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| 00:03.06 | *** join/#asterisk jks (~jks@193.189.93.254) |
| 00:03.50 | *** part/#asterisk Tech_Travis (~Travis@mail.techglia.com) |
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| 00:18.48 | LemensTS | *ping |
| 00:22.30 | *** part/#asterisk DMeloUK (~DominicMe@64.129.95.226) |
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| 00:34.23 | dlynes | *pong |
| 00:34.30 | ryduh | *pong pong |
| 00:34.54 | ryduh | should I be able to ringall in a queue for 3 local members? |
| 00:35.32 | dlynes | ryduh, one would think |
| 00:36.48 | dlynes | Is there a reason why I would get an extraordinarily high UDP 'packets to unknown port received' on an asterisk host? |
| 00:37.14 | dlynes | It's approximately 4% of all UDP packets received |
| 00:38.02 | dlynes | On other machines, it's as low as 0.01% |
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| 00:48.33 | LemensTS | . |
| 00:48.43 | Chainsaw | .. |
| 00:49.17 | LemensTS | If you have your DID's at a provider, and the provider goes down all incoming calls will stop for the customers. How do you guys deal with this, I dont know of a way to do a failover |
| 00:50.38 | bmoraca_work | get a provider with enough redundancy to not go down |
| 00:50.55 | *** join/#asterisk geneticx (~geneticx@c-75-74-66-161.hsd1.fl.comcast.net) |
| 00:51.05 | bmoraca_work | or use SS7 with multiple peers |
| 00:51.35 | bmoraca_work | the former is usually a lot simpler and less expensive than the latter |
| 00:52.30 | bmoraca_work | or, call your provider and tell them to forward your calls to a cell phone or something. they can usually do that even if they can't give you calls |
| 00:52.44 | LemensTS | Yea true. That SS7 looks complicated, but what isn't until you learn it. Ill read up on it. |
| 00:53.07 | bmoraca_work | LemensTS: how many concurrent calls and DIDs are you dealing with? |
| 00:53.37 | LemensTS | Yea I have them on Teleasip now, gonna switch them. Been using voipinnovations, i like them, but doubt the redundancy is that great. |
| 00:53.46 | bmoraca_work | the reason i ask is because SS7 probably isn't the right answer. |
| 00:53.54 | LemensTS | Oh like 30-35 DID's currently |
| 00:53.59 | bmoraca_work | LemensTS: voipinnovations is run by Globalpops |
| 00:54.04 | bmoraca_work | they're a very solid company |
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| 00:54.19 | bmoraca_work | with great prices on toll-free origination |
| 00:54.45 | LemensTS | Yea i havent had any problems with them on outbound |
| 00:55.09 | LemensTS | I got faillover on my end on outbound so its not a big deal like inboud |
| 00:55.13 | bmoraca_work | their prices are decent on outbound, but more expensive than my current provider. my only need for them currently is toll-free origination |
| 00:55.20 | dzup_ | does anyone knows voip provider for mexico and central america with very low rates? |
| 00:55.40 | bmoraca_work | dzup_: "mexico" and "low rates" do not belong in the same sentence together |
| 00:55.50 | bmoraca_work | 60/60 billing is a ripoff |
| 00:56.03 | LemensTS | bmoraca: i wrote a program the checks the rates to the destinations so I just load up different providers anyways |
| 00:56.09 | dzup_ | i know thats why am asking heh |
| 00:56.17 | LemensTS | seems 6/6 is standard now |
| 00:56.34 | bmoraca_work | LemensTS: not to mexico...unless it changed in the last couple months |
| 00:57.03 | LemensTS | bmoraca: nah i was just talking in general. i hadn't looked at mexico in like a year |
| 00:57.09 | bmoraca_work | LemensTS: i thought about doing some LCR, but i get such great prices from my main provider that it just doesn't make sense |
| 00:57.34 | *** join/#asterisk tkrn (~tkrn@216.196.213.246) |
| 00:57.41 | dzup_ | i use flowroute now and am very happy with them. |
| 00:58.00 | bmoraca_work | my average rate throughout the US is less than $0.006/min. i can't really complain too much about it. |
| 00:58.20 | LemensTS | bmoraca: yea thats how it used to be ran, but i wanted to know how much each customer was actualy using a month in case they were costing me money...in I ended up just writing LCR |
| 00:58.36 | dzup_ | bmoraca_work: whos your provider? |
| 00:59.00 | bmoraca_work | dzup_: pacwest. i've been colocating with them for almost 10 years |
| 00:59.51 | dzup_ | hmm |
| 00:59.54 | bmoraca_work | LemensTS: i bill $40/mo for unlimited local/long distance. customers would have to talk A LOT for me to lose money on them, hehe |
| 01:00.12 | bmoraca_work | most of my customers don't even cost me $5/mo |
| 01:00.13 | *** part/#asterisk cjdaniel (~chris@67.23.1.232) |
| 01:00.33 | LemensTS | bmoraca: yea its cool when you see a customers not even make a call |
| 01:00.33 | bmoraca_work | one of my customers costs me about $60/mo, but they pay over $900/mo, so I'm not worried |
| 01:00.46 | dzup_ | what about unternational rates? |
| 01:00.57 | LemensTS | bmoraca: are you doing hosted pbx? |
| 01:01.00 | dzup_ | s/unternational/international/g |
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| 01:01.15 | dzup_ | yes |
| 01:04.06 | bmoraca_work | LemensTS: yes, i do. but i also do pure trunking, though i have many more hosted PBX customers than trunking customers |
| 01:04.54 | LemensTS | It seems hosted pbx makes more money and is easier to sell then residential |
| 01:04.59 | bmoraca_work | so, yes, that monthly take does include equipment, but the equipment costs are pretty low...$80/phone |
| 01:05.06 | *** join/#asterisk m4ck_ (~yoavw@corky.net) |
| 01:05.43 | bmoraca_work | LemensTS: most definitely. residential is too competitive. i'm considering expanding to it just to fill off-hours with traffic because i'm paying for the capacity regardless of whether i use it...but right now, it would be more trouble than it's worth |
| 01:06.14 | *** join/#asterisk aandrade (~aandrade@189.34.124.123) |
| 01:06.31 | m4ck_ | Can anyone please help with a transfer problem? |
| 01:18.38 | *** join/#asterisk infobot (ibot@rikers.org) |
| 01:18.38 | *** topic/#asterisk is Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.6.2.5, 1.6.1.17, 1.6.0.25 (2010/02/25), 1.4.29.1 (2010/02/18), *-Addons 1.6.2.0 (2009/12/18), 1.6.1.2, 1.6.0.4, 1.4.10 (2009/12/02), dahdi-linux 2.2.1 + dahdi-tools 2.2.1 (2010/01/20), Libpri 1.4.10.2 (2009/10/20) -=- Related channels: #asterisknow,#switchvox,#freepbx,#asterisk-dev,#asterisk-bugs |
| 01:18.39 | Katty | ALSO, playpen can be taken outside in the grass |
| 01:18.41 | Katty | for romping |
| 01:18.48 | dlynes | Yummy! Let me sink my nice, razor sharp teeth into that succulent puppy |
| 01:18.56 | Katty | not....exactly |
| 01:19.22 | dlynes | the ferret i had would love a puppy in his cage :) |
| 01:19.32 | dlynes | The bugger loved the rabbits in the cage next to it :( |
| 01:19.58 | dlynes | sucked every last ounce of blood out of the rabbits |
| 01:20.10 | Katty | http://farm5.static.flickr.com/4016/4404666421_64c09a0fa3_b.jpg |
| 01:20.27 | dlynes | wow...three of them |
| 01:20.31 | Katty | four |
| 01:20.45 | carrar | Konbanwa!!!!!! |
| 01:20.46 | dlynes | i guess you feed them cat food? |
| 01:20.56 | Katty | and raw chicken |
| 01:20.58 | dlynes | I used to feed mine fish heads |
| 01:21.10 | dlynes | they loved the fish heads, and I'd get them for free |
| 01:21.28 | dlynes | Picked up a whole garbage bag full every two weeks |
| 01:21.32 | Katty | carrar: herroes |
| 01:22.04 | m4ck_ | dlynes: thanks for the explanation. But I actually tried to add a one-digit extension, just to see if it would transfer to it. It still didn't. |
| 01:22.26 | m4ck_ | dlynes: I created an extension '3', tried to transfer to 3 and got "invalid" as usual. |
| 01:22.28 | dlynes | m4ck_, did you try using includes to make sure that single digit extension got included first? |
| 01:22.45 | Katty | http://farm5.static.flickr.com/4048/4404675643_8f159755c5_b.jpg <- toys. |
| 01:22.56 | dlynes | m4ck_, i.e. include => digit_3_context |
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| 01:22.59 | ghento | j p/ython |
| 01:23.07 | ghento | whoops |
| 01:27.35 | *** join/#asterisk infobot (ibot@rikers.org) |
| 01:27.35 | *** topic/#asterisk is Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.6.2.5, 1.6.1.17, 1.6.0.25 (2010/02/25), 1.4.29.1 (2010/02/18), *-Addons 1.6.2.0 (2009/12/18), 1.6.1.2, 1.6.0.4, 1.4.10 (2009/12/02), dahdi-linux 2.2.1 + dahdi-tools 2.2.1 (2010/01/20), Libpri 1.4.10.2 (2009/10/20) -=- Related channels: #asterisknow,#switchvox,#freepbx,#asterisk-dev,#asterisk-bugs |
| 01:36.13 | *** join/#asterisk infobot (ibot@rikers.org) |
| 01:36.13 | *** topic/#asterisk is Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.6.2.5, 1.6.1.17, 1.6.0.25 (2010/02/25), 1.4.29.1 (2010/02/18), *-Addons 1.6.2.0 (2009/12/18), 1.6.1.2, 1.6.0.4, 1.4.10 (2009/12/02), dahdi-linux 2.2.1 + dahdi-tools 2.2.1 (2010/01/20), Libpri 1.4.10.2 (2009/10/20) -=- Related channels: #asterisknow,#switchvox,#freepbx,#asterisk-dev,#asterisk-bugs |
| 01:36.24 | coppice | dlynes: that name rings a vague bell, but I can't think what it is |
| 01:36.58 | jdoe | it looks like a generic asterisk web interface. |
| 01:37.02 | jdoe | ... like freepbx or whatever. |
| 01:37.07 | dlynes | coppice, something to do with asterisk and telephony |
| 01:37.21 | dlynes | coppice, Asterisk::config perl module is a subproject of it |
| 01:37.28 | dlynes | coppice, seems to be based in Taiwan |
| 01:37.41 | dlynes | coppice, that's why I thought you might know something about it |
| 01:37.55 | dlynes | tankbusta, not a problem |
| 01:38.15 | Corydon76-dig | is banging his head against the wall that is sipp |
| 01:38.20 | coppice | dlynes: well googling only seems to throw up stuff in simplified |
| 01:39.16 | dlynes | coppice, http://cn.freeiris.org/wiki/ |
| 01:39.36 | dlynes | coppice, is that simplified? I thought it was traditional....guess not.... |
| 01:39.42 | coppice | Freeirisæ¯ä¸å°PBX, Freeirisæ¯ä¸å°ç½å
³, Freeirisæ¯å¼å«ä¸å¿, Freeirisä»ä¹é½å¯ä»¥ |
| 01:39.44 | coppice | Sounds like the marketing dept :-) |
| 01:40.02 | dlynes | coppice, yeah...they have commercial support |
| 01:40.56 | coppice | You mean like "Freeirisç±å¥½è
QQ群" |
| 01:41.47 | coppice | QQ is big in China |
| 01:41.58 | dlynes | coppice, nah...more like http://cn.freeiris.org/store.php?action=list&category=support |
| 01:42.05 | dlynes | coppice, yeah..it's like the Chinese msn |
| 01:42.15 | dlynes | coppice, and tons of viruses on it, too |
| 01:43.03 | coppice | well, it sounds like the kind of thing QQ might be interested in supporting |
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| 01:45.52 | Katty | tinkers with Sony Vegas Pro 9 |
| 01:47.23 | coppice | dlynes: well, that site just taught me a new expression - åé³æ¶é¤ - now I've seen it, its obvious what it means :-) |
| 01:56.17 | m4ck_ | dlynes: ok, I narrowed it to a simple configuration and still getting the same problem. Here are the config files: http://corky.net/yoavw/ast/ |
| 01:56.39 | m4ck_ | dlynes: so where did I go wrong? :) |
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| 02:07.39 | dlynes | m4ck_, log? |
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| 02:09.35 | m4ck_ | dlynes: what logging should I enable? Currently it doesn't output anything interesting. |
| 02:10.05 | dlynes | m4ck_, it must output something, or you wouldn't have known you had a problem |
| 02:10.47 | dlynes | m4ck_, i.e. the ten lines or so where it says the call starts, you got an invalid extension, and the call ends |
| 02:10.49 | m4ck_ | dlynes: I know I have a problem because I can't transfer from the extension. As soon as I enter the first digit, it says "I'm sorry..." |
| 02:11.08 | m4ck_ | dlynes: the call doesn't end. |
| 02:11.17 | m4ck_ | dlynes: it just resumes after failing to transfer. |
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| 02:11.20 | dlynes | m4ck_, ok, well...whatever happens |
| 02:11.42 | dlynes | m4ck_, from call to your phone, to when you try to transfer |
| 02:12.27 | m4ck_ | dlynes: I call from an outside line and answer on the ata186. As soon as the call is established, I press '#' on the ata186 extension. |
| 02:12.37 | dlynes | coppice, hrm...my wife just says it means 'echo clear', whatever that means |
| 02:12.49 | m4ck_ | dlynes: the call is paused and asterisk asks for the extension number. |
| 02:13.11 | dlynes | m4ck_, in which context is it asking? |
| 02:13.15 | m4ck_ | dlynes: I try to enter 600 (the test extension). As soon as I enter '6', it says "I'm sorry, that's not a valid extension" and resumes. |
| 02:13.43 | coppice | dlynes: cancel is a better translation than clear |
| 02:13.46 | m4ck_ | dlynes: How do I know which context? I assumed it's the same context as the extension. |
| 02:14.22 | dlynes | coppice, well, my wife's not a telecom person :0 |
| 02:15.14 | coppice | I think generally æ¶é¤ translates better to cancel. Its what you'll see on the typical cancel button of a GUI |
| 02:15.31 | dlynes | m4ck_, i see nothing in your sip configuration telling me what context your sip phone uses |
| 02:15.44 | dlynes | m4ck_, what have you defined for 'context=' in your general section? |
| 02:16.01 | dlynes | coppice, yeah...now she's saying clear it out, not clear :) |
| 02:16.27 | dlynes | coppice, i.e. get rid of it |
| 02:16.31 | m4ck_ | dlynes: actually, in my sip.conf, in the ata186 entry, there's context=out |
| 02:16.43 | dlynes | m4ck_, oh...missed that...sorry |
| 02:16.51 | m4ck_ | dlynes: so I'd assume that the context is 'out', which has a 600 extension |
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| 02:18.03 | dlynes | m4ck_, and your features.conf file? |
| 02:18.24 | m4ck_ | dlynes: sec, I'll upload it too. |
| 02:18.28 | dlynes | m4ck_, you didn't mention you were hitting '#' previously |
| 02:18.39 | coppice | dlynes: tell her Steve says "cancel" :-) |
| 02:18.55 | dlynes | m4ck_, and so it doesn't go to your dialplan...it looks at your features.conf file instead |
| 02:19.41 | coppice | dlynes: A point to the gweilo :-) |
| 02:20.33 | m4ck_ | dlynes: uploaded features.conf |
| 02:20.45 | dlynes | coppice, yang gui zi :) |
| 02:20.56 | m4ck_ | dlynes: in fact, I'm sure something is wrong in my features.conf. I also never managed to get call parking working with it, etc. |
| 02:21.44 | coppice | coppice: ngoh m sik teng pu tong hua faat yam |
| 02:22.41 | dlynes | coppice, quite the mouthful...hehehe |
| 02:23.00 | dlynes | coppice, yang gui zi/lou wei |
| 02:23.15 | dlynes | coppice, yang gui zi means white ghost, lou wei means foreigner |
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| 02:24.37 | dlynes | m4ck_, have you tried hitting '##', and then dialing the extension? |
| 02:24.56 | dlynes | m4ck_, besides...you haven't turned on ',t' or ',T' options to dial when calling ata186 |
| 02:25.33 | m4ck_ | dlynes: ## doesn't matter. For some reason a single '#' immediately triggers the announcement "transfer?" |
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| 02:25.50 | m4ck_ | and re t and T in the dial command, you're right, but I tried to add that and it didn't matter. |
| 02:26.40 | dlynes | m4ck_, have you tried '##', or are you just assuming it doesn't matter? |
| 02:27.00 | m4ck_ | dlynes: of course I tried. |
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| 02:27.20 | dlynes | I don't assume you tried, so of course not |
| 02:27.35 | coppice | dlynes: I'm so used to people saying gweilo, I loose track of the non-offensive terms for us. I'm used to seeing wei guo ren for foreigner at places like immigration counters |
| 02:27.48 | keith4_ | does the Polycom 320/321 have a voicemail button? |
| 02:28.08 | dlynes | coppice, white country person? |
| 02:28.15 | m4ck_ | dlynes: by the way, why does it say "transfer" after one # whereas it should've been ## ? |
| 02:28.48 | dlynes | m4ck_, the default is '#' for blind transfer, but you've overridden it with '##'. I'm guessing it'll use your override and the default |
| 02:29.16 | dlynes | m4ck_, i've never actually overridden it with '##', so I don't know... |
| 02:29.21 | coppice | dlynes: literally outside country person |
| 02:29.48 | dlynes | coppice, oh yeah...not thinking...gui/gwei mean white |
| 02:30.21 | coppice | dlynes: bai means white. gui means ghost or devil |
| 02:30.30 | dlynes | ah |
| 02:31.05 | m4ck_ | dlynes: normally it's a single # ? |
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| 02:31.15 | dlynes | m4ck_, yes...that's the default |
| 02:31.35 | m4ck_ | dlynes: and any idea why the *9 I put there for call parking is ignored even though I added k and K to the dial cmd? |
| 02:38.12 | dlynes | m4ck_, no idea....can you provide a sip debug of the call? |
| 02:38.44 | m4ck_ | dlynes: trying to see sip debug. |
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| 02:43.17 | m4ck_ | btw, how do I put a call on hold? |
| 02:46.21 | dlynes | m4ck_, flash button? |
| 02:46.36 | dlynes | m4ck_, or just hang up for 1/2 second |
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| 02:46.49 | m4ck_ | dlynes: I meant as a dtmf sequence if possible. |
| 02:47.12 | dlynes | m4ck_, no idea...I've never confined myself to trying to do that stuff with analog phones |
| 02:47.20 | dlynes | m4ck_, I've always used a good old sip phone |
| 02:47.37 | m4ck_ | dlynes: yeah, I guess I'm messing with weird stuff :) |
| 02:47.52 | dlynes | m4ck_, nah...lots of people on here do it, I believe |
| 02:47.55 | dlynes | m4ck_, just not me |
| 02:48.26 | dlynes | m4ck_, if my customers are too cheap to spring for a sip phone, i tell them to talk to someone else |
| 02:48.49 | dlynes | m4ck_, and my residential care home customers don't want all the snazzy features |
| 02:49.02 | dlynes | m4ck_, so they're on analog with all that garbage disabled :) |
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| 02:49.35 | dlynes | m4ck_, a lot of them don't even have the patience to learn how to use voicemail, much less call waiting or call transfer |
| 02:51.20 | m4ck_ | dlynes: makes perfect sense. I also wouldn't use that thing for production. I only use it to play with asterisk and figure it out. A learning system per se. |
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| 03:03.02 | dlynes | m4ck_, do you have a sip debug log yet? |
| 03:07.35 | m4ck_ | dlynes: no, I messed something up. Trying to get things working again. Sorry - I'm a newbie in this setup. I feel more at home in the kernel :) |
| 03:08.26 | m4ck_ | dlynes: I should've kept everything in some revision control system (svn or something) |
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| 03:14.02 | m4ck_ | dlynes: got it working with another sip phone (not analog). Same configuration. |
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| 03:14.41 | m4ck_ | wastes time on that analog thing... |
| 03:15.06 | m4ck_ | is sorry for wasting other people's time on the analog system as well. |
| 03:23.55 | m4ck_ | Now I can transfer the incoming call to another extension, but the receiving extension is unable to transfer again. |
| 03:24.24 | m4ck_ | It's symmetric. Each extension can transfer calls it receives, but not calls that were transferred to it. |
| 03:24.32 | m4ck_ | Transfer is not transitive? :) |
| 03:25.00 | mrtelnet | lol |
| 03:25.32 | *** part/#asterisk mrtelnet (~mr.telnet@c-24-12-44-14.hsd1.in.comcast.net) |
| 03:25.33 | *** part/#asterisk CareBear\ (peter@stuge.se) |
| 03:28.25 | *** join/#asterisk voxter (~voxter@66.183.99.78) |
| 03:40.22 | doneir | hrm, are globals vars the only way around random variables being destroyed when 'h' extension is called? |
| 03:40.38 | doneir | asterisk 1.6.17 |
| 03:41.19 | *** join/#asterisk lordmortis (~lordmorti@203-206-117-94.dyn.iinet.net.au) |
| 03:43.52 | Katty | boingboing |
| 03:43.55 | Katty | first video is nearly up! |
| 03:44.14 | LemensTS | doneir: what variables are you talking about |
| 03:45.52 | *** join/#asterisk ruben23 (~ITadmin@122.55.48.243) |
| 03:47.17 | ruben23 | hi |
| 03:47.27 | doneir | i set a variable in my main extension, when it goes to 'h' the variable no longer has a value, some do, but some don't. I read that "Use with great care: Apparently some channel variables get destroyed when the call is hung up, and those variables aren't available anymore (or have inconsistent values) when the h extension is being called." |
