| 00:02.00 | dandate2 | philippines to los angeles, is the data travelling through the pacific ocean or over europe |
| 00:04.47 | drmessano | tracert? |
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| 00:11.39 | R-Guy | dandate2: I would guess (for what thats worth) it would go across the Pacific. |
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| 00:16.28 | R-Guy | Ping time for our traffic from the west coast to central China over the Pacific is 275ms -- Phillippines shouldn't be far off that. |
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| 00:32.39 | dandate2 | yeah from tracert looks like it goes right from manilla to los angeles. good thing i put the pbx in orange county |
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| 00:40.34 | dandate2 | would a DSL installation technician have any ability to affect the ping time and amount of packet loss experienced? |
| 00:40.51 | dandate2 | im going to bribe the guy for the best wiring but dont want to throw the money away if it cant help =/ |
| 00:41.45 | mmattice | "the best wiring"? |
| 00:41.54 | dandate2 | here in the philippines is real hit and miss, a net cafe with a great connection is 16% faster than another cafe 1 block away with the same provider |
| 00:42.08 | mmattice | ah |
| 00:42.23 | dandate2 | yeah when the installation tech comes could i bribe him for the best speed? |
| 00:42.39 | dandate2 | not megabytes up/down but ping times and packetloss |
| 00:43.46 | R-Guy | Probably most important to make sure you get a clean pair from the DSLAM to you modem. |
| 00:44.37 | dandate2 | are there any other terms for this or will the installation tech recognize DSLAM and "clean pair" universally |
| 00:44.42 | WIMPy | Probably most important to get the right configuration. |
| 00:44.58 | R-Guy | <PROTECTED> |
| 00:46.00 | dandate2 | i am so afraid, the center i rent gets 260-280ms to the pbx but this other center a block away gets 280-360ms |
| 00:46.13 | dandate2 | my residential DSL gets 220ms |
| 00:46.20 | dandate2 | i think im going to open a center in my house heh |
| 00:46.30 | WIMPy | Now, that's extreme. |
| 00:47.47 | WIMPy | Migt be a good idea to get a lower data rate. Better than lots of retransmissions. |
| 00:48.12 | dandate2 | you mean less megabytes up/down ? |
| 00:48.20 | WIMPy | yes |
| 00:48.25 | dandate2 | wow thats brilliant |
| 00:48.35 | dandate2 | no wonder my connection is so fast, its only half a meg |
| 00:49.27 | dandate2 | does there need to be any excess bandwidth when only running an ATA adapter |
| 00:50.50 | WIMPy | If you have bad line conditions I'd get a speed set that isn't much higher than needed. |
| 00:51.54 | WIMPy | For voip reliability is essential. |
| 00:53.10 | dandate2 | wow |
| 00:53.23 | dandate2 | every center here with bad lines has huge up/down rates heh |
| 00:53.47 | dandate2 | theres absolutely no redundancy |
| 00:54.40 | WIMPy | ? |
| 00:55.11 | dandate2 | no data centers here |
| 00:55.17 | dandate2 | packet lossi s rampant |
| 00:55.58 | dandate2 | then theres the calls dropping =@ |
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| 03:16.22 | Hectaman | I've got a fire panel that tries to do a ton of dtmf signalling and then dies |
| 03:16.31 | Hectaman | the call that is; outbound over a pri |
| 03:16.50 | Hectaman | inbound on the same 4 port digium card via an Adit 600 |
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| 03:41.56 | CajaCaliente | How is everyone? |
| 03:44.03 | sshock | HotBox: good |
| 03:44.24 | CajaCaliente | Ahhh someone can read my name properly now |
| 03:44.33 | sshock | yep :) |
| 03:45.55 | CajaCaliente | What ver of * are you running sshock? |
| 03:46.31 | sshock | actually I haven't installed it yet, but when I do it will be 1.4.21.2 |
| 03:46.50 | CajaCaliente | Whats your application for it and why are you sticking with the older version? |
| 03:47.07 | sshock | just using what comes with my Debian |
| 03:47.26 | sshock | There's three main reasons I want to try it out: |
| 03:47.47 | sshock | 1. I have a gizmo account and have read that I can set things up with gizmo and google voice to get free incoming and outgoing calls. |
| 03:48.17 | sshock | 2. I want to screen/block calls from certain telemarketers that are starting to bug me and won't stop calling back. |
| 03:49.17 | sshock | 3. Just for fun I'd like to set up a cool menu that can do fun stuff |
| 03:49.26 | CajaCaliente | Nice |
| 03:50.12 | CajaCaliente | I have a small office building that I do work for a financial planner and we provide internet and phone services to the rest of the building. |
| 03:50.13 | sshock | I figure with my Linux and programming experience I ought to be able to figure it all out. Just reading the PDF of the book now... |
| 03:50.44 | sshock | cool, so yours is an actual business application |
| 03:50.45 | CajaCaliente | Install and explore, it's a pretty simple program when you're not using any POTS lines |
| 03:51.08 | sshock | yeah, one of my goals is to avoid POTS completely... |
| 03:51.19 | CajaCaliente | Yea we got some cool stats, integrated phone books (via the Cisco 7960G's API and shit) |
| 03:51.37 | CajaCaliente | POTS are the only thing that gives me a fuckin headache with *, otherwise its smooth sailing all the time. |
| 03:51.49 | sshock | cool, that gives me hope |
| 03:52.07 | CajaCaliente | That and I can't get the landlord to let me ventilate my closet so my PoE switch keeps burning out fans because of the dust and shit. |
| 03:52.11 | sshock | I'm guessing it won't be too hard to connect my * to gizmo |
| 03:52.19 | sshock | haha |
| 03:52.33 | sshock | so is that where your name comes from? |
| 03:53.01 | CajaCaliente | Used to be a big bud smoker |
| 03:53.32 | sshock | ahh |
| 03:53.35 | CajaCaliente | And I lived in a real small room in an apt with other ppl. Whenever they'd open up my door, smoke would pour out :) So I figured its appropriate |
| 03:54.01 | CajaCaliente | Now I'm wasting away my life doing tech consulting and being a US marine corps reservist... *sigh* |
| 03:54.33 | sshock | that doesn't sound so bad |
| 03:55.11 | CajaCaliente | Just no money. Ran out of money to finish engineering school so w/ no degree finding stable jobs are hard. |
| 03:55.23 | sshock | oh; yeah, that sucks |
| 03:55.33 | sshock | think asterisk will run on my Pentium 2 333MHz ? |
| 03:55.58 | CajaCaliente | Hrmmm, probably. Depends on how thinned out the rest of the distro is I'd say. |
| 03:56.31 | CajaCaliente | How well does Debian run on that box as is? |
| 03:56.38 | sshock | pretty good actually |
| 03:57.00 | sshock | I've already got a web server, smtp server, jabber server, mpd server, etc on there and never had any issues |
| 03:57.24 | sshock | and since it just keeps chugging along with no problem, I don't bother to upgrade it (aint broke; don't fix it) |
| 03:58.16 | CajaCaliente | Yea |
| 03:58.24 | CajaCaliente | Are you going to be recording calls? |
| 03:58.33 | sshock | I'm hoping since I'm just using it for home use, which means at most one or two simultaneous calls, it won't be a problem |
| 03:58.38 | sshock | hmm, not normally probably |
| 03:58.57 | sshock | actually if no one answers I think I'll have it forward to a google voice account that will do the voicemail for me |
| 03:59.01 | CajaCaliente | You can start recording w/ a DTMF combination if you're ever interested. |
| 03:59.59 | sshock | I may try it out |
| 04:00.20 | CajaCaliente | We record all the calls for my financial planner and then we have a CPA we do the service for as well. Encrypted and uploaded to off site storage making it immediately over compliant with the various regulatory bodies we have overseeing "us" |
| 04:00.22 | sshock | I've got Vonage right now, and one of the things that bugs me is lack of control and ability to do random stuff |
| 04:00.34 | sshock | cool |
| 04:00.37 | CajaCaliente | Have you looked into OpenPBX and the sort? |
| 04:00.48 | sshock | I've heard about FreePBX |
| 04:01.00 | CajaCaliente | erm that's it :) my bad. had a couple of brews tonight |
| 04:01.02 | sshock | but I was thinking of diving directly into asterisk |
| 04:01.17 | sshock | but we'll see |
| 04:01.26 | drmessano | You wont be able to run a FreePBX based install on that box.. best to go with plain Asterisk |
| 04:02.01 | sshock | ok, so my instincts were probably right on that |
| 04:02.21 | sshock | one thing I hate about Vonage is they charge extra to use a softphone |
| 04:02.34 | CajaCaliente | That's some fuckin horeshit... |
| 04:02.35 | sshock | that's so dumb; once I get this all figured out I'll be able to use any standard SIP client from anywhere |
| 04:02.39 | sshock | yeah |
| 04:02.50 | CajaCaliente | Have you looked into aretta? |
| 04:03.17 | sshock | never heard of it |
| 04:03.19 | drmessano | Flowroute is pretty damn good, and their pricing is second to none |
| 04:03.48 | drmessano | Aretta really doesn't offer anything priced well for residential |
| 04:04.01 | sshock | well, if this thing with gizmo and google voice works, I'll have free incoming and outgoing calls to anywhere US and Canada |
| 04:04.41 | sshock | and if not, the gizmo prices are $4/month for incoming from POTS and $.01/min for outgoing to POTS |
| 04:05.16 | sshock | although since Google bought Gizmo, I'm not being charged the $4/month for incoming calls, and it says my call-in number doesn't expire until 2015 |
| 04:05.29 | *** join/#asterisk mog (~mog@c-68-62-169-225.hsd1.al.comcast.net) |
| 04:05.33 | sshock | so I think the worst I will have is 1cent/min outgoing and that's it |
| 04:06.59 | CajaCaliente | drmessano What are you applications for *? |
| 04:07.15 | drmessano | What do you mean? |
| 04:08.18 | CajaCaliente | How do you use it? |
| 04:09.30 | drmessano | I guess it depends which of the many installs I have done over the years, for various projects, using different features of Asterisk that you're asking about |
| 04:10.18 | carrar | Making calls! |
| 04:10.29 | drmessano | ^^ that's a big one |
| 04:10.44 | drmessano | Taking calls <--- Somewhere in the top5 as well |
| 04:10.58 | carrar | Thats what DND is for |
| 04:11.03 | carrar | DND + VM |
| 04:14.46 | sshock | Here's a dumb question; to test asterisk, am I going to need two boxes, or can I do it with one? |
| 04:15.02 | carrar | You will need 20 boxes |
| 04:15.07 | sshock | I'm guessing I'll need two, because both asterisk and a softphone will want to bind to port 5060 |
| 04:15.13 | sshock | :) |
| 04:15.35 | drmessano | You're gonna run the softphone off that box too? |
| 04:15.52 | carrar | softphones are not SIP Servers |
| 04:15.54 | sshock | just for now; for testing |
| 04:16.05 | carrar | they won't bind to UDP port 5060 |
| 04:16.22 | sshock | oh, maybe it was only doing tcp |
| 04:16.29 | sshock | ok, so maybe I can test it after all |
| 04:17.12 | sshock | (I want to do all my testing from a box behind my firewall before putting it on my firewall/server) |
| 04:17.43 | carrar | Buy a real SIP phone |
| 04:17.57 | carrar | You're gonna want one anyways |
| 04:18.07 | sshock | yeah, not a bad idea, since I need to eventually; read my mind |
| 04:18.22 | sshock | what's a good & cheap IP phone? |
| 04:18.39 | carrar | LinkSys, Cisco, Polycom, Aastra |
| 04:18.53 | sshock | can you find any for under $100? A quick search showed cheapest one at $150 |
| 04:19.06 | drmessano | Yes you can |
| 04:19.11 | carrar | you might have to spend more thenb 10 seconds searching |
| 04:19.51 | sshock | yeah |
| 04:20.52 | sshock | my current (POT) phones are dying out, so I plan to buy IP phones regardless, rather than get a digital<->analog router or PCI card |
| 04:21.22 | carrar | http://cgi.ebay.com/4-Polycom-SoundPoint-IP-430-SIP-Telephones-/270639689453 |
| 04:21.45 | carrar | do some bid sniping |
| 04:21.50 | sshock | wow |
| 04:21.55 | sshock | oh, ebay; of course |
| 04:21.58 | carrar | get 4 for $30 |
| 04:22.19 | p3nguin | For parts or not working |
| 04:22.26 | sshock | that would be sweet |
| 04:22.49 | carrar | ah didn't read that part :) |
| 04:23.14 | sshock | this one seems to only show the base station, not an actual phone |
| 04:23.36 | carrar | http://cgi.ebay.com/Polycom-SoundPoint-IP-430-VoIP-phone-/330477722112 |
| 04:24.03 | sshock | Of course, I also want it to be wireless :) |
| 04:24.14 | carrar | Polycom Spectralink 8020 |
| 04:24.15 | WIMPy | No, you don't. |
| 04:24.20 | sshock | why not? |
| 04:24.20 | carrar | they work great |
| 04:24.26 | WIMPy | If you want wireless, go for dect. |
| 04:24.45 | sshock | hmm, is that going to be expensive? |
| 04:24.53 | WIMPy | wifi is a pita for realtime applications. |
| 04:24.56 | carrar | over $400 |
| 04:25.04 | sshock | hmm, crap; |
| 04:25.10 | sshock | so maybe it would be cheaper to go the digital->analog route |
| 04:25.18 | carrar | Or get a Aastra 57iCT |
| 04:25.23 | carrar | it has a wireless phone with it |
| 04:25.25 | WIMPy | Or digital>digital? |
| 04:25.38 | sshock | well, because normal analog phones aren't that expensive |
| 04:26.13 | WIMPy | No, but then, you probably don't want to use them. |
| 04:26.13 | sshock | of course, then I would lose out on all the benefits of having IP phones |
| 04:26.19 | carrar | doing DA conversions is never the best option |
| 04:26.38 | sshock | well, I'm doing it right now with vonage |
| 04:26.45 | carrar | I'm sorry |
| 04:26.54 | sshock | yeah, and of course they locked down the router, |
| 04:27.04 | sshock | so I can't even re-use it after dropping Vonage |
| 04:27.33 | sshock | a Linksys RTP300 |
| 04:27.45 | mmattice | they have a BOYD plan don't they? |
| 04:28.46 | sshock | boyd? |
| 04:29.24 | carrar | by yourself a cheap inexpensive sip phone |
| 04:29.26 | carrar | http://cgi.ebay.com/Linksys-SPA942-IP-Phone-AC-Power-Supply-/130434654119 |
| 04:29.33 | sshock | oh, BYOD; you spelled the acronym worng :) |
| 04:29.42 | carrar | Those work fine too |
| 04:29.44 | mmattice | dyslexic typing |
| 04:29.52 | sshock | they might, |
| 04:30.09 | sshock | but the point is I already bought this RTP300, and I don't know if there's a way to unlock it from Vonage |
| 04:30.35 | mmattice | you can ask them obviously |
| 04:30.38 | sshock | the voice settings page just says "Please contact your service provider for further information." |
| 04:30.55 | sshock | yeah, I bet they'll just love to tell me how to unlock it |
| 04:31.05 | WIMPy | Doesn't google come up with any howtos? |
| 04:31.26 | sshock | Hmm, there might be something on it; I haven't actually searched for that yet. |
| 04:32.14 | carrar | http://www.dslreports.com/forum/r21123078-Unlock-TUTORIAL-VONAGE-WRTP54GRTP300-WITH-50104 |
| 04:32.33 | carrar | Let me know if there is anything you need googled |
| 04:32.49 | sshock | so, maybe this will be cheaper than buying IP phones... |
