12.1 with sip? But I am thinking I might be better to try and use SGCP/MGCP. Anyone care to comment on those protocols? |
| 18:36.53 | Derkommissar | why does asterisk on the sip invites, sends a from "" "" with nothing else |
| 18:36.54 | Derkommissar | like |
| 18:36.56 | Derkommissar | INVITE sip:9543898047@207.218.174.141 SIP/2.0 |
| 18:36.56 | Derkommissar | Via: SIP/2.0/UDP 208.51.238.10:5060;branch=z9hG4bK5ec8e7c6 |
| 18:36.56 | Derkommissar | From: """" <sip:asterisk@208.51.238.10>;tag=as44c0e1e2 |
| 18:37.04 | PaulTech | I have a iax connection for outbound calls, I signed up for stanaphone.com for a free did (While I learn) |
| 18:37.06 | Derkommissar | how can i change the 4 " |
| 18:37.08 | Derkommissar | ??? |
| 18:37.17 | PaulTech | I run a datacenter in Orlando and setting up in Mexico to sell routes |
| 18:37.26 | PaulTech | Wondering if you have any ideas |
| 18:37.36 | PaulTech | I cant seem to get it to register |
| 18:37.43 | Qwell | Do they suck? |
| 18:37.49 | Qwell | if that, thats likely why |
| 18:37.53 | PaulTech | register=5166875548:xxxxxx@sip.stanaphone.com/5166875548 |
| 18:37.59 | Qwell | register => |
| 18:38.07 | Qwell | with spaces, and = should be => |
| 18:38.16 | PaulTech | Understood |
| 18:38.17 | PaulTech | Reloading |
| 18:38.27 | MinorKing | Can anyone recomend a decent GUI that can be used for things like creating extentions and basic management features? |
| 18:38.36 | Qwell | MinorKing: GUIs suck. :) |
| 18:38.38 | heath__ | if busydetect=no, and i call a busy number, will it just answer and send the busy tone from the telco? or will the call fail to go through? |
| 18:38.41 | MinorKing | Yes |
| 18:38.41 | MinorKing | i know |
| 18:38.51 | MinorKing | Unfortunatly I need one for those less technical then I |
| 18:38.52 | Qwell | MinorKing: They do more harm then good |
| 18:39.09 | MinorKing | I have the initial configuration but need to hand it off to others for just extention creation and dialplans |
| 18:39.47 | PaulTech | It doesnt seem to be registering.. |
| 18:39.55 | Qwell | PaulTech: Do you get errors? |
| 18:40.13 | PaulTech | Name/username Host Dyn Nat ACL Mask Port Status |
| 18:40.13 | PaulTech | stanaphone/5166 204.147.183.18 255.255.255.255 5060 Unmonitored |
| 18:40.21 | PaulTech | Thats the problem Qwell I dont |
| 18:40.25 | PaulTech | I have debug and verbose at 5 |
| 18:40.30 | Qwell | PaulTech: sip show registry |
| 18:40.32 | PaulTech | No output on console |
| 18:40.40 | PaulTech | Nothing |
| 18:40.49 | *** join/#asterisk McUnixJr (~mcmer@McUnixJr.gold.supporter.pdpc) |
| 18:41.54 | PaulTech | <PROTECTED> |
| 18:41.54 | PaulTech | <PROTECTED> |
| 18:41.54 | PaulTech | <PROTECTED> |
| 18:42.02 | PaulTech | Only thing I get on SIP |
| 18:42.28 | pigpen | ok..here is some more info...when I call in using some phones, and I go to type in an extension like...8013, * is only seeing 8 or 80 ...ideas? |
| 18:42.59 | Qwell | pigpen: digittimeout set to something rediculously low? |
| 18:43.15 | pigpen | I have no clue...where would I look or set this feature? |
| 18:43.15 | Qwell | PaulTech: Where in sip.conf are you putting the register line? |
| 18:43.19 | Qwell | It needs to go in [general] |
| 18:43.26 | PaulTech | I take it from-sip-external |
| 18:43.30 | PaulTech | Should be uncommented |
| 18:43.30 | pigpen | in the extentions.conf? |
| 18:43.38 | PaulTech | yes Qwell |
| 18:44.08 | PaulTech | port = 5060 ; Port to bind to (SIP is 5060) |
| 18:44.08 | PaulTech | bindaddr = 0.0.0.0 ; Address to bind to (all addresses on machine) |
| 18:44.08 | PaulTech | disallow=all |
| 18:44.08 | PaulTech | allow=ulaw |
| 18:44.08 | PaulTech | allow=alaw |
| 18:44.08 | PaulTech | context = stanaphone ; Send unknown SIP callers to this context |
| 18:44.10 | PaulTech | callerid = Unknown |
| 18:44.12 | PaulTech | register => 5166875548:xxxxxx@sip.stanaphone.com/5166875548 |
| 18:44.16 | Qwell | ~pastebin |
| 18:44.17 | jbot | extra, extra, read all about it, pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca |
| 18:44.44 | pigpen | ohoh...you make it man... |
| 18:44.49 | pigpen | s/man/mad |
| 18:45.04 | PaulTech | Sorry Qwell |
| 18:45.56 | pigpen | ok..so does the digittimeout go into the extentions.conf? |
| 18:46.14 | Qwell | pigpen: yeah |
| 18:46.18 | pigpen | k |
| 18:46.42 | PaulTech | Ok |
| 18:46.42 | PaulTech | Warning: 392 sip.stanaphone.com:5060 "Noisy feedback tells: pid=33036 req_src_ip=72.29.65.96 req_src_port=5060 in_uri=sip:sip.stanaphone.com out_uri=sip:sip.stanaphone.com via_cnt==1" |
| 18:46.47 | PaulTech | Is that too bad? |
| 18:47.06 | Qwell | dunno, doesn't look like it |
| 18:47.08 | PaulTech | Hmm looks good |
| 18:47.08 | Qwell | its only a warning |
| 18:47.35 | PaulTech | Woot |
| 18:47.54 | Qwell | working? |
| 18:48.16 | PaulTech | Yep, I owe ya a beer Qwell |
| 18:48.18 | Qwell | slePP: You ever get any good passwords in there? :p |
| 18:48.35 | slePP | Qwell: i had a cease & desist notice over one set of passwords.. heh |
| 18:48.39 | Qwell | haha |
| 18:48.43 | Qwell | erm, that sucks |
| 18:48.47 | PaulTech | Tell me if you ever need any type of hosting or did's |
| 18:48.55 | slePP | someone pasted about 1500 compromised passwords to some website, got a notice 'to remove it immediately' attached to the cease & desist |
| 18:48.59 | Qwell | PaulTech: I'm pretty much set there |
| 18:49.13 | Qwell | slePP: nice |
| 18:49.46 | slePP | i'm truly surprised 4 of 13 people don't want to see another draw |
| 18:52.21 | *** join/#asterisk L|NUX (~linux@202.5.145.58) |
| 18:52.46 | PaulTech | Now to figure out queues |
| 18:54.27 | *** part/#asterisk obsidian-studios (~obsidian-@c-66-177-188-197.hsd1.fl.comcast.net) |
| 18:54.51 | PaulTech | Hmmm |
| 18:55.12 | PaulTech | As soon as someone calls it should goto queue 1 and play music till someone picks up |
| 18:55.38 | Ayano | I have a asterisk@home installation and I configured a tdm400p with a trunk on 4. It is still not picking up the calls. Any suggestions? |
| 18:56.55 | PaulTech | Apr 29 14:59:14 WARNING[17882]: Unknown keyword in queue '1': agentannounce at line 15 of queue.conf |
| 18:58.54 | PaulTech | I dont even have a queue.conf |
| 19:00.15 | *** part/#asterisk Gand_DJ (~gandalf@ptr-207-54-104-24.ptr.terago.ca) |
| 19:00.41 | *** join/#asterisk lilwookie (~zoidmeste@modemcable215.87-81-70.mc.videotron.ca) |
| 19:01.58 | *** join/#asterisk jesster (jesster@jesster.org) |
| 19:02.37 | *** join/#asterisk shaon (~shaon@61.68.13.184) |
| 19:03.44 | shaon | can anybody help me setting up an asterisk ivr? |
| 19:03.50 | shaon | pls |
| 19:04.21 | Ayano | I have a asterisk@home installation and I configured a tdm400p with a trunk on 4. It is still not picking up the calls. Any suggestions? |
| 19:04.48 | Ayano | shaon: did you check the examples on the wiki? |
| 19:05.29 | shaon | yes but my issue is bit different than sample wiki |
| 19:05.42 | MikeJ[Laptop] | what? |
| 19:05.47 | pigpen | DigitTimeout(3)...shit..that is short! |
| 19:05.58 | SuPrSluG | shaon:what r u trying to do |
| 19:06.27 | shaon | Ayano !!! my suggestion is to use your asterisk on linux thus u will get more control |
| 19:06.59 | *** join/#asterisk Blackvel (~blackvel@dsl-082-083-173-150.arcor-ip.net) |
| 19:07.17 | Ayano | I'm using linux. I just used asterisk@home because it configures a whole bunch of stuff for you. |
| 19:07.40 | pigpen | what is this asterisk@home....? I keep seeing it... |
| 19:07.58 | PaulTech | Google Says ! http://asteriskathome.sourceforge.net/ |
| 19:08.00 | pigpen | at first I thought is was devs giving home users hell.... |
| 19:08.06 | PaulTech | It's a web GUI |
| 19:08.21 | pigpen | err...no thanks. |
| 19:08.28 | Ayano | it is a cd that you can download that loads centos, and asterisk, and a few web guis in one shot. |