| 03:47.58 | doneir | that from http://www.voip-info.org/wiki/view/Asterisk+h+extension , though i don't know how up to date it is |
| 03:48.23 | Micc | anyone know where I can get a coupon code for fax for asterisk when purchased from digium? |
| 03:48.27 | doneir | i've jumped from 'h' to another extension to check if the variable exists outside of 'h', and it no longer does |
| 03:49.43 | Katty | ruben23: ohai |
| 03:50.05 | ruben23 | hi Micc. |
| 03:50.16 | LemensTS | doneir: what variable is it |
| 03:50.21 | Micc | Hey ruben23, hows it going? |
| 03:50.25 | doneir | a custom one, called 'CONT' |
| 03:50.30 | ruben23 | Micc: i have install and registered my gtalk integration already... |
| 03:50.55 | Micc | ruben23, nice. is it working just like you wanted? |
| 03:51.25 | ruben23 | <PROTECTED> |
| 03:51.40 | Katty | LemensTS: ohai to you too |
| 03:51.58 | ruben23 | i was not able yet to used it for my purpose..ive waited for you...:-D |
| 03:51.59 | Micc | ruben23, lookup JabberSend on voip-info.org |
| 03:53.07 | ruben23 | Micc: i have see this----->http://fuhrmannek.de/projects/asterisk/app_jabber.bef--> any idea..? have you used this..? |
| 03:53.45 | Micc | ruben23, not seen that, but I used this page to help me. http://www.voip-info.org/wiki/view/Asterisk+Jabber |
| 03:53.56 | ruben23 | ive look voip-info.org, hmmm there are script im not familliar how to used it.. |
| 03:53.59 | Micc | ruben23, should be examples of dialplan in there. |
| 03:54.37 | doneir | ok, worked it out. Seems due to me doing a Read(CONT,,1) at the same time as a hangup, the Read() must reset the value before it accepts anything from the suer (not wait for suer input, once input found then change value asi assumed) |
| 03:54.43 | Katty | YA"LL ARE SNOBBY |
| 03:54.54 | doneir | changing this around works fine and the variable is viewable in the 'h' extension |
| 03:55.05 | Micc | ruben23, this is the simplest, http://www.voip-info.org/wiki/view/Asterisk+cmd+JabberSend |
| 03:55.34 | Micc | ruben23, does jabber show stats work? |
| 03:56.14 | ruben23 | Micc: ok so i just add this up on my existing inbound dial plan.. |
| 03:56.35 | Micc | ruben23, I mean do jabber show buddies |
| 03:56.50 | ruben23 | what you mean stats work..? i can see its registered...and i have added buddies.. |
| 03:57.34 | Micc | ruben23, ok, then just add jabbersend in your dialplan |
| 03:57.51 | Micc | and use the account you have registered(logged in as client) |
| 03:58.02 | Micc | just follow the syntax in http://www.voip-info.org/wiki/view/Asterisk+cmd+JabberSend |
| 03:58.18 | ruben23 | ok i will try, but how about with the number or caller ID of the incoming call.. |
| 03:59.07 | ruben23 | to display when the gatlk message a buddy about the incoming call with its number also display.. |
| 04:01.46 | LemensTS | katty: ohai heh |
| 04:02.11 | LemensTS | donier: good job |
| 04:05.01 | Katty | http://www.youtube.com/watch?v=jyQgIbEqc1U |
| 04:05.10 | Katty | ^- teh boys and their new playpen |
| 04:07.29 | jaytee | who's speaking in a thick accent? |
| 04:07.38 | Katty | timestamp? |
| 04:07.57 | jaytee | about 1:22 to 1:28 |
| 04:08.34 | Katty | oh |
| 04:08.41 | Katty | that's Creed Assasin 2 on xbox 360 |
| 04:08.55 | jaytee | what's the white ferret's name? |
| 04:09.16 | Katty | that's Merry |
| 04:09.35 | Katty | BB is the super dark one |
| 04:09.45 | Katty | REF: 2:26 |
| 04:09.48 | jaytee | Riddick!!!! |
| 04:09.56 | Katty | :P |
| 04:10.54 | Katty | at least he listens ;) |
| 04:15.28 | *** join/#asterisk florz (nobody@2001:1a50:503c::1) |
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| 04:20.34 | *** part/#asterisk ruben23 (~ITadmin@122.55.48.243) |
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| 04:34.30 | spenguin[work] | hey Katty |
| 04:39.15 | doneir | would a callback be invalid if say, i made a callback to a user after they hungup? So in 'h' extension, i dial the number they called on (with Dial()), then jumped extension to another context to followup? It seemed to fail on me (and also old the original line the user called on from being free) |
| 04:39.31 | doneir | i take it I should be using .call files? |
| 04:40.09 | *** join/#asterisk Dibri (~gavit@190.98.33.229) |
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| 04:53.59 | doneir | basically, the output i get is http://pastebin.ca/1822281 . The user hangs up, it goes to the extension 'h', it Dial()s and then hangs up straight away - no ring tone comes through on the other phone as it's such a fast hangup |
| 04:54.07 | doneir | i take it you can't do callbacks this way? |
| 05:00.02 | doneir | rarr |
| 05:00.06 | doneir | seems not :) |
| 05:08.53 | *** join/#asterisk file (~file@asterisk/developer-and-muffin-lover/file) |
| 05:08.54 | *** mode/#asterisk [+o file] by ChanServ |
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| 05:34.19 | ghento | Hi all. I notice that some of the .call files in my outgoing spool never seem to go through, and just build up a long list of lines containing "DelayedRetry". What could cause this to occur? |
| 05:47.06 | ChannelZ | busy number, bad dial, who knows |
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| 06:01.37 | *** join/#asterisk Kernel_Core (~I@193.222.108.62) |
| 06:01.41 | Kernel_Core | hi all |
| 06:01.48 | Kernel_Core | anybody familiar with wctdm ? |
| 06:02.33 | Kernel_Core | I have a problem in my wctdm.c ( TDM400P card), it detects the incomming RING after 2-3 rings... hence I never receive CALLERID ! |
| 06:08.45 | kaldemar | for many people it works correctly. how have you configured it? |
| 06:16.49 | Kernel_Core | kaldemar: just default , no special config |
| 06:17.25 | Kernel_Core | kaldemar: if wctdm detects the second and 3rd ring , so it should be able to detect the first ring |
| 06:17.58 | kaldemar | there pretty much is no default. |
| 06:19.03 | Kernel_Core | kaldemar: I mean I configured like the samples for FXO lines. |
| 06:20.16 | Kernel_Core | kaldemar: when I debug it , (modprobe wctdm debug=1 ) I see in my debugging that WCTDM detectcs the ringing after 2-3 rings... |
| 06:24.34 | kaldemar | where are you located? |
| 06:24.52 | Kernel_Core | kaldemar: Iran |
| 06:25.30 | kaldemar | check that you have usecallerid=yes and callerid=asreceived above the channel lines in your config. |
| 06:25.35 | Kernel_Core | kaldemar: I used different opermode when I want to wctdm . |
| 06:25.46 | Kernel_Core | kaldemar: it is already set |
| 06:26.11 | kaldemar | then there's cidsignalling parameter, i have no idea what the correct one is for iranian lines. |
| 06:26.25 | Kernel_Core | kaldemar: I have A800P Card ( openvox card ) , with opvxa1200.c driver , it is similar to wctdm.c |
| 06:27.13 | Kernel_Core | in past I had the similar issue with this card... ring was detected after 2-3 rings |
| 06:27.52 | Kernel_Core | but after I used "fwringdetect=1" everything became Okey ! and the card can detect the first ring and hence callerID works okey |
| 06:28.14 | Kernel_Core | but in my TDM400P card , the story is different |
| 06:28.47 | Kernel_Core | if I use or not use fwringdetect=1 it detects the ring after 2-3 rings... |
| 06:30.13 | kaldemar | there is no fwringdetect parameter for dahdi. |
| 06:31.13 | *** join/#asterisk corretico (~laguilar@201.201.46.106) |
| 06:31.22 | Kernel_Core | kaldemar: yea , but when you want to load wctdm driver , there is . |
| 06:34.56 | Kernel_Core | kaldemar: if you read wctdm.c source file you will see this option there . but it doesn't help |
| 06:35.16 | kaldemar | yes, i just checed. |
| 06:37.03 | Kernel_Core | kaldemar: my problem is this : why dahdi ( wctdm driver) detects the ring late ? after 2-3 rings ? |
| 06:43.15 | Kernel_Core | kaldemar: any news ? |
| 06:44.03 | kaldemar | not really, if the problem really is on the wctdm side and not the asterisk channel driver. |
| 06:49.15 | Kernel_Core | kaldemar: it is on wctdm side , because wctdm provides RING state |
| 06:53.53 | *** part/#asterisk codefreeze-lap (~murf@mail.parsetree.com) |
| 07:19.18 | *** join/#asterisk infobot (ibot@rikers.org) |
| 07:19.18 | *** topic/#asterisk is Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.6.2.5, 1.6.1.17, 1.6.0.25 (2010/02/25), 1.4.29.1 (2010/02/18), *-Addons 1.6.2.0 (2009/12/18), 1.6.1.2, 1.6.0.4, 1.4.10 (2009/12/02), dahdi-linux 2.2.1 + dahdi-tools 2.2.1 (2010/01/20), Libpri 1.4.10.2 (2009/10/20) -=- Related channels: #asterisknow,#switchvox,#freepbx,#asterisk-dev,#asterisk-bugs |
| 07:19.53 | Kernel_Core | tzafrir_laptop: yea I am sure |
| 07:20.25 | Kernel_Core | tzafrir_laptop: I used OPVXA1200 ( A800P ) Openvox card , the driver is similar to wctdm.c |
| 07:21.32 | Kernel_Core | tzafrir_laptop: in past I had the same issue with opvxa1200 driver , but after I used fwringdetect=1 in module parameter , the card detects the incomming first ring and callerid works |
| 07:22.11 | Kernel_Core | but for wctdm it doesn't work |
| 07:23.07 | Kernel_Core | tzafrir_laptop: how wctdm detects the incoming ring event ? |
| 07:24.04 | *** join/#asterisk Mezevenf (~mezevenf@ozbizh67.lnk.telstra.net) |
| 07:24.19 | Mezevenf | hello o/ |
| 07:26.35 | *** join/#asterisk Kernel_Core (~I@193.222.108.62) |
| 07:26.40 | Kernel_Core | ops |
| 07:26.43 | Kernel_Core | just got DC |
| 07:26.51 | Kernel_Core | <Kernel_Core> tzafrir_laptop: in past I had the same issue with opvxa1200 driver , but after I used fwringdetect=1 in module parameter , the card detects the incomming first ring and callerid works |
| 07:26.51 | Kernel_Core | <Kernel_Core> but for wctdm it doesn't work |
| 07:26.51 | Kernel_Core | <Kernel_Core> tzafrir_laptop: how wctdm detects the incoming ring event ? |
| 07:27.30 | tzafrir_laptop | Kernel_Core, can you enable debug logging, to see all the events sent by the driver? |
| 07:27.40 | Kernel_Core | tzafrir_laptop: yea I can |
| 07:27.42 | Kernel_Core | wait ... |
| 07:28.03 | tzafrir_laptop | (ringing is detected as a change of line voltage, basically) |
| 07:28.36 | Kernel_Core | tzafrir_laptop: so why my driver detects it after 2-3 rings ? |
| 07:29.18 | Kernel_Core | tzafrir_laptop: my 15$ cheap chinies made phone works perfectly ( for both callerID and ring) but wctdm driver doesn't detect it ?! |
| 07:30.02 | *** join/#asterisk Ad-Hoc (~nimbus@62.169.216.185) |
| 07:31.04 | Mezevenf | if anyone has a spare moment, I'm trying to setup distinctive rings using GXP2000's and tribox 2.6.2.2 with no luck |
| 07:31.37 | Mezevenf | trixbox* |
| 07:33.07 | *** join/#asterisk gidas (~linas@91-65-10-200-dynip.superkabel.de) |
| 07:34.08 | gidas | is there anyway to set pridialplan variable with asterisk-java api? |
| 07:34.44 | Kernel_Core | tzafrir_laptop: here is my debug , I have 2FXO and 2FXS module installed in my tdm400p card http://www.pastebin.ca/1822364 |
| 07:37.21 | *** join/#asterisk c0rnoTa (~c0rnoTa@178.176.140.137) |
| 07:42.07 | *** part/#asterisk c0rnoTa (~c0rnoTa@178.176.140.137) |
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| 07:47.55 | Kernel_Core | tzafrir_laptop:any idea? |
| 07:49.02 | tzafrir_laptop | Kernel_Core, so the 'ring begin' event was too late? |
| 07:49.11 | Kernel_Core | tzafrir_laptop: exactly |
| 07:49.45 | tzafrir_laptop | any chance that this is the beginning of the second ring? (and the first one was missed) |
| 07:51.24 | Kernel_Core | tzafrir_laptop: first and most of the time the second one missed too |
| 07:53.37 | creativx | man |
| 07:53.58 | creativx | i wish we had some red led indicators that could automagically light up on extension hint status |
| 07:57.13 | *** join/#asterisk tamiel (~tamiel@213.30.183.226) |
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| 08:18.15 | Kernel_Core | tzafrir_laptop: any idea ? |
| 08:19.09 | tzafrir_laptop | Kernel_Core, 2.2.1 ? |
| 08:20.00 | Kernel_Core | tzafrir_laptop: yea , but older version have the same behavour |
| 08:20.37 | tzafrir_laptop | we can't really fix those... |
| 08:20.48 | tzafrir_laptop | was it fixed in later versions? |
| 08:20.55 | Kernel_Core | tzafrir_laptop: no |
| 08:21.19 | Kernel_Core | tzafrir_laptop:what can be wrong ? |
| 08:23.04 | tzafrir_laptop | Kernel_Core, is there any open bug for that? |
| 08:23.19 | Kernel_Core | tzafrir_laptop: no ... |
| 08:24.15 | *** join/#asterisk atis_work (~atis_work@193.238.212.171) |
| 08:25.13 | Kernel_Core | tzafrir_laptop: any suggestion ? |
| 08:26.36 | Kernel_Core | tzafrir_laptop: when I set opermode=AUSTRALIA ( which sets boostringer=1 and fwringdetector ) it detects the ring on second ring |
| 08:26.42 | Kernel_Core | not on the 3rd ring. |
| 08:27.21 | tzafrir_laptop | Kernel_Core, I'm not familiar enough with that specific driver. Looks like a bug |
| 08:28.12 | Kernel_Core | tzafrir_laptop: who is familiar with this driver here ? |
| 08:28.49 | Kernel_Core | tzafrir_laptop: I've seen your patches about wctdm.c and it was the reason I asked you , I thought you are familiar with driver |
| 08:30.27 | *** join/#asterisk joobie (~joobie@CPE-121-214-5-126.lnse3.win.bigpond.net.au) |
| 08:30.47 | joobie | hey guys.. im gona write a script that checks the status of 2 peers to see if they are reachable and their response time |
| 08:30.54 | joobie | is there a quick n easy way to get this from the cli? |
| 08:30.59 | joobie | so i can get the script to poll this |
| 08:31.23 | Kernel_Core | tzafrir_laptop: anyway |
| 08:31.23 | kaldemar | joobie: asterisk -rx 'sip show peers' |
| 08:31.28 | Kernel_Core | thank you for your help . |
| 08:31.31 | *** join/#asterisk codefreeze-lap (~murf@mail.parsetree.com) |
| 08:31.41 | joobie | ty |
| 08:44.11 | *** join/#asterisk codefreeze-lap (~murf@mail.parsetree.com) |
| 08:45.56 | *** join/#asterisk gr0mit (~tim@extrt.txrx.org.uk) |
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| 08:51.17 | *** join/#asterisk Polysics (~Luca@host39-66-dynamic.30-79-r.retail.telecomitalia.it) |
| 08:51.18 | Polysics | hello |
| 08:51.37 | Polysics | on a simple Dial extension, is having Answer before and Hangup after required? |
| 08:52.19 | Polysics | so fa i have exten => 10001,1,Dial(SIP/10001,,r) |
| 08:52.32 | Polysics | and the same for 10002, etc |
| 08:53.07 | Polysics | i am asking because of some clients "locking" after a call, becoming unavailable |
| 08:54.15 | ChannelZ | It's not required no, but it may or may not be what you want |
| 08:54.44 | asteriskuser | hi, has everyone asterisk combine with dialogic cards? |
| 08:55.56 | Polysics | ChannelZ, so far I just need people to call each other |
| 08:56.05 | Polysics | what would Anser and Hangup add? |
| 08:56.30 | *** join/#asterisk angryuser (~angryuser@LPuteaux-151-42-27-99.w193-251.abo.wanadoo.fr) |
| 08:56.36 | ChannelZ | well Answer makes * 'pick up' the channel, even though you're turning around and dialing another device. |
| 08:56.46 | *** join/#asterisk krion (~seb@unaffiliated/krion) |
| 08:57.08 | ChannelZ | But from the calling device's point of view, the call is connected at that instant even though both sides aren't yet connected |
| 08:58.24 | ChannelZ | As for Hangup, it's not really necessary unless autofallthrough is turned off, however once the person physically hangs up the call should be torn down anyway. |
| 09:00.00 | Polysics | so,in a troubleshooting/learning phase, it's better to have em? |
| 09:01.01 | ChannelZ | well Hangup sure, Answer again not necessarily |
| 09:04.10 | *** join/#asterisk romain (~romain@peerfuse.org) |
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| 09:10.10 | joobie | guys the ms that reports when you do 'sip show peers' - what is that? because if i ping it never reflects anything near to what that ms value in asterisk reports |
| 09:12.00 | kaldemar | joobie: it's calculated from a SIP options message, not ping. |
| 09:12.10 | Tim_Toady | its the network rrt plus the time it takes for the peer to reply to the sip notify |
| 09:12.46 | ChannelZ | it's the number of pesos you owe me every time you look |
| 09:14.07 | joobie | ahh |
| 09:14.14 | joobie | so it's like a heartbeat type query |
| 09:14.16 | joobie | sending via sip? |
| 09:14.21 | joobie | measuring the rtt of that? |
| 09:14.39 | joobie | .. the thing is, pennytel for example at times i get 1500ms back from them.. but when i ping, i get aorund 50ms |
| 09:14.42 | joobie | and the calls are fine |
| 09:14.51 | joobie | so it's like that check goes way overboard with its rtt that's not real |
| 09:15.01 | ChannelZ | re: it's the time it takes * to send out a message and for the device to reply |
| 09:15.28 | ChannelZ | it's higher level than a 'ping' so they will never match |
| 09:15.41 | joobie | but 1500ms |
| 09:15.42 | joobie | that's huge |
| 09:15.53 | joobie | if that were actually the case i'm guessing calls will be poor quality |
| 09:15.54 | joobie | but calls are fine |
| 09:15.57 | ChannelZ | Perhaps it's low priority to the device. |
| 09:15.57 | kaldemar | joobie: it's real for SIP |
| 09:16.17 | joobie | my point is - if 1500ms were acurate why am i not seeing poor call quality? |
| 09:16.22 | ChannelZ | the packets going back and forth are 10X larger than a ping too |
| 09:16.42 | joobie | ChannelZ, ping aside - purely looking at the ms reporting in asteirsk and the call quality |
| 09:16.53 | joobie | 1500ms being reported with no side-effect to call quality |
| 09:17.01 | joobie | leads me to believe it's not a useful figure to look at |
| 09:17.02 | kaldemar | joobie: and SIP only carries signaling, not audio |
| 09:17.02 | ChannelZ | you should stop worrying because you're trying to make a conclusion from a bad assumption |
| 09:17.03 | joobie | ? |
| 09:17.50 | joobie | ahh |
| 09:18.01 | joobie | so really that ms is a measure of the channelling speed |
| 09:18.33 | joobie | ChannelZ, im not worried bro,.. just working on some scripts to automate monitoring of my asterisk box |
| 09:18.47 | joobie | i get the sip status .. was wondering if the ms response is worthwhile but it aint really |
| 09:18.58 | ChannelZ | it's a measure of how long it took a SIP options request to reach the device and for the device to reply. |
| 09:19.01 | joobie | is there a factor i can grab from asterisk to monitor the "call quality" of a sip call? |
| 09:19.07 | joobie | yea |
| 09:19.10 | joobie | i understand now |
| 09:19.18 | joobie | <kaldemar> joobie: and SIP only carries signaling, not audio |
| 09:19.43 | joobie | i think that is key - it makes sense why i can get a 1500ms response but have good call quality.. the signaling channel is lagging but not the audio |
| 09:19.55 | joobie | can asterisk somehow report on the audio quality of a sip call? |
| 09:22.04 | *** join/#asterisk nextime (~nextime@unaffiliated/nextime) |
| 09:22.38 | nextime | hello all. Is something changed in how asterisk 1.6.2 manage time based inclusion of context in extension.conf? |
| 09:22.40 | ChannelZ | I don't remember if 'core show channel' shows you any jitter info on a sip call |
| 09:23.38 | nextime | i have something like include => unixmediaopen|09:00-12:59|mon-fri|*|* |
| 09:23.43 | nextime | in one context |
| 09:23.58 | nextime | and some rules in [unixmediaopen] context using s extension |
| 09:24.41 | nextime | when i Goto() to the first context that include unixmediaopen, the s extension is never matched even if the include statement should include the unixmediaopen context |
| 09:24.54 | nextime | it was working before the upgrade to ast 1.6.2 |
| 09:24.56 | *** join/#asterisk af_ (~getsmart@88-149-230-64.dynamic.ngi.it) |
| 09:25.18 | Tim_Toady | joobie check ${RTPAUDIOQOS} |
| 09:27.55 | joobie | thanks Tim_Toady |
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| 10:46.34 | kruemeltee | hello all together :-) |
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| 10:49.20 | Systemt` | hey |
| 10:49.26 | Systemt` | some one can help me please? |
| 10:50.55 | miloux | depends on your question |
| 10:52.31 | Systemt` | i have audio problem on my astrisk |
| 10:54.16 | Systemt` | ? |
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| 11:05.52 | Akiraa | Is there a guide on how to configure port forwarding for multiple SIP devices sitting on the same LAN (behind NAT)? |
