| 00:00.20 | Corydon76-dig | brunner: figure out from your telco bill what service you're getting |
| 00:00.46 | brunner | Corydon76-dig: it's a channelized T1 |
| 00:00.56 | Corydon76-dig | so probably E&M |
| 00:01.07 | *** join/#asterisk SexyKen (n=ken@c-67-160-193-210.hsd1.ca.comcast.net) |
| 00:01.24 | SexyKen | Hey guys - how can I accept a user input and make the caller id that input? |
| 00:01.28 | jaytee | brunner, all 24 channels are voice? |
| 00:01.41 | brunner | jaytee: yes |
| 00:01.46 | brunner | we have a second T1 for data |
| 00:01.48 | jaytee | if so it's 99% probable that it's E&M Wink |
| 00:02.56 | jaytee | brunner, do you do the programming for it or does someone else handle the MAC stuff |
| 00:03.05 | SexyKen | Any ideas? |
| 00:03.26 | *** part/#asterisk eric256 (n=eric256@c-75-71-188-88.hsd1.co.comcast.net) |
| 00:03.57 | brunner | jaytee: someone else did in the past |
| 00:04.20 | brunner | the service contract was terminated by someone else in my organization |
| 00:04.23 | jaytee | SexyKen, I'm always full of ideas. Right now, because of your nick I'm getting all kinds of ideas about SexyBarbie and SexySkipper |
| 00:04.41 | keebler | bmoraca: Still around? |
| 00:04.56 | SexyKen | jaytee: That's good..thanks for the assistance. |
| 00:05.59 | manxpower | brunner: you will *never* get it working if you don't have someone good with the Nortel on hand. |
| 00:06.19 | jaytee | brunner, you're best bet is an immediate cutover. I had the option of migrating users in groups instead of all at once but I'm certified on Nortel systems. |
| 00:06.26 | manxpower | But be prepared for the answer for most of your "Can you set the Nortel to do X" questions be answered as "No." |
| 00:06.33 | rob0 | jaytee: what about Kelly? Too young? |
| 00:06.42 | SexyKen | jaytee: I'm not opposed to you licking my balls...if that's what you mean? |
| 00:06.57 | jaytee | rob0, is she past puberty? what's the age of consent in Arkansas? |
| 00:07.01 | *** join/#asterisk egypcio (n=egypcio@unaffiliated/egypcio) |
| 00:07.01 | [TK]D-Fender | SexyKen: "core show application read" , "core show function CALLERID" |
| 00:07.14 | SexyKen | [TK]D: Thanks...:-) |
| 00:07.17 | drmessano | core show function jayteeneedstogetoutmore |
| 00:07.18 | jaytee | SexyKen, whoa!!!! back off, big fella!!! |
| 00:07.30 | jaytee | drmessano, hahahaaa |
| 00:07.34 | rob0 | Age of consent in AR ... well it depends ... I think it was a lot lower when Clinton was here. |
| 00:07.44 | jaytee | hehe |
| 00:07.46 | brunner | yeah, I think I'll hold off on trying to switch to asterisk for now |
| 00:07.49 | *** join/#asterisk _mwoodj_ (n=mwoodj@pdpc/sponsor/digium/hyper-eye) |
| 00:08.04 | brunner | as I don't know anyone who knows much about nortel |
| 00:08.13 | SexyKen | jaytee: ? By your comment above, I was lead to believe that my name was turning your thoughts into erotic thoughts.... |
| 00:08.24 | brunner | and I can't find any good free online resources |
| 00:08.35 | manxpower | brunner: you won't\ |
| 00:08.52 | edibrac | for an asterisk consultant...should I expect him to have a spare digium or sangoma card to test with? |
| 00:09.12 | [TK]D-Fender | brunner: I'd sooner believe your channelized T1 = FXOLS striaght up |
| 00:09.19 | edibrac | like, like of a spare card might indicate he is not as experienced as he claims? |
| 00:09.20 | jaytee | brunner, for Nortel stuff try Tek-Tips, it has a large Nortel section with some very knowledgeable people that'll answer your forum posts. |
| 00:09.48 | brunner | maybe I'll just buy a book |
| 00:10.03 | [TK]D-Fender | brunner: Maybe you should just REPLACE IT |
| 00:10.37 | Corydon76-dig | brunner: You are in a rather unenviable position |
| 00:10.51 | brunner | [TK]D-Fender: I'd like to, but I don't trust my ability to do it successfully |
| 00:10.59 | jaytee | good luck finding a third party Nortel book |
| 00:11.01 | brunner | rather, I don't trust that I'd be able to |
| 00:11.15 | Corydon76-dig | brunner: If you leave the Nortel, you're gonna get fucked by it, and if you replace it too quickly, you'll get fucked by your staff. |
| 00:11.30 | drmessano | Oh man |
| 00:11.31 | brunner | Corydon76-dig: exactly |
| 00:11.34 | jaytee | "there is no try! only do!" "Master Yoda, I can't!" "That is why you fail!" |
| 00:11.36 | Corydon76-dig | brunner: so the real question is, who would you prefer to be fucked by? |
| 00:11.36 | [TK]D-Fender | "For everything else there's consultants" |
| 00:11.37 | drmessano | Now this is getting dirty |
| 00:11.43 | [TK]D-Fender | ..... |
| 00:11.56 | brunner | Corydon76-dig: lol... how did I know you were going to ask that next? |
| 00:12.52 | Corydon76-dig | brunner: it's a lot worse than it seems |
| 00:12.52 | brunner | jaytee: I really can't risk shutting down the company's phone system |
| 00:12.52 | *** join/#asterisk angler (n=angler@pdpc/sponsor/digium/angler) |
| 00:12.52 | *** mode/#asterisk [+o angler] by ChanServ |
| 00:12.52 | [TK]D-Fender | "For everything else there's consultants" <-- |
| 00:13.10 | brunner | [TK]D-Fender: yeah, that cost $175/hr |
| 00:13.21 | Corydon76-dig | brunner: Your best bet is to intercept all calls and run everything in parallel for a period of time |
| 00:13.25 | brunner | maybe if I go on a date with Corydon76-dig, he'll come to Auburn and set it up for me =] |
| 00:13.26 | Corydon76-dig | 2 phones on every desk |
| 00:13.38 | Corydon76-dig | brunner: I already have a bf |
| 00:13.42 | brunner | aww |
| 00:13.46 | brunner | well there goes the cheap route |
| 00:13.47 | jaytee | Auburn? Auburn where? |
| 00:13.52 | Corydon76-dig | Alabama |
| 00:13.53 | brunner | jaytee: Alabama |
| 00:14.02 | jaytee | Stars fell on it I've heard |
| 00:14.22 | Corydon76-dig | brunner: and then slowly pitch all the Nortel phones, and when you're through, you can just turn off the Nortel |
| 00:14.29 | brunner | Corydon76-dig: you mean using a dual-port card? |
| 00:14.52 | Corydon76-dig | brunner: How many T1s are currently provisioned? |
| 00:15.01 | brunner | two |
| 00:15.10 | brunner | but only one for voice |
| 00:15.10 | Corydon76-dig | Then you need a 4port card |
| 00:15.19 | Corydon76-dig | Oh, okay, then just 2 |
| 00:15.24 | brunner | sorry, that was a stupid reponse on my part |
| 00:15.51 | Corydon76-dig | The other T1 is terminated by a Juniper or a Cisco, then? |
| 00:16.04 | brunner | Cisco, yeah |
| 00:16.04 | Corydon76-dig | data T1? |
| 00:16.08 | brunner | data |
| 00:16.32 | [TK]D-Fender | off to martial arts, BBIAB |
| 00:16.36 | jaytee | I put Asterisk with a dual port TE212P card on the outbound span of my PRI and setup my dialplan to search for any extension in one context and impossible match any 4 digit number it couldn't find to pass to Nortel from Asterisk IP phones. All the Nortel phones are migrated by setting them to a permanent external call forward and Asterisk intercepts it. 4 digit extension dialing both ways. |
| 00:16.46 | Corydon76-dig | Right, so make it dual, figure out the signalling, then one weekend or after hours, you put the Asterisk box between the telco and the Nortel and you just pipe all calls directly through |
| 00:17.51 | Corydon76-dig | brunner: then you start adding phones to the Asterisk system, adding dialplan to ring both the Nortel and the Asterisk phones at the same time |
| 00:18.01 | Corydon76-dig | via DIDs |
| 00:18.03 | brunner | ah, I see |
| 00:18.22 | *** join/#asterisk maddog01 (n=minotaur@dhcp-0-1e-58-0-b-f.cpe.mountaincable.net) |
| 00:18.28 | Corydon76-dig | Once you have all the kinks worked out in your SIP phones, you can start turning off Nortel phones, one at a time |
| 00:18.45 | brunner | well that could be fun |
| 00:18.52 | Corydon76-dig | When you have no more nortel phones, you turn off the Nortel unit |
| 00:18.59 | brunner | there'd be no call transferring between users during the migration, right? |
| 00:19.02 | Corydon76-dig | Nice, slow, progressive steps |
| 00:19.11 | brunner | I mean, from nortel phones to SIP |
| 00:19.13 | *** join/#asterisk erth64net (n=erth64ne@69-30-67-191.dq1sn.easystreet.com) |
| 00:19.31 | jaytee | "There's nothing more satisfying to me to watch someone saying something is impossible being interrupted by someone doing that exact same thing." |
| 00:19.34 | Corydon76-dig | brunner: That's the only tricky part |
| 00:20.31 | Corydon76-dig | brunner: basically, you have to program the Nortel to see those phones as being at another location, so that it transfers the calls out its trunk (which you can then intercept) |
| 00:21.02 | brunner | here's another question: there's only one ethernet port in each office. is there some mini-switch I could get that would allow the computer and the phone to share a jack, should I bootleg it and use the 4-wire phone lines as ethernet, or would I need to wire all new ethernet in every office? |
| 00:21.29 | Corydon76-dig | Many phones have passthrough |
| 00:21.34 | brunner | oh nice. |
| 00:21.39 | Corydon76-dig | Even better if your core routing supports VLANs |
| 00:21.44 | brunner | it does |
| 00:22.20 | Corydon76-dig | You can ensure that the phone tags its packets on a separate VLAN (at least with Polycom phones) |
| 00:22.43 | brunner | hmm... I don't really understand the advantage of that |
| 00:23.07 | Corydon76-dig | You're ensuring that phone bandwidth gets priority |
| 00:23.17 | brunner | I see |
| 00:23.32 | Corydon76-dig | over things like fileshares, which can choke the network and increase latency |
| 00:23.42 | jaytee | jjjjjj jitter |
| 00:23.47 | brunner | how do you feel about voip over a data T1 vs. voice T1? |
| 00:23.52 | lanning | you have to have a switch that can do vlan prioritizing |
| 00:24.11 | brunner | lanning: they're Cisco Catalysts |
| 00:24.27 | lanning | shouldn't be a problem. |
| 00:24.29 | Corydon76-dig | brunner: depends on the termination |
| 00:24.29 | jaytee | 802.1Q trunking |
| 00:24.59 | jaytee | with a VLAN for voice. |
| 00:25.13 | Corydon76-dig | brunner: ideally, you want qos on that data T and you want the termination immediately on the other end |
| 00:25.33 | Corydon76-dig | or a separate data T for voice |
| 00:25.54 | Corydon76-dig | Again, you want to ensure that other data doesn't cause packet delays for voice |
| 00:26.33 | jaytee | externally I prefer PRI to the SS7 for business class lines. The internet is too dodgy |
| 00:26.38 | brunner | is it possible to do this while the T1 is still channelized, rather than switching to PRI first? |
| 00:27.07 | Corydon76-dig | brunner: you can use a channelized T1 for voice, sure |
| 00:27.13 | voxter | what is the ultimate difference between pri and ss7? i know you can get more information about the call from ss7, but is it cheaper, or more flexible any other way, etc? |
| 00:27.48 | manxpower | brunner: if you are not under contract many carriers will upgrade you to PRI by installing a 2nd T-1 then turning off the original channelized T-1 when the PRI is tested and working |
| 00:27.51 | jaytee | ss7 is for the telcos, not for businesses........yet and maybe never |
| 00:27.56 | Corydon76-dig | voxter: data channels on SS7 are concurrent |
| 00:28.03 | *** join/#asterisk CapriCoRN^80 (n=int@207.176.6.65) |
| 00:28.07 | *** join/#asterisk sah-work (n=Bawbatos@65-119-47-100.dia.static.qwest.net) |
| 00:28.17 | manxpower | jaytee: SS7 customer connections are common in some parts of the world. |
| 00:28.25 | jaytee | in the US? |
| 00:28.39 | Corydon76-dig | With a PRI, you have a primary and a backup data channel. The backup never passes data until the primary goes offline (and the backup becomes the primary) |
| 00:28.39 | manxpower | jaytee: not in the USA or Canada |
| 00:28.47 | jaytee | thought that had to licensed and the paper trail was murder |
| 00:29.18 | manxpower | jaytee: as I understand it, carriers can limit customer SS7 connections to not have more permissions than a PRO |
| 00:29.29 | Corydon76-dig | Yeah, in the US, SS7 is only common for carriers, not for premise equipment |
| 00:29.51 | Corydon76-dig | although I wonder if that's more about availability of equipment than anything else |
| 00:30.11 | jaytee | it's more about telcos not wanting to give up control |
| 00:30.38 | Corydon76-dig | If nobody has SS7 equipment, then the telcos aren't anxious to invest resources to learn how to provision it |
| 00:31.23 | jaytee | and yet almost every major telco uses SS7 to interconnect to other telcos here in the US |
| 00:31.25 | *** join/#asterisk CapriCoRN^80 (n=int@207.176.6.79) |
| 00:31.29 | CapriCoRN^80 | hi again |
| 00:31.49 | Corydon76-dig | jaytee: yeah, but only the top tier of techs learn how to do that |
| 00:31.53 | CapriCoRN^80 | [TK]D-Fender: sorry but i didnt figure out whats wrong in my dialup plan |
| 00:31.58 | jaytee | Corydon76-dig, true |
| 00:32.07 | Corydon76-dig | Your average telco tech has no clue about it |
| 00:32.09 | *** part/#asterisk Deeewayne (n=Deeewayn@nat/digium/x-53b548d4e897b2e8) |
| 00:32.10 | jaytee | not much in the way of SS7 expertise out there |
| 00:32.20 | CapriCoRN^80 | [TK]D-Fender: need your help |
| 00:33.35 | Corydon76-dig | Guaranteed, if a CLEC in an area started provisioning SS7 to customers, the ILEC (and other CLECs) in that area would learn to do that PDQ |
| 00:34.04 | jaytee | yep, they sure would |
| 00:34.26 | Corydon76-dig | but it wouldn't spread beyond that area without a CLEC's actions |
| 00:34.52 | *** join/#asterisk Nasra (n=maxshipp@CPE001217b1920e-CM00159a010eda.cpe.net.cable.rogers.com) |
| 00:36.37 | CapriCoRN^80 | .. |
| 00:36.46 | jaytee | ... |
| 00:37.03 | CapriCoRN^80 | just checking that i am still on # |
| 00:37.13 | CapriCoRN^80 | actually my irc is doing some problem |
| 00:37.41 | jaytee | did someone just type a blank line using a hidden nick? my screen is doing some funky things. |
| 00:38.25 | brunner | are there any wifi phones that do SIP video? |
| 00:39.06 | jaytee | hold on a second while I piss my pants laughing |
| 00:39.30 | brunner | I'm not looking to do it. I was just curious. |
| 00:40.43 | *** join/#asterisk Thecops (n=Thecops@206-248-136-187.dsl.teksavvy.com) |
| 00:41.06 | Thecops | someone seen this error before? trying to compile zaptel 1.2 on FC10 zaptel-base.c: In function âzt_registerâ: |
| 00:42.00 | jaytee | wow, is that zaptel 1.2.x on Fedora Core 10? |
| 00:42.17 | Thecops | yes |
| 00:42.22 | Thecops | is it too old ? |
| 00:42.35 | jaytee | why not run Microsoft Word 1.0 on Vista instead? |
| 00:43.05 | Thecops | lol, i think you exaggerated a little bit :p |
| 00:43.23 | jaytee | just a teensy weensy bit |
| 00:43.24 | brunner | wtf? GigE on voip phones? |
| 00:43.40 | jaytee | brunner, yeah! got a problem wid dat? |
| 00:43.48 | Thecops | brunner, i guess it have a bridge to chain your phone with your ethernet pc |
| 00:44.23 | jaytee | so do most Polycom phones but they're only 100mbit and work just fine |
| 00:44.48 | Thecops | jaytee, hrmm. do you know if 1.4 zaptel support 1.2 asterisk? |
| 00:45.06 | jaytee | nope, it's 1.2 or upgrade * |
| 00:45.16 | Thecops | ok |
| 00:45.38 | Thecops | err tough it was gonna be an easy migration :P |
| 00:45.39 | jaytee | but why are you installing 1.2 on a new FC10 install. this can't be a legacy system |
| 00:45.45 | bmoraca | jaytee: some people want to chain their desktop to their phone and not lose gigabit speeds. some buildings were not wired with two cat5e/cat6 drops at each location and some places do not want to pay for that kind of port density |
| 00:45.50 | Thecops | Dont have really time to upgrade the full config |
| 00:46.26 | CapriCoRN^80 | [TK]D-Fender: you there ? |
| 00:46.27 | jaytee | Thecops, do you have to use FC10? |
| 00:46.54 | Thecops | No, thats what im gonna do, take the same release as my other server. but i tough i was able to upgrade OS first then asterisk later |
| 00:47.06 | bmoraca | *cough* use RHEL or CentOS |
| 00:47.07 | jaytee | bmoraca, I understand your point, my point was that MOST people don't require gigabit to their desktops. |
| 00:47.25 | Thecops | bmoraca thanks but im ok with fedora |
| 00:47.27 | *** join/#asterisk mattzerah (n=matt@ozvoip.dsl.onthenet.net) |
| 00:47.39 | brunner | asterisk doesn't support wideband codecs, right? |
| 00:47.48 | jaytee | if zaptel won't compile then you're not really ok with fedora |
| 00:48.06 | jaytee | brunner, almost......soon....very soon. |
| 00:48.18 | Thecops | its a version problem, its clear |
| 00:48.22 | Thecops | not a distrib problem |
| 00:48.33 | Thecops | I had fedora on my old server :p |
| 00:48.34 | Thecops | hehe |
| 00:48.39 | brunner | Corydon76-dig: I don't know why you think I'd sleep with you when your open source pbx can't even support wideband codecs |
| 00:48.42 | jaytee | which VERSION of fedora? |
| 00:48.53 | Thecops | 4 |
| 00:49.00 | Thecops | very old version |
| 00:49.05 | Thecops | dont support my RAID controller |
| 00:49.23 | drmessano | G722? |
| 00:49.27 | Thecops | im gonna just buy some analog phone fuck off |
| 00:49.28 | Thecops | :p |
| 00:49.37 | jaytee | you had 4 on your old server and 10 on the new one and you're sure that it's a version problem with Zaptel and not the version of the distro? sure! whatever! |
| 00:49.46 | jaytee | ~wglwat |
| 00:49.47 | jbot | wglwat is, like, well, good luck with all that |
| 00:50.38 | Thecops | Jaytee, its a distro version problem yes. if i take ubuntu or even CentOS i will get into other problem like this. It depend on what kernel it include into the distrib at the moment the release it..it depend on a bunch of stuff hehe |
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| 00:51.42 | jaytee | try running your 1.2 config on an FC8 or FC7 release instead. |
| 00:51.56 | Thecops | Yes i think i have no choice. |
| 00:52.07 | CapriCoRN^80 | can anyone paste me one line setting of its NAT user from dialup plan |
| 00:52.26 | Thecops | i love so much fiber optic when im about to download stuff |
| 00:52.28 | Thecops | thanks god |
| 00:52.41 | bmoraca | Thecops: fedora core is a desktop architecture. red hat makes RHEL for servers for a reason. basically, that reason is that fedora sucks. |
| 00:54.10 | Thecops | bmoraca, it depend on what you install as package, how you tweak your kernel. How you compile your software. I think each distribution is customizable on your need. it depend on you hehe |
| 00:54.39 | jaytee | ~wglwat |
| 00:54.40 | jbot | i guess wglwat is well, good luck with all that |
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| 00:55.02 | bmoraca | Thecops: right...but why bend over backwards tweeking fedora to act like RHEL when RHEL already acts like RHEL? |
| 00:55.05 | prxtien | has anyone got experience with mp4play or app_rtsp, i keep getting segfaults with both of them |
| 00:55.09 | Mark17 | is it possible to include a file in the extensions.conf? so it would be used like all content from that file was located in extensions.conf (with the context where you include it)? |
| 00:55.37 | brunner | Corydon76-dig: so I just need a TE207P to start tinkering around, right? |
| 00:56.02 | bmoraca | that's an expensive toy |
| 00:56.16 | Thecops | bmoraca I think no one know if zaptel 1.2 will working in my context even if i have RHEL |
| 00:56.22 | brunner | holy shit, I hadn't looked at the price yet |
| 00:56.39 | jaytee | not as expensive as that Legos Robotics kit |
| 00:56.47 | Thecops | lol |
| 00:57.01 | bmoraca | jaytee: i have one of the original Lego Mindstorms kits...that thing was freakin awesome |
| 00:57.13 | brunner | the TE205P is $500 on ebay |
| 00:57.18 | jaytee | they were, and the new ones are incredible too |
| 00:57.27 | bmoraca | brunner: do you need dual T1/E1 connectivity? |
| 00:57.45 | brunner | but I need the echo-cancellation, right? |
| 00:57.46 | jaytee | my TE212P was 1295.00 with HWEC module |
| 00:57.48 | brunner | bmoraca: yes |
| 00:58.01 | jaytee | worth every penny |
| 00:58.03 | edibrac | anyone heard anything bad about the TE121? |
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| 00:58.19 | bmoraca | brunner: then you're up the creek...dual T1 cards ain't cheep |
| 00:58.25 | bmoraca | god i can't spell today |
| 00:58.25 | Thecops | woot FC5 working with my raid controller |
| 00:58.42 | edibrac | i sort of want to shy away from PCI-E since I only have one slot |
| 00:59.01 | brunner | Corydon76-dig: is there any reason I can't buy two single-T1 cards? |
| 00:59.10 | bmoraca | brunner: I've had good luck with the Samgoma A102D two port w/ echo cancel |
| 00:59.22 | bmoraca | brunner: a 2 port will be marginally cheaper than two 1 port |
| 00:59.37 | bmoraca | brunner: also, in that configuration, you'll need the timing backplane/cable |
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| 01:02.36 | bmoraca | brunner: TE207P is $1467 from Voiplink |
| 01:03.34 | dlewis | nice... |
| 01:03.36 | dlewis | NOT!!!!!! |
| 01:04.14 | *** join/#asterisk ZX81 (n=matt@202.49.106.158) |
| 01:05.53 | mattzerah | question: do i stay with zaptel drivers, or update to dahdi ? anyone have an informed position on this ? |
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| 01:09.52 | brunner | okay, so $1000 for the card from ebay, $2000 for 20 business phones, $1000 for the asterisk box, $1000 for consulting when I get stuck. what other expenses would there be? |
| 01:10.09 | *** join/#asterisk [intra]lanman (n=Raymond@freeswitch/developer/intralanman) |
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| 01:14.41 | Thecops | brunner, cabling, power over ethernet, all those good stuff :) |
| 01:14.45 | Thecops | a ethernet switch :) |
| 01:20.16 | prxtien | ast_rtp_write: Don't know how to send format unknown packets with RTP |
| 01:20.19 | prxtien | mm thats an interesting one |
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| 01:25.02 | brunner | oh yeah... PoE.... how is that done with 20 phones? |
| 01:25.18 | brunner | Thecops: I was hoping to be able to use my Cisco Catalyst switches |
| 01:25.41 | *** join/#asterisk maddog01 (n=minotaur@dhcp-0-1e-58-0-b-f.cpe.mountaincable.net) |
| 01:26.14 | Thecops | is it PoE? |
| 01:26.39 | Thecops | You can definately use your Catalyst switches for data communication |
| 01:27.07 | Thecops | also for voice on different vlan |
| 01:27.23 | brunner | I mean, physically |
| 01:27.25 | Thecops | PoE is nice when its integrated to the switch |
| 01:27.31 | dlewis | yup |
| 01:27.52 | brunner | so I'd have to feed everything through a new device that looks like a switch in order to give it power? |
| 01:28.01 | *** join/#asterisk Khratos (n=Khratos@190.166.130.103) |
| 01:28.03 | dlewis | i have one of these: http://www.netgear.com/Products/Switches/AdvancedSmartSwitches/GS748TP.aspx |
| 01:28.19 | Thecops | its better that way brunner |
| 01:28.30 | dlewis | if the switch suppors 802.3af, then most PoE phones will work.. |
| 01:28.34 | dlewis | and PoE devices. |
| 01:28.36 | Thecops | you can put a UPS on the switch then all your network is on UPS |
| 01:28.50 | Khratos | Good evening |
| 01:29.08 | Thecops | Good evening |
| 01:29.11 | dlewis | Thecops: tough to do that, but a good idea... |
| 01:29.37 | Thecops | i did for a call center, its the best |
| 01:29.40 | brunner | I doubt mine supports 802.3af |
| 01:30.53 | brunner | it's about ten years old |
| 01:31.04 | Thecops | 2924? |
| 01:31.09 | brunner | I think so, actually, yeah |
| 01:31.14 | Thecops | lol |
| 01:31.19 | Thecops | I love that switch |
| 01:31.34 | Thecops | but unfortunately it dont support 802.3af |
| 01:31.43 | brunner | it's been grat |
| 01:31.46 | brunner | great* |
| 01:32.01 | brunner | I suppose the new Catalyst switches support it? |
| 01:32.12 | Thecops | Depend on what model you get |
| 01:32.17 | Thecops | but its expensive |
| 01:32.24 | Thecops | its professional |
| 01:33.22 | Thecops | Netgear is less expensive, maybe a good way to go |
| 01:33.22 | eppigy | hello |
| 01:33.27 | brunner | shit, I've got to get something done before 8CT. brb |
| 01:33.40 | eppigy | lol |
| 01:33.59 | Thecops | jaytee awsome all working with FC5 lol |
| 01:34.06 | Thecops | all compiled perfectly |
| 01:34.43 | jaytee | so basically I was totally wrong and it had absolutely nothing to do with too new a kernel then, thank god! |
| 01:34.58 | eppigy | YES |
| 01:35.20 | jaytee | thanks dave, knew I could count on you for moral support :-) |
| 01:35.58 | Thecops | jaytee i guess its a too big difference with zaptel 1.2 and 2.6.27 kernel :) I should be stop being lazy and upgrade my * |
| 01:35.58 | Thecops | lol |
| 01:36.53 | jaytee | read all the upgrade 1.2-1.4 text files and the upgrade file. A LOT of things were changed between 1.2 and 1.4. |
| 01:37.08 | jaytee | alot got changed and alot got deprecated |
| 01:37.14 | Thecops | yes |
| 01:37.18 | Thecops | definately |
| 01:37.36 | Thecops | awsome sangoma kernel module successfully installed |
| 01:38.16 | jaytee | be especially aware of anything in your old dialplan that uses pattern matches and references the old callerid app. it's been deprecated in 1.4 |
| 01:38.26 | Thecops | ok |
| 01:38.53 | jaytee | um, I meant also anything that references the old callerid app, not the pattern matches referencing it. |
| 01:39.23 | *** join/#asterisk DrAk0 (n=luisjose@nelug/coreteam/luisjose) |
| 01:39.25 | jaytee | those are two major differences offhand that would break your dialplan or cause you alot of grief at a minimum |
