00:03.14 | *** join/#asterisk jks (jks@193.189.93.254) |
00:03.22 | russellb | crochat: that is correct. chan_misdn is only for misdn v1 |
00:04.08 | crochat | russellb: Is chan_lcr as good as chan_misdn ? |
00:05.18 | russellb | i have never heard of chan_lcr |
00:08.16 | *** join/#asterisk ManxPower-work (~EWieling@216.186.151.147) |
00:10.11 | crochat | russellb: www.linux-call-router.de |
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00:42.02 | jaytee | russellb, hahahaa, poodle with tin foil hat and Glenn Beck!! Priceless! |
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00:43.55 | russellb | :-) |
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00:54.31 | renvorak | Hi all- I have spent the majority of a couple days trying to solve a problem and am hoping someone here can help. I installed AsteriskNow 1.5 and asterisk-addons about six months ago was able to get MySQL working, but not the uniqueid field which is required to link the files to play in ARI. I understand that I need to pass some variables when recompiling asterisk-addons-1.4.10, but for the life of me, I cannot get it to see asterisk.h despite using |
00:59.20 | Katty | HELLO SUNSHINES |
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01:01.22 | jaytee | hi Katty |
01:01.27 | Katty | hugs jaytee |
01:01.38 | jaytee | thanks, I need that |
01:01.47 | Katty | you alright? (= |
01:01.53 | jaytee | nope |
01:01.57 | Katty | :< |
01:02.02 | Katty | hugs jaytee for good measure. |
01:02.04 | Katty | whatsthematter? |
01:02.15 | jaytee | had an EEG yesterday and have to have an EKG next friday |
01:02.36 | Katty | i'm not sure i know what those are for. |
01:03.01 | jaytee | electro-encephalograph and echo-kardio-graph |
01:03.16 | jaytee | the latter is like an ultrasound of the heart |
01:03.36 | Katty | do you think something is wrong with your heart? |
01:03.36 | jaytee | and the former is just a graphed output of heart rhythms |
01:03.58 | jaytee | the EEG had an irregular rhythm |
01:04.05 | Katty | nods |
01:04.11 | Katty | not a bad idea to get it checked out. |
01:04.25 | Katty | i hope it's nothing, but i'll keep you my thoughts just the same (= |
01:04.28 | jaytee | I've been having dizzy spells every 10 to 15 minutes since wednesday night |
01:04.40 | jaytee | so I went to the doctor friday |
01:05.04 | *** part/#asterisk ruben23 (~AGENT@122.55.48.243) |
01:05.08 | Katty | nods |
01:05.28 | Katty | how is your blood pressure? |
01:05.46 | renvorak | --First time IRC user. Can anyone see me or do I have to select all on the right hand side? thanks. |
01:06.02 | jaytee | my BP was normal when they checked it yesterday |
01:06.13 | jaytee | renvorak, I see you! |
01:06.14 | Katty | renvorak: welcome to a long-time tradition of ircing (= |
01:06.20 | Katty | renvorak: we can see you, just keep on typing! |
01:06.29 | Katty | jaytee: good to hear you BP is okay. |
01:06.43 | Katty | jaytee: are EKGs very specific at all? |
01:06.43 | renvorak | Thanks! |
01:06.56 | jaytee | yw |
01:08.05 | jaytee | Katty, the one they're going to give me will include an amphetamine type drug that stresses the heart so they can see how it's functioning normally and when under stress |
01:08.07 | Katty | jaytee: or is just kind of a generalized test |
01:08.41 | jaytee | EKGs are very specific |
01:10.22 | Katty | amphetamines scare me :< |
01:10.45 | jaytee | same here |
01:10.58 | Katty | that stuff is also addictive. |
01:11.01 | Katty | so be careful |
01:11.31 | Katty | i know it's for a good cause. |
01:11.51 | Katty | just af ew months ago i had a series of dizzy spells. |
01:11.58 | Katty | but it turned out to be a severe sinus infection |
01:12.07 | jaytee | well, I doubt if I'll become addicted from one dose and plus I don't know anyone who deals drugs so I'm fairly safe |
01:12.26 | Katty | nods |
01:12.33 | Katty | one dose is still enough to do something. |
01:13.02 | Katty | keep me posted (= |
01:13.05 | Katty | i want to know everything |
01:13.21 | jaytee | I'm more concerned with the drug giving me a heart attack than getting addicted |
01:13.42 | Katty | i don't think it'll give you a heart attack |
01:13.46 | Katty | the doctors wouldn't take a risk of that |
01:13.54 | jaytee | since I haven't been exercising much I don't think the old ticker can take much stress |
01:14.00 | Katty | they'd give you a low dosage just to see how your heart responds. |
01:14.24 | jaytee | yeah, and being in a hospital that's rated very high up there in cardiac care helps too |
01:14.32 | Katty | most definately |
01:14.35 | Katty | they know what they are doing |