| 11:07.23 | Systemt` | i have the same problem |
| 11:07.26 | Systemt` | no audion : |
| 11:07.27 | Systemt` | :\ |
| 11:08.35 | Systemt` | what router do u have |
| 11:08.37 | Systemt` | ? |
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| 11:10.07 | Akiraa | Systemt`: a 4 port home router behind an ADSL modem (behind NAT) |
| 11:10.35 | Systemt` | what model ? |
| 11:10.41 | Systemt` | mabe i can help u ? |
| 11:11.54 | Systemt` | u need to connect the router and open ports |
| 11:12.03 | Systemt` | but what model is your rauter ? |
| 11:12.28 | *** join/#asterisk arossouw (~arossouw@dsl-146-50-11.telkomadsl.co.za) |
| 11:12.56 | arossouw | hi is there a way to monitor call quality, when making calls via a sip peer? |
| 11:13.16 | Akiraa | Systemt`: the IPPBX uses ports 5060 for SIP messages and ports 10000-10100 for RTP |
| 11:13.28 | Systemt` | ok |
| 11:13.29 | Systemt` | listen |
| 11:13.34 | Akiraa | now, I can forward these ports from each individual machine |
| 11:13.41 | Systemt` | listen |
| 11:13.45 | Akiraa | but not sure how to have 2 phones or more on the same LAN |
| 11:13.52 | Systemt` | so |
| 11:13.58 | Systemt` | aha |
| 11:13.59 | Systemt` | : |
| 11:14.01 | Systemt` | :\ |
| 11:14.20 | Akiraa | say, forward 5060 and 10000-10100 to 192.168.1.2 |
| 11:14.22 | Systemt` | u can open for boute computers |
| 11:14.33 | Systemt` | out |
| 11:14.36 | Systemt` | or open NAT |
| 11:14.41 | kaldemar | Akiraa: just make them register to asterisk and configure them as qualify=yes in sip.conf. |
| 11:15.15 | kaldemar | Akiraa: they will use different ports on the NAT and asterisk will send packets to keep the connection alive. |
| 11:15.21 | kaldemar | ~sipnat |
| 11:15.22 | infobot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 11:15.27 | Systemt` | kaldemar: i have the same problem just my problem is NO audio |
| 11:15.51 | Systemt` | \i have |
| 11:15.56 | Systemt` | IAX2 and SIP |
| 11:16.13 | Systemt` | IAX2 is working good |
| 11:16.17 | Systemt` | but SIP is no audio |
| 11:16.26 | Akiraa | Systemt`: definitely a NAT problem |
| 11:16.40 | Systemt` | yea |
| 11:16.44 | Systemt` | but i open DMZ |
| 11:16.47 | Systemt` | to my server |
| 11:17.26 | kaldemar | Systemt`: look at the same tutorial |
| 11:17.35 | Systemt` | ok |
| 11:17.50 | kaldemar | forwarding ports it not enough. you have to make asterisk aware of the network setup. |
| 11:20.20 | Systemt` | where can i find it ? |
| 11:20.44 | kaldemar | the link 15 lines up |
| 11:20.48 | Systemt` | my asterisk is otking on dhcp |
| 11:23.54 | Akiraa | kaldemar: So with qualify=yes, there is no need for port forwarding on the phone's LAN, right? |
| 11:24.19 | kaldemar | Akiraa: shouldn't be. |
| 11:32.04 | Systemt` | <PROTECTED> |
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| 11:37.14 | Tirael | hi guys |
| 11:38.16 | Tirael | Anybody have connectivity Nortel CS2000 with * ver. 1.6 ? |
| 11:38.36 | kaldemar | Systemt`: pastebin your configuration and a sip debug of a failed call |
| 11:38.40 | kaldemar | ~pb |
| 11:38.41 | infobot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , or apt-get install pastebinit |
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| 11:41.50 | Tirael | Sometimes I can't hear other side, this problem occurs with 50% chance |
| 11:42.14 | Tirael | Occurs only with Nortel CS2000 |
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| 11:58.53 | hellop | howdy boys |
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| 12:00.25 | hellop | Anyone wanna buy some Polycom 501s and a 802.11 wireless sip phone? |
| 12:01.57 | Systemt` | what price? |
| 12:03.29 | hellop | cheaper than the $160 I paid. |
| 12:04.37 | hellop | woah the price has really come down for these |
| 12:05.01 | hellop | $70 each I suppose. |
| 12:06.25 | Systemt` | what about shipping ? |
| 12:15.55 | hellop | $10 |
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| 12:16.21 | hellop | Have 2 501s and one 500 |
| 12:16.32 | hellop | I'll keep my budgetone. ;) |
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| 12:21.07 | lbarth | hello |
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| 12:41.57 | adnc | sometimes my extensions.conf file is busy and i can not open it with my editor. has someone got an idea why this could be the case? |
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| 12:49.04 | leifmadsen | I don't understand the question |
| 12:49.15 | leifmadsen | what editor |
| 12:49.47 | leifmadsen | Asterisk wouldn't lock the file, so it'd either be an external script trying to write to the file, or if you're using vim and didn't exit correctly, there could be a .extensions.conf.tmp file (or whatever vim calls it) |
| 12:49.58 | leifmadsen | additional information required; however, Asterisk wouldn't lock the file |
| 12:50.33 | beek | adnc: when that happens, try: /usr/sbin/lsof | grep extensions.conf to see what has that held open. |
| 12:50.37 | beek | mornin' leifmadsen |
| 12:50.41 | leifmadsen | beek: zup yo? |
| 12:51.13 | leifmadsen | welp, it's time for me to reboot for my morning documentation writing session with Jim |
| 12:52.53 | adnc | beek, ohh, i did look with that but the file does not seem to be open |
| 12:53.11 | adnc | beek i found out that if i look with my regular vim than it works, but vi doesnt. i'm on a debian |
| 12:53.23 | ttwhy | Hi, can someone tell me how to active 16khz speex? I just get the 8 khz version running (using ekiga as softphone) |
| 12:55.12 | beek | adnc: If you use vi in one session to edit the file, then open another session and try to edit it, vi will have it locked in session 1. Perhaps that's what you're experiencing. |
| 13:00.59 | adnc | beek, that wasnt the case, than vi would add a .file.swp |
| 13:01.04 | adnc | strange, well... |
| 13:02.10 | jaytee | mornin beek |
| 13:02.20 | beek | adnc: What is the *exact* error message that you receive? |
| 13:03.14 | beek | mornin' jaytee |
| 13:03.27 | beek | jaytee: How fare the animals? |
| 13:03.46 | jaytee | good I guess, I'm no longer employed there |
| 13:04.22 | beek | Really? When did this change occur? |
| 13:04.41 | jaytee | yesterday |
| 13:05.32 | [TK]D-Fender | jaytee: Whose choice? |
| 13:05.41 | jaytee | not mine |
| 13:05.51 | [TK]D-Fender | jaytee: Sorry to hear... |
| 13:06.02 | jaytee | thanks, both of ya |
| 13:06.41 | jaytee | I should be starting a new job probably next monday or the week after |
| 13:06.48 | beek | That's good news. |
| 13:07.03 | [TK]D-Fender | \o/ |
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| 13:09.59 | dlynes | adnc, could be whatever version of vi you're using is using a different lock file than .extensions.conf.swp |
| 13:10.16 | dlynes | adnc, that's what vim uses, but not necessarily every vi implementation |
| 13:10.29 | dlynes | adnc, There's also nvi, elvis, ... |
| 13:11.21 | [TK]D-Fender | codes with BUTTERFLIES |
| 13:11.23 | dlynes | adnc, also, is vim opening it in read-only mode, or regular mode? |
| 13:13.16 | jaytee | [TK]D-Fender, does that mean that if you hit a key in BUTTERFLIES it causes a typhoon half way across the world? |
| 13:13.38 | [TK]D-Fender | jaytee: http://xkcd.com/378/ |
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| 13:14.42 | jaytee | hehe, yeah I remember that one |
| 13:15.02 | beek | I love that one. |
| 13:15.17 | dlynes | heh |
| 13:15.39 | jaytee | this one is my all time favorite: http://xkcd.com/418/ |
| 13:15.42 | dlynes | just another reason to dislike emacs :) |
| 13:16.07 | beek | dlynes: I don't need another reason! |
| 13:17.23 | jaytee | it was a bug in one of the early versions of GNU Emacs that let Robert Morris's worm cripple the internet back in 1988 |
| 13:17.42 | adnc | dlynes, i use a debian derivate called voyage, where i have my asterisk on it. it comes with a vi clone (i suppose) also i installed vim (the tiny version) unfortunately the installation of vim did not change the binary executable-link from vi to vim |
| 13:18.23 | adnc | ahh, i see that it is elvis-tiny that is causing this problem. unfortunately there is no lockfile .extensins.conf.swp |
| 13:18.56 | adnc | but the installation of vim-common and vim-tiny should repoint to the newly installed vim binary. strange |
| 13:19.40 | adnc | that is the problem vi -> /bin/elvis-tiny |
| 13:19.49 | adnc | vim -> /usr/bin/vim.tiny |
| 13:20.02 | dlynes | adnc, why do you automatically assume that installation of vim should create a symbolic link from vi to vim? |
| 13:20.27 | dlynes | adnc, perhaps that is not voyage's intent |
| 13:20.38 | adnc | dlynes, i thought that would be the case, since vim is a vi clone and installation of vim should replace it. but obviously you are right and it doesnt |
| 13:21.07 | dlynes | adnc, where is your vi symbolic link? is it in /usr/bin or /bin? |
| 13:21.19 | adnc | in /etc/laternatives |
| 13:22.06 | dlynes | adnc, eh? how does /etc/alternatives get into your path? |
| 13:22.27 | ManxPower-work | I use the "joe" editor, the one true editor |
| 13:22.40 | dlynes | adnc, or is there a symbolic link /usr/bin/vi that points to /etc/alternatives/vi? |
| 13:22.52 | adnc | dlynes, it is not in my path, it must be somehow differently managed |
| 13:22.58 | dlynes | adnc, or /bin/vi or that matter? |
| 13:23.06 | tzafrir_laptop | adnc, just install 'vim' |
| 13:23.14 | tzafrir_laptop | vim-common is a helper package |
| 13:23.16 | adnc | yes, /usr/bin/vi is pointint to /etc/alternatives/vi |
| 13:23.34 | dlynes | adnc, anyways... rm -f /etc/alternatives/vi && ln -s /usr/bin/vim-tiny /etc/alternatives/vi |
| 13:23.44 | adnc | tzafrir_laptop, i know, but that packages asks for 25mb of space, and i'm on a cf card where i do not really want to wast that area |
| 13:23.46 | dlynes | adnc, that'll fix your symbolic link for you |
| 13:24.10 | dlynes | adnc, you're trying to run asterisk on cf? |
| 13:24.19 | adnc | dlynes, it is running on cf |
| 13:24.38 | adnc | for the last two weeks |
| 13:24.44 | dlynes | adnc, i'm guessing you're not overly concerned about call quality? or flash memory on cf has gotten a lot faster? |
| 13:25.03 | tzafrir_laptop | adnc, so just use 'update-alternative --config' |
| 13:25.15 | tzafrir_laptop | uses busybox vi on such systems :-) |
| 13:25.28 | adnc | dlynes, i did not understand. |
| 13:25.31 | dlynes | tzafrir_laptop, busybox has vi as a built-in? |
| 13:25.44 | tzafrir_laptop | A rather minimal one, but yes |
| 13:25.51 | adnc | tzafrir_laptop, thats it! thanks |
| 13:26.09 | dlynes | adnc, I remember a few years ago I was using cf, and it was damned slow...has it gotten considerably faster? I don't know if it was a memory speed limitation, or the cf interface... |
| 13:26.16 | adnc | dlynes, i can not get the relation between call quality and flash memory on cf? |
| 13:26.30 | adnc | dlynes, it really works great |
| 13:26.46 | dlynes | adnc, how many simultaneous calls are you handling? |
| 13:27.07 | dlynes | adnc, and do you have full logging enabled? |
| 13:27.13 | adnc | dlynes, it is for my home usage and maximum two simultaneous calls |
| 13:27.22 | adnc | dlynes, no, not full logging |
| 13:27.27 | dlynes | adnc, ah...ok...thought you were using it in a production system |
| 13:27.44 | adnc | dlynes, production for my home ;) |
| 13:27.57 | adnc | i think for large installations this would be not a really good solution |
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| 14:30.36 | Diffen2 | Hello. Are there any tools that get information from asterisk. i mean how many extensions, how many queues and so on? |
| 14:31.28 | ManxPower-work | Diffen2, you can always do a "dialplan show" to get most of that information |
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| 14:32.16 | Diffen2 | ManxPower thank you |
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| 14:32.55 | ManxPower-work | Diffen2, You should read the Asterisk Book |
| 14:32.58 | ManxPower-work | ~book |
| 14:32.59 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
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| 14:34.12 | ManxPower-work | We need the bot to do an on join message "Using a GUI? GTFO!" |
| 14:35.39 | Diffen2 | ManxPower do you know if its possible to export the dialplan show to a file? |
| 14:38.55 | ManxPower-work | Diffen2, Why? It's all there in extensions.conf |
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| 14:39.28 | ManxPower-work | if you really must, then this should work: asterisk -rx "dialplan show" > dialplan.txt |
| 14:39.58 | Diffen2 | thanks man |
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| 15:05.05 | nickfennell | Hey |
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| 15:05.36 | nickfennell | Anyone familiar with RTP stream policy when a call is placed on hold ? |
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| 15:19.57 | aureliheinb | Hello, does SIP MESSAGE(rfc3261) could path trough Asterisk server ???because it doesnot work for me :-( |
| 15:20.31 | ManxPower-work | aureliheinb, That is expected. Only a SIP Proxy would pass all the headers, Asterisk is not a SIP proxy |
| 15:21.32 | aureliheinb | I don't get it ? SIp MESSAGE are standard messages, they are not headers right ? |
| 15:22.21 | Naikrovek | asterisk doesn't forward packets, it's not a proxy |
| 15:22.32 | Naikrovek | asterisk is a source and destination, not a proxy |
| 15:22.40 | Naikrovek | if i'm understanding your question properly |
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| 15:23.00 | aureliheinb | OK I see what you mean but why Asterisk do not support MESSAGE ? |
| 15:23.02 | [TK]D-Fender | aureliheinb: * is not a PROXY and * does not do messaging <- |
| 15:23.30 | aureliheinb | Actually I am using MESSAGE to give some configurations to endpoints . . . |
| 15:23.39 | [TK]D-Fender | aureliheinb: Doesn't matter. * does not do this |
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| 15:25.00 | aureliheinb | do you have any advice to be able to send parameters during an audio/video communication ? |
| 15:26.01 | jpcansa | how can i restrict sip connections to IPs from my country only?? |
| 15:27.38 | [TK]D-Fender | jpcansa: Run a proxy in front that can check because * can't. Or get ready to manually put in a TON of ranges you consider "valid" |
| 15:28.01 | jpcansa | i see |
| 15:33.03 | aureliheinb | so no advice to send parameters by * ??? a header that could contain parameters ??? |
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| 15:33.42 | [TK]D-Fender | aureliheinb: No generic heard while in a call. |
| 15:33.52 | [TK]D-Fender | aureliheinb: An additional header when PLACING a call? Sure |
| 15:35.03 | aureliheinb | add a header cannot make * work in a wierd way ? |
| 15:37.56 | [TK]D-Fender | aureliheinb: ? |
| 15:39.23 | smooth_penguin | jpcansa, iptables has a geoip module |
| 15:41.20 | jpcansa | smooth_penguin, how that works?? |
| 15:41.33 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 15:41.33 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
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| 15:47.11 | ManxPower-work | aureliheinb, Take a look at the SendText application. I do not remember how it sends messages to SIP phones. I've never personally ever gotten it to do anything other than crash my phone. |
| 15:48.11 | ManxPower-work | you can also look at the "sipsak" program (NOT part of Asterisk) that lets you send SIP packets to a phone |
| 15:48.38 | mort_gib | ManxPower-work: I got it to work with Snoms |
| 15:48.48 | jpcansa | what i really need is to secure my * box, someone not auth is connecting via sip and makin calls. the source on the CDR shows this: SIP/113.105.152.34-0000681b |
| 15:48.54 | mort_gib | Annoying, and useless but works all the same |
| 15:49.23 | *** join/#asterisk smooth_penguin (~smoove@59.95.0.242) |
| 15:49.51 | smooth_penguin | jpcansa, http://www.debian-administration.org/articles/518 |
| 15:50.04 | [TK]D-Fender | jpcansa: And who is the call authing as? |
| 15:51.22 | jpcansa | [TK]D-Fender: SIP/113.105.152.34-0000681b is shown as the channel |
| 15:51.26 | *** join/#asterisk Faithful (~Faithful@202.6.145.116) |
| 15:51.40 | [TK]D-Fender | jpcansa: Sometimes the call can match a peer and still look like that. |
| 15:51.44 | [TK]D-Fender | jpcansa: go CONFIRM |
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| 15:52.10 | [TK]D-Fender | jpcansa: and if that is an actually un-authed call, then you'd better look at why you're even allowing it |
| 15:53.22 | jpcansa | i´ll check my contexts |
| 15:53.40 | [TK]D-Fender | jpcansa: go look at your SIP DEBUG |
| 15:53.45 | jpcansa | ok |
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| 15:56.46 | *** join/#asterisk wcselby (~wcselby@216.110.88.194) |
| 15:58.24 | wcselby | o/ |
| 15:58.58 | Skeeter- | ManxPower-work, the coding for the Microbroswing thing is pretty nasty, there is no logic, and the coding is old |
| 15:59.09 | *** join/#asterisk krynnotaur (~krynnotau@unaffiliated/krynnotaur) |
| 15:59.19 | Skeeter- | u gotta try until it work, and when is does, it almost doesnt make any sense |
| 15:59.19 | ManxPower-work | Skeeter-, welcome to my world |
| 15:59.44 | Skeeter- | deveolping a webpage isnt the hard part |
| 16:00.11 | *** join/#asterisk RobH__ (~robh@216.38.133.254) |
| 16:02.04 | Skeeter- | making polycoms to read it is another thing, BUT the potential is there for sure |
| 16:02.18 | *** join/#asterisk Amorsen (~Amorsen@94.127.50.7) |
| 16:02.38 | Skeeter- | just getting the wheater from google to work is pretty hard, ivent got it to work yet |
| 16:03.38 | adnc | when i use Goto does it go back to where it was called after finishing the extension it was asked to go? |
| 16:03.49 | *** join/#asterisk cweagans (~432aa645@gateway/web/freenode/x-usojbuxkswkqazab) |
| 16:03.49 | wcselby | no |
| 16:04.53 | cweagans | is there a way that I can set up one of my SIP extensions to dial out to another number? For example: 208-123-4567 ex.201 == 206-987-6543 |
| 16:05.10 | wcselby | cweagans - sure |
| 16:05.38 | wcselby | cweagans - just set the dialplan for ext 201 to Dial(${TECHNOLOGY}/2069876543) |
| 16:05.39 | Skeeter- | ManxPower-work, what kinda thing did u accomplish |
| 16:05.44 | Diffen2 | Im thinking about do some stress test on my asterisk server. are there any application you can recommend? |
| 16:06.02 | wcselby | where ${TECHNOLOGY} is the call tech used, ie. SIP, or DAHDI/g1, etc |
| 16:06.15 | *** join/#asterisk LemensTS (~LemensTS@adsl-70-238-161-207.dsl.stlsmo.sbcglobal.net) |
| 16:06.25 | ManxPower-work | Diffen2, leave your server unsecured, then just wait. You'll get some very good stress testing. |
| 16:06.25 | LemensTS | . |
| 16:06.30 | wcselby | adnc - I think you want gosub |
| 16:06.41 | cweagans | wcselby: well...I'm not sure what to set there then. We have our SIP trunks with Speakeasy. So would that just be SIP/2089876543? |
| 16:07.00 | Diffen2 | manxpower hehe thats the thing i want to avoid |
| 16:07.01 | cweagans | <-- noob. |
| 16:07.15 | adnc | i would like to set some variables like Set(CALLFILENAME=${EXTEN}) on all extensions do i need to write this for all extensions or can i set this somewhere globaly. since Goto does not get back, i need a different way |
| 16:07.19 | wcselby | cweagans - it would be SIP/(speakeasy_peer)/208... |
| 16:07.38 | ManxPower-work | adnc, you want Gosub |
| 16:07.53 | wcselby | cweagans - you do it like any normal outbound call, just with a predefined phone number instead of using a variable. |
| 16:08.07 | cweagans | huh |
| 16:08.21 | cweagans | well that's easier than I thought it was going to be :) |
| 16:08.22 | aureliheinb | thank you ManxPower-work I am looking at it |
| 16:08.23 | adnc | ManxPower-work, thank you |
| 16:08.29 | cweagans | wcselby: thanks! |
| 16:08.33 | cweagans | wcselby++ |
| 16:08.38 | wcselby | :) |
| 16:10.07 | adnc | core show function Gosub doesnt show anything, strange. |
| 16:10.12 | *** join/#asterisk Warp4 (~robert.wo@firewall-a.buf.ny.i-evolve.net) |
| 16:10.19 | wcselby | core show application gosub |
| 16:10.37 | wcselby | adnc - core show application gosub |
| 16:10.53 | adnc | thanks, but why is gosub an application? |