| 01:39.28 | Thecops | I will probably just make a ghost of that server, and try 1.6 a week-end |
| 01:39.51 | jaytee | 1.6 doesn't use zaptel, it's dahdi only |
| 01:40.11 | Thecops | Oh ok |
| 01:40.20 | Thecops | Sangoma was talking about that i guess its suported |
| 01:41.50 | jaytee | Thecops, most of the changes in DAHDI are in name only and was done due to a trademark issue with another company that had trademarked zaptel. |
| 01:42.41 | *** join/#asterisk docelmo (n=vircuser@pool-141-152-217-234.lyn.east.verizon.net) |
| 01:42.43 | Thecops | oh seriously |
| 01:42.45 | Thecops | thats bullshit |
| 01:43.29 | Thecops | its like NIS server. They got into problem when they had the name YP. It was the acronym for Yellow Page |
| 01:44.03 | jaytee | well, there are also improvements in DAHDI over zaptel. like if you want to change a software based echo canceller from mg2 to one of the others you don't have to recompile DAHDI like you did zaptel. |
| 01:44.47 | Thecops | oh |
| 01:44.57 | Thecops | thats a good feature |
| 01:45.27 | Thecops | i was pissed off about recompiling zaptel to fix my echo problem and trying different echo algorythm |
| 01:45.35 | Thecops | its nice |
| 01:45.49 | jaytee | yeah, dynamic loading by changing a parameter in chan_dahdi.conf and restarting is much nicer than having to recompile. |
| 01:47.10 | *** join/#asterisk CapriCoRN^80 (n=int@207.176.6.79) |
| 01:47.12 | CapriCoRN^80 | hi |
| 01:47.22 | CapriCoRN^80 | [TK]D-Fender: need help :) |
| 01:47.49 | jaytee | Now dammit!!! And don't give me any shit!!! :-) |
| 01:50.11 | icebrew54 | lol |
| 01:51.54 | eppigy | YEAH SON |
| 01:51.55 | CapriCoRN^80 | need some help dialup plan for NAT user |
| 01:52.15 | CapriCoRN^80 | if anyone can paste single line of there NAT user of dialup |
| 01:52.43 | jaytee | ~sipnat |
| 01:52.43 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 01:53.03 | CapriCoRN^80 | jbot: i did that |
| 01:53.03 | jbot | You did that? |
| 01:53.04 | CapriCoRN^80 | :P |
| 01:53.16 | jaytee | jbot botsnack |
| 01:53.16 | jbot | jaytee: thanks |
| 01:53.55 | *** join/#asterisk maddog01 (n=minotaur@dhcp-0-1e-58-0-b-f.cpe.mountaincable.net) |
| 01:55.10 | jaytee | I like the bot in the ubuntu forums. ya go "botsnack" and it goes "Yum! um, er..... I mean apt-get!" |
| 01:55.31 | eppigy | wow |
| 01:55.54 | lanning | funny |
| 01:55.57 | jaytee | plus it knows alot of good Chuck Norris and Mr T one liners |
| 01:56.03 | CapriCoRN^80 | [external] |
| 01:56.03 | CapriCoRN^80 | exten => 401,Dial(SIP/sajid) |
| 01:56.19 | CapriCoRN^80 | this is my nat user dialup plan |
| 01:56.34 | *** join/#asterisk coppice (n=chatzill@201.193.17.210.dyn.pacific.net.hk) |
| 01:56.34 | CapriCoRN^80 | but [TK]D-Finder: told me that there is some problem in it |
| 01:56.37 | eppigy | are you smoking purp by the pound |
| 01:56.39 | CapriCoRN^80 | anyone help me ? |
| 01:56.44 | lanning | "dial plan" |
| 01:56.49 | jaytee | hahahahaaaa purp by the pound hahahahaahaa |
| 01:56.52 | frogonwheels | CapriCoRN^80: read the book |
| 01:56.56 | frogonwheels | ~thebook |
| 01:56.56 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com |
| 01:57.21 | brunner | why is it so hard to make hardware for the nortel phones? |
| 01:57.25 | *** join/#asterisk D3b|4n (n=D3b_4n@unaffiliated/lynxnica) |
| 01:57.36 | eppigy | brunner: depression runs high |
| 01:57.41 | eppigy | among their fabricators |
| 01:57.49 | eppigy | lack of motivation |
| 01:57.58 | D3b|4n | any talk in spanish? |
| 01:57.59 | eppigy | social withdrawal |
| 01:58.00 | frogonwheels | CapriCoRN^80: you are missing the priority. |
| 01:58.06 | eppigy | D3b|4n: TRABAJO |
| 01:58.11 | eppigy | DONDE ESTA |
| 01:58.17 | D3b|4n | nicaragua |
| 01:58.17 | RobH | ~book |
| 01:58.18 | jbot | from memory, book is probably Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://tfot.leifmadsen.com or see ~buybook |
| 01:58.18 | eppigy | NO ME GUSTA |
| 01:58.26 | jaytee | brunner, it's proprietary closed system stuff that requires costly licensing to produce and to create compatible interface systems. |
| 01:58.31 | eppigy | ESTA NOCHE |
| 01:58.38 | CapriCoRN^80 | frogonwheels: please tell me in detail |
| 01:58.40 | frogonwheels | ~x es en No me gusta |
| 01:58.48 | CapriCoRN^80 | i shall be very thanks full |
| 01:58.51 | eppigy | no viva aque |
| 01:58.59 | D3b|4n | necesito un poco de ayuda |
| 01:59.02 | eppigy | CapriCoRN^80: read the book dude |
| 01:59.08 | eppigy | get your mind right |
| 01:59.08 | frogonwheels | CapriCoRN^80: Go through the beginning chapters in the book |
| 01:59.15 | CapriCoRN^80 | oh come on |
| 01:59.27 | jaytee | Chapter 5 especially |
| 01:59.29 | frogonwheels | CapriCoRN^80: Well, you wanted it in detail. |
| 01:59.30 | eppigy | necesito chiva |
| 01:59.40 | lanning | http://www.catb.org/~esr/faqs/smart-questions.html |
| 01:59.43 | *** join/#asterisk mercutioviz (n=michaelc@freeswitch/developer/msc) |
| 01:59.44 | eppigy | un cocaina |
| 01:59.47 | D3b|4n | lol |
| 01:59.49 | frogonwheels | ~x es en un cocaina |
| 01:59.59 | D3b|4n | eppigy puedes ayudarme tengo una consulta |
| 02:00.01 | jaytee | CapriCoRN^80, if you don't understand what a PRIORITY is in the dialplan you need to go read up about it and not waste people's time in here. |
| 02:00.02 | frogonwheels | ;) |
| 02:00.12 | frogonwheels | ~x es en eppigy puedes ayudarme tengo una consulta |
| 02:00.58 | CapriCoRN^80 | jaytee: thanks for telling |
| 02:01.03 | eppigy | ;{ indeed |
| 02:01.08 | frogonwheels | :) |
| 02:01.08 | D3b|4n | need help |
| 02:01.30 | eppigy | no habla espanol me amigo |
| 02:01.48 | frogonwheels | CapriCoRN^80: I gave you the keyword - once you start reading, you'll work it out soon enough. |
| 02:01.48 | eppigy | SMOKE PURP BY THE POUND |
| 02:02.12 | icebrew54 | PURP! |
| 02:02.15 | D3b|4n | :S |
| 02:02.21 | icebrew54 | who got that w0rk? |
| 02:02.34 | jaytee | I saw some 510 Kush that looked like it was neon yellow-green, shit looked outright scary. |
| 02:02.36 | eppigy | true |
| 02:02.53 | eppigy | i like to get chill |
| 02:03.00 | eppigy | and let my veins fill with hate |
| 02:03.35 | jaytee | "all that hate's gonna burn you up inside!" "Nah, it keeps me warm." |
| 02:03.42 | eppigy | yes |
| 02:03.47 | eppigy | it motivates me |
| 02:03.52 | eppigy | gives me focus |
| 02:03.55 | jaytee | and your name is dave! |
| 02:04.02 | eppigy | yes |
| 02:04.05 | eppigy | Dave Casey |
| 02:04.22 | jaytee | he's almost infamous |
| 02:07.28 | *** join/#asterisk n3hxs (n=HAMming@pool-173-66-115-234.washdc.fios.verizon.net) |
| 02:08.41 | CapriCoRN^80 | ok i got that but my real problem is NAT |
| 02:08.51 | CapriCoRN^80 | and i just defined priorities |
| 02:09.03 | CapriCoRN^80 | 1 for dial and 2 for hangup |
| 02:10.12 | frogonwheels | CapriCoRN^80: you need to show us pastebin of (password obfuscated) sip.conf and output from the asterisk console when you try. |
| 02:10.25 | lanning | the client must register to *, and you need to setup sip.conf with NAT options. |
| 02:10.52 | lanning | You must also make sure the NAT device handles SIP and RTP correctly |
| 02:10.59 | CapriCoRN^80 | error 408 from softphone |
| 02:11.02 | CapriCoRN^80 | yea i will do that |
| 02:11.27 | RobH | can anyone suggest what the maximum qualification in ms would be usable for sip? |
| 02:11.41 | RobH | isnt really sure of what range it becomes crap cuz all his users are local |
| 02:11.47 | RobH | all my 'current' users are local |
| 02:11.53 | RobH | cuz i have not enabled the remote ones to use the sysetm |
| 02:13.48 | *** join/#asterisk John-B (n=xpl@84.126.205.1.dyn.user.ono.com) |
| 02:14.16 | keebler | bmoraca: Well, I did it. I went ahead and ordered the EZGO's. |
| 02:14.25 | *** join/#asterisk beek (n=klinebl@pdpc/supporter/professional/beek) |
| 02:15.06 | CapriCoRN^80 | <PROTECTED> |
| 02:15.07 | CapriCoRN^80 | take care |
| 02:15.07 | CapriCoRN^80 | bye |
| 02:20.30 | John-B | hi there ... |
| 02:21.02 | John-B | adad2008*CLI> sip show registry |
| 02:21.03 | John-B | No such command 'sip show' (type 'help' for help) |
| 02:21.03 | John-B | adad2008*CLI> |
| 02:21.27 | John-B | can you help me guys ? |
| 02:22.03 | frogonwheels | John-B: and help sip |
| 02:22.25 | frogonwheels | John-B: is sip module loaded? |
| 02:23.56 | *** join/#asterisk stevetotaro (n=Steve@pool-71-254-231-87.hrbgpa.east.verizon.net) |
| 02:24.45 | *** join/#asterisk JafoJ (n=0wntmin@0wnt.com) |
| 02:25.24 | John-B | frogonwheels : i dont think so.. |
| 02:25.34 | John-B | when i type reload i have some errors... :( |
| 02:25.41 | frogonwheels | John-B: well there ya are :) |
| 02:27.06 | John-B | [Jan 28 11:12:33] NOTICE[32264]: pbx_ael.c:4114 pbx_load_module: AEL load process: compiled config file name '/etc/asterisk/extensions.ael'. |
| 02:27.06 | John-B | [Jan 28 11:12:33] NOTICE[32264]: pbx_ael.c:4117 pbx_load_module: AEL load process: merged config file name '/etc/asterisk/extensions.ael'. |
| 02:27.06 | John-B | [Jan 28 11:12:33] NOTICE[32264]: pbx_ael.c:4120 pbx_load_module: AEL load process: verified config file name '/etc/asterisk/extensions.ael'. |
| 02:27.50 | John-B | [Jan 28 11:12:33] WARNING[32264]: res_smdi.c:748 reload: No SMDI interfaces were specified to listen on, not starting SDMI listener. |
| 02:28.39 | frogonwheels | John-B: hmm. not looking relevent to me. try module load chan_sip.so on the console |
| 02:28.40 | seanbright | John-B: those are not errors, they are notices and warnings |
| 02:30.15 | *** join/#asterisk Avelino (n=Avelino@189-19-159-147.dsl.telesp.net.br) |
| 02:31.46 | John-B | what i most do ? |
| 02:32.11 | frogonwheels | John-B: module load chan_sip.so |
| 02:33.27 | John-B | now is working |
| 02:33.30 | John-B | thanks a lot |
| 02:33.40 | frogonwheels | John-B: need to enable it in your modules.conf now |
| 02:37.58 | *** join/#asterisk riddlebox (n=victoria@75-132-225-75.dhcp.stls.mo.charter.com) |
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| 03:15.10 | prxtien | http://pastebin.com/m17aa5f8e , can anyone help me with this rstp streaming error... i can view the stream no worries from vlc |
| 03:15.15 | prxtien | but asterisk refuses to acknowledge it |
| 03:16.51 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 03:17.16 | *** join/#asterisk Iamnacho (i=Iamnacho@ip68-103-153-140.ks.ok.cox.net) |
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| 03:48.12 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
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| 04:03.46 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 04:03.51 | *** join/#asterisk troubled (n=troubled@unaffiliated/troubled) |
| 04:08.33 | Khratos | ... going to sleep |
| 04:08.44 | *** part/#asterisk Khratos (n=Khratos@190.166.130.103) |
| 04:13.59 | *** join/#asterisk Coolthreads (n=Coolthre@203-97-238-71.cable.telstraclear.net) |
| 04:14.00 | *** join/#asterisk russellb (n=russell@asterisk/digium-open-source-team-lead/russellb) |
| 04:14.00 | *** mode/#asterisk [+o russellb] by ChanServ |
| 04:14.38 | *** join/#asterisk ZenBSDi (n=ZenBSDi@unaffiliated/ZenBSDi) |
| 04:15.35 | ZenBSDi | So, Who wants to join me in starting up a managed pbx company? I just registered the domain mymanagedpbx.com :) |
| 04:15.57 | russellb | step 1) buy domain. step 2) ??? step 3) PROFIT!!1 |
| 04:16.13 | *** join/#asterisk thepacmanfan (n=andrewk@173-22-139-185.client.mchsi.com) |
| 04:16.29 | ZenBSDi | step 1 is done .. step 2 .. I have a dual opteron 1u with 4 gigs of ram and 160 gigs of diskspace tahts going to a colo tomorrow |
| 04:16.57 | thepacmanfan | is there any way to change the caller id on a blind transfer? |
| 04:16.59 | ZenBSDi | probably gonna use packet8 for sip peering for trunks |
| 04:17.22 | [TK]D-Fender | thepacmanfan: In dialplan only |
| 04:17.26 | ZenBSDi | thepacmanfan, in .. |
| 04:17.33 | thepacmanfan | or is there any other way to distinguish already-answered calls from fresh incoming calls? |
| 04:17.38 | ZenBSDi | darn it.. TKD .. let some of us have SOME glory :p |
| 04:18.06 | thepacmanfan | ok, here's how my dialplan is set up |
| 04:18.07 | ZenBSDi | thepacmanfan, its all in the extensions.conf man.. learn how to script your dialplans and you can do almost anything in asterisk |
| 04:18.09 | [TK]D-Fender | thepacmanfan: Go read channelvariables.txt |
| 04:18.57 | ZenBSDi | tkd .. so I finally started playing around with database put's and gets .. man is that helpful!!!! |
| 04:19.46 | jaytee | russellb, :-) |
| 04:19.57 | thepacmanfan | incoming analog calls get a CID set to them, then they ring all my sip phones. once a call is answered, the CSR then will blind xfer it to another phone. |
| 04:20.02 | [TK]D-Fender | ZenBSDi: Step 2.5 learn enough more about * that things like DB's aren't considered major acheivements :p |
| 04:20.34 | jaytee | OMG! you mean we can use daterbasers on this schtuff? |
| 04:20.36 | thepacmanfan | catch is, i need to distinguish an incoming "ringall" call from a blind xfer that's coming from another phone |
| 04:20.48 | [TK]D-Fender | thepacmanfan: Go read channelvariables.txt <--------- |
| 04:21.12 | drmessano | YES! |
| 04:21.16 | drmessano | I GOT SIP WORKING |
| 04:21.29 | jaytee | SIP TCP? With UM? |
| 04:21.42 | ZenBSDi | lol .. exten => s,1,READ(channelvariables.txt) |
| 04:21.44 | drmessano | No, SIP |
| 04:21.50 | drmessano | I cant hook phones up now! |
| 04:21.54 | drmessano | can* |
| 04:21.59 | jaytee | pfffft |
| 04:22.09 | jaytee | had me psyched there for a second |
| 04:22.35 | ZenBSDi | drmessano, congrats.. who's your sip peer? |
| 04:22.37 | drmessano | SIP TCP?? HA |
| 04:22.39 | [TK]D-Fender | jaytee: News!!! Boby Dylan just did a studio show for MTV Unplugged! |
| 04:22.51 | thepacmanfan | [TK]D-Fender: i just skimmed it. how are variables going to do me any good when the phone pretty much bypasses Asterisk on a blind xfer? |
| 04:22.58 | drmessano | Everyone knows Asterisk doesnt support SIP TCP |
| 04:22.59 | [TK]D-Fender | jaytee: They unplugged his ears... he can sing now! |
| 04:23.20 | thepacmanfan | i'm scratching my head here, and i don't know how to get or set variables on a call once it's been answered. |
| 04:23.24 | jaytee | Boby Dylan? the guy with terminal cancer of the adenoids? |
| 04:23.29 | [TK]D-Fender | thepacmanfan: I tell you go read it.. TWICE, then you tell me skimmed it and ask me "what good will it do"? |
| 04:23.40 | [TK]D-Fender | thepacmanfan: Your skimming obviously MISSED STUFF. |
| 04:23.54 | [TK]D-Fender | thepacmanfan: so "duh" GO READ |
| 04:23.59 | thepacmanfan | ok, thanks. |
| 04:24.04 | jaytee | "learnin's hard!" |
| 04:24.10 | ZenBSDi | thepacmanfan, here's a free tip .. when you transfer to say exten 200 .. you should have exten => 200,1,Dial(SIP/200) in your plan.. well if you want to change the number .. then learn how to SET(variables) :p |
| 04:25.02 | ZenBSDi | so now .. for example if I want the callerid to say from 1800playboy exten => 200,1,SET(CALLERID(num)=1800playboy) exten => 200,2,Dial(SIP/200) |
| 04:25.19 | jaytee | man, Marisa Tomei gives me instant wood |
| 04:25.40 | ZenBSDi | I'm probably off in my dialplan cause I'm just coming up with this off top of my head.. but thats the jist of it.. so it IS in the dialplan .. now finally will you READ channelvariables.txt |
| 04:25.44 | ZenBSDi | *huff huff huff* |
| 04:26.03 | thepacmanfan | ZenBSDi: the problem is, that doesn't distinguish between blind transfers, internal calls, or ringall calls. |
| 04:26.12 | thepacmanfan | maybe my ringall isn't set up right... |
| 04:26.16 | ZenBSDi | I think so |
| 04:27.01 | thepacmanfan | hmm |
| 04:27.12 | jaytee | but if you read the damn channel variables file you might (although it's looking less and less likely) be able to figure out how to distinguish between blind transfers, ringall calls and internal calls |
| 04:27.13 | thepacmanfan | however, my ringall is ringing the phones, not hitting the extensions |
| 04:27.31 | thepacmanfan | so that may just be the ticket after all |
| 04:27.50 | ZenBSDi | personally I wouldn't use a ringall in the default context .. I do a [in] to [menu] and if they hit 0, it's a long list of dial(sip/200&sip/201&sip/202) and then in the default have the 2XX,1,Dial(sip/${EXTEN}) so I have more dynamics to work with .. |
| 04:27.59 | jaytee | it's not a winning ticket, that much is for certain |
| 04:28.16 | ZenBSDi | but hey.. I'm an asterisk n00b :p |
| 04:28.27 | thepacmanfan | heh |
| 04:29.20 | jaytee | personally I like to read about stuff before I go asking other people questions that might appear "elementary" |
| 04:29.47 | ZenBSDi | pastebin.ca .. tons of extensions.conf to drool over :p |
| 04:32.30 | ZenBSDi | thepacmanfan, something I do is.. I grab the callerid early .. ;exten => ${MAIN},1,Set(GLOBAL(ICID)=${CALLERID(num)}) so now I have an early global with their phone number for the duration of their call session. I do this cause when I have asterisk forward out to my cell phone.. I want THEIR cid and not my asterisk box's showing up on my cellphone. |
| 04:33.17 | [TK]D-Fender | ZenBSDi: Global.... because its not like ANOTHER call might enter the system at any time and FUBAR that idea ;) |
| 04:33.54 | ZenBSDi | hahah.. I figured I'd throw that out there in hopes someone would point out just that.. |
| 04:34.00 | ZenBSDi | so just set ICID? |
| 04:34.14 | ZenBSDi | I knew there was a reason I commented that out :p |
| 04:34.23 | thepacmanfan | haha |
| 04:34.58 | thepacmanfan | astdb would probably work great for that... i dunno |
| 04:35.11 | thepacmanfan | OTOH, that's probably not a good use for astdb |
| 04:35.31 | ZenBSDi | naw.. astdb is better for like a call forwarding module .. |
| 04:35.53 | *** join/#asterisk xuser (n=jaood@unaffiliated/xuser) |
| 04:36.15 | thepacmanfan | afraid i can't help ya. |
| 04:36.22 | ZenBSDi | prompt "What number do you wish to forward to?" READ(num) and then SET(dbput(agent/ext/number)) |
| 04:36.38 | thepacmanfan | i'm gonna read channelvariables.txt tomorrow when i'm less tired |
| 04:36.57 | D3b|4n | i need help |
| 04:37.18 | D3b|4n | my problem is that if I make a call to a particular number once the recording is stored well. But if I call that same number again the second recording overwrittes the first recording, so i only have the last call |
| 04:37.36 | *** join/#asterisk _gm (n=gmustafa@202.133.78.60) |
| 04:38.19 | [TK]D-Fender | ZenBSDi: SET(dbput(agent/ext/number)) <--- ... pardon? |
| 04:38.44 | [TK]D-Fender | reaches for his ClueBat (tm) |
| 04:39.07 | *** join/#asterisk fish-bulb (n=cstewart@nat/digium/x-e14f3b7de37f5daa) |
| 04:39.08 | thepacmanfan | i just tried your idea of setting the CID for each extension, i.e. 6001,s,1 Set(CALLERID(num)=123), etc, but blind xfers are still retaining the original caller id |
| 04:39.10 | *** join/#asterisk angler (n=angler@pdpc/sponsor/digium/angler) |
| 04:39.10 | *** mode/#asterisk [+o angler] by ChanServ |
| 04:39.12 | *** join/#asterisk mnicholson (n=mnichols@nat/digium/x-77f0d8024e296f96) |
| 04:39.22 | *** join/#asterisk jpeeler (n=jpeeler@asterisk/digium-software-dev/jpeeler) |
| 04:39.26 | [TK]D-Fender | D3b|4n: Go look at what file you are coosing to record under and ask yourself why its the SAME and what you might like it to be so that you can differentiate your recordings. |
| 04:41.46 | D3b|4n | <PROTECTED> |
| 04:42.15 | D3b|4n | so if i make two calls at a different time I should have two recordings |
| 04:42.24 | D3b|4n | but I´m only getting the last one |
| 04:42.24 | eppigy | hello |
| 04:42.46 | D3b|4n | it seems that for some reason the time of the call is not being take in consideration when saving the recording |
| 04:42.56 | [TK]D-Fender | D3b|4n: PASTEBIN is your friend.... |
| 04:43.15 | [TK]D-Fender | D3b|4n: maybe you could show us what you're doing and we can show you where its going wrong. |
| 04:43.17 | [TK]D-Fender | ~pb |
| 04:43.18 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste, or , http://bin.cakephp.org/ |
| 04:43.18 | *** join/#asterisk iWill (n=kvirc@c-24-60-174-6.hsd1.ma.comcast.net) |
| 04:43.29 | eppigy | BEUNO |
| 04:44.10 | *** join/#asterisk `Stevo (n=XXCCZ@CPE-121-209-4-96.sa.bigpond.net.au) |
| 04:44.13 | ZenBSDi | pastebin.ca!!! |
| 04:45.54 | D3b|4n | esta es mi confi http://rafb.net/p/98jOXD65.html |
| 04:45.59 | D3b|4n | this is my confi |
| 04:47.51 | [TK]D-Fender | D3b|4n: Right off I don't trust the contents of your ${Calltime} var |
| 04:49.01 | [TK]D-Fender | D3b|4n: Go look at whats getting executed at CLI. |
| 04:50.51 | *** join/#asterisk terracon (n=greisky@CPE001cf097a750-CM0012254076d6.cpe.net.cable.rogers.com) |
| 04:53.12 | *** part/#asterisk iWill (n=kvirc@c-24-60-174-6.hsd1.ma.comcast.net) |
| 04:54.05 | *** join/#asterisk dmz (n=dmz@adsl-76-246-149-102.dsl.pltn13.sbcglobal.net) |
| 04:54.07 | D3b|4n | http://rafb.net/p/8HmGEE95.html |
| 04:55.13 | D3b|4n | my cli |
| 04:56.11 | [TK]D-Fender | D3b|4n: And the actual file created was? |
| 04:56.45 | D3b|4n | 4804445129_Tue Jan 27 19:46:45 2009_8003_Bayardo Sanchez.wav |
| 04:57.26 | D3b|4n | but i call to 4804445129 erase the old wav and save the new call |
| 04:57.51 | [TK]D-Fender | D3b|4n: Now why would the DATE & TIME be the same? |
| 04:58.18 | [TK]D-Fender | D3b|4n: That'd be the only way, right? |
| 04:59.11 | *** join/#asterisk PaulAviles (n=PaulAvil@dsl-7-36.cofs.net) |
| 04:59.21 | PaulAviles | hello all |
| 04:59.47 | thepacmanfan | ZenBSDi: your suggestion did the trick for me after all. thanks! |
| 04:59.55 | thepacmanfan | one last question tonight. |
| 05:00.10 | PaulAviles | I have a remote phone and I can dial form there to anywhere, but when from the office where the asterisk server is we cannot dial to that extension, says user is not available |
| 05:00.35 | thepacmanfan | on my cisco 7960s, caller id used to read "from asterisk" on one line and "asterisk" on the second line |
| 05:00.51 | thepacmanfan | hold on... i may have just answered my own question. |
| 05:01.24 | *** part/#asterisk ultrav1olet (n=telnet@94.180.4.213) |
| 05:01.41 | drmessano | I have not yet begun to defile myself! |
| 05:01.43 | [TK]D-Fender | PaulAviles: read up : |
| 05:01.45 | [TK]D-Fender | ~sipnat |
| 05:01.46 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 05:01.47 | [TK]D-Fender | ^^^^^^^^^^ |
| 05:03.53 | PaulAviles | jobt, thanks I know that part |
| 05:04.14 | PaulAviles | I think my issue is the dumb netopia router on that remote end |
| 05:04.29 | PaulAviles | the pinholes i have are the basic that crappy unit can support |
| 05:04.39 | thepacmanfan | is there a way to set a blank caller id number value? |
| 05:04.47 | thepacmanfan | <> and < > don't do it. |
| 05:05.47 | PaulAviles | the only issue is for the main office to call the remote |
| 05:06.02 | PaulAviles | how can I check if the traffic is being block in asterisk? |
| 05:06.04 | jaytee | Set(CALLERID(all)="") |
| 05:06.55 | [TK]D-Fender | PaulAviles: so far I don't see you having pastebinned a failed call with SIP debug so we can see what happens.... or your configs |
| 05:07.09 | thepacmanfan | jaytee, i want to send name data, just leave the number blank. |
| 05:07.38 | thepacmanfan | so i'm doing Set(CALLERID(all)=Incoming Call<>) |
| 05:07.39 | jaytee | wow! how about Set(CALLERID(num)="") |
| 05:07.57 | PaulAviles | let me paste then |
| 05:09.08 | PaulAviles | pastebin.com/m6dacc66d |
| 05:09.10 | drmessano | I'll be your huckleberry |
| 05:09.40 | thepacmanfan | jaytee, when i leave it blank, any old CID data is inherited |
| 05:10.23 | jaytee | well, that sure sucks |
| 05:10.34 | thepacmanfan | not the end of the world... just annoying. |
| 05:10.48 | thepacmanfan | it looks like i have to put SOMETHING there. i can do that. |
| 05:12.56 | *** join/#asterisk zafar_ (n=IceChat7@116.71.221.106) |
| 05:13.21 | PaulAviles | http://pastebin.ca/1320324 |
| 05:13.32 | *** part/#asterisk `Stevo (n=XXCCZ@CPE-121-209-4-96.sa.bigpond.net.au) |
| 05:13.38 | PaulAviles | TK, that is with sip debug peer 306 |
| 05:13.58 | PaulAviles | 306/306 70.155.61.252 D N A 5060 OK (186 ms) |
| 05:14.03 | PaulAviles | for sip show peers |
| 05:15.37 | *** join/#asterisk CunningPike (n=arodgers@S01060014bf81366b.vc.shawcable.net) |
| 05:15.54 | jaytee | time to snooze, nite all |
| 05:16.08 | *** part/#asterisk thepacmanfan (n=andrewk@173-22-139-185.client.mchsi.com) |
| 05:16.56 | [TK]D-Fender | PaulAviles: SIP/2.0 404 Not Found <----------- |
| 05:17.13 | PaulAviles | saw that, but it is ip based |
| 05:17.36 | [TK]D-Fender | PaulAviles: Your phone is telling you to "get lost" |
| 05:17.44 | PaulAviles | so that is asterisk trying to contact the remote phone right? |
| 05:17.52 | PaulAviles | on what traffic? |
| 05:18.18 | PaulAviles | that phone can call me, I just can call him |
| 05:18.27 | PaulAviles | you know the 10000 - 20000 range? |
| 05:18.29 | [TK]D-Fender | PaulAviles: LOOK at it. You INVITE out to -- Executing [s@macro-dial:7] Dial("SIP/301-09fd1a80", "SIP/306|15|tr") in new stack |
| 05:18.46 | [TK]D-Fender | To: <sip:306@70.155.61.252:5060;user=phone;transport=udp> <-- IP you are sending to for him |
| 05:18.57 | PaulAviles | yes |
| 05:19.08 | PaulAviles | port 5060 is open and forwarded to the phone |
| 05:19.14 | [TK]D-Fender | <--- SIP read from 70.155.61.252:5060 ---> SIP/2.0 404 Not Found <---- Lookee, thats him! |