| 04:33.00 | carrar | and more problems for you |
| 04:33.03 | carrar | sure |
| 04:33.07 | carrar | have at it |
| 04:33.49 | sshock | of course, I don't want to play around with that until I'm 100% sure I've got asterisk set up and working how I want and ready to drop vonage |
| 04:34.20 | sshock | I'm sure once unlocked Vonage won't help me "fix" it back |
| 04:35.06 | carrar | Why not keep Vonage and their hardware working and buy a sip phone |
| 04:35.07 | WIMPy | OTOH IADs can be bought for 10.95. |
| 04:35.45 | carrar | then if you wanted to you could connect Vonage to your asterisk box |
| 04:36.17 | sshock | IAD? |
| 04:36.45 | mmattice | Improvized Audio Device |
| 04:36.46 | WIMPy | Integrated Access Device - Modem / Router / ATA combo box |
| 04:36.49 | mmattice | ;) |
| 04:36.57 | sshock | oh, ok; I've heard ATA before but not IAD |
| 04:37.30 | WIMPy | NGN termination |
| 04:37.59 | sshock | do they have IADs that can provide multiple lines? |
| 04:38.25 | sshock | I've got cat5 running through my house, so in theory I should be able to have 4 POT phone lines |
| 04:38.46 | WIMPy | Yes. Those el cheapo ones have 4 analogue and 1 ISDN port. |
| 04:38.55 | sshock | sweet |
| 04:38.58 | WIMPy | Sorry. 3+1 |
| 04:39.21 | sshock | because the main feature I don't want to lose by going with analog phones is the ability to have one phone extension for me and a separate one for my wife |
| 04:39.38 | carrar | go SIP |
| 04:39.47 | carrar | POT is so going backwards |
| 04:40.03 | sshock | well, that's what I thought originally too |
| 04:40.11 | carrar | and you were originally correc |
| 04:40.12 | carrar | t |
| 04:40.12 | sshock | but I'm being told a wireless IP phone is sooo expensive |
| 04:40.22 | carrar | You have cat5 |
| 04:40.25 | WIMPy | No, but to shitty. |
| 04:41.46 | sshock | I don't think I could go back to a corded phone, even if it is a sweet IP phone |
| 04:41.54 | sshock | this is for a home setting, not an office |
| 04:41.57 | carrar | Put your computers one wifi channel 1 & your wifi sip phone on 6 or 11 |
| 04:41.57 | WIMPy | It's nice if you're on your way, but the conversation may suffer from quite some glitches. |
| 04:42.24 | WIMPy | If you don't have neighbours... |
| 04:42.32 | sshock | hmm, but to use different channels means buying another wireless AP, right? |
| 04:42.37 | carrar | yup |
| 04:42.48 | carrar | and one that works best with the wifi sip phone |
| 04:42.55 | carrar | some reqire mmw |
| 04:43.10 | sshock | if you say it is worth it |
| 04:43.21 | *** join/#asterisk timahvo1 (~rogue@41.191.224.178) |
| 04:43.54 | sshock | that wouldn't be too bad; I can get a wireless router pretty cheap |
| 04:44.58 | sshock | wouldn't it be possible to have an IP phone with a base station that is hard-wired to the network |
| 04:45.13 | WIMPy | Cheapest and reliable solution is a BRI card in your server and an ISDN dect base. |
| 04:45.38 | WIMPy | My experiences with the cheap SIP dect bases were less than great. |
| 04:46.14 | sshock | hmm... |
| 04:48.52 | carrar | sshock, read this: http://www.polycom.com/global/documents/products/voice/mobile/PLCM-SL-8020-8030.pdf |
| 04:49.22 | sshock | ok |
| 04:50.11 | carrar | SpectraLink 8002 also works great |
| 04:50.48 | sshock | but I still don't think I want to spend that much |
| 04:50.57 | carrar | then buy crap |
| 04:51.02 | carrar | and get crappy results |
| 04:52.03 | ChannelZ | hurray! |
| 04:52.18 | ChannelZ | Won't you take me to |
| 04:52.21 | ChannelZ | Crappy town! |
| 04:52.27 | sshock | :) |
| 04:52.31 | carrar | bus is leaving soon |
| 04:52.36 | carrar | sshock is driving |
| 04:52.56 | carrar | as noted, it's the short bus |
| 04:53.33 | sshock | I can upgrade later; for now I just want to get * working. |
| 04:55.13 | sshock | I may just do all my testing with softphones... |
| 04:55.15 | *** join/#asterisk shamelessn00b (~chatzilla@58-65-172-114.nayatel.pk) |
| 04:55.49 | WIMPy | At least you don't waste money on crap then. |
| 04:56.01 | carrar | Telephoney depot sells the Aastra 57i CT Wireless VoIP phone base station for about $200 |
| 04:56.13 | carrar | SIP phone with awireless handset that also rings with it |
| 04:56.37 | carrar | in 1.9ghz dect rf space |
| 04:56.47 | sshock | hmm, that doesn't sound too bad |
| 04:56.52 | WIMPy | For 172 EUR you get an ISDN dect base incl. 5 mobiles. |
| 04:58.03 | sshock | not bad |
| 04:58.11 | WIMPy | BRI cars is 20.- so you get five reliable phones instead of one. |
| 04:58.55 | carrar | Just get a single story house and yell louder |
| 04:58.59 | carrar | FREE |
| 04:59.14 | sshock | :) |
| 04:59.38 | sshock | so does BRI card mean I'm using ISDN? |
| 05:00.02 | WIMPy | yes |
| 05:00.34 | carrar | and it will cost more in service |
| 05:00.34 | sshock | ISDN uses same connection as normal phones? |
| 05:00.53 | sshock | or does it use cat5 (rj45)? |
| 05:01.24 | carrar | bri is single pair |
| 05:01.26 | WIMPy | 8p4c, so what's usually calles RJ45 |
| 05:01.41 | WIMPy | Two pairs. |
| 05:01.51 | sshock | ok |
| 05:01.52 | carrar | dpends how many lins |
| 05:02.02 | WIMPy | Huh? |
| 05:02.07 | sshock | what do you mean by cost more in service? |
| 05:02.29 | carrar | sip service over the internet will be cheaper then getting ISDN service |
| 05:02.50 | WIMPy | We're not talking about getting an ISDN line. |
| 05:02.55 | sshock | ok, yeah; I'm not interested in any ISDN service |
| 05:03.16 | WIMPy | We're talking about using an ISDN interface to connect dect phones to an Asterisk server. |
| 05:03.34 | carrar | ah |
| 05:04.31 | sshock | ok, so all I'd have to do is plug a BRI card into my * server and connect a CAT5 cable from that to the phone base station |
| 05:04.54 | sshock | and * will know how to talk to that, and the quality will be good? |
| 05:05.03 | WIMPy | More or less. You will probably need a crossover cable. |
| 05:05.12 | sshock | ok |
| 05:05.25 | sshock | so that's not a SIP phone then |
| 05:05.40 | WIMPy | Depends on your card and used software. But they can be made to talk. |
| 05:05.52 | WIMPy | That's not SIP, no. |
| 05:06.05 | WIMPy | That's good old telephony hardware. |
| 05:06.05 | sshock | but it's still going to have more functionality than a normal analog phone |
| 05:06.12 | sshock | or no? |
| 05:06.21 | WIMPy | Yes. |
| 05:06.58 | sshock | ok, I'll definitely keep it in mind for when I'm ready to start buying phones |
| 05:07.14 | WIMPy | Only problem is that it seems impossible to do call transfers with Asterisk, but unless you want to use oter phones as well, the dect base will handle that in your case. |
| 05:07.40 | sshock | ok, I see |
| 05:08.14 | WIMPy | But maybe one day this exotic feature will find it's way... |
| 05:08.19 | drmessano | You can do transfers.. you always have feature codes |
| 05:08.35 | WIMPy | That's not a transfer, that's crap. |
| 05:10.00 | drmessano | If you say so |
| 05:11.26 | sshock | here's a dumb question; don't most IP phones still use standard old RJ12 phone connection for the handset (and headset) jacks to the base? |
| 05:11.48 | sshock | at least, I know my shoretel phone I use at work does |
| 05:11.49 | WIMPy | Yes |
| 05:12.03 | drmessano | RJ12, no |
| 05:12.05 | WIMPy | Why should they be different? |
| 05:12.05 | shamelessn00b | WIMPy: you can do transfers, eg the teco sends you 1234XXXX and you get the last 4 digits from the extension sent and dial it |
| 05:12.11 | drmessano | RJ12 is 6P6C |
| 05:12.37 | WIMPy | shamelessn00b: Pardon? |
| 05:12.59 | sshock | sorry, can't keep my terms straight |
| 05:13.00 | drmessano | Some call the standard 4P4C handset jack an RJ9, but it doesn't technically have an RJ designation |
| 05:13.18 | WIMPy | can never remember those RJ*, but then they are usually only used wrongly anyway. |
| 05:14.24 | *** join/#asterisk WWGD (~WWGD@208.79.14.130) |
| 05:15.08 | sshock | the reason I ask is because, technically doesn't that mean I could connect an analog phone (even a cordless one) to the IP phone base station? |
| 05:15.51 | WIMPy | No |
| 05:15.52 | shamelessn00b | WIMPy: you can use AMI, register an event, when the user presses a button you can redirect them where you want |
| 05:16.01 | WIMPy | Receiver != phone. |
| 05:16.49 | WIMPy | shamelessn00b: Yes, but that would be the same as the inband "feature" stuff and won't work with any phones transfer feature. |
| 05:16.57 | sshock | doh; ok, I think you're right |
| 05:17.31 | sshock | I think I might be able to receive calls with it, but it wouldn't recognize any dialing |
| 05:17.33 | WIMPy | Which indeed seems odd to me as 1.8 got a lot of other less important features added in that direction. |
| 05:17.42 | sshock | maybe |
| 05:17.52 | WIMPy | Not even that. |
| 05:18.05 | sshock | ok |
| 05:18.14 | drmessano | sshock: Nonsense.. even the plugs fit, which they wont, the cordless will expect a line present.. the won't be |
| 05:18.17 | drmessano | there* |
| 05:18.26 | drmessano | sshock: Nonsense.. even IF the plugs fit, which they wont, the cordless will expect a line present.. the won't be |
| 05:18.29 | sshock | alright |
| 05:18.31 | drmessano | GAH |
| 05:18.33 | WIMPy | That's a completely different interface, even if you can get the plug in. |
| 05:19.16 | shamelessn00b | yeah I mean 1.8 got that voice pitch changing feature, who needs it anyways :/ |
| 05:19.54 | WIMPy | shamelessn00b: Oh. Well, I tried that one before. It's fun, but rather senseless. |
| 05:20.45 | sshock | so regular analog cordless phones use 5.8GHz spectrum; what does DECT use? |
| 05:21.40 | shamelessn00b | SRTP and SDES support might be useful |
| 05:21.42 | WIMPy | Depends on country. 1.8-2.5 GHz |
| 05:21.47 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 05:21.59 | WIMPy | Yes. |
| 05:22.18 | sshock | ok |
| 05:22.50 | WIMPy | Default is 1880-1900. |
| 05:23.35 | sshock | isn't it stupid that 2.4GHz has to be shared by everything from WiFi to microwave ovens? |
| 05:23.57 | sshock | seriously, why should my network have to flake out when I go heat up a burrito? |
| 05:24.03 | WIMPy | Use wires. |
| 05:24.27 | sshock | yeah, but no one wants to use wires to talk on the phone at home, now do they? |
| 05:24.57 | WIMPy | I don't want to care about charging batteries. |
| 05:25.16 | sshock | well, yeah that can be annoying, but most base stations I know of also charge |
| 05:25.34 | sshock | so your phone charges when you're not using it |
| 05:25.51 | WIMPy | And you always have to search for the phone, because you left it in some strange place. |
| 05:27.11 | sshock | hehe, well that is true, but you also have base stations with a "find HS" button |
| 05:27.21 | drmessano | WIMPy, do you use one of those 10-yr old Nokia cell phones with the 160x160 display on it because "those smart phones will never last"? |
| 05:27.44 | WIMPy | Actually I don't use a mobile phone any more. |
| 05:28.00 | drmessano | Analog or die? |
| 05:28.29 | WIMPy | No Analog for me. |
| 05:28.44 | WIMPy | I prefer some comfort. |
| 05:28.57 | *** join/#asterisk sipsurfer29 (47ac599b@gateway/web/freenode/ip.71.172.89.155) |
| 05:29.03 | sipsurfer29 | what's a good DID provider? |
| 05:29.11 | WIMPy | That's why I occasionally rant about the voip stuff. |
| 05:30.19 | sipsurfer29 | anyone? |
| 05:30.52 | sshock | sipsurfer29: I used Gizmo which was $4/month for DID number, but then Google bought them out and suspended new user sign-up |
| 05:31.06 | sshock | so that won't work for you |
| 05:31.33 | drmessano | sipsurfer29, Flowroute is pretty awesome |
| 05:31.38 | sipsurfer29 | so what's a good provider where I can receive calls for as little cost as possible? |
| 05:31.41 | sshock | too bad Google Voice won't let you forward calls to any SIP number, cuz then your DID would be free |
| 05:31.43 | sipsurfer29 | flowroute? alright i'll check them out |
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| 05:48.16 | CajaCaliente | Anyone here use agents? |
| 05:49.07 | uqlev | CajaCaliente, the secret ones? |
| 05:49.22 | CajaCaliente | uhhh not sure. Basically here's my issue: |
| 05:50.04 | CajaCaliente | Calls comes in on DAHDI 1 to SIP/ClientA, ClientA does a transfer (not blind) to ClientB by calling them to announce who is on DAHDI 1, then completes the transfer to ClientB |
| 05:50.26 | CajaCaliente | I end up with two call recordings with no definitive way to collate who the end (internal) party was |
| 05:50.52 | CajaCaliente | And I had read something along the lines that using some sort of call queue with agents would work? |
| 05:51.21 | CajaCaliente | I just wasn't sure if that fit my bill because I'm talking about like 2 DID's coming into 3 internal SIP lines |
| 05:51.36 | uqlev | regret I have no idea |
| 05:51.54 | CajaCaliente | Do you use call recording or CDR at all? |
| 05:52.03 | uqlev | no |
| 05:57.41 | *** join/#asterisk Faustov (user@gentoo/user/faustov) |
| 05:57.48 | CajaCaliente | Hola Faustov |
| 05:58.06 | Faustov | hi |
| 05:58.28 | CajaCaliente | What solutions have you used * for? |
| 06:01.09 | *** join/#asterisk wierdo (~jimmy@212.25.51.150) |
| 06:01.45 | CajaCaliente | wierdo: What solutions have you used * for? |
| 06:01.52 | *** join/#asterisk zamba (marius@flage.org) |
| 06:02.37 | wierdo | CajaCaliente, call centers, office PBX ... many things |
| 06:03.10 | CajaCaliente | Have you done much with call recording? |
| 06:03.37 | wierdo | yes |
| 06:04.10 | CajaCaliente | Would you be willing to maybe comment on a predicament I'm having? |
| 06:04.14 | *** join/#asterisk grEvenX (~even@apb9hb.ip.ssc.net) |
| 06:04.44 | wierdo | ok |
| 06:04.48 | CajaCaliente | Calls comes in on DAHDI 1 to SIP/ClientA, ClientA does a transfer (not blind) to ClientB by calling them to announce who is on DAHDI 1, then completes the transfer to ClientB |
| 06:04.57 | CajaCaliente | I end up with two call recordings with no definitive way to collate who the end (internal) party was |
| 06:05.13 | CajaCaliente | And I had read something along the lines that using some sort of call queue with agents would work? Any for sure answer? |
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| 06:05.17 | CajaCaliente | I just wasn't sure if that fit my bill because I'm talking about like 2 DID's coming into 3 internal SIP lines |
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| 06:08.13 | *** part/#asterisk diegoviola (~diego@190.23.48.144) |
| 06:08.23 | wierdo | can you just write in some variable which is the first internal party and include it in the recording name |
| 06:09.14 | CajaCaliente | I suppose that would be a kinda hack to get it done but I had heard there was some sort of clean solution for it |
| 06:10.27 | wierdo | maybe, but i have to run some tests, not shure how it could be done right now |