| 19:08.42 | shaon | i want to receive a call from pstn (zap/1) then it will say a menu then it can dial any extension and disconnect the call but aster conversation it will come back to the main menu should not disconnect |
| 19:08.47 | pigpen | neat idea....but...hmm... |
| 19:10.25 | SuPrSluG | shaon:use goto(s,(whatever priority) |
| 19:11.07 | SuPrSluG | shaon:in the menu |
| 19:11.45 | shaon | SuPrSlug: but when the call is finished zap/1 hangs up |
| 19:12.10 | SuPrSluG | shaon:put the goto before hangup |
| 19:12.35 | shaon | i was thinking to use macro |
| 19:13.14 | sivana | anyone here use Xlite? |
| 19:13.35 | shaon | because i have many sip number and gsm number to dial from PSTN |
| 19:13.57 | shaon | through asterisk |
| 19:16.35 | shaon | *AsTeRiSk GURU* help !!!!!!!!!! |
| 19:21.21 | *** part/#asterisk jackfiber (cico@82.99.197.169) |
| 19:21.39 | shaon | SuPrSluG r u there? |
| 19:21.55 | SuPrSluG | yeah |
| 19:22.05 | shaon | any idea? |
| 19:22.21 | file | neeeeeed fooooooood |
| 19:22.25 | *** part/#asterisk gpearson (~Graham@lrt2.niesc.k12.in.us) |
| 19:25.19 | SuPrSluG | shaon:pastebin ur menu |
| 19:25.49 | *** join/#asterisk MasterYoda (~mnicholso@207.111.174.1) |
| 19:26.14 | MasterYoda | how big is a min or a sec of gsm data |
| 19:26.18 | MasterYoda | also for wav |
| 19:27.28 | Lee__ | gsm is aprox 8kbps |
| 19:27.36 | *** join/#asterisk fosco (fosco@hellfire.frontier.fr) |
| 19:27.38 | fosco | hi |
| 19:27.39 | Lee__ | wav is 64 |
| 19:27.47 | bjohnson | PaulTech: it includes a web gui called AMP.. @ home is an install cd that includes a bunch of software including the Centos Linux distro, asterisk, and AMP |
| 19:27.48 | coppice | GSM is 13.2k bots per second |
| 19:27.54 | PaulTech | Yea bjohnson |
| 19:27.57 | coppice | s/bots/bits |
| 19:27.59 | PaulTech | I have all 3 |
| 19:28.01 | fosco | anyone with a digium card? (TE410P) ? |
| 19:28.01 | Lee__ | sorry, my bad |
| 19:28.04 | PaulTech | without the cd :) |
| 19:28.10 | Lee__ | I guess only speex and g729 can get down to 8 |
| 19:28.16 | PaulTech | and I know how each part was setup so I did this um learning thing :D |
| 19:28.27 | *** join/#asterisk juiceib269 (~juiceib26@out.empireind.com) |
| 19:28.39 | coppice | Lee__ its not you day, is it :-) |
| 19:28.55 | Lee__ | no, it isn't. having problems with AMPs default Meetme rooms |
| 19:29.01 | coppice | G.723.1 is 5 or 6kbps. |
| 19:29.21 | PaulTech | Now if only I could figure out how to force all sip calls into 'Music on Hold' then ring extension iax2/202 |
| 19:29.24 | Lee__ | coppice: woah. is g723 patented? |
| 19:29.27 | bjohnson | shaon: use the special hangup exten |
| 19:29.34 | *** join/#asterisk shaon (~shaon@61.68.13.184) |
| 19:29.54 | coppice | you can go lower, but no popular VoIP codec currently does. AMR might catch on, though. That can go down to 4k bps. |
| 19:30.03 | shaon | Dial plan help!!!!!!!!!!!!! |
| 19:30.08 | bjohnson | shaon: use the special hangup exten |
| 19:30.14 | Lee__ | I have a lot of hope for speex but this dude on the mailing list says it eats CPU |
| 19:30.33 | Lee__ | I haven't been able to try it yet |
| 19:30.34 | bjohnson | Lee__: the more compression, the more cpu needed |
| 19:30.40 | coppice | g.723.1 is patented. Almost any decent codec is smothered in patents. I'm amazed speex managed to steer around them |
| 19:30.42 | bjohnson | general rule |
| 19:30.52 | Lee__ | sure but each algorithm does it differently |
| 19:30.54 | shaon | i am very new can u please give me more detail? |
| 19:31.03 | coppice | speex takes about as much CPU as G.729 |
| 19:31.06 | CoaxD | g.729 is bad with a lossy link |
| 19:31.13 | CoaxD | whereas gsm does fine with that |
| 19:31.18 | SuPrSluG | bjohnson:like h,1,goto(whatever?) |
| 19:31.21 | Lee__ | the ogg project is all about reinventing patented codecs |
| 19:31.23 | akl- | you suffer quality-wise with gsm, though |
| 19:31.33 | bjohnson | shaon: follow one of the billion examples to make an IVR. Also use the hangup extension to catch when a person hangs up and send them to the start of the IVR again |
| 19:31.38 | Lee__ | akl-: I know, I'm using it on an IAX trunk |
| 19:31.49 | bjohnson | SuPrSluG: yes |
| 19:31.57 | *** part/#asterisk MasterYoda (~mnicholso@207.111.174.1) |
| 19:32.01 | PaulTech | Why dont my queues work ! lol |
| 19:32.09 | SuPrSluG | shaon:like h,1,goto(whatever?) |
| 19:32.15 | bjohnson | SuPrSluG: not certain how well it will die if the pstn caller hangs up .. needs to be tested |
| 19:32.21 | Lee__ | coppice: who's the distributor of g723 licences? |
| 19:32.46 | bjohnson | I don't think there is one |
| 19:32.52 | bjohnson | distributor of g723 licences |
| 19:32.53 | coppice | licencing G.723.1 is a huge pain. there is no low cost route for small scale users |
| 19:33.04 | SuPrSluG | bjohnson:well if pstn caller hangsup it's over ain't it? |
| 19:33.20 | shaon | but i have 5 sip extensions do i need to specify all 5 extension rule in incomming context? |
| 19:33.26 | Lee__ | where do you go to start licensing it? |
| 19:33.43 | bjohnson | SuPrSluG: not certain since you're going to an ivr .. I guess make sure there is a timeout in the ivr |
| 19:33.43 | SuPrSluG | bjohnson:no need to send an internal # to the ivr, right? |
| 19:34.02 | bjohnson | SuPrSluG: most ivrs would be s extens |
| 19:34.05 | coppice | Lee__ you can start at the voiceage site |
| 19:34.28 | shaon | yes |
| 19:34.31 | bjohnson | Lee__: and tell them why they should bother tlking to you |
| 19:35.02 | bjohnson | shaon: do i need to specify all 5 extension rule in incomming context? Answer - only if you want to call them |
| 19:35.38 | *** join/#asterisk ikey1 (ikey@220.226.54.63) |
| 19:35.43 | SuPrSluG | bjohnson:it's shaon issue, but thanx for the special h, it's nice to know if a caller gets disconnected they can be put back from whence they came. lol |
| 19:36.10 | coppice | The farce with these patent licence package deals is they won't indemnify you again more patent holders crawling out of the woodwork |
| 19:36.18 | shaon | then it will be a big context can i use macro for this? then there will be no repetation of same lines |
| 19:36.25 | Lee__ | bjohnson: they only deal with multinationals? |
| 19:36.52 | coppice | even MS doesn't have a proper licence for G.723.1 :-) |
| 19:37.48 | shaon | bjohnson: thanks for h,1,foto(wahtever) suggestion |
| 19:37.53 | Lee__ | wow |
| 19:38.03 | Lee__ | huray for speex |
| 19:38.13 | bjohnson | shaon: yes. most people use the stdexten macro, define their local extens in a separate context, and include that context where they want access to those phones |
| 19:38.49 | shaon | i have another problem |
| 19:39.39 | shaon | bjohnson: i have a quicknet phonejack card as a consol |
| 19:40.12 | bjohnson | I can't help you there |
| 19:40.29 | bjohnson | don't have that hardware |
| 19:40.55 | shaon | when i try to dial _393. it only takes one digit after 393(one digit) and dial |
| 19:41.12 | bkw_ | shaon, what channe driver? |
| 19:41.25 | shaon | phone/phone0 |
| 19:41.27 | bkw_ | yep |
| 19:41.30 | bkw_ | it can't do wildcards |
| 19:41.34 | bkw_ | don't use them with chan_phone |
| 19:41.35 | bkw_ | NEXT!!! |
| 19:42.05 | Lee__ | shaon: I can recommend AMP for lots of good defaults, although installing and configuring AMP was one of the most painful tasks involving a computer I have ever done. |
| 19:42.14 | PaulTech | Anyone feel like helping with music-on-hold ? |
| 19:42.25 | PaulTech | AMP was stupidly easy to install |
| 19:42.37 | denon | bjohnson: wha? where? |
| 19:42.45 | denon | damn, he didnt even tell me he was getting those! |
| 19:42.54 | denon | sup b :) |
| 19:43.04 | PaulTech | bkw_, full log shows Starting Music on Hold, mpg123 is installed. Default is set as group and uncommented |
| 19:43.06 | PaulTech | Hit me |