| 05:19.27 | [TK]D-Fender | PaulAviles: Your phone is configured wrong then. |
| 05:19.31 | *** join/#asterisk Qwell (i=north@pdpc/sponsor/digium/Qwell) |
| 05:19.31 | *** mode/#asterisk [+o Qwell] by ChanServ |
| 05:19.41 | [TK]D-Fender | PaulAviles: And you should not be forwarding anything to a remote phone. |
| 05:19.48 | PaulAviles | should both tcp and udp 5060 be enabled? |
| 05:19.54 | [TK]D-Fender | paulaNEITHER |
| 05:19.58 | PaulAviles | huh? |
| 05:20.14 | [TK]D-Fender | PaulAviles: Remote phones should NOT need any forwarding |
| 05:20.35 | PaulAviles | no, .... it is behind a netopia router |
| 05:20.41 | PaulAviles | by default everything is block |
| 05:20.50 | PaulAviles | the phone has a 192.168.1.5/24 ip |
| 05:20.55 | [TK]D-Fender | PaulAviles: Normal NAT keepalaives should be fine |
| 05:21.09 | [TK]D-Fender | PaulAviles: and Yes... I was rather sure it had a private IP. |
| 05:21.34 | PaulAviles | so you are saying that there is nothing that I should do on the netopia side? |
| 05:21.43 | [TK]D-Fender | PaulAviles: Normally no. |
| 05:21.46 | *** join/#asterisk keebler (i=9446c2d5@gateway/web/ajax/mibbit.com/x-add337d1178e7196) |
| 05:22.02 | [TK]D-Fender | PaulAviles: Do you have only 1 SIP device behind that router? |
| 05:22.49 | keebler | [TK]D-Fender: Are you a robot? Haha, You've been on ALL day, or at least everytime I've gotten on. |
| 05:23.09 | PaulAviles | no, other computers too |
| 05:23.36 | *** join/#asterisk mintos (n=mvaliyav@nat/redhat-in/x-3dd90288ff8625d0) |
| 05:23.52 | [TK]D-Fender | PaulAviles: Running SIP? |
| 05:24.06 | PaulAviles | yes |
| 05:24.12 | PaulAviles | cisco 7960's |
| 05:24.20 | [TK]D-Fender | PaulAviles: Already sounding like trouble. |
| 05:24.35 | PaulAviles | should not be... |
| 05:24.42 | [TK]D-Fender | PaulAviles: And what else behind that router has any contact with the outside world? |
| 05:24.53 | PaulAviles | other computers |
| 05:25.41 | John-B | hi there |
| 05:25.52 | [TK]D-Fender | PaulAviles: I'm talking SIP HERE |
| 05:26.34 | PaulAviles | ok |
| 05:27.29 | John-B | [TK]D-Fender : can you help me, i want to rec and afther the asterisk rec ... save xxx.wav and if i call at the same number she replace the .wav who can i make to save 1.xxx.wav 2.xxx.wav |
| 05:27.39 | John-B | [TK]D-Fender : what i most put there : exten = _NXXXXXXXXX,1,Monitor(wav|${EXTEN}_${Calltime}_${CALLERID(num)}_${CALLERID(name)},m) |
| 05:29.08 | John-B | anyone ? |
| 05:29.22 | John-B | :( |
| 05:29.25 | PaulAviles | TK later, thanks for your help need to get zome zzzzzz..... |
| 05:29.32 | [TK]D-Fender | John-B: O RLY? Amazing somebody just left asking about that EXACT piece of code! |
| 05:30.37 | John-B | [TK]D-Fender : yes my friend ... |
| 05:30.59 | John-B | [TK]D-Fender : can you help us ? |
| 05:31.29 | [TK]D-Fender | John-B: Go show me it happening twice |
| 05:31.41 | *** join/#asterisk voxter (n=voxter@S0106001c1025ca09.vc.shawcable.net) |
| 05:33.11 | dan__t | http://www.packtpub.com/asterisk-gateway-interface-programming/book |
| 05:33.19 | dan__t | I really, really want that book, but Amazon doesn't have it. |
| 05:33.56 | [TK]D-Fender | dan__t: Go beat on their door incessently with a wet noodle. I'm sure they'll listen to you then... |
| 05:34.06 | keebler | dan__t: Get the eBook |
| 05:34.11 | dan__t | Always using your head. |
| 05:34.11 | keebler | in PDF |
| 05:34.19 | dan__t | I don't think its available yet. |
| 05:34.27 | dan__t | Besides, I'm a sucker for a real paperback book :/ |
| 05:34.52 | keebler | Ah, "Expected Jan. 2009" |
| 05:34.53 | keebler | Sorry |
| 05:35.52 | dan__t | Yea. THought Amazon would at least have it as a pre-order. |
| 05:35.56 | rob0 | Other than "don't do it", does anyone have a favorite softphone for Linux? |
| 05:36.33 | dan__t | haha |
| 05:37.01 | dan__t | Uh, for testing, I was using something with a pretty flamboyant name.... forget what it is, though... |
| 05:37.06 | dan__t | Something like Flower Power or something |
| 05:37.14 | [TK]D-Fender | ~ekiga |
| 05:37.15 | jbot | [~ekiga] Ekiga is an OSS SIP soft-phone for Windows, MacOSX, and Linux (Requires GTK+ which is included with the binary releases) that can be found at http://www.ekiga.org |
| 05:37.16 | [TK]D-Fender | ~twinkle |
| 05:37.17 | jbot | [~twinkle] Twinkle is an OSS SIP phone for KDE/Qt systems and can be downloaded at http://www.twinklephone.com |
| 05:37.20 | dan__t | Twinkle. |
| 05:37.21 | dan__t | That's it. |
| 05:37.23 | [TK]D-Fender | ~gnophone |
| 05:37.29 | [TK]D-Fender | hmm |
| 05:37.32 | [TK]D-Fender | missed one |
| 05:37.44 | rob0 | yeah I have seen these, I am asking, are they worth the trouble? |
| 05:37.53 | rob0 | linphone? |
| 05:38.22 | dan__t | I cannot comment, sorry. |
| 05:38.52 | rob0 | My son actually WANTS a softphone over a real phone. Go figure. |
| 05:42.15 | drmessano | Stab him |
| 05:42.18 | drmessano | *shrug* |
| 05:42.21 | drmessano | I had nothing |
| 05:42.49 | rob0 | aww, I can't stab my own SON! |
| 05:43.15 | drmessano | Please? |
| 05:44.11 | [TK]D-Fender | rob0: Sure you can Job! |
| 05:44.15 | drmessano | He prefers a softphone.. You cant save him now! |
| 05:44.35 | drmessano | Look |
| 05:44.42 | drmessano | Softphones are the gateway |
| 05:44.47 | drmessano | Next it will be SKYPE |
| 05:44.58 | drmessano | and then, he'll want his own ATA...... |
| 05:45.02 | drmessano | for VONAGE |
| 05:45.21 | *** join/#asterisk the_5th_wheel (n=edd@webster.cybertek.co.za) |
| 05:45.31 | drmessano | rob0: If you dont talk to your kids about Skype, someone else will. |
| 05:46.42 | the_5th_wheel | quick question, the LBO in zapata.conf , What exactly is that, the distance from my device to the NT, to the exanche, or what? |
| 05:46.56 | *** join/#asterisk Gopaul (n=chatzill@59.97.120.233) |
| 05:47.37 | keebler | Vonage started me on the dark path of VOIP, so I can't completely knock it. |
| 05:47.54 | frogonwheels | rob0: I've found voiper on windoze works well. |
| 05:48.38 | drmessano | keebler: HEATHEN |
| 05:49.23 | *** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun) |
| 05:50.14 | [TK]D-Fender | the_5th_wheel: to the smartjack |
| 05:50.38 | keebler | drmessano: I dropped Vonage 2 years ago. |
| 05:51.22 | *** join/#asterisk dlewis_ (i=4579ba4a@about/security/staff/dlewis) |
| 05:51.25 | keebler | drmessano: Just saying, its opened me up to the idea since I already used *nix in my daily life. It wasn't until recently that I could actually make use of Asterisk. |
| 05:56.00 | *** join/#asterisk TrentCreek (n=kvirc@adsl-68-89-74-11.dsl.hrlntx.swbell.net) |
| 05:58.29 | rob0 | :) |
| 05:58.45 | rob0 | Frog, thanks, but no Windows here. |
| 06:01.02 | zafar_ | i am using zoiper on both windows and linux and is working fine |
| 06:01.27 | *** join/#asterisk inam (n=root@116.71.221.106) |
| 06:04.05 | zafar_ | how can i send a voicemail to an email, i have smtp server setup on a public ip |
| 06:05.35 | dlewis | hey |
| 06:05.40 | dlewis | does the Nortel NTDU92AA70 work with asterisk? |
| 06:05.53 | [TK]D-Fender | zafar_: You've already asked in #freepbx. You're in the WRONG CHANNEL for this |
| 06:06.15 | zafar_ | yes but no one there :( |
| 06:06.27 | [TK]D-Fender | zafar_: this is not 2nd level support |
| 06:06.51 | [TK]D-Fender | zafar_: and * uses Sendmail by defulat to send out VM's |
| 06:07.09 | [TK]D-Fender | dlewis: That runs UNISTIM which was previously supported... |
| 06:07.20 | dlewis | ok |
| 06:08.04 | inam | sir chaye ba sku |
| 06:08.08 | [TK]D-Fender | dlewis: Dunno if it still does in 1.6 |
| 06:08.43 | [TK]D-Fender | inam: hukt on fonix werkt 4 u |
| 06:09.05 | dlewis | [TK]D-Fender: no problem, I'll research it. thanks! |
| 06:09.15 | *** join/#asterisk freakazoid0223 (n=matt@pool-71-246-17-74.phlapa.fios.verizon.net) |
| 06:09.42 | [TK]D-Fender | dlewis: If you're considering getting them, DON'T. |
| 06:10.19 | dlewis | [TK]D-Fender: not considering it... Someone is selling there old server with a few of those phones and I was wondering if they worked with asterisk. |
| 06:10.36 | [TK]D-Fender | dlewis: maybe + shit |
| 06:10.45 | [TK]D-Fender | dlewis: Ill advised. |
| 06:10.48 | dlewis | ok |
| 06:10.56 | dlewis | yea, I was gonna buy it. |
| 06:11.30 | *** join/#asterisk ming_zym (n=ming_zym@125.39.45.11) |
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| 06:32.32 | [TK]D-Fender | checkout time. Later all |
| 06:32.41 | *** join/#asterisk botox93 (n=botox93@213.221.82.242) |
| 06:47.00 | *** join/#asterisk Chris-NB (n=chris@85-126-61-10.work.xdsl-line.inode.at) |
| 06:48.25 | the_5th_wheel | [TK]D-Fender: mmm. Any idea what else could cause line lockups? Once a day, I stop being able to recieve calls. Ring requested on channel 0/1 already in use on span 6. Hanging up owner |
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| 07:34.42 | bn43 | hi - I'm trying to get my snom320 phone to record - phone shows recording when I press the button but there is no indication on the console on this and I can't retrieve the recording |
| 07:40.20 | *** join/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 07:40.42 | *** join/#asterisk xrmx__ (n=rm@host1-187-dynamic.31-79-r.retail.telecomitalia.it) |
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| 07:49.33 | fiddur | bn43: snom320 has built-in recording, so it's not done via asterisk. You should find it on the phones own webpage... |
| 07:51.42 | bn43 | I have checked on webpages that say the phone gives a beep to start and end recording - I was hoping to eventually have the recording automatically emailed off to user of the phone via asterisk |
| 07:53.02 | bn43 | do you know how I pull up the recording on the phone? |
| 07:56.09 | fiddur | bn43: No, sorry... I don't have a 320, just seen it mentioned in the joint manual that's also for the 300 (that lacks recording function) |
| 08:03.44 | *** join/#asterisk kerx (n=kerx@adsl-69-104-66-38.dsl.irvnca.pacbell.net) |
| 08:07.17 | *** join/#asterisk andy-x_l (i=425c01d1@gateway/web/ajax/mibbit.com/x-576ac480a08edf27) |
| 08:16.33 | *** join/#asterisk Gary (n=Gary@freenode/staff/colchester-lug.gary) |
| 08:17.45 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 08:18.33 | *** join/#asterisk Micc (n=Michael@c-76-121-255-52.hsd1.wa.comcast.net) |
| 08:18.33 | *** join/#asterisk dssman (n=no@CPE001d7e602900-CM0011aec52a9c.cpe.net.cable.rogers.com) |
| 08:21.20 | *** join/#asterisk pif (n=ldm@zenon.apartia.fr) |
| 08:22.12 | *** join/#asterisk botox93 (n=botox93@213.221.82.242) |
| 08:23.17 | *** join/#asterisk dr0ck (n=dr0ck@nat/digium/x-2f1bec0c2a5a86e2) |
| 08:27.58 | *** join/#asterisk c0rnoTa (n=c0rnoTa@80.251.113.56) |
| 08:31.15 | dan__t | hrm |
| 08:31.22 | dan__t | anyone have a favorite guide to odbc + asterisk? |
| 08:31.43 | Corydon76-dig | The source? |
| 08:31.50 | *** join/#asterisk jql (n=jql@12.9a.344a.static.theplanet.com) |
| 08:31.54 | inam | yah but only for posgresSQL |
| 08:32.02 | angryuser | yes the source of the matrix |
| 08:32.04 | *** join/#asterisk krdian (i=krdian@killer.radom.net) |
| 08:32.28 | krdian | hi |
| 08:33.00 | dan__t | heh |
| 08:33.11 | dan__t | I'm sure I can translate, inam. |
| 08:33.30 | dan__t | So long as 10,000 Maniacs never streams on Internet Radio again.... |
| 08:33.36 | dan__t | Talk about ruining your hackfest. |
| 08:35.15 | dan__t | Wait, Corydon76-dig, you wrote func_odbc iirc? |
| 08:40.22 | *** join/#asterisk joobie (n=joobie@joobie.org) |
| 08:43.22 | *** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 08:47.08 | *** join/#asterisk farkus (i=chatzill@cpe-98-14-94-76.nyc.res.rr.com) |
| 08:54.51 | *** join/#asterisk coppice (n=chatzill@201.193.17.210.dyn.pacific.net.hk) |
| 08:58.12 | dan__t | Well that sucked. |
| 08:59.14 | dan__t | Alright, well, 'odbc show' shows no DSNs. I have one defined, and can access it using isql, but Asterisk doesn't see it. |
| 09:04.41 | *** join/#asterisk siera08 (n=sosoriri@218.207.141.90) |
| 09:05.41 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 09:06.01 | *** join/#asterisk dan__t (n=dant@72.233.89.95) |
| 09:08.09 | dan__t | yea... |
| 09:09.45 | *** join/#asterisk Avelino (n=Avelino@mail.paterno.com.br) |
| 09:15.30 | *** join/#asterisk darkskiez (n=dz@195-11-205-216.suip.mezzonet.net) |
| 09:16.38 | *** part/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 09:18.45 | *** join/#asterisk KermitTheFragger (n=KermitTh@118-197.bbned.dsl.internl.net) |
| 09:28.23 | inam | to work odbc show. u must need to configure odbc file |
| 09:28.41 | inam | for dns and database connectivity |
| 09:37.17 | siera08 | anybody tell me whether asterisk can play MOH loudly or not? |
| 09:37.53 | Maliuta | depends on the files and the settings |
| 09:38.06 | Maliuta | you can always make the files really loud |
| 09:38.13 | siera08 | Maliuta: asterisk has settings about it? |
| 09:38.48 | siera08 | i already have recorded files. i can't rerecord them again. |
| 09:38.57 | Maliuta | remember that you're only going to get voice related frequencies out of it, so you won't get all the sound of any music |
| 09:38.59 | siera08 | i want to play them roudly now. |
| 09:39.07 | Maliuta | and you can always boost the gain in a file |
| 09:40.19 | *** join/#asterisk amessina (n=amessina@2001:470:1f11:68:20e:cff:fe01:d5ec) |
| 09:40.37 | Maliuta | try setting "mode=mp3" |
| 09:41.33 | siera08 | i tried 'mp3' mode. but it;s same. |
| 09:43.14 | siera08 | it's problem for only sound file? |
| 09:44.29 | Maliuta | well it sounds like you didn't record the files properly |
| 09:44.36 | Maliuta | you need to boost the gain in them |
| 09:44.46 | Maliuta | put them into something like audacity |
| 09:45.00 | Maliuta | and thats all I have to say |
| 09:45.41 | siera08 | yes, i have failed to recorded files. |
| 09:46.05 | siera08 | thanku. |
| 09:51.29 | mort_gib | Maliuta: I have the same problem with mp3 |
| 09:52.16 | Maliuta | edit the gain level in the files so that it almost peaks |
| 09:52.38 | Maliuta | you don't want it to peak because it will just get clipped |
| 09:55.50 | siera08 | Maliuta: u know some tool for amplifing the sound volumn? |
| 09:57.56 | Maliuta | siera08: do you know what "audacity" is? |
| 09:59.03 | siera08 | 'audacity = [U] brave but rude or shocking behaviour' in dictionary.. |
| 09:59.10 | siera08 | me is very poor english.. |
| 09:59.31 | Maliuta | try again |
| 09:59.39 | Maliuta | apt-cache search audacity |
| 10:00.21 | *** join/#asterisk [Jasper] (n=jverberk@195-240-174-59.ip.telfort.nl) |
| 10:00.33 | [Jasper] | hej guys, I keep ending up in the [default] context with my trunks |
| 10:00.44 | [Jasper] | for some reason it never goes to its own context... |
| 10:01.07 | Maliuta | [Jasper]: do you have configs in pastebin? |
| 10:01.47 | Maliuta | otherwise you might aswell be saying "the sky is blue, I want it purple, but it's blue" |
| 10:02.43 | siera08 | Maliuta: ah, i see "Audacity: Free Audio Editor and Recorder" |
| 10:02.47 | siera08 | thank u again. |
| 10:04.20 | krdian | :q! |
| 10:04.55 | *** join/#asterisk joelsolanki (i=joelsola@202.160.161.94) |
| 10:05.00 | joelsolanki | Hi all |
| 10:05.13 | Maliuta | obviously [Jasper] just wanted to make random statements |
| 10:05.17 | Maliuta | heads to bed |
| 10:05.42 | joelsolanki | i have installed asterisk from tar.tgz version. right now it is creating cdr at Master.csv |
| 10:05.57 | joelsolanki | but i want to rotate this logs. want to have every day csv files |
| 10:06.01 | joelsolanki | how can i do that ? |
| 10:06.12 | joelsolanki | right now Master.csv has all days cdrs. |
| 10:07.13 | [Jasper] | not really a random statement Maliuta |
| 10:08.18 | [Jasper] | Maliuta http://www.stringed-up.nl/call.txt |
| 10:09.23 | *** join/#asterisk itguru (n=p@host81-134-10-140.in-addr.btopenworld.com) |
| 10:09.53 | *** join/#asterisk joobie (n=joobie@joobie.org) |
| 10:09.58 | itguru | is pulling his hair out trying to get a polycom sound station 6000 to work again :( |
| 10:10.31 | *** join/#asterisk magronez (n=eusei@unaffiliated/magrao/x-2903) |
| 10:10.31 | joobie | sup with it itguru |
| 10:11.04 | coppice | the 6000 is nice, once you've got the latest code on it |
| 10:11.27 | joobie | 6000 sounds like the uber leet model - i've only played with 330 and 600's |
| 10:11.49 | coppice | the 6000 is a conferencing box |
| 10:12.34 | zafar_ | -- <SIP/411-09f64388> Playing 'auth-incorrect' (language 'en') what auth is it talking about |
| 10:12.45 | inam | congratulation zafar ur voicemail start working properly |
| 10:12.52 | itguru | I had it working, tested it for five minutes, and I don't know WHAT my client did to it, but I've been on it for 3 days now, can't figure what changed :( **ANGRY** |
| 10:13.12 | inam | my local voicemail is right now |
| 10:13.16 | inam | working good |
| 10:13.35 | itguru | coppice, joobie - It is a REALLY nice confrence phone - would not look out of place in CTU! *if it still existed* |
| 10:14.02 | *** join/#asterisk Daejeo (n=chatzill@114.201.159.219) |
| 10:16.29 | [Jasper] | Maliuta ? |
| 10:18.17 | Daejeo | can anyone recommend a machine for 50 extens? |
| 10:25.07 | *** join/#asterisk itguru (n=p@host81-134-10-140.in-addr.btopenworld.com) |
| 10:28.39 | joelsolanki | how can i rotate master.csv every day ? |
| 10:28.42 | joelsolanki | any links plz |
| 10:29.50 | beherit | you can use logrotate |
| 10:29.53 | krdian | jpeeler: u can use rasterisk -x "logger rotate" |
| 10:30.12 | *** join/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 10:30.16 | krdian | *joelsolanki |
| 10:30.25 | krdian | sorry jpeeler |
| 10:31.00 | krdian | joelsolanki: or logrotate as beherit sayd |
| 10:31.28 | [Jasper] | can anyone here help me |
| 10:31.42 | [Jasper] | my incoming calls all go to [default] instead of the defined context |
| 10:31.48 | krdian | joelsolanki: logger rotate will rotete all logs |
| 10:31.51 | [Jasper] | http://www.stringed-up.nl/call.txt <- that's a debug |
| 10:32.31 | joelsolanki | hmm got it. |
| 10:32.34 | joelsolanki | i will test out. thanks :) |
| 10:33.08 | *** join/#asterisk DrAk0 (n=luisjose@nelug/coreteam/luisjose) |
| 10:33.26 | krdian | joelsolanki: sorry, mistake... rotate logs ono *.csv.. |
| 10:33.39 | krdian | joelsolanki: so use logrotate :) |
| 10:34.27 | joelsolanki | yedsok |
| 10:36.39 | krdian | [Jasper]: No matching peer for '0613442399' from '81.23.228.129:5060' |
| 10:36.45 | kaldemar | [Jasper]: still can't get calls to match a peer? |
| 10:37.28 | krdian | [Jasper]: did u configure peer/fiend for 81.23.228. |
| 10:37.30 | kaldemar | [Jasper]: are all your incoming calls coming from the same place? as in from the same ip address? |
| 10:44.46 | *** join/#asterisk chi6IT41 (n=chigital@tmo-102-177.customers.d1-online.com) |
| 10:45.36 | [Jasper] | kaldemar no |
| 10:45.47 | [Jasper] | incoming calls you mean from my sip provider? |
| 10:45.51 | [Jasper] | or the actual callers |
| 10:46.00 | *** part/#asterisk gambler1 (n=mene_sun@dragan.eunet.yu) |
| 10:46.04 | [Jasper] | I think it comes from 1 ip I would guess |
| 10:46.23 | kaldemar | that's for you to tell, it's your setup. are you accepting sip calls from more than one place? |
| 10:46.44 | [Jasper] | 2 phone numbers, both from the same provider kaldemar |
| 10:46.54 | [Jasper] | so that's 2 trunks |
| 10:47.05 | kaldemar | then do a peer that matches to the ip the calls are coming from. host=81.23.228.129 will make them match to that ip. |
| 10:48.25 | [Jasper] | hmm |
| 10:48.33 | [Jasper] | at the moment host=budgetphone.nl |
| 10:48.36 | [Jasper] | so I should change that? |
| 10:49.40 | kaldemar | or match by the number that is in the From-header of the SIP messages by type=user, if you want to separate the "trunks". |
| 10:50.22 | kaldemar | yes you whould change it because the calls are clearly not coming from there. |
| 10:50.30 | *** join/#asterisk sergee (n=serg@voip1.west-call.com) |
| 10:53.21 | [Jasper] | hwen I change the host |
| 10:53.23 | [Jasper] | it stops working |
| 10:53.32 | [Jasper] | I can't auth anymore, so that has to be budgetphone.nl I guess |
| 10:54.14 | [Jasper] | which number is in the from-header then kaldemar ? |
| 10:55.36 | kaldemar | by "it" do you mean outgoing calls? if so, leave that as it is and make another peer definition with the ip to match incoming. |
| 10:56.28 | [Jasper] | hmm that is an idea |
| 10:56.36 | [Jasper] | like a outgoing trunk |
| 10:56.38 | [Jasper] | and incoming trunk |
| 10:57.09 | *** join/#asterisk n_akram (n=nakram1@116.71.166.157) |
| 10:57.15 | n_akram | hi room... |
| 10:57.29 | n_akram | anybody installed asterisk 1.6.05? |
| 10:58.05 | [Jasper] | hmm kaldemar apparently my sip provider hsa a host called sip.budgetphone.nl which points to proxy.sipthor.net which has 3 ip adddresses |
| 10:58.08 | [Jasper] | messed up isn't it |
| 10:59.47 | kaldemar | you can try with a single host=sip.budgetphone.nl, but if it doesn't work, try with explicit ip's. |
| 10:59.51 | kaldemar | i'm out > |
| 11:13.46 | *** part/#asterisk n_akram (n=nakram1@116.71.166.157) |
| 11:23.31 | *** join/#asterisk guax (n=guaxinim@unaffiliated/guaxinim) |
| 11:27.44 | [Jasper] | hmm |
| 11:27.49 | [Jasper] | kaldemar you gone? |
| 11:31.17 | [Jasper] | anyone here who can help me? |
| 11:33.07 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 11:37.49 | *** join/#asterisk propellerhead (n=yogurt2u@host15.190-30-186.telecom.net.ar) |
| 11:46.53 | *** join/#asterisk Ast001 (n=uros@cable-94-189-182-105.dynamic.sbb.rs) |
| 11:47.15 | Ast001 | hi can you help me I need to set time on my asterisk server so calldate will be correct |
| 11:47.44 | Ast001 | on linux I already set time but asterisk write time wrong I guess I need to change something in some file to correct that |
| 11:51.29 | *** join/#asterisk quaqo (n=quaqo@83-103-40-166.ip.fastwebnet.it) |
| 11:58.55 | *** join/#asterisk kisu_ (n=kisu@2001:5c0:1100:9900:1d37:23b5:94db:ec35) |
| 11:59.25 | Ast001 | is solution setting timestamp in asterisk.conf ? |
| 12:04.52 | *** join/#asterisk c0rnoTa (n=c0rnoTa@80.251.113.56) |
| 12:14.17 | *** part/#asterisk Ast001 (n=uros@cable-94-189-182-105.dynamic.sbb.rs) |
| 12:16.29 | *** join/#asterisk Coolthreads (n=Coolthre@203-97-238-71.cable.telstraclear.net) |
| 12:20.40 | *** join/#asterisk chi6IT41 (n=chigital@tmo-096-226.customers.d1-online.com) |
| 12:34.00 | *** join/#asterisk path_ (n=path@191-114-21-190.adsl.terra.cl) |
| 12:37.29 | *** join/#asterisk Dr-Linux|home (n=Nothing@119.63.130.34) |
| 12:37.50 | Dr-Linux|home | anybody using 7960 with Asterisk SIP? |
| 12:45.41 | *** join/#asterisk bn43 (n=dhashen@41.28.253.74) |
| 12:46.47 | [Jasper] | damnit |
| 12:47.01 | [Jasper] | asterisk is d to understand |
| 12:48.49 | bn43 | Hi I'm trying to get the record button to work my snom 320 phone - my reading says that the recording is done on the server not on the phone and I can't get it to work - can someone help pls? |
| 12:53.46 | bn43 | apparently there is supposed to be a signal sent by the phone about recording - I don't see that coming up on the console |
| 12:56.16 | *** join/#asterisk [gquit]bombadil (n=dana@CPE-70-94-44-157.wi.res.rr.com) |
| 12:57.10 | bn43 | anyone? |
| 12:59.30 | *** join/#asterisk DarkRift (n=dark@65.92.249.137) |
| 13:00.21 | *** join/#asterisk anonymouz666 (n=anonymou@201.19.167.7) |
| 13:01.22 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 13:02.29 | beek | bn43: That kind of functionality occurs because the phone sends some keystrokes to the server. What you need to do is either find the docs for that phone and see what it sends, then write a dialplan around it, or learn how to the program the phone so that it sends whatever you have Asterisk configured for in your dialplan to do recording. |
| 13:03.51 | bn43 | this is the doc I have found - http://asterisk.snom.com/index.php/Asterisk_1.4/Call_Recording - I think I maybe doing sumthing wrong tho |
| 13:04.34 | bn43 | I've followed John Macnamara's tutorial - he has a template that each phone follows - first line |
| 13:04.56 | bn43 | exten => s,1,Dial(SIP/${MACRO_EXTEN},25) |
| 13:05.28 | bn43 | I've done exten => s,1,Dial(SIP/${MACRO_EXTEN},25,wW) |
| 13:05.45 | bn43 | what am I missing? |
| 13:06.55 | beek | Assuming that you put in what he has, AND you reloaded Asterisk to get the configuration, then I don't know without looking at all of your configs. |
| 13:07.37 | bn43 | yes I have shutdown gracefully and restarted asterisk after changing configs |
| 13:07.41 | beek | bn43: Test it manually with *1 instead of tha tdedicated button and then see if it works. |
| 13:08.19 | bn43 | manual does not work |
| 13:08.22 | *** join/#asterisk eliteschaf (i=c2d02b52@gateway/web/ajax/mibbit.com/x-d6729c94f27e7e52) |
| 13:08.29 | bn43 | *1 that is |
| 13:08.30 | eliteschaf | hey |
| 13:09.02 | bn43 | but someone else on the forum said that the phone records locally and not on the server? |
| 13:11.08 | beek | bn43: I use Polycom phones, not Snom, so I don't know what it does... sorry |
| 13:11.25 | *** join/#asterisk dlewis (i=c7340d67@about/security/staff/dlewis) |
| 13:11.29 | bn43 | I've heard that before! :-) |
| 13:11.35 | bn43 | thanks anyway |
| 13:11.43 | *** join/#asterisk Khratos (n=khratos@190.166.103.146) |
| 13:12.31 | beek | bn43: If I were to work on this problem I'd get it set up so that Asterisk does the recording with *1 (or whatever is in features.conf). THEN I would work on getting the phone to do it with that button. |