| 16:10.57 | bmoraca_work | curious...i just noticed something about 1.6.2...the log files have illegible characters where the color changes in the CLI would be...anyone know how to turn that off or is it fixed in a later version? |
| 16:11.12 | wcselby | adnc - because it's an application....usable from the dialplan |
| 16:11.21 | [TK]D-Fender | adnc: same reason GOT is an application |
| 16:11.22 | bmoraca_work | adnc: because gosub is itself a dialplan application, not a function usable from within another application |
| 16:11.26 | [TK]D-Fender | GOTO* |
| 16:11.34 | adnc | i see |
| 16:12.22 | [TK]D-Fender | adnc: Gosub does not return a value. It is not the structured programming equivalent to a "function call" |
| 16:13.52 | ManxPower-work | Your best bet is to set a variable in your subroutine and access it from your main code. |
| 16:13.55 | adnc | [TK]D-Fender, i need to jump somewhere, set a variable and get back again |
| 16:14.27 | ManxPower-work | adnc, that is one of the most basic things in Asterisk. Why don't you look for examples on the Wiki. |
| 16:14.37 | ManxPower-work | Or even in the extensions.conf.sample |
| 16:14.40 | [TK]D-Fender | adnc: then do it |
| 16:14.44 | *** join/#asterisk tkrn (~tkrn@216.196.213.32) |
| 16:14.48 | [TK]D-Fender | adnc: Variables have no scope in the dialplan <- |
| 16:14.51 | ManxPower-work | or, I suspect the Asterisk Book |
| 16:15.26 | wcselby | [TK]D-Fender - if he's incrementing a global variable or something, he could do it...alhtough why he'd need a gosub to increment a global variable... |
| 16:15.30 | adnc | ManxPower-work, i was suggested using gosub, since goto doesnt come back |
| 16:15.41 | [TK]D-Fender | adnc: So do it |
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| 16:17.47 | ttwhy | Hi, does someone know why a G722 codec could stop every ~30seconds for a short time? cpu usage is totaly low and i cant see any process which should corrupt the call |
| 16:17.54 | bmoraca_work | gosub comes back and can return a value |
| 16:18.37 | bmoraca_work | hrm |
| 16:18.41 | bmoraca_work | i take that back |
| 16:19.20 | bmoraca_work | ahh |
| 16:19.25 | bmoraca_work | GOSUB_RETVAL |
| 16:19.28 | bmoraca_work | that's kind of obtuse |
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| 16:25.59 | wcselby | bmoraca_work - i thought gosub_retval was just a 0 or -1...? |
| 16:26.25 | *** join/#asterisk timeshell (~chatzilla@gw.lusi.on.ca) |
| 16:26.32 | wcselby | ahhh, 1.6 returns an actual value |
| 16:26.40 | wcselby | wish this headache would go away |
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| 16:33.47 | timeshell | bans all headaches with a large troug |
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| 16:36.53 | ChannelZ | is that a blunt instrument? |
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| 16:48.21 | leifmadsen | shhh |
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| 16:49.44 | *** join/#asterisk dinesh___ (~dinesh@84-73-120-175.dclient.hispeed.ch) |
| 16:50.19 | wcselby | is there a way to setup custom hostmasks with freenode? |
| 16:51.16 | dinesh___ | how to detect when a call was unanswered? it seems that either $["${DIALSTATUS}" = "CHANUNAVAIL"] or $["${DIALSTATUS}" = "CONGESTION"] is true in that case |
| 16:51.19 | dinesh___ | which i did not except |
| 16:51.27 | dinesh___ | expect* |
| 16:51.31 | adnc | i use exten => 88,1,VoiceMailMain(s${CALLERIDNUM}) but still it asks for mailboxnumber, also i tried exten => 88,1,VoiceMailMain(${CALLERIDNUM}|s) |
| 16:51.53 | wcselby | adnc - use Voicemail(), not VoicemailMain() |
| 16:52.21 | wcselby | also, adnc, not sure you want to use ${CALLERIDNUM}....unless you've specifically set that variable |
| 16:52.40 | wcselby | ${CALLERID(num)} is the proper way to do it, unless you're using an older version of asterisk |
| 16:53.06 | wcselby | I tend to use Voicemail(${EXTEN},u), because then it goes to the voicemail box that was called, not the voicemail box of the caller... |
| 16:53.52 | wcselby | well, Voicemail(${EXTEN}@default,u) |
| 16:54.37 | ManxPower-work | dinesh___, NOANSWER |
| 16:55.06 | ManxPower-work | that happens when the call is not answered within timeout. CANCEL means caller hung up while the call was still ringing |
| 16:55.29 | dinesh___ | well, there must be some error in my dialplan then |
| 16:55.33 | leifmadsen | exten => voicemail,1,Voicemail(${requestedExtension}@${GLOBAL(voicemail_context)},${IF($[${DIALSTATUS} = BUSY]?b:u)}) |
| 16:56.24 | paulc | That's dialplan ninja action - right there! :-) |
| 16:56.37 | dinesh___ | :P |
| 16:57.53 | dinesh___ | http://pastebin.com/EuR5g3Ae any idea why it it reaches line 11 when the call is simply unanswered? |
| 16:58.12 | dinesh___ | oh, it's BUSY and not "BUSY" ? |
| 16:58.40 | wcselby | or you could use leifmadsen - that's awesome, how long have you been able to use the ${IF()} function like that? |
| 16:59.06 | wcselby | ignore the "or you could use" part of my last comment....lol |
| 16:59.08 | leifmadsen | wcselby: since it was created |
| 16:59.16 | *** join/#asterisk anonymouz666 (~anonymouz@189.24.39.56) |
| 16:59.23 | leifmadsen | wcselby: it works just like all other dialplan functions that can be used inside of applications |
| 16:59.32 | leifmadsen | wcselby: I just have a clever usage of IF() that other people haven't thought of |
| 16:59.34 | idespinner | is there a clear explanation of using conditionals and variables and when one should use $[], ${} or $()? |
| 16:59.39 | dinesh___ | "since I created it" would sound even better :) |
| 16:59.48 | *** join/#asterisk waa (~waa@balrog.credipar.com.br) |
| 17:00.48 | leifmadsen | idespinner: $[ ] <-- expressions for comparing things. ${ } <--- used when you want to read the value of something. $( ) <--- never used |
| 17:01.09 | wcselby | leifmadsen - lol, neato |
| 17:01.45 | idespinner | thank you, that $() was a trick to make sure you werent BSing me |
| 17:01.57 | leifmadsen | :D |
| 17:01.58 | idespinner | jots this down |
| 17:02.03 | leifmadsen | I never BS (except when I do) |
| 17:02.25 | leifmadsen | $[ ] used in things like GotoIf(), ExecIf(), etc... |
| 17:02.50 | idespinner | so... Gotoif(${a}=${b}) is bad... |
| 17:03.10 | idespinner | but GotoIf($[${a}=${b}]) is good? |
| 17:03.16 | leifmadsen | aye |
| 17:03.37 | idespinner | so all the conditionals require a boolean which $[] creates |
| 17:03.45 | leifmadsen | put something like double quotes around your variables if your ${a} or ${b} could potentially be empty |
| 17:03.58 | leifmadsen | yes, $[ ] will return math or booleans |
| 17:04.04 | dinesh___ | oh cool that was it, "BUSY" is wrong, but BUSY works just fine ;) |
| 17:04.09 | leifmadsen | Set(RESULT=$[${RESULT} + 1]) |
| 17:04.28 | leifmadsen | dinesh___: "BUSY" is fine as long as your variable was also wrapped with " " |
| 17:04.33 | idespinner | err ok wait, so if ${result} was null, that wouldnt work without quotes? |
| 17:05.04 | leifmadsen | the " " is not like in programming languages, the double quotes are literal in the string, so you have to have them on both sides because you want to see if the strings are both the same |
| 17:05.18 | leifmadsen | idespinner: well, not quite, let me show you |
| 17:05.28 | leifmadsen | Set(RESULT=$[ + 1]) |
| 17:05.38 | leifmadsen | that's what Asterisk sees if ${RESULT} was not set previously |
| 17:05.41 | leifmadsen | so you could fix that with |
| 17:05.49 | leifmadsen | Set(RESULT=$[0${RESULT} + 1]) |
| 17:05.54 | bmoraca_work | hax |
| 17:05.57 | leifmadsen | because then with ${RESULT} is null, you get |
| 17:06.03 | leifmadsen | Set(RESULT=$[0 + 1]) |
| 17:06.16 | leifmadsen | if you're comparing strings vs. doing math, then the quotes work like this |
| 17:06.37 | leifmadsen | GotoIf($["${RESULT}" = "B"]?...) |
| 17:06.38 | idespinner | ah that makes sense |
| 17:06.44 | leifmadsen | if ${RESULT} is null, Asterisk sees |
| 17:06.54 | leifmadsen | GotoIf($["" = "B"]?...) |
| 17:06.59 | *** join/#asterisk raden_work (~jon@69-179-99-17.stat.centurytel.net) |
| 17:07.08 | leifmadsen | you need the quotes on both sides, because if ${RESULT} was B and you didn't have quotes on both sides |
| 17:07.16 | leifmadsen | GotoIf($[B = "B"]?...) |
| 17:07.23 | leifmadsen | that would return "not equal" |
| 17:07.26 | ManxPower-work | and B is NEVER equal to "B" |
| 17:07.29 | leifmadsen | exactly |
| 17:07.33 | idespinner | what would $[ =B] result in? |
| 17:07.37 | leifmadsen | idespinner: error |
| 17:07.48 | leifmadsen | you can't have a missing value on either side of the operator |
| 17:07.57 | leifmadsen | you'll get a WARNING message on the console |
| 17:08.26 | idespinner | i have to be honest, it make sense but seems a little cludgey |
| 17:08.32 | leifmadsen | not really |
| 17:09.04 | leifmadsen | you could just as easily do: |
| 17:09.14 | leifmadsen | GotoIf($[X${RESULT} = XB]?...) |
| 17:09.20 | leifmadsen | I just find double quotes easier |
| 17:09.31 | leifmadsen | but you must ALWAYS have information on both sides of the operator |
| 17:09.40 | leifmadsen | thats the rule |
| 17:10.40 | idespinner | maybe i'm just hesitant because I feel like the general format of extensions.conf is like a half programming language... |
| 17:10.59 | leifmadsen | its not a programming language, its a scripting language |
| 17:11.16 | leifmadsen | regardless, it's how it is, and has been working and evolving for years |
| 17:12.32 | dinesh___ | hmmm ${DIALSTATUS} returns CONGESTION in case of a unanswered call here |
| 17:12.45 | dinesh___ | perhaps I need to force a timeout on Dial() shorter than the one of my provider |
| 17:12.49 | *** join/#asterisk dkirker-openmobl (~dkirker@openmobl/ceo/dkirker) |
| 17:12.58 | dinesh___ | so that asterisk will generate the unanswered |
| 17:13.23 | ManxPower-work | look at HANGUPCAUSE instead of DIALSTATUS |
| 17:13.30 | angryuser | can someone confirm me that Cause codes (any) for ISDN lines are sent by the provider equipment ? |
| 17:14.14 | ManxPower-work | angryuser, http://www.quintum.com/support/xplatform/network/Q931_Disconnect_Cause_Code_List.pdf |
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| 17:14.41 | angryuser | ManxPower-work, yes i know the list, but it is sent by provider right ? |
| 17:14.50 | ManxPower-work | angryuser, your sense makes no question |
| 17:15.19 | ManxPower-work | Those are the standard Q.931 cause codes. |
| 17:16.27 | carrar | Tatoo them on your ARM! |
| 17:16.38 | leifmadsen | what a neat idea for a tatoo! |
| 17:16.41 | leifmadsen | tattoo* |
| 17:16.50 | angryuser | ManxPower-work, i have one client with really simple problem, but his provider do not admit it, sometimes got cause code 21 (rejected) |
| 17:16.57 | *** join/#asterisk benngard (~benngard@90-230-92-67-no148.tbcn.telia.com) |
| 17:17.08 | angryuser | like 3 times a day for 2-3 min |
| 17:17.49 | angryuser | Some tech came, checked lines, found no logs about call i indicated |
| 17:18.45 | angryuser | Rejected call shoil be logged 'as he says' |
| 17:20.20 | carrar | capture a trace |
| 17:20.23 | carrar | send it to the provider |
| 17:22.36 | ManxPower-work | angryuser, that is VERY common. |
| 17:22.41 | angryuser | carrar, done it, thats why tech came, nothing found, next ? |
| 17:22.46 | ManxPower-work | I always write at least one retry into my dialing scripts. |
| 17:23.08 | carrar | esculate |
| 17:23.17 | *** join/#asterisk coppice (~chatzilla@94.201.17.210.dyn.pacific.net.hk) |
| 17:23.26 | angryuser | ManxPower-work, not common for me, i did around 50+ installs, and have first time this issue. |
| 17:23.45 | ManxPower-work | My internal scripts pretty much retry any call that did not end normally |
| 17:24.01 | ManxPower-work | angryuser, all PRI installs? |
| 17:24.17 | angryuser | ManxPower-work, pri / bri, this one is bri |
| 17:25.15 | ManxPower-work | As I said, it happens on most all PRI carriers I use. Many times it's the far end that is doing something weird. Easy enough to have your dialing script retry the call. |
| 17:25.21 | angryuser | i have less problems with pri, bri has more issues |
| 17:25.56 | ManxPower-work | Out of about 1,000 calls per day we get 2 - 6 calls that fail strangely |
| 17:26.22 | angryuser | acceptable rate |
| 17:26.26 | *** join/#asterisk RobH (~robh@216.38.133.254) |
| 17:26.32 | dinesh___ | do you manage an office voip installation? |
| 17:26.41 | ManxPower-work | angryuser, if you look at your logs I suspect your other systems also have this issue, but nobody reported it. |
| 17:26.42 | dinesh___ | to have 1000 calls/day |
| 17:26.53 | ManxPower-work | dinesh___, I work for a carrier. |
| 17:27.06 | wcselby | i have a client that probably has close to 1000 calls / day |
| 17:27.10 | ManxPower-work | most of those calls are from our cold calls sales people |
| 17:27.26 | wcselby | when you combine inbound / outbound calls |
| 17:27.35 | wcselby | and faxes, etc |
| 17:27.44 | angryuser | this one is weird, its not the same, its like we got "rejected" for 2 min 3 times a day whatever installation, the immediate retry will fail i think |
| 17:28.01 | wcselby | 150-ish person financial company, several departmental call centers, etc |
| 17:28.44 | ManxPower-work | angryuser, I've been using Asterisk with PRIs since 2002 |
| 17:28.47 | wcselby | angryuser - play a message to the caller on the reject code received that says "all circuits are busy now, please try again in a few minutes" |
| 17:29.02 | wcselby | or something along those lines |
| 17:29.13 | ManxPower-work | wcselby, that's what I do if the 2nd try fails |
| 17:29.19 | wcselby | or just say "your carrier sucks, please try again in a few mintues" |
| 17:29.24 | wcselby | or something nicer |
| 17:29.25 | wcselby | :) |
| 17:30.33 | *** join/#asterisk iq (~iq@unaffiliated/iq) |
| 17:30.36 | iq | Hi |
| 17:30.54 | angryuser | wcselby, i am doing that now |
| 17:31.41 | ManxPower-work | (most of my calls that fail with odd cause code work on the 2nd try. I do a .5 second wait between dials. |
| 17:32.30 | *** join/#asterisk Cyberax (~kvirc@81.27.245.2) |
| 17:32.58 | Cyberax | hi all |
| 17:33.00 | angryuser | ManxPower-work, i think i will press the provider again, 5 6 calls a day is fine, not 6 min of T0 down / day its a small client, but whatever |
| 17:33.36 | Cyberax | please help me connect Linksys SPA400 to asterisk 1.6 |
| 17:34.20 | *** join/#asterisk jpeeler (~jpeeler@asterisk/digium-software-dev/jpeeler) |
| 17:35.38 | Cyberax | i get message: |
| 17:35.43 | Cyberax | [Mar 4 20:35:17] WARNING[7951]: chan_sip.c:12673 check_auth: username mismatch, have <Port1>, digest has <> |
| 17:35.43 | Cyberax | [Mar 4 20:35:17] NOTICE[7951]: chan_sip.c:19961 handle_request_invite: Failed to authenticate device 0079157490007 - Port1<sip:0079157490007@192.168.100.152>;tag=9864a8c0-13c4-4b8fef55-dd0d46-3b5e7e96 |
| 17:36.20 | Cyberax | i cannot resolve this problem |
| 17:36.32 | angryuser | Cyberax, set fromusername Port1 on SPA400 |
| 17:36.54 | Cyberax | how do it? |
| 17:36.59 | angryuser | its like 'use usernam' = yes |
| 17:37.06 | angryuser | search |
| 17:37.27 | angryuser | and set the fiel username = Port1 |
| 17:37.31 | angryuser | field* |
| 17:38.13 | Cyberax | i have field Port ID 1 = Port1 |
| 17:38.32 | Cyberax | i cannot find username field |
| 17:39.45 | *** join/#asterisk sgimeno (~chatzilla@81.37.152.168) |
| 17:41.24 | *** join/#asterisk sgimeno (~chatzilla@81.37.152.168) |
| 17:41.36 | *** join/#asterisk outtolunc (~me@c-98-248-96-110.hsd1.ca.comcast.net) |
| 17:42.42 | Cyberax | anyone have spa400 + asterisk |
| 17:42.50 | *** join/#asterisk sgimeno (~chatzilla@81.37.152.168) |
| 17:42.51 | Cyberax | please respond |
| 17:43.25 | wcselby | Cyberax - i don't |
| 17:43.33 | wcselby | i have a pap2t-na, and some spa8000's |
| 17:43.37 | wcselby | nothing in the spa400 line |
| 17:44.40 | paulc | Cyberax can you send a screenshot of your admin/advanced page for the port - then we can tell you exactly which fields to fill in |
| 17:45.04 | Cyberax | how to insert screnn in irc? |
| 17:45.11 | *** join/#asterisk diatonic (~diatonic_@mail.clearwater-research.com) |
| 17:45.34 | ManxPower-work | you don't |
| 17:45.35 | ManxPower-work | ~pb |
| 17:45.36 | infobot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , or apt-get install pastebinit |
| 17:45.43 | paulc | Cyberax: http://imagebin.org/ |
| 17:46.20 | *** join/#asterisk elred_ (~elred_@unaffiliated/elred-/x-5010831) |
| 17:46.29 | Cyberax | ok i do it |
| 17:48.46 | Cyberax | http://imagebin.org/87515 |
| 17:48.48 | Cyberax | here |
| 17:49.50 | paulc | Where do you configure the password for those users though? |
| 17:50.26 | Cyberax | here not user and password only FXO pstn line |
| 17:50.31 | *** join/#asterisk rickross (~rickross@supporter/active/rickross) |
| 17:51.40 | *** join/#asterisk Ad-Hoc (~nimbus@62.1.141.83.dsl.dyn.forthnet.gr) |
| 17:51.51 | *** join/#asterisk waa (~waa@balrog.credipar.com.br) |
| 17:52.19 | Cyberax | other pages for voice mail and line config |
| 17:52.31 | rickross | we're trying to get * working with a "click to call" interface from a CRM called vTiger. We believe the manager.conf is set to allow an account "vtiger" to access, but it repeatedly fails to authenticate when vtiger tries to initiate a call. The message is like: [Mar 4 12:47:57] NOTICE[7732] manager.c: 208.91.135.21 failed to authenticate as 'vtiger' |
| 17:52.46 | rickross | anyone have any experience getting vtiger to work with * ? |
| 17:52.49 | Ad-Hoc | hi |
| 17:53.09 | idespinner | rickross, the authentication will be in /etc/asterisk/manager.conf |
| 17:53.21 | idespinner | make sure the vtiger user is there |
| 17:53.58 | idespinner | and also make sure you have reloaded asterisk |
| 17:53.59 | rickross | idespinner: we believe we have that configured for the account 'vtiger' - double-checking |
| 17:54.43 | idespinner | from the asterisk CLI you can do "manager show user vtiger" |
| 17:55.03 | *** join/#asterisk Firass-z0r (~asadf@c-67-201-205-34.reshall.wwu.edu) |
| 17:55.54 | rickross | ok, it is in a manager_additional.conf that is included by manager.conf |
| 17:56.04 | rickross | checking from the * CLI |
| 17:56.18 | ManxPower-work | rickross, welcome to FreePBX |
| 17:56.32 | ManxPower-work | nothing will be where you expect it to be |
| 17:56.33 | Cyberax | if i delete register then i'm get all calls in one extension |
| 17:56.42 | rickross | rs1*CLI> manager show user vtiger |
| 17:56.42 | rickross | rs1*CLI> |
| 17:56.42 | rickross | <PROTECTED> |
| 17:56.47 | rickross | it is there |
| 17:57.00 | idespinner | rickross, is "secret: <Set>"? |
| 17:57.07 | pyite | anyone know of a good PoE switch (16 ports or 8 ports) that actually has PoE on ALL ports? |
| 17:57.11 | rickross | idespinner: yes, exactly |
| 17:57.20 | *** join/#asterisk ariel_ (~chatzilla@63.214.236.169) |
| 17:57.37 | idespinner | pyite, 3com office connect is good, full gig, layer 3 and poe for around $300 |
| 17:57.47 | pyite | idespinner: but is PoE on ALL ports? that's been the tough part |
| 17:57.49 | idespinner | rickross, I would double check the password... |
| 17:57.52 | *** join/#asterisk RobH (~robh@216.38.133.254) |
| 17:57.54 | idespinner | pyite, yes |
| 17:57.58 | rickross | idespinner, is it possible to get asterisk to display what it thinks the secret is set to? |
| 17:58.03 | pyite | really?!? woop. lemme go look again |
| 17:58.12 | ManxPower-work | the password is sent in cleartext, so you should be able to tcpdump it. |
| 17:58.29 | idespinner | yes, rickross do tcpdump -A |
| 17:58.34 | nickfennell | Is it possible to make asterisk supply whitenoise when a call is placed on hold rather than pausing the RTP stream ? |
| 17:58.48 | nickfennell | or even just to feed silence to the UA |
| 17:58.50 | idespinner | and look for login: vtiger secret:MYPASS |
| 17:58.53 | rickross | ManxPower -work: good idea - I need to get more comfortable with those tcpdump ops |
| 17:58.59 | idespinner | and compare to manager.conf entry |
| 17:59.13 | pyite | idespinner: thanks! that looks perfect actually |
| 17:59.30 | idespinner | tcpdump -A port 5038 |
| 17:59.42 | ManxPower-work | rickross, from memory: tcpdump -X -s 4096 -i INTERFACE port WHATEVERTHEMANAGERPORTIS |