| 19:43.43 | shaon | any idea where i can get a GSM gateway? |
| 19:44.36 | denon | shaon: http://www.voip-info.org/wiki-Asterisk+Connecting+to+the+Cellular+Network |
| 19:44.47 | SuPrSluG | shaon:google asterisk gsm gateway. it's blue something or other |
| 19:44.57 | bkw_ | PaulTech, READ |
| 19:44.58 | bkw_ | no clue |
| 19:45.05 | shaon | thnks a lot |
| 19:45.10 | denon | np |
| 19:45.12 | bkw_ | it hurts to think about such things |
| 19:45.18 | *** join/#asterisk adker (~adker@67-51-234-116.dsl1.glv.ny.frontiernet.net) |
| 19:45.27 | bkw_ | some days I wanna take asterisk and blow it up |
| 19:45.32 | bkw_ | others I make phone calls with it |
| 19:45.34 | denon | rm -rf /usr/src |
| 19:45.43 | bkw_ | haha |
| 19:45.45 | MikeJ[Laptop] | hehe |
| 19:45.47 | bkw_ | asterdrama |
| 19:45.56 | bkw_ | A new book by Spark Mencer |
| 19:45.56 | file | I need fooooooood |
| 19:46.09 | bkw_ | hahahahhahaha |
| 19:46.17 | bkw_ | I don't have CVS commit |
| 19:46.19 | file | I need tacos or I will explode |
| 19:46.26 | PaulTech | <PROTECTED> |
| 19:46.29 | denon | bkw: and you wonder why? :) |
| 19:46.35 | PaulTech | I wonder if my server needs sound installed |
| 19:46.49 | MikeJ[Laptop] | ummmmmm |
| 19:46.53 | CoaxD | PaulTech: Um, no |
| 19:46.58 | *** join/#asterisk viLeR (1000@ip-47-252.telesat.com.co) |
| 19:47.00 | denon | bkw: shopping for a new cell phone .. I hate this, cause the technology I want is always 6mo away |
| 19:47.01 | CoaxD | PaulTech: It just needs mpg123 0.59r, no other version |
| 19:47.02 | bkw_ | PaulTech, you're using voipjet thats your problem |
| 19:47.12 | CoaxD | bkw; dude. heh |
| 19:47.13 | bkw_ | denon, cellphone |
| 19:47.15 | bkw_ | nothing more |
| 19:47.17 | bkw_ | nothing less |
| 19:47.20 | bkw_ | don't get camera phones |
| 19:47.20 | PaulTech | bkw_, doesnt allow for that kinda bitrate ? |
| 19:47.29 | bkw_ | PaulTech, No I just dislike voipjet |
| 19:47.32 | CoaxD | PaulTech: It doesn't have anything to do with voipjet. :P |
| 19:47.38 | denon | 'cause I can put a 1GB mmc in it |
| 19:47.41 | bkw_ | CoaxD, it might |
| 19:47.43 | denon | nfi why I want that .. |
| 19:47.47 | PaulTech | bkw_, I'll buy from anyone I dont like any them much :P |
| 19:47.49 | denon | but it's got an mp3 player and fm radio :) |
| 19:47.49 | CoaxD | PaulTech: bkw just dont like it because the owner of it came in here and started spamming his shit to everyone every 5 minutes |
| 19:47.49 | shaon | Does anybody know how to make asterisk support H323? i tried but endup with a million compiling errors. |
| 19:47.51 | PaulTech | I was just testing |
| 19:48.04 | Qwell | shameless plug in 3 |
| 19:48.05 | CoaxD | PaulTech: When he first got started in voip testing |
| 19:48.13 | PaulTech | Shady ? |
| 19:48.37 | CoaxD | PaulTech: Nah, not shady. just new. He also had some serious issues with his email dealiebob a while back - i helped him with those |
| 19:48.40 | denon | bkw: found out our regional carrier now does unlimited mobile to mobile .. which is cool, but now my plan has more minutes than I'll ever use .. so no need to rig up a phone to the asterisk server :\ |
| 19:48.57 | CoaxD | PaulTech: I.e. several spam tests he was failing due to being a moron, etc |
| 19:48.57 | *** join/#asterisk NK123 ([U2FsdGVkX@cpe-024-163-079-178.nc.res.rr.com) |
| 19:49.04 | PaulTech | We're just learning it now :) |
| 19:49.06 | CoaxD | PaulTech: That said, the guy was certainly NICE.. No doubt about that |
| 19:49.09 | PaulTech | Planning pretty big things |
| 19:49.10 | PaulTech | Oh |
| 19:49.15 | PaulTech | Just new |
| 19:49.17 | CoaxD | yeah |
| 19:49.27 | PaulTech | I just ordered our 3 T1's |
| 19:49.38 | PaulTech | Already got 4Gbps of data backbone |
| 19:49.48 | PaulTech | Could do with a VoIP Expert if anyone is looking for a job in Orlando ;) |
| 19:50.00 | denon | yay, lots of good 4Gbps backbone will do with 4.5Mbps out to the world :) |
| 19:50.11 | PaulTech | denon, t1 is for voice... |
| 19:50.17 | denon | t1 or pri? |
| 19:50.17 | PaulTech | and the 4Gbps if pre-existing |
| 19:50.56 | PaulTech | <PROTECTED> |
| 19:51.07 | PaulTech | 24 |
| 19:51.27 | denon | could be interesting |
| 19:51.42 | CoaxD | denon: I want IAX2 over E1 to be a standard! |
| 19:51.45 | shaon | hacking quintum A800 anyway? i know ip |
| 19:51.47 | PaulTech | Hehe I like IAX2 but I didnt see any non "soft phones" supporting it in a good price rate |
| 19:51.49 | CoaxD | denon: I WANT MY 4 MORE CHANNELS, BITCH! |
| 19:51.58 | CoaxD | Or something |
| 19:52.09 | file | gah a channelized T1? I like PRIs myself |
| 19:52.14 | denon | if you had an E1 .. you'd have to live .. somewhere else |
| 19:52.19 | PaulTech | Europe. |
| 19:52.19 | denon | like france, where the girls dont shave their pits |
| 19:52.26 | PaulTech | and they do shave their pits |
| 19:52.27 | *** join/#asterisk clinthome (~clinthome@va-chrvlle-cad1-bdgrp5-8d-209.chvlva.adelphia.net) |
| 19:52.34 | file | denon: I bet you know that stuff intimately |
| 19:52.38 | PaulTech | been there, its pretty common over there to shave |
| 19:52.39 | denon | PaulTech: well .. ok, but the water is undrinkable |
| 19:52.43 | denon | I know that first hand |
| 19:52.52 | denon | looked like weak tea comming out of the spout in paris |
| 19:52.55 | PaulTech | Actually its harder in american than most of Europe |
| 19:53.09 | denon | it was brown man .. |
| 19:53.17 | PaulTech | Yea, that was the bathroom man |
| 19:53.19 | shaon | any Quicknet Phone jack User? |
| 19:53.27 | denon | shaon: good luck .. nobody uses that crap |
| 19:53.32 | PaulTech | lol |
| 19:53.41 | PaulTech | Playing MPEG stream from fpm-world-mix.mp3 ... |
| 19:53.41 | PaulTech | MPEG 1.0 layer III, 128 kbit/s, 44100 Hz joint-stereo |
| 19:53.42 | PaulTech | [2:18] Decoding of fpm-world-mix.mp3 finished. |
| 19:53.47 | denon | shaon: I know it seems like a good idea to save $5 and use that thing .. but its not |
| 19:53.50 | PaulTech | Ok so my server uses the MP3 |
| 19:53.53 | PaulTech | But the PBX dont play them back |
| 19:54.00 | *** join/#asterisk bajanman (~william@cp66-203-194-32.cp.telus.net) |
| 19:54.06 | shaon | i bought it fro ebay for $5 so using it now |
| 19:54.09 | denon | PaulTech: weird id3? vbr? |
| 19:54.15 | *** join/#asterisk McUnixJr (~mcmer@McUnixJr.gold.supporter.pdpc) [NETSPLIT VICTIM] |
| 19:54.15 | *** join/#asterisk bjohnson (~bjohnson@jecinc.tor.istop.com) |
| 19:54.15 | *** join/#asterisk newl (~newlook@203-59-112-161.dyn.iinet.net.au) [NETSPLIT VICTIM] |
| 19:54.15 | Qwell | BMI is on to him! |
| 19:54.17 | denon | shaon: get what you pay for :) |
| 19:54.21 | PaulTech | denon, stock mp3 it comes with |
| 19:54.30 | denon | huh. |
| 19:54.32 | denon | mpg321? :) |
| 19:54.41 | PaulTech | mpg123 |
| 19:54.49 | PaulTech | -bash: mpg321: command not found |
| 19:54.58 | Qwell | What version? |
| 19:54.58 | PaulTech | High Performance MPEG 1.0/2.0/2.5 Audio Player for Layer 1, 2 and 3. |
| 19:55.09 | PaulTech | Version 0.59r (1999/Jun/15). Written and copyrights by Michael Hipp. |
| 19:55.12 | PaulTech | Uhhh |
| 19:55.33 | shaon | is this possible to do sip over dialup connection with asterisk? |
| 19:56.26 | *** join/#asterisk _Vile (~vile@90.b160.bendtel.net) |
| 19:57.08 | *** join/#asterisk shaon (~shaon@61.68.13.184) |
| 19:57.25 | Blackvel | 1p = 0,01 pounds? |
| 19:57.31 | bjohnson | PaulTech: talk to ManxPower |
| 19:57.35 | shaon | did i miss anything? |
| 19:57.40 | Blackvel | so 1p is lower than 1 euro cent? |
| 19:57.47 | bjohnson | PaulTech: especially if he can do from off-site |
| 19:57.56 | *** join/#asterisk leandro_pt (~leandro@82.155.114.169) |
| 19:58.09 | bjohnson | shaon: yes it's possible .. just not good |
| 19:58.22 | PaulTech | bjohnson, about ? |
| 19:58.29 | PaulTech | The job ? |