| 13:12.54 | *** join/#asterisk SuPrSluG (n=SuPrSluG@firewall-a.buf.ny.i-evolve.net) |
| 13:12.56 | Khratos | Good $TIMEOFDAY :P |
| 13:13.11 | bn43 | ok I'm a noob - how do I get asterisk to do this? |
| 13:13.58 | *** join/#asterisk guax (n=guaxinim@unaffiliated/guaxinim) |
| 13:14.00 | beek | bn43: http://www.voip-info.org/wiki-Asterisk+record+calls |
| 13:23.02 | eliteschaf | guys i've got a problem with the asterisk-gui from digium. if i try to connect to asterisk i get the following error: "Connect attempt from '192.168.0.119' unable to authenticate", here is my http.conf: http://pastebin.com/f3c250c7a |
| 13:23.57 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
| 13:24.24 | bn43 | eliteschaf: I got politely asked to go to the asterisk-gui channel the last time asked something on it..... |
| 13:24.40 | bn43 | :-) |
| 13:24.45 | eliteschaf | kók |
| 13:24.48 | eliteschaf | ok |
| 13:31.28 | *** join/#asterisk n_ghost (n=nakram1@116.71.166.157) |
| 13:31.48 | n_ghost | AMI is documented anywhere??? |
| 13:32.48 | [TK]D-Fender | n_ghost: WIKI & BOOK |
| 13:34.15 | n_ghost | [TK]D-Fender: book is out dated i think... i'll check WKKI... |
| 13:34.41 | *** join/#asterisk ThoMe (i=tm@81.92.168.130) |
| 13:34.44 | ThoMe | Ahoi. |
| 13:34.51 | [TK]D-Fender | Chips. |
| 13:34.56 | ThoMe | Fish? |
| 13:35.27 | ThoMe | Spricht hier wer ein wenig Deutsch? |
| 13:35.42 | eliteschaf | ja |
| 13:35.44 | ThoMe | (geht nur um was übersetzen damit das hier alle verstehen) |
| 13:35.45 | [TK]D-Fender | n_ghost: in your doc/ folder as well |
| 13:36.01 | eliteschaf | und was? |
| 13:36.02 | *** join/#asterisk pawpro (n=IRC@213.166.12.34) |
| 13:36.09 | ThoMe | kann man mit asterisk ein internen ton simmulieren? ihr das sicherlich von firmen, wo man die 0 drückt um ins amt zu gelangen. davor hört man immer "peep peep peep - pause - peeep peep peep" kunde will das gerne. nur wüÃte ich nicht, wie ich das machen soll. :-) |
| 13:36.17 | ThoMe | eliteschaf: das. :/ |
| 13:36.30 | n_ghost | [TK]D-Fender: ok boss... thankyou... |
| 13:37.47 | pawpro | Hi everybody. Could someone tell me how do lets say reject call from AGI with certain SIP Code? Thanks |
| 13:38.10 | pawpro | I'm using * 1.6 and php agi 2.14 |
| 13:38.22 | eliteschaf | thome also der asterisk soll einen ton verschicken? |
| 13:38.30 | eliteschaf | oder kann das alles der asterisk machen? |
| 13:38.45 | [TK]D-Fender | pawpro: you have little control. take a look at the hangupcause options : "core show application hangup" |
| 13:38.51 | ThoMe | eliteschaf: man nimmt den hörer ab und hört ein intern-zeichen |
| 13:38.55 | pawpro | thanks mate |
| 13:39.37 | eliteschaf | aso du meinst du ein gepeepe? |
| 13:40.08 | ThoMe | eliteschaf: ja |
| 13:40.30 | eliteschaf | was will der den damit?! ^^ aber gut ich versuchs mal zu uebersetzen |
| 13:40.40 | ThoMe | eliteschaf: ok :-) |
| 13:40.50 | ThoMe | eliteschaf: er isses gewohnt von der alten anlage |
| 13:40.55 | eliteschaf | hehe |
| 13:41.04 | eliteschaf | musst ihn dazu ermuntern, neues auszuprobieren ;) |
| 13:43.07 | ThoMe | eliteschaf: übersetzt du? |
| 13:43.09 | ThoMe | (bitte) |
| 13:43.28 | eliteschaf | ja muss mir überlegen, wie man das am besten formuliert |
| 13:43.54 | *** join/#asterisk drepan (n=pandre@apcdns2.autopage.co.za) |
| 13:45.18 | drepan | Calls to our remote branch always "die" at 20 secs, error is Maximum retries exceeded. What could be the problem? |
| 13:46.09 | eliteschaf | is there any way to simulate a sound, like you could hear with the old phones, before you dial 0 to talk to the switchboard |
| 13:46.19 | [TK]D-Fender | drepan: Show us the failed call with full core & SIP debug |
| 13:46.37 | drepan | How do I do that, shoudl I google for it |
| 13:46.51 | [TK]D-Fender | eliteschaf: what kind of phone are you using? |
| 13:47.02 | eliteschaf | THome was für ein phone? |
| 13:47.10 | ThoMe | snom 370 |
| 13:47.10 | [TK]D-Fender | drepan: "sip set debug" "set debug 10" |
| 13:47.28 | eliteschaf | [TK]D-Fender: snom 370 |
| 13:47.31 | eliteschaf | ^^ |
| 13:47.42 | [TK]D-Fender | eliteschaf: its your phone's job to give you dialtone. |
| 13:47.51 | *** join/#asterisk kisu (n=kisu@2001:5c0:1100:9900:f1ff:3bdc:54b0:f501) |
| 13:47.55 | esaym | how big of a port range do I need in rtp.conf? I am trying to keep the range as small as possible so that I don't have to forward as many ports |
| 13:48.28 | esaym | I only have a couple of phones |
| 13:48.47 | dlewis | Make sure that /etc/asterisk/rtp.conf shows rtpstart=10000 and rtpend=20000 |
| 13:48.49 | dlewis | ?? rtp |
| 13:48.55 | dlewis | ~ rtp |
| 13:48.56 | jbot | hmm... rtp is The Internet-standard protocol for the transport of real-time data, including audio and video. RTP is used in virtually all voice-over-IP architectures, for videoconferencing, media-on-demand, and other applications. A thin protocol, it supports content identification, timing reconstruction, and detection of lost packets. |
| 13:49.12 | eliteschaf | ThoMe: das sollte dein Phone machen wurde mir gesagt |
| 13:50.01 | RypPn | Has anyone ever come across this with asterisk 1.6.0.5? Outbound dahdi calls thro a sangoma analog card get congestion, circuits busy, until one inbound call has been received. After this point outbound works fine. I have swapped out for asterisk 1.4.23.1 and there is no problem with outbound. I have traces of a working and non-working outbound call if it helps :) |
| 13:50.10 | *** join/#asterisk jpmcallister (n=ec06113@kapla.escelsa.com.br) |
| 13:50.22 | [TK]D-Fender | esaym: indeed RTP.conf Remember 2 ports per simultaneous call (ALL calls, not just "external") |
| 13:50.35 | drepan | [TK]D-Fender: what should I look for? |
| 13:50.49 | [TK]D-Fender | drepan: PASTEBIN |
| 13:50.50 | [TK]D-Fender | ~pb |
| 13:50.51 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste, or , http://bin.cakephp.org/ |
| 13:51.16 | esaym | [TK]D-Fender: calls only use 2 ports? |
| 13:52.16 | [TK]D-Fender | esaym: Each, yes |
| 13:52.25 | [TK]D-Fender | esaym: 1 in, 1 out |
| 13:52.36 | esaym | impressive |
| 13:53.06 | beek | Morning [TK]D-Fender |
| 13:53.11 | [TK]D-Fender | beek: Morning... |
| 13:53.16 | jpmcallister | Hi. I'm using the Busy() command in the dialplan but it is not waiting for the user to hangup as it is supposed to do. Instead, the user is disconnected immediately. Could that be a bug? |
| 13:53.20 | *** join/#asterisk rootforce (n=chatzill@MT2-52.aircanopy.net) |
| 13:53.27 | kerx | anyone know what the causes could be for delay in audio when the call is picked up? |
| 13:53.37 | drepan | [TK]D-Fender: http://pastebin.com/d4499550e |
| 13:54.46 | *** join/#asterisk telnettech (n=telnette@d192-24-95-65.col.wideopenwest.com) |
| 13:55.09 | kerx | also, if someone could please recommend methods for improving call quality for SIP and IAX2 on Asterisk, I would be highly appreciated! |
| 13:55.24 | [TK]D-Fender | drepan: I do not see a complete call in there |
| 13:55.46 | *** join/#asterisk jaytee (n=jforde@unaffiliated/jaytee) |
| 13:56.12 | drepan | [TK]D-Fender: I only started the sip debugging during the call |
| 13:56.21 | drepan | [TK]D-Fender: should I make another? |
| 13:56.37 | beek | Morning jaytee |
| 13:56.54 | jaytee | morning beek |
| 13:57.18 | jaytee | friggin 10" of snow and it's still coming down, roads are a mess. what a fun day |
| 13:57.40 | beek | We were promised 6", got 2" -- nothing to write home about. |
| 13:58.02 | beek | jaytee: I assume that there's a staff 24x7 at the zoo? |
| 13:58.13 | jaytee | yeah |
| 13:59.11 | jpmcallister | anyone know why the call is bein disconnect when I use the Busy() command. It is supposed to wait for the user to hangup |
| 13:59.36 | [TK]D-Fender | drepan: Why are you indicating NAT for a secondary private IP subnet? |
| 14:00.03 | [TK]D-Fender | beek: We're getting it here instead... |
| 14:00.26 | [TK]D-Fender | jpmcallister: Not if you haven't answered the call yet |
| 14:00.40 | drepan | [TK]D-Fender: I have nat=yes on the [general] sip.conf |
| 14:00.56 | [TK]D-Fender | drepan: I see but your call is to a private subnet |
| 14:01.21 | drepan | [TK]D-Fender: for that specific user should I set nat=no |
| 14:01.25 | *** join/#asterisk jmacz (n=jmacz@190.144.75.22) |
| 14:02.26 | jpmcallister | [TK]D-Fend: But the other end is busy. In the dialplan when the the channel been called is busy I indicate it with playtones(busy) and use the command busy() which, as I understand, should wait for the calling user to hangup |
| 14:02.26 | beek | [TK]D-Fender: You're welcome to it. |
| 14:02.45 | *** part/#asterisk eliteschaf (i=c2d02b52@gateway/web/ajax/mibbit.com/x-d6729c94f27e7e52) |
| 14:03.16 | drepan | [TK]D-Fender: made no difference |
| 14:03.37 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 14:03.41 | *** join/#asterisk zeeesh (n=zeeesh@203.215.179.43) |
| 14:04.49 | [TK]D-Fender | drepan: they are PRIVATE. Why are you specifying nat for them? |
| 14:05.07 | *** join/#asterisk [intra]lanman (n=Raymond@freeswitch/developer/intralanman) |
| 14:05.10 | [TK]D-Fender | drepan: perhaps you could give some details abou your setup here... |
| 14:05.23 | drepan | [TK]D-Fender: I have changed it and the call still gets dropped, I have set nat=no |
| 14:06.16 | drepan | I have a Asterisk srv at our branch, connections to the server are via sip softphone. |
| 14:06.34 | drepan | Leased line between the branches, one caller at the branch |
| 14:07.27 | [TK]D-Fender | drepan: looks like faulty networking... they should not be NAT'd |
| 14:07.54 | *** join/#asterisk unasi7 (n=unasi7@213.144.157.100) |
| 14:09.59 | [TK]D-Fender | jpmcallister: No.. if you are being called by a device that can receive OOB call progress then they will get that and not an audio "busy" |
| 14:10.32 | magronez | is away: working |
| 14:11.37 | jpmcallister | [TK]D-Fender: tank you. That clears it to me |
| 14:11.41 | *** join/#asterisk John-B (n=xpl@84.126.205.1.dyn.user.ono.com) |
| 14:13.52 | *** join/#asterisk ocnarf (n=chatzill@125.252.90.5) |
| 14:14.04 | ocnarf | need some quick help |
| 14:14.17 | ocnarf | i changed my limits.conf |
| 14:14.31 | ocnarf | is there anyway it will take effect without rebooting my server |
| 14:14.32 | ocnarf | ? |
| 14:14.43 | jpmcallister | [TK]D-Fender: Now that you mentioned, I realized that I was only getting that behavior when I made the call from a sip softphone. Tank you |
| 14:15.14 | kerx | Anyone know of a good Asterisk VoIP Quality enhancements document if one is available? |
| 14:17.07 | *** join/#asterisk ruben23 (n=IT-ADMIN@124.107.3.178) |
| 14:17.33 | seanbright | jeebus |
| 14:17.47 | seanbright | the hills are alive with the sound of trolling on the -users list |
| 14:17.57 | [TK]D-Fender | ocnarf: Think you're in the wrong channel... |
| 14:18.16 | jaytee | seanbright, thanks for that Julie Andrews moment :-) |
| 14:18.28 | seanbright | my pleasure |
| 14:18.52 | ocnarf | D-Fender: I am getting some "Cannot create socket" error on my asterisk server |
| 14:19.48 | ocnarf | D-Fender: so adjusted my limits.conf. Any idea how can i reload it without rebooting my server? |
| 14:20.29 | kerx | ocnarf, Not possible |
| 14:23.16 | kerx | You know what's interesting, re-wiring your Linksys G router into a ignition for plastic-explosives |
| 14:23.27 | kerx | Now that is troll-like isn't it? |
| 14:23.49 | [TK]D-Fender | kerx: what a sad wastes of newtons.... |
| 14:23.55 | jaytee | is limits.conf only in 1.6? I don't have one on my 1.4 servers. |
| 14:24.09 | kerx | jaytee, I believe he means /etc/security/limits.conf |
| 14:24.11 | [TK]D-Fender | kerx: If you're going out with a bang, take a full-blown (to bits) Cisco out with you! |
| 14:24.15 | jaytee | ah |
| 14:26.49 | kerx | D-Fender, what if I flash the Linksys w/ DD-WRT, take it out of that ugly blue box, and wrap it in some more expensive metal, spray paint it and slap a San Francisco sticket on the front, then cut out the San Fran. Would that be sufficient? |
| 14:27.16 | kerx | Wow, I realized that I should probably end this conversation, before someone takes me seriously... |
| 14:27.29 | ocnarf | thanks kerx and D-Fender |
| 14:27.34 | kerx | We wouldn't want that :-( I'll stop now, and go back to asking or trying to answer #asterisk questions |
| 14:27.41 | ocnarf | i guess i have to reboot it then..:( |
| 14:27.55 | kerx | np ocnarf, yea, you must do it if you are speaking about /etc/security/limits.conf |
| 14:28.26 | kerx | ocnarf, might I ask what settings you've set limits.conf too and if you were actually hitting limits.conf hard limits, what they were? |
| 14:28.35 | kerx | s/too/to |
| 14:29.03 | *** join/#asterisk merlin8282 (n=merlin82@88-122-137-192.rev.libertysurf.net) |
| 14:29.05 | *** join/#asterisk adam000 (n=adam@c-76-97-76-93.hsd1.ga.comcast.net) |
| 14:29.08 | merlin8282 | Hi |
| 14:29.54 | kerx | hi there |
| 14:30.37 | merlin8282 | I have a problem with my FreePBX 2.5.1.1/Asterisk 1.4.22. |
| 14:30.41 | ocnarf | kerx: 1024 |
| 14:30.50 | merlin8282 | Sip phones work correctly, but the analog ones don't. |
| 14:31.10 | kerx | ocnarf: ahh, yea thats a bad one, and must be increased |
| 14:31.12 | [TK]D-Fender | merlin8282: And what's wrong with them? |
| 14:31.18 | merlin8282 | I take the phone, tone is here, but when i press one key it goes away |
| 14:31.23 | ocnarf | Kerx: now i added root soft nofile 65535 and root hard nofile 65535 |
| 14:31.24 | [TK]D-Fender | merlin8282: And what are they attached to? |
| 14:31.45 | [TK]D-Fender | merlin8282: pastebin what you get at CLI at verbose 10. |
| 14:31.47 | [TK]D-Fender | ~pb |
| 14:31.47 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste, or , http://bin.cakephp.org/ |
| 14:31.49 | [TK]D-Fender | ^^^^^^^^^^^ |
| 14:31.50 | merlin8282 | They're attached to a TDM400P, 4x FXS |
| 14:32.01 | vader-- | does the cisco ata 186 take the same config files and use the same tftp loading process as the 7940g phones? |
| 14:33.54 | merlin8282 | http://pastebin.ca/1320632 , almost nothing. |
| 14:35.15 | [TK]D-Fender | merlin8282: Did you connect the molex to the card? |
| 14:35.25 | merlin8282 | sure |
| 14:35.40 | [TK]D-Fender | merlin8282: "core debug 10" try again |
| 14:35.43 | merlin8282 | they did work, when I tried my first extensions.conf, about 10 lines ^^ |
| 14:36.13 | merlin8282 | core set debug 10, you mean ;) |
| 14:36.50 | [TK]D-Fender | merlin8282: ok/fine/sure |
| 14:36.51 | merlin8282 | Hmm, exactly same output |
| 14:37.18 | [TK]D-Fender | merlin8282: I was hoping to see more, but I'm willing to believe that we're in dialplan-error territory |
| 14:37.59 | merlin8282 | mmm. Can it be that I have to configure some context in FreePBX, for the Zap devices ? |
| 14:38.13 | *** join/#asterisk rwaite (n=fieldyca@rrcs-74-218-125-86.central.biz.rr.com) |
| 14:38.36 | merlin8282 | for the moment, it is set to the default "from-internal". |
| 14:40.02 | *** join/#asterisk hesco (n=hesco@c-76-17-99-50.hsd1.ga.comcast.net) |
| 14:41.55 | [TK]D-Fender | merlin8282: make another context with 1 simple exten in it and test |
| 14:43.19 | merlin8282 | Aaah ! I think i've found my error >< |
| 14:43.37 | bn43 | hi is there a way to get the console to detect and display what keys are being pressed on the a phone during a call? |
| 14:43.39 | merlin8282 | I forgot to modify the context in zapata.conf, after installing FreePBX ! |
| 14:44.27 | *** join/#asterisk kamanashisroy (n=kamanash@202.56.7.194) |
| 14:44.31 | [TK]D-Fender | bn43: "core set debug 10" |
| 14:44.53 | *** join/#asterisk The_Boy_Wonder (n=davidvos@nat/digium/x-fe5932ef2bc9c3a8) |
| 14:45.02 | *** join/#asterisk lowtek (n=no@adsl-074-185-089-046.sip.bhm.bellsouth.net) |
| 14:46.04 | bn43 | that did not work - I think that maybe a phone problem - I tried pressing *1 while on a call but nothing showed on the console |
| 14:47.22 | [TK]D-Fender | bn43: Go prove DTMF is working elsewhere |
| 14:48.56 | bn43 | If I access the mailbox on the phone and put in the password does that not prove DTMF? |
| 14:49.13 | merlin8282 | well, didn't work. I try it with a new extentions.conf |
| 14:50.09 | [TK]D-Fender | bn43: Yes |
| 14:50.40 | bn43 | I though so - so why is the console not telling me I am pressing *1 while on a call? |
| 14:50.44 | [TK]D-Fender | merlin8282: Don't forget zapata changes require a restart of * or a reload of chan_zap |
| 14:54.40 | lowtek | Hi guys, Tier 1 support is in the house, TK, go get some coffee, I got it from here, LOL |
| 14:54.56 | eppigy | holla |
| 14:55.12 | [TK]D-Fender | calculates the vertical drop from "tier 1" |
| 14:55.18 | lowtek | lol |
| 14:55.37 | [TK]D-Fender | pushes lowtek off the topp and watches him fall to a grisly demise |
| 15:01.07 | adam000 | Does anyone here use voicepulse? |
| 15:02.34 | *** join/#asterisk putnopvut (n=putnopvu@nat/digium/x-73ab3a5d788d3cf2) |
| 15:02.34 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 15:03.07 | bn43 | I'm reading for recording my phone does notification via SIP Info - does the latest 1.4 Asterisk support this? |
| 15:03.32 | [TK]D-Fender | bn43: Yes... PASTEBIN is your friend... perhaps you should show us what you're doing. |
| 15:03.48 | [TK]D-Fender | bn43: "dtmfmode=info" <- |
| 15:04.02 | [TK]D-Fender | bn43: And you said yo already confirme that Vm navigation works on it. |
| 15:04.06 | [TK]D-Fender | bn43: a moot point. |
| 15:04.09 | lowtek | adam000: What's your question? |
| 15:05.14 | rwaite | anyone know of any softphones for windows besides zoiper and x-lite? |
| 15:05.15 | [TK]D-Fender | lowtek: [10:01]<adam000>Does anyone here use voicepulse? |
| 15:05.21 | [TK]D-Fender | lowtek: Didn't you just read him? :p |
| 15:05.37 | rwaite | i'm leaning toward choosing zoiper but i'd just like alternatives to test with |
| 15:05.47 | [TK]D-Fender | rwaite: Ekiga , Snom |
| 15:06.11 | rwaite | thanks |
| 15:06.52 | adam000 | I'm looking into some different VoIP trunk providers - I'm trying vitelity now & I'm bringing up a voicepulse trunk today - I'm wondering if there's any concensus on which service providers are good/reliable & which ones are crappy. |
| 15:07.05 | bn43 | I'm trying to implement recording from the phone - http://asterisk.snom.com/index.php/Asterisk_1.4/Call_Recording |
| 15:07.06 | rwaite | wow the girl on the ekiga page is hot. |
| 15:07.42 | seanbright | rwaite: she's a russian bride |
| 15:07.58 | rwaite | i'd buy her in a millisecond |
| 15:08.20 | bn43 | which states that features.conf must have automon enable - which I have and DYNAMIC_FEATURES=automon in extensions.conf - which I have |
| 15:08.34 | bn43 | *1 should then manually allow me to record |
| 15:08.46 | *** join/#asterisk coronaXP (n=tochao@220-128-103-58.HINET-IP.hinet.net) |
| 15:08.57 | bn43 | but if I want to use the record button I need to include wW in my contexts |
| 15:09.14 | bn43 | Haven't got to first base yet with *1 |
| 15:09.19 | bn43 | there is no response |
| 15:09.48 | bn43 | driving me nuts - should I pastebin all relevant files? |
| 15:10.30 | *** join/#asterisk moy (n=chatzill@bas1-unionville55-1177733953.dsl.bell.ca) |
| 15:10.40 | merlin8282 | hum, tried to test with an almost empty ext.conf, but same result (and i've restarted asterisk) : http://pastebin.ca/1320660 |
| 15:10.47 | *** join/#asterisk icel (n=dan@75.146.143.126) |
| 15:11.01 | [TK]D-Fender | bn43: Didn't I already ask for that? |
| 15:11.12 | bn43 | sorry - busy doing |
| 15:11.16 | merlin8282 | Only keys 4, 5, * and # don't seem to do anything (but sending DTMF) |
| 15:11.42 | merlin8282 | strange behaviour, i mean |
| 15:11.44 | *** join/#asterisk russellb (n=russell@asterisk/digium-open-source-team-lead/russellb) |
| 15:11.44 | *** mode/#asterisk [+o russellb] by ChanServ |
| 15:11.48 | lowtek | merlin8282: dialplan? |
| 15:11.56 | merlin8282 | lowtek: look at my pastebin |
| 15:12.15 | [TK]D-Fender | rwaite: I have her URI.... I'm ordering now :p |
| 15:13.47 | merlin8282 | and, "sip show peers" returns my sip phone in status "ok" |
| 15:14.01 | merlin8282 | with number 416 |
| 15:15.39 | bn43 | http://pastebin.com/d5a5e98cc |
| 15:15.49 | lowtek | merlin8282: pastebin your extensions.conf |
| 15:15.50 | merlin8282 | dialing 416 from my second sip phone works, but not from analog one. |
| 15:16.09 | lowtek | merlin8282: and sip.conf |
| 15:17.41 | bn43 | another sidenote - I'm looking to replace a pabx where there are some people in the office who are not ours and we need to provide them with a detailed report on calls made and costs - I've had a look at a2billing but that only seems operational for calling cards - what do you guys use? |
| 15:17.46 | merlin8282 | lowtek: http://pastebin.ca/1320670 |
| 15:18.28 | *** join/#asterisk RoyK_ (n=roy@ip-140-26-149-91.dialup.ice.no) |
| 15:18.37 | merlin8282 | but, isn't it simpler to pastebin my "dialplan show" ? |
| 15:18.44 | RoyK_ | does asterisk have a working CNG? |
| 15:19.30 | merlin8282 | (mmm, maybe too long :/) |
| 15:19.42 | RoyK_ | ~pb |
| 15:19.43 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste, or , http://bin.cakephp.org/ |
| 15:20.50 | [TK]D-Fender | bn43: Look at Areski for CDR analysis, etc. a2billing is for CC stuff |
| 15:20.54 | [TK]D-Fender | bn43: and I asked to see the CALL |
| 15:21.28 | [TK]D-Fender | bn43: and your peers do no list the DTMF mode. Please correct, then remove all comments from your confis permanently and repastebin following the next attempt |
| 15:21.43 | [TK]D-Fender | RoyK: No. |
| 15:22.48 | bn43 | oh right - this is the call http://pastebin.com/d28c4b0b2 |
| 15:23.06 | *** join/#asterisk docelm0 (n=vircuser@pool-141-152-220-84.lyn.east.verizon.net) |
| 15:24.18 | *** join/#asterisk Defraz (n=T0tal@72-24-26-22.cpe.cableone.net) |
| 15:26.13 | [TK]D-Fender | bn43: Don't forget you have VERY little time between digits... |
| 15:26.28 | *** join/#asterisk manxpower (n=Administ@router.asteriasgi.com) |
| 15:26.59 | bn43 | [TK]D-Fender: I come across what you have said on google - what do you mean? |
| 15:27.28 | bn43 | doe this mean that I have to punch combination fast in order to take? |
| 15:27.29 | *** join/#asterisk `paul (n=kutimoy@119.93.45.181) |
| 15:27.39 | [TK]D-Fender | bn43: yes. 500ms |
| 15:27.46 | [TK]D-Fender | bn43: dial faster :) |
| 15:28.26 | `paul | when i login as agent how come i recv a "Extension '7009' is not valid for automatic login of agent '7009'" warning? |
| 15:29.22 | *** join/#asterisk Dovid (n=annon@tony09-118-62.inter.net.il) |
| 15:29.36 | bn43 | aha - pounded those keys and got this sox soxio: Can't open input file `/var/spool/asterisk/monitor/auto-1233156522-200-100-out.wav': No such file or directory |
| 15:30.41 | bn43 | asterski - last development 2005? |
| 15:30.48 | Dovid | Hi. I am trying to NoOP out ${CDR(billsec)} on the h extension. As per http://www.voip-info.org/wiki/view/Asterisk+Detailed+Variable+List I have set endbeforehexten=yes in cdr.conf but it still comes up as 0 for some reason |
| 15:30.52 | [TK]D-Fender | bn43: check the WIKI for others... |
| 15:31.17 | bn43 | will do |
| 15:31.48 | [TK]D-Fender | `paul: PASTEBIN your configs & CLI output. |
| 15:31.50 | [TK]D-Fender | ~pb |
| 15:31.51 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste, or , http://bin.cakephp.org/ |
| 15:31.52 | [TK]D-Fender | ^^^^^^^^ |
| 15:32.21 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 15:32.42 | jaytee | [TK]D-Fender, seen Katty lately? |
| 15:32.44 | *** join/#asterisk axisys (n=axisys@155.70.141.45) |
| 15:33.13 | [TK]D-Fender | jaytee: Been about a week |
| 15:33.35 | jaytee | strange, she was in my list of Facebook friends but now she's gone. |
| 15:33.45 | seanbright | she's still twittering |
| 15:33.45 | eppigy | fictional internet persona |
| 15:34.05 | jaytee | fictional like as in "Dave Casey" |
| 15:34.09 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 15:34.16 | eppigy | jaytee: that is a hurtful thing to say |
| 15:34.37 | [TK]D-Fender | jaytee: Maybe she didn't recognize you, I know she's done a lot of housecleaning in FB |
| 15:34.46 | [TK]D-Fender | jaytee: I see a twitter update from 11 hrs ago |
| 15:34.49 | eppigy | so is she a hotty? |
| 15:34.52 | eppigy | or internet fatty |
| 15:34.59 | eppigy | pleas eindulge my curiosity |
| 15:35.02 | eppigy | indulge |
| 15:35.25 | seanbright | she's attractive |
| 15:35.31 | seanbright | that is all i will divulge |
| 15:35.35 | eppigy | I see |
| 15:36.07 | seanbright | but she loses straight dialplan instead of AEL, so she loses a few points |
| 15:36.14 | seanbright | s/loses/uses/ |
| 15:36.22 | eppigy | lol |
| 15:36.35 | Dovid | TK: Did you see my question ? |
| 15:37.07 | bn43 | [TK]D-Fender: seems recording via *1 is working! /var/spool/asterisk/monitor/ shows files but there was no output on console so I did not know it was recording and where |
| 15:37.16 | *** join/#asterisk quaqo (n=quaqo@83-103-40-166.ip.fastwebnet.it) |
| 15:37.21 | bn43 | why is that? |
| 15:37.24 | *** join/#asterisk skirmisha (n=viki@87-126-66-51.btc-net.bg) |
| 15:37.30 | [TK]D-Fender | bn43: max out your verbose & core and you'll see it |
| 15:37.30 | skirmisha | hello guys |
| 15:37.40 | skirmisha | can someone help me with auth in asterisk |
| 15:37.46 | skirmisha | i have strange thing happening |
| 15:37.48 | manxpower | she's a tad high strung, but a cool person. A tad odd like most of us here. |