01:14.57 | Katty | they'll have you right as rain in no time (= |
01:15.26 | jaytee | that last line made me think of The Matrix |
01:17.08 | jaytee | yay, he's quit! |
01:21.22 | jaytee | I love the new fan thing on Facebook: Can this poodle wearing a tinfoil hat get more fans than Glenn Beck? |
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01:52.30 | carrar | Whats facebook |
01:54.40 | carrar | Sounds like some place where you can buy books about cosmetics |
01:55.08 | carrar | or as about as usefull! |
01:59.16 | *** join/#asterisk rue_house (~rue@24.207.119.38) |
01:59.22 | rue_house | sdf |
01:59.38 | rue_house | is there a driver for doing ulaw over a serial port? |
02:05.28 | Katty | carrar: :P |
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02:06.09 | Katty | the dogged help me put away a bag of popcorn. so good of him to save me some calories. |
02:08.30 | rue_house | mmm popcorn, you just worked out my supper problem |
02:08.43 | rue_house | I want a serial port to do ulaw for me |
02:13.20 | renvorak | anyone had experience compliling asterisk-addons in a asterisknow 1.5 environment? craps out when it can't find asterisk.h.... |
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02:45.48 | ManxPower-work | ~asterisknow |
02:45.48 | infobot | somebody said asterisknow was based on Asterisk, but is difficult to support in #asterisk for a number of reasons. Please seek support in #asterisknow instead. |
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03:13.41 | Katty | VS just sent me a catalog. |
03:13.48 | Katty | i wonder if someone's trying to tell me something |
03:14.25 | ChannelZ | you're not friends with John Kerry, are you? |
03:14.34 | ChannelZ | damnit! Edwards. I blew that one |
03:15.41 | Katty | baroo? ^_- |
03:15.47 | Katty | i think that went over my head. |
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03:16.51 | ChannelZ | John Edwards. Adulteror |
03:16.55 | ChannelZ | Maybe I should have said Tiger Woods. |
03:17.06 | ChannelZ | (You're white, right?) |
03:17.19 | Katty | no, i'm orange. |
03:17.53 | smooth_penguin | Im black and white |
03:18.07 | Katty | ^- Zebra |
03:18.23 | renvorak | @infobot: thanks for the quick reply. I have posted it in asterisknow as well, but was hoping to see if anyone else had experience as the problem is more with asterisk-addons... Not to mention #asterisk has 20x the users. Thanks again. |
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03:19.19 | smooth_penguin | no Katty - http://www.ccpo.odu.edu/Research/globec/main_cruises02/lmg0205/pod/gentoo.jpg |
03:20.03 | Katty | you missed it :< |
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03:25.27 | Katty | http://www.natureandwildlife.it/foto/antarctica/11-Antarctica-Emperor-Penguins-Family.JPG <- smooth_penguin came from a happy home. |
03:25.51 | smooth_penguin | :> |
03:29.01 | Katty | bored. |
03:29.47 | smooth_penguin | is going back to bed |
03:30.09 | Katty | ninite |
03:33.51 | smooth_penguin | Katty, hows the critters |
03:35.56 | Katty | well, the ferrets are fine. |
03:36.01 | Katty | riddick had to go to the vet today. |
03:36.07 | Katty | but he is okay |
03:37.09 | smooth_penguin | kk |
03:38.00 | Katty | ear infection :< |
03:38.26 | smooth_penguin | ugh, how did he get that |
03:38.34 | Katty | she thinks he got water is his ear. |
03:38.57 | smooth_penguin | :s |
03:40.24 | smooth_penguin | bye Katty, still need to finish up sleep :s |
03:51.53 | rue_house | anyone know how I could make a chan_rs232 driver? |
03:53.53 | ChannelZ | to what end? But look at the source for one of the channel drivers and have at it.. |
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03:57.03 | Katty | returns with peanutbutter sammich |
03:57.24 | Katty | Dear Universe, thank you for peanut butter and it's amazingness. Love, Katty |
03:57.35 | Katty | its. |
03:57.40 | Katty | it's |
03:57.44 | Katty | its' |
03:57.46 | Katty | one of those. |
04:02.50 | carrar | and |
04:03.00 | carrar | for peanut butter & chocolate ice cream |
04:03.30 | Katty | yesh :>>> |
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04:26.52 | jmcdowell | So, I am looking for a GOOD asterisk from scratch howto |
04:26.57 | jmcdowell | does anyone have any suggestions ? |
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04:47.23 | *** join/#asterisk jcarter1 (~bla@c-68-48-2-59.hsd1.md.comcast.net) |
04:48.39 | jcarter1 | hello - i need some help with setting up a connection from x-lite on my windows machine to the asterisk server running on the same laptop on ubuntu installed on virtual box |