| 06:11.06 | CajaCaliente | No worries, just joined up to maybe help a few souls and be able to bounce some questions against others |
| 06:11.15 | CajaCaliente | Thanks for your willingness though |
| 06:11.15 | wierdo | :) |
| 06:11.20 | wierdo | np |
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| 06:26.24 | J4zen | Can anyone recommend a good book/readup on VoIP(or Asterisk) and SS7/C7? |
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| 07:56.39 | lorenzom | hi all |
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| 07:57.39 | ChannelZ | hi |
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| 08:14.03 | shamelessn00b | hello |
| 08:16.17 | *** join/#asterisk tzafrir_laptop (~tzafrir@local.xorcom.com) |
| 08:17.53 | *** join/#asterisk garymc (~chatzilla@host81-148-79-26.in-addr.btopenworld.com) |
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| 08:30.48 | heffer | tzafrir_laptop: vzaphfc with dahdi 2.4.0 doesn't seem to work. error message is "Unable to receive TEI from network in state 2(Assign awaiting TEI)!", 2.3.0.1 seems okay |
| 08:31.19 | tzafrir_laptop | zaphfc, right? |
| 08:31.25 | heffer | right, from your git |
| 08:31.51 | heffer | versions are asterisk 1.8 rc2, libpri svn r2006, dahdi 2.4.0 |
| 08:33.47 | heffer | it never gets over the stage of waiting for a TEI assignment |
| 08:51.07 | Dovid | hello ev1 |
| 08:51.20 | Dovid | tzafrir: moadim lesimha |
| 08:54.59 | tzafrir_laptop | heffer, is layer 1 down? |
| 08:55.30 | tzafrir_laptop | Can you call out? Call in? Call out possible only shortly after an incoming call? |
| 08:55.57 | tzafrir_laptop | Dovid, hi :-) |
| 08:57.35 | heffer | tzafrir_laptop: i'll check. i can't call in or out though. i'll test the calling out after calling in |
| 09:04.36 | heffer | tzafrir_laptop: the card doesn't react to incoming calls, outgoing calls makes it emit TEI management messages, layer 1 is up and card is in TE mode |
| 09:20.34 | *** join/#asterisk ybit2 (~quassel@unaffiliated/ybit) |
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| 09:46.56 | *** join/#asterisk TobSnyder (~schneider@dslb-088-073-182-034.pools.arcor-ip.net) |
| 09:48.58 | TobSnyder | someone familiar with timing parameters like prewink, preflash, wink, rxwink, rxflash, flash, start etc? |
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| 09:59.44 | *** join/#asterisk rethus1 (~suther@p5087EDE2.dip.t-dialin.net) |
| 09:59.56 | rethus1 | need ConfBridge Dahdi ? |
| 10:04.37 | tzafrir_laptop | TobSnyder, what about them? |
| 10:06.08 | tzafrir_laptop | heffer, what do you have in /etc/dahdi/system.conf ? Also: can you enable pri debug 2 on the span (intense debug) |
| 10:06.21 | tzafrir_laptop | and include a short trace from it, that is |
| 10:07.57 | heffer | tzafrir_laptop: system.conf: http://fpaste.org/qDup/ |
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| 10:14.09 | heffer | tzafrir_laptop: debug log: http://fpaste.org/nXbY/ |
| 10:14.37 | catphish | i just installed 1.8 on a new host, unlike 1.4 it seems that when connecting with "asterisk -vvvvvr", I get no log output, how do I view the log? |
| 10:18.59 | ectospasm | catphish: be sure to enagle console logging in logger.conf |
| 10:19.12 | ectospasm | s/enagle/enable/ |
| 10:19.35 | catphish | i'm using a config from 1.4 - not sure if i'll need to make more extensive changes? |
| 10:19.50 | shamelessn00b | heffer: you system.conf looks suspicious to me |
| 10:19.50 | ectospasm | yeah, that will probably be a bad idea |
| 10:19.59 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 10:20.04 | catphish | i don't even have a logger.conf :( |
| 10:20.26 | ectospasm | that's a problem. I don't know what the default behavior is then. |
| 10:20.45 | catphish | i'll spend more time on a new config |
| 10:20.45 | ectospasm | ...I know FreePBX disables console logging by default. |
| 10:20.51 | heffer | shamelessn00b: that's what is generated by dahdi_genconf |
| 10:21.34 | shamelessn00b | what card do you have |
| 10:24.50 | rethus1 | is "SET CALLERID only for meetme, or can i use it wit ConfBridge or Konference too? |
| 10:25.56 | heffer | shamelessn00b: a HFC-S PCI (the cheap cologne chip ones) |
| 10:26.28 | rethus1 | did somebody know how i can set the User:id in Konference to another calue? |
| 10:26.56 | shamelessn00b | it supports T1? |
| 10:31.30 | tzafrir_laptop | heffer, just another sanity check: that connection is not ptp, it's ptmp, right? |
| 10:31.56 | ruyo | Anyone knows where can I get information on how to interpret mISDNdebugtool output? |
| 10:32.23 | ruyo | I'm looking at mISDNdebugtool.h but the info there doesn't add up to the actual debug output. |
| 10:35.08 | fauxalliance | Good Morning |
| 10:36.08 | heffer | tzafrir_laptop: yes ptmp |
| 10:36.20 | heffer | a standard german ISDN line :D |
| 10:42.05 | *** join/#asterisk c0rnoTa (~c0rnoTa@109.188.77.225) |
| 10:42.39 | c0rnoTa | Hello everyone |
| 10:43.31 | *** join/#asterisk themArt (~themArt@host81-149-124-219.in-addr.btopenworld.com) |
| 10:43.41 | themArt | Mornin |
| 10:44.07 | themArt | Any change with some help with a 'utils.c: fwrite() returned error: Broken pipe' error? |
| 10:47.03 | shamelessn00b | hardhdlc=16 |
| 10:47.46 | fauxalliance | themArt, php probably. what does the generators log indicate? |
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| 10:48.46 | themArt | fauxalliance: yar.. PHP. I'm trying to originate a call. Any simples solutions? Use Perl for example? |
| 10:49.13 | c0rnoTa | I'm using E1 channel to connect to PSTN. Sometimes I get debug messages "Got a FRAME_CONTROL (8) frame on channel DAHDI/". I receive this when call is active and people speaking. The call drops when this message was received. My telco said, that my PBX disconnect the call (send Disconnect frame with cause 16 - normal clear). russellb siad: "control frames are call signaling information ... 8 is Congestion" |
| 10:49.24 | fauxalliance | themArt, start looking at the php logs... pastebin it |
| 10:49.55 | themArt | fauxalliance: Okley dokes. I'll see what I can dig up :-) Ta! |
| 10:50.25 | fauxalliance | themArt, you had it right the first time... use PERL as this is obviously a PHP buggy way of managing sockets... however, not a simple solution./ |
| 10:50.40 | c0rnoTa | I thought I was getting Congestion frame from telco but it's not |
| 10:51.04 | themArt | fauxalliance: ha! I'll take a look. Could be quicker in my case! |
| 10:51.35 | c0rnoTa | So, the question is: What is the initiator of the message in the log? |
| 10:51.46 | tzafrir_laptop | c0rnoTa, and that congestion did not come from the remote end? |
| 10:52.02 | tzafrir_laptop | It can also be "wrong number" and such |
| 10:54.46 | c0rnoTa | tzafrir_laptop: yes. There is no Congestion frame in 'pri debug'. This message appears when call successfully set. |
| 10:56.37 | fauxalliance | themArt, check out the manager.conf read = system,call,log,verbose,command,agent,user,config,originate |
| 10:56.37 | fauxalliance | write = system,call,log,verbose,command,agent,user,config,originate |
| 10:56.56 | themArt | fauxalliance: I've got read: all and write: all |
| 10:57.03 | themArt | fauxalliance: That should work huh? |
| 10:57.10 | fauxalliance | themArt, you will probably still get that broken pipe, but the call should originate properly |
| 10:57.33 | fauxalliance | (debian lenny *1.6) |
| 10:57.47 | fauxalliance | PHP 5.2.6-1+lenny8 with Suhosin-Patch 0.9.6.2 |
| 10:58.19 | themArt | fauxalliance: The exten is SIP and the number is internal. The internal has Monitor and WaitForSilence.. The call originates correctly, but, the call recording file appears empty. |
| 10:58.39 | themArt | fauxalliance: If I call the number via a SIP client, it recird the call quite happily. |
| 10:58.53 | themArt | fauxalliance: (records) |
| 10:59.36 | fauxalliance | another kettle of fish entirely |
| 10:59.54 | themArt | ha! thought it maybe :-( |
| 11:02.24 | *** join/#asterisk Naikrovek (~jjohnson@unaffiliated/naikrovek) |
| 11:02.36 | fauxalliance | Good Morning Naikrovek |
| 11:02.40 | themArt | fauxalliance: thanks anyway :-) |
| 11:02.47 | Naikrovek | morning |
| 11:02.48 | fauxalliance | themArt, GL! |
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| 11:33.02 | rethus1 | did somebody know how i can set the User:id in Konference to another value? |
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| 11:53.21 | TobSnyder | tzafrir_laptop: concerning timing parameters like prewink, preflash, wink, rxwink, rxflash, flash, start etc: just want to know how to configure them, to enable the Hook Flash key (Recall-Key) on analog telephone sets |
| 11:54.39 | tzafrir_laptop | Try reducing the size of rxflash |
| 11:55.11 | TobSnyder | currently those values are not set in any config, so a I guess they are set to default |
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| 11:56.41 | TobSnyder | default of rxfalsh is 1250ms, so what would you set there |
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| 12:02.37 | Naikrovek | oh god i hate my boss, job, career, life, planet, solar system, universe |
| 12:03.11 | *** join/#asterisk [TK]D-Fender (~chatzilla@216.191.106.163) |
| 12:08.01 | fauxalliance | Naikrovek, *grin* it's Monday after all |
| 12:08.32 | shamelessn00b | Naikrovek: who doesnt |
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| 12:25.08 | francispereira | Outbound calls from my asterisk server work but I am unable to hear the voice. Could this be a codecs problem? If so, how do i debug it ? |
| 12:25.24 | drmessano | ~sipnet |
| 12:25.27 | drmessano | ~sipnat |
| 12:25.27 | infobot | it has been said that sipnat is Quick guide on configuring Asterisk + 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 |
| 12:26.10 | [TK]D-Fender | drmessano: Regrettably my server is down... |
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| 12:29.51 | francispereira | [TK]D-Fender, any pointers about debugging my sound problem |
| 12:32.19 | ectospasm | francispereira: as drmessano pointed out, it seems you've got a SIP+NAT issue |
| 12:32.56 | ectospasm | ~sipnat | francispereira |
| 12:33.06 | ectospasm | oh, so redirections don't work |
| 12:33.16 | ectospasm | ~sipnat > francispereira |
| 12:33.21 | ectospasm | nope |
| 12:33.43 | Katty | mew |
| 12:33.57 | stix | Is Beronet okay or is it cheap copies of Digium? |
| 12:34.08 | stix | I am thinking about buying a BRI isdn card |
| 12:34.20 | russellb | you should buy Digium, obviously :-) |
| 12:34.44 | stix | russellb, yes I have recommended that to my customer, but he found something cheaper :) |
| 12:34.59 | stix | and I have never tried Beronet's products |
| 12:35.18 | russellb | stix: tell the customer that Digium makes Asterisk, so you're going to get the best experience with DIgium products |
| 12:35.21 | russellb | works there by the way |
| 12:35.28 | *** join/#asterisk chuckf (~chuckf@ubuntu/member/chuckf) |
| 12:36.17 | stix | russellb, yes I told him that. A distributor of Beronet told my customer that Beronet is making the components for Digiums cards. |
| 12:36.24 | drmessano | O.o |
| 12:36.33 | stix | but I doubt that :) |
| 12:36.54 | russellb | no.. |
| 12:37.31 | drmessano | So there's your out |
| 12:37.42 | nemith | stix: who uses BRI? |
| 12:37.45 | drmessano | The distributors of said cards are a bunch of liars |
| 12:37.52 | russellb | nemith: all of Europe, heh |
| 12:38.01 | nemith | well most... |
| 12:38.21 | russellb | every last one of them! |
| 12:38.30 | stix | nemith, why do you think, that Digium, Sangoma, Beronet and others make BRI cards? :) |
| 12:39.00 | Naikrovek | holy shit nemith is in here |
| 12:39.01 | Naikrovek | whoa |
| 12:39.11 | Naikrovek | now i wonder if he's always been in here |
| 12:39.49 | stix | Has anyone actually tried a card from Beronet? |
| 12:39.53 | Katty | pokes Naikrovek |
| 12:39.58 | stix | anyone in here I mean :) |
| 12:40.15 | Naikrovek | hugs Katty. |
| 12:40.16 | Katty | stix: i have not. i tend to avoid items i can't get support on, should i need it. |
| 12:40.24 | Katty | hugs on Naikrovek |
| 12:40.32 | fauxalliance | wonders why Alan T hasn't recorded prompts yet,,, "you, you and you, in queue, you and you, on hold. " |
| 12:40.33 | stix | Katty, yes okay |
| 12:40.38 | drmessano | Yeah, and since the Beronet distributors are a bunch of liars... |
| 12:40.57 | Katty | stix: a lack of support gives me heart burn. it gives my clients heart burn too |
| 12:41.07 | ectospasm | the Beronet guy I talked to earlier seemed nice enough |
| 12:41.29 | Katty | if only nice paid the bills :< |
| 12:41.40 | ectospasm | I dunno, I wasn't calling for support (-; |
| 12:41.44 | drmessano | ectospasm: Which side are you on, anyway? |
| 12:41.51 | Katty | who says he has to take sides. |
| 12:41.53 | ectospasm | drmessano: I work for Digium (-; |
| 12:41.56 | drmessano | I just did |
| 12:42.04 | Katty | well you don't count ;P |
| 12:42.21 | fauxalliance | counts drmessano twice |
| 12:42.21 | Katty | drmessano: <3 |
| 12:42.38 | drmessano | [08:36] <stix> russellb, yes I told him that. A distributor of Beronet told my customer that Beronet is making the components for Digiums cards. <--- Sounds like they are the enemy and must be stopped |
| 12:43.02 | drmessano | I recommend a full boycott |
| 12:43.12 | ectospasm | sounds like someone is misinformed. Don't attribute to malice what should be considered incompetence |
| 12:44.06 | drmessano | Tell that to Stallman |
| 12:44.12 | ectospasm | or however the quote goes |
| 12:44.24 | Katty | i agree with ectospasm |
| 12:44.30 | Katty | the benefit of the doubt should be given. |
| 12:44.42 | drmessano | Katty: You don't count |
| 12:44.50 | drmessano | Katty: Burn. |
| 12:44.50 | fauxalliance | :D |
| 12:44.51 | Katty | that's okay. i volunteer my opinions anyway. |
| 12:45.14 | drmessano | :) |
| 12:45.43 | [TK]D-Fender | Everyone is entitled to my own opinion ;) |
| 12:46.02 | ectospasm | "If I'd wanted your opinion, I'd have given it to you." |
| 12:46.09 | Katty | okay so on a lighter note |
| 12:46.18 | Katty | my mom's apple tree was overloading, so i brought some apples home |
| 12:46.36 | ectospasm | Katty: did you install Android on any of them? |
| 12:47.08 | Katty | should i make regular ole apple pie, or this new apple pie recipe i found which is..... http://www.food.com/recipe/yummy-crunchy-caramel-apple-pie-31128 |
| 12:47.41 | drmessano | Best thing you can do with an overloaded Apple is hit Open-Apple, Closed-Apple, then press and hold the power button for 5 seconds |
| 12:48.22 | francispereira | ectospasm, [TK]D-Fender drmessano I have another context when incoming calls are forwarded to a mobile number . All those calls work. Do you think it would be a sipnat problem ? |
| 12:48.46 | [TK]D-Fender | francispereira: Dialplan has nothing to do with netowrking issues |