| 15:37.49 | *** join/#asterisk feeds (n=feeds@85-135-242-146.adsl.slovanet.sk) |
| 15:37.53 | manxpower | ~ask |
| 15:37.54 | jbot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 15:38.18 | bn43 | is jbot a person? |
| 15:38.21 | [TK]D-Fender | manxpower: I object! I am not a "tad"! |
| 15:38.26 | [TK]D-Fender | ~jbot |
| 15:38.27 | jbot | i heard jbot is a hack!, or known to have only said one useful thing. a tool, or dating the fembots, or [TK]D-Fender's b*tch, or suck, or a pain in the ass |
| 15:38.28 | bn43 | ok silly question |
| 15:38.32 | skirmisha | first time i get 401 and after that i get ok |
| 15:38.53 | [TK]D-Fender | skirmisha: 401 = GTFO |
| 15:39.06 | jaytee | I enjoy chatting with her and liked her blogs on sleekgeek. She'd actually go to a lot of trouble to write howtos that weren't bad for setting up FOP or using the blacklist feature |
| 15:39.09 | eppigy | i am insane |
| 15:39.13 | jaytee | we know |
| 15:39.18 | skirmisha | what i notice is that in second request i send Authorization: Digest .... |
| 15:39.19 | jaytee | but we won't hold it against you |
| 15:39.25 | skirmisha | any ideas why this is happening |
| 15:39.26 | eppigy | :D |
| 15:39.38 | manxpower | skirmisha: that is normal |
| 15:40.04 | jaytee | 401="You are dead to me!" |
| 15:40.18 | manxpower | Most SIP devices will sent the first request without auth, get rejected, then send again with auth |
| 15:41.08 | skirmisha | where are these options in asterisk to disable it |
| 15:41.17 | skirmisha | is it auth md5? |
| 15:41.20 | manxpower | there are none |
| 15:41.26 | bn43 | right now all I have to do is get the record button to work instead of *1 |
| 15:41.51 | skirmisha | manxpower ok , but sometimes phone does not send second request and thus it can not register |
| 15:42.01 | manxpower | skirmisha: that would be the problem with the phone. |
| 15:42.34 | manxpower | Since you are the only person on the planet with this problem that we are aware of...... |
| 15:42.43 | manxpower | I bet you have some *other* problem. |
| 15:42.58 | skirmisha | no no |
| 15:43.04 | skirmisha | it is not from the phone |
| 15:43.30 | skirmisha | because same phone can register on one of the servers and on the other one can not |
| 15:43.34 | skirmisha | servers has same config |
| 15:43.53 | manxpower | As I said chances are you have some *other* problem. |
| 15:44.17 | skirmisha | ok let me explain again |
| 15:44.31 | skirmisha | same phone register on diff servers with same config |
| 15:44.43 | skirmisha | on first one i do not see second request with auth in it |
| 15:44.49 | skirmisha | on second one i see that request |
| 15:44.57 | skirmisha | could it be some cache? |
| 15:45.08 | manxpower | So it's not the phone. That leaves the network, NAT, routing, Asterisk, and firewalls. |
| 15:45.25 | skirmisha | servers are on same network, just diff ips |
| 15:45.35 | skirmisha | i am connectiong from same place |
| 15:46.02 | skirmisha | shall i look the problem in asterisk config or it is phone problem or cache problem |
| 15:46.31 | manxpower | skirmisha: how are you planning on getting the phone to send the registration with auth the first time it tries? |
| 15:47.11 | merlin8282 | thanks for the help, i have to go. Bye |
| 15:47.16 | manxpower | You are basically doing the VoIP equiv of having cancer and only treating the tummy ache. |
| 15:47.30 | manxpower | find and treat the CAUSE not the symptoms |
| 15:47.36 | bn43 | [TK]D-Fender: what did you mean about enabling DTMF for peers? I assume it was the uncommenting of dtmfmode = rfc2833 ? |
| 15:47.45 | skirmisha | actually i have basic setup of the phone |
| 15:47.55 | [TK]D-Fender | bn43: You told me you were on sip info.... not rfc2833 |
| 15:48.03 | rwaite | pronounces manxpower "mainks power" |
| 15:48.30 | skirmisha | so those requests are send auto |
| 15:48.31 | *** join/#asterisk seanmh (n=johndoe@216.31.95.99) |
| 15:48.57 | bn43 | u lost me there |
| 15:49.39 | *** join/#asterisk AndyML (n=quassel@pool-72-78-117-135.phlapa.fios.verizon.net) |
| 15:49.59 | bn43 | um what I said was does asterisk support sip info - it did not click that sip info was related to dtmf setting |
| 15:50.02 | [TK]D-Fender | [10:03]<bn43>I'm reading for recording my phone does notification via SIP Info - does the latest 1.4 Asterisk support this? |
| 15:50.07 | *** part/#asterisk merlin8282 (n=merlin82@88-122-137-192.rev.libertysurf.net) |
| 15:50.19 | [TK]D-Fender | bn43: It does, you have to TELL * what mode to use... PER DEVICE |
| 15:50.35 | [TK]D-Fender | bn43: rfc2833 != "SIP INFO" |
| 15:51.18 | skirmisha | manxpower what would u advise me? |
| 15:52.05 | bn43 | ok - but I have read that if you change rfc2833 then normal dtmf ie to access mailbox etc will not work? am I talking shite? |
| 15:52.09 | *** part/#asterisk `paul (n=kutimoy@119.93.45.181) |
| 15:53.18 | [TK]D-Fender | bn43: You tell me one thing and contradict it with your actions. You are at a minimum, confused. |
| 15:53.33 | bn43 | yes I am! |
| 15:53.39 | [TK]D-Fender | bn43: Either way it seems to work and I don't recall you telling me what phones you even had. |
| 15:54.04 | [TK]D-Fender | bn43: Now what is this "record" button you're referring to? |
| 15:54.07 | skirmisha | how can i disable Authorization: Digest to be checked in asterisk? |
| 15:54.20 | bn43 | last night I was complaining that my snom320 was requesting password and u told me to chuck it |
| 15:55.13 | [TK]D-Fender | "check" |
| 15:55.29 | bn43 | there is a button on the snom which is labelled record |
| 15:55.52 | *** join/#asterisk ming_zym (n=ming_zym@f0-0-tep-rtr1.corp.cnb.yahoo.com) |
| 15:55.54 | [TK]D-Fender | bn43: Ok, well go read your manual as to what you can program that button to do, because the only triggered recording option is via DTMF |
| 15:56.19 | bn43 | ok |
| 15:57.07 | skirmisha | come on guys |
| 15:57.16 | skirmisha | give me some clue |
| 15:57.56 | *** join/#asterisk CrazyTux (n=brandon@ip68-111-67-4.oc.oc.cox.net) |
| 15:57.57 | telnettech | good morning....i am having difficulties in getting the asterisk to connect to MySQL database on another server. I have verified config and permissions on both servers to allow the connection thru the res_mysql and res_odbc files and the permission table on MySQL server |
| 15:58.15 | jaytee | skirmisha, Colonel Mustard in the drawing room |
| 15:58.25 | skirmisha | is mysql listening on public ip? |
| 15:58.33 | skirmisha | did u add user to mysql table? |
| 15:58.38 | telnettech | i am getting the error no=2003 can connect to MySQL server @ |
| 15:58.44 | lowtek | telnettech: Did you tell mysql to listen on an available IP address and give the other host permission in mysql to connect? |
| 15:58.58 | telnettech | yes |
| 15:59.21 | telnettech | we recently had to reload the OS and MySQL database on the server and since then it is not connecting |
| 15:59.23 | lowtek | telnettech: can you connect with the mysql client from your asterisk server? |
| 15:59.37 | telnettech | no the asterisk server cant connect |
| 15:59.48 | telnettech | i can ssh to the ip of the server though |
| 15:59.53 | lowtek | Can you connect with the "mysql" binary program/client from the asterisk server? |
| 16:00.00 | telnettech | no |
| 16:00.03 | *** join/#asterisk tobias (n=tobias@cpe-069-134-127-101.nc.res.rr.com) |
| 16:00.27 | lowtek | telnettech: So the problem isn't that asterisk can't connect ... |
| 16:00.29 | telnettech | like i said, i can ssh to the server and then do the mysql to connect to database |
| 16:01.00 | lowtek | telnettech: You can ssh to the asterisk server, type the "mysql" command plus args and make a connect to another server's mysql db? |
| 16:01.19 | lowtek | telnettech: What error is asterisk giving you when it tries to connect? |
| 16:01.27 | telnettech | no i am logged onto the asterisk server and can ssh to the mysql server |
| 16:01.34 | eppigy | lol |
| 16:01.35 | *** join/#asterisk sulan (n=sulan@89-253-95-143.customers.ownit.se) |
| 16:01.51 | eppigy | in mysql you stipulate whichy hosts users can connect from |
| 16:01.56 | lowtek | telnettech: that's irrelevant. Can you use the freakin "mysql" command FROM your asterisk server's console to connect to your DB on the OTHER server running the mysql-server? |
| 16:02.09 | telnettech | no i cant |
| 16:02.14 | *** join/#asterisk Firass-z0r (n=asadf@c-67-201-205-34.reshall.wwu.edu) |
| 16:02.19 | lowtek | telnettech: Ok, start there. |
| 16:02.26 | telnettech | i get cant connect to local mysql server |
| 16:02.37 | telnettech | through socket |
| 16:02.41 | eppigy | dog |
| 16:02.51 | lowtek | telnettech: Try #mysql,your problem isn't with asterisk, it's with mysql. |
| 16:03.05 | sulan | Hi guys, I need some help figuring out why DialAsync AMI has a very low level of concurrency (that is, few simultaneous calls showing up on the SIP-side even though a bunch have been queued with DialAsync) |
| 16:04.39 | lowtek | telnettech: First get a successful connection from your mysql client on the asterisk server to the database server, after you establish connectivity, it will be much easier to troubleshoot asterisk (if it doesn't just work because I have a feeling your problem is mysql permissions or that you just don't have the client installed or you compiled asterisk without mysql-dev being installed) |
| 16:04.55 | rwaite | the problem with mysql is mysql |
| 16:05.23 | sulan | s/DialAsync/Originate (async)/g |
| 16:05.31 | telnettech | it worked before the reload so i believe it is on the mysql server |
| 16:05.48 | lowtek | The mysql server was rebuilt? |
| 16:06.11 | telnettech | we had to reload the OS and the mysql server do to other issue so I believe it is on the mysql server but someone else in office thinks it is the asterisk server |
| 16:06.16 | lowtek | telnettech: If your mysql server was rebuilt and your asterisk server hasn't changed then *clearly* the problem exists with your mysql server. |
| 16:06.52 | lowtek | telnettech: Check my.cnf to ensure it's actually listening on the IP addresses you need it to, by default it will only allow local connections. |
| 16:07.38 | *** join/#asterisk ddl (i=erikw@suiko.acc.umu.se) |
| 16:08.09 | ddl | Is it possible to disable the 100 Trying that is sent on a REGISTER? |
| 16:08.34 | telnettech | ok thanks lowtek......sorry to sound like i dont know nothing( i am new to this) what directory is the my.cnf in |
| 16:08.41 | *** join/#asterisk Rico29 (n=Rico@lns-bzn-48f-81-56-220-28.adsl.proxad.net) |
| 16:09.04 | jaytee | it's in /etc |
| 16:09.29 | lowtek | telnettech: or /etc/mysql/my.cnf, what distro? |
| 16:09.34 | Mark17 | is it possible to include a file in the extensions.conf? so it would be used like all content from that file was located in extensions.conf (with the context where you include it)? |
| 16:09.36 | [TK]D-Fender | ddl: Only by direct coding in chan_sip |
| 16:09.45 | *** join/#asterisk Deeewayne (n=Deeewayn@nat/digium/x-6afe2d3a716d4e8c) |
| 16:09.45 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 16:10.10 | [TK]D-Fender | Mark17: include => /etc/asterisk/someotherfile.conf |
| 16:10.21 | [TK]D-Fender | Mark17: #include /etc/asterisk/someotherfile.conf |
| 16:10.23 | [TK]D-Fender | rather |
| 16:10.31 | lowtek | telnettech: Also, if you didn't "restore" your mysql database from a backup, you'll need to respecify permissions using "GRANT" |
| 16:10.55 | telnettech | ok thanks lowtek...i am working on it now |
| 16:11.03 | telnettech | appreciate your help |
| 16:11.25 | ddl | [TK]D-Fender: Ok.. Thanks |
| 16:11.47 | ddl | [TK]D-Fender: Has it always been that way? I can't recall seeing it previously... |
| 16:12.10 | [TK]D-Fender | ddl: Yup |
| 16:12.32 | ddl | [TK]D-Fender: Ok |
| 16:13.05 | *** join/#asterisk erth64net (n=erth64ne@69-30-67-191.dq1sn.easystreet.com) |
| 16:13.19 | *** join/#asterisk hfb (n=hfb@pool-96-247-109-136.lsanca.dsl-w.verizon.net) |
| 16:17.44 | Mark17 | [TK]D-Fender: thnx, i didnt know that that should work without any issues |
| 16:19.00 | fexy | What happened to realtime support in sccp? Is there a better way to having cisco phones being just plug and go using sccp images? |
| 16:19.06 | fexy | err to have |
| 16:19.09 | fexy | and be |
| 16:19.14 | fexy | damn I need more coffee |
| 16:20.02 | bn43 | wanted to get the record button right but enabled dtmf via browser and it now works - win some lose some. |
| 16:20.24 | bn43 | what does bistruffed mean? |
| 16:21.09 | *** join/#asterisk chi6IT41 (n=chigital@tmo-104-178.customers.d1-online.com) |
| 16:21.40 | *** join/#asterisk rootforce (n=chatzill@office.aircanopy.net) |
| 16:23.44 | *** join/#asterisk n3hxs (n=HAMming@63.68.135.4) |
| 16:24.33 | *** join/#asterisk asterisk101 (n=elastixr@196.211.34.2) |
| 16:24.43 | asterisk101 | hi there all |
| 16:24.53 | *** join/#asterisk mascool (n=mascool@c-98-243-123-165.hsd1.mi.comcast.net) |
| 16:25.12 | asterisk101 | i have some sip trunks 5 to be exact need to but them in a outbound group how would one do that? |
| 16:25.36 | bn43 | is there a way for me to email the recordings made to the recorder of the messages? like with voicemail? |
| 16:26.52 | lowtek | bn43: You can run a script when the call is terminated to search for recordings and then build some logic to figure out where to email them. |
| 16:27.13 | asterisk101 | is it possible to add them in a group like adding group=1 in sip.conf settings, will this maybe dial all of them if you point a call to the group? |
| 16:27.25 | lowtek | bn43: Or better yet, store the unique id in the cdr userfield and then allow your users to login to a portal somewhere to review their recordings |
| 16:28.14 | bn43 | newbie here! gonna have to learn how to script - I'm just about discovering bash so its gonna take me awhile |
| 16:28.39 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 16:28.56 | bn43 | I was just assuming there would be a line I would add - cause the voicemail function works very similar |
| 16:29.01 | *** join/#asterisk ReDNeQ (n=ReDNeQ@70.114.226.70) |
| 16:31.46 | asterisk101 | any help would be welcome |
| 16:33.02 | lowtek | bn43: There's no app in asterisk that I know of that will let you review call recordings like voicemail but that's a good idea ... |
| 16:33.09 | [TK]D-Fender | asterisk101: There is no "grouping" with SIP peers. You have to cycle through them manually in dialplan yourself |
| 16:33.23 | asterisk101 | thx |
| 16:33.37 | [TK]D-Fender | bn43: Minimal AGI work for this. |
| 16:35.19 | *** join/#asterisk srd (i=srd@wombat.frictious.net) |
| 16:37.59 | bn43 | [TK]D-Fender: do you mean easy scripting? |
| 16:38.44 | [TK]D-Fender | bn43: Fairly. Scan files in folder. Prompt for date or something, playback the file,etc |
| 16:39.15 | bn43 | one day I'm going to look back at this conversation and say - THATS EASY! |
| 16:39.40 | bn43 | right now code looks alien to me - but I just got to plug away at it |
| 16:39.43 | *** join/#asterisk freakazoid0223 (n=matt@pool-71-246-17-74.phlapa.fios.verizon.net) |
| 16:40.04 | jjshoe | bn43 it's a fairly trivial agi to write |
| 16:40.22 | jjshoe | bn43 if you're not comfortable you can always post a bounty somewhere and have someone do it for you |
| 16:40.29 | jjshoe | bn43 $100 will get you a long way. |
| 16:41.07 | lowtek | bn43: Are you trying to email all the recordings to a single email address? |
| 16:42.20 | [TK]D-Fender | bn43: Oh.. E-MAIL. |
| 16:42.25 | [TK]D-Fender | bn43: thats really easy... |
| 16:43.25 | [TK]D-Fender | bn43: "core show application monitor" <- I know there is a var you can set for the script to call upon completion of a recording. minimal bash to have it e-mailed out from three |
| 16:43.29 | [TK]D-Fender | there* |
| 16:43.50 | manxpower | would the AEL2 case statement support things like "case BUSY | CHANUNAVAIL | CONGESTION:" |
| 16:44.47 | vader-- | have any of you guys worked with the cisco ata 180 series devices? |
| 16:44.48 | *** join/#asterisk CunningPike (n=arodgers@204.239.10.119) |
| 16:44.49 | *** join/#asterisk bmoraca (n=bmoraca@209.60.253.58) |
| 16:44.56 | lowtek | vader--: Yes |
| 16:45.06 | vader-- | i am trying to udnerstand these configurations it wants, i have never worked with Bitmap values |
| 16:45.08 | manxpower | vader--: a long time ago |
| 16:45.19 | lowtek | ... in a galaxy far, fa.. oh |
| 16:45.19 | vader-- | im trying to setup the vlan and opFlags settings |
| 16:47.45 | wonderworld | hi, i am trying to install chan_capi on an ubuntu box (intrepid) there is a package asterisk-chan-capi which i installed, but asterisk didn't seem to utilize it. when i do a "help capi" in the cli, i get : no such command: capi |
| 16:48.02 | wonderworld | do i need to enable capi somehow? |
| 16:48.15 | [TK]D-Fender | wonderworld: What happens when you try to load the module manually? |
| 16:48.19 | *** join/#asterisk mintwork (n=mintone@75.150.132.150) |
| 16:49.34 | wonderworld | modprobe capi loaded it (capi, capifs and kernelcapi are in lsmod) |
| 16:49.45 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 16:50.39 | *** join/#asterisk vgster (n=vgster@94-194-190-189.zone8.bethere.co.uk) |
| 16:50.41 | *** join/#asterisk deadpigeon (n=deadpige@office.xpressamerica.net) |
| 16:50.50 | vader-- | do you guys know much about these bitmap values? |
| 16:51.33 | lowtek | .. these aren't the bitmap values you're looking for, move along ... |
| 16:51.53 | vader-- | ? |
| 16:52.21 | lowtek | .. the force is strong with this one .. |
| 16:53.25 | wonderworld | [TK]D-Fender: never mind, forgot to enable it in modules.conf |
| 16:56.23 | *** join/#asterisk riddlebox (n=user@mscitspubwlgw.wustl.edu) |
| 16:57.27 | *** join/#asterisk Defraz (n=T0tal@fw-poky.fuzecore.com) |
| 16:59.59 | magronez | is back |
| 17:03.07 | *** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun) |
| 17:04.27 | deadpigeon | hi, any reason why dahdi would ignore arguements in chan_dahdi.conf? im seeing errors like: |
| 17:04.46 | deadpigeon | chan_dahdi.c: ignoring signaling, ignoring switchtype, ignoring pridialplan, etcetera. |
| 17:04.47 | *** join/#asterisk stevetotaro (n=Steve@pool-71-254-231-87.hrbgpa.east.verizon.net) |
| 17:05.00 | manxpower | deadpigeon: it can't change those on reload so it ignores CHANGES |
| 17:05.12 | deadpigeon | ah. |
| 17:05.43 | deadpigeon | so i'd have to bring it down completely for it to take those changes? |
| 17:05.55 | manxpower | correct |
| 17:05.57 | deadpigeon | k. i get an invalid arguement when it reads gains... |
| 17:06.10 | manxpower | you seldom need to change those options |
| 17:06.29 | deadpigeon | right, i see. im new to it, trying to get a pri trunk working. |
| 17:06.36 | RoyK | does asterisk support SS/CNG yet? |
| 17:06.54 | deadpigeon | Failed to read gains: Invalid argument |
| 17:07.06 | deadpigeon | should i worry about that? |
| 17:07.08 | manxpower | deadpigeon: what is the gain set to? |
| 17:07.22 | deadpigeon | well, at first it was never set. I wasn't aware I had to set it. I just set tx and rx to 1.0 |
| 17:07.39 | manxpower | deadpigeon: paste the rxgain and txgain |
| 17:07.42 | deadpigeon | so I probably need to reboot the server now. but I didnt even know that had to exist. |
| 17:07.52 | manxpower | you can change gains on the fly |
| 17:08.02 | manxpower | that's why it is not saying it's ignoreing them |
| 17:08.07 | deadpigeon | uh, well I dont know what to set it to. so I did this: |
| 17:08.11 | deadpigeon | rxgain=1.0 |
| 17:08.11 | deadpigeon | txgain=1.0 |
| 17:08.22 | manxpower | remove the .0 |
| 17:08.29 | deadpigeon | k. |
| 17:08.42 | manxpower | and 0 is "no change" 1 means "increase by 1db" |
| 17:08.56 | deadpigeon | I see. Cool. Thanks for the help. |
| 17:09.35 | deadpigeon | I couldn't find a switchtype setting for a Coppercom CSX 1100, so I assumed switchtype=national |
| 17:09.44 | deadpigeon | not much info out there for that particular softswitch. |
| 17:10.07 | manxpower | switchtype is for PRI only |
| 17:10.15 | deadpigeon | right. i am doing pri. |
| 17:10.31 | manxpower | ni2 aka national is the most common |
| 17:10.40 | deadpigeon | we're a clec, we have a coppercom softswitch and im setting up asterisk for office phones. |
| 17:10.51 | deadpigeon | still working on getting the pri's provisioned on the coppercom. |
| 17:14.35 | wonderworld | how do the telcos handle the voice traffic on their side? do they bundle multiple PRIs ? |
| 17:14.52 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 17:15.47 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 17:15.47 | *** join/#asterisk oej (n=olle@80.251.192.2) |
| 17:16.58 | angryuser | wonderworld: depends on operator |
| 17:17.06 | *** join/#asterisk nicoAMG (i=asgalt@201.203.96.42) |
| 17:18.28 | manxpower | [TK]D-Fender: now I'm getting to the "WTF was I thinking!?!?" section of my scripts/ |
| 17:29.47 | *** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 17:29.48 | [TK]D-Fender | manxpower: you = FUBAR'd now.. |
| 17:30.09 | manxpower | I'll figure it out |
| 17:30.13 | *** part/#asterisk RoyK (n=roy@ip-140-26-149-91.dialup.ice.no) |
| 17:33.21 | *** join/#asterisk oej (n=olle@80.251.192.2) |
| 17:40.02 | jazzsunn | sorry for the OT here, but does anyone have a suggestion for an open ip surveillance monitoring solution? |
| 17:45.57 | fexy | you mean open source? |
| 17:46.33 | jazzsunn | yes |
| 17:46.38 | angryuser | jazzsunn: centreon cacti |
| 17:46.42 | *** join/#asterisk keebler (i=9446c2d5@gateway/web/ajax/mibbit.com/x-bde524e9d1bfb545) |
| 17:46.56 | jazzsunn | cacti, as in the graphing software? |
| 17:47.51 | angryuser | jazzsunn: yes, it is collecting data from snmp and building graphs, same as centreon |
| 17:48.23 | angryuser | jazzsunn: but think centreon is more advanced |
| 17:48.34 | angryuser | and user friendly |
| 17:48.39 | *** join/#asterisk dogmeat (n=Bob@unaffiliated/dogmeat) |
| 17:48.52 | jazzsunn | sorry, i was unclear, i meant video surveillance. i only ask because this channel seems to be filled with quite a few astute folks |
| 17:49.36 | *** join/#asterisk nix8n82 (n=nate@63.162.27.243) |
| 17:49.52 | bn43 | [TK]D-Fender: sorry I had to go into a meeting |
| 17:50.11 | *** join/#asterisk Pan3D (n=Pan3D@node2.sensoryresearch.net) |
| 17:50.34 | bn43 | I am teaching myself bash scripting - what where u saying about the script? |
| 17:50.41 | jazzsunn | speaking of meetings, i'm off to one myself. thanks for the input. |
| 17:51.02 | bn43 | I'm looking to e-mail the recording to the recorder of the message |
| 17:51.37 | angryuser | jazzsunn: nope, buti hears something exist, googlz |
| 17:51.47 | angryuser | i heard* |
| 17:52.09 | [TK]D-Fender | bn43: read "monitor"'s instructions to see if you can set the exec script prior to one-touch recording. If so you can call your script and call Sendmail in it, etc to e-mail the file. |
| 17:55.21 | bn43 | ok thanks for the tip - will look it up |
| 17:59.36 | deadpigeon | i dont understand the dynamic span definitions in dahdi system.conf, are these needed? ive configured a TE10P t1 card already, just not sure what the dynamic span is used for |
| 18:03.17 | *** join/#asterisk maddog01 (n=minotaur@dhcp-0-1c-f0-eb-94-24.cpe.quickclic.net) |
| 18:05.00 | *** join/#asterisk xorl (n=xorl@sys-adm.org) |
| 18:05.17 | xorl | So, when asterisk responds back with "CONGESTION" |
| 18:05.28 | xorl | What does that mean, i've looked around but never really figured it out. |
| 18:06.03 | deadpigeon | i assume that means circuits are busy |
| 18:06.15 | deadpigeon | or no working channels. |
| 18:06.18 | mort_gib | <deadpigeon>: Not so |
| 18:06.20 | xorl | hmm, I keep getting it from my PRI provider |
| 18:06.45 | mort_gib | CONGESTION is like Microsofts 8XXX outlook error msg |
| 18:06.50 | xorl | but like 3-4 people from the group aren't on the phone, so i get the CONGESTION from them (they send me emails when this happens) |
| 18:07.06 | xorl | them being my pri provider, not the phones in the group |
| 18:07.06 | mort_gib | a "catch all" for a number of different faults |
| 18:07.25 | *** join/#asterisk maddog01 (n=minotaur@dhcp-0-1c-f0-eb-94-24.cpe.quickclic.net) |
| 18:07.29 | xorl | mort_gib: ah |
| 18:07.43 | xorl | Anyway I can possibly track the cause? |
| 18:07.50 | manxpower | when you get CONGESTION check HANGUPCAUSE, which is more specific |
| 18:08.06 | xorl | I have the asterisk console up with a verbosity of 8 and the sip debug but that doesn't help when the provider emails me the congestion and I don't see anything in my logs related to that |
| 18:08.33 | *** join/#asterisk guaxinim (n=guaxinim@unaffiliated/guaxinim) |
| 18:08.35 | mort_gib | xorl: I read somewhere that given the amount of technologies that asterisk understands they had to use some generic and quite unhelpful error messages |
| 18:08.37 | manxpower | oh, nevermind. You don't have access to that sever |
| 18:08.57 | xorl | manxpower: yeah, I am not the PRI. |
| 18:09.13 | manxpower | HANGUPCAUSE w/PRI gives you the actual Q.831 ISDN disconnect cause code. You can't get more specific than that? |
| 18:09.28 | manxpower | HANGUPCAUSE with SIP is almost as good. |
| 18:09.44 | manxpower | xorl: if you don't have a PRI stop saying "PRI provider" |
| 18:09.45 | xorl | ah, yeah, I am using SIP, so I might be able to do that |
| 18:10.12 | *** join/#asterisk path_ (n=path@76-98-21-190.adsl.terra.cl) |