04:48.57 | jcarter1 | i dont kno which ip adress to use to register with the client |
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05:15.03 | jcarter1 | anyone? |
05:16.02 | ChannelZ | ask virtualbox |
05:16.33 | ChannelZ | There's a whole bunch of modes it runs in for network access |
05:16.48 | jcarter1 | oh |
05:17.01 | jcarter1 | i will give it a try |
05:17.33 | ChannelZ | in NAT mode I have no idea how you get connections into the virtual machine |
05:19.01 | jcarter1 | for what i want to do, it is already enough if my client inside the vm can connect to the server - but virtual box seems to have big trouble with microphone detection, so i cant use my microphone. Now i installed a client in windows, but got no idea how to access the server -_- |
05:22.01 | ChannelZ | I think you have to use Bridging mode in Virtual Box to make a fake interface on the host machine that you can use to connect to the virtual one |
05:24.38 | jcarter1 | you are right |
05:29.36 | ChannelZ | NAT mode (which is is probably default) is only really useful for outgoing traffic out of the virtual machine |
05:30.34 | jcarter1 | #vbox helped me |
05:30.36 | jcarter1 | it works now |
05:30.44 | jcarter1 | i had to change to bridging mode as you said |
05:31.45 | pentanol | hey, anybody use auth over radius? |
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07:50.24 | Whtsup | hello |
07:50.26 | Whtsup | i want to ask |
07:50.41 | Whtsup | how can i add mutiple ip address of host in sip.conf |
07:52.02 | tzafrir_laptop | Whtsup, no |
07:52.12 | tzafrir_laptop | Why do you need that? |
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08:07.09 | Whtsup | hm |
08:07.18 | Whtsup | for a2billing |
08:07.30 | Whtsup | my customer told me add multiple ip |
08:07.33 | Whtsup | how can i add |
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08:10.30 | carrar | SUM |
08:10.31 | carrar | + |
08:10.41 | Whtsup | hmmmmmm |
08:10.45 | carrar | A + C = C |
08:10.52 | Whtsup | ? |
08:10.57 | carrar | A + C = B |
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08:26.02 | tzafrir_laptop | Whtsup, sure. an IP address is merely a 32bit number. |
08:27.22 | tzafrir_laptop | Whtsup, seriously, though, what do you want to do with those multiple IP addresses? |
08:27.41 | tzafrir_laptop | Use a random one? |
08:27.54 | tzafrir_laptop | Use one as a default and another one as a backup? |
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08:56.19 | Godfather_ | o/ |
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11:03.02 | scurb | Anyone know if MixMonitor command is notified in the manager API events? |
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12:12.47 | hyphenex | What's a good VoIP phone I should get? :P |
12:13.11 | benngard | ~phones |
12:13.11 | infobot | While personal preference will dictate which phone works best for you, general consensus on a rough order of quality and suggestibility is as follows: Polycom (any), Aastra 480i, Aastra 5i Series, Linksys SPA-9XX, Snom, Cisco 79XX, everything else. Do not consider Grandstream phones. Ever. |
12:13.47 | hyphenex | Thanks |
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12:41.49 | couponjuan | anyone want to work on a project together? I do html css php mysql and the flash/photoshop. I also have a little bit of a vision too of what I need this for. |
12:42.20 | aceking5 | uh wrong channel? |
12:42.23 | couponjuan | no |
12:42.30 | couponjuan | dynamic .call file |
12:42.50 | couponjuan | can be created and put in the outbound directory right? |
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13:01.57 | hyphenex | hey, what do you think of this one? http://www.ht.com.au/part/V2913-Cisco-Small-Business-Pro-SPA942-4-line-IP-Phone-with-2-port-Switch-VoIP-phone-SIP-v2/detail.hts |
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13:30.41 | puzzled | hyphenex: imo cisco sip firmware sucks, you even need to pay to get new firmware and now that linksys == cisco it's probably only time before they become just as bad |
13:31.07 | puzzled | get a polycom, aastra or snom |
13:40.46 | jmcdowell | Polycom has outstanding quality, but their configs are a litte hard to figure out at first due to 1rst grade documentation. Once you get it, it's easy.. |
13:42.28 | *** join/#asterisk neurosys (~neurosys@c-71-196-20-208.hsd1.fl.comcast.net) |
13:43.07 | jmcdowell | has installed CentOS 5 and asterisk 1.6 and is currently working out the kinks. |
13:43.20 | jmcdowell | still can't dial |
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13:50.27 | tzafrir_laptop | couponjuan, if you ask for "a dynamic call file", you probably want to use "Originate" from the manager interface |