| 19:58.35 | bjohnson | PaulTech: you mentioned looking for voip expert |
| 19:59.00 | shaon | is this posible to do SIP over dialup connection ? |
| 19:59.06 | bjohnson | shaon: yes it's possible .. just not good |
| 19:59.19 | Blackvel | shaon: I wouldnt go below isdn |
| 19:59.22 | file | ooooooooh The Incredibles is on |
| 19:59.47 | shaon | one of my friend got dialup so... |
| 19:59.49 | SuPrSluG | what extension? |
| 20:00.07 | Blackvel | whats a good UK amount for pstn? 1p/min? |
| 20:00.17 | shaon | which codec is best i mean use less bandwidth? |
| 20:00.21 | bjohnson | Blackvel: free is always good |
| 20:00.29 | Blackvel | hehe |
| 20:00.30 | Blackvel | right |
| 20:00.34 | bjohnson | shaon: speex |
| 20:01.01 | shaon | but very few device use that codec |
| 20:01.24 | shaon | any device u know? |
| 20:01.28 | file | g729 |
| 20:01.42 | PaulTech | Ok |
| 20:01.53 | PaulTech | A brother needs some help with music-on-hold |
| 20:01.58 | PaulTech | 15 bucks to the winnar? |
| 20:02.38 | CoaxD | PaulTech: run mpg123 and get version info |
| 20:02.40 | shaon | i got 1 licence from digium 2days back can i use it with other person because it only 1 user registration or i need 2 licence? |
| 20:03.00 | PaulTech | CoaxD, newest |
| 20:03.07 | CoaxD | PaulTech: You *need* 0.59r |
| 20:03.08 | bjohnson | shaon: what device are you talking about? |
| 20:03.12 | PaulTech | Version 0.59r (1999/Jun/15). Written and copyrights by Michael Hipp. |
| 20:03.15 | PaulTech | Well that works out well |
| 20:03.16 | CoaxD | PaulTech: Okay |
| 20:03.22 | CoaxD | PaulTech: And what do you hear on the other end? |
| 20:03.27 | PaulTech | " " |
| 20:03.36 | bjohnson | PaulTech: can you playback() a mp3 |
| 20:03.40 | PaulTech | Yes |
| 20:03.47 | Qwell | its probably just really quiet |
| 20:03.47 | bjohnson | then it's not mpg123 |
| 20:03.47 | PaulTech | I can do MusicOnHold(30) |
| 20:03.56 | PaulTech | or whatever the command is |
| 20:03.59 | shaon | any device use speex codec u know? |
| 20:04.29 | PaulTech | Qwell, I dont have *quitemp3* enabled |
| 20:04.31 | bjohnson | PaulTech: you want to hear moh when on hold? when you dial the internal extension you use the 'm' arg? |
| 20:04.32 | PaulTech | its just mp3 |
| 20:04.49 | PaulTech | bjohnson, sip connections when they are put on hold |
| 20:04.56 | *** join/#asterisk kiokorobert1 (~kiokorobe@196.200.26.42) |
| 20:05.01 | bjohnson | <PROTECTED> |
| 20:05.26 | shaon | how can i use real streaming audio to play with astersik? |
| 20:05.34 | bjohnson | shaon: read the wiki |
| 20:05.36 | *** join/#asterisk jackfiber (cico@82.99.197.169) |
| 20:05.42 | bjohnson | shaon: but it dies frequently |
| 20:05.47 | PaulTech | shaon, |
| 20:05.48 | PaulTech | default => mp3:/var/lib/asterisk/mohmp3-empty,http://www.waixwave.com:8000/ |
| 20:06.00 | shaon | "Real Audio" not mp3 or mpg |
| 20:06.06 | PaulTech | code |
| 20:06.15 | PaulTech | vi realstream.c |
| 20:06.16 | PaulTech | get going |
| 20:06.17 | shaon | from real network |
| 20:06.30 | shido6 | once again |
| 20:06.34 | bjohnson | shaon: check into slimserver support of RA. then pick up the mp3 stream from slin=mserver |
| 20:06.37 | shido6 | for those that missed it b4 |
| 20:06.42 | bjohnson | shaon: read the wiki |
| 20:06.44 | shido6 | FUCK WINDOWS! ok thank you have a nice day |
| 20:06.54 | Blackvel | bjohnson: you can suggest me free voip providers, if you like, especially for UK/US |
| 20:07.13 | PaulTech | bjohnson, I dont understand the question if I make a extension call MusicOnHold(30) it plays fine |
| 20:07.36 | bjohnson | Blackvel: I thought there was sipgate.de and some french one that had lots of minutes for like 9eur/month or something |
| 20:07.52 | shaon | using slimserver can i listen a online radio broadcast? |
| 20:07.57 | PaulTech | exten => 6601,1,WaitMusicOnHold(999) |
| 20:08.11 | bjohnson | PaulTech: to call another extension you use the dial() command. do you use the 'm' arg available with the dial() command |
| 20:08.17 | bjohnson | shaon: yes |
| 20:08.25 | bjohnson | shaon: but it's not a good idea |
| 20:08.28 | PaulTech | Im not calling other extenions this is from outside > inside sip connections |
| 20:08.36 | bjohnson | PaulTech: HOW??? |
| 20:08.47 | bjohnson | you still use dial() |
| 20:08.52 | PaulTech | hehe... |
| 20:08.53 | PaulTech | sorry |
| 20:08.57 | PaulTech | Im still learning man |
| 20:09.19 | bjohnson | go to cli and type show application dial |
| 20:09.25 | bjohnson | look at what the m option does |
| 20:09.29 | shaon | bjohnson, thanks |
| 20:09.30 | bjohnson | then tell us if you're using it |
| 20:09.43 | PaulTech | <PROTECTED> |
| 20:09.50 | bjohnson | wow |
| 20:09.57 | bjohnson | sounds like what you want |
| 20:10.28 | bjohnson | now edit your extensions.conf to use it |
| 20:10.30 | PaulTech | I dont think so... tell me if Im wrong thou, I mean after someone picks up the line then puts on hold |
| 20:11.05 | Blackvel | bjohnson: sipgate is stupid, 2,6ct/uk, 2,3ct/us. nikotel has for 50EUR asset a rate of 1,5ct/min to UK/US |
| 20:11.15 | Blackvel | maybe still too expensive (well now I pay 2,6ct/min) |
| 20:11.21 | shaon | ATA 186 anybody wants to sale? |
| 20:11.31 | Blackvel | nikotel has expensive rates too, 2,9ct/uk |
| 20:11.32 | bjohnson | Blackvel: I haven't looked at euro voip providers |
| 20:11.44 | Blackvel | is babbel.net with 1p any good? |
| 20:11.49 | bjohnson | shaon: forget the ATA 186 .. get something better |
| 20:11.57 | Blackvel | I would love to get BV, but that is then 20$ per month |
| 20:12.09 | shaon | what u suggest? |
| 20:12.36 | Ayano | Can someone give me a hand getting a tdm400p installed. I thought I did everything right, but it is not picking up the calls. |
| 20:12.42 | bjohnson | shaon: Sipura SPA 2000 for 2 fxs port .. you might prefer the SPA 1001 or 3000 models depending on what you want though |
| 20:12.49 | Ayano | I have a fxs on 1, and an fxo on 4. |
| 20:13.02 | bjohnson | Blackvel: what are the NA voip providers to UK? |
| 20:13.33 | leandro_pt | hello.. does any know if there is a "relaxdmtf" for chan_capi? |
| 20:13.36 | *** join/#asterisk Rick_Hunter (~rhunter@05-173.008.popsite.net) |
| 20:13.53 | shaon | Ayano, Did u check zaptel.conf and Zapata.conf and context?/ |
| 20:14.46 | bjohnson | Blackvel: http://www.teliax.com/rates.html looks like most UK is about $0.02 USD (mobile is much more though) |
| 20:15.01 | bjohnson | Blackvel: isn't $0.02 USD about 1p? |
| 20:15.04 | shaon | anybody from Australia? |
| 20:15.04 | *** join/#asterisk zotz (~zotz@24.231.32.109) |
| 20:16.04 | Ayano | shaon: I have the fxo card looking like this... fxsks=4 in zaptel |
| 20:16.12 | shaon | anybody used Addpac AP160? |
| 20:17.15 | Ayano | shaon: and signalling=fxs_ks, context=from-pstn, callerid=""<0>, mailbox=, and channel => 4 |
| 20:17.48 | Ayano | and the pots line going into channel 4 |
| 20:18.32 | shaon | what about zapata? |
| 20:18.56 | shaon | signalling=fxs_ks |
| 20:19.06 | Ayano | yep |
| 20:19.13 | shaon | channel=4 |
| 20:19.37 | Corydon-w | What, no chanel=5 ? |
| 20:19.49 | Ayano | I have channel => 4 |
| 20:19.50 | Ayano | ? |
| 20:19.58 | shaon | did u check context? |
| 20:20.31 | shaon | did u modprobe? |
| 20:20.54 | shaon | ztcfg -vv |
| 20:20.58 | Ayano | I don't know |
| 20:21.05 | shaon | wait |
| 20:22.02 | shaon | run “/etc/init.d/zaptel start” to let the zaptel script to load all zaptel |
| 20:22.02 | shaon | modules |
| 20:22.06 | Corydon-w | Ayano: If you tail /var/log/messages, does it say that it failed to properly initialize the card? |
| 20:22.24 | shaon | run “ztcfg –vv” |
| 20:23.29 | *** join/#asterisk veryhot (~tho@adsl-69-109-159-210.dsl.sndg02.pacbell.net) |
| 20:23.54 | Ayano | corydon-w: no it doesn't |
| 20:24.00 | *** join/#asterisk eKo1 (~bernd@metrored-gw.tropicohn.com) |