| 18:10.14 | manxpower | xorl: if it's incoming calls from your provider to your server there is nothing you can do to diagnose it using HANGUPCAUSE or DIALSTATUS |
| 18:10.19 | mort_gib | manxpower: Will you get the PRI error reported back to your SIP device?? |
| 18:10.22 | *** join/#asterisk synthetiq (n=roger@unl201395.nl.customer.alter.net) |
| 18:11.18 | manxpower | mort_gib: no, but Asterisk seems to follow the SIP -> G.931 cause code translation RFC. |
| 18:11.18 | xorl | manxpower: Ah, yeah, they route the calls to my asterisk system (SIP). |
| 18:11.18 | synthetiq | im using asterisk 1.6 ... i see packets arrive on my inviterface for sip, but when sip debug mode is on ina sterisk, i see it receives nothing... any ideas why? |
| 18:11.49 | manxpower | xorl: all this is only useful for OUTGOING calls from your PBX, not incoming calls |
| 18:11.53 | [TK]D-Fender | synthetiq: Checked that chan_sip is loaded? That its got the port your think it did? |
| 18:12.00 | *** part/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 18:12.03 | [TK]D-Fender | synthetiq: Show us your debug attempt. |
| 18:12.14 | manxpower | [TK]D-Fender: sig debug command would error if chan_sip was not loaded. |
| 18:12.22 | [TK]D-Fender | synthetiq: Global? Or did something silly like assume you could restrict it to an IP and its wrong maybe... |
| 18:12.26 | xorl | manxpower: Thought that much. |
| 18:12.32 | synthetiq | chan sip is loaded |
| 18:12.33 | [TK]D-Fender | manxpower: trust = 0 :) |
| 18:12.35 | xorl | So it's going to be pretty hard to debug the incoming call issues. |
| 18:12.42 | *** part/#asterisk keebler (i=9446c2d5@gateway/web/ajax/mibbit.com/x-bde524e9d1bfb545) |
| 18:12.42 | manxpower | xorl: yup! |
| 18:12.54 | synthetiq | i do a sip set debug on |
| 18:12.58 | xorl | I was going to pick up a 4 channel Digium PRI card to handle all of this and just avoid the provider. |
| 18:13.10 | *** join/#asterisk watchy (n=watchy@76.196.98.139) |
| 18:13.14 | [TK]D-Fender | synthetiq: pastebin it all up |
| 18:13.15 | watchy | hey tk you there |
| 18:13.15 | xorl | We have a DS3 here, so I don't think i'll have any issues. |
| 18:13.19 | watchy | yes i guess you are |
| 18:13.29 | synthetiq | netstat -anp says asterisk is listening on that prot |
| 18:13.35 | watchy | tk: you think its worth my time to make a FOP in delphi/vb/c# for exp? |
| 18:14.08 | [TK]D-Fender | Been done, will be done again... |
| 18:14.16 | *** join/#asterisk nix8n82 (n=nate@63.162.27.243) |
| 18:14.16 | lowtek | LOL so stupid but I couldn't stop watching: http://www.youtube.com/watch?v=Dz04SsZB8xI&feature=dir |
| 18:14.20 | [TK]D-Fender | watchy: If it offers you something its for you to judge if tis worth it |
| 18:14.24 | *** join/#asterisk jjshoe (n=jjshoe@h69-129-142-83.mdsnwi.tisp.static.tds.net) |
| 18:14.26 | watchy | well i kinda want to do it for my own expeirence |
| 18:14.38 | watchy | i think its worth it tk, knowledge is the only reason i dont shoot myself |
| 18:15.00 | synthetiq | [TK]D-Fender: http://pastebin.com/m54ba70e3 |
| 18:15.11 | watchy | learning asterisk has really given me a new life knowledge wise man |
| 18:15.52 | [TK]D-Fender | lowtek: Brilliant |
| 18:16.35 | watchy | thanks for always helping me tk |
| 18:16.37 | [TK]D-Fender | synthetiq: And the CLI testing attempts and proof the packets are arriving... |
| 18:20.07 | Deeewayne | ~thebook |
| 18:20.08 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com |
| 18:20.14 | Deeewayne | thanks jbot! |
| 18:20.33 | *** join/#asterisk loather-work (n=khudson@internal-nat.djnetworks.net) |
| 18:20.34 | watchy | anyone here ever work with SIP intercoms/Speakers in Asterisk? |
| 18:20.43 | watchy | hey tk: do they make SIP Lights? |
| 18:20.44 | lowtek | watchy: in what regard? |
| 18:21.03 | watchy | lowtek: School district i put a system in wants to add their intercom to the * box |
| 18:21.04 | lowtek | watchy: Are you trying to make a loudspeaker system? |
| 18:21.09 | watchy | yea for a school |
| 18:21.13 | lowtek | watchy: It's really easy... |
| 18:21.14 | xorl | hmm, so I am misunderstaning this, routing to the PRI card, I would need to push it through a T-1 or similar interface or could I pump it off one of my catalysts over RJ-45 (cat5e)? |
| 18:21.14 | watchy | they already have the speakers |
| 18:21.21 | watchy | can i use the legacy speakers? |
| 18:21.25 | lowtek | watchy: Get you a Polycom 330 with the mini-din audio jack ... |
| 18:21.31 | watchy | hmm |
| 18:21.38 | watchy | that would broadcast to all speakers though |
| 18:21.40 | synthetiq | [TK]D-Fender: http://pastebin.com/m309e47b9 |
| 18:21.41 | loather-work | xorl: you need T1s or something that can emulate network-side ISDN |
| 18:21.42 | lowtek | watchy: Setup the phone and add the auto-answer parameters to the phone's sip.cfg |
| 18:21.45 | watchy | mind if i /msg you? |
| 18:21.50 | lowtek | lowtek: build the dialplan, let her rip |
| 18:21.51 | xorl | loather-work: ah |
| 18:21.53 | watchy | cause i'll fill this chan up with what i need to tell you |
| 18:21.58 | lowtek | s/lowtek/watchy |
| 18:22.05 | lowtek | watchy: uhm, no thanks. |
| 18:22.14 | watchy | ok. well 1 phone isnt going to work |
| 18:22.25 | watchy | i need every speaker to have its own extension like it is currently |
| 18:22.27 | xorl | loather-work: Would it be possible with a Cisco ISDN BRI Interface card? |
| 18:22.42 | synthetiq | also restarting asterisk was no help |
| 18:22.44 | lowtek | watchy: Then get 1 phone per extension, it will double as a classroom conference phone, lol. |
| 18:23.06 | [TK]D-Fender | synthetiq: thats an ngrep on your server? Where is your phone relative to your server? |
| 18:23.07 | watchy | well they have a wierd setui, the speakers are also connected to the analogs in every class |
| 18:23.12 | loather-work | xorl: no, PRI and BRI are not compatible like that. You'll need a channelized T1 board like a VWIC-1MFT-T1 and a voice carrier module like an NM-HDV in order to do what you want |
| 18:23.19 | lowtek | lowtek: Really, you need a SIP->mini-din device, there aren't any. |
| 18:23.22 | [TK]D-Fender | synthetiq: "netstat -an|grep 5060" |
| 18:23.23 | xorl | loather-work: ah |
| 18:23.25 | synthetiq | the phone is being nat [TK]D-Fender |
| 18:23.32 | *** join/#asterisk maddog01 (n=minotaur@dhcp-0-1c-f0-eb-94-24.cpe.quickclic.net) |
| 18:23.37 | watchy | when they call a class it comes over the intercom, when they pickup the analog it turns off the intercom speaker using relays |
| 18:23.41 | loather-work | xorl: and the cisco BRI interfaces only support CPE termination, not network. It's difficult to find devices that do BRI network-side. |
| 18:23.45 | synthetiq | [TK]D-Fender: udp 0 0 0.0.0.0:5060 0.0.0.0:* |
| 18:24.10 | xorl | loather-work: ah, those Multiflex trunk expansions aren't that expensive |
| 18:24.13 | xorl | might pick one of those up |
| 18:24.15 | lowtek | watchy: The polycom setup I mentioned would work the exact same way, when the handset is liften, the mini-din audio is turned off and it's redirected inside the phone via software to the handset. |
| 18:24.22 | lowtek | s/liften/lifted |
| 18:24.24 | [TK]D-Fender | synthetiq: Try your call again... still no sip debug from *>? |
| 18:24.24 | watchy | yea, ive used the auto answer |
| 18:24.34 | synthetiq | nothing [TK]D-Fender |
| 18:24.51 | [TK]D-Fender | synthetiq: check your firewalls |
| 18:24.52 | loather-work | xorl: yeah, but the NM-HDVs with fully-loaded DSPs aren't cheap. I think out the door I paid $500 for my setup of NM-HDV, VWIC-2MFT-T1-DI. |
| 18:25.05 | xorl | hmmm |
| 18:25.25 | synthetiq | [TK]D-Fender: openfire wall |
| 18:25.32 | loather-work | Then, whatever router you install it in (3640 is a good choice) needs an IOS PLUS featureset to get the entire ISDN stack. |
| 18:25.32 | synthetiq | the packet is arriving |
| 18:25.38 | xorl | well I'll need one of those plus a new digium card, only need the 1 port module for the server |
| 18:25.39 | synthetiq | asterisk does nothing with port 5060 |
| 18:26.18 | watchy | low: i think i'm gonna implent this SIP based speaker system that will connect to the existing legacy speakers |
| 18:26.20 | [TK]D-Fender | synthetiq: Something is screwy |
| 18:26.23 | loather-work | takes SIP in on ethernet and feeds it to the PRI board -- i use it as a dev PRI. |
| 18:26.28 | [TK]D-Fender | synthetiq: Describe your setup. |
| 18:26.41 | watchy | + they have speakers in the hall for the bell etc |
| 18:26.50 | synthetiq | argh |
| 18:26.53 | synthetiq | im here late |
| 18:26.53 | lowtek | There's a SIP based speaker system? |
| 18:26.54 | watchy | we have software that can connect to * and send out bells at certain times |
| 18:26.55 | *** join/#asterisk ChkDigit (n=mike@static24-72-71-175.regina.accesscomm.ca) |
| 18:26.55 | synthetiq | i give up |
| 18:26.58 | watchy | yea lowtek |
| 18:27.00 | synthetiq | im going home |
| 18:27.06 | watchy | you want a url to look at it? |
| 18:27.07 | lowtek | Neat, website? |
| 18:27.08 | synthetiq | gues sill recompile tomrrow |
| 18:27.20 | watchy | hold i gotta find i t, some voipsupply guy emailed me |
| 18:27.23 | [TK]D-Fender | synthetiq: Show me your ngrep attempt including the call from CLI |
| 18:27.23 | *** join/#asterisk killown (n=Yamato@unaffiliated/killown) |
| 18:27.40 | carrar | watchy, there a handfull of SIP paging gateways for asterisk |
| 18:27.53 | watchy | The CyberData SIP-enabled Paging Gateway allows the use of legacy analog zone paging amplifiers when converting to a VOIP system. The Gateway is compatible with most SIP-based IP/PBX servers that comply with the SIP RFC 3261. The Gateway is powered via PoE 802.3af - no external power supply is needed. |
| 18:28.04 | lowtek | rtain times |
| 18:28.05 | lowtek | * ChkDigit (n=mike@static24-72-71-175.regina.accesscomm.ca) has joined #a |
| 18:28.08 | loather-work | anyhow, is there a variable available that I can do something like, "exten => 911,1,noop("extension ${CALLING_EXT} called 911") |
| 18:28.09 | watchy | that work with analog/legacy speakers carrar? |
| 18:28.17 | lowtek | http://www.telephonydepot.com/Catalog/Speakers-Paging |
| 18:28.20 | carrar | that are on a amp yes |
| 18:28.33 | carrar | the paging gateway has a lineout |
| 18:28.38 | watchy | sweet |
| 18:28.40 | carrar | not a speaker out |
| 18:28.58 | watchy | http://www.cyberdata.net/ thats the website lowtek |
| 18:28.58 | carrar | lineout -> paging amp -> speaker |
| 18:29.03 | watchy | carrar: i need a SIP Strobe |
| 18:29.13 | watchy | do they make them? |
| 18:29.16 | carrar | light a flashing light? |
| 18:29.28 | loather-work | a sip strobe light? what on earth would you need something like that for? |
| 18:29.31 | watchy | yea, i have to put them in this .gov company |
| 18:29.36 | carrar | haha |
| 18:29.36 | watchy | they cant have phones in the rooms |
| 18:29.37 | carrar | cool |
| 18:29.42 | carrar | You could make one |
| 18:29.50 | watchy | like i need a bright one that strobes if they dial say 5555 |
| 18:29.56 | watchy | so they know to run for their livers |
| 18:29.57 | watchy | lives |
| 18:30.26 | loather-work | yeah, you could do something wtih an FXS port and a readily-available analog device activator |
| 18:30.27 | [TK]D-Fender | watchy: RadioShacks ells flashers like that you can run off FXS :)_ |
| 18:30.36 | watchy | yea thats what i was thinking |
| 18:30.37 | carrar | yeah |
| 18:30.42 | watchy | i need different colors |
| 18:30.55 | watchy | 1 for run like hell, 1 for other stuff |
| 18:30.59 | loather-work | multiple FXS ports and multiple actuators :) |
| 18:31.00 | carrar | haha |
| 18:31.12 | watchy | also i need it to be EP if thats even possible |
| 18:31.19 | loather-work | ep? |
| 18:31.24 | lowtek | watchy: that's a tall order. good luck! |
| 18:31.24 | watchy | explosion proof |
| 18:31.25 | [TK]D-Fender | loather-work: "core show function CALLERID" |
| 18:31.34 | *** join/#asterisk stevetotaro (n=Steve@pool-71-254-231-87.hrbgpa.east.verizon.net) |
| 18:31.47 | watchy | so when the strobe kicks on, it doesnt blow the facility up cause its caked in explosive dust |
| 18:31.59 | loather-work | [TK]D-Fender: oh, duh. thanks. :) |
| 18:32.00 | watchy | we had to pay $800 each for 50 red analog EP phones |
| 18:32.10 | manxpower | um, there are phones specially designed for explosive enviroments |
| 18:32.11 | lowtek | I want a SIP batphone ... |
| 18:32.14 | carrar | good luck finding a EP rated FXS |
| 18:32.26 | [TK]D-Fender | watchy: Is your wiring EP as well? Kevlar guilded? |
| 18:32.28 | watchy | well the FXS can be mounted outside |
| 18:32.41 | carrar | I am sure they make EP rated strobes |
| 18:32.44 | watchy | they have a bunker and a building outside the bunker where the FXS is |
| 18:32.58 | watchy | so a EP fxs isnt really an isue |
| 18:33.17 | carrar | watchy, where is this at? |
| 18:33.22 | watchy | camden, arkansas |
| 18:33.29 | watchy | we make lots of bombs here |
| 18:33.37 | watchy | missles/grenades etc |
| 18:33.43 | carrar | for which country? :) |
| 18:33.46 | watchy | USA |
| 18:33.53 | carrar | thats good |
| 18:34.14 | watchy | i'm about to do a phone system for a german company that makes explosives for us |
| 18:34.22 | *** join/#asterisk lanning (n=lanning@173.8.187.197) |
| 18:34.26 | watchy | but they have to be based in the US so they bought a big area here to make explosives |
| 18:34.43 | watchy | they cant ship the materials to germany and build them there, so they do it here |
| 18:34.50 | watchy | but the company is still german owned/etc |
| 18:37.14 | watchy | http://www.federalsignal-indust.com/default.asp?pageID=4&lookup=242 |
| 18:37.25 | watchy | that would work, gott figure out how to wire it to a FXS though |
| 18:37.40 | carrar | 2 wires |
| 18:38.15 | watchy | you think the FXS would push enough to current to make it light up? |
| 18:38.24 | carrar | no |
| 18:38.45 | carrar | it could 'trigger' something though |
| 18:39.00 | carrar | unless those are made for it |
| 18:39.09 | carrar | but they don't appear to |
| 18:39.49 | deadpigeon | hm |
| 18:39.55 | deadpigeon | still getting unable to read gains invalid arguement errors |
| 18:39.59 | watchy | im sure they have to make them though i'll just keep lookin |
| 18:41.10 | *** join/#asterisk maddog01 (n=minotaur@dhcp-0-1c-f0-eb-94-24.cpe.quickclic.net) |
| 18:41.13 | carrar | watchy, does it need to connect to asterisk via 2 wires? You can use the USB port to control lots of 'on'off' switches |
| 18:41.32 | carrar | and use relays on those to run 120vac to a device |
| 18:41.55 | carrar | usb logic controllers |
| 18:41.57 | watchy | yea, thats true |
| 18:42.07 | watchy | then i'd need a pc in the facility somewhere |
| 18:42.29 | watchy | im sure they make a phone activated relauy |
| 18:42.29 | carrar | use the asterisk box |
| 18:42.30 | watchy | relay. |
| 18:42.42 | watchy | its about 6000 feet away or further |
| 18:42.43 | carrar | yeah |
| 18:42.50 | carrar | back to the FXS |
| 18:43.00 | watchy | it might be closed to 2 miles away heh |
| 18:43.35 | carrar | http://www.electronic-circuits-diagrams.com/microcontrollers/microcontrollers1.shtml |
| 18:44.42 | carrar | lots of patents for TELEPHONE ACTUATED SWITCHs |
| 18:44.47 | watchy | http://www.quasarelectronics.com/3140-4-channel-dtmf-telephone-relay-switcher.htm |
| 18:44.50 | watchy | that looks useful |
| 18:45.14 | carrar | yeah |
| 18:45.16 | watchy | 3140 - 4-Channel DTMF Telephone Relay SwitcherSimply plugs into your telephone line* and provides 4 independently controllable relay outputs. You can then call your number using a DTMF (tone dialling) telephone from anywhere in the world and remotely turn on/off any of the relays as desired. |
| 18:45.21 | watchy | sweet |
| 18:46.10 | *** join/#asterisk ReDNeQ (n=ReDNeQ@70.114.226.70) |
| 18:46.46 | watchy | $56 for one with 4 relays |
| 18:47.41 | watchy | well they do exist so thats good |
| 18:47.51 | *** join/#asterisk LemensTS (n=customgt@adsl-70-238-154-243.dsl.stlsmo.sbcglobal.net) |
| 18:49.25 | LemensTS | relaxdtmf=yes |
| 18:49.25 | LemensTS | dtmfmode=rfc2833 |
| 18:49.25 | LemensTS | is there another one im missing for sip.conf? having a little dtmf issue..on asterisk 1.2 |
| 18:50.11 | manxpower | LemensTS: the reason DTMF stuff got a significant rewrite in 1.4 chan_sip is because 1.2's DTMF support is so horrid |
| 18:50.40 | manxpower | and relaxdtmf should be ignored when in rfc2833 mode |
| 18:51.20 | LemensTS | Ok thanks |
| 18:51.32 | manxpower | it only makes sense for inband dtms |
| 18:51.39 | manxpower | and dtmf too! |
| 18:52.28 | deadpigeon | failed to read gains, there's probably a lot of reasons why I'd get this error right? |
| 18:56.51 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 18:56.59 | manxpower | Has anyone seen Broadvoice send back a 100 Trying followed by a 403 Forbidden messages? |
| 18:57.42 | *** join/#asterisk maddog01 (n=minotaur@dhcp-0-1c-f0-eb-94-24.cpe.quickclic.net) |
| 18:59.44 | *** join/#asterisk pfn (n=pfnguyen@hanhuy.com) |
| 19:02.27 | eppigy | TRABAJO |
| 19:07.05 | *** join/#asterisk stevetotaro (n=Steve@pool-71-254-231-87.hrbgpa.east.verizon.net) |
| 19:09.41 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 19:13.21 | _ShrikE | FRIJOLES |
| 19:13.37 | [TK]D-Fender | FRITOS |
| 19:13.42 | *** join/#asterisk UQlev (n=kvirc@91.184.220.73) |
| 19:16.18 | watchy | i want mexican for lunch |
| 19:16.18 | watchy | y |
| 19:16.42 | *** join/#asterisk chi6IT41 (n=chigital@tmo-100-55.customers.d1-online.com) |
| 19:21.44 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 19:21.44 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 19:22.02 | lmadsen | anyone know how to determine if there is an echo canceler board on a digium card from the linux console? |
| 19:22.21 | watchy | is there anyway to check zap presence? |
| 19:28.11 | [TK]D-Fender | watchy: Open you case. Do you see a card? |
| 19:28.26 | Juggie | lmadsen, there should be some way to probe the card |
| 19:28.27 | *** join/#asterisk icebrew54 (i=proxy@static-71-117-242-28.ptldor.dsl-w.verizon.net) |
| 19:28.32 | Juggie | does dahdi_cfg not say anything? |
| 19:28.53 | Juggie | try, dahdi_hardware |
| 19:29.08 | *** join/#asterisk anonymouz666 (n=anonymou@201.19.174.121) |
| 19:29.31 | *** join/#asterisk icebrew54 (i=proxy@static-71-117-242-28.ptldor.dsl-w.verizon.net) |
| 19:30.23 | anonymouz666 | anyone in here using the cdr_adaptive_odbc? I added a field, everything works fine, except when there's this field the calldate is totally empty (0000-00-00). When the field is NULL, the calldate get OK. |
| 19:30.43 | anonymouz666 | this.is.a.mistery. |
| 19:33.22 | lmadsen | Juggie: actually I got the info I needed with 'dmesg | grep VPM' |
| 19:33.24 | lmadsen | thanks though! |
| 19:33.30 | *** join/#asterisk Khratos (n=khratos@190.166.103.146) |
| 19:33.49 | Juggie | ah yes, it is detected in dmesg |
| 19:33.50 | Juggie | i forgot |
| 19:35.07 | eppigy | dmesg | kill your family |
| 19:35.16 | icebrew54 | s2067558180 |
| 19:35.31 | icebrew54 | kitty! |
| 19:36.44 | hardwire | kitteh |
| 19:40.52 | *** join/#asterisk keebler (i=9446c2d5@gateway/web/ajax/mibbit.com/x-bde524e9d1bfb545) |
| 19:41.10 | keebler | Has anyone here taken the Cisco IP Telephony Certification? |
| 19:41.15 | *** join/#asterisk maddog01 (n=minotaur@dhcp-0-1c-f0-eb-94-24.cpe.quickclic.net) |
| 19:41.34 | keebler | CIPT? |
| 19:44.13 | lowtek | For what? |
| 19:47.30 | keebler | Well, for the CCIP/CCVP certifications to implement a complete IP/VOIP telephony solution? Or just to have it? I've got authorization and funds to take whatever certifications I want. |
| 19:47.39 | keebler | And can't decide. |
| 19:47.44 | keebler | CCNA for a start, but then what? |
| 19:47.53 | keebler | already have some MCPs, and A+/ |
| 19:48.21 | keebler | Never taking a Windows/Microsoft centric certification program again. |
| 19:48.24 | keebler | Completely useless. |
| 19:49.28 | [TK]D-Fender | keebler: You never expect an employer to care? |
| 19:49.44 | lowtek | LOL, at one point I had 32 MS certs. They had become so easy I would just go take them during lunch. |
| 19:49.46 | [TK]D-Fender | keebler: And CCVP = $$$ |
| 19:50.01 | watchy | hmm |
| 19:50.01 | lowtek | The only one that's actually made me money is my CISSP. |
| 19:50.13 | watchy | can i go from the smartjack to a * box with a quad T1 card |
| 19:50.24 | watchy | then from the quad T1 card to a portmaster dialup server? |
| 19:50.42 | keebler | [TK]D-Fender: Well, I've got my MGIB that I haven't even touched, AND my employer will pay for anything. |
| 19:50.55 | [TK]D-Fender | watchy: Depends what this "portmaster dialup server" talks |
| 19:51.14 | watchy | it has a t1 port |
| 19:51.24 | *** join/#asterisk jasonwoot (n=some@bookit-dev.com) |
| 19:51.24 | [TK]D-Fender | keebler: Certs are for your empoyer, its for your PROSPECTIVE employers |
| 19:51.27 | keebler | [TK]D-Fender: I control the IT department, and we can just write it off as business expenses. |
| 19:51.45 | jasonwoot | Is there a zap equivalent for "sip show inuse"? |
| 19:51.46 | keebler | [TK]D-Fender: Well, they want me to get more training. |
| 19:51.49 | [TK]D-Fender | watchy: Well yes * can talk T1 to it... thats what you bought a card for! |
| 19:52.14 | watchy | tk: I have a full t1, i want to give say 11 channels to * and 12 to the portmaster |
| 19:52.27 | watchy | i can reroute 12 of them through the linux box from the quad t1 to the dialup? |
| 19:52.34 | [TK]D-Fender | watchy: OK/fine/sure, go for it |
| 19:52.40 | watchy | i'm adking |
| 19:52.42 | watchy | asking. |
| 19:52.51 | watchy | is it possible? |
| 19:52.55 | [TK]D-Fender | watchy: OK/fine/sure, go for it <-------- |
| 19:52.58 | *** join/#asterisk erth64net (n=erth64ne@69-30-67-191.dq1sn.easystreet.com) |
| 19:53.09 | watchy | ah. thanks. now to figure it out |
| 19:53.19 | [TK]D-Fender | watchy: You're just taking in channels and calling out to others. How you want to do this is up to you |
| 19:53.45 | keebler | The more I work with Asterisk and get into more complicated networking, the more I think I want to expand and get more involved in the career field. |
| 19:53.49 | watchy | i didnt know if i could make the channels pass correctly and the dialup box still recieve dialtone |
| 19:54.14 | *** join/#asterisk telnettech (n=telnette@d192-24-95-65.col.wideopenwest.com) |
| 19:54.23 | [TK]D-Fender | watchy: Yes it gets tone as appropriate. Dialplan applies here you know, there no "patching" the channels... |
| 19:54.43 | telnettech | can someone tell me where the Elastix gui puts the backup in what directory? |
| 19:54.48 | watchy | ah, so * does need to run on the box to move them over? |
| 19:54.57 | [TK]D-Fender | watchy: Yup |
| 19:55.12 | [TK]D-Fender | watchy: You will in fact be Dial-ing from one to the other |
| 19:55.16 | watchy | i have a 23 chan t1 that is wasted by like 12+ channels for our dialup server |
| 19:55.26 | watchy | oh i see |
| 19:55.30 | [TK]D-Fender | watchy: think of it as no different that * just to a phone... because thats all it is.. another call. |
| 19:55.33 | watchy | when a call comes in * would just forward it to the 2nd port |
| 19:55.38 | [TK]D-Fender | watchy: All calls are just calls... dialplan and all. |
| 19:55.41 | watchy | and the portmaster would answer |
| 19:55.53 | [TK]D-Fender | watchy: In on channel 1 on port 1? out on channel 1 on port 2, etc |
| 19:55.57 | [TK]D-Fender | (figuratively) |
| 19:56.03 | watchy | thanks man. yea thats awesome |
| 19:57.06 | [TK]D-Fender | watchy: You could even be crafty and get a PRI on inbound and spit it out CAS, etc. |
| 19:57.17 | watchy | we have 4 dialup user accounts and decidate a full t to the server for some reason |
| 19:57.26 | [TK]D-Fender | watchy: And be extra sneaky in use of DID> port mapping, providing Busy, etc |
| 19:57.32 | *** join/#asterisk erth64net (n=erth64ne@69-30-67-191.dq1sn.easystreet.com) |
| 19:57.35 | *** join/#asterisk garo (n=garo-irc@d54C3F35B.access.telenet.be) |
| 19:58.26 | jasonwoot | Is there a zap equivalent for "sip show inuse"? |
| 19:58.27 | garo | Can someone call the SIP garo@84.195.243.91 ? (i want to test if it works) |
| 19:58.30 | manxpower | watchy: seems like it would be cheaper to set the 4 people up in a national dialup provider and tell them they have a new phone number. |
| 19:58.43 | [TK]D-Fender | watchy: Most important revelation about * : Every call is jsut a &^$ing call :) |
| 19:58.52 | rwaite | question for anyone who might know: couldnt you get around issues with nat by having the client (behind the nat) send a packet thru a udp port to the server and the packet contains the port number to reply to - the server then replies and the nat router keeps this connection open |
| 19:59.10 | rwaite | this is probably already done, unless there is something i am missing that would prevent it from working |
| 19:59.34 | manxpower | rwaite: the port is changed by the NAT router. |
| 19:59.45 | garo | Or even better, does someone know a website that you can use to call sip numbers ? (so that i don't have to ask you) |
| 20:00.04 | manxpower | i.e. client src port 1234 -> nat router src port 23846 |
| 20:00.06 | rwaite | what if you sent the server a packet saying "quick, try to connect to me on every port right now!" |
| 20:00.22 | rwaite | and the one that gets thru, then you know what port to use? |