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14:14.49 | krefik | anyone know is asterisk-gui can substitute administrative tools like freepbx? |
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14:18.13 | mike3 | I just setup Asterisknow. I'm having some issues where the voice prompts are a little glitchy sounding. How can I fix this? |
14:19.51 | benngard | Message: Thanks for all the fish <- fun logoff message from ami :) |
14:20.40 | n3glv | qos? |
14:21.14 | mike3 | I'm calling on the internal network |
14:21.19 | mike3 | not much activity at al |
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15:28.22 | niekvlessert | hello, i'm trying to get chan_misdn working |
15:28.34 | niekvlessert | anyone with experience on debian etch with that? |
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15:41.17 | puzzled | niekvlessert: nope but if you are not too hung up on debian, just install centos 5.4 on the box/vm and follow http://www.asterisk.org/downloads/yum to get everything including mISDN |
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15:53.59 | Igramul | Hi, is it possible to get the dialed number in a h-extension? (e.g. to write to a fax report the number the fax was sent to) |
15:57.33 | n3glv | yes |
15:57.38 | n3glv | if you had not left |
15:57.39 | n3glv | ;-> |
15:58.24 | *** join/#asterisk hluesea (~hulusikah@88.247.127.66) |
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16:06.09 | *** join/#asterisk niekvlessert (~niek@5ED25657.cable.ziggo.nl) |
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16:20.18 | mike3 | I just setup Asterisknow. I'm having some issues with audio. It seems to be glitchy. Any input? |
16:24.10 | niekvlessert | type of phone/connection/codecs plz |
16:25.19 | *** join/#asterisk danj1980 (~dan@91.109.81.119) |
16:27.57 | danj1980 | Hi. I just had 7 hours of downtime on asterisk because the server power supply failed and needed replacing. I've been googling "asterisk redundancy" and have found loads of questions, but not many answered. |
16:28.41 | danj1980 | Has anyone see a whitepaper on creating a more stable environment? |
16:28.47 | n3glv | danj1980: I think you want high-availability and that's a hard row to hoe |
16:29.12 | n3glv | dual powersupply and raid1 or better is easy |
16:29.14 | *** join/#asterisk CoderForLife (~Miranda@cpe-174-101-155-51.cinci.res.rr.com) |
16:29.19 | n3glv | mirrored servers etc is a mess |
16:29.28 | *** part/#asterisk n3glv (~n3glv@pool-71-182-249-171.pitbpa.east.verizon.net) |
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16:29.58 | danj1980 | The polycom phones we use can register to two servers, so that makes my job a bit easier. |
16:30.27 | *** join/#asterisk xmitter (~xmitter@c-24-21-213-242.hsd1.or.comcast.net) |
16:30.47 | danj1980 | Is it then simply a case of setting up a second server, syncing the config files and mysql? |
16:31.43 | danj1980 | However, internal calls get messy because if extension 101 is on server A, and extension 102 has decided to choose server B, then they cant find each other. |
16:33.21 | danj1980 | Alternatively, we can put Openser in front of both servers, but then if openser went down, it would be the same problem. |
16:34.01 | *** join/#asterisk puzzled_ (~foobar@puzzled.xs4all.nl) |
16:35.01 | danj1980 | By the way, its a purely VOIP based platform. No analog or ISDN lines. |
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17:01.26 | mike3 | niekvlessert, sorry |
17:01.29 | mike3 | um.. let me look |
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17:46.56 | ManxPower-work | ~answers |
17:46.57 | infobot | hmm... 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 |
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18:18.28 | *** join/#asterisk Artur__ (~chatzilla@212.59.229.155) |
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18:19.13 | Artur__ | hi everyone and sorry for my poor english |
18:19.57 | Artur__ | I would like to set my basic dialplan but i have some problem with it . Can anyone help me? |
18:21.11 | fenrus | what do you want your dialplan to do ? |
18:21.17 | *** join/#asterisk xmitter (~xmitter@c-24-21-213-242.hsd1.or.comcast.net) |
18:21.20 | fenrus | have you read any examples? |
18:21.42 | fenrus | you need to be a bit more specific |
18:21.57 | Artur__ | yes i read and some steps from book of asterisk but i stil have litle problem whit it |
18:22.15 | Artur__ | ok can you wait i put hire some of my config |
18:24.09 | Artur__ | [ipfon] |
18:24.10 | Artur__ | exten => 327218816,1,Dial(SIP/artur&SIP/test) |