| 12:50.33 | rethus1 | how can i list event-Listeners that add via a script to AGI? is there a way on the cli to see them? |
| 12:50.47 | francispereira | [TK]D-Fender, but those calls work. They come in to the PRI on to the media gateway and media gateway sends it to the asterisk server from where the server forwards it to a mobile number which is sent back to the media gateway and on the the PRI. Infact there is no firewall inbetween at all. Maybe i am wrong |
| 12:51.36 | [TK]D-Fender | francispereira: I see no configs, no proper description of the call path and most importantly, no SIP DEBUG for a failed call to debug |
| 12:52.34 | [TK]D-Fender | ~pb |
| 12:52.35 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood 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://bin.cakephp.org/ , http://asterisk.pastey.net/ , or install pastebinit with yum or aptitude. |
| 12:52.36 | [TK]D-Fender | ^^^^^^^^^^^^^^ |
| 12:53.16 | *** join/#asterisk rgsteele (~rgsteele@apo.aweber.com) |
| 12:53.37 | *** join/#asterisk catphish (~catphish@gateway.atechmedia.net) |
| 12:53.51 | catphish | is is possible to disable Packet2Packet in 1.4? |
| 12:54.04 | catphish | canreinvite=no doesn't seem to help any more |
| 12:56.38 | francispereira | :) [TK]D-Fender here is what the call log looks like http://pastebin.com/6DgKh15v |
| 12:58.33 | [TK]D-Fender | francispereira: Everything appears to be on a closed local LAN. Is this correct? |
| 12:59.07 | francispereira | yes it is |
| 13:00.17 | [TK]D-Fender | francispereira: dump your firewall |
| 13:00.40 | [TK]D-Fender | (s) |
| 13:00.43 | Naikrovek | ahh, yeah: http://twitter.com/kellyoxford/status/25623360870 |
| 13:00.53 | c0rnoTa | 'm using E1 channel to connect to PSTN. Sometimes I get debug messages "Got a FRAME_CONTROL (8) frame on channel DAHDI/". I receive this when call is active and people speaking. The call drops when this message was received. My telco said, that my PBX disconnect the call (send Disconnect frame with cause 16 - normal clear). russellb siad: "control frames are call signaling information ... 8 is Congestion" So, the question is: What is the initiator of t |
| 13:01.03 | francispereira | i did a iptables -F |
| 13:03.13 | [TK]D-Fender | francispereira: And on whatever system is running X-lite. |
| 13:03.29 | [TK]D-Fender | francispereira: What about INBOUND from that AudioCodes? |
| 13:03.45 | *** join/#asterisk SuPrSluG (~SuPrSluG@host-64-179-106-158.ind.choiceone.net) |
| 13:04.45 | elred_ | c0rnoTa, check http://asteriskpbx.ru/wiki/Q931-cause-codes |
| 13:06.02 | c0rnoTa | elred_: please look to this http://pastebin.mozilla-russia.org/107089 |
| 13:06.19 | c0rnoTa | elred_: line 81 |
| 13:06.45 | francispereira | in bound calls are forwarded to the same server . all inbound calls are forwarded back out through the media gateway, here is my extensions.conf http://pastebin.com/mXsHtruL |
| 13:07.10 | francispereira | x-lite is running on windows 7 |
| 13:07.17 | c0rnoTa | elred_: after that asterisk send hangup and call drops |
| 13:07.21 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 13:07.21 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 13:08.28 | elred_ | c0rnoTa, sorry i am not an expert, i don't know what exactly is the FRAME CONTROL (8) that trigger the hangup |
| 13:08.39 | elred_ | maybe someone here will be able to tell you |
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| 13:12.18 | c0rnoTa | elred_: russellb said: control frames are call signaling information ... 8 is *looks it up...* Congestion. |
| 13:12.55 | c0rnoTa | elred_: thanks for annention |
| 13:12.59 | c0rnoTa | attention |
| 13:13.13 | [TK]D-Fender | francispereira: Add "canreinvite=no" to your gateway peer and retest |
| 13:13.26 | [TK]D-Fender | francispereira: And again, your DIALPLAN has nothing to do with this. |
| 13:14.31 | francispereira | I wanted to show you how inbound calls are handled :) |
| 13:14.42 | francispereira | let me try canreinvire=no |
| 13:15.37 | catphish | i am really confused about canreinvite vs Packet2Packet bridging |
| 13:15.50 | catphish | setting canreinvite=no seems to enable p2p on my system :( |
| 13:16.18 | *** join/#asterisk cesar_CR (~cesar@201.192.86.30) |
| 13:16.43 | leifmadsen | catphish: I'm pretty sure packet2packet means it is going through asterisk still, but is not being transcoded. A native bridge is where packets going between end points directly |
| 13:16.55 | leifmadsen | if I remember correctly -- that is a little bit of a confusing area, I agree |
| 13:17.03 | catphish | ah ok |
| 13:17.22 | leifmadsen | there was a bug that had to do with packet2packet bridging a while ago and file explained it to me, which I've partially forgotten and need to document |
| 13:17.34 | francispereira | [TK]D-Fender, canreinvite=no WORKS ! |
| 13:17.41 | francispereira | hey |
| 13:17.42 | *** join/#asterisk ukine (~ukine@14-145.97-97.tampabay.res.rr.com) |
| 13:17.45 | leifmadsen | there were settings that would allow you to somewhat control how that worked (i.e. packet2packet or native, since the issue only showed with packet2packet I think) |
| 13:17.51 | francispereira | where do i find docs on options like this |
| 13:17.59 | francispereira | the book certainly doesnt have it |
| 13:18.00 | leifmadsen | sip.conf.sample |
| 13:18.01 | catphish | i'm having endless issues with silent calls on a nat |
| 13:18.24 | catphish | too many firewalls :( |
| 13:18.33 | rethus1 | Events in AMI: first i have to enable EVENTS - after this i have to tell, which events should be catched ? |
| 13:20.09 | jamko | catphish: get yourself a block of statics, and put everything on the wan. |
| 13:20.53 | catphish | jamko: that'd be nice |
| 13:21.03 | catphish | but hopefully unnecessary |
| 13:23.35 | francispereira | [TK]D-Fender, thanks ! |
| 13:23.44 | jamko | depends a lot on your firewall. |
| 13:24.54 | kuku | <PROTECTED> |
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| 13:25.13 | leifmadsen | sounds like app_meetme wasn't compiled, likely due to a lack of a DAHDI timing interface |
| 13:25.19 | fauxalliance | catphish, i have four levels of NAT with a bridging router... is that too many? |
| 13:25.25 | leifmadsen | you need to install DAHDI, even if you have no hardware |
| 13:25.50 | catphish | fauxalliance: more than one level of nat is bad design IMO |
| 13:26.57 | fauxalliance | four that work are better than one that doesn't ;-) |
| 13:27.19 | catphish | fauxalliance: how about one that does? |
| 13:27.21 | catphish | :) |
| 13:28.13 | fauxalliance | catphish, we are bridging two demarc's then dyking out two three subnets... clumsy, but hierarchically sensible |
| 13:28.30 | fauxalliance | s/two/to |
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| 13:28.58 | fauxalliance | catphish, how can I invite you to pick a router and whack it with a maul if I only have one ;-) |
| 13:29.34 | fauxalliance | that would be a 'half router, with the works |
| 13:29.37 | fauxalliance | ' ;-) |
| 13:30.17 | fauxalliance | Are we going to be living on IPv4/NAT for the rest of our lives? |
| 13:30.36 | drmessano | No, I like NATing IPv6 |
| 13:30.54 | catphish | painful |
| 13:31.07 | catphish | someone should write into the IPv6 protocol that NAT is banned |
| 13:31.22 | fauxalliance | let's try multiplexing multiple transports over a single TCP connection ;-) |
| 13:31.33 | kuku | leifmadsen: so I have to recompile in order to get it working ? |
| 13:31.51 | [TK]D-Fender | kuku: Yes |
| 13:32.41 | fauxalliance | There is no shortage of IPv4 addresses, because NAT! This is stymieing the adoption of the protocol. |
| 13:34.10 | rethus1 | Events in AMI: first i have to enable EVENTS - after this i have to tell, which events should be catched ? - Right? But how can i tell which event to register ? |
| 13:35.01 | *** join/#asterisk theHub (~karl@69.177.93.21) |
| 13:36.14 | [TK]D-Fender | rethus1: HUH? You get ALL EVENTS. You don't get to pick WHICH. |
| 13:36.24 | [TK]D-Fender | rethus1: It is yoru job to parse out the ones you care about |
| 13:36.39 | rethus1 | ah, ok, thanks |
| 13:37.10 | leifmadsen | kuku: well you need DAHDI installed, then you need to look in menuselect and make sure app_meetme is selected, then you need to compile the module and install it -- yes. |
| 13:37.27 | underdog | https://33ad.org/tmp/down/a1jsvZ/cg-enterprise-phone-systems.pdf <-- Enterprise Phone Systems Comparison Guide...has some cost numbers in it...metions open-source but no costs associated with it |
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| 13:38.06 | underdog | https://33ad.org/tmp/down/ryBxH8/bg-enterprise-phone-systems.pdf <-- buyer's guide from same group |
| 13:38.12 | [TK]D-Fender | underdog: Of course not.. who makes their own solution look bad? :p |
| 13:38.30 | fauxalliance | [TK]D-Fender, Kool-Aid? |
| 13:38.42 | [TK]D-Fender | fauxalliance: DO NOT DRINK! |
| 13:39.52 | Katty | stretches |
| 13:39.59 | Katty | pamples [TK]D-Fender |
| 13:41.03 | bougyman | it's not like the asterisk website mentions much about freeswitch or yate or opensips or ... or anything but asterisk. |
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| 13:41.43 | fauxalliance | bougyman, don't hear much about the Baptists down the the Episcopal either... |
| 13:42.29 | bougyman | fauxalliance: sure you do. |
| 13:43.45 | fauxalliance | bougyman, sotto voce, perhaps |
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| 13:45.44 | fauxalliance | [TK]D-Fender, the Flavor Aid is the one that bites. |
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| 14:06.11 | c0rnoTa | Guys, where i can get answer for my question? The question is: Why channel.c write my "Got a FRAME_CONTROL (8) frame on channel DAHDI/..." When there was no frame from telco received? |
| 14:07.43 | c0rnoTa | After that message call drops. |
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| 14:47.41 | NiugeS | hi all.. using trixbox and was wondering if someone could assist.. I am looking for some help.. At the moment, if for example extension 1003 answers the phone to 0208 123 123, and then transfers the call to 1001, the call is logged as a call between 1001 and 1003. This provides false figures on the reports we generate using the CDR. any suggestions? |
| 14:48.35 | CajaC[a]liente | I have a very similar probably |
| 14:49.48 | *** join/#asterisk bluOxigen (~sfds@unaffiliated/bluOxigen) |
| 14:50.03 | c0rnoTa | NiugeS: blind transfer? |
| 14:50.52 | NiugeS | no ... |
| 14:51.01 | CajaCaliente | c0rnoTa sometimes you don't want a blind xfer |
| 14:51.09 | NiugeS | i aggree.. |
| 14:51.26 | c0rnoTa | NiugeS: i think transfer method is the root of the problem. Have similar too. Because the uniqueid changes too |
| 14:51.36 | NiugeS | but i have just noticed the same problem with the ivr.. if someone calls in the call gets recorded against the ivr however an extension picked it up.. |
| 14:51.36 | c0rnoTa | sometimes |
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| 14:53.32 | NiugeS | does anyone know if there is a fix to it? |
| 14:54.12 | CajaCaliente | I was just talking with wierdo a few hours ago about this very setup. he was going to try some things and get back to me |
| 14:54.18 | c0rnoTa | NiugeS: what version of asterisk-addons? |
| 14:54.49 | rethus1 | can i use "set Callerid" fot setting name and number for an incoming sip-call (sipclient like twinkle) |
| 14:54.57 | WIMPy | What version of Asterisk are you guys using? |
| 14:55.18 | CajaCaliente | 1.6.2.11 |
| 14:55.24 | c0rnoTa | 1.4.30 |
| 14:55.31 | NiugeS | c0rnoTa i'm using trixbox |
| 14:55.37 | NiugeS | or trixbox was installed |
| 14:55.46 | NiugeS | i'm not overly technical but had it instaled.. |
| 14:55.49 | WIMPy | Interesting. I don't have the problem on 1.6.2.9 |
| 14:56.23 | CajaCaliente | WIMPy my issue is kinda self inflicted but I'm unsure of how to resolve it |
| 14:56.40 | WIMPy | I get CDRs for calls from A>B and A>C with both the same ID. |
| 14:57.07 | NiugeS | WIMPy so it preserves the callerid and passes it on correctly? |
| 14:57.11 | CajaCaliente | DAHDI1-1 calls into SIP/Main wanting Alex. SIP/Main calls SIP/Alex, announces whos on the line, then transfers. This results in two separate call recordings that are hard to piece back together as well as disjointed CDR data |
| 14:57.37 | WIMPy | NiugeS: Yes |
| 14:58.50 | WIMPy | CajaCaliente: Well, it's not nice, but not hard either. Look for identical IDs in the CDRs. |
| 15:00.33 | WIMPy | Err, or just put the ID into the filenames. |
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| 15:01.30 | WIMPy | ist still in the process of waking up. |
| 15:02.01 | CajaCaliente | WIMPy negative I have different unique IDs |
| 15:02.25 | WIMPy | Hmm. |
| 15:02.40 | WIMPy | checks on an axfer... |
| 15:03.40 | eugeneoden | is there a trick to get asterisk (1.6.2.13) to propagate codec-changing sip reinvites when directrtpsetup=yes? |
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| 15:05.59 | WIMPy | CajaCaliente: Right. That only seems to work for blind transfers. |
| 15:06.41 | CajaCaliente | Yea thats what I'm getting at. I have been yelled at to use agents or someshit like that but I wasn't sure if my small setup warranted it |
| 15:07.21 | WIMPy | Mine wouldn't, but that might be an issue for me as well. |
| 15:10.09 | WIMPy | It seems rather obvious that it gets a new ID, really. Would probably be best to generate tree CDRs inthat case, I guess. |
| 15:10.23 | CajaCaliente | How would one go about doing so? |
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| 15:18.06 | areay | hi all... i'm using reportholdtime for one of my queues, i was wondering if it's possible to play additional sound files (such as which queue the call is coming from) to the agent when they answer the call... |
| 15:20.01 | WIMPy | CajaCaliente: Fiddle around in the source, I guess. |
| 15:20.31 | CajaCaliente | Its not a huge deal, my client deals with it but I'd love to solve the issue at some point |
| 15:20.59 | WIMPy | CajaCaliente: It sure seems like a good idea. |
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| 15:26.01 | rethus1 | can i use "set Callerid" fot setting name and number for an incoming sip-call (sipclient like twinkle) |
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| 15:30.51 | [TK]D-Fender | rethus1: You can change the calleriD on any call you want |
| 15:30.55 | ChannelZ | yeah |
| 15:31.30 | fauxalliance | [TK]D-Fender, on _most_ trunks. |
| 15:31.56 | [TK]D-Fender | fauxalliance: "incoming sip-call". INCOMING |
| 15:31.58 | rethus1 | [TK]D-Fender: mhh, thats not depends on the conference-plugin i use (meetme, konference )? |
| 15:32.26 | [TK]D-Fender | rethus1: Plugin? WTF is a "plug-in"? Call processing = dialplan = whatever the hell you want to do. |