| 17:59.43 | rickross | idespinner: trying now |
| 17:59.54 | idespinner | ManxPower-work, that works too! |
| 18:00.01 | rickross | ManxPower-work: thx |
| 18:00.34 | *** join/#asterisk RobH_ (~robh@216.38.133.254) |
| 18:01.17 | Corydon76-dig | ManxPower-work: 5038 |
| 18:02.44 | idespinner | nickfennell, i thought thats what Music on hold was |
| 18:02.52 | idespinner | just play back some whitenoise MOH |
| 18:03.50 | nickfennell | hmm that's what I'm thinking |
| 18:03.56 | ManxPower-work | asterisk comes with several silence files" |
| 18:04.08 | nickfennell | I didn't know if there was a better way of doing it than that |
| 18:04.12 | nickfennell | a little "less hacky" |
| 18:06.28 | idespinner | MOH should be built in as a standard feature |
| 18:07.55 | rickross | well, I don't think I know how to read this - http://pastebin.com/01d45jh9 |
| 18:08.17 | rickross | I may need to write the tcpdump output to a file or something |
| 18:09.07 | *** join/#asterisk grEvenX (~even@cC0FD00C3.dhcp.bluecom.no) |
| 18:09.32 | idespinner | rickross, no login attempt there that I saw |
| 18:09.35 | idespinner | should be in plain text |
| 18:09.48 | rickross | line 15 |
| 18:10.27 | rickross | strangely, it seems to have cut off the username to "vtige" |
| 18:12.27 | idespinner | ah yea, i see it now |
| 18:12.31 | idespinner | shouldve searched |
| 18:12.41 | idespinner | try ManxPower's line |
| 18:12.51 | idespinner | tcpdum -X -s ... |
| 18:13.01 | rickross | ok, trying again - one sec |
| 18:15.07 | rickross | http://pastebin.com/JSRSZYkW |
| 18:15.37 | rickross | lines 25-26 show the secret |
| 18:16.00 | rickross | which, duh, we'll now change after this is debugged :) |
| 18:17.39 | rickross | secret is correct - I wonder if ther's a line endings issue with the 0x0A 0x0D from the script issuer? |
| 18:17.46 | *** join/#asterisk atis_work (~atis_work@193.238.212.171) |
| 18:18.15 | idespinner | rickross, you can doublecheck the secret yourself via telnet |
| 18:18.16 | Nugget | telnet is eeeeeeevil! |
| 18:18.34 | idespinner | telnet localhost 5038 |
| 18:18.35 | rickross | idespinner: we're just about to try that |
| 18:20.38 | rickross | http://pastebin.com/37R2VN0i |
| 18:20.42 | rickross | failed - ugh |
| 18:20.46 | raden_work | how can i tell what codecs a channel using |
| 18:21.29 | [TK]D-Fender | raden_work: sip show channel [thechannel] |
| 18:22.16 | raden_work | [TK]D-Fender, thanks |
| 18:22.44 | idespinner | you may want to pastebin your manager.conf... |
| 18:23.15 | raden_work | [Mar 4 12:22:59] WARNING[5050]: chan_sip.c:4339 sip_call: No audio format found to offer |
| 18:24.47 | *** join/#asterisk Tech_Travis (~tech_trav@208.179.137.131) |
| 18:25.05 | Skeeter- | anyone knows how to parse a php file into a html file> |
| 18:25.07 | rickross | ok, the same sequence for another account allowed authentication |
| 18:26.20 | [TK]D-Fender | Skeeter-: Who says PHP inherently has anything to do with HTML? |
| 18:26.37 | dinesh___ | beginners |
| 18:26.43 | Skeeter- | [TK]D-Fender, i have a php output that i wanna parse into html |
| 18:27.08 | raden_work | to use g729 does it have to be allow in general as well as my outbound context ? |
| 18:27.12 | *** join/#asterisk Katty (~asteriska@mail.copi-rite.com) |
| 18:27.18 | Katty | OHAIDERMEILOVELIES |
| 18:27.26 | [TK]D-Fender | Skeeter-: whats to parse? All PHP does is process the stuff in the middle that is a script. Anything outside is just left as dumb text |
| 18:27.33 | [TK]D-Fender | Skeeter-: there is no "parse". |
| 18:27.53 | [TK]D-Fender | raden_work: it has to be allowed by the time your acll hits whatever section it hits |
| 18:28.21 | dinesh___ | would it be difficult to implement a version of Dial() that would work without implicitly answering the call first ? |
| 18:28.37 | dinesh___ | (to call for free, even if it's limited to just 30 seconds, I'd like to see if it's possible) |
| 18:28.44 | [TK]D-Fender | dinesh___: Dial DOESN'T implicitly answer the call first |
| 18:28.44 | dinesh___ | or perhaps it's even already available |
| 18:28.45 | Qwell | dinesh___: Dial doesn't implicitly answer |
| 18:28.52 | [TK]D-Fender | NEXT!@!@!@ |
| 18:28.58 | [TK]D-Fender | (c) BKW |
| 18:29.20 | rickross | idespinner: thanks, man - we had an error in manager.conf - DOH! |
| 18:29.24 | dinesh___ | oh okay, so when the other end answers, that answer gets forwarded |
| 18:29.36 | dinesh___ | so I would need to drop that answer packet |
| 18:30.12 | *** join/#asterisk jelly-bean (~jelly-bea@75-148-103-190-Utah.hfc.comcastbusiness.net) |
| 18:30.19 | [TK]D-Fender | dinesh___: huh? |
| 18:30.23 | jelly-bean | where are asterisk call IDs usually stored? |
| 18:30.30 | jelly-bean | in the database? log files? |
| 18:30.38 | idespinner | rickross, glad to hear, your problem sounded easy enough! |
| 18:30.52 | Skeeter- | [TK]D-Fender, i have a nice php webpage that iw ould like to view using my polycom that can see html |
| 18:31.01 | idespinner | jelly-bean, do you mean the master.csv which has all the cdr hsitory of all calls? |
| 18:31.09 | jelly-bean | idespinner: yes |
| 18:31.11 | jelly-bean | perhaps |
| 18:31.13 | dinesh___ | well I have an incoming extension 1,1,Dial(SIP/home) ; it's possible to have 1,1,Playback(sound, noanswer) that will play "sound" without answering first |
| 18:31.20 | [TK]D-Fender | Skeeter-: ... html is test that gets passed through.... |
| 18:31.26 | dinesh___ | so i'm thinking that it should be possible to get voice as well, without answering |
| 18:31.32 | Katty | Qwell: how is your foot |
| 18:31.32 | jelly-bean | i have an asterisk call_id and i want to use it to find the number the person dialed |
| 18:31.33 | [TK]D-Fender | Skeeter-: and Polycom's don't use HTML |
| 18:31.38 | Qwell | Katty: not broked |
| 18:31.44 | Katty | Qwell: yes you told me that |
| 18:31.46 | Katty | Qwell: but how is it |
| 18:31.47 | Naikrovek | xhtml |
| 18:31.52 | Qwell | Katty: not purple and big |
| 18:31.53 | [TK]D-Fender | ^^ |
| 18:31.56 | Katty | Qwell: :> |
| 18:32.04 | Katty | Qwell: so improvement then, yes? :> |
| 18:32.06 | Qwell | and I gave my drugs away |
| 18:32.07 | Qwell | yes |
| 18:32.09 | *** join/#asterisk arpu (~arpu@chello062178159144.10.14.univie.teleweb.at) |
| 18:32.09 | Skeeter- | [TK]D-Fender, what are they using? |
| 18:32.12 | Katty | YAY!!!!!!!!!!!!!! |
| 18:32.15 | Katty | dances with Qwell |
| 18:32.25 | *** part/#asterisk morex (~m@5ac4bcaa.bb.sky.com) |
| 18:32.38 | Katty | infobot: seen seanmh |
| 18:32.42 | infobot | seanmh <n=johndoe@207.114.199.107> was last seen on IRC in channel #asterisk, 142d 23h 11m 1s ago, saying: 'Katty: how's the 1.6 testing going?'. |
| 18:32.51 | Katty | hrmm. |
| 18:34.42 | [TK]D-Fender | Skeeter-: VERY FINE MANUALS |
| 18:35.03 | Katty | calls symantec again. /sigh |
| 18:37.03 | [TK]D-Fender | Katty: what app? |
| 18:40.18 | raden_work | is there a way if I exceed my g729 license count It will use ulaw ? |
| 18:42.26 | [TK]D-Fender | raden_work: No |
| 18:42.56 | *** join/#asterisk pacmanfan (~pacmanfan@d4-44.rb5.clm.centurytel.net) |
| 18:42.58 | raden_work | that so not cool |
| 18:43.00 | *** part/#asterisk jelly-bean (~jelly-bea@75-148-103-190-Utah.hfc.comcastbusiness.net) |
| 18:43.39 | [TK]D-Fender | raden_work: Lift your skirt, grab your balls, and MAN UP. |
| 18:44.20 | Katty | [TK]D-Fender: endpoint protection |
| 18:45.15 | raden_work | [TK]D-Fender, illl just use 729 on outbound calls then |
| 18:45.16 | Skeeter- | raden_work, u can allow=g729,ulaw i think |
| 18:45.40 | raden_work | Skeeter-, I was told that will always default to ulaw then |
| 18:46.00 | Skeeter- | 2 seperate lines there |
| 18:46.04 | Skeeter- | allow=g729 |
| 18:46.04 | [TK]D-Fender | Katty: SEP : Shouldn't Expect Productivity |
| 18:46.09 | Skeeter- | allow=ulaw |
| 18:46.13 | [TK]D-Fender | Skeeter-: complete waste <- |
| 18:46.18 | Katty | :P |
| 18:46.21 | Skeeter- | [TK]D-Fender, aight ur call |
| 18:46.29 | raden_work | Skeeter-, that is what I thought, but was told it will default to ulaw everytime |
| 18:46.31 | [TK]D-Fender | Skeeter-: it will ALWAYS negotiate the same order and always produce the same reult to an ITSP |
| 18:46.39 | [TK]D-Fender | Skeeter-: because THEIR offer will never change. |
| 18:46.54 | [TK]D-Fender | Skeeter-: So the net negotiation will always yeild the same outcome |
| 18:47.06 | [TK]D-Fender | Skeeter-: Specifying multiple codecs = pointless in that case |
| 18:47.14 | dinesh___ | ManxPower-work: RetryDial() looks cool :) |
| 18:47.44 | raden_work | [TK]D-Fender, will it work for outbound if i run out ? specifying a order that is ? |
| 18:47.54 | [TK]D-Fender | [13:31]<dinesh___>so i'm thinking that it should be possible to get voice as well, without answering <- no |
| 18:48.17 | Katty | on a positive note, i'm starting to understand their accent a bit better |
| 18:48.18 | dddh | do people use tls connections for sip? |
| 18:48.19 | dinesh___ | hm :/ but it's possible to get music though :/ |
| 18:48.27 | dinesh___ | at least 1 way |
| 18:48.43 | *** join/#asterisk korihor (~korihor@201.210.226.98) |
| 18:49.17 | dinesh___ | so the provider of the caller would block any voice from the caller to the callee until the callee answered? |
| 18:49.24 | dinesh___ | that would be a good strategy |
| 18:50.13 | *** join/#asterisk DMeloUK (~DominicMe@64.129.95.226) |
| 18:50.16 | [TK]D-Fender | dinesh___: You're talking about early media. That is send-only from the answerer side. |
| 18:51.52 | *** join/#asterisk mikkel (~mikkel@84-238-113-66.u.parknet.dk) |
| 18:52.55 | dinesh___ | ok thanks |
| 18:54.30 | *** join/#asterisk sulex (~sulex@host-78-14-173-189.cust-adsl.tiscali.it) |
| 18:55.17 | *** join/#asterisk aandrade (~aandrade@201.47.13.214.dynamic.adsl.gvt.net.br) |
| 18:56.28 | *** join/#asterisk hluesea (~hulusikah@88.247.127.66) |
| 18:56.59 | Katty | so what's you guys' favorite app for mounting a virtual cdrom for iso |
| 18:57.05 | Qwell | Katty: mount. |
| 18:57.17 | Katty | for windows. |
| 18:57.22 | Qwell | Debian install CD. |
| 18:57.37 | dinesh___ | alcohol 52% is great |
| 18:57.56 | Katty | Qwell: :P |
| 18:57.59 | Katty | dinesh___: shanks. |
| 18:58.31 | [TK]D-Fender | Katty: Daemon toold |
| 18:58.33 | [TK]D-Fender | tools* |
| 19:04.05 | carrar | windows!@!$!#@ |
| 19:04.27 | carrar | Didn't MSFT go out of biz already? |
| 19:06.36 | idespinner | Katty, use wincdemu |
| 19:06.56 | *** join/#asterisk xuser (~xuser@unaffiliated/xuser) |
| 19:07.14 | idespinner | alchohol 52 is more 'free' version of paid product type thing... |
| 19:07.24 | idespinner | dameon tools is 'adware' |
| 19:07.39 | idespinner | wincdemu <- opensource |
| 19:08.08 | Katty | wow. my symantec rep isn't from india :> |
| 19:08.32 | Qwell | Katty: just because his name is "John", doesn't mean he isn't |
| 19:10.56 | *** join/#asterisk lesouvage (~lesouvage@82.73.69.76) |
| 19:16.44 | Katty | Qwell: he's from texas. |
| 19:17.40 | wcselby | Katty - that makes him awesome, by default |
| 19:17.44 | wcselby | just sayin;.... |
| 19:17.46 | Katty | :> |
| 19:19.22 | lesouvage | What 4 ports isdn bri card do you advice? |
| 19:19.30 | [TK]D-Fender | Katty: What the Texas, North Punjab? |
| 19:19.47 | Kobaz | lesouvage: sangoma |
| 19:20.33 | Katty | [TK]D-Fender: SHUUSH! |
| 19:20.38 | Qwell | lesouvage: Digium b410p. I'm not biased at all. |
| 19:21.39 | [TK]D-Fender | lesouvage: Indeed... Digium blocks his ability to even consciously acknowledge products by other makers ;) |
| 19:22.12 | [TK]D-Fender | sends Qwell the "kill" command |
| 19:22.27 | Qwell | other what now? |
| 19:22.43 | [TK]D-Fender | EXACTLY |
| 19:25.33 | lesouvage | [TK]D-Fender: I expect that the card works and that it is easy to configure. Is Digium the proper choice or should I go for Sangoma or Jungmans. I prefer Digium because they start the Asterisk project years ago. |
| 19:25.47 | Kobaz | any card will work |
| 19:26.00 | Kobaz | well, any card from one of the reputable names will work |
| 19:26.25 | Kobaz | lesouvage: personal preference really... i like the sangoma cards because they give you a lot of tools for debugging problems |
| 19:27.39 | Kobaz | and what's a jungman? i never heard of them |
| 19:27.49 | lesouvage | Kobaz: but if you pick a card without problems you don't need the debug tools. I want to avoid debugging, it is hard to budget and it is boring and stressfull. |
| 19:28.43 | Kobaz | lesouvage: no... the cards don't have any problems... it's for debugging your environment (ie: cabling, voltages, alarms, isdn counters) |
| 19:28.56 | Kobaz | lesouvage: you always need debugging tools |
| 19:29.55 | outtolunc | i think someone should have updated the digium store with the new lunenvox lic pricing, before the email blast |
| 19:32.32 | lesouvage | Kobaz: it is Junghanns see http://www.junghanns.net/en/home.html |
| 19:33.54 | Kobaz | lesouvage: no idea... the three major vendors are sangoma, digium, and rhino |
| 19:34.01 | Kobaz | but don't use rhino, they suck |
| 19:36.37 | *** join/#asterisk miamiseb (~a@208.76.35.132) |
| 19:37.54 | idespinner | whats wrong with rhino? |
| 19:37.55 | *** join/#asterisk DaveCanoe (~Dave@2001:1928:0:ffff::2) |
| 19:38.07 | idespinner | (no experience with anything but digium cards...) |
| 19:38.09 | Kobaz | idespinner: i've had some serious problems with their dsps and firmware |
| 19:38.28 | Kobaz | they are more of a hack slash shop (from the last i dealt with them) |
| 19:38.33 | Kobaz | they may have improved |
| 19:38.44 | Kobaz | i was using rhino t1 cards for a while since they were so cheap |
| 19:38.51 | idespinner | thats too bad, they always touted "american made" |
| 19:39.04 | Kobaz | and then i got my first dual span from them and had nothing but problems... d channel would randomly go away, calls drop, audio problems |
| 19:39.17 | Kobaz | 4 and a half months of "here try this firmware" |
| 19:40.55 | ManxPower-work | I prefer Sangoma |
| 19:41.07 | Kobaz | i popped in my first sangoma card and it worked like a charm |
| 19:41.23 | Kobaz | and... every time i thought a sangoma card was bad... it wasn't |
| 19:41.36 | Kobaz | it was wither a bad cable, or irq conflict, or something stupid |
| 19:41.58 | Kobaz | we actually fried several digium cards doing a standard installation |
| 19:42.06 | Kobaz | this was like 2 years ago though |
| 19:46.24 | *** join/#asterisk chilicuil (~sistemas@unaffiliated/chilicuil) |
| 20:02.57 | raden_work | is there a way to allow g729 passthrough without it taking licensed channels ? |
| 20:03.13 | raden_work | my phones are g729 ready when i dial out I dont see why id need a license |
| 20:04.11 | [TK]D-Fender | raden_work: You don't. You only need it any time * has to transcode anything.. |
| 20:04.31 | raden_work | hmmm |
| 20:04.40 | leifmadsen | or write to disk |
| 20:04.43 | leifmadsen | (i.e. voicemail) |
| 20:04.55 | *** join/#asterisk fifer (~fifer@67.208.108.228) |
| 20:05.02 | [TK]D-Fender | leifmadsen: That shouldn't inherently need to transcode... |
| 20:05.25 | leifmadsen | [TK]D-Fender: but writing to disk requires a license because you're not just blindly passing information across the wire |
| 20:05.33 | leifmadsen | you're not transcoding though |
| 20:05.43 | [TK]D-Fender | leifmadsen: You should if you set your recording codecs accordingly, no? |
| 20:05.52 | leifmadsen | no idea for sure |
| 20:05.56 | [TK]D-Fender | leifmadsen: Figured it was +/- a direct frame dump |
| 20:05.59 | leifmadsen | just pretty sure you need a license to write to disk |
| 20:06.06 | fifer | I'm on *1.4 working to get an openvox a1200p up and most of the install seemed to go fine though I can not verify that zaptel is working |
| 20:06.07 | leifmadsen | could be, I'm going to shutup now |
| 20:06.27 | fifer | the card was detected but the module does not seem to be working, nor is dahdi now. |
| 20:06.48 | fifer | I'm new to the whole dahdi thing so I may have messed something up |
| 20:07.24 | [TK]D-Fender | leifmadsen: Neither of us is 100% on it :) |
| 20:07.32 | [TK]D-Fender | leifmadsen: I am high 90's though ;) |
| 20:09.50 | tzafrir_laptop | fifer, what's the output of: cat /proc/zaptel/* |
| 20:10.10 | tzafrir_laptop | What's the output of: asterisk -rx 'zap show channels' |
| 20:10.12 | tzafrir_laptop | ? |
| 20:10.19 | leifmadsen | [TK]D-Fender: you should try it out and let me know |
| 20:11.41 | *** join/#asterisk italorossi (~italoross@201.76.154.127.intranet.digi.com.br) |
| 20:11.43 | fifer | shows all 12 chanels, shows the 4 fxo ports, lists 1,3,4 as red |
| 20:11.59 | tzafrir_laptop | 'zap show channels'? |
| 20:13.34 | idespinner | fifer, were you installing DAHDI or ZAPTEL? |
| 20:13.34 | fifer | nothing, just the headings |
| 20:13.59 | fifer | I did a fresh ZAPTEL install |
| 20:14.22 | fifer | My understanding that the card is still very new to DAHDI and maybe not ready for prod there |
| 20:14.45 | idespinner | so we can safely ignore "<fifer> I'm new to the whole dahdi thing so I may have messed something up" right? just double checking... |
| 20:15.02 | [TK]D-Fender | fifer: DAHDI = Zaptel. . |
| 20:15.05 | raden_work | <PROTECTED> |
| 20:15.05 | raden_work | <PROTECTED> |
| 20:15.11 | raden_work | but i can dial from my cellphone just fine |
| 20:15.15 | tzafrir_laptop | Or rather: dahdi is the latest version of Zaptel |
| 20:15.23 | fifer | But now the DAHDI app does not work in cli |
| 20:15.40 | tzafrir_laptop | fifer, and anyway, your /etc/asterisk/zapata.conf is misconfigured |
| 20:15.46 | [TK]D-Fender | fifer: You haven't shown us anything ov value in debugging this |
| 20:15.55 | fifer | dahdi show status "no DAHDI interface found" |
| 20:16.05 | idespinner | umm |
| 20:16.30 | idespinner | well i dont think dahdi==zaptel, you either have one or the other, but not both |
| 20:16.35 | [TK]D-Fender | fifer: What ver of *? Did you compile * ASFTER installing DAHDI? |
| 20:16.45 | idespinner | and depending on which asterisk version, you have to have one or the other |
| 20:16.47 | bmoraca_work | you really screwed the pooch, then, cause even with just dahdi dummy, i get something, heh |
| 20:16.52 | sbrath | I have a Queue that people can't seem to transfer calls out of, I have Asterisk 1.6.2 and Queue(c-service,twhr,,,30) , but when they do a transfer it never completes. |
| 20:16.58 | *** join/#asterisk atis_work (~atis_work@193.238.212.171) |
| 20:16.59 | idespinner | some * versions only can do zap, some can only do dahdi |
| 20:17.10 | Kobaz | sbrath: pastebin your logs and console output |
| 20:17.16 | fifer | 1.4.29.1 |
| 20:18.21 | fifer | I installed zaptel-1.4.11.1 yesterday |
| 20:18.29 | fifer | I installed zaptel-1.4.12.1 yesterday |
| 20:18.31 | fifer | actually |
| 20:18.52 | idespinner | yea, thats probably a nogo... i think the latest 1.4.29.x required dahdi... |
| 20:19.35 | fifer | Hm.....a recent post from openvox indicated that the card was working with dahdi but not to use it for production yet. |
| 20:19.38 | ManxPower-work | idespinner, that is INCORRECT! |
| 20:20.03 | fifer | http://bbs.openvox.cn/viewthread.php?tid=587&extra=page%3D1 |
| 20:20.05 | idespinner | ManxPower-work, clarificatoin is always welcome :) |
| 20:20.35 | ManxPower-work | At some point in the 1.4 release cycle chan_zap.so was renamed chan_dahdi.so, however, if you have zaptel installed then te CLI zap commands are available, if DAHDI is available then the CLI dahdi commands are available. |
| 20:20.54 | ManxPower-work | the only time you would notice the chan_dahdi.so is if you looked for it. |
| 20:21.51 | idespinner | ManxPower-work, can you clarify "if you looked for it."? |
| 20:22.05 | ManxPower-work | idespinner, ls /var/lib/asterisk/modules or "core show modules" |
| 20:22.09 | fifer | in cli, zap show chanels gives me the result headings, no errror, but "help zap" says no such command |
| 20:23.15 | raden_work | there a way i can make a phone use ulaw when checking VM and stuff where asterisk would need to encode, and use g729 only when dialing outbound so it passes through ? |
| 20:23.18 | ManxPower-work | fifer, What specific problem are you trying to solve? |
| 20:23.29 | ManxPower-work | raden_work, no. |