18:24.12 | Artur__ | [sipdiscount] |
18:24.14 | Artur__ | exten => 0048327218816,1,Dial (SIP/artur&SIP/test) |
18:24.15 | Artur__ | [incoming] |
18:24.17 | Artur__ | exten => 327218816,1,Dial(SIP/artur,20) |
18:24.18 | Artur__ | exten => 327218816,1,Answer() |
18:24.20 | Artur__ | exten => 327218816,n,Background(main-menu) |
18:24.21 | Artur__ | [test] |
18:24.23 | Artur__ | exten => 101,1,Dial(SIP/artur,20) |
18:24.24 | Artur__ | exten => 101,n,Playback(boston) |
18:24.26 | Artur__ | ;exten => 102,1,Dial(SIP/test,20,m) |
18:24.28 | Artur__ | ;exten => 102,n,Playback(boston) |
18:24.29 | Artur__ | exten => 102,1,Answer() |
18:24.30 | Artur__ | exten => 102,n,Background(main-menu) |
18:24.32 | Artur__ | exten => 102,n,WaitExten() |
18:24.32 | *** mode/#asterisk [+o [TK]D-Fender] by ChanServ |
18:24.33 | neurosys | uh oh |
18:24.33 | Artur__ | exten => 1,1,Playback(digits/1) |
18:24.35 | Artur__ | exten => 2,1,Playback(digits/2) |
18:24.37 | Artur__ | ;exten => _XXXXXXXXX,1,Dial(SIP/${EXTEN}@sipdiscount,10) |
18:24.37 | *** kick/#asterisk [Artur__!~chatzilla@64.235.218.194] by [TK]D-Fender (Artur__) |
18:24.55 | x86 | lol |
18:25.00 | trelane | oops |
18:25.04 | *** join/#asterisk Artur__ (~chatzilla@212.59.229.155) |
18:25.04 | x86 | [TK]D-Fender++ |
18:25.13 | trelane | Artur__, www.pastebin.com |
18:25.24 | [TK]D-Fender | Artur__: Do NOT flood in here again |
18:25.26 | x86 | pastebin.ca > pastebin.com |
18:25.27 | [TK]D-Fender | ~pb |
18:25.28 | 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 |
18:25.29 | [TK]D-Fender | ^^^ |
18:25.52 | x86 | pastebinit? ooo |
18:26.30 | Artur__ | i past config to one of site |
18:26.46 | [TK]D-Fender | Artur__: WHERE'S THE LINK TO IT? |
18:26.57 | [TK]D-Fender | Artur__: Are we supposed to GUESS? |
18:27.35 | Artur__ | http://paste.lisp.org/+219C. |
18:27.36 | trelane | [TK]D-Fender, of course! |
18:27.45 | Artur__ | this is my exten conf |
18:27.52 | Artur__ | extension.conf |
18:28.18 | Artur__ | i would like to set incoing and outgoing first incoming for ipfon |
18:28.31 | neurosys | well his [incoming] context has 2 "1" priorities... |
18:28.38 | Artur__ | i can change test to incoming |
18:28.38 | smooth_penguin | trelane ! |
18:28.45 | trelane | smooth_penguin, I deny everything :) |
18:28.53 | smooth_penguin | heh ;p |
18:29.13 | Artur__ | i know I'm noob with asterisk |
18:29.25 | Artur__ | ok i told what i need |
18:29.55 | smooth_penguin | Artur__, well its basic IRC ethics |
18:30.00 | smooth_penguin | use Pastebin |
18:30.15 | Artur__ | i want to set exten to transfer incoming call to my mobile phone when i not answer the local phone number afret 20 sec |
18:30.53 | Artur__ | pestbin is basic asterisk Qest..? |
18:30.58 | Artur__ | channel |
18:31.14 | Artur__ | ? |
18:31.53 | Artur__ | sorry |
18:32.22 | Artur__ | paste.lips is worst then pastebin? |
18:32.43 | Artur__ | i paste my config to paste.lips and put url |
18:34.13 | smooth_penguin | yah |
18:34.28 | *** join/#asterisk fibres (~no@cpc2-nfds1-0-0-cust1021.lei3.cable.ntl.com) |
18:35.11 | Artur__ | i know there is litle mess some ext I coment becouse i test this |
18:35.22 | *** join/#asterisk rue_house (~rue@24.207.119.38) |
18:35.33 | rue_house | is the D channel part of a T1's 24 channels? |
18:36.00 | x86 | only on a T1/PRI |
18:36.24 | rue_house | you remember how many channels are in an E1? |
18:36.29 | rue_house | not 32 is it? |
18:36.30 | x86 | a T1/CAS has (24) 64kbps channels, none of them B or D channels, but you can think of them all as B channels |
18:36.38 | Artur__ | anyone looking for my extension.conf? |
18:36.40 | x86 | 32 channels E1 yes |
18:36.49 | rue_house | huh |
18:37.03 | x86 | 30B 2D usually on an E1 PRI, iirc |
18:37.20 | rue_house | the ulaw converter chips use a 2.048Mhz clock rate and recieve 8 bits for every 256 clock cycles = 32 channels |
18:37.39 | [TK]D-Fender | rue_house: thats E1, not T1 |
18:37.44 | rue_house | yea |
18:37.46 | Artur__ | can anyone help me in private chat? |
18:37.57 | rue_house | this is just the audio codec chip |
18:38.33 | rue_house | its REALLY easy to make a T1->analog channelbank |
18:38.35 | rue_house | you just need to do frame clocking |
18:38.53 | rue_house | feed the chips the raw t1 stream otherwise |
18:39.11 | rue_house | (chips = MC14LC5480) |
18:39.17 | smooth_penguin | Artur__, whats currently not working? |
18:40.05 | Artur__ | if someone cal to me and i no't answer call should transfer to my mobile phone |
18:40.15 | Artur__ | but nothing hepen |
18:40.18 | rue_house | swear its almost too easy just with discrete logic chips |