| 15:33.00 | *** join/#asterisk ybit (~quassel@unaffiliated/ybit) |
| 15:34.17 | fauxalliance | rethus1, ;-) |
| 15:41.23 | *** join/#asterisk rethus (~suther@p5087E293.dip.t-dialin.net) |
| 15:41.38 | rethus | <PROTECTED> |
| 15:42.00 | R-Guy | Anyone know if there is still a Spanish Asterisk IRC channel? |
| 15:44.14 | fauxalliance | R-Guy, #elastix-es |
| 15:44.34 | R-Guy | fauxalliance: Thanks. |
| 15:45.30 | [TK]D-Fender | rethus: that has NOTHIGN to do with setting callerid. |
| 15:46.12 | fauxalliance | s/any call/any sip call/gc |
| 15:51.01 | *** join/#asterisk FlashDeluxe (~benedict@static-87-79-94-28.netcologne.de) |
| 15:52.29 | rethus | <PROTECTED> |
| 15:55.07 | FlashDeluxe | hi! i have got problems with my asterisk server, if i dial out several numbers, the error appears, that i get a busytone altough the user at the end is not phoning. My sip telephone says "error 503" and dmesg shows the following "" |
| 15:55.08 | FlashDeluxe | [35562.616719] qozap: Unassigning channel 0/2, timeslot 2! |
| 15:55.09 | FlashDeluxe | [35651.047597] qozap: Unassigning channel 0/2, timeslot 2! |
| 15:55.40 | FlashDeluxe | i am using Asterisk 1.4.21.2-BRIstuffed-0.4.0-RC3c |
| 15:56.04 | Qwell | Upgrade. Don't use BRIstuffed. |
| 15:57.22 | FlashDeluxe | why? |
| 15:57.25 | *** join/#asterisk aidinb (~Aidin@unaffiliated/aidinb) |
| 15:59.45 | fauxalliance | I presume Qwell means that the 'BRI-stuffing' kludge is not required in future (current?) releases |
| 16:01.45 | Qwell | No, I'm saying that it breaks a lot of things, and will likely block any upgrade paths. |
| 16:03.42 | *** join/#asterisk eugeneoden (~goden@rrcs-97-77-255-86.sw.biz.rr.com) |
| 16:04.14 | fauxalliance | To elaborate further, the BRI support in Asterisk 1.6 uses the _same directives_ as bristuff used, and thus his migration will not be too painful. |
| 16:05.05 | drmessano | I like a good BRI-stuffed burrito |
| 16:05.16 | *** join/#asterisk iscario (~quassel@24.244.71-86.rev.gaoland.net) |
| 16:05.23 | WIMPy | prefers feta. |
| 16:07.24 | rethus | [TK]D-Fender: did u now the application konference as replacement of meetme? if i do konference list, as user i have only numbers. |
| 16:07.43 | rethus | i search a way to set the usernames there, but with set callerid, it seems not to work. |
| 16:07.49 | FlashDeluxe | thanks for help.. |
| 16:07.50 | rethus | with meetme it worked like a charm |
| 16:07.54 | *** part/#asterisk FlashDeluxe (~benedict@static-87-79-94-28.netcologne.de) |
| 16:07.57 | iscario | hello, i would like to know what is the difference between monitor and mixmonitor (except that mixmonitor is able to launch a script after the recording...) |
| 16:10.08 | fauxalliance | iscario, you can call one in a script |
| 16:10.34 | fauxalliance | the other in dialplan. |
| 16:11.58 | iscario | fauxalliance: so in the dialplan, the role is the same right ? (because both exist as a dialplan application) |
| 16:13.13 | *** join/#asterisk BesticlesWork (~larry@209-58-227-178.static-ip.telepacific.net) |
| 16:13.15 | fauxalliance | iscario, monitor also keeps seperate files for input and output... mixmonitor, gues what, mixes them. |
| 16:13.26 | *** part/#asterisk WWGD (~WWGD@208.79.14.130) |
| 16:13.26 | iscario | ok thanks fauxalliance |
| 16:14.07 | *** join/#asterisk WWGD (~WWGD@208.79.14.130) |
| 16:14.26 | iscario | so i have another question : i can't record a confBridge conference with mixmonitor, is it normal ? |
| 16:18.43 | fauxalliance | iscario, normal, is calling meetme with 'r' — Record conference (records as ${MEETME_RECORDINGFILE} using format ${MEETME_RECORDINGFORMAT}). |
| 16:19.14 | iscario | i do not use meetme fauxalliance, but confBridge |
| 16:19.26 | fauxalliance | my ignorance |
| 16:19.29 | iscario | i did not install dahdi so |
| 16:19.30 | fauxalliance | ? |
| 16:19.43 | fauxalliance | there's your (potential) timing issue |
| 16:20.32 | BesticlesWork | Has anyone deployed asterisk as a outbound dialer here? I am pricing out two dell servers, each will have a 1x1TE412PF. I have never seen asterisk in production, so I don't know if I should put emphasis in CPU & HDD power. |
| 16:20.59 | iscario | timing ? in fact in each time the record stop when confBridge is called fauxalliance |
| 16:21.19 | fauxalliance | BesticlesWork, how many calls per agent? |
| 16:21.54 | fauxalliance | iscario, whats wrong with meetme? |
| 16:22.52 | iscario | i did not try meetme, my question was specific with confbridge, that's why I asked. I guess i'll have to install meetme :) |
| 16:23.03 | iscario | fauxalliance: thx! |
| 16:23.05 | fauxalliance | iscario, or get a 'proxy' to record for you |
| 16:23.27 | iscario | oh, which proxy are you talking abt ? fauxalliance |
| 16:24.00 | fauxalliance | a sony beta cam for all asterisk cares |
| 16:24.59 | *** join/#asterisk Naikrovek (~jjohnson@unaffiliated/naikrovek) |
| 16:29.24 | fauxalliance | BesticlesWork, unsolicited PM's are just a rude as your planned application. "We basically blast the crap out of call lists non stop..." nice |
| 16:29.41 | fauxalliance | hands BesticlesWork a cookie. |
| 16:31.25 | drmessano | Buying servers before ever having used Asterisk.. Hmmm |
| 16:31.56 | [TK]D-Fender | drmessano: I'd like some fries with that please... |
| 16:32.13 | drmessano | Indeed! |
| 16:33.20 | fauxalliance | p00t33n |
| 16:34.13 | *** join/#asterisk bluOxigen (~sfds@unaffiliated/bluOxigen) |
| 16:34.59 | fauxalliance | Lola “Lola, gehst du einkaufen ? Ich brauch Servers!” |
| 16:35.41 | *** join/#asterisk ManxPower (~manxpower@user-24-214-153-32.knology.net) |
| 16:35.55 | *** join/#asterisk Defraz (~Defraz@corp.fuzecore.com) |
| 16:36.38 | ManxPower | I have callevents=yes in sip.conf [general] (Asterisk 1.4.x) and I am not getting a Hold event when a call is put on hold from a phone (Polycom). When put on hold, the device shows Idle instead of Hold. Has anyone had this issue and if so, how did you solve it? |
| 16:37.54 | BesticlesWork | Faux, we're a collection agency. That's what collection agency does. |
| 16:39.00 | fauxalliance | BesticlesWork, I just set up a blacklist for an abortion clinic, we're not debating who's going to hell, just that PM's (unsolicited) are rude, and don't expect too much 'free' help for your venture. |
| 16:39.45 | Qwell | BesticlesWork: No, that's what shadily operated collection agencies do. |
| 16:40.19 | fauxalliance | ~karma |
| 16:40.19 | infobot | fauxalliance has karma of 1 |
| 16:40.45 | *** join/#asterisk Naikrovek (~jjohnson@unaffiliated/naikrovek) |
| 16:41.48 | BesticlesWork | I see. Well, I apologise for the "unsolicited" PM. I haven't used IRC before until I started working on this project. I wasn't aware that the use of PM's are rude. |
| 16:41.57 | Naikrovek | ah well |
| 16:42.02 | Naikrovek | we all learn from our mistakes |
| 16:42.15 | fauxalliance | Qwell, once, one of these collection types kept calling, wondering whey she was getting her personal mobile every time she called ;-) |
| 16:42.23 | Qwell | BesticlesWork: PMing somebody without their permission is just like cal...oh wait. |
| 16:42.35 | WIMPy | It obviousely depends on where you are in IRC land. |
| 16:42.47 | Naikrovek | yeah |
| 16:42.53 | Naikrovek | in a lot of places it's normal |
| 16:42.58 | Naikrovek | but not here |
| 16:42.59 | ManxPower | I thought faking your Caller*ID to fale your identity was illegal in the USA. |
| 16:43.01 | fauxalliance | note's that this sure isn't DALnet |
| 16:43.02 | Naikrovek | freenode ingeneral actually |
| 16:43.26 | p3nguin | There's an act on caller ID spoofing, but I don't know that it ever made it into law. |
| 16:45.13 | Naikrovek | ipv6 prevents spoofing, wonder if we can use that to get around faked callerid somehow |
| 16:45.53 | fauxalliance | Naikrovek, sounds feasable... picture 3g mobile network, hardware serial / DID pairs. |
| 16:46.03 | fauxalliance | spoof that! |
| 16:46.26 | p3nguin | It's not hard. |
| 16:46.54 | p3nguin | It's rather easy to change the IMEI or MEID on most phones. |
| 16:47.22 | *** join/#asterisk Defraz (~Defraz@corp.fuzecore.com) |
| 16:47.28 | fauxalliance | p3nguin, not hard at all... can I ask how you update the pair with the carrier? |
| 16:47.51 | fauxalliance | I.E. you can't do your own ESN swapout. |
| 16:47.54 | p3nguin | There's no reason to update anything with the carrier. They've already made the associations. |
| 16:48.16 | fauxalliance | tangential...just a thought |
| 16:49.13 | *** join/#asterisk bmg505 (~leon@196-209-120-122.dynamic.isadsl.co.za) |
| 16:53.21 | *** join/#asterisk mpe_ (~mpe@0xd99d3f8f.customer.cybercity.dk) |
| 16:55.00 | *** join/#asterisk eugeneoden (~goden@rrcs-97-77-255-86.sw.biz.rr.com) |
| 17:08.14 | *** join/#asterisk idespinner (~idespinne@cpe-76-94-74-210.socal.res.rr.com) |
| 17:09.06 | idespinner | hey folks, having a brain fart here; whats the asterisk app that records dtmf digits into a variable? |
| 17:09.19 | [TK]D-Fender | idespinner: Read() |
| 17:09.40 | idespinner | [TK]D-Fender, thank you, that is it! |
| 17:10.25 | *** join/#asterisk fullstop (~fullstop@64-121-41-67.c3-0.tlg-ubr1.atw-tlg.pa.cable.rcn.com) |
| 17:10.46 | fauxalliance | drmessano, http://memegenerator.net/Y-U-NO/ImageMacro/2797270/You-can-upgrade-away-from-14-Ill-ALWAYS-have-a-face-that-looks-like-an-arse |
| 17:10.55 | kaldemar | m |
| 17:11.16 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 17:11.16 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 17:11.32 | drmessano | lol |
| 17:13.20 | fauxalliance | [TK]D-Fender, http://memegenerator.net/Jared-Leto/ImageMacro/2797334/WTF-Cant-you-see-my-ClueBat |
| 17:13.49 | drmessano | fauxalliance, http://memegenerator.net/SGTHARTMAN/ImageMacro/2797335/TRIXBOX-IS-FOR-SISSIES |
| 17:13.51 | Katty | herroes. |
| 17:16.01 | Qwell | sneakattacks Katty |
| 17:16.29 | fauxalliance | drmessano, http://memegenerator.net/Fat-Guy-at-Computer/ImageMacro/2797384/Hey-Fella-Wanna-see-my-softphone |
| 17:18.37 | *** join/#asterisk citrus (~citrus@72.215.183.28) |
| 17:19.37 | drmessano | haha |
| 17:20.25 | *** join/#asterisk ruyo (~psantos@a81-84-7-119.cpe.netcabo.pt) |
| 17:20.30 | citrus | hey all, i was hoping to see if anyone could recommend a good 1 800 provider. with call quality and reliability being a focus. company is in USA and we expect mainly USA customers calling |
| 17:22.18 | fauxalliance | drmessano, http://memegenerator.net/stupid-bitch/ImageMacro/2797492/I-Used-TRIXBOX-Now-Ill-never-be-like-barbie "see also; "I was told, I didn't listen" |
| 17:23.25 | *** part/#asterisk rethus (~suther@p5087E293.dip.t-dialin.net) |
| 17:23.38 | fauxalliance | and ftfw, http://memegenerator.net/HNNNNNNGGG/ImageMacro/2797520/What-does-this-have-to-do-with-e911-Im-having-a-fucking-hart-attack |
| 17:24.52 | *** join/#asterisk s34n (~chatzilla@ip-208-76-93-125.mvdsl.com) |
| 17:25.21 | s34n | is anybody here doing realtime? |
| 17:25.41 | p3nguin | citrus: VoIP.ms, Flowroute |
| 17:27.52 | drmessano | http://tinyurl.com/2ax3h7q <-- FTW |
| 17:32.10 | *** join/#asterisk ironm (~ironm@84-73-66-195.dclient.hispeed.ch) |
| 17:35.06 | citrus | p3nguin: Thank you |
| 17:35.26 | citrus | i take it trixbox is not very loved in here |
| 17:35.43 | Katty | drmessano: cute. |
| 17:35.43 | Naikrovek | not really, no |
| 17:35.45 | *** join/#asterisk slidesinger (~slidesing@c-68-44-99-50.hsd1.nj.comcast.net) |
| 17:36.05 | Katty | Naikrovek: did you see the pie i'm making tonight? |
| 17:36.18 | Naikrovek | i did not. been fighting issues all day |
| 17:36.21 | *** join/#asterisk BANSAL (~bansal@117.199.120.88) |
| 17:36.41 | citrus | we used trixbox as a pbx at our work. i hate it. as well |
| 17:36.58 | Katty | Naikrovek: http://food.sndimg.com/img/recipes/31/12/8/large/pic6B8FwT.jpg <- Apple pie with a crumb topping, pecans, and caramel sauce |
| 17:38.07 | Naikrovek | ooh |
| 17:38.18 | Naikrovek | not a huge fan of the pecans but looks excellent anyway |
| 17:38.29 | Katty | Naikrovek: not a fan of nuts at all? |
| 17:38.32 | Katty | Naikrovek: in pie |
| 17:38.42 | Naikrovek | not in pie really, no |
| 17:38.43 | Naikrovek | well |
| 17:38.44 | Naikrovek | that's not true |
| 17:38.50 | Naikrovek | maybe just not on top |
| 17:39.01 | Naikrovek | despite my extensive eating career i've not eaten a lot of pie |
| 17:39.08 | [TK]D-Fender | Pecan pie > YOU |
| 17:39.11 | fullstop | How was the Pizza? |
| 17:39.11 | Katty | ah right. |
| 17:39.16 | Katty | fullstop: well. |
| 17:39.34 | Katty | fullstop: it /would/ have been all right i think, but the friends i made it for decided they wanted two pizzas rather than one deep dish |
| 17:39.39 | Katty | fullstop: so it kinda messed it all up |
| 17:39.43 | fullstop | =/ |
| 17:39.51 | Katty | eh, i'll try again |
| 17:40.20 | fullstop | I pickled some jalapeno pepper slices, but I won't know if they are any good for about a month. |
| 17:40.46 | fullstop | We didn't get much rain this season, so they were very hot. |
| 17:40.47 | Katty | hrmm. not a fan of those pickled |
| 17:40.52 | Katty | prefer them freshly sauted. |
| 17:41.03 | Katty | you ever made jelly with them? |
| 17:41.06 | Katty | preserves. whatever. |
| 17:41.08 | *** join/#asterisk deonv (~adium@41.218.75.152) |
| 17:41.10 | fullstop | It's going to frost soon, so I had to use them up. |
| 17:41.32 | fullstop | No.. I've made poppers with them, but my wife hates the way it makes the house smell after frying. |
| 17:41.55 | Katty | ah, yeah...i get that |
| 17:43.04 | s34n | I'm looking for advice on the best way to maintain a phone directory |
| 17:43.18 | p3nguin | Is there no way to get FollowMe() to NOT say "Please hold while I try to locate the person you are calling."? |
| 17:43.27 | bmoraca_work | s34n: i've always used an excel spreadsheet... |
| 17:43.31 | Katty | s34n: ftp. |
| 17:43.42 | fullstop | secretary |
| 17:43.43 | s34n | I would like unskilled people to be able to add and remove numbers, voicemail boxes etc. |
| 17:43.50 | Katty | s34n: excell. |
| 17:43.50 | bmoraca_work | p3nguin: script it yourself :) |
| 17:43.54 | Naikrovek | simple webapp |
| 17:44.08 | Katty | that would work too |
| 17:44.16 | p3nguin | An option would be nice for that. |
| 17:44.34 | s34n | I can build a simple app to parse the * conf files and present them for editing, but I was thinking more along the lines of ldap |
| 17:44.43 | s34n | or something sql-based |
| 17:44.56 | bmoraca_work | p3nguin: it's not hard using astdb and local extensions...it can be as complex as you want it, though |
| 17:45.09 | s34n | so I started looking at realtime stuff |
| 17:45.26 | s34n | I was hoping for some perspective from veterans |
| 17:45.35 | fullstop | bmoraca_work: I was able to get queues up and running with your advice the other day -- thanks. |