| 20:23.37 | fifer | just trying to get this new a1200p installed and working |
| 20:23.40 | ManxPower-work | raden_work, As I said, G729 pass-thru is pretty useless for most things. |
| 20:23.51 | fifer | has 4 fxo modules and two lines connected |
| 20:24.05 | ManxPower-work | fifer, you'll get better help with better questions. |
| 20:24.20 | fifer | zaptel seems happy but not working with *, may have messed up zap/dahdi in my *, not sure |
| 20:24.34 | ManxPower-work | does "zap show channels" show all your configured channels? |
| 20:24.45 | raden_work | ManxPower-work, so if i use g729 i always have to use it ? |
| 20:24.58 | ManxPower-work | raden_work, if you want pass-thru you do. |
| 20:24.59 | fifer | zap show channels just shows the colum headers, no channels |
| 20:25.12 | raden_work | ManxPower-work, well i guess that works |
| 20:25.13 | ManxPower-work | fifer, then you have no channels configured |
| 20:25.36 | ManxPower-work | raden_work, otherwise stop being a cheapskate and spend the money on a couple of G729 licenses. |
| 20:26.01 | *** join/#asterisk iCEBrkr (~icebrkr@72.251.206.106) |
| 20:26.03 | idespinner | raden_work, you may be able to |
| 20:26.08 | fifer | during the zaptel installation I was able to generate both config files |
| 20:26.12 | ManxPower-work | fifer, make sure you don't have any chan_dahdi*.conf files in /etc/asterisk. pastebin your /etc/asterisk/zapata.conf |
| 20:26.13 | idespinner | raden_work, set allow=ulaw,g729 |
| 20:26.16 | *** join/#asterisk chazzm (~chazz@173-24-238-25.client.mchsi.com) |
| 20:26.17 | idespinner | on your phone |
| 20:26.27 | ManxPower-work | fifer, the generated configs files != installed config files. |
| 20:26.28 | fifer | zaptel.conf |
| 20:26.29 | *** join/#asterisk iCEBrkr (~icebrkr@72.251.206.106) |
| 20:26.35 | idespinner | and set allow=g729, disallow all on your trunk |
| 20:26.43 | ManxPower-work | fifer, then pastebin your zapata.conf |
| 20:26.54 | fifer | and zapta-channels.conf |
| 20:26.56 | ManxPower-work | idespinner, that will always use ulaw then |
| 20:27.03 | idespinner | not out the trunk |
| 20:27.08 | ManxPower-work | fifer, zapta-channels.conf IS NOT zapata.conf! |
| 20:27.19 | idespinner | the trunk is allow g729a only |
| 20:27.45 | ManxPower-work | idespinner, The call will still come into asterisk as ulaw, then fail when it tries to transcode to g729 with no licenses |
| 20:27.56 | fifer | http://pastebin.ca/1823069 |
| 20:28.21 | ManxPower-work | fifer, is that file named zaptel.conf |
| 20:28.30 | fifer | YES |
| 20:28.31 | idespinner | if the call isnt answered, there is no rtp stream yet so no codec negotiation has been done yet |
| 20:28.37 | raden_work | ManxPower-work, I got 4 just the way the boss is using the phone he needs about 8 |
| 20:28.51 | ManxPower-work | fifer, then do a module reload chan_dhadi.so |
| 20:28.55 | fifer | both are named properly, not sample or anything |
| 20:29.00 | idespinner | or so i think... |
| 20:29.06 | fifer | ManxPower-work: thanks! Sure, |
| 20:29.08 | ManxPower-work | "both"? There is only one zapata.conf |
| 20:29.12 | *** join/#asterisk frantik (~frantik@190.114.99-84.rev.gaoland.net) |
| 20:29.15 | frantik | hello |
| 20:29.37 | frantik | i'm not at all experienced with asterisk (i manage our linux servers but not all specifics) |
| 20:29.40 | fifer | ok, said it is reloading, no error |
| 20:29.41 | ManxPower-work | raden_work, in Asterisk 1.6 things may have changed somewhat. |
| 20:29.49 | frantik | and i'm having no clue with the beast crash |
| 20:29.50 | ManxPower-work | fifer, check your channels with "zap show channels" |
| 20:30.08 | frantik | at the boot |
| 20:30.13 | frantik | i have a loop of "asterisk died with code 1" |
| 20:30.17 | fifer | still just the headers |
| 20:30.17 | ManxPower-work | fifer, you may have to unload / load chan_dahdi.so, since you are adding channels. |
| 20:30.29 | ManxPower-work | frantik, start asterisk as "asterisk -cvvv" to see where it is failing |
| 20:30.36 | frantik | did that |
| 20:30.42 | ManxPower-work | fifer, "just the headers" means "config file not found" |
| 20:30.48 | ManxPower-work | frantik, well now you see your error message. |
| 20:31.06 | frantik | h/o |
| 20:31.12 | frantik | now thats the problem |
| 20:31.16 | frantik | no* |
| 20:31.47 | fifer | ManxPower-work: Unloaded then reloaded, now change for zap show channels |
| 20:31.48 | ManxPower-work | fifer, Actually, "just the headers" means config file found, nothing found to configure in the file. |
| 20:31.55 | ManxPower-work | fifer, now your zap is configured |
| 20:32.12 | ManxPower-work | did you really mean "now" or did you mean "no" |
| 20:32.34 | fifer | zaptel.conf is in /etc, zapata-channels.conf is in /etc/asterisk |
| 20:33.02 | fifer | my bad, no change |
| 20:33.03 | ManxPower-work | fifer, Asterisk does not use zapata-channels.conf |
| 20:33.17 | fifer | ok |
| 20:33.26 | ManxPower-work | zapata-channels.conf is the automatically generated file that yo are stupposed to rename or import into your config files. |
| 20:33.44 | fifer | ok, missed that step |
| 20:33.56 | frantik | ManxPower-work, i dont see no error message :/ |
| 20:34.07 | ManxPower-work | fifer, delete /etc/asterisk/zapata-channels.conf since you just showed me it is pretty much a duplicated of zapata.conf, right? |
| 20:34.18 | *** join/#asterisk dzup (dzup@support.team.at.shellium.org) |
| 20:34.23 | ManxPower-work | frantik, copy what is output to pastbin.ca |
| 20:34.35 | fifer | I showed you zaptel.conf, I will show the other now.... |
| 20:35.02 | frantik | k |
| 20:35.03 | ManxPower-work | fifer, You still don't get it do you. I'm sorry, I told you your answer. I cannot help you firther. |
| 20:35.09 | frantik | retrieving the output |
| 20:35.43 | *** join/#asterisk kotp (~vgoff@96.2.187.66) |
| 20:35.56 | *** join/#asterisk Akiraaa (~Akiraaaa@79.112.13.42) |
| 20:35.59 | *** join/#asterisk edoceo (~edoceo@c-98-247-254-241.hsd1.wa.comcast.net) |
| 20:36.02 | fifer | ManxPower-work: I greatly appreciate your help but what answer? What should this file be named? OR what file should an include statement be added to include it? |
| 20:36.03 | ManxPower-work | fifer, so I asked for zapata.conf and you showed me zaptel.conf??? Are you deliberatly thing to screw up people helping you? |
| 20:36.15 | Katty | HEYYYYYYYYYYYYYY MARGARITA! |
| 20:36.27 | fifer | I understood you wanted the zaptel.conf |
| 20:36.50 | tzafrir_laptop | fifer, one common way to use zapata-channels.conf is the line #include zapata-channels.conf in zapata.conf |
| 20:36.56 | ManxPower-work | <ManxPower-work> fifer, make sure you don't have any chan_dahdi*.conf files in /etc/asterisk. pastebin your /etc/asterisk/zapata.conf |
| 20:37.03 | tzafrir_laptop | that said, why not just pastebin the file? |
| 20:37.24 | fifer | ManxPower-work: relax, I have spent over 15 years on your end of irc help in the tech world, I'm no newbe to this process, just missing some details and trying to get this figured out |
| 20:37.37 | fifer | I greatly appreciate the help and I'm not trying to be dificult |
| 20:37.37 | ManxPower-work | fifer, nothing I have said since you showed me the wrong file applies., |
| 20:37.43 | fifer | Just a sec and I will show you the right file |
| 20:37.56 | ManxPower-work | show it to tzafrir_laptop |
| 20:38.10 | fifer | I showd you the file I understood you wanted |
| 20:38.13 | fifer | My bad |
| 20:38.15 | fifer | fixing now |
| 20:38.20 | fifer | http://pastebin.ca/1823081 |
| 20:38.30 | sbrath | my Queue transfer inability problem pastebin: http://pastebin.com/9HAiRZgH |
| 20:38.34 | *** join/#asterisk asteriskmonkey (~philip@69.77.169.14) |
| 20:38.38 | ManxPower-work | maybe so, but I've already run out of time to help you on this (should be simple) problem. |
| 20:38.52 | ManxPower-work | It should have taken 2 mins to solve your issue if I had seen the right config files to start with. |
| 20:39.22 | tzafrir_laptop | fifer, what about /etc/asterisk/zapata.conf ? |
| 20:39.27 | asteriskmonkey | anyone know of any incompatibilites that might cause no audio between an asterisk 1.4.x box and an asterisk 1.6.2 box? |
| 20:39.46 | [TK]D-Fender | fifer: where is your zapata.conf? |
| 20:39.56 | fifer | I dont have a zapata.conf |
| 20:40.07 | ManxPower-work | fifer, then asterisk will not load zaptel support. |
| 20:40.08 | fifer | Is that what the zapata-channels.conf needs to be renamed to? |
| 20:40.08 | tzafrir_laptop | asteriskmonkey, those two systems are connected via? SIP? IAX? |
| 20:40.14 | *** join/#asterisk jnfuller (~jnfuller@d64-180-206-233.bchsia.telus.net) |
| 20:40.15 | frantik | ManxPower-work, http://pastebin.ca/1823089 |
| 20:40.37 | *** join/#asterisk mbrevda (~mbrevda@unaffiliated/mbrevda) |
| 20:40.38 | asteriskmonkey | tzafrir_laptop: connected via sip, both public ip, both no fwalls |
| 20:40.40 | tzafrir_laptop | fifer, use configs/chan_dahdi.conf.sample |
| 20:40.48 | tzafrir_laptop | rename it to zapata.conf |
| 20:40.52 | fifer | will do, thanks!! |
| 20:41.00 | Kobaz | sbrath: redo your past without console colors |
| 20:41.00 | mbrevda | anyone know what would cause this? app_voicemail_imapstorage.c: IMAP Error: Connection failed to gmail-imap.l.google.com,993: Connection timed out |
| 20:41.02 | [TK]D-Fender | fifer: http://pastebin.ca/182308 <-- lines 30/31 = wake up and read the BIG PRINT |
| 20:41.03 | frantik | the log leaves me totally clueless and as i specified i'm an asterisk newb |
| 20:41.08 | *** join/#asterisk RobH (~robh@216.38.133.254) |
| 20:41.10 | ManxPower-work | frantik, that only looks like the first part of the file. |
| 20:41.13 | sbrath | How do I turn off the console colors in the logs? |
| 20:41.23 | [TK]D-Fender | fifer: and NO, renaming it will NOT do. |
| 20:41.25 | frantik | thats all i got generated by the thing |
| 20:41.33 | Kobaz | asterisk -n |
| 20:41.52 | frantik | did the asterisk -cccc thing > full.log |
| 20:42.03 | sbrath | That's taken from /var/log/asterisk/full .. How do I change the log to not have colors... or is that -n overall |
| 20:42.22 | ManxPower-work | frantik, how about trying "asterisk -cvvv" instead of -ccc |
| 20:42.26 | frantik | ManxPower-work, is that thing saved anywhere ? |
| 20:42.29 | Kobaz | i'm pretty sure that disables colors for the log too |
| 20:42.33 | tzafrir_laptop | fifer, and add in the end the line: #include zapata-channels.conf |
| 20:42.40 | sbrath | can I change it without restarting? |
| 20:42.48 | ManxPower-work | frantik, sure, but when you are crashing on startup it's pretty pointless. |
| 20:42.56 | Kobaz | try asterisk -rvvvn &> /tmp/log |
| 20:42.59 | frantik | ManxPower-work, i said "thing" because i no longer had the command in buffer but it's what i typed |
| 20:43.05 | ManxPower-work | frantik, try "asterisk -cvvv > mylog.lod 2>&1" |
| 20:43.11 | frantik | k |
| 20:43.17 | *** join/#asterisk ruben23 (~ITadmin@122.55.48.243) |
| 20:43.26 | ManxPower-work | otherwise the ERROR message may not be sent to the log |
| 20:43.40 | Kobaz | ManxPower-work: &> does the same thing |
| 20:43.44 | frantik | getting that file |
| 20:43.46 | ManxPower-work | tzafrir_laptop, is this not documented anywere? |
| 20:43.48 | frantik | in raw mode |
| 20:43.53 | frantik | hod on... |
| 20:44.34 | tzafrir_laptop | ManxPower-work, not sure |
| 20:45.05 | ManxPower-work | didn't you write the genzap stuff? |
| 20:45.50 | sbrath | If I changed the colsole-colors = no in asterisk.conf is their a non re-start way to reload that setting? |
| 20:46.19 | ManxPower-work | sbrath, I thought you were asking about colors in logs, not the console |
| 20:46.40 | Kobaz | sbrath: would help if it was spelled right |
| 20:47.06 | *** join/#asterisk neurosys (~neurosys@c-71-196-20-208.hsd1.fl.comcast.net) |
| 20:47.11 | sbrath | actually it's nocolor=yes .. (tough audience) |
| 20:47.39 | sbrath | ManxPower-work: I was asking about in logs, but I don't want to restart to turn it off. |
| 20:47.41 | *** join/#asterisk Ad-Hoc (~nimbus@62.1.165.106.dsl.dyn.forthnet.gr) |
| 20:48.34 | Skeeter- | ariel_, Can you change the language of the Spectralinks? |
| 20:48.54 | fifer | I now have a zapata.conf ending with an include to zapata-channels.conf, both here in: http://pastebin.ca/1823081 |
| 20:49.00 | frantik | ManxPower-work, raw log scped trough various vpn |
| 20:49.01 | frantik | http://tuxedo.cos.eu/mylog.lod |
| 20:49.02 | *** join/#asterisk jelly-bean (~jelly-bea@75-148-103-190-Utah.hfc.comcastbusiness.net) |
| 20:49.05 | ariel_ | Skeeter-: I do not know, I have not had to do that |
| 20:49.06 | *** part/#asterisk jelly-bean (~jelly-bea@75-148-103-190-Utah.hfc.comcastbusiness.net) |
| 20:49.26 | Skeeter- | ariel_, thats ok, but i dont think that it is possible |
| 20:49.36 | frantik | result of the command you given |
| 20:49.40 | Skeeter- | ariel_, was simply asking, i know you have a lot of backgroun with those |
| 20:49.43 | fifer | I restarted * just to cover bases and did a zap show channels with still now channels listed |
| 20:49.49 | ruben23 | hi |
| 20:49.54 | [TK]D-Fender | fifer: I see no zapata.conf in there |
| 20:50.31 | frantik | maybe its that extension directory that is the issue ? o_0 |
| 20:51.05 | fifer | line 01 indicates the file name, lines 03-40 are the zapata.conf |
| 20:51.13 | dddh | still cannot make everything work |
| 20:51.14 | raden_work | how can i limit my IVR to only repeat / loop 3 times ? |
| 20:51.19 | fifer | http://pastebin.ca/1823095 |
| 20:51.22 | ariel_ | Skeeter-: they have manuals for there phones in English, Francais, Espanol, Deutsch, Italiano maybe they are able too switch |
| 20:51.25 | fifer | it changed the number, my bad |
| 20:51.46 | Skeeter- | ariel_, i raed it |
| 20:52.09 | Skeeter- | ariel_, they simply explain in french how to use an english phone if u get what i mean |
| 20:52.22 | Skeeter- | ariel_, terms are stil the same |
| 20:52.24 | [TK]D-Fender | fifer: And the attempts to verify if the configs are OK, and the card is responsive are where? |
| 20:53.24 | fifer | cat /proc/zaptel/* shows the card and the fxo ports, three of the fxo say RED |
| 20:53.38 | [TK]D-Fender | fifer: PASTEBIn your backup |
| 20:54.13 | *** join/#asterisk Alagar (~alagar_20@122.164.32.198) |
| 20:54.17 | fifer | backup? |
| 20:55.05 | fifer | what can I gleen from zttool? |
| 20:55.24 | fifer | I have been away from zaptel for almost 2 years, very rusty |
| 20:55.32 | sbrath | OK try this: http://pastebin.com/R5SAMAgv |
| 20:56.19 | *** join/#asterisk medicineman (~medicinem@cpe-75-87-82-200.kc.res.rr.com) |
| 20:57.41 | ManxPower-work | frantik, looks like it is actually crashing when loading app_directory_odbc.so |
| 20:58.01 | ManxPower-work | fifer, start by reading the ZAPTEL README |
| 20:58.17 | fifer | headed there now |
| 20:58.25 | ManxPower-work | fifer, should have been the first thing |
| 20:58.27 | *** join/#asterisk sulex (~sulex@host-78-14-173-189.cust-adsl.tiscali.it) |
| 20:58.31 | leifmadsen | raden_work: yes you can! :) |
| 20:58.38 | leifmadsen | raden_work: just use a counter |
| 20:58.47 | leifmadsen | raden_work: show me your existing dialplan for your IVR |
| 20:59.02 | ManxPower-work | So many people giving away fish today. |
| 20:59.02 | frantik | ManxPower-work, how do i disable loading that ? |
| 20:59.23 | miamiseb | rather than teaching to fish? |
| 20:59.26 | fifer | one thing we were talking about at the biginning was wether I messed things up by installing the version of zaptel I did, could I have messed up dahdi in the process? |
| 20:59.31 | fifer | still headed to the readme |
| 20:59.33 | leifmadsen | ManxPower-work: well I want to see what it looks like so I can make sure when I teach him how to fish that I'm doing it in a constructive manner |
| 20:59.35 | ManxPower-work | frantik, in /etc/asterisk/modules.conf noload => app_directory_odbc.so |
| 20:59.40 | frantik | thx |
| 21:00.01 | ManxPower-work | frantik, and you NEED to read the Asterisk Book. |
| 21:00.03 | ManxPower-work | ~answers |
| 21:00.04 | infobot | well, answers is Asterisk Book: http://astbook.asteriskdocs.org & SIP w/ NAT: http://www.aocomputing.net/?p=3 & Wiki: http://www.voip-info.org & docs: /path/to/src/asterisk/doc & 1.4 .vs. 1.6.0 .vs. 1.6.1: : /path/to/src/asterisk/UPGRADE*.txt |
| 21:00.08 | frantik | teaching to fish is fine, but when you only need to learn how to cut the fish, there is a need for that too. |
| 21:00.26 | frantik | ManxPower-work, can't, got 1hour to manage that or revert to backup |
| 21:00.38 | leifmadsen | raden_work: you'll probably want to use a counter along with a GotoIf() |
| 21:00.41 | ManxPower-work | frantik, Huh? |
| 21:00.51 | frantik | i don't have all day to manage it |
| 21:00.51 | miamiseb | he can't take the time to read the book |
| 21:00.55 | frantik | got other tasks awaiting me |
| 21:01.06 | ManxPower-work | oh, then we should toss his sorry ass to the sharks |
| 21:01.10 | leifmadsen | sounds like a problem for a consultant than |
| 21:01.11 | fifer | the main README was what I used to install zaptel yesterday, made sense and it went well |
| 21:01.13 | frantik | the project dev is learning and i am the general sysadmin |
| 21:01.18 | miamiseb | No time for spoon feeding ManxPower-work? |
| 21:01.20 | frantik | so i'm trying |
| 21:01.23 | miamiseb | lol |
| 21:01.25 | fifer | does not talk about the utilities, will look for other readme's |
| 21:01.27 | frantik | but if in the end it takes too long |
| 21:01.31 | frantik | it get reseted |
| 21:01.38 | ManxPower-work | frantik, lack of time is not an excuse for not learning stuff. |
| 21:01.39 | frantik | i'm pragmatic |
| 21:01.54 | frantik | ManxPower-work, when you got work to do, it is |
| 21:02.00 | ManxPower-work | frantik, don't worry, I won't be wasting my time with you again |
| 21:02.03 | frantik | voip not being my particular field of intereset :) |
| 21:02.11 | leifmadsen | Lack of planning on your part does not constitute and emergency on mine..... comes to mind. |
| 21:02.16 | frantik | ManxPower-work, oh thanks already for the help |
| 21:02.19 | miamiseb | haha |
| 21:02.26 | frantik | not saying the help wasnt appreciated |
| 21:02.27 | ManxPower-work | You are looking for free help from the volunteers in this channel because you can't learn what you need to learn. |
| 21:02.33 | leifmadsen | not that it necessarily applies here, I just like that saying |
| 21:02.35 | frantik | i'm saying that if it works its cool |
| 21:02.40 | ManxPower-work | In the real work you hire a consultant, not come here. |
| 21:02.43 | frantik | if not, then i'll apply another solution |
| 21:02.45 | fifer | both zttool and cat/proc/zaptel/* show the card |
| 21:02.46 | ManxPower-work | s/work/world/ |
| 21:02.51 | dinesh___ | well there are user friendly tools available such as sipbroker, freepbx and this kind of stuff |
| 21:02.57 | leifmadsen | ok, we're significantly off topic now |
| 21:03.01 | dinesh___ | i think they have a much smaller learning curve |
| 21:03.03 | frantik | ManxPower-work, in the real world i have a job to do and a list of tasks |
| 21:03.04 | dinesh___ | but i didn't try those |
| 21:03.06 | frantik | and a boss |
| 21:03.08 | miamiseb | frantik: either way, I'm pretty sure they anwser the question, did you get that noload? |
| 21:03.14 | ManxPower-work | frantik, so do all of use in this channel. |
| 21:03.18 | leifmadsen | we can either help, or we can not help -- but berating someone for asking questions is not a solution |
| 21:03.21 | frantik | and its the same thing for every sysadmin can understand |
| 21:03.37 | ManxPower-work | frantik, but you are not a PBX admin. |
| 21:03.55 | Katty | ATTENTION |
| 21:03.56 | frantik | ManxPower-work, general sysadmin |
| 21:03.59 | leifmadsen | ManxPower-work: ok we get it, move on |
| 21:04.00 | Katty | DO YOU KNOW WHERE YOUR CAR KEYS ARE |
| 21:04.01 | *** join/#asterisk corretico (~laguilar@201.201.46.106) |
| 21:04.10 | leifmadsen | Katty: yes, they are with my fiancee at work |
| 21:04.12 | frantik | i manage the server software, appliances, backups, some configs, but not that one |
| 21:04.16 | ariel_ | Katty: yes |
| 21:04.22 | Katty | k |
| 21:04.23 | frantik | its the experimental project of one of our dudes |