18:40.31 | rue_house | so why are they so expensive... |
18:41.03 | niekvlessert | hey guys, it seems there's a xorcom sources.list |
18:41.09 | niekvlessert | can anyone provide it? |
18:41.21 | tzafrir_laptop | I guess I can. |
18:41.32 | tzafrir_laptop | But what do you mean by "xorcom" |
18:41.45 | coppice | rue_house: you have identified your route to unknown riches |
18:42.05 | rue_house | its a 15 year old path! |
18:42.17 | niekvlessert | well i heard there's a way to install a debian and then add some xorcom stuff to your sources.list and then install asterisk with all the crazy isdn stuff in 1 run |
18:42.18 | rue_house | this stuff has been around forever... |
18:42.31 | coppice | ruw_house: so you are rich? |
18:42.40 | rue_house | no |
18:42.46 | Artur__ | smooth i don't understend how to write right exten to make it what should dialplan d when i don't answer after about 10 to 20 sec |
18:43.02 | rue_house | these old chips sure make telephony interface easy |
18:43.11 | tzafrir_laptop | niekvlessert, stuff under http://updates.xorcom.com/rapid and http://updates.xorcom.com/pkg-voip is basically built off the Debian pkg-voip repo |
18:43.29 | rue_house | I want to make a chan_rs232, Im pretty sure I could have an avr bridge it to the codec chip |
18:43.40 | tzafrir_laptop | the pkg-voip one has quite untested packages. The /rapid one has only signed and tested packages |
18:44.08 | rue_house | dosn't mean I will |
18:44.28 | tzafrir_laptop | niekvlessert, Originally there was bristuff in Debian. nowadays (1.6.2) it's gone |
18:44.53 | rue_house | any easy falls apart trying to do voip (cause of echo cancelation) |
18:44.59 | niekvlessert | tzarfrir_laptop: ok, nice, so this is 1.6? |
18:45.04 | tzafrir_laptop | There's only a minor "bri" patch (to replace the "sux" message with something marginally more useful) |
18:45.36 | niekvlessert | yes it is |
18:45.40 | tzafrir_laptop | niekvlessert, yes, 1.6.2 |
18:45.51 | niekvlessert | hmmmm |
18:46.09 | Artur__ | smooth can you help me ? |
18:47.05 | rue_house | where would I start to write a rs232 channel driver? |
18:47.57 | rue_house | ulaw only, raw to serial port |
18:48.20 | niekvlessert | why is it so hard to get chan_misn running?!? |
18:48.25 | niekvlessert | chan_misdn |
18:48.38 | tzafrir_laptop | misdn? it seems quite different |
18:48.49 | tzafrir_laptop | rue_house, how will the signalling be handled? |
18:49.03 | smooth_penguin | Artur__, Im not very sure |
18:49.04 | rue_house | it wont! :) |
18:49.12 | tzafrir_laptop | How do you dial? hang up? |
18:49.14 | rue_house | your always listening and alwasy transmisting |
18:49.22 | rue_house | call it rev 1 :) |
18:49.31 | *** join/#asterisk mykhyggz (~col@evolone.org) |
18:49.34 | rue_house | inband dtmf |
18:49.36 | rue_house | :) |
18:49.38 | Artur__ | smooth heh |
18:49.44 | tzafrir_laptop | rue_house, then look at the "console" channels, I guess |
18:49.50 | rue_house | ok |
18:49.52 | niekvlessert | tzafrir_laptop: trying to get it running with 1.4.29, i can compile misdn but asterisk won't find it so i can't compile chan_misdn |
18:50.23 | tzafrir_laptop | niekvlessert, what is "it"? |
18:50.51 | niekvlessert | tzafrir_laptop: not sure yes, suppserv for sure though |
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18:50.56 | niekvlessert | although i did compile it |
18:50.59 | rue_house | kirby dance!? |
18:51.01 | rue_house | sigh... |
18:51.27 | Artur__ | :( |
18:51.41 | niekvlessert | it can't find misdn either! |
18:51.49 | smooth_penguin | Artur__, well look up on the voip-info website |
18:52.11 | rue_house | tzafrir_laptop, would you say the console is the must cutdown driver there is? |
18:52.14 | niekvlessert | i change menuselect-tree to see what it can & can't find |
18:52.37 | niekvlessert | so now suppserv & misdn is a problem |
18:52.40 | tzafrir_laptop | rue_house, in-band signalling is not nice |
18:52.41 | niekvlessert | nice |
18:52.45 | smooth_penguin | Artur__, http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial |
18:52.51 | tzafrir_laptop | rue_house, are you sure you want to go there? |
18:53.13 | rue_house | I'm sure I can ignore that bit for hte process of having fun |
18:53.37 | Artur__ | thx i read this now |
18:53.42 | rue_house | I had this interesting idea recently, creating parity errors on bytes to do serial sync |
18:53.54 | rue_house | aka, bytes with a parity error are control |
18:53.57 | rue_house | :) |
18:54.35 | rue_house | I could make every 1001'st byte control |
18:54.54 | tzafrir_laptop | rue_house, what speed is your serial channel? |
18:55.18 | rue_house | well, 8Khz 8bit, 64Kbits... so thats.... higher than 2400... |