| 17:45.56 | bmoraca_work | s34n: you can build whatever you want. it doesn't necessarily have to tie in to the asterisk configs. you can build it completely standalone |
| 17:46.02 | bmoraca_work | fullstop: nice. glad i could help. |
| 17:46.38 | s34n | bmoraca_work: but I want it to affect asterisk |
| 17:46.54 | bmoraca_work | fullstop: it would be pretty awesome, though, if you could poll device state across servers without employing AMI |
| 17:47.11 | fullstop | It really feels like IAX2 should be able to do this. |
| 17:47.12 | bmoraca_work | s34n: well, that's a very tall order which goes far beyond "company directory" |
| 17:47.20 | Naikrovek | bmoraca_work: you can with hints after 1.6.2.0 i think |
| 17:48.13 | fullstop | Naikrovek: only at the server level. That is, you can specify IAX2/remote/2611 but it only does the hint for IAX2/remote |
| 17:48.14 | bmoraca_work | Naikrovek: it seems to me that asterisk should be able to subscribe to hints on a remote system. not sure if it actually can, though. |
| 17:48.19 | s34n | bmoraca_work: I was hoping for not so tall |
| 17:48.32 | Naikrovek | there was an update that made it work for BLF |
| 17:48.37 | Naikrovek | i'm sure people in here are using it |
| 17:48.41 | Naikrovek | they should speak up |
| 17:48.54 | fullstop | I am very interested in this if it is true for 1.6 |
| 17:48.56 | bmoraca_work | s34n: you're talking about something like freepbx or other PBX management frontends. |
| 17:50.00 | *** join/#asterisk jaskew (~jdaskew@netblock-75-79-182-110.dslextreme.com) |
| 17:50.41 | bmoraca_work | i'm still not sure it's possible...i don't really see anything in the configs about asterisk subscribing to other servers |
| 17:50.43 | Naikrovek | fullstop: 1.6.1 and later. |
| 17:50.46 | Naikrovek | fullstop: http://svn.digium.com/svn/asterisk/trunk/doc/distributed_devstate.txt |
| 17:50.54 | Naikrovek | took me like 45 seconds to find that |
| 17:51.30 | bmoraca_work | that's more complicated than GROUP_COUNT |
| 17:51.46 | fullstop | Naikrovek: Sorry, I was looking to do it without a new daemon. |
| 17:52.06 | Naikrovek | not a new daemon |
| 17:52.10 | fullstop | One leg of my system is on embedded hardware, and adding such things is difficult. |
| 17:52.12 | fullstop | openais? |
| 17:52.39 | Naikrovek | ah |
| 17:52.44 | Naikrovek | hadn't read that far yet |
| 17:52.49 | Naikrovek | well i guess you're fucked then |
| 17:52.56 | Naikrovek | :) |
| 17:53.05 | fullstop | haha |
| 17:53.05 | bmoraca_work | nah, he got it working for what he needed |
| 17:53.14 | bmoraca_work | or, rather, what he told me he needed :) |
| 17:53.52 | fullstop | yes, it's working... but I'd always prefer a more simple solution. |
| 17:54.16 | Naikrovek | you can also do it over XMPP |
| 17:54.18 | Naikrovek | http://svn.digium.com/svn/asterisk/trunk/doc/distributed_devstate-XMPP.txt |
| 17:54.28 | Naikrovek | you may or may not already have an XMPP server running |
| 17:54.55 | Naikrovek | they're very particular about which server you use though... |
| 17:55.07 | Naikrovek | thought XMPP = XMPP but is willing to be corrected |
| 17:56.42 | Naikrovek | and omg the manual registration |
| 17:56.44 | bmoraca_work | fullstop: you're really only adding 3-4 lines of code...which can be completely generalized so that you only have to add them once. yeah, it's an extra step, but it's not that bad |
| 17:58.14 | fullstop | bmoraca_work: Using a local context has some downsides, but I'm not sure how negatively they will impact me. For example, "show queue qname" shows agents called over the local context as not busy, and the SLA numbers will never calculate correctly. |
| 17:59.00 | *** join/#asterisk Kant (be12315d@gateway/web/freenode/ip.190.18.49.93) |
| 17:59.07 | *** join/#asterisk fifer (~fifer@67.208.108.228) |
| 17:59.20 | fullstop | That is, it shows "in use" only when the phone is ringing. Once it is answered, it goes back to "Not in use" |
| 17:59.49 | bmoraca_work | interesting |
| 17:59.55 | bmoraca_work | that sounds like it should be a bug |
| 18:00.21 | bmoraca_work | or at least "not fully implemented" yet |
| 18:00.49 | fifer | I'm looking for some pointers to troubleshooting fax reception. I have * 1.6.2.13 with fax support compiled in, and Spandsp 0.0.6pre17 |
| 18:01.34 | fifer | I'm getting a success rate of about 1 out of 20. I can see app_fax through debugging keep training down to 2400 bps but still failing. |
| 18:02.10 | bmoraca_work | fifer: what is your transmission medium? |
| 18:02.21 | fifer | The one success was fairly poor quality. Just trying to figure out where to go from here. I'm terminating in * from 2 PRI T-1's |
| 18:02.26 | fifer | No sip involved |
| 18:02.38 | bmoraca_work | might have an echo cancellation issue |
| 18:02.58 | bmoraca_work | echo cancellation should be off for faxing |
| 18:03.11 | ManxPower | The answer to my question is "callevents = yes is documented as telling Asterisk to generate Hold/Unhold Event: in 1.4.x, but it does not actually work" |
| 18:03.28 | *** part/#asterisk Kant (be12315d@gateway/web/freenode/ip.190.18.49.93) |
| 18:03.33 | fifer | I have mg2 set for all the chanels, how can I turn it off for incomming fax? |
| 18:03.50 | ManxPower | fifer, the fax tone will disable the EC. This is per ITU specifications. |
| 18:03.56 | fifer | ah |
| 18:04.17 | ManxPower | no faxes would work on the global PSTN if this was not the case. |
| 18:04.20 | *** join/#asterisk Letoric (~Letoric@tpi-dfw-uc-f1-69-94-238-22.totalprocess.net) |
| 18:04.25 | s34n | bmoraca_work: freepbx is an interesting project, but I would really just like to tie asterisk into my existing provisioning system |
| 18:04.27 | fifer | Right, makes sense |
| 18:05.10 | bmoraca_work | fifer: still might consider checking the dahdi channel when a fax is supposed to be coming in to verify that EC is actually off on it |
| 18:05.28 | fifer | How can I check that? |
| 18:05.29 | s34n | I don't want to maintain my asterisk conf files separate from my normal IT systems if I can avoid it |
| 18:05.36 | fifer | cli? |
| 18:05.44 | bmoraca_work | fifer: "dahdi show channel" in the CLI while a fax is coming in |
| 18:05.49 | bmoraca_work | on that channel |
| 18:05.49 | fifer | thanks! |
| 18:05.51 | fifer | I'll check |
| 18:06.16 | s34n | So I can have my provisioning system rewrite any affected conf files whenever anything changes (ughh) |
| 18:06.17 | bmoraca_work | if the system is overloaded, it might not accurately detect the fax tone |
| 18:06.40 | s34n | or I can try to get asterisk to tie to a common ldap or sql backend |
| 18:08.03 | bmoraca_work | s34n: i believe there was an attempt to get asterisk realtime to work with ldap, but I don't think it was successful. you can have realtime work with a SQL backend (though I was unsuccessful in getting it to work with MSSQL backen) though there are a host of caveats and things that could potentially not work properly |
| 18:08.04 | s34n | or, perhaps there is an api that lets me do granular modifications of asterisk running config without having to bound entire subsystems |
| 18:08.50 | s34n | s/bound/bounce/ |
| 18:09.31 | bmoraca_work | s34n: you can. "sip reload" will reload the sip configurations. "dialplan reload" will reload the dialplan. Both are non-destructive to existing channels. |
| 18:11.27 | fifer | It is always trying to recieve the fax, app_fax always is involved trying to train up, but I checked and EC did NOT get turned off on the active channel the fax was on |
| 18:11.53 | fifer | It did indicate "Fax Handled: yes" but EC was on |
| 18:12.01 | bmoraca_work | s34n: you can also try and use extconfig to store static configurations in an ODBC database |
| 18:12.40 | fifer | Is there a way to manually turn off EC for a channel for one call? Just to test |
| 18:13.30 | fifer | CPU load is very light |
| 18:13.53 | bmoraca_work | i believe faxdetect in chan_dahdi.conf might be necessary |
| 18:14.02 | Deeewayne | fifer, check chan_dahdi.conf and look for 'faxbuffers' |
| 18:14.33 | *** join/#asterisk Tim_Toady (~fuzzy@77.49.122.124.dsl.dyn.forthnet.gr) |
| 18:15.36 | fifer | faxdetect=incoming, no "faxbuffers" |
| 18:17.17 | Deeewayne | enable the fax debug, set verbosity high (10), and pastebin the output |
| 18:19.03 | fifer | Here is the last one I did: http://pastebin.ca/1949752 |
| 18:20.08 | *** join/#asterisk ghenry (~ghenry@pdpc/supporter/monthlybyte/ghenry) |
| 18:20.10 | ghenry | Anyone do Ghana DIDs? |
| 18:20.18 | ghenry | ghenry@surevoip.co.uk if you can |
| 18:23.43 | fifer | is faxdetect a global option or per channel? |
| 18:25.36 | [TK]D-Fender | fifer: Always per channel |
| 18:26.51 | fifer | I'm sure my config needs some work: Here is my chan_dahdi.conf/dahdi-channels.conf: http://pastebin.ca/1949754 |
| 18:27.25 | fifer | I'm guessing I need to add faxdetect to my two T-1 span blocks |
| 18:27.54 | fifer | is reloading dahdi distructive to existing channels? |
| 18:28.04 | fifer | This is our full production system |
| 18:28.30 | fifer | I always assume it is, but thought I would check |
| 18:28.42 | bmoraca_work | i don't believe reloading the asterisk dahdi config is destructive |
| 18:28.51 | bmoraca_work | that said, i've never tried it before :) |
| 18:29.03 | fifer | :-), how brave do I feel today!! |
| 18:29.08 | Katty | stretches |
| 18:29.09 | *** join/#asterisk bsaxon (~bsaxon@12.107.149.61) |
| 18:29.17 | fifer | Well, I have to run to lunch so I can get back for a meeting |
| 18:29.25 | bmoraca_work | just tell them the telco hiccuped and you're on the phone with them to sort it out |
| 18:29.45 | fifer | I'll add faxdetect to my T-1 span blocks and see what that does. |
| 18:29.48 | fifer | Thanks!! |
| 18:29.52 | fifer | bbl |
| 18:29.53 | Deeewayne | fifer, use dahdi_monitor to record the audio of a fax, then use audacity and look for a gap of silence as small as 20-60 ms in the audio |
| 18:30.41 | fifer | @Deeewayne: What would that indicate? |
| 18:30.56 | Deeewayne | you fax killing gap of audio |
| 18:30.58 | Deeewayne | your |
| 18:31.33 | Deeewayne | and then set faxbuffers to the sample value. It might help |
| 18:32.30 | fifer | something like faxbuffers=>6,full |
| 18:32.31 | *** join/#asterisk diatonic (~diatonic_@mail.clearwater-research.com) |
| 18:32.51 | Deeewayne | yeah |
| 18:33.13 | fifer | Thanks! |
| 18:36.50 | R-Guy | Can anyone confirm that "Tono IME" means "MWI tone" ? |
| 18:45.22 | *** join/#asterisk uqlev (~yuriy@91.184.221.31) |
| 18:54.53 | *** part/#asterisk ManxPower (~manxpower@user-24-214-153-32.knology.net) |
| 18:57.02 | *** join/#asterisk imox1234 (~imox1234@p4FC5C510.dip0.t-ipconnect.de) |
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| 19:02.22 | *** join/#asterisk hehol (~Adium@ip-78-94-0-76.unitymediagroup.de) |
| 19:08.31 | Letoric | has anybody had difficulties with $MIXMONITOR_FILENAME not passing through correctly when executing a command as part of app_mixmonitor? |
| 19:09.04 | Letoric | I've tried using ${MIXMONITOR_FILENAME} as well as $MIXMONITOR_FILENAME and both are not providing the correct result |
| 19:09.32 | [TK]D-Fender | Letoric: And we see none of what you are attempting. |
| 19:10.00 | leifmadsen | well $MIXMONITOR_FILENAME definitely wouldn't work |
| 19:10.22 | Letoric | I thought maybe since it was at the command level at that point, it *might* |
| 19:10.26 | Letoric | but alas, I was wrong ;) |
| 19:11.23 | Letoric | [TK]D-Fender: What would be helpful for you to help me in this, the command? The output, which is blank? I'll be glad to provide anything necessary to debug |
| 19:11.45 | Letoric | I hoped it might be a simple 'oh yeah, that was deprecated and you need x' |
| 19:11.54 | [TK]D-Fender | Letoric: Your dialplan and the actual failure at CLI to acheive what you are trying to |
| 19:12.01 | bmoraca_work | wow, the elastix update mirrors are stupid slow right now |
| 19:12.01 | bmoraca_work | uhg |
| 19:12.04 | [TK]D-Fender | PASTEBIN <------------------ |
| 19:12.09 | [TK]D-Fender | ~pb |
| 19:12.09 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood 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://bin.cakephp.org/ , http://asterisk.pastey.net/ , or install pastebinit with yum or aptitude. |
| 19:12.11 | Letoric | yep. Give me a couple to pull it in |
| 19:12.28 | bmoraca_work | 30 kB/s...was downloading from the centos mirrors at over 5 MB/s...ick |
| 19:16.53 | Letoric | [TK]D-Fender: http://pastebin.com/AsMTugQQ |
| 19:17.13 | imox1234 | hello, i have problems to answer a call from 1und1 |
| 19:17.34 | imox1234 | i get all the time a error like this here [Sep 27 21:14:43] NOTICE[8485]: chan_sip.c:20118 handle_request_invite: Sending fake auth rejection for device "anonymous" <sip:493064838338@sip.1und1.de>;tag=as20d4eaba |
| 19:17.44 | imox1234 | and here is my sipdebug http://pastebin.com/zgrHQNWZ |
| 19:17.52 | imox1234 | can somebody help me what is wrong? |
| 19:18.19 | imox1234 | i have this problem only by the shit 1und1 provider sipgate works great |
| 19:20.40 | [TK]D-Fender | Letoric: I see you referencing a variable... where the hell do I see you SET IT? |
| 19:20.50 | *** join/#asterisk lauris (~la@unaffiliated/lauris) |
| 19:21.08 | Letoric | [TK]D-Fender: that variable is innate to the mixmonitor application |
| 19:21.14 | Letoric | there is no setting it |
| 19:21.34 | [TK]D-Fender | Letoric: REALLY... and what does it say it should be filled with? |
| 19:21.45 | Letoric | it says it should contain the file name for the recording |
| 19:21.51 | Letoric | which is set in the statement in that first line |
| 19:22.20 | [TK]D-Fender | Letoric: MixMonitor(${UNIQUEID}-${STRFTIME(${EPOCH},,%G%m%d_%H%M%S)}-${CALLERID(num)}.wav,,mv /var/spool/asterisk/monitor/${MIXMONITOR_FILENAME} /mnt/monitor/${MIXMONITOR_FILENAME}) <-- this does not SET your variable |
| 19:22.32 | [TK]D-Fender | Letoric: That REFERENCES it. You did NOT do a Set() <- |
| 19:22.41 | Letoric | [TK]D-Fender: •The variable MIXMONITOR_FILENAME will contain the name of the file used for recordings |
| 19:22.49 | Letoric | that is from the documentation on the mixmonitor application |
| 19:23.19 | Letoric | I'm not saying I'm doing it right, I'm sure it's an error in my method, but that is what the documentation said, thus why I was relying on it |
| 19:23.24 | [TK]D-Fender | Letoric: You have a timing misconception. it may be populated AFTER the MixMonitor command gets called. |
| 19:23.32 | WIMPy | Letoric: You are calling teh application that sets teh variable. So it can't be set at that point. |
| 19:23.36 | *** join/#asterisk weta (~aross@CPE485b390978ce-CM00222ddf42dd.cpe.net.cable.rogers.com) |
| 19:23.41 | *** join/#asterisk GoatHunter (~a@46-116-82-57.bb.netvision.net.il) |
| 19:23.54 | WIMPy | s/teh/the/g |
| 19:23.58 | Letoric | Ok, pardon my stupidity, but if the command function is part of the mixmonitor application, why would the variable not be stored at that point? |