| 21:04.28 | miamiseb | Katty: out to lunch? |
| 21:04.49 | Katty | buwha? |
| 21:04.57 | fifer | I have * 1.4.29.1 and I installed zaptel-1.4.12.1 yesterday, would that have messed things up? before this card, I had not setup anything other than SIP trunking and phones on this box |
| 21:05.28 | ariel_ | fifer: why zaptel and not dahdi? |
| 21:05.47 | fifer | OpenVox says not to use dahdi with this card in production yet |
| 21:05.53 | frantik | anyway thanks i'll take a closer look to modules |
| 21:06.35 | miamiseb | I plan on using 711u, any reason for me to choose info for DTMF tones vs inband? |
| 21:06.57 | idespinner | miamiseb, most people do rfc2833 or whatever |
| 21:06.59 | *** join/#asterisk jameswf (~james@unaffiliated/jameswf-home) |
| 21:07.02 | idespinner | thats the typical one |
| 21:07.08 | ManxPower-work | miamiseb, no, but there is a reason to use RFC2833 instead of inband. |
| 21:07.09 | miamiseb | the other side is ignoring that |
| 21:07.29 | *** part/#asterisk ManxPower-work (~manxpower@216.186.151.147) |
| 21:07.59 | *** join/#asterisk Wildy (~simba@89.222.134.42) |
| 21:08.12 | idespinner | miamiseb, are you setting this up for a sip trunk or a endpoint(phone)? |
| 21:08.21 | miamiseb | sip trunk |
| 21:08.46 | idespinner | miamiseb, you probably want to double check with your sip trunk provider to make sure you use the same DTMF method as them |
| 21:08.48 | miamiseb | the phone is a cisco phone connected to callmanager via another trunk |
| 21:08.53 | drmessano | Anyone had Gizmo5 "block" their registration due to having qualify set or frequent registration attempts? |
| 21:09.10 | miamiseb | yeah, their IVR uses dtmf tones for navigation and I didn't feel like using the cell |
| 21:09.10 | miamiseb | lol |
| 21:09.16 | *** join/#asterisk nephlite (~jake@72-160-157-250.dyn.centurytel.net) |
| 21:09.39 | dinesh___ | mwerf, gizmo was bought by google |
| 21:09.42 | dinesh___ | :( |
| 21:09.42 | miamiseb | so in order to find out, I have to fix it, or I could be more lazy and just use the cell. |
| 21:09.54 | ariel_ | As far as I know all the Cisco's CM I have use RFC2833 |
| 21:10.00 | fifer | Is there a problem with using the a new version of zaptel with the latest * 1.4? |
| 21:10.27 | fifer | One person said dahdi was required another said no, just trying to make sure |
| 21:10.34 | miamiseb | right, it'll work fine for that leg, if I send it out of a different trunk they get passed, but the termination leg of the call isn't getting em |
| 21:10.43 | russellb | zaptel is supposed to work with any version of 1.4 |
| 21:10.57 | russellb | however, i strongly recommend using DAHDI, as zaptel has not been maintained for a very long time now |
| 21:11.56 | drmessano | wishes SFA could be used with Skype subscriptions #skypeforasterisk #skype #skypein #skypeout |
| 21:12.05 | drmessano | *tweet* |
| 21:12.29 | fifer | I will switch to dahdi as soon as OpenVox says their new compatilbility with it is ready for production |
| 21:12.32 | *** join/#asterisk adnc (~numer@unaffiliated/adnc) |
| 21:12.50 | fifer | unless anyone here has personal experience with an a1200p or a800p and dahdi? |
| 21:12.54 | *** join/#asterisk bjhaid (~herbayjha@41.206.15.3) |
| 21:13.42 | russellb | ugh |
| 21:14.12 | Skeeter- | whats is the fastest way to restore a complete linux system with a RAID1 |
| 21:14.13 | bjhaid | I am really new to asterisk a complete newbie, just completed installation of asterisk, and I have x-lite on my ubuntu 9.10 machine, I do not know what the next steps should be? |
| 21:14.31 | miamiseb | umm, disconnect the one dead drive? |
| 21:14.31 | miamiseb | lol |
| 21:14.43 | drmessano | Skeeter- remove the dead.. damnit |
| 21:14.47 | ariel_ | fifer: have you ever heard of setting up your own little lab for testing before going into production...? |
| 21:14.49 | drmessano | Fine |
| 21:14.59 | Skeeter- | let me rephrase my question |
| 21:15.01 | miamiseb | bjhaid: probably setting up an extension, and then maybe a trunk for external access |
| 21:15.08 | *** join/#asterisk rossand (~aross@CPE000c413a19a3-CM00159a025ad4.cpe.net.cable.rogers.com) |
| 21:15.12 | Skeeter- | restore on 2 other drives |
| 21:15.18 | miamiseb | okay |
| 21:15.32 | ariel_ | remove dead drive put new drive then use mdadm to resync |
| 21:15.33 | drmessano | Image with any number of imaging apps? |
| 21:15.50 | miamiseb | you have several options I'd say, either dd, or tar the filesystem then move it over and reload the bootloader (grub or lilo), assuming its the same hardware |
| 21:16.17 | fifer | ariel_: when the manufacture just announces that they have dahdi compat and says you should NOT use it in prod, I'm not going to argue. I know from personel experience how hard it is to do real world testing in a lab setting and I'm a one man show here. |
| 21:16.18 | miamiseb | or sure there's that, we use acronis, but it really only works for data, imaging the whole drive has been flaky |
| 21:16.20 | bjhaid | miamiseb how do I get that done, and moreover I dont have my xlite running yet |
| 21:16.50 | Skeeter- | miamiseb, yeah those are the method i already use( not the acronis one, watever it is) |
| 21:17.00 | miamiseb | and...? |
| 21:17.09 | miamiseb | is it software or hardware raid 1? |
| 21:17.14 | Skeeter- | software |
| 21:17.15 | drmessano | I've had about 90% success with Acronis.. it's gotten better |
| 21:17.18 | Katty | hmm |
| 21:17.25 | Katty | plays some zelda while symantec has her on hold |
| 21:17.33 | drmessano | But that's still 90% |
| 21:17.36 | Naikrovek | anyone know an indian in hyderabad who is smart, talented, sysadmin hotshot guy who is looking for work? |
| 21:17.37 | miamiseb | bjhaid: http://astbook.asteriskdocs.org/en/2nd_Edition/asterisk-book-html-chunk/asterisk-CHP-4-SECT-7.html |
| 21:17.37 | Skeeter- | Katty, which one? |
| 21:17.41 | *** join/#asterisk The_Boy_Wonder (~vossel@asterisk/batman-developer/dvossel) |
| 21:17.45 | jaytee | i've used Acronis to image a full disk system with LVM on RHEL 5.2 and it works fine. I've even tested the full restore several times. |
| 21:17.47 | Naikrovek | Skeeter-: they're all good |
| 21:17.55 | miamiseb | Skeeter-: and whats the problem you had with the restore? |
| 21:17.57 | *** join/#asterisk chuckf (~chuckf@ubuntu/member/chuckf) |
| 21:18.05 | Katty | Skeeter-: the original zelda, for nes |
| 21:18.07 | bjhaid | thanks |
| 21:18.09 | Skeeter- | Naikrovek, i asked which one, nothing i metioned on quality |
| 21:18.10 | ariel_ | fifer: I as well am my own one man shop, and have many test system in my lab and don't deploy to any of my customer anything that has not been tested and run through my lab. I even have E1's, and a thrunder Bird test unit. |
| 21:18.16 | miamiseb | I've got it on debian, and even building the kernel module had problems. |
| 21:18.31 | jaytee | gotta run, be back later |
| 21:18.32 | Naikrovek | Skeeter-: true |
| 21:18.49 | Skeeter- | miamiseb, i takes about 15 minutes to restore a base debian with all good partion and untar the backup, restore grub |
| 21:18.58 | Skeeter- | miamiseb, wanted sth faster thats it |
| 21:19.26 | dinesh___ | wow gizmo is so expensive now , it used to be much much cheaper |
| 21:19.32 | dinesh___ | and it cannot even call to vietnam anymore |
| 21:19.49 | fifer | ariel_: I'm not currently working as or in an * shop. I am the entire IT department for for a small company. I have set up 8 * systems in the past but I have been out of * for about 1.5 or more years. |
| 21:19.50 | Skeeter- | Katty, can u throw ur sword yet? |
| 21:19.51 | ariel_ | miamiseb: I use lenny here and acronics |
| 21:20.07 | Katty | Skeeter-: erm |
| 21:20.14 | Katty | Skeeter-: as long as you're full health you can throw your sword |
| 21:20.25 | drmessano | Yep |
| 21:20.27 | Skeeter- | Katty, u gotta earn that |
| 21:20.32 | ariel_ | fifer: you are a end user then, call OpenVox and get them to help you out then. |
| 21:20.35 | Skeeter- | Katty, its not a out of the box skill |
| 21:20.36 | Naikrovek | not in the first zelda you don't |
| 21:20.38 | fifer | I am testing things, but I have little time and need to head in the best direction based on the info I have |
| 21:20.40 | drmessano | Erm nope |
| 21:20.42 | miamiseb | only thing faster than 15 minutes (which btw, for a time to recovery from total loss is freaking beautiful) I can imagine would be real-time backup and I'm not experienced with that on linux. In windows, we use double take. |
| 21:20.44 | Skeeter- | really?!? |
| 21:21.11 | Naikrovek | yeah |
| 21:21.35 | Katty | Skeeter-: yes. |
| 21:21.37 | Naikrovek | original NES Zelda you don't need to earn the throwing sword thing. Just have full health |
| 21:21.44 | Naikrovek | you're thinking boomerang or something |
| 21:21.51 | Katty | boomerang is in lvl 1 |
| 21:21.52 | fifer | ariel_: You can call me what you wish, and choose to not help me, but I'm not goign to call OpenVox and get nowhere |
| 21:21.58 | wcselby | or snes zelda |
| 21:21.59 | drmessano | I was up to 8 successive Zelda finishes and then the red flash of doom nuked my saved game |
| 21:22.01 | ariel_ | miamiseb: drbd with hart beat, Xorcom Asterbank with twinstar, failover on our server within 1 minute. |
| 21:22.10 | miamiseb | Also, use zelda as your name and you start on the second quest, always fun. |
| 21:22.20 | Naikrovek | drmessano: emulation! |
| 21:22.21 | Skeeter- | miamiseb, most of our client server have vmware, if a server physicaly crash, and theres another one, it takes about .75 to 1.5 minu to restore |
| 21:22.32 | fifer | I have spent time in the past helping others in this irc channel and I'm not new to *, just new to dahdi and this card |
| 21:22.40 | ariel_ | fifer: I am not calling you anything, but I do know dahdi works on just about every board I have used it on |
| 21:22.40 | drmessano | Naikrovek: 1988-1991 :) |
| 21:22.44 | Naikrovek | aah |
| 21:22.46 | miamiseb | there you go drbd seems to be real time backup for linux |
| 21:22.57 | Katty | emulaton++ |
| 21:23.12 | Naikrovek | ooh |
| 21:23.19 | Naikrovek | when does stargate universe start back up |
| 21:23.26 | ariel_ | I hope soon |
| 21:23.39 | Skeeter- | tv.com got lots on info on shows |
| 21:23.58 | Skeeter- | then check ezrss.it or eztv.it to get the latest release |
| 21:23.59 | miamiseb | Skeeter-: but using vmotion is cheating. =) |
| 21:24.05 | miamiseb | hugs eztv.it |
| 21:24.07 | Skeeter- | miamiseb, why? |
| 21:24.09 | Katty | and btw, NESCafe works with WMC |
| 21:24.17 | miamiseb | cause its licensed, and I'm cheap |
| 21:24.18 | Katty | and therefore xbox 360 with the WMC plugin |
| 21:24.27 | sbrath | So what else could be preventing me from transfering a call in the queue? Could it be in the wrong context? |
| 21:24.40 | Skeeter- | miamiseb, it works pretty nasty good tho |
| 21:24.46 | Skeeter- | miamiseb, support is amazing |
| 21:25.03 | *** join/#asterisk Gugge (~gugge@vlan2.dlxhosting.dk) |
| 21:25.24 | *** join/#asterisk Dibri (~gavit@190.98.33.229) |
| 21:25.32 | miamiseb | Yeah, I know, but our own stuff we just have a hot spare running and its way cheaper. The number of servers we have (even though we are an ISP) is rather small |
| 21:25.37 | sbrath | A call directly to a phone can be transfered, the same call by the same phone but delivered via Queue() trys to transfer but the caller just gets MoH and eventually dropped. I don't see anything in the log as to what it's trying to do with it. |
| 21:25.57 | miamiseb | sbrath: pastebin the logs |
| 21:26.33 | *** join/#asterisk c0rnoTa (~c0rnoTa@nas-8.emserv.ru) |
| 21:26.53 | *** part/#asterisk jnfuller (~jnfuller@d64-180-206-233.bchsia.telus.net) |
| 21:26.54 | *** part/#asterisk c0rnoTa (~c0rnoTa@nas-8.emserv.ru) |
| 21:28.55 | sbrath | http://pastebin.com/R5SAMAgv |
| 21:31.10 | *** join/#asterisk bjhaid (~herbayjha@41.206.15.3) |
| 21:31.27 | [TK]D-Fender | sbrath: And the reason we're not seeing the SIP DEBUG for SIP/4120 is why exactly? |
| 21:31.30 | [TK]D-Fender | BBIAB |
| 21:32.48 | mbrevda | anyone using imap vm and gmail? |
| 21:34.41 | Corydon76-dig | Uh, why exactly are you depending upon a remote IMAP server for voicemail? |
| 21:34.45 | Nugget | I had to stop using imap vm because it was too crashy |
| 21:35.36 | Corydon76-dig | Nugget: I think the bigger problem is that IMAP support is very leaky. |
| 21:36.35 | Katty | likes symantecs on hold musics |
| 21:36.38 | *** part/#asterisk frantik (~frantik@190.114.99-84.rev.gaoland.net) |
| 21:36.48 | Nugget | FlightAware's hold music is awesome |
| 21:37.18 | Nugget | http://macnugget.org/crud/flightaware_hold_music.mp3 |
| 21:37.46 | Katty | telnet |
| 21:37.49 | Katty | :< |
| 21:37.55 | Naikrovek | i know, right |
| 21:38.01 | leifmadsen | Nugget: haha nice |
| 21:38.02 | Naikrovek | it's like expected at this point |
| 21:38.04 | Naikrovek | and it fails us |
| 21:38.07 | Katty | :> |
| 21:38.23 | leifmadsen | your mom fails us |
| 21:38.32 | leifmadsen | looks at Katty :) |
| 21:38.32 | Naikrovek | yeah she failed me too :( |
| 21:38.36 | Katty | your mom's face fails us. |
| 21:38.41 | ellisdee | y0 |
| 21:38.48 | miamiseb | Yeah Naikrovek's mom failed me too! |
| 21:38.50 | Katty | I JUST GOT A TRIFORCE CHUNK |
| 21:39.33 | Katty | where's lvl 2 |
| 21:39.38 | Naikrovek | dungeon 2 |
| 21:39.41 | Katty | yeah |
| 21:39.43 | Katty | where is it |
| 21:39.44 | Naikrovek | intarweb knows |
| 21:39.48 | Katty | i'm too lazy |
| 21:40.00 | Naikrovek | it's easier than hunting all over hte map in-game |
| 21:40.03 | wcselby | katty, it's over there ---> |
| 21:40.14 | Katty | Naikrovek: yeah i looked up a map |
| 21:40.40 | miamiseb | http://www.gamefaqs.com/console/nes/file/563433/27772 |
| 21:40.41 | Katty | http://www.jasonenneking.com/pages/Wii/NES/Zelda/legend_of_zelda_overworld.png |
| 21:40.43 | bmoraca_work | Nugget: what would be even cooler is if that voice overlay was a live feed |
| 21:40.57 | *** join/#asterisk Dibri (~gavit@190.98.33.229) |
| 21:41.09 | Nugget | thought about that, but it's just not practical. we take calls 24 hours a day, even when traffic is light and there's lots of dead air. |
| 21:41.18 | Nugget | I had to trim out a lot of empty when I made the clip |
| 21:42.18 | bmoraca_work | it still would have been cool |
| 21:42.42 | sbrath | [TK]D-Fender : I wasn't doing Sip debug, let me try to get..... |
| 21:43.41 | *** join/#asterisk TheDavidFactor (~chatzilla@nc-71-0-16-133.dhcp.embarqhsd.net) |
| 21:43.52 | Katty | yays, lvl 2 |
| 21:44.45 | *** join/#asterisk defsdoor (~andy@defsdoor.gotadsl.co.uk) |
| 21:45.38 | Naikrovek | Katty: by jove that's suitable for my wide-format printer i dont have |
| 21:45.45 | sbrath | [TK]D-Fender : OK, I think I know why the transfer isn't working. I'm changing the CallerID to something longer, that our Phones are OK with, but when the CallerID is set to the "CUST# 12345 6085551222" the transfer dosen't complete. |
| 21:46.02 | *** join/#asterisk wonderworld (~ww@ip-62-143-22-226.unitymediagroup.de) |
| 21:46.06 | Katty | Naikrovek: it would make a lovely poster |
| 21:46.18 | Naikrovek | yeah |
| 21:46.22 | Naikrovek | katty and i could hang |
| 21:46.25 | Naikrovek | video games |
| 21:46.27 | Naikrovek | phones |
| 21:46.29 | *** join/#asterisk fish-bulb (~cstewart@nat/digium/x-leqhqnriwojqkkwz) |
| 21:47.30 | Katty | moms |
| 21:47.35 | Naikrovek | my mom sucks |
| 21:47.42 | Katty | mine's awesome. |
| 21:47.42 | Naikrovek | terrible person |
| 21:47.53 | Corydon76-dig | fish-bulb: Good afternoon |
| 21:48.42 | fish-bulb | Corydon76-dig: howdy |
| 21:49.59 | *** join/#asterisk rubberneck (~chatzilla@ext-52.sagetelecom.net) |
| 21:51.00 | fish-bulb | KK5G |
| 21:51.10 | sbrath | OK, If I change the CALLERID(num) before I deliver it to a phone, and then that phone attempts a transfer, the transfer fails. |
| 21:51.42 | *** join/#asterisk cweagans (~432aa645@gateway/web/freenode/x-fvelrjstiwlpsutw) |
| 21:52.06 | cweagans | does anybody use speakeasy sip trunks on their asterisk box? |
| 21:52.14 | cweagans | (er, anybody that's here, that is) |
| 21:52.23 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 21:52.51 | leifmadsen | anyone have a recommendation for a small PoE switch that can power Polycom phones? |
| 21:53.01 | TheDavidFactor | does anyone know what's required to make userfield=1 in cdr_mysql.conf take affect on a running system? Asterisk 1.4.0 and it didn't seem to take just by changing it and reloading the module |
| 21:54.19 | [TK]D-Fender | LeHow small? |
| 21:54.30 | rubberneck | I am using Asterisk 1.6.2.1 with DAHDI Version: 2.2.1 in an all SIP environment. I am having an issue with the meetme application. when users call into a conference, all is well and sounds excellent. The probelm is that when the callers hang up the channels never drop and remain in the conference indefinitely. Anyone seen this or have any suggestions? |
| 21:55.21 | [TK]D-Fender | rublook at the SIP debug. this has nothing to do with MeetMe |
| 21:57.01 | rubberneck | [TK]D-Fender: i will have to look, wanted to throw it out here because this server is handling many other calls that are not meetme related, but only the meetme calls have this symptom. |
| 21:57.50 | cweagans | Speakeasy sent me SIP credentials this morning and I can't get them to work (mostly because I'm an asterisk noob). I called them (since they say they support asterisk) and apparently, they only support asterisk via FreePBX or similar. |
| 21:59.30 | wcselby | the freepbx credentials should be the same as asterisk credentials |
| 21:59.43 | wcselby | since freepbx is just a gui front-end for asterisk... |
| 22:00.14 | Naikrovek | folks in here can translate for you |
| 22:00.20 | Katty | YAY! |
| 22:00.33 | Katty | another heart, hunk of triforce, and my blue boomerang |
| 22:02.10 | [TK]D-Fender | rubberneck: I'd immediately recommend upgrading before looking at anything else |
| 22:02.15 | cweagans | wcselby: well, yes. But I'm not sure where to put them (or how to set up my dialplan for dialing out or recieving calls using the trunk) |
| 22:03.01 | [TK]D-Fender | cweagans: Dial(SIP/peeryoumadeinsip.con/thenumberyouwanttodial) |
| 22:03.14 | Naikrovek | leifmadsen: overwhelming options huh |
| 22:03.17 | wcselby | cweagans - get their configs, pastebin them, then ask for some help in here |
| 22:03.30 | LemensTS | Katty: lol u playing nes roms? |
| 22:03.43 | cweagans | wcselby: the configs? like...my sip.conf and extensions.conf? |
| 22:04.17 | [TK]D-Fender | cweagans: I gave you the kind fo formatting you'll need for your dialplan... 1 TSP isn't terribly different from the next |
| 22:04.41 | *** join/#asterisk RobH (~robh@216.38.133.254) |
| 22:05.10 | rubberneck | [TK]D-Fender: upgrading? man I just compiled this a little while ago, what version are we at now? let me look. |
| 22:05.19 | cweagans | [TK]D-Fender : I know the Dial() part of it. I'm not sure how to write the first part of it (exten => ... ) Speakeasy requires some weird things that I'm not sure how to do. |
| 22:08.39 | miamiseb | cweagans: but you have been able to get your trunk to them setup with the supplied credentials? |
| 22:08.54 | cweagans | I don't know. |
| 22:08.57 | cweagans | <--noob. |
| 22:09.02 | miamiseb | sip show peers from console please |
| 22:09.04 | cweagans | how can I find out? |
| 22:09.05 | cweagans | sure |
| 22:09.10 | *** join/#asterisk Dibri (~gavit@190.98.33.229) |
| 22:09.44 | [TK]D-Fender | cweagans: No, they don't |
| 22:10.14 | cweagans | [TK]D-Fender: the email from them says they do.. |
| 22:10.30 | [TK]D-Fender | cweagans: and the part before is the dialplan pattern. This has nothing to do with them. what matters is the # you pass them and the auth and networking setup |
| 22:10.34 | cweagans | miamiseb: yes, the speakeasy trunk is in there |
| 22:10.40 | [TK]D-Fender | cweagans: And I'm telling you "no". |
| 22:10.57 | [TK]D-Fender | cweagans: Do show us your actual attempt with SIP DEBUG enabled <- |
| 22:11.02 | miamiseb | is it qualified? or does it show unmonitored? |
| 22:11.09 | cweagans | unmonitored |
| 22:11.14 | sbrath | So my lesson learned is that Over-riding the CALLERID(num) with something that dosen't look like a telephone number makes Transfer mad. |