18:55.47 | tzafrir_laptop | if it's fast enough (115200) you can use some "extra bandwidth" for signalling |
18:55.49 | rue_house | 4800...9600... 19200.... 38400... |
18:56.00 | tzafrir_laptop | (hmm... sounds like TDM) |
18:56.16 | rue_house | the chip takes in ulaw |
18:56.41 | rue_house | hmm have to be 115K too |
18:57.04 | tzafrir_laptop | 1 B channel and 1 D channel :-) |
18:57.20 | rue_house | or slightly less |
18:57.32 | rue_house | 1/2 rate D channel |
18:59.38 | rue_house | actaully, this is alot like th problem on the hexapod I'm building, muxing control data |
19:00.42 | rue_house | console uses the sound card right? its not audioless? |
19:01.31 | rue_house | ARNING, "Failed to initialize audio system - (%d) %s\n" |
19:01.33 | rue_house | ah good good |
19:09.15 | *** join/#asterisk cuco (~Diego@95.35.58.91) |
19:09.47 | fibres | Can anyone help me? I am having authentication issues with incoming calls on Ast 1.6 I tested the config im using on 1.2 and works ok on 1.6 authentication fails |
19:20.14 | *** join/#asterisk simplydrew_ (~simplydre@pool-74-97-190-109.prvdri.fios.verizon.net) |
19:24.14 | niekvlessert | ./configure can't find mISDN, checked install.log, /usr/bin/ld: cannot find -lmISDN |
19:24.21 | niekvlessert | what does that mean? |
19:27.39 | *** part/#asterisk dezent (dezent@84.243.58.35) |
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19:38.08 | *** join/#asterisk Bullterd (~IceChat7@62.249.223.152) |
19:38.11 | Bullterd | Evening All :) |
19:38.21 | Bullterd | Can anyone advise me on what software to use to setup an outbound proxy? |
19:45.58 | *** join/#asterisk z53102 (~z53102@195.5.125.10) |
19:53.22 | niekvlessert | define proxy |
20:01.03 | *** join/#asterisk Da-Geek (~Da-Geek@gateway.rhev.dageek.co.uk) |
20:01.12 | ManxPower-work | Bullterd: A proxy or a B2BUA? |
20:02.16 | Bullterd | ManxPower-work: A whatnow? |
20:02.50 | ManxPower-work | Bullterd: OpenSER is a SIP Proxy. Asterisk is a SIP Back2BackUserAgent |
20:02.50 | *** join/#asterisk Failrar (~Failrar@5ED66E6D.cable.ziggo.nl) |
20:03.02 | Bullterd | Ah. |
20:03.20 | ManxPower-work | Which means Asterisk basically treats both legs of the calls as different calls and so does not pass SIP stuff directly between the two ends of the call. |
20:03.37 | Bullterd | ManxPower-work: All I know is this solved a big problem for me http://free.sipout.com/ |
20:03.48 | Bullterd | ManxPower-work: its a free service, I dont want to rely on that for buisness, so want to set my own up |
20:04.37 | ManxPower-work | Bullterd: Asterisk is not suitable for that application. |
20:04.38 | [TK]D-Fender | Bullterd: "a big problem"? You haven't describe what problem you even HAVE. |
20:04.54 | p3nguin | bullterd: Take a look at OpenSIPS and see if you think it can help you. |
20:04.58 | ManxPower-work | [TK]D-Fender: ISPs blocking 5060/UDP |
20:05.12 | Bullterd | ManxPower-work: No, Just a general pain with my customers and NAT |
20:06.11 | ManxPower-work | Bullterd: Then your question makes no sense. |
20:06.14 | Bullterd | I wasnt specifically looking for an asterisk related solution, Just a general linux one. I guessed that since it was SIP related thing, here'd be the best place to ask |
20:06.21 | ManxPower-work | Virtually all VoIP PBXs support NA |
20:06.31 | ManxPower-work | T'd clients without problem |
20:06.48 | Bullterd | Thanks for your help |
20:08.07 | niekvlessert | http://pastebin.com/d44a3cdb |
20:08.19 | niekvlessert | misdn trouble still going on |
20:08.29 | niekvlessert | installed 3rd party stuff from svn.digium.com |
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21:16.02 | *** join/#asterisk doug (doug@breakout.telerama.com) |
21:16.21 | doug | hulo! anyone have a nice way of getting text entry from a phonepad? |
21:16.28 | doug | or does that already exist in asterisk? |
21:16.41 | doug | i want to capture the text name of my destinations... |
21:16.55 | doug | in my system, if i'm calling target, it'll say "I don't know this destination, please say the name" |
21:16.57 | *** join/#asterisk russellb_ (~russellb_@asterisk/digium-open-source-team-lead/russellb) |
21:16.57 | *** mode/#asterisk [+o russellb_] by ChanServ |
21:17.01 | doug | then it'll record me saying "TARGET" |
21:17.12 | doug | afterwards, i want it to say "spell the name of the destination you just spoke" |
21:17.17 | doug | and have it take a spelling via dialpad |
21:17.42 | doug | dunno what mechanism, maybe 21 for A, 22 for B, 23 for C, 31 for D etc |
21:18.09 | doug | but somehow have it capture the spelling in a machine-readable way |