| 20:24.05 | Ayano | It doesn't say anything about it. |
| 20:24.07 | Corydon-w | Ayano: do you have dialtone on channel 1? |
| 20:24.12 | Ayano | no |
| 20:24.25 | veryhot | anyone know when nufone.net will open again? |
| 20:24.25 | Ayano | I didn't do anything to set it up though. |
| 20:24.28 | Corydon-w | Ayano: have you run ztcfg ? |
| 20:24.49 | Corydon-w | Ayano: what is the output of: lsmod | grep -c zaptel |
| 20:25.18 | Ayano | Corydon-w: I used asterisk@home for the installation, and then followed thier directions. Do I still need to? |
| 20:25.26 | Corydon-w | Ayano: Please |
| 20:25.38 | *** join/#asterisk Genosse_Darklord (~Miranda@p5088AD6A.dip0.t-ipconnect.de) |
| 20:25.42 | Ayano | just type ztcfg? |
| 20:25.58 | Corydon-w | Yes |
| 20:26.02 | Ayano | k, hold on. |
| 20:26.50 | Ayano | It just went to the next line and didn't display anything. Do I need to run from a spec dir? |
| 20:27.03 | Corydon-w | No, that's usually good |
| 20:27.10 | Corydon-w | Now try: ztcfg -vv |
| 20:27.20 | Corydon-w | Does it say anything about channels being configured? |
| 20:27.56 | Ayano | 2 channels, 1=fxo, 4=fxs |
| 20:28.04 | *** join/#asterisk FarrisG (~jrush@h-68-164-19-170.dllatx37.covad.net) |
| 20:28.10 | Corydon-w | Hmmm |
| 20:28.18 | Corydon-w | Okay, that's good |
| 20:28.47 | Corydon-w | Does your /etc/asterisk/zapata.conf match what's in /etc/zaptel.conf? i.e. same signalling on both? |
| 20:28.52 | FarrisG | does anyone sell a combo headset/handset/speakerphone combo for PC softphones? |
| 20:29.32 | Ayano | Yes, should the signalling be opposite of what the card accually is? |
| 20:29.42 | Corydon-w | Yes |
| 20:29.59 | Corydon-w | Ayano: is Asterisk started? |
| 20:30.09 | Ayano | yes, and sip to sip works |
| 20:30.29 | Corydon-w | Ayano: on the asterisk command line, try: show modules like zap |
| 20:31.10 | bsdfreak | ok |
| 20:31.29 | Corydon-w | Ayano: does chan_zap show up? |
| 20:32.58 | Ayano | It gives help, I dont think its right, hold on. |
| 20:33.07 | bjohnson | FarrisG: err .. you're describing a hardware phone |
| 20:33.46 | shaon | corydon-w: can u please help me with ivr setup? |
| 20:34.09 | Corydon-w | shaon: why me? Just ask the channel |
| 20:34.09 | Ayano | it just gives me usage. are you sure that is the right command? if I just type show modules it works. |
| 20:34.22 | Corydon-w | Okay, you have an older version |
| 20:34.50 | Corydon-w | Ayano: so does chan_zap show up in that long list? |
| 20:35.07 | Corydon-w | Ayano: or you could try: help zap |
| 20:35.36 | veryhot | DID help, know any where I can get unlimited DID quick beside VP? |
| 20:36.22 | *** join/#asterisk JimVanM (~jimvanm@HSE-Toronto-ppp181188.sympatico.ca) |
| 20:36.44 | shaon | help !!!!!!!!! h323 setup on asterisk |
| 20:37.07 | Ayano | k, hold on |
| 20:37.29 | MikeJ[Laptop] | ~rtfw |
| 20:37.30 | jbot | i guess rtfw is Read That F*cking Wiki, the one at http://www.voip-info.org/tiki-index.php |
| 20:37.32 | *** join/#asterisk ruskel (voip@200.102.64.202) |
| 20:38.13 | bjohnson | veryhot: no such thing |
| 20:38.27 | Ayano | if I do a show zap channel and it doesnt show them. |
| 20:38.34 | veryhot | bjohnson: hi |
| 20:38.35 | bjohnson | ~h323 |
| 20:38.36 | jbot | h323 is probably An ITU-T standard for packet-based multimedia communications systems. This standard defines the different multimedia entities that make up a multimedia system - Endpoint, Gateway, Multipoint Conferencing Unit (MCU), and Gatekeeper - and their interaction. This standard is used for many voice-over-IP applications, and is heavily dependent on other ... |
| 20:38.48 | Corydon-w | Ayano: there's no such command. Do: help zap |
| 20:38.59 | veryhot | bjohnson: everyone take so long, some don't even get a # for week. |
| 20:39.22 | Ayano | Hold on, its rebooting. |
| 20:39.39 | Corydon-w | Why are you rebooting? |
| 20:39.49 | bjohnson | veryhot: so? |
| 20:40.40 | veryhot | bjohnson: guess I go and use VP |
| 20:41.19 | shaon | wifi hack any idea? |
| 20:41.37 | bjohnson | veryhot: ok |
| 20:41.47 | bjohnson | shaon: wifi hack? |
| 20:42.10 | shaon | wireless 802.11b wep hack |
| 20:42.20 | bjohnson | geez |
| 20:42.29 | shaon | how to do it? |
| 20:42.39 | Ayano | Call me old fasion, but when I make changes I'm scared one will cause it not to boot, then it makes it easier to find the problem. : ) lol |
| 20:42.40 | veryhot | why? |
| 20:42.50 | PaulTech | You would you recommend for inbound DID, For testing |
| 20:42.53 | PaulTech | Who* |
| 20:43.08 | shaon | bjohnson:i get 3 signals in my laptop |
| 20:43.08 | PaulTech | We're getting our big commit from LVLT but I just want to test with two inbound lines |
| 20:43.17 | bjohnson | PaulTech: any of a thousand per minute providers |
| 20:43.27 | PaulTech | Hehe |
| 20:43.30 | bjohnson | PaulTech: nufone is a usual starting point |
| 20:43.30 | PaulTech | Ok |
| 20:43.32 | veryhot | paultech: I like voicepulse connect |
| 20:43.34 | Himeko | this is not #wephacking |
| 20:43.49 | veryhot | bjohson: nufone still closed |
| 20:43.51 | PaulTech | I like voicepulse already |
| 20:43.59 | bjohnson | voicepulse is a monthly charge .. usually not the best price |
| 20:44.15 | PaulTech | Well this is for a office so unlimited inbound is <3 |
| 20:44.25 | Ayano | corydon-w: ok, I did a zap show channels, and it showed a psedo channel and not the other two. |
| 20:44.35 | veryhot | paultech: I tried so many other DID providers, they suck at giving a did |
| 20:44.41 | kiokorobert1 | anyone tried teliax |
| 20:44.44 | PaulTech | veryhot, bigger from a telco ? |
| 20:44.49 | PaulTech | better rather |
| 20:44.55 | bjohnson | PaulTech: always read the fine print on the "unlimited" |
| 20:44.57 | shaon | himeko: how to hack wep? |
| 20:44.59 | veryhot | paultech: nope, just small provider |
| 20:45.23 | bjohnson | kiokorobert1: teliax gets good reviews |
| 20:45.28 | Corydon-w | Ayano: then almost certainly there's something wrong with your zapata.conf |
| 20:45.35 | PaulTech | bjohnson, been in the IT market awhile |
| 20:45.39 | PaulTech | I understand unlimited |
| 20:45.47 | Corydon-w | Ayano: perhaps you have two different lines that both say [channels] ? |
| 20:45.48 | PaulTech | Thanks thou |
| 20:45.52 | veryhot | paultech: who have you tried? |
| 20:45.54 | Corydon-w | (I've seen that before) |
| 20:46.04 | veryhot | paultech: some place got suck support. |
| 20:46.17 | PaulTech | veryhot, just one testing |
| 20:46.22 | PaulTech | We have a deal with LVLT and BellSouth |
| 20:46.22 | kiokorobert1 | thanks bjohnson |
| 20:46.28 | PaulTech | to provide the DID alone |
| 20:46.33 | veryhot | paultech: some place don't even have DID, but they advertise it. |
| 20:46.41 | Corydon-w | Ayano: try posting your ENTIRE zapata.conf to http://pastebin.ca |
| 20:46.52 | PaulTech | veryhot, yea |
| 20:49.38 | Ayano | Corydon-w: k, hold on. |
| 20:52.28 | Ayano | corydon-w: http://pastebin.ca/10597 |
| 20:52.54 | Ayano | You got it? |
| 20:54.16 | Corydon-w | Well, there's your problem |
| 20:54.25 | Corydon-w | You removed the #, didn't you? |
| 20:54.25 | Ayano | What's the damage |
| 20:54.43 | Ayano | No |
| 20:54.45 | Corydon-w | # does not indicate a comment in zapata.conf |
| 20:54.56 | Ayano | I can add it though. Hold on. |
| 20:55.15 | Corydon-w | You have: "include zapata-channels.conf" instead of "#include zapata-channels.conf" |
| 20:55.29 | Corydon-w | And you'll need to stop and start asterisk after this change |
| 20:55.48 | Ayano | k, trying it. I didn't remove it, that is how it was generated. |
| 20:57.14 | *** join/#asterisk j_vianna (~joaoviann@static-68-236-216-96.nwrk.east.verizon.net) |
| 20:57.22 | *** join/#asterisk r0d3nt (~RatMan@wsip-24-234-241-84.lv.lv.cox.net) |
| 20:57.43 | Ayano | didn't come back up, hold on.... |