| 19:24.29 | WIMPy | You reference it *before* starting MixMonitor. |
| 19:24.36 | Letoric | err |
| 19:24.40 | Letoric | <command> will be executed when the recording is over. Any strings matching ^{X} will be unescaped to ${X} and all variables will be evaluated at the time the application is called. Where <command> is a system (Linux shell) command, see Asterisk cmd System for example values. |
| 19:24.58 | Letoric | I guess I'm really not understanding this one |
| 19:25.04 | GoatHunter | hi, my cell provider seems to block upstream sip packets, tried changing the port, no good, I can hear the side im calling but he cant hear me, server is trixbox 2.6, any suggestions? I tried also using STUN server via the cellphone (running sipdroid) |
| 19:25.21 | Letoric | you want me to set a variable beforehand, then reference that in both places? Because it says that the command won't get the environment variables |
| 19:25.26 | [TK]D-Fender | Letoric: ${MIXMONITOR_FILENAME} <-- does that look properly escaped as per those instructions? |
| 19:25.32 | WIMPy | Ja, should work with ^{}, but not with ${}. |
| 19:26.00 | Letoric | Oh |
| 19:26.21 | Letoric | I didn't understand that part ;) |
| 19:26.53 | Letoric | I'll give that a shot and let you know. Thanks for that |
| 19:29.15 | Letoric | that worked |
| 19:29.19 | Letoric | gah, such a small thing ;) |
| 19:30.39 | Letoric | of course, now I just have to get it to actually DO what it says haha |
| 19:30.56 | Letoric | thanks again guys. As always, you're great |
| 19:31.20 | [TK]D-Fender | tosses Letoric's chickens & eggs into a pan and FRIES BOTH OF THEM. |
| 19:31.28 | [TK]D-Fender | NEXT!@@!@! (c) BKW |
| 19:31.59 | carrar | green eggs? |
| 19:33.10 | tzanger | there's a name I haven't seen in a long time |
| 19:34.01 | *** join/#asterisk ukine (~ukine@14-145.97-97.tampabay.res.rr.com) |
| 19:34.01 | Letoric | tzanger: you know Eggs? From Florida?? |
| 19:34.20 | Letoric | or you referring to Dr Seuss |
| 19:40.41 | *** join/#asterisk t_dot_zilla (~chatzilla@firewall-a.buf.ny.i-evolve.net) |
| 19:43.27 | Katty | OHAI |
| 19:44.25 | paulc | OHAI KATTY! |
| 19:44.56 | Katty | paulc: herroes. |
| 19:44.57 | t_dot_zilla | is there any tool or way of testing for call complettion in asterisk? we'd liek to add an alert to our nagios server if call does not complete |
| 19:44.59 | Katty | paulc: how're you dear |
| 19:45.22 | *** join/#asterisk wcselby (~wcselby@pdpc/supporter/professional/wcselby) |
| 19:45.25 | [TK]D-Fender | t_dot_zilla: what kind fo call? Complete what? Where is the call prior/post? |
| 19:45.26 | wcselby | o/ |
| 19:45.38 | Katty | glomps wcselby |
| 19:45.48 | wcselby | heya Katty |
| 19:45.53 | wcselby | :) |
| 19:47.21 | wcselby | so I've had a fun day. first client has an asterisk that doesn't always respond to invites from his sip provider. upped logging on that one and checking to see what's going on. next client has a phone that just randomly reboots itself during the day. another client called and said inbound calls sometimes just ring and ring a nd ring and ring, and if it does ever get through, it's really faint. |
| 19:47.31 | wcselby | first client is also running asterisk 1.2.10 |
| 19:47.33 | wcselby | which is fun |
| 19:48.06 | wcselby | but they'd rather wait to upgrade until they're ready to go to the fully latest version, instead of me doing an in-place upgrade to latest 1.2 release, just to see if that's causing the issue, etc |
| 19:48.16 | wcselby | and who knows when they'll be ready for that |
| 19:48.38 | t_dot_zilla | [TK]D-Fender: just an internal call to MoH for 3 seconds and then hang up |
| 19:48.41 | wcselby | third client has a trixbox 2.8 box, which is always fun - except I don't have any of the passwords, etc |
| 19:48.54 | wcselby | oh, and my trial version of mirc expired and wouldn't let me on |
| 19:49.06 | wcselby | but - the weather outside if perfect! |
| 19:49.12 | wcselby | so I'm overall in a good mood |
| 19:49.17 | wcselby | and talkative |
| 19:49.29 | p3nguin | People use mIRC? |
| 19:49.51 | *** join/#asterisk Defraz (~Defraz@corp.fuzecore.com) |
| 19:50.18 | *** join/#asterisk [cannibalera] (~cannibale@201-25-254-177.fnsce703.dsl.brasiltelecom.net.br) |
| 19:50.23 | wcselby | p3nguin - i used to not use it, but the client i do use doesn't notify me when someone says my name, so if I'm not looking I miss it |
| 19:50.34 | wcselby | which mirc did just fine |
| 19:50.37 | wcselby | if that makes sense |
| 19:50.43 | p3nguin | What client doesn't have hilighting? |
| 19:51.06 | p3nguin | Kvirc even has a balloon pop-up for it. |
| 19:51.09 | wcselby | vIRC |
| 19:51.18 | wcselby | that's what I use, if it has it I don't know how to enable it |
| 19:52.25 | *** join/#asterisk fireman_biff (~biff@65.48.133.103) |
| 19:54.22 | p3nguin | Settings > Configure KVirc > Interface... Notifier. |
| 19:58.14 | [TK]D-Fender | [15:48]<t_dot_zilla>[TK]D-Fender: just an internal call to MoH for 3 seconds and then hang up <- how is this "incomplete"? |
| 19:58.52 | t_dot_zilla | we want to test to make sure call goes through |
| 19:59.42 | *** part/#asterisk fireman_biff (~biff@65.48.133.103) |
| 19:59.51 | *** join/#asterisk Kobaz (~kobaz@its.kobaz.net) |
| 20:00.19 | [TK]D-Fender | t_dot_zilla: You seem to be saying it hasn't. What does a call that does not "go through" look like? |
| 20:01.21 | fifer | Is "module reload chan_dahdi.so" nondestructive to existing channels? |
| 20:01.34 | [TK]D-Fender | fifer: It will trash existing calls |
| 20:01.37 | fifer | I know that dahdi restart destroyes all existing |
| 20:02.06 | fifer | I was just trying to make sure I was not missing a nondestructive way to load changes :-) |
| 20:02.13 | *** join/#asterisk bcrisp (~bcrisp@wsip-184-191-141-38.ph.ph.cox.net) |
| 20:02.17 | [TK]D-Fender | fifer: There isn't with DAHDI |
| 20:02.27 | fifer | Looks like I'll have to either wait for a calm in the storm or the end of the day |
| 20:02.30 | fifer | thanks! |
| 20:02.42 | fifer | I'm not really surprised, just checking |
| 20:04.27 | *** join/#asterisk timahvo1 (~rogue@41.223.57.76) |
| 20:04.53 | bcrisp | hi all, im receiving the following error: "res_config_mysql.c:581 update_mysql: MySQL RealTime: Updating column 'lastms', but that column does not exist within the table 'sip' (first pair MUST exist)!" . I set enable=no in cdr.conf, is there somewhere else I have to edit to disable cdr and these associated errors? |
| 20:05.22 | Kobaz | registers for astricon |
| 20:05.27 | bcrisp | the idea is to correct the table structure for recording cdr info in mysql, but for the meantime i just want to disable it altogether |
| 20:05.43 | wcselby | bcrisp - that's not a cdr error, it's mysql realtime, which is controlled in extconfig.conf |
| 20:05.57 | Kobaz | yeah |
| 20:05.58 | wcselby | Kobaz - did you get any kind of discount? |
| 20:06.03 | Kobaz | er |
| 20:06.04 | bcrisp | wcselby: awesome, thank you |
| 20:06.14 | Kobaz | wcselby: the yeah was for a different channel |
| 20:06.17 | Kobaz | wcselby: no, i didn't :( |
| 20:06.26 | wcselby | Kobaz - i ask because last year they had a discount for IRC users |
| 20:06.31 | wcselby | but I haven't seen it pop up here yet |
| 20:06.51 | Kobaz | i asked about it too, but Qwell said i was too late |
| 20:07.12 | wcselby | yeah they had another discount that was 100 off if you early registered |
| 20:07.25 | *** join/#asterisk Defraz (~Defraz@corp.fuzecore.com) |
| 20:07.26 | wcselby | the one I used last year was a 20% off, and I registered 3 weeks before the con |
| 20:07.29 | wcselby | so who knows |
| 20:09.02 | bcrisp | wcselby: i made the edits to extconfig and now am receiving other warnings: config.c:2020 find_engine: Realtime mapping for 'sippeers' found to engine 'general', but the engine is not available |
| 20:09.32 | wcselby | bcrisp - i guess maybe you should backup and tell us what you're trying to do |
| 20:10.16 | bcrisp | i was using mysql for recording cdr / realtime.. starting receiving errors of a missing field, now I want to disable whatever is choosing to log to mysql till i can figure out how to fix it |
| 20:10.29 | tzanger | Letoric: I'm referring to bkw |
| 20:11.12 | wcselby | bcrisp - did you recently update? |
| 20:11.23 | wcselby | bcrisp - is your mysql database even online and accessible? |
| 20:11.29 | bcrisp | wcselby: no, i think I may have had this issue in the past but just have greater verbosity now |
| 20:11.57 | wcselby | are you using mysql realtime for sip peers? |
| 20:12.22 | wcselby | realtime isn't just logging, it's for providing registrations and dialplan and all kinds of stuff, depending on how you've set it up |
| 20:12.24 | bcrisp | I was hoping to, but everything is specified in sip.conf |
| 20:12.32 | bcrisp | i was preparing to do so... |
| 20:13.25 | wcselby | bcrisp - ahh, well, verify you've properly setup your database, it looks like you've got missing columns that are expected to be there |
| 20:13.59 | bcrisp | wcselby: right, but my question was how to disassociate * from mysql altogether |
| 20:14.03 | bcrisp | so i can avoid the issues |
| 20:14.23 | wcselby | comment out your entire extconfig.conf file? |
| 20:15.13 | bcrisp | thanks |
| 20:15.22 | wcselby | perhaps also comment out the mysql connect statements in res_config_mysql.conf as well.... |
| 20:15.55 | wcselby | they're not connect statements, it's the database definition, is what I meant |
| 20:15.56 | *** join/#asterisk ferdna (~yup@cpe-24-92-114-97.elp.res.rr.com) |
| 20:15.57 | wcselby | but yeah |
| 20:16.32 | bcrisp | i think that will produce errors.. there are a couple of places that specify mysql as the destination / source (logging, realtime, etc) .. just a lot of config files to look through heh |
| 20:18.56 | *** join/#asterisk deonv (~adium@41.219.109.60) |
| 20:20.38 | ferdna | i have a digium echo canceller card... what colors should the leds on it be? |
| 20:20.44 | ferdna | i only get red |
| 20:21.07 | WIMPy | They are duo-colour? |
| 20:21.12 | ferdna | i have no idea |
| 20:22.05 | WIMPy | I've got 8 green ones. |
| 20:22.38 | ferdna | heck |
| 20:22.48 | bmoraca_work | over 9000 green ones |
| 20:23.07 | WIMPy | 8 on that card. |
| 20:23.29 | WIMPy | Might have some 2000 in my drawers as well. |
| 20:27.40 | ferdna | WIMPy, i only see 6 leds in the card it self |
| 20:29.14 | *** join/#asterisk evil_gordita (robert@ip70-188-50-186.rn.hr.cox.net) |
| 20:31.26 | WIMPy | There are probably different versions. But I didn't see them all until they lit up. |
| 20:32.38 | fifer | well, I was able to restart dahdi with faxdetect=both |
| 20:32.38 | fifer | faxbuffers=>6,full |
| 20:32.51 | fifer | set properly for both T-1 spans |
| 20:33.11 | fifer | When I look at an active channel trying to recieve a fax it still shows EC as on though |
| 20:35.38 | fifer | It does apear to be trying to work at 7,200pbs rather than training down to 2,400 but I still keep getting this error at the end: |
| 20:35.40 | fifer | app_fax.c: Error transmitting fax. result=13: Unexpected message received |
| 20:35.44 | t_dot_zilla | how do you display output of an asterisk command from shell? |
| 20:37.39 | t_dot_zilla | for instance, asterisk -rx "originate zap/1/123456 extension 1@greeting" shows no output to shell |
| 20:38.45 | WIMPy | Can you have multiple host= lines in a peer definition to match on incomming calls? |
| 20:39.07 | bmoraca_work | t_dot_zilla: what are you expecting to see? |
| 20:39.28 | bmoraca_work | WIMPy: i don't believe so. that kind of defeats the purpose of the "host" field... |
| 20:39.34 | t_dot_zilla | i'd like to see some output from it. is there a way to have the xtension print something? like 'OK' ? |
| 20:40.30 | *** join/#asterisk [cannibalera] (~cannibale@201-25-254-177.fnsce703.dsl.brasiltelecom.net.br) |
| 20:43.20 | *** part/#asterisk ironm (~ironm@84-73-66-195.dclient.hispeed.ch) |
| 20:43.36 | *** join/#asterisk ironm (~ironm@84-73-66-195.dclient.hispeed.ch) |
| 20:45.38 | bmoraca_work | t_dot_zilla: use an AMI originate if it's important to verify that the command was successful. |
| 20:47.55 | t_dot_zilla | perl? |
| 20:49.16 | *** join/#asterisk psilikon (~joel@cerberus.vicimarketing.com) |
| 20:49.31 | bmoraca_work | you can use perl if you want |
| 20:50.03 | t_dot_zilla | do i need to install anything on the asterisk end ? module ? |
| 20:50.13 | t_dot_zilla | does it have to reside on the asterisk machine? |
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| 20:58.02 | Katty | hhhhelllllllllllllloooooooooooooooooooooooooo nurse |
| 20:58.18 | Kobaz | yeap |
| 20:58.35 | Katty | infobot: seen jblack? |
| 20:58.42 | infobot | jblack <~jblack@71.181.244.180> was last seen on IRC in channel #asterisk, 70d 3h 6m 34s ago, saying: 'Hi'. |
| 20:58.51 | Katty | oh my. |
| 20:59.16 | fifer | So, one continuing issue I apear to have is that EC is not being turned off for a dahdi channel when an incomming fax is being recieved. |
| 20:59.30 | Katty | i also have a continuing issue. |
| 20:59.34 | Katty | it's called 5 not coming quick enough. |
| 21:00.07 | fifer | faxdetect is set to both but I can see the EC is still on when I look at the in use channel when a fax is comming in |
| 21:00.44 | fifer | I'm also getting a different error now: app_fax.c: Error transmitting fax. result=40: Unexpected DCN after requested retransmiss |
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| 21:21.40 | Letoric | Are there any caveats to using mixmonitor to run a script, instead of a single command? I'm having difficutlies that we're still attempting to debug....but when I execute the same script, with the same command that the verbose output on the call log shows, it works |
| 21:22.03 | Letoric | in a meeting so can't do the pastebin atm, but thought I would throw it out there |
| 21:22.29 | *** join/#asterisk TimeRider (~steve@188-220-33-123.zone11.bethere.co.uk) |
| 21:25.46 | Katty | bored bored bored. |
| 21:28.32 | leifmadsen | Katty: make me a pie! |
| 21:30.18 | wcselby | i'll take a pie |
| 21:30.28 | wcselby | if you're giving them away |
| 21:36.40 | Katty | i'm actually making a pie tonight. |
| 21:36.56 | Katty | http://www.food.com/recipe/yummy-crunchy-caramel-apple-pie-31128 <- that one. |
| 21:37.40 | fifer | I understand there used to be a command to manually turn off EC in a dial plan "ZapEC(off)" is there something currently active in dahdi? I'm on * 1.6.2.13 and dahdi 2.4 |
| 21:39.10 | *** join/#asterisk estr (~eastr@ti0013a380-dhcp2018.bb.online.no) |
| 21:39.10 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 21:39.17 | [TK]D-Fender | \o/ |
| 21:41.50 | *** join/#asterisk kfife (~Miranda@home.chicagoventure.com) |
| 21:42.47 | *** join/#asterisk ningia (~gain@host97-88-dynamic.15-87-r.retail.telecomitalia.it) |