21:18.15 | doug | so it can be automatically added to the directory |
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21:35.03 | ManxPower-work | doug: "core show applications" |
21:44.57 | doug | sure; know which one might do me? |
21:45.25 | doug | or if there's an app that's not normally compiled in...? |
21:46.55 | carrar | So you are asking for speech recognition |
21:47.14 | carrar | look at lumenvox |
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21:56.41 | doug | hm, not speech recognition |
21:56.44 | doug | this would be from the keypad |
21:57.26 | doug | so i'd say the name for the recording, and punch in some numbers for the spelling. |
21:59.43 | ManxPower-work | in 1.6.mumble there should be an application that allows 222=C type of entry |
22:00.16 | doug | know the name? |
22:00.25 | doug | <PROTECTED> |
22:00.42 | doug | if not the name, something i can feed into google to get close... |
22:02.05 | ManxPower-work | doug: I guess I could go track down a 1.6 box and find out for you, but I'm not going to. That sort of stuff is usually left as an exercise to the reader. |
22:03.29 | carrar | You mean people can help themselves? |
22:03.33 | carrar | Thats just crazy talk! |
22:04.41 | doug | i got one, and i just went through the apps |
22:04.46 | doug | which is why i ask. |
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22:08.17 | doug | although now i'm wondering if the sms app is also in 1.4.28 |
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22:09.14 | doug | looks like it. |
22:09.32 | doug | no "alphabet mode" dtmf app tho, or anything close to it. |
22:10.18 | doug | "Works to ETSI ES 201 912 compatible with BT SMS PSTN service in UK." i wonder if that implies no us easy smsing. |
22:11.02 | ManxPower-work | The SMS app only works with European carriers. |
22:11.51 | doug | i'd give my left nut to send/receive sms via asterisk to us phones. |
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22:13.05 | doug | curse the yanks and their monopolies. |
22:13.16 | doug | or duopoloies, i guess. |
22:13.23 | doug | -o |
22:13.33 | ManxPower-work | I was wrong. http://www.voip-info.org/wiki/view/Asterisk+cmd+DTMFToText |
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22:14.48 | carrar | doug, you could setup a SMS gateway |
22:15.16 | ManxPower-work | carrar: Somehow I think that's a tad complex for his needs. |
22:15.23 | carrar | heh |
22:15.50 | ManxPower-work | He would either always need Asterisk at both ends of the call or he would have to import SMS analog sets from Europe. |
22:16.17 | doug | sms gateway; is that something with an ss7 interface? |
22:16.21 | ManxPower-work | Asterisk's SMS uses MODEM TONES to transport the text. |
22:16.32 | carrar | Today youcan however send SMS Mesasges via Email to any phone via asterisk though |
22:16.40 | ManxPower-work | i.e PSTN SMS, not cellular SMS |
22:17.08 | doug | modem tones? that's some kind of asterisk-only texting hack? |
22:17.35 | ManxPower-work | doug: not at all, it's the SMS standard for SMS over PSTN |
22:17.58 | doug | wow, that's twisted. is that the meat of the ETSI ES 201 912 protocol? |
22:18.08 | ManxPower-work | I'm sure you've never heard of it because nobody uses it in the USA. |
22:18.20 | ManxPower-work | doug: I'd have to look it up, but that sounds correct. |
22:18.34 | doug | pstn sms is common in eu? |
22:18.40 | ManxPower-work | doug: yes. |
22:19.03 | doug | how does one receive texts? |
22:19.16 | ManxPower-work | Most european carriers seem to have PSTN SMS <=> Cellular SMS gateways set up. |
22:19.18 | doug | i'm not so good at recognizing modem tones... |
22:19.41 | carrar | doug, the easyest thing to do is buy one of those GMS Modems kits, buy a SIM card and conenct it via Serial to your asterisk box and send SMS commands to your GSM Modem which then can send your SMS message to the SMSC |
22:19.47 | ManxPower-work | doug: you receive texts by receiving a call with an FSK SMS text burst to a PSTN SMS terminal |
22:20.30 | ManxPower-work | I wish they would rename the app to something like EUSMS |
22:20.40 | ManxPower-work | or SMSBUTNOTINUSA |
22:20.56 | doug | oh, needs a terminal. |
22:21.06 | doug | thanks guys, be back later. |
22:21.15 | carrar | http://www.sparkfun.com/commerce/product_info.php?products_id=478 |
22:21.25 | doug | dtmftotext, exactly what i needed (i think). |
22:21.26 | doug | thanks manx |
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23:57.22 | stallione | broke my working asterisk install trying to upgrade freepbx.. :( |
23:57.37 | stallione | Asterisk doesn't start anymore |