| 20:00.23 | manxpower | rwaite: that is called "oh look! i crashed my router!" |
| 20:00.57 | rwaite | manxpower: hmm, but would what im saying even be required? would the server know to use the 23846 port? |
| 20:01.04 | manxpower | rwaite: your idea would work for a few routers if it was the first connection trying to use that port |
| 20:01.26 | manxpower | rwaite: also you are talking about a SIP client we already have support for that |
| 20:01.29 | rwaite | hmm. nat is stupid, i've decided |
| 20:01.43 | rwaite | not sip, i'm actually trying to setup a vpn |
| 20:01.49 | *** join/#asterisk maddog01 (n=minotaur@dhcp-0-1c-f0-eb-94-24.cpe.quickclic.net) |
| 20:01.50 | manxpower | rwaite: no SIP is stupid. It breaks several best practices |
| 20:02.02 | rwaite | i just thought it was a good idea, but i did not know about the port changing |
| 20:02.13 | manxpower | rwaite: you should have no issues with a VPN |
| 20:02.15 | rwaite | i guess thats what lets multiple clients behind nat use the same port |
| 20:02.16 | *** join/#asterisk d00gster (n=doughant@bas1-cooksville01-1279552227.dsl.bell.ca) |
| 20:02.21 | *** join/#asterisk killown (n=Yamato@unaffiliated/killown) |
| 20:02.54 | rwaite | openvpn is nice, i have not got it working yet but when i do i plan on testing sip over it |
| 20:03.03 | rwaite | i hope it doesnt degrade the quality too much |
| 20:04.16 | rwaite | did you know with vmware workstation you can setup virtual networks with virtual segments, etc, and limit their bandwidth and reliability for testing purposes? |
| 20:04.44 | lowtek | SIP works fine over ipsec tunnels, but there is overhead related to the encryption. My recommendation is to use the lowest security settings for the tunnel less encryption time = lower latency. |
| 20:05.22 | rwaite | i see. i am probably going to have to use g729 or gsm, too, i dont think ulaw would work very well |
| 20:07.06 | telnettech | anybody know what directory Elastix gui saves the backup to? |
| 20:07.07 | lowtek | g729 rocks for shitty connections, imho |
| 20:07.47 | *** join/#asterisk erth64net (n=erth64ne@69-30-67-191.dq1sn.easystreet.com) |
| 20:07.49 | rwaite | yeah but i hate how it sounds |
| 20:09.30 | lowtek | gsm doesn't sound much better, best compromise would ge g726 |
| 20:09.42 | lowtek | s/ge/be |
| 20:09.43 | lowtek | imho |
| 20:10.01 | lowtek | Or ilbc, we had great results with that point to point with asterisk 1.2 |
| 20:12.13 | *** join/#asterisk anonymouz666 (n=anonymou@201.19.174.121) |
| 20:12.57 | *** join/#asterisk _aymeric_ (n=aymeric@157.159.47.43) |
| 20:14.01 | *** join/#asterisk killown (n=Yamato@unaffiliated/killown) |
| 20:14.44 | *** join/#asterisk troy- (n=troy@worldnet.tauri.ca) |
| 20:17.28 | *** join/#asterisk `paul (n=kutimoy@119.93.45.181) |
| 20:19.12 | jjshoe | anyone have comments/remarks on asterisk 1.6 and app_flite/ |
| 20:19.26 | jjshoe | s/\//\?/ |
| 20:20.12 | *** join/#asterisk CunningPike (n=arodgers@204.239.10.119) |
| 20:20.54 | jjshoe | this would be about all I can find: http://www.darrensessions.com/app_flite-v06-module-for-asterisk-released/ |
| 20:20.59 | *** join/#asterisk rue_mohr (n=rue@24.207.122.10) |
| 20:21.01 | rue_mohr | ok, so even on my second attempt to install and configure oslec with dahdi, dahdi will not load into asterisk then its turned on in system.conf where to look to try to trace down why? |
| 20:22.54 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 20:23.14 | manxpower | rue_mohr: file a report on bugs.digium.com |
| 20:23.34 | rue_mohr | what shall I include? |
| 20:23.54 | *** join/#asterisk bmoraca (n=bmoraca@209.60.253.58) |
| 20:24.01 | `paul | for an agent in a queue for example... can i have a page that has a "click to show call details" link then view the info (callerid + dialed extension) when his phone rings? |
| 20:24.07 | `paul | is this possible? |
| 20:24.55 | bmoraca | `paul: not with Asterisk alone. however, it's conceivable that one could greate an application that would accomplish that |
| 20:25.16 | `paul | with manager api is it possible? |
| 20:25.17 | rue_mohr | there is a way to modify the incomming calolerid data |
| 20:28.58 | bmoraca | `paul: yes, you would use the manager API as well as various other things. potentially, you could use a database application, etc, and have the dialplan put the appropriate information in it. there are a number of ways it could be done. |
| 20:29.21 | bmoraca | rue_mohr: exten=>whatever,n,Set(CALLERID(name)=whatever) |
| 20:29.45 | rue_mohr | hmm, its an answer but I never asked the question |
| 20:30.11 | bmoraca | rue_mohr: yeah...i see that now |
| 20:30.22 | bmoraca | rue_mohr: read it as "is there a way" rather than "there is a way" |
| 20:30.26 | anonymouz666 | Corydon76-dig: could you please me help me to figure out why cdr_adaptive_odbc always insert calldate (field) value as 0000-00-00 00:00:00 (default datetime field value)? |
| 20:30.37 | Gido-E | bmoraca: yes |
| 20:30.57 | Corydon76-dig | anonymouz666: did you set up an alias? |
| 20:31.02 | rue_mohr | bmoraca, now if you could help me get oslec working, THAT would be great |
| 20:31.12 | anonymouz666 | there is no alias |
| 20:31.38 | anonymouz666 | alias calldate => start ? |
| 20:31.43 | bmoraca | rue_mohr: I already told you that spending the $300 for HWEC is cheaper than paying you to spend a week doing this foolishness. i've only ever used HWEC. |
| 20:31.47 | Corydon76-dig | other way around |
| 20:32.51 | *** join/#asterisk dlewis (i=c7340d67@about/security/staff/dlewis) |
| 20:33.17 | Corydon76-dig | alias start => calldate |
| 20:34.47 | anonymouz666 | Corydon76-dig: wonderful... it works. |
| 20:37.08 | Micc | is g726-32 a high quality codec? |
| 20:42.25 | lowtek | It's pretty close to the quality of g711, so close your users likely wouldn't notice. |
| 20:42.58 | lowtek | But it is compressed so faxing is out. |
| 20:43.07 | lowtek | rather, *reliable* faxing |
| 20:43.26 | bmoraca | reliable faxing doesn't work well on g711 either |
| 20:44.33 | lowtek | g711 is what the PSTN uses in most cases, faxing should work fine or you have another problem like a sh*tty network. |
| 20:47.08 | bmoraca | the internet is a generally shitty network, yes. never had a problem faxing locally, though. |
| 20:47.59 | lowtek | Faxing over the internet isn't that good of an idea, lol. I was pointing out that it's not the codec's fault if you're having problems faxing. |
| 20:48.29 | bmoraca | i'm just pointing out that faxing over sip sucks any way you look at it :) |
| 20:48.42 | `paul | bmoraca: you have any idea on what action to use in manager api (regarding my question a while ago....) |
| 20:49.11 | lowtek | bmoraca: word |
| 20:49.27 | lowtek | paul: missed it, what's the question? |
| 20:49.37 | bmoraca | `paul: there's a whole host of options you would need to use to write something like you're looking at. you would need to monitor the messages, etc, as well as poll the queues. it's a very large undertaking |
| 20:52.25 | wonderworld | what would be the most userfriendly IAX softphone on linux? |
| 20:52.55 | lowtek | Linux is user friendly? |
| 20:53.47 | wonderworld | sure |
| 20:53.50 | bmoraca | `paul: I briefly flirted with the idea of building an operator panel from scratch for my customers, but i'm just one person and had nowhere near the time to appropriately develop it. there are some proprietary apps that will do screenpops and what not based on inbound caller id. you might try one of them out...Snap-a-number is one, and iSymphony is another...both work well in their own way. |
| 20:57.28 | jjshoe | `paul you would need to do a show channels, and show channel <channel> |
| 20:57.33 | *** join/#asterisk savageone (n=savageon@static-66-212-194-134.cpe.metrocast.net) |
| 20:57.36 | savageone | howdie folks! |
| 20:57.42 | Micc | bmoraca, what do you use with your customers? |
| 20:57.43 | [TK]D-Fender | BBIAB |
| 20:57.57 | savageone | anyone working w/ video phones? I know how the software works via asterisk but what do people think about decent models? |
| 20:57.59 | Micc | bmoraca, I'm in the same boat at the moment, contemplating building one myself or using something else. |
| 20:58.00 | savageone | I'm talking home use |
| 20:58.04 | bmoraca | `paul: in fact, using a small coldfusion script and snap-a-number i was able to screenpop client account pages from Microsoft Dynamics CRM |
| 20:58.23 | bmoraca | Micc: for the customers that have wanted to pay for it, I use iSymphony |
| 20:58.42 | Micc | bmoraca, how much is it? |
| 20:59.06 | *** join/#asterisk becks` (n=sdfgsfdg@169-244.104-92.cust.bluewin.ch) |
| 20:59.26 | becks` | hi, what can i do if I need a special character (+) in my user name? ( http://bugs.digium.com/bug_view_advanced_page.php?bug_id=14025 ) |
| 21:00.56 | `paul | is eventwhencalled set in queues.conf???? |
| 21:02.47 | *** join/#asterisk russellb (n=russell@asterisk/digium-open-source-team-lead/russellb) |
| 21:02.47 | *** mode/#asterisk [+o russellb] by ChanServ |
| 21:04.45 | Micc | bmoraca, so I have to buy the server for 600$ before I can have the 19$ client connect to it? |
| 21:05.08 | bmoraca | Micc: yes |
| 21:05.59 | bmoraca | Micc: like I said, it's not cheap, but it does work. the new version even has queue monitoring. |
| 21:06.44 | *** join/#asterisk asheron (n=stephan@95-36-36-255.dsl.alice.nl) |
| 21:06.44 | bmoraca | Micc: you can get a 5 client license for free, though. doesn't have the built-in instant messaging (which is largely worthless anyway)...but it works OK. |
| 21:08.12 | Micc | bmoraca, how much do you charge your customers for it? I'm thinking I could charge 5$ a month or somewhere around there. |
| 21:08.20 | keebler | bmoraca: Ordered the EZGO last night. |
| 21:08.27 | keebler | Wish I didn't haha. |
| 21:08.56 | Micc | bmoraca, do you use anything for customer management? |
| 21:09.41 | *** join/#asterisk Thiago_Lima (n=x@mail01.corp.cma.com.br) |
| 21:10.46 | lowtek | The IronVoice portal is dabombdigitty. |
| 21:11.04 | Micc | lowtek, where can I get that? |
| 21:11.13 | lowtek | http://ironvoice.com |
| 21:11.33 | Thiago_Lima | Hi All.... Any idea what may be happening in these log errors? http://pastebin.com/m57bc33f5 |
| 21:11.57 | Micc | lowtek, that looks like their own proprietary portal. do they let other companies use it? |
| 21:12.39 | bmoraca | Micc: customer management as in what? |
| 21:12.58 | bmoraca | keebler: no good? |
| 21:13.18 | keebler | bmoraca: Well, I really just don't trust the name "EZGO" to be reliable. |
| 21:13.25 | keebler | bmoraca: And I can't find any reviews on them. |
| 21:13.38 | keebler | bmoraca: But they get here tomorrow, so we'll see. |
| 21:13.54 | bmoraca | Micc: I charge $750 for the server and $55 per client. I don't encourage it except for receptionist use, so I try to minimize its use. most people are too stupid to realize its use anyway. |
| 21:14.03 | bmoraca | keebler: nice. let me know. |
| 21:14.05 | keebler | I think I'm going to go with MikroTik though. Cost effective, and pretty robust package. |
| 21:14.26 | keebler | $49 for router board, $49 for Wifi card, $25 for ant. |
| 21:14.45 | bmoraca | Microtik is a software-based solution...similar to PFSense...i don't know that i'd trust it. |
| 21:15.04 | keebler | bmoraca: Well, its debain with a custom GUI. |
| 21:15.10 | keebler | bmoraca: on Routerboard hardware. |
| 21:15.14 | bmoraca | keebler: exactly. |
| 21:15.51 | keebler | bmoraca: Well, can't be any worse than my WRT54Gs. |
| 21:15.54 | Micc | bmoraca, It sounds like your setting it up in conjunction with customer's asterisk server, is that right? |
| 21:16.12 | Micc | bmoraca, I'm providing virtual hosted pbx. Maybe it won't help me there. |
| 21:16.13 | bmoraca | Micc: yes. it's made for use with Asterisk. |
| 21:16.42 | Micc | bmoraca, I have a lot of customers on one server. |
| 21:18.15 | jasonwoot | Frig.... does the linksys 941 not do backlighting? |
| 21:18.20 | jasonwoot | is it just the 942? |
| 21:18.30 | lanning | yes |
| 21:19.20 | *** join/#asterisk WHYS (i=lpfm@137.28.94.209) |
| 21:19.33 | bmoraca | Micc: sounds like a clusterfuck. how do you manage multiple parking lots? i know that isymphony cannot do that. |
| 21:19.47 | lowtek | Asterisk can't do that afaik |
| 21:20.04 | lowtek | Althogh I've seen it faked with meetme |
| 21:20.06 | *** join/#asterisk killown (n=Yamato@unaffiliated/killown) |
| 21:20.11 | lowtek | s/althogh/although |
| 21:20.16 | bmoraca | yeah |
| 21:20.56 | manxpower | You should be able to hack togather multi-parking as well as multi-meetme, etc |
| 21:21.05 | bmoraca | anyone know if using the clock=pit boot option negatively impacts asterisk? |
| 21:21.30 | Micc | bmoraca, I haven't needed more parking lots yet. But I hear asterisk 1.6 can handle multiple parking contexts. |
| 21:22.02 | Micc | Most customers already have an old NEC pbx. |
| 21:22.14 | Micc | We come in with an spa8000 and replace their lines. |
| 21:22.22 | bmoraca | Micc: you let all customers share the same parking lot? sounds like potential for mischief! |
| 21:22.51 | Micc | bmoraca, no. |
| 21:22.52 | bmoraca | Micc: i see. you're just trunking then. that'd work fine. it would also preclude you from using iSymphony for anything useful. |
| 21:23.18 | Micc | bmoraca, we have one customer with 7 aastra phones. |
| 21:23.23 | Micc | they use the parking lot. |
| 21:23.31 | bmoraca | ahhh |
| 21:23.40 | Micc | As soon as we get another customer that needs it we'll get another server with 1.6. |
| 21:23.44 | *** join/#asterisk eneusta1 (n=eldon@static-139-142-157-129.gtcust.grouptelecom.net) |
| 21:23.50 | Micc | or hack it to do multiple parking lots in 1.4.22 |
| 21:24.07 | bmoraca | Micc: from what i've experienced, Asterisk runs really, really well in ESXi |
| 21:24.34 | Micc | bmoraca, its running well in whatever it is from ubiquity right now. |
| 21:24.49 | Micc | bmoraca, $50 a month for an asterisk server isn't bad. |
| 21:25.05 | *** part/#asterisk eneusta1 (n=eldon@static-139-142-157-129.gtcust.grouptelecom.net) |
| 21:25.15 | bmoraca | i already have three cabinets in a colocation...so it's effectively free. |
| 21:25.16 | Micc | bmoraca, we have one of our own servers with a PRI for local calls. |
| 21:25.27 | dlewis | bmoraca: yea, my friend runs his entire environment on ESXi... |
| 21:26.01 | Micc | bmoraca, I'll certainly look into ESXi when we move to our own datacenter. |
| 21:26.46 | bmoraca | i have a PRI in the datacenter as well for inbound/onramp. i'll be looking at sip providers as well, most likely globalpops as we already use them for other things, once our capacity ramps up...right now, my first hosted customer goes live on friday |
| 21:28.09 | Thiago_Lima | Hi All.... Any idea what may be happening in these log errors? http://pastebin.com/m57bc33f5 |
| 21:28.31 | bmoraca | Micc: from the testing i've done, a DL380 G3 with 10gb RAM and 15K drives will support roughly 10 virtual systems. total cost of the box? about $600 :P |
| 21:29.38 | Micc | bmoraca, of each box you mean? |
| 21:30.08 | bmoraca | no...$600 for the single DL380 that supports 10 virtual systems |
| 21:30.11 | *** join/#asterisk erth64net (n=erth64ne@69-30-67-191.dq1sn.easystreet.com) |
| 21:31.00 | *** join/#asterisk voxter (n=voxter@76.77.95.2) |
| 21:31.21 | bmoraca | can't virtualize zaptel hardware, though, so each of the virtual systems trunk out via SIP to a second box which runs linux natively and has my PSTN connectivity in it. it'll also take care of my sip trunking as well. everything will pass through this gateway to get to the virtual systems. |
| 21:32.10 | bmoraca | works extremely well...and at $0.0019 per minute for tier 1 origination, who can argue? |
| 21:32.29 | bmoraca | inbound costs me nothing. |
| 21:33.00 | savageone | anyone here working with affinity voip? |
| 21:37.48 | wonderworld | bmoraca: how do your clients connect? SIP or AIX? |
| 21:39.47 | bmoraca | SIP |
| 21:40.13 | wonderworld | didn't you have trouble on their side with NAT/Routers, etc.? |
| 21:40.57 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 21:41.38 | bmoraca | like i said, my first customer goes live on friday. however, there are many routers that do support sip passthru. the onyl one of note that i've found that does not is the NetGear Rangemax v3. The v2 does support it, but the v3 does not. Most other routers do support it enough to handle 5 or so extensions...which is who my solution is aimed at |
| 21:42.34 | bmoraca | i've had my inhouse service department running off of one for a while and had no problems...then again, we use a pix 515e |
| 21:43.28 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.218.194) |
| 21:44.03 | Micc | bmoraca, where are you located? |
| 21:44.14 | *** join/#asterisk killown (n=Yamato@unaffiliated/killown) |
| 21:44.33 | bmoraca | central california |
| 21:45.25 | Micc | bmoraca, you having any issues with porting? |
| 21:45.41 | Micc | bmoraca, or cease and desist letters from the local ilec? |
| 21:45.52 | bmoraca | haven't yet. we'll find out, though. |
| 21:46.09 | bmoraca | numbers on friday are porting from AT&T and so far everything's been fine |
| 21:46.19 | Micc | bmoraca, isn't ESXi expensive? |
| 21:46.25 | bmoraca | ESXi is free |
| 21:46.44 | Micc | oh, nice! hmmm I think I need to do that DL380 G3 deal. |
| 21:46.51 | bmoraca | Micc: ebay |
| 21:46.52 | lowtek | I can't believe I'm about to say this, but AT&T does a pretty good job with their LNP process. |
| 21:46.54 | Micc | Where do you get 15K rpm drives? |
| 21:47.08 | lowtek | Cincinati Bell however will have you contemplating suicide ... |
| 21:47.16 | bmoraca | Micc: the systems i bought from ebay came with 6 72gb 15K drives each |
| 21:47.46 | bmoraca | lowtek: lol. thankfully, my service provider takes care of that...i just have to get the LOA signed and held on file. |
| 21:47.46 | Micc | bmoraca, nice. I've got a bunch of old 20gb scsci drives in the garage. |
| 21:48.20 | *** join/#asterisk bn43 (n=dhashen@41.27.186.108) |
| 21:48.46 | *** join/#asterisk putnopvut (n=putnopvu@nat/digium/x-0b8c6babe4c70d75) |
| 21:48.46 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 21:48.52 | bmoraca | Micc: i use these things all over the place...i've probably got 15 of them in service right now running webservers and phone servers, etc, off them. they're great little boxes. and iLO is frickin wonderful when they're not hosted locally |
| 21:49.16 | Micc | iLO? |
| 21:49.23 | bmoraca | integrated lights out |
| 21:49.32 | bmoraca | out-of-band management for HP servers |
| 21:50.03 | bmoraca | server can be completely powered off and i can hit the "virtual power button" and turn it on remotely |
| 21:50.25 | bmoraca | if you want to pay for it, there's also an extended license that gives you a VNC-based console |
| 21:50.27 | Micc | wow. Thats nice. |
| 21:50.54 | bmoraca | Micc: 40 minute drive from here to my cabinet...it's a life saver |
| 21:51.39 | Micc | Thats cheaper then spending $50 a month for a vps. |
| 21:52.03 | Micc | I can spend 150$ a month to colocate a 2U and get 10 vps. |
| 21:52.17 | bn43 | hi I got help on vmail notification on my snom phones - whereas previously I was using one extension for mailbox 4242, in my extension I have a different number for each user, however sip.conf shows vmexten=4242 |
| 21:52.49 | bmoraca | Micc: nice. where do you get your pstn access from? sip? |
| 21:53.09 | bn43 | how do get vmexten option to route to each user's mailbox? |
| 21:53.25 | Micc | bmoraca, PRI from Integra and vitelity |
| 21:53.44 | Micc | bmoraca, I'm on the vitelity wholesale program. |
| 21:53.44 | bmoraca | Micc: hooked to your VPS? or does your VPS trunk out somewhere else? |
| 21:54.11 | Micc | bmoraca, I have one real box with the PRI card. |
| 21:54.17 | *** join/#asterisk stevetotaro (n=Steve@pool-71-254-231-87.hrbgpa.east.verizon.net) |
| 21:54.33 | Micc | bmoraca, so the vps trunks out to both vitelity and my other server. |
| 21:54.44 | bmoraca | gotcha |
| 21:54.53 | Micc | bmoraca, local calls go out the PRI, LD calls go out vitelity. |
| 21:55.11 | bmoraca | how are vitelity's rates? |
| 21:55.54 | bmoraca | hrm...$0.0144/min...not bad |
| 21:56.17 | *** join/#asterisk macros73_ (n=cs@67.163.224.69) |
| 21:56.30 | [TK]D-Fender | bn43: make 1 exten that ID's the Caller and passes the box # as appropriate |
| 21:56.54 | *** join/#asterisk defsdoor (n=andy@defsdoor.gotadsl.co.uk) |
| 21:57.29 | Micc | bmoraca, not bad. a little high on the LD rate I think at 1.2 cents a minute. I can't find my rate at the moment, it might be 1.5 |
| 21:58.46 | bmoraca | Micc: that's honestly not bad. my concern is the $8/mo fee per DID...lol |
| 21:59.21 | Micc | bmoraca, I don't pay that. |
| 21:59.28 | Micc | bmoraca, I pay for channels. |
| 21:59.32 | bmoraca | yeah |
| 21:59.39 | Micc | put all my did's on the channels. |
| 22:00.17 | bmoraca | i have unlimited inbound through my PRI, so I'd keep everything there. E911 is also only $0.35/mo |
| 22:00.21 | Micc | Its a pretty lucrative business if you can keep your other expenses down. |
| 22:01.05 | Micc | bmoraca, I pay 1$ a month for e911 and they charge me for cnam lookups. |
| 22:01.06 | *** join/#asterisk path_ (n=path@pc-15-190-86-200.cm.vtr.net) |
| 22:01.12 | bmoraca | my only expense right now is $310/mo for a PRI. two customers will more than cover that at what I aim to charge. |
| 22:02.30 | Micc | bmoraca, I'd rather vitelity handle all the porting and give me the channels. It ends up being cheaper than having my own PRI for inbound. |
| 22:03.08 | dlewis | Micc: how's your experience with vitelity? |
| 22:03.10 | Micc | My customers do about 80% local and 20% LD. |
| 22:03.36 | Micc | dlewis, don't expect to get anything done by talking to someone. If you use their portal it gets done. |
| 22:03.50 | Micc | dlewis, anything else and you'll just be waiting and making a lot of calls to find out status. |
| 22:03.57 | dlewis | true |
| 22:04.02 | dlewis | how's the service otherwise? |
| 22:04.12 | dlewis | i was looking at them |
| 22:04.19 | dlewis | flowroute is the frontrunner in my list right now. |
| 22:04.30 | Micc | dlewis, so far so good. Had a few hickups early on with registartions. Now I'm using IP authentication. |
| 22:04.43 | dlewis | ok |
| 22:05.12 | Micc | dlewis, their portal works pretty well and they are quick to respond to trouble tickets. |
| 22:05.52 | dlewis | Micc: ok |
| 22:05.58 | bmoraca | Micc: my primary customer is dental offices...i'd bet they're 80% inbound, 20% outbound as far as service goes |
| 22:06.10 | Micc | dlewis, the number porting was really nice too. You just upload the signature in jpg or gif. |
| 22:06.41 | Micc | bmoraca, yeah I bet mine are too, although I haven't broken down the amount of inbound vs outbout. |
| 22:07.00 | jjshoe | bmoraca sell them an agi that calls to confirm the next days appointments, then you're talking. |
| 22:07.00 | *** join/#asterisk stevetotaro (n=Steve@pool-71-254-231-87.hrbgpa.east.verizon.net) |
| 22:07.19 | Micc | bmoraca, my average customer has 5 lines and does about 3,000 minutes of dialout. |
| 22:07.36 | bmoraca | jjshoe: i'm working on it...my hangup is the practice management software doesn't have a way to export the appointment information |
| 22:07.43 | bmoraca | Micc: how much do you charge per trunk line? |
| 22:08.09 | Micc | bmoraca, we've been going round and round with our pricing. |
| 22:09.06 | bmoraca | lol |
| 22:09.12 | Micc | bmoraca, its a little complicated, but I think we're charging around 35$ for business line with pbx functions. and $20 for residential with no ld. 25 residential with 500 minutes of LD or 1k, I forget. |
| 22:09.41 | Micc | bmoraca, I think the 35$ business line has 500 minutes per line and a 40 or 45$ line has 1500 minutes. |
| 22:10.05 | Micc | bmoraca, we debated doing unlimited LD, but starting out small one big LD caller could tank you. |
| 22:10.25 | Micc | bmoraca, we just explain to our customers that there is no such thing as unlimited long distance. |
| 22:10.35 | jjshoe | bmoraca there's always a way :) |
| 22:10.49 | Micc | bmoraca, I did get the magic jack working. |
| 22:11.35 | bmoraca | Micc: we're looking at doing a full solution including leasing phones...we're looking at $40/mo per line with 1000 min per line and billing various per month based on the model of phone they want ($20/mo for a Cisco 7940 for instance) |
| 22:11.35 | Micc | bmoraca, but I don't want to let businesses use the magic jack. I'm not sure when they'll figure it out and kick us off. |
| 22:11.35 | bmoraca | jjshoe: no, not with Dentrix. :P |
| 22:11.36 | Micc | bmoraca, yeah that sounds just like us. |
| 22:11.40 | bmoraca | Micc: magic jack? |
| 22:11.57 | Micc | bmoraca, yeah haven't you seen the infomercials late at night? |
| 22:12.10 | Micc | bmoraca, a little usb device that does unlimited long distance for a year for $20. |
| 22:12.10 | jjshoe | bmoraca if I had whatever that was, I would find a way :) |
| 22:12.16 | bmoraca | no...although i do like the sham-wow guy... |
| 22:12.22 | Micc | haha |
| 22:12.35 | Micc | I tried to make a deal with magic jack. They don't even speak the same language. |
| 22:12.47 | bmoraca | Micc: hindu or something? :P |
| 22:12.53 | bmoraca | bad joke, i know |
| 22:12.59 | Micc | bmoraca, non-telecom-ease. |