| 20:57.52 | Corydon-w | Ayano: the problem is, it was not acutally reading your config file |
| 20:59.17 | Ayano | It is saying broken pipe unregistered tor, and zap |
| 20:59.37 | Corydon-w | Ayano: and technically, the callerid on channel 4 should be callerid=asreceived |
| 21:01.25 | Ayano | I made the changes and asterisk wont start. |
| 21:01.43 | Corydon-w | Ayano: you're probably going to need someone to ssh into your machine and figure out the problems... because this piecemeal stuff is getting old |
| 21:02.16 | Ayano | sorry let me try a few things |
| 21:03.18 | JimVanM | Brainstorm request: I have a lamp connected to a Sipura, which I want to "ring" whenever there are calls int the queue |
| 21:03.27 | JimVanM | Any ideas on the best way to do this? |
| 21:03.41 | JimVanM | (it has to stop rining when tere are no calls in queue) |
| 21:03.44 | ThunderDump | mr house |
| 21:04.32 | Ayano | corydon-w: can you go back to the pastebin and check the syntax on the channels for me? |
| 21:04.53 | Corydon-w | Syntax looks fine |
| 21:06.16 | *** join/#asterisk grolloj (~chatzilla@slim-eth0.horizonlive.net) |
| 21:06.28 | Ayano | Okay, I'm going to try a few things. |
| 21:06.35 | *** join/#asterisk DaLion (~DaLion@Toronto-HSE-ppp3771052.sympatico.ca) |
| 21:06.51 | DaLion | Iusing 3 SPA-3000 boxes, they |
| 21:06.52 | DaLion | all set to register once an hour but r SIP protocol seems to be returning a $VAR that |
| 21:06.52 | DaLion | is telling to re-register every 3 minutes. |
| 21:07.04 | DaLion | any way to fix ? |
| 21:13.30 | *** join/#asterisk ramtha (~tk@td9091901.pool.terralink.de) |
| 21:13.49 | Ayano | Corydon-w: I rebooted the server and it came up with asterisk running. I did a stop gracefully, and asterisk -vvvvc. What is the difference? |
| 21:14.05 | ramtha | hi, where i can find the hangup cause, wich asterisk gets if all channels of a span are full?? |
| 21:14.20 | Ayano | Corydon-w: and now the channel is working... Your my hero for today! |
| 21:14.23 | Ayano | Thank you |
| 21:14.24 | ramtha | can i simulate calls? are there any tools? |
| 21:14.31 | Ayano | 7777 |
| 21:15.34 | Ayano | ramtha: i think 7777 simulates and outside call for most systems. |
| 21:16.51 | *** join/#asterisk pussfeller (~todd@t1-rtc-woodlawn.rtcol.com) |
| 21:17.21 | ramtha | Ayano: how can i execute 50 calls at the same minute? |
| 21:17.31 | ramtha | second ;) |
| 21:18.46 | Ayano | You can write a script to do it, or you can write a webpage to do it as well. |
| 21:18.58 | *** join/#asterisk tessier (~treed@210.245.97.143) |
| 21:19.16 | masonc | anyone know how to configure static routes into a linux box? |
| 21:19.18 | ramtha | ok i have no skill for that ;) |
| 21:19.26 | Hydr0p0nx | route add |
| 21:19.43 | masonc | I have eth0 and eth1 |
| 21:19.50 | masonc | the gateway is on eth0 |
| 21:19.52 | Ayano | You can also just manually create 50 call files and drop them in the spooler. |
| 21:20.02 | masonc | but I have some IPs I have to route to through eth1 |
| 21:20.02 | PatrickDK | hmm? static routes? are those possible anymore? |
| 21:20.17 | mike-ff | masonc eth0 and eth1 need to be on different networks |
| 21:20.17 | PatrickDK | oh heh |
| 21:20.28 | PatrickDK | or do proxyarp |
| 21:20.47 | ramtha | masone: route add -net 192.168.0.0 netmask 255.255.255.0 192.168.0.1 |
| 21:20.52 | mike-ff | masonc and then you need to add a default route to your router on eth0 and network routes to the gateways on the eth1 subnet |
| 21:20.55 | ramtha | last ip ist gateway ip |
| 21:21.21 | Ayano | ramtha: You can also just manually create 50 call files and drop them in the spooler. |
| 21:21.44 | ramtha | Ayano: wich cmd i must use? |
| 21:23.09 | Ayano | ramtha: I have a context file somewhere in my e-mail. Give me a little bit. I have to run out for a few minutes. |
| 21:24.06 | ramtha | no problem, thx |
| 21:26.03 | Ayano | ramtha: is this something that has to be done right now? |
| 21:27.10 | ramtha | Ayano: nothing must be done right now ;). i think i dont need id. i only must verify that i do not get hangup cuase 1 if all channels of one group are full |
| 21:27.49 | ramtha | id=it |
| 21:27.53 | *** part/#asterisk eKo1 (~bernd@metrored-gw.tropicohn.com) |
| 21:28.31 | Ayano | Ok, I'll be back on later on. I have to take my daughter to school. Hit me up when I get back on. |
| 21:28.40 | ramtha | ok |
| 21:29.07 | Ayano | Thank you Corydon-w |
| 21:33.19 | *** join/#asterisk Trickyphillips (~Trickyphi@adsl-68-124-57-143.dsl.irvnca.pacbell.net) |
| 21:34.09 | ManxPower | ~docs |
| 21:34.10 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org |
| 21:34.13 | ManxPower | ~mailinglist |
| 21:34.14 | jbot | mailinglist is, like, Search Asterisk mailing lists by prepending site:lists.digium.com to your Google search. Browse the mailing list archive at http://lists.digium.com/ |
| 21:35.27 | *** join/#asterisk clinthome (~clinthome@va-chrvlle-cad1-bdgrp5-8d-209.chvlva.adelphia.net) |
| 21:35.46 | tld | Can I run the extension/sip userlist in a PostgreSQL database? Any anyone know of a resource I can read up on it if it's doable? |
| 21:38.38 | *** part/#asterisk Patrick^ (~patrickm@birch4.mountaincable.net) |
| 21:42.44 | *** join/#asterisk clinthome (~clinthome@va-chrvlle-cad1-bdgrp5-8d-209.chvlva.adelphia.net) |
| 21:50.07 | *** join/#asterisk darby_t (~tom@doa112.neoplus.adsl.tpnet.pl) |
| 21:51.34 | *** join/#asterisk clinthome (~clinthome@va-chrvlle-cad1-bdgrp5-8d-209.chvlva.adelphia.net) |
| 22:02.49 | *** join/#asterisk webrunners (~blake@adsl-67-115-226-12.dsl.lsan03.pacbell.net) |
| 22:03.15 | webrunners | Hello Everyone.... Can anyone possible help with a Call Queue question? |
| 22:07.02 | webrunners | Any Asterisk consultants here? |
| 22:07.08 | *** join/#asterisk gpearson (~Graham@c-67-177-182-16.hsd1.in.comcast.net) |
| 22:07.26 | iq | webrunners, ask the question. I'm sure I wont know the answer but someone else might ;0 |
| 22:08.06 | *** join/#asterisk JerJer[mobile] (~nonyobizn@45.210.5.249) |
| 22:08.16 | *** join/#asterisk FuriousGeorge (~brian@ool-43516ebb.dyn.optonline.net) |
| 22:09.43 | FuriousGeorge | hey all. i know my credentials are right in sip.conf, and this has never happened beforehttp://pastebin.ca/10607 though this is a new setup. can someone transalate that for me? im still learning voipese |
| 22:09.59 | FuriousGeorge | http://pastebin.ca/10607 <--thats the error |
| 22:10.30 | webrunners | I trying to figure out on how to make asterisk not keep ringing a phone that is loged into a call queue when the agent is allready on a call. |
| 22:11.51 | FuriousGeorge | n/m it stopped happening |
| 22:15.58 | FuriousGeorge | actually, this error happens when i try to log in an xlite to my * server. why would that prevent me from authorizing to my sip dialtone provider |
| 22:16.20 | FuriousGeorge | http://pastebin.ca/10607 |
| 22:17.15 | *** join/#asterisk Romik (~romik@1.fix.netvision.net.il) |
| 22:17.15 | *** part/#asterisk Grooby (~Grooby@12.22.232.212) |
| 22:20.23 | *** join/#asterisk rowter (~Drake@201.133.210.80) |
| 22:22.17 | *** join/#asterisk In-Side (~Lowgitek@es-217-129-30-41.netvisao.pt) |
| 22:22.19 | In-Side | Hi |
| 22:22.29 | FuriousGeorge | so does anybody know why i opening xlite makes my registration with sipphone timeout |
| 22:22.33 | In-Side | is any ser magic by there? |
| 22:22.34 | FuriousGeorge | it doesnt make any sense |
| 22:22.48 | In-Side | FuriousGeorge: nat |
| 22:22.50 | In-Side | issue |
| 22:23.00 | FuriousGeorge | * and xlite are on the same side of the nat |
| 22:23.03 | In-Side | maybe... if it was in ser I would have a answer for yo |
| 22:23.13 | FuriousGeorge | its only when i open xlite that i get kicked off |
| 22:23.20 | In-Side | I was confronting the same problem some hours ago.. |
| 22:23.38 | FuriousGeorge | its not like im even trying to make a call |