| 21:46.15 | *** join/#asterisk korcan_ (~johnynum5@ip65-44-169-66.z169-44-65.customer.algx.net) |
| 21:48.45 | fifer | So, one continuing issue I apear to have is that EC is not being turned off for a dahdi channel when an incomming fax is being recieved. |
| 21:49.00 | *** join/#asterisk mpe_ (~mpe@0xd99d3f8f.customer.cybercity.dk) |
| 21:49.18 | [TK]D-Fender | fifer: It'd better not. |
| 21:49.19 | fifer | I'm not sure if this is the main issue I'm having with successfully recieving faxes, but it is the main one I can find |
| 21:50.24 | *** part/#asterisk mpe_ (~mpe@0xd99d3f8f.customer.cybercity.dk) |
| 21:51.00 | fifer | [TK]D-Fender: ? everything I can find and averyone I ask indicate that dahdi will automatically turn off EC when fax tones are detected on that channel |
| 21:52.42 | *** join/#asterisk NickNick (~nicholas@93-97-188-195.zone5.bethere.co.uk) |
| 21:55.06 | [TK]D-Fender | fifer: EC isn't on automatically.. its enabled when needed normally. You can't EC a call that isn't there |
| 21:55.25 | [TK]D-Fender | fifer: Having EC off is REQUIRED |
| 21:55.50 | *** join/#asterisk ningia (~gain@host97-88-dynamic.15-87-r.retail.telecomitalia.it) |
| 21:56.01 | *** part/#asterisk ningia (~gain@host97-88-dynamic.15-87-r.retail.telecomitalia.it) |
| 21:56.08 | *** join/#asterisk ningia (~gain@host97-88-dynamic.15-87-r.retail.telecomitalia.it) |
| 21:56.12 | fifer | When i inspect a dahdi channel that is in use for a fax call EC is still on |
| 21:56.18 | *** part/#asterisk ningia (~gain@host97-88-dynamic.15-87-r.retail.telecomitalia.it) |
| 21:56.28 | *** join/#asterisk ningia (~gain@host97-88-dynamic.15-87-r.retail.telecomitalia.it) |
| 21:56.31 | *** part/#asterisk ningia (~gain@host97-88-dynamic.15-87-r.retail.telecomitalia.it) |
| 21:56.57 | fifer | dahdi show channel |
| 21:58.20 | fifer | This is an example of what I'm getting: http://pastebin.ca/1949752 |
| 21:58.58 | [TK]D-Fender | fifer: If you are in app_fax you are already in too far to debug |
| 21:59.06 | [TK]D-Fender | fifer: You need to see the channel as it comes in |
| 22:00.30 | fifer | This is what someone else in irc asked me to confirm, that EC was off during the call on the in use channel |
| 22:01.11 | [TK]D-Fender | fifer: |
| 22:01.17 | [TK]D-Fender | dahdi show channel X |
| 22:01.18 | fifer | I'm terminating fax from two PRI T-1's and have faxdetect=both and faxbuffers=>6,full |
| 22:01.37 | fifer | that is what I'm doing |
| 22:01.43 | fifer | when the call is active |
| 22:02.22 | fifer | I do a test fax, watch the dahdi channels for the CID of the incomming fax, then when it is connected I check that channel |
| 22:13.15 | *** join/#asterisk Bloudermilk (~Bloudermi@dsl081-234-075.lax1.dsl.speakeasy.net) |
| 22:13.17 | Bloudermilk | Hey all |
| 22:13.40 | Bloudermilk | is the SHELL() application supposed to be built in to Asterisk 1.6.13? |
| 22:14.18 | *** join/#asterisk psykon (~joel@75-121.186-72.tampabay.res.rr.com) |
| 22:17.53 | [TK]D-Fender | Bloudermilk: No such application exists |
| 22:18.19 | Bloudermilk | What is http://www.voip-info.org/wiki/view/Asterisk+func+shell ? |
| 22:18.45 | [TK]D-Fender | Bloudermilk: Exactly what it says |
| 22:19.33 | Bloudermilk | Helpful, thanks |
| 22:19.51 | BesticlesWork | I have attempted to setup Festival on my box. Festival is reporting that it accepted the client, and then disconnects.. no errors. Asterisk is reporting is also reporting no errors. Yet on the phone I hear nothing. I don't know what the next step is to debug my problem. Any suggestions? |
| 22:20.05 | [TK]D-Fender | Bloudermilk: Got a failure to show us? |
| 22:20.33 | Bloudermilk | I think I see my problem. I failed to see the difference between an Application and a Function |
| 22:20.46 | Bloudermilk | SHELL() should be used in conjunction with SET or something |
| 22:21.06 | [TK]D-Fender | bougyman: Congratulations... I was wondering when you would catch that :) |
| 22:21.20 | [TK]D-Fender | bougyman: there is no such APPLICATION. |
| 22:21.25 | [TK]D-Fender | blEven |
| 22:21.29 | WIMPy | just found out that you need a qualify=yes for each IAX peer, even if you have qualify=yes in [general]. |
| 22:21.37 | [TK]D-Fender | Bloudermilk: Dangnammit |
| 22:21.38 | [TK]D-Fender | askdjklashdjklasdh |
| 22:22.07 | Bloudermilk | D-Fender: Your ambiguity mostly translated to rudeness :) |
| 22:22.57 | [TK]D-Fender | [18:17]<[TK]D-Fender>Bloudermilk: No such application exists <- Not ambiguous. Every words was very specific to it's intent. Not an APPLICATION. |
| 22:23.35 | [TK]D-Fender | Bloudermilk: But you seem to have worked past that hurdle... onwards to great success... |
| 22:24.01 | Bloudermilk | D-Fender: And you too |
| 22:25.21 | fifer | [TK]D-Fender: When I do a dahdi show channel x on a channel with a live fax call I should NOT see EC on. Correct? |
| 22:25.39 | fifer | It almost looked like you were indicating otherwise....just trying to make sure I have this right |
| 22:27.31 | [TK]D-Fender | fifer: Having EC enabled on a channel that is trying to fax will fuck it the fuck up. COMPLETELY. Crystal clear enough for you? :) |
| 22:28.46 | fifer | That is NOT what everyone else is telling me so you need to explain |
| 22:29.34 | fifer | Everything I can find says that a key and solid part of Asterisk+dahdi+fax is the detection of fax and the turning off of EC for THAT dahdi chanel for the duration of THAT call |
| 22:30.39 | fifer | I was told to use dahdi show channel x to see if this is working properly and it looks like it is not, which is what I'm trying to figure out |
| 22:32.44 | fifer | the dahdi setting "faxdetection=incomming" is either part or of this or the key to it. This setting is set for both of my T-1 spans |
| 22:34.27 | *** part/#asterisk bsaxon (~bsaxon@12.107.149.61) |
| 22:37.56 | fifer | faxdetect=, bad momory |
| 22:39.42 | *** join/#asterisk [cannibalera] (~cannibale@201-41-231-105.fnsce703.dsl.brasiltelecom.net.br) |
| 22:51.04 | Bloudermilk | Does anyone have any advice when it comes to monitoring the health of asterisk boxes for auto-scaling? |
| 22:51.20 | Bloudermilk | For instance, which properties to monitor |
| 23:00.23 | *** part/#asterisk fracBlend (~fracBlend@unaffiliated/fracblend) |
| 23:05.37 | *** join/#asterisk dandate2 (~gtejkgjke@58.69.25.48) |
| 23:07.53 | dandate2 | I have a DSL with 0.25MB upstream. I am reading that ulaw requires 84kb transmit for the softphone factoring tcp/ip overhead. Does this overhead bear weight for each softphone? Because 84x3 = 252kbps I am afraid that I would push the limits. There would be no computers on this network just an ATA adapter |
| 23:09.34 | bmoraca_work | with SIP, yes, the overhead exists for each channel, not just each device |
| 23:09.43 | bmoraca_work | with IAX, the overhead is once per device |
| 23:10.22 | dandate2 | so with that kind of upstream i could only safely run 2 phones |
| 23:10.25 | bmoraca_work | 256k upstream might be sufficient if there was no other use on that particular network. if there is other use, and you expect three simultaneous calls frequently, then you will run into issues |
| 23:10.46 | dandate2 | right |
| 23:10.49 | dandate2 | well my thing is |
| 23:10.55 | bmoraca_work | dandate2: you could run more phones, but you're likely to only get two simultaneous calls to sound good |
| 23:11.05 | dandate2 | here in the philippines, the more MB you add to your maximum bandwidth limit the slower the ping times |
| 23:11.31 | dandate2 | and the T1 line is $1300/month for 2 MB |
| 23:11.33 | bmoraca_work | as long as the ping time is faster than 150ms, you won't see quality problems caused by latency |
| 23:11.49 | dandate2 | well it just dont get that fast here heh |
| 23:12.16 | dandate2 | my 0.25MB upstream DSL gets 220ms-260ms to los angeles and thats hella fast for the region |
| 23:12.37 | bmoraca_work | you'll likely get quality problems, then |
| 23:12.47 | bmoraca_work | even without the bandwidth congestion |
| 23:13.04 | dandate2 | most net cafes where they have 2+MB connections get 280-360ms to LA |
| 23:13.13 | dandate2 | now thats bad heh |
| 23:13.15 | p3nguin | the T1 line is $1300/month for 2 MB <-- WHAT? |
| 23:13.22 | dandate2 | yeah i got the price chart right here |
| 23:13.30 | *** join/#asterisk nny1 (~Scott@cpe-174-107-201-103.sc.res.rr.com) |
| 23:13.44 | p3nguin | I use more than 2 MegaBytes on IRC every day. |
| 23:13.46 | dandate2 | the 1MB was only $650 /mo but they discontinued it |
| 23:14.04 | nny1 | hmm not sure where to start on this one. configure: error: cannot find install-sh, install.sh, or shtool in `pwd` "."/`pwd` |
| 23:14.07 | dandate2 | now i'm looking at $1300/mo and a $500 installation fee |
| 23:14.14 | nny1 | trying to compile asterisk, get that when I do ./configure |
| 23:14.24 | nny1 | assume something in my build tools, any help appreciated |
| 23:14.31 | dandate2 | but I figure I could just get 4 low bandwidth DSLs for my call center and only spend $80/mo |
| 23:16.14 | dandate2 | but yeah $1300/mo, cuz they got a monopoly with 30 cent per minute call rates to PH |
| 23:17.56 | nny1 | hmm |
| 23:18.20 | nny1 | <PROTECTED> |
| 23:18.33 | nny1 | nothing like working at 7 pm because of a compile error HOORAY! |
| 23:20.54 | *** join/#asterisk [cannibalera] (~cannibale@201-41-231-105.fnsce703.dsl.brasiltelecom.net.br) |
| 23:21.34 | nny1 | intstall.sh is a file generated by autoconf / automake |
| 23:22.54 | nny1 | so asterisk-1.6.2.13 is apparently just missing this key file? I somehow doubt it |
| 23:23.03 | nny1 | and this place is dead and useless as always at this time of day :\ |
| 23:24.39 | nny1 | meh fuck it, time to regress to a known working source file huzzah! |
| 23:28.34 | nny1 | jesus jumping christ on fire in a handbasket. 1.6.2.0 works |
| 23:28.50 | nny1 | yet latest "current" craps the bed on compile. Amazing |
| 23:28.56 | *** join/#asterisk pabelanger (~pabelange@2607:f2c0:a000:166:218:f3ff:fe51:c71) |
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| 23:30.38 | dandate2 | when I check my upstream at speedtest.net, is it just telling me the maximum or my average? |
| 23:31.53 | WIMPy | If that makes a difference you've got bigger troubles than bandwidth. |
| 23:32.20 | *** join/#asterisk Fruchthoernschen (~Fruchthoe@trir-4d0bab0e.pool.mediaWays.net) |
| 23:32.51 | WIMPy | What is a Fruchthoernschen? |
| 23:33.35 | Fruchthoernschen | Hello does someone knows an softphone which works with asterisk and supports soudnservers like pulseaudio? |
| 23:33.54 | nny1 | isn't this some impressive garbage http://www.google.com/#hl=en&expIds=17259,23756,24692,24878,24879,26751&sugexp=ldymls&tok=CyBXcLWiliL3czO3Xxlmag&xhr=t&q=asterisk+1.6.2.13+%22error:+cannot+find+install-sh,+install.sh,%22&cp=19&pf=p&sclient=psy&aq=f&aqi=&aql=&oq=asterisk+1.6.2.13+%22error%3A+cannot+find+install-sh%2C+install.sh%2C%22&gs_rfai=&pbx=1&fp=fa151da40c6c8a2a |
| 23:33.54 | Fruchthoernschen | WIMPy, don't know ;) I needed an Nickname.. ;) |
| 23:35.46 | Fruchthoernschen | there are much soft-phone applications but I can't get them to work or they are out dated. I have successfully teste erika v. 3.2.6 but it don't work with my soundserver/pulseaudio {kde} |
| 23:36.14 | dandate2 | well when it starts the test it begins low and gradually reaches its max |
| 23:36.28 | dandate2 | would that cause problems for voip? |
| 23:36.46 | WIMPy | Actually I thought sound servers were outdated since dmix? |
| 23:37.03 | Fruchthoernschen | oehm |
| 23:37.08 | Fruchthoernschen | good question |
| 23:37.18 | WIMPy | dandate2: That's probably just the test. |
| 23:38.05 | WIMPy | Zoiper will happily play to dmix. |
| 23:41.00 | Fruchthoernschen | WIMPy, sounds good, |
| 23:42.00 | *** join/#asterisk tzafrir (~tzafrir@212.179.75.202) |
| 23:43.07 | nny1 | hmm note to self, slap people who use spaces in file names in linux |
| 23:43.30 | nny1 | kind of hilarious |
| 23:43.51 | nny1 | 1.6.2.0 craps out on make install - space in directory name files are untarred in |
| 23:44.03 | nny1 | 1.6.2.13 craps on ./configure, can't find install-sh due to same reason |
| 23:44.36 | *** join/#asterisk ectospasm (~ectospasm@188.72.223.139) |
| 23:44.48 | nny1 | I always use underscores out of habit and get laughed at for it. Here it is 2010 and I have found two instances were it breaks things |
| 23:48.18 | ferdna | how do you increment your handset volume? |
| 23:48.32 | ferdna | i mean not from the phone, but from the asterisk box |
| 23:51.20 | Fruchthoernschen | WIMPy, can you help my with zoiper to register it at my asterisk? |
| 23:51.47 | Fruchthoernschen | I should use sip to type in my account settings? |
| 23:53.59 | leifmadsen | ferdna: VOLUME() function |
| 23:54.29 | ferdna | leifmadsen, awesome... let me research this function... thank you |
| 23:54.54 | kfife | How many concurrent manager connections do you need to have before something like astmanproxy becomes a good idea? |
| 23:55.00 | leifmadsen | ferdna: no available in 1.4 |
| 23:55.09 | leifmadsen | kfife: hundreds? |
| 23:55.19 | leifmadsen | kfife: astmanproxy is pretty much dead fyi |
| 23:55.25 | kfife | Beautiful. That's what I needed to know. |
| 23:55.26 | leifmadsen | I don't think it's been developed since about 2007 |
| 23:55.31 | ferdna | leifmadsen, rxgain |
| 23:55.44 | leifmadsen | ferdna: then you're talking about analog lines -- you did not specify |
| 23:56.04 | leifmadsen | you just made a general statement, so I gave a general answer |
| 23:56.05 | kfife | leifmadsen: Thanks. Are there things out there to take its place or is it dead because Asterisk doesn't need it anymore. |
| 23:56.15 | ferdna | leifmadsen, thanks you point me to the right direccion |
| 23:56.17 | kfife | i.e. "Hundreds" |
| 23:56.29 | kfife | ...enough for practical purposes |
| 23:56.39 | leifmadsen | kfife: I don't think it's really necessary anymore. The primary reason I think was to create an abstraction layer that was more secure, but you can use TLS now I think to connect to the manager |
| 23:57.11 | leifmadsen | kfife: however, I have not load tested the manager, so I'm not sure how many connections it can handle to be honest -- I personally would figure it'd be somewhere around the same as a channel to be honest |
| 23:57.41 | kfife | Great. I've got a PHP script that could hit hard'ish doing a fleet of astDB lookups. |
| 23:57.52 | leifmadsen | kfife: let me know how it goes :) |
| 23:57.58 | kfife | Thanks. I' |
| 23:58.57 | kfife | leifmadsen: I'll be touching on it briefly at Astricon. It's part of a FFA routine to test the status of a fax. AstDB will return "Trying" "Retrying" "Complete" etc. |
| 23:59.20 | kfife | PHP script used to trigger the fax handed in via PDF, PHP get routine for the external app to lookup the status on demand. |