| 22:13.17 | dlewis | bmoraca: you're thinking leasing phones would hold? |
| 22:13.17 | bmoraca | Micc: i'd love to find a USB FXS adapter that worked with Windows... |
| 22:13.28 | Micc | bmoraca, magic jack works with mac and windows. |
| 22:13.41 | Micc | bmoraca, I think they do advertising in their software though. |
| 22:13.54 | dlewis | Micc: in there softphone, yes |
| 22:14.01 | Micc | bmoraca, I told them I had it working with asterisk and they said, no it doesn't work with asterisk. |
| 22:14.05 | bmoraca | dlewis: small offices HATE capital expenses. they'll pay a higher monthly fee to avoid paying for something outright. |
| 22:14.21 | jjshoe | bmoraca apparently there's an open dental |
| 22:14.37 | dlewis | you gotta love the ignorant. |
| 22:15.07 | Micc | I gotta get back to my day job. |
| 22:15.11 | bn43 | sorry I am new to asterisk and did not understand <[TK]D-Fender>'s answer - I'm not getting how to make vmexten=voicemail route to each users mailbox upon pressing vmail on the phone |
| 22:15.13 | bmoraca | jjshoe: we looked at opendental...(we're Dentrix integrators)...the problem with opendental is that Dentrix is produced by the largest dental supply company |
| 22:15.38 | Micc | bn43, your not alone. I don't think anyone understands TKD's answers. |
| 22:15.59 | WHYS | I am looking to test fax termination *gasp* using * 1.6. Any suggestions as to which way to go, or what to ask to find out what I need to know to make the determination. |
| 22:16.00 | bmoraca | dlewis: it's not that. it's the small business mentality. most won't be able to justify spending $4000 to replace their phone system. but they will pay $250/mo for service and a phone system. |
| 22:16.02 | bn43 | basic problem is that I don't fully understand how exten works - I'm getting there slowly tho |
| 22:16.09 | WHYS | AsterFax / T.38 / ??? |
| 22:17.02 | dlewis | true... |
| 22:17.16 | WHYS | Any favorites here, or things to avoid? |
| 22:17.33 | dlewis | I need to get into this business, lol... |
| 22:17.33 | bn43 | Micc: he's kind enuf to answer questions to us poor noobs tho! |
| 22:17.38 | bmoraca | WHYS: hylafax and iaxtty is all i've used. works well. |
| 22:17.41 | bn43 | :-) |
| 22:17.53 | bn43 | even tho I scratch my head most of the time |
| 22:17.54 | Micc | bn43, if you think mocking you is an answer. |
| 22:18.02 | bmoraca | dlewis: i figure i'll lose money until i get about 10 customers. but that's ok, because i'm not footing the bill :P |
| 22:18.07 | bn43 | ignorance is bliss |
| 22:18.10 | WHYS | Thanks, I'll take a look |
| 22:18.34 | bn43 | I didn't take it that way |
| 22:19.11 | Micc | bn43, maybe its just me he likes to screw with. |
| 22:19.24 | Micc | bn43, or maybe its the stupidity of my questions. |
| 22:19.59 | dlewis | bmoraca: you're using vitelity as well? |
| 22:20.01 | bmoraca | dlewis: also, as I said, I already have the infrastructure in place for it. we use our colocation cabinets for other things which more than pay for the cabinets. |
| 22:20.03 | bn43 | same boat |
| 22:20.35 | bmoraca | dlewis: no, i'd never heard of them until now. i will probably look in to them, though. their rates seem reasonable and for long distance they're a lot better than the datacenter |
| 22:20.42 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.218.194) |
| 22:22.05 | dlewis | cool |
| 22:22.19 | bn43 | [TK]D-Fender: hi-welcome back-I did not quite understand your answer, could you please expand? |
| 22:22.46 | [TK]D-Fender | bn43: read what I said VERY CAREFULLY |
| 22:26.35 | bmoraca | dlewis: we have more than enough data capacity, too...i get about 38Mbps both ways. |
| 22:27.01 | dlewis | nice nice |
| 22:27.28 | *** join/#asterisk ABom9 (n=adamirc@cpe-67-246-182-12.buffalo.res.rr.com) |
| 22:30.23 | *** join/#asterisk martyn-dev1 (n=admin@190.24.134.154) |
| 22:30.26 | martyn-dev1 | Hi,. |
| 22:33.41 | *** join/#asterisk stevetotaro (n=Steve@pool-71-254-231-87.hrbgpa.east.verizon.net) |
| 22:35.15 | *** join/#asterisk SparFux (n=raoul@e182024048.adsl.alicedsl.de) |
| 22:38.03 | martyn-dev1 | My asterisk crashed some minutes ago :( . The last message was: http://asterisk.pastebin.ca/1321151 :( some idea. |
| 22:38.23 | martyn-dev1 | Iwas run asterisk command and asterisk is runnning again. But i want to know why asterisk stoped :( |
| 22:38.35 | *** join/#asterisk dlewis (i=c7340d65@about/security/staff/dlewis) |
| 22:39.19 | *** join/#asterisk nix8n82 (n=nate@63.162.27.243) |
| 22:42.15 | eppigy | because you touch yourself at night |
| 22:43.34 | dlewis | wo |
| 22:43.42 | dlewis | got back in the channel at a bad time i guess |
| 22:43.54 | *** join/#asterisk satish2437 (n=root@122.167.115.41) |
| 22:44.09 | *** join/#asterisk gloin (i=me@unaffiliated/gloin) |
| 22:44.20 | rob0 | Crashterisk |
| 22:44.39 | satish2437 | Hello Everyone, I need help....I am getting HDLC ABORT event from Zaptel all the time....how to resolve it...?? |
| 22:45.46 | satish2437 | Hello Everyone, I need help....I am getting HDLC ABORT event from Zaptel all the time....how to resolve it...?? |
| 22:45.50 | *** join/#asterisk GvJordan (n=gvj@74-44-182-67.dr02.blfd.wv.frontiernet.net) |
| 22:46.04 | eppigy | Hello Everyone, I need help....I am getting HDLC ABORT event from Zaptel all the time....how to resolve it...?? |
| 22:46.27 | *** join/#asterisk trnzmeta (n=bleh@iinet.guard.com.au) |
| 22:46.51 | Mark17 | eppigy: do you have more information? |
| 22:46.53 | GvJordan | Just wonder how can i route calls to the real phone ntwork |
| 22:47.09 | eppigy | Mark17: yes |
| 22:47.16 | Mark17 | GvJordan: there are multiple options, you could use ISDN gateways |
| 22:47.17 | eppigy | what is your query? |
| 22:47.26 | GvJordan | mm |
| 22:47.31 | satish2437 | Please I need help on this.. |
| 22:48.02 | SparFux | notices satish2437 and epigy have the very same problem. |
| 22:48.34 | satish2437 | I need a solution please.. |
| 22:48.47 | Mark17 | satish2437 + eppigy: please provide some information about your configuration, maybe we can detect what is wrong, also mention what you did recently change (use a pastebin like yourpaste.net if possible, replace passwords with something like {PASS} or fake data and not with XXXX or xxxx please |
| 22:49.02 | eppigy | I was just mocking him |
| 22:49.12 | eppigy | by pasting it once again |
| 22:49.17 | eppigy | after he asked twice |
| 22:49.22 | Mark17 | SparFux: i was thinking that it was the same person (because of the same question, it was like copy/paste) |
| 22:49.23 | Mark17 | ah ok |
| 22:49.48 | SparFux | Mark17: LOL |
| 22:49.49 | Mark17 | GvJordan: you could also use an other SIP/AIX/etc. provider for doing it, for example we offer it to our clients |
| 22:50.02 | GvJordan | mm |
| 22:50.09 | rob0 | mixes a solution |
| 22:50.12 | GvJordan | My boss is wanting to set it up |
| 22:50.19 | GvJordan | to route calls for his business |
| 22:50.23 | satish2437 | I have a live SS7 link with signalling on channel number 1. I have configured channel 1 as DCHAN...when I try to read it...I get Exception 6 which is ZT_HDLC_ABORT |
| 22:50.46 | rob0 | needs a more powerful solvent |
| 22:50.55 | rob0 | beer! |
| 22:51.03 | eppigy | synthesize me something nice |
| 22:51.24 | satish2437 | It seems you guys are not at all in a mood to help.... |
| 22:51.42 | Mark17 | GvJordan: to where do most calls go? look for a VOIP provider in your area for the best performance (and most times also the best price) |
| 22:52.00 | Mark17 | satish2437: please provide more information, without more information we cannot solve it |
| 22:52.02 | GvJordan | mm |
| 22:52.15 | GvJordan | so he can't run this himself on asterisk? |
| 22:52.21 | satish2437 | Mark17: what else info do you need |
| 22:52.35 | rob0 | Actually Satish, you need to learn how best to use IRC. Prepare a pastebin with a complete problem description and debugging information, plus what you have tried, and what happened ... et c. |
| 22:52.49 | Mark17 | satish2437: for example did you change something recently would be a small, but still help full bit of information |
| 22:52.54 | SparFux | GvJordan: He can. It was said he could use ISDN Gateway or set it up. Like putting an ISDN card inside his computer and connect it to the ISDN phone network. |
| 22:53.09 | GvJordan | ah |
| 22:53.28 | SparFux | GvJordan: then with asterisk he could use chan_misdn, chan_lcr, chan_capi, or whatever ISDN channel drivers there are. Many options. |
| 22:53.49 | GvJordan | ah thanks |
| 22:54.01 | Mark17 | or use bluetooth and a mobile phone and chan_mobile (watch that not all bluetooth dongles/mobile phone are tested and not all tested dongles/mobile phones works, please see also http://www.voip-info.org/wiki/view/chan_mobile) |
| 22:54.09 | Mark17 | that are the most easy options |
| 22:54.14 | SparFux | GvJordan: A good thing I like it www.linux-call-router.de. That's what I am using, lcr and chan_lcr. |
| 22:54.42 | Mark17 | it depends on if getting ISDN is possible at a location you have with asterisk or where you can setup asterisk |
| 22:54.52 | GvJordan | well |
| 22:54.57 | GvJordan | he is trying to setup this pbx at a datacenter |
| 22:55.14 | satish2437 | Mark17: I have installed zaptel-1.4.12, I have not installed Asterisk as I am writing my own code to read the messages. I have a live SS7 link connected to span 1. The machine has a zaptel compatible T1/E1 card. Now, when I am trying to read on this span on the DCHAN...I am getting Zaptel Event: HDLC_ABORT. |
| 22:56.22 | Mark17 | GvJordan: not at all datacenters you can get ISDN at the location of your server(s) |
| 22:56.45 | *** join/#asterisk RoyK (n=roy@ip-140-26-149-91.dialup.ice.no) |
| 22:56.53 | Mark17 | in general: mobile phones works, but the solution has some limitations and will be less cost effective in most cases |
| 22:57.44 | RoyK | any idea if I can do PLC with g.729 or CFG with asterisk these days? |
| 22:58.03 | Mark17 | satish2437: please provide all logging that might be interesting with this case (use a pastebin like yourpaste.net) and provide your config (please note: replace actual passwords with fake data and dont use XXX or xxx, use a pastebin for this) |
| 22:58.10 | RoyK | .- ... - . .-. .. ... -.- / ... ..- -.-. ... |
| 22:58.15 | eppigy | YEAH |
| 22:58.25 | RoyK | .- ... - . .-. .. ... -.- / ... ..- -.-. -.- ... |
| 22:58.42 | eppigy | SMOKE PURP BY THE POUND |
| 22:59.14 | RoyK | not exactly |
| 22:59.34 | eppigy | your loss |
| 23:00.03 | RoyK | I'm the one that knows morse code - you're the other |
| 23:00.06 | *** join/#asterisk Khratos (n=Khratos@190.166.129.54) |
| 23:00.17 | eppigy | WHOOPTI DOO |
| 23:00.19 | Mark17 | is it possible to do the following (for outbound calls): try calling with outgoing-trunk out-1 and if that fails use outgoing-trunk out-2 and if that fails use outgoing-trunk out-3 etc. etc.? |
| 23:00.44 | RoyK | Mark17: it's all in the sample configs |
| 23:00.46 | eppigy | HEY GUYS LOOK AT MY MORSE CODE PLUGIN |
| 23:00.56 | eppigy | HOW DO YOU SAY VIRGIN IN MORSE |
| 23:01.07 | eppigy | ALLOW ME TO DEMONSTRATE |
| 23:01.10 | RoyK | fucking caps nazi |
| 23:01.16 | eppigy | calm down dog |
| 23:01.35 | RoyK | eppigy: if you do, bitch |
| 23:01.37 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 23:01.47 | eppigy | i dont want any trouble |
| 23:01.51 | *** join/#asterisk maddog01 (n=minotaur@dhcp-0-1e-58-0-b-f.cpe.mountaincable.net) |
| 23:01.51 | eppigy | no nee dfor profanity |
| 23:01.53 | beek | ...- .. .-. --. .. -. |
| 23:02.03 | outtolunc | starts humm'n 'love is in the air.. ' |
| 23:02.09 | eppigy | 8[] |
| 23:02.37 | beek | RoyK: Asterisk does not suck. |
| 23:03.10 | WHYS | where is the troll hunder? |
| 23:03.49 | RoyK | beek: well, it does |
| 23:03.50 | eppigy | ALL THE HUSTLAS THEY GON RIDE TO THIS |
| 23:04.53 | *** part/#asterisk martyn-dev1 (n=admin@190.24.134.154) |
| 23:04.59 | RoyK | alle the caps people should be slaughtered |
| 23:06.47 | Mark17 | RoyK: i will look at the source for the sample configs, dont want to install them on my working asterisk server ;) |
| 23:07.28 | *** join/#asterisk adr|an (n=xpl@unaffiliated/adrianxxx) |
| 23:07.49 | RoyK | Mark17: I've been using asterisk for six years or so. it started out as an experiment and moved into something worse |
| 23:07.53 | *** join/#asterisk obnauticus (n=lol@about/windows/regular/obnauticus) |
| 23:08.26 | Mark17 | i use it for almost a year |
| 23:08.30 | thansen | anyone in here using chan_mobile with a cell phone to send sms? |
| 23:08.37 | Mark17 | in the beginning with some web frontend |
| 23:08.39 | *** join/#asterisk thehar (i=thehar@thehar.xmission.com) |
| 23:08.52 | RoyK | Mark17: look at the code |
| 23:09.00 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 23:09.20 | RoyK | hands anthm a beer |
| 23:09.30 | Mark17 | but for some time now i dont use the web frontend any more (read: it was missing a lot of options and was limited to some old/not updated centos install) |
| 23:09.35 | thehar | go asssterisk |
| 23:09.57 | RoyK | makes some asstrix for thehar |
| 23:10.00 | Mark17 | thansen: i am planning to test it the coming month, but i will probably stick to the xml interface i have with a provider |
| 23:10.05 | thehar | haha |
| 23:10.14 | thehar | RoyK: i'm celebrating the #2 announcement |
| 23:10.38 | RoyK | <PROTECTED> |
| 23:10.39 | thansen | Mark17: do you have a link to your provider? which carier will you test with? |
| 23:10.53 | thehar | yes it is |
| 23:11.50 | RoyK | awaits a stable, scalable asterisk version and suspects it will be released in the beginning of 2020 |
| 23:12.19 | *** join/#asterisk nix8n82 (n=nate@63.162.27.243) |
| 23:12.31 | RoyK | anyone here tried to test the rtp code in asterisk? |
| 23:12.49 | RoyK | it makes a sever die at about 400 concurrent calls |
| 23:13.00 | RoyK | on linux, that is |
| 23:13.19 | *** join/#asterisk keebler (i=9446c2d5@gateway/web/ajax/mibbit.com/x-3e43b7a994703874) |
| 23:13.34 | Mark17 | thansen: i will test it with t-mobile in the netherlands, but i will probably stay at another sms gateway i use for many projects |
| 23:13.36 | thehar | dundi it then |
| 23:14.18 | RoyK | at about 400 concurrent calls, asterisk makes such amounts of system calls, it freezes the whole system |
| 23:14.20 | trnzmeta | guys: good/cheap sms providers in the US? |
| 23:14.35 | trnzmeta | we're using clickatell globally, but one client wants a US local to local |
| 23:14.42 | manxpower | RoyK: isn't his more of a #asterisk-dev thing? |
| 23:14.54 | Qwell | manxpower: #asterisk-fud |
| 23:15.06 | RoyK | manxpower: it's two years since I reported it, so I guess not |
| 23:15.39 | RoyK | manxpower: the rtp stack is still the same, and the answers I got back then was "asterisk is not a soft switch - live with it" |
| 23:15.55 | manxpower | RoyK: then why are you still using Asterisk? |
| 23:15.56 | trnzmeta | we're using clickatell globally, but one client wants a US local to local |
| 23:16.48 | RoyK | manxpower: works well for small installations, but the fact that digium and yourself refuse to listen to actual facts, is not good |
| 23:17.12 | [TK]D-Fender | RoyK: For reference, when was "back then"? And please include the precise version you were using... |
| 23:17.25 | manxpower | RoyK: I have exactly 3 lines of code in Asterisk. I'm not QUALIFIED to look at it. |
| 23:17.34 | *** join/#asterisk JAMMAN2110 (n=James@unaffiliated/jamman2110) |
| 23:17.37 | manxpower | [TK]D-Fender: I guess a bug ID would be helpfl too. |
| 23:18.04 | [TK]D-Fender | manxpower: I suppose it might add something |
| 23:19.06 | manxpower | At least if you file a bug you can be *officially* ignored, rather than unofficially. |
| 23:19.30 | RoyK | [TK]D-Fender: the precise version isn't really relevant - the rtp stack in asterisk hasn't changed much, has it? |
| 23:19.37 | Qwell | well, unless it includes details, which you've been holding back, it will be closed. |
| 23:20.06 | Qwell | There are plenty of people who do 400 simultaneous channels |
| 23:20.52 | RoyK | Qwell: I'll do a new test. Mind, this was all with RTP |
| 23:21.08 | RoyK | and 400 calls |
| 23:21.14 | RoyK | that is 800 channels |
| 23:21.19 | Qwell | well, you have yet to provide any useful information |
| 23:21.31 | thansen | Mark17: did you say who you're currently using? |
| 23:21.46 | RoyK | last I tried, I was told asterisk wasn't a softswitch, so it was irrelevant |
| 23:22.00 | thansen | trnzmeta: when you find a good one let me know :) I'm after the same thing...or gonna go the chan_mobile route |
| 23:22.55 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 23:23.12 | trnzmeta | hmm well clickatell is quite good, however my clients reckon 7c an sms is expensive |
| 23:23.31 | Khratos | This guy is using a 'mystery atmosphere' about details |
| 23:23.39 | adam000 | Does anyone in here have any experience with voicepulse sip/iax trunks? |
| 23:23.51 | *** join/#asterisk maddog01 (n=minotaur@dhcp-0-1e-58-0-b-f.cpe.mountaincable.net) |
| 23:24.05 | Qwell | RoyK: If there are any custom patches, they need to be disclosed. |
| 23:24.32 | Qwell | otherwise, nobody is going to waste any time on it. |
| 23:25.09 | *** join/#asterisk nix8n82 (n=nate@63.162.27.243) |
| 23:25.51 | [TK]D-Fender | RoyK>[TK]D-Fender: the precise version isn't really relevant - the rtp stack in asterisk hasn't changed much, has it? <- Makes me want to beat you over the head with a FUD-Stick |
| 23:26.16 | [TK]D-Fender | "last time I tried it" <- DITTO |
| 23:26.34 | [TK]D-Fender | Holy McFuck how unvalidated can you get? |
| 23:26.41 | Mark17 | thansen: what i use for chan_mobile as a telecom provider yes |
| 23:26.57 | thansen | trnzmeta: yeah, 7c sounds a little steep |
| 23:27.11 | thansen | Mark17: no, who do you currently use as an sms gateway |
| 23:27.12 | Mark17 | for the sms i did not say it (i am working on a new frontend on smsgateway.nu) |
| 23:27.19 | thansen | ah |
| 23:27.33 | Mark17 | one of my domains, i will offer it in the future ;) |
| 23:27.44 | RoyK | [TK]D-Fender: I'll do a new test - I guess it'll turn out like the last one. be seeing you |
| 23:28.32 | [TK]D-Fender | RoyK: Ok.... as long as we've gone from "non-descript point in the past" to "predicting the future".... |
| 23:29.02 | beek | notes "Holy McFuck" for future invective use |
| 23:29.42 | *** join/#asterisk km2 (n=x@cpe-74-64-12-212.nyc.res.rr.com) |
| 23:30.19 | *** join/#asterisk HermesNeto (n=HermesNe@189-94-17-26.3g.claro.net.br) |
| 23:30.20 | Khratos | haha |
| 23:30.42 | rue_mohr | k, so buying hardware echo can, I hope one day soon oslec works so I can give it a try |
| 23:30.53 | [TK]D-Fender | beek: then you'll love this one..... |
| 23:31.03 | rue_mohr | the tdm400P disn't have an option for a echo card does it? |
| 23:31.05 | [TK]D-Fender | thinks NostraDumbAss should sit this one out... |
| 23:31.18 | [TK]D-Fender | module unload chan_clairvoyance.so |
| 23:31.21 | Qwell | rue_mohr: I believe the 410 does |
| 23:31.28 | rue_mohr | ok |
| 23:31.32 | beek | :) |
| 23:31.32 | Qwell | maybe |
| 23:31.43 | [TK]D-Fender | rue_mohr: Echo comes bundled with the hardware ;) |
| 23:32.17 | rue_mohr | would you belive that even with everything I'v been though, asterisk is still the easier solution? |
| 23:33.39 | [TK]D-Fender | rue_mohr: Don't worry... you'll complicate it past the point of no return before long :p |
| 23:33.43 | rue_mohr | I still dont understand why I dont get any echo with my channelbank at home |
| 23:33.44 | [TK]D-Fender | rue_mohr: I have faith |
| 23:34.08 | [TK]D-Fender | rue_mohr: Bridged TDM <- |
| 23:34.24 | [TK]D-Fender | rue_mohr: effectively a straight path |
| 23:34.28 | [TK]D-Fender | (+/-) |
| 23:34.31 | rue_mohr | the company office is more complex than any customer we ever have |
| 23:34.57 | rue_mohr | its bridges nativly on the T1 card and or the channelbank itself? |
| 23:34.59 | Corydon76-dig | ~thebook |
| 23:35.00 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com |
| 23:35.17 | [TK]D-Fender | rue_mohr: You seriously should ahve been WEEKS ago |
| 23:35.22 | keebler | Anyone else use a Mikrotik system with Asterisk? |
| 23:35.26 | [TK]D-Fender | rue_mohr: Not that this hasn't been entertaining... |
| 23:35.39 | [TK]D-Fender | keebler: in what capacity? |
| 23:35.50 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 23:35.58 | rue_mohr | [TK]D-Fender, yes and no, I have spent days in pairs one being sent off in one direction, then being turned around and sent the opposite way |
| 23:36.04 | keebler | [TK]D-Fender: general routing/subnets. |
| 23:36.15 | keebler | via wireless or otherwise. |
| 23:37.20 | rue_mohr | right now I have a customer who wants a system with 4 phones, 2 lines, voicemail and autattendant, the estimates I had to give them today were $3700 and $2800 for new key'd systems |
| 23:37.35 | rue_mohr | the asterisk solution is $1200 |
| 23:38.15 | rue_mohr | and I cant even tell them about it |
| 23:38.20 | rue_mohr | cause I cant get this one working |
| 23:38.37 | rue_mohr | int eh office they think that each * system I sell will take this long to set up |
| 23:39.13 | rue_mohr | I'm no longer able to boast about the great tech support :/ |
| 23:39.41 | Mark17 | rue_mohr: asterisk can be working within 48 hours on request (completely done by us) or can be implemented on our shared voip platform without problems |
| 23:39.55 | Mark17 | as long as the lines in are voip lines and not real lines ;) |
| 23:40.02 | rue_mohr | they are real lines |
| 23:40.44 | Mark17 | than sorry, i dont currently work with real lines (and it might take a day or 2 to learn what is needed) |
| 23:40.53 | rue_mohr | One of the mistakes I made was to not get the echo card, I really thought that one of the free echo cans could do it, the built in ones make no difference at all, oslec dosn't even work with dahdi |
| 23:40.59 | *** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
| 23:41.36 | rue_mohr | alot of my time has been learning the voip phone configs |
| 23:42.13 | rue_mohr | Mark17, you with digium then? |
| 23:42.27 | rue_mohr | can I buy a support plan that will get me help with anything? |
| 23:42.55 | rue_mohr | there isn't much people will help me with here anymore :) |
| 23:43.17 | Mark17 | rue_mohr: no, i have an own company in the netherlands |
| 23:43.31 | rue_mohr | ah |
| 23:43.36 | Mark17 | and i use asterisk for some clients and for my own calls |
| 23:44.25 | Mark17 | and where possible i help opensource projects and opensource software users |
| 23:45.27 | bmoraca | keebler: like I said before...Mikrotik is a software routing solution built off of IPTables. it will probably work, but it'll be a hell of a lot more trouble than a hardware solution. |
| 23:46.03 | rue_mohr | I work for a company that, amongst other things, sells phone systems. I REALLY dont want to have to sell the key'd systems anymore, I spent 4 hours once while tech support for us had 2 technicians arguing about how to do something I could have done in 5 mins with asterisk |
| 23:46.18 | Mark17 | currently i support PHP (nl.php.net) and freenode (grisham.freenode.net) and am planning to support more opensource/opensource related projects as they need it (including new projects if i like them or i think that they are usefull) |
| 23:46.42 | rue_mohr | line 1, 2 -> ring sets 3,4,5 -> voicemail 1 line 3, 4 -> ring sets 3, 4, 5, 7 -> voicemail 2 |
| 23:47.17 | Mark17 | rue_mohr: that is a basic dialplan if you ask me :S |
| 23:48.33 | *** join/#asterisk guax (n=guaxinim@unaffiliated/guaxinim) |
| 23:48.44 | rue_mohr | yea, easy as pie |
| 23:49.10 | rue_mohr | 2 panasonic technicians argueing for 4 hours mostly about if it could evenbe done |
| 23:49.54 | rue_mohr | they managed to do it, but only just barely, its got funny side effects I dont remember, something about internal transfers iirc |
| 23:53.31 | rue_mohr | but this system at our main office is a good challenge, 4 lines, 4 business numbers and a fax line with distinctive ring |
| 23:54.10 | rue_mohr | 6 sets and 5 of them answer for all businesses and need to make sure outgoing calls happen on particular lines |
| 23:54.31 | *** join/#asterisk nightrid3r (n=kvirc@78-20-228-200.access.telenet.be) |
| 23:54.33 | rue_mohr | T1 too expensive and pri unavailable |
| 23:54.39 | [TK]D-Fender | rue_mohr: this EC thing should have taken 3 days tops. t he dialplan bit... thats a freebie. Your SLA deal? Should have really tried to ditch the concept. Failing to should ahve been done in a day... |
| 23:55.25 | rue_mohr | well, oslec does NOT work with dahdi on a TDM800P |
| 23:55.40 | [TK]D-Fender | rue_mohr: who else has taken a swing at it? |
| 23:55.56 | rue_mohr | I'm the only tech for the phone part of the business |
| 23:56.06 | rue_mohr | I'v redone the process twice with different instructions |
| 23:56.31 | rue_mohr | same result, no dahdi channeltypes when echocancelers set to oslec |
| 23:56.33 | keebler | bmoraca: Why will it be troublesome? |
| 23:56.47 | *** join/#asterisk demiv (n=demiv@190.158.82.133) |
| 23:58.59 | rue_mohr | [TK]D-Fender, I could hire rcy to come try for me, I dont suspect the result would be any different but you guys might be more willing to help him for a while |
| 23:59.42 | [TK]D-Fender | rcy? |
| 23:59.53 | rue_mohr | he's been here |
| 23:59.59 | [TK]D-Fender | rue_mohr: Never heard of... |