| 22:11.34 | cweagans | miamiseb: however, the peer details they sent me specify qualify=no |
| 22:11.38 | cweagans | if that matters? |
| 22:12.24 | miamiseb | that means its correct |
| 22:12.29 | miamiseb | whats the trunk called? |
| 22:12.31 | cweagans | speakeasy |
| 22:12.58 | ariel_ | it |
| 22:14.03 | cweagans | miamiseb: do you want sip.conf and extensions.conf to look at? or? |
| 22:14.18 | miamiseb | no, and be careful sending out your sip.conf without munging the user and secret |
| 22:14.23 | cweagans | yeah |
| 22:14.24 | cweagans | I know :) |
| 22:14.35 | cweagans | thanks though :) |
| 22:15.15 | miamiseb | go ahead and pastebin your extensions.conf |
| 22:15.23 | miamiseb | I'm assuming you have an extension (phone) registered? |
| 22:15.28 | cweagans | yeah |
| 22:15.34 | cweagans | 5 of them, actually :) |
| 22:16.47 | cweagans | miamiseb: http://pastebin.com/BuP7gvF4 |
| 22:17.07 | cweagans | also, I like the new look on pastebin.com! Doesn't look so junky =P |
| 22:18.49 | miamiseb | and what happens when you dial a 10 digit string from one of the phones? |
| 22:19.02 | miamiseb | btw, im almost sure its a notoriously bad idea to leave that under your default |
| 22:19.23 | [TK]D-Fender | Where's the FAILED CALL with SIP DEBUG? |
| 22:19.26 | miamiseb | unless your using a firewall to block sip requests from unknown sources, they'd be able to use your speakeasy trunks to dial out to the world |
| 22:19.51 | cweagans | heh, alright. Where should that go then? |
| 22:20.17 | miamiseb | in a context that authenticated users would be put into, likely, internal-calls, or something other context you include from there |
| 22:20.36 | LemensTS | Wouldn't it be nice to allow people on here with a click of a button acccess to your ssh screen session? |
| 22:20.38 | Katty | YAY |
| 22:20.43 | Katty | NEW SWORD! |
| 22:20.44 | Katty | boingboing |
| 22:20.45 | LemensTS | Safely. |
| 22:21.18 | leifmadsen | [TK]D-Fender: I was thinking like 4-5 ports. Just for my home phones so I can save on wall warts |
| 22:21.40 | LemensTS | Experts could save frustation and time. Users could get help easier. |
| 22:21.44 | leifmadsen | rubberneck: that problem sounds like something I saw in the bug tracker -- I'd search for meetme issues (look at both open and closed issues) -- it may already be resolved |
| 22:21.54 | [TK]D-Fender | leifmadsen: I use The 8-port Netgear RP-108 (IIRC) at the office in a few places. 4 PoE, 4 regular ~140$ |
| 22:22.05 | [TK]D-Fender | FP maybe |
| 22:22.14 | leifmadsen | [TK]D-Fender: ah yes, I remember that one now. That sounds like it should be just right up my alley |
| 22:22.20 | leifmadsen | [TK]D-Fender: have polycom phones running off of it? |
| 22:22.38 | fifer | Thanks to all for your help today!!! |
| 22:22.40 | [TK]D-Fender | leifmadsen: Full load of 4 |
| 22:22.50 | fifer | I switched to dahdi and everything seems to be fine now |
| 22:23.33 | fifer | Just need to work out some dial plan things, but it is working at least partialy both outgoing and incomming |
| 22:23.40 | Katty | huzzah! lvl 3 |
| 22:25.57 | Naikrovek | you go, girl |
| 22:26.00 | Naikrovek | still on hold? |
| 22:26.15 | Naikrovek | i seem to recall you saying you were going to play while you were on hold |
| 22:26.59 | cweagans | also, is this post still accurate? http://forums.whirlpool.net.au/forum-replies.cfm?t=679361&r=10616507#r10616507 |
| 22:27.15 | LemensTS | Katty; http://www.cnn.com/video/#/video/us/2010/03/03/dnt.blind.gamer.beats.zelda.wis |
| 22:27.18 | cweagans | (more specifically, is that where the nat=yes and all that needs to go?) |
| 22:27.30 | cweagans | ports are forwarded, but not sure where to stick that config stuff. |
| 22:28.29 | cweagans | (confused because sip_nat.conf doesn't seem to exist. will it be used if I just create it?) |
| 22:28.34 | adnc | i see that an asterisk distribution called gemeinschaft has a page with call volume statistics. has someone got any ideas where i could extract this information directly out of asterisk? |
| 22:29.15 | hardwire | hugs IAXVAR |
| 22:31.01 | cweagans | nvm, found it |
| 22:32.57 | *** join/#asterisk uqlev (~yuriy@91.184.221.31) |
| 22:33.53 | Katty | i sure wish symantec would hurry up |
| 22:33.55 | Katty | i gots to pee |
| 22:39.27 | [TK]D-Fender | cweagans: Rad ---> |
| 22:39.29 | [TK]D-Fender | ~sipnat |
| 22:39.30 | infobot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 22:39.32 | [TK]D-Fender | read* |
| 22:41.31 | cweagans | [TK]D-Fender: thanks :) |
| 22:41.59 | cweagans | also, how can I specify a global caller ID to use when calling out? Speakeasy won't let it go out unless it's the DID they assigned us. |
| 22:42.04 | *** join/#asterisk rossand (~aross@dhcp-233-179.tb-classrooms.carleton.ca) |
| 22:42.50 | [TK]D-Fender | cweagans: Set it before you dial <- |
| 22:43.17 | ruben23 | hi |
| 22:43.20 | cweagans | in my extensions.conf? |
| 22:43.59 | [TK]D-Fender | cweagans: Thats where teh Dial is, isn't it? |
| 22:44.13 | cweagans | thanks :) |
| 22:44.13 | cweagans | lol, okay |
| 22:45.17 | ruben23 | hi i have an existing dial plan for my inbound calls, how do i add up to detect caller ID ----> http://pastebin.com/KX64x8dR |
| 22:45.33 | miamiseb | wow, thatd be annoying, not being able to spoof caller id |
| 22:45.39 | ruben23 | i mean to have the number of the caller. |
| 22:45.45 | miamiseb | how are you going to be able to check other peoples voicemail? |
| 22:46.11 | cweagans | miamiseb: I know, right? |
| 22:46.13 | cweagans | SO annoying =P |
| 22:47.04 | *** join/#asterisk doctorray (~ray@static-71-177-137-76.lsanca.fios.verizon.net) |
| 22:47.35 | doctorray | I choose CPE when I am a client to the telephone company, and NET when I am the telephone company, right? Or is it the other way around? |
| 22:48.46 | *** part/#asterisk mbrevda (~mbrevda@unaffiliated/mbrevda) |
| 22:48.49 | miamiseb | from my recollection, net is when you get timing FROM another source, and cpe is when you generate timing |
| 22:49.27 | miamiseb | generally, the side acting as the ITSP will be CPE and the client side will recieve it via NET |
| 22:50.08 | doctorray | ok, so I had it backwards |
| 22:50.09 | [TK]D-Fender | doctorray: Correct |
| 22:50.17 | doctorray | or,, not? |
| 22:50.22 | [TK]D-Fender | [17:47]<doctorray>I choose CPE when I am a client to the telephone company, and NET when I am the telephone company, right? Or is it the other way around? <--- correct |
| 22:50.58 | miamiseb | two questions there |
| 22:50.59 | miamiseb | which ? |
| 22:51.01 | [TK]D-Fender | ruben23: "core show function CALLERID" |
| 22:51.11 | [TK]D-Fender | doctorray: Former |
| 22:51.13 | doctorray | so putting a new sangoma a101 card in, configuring wanpipe, with a PRI ordered from the telco, I will choose PRI CPE mode. |
| 22:51.17 | *** join/#asterisk bjhaid (~herbayjha@41.206.15.3) |
| 22:51.20 | [TK]D-Fender | doctorray: yes |
| 22:51.47 | doctorray | [TK]D-Fender: How is it that you know everything? I remember you answering questions I posed like, two years ago in here. |
| 22:51.56 | doctorray | [TK]D-Fender: and, thank you. |
| 22:52.01 | carrar | [TK]D-Fender is a BOT |
| 22:52.34 | *** part/#asterisk asteriskmonkey (~philip@69.77.169.14) |
| 22:52.54 | [TK]D-Fender | is the bastard child of ED-209 and M5 |
| 22:53.10 | carrar | heh |
| 22:53.19 | carrar | not five alive? |
| 22:53.35 | [TK]D-Fender | carrar: Not Johnny 5 |
| 22:53.40 | carrar | heh |
| 22:53.40 | [TK]D-Fender | carrar: TOS <- |
| 22:54.41 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 22:55.06 | *** join/#asterisk RobH (~robh@216.38.133.254) |
| 23:01.09 | jaskew | [TK]D-Fender: that would be the M5 Multitronic unit, would it not? built by Daystrom IIRC. |
| 23:01.32 | jaskew | It's sad, but I pulled that from memory, with no help from Google |
| 23:02.36 | [TK]D-Fender | jaskew: :D |
| 23:02.45 | [TK]D-Fender | jaskew: Yes, you are correct. |
| 23:02.49 | miamiseb | [TK]D-Fender: stop showing emotion you bot! |
| 23:02.54 | [TK]D-Fender | jaskew: You are indeed quite sad :p |
| 23:03.01 | *** join/#asterisk eharris (eharris@99-179-7-82.lightspeed.austtx.sbcglobal.net) |
| 23:03.09 | jaskew | Funny - it looked a lot like the Beta-5 |
| 23:04.23 | carrar | Never went into production |
| 23:04.29 | carrar | was fatally flawed |
| 23:04.51 | jaskew | which one? Beta-five worked fine |
| 23:05.01 | carrar | M5 |
| 23:05.04 | MuffinMan | [closed] [Asterisk] Core/General 0000005: SIP re-invites failing with certain proxies reported by jtodd https://issues.asterisk.org/view.php?id=5 |
| 23:05.13 | jaskew | http://supervisor194.com/beta5.html |
| 23:05.20 | carrar | I violated Man and God!! |
| 23:05.21 | carrar | It |
| 23:05.35 | jaskew | glad u fixed that |
| 23:05.43 | carrar | heh |
| 23:07.04 | *** join/#asterisk Systemt` (~lol@89-139-110-13.bb.netvision.net.il) |
| 23:11.14 | miamiseb | bah humbug |
| 23:11.21 | *** join/#asterisk puzzled (~foobar@puzzled.xs4all.nl) |
| 23:11.39 | Systemt` | some one can help me please ? |
| 23:12.10 | miamiseb | I see the dtmf begin and dtmf end, but its not being sent to the remote site, rfc2833 and inband aren't working, moving on to SIP info |
| 23:12.24 | miamiseb | Systemt`: ask your question |
| 23:13.27 | Systemt` | i have audio problem ... |
| 23:13.29 | Systemt` | :\ |
| 23:13.58 | miamiseb | Debugging on new channels is disabled |
| 23:14.05 | miamiseb | too bad that doesn't stop existing channels |
| 23:14.27 | Systemt` | sorry |
| 23:14.34 | Systemt` | i dont undestand .. |
| 23:14.43 | miamiseb | Systemt`: it was speaking about my own problem, what is the audio problem you have. |
| 23:14.50 | miamiseb | Pulling teeth anyone? |
| 23:15.16 | Systemt` | when i call \ some1 calls me ther is no |
| 23:15.19 | Systemt` | audio |
| 23:15.23 | Systemt` | from both sidw.. |
| 23:15.26 | Systemt` | e |
| 23:16.01 | miamiseb | is NAT involved? |
| 23:16.32 | Systemt` | i was making port forwarding... |
| 23:17.01 | miamiseb | thats a yes |
| 23:17.37 | miamiseb | you probabably won't know the RTP ports that your call will trying to establish |
| 23:17.50 | Systemt` | nope : |
| 23:17.58 | Systemt` | im newbie |
| 23:18.36 | miamiseb | right, im not saying because of your knowledge level, I'm saying because its not particular ports, they are negotiated during call setup |
| 23:18.43 | miamiseb | Likely, you'd want to read http://www.voip-info.org/wiki/view/NAT+and+VOIP |
| 23:19.41 | Systemt` | <PROTECTED> |
| 23:19.41 | Systemt` | <PROTECTED> |
| 23:19.44 | Systemt` | this ;) ? |
| 23:19.57 | miamiseb | its not reasonable to forward all those ports |
| 23:20.11 | miamiseb | I guess you could, but itd be really wierd, and at that point, why not just 1 to 1 nat. |
| 23:20.31 | miamiseb | also, it would limit your phones to one |
| 23:21.03 | Systemt` | how can i do that ;\ |
| 23:21.16 | miamiseb | you only want one phone to work behind your router? |
| 23:21.35 | Systemt` | nope |
| 23:21.39 | Systemt` | from Xlite |
| 23:21.46 | Systemt` | i have 5 extintions |
| 23:21.50 | dddh | maybe I should buy some hardware? |
| 23:22.01 | miamiseb | right, thats a softphone, but still a phone |
| 23:22.06 | miamiseb | that would be a problem |
| 23:22.15 | Systemt` | whay ? |
| 23:22.17 | miamiseb | the crux of the matter is explained thusly: Conventional VoIP protocols only deal with the signalling of a telephone connection. The audio traffic is handled by another protocol and to make matters worse, the port on which the audio traffic is sent is random. The NAT router may be able to handle the signalling traffic, but it has no way of knowing that the audio traffic is related to the signalling and should hence be passed to the same device the signalling t |
| 23:23.00 | Systemt` | listen |
| 23:23.04 | miamiseb | read |
| 23:23.07 | Systemt` | i was change my router ... |
| 23:23.21 | Systemt` | from Tplink to D-Link |
| 23:23.32 | *** join/#asterisk nickaugust (~anonymous@34.124.188.72.cfl.res.rr.com) |
| 23:23.33 | Systemt` | and then all the problems starts |
| 23:23.41 | [TK]D-Fender | D-Link are known NAT offenders |
| 23:23.55 | [TK]D-Fender | Your router is very possibly to blame |
| 23:23.57 | Systemt` | i have Dlink Dir-400 |
| 23:24.24 | Systemt` | yea i know |
| 23:24.31 | Systemt` | but how can i figger this problem ? |
| 23:24.36 | miamiseb | if it was working before, it probably because your phones were extensions were using stun or some other workaround |
| 23:24.37 | *** join/#asterisk rare1980_ (~rare1980@115.186.23.249) |
| 23:24.44 | Systemt` | ø÷ãíêä |
| 23:24.46 | Systemt` | resolv |
| 23:24.48 | Systemt` | sorry |
| 23:24.51 | rare1980_ | HEY |
| 23:25.01 | Systemt` | yea |
| 23:25.02 | rare1980_ | i have one issue going on |
| 23:25.05 | Systemt` | but with the IAX |
| 23:25.11 | Systemt` | evry thing good |
| 23:25.14 | Systemt` | but on the SIP |
| 23:25.18 | Systemt` | i have problems |
| 23:25.19 | miamiseb | changing your router or setting up a local sip proxy would be my personal choices, but [TK]D-Fender might know more |
| 23:25.33 | rare1980_ | i have virtual machine installed on ubuntu host. i want access host and virtual machine over pptpd vpn .. i can get connected but after i get connected i can only ping host machine but i can't pinf virtual machine.. plz any help |
| 23:25.37 | Systemt` | [TK]D-Fender |
| 23:25.39 | Systemt` | ? |
| 23:25.42 | [TK]D-Fender | Systemt`: Well I also don't see your configs or SIP DEBUG from failed attempts |
| 23:25.49 | [TK]D-Fender | Systemt`: PASTEBIN is your friend |
| 23:25.50 | [TK]D-Fender | ~pb |
| 23:25.51 | infobot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , or apt-get install pastebinit |
| 23:26.58 | miamiseb | rare1980_: are the host and virtual ips on the same subnet? does your vpn connection setup a proper route to reach the virtual machines IPs? If not, can it reach those IPs naturally through its default gateway? |
| 23:27.47 | Systemt` | [TK]D-Fender: can i pm u ? |
| 23:27.51 | rare1980_ | miamiseb: they are all on same subnet. over the vpn i can access host system. |
| 23:28.13 | rare1980_ | but i can't access virtual machine. and this system as also on same subnet |
| 23:28.39 | miamiseb | virtual machine also ubuntu? iptables -L? |
| 23:28.54 | rare1980_ | all is empty |
| 23:29.00 | [TK]D-Fender | <PROTECTED> |
| 23:29.02 | miamiseb | ip_forwarding enabled? |
| 23:29.10 | rare1980_ | how can i check that |
| 23:29.34 | Systemt` | host=82.166.66.43 |
| 23:29.34 | Systemt` | fromuser=0737000331 |
| 23:29.34 | Systemt` | qualify=yes |
| 23:29.34 | Systemt` | allow=g729 |
| 23:29.34 | Systemt` | allow=ulaw |
| 23:29.34 | Systemt` | allow=alaw |
| 23:29.34 | miamiseb | cat /proc/sys/net/ipv4/ip_forward |
| 23:29.35 | Systemt` | allow=g711 |
| 23:29.35 | Systemt` | dtmfmode=rfc2833 |
| 23:29.40 | miamiseb | ahh |
| 23:29.43 | miamiseb | pastebin! |
| 23:29.47 | Systemt` | type=peer |
| 23:29.47 | Systemt` | canreinvite=no |
| 23:29.47 | Systemt` | insecure=port,invite |
| 23:29.47 | Systemt` | nat=never |
| 23:29.47 | Systemt` | context=from-trunk |
| 23:30.02 | *** join/#asterisk xpot-mobile (~xpot@66.60.101.91) |
| 23:31.23 | rare1980_ | it is 0 |
| 23:31.24 | Systemt` | <PROTECTED> |
| 23:31.25 | Systemt` | <PROTECTED> |
| 23:31.25 | Systemt` | <PROTECTED> |
| 23:31.25 | Systemt` | <PROTECTED> |
| 23:31.25 | Systemt` | <PROTECTED> |
| 23:31.25 | rare1980_ | means off |
| 23:31.25 | Systemt` | <PROTECTED> |
| 23:31.25 | Systemt` | <PROTECTED> |
| 23:31.26 | Systemt` | <PROTECTED> |
| 23:31.27 | Systemt` | <PROTECTED> |
| 23:31.27 | Systemt` | <PROTECTED> |
| 23:31.27 | Systemt` | <PROTECTED> |
| 23:31.28 | Systemt` | <PROTECTED> |
| 23:31.28 | Systemt` | <PROTECTED> |
| 23:31.29 | Systemt` | <PROTECTED> |
| 23:31.31 | miamiseb | dude |
| 23:31.34 | miamiseb | death should follow |
| 23:31.36 | miamiseb | lol |
| 23:31.40 | Systemt` | :\ |
| 23:31.45 | rare1980_ | :) |
| 23:31.46 | miamiseb | that means, pastebin it, and then paste the link |
| 23:31.52 | miamiseb | rare1980_ echo 0 |
| 23:31.55 | miamiseb | or more properly |
| 23:32.03 | miamiseb | echo 1 > /proc/sys/net/ipv4/ip_forward |
| 23:32.04 | miamiseb | errm |
| 23:32.05 | miamiseb | echo 1 |
| 23:32.06 | beek | my eyes are bleeding |
| 23:32.13 | rare1980_ | ok |
| 23:32.17 | miamiseb | nods to beek and chastises Systemt` |
| 23:32.32 | Systemt` | http://pastebin.com/kiMXGBD2 |
| 23:33.17 | rare1980_ | miamiseb: still same |
| 23:33.23 | [TK]D-Fender | Systemt`: show a FAILED CALL with SIP DEBUG enabled |
| 23:33.35 | rare1980_ | :( |
| 23:34.23 | rare1980_ | ops kol |
| 23:34.33 | rare1980_ | man it is working now :D |
| 23:34.36 | *** join/#asterisk ltd_wk (~z@sixified.transact.net.au) |
| 23:34.39 | miamiseb | =) glad it worked |
| 23:34.39 | rare1980_ | thanks miamiseb |
| 23:34.52 | miamiseb | np, linux is easier than * for me |
| 23:34.52 | miamiseb | lol |
| 23:35.43 | miamiseb | Systemt` if you are nat'd you might want to set nat=yes and let it try to determine your externalip |
| 23:36.12 | *** join/#asterisk Faithful (~Faithful@202.6.145.116) |
| 23:36.37 | *** join/#asterisk Dibri (~gavit@190.98.33.229) |
| 23:36.55 | miamiseb | you said it was working before though, so before you start messing about, probably a good idea to make a backup of the conf |
| 23:37.53 | Systemt` | http://pastebin.com/yb5pFjcL |
| 23:37.59 | Systemt` | [TK]D-Fender:http://pastebin.com/yb5pFjcL |
| 23:41.17 | rare1980_ | miamised: i have install elastix on virtual machine.. now with ip forwarding i can ping them but still i can't access them through web browser |
| 23:41.40 | Systemt` | open port 443 |
| 23:41.44 | Systemt` | HTTPS |
| 23:41.48 | ruben23 | hi my asterisk crashed twice on my opeartion, how do isolate and check possible cause of it..where should i start..? |
| 23:43.43 | rare1980_ | system: shall i open this port? |
| 23:44.14 | [TK]D-Fender | Systemt`: next time, try getting the ENTIRE call |
| 23:44.19 | Systemt` | yep |
| 23:44.24 | Systemt` | :? |
| 23:44.28 | Systemt` | TK |
| 23:44.36 | [TK]D-Fender | Systemt`: that is only part of it |
| 23:44.37 | Systemt` | i can bring u access to my box... |
| 23:44.44 | Systemt` | ? |
| 23:44.48 | rare1980_ | how can i allow all port access? i don't wana block any port |
| 23:44.57 | Systemt` | humm |
| 23:45.02 | Systemt` | port forwarding... |
| 23:50.01 | [TK]D-Fender | Systemt`: Contact: <sip:0737000331@192.168.0.102> <--- you have NOT set your system up properly to work from behind NAT |
| 23:50.03 | [TK]D-Fender | ~sipnat |
| 23:50.04 | infobot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 23:50.06 | *** join/#asterisk codefreeze-lap (~murf@mail.parsetree.com) |
| 23:50.06 | [TK]D-Fender | ^^^^^^^^^^^ |
| 23:50.35 | miamiseb | Systemt`: specifically, you have nat=no |
| 23:50.48 | Systemt` | sec please |
| 23:50.52 | miamiseb | also, your qualify isn't working, as that method isn't allowed, but I think thats not really important |
| 23:51.17 | rare1980_ | systemt: wht is the command to port forwarding in ubunut?? |
| 23:54.18 | *** join/#asterisk aandrade (~aandrade@189.34.124.123) |
| 23:54.35 | miamiseb | iptables -J forward |
| 23:54.55 | rare1980_ | miamiseb: thanks let me try |
| 23:55.11 | rare1980_ | soo iptables -J forward will forward all ports? |
| 23:55.17 | miamiseb | nopr |
| 23:55.23 | miamiseb | http://www.linuxhomenetworking.com/wiki/index.php/Quick_HOWTO_:_Ch14_:_Linux_Firewalls_Using_iptables#Port_Forwarding_Type_NAT_.28DHCP_DSL.29 for examples |
| 23:55.41 | rare1980_ | and i have to give this command on host system correct? not on virtual machine? |
| 23:55.51 | miamiseb | whichever one is doing that nat'ing |
| 23:57.08 | rare1980_ | but no is dong nating |
| 23:57.11 | rare1980_ | :S |
| 23:57.19 | miamiseb | then why do you need port forwarding? |
| 23:58.41 | rare1980_ | coz i have installed elastix on virtual machine.. i can ping virtual machine IP over the pptp vpn. |
| 23:58.51 | rare1980_ | but i can access it via browser |
| 23:59.32 | miamiseb | I would setup netcat to listen on some other port on the virtual and test via telnet |
| 23:59.50 | miamiseb | if icmp can get through, no reason tcp shouldn't, outside of firewall issues |