| 22:23.52 | In-Side | now I found the anwer.. but a new problem arise.... i can't find a perfect solution for both problems :S |
| 22:24.10 | In-Side | yes the session timeout |
| 22:24.15 | In-Side | and it hangup |
| 22:24.19 | In-Side | I had same problem |
| 22:24.27 | FuriousGeorge | http://pastebin.ca/10607 <--thats the error i get |
| 22:24.28 | In-Side | it was about the re invite |
| 22:24.33 | FuriousGeorge | reinvite huh |
| 22:24.49 | In-Side | I don't understand much of asterisk to help you sorry |
| 22:24.49 | *** join/#asterisk Sedorox (~Sed@Neptune.client.wlmsprt.pa.sed6.net) |
| 22:25.02 | FuriousGeorge | apparently no one is here today |
| 22:25.15 | In-Side | try to decrease |
| 22:25.21 | In-Side | the registration time |
| 22:25.29 | In-Side | like 300 or something |
| 22:25.37 | FuriousGeorge | in what? xlite? |
| 22:25.48 | In-Side | checkout your configuration to see if you impose a time limit |
| 22:25.49 | *** join/#asterisk bannerman (~bannerman@209.216.176.42) |
| 22:25.52 | In-Side | and sxlite is conform to it |
| 22:26.04 | In-Side | I cant be more explicit sorry |
| 22:26.36 | FuriousGeorge | no, ive done this before and ive never had this problem, its not that. i never needed to set that before |
| 22:27.17 | *** part/#asterisk FarrisG (~jrush@h-68-164-19-170.dllatx37.covad.net) |
| 22:30.03 | FuriousGeorge | it appears when i sip show peers, the user for my xlite is already logged on. then i open xlite and really break everything |
| 22:30.51 | FuriousGeorge | so i guess i have to figure out why this user is being logged on by defaulty |
| 22:30.59 | *** join/#asterisk _SMP_ (~SMP@pandora.burned.net) |
| 22:33.31 | In-Side | sorry i can't help you |
| 22:33.41 | In-Side | I was just wondering |
| 22:34.32 | In-Side | in ser i use this code to turn around that issue |
| 22:34.32 | In-Side | f (method=="INVITE" && client_nat_test("19")) { |
| 22:34.32 | In-Side | <PROTECTED> |
| 22:34.32 | In-Side | <PROTECTED> |
| 22:34.32 | In-Side | <PROTECTED> |
| 22:34.32 | In-Side | }; |
| 22:35.18 | In-Side | so what is the real problem ? |
| 22:35.32 | In-Side | the are multiple logins with same account ? |
| 22:35.40 | In-Side | is that the problem ? |
| 22:35.48 | *** join/#asterisk Lee__ (~Lee__@cpe-69-203-206-248.nyc.res.rr.com) |
| 22:35.56 | Lee__ | allo |
| 22:35.59 | In-Side | allo |
| 22:36.25 | Lee__ | in the US, is getting a 911 POTS line from Verizon the way to deal with 911 service? |
| 22:36.54 | In-Side | huh? |
| 22:37.03 | *** join/#asterisk bjohnson (~bjohnson@66.11.165.65) |
| 22:37.34 | Lee__ | I want my Asterisk boxen to respond to 911 calls even with a power outage |
| 22:38.03 | Lee__ | so I get an FXO card and the lowest level of service from Verizon and only route calls to the FXO if they are 911 |
| 22:38.06 | JerJer[mobile] | Lee__: then power your asterisk box with a UPS and backup generator |
| 22:38.17 | Lee__ | nonono. like the old skool phones |
| 22:38.32 | bjohnson | and make sure your isp doesn't fail |
| 22:38.40 | Nugget | Lee: but yesyes, JerJer's suggestion also solves your problem. |
| 22:38.56 | Lee__ | the FXO cards have a loopback, right? I can get at least one analog phone to maintain 911 service this way. |
| 22:39.04 | *** join/#asterisk likwid-- (likwid@nc-69-68-82-223.dyn.sprint-hsd.net) |
| 22:39.05 | Nugget | why not just get a UPS? |
| 22:39.07 | In-Side | why he will need the isp if the call will be route trought pots ? |
| 22:39.09 | In-Side | rotfl |
| 22:39.15 | In-Side | get an ups |
| 22:39.19 | bjohnson | Lee__: ohhh .. you mean the crap clone cards |
| 22:39.22 | In-Side | and get the asterisk box up |
| 22:39.24 | In-Side | just that |
| 22:39.28 | JerJer[mobile] | Nugget: because a ups will even eventually die if edison does not come back |
| 22:39.36 | Nugget | get a bigger ups. :) |
| 22:39.38 | Lee__ | I don't care who makes the card. The Digium one looks good |
| 22:39.45 | In-Side | JerJer[mobile]: well buy an generator |
| 22:39.46 | Lee__ | UPS is a shoddy solution |
| 22:39.46 | In-Side | rotfl |
| 22:40.00 | In-Side | you have another way |
| 22:40.03 | ManxPower | Anyone here set up Tellabs hardware echocan devices? |
| 22:40.07 | In-Side | use an sipura device |
| 22:40.08 | bjohnson | Lee__: the digium ones don't have a bypass iirc |
| 22:40.12 | In-Side | sipura 3000 |
| 22:40.18 | Nugget | If you think a UPS is a shoddy solution, you're in for a shock when you start messing with the current state of FXO solutions. |
| 22:40.19 | bjohnson | and SPA 3000 is one option |
| 22:40.22 | Lee__ | the analog phone network has the advantage of sending DC on the line. that's why analog phones still have service in a power outage |
| 22:40.23 | In-Side | spa 3000 can dout |
| 22:40.24 | bjohnson | err |
| 22:40.26 | *** join/#asterisk Tall-guy (tall-guy@hssxrg207-195-103-110.sasknet.sk.ca) |
| 22:40.27 | JerJer[mobile] | In-Side: that is what i suggested n the first place, smart guy |
| 22:40.39 | In-Side | sorry i didn't see |
| 22:40.51 | In-Side | anyway he will still need the power source |
| 22:40.52 | In-Side | anyway |
| 22:41.00 | JerJer[mobile] | Lee__: you need a UPS to sustain thru the edison power loss |
| 22:41.14 | Nugget | why do you need an SPA anyway? isn't a phone plugged into the wall what you're talking about? |
| 22:41.16 | In-Side | Lee__: you need eggs to have an omelet |
| 22:41.25 | Nugget | you need a $4 telephone. instant 911 fallback |
| 22:41.28 | In-Side | if he use and spa 3000 |
| 22:41.29 | Lee__ | but not if there's a single analog phone connected to the FXO card and I walk up to it, plug a phone in and use it. |
| 22:41.41 | Nugget | Lee__: just plug the phone into the wall and use it |
| 22:41.44 | In-Side | the outage porwer will be much lower |
| 22:41.59 | bjohnson | leave a phone plugged into the pstn, parallel to the fxo |
| 22:42.03 | Lee__ | yeah, the $4 telephone, but it'd be nice to use that line when there isn't a power outage too, right? |
| 22:42.05 | bjohnson | just turn it's ringer off |
| 22:42.09 | Qwell | Doesn't the SPA 3000 have passthrough? |
| 22:42.12 | Nugget | Lee__: ok, so use it. |
| 22:42.13 | In-Side | yah |
| 22:42.15 | In-Side | no no |
| 22:42.16 | bjohnson | Qwell: yes |
| 22:42.22 | In-Side | well yes it has |
| 22:42.23 | In-Side | rotfl |
| 22:42.25 | Lee__ | bjohnson: that's what I thought, just checking. you guys are really touchy about questions. |
| 22:42.26 | Nugget | doesn't the SPA 3000 require power? :) |
| 22:42.29 | Qwell | is that what he's using, or am I not reading high enough up? |
| 22:42.40 | Qwell | Nugget: No, thats the thing. if the spa3000 loses power, it becomes a passthrough |
| 22:42.40 | In-Side | no spa 300 has pass torught i thkin |
| 22:42.43 | In-Side | let me confirm |
| 22:42.43 | bjohnson | Nugget: not for passthough between the fxs and fxo |
| 22:42.45 | In-Side | ehehe |
| 22:42.47 | Nugget | *nod* |
| 22:42.48 | In-Side | yes it has |
| 22:42.49 | In-Side | rotfl |
| 22:42.50 | In-Side | eheh |
| 22:42.54 | In-Side | I tried and it has |
| 22:42.55 | In-Side | :p |
| 22:42.56 | Nugget | it's still a shoddy solution, if you ask me. :) |
| 22:43.00 | Lee__ | Cool, the sip 3000 sounds cool |
| 22:43.04 | In-Side | I jsut forgot why i obought this |
| 22:43.05 | In-Side | :p |
| 22:43.11 | In-Side | it is cheap |
| 22:43.16 | In-Side | and works nicelly |
| 22:43.23 | In-Side | nice peace of hardware |
| 22:43.28 | In-Side | piece |
| 22:43.31 | ManxPower | The SPA-3k is good if you want local pots service for any reason |
| 22:43.42 | In-Side | and has another advantage |
| 22:43.44 | Lee__ | ManxPower: thanks. |
| 22:43.52 | Qwell | there aren't many ATAs with FXO, are there? |
| 22:44.00 | bjohnson | Qwell: no |
| 22:44.01 | In-Side | ya some |
| 22:44.07 | In-Side | but not so nice as sipura ones |
| 22:44.16 | In-Side | I like then |
| 22:44.31 | In-Side | anyway... there are anybody there that works with ser also? |
| 22:44.46 | In-Side | ser channel seems to have alot of dead people |
| 22:44.47 | In-Side | :s |
| 22:45.04 | In-Side | I can hear the moon there.... |
| 22:45.23 | ManxPower | In-Side, We think SER kills people. |
| 22:45.32 | In-Side | rotfl |
| 22:45.36 | ManxPower | No proof, but the evidence suggests that SER is a mass murderer. |
| 22:45.48 | Qwell | ManxPower: sure, take the easy joke :p |
| 22:45.48 | In-Side | yes.. I start to thinking the same |
| 22:45.49 | In-Side | :s |
| 22:46.03 | ManxPower | Qwell, A SERial killer? |
| 22:46.07 | In-Side | I'm not a english speaker... |
| 22:46.12 | Qwell | No, thats the easy pun |
| 22:46.13 | In-Side | I just understand ehehe |
| 22:46.29 | In-Side | anyway ser is a killer application |
| 22:46.48 | In-Side | but... I thinking the long usage of it my cause brain damage or something |
| 22:46.49 | In-Side | :p |
| 22:47.21 | In-Side | ok... if there are no Ser user here I have a question about asterisk that is hrt my brain |
| 22:47.49 | In-Side | how can I get asterisk to work as brigde and not trying to translate the codecds of my devices ? |
| 22:48.07 | bjohnson | set them to use the same codecs |
| 22:48.14 | In-Side | I use g729 and 723.1 in all of then but * keeps trying to translate it |
| 22:48.33 | In-Side | and of course i don't have the license to use it |
| 22:48.46 | JerJer[mobile] | then don't translate t |
| 22:48.47 | JerJer[mobile] | it |
| 22:48.57 | In-Side | how can i turn off it ? |
| 22:49.11 | JerJer[mobile] | asterisk will pass thru the data |
| 22:49.21 | In-Side | [options] |
| 22:49.21 | In-Side | transcode_via_sln = no |
| 22:49.24 | JerJer[mobile] | if the path stays the same codec all the way thru |
| 22:49.25 | In-Side | I try this |
| 22:49.29 | In-Side | with no success |
| 22:49.34 | JerJer[mobile] | and you don't have any dial modifiers or play any prompts |
| 22:50.09 | In-Side | hows that I can't have dial plans ? |
| 22:50.15 | JerJer[mobile] | In-Side: and your running cvs -head as of at least yesterday? |
| 22:50.17 | JerJer[mobile] | READ |
| 22:50.22 | JerJer[mobile] | dial modifiers |
| 22:50.28 | JerJer[mobile] | |r |
| 22:50.30 | JerJer[mobile] | |m |
| 22:50.33 | JerJer[mobile] | |t |
| 22:50.34 | JerJer[mobile] | |T |
| 22:50.36 | JerJer[mobile] | etc |
| 22:50.36 | Qwell | JerJer[mobile]: I've never seen anything that says so, besides when ordering a DID. Does nufone use SIP? |
| 22:50.43 | In-Side | hmm |
| 22:50.47 | JerJer[mobile] | Qwell: sure |
| 22:50.53 | In-Side | JerJer[mobile]: I really newbie on asterisk sorry |
| 22:51.00 | In-Side | what that really does? |
| 22:51.49 | Qwell | Do you guys terminate yourselves, or use a higher power? I'm fine with PM if you don't want to discuss in the open |
| 22:52.24 | *** join/#asterisk FuriousGeorge (~brian@ool-43516ebb.dyn.optonline.net) |
| 22:53.11 | FuriousGeorge | this sip registration problem appears to be with xlite using oss (using wine) while asterisk tries to do the same. if i start asterisk first, then xlite, sound doesnt work in xlite |
| 22:53.34 | In-Side | lools |
| 22:53.46 | In-Side | you could say it more soon you was using wine |
| 22:53.47 | In-Side | rotfl |
| 22:53.58 | FuriousGeorge | how can i just turn sound off in the console |
| 22:54.08 | Qwell | noload alsa and oss modules? |
| 22:54.12 | Qwell | not sure what that will affect |
| 22:54.33 | *** join/#asterisk lesouvage (~lesouvage@cc341200-a.assen1.dr.home.nl) |
| 22:54.46 | FuriousGeorge | Qwell: where do i put this noload optionj? i see alsa.conf and oss.conf, but they appear to be called by a third file |
| 22:55.00 | Qwell | modules.conf |
| 22:55.07 | FuriousGeorge | thanks |
| 22:56.26 | FuriousGeorge | beautiful. now to make the dialplan work. thanks again, Qwell |
| 22:56.32 | Qwell | works? |
| 22:56.43 | Qwell | paypal.com > |
| 22:56.50 | FuriousGeorge | well, i can use xlite via wine to test my dialplan |
| 22:56.57 | FuriousGeorge | but not EVERYTHING works ;) |
| 22:59.50 | FuriousGeorge | Qwell: if i open asterisk first it still breaks xlite's audio |
| 23:00.12 | Qwell | You didn't do it right then |
| 23:01.17 | FuriousGeorge | k, im just confused, there seems to be another problem. when i open xlite, it automatically starts trying to make a call, if asterisk is already open |
| 23:02.38 | FuriousGeorge | no ive closed xlite and im still getting erros about Maximum retries exceeded on call |
| 23:03.48 | FuriousGeorge | now its working, if i open asterisk first thenopen xlite, all is good |
| 23:04.29 | lesouvage | FutiousGerage: If you want to do it the easy way install Xorcom Rapid, a ready to use Asterisk distribution. |
| 23:05.57 | FuriousGeorge | lesouvage: the point is really to learn this |
| 23:06.12 | FuriousGeorge | i keep getting sidetracked because i need to get better with linux itself |
| 23:09.35 | lesouvage | Does anybody know the exact hight of an tdm400p pci card. It's maybe a kind of strange question but I'm developing an start-up SOHOasterisk box with standard an x100p card but I want to offer the option to upgrade to a n tdm400p card . |
| 23:12.47 | *** join/#asterisk leandro_pt (~leandro_p@bl6-114-169.dsl.telepac.pt) |
| 23:15.29 | FuriousGeorge | im having trouble finding the link for the x-lite for linux beta, anyone got it? |
| 23:16.16 | FuriousGeorge | n/m |
| 23:17.14 | *** join/#asterisk Sedorox (~Sed@Neptune.client.wlmsprt.pa.sed6.net) |
| 23:20.58 | *** part/#asterisk Romik (~romik@1.fix.netvision.net.il) |
| 23:22.06 | leandro_pt | hi.. anyone use chan_capi + dmtf? |
| 23:22.27 | leandro_pt | dtmf |
| 23:28.29 | *** join/#asterisk OloBola (~not@h-66-134-67-154.snvacaid.covad.net) |
| 23:29.56 | OloBola | when where is the message envelope info stored for comedian mail? I can't find callerID info in the "Message Information file". Is it somewhere else? |
| 23:32.57 | *** join/#asterisk tainted- (~identd@65-60-70-243-cust.telepacific.net) |
| 23:33.02 | OloBola | I found it |
| 23:33.34 | *** join/#asterisk jackfiber (cico@82.99.197.169) |
| 23:33.44 | jesster | having trouble getting ring-answer to work with Polycom, anyone try this? I was confused on the wiki. |
| 23:34.01 | jackfiber | can u get any voip phone to work behinf SpeedStream NAT routers (ADSL) ? |
| 23:34.28 | jackfiber | can u get any voip phone to work behind SpeedStream NAT routers (ADSL) ? |
| 23:35.51 | *** join/#asterisk tld (~tld@80.203.70.227) |
| 23:38.51 | *** join/#asterisk JunK-Y (~junky@modemcable174.107-81-70.mc.videotron.ca) |
| 23:41.45 | *** part/#asterisk bkw_ (~bkw_@bkw.developer.and.friend.of.asterisk) |
| 23:45.07 | *** join/#asterisk bkw_ (~brian@bkw.developer.and.friend.of.asterisk) |
| 23:45.08 | *** mode/#asterisk [+o bkw_] by ChanServ |
| 23:45.16 | nwhit | i am having problems with cisco phones hanging up the person after a certain amount of time if you put a person on hold |
| 23:45.19 | nwhit | and they hang up after some time if the other side puts the person on hold |
| 23:45.22 | nwhit | any ideas |
| 23:46.32 | jackfiber | qualify=yes |
| 23:48.56 | nwhit | jackfiber: who's that for? |
| 23:51.42 | jackfiber | it's for qualifying the connection if ur are behind NAT the connection might be dropped by router/ADSl closing UDP port qualify =yes on asterisk side will cause it to be kept open by quality=yes caused 2 seconds |
| 23:51.54 | jackfiber | intervals keep alive packets |
| 23:51.56 | jackfiber | u can set |
| 23:52.01 | jackfiber | qualify=100 |
| 23:52.13 | jackfiber | which sends jeekalive every 100miliseconds |
| 23:52.20 | jackfiber | and any other number |
| 23:52.38 | jackfiber | r u on PPPoE or PPPoA ? |
| 23:52.42 | jackfiber | adsl |
